Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

2460 files changed, 471722 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/BUILD.gn b/third_party/libwebrtc/modules/BUILD.gn
new file mode 100644
index 0000000000..ffb7491b31
--- /dev/null
+++ b/third_party/libwebrtc/modules/BUILD.gn
@@ -0,0 +1,256 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../webrtc.gni")
+import("audio_coding/audio_coding.gni")
+
+group("modules") {
+  deps = [
+    "audio_coding",
+    "audio_device",
+    "audio_mixer",
+    "audio_processing",
+    "congestion_controller",
+    "pacing",
+    "remote_bitrate_estimator",
+    "rtp_rtcp",
+    "utility",
+    "video_coding",
+  ]
+
+  if (rtc_desktop_capture_supported) {
+    deps += [ "desktop_capture" ]
+  }
+}
+
+rtc_source_set("module_api_public") {
+  sources = [ "include/module_common_types_public.h" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_source_set("module_api") {
+  visibility = [ "*" ]
+  sources = [ "include/module_common_types.h" ]
+}
+
+rtc_source_set("module_fec_api") {
+  visibility = [ "*" ]
+  sources = [ "include/module_fec_types.h" ]
+}
+
+if (rtc_include_tests && !build_with_chromium) {
+  modules_tests_resources = [
+    "../resources/audio_coding/testfile16kHz.pcm",
+    "../resources/audio_coding/testfile32kHz.pcm",
+    "../resources/audio_coding/teststereo32kHz.pcm",
+    "../resources/foreman_cif.yuv",
+  ]
+
+  if (is_ios) {
+    bundle_data("modules_tests_bundle_data") {
+      testonly = true
+      sources = modules_tests_resources
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+    }
+  }
+
+  rtc_test("modules_tests") {
+    testonly = true
+
+    deps = [
+      "../test:test_main",
+      "../test:video_test_common",
+      "audio_coding:audio_coding_modules_tests",
+      "rtp_rtcp:rtp_rtcp_modules_tests",
+      "video_coding:video_coding_modules_tests",
+      "//testing/gtest",
+    ]
+
+    if (rtc_desktop_capture_supported) {
+      deps += [ "desktop_capture:desktop_capture_modules_tests" ]
+    }
+
+    data = modules_tests_resources
+
+    if (is_android) {
+      use_default_launcher = false
+      deps += [
+        # NOTE(brandtr): Including Java classes seems only to be possible from
+        # rtc_test targets. Therefore we include this target here, instead of
+        # in video_coding_modules_tests, where it is actually used.
+        "../sdk/android:libjingle_peerconnection_java",
+        "//sdk/android:native_test_jni_onload",
+        "//testing/android/native_test:native_test_support",
+      ]
+      shard_timeout = 900
+    }
+
+    if (is_ios) {
+      deps += [ ":modules_tests_bundle_data" ]
+    }
+  }
+
+  modules_unittests_resources = [
+    "../resources/audio_coding/neteq_opus.rtp",
+    "../resources/audio_coding/neteq_opus_dtx.rtp",
+    "../resources/audio_coding/neteq_universal_new.rtp",
+    "../resources/audio_coding/speech_4_channels_48k_one_second.wav",
+    "../resources/audio_coding/speech_mono_16kHz.pcm",
+    "../resources/audio_coding/speech_mono_32_48kHz.pcm",
+    "../resources/audio_coding/testfile16kHz.pcm",
+    "../resources/audio_coding/testfile32kHz.pcm",
+    "../resources/audio_coding/testfile_fake_stereo_32kHz.pcm",
+    "../resources/audio_coding/teststereo32kHz.pcm",
+    "../resources/audio_device/audio_short16.pcm",
+    "../resources/audio_device/audio_short44.pcm",
+    "../resources/audio_device/audio_short48.pcm",
+    "../resources/audio_processing/agc/agc_audio.pcm",
+    "../resources/audio_processing/agc/agc_no_circular_buffer.dat",
+    "../resources/audio_processing/agc/agc_pitch_gain.dat",
+    "../resources/audio_processing/agc/agc_pitch_lag.dat",
+    "../resources/audio_processing/agc/agc_spectral_peak.dat",
+    "../resources/audio_processing/agc/agc_vad.dat",
+    "../resources/audio_processing/agc/agc_voicing_prob.dat",
+    "../resources/audio_processing/agc/agc_with_circular_buffer.dat",
+    "../resources/audio_processing/output_data_fixed.pb",
+    "../resources/audio_processing/output_data_float.pb",
+    "../resources/audio_processing/output_data_float_avx2.pb",
+    "../resources/audio_processing/output_data_mac.pb",
+    "../resources/audio_processing/transient/ajm-macbook-1-spke16m.pcm",
+    "../resources/audio_processing/transient/audio16kHz.pcm",
+    "../resources/audio_processing/transient/audio32kHz.pcm",
+    "../resources/audio_processing/transient/audio48kHz.pcm",
+    "../resources/audio_processing/transient/audio8kHz.pcm",
+    "../resources/audio_processing/transient/detect16kHz.dat",
+    "../resources/audio_processing/transient/detect32kHz.dat",
+    "../resources/audio_processing/transient/detect48kHz.dat",
+    "../resources/audio_processing/transient/detect8kHz.dat",
+    "../resources/audio_processing/transient/double-utils.dat",
+    "../resources/audio_processing/transient/float-utils.dat",
+    "../resources/audio_processing/transient/suppressed16kHz.pcm",
+    "../resources/audio_processing/transient/suppressed32kHz.pcm",
+    "../resources/audio_processing/transient/suppressed8kHz.pcm",
+    "../resources/audio_processing/transient/wpd0.dat",
+    "../resources/audio_processing/transient/wpd1.dat",
+    "../resources/audio_processing/transient/wpd2.dat",
+    "../resources/audio_processing/transient/wpd3.dat",
+    "../resources/audio_processing/transient/wpd4.dat",
+    "../resources/audio_processing/transient/wpd5.dat",
+    "../resources/audio_processing/transient/wpd6.dat",
+    "../resources/audio_processing/transient/wpd7.dat",
+    "../resources/deflicker_before_cif_short.yuv",
+    "../resources/far16_stereo.pcm",
+    "../resources/far176_stereo.pcm",
+    "../resources/far192_stereo.pcm",
+    "../resources/far22_stereo.pcm",
+    "../resources/far32_stereo.pcm",
+    "../resources/far44_stereo.pcm",
+    "../resources/far48_stereo.pcm",
+    "../resources/far88_stereo.pcm",
+    "../resources/far8_stereo.pcm",
+    "../resources/far96_stereo.pcm",
+    "../resources/foremanColorEnhanced_cif_short.yuv",
+    "../resources/foreman_cif.yuv",
+    "../resources/foreman_cif_short.yuv",
+    "../resources/near16_stereo.pcm",
+    "../resources/near176_stereo.pcm",
+    "../resources/near192_stereo.pcm",
+    "../resources/near22_stereo.pcm",
+    "../resources/near32_stereo.pcm",
+    "../resources/near44_stereo.pcm",
+    "../resources/near48_stereo.pcm",
+    "../resources/near88_stereo.pcm",
+    "../resources/near8_stereo.pcm",
+    "../resources/near96_stereo.pcm",
+    "../resources/ref03.aecdump",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_0_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_0_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_1_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke1_1_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_0_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_0_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_1_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingChoke2_1_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingDelay1_0_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingDelay1_0_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingLoss1_0_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_IncreasingLoss1_0_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_Multi1_1_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_Multi1_1_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_0_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_0_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_1_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyChoke_1_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyDelay_0_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyDelay_0_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyLoss_0_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_SteadyLoss_0_TOF.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_AST.bin",
+    "../resources/remote_bitrate_estimator/VideoSendersTest_BweTest_UnlimitedSpeed_0_TOF.bin",
+    "../resources/short_mixed_mono_48.dat",
+    "../resources/short_mixed_mono_48.pcm",
+    "../resources/short_mixed_mono_48_arm.dat",
+    "../resources/short_mixed_stereo_48.dat",
+    "../resources/short_mixed_stereo_48.pcm",
+    "../resources/voice_engine/audio_tiny48.wav",
+  ]
+  if (is_ios) {
+    bundle_data("modules_unittests_bundle_data") {
+      testonly = true
+      sources = modules_unittests_resources
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+    }
+  }
+
+  rtc_test("modules_unittests") {
+    testonly = true
+    defines = []
+    sources = [ "module_common_types_unittest.cc" ]
+
+    deps = [
+      ":module_api",
+      ":module_api_public",
+      "../test:test_main",
+      "../test:test_support",
+      "audio_coding:audio_coding_unittests",
+      "audio_device:audio_device_unittests",
+      "audio_mixer:audio_mixer_unittests",
+      "audio_processing:audio_processing_unittests",
+      "audio_processing/aec3:aec3_unittests",
+      "audio_processing/ns:ns_unittests",
+      "congestion_controller:congestion_controller_unittests",
+      "pacing:pacing_unittests",
+      "remote_bitrate_estimator:remote_bitrate_estimator_unittests",
+      "rtp_rtcp:rtp_rtcp_unittests",
+      "utility:utility_unittests",
+      "video_coding:video_coding_unittests",
+      "video_coding/timing:timing_unittests",
+    ]
+
+    if (rtc_desktop_capture_supported) {
+      deps += [ "desktop_capture:desktop_capture_unittests" ]
+    }
+
+    data = modules_unittests_resources
+
+    if (is_android) {
+      use_default_launcher = false
+      deps += [
+        "../sdk/android:libjingle_peerconnection_java",
+        "//testing/android/native_test:native_test_support",
+      ]
+      shard_timeout = 900
+    }
+    if (is_ios) {
+      info_plist = "../test/ios/Info.plist"
+      deps += [ ":modules_unittests_bundle_data" ]
+      configs += [ "..:common_objc" ]
+      ldflags = [ "-ObjC" ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/async_audio_processing/BUILD.gn b/third_party/libwebrtc/modules/async_audio_processing/BUILD.gn
new file mode 100644
index 0000000000..7a7ca20df1
--- /dev/null
+++ b/third_party/libwebrtc/modules/async_audio_processing/BUILD.gn
@@ -0,0 +1,43 @@
+# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("async_audio_processing") {
+  sources = [
+    "async_audio_processing.cc",
+    "async_audio_processing.h",
+  ]
+
+  public = [ "async_audio_processing.h" ]
+
+  deps = [
+    "../../api:scoped_refptr",
+    "../../api:sequence_checker",
+    "../../api/audio:audio_frame_api",
+    "../../api/audio:audio_frame_processor",
+    "../../api/task_queue:task_queue",
+    "../../rtc_base:checks",
+    "../../rtc_base:refcount",
+    "../../rtc_base:rtc_task_queue",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("async_audio_processing_test") {
+    testonly = true
+
+    sources = []
+
+    deps = [
+      ":async_audio_processing",
+      "../../api/audio:audio_frame_api",
+      "../../rtc_base:checks",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc
new file mode 100644
index 0000000000..9452f3bcf9
--- /dev/null
+++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc
@@ -0,0 +1,61 @@
+
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/async_audio_processing/async_audio_processing.h"
+
+#include <utility>
+
+#include "api/audio/audio_frame.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AsyncAudioProcessing::Factory::~Factory() = default;
+AsyncAudioProcessing::Factory::Factory(AudioFrameProcessor& frame_processor,
+                                       TaskQueueFactory& task_queue_factory)
+    : frame_processor_(frame_processor),
+      task_queue_factory_(task_queue_factory) {}
+
+std::unique_ptr<AsyncAudioProcessing>
+AsyncAudioProcessing::Factory::CreateAsyncAudioProcessing(
+    AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback) {
+  return std::make_unique<AsyncAudioProcessing>(
+      frame_processor_, task_queue_factory_,
+      std::move(on_frame_processed_callback));
+}
+
+AsyncAudioProcessing::~AsyncAudioProcessing() {
+  frame_processor_.SetSink(nullptr);
+}
+
+AsyncAudioProcessing::AsyncAudioProcessing(
+    AudioFrameProcessor& frame_processor,
+    TaskQueueFactory& task_queue_factory,
+    AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback)
+    : on_frame_processed_callback_(std::move(on_frame_processed_callback)),
+      frame_processor_(frame_processor),
+      task_queue_(task_queue_factory.CreateTaskQueue(
+          "AsyncAudioProcessing",
+          TaskQueueFactory::Priority::NORMAL)) {
+  frame_processor_.SetSink([this](std::unique_ptr<AudioFrame> frame) {
+    task_queue_.PostTask([this, frame = std::move(frame)]() mutable {
+      on_frame_processed_callback_(std::move(frame));
+    });
+  });
+}
+
+void AsyncAudioProcessing::Process(std::unique_ptr<AudioFrame> frame) {
+  task_queue_.PostTask([this, frame = std::move(frame)]() mutable {
+    frame_processor_.Process(std::move(frame));
+  });
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.h b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.h
new file mode 100644
index 0000000000..bbd0f69b1b
--- /dev/null
+++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_ASYNC_AUDIO_PROCESSING_ASYNC_AUDIO_PROCESSING_H_
+#define MODULES_ASYNC_AUDIO_PROCESSING_ASYNC_AUDIO_PROCESSING_H_
+
+#include <memory>
+
+#include "api/audio/audio_frame_processor.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/task_queue.h"
+
+namespace webrtc {
+
+class AudioFrame;
+class TaskQueueFactory;
+
+// Helper class taking care of interactions with AudioFrameProcessor
+// in asynchronous manner. Offloads AudioFrameProcessor::Process calls
+// to a dedicated task queue. Makes sure that it's always safe for
+// AudioFrameProcessor to pass processed frames back to its sink.
+class AsyncAudioProcessing final {
+ public:
+  // Helper class passing AudioFrameProcessor and TaskQueueFactory into
+  // AsyncAudioProcessing constructor.
+  class Factory : public rtc::RefCountInterface {
+   public:
+    Factory(const Factory&) = delete;
+    Factory& operator=(const Factory&) = delete;
+
+    ~Factory();
+    Factory(AudioFrameProcessor& frame_processor,
+            TaskQueueFactory& task_queue_factory);
+
+    std::unique_ptr<AsyncAudioProcessing> CreateAsyncAudioProcessing(
+        AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback);
+
+   private:
+    AudioFrameProcessor& frame_processor_;
+    TaskQueueFactory& task_queue_factory_;
+  };
+
+  AsyncAudioProcessing(const AsyncAudioProcessing&) = delete;
+  AsyncAudioProcessing& operator=(const AsyncAudioProcessing&) = delete;
+
+  ~AsyncAudioProcessing();
+
+  // Creates AsyncAudioProcessing which will pass audio frames to
+  // `frame_processor` on `task_queue_` and reply with processed frames passed
+  // into `on_frame_processed_callback`, which is posted back onto
+  // `task_queue_`. `task_queue_` is created using the provided
+  // `task_queue_factory`.
+  AsyncAudioProcessing(
+      AudioFrameProcessor& frame_processor,
+      TaskQueueFactory& task_queue_factory,
+      AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback);
+
+  // Accepts `frame` for asynchronous processing. Thread-safe.
+  void Process(std::unique_ptr<AudioFrame> frame);
+
+ private:
+  AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback_;
+  AudioFrameProcessor& frame_processor_;
+  rtc::TaskQueue task_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_ASYNC_AUDIO_PROCESSING_ASYNC_AUDIO_PROCESSING_H_
diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build
new file mode 100644
index 0000000000..f599fadae2
--- /dev/null
+++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("async_audio_processing_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/BUILD.gn b/third_party/libwebrtc/modules/audio_coding/BUILD.gn
new file mode 100644
index 0000000000..eac0650a26
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/BUILD.gn
@@ -0,0 +1,1805 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+import("audio_coding.gni")
+if (rtc_enable_protobuf) {
+  import("//third_party/protobuf/proto_library.gni")
+}
+
+visibility = [ ":*" ]
+
+rtc_source_set("audio_coding_module_typedefs") {
+  visibility += [ "*" ]
+  sources = [ "include/audio_coding_module_typedefs.h" ]
+}
+
+rtc_library("audio_coding") {
+  visibility += [ "*" ]
+  sources = [
+    "acm2/acm_receiver.cc",
+    "acm2/acm_receiver.h",
+    "acm2/acm_remixing.cc",
+    "acm2/acm_remixing.h",
+    "acm2/acm_resampler.cc",
+    "acm2/acm_resampler.h",
+    "acm2/audio_coding_module.cc",
+    "acm2/call_statistics.cc",
+    "acm2/call_statistics.h",
+    "include/audio_coding_module.h",
+  ]
+
+  defines = []
+
+  deps = [
+    ":audio_coding_module_typedefs",
+    ":default_neteq_factory",
+    ":neteq",
+    "..:module_api",
+    "..:module_api_public",
+    "../../api:array_view",
+    "../../api:function_view",
+    "../../api/audio:audio_frame_api",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/neteq:neteq_api",
+    "../../common_audio",
+    "../../common_audio:common_audio_c",
+    "../../rtc_base:audio_format_to_string",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
+    "../../system_wrappers:metrics",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("legacy_encoded_audio_frame") {
+  sources = [
+    "codecs/legacy_encoded_audio_frame.cc",
+    "codecs/legacy_encoded_audio_frame.h",
+  ]
+  deps = [
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("webrtc_cng") {
+  visibility += webrtc_default_visibility
+  sources = [
+    "codecs/cng/webrtc_cng.cc",
+    "codecs/cng/webrtc_cng.h",
+  ]
+
+  deps = [
+    "../../api:array_view",
+    "../../common_audio:common_audio_c",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:safe_conversions",
+  ]
+}
+
+rtc_library("audio_encoder_cng") {
+  visibility += [ "*" ]
+  sources = [
+    "codecs/cng/audio_encoder_cng.cc",
+    "codecs/cng/audio_encoder_cng.h",
+  ]
+
+  deps = [
+    ":webrtc_cng",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/units:time_delta",
+    "../../common_audio",
+    "../../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("red") {
+  visibility += [ "*" ]
+  sources = [
+    "codecs/red/audio_encoder_copy_red.cc",
+    "codecs/red/audio_encoder_copy_red.h",
+  ]
+
+  deps = [
+    "../../api:array_view",
+    "../../api:field_trials_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/units:time_delta",
+    "../../common_audio",
+    "../../rtc_base:buffer",
+    "../../rtc_base:byte_order",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("g711") {
+  visibility += [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/g711/audio_decoder_pcm.cc",
+    "codecs/g711/audio_decoder_pcm.h",
+    "codecs/g711/audio_encoder_pcm.cc",
+    "codecs/g711/audio_encoder_pcm.h",
+  ]
+
+  deps = [
+    ":legacy_encoded_audio_frame",
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/units:time_delta",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  public_deps = [ ":g711_c" ]  # no-presubmit-check TODO(webrtc:8603)
+}
+
+rtc_library("g711_c") {
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/g711/g711_interface.c",
+    "codecs/g711/g711_interface.h",
+  ]
+  deps = [ "../third_party/g711:g711_3p" ]
+}
+
+rtc_library("g722") {
+  visibility += [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/g722/audio_decoder_g722.cc",
+    "codecs/g722/audio_decoder_g722.h",
+    "codecs/g722/audio_encoder_g722.cc",
+    "codecs/g722/audio_encoder_g722.h",
+  ]
+
+  deps = [
+    ":legacy_encoded_audio_frame",
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/audio_codecs/g722:audio_encoder_g722_config",
+    "../../api/units:time_delta",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:safe_conversions",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  public_deps = [ ":g722_c" ]  # no-presubmit-check TODO(webrtc:8603)
+}
+
+rtc_library("g722_c") {
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/g722/g722_interface.c",
+    "codecs/g722/g722_interface.h",
+  ]
+  deps = [ "../third_party/g722:g722_3p" ]
+}
+
+rtc_library("ilbc") {
+  visibility += webrtc_default_visibility
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/ilbc/audio_decoder_ilbc.cc",
+    "codecs/ilbc/audio_decoder_ilbc.h",
+    "codecs/ilbc/audio_encoder_ilbc.cc",
+    "codecs/ilbc/audio_encoder_ilbc.h",
+  ]
+
+  deps = [
+    ":legacy_encoded_audio_frame",
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/audio_codecs/ilbc:audio_encoder_ilbc_config",
+    "../../api/units:time_delta",
+    "../../common_audio",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:safe_conversions",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  public_deps = [ ":ilbc_c" ]  # no-presubmit-check TODO(webrtc:8603)
+}
+
+rtc_library("ilbc_c") {
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/ilbc/abs_quant.c",
+    "codecs/ilbc/abs_quant.h",
+    "codecs/ilbc/abs_quant_loop.c",
+    "codecs/ilbc/abs_quant_loop.h",
+    "codecs/ilbc/augmented_cb_corr.c",
+    "codecs/ilbc/augmented_cb_corr.h",
+    "codecs/ilbc/bw_expand.c",
+    "codecs/ilbc/bw_expand.h",
+    "codecs/ilbc/cb_construct.c",
+    "codecs/ilbc/cb_construct.h",
+    "codecs/ilbc/cb_mem_energy.c",
+    "codecs/ilbc/cb_mem_energy.h",
+    "codecs/ilbc/cb_mem_energy_augmentation.c",
+    "codecs/ilbc/cb_mem_energy_augmentation.h",
+    "codecs/ilbc/cb_mem_energy_calc.c",
+    "codecs/ilbc/cb_mem_energy_calc.h",
+    "codecs/ilbc/cb_search.c",
+    "codecs/ilbc/cb_search.h",
+    "codecs/ilbc/cb_search_core.c",
+    "codecs/ilbc/cb_search_core.h",
+    "codecs/ilbc/cb_update_best_index.c",
+    "codecs/ilbc/cb_update_best_index.h",
+    "codecs/ilbc/chebyshev.c",
+    "codecs/ilbc/chebyshev.h",
+    "codecs/ilbc/comp_corr.c",
+    "codecs/ilbc/comp_corr.h",
+    "codecs/ilbc/constants.c",
+    "codecs/ilbc/constants.h",
+    "codecs/ilbc/create_augmented_vec.c",
+    "codecs/ilbc/create_augmented_vec.h",
+    "codecs/ilbc/decode.c",
+    "codecs/ilbc/decode.h",
+    "codecs/ilbc/decode_residual.c",
+    "codecs/ilbc/decode_residual.h",
+    "codecs/ilbc/decoder_interpolate_lsf.c",
+    "codecs/ilbc/decoder_interpolate_lsf.h",
+    "codecs/ilbc/defines.h",
+    "codecs/ilbc/do_plc.c",
+    "codecs/ilbc/do_plc.h",
+    "codecs/ilbc/encode.c",
+    "codecs/ilbc/encode.h",
+    "codecs/ilbc/energy_inverse.c",
+    "codecs/ilbc/energy_inverse.h",
+    "codecs/ilbc/enh_upsample.c",
+    "codecs/ilbc/enh_upsample.h",
+    "codecs/ilbc/enhancer.c",
+    "codecs/ilbc/enhancer.h",
+    "codecs/ilbc/enhancer_interface.c",
+    "codecs/ilbc/enhancer_interface.h",
+    "codecs/ilbc/filtered_cb_vecs.c",
+    "codecs/ilbc/filtered_cb_vecs.h",
+    "codecs/ilbc/frame_classify.c",
+    "codecs/ilbc/frame_classify.h",
+    "codecs/ilbc/gain_dequant.c",
+    "codecs/ilbc/gain_dequant.h",
+    "codecs/ilbc/gain_quant.c",
+    "codecs/ilbc/gain_quant.h",
+    "codecs/ilbc/get_cd_vec.c",
+    "codecs/ilbc/get_cd_vec.h",
+    "codecs/ilbc/get_lsp_poly.c",
+    "codecs/ilbc/get_lsp_poly.h",
+    "codecs/ilbc/get_sync_seq.c",
+    "codecs/ilbc/get_sync_seq.h",
+    "codecs/ilbc/hp_input.c",
+    "codecs/ilbc/hp_input.h",
+    "codecs/ilbc/hp_output.c",
+    "codecs/ilbc/hp_output.h",
+    "codecs/ilbc/ilbc.c",
+    "codecs/ilbc/ilbc.h",
+    "codecs/ilbc/index_conv_dec.c",
+    "codecs/ilbc/index_conv_dec.h",
+    "codecs/ilbc/index_conv_enc.c",
+    "codecs/ilbc/index_conv_enc.h",
+    "codecs/ilbc/init_decode.c",
+    "codecs/ilbc/init_decode.h",
+    "codecs/ilbc/init_encode.c",
+    "codecs/ilbc/init_encode.h",
+    "codecs/ilbc/interpolate.c",
+    "codecs/ilbc/interpolate.h",
+    "codecs/ilbc/interpolate_samples.c",
+    "codecs/ilbc/interpolate_samples.h",
+    "codecs/ilbc/lpc_encode.c",
+    "codecs/ilbc/lpc_encode.h",
+    "codecs/ilbc/lsf_check.c",
+    "codecs/ilbc/lsf_check.h",
+    "codecs/ilbc/lsf_interpolate_to_poly_dec.c",
+    "codecs/ilbc/lsf_interpolate_to_poly_dec.h",
+    "codecs/ilbc/lsf_interpolate_to_poly_enc.c",
+    "codecs/ilbc/lsf_interpolate_to_poly_enc.h",
+    "codecs/ilbc/lsf_to_lsp.c",
+    "codecs/ilbc/lsf_to_lsp.h",
+    "codecs/ilbc/lsf_to_poly.c",
+    "codecs/ilbc/lsf_to_poly.h",
+    "codecs/ilbc/lsp_to_lsf.c",
+    "codecs/ilbc/lsp_to_lsf.h",
+    "codecs/ilbc/my_corr.c",
+    "codecs/ilbc/my_corr.h",
+    "codecs/ilbc/nearest_neighbor.c",
+    "codecs/ilbc/nearest_neighbor.h",
+    "codecs/ilbc/pack_bits.c",
+    "codecs/ilbc/pack_bits.h",
+    "codecs/ilbc/poly_to_lsf.c",
+    "codecs/ilbc/poly_to_lsf.h",
+    "codecs/ilbc/poly_to_lsp.c",
+    "codecs/ilbc/poly_to_lsp.h",
+    "codecs/ilbc/refiner.c",
+    "codecs/ilbc/refiner.h",
+    "codecs/ilbc/simple_interpolate_lsf.c",
+    "codecs/ilbc/simple_interpolate_lsf.h",
+    "codecs/ilbc/simple_lpc_analysis.c",
+    "codecs/ilbc/simple_lpc_analysis.h",
+    "codecs/ilbc/simple_lsf_dequant.c",
+    "codecs/ilbc/simple_lsf_dequant.h",
+    "codecs/ilbc/simple_lsf_quant.c",
+    "codecs/ilbc/simple_lsf_quant.h",
+    "codecs/ilbc/smooth.c",
+    "codecs/ilbc/smooth.h",
+    "codecs/ilbc/smooth_out_data.c",
+    "codecs/ilbc/smooth_out_data.h",
+    "codecs/ilbc/sort_sq.c",
+    "codecs/ilbc/sort_sq.h",
+    "codecs/ilbc/split_vq.c",
+    "codecs/ilbc/split_vq.h",
+    "codecs/ilbc/state_construct.c",
+    "codecs/ilbc/state_construct.h",
+    "codecs/ilbc/state_search.c",
+    "codecs/ilbc/state_search.h",
+    "codecs/ilbc/swap_bytes.c",
+    "codecs/ilbc/swap_bytes.h",
+    "codecs/ilbc/unpack_bits.c",
+    "codecs/ilbc/unpack_bits.h",
+    "codecs/ilbc/vq3.c",
+    "codecs/ilbc/vq3.h",
+    "codecs/ilbc/vq4.c",
+    "codecs/ilbc/vq4.h",
+    "codecs/ilbc/window32_w32.c",
+    "codecs/ilbc/window32_w32.h",
+    "codecs/ilbc/xcorr_coef.c",
+    "codecs/ilbc/xcorr_coef.h",
+  ]
+
+  deps = [
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../common_audio",
+    "../../common_audio:common_audio_c",
+    "../../rtc_base:checks",
+    "../../rtc_base:sanitizer",
+    "../../rtc_base/system:arch",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
+}
+
+rtc_library("isac_vad") {
+  visibility += [ "../audio_processing/vad:*" ]
+  sources = [
+    "codecs/isac/main/source/filter_functions.c",
+    "codecs/isac/main/source/filter_functions.h",
+    "codecs/isac/main/source/isac_vad.c",
+    "codecs/isac/main/source/isac_vad.h",
+    "codecs/isac/main/source/os_specific_inline.h",
+    "codecs/isac/main/source/pitch_estimator.c",
+    "codecs/isac/main/source/pitch_estimator.h",
+    "codecs/isac/main/source/pitch_filter.c",
+    "codecs/isac/main/source/pitch_filter.h",
+    "codecs/isac/main/source/settings.h",
+    "codecs/isac/main/source/structs.h",
+  ]
+  deps = [
+    ":isac_bwinfo",
+    "../../rtc_base:compile_assert_c",
+    "../../rtc_base/system:arch",
+    "../../rtc_base/system:ignore_warnings",
+    "../third_party/fft",
+  ]
+}
+
+rtc_source_set("isac_bwinfo") {
+  sources = [ "codecs/isac/bandwidth_info.h" ]
+  deps = []
+}
+
+rtc_library("pcm16b") {
+  visibility += [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/pcm16b/audio_decoder_pcm16b.cc",
+    "codecs/pcm16b/audio_decoder_pcm16b.h",
+    "codecs/pcm16b/audio_encoder_pcm16b.cc",
+    "codecs/pcm16b/audio_encoder_pcm16b.h",
+    "codecs/pcm16b/pcm16b_common.cc",
+    "codecs/pcm16b/pcm16b_common.h",
+  ]
+
+  deps = [
+    ":g711",
+    ":legacy_encoded_audio_frame",
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+  ]
+  public_deps = [ ":pcm16b_c" ]  # no-presubmit-check TODO(webrtc:8603)
+}
+
+rtc_library("pcm16b_c") {
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/pcm16b/pcm16b.c",
+    "codecs/pcm16b/pcm16b.h",
+  ]
+}
+
+rtc_library("audio_coding_opus_common") {
+  sources = [
+    "codecs/opus/audio_coder_opus_common.cc",
+    "codecs/opus/audio_coder_opus_common.h",
+  ]
+
+  deps = [
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../rtc_base:checks",
+    "../../rtc_base:stringutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("webrtc_opus") {
+  visibility += webrtc_default_visibility
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/opus/audio_decoder_opus.cc",
+    "codecs/opus/audio_decoder_opus.h",
+    "codecs/opus/audio_encoder_opus.cc",
+    "codecs/opus/audio_encoder_opus.h",
+  ]
+
+  deps = [
+    ":audio_coding_opus_common",
+    ":audio_network_adaptor",
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/audio_codecs/opus:audio_encoder_opus_config",
+    "../../common_audio",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:protobuf_utils",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base:stringutils",
+    "../../rtc_base:timeutils",
+    "../../system_wrappers:field_trial",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  public_deps =  # no-presubmit-check TODO(webrtc:8603)
+      [ ":webrtc_opus_wrapper" ]
+
+  defines = audio_codec_defines
+}
+
+rtc_library("webrtc_multiopus") {
+  visibility += webrtc_default_visibility
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/opus/audio_decoder_multi_channel_opus_impl.cc",
+    "codecs/opus/audio_decoder_multi_channel_opus_impl.h",
+    "codecs/opus/audio_encoder_multi_channel_opus_impl.cc",
+    "codecs/opus/audio_encoder_multi_channel_opus_impl.h",
+  ]
+
+  deps = [
+    ":audio_coding_opus_common",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/audio_codecs/opus:audio_decoder_opus_config",
+    "../../api/audio_codecs/opus:audio_encoder_opus_config",
+    "../../api/units:time_delta",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base:stringutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  public_deps =  # no-presubmit-check TODO(webrtc:8603)
+      [ ":webrtc_opus_wrapper" ]
+
+  defines = audio_codec_defines
+}
+
+rtc_library("webrtc_opus_wrapper") {
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "codecs/opus/opus_inst.h",
+    "codecs/opus/opus_interface.cc",
+    "codecs/opus/opus_interface.h",
+  ]
+
+  defines = audio_coding_defines
+
+  deps = [
+    "../../api:array_view",
+    "../../rtc_base:checks",
+    "../../rtc_base:ignore_wundef",
+    "../../system_wrappers:field_trial",
+  ]
+
+  if (rtc_build_opus) {
+    deps += [ rtc_opus_dir ]
+    public_configs = [ "//third_party/opus:opus_config" ]
+  } else if (build_with_mozilla) {
+    public_configs = [ "//third_party/opus:opus_config" ]
+  }
+}
+
+if (rtc_enable_protobuf) {
+  proto_library("ana_debug_dump_proto") {
+    visibility += webrtc_default_visibility
+    sources = [ "audio_network_adaptor/debug_dump.proto" ]
+    link_deps = [ ":ana_config_proto" ]
+    proto_out_dir = "modules/audio_coding/audio_network_adaptor"
+  }
+  proto_library("ana_config_proto") {
+    visibility += [ "*" ]
+    sources = [ "audio_network_adaptor/config.proto" ]
+    proto_out_dir = "modules/audio_coding/audio_network_adaptor"
+  }
+}
+
+rtc_library("audio_network_adaptor_config") {
+  visibility += webrtc_default_visibility
+  sources = [
+    "audio_network_adaptor/audio_network_adaptor_config.cc",
+    "audio_network_adaptor/include/audio_network_adaptor_config.h",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_network_adaptor") {
+  visibility += webrtc_default_visibility
+  sources = [
+    "audio_network_adaptor/audio_network_adaptor_impl.cc",
+    "audio_network_adaptor/audio_network_adaptor_impl.h",
+    "audio_network_adaptor/bitrate_controller.cc",
+    "audio_network_adaptor/bitrate_controller.h",
+    "audio_network_adaptor/channel_controller.cc",
+    "audio_network_adaptor/channel_controller.h",
+    "audio_network_adaptor/controller.cc",
+    "audio_network_adaptor/controller.h",
+    "audio_network_adaptor/controller_manager.cc",
+    "audio_network_adaptor/controller_manager.h",
+    "audio_network_adaptor/debug_dump_writer.cc",
+    "audio_network_adaptor/debug_dump_writer.h",
+    "audio_network_adaptor/dtx_controller.cc",
+    "audio_network_adaptor/dtx_controller.h",
+    "audio_network_adaptor/event_log_writer.cc",
+    "audio_network_adaptor/event_log_writer.h",
+    "audio_network_adaptor/fec_controller_plr_based.cc",
+    "audio_network_adaptor/fec_controller_plr_based.h",
+    "audio_network_adaptor/frame_length_controller.cc",
+    "audio_network_adaptor/frame_length_controller.h",
+    "audio_network_adaptor/frame_length_controller_v2.cc",
+    "audio_network_adaptor/frame_length_controller_v2.h",
+    "audio_network_adaptor/include/audio_network_adaptor.h",
+    "audio_network_adaptor/util/threshold_curve.h",
+  ]
+
+  public_deps =  # no-presubmit-check TODO(webrtc:8603)
+      [ ":audio_network_adaptor_config" ]
+
+  deps = [
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/rtc_event_log",
+    "../../common_audio",
+    "../../logging:rtc_event_audio",
+    "../../rtc_base:checks",
+    "../../rtc_base:ignore_wundef",
+    "../../rtc_base:logging",
+    "../../rtc_base:protobuf_utils",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/system:file_wrapper",
+    "../../system_wrappers",
+    "../../system_wrappers:field_trial",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+
+  if (rtc_enable_protobuf) {
+    deps += [
+      ":ana_config_proto",
+      ":ana_debug_dump_proto",
+    ]
+  }
+}
+
+rtc_library("neteq") {
+  visibility += webrtc_default_visibility
+  sources = [
+    "neteq/accelerate.cc",
+    "neteq/accelerate.h",
+    "neteq/audio_multi_vector.cc",
+    "neteq/audio_multi_vector.h",
+    "neteq/audio_vector.cc",
+    "neteq/audio_vector.h",
+    "neteq/background_noise.cc",
+    "neteq/background_noise.h",
+    "neteq/buffer_level_filter.cc",
+    "neteq/buffer_level_filter.h",
+    "neteq/comfort_noise.cc",
+    "neteq/comfort_noise.h",
+    "neteq/cross_correlation.cc",
+    "neteq/cross_correlation.h",
+    "neteq/decision_logic.cc",
+    "neteq/decision_logic.h",
+    "neteq/decoder_database.cc",
+    "neteq/decoder_database.h",
+    "neteq/delay_manager.cc",
+    "neteq/delay_manager.h",
+    "neteq/dsp_helper.cc",
+    "neteq/dsp_helper.h",
+    "neteq/dtmf_buffer.cc",
+    "neteq/dtmf_buffer.h",
+    "neteq/dtmf_tone_generator.cc",
+    "neteq/dtmf_tone_generator.h",
+    "neteq/expand.cc",
+    "neteq/expand.h",
+    "neteq/expand_uma_logger.cc",
+    "neteq/expand_uma_logger.h",
+    "neteq/histogram.cc",
+    "neteq/histogram.h",
+    "neteq/merge.cc",
+    "neteq/merge.h",
+    "neteq/nack_tracker.cc",
+    "neteq/nack_tracker.h",
+    "neteq/neteq_impl.cc",
+    "neteq/neteq_impl.h",
+    "neteq/normal.cc",
+    "neteq/normal.h",
+    "neteq/packet.cc",
+    "neteq/packet.h",
+    "neteq/packet_arrival_history.cc",
+    "neteq/packet_arrival_history.h",
+    "neteq/packet_buffer.cc",
+    "neteq/packet_buffer.h",
+    "neteq/post_decode_vad.cc",
+    "neteq/post_decode_vad.h",
+    "neteq/preemptive_expand.cc",
+    "neteq/preemptive_expand.h",
+    "neteq/random_vector.cc",
+    "neteq/random_vector.h",
+    "neteq/red_payload_splitter.cc",
+    "neteq/red_payload_splitter.h",
+    "neteq/reorder_optimizer.cc",
+    "neteq/reorder_optimizer.h",
+    "neteq/statistics_calculator.cc",
+    "neteq/statistics_calculator.h",
+    "neteq/sync_buffer.cc",
+    "neteq/sync_buffer.h",
+    "neteq/time_stretch.cc",
+    "neteq/time_stretch.h",
+    "neteq/timestamp_scaler.cc",
+    "neteq/timestamp_scaler.h",
+    "neteq/underrun_optimizer.cc",
+    "neteq/underrun_optimizer.h",
+  ]
+
+  deps = [
+    ":audio_coding_module_typedefs",
+    ":webrtc_cng",
+    "..:module_api_public",
+    "../../api:array_view",
+    "../../api:rtp_headers",
+    "../../api:rtp_packet_info",
+    "../../api:scoped_refptr",
+    "../../api/audio:audio_frame_api",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/neteq:neteq_api",
+    "../../api/neteq:neteq_controller_api",
+    "../../api/neteq:tick_timer",
+    "../../common_audio",
+    "../../common_audio:common_audio_c",
+    "../../rtc_base:audio_format_to_string",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:gtest_prod",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base:sanitizer",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
+    "../../system_wrappers:field_trial",
+    "../../system_wrappers:metrics",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_source_set("default_neteq_factory") {
+  visibility += webrtc_default_visibility
+  sources = [
+    "neteq/default_neteq_factory.cc",
+    "neteq/default_neteq_factory.h",
+  ]
+  deps = [
+    ":neteq",
+    "../../api:scoped_refptr",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/neteq:default_neteq_controller_factory",
+    "../../api/neteq:neteq_api",
+    "../../system_wrappers:system_wrappers",
+  ]
+}
+
+# Although providing only test support, this target must be outside of the
+# rtc_include_tests conditional. The reason is that it supports fuzzer tests
+# that ultimately are built and run as a part of the Chromium ecosystem, which
+# does not set the rtc_include_tests flag.
+rtc_library("neteq_tools_minimal") {
+  visibility += webrtc_default_visibility
+  sources = [
+    "neteq/tools/audio_sink.cc",
+    "neteq/tools/audio_sink.h",
+    "neteq/tools/encode_neteq_input.cc",
+    "neteq/tools/encode_neteq_input.h",
+    "neteq/tools/neteq_input.cc",
+    "neteq/tools/neteq_input.h",
+    "neteq/tools/neteq_test.cc",
+    "neteq/tools/neteq_test.h",
+    "neteq/tools/packet.cc",
+    "neteq/tools/packet.h",
+    "neteq/tools/packet_source.cc",
+    "neteq/tools/packet_source.h",
+  ]
+
+  deps = [
+    ":default_neteq_factory",
+    ":neteq",
+    "../../api:array_view",
+    "../../api:neteq_simulator_api",
+    "../../api:rtp_headers",
+    "../../api/audio:audio_frame_api",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/neteq:custom_neteq_factory",
+    "../../api/neteq:default_neteq_controller_factory",
+    "../../api/neteq:neteq_api",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:copy_on_write_buffer",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:stringutils",
+    "../../system_wrappers",
+    "../rtp_rtcp:rtp_rtcp_format",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  defines = audio_codec_defines
+}
+
+rtc_library("neteq_test_tools") {
+  visibility += webrtc_default_visibility
+  testonly = true
+  sources = [
+    "neteq/tools/audio_checksum.h",
+    "neteq/tools/audio_loop.cc",
+    "neteq/tools/audio_loop.h",
+    "neteq/tools/constant_pcm_packet_source.cc",
+    "neteq/tools/constant_pcm_packet_source.h",
+    "neteq/tools/initial_packet_inserter_neteq_input.cc",
+    "neteq/tools/initial_packet_inserter_neteq_input.h",
+    "neteq/tools/neteq_packet_source_input.cc",
+    "neteq/tools/neteq_packet_source_input.h",
+    "neteq/tools/output_audio_file.h",
+    "neteq/tools/output_wav_file.h",
+    "neteq/tools/rtp_file_source.cc",
+    "neteq/tools/rtp_file_source.h",
+    "neteq/tools/rtp_generator.cc",
+    "neteq/tools/rtp_generator.h",
+  ]
+
+  deps = [
+    ":neteq_tools",
+    ":neteq_tools_minimal",
+    ":pcm16b",
+    "../../api:array_view",
+    "../../api:rtp_headers",
+    "../../common_audio",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:ssl",
+    "../../rtc_base:stringutils",
+    "../../rtc_base/system:arch",
+    "../../test:rtp_test_utils",
+    "../rtp_rtcp:rtp_rtcp_format",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+
+  if (rtc_enable_protobuf) {
+    sources += [
+      "neteq/tools/neteq_event_log_input.cc",
+      "neteq/tools/neteq_event_log_input.h",
+    ]
+    deps += [ ":rtc_event_log_source" ]
+  }
+}
+
+rtc_library("neteq_tools") {
+  visibility += webrtc_default_visibility
+  sources = [
+    "neteq/tools/fake_decode_from_file.cc",
+    "neteq/tools/fake_decode_from_file.h",
+    "neteq/tools/neteq_delay_analyzer.cc",
+    "neteq/tools/neteq_delay_analyzer.h",
+    "neteq/tools/neteq_replacement_input.cc",
+    "neteq/tools/neteq_replacement_input.h",
+    "neteq/tools/neteq_stats_getter.cc",
+    "neteq/tools/neteq_stats_getter.h",
+    "neteq/tools/neteq_stats_plotter.cc",
+    "neteq/tools/neteq_stats_plotter.h",
+  ]
+
+  deps = [
+    ":neteq_input_audio_tools",
+    ":neteq_tools_minimal",
+    "..:module_api_public",
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../rtc_base:checks",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:stringutils",
+    "../../rtc_base:timeutils",
+    "../rtp_rtcp",
+    "../rtp_rtcp:rtp_rtcp_format",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("neteq_input_audio_tools") {
+  visibility += webrtc_default_visibility
+  sources = [
+    "neteq/tools/input_audio_file.cc",
+    "neteq/tools/input_audio_file.h",
+    "neteq/tools/resample_input_audio_file.cc",
+    "neteq/tools/resample_input_audio_file.h",
+  ]
+
+  deps = [
+    "../../common_audio",
+    "../../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+if (rtc_enable_protobuf) {
+  rtc_library("rtc_event_log_source") {
+    testonly = true
+
+    sources = [
+      "neteq/tools/rtc_event_log_source.cc",
+      "neteq/tools/rtc_event_log_source.h",
+    ]
+
+    deps = [
+      ":neteq_tools_minimal",
+      "../../logging:rtc_event_log_parser",
+      "../../rtc_base:checks",
+      "../rtp_rtcp",
+      "../rtp_rtcp:rtp_rtcp_format",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+    public_deps =  # no-presubmit-check TODO(webrtc:8603)
+        [ "../../logging:rtc_event_log_proto" ]
+  }
+
+  # Only used for test purpose. Since we want to use it from chromium
+  # (see audio_coding_modules_tests_shared below), we cannot guard it
+  # under rtc_include_tests.
+  proto_library("neteq_unittest_proto") {
+    testonly = true
+    sources = [ "neteq/neteq_unittest.proto" ]
+    proto_out_dir = "modules/audio_coding/neteq"
+  }
+}
+
+# Allow to re-use some test classes from chromium.
+rtc_library("audio_coding_modules_tests_shared") {
+  testonly = true
+  visibility = []
+  visibility = [ "*" ]
+
+  sources = [
+    "neteq/test/neteq_decoding_test.cc",
+    "neteq/test/neteq_decoding_test.h",
+    "neteq/test/result_sink.cc",
+    "neteq/test/result_sink.h",
+    "test/PCMFile.cc",
+    "test/PCMFile.h",
+    "test/TestStereo.cc",
+    "test/TestStereo.h",
+    "test/opus_test.cc",
+    "test/opus_test.h",
+  ]
+
+  deps = [
+    ":audio_coding",
+    ":audio_coding_module_typedefs",
+    ":default_neteq_factory",
+    ":neteq_test_tools",
+    ":neteq_tools_minimal",
+    ":webrtc_opus_wrapper",
+    "..:module_api",
+    "../../api:rtp_headers",
+    "../../api/audio:audio_frame_api",
+    "../../api/audio_codecs:builtin_audio_decoder_factory",
+    "../../api/audio_codecs:builtin_audio_encoder_factory",
+    "../../api/neteq:neteq_api",
+    "../../rtc_base:checks",
+    "../../rtc_base:ignore_wundef",
+    "../../rtc_base:ssl",
+    "../../rtc_base:stringutils",
+    "../../system_wrappers",
+    "../../test:fileutils",
+    "../../test:test_support",
+    "//testing/gtest",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  defines = audio_coding_defines
+
+  if (rtc_enable_protobuf) {
+    defines += [ "WEBRTC_NETEQ_UNITTEST_BITEXACT" ]
+    deps += [ ":neteq_unittest_proto" ]
+  }
+}
+
+if (rtc_include_tests) {
+  audio_coding_deps = [
+    ":audio_encoder_cng",
+    ":g711",
+    ":g722",
+    ":pcm16b",
+    "../../common_audio",
+    "../../system_wrappers",
+  ]
+  if (rtc_include_ilbc) {
+    audio_coding_deps += [ ":ilbc" ]
+  }
+  if (rtc_include_opus) {
+    audio_coding_deps += [ ":webrtc_opus" ]
+  }
+  if (!build_with_mozilla && !build_with_chromium) {
+    audio_coding_deps += [ ":red" ]
+  }
+
+  rtc_source_set("mocks") {
+    testonly = true
+    sources = [
+      "audio_network_adaptor/mock/mock_audio_network_adaptor.h",
+      "audio_network_adaptor/mock/mock_controller.h",
+      "audio_network_adaptor/mock/mock_controller_manager.h",
+      "audio_network_adaptor/mock/mock_debug_dump_writer.h",
+    ]
+    deps = [
+      ":audio_network_adaptor",
+      "../../test:test_support",
+    ]
+  }
+
+  if (!build_with_chromium) {
+    group("audio_coding_tests") {
+      visibility += webrtc_default_visibility
+      testonly = true
+      public_deps = [  # no-presubmit-check TODO(webrtc:8603)
+        ":acm_receive_test",
+        ":acm_send_test",
+        ":audio_codec_speed_tests",
+        ":audio_decoder_unittests",
+        ":audio_decoder_unittests",
+        ":g711_test",
+        ":g722_test",
+        ":ilbc_test",
+        ":neteq_ilbc_quality_test",
+        ":neteq_opus_quality_test",
+        ":neteq_pcm16b_quality_test",
+        ":neteq_pcmu_quality_test",
+        ":neteq_speed_test",
+        ":rtp_analyze",
+        ":rtp_encode",
+        ":rtp_jitter",
+        ":rtpcat",
+        ":webrtc_opus_fec_test",
+      ]
+      if (rtc_enable_protobuf) {
+        public_deps +=  # no-presubmit-check TODO(webrtc:8603)
+            [ ":neteq_rtpplay" ]
+      }
+    }
+  }
+
+  rtc_library("audio_coding_modules_tests") {
+    testonly = true
+    visibility += webrtc_default_visibility
+
+    sources = [
+      "test/Channel.cc",
+      "test/Channel.h",
+      "test/EncodeDecodeTest.cc",
+      "test/EncodeDecodeTest.h",
+      "test/PacketLossTest.cc",
+      "test/PacketLossTest.h",
+      "test/RTPFile.cc",
+      "test/RTPFile.h",
+      "test/TestAllCodecs.cc",
+      "test/TestAllCodecs.h",
+      "test/TestRedFec.cc",
+      "test/TestRedFec.h",
+      "test/TestVADDTX.cc",
+      "test/TestVADDTX.h",
+      "test/Tester.cc",
+      "test/TwoWayCommunication.cc",
+      "test/TwoWayCommunication.h",
+      "test/target_delay_unittest.cc",
+    ]
+    deps = [
+      ":audio_coding",
+      ":audio_coding_module_typedefs",
+      ":audio_coding_modules_tests_shared",
+      ":audio_encoder_cng",
+      ":pcm16b_c",
+      ":red",
+      ":webrtc_opus_wrapper",
+      "..:module_api",
+      "../../api:rtp_headers",
+      "../../api/audio:audio_frame_api",
+      "../../api/audio_codecs:audio_codecs_api",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
+      "../../api/audio_codecs:builtin_audio_encoder_factory",
+      "../../api/audio_codecs/L16:audio_decoder_L16",
+      "../../api/audio_codecs/L16:audio_encoder_L16",
+      "../../api/audio_codecs/g711:audio_decoder_g711",
+      "../../api/audio_codecs/g711:audio_encoder_g711",
+      "../../api/audio_codecs/g722:audio_decoder_g722",
+      "../../api/audio_codecs/g722:audio_encoder_g722",
+      "../../api/audio_codecs/ilbc:audio_decoder_ilbc",
+      "../../api/audio_codecs/ilbc:audio_encoder_ilbc",
+      "../../api/audio_codecs/opus:audio_decoder_opus",
+      "../../api/audio_codecs/opus:audio_encoder_opus",
+      "../../common_audio",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:stringutils",
+      "../../rtc_base:timeutils",
+      "../../rtc_base/synchronization:mutex",
+      "../../test:fileutils",
+      "../../test:scoped_key_value_config",
+      "../../test:test_support",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+    defines = audio_coding_defines
+  }
+
+  rtc_library("audio_coding_perf_tests") {
+    testonly = true
+    visibility += webrtc_default_visibility
+
+    sources = [
+      "codecs/opus/opus_complexity_unittest.cc",
+      "neteq/test/neteq_performance_unittest.cc",
+    ]
+    deps = [
+      ":neteq_test_support",
+      ":neteq_test_tools",
+      "../../api/audio_codecs/opus:audio_encoder_opus",
+      "../../api/test/metrics:global_metrics_logger_and_exporter",
+      "../../api/test/metrics:metric",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:timeutils",
+      "../../system_wrappers",
+      "../../system_wrappers:field_trial",
+      "../../test:fileutils",
+      "../../test:test_support",
+    ]
+  }
+
+  rtc_library("acm_receive_test") {
+    testonly = true
+    sources = [
+      "acm2/acm_receive_test.cc",
+      "acm2/acm_receive_test.h",
+    ]
+
+    defines = audio_coding_defines
+
+    deps = [
+      ":audio_coding",
+      ":neteq_tools",
+      ":neteq_tools_minimal",
+      "../../api:scoped_refptr",
+      "../../api/audio_codecs:audio_codecs_api",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
+      "../../test:test_support",
+      "//testing/gtest",
+    ]
+
+    deps += audio_coding_deps
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  rtc_library("acm_send_test") {
+    testonly = true
+    sources = [
+      "acm2/acm_send_test.cc",
+      "acm2/acm_send_test.h",
+    ]
+
+    defines = audio_coding_defines
+
+    deps = [
+      ":audio_coding",
+      ":neteq_input_audio_tools",
+      ":neteq_tools",
+      ":neteq_tools_minimal",
+      "../../api/audio:audio_frame_api",
+      "../../api/audio_codecs:audio_codecs_api",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
+      "../../api/audio_codecs:builtin_audio_encoder_factory",
+      "../../rtc_base:checks",
+      "../../rtc_base:stringutils",
+      "../../test:test_support",
+      "//testing/gtest",
+    ]
+    deps += audio_coding_deps
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  if (!build_with_chromium) {
+    audio_decoder_unittests_resources =
+        [ "../../resources/audio_coding/testfile32kHz.pcm" ]
+
+    if (is_ios) {
+      bundle_data("audio_decoder_unittests_bundle_data") {
+        testonly = true
+        sources = audio_decoder_unittests_resources
+        outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+      }
+    }
+
+    rtc_test("audio_decoder_unittests") {
+      testonly = true
+      sources = [ "neteq/audio_decoder_unittest.cc" ]
+
+      defines = neteq_defines
+
+      deps = [
+               ":ilbc",
+               ":neteq",
+               ":neteq_input_audio_tools",
+               ":neteq_tools",
+               "../../api/audio_codecs:audio_codecs_api",
+               "../../api/audio_codecs/opus:audio_encoder_opus",
+               "../../common_audio",
+               "../../rtc_base/system:arch",
+               "../../test:fileutils",
+               "../../test:test_main",
+               "../../test:test_support",
+               "//testing/gtest",
+             ] + audio_coding_deps
+
+      data = audio_decoder_unittests_resources
+
+      if (is_android) {
+        use_default_launcher = false
+        deps += [
+          "//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
+          "//testing/android/native_test:native_test_java",
+          "//testing/android/native_test:native_test_support",
+        ]
+        shard_timeout = 900
+      }
+      if (is_ios) {
+        deps += [ ":audio_decoder_unittests_bundle_data" ]
+      }
+    }
+  }
+
+  if (rtc_enable_protobuf) {
+    rtc_library("neteq_test_factory") {
+      testonly = true
+      visibility += webrtc_default_visibility
+      defines = audio_codec_defines
+      deps = [
+        ":neteq_input_audio_tools",
+        ":neteq_tools",
+        ":neteq_tools_minimal",
+        "../../rtc_base:checks",
+        "../../rtc_base:refcount",
+        "../../test:fileutils",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/strings",
+        "//third_party/abseil-cpp/absl/types:optional",
+      ]
+      sources = [
+        "neteq/tools/neteq_test_factory.cc",
+        "neteq/tools/neteq_test_factory.h",
+      ]
+
+      deps += [
+        ":neteq",
+        ":neteq_test_tools",
+        "../../api/audio_codecs:builtin_audio_decoder_factory",
+        "../../api/neteq:neteq_api",
+        "../../test:audio_test_common",
+        "../../test:field_trial",
+        "../../test:test_support",
+      ]
+    }
+  }
+
+  if (rtc_enable_protobuf && !build_with_chromium) {
+    rtc_executable("neteq_rtpplay") {
+      testonly = true
+      visibility += [ "*" ]
+      defines = []
+      deps = [
+        ":neteq_test_factory",
+        ":neteq_test_tools",
+        ":neteq_tools_minimal",
+        "../../rtc_base:stringutils",
+        "../../system_wrappers:field_trial",
+        "../../test:field_trial",
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/flags:parse",
+        "//third_party/abseil-cpp/absl/strings",
+        "//third_party/abseil-cpp/absl/types:optional",
+      ]
+      sources = [ "neteq/tools/neteq_rtpplay.cc" ]
+    }
+  }
+
+  if (!build_with_chromium) {
+    audio_codec_speed_tests_resources = [
+      "//resources/audio_coding/music_stereo_48kHz.pcm",
+      "//resources/audio_coding/speech_mono_16kHz.pcm",
+      "//resources/audio_coding/speech_mono_32_48kHz.pcm",
+    ]
+
+    if (is_ios) {
+      bundle_data("audio_codec_speed_tests_data") {
+        testonly = true
+        sources = audio_codec_speed_tests_resources
+        outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+      }
+    }
+
+    rtc_test("audio_codec_speed_tests") {
+      testonly = true
+      defines = []
+      deps = [
+        "../../rtc_base:macromagic",
+        "../../test:fileutils",
+      ]
+      sources = [
+        "codecs/opus/opus_speed_test.cc",
+        "codecs/tools/audio_codec_speed_test.cc",
+        "codecs/tools/audio_codec_speed_test.h",
+      ]
+
+      data = audio_codec_speed_tests_resources
+
+      if (is_android) {
+        use_default_launcher = false
+        deps += [
+          "//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
+          "//testing/android/native_test:native_test_java",
+          "//testing/android/native_test:native_test_support",
+        ]
+        shard_timeout = 900
+      }
+
+      if (is_ios) {
+        deps += [ ":audio_codec_speed_tests_data" ]
+      }
+
+      deps += [
+        ":webrtc_opus",
+        "../../rtc_base:checks",
+        "../../test:test_main",
+        "../../test:test_support",
+        "../audio_processing",
+        "//testing/gtest",
+      ]
+    }
+  }
+
+  rtc_library("neteq_test_support") {
+    testonly = true
+    sources = [
+      "neteq/tools/neteq_performance_test.cc",
+      "neteq/tools/neteq_performance_test.h",
+    ]
+
+    deps = [
+      ":default_neteq_factory",
+      ":neteq",
+      ":neteq_test_tools",
+      ":pcm16b",
+      "../../api/audio:audio_frame_api",
+      "../../api/audio_codecs:audio_codecs_api",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
+      "../../api/neteq:neteq_api",
+      "../../rtc_base:checks",
+      "../../system_wrappers",
+      "../../test:fileutils",
+      "../../test:test_support",
+      "//testing/gtest",
+    ]
+  }
+
+  if (!build_with_chromium) {
+    rtc_library("neteq_quality_test_support") {
+      testonly = true
+      sources = [
+        "neteq/tools/neteq_quality_test.cc",
+        "neteq/tools/neteq_quality_test.h",
+      ]
+
+      deps = [
+        ":default_neteq_factory",
+        ":neteq",
+        ":neteq_input_audio_tools",
+        ":neteq_test_tools",
+        ":neteq_tools_minimal",
+        "../../api/audio_codecs:builtin_audio_decoder_factory",
+        "../../api/neteq:neteq_api",
+        "../../rtc_base:checks",
+        "../../rtc_base:stringutils",
+        "../../system_wrappers",
+        "../../test:fileutils",
+        "../../test:test_support",
+        "//testing/gtest",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/strings",
+      ]
+    }
+
+    rtc_executable("rtp_encode") {
+      testonly = true
+
+      deps = [
+        ":audio_coding",
+        ":audio_encoder_cng",
+        ":neteq_input_audio_tools",
+        "../../api/audio:audio_frame_api",
+        "../../api/audio_codecs/L16:audio_encoder_L16",
+        "../../api/audio_codecs/g711:audio_encoder_g711",
+        "../../api/audio_codecs/g722:audio_encoder_g722",
+        "../../api/audio_codecs/ilbc:audio_encoder_ilbc",
+        "../../api/audio_codecs/opus:audio_encoder_opus",
+        "../../rtc_base:safe_conversions",
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/flags:parse",
+        "//third_party/abseil-cpp/absl/memory",
+      ]
+
+      deps += audio_coding_deps
+
+      sources = [ "neteq/tools/rtp_encode.cc" ]
+
+      defines = audio_coding_defines
+    }
+
+    rtc_executable("rtp_jitter") {
+      testonly = true
+
+      deps = [
+        "../../api:array_view",
+        "../../rtc_base:buffer",
+        "../rtp_rtcp:rtp_rtcp_format",
+      ]
+
+      deps += audio_coding_deps
+
+      sources = [ "neteq/tools/rtp_jitter.cc" ]
+
+      defines = audio_coding_defines
+    }
+
+    rtc_executable("rtpcat") {
+      testonly = true
+
+      sources = [ "neteq/tools/rtpcat.cc" ]
+
+      deps = [
+        "../../rtc_base:checks",
+        "../../test:rtp_test_utils",
+        "//testing/gtest",
+      ]
+    }
+
+    rtc_executable("rtp_analyze") {
+      testonly = true
+
+      sources = [ "neteq/tools/rtp_analyze.cc" ]
+
+      deps = [
+        ":neteq",
+        ":neteq_test_tools",
+        ":neteq_tools_minimal",
+        ":pcm16b",
+        "//testing/gtest",
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/flags:parse",
+      ]
+    }
+
+    rtc_executable("neteq_opus_quality_test") {
+      testonly = true
+
+      sources = [ "neteq/test/neteq_opus_quality_test.cc" ]
+
+      deps = [
+        ":neteq",
+        ":neteq_quality_test_support",
+        ":neteq_tools",
+        ":webrtc_opus",
+        "../../test:test_main",
+        "//testing/gtest",
+        "//third_party/abseil-cpp/absl/flags:flag",
+      ]
+    }
+
+    rtc_executable("neteq_speed_test") {
+      testonly = true
+
+      sources = [ "neteq/test/neteq_speed_test.cc" ]
+
+      deps = [
+        ":neteq",
+        ":neteq_test_support",
+        "../../rtc_base:checks",
+        "../../test:test_support",
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/flags:parse",
+      ]
+    }
+
+    rtc_executable("neteq_ilbc_quality_test") {
+      testonly = true
+
+      sources = [ "neteq/test/neteq_ilbc_quality_test.cc" ]
+
+      deps = [
+        ":ilbc",
+        ":neteq",
+        ":neteq_quality_test_support",
+        ":neteq_tools",
+        "../../rtc_base:checks",
+        "../../rtc_base:safe_conversions",
+        "../../test:fileutils",
+        "../../test:test_main",
+        "//testing/gtest",
+        "//third_party/abseil-cpp/absl/flags:flag",
+      ]
+    }
+
+    rtc_executable("neteq_pcmu_quality_test") {
+      testonly = true
+
+      sources = [ "neteq/test/neteq_pcmu_quality_test.cc" ]
+
+      deps = [
+        ":g711",
+        ":neteq",
+        ":neteq_quality_test_support",
+        "../../rtc_base:checks",
+        "../../rtc_base:safe_conversions",
+        "../../test:fileutils",
+        "../../test:test_main",
+        "//testing/gtest",
+        "//third_party/abseil-cpp/absl/flags:flag",
+      ]
+    }
+
+    rtc_executable("neteq_pcm16b_quality_test") {
+      testonly = true
+
+      sources = [ "neteq/test/neteq_pcm16b_quality_test.cc" ]
+
+      deps = [
+        ":neteq",
+        ":neteq_quality_test_support",
+        ":pcm16b",
+        "../../rtc_base:checks",
+        "../../rtc_base:safe_conversions",
+        "../../test:fileutils",
+        "../../test:test_main",
+        "//testing/gtest",
+        "//third_party/abseil-cpp/absl/flags:flag",
+      ]
+    }
+  }
+
+  rtc_executable("g711_test") {
+    testonly = true
+
+    sources = [ "codecs/g711/test/testG711.cc" ]
+
+    deps = [ ":g711" ]
+  }
+
+  rtc_executable("g722_test") {
+    testonly = true
+
+    sources = [ "codecs/g722/test/testG722.cc" ]
+
+    deps = [ ":g722" ]
+  }
+
+  if (!build_with_chromium) {
+    rtc_executable("ilbc_test") {
+      testonly = true
+
+      sources = [ "codecs/ilbc/test/iLBC_test.c" ]
+
+      deps = [ ":ilbc" ]
+    }
+
+    rtc_executable("webrtc_opus_fec_test") {
+      testonly = true
+
+      sources = [ "codecs/opus/opus_fec_test.cc" ]
+
+      deps = [
+        ":webrtc_opus",
+        "../../common_audio",
+        "../../rtc_base:macromagic",
+        "../../test:fileutils",
+        "../../test:test_main",
+        "../../test:test_support",
+        "//testing/gtest",
+      ]
+    }
+
+    rtc_library("audio_coding_unittests") {
+      testonly = true
+      visibility += webrtc_default_visibility
+
+      sources = [
+        "acm2/acm_receiver_unittest.cc",
+        "acm2/acm_remixing_unittest.cc",
+        "acm2/audio_coding_module_unittest.cc",
+        "acm2/call_statistics_unittest.cc",
+        "audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
+        "audio_network_adaptor/bitrate_controller_unittest.cc",
+        "audio_network_adaptor/channel_controller_unittest.cc",
+        "audio_network_adaptor/controller_manager_unittest.cc",
+        "audio_network_adaptor/dtx_controller_unittest.cc",
+        "audio_network_adaptor/event_log_writer_unittest.cc",
+        "audio_network_adaptor/fec_controller_plr_based_unittest.cc",
+        "audio_network_adaptor/frame_length_controller_unittest.cc",
+        "audio_network_adaptor/frame_length_controller_v2_unittest.cc",
+        "audio_network_adaptor/util/threshold_curve_unittest.cc",
+        "codecs/builtin_audio_decoder_factory_unittest.cc",
+        "codecs/builtin_audio_encoder_factory_unittest.cc",
+        "codecs/cng/audio_encoder_cng_unittest.cc",
+        "codecs/cng/cng_unittest.cc",
+        "codecs/ilbc/ilbc_unittest.cc",
+        "codecs/legacy_encoded_audio_frame_unittest.cc",
+        "codecs/opus/audio_decoder_multi_channel_opus_unittest.cc",
+        "codecs/opus/audio_encoder_multi_channel_opus_unittest.cc",
+        "codecs/opus/audio_encoder_opus_unittest.cc",
+        "codecs/opus/opus_bandwidth_unittest.cc",
+        "codecs/opus/opus_unittest.cc",
+        "codecs/red/audio_encoder_copy_red_unittest.cc",
+        "neteq/audio_multi_vector_unittest.cc",
+        "neteq/audio_vector_unittest.cc",
+        "neteq/background_noise_unittest.cc",
+        "neteq/buffer_level_filter_unittest.cc",
+        "neteq/comfort_noise_unittest.cc",
+        "neteq/decision_logic_unittest.cc",
+        "neteq/decoder_database_unittest.cc",
+        "neteq/delay_manager_unittest.cc",
+        "neteq/dsp_helper_unittest.cc",
+        "neteq/dtmf_buffer_unittest.cc",
+        "neteq/dtmf_tone_generator_unittest.cc",
+        "neteq/expand_unittest.cc",
+        "neteq/histogram_unittest.cc",
+        "neteq/merge_unittest.cc",
+        "neteq/mock/mock_buffer_level_filter.h",
+        "neteq/mock/mock_decoder_database.h",
+        "neteq/mock/mock_delay_manager.h",
+        "neteq/mock/mock_dtmf_buffer.h",
+        "neteq/mock/mock_dtmf_tone_generator.h",
+        "neteq/mock/mock_expand.h",
+        "neteq/mock/mock_histogram.h",
+        "neteq/mock/mock_neteq_controller.h",
+        "neteq/mock/mock_packet_buffer.h",
+        "neteq/mock/mock_red_payload_splitter.h",
+        "neteq/mock/mock_statistics_calculator.h",
+        "neteq/nack_tracker_unittest.cc",
+        "neteq/neteq_decoder_plc_unittest.cc",
+        "neteq/neteq_impl_unittest.cc",
+        "neteq/neteq_network_stats_unittest.cc",
+        "neteq/neteq_stereo_unittest.cc",
+        "neteq/neteq_unittest.cc",
+        "neteq/normal_unittest.cc",
+        "neteq/packet_arrival_history_unittest.cc",
+        "neteq/packet_buffer_unittest.cc",
+        "neteq/post_decode_vad_unittest.cc",
+        "neteq/random_vector_unittest.cc",
+        "neteq/red_payload_splitter_unittest.cc",
+        "neteq/reorder_optimizer_unittest.cc",
+        "neteq/statistics_calculator_unittest.cc",
+        "neteq/sync_buffer_unittest.cc",
+        "neteq/time_stretch_unittest.cc",
+        "neteq/timestamp_scaler_unittest.cc",
+        "neteq/tools/input_audio_file_unittest.cc",
+        "neteq/tools/packet_unittest.cc",
+        "neteq/underrun_optimizer_unittest.cc",
+      ]
+
+      deps = [
+        ":acm_receive_test",
+        ":acm_send_test",
+        ":audio_coding",
+        ":audio_coding_module_typedefs",
+        ":audio_coding_modules_tests_shared",
+        ":audio_coding_opus_common",
+        ":audio_encoder_cng",
+        ":audio_network_adaptor",
+        ":default_neteq_factory",
+        ":g711",
+        ":ilbc",
+        ":legacy_encoded_audio_frame",
+        ":mocks",
+        ":neteq",
+        ":neteq_input_audio_tools",
+        ":neteq_test_support",
+        ":neteq_test_tools",
+        ":neteq_tools",
+        ":neteq_tools_minimal",
+        ":pcm16b",
+        ":red",
+        ":webrtc_cng",
+        ":webrtc_opus",
+        "..:module_api",
+        "..:module_api_public",
+        "../../api:array_view",
+        "../../api/audio:audio_frame_api",
+        "../../api/audio_codecs:audio_codecs_api",
+        "../../api/audio_codecs:builtin_audio_decoder_factory",
+        "../../api/audio_codecs:builtin_audio_encoder_factory",
+        "../../api/audio_codecs/opus:audio_decoder_multiopus",
+        "../../api/audio_codecs/opus:audio_decoder_opus",
+        "../../api/audio_codecs/opus:audio_encoder_multiopus",
+        "../../api/audio_codecs/opus:audio_encoder_opus",
+        "../../api/neteq:default_neteq_controller_factory",
+        "../../api/neteq:neteq_api",
+        "../../api/neteq:neteq_controller_api",
+        "../../api/neteq:tick_timer",
+        "../../api/neteq:tick_timer_unittest",
+        "../../api/rtc_event_log",
+        "../../common_audio",
+        "../../common_audio:common_audio_c",
+        "../../common_audio:mock_common_audio",
+        "../../logging:mocks",
+        "../../logging:rtc_event_audio",
+        "../../modules/rtp_rtcp:rtp_rtcp_format",
+        "../../rtc_base:checks",
+        "../../rtc_base:ignore_wundef",
+        "../../rtc_base:macromagic",
+        "../../rtc_base:platform_thread",
+        "../../rtc_base:refcount",
+        "../../rtc_base:rtc_base_tests_utils",
+        "../../rtc_base:rtc_event",
+        "../../rtc_base:safe_conversions",
+        "../../rtc_base:sanitizer",
+        "../../rtc_base:ssl",
+        "../../rtc_base:stringutils",
+        "../../rtc_base:timeutils",
+        "../../rtc_base/synchronization:mutex",
+        "../../rtc_base/system:arch",
+        "../../system_wrappers",
+        "../../test:audio_codec_mocks",
+        "../../test:audio_test_common",
+        "../../test:field_trial",
+        "../../test:fileutils",
+        "../../test:rtc_expect_death",
+        "../../test:rtp_test_utils",
+        "../../test:scoped_key_value_config",
+        "../../test:test_common",
+        "../../test:test_support",
+        "codecs/opus/test",
+        "codecs/opus/test:test_unittest",
+        "//testing/gtest",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/memory",
+        "//third_party/abseil-cpp/absl/strings",
+        "//third_party/abseil-cpp/absl/types:optional",
+      ]
+
+      defines = audio_coding_defines
+
+      if (rtc_enable_protobuf) {
+        defines += [ "WEBRTC_NETEQ_UNITTEST_BITEXACT" ]
+        deps += [
+          ":ana_config_proto",
+          ":neteq_unittest_proto",
+        ]
+      }
+    }
+  }
+}
+
+# For backwards compatibility only! Use
+# webrtc/api/audio_codecs:audio_codecs_api instead.
+# TODO(kwiberg): Remove this.
+rtc_source_set("audio_decoder_interface") {
+  visibility += [ "*" ]
+  sources = [ "codecs/audio_decoder.h" ]
+  deps = [ "../../api/audio_codecs:audio_codecs_api" ]
+}
+
+# For backwards compatibility only! Use
+# webrtc/api/audio_codecs:audio_codecs_api instead.
+# TODO(ossu): Remove this.
+rtc_source_set("audio_encoder_interface") {
+  visibility += [ "*" ]
+  sources = [ "codecs/audio_encoder.h" ]
+  deps = [ "../../api/audio_codecs:audio_codecs_api" ]
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/DEPS b/third_party/libwebrtc/modules/audio_coding/DEPS
new file mode 100644
index 0000000000..3dc9624a4b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/DEPS
@@ -0,0 +1,7 @@
+include_rules = [
+  "+call",
+  "+common_audio",
+  "+logging/rtc_event_log",
+  "+audio_coding/neteq/neteq_unittest.pb.h",  # Different path.
+  "+system_wrappers",
+]
diff --git a/third_party/libwebrtc/modules/audio_coding/OWNERS b/third_party/libwebrtc/modules/audio_coding/OWNERS
new file mode 100644
index 0000000000..c27c2a8d2d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/OWNERS
@@ -0,0 +1,4 @@
+henrik.lundin@webrtc.org
+minyue@webrtc.org
+ivoc@webrtc.org
+jakobi@webrtc.org
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receive_test.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receive_test.cc
new file mode 100644
index 0000000000..8bc76cd2af
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receive_test.cc
@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/acm_receive_test.h"
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+AudioCodingModule::Config MakeAcmConfig(
+    Clock* clock,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
+  AudioCodingModule::Config config;
+  config.clock = clock;
+  config.decoder_factory = std::move(decoder_factory);
+  return config;
+}
+}  // namespace
+
+AcmReceiveTestOldApi::AcmReceiveTestOldApi(
+    PacketSource* packet_source,
+    AudioSink* audio_sink,
+    int output_freq_hz,
+    NumOutputChannels exptected_output_channels,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
+    : clock_(0),
+      acm_(webrtc::AudioCodingModule::Create(
+          MakeAcmConfig(&clock_, std::move(decoder_factory)))),
+      packet_source_(packet_source),
+      audio_sink_(audio_sink),
+      output_freq_hz_(output_freq_hz),
+      exptected_output_channels_(exptected_output_channels) {}
+
+AcmReceiveTestOldApi::~AcmReceiveTestOldApi() = default;
+
+void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
+  acm_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
+                          {104, {"ISAC", 32000, 1}},
+                          {107, {"L16", 8000, 1}},
+                          {108, {"L16", 16000, 1}},
+                          {109, {"L16", 32000, 1}},
+                          {111, {"L16", 8000, 2}},
+                          {112, {"L16", 16000, 2}},
+                          {113, {"L16", 32000, 2}},
+                          {0, {"PCMU", 8000, 1}},
+                          {110, {"PCMU", 8000, 2}},
+                          {8, {"PCMA", 8000, 1}},
+                          {118, {"PCMA", 8000, 2}},
+                          {102, {"ILBC", 8000, 1}},
+                          {9, {"G722", 8000, 1}},
+                          {119, {"G722", 8000, 2}},
+                          {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+                          {13, {"CN", 8000, 1}},
+                          {98, {"CN", 16000, 1}},
+                          {99, {"CN", 32000, 1}}});
+}
+
+// Remaps payload types from ACM's default to those used in the resource file
+// neteq_universal_new.rtp.
+void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
+  acm_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
+                          {104, {"ISAC", 32000, 1}},
+                          {93, {"L16", 8000, 1}},
+                          {94, {"L16", 16000, 1}},
+                          {95, {"L16", 32000, 1}},
+                          {0, {"PCMU", 8000, 1}},
+                          {8, {"PCMA", 8000, 1}},
+                          {102, {"ILBC", 8000, 1}},
+                          {9, {"G722", 8000, 1}},
+                          {120, {"OPUS", 48000, 2}},
+                          {13, {"CN", 8000, 1}},
+                          {98, {"CN", 16000, 1}},
+                          {99, {"CN", 32000, 1}}});
+}
+
+void AcmReceiveTestOldApi::Run() {
+  for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
+       packet = packet_source_->NextPacket()) {
+    // Pull audio until time to insert packet.
+    while (clock_.TimeInMilliseconds() < packet->time_ms()) {
+      AudioFrame output_frame;
+      bool muted;
+      EXPECT_EQ(0,
+                acm_->PlayoutData10Ms(output_freq_hz_, &output_frame, &muted));
+      ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
+      ASSERT_FALSE(muted);
+      const size_t samples_per_block =
+          static_cast<size_t>(output_freq_hz_ * 10 / 1000);
+      EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
+      if (exptected_output_channels_ != kArbitraryChannels) {
+        if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
+          // Don't check number of channels for PLC output, since each test run
+          // usually starts with a short period of mono PLC before decoding the
+          // first packet.
+        } else {
+          EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
+        }
+      }
+      ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
+      clock_.AdvanceTimeMilliseconds(10);
+      AfterGetAudio();
+    }
+
+    EXPECT_EQ(0, acm_->IncomingPacket(
+                     packet->payload(),
+                     static_cast<int32_t>(packet->payload_length_bytes()),
+                     packet->header()))
+        << "Failure when inserting packet:" << std::endl
+        << "  PT = " << static_cast<int>(packet->header().payloadType)
+        << std::endl
+        << "  TS = " << packet->header().timestamp << std::endl
+        << "  SN = " << packet->header().sequenceNumber;
+  }
+}
+
+AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
+    PacketSource* packet_source,
+    AudioSink* audio_sink,
+    int output_freq_hz_1,
+    int output_freq_hz_2,
+    int toggle_period_ms,
+    NumOutputChannels exptected_output_channels)
+    : AcmReceiveTestOldApi(packet_source,
+                           audio_sink,
+                           output_freq_hz_1,
+                           exptected_output_channels,
+                           CreateBuiltinAudioDecoderFactory()),
+      output_freq_hz_1_(output_freq_hz_1),
+      output_freq_hz_2_(output_freq_hz_2),
+      toggle_period_ms_(toggle_period_ms),
+      last_toggle_time_ms_(clock_.TimeInMilliseconds()) {}
+
+void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
+  if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
+    output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
+                          ? output_freq_hz_2_
+                          : output_freq_hz_1_;
+    last_toggle_time_ms_ = clock_.TimeInMilliseconds();
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receive_test.h b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receive_test.h
new file mode 100644
index 0000000000..2095ef9025
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receive_test.h
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
+
+#include <stddef.h>  // for size_t
+
+#include <memory>
+#include <string>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/scoped_refptr.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+class AudioCodingModule;
+class AudioDecoder;
+
+namespace test {
+class AudioSink;
+class PacketSource;
+
+class AcmReceiveTestOldApi {
+ public:
+  enum NumOutputChannels : size_t {
+    kArbitraryChannels = 0,
+    kMonoOutput = 1,
+    kStereoOutput = 2,
+    kQuadOutput = 4
+  };
+
+  AcmReceiveTestOldApi(PacketSource* packet_source,
+                       AudioSink* audio_sink,
+                       int output_freq_hz,
+                       NumOutputChannels exptected_output_channels,
+                       rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
+  virtual ~AcmReceiveTestOldApi();
+
+  AcmReceiveTestOldApi(const AcmReceiveTestOldApi&) = delete;
+  AcmReceiveTestOldApi& operator=(const AcmReceiveTestOldApi&) = delete;
+
+  // Registers the codecs with default parameters from ACM.
+  void RegisterDefaultCodecs();
+
+  // Registers codecs with payload types matching the pre-encoded NetEq test
+  // files.
+  void RegisterNetEqTestCodecs();
+
+  // Runs the test and returns true if successful.
+  void Run();
+
+  AudioCodingModule* get_acm() { return acm_.get(); }
+
+ protected:
+  // Method is called after each block of output audio is received from ACM.
+  virtual void AfterGetAudio() {}
+
+  SimulatedClock clock_;
+  std::unique_ptr<AudioCodingModule> acm_;
+  PacketSource* packet_source_;
+  AudioSink* audio_sink_;
+  int output_freq_hz_;
+  NumOutputChannels exptected_output_channels_;
+};
+
+// This test toggles the output frequency every `toggle_period_ms`. The test
+// starts with `output_freq_hz_1`. Except for the toggling, it does the same
+// thing as AcmReceiveTestOldApi.
+class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
+ public:
+  AcmReceiveTestToggleOutputFreqOldApi(
+      PacketSource* packet_source,
+      AudioSink* audio_sink,
+      int output_freq_hz_1,
+      int output_freq_hz_2,
+      int toggle_period_ms,
+      NumOutputChannels exptected_output_channels);
+
+ protected:
+  void AfterGetAudio() override;
+
+  const int output_freq_hz_1_;
+  const int output_freq_hz_2_;
+  const int toggle_period_ms_;
+  int64_t last_toggle_time_ms_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc
new file mode 100644
index 0000000000..b078af1d2d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -0,0 +1,352 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/acm_receiver.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/acm2/call_statistics.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/audio_format_to_string.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+namespace {
+
+std::unique_ptr<NetEq> CreateNetEq(
+    NetEqFactory* neteq_factory,
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+  if (neteq_factory) {
+    return neteq_factory->CreateNetEq(config, decoder_factory, clock);
+  }
+  return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
+}
+
+}  // namespace
+
+AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
+    : last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
+      neteq_(CreateNetEq(config.neteq_factory,
+                         config.neteq_config,
+                         config.clock,
+                         config.decoder_factory)),
+      clock_(config.clock),
+      resampled_last_output_frame_(true) {
+  RTC_DCHECK(clock_);
+  memset(last_audio_buffer_.get(), 0,
+         sizeof(int16_t) * AudioFrame::kMaxDataSizeSamples);
+}
+
+AcmReceiver::~AcmReceiver() = default;
+
+int AcmReceiver::SetMinimumDelay(int delay_ms) {
+  if (neteq_->SetMinimumDelay(delay_ms))
+    return 0;
+  RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
+  return -1;
+}
+
+int AcmReceiver::SetMaximumDelay(int delay_ms) {
+  if (neteq_->SetMaximumDelay(delay_ms))
+    return 0;
+  RTC_LOG(LS_ERROR) << "AcmReceiver::SetExtraDelay " << delay_ms;
+  return -1;
+}
+
+bool AcmReceiver::SetBaseMinimumDelayMs(int delay_ms) {
+  return neteq_->SetBaseMinimumDelayMs(delay_ms);
+}
+
+int AcmReceiver::GetBaseMinimumDelayMs() const {
+  return neteq_->GetBaseMinimumDelayMs();
+}
+
+absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
+  MutexLock lock(&mutex_);
+  if (!last_decoder_) {
+    return absl::nullopt;
+  }
+  return last_decoder_->sample_rate_hz;
+}
+
+int AcmReceiver::last_output_sample_rate_hz() const {
+  return neteq_->last_output_sample_rate_hz();
+}
+
+int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
+                              rtc::ArrayView<const uint8_t> incoming_payload) {
+  if (incoming_payload.empty()) {
+    neteq_->InsertEmptyPacket(rtp_header);
+    return 0;
+  }
+
+  int payload_type = rtp_header.payloadType;
+  auto format = neteq_->GetDecoderFormat(payload_type);
+  if (format && absl::EqualsIgnoreCase(format->sdp_format.name, "red")) {
+    // This is a RED packet. Get the format of the audio codec.
+    payload_type = incoming_payload[0] & 0x7f;
+    format = neteq_->GetDecoderFormat(payload_type);
+  }
+  if (!format) {
+    RTC_LOG_F(LS_ERROR) << "Payload-type " << payload_type
+                        << " is not registered.";
+    return -1;
+  }
+
+  {
+    MutexLock lock(&mutex_);
+    if (absl::EqualsIgnoreCase(format->sdp_format.name, "cn")) {
+      if (last_decoder_ && last_decoder_->num_channels > 1) {
+        // This is a CNG and the audio codec is not mono, so skip pushing in
+        // packets into NetEq.
+        return 0;
+      }
+    } else {
+      last_decoder_ = DecoderInfo{/*payload_type=*/payload_type,
+                                  /*sample_rate_hz=*/format->sample_rate_hz,
+                                  /*num_channels=*/format->num_channels,
+                                  /*sdp_format=*/std::move(format->sdp_format)};
+    }
+  }  // `mutex_` is released.
+
+  if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) {
+    RTC_LOG(LS_ERROR) << "AcmReceiver::InsertPacket "
+                      << static_cast<int>(rtp_header.payloadType)
+                      << " Failed to insert packet";
+    return -1;
+  }
+  return 0;
+}
+
+int AcmReceiver::GetAudio(int desired_freq_hz,
+                          AudioFrame* audio_frame,
+                          bool* muted) {
+  RTC_DCHECK(muted);
+
+  int current_sample_rate_hz = 0;
+  if (neteq_->GetAudio(audio_frame, muted, &current_sample_rate_hz) !=
+      NetEq::kOK) {
+    RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
+    return -1;
+  }
+
+  RTC_DCHECK_NE(current_sample_rate_hz, 0);
+
+  // Update if resampling is required.
+  const bool need_resampling =
+      (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz);
+
+  // Accessing members, take the lock.
+  MutexLock lock(&mutex_);
+  if (need_resampling && !resampled_last_output_frame_) {
+    // Prime the resampler with the last frame.
+    int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        last_audio_buffer_.get(), current_sample_rate_hz, desired_freq_hz,
+        audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
+        temp_output);
+    if (samples_per_channel_int < 0) {
+      RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - "
+                           "Resampling last_audio_buffer_ failed.";
+      return -1;
+    }
+  }
+
+  // TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output
+  // rate from NetEq changes.
+  if (need_resampling) {
+    // TODO(yujo): handle this more efficiently for muted frames.
+    int samples_per_channel_int = resampler_.Resample10Msec(
+        audio_frame->data(), current_sample_rate_hz, desired_freq_hz,
+        audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
+        audio_frame->mutable_data());
+    if (samples_per_channel_int < 0) {
+      RTC_LOG(LS_ERROR)
+          << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
+      return -1;
+    }
+    audio_frame->samples_per_channel_ =
+        static_cast<size_t>(samples_per_channel_int);
+    audio_frame->sample_rate_hz_ = desired_freq_hz;
+    RTC_DCHECK_EQ(
+        audio_frame->sample_rate_hz_,
+        rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
+    resampled_last_output_frame_ = true;
+  } else {
+    resampled_last_output_frame_ = false;
+    // We might end up here ONLY if codec is changed.
+  }
+
+  // Store current audio in `last_audio_buffer_` for next time.
+  memcpy(last_audio_buffer_.get(), audio_frame->data(),
+         sizeof(int16_t) * audio_frame->samples_per_channel_ *
+             audio_frame->num_channels_);
+
+  call_stats_.DecodedByNetEq(audio_frame->speech_type_, *muted);
+  return 0;
+}
+
+void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+  neteq_->SetCodecs(codecs);
+}
+
+void AcmReceiver::FlushBuffers() {
+  neteq_->FlushBuffers();
+}
+
+void AcmReceiver::RemoveAllCodecs() {
+  MutexLock lock(&mutex_);
+  neteq_->RemoveAllPayloadTypes();
+  last_decoder_ = absl::nullopt;
+}
+
+absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
+  return neteq_->GetPlayoutTimestamp();
+}
+
+int AcmReceiver::FilteredCurrentDelayMs() const {
+  return neteq_->FilteredCurrentDelayMs();
+}
+
+int AcmReceiver::TargetDelayMs() const {
+  return neteq_->TargetDelayMs();
+}
+
+absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
+    const {
+  MutexLock lock(&mutex_);
+  if (!last_decoder_) {
+    return absl::nullopt;
+  }
+  RTC_DCHECK_NE(-1, last_decoder_->payload_type);
+  return std::make_pair(last_decoder_->payload_type, last_decoder_->sdp_format);
+}
+
+void AcmReceiver::GetNetworkStatistics(
+    NetworkStatistics* acm_stat,
+    bool get_and_clear_legacy_stats /* = true */) const {
+  NetEqNetworkStatistics neteq_stat;
+  if (get_and_clear_legacy_stats) {
+    // NetEq function always returns zero, so we don't check the return value.
+    neteq_->NetworkStatistics(&neteq_stat);
+
+    acm_stat->currentExpandRate = neteq_stat.expand_rate;
+    acm_stat->currentSpeechExpandRate = neteq_stat.speech_expand_rate;
+    acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
+    acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
+    acm_stat->currentSecondaryDecodedRate = neteq_stat.secondary_decoded_rate;
+    acm_stat->currentSecondaryDiscardedRate =
+        neteq_stat.secondary_discarded_rate;
+    acm_stat->meanWaitingTimeMs = neteq_stat.mean_waiting_time_ms;
+    acm_stat->maxWaitingTimeMs = neteq_stat.max_waiting_time_ms;
+  } else {
+    neteq_stat = neteq_->CurrentNetworkStatistics();
+    acm_stat->currentExpandRate = 0;
+    acm_stat->currentSpeechExpandRate = 0;
+    acm_stat->currentPreemptiveRate = 0;
+    acm_stat->currentAccelerateRate = 0;
+    acm_stat->currentSecondaryDecodedRate = 0;
+    acm_stat->currentSecondaryDiscardedRate = 0;
+    acm_stat->meanWaitingTimeMs = -1;
+    acm_stat->maxWaitingTimeMs = 1;
+  }
+  acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
+  acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
+  acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found ? true : false;
+
+  NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
+  acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
+  acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
+  acm_stat->silentConcealedSamples =
+      neteq_lifetime_stat.silent_concealed_samples;
+  acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
+  acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
+  acm_stat->jitterBufferTargetDelayMs =
+      neteq_lifetime_stat.jitter_buffer_target_delay_ms;
+  acm_stat->jitterBufferMinimumDelayMs =
+      neteq_lifetime_stat.jitter_buffer_minimum_delay_ms;
+  acm_stat->jitterBufferEmittedCount =
+      neteq_lifetime_stat.jitter_buffer_emitted_count;
+  acm_stat->delayedPacketOutageSamples =
+      neteq_lifetime_stat.delayed_packet_outage_samples;
+  acm_stat->relativePacketArrivalDelayMs =
+      neteq_lifetime_stat.relative_packet_arrival_delay_ms;
+  acm_stat->interruptionCount = neteq_lifetime_stat.interruption_count;
+  acm_stat->totalInterruptionDurationMs =
+      neteq_lifetime_stat.total_interruption_duration_ms;
+  acm_stat->insertedSamplesForDeceleration =
+      neteq_lifetime_stat.inserted_samples_for_deceleration;
+  acm_stat->removedSamplesForAcceleration =
+      neteq_lifetime_stat.removed_samples_for_acceleration;
+  acm_stat->fecPacketsReceived = neteq_lifetime_stat.fec_packets_received;
+  acm_stat->fecPacketsDiscarded = neteq_lifetime_stat.fec_packets_discarded;
+  acm_stat->packetsDiscarded = neteq_lifetime_stat.packets_discarded;
+
+  NetEqOperationsAndState neteq_operations_and_state =
+      neteq_->GetOperationsAndState();
+  acm_stat->packetBufferFlushes =
+      neteq_operations_and_state.packet_buffer_flushes;
+}
+
+int AcmReceiver::EnableNack(size_t max_nack_list_size) {
+  neteq_->EnableNack(max_nack_list_size);
+  return 0;
+}
+
+void AcmReceiver::DisableNack() {
+  neteq_->DisableNack();
+}
+
+std::vector<uint16_t> AcmReceiver::GetNackList(
+    int64_t round_trip_time_ms) const {
+  return neteq_->GetNackList(round_trip_time_ms);
+}
+
+void AcmReceiver::ResetInitialDelay() {
+  neteq_->SetMinimumDelay(0);
+  // TODO(turajs): Should NetEq Buffer be flushed?
+}
+
+uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
+  // Down-cast the time to (32-6)-bit since we only care about
+  // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
+  // We masked 6 most significant bits of 32-bit so there is no overflow in
+  // the conversion from milliseconds to timestamp.
+  const uint32_t now_in_ms =
+      static_cast<uint32_t>(clock_->TimeInMilliseconds() & 0x03ffffff);
+  return static_cast<uint32_t>((decoder_sampling_rate / 1000) * now_in_ms);
+}
+
+void AcmReceiver::GetDecodingCallStatistics(
+    AudioDecodingCallStats* stats) const {
+  MutexLock lock(&mutex_);
+  *stats = call_stats_.GetDecodingStatistics();
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.h b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.h
new file mode 100644
index 0000000000..a61247627f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -0,0 +1,234 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+
+#include <stdint.h>
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+#include <atomic>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/acm2/call_statistics.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class Clock;
+class NetEq;
+struct RTPHeader;
+
+namespace acm2 {
+
+class AcmReceiver {
+ public:
+  // Constructor of the class
+  explicit AcmReceiver(const AudioCodingModule::Config& config);
+
+  // Destructor of the class.
+  ~AcmReceiver();
+
+  //
+  // Inserts a payload with its associated RTP-header into NetEq.
+  //
+  // Input:
+  //   - rtp_header           : RTP header for the incoming payload containing
+  //                            information about payload type, sequence number,
+  //                            timestamp, SSRC and marker bit.
+  //   - incoming_payload     : Incoming audio payload.
+  //   - length_payload       : Length of incoming audio payload in bytes.
+  //
+  // Return value             : 0 if OK.
+  //                           <0 if NetEq returned an error.
+  //
+  int InsertPacket(const RTPHeader& rtp_header,
+                   rtc::ArrayView<const uint8_t> incoming_payload);
+
+  //
+  // Asks NetEq for 10 milliseconds of decoded audio.
+  //
+  // Input:
+  //   -desired_freq_hz       : specifies the sampling rate [Hz] of the output
+  //                            audio. If set -1 indicates to resampling is
+  //                            is required and the audio returned at the
+  //                            sampling rate of the decoder.
+  //
+  // Output:
+  //   -audio_frame           : an audio frame were output data and
+  //                            associated parameters are written to.
+  //   -muted                 : if true, the sample data in audio_frame is not
+  //                            populated, and must be interpreted as all zero.
+  //
+  // Return value             : 0 if OK.
+  //                           -1 if NetEq returned an error.
+  //
+  int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
+
+  // Replace the current set of decoders with the specified set.
+  void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
+
+  //
+  // Sets a minimum delay for packet buffer. The given delay is maintained,
+  // unless channel condition dictates a higher delay.
+  //
+  // Input:
+  //   - delay_ms             : minimum delay in milliseconds.
+  //
+  // Return value             : 0 if OK.
+  //                           <0 if NetEq returned an error.
+  //
+  int SetMinimumDelay(int delay_ms);
+
+  //
+  // Sets a maximum delay [ms] for the packet buffer. The target delay does not
+  // exceed the given value, even if channel condition requires so.
+  //
+  // Input:
+  //   - delay_ms             : maximum delay in milliseconds.
+  //
+  // Return value             : 0 if OK.
+  //                           <0 if NetEq returned an error.
+  //
+  int SetMaximumDelay(int delay_ms);
+
+  // Sets a base minimum delay in milliseconds for the packet buffer.
+  // Base minimum delay sets lower bound minimum delay value which
+  // is set via SetMinimumDelay.
+  //
+  // Returns true if value was successfully set, false overwise.
+  bool SetBaseMinimumDelayMs(int delay_ms);
+
+  // Returns current value of base minimum delay in milliseconds.
+  int GetBaseMinimumDelayMs() const;
+
+  //
+  // Resets the initial delay to zero.
+  //
+  void ResetInitialDelay();
+
+  // Returns the sample rate of the decoder associated with the last incoming
+  // packet. If no packet of a registered non-CNG codec has been received, the
+  // return value is empty. Also, if the decoder was unregistered since the last
+  // packet was inserted, the return value is empty.
+  absl::optional<int> last_packet_sample_rate_hz() const;
+
+  // Returns last_output_sample_rate_hz from the NetEq instance.
+  int last_output_sample_rate_hz() const;
+
+  //
+  // Get the current network statistics from NetEq.
+  //
+  // Output:
+  //   - statistics           : The current network statistics.
+  //
+  void GetNetworkStatistics(NetworkStatistics* statistics,
+                            bool get_and_clear_legacy_stats = true) const;
+
+  //
+  // Flushes the NetEq packet and speech buffers.
+  //
+  void FlushBuffers();
+
+  //
+  // Remove all registered codecs.
+  //
+  void RemoveAllCodecs();
+
+  // Returns the RTP timestamp for the last sample delivered by GetAudio().
+  // The return value will be empty if no valid timestamp is available.
+  absl::optional<uint32_t> GetPlayoutTimestamp();
+
+  // Returns the current total delay from NetEq (packet buffer and sync buffer)
+  // in ms, with smoothing applied to even out short-time fluctuations due to
+  // jitter. The packet buffer part of the delay is not updated during DTX/CNG
+  // periods.
+  //
+  int FilteredCurrentDelayMs() const;
+
+  // Returns the current target delay for NetEq in ms.
+  //
+  int TargetDelayMs() const;
+
+  //
+  // Get payload type and format of the last non-CNG/non-DTMF received payload.
+  // If no non-CNG/non-DTMF packet is received absl::nullopt is returned.
+  //
+  absl::optional<std::pair<int, SdpAudioFormat>> LastDecoder() const;
+
+  //
+  // Enable NACK and set the maximum size of the NACK list. If NACK is already
+  // enabled then the maximum NACK list size is modified accordingly.
+  //
+  // If the sequence number of last received packet is N, the sequence numbers
+  // of NACK list are in the range of [N - `max_nack_list_size`, N).
+  //
+  // `max_nack_list_size` should be positive (none zero) and less than or
+  // equal to `Nack::kNackListSizeLimit`. Otherwise, No change is applied and -1
+  // is returned. 0 is returned at success.
+  //
+  int EnableNack(size_t max_nack_list_size);
+
+  // Disable NACK.
+  void DisableNack();
+
+  //
+  // Get a list of packets to be retransmitted. `round_trip_time_ms` is an
+  // estimate of the round-trip-time (in milliseconds). Missing packets which
+  // will be playout in a shorter time than the round-trip-time (with respect
+  // to the time this API is called) will not be included in the list.
+  //
+  // Negative `round_trip_time_ms` results is an error message and empty list
+  // is returned.
+  //
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
+
+  //
+  // Get statistics of calls to GetAudio().
+  void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
+
+ private:
+  struct DecoderInfo {
+    int payload_type;
+    int sample_rate_hz;
+    int num_channels;
+    SdpAudioFormat sdp_format;
+  };
+
+  uint32_t NowInTimestamp(int decoder_sampling_rate) const;
+
+  mutable Mutex mutex_;
+  absl::optional<DecoderInfo> last_decoder_ RTC_GUARDED_BY(mutex_);
+  ACMResampler resampler_;
+
+  // After construction, this is only ever touched on the thread that calls
+  // AcmReceiver::GetAudio, and only modified in this method.
+  std::unique_ptr<int16_t[]> last_audio_buffer_;
+  CallStatistics call_stats_;
+  const std::unique_ptr<NetEq> neteq_;  // NetEq is thread-safe; no lock needed.
+  Clock* const clock_;
+  std::atomic<bool> resampled_last_output_frame_;
+};
+
+}  // namespace acm2
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
new file mode 100644
index 0000000000..6dd44b696e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -0,0 +1,455 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/acm_receiver.h"
+
+#include <algorithm>  // std::min
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+class AcmReceiverTestOldApi : public AudioPacketizationCallback,
+                              public ::testing::Test {
+ protected:
+  AcmReceiverTestOldApi()
+      : timestamp_(0),
+        packet_sent_(false),
+        last_packet_send_timestamp_(timestamp_),
+        last_frame_type_(AudioFrameType::kEmptyFrame) {
+    config_.decoder_factory = decoder_factory_;
+  }
+
+  ~AcmReceiverTestOldApi() {}
+
+  void SetUp() override {
+    acm_.reset(AudioCodingModule::Create(config_));
+    receiver_.reset(new AcmReceiver(config_));
+    ASSERT_TRUE(receiver_.get() != NULL);
+    ASSERT_TRUE(acm_.get() != NULL);
+    acm_->InitializeReceiver();
+    acm_->RegisterTransportCallback(this);
+
+    rtp_header_.sequenceNumber = 0;
+    rtp_header_.timestamp = 0;
+    rtp_header_.markerBit = false;
+    rtp_header_.ssrc = 0x12345678;  // Arbitrary.
+    rtp_header_.numCSRCs = 0;
+    rtp_header_.payloadType = 0;
+  }
+
+  void TearDown() override {}
+
+  AudioCodecInfo SetEncoder(int payload_type,
+                            const SdpAudioFormat& format,
+                            const std::map<int, int> cng_payload_types = {}) {
+    // Create the speech encoder.
+    absl::optional<AudioCodecInfo> info =
+        encoder_factory_->QueryAudioEncoder(format);
+    RTC_CHECK(info.has_value());
+    std::unique_ptr<AudioEncoder> enc =
+        encoder_factory_->MakeAudioEncoder(payload_type, format, absl::nullopt);
+
+    // If we have a compatible CN specification, stack a CNG on top.
+    auto it = cng_payload_types.find(info->sample_rate_hz);
+    if (it != cng_payload_types.end()) {
+      AudioEncoderCngConfig config;
+      config.speech_encoder = std::move(enc);
+      config.num_channels = 1;
+      config.payload_type = it->second;
+      config.vad_mode = Vad::kVadNormal;
+      enc = CreateComfortNoiseEncoder(std::move(config));
+    }
+
+    // Actually start using the new encoder.
+    acm_->SetEncoder(std::move(enc));
+    return *info;
+  }
+
+  int InsertOnePacketOfSilence(const AudioCodecInfo& info) {
+    // Frame setup according to the codec.
+    AudioFrame frame;
+    frame.sample_rate_hz_ = info.sample_rate_hz;
+    frame.samples_per_channel_ = info.sample_rate_hz / 100;  // 10 ms.
+    frame.num_channels_ = info.num_channels;
+    frame.Mute();
+    packet_sent_ = false;
+    last_packet_send_timestamp_ = timestamp_;
+    int num_10ms_frames = 0;
+    while (!packet_sent_) {
+      frame.timestamp_ = timestamp_;
+      timestamp_ += rtc::checked_cast<uint32_t>(frame.samples_per_channel_);
+      EXPECT_GE(acm_->Add10MsData(frame), 0);
+      ++num_10ms_frames;
+    }
+    return num_10ms_frames;
+  }
+
+  int SendData(AudioFrameType frame_type,
+               uint8_t payload_type,
+               uint32_t timestamp,
+               const uint8_t* payload_data,
+               size_t payload_len_bytes,
+               int64_t absolute_capture_timestamp_ms) override {
+    if (frame_type == AudioFrameType::kEmptyFrame)
+      return 0;
+
+    rtp_header_.payloadType = payload_type;
+    rtp_header_.timestamp = timestamp;
+
+    int ret_val = receiver_->InsertPacket(
+        rtp_header_,
+        rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
+    if (ret_val < 0) {
+      RTC_DCHECK_NOTREACHED();
+      return -1;
+    }
+    rtp_header_.sequenceNumber++;
+    packet_sent_ = true;
+    last_frame_type_ = frame_type;
+    return 0;
+  }
+
+  const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_ =
+      CreateBuiltinAudioEncoderFactory();
+  const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_ =
+      CreateBuiltinAudioDecoderFactory();
+  AudioCodingModule::Config config_;
+  std::unique_ptr<AcmReceiver> receiver_;
+  std::unique_ptr<AudioCodingModule> acm_;
+  RTPHeader rtp_header_;
+  uint32_t timestamp_;
+  bool packet_sent_;  // Set when SendData is called reset when inserting audio.
+  uint32_t last_packet_send_timestamp_;
+  AudioFrameType last_frame_type_;
+};
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_SampleRate DISABLED_SampleRate
+#else
+#define MAYBE_SampleRate SampleRate
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
+  const std::map<int, SdpAudioFormat> codecs = {{0, {"OPUS", 48000, 2}}};
+  receiver_->SetCodecs(codecs);
+
+  constexpr int kOutSampleRateHz = 8000;  // Different than codec sample rate.
+  for (size_t i = 0; i < codecs.size(); ++i) {
+    const int payload_type = rtc::checked_cast<int>(i);
+    const int num_10ms_frames =
+        InsertOnePacketOfSilence(SetEncoder(payload_type, codecs.at(i)));
+    for (int k = 0; k < num_10ms_frames; ++k) {
+      AudioFrame frame;
+      bool muted;
+      EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame, &muted));
+    }
+    EXPECT_EQ(encoder_factory_->QueryAudioEncoder(codecs.at(i))->sample_rate_hz,
+              receiver_->last_output_sample_rate_hz());
+  }
+}
+
+class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
+ protected:
+  AcmReceiverTestFaxModeOldApi() {
+    config_.neteq_config.for_test_no_time_stretching = true;
+  }
+
+  void RunVerifyAudioFrame(const SdpAudioFormat& codec) {
+    // Make sure "fax mode" is enabled. This will avoid delay changes unless the
+    // packet-loss concealment is made. We do this in order to make the
+    // timestamp increments predictable; in normal mode, NetEq may decide to do
+    // accelerate or pre-emptive expand operations after some time, offsetting
+    // the timestamp.
+    EXPECT_TRUE(config_.neteq_config.for_test_no_time_stretching);
+
+    constexpr int payload_type = 17;
+    receiver_->SetCodecs({{payload_type, codec}});
+
+    const AudioCodecInfo info = SetEncoder(payload_type, codec);
+    const int output_sample_rate_hz = info.sample_rate_hz;
+    const size_t output_channels = info.num_channels;
+    const size_t samples_per_ms = rtc::checked_cast<size_t>(
+        rtc::CheckedDivExact(output_sample_rate_hz, 1000));
+    const AudioFrame::VADActivity expected_vad_activity =
+        output_sample_rate_hz > 16000 ? AudioFrame::kVadActive
+                                      : AudioFrame::kVadPassive;
+
+    // Expect the first output timestamp to be 5*fs/8000 samples before the
+    // first inserted timestamp (because of NetEq's look-ahead). (This value is
+    // defined in Expand::overlap_length_.)
+    uint32_t expected_output_ts =
+        last_packet_send_timestamp_ -
+        rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);
+
+    AudioFrame frame;
+    bool muted;
+    EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
+    // Expect timestamp = 0 before first packet is inserted.
+    EXPECT_EQ(0u, frame.timestamp_);
+    for (int i = 0; i < 5; ++i) {
+      const int num_10ms_frames = InsertOnePacketOfSilence(info);
+      for (int k = 0; k < num_10ms_frames; ++k) {
+        EXPECT_EQ(0,
+                  receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
+        EXPECT_EQ(expected_output_ts, frame.timestamp_);
+        expected_output_ts += rtc::checked_cast<uint32_t>(10 * samples_per_ms);
+        EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
+        EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
+        EXPECT_EQ(output_channels, frame.num_channels_);
+        EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
+        EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
+        EXPECT_FALSE(muted);
+      }
+    }
+  }
+};
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyAudioFramePCMU DISABLED_VerifyAudioFramePCMU
+#else
+#define MAYBE_VerifyAudioFramePCMU VerifyAudioFramePCMU
+#endif
+TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFramePCMU) {
+  RunVerifyAudioFrame({"PCMU", 8000, 1});
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_VerifyAudioFrameOpus DISABLED_VerifyAudioFrameOpus
+#else
+#define MAYBE_VerifyAudioFrameOpus VerifyAudioFrameOpus
+#endif
+TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameOpus) {
+  RunVerifyAudioFrame({"opus", 48000, 2});
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
+#else
+#define MAYBE_PostdecodingVad PostdecodingVad
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
+  EXPECT_TRUE(config_.neteq_config.enable_post_decode_vad);
+  constexpr int payload_type = 34;
+  const SdpAudioFormat codec = {"L16", 16000, 1};
+  const AudioCodecInfo info = SetEncoder(payload_type, codec);
+  receiver_->SetCodecs({{payload_type, codec}});
+  constexpr int kNumPackets = 5;
+  AudioFrame frame;
+  for (int n = 0; n < kNumPackets; ++n) {
+    const int num_10ms_frames = InsertOnePacketOfSilence(info);
+    for (int k = 0; k < num_10ms_frames; ++k) {
+      bool muted;
+      ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
+    }
+  }
+  EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
+}
+
+class AcmReceiverTestPostDecodeVadPassiveOldApi : public AcmReceiverTestOldApi {
+ protected:
+  AcmReceiverTestPostDecodeVadPassiveOldApi() {
+    config_.neteq_config.enable_post_decode_vad = false;
+  }
+};
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
+#else
+#define MAYBE_PostdecodingVad PostdecodingVad
+#endif
+TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
+  EXPECT_FALSE(config_.neteq_config.enable_post_decode_vad);
+  constexpr int payload_type = 34;
+  const SdpAudioFormat codec = {"L16", 16000, 1};
+  const AudioCodecInfo info = SetEncoder(payload_type, codec);
+  auto const value = encoder_factory_->QueryAudioEncoder(codec);
+  ASSERT_TRUE(value.has_value());
+  receiver_->SetCodecs({{payload_type, codec}});
+  const int kNumPackets = 5;
+  AudioFrame frame;
+  for (int n = 0; n < kNumPackets; ++n) {
+    const int num_10ms_frames = InsertOnePacketOfSilence(info);
+    for (int k = 0; k < num_10ms_frames; ++k) {
+      bool muted;
+      ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
+    }
+  }
+  EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
+#else
+#define MAYBE_LastAudioCodec LastAudioCodec
+#endif
+#if defined(WEBRTC_CODEC_OPUS)
+TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) {
+  const std::map<int, SdpAudioFormat> codecs = {
+      {0, {"PCMU", 8000, 1}}, {1, {"PCMA", 8000, 1}}, {2, {"L16", 32000, 1}}};
+  const std::map<int, int> cng_payload_types = {
+      {8000, 100}, {16000, 101}, {32000, 102}};
+  {
+    std::map<int, SdpAudioFormat> receive_codecs = codecs;
+    for (const auto& cng_type : cng_payload_types) {
+      receive_codecs.emplace(std::make_pair(
+          cng_type.second, SdpAudioFormat("CN", cng_type.first, 1)));
+    }
+    receiver_->SetCodecs(receive_codecs);
+  }
+
+  // No audio payload is received.
+  EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
+
+  // Start with sending DTX.
+  packet_sent_ = false;
+  InsertOnePacketOfSilence(
+      SetEncoder(0, codecs.at(0), cng_payload_types));  // Enough to test
+                                                        // with one codec.
+  ASSERT_TRUE(packet_sent_);
+  EXPECT_EQ(AudioFrameType::kAudioFrameCN, last_frame_type_);
+
+  // Has received, only, DTX. Last Audio codec is undefined.
+  EXPECT_EQ(absl::nullopt, receiver_->LastDecoder());
+  EXPECT_EQ(absl::nullopt, receiver_->last_packet_sample_rate_hz());
+
+  for (size_t i = 0; i < codecs.size(); ++i) {
+    // Set DTX off to send audio payload.
+    packet_sent_ = false;
+    const int payload_type = rtc::checked_cast<int>(i);
+    const AudioCodecInfo info_without_cng =
+        SetEncoder(payload_type, codecs.at(i));
+    InsertOnePacketOfSilence(info_without_cng);
+
+    // Sanity check if Actually an audio payload received, and it should be
+    // of type "speech."
+    ASSERT_TRUE(packet_sent_);
+    ASSERT_EQ(AudioFrameType::kAudioFrameSpeech, last_frame_type_);
+    EXPECT_EQ(info_without_cng.sample_rate_hz,
+              receiver_->last_packet_sample_rate_hz());
+
+    // Set VAD on to send DTX. Then check if the "Last Audio codec" returns
+    // the expected codec. Encode repeatedly until a DTX is sent.
+    const AudioCodecInfo info_with_cng =
+        SetEncoder(payload_type, codecs.at(i), cng_payload_types);
+    while (last_frame_type_ != AudioFrameType::kAudioFrameCN) {
+      packet_sent_ = false;
+      InsertOnePacketOfSilence(info_with_cng);
+      ASSERT_TRUE(packet_sent_);
+    }
+    EXPECT_EQ(info_with_cng.sample_rate_hz,
+              receiver_->last_packet_sample_rate_hz());
+    EXPECT_EQ(codecs.at(i), receiver_->LastDecoder()->second);
+  }
+}
+#endif
+
+// Check if the statistics are initialized correctly. Before any call to ACM
+// all fields have to be zero.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_InitializedToZero DISABLED_InitializedToZero
+#else
+#define MAYBE_InitializedToZero InitializedToZero
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_InitializedToZero) {
+  AudioDecodingCallStats stats;
+  receiver_->GetDecodingCallStatistics(&stats);
+  EXPECT_EQ(0, stats.calls_to_neteq);
+  EXPECT_EQ(0, stats.calls_to_silence_generator);
+  EXPECT_EQ(0, stats.decoded_normal);
+  EXPECT_EQ(0, stats.decoded_cng);
+  EXPECT_EQ(0, stats.decoded_neteq_plc);
+  EXPECT_EQ(0, stats.decoded_plc_cng);
+  EXPECT_EQ(0, stats.decoded_muted_output);
+}
+
+// Insert some packets and pull audio. Check statistics are valid. Then,
+// simulate packet loss and check if PLC and PLC-to-CNG statistics are
+// correctly updated.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_NetEqCalls DISABLED_NetEqCalls
+#else
+#define MAYBE_NetEqCalls NetEqCalls
+#endif
+TEST_F(AcmReceiverTestOldApi, MAYBE_NetEqCalls) {
+  AudioDecodingCallStats stats;
+  const int kNumNormalCalls = 10;
+  const int kSampleRateHz = 16000;
+  const int kNumSamples10ms = kSampleRateHz / 100;
+  const int kFrameSizeMs = 10;  // Multiple of 10.
+  const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
+  const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
+  const uint8_t kPayloadType = 111;
+  RTPHeader rtp_header;
+  AudioFrame audio_frame;
+  bool muted;
+
+  receiver_->SetCodecs(
+      {{kPayloadType, SdpAudioFormat("L16", kSampleRateHz, 1)}});
+  rtp_header.sequenceNumber = 0xABCD;
+  rtp_header.timestamp = 0xABCDEF01;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.markerBit = false;
+  rtp_header.ssrc = 0x1234;
+  rtp_header.numCSRCs = 0;
+  rtp_header.payload_type_frequency = kSampleRateHz;
+
+  for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) {
+    const uint8_t kPayload[kPayloadSizeBytes] = {0};
+    ASSERT_EQ(0, receiver_->InsertPacket(rtp_header, kPayload));
+    ++rtp_header.sequenceNumber;
+    rtp_header.timestamp += kFrameSizeSamples;
+    ASSERT_EQ(0, receiver_->GetAudio(-1, &audio_frame, &muted));
+    EXPECT_FALSE(muted);
+  }
+  receiver_->GetDecodingCallStatistics(&stats);
+  EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq);
+  EXPECT_EQ(0, stats.calls_to_silence_generator);
+  EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
+  EXPECT_EQ(0, stats.decoded_cng);
+  EXPECT_EQ(0, stats.decoded_neteq_plc);
+  EXPECT_EQ(0, stats.decoded_plc_cng);
+  EXPECT_EQ(0, stats.decoded_muted_output);
+
+  const int kNumPlc = 3;
+  const int kNumPlcCng = 5;
+
+  // Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG.
+  for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) {
+    ASSERT_EQ(0, receiver_->GetAudio(-1, &audio_frame, &muted));
+    EXPECT_FALSE(muted);
+  }
+  receiver_->GetDecodingCallStatistics(&stats);
+  EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq);
+  EXPECT_EQ(0, stats.calls_to_silence_generator);
+  EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
+  EXPECT_EQ(0, stats.decoded_cng);
+  EXPECT_EQ(kNumPlc, stats.decoded_neteq_plc);
+  EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
+  EXPECT_EQ(0, stats.decoded_muted_output);
+  // TODO(henrik.lundin) Add a test with muted state enabled.
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc
new file mode 100644
index 0000000000..13709dbbee
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/acm_remixing.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output) {
+  RTC_DCHECK_EQ(input.num_channels_, 2);
+  RTC_DCHECK_EQ(output.size(), input.samples_per_channel_);
+
+  if (input.muted()) {
+    std::fill(output.begin(), output.begin() + input.samples_per_channel_, 0);
+  } else {
+    const int16_t* const input_data = input.data();
+    for (size_t n = 0; n < input.samples_per_channel_; ++n) {
+      output[n] = rtc::dchecked_cast<int16_t>(
+          (int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
+    }
+  }
+}
+
+void ReMixFrame(const AudioFrame& input,
+                size_t num_output_channels,
+                std::vector<int16_t>* output) {
+  const size_t output_size = num_output_channels * input.samples_per_channel_;
+  RTC_DCHECK(!(input.num_channels_ == 0 && num_output_channels > 0 &&
+               input.samples_per_channel_ > 0));
+
+  if (output->size() != output_size) {
+    output->resize(output_size);
+  }
+
+  // For muted frames, fill the frame with zeros.
+  if (input.muted()) {
+    std::fill(output->begin(), output->end(), 0);
+    return;
+  }
+
+  // Ensure that the special case of zero input channels is handled correctly
+  // (zero samples per channel is already handled correctly in the code below).
+  if (input.num_channels_ == 0) {
+    return;
+  }
+
+  const int16_t* const input_data = input.data();
+  size_t out_index = 0;
+
+  // When upmixing is needed and the input is mono copy the left channel
+  // into the left and right channels, and set any remaining channels to zero.
+  if (input.num_channels_ == 1 && input.num_channels_ < num_output_channels) {
+    for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+      (*output)[out_index++] = input_data[k];
+      (*output)[out_index++] = input_data[k];
+      for (size_t j = 2; j < num_output_channels; ++j) {
+        (*output)[out_index++] = 0;
+      }
+      RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
+    }
+    RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
+    return;
+  }
+
+  size_t in_index = 0;
+
+  // When upmixing is needed and the output is surround, copy the available
+  // channels directly, and set the remaining channels to zero.
+  if (input.num_channels_ < num_output_channels) {
+    for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+      for (size_t j = 0; j < input.num_channels_; ++j) {
+        (*output)[out_index++] = input_data[in_index++];
+      }
+      for (size_t j = input.num_channels_; j < num_output_channels; ++j) {
+        (*output)[out_index++] = 0;
+      }
+      RTC_DCHECK_EQ(in_index, (k + 1) * input.num_channels_);
+      RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
+    }
+    RTC_DCHECK_EQ(in_index, input.samples_per_channel_ * input.num_channels_);
+    RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
+
+    return;
+  }
+
+  // When downmixing is needed, and the input is stereo, average the channels.
+  if (input.num_channels_ == 2) {
+    for (size_t n = 0; n < input.samples_per_channel_; ++n) {
+      (*output)[n] = rtc::dchecked_cast<int16_t>(
+          (int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
+    }
+    return;
+  }
+
+  // When downmixing is needed, and the input is multichannel, drop the surplus
+  // channels.
+  const size_t num_channels_to_drop = input.num_channels_ - num_output_channels;
+  for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+    for (size_t j = 0; j < num_output_channels; ++j) {
+      (*output)[out_index++] = input_data[in_index++];
+    }
+    in_index += num_channels_to_drop;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.h b/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.h
new file mode 100644
index 0000000000..661569b033
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
+
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+
+// Stereo-to-mono downmixing. The length of the output must equal to the number
+// of samples per channel in the input.
+void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output);
+
+// Remixes the interleaved input frame to an interleaved output data vector. The
+// remixed data replaces the data in the output vector which is resized if
+// needed. The remixing supports any combination of input and output channels,
+// as well as any number of samples per channel.
+void ReMixFrame(const AudioFrame& input,
+                size_t num_output_channels,
+                std::vector<int16_t>* output);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing_unittest.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing_unittest.cc
new file mode 100644
index 0000000000..a1a816f727
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing_unittest.cc
@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/acm_remixing.h"
+
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using ::testing::AllOf;
+using ::testing::Each;
+using ::testing::ElementsAreArray;
+using ::testing::SizeIs;
+
+namespace webrtc {
+
+TEST(AcmRemixing, DownMixFrame) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 2;
+    in_data[2 * k + 1] = 0;
+  }
+
+  DownMixFrame(in, out);
+
+  EXPECT_THAT(out, AllOf(SizeIs(480), Each(1)));
+}
+
+TEST(AcmRemixing, DownMixMutedFrame) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 2;
+    in_data[2 * k + 1] = 0;
+  }
+
+  in.Mute();
+
+  DownMixFrame(in, out);
+
+  EXPECT_THAT(out, AllOf(SizeIs(480), Each(0)));
+}
+
+TEST(AcmRemixing, RemixMutedStereoFrameTo6Channels) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 1;
+    in_data[2 * k + 1] = 2;
+  }
+  in.Mute();
+
+  ReMixFrame(in, 6, &out);
+  EXPECT_EQ(6 * 480u, out.size());
+
+  EXPECT_THAT(out, AllOf(SizeIs(in.samples_per_channel_ * 6), Each(0)));
+}
+
+TEST(AcmRemixing, RemixStereoFrameTo6Channels) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 1;
+    in_data[2 * k + 1] = 2;
+  }
+
+  ReMixFrame(in, 6, &out);
+  EXPECT_EQ(6 * 480u, out.size());
+
+  std::vector<int16_t> expected_output(in.samples_per_channel_ * 6);
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    expected_output[6 * k] = 1;
+    expected_output[6 * k + 1] = 2;
+  }
+
+  EXPECT_THAT(out, ElementsAreArray(expected_output));
+}
+
+TEST(AcmRemixing, RemixMonoFrameTo6Channels) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 1;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[k] = 1;
+  }
+
+  ReMixFrame(in, 6, &out);
+  EXPECT_EQ(6 * 480u, out.size());
+
+  std::vector<int16_t> expected_output(in.samples_per_channel_ * 6, 0);
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    expected_output[6 * k] = 1;
+    expected_output[6 * k + 1] = 1;
+  }
+
+  EXPECT_THAT(out, ElementsAreArray(expected_output));
+}
+
+TEST(AcmRemixing, RemixStereoFrameToMono) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 2;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[2 * k] = 2;
+    in_data[2 * k + 1] = 0;
+  }
+
+  ReMixFrame(in, 1, &out);
+  EXPECT_EQ(480u, out.size());
+
+  EXPECT_THAT(out, AllOf(SizeIs(in.samples_per_channel_), Each(1)));
+}
+
+TEST(AcmRemixing, RemixMonoFrameToStereo) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 1;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    in_data[k] = 1;
+  }
+
+  ReMixFrame(in, 2, &out);
+  EXPECT_EQ(960u, out.size());
+
+  EXPECT_THAT(out, AllOf(SizeIs(2 * in.samples_per_channel_), Each(1)));
+}
+
+TEST(AcmRemixing, Remix3ChannelFrameToStereo) {
+  std::vector<int16_t> out(480, 0);
+  AudioFrame in;
+  in.num_channels_ = 3;
+  in.samples_per_channel_ = 480;
+
+  int16_t* const in_data = in.mutable_data();
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    for (size_t j = 0; j < 3; ++j) {
+      in_data[3 * k + j] = j;
+    }
+  }
+
+  ReMixFrame(in, 2, &out);
+  EXPECT_EQ(2 * 480u, out.size());
+
+  std::vector<int16_t> expected_output(in.samples_per_channel_ * 2);
+  for (size_t k = 0; k < in.samples_per_channel_; ++k) {
+    for (size_t j = 0; j < 2; ++j) {
+      expected_output[2 * k + j] = static_cast<int>(j);
+    }
+  }
+
+  EXPECT_THAT(out, ElementsAreArray(expected_output));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.cc
new file mode 100644
index 0000000000..e307c6ca57
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/acm_resampler.h"
+
+#include <string.h>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace acm2 {
+
+ACMResampler::ACMResampler() {}
+
+ACMResampler::~ACMResampler() {}
+
+int ACMResampler::Resample10Msec(const int16_t* in_audio,
+                                 int in_freq_hz,
+                                 int out_freq_hz,
+                                 size_t num_audio_channels,
+                                 size_t out_capacity_samples,
+                                 int16_t* out_audio) {
+  size_t in_length = in_freq_hz * num_audio_channels / 100;
+  if (in_freq_hz == out_freq_hz) {
+    if (out_capacity_samples < in_length) {
+      RTC_DCHECK_NOTREACHED();
+      return -1;
+    }
+    memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
+    return static_cast<int>(in_length / num_audio_channels);
+  }
+
+  if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
+                                    num_audio_channels) != 0) {
+    RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
+                      << out_freq_hz << ", " << num_audio_channels
+                      << ") failed.";
+    return -1;
+  }
+
+  int out_length =
+      resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
+  if (out_length == -1) {
+    RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
+                      << out_audio << ", " << out_capacity_samples
+                      << ") failed.";
+    return -1;
+  }
+
+  return static_cast<int>(out_length / num_audio_channels);
+}
+
+}  // namespace acm2
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.h b/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.h
new file mode 100644
index 0000000000..96ba93a762
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "common_audio/resampler/include/push_resampler.h"
+
+namespace webrtc {
+namespace acm2 {
+
+class ACMResampler {
+ public:
+  ACMResampler();
+  ~ACMResampler();
+
+  int Resample10Msec(const int16_t* in_audio,
+                     int in_freq_hz,
+                     int out_freq_hz,
+                     size_t num_audio_channels,
+                     size_t out_capacity_samples,
+                     int16_t* out_audio);
+
+ private:
+  PushResampler<int16_t> resampler_;
+};
+
+}  // namespace acm2
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_send_test.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_send_test.cc
new file mode 100644
index 0000000000..3e65f94b0d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_send_test.cc
@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/acm_send_test.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_encode.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
+                                     int source_rate_hz,
+                                     int test_duration_ms)
+    : clock_(0),
+      acm_(webrtc::AudioCodingModule::Create([this] {
+        AudioCodingModule::Config config;
+        config.clock = &clock_;
+        config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+        return config;
+      }())),
+      audio_source_(audio_source),
+      source_rate_hz_(source_rate_hz),
+      input_block_size_samples_(
+          static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),
+      codec_registered_(false),
+      test_duration_ms_(test_duration_ms),
+      frame_type_(AudioFrameType::kAudioFrameSpeech),
+      payload_type_(0),
+      timestamp_(0),
+      sequence_number_(0) {
+  input_frame_.sample_rate_hz_ = source_rate_hz_;
+  input_frame_.num_channels_ = 1;
+  input_frame_.samples_per_channel_ = input_block_size_samples_;
+  RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
+  acm_->RegisterTransportCallback(this);
+}
+
+AcmSendTestOldApi::~AcmSendTestOldApi() = default;
+
+bool AcmSendTestOldApi::RegisterCodec(absl::string_view payload_name,
+                                      int clockrate_hz,
+                                      int num_channels,
+                                      int payload_type,
+                                      int frame_size_samples) {
+  SdpAudioFormat format(payload_name, clockrate_hz, num_channels);
+  if (absl::EqualsIgnoreCase(payload_name, "g722")) {
+    RTC_CHECK_EQ(16000, clockrate_hz);
+    format.clockrate_hz = 8000;
+  } else if (absl::EqualsIgnoreCase(payload_name, "opus")) {
+    RTC_CHECK(num_channels == 1 || num_channels == 2);
+    if (num_channels == 2) {
+      format.parameters["stereo"] = "1";
+    }
+    format.num_channels = 2;
+  }
+  format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
+      frame_size_samples, rtc::CheckedDivExact(clockrate_hz, 1000)));
+  auto factory = CreateBuiltinAudioEncoderFactory();
+  acm_->SetEncoder(
+      factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
+  codec_registered_ = true;
+  input_frame_.num_channels_ = num_channels;
+  RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
+  return codec_registered_;
+}
+
+void AcmSendTestOldApi::RegisterExternalCodec(
+    std::unique_ptr<AudioEncoder> external_speech_encoder) {
+  input_frame_.num_channels_ = external_speech_encoder->NumChannels();
+  acm_->SetEncoder(std::move(external_speech_encoder));
+  RTC_DCHECK_LE(input_block_size_samples_ * input_frame_.num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
+  codec_registered_ = true;
+}
+
+std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
+  RTC_DCHECK(codec_registered_);
+  if (filter_.test(static_cast<size_t>(payload_type_))) {
+    // This payload type should be filtered out. Since the payload type is the
+    // same throughout the whole test run, no packet at all will be delivered.
+    // We can just as well signal that the test is over by returning NULL.
+    return nullptr;
+  }
+  // Insert audio and process until one packet is produced.
+  while (clock_.TimeInMilliseconds() < test_duration_ms_) {
+    clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
+    RTC_CHECK(audio_source_->Read(
+        input_block_size_samples_ * input_frame_.num_channels_,
+        input_frame_.mutable_data()));
+    data_to_send_ = false;
+    RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
+    input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
+    if (data_to_send_) {
+      // Encoded packet received.
+      return CreatePacket();
+    }
+  }
+  // Test ended.
+  return nullptr;
+}
+
+// This method receives the callback from ACM when a new packet is produced.
+int32_t AcmSendTestOldApi::SendData(AudioFrameType frame_type,
+                                    uint8_t payload_type,
+                                    uint32_t timestamp,
+                                    const uint8_t* payload_data,
+                                    size_t payload_len_bytes,
+                                    int64_t absolute_capture_timestamp_ms) {
+  // Store the packet locally.
+  frame_type_ = frame_type;
+  payload_type_ = payload_type;
+  timestamp_ = timestamp;
+  last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
+  RTC_DCHECK_EQ(last_payload_vec_.size(), payload_len_bytes);
+  data_to_send_ = true;
+  return 0;
+}
+
+std::unique_ptr<Packet> AcmSendTestOldApi::CreatePacket() {
+  const size_t kRtpHeaderSize = 12;
+  rtc::CopyOnWriteBuffer packet_buffer(last_payload_vec_.size() +
+                                       kRtpHeaderSize);
+  uint8_t* packet_memory = packet_buffer.MutableData();
+  // Populate the header bytes.
+  packet_memory[0] = 0x80;
+  packet_memory[1] = static_cast<uint8_t>(payload_type_);
+  packet_memory[2] = (sequence_number_ >> 8) & 0xFF;
+  packet_memory[3] = (sequence_number_)&0xFF;
+  packet_memory[4] = (timestamp_ >> 24) & 0xFF;
+  packet_memory[5] = (timestamp_ >> 16) & 0xFF;
+  packet_memory[6] = (timestamp_ >> 8) & 0xFF;
+  packet_memory[7] = timestamp_ & 0xFF;
+  // Set SSRC to 0x12345678.
+  packet_memory[8] = 0x12;
+  packet_memory[9] = 0x34;
+  packet_memory[10] = 0x56;
+  packet_memory[11] = 0x78;
+
+  ++sequence_number_;
+
+  // Copy the payload data.
+  memcpy(packet_memory + kRtpHeaderSize, &last_payload_vec_[0],
+         last_payload_vec_.size());
+  auto packet = std::make_unique<Packet>(std::move(packet_buffer),
+                                         clock_.TimeInMilliseconds());
+  RTC_DCHECK(packet);
+  RTC_DCHECK(packet->valid_header());
+  return packet;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_send_test.h b/third_party/libwebrtc/modules/audio_coding/acm2/acm_send_test.h
new file mode 100644
index 0000000000..0bd24705fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_send_test.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/audio_frame.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+class AudioEncoder;
+
+namespace test {
+class InputAudioFile;
+class Packet;
+
+class AcmSendTestOldApi : public AudioPacketizationCallback,
+                          public PacketSource {
+ public:
+  AcmSendTestOldApi(InputAudioFile* audio_source,
+                    int source_rate_hz,
+                    int test_duration_ms);
+  ~AcmSendTestOldApi() override;
+
+  AcmSendTestOldApi(const AcmSendTestOldApi&) = delete;
+  AcmSendTestOldApi& operator=(const AcmSendTestOldApi&) = delete;
+
+  // Registers the send codec. Returns true on success, false otherwise.
+  bool RegisterCodec(absl::string_view payload_name,
+                     int sampling_freq_hz,
+                     int channels,
+                     int payload_type,
+                     int frame_size_samples);
+
+  // Registers an external send codec.
+  void RegisterExternalCodec(
+      std::unique_ptr<AudioEncoder> external_speech_encoder);
+
+  // Inherited from PacketSource.
+  std::unique_ptr<Packet> NextPacket() override;
+
+  // Inherited from AudioPacketizationCallback.
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_len_bytes,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  AudioCodingModule* acm() { return acm_.get(); }
+
+ private:
+  static const int kBlockSizeMs = 10;
+
+  // Creates a Packet object from the last packet produced by ACM (and received
+  // through the SendData method as a callback).
+  std::unique_ptr<Packet> CreatePacket();
+
+  SimulatedClock clock_;
+  std::unique_ptr<AudioCodingModule> acm_;
+  InputAudioFile* audio_source_;
+  int source_rate_hz_;
+  const size_t input_block_size_samples_;
+  AudioFrame input_frame_;
+  bool codec_registered_;
+  int test_duration_ms_;
+  // The following member variables are set whenever SendData() is called.
+  AudioFrameType frame_type_;
+  int payload_type_;
+  uint32_t timestamp_;
+  uint16_t sequence_number_;
+  std::vector<uint8_t> last_payload_vec_;
+  bool data_to_send_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module.cc b/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module.cc
new file mode 100644
index 0000000000..2c186273b6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -0,0 +1,637 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+
+#include <algorithm>
+#include <cstdint>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_remixing.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/include/module_common_types.h"
+#include "modules/include/module_common_types_public.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+// Initial size for the buffer in InputBuffer. This matches 6 channels of 10 ms
+// 48 kHz data.
+constexpr size_t kInitialInputDataBufferSize = 6 * 480;
+
+constexpr int32_t kMaxInputSampleRateHz = 192000;
+
+class AudioCodingModuleImpl final : public AudioCodingModule {
+ public:
+  explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
+  ~AudioCodingModuleImpl() override;
+
+  /////////////////////////////////////////
+  //   Sender
+  //
+
+  void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
+                         modifier) override;
+
+  // Register a transport callback which will be
+  // called to deliver the encoded buffers.
+  int RegisterTransportCallback(AudioPacketizationCallback* transport) override;
+
+  // Add 10 ms of raw (PCM) audio data to the encoder.
+  int Add10MsData(const AudioFrame& audio_frame) override;
+
+  /////////////////////////////////////////
+  // (FEC) Forward Error Correction (codec internal)
+  //
+
+  // Set target packet loss rate
+  int SetPacketLossRate(int loss_rate) override;
+
+  /////////////////////////////////////////
+  //   Receiver
+  //
+
+  // Initialize receiver, resets codec database etc.
+  int InitializeReceiver() override;
+
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
+
+  // Incoming packet from network parsed and ready for decode.
+  int IncomingPacket(const uint8_t* incoming_payload,
+                     const size_t payload_length,
+                     const RTPHeader& rtp_info) override;
+
+  // Get 10 milliseconds of raw audio data to play out, and
+  // automatic resample to the requested frequency if > 0.
+  int PlayoutData10Ms(int desired_freq_hz,
+                      AudioFrame* audio_frame,
+                      bool* muted) override;
+
+  /////////////////////////////////////////
+  //   Statistics
+  //
+
+  int GetNetworkStatistics(NetworkStatistics* statistics) override;
+
+  ANAStats GetANAStats() const override;
+
+  int GetTargetBitrate() const override;
+
+ private:
+  struct InputData {
+    InputData() : buffer(kInitialInputDataBufferSize) {}
+    uint32_t input_timestamp;
+    const int16_t* audio;
+    size_t length_per_channel;
+    size_t audio_channel;
+    // If a re-mix is required (up or down), this buffer will store a re-mixed
+    // version of the input.
+    std::vector<int16_t> buffer;
+  };
+
+  InputData input_data_ RTC_GUARDED_BY(acm_mutex_);
+
+  // This member class writes values to the named UMA histogram, but only if
+  // the value has changed since the last time (and always for the first call).
+  class ChangeLogger {
+   public:
+    explicit ChangeLogger(absl::string_view histogram_name)
+        : histogram_name_(histogram_name) {}
+    // Logs the new value if it is different from the last logged value, or if
+    // this is the first call.
+    void MaybeLog(int value);
+
+   private:
+    int last_value_ = 0;
+    int first_time_ = true;
+    const std::string histogram_name_;
+  };
+
+  int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
+
+  // TODO(bugs.webrtc.org/10739): change `absolute_capture_timestamp_ms` to
+  // int64_t when it always receives a valid value.
+  int Encode(const InputData& input_data,
+             absl::optional<int64_t> absolute_capture_timestamp_ms)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
+
+  int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
+
+  bool HaveValidEncoder(absl::string_view caller_name) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
+
+  // Preprocessing of input audio, including resampling and down-mixing if
+  // required, before pushing audio into encoder's buffer.
+  //
+  // in_frame: input audio-frame
+  // ptr_out: pointer to output audio_frame. If no preprocessing is required
+  //          `ptr_out` will be pointing to `in_frame`, otherwise pointing to
+  //          `preprocess_frame_`.
+  //
+  // Return value:
+  //   -1: if encountering an error.
+  //    0: otherwise.
+  int PreprocessToAddData(const AudioFrame& in_frame,
+                          const AudioFrame** ptr_out)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
+
+  // Change required states after starting to receive the codec corresponding
+  // to `index`.
+  int UpdateUponReceivingCodec(int index);
+
+  mutable Mutex acm_mutex_;
+  rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_mutex_);
+  uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_mutex_);
+  uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_mutex_);
+  acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_mutex_);
+  acm2::AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
+  ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_mutex_);
+
+  // Current encoder stack, provided by a call to RegisterEncoder.
+  std::unique_ptr<AudioEncoder> encoder_stack_ RTC_GUARDED_BY(acm_mutex_);
+
+  // This is to keep track of CN instances where we can send DTMFs.
+  uint8_t previous_pltype_ RTC_GUARDED_BY(acm_mutex_);
+
+  bool receiver_initialized_ RTC_GUARDED_BY(acm_mutex_);
+
+  AudioFrame preprocess_frame_ RTC_GUARDED_BY(acm_mutex_);
+  bool first_10ms_data_ RTC_GUARDED_BY(acm_mutex_);
+
+  bool first_frame_ RTC_GUARDED_BY(acm_mutex_);
+  uint32_t last_timestamp_ RTC_GUARDED_BY(acm_mutex_);
+  uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(acm_mutex_);
+
+  Mutex callback_mutex_;
+  AudioPacketizationCallback* packetization_callback_
+      RTC_GUARDED_BY(callback_mutex_);
+
+  int codec_histogram_bins_log_[static_cast<size_t>(
+      AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)];
+  int number_of_consecutive_empty_packets_;
+};
+
+// Adds a codec usage sample to the histogram.
+void UpdateCodecTypeHistogram(size_t codec_type) {
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.Audio.Encoder.CodecType", static_cast<int>(codec_type),
+      static_cast<int>(
+          webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
+}
+
+void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
+  if (value != last_value_ || first_time_) {
+    first_time_ = false;
+    last_value_ = value;
+    RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, value);
+  }
+}
+
+AudioCodingModuleImpl::AudioCodingModuleImpl(
+    const AudioCodingModule::Config& config)
+    : expected_codec_ts_(0xD87F3F9F),
+      expected_in_ts_(0xD87F3F9F),
+      receiver_(config),
+      bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
+      encoder_stack_(nullptr),
+      previous_pltype_(255),
+      receiver_initialized_(false),
+      first_10ms_data_(false),
+      first_frame_(true),
+      packetization_callback_(NULL),
+      codec_histogram_bins_log_(),
+      number_of_consecutive_empty_packets_(0) {
+  if (InitializeReceiverSafe() < 0) {
+    RTC_LOG(LS_ERROR) << "Cannot initialize receiver";
+  }
+  RTC_LOG(LS_INFO) << "Created";
+}
+
+AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
+
+int32_t AudioCodingModuleImpl::Encode(
+    const InputData& input_data,
+    absl::optional<int64_t> absolute_capture_timestamp_ms) {
+  // TODO(bugs.webrtc.org/10739): add dcheck that
+  // `audio_frame.absolute_capture_timestamp_ms()` always has a value.
+  AudioEncoder::EncodedInfo encoded_info;
+  uint8_t previous_pltype;
+
+  // Check if there is an encoder before.
+  if (!HaveValidEncoder("Process"))
+    return -1;
+
+  if (!first_frame_) {
+    RTC_DCHECK(IsNewerTimestamp(input_data.input_timestamp, last_timestamp_))
+        << "Time should not move backwards";
+  }
+
+  // Scale the timestamp to the codec's RTP timestamp rate.
+  uint32_t rtp_timestamp =
+      first_frame_
+          ? input_data.input_timestamp
+          : last_rtp_timestamp_ +
+                rtc::dchecked_cast<uint32_t>(rtc::CheckedDivExact(
+                    int64_t{input_data.input_timestamp - last_timestamp_} *
+                        encoder_stack_->RtpTimestampRateHz(),
+                    int64_t{encoder_stack_->SampleRateHz()}));
+
+  last_timestamp_ = input_data.input_timestamp;
+  last_rtp_timestamp_ = rtp_timestamp;
+  first_frame_ = false;
+
+  // Clear the buffer before reuse - encoded data will get appended.
+  encode_buffer_.Clear();
+  encoded_info = encoder_stack_->Encode(
+      rtp_timestamp,
+      rtc::ArrayView<const int16_t>(
+          input_data.audio,
+          input_data.audio_channel * input_data.length_per_channel),
+      &encode_buffer_);
+
+  bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000);
+  if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {
+    // Not enough data.
+    return 0;
+  }
+  previous_pltype = previous_pltype_;  // Read it while we have the critsect.
+
+  // Log codec type to histogram once every 500 packets.
+  if (encoded_info.encoded_bytes == 0) {
+    ++number_of_consecutive_empty_packets_;
+  } else {
+    size_t codec_type = static_cast<size_t>(encoded_info.encoder_type);
+    codec_histogram_bins_log_[codec_type] +=
+        number_of_consecutive_empty_packets_ + 1;
+    number_of_consecutive_empty_packets_ = 0;
+    if (codec_histogram_bins_log_[codec_type] >= 500) {
+      codec_histogram_bins_log_[codec_type] -= 500;
+      UpdateCodecTypeHistogram(codec_type);
+    }
+  }
+
+  AudioFrameType frame_type;
+  if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
+    frame_type = AudioFrameType::kEmptyFrame;
+    encoded_info.payload_type = previous_pltype;
+  } else {
+    RTC_DCHECK_GT(encode_buffer_.size(), 0);
+    frame_type = encoded_info.speech ? AudioFrameType::kAudioFrameSpeech
+                                     : AudioFrameType::kAudioFrameCN;
+  }
+
+  {
+    MutexLock lock(&callback_mutex_);
+    if (packetization_callback_) {
+      packetization_callback_->SendData(
+          frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
+          encode_buffer_.data(), encode_buffer_.size(),
+          absolute_capture_timestamp_ms.value_or(-1));
+    }
+  }
+  previous_pltype_ = encoded_info.payload_type;
+  return static_cast<int32_t>(encode_buffer_.size());
+}
+
+/////////////////////////////////////////
+//   Sender
+//
+
+void AudioCodingModuleImpl::ModifyEncoder(
+    rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
+  MutexLock lock(&acm_mutex_);
+  modifier(&encoder_stack_);
+}
+
+// Register a transport callback which will be called to deliver
+// the encoded buffers.
+int AudioCodingModuleImpl::RegisterTransportCallback(
+    AudioPacketizationCallback* transport) {
+  MutexLock lock(&callback_mutex_);
+  packetization_callback_ = transport;
+  return 0;
+}
+
+// Add 10MS of raw (PCM) audio data to the encoder.
+int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
+  MutexLock lock(&acm_mutex_);
+  int r = Add10MsDataInternal(audio_frame, &input_data_);
+  // TODO(bugs.webrtc.org/10739): add dcheck that
+  // `audio_frame.absolute_capture_timestamp_ms()` always has a value.
+  return r < 0
+             ? r
+             : Encode(input_data_, audio_frame.absolute_capture_timestamp_ms());
+}
+
+int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
+                                               InputData* input_data) {
+  if (audio_frame.samples_per_channel_ == 0) {
+    RTC_DCHECK_NOTREACHED();
+    RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, payload length is zero";
+    return -1;
+  }
+
+  if (audio_frame.sample_rate_hz_ > kMaxInputSampleRateHz) {
+    RTC_DCHECK_NOTREACHED();
+    RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, input frequency not valid";
+    return -1;
+  }
+
+  // If the length and frequency matches. We currently just support raw PCM.
+  if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) !=
+      audio_frame.samples_per_channel_) {
+    RTC_LOG(LS_ERROR)
+        << "Cannot Add 10 ms audio, input frequency and length doesn't match";
+    return -1;
+  }
+
+  if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2 &&
+      audio_frame.num_channels_ != 4 && audio_frame.num_channels_ != 6 &&
+      audio_frame.num_channels_ != 8) {
+    RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, invalid number of channels.";
+    return -1;
+  }
+
+  // Do we have a codec registered?
+  if (!HaveValidEncoder("Add10MsData")) {
+    return -1;
+  }
+
+  const AudioFrame* ptr_frame;
+  // Perform a resampling, also down-mix if it is required and can be
+  // performed before resampling (a down mix prior to resampling will take
+  // place if both primary and secondary encoders are mono and input is in
+  // stereo).
+  if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) {
+    return -1;
+  }
+
+  // Check whether we need an up-mix or down-mix?
+  const size_t current_num_channels = encoder_stack_->NumChannels();
+  const bool same_num_channels =
+      ptr_frame->num_channels_ == current_num_channels;
+
+  // TODO(yujo): Skip encode of muted frames.
+  input_data->input_timestamp = ptr_frame->timestamp_;
+  input_data->length_per_channel = ptr_frame->samples_per_channel_;
+  input_data->audio_channel = current_num_channels;
+
+  if (!same_num_channels) {
+    // Remixes the input frame to the output data and in the process resize the
+    // output data if needed.
+    ReMixFrame(*ptr_frame, current_num_channels, &input_data->buffer);
+
+    // For pushing data to primary, point the `ptr_audio` to correct buffer.
+    input_data->audio = input_data->buffer.data();
+    RTC_DCHECK_GE(input_data->buffer.size(),
+                  input_data->length_per_channel * input_data->audio_channel);
+  } else {
+    // When adding data to encoders this pointer is pointing to an audio buffer
+    // with correct number of channels.
+    input_data->audio = ptr_frame->data();
+  }
+
+  return 0;
+}
+
+// Perform a resampling and down-mix if required. We down-mix only if
+// encoder is mono and input is stereo. In case of dual-streaming, both
+// encoders has to be mono for down-mix to take place.
+// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing
+// is required, |*ptr_out| points to `in_frame`.
+// TODO(yujo): Make this more efficient for muted frames.
+int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
+                                               const AudioFrame** ptr_out) {
+  const bool resample =
+      in_frame.sample_rate_hz_ != encoder_stack_->SampleRateHz();
+
+  // This variable is true if primary codec and secondary codec (if exists)
+  // are both mono and input is stereo.
+  // TODO(henrik.lundin): This condition should probably be
+  //   in_frame.num_channels_ > encoder_stack_->NumChannels()
+  const bool down_mix =
+      in_frame.num_channels_ == 2 && encoder_stack_->NumChannels() == 1;
+
+  if (!first_10ms_data_) {
+    expected_in_ts_ = in_frame.timestamp_;
+    expected_codec_ts_ = in_frame.timestamp_;
+    first_10ms_data_ = true;
+  } else if (in_frame.timestamp_ != expected_in_ts_) {
+    RTC_LOG(LS_WARNING) << "Unexpected input timestamp: " << in_frame.timestamp_
+                        << ", expected: " << expected_in_ts_;
+    expected_codec_ts_ +=
+        (in_frame.timestamp_ - expected_in_ts_) *
+        static_cast<uint32_t>(
+            static_cast<double>(encoder_stack_->SampleRateHz()) /
+            static_cast<double>(in_frame.sample_rate_hz_));
+    expected_in_ts_ = in_frame.timestamp_;
+  }
+
+  if (!down_mix && !resample) {
+    // No pre-processing is required.
+    if (expected_in_ts_ == expected_codec_ts_) {
+      // If we've never resampled, we can use the input frame as-is
+      *ptr_out = &in_frame;
+    } else {
+      // Otherwise we'll need to alter the timestamp. Since in_frame is const,
+      // we'll have to make a copy of it.
+      preprocess_frame_.CopyFrom(in_frame);
+      preprocess_frame_.timestamp_ = expected_codec_ts_;
+      *ptr_out = &preprocess_frame_;
+    }
+
+    expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+    expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+    return 0;
+  }
+
+  *ptr_out = &preprocess_frame_;
+  preprocess_frame_.num_channels_ = in_frame.num_channels_;
+  preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
+  std::array<int16_t, AudioFrame::kMaxDataSizeSamples> audio;
+  const int16_t* src_ptr_audio;
+  if (down_mix) {
+    // If a resampling is required, the output of a down-mix is written into a
+    // local buffer, otherwise, it will be written to the output frame.
+    int16_t* dest_ptr_audio =
+        resample ? audio.data() : preprocess_frame_.mutable_data();
+    RTC_DCHECK_GE(audio.size(), preprocess_frame_.samples_per_channel_);
+    RTC_DCHECK_GE(audio.size(), in_frame.samples_per_channel_);
+    DownMixFrame(in_frame,
+                 rtc::ArrayView<int16_t>(
+                     dest_ptr_audio, preprocess_frame_.samples_per_channel_));
+    preprocess_frame_.num_channels_ = 1;
+
+    // Set the input of the resampler to the down-mixed signal.
+    src_ptr_audio = audio.data();
+  } else {
+    // Set the input of the resampler to the original data.
+    src_ptr_audio = in_frame.data();
+  }
+
+  preprocess_frame_.timestamp_ = expected_codec_ts_;
+  preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_;
+  // If it is required, we have to do a resampling.
+  if (resample) {
+    // The result of the resampler is written to output frame.
+    int16_t* dest_ptr_audio = preprocess_frame_.mutable_data();
+
+    int samples_per_channel = resampler_.Resample10Msec(
+        src_ptr_audio, in_frame.sample_rate_hz_, encoder_stack_->SampleRateHz(),
+        preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
+        dest_ptr_audio);
+
+    if (samples_per_channel < 0) {
+      RTC_LOG(LS_ERROR) << "Cannot add 10 ms audio, resampling failed";
+      return -1;
+    }
+    preprocess_frame_.samples_per_channel_ =
+        static_cast<size_t>(samples_per_channel);
+    preprocess_frame_.sample_rate_hz_ = encoder_stack_->SampleRateHz();
+  }
+
+  expected_codec_ts_ +=
+      static_cast<uint32_t>(preprocess_frame_.samples_per_channel_);
+  expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+
+  return 0;
+}
+
+/////////////////////////////////////////
+//   (FEC) Forward Error Correction (codec internal)
+//
+
+int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
+  MutexLock lock(&acm_mutex_);
+  if (HaveValidEncoder("SetPacketLossRate")) {
+    encoder_stack_->OnReceivedUplinkPacketLossFraction(loss_rate / 100.0);
+  }
+  return 0;
+}
+
+/////////////////////////////////////////
+//   Receiver
+//
+
+int AudioCodingModuleImpl::InitializeReceiver() {
+  MutexLock lock(&acm_mutex_);
+  return InitializeReceiverSafe();
+}
+
+// Initialize receiver, resets codec database etc.
+int AudioCodingModuleImpl::InitializeReceiverSafe() {
+  // If the receiver is already initialized then we want to destroy any
+  // existing decoders. After a call to this function, we should have a clean
+  // start-up.
+  if (receiver_initialized_)
+    receiver_.RemoveAllCodecs();
+  receiver_.FlushBuffers();
+
+  receiver_initialized_ = true;
+  return 0;
+}
+
+void AudioCodingModuleImpl::SetReceiveCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  MutexLock lock(&acm_mutex_);
+  receiver_.SetCodecs(codecs);
+}
+
+// Incoming packet from network parsed and ready for decode.
+int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
+                                          const size_t payload_length,
+                                          const RTPHeader& rtp_header) {
+  RTC_DCHECK_EQ(payload_length == 0, incoming_payload == nullptr);
+  return receiver_.InsertPacket(
+      rtp_header,
+      rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
+}
+
+// Get 10 milliseconds of raw audio data to play out.
+// Automatic resample to the requested frequency.
+int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
+                                           AudioFrame* audio_frame,
+                                           bool* muted) {
+  // GetAudio always returns 10 ms, at the requested sample rate.
+  if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
+    RTC_LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
+    return -1;
+  }
+  return 0;
+}
+
+/////////////////////////////////////////
+//   Statistics
+//
+
+// TODO(turajs) change the return value to void. Also change the corresponding
+// NetEq function.
+int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
+  receiver_.GetNetworkStatistics(statistics);
+  return 0;
+}
+
+bool AudioCodingModuleImpl::HaveValidEncoder(
+    absl::string_view caller_name) const {
+  if (!encoder_stack_) {
+    RTC_LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";
+    return false;
+  }
+  return true;
+}
+
+ANAStats AudioCodingModuleImpl::GetANAStats() const {
+  MutexLock lock(&acm_mutex_);
+  if (encoder_stack_)
+    return encoder_stack_->GetANAStats();
+  // If no encoder is set, return default stats.
+  return ANAStats();
+}
+
+int AudioCodingModuleImpl::GetTargetBitrate() const {
+  MutexLock lock(&acm_mutex_);
+  if (!encoder_stack_) {
+    return -1;
+  }
+  return encoder_stack_->GetTargetBitrate();
+}
+
+}  // namespace
+
+AudioCodingModule::Config::Config(
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
+    : neteq_config(),
+      clock(Clock::GetRealTimeClockRaw()),
+      decoder_factory(decoder_factory) {
+  // Post-decode VAD is disabled by default in NetEq, however, Audio
+  // Conference Mixer relies on VAD decisions and fails without them.
+  neteq_config.enable_post_decode_vad = true;
+}
+
+AudioCodingModule::Config::Config(const Config&) = default;
+AudioCodingModule::Config::~Config() = default;
+
+AudioCodingModule* AudioCodingModule::Create(const Config& config) {
+  return new AudioCodingModuleImpl(config);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
new file mode 100644
index 0000000000..f1eb81c015
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -0,0 +1,1278 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/acm2/acm_receive_test.h"
+#include "modules/audio_coding/acm2/acm_send_test.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/neteq/tools/audio_checksum.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/output_audio_file.h"
+#include "modules/audio_coding/neteq/tools/output_wav_file.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "rtc_base/event.h"
+#include "rtc_base/message_digest.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/arch.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/audio_decoder_proxy_factory.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/mock_audio_encoder.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Invoke;
+
+namespace webrtc {
+
+namespace {
+const int kSampleRateHz = 16000;
+const int kNumSamples10ms = kSampleRateHz / 100;
+const int kFrameSizeMs = 10;  // Multiple of 10.
+const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
+const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
+const uint8_t kPayloadType = 111;
+}  // namespace
+
+class RtpData {
+ public:
+  RtpData(int samples_per_packet, uint8_t payload_type)
+      : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {}
+
+  virtual ~RtpData() {}
+
+  void Populate(RTPHeader* rtp_header) {
+    rtp_header->sequenceNumber = 0xABCD;
+    rtp_header->timestamp = 0xABCDEF01;
+    rtp_header->payloadType = payload_type_;
+    rtp_header->markerBit = false;
+    rtp_header->ssrc = 0x1234;
+    rtp_header->numCSRCs = 0;
+
+    rtp_header->payload_type_frequency = kSampleRateHz;
+  }
+
+  void Forward(RTPHeader* rtp_header) {
+    ++rtp_header->sequenceNumber;
+    rtp_header->timestamp += samples_per_packet_;
+  }
+
+ private:
+  int samples_per_packet_;
+  uint8_t payload_type_;
+};
+
+class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
+ public:
+  PacketizationCallbackStubOldApi()
+      : num_calls_(0),
+        last_frame_type_(AudioFrameType::kEmptyFrame),
+        last_payload_type_(-1),
+        last_timestamp_(0) {}
+
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_len_bytes,
+                   int64_t absolute_capture_timestamp_ms) override {
+    MutexLock lock(&mutex_);
+    ++num_calls_;
+    last_frame_type_ = frame_type;
+    last_payload_type_ = payload_type;
+    last_timestamp_ = timestamp;
+    last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes);
+    return 0;
+  }
+
+  int num_calls() const {
+    MutexLock lock(&mutex_);
+    return num_calls_;
+  }
+
+  int last_payload_len_bytes() const {
+    MutexLock lock(&mutex_);
+    return rtc::checked_cast<int>(last_payload_vec_.size());
+  }
+
+  AudioFrameType last_frame_type() const {
+    MutexLock lock(&mutex_);
+    return last_frame_type_;
+  }
+
+  int last_payload_type() const {
+    MutexLock lock(&mutex_);
+    return last_payload_type_;
+  }
+
+  uint32_t last_timestamp() const {
+    MutexLock lock(&mutex_);
+    return last_timestamp_;
+  }
+
+  void SwapBuffers(std::vector<uint8_t>* payload) {
+    MutexLock lock(&mutex_);
+    last_payload_vec_.swap(*payload);
+  }
+
+ private:
+  int num_calls_ RTC_GUARDED_BY(mutex_);
+  AudioFrameType last_frame_type_ RTC_GUARDED_BY(mutex_);
+  int last_payload_type_ RTC_GUARDED_BY(mutex_);
+  uint32_t last_timestamp_ RTC_GUARDED_BY(mutex_);
+  std::vector<uint8_t> last_payload_vec_ RTC_GUARDED_BY(mutex_);
+  mutable Mutex mutex_;
+};
+
+class AudioCodingModuleTestOldApi : public ::testing::Test {
+ protected:
+  AudioCodingModuleTestOldApi()
+      : rtp_utility_(new RtpData(kFrameSizeSamples, kPayloadType)),
+        clock_(Clock::GetRealTimeClock()) {}
+
+  ~AudioCodingModuleTestOldApi() {}
+
+  void TearDown() {}
+
+  void SetUp() {
+    acm_.reset(AudioCodingModule::Create([this] {
+      AudioCodingModule::Config config;
+      config.clock = clock_;
+      config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+      return config;
+    }()));
+
+    rtp_utility_->Populate(&rtp_header_);
+
+    input_frame_.sample_rate_hz_ = kSampleRateHz;
+    input_frame_.num_channels_ = 1;
+    input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000;  // 10 ms.
+    static_assert(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples,
+                  "audio frame too small");
+    input_frame_.Mute();
+
+    ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_));
+
+    SetUpL16Codec();
+  }
+
+  // Set up L16 codec.
+  virtual void SetUpL16Codec() {
+    audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1);
+    pac_size_ = 160;
+  }
+
+  virtual void RegisterCodec() {
+    acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}});
+    acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
+        kPayloadType, *audio_format_, absl::nullopt));
+  }
+
+  virtual void InsertPacketAndPullAudio() {
+    InsertPacket();
+    PullAudio();
+  }
+
+  virtual void InsertPacket() {
+    const uint8_t kPayload[kPayloadSizeBytes] = {0};
+    ASSERT_EQ(0,
+              acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_));
+    rtp_utility_->Forward(&rtp_header_);
+  }
+
+  virtual void PullAudio() {
+    AudioFrame audio_frame;
+    bool muted;
+    ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
+  }
+
+  virtual void InsertAudio() {
+    ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
+    input_frame_.timestamp_ += kNumSamples10ms;
+  }
+
+  virtual void VerifyEncoding() {
+    int last_length = packet_cb_.last_payload_len_bytes();
+    EXPECT_TRUE(last_length == 2 * pac_size_ || last_length == 0)
+        << "Last encoded packet was " << last_length << " bytes.";
+  }
+
+  virtual void InsertAudioAndVerifyEncoding() {
+    InsertAudio();
+    VerifyEncoding();
+  }
+
+  std::unique_ptr<RtpData> rtp_utility_;
+  std::unique_ptr<AudioCodingModule> acm_;
+  PacketizationCallbackStubOldApi packet_cb_;
+  RTPHeader rtp_header_;
+  AudioFrame input_frame_;
+
+  absl::optional<SdpAudioFormat> audio_format_;
+  int pac_size_ = -1;
+
+  Clock* clock_;
+};
+
+class AudioCodingModuleTestOldApiDeathTest
+    : public AudioCodingModuleTestOldApi {};
+
+TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {
+  AudioFrame audio_frame;
+  const int kSampleRateHz = 32000;
+  bool muted;
+  EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame, &muted));
+  ASSERT_FALSE(muted);
+  EXPECT_EQ(0u, audio_frame.timestamp_);
+  EXPECT_GT(audio_frame.num_channels_, 0u);
+  EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),
+            audio_frame.samples_per_channel_);
+  EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
+}
+
+// The below test is temporarily disabled on Windows due to problems
+// with clang debug builds.
+// TODO(tommi): Re-enable when we've figured out what the problem is.
+// http://crbug.com/615050
+#if !defined(WEBRTC_WIN) && defined(__clang__) && RTC_DCHECK_IS_ON && \
+    GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST_F(AudioCodingModuleTestOldApiDeathTest, FailOnZeroDesiredFrequency) {
+  AudioFrame audio_frame;
+  bool muted;
+  RTC_EXPECT_DEATH(acm_->PlayoutData10Ms(0, &audio_frame, &muted),
+                   "dst_sample_rate_hz");
+}
+#endif
+
+// Checks that the transport callback is invoked once for each speech packet.
+// Also checks that the frame type is kAudioFrameSpeech.
+TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) {
+  const int k10MsBlocksPerPacket = 3;
+  pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100;
+  audio_format_->parameters["ptime"] = "30";
+  RegisterCodec();
+  const int kLoops = 10;
+  for (int i = 0; i < kLoops; ++i) {
+    EXPECT_EQ(i / k10MsBlocksPerPacket, packet_cb_.num_calls());
+    if (packet_cb_.num_calls() > 0)
+      EXPECT_EQ(AudioFrameType::kAudioFrameSpeech,
+                packet_cb_.last_frame_type());
+    InsertAudioAndVerifyEncoding();
+  }
+  EXPECT_EQ(kLoops / k10MsBlocksPerPacket, packet_cb_.num_calls());
+  EXPECT_EQ(AudioFrameType::kAudioFrameSpeech, packet_cb_.last_frame_type());
+}
+
+// Introduce this class to set different expectations on the number of encoded
+// bytes. This class expects all encoded packets to be 9 bytes (matching one
+// CNG SID frame) or 0 bytes. This test depends on `input_frame_` containing
+// (near-)zero values. It also introduces a way to register comfort noise with
+// a custom payload type.
+class AudioCodingModuleTestWithComfortNoiseOldApi
+    : public AudioCodingModuleTestOldApi {
+ protected:
+  void RegisterCngCodec(int rtp_payload_type) {
+    acm_->SetReceiveCodecs({{kPayloadType, *audio_format_},
+                            {rtp_payload_type, {"cn", kSampleRateHz, 1}}});
+    acm_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
+      AudioEncoderCngConfig config;
+      config.speech_encoder = std::move(*enc);
+      config.num_channels = 1;
+      config.payload_type = rtp_payload_type;
+      config.vad_mode = Vad::kVadNormal;
+      *enc = CreateComfortNoiseEncoder(std::move(config));
+    });
+  }
+
+  void VerifyEncoding() override {
+    int last_length = packet_cb_.last_payload_len_bytes();
+    EXPECT_TRUE(last_length == 9 || last_length == 0)
+        << "Last encoded packet was " << last_length << " bytes.";
+  }
+
+  void DoTest(int blocks_per_packet, int cng_pt) {
+    const int kLoops = 40;
+    // This array defines the expected frame types, and when they should arrive.
+    // We expect a frame to arrive each time the speech encoder would have
+    // produced a packet, and once every 100 ms the frame should be non-empty,
+    // that is contain comfort noise.
+    const struct {
+      int ix;
+      AudioFrameType type;
+    } expectation[] = {{2, AudioFrameType::kAudioFrameCN},
+                       {5, AudioFrameType::kEmptyFrame},
+                       {8, AudioFrameType::kEmptyFrame},
+                       {11, AudioFrameType::kAudioFrameCN},
+                       {14, AudioFrameType::kEmptyFrame},
+                       {17, AudioFrameType::kEmptyFrame},
+                       {20, AudioFrameType::kAudioFrameCN},
+                       {23, AudioFrameType::kEmptyFrame},
+                       {26, AudioFrameType::kEmptyFrame},
+                       {29, AudioFrameType::kEmptyFrame},
+                       {32, AudioFrameType::kAudioFrameCN},
+                       {35, AudioFrameType::kEmptyFrame},
+                       {38, AudioFrameType::kEmptyFrame}};
+    for (int i = 0; i < kLoops; ++i) {
+      int num_calls_before = packet_cb_.num_calls();
+      EXPECT_EQ(i / blocks_per_packet, num_calls_before);
+      InsertAudioAndVerifyEncoding();
+      int num_calls = packet_cb_.num_calls();
+      if (num_calls == num_calls_before + 1) {
+        EXPECT_EQ(expectation[num_calls - 1].ix, i);
+        EXPECT_EQ(expectation[num_calls - 1].type, packet_cb_.last_frame_type())
+            << "Wrong frame type for lap " << i;
+        EXPECT_EQ(cng_pt, packet_cb_.last_payload_type());
+      } else {
+        EXPECT_EQ(num_calls, num_calls_before);
+      }
+    }
+  }
+};
+
+// Checks that the transport callback is invoked once per frame period of the
+// underlying speech encoder, even when comfort noise is produced.
+// Also checks that the frame type is kAudioFrameCN or kEmptyFrame.
+TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi,
+       TransportCallbackTestForComfortNoiseRegisterCngLast) {
+  const int k10MsBlocksPerPacket = 3;
+  pac_size_ = k10MsBlocksPerPacket * kSampleRateHz / 100;
+  audio_format_->parameters["ptime"] = "30";
+  RegisterCodec();
+  const int kCngPayloadType = 105;
+  RegisterCngCodec(kCngPayloadType);
+  DoTest(k10MsBlocksPerPacket, kCngPayloadType);
+}
+
+// A multi-threaded test for ACM that uses the PCM16b 16 kHz codec.
+class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
+ protected:
+  static const int kNumPackets = 500;
+  static const int kNumPullCalls = 500;
+
+  AudioCodingModuleMtTestOldApi()
+      : AudioCodingModuleTestOldApi(),
+        send_count_(0),
+        insert_packet_count_(0),
+        pull_audio_count_(0),
+        next_insert_packet_time_ms_(0),
+        fake_clock_(new SimulatedClock(0)) {
+    clock_ = fake_clock_.get();
+  }
+
+  void SetUp() {
+    AudioCodingModuleTestOldApi::SetUp();
+    RegisterCodec();  // Must be called before the threads start below.
+    StartThreads();
+  }
+
+  void StartThreads() {
+    quit_.store(false);
+
+    const auto attributes =
+        rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
+    send_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (!quit_.load()) {
+            CbSendImpl();
+          }
+        },
+        "send", attributes);
+    insert_packet_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (!quit_.load()) {
+            CbInsertPacketImpl();
+          }
+        },
+        "insert_packet", attributes);
+    pull_audio_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (!quit_.load()) {
+            CbPullAudioImpl();
+          }
+        },
+        "pull_audio", attributes);
+  }
+
+  void TearDown() {
+    AudioCodingModuleTestOldApi::TearDown();
+    quit_.store(true);
+    pull_audio_thread_.Finalize();
+    send_thread_.Finalize();
+    insert_packet_thread_.Finalize();
+  }
+
+  bool RunTest() { return test_complete_.Wait(TimeDelta::Minutes(10)); }
+
+  virtual bool TestDone() {
+    if (packet_cb_.num_calls() > kNumPackets) {
+      MutexLock lock(&mutex_);
+      if (pull_audio_count_ > kNumPullCalls) {
+        // Both conditions for completion are met. End the test.
+        return true;
+      }
+    }
+    return false;
+  }
+
+  // The send thread doesn't have to care about the current simulated time,
+  // since only the AcmReceiver is using the clock.
+  void CbSendImpl() {
+    SleepMs(1);
+    if (HasFatalFailure()) {
+      // End the test early if a fatal failure (ASSERT_*) has occurred.
+      test_complete_.Set();
+    }
+    ++send_count_;
+    InsertAudioAndVerifyEncoding();
+    if (TestDone()) {
+      test_complete_.Set();
+    }
+  }
+
+  void CbInsertPacketImpl() {
+    SleepMs(1);
+    {
+      MutexLock lock(&mutex_);
+      if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
+        return;
+      }
+      next_insert_packet_time_ms_ += 10;
+    }
+    // Now we're not holding the crit sect when calling ACM.
+    ++insert_packet_count_;
+    InsertPacket();
+  }
+
+  void CbPullAudioImpl() {
+    SleepMs(1);
+    {
+      MutexLock lock(&mutex_);
+      // Don't let the insert thread fall behind.
+      if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) {
+        return;
+      }
+      ++pull_audio_count_;
+    }
+    // Now we're not holding the crit sect when calling ACM.
+    PullAudio();
+    fake_clock_->AdvanceTimeMilliseconds(10);
+  }
+
+  rtc::PlatformThread send_thread_;
+  rtc::PlatformThread insert_packet_thread_;
+  rtc::PlatformThread pull_audio_thread_;
+  // Used to force worker threads to stop looping.
+  std::atomic<bool> quit_;
+
+  rtc::Event test_complete_;
+  int send_count_;
+  int insert_packet_count_;
+  int pull_audio_count_ RTC_GUARDED_BY(mutex_);
+  Mutex mutex_;
+  int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<SimulatedClock> fake_clock_;
+};
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_DoTest DISABLED_DoTest
+#else
+#define MAYBE_DoTest DoTest
+#endif
+TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
+  EXPECT_TRUE(RunTest());
+}
+
+// Disabling all of these tests on iOS until file support has been added.
+// See https://code.google.com/p/webrtc/issues/detail?id=4752 for details.
+#if !defined(WEBRTC_IOS)
+
+// This test verifies bit exactness for the send-side of ACM. The test setup is
+// a chain of three different test classes:
+//
+// test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest
+//
+// The receiver side is driving the test by requesting new packets from
+// AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
+// packet from test::AcmSendTest::NextPacket, which inserts audio from the
+// input file until one packet is produced. (The input file loops indefinitely.)
+// Before passing the packet to the receiver, this test class verifies the
+// packet header and updates a payload checksum with the new payload. The
+// decoded output from the receiver is also verified with a (separate) checksum.
+class AcmSenderBitExactnessOldApi : public ::testing::Test,
+                                    public test::PacketSource {
+ protected:
+  static const int kTestDurationMs = 1000;
+
+  AcmSenderBitExactnessOldApi()
+      : frame_size_rtp_timestamps_(0),
+        packet_count_(0),
+        payload_type_(0),
+        last_sequence_number_(0),
+        last_timestamp_(0),
+        payload_checksum_(rtc::MessageDigestFactory::Create(rtc::DIGEST_MD5)) {}
+
+  // Sets up the test::AcmSendTest object. Returns true on success, otherwise
+  // false.
+  bool SetUpSender(absl::string_view input_file_name, int source_rate) {
+    // Note that `audio_source_` will loop forever. The test duration is set
+    // explicitly by `kTestDurationMs`.
+    audio_source_.reset(new test::InputAudioFile(input_file_name));
+    send_test_.reset(new test::AcmSendTestOldApi(audio_source_.get(),
+                                                 source_rate, kTestDurationMs));
+    return send_test_.get() != NULL;
+  }
+
+  // Registers a send codec in the test::AcmSendTest object. Returns true on
+  // success, false on failure.
+  bool RegisterSendCodec(absl::string_view payload_name,
+                         int sampling_freq_hz,
+                         int channels,
+                         int payload_type,
+                         int frame_size_samples,
+                         int frame_size_rtp_timestamps) {
+    payload_type_ = payload_type;
+    frame_size_rtp_timestamps_ = frame_size_rtp_timestamps;
+    return send_test_->RegisterCodec(payload_name, sampling_freq_hz, channels,
+                                     payload_type, frame_size_samples);
+  }
+
+  void RegisterExternalSendCodec(
+      std::unique_ptr<AudioEncoder> external_speech_encoder,
+      int payload_type) {
+    payload_type_ = payload_type;
+    frame_size_rtp_timestamps_ = rtc::checked_cast<uint32_t>(
+        external_speech_encoder->Num10MsFramesInNextPacket() *
+        external_speech_encoder->RtpTimestampRateHz() / 100);
+    send_test_->RegisterExternalCodec(std::move(external_speech_encoder));
+  }
+
+  // Runs the test. SetUpSender() and RegisterSendCodec() must have been called
+  // before calling this method.
+  void Run(absl::string_view audio_checksum_ref,
+           absl::string_view payload_checksum_ref,
+           int expected_packets,
+           test::AcmReceiveTestOldApi::NumOutputChannels expected_channels,
+           rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr) {
+    if (!decoder_factory) {
+      decoder_factory = CreateBuiltinAudioDecoderFactory();
+    }
+    // Set up the receiver used to decode the packets and verify the decoded
+    // output.
+    test::AudioChecksum audio_checksum;
+    const std::string output_file_name =
+        webrtc::test::OutputPath() +
+        ::testing::UnitTest::GetInstance()
+            ->current_test_info()
+            ->test_case_name() +
+        "_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
+        "_output.wav";
+    const int kOutputFreqHz = 8000;
+    test::OutputWavFile output_file(output_file_name, kOutputFreqHz,
+                                    expected_channels);
+    // Have the output audio sent both to file and to the checksum calculator.
+    test::AudioSinkFork output(&audio_checksum, &output_file);
+    test::AcmReceiveTestOldApi receive_test(this, &output, kOutputFreqHz,
+                                            expected_channels, decoder_factory);
+    ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
+
+    // This is where the actual test is executed.
+    receive_test.Run();
+
+    // Extract and verify the audio checksum.
+    std::string checksum_string = audio_checksum.Finish();
+    ExpectChecksumEq(audio_checksum_ref, checksum_string);
+
+    // Extract and verify the payload checksum.
+    rtc::Buffer checksum_result(payload_checksum_->Size());
+    payload_checksum_->Finish(checksum_result.data(), checksum_result.size());
+    checksum_string = rtc::hex_encode(checksum_result);
+    ExpectChecksumEq(payload_checksum_ref, checksum_string);
+
+    // Verify number of packets produced.
+    EXPECT_EQ(expected_packets, packet_count_);
+
+    // Delete the output file.
+    remove(output_file_name.c_str());
+  }
+
+  // Helper: result must be one the "|"-separated checksums.
+  void ExpectChecksumEq(absl::string_view ref, absl::string_view result) {
+    if (ref.size() == result.size()) {
+      // Only one checksum: clearer message.
+      EXPECT_EQ(ref, result);
+    } else {
+      EXPECT_NE(ref.find(result), absl::string_view::npos)
+          << result << " must be one of these:\n"
+          << ref;
+    }
+  }
+
+  // Inherited from test::PacketSource.
+  std::unique_ptr<test::Packet> NextPacket() override {
+    auto packet = send_test_->NextPacket();
+    if (!packet)
+      return NULL;
+
+    VerifyPacket(packet.get());
+    // TODO(henrik.lundin) Save the packet to file as well.
+
+    // Pass it on to the caller. The caller becomes the owner of `packet`.
+    return packet;
+  }
+
+  // Verifies the packet.
+  void VerifyPacket(const test::Packet* packet) {
+    EXPECT_TRUE(packet->valid_header());
+    // (We can check the header fields even if valid_header() is false.)
+    EXPECT_EQ(payload_type_, packet->header().payloadType);
+    if (packet_count_ > 0) {
+      // This is not the first packet.
+      uint16_t sequence_number_diff =
+          packet->header().sequenceNumber - last_sequence_number_;
+      EXPECT_EQ(1, sequence_number_diff);
+      uint32_t timestamp_diff = packet->header().timestamp - last_timestamp_;
+      EXPECT_EQ(frame_size_rtp_timestamps_, timestamp_diff);
+    }
+    ++packet_count_;
+    last_sequence_number_ = packet->header().sequenceNumber;
+    last_timestamp_ = packet->header().timestamp;
+    // Update the checksum.
+    payload_checksum_->Update(packet->payload(),
+                              packet->payload_length_bytes());
+  }
+
+  void SetUpTest(absl::string_view codec_name,
+                 int codec_sample_rate_hz,
+                 int channels,
+                 int payload_type,
+                 int codec_frame_size_samples,
+                 int codec_frame_size_rtp_timestamps) {
+    ASSERT_TRUE(SetUpSender(
+        channels == 1 ? kTestFileMono32kHz : kTestFileFakeStereo32kHz, 32000));
+    ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels,
+                                  payload_type, codec_frame_size_samples,
+                                  codec_frame_size_rtp_timestamps));
+  }
+
+  void SetUpTestExternalEncoder(
+      std::unique_ptr<AudioEncoder> external_speech_encoder,
+      int payload_type) {
+    ASSERT_TRUE(send_test_);
+    RegisterExternalSendCodec(std::move(external_speech_encoder), payload_type);
+  }
+
+  std::unique_ptr<test::AcmSendTestOldApi> send_test_;
+  std::unique_ptr<test::InputAudioFile> audio_source_;
+  uint32_t frame_size_rtp_timestamps_;
+  int packet_count_;
+  uint8_t payload_type_;
+  uint16_t last_sequence_number_;
+  uint32_t last_timestamp_;
+  std::unique_ptr<rtc::MessageDigest> payload_checksum_;
+  const std::string kTestFileMono32kHz =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const std::string kTestFileFakeStereo32kHz =
+      webrtc::test::ResourcePath("audio_coding/testfile_fake_stereo_32kHz",
+                                 "pcm");
+  const std::string kTestFileQuad48kHz = webrtc::test::ResourcePath(
+      "audio_coding/speech_4_channels_48k_one_second",
+      "wav");
+};
+
+class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {};
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
+  Run(/*audio_checksum_ref=*/"69118ed438ac76252d023e0463819471",
+      /*payload_checksum_ref=*/"c1edd36339ce0326cc4550041ad719a0",
+      /*expected_packets=*/100,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160));
+  Run(/*audio_checksum_ref=*/"f95c87bdd33f631bcf80f4b19445bbd2",
+      /*payload_checksum_ref=*/"ad786526383178b08d80d6eee06e9bad",
+      /*expected_packets=*/100,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320));
+  Run(/*audio_checksum_ref=*/"c50244419c5c3a2f04cc69a022c266a2",
+      /*payload_checksum_ref=*/"5ef82ea885e922263606c6fdbc49f651",
+      /*expected_packets=*/100,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80));
+  Run(/*audio_checksum_ref=*/"4fccf4cc96f1e8e8de4b9fadf62ded9e",
+      /*payload_checksum_ref=*/"62ce5adb0d4965d0a52ec98ae7f98974",
+      /*expected_packets=*/100,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160));
+  Run(/*audio_checksum_ref=*/"e15e388d9d4af8c02a59fe1552fedee3",
+      /*payload_checksum_ref=*/"41ca8edac4b8c71cd54fd9f25ec14870",
+      /*expected_packets=*/100,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_32000khz_10ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320));
+  Run(/*audio_checksum_ref=*/"b240520c0d05003fde7a174ae5957286",
+      /*payload_checksum_ref=*/"50e58502fb04421bf5b857dda4c96879",
+      /*expected_packets=*/100,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcmu_20ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 1, 0, 160, 160));
+  Run(/*audio_checksum_ref=*/"c8d1fc677f33c2022ec5f83c7f302280",
+      /*payload_checksum_ref=*/"8f9b8750bd80fe26b6cbf6659b89f0f9",
+      /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcma_20ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160));
+  Run(/*audio_checksum_ref=*/"47eb60e855eb12d1b0e6da9c975754a4",
+      /*payload_checksum_ref=*/"6ad745e55aa48981bfc790d0eeef2dd1",
+      /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcmu_stereo_20ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 2, 110, 160, 160));
+  Run(/*audio_checksum_ref=*/"6ef2f57d4934714787fd0a834e3ea18e",
+      /*payload_checksum_ref=*/"60b6f25e8d1e74cb679cfe756dd9bca5",
+      /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160));
+  Run(/*audio_checksum_ref=*/"a84d75e098d87ab6b260687eb4b612a2",
+      /*payload_checksum_ref=*/"92b282c83efd20e7eeef52ba40842cf7",
+      /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
+}
+
+#if defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_LINUX) && \
+    defined(WEBRTC_ARCH_X86_64)
+TEST_F(AcmSenderBitExactnessOldApi, Ilbc_30ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240));
+  Run(/*audio_checksum_ref=*/"b14dba0de36efa5ec88a32c0b320b70f",
+      /*payload_checksum_ref=*/"cfae2e9f6aba96e145f2bcdd5050ce78",
+      /*expected_packets=*/33,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
+}
+#endif
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
+TEST_F(AcmSenderBitExactnessOldApi, G722_20ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160));
+  Run(/*audio_checksum_ref=*/"f5264affff25cf2cbd2e1e8a5217f9a3",
+      /*payload_checksum_ref=*/"fc68a87e1380614e658087cb35d5ca10",
+      /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);
+}
+#endif
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
+TEST_F(AcmSenderBitExactnessOldApi, G722_stereo_20ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160));
+  Run(/*audio_checksum_ref=*/"be0b8528ff9db3a2219f55ddd36faf7f",
+      /*payload_checksum_ref=*/"66516152eeaa1e650ad94ff85f668dac",
+      /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
+}
+#endif
+
+namespace {
+// Checksum depends on libopus being compiled with or without SSE.
+const std::string audio_checksum =
+    "6a76fe2ffba057c06eb63239b3c47abe"
+    "|0c4f9d33b4a7379a34ee0c0d5718afe6";
+const std::string payload_checksum =
+    "b43bdf7638b2bc2a5a6f30bdc640b9ed"
+    "|c30d463e7ed10bdd1da9045f80561f27";
+}  // namespace
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
+TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
+  ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
+  Run(audio_checksum, payload_checksum, /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
+}
+#endif
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
+TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms) {
+  const auto config = AudioEncoderOpus::SdpToConfig(
+      SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
+  ASSERT_TRUE(SetUpSender(kTestFileFakeStereo32kHz, 32000));
+  ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(
+      AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120));
+  Run(audio_checksum, payload_checksum, /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
+}
+#endif
+
+// TODO(webrtc:8649): Disabled until the Encoder counterpart of
+// https://webrtc-review.googlesource.com/c/src/+/129768 lands.
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
+TEST_F(AcmSenderBitExactnessNewApi, DISABLED_OpusManyChannels) {
+  constexpr int kNumChannels = 4;
+  constexpr int kOpusPayloadType = 120;
+
+  // Read a 4 channel file at 48kHz.
+  ASSERT_TRUE(SetUpSender(kTestFileQuad48kHz, 48000));
+
+  const auto sdp_format = SdpAudioFormat("multiopus", 48000, kNumChannels,
+                                         {{"channel_mapping", "0,1,2,3"},
+                                          {"coupled_streams", "2"},
+                                          {"num_streams", "2"}});
+  const auto encoder_config =
+      AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+  ASSERT_TRUE(encoder_config.has_value());
+
+  ASSERT_NO_FATAL_FAILURE(
+      SetUpTestExternalEncoder(AudioEncoderMultiChannelOpus::MakeAudioEncoder(
+                                   *encoder_config, kOpusPayloadType),
+                               kOpusPayloadType));
+
+  const auto decoder_config =
+      AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+  const auto opus_decoder =
+      AudioDecoderMultiChannelOpus::MakeAudioDecoder(*decoder_config);
+
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(opus_decoder.get());
+
+  // Set up an EXTERNAL DECODER to parse 4 channels.
+  Run("audio checksum check downstream|8051617907766bec5f4e4a4f7c6d5291",
+      "payload checksum check downstream|b09c52e44b2bdd9a0809e3a5b1623a76",
+      /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kQuadOutput,
+      decoder_factory);
+}
+#endif
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64)
+TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) {
+  auto config = AudioEncoderOpus::SdpToConfig(
+      SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
+  // If not set, default will be kAudio in case of stereo.
+  config->application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
+  ASSERT_TRUE(SetUpSender(kTestFileFakeStereo32kHz, 32000));
+  ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(
+      AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120));
+  const std::string audio_maybe_sse =
+      "1010e60ad34cee73c939edaf563d0593"
+      "|c05b4523d4c3fad2bab96d2a56baa2d0";
+
+  const std::string payload_maybe_sse =
+      "ea48d94e43217793af9b7e15ece94e54"
+      "|bd93c492087093daf662cdd968f6cdda";
+
+  Run(audio_maybe_sse, payload_maybe_sse, /*expected_packets=*/50,
+      /*expected_channels=*/test::AcmReceiveTestOldApi::kStereoOutput);
+}
+#endif
+
+// This test is for verifying the SetBitRate function. The bitrate is changed at
+// the beginning, and the number of generated bytes are checked.
+class AcmSetBitRateTest : public ::testing::Test {
+ protected:
+  static const int kTestDurationMs = 1000;
+
+  // Sets up the test::AcmSendTest object. Returns true on success, otherwise
+  // false.
+  bool SetUpSender() {
+    const std::string input_file_name =
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+    // Note that `audio_source_` will loop forever. The test duration is set
+    // explicitly by `kTestDurationMs`.
+    audio_source_.reset(new test::InputAudioFile(input_file_name));
+    static const int kSourceRateHz = 32000;
+    send_test_.reset(new test::AcmSendTestOldApi(
+        audio_source_.get(), kSourceRateHz, kTestDurationMs));
+    return send_test_.get();
+  }
+
+  // Registers a send codec in the test::AcmSendTest object. Returns true on
+  // success, false on failure.
+  virtual bool RegisterSendCodec(absl::string_view payload_name,
+                                 int sampling_freq_hz,
+                                 int channels,
+                                 int payload_type,
+                                 int frame_size_samples,
+                                 int frame_size_rtp_timestamps) {
+    return send_test_->RegisterCodec(payload_name, sampling_freq_hz, channels,
+                                     payload_type, frame_size_samples);
+  }
+
+  void RegisterExternalSendCodec(
+      std::unique_ptr<AudioEncoder> external_speech_encoder,
+      int payload_type) {
+    send_test_->RegisterExternalCodec(std::move(external_speech_encoder));
+  }
+
+  void RunInner(int min_expected_total_bits, int max_expected_total_bits) {
+    int nr_bytes = 0;
+    while (std::unique_ptr<test::Packet> next_packet =
+               send_test_->NextPacket()) {
+      nr_bytes += rtc::checked_cast<int>(next_packet->payload_length_bytes());
+    }
+    EXPECT_LE(min_expected_total_bits, nr_bytes * 8);
+    EXPECT_GE(max_expected_total_bits, nr_bytes * 8);
+  }
+
+  void SetUpTest(absl::string_view codec_name,
+                 int codec_sample_rate_hz,
+                 int channels,
+                 int payload_type,
+                 int codec_frame_size_samples,
+                 int codec_frame_size_rtp_timestamps) {
+    ASSERT_TRUE(SetUpSender());
+    ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels,
+                                  payload_type, codec_frame_size_samples,
+                                  codec_frame_size_rtp_timestamps));
+  }
+
+  std::unique_ptr<test::AcmSendTestOldApi> send_test_;
+  std::unique_ptr<test::InputAudioFile> audio_source_;
+};
+
+class AcmSetBitRateNewApi : public AcmSetBitRateTest {
+ protected:
+  // Runs the test. SetUpSender() must have been called and a codec must be set
+  // up before calling this method.
+  void Run(int min_expected_total_bits, int max_expected_total_bits) {
+    RunInner(min_expected_total_bits, max_expected_total_bits);
+  }
+};
+
+TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
+  const auto config = AudioEncoderOpus::SdpToConfig(
+      SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "10000"}}));
+  ASSERT_TRUE(SetUpSender());
+  RegisterExternalSendCodec(AudioEncoderOpus::MakeAudioEncoder(*config, 107),
+                            107);
+  RunInner(7000, 12000);
+}
+
+TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
+  const auto config = AudioEncoderOpus::SdpToConfig(
+      SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "50000"}}));
+  ASSERT_TRUE(SetUpSender());
+  RegisterExternalSendCodec(AudioEncoderOpus::MakeAudioEncoder(*config, 107),
+                            107);
+  RunInner(40000, 60000);
+}
+
+// Verify that it works when the data to send is mono and the encoder is set to
+// send surround audio.
+TEST_F(AudioCodingModuleTestOldApi, SendingMultiChannelForMonoInput) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kSamplesPerChannel = kSampleRateHz * 10 / 1000;
+
+  audio_format_ = SdpAudioFormat({"multiopus",
+                                  kSampleRateHz,
+                                  6,
+                                  {{"minptime", "10"},
+                                   {"useinbandfec", "1"},
+                                   {"channel_mapping", "0,4,1,2,3,5"},
+                                   {"num_streams", "4"},
+                                   {"coupled_streams", "2"}}});
+
+  RegisterCodec();
+
+  input_frame_.sample_rate_hz_ = kSampleRateHz;
+  input_frame_.num_channels_ = 1;
+  input_frame_.samples_per_channel_ = kSamplesPerChannel;
+  for (size_t k = 0; k < 10; ++k) {
+    ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
+    input_frame_.timestamp_ += kSamplesPerChannel;
+  }
+}
+
+// Verify that it works when the data to send is stereo and the encoder is set
+// to send surround audio.
+TEST_F(AudioCodingModuleTestOldApi, SendingMultiChannelForStereoInput) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
+
+  audio_format_ = SdpAudioFormat({"multiopus",
+                                  kSampleRateHz,
+                                  6,
+                                  {{"minptime", "10"},
+                                   {"useinbandfec", "1"},
+                                   {"channel_mapping", "0,4,1,2,3,5"},
+                                   {"num_streams", "4"},
+                                   {"coupled_streams", "2"}}});
+
+  RegisterCodec();
+
+  input_frame_.sample_rate_hz_ = kSampleRateHz;
+  input_frame_.num_channels_ = 2;
+  input_frame_.samples_per_channel_ = kSamplesPerChannel;
+  for (size_t k = 0; k < 10; ++k) {
+    ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
+    input_frame_.timestamp_ += kSamplesPerChannel;
+  }
+}
+
+// Verify that it works when the data to send is mono and the encoder is set to
+// send stereo audio.
+TEST_F(AudioCodingModuleTestOldApi, SendingStereoForMonoInput) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
+
+  audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 2);
+
+  RegisterCodec();
+
+  input_frame_.sample_rate_hz_ = kSampleRateHz;
+  input_frame_.num_channels_ = 1;
+  input_frame_.samples_per_channel_ = kSamplesPerChannel;
+  for (size_t k = 0; k < 10; ++k) {
+    ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
+    input_frame_.timestamp_ += kSamplesPerChannel;
+  }
+}
+
+// Verify that it works when the data to send is stereo and the encoder is set
+// to send mono audio.
+TEST_F(AudioCodingModuleTestOldApi, SendingMonoForStereoInput) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kSamplesPerChannel = (kSampleRateHz * 10) / 1000;
+
+  audio_format_ = SdpAudioFormat("L16", kSampleRateHz, 1);
+
+  RegisterCodec();
+
+  input_frame_.sample_rate_hz_ = kSampleRateHz;
+  input_frame_.num_channels_ = 1;
+  input_frame_.samples_per_channel_ = kSamplesPerChannel;
+  for (size_t k = 0; k < 10; ++k) {
+    ASSERT_GE(acm_->Add10MsData(input_frame_), 0);
+    input_frame_.timestamp_ += kSamplesPerChannel;
+  }
+}
+
+// The result on the Android platforms is inconsistent for this test case.
+// On android_rel the result is different from android and android arm64 rel.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
+  DISABLED_OpusFromFormat_48khz_20ms_100kbps
+#else
+#define MAYBE_OpusFromFormat_48khz_20ms_100kbps \
+  OpusFromFormat_48khz_20ms_100kbps
+#endif
+TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
+  const auto config = AudioEncoderOpus::SdpToConfig(
+      SdpAudioFormat("opus", 48000, 2, {{"maxaveragebitrate", "100000"}}));
+  ASSERT_TRUE(SetUpSender());
+  RegisterExternalSendCodec(AudioEncoderOpus::MakeAudioEncoder(*config, 107),
+                            107);
+  RunInner(80000, 120000);
+}
+
+TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
+  AudioEncoderPcmU::Config config;
+  config.frame_size_ms = 20;
+  config.num_channels = 1;
+  config.payload_type = 0;
+  AudioEncoderPcmU encoder(config);
+  auto mock_encoder = std::make_unique<MockAudioEncoder>();
+  // Set expectations on the mock encoder and also delegate the calls to the
+  // real encoder.
+  EXPECT_CALL(*mock_encoder, SampleRateHz())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SampleRateHz));
+  EXPECT_CALL(*mock_encoder, NumChannels())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::NumChannels));
+  EXPECT_CALL(*mock_encoder, RtpTimestampRateHz())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::RtpTimestampRateHz));
+  EXPECT_CALL(*mock_encoder, Num10MsFramesInNextPacket())
+      .Times(AtLeast(1))
+      .WillRepeatedly(
+          Invoke(&encoder, &AudioEncoderPcmU::Num10MsFramesInNextPacket));
+  EXPECT_CALL(*mock_encoder, GetTargetBitrate())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));
+  EXPECT_CALL(*mock_encoder, EncodeImpl(_, _, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(
+          &encoder, static_cast<AudioEncoder::EncodedInfo (AudioEncoder::*)(
+                        uint32_t, rtc::ArrayView<const int16_t>, rtc::Buffer*)>(
+                        &AudioEncoderPcmU::Encode)));
+  ASSERT_TRUE(SetUpSender(kTestFileMono32kHz, 32000));
+  ASSERT_NO_FATAL_FAILURE(
+      SetUpTestExternalEncoder(std::move(mock_encoder), config.payload_type));
+  Run("c8d1fc677f33c2022ec5f83c7f302280", "8f9b8750bd80fe26b6cbf6659b89f0f9",
+      50, test::AcmReceiveTestOldApi::kMonoOutput);
+}
+
+// This test fixture is implemented to run ACM and change the desired output
+// frequency during the call. The input packets are simply PCM16b-wb encoded
+// payloads with a constant value of `kSampleValue`. The test fixture itself
+// acts as PacketSource in between the receive test class and the constant-
+// payload packet source class. The output is both written to file, and analyzed
+// in this test fixture.
+class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test,
+                                          public test::PacketSource,
+                                          public test::AudioSink {
+ protected:
+  static const size_t kTestNumPackets = 50;
+  static const int kEncodedSampleRateHz = 16000;
+  static const size_t kPayloadLenSamples = 30 * kEncodedSampleRateHz / 1000;
+  static const int kPayloadType = 108;  // Default payload type for PCM16b-wb.
+
+  AcmSwitchingOutputFrequencyOldApi()
+      : first_output_(true),
+        num_packets_(0),
+        packet_source_(kPayloadLenSamples,
+                       kSampleValue,
+                       kEncodedSampleRateHz,
+                       kPayloadType),
+        output_freq_2_(0),
+        has_toggled_(false) {}
+
+  void Run(int output_freq_1, int output_freq_2, int toggle_period_ms) {
+    // Set up the receiver used to decode the packets and verify the decoded
+    // output.
+    const std::string output_file_name =
+        webrtc::test::OutputPath() +
+        ::testing::UnitTest::GetInstance()
+            ->current_test_info()
+            ->test_case_name() +
+        "_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() +
+        "_output.pcm";
+    test::OutputAudioFile output_file(output_file_name);
+    // Have the output audio sent both to file and to the WriteArray method in
+    // this class.
+    test::AudioSinkFork output(this, &output_file);
+    test::AcmReceiveTestToggleOutputFreqOldApi receive_test(
+        this, &output, output_freq_1, output_freq_2, toggle_period_ms,
+        test::AcmReceiveTestOldApi::kMonoOutput);
+    ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
+    output_freq_2_ = output_freq_2;
+
+    // This is where the actual test is executed.
+    receive_test.Run();
+
+    // Delete output file.
+    remove(output_file_name.c_str());
+  }
+
+  // Inherited from test::PacketSource.
+  std::unique_ptr<test::Packet> NextPacket() override {
+    // Check if it is time to terminate the test. The packet source is of type
+    // ConstantPcmPacketSource, which is infinite, so we must end the test
+    // "manually".
+    if (num_packets_++ > kTestNumPackets) {
+      EXPECT_TRUE(has_toggled_);
+      return NULL;  // Test ended.
+    }
+
+    // Get the next packet from the source.
+    return packet_source_.NextPacket();
+  }
+
+  // Inherited from test::AudioSink.
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
+    // Skip checking the first output frame, since it has a number of zeros
+    // due to how NetEq is initialized.
+    if (first_output_) {
+      first_output_ = false;
+      return true;
+    }
+    for (size_t i = 0; i < num_samples; ++i) {
+      EXPECT_EQ(kSampleValue, audio[i]);
+    }
+    if (num_samples ==
+        static_cast<size_t>(output_freq_2_ / 100))  // Size of 10 ms frame.
+      has_toggled_ = true;
+    // The return value does not say if the values match the expectation, just
+    // that the method could process the samples.
+    return true;
+  }
+
+  const int16_t kSampleValue = 1000;
+  bool first_output_;
+  size_t num_packets_;
+  test::ConstantPcmPacketSource packet_source_;
+  int output_freq_2_;
+  bool has_toggled_;
+};
+
+TEST_F(AcmSwitchingOutputFrequencyOldApi, TestWithoutToggling) {
+  Run(16000, 16000, 1000);
+}
+
+TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo32Khz) {
+  Run(16000, 32000, 1000);
+}
+
+TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle32KhzTo16Khz) {
+  Run(32000, 16000, 1000);
+}
+
+TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle16KhzTo8Khz) {
+  Run(16000, 8000, 1000);
+}
+
+TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
+  Run(8000, 16000, 1000);
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics.cc b/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics.cc
new file mode 100644
index 0000000000..9f3bdadc88
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/call_statistics.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+void CallStatistics::DecodedByNetEq(AudioFrame::SpeechType speech_type,
+                                    bool muted) {
+  ++decoding_stat_.calls_to_neteq;
+  if (muted) {
+    ++decoding_stat_.decoded_muted_output;
+  }
+  switch (speech_type) {
+    case AudioFrame::kNormalSpeech: {
+      ++decoding_stat_.decoded_normal;
+      break;
+    }
+    case AudioFrame::kPLC: {
+      ++decoding_stat_.decoded_neteq_plc;
+      break;
+    }
+    case AudioFrame::kCodecPLC: {
+      ++decoding_stat_.decoded_codec_plc;
+      break;
+    }
+    case AudioFrame::kCNG: {
+      ++decoding_stat_.decoded_cng;
+      break;
+    }
+    case AudioFrame::kPLCCNG: {
+      ++decoding_stat_.decoded_plc_cng;
+      break;
+    }
+    case AudioFrame::kUndefined: {
+      // If the audio is decoded by NetEq, `kUndefined` is not an option.
+      RTC_DCHECK_NOTREACHED();
+    }
+  }
+}
+
+void CallStatistics::DecodedBySilenceGenerator() {
+  ++decoding_stat_.calls_to_silence_generator;
+}
+
+const AudioDecodingCallStats& CallStatistics::GetDecodingStatistics() const {
+  return decoding_stat_;
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics.h b/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics.h
new file mode 100644
index 0000000000..a2db2a29f4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+#define MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+
+#include "api/audio/audio_frame.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+
+//
+// This class is for book keeping of calls to ACM. It is not useful to log API
+// calls which are supposed to be called every 10ms, e.g. PlayoutData10Ms(),
+// however, it is useful to know the number of such calls in a given time
+// interval. The current implementation covers calls to PlayoutData10Ms() with
+// detailed accounting of the decoded speech type.
+//
+// Thread Safety
+// =============
+// Please note that this class in not thread safe. The class must be protected
+// if different APIs are called from different threads.
+//
+
+namespace webrtc {
+
+namespace acm2 {
+
+class CallStatistics {
+ public:
+  CallStatistics() {}
+  ~CallStatistics() {}
+
+  // Call this method to indicate that NetEq engaged in decoding. `speech_type`
+  // is the audio-type according to NetEq, and `muted` indicates if the decoded
+  // frame was produced in muted state.
+  void DecodedByNetEq(AudioFrame::SpeechType speech_type, bool muted);
+
+  // Call this method to indicate that a decoding call resulted in generating
+  // silence, i.e. call to NetEq is bypassed and the output audio is zero.
+  void DecodedBySilenceGenerator();
+
+  // Get statistics for decoding. The statistics include the number of calls to
+  // NetEq and silence generator, as well as the type of speech pulled of off
+  // NetEq, c.f. declaration of AudioDecodingCallStats for detailed description.
+  const AudioDecodingCallStats& GetDecodingStatistics() const;
+
+ private:
+  // Reset the decoding statistics.
+  void ResetDecodingStatistics();
+
+  AudioDecodingCallStats decoding_stat_;
+};
+
+}  // namespace acm2
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics_unittest.cc b/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
new file mode 100644
index 0000000000..b96977b8e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/acm2/call_statistics.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace acm2 {
+
+TEST(CallStatisticsTest, InitializedZero) {
+  CallStatistics call_stats;
+  AudioDecodingCallStats stats;
+
+  stats = call_stats.GetDecodingStatistics();
+  EXPECT_EQ(0, stats.calls_to_neteq);
+  EXPECT_EQ(0, stats.calls_to_silence_generator);
+  EXPECT_EQ(0, stats.decoded_normal);
+  EXPECT_EQ(0, stats.decoded_cng);
+  EXPECT_EQ(0, stats.decoded_neteq_plc);
+  EXPECT_EQ(0, stats.decoded_plc_cng);
+  EXPECT_EQ(0, stats.decoded_muted_output);
+}
+
+TEST(CallStatisticsTest, AllCalls) {
+  CallStatistics call_stats;
+  AudioDecodingCallStats stats;
+
+  call_stats.DecodedBySilenceGenerator();
+  call_stats.DecodedByNetEq(AudioFrame::kNormalSpeech, false);
+  call_stats.DecodedByNetEq(AudioFrame::kPLC, false);
+  call_stats.DecodedByNetEq(AudioFrame::kCodecPLC, false);
+  call_stats.DecodedByNetEq(AudioFrame::kPLCCNG, true);  // Let this be muted.
+  call_stats.DecodedByNetEq(AudioFrame::kCNG, false);
+
+  stats = call_stats.GetDecodingStatistics();
+  EXPECT_EQ(5, stats.calls_to_neteq);
+  EXPECT_EQ(1, stats.calls_to_silence_generator);
+  EXPECT_EQ(1, stats.decoded_normal);
+  EXPECT_EQ(1, stats.decoded_cng);
+  EXPECT_EQ(1, stats.decoded_neteq_plc);
+  EXPECT_EQ(1, stats.decoded_codec_plc);
+  EXPECT_EQ(1, stats.decoded_plc_cng);
+  EXPECT_EQ(1, stats.decoded_muted_output);
+}
+
+}  // namespace acm2
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_coding.gni b/third_party/libwebrtc/modules/audio_coding/audio_coding.gni
new file mode 100644
index 0000000000..3b147091de
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_coding.gni
@@ -0,0 +1,25 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+audio_codec_defines = []
+if (rtc_include_ilbc) {
+  audio_codec_defines += [ "WEBRTC_CODEC_ILBC" ]
+}
+if (rtc_include_opus) {
+  audio_codec_defines += [ "WEBRTC_CODEC_OPUS" ]
+}
+if (rtc_opus_support_120ms_ptime) {
+  audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
+} else {
+  audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
+}
+
+audio_coding_defines = audio_codec_defines
+neteq_defines = audio_codec_defines
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_coding_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_coding_gn/moz.build
new file mode 100644
index 0000000000..03d99c93b8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_coding_gn/moz.build
@@ -0,0 +1,236 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc",
+    "/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc",
+    "/third_party/libwebrtc/modules/audio_coding/acm2/acm_resampler.cc",
+    "/third_party/libwebrtc/modules/audio_coding/acm2/audio_coding_module.cc",
+    "/third_party/libwebrtc/modules/audio_coding/acm2/call_statistics.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_coding_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_coding_module_typedefs_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_coding_module_typedefs_gn/moz.build
new file mode 100644
index 0000000000..2826ed8642
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_coding_module_typedefs_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_coding_module_typedefs_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_coding_opus_common_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_coding_opus_common_gn/moz.build
new file mode 100644
index 0000000000..bf852e8442
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_coding_opus_common_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_coding_opus_common_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_encoder_cng_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_encoder_cng_gn/moz.build
new file mode 100644
index 0000000000..56b6d50f6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_encoder_cng_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_cng_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc
new file mode 100644
index 0000000000..16fd2a1b9a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+
+AudioEncoderRuntimeConfig::AudioEncoderRuntimeConfig() = default;
+
+AudioEncoderRuntimeConfig::AudioEncoderRuntimeConfig(
+    const AudioEncoderRuntimeConfig& other) = default;
+
+AudioEncoderRuntimeConfig::~AudioEncoderRuntimeConfig() = default;
+
+AudioEncoderRuntimeConfig& AudioEncoderRuntimeConfig::operator=(
+    const AudioEncoderRuntimeConfig& other) = default;
+
+bool AudioEncoderRuntimeConfig::operator==(
+    const AudioEncoderRuntimeConfig& other) const {
+  return bitrate_bps == other.bitrate_bps &&
+         frame_length_ms == other.frame_length_ms &&
+         uplink_packet_loss_fraction == other.uplink_packet_loss_fraction &&
+         enable_fec == other.enable_fec && enable_dtx == other.enable_dtx &&
+         num_channels == other.num_channels;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
new file mode 100644
index 0000000000..64163f9118
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -0,0 +1,171 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kEventLogMinBitrateChangeBps = 5000;
+constexpr float kEventLogMinBitrateChangeFraction = 0.25;
+constexpr float kEventLogMinPacketLossChangeFraction = 0.5;
+}  // namespace
+
+AudioNetworkAdaptorImpl::Config::Config() : event_log(nullptr) {}
+
+AudioNetworkAdaptorImpl::Config::~Config() = default;
+
+AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
+    const Config& config,
+    std::unique_ptr<ControllerManager> controller_manager,
+    std::unique_ptr<DebugDumpWriter> debug_dump_writer)
+    : config_(config),
+      controller_manager_(std::move(controller_manager)),
+      debug_dump_writer_(std::move(debug_dump_writer)),
+      event_log_writer_(
+          config.event_log
+              ? new EventLogWriter(config.event_log,
+                                   kEventLogMinBitrateChangeBps,
+                                   kEventLogMinBitrateChangeFraction,
+                                   kEventLogMinPacketLossChangeFraction)
+              : nullptr) {
+  RTC_DCHECK(controller_manager_);
+}
+
+AudioNetworkAdaptorImpl::~AudioNetworkAdaptorImpl() = default;
+
+void AudioNetworkAdaptorImpl::SetUplinkBandwidth(int uplink_bandwidth_bps) {
+  last_metrics_.uplink_bandwidth_bps = uplink_bandwidth_bps;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+void AudioNetworkAdaptorImpl::SetUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  last_metrics_.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) {
+  last_metrics_.rtt_ms = rtt_ms;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.rtt_ms = rtt_ms;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+void AudioNetworkAdaptorImpl::SetTargetAudioBitrate(
+    int target_audio_bitrate_bps) {
+  last_metrics_.target_audio_bitrate_bps = target_audio_bitrate_bps;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+void AudioNetworkAdaptorImpl::SetOverhead(size_t overhead_bytes_per_packet) {
+  last_metrics_.overhead_bytes_per_packet = overhead_bytes_per_packet;
+  DumpNetworkMetrics();
+
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
+  UpdateNetworkMetrics(network_metrics);
+}
+
+AudioEncoderRuntimeConfig AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
+  AudioEncoderRuntimeConfig config;
+  for (auto& controller :
+       controller_manager_->GetSortedControllers(last_metrics_))
+    controller->MakeDecision(&config);
+
+  // Update ANA stats.
+  auto increment_opt = [](absl::optional<uint32_t>& a) {
+    a = a.value_or(0) + 1;
+  };
+  if (prev_config_) {
+    if (config.bitrate_bps != prev_config_->bitrate_bps) {
+      increment_opt(stats_.bitrate_action_counter);
+    }
+    if (config.enable_dtx != prev_config_->enable_dtx) {
+      increment_opt(stats_.dtx_action_counter);
+    }
+    if (config.enable_fec != prev_config_->enable_fec) {
+      increment_opt(stats_.fec_action_counter);
+    }
+    if (config.frame_length_ms && prev_config_->frame_length_ms) {
+      if (*config.frame_length_ms > *prev_config_->frame_length_ms) {
+        increment_opt(stats_.frame_length_increase_counter);
+      } else if (*config.frame_length_ms < *prev_config_->frame_length_ms) {
+        increment_opt(stats_.frame_length_decrease_counter);
+      }
+    }
+    if (config.num_channels != prev_config_->num_channels) {
+      increment_opt(stats_.channel_action_counter);
+    }
+    if (config.uplink_packet_loss_fraction) {
+      stats_.uplink_packet_loss_fraction = *config.uplink_packet_loss_fraction;
+    }
+  }
+  prev_config_ = config;
+
+  if (debug_dump_writer_)
+    debug_dump_writer_->DumpEncoderRuntimeConfig(config, rtc::TimeMillis());
+
+  if (event_log_writer_)
+    event_log_writer_->MaybeLogEncoderConfig(config);
+
+  return config;
+}
+
+void AudioNetworkAdaptorImpl::StartDebugDump(FILE* file_handle) {
+  debug_dump_writer_ = DebugDumpWriter::Create(file_handle);
+}
+
+void AudioNetworkAdaptorImpl::StopDebugDump() {
+  debug_dump_writer_.reset(nullptr);
+}
+
+ANAStats AudioNetworkAdaptorImpl::GetStats() const {
+  return stats_;
+}
+
+void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {
+  if (debug_dump_writer_)
+    debug_dump_writer_->DumpNetworkMetrics(last_metrics_, rtc::TimeMillis());
+}
+
+void AudioNetworkAdaptorImpl::UpdateNetworkMetrics(
+    const Controller::NetworkMetrics& network_metrics) {
+  for (auto& controller : controller_manager_->GetControllers())
+    controller->UpdateNetworkMetrics(network_metrics);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
new file mode 100644
index 0000000000..664e76bda5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+
+class ControllerManager;
+class EventLogWriter;
+class RtcEventLog;
+
+class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
+ public:
+  struct Config {
+    Config();
+    ~Config();
+    RtcEventLog* event_log;
+  };
+
+  AudioNetworkAdaptorImpl(
+      const Config& config,
+      std::unique_ptr<ControllerManager> controller_manager,
+      std::unique_ptr<DebugDumpWriter> debug_dump_writer = nullptr);
+
+  ~AudioNetworkAdaptorImpl() override;
+
+  AudioNetworkAdaptorImpl(const AudioNetworkAdaptorImpl&) = delete;
+  AudioNetworkAdaptorImpl& operator=(const AudioNetworkAdaptorImpl&) = delete;
+
+  void SetUplinkBandwidth(int uplink_bandwidth_bps) override;
+
+  void SetUplinkPacketLossFraction(float uplink_packet_loss_fraction) override;
+
+  void SetRtt(int rtt_ms) override;
+
+  void SetTargetAudioBitrate(int target_audio_bitrate_bps) override;
+
+  void SetOverhead(size_t overhead_bytes_per_packet) override;
+
+  AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() override;
+
+  void StartDebugDump(FILE* file_handle) override;
+
+  void StopDebugDump() override;
+
+  ANAStats GetStats() const override;
+
+ private:
+  void DumpNetworkMetrics();
+
+  void UpdateNetworkMetrics(const Controller::NetworkMetrics& network_metrics);
+
+  const Config config_;
+
+  std::unique_ptr<ControllerManager> controller_manager_;
+
+  std::unique_ptr<DebugDumpWriter> debug_dump_writer_;
+
+  const std::unique_ptr<EventLogWriter> event_log_writer_;
+
+  Controller::NetworkMetrics last_metrics_;
+
+  absl::optional<AudioEncoderRuntimeConfig> prev_config_;
+
+  ANAStats stats_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
new file mode 100644
index 0000000000..70a50d6de7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -0,0 +1,306 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+
+#include <utility>
+#include <vector>
+
+#include "api/rtc_event_log/rtc_event.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
+#include "rtc_base/fake_clock.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+namespace {
+
+constexpr size_t kNumControllers = 2;
+
+constexpr int64_t kClockInitialTimeMs = 12345678;
+
+MATCHER_P(NetworkMetricsIs, metric, "") {
+  return arg.uplink_bandwidth_bps == metric.uplink_bandwidth_bps &&
+         arg.target_audio_bitrate_bps == metric.target_audio_bitrate_bps &&
+         arg.rtt_ms == metric.rtt_ms &&
+         arg.overhead_bytes_per_packet == metric.overhead_bytes_per_packet &&
+         arg.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction;
+}
+
+MATCHER_P(IsRtcEventAnaConfigEqualTo, config, "") {
+  if (arg->GetType() != RtcEvent::Type::AudioNetworkAdaptation) {
+    return false;
+  }
+  auto ana_event = static_cast<RtcEventAudioNetworkAdaptation*>(arg);
+  return ana_event->config() == config;
+}
+
+MATCHER_P(EncoderRuntimeConfigIs, config, "") {
+  return arg.bitrate_bps == config.bitrate_bps &&
+         arg.frame_length_ms == config.frame_length_ms &&
+         arg.uplink_packet_loss_fraction ==
+             config.uplink_packet_loss_fraction &&
+         arg.enable_fec == config.enable_fec &&
+         arg.enable_dtx == config.enable_dtx &&
+         arg.num_channels == config.num_channels;
+}
+
+struct AudioNetworkAdaptorStates {
+  std::unique_ptr<AudioNetworkAdaptorImpl> audio_network_adaptor;
+  std::vector<std::unique_ptr<MockController>> mock_controllers;
+  std::unique_ptr<MockRtcEventLog> event_log;
+  MockDebugDumpWriter* mock_debug_dump_writer;
+};
+
+AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() {
+  AudioNetworkAdaptorStates states;
+  std::vector<Controller*> controllers;
+  for (size_t i = 0; i < kNumControllers; ++i) {
+    auto controller =
+        std::unique_ptr<MockController>(new NiceMock<MockController>());
+    EXPECT_CALL(*controller, Die());
+    controllers.push_back(controller.get());
+    states.mock_controllers.push_back(std::move(controller));
+  }
+
+  auto controller_manager = std::unique_ptr<MockControllerManager>(
+      new NiceMock<MockControllerManager>());
+
+  EXPECT_CALL(*controller_manager, Die());
+  EXPECT_CALL(*controller_manager, GetControllers())
+      .WillRepeatedly(Return(controllers));
+  EXPECT_CALL(*controller_manager, GetSortedControllers(_))
+      .WillRepeatedly(Return(controllers));
+
+  states.event_log.reset(new NiceMock<MockRtcEventLog>());
+
+  auto debug_dump_writer =
+      std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
+  EXPECT_CALL(*debug_dump_writer, Die());
+  states.mock_debug_dump_writer = debug_dump_writer.get();
+
+  AudioNetworkAdaptorImpl::Config config;
+  config.event_log = states.event_log.get();
+  // AudioNetworkAdaptorImpl governs the lifetime of controller manager.
+  states.audio_network_adaptor.reset(new AudioNetworkAdaptorImpl(
+      config, std::move(controller_manager), std::move(debug_dump_writer)));
+
+  return states;
+}
+
+void SetExpectCallToUpdateNetworkMetrics(
+    const std::vector<std::unique_ptr<MockController>>& controllers,
+    const Controller::NetworkMetrics& check) {
+  for (auto& mock_controller : controllers) {
+    EXPECT_CALL(*mock_controller,
+                UpdateNetworkMetrics(NetworkMetricsIs(check)));
+  }
+}
+
+}  // namespace
+
+TEST(AudioNetworkAdaptorImplTest,
+     UpdateNetworkMetricsIsCalledOnSetUplinkBandwidth) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr int kBandwidth = 16000;
+  Controller::NetworkMetrics check;
+  check.uplink_bandwidth_bps = kBandwidth;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     UpdateNetworkMetricsIsCalledOnSetUplinkPacketLossFraction) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr float kPacketLoss = 0.7f;
+  Controller::NetworkMetrics check;
+  check.uplink_packet_loss_fraction = kPacketLoss;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
+}
+
+TEST(AudioNetworkAdaptorImplTest, UpdateNetworkMetricsIsCalledOnSetRtt) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr int kRtt = 100;
+  Controller::NetworkMetrics check;
+  check.rtt_ms = kRtt;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetRtt(kRtt);
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     UpdateNetworkMetricsIsCalledOnSetTargetAudioBitrate) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr int kTargetAudioBitrate = 15000;
+  Controller::NetworkMetrics check;
+  check.target_audio_bitrate_bps = kTargetAudioBitrate;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
+}
+
+TEST(AudioNetworkAdaptorImplTest, UpdateNetworkMetricsIsCalledOnSetOverhead) {
+  auto states = CreateAudioNetworkAdaptor();
+  constexpr size_t kOverhead = 64;
+  Controller::NetworkMetrics check;
+  check.overhead_bytes_per_packet = kOverhead;
+  SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+  states.audio_network_adaptor->SetOverhead(kOverhead);
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     MakeDecisionIsCalledOnGetEncoderRuntimeConfig) {
+  auto states = CreateAudioNetworkAdaptor();
+  for (auto& mock_controller : states.mock_controllers)
+    EXPECT_CALL(*mock_controller, MakeDecision(_));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
+      "Enabled/");
+  rtc::ScopedFakeClock fake_clock;
+  fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
+  auto states = CreateAudioNetworkAdaptor();
+  AudioEncoderRuntimeConfig config;
+  config.bitrate_bps = 32000;
+  config.enable_fec = true;
+
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config));
+
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpEncoderRuntimeConfig(EncoderRuntimeConfigIs(config),
+                                       kClockInitialTimeMs));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+}
+
+TEST(AudioNetworkAdaptorImplTest,
+     DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
+  rtc::ScopedFakeClock fake_clock;
+  fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
+
+  auto states = CreateAudioNetworkAdaptor();
+
+  constexpr int kBandwidth = 16000;
+  constexpr float kPacketLoss = 0.7f;
+  constexpr int kRtt = 100;
+  constexpr int kTargetAudioBitrate = 15000;
+  constexpr size_t kOverhead = 64;
+
+  Controller::NetworkMetrics check;
+  check.uplink_bandwidth_bps = kBandwidth;
+  int64_t timestamp_check = kClockInitialTimeMs;
+
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(100));
+  timestamp_check += 100;
+  check.uplink_packet_loss_fraction = kPacketLoss;
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(50));
+  timestamp_check += 50;
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(200));
+  timestamp_check += 200;
+  check.rtt_ms = kRtt;
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetRtt(kRtt);
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(150));
+  timestamp_check += 150;
+  check.target_audio_bitrate_bps = kTargetAudioBitrate;
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(50));
+  timestamp_check += 50;
+  check.overhead_bytes_per_packet = kOverhead;
+  EXPECT_CALL(*states.mock_debug_dump_writer,
+              DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+  states.audio_network_adaptor->SetOverhead(kOverhead);
+}
+
+TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
+      "Enabled/");
+  auto states = CreateAudioNetworkAdaptor();
+
+  AudioEncoderRuntimeConfig config;
+  config.bitrate_bps = 32000;
+  config.enable_fec = true;
+
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config));
+
+  EXPECT_CALL(*states.event_log, LogProxy(IsRtcEventAnaConfigEqualTo(config)))
+      .Times(1);
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+}
+
+TEST(AudioNetworkAdaptorImplTest, TestANAStats) {
+  auto states = CreateAudioNetworkAdaptor();
+
+  // Simulate some adaptation, otherwise the stats will not show anything.
+  AudioEncoderRuntimeConfig config1, config2;
+  config1.bitrate_bps = 32000;
+  config1.num_channels = 2;
+  config1.enable_fec = true;
+  config1.enable_dtx = true;
+  config1.frame_length_ms = 120;
+  config1.uplink_packet_loss_fraction = 0.1f;
+  config2.bitrate_bps = 16000;
+  config2.num_channels = 1;
+  config2.enable_fec = false;
+  config2.enable_dtx = false;
+  config2.frame_length_ms = 60;
+  config1.uplink_packet_loss_fraction = 0.1f;
+
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config1));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config2));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+  EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+      .WillOnce(SetArgPointee<0>(config1));
+  states.audio_network_adaptor->GetEncoderRuntimeConfig();
+
+  auto ana_stats = states.audio_network_adaptor->GetStats();
+
+  EXPECT_EQ(ana_stats.bitrate_action_counter, 2u);
+  EXPECT_EQ(ana_stats.channel_action_counter, 2u);
+  EXPECT_EQ(ana_stats.dtx_action_counter, 2u);
+  EXPECT_EQ(ana_stats.fec_action_counter, 2u);
+  EXPECT_EQ(ana_stats.frame_length_increase_counter, 1u);
+  EXPECT_EQ(ana_stats.frame_length_decrease_counter, 1u);
+  EXPECT_EQ(ana_stats.uplink_packet_loss_fraction, 0.1f);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
new file mode 100644
index 0000000000..88ca38d074
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace audio_network_adaptor {
+
+BitrateController::Config::Config(int initial_bitrate_bps,
+                                  int initial_frame_length_ms,
+                                  int fl_increase_overhead_offset,
+                                  int fl_decrease_overhead_offset)
+    : initial_bitrate_bps(initial_bitrate_bps),
+      initial_frame_length_ms(initial_frame_length_ms),
+      fl_increase_overhead_offset(fl_increase_overhead_offset),
+      fl_decrease_overhead_offset(fl_decrease_overhead_offset) {}
+
+BitrateController::Config::~Config() = default;
+
+BitrateController::BitrateController(const Config& config)
+    : config_(config),
+      bitrate_bps_(config_.initial_bitrate_bps),
+      frame_length_ms_(config_.initial_frame_length_ms) {
+  RTC_DCHECK_GT(bitrate_bps_, 0);
+  RTC_DCHECK_GT(frame_length_ms_, 0);
+}
+
+BitrateController::~BitrateController() = default;
+
+void BitrateController::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.target_audio_bitrate_bps)
+    target_audio_bitrate_bps_ = network_metrics.target_audio_bitrate_bps;
+  if (network_metrics.overhead_bytes_per_packet) {
+    RTC_DCHECK_GT(*network_metrics.overhead_bytes_per_packet, 0);
+    overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet;
+  }
+}
+
+void BitrateController::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  // Decision on `bitrate_bps` should not have been made.
+  RTC_DCHECK(!config->bitrate_bps);
+  if (target_audio_bitrate_bps_ && overhead_bytes_per_packet_) {
+    if (config->frame_length_ms)
+      frame_length_ms_ = *config->frame_length_ms;
+    int offset = config->last_fl_change_increase
+                     ? config_.fl_increase_overhead_offset
+                     : config_.fl_decrease_overhead_offset;
+    // Check that
+    // -(*overhead_bytes_per_packet_) <= offset <= (*overhead_bytes_per_packet_)
+    RTC_DCHECK_GE(*overhead_bytes_per_packet_, -offset);
+    RTC_DCHECK_LE(offset, *overhead_bytes_per_packet_);
+    int overhead_rate_bps = static_cast<int>(
+        (*overhead_bytes_per_packet_ + offset) * 8 * 1000 / frame_length_ms_);
+    bitrate_bps_ = std::max(0, *target_audio_bitrate_bps_ - overhead_rate_bps);
+  }
+  config->bitrate_bps = bitrate_bps_;
+}
+
+}  // namespace audio_network_adaptor
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
new file mode 100644
index 0000000000..c1032146cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+namespace audio_network_adaptor {
+
+class BitrateController final : public Controller {
+ public:
+  struct Config {
+    Config(int initial_bitrate_bps,
+           int initial_frame_length_ms,
+           int fl_increase_overhead_offset,
+           int fl_decrease_overhead_offset);
+    ~Config();
+    int initial_bitrate_bps;
+    int initial_frame_length_ms;
+    int fl_increase_overhead_offset;
+    int fl_decrease_overhead_offset;
+  };
+
+  explicit BitrateController(const Config& config);
+
+  ~BitrateController() override;
+
+  BitrateController(const BitrateController&) = delete;
+  BitrateController& operator=(const BitrateController&) = delete;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  const Config config_;
+  int bitrate_bps_;
+  int frame_length_ms_;
+  absl::optional<int> target_audio_bitrate_bps_;
+  absl::optional<size_t> overhead_bytes_per_packet_;
+};
+
+}  // namespace audio_network_adaptor
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
new file mode 100644
index 0000000000..9c593b818b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
@@ -0,0 +1,236 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace audio_network_adaptor {
+
+namespace {
+
+void UpdateNetworkMetrics(
+    BitrateController* controller,
+    const absl::optional<int>& target_audio_bitrate_bps,
+    const absl::optional<size_t>& overhead_bytes_per_packet) {
+  // UpdateNetworkMetrics can accept multiple network metric updates at once.
+  // However, currently, the most used case is to update one metric at a time.
+  // To reflect this fact, we separate the calls.
+  if (target_audio_bitrate_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  if (overhead_bytes_per_packet) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+}
+
+void CheckDecision(BitrateController* controller,
+                   const absl::optional<int>& frame_length_ms,
+                   int expected_bitrate_bps) {
+  AudioEncoderRuntimeConfig config;
+  config.frame_length_ms = frame_length_ms;
+  controller->MakeDecision(&config);
+  EXPECT_EQ(expected_bitrate_bps, config.bitrate_bps);
+}
+
+}  // namespace
+
+// These tests are named AnaBitrateControllerTest to distinguish from
+// BitrateControllerTest in
+// modules/bitrate_controller/bitrate_controller_unittest.cc.
+
+TEST(AnaBitrateControllerTest, OutputInitValueWhenTargetBitrateUnknown) {
+  constexpr int kInitialBitrateBps = 32000;
+  constexpr int kInitialFrameLengthMs = 20;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  BitrateController controller(BitrateController::Config(
+      kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
+  UpdateNetworkMetrics(&controller, absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, OutputInitValueWhenOverheadUnknown) {
+  constexpr int kInitialBitrateBps = 32000;
+  constexpr int kInitialFrameLengthMs = 20;
+  constexpr int kTargetBitrateBps = 48000;
+  BitrateController controller(BitrateController::Config(
+      kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, absl::nullopt);
+  CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, ChangeBitrateOnTargetBitrateChanged) {
+  constexpr int kInitialFrameLengthMs = 20;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  // Frame length unchanged, bitrate changes in accordance with
+  // `metrics.target_audio_bitrate_bps` and `metrics.overhead_bytes_per_packet`.
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kInitialFrameLengthMs, kBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, UpdateMultipleNetworkMetricsAtOnce) {
+  // This test is similar to ChangeBitrateOnTargetBitrateChanged. But instead of
+  // using ::UpdateNetworkMetrics(...), which calls
+  // BitrateController::UpdateNetworkMetrics(...) multiple times, we
+  // we call it only once. This is to verify that
+  // BitrateController::UpdateNetworkMetrics(...) can handle multiple
+  // network updates at once. This is, however, not a common use case in current
+  // audio_network_adaptor_impl.cc.
+  constexpr int kInitialFrameLengthMs = 20;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.target_audio_bitrate_bps = kTargetBitrateBps;
+  network_metrics.overhead_bytes_per_packet = kOverheadBytesPerPacket;
+  controller.UpdateNetworkMetrics(network_metrics);
+  CheckDecision(&controller, kInitialFrameLengthMs, kBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, TreatUnknownFrameLengthAsFrameLengthUnchanged) {
+  constexpr int kInitialFrameLengthMs = 20;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, absl::nullopt, kBitrateBps);
+}
+
+TEST(AnaBitrateControllerTest, IncreaseBitrateOnFrameLengthIncreased) {
+  constexpr int kInitialFrameLengthMs = 20;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, absl::nullopt, kBitrateBps);
+
+  constexpr int kFrameLengthMs = 60;
+  constexpr size_t kPacketOverheadRateDiff =
+      kOverheadBytesPerPacket * 8 * 1000 / 20 -
+      kOverheadBytesPerPacket * 8 * 1000 / 60;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kFrameLengthMs,
+                kBitrateBps + kPacketOverheadRateDiff);
+}
+
+TEST(AnaBitrateControllerTest, DecreaseBitrateOnFrameLengthDecreased) {
+  constexpr int kInitialFrameLengthMs = 60;
+  BitrateController controller(
+      BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
+
+  constexpr int kTargetBitrateBps = 48000;
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
+                                                      1000 /
+                                                      kInitialFrameLengthMs;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, absl::nullopt, kBitrateBps);
+
+  constexpr int kFrameLengthMs = 20;
+  constexpr size_t kPacketOverheadRateDiff =
+      kOverheadBytesPerPacket * 8 * 1000 / 20 -
+      kOverheadBytesPerPacket * 8 * 1000 / 60;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kFrameLengthMs,
+                kBitrateBps - kPacketOverheadRateDiff);
+}
+
+TEST(AnaBitrateControllerTest, BitrateNeverBecomesNegative) {
+  BitrateController controller(BitrateController::Config(32000, 20, 0, 0));
+  constexpr size_t kOverheadBytesPerPacket = 64;
+  constexpr int kFrameLengthMs = 60;
+  // Set a target rate smaller than overhead rate, the bitrate is bounded by 0.
+  constexpr int kTargetBitrateBps =
+      kOverheadBytesPerPacket * 8 * 1000 / kFrameLengthMs - 1;
+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
+  CheckDecision(&controller, kFrameLengthMs, 0);
+}
+
+TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
+  BitrateController controller(BitrateController::Config(32000, 20, 0, 0));
+
+  // Start from an arbitrary overall bitrate.
+  int overall_bitrate = 34567;
+  size_t overhead_bytes_per_packet = 64;
+  int frame_length_ms = 20;
+  int current_bitrate = rtc::checked_cast<int>(
+      overall_bitrate - overhead_bytes_per_packet * 8 * 1000 / frame_length_ms);
+
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: increase overall bitrate.
+  overall_bitrate += 100;
+  current_bitrate += 100;
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: change frame length.
+  frame_length_ms = 60;
+  current_bitrate +=
+      rtc::checked_cast<int>(overhead_bytes_per_packet * 8 * 1000 / 20 -
+                             overhead_bytes_per_packet * 8 * 1000 / 60);
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: change overhead.
+  overhead_bytes_per_packet -= 30;
+  current_bitrate += 30 * 8 * 1000 / frame_length_ms;
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: change frame length.
+  frame_length_ms = 20;
+  current_bitrate -=
+      rtc::checked_cast<int>(overhead_bytes_per_packet * 8 * 1000 / 20 -
+                             overhead_bytes_per_packet * 8 * 1000 / 60);
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+
+  // Next: decrease overall bitrate and frame length.
+  overall_bitrate -= 100;
+  current_bitrate -= 100;
+  frame_length_ms = 60;
+  current_bitrate +=
+      rtc::checked_cast<int>(overhead_bytes_per_packet * 8 * 1000 / 20 -
+                             overhead_bytes_per_packet * 8 * 1000 / 60);
+
+  UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
+  CheckDecision(&controller, frame_length_ms, current_bitrate);
+}
+
+}  // namespace audio_network_adaptor
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
new file mode 100644
index 0000000000..2ef2f4c4d8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+ChannelController::Config::Config(size_t num_encoder_channels,
+                                  size_t intial_channels_to_encode,
+                                  int channel_1_to_2_bandwidth_bps,
+                                  int channel_2_to_1_bandwidth_bps)
+    : num_encoder_channels(num_encoder_channels),
+      intial_channels_to_encode(intial_channels_to_encode),
+      channel_1_to_2_bandwidth_bps(channel_1_to_2_bandwidth_bps),
+      channel_2_to_1_bandwidth_bps(channel_2_to_1_bandwidth_bps) {}
+
+ChannelController::ChannelController(const Config& config)
+    : config_(config), channels_to_encode_(config_.intial_channels_to_encode) {
+  RTC_DCHECK_GT(config_.intial_channels_to_encode, 0lu);
+  // Currently, we require `intial_channels_to_encode` to be <= 2.
+  RTC_DCHECK_LE(config_.intial_channels_to_encode, 2lu);
+  RTC_DCHECK_GE(config_.num_encoder_channels,
+                config_.intial_channels_to_encode);
+}
+
+ChannelController::~ChannelController() = default;
+
+void ChannelController::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.uplink_bandwidth_bps)
+    uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+}
+
+void ChannelController::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  // Decision on `num_channels` should not have been made.
+  RTC_DCHECK(!config->num_channels);
+
+  if (uplink_bandwidth_bps_) {
+    if (channels_to_encode_ == 2 &&
+        *uplink_bandwidth_bps_ <= config_.channel_2_to_1_bandwidth_bps) {
+      channels_to_encode_ = 1;
+    } else if (channels_to_encode_ == 1 &&
+               *uplink_bandwidth_bps_ >= config_.channel_1_to_2_bandwidth_bps) {
+      channels_to_encode_ =
+          std::min(static_cast<size_t>(2), config_.num_encoder_channels);
+    }
+  }
+  config->num_channels = channels_to_encode_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
new file mode 100644
index 0000000000..3cd4bb7dec
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+
+class ChannelController final : public Controller {
+ public:
+  struct Config {
+    Config(size_t num_encoder_channels,
+           size_t intial_channels_to_encode,
+           int channel_1_to_2_bandwidth_bps,
+           int channel_2_to_1_bandwidth_bps);
+    size_t num_encoder_channels;
+    size_t intial_channels_to_encode;
+    // Uplink bandwidth above which the number of encoded channels should switch
+    // from 1 to 2.
+    int channel_1_to_2_bandwidth_bps;
+    // Uplink bandwidth below which the number of encoded channels should switch
+    // from 2 to 1.
+    int channel_2_to_1_bandwidth_bps;
+  };
+
+  explicit ChannelController(const Config& config);
+
+  ~ChannelController() override;
+
+  ChannelController(const ChannelController&) = delete;
+  ChannelController& operator=(const ChannelController&) = delete;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  const Config config_;
+  size_t channels_to_encode_;
+  absl::optional<int> uplink_bandwidth_bps_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
new file mode 100644
index 0000000000..21504bcec0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kNumChannels = 2;
+constexpr int kChannel1To2BandwidthBps = 31000;
+constexpr int kChannel2To1BandwidthBps = 29000;
+constexpr int kMediumBandwidthBps =
+    (kChannel1To2BandwidthBps + kChannel2To1BandwidthBps) / 2;
+
+std::unique_ptr<ChannelController> CreateChannelController(int init_channels) {
+  std::unique_ptr<ChannelController> controller(
+      new ChannelController(ChannelController::Config(
+          kNumChannels, init_channels, kChannel1To2BandwidthBps,
+          kChannel2To1BandwidthBps)));
+  return controller;
+}
+
+void CheckDecision(ChannelController* controller,
+                   const absl::optional<int>& uplink_bandwidth_bps,
+                   size_t expected_num_channels) {
+  if (uplink_bandwidth_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  AudioEncoderRuntimeConfig config;
+  controller->MakeDecision(&config);
+  EXPECT_EQ(expected_num_channels, config.num_channels);
+}
+
+}  // namespace
+
+TEST(ChannelControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
+  constexpr int kInitChannels = 2;
+  auto controller = CreateChannelController(kInitChannels);
+  CheckDecision(controller.get(), absl::nullopt, kInitChannels);
+}
+
+TEST(ChannelControllerTest, SwitchTo2ChannelsOnHighUplinkBandwidth) {
+  constexpr int kInitChannels = 1;
+  auto controller = CreateChannelController(kInitChannels);
+  // Use high bandwidth to check output switch to 2.
+  CheckDecision(controller.get(), kChannel1To2BandwidthBps, 2);
+}
+
+TEST(ChannelControllerTest, SwitchTo1ChannelOnLowUplinkBandwidth) {
+  constexpr int kInitChannels = 2;
+  auto controller = CreateChannelController(kInitChannels);
+  // Use low bandwidth to check output switch to 1.
+  CheckDecision(controller.get(), kChannel2To1BandwidthBps, 1);
+}
+
+TEST(ChannelControllerTest, Maintain1ChannelOnMediumUplinkBandwidth) {
+  constexpr int kInitChannels = 1;
+  auto controller = CreateChannelController(kInitChannels);
+  // Use between-thresholds bandwidth to check output remains at 1.
+  CheckDecision(controller.get(), kMediumBandwidthBps, 1);
+}
+
+TEST(ChannelControllerTest, Maintain2ChannelsOnMediumUplinkBandwidth) {
+  constexpr int kInitChannels = 2;
+  auto controller = CreateChannelController(kInitChannels);
+  // Use between-thresholds bandwidth to check output remains at 2.
+  CheckDecision(controller.get(), kMediumBandwidthBps, 2);
+}
+
+TEST(ChannelControllerTest, CheckBehaviorOnChangingUplinkBandwidth) {
+  constexpr int kInitChannels = 1;
+  auto controller = CreateChannelController(kInitChannels);
+
+  // Use between-thresholds bandwidth to check output remains at 1.
+  CheckDecision(controller.get(), kMediumBandwidthBps, 1);
+
+  // Use high bandwidth to check output switch to 2.
+  CheckDecision(controller.get(), kChannel1To2BandwidthBps, 2);
+
+  // Use between-thresholds bandwidth to check output remains at 2.
+  CheckDecision(controller.get(), kMediumBandwidthBps, 2);
+
+  // Use low bandwidth to check output switch to 1.
+  CheckDecision(controller.get(), kChannel2To1BandwidthBps, 1);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/config.proto b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/config.proto
new file mode 100644
index 0000000000..a815451993
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/config.proto
@@ -0,0 +1,196 @@
+syntax = "proto2";
+
+package webrtc.audio_network_adaptor.config;
+
+option optimize_for = LITE_RUNTIME;
+option java_package = "org.webrtc.AudioNetworkAdaptor";
+option java_outer_classname = "Config";
+option objc_class_prefix = "WANA";
+
+message FecController {
+  message Threshold {
+    // Threshold defines a curve in the bandwidth/packet-loss domain. The
+    // curve is characterized by the two conjunction points: A and B.
+    //
+    // packet ^  |
+    //  loss  | A|
+    //        |   \        A: (low_bandwidth_bps, low_bandwidth_packet_loss)
+    //        |    \       B: (high_bandwidth_bps, high_bandwidth_packet_loss)
+    //        |    B\________
+    //        |---------------> bandwidth
+    optional int32 low_bandwidth_bps = 1;
+    optional float low_bandwidth_packet_loss = 2;
+    optional int32 high_bandwidth_bps = 3;
+    optional float high_bandwidth_packet_loss = 4;
+  }
+
+  // `fec_enabling_threshold` defines a curve, above which FEC should be
+  // enabled. `fec_disabling_threshold` defines a curve, under which FEC
+  // should be disabled. See below
+  //
+  // packet-loss ^   |  |
+  //             |   |  |   FEC
+  //             |    \  \   ON
+  //             | FEC \  \_______ fec_enabling_threshold
+  //             | OFF  \_________ fec_disabling_threshold
+  //             |-----------------> bandwidth
+  optional Threshold fec_enabling_threshold = 1;
+  optional Threshold fec_disabling_threshold = 2;
+
+  // `time_constant_ms` is the time constant for an exponential filter, which
+  // is used for smoothing the packet loss fraction.
+  optional int32 time_constant_ms = 3;
+}
+
+message FecControllerRplrBased {
+  message Threshold {
+    // Threshold defines a curve in the bandwidth/recoverable-packet-loss
+    // domain.
+    // The curve is characterized by the two conjunction points: A and B.
+    //
+    // recoverable ^
+    // packet      |  |
+    //  loss       | A|
+    //             |   \        A: (low_bandwidth_bps,
+    //             |    \           low_bandwidth_recoverable_packet_loss)
+    //             |     \       B: (high_bandwidth_bps,
+    //             |      \          high_bandwidth_recoverable_packet_loss)
+    //             |      B\________
+    //             |---------------> bandwidth
+    optional int32 low_bandwidth_bps = 1;
+    optional float low_bandwidth_recoverable_packet_loss = 2;
+    optional int32 high_bandwidth_bps = 3;
+    optional float high_bandwidth_recoverable_packet_loss = 4;
+  }
+
+  // `fec_enabling_threshold` defines a curve, above which FEC should be
+  // enabled. `fec_disabling_threshold` defines a curve, under which FEC
+  // should be disabled. See below
+  //
+  // packet-loss ^   |  |
+  //             |   |  |   FEC
+  //             |    \  \   ON
+  //             | FEC \  \_______ fec_enabling_threshold
+  //             | OFF  \_________ fec_disabling_threshold
+  //             |-----------------> bandwidth
+  optional Threshold fec_enabling_threshold = 1;
+  optional Threshold fec_disabling_threshold = 2;
+}
+
+message FrameLengthController {
+  // Uplink packet loss fraction below which frame length can increase.
+  optional float fl_increasing_packet_loss_fraction = 1;
+
+  // Uplink packet loss fraction above which frame length should decrease.
+  optional float fl_decreasing_packet_loss_fraction = 2;
+
+  // Uplink bandwidth below which frame length can switch from 20ms to 60ms.
+  optional int32 fl_20ms_to_60ms_bandwidth_bps = 3;
+
+  // Uplink bandwidth above which frame length should switch from 60ms to 20ms.
+  optional int32 fl_60ms_to_20ms_bandwidth_bps = 4;
+
+  // Uplink bandwidth below which frame length can switch from 60ms to 120ms.
+  optional int32 fl_60ms_to_120ms_bandwidth_bps = 5;
+
+  // Uplink bandwidth above which frame length should switch from 120ms to 60ms.
+  optional int32 fl_120ms_to_60ms_bandwidth_bps = 6;
+
+  // Offset to apply to the per-packet overhead when increasing frame length.
+  optional int32 fl_increase_overhead_offset = 7;
+
+  // Offset to apply to the per-packet overhead when decreasing frame length.
+  optional int32 fl_decrease_overhead_offset = 8;
+
+  // Uplink bandwidth below which frame length can switch from 20ms to 40ms. In
+  // current implementation, defining this will invalidate
+  // fl_20ms_to_60ms_bandwidth_bps.
+  optional int32 fl_20ms_to_40ms_bandwidth_bps = 9;
+
+  // Uplink bandwidth above which frame length should switch from 40ms to 20ms.
+  optional int32 fl_40ms_to_20ms_bandwidth_bps = 10;
+
+  // Uplink bandwidth below which frame length can switch from 40ms to 60ms.
+  optional int32 fl_40ms_to_60ms_bandwidth_bps = 11;
+
+  // Uplink bandwidth above which frame length should switch from 60ms to 40ms.
+  // In current implementation, defining this will invalidate
+  // fl_60ms_to_20ms_bandwidth_bps.
+  optional int32 fl_60ms_to_40ms_bandwidth_bps = 12;
+}
+
+message FrameLengthControllerV2 {
+  // FrameLengthControllerV2 chooses the frame length by taking the target
+  // bitrate and subtracting the overhead bitrate to obtain the remaining
+  // bitrate for the payload. The chosen frame length is the shortest possible
+  // where the payload bitrate is more than `min_payload_bitrate_bps`.
+  optional int32 min_payload_bitrate_bps = 1;
+
+  // If true, uses the stable target bitrate to decide the frame length. This
+  // will result in less frame length toggling but spending more time at longer
+  // frame lengths compared to using the normal target bitrate.
+  optional bool use_slow_adaptation = 2;
+}
+
+message ChannelController {
+  // Uplink bandwidth above which the number of encoded channels should switch
+  // from 1 to 2.
+  optional int32 channel_1_to_2_bandwidth_bps = 1;
+
+  // Uplink bandwidth below which the number of encoded channels should switch
+  // from 2 to 1.
+  optional int32 channel_2_to_1_bandwidth_bps = 2;
+}
+
+message DtxController {
+  // Uplink bandwidth below which DTX should be switched on.
+  optional int32 dtx_enabling_bandwidth_bps = 1;
+
+  // Uplink bandwidth above which DTX should be switched off.
+  optional int32 dtx_disabling_bandwidth_bps = 2;
+}
+
+message BitrateController {
+  // Offset to apply to per-packet overhead when the frame length is increased.
+  optional int32 fl_increase_overhead_offset = 1;
+  // Offset to apply to per-packet overhead when the frame length is decreased.
+  optional int32 fl_decrease_overhead_offset = 2;
+}
+
+message Controller {
+  message ScoringPoint {
+    // `ScoringPoint` is a subspace of network condition. It is used for
+    // comparing the significance of controllers.
+    optional int32 uplink_bandwidth_bps = 1;
+    optional float uplink_packet_loss_fraction = 2;
+  }
+
+  // The distance from `scoring_point` to a given network condition defines
+  // the significance of this controller with respect that network condition.
+  // Shorter distance means higher significance. The significances of
+  // controllers determine their order in the processing pipeline. Controllers
+  // without `scoring_point` follow their default order in
+  // `ControllerManager::controllers`.
+  optional ScoringPoint scoring_point = 1;
+
+  oneof controller {
+    FecController fec_controller = 21;
+    FrameLengthController frame_length_controller = 22;
+    ChannelController channel_controller = 23;
+    DtxController dtx_controller = 24;
+    BitrateController bitrate_controller = 25;
+    FecControllerRplrBased fec_controller_rplr_based = 26;
+    FrameLengthControllerV2 frame_length_controller_v2 = 27;
+  }
+}
+
+message ControllerManager {
+  repeated Controller controllers = 1;
+
+  // Least time since last reordering for a new reordering to be made.
+  optional int32 min_reordering_time_ms = 2;
+
+  // Least squared distance from last scoring point for a new reordering to be
+  // made.
+  optional float min_reordering_squared_distance = 3;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller.cc
new file mode 100644
index 0000000000..5e2dc859bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller.cc
@@ -0,0 +1,19 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+
+namespace webrtc {
+
+Controller::NetworkMetrics::NetworkMetrics() = default;
+
+Controller::NetworkMetrics::~NetworkMetrics() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller.h
new file mode 100644
index 0000000000..b70ada01a4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+
+namespace webrtc {
+
+class Controller {
+ public:
+  struct NetworkMetrics {
+    NetworkMetrics();
+    ~NetworkMetrics();
+    absl::optional<int> uplink_bandwidth_bps;
+    absl::optional<float> uplink_packet_loss_fraction;
+    absl::optional<int> target_audio_bitrate_bps;
+    absl::optional<int> rtt_ms;
+    absl::optional<size_t> overhead_bytes_per_packet;
+  };
+
+  virtual ~Controller() = default;
+
+  // Informs network metrics update to this controller. Any non-empty field
+  // indicates an update on the corresponding network metric.
+  virtual void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) = 0;
+
+  virtual void MakeDecision(AudioEncoderRuntimeConfig* config) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
new file mode 100644
index 0000000000..42dd8a8786
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -0,0 +1,454 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+
+#include <cmath>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h"
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+namespace {
+
+#if WEBRTC_ENABLE_PROTOBUF
+
+std::unique_ptr<FecControllerPlrBased> CreateFecControllerPlrBased(
+    const audio_network_adaptor::config::FecController& config,
+    bool initial_fec_enabled) {
+  RTC_CHECK(config.has_fec_enabling_threshold());
+  RTC_CHECK(config.has_fec_disabling_threshold());
+  RTC_CHECK(config.has_time_constant_ms());
+
+  auto& fec_enabling_threshold = config.fec_enabling_threshold();
+  RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_bps());
+  RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_packet_loss());
+  RTC_CHECK(fec_enabling_threshold.has_high_bandwidth_bps());
+  RTC_CHECK(fec_enabling_threshold.has_high_bandwidth_packet_loss());
+
+  auto& fec_disabling_threshold = config.fec_disabling_threshold();
+  RTC_CHECK(fec_disabling_threshold.has_low_bandwidth_bps());
+  RTC_CHECK(fec_disabling_threshold.has_low_bandwidth_packet_loss());
+  RTC_CHECK(fec_disabling_threshold.has_high_bandwidth_bps());
+  RTC_CHECK(fec_disabling_threshold.has_high_bandwidth_packet_loss());
+
+  return std::unique_ptr<FecControllerPlrBased>(
+      new FecControllerPlrBased(FecControllerPlrBased::Config(
+          initial_fec_enabled,
+          ThresholdCurve(fec_enabling_threshold.low_bandwidth_bps(),
+                         fec_enabling_threshold.low_bandwidth_packet_loss(),
+                         fec_enabling_threshold.high_bandwidth_bps(),
+                         fec_enabling_threshold.high_bandwidth_packet_loss()),
+          ThresholdCurve(fec_disabling_threshold.low_bandwidth_bps(),
+                         fec_disabling_threshold.low_bandwidth_packet_loss(),
+                         fec_disabling_threshold.high_bandwidth_bps(),
+                         fec_disabling_threshold.high_bandwidth_packet_loss()),
+          config.time_constant_ms())));
+}
+
+std::unique_ptr<FrameLengthController> CreateFrameLengthController(
+    const audio_network_adaptor::config::FrameLengthController& config,
+    rtc::ArrayView<const int> encoder_frame_lengths_ms,
+    int initial_frame_length_ms,
+    int min_encoder_bitrate_bps) {
+  RTC_CHECK(config.has_fl_increasing_packet_loss_fraction());
+  RTC_CHECK(config.has_fl_decreasing_packet_loss_fraction());
+
+  std::map<FrameLengthController::Config::FrameLengthChange, int>
+      fl_changing_bandwidths_bps;
+
+  if (config.has_fl_20ms_to_60ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(20, 60),
+                       config.fl_20ms_to_60ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_60ms_to_20ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(60, 20),
+                       config.fl_60ms_to_20ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_20ms_to_40ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(20, 40),
+                       config.fl_20ms_to_40ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_40ms_to_20ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(40, 20),
+                       config.fl_40ms_to_20ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_40ms_to_60ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(40, 60),
+                       config.fl_40ms_to_60ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_60ms_to_40ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(
+        std::make_pair(FrameLengthController::Config::FrameLengthChange(60, 40),
+                       config.fl_60ms_to_40ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_60ms_to_120ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(std::make_pair(
+        FrameLengthController::Config::FrameLengthChange(60, 120),
+        config.fl_60ms_to_120ms_bandwidth_bps()));
+  }
+
+  if (config.has_fl_120ms_to_60ms_bandwidth_bps()) {
+    fl_changing_bandwidths_bps.insert(std::make_pair(
+        FrameLengthController::Config::FrameLengthChange(120, 60),
+        config.fl_120ms_to_60ms_bandwidth_bps()));
+  }
+
+  int fl_increase_overhead_offset = 0;
+  if (config.has_fl_increase_overhead_offset()) {
+    fl_increase_overhead_offset = config.fl_increase_overhead_offset();
+  }
+  int fl_decrease_overhead_offset = 0;
+  if (config.has_fl_decrease_overhead_offset()) {
+    fl_decrease_overhead_offset = config.fl_decrease_overhead_offset();
+  }
+
+  FrameLengthController::Config ctor_config(
+      std::set<int>(), initial_frame_length_ms, min_encoder_bitrate_bps,
+      config.fl_increasing_packet_loss_fraction(),
+      config.fl_decreasing_packet_loss_fraction(), fl_increase_overhead_offset,
+      fl_decrease_overhead_offset, std::move(fl_changing_bandwidths_bps));
+
+  for (auto frame_length : encoder_frame_lengths_ms)
+    ctor_config.encoder_frame_lengths_ms.insert(frame_length);
+
+  return std::unique_ptr<FrameLengthController>(
+      new FrameLengthController(ctor_config));
+}
+
+std::unique_ptr<ChannelController> CreateChannelController(
+    const audio_network_adaptor::config::ChannelController& config,
+    size_t num_encoder_channels,
+    size_t intial_channels_to_encode) {
+  RTC_CHECK(config.has_channel_1_to_2_bandwidth_bps());
+  RTC_CHECK(config.has_channel_2_to_1_bandwidth_bps());
+
+  return std::unique_ptr<ChannelController>(new ChannelController(
+      ChannelController::Config(num_encoder_channels, intial_channels_to_encode,
+                                config.channel_1_to_2_bandwidth_bps(),
+                                config.channel_2_to_1_bandwidth_bps())));
+}
+
+std::unique_ptr<DtxController> CreateDtxController(
+    const audio_network_adaptor::config::DtxController& dtx_config,
+    bool initial_dtx_enabled) {
+  RTC_CHECK(dtx_config.has_dtx_enabling_bandwidth_bps());
+  RTC_CHECK(dtx_config.has_dtx_disabling_bandwidth_bps());
+
+  return std::unique_ptr<DtxController>(new DtxController(DtxController::Config(
+      initial_dtx_enabled, dtx_config.dtx_enabling_bandwidth_bps(),
+      dtx_config.dtx_disabling_bandwidth_bps())));
+}
+
+using audio_network_adaptor::BitrateController;
+std::unique_ptr<BitrateController> CreateBitrateController(
+    const audio_network_adaptor::config::BitrateController& bitrate_config,
+    int initial_bitrate_bps,
+    int initial_frame_length_ms) {
+  int fl_increase_overhead_offset = 0;
+  if (bitrate_config.has_fl_increase_overhead_offset()) {
+    fl_increase_overhead_offset = bitrate_config.fl_increase_overhead_offset();
+  }
+  int fl_decrease_overhead_offset = 0;
+  if (bitrate_config.has_fl_decrease_overhead_offset()) {
+    fl_decrease_overhead_offset = bitrate_config.fl_decrease_overhead_offset();
+  }
+  return std::unique_ptr<BitrateController>(
+      new BitrateController(BitrateController::Config(
+          initial_bitrate_bps, initial_frame_length_ms,
+          fl_increase_overhead_offset, fl_decrease_overhead_offset)));
+}
+
+std::unique_ptr<FrameLengthControllerV2> CreateFrameLengthControllerV2(
+    const audio_network_adaptor::config::FrameLengthControllerV2& config,
+    rtc::ArrayView<const int> encoder_frame_lengths_ms) {
+  return std::make_unique<FrameLengthControllerV2>(
+      encoder_frame_lengths_ms, config.min_payload_bitrate_bps(),
+      config.use_slow_adaptation());
+}
+#endif  // WEBRTC_ENABLE_PROTOBUF
+
+}  // namespace
+
+ControllerManagerImpl::Config::Config(int min_reordering_time_ms,
+                                      float min_reordering_squared_distance)
+    : min_reordering_time_ms(min_reordering_time_ms),
+      min_reordering_squared_distance(min_reordering_squared_distance) {}
+
+ControllerManagerImpl::Config::~Config() = default;
+
+std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
+    absl::string_view config_string,
+    size_t num_encoder_channels,
+    rtc::ArrayView<const int> encoder_frame_lengths_ms,
+    int min_encoder_bitrate_bps,
+    size_t intial_channels_to_encode,
+    int initial_frame_length_ms,
+    int initial_bitrate_bps,
+    bool initial_fec_enabled,
+    bool initial_dtx_enabled) {
+  return Create(config_string, num_encoder_channels, encoder_frame_lengths_ms,
+                min_encoder_bitrate_bps, intial_channels_to_encode,
+                initial_frame_length_ms, initial_bitrate_bps,
+                initial_fec_enabled, initial_dtx_enabled, nullptr);
+}
+
+std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
+    absl::string_view config_string,
+    size_t num_encoder_channels,
+    rtc::ArrayView<const int> encoder_frame_lengths_ms,
+    int min_encoder_bitrate_bps,
+    size_t intial_channels_to_encode,
+    int initial_frame_length_ms,
+    int initial_bitrate_bps,
+    bool initial_fec_enabled,
+    bool initial_dtx_enabled,
+    DebugDumpWriter* debug_dump_writer) {
+#if WEBRTC_ENABLE_PROTOBUF
+  audio_network_adaptor::config::ControllerManager controller_manager_config;
+  RTC_CHECK(
+      controller_manager_config.ParseFromString(std::string(config_string)));
+  if (debug_dump_writer)
+    debug_dump_writer->DumpControllerManagerConfig(controller_manager_config,
+                                                   rtc::TimeMillis());
+
+  std::vector<std::unique_ptr<Controller>> controllers;
+  std::map<const Controller*, std::pair<int, float>> scoring_points;
+
+  for (int i = 0; i < controller_manager_config.controllers_size(); ++i) {
+    auto& controller_config = controller_manager_config.controllers(i);
+    std::unique_ptr<Controller> controller;
+    switch (controller_config.controller_case()) {
+      case audio_network_adaptor::config::Controller::kFecController:
+        controller = CreateFecControllerPlrBased(
+            controller_config.fec_controller(), initial_fec_enabled);
+        break;
+      case audio_network_adaptor::config::Controller::kFecControllerRplrBased:
+        // FecControllerRplrBased has been removed and can't be used anymore.
+        RTC_DCHECK_NOTREACHED();
+        continue;
+      case audio_network_adaptor::config::Controller::kFrameLengthController:
+        controller = CreateFrameLengthController(
+            controller_config.frame_length_controller(),
+            encoder_frame_lengths_ms, initial_frame_length_ms,
+            min_encoder_bitrate_bps);
+        break;
+      case audio_network_adaptor::config::Controller::kChannelController:
+        controller = CreateChannelController(
+            controller_config.channel_controller(), num_encoder_channels,
+            intial_channels_to_encode);
+        break;
+      case audio_network_adaptor::config::Controller::kDtxController:
+        controller = CreateDtxController(controller_config.dtx_controller(),
+                                         initial_dtx_enabled);
+        break;
+      case audio_network_adaptor::config::Controller::kBitrateController:
+        controller = CreateBitrateController(
+            controller_config.bitrate_controller(), initial_bitrate_bps,
+            initial_frame_length_ms);
+        break;
+      case audio_network_adaptor::config::Controller::kFrameLengthControllerV2:
+        controller = CreateFrameLengthControllerV2(
+            controller_config.frame_length_controller_v2(),
+            encoder_frame_lengths_ms);
+        break;
+      default:
+        RTC_DCHECK_NOTREACHED();
+    }
+    if (controller_config.has_scoring_point()) {
+      auto& scoring_point = controller_config.scoring_point();
+      RTC_CHECK(scoring_point.has_uplink_bandwidth_bps());
+      RTC_CHECK(scoring_point.has_uplink_packet_loss_fraction());
+      scoring_points[controller.get()] = std::make_pair<int, float>(
+          scoring_point.uplink_bandwidth_bps(),
+          scoring_point.uplink_packet_loss_fraction());
+    }
+    controllers.push_back(std::move(controller));
+  }
+
+  if (scoring_points.size() == 0) {
+    return std::unique_ptr<ControllerManagerImpl>(
+        new ControllerManagerImpl(ControllerManagerImpl::Config(0, 0),
+                                  std::move(controllers), scoring_points));
+  } else {
+    RTC_CHECK(controller_manager_config.has_min_reordering_time_ms());
+    RTC_CHECK(controller_manager_config.has_min_reordering_squared_distance());
+    return std::unique_ptr<ControllerManagerImpl>(new ControllerManagerImpl(
+        ControllerManagerImpl::Config(
+            controller_manager_config.min_reordering_time_ms(),
+            controller_manager_config.min_reordering_squared_distance()),
+        std::move(controllers), scoring_points));
+  }
+
+#else
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+#endif  // WEBRTC_ENABLE_PROTOBUF
+}
+
+ControllerManagerImpl::ControllerManagerImpl(const Config& config)
+    : ControllerManagerImpl(
+          config,
+          std::vector<std::unique_ptr<Controller>>(),
+          std::map<const Controller*, std::pair<int, float>>()) {}
+
+ControllerManagerImpl::ControllerManagerImpl(
+    const Config& config,
+    std::vector<std::unique_ptr<Controller>> controllers,
+    const std::map<const Controller*, std::pair<int, float>>& scoring_points)
+    : config_(config),
+      controllers_(std::move(controllers)),
+      last_reordering_time_ms_(absl::nullopt),
+      last_scoring_point_(0, 0.0) {
+  for (auto& controller : controllers_)
+    default_sorted_controllers_.push_back(controller.get());
+  sorted_controllers_ = default_sorted_controllers_;
+  for (auto& controller_point : scoring_points) {
+    controller_scoring_points_.insert(std::make_pair(
+        controller_point.first, ScoringPoint(controller_point.second.first,
+                                             controller_point.second.second)));
+  }
+}
+
+ControllerManagerImpl::~ControllerManagerImpl() = default;
+
+std::vector<Controller*> ControllerManagerImpl::GetSortedControllers(
+    const Controller::NetworkMetrics& metrics) {
+  if (controller_scoring_points_.size() == 0)
+    return default_sorted_controllers_;
+
+  if (!metrics.uplink_bandwidth_bps || !metrics.uplink_packet_loss_fraction)
+    return sorted_controllers_;
+
+  const int64_t now_ms = rtc::TimeMillis();
+  if (last_reordering_time_ms_ &&
+      now_ms - *last_reordering_time_ms_ < config_.min_reordering_time_ms)
+    return sorted_controllers_;
+
+  ScoringPoint scoring_point(*metrics.uplink_bandwidth_bps,
+                             *metrics.uplink_packet_loss_fraction);
+
+  if (last_reordering_time_ms_ &&
+      last_scoring_point_.SquaredDistanceTo(scoring_point) <
+          config_.min_reordering_squared_distance)
+    return sorted_controllers_;
+
+  // Sort controllers according to the distances of `scoring_point` to the
+  // scoring points of controllers.
+  //
+  // A controller that does not associate with any scoring point
+  // are treated as if
+  // 1) they are less important than any controller that has a scoring point,
+  // 2) they are equally important to any controller that has no scoring point,
+  //    and their relative order will follow `default_sorted_controllers_`.
+  std::vector<Controller*> sorted_controllers(default_sorted_controllers_);
+  std::stable_sort(
+      sorted_controllers.begin(), sorted_controllers.end(),
+      [this, &scoring_point](const Controller* lhs, const Controller* rhs) {
+        auto lhs_scoring_point = controller_scoring_points_.find(lhs);
+        auto rhs_scoring_point = controller_scoring_points_.find(rhs);
+
+        if (lhs_scoring_point == controller_scoring_points_.end())
+          return false;
+
+        if (rhs_scoring_point == controller_scoring_points_.end())
+          return true;
+
+        return lhs_scoring_point->second.SquaredDistanceTo(scoring_point) <
+               rhs_scoring_point->second.SquaredDistanceTo(scoring_point);
+      });
+
+  if (sorted_controllers_ != sorted_controllers) {
+    sorted_controllers_ = sorted_controllers;
+    last_reordering_time_ms_ = now_ms;
+    last_scoring_point_ = scoring_point;
+  }
+  return sorted_controllers_;
+}
+
+std::vector<Controller*> ControllerManagerImpl::GetControllers() const {
+  return default_sorted_controllers_;
+}
+
+ControllerManagerImpl::ScoringPoint::ScoringPoint(
+    int uplink_bandwidth_bps,
+    float uplink_packet_loss_fraction)
+    : uplink_bandwidth_bps(uplink_bandwidth_bps),
+      uplink_packet_loss_fraction(uplink_packet_loss_fraction) {}
+
+namespace {
+
+constexpr int kMinUplinkBandwidthBps = 0;
+constexpr int kMaxUplinkBandwidthBps = 120000;
+
+float NormalizeUplinkBandwidth(int uplink_bandwidth_bps) {
+  uplink_bandwidth_bps =
+      std::min(kMaxUplinkBandwidthBps,
+               std::max(kMinUplinkBandwidthBps, uplink_bandwidth_bps));
+  return static_cast<float>(uplink_bandwidth_bps - kMinUplinkBandwidthBps) /
+         (kMaxUplinkBandwidthBps - kMinUplinkBandwidthBps);
+}
+
+float NormalizePacketLossFraction(float uplink_packet_loss_fraction) {
+  // `uplink_packet_loss_fraction` is seldom larger than 0.3, so we scale it up
+  // by 3.3333f.
+  return std::min(uplink_packet_loss_fraction * 3.3333f, 1.0f);
+}
+
+}  // namespace
+
+float ControllerManagerImpl::ScoringPoint::SquaredDistanceTo(
+    const ScoringPoint& scoring_point) const {
+  float diff_normalized_bitrate_bps =
+      NormalizeUplinkBandwidth(scoring_point.uplink_bandwidth_bps) -
+      NormalizeUplinkBandwidth(uplink_bandwidth_bps);
+  float diff_normalized_packet_loss =
+      NormalizePacketLossFraction(scoring_point.uplink_packet_loss_fraction) -
+      NormalizePacketLossFraction(uplink_packet_loss_fraction);
+  return std::pow(diff_normalized_bitrate_bps, 2) +
+         std::pow(diff_normalized_packet_loss, 2);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
new file mode 100644
index 0000000000..47e8e0f5a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+
+namespace webrtc {
+
+class DebugDumpWriter;
+
+class ControllerManager {
+ public:
+  virtual ~ControllerManager() = default;
+
+  // Sort controllers based on their significance.
+  virtual std::vector<Controller*> GetSortedControllers(
+      const Controller::NetworkMetrics& metrics) = 0;
+
+  virtual std::vector<Controller*> GetControllers() const = 0;
+};
+
+class ControllerManagerImpl final : public ControllerManager {
+ public:
+  struct Config {
+    Config(int min_reordering_time_ms, float min_reordering_squared_distance);
+    ~Config();
+    // Least time since last reordering for a new reordering to be made.
+    int min_reordering_time_ms;
+    // Least squared distance from last scoring point for a new reordering to be
+    // made.
+    float min_reordering_squared_distance;
+  };
+
+  static std::unique_ptr<ControllerManager> Create(
+      absl::string_view config_string,
+      size_t num_encoder_channels,
+      rtc::ArrayView<const int> encoder_frame_lengths_ms,
+      int min_encoder_bitrate_bps,
+      size_t intial_channels_to_encode,
+      int initial_frame_length_ms,
+      int initial_bitrate_bps,
+      bool initial_fec_enabled,
+      bool initial_dtx_enabled);
+
+  static std::unique_ptr<ControllerManager> Create(
+      absl::string_view config_string,
+      size_t num_encoder_channels,
+      rtc::ArrayView<const int> encoder_frame_lengths_ms,
+      int min_encoder_bitrate_bps,
+      size_t intial_channels_to_encode,
+      int initial_frame_length_ms,
+      int initial_bitrate_bps,
+      bool initial_fec_enabled,
+      bool initial_dtx_enabled,
+      DebugDumpWriter* debug_dump_writer);
+
+  explicit ControllerManagerImpl(const Config& config);
+
+  // Dependency injection for testing.
+  ControllerManagerImpl(
+      const Config& config,
+      std::vector<std::unique_ptr<Controller>> controllers,
+      const std::map<const Controller*, std::pair<int, float>>&
+          chracteristic_points);
+
+  ~ControllerManagerImpl() override;
+
+  ControllerManagerImpl(const ControllerManagerImpl&) = delete;
+  ControllerManagerImpl& operator=(const ControllerManagerImpl&) = delete;
+
+  // Sort controllers based on their significance.
+  std::vector<Controller*> GetSortedControllers(
+      const Controller::NetworkMetrics& metrics) override;
+
+  std::vector<Controller*> GetControllers() const override;
+
+ private:
+  // Scoring point is a subset of NetworkMetrics that is used for comparing the
+  // significance of controllers.
+  struct ScoringPoint {
+    // TODO(eladalon): Do we want to experiment with RPLR-based scoring?
+    ScoringPoint(int uplink_bandwidth_bps, float uplink_packet_loss_fraction);
+
+    // Calculate the normalized [0,1] distance between two scoring points.
+    float SquaredDistanceTo(const ScoringPoint& scoring_point) const;
+
+    int uplink_bandwidth_bps;
+    float uplink_packet_loss_fraction;
+  };
+
+  const Config config_;
+
+  std::vector<std::unique_ptr<Controller>> controllers_;
+
+  absl::optional<int64_t> last_reordering_time_ms_;
+  ScoringPoint last_scoring_point_;
+
+  std::vector<Controller*> default_sorted_controllers_;
+
+  std::vector<Controller*> sorted_controllers_;
+
+  // `scoring_points_` saves the scoring points of various
+  // controllers.
+  std::map<const Controller*, ScoringPoint> controller_scoring_points_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
new file mode 100644
index 0000000000..3e6ecf6def
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -0,0 +1,486 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/ignore_wundef.h"
+#include "test/gtest.h"
+
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::NiceMock;
+
+namespace {
+
+constexpr size_t kNumControllers = 4;
+constexpr int kChracteristicBandwithBps[2] = {15000, 0};
+constexpr float kChracteristicPacketLossFraction[2] = {0.2f, 0.0f};
+constexpr int kMinReorderingTimeMs = 200;
+constexpr int kFactor = 100;
+constexpr float kMinReorderingSquareDistance = 1.0f / kFactor / kFactor;
+
+// `kMinUplinkBandwidthBps` and `kMaxUplinkBandwidthBps` are copied from
+// controller_manager.cc
+constexpr int kMinUplinkBandwidthBps = 0;
+constexpr int kMaxUplinkBandwidthBps = 120000;
+constexpr int kMinBandwithChangeBps =
+    (kMaxUplinkBandwidthBps - kMinUplinkBandwidthBps) / kFactor;
+
+struct ControllerManagerStates {
+  std::unique_ptr<ControllerManager> controller_manager;
+  std::vector<MockController*> mock_controllers;
+};
+
+ControllerManagerStates CreateControllerManager() {
+  ControllerManagerStates states;
+  std::vector<std::unique_ptr<Controller>> controllers;
+  std::map<const Controller*, std::pair<int, float>> chracteristic_points;
+  for (size_t i = 0; i < kNumControllers; ++i) {
+    auto controller =
+        std::unique_ptr<MockController>(new NiceMock<MockController>());
+    EXPECT_CALL(*controller, Die());
+    states.mock_controllers.push_back(controller.get());
+    controllers.push_back(std::move(controller));
+  }
+
+  // Assign characteristic points to the last two controllers.
+  chracteristic_points[states.mock_controllers[kNumControllers - 2]] =
+      std::make_pair(kChracteristicBandwithBps[0],
+                     kChracteristicPacketLossFraction[0]);
+  chracteristic_points[states.mock_controllers[kNumControllers - 1]] =
+      std::make_pair(kChracteristicBandwithBps[1],
+                     kChracteristicPacketLossFraction[1]);
+
+  states.controller_manager.reset(new ControllerManagerImpl(
+      ControllerManagerImpl::Config(kMinReorderingTimeMs,
+                                    kMinReorderingSquareDistance),
+      std::move(controllers), chracteristic_points));
+  return states;
+}
+
+// `expected_order` contains the expected indices of all controllers in the
+// vector of controllers returned by GetSortedControllers(). A negative index
+// means that we do not care about its exact place, but we do check that it
+// exists in the vector.
+void CheckControllersOrder(
+    ControllerManagerStates* states,
+    const absl::optional<int>& uplink_bandwidth_bps,
+    const absl::optional<float>& uplink_packet_loss_fraction,
+    const std::vector<int>& expected_order) {
+  RTC_DCHECK_EQ(kNumControllers, expected_order.size());
+  Controller::NetworkMetrics metrics;
+  metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+  metrics.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
+  auto check = states->controller_manager->GetSortedControllers(metrics);
+  EXPECT_EQ(states->mock_controllers.size(), check.size());
+  for (size_t i = 0; i < states->mock_controllers.size(); ++i) {
+    if (expected_order[i] >= 0) {
+      EXPECT_EQ(states->mock_controllers[i], check[expected_order[i]]);
+    } else {
+      EXPECT_NE(check.end(), std::find(check.begin(), check.end(),
+                                       states->mock_controllers[i]));
+    }
+  }
+}
+
+}  // namespace
+
+TEST(ControllerManagerTest, GetControllersReturnAllControllers) {
+  auto states = CreateControllerManager();
+  auto check = states.controller_manager->GetControllers();
+  // Verify that controllers in `check` are one-to-one mapped to those in
+  // `mock_controllers_`.
+  EXPECT_EQ(states.mock_controllers.size(), check.size());
+  for (auto& controller : check)
+    EXPECT_NE(states.mock_controllers.end(),
+              std::find(states.mock_controllers.begin(),
+                        states.mock_controllers.end(), controller));
+}
+
+TEST(ControllerManagerTest, ControllersInDefaultOrderOnEmptyNetworkMetrics) {
+  auto states = CreateControllerManager();
+  // `network_metrics` are empty, and the controllers are supposed to follow the
+  // default order.
+  CheckControllersOrder(&states, absl::nullopt, absl::nullopt, {0, 1, 2, 3});
+}
+
+TEST(ControllerManagerTest, ControllersWithoutCharPointAtEndAndInDefaultOrder) {
+  auto states = CreateControllerManager();
+  CheckControllersOrder(&states, 0, 0.0,
+                        {kNumControllers - 2, kNumControllers - 1, -1, -1});
+}
+
+TEST(ControllerManagerTest, ControllersWithCharPointDependOnNetworkMetrics) {
+  auto states = CreateControllerManager();
+  CheckControllersOrder(&states, kChracteristicBandwithBps[1],
+                        kChracteristicPacketLossFraction[1],
+                        {kNumControllers - 2, kNumControllers - 1, 1, 0});
+}
+
+TEST(ControllerManagerTest, DoNotReorderBeforeMinReordingTime) {
+  rtc::ScopedFakeClock fake_clock;
+  auto states = CreateControllerManager();
+  CheckControllersOrder(&states, kChracteristicBandwithBps[0],
+                        kChracteristicPacketLossFraction[0],
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs - 1));
+  // Move uplink bandwidth and packet loss fraction to the other controller's
+  // characteristic point, which would cause controller manager to reorder the
+  // controllers if time had reached min reordering time.
+  CheckControllersOrder(&states, kChracteristicBandwithBps[1],
+                        kChracteristicPacketLossFraction[1],
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+}
+
+TEST(ControllerManagerTest, ReorderBeyondMinReordingTimeAndMinDistance) {
+  rtc::ScopedFakeClock fake_clock;
+  auto states = CreateControllerManager();
+  constexpr int kBandwidthBps =
+      (kChracteristicBandwithBps[0] + kChracteristicBandwithBps[1]) / 2;
+  constexpr float kPacketLossFraction = (kChracteristicPacketLossFraction[0] +
+                                         kChracteristicPacketLossFraction[1]) /
+                                        2.0f;
+  // Set network metrics to be in the middle between the characteristic points
+  // of two controllers.
+  CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs));
+  // Then let network metrics move a little towards the other controller.
+  CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps - 1,
+                        kPacketLossFraction,
+                        {kNumControllers - 2, kNumControllers - 1, 1, 0});
+}
+
+TEST(ControllerManagerTest, DoNotReorderIfNetworkMetricsChangeTooSmall) {
+  rtc::ScopedFakeClock fake_clock;
+  auto states = CreateControllerManager();
+  constexpr int kBandwidthBps =
+      (kChracteristicBandwithBps[0] + kChracteristicBandwithBps[1]) / 2;
+  constexpr float kPacketLossFraction = (kChracteristicPacketLossFraction[0] +
+                                         kChracteristicPacketLossFraction[1]) /
+                                        2.0f;
+  // Set network metrics to be in the middle between the characteristic points
+  // of two controllers.
+  CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs));
+  // Then let network metrics move a little towards the other controller.
+  CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps + 1,
+                        kPacketLossFraction,
+                        {kNumControllers - 2, kNumControllers - 1, 0, 1});
+}
+
+#if WEBRTC_ENABLE_PROTOBUF
+
+namespace {
+
+void AddBitrateControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  config->add_controllers()->mutable_bitrate_controller();
+}
+
+void AddChannelControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  auto controller_config =
+      config->add_controllers()->mutable_channel_controller();
+  controller_config->set_channel_1_to_2_bandwidth_bps(31000);
+  controller_config->set_channel_2_to_1_bandwidth_bps(29000);
+}
+
+void AddDtxControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  auto controller_config = config->add_controllers()->mutable_dtx_controller();
+  controller_config->set_dtx_enabling_bandwidth_bps(55000);
+  controller_config->set_dtx_disabling_bandwidth_bps(65000);
+}
+
+void AddFecControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  auto controller_config_ext = config->add_controllers();
+  auto controller_config = controller_config_ext->mutable_fec_controller();
+  auto fec_enabling_threshold =
+      controller_config->mutable_fec_enabling_threshold();
+  fec_enabling_threshold->set_low_bandwidth_bps(17000);
+  fec_enabling_threshold->set_low_bandwidth_packet_loss(0.1f);
+  fec_enabling_threshold->set_high_bandwidth_bps(64000);
+  fec_enabling_threshold->set_high_bandwidth_packet_loss(0.05f);
+  auto fec_disabling_threshold =
+      controller_config->mutable_fec_disabling_threshold();
+  fec_disabling_threshold->set_low_bandwidth_bps(15000);
+  fec_disabling_threshold->set_low_bandwidth_packet_loss(0.08f);
+  fec_disabling_threshold->set_high_bandwidth_bps(64000);
+  fec_disabling_threshold->set_high_bandwidth_packet_loss(0.01f);
+  controller_config->set_time_constant_ms(500);
+
+  auto scoring_point = controller_config_ext->mutable_scoring_point();
+  scoring_point->set_uplink_bandwidth_bps(kChracteristicBandwithBps[0]);
+  scoring_point->set_uplink_packet_loss_fraction(
+      kChracteristicPacketLossFraction[0]);
+}
+
+void AddFrameLengthControllerConfig(
+    audio_network_adaptor::config::ControllerManager* config) {
+  auto controller_config_ext = config->add_controllers();
+  auto controller_config =
+      controller_config_ext->mutable_frame_length_controller();
+  controller_config->set_fl_decreasing_packet_loss_fraction(0.05f);
+  controller_config->set_fl_increasing_packet_loss_fraction(0.04f);
+  controller_config->set_fl_20ms_to_40ms_bandwidth_bps(80000);
+  controller_config->set_fl_40ms_to_20ms_bandwidth_bps(88000);
+  controller_config->set_fl_40ms_to_60ms_bandwidth_bps(72000);
+  controller_config->set_fl_60ms_to_40ms_bandwidth_bps(80000);
+
+  auto scoring_point = controller_config_ext->mutable_scoring_point();
+  scoring_point->set_uplink_bandwidth_bps(kChracteristicBandwithBps[1]);
+  scoring_point->set_uplink_packet_loss_fraction(
+      kChracteristicPacketLossFraction[1]);
+}
+
+void AddFrameLengthControllerV2Config(
+    audio_network_adaptor::config::ControllerManager* config) {
+  auto controller =
+      config->add_controllers()->mutable_frame_length_controller_v2();
+  controller->set_min_payload_bitrate_bps(16000);
+  controller->set_use_slow_adaptation(true);
+}
+
+constexpr int kInitialBitrateBps = 24000;
+constexpr size_t kIntialChannelsToEncode = 1;
+constexpr bool kInitialDtxEnabled = true;
+constexpr bool kInitialFecEnabled = true;
+constexpr int kInitialFrameLengthMs = 60;
+constexpr int kMinBitrateBps = 6000;
+
+ControllerManagerStates CreateControllerManager(
+    absl::string_view config_string) {
+  ControllerManagerStates states;
+  constexpr size_t kNumEncoderChannels = 2;
+  const std::vector<int> encoder_frame_lengths_ms = {20, 60};
+  states.controller_manager = ControllerManagerImpl::Create(
+      config_string, kNumEncoderChannels, encoder_frame_lengths_ms,
+      kMinBitrateBps, kIntialChannelsToEncode, kInitialFrameLengthMs,
+      kInitialBitrateBps, kInitialFecEnabled, kInitialDtxEnabled);
+  return states;
+}
+
+enum class ControllerType : int8_t {
+  FEC,
+  CHANNEL,
+  DTX,
+  FRAME_LENGTH,
+  BIT_RATE
+};
+
+void CheckControllersOrder(const std::vector<Controller*>& controllers,
+                           const std::vector<ControllerType>& expected_types) {
+  ASSERT_EQ(expected_types.size(), controllers.size());
+
+  // We also check that the controllers follow the initial settings.
+  AudioEncoderRuntimeConfig encoder_config;
+
+  for (size_t i = 0; i < controllers.size(); ++i) {
+    AudioEncoderRuntimeConfig encoder_config;
+    // We check the order of `controllers` by judging their decisions.
+    controllers[i]->MakeDecision(&encoder_config);
+
+    // Since controllers are not provided with network metrics, they give the
+    // initial values.
+    switch (expected_types[i]) {
+      case ControllerType::FEC:
+        EXPECT_EQ(kInitialFecEnabled, encoder_config.enable_fec);
+        break;
+      case ControllerType::CHANNEL:
+        EXPECT_EQ(kIntialChannelsToEncode, encoder_config.num_channels);
+        break;
+      case ControllerType::DTX:
+        EXPECT_EQ(kInitialDtxEnabled, encoder_config.enable_dtx);
+        break;
+      case ControllerType::FRAME_LENGTH:
+        EXPECT_EQ(kInitialFrameLengthMs, encoder_config.frame_length_ms);
+        break;
+      case ControllerType::BIT_RATE:
+        EXPECT_EQ(kInitialBitrateBps, encoder_config.bitrate_bps);
+    }
+  }
+}
+
+MATCHER_P(ControllerManagerEqual, value, "") {
+  std::string value_string;
+  std::string arg_string;
+  EXPECT_TRUE(arg.SerializeToString(&arg_string));
+  EXPECT_TRUE(value.SerializeToString(&value_string));
+  return arg_string == value_string;
+}
+
+}  // namespace
+
+TEST(ControllerManagerTest, DebugDumpLoggedWhenCreateFromConfigString) {
+  audio_network_adaptor::config::ControllerManager config;
+  config.set_min_reordering_time_ms(kMinReorderingTimeMs);
+  config.set_min_reordering_squared_distance(kMinReorderingSquareDistance);
+
+  AddFecControllerConfig(&config);
+  AddChannelControllerConfig(&config);
+  AddDtxControllerConfig(&config);
+  AddFrameLengthControllerConfig(&config);
+  AddBitrateControllerConfig(&config);
+
+  std::string config_string;
+  config.SerializeToString(&config_string);
+
+  constexpr size_t kNumEncoderChannels = 2;
+  const std::vector<int> encoder_frame_lengths_ms = {20, 60};
+
+  constexpr int64_t kClockInitialTimeMs = 12345678;
+  rtc::ScopedFakeClock fake_clock;
+  fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
+  auto debug_dump_writer =
+      std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
+  EXPECT_CALL(*debug_dump_writer, Die());
+  EXPECT_CALL(*debug_dump_writer,
+              DumpControllerManagerConfig(ControllerManagerEqual(config),
+                                          kClockInitialTimeMs));
+
+  ControllerManagerImpl::Create(config_string, kNumEncoderChannels,
+                                encoder_frame_lengths_ms, kMinBitrateBps,
+                                kIntialChannelsToEncode, kInitialFrameLengthMs,
+                                kInitialBitrateBps, kInitialFecEnabled,
+                                kInitialDtxEnabled, debug_dump_writer.get());
+}
+
+TEST(ControllerManagerTest, CreateFromConfigStringAndCheckDefaultOrder) {
+  audio_network_adaptor::config::ControllerManager config;
+  config.set_min_reordering_time_ms(kMinReorderingTimeMs);
+  config.set_min_reordering_squared_distance(kMinReorderingSquareDistance);
+
+  AddFecControllerConfig(&config);
+  AddChannelControllerConfig(&config);
+  AddDtxControllerConfig(&config);
+  AddFrameLengthControllerConfig(&config);
+  AddBitrateControllerConfig(&config);
+
+  std::string config_string;
+  config.SerializeToString(&config_string);
+
+  auto states = CreateControllerManager(config_string);
+  Controller::NetworkMetrics metrics;
+
+  auto controllers = states.controller_manager->GetSortedControllers(metrics);
+  CheckControllersOrder(
+      controllers,
+      std::vector<ControllerType>{
+          ControllerType::FEC, ControllerType::CHANNEL, ControllerType::DTX,
+          ControllerType::FRAME_LENGTH, ControllerType::BIT_RATE});
+}
+
+TEST(ControllerManagerTest, CreateCharPointFreeConfigAndCheckDefaultOrder) {
+  audio_network_adaptor::config::ControllerManager config;
+
+  // Following controllers have no characteristic points.
+  AddChannelControllerConfig(&config);
+  AddDtxControllerConfig(&config);
+  AddBitrateControllerConfig(&config);
+
+  std::string config_string;
+  config.SerializeToString(&config_string);
+
+  auto states = CreateControllerManager(config_string);
+  Controller::NetworkMetrics metrics;
+
+  auto controllers = states.controller_manager->GetSortedControllers(metrics);
+  CheckControllersOrder(
+      controllers,
+      std::vector<ControllerType>{ControllerType::CHANNEL, ControllerType::DTX,
+                                  ControllerType::BIT_RATE});
+}
+
+TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
+  rtc::ScopedFakeClock fake_clock;
+  audio_network_adaptor::config::ControllerManager config;
+  config.set_min_reordering_time_ms(kMinReorderingTimeMs);
+  config.set_min_reordering_squared_distance(kMinReorderingSquareDistance);
+
+  AddChannelControllerConfig(&config);
+
+  // Internally associated with characteristic point 0.
+  AddFecControllerConfig(&config);
+
+  AddDtxControllerConfig(&config);
+
+  // Internally associated with characteristic point 1.
+  AddFrameLengthControllerConfig(&config);
+
+  AddBitrateControllerConfig(&config);
+
+  std::string config_string;
+  config.SerializeToString(&config_string);
+
+  auto states = CreateControllerManager(config_string);
+
+  Controller::NetworkMetrics metrics;
+  metrics.uplink_bandwidth_bps = kChracteristicBandwithBps[0];
+  metrics.uplink_packet_loss_fraction = kChracteristicPacketLossFraction[0];
+
+  auto controllers = states.controller_manager->GetSortedControllers(metrics);
+  CheckControllersOrder(controllers,
+                        std::vector<ControllerType>{
+                            ControllerType::FEC, ControllerType::FRAME_LENGTH,
+                            ControllerType::CHANNEL, ControllerType::DTX,
+                            ControllerType::BIT_RATE});
+
+  metrics.uplink_bandwidth_bps = kChracteristicBandwithBps[1];
+  metrics.uplink_packet_loss_fraction = kChracteristicPacketLossFraction[1];
+  fake_clock.AdvanceTime(TimeDelta::Millis(kMinReorderingTimeMs - 1));
+  controllers = states.controller_manager->GetSortedControllers(metrics);
+  // Should not reorder since min reordering time is not met.
+  CheckControllersOrder(controllers,
+                        std::vector<ControllerType>{
+                            ControllerType::FEC, ControllerType::FRAME_LENGTH,
+                            ControllerType::CHANNEL, ControllerType::DTX,
+                            ControllerType::BIT_RATE});
+
+  fake_clock.AdvanceTime(TimeDelta::Millis(1));
+  controllers = states.controller_manager->GetSortedControllers(metrics);
+  // Reorder now.
+  CheckControllersOrder(controllers,
+                        std::vector<ControllerType>{
+                            ControllerType::FRAME_LENGTH, ControllerType::FEC,
+                            ControllerType::CHANNEL, ControllerType::DTX,
+                            ControllerType::BIT_RATE});
+}
+
+TEST(ControllerManagerTest, CreateFrameLengthControllerV2) {
+  audio_network_adaptor::config::ControllerManager config;
+  AddFrameLengthControllerV2Config(&config);
+  auto states = CreateControllerManager(config.SerializeAsString());
+  auto controllers = states.controller_manager->GetControllers();
+  EXPECT_TRUE(controllers.size() == 1);
+}
+#endif  // WEBRTC_ENABLE_PROTOBUF
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
new file mode 100644
index 0000000000..3aa6a504f3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
@@ -0,0 +1,42 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.audio_network_adaptor.debug_dump;
+
+import "config.proto";
+
+message NetworkMetrics {
+  optional int32 uplink_bandwidth_bps = 1;
+  optional float uplink_packet_loss_fraction = 2;
+  optional int32 target_audio_bitrate_bps = 3;
+  optional int32 rtt_ms = 4;
+  optional int32 uplink_recoverable_packet_loss_fraction = 5;
+}
+
+message EncoderRuntimeConfig {
+  optional int32 bitrate_bps = 1;
+  optional int32 frame_length_ms = 2;
+  // Note: This is what we tell the encoder. It doesn't have to reflect
+  // the actual NetworkMetrics; it's subject to our decision.
+  optional float uplink_packet_loss_fraction = 3;
+  optional bool enable_fec = 4;
+  optional bool enable_dtx = 5;
+  // Some encoders can encode fewer channels than the actual input to make
+  // better use of the bandwidth. `num_channels` sets the number of channels
+  // to encode.
+  optional uint32 num_channels = 6;
+}
+
+message Event {
+  enum Type {
+    NETWORK_METRICS = 0;
+    ENCODER_RUNTIME_CONFIG = 1;
+    CONTROLLER_MANAGER_CONFIG = 2;
+  }
+  required Type type = 1;
+  required uint32 timestamp = 2;
+  optional NetworkMetrics network_metrics = 3;
+  optional EncoderRuntimeConfig encoder_runtime_config = 4;
+  optional webrtc.audio_network_adaptor.config.ControllerManager
+      controller_manager_config = 5;
+}
+
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
new file mode 100644
index 0000000000..2616706ee5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/system/file_wrapper.h"
+
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+#if WEBRTC_ENABLE_PROTOBUF
+namespace {
+
+using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
+using audio_network_adaptor::debug_dump::Event;
+using audio_network_adaptor::debug_dump::NetworkMetrics;
+
+void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
+  RTC_CHECK(dump_file->is_open());
+  std::string dump_data;
+  event.SerializeToString(&dump_data);
+  int32_t size = rtc::checked_cast<int32_t>(event.ByteSizeLong());
+  dump_file->Write(&size, sizeof(size));
+  dump_file->Write(dump_data.data(), dump_data.length());
+}
+
+}  // namespace
+#endif  // WEBRTC_ENABLE_PROTOBUF
+
+class DebugDumpWriterImpl final : public DebugDumpWriter {
+ public:
+  explicit DebugDumpWriterImpl(FILE* file_handle);
+  ~DebugDumpWriterImpl() override = default;
+
+  void DumpEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& config,
+                                int64_t timestamp) override;
+
+  void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
+                          int64_t timestamp) override;
+
+#if WEBRTC_ENABLE_PROTOBUF
+  void DumpControllerManagerConfig(
+      const audio_network_adaptor::config::ControllerManager&
+          controller_manager_config,
+      int64_t timestamp) override;
+#endif
+
+ private:
+  FileWrapper dump_file_;
+};
+
+DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle) {
+#if WEBRTC_ENABLE_PROTOBUF
+  dump_file_ = FileWrapper(file_handle);
+  RTC_CHECK(dump_file_.is_open());
+#else
+  RTC_DCHECK_NOTREACHED();
+#endif
+}
+
+void DebugDumpWriterImpl::DumpNetworkMetrics(
+    const Controller::NetworkMetrics& metrics,
+    int64_t timestamp) {
+#if WEBRTC_ENABLE_PROTOBUF
+  Event event;
+  event.set_timestamp(timestamp);
+  event.set_type(Event::NETWORK_METRICS);
+  auto dump_metrics = event.mutable_network_metrics();
+
+  if (metrics.uplink_bandwidth_bps)
+    dump_metrics->set_uplink_bandwidth_bps(*metrics.uplink_bandwidth_bps);
+
+  if (metrics.uplink_packet_loss_fraction) {
+    dump_metrics->set_uplink_packet_loss_fraction(
+        *metrics.uplink_packet_loss_fraction);
+  }
+
+  if (metrics.target_audio_bitrate_bps) {
+    dump_metrics->set_target_audio_bitrate_bps(
+        *metrics.target_audio_bitrate_bps);
+  }
+
+  if (metrics.rtt_ms)
+    dump_metrics->set_rtt_ms(*metrics.rtt_ms);
+
+  DumpEventToFile(event, &dump_file_);
+#endif  // WEBRTC_ENABLE_PROTOBUF
+}
+
+void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
+    const AudioEncoderRuntimeConfig& config,
+    int64_t timestamp) {
+#if WEBRTC_ENABLE_PROTOBUF
+  Event event;
+  event.set_timestamp(timestamp);
+  event.set_type(Event::ENCODER_RUNTIME_CONFIG);
+  auto dump_config = event.mutable_encoder_runtime_config();
+
+  if (config.bitrate_bps)
+    dump_config->set_bitrate_bps(*config.bitrate_bps);
+
+  if (config.frame_length_ms)
+    dump_config->set_frame_length_ms(*config.frame_length_ms);
+
+  if (config.uplink_packet_loss_fraction) {
+    dump_config->set_uplink_packet_loss_fraction(
+        *config.uplink_packet_loss_fraction);
+  }
+
+  if (config.enable_fec)
+    dump_config->set_enable_fec(*config.enable_fec);
+
+  if (config.enable_dtx)
+    dump_config->set_enable_dtx(*config.enable_dtx);
+
+  if (config.num_channels)
+    dump_config->set_num_channels(*config.num_channels);
+
+  DumpEventToFile(event, &dump_file_);
+#endif  // WEBRTC_ENABLE_PROTOBUF
+}
+
+#if WEBRTC_ENABLE_PROTOBUF
+void DebugDumpWriterImpl::DumpControllerManagerConfig(
+    const audio_network_adaptor::config::ControllerManager&
+        controller_manager_config,
+    int64_t timestamp) {
+  Event event;
+  event.set_timestamp(timestamp);
+  event.set_type(Event::CONTROLLER_MANAGER_CONFIG);
+  event.mutable_controller_manager_config()->CopyFrom(
+      controller_manager_config);
+  DumpEventToFile(event, &dump_file_);
+}
+#endif  // WEBRTC_ENABLE_PROTOBUF
+
+std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {
+  return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
new file mode 100644
index 0000000000..8fdf2f7728
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
+
+#include <memory>
+
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/system/file_wrapper.h"
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+class DebugDumpWriter {
+ public:
+  static std::unique_ptr<DebugDumpWriter> Create(FILE* file_handle);
+
+  virtual ~DebugDumpWriter() = default;
+
+  virtual void DumpEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& config,
+                                        int64_t timestamp) = 0;
+
+  virtual void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
+                                  int64_t timestamp) = 0;
+
+#if WEBRTC_ENABLE_PROTOBUF
+  virtual void DumpControllerManagerConfig(
+      const audio_network_adaptor::config::ControllerManager&
+          controller_manager_config,
+      int64_t timestamp) = 0;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
new file mode 100644
index 0000000000..b0a7d5d59d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+DtxController::Config::Config(bool initial_dtx_enabled,
+                              int dtx_enabling_bandwidth_bps,
+                              int dtx_disabling_bandwidth_bps)
+    : initial_dtx_enabled(initial_dtx_enabled),
+      dtx_enabling_bandwidth_bps(dtx_enabling_bandwidth_bps),
+      dtx_disabling_bandwidth_bps(dtx_disabling_bandwidth_bps) {}
+
+DtxController::DtxController(const Config& config)
+    : config_(config), dtx_enabled_(config_.initial_dtx_enabled) {}
+
+DtxController::~DtxController() = default;
+
+void DtxController::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.uplink_bandwidth_bps)
+    uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+}
+
+void DtxController::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  // Decision on `enable_dtx` should not have been made.
+  RTC_DCHECK(!config->enable_dtx);
+
+  if (uplink_bandwidth_bps_) {
+    if (dtx_enabled_ &&
+        *uplink_bandwidth_bps_ >= config_.dtx_disabling_bandwidth_bps) {
+      dtx_enabled_ = false;
+    } else if (!dtx_enabled_ &&
+               *uplink_bandwidth_bps_ <= config_.dtx_enabling_bandwidth_bps) {
+      dtx_enabled_ = true;
+    }
+  }
+  config->enable_dtx = dtx_enabled_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
new file mode 100644
index 0000000000..b8a8e476e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+
+class DtxController final : public Controller {
+ public:
+  struct Config {
+    Config(bool initial_dtx_enabled,
+           int dtx_enabling_bandwidth_bps,
+           int dtx_disabling_bandwidth_bps);
+    bool initial_dtx_enabled;
+    // Uplink bandwidth below which DTX should be switched on.
+    int dtx_enabling_bandwidth_bps;
+    // Uplink bandwidth above which DTX should be switched off.
+    int dtx_disabling_bandwidth_bps;
+  };
+
+  explicit DtxController(const Config& config);
+
+  ~DtxController() override;
+
+  DtxController(const DtxController&) = delete;
+  DtxController& operator=(const DtxController&) = delete;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  const Config config_;
+  bool dtx_enabled_;
+  absl::optional<int> uplink_bandwidth_bps_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
new file mode 100644
index 0000000000..567df6f76e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDtxEnablingBandwidthBps = 55000;
+constexpr int kDtxDisablingBandwidthBps = 65000;
+constexpr int kMediumBandwidthBps =
+    (kDtxEnablingBandwidthBps + kDtxDisablingBandwidthBps) / 2;
+
+std::unique_ptr<DtxController> CreateController(int initial_dtx_enabled) {
+  std::unique_ptr<DtxController> controller(new DtxController(
+      DtxController::Config(initial_dtx_enabled, kDtxEnablingBandwidthBps,
+                            kDtxDisablingBandwidthBps)));
+  return controller;
+}
+
+void CheckDecision(DtxController* controller,
+                   const absl::optional<int>& uplink_bandwidth_bps,
+                   bool expected_dtx_enabled) {
+  if (uplink_bandwidth_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  AudioEncoderRuntimeConfig config;
+  controller->MakeDecision(&config);
+  EXPECT_EQ(expected_dtx_enabled, config.enable_dtx);
+}
+
+}  // namespace
+
+TEST(DtxControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
+  constexpr bool kInitialDtxEnabled = true;
+  auto controller = CreateController(kInitialDtxEnabled);
+  CheckDecision(controller.get(), absl::nullopt, kInitialDtxEnabled);
+}
+
+TEST(DtxControllerTest, TurnOnDtxForLowUplinkBandwidth) {
+  auto controller = CreateController(false);
+  CheckDecision(controller.get(), kDtxEnablingBandwidthBps, true);
+}
+
+TEST(DtxControllerTest, TurnOffDtxForHighUplinkBandwidth) {
+  auto controller = CreateController(true);
+  CheckDecision(controller.get(), kDtxDisablingBandwidthBps, false);
+}
+
+TEST(DtxControllerTest, MaintainDtxOffForMediumUplinkBandwidth) {
+  auto controller = CreateController(false);
+  CheckDecision(controller.get(), kMediumBandwidthBps, false);
+}
+
+TEST(DtxControllerTest, MaintainDtxOnForMediumUplinkBandwidth) {
+  auto controller = CreateController(true);
+  CheckDecision(controller.get(), kMediumBandwidthBps, true);
+}
+
+TEST(DtxControllerTest, CheckBehaviorOnChangingUplinkBandwidth) {
+  auto controller = CreateController(false);
+  CheckDecision(controller.get(), kMediumBandwidthBps, false);
+  CheckDecision(controller.get(), kDtxEnablingBandwidthBps, true);
+  CheckDecision(controller.get(), kMediumBandwidthBps, true);
+  CheckDecision(controller.get(), kDtxDisablingBandwidthBps, false);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
new file mode 100644
index 0000000000..0a79484a16
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <cstdlib>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+EventLogWriter::EventLogWriter(RtcEventLog* event_log,
+                               int min_bitrate_change_bps,
+                               float min_bitrate_change_fraction,
+                               float min_packet_loss_change_fraction)
+    : event_log_(event_log),
+      min_bitrate_change_bps_(min_bitrate_change_bps),
+      min_bitrate_change_fraction_(min_bitrate_change_fraction),
+      min_packet_loss_change_fraction_(min_packet_loss_change_fraction) {
+  RTC_DCHECK(event_log_);
+}
+
+EventLogWriter::~EventLogWriter() = default;
+
+void EventLogWriter::MaybeLogEncoderConfig(
+    const AudioEncoderRuntimeConfig& config) {
+  if (last_logged_config_.num_channels != config.num_channels)
+    return LogEncoderConfig(config);
+  if (last_logged_config_.enable_dtx != config.enable_dtx)
+    return LogEncoderConfig(config);
+  if (last_logged_config_.enable_fec != config.enable_fec)
+    return LogEncoderConfig(config);
+  if (last_logged_config_.frame_length_ms != config.frame_length_ms)
+    return LogEncoderConfig(config);
+  if ((!last_logged_config_.bitrate_bps && config.bitrate_bps) ||
+      (last_logged_config_.bitrate_bps && config.bitrate_bps &&
+       std::abs(*last_logged_config_.bitrate_bps - *config.bitrate_bps) >=
+           std::min(static_cast<int>(*last_logged_config_.bitrate_bps *
+                                     min_bitrate_change_fraction_),
+                    min_bitrate_change_bps_))) {
+    return LogEncoderConfig(config);
+  }
+  if ((!last_logged_config_.uplink_packet_loss_fraction &&
+       config.uplink_packet_loss_fraction) ||
+      (last_logged_config_.uplink_packet_loss_fraction &&
+       config.uplink_packet_loss_fraction &&
+       fabs(*last_logged_config_.uplink_packet_loss_fraction -
+            *config.uplink_packet_loss_fraction) >=
+           min_packet_loss_change_fraction_ *
+               *last_logged_config_.uplink_packet_loss_fraction)) {
+    return LogEncoderConfig(config);
+  }
+}
+
+void EventLogWriter::LogEncoderConfig(const AudioEncoderRuntimeConfig& config) {
+  auto config_copy = std::make_unique<AudioEncoderRuntimeConfig>(config);
+  event_log_->Log(
+      std::make_unique<RtcEventAudioNetworkAdaptation>(std::move(config_copy)));
+  last_logged_config_ = config;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
new file mode 100644
index 0000000000..a147311fc7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
+
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+class RtcEventLog;
+
+class EventLogWriter final {
+ public:
+  EventLogWriter(RtcEventLog* event_log,
+                 int min_bitrate_change_bps,
+                 float min_bitrate_change_fraction,
+                 float min_packet_loss_change_fraction);
+  ~EventLogWriter();
+
+  EventLogWriter(const EventLogWriter&) = delete;
+  EventLogWriter& operator=(const EventLogWriter&) = delete;
+
+  void MaybeLogEncoderConfig(const AudioEncoderRuntimeConfig& config);
+
+ private:
+  void LogEncoderConfig(const AudioEncoderRuntimeConfig& config);
+
+  RtcEventLog* const event_log_;
+  const int min_bitrate_change_bps_;
+  const float min_bitrate_change_fraction_;
+  const float min_packet_loss_change_fraction_;
+  AudioEncoderRuntimeConfig last_logged_config_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
new file mode 100644
index 0000000000..2c344534ea
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
@@ -0,0 +1,240 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+
+#include <memory>
+
+#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kMinBitrateChangeBps = 5000;
+constexpr float kMinPacketLossChangeFraction = 0.5;
+constexpr float kMinBitrateChangeFraction = 0.25;
+
+constexpr int kHighBitrateBps = 70000;
+constexpr int kLowBitrateBps = 10000;
+constexpr int kFrameLengthMs = 60;
+constexpr bool kEnableFec = true;
+constexpr bool kEnableDtx = true;
+constexpr float kPacketLossFraction = 0.05f;
+constexpr size_t kNumChannels = 1;
+
+MATCHER_P(IsRtcEventAnaConfigEqualTo, config, "") {
+  if (arg->GetType() != RtcEvent::Type::AudioNetworkAdaptation) {
+    return false;
+  }
+  auto ana_event = static_cast<RtcEventAudioNetworkAdaptation*>(arg);
+  return ana_event->config() == config;
+}
+
+struct EventLogWriterStates {
+  std::unique_ptr<EventLogWriter> event_log_writer;
+  std::unique_ptr<testing::StrictMock<MockRtcEventLog>> event_log;
+  AudioEncoderRuntimeConfig runtime_config;
+};
+
+EventLogWriterStates CreateEventLogWriter() {
+  EventLogWriterStates state;
+  state.event_log.reset(new ::testing::StrictMock<MockRtcEventLog>());
+  state.event_log_writer.reset(new EventLogWriter(
+      state.event_log.get(), kMinBitrateChangeBps, kMinBitrateChangeFraction,
+      kMinPacketLossChangeFraction));
+  state.runtime_config.bitrate_bps = kHighBitrateBps;
+  state.runtime_config.frame_length_ms = kFrameLengthMs;
+  state.runtime_config.uplink_packet_loss_fraction = kPacketLossFraction;
+  state.runtime_config.enable_fec = kEnableFec;
+  state.runtime_config.enable_dtx = kEnableDtx;
+  state.runtime_config.num_channels = kNumChannels;
+  return state;
+}
+}  // namespace
+
+TEST(EventLogWriterTest, FirstConfigIsLogged) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, SameConfigIsNotLogged) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogFecStateChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+
+  state.runtime_config.enable_fec = !kEnableFec;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogDtxStateChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+
+  state.runtime_config.enable_dtx = !kEnableDtx;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogChannelChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+
+  state.runtime_config.num_channels = kNumChannels + 1;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogFrameLengthChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+
+  state.runtime_config.frame_length_ms = 20;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, DoNotLogSmallBitrateChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.bitrate_bps = kHighBitrateBps + kMinBitrateChangeBps - 1;
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogLargeBitrateChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  // At high bitrate, the min fraction rule requires a larger change than the
+  // min change rule. We make sure that the min change rule applies.
+  RTC_DCHECK_GT(kHighBitrateBps * kMinBitrateChangeFraction,
+                kMinBitrateChangeBps);
+  state.runtime_config.bitrate_bps = kHighBitrateBps + kMinBitrateChangeBps;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogMinBitrateChangeFractionOnLowBitrateChange) {
+  auto state = CreateEventLogWriter();
+  state.runtime_config.bitrate_bps = kLowBitrateBps;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  // At high bitrate, the min change rule requires a larger change than the min
+  // fraction rule. We make sure that the min fraction rule applies.
+  state.runtime_config.bitrate_bps =
+      kLowBitrateBps + kLowBitrateBps * kMinBitrateChangeFraction;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, DoNotLogSmallPacketLossFractionChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.uplink_packet_loss_fraction =
+      kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction -
+      0.001f;
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogLargePacketLossFractionChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.uplink_packet_loss_fraction =
+      kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogJustOnceOnMultipleChanges) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.uplink_packet_loss_fraction =
+      kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction;
+  state.runtime_config.frame_length_ms = 20;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+}
+
+TEST(EventLogWriterTest, LogAfterGradualChange) {
+  auto state = CreateEventLogWriter();
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  state.runtime_config.bitrate_bps = kHighBitrateBps + kMinBitrateChangeBps;
+  EXPECT_CALL(*state.event_log,
+              LogProxy(IsRtcEventAnaConfigEqualTo(state.runtime_config)))
+      .Times(1);
+  for (int bitrate_bps = kHighBitrateBps;
+       bitrate_bps <= kHighBitrateBps + kMinBitrateChangeBps; bitrate_bps++) {
+    state.runtime_config.bitrate_bps = bitrate_bps;
+    state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
new file mode 100644
index 0000000000..c5e5fa76e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+
+#include <string>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+class NullSmoothingFilter final : public SmoothingFilter {
+ public:
+  void AddSample(float sample) override { last_sample_ = sample; }
+
+  absl::optional<float> GetAverage() override { return last_sample_; }
+
+  bool SetTimeConstantMs(int time_constant_ms) override {
+    RTC_DCHECK_NOTREACHED();
+    return false;
+  }
+
+ private:
+  absl::optional<float> last_sample_;
+};
+}  // namespace
+
+FecControllerPlrBased::Config::Config(
+    bool initial_fec_enabled,
+    const ThresholdCurve& fec_enabling_threshold,
+    const ThresholdCurve& fec_disabling_threshold,
+    int time_constant_ms)
+    : initial_fec_enabled(initial_fec_enabled),
+      fec_enabling_threshold(fec_enabling_threshold),
+      fec_disabling_threshold(fec_disabling_threshold),
+      time_constant_ms(time_constant_ms) {}
+
+FecControllerPlrBased::FecControllerPlrBased(
+    const Config& config,
+    std::unique_ptr<SmoothingFilter> smoothing_filter)
+    : config_(config),
+      fec_enabled_(config.initial_fec_enabled),
+      packet_loss_smoother_(std::move(smoothing_filter)) {
+  RTC_DCHECK(config_.fec_disabling_threshold <= config_.fec_enabling_threshold);
+}
+
+FecControllerPlrBased::FecControllerPlrBased(const Config& config)
+    : FecControllerPlrBased(
+          config,
+          webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"
+              ? std::unique_ptr<NullSmoothingFilter>(new NullSmoothingFilter())
+              : std::unique_ptr<SmoothingFilter>(
+                    new SmoothingFilterImpl(config.time_constant_ms))) {}
+
+FecControllerPlrBased::~FecControllerPlrBased() = default;
+
+void FecControllerPlrBased::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.uplink_bandwidth_bps)
+    uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+  if (network_metrics.uplink_packet_loss_fraction) {
+    packet_loss_smoother_->AddSample(
+        *network_metrics.uplink_packet_loss_fraction);
+  }
+}
+
+void FecControllerPlrBased::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  RTC_DCHECK(!config->enable_fec);
+  RTC_DCHECK(!config->uplink_packet_loss_fraction);
+
+  const auto& packet_loss = packet_loss_smoother_->GetAverage();
+
+  fec_enabled_ = fec_enabled_ ? !FecDisablingDecision(packet_loss)
+                              : FecEnablingDecision(packet_loss);
+
+  config->enable_fec = fec_enabled_;
+
+  config->uplink_packet_loss_fraction = packet_loss ? *packet_loss : 0.0;
+}
+
+bool FecControllerPlrBased::FecEnablingDecision(
+    const absl::optional<float>& packet_loss) const {
+  if (!uplink_bandwidth_bps_ || !packet_loss) {
+    return false;
+  } else {
+    // Enable when above the curve or exactly on it.
+    return !config_.fec_enabling_threshold.IsBelowCurve(
+        {static_cast<float>(*uplink_bandwidth_bps_), *packet_loss});
+  }
+}
+
+bool FecControllerPlrBased::FecDisablingDecision(
+    const absl::optional<float>& packet_loss) const {
+  if (!uplink_bandwidth_bps_ || !packet_loss) {
+    return false;
+  } else {
+    // Disable when below the curve.
+    return config_.fec_disabling_threshold.IsBelowCurve(
+        {static_cast<float>(*uplink_bandwidth_bps_), *packet_loss});
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
new file mode 100644
index 0000000000..0c57ad1d1e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "common_audio/smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+
+namespace webrtc {
+
+class FecControllerPlrBased final : public Controller {
+ public:
+  struct Config {
+    // `fec_enabling_threshold` defines a curve, above which FEC should be
+    // enabled. `fec_disabling_threshold` defines a curve, under which FEC
+    // should be disabled. See below
+    //
+    // packet-loss ^   |  |
+    //             |   |  |   FEC
+    //             |    \  \   ON
+    //             | FEC \  \_______ fec_enabling_threshold
+    //             | OFF  \_________ fec_disabling_threshold
+    //             |-----------------> bandwidth
+    Config(bool initial_fec_enabled,
+           const ThresholdCurve& fec_enabling_threshold,
+           const ThresholdCurve& fec_disabling_threshold,
+           int time_constant_ms);
+    bool initial_fec_enabled;
+    ThresholdCurve fec_enabling_threshold;
+    ThresholdCurve fec_disabling_threshold;
+    int time_constant_ms;
+  };
+
+  // Dependency injection for testing.
+  FecControllerPlrBased(const Config& config,
+                        std::unique_ptr<SmoothingFilter> smoothing_filter);
+
+  explicit FecControllerPlrBased(const Config& config);
+
+  ~FecControllerPlrBased() override;
+
+  FecControllerPlrBased(const FecControllerPlrBased&) = delete;
+  FecControllerPlrBased& operator=(const FecControllerPlrBased&) = delete;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  bool FecEnablingDecision(const absl::optional<float>& packet_loss) const;
+  bool FecDisablingDecision(const absl::optional<float>& packet_loss) const;
+
+  const Config config_;
+  bool fec_enabled_;
+  absl::optional<int> uplink_bandwidth_bps_;
+  const std::unique_ptr<SmoothingFilter> packet_loss_smoother_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
new file mode 100644
index 0000000000..743b087163
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
@@ -0,0 +1,489 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+
+#include <utility>
+
+#include "common_audio/mocks/mock_smoothing_filter.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace {
+
+// The test uses the following settings:
+//
+// packet-loss ^   |  |
+//             |  A| C|   FEC
+//             |    \  \   ON
+//             | FEC \ D\_______
+//             | OFF B\_________
+//             |-----------------> bandwidth
+//
+// A : (kDisablingBandwidthLow, kDisablingPacketLossAtLowBw)
+// B : (kDisablingBandwidthHigh, kDisablingPacketLossAtHighBw)
+// C : (kEnablingBandwidthLow, kEnablingPacketLossAtLowBw)
+// D : (kEnablingBandwidthHigh, kEnablingPacketLossAtHighBw)
+
+constexpr int kDisablingBandwidthLow = 15000;
+constexpr float kDisablingPacketLossAtLowBw = 0.08f;
+constexpr int kDisablingBandwidthHigh = 64000;
+constexpr float kDisablingPacketLossAtHighBw = 0.01f;
+constexpr int kEnablingBandwidthLow = 17000;
+constexpr float kEnablingPacketLossAtLowBw = 0.1f;
+constexpr int kEnablingBandwidthHigh = 64000;
+constexpr float kEnablingPacketLossAtHighBw = 0.05f;
+
+constexpr float kEpsilon = 1e-5f;
+
+struct FecControllerPlrBasedTestStates {
+  std::unique_ptr<FecControllerPlrBased> controller;
+  MockSmoothingFilter* packet_loss_smoother;
+};
+
+FecControllerPlrBasedTestStates CreateFecControllerPlrBased(
+    bool initial_fec_enabled,
+    const ThresholdCurve& enabling_curve,
+    const ThresholdCurve& disabling_curve) {
+  FecControllerPlrBasedTestStates states;
+  std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
+      new NiceMock<MockSmoothingFilter>());
+  states.packet_loss_smoother = mock_smoothing_filter.get();
+  states.controller.reset(new FecControllerPlrBased(
+      FecControllerPlrBased::Config(initial_fec_enabled, enabling_curve,
+                                    disabling_curve, 0),
+      std::move(mock_smoothing_filter)));
+  return states;
+}
+
+FecControllerPlrBasedTestStates CreateFecControllerPlrBased(
+    bool initial_fec_enabled) {
+  return CreateFecControllerPlrBased(
+      initial_fec_enabled,
+      ThresholdCurve(kEnablingBandwidthLow, kEnablingPacketLossAtLowBw,
+                     kEnablingBandwidthHigh, kEnablingPacketLossAtHighBw),
+      ThresholdCurve(kDisablingBandwidthLow, kDisablingPacketLossAtLowBw,
+                     kDisablingBandwidthHigh, kDisablingPacketLossAtHighBw));
+}
+
+void UpdateNetworkMetrics(FecControllerPlrBasedTestStates* states,
+                          const absl::optional<int>& uplink_bandwidth_bps,
+                          const absl::optional<float>& uplink_packet_loss) {
+  // UpdateNetworkMetrics can accept multiple network metric updates at once.
+  // However, currently, the most used case is to update one metric at a time.
+  // To reflect this fact, we separate the calls.
+  if (uplink_bandwidth_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+    states->controller->UpdateNetworkMetrics(network_metrics);
+  }
+  if (uplink_packet_loss) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_packet_loss_fraction = uplink_packet_loss;
+    EXPECT_CALL(*states->packet_loss_smoother, AddSample(*uplink_packet_loss));
+    states->controller->UpdateNetworkMetrics(network_metrics);
+    // This is called during CheckDecision().
+    EXPECT_CALL(*states->packet_loss_smoother, GetAverage())
+        .WillOnce(Return(*uplink_packet_loss));
+  }
+}
+
+// Checks that the FEC decision and `uplink_packet_loss_fraction` given by
+// `states->controller->MakeDecision` matches `expected_enable_fec` and
+// `expected_uplink_packet_loss_fraction`, respectively.
+void CheckDecision(FecControllerPlrBasedTestStates* states,
+                   bool expected_enable_fec,
+                   float expected_uplink_packet_loss_fraction) {
+  AudioEncoderRuntimeConfig config;
+  states->controller->MakeDecision(&config);
+  EXPECT_EQ(expected_enable_fec, config.enable_fec);
+  EXPECT_EQ(expected_uplink_packet_loss_fraction,
+            config.uplink_packet_loss_fraction);
+}
+
+}  // namespace
+
+TEST(FecControllerPlrBasedTest, OutputInitValueBeforeAnyInputsAreReceived) {
+  for (bool initial_fec_enabled : {false, true}) {
+    auto states = CreateFecControllerPlrBased(initial_fec_enabled);
+    CheckDecision(&states, initial_fec_enabled, 0);
+  }
+}
+
+TEST(FecControllerPlrBasedTest, OutputInitValueWhenUplinkBandwidthUnknown) {
+  // Regardless of the initial FEC state and the packet-loss rate,
+  // the initial FEC state is maintained as long as the BWE is unknown.
+  for (bool initial_fec_enabled : {false, true}) {
+    for (float packet_loss :
+         {kDisablingPacketLossAtLowBw - kEpsilon, kDisablingPacketLossAtLowBw,
+          kDisablingPacketLossAtLowBw + kEpsilon,
+          kEnablingPacketLossAtLowBw - kEpsilon, kEnablingPacketLossAtLowBw,
+          kEnablingPacketLossAtLowBw + kEpsilon}) {
+      auto states = CreateFecControllerPlrBased(initial_fec_enabled);
+      UpdateNetworkMetrics(&states, absl::nullopt, packet_loss);
+      CheckDecision(&states, initial_fec_enabled, packet_loss);
+    }
+  }
+}
+
+TEST(FecControllerPlrBasedTest,
+     OutputInitValueWhenUplinkPacketLossFractionUnknown) {
+  // Regardless of the initial FEC state and the BWE, the initial FEC state
+  // is maintained as long as the packet-loss rate is unknown.
+  for (bool initial_fec_enabled : {false, true}) {
+    for (int bandwidth : {kDisablingBandwidthLow - 1, kDisablingBandwidthLow,
+                          kDisablingBandwidthLow + 1, kEnablingBandwidthLow - 1,
+                          kEnablingBandwidthLow, kEnablingBandwidthLow + 1}) {
+      auto states = CreateFecControllerPlrBased(initial_fec_enabled);
+      UpdateNetworkMetrics(&states, bandwidth, absl::nullopt);
+      CheckDecision(&states, initial_fec_enabled, 0.0);
+    }
+  }
+}
+
+TEST(FecControllerPlrBasedTest, EnableFecForHighBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  UpdateNetworkMetrics(&states, kEnablingBandwidthHigh,
+                       kEnablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kEnablingPacketLossAtHighBw);
+}
+
+TEST(FecControllerPlrBasedTest, UpdateMultipleNetworkMetricsAtOnce) {
+  // This test is similar to EnableFecForHighBandwidth. But instead of
+  // using ::UpdateNetworkMetrics(...), which calls
+  // FecControllerPlrBased::UpdateNetworkMetrics(...) multiple times, we
+  // we call it only once. This is to verify that
+  // FecControllerPlrBased::UpdateNetworkMetrics(...) can handle multiple
+  // network updates at once. This is, however, not a common use case in current
+  // audio_network_adaptor_impl.cc.
+  auto states = CreateFecControllerPlrBased(false);
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.uplink_bandwidth_bps = kEnablingBandwidthHigh;
+  network_metrics.uplink_packet_loss_fraction = kEnablingPacketLossAtHighBw;
+  EXPECT_CALL(*states.packet_loss_smoother, GetAverage())
+      .WillOnce(Return(kEnablingPacketLossAtHighBw));
+  states.controller->UpdateNetworkMetrics(network_metrics);
+  CheckDecision(&states, true, kEnablingPacketLossAtHighBw);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOffForHighBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  constexpr float kPacketLoss = kEnablingPacketLossAtHighBw * 0.99f;
+  UpdateNetworkMetrics(&states, kEnablingBandwidthHigh, kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, EnableFecForMediumBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  constexpr float kPacketLoss =
+      (kEnablingPacketLossAtLowBw + kEnablingPacketLossAtHighBw) / 2.0;
+  UpdateNetworkMetrics(&states,
+                       (kEnablingBandwidthHigh + kEnablingBandwidthLow) / 2,
+                       kPacketLoss);
+  CheckDecision(&states, true, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOffForMediumBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  constexpr float kPacketLoss =
+      kEnablingPacketLossAtLowBw * 0.49f + kEnablingPacketLossAtHighBw * 0.51f;
+  UpdateNetworkMetrics(&states,
+                       (kEnablingBandwidthHigh + kEnablingBandwidthLow) / 2,
+                       kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, EnableFecForLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow,
+                       kEnablingPacketLossAtLowBw);
+  CheckDecision(&states, true, kEnablingPacketLossAtLowBw);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOffForLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  constexpr float kPacketLoss = kEnablingPacketLossAtLowBw * 0.99f;
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow, kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOffForVeryLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(false);
+  // Below `kEnablingBandwidthLow`, no packet loss fraction can cause FEC to
+  // turn on.
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow - 1, 1.0);
+  CheckDecision(&states, false, 1.0);
+}
+
+TEST(FecControllerPlrBasedTest, DisableFecForHighBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  constexpr float kPacketLoss = kDisablingPacketLossAtHighBw - kEpsilon;
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh, kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOnForHighBandwidth) {
+  // Note: Disabling happens when the value is strictly below the threshold.
+  auto states = CreateFecControllerPlrBased(true);
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh,
+                       kDisablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kDisablingPacketLossAtHighBw);
+}
+
+TEST(FecControllerPlrBasedTest, DisableFecOnMediumBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  constexpr float kPacketLoss =
+      (kDisablingPacketLossAtLowBw + kDisablingPacketLossAtHighBw) / 2.0f -
+      kEpsilon;
+  UpdateNetworkMetrics(&states,
+                       (kDisablingBandwidthHigh + kDisablingBandwidthLow) / 2,
+                       kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, MaintainFecOnForMediumBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  constexpr float kPacketLoss = kDisablingPacketLossAtLowBw * 0.51f +
+                                kDisablingPacketLossAtHighBw * 0.49f - kEpsilon;
+  UpdateNetworkMetrics(&states,
+                       (kEnablingBandwidthHigh + kDisablingBandwidthLow) / 2,
+                       kPacketLoss);
+  CheckDecision(&states, true, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, DisableFecForLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  constexpr float kPacketLoss = kDisablingPacketLossAtLowBw - kEpsilon;
+  UpdateNetworkMetrics(&states, kDisablingBandwidthLow, kPacketLoss);
+  CheckDecision(&states, false, kPacketLoss);
+}
+
+TEST(FecControllerPlrBasedTest, DisableFecForVeryLowBandwidth) {
+  auto states = CreateFecControllerPlrBased(true);
+  // Below `kEnablingBandwidthLow`, any packet loss fraction can cause FEC to
+  // turn off.
+  UpdateNetworkMetrics(&states, kDisablingBandwidthLow - 1, 1.0);
+  CheckDecision(&states, false, 1.0);
+}
+
+TEST(FecControllerPlrBasedTest, CheckBehaviorOnChangingNetworkMetrics) {
+  // In this test, we let the network metrics to traverse from 1 to 5.
+  // packet-loss ^ 1 |  |
+  //             |   | 2|
+  //             |    \  \ 3
+  //             |     \4 \_______
+  //             |      \_________
+  //             |---------5-------> bandwidth
+
+  auto states = CreateFecControllerPlrBased(true);
+  UpdateNetworkMetrics(&states, kDisablingBandwidthLow - 1, 1.0);
+  CheckDecision(&states, false, 1.0);
+
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow,
+                       kEnablingPacketLossAtLowBw * 0.99f);
+  CheckDecision(&states, false, kEnablingPacketLossAtLowBw * 0.99f);
+
+  UpdateNetworkMetrics(&states, kEnablingBandwidthHigh,
+                       kEnablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kEnablingPacketLossAtHighBw);
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh,
+                       kDisablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kDisablingPacketLossAtHighBw);
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh + 1, 0.0);
+  CheckDecision(&states, false, 0.0);
+}
+
+TEST(FecControllerPlrBasedTest, CheckBehaviorOnSpecialCurves) {
+  // We test a special configuration, where the points to define the FEC
+  // enabling/disabling curves are placed like the following, otherwise the test
+  // is the same as CheckBehaviorOnChangingNetworkMetrics.
+  //
+  // packet-loss ^   |  |
+  //             |   | C|
+  //             |   |  |
+  //             |   | D|_______
+  //             |  A|___B______
+  //             |-----------------> bandwidth
+
+  constexpr int kEnablingBandwidthHigh = kEnablingBandwidthLow;
+  constexpr float kDisablingPacketLossAtLowBw = kDisablingPacketLossAtHighBw;
+  FecControllerPlrBasedTestStates states;
+  std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
+      new NiceMock<MockSmoothingFilter>());
+  states.packet_loss_smoother = mock_smoothing_filter.get();
+  states.controller.reset(new FecControllerPlrBased(
+      FecControllerPlrBased::Config(
+          true,
+          ThresholdCurve(kEnablingBandwidthLow, kEnablingPacketLossAtLowBw,
+                         kEnablingBandwidthHigh, kEnablingPacketLossAtHighBw),
+          ThresholdCurve(kDisablingBandwidthLow, kDisablingPacketLossAtLowBw,
+                         kDisablingBandwidthHigh, kDisablingPacketLossAtHighBw),
+          0),
+      std::move(mock_smoothing_filter)));
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthLow - 1, 1.0);
+  CheckDecision(&states, false, 1.0);
+
+  UpdateNetworkMetrics(&states, kEnablingBandwidthLow,
+                       kEnablingPacketLossAtHighBw * 0.99f);
+  CheckDecision(&states, false, kEnablingPacketLossAtHighBw * 0.99f);
+
+  UpdateNetworkMetrics(&states, kEnablingBandwidthHigh,
+                       kEnablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kEnablingPacketLossAtHighBw);
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh,
+                       kDisablingPacketLossAtHighBw);
+  CheckDecision(&states, true, kDisablingPacketLossAtHighBw);
+
+  UpdateNetworkMetrics(&states, kDisablingBandwidthHigh + 1, 0.0);
+  CheckDecision(&states, false, 0.0);
+}
+
+TEST(FecControllerPlrBasedTest, SingleThresholdCurveForEnablingAndDisabling) {
+  // Note: To avoid numerical errors, keep kPacketLossAtLowBw and
+  // kPacketLossAthighBw as (negative) integer powers of 2.
+  // This is mostly relevant for the O3 case.
+  constexpr int kBandwidthLow = 10000;
+  constexpr float kPacketLossAtLowBw = 0.25f;
+  constexpr int kBandwidthHigh = 20000;
+  constexpr float kPacketLossAtHighBw = 0.125f;
+  auto curve = ThresholdCurve(kBandwidthLow, kPacketLossAtLowBw, kBandwidthHigh,
+                              kPacketLossAtHighBw);
+
+  // B* stands for "below-curve", O* for "on-curve", and A* for "above-curve".
+  //
+  //                                            //
+  // packet-loss ^                              //
+  //             |    |                         //
+  //             | B1 O1                        //
+  //             |    |                         //
+  //             |    O2                        //
+  //             |     \ A1                     //
+  //             |      \                       //
+  //             |       O3   A2                //
+  //             |     B2 \                     //
+  //             |         \                    //
+  //             |          O4--O5----          //
+  //             |                              //
+  //             |            B3                //
+  //             |-----------------> bandwidth  //
+
+  struct NetworkState {
+    int bandwidth;
+    float packet_loss;
+  };
+
+  std::vector<NetworkState> below{
+      {kBandwidthLow - 1, kPacketLossAtLowBw + 0.1f},  // B1
+      {(kBandwidthLow + kBandwidthHigh) / 2,
+       (kPacketLossAtLowBw + kPacketLossAtHighBw) / 2 - kEpsilon},  // B2
+      {kBandwidthHigh + 1, kPacketLossAtHighBw - kEpsilon}          // B3
+  };
+
+  std::vector<NetworkState> on{
+      {kBandwidthLow, kPacketLossAtLowBw + 0.1f},  // O1
+      {kBandwidthLow, kPacketLossAtLowBw},         // O2
+      {(kBandwidthLow + kBandwidthHigh) / 2,
+       (kPacketLossAtLowBw + kPacketLossAtHighBw) / 2},  // O3
+      {kBandwidthHigh, kPacketLossAtHighBw},             // O4
+      {kBandwidthHigh + 1, kPacketLossAtHighBw},         // O5
+  };
+
+  std::vector<NetworkState> above{
+      {(kBandwidthLow + kBandwidthHigh) / 2,
+       (kPacketLossAtLowBw + kPacketLossAtHighBw) / 2 + kEpsilon},  // A1
+      {kBandwidthHigh + 1, kPacketLossAtHighBw + kEpsilon},         // A2
+  };
+
+  // Test that FEC is turned off whenever we're below the curve, independent
+  // of the starting FEC state.
+  for (NetworkState net_state : below) {
+    for (bool initial_fec_enabled : {false, true}) {
+      auto states =
+          CreateFecControllerPlrBased(initial_fec_enabled, curve, curve);
+      UpdateNetworkMetrics(&states, net_state.bandwidth, net_state.packet_loss);
+      CheckDecision(&states, false, net_state.packet_loss);
+    }
+  }
+
+  // Test that FEC is turned on whenever we're on the curve or above it,
+  // independent of the starting FEC state.
+  for (const std::vector<NetworkState>& states_list : {on, above}) {
+    for (NetworkState net_state : states_list) {
+      for (bool initial_fec_enabled : {false, true}) {
+        auto states =
+            CreateFecControllerPlrBased(initial_fec_enabled, curve, curve);
+        UpdateNetworkMetrics(&states, net_state.bandwidth,
+                             net_state.packet_loss);
+        CheckDecision(&states, true, net_state.packet_loss);
+      }
+    }
+  }
+}
+
+TEST(FecControllerPlrBasedTest, FecAlwaysOff) {
+  ThresholdCurve always_off_curve(0, 1.0f + kEpsilon, 0, 1.0f + kEpsilon);
+  for (bool initial_fec_enabled : {false, true}) {
+    for (int bandwidth : {0, 10000}) {
+      for (float packet_loss : {0.0f, 0.5f, 1.0f}) {
+        auto states = CreateFecControllerPlrBased(
+            initial_fec_enabled, always_off_curve, always_off_curve);
+        UpdateNetworkMetrics(&states, bandwidth, packet_loss);
+        CheckDecision(&states, false, packet_loss);
+      }
+    }
+  }
+}
+
+TEST(FecControllerPlrBasedTest, FecAlwaysOn) {
+  ThresholdCurve always_on_curve(0, 0.0f, 0, 0.0f);
+  for (bool initial_fec_enabled : {false, true}) {
+    for (int bandwidth : {0, 10000}) {
+      for (float packet_loss : {0.0f, 0.5f, 1.0f}) {
+        auto states = CreateFecControllerPlrBased(
+            initial_fec_enabled, always_on_curve, always_on_curve);
+        UpdateNetworkMetrics(&states, bandwidth, packet_loss);
+        CheckDecision(&states, true, packet_loss);
+      }
+    }
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(FecControllerPlrBasedDeathTest, InvalidConfig) {
+  FecControllerPlrBasedTestStates states;
+  std::unique_ptr<MockSmoothingFilter> mock_smoothing_filter(
+      new NiceMock<MockSmoothingFilter>());
+  states.packet_loss_smoother = mock_smoothing_filter.get();
+  EXPECT_DEATH(
+      states.controller.reset(new FecControllerPlrBased(
+          FecControllerPlrBased::Config(
+              true,
+              ThresholdCurve(kDisablingBandwidthLow - 1,
+                             kEnablingPacketLossAtLowBw, kEnablingBandwidthHigh,
+                             kEnablingPacketLossAtHighBw),
+              ThresholdCurve(
+                  kDisablingBandwidthLow, kDisablingPacketLossAtLowBw,
+                  kDisablingBandwidthHigh, kDisablingPacketLossAtHighBw),
+              0),
+          std::move(mock_smoothing_filter))),
+      "Check failed");
+}
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
new file mode 100644
index 0000000000..c47434f9aa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
@@ -0,0 +1,201 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+
+#include <algorithm>
+#include <iterator>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kPreventOveruseMarginBps = 5000;
+
+int OverheadRateBps(size_t overhead_bytes_per_packet, int frame_length_ms) {
+  return static_cast<int>(overhead_bytes_per_packet * 8 * 1000 /
+                          frame_length_ms);
+}
+}  // namespace
+
+FrameLengthController::Config::Config(
+    const std::set<int>& encoder_frame_lengths_ms,
+    int initial_frame_length_ms,
+    int min_encoder_bitrate_bps,
+    float fl_increasing_packet_loss_fraction,
+    float fl_decreasing_packet_loss_fraction,
+    int fl_increase_overhead_offset,
+    int fl_decrease_overhead_offset,
+    std::map<FrameLengthChange, int> fl_changing_bandwidths_bps)
+    : encoder_frame_lengths_ms(encoder_frame_lengths_ms),
+      initial_frame_length_ms(initial_frame_length_ms),
+      min_encoder_bitrate_bps(min_encoder_bitrate_bps),
+      fl_increasing_packet_loss_fraction(fl_increasing_packet_loss_fraction),
+      fl_decreasing_packet_loss_fraction(fl_decreasing_packet_loss_fraction),
+      fl_increase_overhead_offset(fl_increase_overhead_offset),
+      fl_decrease_overhead_offset(fl_decrease_overhead_offset),
+      fl_changing_bandwidths_bps(std::move(fl_changing_bandwidths_bps)) {}
+
+FrameLengthController::Config::Config(const Config& other) = default;
+
+FrameLengthController::Config::~Config() = default;
+
+FrameLengthController::FrameLengthController(const Config& config)
+    : config_(config) {
+  frame_length_ms_ = std::find(config_.encoder_frame_lengths_ms.begin(),
+                               config_.encoder_frame_lengths_ms.end(),
+                               config_.initial_frame_length_ms);
+  // `encoder_frame_lengths_ms` must contain `initial_frame_length_ms`.
+  RTC_DCHECK(frame_length_ms_ != config_.encoder_frame_lengths_ms.end());
+}
+
+FrameLengthController::~FrameLengthController() = default;
+
+void FrameLengthController::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.uplink_bandwidth_bps)
+    uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+  if (network_metrics.uplink_packet_loss_fraction)
+    uplink_packet_loss_fraction_ = network_metrics.uplink_packet_loss_fraction;
+  if (network_metrics.overhead_bytes_per_packet)
+    overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet;
+}
+
+void FrameLengthController::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  // Decision on `frame_length_ms` should not have been made.
+  RTC_DCHECK(!config->frame_length_ms);
+
+  if (FrameLengthIncreasingDecision(*config)) {
+    prev_decision_increase_ = true;
+  } else if (FrameLengthDecreasingDecision(*config)) {
+    prev_decision_increase_ = false;
+  }
+  config->last_fl_change_increase = prev_decision_increase_;
+  config->frame_length_ms = *frame_length_ms_;
+}
+
+FrameLengthController::Config::FrameLengthChange::FrameLengthChange(
+    int from_frame_length_ms,
+    int to_frame_length_ms)
+    : from_frame_length_ms(from_frame_length_ms),
+      to_frame_length_ms(to_frame_length_ms) {}
+
+bool FrameLengthController::Config::FrameLengthChange::operator<(
+    const FrameLengthChange& rhs) const {
+  return from_frame_length_ms < rhs.from_frame_length_ms ||
+         (from_frame_length_ms == rhs.from_frame_length_ms &&
+          to_frame_length_ms < rhs.to_frame_length_ms);
+}
+
+bool FrameLengthController::FrameLengthIncreasingDecision(
+    const AudioEncoderRuntimeConfig& config) {
+  // Increase frame length if
+  // 1. `uplink_bandwidth_bps` is known to be smaller or equal than
+  //    `min_encoder_bitrate_bps` plus `prevent_overuse_margin_bps` plus the
+  //    current overhead rate OR all the following:
+  // 2. longer frame length is available AND
+  // 3. `uplink_bandwidth_bps` is known to be smaller than a threshold AND
+  // 4. `uplink_packet_loss_fraction` is known to be smaller than a threshold.
+
+  // Find next frame length to which a criterion is defined to shift from
+  // current frame length.
+  auto longer_frame_length_ms = std::next(frame_length_ms_);
+  auto increase_threshold = config_.fl_changing_bandwidths_bps.end();
+  while (longer_frame_length_ms != config_.encoder_frame_lengths_ms.end()) {
+    increase_threshold = config_.fl_changing_bandwidths_bps.find(
+        Config::FrameLengthChange(*frame_length_ms_, *longer_frame_length_ms));
+    if (increase_threshold != config_.fl_changing_bandwidths_bps.end())
+      break;
+    longer_frame_length_ms = std::next(longer_frame_length_ms);
+  }
+
+  if (increase_threshold == config_.fl_changing_bandwidths_bps.end())
+    return false;
+
+  // Check that
+  // -(*overhead_bytes_per_packet_) <= offset <= (*overhead_bytes_per_packet_)
+  RTC_DCHECK(
+      !overhead_bytes_per_packet_ ||
+      (overhead_bytes_per_packet_ &&
+       static_cast<size_t>(std::max(0, -config_.fl_increase_overhead_offset)) <=
+           *overhead_bytes_per_packet_ &&
+       static_cast<size_t>(std::max(0, config_.fl_increase_overhead_offset)) <=
+           *overhead_bytes_per_packet_));
+
+  if (uplink_bandwidth_bps_ && overhead_bytes_per_packet_ &&
+      *uplink_bandwidth_bps_ <=
+          config_.min_encoder_bitrate_bps + kPreventOveruseMarginBps +
+              OverheadRateBps(*overhead_bytes_per_packet_ +
+                                  config_.fl_increase_overhead_offset,
+                              *frame_length_ms_)) {
+    frame_length_ms_ = longer_frame_length_ms;
+    return true;
+  }
+
+  if ((uplink_bandwidth_bps_ &&
+       *uplink_bandwidth_bps_ <= increase_threshold->second) &&
+      (uplink_packet_loss_fraction_ &&
+       *uplink_packet_loss_fraction_ <=
+           config_.fl_increasing_packet_loss_fraction)) {
+    frame_length_ms_ = longer_frame_length_ms;
+    return true;
+  }
+  return false;
+}
+
+bool FrameLengthController::FrameLengthDecreasingDecision(
+    const AudioEncoderRuntimeConfig& config) {
+  // Decrease frame length if
+  // 1. shorter frame length is available AND
+  // 2. `uplink_bandwidth_bps` is known to be bigger than
+  // `min_encoder_bitrate_bps` plus `prevent_overuse_margin_bps` plus the
+  // overhead which would be produced with the shorter frame length AND
+  // one or more of the followings:
+  // 3. `uplink_bandwidth_bps` is known to be larger than a threshold,
+  // 4. `uplink_packet_loss_fraction` is known to be larger than a threshold,
+
+  // Find next frame length to which a criterion is defined to shift from
+  // current frame length.
+  auto shorter_frame_length_ms = frame_length_ms_;
+  auto decrease_threshold = config_.fl_changing_bandwidths_bps.end();
+  while (shorter_frame_length_ms != config_.encoder_frame_lengths_ms.begin()) {
+    shorter_frame_length_ms = std::prev(shorter_frame_length_ms);
+    decrease_threshold = config_.fl_changing_bandwidths_bps.find(
+        Config::FrameLengthChange(*frame_length_ms_, *shorter_frame_length_ms));
+    if (decrease_threshold != config_.fl_changing_bandwidths_bps.end())
+      break;
+  }
+
+  if (decrease_threshold == config_.fl_changing_bandwidths_bps.end())
+    return false;
+
+  if (uplink_bandwidth_bps_ && overhead_bytes_per_packet_ &&
+      *uplink_bandwidth_bps_ <=
+          config_.min_encoder_bitrate_bps + kPreventOveruseMarginBps +
+              OverheadRateBps(*overhead_bytes_per_packet_ +
+                                  config_.fl_decrease_overhead_offset,
+                              *shorter_frame_length_ms)) {
+    return false;
+  }
+
+  if ((uplink_bandwidth_bps_ &&
+       *uplink_bandwidth_bps_ >= decrease_threshold->second) ||
+      (uplink_packet_loss_fraction_ &&
+       *uplink_packet_loss_fraction_ >=
+           config_.fl_decreasing_packet_loss_fraction)) {
+    frame_length_ms_ = shorter_frame_length_ms;
+    return true;
+  }
+  return false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
new file mode 100644
index 0000000000..04693f8db7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
+
+#include <stddef.h>
+
+#include <map>
+#include <set>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+
+// Determines target frame length based on the network metrics and the decision
+// of FEC controller.
+class FrameLengthController final : public Controller {
+ public:
+  struct Config {
+    struct FrameLengthChange {
+      FrameLengthChange(int from_frame_length_ms, int to_frame_length_ms);
+      bool operator<(const FrameLengthChange& rhs) const;
+      int from_frame_length_ms;
+      int to_frame_length_ms;
+    };
+    Config(const std::set<int>& encoder_frame_lengths_ms,
+           int initial_frame_length_ms,
+           int min_encoder_bitrate_bps,
+           float fl_increasing_packet_loss_fraction,
+           float fl_decreasing_packet_loss_fraction,
+           int fl_increase_overhead_offset,
+           int fl_decrease_overhead_offset,
+           std::map<FrameLengthChange, int> fl_changing_bandwidths_bps);
+    Config(const Config& other);
+    ~Config();
+    std::set<int> encoder_frame_lengths_ms;
+    int initial_frame_length_ms;
+    int min_encoder_bitrate_bps;
+    // Uplink packet loss fraction below which frame length can increase.
+    float fl_increasing_packet_loss_fraction;
+    // Uplink packet loss fraction below which frame length should decrease.
+    float fl_decreasing_packet_loss_fraction;
+    // Offset to apply to overhead calculation when increasing frame length.
+    int fl_increase_overhead_offset;
+    // Offset to apply to overhead calculation when decreasing frame length.
+    int fl_decrease_overhead_offset;
+    std::map<FrameLengthChange, int> fl_changing_bandwidths_bps;
+  };
+
+  explicit FrameLengthController(const Config& config);
+
+  ~FrameLengthController() override;
+
+  FrameLengthController(const FrameLengthController&) = delete;
+  FrameLengthController& operator=(const FrameLengthController&) = delete;
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  bool FrameLengthIncreasingDecision(const AudioEncoderRuntimeConfig& config);
+
+  bool FrameLengthDecreasingDecision(const AudioEncoderRuntimeConfig& config);
+
+  const Config config_;
+
+  std::set<int>::const_iterator frame_length_ms_;
+
+  absl::optional<int> uplink_bandwidth_bps_;
+
+  absl::optional<float> uplink_packet_loss_fraction_;
+
+  absl::optional<size_t> overhead_bytes_per_packet_;
+
+  // True if the previous frame length decision was an increase, otherwise
+  // false.
+  bool prev_decision_increase_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
new file mode 100644
index 0000000000..23123934dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
@@ -0,0 +1,444 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+
+#include <memory>
+#include <utility>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr float kFlIncreasingPacketLossFraction = 0.04f;
+constexpr float kFlDecreasingPacketLossFraction = 0.05f;
+constexpr int kFlIncreaseOverheadOffset = 0;
+constexpr int kFlDecreaseOverheadOffset = 0;
+constexpr int kMinEncoderBitrateBps = 6000;
+constexpr int kPreventOveruseMarginBps = 5000;
+constexpr size_t kOverheadBytesPerPacket = 20;
+constexpr int kFl20msTo60msBandwidthBps = 40000;
+constexpr int kFl60msTo20msBandwidthBps = 50000;
+constexpr int kFl60msTo120msBandwidthBps = 30000;
+constexpr int kFl120msTo60msBandwidthBps = 40000;
+constexpr int kFl20msTo40msBandwidthBps = 45000;
+constexpr int kFl40msTo20msBandwidthBps = 50000;
+constexpr int kFl40msTo60msBandwidthBps = 40000;
+constexpr int kFl60msTo40msBandwidthBps = 45000;
+
+constexpr int kMediumBandwidthBps =
+    (kFl40msTo20msBandwidthBps + kFl20msTo40msBandwidthBps) / 2;
+constexpr float kMediumPacketLossFraction =
+    (kFlDecreasingPacketLossFraction + kFlIncreasingPacketLossFraction) / 2;
+const std::set<int> kDefaultEncoderFrameLengthsMs = {20, 40, 60, 120};
+
+int VeryLowBitrate(int frame_length_ms) {
+  return kMinEncoderBitrateBps + kPreventOveruseMarginBps +
+         (kOverheadBytesPerPacket * 8 * 1000 / frame_length_ms);
+}
+
+std::unique_ptr<FrameLengthController> CreateController(
+    const std::map<FrameLengthController::Config::FrameLengthChange, int>&
+        frame_length_change_criteria,
+    const std::set<int>& encoder_frame_lengths_ms,
+    int initial_frame_length_ms) {
+  std::unique_ptr<FrameLengthController> controller(
+      new FrameLengthController(FrameLengthController::Config(
+          encoder_frame_lengths_ms, initial_frame_length_ms,
+          kMinEncoderBitrateBps, kFlIncreasingPacketLossFraction,
+          kFlDecreasingPacketLossFraction, kFlIncreaseOverheadOffset,
+          kFlDecreaseOverheadOffset, frame_length_change_criteria)));
+
+  return controller;
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor20msAnd60ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(20, 60),
+       kFl20msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 20),
+       kFl60msTo20msBandwidthBps}};
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor20msAnd40ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(20, 40),
+       kFl20msTo40msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(40, 20),
+       kFl40msTo20msBandwidthBps}};
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor20ms60msAnd120ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(20, 60),
+       kFl20msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 20),
+       kFl60msTo20msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 120),
+       kFl60msTo120msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(120, 60),
+       kFl120msTo60msBandwidthBps}};
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor20ms40ms60msAnd120ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(20, 60),
+       kFl20msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 20),
+       kFl60msTo20msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(20, 40),
+       kFl20msTo40msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(40, 20),
+       kFl40msTo20msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(40, 60),
+       kFl40msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 40),
+       kFl60msTo40msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 120),
+       kFl60msTo120msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(120, 60),
+       kFl120msTo60msBandwidthBps}};
+}
+
+std::map<FrameLengthController::Config::FrameLengthChange, int>
+CreateChangeCriteriaFor40msAnd60ms() {
+  return std::map<FrameLengthController::Config::FrameLengthChange, int>{
+      {FrameLengthController::Config::FrameLengthChange(40, 60),
+       kFl40msTo60msBandwidthBps},
+      {FrameLengthController::Config::FrameLengthChange(60, 40),
+       kFl60msTo40msBandwidthBps}};
+}
+
+void UpdateNetworkMetrics(
+    FrameLengthController* controller,
+    const absl::optional<int>& uplink_bandwidth_bps,
+    const absl::optional<float>& uplink_packet_loss_fraction,
+    const absl::optional<size_t>& overhead_bytes_per_packet) {
+  // UpdateNetworkMetrics can accept multiple network metric updates at once.
+  // However, currently, the most used case is to update one metric at a time.
+  // To reflect this fact, we separate the calls.
+  if (uplink_bandwidth_bps) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  if (uplink_packet_loss_fraction) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+  if (overhead_bytes_per_packet) {
+    Controller::NetworkMetrics network_metrics;
+    network_metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
+    controller->UpdateNetworkMetrics(network_metrics);
+  }
+}
+
+void CheckDecision(FrameLengthController* controller,
+                   int expected_frame_length_ms) {
+  AudioEncoderRuntimeConfig config;
+  controller->MakeDecision(&config);
+  EXPECT_EQ(expected_frame_length_ms, config.frame_length_ms);
+}
+
+}  // namespace
+
+TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
+                       absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkPacketLossFraction) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  UpdateNetworkMetrics(controller.get(), absl::nullopt,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest,
+     Maintain60MsIf20MsNotInReceiverFrameLengthRange) {
+  auto controller =
+      CreateController(CreateChangeCriteriaFor20msAnd60ms(), {60}, 60);
+  // Set FEC on that would cause frame length to decrease if receiver frame
+  // length range included 20ms.
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, IncreaseTo40MsOnMultipleConditions) {
+  // Increase to 40ms frame length if
+  // 1. `uplink_bandwidth_bps` is known to be smaller than a threshold AND
+  // 2. `uplink_packet_loss_fraction` is known to be smaller than a threshold
+  //    AND
+  // 3. FEC is not decided or OFF.
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd40ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  UpdateNetworkMetrics(controller.get(), kFl20msTo40msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+}
+
+TEST(FrameLengthControllerTest, DecreaseTo40MsOnHighUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor40msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 40);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo40msBandwidthBps,
+                       absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+}
+
+TEST(FrameLengthControllerTest, Maintain60MsOnMultipleConditions) {
+  // Maintain 60ms frame length if
+  // 1. `uplink_bandwidth_bps` is at medium level,
+  // 2. `uplink_packet_loss_fraction` is at medium,
+  // 3. FEC is not decided ON.
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
+                       kMediumPacketLossFraction, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, IncreaseTo60MsOnMultipleConditions) {
+  // Increase to 60ms frame length if
+  // 1. `uplink_bandwidth_bps` is known to be smaller than a threshold AND
+  // 2. `uplink_packet_loss_fraction` is known to be smaller than a threshold
+  //    AND
+  // 3. FEC is not decided or OFF.
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, IncreaseTo60MsOnVeryLowUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  // We set packet loss fraction to kFlDecreasingPacketLossFraction, which
+  // should have prevented frame length to increase, if the uplink bandwidth
+  // was not this low.
+  UpdateNetworkMetrics(controller.get(), VeryLowBitrate(20),
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, Maintain60MsOnVeryLowUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  // We set packet loss fraction to FlDecreasingPacketLossFraction, which should
+  // have caused the frame length to decrease, if the uplink bandwidth was not
+  // this low.
+  UpdateNetworkMetrics(controller.get(), VeryLowBitrate(20),
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, UpdateMultipleNetworkMetricsAtOnce) {
+  // This test is similar to IncreaseTo60MsOnMultipleConditions. But instead of
+  // using ::UpdateNetworkMetrics(...), which calls
+  // FrameLengthController::UpdateNetworkMetrics(...) multiple times, we
+  // we call it only once. This is to verify that
+  // FrameLengthController::UpdateNetworkMetrics(...) can handle multiple
+  // network updates at once. This is, however, not a common use case in current
+  // audio_network_adaptor_impl.cc.
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  Controller::NetworkMetrics network_metrics;
+  network_metrics.uplink_bandwidth_bps = kFl20msTo60msBandwidthBps;
+  network_metrics.uplink_packet_loss_fraction = kFlIncreasingPacketLossFraction;
+  controller->UpdateNetworkMetrics(network_metrics);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest,
+     Maintain20MsIf60MsNotInReceiverFrameLengthRange) {
+  auto controller =
+      CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20}, 20);
+  // Use a low uplink bandwidth and a low uplink packet loss fraction that would
+  // cause frame length to increase if receiver frame length included 60ms.
+  UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, Maintain20MsOnMediumUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, Maintain20MsOnMediumUplinkPacketLossFraction) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  // Use a low uplink bandwidth that would cause frame length to increase if
+  // uplink packet loss fraction was low.
+  UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+                       kMediumPacketLossFraction, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, Maintain60MsWhenNo120msCriteriaIsSet) {
+  auto controller = CreateController(CreateChangeCriteriaFor20msAnd60ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, From120MsTo20MsOnHighUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 120);
+  // It takes two steps for frame length to go from 120ms to 20ms.
+  UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
+                       absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
+                       absl::nullopt, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, From120MsTo20MsOnHighUplinkPacketLossFraction) {
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 120);
+  // It takes two steps for frame length to go from 120ms to 20ms.
+  UpdateNetworkMetrics(controller.get(), absl::nullopt,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), absl::nullopt,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+TEST(FrameLengthControllerTest, Maintain120MsOnVeryLowUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 120);
+  // We set packet loss fraction to FlDecreasingPacketLossFraction, which should
+  // have caused the frame length to decrease, if the uplink bandwidth was not
+  // this low.
+  UpdateNetworkMetrics(controller.get(), VeryLowBitrate(60),
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 120);
+}
+
+TEST(FrameLengthControllerTest, From60MsTo120MsOnVeryLowUplinkBandwidth) {
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 60);
+  // We set packet loss fraction to FlDecreasingPacketLossFraction, which should
+  // have prevented frame length to increase, if the uplink bandwidth was not
+  // this low.
+  UpdateNetworkMetrics(controller.get(), VeryLowBitrate(60),
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 120);
+}
+
+TEST(FrameLengthControllerTest, From20MsTo120MsOnMultipleConditions) {
+  // Increase to 120ms frame length if
+  // 1. `uplink_bandwidth_bps` is known to be smaller than a threshold AND
+  // 2. `uplink_packet_loss_fraction` is known to be smaller than a threshold.
+  auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
+                                     kDefaultEncoderFrameLengthsMs, 20);
+  // It takes two steps for frame length to go from 20ms to 120ms.
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 120);
+}
+
+TEST(FrameLengthControllerTest, Stall60MsIf120MsNotInReceiverFrameLengthRange) {
+  auto controller =
+      CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(), {20, 60}, 20);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+}
+
+TEST(FrameLengthControllerTest, CheckBehaviorOnChangingNetworkMetrics) {
+  auto controller =
+      CreateController(CreateChangeCriteriaFor20ms40ms60msAnd120ms(),
+                       kDefaultEncoderFrameLengthsMs, 20);
+  UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+
+  UpdateNetworkMetrics(controller.get(), kFl20msTo40msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo40msBandwidthBps,
+                       kMediumPacketLossFraction, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+
+  UpdateNetworkMetrics(controller.get(), kFl20msTo60msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kMediumPacketLossFraction, kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo120msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 120);
+
+  UpdateNetworkMetrics(controller.get(), kFl120msTo60msBandwidthBps,
+                       kFlIncreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 60);
+
+  UpdateNetworkMetrics(controller.get(), kFl60msTo40msBandwidthBps,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 40);
+
+  UpdateNetworkMetrics(controller.get(), kMediumBandwidthBps,
+                       kFlDecreasingPacketLossFraction,
+                       kOverheadBytesPerPacket);
+  CheckDecision(controller.get(), 20);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc
new file mode 100644
index 0000000000..36fc10ba82
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h"
+
+#include <algorithm>
+
+#include "absl/algorithm/container.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+int OverheadBps(int overhead_bytes_per_packet, int frame_length_ms) {
+  return overhead_bytes_per_packet * 8 * 1000 / frame_length_ms;
+}
+
+}  // namespace
+
+FrameLengthControllerV2::FrameLengthControllerV2(
+    rtc::ArrayView<const int> encoder_frame_lengths_ms,
+    int min_payload_bitrate_bps,
+    bool use_slow_adaptation)
+    : encoder_frame_lengths_ms_(encoder_frame_lengths_ms.begin(),
+                                encoder_frame_lengths_ms.end()),
+      min_payload_bitrate_bps_(min_payload_bitrate_bps),
+      use_slow_adaptation_(use_slow_adaptation) {
+  RTC_CHECK(!encoder_frame_lengths_ms_.empty());
+  absl::c_sort(encoder_frame_lengths_ms_);
+}
+
+void FrameLengthControllerV2::UpdateNetworkMetrics(
+    const NetworkMetrics& network_metrics) {
+  if (network_metrics.target_audio_bitrate_bps) {
+    target_bitrate_bps_ = network_metrics.target_audio_bitrate_bps;
+  }
+  if (network_metrics.overhead_bytes_per_packet) {
+    overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet;
+  }
+  if (network_metrics.uplink_bandwidth_bps) {
+    uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
+  }
+}
+
+void FrameLengthControllerV2::MakeDecision(AudioEncoderRuntimeConfig* config) {
+  if (!target_bitrate_bps_ || !overhead_bytes_per_packet_ ||
+      !uplink_bandwidth_bps_) {
+    return;
+  }
+
+  auto it =
+      absl::c_find_if(encoder_frame_lengths_ms_, [&](int frame_length_ms) {
+        int target = use_slow_adaptation_ ? *uplink_bandwidth_bps_
+                                          : *target_bitrate_bps_;
+        return target -
+                   OverheadBps(*overhead_bytes_per_packet_, frame_length_ms) >
+               min_payload_bitrate_bps_;
+      });
+
+  // Longest frame length is chosen if none match our criteria.
+  config->frame_length_ms = it != encoder_frame_lengths_ms_.end()
+                                ? *it
+                                : encoder_frame_lengths_ms_.back();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
new file mode 100644
index 0000000000..d7102b0b44
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_V2_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_V2_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+
+namespace webrtc {
+
+class FrameLengthControllerV2 final : public Controller {
+ public:
+  FrameLengthControllerV2(rtc::ArrayView<const int> encoder_frame_lengths_ms,
+                          int min_payload_bitrate_bps,
+                          bool use_slow_adaptation);
+
+  void UpdateNetworkMetrics(const NetworkMetrics& network_metrics) override;
+
+  void MakeDecision(AudioEncoderRuntimeConfig* config) override;
+
+ private:
+  std::vector<int> encoder_frame_lengths_ms_;
+  const int min_payload_bitrate_bps_;
+  const bool use_slow_adaptation_;
+
+  absl::optional<int> uplink_bandwidth_bps_;
+  absl::optional<int> target_bitrate_bps_;
+  absl::optional<int> overhead_bytes_per_packet_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_V2_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc
new file mode 100644
index 0000000000..1c88f47c58
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2_unittest.cc
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60, 120};
+constexpr int kMinPayloadBitrateBps = 16000;
+
+}  // namespace
+
+class FrameLengthControllerV2Test : public testing::Test {
+ protected:
+  AudioEncoderRuntimeConfig GetDecision() {
+    AudioEncoderRuntimeConfig config;
+    controller_->MakeDecision(&config);
+    return config;
+  }
+
+  void SetOverhead(int overhead_bytes_per_packet) {
+    overhead_bytes_per_packet_ = overhead_bytes_per_packet;
+    Controller::NetworkMetrics metrics;
+    metrics.overhead_bytes_per_packet = overhead_bytes_per_packet;
+    controller_->UpdateNetworkMetrics(metrics);
+  }
+
+  void SetTargetBitrate(int target_audio_bitrate_bps) {
+    target_audio_bitrate_bps_ = target_audio_bitrate_bps;
+    Controller::NetworkMetrics metrics;
+    metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
+    controller_->UpdateNetworkMetrics(metrics);
+  }
+
+  void SetUplinkBandwidth(int uplink_bandwidth_bps) {
+    Controller::NetworkMetrics metrics;
+    metrics.uplink_bandwidth_bps = uplink_bandwidth_bps;
+    controller_->UpdateNetworkMetrics(metrics);
+  }
+
+  void ExpectFrameLengthDecision(int expected_frame_length_ms) {
+    auto config = GetDecision();
+    EXPECT_EQ(*config.frame_length_ms, expected_frame_length_ms);
+  }
+
+  std::unique_ptr<FrameLengthControllerV2> controller_ =
+      std::make_unique<FrameLengthControllerV2>(kANASupportedFrameLengths,
+                                                kMinPayloadBitrateBps,
+                                                /*use_slow_adaptation=*/false);
+  absl::optional<int> target_audio_bitrate_bps_;
+  absl::optional<int> overhead_bytes_per_packet_;
+};
+
+// Don't return any decision if we haven't received all required network
+// metrics.
+TEST_F(FrameLengthControllerV2Test, RequireNetworkMetrics) {
+  auto config = GetDecision();
+  EXPECT_FALSE(config.bitrate_bps);
+  EXPECT_FALSE(config.frame_length_ms);
+
+  SetOverhead(30);
+  config = GetDecision();
+  EXPECT_FALSE(config.frame_length_ms);
+
+  SetTargetBitrate(32000);
+  config = GetDecision();
+  EXPECT_FALSE(config.frame_length_ms);
+
+  SetUplinkBandwidth(32000);
+  config = GetDecision();
+  EXPECT_TRUE(config.frame_length_ms);
+}
+
+TEST_F(FrameLengthControllerV2Test, UseFastAdaptation) {
+  SetOverhead(50);
+  SetTargetBitrate(50000);
+  SetUplinkBandwidth(50000);
+  ExpectFrameLengthDecision(20);
+
+  SetTargetBitrate(20000);
+  ExpectFrameLengthDecision(120);
+
+  SetTargetBitrate(30000);
+  ExpectFrameLengthDecision(40);
+
+  SetTargetBitrate(25000);
+  ExpectFrameLengthDecision(60);
+}
+
+TEST_F(FrameLengthControllerV2Test, UseSlowAdaptation) {
+  controller_ = std::make_unique<FrameLengthControllerV2>(
+      kANASupportedFrameLengths, kMinPayloadBitrateBps,
+      /*use_slow_adaptation=*/true);
+  SetOverhead(50);
+  SetTargetBitrate(50000);
+  SetUplinkBandwidth(20000);
+  ExpectFrameLengthDecision(120);
+
+  SetUplinkBandwidth(30000);
+  ExpectFrameLengthDecision(40);
+
+  SetUplinkBandwidth(40000);
+  ExpectFrameLengthDecision(20);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
new file mode 100644
index 0000000000..346ed5db1a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+
+namespace webrtc {
+
+// An AudioNetworkAdaptor optimizes the audio experience by suggesting a
+// suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
+// encoder based on network metrics.
+class AudioNetworkAdaptor {
+ public:
+  virtual ~AudioNetworkAdaptor() = default;
+
+  virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
+
+  virtual void SetUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) = 0;
+
+  virtual void SetRtt(int rtt_ms) = 0;
+
+  virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0;
+
+  virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0;
+
+  virtual AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
+
+  virtual void StartDebugDump(FILE* file_handle) = 0;
+
+  virtual void StopDebugDump() = 0;
+
+  virtual ANAStats GetStats() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
new file mode 100644
index 0000000000..bd16292f7e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+struct AudioEncoderRuntimeConfig {
+  AudioEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
+  ~AudioEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig& operator=(const AudioEncoderRuntimeConfig& other);
+  bool operator==(const AudioEncoderRuntimeConfig& other) const;
+  absl::optional<int> bitrate_bps;
+  absl::optional<int> frame_length_ms;
+  // Note: This is what we tell the encoder. It doesn't have to reflect
+  // the actual NetworkMetrics; it's subject to our decision.
+  absl::optional<float> uplink_packet_loss_fraction;
+  absl::optional<bool> enable_fec;
+  absl::optional<bool> enable_dtx;
+
+  // Some encoders can encode fewer channels than the actual input to make
+  // better use of the bandwidth. `num_channels` sets the number of channels
+  // to encode.
+  absl::optional<size_t> num_channels;
+
+  // This is true if the last frame length change was an increase, and otherwise
+  // false.
+  // The value of this boolean is used to apply a different offset to the
+  // per-packet overhead that is reported by the BWE. The exact offset value
+  // is most important right after a frame length change, because the frame
+  // length change affects the overhead. In the steady state, the exact value is
+  // not important because the BWE will compensate.
+  bool last_fl_change_increase = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
new file mode 100644
index 0000000000..26a9061745
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
+
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioNetworkAdaptor : public AudioNetworkAdaptor {
+ public:
+  ~MockAudioNetworkAdaptor() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+
+  MOCK_METHOD(void, SetUplinkBandwidth, (int uplink_bandwidth_bps), (override));
+
+  MOCK_METHOD(void,
+              SetUplinkPacketLossFraction,
+              (float uplink_packet_loss_fraction),
+              (override));
+
+  MOCK_METHOD(void, SetRtt, (int rtt_ms), (override));
+
+  MOCK_METHOD(void,
+              SetTargetAudioBitrate,
+              (int target_audio_bitrate_bps),
+              (override));
+
+  MOCK_METHOD(void,
+              SetOverhead,
+              (size_t overhead_bytes_per_packet),
+              (override));
+
+  MOCK_METHOD(AudioEncoderRuntimeConfig,
+              GetEncoderRuntimeConfig,
+              (),
+              (override));
+
+  MOCK_METHOD(void, StartDebugDump, (FILE * file_handle), (override));
+
+  MOCK_METHOD(void, StopDebugDump, (), (override));
+
+  MOCK_METHOD(ANAStats, GetStats, (), (const, override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h
new file mode 100644
index 0000000000..de554c0517
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
+
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockController : public Controller {
+ public:
+  ~MockController() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(void,
+              UpdateNetworkMetrics,
+              (const NetworkMetrics& network_metrics),
+              (override));
+  MOCK_METHOD(void,
+              MakeDecision,
+              (AudioEncoderRuntimeConfig * config),
+              (override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h
new file mode 100644
index 0000000000..9e2fa466fc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
+
+#include <vector>
+
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockControllerManager : public ControllerManager {
+ public:
+  ~MockControllerManager() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(std::vector<Controller*>,
+              GetSortedControllers,
+              (const Controller::NetworkMetrics& metrics),
+              (override));
+  MOCK_METHOD(std::vector<Controller*>, GetControllers, (), (const, override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
new file mode 100644
index 0000000000..0c6a9efe1d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
+
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDebugDumpWriter : public DebugDumpWriter {
+ public:
+  ~MockDebugDumpWriter() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+
+  MOCK_METHOD(void,
+              DumpEncoderRuntimeConfig,
+              (const AudioEncoderRuntimeConfig& config, int64_t timestamp),
+              (override));
+  MOCK_METHOD(void,
+              DumpNetworkMetrics,
+              (const Controller::NetworkMetrics& metrics, int64_t timestamp),
+              (override));
+#if WEBRTC_ENABLE_PROTOBUF
+  MOCK_METHOD(void,
+              DumpControllerManagerConfig,
+              (const audio_network_adaptor::config::ControllerManager&
+                   controller_manager_config,
+               int64_t timestamp),
+              (override));
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/parse_ana_dump.py b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/parse_ana_dump.py
new file mode 100755
index 0000000000..9c07c18c84
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/parse_ana_dump.py
@@ -0,0 +1,149 @@
+#!/usr/bin/python2
+#  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+#  Use of this source code is governed by a BSD-style license
+#  that can be found in the LICENSE file in the root of the source
+#  tree. An additional intellectual property rights grant can be found
+#  in the file PATENTS.  All contributing project authors may
+#  be found in the AUTHORS file in the root of the source tree.
+
+#  To run this script please copy "out/<build_name>//pyproto/webrtc/modules/
+#  audio_coding/audio_network_adaptor/debug_dump_pb2.py" to this folder.
+#  The you can run this script with:
+#  "python parse_ana_dump.py -m uplink_bandwidth_bps -f dump_file.dat"
+#  You can add as may metrics or decisions to the plot as you like.
+#  form more information call:
+#  "python parse_ana_dump.py --help"
+
+import struct
+from optparse import OptionParser
+
+import matplotlib.pyplot as plt
+
+import debug_dump_pb2
+
+
+def GetNextMessageSize(file_to_parse):
+    data = file_to_parse.read(4)
+    if data == '':
+        return 0
+    return struct.unpack('<I', data)[0]
+
+
+def GetNextMessageFromFile(file_to_parse):
+    message_size = GetNextMessageSize(file_to_parse)
+    if message_size == 0:
+        return None
+    try:
+        event = debug_dump_pb2.Event()
+        event.ParseFromString(file_to_parse.read(message_size))
+    except IOError:
+        print 'Invalid message in file'
+        return None
+    return event
+
+
+def InitMetrics():
+    metrics = {}
+    event = debug_dump_pb2.Event()
+    for metric in event.network_metrics.DESCRIPTOR.fields:
+        metrics[metric.name] = {'time': [], 'value': []}
+    return metrics
+
+
+def InitDecisions():
+    decisions = {}
+    event = debug_dump_pb2.Event()
+    for decision in event.encoder_runtime_config.DESCRIPTOR.fields:
+        decisions[decision.name] = {'time': [], 'value': []}
+    return decisions
+
+
+def ParseAnaDump(dump_file_to_parse):
+    with open(dump_file_to_parse, 'rb') as file_to_parse:
+        metrics = InitMetrics()
+        decisions = InitDecisions()
+        first_time_stamp = None
+        while True:
+            event = GetNextMessageFromFile(file_to_parse)
+            if event is None:
+                break
+            if first_time_stamp is None:
+                first_time_stamp = event.timestamp
+            if event.type == debug_dump_pb2.Event.ENCODER_RUNTIME_CONFIG:
+                for decision in event.encoder_runtime_config.DESCRIPTOR.fields:
+                    if event.encoder_runtime_config.HasField(decision.name):
+                        decisions[decision.name]['time'].append(
+                            event.timestamp - first_time_stamp)
+                        decisions[decision.name]['value'].append(
+                            getattr(event.encoder_runtime_config,
+                                    decision.name))
+            if event.type == debug_dump_pb2.Event.NETWORK_METRICS:
+                for metric in event.network_metrics.DESCRIPTOR.fields:
+                    if event.network_metrics.HasField(metric.name):
+                        metrics[metric.name]['time'].append(event.timestamp -
+                                                            first_time_stamp)
+                        metrics[metric.name]['value'].append(
+                            getattr(event.network_metrics, metric.name))
+    return (metrics, decisions)
+
+
+def main():
+    parser = OptionParser()
+    parser.add_option("-f",
+                      "--dump_file",
+                      dest="dump_file_to_parse",
+                      help="dump file to parse")
+    parser.add_option('-m',
+                      '--metric_plot',
+                      default=[],
+                      type=str,
+                      help='metric key (name of the metric) to plot',
+                      dest='metric_keys',
+                      action='append')
+
+    parser.add_option('-d',
+                      '--decision_plot',
+                      default=[],
+                      type=str,
+                      help='decision key (name of the decision) to plot',
+                      dest='decision_keys',
+                      action='append')
+
+    options = parser.parse_args()[0]
+    if options.dump_file_to_parse is None:
+        print "No dump file to parse is set.\n"
+        parser.print_help()
+        exit()
+    (metrics, decisions) = ParseAnaDump(options.dump_file_to_parse)
+    metric_keys = options.metric_keys
+    decision_keys = options.decision_keys
+    plot_count = len(metric_keys) + len(decision_keys)
+    if plot_count == 0:
+        print "You have to set at least one metric or decision to plot.\n"
+        parser.print_help()
+        exit()
+    plots = []
+    if plot_count == 1:
+        f, mp_plot = plt.subplots()
+        plots.append(mp_plot)
+    else:
+        f, mp_plots = plt.subplots(plot_count, sharex=True)
+        plots.extend(mp_plots.tolist())
+
+    for key in metric_keys:
+        plot = plots.pop()
+        plot.grid(True)
+        plot.set_title(key + " (metric)")
+        plot.plot(metrics[key]['time'], metrics[key]['value'])
+    for key in decision_keys:
+        plot = plots.pop()
+        plot.grid(True)
+        plot.set_title(key + " (decision)")
+        plot.plot(decisions[key]['time'], decisions[key]['value'])
+    f.subplots_adjust(hspace=0.3)
+    plt.show()
+
+
+if __name__ == "__main__":
+    main()
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h
new file mode 100644
index 0000000000..0375386e39
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class ThresholdCurve {
+ public:
+  struct Point {
+    constexpr Point(float x, float y) : x(x), y(y) {}
+    float x;
+    float y;
+  };
+
+  // ThresholdCurve defines a curve. The curve is characterized by the two
+  // conjunction points: A and B. The curve segments the metric space into
+  // three domains - above the curve, on it and below it.
+  //
+  // y-axis ^  |
+  //        | A|
+  //        |   \    A: (a.x, a.y)
+  //        |    \   B: (b.x, b.y)
+  //        |    B\________
+  //        |---------------> bandwidth
+  //
+  // If either a.x == b.x or a.y == b.y, the curve can be defined
+  // by a single point. (We merge the two points into one - either the lower or
+  // the leftmost one - for easier treatment.)
+  //
+  // y-axis ^  |
+  //        |  |
+  //        |  |
+  //        |  |
+  //        | P|__________
+  //        |---------------> bandwidth
+  ThresholdCurve(const Point& left, const Point& right)
+      : a(GetPoint(left, right, true)),
+        b(GetPoint(left, right, false)),
+        slope(b.x - a.x == 0.0f ? 0.0f : (b.y - a.y) / (b.x - a.x)),
+        offset(a.y - slope * a.x) {
+    // TODO(eladalon): We might want to introduce some numerical validations.
+  }
+
+  ThresholdCurve(float a_x, float a_y, float b_x, float b_y)
+      : ThresholdCurve(Point{a_x, a_y}, Point{b_x, b_y}) {}
+
+  // Checks if a point is strictly below the curve.
+  bool IsBelowCurve(const Point& p) const {
+    if (p.x < a.x) {
+      return true;
+    } else if (p.x == a.x) {
+      // In principle, we could merge this into the next else, but to avoid
+      // numerical errors, we treat it separately.
+      return p.y < a.y;
+    } else if (a.x < p.x && p.x < b.x) {
+      return p.y < offset + slope * p.x;
+    } else {  // if (b.x <= p.x)
+      return p.y < b.y;
+    }
+  }
+
+  // Checks if a point is strictly above the curve.
+  bool IsAboveCurve(const Point& p) const {
+    if (p.x <= a.x) {
+      return false;
+    } else if (a.x < p.x && p.x < b.x) {
+      return p.y > offset + slope * p.x;
+    } else {  // if (b.x <= p.x)
+      return p.y > b.y;
+    }
+  }
+
+  bool operator<=(const ThresholdCurve& rhs) const {
+    // This curve is <= the rhs curve if no point from this curve is
+    // above a corresponding point from the rhs curve.
+    return !IsBelowCurve(rhs.a) && !IsBelowCurve(rhs.b) &&
+           !rhs.IsAboveCurve(a) && !rhs.IsAboveCurve(b);
+  }
+
+ private:
+  static const Point& GetPoint(const Point& left,
+                               const Point& right,
+                               bool is_for_left) {
+    RTC_DCHECK_LE(left.x, right.x);
+    RTC_DCHECK_GE(left.y, right.y);
+
+    // Same X-value or Y-value triggers merging both points to the
+    // lower and/or left of the two points, respectively.
+    if (left.x == right.x) {
+      return right;
+    } else if (left.y == right.y) {
+      return left;
+    }
+
+    // If unmerged, boolean flag determines which of the points is desired.
+    return is_for_left ? left : right;
+  }
+
+  const Point a;
+  const Point b;
+  const float slope;
+  const float offset;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc
new file mode 100644
index 0000000000..dc3aec0b18
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc
@@ -0,0 +1,632 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+// A threshold curve divides 2D space into three domains - below, on and above
+// the threshold curve.
+// The curve is defined by two points. Those points, P1 and P2, are ordered so
+// that (P1.x <= P2.x && P1.y >= P2.y).
+// The part of the curve which is between the two points is hereon referred
+// to as the "segment".
+// A "ray" extends from P1 directly upwards into infinity; that's the "vertical
+// ray". Likewise, a "horizontal ray" extends from P2 directly rightwards.
+//
+//  ^   |                         //
+//  |   | vertical ray            //
+//  |   |                         //
+//  |   |                         //
+//  | P1|                         //
+//  |    \                        //
+//  |     \ segment               //
+//  |      \                      //
+//  |       \    horizontal ray   //
+//  |     P2 ------------------   //
+//  *---------------------------> //
+
+namespace webrtc {
+
+namespace {
+enum RelativePosition { kBelow, kOn, kAbove };
+
+void CheckRelativePosition(const ThresholdCurve& curve,
+                           ThresholdCurve::Point point,
+                           RelativePosition pos) {
+  RTC_CHECK(pos == kBelow || pos == kOn || pos == kAbove);
+
+  EXPECT_EQ(pos == kBelow, curve.IsBelowCurve(point));
+  EXPECT_EQ(pos == kAbove, curve.IsAboveCurve(point));
+}
+}  // namespace
+
+// Test that the curve correctly reports the below/above position of points,
+// when the curve is a "normal" one - P1 and P2 are different in both their
+// X and Y values.
+TEST(ThresholdCurveTest, PointPositionToCommonCurve) {
+  // The points (P1-P2) define the curve.           //
+  // All other points are above/below/on the curve. //
+  //                                                //
+  //  ^                                             //
+  //  |     |                                       //
+  //  |  A  F    J  R   V                           //
+  //  |     |                                       //
+  //  |  B  P1   K  S   W                           //
+  //  |      \                                      //
+  //  |       \                                     //
+  //  |        \ L                                  //
+  //  |         \                                   //
+  //  |  C  G    M  T   X                           //
+  //  |           \                                 //
+  //  |          N \                                //
+  //  |             \                               //
+  //  |  D  H    O  P2--Y----------------           //
+  //  |  E  I    Q  U   Z                           //
+  //  *---------------------------------->          //
+  constexpr ThresholdCurve::Point p1{1000, 2000};
+  constexpr ThresholdCurve::Point p2{2000, 1000};
+
+  RTC_CHECK_GT((p1.x + p2.x) / 2, p1.x);
+  RTC_CHECK_LT((p1.x + p2.x) / 2, p2.x);
+  RTC_CHECK_LT((p1.y + p2.y) / 2, p1.y);
+  RTC_CHECK_GT((p1.y + p2.y) / 2, p2.y);
+
+  const ThresholdCurve curve(p1, p2);
+
+  {
+    // All cases where the point lies to the left of P1.
+    constexpr float x = p1.x - 1;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kBelow);           // A
+    CheckRelativePosition(curve, {x, p1.y + 0}, kBelow);           // B
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kBelow);  // C
+    CheckRelativePosition(curve, {x, p2.y + 0}, kBelow);           // D
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);           // E
+  }
+
+  {
+    // All cases where the point has the same x-value as P1.
+    constexpr float x = p1.x;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kOn);              // F
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);              // P1
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kBelow);  // G
+    CheckRelativePosition(curve, {x, p2.y + 0}, kBelow);           // H
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);           // I
+  }
+
+  {
+    // To make sure we're really covering all of the cases, make sure that P1
+    // and P2 were chosen so that L would really be below K, and O would really
+    // be below N. (This would not hold if the Y values are too close together.)
+    RTC_CHECK_LT(((p1.y + p2.y) / 2) + 1, p1.y);
+    RTC_CHECK_LT(p2.y, ((p1.y + p2.y) / 2) - 1);
+
+    // All cases where the point's x-value is between P1 and P2.
+    constexpr float x = (p1.x + p2.x) / 2;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);                 // J
+    CheckRelativePosition(curve, {x, p1.y + 0}, kAbove);                 // K
+    CheckRelativePosition(curve, {x, ((p1.y + p2.y) / 2) + 1}, kAbove);  // L
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kOn);           // M
+    CheckRelativePosition(curve, {x, ((p1.y + p2.y) / 2) - 1}, kBelow);  // N
+    CheckRelativePosition(curve, {x, p2.y + 0}, kBelow);                 // O
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);                 // Q
+  }
+
+  {
+    // All cases where the point has the same x-value as P2.
+    constexpr float x = p2.x;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);           // R
+    CheckRelativePosition(curve, {x, p1.y + 0}, kAbove);           // S
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kAbove);  // T
+    CheckRelativePosition(curve, {x, p2.y + 0}, kOn);              // P2
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);           // U
+  }
+
+  {
+    // All cases where the point lies to the right of P2.
+    constexpr float x = p2.x + 1;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);           // V
+    CheckRelativePosition(curve, {x, p1.y + 0}, kAbove);           // W
+    CheckRelativePosition(curve, {x, (p1.y + p2.y) / 2}, kAbove);  // X
+    CheckRelativePosition(curve, {x, p2.y + 0}, kOn);              // Y
+    CheckRelativePosition(curve, {x, p2.y - 1}, kBelow);           // Z
+  }
+}
+
+// Test that the curve correctly reports the below/above position of points,
+// when the curve is defined by two points with the same Y value.
+TEST(ThresholdCurveTest, PointPositionToCurveWithHorizaontalSegment) {
+  // The points (P1-P2) define the curve.
+  // All other points are above/below/on the curve.
+  //
+  //  ^
+  //  |    |
+  //  |    |
+  //  | A  D   F  I  K
+  //  |    |
+  //  |    |
+  //  | B  P1--G--P2-L--
+  //  | C  E   H  J  M
+  //  *------------------>
+
+  constexpr ThresholdCurve::Point p1{100, 200};
+  constexpr ThresholdCurve::Point p2{p1.x + 1, p1.y};
+
+  RTC_CHECK_GT((p1.x + p2.x) / 2, p1.x);
+  RTC_CHECK_LT((p1.x + p2.x) / 2, p2.x);
+
+  const ThresholdCurve curve(p1, p2);
+
+  {
+    // All cases where the point lies to the left of P1.
+    constexpr float x = p1.x - 1;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kBelow);  // A
+    CheckRelativePosition(curve, {x, p1.y + 0}, kBelow);  // B
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // C
+  }
+
+  {
+    // All cases where the point has the same x-value as P1.
+    constexpr float x = p1.x;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kOn);     // D
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);     // P1
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // E
+  }
+
+  {
+    // All cases where the point's x-value is between P1 and P2.
+    constexpr float x = (p1.x + p2.x) / 2;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);  // F
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);     // G
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // H
+  }
+
+  {
+    // All cases where the point has the same x-value as P2.
+    constexpr float x = p2.x;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);  // I
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);     // P2
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // J
+  }
+
+  {
+    // All cases where the point lies to the right of P2.
+    constexpr float x = p2.x + 1;
+    CheckRelativePosition(curve, {x, p1.y + 1}, kAbove);  // K
+    CheckRelativePosition(curve, {x, p1.y + 0}, kOn);     // L
+    CheckRelativePosition(curve, {x, p1.y - 1}, kBelow);  // M
+  }
+}
+
+// Test that the curve correctly reports the below/above position of points,
+// when the curve is defined by two points with the same X value.
+TEST(ThresholdCurveTest, PointPositionToCurveWithVerticalSegment) {
+  // The points (P1-P2) define the curve.
+  // All other points are above/below/on the curve.
+  //
+  //  ^
+  //  |    |
+  //  | A  B   C
+  //  |    |
+  //  | D  P1  E
+  //  |    |
+  //  | F  G   H
+  //  |    |
+  //  | I  P2--J------
+  //  | K  L   M
+  //  *------------------>
+
+  constexpr ThresholdCurve::Point p1{100, 200};
+  constexpr ThresholdCurve::Point p2{p1.x, p1.y - 1};
+
+  constexpr float left = p1.x - 1;
+  constexpr float on = p1.x;
+  constexpr float right = p1.x + 1;
+
+  RTC_CHECK_LT((p1.y + p2.y) / 2, p1.y);
+  RTC_CHECK_GT((p1.y + p2.y) / 2, p2.y);
+
+  const ThresholdCurve curve(p1, p2);
+
+  {
+    // All cases where the point lies above P1.
+    constexpr float y = p1.y + 1;
+    CheckRelativePosition(curve, {left, y}, kBelow);   // A
+    CheckRelativePosition(curve, {on, y}, kOn);        // B
+    CheckRelativePosition(curve, {right, y}, kAbove);  // C
+  }
+
+  {
+    // All cases where the point has the same y-value as P1.
+    constexpr float y = p1.y;
+    CheckRelativePosition(curve, {left, y}, kBelow);   // D
+    CheckRelativePosition(curve, {on, y}, kOn);        // P1
+    CheckRelativePosition(curve, {right, y}, kAbove);  // E
+  }
+
+  {
+    // All cases where the point's y-value is between P1 and P2.
+    constexpr float y = (p1.y + p2.y) / 2;
+    CheckRelativePosition(curve, {left, y}, kBelow);   // F
+    CheckRelativePosition(curve, {on, y}, kOn);        // G
+    CheckRelativePosition(curve, {right, y}, kAbove);  // H
+  }
+
+  {
+    // All cases where the point has the same y-value as P2.
+    constexpr float y = p2.y;
+    CheckRelativePosition(curve, {left, y}, kBelow);  // I
+    CheckRelativePosition(curve, {on, y}, kOn);       // P2
+    CheckRelativePosition(curve, {right, y}, kOn);    // J
+  }
+
+  {
+    // All cases where the point lies below P2.
+    constexpr float y = p2.y - 1;
+    CheckRelativePosition(curve, {left, y}, kBelow);   // K
+    CheckRelativePosition(curve, {on, y}, kBelow);     // L
+    CheckRelativePosition(curve, {right, y}, kBelow);  // M
+  }
+}
+
+// Test that the curve correctly reports the below/above position of points,
+// when the curve is defined by two points which are identical.
+TEST(ThresholdCurveTest, PointPositionCurveWithNullSegment) {
+  // The points (P1-P2) define the curve.
+  // All other points are above/below/on the curve.
+  //
+  //  ^
+  //  |    |
+  //  | A  D   F
+  //  |    |
+  //  | B  P---G------
+  //  | C  E   H
+  //  *------------------>
+
+  constexpr ThresholdCurve::Point p{100, 200};
+
+  const ThresholdCurve curve(p, p);
+
+  {
+    // All cases where the point lies to the left of P.
+    constexpr float x = p.x - 1;
+    CheckRelativePosition(curve, {x, p.y + 1}, kBelow);  // A
+    CheckRelativePosition(curve, {x, p.y + 0}, kBelow);  // B
+    CheckRelativePosition(curve, {x, p.y - 1}, kBelow);  // C
+  }
+
+  {
+    // All cases where the point has the same x-value as P.
+    constexpr float x = p.x + 0;
+    CheckRelativePosition(curve, {x, p.y + 1}, kOn);     // D
+    CheckRelativePosition(curve, {x, p.y + 0}, kOn);     // P
+    CheckRelativePosition(curve, {x, p.y - 1}, kBelow);  // E
+  }
+
+  {
+    // All cases where the point lies to the right of P.
+    constexpr float x = p.x + 1;
+    CheckRelativePosition(curve, {x, p.y + 1}, kAbove);  // F
+    CheckRelativePosition(curve, {x, p.y + 0}, kOn);     // G
+    CheckRelativePosition(curve, {x, p.y - 1}, kBelow);  // H
+  }
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the two curves have the same projection on the X-axis.
+TEST(ThresholdCurveTest, TwoCurvesSegmentHasSameProjectionAxisX) {
+  //  ^                        //
+  //  | C1 + C2                //
+  //  |  |                     //
+  //  |  |\                    //
+  //  |  | \                   //
+  //  |   \ \                  //
+  //  |    \ \                 //
+  //  |     \ \                //
+  //  |      \ -------- C2     //
+  //  |       --------- C1     //
+  //  *--------------------->  //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  // Same x-values, but higher on Y. (Can be parallel, but doesn't have to be.)
+  constexpr ThresholdCurve::Point c2_left{c1_left.x, c1_left.y + 20};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x, c1_right.y + 10};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the higher curve's projection on the X-axis is a strict subset of the
+// lower curve's projection on the X-axis (on both ends).
+TEST(ThresholdCurveTest, TwoCurvesSegmentOfHigherSubsetProjectionAxisX) {
+  //  ^                       //
+  //  | C1    C2              //
+  //  |  |    |               //
+  //  |  |    |               //
+  //  |   \   |               //
+  //  |    \  |               //
+  //  |     \ \               //
+  //  |      \ \              //
+  //  |       \ --------- C2  //
+  //  |        \              //
+  //  |         \             //
+  //  |          ---------C1  //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{6, 11};
+  constexpr ThresholdCurve::Point c2_right{9, 7};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the higher curve's right point is above lower curve's horizontal ray (meaning
+// the higher curve's projection on the X-axis extends further right than
+// the lower curve's).
+TEST(ThresholdCurveTest,
+     TwoCurvesRightPointOfHigherCurveAboveHorizontalRayOfLower) {
+  //  ^                        //
+  //  | C1 + C2                //
+  //  |  |                     //
+  //  |  |\                    //
+  //  |  | \                   //
+  //  |  |  \                  //
+  //  |  |   \                 //
+  //  |  |    \                //
+  //  |   \    \               //
+  //  |    \    \              //
+  //  |     \    \             //
+  //  |      \    ----- C2     //
+  //  |       --------- C1     //
+  //  *--------------------->  //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{c1_left.x, c1_left.y + 1};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x + 1, c1_right.y + 1};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the higher curve's points are on the lower curve's rays (left point on the
+// veritcal ray, right point on the horizontal ray).
+TEST(ThresholdCurveTest, TwoCurvesPointsOfHigherOnRaysOfLower) {
+  //  ^
+  //  | C1 + C2               //
+  //  |  |                    //
+  //  |  |\                   //
+  //  |  | \                  //
+  //  |   \ \                 //
+  //  |    \ \                //
+  //  |     \ \               //
+  //  |      \ \              //
+  //  |       ----- C1 + C2   //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  // Same x-values, but one of the points is higher on Y (the other isn't).
+  constexpr ThresholdCurve::Point c2_left{c1_left.x, c1_left.y + 2};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x + 3, c1_right.y};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the second curve's segment intersects the first curve's vertical ray.
+TEST(ThresholdCurveTest, SecondCurveCrossesVerticalRayOfFirstCurve) {
+  //  ^                       //
+  //  | C2 C1                 //
+  //  |  | |                  //
+  //  |   \|                  //
+  //  |    |                  //
+  //  |    |\                 //
+  //  |    | \                //
+  //  |     \ \               //
+  //  |      \ \              //
+  //  |       \ \             //
+  //  |        \ ------- C2   //
+  //  |         -------- C1   //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{c1_left.x - 1, c1_left.y + 1};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x, c1_right.y + 1};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the second curve's segment intersects the first curve's horizontal ray.
+TEST(ThresholdCurveTest, SecondCurveCrossesHorizontalRayOfFirstCurve) {
+  //  ^                      //
+  //  | C1 +  C2             //
+  //  |  |                   //
+  //  |  |\                  //
+  //  |  \ \                 //
+  //  |   \ \                //
+  //  |    \ \               //
+  //  |     \ \              //
+  //  |      ----------- C1  //
+  //  |         \            //
+  //  |          ------- C2  //
+  //  *--------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{c1_left.x, c1_left.y + 1};
+  constexpr ThresholdCurve::Point c2_right{c1_right.x + 2, c1_right.y - 1};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// the second curve's segment intersects the first curve's segment.
+TEST(ThresholdCurveTest, TwoCurvesWithCrossingSegments) {
+  //  ^                           //
+  //  | C2 C1                     //
+  //  | |  |                      //
+  //  | |  |                      //
+  //  | |  \                      //
+  //  | |   \                     //
+  //  |  -_  \                    //
+  //  |    -_ \                   //
+  //  |      -_\                  //
+  //  |        -_                 //
+  //  |          \-_              //
+  //  |           \ ---------- C2 //
+  //  |            ----------- C1 //
+  //  |                           //
+  //  |                           //
+  //  *-------------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_right);
+
+  constexpr ThresholdCurve::Point c2_left{4, 9};
+  constexpr ThresholdCurve::Point c2_right{10, 6};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  // The test is structured so that the two curves intersect at (8, 7).
+  RTC_CHECK(!c1_curve.IsAboveCurve({8, 7}));
+  RTC_CHECK(!c1_curve.IsBelowCurve({8, 7}));
+  RTC_CHECK(!c2_curve.IsAboveCurve({8, 7}));
+  RTC_CHECK(!c2_curve.IsBelowCurve({8, 7}));
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_FALSE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// both curves are identical.
+TEST(ThresholdCurveTest, IdenticalCurves) {
+  //  ^                       //
+  //  |  C1 + C2              //
+  //  |  |                    //
+  //  |  |                    //
+  //  |   \                   //
+  //  |    \                  //
+  //  |     \                 //
+  //  |      ------- C1 + C2  //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point left{5, 10};
+  constexpr ThresholdCurve::Point right{10, 5};
+
+  const ThresholdCurve c1_curve(left, right);
+  const ThresholdCurve c2_curve(left, right);
+
+  EXPECT_TRUE(c1_curve <= c2_curve);
+  EXPECT_TRUE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// they are "nearly identical" - the first curve's segment is contained within
+// the second curve's segment, but the second curve's segment extends further
+// to the left (which also produces separate vertical rays for the curves).
+TEST(ThresholdCurveTest, NearlyIdenticalCurvesSecondContinuesOnOtherLeftSide) {
+  //  ^                       //
+  //  | C2 C1                 //
+  //  |  | |                  //
+  //  |  | |                  //
+  //  |   \|                  //
+  //  |    |                  //
+  //  |     \                 //
+  //  |      \                //
+  //  |       \               //
+  //  |        ----- C1 + C2  //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_left);
+
+  constexpr ThresholdCurve::Point c2_left{c1_left.x - 1, c1_left.y + 1};
+  constexpr ThresholdCurve::Point c2_right = c1_right;
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_TRUE(c2_curve <= c1_curve);
+}
+
+// Test that the relative position of two curves is computed correctly when
+// they are "nearly identical" - the first curve's segment is contained within
+// the second curve's segment, but the second curve's segment extends further
+// to the right (which also produces separate horizontal rays for the curves).
+TEST(ThresholdCurveTest, NearlyIdenticalCurvesSecondContinuesOnOtherRightSide) {
+  //  ^                       //
+  //  | C1 + C2               //
+  //  |  |                    //
+  //  |  |                    //
+  //  |   \                   //
+  //  |    \                  //
+  //  |     \                 //
+  //  |      \----------- C1  //
+  //  |       \               //
+  //  |        ---------- C2  //
+  //  *---------------------> //
+
+  constexpr ThresholdCurve::Point c1_left{5, 10};
+  constexpr ThresholdCurve::Point c1_right{10, 5};
+  const ThresholdCurve c1_curve(c1_left, c1_left);
+
+  constexpr ThresholdCurve::Point c2_left = c1_left;
+  constexpr ThresholdCurve::Point c2_right{c1_right.x + 1, c1_right.y - 1};
+  const ThresholdCurve c2_curve(c2_left, c2_right);
+
+  EXPECT_FALSE(c1_curve <= c2_curve);
+  EXPECT_TRUE(c2_curve <= c1_curve);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// The higher-left point must be given as the first point, and the lower-right
+// point must be given as the second.
+// This necessarily produces a non-positive slope.
+TEST(ThresholdCurveDeathTest, WrongOrderPoints) {
+  std::unique_ptr<ThresholdCurve> curve;
+  constexpr ThresholdCurve::Point left{5, 10};
+  constexpr ThresholdCurve::Point right{10, 5};
+  EXPECT_DEATH(curve.reset(new ThresholdCurve(right, left)), "");
+}
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_config_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_config_gn/moz.build
new file mode 100644
index 0000000000..37cf81ad9f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_config_gn/moz.build
@@ -0,0 +1,217 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_network_adaptor_config_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_gn/moz.build
new file mode 100644
index 0000000000..1d0905ebc1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor_gn/moz.build
@@ -0,0 +1,242 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc",
+    "/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_v2.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_network_adaptor_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/audio_decoder.h b/third_party/libwebrtc/modules/audio_coding/codecs/audio_decoder.h
new file mode 100644
index 0000000000..b7b15cdd6e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/audio_decoder.h instead!
+// TODO(kwiberg): Remove it.
+
+#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+#define MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+
+#include "api/audio_codecs/audio_decoder.h"
+
+#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/audio_encoder.h b/third_party/libwebrtc/modules/audio_coding/codecs/audio_encoder.h
new file mode 100644
index 0000000000..010ae6705f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/audio_encoder.h instead!
+// TODO(ossu): Remove it.
+
+#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+#define MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+
+#include "api/audio_codecs/audio_encoder.h"
+
+#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
new file mode 100644
index 0000000000..4a2b261a59
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(AudioDecoderFactoryTest, CreateUnknownDecoder) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreatePcmu) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // PCMu supports 8 kHz, and any number of channels.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreatePcma) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // PCMa supports 8 kHz, and any number of channels.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreateIlbc) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // iLBC supports 8 kHz, 1 channel.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0), absl::nullopt));
+#ifdef WEBRTC_CODEC_ILBC
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1), absl::nullopt));
+#endif
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreateL16) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // L16 supports any clock rate and any number of channels up to 24.
+  const int clockrates[] = {8000, 16000, 32000, 48000};
+  const int num_channels[] = {1, 2, 3, 24};
+  for (int clockrate : clockrates) {
+    EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0),
+                                       absl::nullopt));
+    for (int channels : num_channels) {
+      EXPECT_TRUE(adf->MakeAudioDecoder(
+          SdpAudioFormat("l16", clockrate, channels), absl::nullopt));
+    }
+  }
+}
+
+// Tests that using more channels than the maximum does not work
+TEST(AudioDecoderFactoryTest, MaxNrOfChannels) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  std::vector<std::string> codecs = {
+#ifdef WEBRTC_CODEC_OPUS
+    "opus",
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+    "ilbc",
+#endif
+    "pcmu",
+    "pcma",
+    "l16",
+    "G722",
+    "G711",
+  };
+
+  for (auto codec : codecs) {
+    EXPECT_FALSE(adf->MakeAudioDecoder(
+        SdpAudioFormat(codec, 32000, AudioDecoder::kMaxNumberOfChannels + 1),
+        absl::nullopt));
+  }
+}
+
+TEST(AudioDecoderFactoryTest, CreateG722) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // g722 supports 8 kHz, 1-2 channels.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1), absl::nullopt));
+
+  // g722 actually uses a 16 kHz sample rate instead of the nominal 8 kHz.
+  std::unique_ptr<AudioDecoder> dec =
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt);
+  EXPECT_EQ(16000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTest, CreateOpus) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // Opus supports 48 kHz, 2 channels, and wants a "stereo" parameter whose
+  // value is either "0" or "1".
+  for (int hz : {8000, 16000, 32000, 48000}) {
+    for (int channels : {0, 1, 2, 3}) {
+      for (std::string stereo : {"XX", "0", "1", "2"}) {
+        SdpAudioFormat::Parameters params;
+        if (stereo != "XX") {
+          params["stereo"] = stereo;
+        }
+        const bool good = (hz == 48000 && channels == 2 &&
+                           (stereo == "XX" || stereo == "0" || stereo == "1"));
+        EXPECT_EQ(good,
+                  static_cast<bool>(adf->MakeAudioDecoder(
+                      SdpAudioFormat("opus", hz, channels, std::move(params)),
+                      absl::nullopt)));
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
new file mode 100644
index 0000000000..26ae1eda8a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
@@ -0,0 +1,178 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+
+#include <limits>
+#include <memory>
+#include <vector>
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class AudioEncoderFactoryTest
+    : public ::testing::TestWithParam<rtc::scoped_refptr<AudioEncoderFactory>> {
+};
+
+TEST_P(AudioEncoderFactoryTest, SupportsAtLeastOneFormat) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  EXPECT_FALSE(supported_encoders.empty());
+}
+
+TEST_P(AudioEncoderFactoryTest, CanQueryAllSupportedFormats) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto info = factory->QueryAudioEncoder(spec.format);
+    EXPECT_TRUE(info);
+  }
+}
+
+TEST_P(AudioEncoderFactoryTest, CanConstructAllSupportedEncoders) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto info = factory->QueryAudioEncoder(spec.format);
+    auto encoder = factory->MakeAudioEncoder(127, spec.format, absl::nullopt);
+    EXPECT_TRUE(encoder);
+    EXPECT_EQ(encoder->SampleRateHz(), info->sample_rate_hz);
+    EXPECT_EQ(encoder->NumChannels(), info->num_channels);
+    EXPECT_EQ(encoder->RtpTimestampRateHz(), spec.format.clockrate_hz);
+  }
+}
+
+TEST_P(AudioEncoderFactoryTest, CanRunAllSupportedEncoders) {
+  constexpr int kTestPayloadType = 127;
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto encoder =
+        factory->MakeAudioEncoder(kTestPayloadType, spec.format, absl::nullopt);
+    EXPECT_TRUE(encoder);
+    encoder->Reset();
+    const int num_samples = rtc::checked_cast<int>(
+        encoder->SampleRateHz() * encoder->NumChannels() / 100);
+    rtc::Buffer out;
+    rtc::BufferT<int16_t> audio;
+    audio.SetData(num_samples, [](rtc::ArrayView<int16_t> audio) {
+      for (size_t i = 0; i != audio.size(); ++i) {
+        // Just put some numbers in there, ensure they're within range.
+        audio[i] =
+            static_cast<int16_t>(i & std::numeric_limits<int16_t>::max());
+      }
+      return audio.size();
+    });
+    // This is here to stop the test going forever with a broken encoder.
+    constexpr int kMaxEncodeCalls = 100;
+    int blocks = 0;
+    for (; blocks < kMaxEncodeCalls; ++blocks) {
+      AudioEncoder::EncodedInfo info = encoder->Encode(
+          blocks * encoder->RtpTimestampRateHz() / 100, audio, &out);
+      EXPECT_EQ(info.encoded_bytes, out.size());
+      if (info.encoded_bytes > 0) {
+        EXPECT_EQ(0u, info.encoded_timestamp);
+        EXPECT_EQ(kTestPayloadType, info.payload_type);
+        break;
+      }
+    }
+    ASSERT_LT(blocks, kMaxEncodeCalls);
+    const unsigned int next_timestamp =
+        blocks * encoder->RtpTimestampRateHz() / 100;
+    out.Clear();
+    for (; blocks < kMaxEncodeCalls; ++blocks) {
+      AudioEncoder::EncodedInfo info = encoder->Encode(
+          blocks * encoder->RtpTimestampRateHz() / 100, audio, &out);
+      EXPECT_EQ(info.encoded_bytes, out.size());
+      if (info.encoded_bytes > 0) {
+        EXPECT_EQ(next_timestamp, info.encoded_timestamp);
+        EXPECT_EQ(kTestPayloadType, info.payload_type);
+        break;
+      }
+    }
+    ASSERT_LT(blocks, kMaxEncodeCalls);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(BuiltinAudioEncoderFactoryTest,
+                         AudioEncoderFactoryTest,
+                         ::testing::Values(CreateBuiltinAudioEncoderFactory()));
+
+TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) {
+  using ::testing::ElementsAreArray;
+  // Check that we claim to support the formats we expect from build flags, and
+  // we've ordered them correctly.
+  auto factory = CreateBuiltinAudioEncoderFactory();
+  auto specs = factory->GetSupportedEncoders();
+
+  const std::vector<SdpAudioFormat> supported_formats = [&specs] {
+    std::vector<SdpAudioFormat> formats;
+    formats.reserve(specs.size());
+    for (const auto& spec : specs) {
+      formats.push_back(spec.format);
+    }
+    return formats;
+  }();
+
+  const std::vector<SdpAudioFormat> expected_formats = {
+#ifdef WEBRTC_CODEC_OPUS
+    {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+    {"isac", 16000, 1},
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+    {"isac", 32000, 1},
+#endif
+    {"G722", 8000, 1},
+#ifdef WEBRTC_CODEC_ILBC
+    {"ilbc", 8000, 1},
+#endif
+    {"pcmu", 8000, 1},
+    {"pcma", 8000, 1}
+  };
+
+  ASSERT_THAT(supported_formats, ElementsAreArray(expected_formats));
+}
+
+// Tests that using more channels than the maximum does not work.
+TEST(BuiltinAudioEncoderFactoryTest, MaxNrOfChannels) {
+  rtc::scoped_refptr<AudioEncoderFactory> aef =
+      CreateBuiltinAudioEncoderFactory();
+  std::vector<std::string> codecs = {
+#ifdef WEBRTC_CODEC_OPUS
+    "opus",
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+    "isac",
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+    "ilbc",
+#endif
+    "pcmu",
+    "pcma",
+    "l16",
+    "G722",
+    "G711",
+  };
+
+  for (auto codec : codecs) {
+    EXPECT_FALSE(aef->MakeAudioEncoder(
+        /*payload_type=*/111,
+        /*format=*/
+        SdpAudioFormat(codec, 32000, AudioEncoder::kMaxNumberOfChannels + 1),
+        /*codec_pair_id=*/absl::nullopt));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
new file mode 100644
index 0000000000..7546ac178f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -0,0 +1,322 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kMaxFrameSizeMs = 60;
+
+class AudioEncoderCng final : public AudioEncoder {
+ public:
+  explicit AudioEncoderCng(AudioEncoderCngConfig&& config);
+  ~AudioEncoderCng() override;
+
+  // Not copyable or moveable.
+  AudioEncoderCng(const AudioEncoderCng&) = delete;
+  AudioEncoderCng(AudioEncoderCng&&) = delete;
+  AudioEncoderCng& operator=(const AudioEncoderCng&) = delete;
+  AudioEncoderCng& operator=(AudioEncoderCng&&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+  void Reset() override;
+  bool SetFec(bool enable) override;
+  bool SetDtx(bool enable) override;
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
+      override;
+  void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) override;
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms) override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ private:
+  EncodedInfo EncodePassive(size_t frames_to_encode, rtc::Buffer* encoded);
+  EncodedInfo EncodeActive(size_t frames_to_encode, rtc::Buffer* encoded);
+  size_t SamplesPer10msFrame() const;
+
+  std::unique_ptr<AudioEncoder> speech_encoder_;
+  const int cng_payload_type_;
+  const int num_cng_coefficients_;
+  const int sid_frame_interval_ms_;
+  std::vector<int16_t> speech_buffer_;
+  std::vector<uint32_t> rtp_timestamps_;
+  bool last_frame_active_;
+  std::unique_ptr<Vad> vad_;
+  std::unique_ptr<ComfortNoiseEncoder> cng_encoder_;
+};
+
+AudioEncoderCng::AudioEncoderCng(AudioEncoderCngConfig&& config)
+    : speech_encoder_((static_cast<void>([&] {
+                         RTC_CHECK(config.IsOk()) << "Invalid configuration.";
+                       }()),
+                       std::move(config.speech_encoder))),
+      cng_payload_type_(config.payload_type),
+      num_cng_coefficients_(config.num_cng_coefficients),
+      sid_frame_interval_ms_(config.sid_frame_interval_ms),
+      last_frame_active_(true),
+      vad_(config.vad ? std::unique_ptr<Vad>(config.vad)
+                      : CreateVad(config.vad_mode)),
+      cng_encoder_(new ComfortNoiseEncoder(SampleRateHz(),
+                                           sid_frame_interval_ms_,
+                                           num_cng_coefficients_)) {}
+
+AudioEncoderCng::~AudioEncoderCng() = default;
+
+int AudioEncoderCng::SampleRateHz() const {
+  return speech_encoder_->SampleRateHz();
+}
+
+size_t AudioEncoderCng::NumChannels() const {
+  return 1;
+}
+
+int AudioEncoderCng::RtpTimestampRateHz() const {
+  return speech_encoder_->RtpTimestampRateHz();
+}
+
+size_t AudioEncoderCng::Num10MsFramesInNextPacket() const {
+  return speech_encoder_->Num10MsFramesInNextPacket();
+}
+
+size_t AudioEncoderCng::Max10MsFramesInAPacket() const {
+  return speech_encoder_->Max10MsFramesInAPacket();
+}
+
+int AudioEncoderCng::GetTargetBitrate() const {
+  return speech_encoder_->GetTargetBitrate();
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  RTC_CHECK_EQ(speech_buffer_.size(),
+               rtp_timestamps_.size() * samples_per_10ms_frame);
+  rtp_timestamps_.push_back(rtp_timestamp);
+  RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
+  speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
+  const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
+  if (rtp_timestamps_.size() < frames_to_encode) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_LE(frames_to_encode * 10, kMaxFrameSizeMs)
+      << "Frame size cannot be larger than " << kMaxFrameSizeMs
+      << " ms when using VAD/CNG.";
+
+  // Group several 10 ms blocks per VAD call. Call VAD once or twice using the
+  // following split sizes:
+  // 10 ms = 10 + 0 ms; 20 ms = 20 + 0 ms; 30 ms = 30 + 0 ms;
+  // 40 ms = 20 + 20 ms; 50 ms = 30 + 20 ms; 60 ms = 30 + 30 ms.
+  size_t blocks_in_first_vad_call =
+      (frames_to_encode > 3 ? 3 : frames_to_encode);
+  if (frames_to_encode == 4)
+    blocks_in_first_vad_call = 2;
+  RTC_CHECK_GE(frames_to_encode, blocks_in_first_vad_call);
+  const size_t blocks_in_second_vad_call =
+      frames_to_encode - blocks_in_first_vad_call;
+
+  // Check if all of the buffer is passive speech. Start with checking the first
+  // block.
+  Vad::Activity activity = vad_->VoiceActivity(
+      &speech_buffer_[0], samples_per_10ms_frame * blocks_in_first_vad_call,
+      SampleRateHz());
+  if (activity == Vad::kPassive && blocks_in_second_vad_call > 0) {
+    // Only check the second block if the first was passive.
+    activity = vad_->VoiceActivity(
+        &speech_buffer_[samples_per_10ms_frame * blocks_in_first_vad_call],
+        samples_per_10ms_frame * blocks_in_second_vad_call, SampleRateHz());
+  }
+
+  EncodedInfo info;
+  switch (activity) {
+    case Vad::kPassive: {
+      info = EncodePassive(frames_to_encode, encoded);
+      last_frame_active_ = false;
+      break;
+    }
+    case Vad::kActive: {
+      info = EncodeActive(frames_to_encode, encoded);
+      last_frame_active_ = true;
+      break;
+    }
+    default: {
+      RTC_CHECK_NOTREACHED();
+    }
+  }
+
+  speech_buffer_.erase(
+      speech_buffer_.begin(),
+      speech_buffer_.begin() + frames_to_encode * samples_per_10ms_frame);
+  rtp_timestamps_.erase(rtp_timestamps_.begin(),
+                        rtp_timestamps_.begin() + frames_to_encode);
+  return info;
+}
+
+void AudioEncoderCng::Reset() {
+  speech_encoder_->Reset();
+  speech_buffer_.clear();
+  rtp_timestamps_.clear();
+  last_frame_active_ = true;
+  vad_->Reset();
+  cng_encoder_.reset(new ComfortNoiseEncoder(
+      SampleRateHz(), sid_frame_interval_ms_, num_cng_coefficients_));
+}
+
+bool AudioEncoderCng::SetFec(bool enable) {
+  return speech_encoder_->SetFec(enable);
+}
+
+bool AudioEncoderCng::SetDtx(bool enable) {
+  return speech_encoder_->SetDtx(enable);
+}
+
+bool AudioEncoderCng::SetApplication(Application application) {
+  return speech_encoder_->SetApplication(application);
+}
+
+void AudioEncoderCng::SetMaxPlaybackRate(int frequency_hz) {
+  speech_encoder_->SetMaxPlaybackRate(frequency_hz);
+}
+
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoderCng::ReclaimContainedEncoders() {
+  return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+}
+
+void AudioEncoderCng::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  speech_encoder_->OnReceivedUplinkPacketLossFraction(
+      uplink_packet_loss_fraction);
+}
+
+void AudioEncoderCng::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {
+  speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
+                                             bwe_period_ms);
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderCng::GetFrameLengthRange() const {
+  return speech_encoder_->GetFrameLengthRange();
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
+    size_t frames_to_encode,
+    rtc::Buffer* encoded) {
+  bool force_sid = last_frame_active_;
+  bool output_produced = false;
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  AudioEncoder::EncodedInfo info;
+
+  for (size_t i = 0; i < frames_to_encode; ++i) {
+    // It's important not to pass &info.encoded_bytes directly to
+    // WebRtcCng_Encode(), since later loop iterations may return zero in
+    // that value, in which case we don't want to overwrite any value from
+    // an earlier iteration.
+    size_t encoded_bytes_tmp =
+        cng_encoder_->Encode(rtc::ArrayView<const int16_t>(
+                                 &speech_buffer_[i * samples_per_10ms_frame],
+                                 samples_per_10ms_frame),
+                             force_sid, encoded);
+
+    if (encoded_bytes_tmp > 0) {
+      RTC_CHECK(!output_produced);
+      info.encoded_bytes = encoded_bytes_tmp;
+      output_produced = true;
+      force_sid = false;
+    }
+  }
+
+  info.encoded_timestamp = rtp_timestamps_.front();
+  info.payload_type = cng_payload_type_;
+  info.send_even_if_empty = true;
+  info.speech = false;
+  return info;
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(size_t frames_to_encode,
+                                                        rtc::Buffer* encoded) {
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  AudioEncoder::EncodedInfo info;
+  for (size_t i = 0; i < frames_to_encode; ++i) {
+    info =
+        speech_encoder_->Encode(rtp_timestamps_.front(),
+                                rtc::ArrayView<const int16_t>(
+                                    &speech_buffer_[i * samples_per_10ms_frame],
+                                    samples_per_10ms_frame),
+                                encoded);
+    if (i + 1 == frames_to_encode) {
+      RTC_CHECK_GT(info.encoded_bytes, 0) << "Encoder didn't deliver data.";
+    } else {
+      RTC_CHECK_EQ(info.encoded_bytes, 0)
+          << "Encoder delivered data too early.";
+    }
+  }
+  return info;
+}
+
+size_t AudioEncoderCng::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
+}
+
+}  // namespace
+
+AudioEncoderCngConfig::AudioEncoderCngConfig() = default;
+AudioEncoderCngConfig::AudioEncoderCngConfig(AudioEncoderCngConfig&&) = default;
+AudioEncoderCngConfig::~AudioEncoderCngConfig() = default;
+
+bool AudioEncoderCngConfig::IsOk() const {
+  if (num_channels != 1)
+    return false;
+  if (!speech_encoder)
+    return false;
+  if (num_channels != speech_encoder->NumChannels())
+    return false;
+  if (sid_frame_interval_ms <
+      static_cast<int>(speech_encoder->Max10MsFramesInAPacket() * 10))
+    return false;
+  if (num_cng_coefficients > WEBRTC_CNG_MAX_LPC_ORDER ||
+      num_cng_coefficients <= 0)
+    return false;
+  return true;
+}
+
+std::unique_ptr<AudioEncoder> CreateComfortNoiseEncoder(
+    AudioEncoderCngConfig&& config) {
+  return std::make_unique<AudioEncoderCng>(std::move(config));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
new file mode 100644
index 0000000000..8a1183489f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+#define MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+
+#include <stddef.h>
+
+#include <memory>
+
+#include "api/audio_codecs/audio_encoder.h"
+#include "common_audio/vad/include/vad.h"
+
+namespace webrtc {
+
+struct AudioEncoderCngConfig {
+  // Moveable, not copyable.
+  AudioEncoderCngConfig();
+  AudioEncoderCngConfig(AudioEncoderCngConfig&&);
+  ~AudioEncoderCngConfig();
+
+  bool IsOk() const;
+
+  size_t num_channels = 1;
+  int payload_type = 13;
+  std::unique_ptr<AudioEncoder> speech_encoder;
+  Vad::Aggressiveness vad_mode = Vad::kVadNormal;
+  int sid_frame_interval_ms = 100;
+  int num_cng_coefficients = 8;
+  // The Vad pointer is mainly for testing. If a NULL pointer is passed, the
+  // AudioEncoderCng creates (and destroys) a Vad object internally. If an
+  // object is passed, the AudioEncoderCng assumes ownership of the Vad
+  // object.
+  Vad* vad = nullptr;
+};
+
+std::unique_ptr<AudioEncoder> CreateComfortNoiseEncoder(
+    AudioEncoderCngConfig&& config);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
new file mode 100644
index 0000000000..c688004363
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -0,0 +1,520 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+
+#include <memory>
+#include <vector>
+
+#include "common_audio/vad/mock/mock_vad.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+using ::testing::_;
+using ::testing::Eq;
+using ::testing::InSequence;
+using ::testing::Invoke;
+using ::testing::Not;
+using ::testing::Optional;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+namespace webrtc {
+
+namespace {
+static const size_t kMaxNumSamples = 48 * 10 * 2;  // 10 ms @ 48 kHz stereo.
+static const size_t kMockReturnEncodedBytes = 17;
+static const int kCngPayloadType = 18;
+}  // namespace
+
+class AudioEncoderCngTest : public ::testing::Test {
+ protected:
+  AudioEncoderCngTest()
+      : mock_encoder_owner_(new MockAudioEncoder),
+        mock_encoder_(mock_encoder_owner_.get()),
+        mock_vad_(new MockVad),
+        timestamp_(4711),
+        num_audio_samples_10ms_(0),
+        sample_rate_hz_(8000) {
+    memset(audio_, 0, kMaxNumSamples * 2);
+    EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(1));
+  }
+
+  AudioEncoderCngTest(const AudioEncoderCngTest&) = delete;
+  AudioEncoderCngTest& operator=(const AudioEncoderCngTest&) = delete;
+
+  void TearDown() override {
+    EXPECT_CALL(*mock_vad_, Die()).Times(1);
+    cng_.reset();
+  }
+
+  AudioEncoderCngConfig MakeCngConfig() {
+    AudioEncoderCngConfig config;
+    config.speech_encoder = std::move(mock_encoder_owner_);
+    EXPECT_TRUE(config.speech_encoder);
+
+    // Let the AudioEncoderCng object use a MockVad instead of its internally
+    // created Vad object.
+    config.vad = mock_vad_;
+    config.payload_type = kCngPayloadType;
+
+    return config;
+  }
+
+  void CreateCng(AudioEncoderCngConfig&& config) {
+    num_audio_samples_10ms_ = static_cast<size_t>(10 * sample_rate_hz_ / 1000);
+    ASSERT_LE(num_audio_samples_10ms_, kMaxNumSamples);
+    if (config.speech_encoder) {
+      EXPECT_CALL(*mock_encoder_, SampleRateHz())
+          .WillRepeatedly(Return(sample_rate_hz_));
+      // Max10MsFramesInAPacket() is just used to verify that the SID frame
+      // period is not too small. The return value does not matter that much,
+      // as long as it is smaller than 10.
+      EXPECT_CALL(*mock_encoder_, Max10MsFramesInAPacket())
+          .WillOnce(Return(1u));
+    }
+    cng_ = CreateComfortNoiseEncoder(std::move(config));
+  }
+
+  void Encode() {
+    ASSERT_TRUE(cng_) << "Must call CreateCng() first.";
+    encoded_info_ = cng_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms_),
+        &encoded_);
+    timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
+  }
+
+  // Expect `num_calls` calls to the encoder, all successful. The last call
+  // claims to have encoded `kMockReturnEncodedBytes` bytes, and all the
+  // preceding ones 0 bytes.
+  void ExpectEncodeCalls(size_t num_calls) {
+    InSequence s;
+    AudioEncoder::EncodedInfo info;
+    for (size_t j = 0; j < num_calls - 1; ++j) {
+      EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).WillOnce(Return(info));
+    }
+    info.encoded_bytes = kMockReturnEncodedBytes;
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(
+            Invoke(MockAudioEncoder::FakeEncoding(kMockReturnEncodedBytes)));
+  }
+
+  // Verifies that the cng_ object waits until it has collected
+  // `blocks_per_frame` blocks of audio, and then dispatches all of them to
+  // the underlying codec (speech or cng).
+  void CheckBlockGrouping(size_t blocks_per_frame, bool active_speech) {
+    EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+        .WillRepeatedly(Return(blocks_per_frame));
+    auto config = MakeCngConfig();
+    const int num_cng_coefficients = config.num_cng_coefficients;
+    CreateCng(std::move(config));
+    EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+        .WillRepeatedly(Return(active_speech ? Vad::kActive : Vad::kPassive));
+
+    // Don't expect any calls to the encoder yet.
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).Times(0);
+    for (size_t i = 0; i < blocks_per_frame - 1; ++i) {
+      Encode();
+      EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+    }
+    if (active_speech)
+      ExpectEncodeCalls(blocks_per_frame);
+    Encode();
+    if (active_speech) {
+      EXPECT_EQ(kMockReturnEncodedBytes, encoded_info_.encoded_bytes);
+    } else {
+      EXPECT_EQ(static_cast<size_t>(num_cng_coefficients + 1),
+                encoded_info_.encoded_bytes);
+    }
+  }
+
+  // Verifies that the audio is partitioned into larger blocks before calling
+  // the VAD.
+  void CheckVadInputSize(int input_frame_size_ms,
+                         int expected_first_block_size_ms,
+                         int expected_second_block_size_ms) {
+    const size_t blocks_per_frame =
+        static_cast<size_t>(input_frame_size_ms / 10);
+
+    EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+        .WillRepeatedly(Return(blocks_per_frame));
+
+    // Expect nothing to happen before the last block is sent to cng_.
+    EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)).Times(0);
+    for (size_t i = 0; i < blocks_per_frame - 1; ++i) {
+      Encode();
+    }
+
+    // Let the VAD decision be passive, since an active decision may lead to
+    // early termination of the decision loop.
+    InSequence s;
+    EXPECT_CALL(
+        *mock_vad_,
+        VoiceActivity(_, expected_first_block_size_ms * sample_rate_hz_ / 1000,
+                      sample_rate_hz_))
+        .WillOnce(Return(Vad::kPassive));
+    if (expected_second_block_size_ms > 0) {
+      EXPECT_CALL(*mock_vad_,
+                  VoiceActivity(
+                      _, expected_second_block_size_ms * sample_rate_hz_ / 1000,
+                      sample_rate_hz_))
+          .WillOnce(Return(Vad::kPassive));
+    }
+
+    // With this call to Encode(), `mock_vad_` should be called according to the
+    // above expectations.
+    Encode();
+  }
+
+  // Tests a frame with both active and passive speech. Returns true if the
+  // decision was active speech, false if it was passive.
+  bool CheckMixedActivePassive(Vad::Activity first_type,
+                               Vad::Activity second_type) {
+    // Set the speech encoder frame size to 60 ms, to ensure that the VAD will
+    // be called twice.
+    const size_t blocks_per_frame = 6;
+    EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+        .WillRepeatedly(Return(blocks_per_frame));
+    InSequence s;
+    EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+        .WillOnce(Return(first_type));
+    if (first_type == Vad::kPassive) {
+      // Expect a second call to the VAD only if the first frame was passive.
+      EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+          .WillOnce(Return(second_type));
+    }
+    encoded_info_.payload_type = 0;
+    for (size_t i = 0; i < blocks_per_frame; ++i) {
+      Encode();
+    }
+    return encoded_info_.payload_type != kCngPayloadType;
+  }
+
+  std::unique_ptr<AudioEncoder> cng_;
+  std::unique_ptr<MockAudioEncoder> mock_encoder_owner_;
+  MockAudioEncoder* mock_encoder_;
+  MockVad* mock_vad_;  // Ownership is transferred to `cng_`.
+  uint32_t timestamp_;
+  int16_t audio_[kMaxNumSamples];
+  size_t num_audio_samples_10ms_;
+  rtc::Buffer encoded_;
+  AudioEncoder::EncodedInfo encoded_info_;
+  int sample_rate_hz_;
+};
+
+TEST_F(AudioEncoderCngTest, CreateAndDestroy) {
+  CreateCng(MakeCngConfig());
+}
+
+TEST_F(AudioEncoderCngTest, CheckFrameSizePropagation) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillOnce(Return(17U));
+  EXPECT_EQ(17U, cng_->Num10MsFramesInNextPacket());
+}
+
+TEST_F(AudioEncoderCngTest, CheckTargetAudioBitratePropagation) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_,
+              OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
+  cng_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
+}
+
+TEST_F(AudioEncoderCngTest, CheckPacketLossFractionPropagation) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5));
+  cng_->OnReceivedUplinkPacketLossFraction(0.5);
+}
+
+TEST_F(AudioEncoderCngTest, CheckGetFrameLengthRangePropagation) {
+  CreateCng(MakeCngConfig());
+  auto expected_range =
+      std::make_pair(TimeDelta::Millis(20), TimeDelta::Millis(20));
+  EXPECT_CALL(*mock_encoder_, GetFrameLengthRange())
+      .WillRepeatedly(Return(absl::make_optional(expected_range)));
+  EXPECT_THAT(cng_->GetFrameLengthRange(), Optional(Eq(expected_range)));
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCallsVad) {
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(1U));
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kPassive));
+  Encode();
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects1BlockPassiveSpeech) {
+  CheckBlockGrouping(1, false);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects2BlocksPassiveSpeech) {
+  CheckBlockGrouping(2, false);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects3BlocksPassiveSpeech) {
+  CheckBlockGrouping(3, false);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects1BlockActiveSpeech) {
+  CheckBlockGrouping(1, true);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects2BlocksActiveSpeech) {
+  CheckBlockGrouping(2, true);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects3BlocksActiveSpeech) {
+  CheckBlockGrouping(3, true);
+}
+
+TEST_F(AudioEncoderCngTest, EncodePassive) {
+  const size_t kBlocksPerFrame = 3;
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(kBlocksPerFrame));
+  auto config = MakeCngConfig();
+  const auto sid_frame_interval_ms = config.sid_frame_interval_ms;
+  const auto num_cng_coefficients = config.num_cng_coefficients;
+  CreateCng(std::move(config));
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillRepeatedly(Return(Vad::kPassive));
+  // Expect no calls at all to the speech encoder mock.
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).Times(0);
+  uint32_t expected_timestamp = timestamp_;
+  for (size_t i = 0; i < 100; ++i) {
+    Encode();
+    // Check if it was time to call the cng encoder. This is done once every
+    // `kBlocksPerFrame` calls.
+    if ((i + 1) % kBlocksPerFrame == 0) {
+      // Now check if a SID interval has elapsed.
+      if ((i % (sid_frame_interval_ms / 10)) < kBlocksPerFrame) {
+        // If so, verify that we got a CNG encoding.
+        EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+        EXPECT_FALSE(encoded_info_.speech);
+        EXPECT_EQ(static_cast<size_t>(num_cng_coefficients) + 1,
+                  encoded_info_.encoded_bytes);
+        EXPECT_EQ(expected_timestamp, encoded_info_.encoded_timestamp);
+      }
+      expected_timestamp += rtc::checked_cast<uint32_t>(
+          kBlocksPerFrame * num_audio_samples_10ms_);
+    } else {
+      // Otherwise, expect no output.
+      EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+    }
+  }
+}
+
+// Verifies that the correct action is taken for frames with both active and
+// passive speech.
+TEST_F(AudioEncoderCngTest, MixedActivePassive) {
+  CreateCng(MakeCngConfig());
+
+  // All of the frame is active speech.
+  ExpectEncodeCalls(6);
+  EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kActive));
+  EXPECT_TRUE(encoded_info_.speech);
+
+  // First half of the frame is active speech.
+  ExpectEncodeCalls(6);
+  EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kPassive));
+  EXPECT_TRUE(encoded_info_.speech);
+
+  // Second half of the frame is active speech.
+  ExpectEncodeCalls(6);
+  EXPECT_TRUE(CheckMixedActivePassive(Vad::kPassive, Vad::kActive));
+  EXPECT_TRUE(encoded_info_.speech);
+
+  // All of the frame is passive speech. Expect no calls to `mock_encoder_`.
+  EXPECT_FALSE(CheckMixedActivePassive(Vad::kPassive, Vad::kPassive));
+  EXPECT_FALSE(encoded_info_.speech);
+}
+
+// These tests verify that the audio is partitioned into larger blocks before
+// calling the VAD.
+// The parameters for CheckVadInputSize are:
+// CheckVadInputSize(frame_size, expected_first_block_size,
+//                   expected_second_block_size);
+TEST_F(AudioEncoderCngTest, VadInputSize10Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(10, 10, 0);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize20Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(20, 20, 0);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize30Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(30, 30, 0);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize40Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(40, 20, 20);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize50Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(50, 30, 20);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize60Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(60, 30, 30);
+}
+
+// Verifies that the correct payload type is set when CNG is encoded.
+TEST_F(AudioEncoderCngTest, VerifyCngPayloadType) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).Times(0);
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(1U));
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kPassive));
+  encoded_info_.payload_type = 0;
+  Encode();
+  EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+}
+
+// Verifies that a SID frame is encoded immediately as the signal changes from
+// active speech to passive.
+TEST_F(AudioEncoderCngTest, VerifySidFrameAfterSpeech) {
+  auto config = MakeCngConfig();
+  const auto num_cng_coefficients = config.num_cng_coefficients;
+  CreateCng(std::move(config));
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(1U));
+  // Start with encoding noise.
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .Times(2)
+      .WillRepeatedly(Return(Vad::kPassive));
+  Encode();
+  EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+  EXPECT_EQ(static_cast<size_t>(num_cng_coefficients) + 1,
+            encoded_info_.encoded_bytes);
+  // Encode again, and make sure we got no frame at all (since the SID frame
+  // period is 100 ms by default).
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+
+  // Now encode active speech.
+  encoded_info_.payload_type = 0;
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kActive));
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(
+          Invoke(MockAudioEncoder::FakeEncoding(kMockReturnEncodedBytes)));
+  Encode();
+  EXPECT_EQ(kMockReturnEncodedBytes, encoded_info_.encoded_bytes);
+
+  // Go back to noise again, and verify that a SID frame is emitted.
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kPassive));
+  Encode();
+  EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+  EXPECT_EQ(static_cast<size_t>(num_cng_coefficients) + 1,
+            encoded_info_.encoded_bytes);
+}
+
+// Resetting the CNG should reset both the VAD and the encoder.
+TEST_F(AudioEncoderCngTest, Reset) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, Reset()).Times(1);
+  EXPECT_CALL(*mock_vad_, Reset()).Times(1);
+  cng_->Reset();
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// This test fixture tests various error conditions that makes the
+// AudioEncoderCng die via CHECKs.
+class AudioEncoderCngDeathTest : public AudioEncoderCngTest {
+ protected:
+  AudioEncoderCngDeathTest() : AudioEncoderCngTest() {
+    EXPECT_CALL(*mock_vad_, Die()).Times(1);
+    delete mock_vad_;
+    mock_vad_ = nullptr;
+  }
+
+  // Override AudioEncoderCngTest::TearDown, since that one expects a call to
+  // the destructor of `mock_vad_`. In this case, that object is already
+  // deleted.
+  void TearDown() override { cng_.reset(); }
+
+  AudioEncoderCngConfig MakeCngConfig() {
+    // Don't provide a Vad mock object, since it would leak when the test dies.
+    auto config = AudioEncoderCngTest::MakeCngConfig();
+    config.vad = nullptr;
+    return config;
+  }
+
+  void TryWrongNumCoefficients(int num) {
+    RTC_EXPECT_DEATH(
+        [&] {
+          auto config = MakeCngConfig();
+          config.num_cng_coefficients = num;
+          CreateCng(std::move(config));
+        }(),
+        "Invalid configuration");
+  }
+};
+
+TEST_F(AudioEncoderCngDeathTest, WrongFrameSize) {
+  CreateCng(MakeCngConfig());
+  num_audio_samples_10ms_ *= 2;  // 20 ms frame.
+  RTC_EXPECT_DEATH(Encode(), "");
+  num_audio_samples_10ms_ = 0;  // Zero samples.
+  RTC_EXPECT_DEATH(Encode(), "");
+}
+
+TEST_F(AudioEncoderCngDeathTest, WrongNumCoefficientsA) {
+  TryWrongNumCoefficients(-1);
+}
+
+TEST_F(AudioEncoderCngDeathTest, WrongNumCoefficientsB) {
+  TryWrongNumCoefficients(0);
+}
+
+TEST_F(AudioEncoderCngDeathTest, WrongNumCoefficientsC) {
+  TryWrongNumCoefficients(13);
+}
+
+TEST_F(AudioEncoderCngDeathTest, NullSpeechEncoder) {
+  auto config = MakeCngConfig();
+  config.speech_encoder = nullptr;
+  RTC_EXPECT_DEATH(CreateCng(std::move(config)), "");
+}
+
+TEST_F(AudioEncoderCngDeathTest, StereoEncoder) {
+  EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(2));
+  RTC_EXPECT_DEATH(CreateCng(MakeCngConfig()), "Invalid configuration");
+}
+
+TEST_F(AudioEncoderCngDeathTest, StereoConfig) {
+  RTC_EXPECT_DEATH(
+      [&] {
+        auto config = MakeCngConfig();
+        config.num_channels = 2;
+        CreateCng(std::move(config));
+      }(),
+      "Invalid configuration");
+}
+
+TEST_F(AudioEncoderCngDeathTest, EncoderFrameSizeTooLarge) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(7U));
+  for (int i = 0; i < 6; ++i)
+    Encode();
+  RTC_EXPECT_DEATH(
+      Encode(), "Frame size cannot be larger than 60 ms when using VAD/CNG.");
+}
+
+#endif  // GTEST_HAS_DEATH_TEST
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/cng_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
new file mode 100644
index 0000000000..0e6ab79394
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
@@ -0,0 +1,252 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+enum {
+  kSidShortIntervalUpdate = 1,
+  kSidNormalIntervalUpdate = 100,
+  kSidLongIntervalUpdate = 10000
+};
+
+enum : size_t {
+  kCNGNumParamsLow = 0,
+  kCNGNumParamsNormal = 8,
+  kCNGNumParamsHigh = WEBRTC_CNG_MAX_LPC_ORDER,
+  kCNGNumParamsTooHigh = WEBRTC_CNG_MAX_LPC_ORDER + 1
+};
+
+enum { kNoSid, kForceSid };
+
+class CngTest : public ::testing::Test {
+ protected:
+  virtual void SetUp();
+
+  void TestCngEncode(int sample_rate_hz, int quality);
+
+  int16_t speech_data_[640];  // Max size of CNG internal buffers.
+};
+
+class CngDeathTest : public CngTest {};
+
+void CngTest::SetUp() {
+  FILE* input_file;
+  const std::string file_name =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  input_file = fopen(file_name.c_str(), "rb");
+  ASSERT_TRUE(input_file != NULL);
+  ASSERT_EQ(640, static_cast<int32_t>(
+                     fread(speech_data_, sizeof(int16_t), 640, input_file)));
+  fclose(input_file);
+  input_file = NULL;
+}
+
+void CngTest::TestCngEncode(int sample_rate_hz, int quality) {
+  const size_t num_samples_10ms = rtc::CheckedDivExact(sample_rate_hz, 100);
+  rtc::Buffer sid_data;
+
+  ComfortNoiseEncoder cng_encoder(sample_rate_hz, kSidNormalIntervalUpdate,
+                                  quality);
+  EXPECT_EQ(0U, cng_encoder.Encode(rtc::ArrayView<const int16_t>(
+                                       speech_data_, num_samples_10ms),
+                                   kNoSid, &sid_data));
+  EXPECT_EQ(static_cast<size_t>(quality + 1),
+            cng_encoder.Encode(
+                rtc::ArrayView<const int16_t>(speech_data_, num_samples_10ms),
+                kForceSid, &sid_data));
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Create CNG encoder, init with faulty values, free CNG encoder.
+TEST_F(CngDeathTest, CngInitFail) {
+  // Call with too few parameters.
+  EXPECT_DEATH(
+      {
+        ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate, kCNGNumParamsLow);
+      },
+      "");
+  // Call with too many parameters.
+  EXPECT_DEATH(
+      {
+        ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate,
+                            kCNGNumParamsTooHigh);
+      },
+      "");
+}
+
+// Encode Cng with too long input vector.
+TEST_F(CngDeathTest, CngEncodeTooLong) {
+  rtc::Buffer sid_data;
+
+  // Create encoder.
+  ComfortNoiseEncoder cng_encoder(8000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  // Run encoder with too much data.
+  EXPECT_DEATH(
+      cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 641),
+                         kNoSid, &sid_data),
+      "");
+}
+#endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+TEST_F(CngTest, CngEncode8000) {
+  TestCngEncode(8000, kCNGNumParamsNormal);
+}
+
+TEST_F(CngTest, CngEncode16000) {
+  TestCngEncode(16000, kCNGNumParamsNormal);
+}
+
+TEST_F(CngTest, CngEncode32000) {
+  TestCngEncode(32000, kCNGNumParamsHigh);
+}
+
+TEST_F(CngTest, CngEncode48000) {
+  TestCngEncode(48000, kCNGNumParamsNormal);
+}
+
+TEST_F(CngTest, CngEncode64000) {
+  TestCngEncode(64000, kCNGNumParamsNormal);
+}
+
+// Update SID parameters, for both 9 and 16 parameters.
+TEST_F(CngTest, CngUpdateSid) {
+  rtc::Buffer sid_data;
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // Run normal Encode and UpdateSid.
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+  cng_decoder.UpdateSid(sid_data);
+
+  // Reinit with new length.
+  cng_encoder.Reset(16000, kSidNormalIntervalUpdate, kCNGNumParamsHigh);
+  cng_decoder.Reset();
+
+  // Expect 0 because of unstable parameters after switching length.
+  EXPECT_EQ(0U,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+  EXPECT_EQ(
+      kCNGNumParamsHigh + 1,
+      cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_ + 160, 160),
+                         kForceSid, &sid_data));
+  cng_decoder.UpdateSid(
+      rtc::ArrayView<const uint8_t>(sid_data.data(), kCNGNumParamsNormal + 1));
+}
+
+// Update SID parameters, with wrong parameters or without calling decode.
+TEST_F(CngTest, CngUpdateSidErroneous) {
+  rtc::Buffer sid_data;
+
+  // Encode.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+
+  // First run with valid parameters, then with too many CNG parameters.
+  // The function will operate correctly by only reading the maximum number of
+  // parameters, skipping the extra.
+  EXPECT_EQ(kCNGNumParamsNormal + 1, sid_data.size());
+  cng_decoder.UpdateSid(sid_data);
+
+  // Make sure the input buffer is large enough. Since Encode() appends data, we
+  // need to set the size manually only afterwards, or the buffer will be bigger
+  // than anticipated.
+  sid_data.SetSize(kCNGNumParamsTooHigh + 1);
+  cng_decoder.UpdateSid(sid_data);
+}
+
+// Test to generate cng data, by forcing SID. Both normal and faulty condition.
+TEST_F(CngTest, CngGenerate) {
+  rtc::Buffer sid_data;
+  int16_t out_data[640];
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // Normal Encode.
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+
+  // Normal UpdateSid.
+  cng_decoder.UpdateSid(sid_data);
+
+  // Two normal Generate, one with new_period.
+  EXPECT_TRUE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 640), 1));
+  EXPECT_TRUE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 640), 0));
+
+  // Call Genereate with too much data.
+  EXPECT_FALSE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 641), 0));
+}
+
+// Test automatic SID.
+TEST_F(CngTest, CngAutoSid) {
+  rtc::Buffer sid_data;
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // Normal Encode, 100 msec, where no SID data should be generated.
+  for (int i = 0; i < 10; i++) {
+    EXPECT_EQ(
+        0U, cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kNoSid, &sid_data));
+  }
+
+  // We have reached 100 msec, and SID data should be generated.
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kNoSid, &sid_data));
+}
+
+// Test automatic SID, with very short interval.
+TEST_F(CngTest, CngAutoSidShort) {
+  rtc::Buffer sid_data;
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidShortIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // First call will never generate SID, unless forced to.
+  EXPECT_EQ(0U,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kNoSid, &sid_data));
+
+  // Normal Encode, 100 msec, SID data should be generated all the time.
+  for (int i = 0; i < 10; i++) {
+    EXPECT_EQ(
+        kCNGNumParamsNormal + 1,
+        cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                           kNoSid, &sid_data));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc b/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
new file mode 100644
index 0000000000..48f1b8c296
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
@@ -0,0 +1,436 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+
+#include <algorithm>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kCngMaxOutsizeOrder = 640;
+
+// TODO(ossu): Rename the left-over WebRtcCng according to style guide.
+void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a);
+
+const int32_t WebRtcCng_kDbov[94] = {
+    1081109975, 858756178, 682134279, 541838517, 430397633, 341876992,
+    271562548,  215709799, 171344384, 136103682, 108110997, 85875618,
+    68213428,   54183852,  43039763,  34187699,  27156255,  21570980,
+    17134438,   13610368,  10811100,  8587562,   6821343,   5418385,
+    4303976,    3418770,   2715625,   2157098,   1713444,   1361037,
+    1081110,    858756,    682134,    541839,    430398,    341877,
+    271563,     215710,    171344,    136104,    108111,    85876,
+    68213,      54184,     43040,     34188,     27156,     21571,
+    17134,      13610,     10811,     8588,      6821,      5418,
+    4304,       3419,      2716,      2157,      1713,      1361,
+    1081,       859,       682,       542,       430,       342,
+    272,        216,       171,       136,       108,       86,
+    68,         54,        43,        34,        27,        22,
+    17,         14,        11,        9,         7,         5,
+    4,          3,         3,         2,         2,         1,
+    1,          1,         1,         1};
+
+const int16_t WebRtcCng_kCorrWindow[WEBRTC_CNG_MAX_LPC_ORDER] = {
+    32702, 32636, 32570, 32505, 32439, 32374,
+    32309, 32244, 32179, 32114, 32049, 31985};
+
+}  // namespace
+
+ComfortNoiseDecoder::ComfortNoiseDecoder() {
+  /* Needed to get the right function pointers in SPLIB. */
+  Reset();
+}
+
+void ComfortNoiseDecoder::Reset() {
+  dec_seed_ = 7777; /* For debugging only. */
+  dec_target_energy_ = 0;
+  dec_used_energy_ = 0;
+  for (auto& c : dec_target_reflCoefs_)
+    c = 0;
+  for (auto& c : dec_used_reflCoefs_)
+    c = 0;
+  for (auto& c : dec_filtstate_)
+    c = 0;
+  for (auto& c : dec_filtstateLow_)
+    c = 0;
+  dec_order_ = 5;
+  dec_target_scale_factor_ = 0;
+  dec_used_scale_factor_ = 0;
+}
+
+void ComfortNoiseDecoder::UpdateSid(rtc::ArrayView<const uint8_t> sid) {
+  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER];
+  int32_t targetEnergy;
+  size_t length = sid.size();
+  /* Throw away reflection coefficients of higher order than we can handle. */
+  if (length > (WEBRTC_CNG_MAX_LPC_ORDER + 1))
+    length = WEBRTC_CNG_MAX_LPC_ORDER + 1;
+
+  dec_order_ = static_cast<uint16_t>(length - 1);
+
+  uint8_t sid0 = std::min<uint8_t>(sid[0], 93);
+  targetEnergy = WebRtcCng_kDbov[sid0];
+  /* Take down target energy to 75%. */
+  targetEnergy = targetEnergy >> 1;
+  targetEnergy += targetEnergy >> 2;
+
+  dec_target_energy_ = targetEnergy;
+
+  /* Reconstruct coeffs with tweak for WebRtc implementation of RFC3389. */
+  if (dec_order_ == WEBRTC_CNG_MAX_LPC_ORDER) {
+    for (size_t i = 0; i < (dec_order_); i++) {
+      refCs[i] = sid[i + 1] << 8; /* Q7 to Q15*/
+      dec_target_reflCoefs_[i] = refCs[i];
+    }
+  } else {
+    for (size_t i = 0; i < (dec_order_); i++) {
+      refCs[i] = (sid[i + 1] - 127) * (1 << 8); /* Q7 to Q15. */
+      dec_target_reflCoefs_[i] = refCs[i];
+    }
+  }
+
+  for (size_t i = (dec_order_); i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
+    refCs[i] = 0;
+    dec_target_reflCoefs_[i] = refCs[i];
+  }
+}
+
+bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
+                                   bool new_period) {
+  int16_t excitation[kCngMaxOutsizeOrder];
+  int16_t low[kCngMaxOutsizeOrder];
+  int16_t lpPoly[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t ReflBetaStd = 26214;      /* 0.8 in q15. */
+  int16_t ReflBetaCompStd = 6553;   /* 0.2 in q15. */
+  int16_t ReflBetaNewP = 19661;     /* 0.6 in q15. */
+  int16_t ReflBetaCompNewP = 13107; /* 0.4 in q15. */
+  int16_t Beta, BetaC;              /* These are in Q15. */
+  int32_t targetEnergy;
+  int16_t En;
+  int16_t temp16;
+  const size_t num_samples = out_data.size();
+
+  if (num_samples > kCngMaxOutsizeOrder) {
+    return false;
+  }
+
+  if (new_period) {
+    dec_used_scale_factor_ = dec_target_scale_factor_;
+    Beta = ReflBetaNewP;
+    BetaC = ReflBetaCompNewP;
+  } else {
+    Beta = ReflBetaStd;
+    BetaC = ReflBetaCompStd;
+  }
+
+  /* Calculate new scale factor in Q13 */
+  dec_used_scale_factor_ = rtc::checked_cast<int16_t>(
+      WEBRTC_SPL_MUL_16_16_RSFT(dec_used_scale_factor_, Beta >> 2, 13) +
+      WEBRTC_SPL_MUL_16_16_RSFT(dec_target_scale_factor_, BetaC >> 2, 13));
+
+  dec_used_energy_ = dec_used_energy_ >> 1;
+  dec_used_energy_ += dec_target_energy_ >> 1;
+
+  /* Do the same for the reflection coeffs, albeit in Q15. */
+  for (size_t i = 0; i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
+    dec_used_reflCoefs_[i] =
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_used_reflCoefs_[i], Beta, 15);
+    dec_used_reflCoefs_[i] +=
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_target_reflCoefs_[i], BetaC, 15);
+  }
+
+  /* Compute the polynomial coefficients. */
+  WebRtcCng_K2a16(dec_used_reflCoefs_, WEBRTC_CNG_MAX_LPC_ORDER, lpPoly);
+
+  targetEnergy = dec_used_energy_;
+
+  /* Calculate scaling factor based on filter energy. */
+  En = 8192; /* 1.0 in Q13. */
+  for (size_t i = 0; i < (WEBRTC_CNG_MAX_LPC_ORDER); i++) {
+    /* Floating point value for reference.
+       E *= 1.0 - (dec_used_reflCoefs_[i] / 32768.0) *
+       (dec_used_reflCoefs_[i] / 32768.0);
+     */
+
+    /* Same in fixed point. */
+    /* K(i).^2 in Q15. */
+    temp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_used_reflCoefs_[i],
+                                                dec_used_reflCoefs_[i], 15);
+    /* 1 - K(i).^2 in Q15. */
+    temp16 = 0x7fff - temp16;
+    En = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(En, temp16, 15);
+  }
+
+  /* float scaling= sqrt(E * dec_target_energy_ / (1 << 24)); */
+
+  /* Calculate sqrt(En * target_energy / excitation energy) */
+  targetEnergy = WebRtcSpl_Sqrt(dec_used_energy_);
+
+  En = (int16_t)WebRtcSpl_Sqrt(En) << 6;
+  En = (En * 3) >> 1; /* 1.5 estimates sqrt(2). */
+  dec_used_scale_factor_ = (int16_t)((En * targetEnergy) >> 12);
+
+  /* Generate excitation. */
+  /* Excitation energy per sample is 2.^24 - Q13 N(0,1). */
+  for (size_t i = 0; i < num_samples; i++) {
+    excitation[i] = WebRtcSpl_RandN(&dec_seed_) >> 1;
+  }
+
+  /* Scale to correct energy. */
+  WebRtcSpl_ScaleVector(excitation, excitation, dec_used_scale_factor_,
+                        num_samples, 13);
+
+  /* `lpPoly` - Coefficients in Q12.
+   * `excitation` - Speech samples.
+   * `nst->dec_filtstate` - State preservation.
+   * `out_data` - Filtered speech samples. */
+  WebRtcSpl_FilterAR(lpPoly, WEBRTC_CNG_MAX_LPC_ORDER + 1, excitation,
+                     num_samples, dec_filtstate_, WEBRTC_CNG_MAX_LPC_ORDER,
+                     dec_filtstateLow_, WEBRTC_CNG_MAX_LPC_ORDER,
+                     out_data.data(), low, num_samples);
+
+  return true;
+}
+
+ComfortNoiseEncoder::ComfortNoiseEncoder(int fs, int interval, int quality)
+    : enc_nrOfCoefs_(quality),
+      enc_sampfreq_(fs),
+      enc_interval_(interval),
+      enc_msSinceSid_(0),
+      enc_Energy_(0),
+      enc_reflCoefs_{0},
+      enc_corrVector_{0},
+      enc_seed_(7777) /* For debugging only. */ {
+  RTC_CHECK_GT(quality, 0);
+  RTC_CHECK_LE(quality, WEBRTC_CNG_MAX_LPC_ORDER);
+}
+
+void ComfortNoiseEncoder::Reset(int fs, int interval, int quality) {
+  RTC_CHECK_GT(quality, 0);
+  RTC_CHECK_LE(quality, WEBRTC_CNG_MAX_LPC_ORDER);
+  enc_nrOfCoefs_ = quality;
+  enc_sampfreq_ = fs;
+  enc_interval_ = interval;
+  enc_msSinceSid_ = 0;
+  enc_Energy_ = 0;
+  for (auto& c : enc_reflCoefs_)
+    c = 0;
+  for (auto& c : enc_corrVector_)
+    c = 0;
+  enc_seed_ = 7777; /* For debugging only. */
+}
+
+size_t ComfortNoiseEncoder::Encode(rtc::ArrayView<const int16_t> speech,
+                                   bool force_sid,
+                                   rtc::Buffer* output) {
+  int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int32_t corrVector[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t hanningW[kCngMaxOutsizeOrder];
+  int16_t ReflBeta = 19661;     /* 0.6 in q15. */
+  int16_t ReflBetaComp = 13107; /* 0.4 in q15. */
+  int32_t outEnergy;
+  int outShifts;
+  size_t i;
+  int stab;
+  int acorrScale;
+  size_t index;
+  size_t ind, factor;
+  int32_t* bptr;
+  int32_t blo, bhi;
+  int16_t negate;
+  const int16_t* aptr;
+  int16_t speechBuf[kCngMaxOutsizeOrder];
+
+  const size_t num_samples = speech.size();
+  RTC_CHECK_LE(num_samples, kCngMaxOutsizeOrder);
+
+  for (i = 0; i < num_samples; i++) {
+    speechBuf[i] = speech[i];
+  }
+
+  factor = num_samples;
+
+  /* Calculate energy and a coefficients. */
+  outEnergy = WebRtcSpl_Energy(speechBuf, num_samples, &outShifts);
+  while (outShifts > 0) {
+    /* We can only do 5 shifts without destroying accuracy in
+     * division factor. */
+    if (outShifts > 5) {
+      outEnergy <<= (outShifts - 5);
+      outShifts = 5;
+    } else {
+      factor /= 2;
+      outShifts--;
+    }
+  }
+  outEnergy = WebRtcSpl_DivW32W16(outEnergy, (int16_t)factor);
+
+  if (outEnergy > 1) {
+    /* Create Hanning Window. */
+    WebRtcSpl_GetHanningWindow(hanningW, num_samples / 2);
+    for (i = 0; i < (num_samples / 2); i++)
+      hanningW[num_samples - i - 1] = hanningW[i];
+
+    WebRtcSpl_ElementwiseVectorMult(speechBuf, hanningW, speechBuf, num_samples,
+                                    14);
+
+    WebRtcSpl_AutoCorrelation(speechBuf, num_samples, enc_nrOfCoefs_,
+                              corrVector, &acorrScale);
+
+    if (*corrVector == 0)
+      *corrVector = WEBRTC_SPL_WORD16_MAX;
+
+    /* Adds the bandwidth expansion. */
+    aptr = WebRtcCng_kCorrWindow;
+    bptr = corrVector;
+
+    /* (zzz) lpc16_1 = 17+1+820+2+2 = 842 (ordo2=700). */
+    for (ind = 0; ind < enc_nrOfCoefs_; ind++) {
+      /* The below code multiplies the 16 b corrWindow values (Q15) with
+       * the 32 b corrvector (Q0) and shifts the result down 15 steps. */
+      negate = *bptr < 0;
+      if (negate)
+        *bptr = -*bptr;
+
+      blo = (int32_t)*aptr * (*bptr & 0xffff);
+      bhi = ((blo >> 16) & 0xffff) +
+            ((int32_t)(*aptr++) * ((*bptr >> 16) & 0xffff));
+      blo = (blo & 0xffff) | ((bhi & 0xffff) << 16);
+
+      *bptr = (((bhi >> 16) & 0x7fff) << 17) | ((uint32_t)blo >> 15);
+      if (negate)
+        *bptr = -*bptr;
+      bptr++;
+    }
+    /* End of bandwidth expansion. */
+
+    stab = WebRtcSpl_LevinsonDurbin(corrVector, arCoefs, refCs, enc_nrOfCoefs_);
+
+    if (!stab) {
+      /* Disregard from this frame */
+      return 0;
+    }
+
+  } else {
+    for (i = 0; i < enc_nrOfCoefs_; i++)
+      refCs[i] = 0;
+  }
+
+  if (force_sid) {
+    /* Read instantaneous values instead of averaged. */
+    for (i = 0; i < enc_nrOfCoefs_; i++)
+      enc_reflCoefs_[i] = refCs[i];
+    enc_Energy_ = outEnergy;
+  } else {
+    /* Average history with new values. */
+    for (i = 0; i < enc_nrOfCoefs_; i++) {
+      enc_reflCoefs_[i] =
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(enc_reflCoefs_[i], ReflBeta, 15);
+      enc_reflCoefs_[i] +=
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(refCs[i], ReflBetaComp, 15);
+    }
+    enc_Energy_ = (outEnergy >> 2) + (enc_Energy_ >> 1) + (enc_Energy_ >> 2);
+  }
+
+  if (enc_Energy_ < 1) {
+    enc_Energy_ = 1;
+  }
+
+  if ((enc_msSinceSid_ > (enc_interval_ - 1)) || force_sid) {
+    /* Search for best dbov value. */
+    index = 0;
+    for (i = 1; i < 93; i++) {
+      /* Always round downwards. */
+      if ((enc_Energy_ - WebRtcCng_kDbov[i]) > 0) {
+        index = i;
+        break;
+      }
+    }
+    if ((i == 93) && (index == 0))
+      index = 94;
+
+    const size_t output_coefs = enc_nrOfCoefs_ + 1;
+    output->AppendData(output_coefs, [&](rtc::ArrayView<uint8_t> output) {
+      output[0] = (uint8_t)index;
+
+      /* Quantize coefficients with tweak for WebRtc implementation of
+       * RFC3389. */
+      if (enc_nrOfCoefs_ == WEBRTC_CNG_MAX_LPC_ORDER) {
+        for (i = 0; i < enc_nrOfCoefs_; i++) {
+          /* Q15 to Q7 with rounding. */
+          output[i + 1] = ((enc_reflCoefs_[i] + 128) >> 8);
+        }
+      } else {
+        for (i = 0; i < enc_nrOfCoefs_; i++) {
+          /* Q15 to Q7 with rounding. */
+          output[i + 1] = (127 + ((enc_reflCoefs_[i] + 128) >> 8));
+        }
+      }
+
+      return output_coefs;
+    });
+
+    enc_msSinceSid_ =
+        static_cast<int16_t>((1000 * num_samples) / enc_sampfreq_);
+    return output_coefs;
+  } else {
+    enc_msSinceSid_ +=
+        static_cast<int16_t>((1000 * num_samples) / enc_sampfreq_);
+    return 0;
+  }
+}
+
+namespace {
+/* Values in `k` are Q15, and `a` Q12. */
+void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a) {
+  int16_t any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
+  int16_t* aptr;
+  int16_t* aptr2;
+  int16_t* anyptr;
+  const int16_t* kptr;
+  int m, i;
+
+  kptr = k;
+  *a = 4096; /* i.e., (Word16_MAX >> 3) + 1 */
+  *any = *a;
+  a[1] = (*k + 4) >> 3;
+  for (m = 1; m < useOrder; m++) {
+    kptr++;
+    aptr = a;
+    aptr++;
+    aptr2 = &a[m];
+    anyptr = any;
+    anyptr++;
+
+    any[m + 1] = (*kptr + 4) >> 3;
+    for (i = 0; i < m; i++) {
+      *anyptr++ =
+          (*aptr++) +
+          (int16_t)((((int32_t)(*aptr2--) * (int32_t)*kptr) + 16384) >> 15);
+    }
+
+    aptr = a;
+    anyptr = any;
+    for (i = 0; i < (m + 2); i++) {
+      *aptr++ = *anyptr++;
+    }
+  }
+}
+
+}  // namespace
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.h b/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
new file mode 100644
index 0000000000..7afd243f81
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+#define MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+
+#include <stdint.h>
+
+#include <cstddef>
+
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+
+#define WEBRTC_CNG_MAX_LPC_ORDER 12
+
+namespace webrtc {
+
+class ComfortNoiseDecoder {
+ public:
+  ComfortNoiseDecoder();
+  ~ComfortNoiseDecoder() = default;
+
+  ComfortNoiseDecoder(const ComfortNoiseDecoder&) = delete;
+  ComfortNoiseDecoder& operator=(const ComfortNoiseDecoder&) = delete;
+
+  void Reset();
+
+  // Updates the CN state when a new SID packet arrives.
+  // `sid` is a view of the SID packet without the headers.
+  void UpdateSid(rtc::ArrayView<const uint8_t> sid);
+
+  // Generates comfort noise.
+  // `out_data` will be filled with samples - its size determines the number of
+  // samples generated. When `new_period` is true, CNG history will be reset
+  // before any audio is generated.  Returns `false` if outData is too large -
+  // currently 640 bytes (equalling 10ms at 64kHz).
+  // TODO(ossu): Specify better limits for the size of out_data. Either let it
+  //             be unbounded or limit to 10ms in the current sample rate.
+  bool Generate(rtc::ArrayView<int16_t> out_data, bool new_period);
+
+ private:
+  uint32_t dec_seed_;
+  int32_t dec_target_energy_;
+  int32_t dec_used_energy_;
+  int16_t dec_target_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_used_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_filtstate_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_filtstateLow_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  uint16_t dec_order_;
+  int16_t dec_target_scale_factor_; /* Q29 */
+  int16_t dec_used_scale_factor_;   /* Q29 */
+};
+
+class ComfortNoiseEncoder {
+ public:
+  // Creates a comfort noise encoder.
+  // `fs` selects sample rate: 8000 for narrowband or 16000 for wideband.
+  // `interval` sets the interval at which to generate SID data (in ms).
+  // `quality` selects the number of refl. coeffs. Maximum allowed is 12.
+  ComfortNoiseEncoder(int fs, int interval, int quality);
+  ~ComfortNoiseEncoder() = default;
+
+  ComfortNoiseEncoder(const ComfortNoiseEncoder&) = delete;
+  ComfortNoiseEncoder& operator=(const ComfortNoiseEncoder&) = delete;
+
+  // Resets the comfort noise encoder to its initial state.
+  // Parameters are set as during construction.
+  void Reset(int fs, int interval, int quality);
+
+  // Analyzes background noise from `speech` and appends coefficients to
+  // `output`.  Returns the number of coefficients generated.  If `force_sid` is
+  // true, a SID frame is forced and the internal sid interval counter is reset.
+  // Will fail if the input size is too large (> 640 samples, see
+  // ComfortNoiseDecoder::Generate).
+  size_t Encode(rtc::ArrayView<const int16_t> speech,
+                bool force_sid,
+                rtc::Buffer* output);
+
+ private:
+  size_t enc_nrOfCoefs_;
+  int enc_sampfreq_;
+  int16_t enc_interval_;
+  int16_t enc_msSinceSid_;
+  int32_t enc_Energy_;
+  int16_t enc_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int32_t enc_corrVector_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  uint32_t enc_seed_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
new file mode 100644
index 0000000000..46ac671b30
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+namespace webrtc {
+
+void AudioDecoderPcmU::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmU::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmU::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted =
+      PacketDuration(encoded, encoded_len) *
+      Channels();         // 1 byte per sample per channel
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG711_DecodeU(encoded, encoded_len_adjusted, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+void AudioDecoderPcmA::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmA::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmA::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted =
+      PacketDuration(encoded, encoded_len) *
+      Channels();         // 1 byte per sample per channel
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG711_DecodeA(encoded, encoded_len_adjusted, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
new file mode 100644
index 0000000000..3fa42cba30
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class AudioDecoderPcmU final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1);
+  }
+
+  AudioDecoderPcmU(const AudioDecoderPcmU&) = delete;
+  AudioDecoderPcmU& operator=(const AudioDecoderPcmU&) = delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const size_t num_channels_;
+};
+
+class AudioDecoderPcmA final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1);
+  }
+
+  AudioDecoderPcmA(const AudioDecoderPcmA&) = delete;
+  AudioDecoderPcmA& operator=(const AudioDecoderPcmA&) = delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const size_t num_channels_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
new file mode 100644
index 0000000000..65e2da479d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+bool AudioEncoderPcm::Config::IsOk() const {
+  return (frame_size_ms % 10 == 0) && (num_channels >= 1);
+}
+
+AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
+    : sample_rate_hz_(sample_rate_hz),
+      num_channels_(config.num_channels),
+      payload_type_(config.payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      full_frame_samples_(config.num_channels * config.frame_size_ms *
+                          sample_rate_hz / 1000),
+      first_timestamp_in_buffer_(0) {
+  RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
+  RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
+      << "Frame size must be an integer multiple of 10 ms.";
+  speech_buffer_.reserve(full_frame_samples_);
+}
+
+AudioEncoderPcm::~AudioEncoderPcm() = default;
+
+int AudioEncoderPcm::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioEncoderPcm::NumChannels() const {
+  return num_channels_;
+}
+
+size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderPcm::GetTargetBitrate() const {
+  return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
+                          NumChannels());
+}
+
+AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (speech_buffer_.empty()) {
+    first_timestamp_in_buffer_ = rtp_timestamp;
+  }
+  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
+  if (speech_buffer_.size() < full_frame_samples_) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
+  EncodedInfo info;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoded_bytes = encoded->AppendData(
+      full_frame_samples_ * BytesPerSample(),
+      [&](rtc::ArrayView<uint8_t> encoded) {
+        return EncodeCall(&speech_buffer_[0], full_frame_samples_,
+                          encoded.data());
+      });
+  speech_buffer_.clear();
+  info.encoder_type = GetCodecType();
+  return info;
+}
+
+void AudioEncoderPcm::Reset() {
+  speech_buffer_.clear();
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderPcm::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeA(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmA::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmA;
+}
+
+size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeU(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmU::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmU;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
new file mode 100644
index 0000000000..d50be4b457
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+class AudioEncoderPcm : public AudioEncoder {
+ public:
+  struct Config {
+   public:
+    bool IsOk() const;
+
+    int frame_size_ms;
+    size_t num_channels;
+    int payload_type;
+
+   protected:
+    explicit Config(int pt)
+        : frame_size_ms(20), num_channels(1), payload_type(pt) {}
+  };
+
+  ~AudioEncoderPcm() override;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  AudioEncoderPcm(const Config& config, int sample_rate_hz);
+
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+  virtual size_t EncodeCall(const int16_t* audio,
+                            size_t input_len,
+                            uint8_t* encoded) = 0;
+
+  virtual size_t BytesPerSample() const = 0;
+
+  // Used to set EncodedInfoLeaf::encoder_type in
+  // AudioEncoderPcm::EncodeImpl
+  virtual AudioEncoder::CodecType GetCodecType() const = 0;
+
+ private:
+  const int sample_rate_hz_;
+  const size_t num_channels_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  const size_t full_frame_samples_;
+  std::vector<int16_t> speech_buffer_;
+  uint32_t first_timestamp_in_buffer_;
+};
+
+class AudioEncoderPcmA final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+    Config() : AudioEncoderPcm::Config(8) {}
+  };
+
+  explicit AudioEncoderPcmA(const Config& config)
+      : AudioEncoderPcm(config, kSampleRateHz) {}
+
+  AudioEncoderPcmA(const AudioEncoderPcmA&) = delete;
+  AudioEncoderPcmA& operator=(const AudioEncoderPcmA&) = delete;
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+
+ private:
+  static const int kSampleRateHz = 8000;
+};
+
+class AudioEncoderPcmU final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+    Config() : AudioEncoderPcm::Config(0) {}
+  };
+
+  explicit AudioEncoderPcmU(const Config& config)
+      : AudioEncoderPcm(config, kSampleRateHz) {}
+
+  AudioEncoderPcmU(const AudioEncoderPcmU&) = delete;
+  AudioEncoderPcmU& operator=(const AudioEncoderPcmU&) = delete;
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+
+ private:
+  static const int kSampleRateHz = 8000;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c
new file mode 100644
index 0000000000..5fe1692ccb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include "modules/third_party/g711/g711.h"
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    encoded[n] = linear_to_alaw(speechIn[n]);
+  return len;
+}
+
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    encoded[n] = linear_to_ulaw(speechIn[n]);
+  return len;
+}
+
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    decoded[n] = alaw_to_linear(encoded[n]);
+  *speechType = 1;
+  return len;
+}
+
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    decoded[n] = ulaw_to_linear(encoded[n]);
+  *speechType = 1;
+  return len;
+}
+
+int16_t WebRtcG711_Version(char* version, int16_t lenBytes) {
+  strncpy(version, "2.0.0", lenBytes);
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h
new file mode 100644
index 0000000000..c92e6cc1c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h
@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+// Comfort noise constants
+#define G711_WEBRTC_SPEECH 1
+#define G711_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcG711_EncodeA(...)
+ *
+ * This function encodes a G711 A-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ *      - speechIn           : Input speech vector
+ *      - len                : Samples in speechIn
+ *
+ * Output:
+ *      - encoded            : The encoded data vector
+ *
+ * Return value              : Length (in bytes) of coded data.
+ *                             Always equal to len input parameter.
+ */
+
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG711_EncodeU(...)
+ *
+ * This function encodes a G711 U-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ *      - speechIn           : Input speech vector
+ *      - len                : Samples in speechIn
+ *
+ * Output:
+ *      - encoded            : The encoded data vector
+ *
+ * Return value              : Length (in bytes) of coded data.
+ *                             Always equal to len input parameter.
+ */
+
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG711_DecodeA(...)
+ *
+ * This function decodes a packet G711 A-law frame.
+ *
+ * Input:
+ *      - encoded            : Encoded data
+ *      - len                : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - speechType         : 1 normal, 2 CNG (for G711 it should
+ *                             always return 1 since G711 does not have a
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples in decoded vector
+ *                             -1 - Error
+ */
+
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcG711_DecodeU(...)
+ *
+ * This function decodes a packet G711 U-law frame.
+ *
+ * Input:
+ *      - encoded            : Encoded data
+ *      - len                : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - speechType         : 1 normal, 2 CNG (for G711 it should
+ *                             always return 1 since G711 does not have a
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples in decoded vector
+ *                             -1 - Error
+ */
+
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
+
+/**********************************************************************
+ * WebRtcG711_Version(...)
+ *
+ * This function gives the version string of the G.711 codec.
+ *
+ * Input:
+ *      - lenBytes:     the size of Allocated space (in Bytes) where
+ *                      the version number is written to (in string format).
+ *
+ * Output:
+ *      - version:      Pointer to a buffer where the version number is
+ *                      written to.
+ *
+ */
+
+int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc
new file mode 100644
index 0000000000..f3a42f5d79
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * testG711.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+/* include API */
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G711 1000
+
+/* function for reading audio data from PCM file */
+bool readframe(int16_t* data, FILE* inp, size_t length) {
+  size_t rlen = fread(data, sizeof(int16_t), length, inp);
+  if (rlen >= length)
+    return false;
+  memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+  return true;
+}
+
+int main(int argc, char* argv[]) {
+  char inname[80], outname[40], bitname[40];
+  FILE* inp;
+  FILE* outp;
+  FILE* bitp = NULL;
+  int framecnt;
+  bool endfile;
+
+  size_t framelength = 80;
+
+  /* Runtime statistics */
+  double starttime;
+  double runtime;
+  double length_file;
+
+  size_t stream_len = 0;
+  int16_t shortdata[480];
+  int16_t decoded[480];
+  uint8_t streamdata[1000];
+  int16_t speechType[1];
+  char law[2];
+  char versionNumber[40];
+
+  /* handling wrong input arguments in the command line */
+  if ((argc != 5) && (argc != 6)) {
+    printf("\n\nWrong number of arguments or flag values.\n\n");
+
+    printf("\n");
+    printf("\nG.711 test application\n\n");
+    printf("Usage:\n\n");
+    printf("./testG711.exe framelength law infile outfile \n\n");
+    printf("framelength: Framelength in samples.\n");
+    printf("law        : Coding law, A och u.\n");
+    printf("infile     : Normal speech input file\n");
+    printf("outfile    : Speech output file\n\n");
+    printf("outbits    : Output bitstream file [optional]\n\n");
+    exit(0);
+  }
+
+  /* Get version and print */
+  WebRtcG711_Version(versionNumber, 40);
+
+  printf("-----------------------------------\n");
+  printf("G.711 version: %s\n\n", versionNumber);
+  /* Get frame length */
+  int framelength_int = atoi(argv[1]);
+  if (framelength_int < 0) {
+    printf("  G.722: Invalid framelength %d.\n", framelength_int);
+    exit(1);
+  }
+  framelength = static_cast<size_t>(framelength_int);
+
+  /* Get compression law */
+  strcpy(law, argv[2]);
+
+  /* Get Input and Output files */
+  sscanf(argv[3], "%s", inname);
+  sscanf(argv[4], "%s", outname);
+  if (argc == 6) {
+    sscanf(argv[5], "%s", bitname);
+    if ((bitp = fopen(bitname, "wb")) == NULL) {
+      printf("  G.711: Cannot read file %s.\n", bitname);
+      exit(1);
+    }
+  }
+
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  G.711: Cannot read file %s.\n", inname);
+    exit(1);
+  }
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  G.711: Cannot write file %s.\n", outname);
+    exit(1);
+  }
+  printf("\nInput:  %s\nOutput: %s\n", inname, outname);
+  if (argc == 6) {
+    printf("\nBitfile:  %s\n", bitname);
+  }
+
+  starttime = clock() / (double)CLOCKS_PER_SEC_G711; /* Runtime statistics */
+
+  /* Initialize encoder and decoder */
+  framecnt = 0;
+  endfile = false;
+  while (!endfile) {
+    framecnt++;
+    /* Read speech block */
+    endfile = readframe(shortdata, inp, framelength);
+
+    /* G.711 encoding */
+    if (!strcmp(law, "A")) {
+      /* A-law encoding */
+      stream_len = WebRtcG711_EncodeA(shortdata, framelength, streamdata);
+      if (argc == 6) {
+        /* Write bits to file */
+        if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
+            stream_len) {
+          return -1;
+        }
+      }
+      WebRtcG711_DecodeA(streamdata, stream_len, decoded, speechType);
+    } else if (!strcmp(law, "u")) {
+      /* u-law encoding */
+      stream_len = WebRtcG711_EncodeU(shortdata, framelength, streamdata);
+      if (argc == 6) {
+        /* Write bits to file */
+        if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
+            stream_len) {
+          return -1;
+        }
+      }
+      WebRtcG711_DecodeU(streamdata, stream_len, decoded, speechType);
+    } else {
+      printf("Wrong law mode\n");
+      exit(1);
+    }
+    /* Write coded speech to file */
+    if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) {
+      return -1;
+    }
+  }
+
+  runtime = (double)(clock() / (double)CLOCKS_PER_SEC_G711 - starttime);
+  length_file = ((double)framecnt * (double)framelength / 8000);
+  printf("\n\nLength of speech file: %.1f s\n", length_file);
+  printf("Time to run G.711:      %.2f s (%.2f %% of realtime)\n\n", runtime,
+         (100 * runtime / length_file));
+  printf("---------------------END----------------------\n");
+
+  fclose(inp);
+  fclose(outp);
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
new file mode 100644
index 0000000000..1ecc9bc3d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
@@ -0,0 +1,178 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+
+#include <string.h>
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderG722Impl::AudioDecoderG722Impl() {
+  WebRtcG722_CreateDecoder(&dec_state_);
+  WebRtcG722_DecoderInit(dec_state_);
+}
+
+AudioDecoderG722Impl::~AudioDecoderG722Impl() {
+  WebRtcG722_FreeDecoder(dec_state_);
+}
+
+bool AudioDecoderG722Impl::HasDecodePlc() const {
+  return false;
+}
+
+int AudioDecoderG722Impl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+void AudioDecoderG722Impl::Reset() {
+  WebRtcG722_DecoderInit(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722Impl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
+                                                 timestamp, 8, 16);
+}
+
+int AudioDecoderG722Impl::PacketDuration(const uint8_t* encoded,
+                                         size_t encoded_len) const {
+  // 1/2 encoded byte per sample per channel.
+  return static_cast<int>(2 * encoded_len / Channels());
+}
+
+int AudioDecoderG722Impl::SampleRateHz() const {
+  return 16000;
+}
+
+size_t AudioDecoderG722Impl::Channels() const {
+  return 1;
+}
+
+AudioDecoderG722StereoImpl::AudioDecoderG722StereoImpl() {
+  WebRtcG722_CreateDecoder(&dec_state_left_);
+  WebRtcG722_CreateDecoder(&dec_state_right_);
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+AudioDecoderG722StereoImpl::~AudioDecoderG722StereoImpl() {
+  WebRtcG722_FreeDecoder(dec_state_left_);
+  WebRtcG722_FreeDecoder(dec_state_right_);
+}
+
+int AudioDecoderG722StereoImpl::DecodeInternal(const uint8_t* encoded,
+                                               size_t encoded_len,
+                                               int sample_rate_hz,
+                                               int16_t* decoded,
+                                               SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted = PacketDuration(encoded, encoded_len) *
+                                      Channels() /
+                                      2;  // 1/2 byte per sample per channel
+  int16_t temp_type = 1;  // Default is speech.
+  // De-interleave the bit-stream into two separate payloads.
+  uint8_t* encoded_deinterleaved = new uint8_t[encoded_len_adjusted];
+  SplitStereoPacket(encoded, encoded_len_adjusted, encoded_deinterleaved);
+  // Decode left and right.
+  size_t decoded_len =
+      WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
+                        encoded_len_adjusted / 2, decoded, &temp_type);
+  size_t ret = WebRtcG722_Decode(
+      dec_state_right_, &encoded_deinterleaved[encoded_len_adjusted / 2],
+      encoded_len_adjusted / 2, &decoded[decoded_len], &temp_type);
+  if (ret == decoded_len) {
+    ret += decoded_len;  // Return total number of samples.
+    // Interleave output.
+    for (size_t k = ret / 2; k < ret; k++) {
+      int16_t temp = decoded[k];
+      memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
+              (ret - k - 1) * sizeof(int16_t));
+      decoded[2 * k - ret + 1] = temp;
+    }
+  }
+  *speech_type = ConvertSpeechType(temp_type);
+  delete[] encoded_deinterleaved;
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderG722StereoImpl::PacketDuration(const uint8_t* encoded,
+                                               size_t encoded_len) const {
+  // 1/2 encoded byte per sample per channel. Make sure the length represents
+  // an equal number of bytes per channel. Otherwise, we cannot de-interleave
+  // the encoded data later.
+  return static_cast<int>(2 * (encoded_len / Channels()));
+}
+
+int AudioDecoderG722StereoImpl::SampleRateHz() const {
+  return 16000;
+}
+
+size_t AudioDecoderG722StereoImpl::Channels() const {
+  return 2;
+}
+
+void AudioDecoderG722StereoImpl::Reset() {
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722StereoImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
+                                                 timestamp, 2 * 8, 16);
+}
+
+// Split the stereo packet and place left and right channel after each other
+// in the output array.
+void AudioDecoderG722StereoImpl::SplitStereoPacket(
+    const uint8_t* encoded,
+    size_t encoded_len,
+    uint8_t* encoded_deinterleaved) {
+  // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
+  // where "lx" is 4 bits representing left sample number x, and "rx" right
+  // sample. Two samples fit in one byte, represented with |...|.
+  for (size_t i = 0; i + 1 < encoded_len; i += 2) {
+    uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
+    encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
+    encoded_deinterleaved[i + 1] = right_byte;
+  }
+
+  // Move one byte representing right channel each loop, and place it at the
+  // end of the bytestream vector. After looping the data is reordered to:
+  // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
+  // where N is the total number of samples.
+  for (size_t i = 0; i < encoded_len / 2; i++) {
+    uint8_t right_byte = encoded_deinterleaved[i + 1];
+    memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
+            encoded_len - i - 2);
+    encoded_deinterleaved[encoded_len - 1] = right_byte;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
new file mode 100644
index 0000000000..5872fad5de
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+
+#include "api/audio_codecs/audio_decoder.h"
+
+typedef struct WebRtcG722DecInst G722DecInst;
+
+namespace webrtc {
+
+class AudioDecoderG722Impl final : public AudioDecoder {
+ public:
+  AudioDecoderG722Impl();
+  ~AudioDecoderG722Impl() override;
+
+  AudioDecoderG722Impl(const AudioDecoderG722Impl&) = delete;
+  AudioDecoderG722Impl& operator=(const AudioDecoderG722Impl&) = delete;
+
+  bool HasDecodePlc() const override;
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  G722DecInst* dec_state_;
+};
+
+class AudioDecoderG722StereoImpl final : public AudioDecoder {
+ public:
+  AudioDecoderG722StereoImpl();
+  ~AudioDecoderG722StereoImpl() override;
+
+  AudioDecoderG722StereoImpl(const AudioDecoderG722StereoImpl&) = delete;
+  AudioDecoderG722StereoImpl& operator=(const AudioDecoderG722StereoImpl&) =
+      delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int SampleRateHz() const override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  // Splits the stereo-interleaved payload in `encoded` into separate payloads
+  // for left and right channels. The separated payloads are written to
+  // `encoded_deinterleaved`, which must hold at least `encoded_len` samples.
+  // The left channel starts at offset 0, while the right channel starts at
+  // offset encoded_len / 2 into `encoded_deinterleaved`.
+  void SplitStereoPacket(const uint8_t* encoded,
+                         size_t encoded_len,
+                         uint8_t* encoded_deinterleaved);
+
+  G722DecInst* dec_state_left_;
+  G722DecInst* dec_state_right_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
new file mode 100644
index 0000000000..b7d34ba581
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kSampleRateHz = 16000;
+
+}  // namespace
+
+AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config,
+                                           int payload_type)
+    : num_channels_(config.num_channels),
+      payload_type_(payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      num_10ms_frames_buffered_(0),
+      first_timestamp_in_buffer_(0),
+      encoders_(new EncoderState[num_channels_]),
+      interleave_buffer_(2 * num_channels_) {
+  RTC_CHECK(config.IsOk());
+  const size_t samples_per_channel =
+      kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+  for (size_t i = 0; i < num_channels_; ++i) {
+    encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
+    encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
+  }
+  Reset();
+}
+
+AudioEncoderG722Impl::~AudioEncoderG722Impl() = default;
+
+int AudioEncoderG722Impl::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+size_t AudioEncoderG722Impl::NumChannels() const {
+  return num_channels_;
+}
+
+int AudioEncoderG722Impl::RtpTimestampRateHz() const {
+  // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
+  // codec.
+  return kSampleRateHz / 2;
+}
+
+size_t AudioEncoderG722Impl::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderG722Impl::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderG722Impl::GetTargetBitrate() const {
+  // 4 bits/sample, 16000 samples/s/channel.
+  return static_cast<int>(64000 * NumChannels());
+}
+
+void AudioEncoderG722Impl::Reset() {
+  num_10ms_frames_buffered_ = 0;
+  for (size_t i = 0; i < num_channels_; ++i)
+    RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderG722Impl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+AudioEncoder::EncodedInfo AudioEncoderG722Impl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (num_10ms_frames_buffered_ == 0)
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  // Deinterleave samples and save them in each channel's buffer.
+  const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
+  for (size_t i = 0; i < kSampleRateHz / 100; ++i)
+    for (size_t j = 0; j < num_channels_; ++j)
+      encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
+
+  // If we don't yet have enough samples for a packet, we're done for now.
+  if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+    return EncodedInfo();
+  }
+
+  // Encode each channel separately.
+  RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  num_10ms_frames_buffered_ = 0;
+  const size_t samples_per_channel = SamplesPerChannel();
+  for (size_t i = 0; i < num_channels_; ++i) {
+    const size_t bytes_encoded = WebRtcG722_Encode(
+        encoders_[i].encoder, encoders_[i].speech_buffer.get(),
+        samples_per_channel, encoders_[i].encoded_buffer.data());
+    RTC_CHECK_EQ(bytes_encoded, samples_per_channel / 2);
+  }
+
+  const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;
+  EncodedInfo info;
+  info.encoded_bytes = encoded->AppendData(
+      bytes_to_encode, [&](rtc::ArrayView<uint8_t> encoded) {
+        // Interleave the encoded bytes of the different channels. Each separate
+        // channel and the interleaved stream encodes two samples per byte, most
+        // significant half first.
+        for (size_t i = 0; i < samples_per_channel / 2; ++i) {
+          for (size_t j = 0; j < num_channels_; ++j) {
+            uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
+            interleave_buffer_.data()[j] = two_samples >> 4;
+            interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
+          }
+          for (size_t j = 0; j < num_channels_; ++j)
+            encoded[i * num_channels_ + j] =
+                interleave_buffer_.data()[2 * j] << 4 |
+                interleave_buffer_.data()[2 * j + 1];
+        }
+
+        return bytes_to_encode;
+      });
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kG722;
+  return info;
+}
+
+AudioEncoderG722Impl::EncoderState::EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
+}
+
+AudioEncoderG722Impl::EncoderState::~EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
+}
+
+size_t AudioEncoderG722Impl::SamplesPerChannel() const {
+  return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
new file mode 100644
index 0000000000..a932aa8b7d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioEncoderG722Impl final : public AudioEncoder {
+ public:
+  AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
+  ~AudioEncoderG722Impl() override;
+
+  AudioEncoderG722Impl(const AudioEncoderG722Impl&) = delete;
+  AudioEncoderG722Impl& operator=(const AudioEncoderG722Impl&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  // The encoder state for one channel.
+  struct EncoderState {
+    G722EncInst* encoder;
+    std::unique_ptr<int16_t[]> speech_buffer;  // Queued up for encoding.
+    rtc::Buffer encoded_buffer;                // Already encoded.
+    EncoderState();
+    ~EncoderState();
+  };
+
+  size_t SamplesPerChannel() const;
+
+  const size_t num_channels_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
+  uint32_t first_timestamp_in_buffer_;
+  const std::unique_ptr<EncoderState[]> encoders_;
+  rtc::Buffer interleave_buffer_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.c b/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.c
new file mode 100644
index 0000000000..36ee6d92be
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.c
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/third_party/g722/g722_enc_dec.h"
+
+int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
+{
+    *G722enc_inst=(G722EncInst*)malloc(sizeof(G722EncoderState));
+    if (*G722enc_inst!=NULL) {
+      return(0);
+    } else {
+      return(-1);
+    }
+}
+
+int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
+{
+    // Create and/or reset the G.722 encoder
+    // Bitrate 64 kbps and wideband mode (2)
+    G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
+        (G722EncoderState*) G722enc_inst, 64000, 2);
+    if (G722enc_inst == NULL) {
+        return -1;
+    } else {
+        return 0;
+    }
+}
+
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+{
+    // Free encoder memory
+    return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
+}
+
+size_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded)
+{
+    unsigned char *codechar = (unsigned char*) encoded;
+    // Encode the input speech vector
+    return WebRtc_g722_encode((G722EncoderState*) G722enc_inst, codechar,
+                              speechIn, len);
+}
+
+int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
+{
+    *G722dec_inst=(G722DecInst*)malloc(sizeof(G722DecoderState));
+    if (*G722dec_inst!=NULL) {
+      return(0);
+    } else {
+      return(-1);
+    }
+}
+
+void WebRtcG722_DecoderInit(G722DecInst* inst) {
+  // Create and/or reset the G.722 decoder
+  // Bitrate 64 kbps and wideband mode (2)
+  WebRtc_g722_decode_init((G722DecoderState*)inst, 64000, 2);
+}
+
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+{
+    // Free encoder memory
+    return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
+}
+
+size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+                         const uint8_t *encoded,
+                         size_t len,
+                         int16_t *decoded,
+                         int16_t *speechType)
+{
+    // Decode the G.722 encoder stream
+    *speechType=G722_WEBRTC_SPEECH;
+    return WebRtc_g722_decode((G722DecoderState*) G722dec_inst, decoded,
+                              encoded, len);
+}
+
+int16_t WebRtcG722_Version(char *versionStr, short len)
+{
+    // Get version string
+    char version[30] = "2.0.0\n";
+    if (strlen(version) < (unsigned int)len)
+    {
+        strcpy(versionStr, version);
+        return 0;
+    }
+    else
+    {
+        return -1;
+    }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.h
new file mode 100644
index 0000000000..353de4504f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.h
@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*
+ * Solution to support multiple instances
+ */
+
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/*
+ * Comfort noise constants
+ */
+
+#define G722_WEBRTC_SPEECH 1
+#define G722_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcG722_CreateEncoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance for encoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+int16_t WebRtcG722_CreateEncoder(G722EncInst** G722enc_inst);
+
+/****************************************************************************
+ * WebRtcG722_EncoderInit(...)
+ *
+ * This function initializes a G722 instance
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance, i.e. the user that should receive
+ *                             be initialized
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+
+int16_t WebRtcG722_EncoderInit(G722EncInst* G722enc_inst);
+
+/****************************************************************************
+ * WebRtcG722_FreeEncoder(...)
+ *
+ * Free the memory used for G722 encoder
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance for encoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+int WebRtcG722_FreeEncoder(G722EncInst* G722enc_inst);
+
+/****************************************************************************
+ * WebRtcG722_Encode(...)
+ *
+ * This function encodes G722 encoded data.
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance, i.e. the user that should encode
+ *                              a packet
+ *     - speechIn             : Input speech vector
+ *     - len                  : Samples in speechIn
+ *
+ * Output:
+ *        - encoded           : The encoded data vector
+ *
+ * Return value               : Length (in bytes) of coded data
+ */
+
+size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG722_CreateDecoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ *     - G722dec_inst         : G722 instance for decoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+int16_t WebRtcG722_CreateDecoder(G722DecInst** G722dec_inst);
+
+/****************************************************************************
+ * WebRtcG722_DecoderInit(...)
+ *
+ * This function initializes a G722 instance
+ *
+ * Input:
+ *     - inst      : G722 instance
+ */
+
+void WebRtcG722_DecoderInit(G722DecInst* inst);
+
+/****************************************************************************
+ * WebRtcG722_FreeDecoder(...)
+ *
+ * Free the memory used for G722 decoder
+ *
+ * Input:
+ *     - G722dec_inst         : G722 instance for decoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+
+int WebRtcG722_FreeDecoder(G722DecInst* G722dec_inst);
+
+/****************************************************************************
+ * WebRtcG722_Decode(...)
+ *
+ * This function decodes a packet with G729 frame(s). Output speech length
+ * will be a multiple of 80 samples (80*frames/packet).
+ *
+ * Input:
+ *     - G722dec_inst       : G722 instance, i.e. the user that should decode
+ *                            a packet
+ *     - encoded            : Encoded G722 frame(s)
+ *     - len                : Bytes in encoded vector
+ *
+ * Output:
+ *        - decoded         : The decoded vector
+ *      - speechType        : 1 normal, 2 CNG (Since G722 does not have its own
+ *                            DTX/CNG scheme it should always return 1)
+ *
+ * Return value             : Samples in decoded vector
+ */
+
+size_t WebRtcG722_Decode(G722DecInst* G722dec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcG722_Version(...)
+ *
+ * Get a string with the current version of the codec
+ */
+
+int16_t WebRtcG722_Version(char* versionStr, short len);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/test/testG722.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g722/test/testG722.cc
new file mode 100644
index 0000000000..9f2155d0f7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/test/testG722.cc
@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * testG722.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+/* include API */
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G722 100000
+
+// Forward declaration
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/* function for reading audio data from PCM file */
+bool readframe(int16_t* data, FILE* inp, size_t length) {
+  size_t rlen = fread(data, sizeof(int16_t), length, inp);
+  if (rlen >= length)
+    return false;
+  memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+  return true;
+}
+
+int main(int argc, char* argv[]) {
+  char inname[60], outbit[40], outname[40];
+  FILE *inp, *outbitp, *outp;
+
+  int framecnt;
+  bool endfile;
+  size_t framelength = 160;
+  G722EncInst* G722enc_inst;
+  G722DecInst* G722dec_inst;
+
+  /* Runtime statistics */
+  double starttime;
+  double runtime = 0;
+  double length_file;
+
+  size_t stream_len = 0;
+  int16_t shortdata[960];
+  int16_t decoded[960];
+  uint8_t streamdata[80 * 6];
+  int16_t speechType[1];
+
+  /* handling wrong input arguments in the command line */
+  if (argc != 5) {
+    printf("\n\nWrong number of arguments or flag values.\n\n");
+
+    printf("\n");
+    printf("Usage:\n\n");
+    printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
+    printf("with:\n");
+    printf("framelength  :    Framelength in samples.\n\n");
+    printf("infile       :    Normal speech input file\n\n");
+    printf("outbitfile   :    Bitstream output file\n\n");
+    printf("outspeechfile:    Speech output file\n\n");
+    exit(0);
+  }
+
+  /* Get frame length */
+  int framelength_int = atoi(argv[1]);
+  if (framelength_int < 0) {
+    printf("  G.722: Invalid framelength %d.\n", framelength_int);
+    exit(1);
+  }
+  framelength = static_cast<size_t>(framelength_int);
+
+  /* Get Input and Output files */
+  sscanf(argv[2], "%s", inname);
+  sscanf(argv[3], "%s", outbit);
+  sscanf(argv[4], "%s", outname);
+
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  G.722: Cannot read file %s.\n", inname);
+    exit(1);
+  }
+  if ((outbitp = fopen(outbit, "wb")) == NULL) {
+    printf("  G.722: Cannot write file %s.\n", outbit);
+    exit(1);
+  }
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  G.722: Cannot write file %s.\n", outname);
+    exit(1);
+  }
+  printf("\nInput:%s\nOutput bitstream:%s\nOutput:%s\n", inname, outbit,
+         outname);
+
+  /* Create and init */
+  WebRtcG722_CreateEncoder((G722EncInst**)&G722enc_inst);
+  WebRtcG722_CreateDecoder((G722DecInst**)&G722dec_inst);
+  WebRtcG722_EncoderInit((G722EncInst*)G722enc_inst);
+  WebRtcG722_DecoderInit((G722DecInst*)G722dec_inst);
+
+  /* Initialize encoder and decoder */
+  framecnt = 0;
+  endfile = false;
+  while (!endfile) {
+    framecnt++;
+
+    /* Read speech block */
+    endfile = readframe(shortdata, inp, framelength);
+
+    /* Start clock before call to encoder and decoder */
+    starttime = clock() / (double)CLOCKS_PER_SEC_G722;
+
+    /* G.722 encoding + decoding */
+    stream_len = WebRtcG722_Encode((G722EncInst*)G722enc_inst, shortdata,
+                                   framelength, streamdata);
+    WebRtcG722_Decode(G722dec_inst, streamdata, stream_len, decoded,
+                      speechType);
+
+    /* Stop clock after call to encoder and decoder */
+    runtime += (double)((clock() / (double)CLOCKS_PER_SEC_G722) - starttime);
+
+    /* Write coded bits to file */
+    if (fwrite(streamdata, sizeof(short), stream_len / 2, outbitp) !=
+        stream_len / 2) {
+      return -1;
+    }
+    /* Write coded speech to file */
+    if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) {
+      return -1;
+    }
+  }
+
+  WebRtcG722_FreeEncoder((G722EncInst*)G722enc_inst);
+  WebRtcG722_FreeDecoder((G722DecInst*)G722dec_inst);
+
+  length_file = ((double)framecnt * (double)framelength / 16000);
+  printf("\n\nLength of speech file: %.1f s\n", length_file);
+  printf("Time to run G.722:      %.2f s (%.2f %% of realtime)\n\n", runtime,
+         (100 * runtime / length_file));
+  printf("---------------------END----------------------\n");
+
+  fclose(inp);
+  fclose(outbitp);
+  fclose(outp);
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
new file mode 100644
index 0000000000..77da78ba7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant.h"
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant_loop.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+                                   and idxVec, uses state_first as
+                                   input) */
+    int16_t *in,     /* (i) vector to encode */
+    int16_t *weightDenum   /* (i) denominator of synthesis filter */
+                            ) {
+  int16_t *syntOut;
+  size_t quantLen[2];
+
+  /* Stack based */
+  int16_t syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
+  int16_t in_weightedVec[STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t *in_weighted = &in_weightedVec[LPC_FILTERORDER];
+
+  /* Initialize the buffers */
+  WebRtcSpl_MemSetW16(syntOutBuf, 0, LPC_FILTERORDER+STATE_SHORT_LEN_30MS);
+  syntOut = &syntOutBuf[LPC_FILTERORDER];
+  /* Start with zero state */
+  WebRtcSpl_MemSetW16(in_weightedVec, 0, LPC_FILTERORDER);
+
+  /* Perform the quantization loop in two sections of length quantLen[i],
+     where the perceptual weighting filter is updated at the subframe
+     border */
+
+  if (iLBC_encbits->state_first) {
+    quantLen[0]=SUBL;
+    quantLen[1]=iLBCenc_inst->state_short_len-SUBL;
+  } else {
+    quantLen[0]=iLBCenc_inst->state_short_len-SUBL;
+    quantLen[1]=SUBL;
+  }
+
+  /* Calculate the weighted residual, switch perceptual weighting
+     filter at the subframe border */
+  WebRtcSpl_FilterARFastQ12(
+      in, in_weighted,
+      weightDenum, LPC_FILTERORDER+1, quantLen[0]);
+  WebRtcSpl_FilterARFastQ12(
+      &in[quantLen[0]], &in_weighted[quantLen[0]],
+      &weightDenum[LPC_FILTERORDER+1], LPC_FILTERORDER+1, quantLen[1]);
+
+  WebRtcIlbcfix_AbsQuantLoop(
+      syntOut,
+      in_weighted,
+      weightDenum,
+      quantLen,
+      iLBC_encbits->idxVec);
+
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
new file mode 100644
index 0000000000..c72e29cf29
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits* iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+                                   and idxVec, uses state_first as
+                                   input) */
+    int16_t* in,             /* (i) vector to encode */
+    int16_t* weightDenum     /* (i) denominator of synthesis filter */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
new file mode 100644
index 0000000000..cf9266299d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant_loop.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/sort_sq.h"
+
+void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
+                                int16_t *weightDenumIN, size_t *quantLenIN,
+                                int16_t *idxVecIN ) {
+  size_t k1, k2;
+  int16_t index;
+  int32_t toQW32;
+  int32_t toQ32;
+  int16_t tmp16a;
+  int16_t xq;
+
+  int16_t *syntOut   = syntOutIN;
+  int16_t *in_weighted  = in_weightedIN;
+  int16_t *weightDenum  = weightDenumIN;
+  size_t *quantLen  = quantLenIN;
+  int16_t *idxVec   = idxVecIN;
+
+  for(k1=0;k1<2;k1++) {
+    for(k2=0;k2<quantLen[k1];k2++){
+
+      /* Filter to get the predicted value */
+      WebRtcSpl_FilterARFastQ12(
+          syntOut, syntOut,
+          weightDenum, LPC_FILTERORDER+1, 1);
+
+      /* the quantizer */
+      toQW32 = (int32_t)(*in_weighted) - (int32_t)(*syntOut);
+
+      toQ32 = (((int32_t)toQW32)<<2);
+
+      if (toQ32 > 32767) {
+        toQ32 = (int32_t) 32767;
+      } else if (toQ32 < -32768) {
+        toQ32 = (int32_t) -32768;
+      }
+
+      /* Quantize the state */
+      if (toQW32<(-7577)) {
+        /* To prevent negative overflow */
+        index=0;
+      } else if (toQW32>8151) {
+        /* To prevent positive overflow */
+        index=7;
+      } else {
+        /* Find the best quantization index
+           (state_sq3Tbl is in Q13 and toQ is in Q11)
+        */
+        WebRtcIlbcfix_SortSq(&xq, &index,
+                             (int16_t)toQ32,
+                             WebRtcIlbcfix_kStateSq3, 8);
+      }
+
+      /* Store selected index */
+      (*idxVec++) = index;
+
+      /* Compute decoded sample and update of the prediction filter */
+      tmp16a = ((WebRtcIlbcfix_kStateSq3[index] + 2 ) >> 2);
+
+      *syntOut     = (int16_t) (tmp16a + (int32_t)(*in_weighted) - toQW32);
+
+      syntOut++; in_weighted++;
+    }
+    /* Update perceptual weighting filter at subframe border */
+    weightDenum += 11;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
new file mode 100644
index 0000000000..841d73b9fb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuantLoop(int16_t* syntOutIN,
+                                int16_t* in_weightedIN,
+                                int16_t* weightDenumIN,
+                                size_t* quantLenIN,
+                                int16_t* idxVecIN);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
new file mode 100644
index 0000000000..57b5abbe23
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+AudioDecoderIlbcImpl::AudioDecoderIlbcImpl() {
+  WebRtcIlbcfix_DecoderCreate(&dec_state_);
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+AudioDecoderIlbcImpl::~AudioDecoderIlbcImpl() {
+  WebRtcIlbcfix_DecoderFree(dec_state_);
+}
+
+bool AudioDecoderIlbcImpl::HasDecodePlc() const {
+  return true;
+}
+
+int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+size_t AudioDecoderIlbcImpl::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
+}
+
+void AudioDecoderIlbcImpl::Reset() {
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+  size_t bytes_per_frame;
+  int timestamps_per_frame;
+  if (payload.size() >= 950) {
+    RTC_LOG(LS_WARNING)
+        << "AudioDecoderIlbcImpl::ParsePayload: Payload too large";
+    return results;
+  }
+  if (payload.size() % 38 == 0) {
+    // 20 ms frames.
+    bytes_per_frame = 38;
+    timestamps_per_frame = 160;
+  } else if (payload.size() % 50 == 0) {
+    // 30 ms frames.
+    bytes_per_frame = 50;
+    timestamps_per_frame = 240;
+  } else {
+    RTC_LOG(LS_WARNING)
+        << "AudioDecoderIlbcImpl::ParsePayload: Invalid payload";
+    return results;
+  }
+
+  RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame);
+  if (payload.size() == bytes_per_frame) {
+    std::unique_ptr<EncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(this, std::move(payload)));
+    results.emplace_back(timestamp, 0, std::move(frame));
+  } else {
+    size_t byte_offset;
+    uint32_t timestamp_offset;
+    for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
+         byte_offset += bytes_per_frame,
+        timestamp_offset += timestamps_per_frame) {
+      std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
+          this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
+      results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
+    }
+  }
+
+  return results;
+}
+
+int AudioDecoderIlbcImpl::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderIlbcImpl::Channels() const {
+  return 1;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
new file mode 100644
index 0000000000..46ba755148
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+
+typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
+
+namespace webrtc {
+
+class AudioDecoderIlbcImpl final : public AudioDecoder {
+ public:
+  AudioDecoderIlbcImpl();
+  ~AudioDecoderIlbcImpl() override;
+
+  AudioDecoderIlbcImpl(const AudioDecoderIlbcImpl&) = delete;
+  AudioDecoderIlbcImpl& operator=(const AudioDecoderIlbcImpl&) = delete;
+
+  bool HasDecodePlc() const override;
+  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  IlbcDecoderInstance* dec_state_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
new file mode 100644
index 0000000000..9fbf42ceeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+
+#include <algorithm>
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kSampleRateHz = 8000;
+
+int GetIlbcBitrate(int ptime) {
+  switch (ptime) {
+    case 20:
+    case 40:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 30:
+    case 60:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
+}  // namespace
+
+AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
+                                           int payload_type)
+    : frame_size_ms_(config.frame_size_ms),
+      payload_type_(payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      encoder_(nullptr) {
+  RTC_CHECK(config.IsOk());
+  Reset();
+}
+
+AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+}
+
+int AudioEncoderIlbcImpl::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+size_t AudioEncoderIlbcImpl::NumChannels() const {
+  return 1;
+}
+
+size_t AudioEncoderIlbcImpl::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderIlbcImpl::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderIlbcImpl::GetTargetBitrate() const {
+  return GetIlbcBitrate(rtc::dchecked_cast<int>(num_10ms_frames_per_packet_) *
+                        10);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  // Save timestamp if starting a new packet.
+  if (num_10ms_frames_buffered_ == 0)
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  // Buffer input.
+  std::copy(audio.cbegin(), audio.cend(),
+            input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
+
+  // If we don't yet have enough buffered input for a whole packet, we're done
+  // for now.
+  if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+    return EncodedInfo();
+  }
+
+  // Encode buffered input.
+  RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  num_10ms_frames_buffered_ = 0;
+  size_t encoded_bytes = encoded->AppendData(
+      RequiredOutputSizeBytes(), [&](rtc::ArrayView<uint8_t> encoded) {
+        const int r = WebRtcIlbcfix_Encode(
+            encoder_, input_buffer_,
+            kSampleRateHz / 100 * num_10ms_frames_per_packet_, encoded.data());
+        RTC_CHECK_GE(r, 0);
+
+        return static_cast<size_t>(r);
+      });
+
+  RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
+
+  EncodedInfo info;
+  info.encoded_bytes = encoded_bytes;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kIlbc;
+  return info;
+}
+
+void AudioEncoderIlbcImpl::Reset() {
+  if (encoder_)
+    RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
+  const int encoder_frame_size_ms =
+      frame_size_ms_ > 30 ? frame_size_ms_ / 2 : frame_size_ms_;
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+  num_10ms_frames_buffered_ = 0;
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderIlbcImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
+  switch (num_10ms_frames_per_packet_) {
+    case 2:
+      return 38;
+    case 3:
+      return 50;
+    case 4:
+      return 2 * 38;
+    case 6:
+      return 2 * 50;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
new file mode 100644
index 0000000000..c8dfa2ca6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+namespace webrtc {
+
+class AudioEncoderIlbcImpl final : public AudioEncoder {
+ public:
+  AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
+  ~AudioEncoderIlbcImpl() override;
+
+  AudioEncoderIlbcImpl(const AudioEncoderIlbcImpl&) = delete;
+  AudioEncoderIlbcImpl& operator=(const AudioEncoderIlbcImpl&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ private:
+  size_t RequiredOutputSizeBytes() const;
+
+  static constexpr size_t kMaxSamplesPerPacket = 480;
+  const int frame_size_ms_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
+  uint32_t first_timestamp_in_buffer_;
+  int16_t input_buffer_[kMaxSamplesPerPacket];
+  IlbcEncoderInstance* encoder_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
new file mode 100644
index 0000000000..c915a2f9f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/augmented_cb_corr.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+    int16_t *target,   /* (i) Target vector */
+    int16_t *buffer,   /* (i) Memory buffer */
+    int16_t *interpSamples, /* (i) buffer with
+                                     interpolated samples */
+    int32_t *crossDot,  /* (o) The cross correlation between
+                                 the target and the Augmented
+                                 vector */
+    size_t low,    /* (i) Lag to start from (typically
+                             20) */
+    size_t high,   /* (i) Lag to end at (typically 39) */
+    int scale)   /* (i) Scale factor to use for
+                              the crossDot */
+{
+  size_t lagcount;
+  size_t ilow;
+  int16_t *targetPtr;
+  int32_t *crossDotPtr;
+  int16_t *iSPtr=interpSamples;
+
+  /* Calculate the correlation between the target and the
+     interpolated codebook. The correlation is calculated in
+     3 sections with the interpolated part in the middle */
+  crossDotPtr=crossDot;
+  for (lagcount=low; lagcount<=high; lagcount++) {
+
+    ilow = lagcount - 4;
+
+    /* Compute dot product for the first (lagcount-4) samples */
+    (*crossDotPtr) = WebRtcSpl_DotProductWithScale(target, buffer-lagcount, ilow, scale);
+
+    /* Compute dot product on the interpolated samples */
+    (*crossDotPtr) += WebRtcSpl_DotProductWithScale(target+ilow, iSPtr, 4, scale);
+    targetPtr = target + lagcount;
+    iSPtr += lagcount-ilow;
+
+    /* Compute dot product for the remaining samples */
+    (*crossDotPtr) += WebRtcSpl_DotProductWithScale(targetPtr, buffer-lagcount, SUBL-lagcount, scale);
+    crossDotPtr++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
new file mode 100644
index 0000000000..2e9612e51a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Calculate correlation between target and Augmented codebooks
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+    int16_t* target,        /* (i) Target vector */
+    int16_t* buffer,        /* (i) Memory buffer */
+    int16_t* interpSamples, /* (i) buffer with
+                                           interpolated samples */
+    int32_t* crossDot,      /* (o) The cross correlation between
+                                               the target and the Augmented
+                                               vector */
+    size_t low,             /* (i) Lag to start from (typically
+                                                            20) */
+    size_t high,            /* (i) Lag to end at (typically 39 */
+    int scale);             /* (i) Scale factor to use for the crossDot */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c
new file mode 100644
index 0000000000..1a9b882adf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+/* The output is in the same domain as the input */
+void WebRtcIlbcfix_BwExpand(
+    int16_t *out, /* (o) the bandwidth expanded lpc coefficients */
+    int16_t *in,  /* (i) the lpc coefficients before bandwidth
+                                   expansion */
+    int16_t *coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
+                            ) {
+  int i;
+
+  out[0] = in[0];
+  for (i = 1; i < length; i++) {
+    /* out[i] = coef[i] * in[i] with rounding.
+       in[] and out[] are in Q12 and coef[] is in Q15
+    */
+    out[i] = (int16_t)((coef[i] * in[i] + 16384) >> 15);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.h
new file mode 100644
index 0000000000..ff9b0b302e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_BwExpand(
+    int16_t* out,  /* (o) the bandwidth expanded lpc coefficients */
+    int16_t* in,   /* (i) the lpc coefficients before bandwidth
+                                    expansion */
+    int16_t* coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
new file mode 100644
index 0000000000..1e9a7040c7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/gain_dequant.h"
+#include "modules/audio_coding/codecs/ilbc/get_cd_vec.h"
+#include "rtc_base/sanitizer.h"
+
+// An arithmetic operation that is allowed to overflow. (It's still undefined
+// behavior, so not a good idea; this just makes UBSan ignore the violation, so
+// that our old code can continue to do what it's always been doing.)
+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
+    OverflowingAddS32S32ToS32(int32_t a, int32_t b) {
+  return a + b;
+}
+
+/*----------------------------------------------------------------*
+ *  Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_CbConstruct(
+    int16_t* decvector,        /* (o) Decoded vector */
+    const int16_t* index,      /* (i) Codebook indices */
+    const int16_t* gain_index, /* (i) Gain quantization indices */
+    int16_t* mem,              /* (i) Buffer for codevector construction */
+    size_t lMem,               /* (i) Length of buffer */
+    size_t veclen) {           /* (i) Length of vector */
+  size_t j;
+  int16_t gain[CB_NSTAGES];
+  /* Stack based */
+  int16_t cbvec0[SUBL];
+  int16_t cbvec1[SUBL];
+  int16_t cbvec2[SUBL];
+  int32_t a32;
+  int16_t *gainPtr;
+
+  /* gain de-quantization */
+
+  gain[0] = WebRtcIlbcfix_GainDequant(gain_index[0], 16384, 0);
+  gain[1] = WebRtcIlbcfix_GainDequant(gain_index[1], gain[0], 1);
+  gain[2] = WebRtcIlbcfix_GainDequant(gain_index[2], gain[1], 2);
+
+  /* codebook vector construction and construction of total vector */
+
+  /* Stack based */
+  if (!WebRtcIlbcfix_GetCbVec(cbvec0, mem, (size_t)index[0], lMem, veclen))
+    return false;  // Failure.
+  if (!WebRtcIlbcfix_GetCbVec(cbvec1, mem, (size_t)index[1], lMem, veclen))
+    return false;  // Failure.
+  if (!WebRtcIlbcfix_GetCbVec(cbvec2, mem, (size_t)index[2], lMem, veclen))
+    return false;  // Failure.
+
+  gainPtr = &gain[0];
+  for (j=0;j<veclen;j++) {
+    a32 = (*gainPtr++) * cbvec0[j];
+    a32 += (*gainPtr++) * cbvec1[j];
+    a32 = OverflowingAddS32S32ToS32(a32, (*gainPtr) * cbvec2[j]);
+    gainPtr -= 2;
+    decvector[j] = (int16_t)((a32 + 8192) >> 14);
+  }
+
+  return true;  // Success.
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
new file mode 100644
index 0000000000..8f7c663164
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+
+#include <stdbool.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+// Returns true on success, false on failure.
+ABSL_MUST_USE_RESULT
+bool WebRtcIlbcfix_CbConstruct(
+    int16_t* decvector,        /* (o) Decoded vector */
+    const int16_t* index,      /* (i) Codebook indices */
+    const int16_t* gain_index, /* (i) Gain quantization indices */
+    int16_t* mem,              /* (i) Buffer for codevector construction */
+    size_t lMem,               /* (i) Length of buffer */
+    size_t veclen              /* (i) Length of vector */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
new file mode 100644
index 0000000000..21e4197607
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergy.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy.h"
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Function WebRtcIlbcfix_CbMemEnergy computes the energy of all
+ * the vectors in the codebook memory that will be used in the
+ * following search for the best match.
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbMemEnergy(
+    size_t range,
+    int16_t *CB,   /* (i) The CB memory (1:st section) */
+    int16_t *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
+    size_t lMem,   /* (i) Length of the CB memory */
+    size_t lTarget,   /* (i) Length of the target vector */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
+                               ) {
+  int16_t *ppi, *ppo, *pp;
+  int32_t energy, tmp32;
+
+  /* Compute the energy and store it in a vector. Also the
+   * corresponding shift values are stored. The energy values
+   * are reused in all three stages. */
+
+  /* Calculate the energy in the first block of 'lTarget' sampels. */
+  ppi = CB+lMem-lTarget-1;
+  ppo = CB+lMem-1;
+
+  pp=CB+lMem-lTarget;
+  energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+  /* Normalize the energy and store the number of shifts */
+  energyShifts[0] = (int16_t)WebRtcSpl_NormW32(energy);
+  tmp32 = energy << energyShifts[0];
+  energyW16[0] = (int16_t)(tmp32 >> 16);
+
+  /* Compute the energy of the rest of the cb memory
+   * by step wise adding and subtracting the next
+   * sample and the last sample respectively. */
+  WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, 0);
+
+  /* Next, precompute the energy values for the filtered cb section */
+  energy=0;
+  pp=filteredCB+lMem-lTarget;
+
+  energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+  /* Normalize the energy and store the number of shifts */
+  energyShifts[base_size] = (int16_t)WebRtcSpl_NormW32(energy);
+  tmp32 = energy << energyShifts[base_size];
+  energyW16[base_size] = (int16_t)(tmp32 >> 16);
+
+  ppi = filteredCB + lMem - 1 - lTarget;
+  ppo = filteredCB + lMem - 1;
+
+  WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, base_size);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
new file mode 100644
index 0000000000..17ec337dc6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergy.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbMemEnergy(
+    size_t range,
+    int16_t* CB,           /* (i) The CB memory (1:st section) */
+    int16_t* filteredCB,   /* (i) The filtered CB memory (2:nd section) */
+    size_t lMem,           /* (i) Length of the CB memory */
+    size_t lTarget,        /* (i) Length of the target vector */
+    int16_t* energyW16,    /* (o) Energy in the CB vectors */
+    int16_t* energyShifts, /* (o) Shift value of the energy */
+    int scale,             /* (i) The scaling of all energy values */
+    size_t base_size /* (i) Index to where energy values should be stored */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
new file mode 100644
index 0000000000..0619bbe422
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyAugmentation.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbMemEnergyAugmentation(
+    int16_t *interpSamples, /* (i) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size,  /* (i) Index to where energy values should be stored */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts /* (o) Shift value of the energy */
+                                           ){
+  int32_t energy, tmp32;
+  int16_t *ppe, *pp, *interpSamplesPtr;
+  int16_t *CBmemPtr;
+  size_t lagcount;
+  int16_t *enPtr=&energyW16[base_size-20];
+  int16_t *enShPtr=&energyShifts[base_size-20];
+  int32_t nrjRecursive;
+
+  CBmemPtr = CBmem+147;
+  interpSamplesPtr = interpSamples;
+
+  /* Compute the energy for the first (low-5) noninterpolated samples */
+  nrjRecursive = WebRtcSpl_DotProductWithScale( CBmemPtr-19, CBmemPtr-19, 15, scale);
+  ppe = CBmemPtr - 20;
+
+  for (lagcount=20; lagcount<=39; lagcount++) {
+
+    /* Update the energy recursively to save complexity */
+    nrjRecursive += (*ppe * *ppe) >> scale;
+    ppe--;
+    energy = nrjRecursive;
+
+    /* interpolation */
+    energy += WebRtcSpl_DotProductWithScale(interpSamplesPtr, interpSamplesPtr, 4, scale);
+    interpSamplesPtr += 4;
+
+    /* Compute energy for the remaining samples */
+    pp = CBmemPtr - lagcount;
+    energy += WebRtcSpl_DotProductWithScale(pp, pp, SUBL-lagcount, scale);
+
+    /* Normalize the energy and store the number of shifts */
+    (*enShPtr) = (int16_t)WebRtcSpl_NormW32(energy);
+    tmp32 = energy << *enShPtr;
+    *enPtr = (int16_t)(tmp32 >> 16);
+    enShPtr++;
+    enPtr++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
new file mode 100644
index 0000000000..d7b7a0d97e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyAugmentation.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbMemEnergyAugmentation(
+    int16_t* interpSamples, /* (i) The interpolated samples */
+    int16_t* CBmem,         /* (i) The CB memory */
+    int scale,              /* (i) The scaling of all energy values */
+    size_t base_size,   /* (i) Index to where energy values should be stored */
+    int16_t* energyW16, /* (o) Energy in the CB vectors */
+    int16_t* energyShifts /* (o) Shift value of the energy */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
new file mode 100644
index 0000000000..58c0c5fe6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyCalc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* Compute the energy of the rest of the cb memory
+ * by step wise adding and subtracting the next
+ * sample and the last sample respectively */
+void WebRtcIlbcfix_CbMemEnergyCalc(
+    int32_t energy,   /* (i) input start energy */
+    size_t range,   /* (i) number of iterations */
+    int16_t *ppi,   /* (i) input pointer 1 */
+    int16_t *ppo,   /* (i) input pointer 2 */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
+                                   )
+{
+  size_t j;
+  int16_t shft;
+  int32_t tmp;
+  int16_t *eSh_ptr;
+  int16_t *eW16_ptr;
+
+
+  eSh_ptr  = &energyShifts[1+base_size];
+  eW16_ptr = &energyW16[1+base_size];
+
+  for (j = 0; j + 1 < range; j++) {
+
+    /* Calculate next energy by a +/-
+       operation on the edge samples */
+    tmp = (*ppi) * (*ppi) - (*ppo) * (*ppo);
+    energy += tmp >> scale;
+    energy = WEBRTC_SPL_MAX(energy, 0);
+
+    ppi--;
+    ppo--;
+
+    /* Normalize the energy into a int16_t and store
+       the number of shifts */
+
+    shft = (int16_t)WebRtcSpl_NormW32(energy);
+    *eSh_ptr++ = shft;
+
+    tmp = energy << shft;
+    *eW16_ptr++ = (int16_t)(tmp >> 16);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
new file mode 100644
index 0000000000..1d1e8d62b9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyCalc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbMemEnergyCalc(
+    int32_t energy,        /* (i) input start energy */
+    size_t range,          /* (i) number of iterations */
+    int16_t* ppi,          /* (i) input pointer 1 */
+    int16_t* ppo,          /* (i) input pointer 2 */
+    int16_t* energyW16,    /* (o) Energy in the CB vectors */
+    int16_t* energyShifts, /* (o) Shift value of the energy */
+    int scale,             /* (i) The scaling of all energy values */
+    size_t base_size /* (i) Index to where energy values should be stored */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c
new file mode 100644
index 0000000000..24b5292354
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -0,0 +1,405 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearch.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_search.h"
+
+#include "modules/audio_coding/codecs/ilbc/augmented_cb_corr.h"
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy.h"
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h"
+#include "modules/audio_coding/codecs/ilbc/cb_search_core.h"
+#include "modules/audio_coding/codecs/ilbc/cb_update_best_index.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/energy_inverse.h"
+#include "modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h"
+#include "modules/audio_coding/codecs/ilbc/gain_quant.h"
+#include "modules/audio_coding/codecs/ilbc/interpolate_samples.h"
+
+/*----------------------------------------------------------------*
+ *  Search routine for codebook encoding and gain quantization.
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbSearch(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) the encoder state structure */
+    int16_t *index,  /* (o) Codebook indices */
+    int16_t *gain_index, /* (o) Gain quantization indices */
+    int16_t *intarget, /* (i) Target vector for encoding */
+    int16_t *decResidual,/* (i) Decoded residual for codebook construction */
+    size_t lMem,  /* (i) Length of buffer */
+    size_t lTarget,  /* (i) Length of vector */
+    int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */
+    size_t block  /* (i) the subblock number */
+                            ) {
+  size_t i, range;
+  int16_t ii, j, stage;
+  int16_t *pp;
+  int16_t tmp;
+  int scale;
+  int16_t bits, temp1, temp2;
+  size_t base_size;
+  int32_t codedEner, targetEner;
+  int16_t gains[CB_NSTAGES+1];
+  int16_t *cb_vecPtr;
+  size_t indexOffset, sInd, eInd;
+  int32_t CritMax=0;
+  int16_t shTotMax=WEBRTC_SPL_WORD16_MIN;
+  size_t bestIndex=0;
+  int16_t bestGain=0;
+  size_t indexNew;
+  int16_t CritNewSh;
+  int32_t CritNew;
+  int32_t *cDotPtr;
+  size_t noOfZeros;
+  int16_t *gainPtr;
+  int32_t t32, tmpW32;
+  int16_t *WebRtcIlbcfix_kGainSq5_ptr;
+  /* Stack based */
+  int16_t CBbuf[CB_MEML+LPC_FILTERORDER+CB_HALFFILTERLEN];
+  int32_t cDot[128];
+  int32_t Crit[128];
+  int16_t targetVec[SUBL+LPC_FILTERORDER];
+  int16_t cbvectors[CB_MEML + 1];  /* Adding one extra position for
+                                            Coverity warnings. */
+  int16_t codedVec[SUBL];
+  int16_t interpSamples[20*4];
+  int16_t interpSamplesFilt[20*4];
+  int16_t energyW16[CB_EXPAND*128];
+  int16_t energyShifts[CB_EXPAND*128];
+  int16_t *inverseEnergy=energyW16;   /* Reuse memory */
+  int16_t *inverseEnergyShifts=energyShifts; /* Reuse memory */
+  int16_t *buf = &CBbuf[LPC_FILTERORDER];
+  int16_t *target = &targetVec[LPC_FILTERORDER];
+  int16_t *aug_vec = (int16_t*)cDot;   /* length [SUBL], reuse memory */
+
+  /* Determine size of codebook sections */
+
+  base_size=lMem-lTarget+1;
+  if (lTarget==SUBL) {
+    base_size=lMem-19;
+  }
+
+  /* weighting of the CB memory */
+  noOfZeros=lMem-WebRtcIlbcfix_kFilterRange[block];
+  WebRtcSpl_MemSetW16(&buf[-LPC_FILTERORDER], 0, noOfZeros+LPC_FILTERORDER);
+  WebRtcSpl_FilterARFastQ12(
+      decResidual+noOfZeros, buf+noOfZeros,
+      weightDenum, LPC_FILTERORDER+1, WebRtcIlbcfix_kFilterRange[block]);
+
+  /* weighting of the target vector */
+  WEBRTC_SPL_MEMCPY_W16(&target[-LPC_FILTERORDER], buf+noOfZeros+WebRtcIlbcfix_kFilterRange[block]-LPC_FILTERORDER, LPC_FILTERORDER);
+  WebRtcSpl_FilterARFastQ12(
+      intarget, target,
+      weightDenum, LPC_FILTERORDER+1, lTarget);
+
+  /* Store target, towards the end codedVec is calculated as
+     the initial target minus the remaining target */
+  WEBRTC_SPL_MEMCPY_W16(codedVec, target, lTarget);
+
+  /* Find the highest absolute value to calculate proper
+     vector scale factor (so that it uses 12 bits) */
+  temp1 = WebRtcSpl_MaxAbsValueW16(buf, lMem);
+  temp2 = WebRtcSpl_MaxAbsValueW16(target, lTarget);
+
+  if ((temp1>0)&&(temp2>0)) {
+    temp1 = WEBRTC_SPL_MAX(temp1, temp2);
+    scale = WebRtcSpl_GetSizeInBits((uint32_t)(temp1 * temp1));
+  } else {
+    /* temp1 or temp2 is negative (maximum was -32768) */
+    scale = 30;
+  }
+
+  /* Scale to so that a mul-add 40 times does not overflow */
+  scale = scale - 25;
+  scale = WEBRTC_SPL_MAX(0, scale);
+
+  /* Compute energy of the original target */
+  targetEner = WebRtcSpl_DotProductWithScale(target, target, lTarget, scale);
+
+  /* Prepare search over one more codebook section. This section
+     is created by filtering the original buffer with a filter. */
+  WebRtcIlbcfix_FilteredCbVecs(cbvectors, buf, lMem, WebRtcIlbcfix_kFilterRange[block]);
+
+  range = WebRtcIlbcfix_kSearchRange[block][0];
+
+  if(lTarget == SUBL) {
+    /* Create the interpolated samples and store them for use in all stages */
+
+    /* First section, non-filtered half of the cb */
+    WebRtcIlbcfix_InterpolateSamples(interpSamples, buf, lMem);
+
+    /* Second section, filtered half of the cb */
+    WebRtcIlbcfix_InterpolateSamples(interpSamplesFilt, cbvectors, lMem);
+
+    /* Compute the CB vectors' energies for the first cb section (non-filtered) */
+    WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamples, buf,
+                                          scale, 20, energyW16, energyShifts);
+
+    /* Compute the CB vectors' energies for the second cb section (filtered cb) */
+    WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamplesFilt, cbvectors, scale,
+                                          base_size + 20, energyW16,
+                                          energyShifts);
+
+    /* Compute the CB vectors' energies and store them in the vector
+     * energyW16. Also the corresponding shift values are stored. The
+     * energy values are used in all three stages. */
+    WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
+                              lTarget, energyW16+20, energyShifts+20, scale, base_size);
+
+  } else {
+    /* Compute the CB vectors' energies and store them in the vector
+     * energyW16. Also the corresponding shift values are stored. The
+     * energy values are used in all three stages. */
+    WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
+                              lTarget, energyW16, energyShifts, scale, base_size);
+
+    /* Set the energy positions 58-63 and 122-127 to zero
+       (otherwise they are uninitialized) */
+    WebRtcSpl_MemSetW16(energyW16+range, 0, (base_size-range));
+    WebRtcSpl_MemSetW16(energyW16+range+base_size, 0, (base_size-range));
+  }
+
+  /* Calculate Inverse Energy (energyW16 is already normalized
+     and will contain the inverse energy in Q29 after this call */
+  WebRtcIlbcfix_EnergyInverse(energyW16, base_size*CB_EXPAND);
+
+  /* The gain value computed in the previous stage is used
+   * as an upper limit to what the next stage gain value
+   * is allowed to be. In stage 0, 16384 (1.0 in Q14) is used as
+   * the upper limit. */
+  gains[0] = 16384;
+
+  for (stage=0; stage<CB_NSTAGES; stage++) {
+
+    /* Set up memories */
+    range = WebRtcIlbcfix_kSearchRange[block][stage];
+
+    /* initialize search measures */
+    CritMax=0;
+    shTotMax=-100;
+    bestIndex=0;
+    bestGain=0;
+
+    /* loop over lags 40+ in the first codebook section, full search */
+    cb_vecPtr = buf+lMem-lTarget;
+
+    /* Calculate all the cross correlations (augmented part of CB) */
+    if (lTarget==SUBL) {
+      WebRtcIlbcfix_AugmentedCbCorr(target, buf+lMem,
+                                    interpSamples, cDot,
+                                    20, 39, scale);
+      cDotPtr=&cDot[20];
+    } else {
+      cDotPtr=cDot;
+    }
+    /* Calculate all the cross correlations (main part of CB) */
+    WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, range, scale, -1);
+
+    /* Adjust the search range for the augmented vectors */
+    if (lTarget==SUBL) {
+      range=WebRtcIlbcfix_kSearchRange[block][stage]+20;
+    } else {
+      range=WebRtcIlbcfix_kSearchRange[block][stage];
+    }
+
+    indexOffset=0;
+
+    /* Search for best index in this part of the vector */
+    WebRtcIlbcfix_CbSearchCore(
+        cDot, range, stage, inverseEnergy,
+        inverseEnergyShifts, Crit,
+        &indexNew, &CritNew, &CritNewSh);
+
+    /* Update the global best index and the corresponding gain */
+    WebRtcIlbcfix_CbUpdateBestIndex(
+        CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew+indexOffset],
+        inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
+        &CritMax, &shTotMax, &bestIndex, &bestGain);
+
+    sInd = ((CB_RESRANGE >> 1) > bestIndex) ?
+        0 : (bestIndex - (CB_RESRANGE >> 1));
+    eInd=sInd+CB_RESRANGE;
+    if (eInd>=range) {
+      eInd=range-1;
+      sInd=eInd-CB_RESRANGE;
+    }
+
+    range = WebRtcIlbcfix_kSearchRange[block][stage];
+
+    if (lTarget==SUBL) {
+      i=sInd;
+      if (sInd<20) {
+        WebRtcIlbcfix_AugmentedCbCorr(target, cbvectors + lMem,
+                                      interpSamplesFilt, cDot, sInd + 20,
+                                      WEBRTC_SPL_MIN(39, (eInd + 20)), scale);
+        i=20;
+        cDotPtr = &cDot[20 - sInd];
+      } else {
+        cDotPtr = cDot;
+      }
+
+      cb_vecPtr = cbvectors+lMem-20-i;
+
+      /* Calculate the cross correlations (main part of the filtered CB) */
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget,
+                                 eInd - i + 1, scale, -1);
+
+    } else {
+      cDotPtr = cDot;
+      cb_vecPtr = cbvectors+lMem-lTarget-sInd;
+
+      /* Calculate the cross correlations (main part of the filtered CB) */
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget,
+                                 eInd - sInd + 1, scale, -1);
+
+    }
+
+    /* Adjust the search range for the augmented vectors */
+    indexOffset=base_size+sInd;
+
+    /* Search for best index in this part of the vector */
+    WebRtcIlbcfix_CbSearchCore(
+        cDot, eInd-sInd+1, stage, inverseEnergy+indexOffset,
+        inverseEnergyShifts+indexOffset, Crit,
+        &indexNew, &CritNew, &CritNewSh);
+
+    /* Update the global best index and the corresponding gain */
+    WebRtcIlbcfix_CbUpdateBestIndex(
+        CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew],
+        inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
+        &CritMax, &shTotMax, &bestIndex, &bestGain);
+
+    index[stage] = (int16_t)bestIndex;
+
+
+    bestGain = WebRtcIlbcfix_GainQuant(bestGain,
+                                       (int16_t)WEBRTC_SPL_ABS_W16(gains[stage]), stage, &gain_index[stage]);
+
+    /* Extract the best (according to measure) codebook vector
+       Also adjust the index, so that the augmented vectors are last.
+       Above these vectors were first...
+    */
+
+    if(lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) {
+
+      if((size_t)index[stage]<base_size) {
+        pp=buf+lMem-lTarget-index[stage];
+      } else {
+        pp=cbvectors+lMem-lTarget-
+            index[stage]+base_size;
+      }
+
+    } else {
+
+      if ((size_t)index[stage]<base_size) {
+        if (index[stage]>=20) {
+          /* Adjust index and extract vector */
+          index[stage]-=20;
+          pp=buf+lMem-lTarget-index[stage];
+        } else {
+          /* Adjust index and extract vector */
+          index[stage]+=(int16_t)(base_size-20);
+
+          WebRtcIlbcfix_CreateAugmentedVec(index[stage]-base_size+40,
+                                           buf+lMem, aug_vec);
+          pp = aug_vec;
+
+        }
+      } else {
+
+        if ((index[stage] - base_size) >= 20) {
+          /* Adjust index and extract vector */
+          index[stage]-=20;
+          pp=cbvectors+lMem-lTarget-
+              index[stage]+base_size;
+        } else {
+          /* Adjust index and extract vector */
+          index[stage]+=(int16_t)(base_size-20);
+          WebRtcIlbcfix_CreateAugmentedVec(index[stage]-2*base_size+40,
+                                           cbvectors+lMem, aug_vec);
+          pp = aug_vec;
+        }
+      }
+    }
+
+    /* Subtract the best codebook vector, according
+       to measure, from the target vector */
+
+    WebRtcSpl_AddAffineVectorToVector(target, pp, (int16_t)(-bestGain),
+                                      (int32_t)8192, (int16_t)14, lTarget);
+
+    /* record quantized gain */
+    gains[stage+1] = bestGain;
+
+  } /* end of Main Loop. for (stage=0;... */
+
+  /* Calculte the coded vector (original target - what's left) */
+  for (i=0;i<lTarget;i++) {
+    codedVec[i]-=target[i];
+  }
+
+  /* Gain adjustment for energy matching */
+  codedEner = WebRtcSpl_DotProductWithScale(codedVec, codedVec, lTarget, scale);
+
+  j=gain_index[0];
+
+  temp1 = (int16_t)WebRtcSpl_NormW32(codedEner);
+  temp2 = (int16_t)WebRtcSpl_NormW32(targetEner);
+
+  if(temp1 < temp2) {
+    bits = 16 - temp1;
+  } else {
+    bits = 16 - temp2;
+  }
+
+  tmp = (int16_t)((gains[1] * gains[1]) >> 14);
+
+  targetEner = (int16_t)WEBRTC_SPL_SHIFT_W32(targetEner, -bits) * tmp;
+
+  tmpW32 = ((int32_t)(gains[1]-1))<<1;
+
+  /* Pointer to the table that contains
+     gain_sq5TblFIX * gain_sq5TblFIX in Q14 */
+  gainPtr=(int16_t*)WebRtcIlbcfix_kGainSq5Sq+gain_index[0];
+  temp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(codedEner, -bits);
+
+  WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[j];
+
+  /* targetEner and codedEner are in Q(-2*scale) */
+  for (ii=gain_index[0];ii<32;ii++) {
+
+    /* Change the index if
+       (codedEnergy*gainTbl[i]*gainTbl[i])<(targetEn*gain[0]*gain[0]) AND
+       gainTbl[i] < 2*gain[0]
+    */
+
+    t32 = temp1 * *gainPtr;
+    t32 = t32 - targetEner;
+    if (t32 < 0) {
+      if ((*WebRtcIlbcfix_kGainSq5_ptr) < tmpW32) {
+        j=ii;
+        WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[ii];
+      }
+    }
+    gainPtr++;
+  }
+  gain_index[0]=j;
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.h
new file mode 100644
index 0000000000..84a52c7868
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearch.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbSearch(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) the encoder state structure */
+    int16_t* index,       /* (o) Codebook indices */
+    int16_t* gain_index,  /* (o) Gain quantization indices */
+    int16_t* intarget,    /* (i) Target vector for encoding */
+    int16_t* decResidual, /* (i) Decoded residual for codebook construction */
+    size_t lMem,          /* (i) Length of buffer */
+    size_t lTarget,       /* (i) Length of vector */
+    int16_t* weightDenum, /* (i) weighting filter coefficients in Q12 */
+    size_t block          /* (i) the subblock number */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
new file mode 100644
index 0000000000..a75e5b0ab8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearchCore.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_search_core.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbSearchCore(
+    int32_t *cDot,    /* (i) Cross Correlation */
+    size_t range,    /* (i) Search range */
+    int16_t stage,    /* (i) Stage of this search */
+    int16_t *inverseEnergy,  /* (i) Inversed energy */
+    int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy
+                                           with the offset 2*16-29 */
+    int32_t *Crit,    /* (o) The criteria */
+    size_t *bestIndex,   /* (o) Index that corresponds to
+                                                   maximum criteria (in this
+                                                   vector) */
+    int32_t *bestCrit,   /* (o) Value of critera for the
+                                                   chosen index */
+    int16_t *bestCritSh)   /* (o) The domain of the chosen
+                                                   criteria */
+{
+  int32_t maxW32, tmp32;
+  int16_t max, sh, tmp16;
+  size_t i;
+  int32_t *cDotPtr;
+  int16_t cDotSqW16;
+  int16_t *inverseEnergyPtr;
+  int32_t *critPtr;
+  int16_t *inverseEnergyShiftPtr;
+
+  /* Don't allow negative values for stage 0 */
+  if (stage==0) {
+    cDotPtr=cDot;
+    for (i=0;i<range;i++) {
+      *cDotPtr=WEBRTC_SPL_MAX(0, (*cDotPtr));
+      cDotPtr++;
+    }
+  }
+
+  /* Normalize cDot to int16_t, calculate the square of cDot and store the upper int16_t */
+  maxW32 = WebRtcSpl_MaxAbsValueW32(cDot, range);
+
+  sh = (int16_t)WebRtcSpl_NormW32(maxW32);
+  cDotPtr = cDot;
+  inverseEnergyPtr = inverseEnergy;
+  critPtr = Crit;
+  inverseEnergyShiftPtr=inverseEnergyShift;
+  max=WEBRTC_SPL_WORD16_MIN;
+
+  for (i=0;i<range;i++) {
+    /* Calculate cDot*cDot and put the result in a int16_t */
+    tmp32 = *cDotPtr << sh;
+    tmp16 = (int16_t)(tmp32 >> 16);
+    cDotSqW16 = (int16_t)(((int32_t)(tmp16)*(tmp16))>>16);
+
+    /* Calculate the criteria (cDot*cDot/energy) */
+    *critPtr = cDotSqW16 * *inverseEnergyPtr;
+
+    /* Extract the maximum shift value under the constraint
+       that the criteria is not zero */
+    if ((*critPtr)!=0) {
+      max = WEBRTC_SPL_MAX((*inverseEnergyShiftPtr), max);
+    }
+
+    inverseEnergyPtr++;
+    inverseEnergyShiftPtr++;
+    critPtr++;
+    cDotPtr++;
+  }
+
+  /* If no max shifts still at initialization value, set shift to zero */
+  if (max==WEBRTC_SPL_WORD16_MIN) {
+    max = 0;
+  }
+
+  /* Modify the criterias, so that all of them use the same Q domain */
+  critPtr=Crit;
+  inverseEnergyShiftPtr=inverseEnergyShift;
+  for (i=0;i<range;i++) {
+    /* Guarantee that the shift value is less than 16
+       in order to simplify for DSP's (and guard against >31) */
+    tmp16 = WEBRTC_SPL_MIN(16, max-(*inverseEnergyShiftPtr));
+
+    (*critPtr)=WEBRTC_SPL_SHIFT_W32((*critPtr),-tmp16);
+    critPtr++;
+    inverseEnergyShiftPtr++;
+  }
+
+  /* Find the index of the best value */
+  *bestIndex = WebRtcSpl_MaxIndexW32(Crit, range);
+  *bestCrit = Crit[*bestIndex];
+
+  /* Calculate total shifts of this criteria */
+  *bestCritSh = 32 - 2*sh + max;
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
new file mode 100644
index 0000000000..5da70e0988
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearchCore.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbSearchCore(
+    int32_t* cDot,               /* (i) Cross Correlation */
+    size_t range,                /* (i) Search range */
+    int16_t stage,               /* (i) Stage of this search */
+    int16_t* inverseEnergy,      /* (i) Inversed energy */
+    int16_t* inverseEnergyShift, /* (i) Shifts of inversed energy
+                                          with the offset 2*16-29 */
+    int32_t* Crit,               /* (o) The criteria */
+    size_t* bestIndex,           /* (o) Index that corresponds to
+                                           maximum criteria (in this
+                                           vector) */
+    int32_t* bestCrit, /* (o) Value of critera for the
+                                chosen index */
+    int16_t* bestCritSh); /* (o) The domain of the chosen
+                                   criteria */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
new file mode 100644
index 0000000000..d6fa4d93d4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbUpdateBestIndex.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_update_best_index.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbUpdateBestIndex(
+    int32_t CritNew,    /* (i) New Potentially best Criteria */
+    int16_t CritNewSh,   /* (i) Shift value of above Criteria */
+    size_t IndexNew,   /* (i) Index of new Criteria */
+    int32_t cDotNew,    /* (i) Cross dot of new index */
+    int16_t invEnergyNew,  /* (i) Inversed energy new index */
+    int16_t energyShiftNew,  /* (i) Energy shifts of new index */
+    int32_t *CritMax,   /* (i/o) Maximum Criteria (so far) */
+    int16_t *shTotMax,   /* (i/o) Shifts of maximum criteria */
+    size_t *bestIndex,   /* (i/o) Index that corresponds to
+                                                   maximum criteria */
+    int16_t *bestGain)   /* (i/o) Gain in Q14 that corresponds
+                                                   to maximum criteria */
+{
+  int16_t shOld, shNew, tmp16;
+  int16_t scaleTmp;
+  int32_t gainW32;
+
+  /* Normalize the new and old Criteria to the same domain */
+  if (CritNewSh>(*shTotMax)) {
+    shOld=WEBRTC_SPL_MIN(31,CritNewSh-(*shTotMax));
+    shNew=0;
+  } else {
+    shOld=0;
+    shNew=WEBRTC_SPL_MIN(31,(*shTotMax)-CritNewSh);
+  }
+
+  /* Compare the two criterias. If the new one is better,
+     calculate the gain and store this index as the new best one
+  */
+
+  if ((CritNew >> shNew) > (*CritMax >> shOld)) {
+
+    tmp16 = (int16_t)WebRtcSpl_NormW32(cDotNew);
+    tmp16 = 16 - tmp16;
+
+    /* Calculate the gain in Q14
+       Compensate for inverseEnergyshift in Q29 and that the energy
+       value was stored in a int16_t (shifted down 16 steps)
+       => 29-14+16 = 31 */
+
+    scaleTmp = -energyShiftNew-tmp16+31;
+    scaleTmp = WEBRTC_SPL_MIN(31, scaleTmp);
+
+    gainW32 = ((int16_t)WEBRTC_SPL_SHIFT_W32(cDotNew, -tmp16) * invEnergyNew) >>
+        scaleTmp;
+
+    /* Check if criteria satisfies Gain criteria (max 1.3)
+       if it is larger set the gain to 1.3
+       (slightly different from FLP version)
+    */
+    if (gainW32>21299) {
+      *bestGain=21299;
+    } else if (gainW32<-21299) {
+      *bestGain=-21299;
+    } else {
+      *bestGain=(int16_t)gainW32;
+    }
+
+    *CritMax=CritNew;
+    *shTotMax=CritNewSh;
+    *bestIndex = IndexNew;
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
new file mode 100644
index 0000000000..1a95d531e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbUpdateBestIndex.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbUpdateBestIndex(
+    int32_t CritNew,        /* (i) New Potentially best Criteria */
+    int16_t CritNewSh,      /* (i) Shift value of above Criteria */
+    size_t IndexNew,        /* (i) Index of new Criteria */
+    int32_t cDotNew,        /* (i) Cross dot of new index */
+    int16_t invEnergyNew,   /* (i) Inversed energy new index */
+    int16_t energyShiftNew, /* (i) Energy shifts of new index */
+    int32_t* CritMax,       /* (i/o) Maximum Criteria (so far) */
+    int16_t* shTotMax,      /* (i/o) Shifts of maximum criteria */
+    size_t* bestIndex,      /* (i/o) Index that corresponds to
+                                      maximum criteria */
+    int16_t* bestGain);     /* (i/o) Gain in Q14 that corresponds
+                                     to maximum criteria */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c
new file mode 100644
index 0000000000..b4eee66219
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Chebyshev.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/chebyshev.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*------------------------------------------------------------------*
+ *  Calculate the Chevyshev polynomial series
+ *  F(w) = 2*exp(-j5w)*C(x)
+ *   C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
+ *   T_i(x) is the i:th order Chebyshev polynomial
+ *------------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_Chebyshev(
+    /* (o) Result of C(x) */
+    int16_t x,  /* (i) Value to the Chevyshev polynomial */
+    int16_t *f  /* (i) The coefficients in the polynomial */
+                                      ) {
+  int16_t b1_high, b1_low; /* Use the high, low format to increase the accuracy */
+  int32_t b2;
+  int32_t tmp1W32;
+  int32_t tmp2W32;
+  int i;
+
+  b2 = (int32_t)0x1000000; /* b2 = 1.0 (Q23) */
+  /* Calculate b1 = 2*x + f[1] */
+  tmp1W32 = (x << 10) + (f[1] << 14);
+
+  for (i = 2; i < 5; i++) {
+    tmp2W32 = tmp1W32;
+
+    /* Split b1 (in tmp1W32) into a high and low part */
+    b1_high = (int16_t)(tmp1W32 >> 16);
+    b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1);
+
+    /* Calculate 2*x*b1-b2+f[i] */
+    tmp1W32 = ((b1_high * x + ((b1_low * x) >> 15)) << 2) - b2 + (f[i] << 14);
+
+    /* Update b2 for next round */
+    b2 = tmp2W32;
+  }
+
+  /* Split b1 (in tmp1W32) into a high and low part */
+  b1_high = (int16_t)(tmp1W32 >> 16);
+  b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1);
+
+  /* tmp1W32 = x*b1 - b2 + f[i]/2 */
+  tmp1W32 = ((b1_high * x) << 1) + (((b1_low * x) >> 15) << 1) -
+      b2 + (f[i] << 13);
+
+  /* Handle overflows and set to maximum or minimum int16_t instead */
+  if (tmp1W32>((int32_t)33553408)) {
+    return(WEBRTC_SPL_WORD16_MAX);
+  } else if (tmp1W32<((int32_t)-33554432)) {
+    return(WEBRTC_SPL_WORD16_MIN);
+  } else {
+    return (int16_t)(tmp1W32 >> 10);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.h
new file mode 100644
index 0000000000..7e7742c5cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Chebyshev.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*------------------------------------------------------------------*
+ *  Calculate the Chevyshev polynomial series
+ *  F(w) = 2*exp(-j5w)*C(x)
+ *   C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
+ *   T_i(x) is the i:th order Chebyshev polynomial
+ *------------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_Chebyshev(
+    /* (o) Result of C(x) */
+    int16_t x, /* (i) Value to the Chevyshev polynomial */
+    int16_t* f /* (i) The coefficients in the polynomial */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
new file mode 100644
index 0000000000..452bc78e3b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CompCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/comp_corr.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Compute cross correlation and pitch gain for pitch prediction
+ *  of last subframe at given lag.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CompCorr(
+    int32_t *corr, /* (o) cross correlation */
+    int32_t *ener, /* (o) energy */
+    int16_t *buffer, /* (i) signal buffer */
+    size_t lag,  /* (i) pitch lag */
+    size_t bLen, /* (i) length of buffer */
+    size_t sRange, /* (i) correlation search length */
+    int16_t scale /* (i) number of rightshifts to use */
+                            ){
+  int16_t *w16ptr;
+
+  w16ptr=&buffer[bLen-sRange-lag];
+
+  /* Calculate correlation and energy */
+  (*corr)=WebRtcSpl_DotProductWithScale(&buffer[bLen-sRange], w16ptr, sRange, scale);
+  (*ener)=WebRtcSpl_DotProductWithScale(w16ptr, w16ptr, sRange, scale);
+
+  /* For zero energy set the energy to 0 in order to avoid potential
+     problems for coming divisions */
+  if (*ener == 0) {
+    *corr = 0;
+    *ener = 1;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
new file mode 100644
index 0000000000..010c6a1ce5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CompCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Compute cross correlation and pitch gain for pitch prediction
+ *  of last subframe at given lag.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CompCorr(int32_t* corr,   /* (o) cross correlation */
+                            int32_t* ener,   /* (o) energy */
+                            int16_t* buffer, /* (i) signal buffer */
+                            size_t lag,      /* (i) pitch lag */
+                            size_t bLen,     /* (i) length of buffer */
+                            size_t sRange,   /* (i) correlation search length */
+                            int16_t scale /* (i) number of rightshifts to use */
+                            );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m
new file mode 100644
index 0000000000..4bda83622f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m
@@ -0,0 +1,57 @@
+% % Copyright(c) 2011 The WebRTC project authors.All Rights Reserved.%
+        % Use of this source code is governed by a BSD
+    -
+    style license % that can be found in the LICENSE file in the root of the source
+        % tree.An additional intellectual property rights grant can be found
+        % in the file PATENTS.All contributing project authors may
+        % be found in the AUTHORS file in the root of the source tree.%
+
+        clear;
+pack;
+%
+% Enter the path to YOUR executable and remember to define the perprocessor
+% variable PRINT_MIPS te get the instructions printed to the screen.
+%
+command = '!iLBCtest.exe 30 speechAndBGnoise.pcm out1.bit out1.pcm tlm10_30ms.dat';
+cout=' > st.txt';   %saves to matlab variable 'st'
+eval(strcat(command,cout));
+if(length(cout)>3)
+    load st.txt
+else
+    disp('No cout file to load')
+end
+
+% initialize vector to zero
+index = find(st(1:end,1)==-1);
+indexnonzero = find(st(1:end,1)>0);
+frames = length(index)-indexnonzero(1)+1;
+start = indexnonzero(1) - 1;
+functionOrder=max(st(:,2));
+new=zeros(frames,functionOrder);
+
+for i = 1:frames,
+    for j = index(start-1+i)+1:(index(start+i)-1),
+        new(i,st(j,2)) = new(i,st(j,2)) + st(j,1);
+    end
+end
+
+result=zeros(functionOrder,3);
+for i=1:functionOrder
+    nonzeroelements = find(new(1:end,i)>0);
+    result(i,1)=i;
+    
+    % Compute each function's mean complexity
+    % result(i,2)=(sum(new(nonzeroelements,i))/(length(nonzeroelements)*0.03))/1000000;
+    
+    % Compute each function's maximum complexity in encoding
+    % and decoding respectively and then add it together:
+    % result(i,3)=(max(new(1:end,i))/0.03)/1000000;
+    result(i,3)=(max(new(1:size(new,1)/2,i))/0.03)/1000000 + (max(new(size(new,1)/2+1:end,i))/0.03)/1000000;
+end
+
+result
+
+% Compute maximum complexity for a single frame (enc/dec separately and together)
+maxEncComplexityInAFrame = (max(sum(new(1:size(new,1)/2,:),2))/0.03)/1000000
+maxDecComplexityInAFrame = (max(sum(new(size(new,1)/2+1:end,:),2))/0.03)/1000000
+totalComplexity = maxEncComplexityInAFrame + maxDecComplexityInAFrame
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c
new file mode 100644
index 0000000000..22f2acb330
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c
@@ -0,0 +1,667 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* HP Filters {b[0] b[1] b[2] -a[1] -a[2]} */
+
+const int16_t WebRtcIlbcfix_kHpInCoefs[5] = {3798, -7596, 3798, 7807, -3733};
+const int16_t WebRtcIlbcfix_kHpOutCoefs[5] = {3849, -7699, 3849, 7918, -3833};
+
+/* Window in Q11 to window the energies of the 5 choises (3 for 20ms) in the choise for
+   the 80 sample start state
+*/
+const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[NSUB_MAX-1]= {
+  1638, 1843, 2048, 1843, 1638
+};
+
+/* LP Filter coeffs used for downsampling */
+const int16_t WebRtcIlbcfix_kLpFiltCoefs[FILTERORDER_DS_PLUS1]= {
+  -273, 512, 1297, 1696, 1297, 512, -273
+};
+
+/* Constants used in the LPC calculations */
+
+/* Hanning LPC window (in Q15) */
+const int16_t WebRtcIlbcfix_kLpcWin[BLOCKL_MAX] = {
+  6, 22, 50, 89, 139, 200, 272, 355, 449, 554, 669, 795,
+  932, 1079, 1237, 1405, 1583, 1771, 1969, 2177, 2395, 2622, 2858, 3104,
+  3359, 3622, 3894, 4175, 4464, 4761, 5066, 5379, 5699, 6026, 6361, 6702,
+  7050, 7404, 7764, 8130, 8502, 8879, 9262, 9649, 10040, 10436, 10836, 11240,
+  11647, 12058, 12471, 12887, 13306, 13726, 14148, 14572, 14997, 15423, 15850, 16277,
+  16704, 17131, 17558, 17983, 18408, 18831, 19252, 19672, 20089, 20504, 20916, 21325,
+  21730, 22132, 22530, 22924, 23314, 23698, 24078, 24452, 24821, 25185, 25542, 25893,
+  26238, 26575, 26906, 27230, 27547, 27855, 28156, 28450, 28734, 29011, 29279, 29538,
+  29788, 30029, 30261, 30483, 30696, 30899, 31092, 31275, 31448, 31611, 31764, 31906,
+  32037, 32158, 32268, 32367, 32456, 32533, 32600, 32655, 32700, 32733, 32755, 32767,
+  32767, 32755, 32733, 32700, 32655, 32600, 32533, 32456, 32367, 32268, 32158, 32037,
+  31906, 31764, 31611, 31448, 31275, 31092, 30899, 30696, 30483, 30261, 30029, 29788,
+  29538, 29279, 29011, 28734, 28450, 28156, 27855, 27547, 27230, 26906, 26575, 26238,
+  25893, 25542, 25185, 24821, 24452, 24078, 23698, 23314, 22924, 22530, 22132, 21730,
+  21325, 20916, 20504, 20089, 19672, 19252, 18831, 18408, 17983, 17558, 17131, 16704,
+  16277, 15850, 15423, 14997, 14572, 14148, 13726, 13306, 12887, 12471, 12058, 11647,
+  11240, 10836, 10436, 10040, 9649, 9262, 8879, 8502, 8130, 7764, 7404, 7050,
+  6702, 6361, 6026, 5699, 5379, 5066, 4761, 4464, 4175, 3894, 3622, 3359,
+  3104, 2858, 2622, 2395, 2177, 1969, 1771, 1583, 1405, 1237, 1079, 932,
+  795, 669, 554, 449, 355, 272, 200, 139, 89, 50, 22, 6
+};
+
+/* Asymmetric LPC window (in Q15)*/
+const int16_t WebRtcIlbcfix_kLpcAsymWin[BLOCKL_MAX] = {
+  2, 7, 15, 27, 42, 60, 81, 106, 135, 166, 201, 239,
+  280, 325, 373, 424, 478, 536, 597, 661, 728, 798, 872, 949,
+  1028, 1111, 1197, 1287, 1379, 1474, 1572, 1674, 1778, 1885, 1995, 2108,
+  2224, 2343, 2465, 2589, 2717, 2847, 2980, 3115, 3254, 3395, 3538, 3684,
+  3833, 3984, 4138, 4295, 4453, 4615, 4778, 4944, 5112, 5283, 5456, 5631,
+  5808, 5987, 6169, 6352, 6538, 6725, 6915, 7106, 7300, 7495, 7692, 7891,
+  8091, 8293, 8497, 8702, 8909, 9118, 9328, 9539, 9752, 9966, 10182, 10398,
+  10616, 10835, 11055, 11277, 11499, 11722, 11947, 12172, 12398, 12625, 12852, 13080,
+  13309, 13539, 13769, 14000, 14231, 14463, 14695, 14927, 15160, 15393, 15626, 15859,
+  16092, 16326, 16559, 16792, 17026, 17259, 17492, 17725, 17957, 18189, 18421, 18653,
+  18884, 19114, 19344, 19573, 19802, 20030, 20257, 20483, 20709, 20934, 21157, 21380,
+  21602, 21823, 22042, 22261, 22478, 22694, 22909, 23123, 23335, 23545, 23755, 23962,
+  24168, 24373, 24576, 24777, 24977, 25175, 25371, 25565, 25758, 25948, 26137, 26323,
+  26508, 26690, 26871, 27049, 27225, 27399, 27571, 27740, 27907, 28072, 28234, 28394,
+  28552, 28707, 28860, 29010, 29157, 29302, 29444, 29584, 29721, 29855, 29987, 30115,
+  30241, 30364, 30485, 30602, 30717, 30828, 30937, 31043, 31145, 31245, 31342, 31436,
+  31526, 31614, 31699, 31780, 31858, 31933, 32005, 32074, 32140, 32202, 32261, 32317,
+  32370, 32420, 32466, 32509, 32549, 32585, 32618, 32648, 32675, 32698, 32718, 32734,
+  32748, 32758, 32764, 32767, 32767, 32667, 32365, 31863, 31164, 30274, 29197, 27939,
+  26510, 24917, 23170, 21281, 19261, 17121, 14876, 12540, 10126, 7650, 5126, 2571
+};
+
+/* Lag window for LPC (Q31) */
+const int32_t WebRtcIlbcfix_kLpcLagWin[LPC_FILTERORDER + 1]={
+  2147483647,   2144885453,   2137754373,   2125918626,   2109459810,
+  2088483140,   2063130336,   2033564590,   1999977009,   1962580174,
+  1921610283};
+
+/* WebRtcIlbcfix_kLpcChirpSyntDenum vector in Q15 corresponding
+ * floating point vector {1 0.9025 0.9025^2 0.9025^3 ...}
+ */
+const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[LPC_FILTERORDER + 1] = {
+  32767, 29573, 26690, 24087,
+  21739, 19619, 17707, 15980,
+  14422, 13016, 11747};
+
+/* WebRtcIlbcfix_kLpcChirpWeightDenum in Q15 corresponding to
+ * floating point vector {1 0.4222 0.4222^2... }
+ */
+const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[LPC_FILTERORDER + 1] = {
+  32767, 13835, 5841, 2466, 1041, 440,
+  186, 78,  33,  14,  6};
+
+/* LSF quantization Q13 domain */
+const int16_t WebRtcIlbcfix_kLsfCb[64 * 3 + 128 * 3 + 128 * 4] = {
+  1273,       2238,       3696,
+  3199,       5309,       8209,
+  3606,       5671,       7829,
+  2815,       5262,       8778,
+  2608,       4027,       5493,
+  1582,       3076,       5945,
+  2983,       4181,       5396,
+  2437,       4322,       6902,
+  1861,       2998,       4613,
+  2007,       3250,       5214,
+  1388,       2459,       4262,
+  2563,       3805,       5269,
+  2036,       3522,       5129,
+  1935,       4025,       6694,
+  2744,       5121,       7338,
+  2810,       4248,       5723,
+  3054,       5405,       7745,
+  1449,       2593,       4763,
+  3411,       5128,       6596,
+  2484,       4659,       7496,
+  1668,       2879,       4818,
+  1812,       3072,       5036,
+  1638,       2649,       3900,
+  2464,       3550,       4644,
+  1853,       2900,       4158,
+  2458,       4163,       5830,
+  2556,       4036,       6254,
+  2703,       4432,       6519,
+  3062,       4953,       7609,
+  1725,       3703,       6187,
+  2221,       3877,       5427,
+  2339,       3579,       5197,
+  2021,       4633,       7037,
+  2216,       3328,       4535,
+  2961,       4739,       6667,
+  2807,       3955,       5099,
+  2788,       4501,       6088,
+  1642,       2755,       4431,
+  3341,       5282,       7333,
+  2414,       3726,       5727,
+  1582,       2822,       5269,
+  2259,       3447,       4905,
+  3117,       4986,       7054,
+  1825,       3491,       5542,
+  3338,       5736,       8627,
+  1789,       3090,       5488,
+  2566,       3720,       4923,
+  2846,       4682,       7161,
+  1950,       3321,       5976,
+  1834,       3383,       6734,
+  3238,       4769,       6094,
+  2031,       3978,       5903,
+  1877,       4068,       7436,
+  2131,       4644,       8296,
+  2764,       5010,       8013,
+  2194,       3667,       6302,
+  2053,       3127,       4342,
+  3523,       6595,      10010,
+  3134,       4457,       5748,
+  3142,       5819,       9414,
+  2223,       4334,       6353,
+  2022,       3224,       4822,
+  2186,       3458,       5544,
+  2552,       4757,       6870,
+  10905,      12917,      14578,
+  9503,      11485,      14485,
+  9518,      12494,      14052,
+  6222,       7487,       9174,
+  7759,       9186,      10506,
+  8315,      12755,      14786,
+  9609,      11486,      13866,
+  8909,      12077,      13643,
+  7369,       9054,      11520,
+  9408,      12163,      14715,
+  6436,       9911,      12843,
+  7109,       9556,      11884,
+  7557,      10075,      11640,
+  6482,       9202,      11547,
+  6463,       7914,      10980,
+  8611,      10427,      12752,
+  7101,       9676,      12606,
+  7428,      11252,      13172,
+  10197,      12955,      15842,
+  7487,      10955,      12613,
+  5575,       7858,      13621,
+  7268,      11719,      14752,
+  7476,      11744,      13795,
+  7049,       8686,      11922,
+  8234,      11314,      13983,
+  6560,      11173,      14984,
+  6405,       9211,      12337,
+  8222,      12054,      13801,
+  8039,      10728,      13255,
+  10066,      12733,      14389,
+  6016,       7338,      10040,
+  6896,       8648,      10234,
+  7538,       9170,      12175,
+  7327,      12608,      14983,
+  10516,      12643,      15223,
+  5538,       7644,      12213,
+  6728,      12221,      14253,
+  7563,       9377,      12948,
+  8661,      11023,      13401,
+  7280,       8806,      11085,
+  7723,       9793,      12333,
+  12225,      14648,      16709,
+  8768,      13389,      15245,
+  10267,      12197,      13812,
+  5301,       7078,      11484,
+  7100,      10280,      11906,
+  8716,      12555,      14183,
+  9567,      12464,      15434,
+  7832,      12305,      14300,
+  7608,      10556,      12121,
+  8913,      11311,      12868,
+  7414,       9722,      11239,
+  8666,      11641,      13250,
+  9079,      10752,      12300,
+  8024,      11608,      13306,
+  10453,      13607,      16449,
+  8135,       9573,      10909,
+  6375,       7741,      10125,
+  10025,      12217,      14874,
+  6985,      11063,      14109,
+  9296,      13051,      14642,
+  8613,      10975,      12542,
+  6583,      10414,      13534,
+  6191,       9368,      13430,
+  5742,       6859,       9260,
+  7723,       9813,      13679,
+  8137,      11291,      12833,
+  6562,       8973,      10641,
+  6062,       8462,      11335,
+  6928,       8784,      12647,
+  7501,       8784,      10031,
+  8372,      10045,      12135,
+  8191,       9864,      12746,
+  5917,       7487,      10979,
+  5516,       6848,      10318,
+  6819,       9899,      11421,
+  7882,      12912,      15670,
+  9558,      11230,      12753,
+  7752,       9327,      11472,
+  8479,       9980,      11358,
+  11418,      14072,      16386,
+  7968,      10330,      14423,
+  8423,      10555,      12162,
+  6337,      10306,      14391,
+  8850,      10879,      14276,
+  6750,      11885,      15710,
+  7037,       8328,       9764,
+  6914,       9266,      13476,
+  9746,      13949,      15519,
+  11032,      14444,      16925,
+  8032,      10271,      11810,
+  10962,      13451,      15833,
+  10021,      11667,      13324,
+  6273,       8226,      12936,
+  8543,      10397,      13496,
+  7936,      10302,      12745,
+  6769,       8138,      10446,
+  6081,       7786,      11719,
+  8637,      11795,      14975,
+  8790,      10336,      11812,
+  7040,       8490,      10771,
+  7338,      10381,      13153,
+  6598,       7888,       9358,
+  6518,       8237,      12030,
+  9055,      10763,      12983,
+  6490,      10009,      12007,
+  9589,      12023,      13632,
+  6867,       9447,      10995,
+  7930,       9816,      11397,
+  10241,      13300,      14939,
+  5830,       8670,      12387,
+  9870,      11915,      14247,
+  9318,      11647,      13272,
+  6721,      10836,      12929,
+  6543,       8233,       9944,
+  8034,      10854,      12394,
+  9112,      11787,      14218,
+  9302,      11114,      13400,
+  9022,      11366,      13816,
+  6962,      10461,      12480,
+  11288,      13333,      15222,
+  7249,       8974,      10547,
+  10566,      12336,      14390,
+  6697,      11339,      13521,
+  11851,      13944,      15826,
+  6847,       8381,      11349,
+  7509,       9331,      10939,
+  8029,       9618,      11909,
+  13973,      17644,      19647,      22474,
+  14722,      16522,      20035,      22134,
+  16305,      18179,      21106,      23048,
+  15150,      17948,      21394,      23225,
+  13582,      15191,      17687,      22333,
+  11778,      15546,      18458,      21753,
+  16619,      18410,      20827,      23559,
+  14229,      15746,      17907,      22474,
+  12465,      15327,      20700,      22831,
+  15085,      16799,      20182,      23410,
+  13026,      16935,      19890,      22892,
+  14310,      16854,      19007,      22944,
+  14210,      15897,      18891,      23154,
+  14633,      18059,      20132,      22899,
+  15246,      17781,      19780,      22640,
+  16396,      18904,      20912,      23035,
+  14618,      17401,      19510,      21672,
+  15473,      17497,      19813,      23439,
+  18851,      20736,      22323,      23864,
+  15055,      16804,      18530,      20916,
+  16490,      18196,      19990,      21939,
+  11711,      15223,      21154,      23312,
+  13294,      15546,      19393,      21472,
+  12956,      16060,      20610,      22417,
+  11628,      15843,      19617,      22501,
+  14106,      16872,      19839,      22689,
+  15655,      18192,      20161,      22452,
+  12953,      15244,      20619,      23549,
+  15322,      17193,      19926,      21762,
+  16873,      18676,      20444,      22359,
+  14874,      17871,      20083,      21959,
+  11534,      14486,      19194,      21857,
+  17766,      19617,      21338,      23178,
+  13404,      15284,      19080,      23136,
+  15392,      17527,      19470,      21953,
+  14462,      16153,      17985,      21192,
+  17734,      19750,      21903,      23783,
+  16973,      19096,      21675,      23815,
+  16597,      18936,      21257,      23461,
+  15966,      17865,      20602,      22920,
+  15416,      17456,      20301,      22972,
+  18335,      20093,      21732,      23497,
+  15548,      17217,      20679,      23594,
+  15208,      16995,      20816,      22870,
+  13890,      18015,      20531,      22468,
+  13211,      15377,      19951,      22388,
+  12852,      14635,      17978,      22680,
+  16002,      17732,      20373,      23544,
+  11373,      14134,      19534,      22707,
+  17329,      19151,      21241,      23462,
+  15612,      17296,      19362,      22850,
+  15422,      19104,      21285,      23164,
+  13792,      17111,      19349,      21370,
+  15352,      17876,      20776,      22667,
+  15253,      16961,      18921,      22123,
+  14108,      17264,      20294,      23246,
+  15785,      17897,      20010,      21822,
+  17399,      19147,      20915,      22753,
+  13010,      15659,      18127,      20840,
+  16826,      19422,      22218,      24084,
+  18108,      20641,      22695,      24237,
+  18018,      20273,      22268,      23920,
+  16057,      17821,      21365,      23665,
+  16005,      17901,      19892,      23016,
+  13232,      16683,      21107,      23221,
+  13280,      16615,      19915,      21829,
+  14950,      18575,      20599,      22511,
+  16337,      18261,      20277,      23216,
+  14306,      16477,      21203,      23158,
+  12803,      17498,      20248,      22014,
+  14327,      17068,      20160,      22006,
+  14402,      17461,      21599,      23688,
+  16968,      18834,      20896,      23055,
+  15070,      17157,      20451,      22315,
+  15419,      17107,      21601,      23946,
+  16039,      17639,      19533,      21424,
+  16326,      19261,      21745,      23673,
+  16489,      18534,      21658,      23782,
+  16594,      18471,      20549,      22807,
+  18973,      21212,      22890,      24278,
+  14264,      18674,      21123,      23071,
+  15117,      16841,      19239,      23118,
+  13762,      15782,      20478,      23230,
+  14111,      15949,      20058,      22354,
+  14990,      16738,      21139,      23492,
+  13735,      16971,      19026,      22158,
+  14676,      17314,      20232,      22807,
+  16196,      18146,      20459,      22339,
+  14747,      17258,      19315,      22437,
+  14973,      17778,      20692,      23367,
+  15715,      17472,      20385,      22349,
+  15702,      18228,      20829,      23410,
+  14428,      16188,      20541,      23630,
+  16824,      19394,      21365,      23246,
+  13069,      16392,      18900,      21121,
+  12047,      16640,      19463,      21689,
+  14757,      17433,      19659,      23125,
+  15185,      16930,      19900,      22540,
+  16026,      17725,      19618,      22399,
+  16086,      18643,      21179,      23472,
+  15462,      17248,      19102,      21196,
+  17368,      20016,      22396,      24096,
+  12340,      14475,      19665,      23362,
+  13636,      16229,      19462,      22728,
+  14096,      16211,      19591,      21635,
+  12152,      14867,      19943,      22301,
+  14492,      17503,      21002,      22728,
+  14834,      16788,      19447,      21411,
+  14650,      16433,      19326,      22308,
+  14624,      16328,      19659,      23204,
+  13888,      16572,      20665,      22488,
+  12977,      16102,      18841,      22246,
+  15523,      18431,      21757,      23738,
+  14095,      16349,      18837,      20947,
+  13266,      17809,      21088,      22839,
+  15427,      18190,      20270,      23143,
+  11859,      16753,      20935,      22486,
+  12310,      17667,      21736,      23319,
+  14021,      15926,      18702,      22002,
+  12286,      15299,      19178,      21126,
+  15703,      17491,      21039,      23151,
+  12272,      14018,      18213,      22570,
+  14817,      16364,      18485,      22598,
+  17109,      19683,      21851,      23677,
+  12657,      14903,      19039,      22061,
+  14713,      16487,      20527,      22814,
+  14635,      16726,      18763,      21715,
+  15878,      18550,      20718,      22906
+};
+
+const int16_t WebRtcIlbcfix_kLsfDimCb[LSF_NSPLIT] = {3, 3, 4};
+const int16_t WebRtcIlbcfix_kLsfSizeCb[LSF_NSPLIT] = {64,128,128};
+
+const int16_t WebRtcIlbcfix_kLsfMean[LPC_FILTERORDER] = {
+  2308,       3652,       5434,       7885,
+  10255,      12559,      15160,      17513,
+  20328,      22752};
+
+const int16_t WebRtcIlbcfix_kLspMean[LPC_FILTERORDER] = {
+  31476, 29565, 25819, 18725, 10276,
+  1236, -9049, -17600, -25884, -30618
+};
+
+/* Q14 */
+const int16_t WebRtcIlbcfix_kLsfWeight20ms[4] = {12288, 8192, 4096, 0};
+const int16_t WebRtcIlbcfix_kLsfWeight30ms[6] = {8192, 16384, 10923, 5461, 0, 0};
+
+/*
+   cos(x) in Q15
+   WebRtcIlbcfix_kCos[i] = cos(pi*i/64.0)
+   used in WebRtcIlbcfix_Lsp2Lsf()
+*/
+
+const int16_t WebRtcIlbcfix_kCos[64] = {
+  32767,  32729,  32610,  32413,  32138,  31786,  31357,  30853,
+  30274,  29622,  28899,  28106,  27246,  26320,  25330,  24279,
+  23170,  22006,  20788,  19520,  18205,  16846,  15447,  14010,
+  12540,  11039,   9512,   7962,   6393,   4808,   3212,   1608,
+  0,  -1608,  -3212,  -4808,  -6393,  -7962,  -9512, -11039,
+  -12540, -14010, -15447, -16846, -18205, -19520, -20788, -22006,
+  -23170, -24279, -25330, -26320, -27246, -28106, -28899, -29622,
+  -30274, -30853, -31357, -31786, -32138, -32413, -32610, -32729
+};
+
+/*
+   Derivative in Q19, used to interpolate between the
+   WebRtcIlbcfix_kCos[] values to get a more exact y = cos(x)
+*/
+const int16_t WebRtcIlbcfix_kCosDerivative[64] = {
+  -632,  -1893,  -3150,  -4399,  -5638,  -6863,  -8072,  -9261,
+  -10428, -11570, -12684, -13767, -14817, -15832, -16808, -17744,
+  -18637, -19486, -20287, -21039, -21741, -22390, -22986, -23526,
+  -24009, -24435, -24801, -25108, -25354, -25540, -25664, -25726,
+  -25726, -25664, -25540, -25354, -25108, -24801, -24435, -24009,
+  -23526, -22986, -22390, -21741, -21039, -20287, -19486, -18637,
+  -17744, -16808, -15832, -14817, -13767, -12684, -11570, -10428,
+  -9261,  -8072,  -6863,  -5638,  -4399,  -3150,  -1893,   -632};
+
+/*
+  Table in Q15, used for a2lsf conversion
+  WebRtcIlbcfix_kCosGrid[i] = cos((2*pi*i)/(float)(2*COS_GRID_POINTS));
+*/
+
+const int16_t WebRtcIlbcfix_kCosGrid[COS_GRID_POINTS + 1] = {
+  32760, 32723, 32588, 32364, 32051, 31651, 31164, 30591,
+  29935, 29196, 28377, 27481, 26509, 25465, 24351, 23170,
+  21926, 20621, 19260, 17846, 16384, 14876, 13327, 11743,
+  10125, 8480, 6812, 5126, 3425, 1714, 0, -1714, -3425,
+  -5126, -6812, -8480, -10125, -11743, -13327, -14876,
+  -16384, -17846, -19260, -20621, -21926, -23170, -24351,
+  -25465, -26509, -27481, -28377, -29196, -29935, -30591,
+  -31164, -31651, -32051, -32364, -32588, -32723, -32760
+};
+
+/*
+   Derivative of y = acos(x) in Q12
+   used in WebRtcIlbcfix_Lsp2Lsf()
+*/
+
+const int16_t WebRtcIlbcfix_kAcosDerivative[64] = {
+  -26887, -8812, -5323, -3813, -2979, -2444, -2081, -1811,
+  -1608, -1450, -1322, -1219, -1132, -1059, -998, -946,
+  -901, -861, -827, -797, -772, -750, -730, -713,
+  -699, -687, -677, -668, -662, -657, -654, -652,
+  -652, -654, -657, -662, -668, -677, -687, -699,
+  -713, -730, -750, -772, -797, -827, -861, -901,
+  -946, -998, -1059, -1132, -1219, -1322, -1450, -1608,
+  -1811, -2081, -2444, -2979, -3813, -5323, -8812, -26887
+};
+
+
+/* Tables for quantization of start state */
+
+/* State quantization tables */
+const int16_t WebRtcIlbcfix_kStateSq3[8] = { /* Values in Q13 */
+  -30473, -17838, -9257, -2537,
+  3639, 10893, 19958, 32636
+};
+
+/* This table defines the limits for the selection of the freqg
+   less or equal than value 0 => index = 0
+   less or equal than value k => index = k
+*/
+const int32_t WebRtcIlbcfix_kChooseFrgQuant[64] = {
+  118, 163, 222, 305, 425, 604,
+  851, 1174, 1617, 2222, 3080, 4191,
+  5525, 7215, 9193, 11540, 14397, 17604,
+  21204, 25209, 29863, 35720, 42531, 50375,
+  59162, 68845, 80108, 93754, 110326, 129488,
+  150654, 174328, 201962, 233195, 267843, 308239,
+  354503, 405988, 464251, 531550, 608652, 697516,
+  802526, 928793, 1080145, 1258120, 1481106, 1760881,
+  2111111, 2546619, 3078825, 3748642, 4563142, 5573115,
+  6887601, 8582108, 10797296, 14014513, 18625760, 25529599,
+  37302935, 58819185, 109782723, WEBRTC_SPL_WORD32_MAX
+};
+
+const int16_t WebRtcIlbcfix_kScale[64] = {
+  /* Values in Q16 */
+  29485, 25003, 21345, 18316, 15578, 13128, 10973, 9310, 7955,
+  6762, 5789, 4877, 4255, 3699, 3258, 2904, 2595, 2328,
+  2123, 1932, 1785, 1631, 1493, 1370, 1260, 1167, 1083,
+  /* Values in Q21 */
+  32081, 29611, 27262, 25229, 23432, 21803, 20226, 18883, 17609,
+  16408, 15311, 14327, 13390, 12513, 11693, 10919, 10163, 9435,
+  8739, 8100, 7424, 6813, 6192, 5648, 5122, 4639, 4207, 3798,
+  3404, 3048, 2706, 2348, 2036, 1713, 1393, 1087, 747
+};
+
+/*frgq in fixpoint, but already computed like this:
+  for(i=0; i<64; i++){
+  a = (pow(10,frgq[i])/4.5);
+  WebRtcIlbcfix_kFrgQuantMod[i] = round(a);
+  }
+
+  Value 0 :36 in Q8
+  37:58 in Q5
+  59:63 in Q3
+*/
+const int16_t WebRtcIlbcfix_kFrgQuantMod[64] = {
+  /* First 37 values in Q8 */
+  569, 671, 786, 916, 1077, 1278,
+  1529, 1802, 2109, 2481, 2898, 3440,
+  3943, 4535, 5149, 5778, 6464, 7208,
+  7904, 8682, 9397, 10285, 11240, 12246,
+  13313, 14382, 15492, 16735, 18131, 19693,
+  21280, 22912, 24624, 26544, 28432, 30488,
+  32720,
+  /* 22 values in Q5 */
+  4383, 4684, 5012, 5363, 5739, 6146,
+  6603, 7113, 7679, 8285, 9040, 9850,
+  10838, 11882, 13103, 14467, 15950, 17669,
+  19712, 22016, 24800, 28576,
+  /* 5 values in Q3 */
+  8240, 9792, 12040, 15440, 22472
+};
+
+/* Constants for codebook search and creation */
+
+/* Expansion filter to get additional cb section.
+ * Q12 and reversed compared to flp
+ */
+const int16_t WebRtcIlbcfix_kCbFiltersRev[CB_FILTERLEN]={
+  -140, 446, -755, 3302, 2922, -590, 343, -138};
+
+/* Weighting coefficients for short lags.
+ * [0.2 0.4 0.6 0.8] in Q15 */
+const int16_t WebRtcIlbcfix_kAlpha[4]={
+  6554, 13107, 19661, 26214};
+
+/* Ranges for search and filters at different subframes */
+
+const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={
+  {58,58,58}, {108,44,44}, {108,108,108}, {108,108,108}, {108,108,108}};
+
+const size_t WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147};
+
+/* Gain Quantization for the codebook gains of the 3 stages */
+
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq3[9]={
+  -16384, -10813, -5407, 0, 4096, 8192,
+  12288, 16384, 32767};
+
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq4[17]={
+  -17203, -14746, -12288, -9830, -7373, -4915,
+  -2458, 0, 2458, 4915, 7373, 9830,
+  12288, 14746, 17203, 19661, 32767};
+
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq5[33]={
+  614,        1229,        1843,        2458,        3072,       3686,
+  4301,        4915,        5530,        6144,        6758,        7373,
+  7987,        8602,        9216,        9830,       10445,       11059,
+  11674,       12288,       12902,       13517,       14131,       14746,
+  15360,       15974,       16589,       17203,       17818,       18432,
+  19046,       19661,    32767};
+
+/* Q14 gain_sq5Tbl squared in Q14 */
+const int16_t WebRtcIlbcfix_kGainSq5Sq[32] = {
+  23,   92,    207,  368,  576,  829,
+  1129,  1474,   1866,  2304,  2787,  3317,
+  3893,  4516,   5184,  5897,  6658,  7464,
+  8318,  9216,   10160,  11151,  12187,  13271,
+  14400,  15574,   16796,  18062,  19377,  20736,
+  22140,  23593
+};
+
+const int16_t* const WebRtcIlbcfix_kGain[3] =
+{WebRtcIlbcfix_kGainSq5, WebRtcIlbcfix_kGainSq4, WebRtcIlbcfix_kGainSq3};
+
+
+/* Tables for the Enhancer, using upsamling factor 4 (ENH_UPS0 = 4) */
+
+const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1]={
+  {0,    0,    0, 4096,    0,  0,   0},
+  {64, -315, 1181, 3531, -436, 77, -64},
+  {97, -509, 2464, 2464, -509, 97, -97},
+  {77, -436, 3531, 1181, -315, 64, -77}
+};
+
+const int16_t WebRtcIlbcfix_kEnhWt[3] = {
+  4800, 16384, 27968 /* Q16 */
+};
+
+const size_t WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = {
+  160, 480, 800, 1120, 1440, 1760, 2080, 2400  /* Q(-2) */
+};
+
+/* PLC table */
+
+const int16_t WebRtcIlbcfix_kPlcPerSqr[6] = { /* Grid points for square of periodiciy in Q15 */
+  839, 1343, 2048, 2998, 4247, 5849
+};
+
+const int16_t WebRtcIlbcfix_kPlcPitchFact[6] = { /* Value of y=(x^4-0.4)/(0.7-0.4) in grid points in Q15 */
+  0, 5462, 10922, 16384, 21846, 27306
+};
+
+const int16_t WebRtcIlbcfix_kPlcPfSlope[6] = { /* Slope of y=(x^4-0.4)/(0.7-0.4) in Q11 */
+  26667, 18729, 13653, 10258, 7901, 6214
+};
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.h
new file mode 100644
index 0000000000..a8645c00db
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* high pass filters */
+
+extern const int16_t WebRtcIlbcfix_kHpInCoefs[];
+extern const int16_t WebRtcIlbcfix_kHpOutCoefs[];
+
+/* Window for start state decision */
+extern const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[];
+
+/* low pass filter used for downsampling */
+extern const int16_t WebRtcIlbcfix_kLpFiltCoefs[];
+
+/* LPC analysis and quantization */
+
+extern const int16_t WebRtcIlbcfix_kLpcWin[];
+extern const int16_t WebRtcIlbcfix_kLpcAsymWin[];
+extern const int32_t WebRtcIlbcfix_kLpcLagWin[];
+extern const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[];
+extern const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[];
+extern const int16_t WebRtcIlbcfix_kLsfDimCb[];
+extern const int16_t WebRtcIlbcfix_kLsfSizeCb[];
+extern const int16_t WebRtcIlbcfix_kLsfCb[];
+extern const int16_t WebRtcIlbcfix_kLsfWeight20ms[];
+extern const int16_t WebRtcIlbcfix_kLsfWeight30ms[];
+extern const int16_t WebRtcIlbcfix_kLsfMean[];
+extern const int16_t WebRtcIlbcfix_kLspMean[];
+extern const int16_t WebRtcIlbcfix_kCos[];
+extern const int16_t WebRtcIlbcfix_kCosDerivative[];
+extern const int16_t WebRtcIlbcfix_kCosGrid[];
+extern const int16_t WebRtcIlbcfix_kAcosDerivative[];
+
+/* state quantization tables */
+
+extern const int16_t WebRtcIlbcfix_kStateSq3[];
+extern const int32_t WebRtcIlbcfix_kChooseFrgQuant[];
+extern const int16_t WebRtcIlbcfix_kScale[];
+extern const int16_t WebRtcIlbcfix_kFrgQuantMod[];
+
+/* Ranges for search and filters at different subframes */
+
+extern const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES];
+extern const size_t WebRtcIlbcfix_kFilterRange[];
+
+/* gain quantization tables */
+
+extern const int16_t WebRtcIlbcfix_kGainSq3[];
+extern const int16_t WebRtcIlbcfix_kGainSq4[];
+extern const int16_t WebRtcIlbcfix_kGainSq5[];
+extern const int16_t WebRtcIlbcfix_kGainSq5Sq[];
+extern const int16_t* const WebRtcIlbcfix_kGain[];
+
+/* adaptive codebook definitions */
+
+extern const int16_t WebRtcIlbcfix_kCbFiltersRev[];
+extern const int16_t WebRtcIlbcfix_kAlpha[];
+
+/* enhancer definitions */
+
+extern const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0]
+                                                 [ENH_FLO_MULT2_PLUS1];
+extern const int16_t WebRtcIlbcfix_kEnhWt[];
+extern const size_t WebRtcIlbcfix_kEnhPlocs[];
+
+/* PLC tables */
+
+extern const int16_t WebRtcIlbcfix_kPlcPerSqr[];
+extern const int16_t WebRtcIlbcfix_kPlcPitchFact[];
+extern const int16_t WebRtcIlbcfix_kPlcPfSlope[];
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
new file mode 100644
index 0000000000..7e21faee6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CreateAugmentedVec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "rtc_base/sanitizer.h"
+
+/*----------------------------------------------------------------*
+ *  Recreate a specific codebook vector from the augmented part.
+ *
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CreateAugmentedVec(
+    size_t index,          /* (i) Index for the augmented vector to be
+                              created */
+    const int16_t* buffer, /* (i) Pointer to the end of the codebook memory
+                              that is used for creation of the augmented
+                              codebook */
+    int16_t* cbVec) {      /* (o) The constructed codebook vector */
+  size_t ilow;
+  const int16_t *ppo, *ppi;
+  int16_t cbVecTmp[4];
+  /* Interpolation starts 4 elements before cbVec+index, but must not start
+     outside `cbVec`; clamping interp_len to stay within `cbVec`.
+   */
+  size_t interp_len = WEBRTC_SPL_MIN(index, 4);
+
+  rtc_MsanCheckInitialized(buffer - index - interp_len, sizeof(buffer[0]),
+                           index + interp_len);
+
+  ilow = index - interp_len;
+
+  /* copy the first noninterpolated part */
+  ppo = buffer-index;
+  WEBRTC_SPL_MEMCPY_W16(cbVec, ppo, index);
+
+  /* interpolation */
+  ppo = buffer - interp_len;
+  ppi = buffer - index - interp_len;
+
+  /* perform cbVec[ilow+k] = ((ppi[k]*alphaTbl[k])>>15) +
+                             ((ppo[k]*alphaTbl[interp_len-1-k])>>15);
+     for k = 0..interp_len-1
+  */
+  WebRtcSpl_ElementwiseVectorMult(&cbVec[ilow], ppi, WebRtcIlbcfix_kAlpha,
+                                  interp_len, 15);
+  WebRtcSpl_ReverseOrderMultArrayElements(
+      cbVecTmp, ppo, &WebRtcIlbcfix_kAlpha[interp_len - 1], interp_len, 15);
+  WebRtcSpl_AddVectorsAndShift(&cbVec[ilow], &cbVec[ilow], cbVecTmp, interp_len,
+                               0);
+
+  /* copy the second noninterpolated part */
+  ppo = buffer - index;
+  /* `tempbuff2` is declared in WebRtcIlbcfix_GetCbVec and is SUBL+5 elements
+     long. `buffer` points one element past the end of that vector, i.e., at
+     tempbuff2+SUBL+5. Since ppo=buffer-index, we cannot read any more than
+     `index` elements from `ppo`.
+
+     `cbVec` is declared to be SUBL elements long in WebRtcIlbcfix_CbConstruct.
+     Therefore, we can only write SUBL-index elements to cbVec+index.
+
+     These two conditions limit the number of elements to copy.
+   */
+  WEBRTC_SPL_MEMCPY_W16(cbVec+index, ppo, WEBRTC_SPL_MIN(SUBL-index, index));
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
new file mode 100644
index 0000000000..d7e5be1c2f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CreateAugmentedVec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Recreate a specific codebook vector from the augmented part.
+ *
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CreateAugmentedVec(
+    size_t index, /* (i) Index for the augmented vector to be
+                     created */
+    const int16_t* buffer, /* (i) Pointer to the end of the codebook memory
+                              that is used for creation of the augmented
+                              codebook */
+    int16_t* cbVec);       /* (o) The construced codebook vector */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c
new file mode 100644
index 0000000000..d7621d5b65
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c
@@ -0,0 +1,261 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Decode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/decode_residual.h"
+#include "modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h"
+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+#include "rtc_base/system/arch.h"
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
+#endif
+
+/*----------------------------------------------------------------*
+ *  main decoder function
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_DecodeImpl(
+    int16_t *decblock,    /* (o) decoded signal block */
+    const uint16_t *bytes, /* (i) encoded signal bits */
+    IlbcDecoder *iLBCdec_inst, /* (i/o) the decoder state
+                                           structure */
+    int16_t mode      /* (i) 0: bad packet, PLC,
+                                                                   1: normal */
+                           ) {
+  const int old_mode = iLBCdec_inst->mode;
+  const int old_use_enhancer = iLBCdec_inst->use_enhancer;
+
+  size_t i;
+  int16_t order_plus_one;
+
+  int16_t last_bit;
+  int16_t *data;
+  /* Stack based */
+  int16_t decresidual[BLOCKL_MAX];
+  int16_t PLCresidual[BLOCKL_MAX + LPC_FILTERORDER];
+  int16_t syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
+  int16_t PLClpc[LPC_FILTERORDER + 1];
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+  uint16_t swapped[NO_OF_WORDS_30MS];
+#endif
+  iLBC_bits *iLBCbits_inst = (iLBC_bits*)PLCresidual;
+
+  /* Reuse some buffers that are non overlapping in order to save stack memory */
+  data = &PLCresidual[LPC_FILTERORDER];
+
+  if (mode) { /* the data are good */
+
+    /* decode data */
+
+    /* Unpacketize bits into parameters */
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+    WebRtcIlbcfix_SwapBytes(bytes, iLBCdec_inst->no_of_words, swapped);
+    last_bit = WebRtcIlbcfix_UnpackBits(swapped, iLBCbits_inst, iLBCdec_inst->mode);
+#else
+    last_bit = WebRtcIlbcfix_UnpackBits(bytes, iLBCbits_inst, iLBCdec_inst->mode);
+#endif
+
+    /* Check for bit errors */
+    if (iLBCbits_inst->startIdx<1)
+      mode = 0;
+    if ((iLBCdec_inst->mode==20) && (iLBCbits_inst->startIdx>3))
+      mode = 0;
+    if ((iLBCdec_inst->mode==30) && (iLBCbits_inst->startIdx>5))
+      mode = 0;
+    if (last_bit==1)
+      mode = 0;
+
+    if (mode) { /* No bit errors was detected, continue decoding */
+      /* Stack based */
+      int16_t lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
+      int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+
+      /* adjust index */
+      WebRtcIlbcfix_IndexConvDec(iLBCbits_inst->cb_index);
+
+      /* decode the lsf */
+      WebRtcIlbcfix_SimpleLsfDeQ(lsfdeq, (int16_t*)(iLBCbits_inst->lsf), iLBCdec_inst->lpc_n);
+      WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCdec_inst->lpc_n);
+      WebRtcIlbcfix_DecoderInterpolateLsp(syntdenum, weightdenum,
+                                          lsfdeq, LPC_FILTERORDER, iLBCdec_inst);
+
+      /* Decode the residual using the cb and gain indexes */
+      if (!WebRtcIlbcfix_DecodeResidual(iLBCdec_inst, iLBCbits_inst,
+                                        decresidual, syntdenum))
+        goto error;
+
+      /* preparing the plc for a future loss! */
+      WebRtcIlbcfix_DoThePlc(
+          PLCresidual, PLClpc, 0, decresidual,
+          syntdenum + (LPC_FILTERORDER + 1) * (iLBCdec_inst->nsub - 1),
+          iLBCdec_inst->last_lag, iLBCdec_inst);
+
+      /* Use the output from doThePLC */
+      WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
+    }
+
+  }
+
+  if (mode == 0) {
+    /* the data is bad (either a PLC call
+     * was made or a bit error was detected)
+     */
+
+    /* packet loss conceal */
+
+    WebRtcIlbcfix_DoThePlc(PLCresidual, PLClpc, 1, decresidual, syntdenum,
+                           iLBCdec_inst->last_lag, iLBCdec_inst);
+
+    WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
+
+    order_plus_one = LPC_FILTERORDER + 1;
+
+    for (i = 0; i < iLBCdec_inst->nsub; i++) {
+      WEBRTC_SPL_MEMCPY_W16(syntdenum+(i*order_plus_one),
+                            PLClpc, order_plus_one);
+    }
+  }
+
+  if ((*iLBCdec_inst).use_enhancer == 1) { /* Enhancer activated */
+
+    /* Update the filter and filter coefficients if there was a packet loss */
+    if (iLBCdec_inst->prev_enh_pl==2) {
+      for (i=0;i<iLBCdec_inst->nsub;i++) {
+        WEBRTC_SPL_MEMCPY_W16(&(iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)]),
+                              syntdenum, (LPC_FILTERORDER+1));
+      }
+    }
+
+    /* post filtering */
+    (*iLBCdec_inst).last_lag =
+        WebRtcIlbcfix_EnhancerInterface(data, decresidual, iLBCdec_inst);
+
+    /* synthesis filtering */
+
+    /* Set up the filter state */
+    WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
+
+    if (iLBCdec_inst->mode==20) {
+      /* Enhancer has 40 samples delay */
+      i=0;
+      WebRtcSpl_FilterARFastQ12(
+          data, data,
+          iLBCdec_inst->old_syntdenum + (i+iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1),
+          LPC_FILTERORDER+1, SUBL);
+
+      for (i=1; i < iLBCdec_inst->nsub; i++) {
+        WebRtcSpl_FilterARFastQ12(
+            data+i*SUBL, data+i*SUBL,
+            syntdenum+(i-1)*(LPC_FILTERORDER+1),
+            LPC_FILTERORDER+1, SUBL);
+      }
+
+    } else if (iLBCdec_inst->mode==30) {
+      /* Enhancer has 80 samples delay */
+      for (i=0; i < 2; i++) {
+        WebRtcSpl_FilterARFastQ12(
+            data+i*SUBL, data+i*SUBL,
+            iLBCdec_inst->old_syntdenum + (i+4)*(LPC_FILTERORDER+1),
+            LPC_FILTERORDER+1, SUBL);
+      }
+      for (i=2; i < iLBCdec_inst->nsub; i++) {
+        WebRtcSpl_FilterARFastQ12(
+            data+i*SUBL, data+i*SUBL,
+            syntdenum+(i-2)*(LPC_FILTERORDER+1),
+            LPC_FILTERORDER+1, SUBL);
+      }
+    }
+
+    /* Save the filter state */
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+
+  } else { /* Enhancer not activated */
+    size_t lag;
+
+    /* Find last lag (since the enhancer is not called to give this info) */
+    lag = 20;
+    if (iLBCdec_inst->mode==20) {
+      lag = WebRtcIlbcfix_XcorrCoef(
+          &decresidual[iLBCdec_inst->blockl-60],
+          &decresidual[iLBCdec_inst->blockl-60-lag],
+          60,
+          80, lag, -1);
+    } else {
+      lag = WebRtcIlbcfix_XcorrCoef(
+          &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL],
+          &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL-lag],
+          ENH_BLOCKL,
+          100, lag, -1);
+    }
+
+    /* Store lag (it is needed if next packet is lost) */
+    (*iLBCdec_inst).last_lag = lag;
+
+    /* copy data and run synthesis filter */
+    WEBRTC_SPL_MEMCPY_W16(data, decresidual, iLBCdec_inst->blockl);
+
+    /* Set up the filter state */
+    WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
+
+    for (i=0; i < iLBCdec_inst->nsub; i++) {
+      WebRtcSpl_FilterARFastQ12(
+          data+i*SUBL, data+i*SUBL,
+          syntdenum + i*(LPC_FILTERORDER+1),
+          LPC_FILTERORDER+1, SUBL);
+    }
+
+    /* Save the filter state */
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+  }
+
+  WEBRTC_SPL_MEMCPY_W16(decblock,data,iLBCdec_inst->blockl);
+
+  /* High pass filter the signal (with upscaling a factor 2 and saturation) */
+  WebRtcIlbcfix_HpOutput(decblock, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
+                         iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+                         iLBCdec_inst->blockl);
+
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->old_syntdenum,
+                        syntdenum, iLBCdec_inst->nsub*(LPC_FILTERORDER+1));
+
+  iLBCdec_inst->prev_enh_pl=0;
+
+  if (mode==0) { /* PLC was used */
+    iLBCdec_inst->prev_enh_pl=1;
+  }
+
+  return 0;  // Success.
+
+error:
+  // The decoder got sick from eating that data. Reset it and return.
+  WebRtcIlbcfix_InitDecode(iLBCdec_inst, old_mode, old_use_enhancer);
+  return -1;  // Error
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.h
new file mode 100644
index 0000000000..a7d2910115
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Decode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  main decoder function
+ *---------------------------------------------------------------*/
+
+// Returns 0 on success, -1 on error.
+ABSL_MUST_USE_RESULT
+int WebRtcIlbcfix_DecodeImpl(
+    int16_t* decblock,         /* (o) decoded signal block */
+    const uint16_t* bytes,     /* (i) encoded signal bits */
+    IlbcDecoder* iLBCdec_inst, /* (i/o) the decoder state
+                                           structure */
+    int16_t mode               /* (i) 0: bad packet, PLC,
+                                      1: normal */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
new file mode 100644
index 0000000000..a9668e2889
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecodeResidual.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/decode_residual.h"
+
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+
+/*----------------------------------------------------------------*
+ *  frame residual decoder function (subrutine to iLBC_decode)
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_DecodeResidual(
+    IlbcDecoder *iLBCdec_inst,
+    /* (i/o) the decoder state structure */
+    iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
+                                for the decoding  */
+    int16_t *decresidual,  /* (o) decoded residual frame */
+    int16_t *syntdenum   /* (i) the decoded synthesis filter
+                                  coefficients */
+                                  ) {
+  size_t meml_gotten, diff, start_pos;
+  size_t subcount, subframe;
+  int16_t *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */
+  int16_t *memVec = iLBCdec_inst->prevResidual;  /* Memory for codebook and filter state (reuse memory in state) */
+  int16_t *mem = &memVec[CB_HALFFILTERLEN];   /* Memory for codebook */
+
+  diff = STATE_LEN - iLBCdec_inst->state_short_len;
+
+  if (iLBC_encbits->state_first == 1) {
+    start_pos = (iLBC_encbits->startIdx-1)*SUBL;
+  } else {
+    start_pos = (iLBC_encbits->startIdx-1)*SUBL + diff;
+  }
+
+  /* decode scalar part of start state */
+
+  WebRtcIlbcfix_StateConstruct(iLBC_encbits->idxForMax,
+                               iLBC_encbits->idxVec, &syntdenum[(iLBC_encbits->startIdx-1)*(LPC_FILTERORDER+1)],
+                               &decresidual[start_pos], iLBCdec_inst->state_short_len
+                               );
+
+  if (iLBC_encbits->state_first) { /* put adaptive part in the end */
+
+    /* setup memory */
+
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCdec_inst->state_short_len);
+    WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCdec_inst->state_short_len, decresidual+start_pos,
+                          iLBCdec_inst->state_short_len);
+
+    /* construct decoded vector */
+
+    if (!WebRtcIlbcfix_CbConstruct(
+            &decresidual[start_pos + iLBCdec_inst->state_short_len],
+            iLBC_encbits->cb_index, iLBC_encbits->gain_index,
+            mem + CB_MEML - ST_MEM_L_TBL, ST_MEM_L_TBL, diff))
+      return false;  // Error.
+
+  }
+  else {/* put adaptive part in the beginning */
+
+    /* setup memory */
+
+    meml_gotten = iLBCdec_inst->state_short_len;
+    WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
+                                  decresidual+start_pos, meml_gotten);
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
+
+    /* construct decoded vector */
+
+    if (!WebRtcIlbcfix_CbConstruct(reverseDecresidual, iLBC_encbits->cb_index,
+                                   iLBC_encbits->gain_index,
+                                   mem + CB_MEML - ST_MEM_L_TBL, ST_MEM_L_TBL,
+                                   diff))
+      return false;  // Error.
+
+    /* get decoded residual from reversed vector */
+
+    WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1],
+                                  reverseDecresidual, diff);
+  }
+
+  /* counter for predicted subframes */
+
+  subcount=1;
+
+  /* forward prediction of subframes */
+
+  if (iLBCdec_inst->nsub > iLBC_encbits->startIdx + 1) {
+
+    /* setup memory */
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
+    WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
+                          decresidual+(iLBC_encbits->startIdx-1)*SUBL, STATE_LEN);
+
+    /* loop over subframes to encode */
+
+    size_t Nfor = iLBCdec_inst->nsub - iLBC_encbits->startIdx - 1;
+    for (subframe=0; subframe<Nfor; subframe++) {
+
+      /* construct decoded vector */
+      if (!WebRtcIlbcfix_CbConstruct(
+              &decresidual[(iLBC_encbits->startIdx + 1 + subframe) * SUBL],
+              iLBC_encbits->cb_index + subcount * CB_NSTAGES,
+              iLBC_encbits->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+              SUBL))
+        return false;  // Error;
+
+      /* update memory */
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &decresidual[(iLBC_encbits->startIdx+1+subframe)*SUBL], SUBL);
+
+      subcount++;
+    }
+
+  }
+
+  /* backward prediction of subframes */
+
+  if (iLBC_encbits->startIdx > 1) {
+
+    /* setup memory */
+
+    meml_gotten = SUBL*(iLBCdec_inst->nsub+1-iLBC_encbits->startIdx);
+    if( meml_gotten > CB_MEML ) {
+      meml_gotten=CB_MEML;
+    }
+
+    WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
+                                  decresidual+(iLBC_encbits->startIdx-1)*SUBL, meml_gotten);
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
+
+    /* loop over subframes to decode */
+
+    size_t Nback = iLBC_encbits->startIdx - 1;
+    for (subframe=0; subframe<Nback; subframe++) {
+
+      /* construct decoded vector */
+      if (!WebRtcIlbcfix_CbConstruct(
+              &reverseDecresidual[subframe * SUBL],
+              iLBC_encbits->cb_index + subcount * CB_NSTAGES,
+              iLBC_encbits->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+              SUBL))
+        return false;  // Error.
+
+      /* update memory */
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &reverseDecresidual[subframe*SUBL], SUBL);
+
+      subcount++;
+    }
+
+    /* get decoded residual from reversed vector */
+    WebRtcSpl_MemCpyReversedOrder(decresidual+SUBL*Nback-1,
+                                  reverseDecresidual, SUBL*Nback);
+  }
+
+  return true;  // Success.
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
new file mode 100644
index 0000000000..d079577661
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecodeResidual.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+
+#include <stdbool.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  frame residual decoder function (subrutine to iLBC_decode)
+ *---------------------------------------------------------------*/
+
+// Returns true on success, false on failure. In case of failure, the decoder
+// state may be corrupted and needs resetting.
+ABSL_MUST_USE_RESULT
+bool WebRtcIlbcfix_DecodeResidual(
+    IlbcDecoder* iLBCdec_inst, /* (i/o) the decoder state structure */
+    iLBC_bits* iLBC_encbits,   /* (i/o) Encoded bits, which are used
+                                        for the decoding  */
+    int16_t* decresidual,      /* (o) decoded residual frame */
+    int16_t* syntdenum         /* (i) the decoded synthesis filter
+                                                         coefficients */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
new file mode 100644
index 0000000000..d96bb9b2e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecoderInterpolateLsp.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h"
+
+/*----------------------------------------------------------------*
+ *  obtain synthesis and weighting filters form lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecoderInterpolateLsp(
+    int16_t *syntdenum,  /* (o) synthesis filter coefficients */
+    int16_t *weightdenum, /* (o) weighting denumerator
+                                   coefficients */
+    int16_t *lsfdeq,   /* (i) dequantized lsf coefficients */
+    int16_t length,   /* (i) length of lsf coefficient vector */
+    IlbcDecoder *iLBCdec_inst
+    /* (i) the decoder state structure */
+                                          ){
+  size_t i;
+  int pos, lp_length;
+  int16_t  lp[LPC_FILTERORDER + 1], *lsfdeq2;
+
+  lsfdeq2 = lsfdeq + length;
+  lp_length = length + 1;
+
+  if (iLBCdec_inst->mode==30) {
+    /* subframe 1: Interpolation between old and first LSF */
+
+    WebRtcIlbcfix_LspInterpolate2PolyDec(lp, (*iLBCdec_inst).lsfdeqold, lsfdeq,
+                                         WebRtcIlbcfix_kLsfWeight30ms[0], length);
+    WEBRTC_SPL_MEMCPY_W16(syntdenum,lp,lp_length);
+    WebRtcIlbcfix_BwExpand(weightdenum, lp, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
+
+    /* subframes 2 to 6: interpolation between first and last LSF */
+
+    pos = lp_length;
+    for (i = 1; i < 6; i++) {
+      WebRtcIlbcfix_LspInterpolate2PolyDec(lp, lsfdeq, lsfdeq2,
+                                           WebRtcIlbcfix_kLsfWeight30ms[i], length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum + pos,lp,lp_length);
+      WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
+      pos += lp_length;
+    }
+  } else { /* iLBCdec_inst->mode=20 */
+    /* subframes 1 to 4: interpolation between old and new LSF */
+    pos = 0;
+    for (i = 0; i < iLBCdec_inst->nsub; i++) {
+      WebRtcIlbcfix_LspInterpolate2PolyDec(lp, iLBCdec_inst->lsfdeqold, lsfdeq,
+                                           WebRtcIlbcfix_kLsfWeight20ms[i], length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum+pos,lp,lp_length);
+      WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
+      pos += lp_length;
+    }
+  }
+
+  /* update memory */
+
+  if (iLBCdec_inst->mode==30) {
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq2, length);
+  } else {
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq, length);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
new file mode 100644
index 0000000000..8b08114467
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecoderInterpolateLsp.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  obtain synthesis and weighting filters form lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecoderInterpolateLsp(
+    int16_t* syntdenum, /* (o) synthesis filter coefficients */
+    int16_t* weightdenum, /* (o) weighting denumerator
+                                   coefficients */
+    int16_t* lsfdeq, /* (i) dequantized lsf coefficients */
+    int16_t length,  /* (i) length of lsf coefficient vector */
+    IlbcDecoder* iLBCdec_inst
+    /* (i) the decoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/defines.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/defines.h
new file mode 100644
index 0000000000..64135c4887
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/defines.h
@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ define.h
+
+******************************************************************/
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+
+#include <stdint.h>
+#include <string.h>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+/* general codec settings */
+
+#define FS 8000
+#define BLOCKL_20MS 160
+#define BLOCKL_30MS 240
+#define BLOCKL_MAX 240
+#define NSUB_20MS 4
+#define NSUB_30MS 6
+#define NSUB_MAX 6
+#define NASUB_20MS 2
+#define NASUB_30MS 4
+#define NASUB_MAX 4
+#define SUBL 40
+#define STATE_LEN 80
+#define STATE_SHORT_LEN_30MS 58
+#define STATE_SHORT_LEN_20MS 57
+
+/* LPC settings */
+
+#define LPC_FILTERORDER 10
+#define LPC_LOOKBACK 60
+#define LPC_N_20MS 1
+#define LPC_N_30MS 2
+#define LPC_N_MAX 2
+#define LPC_ASYMDIFF 20
+#define LSF_NSPLIT 3
+#define LSF_NUMBER_OF_STEPS 4
+#define LPC_HALFORDER 5
+#define COS_GRID_POINTS 60
+
+/* cb settings */
+
+#define CB_NSTAGES 3
+#define CB_EXPAND 2
+#define CB_MEML 147
+#define CB_FILTERLEN (2 * 4)
+#define CB_HALFFILTERLEN 4
+#define CB_RESRANGE 34
+#define CB_MAXGAIN_FIXQ6 83 /* error = -0.24% */
+#define CB_MAXGAIN_FIXQ14 21299
+
+/* enhancer */
+
+#define ENH_BLOCKL 80 /* block length */
+#define ENH_BLOCKL_HALF (ENH_BLOCKL / 2)
+#define ENH_HL                                                         \
+  3 /* 2*ENH_HL+1 is number blocks                                     \
+                                                        in said second \
+       sequence */
+#define ENH_SLOP                    \
+  2 /* max difference estimated and \
+                                                       correct pitch period */
+#define ENH_PLOCSL                                                          \
+  8 /* pitch-estimates and                                                  \
+                                                     pitch-locations buffer \
+       length */
+#define ENH_OVERHANG 2
+#define ENH_UPS0 4 /* upsampling rate */
+#define ENH_FL0 3  /* 2*FLO+1 is the length of each filter */
+#define ENH_FLO_MULT2_PLUS1 7
+#define ENH_VECTL (ENH_BLOCKL + 2 * ENH_FL0)
+#define ENH_CORRDIM (2 * ENH_SLOP + 1)
+#define ENH_NBLOCKS (BLOCKL / ENH_BLOCKL)
+#define ENH_NBLOCKS_EXTRA 5
+#define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS+ENH_NBLOCKS_EXTRA */
+#define ENH_BUFL (ENH_NBLOCKS_TOT) * ENH_BLOCKL
+#define ENH_BUFL_FILTEROVERHEAD 3
+#define ENH_A0 819                      /* Q14 */
+#define ENH_A0_MINUS_A0A0DIV4 848256041 /* Q34 */
+#define ENH_A0DIV2 26843546             /* Q30 */
+
+/* PLC */
+
+/* Down sampling */
+
+#define FILTERORDER_DS_PLUS1 7
+#define DELAY_DS 3
+#define FACTOR_DS 2
+
+/* bit stream defs */
+
+#define NO_OF_BYTES_20MS 38
+#define NO_OF_BYTES_30MS 50
+#define NO_OF_WORDS_20MS 19
+#define NO_OF_WORDS_30MS 25
+#define STATE_BITS 3
+#define BYTE_LEN 8
+#define ULP_CLASSES 3
+
+/* help parameters */
+
+#define TWO_PI_FIX 25736 /* Q12 */
+
+/* Constants for codebook search and creation */
+
+#define ST_MEM_L_TBL 85
+#define MEM_LF_TBL 147
+
+/* Struct for the bits */
+typedef struct iLBC_bits_t_ {
+  int16_t lsf[LSF_NSPLIT * LPC_N_MAX];
+  int16_t cb_index[CB_NSTAGES * (NASUB_MAX + 1)];   /* First CB_NSTAGES values
+                                                       contains extra CB index */
+  int16_t gain_index[CB_NSTAGES * (NASUB_MAX + 1)]; /* First CB_NSTAGES values
+                                                       contains extra CB gain */
+  size_t idxForMax;
+  int16_t state_first;
+  int16_t idxVec[STATE_SHORT_LEN_30MS];
+  int16_t firstbits;
+  size_t startIdx;
+} iLBC_bits;
+
+/* type definition encoder instance */
+typedef struct IlbcEncoder_ {
+  /* flag for frame size mode */
+  int16_t mode;
+
+  /* basic parameters for different frame sizes */
+  size_t blockl;
+  size_t nsub;
+  int16_t nasub;
+  size_t no_of_bytes, no_of_words;
+  int16_t lpc_n;
+  size_t state_short_len;
+
+  /* analysis filter state */
+  int16_t anaMem[LPC_FILTERORDER];
+
+  /* Fix-point old lsf parameters for interpolation */
+  int16_t lsfold[LPC_FILTERORDER];
+  int16_t lsfdeqold[LPC_FILTERORDER];
+
+  /* signal buffer for LP analysis */
+  int16_t lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
+
+  /* state of input HP filter */
+  int16_t hpimemx[2];
+  int16_t hpimemy[4];
+
+#ifdef SPLIT_10MS
+  int16_t weightdenumbuf[66];
+  int16_t past_samples[160];
+  uint16_t bytes[25];
+  int16_t section;
+  int16_t Nfor_flag;
+  int16_t Nback_flag;
+  int16_t start_pos;
+  size_t diff;
+#endif
+
+} IlbcEncoder;
+
+/* type definition decoder instance */
+typedef struct IlbcDecoder_ {
+  /* flag for frame size mode */
+  int16_t mode;
+
+  /* basic parameters for different frame sizes */
+  size_t blockl;
+  size_t nsub;
+  int16_t nasub;
+  size_t no_of_bytes, no_of_words;
+  int16_t lpc_n;
+  size_t state_short_len;
+
+  /* synthesis filter state */
+  int16_t syntMem[LPC_FILTERORDER];
+
+  /* old LSF for interpolation */
+  int16_t lsfdeqold[LPC_FILTERORDER];
+
+  /* pitch lag estimated in enhancer and used in PLC */
+  size_t last_lag;
+
+  /* PLC state information */
+  int consPLICount, prev_enh_pl;
+  int16_t perSquare;
+
+  int16_t prevScale, prevPLI;
+  size_t prevLag;
+  int16_t prevLpc[LPC_FILTERORDER + 1];
+  int16_t prevResidual[NSUB_MAX * SUBL];
+  int16_t seed;
+
+  /* previous synthesis filter parameters */
+
+  int16_t old_syntdenum[(LPC_FILTERORDER + 1) * NSUB_MAX];
+
+  /* state of output HP filter */
+  int16_t hpimemx[2];
+  int16_t hpimemy[4];
+
+  /* enhancer state information */
+  int use_enhancer;
+  int16_t enh_buf[ENH_BUFL + ENH_BUFL_FILTEROVERHEAD];
+  size_t enh_period[ENH_NBLOCKS_TOT];
+
+} IlbcDecoder;
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c
new file mode 100644
index 0000000000..9ca6ca48e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c
@@ -0,0 +1,309 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DoThePlc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/comp_corr.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Packet loss concealment routine. Conceals a residual signal
+ *  and LP parameters. If no packet loss, update state.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DoThePlc(
+    int16_t *PLCresidual,  /* (o) concealed residual */
+    int16_t *PLClpc,    /* (o) concealed LP parameters */
+    int16_t PLI,     /* (i) packet loss indicator
+                                                           0 - no PL, 1 = PL */
+    int16_t *decresidual,  /* (i) decoded residual */
+    int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
+    size_t inlag,    /* (i) pitch lag */
+    IlbcDecoder *iLBCdec_inst
+    /* (i/o) decoder instance */
+                            ){
+  size_t i;
+  int32_t cross, ener, cross_comp, ener_comp = 0;
+  int32_t measure, maxMeasure, energy;
+  int32_t noise_energy_threshold_30dB;
+  int16_t max, crossSquareMax, crossSquare;
+  size_t j, lag, randlag;
+  int16_t tmp1, tmp2;
+  int16_t shift1, shift2, shift3, shiftMax;
+  int16_t scale3;
+  size_t corrLen;
+  int32_t tmpW32, tmp2W32;
+  int16_t use_gain;
+  int16_t tot_gain;
+  int16_t max_perSquare;
+  int16_t scale1, scale2;
+  int16_t totscale;
+  int32_t nom;
+  int16_t denom;
+  int16_t pitchfact;
+  size_t use_lag;
+  int ind;
+  int16_t randvec[BLOCKL_MAX];
+
+  /* Packet Loss */
+  if (PLI == 1) {
+
+    (*iLBCdec_inst).consPLICount += 1;
+
+    /* if previous frame not lost,
+       determine pitch pred. gain */
+
+    if (iLBCdec_inst->prevPLI != 1) {
+
+      /* Maximum 60 samples are correlated, preserve as high accuracy
+         as possible without getting overflow */
+      max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual,
+                                     iLBCdec_inst->blockl);
+      scale3 = (WebRtcSpl_GetSizeInBits(max)<<1) - 25;
+      if (scale3 < 0) {
+        scale3 = 0;
+      }
+
+      /* Store scale for use when interpolating between the
+       * concealment and the received packet */
+      iLBCdec_inst->prevScale = scale3;
+
+      /* Search around the previous lag +/-3 to find the
+         best pitch period */
+      lag = inlag - 3;
+
+      /* Guard against getting outside the frame */
+      corrLen = (size_t)WEBRTC_SPL_MIN(60, iLBCdec_inst->blockl-(inlag+3));
+
+      WebRtcIlbcfix_CompCorr( &cross, &ener,
+                              iLBCdec_inst->prevResidual, lag, iLBCdec_inst->blockl, corrLen, scale3);
+
+      /* Normalize and store cross^2 and the number of shifts */
+      shiftMax = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross))-15;
+      crossSquareMax = (int16_t)((
+          (int16_t)WEBRTC_SPL_SHIFT_W32(cross, -shiftMax) *
+          (int16_t)WEBRTC_SPL_SHIFT_W32(cross, -shiftMax)) >> 15);
+
+      for (j=inlag-2;j<=inlag+3;j++) {
+        WebRtcIlbcfix_CompCorr( &cross_comp, &ener_comp,
+                                iLBCdec_inst->prevResidual, j, iLBCdec_inst->blockl, corrLen, scale3);
+
+        /* Use the criteria (corr*corr)/energy to compare if
+           this lag is better or not. To avoid the division,
+           do a cross multiplication */
+        shift1 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross_comp))-15;
+        crossSquare = (int16_t)((
+            (int16_t)WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1) *
+            (int16_t)WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1)) >> 15);
+
+        shift2 = WebRtcSpl_GetSizeInBits(ener)-15;
+        measure = (int16_t)WEBRTC_SPL_SHIFT_W32(ener, -shift2) * crossSquare;
+
+        shift3 = WebRtcSpl_GetSizeInBits(ener_comp)-15;
+        maxMeasure = (int16_t)WEBRTC_SPL_SHIFT_W32(ener_comp, -shift3) *
+            crossSquareMax;
+
+        /* Calculate shift value, so that the two measures can
+           be put in the same Q domain */
+        if(2 * shiftMax + shift3 > 2 * shift1 + shift2) {
+          tmp1 =
+              WEBRTC_SPL_MIN(31, 2 * shiftMax + shift3 - 2 * shift1 - shift2);
+          tmp2 = 0;
+        } else {
+          tmp1 = 0;
+          tmp2 =
+              WEBRTC_SPL_MIN(31, 2 * shift1 + shift2 - 2 * shiftMax - shift3);
+        }
+
+        if ((measure>>tmp1) > (maxMeasure>>tmp2)) {
+          /* New lag is better => record lag, measure and domain */
+          lag = j;
+          crossSquareMax = crossSquare;
+          cross = cross_comp;
+          shiftMax = shift1;
+          ener = ener_comp;
+        }
+      }
+
+      /* Calculate the periodicity for the lag with the maximum correlation.
+
+         Definition of the periodicity:
+         abs(corr(vec1, vec2))/(sqrt(energy(vec1))*sqrt(energy(vec2)))
+
+         Work in the Square domain to simplify the calculations
+         max_perSquare is less than 1 (in Q15)
+      */
+      tmp2W32=WebRtcSpl_DotProductWithScale(&iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
+                                            &iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
+                                            corrLen, scale3);
+
+      if ((tmp2W32>0)&&(ener_comp>0)) {
+        /* norm energies to int16_t, compute the product of the energies and
+           use the upper int16_t as the denominator */
+
+        scale1=(int16_t)WebRtcSpl_NormW32(tmp2W32)-16;
+        tmp1=(int16_t)WEBRTC_SPL_SHIFT_W32(tmp2W32, scale1);
+
+        scale2=(int16_t)WebRtcSpl_NormW32(ener)-16;
+        tmp2=(int16_t)WEBRTC_SPL_SHIFT_W32(ener, scale2);
+        denom = (int16_t)((tmp1 * tmp2) >> 16);  /* in Q(scale1+scale2-16) */
+
+        /* Square the cross correlation and norm it such that max_perSquare
+           will be in Q15 after the division */
+
+        totscale = scale1+scale2-1;
+        tmp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, (totscale>>1));
+        tmp2 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, totscale-(totscale>>1));
+
+        nom = tmp1 * tmp2;
+        max_perSquare = (int16_t)WebRtcSpl_DivW32W16(nom, denom);
+
+      } else {
+        max_perSquare = 0;
+      }
+    }
+
+    /* previous frame lost, use recorded lag and gain */
+
+    else {
+      lag = iLBCdec_inst->prevLag;
+      max_perSquare = iLBCdec_inst->perSquare;
+    }
+
+    /* Attenuate signal and scale down pitch pred gain if
+       several frames lost consecutively */
+
+    use_gain = 32767;   /* 1.0 in Q15 */
+
+    if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>320) {
+      use_gain = 29491;  /* 0.9 in Q15 */
+    } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>640) {
+      use_gain = 22938;  /* 0.7 in Q15 */
+    } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>960) {
+      use_gain = 16384;  /* 0.5 in Q15 */
+    } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>1280) {
+      use_gain = 0;   /* 0.0 in Q15 */
+    }
+
+    /* Compute mixing factor of picth repeatition and noise:
+       for max_per>0.7 set periodicity to 1.0
+       0.4<max_per<0.7 set periodicity to (maxper-0.4)/0.7-0.4)
+       max_per<0.4 set periodicity to 0.0
+    */
+
+    if (max_perSquare>7868) { /* periodicity > 0.7  (0.7^4=0.2401 in Q15) */
+      pitchfact = 32767;
+    } else if (max_perSquare>839) { /* 0.4 < periodicity < 0.7 (0.4^4=0.0256 in Q15) */
+      /* find best index and interpolate from that */
+      ind = 5;
+      while ((max_perSquare<WebRtcIlbcfix_kPlcPerSqr[ind])&&(ind>0)) {
+        ind--;
+      }
+      /* pitch fact is approximated by first order */
+      tmpW32 = (int32_t)WebRtcIlbcfix_kPlcPitchFact[ind] +
+          ((WebRtcIlbcfix_kPlcPfSlope[ind] *
+              (max_perSquare - WebRtcIlbcfix_kPlcPerSqr[ind])) >> 11);
+
+      pitchfact = (int16_t)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */
+
+    } else { /* periodicity < 0.4 */
+      pitchfact = 0;
+    }
+
+    /* avoid repetition of same pitch cycle (buzzyness) */
+    use_lag = lag;
+    if (lag<80) {
+      use_lag = 2*lag;
+    }
+
+    /* compute concealed residual */
+    noise_energy_threshold_30dB = (int32_t)iLBCdec_inst->blockl * 900;
+    energy = 0;
+    for (i=0; i<iLBCdec_inst->blockl; i++) {
+
+      /* noise component -  52 < randlagFIX < 117 */
+      iLBCdec_inst->seed = (int16_t)(iLBCdec_inst->seed * 31821 + 13849);
+      randlag = 53 + (iLBCdec_inst->seed & 63);
+      if (randlag > i) {
+        randvec[i] =
+            iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - randlag];
+      } else {
+        randvec[i] = iLBCdec_inst->prevResidual[i - randlag];
+      }
+
+      /* pitch repeatition component */
+      if (use_lag > i) {
+        PLCresidual[i] =
+            iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - use_lag];
+      } else {
+        PLCresidual[i] = PLCresidual[i - use_lag];
+      }
+
+      /* Attinuate total gain for each 10 ms */
+      if (i<80) {
+        tot_gain=use_gain;
+      } else if (i<160) {
+        tot_gain = (int16_t)((31130 * use_gain) >> 15);  /* 0.95*use_gain */
+      } else {
+        tot_gain = (int16_t)((29491 * use_gain) >> 15);  /* 0.9*use_gain */
+      }
+
+
+      /* mix noise and pitch repeatition */
+      PLCresidual[i] = (int16_t)((tot_gain *
+          ((pitchfact * PLCresidual[i] + (32767 - pitchfact) * randvec[i] +
+              16384) >> 15)) >> 15);
+
+      /* Compute energy until threshold for noise energy is reached */
+      if (energy < noise_energy_threshold_30dB) {
+        energy += PLCresidual[i] * PLCresidual[i];
+      }
+    }
+
+    /* less than 30 dB, use only noise */
+    if (energy < noise_energy_threshold_30dB) {
+      for (i=0; i<iLBCdec_inst->blockl; i++) {
+        PLCresidual[i] = randvec[i];
+      }
+    }
+
+    /* use the old LPC */
+    WEBRTC_SPL_MEMCPY_W16(PLClpc, (*iLBCdec_inst).prevLpc, LPC_FILTERORDER+1);
+
+    /* Update state in case there are multiple frame losses */
+    iLBCdec_inst->prevLag = lag;
+    iLBCdec_inst->perSquare = max_perSquare;
+  }
+
+  /* no packet loss, copy input */
+
+  else {
+    WEBRTC_SPL_MEMCPY_W16(PLCresidual, decresidual, iLBCdec_inst->blockl);
+    WEBRTC_SPL_MEMCPY_W16(PLClpc, lpc, (LPC_FILTERORDER+1));
+    iLBCdec_inst->consPLICount = 0;
+  }
+
+  /* update state */
+  iLBCdec_inst->prevPLI = PLI;
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevLpc, PLClpc, (LPC_FILTERORDER+1));
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevResidual, PLCresidual, iLBCdec_inst->blockl);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.h
new file mode 100644
index 0000000000..c19c4eca32
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DoThePlc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Packet loss concealment routine. Conceals a residual signal
+ *  and LP parameters. If no packet loss, update state.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DoThePlc(
+    int16_t* PLCresidual, /* (o) concealed residual */
+    int16_t* PLClpc,      /* (o) concealed LP parameters */
+    int16_t PLI,          /* (i) packet loss indicator
+                                                                0 - no PL, 1 = PL */
+    int16_t* decresidual, /* (i) decoded residual */
+    int16_t* lpc,         /* (i) decoded LPC (only used for no PL) */
+    size_t inlag,         /* (i) pitch lag */
+    IlbcDecoder* iLBCdec_inst
+    /* (i/o) decoder instance */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c
new file mode 100644
index 0000000000..8e536221cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c
@@ -0,0 +1,517 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Encode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+#include "modules/audio_coding/codecs/ilbc/cb_search.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/frame_classify.h"
+#include "modules/audio_coding/codecs/ilbc/hp_input.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_enc.h"
+#include "modules/audio_coding/codecs/ilbc/lpc_encode.h"
+#include "modules/audio_coding/codecs/ilbc/pack_bits.h"
+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
+#include "modules/audio_coding/codecs/ilbc/state_search.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/arch.h"
+
+#ifdef SPLIT_10MS
+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+#endif
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
+#endif
+
+/*----------------------------------------------------------------*
+ *  main encoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EncodeImpl(
+    uint16_t *bytes,     /* (o) encoded data bits iLBC */
+    const int16_t *block, /* (i) speech vector to encode */
+    IlbcEncoder *iLBCenc_inst /* (i/o) the general encoder
+                                     state */
+                          ){
+  size_t n, meml_gotten, Nfor;
+  size_t diff, start_pos;
+  size_t index;
+  size_t subcount, subframe;
+  size_t start_count, end_count;
+  int16_t *residual;
+  int32_t en1, en2;
+  int16_t scale, max;
+  int16_t *syntdenum;
+  int16_t *decresidual;
+  int16_t *reverseResidual;
+  int16_t *reverseDecresidual;
+  /* Stack based */
+  int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+  int16_t dataVec[BLOCKL_MAX + LPC_FILTERORDER];
+  int16_t memVec[CB_MEML+CB_FILTERLEN];
+  int16_t bitsMemory[sizeof(iLBC_bits)/sizeof(int16_t)];
+  iLBC_bits *iLBCbits_inst = (iLBC_bits*)bitsMemory;
+
+
+#ifdef SPLIT_10MS
+  int16_t *weightdenumbuf = iLBCenc_inst->weightdenumbuf;
+  int16_t last_bit;
+#endif
+
+  int16_t *data = &dataVec[LPC_FILTERORDER];
+  int16_t *mem = &memVec[CB_HALFFILTERLEN];
+
+  /* Reuse som buffers to save stack memory */
+  residual = &iLBCenc_inst->lpc_buffer[LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl];
+  syntdenum = mem;      /* syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX] and mem are used non overlapping in the code */
+  decresidual = residual;     /* Already encoded residual is overwritten by the decoded version */
+  reverseResidual = data;     /* data and reverseResidual are used non overlapping in the code */
+  reverseDecresidual = reverseResidual; /* Already encoded residual is overwritten by the decoded version */
+
+#ifdef SPLIT_10MS
+
+  WebRtcSpl_MemSetW16 (  (int16_t *) iLBCbits_inst, 0,
+                         sizeof(iLBC_bits) / sizeof(int16_t)  );
+
+  start_pos = iLBCenc_inst->start_pos;
+  diff = iLBCenc_inst->diff;
+
+  if (iLBCenc_inst->section != 0){
+    WEBRTC_SPL_MEMCPY_W16 (weightdenum, weightdenumbuf,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+    /* Un-Packetize the frame into parameters */
+    last_bit = WebRtcIlbcfix_UnpackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+    if (last_bit)
+      return;
+    /* adjust index */
+    WebRtcIlbcfix_IndexConvDec (iLBCbits_inst->cb_index);
+
+    if (iLBCenc_inst->section == 1){
+      /* Save first 80 samples of a 160/240 sample frame for 20/30msec */
+      WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples, block, 80);
+    }
+    else{ // iLBCenc_inst->section == 2 AND mode = 30ms
+      /* Save second 80 samples of a 240 sample frame for 30msec */
+      WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples + 80, block, 80);
+    }
+  }
+  else{ // iLBCenc_inst->section == 0
+    /* form a complete frame of 160/240 for 20msec/30msec mode */
+    WEBRTC_SPL_MEMCPY_W16 (data + (iLBCenc_inst->mode * 8) - 80, block, 80);
+    WEBRTC_SPL_MEMCPY_W16 (data, iLBCenc_inst->past_samples,
+                           (iLBCenc_inst->mode * 8) - 80);
+    iLBCenc_inst->Nfor_flag = 0;
+    iLBCenc_inst->Nback_flag = 0;
+#else
+    /* copy input block to data*/
+    WEBRTC_SPL_MEMCPY_W16(data,block,iLBCenc_inst->blockl);
+#endif
+
+    /* high pass filtering of input signal and scale down the residual (*0.5) */
+    WebRtcIlbcfix_HpInput(data, (int16_t*)WebRtcIlbcfix_kHpInCoefs,
+                          iLBCenc_inst->hpimemy, iLBCenc_inst->hpimemx,
+                          iLBCenc_inst->blockl);
+
+    /* LPC of hp filtered input data */
+    WebRtcIlbcfix_LpcEncode(syntdenum, weightdenum, iLBCbits_inst->lsf, data,
+                            iLBCenc_inst);
+
+    /* Set up state */
+    WEBRTC_SPL_MEMCPY_W16(dataVec, iLBCenc_inst->anaMem, LPC_FILTERORDER);
+
+    /* inverse filter to get residual */
+    for (n=0; n<iLBCenc_inst->nsub; n++ ) {
+      WebRtcSpl_FilterMAFastQ12(
+          &data[n*SUBL], &residual[n*SUBL],
+          &syntdenum[n*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, SUBL);
+    }
+
+    /* Copy the state for next frame */
+    WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->anaMem, &data[iLBCenc_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+
+    /* find state location */
+
+    iLBCbits_inst->startIdx = WebRtcIlbcfix_FrameClassify(iLBCenc_inst,residual);
+
+    /* check if state should be in first or last part of the
+       two subframes */
+
+    index = (iLBCbits_inst->startIdx-1)*SUBL;
+    max=WebRtcSpl_MaxAbsValueW16(&residual[index], 2*SUBL);
+    scale = WebRtcSpl_GetSizeInBits((uint32_t)(max * max));
+
+    /* Scale to maximum 25 bits so that the MAC won't cause overflow */
+    scale = scale - 25;
+    if(scale < 0) {
+      scale = 0;
+    }
+
+    diff = STATE_LEN - iLBCenc_inst->state_short_len;
+    en1=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
+                                      iLBCenc_inst->state_short_len, scale);
+    index += diff;
+    en2=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
+                                      iLBCenc_inst->state_short_len, scale);
+    if (en1 > en2) {
+      iLBCbits_inst->state_first = 1;
+      start_pos = (iLBCbits_inst->startIdx-1)*SUBL;
+    } else {
+      iLBCbits_inst->state_first = 0;
+      start_pos = (iLBCbits_inst->startIdx-1)*SUBL + diff;
+    }
+
+    /* scalar quantization of state */
+
+    WebRtcIlbcfix_StateSearch(iLBCenc_inst, iLBCbits_inst, &residual[start_pos],
+                              &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+                              &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)]);
+
+    WebRtcIlbcfix_StateConstruct(iLBCbits_inst->idxForMax, iLBCbits_inst->idxVec,
+                                 &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+                                 &decresidual[start_pos], iLBCenc_inst->state_short_len
+                                 );
+
+    /* predictive quantization in state */
+
+    if (iLBCbits_inst->state_first) { /* put adaptive part in the end */
+
+      /* setup memory */
+
+      WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCenc_inst->state_short_len,
+                            decresidual+start_pos, iLBCenc_inst->state_short_len);
+
+      /* encode subframes */
+
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+                             &residual[start_pos+iLBCenc_inst->state_short_len],
+                             mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
+                             &weightdenum[iLBCbits_inst->startIdx*(LPC_FILTERORDER+1)], 0);
+
+      /* construct decoded vector */
+
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+          &decresidual[start_pos + iLBCenc_inst->state_short_len],
+          iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+          mem + CB_MEML - ST_MEM_L_TBL, ST_MEM_L_TBL, diff));
+
+    }
+    else { /* put adaptive part in the beginning */
+
+      /* create reversed vectors for prediction */
+
+      WebRtcSpl_MemCpyReversedOrder(&reverseResidual[diff-1],
+                                    &residual[(iLBCbits_inst->startIdx+1)*SUBL-STATE_LEN], diff);
+
+      /* setup memory */
+
+      meml_gotten = iLBCenc_inst->state_short_len;
+      WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[start_pos], meml_gotten);
+      WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
+
+      /* encode subframes */
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+                             reverseResidual, mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
+                             &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+                             0);
+
+      /* construct decoded vector */
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+            reverseDecresidual, iLBCbits_inst->cb_index,
+            iLBCbits_inst->gain_index, mem + CB_MEML - ST_MEM_L_TBL,
+            ST_MEM_L_TBL, diff));
+
+      /* get decoded residual from reversed vector */
+
+      WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1], reverseDecresidual, diff);
+    }
+
+#ifdef SPLIT_10MS
+    iLBCenc_inst->start_pos = start_pos;
+    iLBCenc_inst->diff = diff;
+    iLBCenc_inst->section++;
+    /* adjust index */
+    WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
+    /* Packetize the parameters into the frame */
+    WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+    WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+    return;
+  }
+#endif
+
+  /* forward prediction of subframes */
+
+  Nfor = iLBCenc_inst->nsub-iLBCbits_inst->startIdx-1;
+
+  /* counter for predicted subframes */
+#ifdef SPLIT_10MS
+  if (iLBCenc_inst->mode == 20)
+  {
+    subcount = 1;
+  }
+  if (iLBCenc_inst->mode == 30)
+  {
+    if (iLBCenc_inst->section == 1)
+    {
+      subcount = 1;
+    }
+    if (iLBCenc_inst->section == 2)
+    {
+      subcount = 3;
+    }
+  }
+#else
+  subcount=1;
+#endif
+
+  if( Nfor > 0 ){
+
+    /* setup memory */
+
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
+    WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
+                          decresidual+(iLBCbits_inst->startIdx-1)*SUBL, STATE_LEN);
+
+#ifdef SPLIT_10MS
+    if (iLBCenc_inst->Nfor_flag > 0)
+    {
+      for (subframe = 0; subframe < WEBRTC_SPL_MIN (Nfor, 2); subframe++)
+      {
+        /* update memory */
+        WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
+        WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
+                               &decresidual[(iLBCbits_inst->startIdx + 1 +
+                                             subframe) * SUBL], SUBL);
+      }
+    }
+
+    iLBCenc_inst->Nfor_flag++;
+
+    if (iLBCenc_inst->mode == 20)
+    {
+      start_count = 0;
+      end_count = Nfor;
+    }
+    if (iLBCenc_inst->mode == 30)
+    {
+      if (iLBCenc_inst->section == 1)
+      {
+        start_count = 0;
+        end_count = WEBRTC_SPL_MIN (Nfor, (size_t)2);
+      }
+      if (iLBCenc_inst->section == 2)
+      {
+        start_count = WEBRTC_SPL_MIN (Nfor, (size_t)2);
+        end_count = Nfor;
+      }
+    }
+#else
+    start_count = 0;
+    end_count = Nfor;
+#endif
+
+    /* loop over subframes to encode */
+
+    for (subframe = start_count; subframe < end_count; subframe++){
+
+      /* encode subframe */
+
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+                             iLBCbits_inst->gain_index+subcount*CB_NSTAGES,
+                             &residual[(iLBCbits_inst->startIdx+1+subframe)*SUBL],
+                             mem, MEM_LF_TBL, SUBL,
+                             &weightdenum[(iLBCbits_inst->startIdx+1+subframe)*(LPC_FILTERORDER+1)],
+                             subcount);
+
+      /* construct decoded vector */
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+            &decresidual[(iLBCbits_inst->startIdx + 1 + subframe) * SUBL],
+            iLBCbits_inst->cb_index + subcount * CB_NSTAGES,
+            iLBCbits_inst->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+            SUBL));
+
+      /* update memory */
+
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &decresidual[(iLBCbits_inst->startIdx+1+subframe)*SUBL], SUBL);
+
+      subcount++;
+    }
+  }
+
+#ifdef SPLIT_10MS
+  if ((iLBCenc_inst->section == 1) &&
+      (iLBCenc_inst->mode == 30) && (Nfor > 0) && (end_count == 2))
+  {
+    iLBCenc_inst->section++;
+    /* adjust index */
+    WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
+    /* Packetize the parameters into the frame */
+    WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+    WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+    return;
+  }
+#endif
+
+  /* backward prediction of subframes */
+
+  if (iLBCbits_inst->startIdx > 1) {
+
+    /* create reverse order vectors
+       (The decresidual does not need to be copied since it is
+       contained in the same vector as the residual)
+    */
+
+    size_t Nback = iLBCbits_inst->startIdx - 1;
+    WebRtcSpl_MemCpyReversedOrder(&reverseResidual[Nback*SUBL-1], residual, Nback*SUBL);
+
+    /* setup memory */
+
+    meml_gotten = SUBL*(iLBCenc_inst->nsub+1-iLBCbits_inst->startIdx);
+    if( meml_gotten > CB_MEML ) {
+      meml_gotten=CB_MEML;
+    }
+
+    WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[Nback*SUBL], meml_gotten);
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
+
+#ifdef SPLIT_10MS
+    if (iLBCenc_inst->Nback_flag > 0)
+    {
+      for (subframe = 0; subframe < WEBRTC_SPL_MAX (2 - Nfor, 0); subframe++)
+      {
+        /* update memory */
+        WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
+        WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
+                               &reverseDecresidual[subframe * SUBL], SUBL);
+      }
+    }
+
+    iLBCenc_inst->Nback_flag++;
+
+
+    if (iLBCenc_inst->mode == 20)
+    {
+      start_count = 0;
+      end_count = Nback;
+    }
+    if (iLBCenc_inst->mode == 30)
+    {
+      if (iLBCenc_inst->section == 1)
+      {
+        start_count = 0;
+        end_count = (Nfor >= 2) ? 0 : (2 - NFor);
+      }
+      if (iLBCenc_inst->section == 2)
+      {
+        start_count = (Nfor >= 2) ? 0 : (2 - NFor);
+        end_count = Nback;
+      }
+    }
+#else
+    start_count = 0;
+    end_count = Nback;
+#endif
+
+    /* loop over subframes to encode */
+
+    for (subframe = start_count; subframe < end_count; subframe++){
+
+      /* encode subframe */
+
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+                             iLBCbits_inst->gain_index+subcount*CB_NSTAGES, &reverseResidual[subframe*SUBL],
+                             mem, MEM_LF_TBL, SUBL,
+                             &weightdenum[(iLBCbits_inst->startIdx-2-subframe)*(LPC_FILTERORDER+1)],
+                             subcount);
+
+      /* construct decoded vector */
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+            &reverseDecresidual[subframe * SUBL],
+            iLBCbits_inst->cb_index + subcount * CB_NSTAGES,
+            iLBCbits_inst->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+            SUBL));
+
+      /* update memory */
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &reverseDecresidual[subframe*SUBL], SUBL);
+
+      subcount++;
+
+    }
+
+    /* get decoded residual from reversed vector */
+
+    WebRtcSpl_MemCpyReversedOrder(&decresidual[SUBL*Nback-1], reverseDecresidual, SUBL*Nback);
+  }
+  /* end encoding part */
+
+  /* adjust index */
+
+  WebRtcIlbcfix_IndexConvEnc(iLBCbits_inst->cb_index);
+
+  /* Packetize the parameters into the frame */
+
+#ifdef SPLIT_10MS
+  if( (iLBCenc_inst->mode==30) && (iLBCenc_inst->section==1) ){
+    WebRtcIlbcfix_PackBits(iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+  }
+  else{
+    WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
+  }
+#else
+  WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
+#endif
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+  /* Swap bytes for LITTLE ENDIAN since the packbits()
+     function assumes BIG_ENDIAN machine */
+#ifdef SPLIT_10MS
+  if (( (iLBCenc_inst->section == 1) && (iLBCenc_inst->mode == 20) ) ||
+      ( (iLBCenc_inst->section == 2) && (iLBCenc_inst->mode == 30) )){
+    WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words, bytes);
+  }
+#else
+  WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words, bytes);
+#endif
+#endif
+
+#ifdef SPLIT_10MS
+  if (subcount == (iLBCenc_inst->nsub - 1))
+  {
+    iLBCenc_inst->section = 0;
+  }
+  else
+  {
+    iLBCenc_inst->section++;
+    WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+  }
+#endif
+
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.h
new file mode 100644
index 0000000000..bc3e187d92
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Encode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  main encoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EncodeImpl(
+    uint16_t* bytes,      /* (o) encoded data bits iLBC */
+    const int16_t* block, /* (i) speech vector to encode */
+    IlbcEncoder* iLBCenc_inst /* (i/o) the general encoder
+                                           state */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
new file mode 100644
index 0000000000..7f00254aea
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnergyInverse.c
+
+******************************************************************/
+
+/* Inverses the in vector in into Q29 domain */
+
+#include "modules/audio_coding/codecs/ilbc/energy_inverse.h"
+
+void WebRtcIlbcfix_EnergyInverse(
+    int16_t *energy,    /* (i/o) Energy and inverse
+                                                           energy (in Q29) */
+    size_t noOfEnergies)  /* (i)   The length of the energy
+                                   vector */
+{
+  int32_t Nom=(int32_t)0x1FFFFFFF;
+  int16_t *energyPtr;
+  size_t i;
+
+  /* Set the minimum energy value to 16384 to avoid overflow */
+  energyPtr=energy;
+  for (i=0; i<noOfEnergies; i++) {
+    (*energyPtr)=WEBRTC_SPL_MAX((*energyPtr),16384);
+    energyPtr++;
+  }
+
+  /* Calculate inverse energy in Q29 */
+  energyPtr=energy;
+  for (i=0; i<noOfEnergies; i++) {
+    (*energyPtr) = (int16_t)WebRtcSpl_DivW32W16(Nom, (*energyPtr));
+    energyPtr++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
new file mode 100644
index 0000000000..15391cf230
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnergyInverse.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* Inverses the in vector in into Q29 domain */
+
+void WebRtcIlbcfix_EnergyInverse(
+    int16_t*
+        energy, /* (i/o) Energy and inverse
+                                                          energy (in Q29) */
+    size_t noOfEnergies); /* (i)   The length of the energy
+                                 vector */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c
new file mode 100644
index 0000000000..cd3d0a4db1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhUpsample.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/enh_upsample.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * upsample finite array assuming zeros outside bounds
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EnhUpsample(
+    int32_t *useq1, /* (o) upsampled output sequence */
+    int16_t *seq1 /* (i) unupsampled sequence */
+                                ){
+  int j;
+  int32_t *pu1, *pu11;
+  int16_t *ps, *w16tmp;
+  const int16_t *pp;
+
+  /* filtering: filter overhangs left side of sequence */
+  pu1=useq1;
+  for (j=0;j<ENH_UPS0; j++) {
+    pu11=pu1;
+    /* i = 2 */
+    pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+    ps=seq1+2;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+    /* i = 3 */
+    pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+    ps=seq1+3;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+    /* i = 4 */
+    pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+    ps=seq1+4;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu1++;
+  }
+
+  /* filtering: simple convolution=inner products
+     (not needed since the sequence is so short)
+  */
+
+  /* filtering: filter overhangs right side of sequence */
+
+  /* Code with loops, which is equivivalent to the expanded version below
+
+     filterlength = 5;
+     hf1 = 2;
+     for(j=0;j<ENH_UPS0; j++){
+     pu = useq1 + (filterlength-hfl)*ENH_UPS0 + j;
+     for(i=1; i<=hfl; i++){
+     *pu=0;
+     pp = polyp[j]+i;
+     ps = seq1+dim1-1;
+     for(k=0;k<filterlength-i;k++) {
+     *pu += (*ps--) * *pp++;
+     }
+     pu+=ENH_UPS0;
+     }
+     }
+  */
+  pu1 = useq1 + 12;
+  w16tmp = seq1+4;
+  for (j=0;j<ENH_UPS0; j++) {
+    pu11 = pu1;
+    /* i = 1 */
+    pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+2;
+    ps = w16tmp;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+    /* i = 2 */
+    pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+3;
+    ps = w16tmp;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+
+    pu1++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h
new file mode 100644
index 0000000000..b427eca50a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhUpsample.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * upsample finite array assuming zeros outside bounds
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EnhUpsample(
+    int32_t* useq1, /* (o) upsampled output sequence */
+    int16_t* seq1   /* (i) unupsampled sequence */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c
new file mode 100644
index 0000000000..bd4e60015c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Enhancer.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/enhancer.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/get_sync_seq.h"
+#include "modules/audio_coding/codecs/ilbc/smooth.h"
+
+/*----------------------------------------------------------------*
+ * perform enhancement on idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Enhancer(
+    int16_t *odata,   /* (o) smoothed block, dimension blockl */
+    int16_t *idata,   /* (i) data buffer used for enhancing */
+    size_t idatal,   /* (i) dimension idata */
+    size_t centerStartPos, /* (i) first sample current block within idata */
+    size_t *period,   /* (i) pitch period array (pitch bward-in time) */
+    const size_t *plocs,   /* (i) locations where period array values valid */
+    size_t periodl   /* (i) dimension of period and plocs */
+                            ){
+  /* Stack based */
+  int16_t surround[ENH_BLOCKL];
+
+  WebRtcSpl_MemSetW16(surround, 0, ENH_BLOCKL);
+
+  /* get said second sequence of segments */
+
+  WebRtcIlbcfix_GetSyncSeq(idata, idatal, centerStartPos, period, plocs,
+                           periodl, ENH_HL, surround);
+
+  /* compute the smoothed output from said second sequence */
+
+  WebRtcIlbcfix_Smooth(odata, idata + centerStartPos, surround);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.h
new file mode 100644
index 0000000000..386949347a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Enhancer.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * perform enhancement on idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Enhancer(
+    int16_t* odata,        /* (o) smoothed block, dimension blockl */
+    int16_t* idata,        /* (i) data buffer used for enhancing */
+    size_t idatal,         /* (i) dimension idata */
+    size_t centerStartPos, /* (i) first sample current block within idata */
+    size_t* period,        /* (i) pitch period array (pitch bward-in time) */
+    const size_t* plocs,   /* (i) locations where period array values valid */
+    size_t periodl         /* (i) dimension of period and plocs */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
new file mode 100644
index 0000000000..ca23e19ae3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -0,0 +1,382 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhancerInterface.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/enhancer.h"
+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+
+
+
+/*----------------------------------------------------------------*
+ * interface for enhancer
+ *---------------------------------------------------------------*/
+
+size_t  // (o) Estimated lag in end of in[]
+    WebRtcIlbcfix_EnhancerInterface(
+        int16_t* out,                 // (o) enhanced signal
+        const int16_t* in,            // (i) unenhanced signal
+        IlbcDecoder* iLBCdec_inst) {  // (i) buffers etc
+  size_t iblock;
+  size_t lag=20, tlag=20;
+  size_t inLen=iLBCdec_inst->blockl+120;
+  int16_t scale, scale1;
+  size_t plc_blockl;
+  int16_t *enh_buf;
+  size_t *enh_period;
+  int32_t tmp1, tmp2, max;
+  size_t new_blocks;
+  int16_t *enh_bufPtr1;
+  size_t i;
+  size_t k;
+  int16_t EnChange;
+  int16_t SqrtEnChange;
+  int16_t inc;
+  int16_t win;
+  int16_t *tmpW16ptr;
+  size_t startPos;
+  int16_t *plc_pred;
+  const int16_t *target, *regressor;
+  int16_t max16;
+  int shifts;
+  int32_t ener;
+  int16_t enerSh;
+  int16_t corrSh;
+  size_t ind;
+  int16_t sh;
+  size_t start, stop;
+  /* Stack based */
+  int16_t totsh[3];
+  int16_t downsampled[(BLOCKL_MAX+120)>>1]; /* length 180 */
+  int32_t corr32[50];
+  int32_t corrmax[3];
+  int16_t corr16[3];
+  int16_t en16[3];
+  size_t lagmax[3];
+
+  plc_pred = downsampled; /* Reuse memory since plc_pred[ENH_BLOCKL] and
+                              downsampled are non overlapping */
+  enh_buf=iLBCdec_inst->enh_buf;
+  enh_period=iLBCdec_inst->enh_period;
+
+  /* Copy in the new data into the enhancer buffer */
+  memmove(enh_buf, &enh_buf[iLBCdec_inst->blockl],
+          (ENH_BUFL - iLBCdec_inst->blockl) * sizeof(*enh_buf));
+
+  WEBRTC_SPL_MEMCPY_W16(&enh_buf[ENH_BUFL-iLBCdec_inst->blockl], in,
+                        iLBCdec_inst->blockl);
+
+  /* Set variables that are dependent on frame size */
+  if (iLBCdec_inst->mode==30) {
+    plc_blockl=ENH_BLOCKL;
+    new_blocks=3;
+    startPos=320;  /* Start position for enhancement
+                     (640-new_blocks*ENH_BLOCKL-80) */
+  } else {
+    plc_blockl=40;
+    new_blocks=2;
+    startPos=440;  /* Start position for enhancement
+                    (640-new_blocks*ENH_BLOCKL-40) */
+  }
+
+  /* Update the pitch prediction for each enhancer block, move the old ones */
+  memmove(enh_period, &enh_period[new_blocks],
+          (ENH_NBLOCKS_TOT - new_blocks) * sizeof(*enh_period));
+
+  WebRtcSpl_DownsampleFast(
+      enh_buf+ENH_BUFL-inLen,    /* Input samples */
+      inLen + ENH_BUFL_FILTEROVERHEAD,
+      downsampled,
+      inLen / 2,
+      (int16_t*)WebRtcIlbcfix_kLpFiltCoefs,  /* Coefficients in Q12 */
+      FILTERORDER_DS_PLUS1,    /* Length of filter (order-1) */
+      FACTOR_DS,
+      DELAY_DS);
+
+  /* Estimate the pitch in the down sampled domain. */
+  for(iblock = 0; iblock<new_blocks; iblock++){
+
+    /* references */
+    target = downsampled + 60 + iblock * ENH_BLOCKL_HALF;
+    regressor = target - 10;
+
+    /* scaling */
+    max16 = WebRtcSpl_MaxAbsValueW16(&regressor[-50], ENH_BLOCKL_HALF + 50 - 1);
+    shifts = WebRtcSpl_GetSizeInBits((uint32_t)(max16 * max16)) - 25;
+    shifts = WEBRTC_SPL_MAX(0, shifts);
+
+    /* compute cross correlation */
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, ENH_BLOCKL_HALF, 50,
+                               shifts, -1);
+
+    /* Find 3 highest correlations that should be compared for the
+       highest (corr*corr)/ener */
+
+    for (i=0;i<2;i++) {
+      lagmax[i] = WebRtcSpl_MaxIndexW32(corr32, 50);
+      corrmax[i] = corr32[lagmax[i]];
+      start = WEBRTC_SPL_MAX(2, lagmax[i]) - 2;
+      stop = WEBRTC_SPL_MIN(47, lagmax[i]) + 2;
+      for (k = start; k <= stop; k++) {
+        corr32[k] = 0;
+      }
+    }
+    lagmax[2] = WebRtcSpl_MaxIndexW32(corr32, 50);
+    corrmax[2] = corr32[lagmax[2]];
+
+    /* Calculate normalized corr^2 and ener */
+    for (i=0;i<3;i++) {
+      corrSh = 15-WebRtcSpl_GetSizeInBits(corrmax[i]);
+      ener = WebRtcSpl_DotProductWithScale(regressor - lagmax[i],
+                                           regressor - lagmax[i],
+                                           ENH_BLOCKL_HALF, shifts);
+      enerSh = 15-WebRtcSpl_GetSizeInBits(ener);
+      corr16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(corrmax[i], corrSh);
+      corr16[i] = (int16_t)((corr16[i] * corr16[i]) >> 16);
+      en16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(ener, enerSh);
+      totsh[i] = enerSh - 2 * corrSh;
+    }
+
+    /* Compare lagmax[0..3] for the (corr^2)/ener criteria */
+    ind = 0;
+    for (i=1; i<3; i++) {
+      if (totsh[ind] > totsh[i]) {
+        sh = WEBRTC_SPL_MIN(31, totsh[ind]-totsh[i]);
+        if (corr16[ind] * en16[i] < (corr16[i] * en16[ind]) >> sh) {
+          ind = i;
+        }
+      } else {
+        sh = WEBRTC_SPL_MIN(31, totsh[i]-totsh[ind]);
+        if ((corr16[ind] * en16[i]) >> sh < corr16[i] * en16[ind]) {
+          ind = i;
+        }
+      }
+    }
+
+    lag = lagmax[ind] + 10;
+
+    /* Store the estimated lag in the non-downsampled domain */
+    enh_period[ENH_NBLOCKS_TOT - new_blocks + iblock] = lag * 8;
+
+    /* Store the estimated lag for backward PLC */
+    if (iLBCdec_inst->prev_enh_pl==1) {
+      if (!iblock) {
+        tlag = lag * 2;
+      }
+    } else {
+      if (iblock==1) {
+        tlag = lag * 2;
+      }
+    }
+
+    lag *= 2;
+  }
+
+  if ((iLBCdec_inst->prev_enh_pl==1)||(iLBCdec_inst->prev_enh_pl==2)) {
+
+    /* Calculate the best lag of the new frame
+       This is used to interpolate backwards and mix with the PLC'd data
+    */
+
+    /* references */
+    target=in;
+    regressor=in+tlag-1;
+
+    /* scaling */
+    // Note that this is not abs-max, so we will take the absolute value below.
+    max16 = WebRtcSpl_MaxAbsElementW16(regressor, plc_blockl + 3 - 1);
+    const int16_t max_target =
+        WebRtcSpl_MaxAbsElementW16(target, plc_blockl + 3 - 1);
+    const int64_t max_val = plc_blockl * abs(max16 * max_target);
+    const int32_t factor = max_val >> 31;
+    shifts = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+
+    /* compute cross correlation */
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, plc_blockl, 3, shifts,
+                               1);
+
+    /* find lag */
+    lag=WebRtcSpl_MaxIndexW32(corr32, 3);
+    lag+=tlag-1;
+
+    /* Copy the backward PLC to plc_pred */
+
+    if (iLBCdec_inst->prev_enh_pl==1) {
+      if (lag>plc_blockl) {
+        WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-plc_blockl], plc_blockl);
+      } else {
+        WEBRTC_SPL_MEMCPY_W16(&plc_pred[plc_blockl-lag], in, lag);
+        WEBRTC_SPL_MEMCPY_W16(
+            plc_pred, &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl+lag],
+            (plc_blockl-lag));
+      }
+    } else {
+      size_t pos;
+
+      pos = plc_blockl;
+
+      while (lag<pos) {
+        WEBRTC_SPL_MEMCPY_W16(&plc_pred[pos-lag], in, lag);
+        pos = pos - lag;
+      }
+      WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-pos], pos);
+
+    }
+
+    if (iLBCdec_inst->prev_enh_pl==1) {
+      /* limit energy change
+         if energy in backward PLC is more than 4 times higher than the forward
+         PLC, then reduce the energy in the backward PLC vector:
+         sample 1...len-16 set energy of the to 4 times forward PLC
+         sample len-15..len interpolate between 4 times fw PLC and bw PLC energy
+
+         Note: Compared to floating point code there is a slight change,
+         the window is 16 samples long instead of 10 samples to simplify the
+         calculations
+      */
+
+      max=WebRtcSpl_MaxAbsValueW16(
+          &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl], plc_blockl);
+      max16=WebRtcSpl_MaxAbsValueW16(plc_pred, plc_blockl);
+      max = WEBRTC_SPL_MAX(max, max16);
+      scale=22-(int16_t)WebRtcSpl_NormW32(max);
+      scale=WEBRTC_SPL_MAX(scale,0);
+
+      tmp2 = WebRtcSpl_DotProductWithScale(
+          &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
+          &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
+          plc_blockl, scale);
+      tmp1 = WebRtcSpl_DotProductWithScale(plc_pred, plc_pred,
+                                           plc_blockl, scale);
+
+      /* Check the energy difference */
+      if ((tmp1>0)&&((tmp1>>2)>tmp2)) {
+        /* EnChange is now guaranteed to be <0.5
+           Calculate EnChange=tmp2/tmp1 in Q16
+        */
+
+        scale1=(int16_t)WebRtcSpl_NormW32(tmp1);
+        tmp1=WEBRTC_SPL_SHIFT_W32(tmp1, (scale1-16)); /* using 15 bits */
+
+        tmp2=WEBRTC_SPL_SHIFT_W32(tmp2, (scale1));
+        EnChange = (int16_t)WebRtcSpl_DivW32W16(tmp2,
+                                                      (int16_t)tmp1);
+
+        /* Calculate the Sqrt of the energy in Q15 ((14+16)/2) */
+        SqrtEnChange = (int16_t)WebRtcSpl_SqrtFloor(EnChange << 14);
+
+
+        /* Multiply first part of vector with 2*SqrtEnChange */
+        WebRtcSpl_ScaleVector(plc_pred, plc_pred, SqrtEnChange, plc_blockl-16,
+                              14);
+
+        /* Calculate increase parameter for window part (16 last samples) */
+        /* (1-2*SqrtEnChange)/16 in Q15 */
+        inc = 2048 - (SqrtEnChange >> 3);
+
+        win=0;
+        tmpW16ptr=&plc_pred[plc_blockl-16];
+
+        for (i=16;i>0;i--) {
+          *tmpW16ptr = (int16_t)(
+              (*tmpW16ptr * (SqrtEnChange + (win >> 1))) >> 14);
+          /* multiply by (2.0*SqrtEnChange+win) */
+
+          win += inc;
+          tmpW16ptr++;
+        }
+      }
+
+      /* Make the linear interpolation between the forward PLC'd data
+         and the backward PLC'd data (from the new frame)
+      */
+
+      if (plc_blockl==40) {
+        inc=400; /* 1/41 in Q14 */
+      } else { /* plc_blockl==80 */
+        inc=202; /* 1/81 in Q14 */
+      }
+      win=0;
+      enh_bufPtr1=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
+      for (i=0; i<plc_blockl; i++) {
+        win+=inc;
+        *enh_bufPtr1 = (int16_t)((*enh_bufPtr1 * win) >> 14);
+        *enh_bufPtr1 += (int16_t)(
+            ((16384 - win) * plc_pred[plc_blockl - 1 - i]) >> 14);
+        enh_bufPtr1--;
+      }
+    } else {
+      int16_t *synt = &downsampled[LPC_FILTERORDER];
+
+      enh_bufPtr1=&enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl];
+      WEBRTC_SPL_MEMCPY_W16(enh_bufPtr1, plc_pred, plc_blockl);
+
+      /* Clear fileter memory */
+      WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
+      WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
+      WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
+
+      /* Initialize filter memory by filtering through 2 lags */
+      WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], iLBCdec_inst->syntMem,
+                            LPC_FILTERORDER);
+      WebRtcSpl_FilterARFastQ12(
+          enh_bufPtr1,
+          synt,
+          &iLBCdec_inst->old_syntdenum[
+              (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, lag);
+
+      WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], &synt[lag-LPC_FILTERORDER],
+                            LPC_FILTERORDER);
+      WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
+                             iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+                             lag);
+      WebRtcSpl_FilterARFastQ12(
+          enh_bufPtr1, synt,
+          &iLBCdec_inst->old_syntdenum[
+              (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, lag);
+
+      WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &synt[lag-LPC_FILTERORDER],
+                            LPC_FILTERORDER);
+      WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
+                             iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+                             lag);
+    }
+  }
+
+
+  /* Perform enhancement block by block */
+
+  for (iblock = 0; iblock<new_blocks; iblock++) {
+    WebRtcIlbcfix_Enhancer(out + iblock * ENH_BLOCKL,
+                           enh_buf,
+                           ENH_BUFL,
+                           iblock * ENH_BLOCKL + startPos,
+                           enh_period,
+                           WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT);
+  }
+
+  return (lag);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
new file mode 100644
index 0000000000..5022a47c3a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhancerInterface.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * interface for enhancer
+ *---------------------------------------------------------------*/
+
+size_t  // (o) Estimated lag in end of in[]
+WebRtcIlbcfix_EnhancerInterface(int16_t* out,       // (o) enhanced signal
+                                const int16_t* in,  // (i) unenhanced signal
+                                IlbcDecoder* iLBCdec_inst);  // (i) buffers etc
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
new file mode 100644
index 0000000000..6b4f30c96b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FilteredCbVecs.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Construct an additional codebook vector by filtering the
+ *  initial codebook buffer. This vector is then used to expand
+ *  the codebook with an additional section.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_FilteredCbVecs(
+    int16_t *cbvectors, /* (o) Codebook vector for the higher section */
+    int16_t *CBmem,  /* (i) Codebook memory that is filtered to create a
+                                           second CB section */
+    size_t lMem,  /* (i) Length of codebook memory */
+    size_t samples    /* (i) Number of samples to filter */
+                                  ) {
+
+  /* Set up the memory, start with zero state */
+  WebRtcSpl_MemSetW16(CBmem+lMem, 0, CB_HALFFILTERLEN);
+  WebRtcSpl_MemSetW16(CBmem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
+  WebRtcSpl_MemSetW16(cbvectors, 0, lMem-samples);
+
+  /* Filter to obtain the filtered CB memory */
+
+  WebRtcSpl_FilterMAFastQ12(
+      CBmem+CB_HALFFILTERLEN+lMem-samples, cbvectors+lMem-samples,
+      (int16_t*)WebRtcIlbcfix_kCbFiltersRev, CB_FILTERLEN, samples);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
new file mode 100644
index 0000000000..661262e42e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FilteredCbVecs.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Construct an additional codebook vector by filtering the
+ *  initial codebook buffer. This vector is then used to expand
+ *  the codebook with an additional section.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_FilteredCbVecs(
+    int16_t* cbvectors, /* (o) Codebook vector for the higher section */
+    int16_t* CBmem,     /* (i) Codebook memory that is filtered to create a
+                                              second CB section */
+    size_t lMem,        /* (i) Length of codebook memory */
+    size_t samples      /* (i) Number of samples to filter */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
new file mode 100644
index 0000000000..c1084b1645
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FrameClassify.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/frame_classify.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Classification of subframes to localize start state
+ *---------------------------------------------------------------*/
+
+size_t WebRtcIlbcfix_FrameClassify(
+    /* (o) Index to the max-energy sub frame */
+    IlbcEncoder *iLBCenc_inst,
+    /* (i/o) the encoder state structure */
+    int16_t *residualFIX /* (i) lpc residual signal */
+                                                ){
+  int16_t max, scale;
+  int32_t ssqEn[NSUB_MAX-1];
+  int16_t *ssqPtr;
+  int32_t *seqEnPtr;
+  int32_t maxW32;
+  int16_t scale1;
+  size_t pos;
+  size_t n;
+
+  /*
+    Calculate the energy of each of the 80 sample blocks
+    in the draft the 4 first and last samples are windowed with 1/5...4/5
+    and 4/5...1/5 respectively. To simplify for the fixpoint we have changed
+    this to 0 0 1 1 and 1 1 0 0
+  */
+
+  max = WebRtcSpl_MaxAbsValueW16(residualFIX, iLBCenc_inst->blockl);
+  scale = WebRtcSpl_GetSizeInBits((uint32_t)(max * max));
+
+  /* Scale to maximum 24 bits so that it won't overflow for 76 samples */
+  scale = scale-24;
+  scale1 = WEBRTC_SPL_MAX(0, scale);
+
+  /* Calculate energies */
+  ssqPtr=residualFIX + 2;
+  seqEnPtr=ssqEn;
+  for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
+    (*seqEnPtr) = WebRtcSpl_DotProductWithScale(ssqPtr, ssqPtr, 76, scale1);
+    ssqPtr += 40;
+    seqEnPtr++;
+  }
+
+  /* Scale to maximum 20 bits in order to allow for the 11 bit window */
+  maxW32 = WebRtcSpl_MaxValueW32(ssqEn, iLBCenc_inst->nsub - 1);
+  scale = WebRtcSpl_GetSizeInBits(maxW32) - 20;
+  scale1 = WEBRTC_SPL_MAX(0, scale);
+
+  /* Window each 80 block with the ssqEn_winTbl window to give higher probability for
+     the blocks in the middle
+  */
+  seqEnPtr=ssqEn;
+  if (iLBCenc_inst->mode==20) {
+    ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin+1;
+  } else {
+    ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin;
+  }
+  for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
+    (*seqEnPtr)=WEBRTC_SPL_MUL(((*seqEnPtr)>>scale1), (*ssqPtr));
+    seqEnPtr++;
+    ssqPtr++;
+  }
+
+  /* Extract the best choise of start state */
+  pos = WebRtcSpl_MaxIndexW32(ssqEn, iLBCenc_inst->nsub - 1) + 1;
+
+  return(pos);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
new file mode 100644
index 0000000000..7615106d70
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FrameClassify.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+size_t WebRtcIlbcfix_FrameClassify(
+    /* (o) Index to the max-energy sub frame */
+    IlbcEncoder* iLBCenc_inst,
+    /* (i/o) the encoder state structure */
+    int16_t* residualFIX /* (i) lpc residual signal */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c
new file mode 100644
index 0000000000..1357dece33
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainDequant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/gain_dequant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  decoder for quantized gains in the gain-shape coding of
+ *  residual
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_GainDequant(
+    /* (o) quantized gain value (Q14) */
+    int16_t index, /* (i) quantization index */
+    int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
+    int16_t stage /* (i) The stage of the search */
+                                                ){
+  int16_t scale;
+  const int16_t *gain;
+
+  /* obtain correct scale factor */
+
+  scale=WEBRTC_SPL_ABS_W16(maxIn);
+  scale = WEBRTC_SPL_MAX(1638, scale);  /* if lower than 0.1, set it to 0.1 */
+
+  /* select the quantization table and return the decoded value */
+  gain = WebRtcIlbcfix_kGain[stage];
+
+  return (int16_t)((scale * gain[index] + 8192) >> 14);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h
new file mode 100644
index 0000000000..2b97550b6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainDequant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  decoder for quantized gains in the gain-shape coding of
+ *  residual
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_GainDequant(
+    /* (o) quantized gain value (Q14) */
+    int16_t index, /* (i) quantization index */
+    int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
+    int16_t stage  /* (i) The stage of the search */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c
new file mode 100644
index 0000000000..9a6d49d51a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainQuant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/gain_quant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  quantizer for the gain in the gain-shape coding of residual
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
+    int16_t gain, /* (i) gain value Q14 */
+    int16_t maxIn, /* (i) maximum of gain value Q14 */
+    int16_t stage, /* (i) The stage of the search */
+    int16_t *index /* (o) quantization index */
+                                        ) {
+
+  int16_t scale, cblen;
+  int32_t gainW32, measure1, measure2;
+  const int16_t *cbPtr, *cb;
+  int loc, noMoves, noChecks, i;
+
+  /* ensure a lower bound (0.1) on the scaling factor */
+
+  scale = WEBRTC_SPL_MAX(1638, maxIn);
+
+  /* select the quantization table and calculate
+     the length of the table and the number of
+     steps in the binary search that are needed */
+  cb = WebRtcIlbcfix_kGain[stage];
+  cblen = 32>>stage;
+  noChecks = 4-stage;
+
+  /* Multiply the gain with 2^14 to make the comparison
+     easier and with higher precision */
+  gainW32 = gain << 14;
+
+  /* Do a binary search, starting in the middle of the CB
+     loc - defines the current position in the table
+     noMoves - defines the number of steps to move in the CB in order
+     to get next CB location
+  */
+
+  loc = cblen>>1;
+  noMoves = loc;
+  cbPtr = cb + loc; /* Centre of CB */
+
+  for (i=noChecks;i>0;i--) {
+    noMoves>>=1;
+    measure1 = scale * *cbPtr;
+
+    /* Move up if gain is larger, otherwise move down in table */
+    measure1 = measure1 - gainW32;
+
+    if (0>measure1) {
+      cbPtr+=noMoves;
+      loc+=noMoves;
+    } else {
+      cbPtr-=noMoves;
+      loc-=noMoves;
+    }
+  }
+
+  /* Check which value is the closest one: loc-1, loc or loc+1 */
+
+  measure1 = scale * *cbPtr;
+  if (gainW32>measure1) {
+    /* Check against value above loc */
+    measure2 = scale * cbPtr[1];
+    if ((measure2-gainW32)<(gainW32-measure1)) {
+      loc+=1;
+    }
+  } else {
+    /* Check against value below loc */
+    measure2 = scale * cbPtr[-1];
+    if ((gainW32-measure2)<=(measure1-gainW32)) {
+      loc-=1;
+    }
+  }
+
+  /* Guard against getting outside the table. The calculation above can give a location
+     which is one above the maximum value (in very rare cases) */
+  loc=WEBRTC_SPL_MIN(loc, (cblen-1));
+  *index=loc;
+
+  /* Calculate and return the quantized gain value (in Q14) */
+  return (int16_t)((scale * cb[loc] + 8192) >> 14);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.h
new file mode 100644
index 0000000000..761f7d2f79
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainQuant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  quantizer for the gain in the gain-shape coding of residual
+ *---------------------------------------------------------------*/
+
+int16_t
+WebRtcIlbcfix_GainQuant(               /* (o) quantized gain value */
+                        int16_t gain,  /* (i) gain value Q14 */
+                        int16_t maxIn, /* (i) maximum of gain value Q14 */
+                        int16_t stage, /* (i) The stage of the search */
+                        int16_t* index /* (o) quantization index */
+                        );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
new file mode 100644
index 0000000000..e9cd2008e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetCbVec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/get_cd_vec.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Construct codebook vector for given index.
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_GetCbVec(
+    int16_t *cbvec,   /* (o) Constructed codebook vector */
+    int16_t *mem,   /* (i) Codebook buffer */
+    size_t index,   /* (i) Codebook index */
+    size_t lMem,   /* (i) Length of codebook buffer */
+    size_t cbveclen   /* (i) Codebook vector length */
+                            ){
+  size_t k, base_size;
+  size_t lag;
+  /* Stack based */
+  int16_t tempbuff2[SUBL+5];
+
+  /* Determine size of codebook sections */
+
+  base_size=lMem-cbveclen+1;
+
+  if (cbveclen==SUBL) {
+    base_size += cbveclen / 2;
+  }
+
+  /* No filter -> First codebook section */
+
+  if (index<lMem-cbveclen+1) {
+
+    /* first non-interpolated vectors */
+
+    k=index+cbveclen;
+    /* get vector */
+    WEBRTC_SPL_MEMCPY_W16(cbvec, mem+lMem-k, cbveclen);
+
+  } else if (index < base_size) {
+
+    /* Calculate lag */
+
+    k = (2 * (index - (lMem - cbveclen + 1))) + cbveclen;
+
+    lag = k / 2;
+
+    WebRtcIlbcfix_CreateAugmentedVec(lag, mem+lMem, cbvec);
+
+  }
+
+  /* Higher codebbok section based on filtering */
+
+  else {
+
+    size_t memIndTest;
+
+    /* first non-interpolated vectors */
+
+    if (index-base_size<lMem-cbveclen+1) {
+
+      /* Set up filter memory, stuff zeros outside memory buffer */
+
+      memIndTest = lMem-(index-base_size+cbveclen);
+
+      WebRtcSpl_MemSetW16(mem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
+      WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
+
+      /* do filtering to get the codebook vector */
+
+      WebRtcSpl_FilterMAFastQ12(
+          &mem[memIndTest+4], cbvec, (int16_t*)WebRtcIlbcfix_kCbFiltersRev,
+          CB_FILTERLEN, cbveclen);
+    }
+
+    /* interpolated vectors */
+
+    else {
+      if (cbveclen < SUBL) {
+        // We're going to fill in cbveclen + 5 elements of tempbuff2 in
+        // WebRtcSpl_FilterMAFastQ12, less than the SUBL + 5 elements we'll be
+        // using in WebRtcIlbcfix_CreateAugmentedVec. This error is caused by
+        // bad values in `index` (which come from the encoded stream). Tell the
+        // caller that things went south, and that the decoder state is now
+        // corrupt (because it's half-way through an update that we can't
+        // complete).
+        return false;
+      }
+
+      /* Stuff zeros outside memory buffer  */
+      memIndTest = lMem-cbveclen-CB_FILTERLEN;
+      WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
+
+      /* do filtering */
+      WebRtcSpl_FilterMAFastQ12(
+          &mem[memIndTest+7], tempbuff2, (int16_t*)WebRtcIlbcfix_kCbFiltersRev,
+          CB_FILTERLEN, cbveclen+5);
+
+      /* Calculate lag index */
+      lag = (cbveclen<<1)-20+index-base_size-lMem-1;
+
+      WebRtcIlbcfix_CreateAugmentedVec(lag, tempbuff2+SUBL+5, cbvec);
+    }
+  }
+
+  return true;  // Success.
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
new file mode 100644
index 0000000000..99537dd0f7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetCbVec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+
+#include <stdbool.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+// Returns true on success, false on failure. In case of failure, the decoder
+// state may be corrupted and needs resetting.
+ABSL_MUST_USE_RESULT
+bool WebRtcIlbcfix_GetCbVec(
+    int16_t* cbvec, /* (o) Constructed codebook vector */
+    int16_t* mem,   /* (i) Codebook buffer */
+    size_t index,   /* (i) Codebook index */
+    size_t lMem,    /* (i) Length of codebook buffer */
+    size_t cbveclen /* (i) Codebook vector length */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
new file mode 100644
index 0000000000..e0fb21caf0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetLspPoly.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/get_lsp_poly.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * Construct the polynomials F1(z) and F2(z) from the LSP
+ * (Computations are done in Q24)
+ *
+ * The expansion is performed using the following recursion:
+ *
+ * f[0] = 1;
+ * tmp = -2.0 * lsp[0];
+ * f[1] = tmp;
+ * for (i=2; i<=5; i++) {
+ *    b = -2.0 * lsp[2*i-2];
+ *    f[i] = tmp*f[i-1] + 2.0*f[i-2];
+ *    for (j=i; j>=2; j--) {
+ *       f[j] = f[j] + tmp*f[j-1] + f[j-2];
+ *    }
+ *    f[i] = f[i] + tmp;
+ * }
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetLspPoly(
+    int16_t *lsp, /* (i) LSP in Q15 */
+    int32_t *f)  /* (o) polonymial in Q24 */
+{
+  int32_t tmpW32;
+  int i, j;
+  int16_t high, low;
+  int16_t *lspPtr;
+  int32_t *fPtr;
+
+  lspPtr = lsp;
+  fPtr = f;
+  /* f[0] = 1.0 (Q24) */
+  (*fPtr) = (int32_t)16777216;
+  fPtr++;
+
+  (*fPtr) = WEBRTC_SPL_MUL((*lspPtr), -1024);
+  fPtr++;
+  lspPtr+=2;
+
+  for(i=2; i<=5; i++)
+  {
+    (*fPtr) = fPtr[-2];
+
+    for(j=i; j>1; j--)
+    {
+      /* Compute f[j] = f[j] + tmp*f[j-1] + f[j-2]; */
+      high = (int16_t)(fPtr[-1] >> 16);
+      low = (int16_t)((fPtr[-1] & 0xffff) >> 1);
+
+      tmpW32 = 4 * high * *lspPtr + 4 * ((low * *lspPtr) >> 15);
+
+      (*fPtr) += fPtr[-2];
+      (*fPtr) -= tmpW32;
+      fPtr--;
+    }
+    *fPtr -= *lspPtr * (1 << 10);
+
+    fPtr+=i;
+    lspPtr+=2;
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
new file mode 100644
index 0000000000..70c9c4d4b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetLspPoly.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * Construct the polynomials F1(z) and F2(z) from the LSP
+ * (Computations are done in Q24)
+ *
+ * The expansion is performed using the following recursion:
+ *
+ * f[0] = 1;
+ * tmp = -2.0 * lsp[0];
+ * f[1] = tmp;
+ * for (i=2; i<=5; i++) {
+ *    b = -2.0 * lsp[2*i-2];
+ *    f[i] = tmp*f[i-1] + 2.0*f[i-2];
+ *    for (j=i; j>=2; j--) {
+ *       f[j] = f[j] + tmp*f[j-1] + f[j-2];
+ *    }
+ *    f[i] = f[i] + tmp;
+ * }
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetLspPoly(int16_t* lsp, /* (i) LSP in Q15 */
+                              int32_t* f);  /* (o) polonymial in Q24 */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
new file mode 100644
index 0000000000..68a569a40a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetSyncSeq.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/get_sync_seq.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/nearest_neighbor.h"
+#include "modules/audio_coding/codecs/ilbc/refiner.h"
+
+/*----------------------------------------------------------------*
+ * get the pitch-synchronous sample sequence
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetSyncSeq(
+    int16_t *idata,   /* (i) original data */
+    size_t idatal,   /* (i) dimension of data */
+    size_t centerStartPos, /* (i) where current block starts */
+    size_t *period,   /* (i) rough-pitch-period array       (Q-2) */
+    const size_t *plocs, /* (i) where periods of period array are taken (Q-2) */
+    size_t periodl,   /* (i) dimension period array */
+    size_t hl,    /* (i) 2*hl+1 is the number of sequences */
+    int16_t *surround  /* (i/o) The contribution from this sequence
+                                summed with earlier contributions */
+                              ){
+  size_t i, centerEndPos, q;
+  /* Stack based */
+  size_t lagBlock[2 * ENH_HL + 1];
+  size_t blockStartPos[2 * ENH_HL + 1]; /* The position to search around (Q2) */
+  size_t plocs2[ENH_PLOCSL];
+
+  centerEndPos = centerStartPos + ENH_BLOCKL - 1;
+
+  /* present (find predicted lag from this position) */
+
+  WebRtcIlbcfix_NearestNeighbor(lagBlock + hl,
+                                plocs,
+                                2 * (centerStartPos + centerEndPos),
+                                periodl);
+
+  blockStartPos[hl] = 4 * centerStartPos;
+
+  /* past (find predicted position and perform a refined
+     search to find the best sequence) */
+
+  for (q = hl; q > 0; q--) {
+    size_t qq = q - 1;
+    size_t period_q = period[lagBlock[q]];
+    /* Stop if this sequence would be outside the buffer; that means all
+       further-past sequences would also be outside the buffer. */
+    if (blockStartPos[q] < period_q + (4 * ENH_OVERHANG))
+      break;
+    blockStartPos[qq] = blockStartPos[q] - period_q;
+
+    size_t value = blockStartPos[qq] + 4 * ENH_BLOCKL_HALF;
+    value = (value > period_q) ? (value - period_q) : 0;
+    WebRtcIlbcfix_NearestNeighbor(lagBlock + qq, plocs, value, periodl);
+
+    /* Find the best possible sequence in the 4 times upsampled
+        domain around blockStartPos+q */
+    WebRtcIlbcfix_Refiner(blockStartPos + qq, idata, idatal, centerStartPos,
+                          blockStartPos[qq], surround,
+                          WebRtcIlbcfix_kEnhWt[qq]);
+  }
+
+  /* future (find predicted position and perform a refined
+     search to find the best sequence) */
+
+  for (i = 0; i < periodl; i++) {
+    plocs2[i] = plocs[i] - period[i];
+  }
+
+  for (q = hl + 1; q <= (2 * hl); q++) {
+
+    WebRtcIlbcfix_NearestNeighbor(
+        lagBlock + q,
+        plocs2,
+        blockStartPos[q - 1] + 4 * ENH_BLOCKL_HALF,
+        periodl);
+
+    blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
+
+    if (blockStartPos[q] + 4 * (ENH_BLOCKL + ENH_OVERHANG) < 4 * idatal) {
+
+      /* Find the best possible sequence in the 4 times upsampled
+         domain around blockStartPos+q */
+      WebRtcIlbcfix_Refiner(blockStartPos + q, idata, idatal, centerStartPos,
+                            blockStartPos[q], surround,
+                            WebRtcIlbcfix_kEnhWt[2 * hl - q]);
+
+    } else {
+      /* Don't add anything since this sequence would
+         be outside the buffer */
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
new file mode 100644
index 0000000000..90962fa063
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetSyncSeq.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * get the pitch-synchronous sample sequence
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetSyncSeq(
+    int16_t* idata,        /* (i) original data */
+    size_t idatal,         /* (i) dimension of data */
+    size_t centerStartPos, /* (i) where current block starts */
+    size_t* period,        /* (i) rough-pitch-period array       (Q-2) */
+    const size_t* plocs, /* (i) where periods of period array are taken (Q-2) */
+    size_t periodl,      /* (i) dimension period array */
+    size_t hl,           /* (i) 2*hl+1 is the number of sequences */
+    int16_t* surround    /* (i/o) The contribution from this sequence
+                                  summed with earlier contributions */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c
new file mode 100644
index 0000000000..be582f2e23
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpInput.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/hp_input.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  high-pass filter of input with *0.5 and saturation
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_HpInput(
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
+                                                                   {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                                                                   is assumed to be 1.0 */
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                                                                   yhi[n-2] ylow[n-2] */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len)      /* (i)   Number of samples to filter */
+{
+  size_t i;
+  int32_t tmpW32;
+  int32_t tmpW32b;
+
+  for (i=0; i<len; i++) {
+
+    /*
+        y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
+        + (-a[1])*y[i-1] + (-a[2])*y[i-2];
+    */
+
+    tmpW32 = y[1] * ba[3];  /* (-a[1])*y[i-1] (low part) */
+    tmpW32 += y[3] * ba[4];  /* (-a[2])*y[i-2] (low part) */
+    tmpW32 = (tmpW32>>15);
+    tmpW32 += y[0] * ba[3];  /* (-a[1])*y[i-1] (high part) */
+    tmpW32 += y[2] * ba[4];  /* (-a[2])*y[i-2] (high part) */
+    tmpW32 = (tmpW32<<1);
+
+    tmpW32 += signal[i] * ba[0];  /* b[0]*x[0] */
+    tmpW32 += x[0] * ba[1];  /* b[1]*x[i-1] */
+    tmpW32 += x[1] * ba[2];  /* b[2]*x[i-2] */
+
+    /* Update state (input part) */
+    x[1] = x[0];
+    x[0] = signal[i];
+
+    /* Rounding in Q(12+1), i.e. add 2^12 */
+    tmpW32b = tmpW32 + 4096;
+
+    /* Saturate (to 2^28) so that the HP filtered signal does not overflow */
+    tmpW32b = WEBRTC_SPL_SAT((int32_t)268435455, tmpW32b, (int32_t)-268435456);
+
+    /* Convert back to Q0 and multiply with 0.5 */
+    signal[i] = (int16_t)(tmpW32b >> 13);
+
+    /* Update state (filtered part) */
+    y[2] = y[0];
+    y[3] = y[1];
+
+    /* upshift tmpW32 by 3 with saturation */
+    if (tmpW32>268435455) {
+      tmpW32 = WEBRTC_SPL_WORD32_MAX;
+    } else if (tmpW32<-268435456) {
+      tmpW32 = WEBRTC_SPL_WORD32_MIN;
+    } else {
+      tmpW32 <<= 3;
+    }
+
+    y[0] = (int16_t)(tmpW32 >> 16);
+    y[1] = (int16_t)((tmpW32 - (y[0] << 16)) >> 1);
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.h
new file mode 100644
index 0000000000..9143d8efed
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpInput.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+// clang-format off
+// Bad job here. https://bugs.llvm.org/show_bug.cgi?id=34274
+void WebRtcIlbcfix_HpInput(
+    int16_t* signal, /* (i/o) signal vector */
+    int16_t* ba,     /* (i)   B- and A-coefficients (2:nd order)
+                              {b[0] b[1] b[2] -a[1] -a[2]}
+                              a[0] is assumed to be 1.0 */
+    int16_t* y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                              yhi[n-2] ylow[n-2] */
+    int16_t* x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len);     /* (i)   Number of samples to filter */
+// clang-format on
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c
new file mode 100644
index 0000000000..cc5f6dcd37
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpOutput.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  high-pass filter of output and *2 with saturation
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_HpOutput(
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
+                                                                   {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                                                                   is assumed to be 1.0 */
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                                                                   yhi[n-2] ylow[n-2] */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len)      /* (i)   Number of samples to filter */
+{
+  size_t i;
+  int32_t tmpW32;
+  int32_t tmpW32b;
+
+  for (i=0; i<len; i++) {
+
+    /*
+      y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
+      + (-a[1])*y[i-1] + (-a[2])*y[i-2];
+    */
+
+    tmpW32 = y[1] * ba[3];  /* (-a[1])*y[i-1] (low part) */
+    tmpW32 += y[3] * ba[4];  /* (-a[2])*y[i-2] (low part) */
+    tmpW32 = (tmpW32>>15);
+    tmpW32 += y[0] * ba[3];  /* (-a[1])*y[i-1] (high part) */
+    tmpW32 += y[2] * ba[4];  /* (-a[2])*y[i-2] (high part) */
+    tmpW32 *= 2;
+
+    tmpW32 += signal[i] * ba[0];  /* b[0]*x[0] */
+    tmpW32 += x[0] * ba[1];  /* b[1]*x[i-1] */
+    tmpW32 += x[1] * ba[2];  /* b[2]*x[i-2] */
+
+    /* Update state (input part) */
+    x[1] = x[0];
+    x[0] = signal[i];
+
+    /* Rounding in Q(12-1), i.e. add 2^10 */
+    tmpW32b = tmpW32 + 1024;
+
+    /* Saturate (to 2^26) so that the HP filtered signal does not overflow */
+    tmpW32b = WEBRTC_SPL_SAT((int32_t)67108863, tmpW32b, (int32_t)-67108864);
+
+    /* Convert back to Q0 and multiply with 2 */
+    signal[i] = (int16_t)(tmpW32b >> 11);
+
+    /* Update state (filtered part) */
+    y[2] = y[0];
+    y[3] = y[1];
+
+    /* upshift tmpW32 by 3 with saturation */
+    if (tmpW32>268435455) {
+      tmpW32 = WEBRTC_SPL_WORD32_MAX;
+    } else if (tmpW32<-268435456) {
+      tmpW32 = WEBRTC_SPL_WORD32_MIN;
+    } else {
+      tmpW32 *= 8;
+    }
+
+    y[0] = (int16_t)(tmpW32 >> 16);
+    y[1] = (int16_t)((tmpW32 & 0xffff) >> 1);
+
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.h
new file mode 100644
index 0000000000..6d1bd3cd88
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpOutput.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+// clang-format off
+// Bad job here. https://bugs.llvm.org/show_bug.cgi?id=34274
+void WebRtcIlbcfix_HpOutput(
+    int16_t* signal, /* (i/o) signal vector */
+    int16_t* ba,     /* (i)   B- and A-coefficients (2:nd order)
+                              {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                              is assumed to be 1.0 */
+    int16_t* y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                              yhi[n-2] ylow[n-2] */
+    int16_t* x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len);     /* (i)   Number of samples to filter */
+// clang-format on
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c
new file mode 100644
index 0000000000..ba6c3e46c3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -0,0 +1,288 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBCInterface.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+#include <stdlib.h>
+
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/init_encode.h"
+#include "rtc_base/checks.h"
+
+int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
+                                    int16_t* ILBCENC_inst_Addr,
+                                    int16_t* size) {
+  *iLBC_encinst=(IlbcEncoderInstance*)ILBCENC_inst_Addr;
+  *size=sizeof(IlbcEncoder)/sizeof(int16_t);
+  if (*iLBC_encinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance** iLBC_decinst,
+                                    int16_t* ILBCDEC_inst_Addr,
+                                    int16_t* size) {
+  *iLBC_decinst=(IlbcDecoderInstance*)ILBCDEC_inst_Addr;
+  *size=sizeof(IlbcDecoder)/sizeof(int16_t);
+  if (*iLBC_decinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance **iLBC_encinst) {
+  *iLBC_encinst=(IlbcEncoderInstance*)malloc(sizeof(IlbcEncoder));
+  if (*iLBC_encinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance **iLBC_decinst) {
+  *iLBC_decinst=(IlbcDecoderInstance*)malloc(sizeof(IlbcDecoder));
+  if (*iLBC_decinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance *iLBC_encinst) {
+  free(iLBC_encinst);
+  return(0);
+}
+
+int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance *iLBC_decinst) {
+  free(iLBC_decinst);
+  return(0);
+}
+
+int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
+                                  int16_t mode) {
+  if ((mode==20)||(mode==30)) {
+    WebRtcIlbcfix_InitEncode((IlbcEncoder*) iLBCenc_inst, mode);
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded) {
+  size_t pos = 0;
+  size_t encpos = 0;
+
+  if ((len != ((IlbcEncoder*)iLBCenc_inst)->blockl) &&
+#ifdef SPLIT_10MS
+      (len != 80) &&
+#endif
+      (len != 2*((IlbcEncoder*)iLBCenc_inst)->blockl) &&
+      (len != 3*((IlbcEncoder*)iLBCenc_inst)->blockl))
+  {
+    /* A maximum of 3 frames/packet is allowed */
+    return(-1);
+  } else {
+
+    /* call encoder */
+    while (pos<len) {
+      WebRtcIlbcfix_EncodeImpl((uint16_t*)&encoded[2 * encpos], &speechIn[pos],
+                               (IlbcEncoder*)iLBCenc_inst);
+#ifdef SPLIT_10MS
+      pos += 80;
+      if(((IlbcEncoder*)iLBCenc_inst)->section == 0)
+#else
+        pos += ((IlbcEncoder*)iLBCenc_inst)->blockl;
+#endif
+      encpos += ((IlbcEncoder*)iLBCenc_inst)->no_of_words;
+    }
+    return (int)(encpos*2);
+  }
+}
+
+int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
+                                  int16_t mode) {
+  if ((mode==20)||(mode==30)) {
+    WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, mode, 1);
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst) {
+  WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 20, 1);
+}
+void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst) {
+  WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 30, 1);
+}
+
+
+int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType)
+{
+  size_t i=0;
+  /* Allow for automatic switching between the frame sizes
+     (although you do get some discontinuity) */
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
+    /* ok, do nothing */
+  } else {
+    /* Test if the mode has changed */
+    if (((IlbcDecoder*)iLBCdec_inst)->mode==20) {
+      if ((len==NO_OF_BYTES_30MS)||
+          (len==2*NO_OF_BYTES_30MS)||
+          (len==3*NO_OF_BYTES_30MS)) {
+        WebRtcIlbcfix_InitDecode(
+            ((IlbcDecoder*)iLBCdec_inst), 30,
+            ((IlbcDecoder*)iLBCdec_inst)->use_enhancer);
+      } else {
+        /* Unsupported frame length */
+        return(-1);
+      }
+    } else {
+      if ((len==NO_OF_BYTES_20MS)||
+          (len==2*NO_OF_BYTES_20MS)||
+          (len==3*NO_OF_BYTES_20MS)) {
+        WebRtcIlbcfix_InitDecode(
+            ((IlbcDecoder*)iLBCdec_inst), 20,
+            ((IlbcDecoder*)iLBCdec_inst)->use_enhancer);
+      } else {
+        /* Unsupported frame length */
+        return(-1);
+      }
+    }
+  }
+
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
+    if (WebRtcIlbcfix_DecodeImpl(
+            &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
+            (const uint16_t*)&encoded
+                [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+            (IlbcDecoder*)iLBCdec_inst, 1) == -1)
+      return -1;
+    i++;
+  }
+  /* iLBC does not support VAD/CNG yet */
+  *speechType=1;
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
+{
+  size_t i=0;
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
+    /* ok, do nothing */
+  } else {
+    return(-1);
+  }
+
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
+    if (!WebRtcIlbcfix_DecodeImpl(
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
+        (const uint16_t*)&encoded
+            [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+        (IlbcDecoder*)iLBCdec_inst, 1))
+      return -1;
+    i++;
+  }
+  /* iLBC does not support VAD/CNG yet */
+  *speechType=1;
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
+{
+  size_t i=0;
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
+    /* ok, do nothing */
+  } else {
+    return(-1);
+  }
+
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
+    if (!WebRtcIlbcfix_DecodeImpl(
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
+        (const uint16_t*)&encoded
+            [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+        (IlbcDecoder*)iLBCdec_inst, 1))
+      return -1;
+    i++;
+  }
+  /* iLBC does not support VAD/CNG yet */
+  *speechType=1;
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
+                               int16_t* decoded,
+                               size_t noOfLostFrames) {
+  size_t i;
+  uint16_t dummy;
+
+  for (i=0;i<noOfLostFrames;i++) {
+    // PLC decoding shouldn't fail, because there is no external input data
+    // that can be bad.
+    int result = WebRtcIlbcfix_DecodeImpl(
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl], &dummy,
+        (IlbcDecoder*)iLBCdec_inst, 0);
+    RTC_CHECK_EQ(result, 0);
+  }
+  return (noOfLostFrames*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
+                              int16_t* decoded,
+                              size_t noOfLostFrames) {
+  /* Two input parameters not used, but needed for function pointers in NetEQ */
+  (void)(decoded = NULL);
+  (void)(noOfLostFrames = 0);
+
+  WebRtcSpl_MemSetW16(((IlbcDecoder*)iLBCdec_inst)->enh_buf, 0, ENH_BUFL);
+  ((IlbcDecoder*)iLBCdec_inst)->prev_enh_pl = 2;
+
+  return (0);
+}
+
+void WebRtcIlbcfix_version(char *version)
+{
+  strcpy((char*)version, "1.1.1");
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.h
new file mode 100644
index 0000000000..de8cfde111
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.h
@@ -0,0 +1,251 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * ilbc.h
+ *
+ * This header file contains all of the API's for iLBC.
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*
+ * Solution to support multiple instances
+ * Customer has to cast instance to proper type
+ */
+
+typedef struct iLBC_encinst_t_ IlbcEncoderInstance;
+
+typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
+
+/*
+ * Comfort noise constants
+ */
+
+#define ILBC_SPEECH 1
+#define ILBC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcIlbcfix_XxxAssign(...)
+ *
+ * These functions assigns the encoder/decoder instance to the specified
+ * memory location
+ *
+ * Input:
+ *     - XXX_xxxinst       : Pointer to created instance that should be
+ *                           assigned
+ *     - ILBCXXX_inst_Addr : Pointer to the desired memory space
+ *     - size              : The size that this structure occupies (in Word16)
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
+                                    int16_t* ILBCENC_inst_Addr,
+                                    int16_t* size);
+int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance** iLBC_decinst,
+                                    int16_t* ILBCDEC_inst_Addr,
+                                    int16_t* size);
+
+/****************************************************************************
+ * WebRtcIlbcfix_XxxAssign(...)
+ *
+ * These functions create a instance to the specified structure
+ *
+ * Input:
+ *      - XXX_inst        : Pointer to created instance that should be created
+ *
+ * Return value           :  0 - Ok
+ *                          -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance** iLBC_encinst);
+int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance** iLBC_decinst);
+
+/****************************************************************************
+ * WebRtcIlbcfix_XxxFree(...)
+ *
+ * These functions frees the dynamic memory of a specified instance
+ *
+ * Input:
+ *      - XXX_inst          : Pointer to created instance that should be freed
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance* iLBC_encinst);
+int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance* iLBC_decinst);
+
+/****************************************************************************
+ * WebRtcIlbcfix_EncoderInit(...)
+ *
+ * This function initializes a iLBC instance
+ *
+ * Input:
+ *      - iLBCenc_inst      : iLBC instance, i.e. the user that should receive
+ *                            be initialized
+ *      - frameLen          : The frame length of the codec 20/30 (ms)
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
+                                  int16_t frameLen);
+
+/****************************************************************************
+ * WebRtcIlbcfix_Encode(...)
+ *
+ * This function encodes one iLBC frame. Input speech length has be a
+ * multiple of the frame length.
+ *
+ * Input:
+ *      - iLBCenc_inst      : iLBC instance, i.e. the user that should encode
+ *                            a package
+ *      - speechIn          : Input speech vector
+ *      - len               : Samples in speechIn (160, 240, 320 or 480)
+ *
+ * Output:
+ *  - encoded               : The encoded data vector
+ *
+ * Return value             : >0 - Length (in bytes) of coded data
+ *                            -1 - Error
+ */
+
+int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcIlbcfix_DecoderInit(...)
+ *
+ * This function initializes a iLBC instance with either 20 or 30 ms frames
+ * Alternatively the WebRtcIlbcfix_DecoderInit_XXms can be used. Then it's
+ * not needed to specify the frame length with a variable.
+ *
+ * Input:
+ *      - IlbcDecoderInstance : iLBC decoder instance
+ *      - frameLen            : The frame length of the codec 20/30 (ms)
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
+                                  int16_t frameLen);
+void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst);
+void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst);
+
+/****************************************************************************
+ * WebRtcIlbcfix_Decode(...)
+ *
+ * This function decodes a packet with iLBC frame(s). Output speech length
+ * will be a multiple of 160 or 240 samples ((160 or 240)*frames/packet).
+ *
+ * Input:
+ *      - iLBCdec_inst      : iLBC instance, i.e. the user that should decode
+ *                            a packet
+ *      - encoded           : Encoded iLBC frame(s)
+ *      - len               : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded           : The decoded vector
+ *      - speechType        : 1 normal, 2 CNG
+ *
+ * Return value             : >0 - Samples in decoded vector
+ *                            -1 - Error
+ */
+
+int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType);
+int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType);
+int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcIlbcfix_DecodePlc(...)
+ *
+ * This function conducts PLC for iLBC frame(s). Output speech length
+ * will be a multiple of 160 or 240 samples.
+ *
+ * Input:
+ *      - iLBCdec_inst      : iLBC instance, i.e. the user that should perform
+ *                            a PLC
+ *      - noOfLostFrames    : Number of PLC frames to produce
+ *
+ * Output:
+ *      - decoded           : The "decoded" vector
+ *
+ * Return value             : Samples in decoded PLC vector
+ */
+
+size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
+                               int16_t* decoded,
+                               size_t noOfLostFrames);
+
+/****************************************************************************
+ * WebRtcIlbcfix_NetEqPlc(...)
+ *
+ * This function updates the decoder when a packet loss has occured, but it
+ * does not produce any PLC data. Function can be used if another PLC method
+ * is used (i.e NetEq).
+ *
+ * Input:
+ *      - iLBCdec_inst      : iLBC instance that should be updated
+ *      - noOfLostFrames    : Number of lost frames
+ *
+ * Output:
+ *      - decoded           : The "decoded" vector (nothing in this case)
+ *
+ * Return value             : Samples in decoded PLC vector
+ */
+
+size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
+                              int16_t* decoded,
+                              size_t noOfLostFrames);
+
+/****************************************************************************
+ * WebRtcIlbcfix_version(...)
+ *
+ * This function returns the version number of iLBC
+ *
+ * Output:
+ *      - version           : Version number of iLBC (maximum 20 char)
+ */
+
+void WebRtcIlbcfix_version(char* version);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
new file mode 100644
index 0000000000..689292f131
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
@@ -0,0 +1,140 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(IlbcTest, BadPacket) {
+  // Get a good packet.
+  AudioEncoderIlbcConfig config;
+  config.frame_size_ms = 20;  // We need 20 ms rather than the default 30 ms;
+                              // otherwise, all possible values of cb_index[2]
+                              // are valid.
+  AudioEncoderIlbcImpl encoder(config, 102);
+  std::vector<int16_t> samples(encoder.SampleRateHz() / 100, 4711);
+  rtc::Buffer packet;
+  int num_10ms_chunks = 0;
+  while (packet.size() == 0) {
+    encoder.Encode(0, samples, &packet);
+    num_10ms_chunks += 1;
+  }
+
+  // Break the packet by setting all bits of the unsigned 7-bit number
+  // cb_index[2] to 1, giving it a value of 127. For a 20 ms packet, this is
+  // too large.
+  EXPECT_EQ(38u, packet.size());
+  rtc::Buffer bad_packet(packet.data(), packet.size());
+  bad_packet[29] |= 0x3f;  // Bits 1-6.
+  bad_packet[30] |= 0x80;  // Bit 0.
+
+  // Decode the bad packet. We expect the decoder to respond by returning -1.
+  AudioDecoderIlbcImpl decoder;
+  std::vector<int16_t> decoded_samples(num_10ms_chunks * samples.size());
+  AudioDecoder::SpeechType speech_type;
+  EXPECT_EQ(-1, decoder.Decode(bad_packet.data(), bad_packet.size(),
+                               encoder.SampleRateHz(),
+                               sizeof(int16_t) * decoded_samples.size(),
+                               decoded_samples.data(), &speech_type));
+
+  // Decode the good packet. This should work, because the failed decoding
+  // should not have left the decoder in a broken state.
+  EXPECT_EQ(static_cast<int>(decoded_samples.size()),
+            decoder.Decode(packet.data(), packet.size(), encoder.SampleRateHz(),
+                           sizeof(int16_t) * decoded_samples.size(),
+                           decoded_samples.data(), &speech_type));
+}
+
+class SplitIlbcTest : public ::testing::TestWithParam<std::pair<int, int> > {
+ protected:
+  virtual void SetUp() {
+    const std::pair<int, int> parameters = GetParam();
+    num_frames_ = parameters.first;
+    frame_length_ms_ = parameters.second;
+    frame_length_bytes_ = (frame_length_ms_ == 20) ? 38 : 50;
+  }
+  size_t num_frames_;
+  int frame_length_ms_;
+  size_t frame_length_bytes_;
+};
+
+TEST_P(SplitIlbcTest, NumFrames) {
+  AudioDecoderIlbcImpl decoder;
+  const size_t frame_length_samples = frame_length_ms_ * 8;
+  const auto generate_payload = [](size_t payload_length_bytes) {
+    rtc::Buffer payload(payload_length_bytes);
+    // Fill payload with increasing integers {0, 1, 2, ...}.
+    for (size_t i = 0; i < payload.size(); ++i) {
+      payload[i] = static_cast<uint8_t>(i);
+    }
+    return payload;
+  };
+
+  const auto results = decoder.ParsePayload(
+      generate_payload(frame_length_bytes_ * num_frames_), 0);
+  EXPECT_EQ(num_frames_, results.size());
+
+  size_t frame_num = 0;
+  uint8_t payload_value = 0;
+  for (const auto& result : results) {
+    EXPECT_EQ(frame_length_samples * frame_num, result.timestamp);
+    const LegacyEncodedAudioFrame* frame =
+        static_cast<const LegacyEncodedAudioFrame*>(result.frame.get());
+    const rtc::Buffer& payload = frame->payload();
+    EXPECT_EQ(frame_length_bytes_, payload.size());
+    for (size_t i = 0; i < payload.size(); ++i, ++payload_value) {
+      EXPECT_EQ(payload_value, payload[i]);
+    }
+    ++frame_num;
+  }
+}
+
+// Test 1 through 5 frames of 20 and 30 ms size.
+// Also test the maximum number of frames in one packet for 20 and 30 ms.
+// The maximum is defined by the largest payload length that can be uniquely
+// resolved to a frame size of either 38 bytes (20 ms) or 50 bytes (30 ms).
+INSTANTIATE_TEST_SUITE_P(
+    IlbcTest,
+    SplitIlbcTest,
+    ::testing::Values(std::pair<int, int>(1, 20),  // 1 frame, 20 ms.
+                      std::pair<int, int>(2, 20),  // 2 frames, 20 ms.
+                      std::pair<int, int>(3, 20),  // And so on.
+                      std::pair<int, int>(4, 20),
+                      std::pair<int, int>(5, 20),
+                      std::pair<int, int>(24, 20),
+                      std::pair<int, int>(1, 30),
+                      std::pair<int, int>(2, 30),
+                      std::pair<int, int>(3, 30),
+                      std::pair<int, int>(4, 30),
+                      std::pair<int, int>(5, 30),
+                      std::pair<int, int>(18, 30)));
+
+// Test too large payload size.
+TEST(IlbcTest, SplitTooLargePayload) {
+  AudioDecoderIlbcImpl decoder;
+  constexpr size_t kPayloadLengthBytes = 950;
+  const auto results =
+      decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0);
+  EXPECT_TRUE(results.empty());
+}
+
+// Payload not an integer number of frames.
+TEST(IlbcTest, SplitUnevenPayload) {
+  AudioDecoderIlbcImpl decoder;
+  constexpr size_t kPayloadLengthBytes = 39;  // Not an even number of frames.
+  const auto results =
+      decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0);
+  EXPECT_TRUE(results.empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c
new file mode 100644
index 0000000000..d78f81a897
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvDec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_IndexConvDec(
+    int16_t *index   /* (i/o) Codebook indexes */
+                                ){
+  int k;
+
+  for (k=4;k<6;k++) {
+    /* Readjust the second and third codebook index for the first 40 sample
+       so that they look the same as the first (in terms of lag)
+    */
+    if ((index[k]>=44)&&(index[k]<108)) {
+      index[k]+=64;
+    } else if ((index[k]>=108)&&(index[k]<128)) {
+      index[k]+=128;
+    } else {
+      /* ERROR */
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h
new file mode 100644
index 0000000000..4f08ce04df
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvDec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_IndexConvDec(int16_t* index /* (i/o) Codebook indexes */
+                                );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c
new file mode 100644
index 0000000000..83144150b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ IiLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvEnc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/index_conv_enc.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Convert the codebook indexes to make the search easier
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_IndexConvEnc(
+    int16_t *index   /* (i/o) Codebook indexes */
+                                ){
+  int k;
+
+  for (k=4;k<6;k++) {
+    /* Readjust the second and third codebook index so that it is
+       packetized into 7 bits (before it was put in lag-wise the same
+       way as for the first codebook which uses 8 bits)
+    */
+    if ((index[k]>=108)&&(index[k]<172)) {
+      index[k]-=64;
+    } else if (index[k]>=236) {
+      index[k]-=128;
+    } else {
+      /* ERROR */
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h
new file mode 100644
index 0000000000..4fbf98084e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvEnc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Convert the codebook indexes to make the search easier
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_IndexConvEnc(int16_t* index /* (i/o) Codebook indexes */
+                                );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c
new file mode 100644
index 0000000000..3eb41e33b0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+	iLBC Speech Coder ANSI-C Source Code
+
+	WebRtcIlbcfix_InitDecode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of decoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
+    IlbcDecoder *iLBCdec_inst,  /* (i/o) Decoder instance */
+    int16_t mode,  /* (i) frame size mode */
+    int use_enhancer) {  /* (i) 1: use enhancer, 0: no enhancer */
+  int i;
+
+  iLBCdec_inst->mode = mode;
+
+  /* Set all the variables that are dependent on the frame size mode */
+  if (mode==30) {
+    iLBCdec_inst->blockl = BLOCKL_30MS;
+    iLBCdec_inst->nsub = NSUB_30MS;
+    iLBCdec_inst->nasub = NASUB_30MS;
+    iLBCdec_inst->lpc_n = LPC_N_30MS;
+    iLBCdec_inst->no_of_bytes = NO_OF_BYTES_30MS;
+    iLBCdec_inst->no_of_words = NO_OF_WORDS_30MS;
+    iLBCdec_inst->state_short_len=STATE_SHORT_LEN_30MS;
+  }
+  else if (mode==20) {
+    iLBCdec_inst->blockl = BLOCKL_20MS;
+    iLBCdec_inst->nsub = NSUB_20MS;
+    iLBCdec_inst->nasub = NASUB_20MS;
+    iLBCdec_inst->lpc_n = LPC_N_20MS;
+    iLBCdec_inst->no_of_bytes = NO_OF_BYTES_20MS;
+    iLBCdec_inst->no_of_words = NO_OF_WORDS_20MS;
+    iLBCdec_inst->state_short_len=STATE_SHORT_LEN_20MS;
+  }
+  else {
+    return(-1);
+  }
+
+  /* Reset all the previous LSF to mean LSF */
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+
+  /* Clear the synthesis filter memory */
+  WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
+
+  /* Set the old synthesis filter to {1.0 0.0 ... 0.0} */
+  WebRtcSpl_MemSetW16(iLBCdec_inst->old_syntdenum, 0, ((LPC_FILTERORDER + 1)*NSUB_MAX));
+  for (i=0; i<NSUB_MAX; i++) {
+    iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)] = 4096;
+  }
+
+  /* Clear the variables that are used for the PLC */
+  iLBCdec_inst->last_lag = 20;
+  iLBCdec_inst->consPLICount = 0;
+  iLBCdec_inst->prevPLI = 0;
+  iLBCdec_inst->perSquare = 0;
+  iLBCdec_inst->prevLag = 120;
+  iLBCdec_inst->prevLpc[0] = 4096;
+  WebRtcSpl_MemSetW16(iLBCdec_inst->prevLpc+1, 0, LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(iLBCdec_inst->prevResidual, 0, BLOCKL_MAX);
+
+  /* Initialize the seed for the random number generator */
+  iLBCdec_inst->seed = 777;
+
+  /* Set the filter state of the HP filter to 0 */
+  WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
+  WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
+
+  /* Set the variables that are used in the ehnahcer */
+  iLBCdec_inst->use_enhancer = use_enhancer;
+  WebRtcSpl_MemSetW16(iLBCdec_inst->enh_buf, 0, (ENH_BUFL+ENH_BUFL_FILTEROVERHEAD));
+  for (i=0;i<ENH_NBLOCKS_TOT;i++) {
+    iLBCdec_inst->enh_period[i]=160; /* Q(-4) */
+  }
+
+  iLBCdec_inst->prev_enh_pl = 0;
+
+  return (int)(iLBCdec_inst->blockl);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.h
new file mode 100644
index 0000000000..a2b7b91287
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitDecode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of decoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitDecode(/* (o) Number of decoded samples */
+                             IlbcDecoder*
+                                 iLBCdec_inst, /* (i/o) Decoder instance */
+                             int16_t mode,     /* (i) frame size mode */
+                             int use_enhancer  /* (i) 1 to use enhancer
+                                                  0 to run without enhancer */
+                             );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c
new file mode 100644
index 0000000000..aa858e94bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitEncode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/init_encode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of encoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
+    IlbcEncoder *iLBCenc_inst,  /* (i/o) Encoder instance */
+    int16_t mode) {  /* (i) frame size mode */
+  iLBCenc_inst->mode = mode;
+
+  /* Set all the variables that are dependent on the frame size mode */
+  if (mode==30) {
+    iLBCenc_inst->blockl = BLOCKL_30MS;
+    iLBCenc_inst->nsub = NSUB_30MS;
+    iLBCenc_inst->nasub = NASUB_30MS;
+    iLBCenc_inst->lpc_n = LPC_N_30MS;
+    iLBCenc_inst->no_of_bytes = NO_OF_BYTES_30MS;
+    iLBCenc_inst->no_of_words = NO_OF_WORDS_30MS;
+    iLBCenc_inst->state_short_len=STATE_SHORT_LEN_30MS;
+  }
+  else if (mode==20) {
+    iLBCenc_inst->blockl = BLOCKL_20MS;
+    iLBCenc_inst->nsub = NSUB_20MS;
+    iLBCenc_inst->nasub = NASUB_20MS;
+    iLBCenc_inst->lpc_n = LPC_N_20MS;
+    iLBCenc_inst->no_of_bytes = NO_OF_BYTES_20MS;
+    iLBCenc_inst->no_of_words = NO_OF_WORDS_20MS;
+    iLBCenc_inst->state_short_len=STATE_SHORT_LEN_20MS;
+  }
+  else {
+    return(-1);
+  }
+
+  /* Clear the buffers and set the previous LSF and LSP to the mean value */
+  WebRtcSpl_MemSetW16(iLBCenc_inst->anaMem, 0, LPC_FILTERORDER);
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(iLBCenc_inst->lpc_buffer, 0, LPC_LOOKBACK + BLOCKL_MAX);
+
+  /* Set the filter state of the HP filter to 0 */
+  WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemx, 0, 2);
+  WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemy, 0, 4);
+
+#ifdef SPLIT_10MS
+  /*Zeroing the past samples for 10msec Split*/
+  WebRtcSpl_MemSetW16(iLBCenc_inst->past_samples,0,160);
+  iLBCenc_inst->section = 0;
+#endif
+
+  return (int)(iLBCenc_inst->no_of_bytes);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.h
new file mode 100644
index 0000000000..4ada6a30c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitEncode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of encoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitEncode(/* (o) Number of bytes encoded */
+                             IlbcEncoder*
+                                 iLBCenc_inst, /* (i/o) Encoder instance */
+                             int16_t mode      /* (i) frame size mode */
+                             );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c
new file mode 100644
index 0000000000..17ed244bd4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Interpolate.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  interpolation between vectors
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Interpolate(
+    int16_t *out, /* (o) output vector */
+    int16_t *in1, /* (i) first input vector */
+    int16_t *in2, /* (i) second input vector */
+    int16_t coef, /* (i) weight coefficient in Q14 */
+    int16_t length)  /* (i) number of sample is vectors */
+{
+  int i;
+  int16_t invcoef;
+
+  /*
+    Performs the operation out[i] = in[i]*coef + (1-coef)*in2[i] (with rounding)
+  */
+
+  invcoef = 16384 - coef; /* 16384 = 1.0 (Q14)*/
+  for (i = 0; i < length; i++) {
+    out[i] = (int16_t)((coef * in1[i] + invcoef * in2[i] + 8192) >> 14);
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.h
new file mode 100644
index 0000000000..892082b75c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Interpolate.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  interpolation between vectors
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Interpolate(
+    int16_t* out,    /* (o) output vector */
+    int16_t* in1,    /* (i) first input vector */
+    int16_t* in2,    /* (i) second input vector */
+    int16_t coef,    /* (i) weight coefficient in Q14 */
+    int16_t length); /* (i) number of sample is vectors */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
new file mode 100644
index 0000000000..6dddd6fb86
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InterpolateSamples.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/interpolate_samples.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_InterpolateSamples(
+    int16_t *interpSamples, /* (o) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    size_t lMem    /* (i) Length of the CB memory */
+                                      ) {
+  int16_t *ppi, *ppo, i, j, temp1, temp2;
+  int16_t *tmpPtr;
+
+  /* Calculate the 20 vectors of interpolated samples (4 samples each)
+     that are used in the codebooks for lag 20 to 39 */
+  tmpPtr = interpSamples;
+  for (j=0; j<20; j++) {
+    temp1 = 0;
+    temp2 = 3;
+    ppo = CBmem+lMem-4;
+    ppi = CBmem+lMem-j-24;
+    for (i=0; i<4; i++) {
+
+      *tmpPtr++ = (int16_t)((WebRtcIlbcfix_kAlpha[temp2] * *ppo) >> 15) +
+          (int16_t)((WebRtcIlbcfix_kAlpha[temp1] * *ppi) >> 15);
+
+      ppo++;
+      ppi++;
+      temp1++;
+      temp2--;
+    }
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
new file mode 100644
index 0000000000..bc665d7854
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InterpolateSamples.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Construct the interpolated samples for the Augmented CB
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_InterpolateSamples(
+    int16_t* interpSamples, /* (o) The interpolated samples */
+    int16_t* CBmem,         /* (i) The CB memory */
+    size_t lMem             /* (i) Length of the CB memory */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
new file mode 100644
index 0000000000..89f6d29724
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LpcEncode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lpc_encode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+#include "modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h"
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_quant.h"
+
+/*----------------------------------------------------------------*
+ *  lpc encoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LpcEncode(
+    int16_t *syntdenum,  /* (i/o) synthesis filter coefficients
+                                           before/after encoding */
+    int16_t *weightdenum, /* (i/o) weighting denumerator coefficients
+                                   before/after encoding */
+    int16_t *lsf_index,  /* (o) lsf quantization index */
+    int16_t *data,   /* (i) Speech to do LPC analysis on */
+    IlbcEncoder *iLBCenc_inst
+    /* (i/o) the encoder state structure */
+                              ) {
+  /* Stack based */
+  int16_t lsf[LPC_FILTERORDER * LPC_N_MAX];
+  int16_t lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
+
+  /* Calculate LSF's from the input speech */
+  WebRtcIlbcfix_SimpleLpcAnalysis(lsf, data, iLBCenc_inst);
+
+  /* Quantize the LSF's */
+  WebRtcIlbcfix_SimpleLsfQ(lsfdeq, lsf_index, lsf, iLBCenc_inst->lpc_n);
+
+  /* Stableize the LSF's if needed */
+  WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCenc_inst->lpc_n);
+
+  /* Calculate the synthesis and weighting filter coefficients from
+     the optimal LSF and the dequantized LSF */
+  WebRtcIlbcfix_SimpleInterpolateLsf(syntdenum, weightdenum,
+                                     lsf, lsfdeq, iLBCenc_inst->lsfold,
+                                     iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
new file mode 100644
index 0000000000..a67b77acbf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LpcEncode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc encoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LpcEncode(
+    int16_t* syntdenum,   /* (i/o) synthesis filter coefficients
+                                   before/after encoding */
+    int16_t* weightdenum, /* (i/o) weighting denumerator coefficients
+                                   before/after encoding */
+    int16_t* lsf_index,   /* (o) lsf quantization index */
+    int16_t* data,        /* (i) Speech to do LPC analysis on */
+    IlbcEncoder* iLBCenc_inst
+    /* (i/o) the encoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c
new file mode 100644
index 0000000000..9f0e19a2d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfCheck.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  check for stability of lsf coefficients
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_LsfCheck(
+    int16_t *lsf, /* LSF parameters */
+    int dim, /* dimension of LSF */
+    int NoAn)  /* No of analysis per frame */
+{
+  int k,n,m, Nit=2, change=0,pos;
+  const int16_t eps=319;  /* 0.039 in Q13 (50 Hz)*/
+  const int16_t eps2=160;  /* eps/2.0 in Q13;*/
+  const int16_t maxlsf=25723; /* 3.14; (4000 Hz)*/
+  const int16_t minlsf=82;  /* 0.01; (0 Hz)*/
+
+  /* LSF separation check*/
+  for (n=0;n<Nit;n++) {  /* Run through a 2 times */
+    for (m=0;m<NoAn;m++) { /* Number of analyses per frame */
+      for (k=0;k<(dim-1);k++) {
+        pos=m*dim+k;
+
+        /* Seperate coefficients with a safety margin of 50 Hz */
+        if ((lsf[pos+1]-lsf[pos])<eps) {
+
+          if (lsf[pos+1]<lsf[pos]) {
+            lsf[pos+1]= lsf[pos]+eps2;
+            lsf[pos]= lsf[pos+1]-eps2;
+          } else {
+            lsf[pos]-=eps2;
+            lsf[pos+1]+=eps2;
+          }
+          change=1;
+        }
+
+        /* Limit minimum and maximum LSF */
+        if (lsf[pos]<minlsf) {
+          lsf[pos]=minlsf;
+          change=1;
+        }
+
+        if (lsf[pos]>maxlsf) {
+          lsf[pos]=maxlsf;
+          change=1;
+        }
+      }
+    }
+  }
+
+  return change;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.h
new file mode 100644
index 0000000000..9ba90a31e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfCheck.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  check for stability of lsf coefficients
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_LsfCheck(int16_t* lsf, /* LSF parameters */
+                           int dim,      /* dimension of LSF */
+                           int NoAn);    /* No of analysis per frame */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
new file mode 100644
index 0000000000..04de5e7e6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LspInterpolate2PolyDec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
+
+/*----------------------------------------------------------------*
+ *  interpolation of lsf coefficients for the decoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LspInterpolate2PolyDec(
+    int16_t *a,   /* (o) lpc coefficients Q12 */
+    int16_t *lsf1,  /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2,  /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
+                                   lsf1 and lsf2 Q14 */
+    int16_t length  /* (i) length of coefficient vectors */
+                                          ){
+  int16_t lsftmp[LPC_FILTERORDER];
+
+  /* interpolate LSF */
+  WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
+
+  /* Compute the filter coefficients from the LSF */
+  WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
new file mode 100644
index 0000000000..6cc9d9746d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LspInterpolate2PolyDec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  interpolation of lsf coefficients for the decoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LspInterpolate2PolyDec(
+    int16_t* a,    /* (o) lpc coefficients Q12 */
+    int16_t* lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t* lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
+                                   lsf1 and lsf2 Q14 */
+    int16_t length /* (i) length of coefficient vectors */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
new file mode 100644
index 0000000000..618821216c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator and conversion from lsf to a coefficients
+ *  (subrutine to SimpleInterpolateLSF)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
+    int16_t *a,  /* (o) lpc coefficients Q12 */
+    int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef, /* (i) weighting coefficient to use between
+                           lsf1 and lsf2 Q14 */
+    int16_t length /* (i) length of coefficient vectors */
+                                          ) {
+  /* Stack based */
+  int16_t lsftmp[LPC_FILTERORDER];
+
+  /* interpolate LSF */
+  WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
+
+  /* Compute the filter coefficients from the LSF */
+  WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
new file mode 100644
index 0000000000..b278a10f4b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator and conversion from lsf to a coefficients
+ *  (subrutine to SimpleInterpolateLSF)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
+    int16_t* a,    /* (o) lpc coefficients Q12 */
+    int16_t* lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t* lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
+                            lsf1 and lsf2 Q14 */
+    int16_t length /* (i) length of coefficient vectors */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
new file mode 100644
index 0000000000..ee8292f394
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Lsp.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_to_lsp.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  conversion from lsf to lsp coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Lsp(
+    int16_t *lsf, /* (i) lsf in Q13 values between 0 and pi */
+    int16_t *lsp, /* (o) lsp in Q15 values between -1 and 1 */
+    int16_t m  /* (i) number of coefficients */
+                           ) {
+  int16_t i, k;
+  int16_t diff; /* difference, which is used for the
+                           linear approximation (Q8) */
+  int16_t freq; /* normalized frequency in Q15 (0..1) */
+  int32_t tmpW32;
+
+  for(i=0; i<m; i++)
+  {
+    freq = (int16_t)((lsf[i] * 20861) >> 15);
+    /* 20861: 1.0/(2.0*PI) in Q17 */
+    /*
+       Upper 8 bits give the index k and
+       Lower 8 bits give the difference, which needs
+       to be approximated linearly
+    */
+    k = freq >> 8;
+    diff = (freq&0x00ff);
+
+    /* Guard against getting outside table */
+
+    if (k>63) {
+      k = 63;
+    }
+
+    /* Calculate linear approximation */
+    tmpW32 = WebRtcIlbcfix_kCosDerivative[k] * diff;
+    lsp[i] = WebRtcIlbcfix_kCos[k] + (int16_t)(tmpW32 >> 12);
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
new file mode 100644
index 0000000000..6bc6c44dbd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Lsp.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  conversion from lsf to lsp coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Lsp(
+    int16_t* lsf, /* (i) lsf in Q13 values between 0 and pi */
+    int16_t* lsp, /* (o) lsp in Q15 values between -1 and 1 */
+    int16_t m     /* (i) number of coefficients */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
new file mode 100644
index 0000000000..8ca91d82f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Poly.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/get_lsp_poly.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_to_lsp.h"
+
+void WebRtcIlbcfix_Lsf2Poly(
+    int16_t *a,     /* (o) predictor coefficients (order = 10) in Q12 */
+    int16_t *lsf    /* (i) line spectral frequencies in Q13 */
+                            ) {
+  int32_t f[2][6]; /* f[0][] and f[1][] corresponds to
+                            F1(z) and F2(z) respectivly */
+  int32_t *f1ptr, *f2ptr;
+  int16_t *a1ptr, *a2ptr;
+  int32_t tmpW32;
+  int16_t lsp[10];
+  int i;
+
+  /* Convert lsf to lsp */
+  WebRtcIlbcfix_Lsf2Lsp(lsf, lsp, LPC_FILTERORDER);
+
+  /* Get F1(z) and F2(z) from the lsp */
+  f1ptr=f[0];
+  f2ptr=f[1];
+  WebRtcIlbcfix_GetLspPoly(&lsp[0],f1ptr);
+  WebRtcIlbcfix_GetLspPoly(&lsp[1],f2ptr);
+
+  /* for i = 5 down to 1
+     Compute f1[i] += f1[i-1];
+     and     f2[i] += f2[i-1];
+  */
+  f1ptr=&f[0][5];
+  f2ptr=&f[1][5];
+  for (i=5; i>0; i--)
+  {
+    (*f1ptr) += (*(f1ptr-1));
+    (*f2ptr) -= (*(f2ptr-1));
+    f1ptr--;
+    f2ptr--;
+  }
+
+  /* Get the A(z) coefficients
+     a[0] = 1.0
+     for i = 1 to 5
+     a[i] = (f1[i] + f2[i] + round)>>13;
+     for i = 1 to 5
+     a[11-i] = (f1[i] - f2[i] + round)>>13;
+  */
+  a[0]=4096;
+  a1ptr=&a[1];
+  a2ptr=&a[10];
+  f1ptr=&f[0][1];
+  f2ptr=&f[1][1];
+  for (i=5; i>0; i--)
+  {
+    tmpW32 = (*f1ptr) + (*f2ptr);
+    *a1ptr = (int16_t)((tmpW32 + 4096) >> 13);
+
+    tmpW32 = (*f1ptr) - (*f2ptr);
+    *a2ptr = (int16_t)((tmpW32 + 4096) >> 13);
+
+    a1ptr++;
+    a2ptr--;
+    f1ptr++;
+    f2ptr++;
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
new file mode 100644
index 0000000000..f26d3a8d2d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Poly.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Convert from LSF coefficients to A coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Poly(
+    int16_t* a,  /* (o) predictor coefficients (order = 10) in Q12 */
+    int16_t* lsf /* (i) line spectral frequencies in Q13 */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
new file mode 100644
index 0000000000..227f4d45b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsp2Lsf.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsp_to_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  conversion from LSP coefficients to LSF coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsp2Lsf(
+    int16_t *lsp, /* (i) lsp vector -1...+1 in Q15 */
+    int16_t *lsf, /* (o) Lsf vector 0...Pi in Q13
+                           (ordered, so that lsf[i]<lsf[i+1]) */
+    int16_t m  /* (i) Number of coefficients */
+                           )
+{
+  int16_t i, k;
+  int16_t diff; /* diff between table value and desired value (Q15) */
+  int16_t freq; /* lsf/(2*pi) (Q16) */
+  int16_t *lspPtr, *lsfPtr, *cosTblPtr;
+  int16_t tmp;
+
+  /* set the index to maximum index value in WebRtcIlbcfix_kCos */
+  k = 63;
+
+  /*
+     Start with the highest LSP and then work the way down
+     For each LSP the lsf is calculated by first order approximation
+     of the acos(x) function
+  */
+  lspPtr = &lsp[9];
+  lsfPtr = &lsf[9];
+  cosTblPtr=(int16_t*)&WebRtcIlbcfix_kCos[k];
+  for(i=m-1; i>=0; i--)
+  {
+    /*
+       locate value in the table, which is just above lsp[i],
+       basically an approximation to acos(x)
+    */
+    while( (((int32_t)(*cosTblPtr)-(*lspPtr)) < 0)&&(k>0) )
+    {
+      k-=1;
+      cosTblPtr--;
+    }
+
+    /* Calculate diff, which is used in the linear approximation of acos(x) */
+    diff = (*lspPtr)-(*cosTblPtr);
+
+    /*
+       The linear approximation of acos(lsp[i]) :
+       acos(lsp[i])= k*512 + (WebRtcIlbcfix_kAcosDerivative[ind]*offset >> 11)
+    */
+
+    /* tmp (linear offset) in Q16 */
+    tmp = (int16_t)((WebRtcIlbcfix_kAcosDerivative[k] * diff) >> 11);
+
+    /* freq in Q16 */
+    freq = (k << 9) + tmp;
+
+    /* lsf = freq*2*pi */
+    (*lsfPtr) = (int16_t)(((int32_t)freq*25736)>>15);
+
+    lsfPtr--;
+    lspPtr--;
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
new file mode 100644
index 0000000000..c2f4b7692d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsp2Lsf.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  conversion from LSP coefficients to LSF coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsp2Lsf(
+    int16_t* lsp, /* (i) lsp vector -1...+1 in Q15 */
+    int16_t* lsf, /* (o) Lsf vector 0...Pi in Q13
+                           (ordered, so that lsf[i]<lsf[i+1]) */
+    int16_t m     /* (i) Number of coefficients */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c
new file mode 100644
index 0000000000..9b870e0ef0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_MyCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/my_corr.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * compute cross correlation between sequences
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_MyCorr(
+    int32_t* corr,  /* (o) correlation of seq1 and seq2 */
+    const int16_t* seq1,  /* (i) first sequence */
+    size_t dim1,  /* (i) dimension first seq1 */
+    const int16_t* seq2, /* (i) second sequence */
+    size_t dim2   /* (i) dimension seq2 */
+                          ){
+  uint32_t max1, max2;
+  size_t loops;
+  int right_shift;
+
+  // Calculate a right shift that will let us sum dim2 pairwise products of
+  // values from the two sequences without overflowing an int32_t. (The +1 in
+  // max1 and max2 are because WebRtcSpl_MaxAbsValueW16 will return 2**15 - 1
+  // if the input array contains -2**15.)
+  max1 = WebRtcSpl_MaxAbsValueW16(seq1, dim1) + 1;
+  max2 = WebRtcSpl_MaxAbsValueW16(seq2, dim2) + 1;
+  right_shift =
+      (64 - 31) - WebRtcSpl_CountLeadingZeros64((max1 * max2) * (uint64_t)dim2);
+  if (right_shift < 0) {
+    right_shift = 0;
+  }
+
+  loops=dim1-dim2+1;
+
+  /* Calculate the cross correlations */
+  WebRtcSpl_CrossCorrelation(corr, seq2, seq1, dim2, loops, right_shift, 1);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.h
new file mode 100644
index 0000000000..c0c2fa4a48
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_MyCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * compute cross correlation between sequences
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_MyCorr(int32_t* corr, /* (o) correlation of seq1 and seq2 */
+                          const int16_t* seq1, /* (i) first sequence */
+                          size_t dim1,         /* (i) dimension first seq1 */
+                          const int16_t* seq2, /* (i) second sequence */
+                          size_t dim2          /* (i) dimension seq2 */
+                          );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
new file mode 100644
index 0000000000..1ecdd96d5a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_NearestNeighbor.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/nearest_neighbor.h"
+
+void WebRtcIlbcfix_NearestNeighbor(size_t* index,
+                                   const size_t* array,
+                                   size_t value,
+                                   size_t arlength) {
+  size_t i;
+  size_t min_diff = (size_t)-1;
+  for (i = 0; i < arlength; i++) {
+    const size_t diff =
+        (array[i] < value) ? (value - array[i]) : (array[i] - value);
+    if (diff < min_diff) {
+      *index = i;
+      min_diff = diff;
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
new file mode 100644
index 0000000000..704cf2a37d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_NearestNeighbor.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * Find index in array such that the array element with said
+ * index is the element of said array closest to "value"
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_NearestNeighbor(
+    size_t* index,       /* (o) index of array element closest to value */
+    const size_t* array, /* (i) data array (Q2) */
+    size_t value,        /* (i) value (Q2) */
+    size_t arlength      /* (i) dimension of data array (==ENH_NBLOCKS_TOT) */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c
new file mode 100644
index 0000000000..dd44eb8fb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c
@@ -0,0 +1,253 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_PackBits.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/pack_bits.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_PackBits(
+    uint16_t *bitstream,   /* (o) The packetized bitstream */
+    iLBC_bits *enc_bits,  /* (i) Encoded bits */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
+                             ){
+  uint16_t *bitstreamPtr;
+  int i, k;
+  int16_t *tmpPtr;
+
+  bitstreamPtr=bitstream;
+
+  /* Class 1 bits of ULP */
+  /* First int16_t */
+  (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[0])<<10;   /* Bit 0..5  */
+  (*bitstreamPtr) |= (enc_bits->lsf[1])<<3;     /* Bit 6..12 */
+  (*bitstreamPtr) |= (enc_bits->lsf[2]&0x70)>>4;    /* Bit 13..15 */
+  bitstreamPtr++;
+  /* Second int16_t */
+  (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[2]&0xF)<<12;  /* Bit 0..3  */
+
+  if (mode==20) {
+    (*bitstreamPtr) |= (enc_bits->startIdx)<<10;    /* Bit 4..5  */
+    (*bitstreamPtr) |= (enc_bits->state_first)<<9;    /* Bit 6  */
+    (*bitstreamPtr) |= (enc_bits->idxForMax)<<3;    /* Bit 7..12 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[0])&0x70)>>4;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    (*bitstreamPtr) = ((enc_bits->cb_index[0])&0xE)<<12;  /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x18)<<8;  /* Bit 3..4  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x8)<<7;  /* Bit 5  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0xFE)<<2;  /* Bit 6..12 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x10)>>2;  /* Bit 13  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x8)>>2;  /* Bit 14  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x10)>>4;  /* Bit 15  */
+  } else { /* mode==30 */
+    (*bitstreamPtr) |= (enc_bits->lsf[3])<<6;     /* Bit 4..9  */
+    (*bitstreamPtr) |= (enc_bits->lsf[4]&0x7E)>>1;    /* Bit 10..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[4]&0x1)<<15;  /* Bit 0  */
+    (*bitstreamPtr) |= (enc_bits->lsf[5])<<8;     /* Bit 1..7  */
+    (*bitstreamPtr) |= (enc_bits->startIdx)<<5;     /* Bit 8..10 */
+    (*bitstreamPtr) |= (enc_bits->state_first)<<4;    /* Bit 11  */
+    (*bitstreamPtr) |= ((enc_bits->idxForMax)&0x3C)>>2;   /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 4:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->idxForMax&0x3)<<14; /* Bit 0..1  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x78)<<7;   /* Bit 2..5  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x10)<<5;  /* Bit 6  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x8)<<5;  /* Bit 7  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[3]&0xFC);   /* Bit 8..13 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[3]&0x10)>>3;  /* Bit 14  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x8)>>3;  /* Bit 15  */
+  }
+  /* Class 2 bits of ULP */
+  /* 4:th to 6:th int16_t for 20 ms case
+     5:th to 7:th int16_t for 30 ms case */
+  bitstreamPtr++;
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<3; k++) {
+    (*bitstreamPtr) = 0;
+    for (i=15; i>=0; i--) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 7:th int16_t */
+    (*bitstreamPtr) = 0;
+    for (i=15; i>6; i--) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4)<<4;  /* Bit 9  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<2;  /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x4)<<1;  /* Bit 12  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x8)>>1;  /* Bit 13  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)>>2;  /* Bit 14..15 */
+
+  } else { /* mode==30 */
+    /* 8:th int16_t */
+    (*bitstreamPtr) = 0;
+    for (i=15; i>5; i--) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x6)<<3;   /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x8);   /* Bit 12  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4);   /* Bit 13  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[3]&0x2);    /* Bit 14  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[6]&0x80)>>7;   /* Bit 15  */
+    bitstreamPtr++;
+    /* 9:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->cb_index[6]&0x7E)<<9;/* Bit 0..5  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[9]&0xFE)<<2;   /* Bit 6..12 */
+    (*bitstreamPtr) |= (enc_bits->cb_index[12]&0xE0)>>5;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 10:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->cb_index[12]&0x1E)<<11;/* Bit 0..3 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<8;  /* Bit 4..5  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x6)<<7;  /* Bit 6..7  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x18)<<3;  /* Bit 8..9  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)<<2;  /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[9]&0x10)>>1;  /* Bit 12  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[10]&0x8)>>1;  /* Bit 13  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[12]&0x10)>>3;  /* Bit 14  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[13]&0x8)>>3;  /* Bit 15  */
+  }
+  bitstreamPtr++;
+  /* Class 3 bits of ULP */
+  /*  8:th to 14:th int16_t for 20 ms case
+      11:th to 17:th int16_t for 30 ms case */
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<7; k++) {
+    (*bitstreamPtr) = 0;
+    for (i=14; i>=0; i-=2) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x3))<<i; /* Bit 15-i..14-i*/
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 15:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+    (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<13;  /* Bit 2  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<6;   /* Bit 3..9  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x7E)>>1;  /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 16:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[2])&0x1))<<15;
+    /* Bit 0  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<12;  /* Bit 1..3  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<10;  /* Bit 4..5  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[2]))<<7;   /* Bit 6..8  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<6;  /* Bit 9  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x7E)>>1;  /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 17:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[4])&0x1))<<15;
+    /* Bit 0  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[5])<<8;    /* Bit 1..7  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[6]);     /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 18:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[7]))<<8; /* Bit 0..7  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[8]);     /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->gain_index[3])&0x3))<<14;
+    /* Bit 0..1  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x3)<<12;  /* Bit 2..3  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[5]))<<9;   /* Bit 4..6  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<6;  /* Bit 7..9  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<4;  /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[8])<<1;   /* Bit 12..14 */
+  } else { /* mode==30 */
+    /* 18:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+    (*bitstreamPtr) |= (((enc_bits->idxVec[57])&0x3))<<12;  /* Bit 2..3  */
+    (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<11;  /* Bit 4  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<4;   /* Bit 5..11 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x78)>>3;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[2])&0x7)<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<10;  /* Bit 3..5  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<8;  /* Bit 6..7  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[2])&0x7)<<5;  /* Bit 8..10 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<4;  /* Bit 11  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x78)>>3;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 20:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[4])&0x7)<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[5]))<<6;   /* Bit 3..9  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[6])&0x1)<<5;  /* Bit 10  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[7])&0xF8)>>3;  /* Bit 11..15 */
+    bitstreamPtr++;
+    /* 21:st int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[7])&0x7)<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[8]))<<5;   /* Bit 3..10 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[9])&0x1)<<4;  /* Bit 11  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[10])&0xF0)>>4;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 22:nd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[10])&0xF)<<12;
+    /* Bit 0..3  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[11]))<<4;   /* Bit 4..11 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[12])&0x1)<<3;  /* Bit 12  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[13])&0xE0)>>5;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 23:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[13])&0x1F)<<11;
+    /* Bit 0..4  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[14]))<<3;   /* Bit 5..12 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x3)<<1;  /* Bit 13..14 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x1);   /* Bit 15  */
+    bitstreamPtr++;
+    /* 24:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->gain_index[5]))<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<10;  /* Bit 3..5  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<8;  /* Bit 6..7  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[8]))<<5;   /* Bit 8..10 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[9])&0xF)<<1;  /* Bit 11..14 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[10])&0x4)>>2;  /* Bit 15  */
+    bitstreamPtr++;
+    /* 25:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->gain_index[10])&0x3)<<14;
+    /* Bit 0..1  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[11]))<<11;  /* Bit 2..4  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[12])&0xF)<<7;  /* Bit 5..8  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[13])&0x7)<<4;  /* Bit 9..11 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[14]))<<1;   /* Bit 12..14 */
+  }
+  /* Last bit is automatically zero */
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.h
new file mode 100644
index 0000000000..8dcf41ce08
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_PackBits.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_PackBits(
+    uint16_t* bitstream, /* (o) The packetized bitstream */
+    iLBC_bits* enc_bits, /* (i) Encoded bits */
+    int16_t mode         /* (i) Codec mode (20 or 30) */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
new file mode 100644
index 0000000000..7192eaab49
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsf.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/lsp_to_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsp.h"
+
+void WebRtcIlbcfix_Poly2Lsf(
+    int16_t *lsf,   /* (o) lsf coefficients (Q13) */
+    int16_t *a    /* (i) A coefficients (Q12) */
+                            ) {
+  int16_t lsp[10];
+  WebRtcIlbcfix_Poly2Lsp(a, lsp, (int16_t*)WebRtcIlbcfix_kLspMean);
+  WebRtcIlbcfix_Lsp2Lsf(lsp, lsf, 10);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
new file mode 100644
index 0000000000..363e392bb2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsf.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  conversion from lpc coefficients to lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsf(int16_t* lsf, /* (o) lsf coefficients (Q13) */
+                            int16_t* a    /* (i) A coefficients (Q12) */
+                            );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
new file mode 100644
index 0000000000..ad0ecd70ab
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
@@ -0,0 +1,159 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsp.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsp.h"
+
+#include "modules/audio_coding/codecs/ilbc/chebyshev.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+/*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsp coefficients
+ * function is only for 10:th order LPC
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsp(
+    int16_t *a,  /* (o) A coefficients in Q12 */
+    int16_t *lsp, /* (i) LSP coefficients in Q15 */
+    int16_t *old_lsp /* (i) old LSP coefficients that are used if the new
+                              coefficients turn out to be unstable */
+                            ) {
+  int16_t f[2][6]; /* f[0][] represents f1 and f[1][] represents f2 */
+  int16_t *a_i_ptr, *a_10mi_ptr;
+  int16_t *f1ptr, *f2ptr;
+  int32_t tmpW32;
+  int16_t x, y, xlow, ylow, xmid, ymid, xhigh, yhigh, xint;
+  int16_t shifts, sign;
+  int i, j;
+  int foundFreqs;
+  int fi_select;
+
+  /*
+     Calculate the two polynomials f1(z) and f2(z)
+     (the sum and the diff polynomial)
+     f1[0] = f2[0] = 1.0;
+     f1[i+1] = a[i+1] + a[10-i] - f1[i];
+     f2[i+1] = a[i+1] - a[10-i] - f1[i];
+  */
+
+  a_i_ptr = a + 1;
+  a_10mi_ptr = a + 10;
+  f1ptr = f[0];
+  f2ptr = f[1];
+  (*f1ptr) = 1024; /* 1.0 in Q10 */
+  (*f2ptr) = 1024; /* 1.0 in Q10 */
+  for (i = 0; i < 5; i++) {
+    *(f1ptr + 1) =
+        (int16_t)((((int32_t)(*a_i_ptr) + *a_10mi_ptr) >> 2) - *f1ptr);
+    *(f2ptr + 1) =
+        (int16_t)((((int32_t)(*a_i_ptr) - *a_10mi_ptr) >> 2) + *f2ptr);
+    a_i_ptr++;
+    a_10mi_ptr--;
+    f1ptr++;
+    f2ptr++;
+  }
+
+  /*
+    find the LSPs using the Chebychev pol. evaluation
+  */
+
+  fi_select = 0; /* selector between f1 and f2, start with f1 */
+
+  foundFreqs = 0;
+
+  xlow = WebRtcIlbcfix_kCosGrid[0];
+  ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+
+  /*
+     Iterate until all the 10 LSP's have been found or
+     all the grid points have been tried. If the 10 LSP's can
+     not be found, set the LSP vector to previous LSP
+  */
+
+  for (j = 1; j < COS_GRID_POINTS && foundFreqs < 10; j++) {
+    xhigh = xlow;
+    yhigh = ylow;
+    xlow = WebRtcIlbcfix_kCosGrid[j];
+    ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+
+    if (ylow * yhigh <= 0) {
+      /* Run 4 times to reduce the interval */
+      for (i = 0; i < 4; i++) {
+        /* xmid =(xlow + xhigh)/2 */
+        xmid = (xlow >> 1) + (xhigh >> 1);
+        ymid = WebRtcIlbcfix_Chebyshev(xmid, f[fi_select]);
+
+        if (ylow * ymid <= 0) {
+          yhigh = ymid;
+          xhigh = xmid;
+        } else {
+          ylow = ymid;
+          xlow = xmid;
+        }
+      }
+
+      /*
+        Calculater xint by linear interpolation:
+        xint = xlow - ylow*(xhigh-xlow)/(yhigh-ylow);
+      */
+
+      x = xhigh - xlow;
+      y = yhigh - ylow;
+
+      if (y == 0) {
+        xint = xlow;
+      } else {
+        sign = y;
+        y = WEBRTC_SPL_ABS_W16(y);
+        shifts = (int16_t)WebRtcSpl_NormW32(y)-16;
+        y <<= shifts;
+        y = (int16_t)WebRtcSpl_DivW32W16(536838144, y); /* 1/(yhigh-ylow) */
+
+        tmpW32 = (x * y) >> (19 - shifts);
+
+        /* y=(xhigh-xlow)/(yhigh-ylow) */
+        y = (int16_t)(tmpW32&0xFFFF);
+
+        if (sign < 0) {
+          y = -y;
+        }
+        /* tmpW32 = ylow*(xhigh-xlow)/(yhigh-ylow) */
+        tmpW32 = (ylow * y) >> 10;
+        xint = xlow-(int16_t)(tmpW32&0xFFFF);
+      }
+
+      /* Store the calculated lsp */
+      lsp[foundFreqs] = (int16_t)xint;
+      foundFreqs++;
+
+      /* if needed, set xlow and ylow for next recursion */
+      if (foundFreqs<10) {
+        xlow = xint;
+        /* Swap between f1 and f2 (f[0][] and f[1][]) */
+        fi_select = ((fi_select+1)&0x1);
+
+        ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+      }
+    }
+  }
+
+  /* Check if M roots found, if not then use the old LSP */
+  if (foundFreqs < 10) {
+    WEBRTC_SPL_MEMCPY_W16(lsp, old_lsp, 10);
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
new file mode 100644
index 0000000000..928ee4efdb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsp.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsp coefficients
+ * function is only for 10:th order LPC
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsp(
+    int16_t* a,      /* (o) A coefficients in Q12 */
+    int16_t* lsp,    /* (i) LSP coefficients in Q15 */
+    int16_t* old_lsp /* (i) old LSP coefficients that are used if the new
+                              coefficients turn out to be unstable */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c
new file mode 100644
index 0000000000..5bdab7a4b0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Refiner.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/refiner.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/enh_upsample.h"
+#include "modules/audio_coding/codecs/ilbc/my_corr.h"
+
+/*----------------------------------------------------------------*
+ * find segment starting near idata+estSegPos that has highest
+ * correlation with idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
+ * resolution of ENH_UPSO times the original of the original
+ * sampling rate
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Refiner(
+    size_t *updStartPos, /* (o) updated start point (Q-2) */
+    int16_t *idata,   /* (i) original data buffer */
+    size_t idatal,   /* (i) dimension of idata */
+    size_t centerStartPos, /* (i) beginning center segment */
+    size_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
+    int16_t *surround,  /* (i/o) The contribution from this sequence
+                                           summed with earlier contributions */
+    int16_t gain    /* (i) Gain to use for this sequence */
+                           ){
+  size_t estSegPosRounded, searchSegStartPos, searchSegEndPos, corrdim;
+  size_t tloc, tloc2, i;
+
+  int32_t maxtemp, scalefact;
+  int16_t *filtStatePtr, *polyPtr;
+  /* Stack based */
+  int16_t filt[7];
+  int32_t corrVecUps[ENH_CORRDIM*ENH_UPS0];
+  int32_t corrVecTemp[ENH_CORRDIM];
+  int16_t vect[ENH_VECTL];
+  int16_t corrVec[ENH_CORRDIM];
+
+  /* defining array bounds */
+
+  estSegPosRounded = (estSegPos - 2) >> 2;
+
+  searchSegStartPos =
+      (estSegPosRounded < ENH_SLOP) ? 0 : (estSegPosRounded - ENH_SLOP);
+
+  searchSegEndPos = estSegPosRounded + ENH_SLOP;
+  if ((searchSegEndPos + ENH_BLOCKL) >= idatal) {
+    searchSegEndPos = idatal - ENH_BLOCKL - 1;
+  }
+
+  corrdim = searchSegEndPos + 1 - searchSegStartPos;
+
+  /* compute upsampled correlation and find
+     location of max */
+
+  WebRtcIlbcfix_MyCorr(corrVecTemp, idata + searchSegStartPos,
+                       corrdim + ENH_BLOCKL - 1, idata + centerStartPos,
+                       ENH_BLOCKL);
+
+  /* Calculate the rescaling factor for the correlation in order to
+     put the correlation in a int16_t vector instead */
+  maxtemp = WebRtcSpl_MaxAbsValueW32(corrVecTemp, corrdim);
+
+  scalefact = WebRtcSpl_GetSizeInBits(maxtemp) - 15;
+
+  if (scalefact > 0) {
+    for (i = 0; i < corrdim; i++) {
+      corrVec[i] = (int16_t)(corrVecTemp[i] >> scalefact);
+    }
+  } else {
+    for (i = 0; i < corrdim; i++) {
+      corrVec[i] = (int16_t)corrVecTemp[i];
+    }
+  }
+  /* In order to guarantee that all values are initialized */
+  for (i = corrdim; i < ENH_CORRDIM; i++) {
+    corrVec[i] = 0;
+  }
+
+  /* Upsample the correlation */
+  WebRtcIlbcfix_EnhUpsample(corrVecUps, corrVec);
+
+  /* Find maximum */
+  tloc = WebRtcSpl_MaxIndexW32(corrVecUps, ENH_UPS0 * corrdim);
+
+  /* make vector can be upsampled without ever running outside
+     bounds */
+  *updStartPos = searchSegStartPos * 4 + tloc + 4;
+
+  tloc2 = (tloc + 3) >> 2;
+
+  /* initialize the vector to be filtered, stuff with zeros
+     when data is outside idata buffer */
+  if (ENH_FL0 > (searchSegStartPos + tloc2)) {
+    const size_t st = ENH_FL0 - searchSegStartPos - tloc2;
+    WebRtcSpl_MemSetW16(vect, 0, st);
+    WEBRTC_SPL_MEMCPY_W16(&vect[st], idata, ENH_VECTL - st);
+  } else {
+    const size_t st = searchSegStartPos + tloc2 - ENH_FL0;
+    if ((st + ENH_VECTL) > idatal) {
+      const size_t en = st + ENH_VECTL - idatal;
+      WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL - en);
+      WebRtcSpl_MemSetW16(&vect[ENH_VECTL - en], 0, en);
+    } else {
+      WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL);
+    }
+  }
+
+  /* compute the segment (this is actually a convolution) */
+  filtStatePtr = filt + 6;
+  polyPtr = (int16_t*)WebRtcIlbcfix_kEnhPolyPhaser[tloc2 * ENH_UPS0 - tloc];
+  for (i = 0; i < 7; i++) {
+    *filtStatePtr-- = *polyPtr++;
+  }
+
+  WebRtcSpl_FilterMAFastQ12(&vect[6], vect, filt, ENH_FLO_MULT2_PLUS1,
+                            ENH_BLOCKL);
+
+  /* Add the contribution from this vector (scaled with gain) to the total
+     surround vector */
+  WebRtcSpl_AddAffineVectorToVector(surround, vect, gain, 32768, 16,
+                                    ENH_BLOCKL);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.h
new file mode 100644
index 0000000000..564c9d96e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Refiner.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * find segment starting near idata+estSegPos that has highest
+ * correlation with idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
+ * resolution of ENH_UPSO times the original of the original
+ * sampling rate
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Refiner(
+    size_t* updStartPos,   /* (o) updated start point (Q-2) */
+    int16_t* idata,        /* (i) original data buffer */
+    size_t idatal,         /* (i) dimension of idata */
+    size_t centerStartPos, /* (i) beginning center segment */
+    size_t estSegPos,      /* (i) estimated beginning other segment (Q-2) */
+    int16_t* surround,     /* (i/o) The contribution from this sequence
+                                    summed with earlier contributions */
+    int16_t gain           /* (i) Gain to use for this sequence */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
new file mode 100644
index 0000000000..7343530a5e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleInterpolateLsf.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h"
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleInterpolateLsf(
+    int16_t *syntdenum, /* (o) the synthesis filter denominator
+                                   resulting from the quantized
+                                   interpolated lsf Q12 */
+    int16_t *weightdenum, /* (o) the weighting filter denominator
+                                   resulting from the unquantized
+                                   interpolated lsf Q12 */
+    int16_t *lsf,  /* (i) the unquantized lsf coefficients Q13 */
+    int16_t *lsfdeq,  /* (i) the dequantized lsf coefficients Q13 */
+    int16_t *lsfold,  /* (i) the unquantized lsf coefficients of
+                                           the previous signal frame Q13 */
+    int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the
+                                   previous signal frame Q13 */
+    int16_t length,  /* (i) should equate FILTERORDER */
+    IlbcEncoder *iLBCenc_inst
+    /* (i/o) the encoder state structure */
+                                        ) {
+  size_t i;
+  int pos, lp_length;
+
+  int16_t *lsf2, *lsfdeq2;
+  /* Stack based */
+  int16_t lp[LPC_FILTERORDER + 1];
+
+  lsf2 = lsf + length;
+  lsfdeq2 = lsfdeq + length;
+  lp_length = length + 1;
+
+  if (iLBCenc_inst->mode==30) {
+    /* subframe 1: Interpolation between old and first set of
+       lsf coefficients */
+
+    /* Calculate Analysis/Syntehsis filter from quantized LSF */
+    WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
+                                         WebRtcIlbcfix_kLsfWeight30ms[0],
+                                         length);
+    WEBRTC_SPL_MEMCPY_W16(syntdenum, lp, lp_length);
+
+    /* Calculate Weighting filter from quantized LSF */
+    WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
+                                         WebRtcIlbcfix_kLsfWeight30ms[0],
+                                         length);
+    WebRtcIlbcfix_BwExpand(weightdenum, lp,
+                           (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                           (int16_t)lp_length);
+
+    /* subframe 2 to 6: Interpolation between first and second
+       set of lsf coefficients */
+
+    pos = lp_length;
+    for (i = 1; i < iLBCenc_inst->nsub; i++) {
+
+      /* Calculate Analysis/Syntehsis filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeq, lsfdeq2,
+                                           WebRtcIlbcfix_kLsfWeight30ms[i],
+                                           length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
+
+      /* Calculate Weighting filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsf, lsf2,
+                                           WebRtcIlbcfix_kLsfWeight30ms[i],
+                                           length);
+      WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                             (int16_t)lp_length);
+
+      pos += lp_length;
+    }
+
+    /* update memory */
+
+    WEBRTC_SPL_MEMCPY_W16(lsfold, lsf2, length);
+    WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq2, length);
+
+  } else { /* iLBCenc_inst->mode==20 */
+    pos = 0;
+    for (i = 0; i < iLBCenc_inst->nsub; i++) {
+
+      /* Calculate Analysis/Syntehsis filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
+                                           WebRtcIlbcfix_kLsfWeight20ms[i],
+                                           length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
+
+      /* Calculate Weighting filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
+                                           WebRtcIlbcfix_kLsfWeight20ms[i],
+                                           length);
+      WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                             (int16_t)lp_length);
+
+      pos += lp_length;
+    }
+
+    /* update memory */
+
+    WEBRTC_SPL_MEMCPY_W16(lsfold, lsf, length);
+    WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq, length);
+
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
new file mode 100644
index 0000000000..ee53e4bd08
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleInterpolateLsf.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleInterpolateLsf(
+    int16_t* syntdenum,   /* (o) the synthesis filter denominator
+                                     resulting from the quantized
+                                     interpolated lsf Q12 */
+    int16_t* weightdenum, /* (o) the weighting filter denominator
+                                   resulting from the unquantized
+                                   interpolated lsf Q12 */
+    int16_t* lsf,         /* (i) the unquantized lsf coefficients Q13 */
+    int16_t* lsfdeq,      /* (i) the dequantized lsf coefficients Q13 */
+    int16_t* lsfold,      /* (i) the unquantized lsf coefficients of
+                                               the previous signal frame Q13 */
+    int16_t* lsfdeqold,   /* (i) the dequantized lsf coefficients of the
+                                     previous signal frame Q13 */
+    int16_t length,       /* (i) should equate FILTERORDER */
+    IlbcEncoder* iLBCenc_inst
+    /* (i/o) the encoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
new file mode 100644
index 0000000000..fdc4553d95
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLpcAnalysis.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/window32_w32.h"
+
+/*----------------------------------------------------------------*
+ *  lpc analysis (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLpcAnalysis(
+    int16_t *lsf,   /* (o) lsf coefficients */
+    int16_t *data,   /* (i) new block of speech */
+    IlbcEncoder *iLBCenc_inst
+    /* (i/o) the encoder state structure */
+                                     ) {
+  int k;
+  int scale;
+  size_t is;
+  int16_t stability;
+  /* Stack based */
+  int16_t A[LPC_FILTERORDER + 1];
+  int32_t R[LPC_FILTERORDER + 1];
+  int16_t windowedData[BLOCKL_MAX];
+  int16_t rc[LPC_FILTERORDER];
+
+  is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer+is,data,iLBCenc_inst->blockl);
+
+  /* No lookahead, last window is asymmetric */
+
+  for (k = 0; k < iLBCenc_inst->lpc_n; k++) {
+
+    is = LPC_LOOKBACK;
+
+    if (k < (iLBCenc_inst->lpc_n - 1)) {
+
+      /* Hanning table WebRtcIlbcfix_kLpcWin[] is in Q15-domain so the output is right-shifted 15 */
+      WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer, WebRtcIlbcfix_kLpcWin, BLOCKL_MAX, 15);
+    } else {
+
+      /* Hanning table WebRtcIlbcfix_kLpcAsymWin[] is in Q15-domain so the output is right-shifted 15 */
+      WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer+is, WebRtcIlbcfix_kLpcAsymWin, BLOCKL_MAX, 15);
+    }
+
+    /* Compute autocorrelation */
+    WebRtcSpl_AutoCorrelation(windowedData, BLOCKL_MAX, LPC_FILTERORDER, R, &scale);
+
+    /* Window autocorrelation vector */
+    WebRtcIlbcfix_Window32W32(R, R, WebRtcIlbcfix_kLpcLagWin, LPC_FILTERORDER + 1 );
+
+    /* Calculate the A coefficients from the Autocorrelation using Levinson Durbin algorithm */
+    stability=WebRtcSpl_LevinsonDurbin(R, A, rc, LPC_FILTERORDER);
+
+    /*
+       Set the filter to {1.0, 0.0, 0.0,...} if filter from Levinson Durbin algorithm is unstable
+       This should basically never happen...
+    */
+    if (stability!=1) {
+      A[0]=4096;
+      WebRtcSpl_MemSetW16(&A[1], 0, LPC_FILTERORDER);
+    }
+
+    /* Bandwidth expand the filter coefficients */
+    WebRtcIlbcfix_BwExpand(A, A, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, LPC_FILTERORDER+1);
+
+    /* Convert from A to LSF representation */
+    WebRtcIlbcfix_Poly2Lsf(lsf + k*LPC_FILTERORDER, A);
+  }
+
+  is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer,
+                        iLBCenc_inst->lpc_buffer+LPC_LOOKBACK+BLOCKL_MAX-is, is);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
new file mode 100644
index 0000000000..b5c839ba2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLpcAnalysis.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc analysis (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLpcAnalysis(
+    int16_t* lsf,  /* (o) lsf coefficients */
+    int16_t* data, /* (i) new block of speech */
+    IlbcEncoder* iLBCenc_inst
+    /* (i/o) the encoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
new file mode 100644
index 0000000000..e7494ceb59
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfDeQ.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  obtain dequantized lsf coefficients from quantization index
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfDeQ(
+    int16_t *lsfdeq,  /* (o) dequantized lsf coefficients */
+    int16_t *index,  /* (i) quantization index */
+    int16_t lpc_n  /* (i) number of LPCs */
+                                ){
+  int i, j, pos, cb_pos;
+
+  /* decode first LSF */
+
+  pos = 0;
+  cb_pos = 0;
+  for (i = 0; i < LSF_NSPLIT; i++) {
+    for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
+      lsfdeq[pos + j] = WebRtcIlbcfix_kLsfCb[cb_pos + j + index[i] *
+                                             WebRtcIlbcfix_kLsfDimCb[i]];
+    }
+    pos += WebRtcIlbcfix_kLsfDimCb[i];
+    cb_pos += WebRtcIlbcfix_kLsfSizeCb[i] * WebRtcIlbcfix_kLsfDimCb[i];
+  }
+
+  if (lpc_n>1) {
+    /* decode last LSF */
+    pos = 0;
+    cb_pos = 0;
+    for (i = 0; i < LSF_NSPLIT; i++) {
+      for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
+        lsfdeq[LPC_FILTERORDER + pos + j] = WebRtcIlbcfix_kLsfCb[
+            cb_pos + index[LSF_NSPLIT + i] * WebRtcIlbcfix_kLsfDimCb[i] + j];
+      }
+      pos += WebRtcIlbcfix_kLsfDimCb[i];
+      cb_pos += WebRtcIlbcfix_kLsfSizeCb[i] * WebRtcIlbcfix_kLsfDimCb[i];
+    }
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
new file mode 100644
index 0000000000..6d97d3df33
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfDeQ.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  obtain dequantized lsf coefficients from quantization index
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfDeQ(
+    int16_t* lsfdeq, /* (o) dequantized lsf coefficients */
+    int16_t* index,  /* (i) quantization index */
+    int16_t lpc_n    /* (i) number of LPCs */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
new file mode 100644
index 0000000000..1291d1442e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfQ.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_quant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/split_vq.h"
+
+/*----------------------------------------------------------------*
+ *  lsf quantizer (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfQ(
+    int16_t *lsfdeq, /* (o) dequantized lsf coefficients
+                                   (dimension FILTERORDER) Q13 */
+    int16_t *index, /* (o) quantization index */
+    int16_t *lsf, /* (i) the lsf coefficient vector to be
+                           quantized (dimension FILTERORDER) Q13 */
+    int16_t lpc_n /* (i) number of lsf sets to quantize */
+                              ){
+
+  /* Quantize first LSF with memoryless split VQ */
+  WebRtcIlbcfix_SplitVq( lsfdeq, index, lsf,
+                         (int16_t*)WebRtcIlbcfix_kLsfCb, (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb);
+
+  if (lpc_n==2) {
+    /* Quantize second LSF with memoryless split VQ */
+    WebRtcIlbcfix_SplitVq( lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT,
+                           lsf + LPC_FILTERORDER, (int16_t*)WebRtcIlbcfix_kLsfCb,
+                           (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb);
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
new file mode 100644
index 0000000000..66b553213a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfQ.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  lsf quantizer (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfQ(
+    int16_t* lsfdeq, /* (o) dequantized lsf coefficients
+                                   (dimension FILTERORDER) Q13 */
+    int16_t* index,  /* (o) quantization index */
+    int16_t* lsf,    /* (i) the lsf coefficient vector to be
+                              quantized (dimension FILTERORDER) Q13 */
+    int16_t lpc_n    /* (i) number of lsf sets to quantize */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c
new file mode 100644
index 0000000000..631b2f432a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c
@@ -0,0 +1,212 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/smooth.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/smooth_out_data.h"
+
+/*----------------------------------------------------------------*
+ * find the smoothed output data
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Smooth(
+    int16_t *odata,   /* (o) smoothed output */
+    int16_t *current,  /* (i) the un enhanced residual for
+                                this block */
+    int16_t *surround  /* (i) The approximation from the
+                                surrounding sequences */
+                          ) {
+  int16_t scale, scale1, scale2;
+  int16_t A, B, C, denomW16;
+  int32_t B_W32, denom, num;
+  int32_t errs;
+  int32_t w00,w10,w11, endiff, crit;
+  int32_t w00prim, w10prim, w11_div_w00;
+  int16_t w11prim;
+  int16_t bitsw00, bitsw10, bitsw11;
+  int32_t w11w00, w10w10, w00w00;
+  uint32_t max1, max2, max12;
+
+  /* compute some inner products (ensure no overflow by first calculating proper scale factor) */
+
+  w00 = w10 = w11 = 0;
+
+  // Calculate a right shift that will let us sum ENH_BLOCKL pairwise products
+  // of values from the two sequences without overflowing an int32_t. (The +1
+  // in max1 and max2 are because WebRtcSpl_MaxAbsValueW16 will return 2**15 -
+  // 1 if the input array contains -2**15.)
+  max1 = WebRtcSpl_MaxAbsValueW16(current, ENH_BLOCKL) + 1;
+  max2 = WebRtcSpl_MaxAbsValueW16(surround, ENH_BLOCKL) + 1;
+  max12 = WEBRTC_SPL_MAX(max1, max2);
+  scale = (64 - 31) -
+          WebRtcSpl_CountLeadingZeros64((max12 * max12) * (uint64_t)ENH_BLOCKL);
+  scale=WEBRTC_SPL_MAX(0, scale);
+
+  w00=WebRtcSpl_DotProductWithScale(current,current,ENH_BLOCKL,scale);
+  w11=WebRtcSpl_DotProductWithScale(surround,surround,ENH_BLOCKL,scale);
+  w10=WebRtcSpl_DotProductWithScale(surround,current,ENH_BLOCKL,scale);
+
+  if (w00<0) w00 = WEBRTC_SPL_WORD32_MAX;
+  if (w11<0) w11 = WEBRTC_SPL_WORD32_MAX;
+
+  /* Rescale w00 and w11 to w00prim and w11prim, so that w00prim/w11prim
+     is in Q16 */
+
+  bitsw00 = WebRtcSpl_GetSizeInBits(w00);
+  bitsw11 = WebRtcSpl_GetSizeInBits(w11);
+  bitsw10 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(w10));
+  scale1 = 31 - bitsw00;
+  scale2 = 15 - bitsw11;
+
+  if (scale2>(scale1-16)) {
+    scale2 = scale1 - 16;
+  } else {
+    scale1 = scale2 + 16;
+  }
+
+  w00prim = w00 << scale1;
+  w11prim = (int16_t) WEBRTC_SPL_SHIFT_W32(w11, scale2);
+
+  /* Perform C = sqrt(w11/w00) (C is in Q11 since (16+6)/2=11) */
+  if (w11prim>64) {
+    endiff = WebRtcSpl_DivW32W16(w00prim, w11prim) << 6;
+    C = (int16_t)WebRtcSpl_SqrtFloor(endiff); /* C is in Q11 */
+  } else {
+    C = 1;
+  }
+
+  /* first try enhancement without power-constraint */
+
+  errs = WebRtcIlbcfix_Smooth_odata(odata, current, surround, C);
+
+
+
+  /* if constraint violated by first try, add constraint */
+
+  if ( (6-scale+scale1) > 31) {
+    crit=0;
+  } else {
+    /* crit = 0.05 * w00 (Result in Q-6) */
+    crit = WEBRTC_SPL_SHIFT_W32(
+        WEBRTC_SPL_MUL(ENH_A0, w00prim >> 14),
+        -(6-scale+scale1));
+  }
+
+  if (errs > crit) {
+
+    if( w00 < 1) {
+      w00=1;
+    }
+
+    /* Calculate w11*w00, w10*w10 and w00*w00 in the same Q domain */
+
+    scale1 = bitsw00-15;
+    scale2 = bitsw11-15;
+
+    if (scale2>scale1) {
+      scale = scale2;
+    } else {
+      scale = scale1;
+    }
+
+    w11w00 = (int16_t)WEBRTC_SPL_SHIFT_W32(w11, -scale) *
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale);
+
+    w10w10 = (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale) *
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale);
+
+    w00w00 = (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale) *
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale);
+
+    /* Calculate (w11*w00-w10*w10)/(w00*w00) in Q16 */
+    if (w00w00>65536) {
+      endiff = (w11w00-w10w10);
+      endiff = WEBRTC_SPL_MAX(0, endiff);
+      /* denom is in Q16 */
+      denom = WebRtcSpl_DivW32W16(endiff, (int16_t)(w00w00 >> 16));
+    } else {
+      denom = 65536;
+    }
+
+    if( denom > 7){ /* eliminates numerical problems
+                       for if smooth */
+
+      scale=WebRtcSpl_GetSizeInBits(denom)-15;
+
+      if (scale>0) {
+        /* denomW16 is in Q(16+scale) */
+        denomW16 = (int16_t)(denom >> scale);
+
+        /* num in Q(34-scale) */
+        num = ENH_A0_MINUS_A0A0DIV4 >> scale;
+      } else {
+        /* denomW16 is in Q16 */
+        denomW16=(int16_t)denom;
+
+        /* num in Q34 */
+        num=ENH_A0_MINUS_A0A0DIV4;
+      }
+
+      /* A sqrt( (ENH_A0-(ENH_A0^2)/4)*(w00*w00)/(w11*w00 + w10*w10) ) in Q9 */
+      A = (int16_t)WebRtcSpl_SqrtFloor(WebRtcSpl_DivW32W16(num, denomW16));
+
+      /* B_W32 is in Q30 ( B = 1 - ENH_A0/2 - A * w10/w00 ) */
+      scale1 = 31-bitsw10;
+      scale2 = 21-scale1;
+      w10prim = w10 == 0 ? 0 : w10 * (1 << scale1);
+      w00prim = WEBRTC_SPL_SHIFT_W32(w00, -scale2);
+      scale = bitsw00-scale2-15;
+
+      if (scale>0) {
+        w10prim >>= scale;
+        w00prim >>= scale;
+      }
+
+      if ((w00prim>0)&&(w10prim>0)) {
+        w11_div_w00=WebRtcSpl_DivW32W16(w10prim, (int16_t)w00prim);
+
+        if (WebRtcSpl_GetSizeInBits(w11_div_w00)+WebRtcSpl_GetSizeInBits(A)>31) {
+          B_W32 = 0;
+        } else {
+          B_W32 = (int32_t)1073741824 - (int32_t)ENH_A0DIV2 -
+              WEBRTC_SPL_MUL(A, w11_div_w00);
+        }
+        B = (int16_t)(B_W32 >> 16);  /* B in Q14. */
+      } else {
+        /* No smoothing */
+        A = 0;
+        B = 16384; /* 1 in Q14 */
+      }
+    }
+    else{ /* essentially no difference between cycles;
+             smoothing not needed */
+
+      A = 0;
+      B = 16384; /* 1 in Q14 */
+    }
+
+    /* create smoothed sequence */
+
+    WebRtcSpl_ScaleAndAddVectors(surround, A, 9,
+                                current, B, 14,
+                                odata, ENH_BLOCKL);
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.h
new file mode 100644
index 0000000000..c8752be64f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * find the smoothed output data
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Smooth(int16_t* odata,   /* (o) smoothed output */
+                          int16_t* current, /* (i) the un enhanced residual for
+                                                     this block */
+                          int16_t* surround /* (i) The approximation from the
+                                                     surrounding sequences */
+                          );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
new file mode 100644
index 0000000000..9f952bfb93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth_odata.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/smooth_out_data.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "rtc_base/sanitizer.h"
+
+// An s32 + s32 -> s32 addition that's allowed to overflow. (It's still
+// undefined behavior, so not a good idea; this just makes UBSan ignore the
+// violation, so that our old code can continue to do what it's always been
+// doing.)
+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
+    OverflowingAdd_S32_S32_To_S32(int32_t a, int32_t b) {
+  return a + b;
+}
+
+int32_t WebRtcIlbcfix_Smooth_odata(
+    int16_t *odata,
+    int16_t *psseq,
+    int16_t *surround,
+    int16_t C)
+{
+  int i;
+
+  int16_t err;
+  int32_t errs;
+
+  for(i=0;i<80;i++) {
+    odata[i]= (int16_t)((C * surround[i] + 1024) >> 11);
+  }
+
+  errs=0;
+  for(i=0;i<80;i++) {
+    err = (psseq[i] - odata[i]) >> 3;
+    errs = OverflowingAdd_S32_S32_To_S32(errs, err * err);  // errs in Q-6
+  }
+
+  return errs;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h
new file mode 100644
index 0000000000..318e7b04a2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth_odata.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * help function to WebRtcIlbcfix_Smooth()
+ *---------------------------------------------------------------*/
+
+int32_t WebRtcIlbcfix_Smooth_odata(int16_t* odata,
+                                   int16_t* psseq,
+                                   int16_t* surround,
+                                   int16_t C);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c
new file mode 100644
index 0000000000..c3a24750f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SortSq.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/sort_sq.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  scalar quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SortSq(
+    int16_t *xq,   /* (o) the quantized value */
+    int16_t *index,  /* (o) the quantization index */
+    int16_t x,   /* (i) the value to quantize */
+    const int16_t *cb, /* (i) the quantization codebook */
+    int16_t cb_size  /* (i) the size of the quantization codebook */
+                          ){
+  int i;
+
+  if (x <= cb[0]) {
+    *index = 0;
+    *xq = cb[0];
+  } else {
+    i = 0;
+    while ((x > cb[i]) && (i < (cb_size-1))) {
+      i++;
+    }
+
+    if (x > (((int32_t)cb[i] + cb[i - 1] + 1) >> 1)) {
+      *index = i;
+      *xq = cb[i];
+    } else {
+      *index = i - 1;
+      *xq = cb[i - 1];
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.h
new file mode 100644
index 0000000000..02028dae93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SortSq.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  scalar quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SortSq(
+    int16_t* xq,       /* (o) the quantized value */
+    int16_t* index,    /* (o) the quantization index */
+    int16_t x,         /* (i) the value to quantize */
+    const int16_t* cb, /* (i) the quantization codebook */
+    int16_t cb_size    /* (i) the size of the quantization codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c
new file mode 100644
index 0000000000..c1f04d2287
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SplitVq.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/split_vq.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/vq3.h"
+#include "modules/audio_coding/codecs/ilbc/vq4.h"
+
+/*----------------------------------------------------------------*
+ *  split vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SplitVq(
+    int16_t *qX,  /* (o) the quantized vector in Q13 */
+    int16_t *index, /* (o) a vector of indexes for all vector
+                                   codebooks in the split */
+    int16_t *X,  /* (i) the vector to quantize */
+    int16_t *CB,  /* (i) the quantizer codebook in Q13 */
+    int16_t *dim, /* (i) the dimension of X and qX */
+    int16_t *cbsize /* (i) the number of vectors in the codebook */
+                           ) {
+
+  int16_t *qXPtr, *indexPtr, *CBPtr, *XPtr;
+
+  /* Quantize X with the 3 vectror quantization tables */
+
+  qXPtr=qX;
+  indexPtr=index;
+  CBPtr=CB;
+  XPtr=X;
+  WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[0]);
+
+  qXPtr+=3;
+  indexPtr+=1;
+  CBPtr+=(dim[0]*cbsize[0]);
+  XPtr+=3;
+  WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[1]);
+
+  qXPtr+=3;
+  indexPtr+=1;
+  CBPtr+=(dim[1]*cbsize[1]);
+  XPtr+=3;
+  WebRtcIlbcfix_Vq4(qXPtr, indexPtr, CBPtr, XPtr, cbsize[2]);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.h
new file mode 100644
index 0000000000..e4b02a2bc2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SplitVq.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  split vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SplitVq(
+    int16_t* qX,    /* (o) the quantized vector in Q13 */
+    int16_t* index, /* (o) a vector of indexes for all vector
+                                   codebooks in the split */
+    int16_t* X,     /* (i) the vector to quantize */
+    int16_t* CB,    /* (i) the quantizer codebook in Q13 */
+    int16_t* dim,   /* (i) the dimension of X and qX */
+    int16_t* cbsize /* (i) the number of vectors in the codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c
new file mode 100644
index 0000000000..c58086c03b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateConstruct.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  decoding of the start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateConstruct(
+    size_t idxForMax,   /* (i) 6-bit index for the quantization of
+                                           max amplitude */
+    int16_t *idxVec,   /* (i) vector of quantization indexes */
+    int16_t *syntDenum,  /* (i) synthesis filter denumerator */
+    int16_t *Out_fix,  /* (o) the decoded state vector */
+    size_t len    /* (i) length of a state vector */
+                                  ) {
+  size_t k;
+  int16_t maxVal;
+  int16_t *tmp1, *tmp2, *tmp3;
+  /* Stack based */
+  int16_t numerator[1+LPC_FILTERORDER];
+  int16_t sampleValVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t sampleMaVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t *sampleVal = &sampleValVec[LPC_FILTERORDER];
+  int16_t *sampleMa = &sampleMaVec[LPC_FILTERORDER];
+  int16_t *sampleAr = &sampleValVec[LPC_FILTERORDER];
+
+  /* initialization of coefficients */
+
+  for (k=0; k<LPC_FILTERORDER+1; k++){
+    numerator[k] = syntDenum[LPC_FILTERORDER-k];
+  }
+
+  /* decoding of the maximum value */
+
+  maxVal = WebRtcIlbcfix_kFrgQuantMod[idxForMax];
+
+  /* decoding of the sample values */
+  tmp1 = sampleVal;
+  tmp2 = &idxVec[len-1];
+
+  if (idxForMax<37) {
+    for(k=0; k<len; k++){
+      /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 2097152 (= 0.5 << 22)
+        maxVal is in Q8 and result is in Q(-1) */
+      *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 2097152) >>
+          22);
+      tmp1++;
+      tmp2--;
+    }
+  } else if (idxForMax<59) {
+    for(k=0; k<len; k++){
+      /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 262144 (= 0.5 << 19)
+        maxVal is in Q5 and result is in Q(-1) */
+      *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 262144) >>
+          19);
+      tmp1++;
+      tmp2--;
+    }
+  } else {
+    for(k=0; k<len; k++){
+      /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 65536 (= 0.5 << 17)
+        maxVal is in Q3 and result is in Q(-1) */
+      *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 65536) >>
+          17);
+      tmp1++;
+      tmp2--;
+    }
+  }
+
+  /* Set the rest of the data to zero */
+  WebRtcSpl_MemSetW16(&sampleVal[len], 0, len);
+
+  /* circular convolution with all-pass filter */
+
+  /* Set the state to zero */
+  WebRtcSpl_MemSetW16(sampleValVec, 0, (LPC_FILTERORDER));
+
+  /* Run MA filter + AR filter */
+  WebRtcSpl_FilterMAFastQ12(
+      sampleVal, sampleMa,
+      numerator, LPC_FILTERORDER+1, len + LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER));
+  WebRtcSpl_FilterARFastQ12(
+      sampleMa, sampleAr,
+      syntDenum, LPC_FILTERORDER+1, 2 * len);
+
+  tmp1 = &sampleAr[len-1];
+  tmp2 = &sampleAr[2*len-1];
+  tmp3 = Out_fix;
+  for(k=0;k<len;k++){
+    (*tmp3) = (*tmp1) + (*tmp2);
+    tmp1--;
+    tmp2--;
+    tmp3++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.h
new file mode 100644
index 0000000000..4c3011937d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateConstruct.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Generate the start state from the quantized indexes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateConstruct(
+    size_t idxForMax,   /* (i) 6-bit index for the quantization of
+                                           max amplitude */
+    int16_t* idxVec,    /* (i) vector of quantization indexes */
+    int16_t* syntDenum, /* (i) synthesis filter denumerator */
+    int16_t* Out_fix,   /* (o) the decoded state vector */
+    size_t len          /* (i) length of a state vector */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c
new file mode 100644
index 0000000000..7227ac9d45
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateSearch.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/state_search.h"
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  encoding of start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateSearch(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
+                               and idxVec, input state_first) */
+    int16_t *residual,   /* (i) target residual vector */
+    int16_t *syntDenum,  /* (i) lpc synthesis filter */
+    int16_t *weightDenum  /* (i) weighting filter denuminator */
+                               ) {
+  size_t k, index;
+  int16_t maxVal;
+  int16_t scale, shift;
+  int32_t maxValsq;
+  int16_t scaleRes;
+  int16_t max;
+  int i;
+  /* Stack based */
+  int16_t numerator[1+LPC_FILTERORDER];
+  int16_t residualLongVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t sampleMa[2*STATE_SHORT_LEN_30MS];
+  int16_t *residualLong = &residualLongVec[LPC_FILTERORDER];
+  int16_t *sampleAr = residualLong;
+
+  /* Scale to maximum 12 bits to avoid saturation in circular convolution filter */
+  max = WebRtcSpl_MaxAbsValueW16(residual, iLBCenc_inst->state_short_len);
+  scaleRes = WebRtcSpl_GetSizeInBits(max)-12;
+  scaleRes = WEBRTC_SPL_MAX(0, scaleRes);
+  /* Set up the filter coefficients for the circular convolution */
+  for (i=0; i<LPC_FILTERORDER+1; i++) {
+    numerator[i] = (syntDenum[LPC_FILTERORDER-i]>>scaleRes);
+  }
+
+  /* Copy the residual to a temporary buffer that we can filter
+   * and set the remaining samples to zero.
+   */
+  WEBRTC_SPL_MEMCPY_W16(residualLong, residual, iLBCenc_inst->state_short_len);
+  WebRtcSpl_MemSetW16(residualLong + iLBCenc_inst->state_short_len, 0, iLBCenc_inst->state_short_len);
+
+  /* Run the Zero-Pole filter (Ciurcular convolution) */
+  WebRtcSpl_MemSetW16(residualLongVec, 0, LPC_FILTERORDER);
+  WebRtcSpl_FilterMAFastQ12(residualLong, sampleMa, numerator,
+                            LPC_FILTERORDER + 1,
+                            iLBCenc_inst->state_short_len + LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(&sampleMa[iLBCenc_inst->state_short_len + LPC_FILTERORDER], 0, iLBCenc_inst->state_short_len - LPC_FILTERORDER);
+
+  WebRtcSpl_FilterARFastQ12(
+      sampleMa, sampleAr,
+      syntDenum, LPC_FILTERORDER+1, 2 * iLBCenc_inst->state_short_len);
+
+  for(k=0;k<iLBCenc_inst->state_short_len;k++){
+    sampleAr[k] += sampleAr[k+iLBCenc_inst->state_short_len];
+  }
+
+  /* Find maximum absolute value in the vector */
+  maxVal=WebRtcSpl_MaxAbsValueW16(sampleAr, iLBCenc_inst->state_short_len);
+
+  /* Find the best index */
+
+  if ((((int32_t)maxVal)<<scaleRes)<23170) {
+    maxValsq=((int32_t)maxVal*maxVal)<<(2+2*scaleRes);
+  } else {
+    maxValsq=(int32_t)WEBRTC_SPL_WORD32_MAX;
+  }
+
+  index=0;
+  for (i=0;i<63;i++) {
+
+    if (maxValsq>=WebRtcIlbcfix_kChooseFrgQuant[i]) {
+      index=i+1;
+    } else {
+      i=63;
+    }
+  }
+  iLBC_encbits->idxForMax=index;
+
+  /* Rescale the vector before quantization */
+  scale=WebRtcIlbcfix_kScale[index];
+
+  if (index<27) { /* scale table is in Q16, fout[] is in Q(-1) and we want the result to be in Q11 */
+    shift=4;
+  } else { /* scale table is in Q21, fout[] is in Q(-1) and we want the result to be in Q11 */
+    shift=9;
+  }
+
+  /* Set up vectors for AbsQuant and rescale it with the scale factor */
+  WebRtcSpl_ScaleVectorWithSat(sampleAr, sampleAr, scale,
+                              iLBCenc_inst->state_short_len, (int16_t)(shift-scaleRes));
+
+  /* Quantize the values in fout[] */
+  WebRtcIlbcfix_AbsQuant(iLBCenc_inst, iLBC_encbits, sampleAr, weightDenum);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.h
new file mode 100644
index 0000000000..6469138a0e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateSearch.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  encoding of start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateSearch(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits* iLBC_encbits, /* (i/o) Encoded bits (output idxForMax
+                                and idxVec, input state_first) */
+    int16_t* residual,       /* (i) target residual vector */
+    int16_t* syntDenum,      /* (i) lpc synthesis filter */
+    int16_t* weightDenum     /* (i) weighting filter denuminator */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
new file mode 100644
index 0000000000..bbafc1a2ed
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SwapBytes.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
+
+/*----------------------------------------------------------------*
+ * Swap bytes (to simplify operations on Little Endian machines)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SwapBytes(
+    const uint16_t* input,   /* (i) the sequence to swap */
+    size_t wordLength,      /* (i) number or uint16_t to swap */
+    uint16_t* output         /* (o) the swapped sequence */
+                              ) {
+  size_t k;
+  for (k = wordLength; k > 0; k--) {
+    *output++ = (*input >> 8)|(*input << 8);
+    input++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
new file mode 100644
index 0000000000..c59bf3068a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SwapBytes.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * Swap bytes (to simplify operations on Little Endian machines)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SwapBytes(
+    const uint16_t* input, /* (i) the sequence to swap */
+    size_t wordLength,     /* (i) number or uint16_t to swap */
+    uint16_t* output       /* (o) the swapped sequence */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/empty.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/empty.cc
new file mode 100644
index 0000000000..e69de29bb2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/empty.cc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
new file mode 100644
index 0000000000..e0ca075eda
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
@@ -0,0 +1,238 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+	iLBC Speech Coder ANSI-C Source Code
+
+        iLBC_test.c
+
+******************************************************************/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+/*---------------------------------------------------------------*
+ *  Main program to test iLBC encoding and decoding
+ *
+ *  Usage:
+ *	  exefile_name.exe <infile> <bytefile> <outfile> <channel>
+ *
+ *    <infile>   : Input file, speech for encoder (16-bit pcm file)
+ *    <bytefile> : Bit stream output from the encoder
+ *    <outfile>  : Output file, decoded speech (16-bit pcm file)
+ *    <channel>  : Bit error file, optional (16-bit)
+ *                     1 - Packet received correctly
+ *                     0 - Packet Lost
+ *
+ *--------------------------------------------------------------*/
+
+#define BLOCKL_MAX			240
+#define ILBCNOOFWORDS_MAX	25
+
+
+int main(int argc, char* argv[])
+{
+
+  FILE *ifileid,*efileid,*ofileid, *cfileid;
+  int16_t data[BLOCKL_MAX];
+  uint8_t encoded_data[2 * ILBCNOOFWORDS_MAX];
+  int16_t decoded_data[BLOCKL_MAX];
+  int len_int, mode;
+  short pli;
+  int blockcount = 0;
+  size_t frameLen, len, len_i16s;
+  int16_t speechType;
+  IlbcEncoderInstance *Enc_Inst;
+  IlbcDecoderInstance *Dec_Inst;
+
+#ifdef __ILBC_WITH_40BITACC
+  /* Doublecheck that long long exists */
+  if (sizeof(long)>=sizeof(long long)) {
+    fprintf(stderr, "40-bit simulation is not be supported on this platform\n");
+    exit(0);
+  }
+#endif
+
+  /* get arguments and open files */
+
+  if ((argc!=5) && (argc!=6)) {
+    fprintf(stderr,
+            "\n*-----------------------------------------------*\n");
+    fprintf(stderr,
+            "   %s <20,30> input encoded decoded (channel)\n\n",
+            argv[0]);
+    fprintf(stderr,
+            "   mode    : Frame size for the encoding/decoding\n");
+    fprintf(stderr,
+            "                 20 - 20 ms\n");
+    fprintf(stderr,
+            "                 30 - 30 ms\n");
+    fprintf(stderr,
+            "   input   : Speech for encoder (16-bit pcm file)\n");
+    fprintf(stderr,
+            "   encoded : Encoded bit stream\n");
+    fprintf(stderr,
+            "   decoded : Decoded speech (16-bit pcm file)\n");
+    fprintf(stderr,
+            "   channel : Packet loss pattern, optional (16-bit)\n");
+    fprintf(stderr,
+            "                  1 - Packet received correctly\n");
+    fprintf(stderr,
+            "                  0 - Packet Lost\n");
+    fprintf(stderr,
+            "*-----------------------------------------------*\n\n");
+    exit(1);
+  }
+  mode=atoi(argv[1]);
+  if (mode != 20 && mode != 30) {
+    fprintf(stderr,"Wrong mode %s, must be 20, or 30\n",
+            argv[1]);
+    exit(2);
+  }
+  if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+    exit(2);}
+  if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open encoded file file %s\n",
+            argv[3]); exit(1);}
+  if ( (ofileid=fopen(argv[4],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open decoded file %s\n",
+            argv[4]); exit(1);}
+  if (argc==6) {
+    if( (cfileid=fopen(argv[5],"rb")) == NULL) {
+      fprintf(stderr, "Cannot open channel file %s\n",
+              argv[5]);
+      exit(1);
+    }
+  } else {
+    cfileid=NULL;
+  }
+
+  /* print info */
+
+  fprintf(stderr, "\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*      iLBC test program                            *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,"\nMode           : %2d ms\n", mode);
+  fprintf(stderr,"Input file     : %s\n", argv[2]);
+  fprintf(stderr,"Encoded file   : %s\n", argv[3]);
+  fprintf(stderr,"Output file    : %s\n", argv[4]);
+  if (argc==6) {
+    fprintf(stderr,"Channel file   : %s\n", argv[5]);
+  }
+  fprintf(stderr,"\n");
+
+  /* Create structs */
+  WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
+  WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
+
+
+  /* Initialization */
+
+  WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
+  WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
+  frameLen = (size_t)(mode*8);
+
+  /* loop over input blocks */
+
+  while (fread(data,sizeof(int16_t),frameLen,ifileid) == frameLen) {
+
+    blockcount++;
+
+    /* encoding */
+
+    fprintf(stderr, "--- Encoding block %i --- ",blockcount);
+    len_int = WebRtcIlbcfix_Encode(Enc_Inst, data, frameLen, encoded_data);
+    if (len_int < 0) {
+      fprintf(stderr, "Error encoding\n");
+      exit(0);
+    }
+    len = (size_t)len_int;
+    fprintf(stderr, "\r");
+
+    /* write byte file */
+
+    len_i16s = (len + 1) / sizeof(int16_t);
+    if (fwrite(encoded_data, sizeof(int16_t), len_i16s, efileid) != len_i16s) {
+      return -1;
+    }
+
+    /* get channel data if provided */
+    if (argc==6) {
+      if (fread(&pli, sizeof(int16_t), 1, cfileid)) {
+        if ((pli!=0)&&(pli!=1)) {
+          fprintf(stderr, "Error in channel file\n");
+          exit(0);
+        }
+        if (pli==0) {
+          /* Packet loss -> remove info from frame */
+          memset(encoded_data, 0,
+                 sizeof(int16_t)*ILBCNOOFWORDS_MAX);
+        }
+      } else {
+        fprintf(stderr, "Error. Channel file too short\n");
+        exit(0);
+      }
+    } else {
+      pli=1;
+    }
+
+    /* decoding */
+
+    fprintf(stderr, "--- Decoding block %i --- ",blockcount);
+    if (pli==1) {
+      len_int=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data,
+                                   len, decoded_data,&speechType);
+      if (len_int < 0) {
+        fprintf(stderr, "Error decoding\n");
+        exit(0);
+      }
+      len = (size_t)len_int;
+    } else {
+      len=WebRtcIlbcfix_DecodePlc(Dec_Inst, decoded_data, 1);
+    }
+    fprintf(stderr, "\r");
+
+    /* write output file */
+
+    if (fwrite(decoded_data, sizeof(int16_t), len, ofileid) != len) {
+      return -1;
+    }
+  }
+
+  /* close files */
+
+  fclose(ifileid);  fclose(efileid); fclose(ofileid);
+  if (argc==6) {
+    fclose(cfileid);
+  }
+
+  /* Free structs */
+  WebRtcIlbcfix_EncoderFree(Enc_Inst);
+  WebRtcIlbcfix_DecoderFree(Dec_Inst);
+
+
+  printf("\nDone with simulation\n\n");
+
+  return(0);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
new file mode 100644
index 0000000000..132f3bdb37
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -0,0 +1,215 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+iLBC Speech Coder ANSI-C Source Code
+
+iLBC_test.c
+
+******************************************************************/
+
+#include <math.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <time.h>
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+//#define JUNK_DATA
+#ifdef JUNK_DATA
+#define SEED_FILE "randseed.txt"
+#endif
+
+
+/*----------------------------------------------------------------*
+*  Main program to test iLBC encoding and decoding
+*
+*  Usage:
+*		exefile_name.exe <infile> <bytefile> <outfile>
+*
+*---------------------------------------------------------------*/
+
+int main(int argc, char* argv[])
+{
+  FILE *ifileid,*efileid,*ofileid, *chfileid;
+  short encoded_data[55], data[240], speechType;
+  int len_int, mode;
+  short pli;
+  size_t len, readlen;
+  int blockcount = 0;
+
+  IlbcEncoderInstance *Enc_Inst;
+  IlbcDecoderInstance *Dec_Inst;
+#ifdef JUNK_DATA
+  size_t i;
+  FILE *seedfile;
+  unsigned int random_seed = (unsigned int) time(NULL);//1196764538
+#endif
+
+  /* Create structs */
+  WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
+  WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
+
+  /* get arguments and open files */
+
+  if (argc != 6 ) {
+    fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
+            argv[0]);
+    fprintf(stderr, "Example:\n");
+    fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
+    exit(1);
+  }
+  mode=atoi(argv[1]);
+  if (mode != 20 && mode != 30) {
+    fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
+    exit(2);
+  }
+  if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+    exit(2);}
+  if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open channelfile file %s\n",
+            argv[3]); exit(3);}
+  if( (ofileid=fopen(argv[4],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open output file %s\n",
+            argv[4]); exit(3);}
+  if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
+    exit(2);
+  }
+  /* print info */
+  fprintf(stderr, "\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*      iLBCtest                                     *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+		"*---------------------------------------------------*\n");
+#ifdef SPLIT_10MS
+  fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
+#else
+  fprintf(stderr,"\nMode          : %2d ms\n", mode);
+#endif
+  fprintf(stderr,"\nInput file    : %s\n", argv[2]);
+  fprintf(stderr,"Coded file    : %s\n", argv[3]);
+  fprintf(stderr,"Output file   : %s\n\n", argv[4]);
+  fprintf(stderr,"Channel file  : %s\n\n", argv[5]);
+
+#ifdef JUNK_DATA
+  srand(random_seed);
+
+  if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+    fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+  }
+  else {
+    fprintf(seedfile, "%u\n", random_seed);
+    fclose(seedfile);
+  }
+#endif
+
+  /* Initialization */
+  WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
+  WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
+
+  /* loop over input blocks */
+#ifdef SPLIT_10MS
+  readlen = 80;
+#else
+  readlen = (size_t)(mode << 3);
+#endif
+  while(fread(data, sizeof(short), readlen, ifileid) == readlen) {
+    blockcount++;
+
+    /* encoding */
+    fprintf(stderr, "--- Encoding block %i --- ",blockcount);
+    len_int=WebRtcIlbcfix_Encode(Enc_Inst, data, readlen, encoded_data);
+    if (len_int < 0) {
+      fprintf(stderr, "Error encoding\n");
+      exit(0);
+    }
+    len = (size_t)len_int;
+    fprintf(stderr, "\r");
+
+#ifdef JUNK_DATA
+    for ( i = 0; i < len; i++) {
+      encoded_data[i] = (short) (encoded_data[i] + (short) rand());
+    }
+#endif
+    /* write byte file */
+    if(len != 0){ //len may be 0 in 10ms split case
+      fwrite(encoded_data,1,len,efileid);
+
+      /* get channel data if provided */
+      if (argc==6) {
+        if (fread(&pli, sizeof(int16_t), 1, chfileid)) {
+          if ((pli!=0)&&(pli!=1)) {
+            fprintf(stderr, "Error in channel file\n");
+            exit(0);
+          }
+          if (pli==0) {
+            /* Packet loss -> remove info from frame */
+            memset(encoded_data, 0, sizeof(int16_t)*25);
+          }
+        } else {
+          fprintf(stderr, "Error. Channel file too short\n");
+          exit(0);
+        }
+      } else {
+        pli=1;
+      }
+
+      /* decoding */
+      fprintf(stderr, "--- Decoding block %i --- ",blockcount);
+      if (pli==1) {
+        len_int = WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data,
+                                       &speechType);
+        if (len_int < 0) {
+          fprintf(stderr, "Error decoding\n");
+          exit(0);
+        }
+        len = (size_t)len_int;
+      } else {
+        len=WebRtcIlbcfix_DecodePlc(Dec_Inst, data, 1);
+      }
+      fprintf(stderr, "\r");
+
+      /* write output file */
+      fwrite(data,sizeof(short),len,ofileid);
+    }
+  }
+
+#ifdef JUNK_DATA
+  if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+    fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+  }
+  else {
+    fprintf(seedfile, "ok\n\n");
+    fclose(seedfile);
+  }
+#endif
+
+  /* free structs */
+  WebRtcIlbcfix_EncoderFree(Enc_Inst);
+  WebRtcIlbcfix_DecoderFree(Dec_Inst);
+
+  /* close files */
+  fclose(ifileid);
+  fclose(efileid);
+  fclose(ofileid);
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
new file mode 100644
index 0000000000..a62a42edf6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
@@ -0,0 +1,343 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+	iLBC Speech Coder ANSI-C Source Code
+
+        iLBC_test.c
+
+******************************************************************/
+
+#include <math.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/nit_encode.h"
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+#define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS)/2
+
+/* Runtime statistics */
+#include <time.h>
+/* #define CLOCKS_PER_SEC  1000 */
+
+/*----------------------------------------------------------------*
+ *  Encoder interface function
+ *---------------------------------------------------------------*/
+
+short encode(                         /* (o) Number of bytes encoded */
+    IlbcEncoder *iLBCenc_inst,    /* (i/o) Encoder instance */
+    int16_t *encoded_data,      /* (o) The encoded bytes */
+    int16_t *data               /* (i) The signal block to encode */
+                                                        ){
+
+  /* do the actual encoding */
+  WebRtcIlbcfix_Encode((uint16_t *)encoded_data, data, iLBCenc_inst);
+
+  return (iLBCenc_inst->no_of_bytes);
+}
+
+/*----------------------------------------------------------------*
+ *  Decoder interface function
+ *---------------------------------------------------------------*/
+
+short decode( /* (o) Number of decoded samples */
+    IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
+    short *decoded_data, /* (o) Decoded signal block */
+    short *encoded_data, /* (i) Encoded bytes */
+    short mode           /* (i) 0=PL, 1=Normal */
+              ){
+
+  /* check if mode is valid */
+
+  if (mode<0 || mode>1) {
+    printf("\nERROR - Wrong mode - 0, 1 allowed\n"); exit(3);}
+
+  /* do actual decoding of block */
+
+  WebRtcIlbcfix_Decode(decoded_data, (uint16_t *)encoded_data,
+                       iLBCdec_inst, mode);
+
+  return (iLBCdec_inst->blockl);
+}
+
+/*----------------------------------------------------------------*
+ *  Main program to test iLBC encoding and decoding
+ *
+ *  Usage:
+ *		exefile_name.exe <infile> <bytefile> <outfile> <channelfile>
+ *
+ *---------------------------------------------------------------*/
+
+#define MAXFRAMES   10000
+#define MAXFILELEN (BLOCKL_MAX*MAXFRAMES)
+
+int main(int argc, char* argv[])
+{
+
+  /* Runtime statistics */
+
+  float starttime1, starttime2;
+  float runtime1, runtime2;
+  float outtime;
+
+  FILE *ifileid,*efileid,*ofileid, *chfileid;
+  short *inputdata, *encodeddata, *decodeddata;
+  short *channeldata;
+  int blockcount = 0, noOfBlocks=0, i, noOfLostBlocks=0;
+  short mode;
+  IlbcEncoder Enc_Inst;
+  IlbcDecoder Dec_Inst;
+
+  short frameLen;
+  short count;
+#ifdef SPLIT_10MS
+  short size;
+#endif
+
+  inputdata=(short*) malloc(MAXFILELEN*sizeof(short));
+  if (inputdata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    exit(0);
+  }
+  encodeddata=(short*) malloc(ILBCNOOFWORDS_MAX*MAXFRAMES*sizeof(short));
+  if (encodeddata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    free(inputdata);
+    exit(0);
+  }
+  decodeddata=(short*) malloc(MAXFILELEN*sizeof(short));
+  if (decodeddata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    free(inputdata);
+    free(encodeddata);
+    exit(0);
+  }
+  channeldata=(short*) malloc(MAXFRAMES*sizeof(short));
+  if (channeldata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    free(inputdata);
+    free(encodeddata);
+    free(decodeddata);
+    exit(0);
+  }
+
+  /* get arguments and open files */
+
+  if (argc != 6 ) {
+    fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
+            argv[0]);
+    fprintf(stderr, "Example:\n");
+    fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
+    exit(1);
+  }
+  mode=atoi(argv[1]);
+  if (mode != 20 && mode != 30) {
+    fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
+    exit(2);
+  }
+  if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+    exit(2);}
+  if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open channelfile file %s\n",
+            argv[3]); exit(3);}
+  if( (ofileid=fopen(argv[4],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open output file %s\n",
+            argv[4]); exit(3);}
+  if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
+    exit(2);}
+
+
+  /* print info */
+#ifndef PRINT_MIPS
+  fprintf(stderr, "\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*      iLBCtest                                     *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+#ifdef SPLIT_10MS
+  fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
+#else
+  fprintf(stderr,"\nMode          : %2d ms\n", mode);
+#endif
+  fprintf(stderr,"\nInput file    : %s\n", argv[2]);
+  fprintf(stderr,"Coded file    : %s\n", argv[3]);
+  fprintf(stderr,"Output file   : %s\n\n", argv[4]);
+  fprintf(stderr,"Channel file  : %s\n\n", argv[5]);
+#endif
+
+  /* Initialization */
+
+  WebRtcIlbcfix_EncoderInit(&Enc_Inst, mode);
+  WebRtcIlbcfix_DecoderInit(&Dec_Inst, mode, 1);
+
+  /* extract the input file and channel file */
+
+#ifdef SPLIT_10MS
+  frameLen = (mode==20)? 80:160;
+  fread(Enc_Inst.past_samples, sizeof(short), frameLen, ifileid);
+  Enc_Inst.section = 0;
+
+  while( fread(&inputdata[noOfBlocks*80], sizeof(short),
+               80, ifileid) == 80 ) {
+    noOfBlocks++;
+  }
+
+  noOfBlocks += frameLen/80;
+  frameLen = 80;
+#else
+  frameLen = Enc_Inst.blockl;
+
+  while( fread(&inputdata[noOfBlocks*Enc_Inst.blockl],sizeof(short),
+               Enc_Inst.blockl,ifileid)==(uint16_t)Enc_Inst.blockl){
+    noOfBlocks++;
+  }
+#endif
+
+
+  while ((fread(&channeldata[blockcount],sizeof(short), 1,chfileid)==1)
+            && ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) )) {
+    blockcount++;
+  }
+
+  if ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) ) {
+    fprintf(stderr,"Channel file %s is too short\n", argv[4]);
+    free(inputdata);
+    free(encodeddata);
+    free(decodeddata);
+    free(channeldata);
+    exit(0);
+  }
+
+  count=0;
+
+  /* Runtime statistics */
+
+  starttime1 = clock()/(float)CLOCKS_PER_SEC;
+
+  /* Encoding loop */
+#ifdef PRINT_MIPS
+  printf("-1 -1\n");
+#endif
+
+#ifdef SPLIT_10MS
+  /* "Enc_Inst.section != 0" is to make sure we run through full
+     lengths of all vectors for 10ms split mode.
+  */
+  //   while( (count < noOfBlocks) || (Enc_Inst.section != 0) )    {
+  while( count < blockcount * (Enc_Inst.blockl/frameLen) )    {
+
+    encode(&Enc_Inst, &encodeddata[Enc_Inst.no_of_words *
+                                   (count/(Enc_Inst.nsub/2))],
+           &inputdata[frameLen * count] );
+#else
+    while (count < noOfBlocks) {
+      encode( &Enc_Inst, &encodeddata[Enc_Inst.no_of_words * count],
+              &inputdata[frameLen * count] );
+#endif
+
+#ifdef PRINT_MIPS
+      printf("-1 -1\n");
+#endif
+
+      count++;
+    }
+
+    count=0;
+
+    /* Runtime statistics */
+
+    starttime2=clock()/(float)CLOCKS_PER_SEC;
+    runtime1 = (float)(starttime2-starttime1);
+
+    /* Decoding loop */
+
+    while (count < blockcount) {
+      if (channeldata[count]==1) {
+        /* Normal decoding */
+        decode(&Dec_Inst, &decodeddata[count * Dec_Inst.blockl],
+               &encodeddata[Dec_Inst.no_of_words * count], 1);
+      } else if (channeldata[count]==0) {
+        /* PLC */
+        short emptydata[ILBCNOOFWORDS_MAX];
+        memset(emptydata, 0, Dec_Inst.no_of_words*sizeof(short));
+        decode(&Dec_Inst, &decodeddata[count*Dec_Inst.blockl],
+               emptydata, 0);
+        noOfLostBlocks++;
+      } else {
+        printf("Error in channel file (values have to be either 1 or 0)\n");
+        exit(0);
+      }
+#ifdef PRINT_MIPS
+      printf("-1 -1\n");
+#endif
+
+      count++;
+    }
+
+    /* Runtime statistics */
+
+    runtime2 = (float)(clock()/(float)CLOCKS_PER_SEC-starttime2);
+
+    outtime = (float)((float)blockcount*
+                      (float)mode/1000.0);
+
+#ifndef PRINT_MIPS
+    printf("\nLength of speech file: %.1f s\n", outtime);
+    printf("Lost frames          : %.1f%%\n\n", 100*(float)noOfLostBlocks/(float)blockcount);
+
+    printf("Time to run iLBC_encode+iLBC_decode:");
+    printf(" %.1f s (%.1f%% of realtime)\n", runtime1+runtime2,
+           (100*(runtime1+runtime2)/outtime));
+
+    printf("Time in iLBC_encode                :");
+    printf(" %.1f s (%.1f%% of total runtime)\n",
+           runtime1, 100.0*runtime1/(runtime1+runtime2));
+
+    printf("Time in iLBC_decode                :");
+    printf(" %.1f s (%.1f%% of total runtime)\n\n",
+           runtime2, 100.0*runtime2/(runtime1+runtime2));
+#endif
+
+    /* Write data to files */
+    for (i=0; i<blockcount; i++) {
+      fwrite(&encodeddata[i*Enc_Inst.no_of_words], sizeof(short),
+             Enc_Inst.no_of_words, efileid);
+    }
+    for (i=0;i<blockcount;i++) {
+      fwrite(&decodeddata[i*Enc_Inst.blockl],sizeof(short),Enc_Inst.blockl,ofileid);
+    }
+
+    /* return memory and close files */
+
+    free(inputdata);
+    free(encodeddata);
+    free(decodeddata);
+    free(channeldata);
+    fclose(ifileid);  fclose(efileid); fclose(ofileid);
+    return(0);
+  }
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c
new file mode 100644
index 0000000000..a9a0147b9d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c
@@ -0,0 +1,241 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_UnpackBits.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
+    const uint16_t *bitstream,    /* (i) The packatized bitstream */
+    iLBC_bits *enc_bits,  /* (o) Paramerers from bitstream */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
+                                        ) {
+  const uint16_t *bitstreamPtr;
+  int i, k;
+  int16_t *tmpPtr;
+
+  bitstreamPtr=bitstream;
+
+  /* First int16_t */
+  enc_bits->lsf[0]  =  (*bitstreamPtr)>>10;       /* Bit 0..5  */
+  enc_bits->lsf[1]  = ((*bitstreamPtr)>>3)&0x7F;      /* Bit 6..12 */
+  enc_bits->lsf[2]  = ((*bitstreamPtr)&0x7)<<4;      /* Bit 13..15 */
+  bitstreamPtr++;
+  /* Second int16_t */
+  enc_bits->lsf[2] |= ((*bitstreamPtr)>>12)&0xF;      /* Bit 0..3  */
+
+  if (mode==20) {
+    enc_bits->startIdx             = ((*bitstreamPtr)>>10)&0x3;  /* Bit 4..5  */
+    enc_bits->state_first          = ((*bitstreamPtr)>>9)&0x1;  /* Bit 6  */
+    enc_bits->idxForMax            = ((*bitstreamPtr)>>3)&0x3F;  /* Bit 7..12 */
+    enc_bits->cb_index[0]          = ((*bitstreamPtr)&0x7)<<4;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>12)&0xE;  /* Bit 0..2  */
+    enc_bits->gain_index[0]        = ((*bitstreamPtr)>>8)&0x18;  /* Bit 3..4  */
+    enc_bits->gain_index[1]        = ((*bitstreamPtr)>>7)&0x8;  /* Bit 5  */
+    enc_bits->cb_index[3]          = ((*bitstreamPtr)>>2)&0xFE;  /* Bit 6..12 */
+    enc_bits->gain_index[3]        = ((*bitstreamPtr)<<2)&0x10;  /* Bit 13  */
+    enc_bits->gain_index[4]        = ((*bitstreamPtr)<<2)&0x8;  /* Bit 14  */
+    enc_bits->gain_index[6]        = ((*bitstreamPtr)<<4)&0x10;  /* Bit 15  */
+  } else { /* mode==30 */
+    enc_bits->lsf[3]               = ((*bitstreamPtr)>>6)&0x3F;  /* Bit 4..9  */
+    enc_bits->lsf[4]               = ((*bitstreamPtr)<<1)&0x7E;  /* Bit 10..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    enc_bits->lsf[4]              |= ((*bitstreamPtr)>>15)&0x1;  /* Bit 0  */
+    enc_bits->lsf[5]               = ((*bitstreamPtr)>>8)&0x7F;  /* Bit 1..7  */
+    enc_bits->startIdx             = ((*bitstreamPtr)>>5)&0x7;  /* Bit 8..10 */
+    enc_bits->state_first          = ((*bitstreamPtr)>>4)&0x1;  /* Bit 11  */
+    enc_bits->idxForMax            = ((*bitstreamPtr)<<2)&0x3C;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 4:th int16_t */
+    enc_bits->idxForMax           |= ((*bitstreamPtr)>>14)&0x3;  /* Bit 0..1  */
+    enc_bits->cb_index[0]        = ((*bitstreamPtr)>>7)&0x78;  /* Bit 2..5  */
+    enc_bits->gain_index[0]        = ((*bitstreamPtr)>>5)&0x10;  /* Bit 6  */
+    enc_bits->gain_index[1]        = ((*bitstreamPtr)>>5)&0x8;  /* Bit 7  */
+    enc_bits->cb_index[3]          = ((*bitstreamPtr))&0xFC;  /* Bit 8..13 */
+    enc_bits->gain_index[3]        = ((*bitstreamPtr)<<3)&0x10;  /* Bit 14  */
+    enc_bits->gain_index[4]        = ((*bitstreamPtr)<<3)&0x8;  /* Bit 15  */
+  }
+  /* Class 2 bits of ULP */
+  /* 4:th to 6:th int16_t for 20 ms case
+     5:th to 7:th int16_t for 30 ms case */
+  bitstreamPtr++;
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<3; k++) {
+    for (i=15; i>=0; i--) {
+      (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 7:th int16_t */
+    for (i=15; i>6; i--) {
+      (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>4)&0x4; /* Bit 9  */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>1)&0x4; /* Bit 12  */
+    enc_bits->gain_index[6]       |= ((*bitstreamPtr)<<1)&0x8; /* Bit 13  */
+    enc_bits->gain_index[7]        = ((*bitstreamPtr)<<2)&0xC; /* Bit 14..15 */
+
+  } else { /* mode==30 */
+    /* 8:th int16_t */
+    for (i=15; i>5; i--) {
+      (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>3)&0x6; /* Bit 10..11 */
+    enc_bits->gain_index[0]       |= ((*bitstreamPtr))&0x8;  /* Bit 12  */
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr))&0x4;  /* Bit 13  */
+    enc_bits->cb_index[3]         |= ((*bitstreamPtr))&0x2;  /* Bit 14  */
+    enc_bits->cb_index[6]          = ((*bitstreamPtr)<<7)&0x80; /* Bit 15  */
+    bitstreamPtr++;
+    /* 9:th int16_t */
+    enc_bits->cb_index[6]         |= ((*bitstreamPtr)>>9)&0x7E; /* Bit 0..5  */
+    enc_bits->cb_index[9]          = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */
+    enc_bits->cb_index[12]         = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 10:th int16_t */
+    enc_bits->cb_index[12]         |= ((*bitstreamPtr)>>11)&0x1E;/* Bit 0..3 */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>8)&0xC; /* Bit 4..5  */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>7)&0x6; /* Bit 6..7  */
+    enc_bits->gain_index[6]        = ((*bitstreamPtr)>>3)&0x18; /* Bit 8..9  */
+    enc_bits->gain_index[7]        = ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
+    enc_bits->gain_index[9]        = ((*bitstreamPtr)<<1)&0x10; /* Bit 12  */
+    enc_bits->gain_index[10]       = ((*bitstreamPtr)<<1)&0x8; /* Bit 13  */
+    enc_bits->gain_index[12]       = ((*bitstreamPtr)<<3)&0x10; /* Bit 14  */
+    enc_bits->gain_index[13]       = ((*bitstreamPtr)<<3)&0x8; /* Bit 15  */
+  }
+  bitstreamPtr++;
+  /* Class 3 bits of ULP */
+  /*  8:th to 14:th int16_t for 20 ms case
+      11:th to 17:th int16_t for 30 ms case */
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<7; k++) {
+    for (i=14; i>=0; i-=2) {
+      (*tmpPtr)                 |= ((*bitstreamPtr)>>i)&0x3; /* Bit 15-i..14-i*/
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 15:th int16_t */
+    enc_bits->idxVec[56]          |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>13)&0x1; /* Bit 2  */
+    enc_bits->cb_index[1]          = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9  */
+    enc_bits->cb_index[2]          = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 16:th int16_t */
+    enc_bits->cb_index[2]         |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0  */
+    enc_bits->gain_index[0]       |= ((*bitstreamPtr)>>12)&0x7; /* Bit 1..3  */
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5  */
+    enc_bits->gain_index[2]        = ((*bitstreamPtr)>>7)&0x7; /* Bit 6..8  */
+    enc_bits->cb_index[3]         |= ((*bitstreamPtr)>>6)&0x1; /* Bit 9  */
+    enc_bits->cb_index[4]          = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 17:th int16_t */
+    enc_bits->cb_index[4]         |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0  */
+    enc_bits->cb_index[5]          = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7  */
+    enc_bits->cb_index[6]          = ((*bitstreamPtr))&0xFF; /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 18:th int16_t */
+    enc_bits->cb_index[7]          = (*bitstreamPtr)>>8;  /* Bit 0..7  */
+    enc_bits->cb_index[8]          = (*bitstreamPtr)&0xFF;  /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3  */
+    enc_bits->gain_index[5]        = ((*bitstreamPtr)>>9)&0x7; /* Bit 4..6  */
+    enc_bits->gain_index[6]       |= ((*bitstreamPtr)>>6)&0x7; /* Bit 7..9  */
+    enc_bits->gain_index[7]       |= ((*bitstreamPtr)>>4)&0x3; /* Bit 10..11 */
+    enc_bits->gain_index[8]        = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
+  } else { /* mode==30 */
+    /* 18:th int16_t */
+    enc_bits->idxVec[56]          |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->idxVec[57]          |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3  */
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>11)&1; /* Bit 4  */
+    enc_bits->cb_index[1]          = ((*bitstreamPtr)>>4)&0x7F; /* Bit 5..11 */
+    enc_bits->cb_index[2]          = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    enc_bits->cb_index[2]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->gain_index[0]       |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5  */
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7  */
+    enc_bits->gain_index[2]        = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
+    enc_bits->cb_index[3]         |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11  */
+    enc_bits->cb_index[4]          = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 20:th int16_t */
+    enc_bits->cb_index[4]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->cb_index[5]          = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9  */
+    enc_bits->cb_index[6]         |= ((*bitstreamPtr)>>5)&0x1; /* Bit 10  */
+    enc_bits->cb_index[7]          = ((*bitstreamPtr)<<3)&0xF8; /* Bit 11..15 */
+    bitstreamPtr++;
+    /* 21:st int16_t */
+    enc_bits->cb_index[7]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->cb_index[8]          = ((*bitstreamPtr)>>5)&0xFF; /* Bit 3..10 */
+    enc_bits->cb_index[9]         |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11  */
+    enc_bits->cb_index[10]         = ((*bitstreamPtr)<<4)&0xF0; /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 22:nd int16_t */
+    enc_bits->cb_index[10]        |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3  */
+    enc_bits->cb_index[11]         = ((*bitstreamPtr)>>4)&0xFF; /* Bit 4..11 */
+    enc_bits->cb_index[12]        |= ((*bitstreamPtr)>>3)&0x1; /* Bit 12  */
+    enc_bits->cb_index[13]         = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 23:rd int16_t */
+    enc_bits->cb_index[13]        |= ((*bitstreamPtr)>>11)&0x1F;/* Bit 0..4  */
+    enc_bits->cb_index[14]         = ((*bitstreamPtr)>>3)&0xFF; /* Bit 5..12 */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>1)&0x3; /* Bit 13..14 */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)&0x1);  /* Bit 15  */
+    bitstreamPtr++;
+    /* 24:rd int16_t */
+    enc_bits->gain_index[5]        = ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->gain_index[6]       |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5  */
+    enc_bits->gain_index[7]       |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7  */
+    enc_bits->gain_index[8]        = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
+    enc_bits->gain_index[9]       |= ((*bitstreamPtr)>>1)&0xF; /* Bit 11..14 */
+    enc_bits->gain_index[10]      |= ((*bitstreamPtr)<<2)&0x4; /* Bit 15  */
+    bitstreamPtr++;
+    /* 25:rd int16_t */
+    enc_bits->gain_index[10]      |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->gain_index[11]       = ((*bitstreamPtr)>>11)&0x7; /* Bit 2..4  */
+    enc_bits->gain_index[12]      |= ((*bitstreamPtr)>>7)&0xF; /* Bit 5..8  */
+    enc_bits->gain_index[13]      |= ((*bitstreamPtr)>>4)&0x7; /* Bit 9..11 */
+    enc_bits->gain_index[14]       = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
+  }
+  /* Last bit should be zero, otherwise it's an "empty" frame */
+  if (((*bitstreamPtr)&0x1) == 1) {
+    return(1);
+  } else {
+    return(0);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h
new file mode 100644
index 0000000000..1a63280e6b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_UnpackBits.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+int16_t
+WebRtcIlbcfix_UnpackBits(/* (o) "Empty" frame indicator */
+                         const uint16_t*
+                             bitstream, /* (i) The packatized bitstream */
+                         iLBC_bits*
+                             enc_bits, /* (o) Paramerers from bitstream */
+                         int16_t mode  /* (i) Codec mode (20 or 30) */
+                         );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c
new file mode 100644
index 0000000000..d9375fb995
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq3.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/vq3.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+/*----------------------------------------------------------------*
+ *  vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq3(
+    int16_t *Xq, /* quantized vector (Q13) */
+    int16_t *index,
+    int16_t *CB, /* codebook in Q13 */
+    int16_t *X,  /* vector to quantize (Q13) */
+    int16_t n_cb
+                       ){
+  int16_t i, j;
+  int16_t pos, minindex=0;
+  int16_t tmp;
+  int32_t dist, mindist;
+
+  pos = 0;
+  mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
+
+  /* Find the codebook with the lowest square distance */
+  for (j = 0; j < n_cb; j++) {
+    tmp = X[0] - CB[pos];
+    dist = tmp * tmp;
+    for (i = 1; i < 3; i++) {
+      tmp = X[i] - CB[pos + i];
+      dist += tmp * tmp;
+    }
+
+    if (dist < mindist) {
+      mindist = dist;
+      minindex = j;
+    }
+    pos += 3;
+  }
+
+  /* Store the quantized codebook and the index */
+  for (i = 0; i < 3; i++) {
+    Xq[i] = CB[minindex*3 + i];
+  }
+  *index = minindex;
+
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.h
new file mode 100644
index 0000000000..c946478a1a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq3.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Vector quantization of order 3 (based on MSE)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq3(
+    int16_t* Xq,    /* (o) the quantized vector (Q13) */
+    int16_t* index, /* (o) the quantization index */
+    int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
+    int16_t* X,     /* (i) the vector to quantize (Q13) */
+    int16_t n_cb    /* (i) the number of vectors in the codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c
new file mode 100644
index 0000000000..c9a65aec2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq4.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/vq4.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+/*----------------------------------------------------------------*
+ *  vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq4(
+    int16_t *Xq, /* quantized vector (Q13) */
+    int16_t *index,
+    int16_t *CB, /* codebook in Q13 */
+    int16_t *X,  /* vector to quantize (Q13) */
+    int16_t n_cb
+                       ){
+  int16_t i, j;
+  int16_t pos, minindex=0;
+  int16_t tmp;
+  int32_t dist, mindist;
+
+  pos = 0;
+  mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
+
+  /* Find the codebook with the lowest square distance */
+  for (j = 0; j < n_cb; j++) {
+    tmp = X[0] - CB[pos];
+    dist = tmp * tmp;
+    for (i = 1; i < 4; i++) {
+      tmp = X[i] - CB[pos + i];
+      dist += tmp * tmp;
+    }
+
+    if (dist < mindist) {
+      mindist = dist;
+      minindex = j;
+    }
+    pos += 4;
+  }
+
+  /* Store the quantized codebook and the index */
+  for (i = 0; i < 4; i++) {
+    Xq[i] = CB[minindex*4 + i];
+  }
+  *index = minindex;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.h
new file mode 100644
index 0000000000..6d14830c03
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq4.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Vector quantization of order 4 (based on MSE)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq4(
+    int16_t* Xq,    /* (o) the quantized vector (Q13) */
+    int16_t* index, /* (o) the quantization index */
+    int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
+    int16_t* X,     /* (i) the vector to quantize (Q13) */
+    int16_t n_cb    /* (i) the number of vectors in the codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
new file mode 100644
index 0000000000..e82d167220
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Window32W32.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/window32_w32.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  window multiplication
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Window32W32(
+    int32_t *z,    /* Output */
+    int32_t *x,    /* Input (same domain as Output)*/
+    const int32_t  *y,  /* Q31 Window */
+    size_t N     /* length to process */
+                               ) {
+  size_t i;
+  int16_t x_low, x_hi, y_low, y_hi;
+  int16_t left_shifts;
+  int32_t temp;
+
+  left_shifts = (int16_t)WebRtcSpl_NormW32(x[0]);
+  WebRtcSpl_VectorBitShiftW32(x, N, x, (int16_t)(-left_shifts));
+
+
+  /* The double precision numbers use a special representation:
+   * w32 = hi<<16 + lo<<1
+   */
+  for (i = 0; i < N; i++) {
+    /* Extract higher bytes */
+    x_hi = (int16_t)(x[i] >> 16);
+    y_hi = (int16_t)(y[i] >> 16);
+
+    /* Extract lower bytes, defined as (w32 - hi<<16)>>1 */
+    x_low = (int16_t)((x[i] - (x_hi << 16)) >> 1);
+
+    y_low = (int16_t)((y[i] - (y_hi << 16)) >> 1);
+
+    /* Calculate z by a 32 bit multiplication using both low and high from x and y */
+    temp = ((x_hi * y_hi) << 1) + ((x_hi * y_low) >> 14);
+
+    z[i] = temp + ((x_low * y_hi) >> 14);
+  }
+
+  WebRtcSpl_VectorBitShiftW32(z, N, z, left_shifts);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
new file mode 100644
index 0000000000..15d72c5ba2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Window32W32.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  window multiplication
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Window32W32(int32_t* z, /* Output */
+                               int32_t* x, /* Input (same domain as Output)*/
+                               const int32_t* y, /* Q31 Window */
+                               size_t N          /* length to process */
+                               );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
new file mode 100644
index 0000000000..9dc880b37e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
@@ -0,0 +1,142 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_XcorrCoef.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * cross correlation which finds the optimal lag for the
+ * crossCorr*crossCorr/(energy) criteria
+ *---------------------------------------------------------------*/
+
+size_t WebRtcIlbcfix_XcorrCoef(
+    int16_t *target,  /* (i) first array */
+    int16_t *regressor, /* (i) second array */
+    size_t subl,  /* (i) dimension arrays */
+    size_t searchLen, /* (i) the search lenght */
+    size_t offset,  /* (i) samples offset between arrays */
+    int16_t step   /* (i) +1 or -1 */
+                            ){
+  size_t k;
+  size_t maxlag;
+  int16_t pos;
+  int16_t max;
+  int16_t crossCorrScale, Energyscale;
+  int16_t crossCorrSqMod, crossCorrSqMod_Max;
+  int32_t crossCorr, Energy;
+  int16_t crossCorrmod, EnergyMod, EnergyMod_Max;
+  int16_t *tp, *rp;
+  int16_t *rp_beg, *rp_end;
+  int16_t totscale, totscale_max;
+  int16_t scalediff;
+  int32_t newCrit, maxCrit;
+  int shifts;
+
+  /* Initializations, to make sure that the first one is selected */
+  crossCorrSqMod_Max=0;
+  EnergyMod_Max=WEBRTC_SPL_WORD16_MAX;
+  totscale_max=-500;
+  maxlag=0;
+  pos=0;
+
+  /* Find scale value and start position */
+  if (step==1) {
+    max=WebRtcSpl_MaxAbsValueW16(regressor, subl + searchLen - 1);
+    rp_beg = regressor;
+    rp_end = regressor + subl;
+  } else { /* step==-1 */
+    max = WebRtcSpl_MaxAbsValueW16(regressor - searchLen, subl + searchLen - 1);
+    rp_beg = regressor - 1;
+    rp_end = regressor + subl - 1;
+  }
+
+  /* Introduce a scale factor on the Energy in int32_t in
+     order to make sure that the calculation does not
+     overflow */
+
+  if (max>5000) {
+    shifts=2;
+  } else {
+    shifts=0;
+  }
+
+  /* Calculate the first energy, then do a +/- to get the other energies */
+  Energy=WebRtcSpl_DotProductWithScale(regressor, regressor, subl, shifts);
+
+  for (k=0;k<searchLen;k++) {
+    tp = target;
+    rp = &regressor[pos];
+
+    crossCorr=WebRtcSpl_DotProductWithScale(tp, rp, subl, shifts);
+
+    if ((Energy>0)&&(crossCorr>0)) {
+
+      /* Put cross correlation and energy on 16 bit word */
+      crossCorrScale=(int16_t)WebRtcSpl_NormW32(crossCorr)-16;
+      crossCorrmod=(int16_t)WEBRTC_SPL_SHIFT_W32(crossCorr, crossCorrScale);
+      Energyscale=(int16_t)WebRtcSpl_NormW32(Energy)-16;
+      EnergyMod=(int16_t)WEBRTC_SPL_SHIFT_W32(Energy, Energyscale);
+
+      /* Square cross correlation and store upper int16_t */
+      crossCorrSqMod = (int16_t)((crossCorrmod * crossCorrmod) >> 16);
+
+      /* Calculate the total number of (dynamic) right shifts that have
+         been performed on (crossCorr*crossCorr)/energy
+      */
+      totscale=Energyscale-(crossCorrScale<<1);
+
+      /* Calculate the shift difference in order to be able to compare the two
+         (crossCorr*crossCorr)/energy in the same domain
+      */
+      scalediff=totscale-totscale_max;
+      scalediff=WEBRTC_SPL_MIN(scalediff,31);
+      scalediff=WEBRTC_SPL_MAX(scalediff,-31);
+
+      /* Compute the cross multiplication between the old best criteria
+         and the new one to be able to compare them without using a
+         division */
+
+      if (scalediff<0) {
+        newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max)>>(-scalediff);
+        maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod);
+      } else {
+        newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max);
+        maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod)>>scalediff;
+      }
+
+      /* Store the new lag value if the new criteria is larger
+         than previous largest criteria */
+
+      if (newCrit > maxCrit) {
+        crossCorrSqMod_Max = crossCorrSqMod;
+        EnergyMod_Max = EnergyMod;
+        totscale_max = totscale;
+        maxlag = k;
+      }
+    }
+    pos+=step;
+
+    /* Do a +/- to get the next energy */
+    Energy += step * ((*rp_end * *rp_end - *rp_beg * *rp_beg) >> shifts);
+    rp_beg+=step;
+    rp_end+=step;
+  }
+
+  return(maxlag+offset);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
new file mode 100644
index 0000000000..3be5a296b5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_XcorrCoef.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * cross correlation which finds the optimal lag for the
+ * crossCorr*crossCorr/(energy) criteria
+ *---------------------------------------------------------------*/
+
+size_t WebRtcIlbcfix_XcorrCoef(
+    int16_t* target,    /* (i) first array */
+    int16_t* regressor, /* (i) second array */
+    size_t subl,        /* (i) dimension arrays */
+    size_t searchLen,   /* (i) the search lenght */
+    size_t offset,      /* (i) samples offset between arrays */
+    int16_t step        /* (i) +1 or -1 */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/bandwidth_info.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
new file mode 100644
index 0000000000..c3830a5f7c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+
+#include <stdint.h>
+
+typedef struct {
+  int in_use;
+  int32_t send_bw_avg;
+  int32_t send_max_delay_avg;
+  int16_t bottleneck_idx;
+  int16_t jitter_info;
+} IsacBandwidthInfo;
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
new file mode 100644
index 0000000000..a4f297c5a1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
@@ -0,0 +1,195 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory.h>
+#include <string.h>
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
+
+static void WebRtcIsac_AllPoleFilter(double* InOut,
+                                     double* Coef,
+                                     size_t lengthInOut,
+                                     int orderCoef) {
+  /* the state of filter is assumed to be in InOut[-1] to InOut[-orderCoef] */
+  double scal;
+  double sum;
+  size_t n;
+  int k;
+
+  //if (fabs(Coef[0]-1.0)<0.001) {
+  if ( (Coef[0] > 0.9999) && (Coef[0] < 1.0001) )
+  {
+    for(n = 0; n < lengthInOut; n++)
+    {
+      sum = Coef[1] * InOut[-1];
+      for(k = 2; k <= orderCoef; k++){
+        sum += Coef[k] * InOut[-k];
+      }
+      *InOut++ -= sum;
+    }
+  }
+  else
+  {
+    scal = 1.0 / Coef[0];
+    for(n=0;n<lengthInOut;n++)
+    {
+      *InOut *= scal;
+      for(k=1;k<=orderCoef;k++){
+        *InOut -= scal*Coef[k]*InOut[-k];
+      }
+      InOut++;
+    }
+  }
+}
+
+static void WebRtcIsac_AllZeroFilter(double* In,
+                                     double* Coef,
+                                     size_t lengthInOut,
+                                     int orderCoef,
+                                     double* Out) {
+  /* the state of filter is assumed to be in In[-1] to In[-orderCoef] */
+
+  size_t n;
+  int k;
+  double tmp;
+
+  for(n = 0; n < lengthInOut; n++)
+  {
+    tmp = In[0] * Coef[0];
+
+    for(k = 1; k <= orderCoef; k++){
+      tmp += Coef[k] * In[-k];
+    }
+
+    *Out++ = tmp;
+    In++;
+  }
+}
+
+static void WebRtcIsac_ZeroPoleFilter(double* In,
+                                      double* ZeroCoef,
+                                      double* PoleCoef,
+                                      size_t lengthInOut,
+                                      int orderCoef,
+                                      double* Out) {
+  /* the state of the zero section is assumed to be in In[-1] to In[-orderCoef] */
+  /* the state of the pole section is assumed to be in Out[-1] to Out[-orderCoef] */
+
+  WebRtcIsac_AllZeroFilter(In,ZeroCoef,lengthInOut,orderCoef,Out);
+  WebRtcIsac_AllPoleFilter(Out,PoleCoef,lengthInOut,orderCoef);
+}
+
+
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order) {
+  size_t  lag, n;
+  double sum, prod;
+  const double *x_lag;
+
+  for (lag = 0; lag <= order; lag++)
+  {
+    sum = 0.0f;
+    x_lag = &x[lag];
+    prod = x[0] * x_lag[0];
+    for (n = 1; n < N - lag; n++) {
+      sum += prod;
+      prod = x[n] * x_lag[n];
+    }
+    sum += prod;
+    r[lag] = sum;
+  }
+
+}
+
+static void WebRtcIsac_BwExpand(double* out,
+                                double* in,
+                                double coef,
+                                size_t length) {
+  size_t i;
+  double  chirp;
+
+  chirp = coef;
+
+  out[0] = in[0];
+  for (i = 1; i < length; i++) {
+    out[i] = chirp * in[i];
+    chirp *= coef;
+  }
+}
+
+void WebRtcIsac_WeightingFilter(const double* in,
+                                double* weiout,
+                                double* whiout,
+                                WeightFiltstr* wfdata) {
+  double  tmpbuffer[PITCH_FRAME_LEN + PITCH_WLPCBUFLEN];
+  double  corr[PITCH_WLPCORDER+1], rc[PITCH_WLPCORDER+1];
+  double apol[PITCH_WLPCORDER+1], apolr[PITCH_WLPCORDER+1];
+  double  rho=0.9, *inp, *dp, *dp2;
+  double  whoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+  double  weoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+  double  *weo, *who, opol[PITCH_WLPCORDER+1], ext[PITCH_WLPCWINLEN];
+  int     k, n, endpos, start;
+
+  /* Set up buffer and states */
+  memcpy(tmpbuffer, wfdata->buffer, sizeof(double) * PITCH_WLPCBUFLEN);
+  memcpy(tmpbuffer+PITCH_WLPCBUFLEN, in, sizeof(double) * PITCH_FRAME_LEN);
+  memcpy(wfdata->buffer, tmpbuffer+PITCH_FRAME_LEN, sizeof(double) * PITCH_WLPCBUFLEN);
+
+  dp=weoutbuf;
+  dp2=whoutbuf;
+  for (k=0;k<PITCH_WLPCORDER;k++) {
+    *dp++ = wfdata->weostate[k];
+    *dp2++ = wfdata->whostate[k];
+    opol[k]=0.0;
+  }
+  opol[0]=1.0;
+  opol[PITCH_WLPCORDER]=0.0;
+  weo=dp;
+  who=dp2;
+
+  endpos=PITCH_WLPCBUFLEN + PITCH_SUBFRAME_LEN;
+  inp=tmpbuffer + PITCH_WLPCBUFLEN;
+
+  for (n=0; n<PITCH_SUBFRAMES; n++) {
+    /* Windowing */
+    start=endpos-PITCH_WLPCWINLEN;
+    for (k=0; k<PITCH_WLPCWINLEN; k++) {
+      ext[k]=wfdata->window[k]*tmpbuffer[start+k];
+    }
+
+    /* Get LPC polynomial */
+    WebRtcIsac_AutoCorr(corr, ext, PITCH_WLPCWINLEN, PITCH_WLPCORDER);
+    corr[0]=1.01*corr[0]+1.0; /* White noise correction */
+    WebRtcIsac_LevDurb(apol, rc, corr, PITCH_WLPCORDER);
+    WebRtcIsac_BwExpand(apolr, apol, rho, PITCH_WLPCORDER+1);
+
+    /* Filtering */
+    WebRtcIsac_ZeroPoleFilter(inp, apol, apolr, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, weo);
+    WebRtcIsac_ZeroPoleFilter(inp, apolr, opol, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, who);
+
+    inp+=PITCH_SUBFRAME_LEN;
+    endpos+=PITCH_SUBFRAME_LEN;
+    weo+=PITCH_SUBFRAME_LEN;
+    who+=PITCH_SUBFRAME_LEN;
+  }
+
+  /* Export filter states */
+  for (k=0;k<PITCH_WLPCORDER;k++) {
+    wfdata->weostate[k]=weoutbuf[PITCH_FRAME_LEN+k];
+    wfdata->whostate[k]=whoutbuf[PITCH_FRAME_LEN+k];
+  }
+
+  /* Export output data */
+  memcpy(weiout, weoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
+  memcpy(whiout, whoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.h
new file mode 100644
index 0000000000..a747a7f549
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
+
+#include <stddef.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
+
+void WebRtcIsac_WeightingFilter(const double* in,
+                                double* weiout,
+                                double* whiout,
+                                WeightFiltstr* wfdata);
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c
new file mode 100644
index 0000000000..57cf0c39da
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c
@@ -0,0 +1,409 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
+
+#include <math.h>
+
+void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata) {
+  int k;
+
+  for (k = 0; k < PITCH_BUFFSIZE; k++) {
+    pitchfiltdata->ubuf[k] = 0.0;
+  }
+  pitchfiltdata->ystate[0] = 0.0;
+  for (k = 1; k < (PITCH_DAMPORDER); k++) {
+    pitchfiltdata->ystate[k] = 0.0;
+  }
+  pitchfiltdata->oldlagp[0] = 50.0;
+  pitchfiltdata->oldgainp[0] = 0.0;
+}
+
+static void WebRtcIsac_InitWeightingFilter(WeightFiltstr* wfdata) {
+  int k;
+  double t, dtmp, dtmp2, denum, denum2;
+
+  for (k = 0; k < PITCH_WLPCBUFLEN; k++)
+    wfdata->buffer[k] = 0.0;
+
+  for (k = 0; k < PITCH_WLPCORDER; k++) {
+    wfdata->istate[k] = 0.0;
+    wfdata->weostate[k] = 0.0;
+    wfdata->whostate[k] = 0.0;
+  }
+
+  /* next part should be in Matlab, writing to a global table */
+  t = 0.5;
+  denum = 1.0 / ((double)PITCH_WLPCWINLEN);
+  denum2 = denum * denum;
+  for (k = 0; k < PITCH_WLPCWINLEN; k++) {
+    dtmp = PITCH_WLPCASYM * t * denum + (1 - PITCH_WLPCASYM) * t * t * denum2;
+    dtmp *= 3.14159265;
+    dtmp2 = sin(dtmp);
+    wfdata->window[k] = dtmp2 * dtmp2;
+    t++;
+  }
+}
+
+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* State) {
+  int k;
+
+  for (k = 0; k < PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
+                      PITCH_FRAME_LEN / 2 + 2;
+       k++)
+    State->dec_buffer[k] = 0.0;
+  for (k = 0; k < 2 * ALLPASSSECTIONS + 1; k++)
+    State->decimator_state[k] = 0.0;
+  for (k = 0; k < 2; k++)
+    State->hp_state[k] = 0.0;
+  for (k = 0; k < QLOOKAHEAD; k++)
+    State->whitened_buf[k] = 0.0;
+  for (k = 0; k < QLOOKAHEAD; k++)
+    State->inbuf[k] = 0.0;
+
+  WebRtcIsac_InitPitchFilter(&(State->PFstr_wght));
+
+  WebRtcIsac_InitPitchFilter(&(State->PFstr));
+
+  WebRtcIsac_InitWeightingFilter(&(State->Wghtstr));
+}
+
+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata) {
+  int k;
+
+  for (k = 0; k < QLOOKAHEAD; k++) {
+    prefiltdata->INLABUF1[k] = 0;
+    prefiltdata->INLABUF2[k] = 0;
+
+    prefiltdata->INLABUF1_float[k] = 0;
+    prefiltdata->INLABUF2_float[k] = 0;
+  }
+  for (k = 0; k < 2 * (QORDER - 1); k++) {
+    prefiltdata->INSTAT1[k] = 0;
+    prefiltdata->INSTAT2[k] = 0;
+    prefiltdata->INSTATLA1[k] = 0;
+    prefiltdata->INSTATLA2[k] = 0;
+
+    prefiltdata->INSTAT1_float[k] = 0;
+    prefiltdata->INSTAT2_float[k] = 0;
+    prefiltdata->INSTATLA1_float[k] = 0;
+    prefiltdata->INSTATLA2_float[k] = 0;
+  }
+
+  /* High pass filter states */
+  prefiltdata->HPstates[0] = 0.0;
+  prefiltdata->HPstates[1] = 0.0;
+
+  prefiltdata->HPstates_float[0] = 0.0f;
+  prefiltdata->HPstates_float[1] = 0.0f;
+
+  return;
+}
+
+double WebRtcIsac_LevDurb(double* a, double* k, double* r, size_t order) {
+  const double LEVINSON_EPS = 1.0e-10;
+
+  double sum, alpha;
+  size_t m, m_h, i;
+  alpha = 0;  // warning -DH
+  a[0] = 1.0;
+  if (r[0] < LEVINSON_EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
+    for (i = 0; i < order; i++) {
+      k[i] = 0;
+      a[i + 1] = 0;
+    }
+  } else {
+    a[1] = k[0] = -r[1] / r[0];
+    alpha = r[0] + r[1] * k[0];
+    for (m = 1; m < order; m++) {
+      sum = r[m + 1];
+      for (i = 0; i < m; i++) {
+        sum += a[i + 1] * r[m - i];
+      }
+      k[m] = -sum / alpha;
+      alpha += k[m] * sum;
+      m_h = (m + 1) >> 1;
+      for (i = 0; i < m_h; i++) {
+        sum = a[i + 1] + k[m] * a[m - i];
+        a[m - i] += k[m] * a[i + 1];
+        a[i + 1] = sum;
+      }
+      a[m + 1] = k[m];
+    }
+  }
+  return alpha;
+}
+
+/* The upper channel all-pass filter factors */
+const float WebRtcIsac_kUpperApFactorsFloat[2] = {0.03470000000000f,
+                                                  0.38260000000000f};
+
+/* The lower channel all-pass filter factors */
+const float WebRtcIsac_kLowerApFactorsFloat[2] = {0.15440000000000f,
+                                                  0.74400000000000f};
+
+/* This function performs all-pass filtering--a series of first order all-pass
+ * sections are used to filter the input in a cascade manner.
+ * The input is overwritten!!
+ */
+void WebRtcIsac_AllPassFilter2Float(float* InOut,
+                                    const float* APSectionFactors,
+                                    int lengthInOut,
+                                    int NumberOfSections,
+                                    float* FilterState) {
+  int n, j;
+  float temp;
+  for (j = 0; j < NumberOfSections; j++) {
+    for (n = 0; n < lengthInOut; n++) {
+      temp = FilterState[j] + APSectionFactors[j] * InOut[n];
+      FilterState[j] = -APSectionFactors[j] * temp + InOut[n];
+      InOut[n] = temp;
+    }
+  }
+}
+
+/* The number of composite all-pass filter factors */
+#define NUMBEROFCOMPOSITEAPSECTIONS 4
+
+/* Function WebRtcIsac_SplitAndFilter
+ * This function creates low-pass and high-pass decimated versions of part of
+ the input signal, and part of the signal in the input 'lookahead buffer'.
+
+ INPUTS:
+ in: a length FRAMESAMPLES array of input samples
+ prefiltdata: input data structure containing the filterbank states
+ and lookahead samples from the previous encoding
+ iteration.
+ OUTPUTS:
+ LP: a FRAMESAMPLES_HALF array of low-pass filtered samples that
+ have been phase equalized.  The first QLOOKAHEAD samples are
+ based on the samples in the two prefiltdata->INLABUFx arrays
+ each of length QLOOKAHEAD.
+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
+ array in[].
+ HP: a FRAMESAMPLES_HALF array of high-pass filtered samples that
+ have been phase equalized.  The first QLOOKAHEAD samples are
+ based on the samples in the two prefiltdata->INLABUFx arrays
+ each of length QLOOKAHEAD.
+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
+ array in[].
+
+ LP_la: a FRAMESAMPLES_HALF array of low-pass filtered samples.
+ These samples are not phase equalized. They are computed
+ from the samples in the in[] array.
+ HP_la: a FRAMESAMPLES_HALF array of high-pass filtered samples
+ that are not phase equalized. They are computed from
+ the in[] vector.
+ prefiltdata: this input data structure's filterbank state and
+ lookahead sample buffers are updated for the next
+ encoding iteration.
+*/
+void WebRtcIsac_SplitAndFilterFloat(float* pin,
+                                    float* LP,
+                                    float* HP,
+                                    double* LP_la,
+                                    double* HP_la,
+                                    PreFiltBankstr* prefiltdata) {
+  int k, n;
+  float CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
+  float ForTransform_CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
+  float ForTransform_CompositeAPFilterState2[NUMBEROFCOMPOSITEAPSECTIONS];
+  float tempinoutvec[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
+  float tempin_ch1[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
+  float tempin_ch2[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
+  float in[FRAMESAMPLES];
+  float ftmp;
+
+  /* HPstcoeff_in = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
+  static const float kHpStCoefInFloat[4] = {
+      -1.94895953203325f, 0.94984516000000f, -0.05101826139794f,
+      0.05015484000000f};
+
+  /* The composite all-pass filter factors */
+  static const float WebRtcIsac_kCompositeApFactorsFloat[4] = {
+      0.03470000000000f, 0.15440000000000f, 0.38260000000000f,
+      0.74400000000000f};
+
+  // The matrix for transforming the backward composite state to upper channel
+  // state.
+  static const float WebRtcIsac_kTransform1Float[8] = {
+      -0.00158678506084f, 0.00127157815343f, -0.00104805672709f,
+      0.00084837248079f,  0.00134467983258f, -0.00107756549387f,
+      0.00088814793277f,  -0.00071893072525f};
+
+  // The matrix for transforming the backward composite state to lower channel
+  // state.
+  static const float WebRtcIsac_kTransform2Float[8] = {
+      -0.00170686041697f, 0.00136780109829f, -0.00112736532350f,
+      0.00091257055385f,  0.00103094281812f, -0.00082615076557f,
+      0.00068092756088f,  -0.00055119165484f};
+
+  /* High pass filter */
+
+  for (k = 0; k < FRAMESAMPLES; k++) {
+    in[k] = pin[k] + kHpStCoefInFloat[2] * prefiltdata->HPstates_float[0] +
+            kHpStCoefInFloat[3] * prefiltdata->HPstates_float[1];
+    ftmp = pin[k] - kHpStCoefInFloat[0] * prefiltdata->HPstates_float[0] -
+           kHpStCoefInFloat[1] * prefiltdata->HPstates_float[1];
+    prefiltdata->HPstates_float[1] = prefiltdata->HPstates_float[0];
+    prefiltdata->HPstates_float[0] = ftmp;
+  }
+
+  /* First Channel */
+
+  /*initial state of composite filter is zero */
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    CompositeAPFilterState[k] = 0.0;
+  }
+  /* put every other sample of input into a temporary vector in reverse
+   * (backward) order*/
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempinoutvec[k] = in[FRAMESAMPLES - 1 - 2 * k];
+  }
+
+  /* now all-pass filter the backwards vector.  Output values overwrite the
+   * input vector. */
+  WebRtcIsac_AllPassFilter2Float(
+      tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat, FRAMESAMPLES_HALF,
+      NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  /* save the backwards filtered output for later forward filtering,
+     but write it in forward order*/
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempin_ch1[FRAMESAMPLES_HALF + QLOOKAHEAD - 1 - k] = tempinoutvec[k];
+  }
+
+  /* save the backwards filter state  becaue it will be transformed
+     later into a forward state */
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    ForTransform_CompositeAPFilterState[k] = CompositeAPFilterState[k];
+  }
+
+  /* now backwards filter the samples in the lookahead buffer. The samples were
+     placed there in the encoding of the previous frame.  The output samples
+     overwrite the input samples */
+  WebRtcIsac_AllPassFilter2Float(
+      prefiltdata->INLABUF1_float, WebRtcIsac_kCompositeApFactorsFloat,
+      QLOOKAHEAD, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  /* save the output, but write it in forward order */
+  /* write the lookahead samples for the next encoding iteration. Every other
+     sample at the end of the input frame is written in reverse order for the
+     lookahead length. Exported in the prefiltdata structure. */
+  for (k = 0; k < QLOOKAHEAD; k++) {
+    tempin_ch1[QLOOKAHEAD - 1 - k] = prefiltdata->INLABUF1_float[k];
+    prefiltdata->INLABUF1_float[k] = in[FRAMESAMPLES - 1 - 2 * k];
+  }
+
+  /* Second Channel.  This is exactly like the first channel, except that the
+     even samples are now filtered instead (lower channel). */
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    CompositeAPFilterState[k] = 0.0;
+  }
+
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempinoutvec[k] = in[FRAMESAMPLES - 2 - 2 * k];
+  }
+
+  WebRtcIsac_AllPassFilter2Float(
+      tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat, FRAMESAMPLES_HALF,
+      NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempin_ch2[FRAMESAMPLES_HALF + QLOOKAHEAD - 1 - k] = tempinoutvec[k];
+  }
+
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    ForTransform_CompositeAPFilterState2[k] = CompositeAPFilterState[k];
+  }
+
+  WebRtcIsac_AllPassFilter2Float(
+      prefiltdata->INLABUF2_float, WebRtcIsac_kCompositeApFactorsFloat,
+      QLOOKAHEAD, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  for (k = 0; k < QLOOKAHEAD; k++) {
+    tempin_ch2[QLOOKAHEAD - 1 - k] = prefiltdata->INLABUF2_float[k];
+    prefiltdata->INLABUF2_float[k] = in[FRAMESAMPLES - 2 - 2 * k];
+  }
+
+  /* Transform filter states from backward to forward */
+  /*At this point, each of the states of the backwards composite filters for the
+    two channels are transformed into forward filtering states for the
+    corresponding forward channel filters.  Each channel's forward filtering
+    state from the previous
+    encoding iteration is added to the transformed state to get a proper forward
+    state */
+
+  /* So the existing NUMBEROFCOMPOSITEAPSECTIONS x 1 (4x1) state vector is
+     multiplied by a NUMBEROFCHANNELAPSECTIONSxNUMBEROFCOMPOSITEAPSECTIONS (2x4)
+     transform matrix to get the new state that is added to the previous 2x1
+     input state */
+
+  for (k = 0; k < NUMBEROFCHANNELAPSECTIONS; k++) { /* k is row variable */
+    for (n = 0; n < NUMBEROFCOMPOSITEAPSECTIONS;
+         n++) { /* n is column variable */
+      prefiltdata->INSTAT1_float[k] +=
+          ForTransform_CompositeAPFilterState[n] *
+          WebRtcIsac_kTransform1Float[k * NUMBEROFCHANNELAPSECTIONS + n];
+      prefiltdata->INSTAT2_float[k] +=
+          ForTransform_CompositeAPFilterState2[n] *
+          WebRtcIsac_kTransform2Float[k * NUMBEROFCHANNELAPSECTIONS + n];
+    }
+  }
+
+  /*obtain polyphase components by forward all-pass filtering through each
+   * channel */
+  /* the backward filtered samples are now forward filtered with the
+   * corresponding channel filters */
+  /* The all pass filtering automatically updates the filter states which are
+     exported in the prefiltdata structure */
+  WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kUpperApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTAT1_float);
+  WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kLowerApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTAT2_float);
+
+  /* Now Construct low-pass and high-pass signals as combinations of polyphase
+   * components */
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    LP[k] = 0.5f * (tempin_ch1[k] + tempin_ch2[k]); /* low pass signal*/
+    HP[k] = 0.5f * (tempin_ch1[k] - tempin_ch2[k]); /* high pass signal*/
+  }
+
+  /* Lookahead LP and HP signals */
+  /* now create low pass and high pass signals of the input vector.  However, no
+     backwards filtering is performed, and hence no phase equalization is
+     involved. Also, the input contains some samples that are lookahead samples.
+     The high pass and low pass signals that are created are used outside this
+     function for analysis (not encoding) purposes */
+
+  /* set up input */
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempin_ch1[k] = in[2 * k + 1];
+    tempin_ch2[k] = in[2 * k];
+  }
+
+  /* the input filter states are passed in and updated by the all-pass filtering
+     routine and exported in the prefiltdata structure*/
+  WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kUpperApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTATLA1_float);
+  WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kLowerApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTATLA2_float);
+
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    LP_la[k] = (float)(0.5f * (tempin_ch1[k] + tempin_ch2[k]));  /*low pass */
+    HP_la[k] = (double)(0.5f * (tempin_ch1[k] - tempin_ch2[k])); /* high pass */
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.h
new file mode 100644
index 0000000000..1aecfc4046
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
+
+#include <stddef.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata);
+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* state);
+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata);
+
+double WebRtcIsac_LevDurb(double* a, double* k, double* r, size_t order);
+
+/* The number of all-pass filter factors in an upper or lower channel*/
+#define NUMBEROFCHANNELAPSECTIONS 2
+
+/* The upper channel all-pass filter factors */
+extern const float WebRtcIsac_kUpperApFactorsFloat[2];
+
+/* The lower channel all-pass filter factors */
+extern const float WebRtcIsac_kLowerApFactorsFloat[2];
+
+void WebRtcIsac_AllPassFilter2Float(float* InOut,
+                                    const float* APSectionFactors,
+                                    int lengthInOut,
+                                    int NumberOfSections,
+                                    float* FilterState);
+void WebRtcIsac_SplitAndFilterFloat(float* in,
+                                    float* LP,
+                                    float* HP,
+                                    double* LP_la,
+                                    double* HP_la,
+                                    PreFiltBankstr* prefiltdata);
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
new file mode 100644
index 0000000000..fe9afa4ba2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+
+#include <math.h>
+
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_POSIX)
+#define WebRtcIsac_lrint lrint
+#elif (defined(WEBRTC_ARCH_X86) && defined(WIN32))
+static __inline long int WebRtcIsac_lrint(double x_dbl) {
+  long int x_int;
+
+  __asm {
+    fld x_dbl
+    fistp x_int
+  }
+  ;
+
+  return x_int;
+}
+#else  // Do a slow but correct implementation of lrint
+
+static __inline long int WebRtcIsac_lrint(double x_dbl) {
+  long int x_int;
+  x_int = (long int)floor(x_dbl + 0.499999999999);
+  return x_int;
+}
+
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
new file mode 100644
index 0000000000..8a19ac1710
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
@@ -0,0 +1,695 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+
+#include <math.h>
+#include <memory.h>
+#include <string.h>
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+
+#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h"
+#include "modules/audio_coding/codecs/isac/main/source/pitch_filter.h"
+#include "rtc_base/system/ignore_warnings.h"
+
+static const double kInterpolWin[8] = {-0.00067556028640,  0.02184247643159, -0.12203175715679,  0.60086484101160,
+                                       0.60086484101160, -0.12203175715679,  0.02184247643159, -0.00067556028640};
+
+/* interpolation filter */
+__inline static void IntrepolFilter(double *data_ptr, double *intrp)
+{
+  *intrp = kInterpolWin[0] * data_ptr[-3];
+  *intrp += kInterpolWin[1] * data_ptr[-2];
+  *intrp += kInterpolWin[2] * data_ptr[-1];
+  *intrp += kInterpolWin[3] * data_ptr[0];
+  *intrp += kInterpolWin[4] * data_ptr[1];
+  *intrp += kInterpolWin[5] * data_ptr[2];
+  *intrp += kInterpolWin[6] * data_ptr[3];
+  *intrp += kInterpolWin[7] * data_ptr[4];
+}
+
+
+/* 2D parabolic interpolation */
+/* probably some 0.5 factors can be eliminated, and the square-roots can be removed from the Cholesky fact. */
+__inline static void Intrpol2D(double T[3][3], double *x, double *y, double *peak_val)
+{
+  double c, b[2], A[2][2];
+  double t1, t2, d;
+  double delta1, delta2;
+
+
+  // double T[3][3] = {{-1.25, -.25,-.25}, {-.25, .75, .75}, {-.25, .75, .75}};
+  // should result in: delta1 = 0.5;  delta2 = 0.0;  peak_val = 1.0
+
+  c = T[1][1];
+  b[0] = 0.5 * (T[1][2] + T[2][1] - T[0][1] - T[1][0]);
+  b[1] = 0.5 * (T[1][0] + T[2][1] - T[0][1] - T[1][2]);
+  A[0][1] = -0.5 * (T[0][1] + T[2][1] - T[1][0] - T[1][2]);
+  t1 = 0.5 * (T[0][0] + T[2][2]) - c;
+  t2 = 0.5 * (T[2][0] + T[0][2]) - c;
+  d = (T[0][1] + T[1][2] + T[1][0] + T[2][1]) - 4.0 * c - t1 - t2;
+  A[0][0] = -t1 - 0.5 * d;
+  A[1][1] = -t2 - 0.5 * d;
+
+  /* deal with singularities or ill-conditioned cases */
+  if ( (A[0][0] < 1e-7) || ((A[0][0] * A[1][1] - A[0][1] * A[0][1]) < 1e-7) ) {
+    *peak_val = T[1][1];
+    return;
+  }
+
+  /* Cholesky decomposition: replace A by upper-triangular factor */
+  A[0][0] = sqrt(A[0][0]);
+  A[0][1] = A[0][1] / A[0][0];
+  A[1][1] = sqrt(A[1][1] - A[0][1] * A[0][1]);
+
+  /* compute [x; y] = -0.5 * inv(A) * b */
+  t1 = b[0] / A[0][0];
+  t2 = (b[1] - t1 * A[0][1]) / A[1][1];
+  delta2 = t2 / A[1][1];
+  delta1 = 0.5 * (t1 - delta2 * A[0][1]) / A[0][0];
+  delta2 *= 0.5;
+
+  /* limit norm */
+  t1 = delta1 * delta1 + delta2 * delta2;
+  if (t1 > 1.0) {
+    delta1 /= t1;
+    delta2 /= t1;
+  }
+
+  *peak_val = 0.5 * (b[0] * delta1 + b[1] * delta2) + c;
+
+  *x += delta1;
+  *y += delta2;
+}
+
+
+static void PCorr(const double *in, double *outcorr)
+{
+  double sum, ysum, prod;
+  const double *x, *inptr;
+  int k, n;
+
+  //ysum = 1e-6;          /* use this with float (i.s.o. double)! */
+  ysum = 1e-13;
+  sum = 0.0;
+  x = in + PITCH_MAX_LAG/2 + 2;
+  for (n = 0; n < PITCH_CORR_LEN2; n++) {
+    ysum += in[n] * in[n];
+    sum += x[n] * in[n];
+  }
+
+  outcorr += PITCH_LAG_SPAN2 - 1;     /* index of last element in array */
+  *outcorr = sum / sqrt(ysum);
+
+  for (k = 1; k < PITCH_LAG_SPAN2; k++) {
+    ysum -= in[k-1] * in[k-1];
+    ysum += in[PITCH_CORR_LEN2 + k - 1] * in[PITCH_CORR_LEN2 + k - 1];
+    sum = 0.0;
+    inptr = &in[k];
+    prod = x[0] * inptr[0];
+    for (n = 1; n < PITCH_CORR_LEN2; n++) {
+      sum += prod;
+      prod = x[n] * inptr[n];
+    }
+    sum += prod;
+    outcorr--;
+    *outcorr = sum / sqrt(ysum);
+  }
+}
+
+static void WebRtcIsac_AllpassFilterForDec(double* InOut,
+                                           const double* APSectionFactors,
+                                           size_t lengthInOut,
+                                           double* FilterState) {
+  // This performs all-pass filtering--a series of first order all-pass
+  // sections are used to filter the input in a cascade manner.
+  size_t n, j;
+  double temp;
+  for (j = 0; j < ALLPASSSECTIONS; j++) {
+    for (n = 0; n < lengthInOut; n += 2) {
+      temp = InOut[n];  // store input
+      InOut[n] = FilterState[j] + APSectionFactors[j] * temp;
+      FilterState[j] = -APSectionFactors[j] * InOut[n] + temp;
+    }
+  }
+}
+
+static void WebRtcIsac_DecimateAllpass(
+    const double* in,
+    double* state_in,  // array of size: 2*ALLPASSSECTIONS+1
+    size_t N,          // number of input samples
+    double* out) {     // array of size N/2
+
+  static const double APupper[ALLPASSSECTIONS] = {0.0347, 0.3826};
+  static const double APlower[ALLPASSSECTIONS] = {0.1544, 0.744};
+
+  size_t n;
+  double data_vec[PITCH_FRAME_LEN];
+
+  /* copy input */
+  memcpy(data_vec + 1, in, sizeof(double) * (N - 1));
+
+  data_vec[0] = state_in[2 * ALLPASSSECTIONS];  // the z^(-1) state
+  state_in[2 * ALLPASSSECTIONS] = in[N - 1];
+
+  WebRtcIsac_AllpassFilterForDec(data_vec + 1, APupper, N, state_in);
+  WebRtcIsac_AllpassFilterForDec(data_vec, APlower, N,
+                                 state_in + ALLPASSSECTIONS);
+
+  for (n = 0; n < N / 2; n++)
+    out[n] = data_vec[2 * n] + data_vec[2 * n + 1];
+}
+
+RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
+
+static void WebRtcIsac_InitializePitch(const double* in,
+                                       const double old_lag,
+                                       const double old_gain,
+                                       PitchAnalysisStruct* State,
+                                       double* lags) {
+  double buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
+  double ratio, log_lag, gain_bias;
+  double bias;
+  double corrvec1[PITCH_LAG_SPAN2];
+  double corrvec2[PITCH_LAG_SPAN2];
+  int m, k;
+  // Allocating 10 extra entries at the begining of the CorrSurf
+  double corrSurfBuff[10 + (2*PITCH_BW+3)*(PITCH_LAG_SPAN2+4)];
+  double* CorrSurf[2*PITCH_BW+3];
+  double *CorrSurfPtr1, *CorrSurfPtr2;
+  double LagWin[3] = {0.2, 0.5, 0.98};
+  int ind1, ind2, peaks_ind, peak, max_ind;
+  int peaks[PITCH_MAX_NUM_PEAKS];
+  double adj, gain_tmp;
+  double corr, corr_max;
+  double intrp_a, intrp_b, intrp_c, intrp_d;
+  double peak_vals[PITCH_MAX_NUM_PEAKS];
+  double lags1[PITCH_MAX_NUM_PEAKS];
+  double lags2[PITCH_MAX_NUM_PEAKS];
+  double T[3][3];
+  int row;
+
+  for(k = 0; k < 2*PITCH_BW+3; k++)
+  {
+    CorrSurf[k] = &corrSurfBuff[10 + k * (PITCH_LAG_SPAN2+4)];
+  }
+  /* reset CorrSurf matrix */
+  memset(corrSurfBuff, 0, sizeof(double) * (10 + (2*PITCH_BW+3) * (PITCH_LAG_SPAN2+4)));
+
+  //warnings -DH
+  max_ind = 0;
+  peak = 0;
+
+  /* copy old values from state buffer */
+  memcpy(buf_dec, State->dec_buffer, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
+
+  /* decimation; put result after the old values */
+  WebRtcIsac_DecimateAllpass(in, State->decimator_state, PITCH_FRAME_LEN,
+                             &buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2]);
+
+  /* low-pass filtering */
+  for (k = PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2; k++)
+    buf_dec[k] += 0.75 * buf_dec[k-1] - 0.25 * buf_dec[k-2];
+
+  /* copy end part back into state buffer */
+  memcpy(State->dec_buffer, buf_dec+PITCH_FRAME_LEN/2, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
+
+  /* compute correlation for first and second half of the frame */
+  PCorr(buf_dec, corrvec1);
+  PCorr(buf_dec + PITCH_CORR_STEP2, corrvec2);
+
+  /* bias towards pitch lag of previous frame */
+  log_lag = log(0.5 * old_lag);
+  gain_bias = 4.0 * old_gain * old_gain;
+  if (gain_bias > 0.8) gain_bias = 0.8;
+  for (k = 0; k < PITCH_LAG_SPAN2; k++)
+  {
+    ratio = log((double) (k + (PITCH_MIN_LAG/2-2))) - log_lag;
+    bias = 1.0 + gain_bias * exp(-5.0 * ratio * ratio);
+    corrvec1[k] *= bias;
+  }
+
+  /* taper correlation functions */
+  for (k = 0; k < 3; k++) {
+    gain_tmp = LagWin[k];
+    corrvec1[k] *= gain_tmp;
+    corrvec2[k] *= gain_tmp;
+    corrvec1[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
+    corrvec2[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
+  }
+
+  corr_max = 0.0;
+  /* fill middle row of correlation surface */
+  ind1 = 0;
+  ind2 = 0;
+  CorrSurfPtr1 = &CorrSurf[PITCH_BW][2];
+  for (k = 0; k < PITCH_LAG_SPAN2; k++) {
+    corr = corrvec1[ind1++] + corrvec2[ind2++];
+    CorrSurfPtr1[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+    }
+  }
+  /* fill first and last rows of correlation surface */
+  ind1 = 0;
+  ind2 = PITCH_BW;
+  CorrSurfPtr1 = &CorrSurf[0][2];
+  CorrSurfPtr2 = &CorrSurf[2*PITCH_BW][PITCH_BW+2];
+  for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW; k++) {
+    ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+    adj = 0.2 * ratio * (2.0 - ratio);   /* adjustment factor; inverse parabola as a function of ratio */
+    corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+    CorrSurfPtr1[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+    }
+    corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+    CorrSurfPtr2[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+    }
+  }
+  /* fill second and next to last rows of correlation surface */
+  ind1 = 0;
+  ind2 = PITCH_BW-1;
+  CorrSurfPtr1 = &CorrSurf[1][2];
+  CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-1][PITCH_BW+1];
+  for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+1; k++) {
+    ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+    adj = 0.9 * ratio * (2.0 - ratio);   /* adjustment factor; inverse parabola as a function of ratio */
+    corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+    CorrSurfPtr1[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+    }
+    corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+    CorrSurfPtr2[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+    }
+  }
+  /* fill remainder of correlation surface */
+  for (m = 2; m < PITCH_BW; m++) {
+    ind1 = 0;
+    ind2 = PITCH_BW - m;         /* always larger than ind1 */
+    CorrSurfPtr1 = &CorrSurf[m][2];
+    CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-m][PITCH_BW+2-m];
+    for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+m; k++) {
+      ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+      adj = ratio * (2.0 - ratio);    /* adjustment factor; inverse parabola as a function of ratio */
+      corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+      CorrSurfPtr1[k] = corr;
+      if (corr > corr_max) {
+        corr_max = corr;  /* update maximum */
+        max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+      }
+      corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+      CorrSurfPtr2[k] = corr;
+      if (corr > corr_max) {
+        corr_max = corr;  /* update maximum */
+        max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+      }
+    }
+  }
+
+  /* threshold value to qualify as a peak */
+  corr_max *= 0.6;
+
+  peaks_ind = 0;
+  /* find peaks */
+  for (m = 1; m < PITCH_BW+1; m++) {
+    if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+    CorrSurfPtr1 = &CorrSurf[m][2];
+    for (k = 2; k < PITCH_LAG_SPAN2-PITCH_BW-2+m; k++) {
+      corr = CorrSurfPtr1[k];
+      if (corr > corr_max) {
+        if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
+          if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
+            /* found a peak; store index into matrix */
+            peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+            if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+          }
+        }
+      }
+    }
+  }
+  for (m = PITCH_BW+1; m < 2*PITCH_BW; m++) {
+    if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+    CorrSurfPtr1 = &CorrSurf[m][2];
+    for (k = 2+m-PITCH_BW; k < PITCH_LAG_SPAN2-2; k++) {
+      corr = CorrSurfPtr1[k];
+      if (corr > corr_max) {
+        if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
+          if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
+            /* found a peak; store index into matrix */
+            peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+            if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+          }
+        }
+      }
+    }
+  }
+
+  if (peaks_ind > 0) {
+    /* examine each peak */
+    CorrSurfPtr1 = &CorrSurf[0][0];
+    for (k = 0; k < peaks_ind; k++) {
+      peak = peaks[k];
+
+      /* compute four interpolated values around current peak */
+      IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)], &intrp_a);
+      IntrepolFilter(&CorrSurfPtr1[peak - 1            ], &intrp_b);
+      IntrepolFilter(&CorrSurfPtr1[peak                ], &intrp_c);
+      IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)], &intrp_d);
+
+      /* determine maximum of the interpolated values */
+      corr = CorrSurfPtr1[peak];
+      corr_max = intrp_a;
+      if (intrp_b > corr_max) corr_max = intrp_b;
+      if (intrp_c > corr_max) corr_max = intrp_c;
+      if (intrp_d > corr_max) corr_max = intrp_d;
+
+      /* determine where the peak sits and fill a 3x3 matrix around it */
+      row = peak / (PITCH_LAG_SPAN2+4);
+      lags1[k] = (double) ((peak - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
+      lags2[k] = (double) (lags1[k] + PITCH_BW - row);
+      if ( corr > corr_max ) {
+        T[0][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+        T[2][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+        T[1][1] = corr;
+        T[0][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+        T[2][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+        T[1][0] = intrp_a;
+        T[0][1] = intrp_b;
+        T[2][1] = intrp_c;
+        T[1][2] = intrp_d;
+      } else {
+        if (intrp_a == corr_max) {
+          lags1[k] -= 0.5;
+          lags2[k] += 0.5;
+          IntrepolFilter(&CorrSurfPtr1[peak - 2*(PITCH_LAG_SPAN2+5)], &T[0][0]);
+          IntrepolFilter(&CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)], &T[2][0]);
+          T[1][1] = intrp_a;
+          T[0][2] = intrp_b;
+          T[2][2] = intrp_c;
+          T[1][0] = CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)];
+          T[0][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+          T[2][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+          T[1][2] = corr;
+        } else if (intrp_b == corr_max) {
+          lags1[k] -= 0.5;
+          lags2[k] -= 0.5;
+          IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+6)], &T[0][0]);
+          T[2][0] = intrp_a;
+          T[1][1] = intrp_b;
+          IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+3)], &T[0][2]);
+          T[2][2] = intrp_d;
+          T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+          T[0][1] = CorrSurfPtr1[peak - 1];
+          T[2][1] = corr;
+          T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+        } else if (intrp_c == corr_max) {
+          lags1[k] += 0.5;
+          lags2[k] += 0.5;
+          T[0][0] = intrp_a;
+          IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)], &T[2][0]);
+          T[1][1] = intrp_c;
+          T[0][2] = intrp_d;
+          IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)], &T[2][2]);
+          T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+          T[0][1] = corr;
+          T[2][1] = CorrSurfPtr1[peak + 1];
+          T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+        } else {
+          lags1[k] += 0.5;
+          lags2[k] -= 0.5;
+          T[0][0] = intrp_b;
+          T[2][0] = intrp_c;
+          T[1][1] = intrp_d;
+          IntrepolFilter(&CorrSurfPtr1[peak + 2*(PITCH_LAG_SPAN2+4)], &T[0][2]);
+          IntrepolFilter(&CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)], &T[2][2]);
+          T[1][0] = corr;
+          T[0][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+          T[2][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+          T[1][2] = CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)];
+        }
+      }
+
+      /* 2D parabolic interpolation gives more accurate lags and peak value */
+      Intrpol2D(T, &lags1[k], &lags2[k], &peak_vals[k]);
+    }
+
+    /* determine the highest peak, after applying a bias towards short lags */
+    corr_max = 0.0;
+    for (k = 0; k < peaks_ind; k++) {
+      corr = peak_vals[k] * pow(PITCH_PEAK_DECAY, log(lags1[k] + lags2[k]));
+      if (corr > corr_max) {
+        corr_max = corr;
+        peak = k;
+      }
+    }
+
+    lags1[peak] *= 2.0;
+    lags2[peak] *= 2.0;
+
+    if (lags1[peak] < (double) PITCH_MIN_LAG) lags1[peak] = (double) PITCH_MIN_LAG;
+    if (lags2[peak] < (double) PITCH_MIN_LAG) lags2[peak] = (double) PITCH_MIN_LAG;
+    if (lags1[peak] > (double) PITCH_MAX_LAG) lags1[peak] = (double) PITCH_MAX_LAG;
+    if (lags2[peak] > (double) PITCH_MAX_LAG) lags2[peak] = (double) PITCH_MAX_LAG;
+
+    /* store lags of highest peak in output array */
+    lags[0] = lags1[peak];
+    lags[1] = lags1[peak];
+    lags[2] = lags2[peak];
+    lags[3] = lags2[peak];
+  }
+  else
+  {
+    row = max_ind / (PITCH_LAG_SPAN2+4);
+    lags1[0] = (double) ((max_ind - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
+    lags2[0] = (double) (lags1[0] + PITCH_BW - row);
+
+    if (lags1[0] < (double) PITCH_MIN_LAG) lags1[0] = (double) PITCH_MIN_LAG;
+    if (lags2[0] < (double) PITCH_MIN_LAG) lags2[0] = (double) PITCH_MIN_LAG;
+    if (lags1[0] > (double) PITCH_MAX_LAG) lags1[0] = (double) PITCH_MAX_LAG;
+    if (lags2[0] > (double) PITCH_MAX_LAG) lags2[0] = (double) PITCH_MAX_LAG;
+
+    /* store lags of highest peak in output array */
+    lags[0] = lags1[0];
+    lags[1] = lags1[0];
+    lags[2] = lags2[0];
+    lags[3] = lags2[0];
+  }
+}
+
+RTC_POP_IGNORING_WFRAME_LARGER_THAN()
+
+/* create weighting matrix by orthogonalizing a basis of polynomials of increasing order
+ * t = (0:4)';
+ * A = [t.^0, t.^1, t.^2, t.^3, t.^4];
+ * [Q, dummy] = qr(A);
+ * P.Weight = Q * diag([0, .1, .5, 1, 1]) * Q'; */
+static const double kWeight[5][5] = {
+  { 0.29714285714286,  -0.30857142857143,  -0.05714285714286,   0.05142857142857,  0.01714285714286},
+  {-0.30857142857143,   0.67428571428571,  -0.27142857142857,  -0.14571428571429,  0.05142857142857},
+  {-0.05714285714286,  -0.27142857142857,   0.65714285714286,  -0.27142857142857, -0.05714285714286},
+  { 0.05142857142857,  -0.14571428571429,  -0.27142857142857,   0.67428571428571, -0.30857142857143},
+  { 0.01714285714286,   0.05142857142857,  -0.05714285714286,  -0.30857142857143,  0.29714285714286}
+};
+
+/* second order high-pass filter */
+static void WebRtcIsac_Highpass(const double* in,
+                         double* out,
+                         double* state,
+                         size_t N) {
+  /* create high-pass filter ocefficients
+   * z = 0.998 * exp(j*2*pi*35/8000);
+   * p = 0.94 * exp(j*2*pi*140/8000);
+   * HP_b = [1, -2*real(z), abs(z)^2];
+   * HP_a = [1, -2*real(p), abs(p)^2]; */
+  static const double a_coef[2] = { 1.86864659625574, -0.88360000000000};
+  static const double b_coef[2] = {-1.99524591718270,  0.99600400000000};
+
+  size_t k;
+
+  for (k=0; k<N; k++) {
+    *out = *in + state[1];
+    state[1] = state[0] + b_coef[0] * *in + a_coef[0] * *out;
+    state[0] = b_coef[1] * *in++ + a_coef[1] * *out++;
+  }
+}
+
+RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
+
+void WebRtcIsac_PitchAnalysis(const double *in,               /* PITCH_FRAME_LEN samples */
+                              double *out,                    /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+                              PitchAnalysisStruct *State,
+                              double *lags,
+                              double *gains)
+{
+  double HPin[PITCH_FRAME_LEN];
+  double Weighted[PITCH_FRAME_LEN];
+  double Whitened[PITCH_FRAME_LEN + QLOOKAHEAD];
+  double inbuf[PITCH_FRAME_LEN + QLOOKAHEAD];
+  double out_G[PITCH_FRAME_LEN + QLOOKAHEAD];          // could be removed by using out instead
+  double out_dG[4][PITCH_FRAME_LEN + QLOOKAHEAD];
+  double old_lag, old_gain;
+  double nrg_wht, tmp;
+  double Wnrg, Wfluct, Wgain;
+  double H[4][4];
+  double grad[4];
+  double dG[4];
+  int k, m, n, iter;
+
+  /* high pass filtering using second order pole-zero filter */
+  WebRtcIsac_Highpass(in, HPin, State->hp_state, PITCH_FRAME_LEN);
+
+  /* copy from state into buffer */
+  memcpy(Whitened, State->whitened_buf, sizeof(double) * QLOOKAHEAD);
+
+  /* compute weighted and whitened signals */
+  WebRtcIsac_WeightingFilter(HPin, &Weighted[0], &Whitened[QLOOKAHEAD], &(State->Wghtstr));
+
+  /* copy from buffer into state */
+  memcpy(State->whitened_buf, Whitened+PITCH_FRAME_LEN, sizeof(double) * QLOOKAHEAD);
+
+  old_lag = State->PFstr_wght.oldlagp[0];
+  old_gain = State->PFstr_wght.oldgainp[0];
+
+  /* inital pitch estimate */
+  WebRtcIsac_InitializePitch(Weighted, old_lag, old_gain, State, lags);
+
+
+  /* Iterative optimization of lags - to be done */
+
+  /* compute energy of whitened signal */
+  nrg_wht = 0.0;
+  for (k = 0; k < PITCH_FRAME_LEN + QLOOKAHEAD; k++)
+    nrg_wht += Whitened[k] * Whitened[k];
+
+
+  /* Iterative optimization of gains */
+
+  /* set weights for energy, gain fluctiation, and spectral gain penalty functions */
+  Wnrg = 1.0 / nrg_wht;
+  Wgain = 0.005;
+  Wfluct = 3.0;
+
+  /* set initial gains */
+  for (k = 0; k < 4; k++)
+    gains[k] = PITCH_MAX_GAIN_06;
+
+  /* two iterations should be enough */
+  for (iter = 0; iter < 2; iter++) {
+    /* compute Jacobian of pre-filter output towards gains */
+    WebRtcIsac_PitchfilterPre_gains(Whitened, out_G, out_dG, &(State->PFstr_wght), lags, gains);
+
+    /* gradient and approximate Hessian (lower triangle) for minimizing the filter's output power */
+    for (k = 0; k < 4; k++) {
+      tmp = 0.0;
+      for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
+        tmp += out_G[n] * out_dG[k][n];
+      grad[k] = tmp * Wnrg;
+    }
+    for (k = 0; k < 4; k++) {
+      for (m = 0; m <= k; m++) {
+        tmp = 0.0;
+        for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
+          tmp += out_dG[m][n] * out_dG[k][n];
+        H[k][m] = tmp * Wnrg;
+      }
+    }
+
+    /* add gradient and Hessian (lower triangle) for dampening fast gain changes */
+    for (k = 0; k < 4; k++) {
+      tmp = kWeight[k+1][0] * old_gain;
+      for (m = 0; m < 4; m++)
+        tmp += kWeight[k+1][m+1] * gains[m];
+      grad[k] += tmp * Wfluct;
+    }
+    for (k = 0; k < 4; k++) {
+      for (m = 0; m <= k; m++) {
+        H[k][m] += kWeight[k+1][m+1] * Wfluct;
+      }
+    }
+
+    /* add gradient and Hessian for dampening gain */
+    for (k = 0; k < 3; k++) {
+      tmp = 1.0 / (1 - gains[k]);
+      grad[k] += tmp * tmp * Wgain;
+      H[k][k] += 2.0 * tmp * (tmp * tmp * Wgain);
+    }
+    tmp = 1.0 / (1 - gains[3]);
+    grad[3] += 1.33 * (tmp * tmp * Wgain);
+    H[3][3] += 2.66 * tmp * (tmp * tmp * Wgain);
+
+
+    /* compute Cholesky factorization of Hessian
+     * by overwritting the upper triangle; scale factors on diagonal
+     * (for non pc-platforms store the inverse of the diagonals seperately to minimize divisions) */
+    H[0][1] = H[1][0] / H[0][0];
+    H[0][2] = H[2][0] / H[0][0];
+    H[0][3] = H[3][0] / H[0][0];
+    H[1][1] -= H[0][0] * H[0][1] * H[0][1];
+    H[1][2] = (H[2][1] - H[0][1] * H[2][0]) / H[1][1];
+    H[1][3] = (H[3][1] - H[0][1] * H[3][0]) / H[1][1];
+    H[2][2] -= H[0][0] * H[0][2] * H[0][2] + H[1][1] * H[1][2] * H[1][2];
+    H[2][3] = (H[3][2] - H[0][2] * H[3][0] - H[1][2] * H[1][1] * H[1][3]) / H[2][2];
+    H[3][3] -= H[0][0] * H[0][3] * H[0][3] + H[1][1] * H[1][3] * H[1][3] + H[2][2] * H[2][3] * H[2][3];
+
+    /* Compute update as  delta_gains = -inv(H) * grad */
+    /* copy and negate */
+    for (k = 0; k < 4; k++)
+      dG[k] = -grad[k];
+    /* back substitution */
+    dG[1] -= dG[0] * H[0][1];
+    dG[2] -= dG[0] * H[0][2] + dG[1] * H[1][2];
+    dG[3] -= dG[0] * H[0][3] + dG[1] * H[1][3] + dG[2] * H[2][3];
+    /* scale */
+    for (k = 0; k < 4; k++)
+      dG[k] /= H[k][k];
+    /* back substitution */
+    dG[2] -= dG[3] * H[2][3];
+    dG[1] -= dG[3] * H[1][3] + dG[2] * H[1][2];
+    dG[0] -= dG[3] * H[0][3] + dG[2] * H[0][2] + dG[1] * H[0][1];
+
+    /* update gains and check range */
+    for (k = 0; k < 4; k++) {
+      gains[k] += dG[k];
+      if (gains[k] > PITCH_MAX_GAIN)
+        gains[k] = PITCH_MAX_GAIN;
+      else if (gains[k] < 0.0)
+        gains[k] = 0.0;
+    }
+  }
+
+  /* update state for next frame */
+  WebRtcIsac_PitchfilterPre(Whitened, out, &(State->PFstr_wght), lags, gains);
+
+  /* concatenate previous input's end and current input */
+  memcpy(inbuf, State->inbuf, sizeof(double) * QLOOKAHEAD);
+  memcpy(inbuf+QLOOKAHEAD, in, sizeof(double) * PITCH_FRAME_LEN);
+
+  /* lookahead pitch filtering for masking analysis */
+  WebRtcIsac_PitchfilterPre_la(inbuf, out, &(State->PFstr), lags, gains);
+
+  /* store last part of input */
+  for (k = 0; k < QLOOKAHEAD; k++)
+    State->inbuf[k] = inbuf[k + PITCH_FRAME_LEN];
+}
+
+RTC_POP_IGNORING_WFRAME_LARGER_THAN()
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
new file mode 100644
index 0000000000..4ab78c20ad
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_estimator.h
+ *
+ * Pitch functions
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_PitchAnalysis(
+    const double* in, /* PITCH_FRAME_LEN samples */
+    double* out,      /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+    PitchAnalysisStruct* State,
+    double* lags,
+    double* gains);
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
new file mode 100644
index 0000000000..bf03dfff2e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
@@ -0,0 +1,388 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+#include <memory.h>
+#include <stdlib.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
+#include "rtc_base/compile_assert_c.h"
+
+/*
+ * We are implementing the following filters;
+ *
+ * Pre-filtering:
+ *   y(z) = x(z) + damper(z) * gain * (x(z) + y(z)) * z ^ (-lag);
+ *
+ * Post-filtering:
+ *   y(z) = x(z) - damper(z) * gain * (x(z) + y(z)) * z ^ (-lag);
+ *
+ * Note that `lag` is a floating number so we perform an interpolation to
+ * obtain the correct `lag`.
+ *
+ */
+
+static const double kDampFilter[PITCH_DAMPORDER] = {-0.07, 0.25, 0.64, 0.25,
+    -0.07};
+
+/* interpolation coefficients; generated by design_pitch_filter.m */
+static const double kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
+    {-0.02239172458614,  0.06653315052934, -0.16515880017569,  0.60701333734125,
+     0.64671399919202, -0.20249000396417,  0.09926548334755, -0.04765933793109,
+     0.01754159521746},
+    {-0.01985640750434,  0.05816126837866, -0.13991265473714,  0.44560418147643,
+     0.79117042386876, -0.20266133815188,  0.09585268418555, -0.04533310458084,
+     0.01654127246314},
+    {-0.01463300534216,  0.04229888475060, -0.09897034715253,  0.28284326017787,
+     0.90385267956632, -0.16976950138649,  0.07704272393639, -0.03584218578311,
+     0.01295781500709},
+    {-0.00764851320885,  0.02184035544377, -0.04985561057281,  0.13083306574393,
+     0.97545011664662, -0.10177807997561,  0.04400901776474, -0.02010737175166,
+     0.00719783432422},
+    {-0.00000000000000,  0.00000000000000, -0.00000000000001,  0.00000000000001,
+     0.99999999999999,  0.00000000000001, -0.00000000000001,  0.00000000000000,
+     -0.00000000000000},
+    {0.00719783432422, -0.02010737175166,  0.04400901776474, -0.10177807997562,
+     0.97545011664663,  0.13083306574393, -0.04985561057280,  0.02184035544377,
+     -0.00764851320885},
+    {0.01295781500710, -0.03584218578312,  0.07704272393640, -0.16976950138650,
+     0.90385267956634,  0.28284326017785, -0.09897034715252,  0.04229888475059,
+     -0.01463300534216},
+    {0.01654127246315, -0.04533310458085,  0.09585268418557, -0.20266133815190,
+     0.79117042386878,  0.44560418147640, -0.13991265473712,  0.05816126837865,
+     -0.01985640750433}
+};
+
+/*
+ * Enumerating the operation of the filter.
+ * iSAC has 4 different pitch-filter which are very similar in their structure.
+ *
+ * kPitchFilterPre     : In this mode the filter is operating as pitch
+ *                       pre-filter. This is used at the encoder.
+ * kPitchFilterPost    : In this mode the filter is operating as pitch
+ *                       post-filter. This is the inverse of pre-filter and used
+ *                       in the decoder.
+ * kPitchFilterPreLa   : This is, in structure, similar to pre-filtering but
+ *                       utilizing 3 millisecond lookahead. It is used to
+ *                       obtain the signal for LPC analysis.
+ * kPitchFilterPreGain : This is, in structure, similar to pre-filtering but
+ *                       differential changes in gain is considered. This is
+ *                       used to find the optimal gain.
+ */
+typedef enum {
+  kPitchFilterPre, kPitchFilterPost, kPitchFilterPreLa, kPitchFilterPreGain
+} PitchFilterOperation;
+
+/*
+ * Structure with parameters used for pitch-filtering.
+ * buffer           : a buffer where the sum of previous inputs and outputs
+ *                    are stored.
+ * damper_state     : the state of the damping filter. The filter is defined by
+ *                    `kDampFilter`.
+ * interpol_coeff   : pointer to a set of coefficient which are used to utilize
+ *                    fractional pitch by interpolation.
+ * gain             : pitch-gain to be applied to the current segment of input.
+ * lag              : pitch-lag for the current segment of input.
+ * lag_offset       : the offset of lag w.r.t. current sample.
+ * sub_frame        : sub-frame index, there are 4 pitch sub-frames in an iSAC
+ *                    frame.
+ *                    This specifies the usage of the filter. See
+ *                    'PitchFilterOperation' for operational modes.
+ * num_samples      : number of samples to be processed in each segment.
+ * index            : index of the input and output sample.
+ * damper_state_dg  : state of damping filter for different trial gains.
+ * gain_mult        : differential changes to gain.
+ */
+typedef struct {
+  double buffer[PITCH_INTBUFFSIZE + QLOOKAHEAD];
+  double damper_state[PITCH_DAMPORDER];
+  const double *interpol_coeff;
+  double gain;
+  double lag;
+  int lag_offset;
+
+  int sub_frame;
+  PitchFilterOperation mode;
+  int num_samples;
+  int index;
+
+  double damper_state_dg[4][PITCH_DAMPORDER];
+  double gain_mult[4];
+} PitchFilterParam;
+
+/**********************************************************************
+ * FilterSegment()
+ * Filter one segment, a quarter of a frame.
+ *
+ * Inputs
+ *   in_data      : pointer to the input signal of 30 ms at 8 kHz sample-rate.
+ *   filter_param : pitch filter parameters.
+ *
+ * Outputs
+ *   out_data     : pointer to a buffer where the filtered signal is written to.
+ *   out_dg       : [only used in kPitchFilterPreGain] pointer to a buffer
+ *                  where the output of different gain values (differential
+ *                  change to gain) is written.
+ */
+static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
+                          double* out_data,
+                          double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD]) {
+  int n;
+  int m;
+  int j;
+  double sum;
+  double sum2;
+  /* Index of `parameters->buffer` where the output is written to. */
+  int pos = parameters->index + PITCH_BUFFSIZE;
+  /* Index of `parameters->buffer` where samples are read for fractional-lag
+   * computation. */
+  int pos_lag = pos - parameters->lag_offset;
+
+  for (n = 0; n < parameters->num_samples; ++n) {
+    /* Shift low pass filter states. */
+    for (m = PITCH_DAMPORDER - 1; m > 0; --m) {
+      parameters->damper_state[m] = parameters->damper_state[m - 1];
+    }
+    /* Filter to get fractional pitch. */
+    sum = 0.0;
+    for (m = 0; m < PITCH_FRACORDER; ++m) {
+      sum += parameters->buffer[pos_lag + m] * parameters->interpol_coeff[m];
+    }
+    /* Multiply with gain. */
+    parameters->damper_state[0] = parameters->gain * sum;
+
+    if (parameters->mode == kPitchFilterPreGain) {
+      int lag_index = parameters->index - parameters->lag_offset;
+      int m_tmp = (lag_index < 0) ? -lag_index : 0;
+      /* Update the damper state for the new sample. */
+      for (m = PITCH_DAMPORDER - 1; m > 0; --m) {
+        for (j = 0; j < 4; ++j) {
+          parameters->damper_state_dg[j][m] =
+              parameters->damper_state_dg[j][m - 1];
+        }
+      }
+
+      for (j = 0; j < parameters->sub_frame + 1; ++j) {
+        /* Filter for fractional pitch. */
+        sum2 = 0.0;
+        for (m = PITCH_FRACORDER-1; m >= m_tmp; --m) {
+          /* `lag_index + m` is always larger than or equal to zero, see how
+           * m_tmp is computed. This is equivalent to assume samples outside
+           * `out_dg[j]` are zero. */
+          sum2 += out_dg[j][lag_index + m] * parameters->interpol_coeff[m];
+        }
+        /* Add the contribution of differential gain change. */
+        parameters->damper_state_dg[j][0] = parameters->gain_mult[j] * sum +
+            parameters->gain * sum2;
+      }
+
+      /* Filter with damping filter, and store the results. */
+      for (j = 0; j < parameters->sub_frame + 1; ++j) {
+        sum = 0.0;
+        for (m = 0; m < PITCH_DAMPORDER; ++m) {
+          sum -= parameters->damper_state_dg[j][m] * kDampFilter[m];
+        }
+        out_dg[j][parameters->index] = sum;
+      }
+    }
+    /* Filter with damping filter. */
+    sum = 0.0;
+    for (m = 0; m < PITCH_DAMPORDER; ++m) {
+      sum += parameters->damper_state[m] * kDampFilter[m];
+    }
+
+    /* Subtract from input and update buffer. */
+    out_data[parameters->index] = in_data[parameters->index] - sum;
+    parameters->buffer[pos] = in_data[parameters->index] +
+        out_data[parameters->index];
+
+    ++parameters->index;
+    ++pos;
+    ++pos_lag;
+  }
+  return;
+}
+
+/* Update filter parameters based on the pitch-gains and pitch-lags. */
+static void Update(PitchFilterParam* parameters) {
+  double fraction;
+  int fraction_index;
+  /* Compute integer lag-offset. */
+  parameters->lag_offset = WebRtcIsac_lrint(parameters->lag + PITCH_FILTDELAY +
+                                            0.5);
+  /* Find correct set of coefficients for computing fractional pitch. */
+  fraction = parameters->lag_offset - (parameters->lag + PITCH_FILTDELAY);
+  fraction_index = WebRtcIsac_lrint(PITCH_FRACS * fraction - 0.5);
+  parameters->interpol_coeff = kIntrpCoef[fraction_index];
+
+  if (parameters->mode == kPitchFilterPreGain) {
+    /* If in this mode make a differential change to pitch gain. */
+    parameters->gain_mult[parameters->sub_frame] += 0.2;
+    if (parameters->gain_mult[parameters->sub_frame] > 1.0) {
+      parameters->gain_mult[parameters->sub_frame] = 1.0;
+    }
+    if (parameters->sub_frame > 0) {
+      parameters->gain_mult[parameters->sub_frame - 1] -= 0.2;
+    }
+  }
+}
+
+/******************************************************************************
+ * FilterFrame()
+ * Filter a frame of 30 millisecond, given pitch-lags and pitch-gains.
+ *
+ * Inputs
+ *   in_data     : pointer to the input signal of 30 ms at 8 kHz sample-rate.
+ *   lags        : pointer to pitch-lags, 4 lags per frame.
+ *   gains       : pointer to pitch-gians, 4 gains per frame.
+ *   mode        : defining the functionality of the filter. It takes the
+ *                 following values.
+ *                 kPitchFilterPre:     Pitch pre-filter, used at encoder.
+ *                 kPitchFilterPost:    Pitch post-filter, used at decoder.
+ *                 kPitchFilterPreLa:   Pitch pre-filter with lookahead.
+ *                 kPitchFilterPreGain: Pitch pre-filter used to otain optimal
+ *                                      pitch-gains.
+ *
+ * Outputs
+ *   out_data    : pointer to a buffer where the filtered signal is written to.
+ *   out_dg      : [only used in kPitchFilterPreGain] pointer to a buffer
+ *                 where the output of different gain values (differential
+ *                 change to gain) is written.
+ */
+static void FilterFrame(const double* in_data, PitchFiltstr* filter_state,
+                        double* lags, double* gains, PitchFilterOperation mode,
+                        double* out_data,
+                        double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD]) {
+  PitchFilterParam filter_parameters;
+  double gain_delta, lag_delta;
+  double old_lag, old_gain;
+  int n;
+  int m;
+  const double kEnhancer = 1.3;
+
+  /* Set up buffer and states. */
+  filter_parameters.index = 0;
+  filter_parameters.lag_offset = 0;
+  filter_parameters.mode = mode;
+  /* Copy states to local variables. */
+  memcpy(filter_parameters.buffer, filter_state->ubuf,
+         sizeof(filter_state->ubuf));
+  RTC_COMPILE_ASSERT(sizeof(filter_parameters.buffer) >=
+                 sizeof(filter_state->ubuf));
+  memset(filter_parameters.buffer +
+             sizeof(filter_state->ubuf) / sizeof(filter_state->ubuf[0]),
+         0, sizeof(filter_parameters.buffer) - sizeof(filter_state->ubuf));
+  memcpy(filter_parameters.damper_state, filter_state->ystate,
+         sizeof(filter_state->ystate));
+
+  if (mode == kPitchFilterPreGain) {
+    /* Clear buffers. */
+    memset(filter_parameters.gain_mult, 0, sizeof(filter_parameters.gain_mult));
+    memset(filter_parameters.damper_state_dg, 0,
+           sizeof(filter_parameters.damper_state_dg));
+    for (n = 0; n < PITCH_SUBFRAMES; ++n) {
+      //memset(out_dg[n], 0, sizeof(double) * (PITCH_FRAME_LEN + QLOOKAHEAD));
+      memset(out_dg[n], 0, sizeof(out_dg[n]));
+    }
+  } else if (mode == kPitchFilterPost) {
+    /* Make output more periodic. Negative sign is to change the structure
+     * of the filter. */
+    for (n = 0; n < PITCH_SUBFRAMES; ++n) {
+      gains[n] *= -kEnhancer;
+    }
+  }
+
+  old_lag = *filter_state->oldlagp;
+  old_gain = *filter_state->oldgainp;
+
+  /* No interpolation if pitch lag step is big. */
+  if ((lags[0] > (PITCH_UPSTEP * old_lag)) ||
+      (lags[0] < (PITCH_DOWNSTEP * old_lag))) {
+    old_lag = lags[0];
+    old_gain = gains[0];
+
+    if (mode == kPitchFilterPreGain) {
+      filter_parameters.gain_mult[0] = 1.0;
+    }
+  }
+
+  filter_parameters.num_samples = PITCH_UPDATE;
+  for (m = 0; m < PITCH_SUBFRAMES; ++m) {
+    /* Set the sub-frame value. */
+    filter_parameters.sub_frame = m;
+    /* Calculate interpolation steps for pitch-lag and pitch-gain. */
+    lag_delta = (lags[m] - old_lag) / PITCH_GRAN_PER_SUBFRAME;
+    filter_parameters.lag = old_lag;
+    gain_delta = (gains[m] - old_gain) / PITCH_GRAN_PER_SUBFRAME;
+    filter_parameters.gain = old_gain;
+    /* Store for the next sub-frame. */
+    old_lag = lags[m];
+    old_gain = gains[m];
+
+    for (n = 0; n < PITCH_GRAN_PER_SUBFRAME; ++n) {
+      /* Step-wise interpolation of pitch gains and lags. As pitch-lag changes,
+       * some parameters of filter need to be update. */
+      filter_parameters.gain += gain_delta;
+      filter_parameters.lag += lag_delta;
+      /* Update parameters according to new lag value. */
+      Update(&filter_parameters);
+      /* Filter a segment of input. */
+      FilterSegment(in_data, &filter_parameters, out_data, out_dg);
+    }
+  }
+
+  if (mode != kPitchFilterPreGain) {
+    /* Export buffer and states. */
+    memcpy(filter_state->ubuf, &filter_parameters.buffer[PITCH_FRAME_LEN],
+           sizeof(filter_state->ubuf));
+    memcpy(filter_state->ystate, filter_parameters.damper_state,
+           sizeof(filter_state->ystate));
+
+    /* Store for the next frame. */
+    *filter_state->oldlagp = old_lag;
+    *filter_state->oldgainp = old_gain;
+  }
+
+  if ((mode == kPitchFilterPreGain) || (mode == kPitchFilterPreLa)) {
+    /* Filter the lookahead segment, this is treated as the last sub-frame. So
+     * set `pf_param` to last sub-frame. */
+    filter_parameters.sub_frame = PITCH_SUBFRAMES - 1;
+    filter_parameters.num_samples = QLOOKAHEAD;
+    FilterSegment(in_data, &filter_parameters, out_data, out_dg);
+  }
+}
+
+void WebRtcIsac_PitchfilterPre(double* in_data, double* out_data,
+                               PitchFiltstr* pf_state, double* lags,
+                               double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPre, out_data, NULL);
+}
+
+void WebRtcIsac_PitchfilterPre_la(double* in_data, double* out_data,
+                                  PitchFiltstr* pf_state, double* lags,
+                                  double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPreLa, out_data,
+              NULL);
+}
+
+void WebRtcIsac_PitchfilterPre_gains(
+    double* in_data, double* out_data,
+    double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD], PitchFiltstr *pf_state,
+    double* lags, double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPreGain, out_data,
+              out_dg);
+}
+
+void WebRtcIsac_PitchfilterPost(double* in_data, double* out_data,
+                                PitchFiltstr* pf_state, double* lags,
+                                double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPost, out_data, NULL);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.h
new file mode 100644
index 0000000000..9a232de87b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_PitchfilterPre(double* indat,
+                               double* outdat,
+                               PitchFiltstr* pfp,
+                               double* lags,
+                               double* gains);
+
+void WebRtcIsac_PitchfilterPost(double* indat,
+                                double* outdat,
+                                PitchFiltstr* pfp,
+                                double* lags,
+                                double* gains);
+
+void WebRtcIsac_PitchfilterPre_la(double* indat,
+                                  double* outdat,
+                                  PitchFiltstr* pfp,
+                                  double* lags,
+                                  double* gains);
+
+void WebRtcIsac_PitchfilterPre_gains(
+    double* indat,
+    double* outdat,
+    double out_dG[][PITCH_FRAME_LEN + QLOOKAHEAD],
+    PitchFiltstr* pfp,
+    double* lags,
+    double* gains);
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/settings.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/settings.h
new file mode 100644
index 0000000000..abce90c4f5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/settings.h
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * settings.h
+ *
+ * Declaration of #defines used in the iSAC codec
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+
+/* sampling frequency (Hz) */
+#define FS 16000
+
+/* number of samples per frame (either 320 (20ms), 480 (30ms) or 960 (60ms)) */
+#define INITIAL_FRAMESAMPLES 960
+
+/* do not modify the following; this will have to be modified if we
+ * have a 20ms framesize option */
+/**********************************************************************/
+/* miliseconds */
+#define FRAMESIZE 30
+/* number of samples per frame processed in the encoder, 480 */
+#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
+#define FRAMESAMPLES_HALF 240
+#define FRAMESAMPLES_QUARTER 120
+/**********************************************************************/
+
+/* max number of samples per frame (= 60 ms frame) */
+#define MAX_FRAMESAMPLES 960
+#define MAX_SWBFRAMESAMPLES (MAX_FRAMESAMPLES * 2)
+/* number of samples per 10ms frame */
+#define FRAMESAMPLES_10ms ((10 * FS) / 1000)
+#define SWBFRAMESAMPLES_10ms (FRAMESAMPLES_10ms * 2)
+/* number of samples in 30 ms frame */
+#define FRAMESAMPLES_30ms 480
+/* number of subframes */
+#define SUBFRAMES 6
+/* length of a subframe */
+#define UPDATE 80
+/* length of half a subframe (low/high band) */
+#define HALF_SUBFRAMELEN (UPDATE / 2)
+/* samples of look ahead (in a half-band, so actually
+ * half the samples of look ahead @ FS) */
+#define QLOOKAHEAD 24 /* 3 ms */
+/* order of AR model in spectral entropy coder */
+#define AR_ORDER 6
+/* order of LP model in spectral entropy coder */
+#define LP_ORDER 0
+
+/* window length (masking analysis) */
+#define WINLEN 256
+/* order of low-band pole filter used to approximate masking curve */
+#define ORDERLO 12
+/* order of hi-band pole filter used to approximate masking curve */
+#define ORDERHI 6
+
+#define UB_LPC_ORDER 4
+#define UB_LPC_VEC_PER_FRAME 2
+#define UB16_LPC_VEC_PER_FRAME 4
+#define UB_ACTIVE_SUBFRAMES 2
+#define UB_MAX_LPC_ORDER 6
+#define UB_INTERPOL_SEGMENTS 1
+#define UB16_INTERPOL_SEGMENTS 3
+#define LB_TOTAL_DELAY_SAMPLES 48
+enum ISACBandwidth { isac8kHz = 8, isac12kHz = 12, isac16kHz = 16 };
+enum ISACBand {
+  kIsacLowerBand = 0,
+  kIsacUpperBand12 = 1,
+  kIsacUpperBand16 = 2
+};
+enum IsacSamplingRate { kIsacWideband = 16, kIsacSuperWideband = 32 };
+#define UB_LPC_GAIN_DIM SUBFRAMES
+#define FB_STATE_SIZE_WORD32 6
+
+/* order for post_filter_bank */
+#define POSTQORDER 3
+/* order for pre-filterbank */
+#define QORDER 3
+/* another order */
+#define QORDER_ALL (POSTQORDER + QORDER - 1)
+/* for decimator */
+#define ALLPASSSECTIONS 2
+
+/* array size for byte stream in number of bytes. */
+/* The old maximum size still needed for the decoding */
+#define STREAM_SIZE_MAX 600
+#define STREAM_SIZE_MAX_30 200 /* 200 bytes=53.4 kbps @ 30 ms.framelength */
+#define STREAM_SIZE_MAX_60 400 /* 400 bytes=53.4 kbps @ 60 ms.framelength */
+
+/* storage size for bit counts */
+#define BIT_COUNTER_SIZE 30
+/* maximum order of any AR model or filter */
+#define MAX_AR_MODEL_ORDER 12  // 50
+
+/* For pitch analysis */
+#define PITCH_FRAME_LEN (FRAMESAMPLES_HALF) /* 30 ms  */
+#define PITCH_MAX_LAG 140                   /* 57 Hz  */
+#define PITCH_MIN_LAG 20                    /* 400 Hz */
+#define PITCH_MAX_GAIN 0.45
+#define PITCH_MAX_GAIN_06 0.27 /* PITCH_MAX_GAIN*0.6 */
+#define PITCH_MAX_GAIN_Q12 1843
+#define PITCH_LAG_SPAN2 (PITCH_MAX_LAG / 2 - PITCH_MIN_LAG / 2 + 5)
+#define PITCH_CORR_LEN2 60 /* 15 ms  */
+#define PITCH_CORR_STEP2 (PITCH_FRAME_LEN / 4)
+#define PITCH_BW 11 /* half the band width of correlation surface */
+#define PITCH_SUBFRAMES 4
+#define PITCH_GRAN_PER_SUBFRAME 5
+#define PITCH_SUBFRAME_LEN (PITCH_FRAME_LEN / PITCH_SUBFRAMES)
+#define PITCH_UPDATE (PITCH_SUBFRAME_LEN / PITCH_GRAN_PER_SUBFRAME)
+/* maximum number of peaks to be examined in correlation surface */
+#define PITCH_MAX_NUM_PEAKS 10
+#define PITCH_PEAK_DECAY 0.85
+/* For weighting filter */
+#define PITCH_WLPCORDER 6
+#define PITCH_WLPCWINLEN PITCH_FRAME_LEN
+#define PITCH_WLPCASYM 0.3 /* asymmetry parameter */
+#define PITCH_WLPCBUFLEN PITCH_WLPCWINLEN
+/* For pitch filter */
+/* Extra 50 for fraction and LP filters */
+#define PITCH_BUFFSIZE (PITCH_MAX_LAG + 50)
+#define PITCH_INTBUFFSIZE (PITCH_FRAME_LEN + PITCH_BUFFSIZE)
+/* Max rel. step for interpolation */
+#define PITCH_UPSTEP 1.5
+/* Max rel. step for interpolation */
+#define PITCH_DOWNSTEP 0.67
+#define PITCH_FRACS 8
+#define PITCH_FRACORDER 9
+#define PITCH_DAMPORDER 5
+#define PITCH_FILTDELAY 1.5f
+/* stepsize for quantization of the pitch Gain */
+#define PITCH_GAIN_STEPSIZE 0.125
+
+/* Order of high pass filter */
+#define HPORDER 2
+
+/* some mathematical constants */
+/* log2(exp) */
+#define LOG2EXP 1.44269504088896
+#define PI 3.14159265358979
+
+/* Maximum number of iterations allowed to limit payload size */
+#define MAX_PAYLOAD_LIMIT_ITERATION 5
+
+/* Redundant Coding */
+#define RCU_BOTTLENECK_BPS 16000
+#define RCU_TRANSCODING_SCALE 0.40f
+#define RCU_TRANSCODING_SCALE_INVERSE 2.5f
+
+#define RCU_TRANSCODING_SCALE_UB 0.50f
+#define RCU_TRANSCODING_SCALE_UB_INVERSE 2.0f
+
+/* Define Error codes */
+/* 6000 General */
+#define ISAC_MEMORY_ALLOCATION_FAILED 6010
+#define ISAC_MODE_MISMATCH 6020
+#define ISAC_DISALLOWED_BOTTLENECK 6030
+#define ISAC_DISALLOWED_FRAME_LENGTH 6040
+#define ISAC_UNSUPPORTED_SAMPLING_FREQUENCY 6050
+
+/* 6200 Bandwidth estimator */
+#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
+/* 6400 Encoder */
+#define ISAC_ENCODER_NOT_INITIATED 6410
+#define ISAC_DISALLOWED_CODING_MODE 6420
+#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
+#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
+#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
+#define ISAC_DISALLOWED_ENCODER_BANDWIDTH 6460
+/* 6600 Decoder */
+#define ISAC_DECODER_NOT_INITIATED 6610
+#define ISAC_EMPTY_PACKET 6620
+#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
+#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
+#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
+#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
+#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
+#define ISAC_RANGE_ERROR_DECODE_LPC 6680
+#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
+#define ISAC_LENGTH_MISMATCH 6730
+#define ISAC_RANGE_ERROR_DECODE_BANDWITH 6740
+#define ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER 6750
+#define ISAC_DISALLOWED_LPC_MODEL 6760
+/* 6800 Call setup formats */
+#define ISAC_INCOMPATIBLE_FORMATS 6810
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/structs.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/structs.h
new file mode 100644
index 0000000000..6861ca42bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/structs.h
@@ -0,0 +1,448 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * structs.h
+ *
+ * This header file contains all the structs used in the ISAC codec
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+
+#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
+#include "modules/third_party/fft/fft.h"
+
+typedef struct Bitstreamstruct {
+  uint8_t stream[STREAM_SIZE_MAX];
+  uint32_t W_upper;
+  uint32_t streamval;
+  uint32_t stream_index;
+
+} Bitstr;
+
+typedef struct {
+  double DataBufferLo[WINLEN];
+  double DataBufferHi[WINLEN];
+
+  double CorrBufLo[ORDERLO + 1];
+  double CorrBufHi[ORDERHI + 1];
+
+  float PreStateLoF[ORDERLO + 1];
+  float PreStateLoG[ORDERLO + 1];
+  float PreStateHiF[ORDERHI + 1];
+  float PreStateHiG[ORDERHI + 1];
+  float PostStateLoF[ORDERLO + 1];
+  float PostStateLoG[ORDERLO + 1];
+  float PostStateHiF[ORDERHI + 1];
+  float PostStateHiG[ORDERHI + 1];
+
+  double OldEnergy;
+
+} MaskFiltstr;
+
+typedef struct {
+  // state vectors for each of the two analysis filters
+  double INSTAT1[2 * (QORDER - 1)];
+  double INSTAT2[2 * (QORDER - 1)];
+  double INSTATLA1[2 * (QORDER - 1)];
+  double INSTATLA2[2 * (QORDER - 1)];
+  double INLABUF1[QLOOKAHEAD];
+  double INLABUF2[QLOOKAHEAD];
+
+  float INSTAT1_float[2 * (QORDER - 1)];
+  float INSTAT2_float[2 * (QORDER - 1)];
+  float INSTATLA1_float[2 * (QORDER - 1)];
+  float INSTATLA2_float[2 * (QORDER - 1)];
+  float INLABUF1_float[QLOOKAHEAD];
+  float INLABUF2_float[QLOOKAHEAD];
+
+  /* High pass filter */
+  double HPstates[HPORDER];
+  float HPstates_float[HPORDER];
+
+} PreFiltBankstr;
+
+typedef struct {
+  // state vectors for each of the two analysis filters
+  double STATE_0_LOWER[2 * POSTQORDER];
+  double STATE_0_UPPER[2 * POSTQORDER];
+
+  /* High pass filter */
+  double HPstates1[HPORDER];
+  double HPstates2[HPORDER];
+
+  float STATE_0_LOWER_float[2 * POSTQORDER];
+  float STATE_0_UPPER_float[2 * POSTQORDER];
+
+  float HPstates1_float[HPORDER];
+  float HPstates2_float[HPORDER];
+
+} PostFiltBankstr;
+
+typedef struct {
+  // data buffer for pitch filter
+  double ubuf[PITCH_BUFFSIZE];
+
+  // low pass state vector
+  double ystate[PITCH_DAMPORDER];
+
+  // old lag and gain
+  double oldlagp[1];
+  double oldgainp[1];
+
+} PitchFiltstr;
+
+typedef struct {
+  // data buffer
+  double buffer[PITCH_WLPCBUFLEN];
+
+  // state vectors
+  double istate[PITCH_WLPCORDER];
+  double weostate[PITCH_WLPCORDER];
+  double whostate[PITCH_WLPCORDER];
+
+  // LPC window   -> should be a global array because constant
+  double window[PITCH_WLPCWINLEN];
+
+} WeightFiltstr;
+
+typedef struct {
+  // for inital estimator
+  double dec_buffer[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
+                    PITCH_FRAME_LEN / 2 + 2];
+  double decimator_state[2 * ALLPASSSECTIONS + 1];
+  double hp_state[2];
+
+  double whitened_buf[QLOOKAHEAD];
+
+  double inbuf[QLOOKAHEAD];
+
+  PitchFiltstr PFstr_wght;
+  PitchFiltstr PFstr;
+  WeightFiltstr Wghtstr;
+
+} PitchAnalysisStruct;
+
+/* Have instance of struct together with other iSAC structs */
+typedef struct {
+  /* Previous frame length (in ms)                                    */
+  int32_t prev_frame_length;
+
+  /* Previous RTP timestamp from received
+     packet (in samples relative beginning)                           */
+  int32_t prev_rec_rtp_number;
+
+  /* Send timestamp for previous packet (in ms using timeGetTime())   */
+  uint32_t prev_rec_send_ts;
+
+  /* Arrival time for previous packet (in ms using timeGetTime())     */
+  uint32_t prev_rec_arr_ts;
+
+  /* rate of previous packet, derived from RTP timestamps (in bits/s) */
+  float prev_rec_rtp_rate;
+
+  /* Time sinse the last update of the BN estimate (in ms)            */
+  uint32_t last_update_ts;
+
+  /* Time sinse the last reduction (in ms)                            */
+  uint32_t last_reduction_ts;
+
+  /* How many times the estimate was update in the beginning          */
+  int32_t count_tot_updates_rec;
+
+  /* The estimated bottle neck rate from there to here (in bits/s)    */
+  int32_t rec_bw;
+  float rec_bw_inv;
+  float rec_bw_avg;
+  float rec_bw_avg_Q;
+
+  /* The estimated mean absolute jitter value,
+     as seen on this side (in ms)                                     */
+  float rec_jitter;
+  float rec_jitter_short_term;
+  float rec_jitter_short_term_abs;
+  float rec_max_delay;
+  float rec_max_delay_avg_Q;
+
+  /* (assumed) bitrate for headers (bps)                              */
+  float rec_header_rate;
+
+  /* The estimated bottle neck rate from here to there (in bits/s)    */
+  float send_bw_avg;
+
+  /* The estimated mean absolute jitter value, as seen on
+     the other siee (in ms)                                           */
+  float send_max_delay_avg;
+
+  // number of packets received since last update
+  int num_pkts_rec;
+
+  int num_consec_rec_pkts_over_30k;
+
+  // flag for marking that a high speed network has been
+  // detected downstream
+  int hsn_detect_rec;
+
+  int num_consec_snt_pkts_over_30k;
+
+  // flag for marking that a high speed network has
+  // been detected upstream
+  int hsn_detect_snd;
+
+  uint32_t start_wait_period;
+
+  int in_wait_period;
+
+  int change_to_WB;
+
+  uint32_t senderTimestamp;
+  uint32_t receiverTimestamp;
+  // enum IsacSamplingRate incomingStreamSampFreq;
+  uint16_t numConsecLatePkts;
+  float consecLatency;
+  int16_t inWaitLatePkts;
+
+  IsacBandwidthInfo external_bw_info;
+} BwEstimatorstr;
+
+typedef struct {
+  /* boolean, flags if previous packet exceeded B.N. */
+  int PrevExceed;
+  /* ms */
+  int ExceedAgo;
+  /* packets left to send in current burst */
+  int BurstCounter;
+  /* packets */
+  int InitCounter;
+  /* ms remaining in buffer when next packet will be sent */
+  double StillBuffered;
+
+} RateModel;
+
+/* The following strutc is used to store data from encoding, to make it
+   fast and easy to construct a new bitstream with a different Bandwidth
+   estimate. All values (except framelength and minBytes) is double size to
+   handle 60 ms of data.
+*/
+typedef struct {
+  /* Used to keep track of if it is first or second part of 60 msec packet */
+  int startIdx;
+
+  /* Frame length in samples */
+  int16_t framelength;
+
+  /* Pitch Gain */
+  int pitchGain_index[2];
+
+  /* Pitch Lag */
+  double meanGain[2];
+  int pitchIndex[PITCH_SUBFRAMES * 2];
+
+  /* LPC */
+  int LPCindex_s[108 * 2]; /* KLT_ORDER_SHAPE = 108 */
+  int LPCindex_g[12 * 2];  /* KLT_ORDER_GAIN = 12 */
+  double LPCcoeffs_lo[(ORDERLO + 1) * SUBFRAMES * 2];
+  double LPCcoeffs_hi[(ORDERHI + 1) * SUBFRAMES * 2];
+
+  /* Encode Spec */
+  int16_t fre[FRAMESAMPLES];
+  int16_t fim[FRAMESAMPLES];
+  int16_t AvgPitchGain[2];
+
+  /* Used in adaptive mode only */
+  int minBytes;
+
+} IsacSaveEncoderData;
+
+typedef struct {
+  int indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+  double lpcGain[SUBFRAMES << 1];
+  int lpcGainIndex[SUBFRAMES << 1];
+
+  Bitstr bitStreamObj;
+
+  int16_t realFFT[FRAMESAMPLES_HALF];
+  int16_t imagFFT[FRAMESAMPLES_HALF];
+} ISACUBSaveEncDataStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PreFiltBankstr prefiltbankstr_obj;
+  PitchFiltstr pitchfiltstr_obj;
+  PitchAnalysisStruct pitchanalysisstr_obj;
+  FFTstr fftstr_obj;
+  IsacSaveEncoderData SaveEnc_obj;
+
+  int buffer_index;
+  int16_t current_framesamples;
+
+  float data_buffer_float[FRAMESAMPLES_30ms];
+
+  int frame_nb;
+  double bottleneck;
+  int16_t new_framelength;
+  double s2nr;
+
+  /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t payloadLimitBytes30;
+  /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t payloadLimitBytes60;
+  /* Maximum allowed number of bits for both 30 and 60 msec packet */
+  int16_t maxPayloadBytes;
+  /* Maximum allowed rate in bytes per 30 msec packet */
+  int16_t maxRateInBytes;
+
+  /*---
+    If set to 1 iSAC will not adapt the frame-size, if used in
+    channel-adaptive mode. The initial value will be used for all rates.
+    ---*/
+  int16_t enforceFrameSize;
+
+  /*-----
+    This records the BWE index the encoder injected into the bit-stream.
+    It will be used in RCU. The same BWE index of main payload will be in
+    the redundant payload. We can not retrieve it from BWE because it is
+    a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
+    called only once per each encode.
+    -----*/
+  int16_t lastBWIdx;
+} ISACLBEncStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PreFiltBankstr prefiltbankstr_obj;
+  FFTstr fftstr_obj;
+  ISACUBSaveEncDataStruct SaveEnc_obj;
+
+  int buffer_index;
+  float data_buffer_float[MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES];
+  double bottleneck;
+  /* Maximum allowed number of bits for a 30 msec packet */
+  // int16_t        payloadLimitBytes30;
+  /* Maximum allowed number of bits for both 30 and 60 msec packet */
+  // int16_t        maxPayloadBytes;
+  int16_t maxPayloadSizeBytes;
+
+  double lastLPCVec[UB_LPC_ORDER];
+  int16_t numBytesUsed;
+  int16_t lastJitterInfo;
+} ISACUBEncStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PostFiltBankstr postfiltbankstr_obj;
+  PitchFiltstr pitchfiltstr_obj;
+  FFTstr fftstr_obj;
+
+} ISACLBDecStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PostFiltBankstr postfiltbankstr_obj;
+  FFTstr fftstr_obj;
+
+} ISACUBDecStruct;
+
+typedef struct {
+  ISACLBEncStruct ISACencLB_obj;
+  ISACLBDecStruct ISACdecLB_obj;
+} ISACLBStruct;
+
+typedef struct {
+  ISACUBEncStruct ISACencUB_obj;
+  ISACUBDecStruct ISACdecUB_obj;
+} ISACUBStruct;
+
+/*
+  This struct is used to take a snapshot of the entropy coder and LPC gains
+  right before encoding LPC gains. This allows us to go back to that state
+  if we like to limit the payload size.
+*/
+typedef struct {
+  /* 6 lower-band & 6 upper-band */
+  double loFiltGain[SUBFRAMES];
+  double hiFiltGain[SUBFRAMES];
+  /* Upper boundary of interval W */
+  uint32_t W_upper;
+  uint32_t streamval;
+  /* Index to the current position in bytestream */
+  uint32_t stream_index;
+  uint8_t stream[3];
+} transcode_obj;
+
+typedef struct {
+  // TODO(kwiberg): The size of these tables could be reduced by storing floats
+  // instead of doubles, and by making use of the identity cos(x) =
+  // sin(x+pi/2). They could also be made global constants that we fill in at
+  // compile time.
+  double costab1[FRAMESAMPLES_HALF];
+  double sintab1[FRAMESAMPLES_HALF];
+  double costab2[FRAMESAMPLES_QUARTER];
+  double sintab2[FRAMESAMPLES_QUARTER];
+} TransformTables;
+
+typedef struct {
+  // lower-band codec instance
+  ISACLBStruct instLB;
+  // upper-band codec instance
+  ISACUBStruct instUB;
+
+  // Bandwidth Estimator and model for the rate.
+  BwEstimatorstr bwestimator_obj;
+  RateModel rate_data_obj;
+  double MaxDelay;
+
+  /* 0 = adaptive; 1 = instantaneous */
+  int16_t codingMode;
+
+  // overall bottleneck of the codec
+  int32_t bottleneck;
+
+  // QMF Filter state
+  int32_t analysisFBState1[FB_STATE_SIZE_WORD32];
+  int32_t analysisFBState2[FB_STATE_SIZE_WORD32];
+  int32_t synthesisFBState1[FB_STATE_SIZE_WORD32];
+  int32_t synthesisFBState2[FB_STATE_SIZE_WORD32];
+
+  // Error Code
+  int16_t errorCode;
+
+  // bandwidth of the encoded audio 8, 12 or 16 kHz
+  enum ISACBandwidth bandwidthKHz;
+  // Sampling rate of audio, encoder and decode,  8 or 16 kHz
+  enum IsacSamplingRate encoderSamplingRateKHz;
+  enum IsacSamplingRate decoderSamplingRateKHz;
+  // Flag to keep track of initializations, lower & upper-band
+  // encoder and decoder.
+  int16_t initFlag;
+
+  // Flag to to indicate signal bandwidth switch
+  int16_t resetFlag_8kHz;
+
+  // Maximum allowed rate, measured in Bytes per 30 ms.
+  int16_t maxRateBytesPer30Ms;
+  // Maximum allowed payload-size, measured in Bytes.
+  int16_t maxPayloadSizeBytes;
+  /* The expected sampling rate of the input signal. Valid values are 16000
+   * and 32000. This is not the operation sampling rate of the codec. */
+  uint16_t in_sample_rate_hz;
+
+  // Trig tables for WebRtcIsac_Time2Spec and WebRtcIsac_Spec2time.
+  TransformTables transform_tables;
+} ISACMainStruct;
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
new file mode 100644
index 0000000000..dacf325082
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
+                                                 rtc::Buffer&& payload)
+    : decoder_(decoder), payload_(std::move(payload)) {}
+
+LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default;
+
+size_t LegacyEncodedAudioFrame::Duration() const {
+  const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+  return (ret < 0) ? 0 : static_cast<size_t>(ret);
+}
+
+absl::optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
+LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
+  AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+  const int ret = decoder_->Decode(
+      payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+      decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+
+  if (ret < 0)
+    return absl::nullopt;
+
+  return DecodeResult{static_cast<size_t>(ret), speech_type};
+}
+
+std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
+    AudioDecoder* decoder,
+    rtc::Buffer&& payload,
+    uint32_t timestamp,
+    size_t bytes_per_ms,
+    uint32_t timestamps_per_ms) {
+  RTC_DCHECK(payload.data());
+  std::vector<AudioDecoder::ParseResult> results;
+  size_t split_size_bytes = payload.size();
+
+  // Find a "chunk size" >= 20 ms and < 40 ms.
+  const size_t min_chunk_size = bytes_per_ms * 20;
+  if (min_chunk_size >= payload.size()) {
+    std::unique_ptr<LegacyEncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(decoder, std::move(payload)));
+    results.emplace_back(timestamp, 0, std::move(frame));
+  } else {
+    // Reduce the split size by half as long as `split_size_bytes` is at least
+    // twice the minimum chunk size (so that the resulting size is at least as
+    // large as the minimum chunk size).
+    while (split_size_bytes >= 2 * min_chunk_size) {
+      split_size_bytes /= 2;
+    }
+
+    const uint32_t timestamps_per_chunk = static_cast<uint32_t>(
+        split_size_bytes * timestamps_per_ms / bytes_per_ms);
+    size_t byte_offset;
+    uint32_t timestamp_offset;
+    for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
+         byte_offset += split_size_bytes,
+        timestamp_offset += timestamps_per_chunk) {
+      split_size_bytes =
+          std::min(split_size_bytes, payload.size() - byte_offset);
+      rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes);
+      std::unique_ptr<LegacyEncodedAudioFrame> frame(
+          new LegacyEncodedAudioFrame(decoder, std::move(new_payload)));
+      results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
+    }
+  }
+
+  return results;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
new file mode 100644
index 0000000000..21da1367ed
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+#define MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
+ public:
+  LegacyEncodedAudioFrame(AudioDecoder* decoder, rtc::Buffer&& payload);
+  ~LegacyEncodedAudioFrame() override;
+
+  static std::vector<AudioDecoder::ParseResult> SplitBySamples(
+      AudioDecoder* decoder,
+      rtc::Buffer&& payload,
+      uint32_t timestamp,
+      size_t bytes_per_ms,
+      uint32_t timestamps_per_ms);
+
+  size_t Duration() const override;
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override;
+
+  // For testing:
+  const rtc::Buffer& payload() const { return payload_; }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
new file mode 100644
index 0000000000..f81aeeea80
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+enum class NetEqDecoder {
+  kDecoderPCMu,
+  kDecoderPCMa,
+  kDecoderPCMu_2ch,
+  kDecoderPCMa_2ch,
+  kDecoderPCM16B,
+  kDecoderPCM16Bwb,
+  kDecoderPCM16Bswb32kHz,
+  kDecoderPCM16Bswb48kHz,
+  kDecoderPCM16B_2ch,
+  kDecoderPCM16Bwb_2ch,
+  kDecoderPCM16Bswb32kHz_2ch,
+  kDecoderPCM16Bswb48kHz_2ch,
+  kDecoderPCM16B_5ch,
+  kDecoderG722,
+};
+
+class SplitBySamplesTest : public ::testing::TestWithParam<NetEqDecoder> {
+ protected:
+  virtual void SetUp() {
+    decoder_type_ = GetParam();
+    switch (decoder_type_) {
+      case NetEqDecoder::kDecoderPCMu:
+      case NetEqDecoder::kDecoderPCMa:
+        bytes_per_ms_ = 8;
+        samples_per_ms_ = 8;
+        break;
+      case NetEqDecoder::kDecoderPCMu_2ch:
+      case NetEqDecoder::kDecoderPCMa_2ch:
+        bytes_per_ms_ = 2 * 8;
+        samples_per_ms_ = 8;
+        break;
+      case NetEqDecoder::kDecoderG722:
+        bytes_per_ms_ = 8;
+        samples_per_ms_ = 16;
+        break;
+      case NetEqDecoder::kDecoderPCM16B:
+        bytes_per_ms_ = 16;
+        samples_per_ms_ = 8;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bwb:
+        bytes_per_ms_ = 32;
+        samples_per_ms_ = 16;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bswb32kHz:
+        bytes_per_ms_ = 64;
+        samples_per_ms_ = 32;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bswb48kHz:
+        bytes_per_ms_ = 96;
+        samples_per_ms_ = 48;
+        break;
+      case NetEqDecoder::kDecoderPCM16B_2ch:
+        bytes_per_ms_ = 2 * 16;
+        samples_per_ms_ = 8;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bwb_2ch:
+        bytes_per_ms_ = 2 * 32;
+        samples_per_ms_ = 16;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
+        bytes_per_ms_ = 2 * 64;
+        samples_per_ms_ = 32;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
+        bytes_per_ms_ = 2 * 96;
+        samples_per_ms_ = 48;
+        break;
+      case NetEqDecoder::kDecoderPCM16B_5ch:
+        bytes_per_ms_ = 5 * 16;
+        samples_per_ms_ = 8;
+        break;
+      default:
+        RTC_DCHECK_NOTREACHED();
+        break;
+    }
+  }
+  size_t bytes_per_ms_;
+  int samples_per_ms_;
+  NetEqDecoder decoder_type_;
+};
+
+// Test splitting sample-based payloads.
+TEST_P(SplitBySamplesTest, PayloadSizes) {
+  constexpr uint32_t kBaseTimestamp = 0x12345678;
+  struct ExpectedSplit {
+    size_t payload_size_ms;
+    size_t num_frames;
+    // For simplicity. We only expect up to two packets per split.
+    size_t frame_sizes[2];
+  };
+  // The payloads are expected to be split as follows:
+  // 10 ms -> 10 ms
+  // 20 ms -> 20 ms
+  // 30 ms -> 30 ms
+  // 40 ms -> 20 + 20 ms
+  // 50 ms -> 25 + 25 ms
+  // 60 ms -> 30 + 30 ms
+  ExpectedSplit expected_splits[] = {{10, 1, {10}},     {20, 1, {20}},
+                                     {30, 1, {30}},     {40, 2, {20, 20}},
+                                     {50, 2, {25, 25}}, {60, 2, {30, 30}}};
+
+  for (const auto& expected_split : expected_splits) {
+    // The payload values are set to steadily increase (modulo 256), so that the
+    // resulting frames can be checked and we can be reasonably certain no
+    // sample was missed or repeated.
+    const auto generate_payload = [](size_t num_bytes) {
+      rtc::Buffer payload(num_bytes);
+      uint8_t value = 0;
+      // Allow wrap-around of value in counter below.
+      for (size_t i = 0; i != payload.size(); ++i, ++value) {
+        payload[i] = value;
+      }
+      return payload;
+    };
+
+    const auto results = LegacyEncodedAudioFrame::SplitBySamples(
+        nullptr,
+        generate_payload(expected_split.payload_size_ms * bytes_per_ms_),
+        kBaseTimestamp, bytes_per_ms_, samples_per_ms_);
+
+    EXPECT_EQ(expected_split.num_frames, results.size());
+    uint32_t expected_timestamp = kBaseTimestamp;
+    uint8_t value = 0;
+    for (size_t i = 0; i != expected_split.num_frames; ++i) {
+      const auto& result = results[i];
+      const LegacyEncodedAudioFrame* frame =
+          static_cast<const LegacyEncodedAudioFrame*>(result.frame.get());
+      const size_t length_bytes = expected_split.frame_sizes[i] * bytes_per_ms_;
+      EXPECT_EQ(length_bytes, frame->payload().size());
+      EXPECT_EQ(expected_timestamp, result.timestamp);
+      const rtc::Buffer& payload = frame->payload();
+      // Allow wrap-around of value in counter below.
+      for (size_t i = 0; i != payload.size(); ++i, ++value) {
+        ASSERT_EQ(value, payload[i]);
+      }
+
+      expected_timestamp += rtc::checked_cast<uint32_t>(
+          expected_split.frame_sizes[i] * samples_per_ms_);
+    }
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    LegacyEncodedAudioFrame,
+    SplitBySamplesTest,
+    ::testing::Values(NetEqDecoder::kDecoderPCMu,
+                      NetEqDecoder::kDecoderPCMa,
+                      NetEqDecoder::kDecoderPCMu_2ch,
+                      NetEqDecoder::kDecoderPCMa_2ch,
+                      NetEqDecoder::kDecoderG722,
+                      NetEqDecoder::kDecoderPCM16B,
+                      NetEqDecoder::kDecoderPCM16Bwb,
+                      NetEqDecoder::kDecoderPCM16Bswb32kHz,
+                      NetEqDecoder::kDecoderPCM16Bswb48kHz,
+                      NetEqDecoder::kDecoderPCM16B_2ch,
+                      NetEqDecoder::kDecoderPCM16Bwb_2ch,
+                      NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch,
+                      NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch,
+                      NetEqDecoder::kDecoderPCM16B_5ch));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/DEPS b/third_party/libwebrtc/modules/audio_coding/codecs/opus/DEPS
new file mode 100644
index 0000000000..c2530726ad
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/DEPS
@@ -0,0 +1,5 @@
+specific_include_rules = {
+  "opus_inst\.h": [
+    "+third_party/opus",
+  ],
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc
new file mode 100644
index 0000000000..03c02186d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+                                               absl::string_view param) {
+  auto it = format.parameters.find(std::string(param));
+  if (it == format.parameters.end())
+    return absl::nullopt;
+
+  return it->second;
+}
+
+// Parses a comma-separated string "1,2,0,6" into a std::vector<unsigned char>.
+template <>
+absl::optional<std::vector<unsigned char>> GetFormatParameter(
+    const SdpAudioFormat& format,
+    absl::string_view param) {
+  std::vector<unsigned char> result;
+  const std::string comma_separated_list =
+      GetFormatParameter(format, param).value_or("");
+  size_t pos = 0;
+  while (pos < comma_separated_list.size()) {
+    const size_t next_comma = comma_separated_list.find(',', pos);
+    const size_t distance_to_next_comma = next_comma == std::string::npos
+                                              ? std::string::npos
+                                              : (next_comma - pos);
+    auto substring_with_number =
+        comma_separated_list.substr(pos, distance_to_next_comma);
+    auto conv = rtc::StringToNumber<int>(substring_with_number);
+    if (!conv.has_value()) {
+      return absl::nullopt;
+    }
+    result.push_back(*conv);
+    pos += substring_with_number.size() + 1;
+  }
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
new file mode 100644
index 0000000000..5ebb51b577
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+                                               absl::string_view param);
+
+template <typename T>
+absl::optional<T> GetFormatParameter(const SdpAudioFormat& format,
+                                     absl::string_view param) {
+  return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
+}
+
+template <>
+absl::optional<std::vector<unsigned char>> GetFormatParameter(
+    const SdpAudioFormat& format,
+    absl::string_view param);
+
+class OpusFrame : public AudioDecoder::EncodedAudioFrame {
+ public:
+  OpusFrame(AudioDecoder* decoder,
+            rtc::Buffer&& payload,
+            bool is_primary_payload)
+      : decoder_(decoder),
+        payload_(std::move(payload)),
+        is_primary_payload_(is_primary_payload) {}
+
+  size_t Duration() const override {
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    } else {
+      ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
+    }
+    return (ret < 0) ? 0 : static_cast<size_t>(ret);
+  }
+
+  bool IsDtxPacket() const override { return payload_.size() <= 2; }
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->Decode(
+          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    } else {
+      ret = decoder_->DecodeRedundant(
+          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    }
+
+    if (ret < 0)
+      return absl::nullopt;
+
+    return DecodeResult{static_cast<size_t>(ret), speech_type};
+  }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+  const bool is_primary_payload_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
new file mode 100644
index 0000000000..285ea89959
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+std::unique_ptr<AudioDecoderMultiChannelOpusImpl>
+AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(
+    AudioDecoderMultiChannelOpusConfig config) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  // Fill the pointer with a working decoder through the C interface. This
+  // allocates memory.
+  OpusDecInst* dec_state = nullptr;
+  const int error = WebRtcOpus_MultistreamDecoderCreate(
+      &dec_state, config.num_channels, config.num_streams,
+      config.coupled_streams, config.channel_mapping.data());
+  if (error != 0) {
+    return nullptr;
+  }
+
+  // Pass the ownership to DecoderImpl. Not using 'make_unique' because the
+  // c-tor is private.
+  return std::unique_ptr<AudioDecoderMultiChannelOpusImpl>(
+      new AudioDecoderMultiChannelOpusImpl(dec_state, config));
+}
+
+AudioDecoderMultiChannelOpusImpl::AudioDecoderMultiChannelOpusImpl(
+    OpusDecInst* dec_state,
+    AudioDecoderMultiChannelOpusConfig config)
+    : dec_state_(dec_state), config_(config) {
+  RTC_DCHECK(dec_state);
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderMultiChannelOpusImpl::~AudioDecoderMultiChannelOpusImpl() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+absl::optional<AudioDecoderMultiChannelOpusConfig>
+AudioDecoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) {
+  AudioDecoderMultiChannelOpusConfig config;
+  config.num_channels = format.num_channels;
+  auto num_streams = GetFormatParameter<int>(format, "num_streams");
+  if (!num_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.num_streams = *num_streams;
+
+  auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams");
+  if (!coupled_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.coupled_streams = *coupled_streams;
+
+  auto channel_mapping =
+      GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping");
+  if (!channel_mapping.has_value()) {
+    return absl::nullopt;
+  }
+  config.channel_mapping = *channel_mapping;
+  if (!config.IsOk()) {
+    return absl::nullopt;
+  }
+  return config;
+}
+
+std::vector<AudioDecoder::ParseResult>
+AudioDecoderMultiChannelOpusImpl::ParsePayload(rtc::Buffer&& payload,
+                                               uint32_t timestamp) {
+  std::vector<ParseResult> results;
+
+  if (PacketHasFec(payload.data(), payload.size())) {
+    const int duration =
+        PacketDurationRedundant(payload.data(), payload.size());
+    RTC_DCHECK_GE(duration, 0);
+    rtc::Buffer payload_copy(payload.data(), payload.size());
+    std::unique_ptr<EncodedAudioFrame> fec_frame(
+        new OpusFrame(this, std::move(payload_copy), false));
+    results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
+  }
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OpusFrame(this, std::move(payload), true));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoderMultiChannelOpusImpl::DecodeInternal(const uint8_t* encoded,
+                                                     size_t encoded_len,
+                                                     int sample_rate_hz,
+                                                     int16_t* decoded,
+                                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(
+        config_.num_channels);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderMultiChannelOpusImpl::DecodeRedundantInternal(
+    const uint8_t* encoded,
+    size_t encoded_len,
+    int sample_rate_hz,
+    int16_t* decoded,
+    SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(
+        config_.num_channels);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderMultiChannelOpusImpl::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderMultiChannelOpusImpl::PacketDuration(const uint8_t* encoded,
+                                                     size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderMultiChannelOpusImpl::PacketDurationRedundant(
+    const uint8_t* encoded,
+    size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len, 48000);
+}
+
+bool AudioDecoderMultiChannelOpusImpl::PacketHasFec(const uint8_t* encoded,
+                                                    size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+int AudioDecoderMultiChannelOpusImpl::SampleRateHz() const {
+  return 48000;
+}
+
+size_t AudioDecoderMultiChannelOpusImpl::Channels() const {
+  return config_.num_channels;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
new file mode 100644
index 0000000000..2ff47a8a53
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoderMultiChannelOpusImpl final : public AudioDecoder {
+ public:
+  static std::unique_ptr<AudioDecoderMultiChannelOpusImpl> MakeAudioDecoder(
+      AudioDecoderMultiChannelOpusConfig config);
+
+  ~AudioDecoderMultiChannelOpusImpl() override;
+
+  AudioDecoderMultiChannelOpusImpl(const AudioDecoderMultiChannelOpusImpl&) =
+      delete;
+  AudioDecoderMultiChannelOpusImpl& operator=(
+      const AudioDecoderMultiChannelOpusImpl&) = delete;
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+  static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  AudioDecoderMultiChannelOpusImpl(OpusDecInst* dec_state,
+                                   AudioDecoderMultiChannelOpusConfig config);
+
+  OpusDecInst* dec_state_;
+  const AudioDecoderMultiChannelOpusConfig config_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc
new file mode 100644
index 0000000000..57e2107f3c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
+
+TEST(AudioDecoderMultiOpusTest, GetFormatParameter) {
+  const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                  {{"channel_mapping", "0,1,2,3"},
+                                   {"coupled_streams", "2"},
+                                   {"num_streams", "2"}});
+
+  EXPECT_EQ(GetFormatParameter(sdp_format, "channel_mapping"),
+            absl::optional<std::string>("0,1,2,3"));
+
+  EXPECT_EQ(GetFormatParameter<int>(sdp_format, "coupled_streams"),
+            absl::optional<int>(2));
+
+  EXPECT_EQ(GetFormatParameter(sdp_format, "missing"), absl::nullopt);
+
+  EXPECT_EQ(GetFormatParameter<int>(sdp_format, "channel_mapping"),
+            absl::nullopt);
+}
+
+TEST(AudioDecoderMultiOpusTest, InvalidChannelMappings) {
+  {
+    // Can't use channel 3 if there are only 2 channels.
+    const SdpAudioFormat sdp_format("multiopus", 48000, 2,
+                                    {{"channel_mapping", "3,0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+  {
+    // The mapping is too long. There are only 5 channels, but 6 elements in the
+    // mapping.
+    const SdpAudioFormat sdp_format("multiopus", 48000, 5,
+                                    {{"channel_mapping", "0,1,2,3,4,5"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+  {
+    // The mapping doesn't parse correctly.
+    const SdpAudioFormat sdp_format(
+        "multiopus", 48000, 5,
+        {{"channel_mapping", "0,1,two,3,4"}, {"coupled_streams", "0"}});
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+}
+
+TEST(AudioDecoderMultiOpusTest, ValidSdpToConfigProducesCorrectConfig) {
+  const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                  {{"channel_mapping", "3,1,2,0"},
+                                   {"coupled_streams", "2"},
+                                   {"num_streams", "2"}});
+
+  const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+      AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+  ASSERT_TRUE(decoder_config.has_value());
+  EXPECT_TRUE(decoder_config->IsOk());
+  EXPECT_EQ(decoder_config->coupled_streams, 2);
+  EXPECT_THAT(decoder_config->channel_mapping,
+              ::testing::ContainerEq(std::vector<unsigned char>({3, 1, 2, 0})));
+}
+
+TEST(AudioDecoderMultiOpusTest, InvalidSdpToConfigDoesNotProduceConfig) {
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                    {{"channel_mapping", "0,1,2,3"},
+                                     {"coupled_stream", "2"},
+                                     {"num_streams", "2"}});
+
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                    {{"channel_mapping", "0,1,2 3"},
+                                     {"coupled_streams", "2"},
+                                     {"num_streams", "2"}});
+
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+}
+
+TEST(AudioDecoderMultiOpusTest, CodecsCanBeCreated) {
+  const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                  {{"channel_mapping", "0,1,2,3"},
+                                   {"coupled_streams", "2"},
+                                   {"num_streams", "2"}});
+
+  const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+      AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+  ASSERT_TRUE(decoder_config.has_value());
+
+  const std::unique_ptr<AudioDecoder> opus_decoder =
+      AudioDecoderMultiChannelOpus::MakeAudioDecoder(*decoder_config);
+
+  EXPECT_TRUE(opus_decoder);
+}
+
+TEST(AudioDecoderMultiOpusTest, AdvertisedCodecsCanBeCreated) {
+  std::vector<AudioCodecSpec> specs;
+  AudioDecoderMultiChannelOpus::AppendSupportedDecoders(&specs);
+
+  EXPECT_FALSE(specs.empty());
+
+  for (const AudioCodecSpec& spec : specs) {
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(spec.format);
+    ASSERT_TRUE(decoder_config.has_value());
+
+    const std::unique_ptr<AudioDecoder> opus_decoder =
+        AudioDecoderMultiChannelOpus::MakeAudioDecoder(*decoder_config);
+
+    EXPECT_TRUE(opus_decoder);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
new file mode 100644
index 0000000000..cff9685548
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels,
+                                           int sample_rate_hz)
+    : channels_{num_channels}, sample_rate_hz_{sample_rate_hz} {
+  RTC_DCHECK(num_channels == 1 || num_channels == 2);
+  RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000);
+  const int error =
+      WebRtcOpus_DecoderCreate(&dec_state_, channels_, sample_rate_hz_);
+  RTC_DCHECK(error == 0);
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderOpusImpl::~AudioDecoderOpusImpl() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+
+  if (PacketHasFec(payload.data(), payload.size())) {
+    const int duration =
+        PacketDurationRedundant(payload.data(), payload.size());
+    RTC_DCHECK_GE(duration, 0);
+    rtc::Buffer payload_copy(payload.data(), payload.size());
+    std::unique_ptr<EncodedAudioFrame> fec_frame(
+        new OpusFrame(this, std::move(payload_copy), false));
+    results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
+  }
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OpusFrame(this, std::move(payload), true));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded,
+                                                  size_t encoded_len,
+                                                  int sample_rate_hz,
+                                                  int16_t* decoded,
+                                                  SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderOpusImpl::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderOpusImpl::PacketDuration(const uint8_t* encoded,
+                                         size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded,
+                                                  size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len, sample_rate_hz_);
+}
+
+bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
+                                        size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+int AudioDecoderOpusImpl::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioDecoderOpusImpl::Channels() const {
+  return channels_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
new file mode 100644
index 0000000000..e8fd0440bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoderOpusImpl final : public AudioDecoder {
+ public:
+  explicit AudioDecoderOpusImpl(size_t num_channels,
+                                int sample_rate_hz = 48000);
+  ~AudioDecoderOpusImpl() override;
+
+  AudioDecoderOpusImpl(const AudioDecoderOpusImpl&) = delete;
+  AudioDecoderOpusImpl& operator=(const AudioDecoderOpusImpl&) = delete;
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  OpusDecInst* dec_state_;
+  const size_t channels_;
+  const int sample_rate_hz_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
new file mode 100644
index 0000000000..38a11c123d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
@@ -0,0 +1,366 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * LEFT TO DO:
+ * - WRITE TESTS for the stuff in this  file.
+ * - Check the creation, maybe make it safer by returning an empty optional or
+ *   unique_ptr. --- It looks OK, but RecreateEncoderInstance can perhaps crash
+ *   on a valid config. Can run it in the fuzzer for some time. Should prbl also
+ *   fuzz the config.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+namespace {
+
+// Recommended bitrates for one channel:
+// 8-12 kb/s for NB speech,
+// 16-20 kb/s for WB speech,
+// 28-40 kb/s for FB speech,
+// 48-64 kb/s for FB mono music, and
+// 64-128 kb/s for FB stereo music.
+// The current implementation multiplies these values by the number of channels.
+constexpr int kOpusBitrateNbBps = 12000;
+constexpr int kOpusBitrateWbBps = 20000;
+constexpr int kOpusBitrateFbBps = 32000;
+
+constexpr int kDefaultMaxPlaybackRate = 48000;
+// These two lists must be sorted from low to high
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120};
+#else
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
+#endif
+
+int GetBitrateBps(const AudioEncoderMultiChannelOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return config.bitrate_bps;
+}
+int GetMaxPlaybackRate(const SdpAudioFormat& format) {
+  const auto param = GetFormatParameter<int>(format, "maxplaybackrate");
+  if (param && *param >= 8000) {
+    return std::min(*param, kDefaultMaxPlaybackRate);
+  }
+  return kDefaultMaxPlaybackRate;
+}
+
+int GetFrameSizeMs(const SdpAudioFormat& format) {
+  const auto ptime = GetFormatParameter<int>(format, "ptime");
+  if (ptime.has_value()) {
+    // Pick the next highest supported frame length from
+    // kOpusSupportedFrameLengths.
+    for (const int supported_frame_length : kOpusSupportedFrameLengths) {
+      if (supported_frame_length >= *ptime) {
+        return supported_frame_length;
+      }
+    }
+    // If none was found, return the largest supported frame length.
+    return *(std::end(kOpusSupportedFrameLengths) - 1);
+  }
+
+  return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+}
+
+int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
+  const int bitrate = [&] {
+    if (max_playback_rate <= 8000) {
+      return kOpusBitrateNbBps * rtc::dchecked_cast<int>(num_channels);
+    } else if (max_playback_rate <= 16000) {
+      return kOpusBitrateWbBps * rtc::dchecked_cast<int>(num_channels);
+    } else {
+      return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
+    }
+  }();
+  RTC_DCHECK_GE(bitrate, AudioEncoderMultiChannelOpusConfig::kMinBitrateBps);
+  return bitrate;
+}
+
+// Get the maxaveragebitrate parameter in string-form, so we can properly figure
+// out how invalid it is and accurately log invalid values.
+int CalculateBitrate(int max_playback_rate_hz,
+                     size_t num_channels,
+                     absl::optional<std::string> bitrate_param) {
+  const int default_bitrate =
+      CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
+
+  if (bitrate_param) {
+    const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
+    if (bitrate) {
+      const int chosen_bitrate =
+          std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                   std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
+      if (bitrate != chosen_bitrate) {
+        RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
+                            << " clamped to " << chosen_bitrate;
+      }
+      return chosen_bitrate;
+    }
+    RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate \"" << *bitrate_param
+                        << "\" replaced by default bitrate " << default_bitrate;
+  }
+
+  return default_bitrate;
+}
+
+}  // namespace
+
+std::unique_ptr<AudioEncoder>
+AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config,
+    int payload_type) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderMultiChannelOpusImpl>(config,
+                                                            payload_type);
+}
+
+AudioEncoderMultiChannelOpusImpl::AudioEncoderMultiChannelOpusImpl(
+    const AudioEncoderMultiChannelOpusConfig& config,
+    int payload_type)
+    : payload_type_(payload_type), inst_(nullptr) {
+  RTC_DCHECK(0 <= payload_type && payload_type <= 127);
+
+  RTC_CHECK(RecreateEncoderInstance(config));
+}
+
+AudioEncoderMultiChannelOpusImpl::~AudioEncoderMultiChannelOpusImpl() {
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+}
+
+size_t AudioEncoderMultiChannelOpusImpl::SufficientOutputBufferSize() const {
+  // Calculate the number of bytes we expect the encoder to produce,
+  // then multiply by two to give a wide margin for error.
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
+  return 2 * approx_encoded_bytes;
+}
+
+void AudioEncoderMultiChannelOpusImpl::Reset() {
+  RTC_CHECK(RecreateEncoderInstance(config_));
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderMultiChannelOpusImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(config_.frame_size_ms),
+           TimeDelta::Millis(config_.frame_size_ms)}};
+}
+
+// If the given config is OK, recreate the Opus encoder instance with those
+// settings, save the config, and return true. Otherwise, do nothing and return
+// false.
+bool AudioEncoderMultiChannelOpusImpl::RecreateEncoderInstance(
+    const AudioEncoderMultiChannelOpusConfig& config) {
+  if (!config.IsOk())
+    return false;
+  config_ = config;
+  if (inst_)
+    RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+  input_buffer_.clear();
+  input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_MultistreamEncoderCreate(
+             &inst_, config.num_channels,
+             config.application ==
+                     AudioEncoderMultiChannelOpusConfig::ApplicationMode::kVoip
+                 ? 0
+                 : 1,
+             config.num_streams, config.coupled_streams,
+             config.channel_mapping.data()));
+  const int bitrate = GetBitrateBps(config);
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, bitrate));
+  RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
+  if (config.fec_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus enable FEC";
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus disable FEC";
+  }
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
+  RTC_LOG(LS_VERBOSE) << "Set Opus playback rate to "
+                      << config.max_playback_rate_hz << " hz.";
+
+  // Use the DEFAULT complexity.
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetComplexity(inst_, AudioEncoderOpusConfig().complexity));
+  RTC_LOG(LS_VERBOSE) << "Set Opus coding complexity to "
+                      << AudioEncoderOpusConfig().complexity;
+
+  if (config.dtx_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus enable DTX";
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus disable DTX";
+  }
+
+  if (config.cbr_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus enable CBR";
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus disable CBR";
+  }
+  num_channels_to_encode_ = NumChannels();
+  next_frame_length_ms_ = config_.frame_size_ms;
+  RTC_LOG(LS_VERBOSE) << "Set Opus frame length to " << config_.frame_size_ms
+                      << " ms";
+  return true;
+}
+
+absl::optional<AudioEncoderMultiChannelOpusConfig>
+AudioEncoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "multiopus") ||
+      format.clockrate_hz != 48000) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderMultiChannelOpusConfig config;
+  config.num_channels = format.num_channels;
+  config.frame_size_ms = GetFrameSizeMs(format);
+  config.max_playback_rate_hz = GetMaxPlaybackRate(format);
+  config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
+  config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
+  config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
+  config.bitrate_bps =
+      CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
+                       GetFormatParameter(format, "maxaveragebitrate"));
+  config.application =
+      config.num_channels == 1
+          ? AudioEncoderMultiChannelOpusConfig::ApplicationMode::kVoip
+          : AudioEncoderMultiChannelOpusConfig::ApplicationMode::kAudio;
+
+  config.supported_frame_lengths_ms.clear();
+  std::copy(std::begin(kOpusSupportedFrameLengths),
+            std::end(kOpusSupportedFrameLengths),
+            std::back_inserter(config.supported_frame_lengths_ms));
+
+  auto num_streams = GetFormatParameter<int>(format, "num_streams");
+  if (!num_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.num_streams = *num_streams;
+
+  auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams");
+  if (!coupled_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.coupled_streams = *coupled_streams;
+
+  auto channel_mapping =
+      GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping");
+  if (!channel_mapping.has_value()) {
+    return absl::nullopt;
+  }
+  config.channel_mapping = *channel_mapping;
+
+  if (!config.IsOk()) {
+    return absl::nullopt;
+  }
+  return config;
+}
+
+AudioCodecInfo AudioEncoderMultiChannelOpusImpl::QueryAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  AudioCodecInfo info(48000, config.num_channels, config.bitrate_bps,
+                      AudioEncoderOpusConfig::kMinBitrateBps,
+                      AudioEncoderOpusConfig::kMaxBitrateBps);
+  info.allow_comfort_noise = false;
+  info.supports_network_adaption = false;
+  return info;
+}
+
+size_t AudioEncoderMultiChannelOpusImpl::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
+}
+size_t AudioEncoderMultiChannelOpusImpl::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(48000, 100) * config_.num_channels;
+}
+int AudioEncoderMultiChannelOpusImpl::SampleRateHz() const {
+  return 48000;
+}
+size_t AudioEncoderMultiChannelOpusImpl::NumChannels() const {
+  return config_.num_channels;
+}
+size_t AudioEncoderMultiChannelOpusImpl::Num10MsFramesInNextPacket() const {
+  return Num10msFramesPerPacket();
+}
+size_t AudioEncoderMultiChannelOpusImpl::Max10MsFramesInAPacket() const {
+  return Num10msFramesPerPacket();
+}
+int AudioEncoderMultiChannelOpusImpl::GetTargetBitrate() const {
+  return GetBitrateBps(config_);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderMultiChannelOpusImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (input_buffer_.empty())
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
+  if (input_buffer_.size() <
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
+
+  const size_t max_encoded_bytes = SufficientOutputBufferSize();
+  EncodedInfo info;
+  info.encoded_bytes = encoded->AppendData(
+      max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
+        int status = WebRtcOpus_Encode(
+            inst_, &input_buffer_[0],
+            rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
+            rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
+
+        RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
+
+        return static_cast<size_t>(status);
+      });
+  input_buffer_.clear();
+
+  // Will use new packet size for next encoding.
+  config_.frame_size_ms = next_frame_length_ms_;
+
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.send_even_if_empty = true;  // Allows Opus to send empty packets.
+
+  info.speech = true;
+  info.encoder_type = CodecType::kOther;
+
+  return info;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
new file mode 100644
index 0000000000..8a7210515c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class AudioEncoderMultiChannelOpusImpl final : public AudioEncoder {
+ public:
+  AudioEncoderMultiChannelOpusImpl(
+      const AudioEncoderMultiChannelOpusConfig& config,
+      int payload_type);
+  ~AudioEncoderMultiChannelOpusImpl() override;
+
+  AudioEncoderMultiChannelOpusImpl(const AudioEncoderMultiChannelOpusImpl&) =
+      delete;
+  AudioEncoderMultiChannelOpusImpl& operator=(
+      const AudioEncoderMultiChannelOpusImpl&) = delete;
+
+  // Static interface for use by BuiltinAudioEncoderFactory.
+  static constexpr const char* GetPayloadName() { return "multiopus"; }
+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format);
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  static absl::optional<AudioEncoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+  static AudioCodecInfo QueryAudioEncoder(
+      const AudioEncoderMultiChannelOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderMultiChannelOpusConfig&,
+      int payload_type);
+
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
+  bool RecreateEncoderInstance(
+      const AudioEncoderMultiChannelOpusConfig& config);
+  void SetFrameLength(int frame_length_ms);
+  void SetNumChannelsToEncode(size_t num_channels_to_encode);
+  void SetProjectedPacketLossRate(float fraction);
+
+  AudioEncoderMultiChannelOpusConfig config_;
+  const int payload_type_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  size_t num_channels_to_encode_;
+  int next_frame_length_ms_;
+
+  friend struct AudioEncoderMultiChannelOpus;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc
new file mode 100644
index 0000000000..92f6f2c169
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+
+#include "test/gmock.h"
+
+namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace {
+constexpr int kOpusPayloadType = 120;
+}  // namespace
+
+TEST(AudioEncoderMultiOpusTest, CheckConfigValidity) {
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 2,
+                                    {{"channel_mapping", "3,0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // Maps input channel 0 to coded channel 3, which doesn't exist.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 2,
+                                    {{"channel_mapping", "0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // The mapping is too short.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,0,0"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "1"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // Coded channel 0 comes from both input channels 0, 1 and 2.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,255,255"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "1"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+    ASSERT_TRUE(encoder_config.has_value());
+
+    // This is fine, because channels 1, 2 are set to be ignored.
+    EXPECT_TRUE(encoder_config->IsOk());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,255,255"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // This is NOT fine, because channels nothing says how coded channel 1
+    // should be coded.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+}
+
+TEST(AudioEncoderMultiOpusTest, ConfigValuesAreParsedCorrectly) {
+  SdpAudioFormat sdp_format({"multiopus",
+                             48000,
+                             6,
+                             {{"minptime", "10"},
+                              {"useinbandfec", "1"},
+                              {"channel_mapping", "0,4,1,2,3,5"},
+                              {"num_streams", "4"},
+                              {"coupled_streams", "2"}}});
+  const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+      AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+  ASSERT_TRUE(encoder_config.has_value());
+
+  EXPECT_EQ(encoder_config->coupled_streams, 2);
+  EXPECT_EQ(encoder_config->num_streams, 4);
+  EXPECT_THAT(
+      encoder_config->channel_mapping,
+      testing::ContainerEq(std::vector<unsigned char>({0, 4, 1, 2, 3, 5})));
+}
+
+TEST(AudioEncoderMultiOpusTest, CreateFromValidConfig) {
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,255,255"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+    ASSERT_FALSE(encoder_config.has_value());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "1,255,0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "1"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+    ASSERT_TRUE(encoder_config.has_value());
+
+    EXPECT_THAT(encoder_config->channel_mapping,
+                testing::ContainerEq(std::vector<unsigned char>({1, 255, 0})));
+
+    EXPECT_TRUE(encoder_config->IsOk());
+
+    const std::unique_ptr<AudioEncoder> opus_encoder =
+        AudioEncoderMultiChannelOpus::MakeAudioEncoder(*encoder_config,
+                                                       kOpusPayloadType);
+
+    // Creating an encoder from a valid config should work.
+    EXPECT_TRUE(opus_encoder);
+  }
+}
+
+TEST(AudioEncoderMultiOpusTest, AdvertisedCodecsCanBeCreated) {
+  std::vector<AudioCodecSpec> specs;
+  AudioEncoderMultiChannelOpus::AppendSupportedEncoders(&specs);
+
+  EXPECT_FALSE(specs.empty());
+
+  for (const AudioCodecSpec& spec : specs) {
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(spec.format);
+    ASSERT_TRUE(encoder_config.has_value());
+
+    const std::unique_ptr<AudioEncoder> opus_encoder =
+        AudioEncoderMultiChannelOpus::MakeAudioEncoder(*encoder_config,
+                                                       kOpusPayloadType);
+
+    EXPECT_TRUE(opus_encoder);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
new file mode 100644
index 0000000000..17e0e33b1d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -0,0 +1,824 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+
+#include <algorithm>
+#include <iterator>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/exp_filter.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/string_to_number.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+// Codec parameters for Opus.
+// draft-spittka-payload-rtp-opus-03
+
+// Recommended bitrates:
+// 8-12 kb/s for NB speech,
+// 16-20 kb/s for WB speech,
+// 28-40 kb/s for FB speech,
+// 48-64 kb/s for FB mono music, and
+// 64-128 kb/s for FB stereo music.
+// The current implementation applies the following values to mono signals,
+// and multiplies them by 2 for stereo.
+constexpr int kOpusBitrateNbBps = 12000;
+constexpr int kOpusBitrateWbBps = 20000;
+constexpr int kOpusBitrateFbBps = 32000;
+
+constexpr int kRtpTimestampRateHz = 48000;
+constexpr int kDefaultMaxPlaybackRate = 48000;
+
+// These two lists must be sorted from low to high
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60, 120};
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120};
+#else
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60};
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
+#endif
+
+// PacketLossFractionSmoother uses an exponential filter with a time constant
+// of -1.0 / ln(0.9999) = 10000 ms.
+constexpr float kAlphaForPacketLossFractionSmoother = 0.9999f;
+constexpr float kMaxPacketLossFraction = 0.2f;
+
+int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
+  const int bitrate = [&] {
+    if (max_playback_rate <= 8000) {
+      return kOpusBitrateNbBps * rtc::dchecked_cast<int>(num_channels);
+    } else if (max_playback_rate <= 16000) {
+      return kOpusBitrateWbBps * rtc::dchecked_cast<int>(num_channels);
+    } else {
+      return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
+    }
+  }();
+  RTC_DCHECK_GE(bitrate, AudioEncoderOpusConfig::kMinBitrateBps);
+  RTC_DCHECK_LE(bitrate, AudioEncoderOpusConfig::kMaxBitrateBps);
+  return bitrate;
+}
+
+// Get the maxaveragebitrate parameter in string-form, so we can properly figure
+// out how invalid it is and accurately log invalid values.
+int CalculateBitrate(int max_playback_rate_hz,
+                     size_t num_channels,
+                     absl::optional<std::string> bitrate_param) {
+  const int default_bitrate =
+      CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
+
+  if (bitrate_param) {
+    const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
+    if (bitrate) {
+      const int chosen_bitrate =
+          std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                   std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
+      if (bitrate != chosen_bitrate) {
+        RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
+                            << " clamped to " << chosen_bitrate;
+      }
+      return chosen_bitrate;
+    }
+    RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate \"" << *bitrate_param
+                        << "\" replaced by default bitrate " << default_bitrate;
+  }
+
+  return default_bitrate;
+}
+
+int GetChannelCount(const SdpAudioFormat& format) {
+  const auto param = GetFormatParameter(format, "stereo");
+  if (param == "1") {
+    return 2;
+  } else {
+    return 1;
+  }
+}
+
+int GetMaxPlaybackRate(const SdpAudioFormat& format) {
+  const auto param = GetFormatParameter<int>(format, "maxplaybackrate");
+  if (param && *param >= 8000) {
+    return std::min(*param, kDefaultMaxPlaybackRate);
+  }
+  return kDefaultMaxPlaybackRate;
+}
+
+int GetFrameSizeMs(const SdpAudioFormat& format) {
+  const auto ptime = GetFormatParameter<int>(format, "ptime");
+  if (ptime) {
+    // Pick the next highest supported frame length from
+    // kOpusSupportedFrameLengths.
+    for (const int supported_frame_length : kOpusSupportedFrameLengths) {
+      if (supported_frame_length >= *ptime) {
+        return supported_frame_length;
+      }
+    }
+    // If none was found, return the largest supported frame length.
+    return *(std::end(kOpusSupportedFrameLengths) - 1);
+  }
+
+  return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+}
+
+void FindSupportedFrameLengths(int min_frame_length_ms,
+                               int max_frame_length_ms,
+                               std::vector<int>* out) {
+  out->clear();
+  std::copy_if(std::begin(kANASupportedFrameLengths),
+               std::end(kANASupportedFrameLengths), std::back_inserter(*out),
+               [&](int frame_length_ms) {
+                 return frame_length_ms >= min_frame_length_ms &&
+                        frame_length_ms <= max_frame_length_ms;
+               });
+  RTC_DCHECK(std::is_sorted(out->begin(), out->end()));
+}
+
+int GetBitrateBps(const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return *config.bitrate_bps;
+}
+
+std::vector<float> GetBitrateMultipliers() {
+  constexpr char kBitrateMultipliersName[] =
+      "WebRTC-Audio-OpusBitrateMultipliers";
+  const bool use_bitrate_multipliers =
+      webrtc::field_trial::IsEnabled(kBitrateMultipliersName);
+  if (use_bitrate_multipliers) {
+    const std::string field_trial_string =
+        webrtc::field_trial::FindFullName(kBitrateMultipliersName);
+    std::vector<std::string> pieces;
+    rtc::tokenize(field_trial_string, '-', &pieces);
+    if (pieces.size() < 2 || pieces[0] != "Enabled") {
+      RTC_LOG(LS_WARNING) << "Invalid parameters for "
+                          << kBitrateMultipliersName
+                          << ", not using custom values.";
+      return std::vector<float>();
+    }
+    std::vector<float> multipliers(pieces.size() - 1);
+    for (size_t i = 1; i < pieces.size(); i++) {
+      if (!rtc::FromString(pieces[i], &multipliers[i - 1])) {
+        RTC_LOG(LS_WARNING)
+            << "Invalid parameters for " << kBitrateMultipliersName
+            << ", not using custom values.";
+        return std::vector<float>();
+      }
+    }
+    RTC_LOG(LS_INFO) << "Using custom bitrate multipliers: "
+                     << field_trial_string;
+    return multipliers;
+  }
+  return std::vector<float>();
+}
+
+int GetMultipliedBitrate(int bitrate, const std::vector<float>& multipliers) {
+  // The multipliers are valid from 5 kbps.
+  const size_t bitrate_kbps = static_cast<size_t>(bitrate / 1000);
+  if (bitrate_kbps < 5 || bitrate_kbps >= multipliers.size() + 5) {
+    return bitrate;
+  }
+  return static_cast<int>(multipliers[bitrate_kbps - 5] * bitrate);
+}
+}  // namespace
+
+void AudioEncoderOpusImpl::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"opus",
+                              kRtpTimestampRateHz,
+                              2,
+                              {{"minptime", "10"}, {"useinbandfec", "1"}}};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderOpusImpl::QueryAudioEncoder(
+    const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  AudioCodecInfo info(config.sample_rate_hz, config.num_channels,
+                      *config.bitrate_bps,
+                      AudioEncoderOpusConfig::kMinBitrateBps,
+                      AudioEncoderOpusConfig::kMaxBitrateBps);
+  info.allow_comfort_noise = false;
+  info.supports_network_adaption = true;
+  return info;
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
+    const AudioEncoderOpusConfig& config,
+    int payload_type) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderOpusImpl>(config, payload_type);
+}
+
+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "opus") ||
+      format.clockrate_hz != kRtpTimestampRateHz) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderOpusConfig config;
+  config.num_channels = GetChannelCount(format);
+  config.frame_size_ms = GetFrameSizeMs(format);
+  config.max_playback_rate_hz = GetMaxPlaybackRate(format);
+  config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
+  config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
+  config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
+  config.bitrate_bps =
+      CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
+                       GetFormatParameter(format, "maxaveragebitrate"));
+  config.application = config.num_channels == 1
+                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+
+  constexpr int kMinANAFrameLength = kANASupportedFrameLengths[0];
+  constexpr int kMaxANAFrameLength =
+      kANASupportedFrameLengths[arraysize(kANASupportedFrameLengths) - 1];
+
+  // For now, minptime and maxptime are only used with ANA. If ptime is outside
+  // of this range, it will get adjusted once ANA takes hold. Ideally, we'd know
+  // if ANA was to be used when setting up the config, and adjust accordingly.
+  const int min_frame_length_ms =
+      GetFormatParameter<int>(format, "minptime").value_or(kMinANAFrameLength);
+  const int max_frame_length_ms =
+      GetFormatParameter<int>(format, "maxptime").value_or(kMaxANAFrameLength);
+
+  FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
+                            &config.supported_frame_lengths_ms);
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+absl::optional<int> AudioEncoderOpusImpl::GetNewComplexity(
+    const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  const int bitrate_bps = GetBitrateBps(config);
+  if (bitrate_bps >= config.complexity_threshold_bps -
+                         config.complexity_threshold_window_bps &&
+      bitrate_bps <= config.complexity_threshold_bps +
+                         config.complexity_threshold_window_bps) {
+    // Within the hysteresis window; make no change.
+    return absl::nullopt;
+  } else {
+    return bitrate_bps <= config.complexity_threshold_bps
+               ? config.low_rate_complexity
+               : config.complexity;
+  }
+}
+
+absl::optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
+    const AudioEncoderOpusConfig& config,
+    OpusEncInst* inst) {
+  constexpr int kMinWidebandBitrate = 8000;
+  constexpr int kMaxNarrowbandBitrate = 9000;
+  constexpr int kAutomaticThreshold = 11000;
+  RTC_DCHECK(config.IsOk());
+  const int bitrate = GetBitrateBps(config);
+  if (bitrate > kAutomaticThreshold) {
+    return absl::optional<int>(OPUS_AUTO);
+  }
+  const int bandwidth = WebRtcOpus_GetBandwidth(inst);
+  RTC_DCHECK_GE(bandwidth, 0);
+  if (bitrate > kMaxNarrowbandBitrate && bandwidth < OPUS_BANDWIDTH_WIDEBAND) {
+    return absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND);
+  } else if (bitrate < kMinWidebandBitrate &&
+             bandwidth > OPUS_BANDWIDTH_NARROWBAND) {
+    return absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND);
+  }
+  return absl::optional<int>();
+}
+
+class AudioEncoderOpusImpl::PacketLossFractionSmoother {
+ public:
+  explicit PacketLossFractionSmoother()
+      : last_sample_time_ms_(rtc::TimeMillis()),
+        smoother_(kAlphaForPacketLossFractionSmoother) {}
+
+  // Gets the smoothed packet loss fraction.
+  float GetAverage() const {
+    float value = smoother_.filtered();
+    return (value == rtc::ExpFilter::kValueUndefined) ? 0.0f : value;
+  }
+
+  // Add new observation to the packet loss fraction smoother.
+  void AddSample(float packet_loss_fraction) {
+    int64_t now_ms = rtc::TimeMillis();
+    smoother_.Apply(static_cast<float>(now_ms - last_sample_time_ms_),
+                    packet_loss_fraction);
+    last_sample_time_ms_ = now_ms;
+  }
+
+ private:
+  int64_t last_sample_time_ms_;
+
+  // An exponential filter is used to smooth the packet loss fraction.
+  rtc::ExpFilter smoother_;
+};
+
+AudioEncoderOpusImpl::AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config,
+                                           int payload_type)
+    : AudioEncoderOpusImpl(
+          config,
+          payload_type,
+          [this](absl::string_view config_string, RtcEventLog* event_log) {
+            return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
+          },
+          // We choose 5sec as initial time constant due to empirical data.
+          std::make_unique<SmoothingFilterImpl>(5000)) {}
+
+AudioEncoderOpusImpl::AudioEncoderOpusImpl(
+    const AudioEncoderOpusConfig& config,
+    int payload_type,
+    const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
+    std::unique_ptr<SmoothingFilter> bitrate_smoother)
+    : payload_type_(payload_type),
+      use_stable_target_for_adaptation_(!webrtc::field_trial::IsDisabled(
+          "WebRTC-Audio-StableTargetAdaptation")),
+      adjust_bandwidth_(
+          webrtc::field_trial::IsEnabled("WebRTC-AdjustOpusBandwidth")),
+      bitrate_changed_(true),
+      bitrate_multipliers_(GetBitrateMultipliers()),
+      packet_loss_rate_(0.0),
+      inst_(nullptr),
+      packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
+      audio_network_adaptor_creator_(audio_network_adaptor_creator),
+      bitrate_smoother_(std::move(bitrate_smoother)),
+      consecutive_dtx_frames_(0) {
+  RTC_DCHECK(0 <= payload_type && payload_type <= 127);
+
+  // Sanity check of the redundant payload type field that we want to get rid
+  // of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
+
+  RTC_CHECK(RecreateEncoderInstance(config));
+  SetProjectedPacketLossRate(packet_loss_rate_);
+}
+
+AudioEncoderOpusImpl::AudioEncoderOpusImpl(int payload_type,
+                                           const SdpAudioFormat& format)
+    : AudioEncoderOpusImpl(*SdpToConfig(format), payload_type) {}
+
+AudioEncoderOpusImpl::~AudioEncoderOpusImpl() {
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+}
+
+int AudioEncoderOpusImpl::SampleRateHz() const {
+  return config_.sample_rate_hz;
+}
+
+size_t AudioEncoderOpusImpl::NumChannels() const {
+  return config_.num_channels;
+}
+
+int AudioEncoderOpusImpl::RtpTimestampRateHz() const {
+  return kRtpTimestampRateHz;
+}
+
+size_t AudioEncoderOpusImpl::Num10MsFramesInNextPacket() const {
+  return Num10msFramesPerPacket();
+}
+
+size_t AudioEncoderOpusImpl::Max10MsFramesInAPacket() const {
+  return Num10msFramesPerPacket();
+}
+
+int AudioEncoderOpusImpl::GetTargetBitrate() const {
+  return GetBitrateBps(config_);
+}
+
+void AudioEncoderOpusImpl::Reset() {
+  RTC_CHECK(RecreateEncoderInstance(config_));
+}
+
+bool AudioEncoderOpusImpl::SetFec(bool enable) {
+  if (enable) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+  }
+  config_.fec_enabled = enable;
+  return true;
+}
+
+bool AudioEncoderOpusImpl::SetDtx(bool enable) {
+  if (enable) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+  }
+  config_.dtx_enabled = enable;
+  return true;
+}
+
+bool AudioEncoderOpusImpl::GetDtx() const {
+  return config_.dtx_enabled;
+}
+
+bool AudioEncoderOpusImpl::SetApplication(Application application) {
+  auto conf = config_;
+  switch (application) {
+    case Application::kSpeech:
+      conf.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
+      break;
+    case Application::kAudio:
+      conf.application = AudioEncoderOpusConfig::ApplicationMode::kAudio;
+      break;
+  }
+  return RecreateEncoderInstance(conf);
+}
+
+void AudioEncoderOpusImpl::SetMaxPlaybackRate(int frequency_hz) {
+  auto conf = config_;
+  conf.max_playback_rate_hz = frequency_hz;
+  RTC_CHECK(RecreateEncoderInstance(conf));
+}
+
+bool AudioEncoderOpusImpl::EnableAudioNetworkAdaptor(
+    const std::string& config_string,
+    RtcEventLog* event_log) {
+  audio_network_adaptor_ =
+      audio_network_adaptor_creator_(config_string, event_log);
+  return audio_network_adaptor_.get() != nullptr;
+}
+
+void AudioEncoderOpusImpl::DisableAudioNetworkAdaptor() {
+  audio_network_adaptor_.reset(nullptr);
+}
+
+void AudioEncoderOpusImpl::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  if (audio_network_adaptor_) {
+    audio_network_adaptor_->SetUplinkPacketLossFraction(
+        uplink_packet_loss_fraction);
+    ApplyAudioNetworkAdaptor();
+  }
+  packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction);
+  float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage();
+  SetProjectedPacketLossRate(average_fraction_loss);
+}
+
+void AudioEncoderOpusImpl::OnReceivedTargetAudioBitrate(
+    int target_audio_bitrate_bps) {
+  SetTargetBitrate(target_audio_bitrate_bps);
+}
+
+void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms,
+    absl::optional<int64_t> stable_target_bitrate_bps) {
+  if (audio_network_adaptor_) {
+    audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
+    if (use_stable_target_for_adaptation_) {
+      if (stable_target_bitrate_bps)
+        audio_network_adaptor_->SetUplinkBandwidth(*stable_target_bitrate_bps);
+    } else {
+      // We give smoothed bitrate allocation to audio network adaptor as
+      // the uplink bandwidth.
+      // The BWE spikes should not affect the bitrate smoother more than 25%.
+      // To simplify the calculations we use a step response as input signal.
+      // The step response of an exponential filter is
+      // u(t) = 1 - e^(-t / time_constant).
+      // In order to limit the affect of a BWE spike within 25% of its value
+      // before
+      // the next BWE update, we would choose a time constant that fulfills
+      // 1 - e^(-bwe_period_ms / time_constant) < 0.25
+      // Then 4 * bwe_period_ms is a good choice.
+      if (bwe_period_ms)
+        bitrate_smoother_->SetTimeConstantMs(*bwe_period_ms * 4);
+      bitrate_smoother_->AddSample(target_audio_bitrate_bps);
+    }
+
+    ApplyAudioNetworkAdaptor();
+  } else {
+    if (!overhead_bytes_per_packet_) {
+      RTC_LOG(LS_INFO)
+          << "AudioEncoderOpusImpl: Overhead unknown, target audio bitrate "
+          << target_audio_bitrate_bps << " bps is ignored.";
+      return;
+    }
+    const int overhead_bps = static_cast<int>(
+        *overhead_bytes_per_packet_ * 8 * 100 / Num10MsFramesInNextPacket());
+    SetTargetBitrate(
+        std::min(AudioEncoderOpusConfig::kMaxBitrateBps,
+                 std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                          target_audio_bitrate_bps - overhead_bps)));
+  }
+}
+void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {
+  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, bwe_period_ms,
+                            absl::nullopt);
+}
+
+void AudioEncoderOpusImpl::OnReceivedUplinkAllocation(
+    BitrateAllocationUpdate update) {
+  OnReceivedUplinkBandwidth(update.target_bitrate.bps(), update.bwe_period.ms(),
+                            update.stable_target_bitrate.bps());
+}
+
+void AudioEncoderOpusImpl::OnReceivedRtt(int rtt_ms) {
+  if (!audio_network_adaptor_)
+    return;
+  audio_network_adaptor_->SetRtt(rtt_ms);
+  ApplyAudioNetworkAdaptor();
+}
+
+void AudioEncoderOpusImpl::OnReceivedOverhead(
+    size_t overhead_bytes_per_packet) {
+  if (audio_network_adaptor_) {
+    audio_network_adaptor_->SetOverhead(overhead_bytes_per_packet);
+    ApplyAudioNetworkAdaptor();
+  } else {
+    overhead_bytes_per_packet_ = overhead_bytes_per_packet;
+  }
+}
+
+void AudioEncoderOpusImpl::SetReceiverFrameLengthRange(
+    int min_frame_length_ms,
+    int max_frame_length_ms) {
+  // Ensure that `SetReceiverFrameLengthRange` is called before
+  // `EnableAudioNetworkAdaptor`, otherwise we need to recreate
+  // `audio_network_adaptor_`, which is not a needed use case.
+  RTC_DCHECK(!audio_network_adaptor_);
+  FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
+                            &config_.supported_frame_lengths_ms);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  MaybeUpdateUplinkBandwidth();
+
+  if (input_buffer_.empty())
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
+  if (input_buffer_.size() <
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
+
+  const size_t max_encoded_bytes = SufficientOutputBufferSize();
+  EncodedInfo info;
+  info.encoded_bytes = encoded->AppendData(
+      max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
+        int status = WebRtcOpus_Encode(
+            inst_, &input_buffer_[0],
+            rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
+            rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
+
+        RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
+
+        return static_cast<size_t>(status);
+      });
+  input_buffer_.clear();
+
+  bool dtx_frame = (info.encoded_bytes <= 2);
+
+  // Will use new packet size for next encoding.
+  config_.frame_size_ms = next_frame_length_ms_;
+
+  if (adjust_bandwidth_ && bitrate_changed_) {
+    const auto bandwidth = GetNewBandwidth(config_, inst_);
+    if (bandwidth) {
+      RTC_CHECK_EQ(0, WebRtcOpus_SetBandwidth(inst_, *bandwidth));
+    }
+    bitrate_changed_ = false;
+  }
+
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.send_even_if_empty = true;  // Allows Opus to send empty packets.
+  // After 20 DTX frames (MAX_CONSECUTIVE_DTX) Opus will send a frame
+  // coding the background noise. Avoid flagging this frame as speech
+  // (even though there is a probability of the frame being speech).
+  info.speech = !dtx_frame && (consecutive_dtx_frames_ != 20);
+  info.encoder_type = CodecType::kOpus;
+
+  // Increase or reset DTX counter.
+  consecutive_dtx_frames_ = (dtx_frame) ? (consecutive_dtx_frames_ + 1) : (0);
+
+  return info;
+}
+
+size_t AudioEncoderOpusImpl::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
+}
+
+size_t AudioEncoderOpusImpl::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(config_.sample_rate_hz, 100) *
+         config_.num_channels;
+}
+
+size_t AudioEncoderOpusImpl::SufficientOutputBufferSize() const {
+  // Calculate the number of bytes we expect the encoder to produce,
+  // then multiply by two to give a wide margin for error.
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
+  return 2 * approx_encoded_bytes;
+}
+
+// If the given config is OK, recreate the Opus encoder instance with those
+// settings, save the config, and return true. Otherwise, do nothing and return
+// false.
+bool AudioEncoderOpusImpl::RecreateEncoderInstance(
+    const AudioEncoderOpusConfig& config) {
+  if (!config.IsOk())
+    return false;
+  config_ = config;
+  if (inst_)
+    RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+  input_buffer_.clear();
+  input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(
+                      &inst_, config.num_channels,
+                      config.application ==
+                              AudioEncoderOpusConfig::ApplicationMode::kVoip
+                          ? 0
+                          : 1,
+                      config.sample_rate_hz));
+  const int bitrate = GetBitrateBps(config);
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, bitrate));
+  RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
+  if (config.fec_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+  }
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
+  // Use the default complexity if the start bitrate is within the hysteresis
+  // window.
+  complexity_ = GetNewComplexity(config).value_or(config.complexity);
+  RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
+  bitrate_changed_ = true;
+  if (config.dtx_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+  }
+  RTC_CHECK_EQ(0,
+               WebRtcOpus_SetPacketLossRate(
+                   inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  if (config.cbr_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_));
+  }
+  num_channels_to_encode_ = NumChannels();
+  next_frame_length_ms_ = config_.frame_size_ms;
+  return true;
+}
+
+void AudioEncoderOpusImpl::SetFrameLength(int frame_length_ms) {
+  if (next_frame_length_ms_ != frame_length_ms) {
+    RTC_LOG(LS_VERBOSE) << "Update Opus frame length "
+                        << "from " << next_frame_length_ms_ << " ms "
+                        << "to " << frame_length_ms << " ms.";
+  }
+  next_frame_length_ms_ = frame_length_ms;
+}
+
+void AudioEncoderOpusImpl::SetNumChannelsToEncode(
+    size_t num_channels_to_encode) {
+  RTC_DCHECK_GT(num_channels_to_encode, 0);
+  RTC_DCHECK_LE(num_channels_to_encode, config_.num_channels);
+
+  if (num_channels_to_encode_ == num_channels_to_encode)
+    return;
+
+  RTC_CHECK_EQ(0, WebRtcOpus_SetForceChannels(inst_, num_channels_to_encode));
+  num_channels_to_encode_ = num_channels_to_encode;
+}
+
+void AudioEncoderOpusImpl::SetProjectedPacketLossRate(float fraction) {
+  fraction = std::min(std::max(fraction, 0.0f), kMaxPacketLossFraction);
+  if (packet_loss_rate_ != fraction) {
+    packet_loss_rate_ = fraction;
+    RTC_CHECK_EQ(
+        0, WebRtcOpus_SetPacketLossRate(
+               inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  }
+}
+
+void AudioEncoderOpusImpl::SetTargetBitrate(int bits_per_second) {
+  const int new_bitrate = rtc::SafeClamp<int>(
+      bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
+      AudioEncoderOpusConfig::kMaxBitrateBps);
+  if (config_.bitrate_bps && *config_.bitrate_bps != new_bitrate) {
+    config_.bitrate_bps = new_bitrate;
+    RTC_DCHECK(config_.IsOk());
+    const int bitrate = GetBitrateBps(config_);
+    RTC_CHECK_EQ(
+        0, WebRtcOpus_SetBitRate(
+               inst_, GetMultipliedBitrate(bitrate, bitrate_multipliers_)));
+    RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
+    bitrate_changed_ = true;
+  }
+
+  const auto new_complexity = GetNewComplexity(config_);
+  if (new_complexity && complexity_ != *new_complexity) {
+    complexity_ = *new_complexity;
+    RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
+  }
+}
+
+void AudioEncoderOpusImpl::ApplyAudioNetworkAdaptor() {
+  auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
+
+  if (config.bitrate_bps)
+    SetTargetBitrate(*config.bitrate_bps);
+  if (config.frame_length_ms)
+    SetFrameLength(*config.frame_length_ms);
+  if (config.enable_dtx)
+    SetDtx(*config.enable_dtx);
+  if (config.num_channels)
+    SetNumChannelsToEncode(*config.num_channels);
+}
+
+std::unique_ptr<AudioNetworkAdaptor>
+AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator(
+    absl::string_view config_string,
+    RtcEventLog* event_log) const {
+  AudioNetworkAdaptorImpl::Config config;
+  config.event_log = event_log;
+  return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
+      config, ControllerManagerImpl::Create(
+                  config_string, NumChannels(), supported_frame_lengths_ms(),
+                  AudioEncoderOpusConfig::kMinBitrateBps,
+                  num_channels_to_encode_, next_frame_length_ms_,
+                  GetTargetBitrate(), config_.fec_enabled, GetDtx())));
+}
+
+void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
+  if (audio_network_adaptor_ && !use_stable_target_for_adaptation_) {
+    int64_t now_ms = rtc::TimeMillis();
+    if (!bitrate_smoother_last_update_time_ ||
+        now_ms - *bitrate_smoother_last_update_time_ >=
+            config_.uplink_bandwidth_update_interval_ms) {
+      absl::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
+      if (smoothed_bitrate)
+        audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
+      bitrate_smoother_last_update_time_ = now_ms;
+    }
+  }
+}
+
+ANAStats AudioEncoderOpusImpl::GetANAStats() const {
+  if (audio_network_adaptor_) {
+    return audio_network_adaptor_->GetStats();
+  }
+  return ANAStats();
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta> >
+AudioEncoderOpusImpl::GetFrameLengthRange() const {
+  if (audio_network_adaptor_) {
+    if (config_.supported_frame_lengths_ms.empty()) {
+      return absl::nullopt;
+    }
+    return {{TimeDelta::Millis(config_.supported_frame_lengths_ms.front()),
+             TimeDelta::Millis(config_.supported_frame_lengths_ms.back())}};
+  } else {
+    return {{TimeDelta::Millis(config_.frame_size_ms),
+             TimeDelta::Millis(config_.frame_size_ms)}};
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
new file mode 100644
index 0000000000..8c5c235016
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include <functional>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "common_audio/smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class AudioEncoderOpusImpl final : public AudioEncoder {
+ public:
+  // Returns empty if the current bitrate falls within the hysteresis window,
+  // defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
+  // Otherwise, returns the current complexity depending on whether the
+  // current bitrate is above or below complexity_threshold_bps.
+  static absl::optional<int> GetNewComplexity(
+      const AudioEncoderOpusConfig& config);
+
+  // Returns OPUS_AUTO if the the current bitrate is above wideband threshold.
+  // Returns empty if it is below, but bandwidth coincides with the desired one.
+  // Otherwise returns the desired bandwidth.
+  static absl::optional<int> GetNewBandwidth(
+      const AudioEncoderOpusConfig& config,
+      OpusEncInst* inst);
+
+  using AudioNetworkAdaptorCreator =
+      std::function<std::unique_ptr<AudioNetworkAdaptor>(absl::string_view,
+                                                         RtcEventLog*)>;
+
+  AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config, int payload_type);
+
+  // Dependency injection for testing.
+  AudioEncoderOpusImpl(
+      const AudioEncoderOpusConfig& config,
+      int payload_type,
+      const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
+      std::unique_ptr<SmoothingFilter> bitrate_smoother);
+
+  AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format);
+  ~AudioEncoderOpusImpl() override;
+
+  AudioEncoderOpusImpl(const AudioEncoderOpusImpl&) = delete;
+  AudioEncoderOpusImpl& operator=(const AudioEncoderOpusImpl&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  bool SetFec(bool enable) override;
+
+  // Set Opus DTX. Once enabled, Opus stops transmission, when it detects
+  // voice being inactive. During that, it still sends 2 packets (one for
+  // content, one for signaling) about every 400 ms.
+  bool SetDtx(bool enable) override;
+  bool GetDtx() const override;
+
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                 RtcEventLog* event_log) override;
+  void DisableAudioNetworkAdaptor() override;
+  void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) override;
+  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms) override;
+  void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
+  void OnReceivedRtt(int rtt_ms) override;
+  void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
+  void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                   int max_frame_length_ms) override;
+  ANAStats GetANAStats() const override;
+  absl::optional<std::pair<TimeDelta, TimeDelta> > GetFrameLengthRange()
+      const override;
+  rtc::ArrayView<const int> supported_frame_lengths_ms() const {
+    return config_.supported_frame_lengths_ms;
+  }
+
+  // Getters for testing.
+  float packet_loss_rate() const { return packet_loss_rate_; }
+  AudioEncoderOpusConfig::ApplicationMode application() const {
+    return config_.application;
+  }
+  bool fec_enabled() const { return config_.fec_enabled; }
+  size_t num_channels_to_encode() const { return num_channels_to_encode_; }
+  int next_frame_length_ms() const { return next_frame_length_ms_; }
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  class PacketLossFractionSmoother;
+
+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderOpusConfig&,
+      int payload_type);
+
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
+  bool RecreateEncoderInstance(const AudioEncoderOpusConfig& config);
+  void SetFrameLength(int frame_length_ms);
+  void SetNumChannelsToEncode(size_t num_channels_to_encode);
+  void SetProjectedPacketLossRate(float fraction);
+
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms,
+      absl::optional<int64_t> link_capacity_allocation);
+
+  // TODO(minyue): remove "override" when we can deprecate
+  // `AudioEncoder::SetTargetBitrate`.
+  void SetTargetBitrate(int target_bps) override;
+
+  void ApplyAudioNetworkAdaptor();
+  std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
+      absl::string_view config_string,
+      RtcEventLog* event_log) const;
+
+  void MaybeUpdateUplinkBandwidth();
+
+  AudioEncoderOpusConfig config_;
+  const int payload_type_;
+  const bool use_stable_target_for_adaptation_;
+  const bool adjust_bandwidth_;
+  bool bitrate_changed_;
+  // A multiplier for bitrates at 5 kbps and higher. The target bitrate
+  // will be multiplied by these multipliers, each multiplier is applied to a
+  // 1 kbps range.
+  std::vector<float> bitrate_multipliers_;
+  float packet_loss_rate_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  size_t num_channels_to_encode_;
+  int next_frame_length_ms_;
+  int complexity_;
+  std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
+  const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
+  std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
+  absl::optional<size_t> overhead_bytes_per_packet_;
+  const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
+  absl::optional<int64_t> bitrate_smoother_last_update_time_;
+  int consecutive_dtx_frames_;
+
+  friend struct AudioEncoderOpus;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
new file mode 100644
index 0000000000..a2ebe43bbe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -0,0 +1,914 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+
+#include <array>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/mocks/mock_smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fake_clock.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace {
+
+constexpr int kDefaultOpusPayloadType = 105;
+constexpr int kDefaultOpusRate = 32000;
+constexpr int kDefaultOpusPacSize = 960;
+constexpr int64_t kInitialTimeUs = 12345678;
+
+AudioEncoderOpusConfig CreateConfigWithParameters(
+    const SdpAudioFormat::Parameters& params) {
+  const SdpAudioFormat format("opus", 48000, 2, params);
+  return *AudioEncoderOpus::SdpToConfig(format);
+}
+
+struct AudioEncoderOpusStates {
+  MockAudioNetworkAdaptor* mock_audio_network_adaptor;
+  MockSmoothingFilter* mock_bitrate_smoother;
+  std::unique_ptr<AudioEncoderOpusImpl> encoder;
+  std::unique_ptr<rtc::ScopedFakeClock> fake_clock;
+  AudioEncoderOpusConfig config;
+};
+
+std::unique_ptr<AudioEncoderOpusStates> CreateCodec(int sample_rate_hz,
+                                                    size_t num_channels) {
+  std::unique_ptr<AudioEncoderOpusStates> states =
+      std::make_unique<AudioEncoderOpusStates>();
+  states->mock_audio_network_adaptor = nullptr;
+  states->fake_clock.reset(new rtc::ScopedFakeClock());
+  states->fake_clock->SetTime(Timestamp::Micros(kInitialTimeUs));
+
+  MockAudioNetworkAdaptor** mock_ptr = &states->mock_audio_network_adaptor;
+  AudioEncoderOpusImpl::AudioNetworkAdaptorCreator creator =
+      [mock_ptr](absl::string_view, RtcEventLog* event_log) {
+        std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
+            new NiceMock<MockAudioNetworkAdaptor>());
+        EXPECT_CALL(*adaptor, Die());
+        *mock_ptr = adaptor.get();
+        return adaptor;
+      };
+
+  AudioEncoderOpusConfig config;
+  config.frame_size_ms = rtc::CheckedDivExact(kDefaultOpusPacSize, 48);
+  config.sample_rate_hz = sample_rate_hz;
+  config.num_channels = num_channels;
+  config.bitrate_bps = kDefaultOpusRate;
+  config.application = num_channels == 1
+                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+  config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
+  states->config = config;
+
+  std::unique_ptr<MockSmoothingFilter> bitrate_smoother(
+      new MockSmoothingFilter());
+  states->mock_bitrate_smoother = bitrate_smoother.get();
+
+  states->encoder.reset(
+      new AudioEncoderOpusImpl(states->config, kDefaultOpusPayloadType, creator,
+                               std::move(bitrate_smoother)));
+  return states;
+}
+
+AudioEncoderRuntimeConfig CreateEncoderRuntimeConfig() {
+  constexpr int kBitrate = 40000;
+  constexpr int kFrameLength = 60;
+  constexpr bool kEnableDtx = false;
+  constexpr size_t kNumChannels = 1;
+  AudioEncoderRuntimeConfig config;
+  config.bitrate_bps = kBitrate;
+  config.frame_length_ms = kFrameLength;
+  config.enable_dtx = kEnableDtx;
+  config.num_channels = kNumChannels;
+  return config;
+}
+
+void CheckEncoderRuntimeConfig(const AudioEncoderOpusImpl* encoder,
+                               const AudioEncoderRuntimeConfig& config) {
+  EXPECT_EQ(*config.bitrate_bps, encoder->GetTargetBitrate());
+  EXPECT_EQ(*config.frame_length_ms, encoder->next_frame_length_ms());
+  EXPECT_EQ(*config.enable_dtx, encoder->GetDtx());
+  EXPECT_EQ(*config.num_channels, encoder->num_channels_to_encode());
+}
+
+// Create 10ms audio data blocks for a total packet size of "packet_size_ms".
+std::unique_ptr<test::AudioLoop> Create10msAudioBlocks(
+    const std::unique_ptr<AudioEncoderOpusImpl>& encoder,
+    int packet_size_ms) {
+  const std::string file_name =
+      test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+
+  std::unique_ptr<test::AudioLoop> speech_data(new test::AudioLoop());
+  int audio_samples_per_ms =
+      rtc::CheckedDivExact(encoder->SampleRateHz(), 1000);
+  if (!speech_data->Init(
+          file_name,
+          packet_size_ms * audio_samples_per_ms *
+              encoder->num_channels_to_encode(),
+          10 * audio_samples_per_ms * encoder->num_channels_to_encode()))
+    return nullptr;
+  return speech_data;
+}
+
+}  // namespace
+
+class AudioEncoderOpusTest : public ::testing::TestWithParam<int> {
+ protected:
+  int sample_rate_hz_{GetParam()};
+};
+INSTANTIATE_TEST_SUITE_P(Param,
+                         AudioEncoderOpusTest,
+                         ::testing::Values(16000, 48000));
+
+TEST_P(AudioEncoderOpusTest, DefaultApplicationModeMono) {
+  auto states = CreateCodec(sample_rate_hz_, 1);
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, ChangeApplicationMode) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  EXPECT_TRUE(
+      states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  // Trigger a reset.
+  states->encoder->Reset();
+  // Verify that the mode is still kAudio.
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
+            states->encoder->application());
+
+  // Now change to kVoip.
+  EXPECT_TRUE(
+      states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+
+  // Trigger a reset again.
+  states->encoder->Reset();
+  // Verify that the mode is still kVoip.
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, ToggleDtx) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  // Enable DTX
+  EXPECT_TRUE(states->encoder->SetDtx(true));
+  EXPECT_TRUE(states->encoder->GetDtx());
+  // Turn off DTX.
+  EXPECT_TRUE(states->encoder->SetDtx(false));
+  EXPECT_FALSE(states->encoder->GetDtx());
+}
+
+TEST_P(AudioEncoderOpusTest,
+       OnReceivedUplinkBandwidthWithoutAudioNetworkAdaptor) {
+  auto states = CreateCodec(sample_rate_hz_, 1);
+  // Constants are replicated from audio_states->encoderopus.cc.
+  const int kMinBitrateBps = 6000;
+  const int kMaxBitrateBps = 510000;
+  const int kOverheadBytesPerPacket = 64;
+  states->encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
+  const int kOverheadBps = 8 * kOverheadBytesPerPacket *
+                           rtc::CheckedDivExact(48000, kDefaultOpusPacSize);
+  // Set a too low bitrate.
+  states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps - 1,
+                                             absl::nullopt);
+  EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
+  // Set a too high bitrate.
+  states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps + 1,
+                                             absl::nullopt);
+  EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
+  // Set the minimum rate.
+  states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps,
+                                             absl::nullopt);
+  EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
+  // Set the maximum rate.
+  states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps,
+                                             absl::nullopt);
+  EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
+  // Set rates from kMaxBitrateBps up to 32000 bps.
+  for (int rate = kMinBitrateBps + kOverheadBps; rate <= 32000 + kOverheadBps;
+       rate += 1000) {
+    states->encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
+    EXPECT_EQ(rate - kOverheadBps, states->encoder->GetTargetBitrate());
+  }
+}
+
+TEST_P(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  // Before calling to `SetReceiverFrameLengthRange`,
+  // `supported_frame_lengths_ms` should contain only the frame length being
+  // used.
+  using ::testing::ElementsAre;
+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+              ElementsAre(states->encoder->next_frame_length_ms()));
+  states->encoder->SetReceiverFrameLengthRange(0, 12345);
+  states->encoder->SetReceiverFrameLengthRange(21, 60);
+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+              ElementsAre(40, 60));
+  states->encoder->SetReceiverFrameLengthRange(20, 59);
+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+              ElementsAre(20, 40));
+}
+
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any packet loss fraction is fine.
+  constexpr float kUplinkPacketLoss = 0.1f;
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetUplinkPacketLossFraction(kUplinkPacketLoss));
+  states->encoder->OnReceivedUplinkPacketLossFraction(kUplinkPacketLoss);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any target audio bitrate is fine.
+  constexpr int kTargetAudioBitrate = 30000;
+  constexpr int64_t kProbingIntervalMs = 3000;
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetTargetAudioBitrate(kTargetAudioBitrate));
+  EXPECT_CALL(*states->mock_bitrate_smoother,
+              SetTimeConstantMs(kProbingIntervalMs * 4));
+  EXPECT_CALL(*states->mock_bitrate_smoother, AddSample(kTargetAudioBitrate));
+  states->encoder->OnReceivedUplinkBandwidth(kTargetAudioBitrate,
+                                             kProbingIntervalMs);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkAllocation) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  BitrateAllocationUpdate update;
+  update.target_bitrate = DataRate::BitsPerSec(30000);
+  update.stable_target_bitrate = DataRate::BitsPerSec(20000);
+  update.bwe_period = TimeDelta::Millis(200);
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetTargetAudioBitrate(update.target_bitrate.bps()));
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetUplinkBandwidth(update.stable_target_bitrate.bps()));
+  states->encoder->OnReceivedUplinkAllocation(update);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any rtt is fine.
+  constexpr int kRtt = 30;
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetRtt(kRtt));
+  states->encoder->OnReceivedRtt(kRtt);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedOverhead) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any overhead is fine.
+  constexpr size_t kOverhead = 64;
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead));
+  states->encoder->OnReceivedOverhead(kOverhead);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest,
+       PacketLossFractionSmoothedOnSetUplinkPacketLossFraction) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  // The values are carefully chosen so that if no smoothing is made, the test
+  // will fail.
+  constexpr float kPacketLossFraction_1 = 0.02f;
+  constexpr float kPacketLossFraction_2 = 0.198f;
+  // `kSecondSampleTimeMs` is chosen to ease the calculation since
+  // 0.9999 ^ 6931 = 0.5.
+  constexpr int64_t kSecondSampleTimeMs = 6931;
+
+  // First time, no filtering.
+  states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
+  EXPECT_FLOAT_EQ(0.02f, states->encoder->packet_loss_rate());
+
+  states->fake_clock->AdvanceTime(TimeDelta::Millis(kSecondSampleTimeMs));
+  states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
+
+  // Now the output of packet loss fraction smoother should be
+  // (0.02 + 0.198) / 2 = 0.109.
+  EXPECT_NEAR(0.109f, states->encoder->packet_loss_rate(), 0.001);
+}
+
+TEST_P(AudioEncoderOpusTest, PacketLossRateUpperBounded) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  states->encoder->OnReceivedUplinkPacketLossFraction(0.5);
+  EXPECT_FLOAT_EQ(0.2f, states->encoder->packet_loss_rate());
+}
+
+TEST_P(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusRate * 2,
+                                             absl::nullopt);
+
+  // Since `OnReceivedOverhead` has not been called, the codec bitrate should
+  // not change.
+  EXPECT_EQ(kDefaultOpusRate, states->encoder->GetTargetBitrate());
+}
+
+// Verifies that the complexity adaptation in the config works as intended.
+TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
+  AudioEncoderOpusConfig config;
+  config.low_rate_complexity = 8;
+  config.complexity = 6;
+
+  // Bitrate within hysteresis window. Expect empty output.
+  config.bitrate_bps = 12500;
+  EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+
+  // Bitrate below hysteresis window. Expect higher complexity.
+  config.bitrate_bps = 10999;
+  EXPECT_EQ(8, AudioEncoderOpusImpl::GetNewComplexity(config));
+
+  // Bitrate within hysteresis window. Expect empty output.
+  config.bitrate_bps = 12500;
+  EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+
+  // Bitrate above hysteresis window. Expect lower complexity.
+  config.bitrate_bps = 14001;
+  EXPECT_EQ(6, AudioEncoderOpusImpl::GetNewComplexity(config));
+}
+
+// Verifies that the bandwidth adaptation in the config works as intended.
+TEST_P(AudioEncoderOpusTest, ConfigBandwidthAdaptation) {
+  AudioEncoderOpusConfig config;
+  const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);
+  const std::vector<int16_t> silence(
+      opus_rate_khz * config.frame_size_ms * config.num_channels, 0);
+  constexpr size_t kMaxBytes = 1000;
+  uint8_t bitstream[kMaxBytes];
+
+  OpusEncInst* inst;
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(
+                   &inst, config.num_channels,
+                   config.application ==
+                           AudioEncoderOpusConfig::ApplicationMode::kVoip
+                       ? 0
+                       : 1,
+                   sample_rate_hz_));
+
+  // Bitrate below minmum wideband. Expect narrowband.
+  config.bitrate_bps = absl::optional<int>(7999);
+  auto bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
+  WebRtcOpus_SetBandwidth(inst, *bandwidth);
+  // It is necessary to encode here because Opus has some logic in the encoder
+  // that goes from the user-set bandwidth to the used and returned one.
+  WebRtcOpus_Encode(inst, silence.data(),
+                    rtc::CheckedDivExact(silence.size(), config.num_channels),
+                    kMaxBytes, bitstream);
+
+  // Bitrate not yet above maximum narrowband. Expect empty.
+  config.bitrate_bps = absl::optional<int>(9000);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(), bandwidth);
+
+  // Bitrate above maximum narrowband. Expect wideband.
+  config.bitrate_bps = absl::optional<int>(9001);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
+  WebRtcOpus_SetBandwidth(inst, *bandwidth);
+  // It is necessary to encode here because Opus has some logic in the encoder
+  // that goes from the user-set bandwidth to the used and returned one.
+  WebRtcOpus_Encode(inst, silence.data(),
+                    rtc::CheckedDivExact(silence.size(), config.num_channels),
+                    kMaxBytes, bitstream);
+
+  // Bitrate not yet below minimum wideband. Expect empty.
+  config.bitrate_bps = absl::optional<int>(8000);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(), bandwidth);
+
+  // Bitrate above automatic threshold. Expect automatic.
+  config.bitrate_bps = absl::optional<int>(12001);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(OPUS_AUTO), bandwidth);
+
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(inst));
+}
+
+TEST_P(AudioEncoderOpusTest, EmptyConfigDoesNotAffectEncoderSettings) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig empty_config;
+
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config))
+      .WillOnce(Return(empty_config));
+
+  constexpr size_t kOverhead = 64;
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead))
+      .Times(2);
+  states->encoder->OnReceivedOverhead(kOverhead);
+  states->encoder->OnReceivedOverhead(kOverhead);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+  const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);
+  const std::vector<int16_t> audio(opus_rate_khz * 10 * 2, 0);
+  rtc::Buffer encoded;
+  EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
+      .WillOnce(Return(50000));
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(50000));
+  states->encoder->Encode(
+      0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+
+  // Repeat update uplink bandwidth tests.
+  for (int i = 0; i < 5; i++) {
+    // Don't update till it is time to update again.
+    states->fake_clock->AdvanceTime(TimeDelta::Millis(
+        states->config.uplink_bandwidth_update_interval_ms - 1));
+    states->encoder->Encode(
+        0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+
+    // Update when it is time to update.
+    EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
+        .WillOnce(Return(40000));
+    EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(40000));
+    states->fake_clock->AdvanceTime(TimeDelta::Millis(1));
+    states->encoder->Encode(
+        0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+  }
+}
+
+TEST_P(AudioEncoderOpusTest, EncodeAtMinBitrate) {
+  auto states = CreateCodec(sample_rate_hz_, 1);
+  constexpr int kNumPacketsToEncode = 2;
+  auto audio_frames =
+      Create10msAudioBlocks(states->encoder, kNumPacketsToEncode * 20);
+  ASSERT_TRUE(audio_frames) << "Create10msAudioBlocks failed";
+  rtc::Buffer encoded;
+  uint32_t rtp_timestamp = 12345;  // Just a number not important to this test.
+
+  states->encoder->OnReceivedUplinkBandwidth(0, absl::nullopt);
+  for (int packet_index = 0; packet_index < kNumPacketsToEncode;
+       packet_index++) {
+    // Make sure we are not encoding before we have enough data for
+    // a 20ms packet.
+    for (int index = 0; index < 1; index++) {
+      states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
+                              &encoded);
+      EXPECT_EQ(0u, encoded.size());
+    }
+
+    // Should encode now.
+    states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
+                            &encoded);
+    EXPECT_GT(encoded.size(), 0u);
+    encoded.Clear();
+  }
+}
+
+TEST(AudioEncoderOpusTest, TestConfigDefaults) {
+  const auto config_opt = AudioEncoderOpus::SdpToConfig({"opus", 48000, 2});
+  ASSERT_TRUE(config_opt);
+  EXPECT_EQ(48000, config_opt->max_playback_rate_hz);
+  EXPECT_EQ(1u, config_opt->num_channels);
+  EXPECT_FALSE(config_opt->fec_enabled);
+  EXPECT_FALSE(config_opt->dtx_enabled);
+  EXPECT_EQ(20, config_opt->frame_size_ms);
+}
+
+TEST(AudioEncoderOpusTest, TestConfigFromParams) {
+  const auto config1 = CreateConfigWithParameters({{"stereo", "0"}});
+  EXPECT_EQ(1U, config1.num_channels);
+
+  const auto config2 = CreateConfigWithParameters({{"stereo", "1"}});
+  EXPECT_EQ(2U, config2.num_channels);
+
+  const auto config3 = CreateConfigWithParameters({{"useinbandfec", "0"}});
+  EXPECT_FALSE(config3.fec_enabled);
+
+  const auto config4 = CreateConfigWithParameters({{"useinbandfec", "1"}});
+  EXPECT_TRUE(config4.fec_enabled);
+
+  const auto config5 = CreateConfigWithParameters({{"usedtx", "0"}});
+  EXPECT_FALSE(config5.dtx_enabled);
+
+  const auto config6 = CreateConfigWithParameters({{"usedtx", "1"}});
+  EXPECT_TRUE(config6.dtx_enabled);
+
+  const auto config7 = CreateConfigWithParameters({{"cbr", "0"}});
+  EXPECT_FALSE(config7.cbr_enabled);
+
+  const auto config8 = CreateConfigWithParameters({{"cbr", "1"}});
+  EXPECT_TRUE(config8.cbr_enabled);
+
+  const auto config9 =
+      CreateConfigWithParameters({{"maxplaybackrate", "12345"}});
+  EXPECT_EQ(12345, config9.max_playback_rate_hz);
+
+  const auto config10 =
+      CreateConfigWithParameters({{"maxaveragebitrate", "96000"}});
+  EXPECT_EQ(96000, config10.bitrate_bps);
+
+  const auto config11 = CreateConfigWithParameters({{"maxptime", "40"}});
+  for (int frame_length : config11.supported_frame_lengths_ms) {
+    EXPECT_LE(frame_length, 40);
+  }
+
+  const auto config12 = CreateConfigWithParameters({{"minptime", "40"}});
+  for (int frame_length : config12.supported_frame_lengths_ms) {
+    EXPECT_GE(frame_length, 40);
+  }
+
+  const auto config13 = CreateConfigWithParameters({{"ptime", "40"}});
+  EXPECT_EQ(40, config13.frame_size_ms);
+
+  constexpr int kMinSupportedFrameLength = 10;
+  constexpr int kMaxSupportedFrameLength =
+      WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
+
+  const auto config14 = CreateConfigWithParameters({{"ptime", "1"}});
+  EXPECT_EQ(kMinSupportedFrameLength, config14.frame_size_ms);
+
+  const auto config15 = CreateConfigWithParameters({{"ptime", "2000"}});
+  EXPECT_EQ(kMaxSupportedFrameLength, config15.frame_size_ms);
+}
+
+TEST(AudioEncoderOpusTest, TestConfigFromInvalidParams) {
+  const webrtc::SdpAudioFormat format("opus", 48000, 2);
+  const auto default_config = *AudioEncoderOpus::SdpToConfig(format);
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60, 120});
+#else
+  const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60});
+#endif
+
+  AudioEncoderOpusConfig config;
+  config = CreateConfigWithParameters({{"stereo", "invalid"}});
+  EXPECT_EQ(default_config.num_channels, config.num_channels);
+
+  config = CreateConfigWithParameters({{"useinbandfec", "invalid"}});
+  EXPECT_EQ(default_config.fec_enabled, config.fec_enabled);
+
+  config = CreateConfigWithParameters({{"usedtx", "invalid"}});
+  EXPECT_EQ(default_config.dtx_enabled, config.dtx_enabled);
+
+  config = CreateConfigWithParameters({{"cbr", "invalid"}});
+  EXPECT_EQ(default_config.dtx_enabled, config.dtx_enabled);
+
+  config = CreateConfigWithParameters({{"maxplaybackrate", "0"}});
+  EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
+
+  config = CreateConfigWithParameters({{"maxplaybackrate", "-23"}});
+  EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
+
+  config = CreateConfigWithParameters({{"maxplaybackrate", "not a number!"}});
+  EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "0"}});
+  EXPECT_EQ(6000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "-1000"}});
+  EXPECT_EQ(6000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "1024000"}});
+  EXPECT_EQ(510000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "not a number!"}});
+  EXPECT_EQ(default_config.bitrate_bps, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxptime", "invalid"}});
+  EXPECT_EQ(default_supported_frame_lengths_ms,
+            config.supported_frame_lengths_ms);
+
+  config = CreateConfigWithParameters({{"minptime", "invalid"}});
+  EXPECT_EQ(default_supported_frame_lengths_ms,
+            config.supported_frame_lengths_ms);
+
+  config = CreateConfigWithParameters({{"ptime", "invalid"}});
+  EXPECT_EQ(default_supported_frame_lengths_ms,
+            config.supported_frame_lengths_ms);
+}
+
+TEST(AudioEncoderOpusTest, GetFrameLenghtRange) {
+  AudioEncoderOpusConfig config =
+      CreateConfigWithParameters({{"maxptime", "10"}, {"ptime", "10"}});
+  std::unique_ptr<AudioEncoder> encoder =
+      AudioEncoderOpus::MakeAudioEncoder(config, kDefaultOpusPayloadType);
+  auto ptime = webrtc::TimeDelta::Millis(10);
+  absl::optional<std::pair<webrtc::TimeDelta, webrtc::TimeDelta>> range = {
+      {ptime, ptime}};
+  EXPECT_EQ(encoder->GetFrameLengthRange(), range);
+}
+
+// Test that bitrate will be overridden by the "maxaveragebitrate" parameter.
+// Also test that the "maxaveragebitrate" can't be set to values outside the
+// range of 6000 and 510000
+TEST(AudioEncoderOpusTest, SetSendCodecOpusMaxAverageBitrate) {
+  // Ignore if less than 6000.
+  const auto config1 = AudioEncoderOpus::SdpToConfig(
+      {"opus", 48000, 2, {{"maxaveragebitrate", "5999"}}});
+  EXPECT_EQ(6000, config1->bitrate_bps);
+
+  // Ignore if larger than 510000.
+  const auto config2 = AudioEncoderOpus::SdpToConfig(
+      {"opus", 48000, 2, {{"maxaveragebitrate", "510001"}}});
+  EXPECT_EQ(510000, config2->bitrate_bps);
+
+  const auto config3 = AudioEncoderOpus::SdpToConfig(
+      {"opus", 48000, 2, {{"maxaveragebitrate", "200000"}}});
+  EXPECT_EQ(200000, config3->bitrate_bps);
+}
+
+// Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateNb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "8000"}});
+  EXPECT_EQ(8000, config.max_playback_rate_hz);
+  EXPECT_EQ(12000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "8000"}, {"stereo", "1"}});
+  EXPECT_EQ(8000, config.max_playback_rate_hz);
+  EXPECT_EQ(24000, config.bitrate_bps);
+}
+
+// Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateMb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "8001"}});
+  EXPECT_EQ(8001, config.max_playback_rate_hz);
+  EXPECT_EQ(20000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "8001"}, {"stereo", "1"}});
+  EXPECT_EQ(8001, config.max_playback_rate_hz);
+  EXPECT_EQ(40000, config.bitrate_bps);
+}
+
+// Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateWb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "12001"}});
+  EXPECT_EQ(12001, config.max_playback_rate_hz);
+  EXPECT_EQ(20000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "12001"}, {"stereo", "1"}});
+  EXPECT_EQ(12001, config.max_playback_rate_hz);
+  EXPECT_EQ(40000, config.bitrate_bps);
+}
+
+// Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateSwb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "16001"}});
+  EXPECT_EQ(16001, config.max_playback_rate_hz);
+  EXPECT_EQ(32000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "16001"}, {"stereo", "1"}});
+  EXPECT_EQ(16001, config.max_playback_rate_hz);
+  EXPECT_EQ(64000, config.bitrate_bps);
+}
+
+// Test 24000 < maxplaybackrate triggers Opus full band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateFb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "24001"}});
+  EXPECT_EQ(24001, config.max_playback_rate_hz);
+  EXPECT_EQ(32000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "24001"}, {"stereo", "1"}});
+  EXPECT_EQ(24001, config.max_playback_rate_hz);
+  EXPECT_EQ(64000, config.bitrate_bps);
+}
+
+TEST_P(AudioEncoderOpusTest, OpusFlagDtxAsNonSpeech) {
+  // Create encoder with DTX enabled.
+  AudioEncoderOpusConfig config;
+  config.dtx_enabled = true;
+  config.sample_rate_hz = sample_rate_hz_;
+  constexpr int payload_type = 17;
+  const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+
+  // Open file containing speech and silence.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  test::AudioLoop audio_loop;
+  // Use the file as if it were sampled at our desired input rate.
+  const size_t max_loop_length_samples =
+      sample_rate_hz_ * 10;  // Max 10 second loop.
+  const size_t input_block_size_samples =
+      10 * sample_rate_hz_ / 1000;  // 10 ms.
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, max_loop_length_samples,
+                              input_block_size_samples));
+
+  // Encode.
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  int nonspeech_frames = 0;
+  int max_nonspeech_frames = 0;
+  int dtx_frames = 0;
+  int max_dtx_frames = 0;
+  uint32_t rtp_timestamp = 0u;
+  for (size_t i = 0; i < 500; ++i) {
+    encoded.Clear();
+
+    // Every second call to the encoder will generate an Opus packet.
+    for (int j = 0; j < 2; j++) {
+      info =
+          encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
+      rtp_timestamp += input_block_size_samples;
+    }
+
+    // Bookkeeping of number of DTX frames.
+    if (info.encoded_bytes <= 2) {
+      ++dtx_frames;
+    } else {
+      if (dtx_frames > max_dtx_frames)
+        max_dtx_frames = dtx_frames;
+      dtx_frames = 0;
+    }
+
+    // Bookkeeping of number of non-speech frames.
+    if (info.speech == 0) {
+      ++nonspeech_frames;
+    } else {
+      if (nonspeech_frames > max_nonspeech_frames)
+        max_nonspeech_frames = nonspeech_frames;
+      nonspeech_frames = 0;
+    }
+  }
+
+  // Maximum number of consecutive non-speech packets should exceed 15.
+  EXPECT_GT(max_nonspeech_frames, 15);
+}
+
+TEST(AudioEncoderOpusTest, OpusDtxFilteringHighEnergyRefreshPackets) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-OpusAvoidNoisePumpingDuringDtx/Enabled/");
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile16kHz", "pcm");
+  constexpr int kSampleRateHz = 16000;
+  AudioEncoderOpusConfig config;
+  config.dtx_enabled = true;
+  config.sample_rate_hz = kSampleRateHz;
+  constexpr int payload_type = 17;
+  const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+  test::AudioLoop audio_loop;
+  constexpr size_t kMaxLoopLengthSaples = kSampleRateHz * 11.6f;
+  constexpr size_t kInputBlockSizeSamples = kSampleRateHz / 100;
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSaples,
+                              kInputBlockSizeSamples));
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  // Encode the audio file and store the last part that corresponds to silence.
+  constexpr size_t kSilenceDurationSamples = kSampleRateHz * 0.2f;
+  std::array<int16_t, kSilenceDurationSamples> silence;
+  uint32_t rtp_timestamp = 0;
+  bool last_packet_dtx_frame = false;
+  bool opus_entered_dtx = false;
+  bool silence_filled = false;
+  size_t timestamp_start_silence = 0;
+  while (!silence_filled && rtp_timestamp < kMaxLoopLengthSaples) {
+    encoded.Clear();
+    // Every second call to the encoder will generate an Opus packet.
+    for (int j = 0; j < 2; j++) {
+      auto next_frame = audio_loop.GetNextBlock();
+      info = encoder->Encode(rtp_timestamp, next_frame, &encoded);
+      if (opus_entered_dtx) {
+        size_t silence_frame_start = rtp_timestamp - timestamp_start_silence;
+        silence_filled = silence_frame_start >= kSilenceDurationSamples;
+        if (!silence_filled) {
+          std::copy(next_frame.begin(), next_frame.end(),
+                    silence.begin() + silence_frame_start);
+        }
+      }
+      rtp_timestamp += kInputBlockSizeSamples;
+    }
+    EXPECT_TRUE(info.encoded_bytes > 0 || last_packet_dtx_frame);
+    last_packet_dtx_frame = info.encoded_bytes > 0 ? info.encoded_bytes <= 2
+                                                   : last_packet_dtx_frame;
+    if (info.encoded_bytes <= 2 && !opus_entered_dtx) {
+      timestamp_start_silence = rtp_timestamp;
+    }
+    opus_entered_dtx = info.encoded_bytes <= 2;
+  }
+
+  EXPECT_TRUE(silence_filled);
+  // The copied 200 ms of silence is used for creating 6 bursts that are fed to
+  // the encoder, the first three ones with a larger energy and the last three
+  // with a lower energy. This test verifies that the encoder just sends refresh
+  // DTX packets during the last bursts.
+  int number_non_empty_packets_during_increase = 0;
+  int number_non_empty_packets_during_decrease = 0;
+  for (size_t burst = 0; burst < 6; ++burst) {
+    uint32_t rtp_timestamp_start = rtp_timestamp;
+    const bool increase_noise = burst < 3;
+    const float gain = increase_noise ? 1.4f : 0.0f;
+    while (rtp_timestamp < rtp_timestamp_start + kSilenceDurationSamples) {
+      encoded.Clear();
+      // Every second call to the encoder will generate an Opus packet.
+      for (int j = 0; j < 2; j++) {
+        std::array<int16_t, kInputBlockSizeSamples> silence_frame;
+        size_t silence_frame_start = rtp_timestamp - rtp_timestamp_start;
+        std::transform(
+            silence.begin() + silence_frame_start,
+            silence.begin() + silence_frame_start + kInputBlockSizeSamples,
+            silence_frame.begin(), [gain](float s) { return gain * s; });
+        info = encoder->Encode(rtp_timestamp, silence_frame, &encoded);
+        rtp_timestamp += kInputBlockSizeSamples;
+      }
+      EXPECT_TRUE(info.encoded_bytes > 0 || last_packet_dtx_frame);
+      last_packet_dtx_frame = info.encoded_bytes > 0 ? info.encoded_bytes <= 2
+                                                     : last_packet_dtx_frame;
+      // Tracking the number of non empty packets.
+      if (increase_noise && info.encoded_bytes > 2) {
+        number_non_empty_packets_during_increase++;
+      }
+      if (!increase_noise && info.encoded_bytes > 2) {
+        number_non_empty_packets_during_decrease++;
+      }
+    }
+  }
+  // Check that the refresh DTX packets are just sent during the decrease energy
+  // region.
+  EXPECT_EQ(number_non_empty_packets_during_increase, 0);
+  EXPECT_GT(number_non_empty_packets_during_decrease, 0);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
new file mode 100644
index 0000000000..38b60c6187
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "common_audio/include/audio_util.h"
+#include "common_audio/window_generator.h"
+#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kNumChannels = 1u;
+constexpr int kSampleRateHz = 48000;
+constexpr size_t kMaxLoopLengthSamples = kSampleRateHz * 50;  // 50 seconds.
+constexpr size_t kInputBlockSizeSamples = 10 * kSampleRateHz / 1000;   // 10 ms
+constexpr size_t kOutputBlockSizeSamples = 20 * kSampleRateHz / 1000;  // 20 ms
+constexpr size_t kFftSize = 1024;
+constexpr size_t kNarrowbandSize = 4000 * kFftSize / kSampleRateHz;
+constexpr float kKbdAlpha = 1.5f;
+
+class PowerRatioEstimator : public LappedTransform::Callback {
+ public:
+  PowerRatioEstimator() : low_pow_(0.f), high_pow_(0.f) {
+    WindowGenerator::KaiserBesselDerived(kKbdAlpha, kFftSize, window_);
+    transform_.reset(new LappedTransform(kNumChannels, 0u,
+                                         kInputBlockSizeSamples, window_,
+                                         kFftSize, kFftSize / 2, this));
+  }
+
+  void ProcessBlock(float* data) { transform_->ProcessChunk(&data, nullptr); }
+
+  float PowerRatio() { return high_pow_ / low_pow_; }
+
+ protected:
+  void ProcessAudioBlock(const std::complex<float>* const* input,
+                         size_t num_input_channels,
+                         size_t num_freq_bins,
+                         size_t num_output_channels,
+                         std::complex<float>* const* output) override {
+    float low_pow = 0.f;
+    float high_pow = 0.f;
+    for (size_t i = 0u; i < num_input_channels; ++i) {
+      for (size_t j = 0u; j < kNarrowbandSize; ++j) {
+        float low_mag = std::abs(input[i][j]);
+        low_pow += low_mag * low_mag;
+        float high_mag = std::abs(input[i][j + kNarrowbandSize]);
+        high_pow += high_mag * high_mag;
+      }
+    }
+    low_pow_ += low_pow / (num_input_channels * kFftSize);
+    high_pow_ += high_pow / (num_input_channels * kFftSize);
+  }
+
+ private:
+  std::unique_ptr<LappedTransform> transform_;
+  float window_[kFftSize];
+  float low_pow_;
+  float high_pow_;
+};
+
+float EncodedPowerRatio(AudioEncoder* encoder,
+                        AudioDecoder* decoder,
+                        test::AudioLoop* audio_loop) {
+  // Encode and decode.
+  uint32_t rtp_timestamp = 0u;
+  constexpr size_t kBufferSize = 500;
+  rtc::Buffer encoded(kBufferSize);
+  std::vector<int16_t> decoded(kOutputBlockSizeSamples);
+  std::vector<float> decoded_float(kOutputBlockSizeSamples);
+  AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+  PowerRatioEstimator power_ratio_estimator;
+  for (size_t i = 0; i < 1000; ++i) {
+    encoded.Clear();
+    AudioEncoder::EncodedInfo encoder_info =
+        encoder->Encode(rtp_timestamp, audio_loop->GetNextBlock(), &encoded);
+    rtp_timestamp += kInputBlockSizeSamples;
+    if (encoded.size() > 0) {
+      int decoder_info = decoder->Decode(
+          encoded.data(), encoded.size(), kSampleRateHz,
+          decoded.size() * sizeof(decoded[0]), decoded.data(), &speech_type);
+      if (decoder_info > 0) {
+        S16ToFloat(decoded.data(), decoded.size(), decoded_float.data());
+        power_ratio_estimator.ProcessBlock(decoded_float.data());
+      }
+    }
+  }
+  return power_ratio_estimator.PowerRatio();
+}
+
+}  // namespace
+
+// TODO(ivoc): Remove this test, WebRTC-AdjustOpusBandwidth is obsolete.
+TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-AdjustOpusBandwidth/Enabled/");
+
+  constexpr float kMaxNarrowbandRatio = 0.0035f;
+  constexpr float kMinWidebandRatio = 0.01f;
+
+  // Create encoder.
+  AudioEncoderOpusConfig enc_config;
+  enc_config.bitrate_bps = absl::optional<int>(7999);
+  enc_config.num_channels = kNumChannels;
+  constexpr int payload_type = 17;
+  auto encoder = AudioEncoderOpus::MakeAudioEncoder(enc_config, payload_type);
+
+  // Create decoder.
+  AudioDecoderOpus::Config dec_config;
+  dec_config.num_channels = kNumChannels;
+  auto decoder = AudioDecoderOpus::MakeAudioDecoder(dec_config);
+
+  // Open speech file.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
+  test::AudioLoop audio_loop;
+  EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
+  ASSERT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                              kInputBlockSizeSamples));
+
+  EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMaxNarrowbandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(9000);
+  EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMaxNarrowbandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(9001);
+  EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMinWidebandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(8000);
+  EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMinWidebandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(12001);
+  EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMinWidebandRatio);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
new file mode 100644
index 0000000000..e8c131092c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/time_utils.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::GetGlobalMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+int64_t RunComplexityTest(const AudioEncoderOpusConfig& config) {
+  // Create encoder.
+  constexpr int payload_type = 17;
+  const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+  // Open speech file.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
+  test::AudioLoop audio_loop;
+  constexpr int kSampleRateHz = 48000;
+  EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
+  constexpr size_t kMaxLoopLengthSamples =
+      kSampleRateHz * 10;  // 10 second loop.
+  constexpr size_t kInputBlockSizeSamples =
+      10 * kSampleRateHz / 1000;  // 60 ms.
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                              kInputBlockSizeSamples));
+  // Encode.
+  const int64_t start_time_ms = rtc::TimeMillis();
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  uint32_t rtp_timestamp = 0u;
+  for (size_t i = 0; i < 10000; ++i) {
+    encoded.Clear();
+    info = encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
+    rtp_timestamp += kInputBlockSizeSamples;
+  }
+  return rtc::TimeMillis() - start_time_ms;
+}
+
+// This test encodes an audio file using Opus twice with different bitrates
+// (~11 kbps and 15.5 kbps). The runtime for each is measured, and the ratio
+// between the two is calculated and tracked. This test explicitly sets the
+// low_rate_complexity to 9. When running on desktop platforms, this is the same
+// as the regular complexity, and the expectation is that the resulting ratio
+// should be less than 100% (since the encoder runs faster at lower bitrates,
+// given a fixed complexity setting). On the other hand, when running on
+// mobiles, the regular complexity is 5, and we expect the resulting ratio to
+// be higher, since we have explicitly asked for a higher complexity setting at
+// the lower rate.
+TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_On) {
+  // Create config.
+  AudioEncoderOpusConfig config;
+  // The limit -- including the hysteresis window -- at which the complexity
+  // shuold be increased.
+  config.bitrate_bps = 11000 - 1;
+  config.low_rate_complexity = 9;
+  int64_t runtime_10999bps = RunComplexityTest(config);
+
+  config.bitrate_bps = 15500;
+  int64_t runtime_15500bps = RunComplexityTest(config);
+
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "opus_encoding_complexity_ratio", "adaptation_on",
+      100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+// This test is identical to the one above, but without the complexity
+// adaptation enabled (neither on desktop, nor on mobile). The expectation is
+// that the resulting ratio is less than 100% at all times.
+TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_Off) {
+  // Create config.
+  AudioEncoderOpusConfig config;
+  // The limit -- including the hysteresis window -- at which the complexity
+  // shuold be increased (but not in this test since complexity adaptation is
+  // disabled).
+  config.bitrate_bps = 11000 - 1;
+  int64_t runtime_10999bps = RunComplexityTest(config);
+
+  config.bitrate_bps = 15500;
+  int64_t runtime_15500bps = RunComplexityTest(config);
+
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "opus_encoding_complexity_ratio", "adaptation_off",
+      100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
new file mode 100644
index 0000000000..815f26e31c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -0,0 +1,248 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using std::get;
+using std::string;
+using std::tuple;
+using ::testing::TestWithParam;
+
+namespace webrtc {
+
+// Define coding parameter as <channels, bit_rate, filename, extension>.
+typedef tuple<size_t, int, string, string> coding_param;
+typedef struct mode mode;
+
+struct mode {
+  bool fec;
+  uint8_t target_packet_loss_rate;
+};
+
+const int kOpusBlockDurationMs = 20;
+const int kOpusSamplingKhz = 48;
+
+class OpusFecTest : public TestWithParam<coding_param> {
+ protected:
+  OpusFecTest();
+
+  void SetUp() override;
+  void TearDown() override;
+
+  virtual void EncodeABlock();
+
+  virtual void DecodeABlock(bool lost_previous, bool lost_current);
+
+  int block_duration_ms_;
+  int sampling_khz_;
+  size_t block_length_sample_;
+
+  size_t channels_;
+  int bit_rate_;
+
+  size_t data_pointer_;
+  size_t loop_length_samples_;
+  size_t max_bytes_;
+  size_t encoded_bytes_;
+
+  WebRtcOpusEncInst* opus_encoder_;
+  WebRtcOpusDecInst* opus_decoder_;
+
+  string in_filename_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<int16_t[]> out_data_;
+  std::unique_ptr<uint8_t[]> bit_stream_;
+};
+
+void OpusFecTest::SetUp() {
+  channels_ = get<0>(GetParam());
+  bit_rate_ = get<1>(GetParam());
+  printf("Coding %zu channel signal at %d bps.\n", channels_, bit_rate_);
+
+  in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
+
+  FILE* fp = fopen(in_filename_.c_str(), "rb");
+  ASSERT_FALSE(fp == NULL);
+
+  // Obtain file size.
+  fseek(fp, 0, SEEK_END);
+  loop_length_samples_ = ftell(fp) / sizeof(int16_t);
+  rewind(fp);
+
+  // Allocate memory to contain the whole file.
+  in_data_.reset(
+      new int16_t[loop_length_samples_ + block_length_sample_ * channels_]);
+
+  // Copy the file into the buffer.
+  ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
+            loop_length_samples_);
+  fclose(fp);
+
+  // The audio will be used in a looped manner. To ease the acquisition of an
+  // audio frame that crosses the end of the excerpt, we add an extra block
+  // length of samples to the end of the array, starting over again from the
+  // beginning of the array. Audio frames cross the end of the excerpt always
+  // appear as a continuum of memory.
+  memcpy(&in_data_[loop_length_samples_], &in_data_[0],
+         block_length_sample_ * channels_ * sizeof(int16_t));
+
+  // Maximum number of bytes in output bitstream.
+  max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t);
+
+  out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]);
+  bit_stream_.reset(new uint8_t[max_bytes_]);
+
+  // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
+  int app = channels_ == 1 ? 0 : 1;
+
+  // Create encoder memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
+  EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
+  // Set bitrate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
+}
+
+void OpusFecTest::TearDown() {
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+OpusFecTest::OpusFecTest()
+    : block_duration_ms_(kOpusBlockDurationMs),
+      sampling_khz_(kOpusSamplingKhz),
+      block_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * sampling_khz_)),
+      data_pointer_(0),
+      max_bytes_(0),
+      encoded_bytes_(0),
+      opus_encoder_(NULL),
+      opus_decoder_(NULL) {}
+
+void OpusFecTest::EncodeABlock() {
+  int value =
+      WebRtcOpus_Encode(opus_encoder_, &in_data_[data_pointer_],
+                        block_length_sample_, max_bytes_, &bit_stream_[0]);
+  EXPECT_GT(value, 0);
+
+  encoded_bytes_ = static_cast<size_t>(value);
+}
+
+void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
+  int16_t audio_type;
+  int value_1 = 0, value_2 = 0;
+
+  if (lost_previous) {
+    // Decode previous frame.
+    if (!lost_current &&
+        WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_) == 1) {
+      value_1 =
+          WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], encoded_bytes_,
+                               &out_data_[0], &audio_type);
+    } else {
+      // Call decoder PLC.
+      while (value_1 < static_cast<int>(block_length_sample_)) {
+        int ret = WebRtcOpus_Decode(opus_decoder_, NULL, 0, &out_data_[value_1],
+                                    &audio_type);
+        EXPECT_EQ(ret, sampling_khz_ * 10);  // Should return 10 ms of samples.
+        value_1 += ret;
+      }
+    }
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_1);
+  }
+
+  if (!lost_current) {
+    // Decode current frame.
+    value_2 = WebRtcOpus_Decode(opus_decoder_, &bit_stream_[0], encoded_bytes_,
+                                &out_data_[value_1 * channels_], &audio_type);
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_2);
+  }
+}
+
+TEST_P(OpusFecTest, RandomPacketLossTest) {
+  const int kDurationMs = 200000;
+  int time_now_ms, fec_frames;
+  int actual_packet_loss_rate;
+  bool lost_current, lost_previous;
+  mode mode_set[3] = {{true, 0}, {false, 0}, {true, 50}};
+
+  lost_current = false;
+  for (int i = 0; i < 3; i++) {
+    if (mode_set[i].fec) {
+      EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+      EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(
+                       opus_encoder_, mode_set[i].target_packet_loss_rate));
+      printf("FEC is ON, target at packet loss rate %d percent.\n",
+             mode_set[i].target_packet_loss_rate);
+    } else {
+      EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
+      printf("FEC is OFF.\n");
+    }
+    // In this test, we let the target packet loss rate match the actual rate.
+    actual_packet_loss_rate = mode_set[i].target_packet_loss_rate;
+    // Run every mode a certain time.
+    time_now_ms = 0;
+    fec_frames = 0;
+    while (time_now_ms < kDurationMs) {
+      // Encode & decode.
+      EncodeABlock();
+
+      // Check if payload has FEC.
+      int fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
+
+      // If FEC is disabled or the target packet loss rate is set to 0, there
+      // should be no FEC in the bit stream.
+      if (!mode_set[i].fec || mode_set[i].target_packet_loss_rate == 0) {
+        EXPECT_EQ(fec, 0);
+      } else if (fec == 1) {
+        fec_frames++;
+      }
+
+      lost_previous = lost_current;
+      lost_current = rand() < actual_packet_loss_rate * (RAND_MAX / 100);
+      DecodeABlock(lost_previous, lost_current);
+
+      time_now_ms += block_duration_ms_;
+
+      // `data_pointer_` is incremented and wrapped across
+      // `loop_length_samples_`.
+      data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
+                      loop_length_samples_;
+    }
+    if (mode_set[i].fec) {
+      printf("%.2f percent frames has FEC.\n",
+             static_cast<float>(fec_frames) * block_duration_ms_ / 2000);
+    }
+  }
+}
+
+const coding_param param_set[] = {
+    std::make_tuple(1,
+                    64000,
+                    string("audio_coding/testfile32kHz"),
+                    string("pcm")),
+    std::make_tuple(1,
+                    32000,
+                    string("audio_coding/testfile32kHz"),
+                    string("pcm")),
+    std::make_tuple(2,
+                    64000,
+                    string("audio_coding/teststereo32kHz"),
+                    string("pcm"))};
+
+// 64 kbps, stereo
+INSTANTIATE_TEST_SUITE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h
new file mode 100644
index 0000000000..92c5c354a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+
+#include <stddef.h>
+
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "third_party/opus/src/include/opus.h"
+#include "third_party/opus/src/include/opus_multistream.h"
+RTC_POP_IGNORING_WUNDEF()
+
+struct WebRtcOpusEncInst {
+  OpusEncoder* encoder;
+  OpusMSEncoder* multistream_encoder;
+  size_t channels;
+  int in_dtx_mode;
+  bool avoid_noise_pumping_during_dtx;
+  int sample_rate_hz;
+  float smooth_energy_non_active_frames;
+};
+
+struct WebRtcOpusDecInst {
+  OpusDecoder* decoder;
+  OpusMSDecoder* multistream_decoder;
+  int prev_decoded_samples;
+  bool plc_use_prev_decoded_samples;
+  size_t channels;
+  int in_dtx_mode;
+  int sample_rate_hz;
+};
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc
new file mode 100644
index 0000000000..67d8619b34
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc
@@ -0,0 +1,881 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+#include <cstdlib>
+
+#include <numeric>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+enum {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  /* Maximum supported frame size in WebRTC is 120 ms. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 120,
+#else
+  /* Maximum supported frame size in WebRTC is 60 ms. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 60,
+#endif
+
+  /* The format allows up to 120 ms frames. Since we don't control the other
+   * side, we must allow for packets of that size. NetEq is currently limited
+   * to 60 ms on the receive side. */
+  kWebRtcOpusMaxDecodeFrameSizeMs = 120,
+
+  // Duration of audio that each call to packet loss concealment covers.
+  kWebRtcOpusPlcFrameSizeMs = 10,
+};
+
+constexpr char kPlcUsePrevDecodedSamplesFieldTrial[] =
+    "WebRTC-Audio-OpusPlcUsePrevDecodedSamples";
+
+constexpr char kAvoidNoisePumpingDuringDtxFieldTrial[] =
+    "WebRTC-Audio-OpusAvoidNoisePumpingDuringDtx";
+
+constexpr char kSetSignalVoiceWithDtxFieldTrial[] =
+    "WebRTC-Audio-OpusSetSignalVoiceWithDtx";
+
+static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) {
+  RTC_DCHECK_GT(frame_size_ms, 0);
+  RTC_DCHECK_EQ(frame_size_ms % 10, 0);
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  RTC_DCHECK_EQ(sample_rate_hz % 1000, 0);
+  return frame_size_ms * (sample_rate_hz / 1000);
+}
+
+// Maximum sample count per channel.
+static int MaxFrameSizePerChannel(int sample_rate_hz) {
+  return FrameSizePerChannel(kWebRtcOpusMaxDecodeFrameSizeMs, sample_rate_hz);
+}
+
+// Default sample count per channel.
+static int DefaultFrameSizePerChannel(int sample_rate_hz) {
+  return FrameSizePerChannel(20, sample_rate_hz);
+}
+
+// Returns true if the `encoded` payload corresponds to a refresh DTX packet
+// whose energy is larger than the expected for non activity packets.
+static bool WebRtcOpus_IsHighEnergyRefreshDtxPacket(
+    OpusEncInst* inst,
+    rtc::ArrayView<const int16_t> frame,
+    rtc::ArrayView<const uint8_t> encoded) {
+  if (encoded.size() <= 2) {
+    return false;
+  }
+  int number_frames =
+      frame.size() / DefaultFrameSizePerChannel(inst->sample_rate_hz);
+  if (number_frames > 0 &&
+      WebRtcOpus_PacketHasVoiceActivity(encoded.data(), encoded.size()) == 0) {
+    const float average_frame_energy =
+        std::accumulate(frame.begin(), frame.end(), 0.0f,
+                        [](float a, int32_t b) { return a + b * b; }) /
+        number_frames;
+    if (WebRtcOpus_GetInDtx(inst) == 1 &&
+        average_frame_energy >= inst->smooth_energy_non_active_frames * 0.5f) {
+      // This is a refresh DTX packet as the encoder is in DTX and has
+      // produced a payload > 2 bytes. This refresh packet has a higher energy
+      // than the smooth energy of non activity frames (with a 3 dB negative
+      // margin) and, therefore, it is flagged as a high energy refresh DTX
+      // packet.
+      return true;
+    }
+    // The average energy is tracked in a similar way as the modeling of the
+    // comfort noise in the Silk decoder in Opus
+    // (third_party/opus/src/silk/CNG.c).
+    if (average_frame_energy < inst->smooth_energy_non_active_frames * 0.5f) {
+      inst->smooth_energy_non_active_frames = average_frame_energy;
+    } else {
+      inst->smooth_energy_non_active_frames +=
+          (average_frame_energy - inst->smooth_energy_non_active_frames) *
+          0.25f;
+    }
+  }
+  return false;
+}
+
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
+                                 size_t channels,
+                                 int32_t application,
+                                 int sample_rate_hz) {
+  int opus_app;
+  if (!inst)
+    return -1;
+
+  switch (application) {
+    case 0:
+      opus_app = OPUS_APPLICATION_VOIP;
+      break;
+    case 1:
+      opus_app = OPUS_APPLICATION_AUDIO;
+      break;
+    default:
+      return -1;
+  }
+
+  OpusEncInst* state =
+      reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst)));
+  RTC_DCHECK(state);
+
+  int error;
+  state->encoder = opus_encoder_create(
+      sample_rate_hz, static_cast<int>(channels), opus_app, &error);
+
+  if (error != OPUS_OK || (!state->encoder && !state->multistream_encoder)) {
+    WebRtcOpus_EncoderFree(state);
+    return -1;
+  }
+
+  state->in_dtx_mode = 0;
+  state->channels = channels;
+  state->sample_rate_hz = sample_rate_hz;
+  state->smooth_energy_non_active_frames = 0.0f;
+  state->avoid_noise_pumping_during_dtx =
+      webrtc::field_trial::IsEnabled(kAvoidNoisePumpingDuringDtxFieldTrial);
+
+  *inst = state;
+  return 0;
+}
+
+int16_t WebRtcOpus_MultistreamEncoderCreate(
+    OpusEncInst** inst,
+    size_t channels,
+    int32_t application,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping) {
+  int opus_app;
+  if (!inst)
+    return -1;
+
+  switch (application) {
+    case 0:
+      opus_app = OPUS_APPLICATION_VOIP;
+      break;
+    case 1:
+      opus_app = OPUS_APPLICATION_AUDIO;
+      break;
+    default:
+      return -1;
+  }
+
+  OpusEncInst* state =
+      reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst)));
+  RTC_DCHECK(state);
+
+  int error;
+  const int sample_rate_hz = 48000;
+  state->multistream_encoder = opus_multistream_encoder_create(
+      sample_rate_hz, channels, streams, coupled_streams, channel_mapping,
+      opus_app, &error);
+
+  if (error != OPUS_OK || (!state->encoder && !state->multistream_encoder)) {
+    WebRtcOpus_EncoderFree(state);
+    return -1;
+  }
+
+  state->in_dtx_mode = 0;
+  state->channels = channels;
+  state->sample_rate_hz = sample_rate_hz;
+  state->smooth_energy_non_active_frames = 0.0f;
+  state->avoid_noise_pumping_during_dtx = false;
+
+  *inst = state;
+  return 0;
+}
+
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
+  if (inst) {
+    if (inst->encoder) {
+      opus_encoder_destroy(inst->encoder);
+    } else {
+      opus_multistream_encoder_destroy(inst->multistream_encoder);
+    }
+    free(inst);
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      size_t samples,
+                      size_t length_encoded_buffer,
+                      uint8_t* encoded) {
+  int res;
+
+  if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
+    return -1;
+  }
+
+  if (inst->encoder) {
+    res = opus_encode(inst->encoder, (const opus_int16*)audio_in,
+                      static_cast<int>(samples), encoded,
+                      static_cast<opus_int32>(length_encoded_buffer));
+  } else {
+    res = opus_multistream_encode(
+        inst->multistream_encoder, (const opus_int16*)audio_in,
+        static_cast<int>(samples), encoded,
+        static_cast<opus_int32>(length_encoded_buffer));
+  }
+
+  if (res <= 0) {
+    return -1;
+  }
+
+  if (res <= 2) {
+    // Indicates DTX since the packet has nothing but a header. In principle,
+    // there is no need to send this packet. However, we do transmit the first
+    // occurrence to let the decoder know that the encoder enters DTX mode.
+    if (inst->in_dtx_mode) {
+      return 0;
+    } else {
+      inst->in_dtx_mode = 1;
+      return res;
+    }
+  }
+
+  if (inst->avoid_noise_pumping_during_dtx && WebRtcOpus_GetUseDtx(inst) == 1 &&
+      WebRtcOpus_IsHighEnergyRefreshDtxPacket(
+          inst, rtc::MakeArrayView(audio_in, samples),
+          rtc::MakeArrayView(encoded, res))) {
+    // This packet is a high energy refresh DTX packet. For avoiding an increase
+    // of the energy in the DTX region at the decoder, this packet is
+    // substituted by a TOC byte with one empty frame.
+    // The number of frames described in the TOC byte
+    // (https://tools.ietf.org/html/rfc6716#section-3.1) are overwritten to
+    // always indicate one frame (last two bits equal to 0).
+    encoded[0] = encoded[0] & 0b11111100;
+    inst->in_dtx_mode = 1;
+    // The payload is just the TOC byte and has 1 byte as length.
+    return 1;
+  }
+  inst->in_dtx_mode = 0;
+  return res;
+}
+
+#define ENCODER_CTL(inst, vargs)                \
+  (inst->encoder                                \
+       ? opus_encoder_ctl(inst->encoder, vargs) \
+       : opus_multistream_encoder_ctl(inst->multistream_encoder, vargs))
+
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_BITRATE(rate));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_PACKET_LOSS_PERC(loss_rate));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz) {
+  opus_int32 set_bandwidth;
+
+  if (!inst)
+    return -1;
+
+  if (frequency_hz <= 8000) {
+    set_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+  } else if (frequency_hz <= 12000) {
+    set_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+  } else if (frequency_hz <= 16000) {
+    set_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+  } else if (frequency_hz <= 24000) {
+    set_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
+  } else {
+    set_bandwidth = OPUS_BANDWIDTH_FULLBAND;
+  }
+  return ENCODER_CTL(inst, OPUS_SET_MAX_BANDWIDTH(set_bandwidth));
+}
+
+int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
+                                      int32_t* result_hz) {
+  if (inst->encoder) {
+    if (opus_encoder_ctl(inst->encoder, OPUS_GET_MAX_BANDWIDTH(result_hz)) ==
+        OPUS_OK) {
+      return 0;
+    }
+    return -1;
+  }
+
+  opus_int32 max_bandwidth;
+  int s;
+  int ret;
+
+  max_bandwidth = 0;
+  ret = OPUS_OK;
+  s = 0;
+  while (ret == OPUS_OK) {
+    OpusEncoder* enc;
+    opus_int32 bandwidth;
+
+    ret = ENCODER_CTL(inst, OPUS_MULTISTREAM_GET_ENCODER_STATE(s, &enc));
+    if (ret == OPUS_BAD_ARG)
+      break;
+    if (ret != OPUS_OK)
+      return -1;
+    if (opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&bandwidth)) != OPUS_OK)
+      return -1;
+
+    if (max_bandwidth != 0 && max_bandwidth != bandwidth)
+      return -1;
+
+    max_bandwidth = bandwidth;
+    s++;
+  }
+  *result_hz = max_bandwidth;
+  return 0;
+}
+
+int16_t WebRtcOpus_EnableFec(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_INBAND_FEC(1));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_INBAND_FEC(0));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst) {
+  if (inst) {
+    if (webrtc::field_trial::IsEnabled(kSetSignalVoiceWithDtxFieldTrial)) {
+      int ret = ENCODER_CTL(inst, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
+      if (ret != OPUS_OK) {
+        return ret;
+      }
+    }
+    return ENCODER_CTL(inst, OPUS_SET_DTX(1));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) {
+  if (inst) {
+    if (webrtc::field_trial::IsEnabled(kSetSignalVoiceWithDtxFieldTrial)) {
+      int ret = ENCODER_CTL(inst, OPUS_SET_SIGNAL(OPUS_AUTO));
+      if (ret != OPUS_OK) {
+        return ret;
+      }
+    }
+    return ENCODER_CTL(inst, OPUS_SET_DTX(0));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_GetUseDtx(OpusEncInst* inst) {
+  if (inst) {
+    opus_int32 use_dtx;
+    if (ENCODER_CTL(inst, OPUS_GET_DTX(&use_dtx)) == 0) {
+      return use_dtx;
+    }
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_VBR(0));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_DisableCbr(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_VBR(1));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_COMPLEXITY(complexity));
+  } else {
+    return -1;
+  }
+}
+
+int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst) {
+  if (!inst) {
+    return -1;
+  }
+  int32_t bandwidth;
+  if (ENCODER_CTL(inst, OPUS_GET_BANDWIDTH(&bandwidth)) == 0) {
+    return bandwidth;
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_BANDWIDTH(bandwidth));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels) {
+  if (!inst)
+    return -1;
+  if (num_channels == 0) {
+    return ENCODER_CTL(inst, OPUS_SET_FORCE_CHANNELS(OPUS_AUTO));
+  } else if (num_channels == 1 || num_channels == 2) {
+    return ENCODER_CTL(inst, OPUS_SET_FORCE_CHANNELS(num_channels));
+  } else {
+    return -1;
+  }
+}
+
+int32_t WebRtcOpus_GetInDtx(OpusEncInst* inst) {
+  if (!inst) {
+    return -1;
+  }
+#ifdef OPUS_GET_IN_DTX
+  int32_t in_dtx;
+  if (ENCODER_CTL(inst, OPUS_GET_IN_DTX(&in_dtx)) == 0) {
+    return in_dtx;
+  }
+#endif
+  return -1;
+}
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
+                                 size_t channels,
+                                 int sample_rate_hz) {
+  int error;
+  OpusDecInst* state;
+
+  if (inst != NULL) {
+    // Create Opus decoder state.
+    state = reinterpret_cast<OpusDecInst*>(calloc(1, sizeof(OpusDecInst)));
+    if (state == NULL) {
+      return -1;
+    }
+
+    state->decoder =
+        opus_decoder_create(sample_rate_hz, static_cast<int>(channels), &error);
+    if (error == OPUS_OK && state->decoder) {
+      // Creation of memory all ok.
+      state->channels = channels;
+      state->sample_rate_hz = sample_rate_hz;
+      state->plc_use_prev_decoded_samples =
+          webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
+      if (state->plc_use_prev_decoded_samples) {
+        state->prev_decoded_samples =
+            DefaultFrameSizePerChannel(state->sample_rate_hz);
+      }
+      state->in_dtx_mode = 0;
+      *inst = state;
+      return 0;
+    }
+
+    // If memory allocation was unsuccessful, free the entire state.
+    if (state->decoder) {
+      opus_decoder_destroy(state->decoder);
+    }
+    free(state);
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_MultistreamDecoderCreate(
+    OpusDecInst** inst,
+    size_t channels,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping) {
+  int error;
+  OpusDecInst* state;
+
+  if (inst != NULL) {
+    // Create Opus decoder state.
+    state = reinterpret_cast<OpusDecInst*>(calloc(1, sizeof(OpusDecInst)));
+    if (state == NULL) {
+      return -1;
+    }
+
+    // Create new memory, always at 48000 Hz.
+    state->multistream_decoder = opus_multistream_decoder_create(
+        48000, channels, streams, coupled_streams, channel_mapping, &error);
+
+    if (error == OPUS_OK && state->multistream_decoder) {
+      // Creation of memory all ok.
+      state->channels = channels;
+      state->sample_rate_hz = 48000;
+      state->plc_use_prev_decoded_samples =
+          webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
+      if (state->plc_use_prev_decoded_samples) {
+        state->prev_decoded_samples =
+            DefaultFrameSizePerChannel(state->sample_rate_hz);
+      }
+      state->in_dtx_mode = 0;
+      *inst = state;
+      return 0;
+    }
+
+    // If memory allocation was unsuccessful, free the entire state.
+    opus_multistream_decoder_destroy(state->multistream_decoder);
+    free(state);
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
+  if (inst) {
+    if (inst->decoder) {
+      opus_decoder_destroy(inst->decoder);
+    } else if (inst->multistream_decoder) {
+      opus_multistream_decoder_destroy(inst->multistream_decoder);
+    }
+    free(inst);
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
+  return inst->channels;
+}
+
+void WebRtcOpus_DecoderInit(OpusDecInst* inst) {
+  if (inst->decoder) {
+    opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
+  } else {
+    opus_multistream_decoder_ctl(inst->multistream_decoder, OPUS_RESET_STATE);
+  }
+  inst->in_dtx_mode = 0;
+}
+
+/* For decoder to determine if it is to output speech or comfort noise. */
+static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
+  // Audio type becomes comfort noise if `encoded_byte` is 1 and keeps
+  // to be so if the following `encoded_byte` are 0 or 1.
+  if (encoded_bytes == 0 && inst->in_dtx_mode) {
+    return 2;  // Comfort noise.
+  } else if (encoded_bytes == 1 || encoded_bytes == 2) {
+    // TODO(henrik.lundin): There is a slight risk that a 2-byte payload is in
+    // fact a 1-byte TOC with a 1-byte payload. That will be erroneously
+    // interpreted as comfort noise output, but such a payload is probably
+    // faulty anyway.
+
+    // TODO(webrtc:10218): This is wrong for multistream opus. Then are several
+    // single-stream packets glued together with some packet size bytes in
+    // between. See https://tools.ietf.org/html/rfc6716#appendix-B
+    inst->in_dtx_mode = 1;
+    return 2;  // Comfort noise.
+  } else {
+    inst->in_dtx_mode = 0;
+    return 0;  // Speech.
+  }
+}
+
+/* `frame_size` is set to maximum Opus frame size in the normal case, and
+ * is set to the number of samples needed for PLC in case of losses.
+ * It is up to the caller to make sure the value is correct. */
+static int DecodeNative(OpusDecInst* inst,
+                        const uint8_t* encoded,
+                        size_t encoded_bytes,
+                        int frame_size,
+                        int16_t* decoded,
+                        int16_t* audio_type,
+                        int decode_fec) {
+  int res = -1;
+  if (inst->decoder) {
+    res = opus_decode(
+        inst->decoder, encoded, static_cast<opus_int32>(encoded_bytes),
+        reinterpret_cast<opus_int16*>(decoded), frame_size, decode_fec);
+  } else {
+    res = opus_multistream_decode(inst->multistream_decoder, encoded,
+                                  static_cast<opus_int32>(encoded_bytes),
+                                  reinterpret_cast<opus_int16*>(decoded),
+                                  frame_size, decode_fec);
+  }
+
+  if (res <= 0)
+    return -1;
+
+  *audio_type = DetermineAudioType(inst, encoded_bytes);
+
+  return res;
+}
+
+static int DecodePlc(OpusDecInst* inst, int16_t* decoded) {
+  int16_t audio_type = 0;
+  int decoded_samples;
+  int plc_samples =
+      FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
+
+  if (inst->plc_use_prev_decoded_samples) {
+    /* The number of samples we ask for is `number_of_lost_frames` times
+     * `prev_decoded_samples_`. Limit the number of samples to maximum
+     * `MaxFrameSizePerChannel()`. */
+    plc_samples = inst->prev_decoded_samples;
+    const int max_samples_per_channel =
+        MaxFrameSizePerChannel(inst->sample_rate_hz);
+    plc_samples = plc_samples <= max_samples_per_channel
+                      ? plc_samples
+                      : max_samples_per_channel;
+  }
+  decoded_samples =
+      DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0);
+  if (decoded_samples < 0) {
+    return -1;
+  }
+
+  return decoded_samples;
+}
+
+int WebRtcOpus_Decode(OpusDecInst* inst,
+                      const uint8_t* encoded,
+                      size_t encoded_bytes,
+                      int16_t* decoded,
+                      int16_t* audio_type) {
+  int decoded_samples;
+
+  if (encoded_bytes == 0) {
+    *audio_type = DetermineAudioType(inst, encoded_bytes);
+    decoded_samples = DecodePlc(inst, decoded);
+  } else {
+    decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
+                                   MaxFrameSizePerChannel(inst->sample_rate_hz),
+                                   decoded, audio_type, 0);
+  }
+  if (decoded_samples < 0) {
+    return -1;
+  }
+
+  if (inst->plc_use_prev_decoded_samples) {
+    /* Update decoded sample memory, to be used by the PLC in case of losses. */
+    inst->prev_decoded_samples = decoded_samples;
+  }
+
+  return decoded_samples;
+}
+
+int WebRtcOpus_DecodeFec(OpusDecInst* inst,
+                         const uint8_t* encoded,
+                         size_t encoded_bytes,
+                         int16_t* decoded,
+                         int16_t* audio_type) {
+  int decoded_samples;
+  int fec_samples;
+
+  if (WebRtcOpus_PacketHasFec(encoded, encoded_bytes) != 1) {
+    return 0;
+  }
+
+  fec_samples =
+      opus_packet_get_samples_per_frame(encoded, inst->sample_rate_hz);
+
+  decoded_samples = DecodeNative(inst, encoded, encoded_bytes, fec_samples,
+                                 decoded, audio_type, 1);
+  if (decoded_samples < 0) {
+    return -1;
+  }
+
+  return decoded_samples;
+}
+
+int WebRtcOpus_DurationEst(OpusDecInst* inst,
+                           const uint8_t* payload,
+                           size_t payload_length_bytes) {
+  if (payload_length_bytes == 0) {
+    // WebRtcOpus_Decode calls PLC when payload length is zero. So we return
+    // PLC duration correspondingly.
+    return WebRtcOpus_PlcDuration(inst);
+  }
+
+  int frames, samples;
+  frames = opus_packet_get_nb_frames(
+      payload, static_cast<opus_int32>(payload_length_bytes));
+  if (frames < 0) {
+    /* Invalid payload data. */
+    return 0;
+  }
+  samples =
+      frames * opus_packet_get_samples_per_frame(payload, inst->sample_rate_hz);
+  if (samples > 120 * inst->sample_rate_hz / 1000) {
+    // More than 120 ms' worth of samples.
+    return 0;
+  }
+  return samples;
+}
+
+int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
+  if (inst->plc_use_prev_decoded_samples) {
+    /* The number of samples we ask for is `number_of_lost_frames` times
+     * `prev_decoded_samples_`. Limit the number of samples to maximum
+     * `MaxFrameSizePerChannel()`. */
+    const int plc_samples = inst->prev_decoded_samples;
+    const int max_samples_per_channel =
+        MaxFrameSizePerChannel(inst->sample_rate_hz);
+    return plc_samples <= max_samples_per_channel ? plc_samples
+                                                  : max_samples_per_channel;
+  }
+  return FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
+}
+
+int WebRtcOpus_FecDurationEst(const uint8_t* payload,
+                              size_t payload_length_bytes,
+                              int sample_rate_hz) {
+  if (WebRtcOpus_PacketHasFec(payload, payload_length_bytes) != 1) {
+    return 0;
+  }
+  const int samples =
+      opus_packet_get_samples_per_frame(payload, sample_rate_hz);
+  const int samples_per_ms = sample_rate_hz / 1000;
+  if (samples < 10 * samples_per_ms || samples > 120 * samples_per_ms) {
+    /* Invalid payload duration. */
+    return 0;
+  }
+  return samples;
+}
+
+int WebRtcOpus_NumSilkFrames(const uint8_t* payload) {
+  // For computing the payload length in ms, the sample rate is not important
+  // since it cancels out. We use 48 kHz, but any valid sample rate would work.
+  int payload_length_ms =
+      opus_packet_get_samples_per_frame(payload, 48000) / 48;
+  if (payload_length_ms < 10)
+    payload_length_ms = 10;
+
+  int silk_frames;
+  switch (payload_length_ms) {
+    case 10:
+    case 20:
+      silk_frames = 1;
+      break;
+    case 40:
+      silk_frames = 2;
+      break;
+    case 60:
+      silk_frames = 3;
+      break;
+    default:
+      return 0;  // It is actually even an invalid packet.
+  }
+  return silk_frames;
+}
+
+// This method is based on Definition of the Opus Audio Codec
+// (https://tools.ietf.org/html/rfc6716). Basically, this method is based on
+// parsing the LP layer of an Opus packet, particularly the LBRR flag.
+int WebRtcOpus_PacketHasFec(const uint8_t* payload,
+                            size_t payload_length_bytes) {
+  if (payload == NULL || payload_length_bytes == 0)
+    return 0;
+
+  // In CELT_ONLY mode, packets should not have FEC.
+  if (payload[0] & 0x80)
+    return 0;
+
+  int silk_frames = WebRtcOpus_NumSilkFrames(payload);
+  if (silk_frames == 0)
+    return 0;  // Not valid.
+
+  const int channels = opus_packet_get_nb_channels(payload);
+  RTC_DCHECK(channels == 1 || channels == 2);
+
+  // Max number of frames in an Opus packet is 48.
+  opus_int16 frame_sizes[48];
+  const unsigned char* frame_data[48];
+
+  // Parse packet to get the frames. But we only care about the first frame,
+  // since we can only decode the FEC from the first one.
+  if (opus_packet_parse(payload, static_cast<opus_int32>(payload_length_bytes),
+                        NULL, frame_data, frame_sizes, NULL) < 0) {
+    return 0;
+  }
+
+  if (frame_sizes[0] < 1) {
+    return 0;
+  }
+
+  // A frame starts with the LP layer. The LP layer begins with two to eight
+  // header bits.These consist of one VAD bit per SILK frame (up to 3),
+  // followed by a single flag indicating the presence of LBRR frames.
+  // For a stereo packet, these first flags correspond to the mid channel, and
+  // a second set of flags is included for the side channel. Because these are
+  // the first symbols decoded by the range coder and because they are coded
+  // as binary values with uniform probability, they can be extracted directly
+  // from the most significant bits of the first byte of compressed data.
+  for (int n = 0; n < channels; n++) {
+    // The LBRR bit for channel 1 is on the (`silk_frames` + 1)-th bit, and
+    // that of channel 2 is on the |(`silk_frames` + 1) * 2 + 1|-th bit.
+    if (frame_data[0][0] & (0x80 >> ((n + 1) * (silk_frames + 1) - 1)))
+      return 1;
+  }
+
+  return 0;
+}
+
+int WebRtcOpus_PacketHasVoiceActivity(const uint8_t* payload,
+                                      size_t payload_length_bytes) {
+  if (payload == NULL || payload_length_bytes == 0)
+    return 0;
+
+  // In CELT_ONLY mode we can not determine whether there is VAD.
+  if (payload[0] & 0x80)
+    return -1;
+
+  int silk_frames = WebRtcOpus_NumSilkFrames(payload);
+  if (silk_frames == 0)
+    return -1;
+
+  const int channels = opus_packet_get_nb_channels(payload);
+  RTC_DCHECK(channels == 1 || channels == 2);
+
+  // Max number of frames in an Opus packet is 48.
+  opus_int16 frame_sizes[48];
+  const unsigned char* frame_data[48];
+
+  // Parse packet to get the frames.
+  int frames =
+      opus_packet_parse(payload, static_cast<opus_int32>(payload_length_bytes),
+                        NULL, frame_data, frame_sizes, NULL);
+  if (frames < 0)
+    return -1;
+
+  // Iterate over all Opus frames which may contain multiple SILK frames.
+  for (int frame = 0; frame < frames; frame++) {
+    if (frame_sizes[frame] < 1) {
+      continue;
+    }
+    if (frame_data[frame][0] >> (8 - silk_frames))
+      return 1;
+    if (channels == 2 &&
+        (frame_data[frame][0] << (silk_frames + 1)) >> (8 - silk_frames))
+      return 1;
+  }
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.h
new file mode 100644
index 0000000000..89159ce1c0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.h
@@ -0,0 +1,547 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+// Opaque wrapper types for the codec state.
+typedef struct WebRtcOpusEncInst OpusEncInst;
+typedef struct WebRtcOpusDecInst OpusDecInst;
+
+/****************************************************************************
+ * WebRtcOpus_EncoderCreate(...)
+ *
+ * This function creates an Opus encoder that encodes mono or stereo.
+ *
+ * Input:
+ *      - channels           : number of channels; 1 or 2.
+ *      - application        : 0 - VOIP applications.
+ *                                 Favor speech intelligibility.
+ *                             1 - Audio applications.
+ *                                 Favor faithfulness to the original input.
+ *      - sample_rate_hz     : sample rate of input audio
+ *
+ * Output:
+ *      - inst               : a pointer to Encoder context that is created
+ *                             if success.
+ *
+ * Return value              : 0 - Success
+ *                            -1 - Error
+ */
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
+                                 size_t channels,
+                                 int32_t application,
+                                 int sample_rate_hz);
+
+/****************************************************************************
+ * WebRtcOpus_MultistreamEncoderCreate(...)
+ *
+ * This function creates an Opus encoder with any supported channel count.
+ *
+ * Input:
+ *      - channels           : number of channels in the input of the encoder.
+ *      - application        : 0 - VOIP applications.
+ *                                 Favor speech intelligibility.
+ *                             1 - Audio applications.
+ *                                 Favor faithfulness to the original input.
+ *      - streams            : number of streams, as described in RFC 7845.
+ *      - coupled_streams    : number of coupled streams, as described in
+ *                             RFC 7845.
+ *      - channel_mapping    : the channel mapping; pointer to array of
+ *                             `channel` bytes, as described in RFC 7845.
+ *
+ * Output:
+ *      - inst               : a pointer to Encoder context that is created
+ *                             if success.
+ *
+ * Return value              : 0 - Success
+ *                            -1 - Error
+ */
+int16_t WebRtcOpus_MultistreamEncoderCreate(
+    OpusEncInst** inst,
+    size_t channels,
+    int32_t application,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping);
+
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Encode(...)
+ *
+ * This function encodes audio as a series of Opus frames and inserts
+ * it into a packet. Input buffer can be any length.
+ *
+ * Input:
+ *      - inst                  : Encoder context
+ *      - audio_in              : Input speech data buffer
+ *      - samples               : Samples per channel in audio_in
+ *      - length_encoded_buffer : Output buffer size
+ *
+ * Output:
+ *      - encoded               : Output compressed data buffer
+ *
+ * Return value                 : >=0 - Length (in bytes) of coded data
+ *                                -1 - Error
+ */
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      size_t samples,
+                      size_t length_encoded_buffer,
+                      uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcOpus_SetBitRate(...)
+ *
+ * This function adjusts the target bitrate of the encoder.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - rate               : New target bitrate
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate);
+
+/****************************************************************************
+ * WebRtcOpus_SetPacketLossRate(...)
+ *
+ * This function configures the encoder's expected packet loss percentage.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - loss_rate          : loss percentage in the range 0-100, inclusive.
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate);
+
+/****************************************************************************
+ * WebRtcOpus_SetMaxPlaybackRate(...)
+ *
+ * Configures the maximum playback rate for encoding. Due to hardware
+ * limitations, the receiver may render audio up to a playback rate. Opus
+ * encoder can use this information to optimize for network usage and encoding
+ * complexity. This will affect the audio bandwidth in the coded audio. However,
+ * the input/output sample rate is not affected.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - frequency_hz       : Maximum playback rate in Hz.
+ *                             This parameter can take any value. The relation
+ *                             between the value and the Opus internal mode is
+ *                             as following:
+ *                             frequency_hz <= 8000           narrow band
+ *                             8000 < frequency_hz <= 12000   medium band
+ *                             12000 < frequency_hz <= 16000  wide band
+ *                             16000 < frequency_hz <= 24000  super wide band
+ *                             frequency_hz > 24000           full band
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz);
+
+/****************************************************************************
+ * WebRtcOpus_GetMaxPlaybackRate(...)
+ *
+ * Queries the maximum playback rate for encoding. If different single-stream
+ * encoders have different maximum playback rates, this function fails.
+ *
+ * Input:
+ *      - inst               : Encoder context.
+ * Output:
+ *      - result_hz          : The maximum playback rate in Hz.
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
+                                      int32_t* result_hz);
+
+/* TODO(minyue): Check whether an API to check the FEC and the packet loss rate
+ * is needed. It might not be very useful since there are not many use cases and
+ * the caller can always maintain the states. */
+
+/****************************************************************************
+ * WebRtcOpus_EnableFec()
+ *
+ * This function enables FEC for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_EnableFec(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DisableFec()
+ *
+ * This function disables FEC for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DisableFec(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_EnableDtx()
+ *
+ * This function enables Opus internal DTX for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DisableDtx()
+ *
+ * This function disables Opus internal DTX for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_GetUseDtx()
+ *
+ * This function gets the DTX configuration used for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Encoder does not use DTX.
+ *                              1 - Encoder uses DTX.
+ *                             -1 - Error.
+ */
+int16_t WebRtcOpus_GetUseDtx(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_EnableCbr()
+ *
+ * This function enables CBR for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DisableCbr()
+ *
+ * This function disables CBR for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DisableCbr(OpusEncInst* inst);
+
+/*
+ * WebRtcOpus_SetComplexity(...)
+ *
+ * This function adjusts the computational complexity. The effect is the same as
+ * calling the complexity setting of Opus as an Opus encoder related CTL.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - complexity         : New target complexity (0-10, inclusive)
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity);
+
+/*
+ * WebRtcOpus_GetBandwidth(...)
+ *
+ * This function returns the current bandwidth.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  Bandwidth - Success
+ *                             -1 - Error
+ */
+int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst);
+
+/*
+ * WebRtcOpus_SetBandwidth(...)
+ *
+ * By default Opus decides which bandwidth to encode the signal in depending on
+ * the the bitrate. This function overrules the previous setting and forces the
+ * encoder to encode in narrowband/wideband/fullband/etc.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - bandwidth          : New target bandwidth. Valid values are:
+ *                             OPUS_BANDWIDTH_NARROWBAND
+ *                             OPUS_BANDWIDTH_MEDIUMBAND
+ *                             OPUS_BANDWIDTH_WIDEBAND
+ *                             OPUS_BANDWIDTH_SUPERWIDEBAND
+ *                             OPUS_BANDWIDTH_FULLBAND
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth);
+
+/*
+ * WebRtcOpus_GetInDtx(...)
+ *
+ * Gets the DTX state of the encoder.
+ *
+ * Input:
+ *      - inst   : Encoder context
+ *
+ * Return value  : -1 - Error.
+ *                 1  - Last encoded frame was comfort noise update during DTX.
+ *                 0  - Last encoded frame was encoded with encoder not in DTX.
+ */
+int32_t WebRtcOpus_GetInDtx(OpusEncInst* inst);
+
+/*
+ * WebRtcOpus_SetForceChannels(...)
+ *
+ * If the encoder is initialized as a stereo encoder, Opus will by default
+ * decide whether to encode in mono or stereo based on the bitrate. This
+ * function overrules the previous setting, and forces the encoder to encode
+ * in auto/mono/stereo.
+ *
+ * If the Encoder is initialized as a mono encoder, and one tries to force
+ * stereo, the function will return an error.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - num_channels       : 0 - Not forced
+ *                             1 - Mono
+ *                             2 - Stereo
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels);
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
+                                 size_t channels,
+                                 int sample_rate_hz);
+
+/****************************************************************************
+ * WebRtcOpus_MultistreamDecoderCreate(...)
+ *
+ * This function creates an Opus decoder with any supported channel count.
+ *
+ * Input:
+ *      - channels           : number of output channels that the decoder
+ *                             will produce.
+ *      - streams            : number of encoded streams, as described in
+ *                             RFC 7845.
+ *      - coupled_streams    : number of coupled streams, as described in
+ *                             RFC 7845.
+ *      - channel_mapping    : the channel mapping; pointer to array of
+ *                             `channel` bytes, as described in RFC 7845.
+ *
+ * Output:
+ *      - inst               : a pointer to a Decoder context that is created
+ *                             if success.
+ *
+ * Return value              : 0 - Success
+ *                            -1 - Error
+ */
+int16_t WebRtcOpus_MultistreamDecoderCreate(
+    OpusDecInst** inst,
+    size_t channels,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping);
+
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DecoderChannels(...)
+ *
+ * This function returns the number of channels created for Opus decoder.
+ */
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DecoderInit(...)
+ *
+ * This function resets state of the decoder.
+ *
+ * Input:
+ *      - inst               : Decoder context
+ */
+void WebRtcOpus_DecoderInit(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Decode(...)
+ *
+ * This function decodes an Opus packet into one or more audio frames at the
+ * ACM interface's sampling rate (32 kHz).
+ *
+ * Input:
+ *      - inst               : Decoder context
+ *      - encoded            : Encoded data
+ *      - encoded_bytes      : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - audio_type         : 1 normal, 2 CNG (for Opus it should
+ *                             always return 1 since we're not using Opus's
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples per channel in decoded vector
+ *                             -1 - Error
+ */
+int WebRtcOpus_Decode(OpusDecInst* inst,
+                      const uint8_t* encoded,
+                      size_t encoded_bytes,
+                      int16_t* decoded,
+                      int16_t* audio_type);
+
+/****************************************************************************
+ * WebRtcOpus_DecodeFec(...)
+ *
+ * This function decodes the FEC data from an Opus packet into one or more audio
+ * frames at the ACM interface's sampling rate (32 kHz).
+ *
+ * Input:
+ *      - inst               : Decoder context
+ *      - encoded            : Encoded data
+ *      - encoded_bytes      : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector (previous frame)
+ *
+ * Return value              : >0 - Samples per channel in decoded vector
+ *                              0 - No FEC data in the packet
+ *                             -1 - Error
+ */
+int WebRtcOpus_DecodeFec(OpusDecInst* inst,
+                         const uint8_t* encoded,
+                         size_t encoded_bytes,
+                         int16_t* decoded,
+                         int16_t* audio_type);
+
+/****************************************************************************
+ * WebRtcOpus_DurationEst(...)
+ *
+ * This function calculates the duration of an opus packet.
+ * Input:
+ *        - inst                 : Decoder context
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *
+ * Return value                  : The duration of the packet, in samples per
+ *                                 channel.
+ */
+int WebRtcOpus_DurationEst(OpusDecInst* inst,
+                           const uint8_t* payload,
+                           size_t payload_length_bytes);
+
+/****************************************************************************
+ * WebRtcOpus_PlcDuration(...)
+ *
+ * This function calculates the duration of a frame returned by packet loss
+ * concealment (PLC).
+ *
+ * Input:
+ *        - inst                 : Decoder context
+ *
+ * Return value                  : The duration of a frame returned by PLC, in
+ *                                 samples per channel.
+ */
+int WebRtcOpus_PlcDuration(OpusDecInst* inst);
+
+/* TODO(minyue): Check whether it is needed to add a decoder context to the
+ * arguments, like WebRtcOpus_DurationEst(...). In fact, the packet itself tells
+ * the duration. The decoder context in WebRtcOpus_DurationEst(...) is not used.
+ * So it may be advisable to remove it from WebRtcOpus_DurationEst(...). */
+
+/****************************************************************************
+ * WebRtcOpus_FecDurationEst(...)
+ *
+ * This function calculates the duration of the FEC data within an opus packet.
+ * Input:
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *        - sample_rate_hz       : Sample rate of output audio
+ *
+ * Return value                  : >0 - The duration of the FEC data in the
+ *                                 packet in samples per channel.
+ *                                  0 - No FEC data in the packet.
+ */
+int WebRtcOpus_FecDurationEst(const uint8_t* payload,
+                              size_t payload_length_bytes,
+                              int sample_rate_hz);
+
+/****************************************************************************
+ * WebRtcOpus_PacketHasFec(...)
+ *
+ * This function detects if an opus packet has FEC.
+ * Input:
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *
+ * Return value                  : 0 - the packet does NOT contain FEC.
+ *                                 1 - the packet contains FEC.
+ */
+int WebRtcOpus_PacketHasFec(const uint8_t* payload,
+                            size_t payload_length_bytes);
+
+/****************************************************************************
+ * WebRtcOpus_PacketHasVoiceActivity(...)
+ *
+ * This function returns the SILK VAD information encoded in the opus packet.
+ * For CELT-only packets that do not have VAD information, it returns -1.
+ * Input:
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *
+ * Return value                  : 0 - no frame had the VAD flag set.
+ *                                 1 - at least one frame had the VAD flag set.
+ *                                -1 - VAD status could not be determined.
+ */
+int WebRtcOpus_PacketHasVoiceActivity(const uint8_t* payload,
+                                      size_t payload_length_bytes);
+
+#ifdef __cplusplus
+}  // extern "C"
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
new file mode 100644
index 0000000000..4477e8a5f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+
+using ::std::string;
+
+namespace webrtc {
+
+static const int kOpusBlockDurationMs = 20;
+static const int kOpusSamplingKhz = 48;
+
+class OpusSpeedTest : public AudioCodecSpeedTest {
+ protected:
+  OpusSpeedTest();
+  void SetUp() override;
+  void TearDown() override;
+  float EncodeABlock(int16_t* in_data,
+                     uint8_t* bit_stream,
+                     size_t max_bytes,
+                     size_t* encoded_bytes) override;
+  float DecodeABlock(const uint8_t* bit_stream,
+                     size_t encoded_bytes,
+                     int16_t* out_data) override;
+  WebRtcOpusEncInst* opus_encoder_;
+  WebRtcOpusDecInst* opus_decoder_;
+};
+
+OpusSpeedTest::OpusSpeedTest()
+    : AudioCodecSpeedTest(kOpusBlockDurationMs,
+                          kOpusSamplingKhz,
+                          kOpusSamplingKhz),
+      opus_encoder_(NULL),
+      opus_decoder_(NULL) {}
+
+void OpusSpeedTest::SetUp() {
+  AudioCodecSpeedTest::SetUp();
+  // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
+  int app = channels_ == 1 ? 0 : 1;
+  /* Create encoder memory. */
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
+  EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
+  /* Set bitrate. */
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
+}
+
+void OpusSpeedTest::TearDown() {
+  AudioCodecSpeedTest::TearDown();
+  /* Free memory. */
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+float OpusSpeedTest::EncodeABlock(int16_t* in_data,
+                                  uint8_t* bit_stream,
+                                  size_t max_bytes,
+                                  size_t* encoded_bytes) {
+  clock_t clocks = clock();
+  int value = WebRtcOpus_Encode(opus_encoder_, in_data, input_length_sample_,
+                                max_bytes, bit_stream);
+  clocks = clock() - clocks;
+  EXPECT_GT(value, 0);
+  *encoded_bytes = static_cast<size_t>(value);
+  return 1000.0 * clocks / CLOCKS_PER_SEC;
+}
+
+float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
+                                  size_t encoded_bytes,
+                                  int16_t* out_data) {
+  int value;
+  int16_t audio_type;
+  clock_t clocks = clock();
+  value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
+                            &audio_type);
+  clocks = clock() - clocks;
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
+  return 1000.0 * clocks / CLOCKS_PER_SEC;
+}
+
+/* Test audio length in second. */
+constexpr size_t kDurationSec = 400;
+
+#define ADD_TEST(complexity)                                           \
+  TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) {           \
+    /* Set complexity. */                                              \
+    printf("Setting complexity to %d ...\n", complexity);              \
+    EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
+    EncodeDecode(kDurationSec);                                        \
+  }
+
+ADD_TEST(10)
+ADD_TEST(9)
+ADD_TEST(8)
+ADD_TEST(7)
+ADD_TEST(6)
+ADD_TEST(5)
+ADD_TEST(4)
+ADD_TEST(3)
+ADD_TEST(2)
+ADD_TEST(1)
+ADD_TEST(0)
+
+#define ADD_BANDWIDTH_TEST(bandwidth)                                \
+  TEST_P(OpusSpeedTest, OpusSetBandwidthTest##bandwidth) {           \
+    /* Set bandwidth. */                                             \
+    printf("Setting bandwidth to %d ...\n", bandwidth);              \
+    EXPECT_EQ(0, WebRtcOpus_SetBandwidth(opus_encoder_, bandwidth)); \
+    EncodeDecode(kDurationSec);                                      \
+  }
+
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_NARROWBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_MEDIUMBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_WIDEBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_SUPERWIDEBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_FULLBAND)
+
+// List all test cases: (channel, bit rat, filename, extension).
+const coding_param param_set[] = {
+    std::make_tuple(1,
+                    64000,
+                    string("audio_coding/speech_mono_32_48kHz"),
+                    string("pcm"),
+                    true),
+    std::make_tuple(1,
+                    32000,
+                    string("audio_coding/speech_mono_32_48kHz"),
+                    string("pcm"),
+                    true),
+    std::make_tuple(2,
+                    64000,
+                    string("audio_coding/music_stereo_48kHz"),
+                    string("pcm"),
+                    true)};
+
+INSTANTIATE_TEST_SUITE_P(AllTest,
+                         OpusSpeedTest,
+                         ::testing::ValuesIn(param_set));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
new file mode 100644
index 0000000000..4a9156ad58
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -0,0 +1,979 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+// Equivalent to SDP params
+// {{"channel_mapping", "0,1,2,3"}, {"coupled_streams", "2"}}.
+constexpr unsigned char kQuadChannelMapping[] = {0, 1, 2, 3};
+constexpr int kQuadTotalStreams = 2;
+constexpr int kQuadCoupledStreams = 2;
+
+constexpr unsigned char kStereoChannelMapping[] = {0, 1};
+constexpr int kStereoTotalStreams = 1;
+constexpr int kStereoCoupledStreams = 1;
+
+constexpr unsigned char kMonoChannelMapping[] = {0};
+constexpr int kMonoTotalStreams = 1;
+constexpr int kMonoCoupledStreams = 0;
+
+void CreateSingleOrMultiStreamEncoder(WebRtcOpusEncInst** opus_encoder,
+                                      int channels,
+                                      int application,
+                                      bool use_multistream,
+                                      int encoder_sample_rate_hz) {
+  EXPECT_TRUE(channels == 1 || channels == 2 || use_multistream);
+  if (use_multistream) {
+    EXPECT_EQ(encoder_sample_rate_hz, 48000);
+    if (channels == 1) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
+                       opus_encoder, channels, application, kMonoTotalStreams,
+                       kMonoCoupledStreams, kMonoChannelMapping));
+    } else if (channels == 2) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
+                       opus_encoder, channels, application, kStereoTotalStreams,
+                       kStereoCoupledStreams, kStereoChannelMapping));
+    } else if (channels == 4) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
+                       opus_encoder, channels, application, kQuadTotalStreams,
+                       kQuadCoupledStreams, kQuadChannelMapping));
+    } else {
+      EXPECT_TRUE(false) << channels;
+    }
+  } else {
+    EXPECT_EQ(0, WebRtcOpus_EncoderCreate(opus_encoder, channels, application,
+                                          encoder_sample_rate_hz));
+  }
+}
+
+void CreateSingleOrMultiStreamDecoder(WebRtcOpusDecInst** opus_decoder,
+                                      int channels,
+                                      bool use_multistream,
+                                      int decoder_sample_rate_hz) {
+  EXPECT_TRUE(channels == 1 || channels == 2 || use_multistream);
+  if (use_multistream) {
+    EXPECT_EQ(decoder_sample_rate_hz, 48000);
+    if (channels == 1) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
+                       opus_decoder, channels, kMonoTotalStreams,
+                       kMonoCoupledStreams, kMonoChannelMapping));
+    } else if (channels == 2) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
+                       opus_decoder, channels, kStereoTotalStreams,
+                       kStereoCoupledStreams, kStereoChannelMapping));
+    } else if (channels == 4) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
+                       opus_decoder, channels, kQuadTotalStreams,
+                       kQuadCoupledStreams, kQuadChannelMapping));
+    } else {
+      EXPECT_TRUE(false) << channels;
+    }
+  } else {
+    EXPECT_EQ(0, WebRtcOpus_DecoderCreate(opus_decoder, channels,
+                                          decoder_sample_rate_hz));
+  }
+}
+
+int SamplesPerChannel(int sample_rate_hz, int duration_ms) {
+  const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz, 1000);
+  return samples_per_ms * duration_ms;
+}
+
+using test::AudioLoop;
+using ::testing::Combine;
+using ::testing::TestWithParam;
+using ::testing::Values;
+
+// Maximum number of bytes in output bitstream.
+const size_t kMaxBytes = 2000;
+
+class OpusTest
+    : public TestWithParam<::testing::tuple<size_t, int, bool, int, int>> {
+ protected:
+  OpusTest() = default;
+
+  void TestDtxEffect(bool dtx, int block_length_ms);
+
+  void TestCbrEffect(bool dtx, int block_length_ms);
+
+  // Prepare `speech_data_` for encoding, read from a hard-coded file.
+  // After preparation, `speech_data_.GetNextBlock()` returns a pointer to a
+  // block of `block_length_ms` milliseconds. The data is looped every
+  // `loop_length_ms` milliseconds.
+  void PrepareSpeechData(int block_length_ms, int loop_length_ms);
+
+  int EncodeDecode(WebRtcOpusEncInst* encoder,
+                   rtc::ArrayView<const int16_t> input_audio,
+                   WebRtcOpusDecInst* decoder,
+                   int16_t* output_audio,
+                   int16_t* audio_type);
+
+  void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
+                          opus_int32 expect,
+                          int32_t set);
+
+  void CheckAudioBounded(const int16_t* audio,
+                         size_t samples,
+                         size_t channels,
+                         uint16_t bound) const;
+
+  WebRtcOpusEncInst* opus_encoder_ = nullptr;
+  WebRtcOpusDecInst* opus_decoder_ = nullptr;
+  AudioLoop speech_data_;
+  uint8_t bitstream_[kMaxBytes];
+  size_t encoded_bytes_ = 0;
+  const size_t channels_{std::get<0>(GetParam())};
+  const int application_{std::get<1>(GetParam())};
+  const bool use_multistream_{std::get<2>(GetParam())};
+  const int encoder_sample_rate_hz_{std::get<3>(GetParam())};
+  const int decoder_sample_rate_hz_{std::get<4>(GetParam())};
+};
+
+}  // namespace
+
+// Singlestream: Try all combinations.
+INSTANTIATE_TEST_SUITE_P(Singlestream,
+                         OpusTest,
+                         testing::Combine(testing::Values(1, 2),
+                                          testing::Values(0, 1),
+                                          testing::Values(false),
+                                          testing::Values(16000, 48000),
+                                          testing::Values(16000, 48000)));
+
+// Multistream: Some representative cases (only 48 kHz for now).
+INSTANTIATE_TEST_SUITE_P(
+    Multistream,
+    OpusTest,
+    testing::Values(std::make_tuple(1, 0, true, 48000, 48000),
+                    std::make_tuple(2, 1, true, 48000, 48000),
+                    std::make_tuple(4, 0, true, 48000, 48000),
+                    std::make_tuple(4, 1, true, 48000, 48000)));
+
+void OpusTest::PrepareSpeechData(int block_length_ms, int loop_length_ms) {
+  std::map<int, std::string> channel_to_basename = {
+      {1, "audio_coding/testfile32kHz"},
+      {2, "audio_coding/teststereo32kHz"},
+      {4, "audio_coding/speech_4_channels_48k_one_second"}};
+  std::map<int, std::string> channel_to_suffix = {
+      {1, "pcm"}, {2, "pcm"}, {4, "wav"}};
+  const std::string file_name = webrtc::test::ResourcePath(
+      channel_to_basename[channels_], channel_to_suffix[channels_]);
+  if (loop_length_ms < block_length_ms) {
+    loop_length_ms = block_length_ms;
+  }
+  const int sample_rate_khz =
+      rtc::CheckedDivExact(encoder_sample_rate_hz_, 1000);
+  EXPECT_TRUE(speech_data_.Init(file_name,
+                                loop_length_ms * sample_rate_khz * channels_,
+                                block_length_ms * sample_rate_khz * channels_));
+}
+
+void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
+                                  opus_int32 expect,
+                                  int32_t set) {
+  opus_int32 bandwidth;
+  EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, set));
+  EXPECT_EQ(0, WebRtcOpus_GetMaxPlaybackRate(opus_encoder_, &bandwidth));
+  EXPECT_EQ(expect, bandwidth);
+}
+
+void OpusTest::CheckAudioBounded(const int16_t* audio,
+                                 size_t samples,
+                                 size_t channels,
+                                 uint16_t bound) const {
+  for (size_t i = 0; i < samples; ++i) {
+    for (size_t c = 0; c < channels; ++c) {
+      ASSERT_GE(audio[i * channels + c], -bound);
+      ASSERT_LE(audio[i * channels + c], bound);
+    }
+  }
+}
+
+int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
+                           rtc::ArrayView<const int16_t> input_audio,
+                           WebRtcOpusDecInst* decoder,
+                           int16_t* output_audio,
+                           int16_t* audio_type) {
+  const int input_samples_per_channel =
+      rtc::CheckedDivExact(input_audio.size(), channels_);
+  int encoded_bytes_int =
+      WebRtcOpus_Encode(encoder, input_audio.data(), input_samples_per_channel,
+                        kMaxBytes, bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
+  if (encoded_bytes_ != 0) {
+    int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
+    int act_len = WebRtcOpus_Decode(decoder, bitstream_, encoded_bytes_,
+                                    output_audio, audio_type);
+    EXPECT_EQ(est_len, act_len);
+    return act_len;
+  } else {
+    int total_dtx_len = 0;
+    const int output_samples_per_channel = input_samples_per_channel *
+                                           decoder_sample_rate_hz_ /
+                                           encoder_sample_rate_hz_;
+    while (total_dtx_len < output_samples_per_channel) {
+      int est_len = WebRtcOpus_DurationEst(decoder, NULL, 0);
+      int act_len = WebRtcOpus_Decode(decoder, NULL, 0,
+                                      &output_audio[total_dtx_len * channels_],
+                                      audio_type);
+      EXPECT_EQ(est_len, act_len);
+      total_dtx_len += act_len;
+    }
+    return total_dtx_len;
+  }
+}
+
+// Test if encoder/decoder can enter DTX mode properly and do not enter DTX when
+// they should not. This test is signal dependent.
+void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
+  PrepareSpeechData(block_length_ms, 2000);
+  const size_t input_samples =
+      rtc::CheckedDivExact(encoder_sample_rate_hz_, 1000) * block_length_ms;
+  const size_t output_samples =
+      rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Set input audio as silence.
+  std::vector<int16_t> silence(input_samples * channels_, 0);
+
+  // Setting DTX.
+  EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_)
+                   : WebRtcOpus_DisableDtx(opus_encoder_));
+
+  int16_t audio_type;
+  int16_t* output_data_decode = new int16_t[output_samples * channels_];
+
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_EQ(output_samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+                  output_data_decode, &audio_type)));
+    // If not DTX, it should never enter DTX mode. If DTX, we do not care since
+    // whether it enters DTX depends on the signal type.
+    if (!dtx) {
+      EXPECT_GT(encoded_bytes_, 1U);
+      EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(0, audio_type);  // Speech.
+    }
+  }
+
+  // We input some silent segments. In DTX mode, the encoder will stop sending.
+  // However, DTX may happen after a while.
+  for (int i = 0; i < 30; ++i) {
+    EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                  opus_encoder_, silence, opus_decoder_,
+                                  output_data_decode, &audio_type)));
+    if (!dtx) {
+      EXPECT_GT(encoded_bytes_, 1U);
+      EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(0, audio_type);  // Speech.
+    } else if (encoded_bytes_ == 1) {
+      EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(2, audio_type);  // Comfort noise.
+      break;
+    }
+  }
+
+  // When Opus is in DTX, it wakes up in a regular basis. It sends two packets,
+  // one with an arbitrary size and the other of 1-byte, then stops sending for
+  // a certain number of frames.
+
+  // `max_dtx_frames` is the maximum number of frames Opus can stay in DTX.
+  // TODO(kwiberg): Why does this number depend on the encoding sample rate?
+  const int max_dtx_frames =
+      (encoder_sample_rate_hz_ == 16000 ? 800 : 400) / block_length_ms + 1;
+
+  // We run `kRunTimeMs` milliseconds of pure silence.
+  const int kRunTimeMs = 4500;
+
+  // We check that, after a `kCheckTimeMs` milliseconds (given that the CNG in
+  // Opus needs time to adapt), the absolute values of DTX decoded signal are
+  // bounded by `kOutputValueBound`.
+  const int kCheckTimeMs = 4000;
+
+#if defined(OPUS_FIXED_POINT)
+  // Fixed-point Opus generates a random (comfort) noise, which has a less
+  // predictable value bound than its floating-point Opus. This value depends on
+  // input signal, and the time window for checking the output values (between
+  // `kCheckTimeMs` and `kRunTimeMs`).
+  const uint16_t kOutputValueBound = 30;
+
+#else
+  const uint16_t kOutputValueBound = 2;
+#endif
+
+  int time = 0;
+  while (time < kRunTimeMs) {
+    // DTX mode is maintained for maximum `max_dtx_frames` frames.
+    int i = 0;
+    for (; i < max_dtx_frames; ++i) {
+      time += block_length_ms;
+      EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                    opus_encoder_, silence, opus_decoder_,
+                                    output_data_decode, &audio_type)));
+      if (dtx) {
+        if (encoded_bytes_ > 1)
+          break;
+        EXPECT_EQ(0U, encoded_bytes_)  // Send 0 byte.
+            << "Opus should have entered DTX mode.";
+        EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+        EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+        EXPECT_EQ(2, audio_type);  // Comfort noise.
+        if (time >= kCheckTimeMs) {
+          CheckAudioBounded(output_data_decode, output_samples, channels_,
+                            kOutputValueBound);
+        }
+      } else {
+        EXPECT_GT(encoded_bytes_, 1U);
+        EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+        EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+        EXPECT_EQ(0, audio_type);  // Speech.
+      }
+    }
+
+    if (dtx) {
+      // With DTX, Opus must stop transmission for some time.
+      EXPECT_GT(i, 1);
+    }
+
+    // We expect a normal payload.
+    EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+    EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+    EXPECT_EQ(0, audio_type);  // Speech.
+
+    // Enters DTX again immediately.
+    time += block_length_ms;
+    EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                  opus_encoder_, silence, opus_decoder_,
+                                  output_data_decode, &audio_type)));
+    if (dtx) {
+      EXPECT_EQ(1U, encoded_bytes_);  // Send 1 byte.
+      EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(2, audio_type);  // Comfort noise.
+      if (time >= kCheckTimeMs) {
+        CheckAudioBounded(output_data_decode, output_samples, channels_,
+                          kOutputValueBound);
+      }
+    } else {
+      EXPECT_GT(encoded_bytes_, 1U);
+      EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(0, audio_type);  // Speech.
+    }
+  }
+
+  silence[0] = 10000;
+  if (dtx) {
+    // Verify that encoder/decoder can jump out from DTX mode.
+    EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                  opus_encoder_, silence, opus_decoder_,
+                                  output_data_decode, &audio_type)));
+    EXPECT_GT(encoded_bytes_, 1U);
+    EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+    EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+    EXPECT_EQ(0, audio_type);  // Speech.
+  }
+
+  // Free memory.
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+// Test if CBR does what we expect.
+void OpusTest::TestCbrEffect(bool cbr, int block_length_ms) {
+  PrepareSpeechData(block_length_ms, 2000);
+  const size_t output_samples =
+      rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
+
+  int32_t max_pkt_size_diff = 0;
+  int32_t prev_pkt_size = 0;
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Setting CBR.
+  EXPECT_EQ(0, cbr ? WebRtcOpus_EnableCbr(opus_encoder_)
+                   : WebRtcOpus_DisableCbr(opus_encoder_));
+
+  int16_t audio_type;
+  std::vector<int16_t> audio_out(output_samples * channels_);
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_EQ(output_samples,
+              static_cast<size_t>(
+                  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                               opus_decoder_, audio_out.data(), &audio_type)));
+
+    if (prev_pkt_size > 0) {
+      int32_t diff = std::abs((int32_t)encoded_bytes_ - prev_pkt_size);
+      max_pkt_size_diff = std::max(max_pkt_size_diff, diff);
+    }
+    prev_pkt_size = rtc::checked_cast<int32_t>(encoded_bytes_);
+  }
+
+  if (cbr) {
+    EXPECT_EQ(max_pkt_size_diff, 0);
+  } else {
+    EXPECT_GT(max_pkt_size_diff, 0);
+  }
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+// Test failing Create.
+TEST(OpusTest, OpusCreateFail) {
+  WebRtcOpusEncInst* opus_encoder;
+  WebRtcOpusDecInst* opus_decoder;
+
+  // Test to see that an invalid pointer is caught.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(NULL, 1, 0, 48000));
+  // Invalid channel number.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 257, 0, 48000));
+  // Invalid applciation mode.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 2, 48000));
+  // Invalid sample rate.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 0, 12345));
+
+  EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(NULL, 1, 48000));
+  // Invalid channel number.
+  EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 257, 48000));
+  // Invalid sample rate.
+  EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 1, 12345));
+}
+
+// Test failing Free.
+TEST(OpusTest, OpusFreeFail) {
+  // Test to see that an invalid pointer is caught.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderFree(NULL));
+  EXPECT_EQ(-1, WebRtcOpus_DecoderFree(NULL));
+}
+
+// Test normal Create and Free.
+TEST_P(OpusTest, OpusCreateFree) {
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+  EXPECT_TRUE(opus_encoder_ != NULL);
+  EXPECT_TRUE(opus_decoder_ != NULL);
+  // Free encoder and decoder memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+#define ENCODER_CTL(inst, vargs)               \
+  inst->encoder                                \
+      ? opus_encoder_ctl(inst->encoder, vargs) \
+      : opus_multistream_encoder_ctl(inst->multistream_encoder, vargs)
+
+TEST_P(OpusTest, OpusEncodeDecode) {
+  PrepareSpeechData(20, 20);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Check number of channels for decoder.
+  EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
+
+  // Check application mode.
+  opus_int32 app;
+  ENCODER_CTL(opus_encoder_, OPUS_GET_APPLICATION(&app));
+  EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO,
+            app);
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  int16_t* output_data_decode =
+      new int16_t[decode_samples_per_channel * channels_];
+  EXPECT_EQ(decode_samples_per_channel,
+            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                         opus_decoder_, output_data_decode, &audio_type));
+
+  // Free memory.
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusSetBitRate) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
+
+  // Create encoder memory, try with different bitrates.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 30000));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 300000));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 600000));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusSetComplexity) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 9));
+
+  // Create encoder memory, try with different complexities.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 0));
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 10));
+  EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 11));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusSetBandwidth) {
+  if (channels_ > 2) {
+    // TODO(webrtc:10217): investigate why multi-stream Opus reports
+    // narrowband when it's configured with FULLBAND.
+    return;
+  }
+  PrepareSpeechData(20, 20);
+
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  std::unique_ptr<int16_t[]> output_data_decode(
+      new int16_t[decode_samples_per_channel * channels_]());
+
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1,
+            WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND));
+  EXPECT_EQ(-1, WebRtcOpus_GetBandwidth(opus_encoder_));
+
+  // Create encoder memory, try with different bandwidths.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  EXPECT_EQ(-1, WebRtcOpus_SetBandwidth(opus_encoder_,
+                                        OPUS_BANDWIDTH_NARROWBAND - 1));
+  EXPECT_EQ(0,
+            WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND));
+  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+               output_data_decode.get(), &audio_type);
+  EXPECT_EQ(OPUS_BANDWIDTH_NARROWBAND, WebRtcOpus_GetBandwidth(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_FULLBAND));
+  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+               output_data_decode.get(), &audio_type);
+  EXPECT_EQ(encoder_sample_rate_hz_ == 16000 ? OPUS_BANDWIDTH_WIDEBAND
+                                             : OPUS_BANDWIDTH_FULLBAND,
+            WebRtcOpus_GetBandwidth(opus_encoder_));
+  EXPECT_EQ(
+      -1, WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_FULLBAND + 1));
+  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+               output_data_decode.get(), &audio_type);
+  EXPECT_EQ(encoder_sample_rate_hz_ == 16000 ? OPUS_BANDWIDTH_WIDEBAND
+                                             : OPUS_BANDWIDTH_FULLBAND,
+            WebRtcOpus_GetBandwidth(opus_encoder_));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusForceChannels) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
+
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  ASSERT_NE(nullptr, opus_encoder_);
+
+  if (channels_ >= 2) {
+    EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 3));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 2));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0));
+  } else {
+    EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 2));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0));
+  }
+
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+// Encode and decode one frame, initialize the decoder and
+// decode once more.
+TEST_P(OpusTest, OpusDecodeInit) {
+  PrepareSpeechData(20, 20);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  int16_t* output_data_decode =
+      new int16_t[decode_samples_per_channel * channels_];
+  EXPECT_EQ(decode_samples_per_channel,
+            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                         opus_decoder_, output_data_decode, &audio_type));
+
+  WebRtcOpus_DecoderInit(opus_decoder_);
+
+  EXPECT_EQ(decode_samples_per_channel,
+            WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
+                              output_data_decode, &audio_type));
+
+  // Free memory.
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusEnableDisableFec) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_EnableFec(opus_encoder_));
+  EXPECT_EQ(-1, WebRtcOpus_DisableFec(opus_encoder_));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusEnableDisableDtx) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_EnableDtx(opus_encoder_));
+  EXPECT_EQ(-1, WebRtcOpus_DisableDtx(opus_encoder_));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  opus_int32 dtx;
+
+  // DTX is off by default.
+  ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
+  EXPECT_EQ(0, dtx);
+
+  // Test to enable DTX.
+  EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
+  ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
+  EXPECT_EQ(1, dtx);
+
+  // Test to disable DTX.
+  EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_encoder_));
+  ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
+  EXPECT_EQ(0, dtx);
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusDtxOff) {
+  TestDtxEffect(false, 10);
+  TestDtxEffect(false, 20);
+  TestDtxEffect(false, 40);
+}
+
+TEST_P(OpusTest, OpusDtxOn) {
+  if (channels_ > 2 || application_ != 0) {
+    // DTX does not work with OPUS_APPLICATION_AUDIO at low complexity settings.
+    // TODO(webrtc:10218): adapt the test to the sizes and order of multi-stream
+    // DTX packets.
+    return;
+  }
+  TestDtxEffect(true, 10);
+  TestDtxEffect(true, 20);
+  TestDtxEffect(true, 40);
+}
+
+TEST_P(OpusTest, OpusCbrOff) {
+  TestCbrEffect(false, 10);
+  TestCbrEffect(false, 20);
+  TestCbrEffect(false, 40);
+}
+
+TEST_P(OpusTest, OpusCbrOn) {
+  TestCbrEffect(true, 10);
+  TestCbrEffect(true, 20);
+  TestCbrEffect(true, 40);
+}
+
+TEST_P(OpusTest, OpusSetPacketLossRate) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
+  EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, -1));
+  EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 101));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusSetMaxPlaybackRate) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, 20000));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 48000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 24001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 24000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 16001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 16000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 12001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 12000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 8001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 8000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 4000);
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+// Test PLC.
+TEST_P(OpusTest, OpusDecodePlc) {
+  PrepareSpeechData(20, 20);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Check number of channels for decoder.
+  EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  int16_t* output_data_decode =
+      new int16_t[decode_samples_per_channel * channels_];
+  EXPECT_EQ(decode_samples_per_channel,
+            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                         opus_decoder_, output_data_decode, &audio_type));
+
+  // Call decoder PLC.
+  constexpr int kPlcDurationMs = 10;
+  const int plc_samples = decoder_sample_rate_hz_ * kPlcDurationMs / 1000;
+  int16_t* plc_buffer = new int16_t[plc_samples * channels_];
+  EXPECT_EQ(plc_samples,
+            WebRtcOpus_Decode(opus_decoder_, NULL, 0, plc_buffer, &audio_type));
+
+  // Free memory.
+  delete[] plc_buffer;
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+// Duration estimation.
+TEST_P(OpusTest, OpusDurationEstimation) {
+  PrepareSpeechData(20, 20);
+
+  // Create.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // 10 ms. We use only first 10 ms of a 20 ms block.
+  auto speech_block = speech_data_.GetNextBlock();
+  int encoded_bytes_int = WebRtcOpus_Encode(
+      opus_encoder_, speech_block.data(),
+      rtc::CheckedDivExact(speech_block.size(), 2 * channels_), kMaxBytes,
+      bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/10),
+            WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
+                                   static_cast<size_t>(encoded_bytes_int)));
+
+  // 20 ms
+  speech_block = speech_data_.GetNextBlock();
+  encoded_bytes_int =
+      WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+                        rtc::CheckedDivExact(speech_block.size(), channels_),
+                        kMaxBytes, bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20),
+            WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
+                                   static_cast<size_t>(encoded_bytes_int)));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusDecodeRepacketized) {
+  if (channels_ > 2) {
+    // As per the Opus documentation
+    // https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__repacketizer.html#details,
+    // multiple streams are not supported.
+    return;
+  }
+  constexpr size_t kPackets = 6;
+
+  PrepareSpeechData(20, 20 * kPackets);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  ASSERT_NE(nullptr, opus_encoder_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+  ASSERT_NE(nullptr, opus_decoder_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Check number of channels for decoder.
+  EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  std::unique_ptr<int16_t[]> output_data_decode(
+      new int16_t[kPackets * decode_samples_per_channel * channels_]);
+  OpusRepacketizer* rp = opus_repacketizer_create();
+
+  size_t num_packets = 0;
+  constexpr size_t kMaxCycles = 100;
+  for (size_t idx = 0; idx < kMaxCycles; ++idx) {
+    auto speech_block = speech_data_.GetNextBlock();
+    encoded_bytes_ =
+        WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+                          rtc::CheckedDivExact(speech_block.size(), channels_),
+                          kMaxBytes, bitstream_);
+    if (opus_repacketizer_cat(rp, bitstream_,
+                              rtc::checked_cast<opus_int32>(encoded_bytes_)) ==
+        OPUS_OK) {
+      ++num_packets;
+      if (num_packets == kPackets) {
+        break;
+      }
+    } else {
+      // Opus repacketizer cannot guarantee a success. We try again if it fails.
+      opus_repacketizer_init(rp);
+      num_packets = 0;
+    }
+  }
+  EXPECT_EQ(kPackets, num_packets);
+
+  encoded_bytes_ = opus_repacketizer_out(rp, bitstream_, kMaxBytes);
+
+  EXPECT_EQ(decode_samples_per_channel * kPackets,
+            static_cast<size_t>(WebRtcOpus_DurationEst(
+                opus_decoder_, bitstream_, encoded_bytes_)));
+
+  EXPECT_EQ(decode_samples_per_channel * kPackets,
+            static_cast<size_t>(
+                WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
+                                  output_data_decode.get(), &audio_type)));
+
+  // Free memory.
+  opus_repacketizer_destroy(rp);
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST(OpusVadTest, CeltUnknownStatus) {
+  const uint8_t celt[] = {0x80};
+  EXPECT_EQ(WebRtcOpus_PacketHasVoiceActivity(celt, 1), -1);
+}
+
+TEST(OpusVadTest, Mono20msVadSet) {
+  uint8_t silk20msMonoVad[] = {0x78, 0x80};
+  EXPECT_TRUE(WebRtcOpus_PacketHasVoiceActivity(silk20msMonoVad, 2));
+}
+
+TEST(OpusVadTest, Mono20MsVadUnset) {
+  uint8_t silk20msMonoSilence[] = {0x78, 0x00};
+  EXPECT_FALSE(WebRtcOpus_PacketHasVoiceActivity(silk20msMonoSilence, 2));
+}
+
+TEST(OpusVadTest, Stereo20MsVadOnSideChannel) {
+  uint8_t silk20msStereoVadSideChannel[] = {0x78 | 0x04, 0x20};
+  EXPECT_TRUE(
+      WebRtcOpus_PacketHasVoiceActivity(silk20msStereoVadSideChannel, 2));
+}
+
+TEST(OpusVadTest, TwoOpusMonoFramesVadOnSecond) {
+  uint8_t twoMonoFrames[] = {0x78 | 0x1, 0x00, 0x80};
+  EXPECT_TRUE(WebRtcOpus_PacketHasVoiceActivity(twoMonoFrames, 3));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn
new file mode 100644
index 0000000000..8bc0bf5e0e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn
@@ -0,0 +1,55 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../../../webrtc.gni")
+
+visibility = [
+  ":*",
+  "../../../:*",
+]
+
+if (rtc_include_tests) {
+  rtc_library("test") {
+    testonly = true
+
+    sources = [
+      "audio_ring_buffer.cc",
+      "audio_ring_buffer.h",
+      "blocker.cc",
+      "blocker.h",
+      "lapped_transform.cc",
+      "lapped_transform.h",
+    ]
+
+    deps = [
+      "../../../../../common_audio",
+      "../../../../../common_audio:common_audio_c",
+      "../../../../../rtc_base:checks",
+      "../../../../../rtc_base/memory:aligned_malloc",
+    ]
+  }
+
+  rtc_library("test_unittest") {
+    testonly = true
+
+    sources = [
+      "audio_ring_buffer_unittest.cc",
+      "blocker_unittest.cc",
+      "lapped_transform_unittest.cc",
+    ]
+
+    deps = [
+      ":test",
+      "../../../../../common_audio",
+      "../../../../../common_audio:common_audio_c",
+      "../../../../../rtc_base:macromagic",
+      "../../../../../test:test_support",
+      "//testing/gtest",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc
new file mode 100644
index 0000000000..2a71b43d2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+#include "common_audio/ring_buffer.h"
+#include "rtc_base/checks.h"
+
+// This is a simple multi-channel wrapper over the ring_buffer.h C interface.
+
+namespace webrtc {
+
+AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) {
+  buffers_.reserve(channels);
+  for (size_t i = 0; i < channels; ++i)
+    buffers_.push_back(WebRtc_CreateBuffer(max_frames, sizeof(float)));
+}
+
+AudioRingBuffer::~AudioRingBuffer() {
+  for (auto* buf : buffers_)
+    WebRtc_FreeBuffer(buf);
+}
+
+void AudioRingBuffer::Write(const float* const* data,
+                            size_t channels,
+                            size_t frames) {
+  RTC_DCHECK_EQ(buffers_.size(), channels);
+  for (size_t i = 0; i < channels; ++i) {
+    const size_t written = WebRtc_WriteBuffer(buffers_[i], data[i], frames);
+    RTC_CHECK_EQ(written, frames);
+  }
+}
+
+void AudioRingBuffer::Read(float* const* data, size_t channels, size_t frames) {
+  RTC_DCHECK_EQ(buffers_.size(), channels);
+  for (size_t i = 0; i < channels; ++i) {
+    const size_t read =
+        WebRtc_ReadBuffer(buffers_[i], nullptr, data[i], frames);
+    RTC_CHECK_EQ(read, frames);
+  }
+}
+
+size_t AudioRingBuffer::ReadFramesAvailable() const {
+  // All buffers have the same amount available.
+  return WebRtc_available_read(buffers_[0]);
+}
+
+size_t AudioRingBuffer::WriteFramesAvailable() const {
+  // All buffers have the same amount available.
+  return WebRtc_available_write(buffers_[0]);
+}
+
+void AudioRingBuffer::MoveReadPositionForward(size_t frames) {
+  for (auto* buf : buffers_) {
+    const size_t moved =
+        static_cast<size_t>(WebRtc_MoveReadPtr(buf, static_cast<int>(frames)));
+    RTC_CHECK_EQ(moved, frames);
+  }
+}
+
+void AudioRingBuffer::MoveReadPositionBackward(size_t frames) {
+  for (auto* buf : buffers_) {
+    const size_t moved = static_cast<size_t>(
+        -WebRtc_MoveReadPtr(buf, -static_cast<int>(frames)));
+    RTC_CHECK_EQ(moved, frames);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
new file mode 100644
index 0000000000..a280ca2410
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+struct RingBuffer;
+
+namespace webrtc {
+
+// A ring buffer tailored for float deinterleaved audio. Any operation that
+// cannot be performed as requested will cause a crash (e.g. insufficient data
+// in the buffer to fulfill a read request.)
+class AudioRingBuffer final {
+ public:
+  // Specify the number of channels and maximum number of frames the buffer will
+  // contain.
+  AudioRingBuffer(size_t channels, size_t max_frames);
+  ~AudioRingBuffer();
+
+  // Copies `data` to the buffer and advances the write pointer. `channels` must
+  // be the same as at creation time.
+  void Write(const float* const* data, size_t channels, size_t frames);
+
+  // Copies from the buffer to `data` and advances the read pointer. `channels`
+  // must be the same as at creation time.
+  void Read(float* const* data, size_t channels, size_t frames);
+
+  size_t ReadFramesAvailable() const;
+  size_t WriteFramesAvailable() const;
+
+  // Moves the read position. The forward version advances the read pointer
+  // towards the write pointer and the backward verison withdraws the read
+  // pointer away from the write pointer (i.e. flushing and stuffing the buffer
+  // respectively.)
+  void MoveReadPositionForward(size_t frames);
+  void MoveReadPositionBackward(size_t frames);
+
+ private:
+  // TODO(kwiberg): Use std::vector<std::unique_ptr<RingBuffer>> instead.
+  std::vector<RingBuffer*> buffers_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
new file mode 100644
index 0000000000..6dbc8ee9fe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+#include <memory>
+
+#include "common_audio/channel_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class AudioRingBufferTest
+    : public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
+};
+
+void ReadAndWriteTest(const ChannelBuffer<float>& input,
+                      size_t num_write_chunk_frames,
+                      size_t num_read_chunk_frames,
+                      size_t buffer_frames,
+                      ChannelBuffer<float>* output) {
+  const size_t num_channels = input.num_channels();
+  const size_t total_frames = input.num_frames();
+  AudioRingBuffer buf(num_channels, buffer_frames);
+  std::unique_ptr<float*[]> slice(new float*[num_channels]);
+
+  size_t input_pos = 0;
+  size_t output_pos = 0;
+  while (input_pos + buf.WriteFramesAvailable() < total_frames) {
+    // Write until the buffer is as full as possible.
+    while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
+      buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+                num_write_chunk_frames);
+      input_pos += num_write_chunk_frames;
+    }
+    // Read until the buffer is as empty as possible.
+    while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
+      EXPECT_LT(output_pos, total_frames);
+      buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+               num_read_chunk_frames);
+      output_pos += num_read_chunk_frames;
+    }
+  }
+
+  // Write and read the last bit.
+  if (input_pos < total_frames) {
+    buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+              total_frames - input_pos);
+  }
+  if (buf.ReadFramesAvailable()) {
+    buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+             buf.ReadFramesAvailable());
+  }
+  EXPECT_EQ(0u, buf.ReadFramesAvailable());
+}
+
+TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
+  const size_t kFrames = 5000;
+  const size_t num_channels = ::testing::get<3>(GetParam());
+
+  // Initialize the input data to an increasing sequence.
+  ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
+  for (size_t i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < kFrames; ++j)
+      input.channels()[i][j] = (i + 1) * (j + 1);
+
+  ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels));
+  ReadAndWriteTest(input, ::testing::get<0>(GetParam()),
+                   ::testing::get<1>(GetParam()), ::testing::get<2>(GetParam()),
+                   &output);
+
+  // Verify the read data matches the input.
+  for (size_t i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < kFrames; ++j)
+      EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioRingBufferTest,
+    AudioRingBufferTest,
+    ::testing::Combine(::testing::Values(10, 20, 42),  // num_write_chunk_frames
+                       ::testing::Values(1, 10, 17),   // num_read_chunk_frames
+                       ::testing::Values(100, 256),    // buffer_frames
+                       ::testing::Values(1, 4)));      // num_channels
+
+TEST_F(AudioRingBufferTest, MoveReadPosition) {
+  const size_t kNumChannels = 1;
+  const float kInputArray[] = {1, 2, 3, 4};
+  const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray);
+  ChannelBuffer<float> input(kNumFrames, kNumChannels);
+  input.SetDataForTesting(kInputArray, kNumFrames);
+  AudioRingBuffer buf(kNumChannels, kNumFrames);
+  buf.Write(input.channels(), kNumChannels, kNumFrames);
+
+  buf.MoveReadPositionForward(3);
+  ChannelBuffer<float> output(1, kNumChannels);
+  buf.Read(output.channels(), kNumChannels, 1);
+  EXPECT_EQ(4, output.channels()[0][0]);
+  buf.MoveReadPositionBackward(3);
+  buf.Read(output.channels(), kNumChannels, 1);
+  EXPECT_EQ(2, output.channels()[0][0]);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc
new file mode 100644
index 0000000000..33406cead9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc
@@ -0,0 +1,215 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+
+#include <string.h>
+
+#include "rtc_base/checks.h"
+
+namespace {
+
+// Adds `a` and `b` frame by frame into `result` (basically matrix addition).
+void AddFrames(const float* const* a,
+               size_t a_start_index,
+               const float* const* b,
+               int b_start_index,
+               size_t num_frames,
+               size_t num_channels,
+               float* const* result,
+               size_t result_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    for (size_t j = 0; j < num_frames; ++j) {
+      result[i][j + result_start_index] =
+          a[i][j + a_start_index] + b[i][j + b_start_index];
+    }
+  }
+}
+
+// Copies `src` into `dst` channel by channel.
+void CopyFrames(const float* const* src,
+                size_t src_start_index,
+                size_t num_frames,
+                size_t num_channels,
+                float* const* dst,
+                size_t dst_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    memcpy(&dst[i][dst_start_index], &src[i][src_start_index],
+           num_frames * sizeof(dst[i][dst_start_index]));
+  }
+}
+
+// Moves `src` into `dst` channel by channel.
+void MoveFrames(const float* const* src,
+                size_t src_start_index,
+                size_t num_frames,
+                size_t num_channels,
+                float* const* dst,
+                size_t dst_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    memmove(&dst[i][dst_start_index], &src[i][src_start_index],
+            num_frames * sizeof(dst[i][dst_start_index]));
+  }
+}
+
+void ZeroOut(float* const* buffer,
+             size_t starting_idx,
+             size_t num_frames,
+             size_t num_channels) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    memset(&buffer[i][starting_idx], 0,
+           num_frames * sizeof(buffer[i][starting_idx]));
+  }
+}
+
+// Pointwise multiplies each channel of `frames` with `window`. Results are
+// stored in `frames`.
+void ApplyWindow(const float* window,
+                 size_t num_frames,
+                 size_t num_channels,
+                 float* const* frames) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    for (size_t j = 0; j < num_frames; ++j) {
+      frames[i][j] = frames[i][j] * window[j];
+    }
+  }
+}
+
+size_t gcd(size_t a, size_t b) {
+  size_t tmp;
+  while (b) {
+    tmp = a;
+    a = b;
+    b = tmp % b;
+  }
+  return a;
+}
+
+}  // namespace
+
+namespace webrtc {
+
+Blocker::Blocker(size_t chunk_size,
+                 size_t block_size,
+                 size_t num_input_channels,
+                 size_t num_output_channels,
+                 const float* window,
+                 size_t shift_amount,
+                 BlockerCallback* callback)
+    : chunk_size_(chunk_size),
+      block_size_(block_size),
+      num_input_channels_(num_input_channels),
+      num_output_channels_(num_output_channels),
+      initial_delay_(block_size_ - gcd(chunk_size, shift_amount)),
+      frame_offset_(0),
+      input_buffer_(num_input_channels_, chunk_size_ + initial_delay_),
+      output_buffer_(chunk_size_ + initial_delay_, num_output_channels_),
+      input_block_(block_size_, num_input_channels_),
+      output_block_(block_size_, num_output_channels_),
+      window_(new float[block_size_]),
+      shift_amount_(shift_amount),
+      callback_(callback) {
+  RTC_CHECK_LE(num_output_channels_, num_input_channels_);
+  RTC_CHECK_LE(shift_amount_, block_size_);
+
+  memcpy(window_.get(), window, block_size_ * sizeof(*window_.get()));
+  input_buffer_.MoveReadPositionBackward(initial_delay_);
+}
+
+Blocker::~Blocker() = default;
+
+// When block_size < chunk_size the input and output buffers look like this:
+//
+//                      delay*             chunk_size    chunk_size + delay*
+//  buffer: <-------------|---------------------|---------------|>
+//                _a_              _b_                 _c_
+//
+// On each call to ProcessChunk():
+// 1. New input gets read into sections _b_ and _c_ of the input buffer.
+// 2. We block starting from frame_offset.
+// 3. We block until we reach a block `bl` that doesn't contain any frames
+//    from sections _a_ or _b_ of the input buffer.
+// 4. We window the current block, fire the callback for processing, window
+//    again, and overlap/add to the output buffer.
+// 5. We copy sections _a_ and _b_ of the output buffer into output.
+// 6. For both the input and the output buffers, we copy section _c_ into
+//    section _a_.
+// 7. We set the new frame_offset to be the difference between the first frame
+//    of `bl` and the border between sections _b_ and _c_.
+//
+// When block_size > chunk_size the input and output buffers look like this:
+//
+//                   chunk_size               delay*       chunk_size + delay*
+//  buffer: <-------------|---------------------|---------------|>
+//                _a_              _b_                 _c_
+//
+// On each call to ProcessChunk():
+// The procedure is the same as above, except for:
+// 1. New input gets read into section _c_ of the input buffer.
+// 3. We block until we reach a block `bl` that doesn't contain any frames
+//    from section _a_ of the input buffer.
+// 5. We copy section _a_ of the output buffer into output.
+// 6. For both the input and the output buffers, we copy sections _b_ and _c_
+//    into section _a_ and _b_.
+// 7. We set the new frame_offset to be the difference between the first frame
+//    of `bl` and the border between sections _a_ and _b_.
+//
+// * delay here refers to inintial_delay_
+//
+// TODO(claguna): Look at using ring buffers to eliminate some copies.
+void Blocker::ProcessChunk(const float* const* input,
+                           size_t chunk_size,
+                           size_t num_input_channels,
+                           size_t num_output_channels,
+                           float* const* output) {
+  RTC_CHECK_EQ(chunk_size, chunk_size_);
+  RTC_CHECK_EQ(num_input_channels, num_input_channels_);
+  RTC_CHECK_EQ(num_output_channels, num_output_channels_);
+
+  input_buffer_.Write(input, num_input_channels, chunk_size_);
+  size_t first_frame_in_block = frame_offset_;
+
+  // Loop through blocks.
+  while (first_frame_in_block < chunk_size_) {
+    input_buffer_.Read(input_block_.channels(), num_input_channels,
+                       block_size_);
+    input_buffer_.MoveReadPositionBackward(block_size_ - shift_amount_);
+
+    ApplyWindow(window_.get(), block_size_, num_input_channels_,
+                input_block_.channels());
+    callback_->ProcessBlock(input_block_.channels(), block_size_,
+                            num_input_channels_, num_output_channels_,
+                            output_block_.channels());
+    ApplyWindow(window_.get(), block_size_, num_output_channels_,
+                output_block_.channels());
+
+    AddFrames(output_buffer_.channels(), first_frame_in_block,
+              output_block_.channels(), 0, block_size_, num_output_channels_,
+              output_buffer_.channels(), first_frame_in_block);
+
+    first_frame_in_block += shift_amount_;
+  }
+
+  // Copy output buffer to output
+  CopyFrames(output_buffer_.channels(), 0, chunk_size_, num_output_channels_,
+             output, 0);
+
+  // Copy output buffer [chunk_size_, chunk_size_ + initial_delay]
+  // to output buffer [0, initial_delay], zero the rest.
+  MoveFrames(output_buffer_.channels(), chunk_size, initial_delay_,
+             num_output_channels_, output_buffer_.channels(), 0);
+  ZeroOut(output_buffer_.channels(), initial_delay_, chunk_size_,
+          num_output_channels_);
+
+  // Calculate new starting frames.
+  frame_offset_ = first_frame_in_block - chunk_size_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h
new file mode 100644
index 0000000000..59b7e29621
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
+
+#include <memory>
+
+#include "common_audio/channel_buffer.h"
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+namespace webrtc {
+
+// The callback function to process audio in the time domain. Input has already
+// been windowed, and output will be windowed. The number of input channels
+// must be >= the number of output channels.
+class BlockerCallback {
+ public:
+  virtual ~BlockerCallback() {}
+
+  virtual void ProcessBlock(const float* const* input,
+                            size_t num_frames,
+                            size_t num_input_channels,
+                            size_t num_output_channels,
+                            float* const* output) = 0;
+};
+
+// The main purpose of Blocker is to abstract away the fact that often we
+// receive a different number of audio frames than our transform takes. For
+// example, most FFTs work best when the fft-size is a power of 2, but suppose
+// we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames
+// of audio, which is not a power of 2. Blocker allows us to specify the
+// transform and all other necessary processing via the Process() callback
+// function without any constraints on the transform-size
+// (read: `block_size_`) or received-audio-size (read: `chunk_size_`).
+// We handle this for the multichannel audio case, allowing for different
+// numbers of input and output channels (for example, beamforming takes 2 or
+// more input channels and returns 1 output channel). Audio signals are
+// represented as deinterleaved floats in the range [-1, 1].
+//
+// Blocker is responsible for:
+// - blocking audio while handling potential discontinuities on the edges
+//   of chunks
+// - windowing blocks before sending them to Process()
+// - windowing processed blocks, and overlap-adding them together before
+//   sending back a processed chunk
+//
+// To use blocker:
+// 1. Impelment a BlockerCallback object `bc`.
+// 2. Instantiate a Blocker object `b`, passing in `bc`.
+// 3. As you receive audio, call b.ProcessChunk() to get processed audio.
+//
+// A small amount of delay is added to the first received chunk to deal with
+// the difference in chunk/block sizes. This delay is <= chunk_size.
+//
+// Ownership of window is retained by the caller.  That is, Blocker makes a
+// copy of window and does not attempt to delete it.
+class Blocker {
+ public:
+  Blocker(size_t chunk_size,
+          size_t block_size,
+          size_t num_input_channels,
+          size_t num_output_channels,
+          const float* window,
+          size_t shift_amount,
+          BlockerCallback* callback);
+  ~Blocker();
+
+  void ProcessChunk(const float* const* input,
+                    size_t chunk_size,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output);
+
+  size_t initial_delay() const { return initial_delay_; }
+
+ private:
+  const size_t chunk_size_;
+  const size_t block_size_;
+  const size_t num_input_channels_;
+  const size_t num_output_channels_;
+
+  // The number of frames of delay to add at the beginning of the first chunk.
+  const size_t initial_delay_;
+
+  // The frame index into the input buffer where the first block should be read
+  // from. This is necessary because shift_amount_ is not necessarily a
+  // multiple of chunk_size_, so blocks won't line up at the start of the
+  // buffer.
+  size_t frame_offset_;
+
+  // Since blocks nearly always overlap, there are certain blocks that require
+  // frames from the end of one chunk and the beginning of the next chunk. The
+  // input and output buffers are responsible for saving those frames between
+  // calls to ProcessChunk().
+  //
+  // Both contain |initial delay| + `chunk_size` frames. The input is a fairly
+  // standard FIFO, but due to the overlap-add it's harder to use an
+  // AudioRingBuffer for the output.
+  AudioRingBuffer input_buffer_;
+  ChannelBuffer<float> output_buffer_;
+
+  // Space for the input block (can't wrap because of windowing).
+  ChannelBuffer<float> input_block_;
+
+  // Space for the output block (can't wrap because of overlap/add).
+  ChannelBuffer<float> output_block_;
+
+  std::unique_ptr<float[]> window_;
+
+  // The amount of frames between the start of contiguous blocks. For example,
+  // `shift_amount_` = `block_size_` / 2 for a Hann window.
+  size_t shift_amount_;
+
+  BlockerCallback* callback_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
new file mode 100644
index 0000000000..9c8e789ba9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
@@ -0,0 +1,293 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+
+#include <memory>
+
+#include "rtc_base/arraysize.h"
+#include "test/gtest.h"
+
+namespace {
+
+// Callback Function to add 3 to every sample in the signal.
+class PlusThreeBlockerCallback : public webrtc::BlockerCallback {
+ public:
+  void ProcessBlock(const float* const* input,
+                    size_t num_frames,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output) override {
+    for (size_t i = 0; i < num_output_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        output[i][j] = input[i][j] + 3;
+      }
+    }
+  }
+};
+
+// No-op Callback Function.
+class CopyBlockerCallback : public webrtc::BlockerCallback {
+ public:
+  void ProcessBlock(const float* const* input,
+                    size_t num_frames,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output) override {
+    for (size_t i = 0; i < num_output_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        output[i][j] = input[i][j];
+      }
+    }
+  }
+};
+
+}  // namespace
+
+namespace webrtc {
+
+// Tests blocking with a window that multiplies the signal by 2, a callback
+// that adds 3 to each sample in the signal, and different combinations of chunk
+// size, block size, and shift amount.
+class BlockerTest : public ::testing::Test {
+ protected:
+  void RunTest(Blocker* blocker,
+               size_t chunk_size,
+               size_t num_frames,
+               const float* const* input,
+               float* const* input_chunk,
+               float* const* output,
+               float* const* output_chunk,
+               size_t num_input_channels,
+               size_t num_output_channels) {
+    size_t start = 0;
+    size_t end = chunk_size - 1;
+    while (end < num_frames) {
+      CopyTo(input_chunk, 0, start, num_input_channels, chunk_size, input);
+      blocker->ProcessChunk(input_chunk, chunk_size, num_input_channels,
+                            num_output_channels, output_chunk);
+      CopyTo(output, start, 0, num_output_channels, chunk_size, output_chunk);
+
+      start += chunk_size;
+      end += chunk_size;
+    }
+  }
+
+  void ValidateSignalEquality(const float* const* expected,
+                              const float* const* actual,
+                              size_t num_channels,
+                              size_t num_frames) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        EXPECT_FLOAT_EQ(expected[i][j], actual[i][j]);
+      }
+    }
+  }
+
+  void ValidateInitialDelay(const float* const* output,
+                            size_t num_channels,
+                            size_t num_frames,
+                            size_t initial_delay) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        if (j < initial_delay) {
+          EXPECT_FLOAT_EQ(output[i][j], 0.f);
+        } else {
+          EXPECT_GT(output[i][j], 0.f);
+        }
+      }
+    }
+  }
+
+  static void CopyTo(float* const* dst,
+                     size_t start_index_dst,
+                     size_t start_index_src,
+                     size_t num_channels,
+                     size_t num_frames,
+                     const float* const* src) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      memcpy(&dst[i][start_index_dst], &src[i][start_index_src],
+             num_frames * sizeof(float));
+    }
+  }
+};
+
+TEST_F(BlockerTest, TestBlockerMutuallyPrimeChunkandBlockSize) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 10;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 5;
+  const size_t kShiftAmount = 2;
+
+  const float kInput[kNumInputChannels][kNumFrames] = {
+      {1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+      {2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+      {3, 3, 3, 3, 3, 3, 3, 3, 3, 3}};
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput));
+
+  const float kExpectedOutput[kNumInputChannels][kNumFrames] = {
+      {6, 6, 12, 20, 20, 20, 20, 20, 20, 20},
+      {6, 6, 12, 28, 28, 28, 28, 28, 28, 28}};
+  ChannelBuffer<float> expected_output_cb(kNumFrames, kNumInputChannels);
+  expected_output_cb.SetDataForTesting(
+      kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput));
+
+  const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f};
+
+  ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels);
+  ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels);
+  ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
+
+  PlusThreeBlockerCallback callback;
+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
+                  kWindow, kShiftAmount, &callback);
+
+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
+          input_chunk_cb.channels(), actual_output_cb.channels(),
+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+  ValidateSignalEquality(expected_output_cb.channels(),
+                         actual_output_cb.channels(), kNumOutputChannels,
+                         kNumFrames);
+}
+
+TEST_F(BlockerTest, TestBlockerMutuallyPrimeShiftAndBlockSize) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 12;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 6;
+  const size_t kShiftAmount = 3;
+
+  const float kInput[kNumInputChannels][kNumFrames] = {
+      {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+      {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+      {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}};
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput));
+
+  const float kExpectedOutput[kNumOutputChannels][kNumFrames] = {
+      {6, 10, 10, 20, 10, 10, 20, 10, 10, 20, 10, 10},
+      {6, 14, 14, 28, 14, 14, 28, 14, 14, 28, 14, 14}};
+  ChannelBuffer<float> expected_output_cb(kNumFrames, kNumOutputChannels);
+  expected_output_cb.SetDataForTesting(
+      kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput));
+
+  const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f};
+
+  ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels);
+  ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels);
+  ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
+
+  PlusThreeBlockerCallback callback;
+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
+                  kWindow, kShiftAmount, &callback);
+
+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
+          input_chunk_cb.channels(), actual_output_cb.channels(),
+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+  ValidateSignalEquality(expected_output_cb.channels(),
+                         actual_output_cb.channels(), kNumOutputChannels,
+                         kNumFrames);
+}
+
+TEST_F(BlockerTest, TestBlockerNoOverlap) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 12;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 4;
+  const size_t kShiftAmount = 4;
+
+  const float kInput[kNumInputChannels][kNumFrames] = {
+      {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+      {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+      {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}};
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput));
+
+  const float kExpectedOutput[kNumOutputChannels][kNumFrames] = {
+      {10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10},
+      {14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14}};
+  ChannelBuffer<float> expected_output_cb(kNumFrames, kNumOutputChannels);
+  expected_output_cb.SetDataForTesting(
+      kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput));
+
+  const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f};
+
+  ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels);
+  ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels);
+  ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
+
+  PlusThreeBlockerCallback callback;
+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
+                  kWindow, kShiftAmount, &callback);
+
+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
+          input_chunk_cb.channels(), actual_output_cb.channels(),
+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+  ValidateSignalEquality(expected_output_cb.channels(),
+                         actual_output_cb.channels(), kNumOutputChannels,
+                         kNumFrames);
+}
+
+TEST_F(BlockerTest, InitialDelaysAreMinimum) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 1280;
+  const size_t kChunkSize[] = {80,  80,  80,  80,  80,  80,
+                               160, 160, 160, 160, 160, 160};
+  const size_t kBlockSize[] = {64,  64,  64,  128, 128, 128,
+                               128, 128, 128, 256, 256, 256};
+  const size_t kShiftAmount[] = {16, 32, 64,  32, 64,  128,
+                                 32, 64, 128, 64, 128, 256};
+  const size_t kInitialDelay[] = {48, 48, 48, 112, 112, 112,
+                                  96, 96, 96, 224, 224, 224};
+
+  float input[kNumInputChannels][kNumFrames];
+  for (size_t i = 0; i < kNumInputChannels; ++i) {
+    for (size_t j = 0; j < kNumFrames; ++j) {
+      input[i][j] = i + 1;
+    }
+  }
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(input[0], sizeof(input) / sizeof(**input));
+
+  ChannelBuffer<float> output_cb(kNumFrames, kNumOutputChannels);
+
+  CopyBlockerCallback callback;
+
+  for (size_t i = 0; i < arraysize(kChunkSize); ++i) {
+    std::unique_ptr<float[]> window(new float[kBlockSize[i]]);
+    for (size_t j = 0; j < kBlockSize[i]; ++j) {
+      window[j] = 1.f;
+    }
+
+    ChannelBuffer<float> input_chunk_cb(kChunkSize[i], kNumInputChannels);
+    ChannelBuffer<float> output_chunk_cb(kChunkSize[i], kNumOutputChannels);
+
+    Blocker blocker(kChunkSize[i], kBlockSize[i], kNumInputChannels,
+                    kNumOutputChannels, window.get(), kShiftAmount[i],
+                    &callback);
+
+    RunTest(&blocker, kChunkSize[i], kNumFrames, input_cb.channels(),
+            input_chunk_cb.channels(), output_cb.channels(),
+            output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+    ValidateInitialDelay(output_cb.channels(), kNumOutputChannels, kNumFrames,
+                         kInitialDelay[i]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc
new file mode 100644
index 0000000000..b1a6526bba
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
+
+#include <algorithm>
+#include <cstdlib>
+#include <cstring>
+
+#include "common_audio/real_fourier.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void LappedTransform::BlockThunk::ProcessBlock(const float* const* input,
+                                               size_t num_frames,
+                                               size_t num_input_channels,
+                                               size_t num_output_channels,
+                                               float* const* output) {
+  RTC_CHECK_EQ(num_input_channels, parent_->num_in_channels_);
+  RTC_CHECK_EQ(num_output_channels, parent_->num_out_channels_);
+  RTC_CHECK_EQ(parent_->block_length_, num_frames);
+
+  for (size_t i = 0; i < num_input_channels; ++i) {
+    memcpy(parent_->real_buf_.Row(i), input[i], num_frames * sizeof(*input[0]));
+    parent_->fft_->Forward(parent_->real_buf_.Row(i),
+                           parent_->cplx_pre_.Row(i));
+  }
+
+  size_t block_length =
+      RealFourier::ComplexLength(RealFourier::FftOrder(num_frames));
+  RTC_CHECK_EQ(parent_->cplx_length_, block_length);
+  parent_->block_processor_->ProcessAudioBlock(
+      parent_->cplx_pre_.Array(), num_input_channels, parent_->cplx_length_,
+      num_output_channels, parent_->cplx_post_.Array());
+
+  for (size_t i = 0; i < num_output_channels; ++i) {
+    parent_->fft_->Inverse(parent_->cplx_post_.Row(i),
+                           parent_->real_buf_.Row(i));
+    memcpy(output[i], parent_->real_buf_.Row(i),
+           num_frames * sizeof(*input[0]));
+  }
+}
+
+LappedTransform::LappedTransform(size_t num_in_channels,
+                                 size_t num_out_channels,
+                                 size_t chunk_length,
+                                 const float* window,
+                                 size_t block_length,
+                                 size_t shift_amount,
+                                 Callback* callback)
+    : blocker_callback_(this),
+      num_in_channels_(num_in_channels),
+      num_out_channels_(num_out_channels),
+      block_length_(block_length),
+      chunk_length_(chunk_length),
+      block_processor_(callback),
+      blocker_(chunk_length_,
+               block_length_,
+               num_in_channels_,
+               num_out_channels_,
+               window,
+               shift_amount,
+               &blocker_callback_),
+      fft_(RealFourier::Create(RealFourier::FftOrder(block_length_))),
+      cplx_length_(RealFourier::ComplexLength(fft_->order())),
+      real_buf_(num_in_channels,
+                block_length_,
+                RealFourier::kFftBufferAlignment),
+      cplx_pre_(num_in_channels,
+                cplx_length_,
+                RealFourier::kFftBufferAlignment),
+      cplx_post_(num_out_channels,
+                 cplx_length_,
+                 RealFourier::kFftBufferAlignment) {
+  RTC_CHECK(num_in_channels_ > 0);
+  RTC_CHECK_GT(block_length_, 0);
+  RTC_CHECK_GT(chunk_length_, 0);
+  RTC_CHECK(block_processor_);
+
+  // block_length_ power of 2?
+  RTC_CHECK_EQ(0, block_length_ & (block_length_ - 1));
+}
+
+LappedTransform::~LappedTransform() = default;
+
+void LappedTransform::ProcessChunk(const float* const* in_chunk,
+                                   float* const* out_chunk) {
+  blocker_.ProcessChunk(in_chunk, chunk_length_, num_in_channels_,
+                        num_out_channels_, out_chunk);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h
new file mode 100644
index 0000000000..bb25c34a9e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h
@@ -0,0 +1,175 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
+
+#include <complex>
+#include <memory>
+
+#include "common_audio/real_fourier.h"
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+#include "rtc_base/memory/aligned_malloc.h"
+
+namespace webrtc {
+
+// Wrapper class for aligned arrays. Every row (and the first dimension) are
+// aligned to the given byte alignment.
+template <typename T>
+class AlignedArray {
+ public:
+  AlignedArray(size_t rows, size_t cols, size_t alignment)
+      : rows_(rows), cols_(cols) {
+    RTC_CHECK_GT(alignment, 0);
+    head_row_ =
+        static_cast<T**>(AlignedMalloc(rows_ * sizeof(*head_row_), alignment));
+    for (size_t i = 0; i < rows_; ++i) {
+      head_row_[i] = static_cast<T*>(
+          AlignedMalloc(cols_ * sizeof(**head_row_), alignment));
+    }
+  }
+
+  ~AlignedArray() {
+    for (size_t i = 0; i < rows_; ++i) {
+      AlignedFree(head_row_[i]);
+    }
+    AlignedFree(head_row_);
+  }
+
+  T* const* Array() { return head_row_; }
+
+  const T* const* Array() const { return head_row_; }
+
+  T* Row(size_t row) {
+    RTC_CHECK_LE(row, rows_);
+    return head_row_[row];
+  }
+
+  const T* Row(size_t row) const {
+    RTC_CHECK_LE(row, rows_);
+    return head_row_[row];
+  }
+
+ private:
+  size_t rows_;
+  size_t cols_;
+  T** head_row_;
+};
+
+// Helper class for audio processing modules which operate on frequency domain
+// input derived from the windowed time domain audio stream.
+//
+// The input audio chunk is sliced into possibly overlapping blocks, multiplied
+// by a window and transformed with an FFT implementation. The transformed data
+// is supplied to the given callback for processing. The processed output is
+// then inverse transformed into the time domain and spliced back into a chunk
+// which constitutes the final output of this processing module.
+class LappedTransform {
+ public:
+  class Callback {
+   public:
+    virtual ~Callback() {}
+
+    virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
+                                   size_t num_in_channels,
+                                   size_t frames,
+                                   size_t num_out_channels,
+                                   std::complex<float>* const* out_block) = 0;
+  };
+
+  // Construct a transform instance. `chunk_length` is the number of samples in
+  // each channel. `window` defines the window, owned by the caller (a copy is
+  // made internally); `window` should have length equal to `block_length`.
+  // `block_length` defines the length of a block, in samples.
+  // `shift_amount` is in samples. `callback` is the caller-owned audio
+  // processing function called for each block of the input chunk.
+  LappedTransform(size_t num_in_channels,
+                  size_t num_out_channels,
+                  size_t chunk_length,
+                  const float* window,
+                  size_t block_length,
+                  size_t shift_amount,
+                  Callback* callback);
+  ~LappedTransform();
+
+  // Main audio processing helper method. Internally slices `in_chunk` into
+  // blocks, transforms them to frequency domain, calls the callback for each
+  // block and returns a de-blocked time domain chunk of audio through
+  // `out_chunk`. Both buffers are caller-owned.
+  void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
+
+  // Get the chunk length.
+  //
+  // The chunk length is the number of samples per channel that must be passed
+  // to ProcessChunk via the parameter in_chunk.
+  //
+  // Returns the same chunk_length passed to the LappedTransform constructor.
+  size_t chunk_length() const { return chunk_length_; }
+
+  // Get the number of input channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // in_chunk.
+  //
+  // Returns the same num_in_channels passed to the LappedTransform constructor.
+  size_t num_in_channels() const { return num_in_channels_; }
+
+  // Get the number of output channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // out_chunk.
+  //
+  // Returns the same num_out_channels passed to the LappedTransform
+  // constructor.
+  size_t num_out_channels() const { return num_out_channels_; }
+
+  // Returns the initial delay.
+  //
+  // This is the delay introduced by the `blocker_` to be able to get and return
+  // chunks of `chunk_length`, but process blocks of `block_length`.
+  size_t initial_delay() const { return blocker_.initial_delay(); }
+
+ private:
+  // Internal middleware callback, given to the blocker. Transforms each block
+  // and hands it over to the processing method given at construction time.
+  class BlockThunk : public BlockerCallback {
+   public:
+    explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
+
+    void ProcessBlock(const float* const* input,
+                      size_t num_frames,
+                      size_t num_input_channels,
+                      size_t num_output_channels,
+                      float* const* output) override;
+
+   private:
+    LappedTransform* const parent_;
+  } blocker_callback_;
+
+  const size_t num_in_channels_;
+  const size_t num_out_channels_;
+
+  const size_t block_length_;
+  const size_t chunk_length_;
+
+  Callback* const block_processor_;
+  Blocker blocker_;
+
+  // TODO(alessiob): Replace RealFourier with a different FFT library.
+  std::unique_ptr<RealFourier> fft_;
+  const size_t cplx_length_;
+  AlignedArray<float> real_buf_;
+  AlignedArray<std::complex<float> > cplx_pre_;
+  AlignedArray<std::complex<float> > cplx_post_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
new file mode 100644
index 0000000000..1003ed52e5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstring>
+
+#include "test/gtest.h"
+
+using std::complex;
+
+namespace {
+
+class NoopCallback : public webrtc::LappedTransform::Callback {
+ public:
+  NoopCallback() : block_num_(0) {}
+
+  void ProcessAudioBlock(const complex<float>* const* in_block,
+                         size_t in_channels,
+                         size_t frames,
+                         size_t out_channels,
+                         complex<float>* const* out_block) override {
+    RTC_CHECK_EQ(in_channels, out_channels);
+    for (size_t i = 0; i < out_channels; ++i) {
+      memcpy(out_block[i], in_block[i], sizeof(**in_block) * frames);
+    }
+    ++block_num_;
+  }
+
+  size_t block_num() { return block_num_; }
+
+ private:
+  size_t block_num_;
+};
+
+class FftCheckerCallback : public webrtc::LappedTransform::Callback {
+ public:
+  FftCheckerCallback() : block_num_(0) {}
+
+  void ProcessAudioBlock(const complex<float>* const* in_block,
+                         size_t in_channels,
+                         size_t frames,
+                         size_t out_channels,
+                         complex<float>* const* out_block) override {
+    RTC_CHECK_EQ(in_channels, out_channels);
+
+    size_t full_length = (frames - 1) * 2;
+    ++block_num_;
+
+    if (block_num_ > 0) {
+      ASSERT_NEAR(in_block[0][0].real(), static_cast<float>(full_length),
+                  1e-5f);
+      ASSERT_NEAR(in_block[0][0].imag(), 0.0f, 1e-5f);
+      for (size_t i = 1; i < frames; ++i) {
+        ASSERT_NEAR(in_block[0][i].real(), 0.0f, 1e-5f);
+        ASSERT_NEAR(in_block[0][i].imag(), 0.0f, 1e-5f);
+      }
+    }
+  }
+
+  size_t block_num() { return block_num_; }
+
+ private:
+  size_t block_num_;
+};
+
+void SetFloatArray(float value, int rows, int cols, float* const* array) {
+  for (int i = 0; i < rows; ++i) {
+    for (int j = 0; j < cols; ++j) {
+      array[i][j] = value;
+    }
+  }
+}
+
+}  // namespace
+
+namespace webrtc {
+
+TEST(LappedTransformTest, Windowless) {
+  const size_t kChannels = 3;
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  const size_t kShiftAmount = 64;
+  NoopCallback noop;
+
+  // Rectangular window.
+  float window[kBlockLength];
+  std::fill(window, &window[kBlockLength], 1.0f);
+
+  LappedTransform trans(kChannels, kChannels, kChunkLength, window,
+                        kBlockLength, kShiftAmount, &noop);
+  float in_buffer[kChannels][kChunkLength];
+  float* in_chunk[kChannels];
+  float out_buffer[kChannels][kChunkLength];
+  float* out_chunk[kChannels];
+
+  in_chunk[0] = in_buffer[0];
+  in_chunk[1] = in_buffer[1];
+  in_chunk[2] = in_buffer[2];
+  out_chunk[0] = out_buffer[0];
+  out_chunk[1] = out_buffer[1];
+  out_chunk[2] = out_buffer[2];
+  SetFloatArray(2.0f, kChannels, kChunkLength, in_chunk);
+  SetFloatArray(-1.0f, kChannels, kChunkLength, out_chunk);
+
+  trans.ProcessChunk(in_chunk, out_chunk);
+
+  for (size_t i = 0; i < kChannels; ++i) {
+    for (size_t j = 0; j < kChunkLength; ++j) {
+      ASSERT_NEAR(out_chunk[i][j], 2.0f, 1e-5f);
+    }
+  }
+
+  ASSERT_EQ(kChunkLength / kBlockLength, noop.block_num());
+}
+
+TEST(LappedTransformTest, IdentityProcessor) {
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  const size_t kShiftAmount = 32;
+  NoopCallback noop;
+
+  // Identity window for |overlap = block_size / 2|.
+  float window[kBlockLength];
+  std::fill(window, &window[kBlockLength], std::sqrt(0.5f));
+
+  LappedTransform trans(1, 1, kChunkLength, window, kBlockLength, kShiftAmount,
+                        &noop);
+  float in_buffer[kChunkLength];
+  float* in_chunk = in_buffer;
+  float out_buffer[kChunkLength];
+  float* out_chunk = out_buffer;
+
+  SetFloatArray(2.0f, 1, kChunkLength, &in_chunk);
+  SetFloatArray(-1.0f, 1, kChunkLength, &out_chunk);
+
+  trans.ProcessChunk(&in_chunk, &out_chunk);
+
+  for (size_t i = 0; i < kChunkLength; ++i) {
+    ASSERT_NEAR(out_chunk[i], (i < kBlockLength - kShiftAmount) ? 0.0f : 2.0f,
+                1e-5f);
+  }
+
+  ASSERT_EQ(kChunkLength / kShiftAmount, noop.block_num());
+}
+
+TEST(LappedTransformTest, Callbacks) {
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  FftCheckerCallback call;
+
+  // Rectangular window.
+  float window[kBlockLength];
+  std::fill(window, &window[kBlockLength], 1.0f);
+
+  LappedTransform trans(1, 1, kChunkLength, window, kBlockLength, kBlockLength,
+                        &call);
+  float in_buffer[kChunkLength];
+  float* in_chunk = in_buffer;
+  float out_buffer[kChunkLength];
+  float* out_chunk = out_buffer;
+
+  SetFloatArray(1.0f, 1, kChunkLength, &in_chunk);
+  SetFloatArray(-1.0f, 1, kChunkLength, &out_chunk);
+
+  trans.ProcessChunk(&in_chunk, &out_chunk);
+
+  ASSERT_EQ(kChunkLength / kBlockLength, call.block_num());
+}
+
+TEST(LappedTransformTest, chunk_length) {
+  const size_t kBlockLength = 64;
+  FftCheckerCallback call;
+  const float window[kBlockLength] = {};
+
+  // Make sure that chunk_length returns the same value passed to the
+  // LappedTransform constructor.
+  {
+    const size_t kExpectedChunkLength = 512;
+    const LappedTransform trans(1, 1, kExpectedChunkLength, window,
+                                kBlockLength, kBlockLength, &call);
+
+    EXPECT_EQ(kExpectedChunkLength, trans.chunk_length());
+  }
+  {
+    const size_t kExpectedChunkLength = 160;
+    const LappedTransform trans(1, 1, kExpectedChunkLength, window,
+                                kBlockLength, kBlockLength, &call);
+
+    EXPECT_EQ(kExpectedChunkLength, trans.chunk_length());
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
new file mode 100644
index 0000000000..7761efe8b3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels)
+    : sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) {
+  RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+             sample_rate_hz == 32000 || sample_rate_hz == 48000)
+      << "Unsupported sample rate " << sample_rate_hz;
+  RTC_DCHECK_GE(num_channels, 1);
+}
+
+void AudioDecoderPcm16B::Reset() {}
+
+int AudioDecoderPcm16B::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioDecoderPcm16B::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
+                                       size_t encoded_len,
+                                       int sample_rate_hz,
+                                       int16_t* decoded,
+                                       SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted =
+      PacketDuration(encoded, encoded_len) * 2 *
+      Channels();  // 2 bytes per sample per channel
+  size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len_adjusted, decoded);
+  *speech_type = ConvertSpeechType(1);
+  return static_cast<int>(ret);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz_, 1000);
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, samples_per_ms * 2 * num_channels_,
+      samples_per_ms);
+}
+
+int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
+                                       size_t encoded_len) const {
+  // Two encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / (2 * Channels()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
new file mode 100644
index 0000000000..6f50161d3f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoderPcm16B final : public AudioDecoder {
+ public:
+  AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels);
+
+  AudioDecoderPcm16B(const AudioDecoderPcm16B&) = delete;
+  AudioDecoderPcm16B& operator=(const AudioDecoderPcm16B&) = delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const int sample_rate_hz_;
+  const size_t num_channels_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
new file mode 100644
index 0000000000..9445b1ee3e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
+                                      size_t input_len,
+                                      uint8_t* encoded) {
+  return WebRtcPcm16b_Encode(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcm16B::BytesPerSample() const {
+  return 2;
+}
+
+AudioEncoder::CodecType AudioEncoderPcm16B::GetCodecType() const {
+  return CodecType::kOther;
+}
+
+bool AudioEncoderPcm16B::Config::IsOk() const {
+  if ((sample_rate_hz != 8000) && (sample_rate_hz != 16000) &&
+      (sample_rate_hz != 32000) && (sample_rate_hz != 48000))
+    return false;
+  return AudioEncoderPcm::Config::IsOk();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
new file mode 100644
index 0000000000..c363b40b3f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
+
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+
+namespace webrtc {
+
+class AudioEncoderPcm16B final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+   public:
+    Config() : AudioEncoderPcm::Config(107), sample_rate_hz(8000) {}
+    bool IsOk() const;
+
+    int sample_rate_hz;
+  };
+
+  explicit AudioEncoderPcm16B(const Config& config)
+      : AudioEncoderPcm(config, config.sample_rate_hz) {}
+
+  AudioEncoderPcm16B(const AudioEncoderPcm16B&) = delete;
+  AudioEncoderPcm16B& operator=(const AudioEncoderPcm16B&) = delete;
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
new file mode 100644
index 0000000000..2f6dce5f41
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+
+size_t WebRtcPcm16b_Encode(const int16_t* speech,
+                           size_t len,
+                           uint8_t* encoded) {
+  size_t i;
+  for (i = 0; i < len; ++i) {
+    uint16_t s = speech[i];
+    encoded[2 * i] = s >> 8;
+    encoded[2 * i + 1] = s;
+  }
+  return 2 * len;
+}
+
+size_t WebRtcPcm16b_Decode(const uint8_t* encoded,
+                           size_t len,
+                           int16_t* speech) {
+  size_t i;
+  for (i = 0; i < len / 2; ++i)
+    speech[i] = encoded[2 * i] << 8 | encoded[2 * i + 1];
+  return len / 2;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
new file mode 100644
index 0000000000..75d1efda3b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
+/*
+ * Define the fixpoint numeric formats
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcPcm16b_Encode(...)
+ *
+ * "Encode" a sample vector to 16 bit linear (Encoded standard is big endian)
+ *
+ * Input:
+ *              - speech        : Input speech vector
+ *              - len           : Number of samples in speech vector
+ *
+ * Output:
+ *              - encoded       : Encoded data vector (big endian 16 bit)
+ *
+ * Returned value               : Length (in bytes) of coded data.
+ *                                Always equal to twice the len input parameter.
+ */
+
+size_t WebRtcPcm16b_Encode(const int16_t* speech, size_t len, uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcPcm16b_Decode(...)
+ *
+ * "Decode" a vector to 16 bit linear (Encoded standard is big endian)
+ *
+ * Input:
+ *              - encoded       : Encoded data vector (big endian 16 bit)
+ *              - len           : Number of bytes in encoded
+ *
+ * Output:
+ *              - speech        : Decoded speech vector
+ *
+ * Returned value               : Samples in speech
+ */
+
+size_t WebRtcPcm16b_Decode(const uint8_t* encoded, size_t len, int16_t* speech);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
new file mode 100644
index 0000000000..ecf91b45ac
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+
+#include <stdint.h>
+
+#include <initializer_list>
+
+namespace webrtc {
+
+void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs) {
+  for (uint8_t num_channels : {1, 2}) {
+    for (int sample_rate_hz : {8000, 16000, 32000}) {
+      specs->push_back(
+          {{"L16", sample_rate_hz, num_channels},
+           {sample_rate_hz, num_channels, sample_rate_hz * num_channels * 16}});
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
new file mode 100644
index 0000000000..3fae717ff3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
+
+#include <vector>
+
+#include "api/audio_codecs/audio_format.h"
+
+namespace webrtc {
+void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs);
+}
+
+#endif  // MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
new file mode 100644
index 0000000000..724bba52d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -0,0 +1,272 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+
+#include <string.h>
+
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/byte_order.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+static constexpr const int kRedMaxPacketSize =
+    1 << 10;  // RED packets must be less than 1024 bytes to fit the 10 bit
+              // block length.
+static constexpr const size_t kRedMaxTimestampDelta =
+    1 << 14;  // RED packets can encode a timestamp delta of 14 bits.
+static constexpr const size_t kAudioMaxRtpPacketLen =
+    1200;  // The typical MTU is 1200 bytes.
+
+static constexpr size_t kRedHeaderLength = 4;  // 4 bytes RED header.
+static constexpr size_t kRedLastHeaderLength =
+    1;  // reduced size for last RED header.
+
+static constexpr size_t kRedNumberOfRedundantEncodings =
+    1;  // The level of redundancy we support.
+
+AudioEncoderCopyRed::Config::Config() = default;
+AudioEncoderCopyRed::Config::Config(Config&&) = default;
+AudioEncoderCopyRed::Config::~Config() = default;
+
+size_t GetMaxRedundancyFromFieldTrial(const FieldTrialsView& field_trials) {
+  const std::string red_trial =
+      field_trials.Lookup("WebRTC-Audio-Red-For-Opus");
+  size_t redundancy = 0;
+  if (sscanf(red_trial.c_str(), "Enabled-%zu", &redundancy) != 1 ||
+      redundancy > 9) {
+    return kRedNumberOfRedundantEncodings;
+  }
+  return redundancy;
+}
+
+AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config,
+                                         const FieldTrialsView& field_trials)
+    : speech_encoder_(std::move(config.speech_encoder)),
+      primary_encoded_(0, kAudioMaxRtpPacketLen),
+      max_packet_length_(kAudioMaxRtpPacketLen),
+      red_payload_type_(config.payload_type) {
+  RTC_CHECK(speech_encoder_) << "Speech encoder not provided.";
+
+  auto number_of_redundant_encodings =
+      GetMaxRedundancyFromFieldTrial(field_trials);
+  for (size_t i = 0; i < number_of_redundant_encodings; i++) {
+    std::pair<EncodedInfo, rtc::Buffer> redundant;
+    redundant.second.EnsureCapacity(kAudioMaxRtpPacketLen);
+    redundant_encodings_.push_front(std::move(redundant));
+  }
+}
+
+AudioEncoderCopyRed::~AudioEncoderCopyRed() = default;
+
+int AudioEncoderCopyRed::SampleRateHz() const {
+  return speech_encoder_->SampleRateHz();
+}
+
+size_t AudioEncoderCopyRed::NumChannels() const {
+  return speech_encoder_->NumChannels();
+}
+
+int AudioEncoderCopyRed::RtpTimestampRateHz() const {
+  return speech_encoder_->RtpTimestampRateHz();
+}
+
+size_t AudioEncoderCopyRed::Num10MsFramesInNextPacket() const {
+  return speech_encoder_->Num10MsFramesInNextPacket();
+}
+
+size_t AudioEncoderCopyRed::Max10MsFramesInAPacket() const {
+  return speech_encoder_->Max10MsFramesInAPacket();
+}
+
+int AudioEncoderCopyRed::GetTargetBitrate() const {
+  return speech_encoder_->GetTargetBitrate();
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  primary_encoded_.Clear();
+  EncodedInfo info =
+      speech_encoder_->Encode(rtp_timestamp, audio, &primary_encoded_);
+  RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
+  RTC_DCHECK_EQ(primary_encoded_.size(), info.encoded_bytes);
+
+  if (info.encoded_bytes == 0 || info.encoded_bytes >= kRedMaxPacketSize) {
+    return info;
+  }
+  RTC_DCHECK_GT(max_packet_length_, info.encoded_bytes);
+
+  size_t header_length_bytes = kRedLastHeaderLength;
+  size_t bytes_available = max_packet_length_ - info.encoded_bytes;
+  auto it = redundant_encodings_.begin();
+
+  // Determine how much redundancy we can fit into our packet by
+  // iterating forward. This is determined both by the length as well
+  // as the timestamp difference. The latter can occur with opus DTX which
+  // has timestamp gaps of 400ms which exceeds REDs timestamp delta field size.
+  for (; it != redundant_encodings_.end(); it++) {
+    if (bytes_available < kRedHeaderLength + it->first.encoded_bytes) {
+      break;
+    }
+    if (it->first.encoded_bytes == 0) {
+      break;
+    }
+    if (rtp_timestamp - it->first.encoded_timestamp >= kRedMaxTimestampDelta) {
+      break;
+    }
+    bytes_available -= kRedHeaderLength + it->first.encoded_bytes;
+    header_length_bytes += kRedHeaderLength;
+  }
+
+  // Allocate room for RFC 2198 header.
+  encoded->SetSize(header_length_bytes);
+
+  // Iterate backwards and append the data.
+  size_t header_offset = 0;
+  while (it-- != redundant_encodings_.begin()) {
+    encoded->AppendData(it->second);
+
+    const uint32_t timestamp_delta =
+        info.encoded_timestamp - it->first.encoded_timestamp;
+    encoded->data()[header_offset] = it->first.payload_type | 0x80;
+    rtc::SetBE16(static_cast<uint8_t*>(encoded->data()) + header_offset + 1,
+                 (timestamp_delta << 2) | (it->first.encoded_bytes >> 8));
+    encoded->data()[header_offset + 3] = it->first.encoded_bytes & 0xff;
+    header_offset += kRedHeaderLength;
+    info.redundant.push_back(it->first);
+  }
+
+  // `info` will be implicitly cast to an EncodedInfoLeaf struct, effectively
+  // discarding the (empty) vector of redundant information. This is
+  // intentional.
+  if (header_length_bytes > kRedHeaderLength) {
+    info.redundant.push_back(info);
+    RTC_DCHECK_EQ(info.speech,
+                  info.redundant[info.redundant.size() - 1].speech);
+  }
+
+  encoded->AppendData(primary_encoded_);
+  RTC_DCHECK_EQ(header_offset, header_length_bytes - 1);
+  encoded->data()[header_offset] = info.payload_type;
+
+  // Shift the redundant encodings.
+  auto rit = redundant_encodings_.rbegin();
+  for (auto next = std::next(rit); next != redundant_encodings_.rend();
+       rit++, next = std::next(rit)) {
+    rit->first = next->first;
+    rit->second.SetData(next->second);
+  }
+  it = redundant_encodings_.begin();
+  if (it != redundant_encodings_.end()) {
+    it->first = info;
+    it->second.SetData(primary_encoded_);
+  }
+
+  // Update main EncodedInfo.
+  info.payload_type = red_payload_type_;
+  info.encoded_bytes = encoded->size();
+  return info;
+}
+
+void AudioEncoderCopyRed::Reset() {
+  speech_encoder_->Reset();
+  auto number_of_redundant_encodings = redundant_encodings_.size();
+  redundant_encodings_.clear();
+  for (size_t i = 0; i < number_of_redundant_encodings; i++) {
+    std::pair<EncodedInfo, rtc::Buffer> redundant;
+    redundant.second.EnsureCapacity(kAudioMaxRtpPacketLen);
+    redundant_encodings_.push_front(std::move(redundant));
+  }
+}
+
+bool AudioEncoderCopyRed::SetFec(bool enable) {
+  return speech_encoder_->SetFec(enable);
+}
+
+bool AudioEncoderCopyRed::SetDtx(bool enable) {
+  return speech_encoder_->SetDtx(enable);
+}
+
+bool AudioEncoderCopyRed::GetDtx() const {
+  return speech_encoder_->GetDtx();
+}
+
+bool AudioEncoderCopyRed::SetApplication(Application application) {
+  return speech_encoder_->SetApplication(application);
+}
+
+void AudioEncoderCopyRed::SetMaxPlaybackRate(int frequency_hz) {
+  speech_encoder_->SetMaxPlaybackRate(frequency_hz);
+}
+
+bool AudioEncoderCopyRed::EnableAudioNetworkAdaptor(
+    const std::string& config_string,
+    RtcEventLog* event_log) {
+  return speech_encoder_->EnableAudioNetworkAdaptor(config_string, event_log);
+}
+
+void AudioEncoderCopyRed::DisableAudioNetworkAdaptor() {
+  speech_encoder_->DisableAudioNetworkAdaptor();
+}
+
+void AudioEncoderCopyRed::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  speech_encoder_->OnReceivedUplinkPacketLossFraction(
+      uplink_packet_loss_fraction);
+}
+
+void AudioEncoderCopyRed::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {
+  speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
+                                             bwe_period_ms);
+}
+
+void AudioEncoderCopyRed::OnReceivedUplinkAllocation(
+    BitrateAllocationUpdate update) {
+  speech_encoder_->OnReceivedUplinkAllocation(update);
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderCopyRed::GetFrameLengthRange() const {
+  return speech_encoder_->GetFrameLengthRange();
+}
+
+void AudioEncoderCopyRed::OnReceivedRtt(int rtt_ms) {
+  speech_encoder_->OnReceivedRtt(rtt_ms);
+}
+
+void AudioEncoderCopyRed::OnReceivedOverhead(size_t overhead_bytes_per_packet) {
+  max_packet_length_ = kAudioMaxRtpPacketLen - overhead_bytes_per_packet;
+  return speech_encoder_->OnReceivedOverhead(overhead_bytes_per_packet);
+}
+
+void AudioEncoderCopyRed::SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                                      int max_frame_length_ms) {
+  return speech_encoder_->SetReceiverFrameLengthRange(min_frame_length_ms,
+                                                      max_frame_length_ms);
+}
+
+ANAStats AudioEncoderCopyRed::GetANAStats() const {
+  return speech_encoder_->GetANAStats();
+}
+
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoderCopyRed::ReclaimContainedEncoders() {
+  return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
new file mode 100644
index 0000000000..359b5eaa17
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
+#define MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/field_trials_view.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+// This class implements redundant audio coding as described in
+//   https://tools.ietf.org/html/rfc2198
+// The class object will have an underlying AudioEncoder object that performs
+// the actual encodings. The current class will gather the N latest encodings
+// from the underlying codec into one packet. Currently N is hard-coded to 2.
+
+class AudioEncoderCopyRed final : public AudioEncoder {
+ public:
+  struct Config {
+    Config();
+    Config(Config&&);
+    ~Config();
+    int payload_type;
+    std::unique_ptr<AudioEncoder> speech_encoder;
+  };
+
+  AudioEncoderCopyRed(Config&& config, const FieldTrialsView& field_trials);
+
+  ~AudioEncoderCopyRed() override;
+
+  AudioEncoderCopyRed(const AudioEncoderCopyRed&) = delete;
+  AudioEncoderCopyRed& operator=(const AudioEncoderCopyRed&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  bool SetFec(bool enable) override;
+
+  bool SetDtx(bool enable) override;
+  bool GetDtx() const override;
+
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                 RtcEventLog* event_log) override;
+  void DisableAudioNetworkAdaptor() override;
+  void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) override;
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms) override;
+  void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
+  void OnReceivedRtt(int rtt_ms) override;
+  void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
+  void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                   int max_frame_length_ms) override;
+  ANAStats GetANAStats() const override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
+      override;
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  std::unique_ptr<AudioEncoder> speech_encoder_;
+  rtc::Buffer primary_encoded_;
+  size_t max_packet_length_;
+  int red_payload_type_;
+  std::list<std::pair<EncodedInfo, rtc::Buffer>> redundant_encodings_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
new file mode 100644
index 0000000000..795a996624
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -0,0 +1,641 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+
+#include <memory>
+#include <vector>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/scoped_key_value_config.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+using ::testing::_;
+using ::testing::Eq;
+using ::testing::InSequence;
+using ::testing::Invoke;
+using ::testing::MockFunction;
+using ::testing::Not;
+using ::testing::Optional;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+namespace webrtc {
+
+namespace {
+static const size_t kMaxNumSamples = 48 * 10 * 2;  // 10 ms @ 48 kHz stereo.
+static const size_t kRedLastHeaderLength =
+    1;  // 1 byte RED header for the last element.
+}
+
+class AudioEncoderCopyRedTest : public ::testing::Test {
+ protected:
+  AudioEncoderCopyRedTest()
+      : mock_encoder_(new MockAudioEncoder),
+        timestamp_(4711),
+        sample_rate_hz_(16000),
+        num_audio_samples_10ms(sample_rate_hz_ / 100),
+        red_payload_type_(200) {
+    AudioEncoderCopyRed::Config config;
+    config.payload_type = red_payload_type_;
+    config.speech_encoder = std::unique_ptr<AudioEncoder>(mock_encoder_);
+    red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials_));
+    memset(audio_, 0, sizeof(audio_));
+    EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(1U));
+    EXPECT_CALL(*mock_encoder_, SampleRateHz())
+        .WillRepeatedly(Return(sample_rate_hz_));
+  }
+
+  void TearDown() override { red_.reset(); }
+
+  void Encode() {
+    ASSERT_TRUE(red_.get() != NULL);
+    encoded_.Clear();
+    encoded_info_ = red_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
+        &encoded_);
+    timestamp_ += rtc::checked_cast<uint32_t>(num_audio_samples_10ms);
+  }
+
+  test::ScopedKeyValueConfig field_trials_;
+  MockAudioEncoder* mock_encoder_;
+  std::unique_ptr<AudioEncoderCopyRed> red_;
+  uint32_t timestamp_;
+  int16_t audio_[kMaxNumSamples];
+  const int sample_rate_hz_;
+  size_t num_audio_samples_10ms;
+  rtc::Buffer encoded_;
+  AudioEncoder::EncodedInfo encoded_info_;
+  const int red_payload_type_;
+};
+
+TEST_F(AudioEncoderCopyRedTest, CreateAndDestroy) {}
+
+TEST_F(AudioEncoderCopyRedTest, CheckSampleRatePropagation) {
+  EXPECT_CALL(*mock_encoder_, SampleRateHz()).WillOnce(Return(17));
+  EXPECT_EQ(17, red_->SampleRateHz());
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckNumChannelsPropagation) {
+  EXPECT_CALL(*mock_encoder_, NumChannels()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->NumChannels());
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckFrameSizePropagation) {
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->Num10MsFramesInNextPacket());
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckMaxFrameSizePropagation) {
+  EXPECT_CALL(*mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->Max10MsFramesInAPacket());
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) {
+  EXPECT_CALL(*mock_encoder_,
+              OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
+  red_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) {
+  EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5));
+  red_->OnReceivedUplinkPacketLossFraction(0.5);
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckGetFrameLengthRangePropagation) {
+  auto expected_range =
+      std::make_pair(TimeDelta::Millis(20), TimeDelta::Millis(20));
+  EXPECT_CALL(*mock_encoder_, GetFrameLengthRange())
+      .WillRepeatedly(Return(absl::make_optional(expected_range)));
+  EXPECT_THAT(red_->GetFrameLengthRange(), Optional(Eq(expected_range)));
+}
+
+// Checks that the an Encode() call is immediately propagated to the speech
+// encoder.
+TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) {
+  // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction
+  // check ensures that exactly one call to EncodeImpl happens in each
+  // Encode call.
+  InSequence s;
+  MockFunction<void(int check_point_id)> check;
+  for (int i = 1; i <= 6; ++i) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillRepeatedly(Return(AudioEncoder::EncodedInfo()));
+    EXPECT_CALL(check, Call(i));
+    Encode();
+    check.Call(i);
+  }
+}
+
+// Checks that no output is produced if the underlying codec doesn't emit any
+// new data, even if the RED codec is loaded with a secondary encoding.
+TEST_F(AudioEncoderCopyRedTest, CheckNoOutput) {
+  static const size_t kEncodedSize = 17;
+  static const size_t kHeaderLenBytes = 5;
+  {
+    InSequence s;
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(kEncodedSize)))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(0)))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(kEncodedSize)));
+  }
+
+  // Start with one Encode() call that will produce output.
+  Encode();
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  EXPECT_EQ(0u, encoded_info_.redundant.size());
+  EXPECT_EQ(kEncodedSize + kRedLastHeaderLength, encoded_info_.encoded_bytes);
+
+  // Next call to the speech encoder will not produce any output.
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+
+  // Final call to the speech encoder will produce output.
+  Encode();
+  EXPECT_EQ(2 * kEncodedSize + kHeaderLenBytes, encoded_info_.encoded_bytes);
+  ASSERT_EQ(2u, encoded_info_.redundant.size());
+}
+
+// Checks that the correct payload sizes are populated into the redundancy
+// information for a redundancy level of 1.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes1) {
+  // Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
+  // of calls.
+  static const int kNumPackets = 10;
+  InSequence s;
+  for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size)));
+  }
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.redundant.size());
+  EXPECT_EQ(kRedLastHeaderLength + 1u, encoded_info_.encoded_bytes);
+
+  for (size_t i = 2; i <= kNumPackets; ++i) {
+    Encode();
+    ASSERT_EQ(2u, encoded_info_.redundant.size());
+    EXPECT_EQ(i, encoded_info_.redundant[1].encoded_bytes);
+    EXPECT_EQ(i - 1, encoded_info_.redundant[0].encoded_bytes);
+    EXPECT_EQ(5 + i + (i - 1), encoded_info_.encoded_bytes);
+  }
+}
+
+// Checks that the correct payload sizes are populated into the redundancy
+// information for a redundancy level of 0.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes0) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-0/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
+
+  // Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
+  // of calls.
+  static const int kNumPackets = 10;
+  InSequence s;
+  for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size)));
+  }
+
+  for (size_t i = 1; i <= kNumPackets; ++i) {
+    Encode();
+    ASSERT_EQ(0u, encoded_info_.redundant.size());
+    EXPECT_EQ(1 + i, encoded_info_.encoded_bytes);
+  }
+}
+// Checks that the correct payload sizes are populated into the redundancy
+// information for a redundancy level of 2.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes2) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-2/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
+
+  // Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
+  // of calls.
+  static const int kNumPackets = 10;
+  InSequence s;
+  for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size)));
+  }
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.redundant.size());
+  EXPECT_EQ(kRedLastHeaderLength + 1u, encoded_info_.encoded_bytes);
+
+  // Second call is also special since it does not include a tertiary
+  // payload.
+  Encode();
+  EXPECT_EQ(2u, encoded_info_.redundant.size());
+  EXPECT_EQ(8u, encoded_info_.encoded_bytes);
+
+  for (size_t i = 3; i <= kNumPackets; ++i) {
+    Encode();
+    ASSERT_EQ(3u, encoded_info_.redundant.size());
+    EXPECT_EQ(i, encoded_info_.redundant[2].encoded_bytes);
+    EXPECT_EQ(i - 1, encoded_info_.redundant[1].encoded_bytes);
+    EXPECT_EQ(i - 2, encoded_info_.redundant[0].encoded_bytes);
+    EXPECT_EQ(9 + i + (i - 1) + (i - 2), encoded_info_.encoded_bytes);
+  }
+}
+
+// Checks that the correct payload sizes are populated into the redundancy
+// information for a redundancy level of 3.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes3) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-3/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials_));
+
+  // Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
+  // of calls.
+  static const int kNumPackets = 10;
+  InSequence s;
+  for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size)));
+  }
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.redundant.size());
+  EXPECT_EQ(kRedLastHeaderLength + 1u, encoded_info_.encoded_bytes);
+
+  // Second call is also special since it does not include a tertiary
+  // payload.
+  Encode();
+  EXPECT_EQ(2u, encoded_info_.redundant.size());
+  EXPECT_EQ(8u, encoded_info_.encoded_bytes);
+
+  // Third call is also special since it does not include a quaternary
+  // payload.
+  Encode();
+  EXPECT_EQ(3u, encoded_info_.redundant.size());
+  EXPECT_EQ(15u, encoded_info_.encoded_bytes);
+
+  for (size_t i = 4; i <= kNumPackets; ++i) {
+    Encode();
+    ASSERT_EQ(4u, encoded_info_.redundant.size());
+    EXPECT_EQ(i, encoded_info_.redundant[3].encoded_bytes);
+    EXPECT_EQ(i - 1, encoded_info_.redundant[2].encoded_bytes);
+    EXPECT_EQ(i - 2, encoded_info_.redundant[1].encoded_bytes);
+    EXPECT_EQ(i - 3, encoded_info_.redundant[0].encoded_bytes);
+    EXPECT_EQ(13 + i + (i - 1) + (i - 2) + (i - 3),
+              encoded_info_.encoded_bytes);
+  }
+}
+
+// Checks that the correct timestamps are returned.
+TEST_F(AudioEncoderCopyRedTest, CheckTimestamps) {
+  uint32_t primary_timestamp = timestamp_;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 17;
+  info.encoded_timestamp = timestamp_;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(primary_timestamp, encoded_info_.encoded_timestamp);
+
+  uint32_t secondary_timestamp = primary_timestamp;
+  primary_timestamp = timestamp_;
+  info.encoded_timestamp = timestamp_;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+
+  Encode();
+  ASSERT_EQ(2u, encoded_info_.redundant.size());
+  EXPECT_EQ(primary_timestamp, encoded_info_.redundant[1].encoded_timestamp);
+  EXPECT_EQ(secondary_timestamp, encoded_info_.redundant[0].encoded_timestamp);
+  EXPECT_EQ(primary_timestamp, encoded_info_.encoded_timestamp);
+}
+
+// Checks that the primary and secondary payloads are written correctly.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloads) {
+  // Let the mock encoder write payloads with increasing values. The first
+  // payload will have values 0, 1, 2, ..., kPayloadLenBytes - 1.
+  static const size_t kPayloadLenBytes = 5;
+  static const size_t kHeaderLenBytes = 5;
+  uint8_t payload[kPayloadLenBytes];
+  for (uint8_t i = 0; i < kPayloadLenBytes; ++i) {
+    payload[i] = i;
+  }
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillRepeatedly(Invoke(MockAudioEncoder::CopyEncoding(payload)));
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(kRedLastHeaderLength + kPayloadLenBytes,
+            encoded_info_.encoded_bytes);
+  for (size_t i = 0; i < kPayloadLenBytes; ++i) {
+    EXPECT_EQ(i, encoded_.data()[kRedLastHeaderLength + i]);
+  }
+
+  for (int j = 0; j < 1; ++j) {
+    // Increment all values of the payload by 10.
+    for (size_t i = 0; i < kPayloadLenBytes; ++i)
+      payload[i] += 10;
+
+    Encode();
+    ASSERT_EQ(2u, encoded_info_.redundant.size());
+    EXPECT_EQ(kPayloadLenBytes, encoded_info_.redundant[0].encoded_bytes);
+    EXPECT_EQ(kPayloadLenBytes, encoded_info_.redundant[1].encoded_bytes);
+    for (size_t i = 0; i < kPayloadLenBytes; ++i) {
+      // Check secondary payload.
+      EXPECT_EQ(j * 10 + i, encoded_.data()[kHeaderLenBytes + i]);
+
+      // Check primary payload.
+      EXPECT_EQ((j + 1) * 10 + i,
+                encoded_.data()[kHeaderLenBytes + i + kPayloadLenBytes]);
+    }
+  }
+}
+
+// Checks correct propagation of payload type.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadType) {
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 17;
+  info.payload_type = primary_payload_type;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  ASSERT_EQ(0u, encoded_info_.redundant.size());
+
+  const int secondary_payload_type = red_payload_type_ + 2;
+  info.payload_type = secondary_payload_type;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+
+  Encode();
+  ASSERT_EQ(2u, encoded_info_.redundant.size());
+  EXPECT_EQ(secondary_payload_type, encoded_info_.redundant[1].payload_type);
+  EXPECT_EQ(primary_payload_type, encoded_info_.redundant[0].payload_type);
+  EXPECT_EQ(red_payload_type_, encoded_info_.payload_type);
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header) {
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 10;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Second call will produce a redundant encoding.
+
+  EXPECT_EQ(encoded_.size(),
+            5u + 2 * 10u);  // header size + two encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type | 0x80);
+
+  uint32_t timestamp_delta = encoded_info_.encoded_timestamp -
+                             encoded_info_.redundant[0].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[1], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[2] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[2] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[3], encoded_info_.redundant[1].encoded_bytes & 0xff);
+  EXPECT_EQ(encoded_[4], primary_payload_type);
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Third call will produce a redundant encoding with double
+             // redundancy.
+
+  EXPECT_EQ(encoded_.size(),
+            5u + 2 * 10u);  // header size + two encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type | 0x80);
+
+  timestamp_delta = encoded_info_.encoded_timestamp -
+                    encoded_info_.redundant[0].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[1], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[2] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[2] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[3], encoded_info_.redundant[1].encoded_bytes & 0xff);
+
+  EXPECT_EQ(encoded_[4], primary_payload_type | 0x80);
+  timestamp_delta = encoded_info_.encoded_timestamp -
+                    encoded_info_.redundant[1].encoded_timestamp;
+}
+
+// Variant with a redundancy of 0.
+TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header0) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-0/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
+
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 10;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Second call will not produce a redundant encoding.
+
+  EXPECT_EQ(encoded_.size(),
+            1u + 1 * 10u);  // header size + one encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type);
+}
+// Variant with a redundancy of 2.
+TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header2) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-2/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
+
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 10;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Second call will produce a redundant encoding.
+
+  EXPECT_EQ(encoded_.size(),
+            5u + 2 * 10u);  // header size + two encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type | 0x80);
+
+  uint32_t timestamp_delta = encoded_info_.encoded_timestamp -
+                             encoded_info_.redundant[0].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[1], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[2] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[2] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[3], encoded_info_.redundant[1].encoded_bytes & 0xff);
+  EXPECT_EQ(encoded_[4], primary_payload_type);
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Third call will produce a redundant encoding with double
+             // redundancy.
+
+  EXPECT_EQ(encoded_.size(),
+            9u + 3 * 10u);  // header size + three encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type | 0x80);
+
+  timestamp_delta = encoded_info_.encoded_timestamp -
+                    encoded_info_.redundant[0].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[1], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[2] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[2] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[3], encoded_info_.redundant[1].encoded_bytes & 0xff);
+
+  EXPECT_EQ(encoded_[4], primary_payload_type | 0x80);
+  timestamp_delta = encoded_info_.encoded_timestamp -
+                    encoded_info_.redundant[1].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[5], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[6] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[6] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[7], encoded_info_.redundant[1].encoded_bytes & 0xff);
+  EXPECT_EQ(encoded_[8], primary_payload_type);
+}
+
+TEST_F(AudioEncoderCopyRedTest, RespectsPayloadMTU) {
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 600;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  info.encoded_bytes = 500;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Second call will produce a redundant encoding.
+
+  EXPECT_EQ(encoded_.size(), 5u + 600u + 500u);
+
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  info.encoded_bytes = 400;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Third call will drop the oldest packet.
+  EXPECT_EQ(encoded_.size(), 5u + 500u + 400u);
+}
+
+TEST_F(AudioEncoderCopyRedTest, LargeTimestampGap) {
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 100;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  // Update timestamp to simulate a 400ms gap like the one
+  // opus DTX causes.
+  timestamp_ += 19200;
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  info.encoded_bytes = 200;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+
+  // The old packet will be dropped.
+  EXPECT_EQ(encoded_.size(), 1u + 200u);
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// This test fixture tests various error conditions that makes the
+// AudioEncoderCng die via CHECKs.
+class AudioEncoderCopyRedDeathTest : public AudioEncoderCopyRedTest {
+ protected:
+  AudioEncoderCopyRedDeathTest() : AudioEncoderCopyRedTest() {}
+};
+
+TEST_F(AudioEncoderCopyRedDeathTest, WrongFrameSize) {
+  num_audio_samples_10ms *= 2;  // 20 ms frame.
+  RTC_EXPECT_DEATH(Encode(), "");
+  num_audio_samples_10ms = 0;  // Zero samples.
+  RTC_EXPECT_DEATH(Encode(), "");
+}
+
+TEST_F(AudioEncoderCopyRedDeathTest, NullSpeechEncoder) {
+  test::ScopedKeyValueConfig field_trials;
+  AudioEncoderCopyRed* red = NULL;
+  AudioEncoderCopyRed::Config config;
+  config.speech_encoder = NULL;
+  RTC_EXPECT_DEATH(
+      red = new AudioEncoderCopyRed(std::move(config), field_trials),
+      "Speech encoder not provided.");
+  // The delete operation is needed to avoid leak reports from memcheck.
+  delete red;
+}
+
+#endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
new file mode 100644
index 0000000000..537e6fcede
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using ::std::get;
+
+namespace webrtc {
+
+AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
+                                         int input_sampling_khz,
+                                         int output_sampling_khz)
+    : block_duration_ms_(block_duration_ms),
+      input_sampling_khz_(input_sampling_khz),
+      output_sampling_khz_(output_sampling_khz),
+      input_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
+      output_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
+      data_pointer_(0),
+      loop_length_samples_(0),
+      max_bytes_(0),
+      encoded_bytes_(0),
+      encoding_time_ms_(0.0),
+      decoding_time_ms_(0.0),
+      out_file_(NULL) {}
+
+void AudioCodecSpeedTest::SetUp() {
+  channels_ = get<0>(GetParam());
+  bit_rate_ = get<1>(GetParam());
+  in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
+  save_out_data_ = get<4>(GetParam());
+
+  FILE* fp = fopen(in_filename_.c_str(), "rb");
+  RTC_DCHECK(fp);
+
+  // Obtain file size.
+  fseek(fp, 0, SEEK_END);
+  loop_length_samples_ = ftell(fp) / sizeof(int16_t);
+  rewind(fp);
+
+  // Allocate memory to contain the whole file.
+  in_data_.reset(
+      new int16_t[loop_length_samples_ + input_length_sample_ * channels_]);
+
+  data_pointer_ = 0;
+
+  // Copy the file into the buffer.
+  ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
+            loop_length_samples_);
+  fclose(fp);
+
+  // Add an extra block length of samples to the end of the array, starting
+  // over again from the beginning of the array. This is done to simplify
+  // the reading process when reading over the end of the loop.
+  memcpy(&in_data_[loop_length_samples_], &in_data_[0],
+         input_length_sample_ * channels_ * sizeof(int16_t));
+
+  max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
+  out_data_.reset(new int16_t[output_length_sample_ * channels_]);
+  bit_stream_.reset(new uint8_t[max_bytes_]);
+
+  if (save_out_data_) {
+    std::string out_filename =
+        ::testing::UnitTest::GetInstance()->current_test_info()->name();
+
+    // Erase '/'
+    size_t found;
+    while ((found = out_filename.find('/')) != std::string::npos)
+      out_filename.replace(found, 1, "_");
+
+    out_filename = test::OutputPath() + out_filename + ".pcm";
+
+    out_file_ = fopen(out_filename.c_str(), "wb");
+    RTC_DCHECK(out_file_);
+
+    printf("Output to be saved in %s.\n", out_filename.c_str());
+  }
+}
+
+void AudioCodecSpeedTest::TearDown() {
+  if (save_out_data_) {
+    fclose(out_file_);
+  }
+}
+
+void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
+  size_t time_now_ms = 0;
+  float time_ms;
+
+  printf("Coding %d kHz-sampled %zu-channel audio at %d bps ...\n",
+         input_sampling_khz_, channels_, bit_rate_);
+
+  while (time_now_ms < audio_duration_sec * 1000) {
+    // Encode & decode.
+    time_ms = EncodeABlock(&in_data_[data_pointer_], &bit_stream_[0],
+                           max_bytes_, &encoded_bytes_);
+    encoding_time_ms_ += time_ms;
+    time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]);
+    decoding_time_ms_ += time_ms;
+    if (save_out_data_) {
+      fwrite(&out_data_[0], sizeof(int16_t), output_length_sample_ * channels_,
+             out_file_);
+    }
+    data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
+                    loop_length_samples_;
+    time_now_ms += block_duration_ms_;
+  }
+
+  printf("Encoding: %.2f%% real time,\nDecoding: %.2f%% real time.\n",
+         (encoding_time_ms_ / audio_duration_sec) / 10.0,
+         (decoding_time_ms_ / audio_duration_sec) / 10.0);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
new file mode 100644
index 0000000000..c5f1d7c259
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+#define MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+
+#include <memory>
+#include <string>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Define coding parameter as
+// <channels, bit_rate, file_name, extension, if_save_output>.
+typedef std::tuple<size_t, int, std::string, std::string, bool> coding_param;
+
+class AudioCodecSpeedTest : public ::testing::TestWithParam<coding_param> {
+ protected:
+  AudioCodecSpeedTest(int block_duration_ms,
+                      int input_sampling_khz,
+                      int output_sampling_khz);
+  virtual void SetUp();
+  virtual void TearDown();
+
+  // EncodeABlock(...) does the following:
+  // 1. encodes a block of audio, saved in `in_data`,
+  // 2. save the bit stream to `bit_stream` of `max_bytes` bytes in size,
+  // 3. assign `encoded_bytes` with the length of the bit stream (in bytes),
+  // 4. return the cost of time (in millisecond) spent on actual encoding.
+  virtual float EncodeABlock(int16_t* in_data,
+                             uint8_t* bit_stream,
+                             size_t max_bytes,
+                             size_t* encoded_bytes) = 0;
+
+  // DecodeABlock(...) does the following:
+  // 1. decodes the bit stream in `bit_stream` with a length of `encoded_bytes`
+  // (in bytes),
+  // 2. save the decoded audio in `out_data`,
+  // 3. return the cost of time (in millisecond) spent on actual decoding.
+  virtual float DecodeABlock(const uint8_t* bit_stream,
+                             size_t encoded_bytes,
+                             int16_t* out_data) = 0;
+
+  // Encoding and decode an audio of `audio_duration` (in seconds) and
+  // record the runtime for encoding and decoding separately.
+  void EncodeDecode(size_t audio_duration);
+
+  int block_duration_ms_;
+  int input_sampling_khz_;
+  int output_sampling_khz_;
+
+  // Number of samples-per-channel in a frame.
+  size_t input_length_sample_;
+
+  // Expected output number of samples-per-channel in a frame.
+  size_t output_length_sample_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<int16_t[]> out_data_;
+  size_t data_pointer_;
+  size_t loop_length_samples_;
+  std::unique_ptr<uint8_t[]> bit_stream_;
+
+  // Maximum number of bytes in output bitstream for a frame of audio.
+  size_t max_bytes_;
+
+  size_t encoded_bytes_;
+  float encoding_time_ms_;
+  float decoding_time_ms_;
+  FILE* out_file_;
+
+  size_t channels_;
+
+  // Bit rate is in bit-per-second.
+  int bit_rate_;
+
+  std::string in_filename_;
+
+  // Determines whether to save the output to file.
+  bool save_out_data_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/default_neteq_factory_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/default_neteq_factory_gn/moz.build
new file mode 100644
index 0000000000..206a4ae313
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/default_neteq_factory_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("default_neteq_factory_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/g3doc/index.md b/third_party/libwebrtc/modules/audio_coding/g3doc/index.md
new file mode 100644
index 0000000000..467563e696
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/g3doc/index.md
@@ -0,0 +1,32 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'hlundin' reviewed: '2021-04-13'} *-->
+
+# The WebRTC Audio Coding Module
+
+WebRTC audio coding module can handle both audio sending and receiving. Folder
+[`acm2`][acm2] contains implementations of the APIs.
+
+*   Audio Sending Audio frames, each of which should always contain 10 ms worth
+    of data, are provided to the audio coding module through
+    [`Add10MsData()`][Add10MsData]. The audio coding module uses a provided
+    audio encoder to encoded audio frames and deliver the data to a
+    pre-registered audio packetization callback, which is supposed to wrap the
+    encoded audio into RTP packets and send them over a transport. Built-in
+    audio codecs are included the [`codecs`][codecs] folder. The
+    [audio network adaptor][ANA] provides an add-on functionality to an audio
+    encoder (currently limited to Opus) to make the audio encoder adaptive to
+    network conditions (bandwidth, packet loss rate, etc).
+
+*   Audio Receiving Audio packets are provided to the audio coding module
+    through [`IncomingPacket()`][IncomingPacket], and are processed by an audio
+    jitter buffer ([NetEq][NetEq]), which includes decoding of the packets.
+    Audio decoders are provided by an audio decoder factory. Decoded audio
+    samples should be queried by calling [`PlayoutData10Ms()`][PlayoutData10Ms].
+
+[acm2]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/acm2/;drc=854d59f7501aac9e9bccfa7b4d1f7f4db7842719
+[Add10MsData]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=136;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
+[codecs]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/codecs/;drc=883fea1548d58e0080f98d66fab2e0c744dfb556
+[ANA]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/audio_network_adaptor/;drc=1f99551775cd876c116d1d90cba94c8a4670d184
+[IncomingPacket]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=192;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
+[NetEq]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/neteq/;drc=213dc2cfc5f1b360b1c6fc51d393491f5de49d3d
+[PlayoutData10Ms]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/include/audio_coding_module.h;l=216;drc=d82a02c837d33cdfd75121e40dcccd32515e42d6
diff --git a/third_party/libwebrtc/modules/audio_coding/g711_c_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/g711_c_gn/moz.build
new file mode 100644
index 0000000000..8322910094
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/g711_c_gn/moz.build
@@ -0,0 +1,217 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("g711_c_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/g711_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/g711_gn/moz.build
new file mode 100644
index 0000000000..3a21cddc41
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/g711_gn/moz.build
@@ -0,0 +1,226 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("g711_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/g722_c_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/g722_c_gn/moz.build
new file mode 100644
index 0000000000..dc05788b80
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/g722_c_gn/moz.build
@@ -0,0 +1,217 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.c"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("g722_c_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/g722_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/g722_gn/moz.build
new file mode 100644
index 0000000000..910eef267f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/g722_gn/moz.build
@@ -0,0 +1,226 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("g722_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/ilbc_c_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/ilbc_c_gn/moz.build
new file mode 100644
index 0000000000..1ff9bdf07a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/ilbc_c_gn/moz.build
@@ -0,0 +1,300 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("ilbc_c_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/ilbc_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/ilbc_gn/moz.build
new file mode 100644
index 0000000000..4a7279386e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/ilbc_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("ilbc_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h
new file mode 100644
index 0000000000..8b518fb979
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module.h
@@ -0,0 +1,246 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/function_view.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_factory.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+// forward declarations
+class AudioDecoder;
+class AudioEncoder;
+class AudioFrame;
+struct RTPHeader;
+
+// Callback class used for sending data ready to be packetized
+class AudioPacketizationCallback {
+ public:
+  virtual ~AudioPacketizationCallback() {}
+
+  virtual int32_t SendData(AudioFrameType frame_type,
+                           uint8_t payload_type,
+                           uint32_t timestamp,
+                           const uint8_t* payload_data,
+                           size_t payload_len_bytes,
+                           int64_t absolute_capture_timestamp_ms) {
+    // TODO(bugs.webrtc.org/10739): Deprecate the old SendData and make this one
+    // pure virtual.
+    return SendData(frame_type, payload_type, timestamp, payload_data,
+                    payload_len_bytes);
+  }
+  virtual int32_t SendData(AudioFrameType frame_type,
+                           uint8_t payload_type,
+                           uint32_t timestamp,
+                           const uint8_t* payload_data,
+                           size_t payload_len_bytes) {
+    RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
+    return -1;
+  }
+};
+
+class AudioCodingModule {
+ protected:
+  AudioCodingModule() {}
+
+ public:
+  struct Config {
+    explicit Config(
+        rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr);
+    Config(const Config&);
+    ~Config();
+
+    NetEq::Config neteq_config;
+    Clock* clock;
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
+    NetEqFactory* neteq_factory = nullptr;
+  };
+
+  static AudioCodingModule* Create(const Config& config);
+  virtual ~AudioCodingModule() = default;
+
+  ///////////////////////////////////////////////////////////////////////////
+  //   Sender
+  //
+
+  // `modifier` is called exactly once with one argument: a pointer to the
+  // unique_ptr that holds the current encoder (which is null if there is no
+  // current encoder). For the duration of the call, `modifier` has exclusive
+  // access to the unique_ptr; it may call the encoder, steal the encoder and
+  // replace it with another encoder or with nullptr, etc.
+  virtual void ModifyEncoder(
+      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
+
+  // Utility method for simply replacing the existing encoder with a new one.
+  void SetEncoder(std::unique_ptr<AudioEncoder> new_encoder) {
+    ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+      *encoder = std::move(new_encoder);
+    });
+  }
+
+  // int32_t RegisterTransportCallback()
+  // Register a transport callback which will be called to deliver
+  // the encoded buffers whenever Process() is called and a
+  // bit-stream is ready.
+  //
+  // Input:
+  //   -transport          : pointer to the callback class
+  //                         transport->SendData() is called whenever
+  //                         Process() is called and bit-stream is ready
+  //                         to deliver.
+  //
+  // Return value:
+  //   -1 if the transport callback could not be registered
+  //    0 if registration is successful.
+  //
+  virtual int32_t RegisterTransportCallback(
+      AudioPacketizationCallback* transport) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  // int32_t Add10MsData()
+  // Add 10MS of raw (PCM) audio data and encode it. If the sampling
+  // frequency of the audio does not match the sampling frequency of the
+  // current encoder ACM will resample the audio. If an encoded packet was
+  // produced, it will be delivered via the callback object registered using
+  // RegisterTransportCallback, and the return value from this function will
+  // be the number of bytes encoded.
+  //
+  // Input:
+  //   -audio_frame        : the input audio frame, containing raw audio
+  //                         sampling frequency etc.
+  //
+  // Return value:
+  //   >= 0   number of bytes encoded.
+  //     -1   some error occurred.
+  //
+  virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  // int SetPacketLossRate()
+  // Sets expected packet loss rate for encoding. Some encoders provide packet
+  // loss gnostic encoding to make stream less sensitive to packet losses,
+  // through e.g., FEC. No effects on codecs that do not provide such encoding.
+  //
+  // Input:
+  //   -packet_loss_rate   : expected packet loss rate (0 -- 100 inclusive).
+  //
+  // Return value
+  //   -1 if failed to set packet loss rate,
+  //   0 if succeeded.
+  //
+  // This is only used in test code that rely on old ACM APIs.
+  // TODO(minyue): Remove it when possible.
+  virtual int SetPacketLossRate(int packet_loss_rate) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  //   Receiver
+  //
+
+  ///////////////////////////////////////////////////////////////////////////
+  // int32_t InitializeReceiver()
+  // Any decoder-related state of ACM will be initialized to the
+  // same state when ACM is created. This will not interrupt or
+  // effect encoding functionality of ACM. ACM would lose all the
+  // decoding-related settings by calling this function.
+  // For instance, all registered codecs are deleted and have to be
+  // registered again.
+  //
+  // Return value:
+  //   -1 if failed to initialize,
+  //    0 if succeeded.
+  //
+  virtual int32_t InitializeReceiver() = 0;
+
+  // Replace any existing decoders with the given payload type -> decoder map.
+  virtual void SetReceiveCodecs(
+      const std::map<int, SdpAudioFormat>& codecs) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  // int32_t IncomingPacket()
+  // Call this function to insert a parsed RTP packet into ACM.
+  //
+  // Inputs:
+  //   -incoming_payload   : received payload.
+  //   -payload_len_bytes  : the length of payload in bytes.
+  //   -rtp_info           : the relevant information retrieved from RTP
+  //                         header.
+  //
+  // Return value:
+  //   -1 if failed to push in the payload
+  //    0 if payload is successfully pushed in.
+  //
+  virtual int32_t IncomingPacket(const uint8_t* incoming_payload,
+                                 size_t payload_len_bytes,
+                                 const RTPHeader& rtp_header) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  // int32_t PlayoutData10Ms(
+  // Get 10 milliseconds of raw audio data for playout, at the given sampling
+  // frequency. ACM will perform a resampling if required.
+  //
+  // Input:
+  //   -desired_freq_hz    : the desired sampling frequency, in Hertz, of the
+  //                         output audio. If set to -1, the function returns
+  //                         the audio at the current sampling frequency.
+  //
+  // Output:
+  //   -audio_frame        : output audio frame which contains raw audio data
+  //                         and other relevant parameters.
+  //   -muted              : if true, the sample data in audio_frame is not
+  //                         populated, and must be interpreted as all zero.
+  //
+  // Return value:
+  //   -1 if the function fails,
+  //    0 if the function succeeds.
+  //
+  virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
+                                  AudioFrame* audio_frame,
+                                  bool* muted) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  //   statistics
+  //
+
+  ///////////////////////////////////////////////////////////////////////////
+  // int32_t  GetNetworkStatistics()
+  // Get network statistics. Note that the internal statistics of NetEq are
+  // reset by this call.
+  //
+  // Input:
+  //   -network_statistics : a structure that contains network statistics.
+  //
+  // Return value:
+  //   -1 if failed to set the network statistics,
+  //    0 if statistics are set successfully.
+  //
+  virtual int32_t GetNetworkStatistics(
+      NetworkStatistics* network_statistics) = 0;
+
+  virtual ANAStats GetANAStats() const = 0;
+
+  virtual int GetTargetBitrate() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module_typedefs.h b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
new file mode 100644
index 0000000000..9d2fcfe22e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -0,0 +1,137 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+
+#include <map>
+
+namespace webrtc {
+
+///////////////////////////////////////////////////////////////////////////
+// enum ACMVADMode
+// An enumerator for aggressiveness of VAD
+// -VADNormal                : least aggressive mode.
+// -VADLowBitrate            : more aggressive than "VADNormal" to save on
+//                             bit-rate.
+// -VADAggr                  : an aggressive mode.
+// -VADVeryAggr              : the most agressive mode.
+//
+enum ACMVADMode {
+  VADNormal = 0,
+  VADLowBitrate = 1,
+  VADAggr = 2,
+  VADVeryAggr = 3
+};
+
+enum class AudioFrameType {
+  kEmptyFrame = 0,
+  kAudioFrameSpeech = 1,
+  kAudioFrameCN = 2,
+};
+
+///////////////////////////////////////////////////////////////////////////
+//
+// Enumeration of Opus mode for intended application.
+//
+// kVoip              : optimized for voice signals.
+// kAudio             : optimized for non-voice signals like music.
+//
+enum OpusApplicationMode {
+  kVoip = 0,
+  kAudio = 1,
+};
+
+// Statistics for calls to AudioCodingModule::PlayoutData10Ms().
+struct AudioDecodingCallStats {
+  AudioDecodingCallStats()
+      : calls_to_silence_generator(0),
+        calls_to_neteq(0),
+        decoded_normal(0),
+        decoded_neteq_plc(0),
+        decoded_codec_plc(0),
+        decoded_cng(0),
+        decoded_plc_cng(0),
+        decoded_muted_output(0) {}
+
+  int calls_to_silence_generator;  // Number of calls where silence generated,
+                                   // and NetEq was disengaged from decoding.
+  int calls_to_neteq;              // Number of calls to NetEq.
+  int decoded_normal;  // Number of calls where audio RTP packet decoded.
+  int decoded_neteq_plc;  // Number of calls resulted in NetEq PLC.
+  int decoded_codec_plc;  // Number of calls resulted in codec PLC.
+  int decoded_cng;  // Number of calls where comfort noise generated due to DTX.
+  int decoded_plc_cng;       // Number of calls resulted where PLC faded to CNG.
+  int decoded_muted_output;  // Number of calls returning a muted state output.
+};
+
+// NETEQ statistics.
+struct NetworkStatistics {
+  // current jitter buffer size in ms
+  uint16_t currentBufferSize;
+  // preferred (optimal) buffer size in ms
+  uint16_t preferredBufferSize;
+  // adding extra delay due to "peaky jitter"
+  bool jitterPeaksFound;
+  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
+  uint64_t totalSamplesReceived;
+  uint64_t concealedSamples;
+  uint64_t silentConcealedSamples;
+  uint64_t concealmentEvents;
+  uint64_t jitterBufferDelayMs;
+  uint64_t jitterBufferTargetDelayMs;
+  uint64_t jitterBufferMinimumDelayMs;
+  uint64_t jitterBufferEmittedCount;
+  uint64_t insertedSamplesForDeceleration;
+  uint64_t removedSamplesForAcceleration;
+  uint64_t fecPacketsReceived;
+  uint64_t fecPacketsDiscarded;
+  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats
+  uint64_t packetsDiscarded;
+  // Stats below DO NOT correspond directly to anything in the WebRTC stats
+  // fraction (of original stream) of synthesized audio inserted through
+  // expansion (in Q14)
+  uint16_t currentExpandRate;
+  // fraction (of original stream) of synthesized speech inserted through
+  // expansion (in Q14)
+  uint16_t currentSpeechExpandRate;
+  // fraction of synthesized speech inserted through pre-emptive expansion
+  // (in Q14)
+  uint16_t currentPreemptiveRate;
+  // fraction of data removed through acceleration (in Q14)
+  uint16_t currentAccelerateRate;
+  // fraction of data coming from secondary decoding (in Q14)
+  uint16_t currentSecondaryDecodedRate;
+  // Fraction of secondary data, including FEC and RED, that is discarded (in
+  // Q14). Discarding of secondary data can be caused by the reception of the
+  // primary data, obsoleting the secondary data. It can also be caused by early
+  // or late arrival of secondary data.
+  uint16_t currentSecondaryDiscardedRate;
+  // average packet waiting time in the jitter buffer (ms)
+  int meanWaitingTimeMs;
+  // max packet waiting time in the jitter buffer (ms)
+  int maxWaitingTimeMs;
+  // count of the number of buffer flushes
+  uint64_t packetBufferFlushes;
+  // number of samples expanded due to delayed packets
+  uint64_t delayedPacketOutageSamples;
+  // arrival delay of incoming packets
+  uint64_t relativePacketArrivalDelayMs;
+  // number of audio interruptions
+  int32_t interruptionCount;
+  // total duration of audio interruptions
+  int32_t totalInterruptionDurationMs;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/isac_bwinfo_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/isac_bwinfo_gn/moz.build
new file mode 100644
index 0000000000..bb9cdf4460
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/isac_bwinfo_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("isac_bwinfo_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/isac_vad_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/isac_vad_gn/moz.build
new file mode 100644
index 0000000000..86f566663a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/isac_vad_gn/moz.build
@@ -0,0 +1,220 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("isac_vad_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/legacy_encoded_audio_frame_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/legacy_encoded_audio_frame_gn/moz.build
new file mode 100644
index 0000000000..b642ffbe95
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/legacy_encoded_audio_frame_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("legacy_encoded_audio_frame_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc
new file mode 100644
index 0000000000..f4ef6cdccb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/accelerate.h"
+
+
+#include "api/array_view.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+namespace webrtc {
+
+Accelerate::ReturnCodes Accelerate::Process(const int16_t* input,
+                                            size_t input_length,
+                                            bool fast_accelerate,
+                                            AudioMultiVector* output,
+                                            size_t* length_change_samples) {
+  // Input length must be (almost) 30 ms.
+  static const size_t k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
+  if (num_channels_ == 0 ||
+      input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) {
+    // Length of input data too short to do accelerate. Simply move all data
+    // from input to output.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kError;
+  }
+  return TimeStretch::Process(input, input_length, fast_accelerate, output,
+                              length_change_samples);
+}
+
+void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/,
+                                               int16_t* best_correlation,
+                                               size_t* /*peak_index*/) const {
+  // When the signal does not contain any active speech, the correlation does
+  // not matter. Simply set it to zero.
+  *best_correlation = 0;
+}
+
+Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
+    const int16_t* input,
+    size_t input_length,
+    size_t peak_index,
+    int16_t best_correlation,
+    bool active_speech,
+    bool fast_mode,
+    AudioMultiVector* output) const {
+  // Check for strong correlation or passive speech.
+  // Use 8192 (0.5 in Q14) in fast mode.
+  const int correlation_threshold = fast_mode ? 8192 : kCorrelationThreshold;
+  if ((best_correlation > correlation_threshold) || !active_speech) {
+    // Do accelerate operation by overlap add.
+
+    // Pre-calculate common multiplication with `fs_mult_`.
+    // 120 corresponds to 15 ms.
+    size_t fs_mult_120 = fs_mult_ * 120;
+
+    if (fast_mode) {
+      // Fit as many multiples of `peak_index` as possible in fs_mult_120.
+      // TODO(henrik.lundin) Consider finding multiple correlation peaks and
+      // pick the one with the longest correlation lag in this case.
+      peak_index = (fs_mult_120 / peak_index) * peak_index;
+    }
+
+    RTC_DCHECK_GE(fs_mult_120, peak_index);  // Should be handled in Process().
+    // Copy first part; 0 to 15 ms.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, fs_mult_120 * num_channels_));
+    // Copy the `peak_index` starting at 15 ms to `temp_vector`.
+    AudioMultiVector temp_vector(num_channels_);
+    temp_vector.PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[fs_mult_120 * num_channels_], peak_index * num_channels_));
+    // Cross-fade `temp_vector` onto the end of `output`.
+    output->CrossFade(temp_vector, peak_index);
+    // Copy the last unmodified part, 15 ms + pitch period until the end.
+    output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[(fs_mult_120 + peak_index) * num_channels_],
+        input_length - (fs_mult_120 + peak_index) * num_channels_));
+
+    if (active_speech) {
+      return kSuccess;
+    } else {
+      return kSuccessLowEnergy;
+    }
+  } else {
+    // Accelerate not allowed. Simply move all data from decoded to outData.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kNoStretch;
+  }
+}
+
+Accelerate* AccelerateFactory::Create(
+    int sample_rate_hz,
+    size_t num_channels,
+    const BackgroundNoise& background_noise) const {
+  return new Accelerate(sample_rate_hz, num_channels, background_noise);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.h b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.h
new file mode 100644
index 0000000000..01fe874d54
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
+#define MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/neteq/time_stretch.h"
+
+namespace webrtc {
+
+class AudioMultiVector;
+class BackgroundNoise;
+
+// This class implements the Accelerate operation. Most of the work is done
+// in the base class TimeStretch, which is shared with the PreemptiveExpand
+// operation. In the Accelerate class, the operations that are specific to
+// Accelerate are implemented.
+class Accelerate : public TimeStretch {
+ public:
+  Accelerate(int sample_rate_hz,
+             size_t num_channels,
+             const BackgroundNoise& background_noise)
+      : TimeStretch(sample_rate_hz, num_channels, background_noise) {}
+
+  Accelerate(const Accelerate&) = delete;
+  Accelerate& operator=(const Accelerate&) = delete;
+
+  // This method performs the actual Accelerate operation. The samples are
+  // read from `input`, of length `input_length` elements, and are written to
+  // `output`. The number of samples removed through time-stretching is
+  // is provided in the output `length_change_samples`. The method returns
+  // the outcome of the operation as an enumerator value. If `fast_accelerate`
+  // is true, the algorithm will relax the requirements on finding strong
+  // correlations, and may remove multiple pitch periods if possible.
+  ReturnCodes Process(const int16_t* input,
+                      size_t input_length,
+                      bool fast_accelerate,
+                      AudioMultiVector* output,
+                      size_t* length_change_samples);
+
+ protected:
+  // Sets the parameters `best_correlation` and `peak_index` to suitable
+  // values when the signal contains no active speech.
+  void SetParametersForPassiveSpeech(size_t len,
+                                     int16_t* best_correlation,
+                                     size_t* peak_index) const override;
+
+  // Checks the criteria for performing the time-stretching operation and,
+  // if possible, performs the time-stretching.
+  ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
+                                      size_t input_length,
+                                      size_t peak_index,
+                                      int16_t best_correlation,
+                                      bool active_speech,
+                                      bool fast_mode,
+                                      AudioMultiVector* output) const override;
+};
+
+struct AccelerateFactory {
+  AccelerateFactory() {}
+  virtual ~AccelerateFactory() {}
+
+  virtual Accelerate* Create(int sample_rate_hz,
+                             size_t num_channels,
+                             const BackgroundNoise& background_noise) const;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
new file mode 100644
index 0000000000..fef3c3c1e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -0,0 +1,526 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+
+#include <array>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "rtc_base/system/arch.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kOverheadBytesPerPacket = 50;
+
+// The absolute difference between the input and output (the first channel) is
+// compared vs `tolerance`. The parameter `delay` is used to correct for codec
+// delays.
+void CompareInputOutput(const std::vector<int16_t>& input,
+                        const std::vector<int16_t>& output,
+                        size_t num_samples,
+                        size_t channels,
+                        int tolerance,
+                        int delay) {
+  ASSERT_LE(num_samples, input.size());
+  ASSERT_LE(num_samples * channels, output.size());
+  for (unsigned int n = 0; n < num_samples - delay; ++n) {
+    ASSERT_NEAR(input[n], output[channels * n + delay], tolerance)
+        << "Exit test on first diff; n = " << n;
+  }
+}
+
+// The absolute difference between the first two channels in `output` is
+// compared vs `tolerance`.
+void CompareTwoChannels(const std::vector<int16_t>& output,
+                        size_t samples_per_channel,
+                        size_t channels,
+                        int tolerance) {
+  ASSERT_GE(channels, 2u);
+  ASSERT_LE(samples_per_channel * channels, output.size());
+  for (unsigned int n = 0; n < samples_per_channel; ++n)
+    ASSERT_NEAR(output[channels * n], output[channels * n + 1], tolerance)
+        << "Stereo samples differ.";
+}
+
+// Calculates mean-squared error between input and output (the first channel).
+// The parameter `delay` is used to correct for codec delays.
+double MseInputOutput(const std::vector<int16_t>& input,
+                      const std::vector<int16_t>& output,
+                      size_t num_samples,
+                      size_t channels,
+                      int delay) {
+  RTC_DCHECK_LT(delay, static_cast<int>(num_samples));
+  RTC_DCHECK_LE(num_samples, input.size());
+  RTC_DCHECK_LE(num_samples * channels, output.size());
+  if (num_samples == 0)
+    return 0.0;
+  double squared_sum = 0.0;
+  for (unsigned int n = 0; n < num_samples - delay; ++n) {
+    squared_sum += (input[n] - output[channels * n + delay]) *
+                   (input[n] - output[channels * n + delay]);
+  }
+  return squared_sum / (num_samples - delay);
+}
+}  // namespace
+
+class AudioDecoderTest : public ::testing::Test {
+ protected:
+  AudioDecoderTest()
+      : input_audio_(
+            webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+            32000),
+        codec_input_rate_hz_(32000),  // Legacy default value.
+        frame_size_(0),
+        data_length_(0),
+        channels_(1),
+        payload_type_(17),
+        decoder_(NULL) {}
+
+  ~AudioDecoderTest() override {}
+
+  void SetUp() override {
+    if (audio_encoder_)
+      codec_input_rate_hz_ = audio_encoder_->SampleRateHz();
+    // Create arrays.
+    ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0";
+  }
+
+  void TearDown() override {
+    delete decoder_;
+    decoder_ = NULL;
+  }
+
+  virtual void InitEncoder() {}
+
+  // TODO(henrik.lundin) Change return type to size_t once most/all overriding
+  // implementations are gone.
+  virtual int EncodeFrame(const int16_t* input,
+                          size_t input_len_samples,
+                          rtc::Buffer* output) {
+    AudioEncoder::EncodedInfo encoded_info;
+    const size_t samples_per_10ms = audio_encoder_->SampleRateHz() / 100;
+    RTC_CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),
+                 input_len_samples);
+    std::unique_ptr<int16_t[]> interleaved_input(
+        new int16_t[channels_ * samples_per_10ms]);
+    for (size_t i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
+      EXPECT_EQ(0u, encoded_info.encoded_bytes);
+
+      // Duplicate the mono input signal to however many channels the test
+      // wants.
+      test::InputAudioFile::DuplicateInterleaved(input + i * samples_per_10ms,
+                                                 samples_per_10ms, channels_,
+                                                 interleaved_input.get());
+
+      encoded_info =
+          audio_encoder_->Encode(0,
+                                 rtc::ArrayView<const int16_t>(
+                                     interleaved_input.get(),
+                                     audio_encoder_->NumChannels() *
+                                         audio_encoder_->SampleRateHz() / 100),
+                                 output);
+    }
+    EXPECT_EQ(payload_type_, encoded_info.payload_type);
+    return static_cast<int>(encoded_info.encoded_bytes);
+  }
+
+  // Encodes and decodes audio. The absolute difference between the input and
+  // output is compared vs `tolerance`, and the mean-squared error is compared
+  // with `mse`. The encoded stream should contain `expected_bytes`. For stereo
+  // audio, the absolute difference between the two channels is compared vs
+  // `channel_diff_tolerance`.
+  void EncodeDecodeTest(size_t expected_bytes,
+                        int tolerance,
+                        double mse,
+                        int delay = 0,
+                        int channel_diff_tolerance = 0) {
+    ASSERT_GE(tolerance, 0) << "Test must define a tolerance >= 0";
+    ASSERT_GE(channel_diff_tolerance, 0)
+        << "Test must define a channel_diff_tolerance >= 0";
+    size_t processed_samples = 0u;
+    size_t encoded_bytes = 0u;
+    InitEncoder();
+    std::vector<int16_t> input;
+    std::vector<int16_t> decoded;
+    while (processed_samples + frame_size_ <= data_length_) {
+      // Extend input vector with `frame_size_`.
+      input.resize(input.size() + frame_size_, 0);
+      // Read from input file.
+      ASSERT_GE(input.size() - processed_samples, frame_size_);
+      ASSERT_TRUE(input_audio_.Read(frame_size_, codec_input_rate_hz_,
+                                    &input[processed_samples]));
+      rtc::Buffer encoded;
+      size_t enc_len =
+          EncodeFrame(&input[processed_samples], frame_size_, &encoded);
+      // Make sure that frame_size_ * channels_ samples are allocated and free.
+      decoded.resize((processed_samples + frame_size_) * channels_, 0);
+
+      const std::vector<AudioDecoder::ParseResult> parse_result =
+          decoder_->ParsePayload(std::move(encoded), /*timestamp=*/0);
+      RTC_CHECK_EQ(parse_result.size(), size_t{1});
+      auto decode_result = parse_result[0].frame->Decode(
+          rtc::ArrayView<int16_t>(&decoded[processed_samples * channels_],
+                                  frame_size_ * channels_ * sizeof(int16_t)));
+      RTC_CHECK(decode_result.has_value());
+      EXPECT_EQ(frame_size_ * channels_, decode_result->num_decoded_samples);
+      encoded_bytes += enc_len;
+      processed_samples += frame_size_;
+    }
+    // For some codecs it doesn't make sense to check expected number of bytes,
+    // since the number can vary for different platforms. Opus is such a codec.
+    // In this case expected_bytes is set to 0.
+    if (expected_bytes) {
+      EXPECT_EQ(expected_bytes, encoded_bytes);
+    }
+    CompareInputOutput(input, decoded, processed_samples, channels_, tolerance,
+                       delay);
+    if (channels_ == 2)
+      CompareTwoChannels(decoded, processed_samples, channels_,
+                         channel_diff_tolerance);
+    EXPECT_LE(
+        MseInputOutput(input, decoded, processed_samples, channels_, delay),
+        mse);
+  }
+
+  // Encodes a payload and decodes it twice with decoder re-init before each
+  // decode. Verifies that the decoded result is the same.
+  void ReInitTest() {
+    InitEncoder();
+    std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);
+    ASSERT_TRUE(
+        input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
+    std::array<rtc::Buffer, 2> encoded;
+    EncodeFrame(input.get(), frame_size_, &encoded[0]);
+    // Make a copy.
+    encoded[1].SetData(encoded[0].data(), encoded[0].size());
+
+    std::array<std::vector<int16_t>, 2> outputs;
+    for (size_t i = 0; i < outputs.size(); ++i) {
+      outputs[i].resize(frame_size_ * channels_);
+      decoder_->Reset();
+      const std::vector<AudioDecoder::ParseResult> parse_result =
+          decoder_->ParsePayload(std::move(encoded[i]), /*timestamp=*/0);
+      RTC_CHECK_EQ(parse_result.size(), size_t{1});
+      auto decode_result = parse_result[0].frame->Decode(outputs[i]);
+      RTC_CHECK(decode_result.has_value());
+      EXPECT_EQ(frame_size_ * channels_, decode_result->num_decoded_samples);
+    }
+    EXPECT_EQ(outputs[0], outputs[1]);
+  }
+
+  // Call DecodePlc and verify that the correct number of samples is produced.
+  void DecodePlcTest() {
+    InitEncoder();
+    std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);
+    ASSERT_TRUE(
+        input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
+    rtc::Buffer encoded;
+    EncodeFrame(input.get(), frame_size_, &encoded);
+    decoder_->Reset();
+    std::vector<int16_t> output(frame_size_ * channels_);
+    const std::vector<AudioDecoder::ParseResult> parse_result =
+        decoder_->ParsePayload(std::move(encoded), /*timestamp=*/0);
+    RTC_CHECK_EQ(parse_result.size(), size_t{1});
+    auto decode_result = parse_result[0].frame->Decode(output);
+    RTC_CHECK(decode_result.has_value());
+    EXPECT_EQ(frame_size_ * channels_, decode_result->num_decoded_samples);
+    // Call DecodePlc and verify that we get one frame of data.
+    // (Overwrite the output from the above Decode call, but that does not
+    // matter.)
+    size_t dec_len =
+        decoder_->DecodePlc(/*num_frames=*/1, /*decoded=*/output.data());
+    EXPECT_EQ(frame_size_ * channels_, dec_len);
+  }
+
+  test::ResampleInputAudioFile input_audio_;
+  int codec_input_rate_hz_;
+  size_t frame_size_;
+  size_t data_length_;
+  size_t channels_;
+  const int payload_type_;
+  AudioDecoder* decoder_;
+  std::unique_ptr<AudioEncoder> audio_encoder_;
+};
+
+class AudioDecoderPcmUTest : public AudioDecoderTest {
+ protected:
+  AudioDecoderPcmUTest() : AudioDecoderTest() {
+    frame_size_ = 160;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderPcmU(1);
+    AudioEncoderPcmU::Config config;
+    config.frame_size_ms = static_cast<int>(frame_size_ / 8);
+    config.payload_type = payload_type_;
+    audio_encoder_.reset(new AudioEncoderPcmU(config));
+  }
+};
+
+class AudioDecoderPcmATest : public AudioDecoderTest {
+ protected:
+  AudioDecoderPcmATest() : AudioDecoderTest() {
+    frame_size_ = 160;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderPcmA(1);
+    AudioEncoderPcmA::Config config;
+    config.frame_size_ms = static_cast<int>(frame_size_ / 8);
+    config.payload_type = payload_type_;
+    audio_encoder_.reset(new AudioEncoderPcmA(config));
+  }
+};
+
+class AudioDecoderPcm16BTest : public AudioDecoderTest {
+ protected:
+  AudioDecoderPcm16BTest() : AudioDecoderTest() {
+    codec_input_rate_hz_ = 16000;
+    frame_size_ = 20 * codec_input_rate_hz_ / 1000;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderPcm16B(codec_input_rate_hz_, 1);
+    RTC_DCHECK(decoder_);
+    AudioEncoderPcm16B::Config config;
+    config.sample_rate_hz = codec_input_rate_hz_;
+    config.frame_size_ms =
+        static_cast<int>(frame_size_ / (config.sample_rate_hz / 1000));
+    config.payload_type = payload_type_;
+    audio_encoder_.reset(new AudioEncoderPcm16B(config));
+  }
+};
+
+class AudioDecoderIlbcTest : public AudioDecoderTest {
+ protected:
+  AudioDecoderIlbcTest() : AudioDecoderTest() {
+    codec_input_rate_hz_ = 8000;
+    frame_size_ = 240;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderIlbcImpl;
+    RTC_DCHECK(decoder_);
+    AudioEncoderIlbcConfig config;
+    config.frame_size_ms = 30;
+    audio_encoder_.reset(new AudioEncoderIlbcImpl(config, payload_type_));
+  }
+
+  // Overload the default test since iLBC's function WebRtcIlbcfix_NetEqPlc does
+  // not return any data. It simply resets a few states and returns 0.
+  void DecodePlcTest() {
+    InitEncoder();
+    std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);
+    ASSERT_TRUE(
+        input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
+    rtc::Buffer encoded;
+    size_t enc_len = EncodeFrame(input.get(), frame_size_, &encoded);
+    AudioDecoder::SpeechType speech_type;
+    decoder_->Reset();
+    std::unique_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
+    size_t dec_len = decoder_->Decode(
+        encoded.data(), enc_len, codec_input_rate_hz_,
+        frame_size_ * channels_ * sizeof(int16_t), output.get(), &speech_type);
+    EXPECT_EQ(frame_size_, dec_len);
+    // Simply call DecodePlc and verify that we get 0 as return value.
+    EXPECT_EQ(0U, decoder_->DecodePlc(1, output.get()));
+  }
+};
+
+class AudioDecoderG722Test : public AudioDecoderTest {
+ protected:
+  AudioDecoderG722Test() : AudioDecoderTest() {
+    codec_input_rate_hz_ = 16000;
+    frame_size_ = 160;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderG722Impl;
+    RTC_DCHECK(decoder_);
+    AudioEncoderG722Config config;
+    config.frame_size_ms = 10;
+    config.num_channels = 1;
+    audio_encoder_.reset(new AudioEncoderG722Impl(config, payload_type_));
+  }
+};
+
+class AudioDecoderG722StereoTest : public AudioDecoderTest {
+ protected:
+  AudioDecoderG722StereoTest() : AudioDecoderTest() {
+    channels_ = 2;
+    codec_input_rate_hz_ = 16000;
+    frame_size_ = 160;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderG722StereoImpl;
+    RTC_DCHECK(decoder_);
+    AudioEncoderG722Config config;
+    config.frame_size_ms = 10;
+    config.num_channels = 2;
+    audio_encoder_.reset(new AudioEncoderG722Impl(config, payload_type_));
+  }
+};
+
+class AudioDecoderOpusTest
+    : public AudioDecoderTest,
+      public testing::WithParamInterface<std::tuple<int, int>> {
+ protected:
+  AudioDecoderOpusTest() : AudioDecoderTest() {
+    channels_ = opus_num_channels_;
+    codec_input_rate_hz_ = opus_sample_rate_hz_;
+    frame_size_ = rtc::CheckedDivExact(opus_sample_rate_hz_, 100);
+    data_length_ = 10 * frame_size_;
+    decoder_ =
+        new AudioDecoderOpusImpl(opus_num_channels_, opus_sample_rate_hz_);
+    AudioEncoderOpusConfig config;
+    config.frame_size_ms = 10;
+    config.sample_rate_hz = opus_sample_rate_hz_;
+    config.num_channels = opus_num_channels_;
+    config.application = opus_num_channels_ == 1
+                             ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                             : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+    audio_encoder_ = AudioEncoderOpus::MakeAudioEncoder(config, payload_type_);
+    audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket);
+  }
+  const int opus_sample_rate_hz_{std::get<0>(GetParam())};
+  const int opus_num_channels_{std::get<1>(GetParam())};
+};
+
+INSTANTIATE_TEST_SUITE_P(Param,
+                         AudioDecoderOpusTest,
+                         testing::Combine(testing::Values(16000, 48000),
+                                          testing::Values(1, 2)));
+
+TEST_F(AudioDecoderPcmUTest, EncodeDecode) {
+  int tolerance = 251;
+  double mse = 1734.0;
+  EncodeDecodeTest(data_length_, tolerance, mse);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+namespace {
+int SetAndGetTargetBitrate(AudioEncoder* audio_encoder, int rate) {
+  audio_encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
+  return audio_encoder->GetTargetBitrate();
+}
+void TestSetAndGetTargetBitratesWithFixedCodec(AudioEncoder* audio_encoder,
+                                               int fixed_rate) {
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, 32000));
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate - 1));
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate));
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate + 1));
+}
+}  // namespace
+
+TEST_F(AudioDecoderPcmUTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
+}
+
+TEST_F(AudioDecoderPcmATest, EncodeDecode) {
+  int tolerance = 308;
+  double mse = 1931.0;
+  EncodeDecodeTest(data_length_, tolerance, mse);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_F(AudioDecoderPcmATest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
+}
+
+TEST_F(AudioDecoderPcm16BTest, EncodeDecode) {
+  int tolerance = 0;
+  double mse = 0.0;
+  EncodeDecodeTest(2 * data_length_, tolerance, mse);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_F(AudioDecoderPcm16BTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(),
+                                            codec_input_rate_hz_ * 16);
+}
+
+TEST_F(AudioDecoderIlbcTest, EncodeDecode) {
+  int tolerance = 6808;
+  double mse = 2.13e6;
+  int delay = 80;  // Delay from input to output.
+  EncodeDecodeTest(500, tolerance, mse, delay);
+  ReInitTest();
+  EXPECT_TRUE(decoder_->HasDecodePlc());
+  DecodePlcTest();
+}
+
+TEST_F(AudioDecoderIlbcTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 13333);
+}
+
+TEST_F(AudioDecoderG722Test, EncodeDecode) {
+  int tolerance = 6176;
+  double mse = 238630.0;
+  int delay = 22;  // Delay from input to output.
+  EncodeDecodeTest(data_length_ / 2, tolerance, mse, delay);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_F(AudioDecoderG722Test, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
+}
+
+TEST_F(AudioDecoderG722StereoTest, EncodeDecode) {
+  int tolerance = 6176;
+  int channel_diff_tolerance = 0;
+  double mse = 238630.0;
+  int delay = 22;  // Delay from input to output.
+  EncodeDecodeTest(data_length_, tolerance, mse, delay, channel_diff_tolerance);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_F(AudioDecoderG722StereoTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 128000);
+}
+
+// TODO(http://bugs.webrtc.org/12518): Enable the test after Opus has been
+// updated.
+TEST_P(AudioDecoderOpusTest, DISABLED_EncodeDecode) {
+  constexpr int tolerance = 6176;
+  constexpr int channel_diff_tolerance = 6;
+  constexpr double mse = 238630.0;
+  constexpr int delay = 22;  // Delay from input to output.
+  EncodeDecodeTest(0, tolerance, mse, delay, channel_diff_tolerance);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_P(AudioDecoderOpusTest, SetTargetBitrate) {
+  const int overhead_rate =
+      8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_;
+  EXPECT_EQ(6000,
+            SetAndGetTargetBitrate(audio_encoder_.get(), 5999 + overhead_rate));
+  EXPECT_EQ(6000,
+            SetAndGetTargetBitrate(audio_encoder_.get(), 6000 + overhead_rate));
+  EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
+                                          32000 + overhead_rate));
+  EXPECT_EQ(510000, SetAndGetTargetBitrate(audio_encoder_.get(),
+                                           510000 + overhead_rate));
+  EXPECT_EQ(510000, SetAndGetTargetBitrate(audio_encoder_.get(),
+                                           511000 + overhead_rate));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.cc
new file mode 100644
index 0000000000..14ae94649b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.cc
@@ -0,0 +1,222 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioMultiVector::AudioMultiVector(size_t N) {
+  RTC_DCHECK_GT(N, 0);
+  if (N < 1)
+    N = 1;
+  for (size_t n = 0; n < N; ++n) {
+    channels_.push_back(new AudioVector);
+  }
+  num_channels_ = N;
+}
+
+AudioMultiVector::AudioMultiVector(size_t N, size_t initial_size) {
+  RTC_DCHECK_GT(N, 0);
+  if (N < 1)
+    N = 1;
+  for (size_t n = 0; n < N; ++n) {
+    channels_.push_back(new AudioVector(initial_size));
+  }
+  num_channels_ = N;
+}
+
+AudioMultiVector::~AudioMultiVector() {
+  std::vector<AudioVector*>::iterator it = channels_.begin();
+  while (it != channels_.end()) {
+    delete (*it);
+    ++it;
+  }
+}
+
+void AudioMultiVector::Clear() {
+  for (size_t i = 0; i < num_channels_; ++i) {
+    channels_[i]->Clear();
+  }
+}
+
+void AudioMultiVector::Zeros(size_t length) {
+  for (size_t i = 0; i < num_channels_; ++i) {
+    channels_[i]->Clear();
+    channels_[i]->Extend(length);
+  }
+}
+
+void AudioMultiVector::CopyTo(AudioMultiVector* copy_to) const {
+  if (copy_to) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->CopyTo(&(*copy_to)[i]);
+    }
+  }
+}
+
+void AudioMultiVector::PushBackInterleaved(
+    rtc::ArrayView<const int16_t> append_this) {
+  RTC_DCHECK_EQ(append_this.size() % num_channels_, 0);
+  if (append_this.empty()) {
+    return;
+  }
+  if (num_channels_ == 1) {
+    // Special case to avoid extra allocation and data shuffling.
+    channels_[0]->PushBack(append_this.data(), append_this.size());
+    return;
+  }
+  size_t length_per_channel = append_this.size() / num_channels_;
+  int16_t* temp_array = new int16_t[length_per_channel];  // Temporary storage.
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    // Copy elements to `temp_array`.
+    for (size_t i = 0; i < length_per_channel; ++i) {
+      temp_array[i] = append_this[channel + i * num_channels_];
+    }
+    channels_[channel]->PushBack(temp_array, length_per_channel);
+  }
+  delete[] temp_array;
+}
+
+void AudioMultiVector::PushBack(const AudioMultiVector& append_this) {
+  RTC_DCHECK_EQ(num_channels_, append_this.num_channels_);
+  if (num_channels_ == append_this.num_channels_) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->PushBack(append_this[i]);
+    }
+  }
+}
+
+void AudioMultiVector::PushBackFromIndex(const AudioMultiVector& append_this,
+                                         size_t index) {
+  RTC_DCHECK_LT(index, append_this.Size());
+  index = std::min(index, append_this.Size() - 1);
+  size_t length = append_this.Size() - index;
+  RTC_DCHECK_EQ(num_channels_, append_this.num_channels_);
+  if (num_channels_ == append_this.num_channels_) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->PushBack(append_this[i], length, index);
+    }
+  }
+}
+
+void AudioMultiVector::PopFront(size_t length) {
+  for (size_t i = 0; i < num_channels_; ++i) {
+    channels_[i]->PopFront(length);
+  }
+}
+
+void AudioMultiVector::PopBack(size_t length) {
+  for (size_t i = 0; i < num_channels_; ++i) {
+    channels_[i]->PopBack(length);
+  }
+}
+
+size_t AudioMultiVector::ReadInterleaved(size_t length,
+                                         int16_t* destination) const {
+  return ReadInterleavedFromIndex(0, length, destination);
+}
+
+size_t AudioMultiVector::ReadInterleavedFromIndex(size_t start_index,
+                                                  size_t length,
+                                                  int16_t* destination) const {
+  RTC_DCHECK(destination);
+  size_t index = 0;  // Number of elements written to `destination` so far.
+  RTC_DCHECK_LE(start_index, Size());
+  start_index = std::min(start_index, Size());
+  if (length + start_index > Size()) {
+    length = Size() - start_index;
+  }
+  if (num_channels_ == 1) {
+    // Special case to avoid the nested for loop below.
+    (*this)[0].CopyTo(length, start_index, destination);
+    return length;
+  }
+  for (size_t i = 0; i < length; ++i) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      destination[index] = (*this)[channel][i + start_index];
+      ++index;
+    }
+  }
+  return index;
+}
+
+size_t AudioMultiVector::ReadInterleavedFromEnd(size_t length,
+                                                int16_t* destination) const {
+  length = std::min(length, Size());  // Cannot read more than Size() elements.
+  return ReadInterleavedFromIndex(Size() - length, length, destination);
+}
+
+void AudioMultiVector::OverwriteAt(const AudioMultiVector& insert_this,
+                                   size_t length,
+                                   size_t position) {
+  RTC_DCHECK_EQ(num_channels_, insert_this.num_channels_);
+  // Cap `length` at the length of `insert_this`.
+  RTC_DCHECK_LE(length, insert_this.Size());
+  length = std::min(length, insert_this.Size());
+  if (num_channels_ == insert_this.num_channels_) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->OverwriteAt(insert_this[i], length, position);
+    }
+  }
+}
+
+void AudioMultiVector::CrossFade(const AudioMultiVector& append_this,
+                                 size_t fade_length) {
+  RTC_DCHECK_EQ(num_channels_, append_this.num_channels_);
+  if (num_channels_ == append_this.num_channels_) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->CrossFade(append_this[i], fade_length);
+    }
+  }
+}
+
+size_t AudioMultiVector::Channels() const {
+  return num_channels_;
+}
+
+size_t AudioMultiVector::Size() const {
+  RTC_DCHECK(channels_[0]);
+  return channels_[0]->Size();
+}
+
+void AudioMultiVector::AssertSize(size_t required_size) {
+  if (Size() < required_size) {
+    size_t extend_length = required_size - Size();
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      channels_[channel]->Extend(extend_length);
+    }
+  }
+}
+
+bool AudioMultiVector::Empty() const {
+  RTC_DCHECK(channels_[0]);
+  return channels_[0]->Empty();
+}
+
+void AudioMultiVector::CopyChannel(size_t from_channel, size_t to_channel) {
+  RTC_DCHECK_LT(from_channel, num_channels_);
+  RTC_DCHECK_LT(to_channel, num_channels_);
+  channels_[from_channel]->CopyTo(channels_[to_channel]);
+}
+
+const AudioVector& AudioMultiVector::operator[](size_t index) const {
+  return *(channels_[index]);
+}
+
+AudioVector& AudioMultiVector::operator[](size_t index) {
+  return *(channels_[index]);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.h b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.h
new file mode 100644
index 0000000000..715ec6dfc7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.h
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
+
+#include <stdint.h>
+#include <string.h>
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+
+namespace webrtc {
+
+class AudioMultiVector {
+ public:
+  // Creates an empty AudioMultiVector with `N` audio channels. `N` must be
+  // larger than 0.
+  explicit AudioMultiVector(size_t N);
+
+  // Creates an AudioMultiVector with `N` audio channels, each channel having
+  // an initial size. `N` must be larger than 0.
+  AudioMultiVector(size_t N, size_t initial_size);
+
+  virtual ~AudioMultiVector();
+
+  AudioMultiVector(const AudioMultiVector&) = delete;
+  AudioMultiVector& operator=(const AudioMultiVector&) = delete;
+
+  // Deletes all values and make the vector empty.
+  virtual void Clear();
+
+  // Clears the vector and inserts `length` zeros into each channel.
+  virtual void Zeros(size_t length);
+
+  // Copies all values from this vector to `copy_to`. Any contents in `copy_to`
+  // are deleted. After the operation is done, `copy_to` will be an exact
+  // replica of this object. The source and the destination must have the same
+  // number of channels.
+  virtual void CopyTo(AudioMultiVector* copy_to) const;
+
+  // Appends the contents of `append_this` to the end of this object. The array
+  // is assumed to be channel-interleaved. The length must be an even multiple
+  // of this object's number of channels. The length of this object is increased
+  // with the length of the array divided by the number of channels.
+  void PushBackInterleaved(rtc::ArrayView<const int16_t> append_this);
+
+  // Appends the contents of AudioMultiVector `append_this` to this object. The
+  // length of this object is increased with the length of `append_this`.
+  virtual void PushBack(const AudioMultiVector& append_this);
+
+  // Appends the contents of AudioMultiVector `append_this` to this object,
+  // taken from `index` up until the end of `append_this`. The length of this
+  // object is increased.
+  virtual void PushBackFromIndex(const AudioMultiVector& append_this,
+                                 size_t index);
+
+  // Removes `length` elements from the beginning of this object, from each
+  // channel.
+  virtual void PopFront(size_t length);
+
+  // Removes `length` elements from the end of this object, from each
+  // channel.
+  virtual void PopBack(size_t length);
+
+  // Reads `length` samples from each channel and writes them interleaved to
+  // `destination`. The total number of elements written to `destination` is
+  // returned, i.e., `length` * number of channels. If the AudioMultiVector
+  // contains less than `length` samples per channel, this is reflected in the
+  // return value.
+  virtual size_t ReadInterleaved(size_t length, int16_t* destination) const;
+
+  // Like ReadInterleaved() above, but reads from `start_index` instead of from
+  // the beginning.
+  virtual size_t ReadInterleavedFromIndex(size_t start_index,
+                                          size_t length,
+                                          int16_t* destination) const;
+
+  // Like ReadInterleaved() above, but reads from the end instead of from
+  // the beginning.
+  virtual size_t ReadInterleavedFromEnd(size_t length,
+                                        int16_t* destination) const;
+
+  // Overwrites each channel in this AudioMultiVector with values taken from
+  // `insert_this`. The values are taken from the beginning of `insert_this` and
+  // are inserted starting at `position`. `length` values are written into each
+  // channel. If `length` and `position` are selected such that the new data
+  // extends beyond the end of the current AudioVector, the vector is extended
+  // to accommodate the new data. `length` is limited to the length of
+  // `insert_this`.
+  virtual void OverwriteAt(const AudioMultiVector& insert_this,
+                           size_t length,
+                           size_t position);
+
+  // Appends `append_this` to the end of the current vector. Lets the two
+  // vectors overlap by `fade_length` samples (per channel), and cross-fade
+  // linearly in this region.
+  virtual void CrossFade(const AudioMultiVector& append_this,
+                         size_t fade_length);
+
+  // Returns the number of channels.
+  virtual size_t Channels() const;
+
+  // Returns the number of elements per channel in this AudioMultiVector.
+  virtual size_t Size() const;
+
+  // Verify that each channel can hold at least `required_size` elements. If
+  // not, extend accordingly.
+  virtual void AssertSize(size_t required_size);
+
+  virtual bool Empty() const;
+
+  // Copies the data between two channels in the AudioMultiVector. The method
+  // does not add any new channel. Thus, `from_channel` and `to_channel` must
+  // both be valid channel numbers.
+  virtual void CopyChannel(size_t from_channel, size_t to_channel);
+
+  // Accesses and modifies a channel (i.e., an AudioVector object) of this
+  // AudioMultiVector.
+  const AudioVector& operator[](size_t index) const;
+  AudioVector& operator[](size_t index);
+
+ protected:
+  std::vector<AudioVector*> channels_;
+  size_t num_channels_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
new file mode 100644
index 0000000000..386c3d48a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
@@ -0,0 +1,323 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+#include <stdlib.h>
+
+#include <string>
+#include <vector>
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// This is a value-parameterized test. The test cases are instantiated with
+// different values for the test parameter, which is used to determine the
+// number of channels in the AudioMultiBuffer. Note that it is not possible
+// to combine typed testing with value-parameterized testing, and since the
+// tests for AudioVector already covers a number of different type parameters,
+// this test focuses on testing different number of channels, and keeping the
+// value type constant.
+
+class AudioMultiVectorTest : public ::testing::TestWithParam<size_t> {
+ protected:
+  AudioMultiVectorTest()
+      : num_channels_(GetParam()),  // Get the test parameter.
+        array_interleaved_(num_channels_ * array_length()) {}
+
+  ~AudioMultiVectorTest() = default;
+
+  virtual void SetUp() {
+    // Populate test arrays.
+    for (size_t i = 0; i < array_length(); ++i) {
+      array_[i] = static_cast<int16_t>(i);
+    }
+    int16_t* ptr = array_interleaved_.data();
+    // Write 100, 101, 102, ... for first channel.
+    // Write 200, 201, 202, ... for second channel.
+    // And so on.
+    for (size_t i = 0; i < array_length(); ++i) {
+      for (size_t j = 1; j <= num_channels_; ++j) {
+        *ptr = rtc::checked_cast<int16_t>(j * 100 + i);
+        ++ptr;
+      }
+    }
+  }
+
+  size_t array_length() const { return sizeof(array_) / sizeof(array_[0]); }
+
+  const size_t num_channels_;
+  int16_t array_[10];
+  std::vector<int16_t> array_interleaved_;
+};
+
+// Create and destroy AudioMultiVector objects, both empty and with a predefined
+// length.
+TEST_P(AudioMultiVectorTest, CreateAndDestroy) {
+  AudioMultiVector vec1(num_channels_);
+  EXPECT_TRUE(vec1.Empty());
+  EXPECT_EQ(num_channels_, vec1.Channels());
+  EXPECT_EQ(0u, vec1.Size());
+
+  size_t initial_size = 17;
+  AudioMultiVector vec2(num_channels_, initial_size);
+  EXPECT_FALSE(vec2.Empty());
+  EXPECT_EQ(num_channels_, vec2.Channels());
+  EXPECT_EQ(initial_size, vec2.Size());
+}
+
+// Test the subscript operator [] for getting and setting.
+TEST_P(AudioMultiVectorTest, SubscriptOperator) {
+  AudioMultiVector vec(num_channels_, array_length());
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < array_length(); ++i) {
+      vec[channel][i] = static_cast<int16_t>(i);
+      // Make sure to use the const version.
+      const AudioVector& audio_vec = vec[channel];
+      EXPECT_EQ(static_cast<int16_t>(i), audio_vec[i]);
+    }
+  }
+}
+
+// Test the PushBackInterleaved method and the CopyFrom method. The Clear
+// method is also invoked.
+TEST_P(AudioMultiVectorTest, PushBackInterleavedAndCopy) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  AudioMultiVector vec_copy(num_channels_);
+  vec.CopyTo(&vec_copy);  // Copy from `vec` to `vec_copy`.
+  ASSERT_EQ(num_channels_, vec.Channels());
+  ASSERT_EQ(array_length(), vec.Size());
+  ASSERT_EQ(num_channels_, vec_copy.Channels());
+  ASSERT_EQ(array_length(), vec_copy.Size());
+  for (size_t channel = 0; channel < vec.Channels(); ++channel) {
+    for (size_t i = 0; i < array_length(); ++i) {
+      EXPECT_EQ(static_cast<int16_t>((channel + 1) * 100 + i), vec[channel][i]);
+      EXPECT_EQ(vec[channel][i], vec_copy[channel][i]);
+    }
+  }
+
+  // Clear `vec` and verify that it is empty.
+  vec.Clear();
+  EXPECT_TRUE(vec.Empty());
+
+  // Now copy the empty vector and verify that the copy becomes empty too.
+  vec.CopyTo(&vec_copy);
+  EXPECT_TRUE(vec_copy.Empty());
+}
+
+// Try to copy to a NULL pointer. Nothing should happen.
+TEST_P(AudioMultiVectorTest, CopyToNull) {
+  AudioMultiVector vec(num_channels_);
+  AudioMultiVector* vec_copy = NULL;
+  vec.PushBackInterleaved(array_interleaved_);
+  vec.CopyTo(vec_copy);
+}
+
+// Test the PushBack method with another AudioMultiVector as input argument.
+TEST_P(AudioMultiVectorTest, PushBackVector) {
+  AudioMultiVector vec1(num_channels_, array_length());
+  AudioMultiVector vec2(num_channels_, array_length());
+  // Set the first vector to [0, 1, ..., array_length() - 1] +
+  //   100 * channel_number.
+  // Set the second vector to [array_length(), array_length() + 1, ...,
+  //   2 * array_length() - 1] + 100 * channel_number.
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < array_length(); ++i) {
+      vec1[channel][i] = static_cast<int16_t>(i + 100 * channel);
+      vec2[channel][i] =
+          static_cast<int16_t>(i + 100 * channel + array_length());
+    }
+  }
+  // Append vec2 to the back of vec1.
+  vec1.PushBack(vec2);
+  ASSERT_EQ(2u * array_length(), vec1.Size());
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < 2 * array_length(); ++i) {
+      EXPECT_EQ(static_cast<int16_t>(i + 100 * channel), vec1[channel][i]);
+    }
+  }
+}
+
+// Test the PushBackFromIndex method.
+TEST_P(AudioMultiVectorTest, PushBackFromIndex) {
+  AudioMultiVector vec1(num_channels_);
+  vec1.PushBackInterleaved(array_interleaved_);
+  AudioMultiVector vec2(num_channels_);
+
+  // Append vec1 to the back of vec2 (which is empty). Read vec1 from the second
+  // last element.
+  vec2.PushBackFromIndex(vec1, array_length() - 2);
+  ASSERT_EQ(2u, vec2.Size());
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < 2; ++i) {
+      EXPECT_EQ(array_interleaved_[channel +
+                                   num_channels_ * (array_length() - 2 + i)],
+                vec2[channel][i]);
+    }
+  }
+}
+
+// Starts with pushing some values to the vector, then test the Zeros method.
+TEST_P(AudioMultiVectorTest, Zeros) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  vec.Zeros(2 * array_length());
+  ASSERT_EQ(num_channels_, vec.Channels());
+  ASSERT_EQ(2u * array_length(), vec.Size());
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < 2 * array_length(); ++i) {
+      EXPECT_EQ(0, vec[channel][i]);
+    }
+  }
+}
+
+// Test the ReadInterleaved method
+TEST_P(AudioMultiVectorTest, ReadInterleaved) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  int16_t* output = new int16_t[array_interleaved_.size()];
+  // Read 5 samples.
+  size_t read_samples = 5;
+  EXPECT_EQ(num_channels_ * read_samples,
+            vec.ReadInterleaved(read_samples, output));
+  EXPECT_EQ(0, memcmp(array_interleaved_.data(), output,
+                      read_samples * sizeof(int16_t)));
+
+  // Read too many samples. Expect to get all samples from the vector.
+  EXPECT_EQ(array_interleaved_.size(),
+            vec.ReadInterleaved(array_length() + 1, output));
+  EXPECT_EQ(0, memcmp(array_interleaved_.data(), output,
+                      read_samples * sizeof(int16_t)));
+
+  delete[] output;
+}
+
+// Test the PopFront method.
+TEST_P(AudioMultiVectorTest, PopFront) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  vec.PopFront(1);  // Remove one element from each channel.
+  ASSERT_EQ(array_length() - 1u, vec.Size());
+  // Let `ptr` point to the second element of the first channel in the
+  // interleaved array.
+  int16_t* ptr = &array_interleaved_[num_channels_];
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      EXPECT_EQ(*ptr, vec[channel][i]);
+      ++ptr;
+    }
+  }
+  vec.PopFront(array_length());  // Remove more elements than vector size.
+  EXPECT_EQ(0u, vec.Size());
+}
+
+// Test the PopBack method.
+TEST_P(AudioMultiVectorTest, PopBack) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  vec.PopBack(1);  // Remove one element from each channel.
+  ASSERT_EQ(array_length() - 1u, vec.Size());
+  // Let `ptr` point to the first element of the first channel in the
+  // interleaved array.
+  int16_t* ptr = array_interleaved_.data();
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      EXPECT_EQ(*ptr, vec[channel][i]);
+      ++ptr;
+    }
+  }
+  vec.PopBack(array_length());  // Remove more elements than vector size.
+  EXPECT_EQ(0u, vec.Size());
+}
+
+// Test the AssertSize method.
+TEST_P(AudioMultiVectorTest, AssertSize) {
+  AudioMultiVector vec(num_channels_, array_length());
+  EXPECT_EQ(array_length(), vec.Size());
+  // Start with asserting with smaller sizes than already allocated.
+  vec.AssertSize(0);
+  vec.AssertSize(array_length() - 1);
+  // Nothing should have changed.
+  EXPECT_EQ(array_length(), vec.Size());
+  // Assert with one element longer than already allocated.
+  vec.AssertSize(array_length() + 1);
+  // Expect vector to have grown.
+  EXPECT_EQ(array_length() + 1, vec.Size());
+  // Also check the individual AudioVectors.
+  for (size_t channel = 0; channel < vec.Channels(); ++channel) {
+    EXPECT_EQ(array_length() + 1u, vec[channel].Size());
+  }
+}
+
+// Test the PushBack method with another AudioMultiVector as input argument.
+TEST_P(AudioMultiVectorTest, OverwriteAt) {
+  AudioMultiVector vec1(num_channels_);
+  vec1.PushBackInterleaved(array_interleaved_);
+  AudioMultiVector vec2(num_channels_);
+  vec2.Zeros(3);  // 3 zeros in each channel.
+  // Overwrite vec2 at position 5.
+  vec1.OverwriteAt(vec2, 3, 5);
+  // Verify result.
+  // Length remains the same.
+  ASSERT_EQ(array_length(), vec1.Size());
+  int16_t* ptr = array_interleaved_.data();
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      if (i >= 5 && i <= 7) {
+        // Elements 5, 6, 7 should have been replaced with zeros.
+        EXPECT_EQ(0, vec1[channel][i]);
+      } else {
+        EXPECT_EQ(*ptr, vec1[channel][i]);
+      }
+      ++ptr;
+    }
+  }
+}
+
+// Test the CopyChannel method, when the test is instantiated with at least two
+// channels.
+TEST_P(AudioMultiVectorTest, CopyChannel) {
+  if (num_channels_ < 2)
+    return;
+
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  // Create a reference copy.
+  AudioMultiVector ref(num_channels_);
+  ref.PushBack(vec);
+  // Copy from first to last channel.
+  vec.CopyChannel(0, num_channels_ - 1);
+  // Verify that the first and last channels are identical; the others should
+  // be left untouched.
+  for (size_t i = 0; i < array_length(); ++i) {
+    // Verify that all but the last channel are untouched.
+    for (size_t channel = 0; channel < num_channels_ - 1; ++channel) {
+      EXPECT_EQ(ref[channel][i], vec[channel][i]);
+    }
+    // Verify that the last and the first channels are identical.
+    EXPECT_EQ(vec[0][i], vec[num_channels_ - 1][i]);
+  }
+}
+
+TEST_P(AudioMultiVectorTest, PushBackEmptyArray) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved({});
+  EXPECT_TRUE(vec.Empty());
+}
+
+INSTANTIATE_TEST_SUITE_P(TestNumChannels,
+                         AudioMultiVectorTest,
+                         ::testing::Values(static_cast<size_t>(1),
+                                           static_cast<size_t>(2),
+                                           static_cast<size_t>(5)));
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.cc
new file mode 100644
index 0000000000..10e8936447
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.cc
@@ -0,0 +1,381 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/audio_vector.h"
+
+
+#include <algorithm>
+#include <memory>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioVector::AudioVector() : AudioVector(kDefaultInitialSize) {
+  Clear();
+}
+
+AudioVector::AudioVector(size_t initial_size)
+    : array_(new int16_t[initial_size + 1]),
+      capacity_(initial_size + 1),
+      begin_index_(0),
+      end_index_(capacity_ - 1) {
+  memset(array_.get(), 0, capacity_ * sizeof(int16_t));
+}
+
+AudioVector::~AudioVector() = default;
+
+void AudioVector::Clear() {
+  end_index_ = begin_index_ = 0;
+}
+
+void AudioVector::CopyTo(AudioVector* copy_to) const {
+  RTC_DCHECK(copy_to);
+  copy_to->Reserve(Size());
+  CopyTo(Size(), 0, copy_to->array_.get());
+  copy_to->begin_index_ = 0;
+  copy_to->end_index_ = Size();
+}
+
+void AudioVector::CopyTo(size_t length,
+                         size_t position,
+                         int16_t* copy_to) const {
+  if (length == 0)
+    return;
+  length = std::min(length, Size() - position);
+  const size_t copy_index = (begin_index_ + position) % capacity_;
+  const size_t first_chunk_length = std::min(length, capacity_ - copy_index);
+  memcpy(copy_to, &array_[copy_index], first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(&copy_to[first_chunk_length], array_.get(),
+           remaining_length * sizeof(int16_t));
+  }
+}
+
+void AudioVector::PushFront(const AudioVector& prepend_this) {
+  const size_t length = prepend_this.Size();
+  if (length == 0)
+    return;
+
+  // Although the subsequent calling to PushFront does Reserve in it, it is
+  // always more efficient to do a big Reserve first.
+  Reserve(Size() + length);
+
+  const size_t first_chunk_length =
+      std::min(length, prepend_this.capacity_ - prepend_this.begin_index_);
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0)
+    PushFront(prepend_this.array_.get(), remaining_length);
+  PushFront(&prepend_this.array_[prepend_this.begin_index_],
+            first_chunk_length);
+}
+
+void AudioVector::PushFront(const int16_t* prepend_this, size_t length) {
+  if (length == 0)
+    return;
+  Reserve(Size() + length);
+  const size_t first_chunk_length = std::min(length, begin_index_);
+  memcpy(&array_[begin_index_ - first_chunk_length],
+         &prepend_this[length - first_chunk_length],
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(&array_[capacity_ - remaining_length], prepend_this,
+           remaining_length * sizeof(int16_t));
+  }
+  begin_index_ = (begin_index_ + capacity_ - length) % capacity_;
+}
+
+void AudioVector::PushBack(const AudioVector& append_this) {
+  PushBack(append_this, append_this.Size(), 0);
+}
+
+void AudioVector::PushBack(const AudioVector& append_this,
+                           size_t length,
+                           size_t position) {
+  RTC_DCHECK_LE(position, append_this.Size());
+  RTC_DCHECK_LE(length, append_this.Size() - position);
+
+  if (length == 0)
+    return;
+
+  // Although the subsequent calling to PushBack does Reserve in it, it is
+  // always more efficient to do a big Reserve first.
+  Reserve(Size() + length);
+
+  const size_t start_index =
+      (append_this.begin_index_ + position) % append_this.capacity_;
+  const size_t first_chunk_length =
+      std::min(length, append_this.capacity_ - start_index);
+  PushBack(&append_this.array_[start_index], first_chunk_length);
+
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0)
+    PushBack(append_this.array_.get(), remaining_length);
+}
+
+void AudioVector::PushBack(const int16_t* append_this, size_t length) {
+  if (length == 0)
+    return;
+  Reserve(Size() + length);
+  const size_t first_chunk_length = std::min(length, capacity_ - end_index_);
+  memcpy(&array_[end_index_], append_this,
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(array_.get(), &append_this[first_chunk_length],
+           remaining_length * sizeof(int16_t));
+  }
+  end_index_ = (end_index_ + length) % capacity_;
+}
+
+void AudioVector::PopFront(size_t length) {
+  if (length == 0)
+    return;
+  length = std::min(length, Size());
+  begin_index_ = (begin_index_ + length) % capacity_;
+}
+
+void AudioVector::PopBack(size_t length) {
+  if (length == 0)
+    return;
+  // Never remove more than what is in the array.
+  length = std::min(length, Size());
+  end_index_ = (end_index_ + capacity_ - length) % capacity_;
+}
+
+void AudioVector::Extend(size_t extra_length) {
+  if (extra_length == 0)
+    return;
+  InsertZerosByPushBack(extra_length, Size());
+}
+
+void AudioVector::InsertAt(const int16_t* insert_this,
+                           size_t length,
+                           size_t position) {
+  if (length == 0)
+    return;
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  // When inserting to a position closer to the beginning, it is more efficient
+  // to insert by pushing front than to insert by pushing back, since less data
+  // will be moved, vice versa.
+  if (position <= Size() - position) {
+    InsertByPushFront(insert_this, length, position);
+  } else {
+    InsertByPushBack(insert_this, length, position);
+  }
+}
+
+void AudioVector::InsertZerosAt(size_t length, size_t position) {
+  if (length == 0)
+    return;
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  // When inserting to a position closer to the beginning, it is more efficient
+  // to insert by pushing front than to insert by pushing back, since less data
+  // will be moved, vice versa.
+  if (position <= Size() - position) {
+    InsertZerosByPushFront(length, position);
+  } else {
+    InsertZerosByPushBack(length, position);
+  }
+}
+
+void AudioVector::OverwriteAt(const AudioVector& insert_this,
+                              size_t length,
+                              size_t position) {
+  RTC_DCHECK_LE(length, insert_this.Size());
+  if (length == 0)
+    return;
+
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  // Although the subsequent calling to OverwriteAt does Reserve in it, it is
+  // always more efficient to do a big Reserve first.
+  size_t new_size = std::max(Size(), position + length);
+  Reserve(new_size);
+
+  const size_t first_chunk_length =
+      std::min(length, insert_this.capacity_ - insert_this.begin_index_);
+  OverwriteAt(&insert_this.array_[insert_this.begin_index_], first_chunk_length,
+              position);
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    OverwriteAt(insert_this.array_.get(), remaining_length,
+                position + first_chunk_length);
+  }
+}
+
+void AudioVector::OverwriteAt(const int16_t* insert_this,
+                              size_t length,
+                              size_t position) {
+  if (length == 0)
+    return;
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  size_t new_size = std::max(Size(), position + length);
+  Reserve(new_size);
+
+  const size_t overwrite_index = (begin_index_ + position) % capacity_;
+  const size_t first_chunk_length =
+      std::min(length, capacity_ - overwrite_index);
+  memcpy(&array_[overwrite_index], insert_this,
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(array_.get(), &insert_this[first_chunk_length],
+           remaining_length * sizeof(int16_t));
+  }
+
+  end_index_ = (begin_index_ + new_size) % capacity_;
+}
+
+void AudioVector::CrossFade(const AudioVector& append_this,
+                            size_t fade_length) {
+  // Fade length cannot be longer than the current vector or `append_this`.
+  RTC_DCHECK_LE(fade_length, Size());
+  RTC_DCHECK_LE(fade_length, append_this.Size());
+  fade_length = std::min(fade_length, Size());
+  fade_length = std::min(fade_length, append_this.Size());
+  size_t position = Size() - fade_length + begin_index_;
+  // Cross fade the overlapping regions.
+  // `alpha` is the mixing factor in Q14.
+  // TODO(hlundin): Consider skipping +1 in the denominator to produce a
+  // smoother cross-fade, in particular at the end of the fade.
+  int alpha_step = 16384 / (static_cast<int>(fade_length) + 1);
+  int alpha = 16384;
+  for (size_t i = 0; i < fade_length; ++i) {
+    alpha -= alpha_step;
+    array_[(position + i) % capacity_] =
+        (alpha * array_[(position + i) % capacity_] +
+         (16384 - alpha) * append_this[i] + 8192) >>
+        14;
+  }
+  RTC_DCHECK_GE(alpha, 0);  // Verify that the slope was correct.
+  // Append what is left of `append_this`.
+  size_t samples_to_push_back = append_this.Size() - fade_length;
+  if (samples_to_push_back > 0)
+    PushBack(append_this, samples_to_push_back, fade_length);
+}
+
+// Returns the number of elements in this AudioVector.
+size_t AudioVector::Size() const {
+  return (end_index_ + capacity_ - begin_index_) % capacity_;
+}
+
+// Returns true if this AudioVector is empty.
+bool AudioVector::Empty() const {
+  return begin_index_ == end_index_;
+}
+
+void AudioVector::Reserve(size_t n) {
+  if (capacity_ > n)
+    return;
+  const size_t length = Size();
+  // Reserve one more sample to remove the ambiguity between empty vector and
+  // full vector. Therefore `begin_index_` == `end_index_` indicates empty
+  // vector, and `begin_index_` == (`end_index_` + 1) % capacity indicates
+  // full vector.
+  std::unique_ptr<int16_t[]> temp_array(new int16_t[n + 1]);
+  CopyTo(length, 0, temp_array.get());
+  array_.swap(temp_array);
+  begin_index_ = 0;
+  end_index_ = length;
+  capacity_ = n + 1;
+}
+
+void AudioVector::InsertByPushBack(const int16_t* insert_this,
+                                   size_t length,
+                                   size_t position) {
+  const size_t move_chunk_length = Size() - position;
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (move_chunk_length > 0) {
+    // TODO(minyue): see if it is possible to avoid copying to a buffer.
+    temp_array.reset(new int16_t[move_chunk_length]);
+    CopyTo(move_chunk_length, position, temp_array.get());
+    PopBack(move_chunk_length);
+  }
+
+  Reserve(Size() + length + move_chunk_length);
+  PushBack(insert_this, length);
+  if (move_chunk_length > 0)
+    PushBack(temp_array.get(), move_chunk_length);
+}
+
+void AudioVector::InsertByPushFront(const int16_t* insert_this,
+                                    size_t length,
+                                    size_t position) {
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (position > 0) {
+    // TODO(minyue): see if it is possible to avoid copying to a buffer.
+    temp_array.reset(new int16_t[position]);
+    CopyTo(position, 0, temp_array.get());
+    PopFront(position);
+  }
+
+  Reserve(Size() + length + position);
+  PushFront(insert_this, length);
+  if (position > 0)
+    PushFront(temp_array.get(), position);
+}
+
+void AudioVector::InsertZerosByPushBack(size_t length, size_t position) {
+  const size_t move_chunk_length = Size() - position;
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (move_chunk_length > 0) {
+    temp_array.reset(new int16_t[move_chunk_length]);
+    CopyTo(move_chunk_length, position, temp_array.get());
+    PopBack(move_chunk_length);
+  }
+
+  Reserve(Size() + length + move_chunk_length);
+
+  const size_t first_zero_chunk_length =
+      std::min(length, capacity_ - end_index_);
+  memset(&array_[end_index_], 0, first_zero_chunk_length * sizeof(int16_t));
+  const size_t remaining_zero_length = length - first_zero_chunk_length;
+  if (remaining_zero_length > 0)
+    memset(array_.get(), 0, remaining_zero_length * sizeof(int16_t));
+  end_index_ = (end_index_ + length) % capacity_;
+
+  if (move_chunk_length > 0)
+    PushBack(temp_array.get(), move_chunk_length);
+}
+
+void AudioVector::InsertZerosByPushFront(size_t length, size_t position) {
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (position > 0) {
+    temp_array.reset(new int16_t[position]);
+    CopyTo(position, 0, temp_array.get());
+    PopFront(position);
+  }
+
+  Reserve(Size() + length + position);
+
+  const size_t first_zero_chunk_length = std::min(length, begin_index_);
+  memset(&array_[begin_index_ - first_zero_chunk_length], 0,
+         first_zero_chunk_length * sizeof(int16_t));
+  const size_t remaining_zero_length = length - first_zero_chunk_length;
+  if (remaining_zero_length > 0)
+    memset(&array_[capacity_ - remaining_zero_length], 0,
+           remaining_zero_length * sizeof(int16_t));
+  begin_index_ = (begin_index_ + capacity_ - length) % capacity_;
+
+  if (position > 0)
+    PushFront(temp_array.get(), position);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.h b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.h
new file mode 100644
index 0000000000..d68f3ec6be
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.h
@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
+
+#include <string.h>
+
+#include <cstdint>
+#include <memory>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class AudioVector {
+ public:
+  // Creates an empty AudioVector.
+  AudioVector();
+
+  // Creates an AudioVector with an initial size.
+  explicit AudioVector(size_t initial_size);
+
+  virtual ~AudioVector();
+
+  AudioVector(const AudioVector&) = delete;
+  AudioVector& operator=(const AudioVector&) = delete;
+
+  // Deletes all values and make the vector empty.
+  virtual void Clear();
+
+  // Copies all values from this vector to `copy_to`. Any contents in `copy_to`
+  // are deleted before the copy operation. After the operation is done,
+  // `copy_to` will be an exact replica of this object.
+  virtual void CopyTo(AudioVector* copy_to) const;
+
+  // Copies `length` values from `position` in this vector to `copy_to`.
+  virtual void CopyTo(size_t length, size_t position, int16_t* copy_to) const;
+
+  // Prepends the contents of AudioVector `prepend_this` to this object. The
+  // length of this object is increased with the length of `prepend_this`.
+  virtual void PushFront(const AudioVector& prepend_this);
+
+  // Same as above, but with an array `prepend_this` with `length` elements as
+  // source.
+  virtual void PushFront(const int16_t* prepend_this, size_t length);
+
+  // Same as PushFront but will append to the end of this object.
+  virtual void PushBack(const AudioVector& append_this);
+
+  // Appends a segment of `append_this` to the end of this object. The segment
+  // starts from `position` and has `length` samples.
+  virtual void PushBack(const AudioVector& append_this,
+                        size_t length,
+                        size_t position);
+
+  // Same as PushFront but will append to the end of this object.
+  virtual void PushBack(const int16_t* append_this, size_t length);
+
+  // Removes `length` elements from the beginning of this object.
+  virtual void PopFront(size_t length);
+
+  // Removes `length` elements from the end of this object.
+  virtual void PopBack(size_t length);
+
+  // Extends this object with `extra_length` elements at the end. The new
+  // elements are initialized to zero.
+  virtual void Extend(size_t extra_length);
+
+  // Inserts `length` elements taken from the array `insert_this` and insert
+  // them at `position`. The length of the AudioVector is increased by `length`.
+  // `position` = 0 means that the new values are prepended to the vector.
+  // `position` = Size() means that the new values are appended to the vector.
+  virtual void InsertAt(const int16_t* insert_this,
+                        size_t length,
+                        size_t position);
+
+  // Like InsertAt, but inserts `length` zero elements at `position`.
+  virtual void InsertZerosAt(size_t length, size_t position);
+
+  // Overwrites `length` elements of this AudioVector starting from `position`
+  // with first values in `AudioVector`. The definition of `position`
+  // is the same as for InsertAt(). If `length` and `position` are selected
+  // such that the new data extends beyond the end of the current AudioVector,
+  // the vector is extended to accommodate the new data.
+  virtual void OverwriteAt(const AudioVector& insert_this,
+                           size_t length,
+                           size_t position);
+
+  // Overwrites `length` elements of this AudioVector with values taken from the
+  // array `insert_this`, starting at `position`. The definition of `position`
+  // is the same as for InsertAt(). If `length` and `position` are selected
+  // such that the new data extends beyond the end of the current AudioVector,
+  // the vector is extended to accommodate the new data.
+  virtual void OverwriteAt(const int16_t* insert_this,
+                           size_t length,
+                           size_t position);
+
+  // Appends `append_this` to the end of the current vector. Lets the two
+  // vectors overlap by `fade_length` samples, and cross-fade linearly in this
+  // region.
+  virtual void CrossFade(const AudioVector& append_this, size_t fade_length);
+
+  // Returns the number of elements in this AudioVector.
+  virtual size_t Size() const;
+
+  // Returns true if this AudioVector is empty.
+  virtual bool Empty() const;
+
+  // Accesses and modifies an element of AudioVector.
+  inline const int16_t& operator[](size_t index) const {
+    return array_[WrapIndex(index, begin_index_, capacity_)];
+  }
+
+  inline int16_t& operator[](size_t index) {
+    return array_[WrapIndex(index, begin_index_, capacity_)];
+  }
+
+ private:
+  static const size_t kDefaultInitialSize = 10;
+
+  // This method is used by the [] operators to calculate an index within the
+  // capacity of the array, but without using the modulo operation (%).
+  static inline size_t WrapIndex(size_t index,
+                                 size_t begin_index,
+                                 size_t capacity) {
+    RTC_DCHECK_LT(index, capacity);
+    RTC_DCHECK_LT(begin_index, capacity);
+    size_t ix = begin_index + index;
+    RTC_DCHECK_GE(ix, index);  // Check for overflow.
+    if (ix >= capacity) {
+      ix -= capacity;
+    }
+    RTC_DCHECK_LT(ix, capacity);
+    return ix;
+  }
+
+  void Reserve(size_t n);
+
+  void InsertByPushBack(const int16_t* insert_this,
+                        size_t length,
+                        size_t position);
+
+  void InsertByPushFront(const int16_t* insert_this,
+                         size_t length,
+                         size_t position);
+
+  void InsertZerosByPushBack(size_t length, size_t position);
+
+  void InsertZerosByPushFront(size_t length, size_t position);
+
+  std::unique_ptr<int16_t[]> array_;
+
+  size_t capacity_;  // Allocated number of samples in the array.
+
+  // The index of the first sample in `array_`, except when
+  // |begin_index_ == end_index_|, which indicates an empty buffer.
+  size_t begin_index_;
+
+  // The index of the sample after the last sample in `array_`.
+  size_t end_index_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
new file mode 100644
index 0000000000..ae9dd88606
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
@@ -0,0 +1,384 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/audio_vector.h"
+
+#include <stdlib.h>
+
+#include <string>
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class AudioVectorTest : public ::testing::Test {
+ protected:
+  virtual void SetUp() {
+    // Populate test array.
+    for (size_t i = 0; i < array_length(); ++i) {
+      array_[i] = rtc::checked_cast<int16_t>(i);
+    }
+  }
+
+  size_t array_length() const { return sizeof(array_) / sizeof(array_[0]); }
+
+  int16_t array_[10];
+};
+
+// Create and destroy AudioVector objects, both empty and with a predefined
+// length.
+TEST_F(AudioVectorTest, CreateAndDestroy) {
+  AudioVector vec1;
+  EXPECT_TRUE(vec1.Empty());
+  EXPECT_EQ(0u, vec1.Size());
+
+  size_t initial_size = 17;
+  AudioVector vec2(initial_size);
+  EXPECT_FALSE(vec2.Empty());
+  EXPECT_EQ(initial_size, vec2.Size());
+}
+
+// Test the subscript operator [] for getting and setting.
+TEST_F(AudioVectorTest, SubscriptOperator) {
+  AudioVector vec(array_length());
+  for (size_t i = 0; i < array_length(); ++i) {
+    vec[i] = static_cast<int16_t>(i);
+    const int16_t& value = vec[i];  // Make sure to use the const version.
+    EXPECT_EQ(static_cast<int16_t>(i), value);
+  }
+}
+
+// Test the PushBack method and the CopyFrom method. The Clear method is also
+// invoked.
+TEST_F(AudioVectorTest, PushBackAndCopy) {
+  AudioVector vec;
+  AudioVector vec_copy;
+  vec.PushBack(array_, array_length());
+  vec.CopyTo(&vec_copy);  // Copy from `vec` to `vec_copy`.
+  ASSERT_EQ(array_length(), vec.Size());
+  ASSERT_EQ(array_length(), vec_copy.Size());
+  for (size_t i = 0; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[i]);
+    EXPECT_EQ(array_[i], vec_copy[i]);
+  }
+
+  // Clear `vec` and verify that it is empty.
+  vec.Clear();
+  EXPECT_TRUE(vec.Empty());
+
+  // Now copy the empty vector and verify that the copy becomes empty too.
+  vec.CopyTo(&vec_copy);
+  EXPECT_TRUE(vec_copy.Empty());
+}
+
+// Test the PushBack method with another AudioVector as input argument.
+TEST_F(AudioVectorTest, PushBackVector) {
+  static const size_t kLength = 10;
+  AudioVector vec1(kLength);
+  AudioVector vec2(kLength);
+  // Set the first vector to [0, 1, ..., kLength - 1].
+  // Set the second vector to [kLength, kLength + 1, ..., 2 * kLength - 1].
+  for (size_t i = 0; i < kLength; ++i) {
+    vec1[i] = static_cast<int16_t>(i);
+    vec2[i] = static_cast<int16_t>(i + kLength);
+  }
+  // Append vec2 to the back of vec1.
+  vec1.PushBack(vec2);
+  ASSERT_EQ(2 * kLength, vec1.Size());
+  for (size_t i = 0; i < 2 * kLength; ++i) {
+    EXPECT_EQ(static_cast<int16_t>(i), vec1[i]);
+  }
+}
+
+// Test the PushFront method.
+TEST_F(AudioVectorTest, PushFront) {
+  AudioVector vec;
+  vec.PushFront(array_, array_length());
+  ASSERT_EQ(array_length(), vec.Size());
+  for (size_t i = 0; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[i]);
+  }
+}
+
+// Test the PushFront method with another AudioVector as input argument.
+TEST_F(AudioVectorTest, PushFrontVector) {
+  static const size_t kLength = 10;
+  AudioVector vec1(kLength);
+  AudioVector vec2(kLength);
+  // Set the first vector to [0, 1, ..., kLength - 1].
+  // Set the second vector to [kLength, kLength + 1, ..., 2 * kLength - 1].
+  for (size_t i = 0; i < kLength; ++i) {
+    vec1[i] = static_cast<int16_t>(i);
+    vec2[i] = static_cast<int16_t>(i + kLength);
+  }
+  // Prepend vec1 to the front of vec2.
+  vec2.PushFront(vec1);
+  ASSERT_EQ(2 * kLength, vec2.Size());
+  for (size_t i = 0; i < 2 * kLength; ++i) {
+    EXPECT_EQ(static_cast<int16_t>(i), vec2[i]);
+  }
+}
+
+// Test the PopFront method.
+TEST_F(AudioVectorTest, PopFront) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  vec.PopFront(1);  // Remove one element.
+  EXPECT_EQ(array_length() - 1u, vec.Size());
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    EXPECT_EQ(static_cast<int16_t>(i + 1), vec[i]);
+  }
+  vec.PopFront(array_length());  // Remove more elements than vector size.
+  EXPECT_EQ(0u, vec.Size());
+}
+
+// Test the PopBack method.
+TEST_F(AudioVectorTest, PopBack) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  vec.PopBack(1);  // Remove one element.
+  EXPECT_EQ(array_length() - 1u, vec.Size());
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    EXPECT_EQ(static_cast<int16_t>(i), vec[i]);
+  }
+  vec.PopBack(array_length());  // Remove more elements than vector size.
+  EXPECT_EQ(0u, vec.Size());
+}
+
+// Test the Extend method.
+TEST_F(AudioVectorTest, Extend) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  vec.Extend(5);  // Extend with 5 elements, which should all be zeros.
+  ASSERT_EQ(array_length() + 5u, vec.Size());
+  // Verify that all are zero.
+  for (size_t i = array_length(); i < array_length() + 5; ++i) {
+    EXPECT_EQ(0, vec[i]);
+  }
+}
+
+// Test the InsertAt method with an insert position in the middle of the vector.
+TEST_F(AudioVectorTest, InsertAt) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position = 5;
+  vec.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {0, 1, ..., `insert_position` - 1, 100, 101, ..., 100 + kNewLength - 1,
+  //  `insert_position`, `insert_position` + 1, ..., kLength - 1}.
+  size_t pos = 0;
+  for (int i = 0; i < insert_position; ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+  for (size_t i = insert_position; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+}
+
+// Test the InsertZerosAt method with an insert position in the middle of the
+// vector. Use the InsertAt method as reference.
+TEST_F(AudioVectorTest, InsertZerosAt) {
+  AudioVector vec;
+  AudioVector vec_ref;
+  vec.PushBack(array_, array_length());
+  vec_ref.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int insert_position = 5;
+  vec.InsertZerosAt(kNewLength, insert_position);
+  int16_t new_array[kNewLength] = {0};  // All zero elements.
+  vec_ref.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vectors are identical.
+  ASSERT_EQ(vec_ref.Size(), vec.Size());
+  for (size_t i = 0; i < vec.Size(); ++i) {
+    EXPECT_EQ(vec_ref[i], vec[i]);
+  }
+}
+
+// Test the InsertAt method with an insert position at the start of the vector.
+TEST_F(AudioVectorTest, InsertAtBeginning) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position = 0;
+  vec.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {100, 101, ..., 100 + kNewLength - 1,
+  //  0, 1, ..., kLength - 1}.
+  size_t pos = 0;
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+  for (size_t i = insert_position; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+}
+
+// Test the InsertAt method with an insert position at the end of the vector.
+TEST_F(AudioVectorTest, InsertAtEnd) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position = rtc::checked_cast<int>(array_length());
+  vec.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {0, 1, ..., kLength - 1, 100, 101, ..., 100 + kNewLength - 1 }.
+  size_t pos = 0;
+  for (size_t i = 0; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+}
+
+// Test the InsertAt method with an insert position beyond the end of the
+// vector. Verify that a position beyond the end of the vector does not lead to
+// an error. The expected outcome is the same as if the vector end was used as
+// input position. That is, the input position should be capped at the maximum
+// allowed value.
+TEST_F(AudioVectorTest, InsertBeyondEnd) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position =
+      rtc::checked_cast<int>(array_length() + 10);  // Too large.
+  vec.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {0, 1, ..., kLength - 1, 100, 101, ..., 100 + kNewLength - 1 }.
+  size_t pos = 0;
+  for (size_t i = 0; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+}
+
+// Test the OverwriteAt method with a position such that all of the new values
+// fit within the old vector.
+TEST_F(AudioVectorTest, OverwriteAt) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  size_t insert_position = 2;
+  vec.OverwriteAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {0, ..., `insert_position` - 1, 100, 101, ..., 100 + kNewLength - 1,
+  //  `insert_position`, `insert_position` + 1, ..., kLength - 1}.
+  size_t pos = 0;
+  for (pos = 0; pos < insert_position; ++pos) {
+    EXPECT_EQ(array_[pos], vec[pos]);
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+  for (; pos < array_length(); ++pos) {
+    EXPECT_EQ(array_[pos], vec[pos]);
+  }
+}
+
+// Test the OverwriteAt method with a position such that some of the new values
+// extend beyond the end of the current vector. This is valid, and the vector is
+// expected to expand to accommodate the new values.
+TEST_F(AudioVectorTest, OverwriteBeyondEnd) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position = rtc::checked_cast<int>(array_length() - 2);
+  vec.OverwriteAt(new_array, kNewLength, insert_position);
+  ASSERT_EQ(array_length() - 2u + kNewLength, vec.Size());
+  // Verify that the vector looks as follows:
+  // {0, ..., `insert_position` - 1, 100, 101, ..., 100 + kNewLength - 1,
+  //  `insert_position`, `insert_position` + 1, ..., kLength - 1}.
+  int pos = 0;
+  for (pos = 0; pos < insert_position; ++pos) {
+    EXPECT_EQ(array_[pos], vec[pos]);
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+  // Verify that we checked to the end of `vec`.
+  EXPECT_EQ(vec.Size(), static_cast<size_t>(pos));
+}
+
+TEST_F(AudioVectorTest, CrossFade) {
+  static const size_t kLength = 100;
+  static const size_t kFadeLength = 10;
+  AudioVector vec1(kLength);
+  AudioVector vec2(kLength);
+  // Set all vector elements to 0 in `vec1` and 100 in `vec2`.
+  for (size_t i = 0; i < kLength; ++i) {
+    vec1[i] = 0;
+    vec2[i] = 100;
+  }
+  vec1.CrossFade(vec2, kFadeLength);
+  ASSERT_EQ(2 * kLength - kFadeLength, vec1.Size());
+  // First part untouched.
+  for (size_t i = 0; i < kLength - kFadeLength; ++i) {
+    EXPECT_EQ(0, vec1[i]);
+  }
+  // Check mixing zone.
+  for (size_t i = 0; i < kFadeLength; ++i) {
+    EXPECT_NEAR((i + 1) * 100 / (kFadeLength + 1),
+                vec1[kLength - kFadeLength + i], 1);
+  }
+  // Second part untouched.
+  for (size_t i = kLength; i < vec1.Size(); ++i) {
+    EXPECT_EQ(100, vec1[i]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc
new file mode 100644
index 0000000000..2c95d3b390
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc
@@ -0,0 +1,309 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/background_noise.h"
+
+#include <string.h>  // memcpy
+
+#include <algorithm>  // min, max
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/post_decode_vad.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kMaxSampleRate = 48000;
+
+}  // namespace
+
+// static
+constexpr size_t BackgroundNoise::kMaxLpcOrder;
+
+BackgroundNoise::BackgroundNoise(size_t num_channels)
+    : num_channels_(num_channels),
+      channel_parameters_(new ChannelParameters[num_channels_]) {
+  Reset();
+}
+
+BackgroundNoise::~BackgroundNoise() {}
+
+void BackgroundNoise::Reset() {
+  initialized_ = false;
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    channel_parameters_[channel].Reset();
+  }
+}
+
+bool BackgroundNoise::Update(const AudioMultiVector& input,
+                             const PostDecodeVad& vad) {
+  bool filter_params_saved = false;
+  if (vad.running() && vad.active_speech()) {
+    // Do not update the background noise parameters if we know that the signal
+    // is active speech.
+    return filter_params_saved;
+  }
+
+  int32_t auto_correlation[kMaxLpcOrder + 1];
+  int16_t fiter_output[kMaxLpcOrder + kResidualLength];
+  int16_t reflection_coefficients[kMaxLpcOrder];
+  int16_t lpc_coefficients[kMaxLpcOrder + 1];
+
+  for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
+    ChannelParameters& parameters = channel_parameters_[channel_ix];
+    int16_t temp_signal_array[kVecLen + kMaxLpcOrder] = {0};
+    int16_t* temp_signal = &temp_signal_array[kMaxLpcOrder];
+    RTC_DCHECK_GE(input.Size(), kVecLen);
+    input[channel_ix].CopyTo(kVecLen, input.Size() - kVecLen, temp_signal);
+    int32_t sample_energy =
+        CalculateAutoCorrelation(temp_signal, kVecLen, auto_correlation);
+
+    if ((!vad.running() &&
+         sample_energy < parameters.energy_update_threshold) ||
+        (vad.running() && !vad.active_speech())) {
+      // Generate LPC coefficients.
+      if (auto_correlation[0] <= 0) {
+        // Center value in auto-correlation is not positive. Do not update.
+        return filter_params_saved;
+      }
+
+      // Regardless of whether the filter is actually updated or not,
+      // update energy threshold levels, since we have in fact observed
+      // a low energy signal.
+      if (sample_energy < parameters.energy_update_threshold) {
+        // Never go under 1.0 in average sample energy.
+        parameters.energy_update_threshold = std::max(sample_energy, 1);
+        parameters.low_energy_update_threshold = 0;
+      }
+
+      // Only update BGN if filter is stable, i.e., if return value from
+      // Levinson-Durbin function is 1.
+      if (WebRtcSpl_LevinsonDurbin(auto_correlation, lpc_coefficients,
+                                   reflection_coefficients,
+                                   kMaxLpcOrder) != 1) {
+        return filter_params_saved;
+      }
+
+      // Generate the CNG gain factor by looking at the energy of the residual.
+      WebRtcSpl_FilterMAFastQ12(temp_signal + kVecLen - kResidualLength,
+                                fiter_output, lpc_coefficients,
+                                kMaxLpcOrder + 1, kResidualLength);
+      int32_t residual_energy = WebRtcSpl_DotProductWithScale(
+          fiter_output, fiter_output, kResidualLength, 0);
+
+      // Check spectral flatness.
+      // Comparing the residual variance with the input signal variance tells
+      // if the spectrum is flat or not.
+      // If 5 * residual_energy >= 16 * sample_energy, the spectrum is flat
+      // enough.  Also ensure that the energy is non-zero.
+      if ((sample_energy > 0) &&
+          (int64_t{5} * residual_energy >= int64_t{16} * sample_energy)) {
+        // Spectrum is flat enough; save filter parameters.
+        // `temp_signal` + `kVecLen` - `kMaxLpcOrder` points at the first of the
+        // `kMaxLpcOrder` samples in the residual signal, which will form the
+        // filter state for the next noise generation.
+        SaveParameters(channel_ix, lpc_coefficients,
+                       temp_signal + kVecLen - kMaxLpcOrder, sample_energy,
+                       residual_energy);
+        filter_params_saved = true;
+      }
+    } else {
+      // Will only happen if post-decode VAD is disabled and `sample_energy` is
+      // not low enough. Increase the threshold for update so that it increases
+      // by a factor 4 in 4 seconds.
+      IncrementEnergyThreshold(channel_ix, sample_energy);
+    }
+  }
+  return filter_params_saved;
+}
+
+void BackgroundNoise::GenerateBackgroundNoise(
+    rtc::ArrayView<const int16_t> random_vector,
+    size_t channel,
+    int mute_slope,
+    bool too_many_expands,
+    size_t num_noise_samples,
+    int16_t* buffer) {
+  constexpr size_t kNoiseLpcOrder = kMaxLpcOrder;
+  int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125];
+  RTC_DCHECK_LE(num_noise_samples, (kMaxSampleRate / 8000 * 125));
+  RTC_DCHECK_GE(random_vector.size(), num_noise_samples);
+  int16_t* noise_samples = &buffer[kNoiseLpcOrder];
+  if (initialized()) {
+    // Use background noise parameters.
+    memcpy(noise_samples - kNoiseLpcOrder, FilterState(channel),
+           sizeof(int16_t) * kNoiseLpcOrder);
+
+    int dc_offset = 0;
+    if (ScaleShift(channel) > 1) {
+      dc_offset = 1 << (ScaleShift(channel) - 1);
+    }
+
+    // Scale random vector to correct energy level.
+    WebRtcSpl_AffineTransformVector(scaled_random_vector, random_vector.data(),
+                                    Scale(channel), dc_offset,
+                                    ScaleShift(channel), num_noise_samples);
+
+    WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_samples,
+                              Filter(channel), kNoiseLpcOrder + 1,
+                              num_noise_samples);
+
+    SetFilterState(
+        channel,
+        {&(noise_samples[num_noise_samples - kNoiseLpcOrder]), kNoiseLpcOrder});
+
+    // Unmute the background noise.
+    int16_t bgn_mute_factor = MuteFactor(channel);
+    if (bgn_mute_factor < 16384) {
+      WebRtcSpl_AffineTransformVector(noise_samples, noise_samples,
+                                      bgn_mute_factor, 8192, 14,
+                                      num_noise_samples);
+    }
+    // Update mute_factor in BackgroundNoise class.
+    SetMuteFactor(channel, bgn_mute_factor);
+  } else {
+    // BGN parameters have not been initialized; use zero noise.
+    memset(noise_samples, 0, sizeof(int16_t) * num_noise_samples);
+  }
+}
+
+int32_t BackgroundNoise::Energy(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].energy;
+}
+
+void BackgroundNoise::SetMuteFactor(size_t channel, int16_t value) {
+  RTC_DCHECK_LT(channel, num_channels_);
+  channel_parameters_[channel].mute_factor = value;
+}
+
+int16_t BackgroundNoise::MuteFactor(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].mute_factor;
+}
+
+const int16_t* BackgroundNoise::Filter(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].filter;
+}
+
+const int16_t* BackgroundNoise::FilterState(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].filter_state;
+}
+
+void BackgroundNoise::SetFilterState(size_t channel,
+                                     rtc::ArrayView<const int16_t> input) {
+  RTC_DCHECK_LT(channel, num_channels_);
+  size_t length = std::min(input.size(), kMaxLpcOrder);
+  memcpy(channel_parameters_[channel].filter_state, input.data(),
+         length * sizeof(int16_t));
+}
+
+int16_t BackgroundNoise::Scale(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].scale;
+}
+int16_t BackgroundNoise::ScaleShift(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].scale_shift;
+}
+
+int32_t BackgroundNoise::CalculateAutoCorrelation(
+    const int16_t* signal,
+    size_t length,
+    int32_t* auto_correlation) const {
+  static const int kCorrelationStep = -1;
+  const int correlation_scale =
+      CrossCorrelationWithAutoShift(signal, signal, length, kMaxLpcOrder + 1,
+                                    kCorrelationStep, auto_correlation);
+
+  // Number of shifts to normalize energy to energy/sample.
+  int energy_sample_shift = kLogVecLen - correlation_scale;
+  return auto_correlation[0] >> energy_sample_shift;
+}
+
+void BackgroundNoise::IncrementEnergyThreshold(size_t channel,
+                                               int32_t sample_energy) {
+  // TODO(hlundin): Simplify the below threshold update. What this code
+  // does is simply "threshold += (increment * threshold) >> 16", but due
+  // to the limited-width operations, it is not exactly the same. The
+  // difference should be inaudible, but bit-exactness would not be
+  // maintained.
+  RTC_DCHECK_LT(channel, num_channels_);
+  ChannelParameters& parameters = channel_parameters_[channel];
+  int32_t temp_energy =
+      (kThresholdIncrement * parameters.low_energy_update_threshold) >> 16;
+  temp_energy +=
+      kThresholdIncrement * (parameters.energy_update_threshold & 0xFF);
+  temp_energy +=
+      (kThresholdIncrement * ((parameters.energy_update_threshold >> 8) & 0xFF))
+      << 8;
+  parameters.low_energy_update_threshold += temp_energy;
+
+  parameters.energy_update_threshold +=
+      kThresholdIncrement * (parameters.energy_update_threshold >> 16);
+  parameters.energy_update_threshold +=
+      parameters.low_energy_update_threshold >> 16;
+  parameters.low_energy_update_threshold =
+      parameters.low_energy_update_threshold & 0x0FFFF;
+
+  // Update maximum energy.
+  // Decrease by a factor 1/1024 each time.
+  parameters.max_energy = parameters.max_energy - (parameters.max_energy >> 10);
+  if (sample_energy > parameters.max_energy) {
+    parameters.max_energy = sample_energy;
+  }
+
+  // Set `energy_update_threshold` to no less than 60 dB lower than
+  // `max_energy_`. Adding 524288 assures proper rounding.
+  int32_t energy_update_threshold = (parameters.max_energy + 524288) >> 20;
+  if (energy_update_threshold > parameters.energy_update_threshold) {
+    parameters.energy_update_threshold = energy_update_threshold;
+  }
+}
+
+void BackgroundNoise::SaveParameters(size_t channel,
+                                     const int16_t* lpc_coefficients,
+                                     const int16_t* filter_state,
+                                     int32_t sample_energy,
+                                     int32_t residual_energy) {
+  RTC_DCHECK_LT(channel, num_channels_);
+  ChannelParameters& parameters = channel_parameters_[channel];
+  memcpy(parameters.filter, lpc_coefficients,
+         (kMaxLpcOrder + 1) * sizeof(int16_t));
+  memcpy(parameters.filter_state, filter_state, kMaxLpcOrder * sizeof(int16_t));
+  // Save energy level and update energy threshold levels.
+  // Never get under 1.0 in average sample energy.
+  parameters.energy = std::max(sample_energy, 1);
+  parameters.energy_update_threshold = parameters.energy;
+  parameters.low_energy_update_threshold = 0;
+
+  // Normalize residual_energy to 29 or 30 bits before sqrt.
+  int16_t norm_shift = WebRtcSpl_NormW32(residual_energy) - 1;
+  if (norm_shift & 0x1) {
+    norm_shift -= 1;  // Even number of shifts required.
+  }
+  residual_energy = WEBRTC_SPL_SHIFT_W32(residual_energy, norm_shift);
+
+  // Calculate scale and shift factor.
+  parameters.scale = static_cast<int16_t>(WebRtcSpl_SqrtFloor(residual_energy));
+  // Add 13 to the `scale_shift_`, since the random numbers table is in
+  // Q13.
+  // TODO(hlundin): Move the "13" to where the `scale_shift_` is used?
+  parameters.scale_shift =
+      static_cast<int16_t>(13 + ((kLogResidualLength + norm_shift) / 2));
+
+  initialized_ = true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h
new file mode 100644
index 0000000000..8e6d5890a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
+#define MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
+
+#include <string.h>  // size_t
+
+#include <memory>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class AudioMultiVector;
+class PostDecodeVad;
+
+// This class handles estimation of background noise parameters.
+class BackgroundNoise {
+ public:
+  // TODO(hlundin): For 48 kHz support, increase kMaxLpcOrder to 10.
+  // Will work anyway, but probably sound a little worse.
+  static constexpr size_t kMaxLpcOrder = 8;  // 32000 / 8000 + 4.
+
+  explicit BackgroundNoise(size_t num_channels);
+  virtual ~BackgroundNoise();
+
+  BackgroundNoise(const BackgroundNoise&) = delete;
+  BackgroundNoise& operator=(const BackgroundNoise&) = delete;
+
+  void Reset();
+
+  // Updates the parameter estimates based on the signal currently in the
+  // `sync_buffer`, and on the latest decision in `vad` if it is running.
+  // Returns true if the filter parameters are updated.
+  bool Update(const AudioMultiVector& sync_buffer, const PostDecodeVad& vad);
+
+  // Generates background noise given a random vector and writes the output to
+  // `buffer`.
+  void GenerateBackgroundNoise(rtc::ArrayView<const int16_t> random_vector,
+                               size_t channel,
+                               int mute_slope,
+                               bool too_many_expands,
+                               size_t num_noise_samples,
+                               int16_t* buffer);
+
+  // Returns `energy_` for `channel`.
+  int32_t Energy(size_t channel) const;
+
+  // Sets the value of `mute_factor_` for `channel` to `value`.
+  void SetMuteFactor(size_t channel, int16_t value);
+
+  // Returns `mute_factor_` for `channel`.
+  int16_t MuteFactor(size_t channel) const;
+
+  // Returns a pointer to `filter_` for `channel`.
+  const int16_t* Filter(size_t channel) const;
+
+  // Returns a pointer to `filter_state_` for `channel`.
+  const int16_t* FilterState(size_t channel) const;
+
+  // Copies `input` to the filter state. Will not copy more than `kMaxLpcOrder`
+  // elements.
+  void SetFilterState(size_t channel, rtc::ArrayView<const int16_t> input);
+
+  // Returns `scale_` for `channel`.
+  int16_t Scale(size_t channel) const;
+
+  // Returns `scale_shift_` for `channel`.
+  int16_t ScaleShift(size_t channel) const;
+
+  // Accessors.
+  bool initialized() const { return initialized_; }
+
+ private:
+  static const int kThresholdIncrement = 229;  // 0.0035 in Q16.
+  static const size_t kVecLen = 256;
+  static const int kLogVecLen = 8;  // log2(kVecLen).
+  static const size_t kResidualLength = 64;
+  static const int16_t kLogResidualLength = 6;  // log2(kResidualLength)
+
+  struct ChannelParameters {
+    // Constructor.
+    ChannelParameters() { Reset(); }
+
+    void Reset() {
+      energy = 2500;
+      max_energy = 0;
+      energy_update_threshold = 500000;
+      low_energy_update_threshold = 0;
+      memset(filter_state, 0, sizeof(filter_state));
+      memset(filter, 0, sizeof(filter));
+      filter[0] = 4096;
+      mute_factor = 0;
+      scale = 20000;
+      scale_shift = 24;
+    }
+
+    int32_t energy;
+    int32_t max_energy;
+    int32_t energy_update_threshold;
+    int32_t low_energy_update_threshold;
+    int16_t filter_state[kMaxLpcOrder];
+    int16_t filter[kMaxLpcOrder + 1];
+    int16_t mute_factor;
+    int16_t scale;
+    int16_t scale_shift;
+  };
+
+  int32_t CalculateAutoCorrelation(const int16_t* signal,
+                                   size_t length,
+                                   int32_t* auto_correlation) const;
+
+  // Increments the energy threshold by a factor 1 + `kThresholdIncrement`.
+  void IncrementEnergyThreshold(size_t channel, int32_t sample_energy);
+
+  // Updates the filter parameters.
+  void SaveParameters(size_t channel,
+                      const int16_t* lpc_coefficients,
+                      const int16_t* filter_state,
+                      int32_t sample_energy,
+                      int32_t residual_energy);
+
+  size_t num_channels_;
+  std::unique_ptr<ChannelParameters[]> channel_parameters_;
+  bool initialized_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise_unittest.cc
new file mode 100644
index 0000000000..e32492f57e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise_unittest.cc
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for BackgroundNoise class.
+
+#include "modules/audio_coding/neteq/background_noise.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(BackgroundNoise, CreateAndDestroy) {
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc
new file mode 100644
index 0000000000..2c42d0d13f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+
+#include <stdint.h>
+
+#include <algorithm>
+
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+BufferLevelFilter::BufferLevelFilter() {
+  Reset();
+}
+
+void BufferLevelFilter::Reset() {
+  filtered_current_level_ = 0;
+  level_factor_ = 253;
+}
+
+void BufferLevelFilter::Update(size_t buffer_size_samples,
+                               int time_stretched_samples) {
+  // Filter:
+  // `filtered_current_level_` = `level_factor_` * `filtered_current_level_` +
+  //                            (1 - `level_factor_`) * `buffer_size_samples`
+  // `level_factor_` and `filtered_current_level_` are in Q8.
+  // `buffer_size_samples` is in Q0.
+  const int64_t filtered_current_level =
+      (level_factor_ * int64_t{filtered_current_level_} >> 8) +
+      (256 - level_factor_) * rtc::dchecked_cast<int64_t>(buffer_size_samples);
+
+  // Account for time-scale operations (accelerate and pre-emptive expand) and
+  // make sure that the filtered value remains non-negative.
+  filtered_current_level_ = rtc::saturated_cast<int>(std::max<int64_t>(
+      0, filtered_current_level - int64_t{time_stretched_samples} * (1 << 8)));
+}
+
+void BufferLevelFilter::SetFilteredBufferLevel(int buffer_size_samples) {
+  filtered_current_level_ =
+      rtc::saturated_cast<int>(int64_t{buffer_size_samples} * 256);
+}
+
+void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level_ms) {
+  if (target_buffer_level_ms <= 20) {
+    level_factor_ = 251;
+  } else if (target_buffer_level_ms <= 60) {
+    level_factor_ = 252;
+  } else if (target_buffer_level_ms <= 140) {
+    level_factor_ = 253;
+  } else {
+    level_factor_ = 254;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.h b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.h
new file mode 100644
index 0000000000..ced36da9c2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+class BufferLevelFilter {
+ public:
+  BufferLevelFilter();
+  virtual ~BufferLevelFilter() {}
+
+  BufferLevelFilter(const BufferLevelFilter&) = delete;
+  BufferLevelFilter& operator=(const BufferLevelFilter&) = delete;
+
+  virtual void Reset();
+
+  // Updates the filter. Current buffer size is `buffer_size_samples`.
+  // `time_stretched_samples` is subtracted from the filtered value (thus
+  // bypassing the filter operation).
+  virtual void Update(size_t buffer_size_samples, int time_stretched_samples);
+
+  // Set the filtered buffer level to a particular value directly. This should
+  // only be used in case of large changes in buffer size, such as buffer
+  // flushes.
+  virtual void SetFilteredBufferLevel(int buffer_size_samples);
+
+  // The target level is used to select the appropriate filter coefficient.
+  virtual void SetTargetBufferLevel(int target_buffer_level_ms);
+
+  // Returns filtered current level in number of samples.
+  virtual int filtered_current_level() const {
+    // Round to nearest whole sample.
+    return (int64_t{filtered_current_level_} + (1 << 7)) >> 8;
+  }
+
+ private:
+  int level_factor_;  // Filter factor for the buffer level filter in Q8.
+  int filtered_current_level_;  // Filtered current buffer level in Q8.
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
new file mode 100644
index 0000000000..6773e96f58
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for BufferLevelFilter class.
+
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+
+#include <math.h>  // Access to pow function.
+
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(BufferLevelFilter, CreateAndDestroy) {
+  BufferLevelFilter* filter = new BufferLevelFilter();
+  EXPECT_EQ(0, filter->filtered_current_level());
+  delete filter;
+}
+
+TEST(BufferLevelFilter, ConvergenceTest) {
+  BufferLevelFilter filter;
+  for (int times = 10; times <= 50; times += 10) {
+    for (int value = 100; value <= 200; value += 10) {
+      filter.Reset();
+      filter.SetTargetBufferLevel(20);  // Makes filter coefficient 251/256.
+      rtc::StringBuilder ss;
+      ss << "times = " << times << ", value = " << value;
+      SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+      for (int i = 0; i < times; ++i) {
+        filter.Update(value, 0 /* time_stretched_samples */);
+      }
+      // Expect the filtered value to be (theoretically)
+      // (1 - (251/256) ^ `times`) * `value`.
+      double expected_value_double = (1 - pow(251.0 / 256.0, times)) * value;
+      int expected_value = static_cast<int>(expected_value_double);
+
+      // The actual value may differ slightly from the expected value due to
+      // intermediate-stage rounding errors in the filter implementation.
+      // This is why we have to use EXPECT_NEAR with a tolerance of +/-1.
+      EXPECT_NEAR(expected_value, filter.filtered_current_level(), 1);
+    }
+  }
+}
+
+// Verify that target buffer level impacts on the filter convergence.
+TEST(BufferLevelFilter, FilterFactor) {
+  BufferLevelFilter filter;
+  // Update 10 times with value 100.
+  const int kTimes = 10;
+  const int kValue = 100;
+
+  filter.SetTargetBufferLevel(60);  // Makes filter coefficient 252/256.
+  for (int i = 0; i < kTimes; ++i) {
+    filter.Update(kValue, 0 /* time_stretched_samples */);
+  }
+  // Expect the filtered value to be
+  // (1 - (252/256) ^ `kTimes`) * `kValue`.
+  int expected_value = 15;
+  EXPECT_EQ(expected_value, filter.filtered_current_level());
+
+  filter.Reset();
+  filter.SetTargetBufferLevel(140);  // Makes filter coefficient 253/256.
+  for (int i = 0; i < kTimes; ++i) {
+    filter.Update(kValue, 0 /* time_stretched_samples */);
+  }
+  // Expect the filtered value to be
+  // (1 - (253/256) ^ `kTimes`) * `kValue`.
+  expected_value = 11;
+  EXPECT_EQ(expected_value, filter.filtered_current_level());
+
+  filter.Reset();
+  filter.SetTargetBufferLevel(160);  // Makes filter coefficient 254/256.
+  for (int i = 0; i < kTimes; ++i) {
+    filter.Update(kValue, 0 /* time_stretched_samples */);
+  }
+  // Expect the filtered value to be
+  // (1 - (254/256) ^ `kTimes`) * `kValue`.
+  expected_value = 8;
+  EXPECT_EQ(expected_value, filter.filtered_current_level());
+}
+
+TEST(BufferLevelFilter, TimeStretchedSamples) {
+  BufferLevelFilter filter;
+  filter.SetTargetBufferLevel(20);  // Makes filter coefficient 251/256.
+  // Update 10 times with value 100.
+  const int kTimes = 10;
+  const int kValue = 100;
+  const int kTimeStretchedSamples = 3;
+  for (int i = 0; i < kTimes; ++i) {
+    filter.Update(kValue, 0);
+  }
+  // Expect the filtered value to be
+  // (1 - (251/256) ^ `kTimes`) * `kValue`.
+  const int kExpectedValue = 18;
+  EXPECT_EQ(kExpectedValue, filter.filtered_current_level());
+
+  // Update filter again, now with non-zero value for packet length.
+  // Set the current filtered value to be the input, in order to isolate the
+  // impact of `kTimeStretchedSamples`.
+  filter.Update(filter.filtered_current_level(), kTimeStretchedSamples);
+  EXPECT_EQ(kExpectedValue - kTimeStretchedSamples,
+            filter.filtered_current_level());
+  // Try negative value and verify that we come back to the previous result.
+  filter.Update(filter.filtered_current_level(), -kTimeStretchedSamples);
+  EXPECT_EQ(kExpectedValue, filter.filtered_current_level());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.cc b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.cc
new file mode 100644
index 0000000000..a2ce888f45
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.cc
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/comfort_noise.h"
+
+
+#include <cstdint>
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+void ComfortNoise::Reset() {
+  first_call_ = true;
+}
+
+int ComfortNoise::UpdateParameters(const Packet& packet) {
+  // Get comfort noise decoder.
+  if (decoder_database_->SetActiveCngDecoder(packet.payload_type) != kOK) {
+    return kUnknownPayloadType;
+  }
+  ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+  RTC_DCHECK(cng_decoder);
+  cng_decoder->UpdateSid(packet.payload);
+  return kOK;
+}
+
+int ComfortNoise::Generate(size_t requested_length, AudioMultiVector* output) {
+  // TODO(hlundin): Change to an enumerator and skip assert.
+  RTC_DCHECK(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
+             fs_hz_ == 48000);
+  // Not adapted for multi-channel yet.
+  if (output->Channels() != 1) {
+    RTC_LOG(LS_ERROR) << "No multi-channel support";
+    return kMultiChannelNotSupported;
+  }
+
+  size_t number_of_samples = requested_length;
+  bool new_period = false;
+  if (first_call_) {
+    // Generate noise and overlap slightly with old data.
+    number_of_samples = requested_length + overlap_length_;
+    new_period = true;
+  }
+  output->AssertSize(number_of_samples);
+  // Get the decoder from the database.
+  ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+  if (!cng_decoder) {
+    RTC_LOG(LS_ERROR) << "Unknwown payload type";
+    return kUnknownPayloadType;
+  }
+
+  std::unique_ptr<int16_t[]> temp(new int16_t[number_of_samples]);
+  if (!cng_decoder->Generate(
+          rtc::ArrayView<int16_t>(temp.get(), number_of_samples), new_period)) {
+    // Error returned.
+    output->Zeros(requested_length);
+    RTC_LOG(LS_ERROR)
+        << "ComfortNoiseDecoder::Genererate failed to generate comfort noise";
+    return kInternalError;
+  }
+  (*output)[0].OverwriteAt(temp.get(), number_of_samples, 0);
+
+  if (first_call_) {
+    // Set tapering window parameters. Values are in Q15.
+    int16_t muting_window;              // Mixing factor for overlap data.
+    int16_t muting_window_increment;    // Mixing factor increment (negative).
+    int16_t unmuting_window;            // Mixing factor for comfort noise.
+    int16_t unmuting_window_increment;  // Mixing factor increment.
+    if (fs_hz_ == 8000) {
+      muting_window = DspHelper::kMuteFactorStart8kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement8kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart8kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement8kHz;
+    } else if (fs_hz_ == 16000) {
+      muting_window = DspHelper::kMuteFactorStart16kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement16kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart16kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement16kHz;
+    } else if (fs_hz_ == 32000) {
+      muting_window = DspHelper::kMuteFactorStart32kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement32kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart32kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement32kHz;
+    } else {  // fs_hz_ == 48000
+      muting_window = DspHelper::kMuteFactorStart48kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement48kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart48kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement48kHz;
+    }
+
+    // Do overlap-add between new vector and overlap.
+    size_t start_ix = sync_buffer_->Size() - overlap_length_;
+    for (size_t i = 0; i < overlap_length_; i++) {
+      /* overlapVec[i] = WinMute * overlapVec[i] + WinUnMute * outData[i] */
+      // The expression (*output)[0][i] is the i-th element in the first
+      // channel.
+      (*sync_buffer_)[0][start_ix + i] =
+          (((*sync_buffer_)[0][start_ix + i] * muting_window) +
+           ((*output)[0][i] * unmuting_window) + 16384) >>
+          15;
+      muting_window += muting_window_increment;
+      unmuting_window += unmuting_window_increment;
+    }
+    // Remove `overlap_length_` samples from the front of `output` since they
+    // were mixed into `sync_buffer_` above.
+    output->PopFront(overlap_length_);
+  }
+  first_call_ = false;
+  return kOK;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.h b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.h
new file mode 100644
index 0000000000..31fcee31d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
+#define MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+
+// Forward declarations.
+class AudioMultiVector;
+class DecoderDatabase;
+class SyncBuffer;
+struct Packet;
+
+// This class acts as an interface to the CNG generator.
+class ComfortNoise {
+ public:
+  enum ReturnCodes {
+    kOK = 0,
+    kUnknownPayloadType,
+    kInternalError,
+    kMultiChannelNotSupported
+  };
+
+  ComfortNoise(int fs_hz,
+               DecoderDatabase* decoder_database,
+               SyncBuffer* sync_buffer)
+      : fs_hz_(fs_hz),
+        first_call_(true),
+        overlap_length_(5 * fs_hz_ / 8000),
+        decoder_database_(decoder_database),
+        sync_buffer_(sync_buffer) {}
+
+  ComfortNoise(const ComfortNoise&) = delete;
+  ComfortNoise& operator=(const ComfortNoise&) = delete;
+
+  // Resets the state. Should be called before each new comfort noise period.
+  void Reset();
+
+  // Update the comfort noise generator with the parameters in `packet`.
+  int UpdateParameters(const Packet& packet);
+
+  // Generates `requested_length` samples of comfort noise and writes to
+  // `output`. If this is the first in call after Reset (or first after creating
+  // the object), it will also mix in comfort noise at the end of the
+  // SyncBuffer object provided in the constructor.
+  int Generate(size_t requested_length, AudioMultiVector* output);
+
+  // Returns the last error code that was produced by the comfort noise
+  // decoder. Returns 0 if no error has been encountered since the last reset.
+  int internal_error_code() { return internal_error_code_; }
+
+ private:
+  int fs_hz_;
+  bool first_call_;
+  size_t overlap_length_;
+  DecoderDatabase* decoder_database_;
+  SyncBuffer* sync_buffer_;
+  int internal_error_code_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc
new file mode 100644
index 0000000000..b436800061
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for ComfortNoise class.
+
+#include "modules/audio_coding/neteq/comfort_noise.h"
+
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(ComfortNoise, CreateAndDestroy) {
+  int fs = 8000;
+  MockDecoderDatabase db;
+  SyncBuffer sync_buffer(1, 1000);
+  ComfortNoise cn(fs, &db, &sync_buffer);
+  EXPECT_CALL(db, Die());  // Called when `db` goes out of scope.
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.cc b/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.cc
new file mode 100644
index 0000000000..37ed9374f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.cc
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/cross_correlation.h"
+
+#include <cstdlib>
+#include <limits>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+// This function decides the overflow-protecting scaling and calls
+// WebRtcSpl_CrossCorrelation.
+int CrossCorrelationWithAutoShift(const int16_t* sequence_1,
+                                  const int16_t* sequence_2,
+                                  size_t sequence_1_length,
+                                  size_t cross_correlation_length,
+                                  int cross_correlation_step,
+                                  int32_t* cross_correlation) {
+  // Find the element that has the maximum absolute value of sequence_1 and 2.
+  // Note that these values may be negative.
+  const int16_t max_1 =
+      WebRtcSpl_MaxAbsElementW16(sequence_1, sequence_1_length);
+  const int sequence_2_shift =
+      cross_correlation_step * (static_cast<int>(cross_correlation_length) - 1);
+  const int16_t* sequence_2_start =
+      sequence_2_shift >= 0 ? sequence_2 : sequence_2 + sequence_2_shift;
+  const size_t sequence_2_length =
+      sequence_1_length + std::abs(sequence_2_shift);
+  const int16_t max_2 =
+      WebRtcSpl_MaxAbsElementW16(sequence_2_start, sequence_2_length);
+
+  // In order to avoid overflow when computing the sum we should scale the
+  // samples so that (in_vector_length * max_1 * max_2) will not overflow.
+  const int64_t max_value =
+      abs(max_1 * max_2) * static_cast<int64_t>(sequence_1_length);
+  const int32_t factor = max_value >> 31;
+  const int scaling = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+
+  WebRtcSpl_CrossCorrelation(cross_correlation, sequence_1, sequence_2,
+                             sequence_1_length, cross_correlation_length,
+                             scaling, cross_correlation_step);
+
+  return scaling;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.h b/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.h
new file mode 100644
index 0000000000..5082ce6a30
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+#define MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// The function calculates the cross-correlation between two sequences
+// `sequence_1` and `sequence_2`. `sequence_1` is taken as reference, with
+// `sequence_1_length` as its length. `sequence_2` slides for the calculation of
+// cross-correlation. The result will be saved in `cross_correlation`.
+// `cross_correlation_length` correlation points are calculated.
+// The corresponding lag starts from 0, and increases with a step of
+// `cross_correlation_step`. The result is without normalization. To avoid
+// overflow, the result will be right shifted. The amount of shifts will be
+// returned.
+//
+// Input:
+//     - sequence_1     : First sequence (reference).
+//     - sequence_2     : Second sequence (sliding during calculation).
+//     - sequence_1_length : Length of `sequence_1`.
+//     - cross_correlation_length : Number of cross-correlations to calculate.
+//     - cross_correlation_step : Step in the lag for the cross-correlation.
+//
+// Output:
+//      - cross_correlation : The cross-correlation in Q(-right_shifts)
+//
+// Return:
+//      Number of right shifts in cross_correlation.
+
+int CrossCorrelationWithAutoShift(const int16_t* sequence_1,
+                                  const int16_t* sequence_2,
+                                  size_t sequence_1_length,
+                                  size_t cross_correlation_length,
+                                  int cross_correlation_step,
+                                  int32_t* cross_correlation);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
new file mode 100644
index 0000000000..91b0252d2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
@@ -0,0 +1,515 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/decision_logic.h"
+
+#include <stdio.h>
+
+#include <cstdint>
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "modules/audio_coding/neteq/packet_arrival_history.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kPostponeDecodingLevel = 50;
+constexpr int kTargetLevelWindowMs = 100;
+constexpr int kMaxWaitForPacketTicks = 10;
+// The granularity of delay adjustments (accelerate/preemptive expand) is 15ms,
+// but round up since the clock has a granularity of 10ms.
+constexpr int kDelayAdjustmentGranularityMs = 20;
+
+std::unique_ptr<DelayManager> CreateDelayManager(
+    const NetEqController::Config& neteq_config) {
+  DelayManager::Config config;
+  config.max_packets_in_buffer = neteq_config.max_packets_in_buffer;
+  config.base_minimum_delay_ms = neteq_config.base_min_delay_ms;
+  config.Log();
+  return std::make_unique<DelayManager>(config, neteq_config.tick_timer);
+}
+
+bool IsTimestretch(NetEq::Mode mode) {
+  return mode == NetEq::Mode::kAccelerateSuccess ||
+         mode == NetEq::Mode::kAccelerateLowEnergy ||
+         mode == NetEq::Mode::kPreemptiveExpandSuccess ||
+         mode == NetEq::Mode::kPreemptiveExpandLowEnergy;
+}
+
+bool IsCng(NetEq::Mode mode) {
+  return mode == NetEq::Mode::kRfc3389Cng ||
+         mode == NetEq::Mode::kCodecInternalCng;
+}
+
+bool IsExpand(NetEq::Mode mode) {
+  return mode == NetEq::Mode::kExpand || mode == NetEq::Mode::kCodecPlc;
+}
+
+}  // namespace
+
+DecisionLogic::Config::Config() {
+  StructParametersParser::Create(
+      "enable_stable_playout_delay", &enable_stable_playout_delay,  //
+      "reinit_after_expands", &reinit_after_expands,                //
+      "packet_history_size_ms", &packet_history_size_ms,            //
+      "cng_timeout_ms", &cng_timeout_ms,                            //
+      "deceleration_target_level_offset_ms",
+      &deceleration_target_level_offset_ms)
+      ->Parse(webrtc::field_trial::FindFullName(
+          "WebRTC-Audio-NetEqDecisionLogicConfig"));
+  RTC_LOG(LS_INFO) << "NetEq decision logic config:"
+                   << " enable_stable_playout_delay="
+                   << enable_stable_playout_delay
+                   << " reinit_after_expands=" << reinit_after_expands
+                   << " packet_history_size_ms=" << packet_history_size_ms
+                   << " cng_timeout_ms=" << cng_timeout_ms.value_or(-1)
+                   << " deceleration_target_level_offset_ms="
+                   << deceleration_target_level_offset_ms;
+}
+
+DecisionLogic::DecisionLogic(NetEqController::Config config)
+    : DecisionLogic(config,
+                    CreateDelayManager(config),
+                    std::make_unique<BufferLevelFilter>()) {}
+
+DecisionLogic::DecisionLogic(
+    NetEqController::Config config,
+    std::unique_ptr<DelayManager> delay_manager,
+    std::unique_ptr<BufferLevelFilter> buffer_level_filter)
+    : delay_manager_(std::move(delay_manager)),
+      buffer_level_filter_(std::move(buffer_level_filter)),
+      packet_arrival_history_(config_.packet_history_size_ms),
+      tick_timer_(config.tick_timer),
+      disallow_time_stretching_(!config.allow_time_stretching),
+      timescale_countdown_(
+          tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1)) {}
+
+DecisionLogic::~DecisionLogic() = default;
+
+void DecisionLogic::SoftReset() {
+  packet_length_samples_ = 0;
+  sample_memory_ = 0;
+  prev_time_scale_ = false;
+  timescale_countdown_ =
+      tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1);
+  time_stretched_cn_samples_ = 0;
+  delay_manager_->Reset();
+  buffer_level_filter_->Reset();
+  packet_arrival_history_.Reset();
+  last_playout_delay_ms_ = 0;
+}
+
+void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
+  // TODO(hlundin): Change to an enumerator and skip assert.
+  RTC_DCHECK(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 ||
+             fs_hz == 48000);
+  sample_rate_khz_ = fs_hz / 1000;
+  output_size_samples_ = output_size_samples;
+  packet_arrival_history_.set_sample_rate(fs_hz);
+}
+
+NetEq::Operation DecisionLogic::GetDecision(const NetEqStatus& status,
+                                            bool* reset_decoder) {
+  // If last mode was CNG (or Expand, since this could be covering up for
+  // a lost CNG packet), remember that CNG is on. This is needed if comfort
+  // noise is interrupted by DTMF.
+  if (status.last_mode == NetEq::Mode::kRfc3389Cng) {
+    cng_state_ = kCngRfc3389On;
+  } else if (status.last_mode == NetEq::Mode::kCodecInternalCng) {
+    cng_state_ = kCngInternalOn;
+  }
+
+  if (IsExpand(status.last_mode)) {
+    ++num_consecutive_expands_;
+  } else {
+    num_consecutive_expands_ = 0;
+  }
+
+  if (!IsExpand(status.last_mode) && !IsCng(status.last_mode)) {
+    last_playout_delay_ms_ = GetPlayoutDelayMs(status);
+  }
+
+  prev_time_scale_ = prev_time_scale_ && IsTimestretch(status.last_mode);
+  if (prev_time_scale_) {
+    timescale_countdown_ = tick_timer_->GetNewCountdown(kMinTimescaleInterval);
+  }
+  if (!IsCng(status.last_mode)) {
+    FilterBufferLevel(status.packet_buffer_info.span_samples);
+  }
+
+  // Guard for errors, to avoid getting stuck in error mode.
+  if (status.last_mode == NetEq::Mode::kError) {
+    if (!status.next_packet) {
+      return NetEq::Operation::kExpand;
+    } else {
+      // Use kUndefined to flag for a reset.
+      return NetEq::Operation::kUndefined;
+    }
+  }
+
+  if (status.next_packet && status.next_packet->is_cng) {
+    return CngOperation(status);
+  }
+
+  // Handle the case with no packet at all available (except maybe DTMF).
+  if (!status.next_packet) {
+    return NoPacket(status);
+  }
+
+  // If the expand period was very long, reset NetEQ since it is likely that the
+  // sender was restarted.
+  if (num_consecutive_expands_ > config_.reinit_after_expands) {
+    *reset_decoder = true;
+    return NetEq::Operation::kNormal;
+  }
+
+  // Make sure we don't restart audio too soon after an expansion to avoid
+  // running out of data right away again. We should only wait if there are no
+  // DTX or CNG packets in the buffer (otherwise we should just play out what we
+  // have, since we cannot know the exact duration of DTX or CNG packets), and
+  // if the mute factor is low enough (otherwise the expansion was short enough
+  // to not be noticable).
+  // Note that the MuteFactor is in Q14, so a value of 16384 corresponds to 1.
+  const int target_level_samples = TargetLevelMs() * sample_rate_khz_;
+  if (!config_.enable_stable_playout_delay && IsExpand(status.last_mode) &&
+      status.expand_mutefactor < 16384 / 2 &&
+      status.packet_buffer_info.span_samples <
+          static_cast<size_t>(target_level_samples * kPostponeDecodingLevel /
+                              100) &&
+      !status.packet_buffer_info.dtx_or_cng) {
+    return NetEq::Operation::kExpand;
+  }
+
+  const uint32_t five_seconds_samples =
+      static_cast<uint32_t>(5000 * sample_rate_khz_);
+  // Check if the required packet is available.
+  if (status.target_timestamp == status.next_packet->timestamp) {
+    return ExpectedPacketAvailable(status);
+  }
+  if (!PacketBuffer::IsObsoleteTimestamp(status.next_packet->timestamp,
+                                         status.target_timestamp,
+                                         five_seconds_samples)) {
+    return FuturePacketAvailable(status);
+  }
+  // This implies that available_timestamp < target_timestamp, which can
+  // happen when a new stream or codec is received. Signal for a reset.
+  return NetEq::Operation::kUndefined;
+}
+
+void DecisionLogic::NotifyMutedState() {
+  ++num_consecutive_expands_;
+}
+
+int DecisionLogic::TargetLevelMs() const {
+  int target_delay_ms = delay_manager_->TargetDelayMs();
+  if (!config_.enable_stable_playout_delay) {
+    target_delay_ms =
+        std::max(target_delay_ms,
+                 static_cast<int>(packet_length_samples_ / sample_rate_khz_));
+  }
+  return target_delay_ms;
+}
+
+int DecisionLogic::UnlimitedTargetLevelMs() const {
+  return delay_manager_->UnlimitedTargetLevelMs();
+}
+
+int DecisionLogic::GetFilteredBufferLevel() const {
+  if (config_.enable_stable_playout_delay) {
+    return last_playout_delay_ms_ * sample_rate_khz_;
+  }
+  return buffer_level_filter_->filtered_current_level();
+}
+
+absl::optional<int> DecisionLogic::PacketArrived(
+    int fs_hz,
+    bool should_update_stats,
+    const PacketArrivedInfo& info) {
+  buffer_flush_ = buffer_flush_ || info.buffer_flush;
+  if (!should_update_stats || info.is_cng_or_dtmf) {
+    return absl::nullopt;
+  }
+  if (info.packet_length_samples > 0 && fs_hz > 0 &&
+      info.packet_length_samples != packet_length_samples_) {
+    packet_length_samples_ = info.packet_length_samples;
+    delay_manager_->SetPacketAudioLength(packet_length_samples_ * 1000 / fs_hz);
+  }
+  int64_t time_now_ms = tick_timer_->ticks() * tick_timer_->ms_per_tick();
+  packet_arrival_history_.Insert(info.main_timestamp, time_now_ms);
+  if (packet_arrival_history_.size() < 2) {
+    // No meaningful delay estimate unless at least 2 packets have arrived.
+    return absl::nullopt;
+  }
+  int arrival_delay_ms =
+      packet_arrival_history_.GetDelayMs(info.main_timestamp, time_now_ms);
+  bool reordered =
+      !packet_arrival_history_.IsNewestRtpTimestamp(info.main_timestamp);
+  delay_manager_->Update(arrival_delay_ms, reordered);
+  return arrival_delay_ms;
+}
+
+void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples) {
+  buffer_level_filter_->SetTargetBufferLevel(TargetLevelMs());
+
+  int time_stretched_samples = time_stretched_cn_samples_;
+  if (prev_time_scale_) {
+    time_stretched_samples += sample_memory_;
+  }
+
+  if (buffer_flush_) {
+    buffer_level_filter_->SetFilteredBufferLevel(buffer_size_samples);
+    buffer_flush_ = false;
+  } else {
+    buffer_level_filter_->Update(buffer_size_samples, time_stretched_samples);
+  }
+  prev_time_scale_ = false;
+  time_stretched_cn_samples_ = 0;
+}
+
+NetEq::Operation DecisionLogic::CngOperation(
+    NetEqController::NetEqStatus status) {
+  // Signed difference between target and available timestamp.
+  int32_t timestamp_diff = static_cast<int32_t>(
+      static_cast<uint32_t>(status.generated_noise_samples +
+                            status.target_timestamp) -
+      status.next_packet->timestamp);
+  int optimal_level_samp = TargetLevelMs() * sample_rate_khz_;
+  const int64_t excess_waiting_time_samp =
+      -static_cast<int64_t>(timestamp_diff) - optimal_level_samp;
+
+  if (excess_waiting_time_samp > optimal_level_samp / 2) {
+    // The waiting time for this packet will be longer than 1.5
+    // times the wanted buffer delay. Apply fast-forward to cut the
+    // waiting time down to the optimal.
+    noise_fast_forward_ = rtc::saturated_cast<size_t>(noise_fast_forward_ +
+                                                      excess_waiting_time_samp);
+    timestamp_diff =
+        rtc::saturated_cast<int32_t>(timestamp_diff + excess_waiting_time_samp);
+  }
+
+  if (timestamp_diff < 0 && status.last_mode == NetEq::Mode::kRfc3389Cng) {
+    // Not time to play this packet yet. Wait another round before using this
+    // packet. Keep on playing CNG from previous CNG parameters.
+    return NetEq::Operation::kRfc3389CngNoPacket;
+  } else {
+    // Otherwise, go for the CNG packet now.
+    noise_fast_forward_ = 0;
+    return NetEq::Operation::kRfc3389Cng;
+  }
+}
+
+NetEq::Operation DecisionLogic::NoPacket(NetEqController::NetEqStatus status) {
+  if (cng_state_ == kCngRfc3389On) {
+    // Keep on playing comfort noise.
+    return NetEq::Operation::kRfc3389CngNoPacket;
+  } else if (cng_state_ == kCngInternalOn) {
+    // Stop CNG after a timeout.
+    if (config_.cng_timeout_ms &&
+        status.generated_noise_samples >
+            static_cast<size_t>(*config_.cng_timeout_ms * sample_rate_khz_)) {
+      return NetEq::Operation::kExpand;
+    }
+    return NetEq::Operation::kCodecInternalCng;
+  } else if (status.play_dtmf) {
+    return NetEq::Operation::kDtmf;
+  } else {
+    // Nothing to play, do expand.
+    return NetEq::Operation::kExpand;
+  }
+}
+
+NetEq::Operation DecisionLogic::ExpectedPacketAvailable(
+    NetEqController::NetEqStatus status) {
+  if (!disallow_time_stretching_ && status.last_mode != NetEq::Mode::kExpand &&
+      !status.play_dtmf) {
+    if (config_.enable_stable_playout_delay) {
+      const int playout_delay_ms = GetPlayoutDelayMs(status);
+      if (playout_delay_ms >= HighThreshold() << 2) {
+        return NetEq::Operation::kFastAccelerate;
+      }
+      if (TimescaleAllowed()) {
+        if (playout_delay_ms >= HighThreshold()) {
+          return NetEq::Operation::kAccelerate;
+        }
+        if (playout_delay_ms < LowThreshold()) {
+          return NetEq::Operation::kPreemptiveExpand;
+        }
+      }
+    } else {
+      const int target_level_samples = TargetLevelMs() * sample_rate_khz_;
+      const int low_limit = std::max(
+          target_level_samples * 3 / 4,
+          target_level_samples -
+              config_.deceleration_target_level_offset_ms * sample_rate_khz_);
+      const int high_limit = std::max(
+          target_level_samples,
+          low_limit + kDelayAdjustmentGranularityMs * sample_rate_khz_);
+
+      const int buffer_level_samples =
+          buffer_level_filter_->filtered_current_level();
+      if (buffer_level_samples >= high_limit << 2)
+        return NetEq::Operation::kFastAccelerate;
+      if (TimescaleAllowed()) {
+        if (buffer_level_samples >= high_limit)
+          return NetEq::Operation::kAccelerate;
+        if (buffer_level_samples < low_limit)
+          return NetEq::Operation::kPreemptiveExpand;
+      }
+    }
+  }
+  return NetEq::Operation::kNormal;
+}
+
+NetEq::Operation DecisionLogic::FuturePacketAvailable(
+    NetEqController::NetEqStatus status) {
+  // Required packet is not available, but a future packet is.
+  // Check if we should continue with an ongoing expand because the new packet
+  // is too far into the future.
+  if (IsExpand(status.last_mode) && ShouldContinueExpand(status)) {
+    if (status.play_dtmf) {
+      // Still have DTMF to play, so do not do expand.
+      return NetEq::Operation::kDtmf;
+    } else {
+      // Nothing to play.
+      return NetEq::Operation::kExpand;
+    }
+  }
+
+  if (status.last_mode == NetEq::Mode::kCodecPlc) {
+    return NetEq::Operation::kNormal;
+  }
+
+  // If previous was comfort noise, then no merge is needed.
+  if (IsCng(status.last_mode)) {
+    uint32_t timestamp_leap =
+        status.next_packet->timestamp - status.target_timestamp;
+    const bool generated_enough_noise =
+        status.generated_noise_samples >= timestamp_leap;
+
+    int playout_delay_ms = GetNextPacketDelayMs(status);
+    const bool above_target_delay = playout_delay_ms > HighThresholdCng();
+    const bool below_target_delay = playout_delay_ms < LowThresholdCng();
+    // Keep the delay same as before CNG, but make sure that it is within the
+    // target window.
+    if ((generated_enough_noise && !below_target_delay) || above_target_delay) {
+      time_stretched_cn_samples_ =
+          timestamp_leap - status.generated_noise_samples;
+      return NetEq::Operation::kNormal;
+    }
+
+    if (status.last_mode == NetEq::Mode::kRfc3389Cng) {
+      return NetEq::Operation::kRfc3389CngNoPacket;
+    }
+    return NetEq::Operation::kCodecInternalCng;
+  }
+
+  // Do not merge unless we have done an expand before.
+  if (status.last_mode == NetEq::Mode::kExpand) {
+    return NetEq::Operation::kMerge;
+  } else if (status.play_dtmf) {
+    // Play DTMF instead of expand.
+    return NetEq::Operation::kDtmf;
+  } else {
+    return NetEq::Operation::kExpand;
+  }
+}
+
+bool DecisionLogic::UnderTargetLevel() const {
+  return buffer_level_filter_->filtered_current_level() <
+         TargetLevelMs() * sample_rate_khz_;
+}
+
+bool DecisionLogic::ReinitAfterExpands(uint32_t timestamp_leap) const {
+  return timestamp_leap >= static_cast<uint32_t>(output_size_samples_ *
+                                                 config_.reinit_after_expands);
+}
+
+bool DecisionLogic::PacketTooEarly(uint32_t timestamp_leap) const {
+  return timestamp_leap >
+         static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
+}
+
+bool DecisionLogic::MaxWaitForPacket() const {
+  return num_consecutive_expands_ >= kMaxWaitForPacketTicks;
+}
+
+bool DecisionLogic::ShouldContinueExpand(
+    NetEqController::NetEqStatus status) const {
+  uint32_t timestamp_leap =
+      status.next_packet->timestamp - status.target_timestamp;
+  if (config_.enable_stable_playout_delay) {
+    return GetNextPacketDelayMs(status) < HighThreshold() &&
+           PacketTooEarly(timestamp_leap);
+  }
+  return !ReinitAfterExpands(timestamp_leap) && !MaxWaitForPacket() &&
+         PacketTooEarly(timestamp_leap) && UnderTargetLevel();
+}
+
+int DecisionLogic::GetNextPacketDelayMs(
+    NetEqController::NetEqStatus status) const {
+  if (config_.enable_stable_playout_delay) {
+    return packet_arrival_history_.GetDelayMs(
+        status.next_packet->timestamp,
+        tick_timer_->ticks() * tick_timer_->ms_per_tick());
+  }
+  return status.packet_buffer_info.span_samples / sample_rate_khz_;
+}
+
+int DecisionLogic::GetPlayoutDelayMs(
+    NetEqController::NetEqStatus status) const {
+  uint32_t playout_timestamp =
+      status.target_timestamp - status.sync_buffer_samples;
+  return packet_arrival_history_.GetDelayMs(
+      playout_timestamp, tick_timer_->ticks() * tick_timer_->ms_per_tick());
+}
+
+int DecisionLogic::LowThreshold() const {
+  int target_delay_ms = TargetLevelMs();
+  return std::max(
+      target_delay_ms * 3 / 4,
+      target_delay_ms - config_.deceleration_target_level_offset_ms);
+}
+
+int DecisionLogic::HighThreshold() const {
+  if (config_.enable_stable_playout_delay) {
+    return std::max(TargetLevelMs(), packet_arrival_history_.GetMaxDelayMs()) +
+           kDelayAdjustmentGranularityMs;
+  }
+  return std::max(TargetLevelMs(),
+                  LowThreshold() + kDelayAdjustmentGranularityMs);
+}
+
+int DecisionLogic::LowThresholdCng() const {
+  if (config_.enable_stable_playout_delay) {
+    return LowThreshold();
+  }
+  return std::max(0, TargetLevelMs() - kTargetLevelWindowMs / 2);
+}
+
+int DecisionLogic::HighThresholdCng() const {
+  if (config_.enable_stable_playout_delay) {
+    return HighThreshold();
+  }
+  return TargetLevelMs() + kTargetLevelWindowMs / 2;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h
new file mode 100644
index 0000000000..8d1ff4d622
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h
@@ -0,0 +1,202 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+
+#include <memory>
+
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/packet_arrival_history.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+// This is the class for the decision tree implementation.
+class DecisionLogic : public NetEqController {
+ public:
+  DecisionLogic(NetEqController::Config config);
+  DecisionLogic(NetEqController::Config config,
+                std::unique_ptr<DelayManager> delay_manager,
+                std::unique_ptr<BufferLevelFilter> buffer_level_filter);
+
+  ~DecisionLogic() override;
+
+  DecisionLogic(const DecisionLogic&) = delete;
+  DecisionLogic& operator=(const DecisionLogic&) = delete;
+
+  // Not used.
+  void Reset() override {}
+
+  // Resets parts of the state. Typically done when switching codecs.
+  void SoftReset() override;
+
+  // Sets the sample rate and the output block size.
+  void SetSampleRate(int fs_hz, size_t output_size_samples) override;
+
+  // Given info about the latest received packet, and current jitter buffer
+  // status, returns the operation. `target_timestamp` and `expand_mutefactor`
+  // are provided for reference. `last_packet_samples` is the number of samples
+  // obtained from the last decoded frame. If there is a packet available, it
+  // should be supplied in `packet`; otherwise it should be NULL. The mode
+  // resulting from the last call to NetEqImpl::GetAudio is supplied in
+  // `last_mode`. If there is a DTMF event to play, `play_dtmf` should be set to
+  // true. The output variable `reset_decoder` will be set to true if a reset is
+  // required; otherwise it is left unchanged (i.e., it can remain true if it
+  // was true before the call).
+  NetEq::Operation GetDecision(const NetEqController::NetEqStatus& status,
+                               bool* reset_decoder) override;
+
+  // These methods test the `cng_state_` for different conditions.
+  bool CngRfc3389On() const override { return cng_state_ == kCngRfc3389On; }
+  bool CngOff() const override { return cng_state_ == kCngOff; }
+
+  // Resets the `cng_state_` to kCngOff.
+  void SetCngOff() override { cng_state_ = kCngOff; }
+
+  void ExpandDecision(NetEq::Operation operation) override {}
+
+  // Adds `value` to `sample_memory_`.
+  void AddSampleMemory(int32_t value) override { sample_memory_ += value; }
+
+  int TargetLevelMs() const override;
+
+  int UnlimitedTargetLevelMs() const override;
+
+  absl::optional<int> PacketArrived(int fs_hz,
+                                    bool should_update_stats,
+                                    const PacketArrivedInfo& info) override;
+
+  void RegisterEmptyPacket() override {}
+
+  void NotifyMutedState() override;
+
+  bool SetMaximumDelay(int delay_ms) override {
+    return delay_manager_->SetMaximumDelay(delay_ms);
+  }
+  bool SetMinimumDelay(int delay_ms) override {
+    return delay_manager_->SetMinimumDelay(delay_ms);
+  }
+  bool SetBaseMinimumDelay(int delay_ms) override {
+    return delay_manager_->SetBaseMinimumDelay(delay_ms);
+  }
+  int GetBaseMinimumDelay() const override {
+    return delay_manager_->GetBaseMinimumDelay();
+  }
+  bool PeakFound() const override { return false; }
+
+  int GetFilteredBufferLevel() const override;
+
+  // Accessors and mutators.
+  void set_sample_memory(int32_t value) override { sample_memory_ = value; }
+  size_t noise_fast_forward() const override { return noise_fast_forward_; }
+  size_t packet_length_samples() const override {
+    return packet_length_samples_;
+  }
+  void set_packet_length_samples(size_t value) override {
+    packet_length_samples_ = value;
+  }
+  void set_prev_time_scale(bool value) override { prev_time_scale_ = value; }
+
+ private:
+  // The value 5 sets maximum time-stretch rate to about 100 ms/s.
+  static const int kMinTimescaleInterval = 5;
+
+  enum CngState { kCngOff, kCngRfc3389On, kCngInternalOn };
+
+  // Updates the `buffer_level_filter_` with the current buffer level
+  // `buffer_size_samples`.
+  void FilterBufferLevel(size_t buffer_size_samples);
+
+  // Returns the operation given that the next available packet is a comfort
+  // noise payload (RFC 3389 only, not codec-internal).
+  virtual NetEq::Operation CngOperation(NetEqController::NetEqStatus status);
+
+  // Returns the operation given that no packets are available (except maybe
+  // a DTMF event, flagged by setting `play_dtmf` true).
+  virtual NetEq::Operation NoPacket(NetEqController::NetEqStatus status);
+
+  // Returns the operation to do given that the expected packet is available.
+  virtual NetEq::Operation ExpectedPacketAvailable(
+      NetEqController::NetEqStatus status);
+
+  // Returns the operation to do given that the expected packet is not
+  // available, but a packet further into the future is at hand.
+  virtual NetEq::Operation FuturePacketAvailable(
+      NetEqController::NetEqStatus status);
+
+  // Checks if enough time has elapsed since the last successful timescale
+  // operation was done (i.e., accelerate or preemptive expand).
+  bool TimescaleAllowed() const {
+    return !timescale_countdown_ || timescale_countdown_->Finished();
+  }
+
+  // Checks if the current (filtered) buffer level is under the target level.
+  bool UnderTargetLevel() const;
+
+  // Checks if `timestamp_leap` is so long into the future that a reset due
+  // to exceeding kReinitAfterExpands will be done.
+  bool ReinitAfterExpands(uint32_t timestamp_leap) const;
+
+  // Checks if we still have not done enough expands to cover the distance from
+  // the last decoded packet to the next available packet, the distance beeing
+  // conveyed in `timestamp_leap`.
+  bool PacketTooEarly(uint32_t timestamp_leap) const;
+
+  bool MaxWaitForPacket() const;
+
+  bool ShouldContinueExpand(NetEqController::NetEqStatus status) const;
+
+  int GetNextPacketDelayMs(NetEqController::NetEqStatus status) const;
+  int GetPlayoutDelayMs(NetEqController::NetEqStatus status) const;
+
+  int LowThreshold() const;
+  int HighThreshold() const;
+  int LowThresholdCng() const;
+  int HighThresholdCng() const;
+
+  // Runtime configurable options through field trial
+  // WebRTC-Audio-NetEqDecisionLogicConfig.
+  struct Config {
+    Config();
+
+    bool enable_stable_playout_delay = false;
+    int reinit_after_expands = 100;
+    int deceleration_target_level_offset_ms = 85;
+    int packet_history_size_ms = 2000;
+    absl::optional<int> cng_timeout_ms;
+  };
+  Config config_;
+  std::unique_ptr<DelayManager> delay_manager_;
+  std::unique_ptr<BufferLevelFilter> buffer_level_filter_;
+  PacketArrivalHistory packet_arrival_history_;
+  const TickTimer* tick_timer_;
+  int sample_rate_khz_;
+  size_t output_size_samples_;
+  CngState cng_state_ = kCngOff;  // Remember if comfort noise is interrupted by
+                                  // other event (e.g., DTMF).
+  size_t noise_fast_forward_ = 0;
+  size_t packet_length_samples_ = 0;
+  int sample_memory_ = 0;
+  bool prev_time_scale_ = false;
+  bool disallow_time_stretching_;
+  std::unique_ptr<TickTimer::Countdown> timescale_countdown_;
+  int num_consecutive_expands_ = 0;
+  int time_stretched_cn_samples_ = 0;
+  bool buffer_flush_ = false;
+  int last_playout_delay_ms_ = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
new file mode 100644
index 0000000000..6150c9a6db
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -0,0 +1,219 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for DecisionLogic class and derived classes.
+
+#include "modules/audio_coding/neteq/decision_logic.h"
+
+#include "api/neteq/neteq_controller.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/mock/mock_buffer_level_filter.h"
+#include "modules/audio_coding/neteq/mock/mock_delay_manager.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kSampleRate = 8000;
+constexpr int kSamplesPerMs = kSampleRate / 1000;
+constexpr int kOutputSizeSamples = kSamplesPerMs * 10;
+constexpr int kMinTimescaleInterval = 5;
+
+NetEqController::NetEqStatus CreateNetEqStatus(NetEq::Mode last_mode,
+                                               int current_delay_ms) {
+  NetEqController::NetEqStatus status;
+  status.play_dtmf = false;
+  status.last_mode = last_mode;
+  status.target_timestamp = 1234;
+  status.generated_noise_samples = 0;
+  status.expand_mutefactor = 0;
+  status.packet_buffer_info.num_samples = current_delay_ms * kSamplesPerMs;
+  status.packet_buffer_info.span_samples = current_delay_ms * kSamplesPerMs;
+  status.packet_buffer_info.span_samples_no_dtx =
+      current_delay_ms * kSamplesPerMs;
+  status.packet_buffer_info.dtx_or_cng = false;
+  status.next_packet = {status.target_timestamp, false, false};
+  return status;
+}
+
+using ::testing::Return;
+
+}  // namespace
+
+class DecisionLogicTest : public ::testing::Test {
+ protected:
+  DecisionLogicTest() {
+    test::ScopedFieldTrials trials(
+        "WebRTC-Audio-NetEqDecisionLogicConfig/cng_timeout_ms:1000/");
+    NetEqController::Config config;
+    config.tick_timer = &tick_timer_;
+    config.allow_time_stretching = true;
+    auto delay_manager = std::make_unique<MockDelayManager>(
+        DelayManager::Config(), config.tick_timer);
+    mock_delay_manager_ = delay_manager.get();
+    auto buffer_level_filter = std::make_unique<MockBufferLevelFilter>();
+    mock_buffer_level_filter_ = buffer_level_filter.get();
+    decision_logic_ = std::make_unique<DecisionLogic>(
+        config, std::move(delay_manager), std::move(buffer_level_filter));
+    decision_logic_->SetSampleRate(kSampleRate, kOutputSizeSamples);
+  }
+
+  TickTimer tick_timer_;
+  std::unique_ptr<DecisionLogic> decision_logic_;
+  MockDelayManager* mock_delay_manager_;
+  MockBufferLevelFilter* mock_buffer_level_filter_;
+};
+
+TEST_F(DecisionLogicTest, NormalOperation) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(100));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(90 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 100), &reset_decoder),
+            NetEq::Operation::kNormal);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, Accelerate) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(100));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(110 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 100), &reset_decoder),
+            NetEq::Operation::kAccelerate);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, FastAccelerate) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(100));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(400 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 100), &reset_decoder),
+            NetEq::Operation::kFastAccelerate);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, PreemptiveExpand) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(100));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(50 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 100), &reset_decoder),
+            NetEq::Operation::kPreemptiveExpand);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, DecelerationTargetLevelOffset) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(500));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(400 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 400), &reset_decoder),
+            NetEq::Operation::kPreemptiveExpand);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, PostponeDecodeAfterExpand) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(500));
+
+  // Below 50% target delay threshold.
+  bool reset_decoder = false;
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kExpand, 200), &reset_decoder),
+            NetEq::Operation::kExpand);
+  EXPECT_FALSE(reset_decoder);
+
+  // Above 50% target delay threshold.
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kExpand, 250), &reset_decoder),
+            NetEq::Operation::kNormal);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, TimeStrechComfortNoise) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(500));
+
+  {
+    bool reset_decoder = false;
+    // Below target window.
+    auto status = CreateNetEqStatus(NetEq::Mode::kCodecInternalCng, 400);
+    status.generated_noise_samples = 400 * kSamplesPerMs;
+    status.next_packet->timestamp =
+        status.target_timestamp + 400 * kSamplesPerMs;
+    EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder),
+              NetEq::Operation::kCodecInternalCng);
+    EXPECT_FALSE(reset_decoder);
+  }
+
+  {
+    bool reset_decoder = false;
+    // Above target window.
+    auto status = CreateNetEqStatus(NetEq::Mode::kCodecInternalCng, 600);
+    status.generated_noise_samples = 200 * kSamplesPerMs;
+    status.next_packet->timestamp =
+        status.target_timestamp + 400 * kSamplesPerMs;
+    EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder),
+              NetEq::Operation::kNormal);
+    EXPECT_FALSE(reset_decoder);
+
+    // The buffer level filter should be adjusted with the number of samples
+    // that was skipped.
+    int timestamp_leap = status.next_packet->timestamp -
+                         status.target_timestamp -
+                         status.generated_noise_samples;
+    EXPECT_CALL(*mock_buffer_level_filter_,
+                Update(400 * kSamplesPerMs, timestamp_leap));
+    EXPECT_EQ(decision_logic_->GetDecision(
+                  CreateNetEqStatus(NetEq::Mode::kNormal, 400), &reset_decoder),
+              NetEq::Operation::kNormal);
+    EXPECT_FALSE(reset_decoder);
+  }
+}
+
+TEST_F(DecisionLogicTest, CngTimeout) {
+  auto status = CreateNetEqStatus(NetEq::Mode::kCodecInternalCng, 0);
+  status.next_packet = absl::nullopt;
+  status.generated_noise_samples = kSamplesPerMs * 500;
+  bool reset_decoder = false;
+  EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder),
+            NetEq::Operation::kCodecInternalCng);
+  status.generated_noise_samples = kSamplesPerMs * 1010;
+  EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder),
+            NetEq::Operation::kExpand);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.cc
new file mode 100644
index 0000000000..3447ced1da
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -0,0 +1,285 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/decoder_database.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <list>
+#include <type_traits>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/audio_format_to_string.h"
+
+namespace webrtc {
+
+DecoderDatabase::DecoderDatabase(
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+    absl::optional<AudioCodecPairId> codec_pair_id)
+    : active_decoder_type_(-1),
+      active_cng_decoder_type_(-1),
+      decoder_factory_(decoder_factory),
+      codec_pair_id_(codec_pair_id) {}
+
+DecoderDatabase::~DecoderDatabase() = default;
+
+DecoderDatabase::DecoderInfo::DecoderInfo(
+    const SdpAudioFormat& audio_format,
+    absl::optional<AudioCodecPairId> codec_pair_id,
+    AudioDecoderFactory* factory,
+    absl::string_view codec_name)
+    : name_(codec_name),
+      audio_format_(audio_format),
+      codec_pair_id_(codec_pair_id),
+      factory_(factory),
+      cng_decoder_(CngDecoder::Create(audio_format)),
+      subtype_(SubtypeFromFormat(audio_format)) {}
+
+DecoderDatabase::DecoderInfo::DecoderInfo(
+    const SdpAudioFormat& audio_format,
+    absl::optional<AudioCodecPairId> codec_pair_id,
+    AudioDecoderFactory* factory)
+    : DecoderInfo(audio_format, codec_pair_id, factory, audio_format.name) {}
+
+DecoderDatabase::DecoderInfo::DecoderInfo(DecoderInfo&&) = default;
+DecoderDatabase::DecoderInfo::~DecoderInfo() = default;
+
+AudioDecoder* DecoderDatabase::DecoderInfo::GetDecoder() const {
+  if (subtype_ != Subtype::kNormal) {
+    // These are handled internally, so they have no AudioDecoder objects.
+    return nullptr;
+  }
+  if (!decoder_) {
+    // TODO(ossu): Keep a check here for now, since a number of tests create
+    // DecoderInfos without factories.
+    RTC_DCHECK(factory_);
+    decoder_ = factory_->MakeAudioDecoder(audio_format_, codec_pair_id_);
+  }
+  RTC_DCHECK(decoder_) << "Failed to create: " << rtc::ToString(audio_format_);
+  return decoder_.get();
+}
+
+bool DecoderDatabase::DecoderInfo::IsType(absl::string_view name) const {
+  return absl::EqualsIgnoreCase(audio_format_.name, name);
+}
+
+absl::optional<DecoderDatabase::DecoderInfo::CngDecoder>
+DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "CN")) {
+    // CN has a 1:1 RTP clock rate to sample rate ratio.
+    const int sample_rate_hz = format.clockrate_hz;
+    RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+               sample_rate_hz == 32000 || sample_rate_hz == 48000);
+    return DecoderDatabase::DecoderInfo::CngDecoder{sample_rate_hz};
+  } else {
+    return absl::nullopt;
+  }
+}
+
+DecoderDatabase::DecoderInfo::Subtype
+DecoderDatabase::DecoderInfo::SubtypeFromFormat(const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "CN")) {
+    return Subtype::kComfortNoise;
+  } else if (absl::EqualsIgnoreCase(format.name, "telephone-event")) {
+    return Subtype::kDtmf;
+  } else if (absl::EqualsIgnoreCase(format.name, "red")) {
+    return Subtype::kRed;
+  }
+
+  return Subtype::kNormal;
+}
+
+bool DecoderDatabase::Empty() const {
+  return decoders_.empty();
+}
+
+int DecoderDatabase::Size() const {
+  return static_cast<int>(decoders_.size());
+}
+
+std::vector<int> DecoderDatabase::SetCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  // First collect all payload types that we'll remove or reassign, then remove
+  // them from the database.
+  std::vector<int> changed_payload_types;
+  for (const std::pair<uint8_t, const DecoderInfo&> kv : decoders_) {
+    auto i = codecs.find(kv.first);
+    if (i == codecs.end() || i->second != kv.second.GetFormat()) {
+      changed_payload_types.push_back(kv.first);
+    }
+  }
+  for (int pl_type : changed_payload_types) {
+    Remove(pl_type);
+  }
+
+  // Enter the new and changed payload type mappings into the database.
+  for (const auto& kv : codecs) {
+    const int& rtp_payload_type = kv.first;
+    const SdpAudioFormat& audio_format = kv.second;
+    RTC_DCHECK_GE(rtp_payload_type, 0);
+    RTC_DCHECK_LE(rtp_payload_type, 0x7f);
+    if (decoders_.count(rtp_payload_type) == 0) {
+      decoders_.insert(std::make_pair(
+          rtp_payload_type,
+          DecoderInfo(audio_format, codec_pair_id_, decoder_factory_.get())));
+    } else {
+      // The mapping for this payload type hasn't changed.
+    }
+  }
+
+  return changed_payload_types;
+}
+
+int DecoderDatabase::RegisterPayload(int rtp_payload_type,
+                                     const SdpAudioFormat& audio_format) {
+  if (rtp_payload_type < 0 || rtp_payload_type > 0x7f) {
+    return kInvalidRtpPayloadType;
+  }
+  const auto ret = decoders_.insert(std::make_pair(
+      rtp_payload_type,
+      DecoderInfo(audio_format, codec_pair_id_, decoder_factory_.get())));
+  if (ret.second == false) {
+    // Database already contains a decoder with type `rtp_payload_type`.
+    return kDecoderExists;
+  }
+  return kOK;
+}
+
+int DecoderDatabase::Remove(uint8_t rtp_payload_type) {
+  if (decoders_.erase(rtp_payload_type) == 0) {
+    // No decoder with that `rtp_payload_type`.
+    return kDecoderNotFound;
+  }
+  if (active_decoder_type_ == rtp_payload_type) {
+    active_decoder_type_ = -1;  // No active decoder.
+  }
+  if (active_cng_decoder_type_ == rtp_payload_type) {
+    active_cng_decoder_type_ = -1;  // No active CNG decoder.
+  }
+  return kOK;
+}
+
+void DecoderDatabase::RemoveAll() {
+  decoders_.clear();
+  active_decoder_type_ = -1;      // No active decoder.
+  active_cng_decoder_type_ = -1;  // No active CNG decoder.
+}
+
+const DecoderDatabase::DecoderInfo* DecoderDatabase::GetDecoderInfo(
+    uint8_t rtp_payload_type) const {
+  DecoderMap::const_iterator it = decoders_.find(rtp_payload_type);
+  if (it == decoders_.end()) {
+    // Decoder not found.
+    return NULL;
+  }
+  return &it->second;
+}
+
+int DecoderDatabase::SetActiveDecoder(uint8_t rtp_payload_type,
+                                      bool* new_decoder) {
+  // Check that `rtp_payload_type` exists in the database.
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  if (!info) {
+    // Decoder not found.
+    return kDecoderNotFound;
+  }
+  RTC_CHECK(!info->IsComfortNoise());
+  RTC_DCHECK(new_decoder);
+  *new_decoder = false;
+  if (active_decoder_type_ < 0) {
+    // This is the first active decoder.
+    *new_decoder = true;
+  } else if (active_decoder_type_ != rtp_payload_type) {
+    // Moving from one active decoder to another. Delete the first one.
+    const DecoderInfo* old_info = GetDecoderInfo(active_decoder_type_);
+    RTC_DCHECK(old_info);
+    old_info->DropDecoder();
+    *new_decoder = true;
+  }
+  active_decoder_type_ = rtp_payload_type;
+  return kOK;
+}
+
+AudioDecoder* DecoderDatabase::GetActiveDecoder() const {
+  if (active_decoder_type_ < 0) {
+    // No active decoder.
+    return NULL;
+  }
+  return GetDecoder(active_decoder_type_);
+}
+
+int DecoderDatabase::SetActiveCngDecoder(uint8_t rtp_payload_type) {
+  // Check that `rtp_payload_type` exists in the database.
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  if (!info) {
+    // Decoder not found.
+    return kDecoderNotFound;
+  }
+  if (active_cng_decoder_type_ >= 0 &&
+      active_cng_decoder_type_ != rtp_payload_type) {
+    // Moving from one active CNG decoder to another. Delete the first one.
+    RTC_DCHECK(active_cng_decoder_);
+    active_cng_decoder_.reset();
+  }
+  active_cng_decoder_type_ = rtp_payload_type;
+  return kOK;
+}
+
+ComfortNoiseDecoder* DecoderDatabase::GetActiveCngDecoder() const {
+  if (active_cng_decoder_type_ < 0) {
+    // No active CNG decoder.
+    return NULL;
+  }
+  if (!active_cng_decoder_) {
+    active_cng_decoder_.reset(new ComfortNoiseDecoder);
+  }
+  return active_cng_decoder_.get();
+}
+
+AudioDecoder* DecoderDatabase::GetDecoder(uint8_t rtp_payload_type) const {
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  return info ? info->GetDecoder() : nullptr;
+}
+
+bool DecoderDatabase::IsComfortNoise(uint8_t rtp_payload_type) const {
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  return info && info->IsComfortNoise();
+}
+
+bool DecoderDatabase::IsDtmf(uint8_t rtp_payload_type) const {
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  return info && info->IsDtmf();
+}
+
+bool DecoderDatabase::IsRed(uint8_t rtp_payload_type) const {
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  return info && info->IsRed();
+}
+
+int DecoderDatabase::CheckPayloadTypes(const PacketList& packet_list) const {
+  PacketList::const_iterator it;
+  for (it = packet_list.begin(); it != packet_list.end(); ++it) {
+    if (!GetDecoderInfo(it->payload_type)) {
+      // Payload type is not found.
+      RTC_LOG(LS_WARNING) << "CheckPayloadTypes: unknown RTP payload type "
+                          << static_cast<int>(it->payload_type);
+      return kDecoderNotFound;
+    }
+  }
+  return kOK;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.h b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.h
new file mode 100644
index 0000000000..8cf2019135
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.h
@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
+#define MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/scoped_refptr.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+class DecoderDatabase {
+ public:
+  enum DatabaseReturnCodes {
+    kOK = 0,
+    kInvalidRtpPayloadType = -1,
+    kCodecNotSupported = -2,
+    kInvalidSampleRate = -3,
+    kDecoderExists = -4,
+    kDecoderNotFound = -5,
+    kInvalidPointer = -6
+  };
+
+  // Class that stores decoder info in the database.
+  class DecoderInfo {
+   public:
+    DecoderInfo(const SdpAudioFormat& audio_format,
+                absl::optional<AudioCodecPairId> codec_pair_id,
+                AudioDecoderFactory* factory,
+                absl::string_view codec_name);
+    explicit DecoderInfo(const SdpAudioFormat& audio_format,
+                         absl::optional<AudioCodecPairId> codec_pair_id,
+                         AudioDecoderFactory* factory = nullptr);
+    DecoderInfo(DecoderInfo&&);
+    ~DecoderInfo();
+
+    // Get the AudioDecoder object, creating it first if necessary.
+    AudioDecoder* GetDecoder() const;
+
+    // Delete the AudioDecoder object, unless it's external. (This means we can
+    // always recreate it later if we need it.)
+    void DropDecoder() const { decoder_.reset(); }
+
+    int SampleRateHz() const {
+      if (IsDtmf()) {
+        // DTMF has a 1:1 mapping between clock rate and sample rate.
+        return audio_format_.clockrate_hz;
+      }
+      const AudioDecoder* decoder = GetDecoder();
+      RTC_DCHECK_EQ(1, !!decoder + !!cng_decoder_);
+      return decoder ? decoder->SampleRateHz() : cng_decoder_->sample_rate_hz;
+    }
+
+    const SdpAudioFormat& GetFormat() const { return audio_format_; }
+
+    // Returns true if the decoder's format is comfort noise.
+    bool IsComfortNoise() const {
+      RTC_DCHECK_EQ(!!cng_decoder_, subtype_ == Subtype::kComfortNoise);
+      return subtype_ == Subtype::kComfortNoise;
+    }
+
+    // Returns true if the decoder's format is DTMF.
+    bool IsDtmf() const { return subtype_ == Subtype::kDtmf; }
+
+    // Returns true if the decoder's format is RED.
+    bool IsRed() const { return subtype_ == Subtype::kRed; }
+
+    // Returns true if the decoder's format is named `name`.
+    bool IsType(absl::string_view name) const;
+
+    const std::string& get_name() const { return name_; }
+
+   private:
+    // TODO(ossu): `name_` is kept here while we retain the old external
+    //             decoder interface. Remove this once using an
+    //             AudioDecoderFactory has supplanted the old functionality.
+    const std::string name_;
+
+    const SdpAudioFormat audio_format_;
+    const absl::optional<AudioCodecPairId> codec_pair_id_;
+    AudioDecoderFactory* const factory_;
+    mutable std::unique_ptr<AudioDecoder> decoder_;
+
+    // Set iff this is a comfort noise decoder.
+    struct CngDecoder {
+      static absl::optional<CngDecoder> Create(const SdpAudioFormat& format);
+      int sample_rate_hz;
+    };
+    const absl::optional<CngDecoder> cng_decoder_;
+
+    enum class Subtype : int8_t { kNormal, kComfortNoise, kDtmf, kRed };
+
+    static Subtype SubtypeFromFormat(const SdpAudioFormat& format);
+
+    const Subtype subtype_;
+  };
+
+  // Maximum value for 8 bits, and an invalid RTP payload type (since it is
+  // only 7 bits).
+  static const uint8_t kRtpPayloadTypeError = 0xFF;
+
+  DecoderDatabase(
+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+      absl::optional<AudioCodecPairId> codec_pair_id);
+
+  virtual ~DecoderDatabase();
+
+  DecoderDatabase(const DecoderDatabase&) = delete;
+  DecoderDatabase& operator=(const DecoderDatabase&) = delete;
+
+  // Returns true if the database is empty.
+  virtual bool Empty() const;
+
+  // Returns the number of decoders registered in the database.
+  virtual int Size() const;
+
+  // Replaces the existing set of decoders with the given set. Returns the
+  // payload types that were reassigned or removed while doing so.
+  virtual std::vector<int> SetCodecs(
+      const std::map<int, SdpAudioFormat>& codecs);
+
+  // Registers a decoder for the given payload type. Returns kOK on success;
+  // otherwise an error code.
+  virtual int RegisterPayload(int rtp_payload_type,
+                              const SdpAudioFormat& audio_format);
+
+  // Removes the entry for `rtp_payload_type` from the database.
+  // Returns kDecoderNotFound or kOK depending on the outcome of the operation.
+  virtual int Remove(uint8_t rtp_payload_type);
+
+  // Remove all entries.
+  virtual void RemoveAll();
+
+  // Returns a pointer to the DecoderInfo struct for `rtp_payload_type`. If
+  // no decoder is registered with that `rtp_payload_type`, NULL is returned.
+  virtual const DecoderInfo* GetDecoderInfo(uint8_t rtp_payload_type) const;
+
+  // Sets the active decoder to be `rtp_payload_type`. If this call results in a
+  // change of active decoder, `new_decoder` is set to true. The previous active
+  // decoder's AudioDecoder object is deleted.
+  virtual int SetActiveDecoder(uint8_t rtp_payload_type, bool* new_decoder);
+
+  // Returns the current active decoder, or NULL if no active decoder exists.
+  virtual AudioDecoder* GetActiveDecoder() const;
+
+  // Sets the active comfort noise decoder to be `rtp_payload_type`. If this
+  // call results in a change of active comfort noise decoder, the previous
+  // active decoder's AudioDecoder object is deleted.
+  virtual int SetActiveCngDecoder(uint8_t rtp_payload_type);
+
+  // Returns the current active comfort noise decoder, or NULL if no active
+  // comfort noise decoder exists.
+  virtual ComfortNoiseDecoder* GetActiveCngDecoder() const;
+
+  // The following are utility methods: they will look up DecoderInfo through
+  // GetDecoderInfo and call the respective method on that info object, if it
+  // exists.
+
+  // Returns a pointer to the AudioDecoder object associated with
+  // `rtp_payload_type`, or NULL if none is registered. If the AudioDecoder
+  // object does not exist for that decoder, the object is created.
+  AudioDecoder* GetDecoder(uint8_t rtp_payload_type) const;
+
+  // Returns true if `rtp_payload_type` is registered as comfort noise.
+  bool IsComfortNoise(uint8_t rtp_payload_type) const;
+
+  // Returns true if `rtp_payload_type` is registered as DTMF.
+  bool IsDtmf(uint8_t rtp_payload_type) const;
+
+  // Returns true if `rtp_payload_type` is registered as RED.
+  bool IsRed(uint8_t rtp_payload_type) const;
+
+  // Returns kOK if all packets in `packet_list` carry payload types that are
+  // registered in the database. Otherwise, returns kDecoderNotFound.
+  int CheckPayloadTypes(const PacketList& packet_list) const;
+
+ private:
+  typedef std::map<uint8_t, DecoderInfo> DecoderMap;
+
+  DecoderMap decoders_;
+  int active_decoder_type_;
+  int active_cng_decoder_type_;
+  mutable std::unique_ptr<ComfortNoiseDecoder> active_cng_decoder_;
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  const absl::optional<AudioCodecPairId> codec_pair_id_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
new file mode 100644
index 0000000000..445c21924b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -0,0 +1,227 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/decoder_database.h"
+
+#include <stdlib.h>
+
+#include <string>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/mock_audio_decoder_factory.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+
+namespace webrtc {
+
+TEST(DecoderDatabase, CreateAndDestroy) {
+  DecoderDatabase db(rtc::make_ref_counted<MockAudioDecoderFactory>(),
+                     absl::nullopt);
+  EXPECT_EQ(0, db.Size());
+  EXPECT_TRUE(db.Empty());
+}
+
+TEST(DecoderDatabase, InsertAndRemove) {
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  DecoderDatabase db(factory, absl::nullopt);
+  const uint8_t kPayloadType = 0;
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, SdpAudioFormat(kCodecName, 8000, 1)));
+  EXPECT_EQ(1, db.Size());
+  EXPECT_FALSE(db.Empty());
+  EXPECT_EQ(DecoderDatabase::kOK, db.Remove(kPayloadType));
+  EXPECT_EQ(0, db.Size());
+  EXPECT_TRUE(db.Empty());
+}
+
+TEST(DecoderDatabase, InsertAndRemoveAll) {
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  DecoderDatabase db(factory, absl::nullopt);
+  const std::string kCodecName1 = "Robert\'); DROP TABLE Students;";
+  const std::string kCodecName2 = "https://xkcd.com/327/";
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(0, SdpAudioFormat(kCodecName1, 8000, 1)));
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(1, SdpAudioFormat(kCodecName2, 8000, 1)));
+  EXPECT_EQ(2, db.Size());
+  EXPECT_FALSE(db.Empty());
+  db.RemoveAll();
+  EXPECT_EQ(0, db.Size());
+  EXPECT_TRUE(db.Empty());
+}
+
+TEST(DecoderDatabase, GetDecoderInfo) {
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  auto* decoder = new MockAudioDecoder;
+  EXPECT_CALL(*factory, MakeAudioDecoderMock(_, _, _))
+      .WillOnce(Invoke([decoder](const SdpAudioFormat& format,
+                                 absl::optional<AudioCodecPairId> codec_pair_id,
+                                 std::unique_ptr<AudioDecoder>* dec) {
+        EXPECT_EQ("pcmu", format.name);
+        dec->reset(decoder);
+      }));
+  DecoderDatabase db(factory, absl::nullopt);
+  const uint8_t kPayloadType = 0;
+  const std::string kCodecName = "pcmu";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, SdpAudioFormat(kCodecName, 8000, 1)));
+  const DecoderDatabase::DecoderInfo* info;
+  info = db.GetDecoderInfo(kPayloadType);
+  ASSERT_TRUE(info != NULL);
+  EXPECT_TRUE(info->IsType("pcmu"));
+  EXPECT_EQ(kCodecName, info->get_name());
+  EXPECT_EQ(decoder, db.GetDecoder(kPayloadType));
+  info = db.GetDecoderInfo(kPayloadType + 1);  // Other payload type.
+  EXPECT_TRUE(info == NULL);                   // Should not be found.
+}
+
+TEST(DecoderDatabase, GetDecoder) {
+  DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), absl::nullopt);
+  const uint8_t kPayloadType = 0;
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(kPayloadType, SdpAudioFormat("l16", 8000, 1)));
+  AudioDecoder* dec = db.GetDecoder(kPayloadType);
+  ASSERT_TRUE(dec != NULL);
+}
+
+TEST(DecoderDatabase, TypeTests) {
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  DecoderDatabase db(factory, absl::nullopt);
+  const uint8_t kPayloadTypePcmU = 0;
+  const uint8_t kPayloadTypeCng = 13;
+  const uint8_t kPayloadTypeDtmf = 100;
+  const uint8_t kPayloadTypeRed = 101;
+  const uint8_t kPayloadNotUsed = 102;
+  // Load into database.
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypePcmU, SdpAudioFormat("pcmu", 8000, 1)));
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(kPayloadTypeCng, SdpAudioFormat("cn", 8000, 1)));
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(kPayloadTypeDtmf,
+                               SdpAudioFormat("telephone-event", 8000, 1)));
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypeRed, SdpAudioFormat("red", 8000, 1)));
+  EXPECT_EQ(4, db.Size());
+  // Test.
+  EXPECT_FALSE(db.IsComfortNoise(kPayloadNotUsed));
+  EXPECT_FALSE(db.IsDtmf(kPayloadNotUsed));
+  EXPECT_FALSE(db.IsRed(kPayloadNotUsed));
+  EXPECT_FALSE(db.IsComfortNoise(kPayloadTypePcmU));
+  EXPECT_FALSE(db.IsDtmf(kPayloadTypePcmU));
+  EXPECT_FALSE(db.IsRed(kPayloadTypePcmU));
+  EXPECT_TRUE(db.IsComfortNoise(kPayloadTypeCng));
+  EXPECT_TRUE(db.IsDtmf(kPayloadTypeDtmf));
+  EXPECT_TRUE(db.IsRed(kPayloadTypeRed));
+}
+
+TEST(DecoderDatabase, CheckPayloadTypes) {
+  constexpr int kNumPayloads = 10;
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  DecoderDatabase db(factory, absl::nullopt);
+  // Load a number of payloads into the database. Payload types are 0, 1, ...,
+  // while the decoder type is the same for all payload types (this does not
+  // matter for the test).
+  for (uint8_t payload_type = 0; payload_type < kNumPayloads; ++payload_type) {
+    EXPECT_EQ(
+        DecoderDatabase::kOK,
+        db.RegisterPayload(payload_type, SdpAudioFormat("pcmu", 8000, 1)));
+  }
+  PacketList packet_list;
+  for (int i = 0; i < kNumPayloads + 1; ++i) {
+    // Create packet with payload type `i`. The last packet will have a payload
+    // type that is not registered in the decoder database.
+    Packet packet;
+    packet.payload_type = i;
+    packet_list.push_back(std::move(packet));
+  }
+
+  // Expect to return false, since the last packet is of an unknown type.
+  EXPECT_EQ(DecoderDatabase::kDecoderNotFound,
+            db.CheckPayloadTypes(packet_list));
+
+  packet_list.pop_back();  // Remove the unknown one.
+
+  EXPECT_EQ(DecoderDatabase::kOK, db.CheckPayloadTypes(packet_list));
+
+  // Delete all packets.
+  PacketList::iterator it = packet_list.begin();
+  while (it != packet_list.end()) {
+    it = packet_list.erase(it);
+  }
+}
+
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#define IF_ISAC(x) x
+#else
+#define IF_ISAC(x) DISABLED_##x
+#endif
+
+// Test the methods for setting and getting active speech and CNG decoders.
+TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
+  DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), absl::nullopt);
+  // Load payload types.
+  ASSERT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(0, SdpAudioFormat("pcmu", 8000, 1)));
+  ASSERT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(103, SdpAudioFormat("isac", 16000, 1)));
+  ASSERT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(13, SdpAudioFormat("cn", 8000, 1)));
+  // Verify that no decoders are active from the start.
+  EXPECT_EQ(NULL, db.GetActiveDecoder());
+  EXPECT_EQ(NULL, db.GetActiveCngDecoder());
+
+  // Set active speech codec.
+  bool changed;  // Should be true when the active decoder changed.
+  EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveDecoder(0, &changed));
+  EXPECT_TRUE(changed);
+  AudioDecoder* decoder = db.GetActiveDecoder();
+  ASSERT_FALSE(decoder == NULL);  // Should get a decoder here.
+
+  // Set the same again. Expect no change.
+  EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveDecoder(0, &changed));
+  EXPECT_FALSE(changed);
+  decoder = db.GetActiveDecoder();
+  ASSERT_FALSE(decoder == NULL);  // Should get a decoder here.
+
+  // Change active decoder.
+  EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveDecoder(103, &changed));
+  EXPECT_TRUE(changed);
+  decoder = db.GetActiveDecoder();
+  ASSERT_FALSE(decoder == NULL);  // Should get a decoder here.
+
+  // Remove the active decoder, and verify that the active becomes NULL.
+  EXPECT_EQ(DecoderDatabase::kOK, db.Remove(103));
+  EXPECT_EQ(NULL, db.GetActiveDecoder());
+
+  // Set active CNG codec.
+  EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveCngDecoder(13));
+  ComfortNoiseDecoder* cng = db.GetActiveCngDecoder();
+  ASSERT_FALSE(cng == NULL);  // Should get a decoder here.
+
+  // Remove the active CNG decoder, and verify that the active becomes NULL.
+  EXPECT_EQ(DecoderDatabase::kOK, db.Remove(13));
+  EXPECT_EQ(NULL, db.GetActiveCngDecoder());
+
+  // Try to set non-existing codecs as active.
+  EXPECT_EQ(DecoderDatabase::kDecoderNotFound,
+            db.SetActiveDecoder(17, &changed));
+  EXPECT_EQ(DecoderDatabase::kDecoderNotFound, db.SetActiveCngDecoder(17));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.cc b/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.cc
new file mode 100644
index 0000000000..487450fe0f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.cc
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+
+#include <utility>
+
+#include "modules/audio_coding/neteq/neteq_impl.h"
+
+namespace webrtc {
+
+DefaultNetEqFactory::DefaultNetEqFactory() = default;
+DefaultNetEqFactory::~DefaultNetEqFactory() = default;
+
+std::unique_ptr<NetEq> DefaultNetEqFactory::CreateNetEq(
+    const NetEq::Config& config,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+    Clock* clock) const {
+  return std::make_unique<NetEqImpl>(
+      config, NetEqImpl::Dependencies(config, clock, decoder_factory,
+                                      controller_factory_));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.h b/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.h
new file mode 100644
index 0000000000..24d2bae419
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_DEFAULT_NETEQ_FACTORY_H_
+#define MODULES_AUDIO_CODING_NETEQ_DEFAULT_NETEQ_FACTORY_H_
+
+#include <memory>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/neteq/default_neteq_controller_factory.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/scoped_refptr.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class DefaultNetEqFactory : public NetEqFactory {
+ public:
+  DefaultNetEqFactory();
+  ~DefaultNetEqFactory() override;
+  DefaultNetEqFactory(const DefaultNetEqFactory&) = delete;
+  DefaultNetEqFactory& operator=(const DefaultNetEqFactory&) = delete;
+
+  std::unique_ptr<NetEq> CreateNetEq(
+      const NetEq::Config& config,
+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+      Clock* clock) const override;
+
+ private:
+  const DefaultNetEqControllerFactory controller_factory_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DEFAULT_NETEQ_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc
new file mode 100644
index 0000000000..bf3a0f18a1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -0,0 +1,207 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/delay_manager.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <algorithm>
+#include <memory>
+#include <numeric>
+#include <string>
+
+#include "modules/include/module_common_types_public.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kMinBaseMinimumDelayMs = 0;
+constexpr int kMaxBaseMinimumDelayMs = 10000;
+constexpr int kStartDelayMs = 80;
+
+std::unique_ptr<ReorderOptimizer> MaybeCreateReorderOptimizer(
+    const DelayManager::Config& config) {
+  if (!config.use_reorder_optimizer) {
+    return nullptr;
+  }
+  return std::make_unique<ReorderOptimizer>(
+      (1 << 15) * config.reorder_forget_factor, config.ms_per_loss_percent,
+      config.start_forget_weight);
+}
+
+}  // namespace
+
+DelayManager::Config::Config() {
+  StructParametersParser::Create(                       //
+      "quantile", &quantile,                            //
+      "forget_factor", &forget_factor,                  //
+      "start_forget_weight", &start_forget_weight,      //
+      "resample_interval_ms", &resample_interval_ms,    //
+      "use_reorder_optimizer", &use_reorder_optimizer,  //
+      "reorder_forget_factor", &reorder_forget_factor,  //
+      "ms_per_loss_percent", &ms_per_loss_percent)
+      ->Parse(webrtc::field_trial::FindFullName(
+          "WebRTC-Audio-NetEqDelayManagerConfig"));
+}
+
+void DelayManager::Config::Log() {
+  RTC_LOG(LS_INFO) << "Delay manager config:"
+                      " quantile="
+                   << quantile << " forget_factor=" << forget_factor
+                   << " start_forget_weight=" << start_forget_weight.value_or(0)
+                   << " resample_interval_ms="
+                   << resample_interval_ms.value_or(0)
+                   << " use_reorder_optimizer=" << use_reorder_optimizer
+                   << " reorder_forget_factor=" << reorder_forget_factor
+                   << " ms_per_loss_percent=" << ms_per_loss_percent;
+}
+
+DelayManager::DelayManager(const Config& config, const TickTimer* tick_timer)
+    : max_packets_in_buffer_(config.max_packets_in_buffer),
+      underrun_optimizer_(tick_timer,
+                          (1 << 30) * config.quantile,
+                          (1 << 15) * config.forget_factor,
+                          config.start_forget_weight,
+                          config.resample_interval_ms),
+      reorder_optimizer_(MaybeCreateReorderOptimizer(config)),
+      base_minimum_delay_ms_(config.base_minimum_delay_ms),
+      effective_minimum_delay_ms_(config.base_minimum_delay_ms),
+      minimum_delay_ms_(0),
+      maximum_delay_ms_(0),
+      target_level_ms_(kStartDelayMs) {
+  RTC_DCHECK_GE(base_minimum_delay_ms_, 0);
+
+  Reset();
+}
+
+DelayManager::~DelayManager() {}
+
+void DelayManager::Update(int arrival_delay_ms, bool reordered) {
+  if (!reorder_optimizer_ || !reordered) {
+    underrun_optimizer_.Update(arrival_delay_ms);
+  }
+  target_level_ms_ =
+      underrun_optimizer_.GetOptimalDelayMs().value_or(kStartDelayMs);
+  if (reorder_optimizer_) {
+    reorder_optimizer_->Update(arrival_delay_ms, reordered, target_level_ms_);
+    target_level_ms_ = std::max(
+        target_level_ms_, reorder_optimizer_->GetOptimalDelayMs().value_or(0));
+  }
+  unlimited_target_level_ms_ = target_level_ms_;
+  target_level_ms_ = std::max(target_level_ms_, effective_minimum_delay_ms_);
+  if (maximum_delay_ms_ > 0) {
+    target_level_ms_ = std::min(target_level_ms_, maximum_delay_ms_);
+  }
+  if (packet_len_ms_ > 0) {
+    // Limit to 75% of maximum buffer size.
+    target_level_ms_ = std::min(
+        target_level_ms_, 3 * max_packets_in_buffer_ * packet_len_ms_ / 4);
+  }
+}
+
+int DelayManager::SetPacketAudioLength(int length_ms) {
+  if (length_ms <= 0) {
+    RTC_LOG_F(LS_ERROR) << "length_ms = " << length_ms;
+    return -1;
+  }
+  packet_len_ms_ = length_ms;
+  return 0;
+}
+
+void DelayManager::Reset() {
+  packet_len_ms_ = 0;
+  underrun_optimizer_.Reset();
+  target_level_ms_ = kStartDelayMs;
+  if (reorder_optimizer_) {
+    reorder_optimizer_->Reset();
+  }
+}
+
+int DelayManager::TargetDelayMs() const {
+  return target_level_ms_;
+}
+
+int DelayManager::UnlimitedTargetLevelMs() const {
+  return unlimited_target_level_ms_;
+}
+
+bool DelayManager::IsValidMinimumDelay(int delay_ms) const {
+  return 0 <= delay_ms && delay_ms <= MinimumDelayUpperBound();
+}
+
+bool DelayManager::IsValidBaseMinimumDelay(int delay_ms) const {
+  return kMinBaseMinimumDelayMs <= delay_ms &&
+         delay_ms <= kMaxBaseMinimumDelayMs;
+}
+
+bool DelayManager::SetMinimumDelay(int delay_ms) {
+  if (!IsValidMinimumDelay(delay_ms)) {
+    return false;
+  }
+
+  minimum_delay_ms_ = delay_ms;
+  UpdateEffectiveMinimumDelay();
+  return true;
+}
+
+bool DelayManager::SetMaximumDelay(int delay_ms) {
+  // If `delay_ms` is zero then it unsets the maximum delay and target level is
+  // unconstrained by maximum delay.
+  if (delay_ms != 0 && delay_ms < minimum_delay_ms_) {
+    // Maximum delay shouldn't be less than minimum delay or less than a packet.
+    return false;
+  }
+
+  maximum_delay_ms_ = delay_ms;
+  UpdateEffectiveMinimumDelay();
+  return true;
+}
+
+bool DelayManager::SetBaseMinimumDelay(int delay_ms) {
+  if (!IsValidBaseMinimumDelay(delay_ms)) {
+    return false;
+  }
+
+  base_minimum_delay_ms_ = delay_ms;
+  UpdateEffectiveMinimumDelay();
+  return true;
+}
+
+int DelayManager::GetBaseMinimumDelay() const {
+  return base_minimum_delay_ms_;
+}
+
+void DelayManager::UpdateEffectiveMinimumDelay() {
+  // Clamp `base_minimum_delay_ms_` into the range which can be effectively
+  // used.
+  const int base_minimum_delay_ms =
+      rtc::SafeClamp(base_minimum_delay_ms_, 0, MinimumDelayUpperBound());
+  effective_minimum_delay_ms_ =
+      std::max(minimum_delay_ms_, base_minimum_delay_ms);
+}
+
+int DelayManager::MinimumDelayUpperBound() const {
+  // Choose the lowest possible bound discarding 0 cases which mean the value
+  // is not set and unconstrained.
+  int q75 = max_packets_in_buffer_ * packet_len_ms_ * 3 / 4;
+  q75 = q75 > 0 ? q75 : kMaxBaseMinimumDelayMs;
+  const int maximum_delay_ms =
+      maximum_delay_ms_ > 0 ? maximum_delay_ms_ : kMaxBaseMinimumDelayMs;
+  return std::min(maximum_delay_ms, q75);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.h b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.h
new file mode 100644
index 0000000000..a333681535
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.h
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
+
+#include <string.h>  // Provide access to size_t.
+
+#include <deque>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/histogram.h"
+#include "modules/audio_coding/neteq/reorder_optimizer.h"
+#include "modules/audio_coding/neteq/underrun_optimizer.h"
+
+namespace webrtc {
+
+class DelayManager {
+ public:
+  struct Config {
+    Config();
+    void Log();
+
+    // Options that can be configured via field trial.
+    double quantile = 0.95;
+    double forget_factor = 0.983;
+    absl::optional<double> start_forget_weight = 2;
+    absl::optional<int> resample_interval_ms = 500;
+
+    bool use_reorder_optimizer = true;
+    double reorder_forget_factor = 0.9993;
+    int ms_per_loss_percent = 20;
+
+    // Options that are externally populated.
+    int max_packets_in_buffer = 200;
+    int base_minimum_delay_ms = 0;
+  };
+
+  DelayManager(const Config& config, const TickTimer* tick_timer);
+
+  virtual ~DelayManager();
+
+  DelayManager(const DelayManager&) = delete;
+  DelayManager& operator=(const DelayManager&) = delete;
+
+  // Updates the delay manager that a new packet arrived with delay
+  // `arrival_delay_ms`. This updates the statistics and a new target buffer
+  // level is calculated. The `reordered` flag indicates if the packet was
+  // reordered.
+  virtual void Update(int arrival_delay_ms, bool reordered);
+
+  // Resets all state.
+  virtual void Reset();
+
+  // Gets the target buffer level in milliseconds. If a minimum or maximum delay
+  // has been set, the target delay reported here also respects the configured
+  // min/max delay.
+  virtual int TargetDelayMs() const;
+
+  // Reports the target delay that would be used if no minimum/maximum delay
+  // would be set.
+  virtual int UnlimitedTargetLevelMs() const;
+
+  // Notifies the DelayManager of how much audio data is carried in each packet.
+  virtual int SetPacketAudioLength(int length_ms);
+
+  // Accessors and mutators.
+  // Assuming `delay` is in valid range.
+  virtual bool SetMinimumDelay(int delay_ms);
+  virtual bool SetMaximumDelay(int delay_ms);
+  virtual bool SetBaseMinimumDelay(int delay_ms);
+  virtual int GetBaseMinimumDelay() const;
+
+  // These accessors are only intended for testing purposes.
+  int effective_minimum_delay_ms_for_test() const {
+    return effective_minimum_delay_ms_;
+  }
+
+ private:
+  // Provides value which minimum delay can't exceed based on current buffer
+  // size and given `maximum_delay_ms_`. Lower bound is a constant 0.
+  int MinimumDelayUpperBound() const;
+
+  // Updates `effective_minimum_delay_ms_` delay based on current
+  // `minimum_delay_ms_`, `base_minimum_delay_ms_` and `maximum_delay_ms_`
+  // and buffer size.
+  void UpdateEffectiveMinimumDelay();
+
+  // Makes sure that `delay_ms` is less than maximum delay, if any maximum
+  // is set. Also, if possible check `delay_ms` to be less than 75% of
+  // `max_packets_in_buffer_`.
+  bool IsValidMinimumDelay(int delay_ms) const;
+
+  bool IsValidBaseMinimumDelay(int delay_ms) const;
+
+  // TODO(jakobi): set maximum buffer delay instead of number of packets.
+  const int max_packets_in_buffer_;
+  UnderrunOptimizer underrun_optimizer_;
+  std::unique_ptr<ReorderOptimizer> reorder_optimizer_;
+
+  int base_minimum_delay_ms_;
+  int effective_minimum_delay_ms_;  // Used as lower bound for target delay.
+  int minimum_delay_ms_;            // Externally set minimum delay.
+  int maximum_delay_ms_;            // Externally set maximum allowed delay.
+
+  int packet_len_ms_ = 0;
+  int target_level_ms_ = 0;  // Currently preferred buffer level.
+  int unlimited_target_level_ms_ = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
new file mode 100644
index 0000000000..da5f53188c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -0,0 +1,246 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for DelayManager class.
+
+#include "modules/audio_coding/neteq/delay_manager.h"
+
+#include <math.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/histogram.h"
+#include "modules/audio_coding/neteq/mock/mock_histogram.h"
+#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
+#include "rtc_base/checks.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kMaxNumberOfPackets = 200;
+constexpr int kTimeStepMs = 10;
+constexpr int kFrameSizeMs = 20;
+constexpr int kMaxBufferSizeMs = kMaxNumberOfPackets * kFrameSizeMs;
+
+}  // namespace
+
+class DelayManagerTest : public ::testing::Test {
+ protected:
+  DelayManagerTest();
+  virtual void SetUp();
+  void Update(int delay);
+  void IncreaseTime(int inc_ms);
+
+  TickTimer tick_timer_;
+  DelayManager dm_;
+};
+
+DelayManagerTest::DelayManagerTest()
+    : dm_(DelayManager::Config(), &tick_timer_) {}
+
+void DelayManagerTest::SetUp() {
+  dm_.SetPacketAudioLength(kFrameSizeMs);
+}
+
+void DelayManagerTest::Update(int delay) {
+  dm_.Update(delay, false);
+}
+
+void DelayManagerTest::IncreaseTime(int inc_ms) {
+  for (int t = 0; t < inc_ms; t += kTimeStepMs) {
+    tick_timer_.Increment();
+  }
+}
+
+TEST_F(DelayManagerTest, CreateAndDestroy) {
+  // Nothing to do here. The test fixture creates and destroys the DelayManager
+  // object.
+}
+
+TEST_F(DelayManagerTest, UpdateNormal) {
+  for (int i = 0; i < 50; ++i) {
+    Update(0);
+    IncreaseTime(kFrameSizeMs);
+  }
+  EXPECT_EQ(20, dm_.TargetDelayMs());
+}
+
+TEST_F(DelayManagerTest, MaxDelay) {
+  Update(0);
+  const int kMaxDelayMs = 60;
+  EXPECT_GT(dm_.TargetDelayMs(), kMaxDelayMs);
+  EXPECT_TRUE(dm_.SetMaximumDelay(kMaxDelayMs));
+  Update(0);
+  EXPECT_EQ(kMaxDelayMs, dm_.TargetDelayMs());
+}
+
+TEST_F(DelayManagerTest, MinDelay) {
+  Update(0);
+  int kMinDelayMs = 7 * kFrameSizeMs;
+  EXPECT_LT(dm_.TargetDelayMs(), kMinDelayMs);
+  dm_.SetMinimumDelay(kMinDelayMs);
+  IncreaseTime(kFrameSizeMs);
+  Update(0);
+  EXPECT_EQ(kMinDelayMs, dm_.TargetDelayMs());
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayCheckValidRange) {
+  // Base minimum delay should be between [0, 10000] milliseconds.
+  EXPECT_FALSE(dm_.SetBaseMinimumDelay(-1));
+  EXPECT_FALSE(dm_.SetBaseMinimumDelay(10001));
+  EXPECT_EQ(dm_.GetBaseMinimumDelay(), 0);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(7999));
+  EXPECT_EQ(dm_.GetBaseMinimumDelay(), 7999);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayLowerThanMinimumDelay) {
+  constexpr int kBaseMinimumDelayMs = 100;
+  constexpr int kMinimumDelayMs = 200;
+
+  // Base minimum delay sets lower bound on minimum. That is why when base
+  // minimum delay is lower than minimum delay we use minimum delay.
+  RTC_DCHECK_LT(kBaseMinimumDelayMs, kMinimumDelayMs);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMinimumDelayMs);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayGreaterThanMinimumDelay) {
+  constexpr int kBaseMinimumDelayMs = 70;
+  constexpr int kMinimumDelayMs = 30;
+
+  // Base minimum delay sets lower bound on minimum. That is why when base
+  // minimum delay is greater than minimum delay we use base minimum delay.
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMinimumDelayMs);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kBaseMinimumDelayMs);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayGreaterThanBufferSize) {
+  constexpr int kBaseMinimumDelayMs = kMaxBufferSizeMs + 1;
+  constexpr int kMinimumDelayMs = 12;
+  constexpr int kMaximumDelayMs = 20;
+  constexpr int kMaxBufferSizeMsQ75 = 3 * kMaxBufferSizeMs / 4;
+
+  EXPECT_TRUE(dm_.SetMaximumDelay(kMaximumDelayMs));
+
+  // Base minimum delay is greater than minimum delay, that is why we clamp
+  // it to current the highest possible value which is maximum delay.
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMinimumDelayMs);
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMaxBufferSizeMs);
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMaximumDelayMs);
+  RTC_DCHECK_LT(kMaximumDelayMs, kMaxBufferSizeMsQ75);
+
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+
+  // Unset maximum value.
+  EXPECT_TRUE(dm_.SetMaximumDelay(0));
+
+  // With maximum value unset, the highest possible value now is 75% of
+  // currently possible maximum buffer size.
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMaxBufferSizeMsQ75);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayGreaterThanMaximumDelay) {
+  constexpr int kMaximumDelayMs = 400;
+  constexpr int kBaseMinimumDelayMs = kMaximumDelayMs + 1;
+  constexpr int kMinimumDelayMs = 20;
+
+  // Base minimum delay is greater than minimum delay, that is why we clamp
+  // it to current the highest possible value which is kMaximumDelayMs.
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMinimumDelayMs);
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMaximumDelayMs);
+  RTC_DCHECK_LT(kMaximumDelayMs, kMaxBufferSizeMs);
+
+  EXPECT_TRUE(dm_.SetMaximumDelay(kMaximumDelayMs));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMaximumDelayMs);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayLowerThanMaxSize) {
+  constexpr int kMaximumDelayMs = 400;
+  constexpr int kBaseMinimumDelayMs = kMaximumDelayMs - 1;
+  constexpr int kMinimumDelayMs = 20;
+
+  // Base minimum delay is greater than minimum delay, and lower than maximum
+  // delays that is why it is used.
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMinimumDelayMs);
+  RTC_DCHECK_LT(kBaseMinimumDelayMs, kMaximumDelayMs);
+
+  EXPECT_TRUE(dm_.SetMaximumDelay(kMaximumDelayMs));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kBaseMinimumDelayMs);
+}
+
+TEST_F(DelayManagerTest, MinimumDelayMemorization) {
+  // Check that when we increase base minimum delay to value higher than
+  // minimum delay then minimum delay is still memorized. This allows to
+  // restore effective minimum delay to memorized minimum delay value when we
+  // decrease base minimum delay.
+  constexpr int kBaseMinimumDelayMsLow = 10;
+  constexpr int kMinimumDelayMs = 20;
+  constexpr int kBaseMinimumDelayMsHigh = 30;
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMsLow));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  // Minimum delay is used as it is higher than base minimum delay.
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMinimumDelayMs);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMsHigh));
+  // Base minimum delay is used as it is now higher than minimum delay.
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kBaseMinimumDelayMsHigh);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMsLow));
+  // Check that minimum delay is memorized and is used again.
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMinimumDelayMs);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelay) {
+  // First packet arrival.
+  Update(0);
+
+  constexpr int kBaseMinimumDelayMs = 7 * kFrameSizeMs;
+  EXPECT_LT(dm_.TargetDelayMs(), kBaseMinimumDelayMs);
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_EQ(dm_.GetBaseMinimumDelay(), kBaseMinimumDelayMs);
+
+  IncreaseTime(kFrameSizeMs);
+  Update(0);
+  EXPECT_EQ(dm_.GetBaseMinimumDelay(), kBaseMinimumDelayMs);
+  EXPECT_EQ(kBaseMinimumDelayMs, dm_.TargetDelayMs());
+}
+
+TEST_F(DelayManagerTest, Failures) {
+  // Wrong packet size.
+  EXPECT_EQ(-1, dm_.SetPacketAudioLength(0));
+  EXPECT_EQ(-1, dm_.SetPacketAudioLength(-1));
+
+  // Minimum delay higher than a maximum delay is not accepted.
+  EXPECT_TRUE(dm_.SetMaximumDelay(20));
+  EXPECT_FALSE(dm_.SetMinimumDelay(40));
+
+  // Maximum delay less than minimum delay is not accepted.
+  EXPECT_TRUE(dm_.SetMaximumDelay(100));
+  EXPECT_TRUE(dm_.SetMinimumDelay(80));
+  EXPECT_FALSE(dm_.SetMaximumDelay(60));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.cc
new file mode 100644
index 0000000000..a979f94214
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.cc
@@ -0,0 +1,373 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/dsp_helper.h"
+
+#include <string.h>  // Access to memset.
+
+#include <algorithm>  // Access to min, max.
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+// Table of constants used in method DspHelper::ParabolicFit().
+const int16_t DspHelper::kParabolaCoefficients[17][3] = {
+    {120, 32, 64},   {140, 44, 75},   {150, 50, 80},   {160, 57, 85},
+    {180, 72, 96},   {200, 89, 107},  {210, 98, 112},  {220, 108, 117},
+    {240, 128, 128}, {260, 150, 139}, {270, 162, 144}, {280, 174, 149},
+    {300, 200, 160}, {320, 228, 171}, {330, 242, 176}, {340, 257, 181},
+    {360, 288, 192}};
+
+// Filter coefficients used when downsampling from the indicated sample rates
+// (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12. The corresponding Q0
+// values are provided in the comments before each array.
+
+// Q0 values: {0.3, 0.4, 0.3}.
+const int16_t DspHelper::kDownsample8kHzTbl[3] = {1229, 1638, 1229};
+
+// Q0 values: {0.15, 0.2, 0.3, 0.2, 0.15}.
+const int16_t DspHelper::kDownsample16kHzTbl[5] = {614, 819, 1229, 819, 614};
+
+// Q0 values: {0.1425, 0.1251, 0.1525, 0.1628, 0.1525, 0.1251, 0.1425}.
+const int16_t DspHelper::kDownsample32kHzTbl[7] = {584, 512, 625, 667,
+                                                   625, 512, 584};
+
+// Q0 values: {0.2487, 0.0952, 0.1042, 0.1074, 0.1042, 0.0952, 0.2487}.
+const int16_t DspHelper::kDownsample48kHzTbl[7] = {1019, 390, 427, 440,
+                                                   427,  390, 1019};
+
+int DspHelper::RampSignal(const int16_t* input,
+                          size_t length,
+                          int factor,
+                          int increment,
+                          int16_t* output) {
+  int factor_q20 = (factor << 6) + 32;
+  // TODO(hlundin): Add 32 to factor_q20 when converting back to Q14?
+  for (size_t i = 0; i < length; ++i) {
+    output[i] = (factor * input[i] + 8192) >> 14;
+    factor_q20 += increment;
+    factor_q20 = std::max(factor_q20, 0);  // Never go negative.
+    factor = std::min(factor_q20 >> 6, 16384);
+  }
+  return factor;
+}
+
+int DspHelper::RampSignal(int16_t* signal,
+                          size_t length,
+                          int factor,
+                          int increment) {
+  return RampSignal(signal, length, factor, increment, signal);
+}
+
+int DspHelper::RampSignal(AudioVector* signal,
+                          size_t start_index,
+                          size_t length,
+                          int factor,
+                          int increment) {
+  int factor_q20 = (factor << 6) + 32;
+  // TODO(hlundin): Add 32 to factor_q20 when converting back to Q14?
+  for (size_t i = start_index; i < start_index + length; ++i) {
+    (*signal)[i] = (factor * (*signal)[i] + 8192) >> 14;
+    factor_q20 += increment;
+    factor_q20 = std::max(factor_q20, 0);  // Never go negative.
+    factor = std::min(factor_q20 >> 6, 16384);
+  }
+  return factor;
+}
+
+int DspHelper::RampSignal(AudioMultiVector* signal,
+                          size_t start_index,
+                          size_t length,
+                          int factor,
+                          int increment) {
+  RTC_DCHECK_LE(start_index + length, signal->Size());
+  if (start_index + length > signal->Size()) {
+    // Wrong parameters. Do nothing and return the scale factor unaltered.
+    return factor;
+  }
+  int end_factor = 0;
+  // Loop over the channels, starting at the same `factor` each time.
+  for (size_t channel = 0; channel < signal->Channels(); ++channel) {
+    end_factor =
+        RampSignal(&(*signal)[channel], start_index, length, factor, increment);
+  }
+  return end_factor;
+}
+
+void DspHelper::PeakDetection(int16_t* data,
+                              size_t data_length,
+                              size_t num_peaks,
+                              int fs_mult,
+                              size_t* peak_index,
+                              int16_t* peak_value) {
+  size_t min_index = 0;
+  size_t max_index = 0;
+
+  for (size_t i = 0; i <= num_peaks - 1; i++) {
+    if (num_peaks == 1) {
+      // Single peak.  The parabola fit assumes that an extra point is
+      // available; worst case it gets a zero on the high end of the signal.
+      // TODO(hlundin): This can potentially get much worse. It breaks the
+      // API contract, that the length of `data` is `data_length`.
+      data_length++;
+    }
+
+    peak_index[i] = WebRtcSpl_MaxIndexW16(data, data_length - 1);
+
+    if (i != num_peaks - 1) {
+      min_index = (peak_index[i] > 2) ? (peak_index[i] - 2) : 0;
+      max_index = std::min(data_length - 1, peak_index[i] + 2);
+    }
+
+    if ((peak_index[i] != 0) && (peak_index[i] != (data_length - 2))) {
+      ParabolicFit(&data[peak_index[i] - 1], fs_mult, &peak_index[i],
+                   &peak_value[i]);
+    } else {
+      if (peak_index[i] == data_length - 2) {
+        if (data[peak_index[i]] > data[peak_index[i] + 1]) {
+          ParabolicFit(&data[peak_index[i] - 1], fs_mult, &peak_index[i],
+                       &peak_value[i]);
+        } else if (data[peak_index[i]] <= data[peak_index[i] + 1]) {
+          // Linear approximation.
+          peak_value[i] = (data[peak_index[i]] + data[peak_index[i] + 1]) >> 1;
+          peak_index[i] = (peak_index[i] * 2 + 1) * fs_mult;
+        }
+      } else {
+        peak_value[i] = data[peak_index[i]];
+        peak_index[i] = peak_index[i] * 2 * fs_mult;
+      }
+    }
+
+    if (i != num_peaks - 1) {
+      memset(&data[min_index], 0,
+             sizeof(data[0]) * (max_index - min_index + 1));
+    }
+  }
+}
+
+void DspHelper::ParabolicFit(int16_t* signal_points,
+                             int fs_mult,
+                             size_t* peak_index,
+                             int16_t* peak_value) {
+  uint16_t fit_index[13];
+  if (fs_mult == 1) {
+    fit_index[0] = 0;
+    fit_index[1] = 8;
+    fit_index[2] = 16;
+  } else if (fs_mult == 2) {
+    fit_index[0] = 0;
+    fit_index[1] = 4;
+    fit_index[2] = 8;
+    fit_index[3] = 12;
+    fit_index[4] = 16;
+  } else if (fs_mult == 4) {
+    fit_index[0] = 0;
+    fit_index[1] = 2;
+    fit_index[2] = 4;
+    fit_index[3] = 6;
+    fit_index[4] = 8;
+    fit_index[5] = 10;
+    fit_index[6] = 12;
+    fit_index[7] = 14;
+    fit_index[8] = 16;
+  } else {
+    fit_index[0] = 0;
+    fit_index[1] = 1;
+    fit_index[2] = 3;
+    fit_index[3] = 4;
+    fit_index[4] = 5;
+    fit_index[5] = 7;
+    fit_index[6] = 8;
+    fit_index[7] = 9;
+    fit_index[8] = 11;
+    fit_index[9] = 12;
+    fit_index[10] = 13;
+    fit_index[11] = 15;
+    fit_index[12] = 16;
+  }
+
+  //  num = -3 * signal_points[0] + 4 * signal_points[1] - signal_points[2];
+  //  den =      signal_points[0] - 2 * signal_points[1] + signal_points[2];
+  int32_t num =
+      (signal_points[0] * -3) + (signal_points[1] * 4) - signal_points[2];
+  int32_t den = signal_points[0] + (signal_points[1] * -2) + signal_points[2];
+  int32_t temp = num * 120;
+  int flag = 1;
+  int16_t stp = kParabolaCoefficients[fit_index[fs_mult]][0] -
+                kParabolaCoefficients[fit_index[fs_mult - 1]][0];
+  int16_t strt = (kParabolaCoefficients[fit_index[fs_mult]][0] +
+                  kParabolaCoefficients[fit_index[fs_mult - 1]][0]) /
+                 2;
+  int16_t lmt;
+  if (temp < -den * strt) {
+    lmt = strt - stp;
+    while (flag) {
+      if ((flag == fs_mult) || (temp > -den * lmt)) {
+        *peak_value =
+            (den * kParabolaCoefficients[fit_index[fs_mult - flag]][1] +
+             num * kParabolaCoefficients[fit_index[fs_mult - flag]][2] +
+             signal_points[0] * 256) /
+            256;
+        *peak_index = *peak_index * 2 * fs_mult - flag;
+        flag = 0;
+      } else {
+        flag++;
+        lmt -= stp;
+      }
+    }
+  } else if (temp > -den * (strt + stp)) {
+    lmt = strt + 2 * stp;
+    while (flag) {
+      if ((flag == fs_mult) || (temp < -den * lmt)) {
+        int32_t temp_term_1 =
+            den * kParabolaCoefficients[fit_index[fs_mult + flag]][1];
+        int32_t temp_term_2 =
+            num * kParabolaCoefficients[fit_index[fs_mult + flag]][2];
+        int32_t temp_term_3 = signal_points[0] * 256;
+        *peak_value = (temp_term_1 + temp_term_2 + temp_term_3) / 256;
+        *peak_index = *peak_index * 2 * fs_mult + flag;
+        flag = 0;
+      } else {
+        flag++;
+        lmt += stp;
+      }
+    }
+  } else {
+    *peak_value = signal_points[1];
+    *peak_index = *peak_index * 2 * fs_mult;
+  }
+}
+
+size_t DspHelper::MinDistortion(const int16_t* signal,
+                                size_t min_lag,
+                                size_t max_lag,
+                                size_t length,
+                                int32_t* distortion_value) {
+  size_t best_index = 0;
+  int32_t min_distortion = WEBRTC_SPL_WORD32_MAX;
+  for (size_t i = min_lag; i <= max_lag; i++) {
+    int32_t sum_diff = 0;
+    const int16_t* data1 = signal;
+    const int16_t* data2 = signal - i;
+    for (size_t j = 0; j < length; j++) {
+      sum_diff += WEBRTC_SPL_ABS_W32(data1[j] - data2[j]);
+    }
+    // Compare with previous minimum.
+    if (sum_diff < min_distortion) {
+      min_distortion = sum_diff;
+      best_index = i;
+    }
+  }
+  *distortion_value = min_distortion;
+  return best_index;
+}
+
+void DspHelper::CrossFade(const int16_t* input1,
+                          const int16_t* input2,
+                          size_t length,
+                          int16_t* mix_factor,
+                          int16_t factor_decrement,
+                          int16_t* output) {
+  int16_t factor = *mix_factor;
+  int16_t complement_factor = 16384 - factor;
+  for (size_t i = 0; i < length; i++) {
+    output[i] =
+        (factor * input1[i] + complement_factor * input2[i] + 8192) >> 14;
+    factor -= factor_decrement;
+    complement_factor += factor_decrement;
+  }
+  *mix_factor = factor;
+}
+
+void DspHelper::UnmuteSignal(const int16_t* input,
+                             size_t length,
+                             int16_t* factor,
+                             int increment,
+                             int16_t* output) {
+  uint16_t factor_16b = *factor;
+  int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
+  for (size_t i = 0; i < length; i++) {
+    output[i] = (factor_16b * input[i] + 8192) >> 14;
+    factor_32b = std::max(factor_32b + increment, 0);
+    factor_16b = std::min(16384, factor_32b >> 6);
+  }
+  *factor = factor_16b;
+}
+
+void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) {
+  int32_t factor = (16384 << 6) + 32;
+  for (size_t i = 0; i < length; i++) {
+    signal[i] = ((factor >> 6) * signal[i] + 8192) >> 14;
+    factor -= mute_slope;
+  }
+}
+
+int DspHelper::DownsampleTo4kHz(const int16_t* input,
+                                size_t input_length,
+                                size_t output_length,
+                                int input_rate_hz,
+                                bool compensate_delay,
+                                int16_t* output) {
+  // Set filter parameters depending on input frequency.
+  // NOTE: The phase delay values are wrong compared to the true phase delay
+  // of the filters. However, the error is preserved (through the +1 term) for
+  // consistency.
+  const int16_t* filter_coefficients;  // Filter coefficients.
+  size_t filter_length;                // Number of coefficients.
+  size_t filter_delay;                 // Phase delay in samples.
+  int16_t factor;                      // Conversion rate (inFsHz / 8000).
+  switch (input_rate_hz) {
+    case 8000: {
+      filter_length = 3;
+      factor = 2;
+      filter_coefficients = kDownsample8kHzTbl;
+      filter_delay = 1 + 1;
+      break;
+    }
+    case 16000: {
+      filter_length = 5;
+      factor = 4;
+      filter_coefficients = kDownsample16kHzTbl;
+      filter_delay = 2 + 1;
+      break;
+    }
+    case 32000: {
+      filter_length = 7;
+      factor = 8;
+      filter_coefficients = kDownsample32kHzTbl;
+      filter_delay = 3 + 1;
+      break;
+    }
+    case 48000: {
+      filter_length = 7;
+      factor = 12;
+      filter_coefficients = kDownsample48kHzTbl;
+      filter_delay = 3 + 1;
+      break;
+    }
+    default: {
+      RTC_DCHECK_NOTREACHED();
+      return -1;
+    }
+  }
+
+  if (!compensate_delay) {
+    // Disregard delay compensation.
+    filter_delay = 0;
+  }
+
+  // Returns -1 if input signal is too short; 0 otherwise.
+  return WebRtcSpl_DownsampleFast(
+      &input[filter_length - 1], input_length - filter_length + 1, output,
+      output_length, filter_coefficients, filter_length, factor, filter_delay);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.h b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.h
new file mode 100644
index 0000000000..4aead7df18
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.h
@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
+
+#include <stdint.h>
+#include <string.h>
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+
+namespace webrtc {
+
+// This class contains various signal processing functions, all implemented as
+// static methods.
+class DspHelper {
+ public:
+  // Filter coefficients used when downsampling from the indicated sample rates
+  // (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12.
+  static const int16_t kDownsample8kHzTbl[3];
+  static const int16_t kDownsample16kHzTbl[5];
+  static const int16_t kDownsample32kHzTbl[7];
+  static const int16_t kDownsample48kHzTbl[7];
+
+  // Constants used to mute and unmute over 5 samples. The coefficients are
+  // in Q15.
+  static const int kMuteFactorStart8kHz = 27307;
+  static const int kMuteFactorIncrement8kHz = -5461;
+  static const int kUnmuteFactorStart8kHz = 5461;
+  static const int kUnmuteFactorIncrement8kHz = 5461;
+  static const int kMuteFactorStart16kHz = 29789;
+  static const int kMuteFactorIncrement16kHz = -2979;
+  static const int kUnmuteFactorStart16kHz = 2979;
+  static const int kUnmuteFactorIncrement16kHz = 2979;
+  static const int kMuteFactorStart32kHz = 31208;
+  static const int kMuteFactorIncrement32kHz = -1560;
+  static const int kUnmuteFactorStart32kHz = 1560;
+  static const int kUnmuteFactorIncrement32kHz = 1560;
+  static const int kMuteFactorStart48kHz = 31711;
+  static const int kMuteFactorIncrement48kHz = -1057;
+  static const int kUnmuteFactorStart48kHz = 1057;
+  static const int kUnmuteFactorIncrement48kHz = 1057;
+
+  // Multiplies the signal with a gradually changing factor.
+  // The first sample is multiplied with `factor` (in Q14). For each sample,
+  // `factor` is increased (additive) by the `increment` (in Q20), which can
+  // be negative. Returns the scale factor after the last increment.
+  static int RampSignal(const int16_t* input,
+                        size_t length,
+                        int factor,
+                        int increment,
+                        int16_t* output);
+
+  // Same as above, but with the samples of `signal` being modified in-place.
+  static int RampSignal(int16_t* signal,
+                        size_t length,
+                        int factor,
+                        int increment);
+
+  // Same as above, but processes `length` samples from `signal`, starting at
+  // `start_index`.
+  static int RampSignal(AudioVector* signal,
+                        size_t start_index,
+                        size_t length,
+                        int factor,
+                        int increment);
+
+  // Same as above, but for an AudioMultiVector.
+  static int RampSignal(AudioMultiVector* signal,
+                        size_t start_index,
+                        size_t length,
+                        int factor,
+                        int increment);
+
+  // Peak detection with parabolic fit. Looks for `num_peaks` maxima in `data`,
+  // having length `data_length` and sample rate multiplier `fs_mult`. The peak
+  // locations and values are written to the arrays `peak_index` and
+  // `peak_value`, respectively. Both arrays must hold at least `num_peaks`
+  // elements.
+  static void PeakDetection(int16_t* data,
+                            size_t data_length,
+                            size_t num_peaks,
+                            int fs_mult,
+                            size_t* peak_index,
+                            int16_t* peak_value);
+
+  // Estimates the height and location of a maximum. The three values in the
+  // array `signal_points` are used as basis for a parabolic fit, which is then
+  // used to find the maximum in an interpolated signal. The `signal_points` are
+  // assumed to be from a 4 kHz signal, while the maximum, written to
+  // `peak_index` and `peak_value` is given in the full sample rate, as
+  // indicated by the sample rate multiplier `fs_mult`.
+  static void ParabolicFit(int16_t* signal_points,
+                           int fs_mult,
+                           size_t* peak_index,
+                           int16_t* peak_value);
+
+  // Calculates the sum-abs-diff for `signal` when compared to a displaced
+  // version of itself. Returns the displacement lag that results in the minimum
+  // distortion. The resulting distortion is written to `distortion_value`.
+  // The values of `min_lag` and `max_lag` are boundaries for the search.
+  static size_t MinDistortion(const int16_t* signal,
+                              size_t min_lag,
+                              size_t max_lag,
+                              size_t length,
+                              int32_t* distortion_value);
+
+  // Mixes `length` samples from `input1` and `input2` together and writes the
+  // result to `output`. The gain for `input1` starts at `mix_factor` (Q14) and
+  // is decreased by `factor_decrement` (Q14) for each sample. The gain for
+  // `input2` is the complement 16384 - mix_factor.
+  static void CrossFade(const int16_t* input1,
+                        const int16_t* input2,
+                        size_t length,
+                        int16_t* mix_factor,
+                        int16_t factor_decrement,
+                        int16_t* output);
+
+  // Scales `input` with an increasing gain. Applies `factor` (Q14) to the first
+  // sample and increases the gain by `increment` (Q20) for each sample. The
+  // result is written to `output`. `length` samples are processed.
+  static void UnmuteSignal(const int16_t* input,
+                           size_t length,
+                           int16_t* factor,
+                           int increment,
+                           int16_t* output);
+
+  // Starts at unity gain and gradually fades out `signal`. For each sample,
+  // the gain is reduced by `mute_slope` (Q14). `length` samples are processed.
+  static void MuteSignal(int16_t* signal, int mute_slope, size_t length);
+
+  // Downsamples `input` from `sample_rate_hz` to 4 kHz sample rate. The input
+  // has `input_length` samples, and the method will write `output_length`
+  // samples to `output`. Compensates for the phase delay of the downsampling
+  // filters if `compensate_delay` is true. Returns -1 if the input is too short
+  // to produce `output_length` samples, otherwise 0.
+  static int DownsampleTo4kHz(const int16_t* input,
+                              size_t input_length,
+                              size_t output_length,
+                              int input_rate_hz,
+                              bool compensate_delay,
+                              int16_t* output);
+
+  DspHelper(const DspHelper&) = delete;
+  DspHelper& operator=(const DspHelper&) = delete;
+
+ private:
+  // Table of constants used in method DspHelper::ParabolicFit().
+  static const int16_t kParabolaCoefficients[17][3];
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc
new file mode 100644
index 0000000000..09247417d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/dsp_helper.h"
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(DspHelper, RampSignalArray) {
+  static const int kLen = 100;
+  int16_t input[kLen];
+  int16_t output[kLen];
+  // Fill input with 1000.
+  for (int i = 0; i < kLen; ++i) {
+    input[i] = 1000;
+  }
+  int start_factor = 0;
+  // Ramp from 0 to 1 (in Q14) over the array. Note that `increment` is in Q20,
+  // while the factor is in Q14, hence the shift by 6.
+  int increment = (16384 << 6) / kLen;
+
+  // Test first method.
+  int stop_factor =
+      DspHelper::RampSignal(input, kLen, start_factor, increment, output);
+  EXPECT_EQ(16383, stop_factor);  // Almost reach 1 in Q14.
+  for (int i = 0; i < kLen; ++i) {
+    EXPECT_EQ(1000 * i / kLen, output[i]);
+  }
+
+  // Test second method. (Note that this modifies `input`.)
+  stop_factor = DspHelper::RampSignal(input, kLen, start_factor, increment);
+  EXPECT_EQ(16383, stop_factor);  // Almost reach 1 in Q14.
+  for (int i = 0; i < kLen; ++i) {
+    EXPECT_EQ(1000 * i / kLen, input[i]);
+  }
+}
+
+TEST(DspHelper, RampSignalAudioMultiVector) {
+  static const int kLen = 100;
+  static const int kChannels = 5;
+  AudioMultiVector input(kChannels, kLen * 3);
+  // Fill input with 1000.
+  for (int i = 0; i < kLen * 3; ++i) {
+    for (int channel = 0; channel < kChannels; ++channel) {
+      input[channel][i] = 1000;
+    }
+  }
+  // We want to start ramping at `start_index` and keep ramping for `kLen`
+  // samples.
+  int start_index = kLen;
+  int start_factor = 0;
+  // Ramp from 0 to 1 (in Q14) in `kLen` samples. Note that `increment` is in
+  // Q20, while the factor is in Q14, hence the shift by 6.
+  int increment = (16384 << 6) / kLen;
+
+  int stop_factor =
+      DspHelper::RampSignal(&input, start_index, kLen, start_factor, increment);
+  EXPECT_EQ(16383, stop_factor);  // Almost reach 1 in Q14.
+  // Verify that the first `kLen` samples are left untouched.
+  int i;
+  for (i = 0; i < kLen; ++i) {
+    for (int channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(1000, input[channel][i]);
+    }
+  }
+  // Verify that the next block of `kLen` samples are ramped.
+  for (; i < 2 * kLen; ++i) {
+    for (int channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(1000 * (i - kLen) / kLen, input[channel][i]);
+    }
+  }
+  // Verify the last `kLen` samples are left untouched.
+  for (; i < 3 * kLen; ++i) {
+    for (int channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(1000, input[channel][i]);
+    }
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.cc
new file mode 100644
index 0000000000..115bfcf97b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.cc
@@ -0,0 +1,246 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
+
+#include <algorithm>  // max
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+// Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
+// longer required, this #define should be removed (and the code that it
+// enables).
+#define LEGACY_BITEXACT
+
+namespace webrtc {
+
+DtmfBuffer::DtmfBuffer(int fs_hz) {
+  SetSampleRate(fs_hz);
+}
+
+DtmfBuffer::~DtmfBuffer() = default;
+
+void DtmfBuffer::Flush() {
+  buffer_.clear();
+}
+
+// The ParseEvent method parses 4 bytes from `payload` according to this format
+// from RFC 4733:
+//
+//  0                   1                   2                   3
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |     event     |E|R| volume    |          duration             |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Legend (adapted from RFC 4733)
+// - event:    The event field is a number between 0 and 255 identifying a
+//             specific telephony event. The buffer will not accept any event
+//             numbers larger than 15.
+// - E:        If set to a value of one, the "end" bit indicates that this
+//             packet contains the end of the event.  For long-lasting events
+//             that have to be split into segments, only the final packet for
+//             the final segment will have the E bit set.
+// - R:        Reserved.
+// - volume:   For DTMF digits and other events representable as tones, this
+//             field describes the power level of the tone, expressed in dBm0
+//             after dropping the sign.  Power levels range from 0 to -63 dBm0.
+//             Thus, larger values denote lower volume. The buffer discards
+//             values larger than 36 (i.e., lower than -36 dBm0).
+// - duration: The duration field indicates the duration of the event or segment
+//             being reported, in timestamp units, expressed as an unsigned
+//             integer in network byte order.  For a non-zero value, the event
+//             or segment began at the instant identified by the RTP timestamp
+//             and has so far lasted as long as indicated by this parameter.
+//             The event may or may not have ended.  If the event duration
+//             exceeds the maximum representable by the duration field, the
+//             event is split into several contiguous segments. The buffer will
+//             discard zero-duration events.
+//
+int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp,
+                           const uint8_t* payload,
+                           size_t payload_length_bytes,
+                           DtmfEvent* event) {
+  RTC_CHECK(payload);
+  RTC_CHECK(event);
+  if (payload_length_bytes < 4) {
+    RTC_LOG(LS_WARNING) << "ParseEvent payload too short";
+    return kPayloadTooShort;
+  }
+
+  event->event_no = payload[0];
+  event->end_bit = ((payload[1] & 0x80) != 0);
+  event->volume = (payload[1] & 0x3F);
+  event->duration = payload[2] << 8 | payload[3];
+  event->timestamp = rtp_timestamp;
+  return kOK;
+}
+
+// Inserts a DTMF event into the buffer. The event should be parsed from the
+// bit stream using the ParseEvent method above before inserting it in the
+// buffer.
+// DTMF events can be quite long, and in most cases the duration of the event
+// is not known when the first packet describing it is sent. To deal with that,
+// the RFC 4733 specifies that multiple packets are sent for one and the same
+// event as it is being created (typically, as the user is pressing the key).
+// These packets will all share the same start timestamp and event number,
+// while the duration will be the cumulative duration from the start. When
+// inserting a new event, the InsertEvent method tries to find a matching event
+// already in the buffer. If so, the new event is simply merged with the
+// existing one.
+int DtmfBuffer::InsertEvent(const DtmfEvent& event) {
+  if (event.event_no < 0 || event.event_no > 15 || event.volume < 0 ||
+      event.volume > 63 || event.duration <= 0 || event.duration > 65535) {
+    RTC_LOG(LS_WARNING) << "InsertEvent invalid parameters";
+    return kInvalidEventParameters;
+  }
+  DtmfList::iterator it = buffer_.begin();
+  while (it != buffer_.end()) {
+    if (MergeEvents(it, event)) {
+      // A matching event was found and the new event was merged.
+      return kOK;
+    }
+    ++it;
+  }
+  buffer_.push_back(event);
+  // Sort the buffer using CompareEvents to rank the events.
+  buffer_.sort(CompareEvents);
+  return kOK;
+}
+
+bool DtmfBuffer::GetEvent(uint32_t current_timestamp, DtmfEvent* event) {
+  DtmfList::iterator it = buffer_.begin();
+  while (it != buffer_.end()) {
+    // `event_end` is an estimate of where the current event ends. If the end
+    // bit is set, we know that the event ends at `timestamp` + `duration`.
+    uint32_t event_end = it->timestamp + it->duration;
+#ifdef LEGACY_BITEXACT
+    bool next_available = false;
+#endif
+    if (!it->end_bit) {
+      // If the end bit is not set, we allow extrapolation of the event for
+      // some time.
+      event_end += max_extrapolation_samples_;
+      DtmfList::iterator next = it;
+      ++next;
+      if (next != buffer_.end()) {
+        // If there is a next event in the buffer, we will not extrapolate over
+        // the start of that new event.
+        event_end = std::min(event_end, next->timestamp);
+#ifdef LEGACY_BITEXACT
+        next_available = true;
+#endif
+      }
+    }
+    if (current_timestamp >= it->timestamp &&
+        current_timestamp <= event_end) {  // TODO(hlundin): Change to <.
+      // Found a matching event.
+      if (event) {
+        event->event_no = it->event_no;
+        event->end_bit = it->end_bit;
+        event->volume = it->volume;
+        event->duration = it->duration;
+        event->timestamp = it->timestamp;
+      }
+#ifdef LEGACY_BITEXACT
+      if (it->end_bit && current_timestamp + frame_len_samples_ >= event_end) {
+        // We are done playing this. Erase the event.
+        buffer_.erase(it);
+      }
+#endif
+      return true;
+    } else if (current_timestamp > event_end) {  // TODO(hlundin): Change to >=.
+// Erase old event. Operation returns a valid pointer to the next element
+// in the list.
+#ifdef LEGACY_BITEXACT
+      if (!next_available) {
+        if (event) {
+          event->event_no = it->event_no;
+          event->end_bit = it->end_bit;
+          event->volume = it->volume;
+          event->duration = it->duration;
+          event->timestamp = it->timestamp;
+        }
+        it = buffer_.erase(it);
+        return true;
+      } else {
+        it = buffer_.erase(it);
+      }
+#else
+      it = buffer_.erase(it);
+#endif
+    } else {
+      ++it;
+    }
+  }
+  return false;
+}
+
+size_t DtmfBuffer::Length() const {
+  return buffer_.size();
+}
+
+bool DtmfBuffer::Empty() const {
+  return buffer_.empty();
+}
+
+int DtmfBuffer::SetSampleRate(int fs_hz) {
+  if (fs_hz != 8000 &&
+      fs_hz != 16000 &&
+      fs_hz != 32000 &&
+      fs_hz != 44100 &&
+      fs_hz != 48000) {
+    return kInvalidSampleRate;
+  }
+  max_extrapolation_samples_ = 7 * fs_hz / 100;
+  frame_len_samples_ = fs_hz / 100;
+  return kOK;
+}
+
+// The method returns true if the two events are considered to be the same.
+// The are defined as equal if they share the same timestamp and event number.
+// The special case with long-lasting events that have to be split into segments
+// is not handled in this method. These will be treated as separate events in
+// the buffer.
+bool DtmfBuffer::SameEvent(const DtmfEvent& a, const DtmfEvent& b) {
+  return (a.event_no == b.event_no) && (a.timestamp == b.timestamp);
+}
+
+bool DtmfBuffer::MergeEvents(DtmfList::iterator it, const DtmfEvent& event) {
+  if (SameEvent(*it, event)) {
+    if (!it->end_bit) {
+      // Do not extend the duration of an event for which the end bit was
+      // already received.
+      it->duration = std::max(event.duration, it->duration);
+    }
+    if (event.end_bit) {
+      it->end_bit = true;
+    }
+    return true;
+  } else {
+    return false;
+  }
+}
+
+// Returns true if `a` goes before `b` in the sorting order ("`a` < `b`").
+// The events are ranked using their start timestamp (taking wrap-around into
+// account). In the unlikely situation that two events share the same start
+// timestamp, the event number is used to rank the two. Note that packets
+// that belong to the same events, and therefore sharing the same start
+// timestamp, have already been merged before the sort method is called.
+bool DtmfBuffer::CompareEvents(const DtmfEvent& a, const DtmfEvent& b) {
+  if (a.timestamp == b.timestamp) {
+    return a.event_no < b.event_no;
+  }
+  // Take wrap-around into account.
+  return (static_cast<uint32_t>(b.timestamp - a.timestamp) < 0xFFFFFFFF / 2);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.h
new file mode 100644
index 0000000000..62b751525c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.h
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+
+namespace webrtc {
+
+struct DtmfEvent {
+  uint32_t timestamp;
+  int event_no;
+  int volume;
+  int duration;
+  bool end_bit;
+
+  // Constructors
+  DtmfEvent()
+      : timestamp(0), event_no(0), volume(0), duration(0), end_bit(false) {}
+  DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end)
+      : timestamp(ts), event_no(ev), volume(vol), duration(dur), end_bit(end) {}
+};
+
+// This is the buffer holding DTMF events while waiting for them to be played.
+class DtmfBuffer {
+ public:
+  enum BufferReturnCodes {
+    kOK = 0,
+    kInvalidPointer,
+    kPayloadTooShort,
+    kInvalidEventParameters,
+    kInvalidSampleRate
+  };
+
+  // Set up the buffer for use at sample rate `fs_hz`.
+  explicit DtmfBuffer(int fs_hz);
+
+  virtual ~DtmfBuffer();
+
+  DtmfBuffer(const DtmfBuffer&) = delete;
+  DtmfBuffer& operator=(const DtmfBuffer&) = delete;
+
+  // Flushes the buffer.
+  virtual void Flush();
+
+  // Static method to parse 4 bytes from `payload` as a DTMF event (RFC 4733)
+  // and write the parsed information into the struct `event`. Input variable
+  // `rtp_timestamp` is simply copied into the struct.
+  static int ParseEvent(uint32_t rtp_timestamp,
+                        const uint8_t* payload,
+                        size_t payload_length_bytes,
+                        DtmfEvent* event);
+
+  // Inserts `event` into the buffer. The method looks for a matching event and
+  // merges the two if a match is found.
+  virtual int InsertEvent(const DtmfEvent& event);
+
+  // Checks if a DTMF event should be played at time `current_timestamp`. If so,
+  // the method returns true; otherwise false. The parameters of the event to
+  // play will be written to `event`.
+  virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);
+
+  // Number of events in the buffer.
+  virtual size_t Length() const;
+
+  virtual bool Empty() const;
+
+  // Set a new sample rate.
+  virtual int SetSampleRate(int fs_hz);
+
+ private:
+  typedef std::list<DtmfEvent> DtmfList;
+
+  int max_extrapolation_samples_;
+  int frame_len_samples_;  // TODO(hlundin): Remove this later.
+
+  // Compares two events and returns true if they are the same.
+  static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b);
+
+  // Merges `event` to the event pointed out by `it`. The method checks that
+  // the two events are the same (using the SameEvent method), and merges them
+  // if that was the case, returning true. If the events are not the same, false
+  // is returned.
+  bool MergeEvents(DtmfList::iterator it, const DtmfEvent& event);
+
+  // Method used by the sort algorithm to rank events in the buffer.
+  static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b);
+
+  DtmfList buffer_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
new file mode 100644
index 0000000000..83745b6c09
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
@@ -0,0 +1,297 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
+
+#ifdef WIN32
+#include <winsock2.h>  // ntohl()
+#else
+#include <arpa/inet.h>  // ntohl()
+#endif
+
+#include <iostream>
+
+#include "test/gtest.h"
+
+// Modify the tests so that they pass with the modifications done to DtmfBuffer
+// for backwards bit-exactness. Once bit-exactness is no longer required, this
+// #define should be removed (and the code that it enables).
+#define LEGACY_BITEXACT
+
+namespace webrtc {
+
+static int sample_rate_hz = 8000;
+
+static uint32_t MakeDtmfPayload(int event, bool end, int volume, int duration) {
+  uint32_t payload = 0;
+  //  0                   1                   2                   3
+  //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |     event     |E|R| volume    |          duration             |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  payload |= (event & 0x00FF) << 24;
+  payload |= (end ? 0x00800000 : 0x00000000);
+  payload |= (volume & 0x003F) << 16;
+  payload |= (duration & 0xFFFF);
+  payload = ntohl(payload);
+  return payload;
+}
+
+static bool EqualEvents(const DtmfEvent& a, const DtmfEvent& b) {
+  return (a.duration == b.duration && a.end_bit == b.end_bit &&
+          a.event_no == b.event_no && a.timestamp == b.timestamp &&
+          a.volume == b.volume);
+}
+
+TEST(DtmfBuffer, CreateAndDestroy) {
+  DtmfBuffer* buffer = new DtmfBuffer(sample_rate_hz);
+  delete buffer;
+}
+
+// Test the event parser.
+TEST(DtmfBuffer, ParseEvent) {
+  int event_no = 7;
+  bool end_bit = true;
+  int volume = 17;
+  int duration = 4711;
+  uint32_t timestamp = 0x12345678;
+  uint32_t payload = MakeDtmfPayload(event_no, end_bit, volume, duration);
+  uint8_t* payload_ptr = reinterpret_cast<uint8_t*>(&payload);
+  DtmfEvent event;
+  EXPECT_EQ(DtmfBuffer::kOK, DtmfBuffer::ParseEvent(timestamp, payload_ptr,
+                                                    sizeof(payload), &event));
+  EXPECT_EQ(duration, event.duration);
+  EXPECT_EQ(end_bit, event.end_bit);
+  EXPECT_EQ(event_no, event.event_no);
+  EXPECT_EQ(timestamp, event.timestamp);
+  EXPECT_EQ(volume, event.volume);
+
+  EXPECT_EQ(DtmfBuffer::kPayloadTooShort,
+            DtmfBuffer::ParseEvent(timestamp, payload_ptr, 3, &event));
+}
+
+TEST(DtmfBuffer, SimpleInsertAndGet) {
+  int event_no = 7;
+  bool end_bit = true;
+  int volume = 17;
+  int duration = 4711;
+  uint32_t timestamp = 0x12345678;
+  DtmfEvent event(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+  EXPECT_EQ(1u, buffer.Length());
+  EXPECT_FALSE(buffer.Empty());
+  DtmfEvent out_event;
+  // Too early to get event.
+  EXPECT_FALSE(buffer.GetEvent(timestamp - 10, &out_event));
+  EXPECT_EQ(1u, buffer.Length());
+  EXPECT_FALSE(buffer.Empty());
+  // Get the event at its starting timestamp.
+  EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event));
+  EXPECT_TRUE(EqualEvents(event, out_event));
+  EXPECT_EQ(1u, buffer.Length());
+  EXPECT_FALSE(buffer.Empty());
+  // Get the event some time into the event.
+  EXPECT_TRUE(buffer.GetEvent(timestamp + duration / 2, &out_event));
+  EXPECT_TRUE(EqualEvents(event, out_event));
+  EXPECT_EQ(1u, buffer.Length());
+  EXPECT_FALSE(buffer.Empty());
+// Give a "current" timestamp after the event has ended.
+#ifdef LEGACY_BITEXACT
+  EXPECT_TRUE(buffer.GetEvent(timestamp + duration + 10, &out_event));
+#endif
+  EXPECT_FALSE(buffer.GetEvent(timestamp + duration + 10, &out_event));
+  EXPECT_EQ(0u, buffer.Length());
+  EXPECT_TRUE(buffer.Empty());
+}
+
+TEST(DtmfBuffer, MergingPackets) {
+  int event_no = 0;
+  bool end_bit = false;
+  int volume = 17;
+  int duration = 80;
+  uint32_t timestamp = 0x12345678;
+  DtmfEvent event(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+
+  event.duration += 80;
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+
+  event.duration += 80;
+  event.end_bit = true;
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+
+  EXPECT_EQ(1u, buffer.Length());
+
+  DtmfEvent out_event;
+  EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event));
+  EXPECT_TRUE(EqualEvents(event, out_event));
+}
+
+// This test case inserts one shorter event completely overlapped by one longer
+// event. The expected outcome is that only the longer event is played.
+TEST(DtmfBuffer, OverlappingEvents) {
+  int event_no = 0;
+  bool end_bit = true;
+  int volume = 1;
+  int duration = 80;
+  uint32_t timestamp = 0x12345678 + 80;
+  DtmfEvent short_event(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(short_event));
+
+  event_no = 10;
+  end_bit = false;
+  timestamp = 0x12345678;
+  DtmfEvent long_event(timestamp, event_no, volume, duration, end_bit);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(long_event));
+
+  long_event.duration += 80;
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(long_event));
+
+  long_event.duration += 80;
+  long_event.end_bit = true;
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(long_event));
+
+  EXPECT_EQ(2u, buffer.Length());
+
+  DtmfEvent out_event;
+  // Expect to get the long event.
+  EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event));
+  EXPECT_TRUE(EqualEvents(long_event, out_event));
+// Expect no more events.
+#ifdef LEGACY_BITEXACT
+  EXPECT_TRUE(
+      buffer.GetEvent(timestamp + long_event.duration + 10, &out_event));
+  EXPECT_TRUE(EqualEvents(long_event, out_event));
+  EXPECT_TRUE(
+      buffer.GetEvent(timestamp + long_event.duration + 10, &out_event));
+  EXPECT_TRUE(EqualEvents(short_event, out_event));
+#else
+  EXPECT_FALSE(
+      buffer.GetEvent(timestamp + long_event.duration + 10, &out_event));
+#endif
+  EXPECT_TRUE(buffer.Empty());
+}
+
+TEST(DtmfBuffer, ExtrapolationTime) {
+  int event_no = 0;
+  bool end_bit = false;
+  int volume = 1;
+  int duration = 80;
+  uint32_t timestamp = 0x12345678;
+  DtmfEvent event1(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event1));
+  EXPECT_EQ(1u, buffer.Length());
+
+  DtmfEvent out_event;
+  // Get the event at the start.
+  EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event));
+  EXPECT_TRUE(EqualEvents(event1, out_event));
+  // Also get the event 100 samples after the end of the event (since we're
+  // missing the end bit).
+  uint32_t timestamp_now = timestamp + duration + 100;
+  EXPECT_TRUE(buffer.GetEvent(timestamp_now, &out_event));
+  EXPECT_TRUE(EqualEvents(event1, out_event));
+  // Insert another event starting back-to-back with the previous event.
+  timestamp += duration;
+  event_no = 1;
+  DtmfEvent event2(timestamp, event_no, volume, duration, end_bit);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event2));
+  EXPECT_EQ(2u, buffer.Length());
+  // Now we expect to get the new event when supplying `timestamp_now`.
+  EXPECT_TRUE(buffer.GetEvent(timestamp_now, &out_event));
+  EXPECT_TRUE(EqualEvents(event2, out_event));
+  // Expect the the first event to be erased now.
+  EXPECT_EQ(1u, buffer.Length());
+  // Move `timestamp_now` to more than 560 samples after the end of the second
+  // event. Expect that event to be erased.
+  timestamp_now = timestamp + duration + 600;
+#ifdef LEGACY_BITEXACT
+  EXPECT_TRUE(buffer.GetEvent(timestamp_now, &out_event));
+#endif
+  EXPECT_FALSE(buffer.GetEvent(timestamp_now, &out_event));
+  EXPECT_TRUE(buffer.Empty());
+}
+
+TEST(DtmfBuffer, TimestampWraparound) {
+  int event_no = 0;
+  bool end_bit = true;
+  int volume = 1;
+  int duration = 80;
+  uint32_t timestamp1 = 0xFFFFFFFF - duration;
+  DtmfEvent event1(timestamp1, event_no, volume, duration, end_bit);
+  uint32_t timestamp2 = 0;
+  DtmfEvent event2(timestamp2, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event1));
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event2));
+  EXPECT_EQ(2u, buffer.Length());
+  DtmfEvent out_event;
+  EXPECT_TRUE(buffer.GetEvent(timestamp1, &out_event));
+  EXPECT_TRUE(EqualEvents(event1, out_event));
+#ifdef LEGACY_BITEXACT
+  EXPECT_EQ(1u, buffer.Length());
+#else
+  EXPECT_EQ(2u, buffer.Length());
+#endif
+
+  buffer.Flush();
+  // Reverse the insert order. Expect same results.
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event2));
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event1));
+  EXPECT_EQ(2u, buffer.Length());
+  EXPECT_TRUE(buffer.GetEvent(timestamp1, &out_event));
+  EXPECT_TRUE(EqualEvents(event1, out_event));
+#ifdef LEGACY_BITEXACT
+  EXPECT_EQ(1u, buffer.Length());
+#else
+  EXPECT_EQ(2u, buffer.Length());
+#endif
+}
+
+TEST(DtmfBuffer, InvalidEvents) {
+  int event_no = 0;
+  bool end_bit = true;
+  int volume = 1;
+  int duration = 80;
+  uint32_t timestamp = 0x12345678;
+  DtmfEvent event(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+
+  // Invalid event number.
+  event.event_no = -1;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.event_no = 16;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.event_no = 0;  // Valid value;
+
+  // Invalid volume.
+  event.volume = -1;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.volume = 64;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.volume = 0;  // Valid value;
+
+  // Invalid duration.
+  event.duration = -1;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.duration = 0;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.duration = 0xFFFF + 1;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.duration = 1;  // Valid value;
+
+  // Finish with a valid event, just to verify that all is ok.
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
new file mode 100644
index 0000000000..9061e27c67
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
@@ -0,0 +1,215 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This class provides a generator for DTMF tones. The tone generation is based
+// on a sinusoid recursion. Each sinusoid is generated using a recursion
+// formula; x[n] = a * x[n-1] - x[n-2], where the coefficient
+// a = 2*cos(2*pi*f/fs). The recursion is started with x[-1] = 0 and
+// x[-2] = sin(2*pi*f/fs). (Note that with this initialization, the resulting
+// sinusoid gets a "negative" rotation; x[n] = sin(-2*pi*f/fs * n + phi), but
+// kept this way due to historical reasons.)
+// TODO(hlundin): Change to positive rotation?
+//
+// Each key on the telephone keypad corresponds to an "event", 0-15. Each event
+// is mapped to a tone pair, with a low and a high frequency. There are four
+// low and four high frequencies, each corresponding to a row and column,
+// respectively, on the keypad as illustrated below.
+//
+//          1209 Hz  1336 Hz  1477 Hz  1633 Hz
+// 697 Hz      1        2        3       12
+// 770 Hz      4        5        6       13
+// 852 Hz      7        8        9       14
+// 941 Hz     10        0       11       15
+
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
+
+#include "modules/audio_coding/neteq/audio_vector.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// The filter coefficient a = 2*cos(2*pi*f/fs) for the low frequency tone, for
+// sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0 through 15.
+// Values are in Q14.
+const int DtmfToneGenerator::kCoeff1[4][16] = {
+    {24219, 27980, 27980, 27980, 26956, 26956, 26956, 25701, 25701, 25701,
+     24219, 24219, 27980, 26956, 25701, 24219},
+    {30556, 31548, 31548, 31548, 31281, 31281, 31281, 30951, 30951, 30951,
+     30556, 30556, 31548, 31281, 30951, 30556},
+    {32210, 32462, 32462, 32462, 32394, 32394, 32394, 32311, 32311, 32311,
+     32210, 32210, 32462, 32394, 32311, 32210},
+    {32520, 32632, 32632, 32632, 32602, 32602, 32602, 32564, 32564, 32564,
+     32520, 32520, 32632, 32602, 32564, 32520}};
+
+// The filter coefficient a = 2*cos(2*pi*f/fs) for the high frequency tone, for
+// sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0 through 15.
+// Values are in Q14.
+const int DtmfToneGenerator::kCoeff2[4][16] = {
+    {16325, 19073, 16325, 13085, 19073, 16325, 13085, 19073, 16325, 13085,
+     19073, 13085, 9315, 9315, 9315, 9315},
+    {28361, 29144, 28361, 27409, 29144, 28361, 27409, 29144, 28361, 27409,
+     29144, 27409, 26258, 26258, 26258, 26258},
+    {31647, 31849, 31647, 31400, 31849, 31647, 31400, 31849, 31647, 31400,
+     31849, 31400, 31098, 31098, 31098, 31098},
+    {32268, 32359, 32268, 32157, 32359, 32268, 32157, 32359, 32268, 32157,
+     32359, 32157, 32022, 32022, 32022, 32022}};
+
+// The initialization value x[-2] = sin(2*pi*f/fs) for the low frequency tone,
+// for sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0-15.
+// Values are in Q14.
+const int DtmfToneGenerator::kInitValue1[4][16] = {
+    {11036, 8528, 8528, 8528, 9315, 9315, 9315, 10163, 10163, 10163, 11036,
+     11036, 8528, 9315, 10163, 11036},
+    {5918, 4429, 4429, 4429, 4879, 4879, 4879, 5380, 5380, 5380, 5918, 5918,
+     4429, 4879, 5380, 5918},
+    {3010, 2235, 2235, 2235, 2468, 2468, 2468, 2728, 2728, 2728, 3010, 3010,
+     2235, 2468, 2728, 3010},
+    {2013, 1493, 1493, 1493, 1649, 1649, 1649, 1823, 1823, 1823, 2013, 2013,
+     1493, 1649, 1823, 2013}};
+
+// The initialization value x[-2] = sin(2*pi*f/fs) for the high frequency tone,
+// for sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0-15.
+// Values are in Q14.
+const int DtmfToneGenerator::kInitValue2[4][16] = {
+    {14206, 13323, 14206, 15021, 13323, 14206, 15021, 13323, 14206, 15021,
+     13323, 15021, 15708, 15708, 15708, 15708},
+    {8207, 7490, 8207, 8979, 7490, 8207, 8979, 7490, 8207, 8979, 7490, 8979,
+     9801, 9801, 9801, 9801},
+    {4249, 3853, 4249, 4685, 3853, 4249, 4685, 3853, 4249, 4685, 3853, 4685,
+     5164, 5164, 5164, 5164},
+    {2851, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 3148,
+     3476, 3476, 3476, 3476}};
+
+// Amplitude multipliers for volume values 0 through 63, corresponding to
+// 0 dBm0 through -63 dBm0. Values are in Q14.
+// for a in range(0, 64):
+//   print round(16141.0 * 10**(-float(a)/20))
+const int DtmfToneGenerator::kAmplitude[64] = {
+    16141, 14386, 12821, 11427, 10184, 9077, 8090, 7210, 6426, 5727, 5104,
+    4549,  4054,  3614,  3221,  2870,  2558, 2280, 2032, 1811, 1614, 1439,
+    1282,  1143,  1018,  908,   809,   721,  643,  573,  510,  455,  405,
+    361,   322,   287,   256,   228,   203,  181,  161,  144,  128,  114,
+    102,   91,    81,    72,    64,    57,   51,   45,   41,   36,   32,
+    29,    26,    23,    20,    18,    16,   14,   13,   11};
+
+// Constructor.
+DtmfToneGenerator::DtmfToneGenerator()
+    : initialized_(false), coeff1_(0), coeff2_(0), amplitude_(0) {}
+
+// Initialize the DTMF generator with sample rate fs Hz (8000, 16000, 32000,
+// 48000), event (0-15) and attenuation (0-36 dB).
+// Returns 0 on success, otherwise an error code.
+int DtmfToneGenerator::Init(int fs, int event, int attenuation) {
+  initialized_ = false;
+  size_t fs_index;
+  if (fs == 8000) {
+    fs_index = 0;
+  } else if (fs == 16000) {
+    fs_index = 1;
+  } else if (fs == 32000) {
+    fs_index = 2;
+  } else if (fs == 48000) {
+    fs_index = 3;
+  } else {
+    RTC_DCHECK_NOTREACHED();
+    fs_index = 1;  // Default to 8000 Hz.
+  }
+
+  if (event < 0 || event > 15) {
+    return kParameterError;  // Invalid event number.
+  }
+
+  if (attenuation < 0 || attenuation > 63) {
+    return kParameterError;  // Invalid attenuation.
+  }
+
+  // Look up oscillator coefficient for low and high frequencies.
+  RTC_DCHECK_LE(0, fs_index);
+  RTC_DCHECK_GT(arraysize(kCoeff1), fs_index);
+  RTC_DCHECK_GT(arraysize(kCoeff2), fs_index);
+  RTC_DCHECK_LE(0, event);
+  RTC_DCHECK_GT(arraysize(kCoeff1[fs_index]), event);
+  RTC_DCHECK_GT(arraysize(kCoeff2[fs_index]), event);
+  coeff1_ = kCoeff1[fs_index][event];
+  coeff2_ = kCoeff2[fs_index][event];
+
+  // Look up amplitude multiplier.
+  RTC_DCHECK_LE(0, attenuation);
+  RTC_DCHECK_GT(arraysize(kAmplitude), attenuation);
+  amplitude_ = kAmplitude[attenuation];
+
+  // Initialize sample history.
+  RTC_DCHECK_LE(0, fs_index);
+  RTC_DCHECK_GT(arraysize(kInitValue1), fs_index);
+  RTC_DCHECK_GT(arraysize(kInitValue2), fs_index);
+  RTC_DCHECK_LE(0, event);
+  RTC_DCHECK_GT(arraysize(kInitValue1[fs_index]), event);
+  RTC_DCHECK_GT(arraysize(kInitValue2[fs_index]), event);
+  sample_history1_[0] = kInitValue1[fs_index][event];
+  sample_history1_[1] = 0;
+  sample_history2_[0] = kInitValue2[fs_index][event];
+  sample_history2_[1] = 0;
+
+  initialized_ = true;
+  return 0;
+}
+
+// Reset tone generator to uninitialized state.
+void DtmfToneGenerator::Reset() {
+  initialized_ = false;
+}
+
+// Generate num_samples of DTMF signal and write to `output`.
+int DtmfToneGenerator::Generate(size_t num_samples, AudioMultiVector* output) {
+  if (!initialized_) {
+    return kNotInitialized;
+  }
+
+  if (!output) {
+    return kParameterError;
+  }
+
+  output->AssertSize(num_samples);
+  for (size_t i = 0; i < num_samples; ++i) {
+    // Use recursion formula y[n] = a * y[n - 1] - y[n - 2].
+    int16_t temp_val_low =
+        ((coeff1_ * sample_history1_[1] + 8192) >> 14) - sample_history1_[0];
+    int16_t temp_val_high =
+        ((coeff2_ * sample_history2_[1] + 8192) >> 14) - sample_history2_[0];
+
+    // Update recursion memory.
+    sample_history1_[0] = sample_history1_[1];
+    sample_history1_[1] = temp_val_low;
+    sample_history2_[0] = sample_history2_[1];
+    sample_history2_[1] = temp_val_high;
+
+    // Attenuate the low frequency tone 3 dB.
+    int32_t temp_val =
+        kAmpMultiplier * temp_val_low + temp_val_high * (1 << 15);
+    // Normalize the signal to Q14 with proper rounding.
+    temp_val = (temp_val + 16384) >> 15;
+    // Scale the signal to correct volume.
+    (*output)[0][i] =
+        static_cast<int16_t>((temp_val * amplitude_ + 8192) >> 14);
+  }
+  // Copy first channel to all other channels.
+  for (size_t channel = 1; channel < output->Channels(); ++channel) {
+    output->CopyChannel(0, channel);
+  }
+
+  return static_cast<int>(num_samples);
+}
+
+bool DtmfToneGenerator::initialized() const {
+  return initialized_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.h b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
new file mode 100644
index 0000000000..35114f4f49
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+namespace webrtc {
+
+// This class provides a generator for DTMF tones.
+class DtmfToneGenerator {
+ public:
+  enum ReturnCodes {
+    kNotInitialized = -1,
+    kParameterError = -2,
+  };
+
+  DtmfToneGenerator();
+  virtual ~DtmfToneGenerator() {}
+
+  DtmfToneGenerator(const DtmfToneGenerator&) = delete;
+  DtmfToneGenerator& operator=(const DtmfToneGenerator&) = delete;
+
+  virtual int Init(int fs, int event, int attenuation);
+  virtual void Reset();
+  virtual int Generate(size_t num_samples, AudioMultiVector* output);
+  virtual bool initialized() const;
+
+ private:
+  static const int kCoeff1[4][16];  // 1st oscillator model coefficient table.
+  static const int kCoeff2[4][16];  // 2nd oscillator model coefficient table.
+  static const int kInitValue1[4][16];  // Initialization for 1st oscillator.
+  static const int kInitValue2[4][16];  // Initialization for 2nd oscillator.
+  static const int kAmplitude[64];      // Amplitude for 0 through -63 dBm0.
+  static const int16_t kAmpMultiplier = 23171;  // 3 dB attenuation (in Q15).
+
+  bool initialized_;            // True if generator is initialized properly.
+  int coeff1_;                  // 1st oscillator coefficient for this event.
+  int coeff2_;                  // 2nd oscillator coefficient for this event.
+  int amplitude_;               // Amplitude for this event.
+  int16_t sample_history1_[2];  // Last 2 samples for the 1st oscillator.
+  int16_t sample_history2_[2];  // Last 2 samples for the 2nd oscillator.
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
new file mode 100644
index 0000000000..e843706dd3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
@@ -0,0 +1,180 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for DtmfToneGenerator class.
+
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
+
+#include <math.h>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class DtmfToneGeneratorTest : public ::testing::Test {
+ protected:
+  static const double kLowFreqHz[16];
+  static const double kHighFreqHz[16];
+  // This is the attenuation applied to all cases.
+  const double kBaseAttenuation = 16141.0 / 16384.0;
+  const double k3dbAttenuation = 23171.0 / 32768;
+  const int kNumSamples = 10;
+
+  void TestAllTones(int fs_hz, int channels) {
+    AudioMultiVector signal(channels);
+
+    for (int event = 0; event <= 15; ++event) {
+      rtc::StringBuilder ss;
+      ss << "Checking event " << event << " at sample rate " << fs_hz;
+      SCOPED_TRACE(ss.str());
+      const int kAttenuation = 0;
+      ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, kAttenuation));
+      EXPECT_TRUE(tone_gen_.initialized());
+      EXPECT_EQ(kNumSamples, tone_gen_.Generate(kNumSamples, &signal));
+
+      double f1 = kLowFreqHz[event];
+      double f2 = kHighFreqHz[event];
+      const double pi = 3.14159265358979323846;
+
+      for (int n = 0; n < kNumSamples; ++n) {
+        double x = k3dbAttenuation * sin(2.0 * pi * f1 / fs_hz * (-n - 1)) +
+                   sin(2.0 * pi * f2 / fs_hz * (-n - 1));
+        x *= kBaseAttenuation;
+        x = ldexp(x, 14);  // Scale to Q14.
+        for (int channel = 0; channel < channels; ++channel) {
+          EXPECT_NEAR(x, static_cast<double>(signal[channel][n]), 25);
+        }
+      }
+
+      tone_gen_.Reset();
+      EXPECT_FALSE(tone_gen_.initialized());
+    }
+  }
+
+  void TestAmplitudes(int fs_hz, int channels) {
+    AudioMultiVector signal(channels);
+    AudioMultiVector ref_signal(channels);
+
+    const int event_vec[] = {0, 4, 9, 13};  // Test a few events.
+    for (int e = 0; e < 4; ++e) {
+      int event = event_vec[e];
+      // Create full-scale reference.
+      ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, 0));  // 0 attenuation.
+      EXPECT_EQ(kNumSamples, tone_gen_.Generate(kNumSamples, &ref_signal));
+      // Test every 5 steps (to save time).
+      for (int attenuation = 1; attenuation <= 63; attenuation += 5) {
+        rtc::StringBuilder ss;
+        ss << "Checking event " << event << " at sample rate " << fs_hz;
+        ss << "; attenuation " << attenuation;
+        SCOPED_TRACE(ss.str());
+        ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, attenuation));
+        EXPECT_EQ(kNumSamples, tone_gen_.Generate(kNumSamples, &signal));
+        for (int n = 0; n < kNumSamples; ++n) {
+          double attenuation_factor =
+              DbToRatio(-static_cast<float>(attenuation));
+          // Verify that the attenuation is correct.
+          for (int channel = 0; channel < channels; ++channel) {
+            EXPECT_NEAR(attenuation_factor * ref_signal[channel][n],
+                        signal[channel][n], 2);
+          }
+        }
+
+        tone_gen_.Reset();
+      }
+    }
+  }
+
+  DtmfToneGenerator tone_gen_;
+};
+
+// Low and high frequencies for events 0 through 15.
+const double DtmfToneGeneratorTest::kLowFreqHz[16] = {
+    941.0, 697.0, 697.0, 697.0, 770.0, 770.0, 770.0, 852.0,
+    852.0, 852.0, 941.0, 941.0, 697.0, 770.0, 852.0, 941.0};
+const double DtmfToneGeneratorTest::kHighFreqHz[16] = {
+    1336.0, 1209.0, 1336.0, 1477.0, 1209.0, 1336.0, 1477.0, 1209.0,
+    1336.0, 1477.0, 1209.0, 1477.0, 1633.0, 1633.0, 1633.0, 1633.0};
+
+TEST_F(DtmfToneGeneratorTest, Test8000Mono) {
+  TestAllTones(8000, 1);
+  TestAmplitudes(8000, 1);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test16000Mono) {
+  TestAllTones(16000, 1);
+  TestAmplitudes(16000, 1);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test32000Mono) {
+  TestAllTones(32000, 1);
+  TestAmplitudes(32000, 1);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test48000Mono) {
+  TestAllTones(48000, 1);
+  TestAmplitudes(48000, 1);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test8000Stereo) {
+  TestAllTones(8000, 2);
+  TestAmplitudes(8000, 2);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test16000Stereo) {
+  TestAllTones(16000, 2);
+  TestAmplitudes(16000, 2);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test32000Stereo) {
+  TestAllTones(32000, 2);
+  TestAmplitudes(32000, 2);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test48000Stereo) {
+  TestAllTones(48000, 2);
+  TestAmplitudes(48000, 2);
+}
+
+TEST(DtmfToneGenerator, TestErrors) {
+  DtmfToneGenerator tone_gen;
+  const int kNumSamples = 10;
+  AudioMultiVector signal(1);  // One channel.
+
+  // Try to generate tones without initializing.
+  EXPECT_EQ(DtmfToneGenerator::kNotInitialized,
+            tone_gen.Generate(kNumSamples, &signal));
+
+  const int fs = 16000;       // Valid sample rate.
+  const int event = 7;        // Valid event.
+  const int attenuation = 0;  // Valid attenuation.
+  // Initialize with invalid event -1.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError,
+            tone_gen.Init(fs, -1, attenuation));
+  // Initialize with invalid event 16.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError,
+            tone_gen.Init(fs, 16, attenuation));
+  // Initialize with invalid attenuation -1.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError, tone_gen.Init(fs, event, -1));
+  // Initialize with invalid attenuation 64.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError, tone_gen.Init(fs, event, 64));
+  EXPECT_FALSE(tone_gen.initialized());  // Should still be uninitialized.
+
+  // Initialize with valid parameters.
+  ASSERT_EQ(0, tone_gen.Init(fs, event, attenuation));
+  EXPECT_TRUE(tone_gen.initialized());
+  // NULL pointer to destination.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError,
+            tone_gen.Generate(kNumSamples, NULL));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand.cc b/third_party/libwebrtc/modules/audio_coding/neteq/expand.cc
new file mode 100644
index 0000000000..9c3274609f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand.cc
@@ -0,0 +1,888 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/expand.h"
+
+#include <string.h>  // memset
+
+#include <algorithm>  // min, max
+#include <limits>     // numeric_limits<T>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+Expand::Expand(BackgroundNoise* background_noise,
+               SyncBuffer* sync_buffer,
+               RandomVector* random_vector,
+               StatisticsCalculator* statistics,
+               int fs,
+               size_t num_channels)
+    : random_vector_(random_vector),
+      sync_buffer_(sync_buffer),
+      first_expand_(true),
+      fs_hz_(fs),
+      num_channels_(num_channels),
+      consecutive_expands_(0),
+      background_noise_(background_noise),
+      statistics_(statistics),
+      overlap_length_(5 * fs / 8000),
+      lag_index_direction_(0),
+      current_lag_index_(0),
+      stop_muting_(false),
+      expand_duration_samples_(0),
+      channel_parameters_(new ChannelParameters[num_channels_]) {
+  RTC_DCHECK(fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000);
+  RTC_DCHECK_LE(fs,
+                static_cast<int>(kMaxSampleRate));  // Should not be possible.
+  RTC_DCHECK_GT(num_channels_, 0);
+  memset(expand_lags_, 0, sizeof(expand_lags_));
+  Reset();
+}
+
+Expand::~Expand() = default;
+
+void Expand::Reset() {
+  first_expand_ = true;
+  consecutive_expands_ = 0;
+  max_lag_ = 0;
+  for (size_t ix = 0; ix < num_channels_; ++ix) {
+    channel_parameters_[ix].expand_vector0.Clear();
+    channel_parameters_[ix].expand_vector1.Clear();
+  }
+}
+
+int Expand::Process(AudioMultiVector* output) {
+  int16_t random_vector[kMaxSampleRate / 8000 * 120 + 30];
+  int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125];
+  static const int kTempDataSize = 3600;
+  int16_t temp_data[kTempDataSize];  // TODO(hlundin) Remove this.
+  int16_t* voiced_vector_storage = temp_data;
+  int16_t* voiced_vector = &voiced_vector_storage[overlap_length_];
+  static const size_t kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+  int16_t unvoiced_array_memory[kNoiseLpcOrder + kMaxSampleRate / 8000 * 125];
+  int16_t* unvoiced_vector = unvoiced_array_memory + kUnvoicedLpcOrder;
+  int16_t* noise_vector = unvoiced_array_memory + kNoiseLpcOrder;
+
+  int fs_mult = fs_hz_ / 8000;
+
+  if (first_expand_) {
+    // Perform initial setup if this is the first expansion since last reset.
+    AnalyzeSignal(random_vector);
+    first_expand_ = false;
+    expand_duration_samples_ = 0;
+  } else {
+    // This is not the first expansion, parameters are already estimated.
+    // Extract a noise segment.
+    size_t rand_length = max_lag_;
+    // This only applies to SWB where length could be larger than 256.
+    RTC_DCHECK_LE(rand_length, kMaxSampleRate / 8000 * 120 + 30);
+    GenerateRandomVector(2, rand_length, random_vector);
+  }
+
+  // Generate signal.
+  UpdateLagIndex();
+
+  // Voiced part.
+  // Generate a weighted vector with the current lag.
+  size_t expansion_vector_length = max_lag_ + overlap_length_;
+  size_t current_lag = expand_lags_[current_lag_index_];
+  // Copy lag+overlap data.
+  size_t expansion_vector_position =
+      expansion_vector_length - current_lag - overlap_length_;
+  size_t temp_length = current_lag + overlap_length_;
+  for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
+    ChannelParameters& parameters = channel_parameters_[channel_ix];
+    if (current_lag_index_ == 0) {
+      // Use only expand_vector0.
+      RTC_DCHECK_LE(expansion_vector_position + temp_length,
+                    parameters.expand_vector0.Size());
+      parameters.expand_vector0.CopyTo(temp_length, expansion_vector_position,
+                                       voiced_vector_storage);
+    } else if (current_lag_index_ == 1) {
+      std::unique_ptr<int16_t[]> temp_0(new int16_t[temp_length]);
+      parameters.expand_vector0.CopyTo(temp_length, expansion_vector_position,
+                                       temp_0.get());
+      std::unique_ptr<int16_t[]> temp_1(new int16_t[temp_length]);
+      parameters.expand_vector1.CopyTo(temp_length, expansion_vector_position,
+                                       temp_1.get());
+      // Mix 3/4 of expand_vector0 with 1/4 of expand_vector1.
+      WebRtcSpl_ScaleAndAddVectorsWithRound(temp_0.get(), 3, temp_1.get(), 1, 2,
+                                            voiced_vector_storage, temp_length);
+    } else if (current_lag_index_ == 2) {
+      // Mix 1/2 of expand_vector0 with 1/2 of expand_vector1.
+      RTC_DCHECK_LE(expansion_vector_position + temp_length,
+                    parameters.expand_vector0.Size());
+      RTC_DCHECK_LE(expansion_vector_position + temp_length,
+                    parameters.expand_vector1.Size());
+
+      std::unique_ptr<int16_t[]> temp_0(new int16_t[temp_length]);
+      parameters.expand_vector0.CopyTo(temp_length, expansion_vector_position,
+                                       temp_0.get());
+      std::unique_ptr<int16_t[]> temp_1(new int16_t[temp_length]);
+      parameters.expand_vector1.CopyTo(temp_length, expansion_vector_position,
+                                       temp_1.get());
+      WebRtcSpl_ScaleAndAddVectorsWithRound(temp_0.get(), 1, temp_1.get(), 1, 1,
+                                            voiced_vector_storage, temp_length);
+    }
+
+    // Get tapering window parameters. Values are in Q15.
+    int16_t muting_window, muting_window_increment;
+    int16_t unmuting_window, unmuting_window_increment;
+    if (fs_hz_ == 8000) {
+      muting_window = DspHelper::kMuteFactorStart8kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement8kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart8kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement8kHz;
+    } else if (fs_hz_ == 16000) {
+      muting_window = DspHelper::kMuteFactorStart16kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement16kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart16kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement16kHz;
+    } else if (fs_hz_ == 32000) {
+      muting_window = DspHelper::kMuteFactorStart32kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement32kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart32kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement32kHz;
+    } else {  // fs_ == 48000
+      muting_window = DspHelper::kMuteFactorStart48kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement48kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart48kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement48kHz;
+    }
+
+    // Smooth the expanded if it has not been muted to a low amplitude and
+    // `current_voice_mix_factor` is larger than 0.5.
+    if ((parameters.mute_factor > 819) &&
+        (parameters.current_voice_mix_factor > 8192)) {
+      size_t start_ix = sync_buffer_->Size() - overlap_length_;
+      for (size_t i = 0; i < overlap_length_; i++) {
+        // Do overlap add between new vector and overlap.
+        (*sync_buffer_)[channel_ix][start_ix + i] =
+            (((*sync_buffer_)[channel_ix][start_ix + i] * muting_window) +
+             (((parameters.mute_factor * voiced_vector_storage[i]) >> 14) *
+              unmuting_window) +
+             16384) >>
+            15;
+        muting_window += muting_window_increment;
+        unmuting_window += unmuting_window_increment;
+      }
+    } else if (parameters.mute_factor == 0) {
+      // The expanded signal will consist of only comfort noise if
+      // mute_factor = 0. Set the output length to 15 ms for best noise
+      // production.
+      // TODO(hlundin): This has been disabled since the length of
+      // parameters.expand_vector0 and parameters.expand_vector1 no longer
+      // match with expand_lags_, causing invalid reads and writes. Is it a good
+      // idea to enable this again, and solve the vector size problem?
+      //      max_lag_ = fs_mult * 120;
+      //      expand_lags_[0] = fs_mult * 120;
+      //      expand_lags_[1] = fs_mult * 120;
+      //      expand_lags_[2] = fs_mult * 120;
+    }
+
+    // Unvoiced part.
+    // Filter `scaled_random_vector` through `ar_filter_`.
+    memcpy(unvoiced_vector - kUnvoicedLpcOrder, parameters.ar_filter_state,
+           sizeof(int16_t) * kUnvoicedLpcOrder);
+    int32_t add_constant = 0;
+    if (parameters.ar_gain_scale > 0) {
+      add_constant = 1 << (parameters.ar_gain_scale - 1);
+    }
+    WebRtcSpl_AffineTransformVector(scaled_random_vector, random_vector,
+                                    parameters.ar_gain, add_constant,
+                                    parameters.ar_gain_scale, current_lag);
+    WebRtcSpl_FilterARFastQ12(scaled_random_vector, unvoiced_vector,
+                              parameters.ar_filter, kUnvoicedLpcOrder + 1,
+                              current_lag);
+    memcpy(parameters.ar_filter_state,
+           &(unvoiced_vector[current_lag - kUnvoicedLpcOrder]),
+           sizeof(int16_t) * kUnvoicedLpcOrder);
+
+    // Combine voiced and unvoiced contributions.
+
+    // Set a suitable cross-fading slope.
+    // For lag =
+    //   <= 31 * fs_mult            => go from 1 to 0 in about 8 ms;
+    //  (>= 31 .. <= 63) * fs_mult  => go from 1 to 0 in about 16 ms;
+    //   >= 64 * fs_mult            => go from 1 to 0 in about 32 ms.
+    // temp_shift = getbits(max_lag_) - 5.
+    int temp_shift =
+        (31 - WebRtcSpl_NormW32(rtc::dchecked_cast<int32_t>(max_lag_))) - 5;
+    int16_t mix_factor_increment = 256 >> temp_shift;
+    if (stop_muting_) {
+      mix_factor_increment = 0;
+    }
+
+    // Create combined signal by shifting in more and more of unvoiced part.
+    temp_shift = 8 - temp_shift;  // = getbits(mix_factor_increment).
+    size_t temp_length =
+        (parameters.current_voice_mix_factor - parameters.voice_mix_factor) >>
+        temp_shift;
+    temp_length = std::min(temp_length, current_lag);
+    DspHelper::CrossFade(voiced_vector, unvoiced_vector, temp_length,
+                         &parameters.current_voice_mix_factor,
+                         mix_factor_increment, temp_data);
+
+    // End of cross-fading period was reached before end of expanded signal
+    // path. Mix the rest with a fixed mixing factor.
+    if (temp_length < current_lag) {
+      if (mix_factor_increment != 0) {
+        parameters.current_voice_mix_factor = parameters.voice_mix_factor;
+      }
+      int16_t temp_scale = 16384 - parameters.current_voice_mix_factor;
+      WebRtcSpl_ScaleAndAddVectorsWithRound(
+          voiced_vector + temp_length, parameters.current_voice_mix_factor,
+          unvoiced_vector + temp_length, temp_scale, 14,
+          temp_data + temp_length, current_lag - temp_length);
+    }
+
+    // Select muting slope depending on how many consecutive expands we have
+    // done.
+    if (consecutive_expands_ == 3) {
+      // Let the mute factor decrease from 1.0 to 0.95 in 6.25 ms.
+      // mute_slope = 0.0010 / fs_mult in Q20.
+      parameters.mute_slope = std::max(parameters.mute_slope, 1049 / fs_mult);
+    }
+    if (consecutive_expands_ == 7) {
+      // Let the mute factor decrease from 1.0 to 0.90 in 6.25 ms.
+      // mute_slope = 0.0020 / fs_mult in Q20.
+      parameters.mute_slope = std::max(parameters.mute_slope, 2097 / fs_mult);
+    }
+
+    // Mute segment according to slope value.
+    if ((consecutive_expands_ != 0) || !parameters.onset) {
+      // Mute to the previous level, then continue with the muting.
+      WebRtcSpl_AffineTransformVector(
+          temp_data, temp_data, parameters.mute_factor, 8192, 14, current_lag);
+
+      if (!stop_muting_) {
+        DspHelper::MuteSignal(temp_data, parameters.mute_slope, current_lag);
+
+        // Shift by 6 to go from Q20 to Q14.
+        // TODO(hlundin): Adding 8192 before shifting 6 steps seems wrong.
+        // Legacy.
+        int16_t gain = static_cast<int16_t>(
+            16384 - (((current_lag * parameters.mute_slope) + 8192) >> 6));
+        gain = ((gain * parameters.mute_factor) + 8192) >> 14;
+
+        // Guard against getting stuck with very small (but sometimes audible)
+        // gain.
+        if ((consecutive_expands_ > 3) && (gain >= parameters.mute_factor)) {
+          parameters.mute_factor = 0;
+        } else {
+          parameters.mute_factor = gain;
+        }
+      }
+    }
+
+    // Background noise part.
+    background_noise_->GenerateBackgroundNoise(
+        random_vector, channel_ix, channel_parameters_[channel_ix].mute_slope,
+        TooManyExpands(), current_lag, unvoiced_array_memory);
+
+    // Add background noise to the combined voiced-unvoiced signal.
+    for (size_t i = 0; i < current_lag; i++) {
+      temp_data[i] = temp_data[i] + noise_vector[i];
+    }
+    if (channel_ix == 0) {
+      output->AssertSize(current_lag);
+    } else {
+      RTC_DCHECK_EQ(output->Size(), current_lag);
+    }
+    (*output)[channel_ix].OverwriteAt(temp_data, current_lag, 0);
+  }
+
+  // Increase call number and cap it.
+  consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands
+                             ? kMaxConsecutiveExpands
+                             : consecutive_expands_ + 1;
+  expand_duration_samples_ += output->Size();
+  // Clamp the duration counter at 2 seconds.
+  expand_duration_samples_ = std::min(expand_duration_samples_,
+                                      rtc::dchecked_cast<size_t>(fs_hz_ * 2));
+  return 0;
+}
+
+void Expand::SetParametersForNormalAfterExpand() {
+  current_lag_index_ = 0;
+  lag_index_direction_ = 0;
+  stop_muting_ = true;  // Do not mute signal any more.
+  statistics_->LogDelayedPacketOutageEvent(expand_duration_samples_, fs_hz_);
+  statistics_->EndExpandEvent(fs_hz_);
+}
+
+void Expand::SetParametersForMergeAfterExpand() {
+  current_lag_index_ = -1;  /* out of the 3 possible ones */
+  lag_index_direction_ = 1; /* make sure we get the "optimal" lag */
+  stop_muting_ = true;
+  statistics_->EndExpandEvent(fs_hz_);
+}
+
+bool Expand::Muted() const {
+  if (first_expand_ || stop_muting_)
+    return false;
+  RTC_DCHECK(channel_parameters_);
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    if (channel_parameters_[ch].mute_factor != 0)
+      return false;
+  }
+  return true;
+}
+
+size_t Expand::overlap_length() const {
+  return overlap_length_;
+}
+
+void Expand::InitializeForAnExpandPeriod() {
+  lag_index_direction_ = 1;
+  current_lag_index_ = -1;
+  stop_muting_ = false;
+  random_vector_->set_seed_increment(1);
+  consecutive_expands_ = 0;
+  for (size_t ix = 0; ix < num_channels_; ++ix) {
+    channel_parameters_[ix].current_voice_mix_factor = 16384;  // 1.0 in Q14.
+    channel_parameters_[ix].mute_factor = 16384;               // 1.0 in Q14.
+    // Start with 0 gain for background noise.
+    background_noise_->SetMuteFactor(ix, 0);
+  }
+}
+
+bool Expand::TooManyExpands() {
+  return consecutive_expands_ >= kMaxConsecutiveExpands;
+}
+
+void Expand::AnalyzeSignal(int16_t* random_vector) {
+  int32_t auto_correlation[kUnvoicedLpcOrder + 1];
+  int16_t reflection_coeff[kUnvoicedLpcOrder];
+  int16_t correlation_vector[kMaxSampleRate / 8000 * 102];
+  size_t best_correlation_index[kNumCorrelationCandidates];
+  int16_t best_correlation[kNumCorrelationCandidates];
+  size_t best_distortion_index[kNumCorrelationCandidates];
+  int16_t best_distortion[kNumCorrelationCandidates];
+  int32_t correlation_vector2[(99 * kMaxSampleRate / 8000) + 1];
+  int32_t best_distortion_w32[kNumCorrelationCandidates];
+  static const size_t kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+  int16_t unvoiced_array_memory[kNoiseLpcOrder + kMaxSampleRate / 8000 * 125];
+  int16_t* unvoiced_vector = unvoiced_array_memory + kUnvoicedLpcOrder;
+
+  int fs_mult = fs_hz_ / 8000;
+
+  // Pre-calculate common multiplications with fs_mult.
+  size_t fs_mult_4 = static_cast<size_t>(fs_mult * 4);
+  size_t fs_mult_20 = static_cast<size_t>(fs_mult * 20);
+  size_t fs_mult_120 = static_cast<size_t>(fs_mult * 120);
+  size_t fs_mult_dist_len = fs_mult * kDistortionLength;
+  size_t fs_mult_lpc_analysis_len = fs_mult * kLpcAnalysisLength;
+
+  const size_t signal_length = static_cast<size_t>(256 * fs_mult);
+
+  const size_t audio_history_position = sync_buffer_->Size() - signal_length;
+  std::unique_ptr<int16_t[]> audio_history(new int16_t[signal_length]);
+  (*sync_buffer_)[0].CopyTo(signal_length, audio_history_position,
+                            audio_history.get());
+
+  // Initialize.
+  InitializeForAnExpandPeriod();
+
+  // Calculate correlation in downsampled domain (4 kHz sample rate).
+  size_t correlation_length = 51;  // TODO(hlundin): Legacy bit-exactness.
+  // If it is decided to break bit-exactness `correlation_length` should be
+  // initialized to the return value of Correlation().
+  Correlation(audio_history.get(), signal_length, correlation_vector);
+
+  // Find peaks in correlation vector.
+  DspHelper::PeakDetection(correlation_vector, correlation_length,
+                           kNumCorrelationCandidates, fs_mult,
+                           best_correlation_index, best_correlation);
+
+  // Adjust peak locations; cross-correlation lags start at 2.5 ms
+  // (20 * fs_mult samples).
+  best_correlation_index[0] += fs_mult_20;
+  best_correlation_index[1] += fs_mult_20;
+  best_correlation_index[2] += fs_mult_20;
+
+  // Calculate distortion around the `kNumCorrelationCandidates` best lags.
+  int distortion_scale = 0;
+  for (size_t i = 0; i < kNumCorrelationCandidates; i++) {
+    size_t min_index =
+        std::max(fs_mult_20, best_correlation_index[i] - fs_mult_4);
+    size_t max_index =
+        std::min(fs_mult_120 - 1, best_correlation_index[i] + fs_mult_4);
+    best_distortion_index[i] = DspHelper::MinDistortion(
+        &(audio_history[signal_length - fs_mult_dist_len]), min_index,
+        max_index, fs_mult_dist_len, &best_distortion_w32[i]);
+    distortion_scale = std::max(16 - WebRtcSpl_NormW32(best_distortion_w32[i]),
+                                distortion_scale);
+  }
+  // Shift the distortion values to fit in 16 bits.
+  WebRtcSpl_VectorBitShiftW32ToW16(best_distortion, kNumCorrelationCandidates,
+                                   best_distortion_w32, distortion_scale);
+
+  // Find the maximizing index `i` of the cost function
+  // f[i] = best_correlation[i] / best_distortion[i].
+  int32_t best_ratio = std::numeric_limits<int32_t>::min();
+  size_t best_index = std::numeric_limits<size_t>::max();
+  for (size_t i = 0; i < kNumCorrelationCandidates; ++i) {
+    int32_t ratio;
+    if (best_distortion[i] > 0) {
+      ratio = (best_correlation[i] * (1 << 16)) / best_distortion[i];
+    } else if (best_correlation[i] == 0) {
+      ratio = 0;  // No correlation set result to zero.
+    } else {
+      ratio = std::numeric_limits<int32_t>::max();  // Denominator is zero.
+    }
+    if (ratio > best_ratio) {
+      best_index = i;
+      best_ratio = ratio;
+    }
+  }
+
+  size_t distortion_lag = best_distortion_index[best_index];
+  size_t correlation_lag = best_correlation_index[best_index];
+  max_lag_ = std::max(distortion_lag, correlation_lag);
+
+  // Calculate the exact best correlation in the range between
+  // `correlation_lag` and `distortion_lag`.
+  correlation_length = std::max(std::min(distortion_lag + 10, fs_mult_120),
+                                static_cast<size_t>(60 * fs_mult));
+
+  size_t start_index = std::min(distortion_lag, correlation_lag);
+  size_t correlation_lags = static_cast<size_t>(
+      WEBRTC_SPL_ABS_W16((distortion_lag - correlation_lag)) + 1);
+  RTC_DCHECK_LE(correlation_lags, static_cast<size_t>(99 * fs_mult + 1));
+
+  for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
+    ChannelParameters& parameters = channel_parameters_[channel_ix];
+    if (channel_ix > 0) {
+      // When channel_ix == 0, audio_history contains the correct audio. For the
+      // other cases, we will have to copy the correct channel into
+      // audio_history.
+      (*sync_buffer_)[channel_ix].CopyTo(signal_length, audio_history_position,
+                                         audio_history.get());
+    }
+
+    // Calculate suitable scaling.
+    int16_t signal_max = WebRtcSpl_MaxAbsValueW16(
+        &audio_history[signal_length - correlation_length - start_index -
+                       correlation_lags],
+        correlation_length + start_index + correlation_lags - 1);
+    int correlation_scale =
+        (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
+        (31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31;
+    correlation_scale = std::max(0, correlation_scale);
+
+    // Calculate the correlation, store in `correlation_vector2`.
+    WebRtcSpl_CrossCorrelation(
+        correlation_vector2,
+        &(audio_history[signal_length - correlation_length]),
+        &(audio_history[signal_length - correlation_length - start_index]),
+        correlation_length, correlation_lags, correlation_scale, -1);
+
+    // Find maximizing index.
+    best_index = WebRtcSpl_MaxIndexW32(correlation_vector2, correlation_lags);
+    int32_t max_correlation = correlation_vector2[best_index];
+    // Compensate index with start offset.
+    best_index = best_index + start_index;
+
+    // Calculate energies.
+    int32_t energy1 = WebRtcSpl_DotProductWithScale(
+        &(audio_history[signal_length - correlation_length]),
+        &(audio_history[signal_length - correlation_length]),
+        correlation_length, correlation_scale);
+    int32_t energy2 = WebRtcSpl_DotProductWithScale(
+        &(audio_history[signal_length - correlation_length - best_index]),
+        &(audio_history[signal_length - correlation_length - best_index]),
+        correlation_length, correlation_scale);
+
+    // Calculate the correlation coefficient between the two portions of the
+    // signal.
+    int32_t corr_coefficient;
+    if ((energy1 > 0) && (energy2 > 0)) {
+      int energy1_scale = std::max(16 - WebRtcSpl_NormW32(energy1), 0);
+      int energy2_scale = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+      // Make sure total scaling is even (to simplify scale factor after sqrt).
+      if ((energy1_scale + energy2_scale) & 1) {
+        // If sum is odd, add 1 to make it even.
+        energy1_scale += 1;
+      }
+      int32_t scaled_energy1 = energy1 >> energy1_scale;
+      int32_t scaled_energy2 = energy2 >> energy2_scale;
+      int16_t sqrt_energy_product = static_cast<int16_t>(
+          WebRtcSpl_SqrtFloor(scaled_energy1 * scaled_energy2));
+      // Calculate max_correlation / sqrt(energy1 * energy2) in Q14.
+      int cc_shift = 14 - (energy1_scale + energy2_scale) / 2;
+      max_correlation = WEBRTC_SPL_SHIFT_W32(max_correlation, cc_shift);
+      corr_coefficient =
+          WebRtcSpl_DivW32W16(max_correlation, sqrt_energy_product);
+      // Cap at 1.0 in Q14.
+      corr_coefficient = std::min(16384, corr_coefficient);
+    } else {
+      corr_coefficient = 0;
+    }
+
+    // Extract the two vectors expand_vector0 and expand_vector1 from
+    // `audio_history`.
+    size_t expansion_length = max_lag_ + overlap_length_;
+    const int16_t* vector1 = &(audio_history[signal_length - expansion_length]);
+    const int16_t* vector2 = vector1 - distortion_lag;
+    // Normalize the second vector to the same energy as the first.
+    energy1 = WebRtcSpl_DotProductWithScale(vector1, vector1, expansion_length,
+                                            correlation_scale);
+    energy2 = WebRtcSpl_DotProductWithScale(vector2, vector2, expansion_length,
+                                            correlation_scale);
+    // Confirm that amplitude ratio sqrt(energy1 / energy2) is within 0.5 - 2.0,
+    // i.e., energy1 / energy2 is within 0.25 - 4.
+    int16_t amplitude_ratio;
+    if ((energy1 / 4 < energy2) && (energy1 > energy2 / 4)) {
+      // Energy constraint fulfilled. Use both vectors and scale them
+      // accordingly.
+      int32_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+      int32_t scaled_energy1 = scaled_energy2 - 13;
+      // Calculate scaled_energy1 / scaled_energy2 in Q13.
+      int32_t energy_ratio =
+          WebRtcSpl_DivW32W16(WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
+                              static_cast<int16_t>(energy2 >> scaled_energy2));
+      // Calculate sqrt ratio in Q13 (sqrt of en1/en2 in Q26).
+      amplitude_ratio =
+          static_cast<int16_t>(WebRtcSpl_SqrtFloor(energy_ratio << 13));
+      // Copy the two vectors and give them the same energy.
+      parameters.expand_vector0.Clear();
+      parameters.expand_vector0.PushBack(vector1, expansion_length);
+      parameters.expand_vector1.Clear();
+      if (parameters.expand_vector1.Size() < expansion_length) {
+        parameters.expand_vector1.Extend(expansion_length -
+                                         parameters.expand_vector1.Size());
+      }
+      std::unique_ptr<int16_t[]> temp_1(new int16_t[expansion_length]);
+      WebRtcSpl_AffineTransformVector(
+          temp_1.get(), const_cast<int16_t*>(vector2), amplitude_ratio, 4096,
+          13, expansion_length);
+      parameters.expand_vector1.OverwriteAt(temp_1.get(), expansion_length, 0);
+    } else {
+      // Energy change constraint not fulfilled. Only use last vector.
+      parameters.expand_vector0.Clear();
+      parameters.expand_vector0.PushBack(vector1, expansion_length);
+      // Copy from expand_vector0 to expand_vector1.
+      parameters.expand_vector0.CopyTo(&parameters.expand_vector1);
+      // Set the energy_ratio since it is used by muting slope.
+      if ((energy1 / 4 < energy2) || (energy2 == 0)) {
+        amplitude_ratio = 4096;  // 0.5 in Q13.
+      } else {
+        amplitude_ratio = 16384;  // 2.0 in Q13.
+      }
+    }
+
+    // Set the 3 lag values.
+    if (distortion_lag == correlation_lag) {
+      expand_lags_[0] = distortion_lag;
+      expand_lags_[1] = distortion_lag;
+      expand_lags_[2] = distortion_lag;
+    } else {
+      // `distortion_lag` and `correlation_lag` are not equal; use different
+      // combinations of the two.
+      // First lag is `distortion_lag` only.
+      expand_lags_[0] = distortion_lag;
+      // Second lag is the average of the two.
+      expand_lags_[1] = (distortion_lag + correlation_lag) / 2;
+      // Third lag is the average again, but rounding towards `correlation_lag`.
+      if (distortion_lag > correlation_lag) {
+        expand_lags_[2] = (distortion_lag + correlation_lag - 1) / 2;
+      } else {
+        expand_lags_[2] = (distortion_lag + correlation_lag + 1) / 2;
+      }
+    }
+
+    // Calculate the LPC and the gain of the filters.
+
+    // Calculate kUnvoicedLpcOrder + 1 lags of the auto-correlation function.
+    size_t temp_index =
+        signal_length - fs_mult_lpc_analysis_len - kUnvoicedLpcOrder;
+    // Copy signal to temporary vector to be able to pad with leading zeros.
+    int16_t* temp_signal =
+        new int16_t[fs_mult_lpc_analysis_len + kUnvoicedLpcOrder];
+    memset(temp_signal, 0,
+           sizeof(int16_t) * (fs_mult_lpc_analysis_len + kUnvoicedLpcOrder));
+    memcpy(&temp_signal[kUnvoicedLpcOrder],
+           &audio_history[temp_index + kUnvoicedLpcOrder],
+           sizeof(int16_t) * fs_mult_lpc_analysis_len);
+    CrossCorrelationWithAutoShift(
+        &temp_signal[kUnvoicedLpcOrder], &temp_signal[kUnvoicedLpcOrder],
+        fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1, -1, auto_correlation);
+    delete[] temp_signal;
+
+    // Verify that variance is positive.
+    if (auto_correlation[0] > 0) {
+      // Estimate AR filter parameters using Levinson-Durbin algorithm;
+      // kUnvoicedLpcOrder + 1 filter coefficients.
+      int16_t stability =
+          WebRtcSpl_LevinsonDurbin(auto_correlation, parameters.ar_filter,
+                                   reflection_coeff, kUnvoicedLpcOrder);
+
+      // Keep filter parameters only if filter is stable.
+      if (stability != 1) {
+        // Set first coefficient to 4096 (1.0 in Q12).
+        parameters.ar_filter[0] = 4096;
+        // Set remaining `kUnvoicedLpcOrder` coefficients to zero.
+        WebRtcSpl_MemSetW16(parameters.ar_filter + 1, 0, kUnvoicedLpcOrder);
+      }
+    }
+
+    if (channel_ix == 0) {
+      // Extract a noise segment.
+      size_t noise_length;
+      if (distortion_lag < 40) {
+        noise_length = 2 * distortion_lag + 30;
+      } else {
+        noise_length = distortion_lag + 30;
+      }
+      if (noise_length <= RandomVector::kRandomTableSize) {
+        memcpy(random_vector, RandomVector::kRandomTable,
+               sizeof(int16_t) * noise_length);
+      } else {
+        // Only applies to SWB where length could be larger than
+        // `kRandomTableSize`.
+        memcpy(random_vector, RandomVector::kRandomTable,
+               sizeof(int16_t) * RandomVector::kRandomTableSize);
+        RTC_DCHECK_LE(noise_length, kMaxSampleRate / 8000 * 120 + 30);
+        random_vector_->IncreaseSeedIncrement(2);
+        random_vector_->Generate(
+            noise_length - RandomVector::kRandomTableSize,
+            &random_vector[RandomVector::kRandomTableSize]);
+      }
+    }
+
+    // Set up state vector and calculate scale factor for unvoiced filtering.
+    memcpy(parameters.ar_filter_state,
+           &(audio_history[signal_length - kUnvoicedLpcOrder]),
+           sizeof(int16_t) * kUnvoicedLpcOrder);
+    memcpy(unvoiced_vector - kUnvoicedLpcOrder,
+           &(audio_history[signal_length - 128 - kUnvoicedLpcOrder]),
+           sizeof(int16_t) * kUnvoicedLpcOrder);
+    WebRtcSpl_FilterMAFastQ12(&audio_history[signal_length - 128],
+                              unvoiced_vector, parameters.ar_filter,
+                              kUnvoicedLpcOrder + 1, 128);
+    const int unvoiced_max_abs = [&] {
+      const int16_t max_abs = WebRtcSpl_MaxAbsValueW16(unvoiced_vector, 128);
+      // Since WebRtcSpl_MaxAbsValueW16 returns 2^15 - 1 when the input contains
+      // -2^15, we have to conservatively bump the return value by 1
+      // if it is 2^15 - 1.
+      return max_abs == WEBRTC_SPL_WORD16_MAX ? max_abs + 1 : max_abs;
+    }();
+    // Pick the smallest n such that 2^n > unvoiced_max_abs; then the maximum
+    // value of the dot product is less than 2^7 * 2^(2*n) = 2^(2*n + 7), so to
+    // prevent overflows we want 2n + 7 <= 31, which means we should shift by
+    // 2n + 7 - 31 bits, if this value is greater than zero.
+    int unvoiced_prescale =
+        std::max(0, 2 * WebRtcSpl_GetSizeInBits(unvoiced_max_abs) - 24);
+
+    int32_t unvoiced_energy = WebRtcSpl_DotProductWithScale(
+        unvoiced_vector, unvoiced_vector, 128, unvoiced_prescale);
+
+    // Normalize `unvoiced_energy` to 28 or 29 bits to preserve sqrt() accuracy.
+    int16_t unvoiced_scale = WebRtcSpl_NormW32(unvoiced_energy) - 3;
+    // Make sure we do an odd number of shifts since we already have 7 shifts
+    // from dividing with 128 earlier. This will make the total scale factor
+    // even, which is suitable for the sqrt.
+    unvoiced_scale += ((unvoiced_scale & 0x1) ^ 0x1);
+    unvoiced_energy = WEBRTC_SPL_SHIFT_W32(unvoiced_energy, unvoiced_scale);
+    int16_t unvoiced_gain =
+        static_cast<int16_t>(WebRtcSpl_SqrtFloor(unvoiced_energy));
+    parameters.ar_gain_scale =
+        13 + (unvoiced_scale + 7 - unvoiced_prescale) / 2;
+    parameters.ar_gain = unvoiced_gain;
+
+    // Calculate voice_mix_factor from corr_coefficient.
+    // Let x = corr_coefficient. Then, we compute:
+    // if (x > 0.48)
+    //   voice_mix_factor = (-5179 + 19931x - 16422x^2 + 5776x^3) / 4096;
+    // else
+    //   voice_mix_factor = 0;
+    if (corr_coefficient > 7875) {
+      int16_t x1, x2, x3;
+      // `corr_coefficient` is in Q14.
+      x1 = static_cast<int16_t>(corr_coefficient);
+      x2 = (x1 * x1) >> 14;  // Shift 14 to keep result in Q14.
+      x3 = (x1 * x2) >> 14;
+      static const int kCoefficients[4] = {-5179, 19931, -16422, 5776};
+      int32_t temp_sum = kCoefficients[0] * 16384;
+      temp_sum += kCoefficients[1] * x1;
+      temp_sum += kCoefficients[2] * x2;
+      temp_sum += kCoefficients[3] * x3;
+      parameters.voice_mix_factor =
+          static_cast<int16_t>(std::min(temp_sum / 4096, 16384));
+      parameters.voice_mix_factor =
+          std::max(parameters.voice_mix_factor, static_cast<int16_t>(0));
+    } else {
+      parameters.voice_mix_factor = 0;
+    }
+
+    // Calculate muting slope. Reuse value from earlier scaling of
+    // `expand_vector0` and `expand_vector1`.
+    int16_t slope = amplitude_ratio;
+    if (slope > 12288) {
+      // slope > 1.5.
+      // Calculate (1 - (1 / slope)) / distortion_lag =
+      // (slope - 1) / (distortion_lag * slope).
+      // `slope` is in Q13, so 1 corresponds to 8192. Shift up to Q25 before
+      // the division.
+      // Shift the denominator from Q13 to Q5 before the division. The result of
+      // the division will then be in Q20.
+      int16_t denom =
+          rtc::saturated_cast<int16_t>((distortion_lag * slope) >> 8);
+      int temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12, denom);
+      if (slope > 14746) {
+        // slope > 1.8.
+        // Divide by 2, with proper rounding.
+        parameters.mute_slope = (temp_ratio + 1) / 2;
+      } else {
+        // Divide by 8, with proper rounding.
+        parameters.mute_slope = (temp_ratio + 4) / 8;
+      }
+      parameters.onset = true;
+    } else {
+      // Calculate (1 - slope) / distortion_lag.
+      // Shift `slope` by 7 to Q20 before the division. The result is in Q20.
+      parameters.mute_slope = WebRtcSpl_DivW32W16(
+          (8192 - slope) * 128, static_cast<int16_t>(distortion_lag));
+      if (parameters.voice_mix_factor <= 13107) {
+        // Make sure the mute factor decreases from 1.0 to 0.9 in no more than
+        // 6.25 ms.
+        // mute_slope >= 0.005 / fs_mult in Q20.
+        parameters.mute_slope = std::max(5243 / fs_mult, parameters.mute_slope);
+      } else if (slope > 8028) {
+        parameters.mute_slope = 0;
+      }
+      parameters.onset = false;
+    }
+  }
+}
+
+Expand::ChannelParameters::ChannelParameters()
+    : mute_factor(16384),
+      ar_gain(0),
+      ar_gain_scale(0),
+      voice_mix_factor(0),
+      current_voice_mix_factor(0),
+      onset(false),
+      mute_slope(0) {
+  memset(ar_filter, 0, sizeof(ar_filter));
+  memset(ar_filter_state, 0, sizeof(ar_filter_state));
+}
+
+void Expand::Correlation(const int16_t* input,
+                         size_t input_length,
+                         int16_t* output) const {
+  // Set parameters depending on sample rate.
+  const int16_t* filter_coefficients;
+  size_t num_coefficients;
+  int16_t downsampling_factor;
+  if (fs_hz_ == 8000) {
+    num_coefficients = 3;
+    downsampling_factor = 2;
+    filter_coefficients = DspHelper::kDownsample8kHzTbl;
+  } else if (fs_hz_ == 16000) {
+    num_coefficients = 5;
+    downsampling_factor = 4;
+    filter_coefficients = DspHelper::kDownsample16kHzTbl;
+  } else if (fs_hz_ == 32000) {
+    num_coefficients = 7;
+    downsampling_factor = 8;
+    filter_coefficients = DspHelper::kDownsample32kHzTbl;
+  } else {  // fs_hz_ == 48000.
+    num_coefficients = 7;
+    downsampling_factor = 12;
+    filter_coefficients = DspHelper::kDownsample48kHzTbl;
+  }
+
+  // Correlate from lag 10 to lag 60 in downsampled domain.
+  // (Corresponds to 20-120 for narrow-band, 40-240 for wide-band, and so on.)
+  static const size_t kCorrelationStartLag = 10;
+  static const size_t kNumCorrelationLags = 54;
+  static const size_t kCorrelationLength = 60;
+  // Downsample to 4 kHz sample rate.
+  static const size_t kDownsampledLength =
+      kCorrelationStartLag + kNumCorrelationLags + kCorrelationLength;
+  int16_t downsampled_input[kDownsampledLength];
+  static const size_t kFilterDelay = 0;
+  WebRtcSpl_DownsampleFast(
+      input + input_length - kDownsampledLength * downsampling_factor,
+      kDownsampledLength * downsampling_factor, downsampled_input,
+      kDownsampledLength, filter_coefficients, num_coefficients,
+      downsampling_factor, kFilterDelay);
+
+  // Normalize `downsampled_input` to using all 16 bits.
+  int16_t max_value =
+      WebRtcSpl_MaxAbsValueW16(downsampled_input, kDownsampledLength);
+  int16_t norm_shift = 16 - WebRtcSpl_NormW32(max_value);
+  WebRtcSpl_VectorBitShiftW16(downsampled_input, kDownsampledLength,
+                              downsampled_input, norm_shift);
+
+  int32_t correlation[kNumCorrelationLags];
+  CrossCorrelationWithAutoShift(
+      &downsampled_input[kDownsampledLength - kCorrelationLength],
+      &downsampled_input[kDownsampledLength - kCorrelationLength -
+                         kCorrelationStartLag],
+      kCorrelationLength, kNumCorrelationLags, -1, correlation);
+
+  // Normalize and move data from 32-bit to 16-bit vector.
+  int32_t max_correlation =
+      WebRtcSpl_MaxAbsValueW32(correlation, kNumCorrelationLags);
+  int16_t norm_shift2 = static_cast<int16_t>(
+      std::max(18 - WebRtcSpl_NormW32(max_correlation), 0));
+  WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation,
+                                   norm_shift2);
+}
+
+void Expand::UpdateLagIndex() {
+  current_lag_index_ = current_lag_index_ + lag_index_direction_;
+  // Change direction if needed.
+  if (current_lag_index_ <= 0) {
+    lag_index_direction_ = 1;
+  }
+  if (current_lag_index_ >= kNumLags - 1) {
+    lag_index_direction_ = -1;
+  }
+}
+
+Expand* ExpandFactory::Create(BackgroundNoise* background_noise,
+                              SyncBuffer* sync_buffer,
+                              RandomVector* random_vector,
+                              StatisticsCalculator* statistics,
+                              int fs,
+                              size_t num_channels) const {
+  return new Expand(background_noise, sync_buffer, random_vector, statistics,
+                    fs, num_channels);
+}
+
+void Expand::GenerateRandomVector(int16_t seed_increment,
+                                  size_t length,
+                                  int16_t* random_vector) {
+  // TODO(turajs): According to hlundin The loop should not be needed. Should be
+  // just as good to generate all of the vector in one call.
+  size_t samples_generated = 0;
+  const size_t kMaxRandSamples = RandomVector::kRandomTableSize;
+  while (samples_generated < length) {
+    size_t rand_length = std::min(length - samples_generated, kMaxRandSamples);
+    random_vector_->IncreaseSeedIncrement(seed_increment);
+    random_vector_->Generate(rand_length, &random_vector[samples_generated]);
+    samples_generated += rand_length;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand.h b/third_party/libwebrtc/modules/audio_coding/neteq/expand.h
new file mode 100644
index 0000000000..2e64583ec2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand.h
@@ -0,0 +1,154 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
+#define MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
+
+
+#include <memory>
+
+#include "modules/audio_coding/neteq/audio_vector.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class AudioMultiVector;
+class BackgroundNoise;
+class RandomVector;
+class StatisticsCalculator;
+class SyncBuffer;
+
+// This class handles extrapolation of audio data from the sync_buffer to
+// produce packet-loss concealment.
+// TODO(hlundin): Refactor this class to divide the long methods into shorter
+// ones.
+class Expand {
+ public:
+  Expand(BackgroundNoise* background_noise,
+         SyncBuffer* sync_buffer,
+         RandomVector* random_vector,
+         StatisticsCalculator* statistics,
+         int fs,
+         size_t num_channels);
+
+  virtual ~Expand();
+
+  Expand(const Expand&) = delete;
+  Expand& operator=(const Expand&) = delete;
+
+  // Resets the object.
+  virtual void Reset();
+
+  // The main method to produce concealment data. The data is appended to the
+  // end of `output`.
+  virtual int Process(AudioMultiVector* output);
+
+  // Prepare the object to do extra expansion during normal operation following
+  // a period of expands.
+  virtual void SetParametersForNormalAfterExpand();
+
+  // Prepare the object to do extra expansion during merge operation following
+  // a period of expands.
+  virtual void SetParametersForMergeAfterExpand();
+
+  // Returns the mute factor for `channel`.
+  int16_t MuteFactor(size_t channel) const {
+    RTC_DCHECK_LT(channel, num_channels_);
+    return channel_parameters_[channel].mute_factor;
+  }
+
+  // Returns true if expansion has been faded down to zero amplitude (for all
+  // channels); false otherwise.
+  bool Muted() const;
+
+  // Accessors and mutators.
+  virtual size_t overlap_length() const;
+  size_t max_lag() const { return max_lag_; }
+
+ protected:
+  static const int kMaxConsecutiveExpands = 200;
+  void GenerateRandomVector(int16_t seed_increment,
+                            size_t length,
+                            int16_t* random_vector);
+
+  // Initializes member variables at the beginning of an expand period.
+  void InitializeForAnExpandPeriod();
+
+  bool TooManyExpands();
+
+  // Analyzes the signal history in `sync_buffer_`, and set up all parameters
+  // necessary to produce concealment data.
+  void AnalyzeSignal(int16_t* random_vector);
+
+  RandomVector* const random_vector_;
+  SyncBuffer* const sync_buffer_;
+  bool first_expand_;
+  const int fs_hz_;
+  const size_t num_channels_;
+  int consecutive_expands_;
+
+ private:
+  static const size_t kUnvoicedLpcOrder = 6;
+  static const size_t kNumCorrelationCandidates = 3;
+  static const size_t kDistortionLength = 20;
+  static const size_t kLpcAnalysisLength = 160;
+  static const size_t kMaxSampleRate = 48000;
+  static const int kNumLags = 3;
+
+  struct ChannelParameters {
+    ChannelParameters();
+    int16_t mute_factor;
+    int16_t ar_filter[kUnvoicedLpcOrder + 1];
+    int16_t ar_filter_state[kUnvoicedLpcOrder];
+    int16_t ar_gain;
+    int16_t ar_gain_scale;
+    int16_t voice_mix_factor;         /* Q14 */
+    int16_t current_voice_mix_factor; /* Q14 */
+    AudioVector expand_vector0;
+    AudioVector expand_vector1;
+    bool onset;
+    int mute_slope; /* Q20 */
+  };
+
+  // Calculate the auto-correlation of `input`, with length `input_length`
+  // samples. The correlation is calculated from a downsampled version of
+  // `input`, and is written to `output`.
+  void Correlation(const int16_t* input,
+                   size_t input_length,
+                   int16_t* output) const;
+
+  void UpdateLagIndex();
+
+  BackgroundNoise* const background_noise_;
+  StatisticsCalculator* const statistics_;
+  const size_t overlap_length_;
+  size_t max_lag_;
+  size_t expand_lags_[kNumLags];
+  int lag_index_direction_;
+  int current_lag_index_;
+  bool stop_muting_;
+  size_t expand_duration_samples_;
+  std::unique_ptr<ChannelParameters[]> channel_parameters_;
+};
+
+struct ExpandFactory {
+  ExpandFactory() {}
+  virtual ~ExpandFactory() {}
+
+  virtual Expand* Create(BackgroundNoise* background_noise,
+                         SyncBuffer* sync_buffer,
+                         RandomVector* random_vector,
+                         StatisticsCalculator* statistics,
+                         int fs,
+                         size_t num_channels) const;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.cc b/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.cc
new file mode 100644
index 0000000000..a91358b489
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.cc
@@ -0,0 +1,71 @@
+/*  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/expand_uma_logger.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+std::unique_ptr<TickTimer::Countdown> GetNewCountdown(
+    const TickTimer& tick_timer,
+    int logging_period_s) {
+  return tick_timer.GetNewCountdown((logging_period_s * 1000) /
+                                    tick_timer.ms_per_tick());
+}
+}  // namespace
+
+ExpandUmaLogger::ExpandUmaLogger(absl::string_view uma_name,
+                                 int logging_period_s,
+                                 const TickTimer* tick_timer)
+    : uma_name_(uma_name),
+      logging_period_s_(logging_period_s),
+      tick_timer_(*tick_timer),
+      timer_(GetNewCountdown(tick_timer_, logging_period_s_)) {
+  RTC_DCHECK(tick_timer);
+  RTC_DCHECK_GT(logging_period_s_, 0);
+}
+
+ExpandUmaLogger::~ExpandUmaLogger() = default;
+
+void ExpandUmaLogger::UpdateSampleCounter(uint64_t samples,
+                                          int sample_rate_hz) {
+  if ((last_logged_value_ && *last_logged_value_ > samples) ||
+      sample_rate_hz_ != sample_rate_hz) {
+    // Sanity checks. The incremental counter moved backwards, or sample rate
+    // changed.
+    last_logged_value_.reset();
+  }
+  last_value_ = samples;
+  sample_rate_hz_ = sample_rate_hz;
+  if (!last_logged_value_) {
+    last_logged_value_ = absl::optional<uint64_t>(samples);
+  }
+
+  if (!timer_->Finished()) {
+    // Not yet time to log.
+    return;
+  }
+
+  RTC_DCHECK(last_logged_value_);
+  RTC_DCHECK_GE(last_value_, *last_logged_value_);
+  const uint64_t diff = last_value_ - *last_logged_value_;
+  last_logged_value_ = absl::optional<uint64_t>(last_value_);
+  // Calculate rate in percent.
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  const int rate = (100 * diff) / (sample_rate_hz * logging_period_s_);
+  RTC_DCHECK_GE(rate, 0);
+  RTC_DCHECK_LE(rate, 100);
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE(uma_name_, rate);
+  timer_ = GetNewCountdown(tick_timer_, logging_period_s_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.h b/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.h
new file mode 100644
index 0000000000..cc5c20a886
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.h
@@ -0,0 +1,57 @@
+/*  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
+#define MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+
+namespace webrtc {
+
+// This class is used to periodically log values to a UMA histogram. The caller
+// is expected to update this class with an incremental sample counter which
+// counts expand samples. At the end of each logging period, the class will
+// calculate the fraction of samples that were expand samples during that period
+// and report that in percent. The logging period must be strictly positive.
+// Does not take ownership of tick_timer and the pointer must refer to a valid
+// object that outlives the one constructed.
+class ExpandUmaLogger {
+ public:
+  ExpandUmaLogger(absl::string_view uma_name,
+                  int logging_period_s,
+                  const TickTimer* tick_timer);
+
+  ~ExpandUmaLogger();
+
+  ExpandUmaLogger(const ExpandUmaLogger&) = delete;
+  ExpandUmaLogger& operator=(const ExpandUmaLogger&) = delete;
+
+  // In this call, value should be an incremental sample counter. The sample
+  // rate must be strictly positive.
+  void UpdateSampleCounter(uint64_t value, int sample_rate_hz);
+
+ private:
+  const std::string uma_name_;
+  const int logging_period_s_;
+  const TickTimer& tick_timer_;
+  std::unique_ptr<TickTimer::Countdown> timer_;
+  absl::optional<uint64_t> last_logged_value_;
+  uint64_t last_value_ = 0;
+  int sample_rate_hz_ = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/expand_unittest.cc
new file mode 100644
index 0000000000..9355fce5e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand_unittest.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for Expand class.
+
+#include "modules/audio_coding/neteq/expand.h"
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TEST(Expand, CreateAndDestroy) {
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
+}
+
+TEST(Expand, CreateUsingFactory) {
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  ExpandFactory expand_factory;
+  Expand* expand = expand_factory.Create(&bgn, &sync_buffer, &random_vector,
+                                         &statistics, fs, channels);
+  EXPECT_TRUE(expand != NULL);
+  delete expand;
+}
+
+namespace {
+class FakeStatisticsCalculator : public StatisticsCalculator {
+ public:
+  void LogDelayedPacketOutageEvent(int num_samples, int fs_hz) override {
+    last_outage_duration_samples_ = num_samples;
+  }
+
+  int last_outage_duration_samples() const {
+    return last_outage_duration_samples_;
+  }
+
+ private:
+  int last_outage_duration_samples_ = 0;
+};
+
+// This is the same size that is given to the SyncBuffer object in NetEq.
+const size_t kNetEqSyncBufferLengthMs = 720;
+}  // namespace
+
+class ExpandTest : public ::testing::Test {
+ protected:
+  ExpandTest()
+      : input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+                    32000),
+        test_sample_rate_hz_(32000),
+        num_channels_(1),
+        background_noise_(num_channels_),
+        sync_buffer_(num_channels_,
+                     kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
+        expand_(&background_noise_,
+                &sync_buffer_,
+                &random_vector_,
+                &statistics_,
+                test_sample_rate_hz_,
+                num_channels_) {
+    input_file_.set_output_rate_hz(test_sample_rate_hz_);
+  }
+
+  void SetUp() override {
+    // Fast-forward the input file until there is speech (about 1.1 second into
+    // the file).
+    const int speech_start_samples =
+        static_cast<int>(test_sample_rate_hz_ * 1.1f);
+    ASSERT_TRUE(input_file_.Seek(speech_start_samples));
+
+    // Pre-load the sync buffer with speech data.
+    std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
+    ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
+    sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
+    ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
+  }
+
+  test::ResampleInputAudioFile input_file_;
+  int test_sample_rate_hz_;
+  size_t num_channels_;
+  BackgroundNoise background_noise_;
+  SyncBuffer sync_buffer_;
+  RandomVector random_vector_;
+  FakeStatisticsCalculator statistics_;
+  Expand expand_;
+};
+
+// This test calls the expand object to produce concealment data a few times,
+// and then ends by calling SetParametersForNormalAfterExpand. This simulates
+// the situation where the packet next up for decoding was just delayed, not
+// lost.
+TEST_F(ExpandTest, DelayedPacketOutage) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    EXPECT_EQ(0, statistics_.last_outage_duration_samples());
+  }
+  expand_.SetParametersForNormalAfterExpand();
+  // Convert `sum_output_len_samples` to milliseconds.
+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
+            statistics_.last_outage_duration_samples());
+}
+
+// This test is similar to DelayedPacketOutage, but ends by calling
+// SetParametersForMergeAfterExpand. This simulates the situation where the
+// packet next up for decoding was actually lost (or at least a later packet
+// arrived before it).
+TEST_F(ExpandTest, LostPacketOutage) {
+  AudioMultiVector output(num_channels_);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    EXPECT_EQ(0, statistics_.last_outage_duration_samples());
+  }
+  expand_.SetParametersForMergeAfterExpand();
+  EXPECT_EQ(0, statistics_.last_outage_duration_samples());
+}
+
+// This test is similar to the DelayedPacketOutage test above, but with the
+// difference that Expand::Reset() is called after 5 calls to Expand::Process().
+// This should reset the statistics, and will in the end lead to an outage of
+// 5 periods instead of 10.
+TEST_F(ExpandTest, CheckOutageStatsAfterReset) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    if (i == 5) {
+      expand_.Reset();
+      sum_output_len_samples = 0;
+    }
+    EXPECT_EQ(0, statistics_.last_outage_duration_samples());
+  }
+  expand_.SetParametersForNormalAfterExpand();
+  // Convert `sum_output_len_samples` to milliseconds.
+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
+            statistics_.last_outage_duration_samples());
+}
+
+namespace {
+// Runs expand until Muted() returns true. Times out after 1000 calls.
+void ExpandUntilMuted(size_t num_channels, Expand* expand) {
+  EXPECT_FALSE(expand->Muted()) << "Instance is muted from the start";
+  AudioMultiVector output(num_channels);
+  int num_calls = 0;
+  while (!expand->Muted()) {
+    ASSERT_LT(num_calls++, 1000) << "Test timed out";
+    EXPECT_EQ(0, expand->Process(&output));
+  }
+}
+}  // namespace
+
+// Verifies that Muted() returns true after a long expand period. Also verifies
+// that Muted() is reset to false after calling Reset(),
+// SetParametersForMergeAfterExpand() and SetParametersForNormalAfterExpand().
+TEST_F(ExpandTest, Muted) {
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.Reset();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.SetParametersForMergeAfterExpand();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+
+  expand_.Reset();  // Must reset in order to start a new expand period.
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.SetParametersForNormalAfterExpand();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/g3doc/index.md b/third_party/libwebrtc/modules/audio_coding/neteq/g3doc/index.md
new file mode 100644
index 0000000000..40e76e2742
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/g3doc/index.md
@@ -0,0 +1,102 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'jakobi' reviewed: '2021-04-13'} *-->
+
+# NetEq
+
+NetEq is the audio jitter buffer and packet loss concealer. The jitter buffer is
+an adaptive jitter buffer, meaning that the buffering delay is continuously
+optimized based on the network conditions. Its main goal is to ensure a smooth
+playout of incoming audio packets from the network with a low amount of audio
+artifacts (alterations to the original content of the packets) while at the same
+time keep the delay as low as possible.
+
+## API
+
+At a high level, the NetEq API has two main functions:
+[`InsertPacket`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=198;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72)
+and
+[`GetAudio`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=219;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72).
+
+### InsertPacket
+
+[`InsertPacket`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=198;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72)
+delivers an RTP packet from the network to NetEq where the following happens:
+
+1.  The packet is discarded if it is too late for playout (for example if it was
+    reordered). Otherwize it is put into the packet buffer where it is stored
+    until it is time for playout. If the buffer is full, discard all the
+    existing packets (this should be rare).
+2.  The interarrival time between packets is analyzed and statistics is updated
+    which is used to derive a new target playout delay. The interarrival time is
+    measured in the number of GetAudio ‘ticks’ and thus clock drift between the
+    sender and receiver can be accounted for.
+
+### GetAudio
+
+[`GetAudio`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=219;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72)
+pulls 10 ms of audio from NetEq for playout. A much simplified decision logic is
+as follows:
+
+1.  If there is 10 ms audio in the sync buffer then return that.
+2.  If the next packet is available (based on RTP timestamp) in the packet
+    buffer then decode it and append the result to the sync buffer.
+    1.  Compare the current delay estimate (filtered buffer level) with the
+        target delay and time stretch (accelerate or decelerate) the contents of
+        the sync buffer if the buffer level is too high or too low.
+    2.  Return 10 ms of audio from the sync buffer.
+3.  If the last decoded packet was a discontinuous transmission (DTX) packet
+    then generate comfort noise.
+4.  If there is no available packet for decoding due to the next packet having
+    not arrived or been lost then generate packet loss concealment by
+    extrapolating the remaining audio in the sync buffer or by asking the
+    decoder to produce it.
+
+In summary, the output is the result one of the following operations:
+
+*   Normal: audio decoded from a packet.
+*   Acceleration: accelerated playout of a decoded packet.
+*   Preemptive expand: decelerated playout of a decoded packet.
+*   Expand: packet loss concealment generated by NetEq or the decoder.
+*   Merge: audio stitched together from packet loss concealment to decoded data
+    in case of a loss.
+*   Comfort noise (CNG): comfort noise generated by NetEq or the decoder between
+    talk spurts due to discontinuous transmission of packets (DTX).
+
+## Statistics
+
+There are a number of functions that can be used to query the internal state of
+NetEq, statistics about the type of audio output and latency metrics such as how
+long time packets have waited in the buffer.
+
+*   [`NetworkStatistics`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=273;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72):
+    instantaneous values or stats averaged over the duration since last call to
+    this function.
+*   [`GetLifetimeStatistics`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=280;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72):
+    cumulative stats that persist over the lifetime of the class.
+*   [`GetOperationsAndState`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=284;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72):
+    information about the internal state of NetEq (is only inteded to be used
+    for testing and debugging).
+
+## Tests and tools
+
+*   [`neteq_rtpplay`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc;drc=cee751abff598fc19506f77de08bea7c61b9dcca):
+    Simulate NetEq behavior based on either an RTP dump, a PCAP file or an RTC
+    event log. A replacement audio file can also be used instead of the original
+    payload. Outputs aggregated statistics and optionally an audio file to
+    listen to.
+*   [`neteq_speed_test`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc;drc=2ab97f6f8e27b47c0d9beeb8b6ca5387bda9f55c):
+    Measure performance of NetEq, used on perf bots.
+*   Unit tests including bit exactness tests where RTP file is used as an input
+    to NetEq, the output is concatenated and a checksum is calculated and
+    compared against a reference.
+
+## Other responsibilities
+
+*   Dual-tone multi-frequency signaling (DTMF): receive telephone events and
+    produce dual tone waveforms.
+*   Forward error correction (RED or codec inband FEC): split inserted packets
+    and prioritize the payloads.
+*   NACK (negative acknowledgement): keep track of lost packets and generate a
+    list of packets to NACK.
+*   Audio/video sync: NetEq can be instructed to increase the latency in order
+    to keep audio and video in sync.
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/histogram.cc b/third_party/libwebrtc/modules/audio_coding/neteq/histogram.cc
new file mode 100644
index 0000000000..e4b7f10379
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/histogram.cc
@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/histogram.h"
+
+#include <algorithm>
+#include <cstdlib>
+#include <numeric>
+
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+Histogram::Histogram(size_t num_buckets,
+                     int forget_factor,
+                     absl::optional<double> start_forget_weight)
+    : buckets_(num_buckets, 0),
+      forget_factor_(0),
+      base_forget_factor_(forget_factor),
+      add_count_(0),
+      start_forget_weight_(start_forget_weight) {
+  RTC_DCHECK_LT(base_forget_factor_, 1 << 15);
+}
+
+Histogram::~Histogram() {}
+
+// Each element in the vector is first multiplied by the forgetting factor
+// `forget_factor_`. Then the vector element indicated by `iat_packets` is then
+// increased (additive) by 1 - `forget_factor_`. This way, the probability of
+// `value` is slightly increased, while the sum of the histogram remains
+// constant (=1).
+// Due to inaccuracies in the fixed-point arithmetic, the histogram may no
+// longer sum up to 1 (in Q30) after the update. To correct this, a correction
+// term is added or subtracted from the first element (or elements) of the
+// vector.
+// The forgetting factor `forget_factor_` is also updated. When the DelayManager
+// is reset, the factor is set to 0 to facilitate rapid convergence in the
+// beginning. With each update of the histogram, the factor is increased towards
+// the steady-state value `base_forget_factor_`.
+void Histogram::Add(int value) {
+  RTC_DCHECK(value >= 0);
+  RTC_DCHECK(value < static_cast<int>(buckets_.size()));
+  int vector_sum = 0;  // Sum up the vector elements as they are processed.
+  // Multiply each element in `buckets_` with `forget_factor_`.
+  for (int& bucket : buckets_) {
+    bucket = (static_cast<int64_t>(bucket) * forget_factor_) >> 15;
+    vector_sum += bucket;
+  }
+
+  // Increase the probability for the currently observed inter-arrival time
+  // by 1 - `forget_factor_`. The factor is in Q15, `buckets_` in Q30.
+  // Thus, left-shift 15 steps to obtain result in Q30.
+  buckets_[value] += (32768 - forget_factor_) << 15;
+  vector_sum += (32768 - forget_factor_) << 15;  // Add to vector sum.
+
+  // `buckets_` should sum up to 1 (in Q30), but it may not due to
+  // fixed-point rounding errors.
+  vector_sum -= 1 << 30;  // Should be zero. Compensate if not.
+  if (vector_sum != 0) {
+    // Modify a few values early in `buckets_`.
+    int flip_sign = vector_sum > 0 ? -1 : 1;
+    for (int& bucket : buckets_) {
+      // Add/subtract 1/16 of the element, but not more than `vector_sum`.
+      int correction = flip_sign * std::min(std::abs(vector_sum), bucket >> 4);
+      bucket += correction;
+      vector_sum += correction;
+      if (std::abs(vector_sum) == 0) {
+        break;
+      }
+    }
+  }
+  RTC_DCHECK(vector_sum == 0);  // Verify that the above is correct.
+
+  ++add_count_;
+
+  // Update `forget_factor_` (changes only during the first seconds after a
+  // reset). The factor converges to `base_forget_factor_`.
+  if (start_forget_weight_) {
+    if (forget_factor_ != base_forget_factor_) {
+      int old_forget_factor = forget_factor_;
+      int forget_factor =
+          (1 << 15) * (1 - start_forget_weight_.value() / (add_count_ + 1));
+      forget_factor_ =
+          std::max(0, std::min(base_forget_factor_, forget_factor));
+      // The histogram is updated recursively by forgetting the old histogram
+      // with `forget_factor_` and adding a new sample multiplied by |1 -
+      // forget_factor_|. We need to make sure that the effective weight on the
+      // new sample is no smaller than those on the old samples, i.e., to
+      // satisfy the following DCHECK.
+      RTC_DCHECK_GE((1 << 15) - forget_factor_,
+                    ((1 << 15) - old_forget_factor) * forget_factor_ >> 15);
+    }
+  } else {
+    forget_factor_ += (base_forget_factor_ - forget_factor_ + 3) >> 2;
+  }
+}
+
+int Histogram::Quantile(int probability) {
+  // Find the bucket for which the probability of observing an
+  // inter-arrival time larger than or equal to `index` is larger than or
+  // equal to `probability`. The sought probability is estimated using
+  // the histogram as the reverse cumulant PDF, i.e., the sum of elements from
+  // the end up until `index`. Now, since the sum of all elements is 1
+  // (in Q30) by definition, and since the solution is often a low value for
+  // `iat_index`, it is more efficient to start with `sum` = 1 and subtract
+  // elements from the start of the histogram.
+  int inverse_probability = (1 << 30) - probability;
+  size_t index = 0;        // Start from the beginning of `buckets_`.
+  int sum = 1 << 30;       // Assign to 1 in Q30.
+  sum -= buckets_[index];
+
+  while ((sum > inverse_probability) && (index < buckets_.size() - 1)) {
+    // Subtract the probabilities one by one until the sum is no longer greater
+    // than `inverse_probability`.
+    ++index;
+    sum -= buckets_[index];
+  }
+  return static_cast<int>(index);
+}
+
+// Set the histogram vector to an exponentially decaying distribution
+// buckets_[i] = 0.5^(i+1), i = 0, 1, 2, ...
+// buckets_ is in Q30.
+void Histogram::Reset() {
+  // Set temp_prob to (slightly more than) 1 in Q14. This ensures that the sum
+  // of buckets_ is 1.
+  uint16_t temp_prob = 0x4002;  // 16384 + 2 = 100000000000010 binary.
+  for (int& bucket : buckets_) {
+    temp_prob >>= 1;
+    bucket = temp_prob << 16;
+  }
+  forget_factor_ = 0;  // Adapt the histogram faster for the first few packets.
+  add_count_ = 0;
+}
+
+int Histogram::NumBuckets() const {
+  return buckets_.size();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/histogram.h b/third_party/libwebrtc/modules/audio_coding/neteq/histogram.h
new file mode 100644
index 0000000000..265a10e00a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/histogram.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_HISTOGRAM_H_
+#define MODULES_AUDIO_CODING_NETEQ_HISTOGRAM_H_
+
+#include <string.h>  // Provide access to size_t.
+
+#include <vector>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+class Histogram {
+ public:
+  // Creates histogram with capacity `num_buckets` and `forget_factor` in Q15.
+  Histogram(size_t num_buckets,
+            int forget_factor,
+            absl::optional<double> start_forget_weight = absl::nullopt);
+
+  virtual ~Histogram();
+
+  // Resets the histogram to the default start distribution.
+  virtual void Reset();
+
+  // Add entry in bucket `index`.
+  virtual void Add(int index);
+
+  // Calculates the quantile at `probability` (in Q30) of the histogram
+  // distribution.
+  virtual int Quantile(int probability);
+
+  // Returns the number of buckets in the histogram.
+  virtual int NumBuckets() const;
+
+  // Returns the probability for each bucket in Q30.
+  const std::vector<int>& buckets() const { return buckets_; }
+
+  // Accessors only intended for testing purposes.
+  int base_forget_factor_for_testing() const { return base_forget_factor_; }
+  int forget_factor_for_testing() const { return forget_factor_; }
+  absl::optional<double> start_forget_weight_for_testing() const {
+    return start_forget_weight_;
+  }
+
+ private:
+  std::vector<int> buckets_;
+  int forget_factor_;  // Q15
+  const int base_forget_factor_;
+  int add_count_;
+  const absl::optional<double> start_forget_weight_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_HISTOGRAM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/histogram_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/histogram_unittest.cc
new file mode 100644
index 0000000000..e30a2956dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/histogram_unittest.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/histogram.h"
+
+#include <cmath>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(HistogramTest, Initialization) {
+  Histogram histogram(65, 32440);
+  histogram.Reset();
+  const auto& buckets = histogram.buckets();
+  double sum = 0.0;
+  for (size_t i = 0; i < buckets.size(); i++) {
+    EXPECT_NEAR(ldexp(std::pow(0.5, static_cast<int>(i + 1)), 30), buckets[i],
+                65537);
+    // Tolerance 65537 in Q30 corresponds to a delta of approximately 0.00006.
+    sum += buckets[i];
+  }
+  EXPECT_EQ(1 << 30, static_cast<int>(sum));  // Should be 1 in Q30.
+}
+
+TEST(HistogramTest, Add) {
+  Histogram histogram(10, 32440);
+  histogram.Reset();
+  const std::vector<int> before = histogram.buckets();
+  const int index = 5;
+  histogram.Add(index);
+  const std::vector<int> after = histogram.buckets();
+  EXPECT_GT(after[index], before[index]);
+  int sum = 0;
+  for (int bucket : after) {
+    sum += bucket;
+  }
+  EXPECT_EQ(1 << 30, sum);
+}
+
+TEST(HistogramTest, ForgetFactor) {
+  Histogram histogram(10, 32440);
+  histogram.Reset();
+  const std::vector<int> before = histogram.buckets();
+  const int index = 4;
+  histogram.Add(index);
+  const std::vector<int> after = histogram.buckets();
+  for (int i = 0; i < histogram.NumBuckets(); ++i) {
+    if (i != index) {
+      EXPECT_LT(after[i], before[i]);
+    }
+  }
+}
+
+TEST(HistogramTest, ReachSteadyStateForgetFactor) {
+  static constexpr int kSteadyStateForgetFactor = (1 << 15) * 0.9993;
+  Histogram histogram(100, kSteadyStateForgetFactor, 1.0);
+  histogram.Reset();
+  int n = (1 << 15) / ((1 << 15) - kSteadyStateForgetFactor);
+  for (int i = 0; i < n; ++i) {
+    histogram.Add(0);
+  }
+  EXPECT_EQ(histogram.forget_factor_for_testing(), kSteadyStateForgetFactor);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/merge.cc b/third_party/libwebrtc/modules/audio_coding/neteq/merge.cc
new file mode 100644
index 0000000000..0aec6d2597
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/merge.cc
@@ -0,0 +1,391 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/merge.h"
+
+#include <string.h>  // memmove, memcpy, memset, size_t
+
+#include <algorithm>  // min, max
+#include <memory>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+Merge::Merge(int fs_hz,
+             size_t num_channels,
+             Expand* expand,
+             SyncBuffer* sync_buffer)
+    : fs_hz_(fs_hz),
+      num_channels_(num_channels),
+      fs_mult_(fs_hz_ / 8000),
+      timestamps_per_call_(static_cast<size_t>(fs_hz_ / 100)),
+      expand_(expand),
+      sync_buffer_(sync_buffer),
+      expanded_(num_channels_) {
+  RTC_DCHECK_GT(num_channels_, 0);
+}
+
+Merge::~Merge() = default;
+
+size_t Merge::Process(int16_t* input,
+                      size_t input_length,
+                      AudioMultiVector* output) {
+  // TODO(hlundin): Change to an enumerator and skip assert.
+  RTC_DCHECK(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
+             fs_hz_ == 48000);
+  RTC_DCHECK_LE(fs_hz_, kMaxSampleRate);  // Should not be possible.
+  if (input_length == 0) {
+    return 0;
+  }
+
+  size_t old_length;
+  size_t expand_period;
+  // Get expansion data to overlap and mix with.
+  size_t expanded_length = GetExpandedSignal(&old_length, &expand_period);
+
+  // Transfer input signal to an AudioMultiVector.
+  AudioMultiVector input_vector(num_channels_);
+  input_vector.PushBackInterleaved(
+      rtc::ArrayView<const int16_t>(input, input_length));
+  size_t input_length_per_channel = input_vector.Size();
+  RTC_DCHECK_EQ(input_length_per_channel, input_length / num_channels_);
+
+  size_t best_correlation_index = 0;
+  size_t output_length = 0;
+
+  std::unique_ptr<int16_t[]> input_channel(
+      new int16_t[input_length_per_channel]);
+  std::unique_ptr<int16_t[]> expanded_channel(new int16_t[expanded_length]);
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    input_vector[channel].CopyTo(input_length_per_channel, 0,
+                                 input_channel.get());
+    expanded_[channel].CopyTo(expanded_length, 0, expanded_channel.get());
+
+    const int16_t new_mute_factor = std::min<int16_t>(
+        16384, SignalScaling(input_channel.get(), input_length_per_channel,
+                             expanded_channel.get()));
+
+    if (channel == 0) {
+      // Downsample, correlate, and find strongest correlation period for the
+      // reference (i.e., first) channel only.
+      // Downsample to 4kHz sample rate.
+      Downsample(input_channel.get(), input_length_per_channel,
+                 expanded_channel.get(), expanded_length);
+
+      // Calculate the lag of the strongest correlation period.
+      best_correlation_index = CorrelateAndPeakSearch(
+          old_length, input_length_per_channel, expand_period);
+    }
+
+    temp_data_.resize(input_length_per_channel + best_correlation_index);
+    int16_t* decoded_output = temp_data_.data() + best_correlation_index;
+
+    // Mute the new decoded data if needed (and unmute it linearly).
+    // This is the overlapping part of expanded_signal.
+    size_t interpolation_length =
+        std::min(kMaxCorrelationLength * fs_mult_,
+                 expanded_length - best_correlation_index);
+    interpolation_length =
+        std::min(interpolation_length, input_length_per_channel);
+
+    RTC_DCHECK_LE(new_mute_factor, 16384);
+    int16_t mute_factor =
+        std::max(expand_->MuteFactor(channel), new_mute_factor);
+    RTC_DCHECK_GE(mute_factor, 0);
+
+    if (mute_factor < 16384) {
+      // Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
+      // and so on, or as fast as it takes to come back to full gain within the
+      // frame length.
+      const int back_to_fullscale_inc = static_cast<int>(
+          ((16384 - mute_factor) << 6) / input_length_per_channel);
+      const int increment = std::max(4194 / fs_mult_, back_to_fullscale_inc);
+      mute_factor = static_cast<int16_t>(DspHelper::RampSignal(
+          input_channel.get(), interpolation_length, mute_factor, increment));
+      DspHelper::UnmuteSignal(&input_channel[interpolation_length],
+                              input_length_per_channel - interpolation_length,
+                              &mute_factor, increment,
+                              &decoded_output[interpolation_length]);
+    } else {
+      // No muting needed.
+      memmove(
+          &decoded_output[interpolation_length],
+          &input_channel[interpolation_length],
+          sizeof(int16_t) * (input_length_per_channel - interpolation_length));
+    }
+
+    // Do overlap and mix linearly.
+    int16_t increment =
+        static_cast<int16_t>(16384 / (interpolation_length + 1));  // In Q14.
+    int16_t local_mute_factor = 16384 - increment;
+    memmove(temp_data_.data(), expanded_channel.get(),
+            sizeof(int16_t) * best_correlation_index);
+    DspHelper::CrossFade(&expanded_channel[best_correlation_index],
+                         input_channel.get(), interpolation_length,
+                         &local_mute_factor, increment, decoded_output);
+
+    output_length = best_correlation_index + input_length_per_channel;
+    if (channel == 0) {
+      RTC_DCHECK(output->Empty());  // Output should be empty at this point.
+      output->AssertSize(output_length);
+    } else {
+      RTC_DCHECK_EQ(output->Size(), output_length);
+    }
+    (*output)[channel].OverwriteAt(temp_data_.data(), output_length, 0);
+  }
+
+  // Copy back the first part of the data to `sync_buffer_` and remove it from
+  // `output`.
+  sync_buffer_->ReplaceAtIndex(*output, old_length, sync_buffer_->next_index());
+  output->PopFront(old_length);
+
+  // Return new added length. `old_length` samples were borrowed from
+  // `sync_buffer_`.
+  RTC_DCHECK_GE(output_length, old_length);
+  return output_length - old_length;
+}
+
+size_t Merge::GetExpandedSignal(size_t* old_length, size_t* expand_period) {
+  // Check how much data that is left since earlier.
+  *old_length = sync_buffer_->FutureLength();
+  // Should never be less than overlap_length.
+  RTC_DCHECK_GE(*old_length, expand_->overlap_length());
+  // Generate data to merge the overlap with using expand.
+  expand_->SetParametersForMergeAfterExpand();
+
+  if (*old_length >= 210 * kMaxSampleRate / 8000) {
+    // TODO(hlundin): Write test case for this.
+    // The number of samples available in the sync buffer is more than what fits
+    // in expanded_signal. Keep the first 210 * kMaxSampleRate / 8000 samples,
+    // but shift them towards the end of the buffer. This is ok, since all of
+    // the buffer will be expand data anyway, so as long as the beginning is
+    // left untouched, we're fine.
+    size_t length_diff = *old_length - 210 * kMaxSampleRate / 8000;
+    sync_buffer_->InsertZerosAtIndex(length_diff, sync_buffer_->next_index());
+    *old_length = 210 * kMaxSampleRate / 8000;
+    // This is the truncated length.
+  }
+  // This assert should always be true thanks to the if statement above.
+  RTC_DCHECK_GE(210 * kMaxSampleRate / 8000, *old_length);
+
+  AudioMultiVector expanded_temp(num_channels_);
+  expand_->Process(&expanded_temp);
+  *expand_period = expanded_temp.Size();  // Samples per channel.
+
+  expanded_.Clear();
+  // Copy what is left since earlier into the expanded vector.
+  expanded_.PushBackFromIndex(*sync_buffer_, sync_buffer_->next_index());
+  RTC_DCHECK_EQ(expanded_.Size(), *old_length);
+  RTC_DCHECK_GT(expanded_temp.Size(), 0);
+  // Do "ugly" copy and paste from the expanded in order to generate more data
+  // to correlate (but not interpolate) with.
+  const size_t required_length = static_cast<size_t>((120 + 80 + 2) * fs_mult_);
+  if (expanded_.Size() < required_length) {
+    while (expanded_.Size() < required_length) {
+      // Append one more pitch period each time.
+      expanded_.PushBack(expanded_temp);
+    }
+    // Trim the length to exactly `required_length`.
+    expanded_.PopBack(expanded_.Size() - required_length);
+  }
+  RTC_DCHECK_GE(expanded_.Size(), required_length);
+  return required_length;
+}
+
+int16_t Merge::SignalScaling(const int16_t* input,
+                             size_t input_length,
+                             const int16_t* expanded_signal) const {
+  // Adjust muting factor if new vector is more or less of the BGN energy.
+  const auto mod_input_length = rtc::SafeMin<size_t>(
+      64 * rtc::dchecked_cast<size_t>(fs_mult_), input_length);
+
+  // Missing input, do no muting
+  if (mod_input_length == 0) {
+    return 16384;
+  }
+
+  const int16_t expanded_max =
+      WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
+  int32_t factor =
+      (expanded_max * expanded_max) / (std::numeric_limits<int32_t>::max() /
+                                       static_cast<int32_t>(mod_input_length));
+  const int expanded_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+  int32_t energy_expanded = WebRtcSpl_DotProductWithScale(
+      expanded_signal, expanded_signal, mod_input_length, expanded_shift);
+
+  // Calculate energy of input signal.
+  const int16_t input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
+  factor = (input_max * input_max) / (std::numeric_limits<int32_t>::max() /
+                                      static_cast<int32_t>(mod_input_length));
+  const int input_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+  int32_t energy_input = WebRtcSpl_DotProductWithScale(
+      input, input, mod_input_length, input_shift);
+
+  // Align to the same Q-domain.
+  if (input_shift > expanded_shift) {
+    energy_expanded = energy_expanded >> (input_shift - expanded_shift);
+  } else {
+    energy_input = energy_input >> (expanded_shift - input_shift);
+  }
+
+  // Calculate muting factor to use for new frame.
+  int16_t mute_factor;
+  if (energy_input > energy_expanded) {
+    // Normalize `energy_input` to 14 bits.
+    int16_t temp_shift = WebRtcSpl_NormW32(energy_input) - 17;
+    energy_input = WEBRTC_SPL_SHIFT_W32(energy_input, temp_shift);
+    // Put `energy_expanded` in a domain 14 higher, so that
+    // energy_expanded / energy_input is in Q14.
+    energy_expanded = WEBRTC_SPL_SHIFT_W32(energy_expanded, temp_shift + 14);
+    // Calculate sqrt(energy_expanded / energy_input) in Q14.
+    mute_factor = static_cast<int16_t>(
+        WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14));
+  } else {
+    // Set to 1 (in Q14) when `expanded` has higher energy than `input`.
+    mute_factor = 16384;
+  }
+
+  return mute_factor;
+}
+
+// TODO(hlundin): There are some parameter values in this method that seem
+// strange. Compare with Expand::Correlation.
+void Merge::Downsample(const int16_t* input,
+                       size_t input_length,
+                       const int16_t* expanded_signal,
+                       size_t expanded_length) {
+  const int16_t* filter_coefficients;
+  size_t num_coefficients;
+  int decimation_factor = fs_hz_ / 4000;
+  static const size_t kCompensateDelay = 0;
+  size_t length_limit = static_cast<size_t>(fs_hz_ / 100);  // 10 ms in samples.
+  if (fs_hz_ == 8000) {
+    filter_coefficients = DspHelper::kDownsample8kHzTbl;
+    num_coefficients = 3;
+  } else if (fs_hz_ == 16000) {
+    filter_coefficients = DspHelper::kDownsample16kHzTbl;
+    num_coefficients = 5;
+  } else if (fs_hz_ == 32000) {
+    filter_coefficients = DspHelper::kDownsample32kHzTbl;
+    num_coefficients = 7;
+  } else {  // fs_hz_ == 48000
+    filter_coefficients = DspHelper::kDownsample48kHzTbl;
+    num_coefficients = 7;
+  }
+  size_t signal_offset = num_coefficients - 1;
+  WebRtcSpl_DownsampleFast(
+      &expanded_signal[signal_offset], expanded_length - signal_offset,
+      expanded_downsampled_, kExpandDownsampLength, filter_coefficients,
+      num_coefficients, decimation_factor, kCompensateDelay);
+  if (input_length <= length_limit) {
+    // Not quite long enough, so we have to cheat a bit.
+    // If the input is shorter than the offset, we consider the input to be 0
+    // length. This will cause us to skip the downsampling since it makes no
+    // sense anyway, and input_downsampled_ will be filled with zeros. This is
+    // clearly a pathological case, and the signal quality will suffer, but
+    // there is not much we can do.
+    const size_t temp_len =
+        input_length > signal_offset ? input_length - signal_offset : 0;
+    // TODO(hlundin): Should `downsamp_temp_len` be corrected for round-off
+    // errors? I.e., (temp_len + decimation_factor - 1) / decimation_factor?
+    size_t downsamp_temp_len = temp_len / decimation_factor;
+    if (downsamp_temp_len > 0) {
+      WebRtcSpl_DownsampleFast(&input[signal_offset], temp_len,
+                               input_downsampled_, downsamp_temp_len,
+                               filter_coefficients, num_coefficients,
+                               decimation_factor, kCompensateDelay);
+    }
+    memset(&input_downsampled_[downsamp_temp_len], 0,
+           sizeof(int16_t) * (kInputDownsampLength - downsamp_temp_len));
+  } else {
+    WebRtcSpl_DownsampleFast(
+        &input[signal_offset], input_length - signal_offset, input_downsampled_,
+        kInputDownsampLength, filter_coefficients, num_coefficients,
+        decimation_factor, kCompensateDelay);
+  }
+}
+
+size_t Merge::CorrelateAndPeakSearch(size_t start_position,
+                                     size_t input_length,
+                                     size_t expand_period) const {
+  // Calculate correlation without any normalization.
+  const size_t max_corr_length = kMaxCorrelationLength;
+  size_t stop_position_downsamp =
+      std::min(max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
+
+  int32_t correlation[kMaxCorrelationLength];
+  CrossCorrelationWithAutoShift(input_downsampled_, expanded_downsampled_,
+                                kInputDownsampLength, stop_position_downsamp, 1,
+                                correlation);
+
+  // Normalize correlation to 14 bits and copy to a 16-bit array.
+  const size_t pad_length = expand_->overlap_length() - 1;
+  const size_t correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength;
+  std::unique_ptr<int16_t[]> correlation16(
+      new int16_t[correlation_buffer_size]);
+  memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t));
+  int16_t* correlation_ptr = &correlation16[pad_length];
+  int32_t max_correlation =
+      WebRtcSpl_MaxAbsValueW32(correlation, stop_position_downsamp);
+  int norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
+  WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
+                                   correlation, norm_shift);
+
+  // Calculate allowed starting point for peak finding.
+  // The peak location bestIndex must fulfill two criteria:
+  // (1) w16_bestIndex + input_length <
+  //     timestamps_per_call_ + expand_->overlap_length();
+  // (2) w16_bestIndex + input_length < start_position.
+  size_t start_index = timestamps_per_call_ + expand_->overlap_length();
+  start_index = std::max(start_position, start_index);
+  start_index = (input_length > start_index) ? 0 : (start_index - input_length);
+  // Downscale starting index to 4kHz domain. (fs_mult_ * 2 = fs_hz_ / 4000.)
+  size_t start_index_downsamp = start_index / (fs_mult_ * 2);
+
+  // Calculate a modified `stop_position_downsamp` to account for the increased
+  // start index `start_index_downsamp` and the effective array length.
+  size_t modified_stop_pos =
+      std::min(stop_position_downsamp,
+               kMaxCorrelationLength + pad_length - start_index_downsamp);
+  size_t best_correlation_index;
+  int16_t best_correlation;
+  static const size_t kNumCorrelationCandidates = 1;
+  DspHelper::PeakDetection(&correlation_ptr[start_index_downsamp],
+                           modified_stop_pos, kNumCorrelationCandidates,
+                           fs_mult_, &best_correlation_index,
+                           &best_correlation);
+  // Compensate for modified start index.
+  best_correlation_index += start_index;
+
+  // Ensure that underrun does not occur for 10ms case => we have to get at
+  // least 10ms + overlap . (This should never happen thanks to the above
+  // modification of peak-finding starting point.)
+  while (((best_correlation_index + input_length) <
+          (timestamps_per_call_ + expand_->overlap_length())) ||
+         ((best_correlation_index + input_length) < start_position)) {
+    RTC_DCHECK_NOTREACHED();                  // Should never happen.
+    best_correlation_index += expand_period;  // Jump one lag ahead.
+  }
+  return best_correlation_index;
+}
+
+size_t Merge::RequiredFutureSamples() {
+  return fs_hz_ / 100 * num_channels_;  // 10 ms.
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/merge.h b/third_party/libwebrtc/modules/audio_coding/neteq/merge.h
new file mode 100644
index 0000000000..2f27106bfe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/merge.h
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MERGE_H_
+#define MODULES_AUDIO_CODING_NETEQ_MERGE_H_
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class Expand;
+class SyncBuffer;
+
+// This class handles the transition from expansion to normal operation.
+// When a packet is not available for decoding when needed, the expand operation
+// is called to generate extrapolation data. If the missing packet arrives,
+// i.e., it was just delayed, it can be decoded and appended directly to the
+// end of the expanded data (thanks to how the Expand class operates). However,
+// if a later packet arrives instead, the loss is a fact, and the new data must
+// be stitched together with the end of the expanded data. This stitching is
+// what the Merge class does.
+class Merge {
+ public:
+  Merge(int fs_hz,
+        size_t num_channels,
+        Expand* expand,
+        SyncBuffer* sync_buffer);
+  virtual ~Merge();
+
+  Merge(const Merge&) = delete;
+  Merge& operator=(const Merge&) = delete;
+
+  // The main method to produce the audio data. The decoded data is supplied in
+  // `input`, having `input_length` samples in total for all channels
+  // (interleaved). The result is written to `output`. The number of channels
+  // allocated in `output` defines the number of channels that will be used when
+  // de-interleaving `input`.
+  virtual size_t Process(int16_t* input,
+                         size_t input_length,
+                         AudioMultiVector* output);
+
+  virtual size_t RequiredFutureSamples();
+
+ protected:
+  const int fs_hz_;
+  const size_t num_channels_;
+
+ private:
+  static const int kMaxSampleRate = 48000;
+  static const size_t kExpandDownsampLength = 100;
+  static const size_t kInputDownsampLength = 40;
+  static const size_t kMaxCorrelationLength = 60;
+
+  // Calls `expand_` to get more expansion data to merge with. The data is
+  // written to `expanded_signal_`. Returns the length of the expanded data,
+  // while `expand_period` will be the number of samples in one expansion period
+  // (typically one pitch period). The value of `old_length` will be the number
+  // of samples that were taken from the `sync_buffer_`.
+  size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
+
+  // Analyzes `input` and `expanded_signal` and returns muting factor (Q14) to
+  // be used on the new data.
+  int16_t SignalScaling(const int16_t* input,
+                        size_t input_length,
+                        const int16_t* expanded_signal) const;
+
+  // Downsamples `input` (`input_length` samples) and `expanded_signal` to
+  // 4 kHz sample rate. The downsampled signals are written to
+  // `input_downsampled_` and `expanded_downsampled_`, respectively.
+  void Downsample(const int16_t* input,
+                  size_t input_length,
+                  const int16_t* expanded_signal,
+                  size_t expanded_length);
+
+  // Calculates cross-correlation between `input_downsampled_` and
+  // `expanded_downsampled_`, and finds the correlation maximum. The maximizing
+  // lag is returned.
+  size_t CorrelateAndPeakSearch(size_t start_position,
+                                size_t input_length,
+                                size_t expand_period) const;
+
+  const int fs_mult_;  // fs_hz_ / 8000.
+  const size_t timestamps_per_call_;
+  Expand* expand_;
+  SyncBuffer* sync_buffer_;
+  int16_t expanded_downsampled_[kExpandDownsampLength];
+  int16_t input_downsampled_[kInputDownsampLength];
+  AudioMultiVector expanded_;
+  std::vector<int16_t> temp_data_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MERGE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/merge_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/merge_unittest.cc
new file mode 100644
index 0000000000..d5a55eb056
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/merge_unittest.cc
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for Merge class.
+
+#include "modules/audio_coding/neteq/merge.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TEST(Merge, CreateAndDestroy) {
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
+  Merge merge(fs, channels, &expand, &sync_buffer);
+}
+
+namespace {
+// This is the same size that is given to the SyncBuffer object in NetEq.
+const size_t kNetEqSyncBufferLengthMs = 720;
+}  // namespace
+
+class MergeTest : public testing::TestWithParam<size_t> {
+ protected:
+  MergeTest()
+      : input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+                    32000),
+        test_sample_rate_hz_(8000),
+        num_channels_(1),
+        background_noise_(num_channels_),
+        sync_buffer_(num_channels_,
+                     kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
+        expand_(&background_noise_,
+                &sync_buffer_,
+                &random_vector_,
+                &statistics_,
+                test_sample_rate_hz_,
+                num_channels_),
+        merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) {
+    input_file_.set_output_rate_hz(test_sample_rate_hz_);
+  }
+
+  void SetUp() override {
+    // Fast-forward the input file until there is speech (about 1.1 second into
+    // the file).
+    const int speech_start_samples =
+        static_cast<int>(test_sample_rate_hz_ * 1.1f);
+    ASSERT_TRUE(input_file_.Seek(speech_start_samples));
+
+    // Pre-load the sync buffer with speech data.
+    std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
+    ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
+    sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
+    // Move index such that the sync buffer appears to have 5 ms left to play.
+    sync_buffer_.set_next_index(sync_buffer_.next_index() -
+                                test_sample_rate_hz_ * 5 / 1000);
+    ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
+    ASSERT_GT(sync_buffer_.FutureLength(), 0u);
+  }
+
+  test::ResampleInputAudioFile input_file_;
+  int test_sample_rate_hz_;
+  size_t num_channels_;
+  BackgroundNoise background_noise_;
+  SyncBuffer sync_buffer_;
+  RandomVector random_vector_;
+  StatisticsCalculator statistics_;
+  Expand expand_;
+  Merge merge_;
+};
+
+TEST_P(MergeTest, Process) {
+  AudioMultiVector output(num_channels_);
+  // Start by calling Expand once, to prime the state.
+  EXPECT_EQ(0, expand_.Process(&output));
+  EXPECT_GT(output.Size(), 0u);
+  output.Clear();
+  // Now call Merge, but with a very short decoded input. Try different length
+  // if the input.
+  const size_t input_len = GetParam();
+  std::vector<int16_t> input(input_len, 17);
+  merge_.Process(input.data(), input_len, &output);
+  EXPECT_GT(output.Size(), 0u);
+}
+
+// Instantiate with values for the input length that are interesting in
+// Merge::Downsample. Why are these values interesting?
+// - In 8000 Hz sample rate, signal_offset in Merge::Downsample will be 2, so
+//   the values 1, 2, 3 are just around that value.
+// - Also in 8000 Hz, the variable length_limit in the same method will be 80,
+//   so values 80 and 81 will be on either side of the branch point
+//   "input_length <= length_limit".
+// - Finally, 160 is simply 20 ms in 8000 Hz, which is a common packet size.
+INSTANTIATE_TEST_SUITE_P(DifferentInputLengths,
+                         MergeTest,
+                         testing::Values(1, 2, 3, 80, 81, 160));
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
new file mode 100644
index 0000000000..503f6ac6bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockBufferLevelFilter : public BufferLevelFilter {
+ public:
+  MOCK_METHOD(void,
+              Update,
+              (size_t buffer_size_samples, int time_stretched_samples));
+  MOCK_METHOD(int, filtered_current_level, (), (const));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
new file mode 100644
index 0000000000..2394120e99
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
+
+#include <string>
+
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDecoderDatabase : public DecoderDatabase {
+ public:
+  explicit MockDecoderDatabase(
+      rtc::scoped_refptr<AudioDecoderFactory> factory = nullptr)
+      : DecoderDatabase(factory, absl::nullopt) {}
+  ~MockDecoderDatabase() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(bool, Empty, (), (const, override));
+  MOCK_METHOD(int, Size, (), (const, override));
+  MOCK_METHOD(int,
+              RegisterPayload,
+              (int rtp_payload_type, const SdpAudioFormat& audio_format),
+              (override));
+  MOCK_METHOD(int, Remove, (uint8_t rtp_payload_type), (override));
+  MOCK_METHOD(void, RemoveAll, (), (override));
+  MOCK_METHOD(const DecoderInfo*,
+              GetDecoderInfo,
+              (uint8_t rtp_payload_type),
+              (const, override));
+  MOCK_METHOD(int,
+              SetActiveDecoder,
+              (uint8_t rtp_payload_type, bool* new_decoder),
+              (override));
+  MOCK_METHOD(AudioDecoder*, GetActiveDecoder, (), (const, override));
+  MOCK_METHOD(int, SetActiveCngDecoder, (uint8_t rtp_payload_type), (override));
+  MOCK_METHOD(ComfortNoiseDecoder*, GetActiveCngDecoder, (), (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
new file mode 100644
index 0000000000..d783f8743b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDelayManager : public DelayManager {
+ public:
+  MockDelayManager(const MockDelayManager::Config& config,
+                   const TickTimer* tick_timer)
+      : DelayManager(config, tick_timer) {}
+  MOCK_METHOD(int, TargetDelayMs, (), (const));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
new file mode 100644
index 0000000000..c60c56d36b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
+
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDtmfBuffer : public DtmfBuffer {
+ public:
+  MockDtmfBuffer(int fs) : DtmfBuffer(fs) {}
+  ~MockDtmfBuffer() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(void, Flush, (), (override));
+  MOCK_METHOD(int, InsertEvent, (const DtmfEvent& event), (override));
+  MOCK_METHOD(bool,
+              GetEvent,
+              (uint32_t current_timestamp, DtmfEvent* event),
+              (override));
+  MOCK_METHOD(size_t, Length, (), (const, override));
+  MOCK_METHOD(bool, Empty, (), (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
new file mode 100644
index 0000000000..60de167c29
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
+
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDtmfToneGenerator : public DtmfToneGenerator {
+ public:
+  ~MockDtmfToneGenerator() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(int, Init, (int fs, int event, int attenuation), (override));
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(int,
+              Generate,
+              (size_t num_samples, AudioMultiVector* output),
+              (override));
+  MOCK_METHOD(bool, initialized, (), (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_expand.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_expand.h
new file mode 100644
index 0000000000..9d66779021
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_expand.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
+
+#include "modules/audio_coding/neteq/expand.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockExpand : public Expand {
+ public:
+  MockExpand(BackgroundNoise* background_noise,
+             SyncBuffer* sync_buffer,
+             RandomVector* random_vector,
+             StatisticsCalculator* statistics,
+             int fs,
+             size_t num_channels)
+      : Expand(background_noise,
+               sync_buffer,
+               random_vector,
+               statistics,
+               fs,
+               num_channels) {}
+  ~MockExpand() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(int, Process, (AudioMultiVector * output), (override));
+  MOCK_METHOD(void, SetParametersForNormalAfterExpand, (), (override));
+  MOCK_METHOD(void, SetParametersForMergeAfterExpand, (), (override));
+  MOCK_METHOD(size_t, overlap_length, (), (const, override));
+};
+
+}  // namespace webrtc
+
+namespace webrtc {
+
+class MockExpandFactory : public ExpandFactory {
+ public:
+  MOCK_METHOD(Expand*,
+              Create,
+              (BackgroundNoise * background_noise,
+               SyncBuffer* sync_buffer,
+               RandomVector* random_vector,
+               StatisticsCalculator* statistics,
+               int fs,
+               size_t num_channels),
+              (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_histogram.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_histogram.h
new file mode 100644
index 0000000000..03abbc1d4b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_histogram.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_HISTOGRAM_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_HISTOGRAM_H_
+
+#include "modules/audio_coding/neteq/histogram.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockHistogram : public Histogram {
+ public:
+  MockHistogram(size_t num_buckets, int forget_factor)
+      : Histogram(num_buckets, forget_factor) {}
+  virtual ~MockHistogram() {}
+
+  MOCK_METHOD(void, Add, (int), (override));
+  MOCK_METHOD(int, Quantile, (int), (override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_HISTOGRAM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_neteq_controller.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_neteq_controller.h
new file mode 100644
index 0000000000..6d88e09216
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_neteq_controller.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_
+
+#include "api/neteq/neteq_controller.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockNetEqController : public NetEqController {
+ public:
+  MockNetEqController() = default;
+  ~MockNetEqController() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(void, SoftReset, (), (override));
+  MOCK_METHOD(NetEq::Operation,
+              GetDecision,
+              (const NetEqStatus& neteq_status, bool* reset_decoder),
+              (override));
+  MOCK_METHOD(void, RegisterEmptyPacket, (), (override));
+  MOCK_METHOD(void,
+              SetSampleRate,
+              (int fs_hz, size_t output_size_samples),
+              (override));
+  MOCK_METHOD(bool, SetMaximumDelay, (int delay_ms), (override));
+  MOCK_METHOD(bool, SetMinimumDelay, (int delay_ms), (override));
+  MOCK_METHOD(bool, SetBaseMinimumDelay, (int delay_ms), (override));
+  MOCK_METHOD(int, GetBaseMinimumDelay, (), (const, override));
+  MOCK_METHOD(bool, CngRfc3389On, (), (const, override));
+  MOCK_METHOD(bool, CngOff, (), (const, override));
+  MOCK_METHOD(void, SetCngOff, (), (override));
+  MOCK_METHOD(void, ExpandDecision, (NetEq::Operation operation), (override));
+  MOCK_METHOD(void, AddSampleMemory, (int32_t value), (override));
+  MOCK_METHOD(int, TargetLevelMs, (), (const, override));
+  MOCK_METHOD(absl::optional<int>,
+              PacketArrived,
+              (int fs_hz,
+               bool should_update_stats,
+               const PacketArrivedInfo& info),
+              (override));
+  MOCK_METHOD(void, NotifyMutedState, (), (override));
+  MOCK_METHOD(bool, PeakFound, (), (const, override));
+  MOCK_METHOD(int, GetFilteredBufferLevel, (), (const, override));
+  MOCK_METHOD(void, set_sample_memory, (int32_t value), (override));
+  MOCK_METHOD(size_t, noise_fast_forward, (), (const, override));
+  MOCK_METHOD(size_t, packet_length_samples, (), (const, override));
+  MOCK_METHOD(void, set_packet_length_samples, (size_t value), (override));
+  MOCK_METHOD(void, set_prev_time_scale, (bool value), (override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
new file mode 100644
index 0000000000..48357ea466
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockPacketBuffer : public PacketBuffer {
+ public:
+  MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
+      : PacketBuffer(max_number_of_packets, tick_timer) {}
+  ~MockPacketBuffer() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(void, Flush, (StatisticsCalculator * stats), (override));
+  MOCK_METHOD(void,
+              PartialFlush,
+              (int target_level_ms,
+               size_t sample_rate,
+               size_t last_decoded_length,
+               StatisticsCalculator* stats),
+              (override));
+  MOCK_METHOD(bool, Empty, (), (const, override));
+  MOCK_METHOD(int,
+              InsertPacket,
+              (Packet && packet,
+               StatisticsCalculator* stats,
+               size_t last_decoded_length,
+               size_t sample_rate,
+               int target_level_ms,
+               const DecoderDatabase& decoder_database),
+              (override));
+  MOCK_METHOD(int,
+              InsertPacketList,
+              (PacketList * packet_list,
+               const DecoderDatabase& decoder_database,
+               absl::optional<uint8_t>* current_rtp_payload_type,
+               absl::optional<uint8_t>* current_cng_rtp_payload_type,
+               StatisticsCalculator* stats,
+               size_t last_decoded_length,
+               size_t sample_rate,
+               int target_level_ms),
+              (override));
+  MOCK_METHOD(int,
+              NextTimestamp,
+              (uint32_t * next_timestamp),
+              (const, override));
+  MOCK_METHOD(int,
+              NextHigherTimestamp,
+              (uint32_t timestamp, uint32_t* next_timestamp),
+              (const, override));
+  MOCK_METHOD(const Packet*, PeekNextPacket, (), (const, override));
+  MOCK_METHOD(absl::optional<Packet>, GetNextPacket, (), (override));
+  MOCK_METHOD(int,
+              DiscardNextPacket,
+              (StatisticsCalculator * stats),
+              (override));
+  MOCK_METHOD(void,
+              DiscardOldPackets,
+              (uint32_t timestamp_limit,
+               uint32_t horizon_samples,
+               StatisticsCalculator* stats),
+              (override));
+  MOCK_METHOD(void,
+              DiscardAllOldPackets,
+              (uint32_t timestamp_limit, StatisticsCalculator* stats),
+              (override));
+  MOCK_METHOD(size_t, NumPacketsInBuffer, (), (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
new file mode 100644
index 0000000000..9daf571a80
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
+
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRedPayloadSplitter : public RedPayloadSplitter {
+ public:
+  MOCK_METHOD(bool, SplitRed, (PacketList * packet_list), (override));
+  MOCK_METHOD(void,
+              CheckRedPayloads,
+              (PacketList * packet_list,
+               const DecoderDatabase& decoder_database),
+              (override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_statistics_calculator.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
new file mode 100644
index 0000000000..f8812478d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
+
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockStatisticsCalculator : public StatisticsCalculator {
+ public:
+  MOCK_METHOD(void, PacketsDiscarded, (size_t num_packets), (override));
+  MOCK_METHOD(void,
+              SecondaryPacketsDiscarded,
+              (size_t num_packets),
+              (override));
+  MOCK_METHOD(void, RelativePacketArrivalDelay, (size_t delay_ms), (override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.cc b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.cc
new file mode 100644
index 0000000000..04cc5b52e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.cc
@@ -0,0 +1,267 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/nack_tracker.h"
+
+#include <cstdint>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+const int kDefaultSampleRateKhz = 48;
+const int kMaxPacketSizeMs = 120;
+constexpr char kNackTrackerConfigFieldTrial[] =
+    "WebRTC-Audio-NetEqNackTrackerConfig";
+
+}  // namespace
+
+NackTracker::Config::Config() {
+  auto parser = StructParametersParser::Create(
+      "packet_loss_forget_factor", &packet_loss_forget_factor,
+      "ms_per_loss_percent", &ms_per_loss_percent, "never_nack_multiple_times",
+      &never_nack_multiple_times, "require_valid_rtt", &require_valid_rtt,
+      "max_loss_rate", &max_loss_rate);
+  parser->Parse(
+      webrtc::field_trial::FindFullName(kNackTrackerConfigFieldTrial));
+  RTC_LOG(LS_INFO) << "Nack tracker config:"
+                      " packet_loss_forget_factor="
+                   << packet_loss_forget_factor
+                   << " ms_per_loss_percent=" << ms_per_loss_percent
+                   << " never_nack_multiple_times=" << never_nack_multiple_times
+                   << " require_valid_rtt=" << require_valid_rtt
+                   << " max_loss_rate=" << max_loss_rate;
+}
+
+NackTracker::NackTracker()
+    : sequence_num_last_received_rtp_(0),
+      timestamp_last_received_rtp_(0),
+      any_rtp_received_(false),
+      sequence_num_last_decoded_rtp_(0),
+      timestamp_last_decoded_rtp_(0),
+      any_rtp_decoded_(false),
+      sample_rate_khz_(kDefaultSampleRateKhz),
+      max_nack_list_size_(kNackListSizeLimit) {}
+
+NackTracker::~NackTracker() = default;
+
+void NackTracker::UpdateSampleRate(int sample_rate_hz) {
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  sample_rate_khz_ = sample_rate_hz / 1000;
+}
+
+void NackTracker::UpdateLastReceivedPacket(uint16_t sequence_number,
+                                           uint32_t timestamp) {
+  // Just record the value of sequence number and timestamp if this is the
+  // first packet.
+  if (!any_rtp_received_) {
+    sequence_num_last_received_rtp_ = sequence_number;
+    timestamp_last_received_rtp_ = timestamp;
+    any_rtp_received_ = true;
+    // If no packet is decoded, to have a reasonable estimate of time-to-play
+    // use the given values.
+    if (!any_rtp_decoded_) {
+      sequence_num_last_decoded_rtp_ = sequence_number;
+      timestamp_last_decoded_rtp_ = timestamp;
+    }
+    return;
+  }
+
+  if (sequence_number == sequence_num_last_received_rtp_)
+    return;
+
+  // Received RTP should not be in the list.
+  nack_list_.erase(sequence_number);
+
+  // If this is an old sequence number, no more action is required, return.
+  if (IsNewerSequenceNumber(sequence_num_last_received_rtp_, sequence_number))
+    return;
+
+  UpdatePacketLossRate(sequence_number - sequence_num_last_received_rtp_ - 1);
+
+  UpdateList(sequence_number, timestamp);
+
+  sequence_num_last_received_rtp_ = sequence_number;
+  timestamp_last_received_rtp_ = timestamp;
+  LimitNackListSize();
+}
+
+absl::optional<int> NackTracker::GetSamplesPerPacket(
+    uint16_t sequence_number_current_received_rtp,
+    uint32_t timestamp_current_received_rtp) const {
+  uint32_t timestamp_increase =
+      timestamp_current_received_rtp - timestamp_last_received_rtp_;
+  uint16_t sequence_num_increase =
+      sequence_number_current_received_rtp - sequence_num_last_received_rtp_;
+
+  int samples_per_packet = timestamp_increase / sequence_num_increase;
+  if (samples_per_packet == 0 ||
+      samples_per_packet > kMaxPacketSizeMs * sample_rate_khz_) {
+    // Not a valid samples per packet.
+    return absl::nullopt;
+  }
+  return samples_per_packet;
+}
+
+void NackTracker::UpdateList(uint16_t sequence_number_current_received_rtp,
+                             uint32_t timestamp_current_received_rtp) {
+  if (!IsNewerSequenceNumber(sequence_number_current_received_rtp,
+                             sequence_num_last_received_rtp_ + 1)) {
+    return;
+  }
+  RTC_DCHECK(!any_rtp_decoded_ ||
+             IsNewerSequenceNumber(sequence_number_current_received_rtp,
+                                   sequence_num_last_decoded_rtp_));
+
+  absl::optional<int> samples_per_packet = GetSamplesPerPacket(
+      sequence_number_current_received_rtp, timestamp_current_received_rtp);
+  if (!samples_per_packet) {
+    return;
+  }
+
+  for (uint16_t n = sequence_num_last_received_rtp_ + 1;
+       IsNewerSequenceNumber(sequence_number_current_received_rtp, n); ++n) {
+    uint32_t timestamp = EstimateTimestamp(n, *samples_per_packet);
+    NackElement nack_element(TimeToPlay(timestamp), timestamp);
+    nack_list_.insert(nack_list_.end(), std::make_pair(n, nack_element));
+  }
+}
+
+uint32_t NackTracker::EstimateTimestamp(uint16_t sequence_num,
+                                        int samples_per_packet) {
+  uint16_t sequence_num_diff = sequence_num - sequence_num_last_received_rtp_;
+  return sequence_num_diff * samples_per_packet + timestamp_last_received_rtp_;
+}
+
+void NackTracker::UpdateEstimatedPlayoutTimeBy10ms() {
+  while (!nack_list_.empty() &&
+         nack_list_.begin()->second.time_to_play_ms <= 10)
+    nack_list_.erase(nack_list_.begin());
+
+  for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end(); ++it)
+    it->second.time_to_play_ms -= 10;
+}
+
+void NackTracker::UpdateLastDecodedPacket(uint16_t sequence_number,
+                                          uint32_t timestamp) {
+  if (IsNewerSequenceNumber(sequence_number, sequence_num_last_decoded_rtp_) ||
+      !any_rtp_decoded_) {
+    sequence_num_last_decoded_rtp_ = sequence_number;
+    timestamp_last_decoded_rtp_ = timestamp;
+    // Packets in the list with sequence numbers less than the
+    // sequence number of the decoded RTP should be removed from the lists.
+    // They will be discarded by the jitter buffer if they arrive.
+    nack_list_.erase(nack_list_.begin(),
+                     nack_list_.upper_bound(sequence_num_last_decoded_rtp_));
+
+    // Update estimated time-to-play.
+    for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end();
+         ++it)
+      it->second.time_to_play_ms = TimeToPlay(it->second.estimated_timestamp);
+  } else {
+    RTC_DCHECK_EQ(sequence_number, sequence_num_last_decoded_rtp_);
+
+    // Same sequence number as before. 10 ms is elapsed, update estimations for
+    // time-to-play.
+    UpdateEstimatedPlayoutTimeBy10ms();
+
+    // Update timestamp for better estimate of time-to-play, for packets which
+    // are added to NACK list later on.
+    timestamp_last_decoded_rtp_ += sample_rate_khz_ * 10;
+  }
+  any_rtp_decoded_ = true;
+}
+
+NackTracker::NackList NackTracker::GetNackList() const {
+  return nack_list_;
+}
+
+void NackTracker::Reset() {
+  nack_list_.clear();
+
+  sequence_num_last_received_rtp_ = 0;
+  timestamp_last_received_rtp_ = 0;
+  any_rtp_received_ = false;
+  sequence_num_last_decoded_rtp_ = 0;
+  timestamp_last_decoded_rtp_ = 0;
+  any_rtp_decoded_ = false;
+  sample_rate_khz_ = kDefaultSampleRateKhz;
+}
+
+void NackTracker::SetMaxNackListSize(size_t max_nack_list_size) {
+  RTC_CHECK_GT(max_nack_list_size, 0);
+  // Ugly hack to get around the problem of passing static consts by reference.
+  const size_t kNackListSizeLimitLocal = NackTracker::kNackListSizeLimit;
+  RTC_CHECK_LE(max_nack_list_size, kNackListSizeLimitLocal);
+
+  max_nack_list_size_ = max_nack_list_size;
+  LimitNackListSize();
+}
+
+void NackTracker::LimitNackListSize() {
+  uint16_t limit = sequence_num_last_received_rtp_ -
+                   static_cast<uint16_t>(max_nack_list_size_) - 1;
+  nack_list_.erase(nack_list_.begin(), nack_list_.upper_bound(limit));
+}
+
+int64_t NackTracker::TimeToPlay(uint32_t timestamp) const {
+  uint32_t timestamp_increase = timestamp - timestamp_last_decoded_rtp_;
+  return timestamp_increase / sample_rate_khz_;
+}
+
+// We don't erase elements with time-to-play shorter than round-trip-time.
+std::vector<uint16_t> NackTracker::GetNackList(int64_t round_trip_time_ms) {
+  RTC_DCHECK_GE(round_trip_time_ms, 0);
+  std::vector<uint16_t> sequence_numbers;
+  if (round_trip_time_ms == 0) {
+    if (config_.require_valid_rtt) {
+      return sequence_numbers;
+    } else {
+      round_trip_time_ms = config_.default_rtt_ms;
+    }
+  }
+  if (packet_loss_rate_ >
+      static_cast<uint32_t>(config_.max_loss_rate * (1 << 30))) {
+    return sequence_numbers;
+  }
+  // The estimated packet loss is between 0 and 1, so we need to multiply by 100
+  // here.
+  int max_wait_ms =
+      100.0 * config_.ms_per_loss_percent * packet_loss_rate_ / (1 << 30);
+  for (NackList::const_iterator it = nack_list_.begin(); it != nack_list_.end();
+       ++it) {
+    int64_t time_since_packet_ms =
+        (timestamp_last_received_rtp_ - it->second.estimated_timestamp) /
+        sample_rate_khz_;
+    if (it->second.time_to_play_ms > round_trip_time_ms ||
+        time_since_packet_ms + round_trip_time_ms < max_wait_ms)
+      sequence_numbers.push_back(it->first);
+  }
+  if (config_.never_nack_multiple_times) {
+    nack_list_.clear();
+  }
+  return sequence_numbers;
+}
+
+void NackTracker::UpdatePacketLossRate(int packets_lost) {
+  const uint64_t alpha_q30 = (1 << 30) * config_.packet_loss_forget_factor;
+  // Exponential filter.
+  packet_loss_rate_ = (alpha_q30 * packet_loss_rate_) >> 30;
+  for (int i = 0; i < packets_lost; ++i) {
+    packet_loss_rate_ =
+        ((alpha_q30 * packet_loss_rate_) >> 30) + ((1 << 30) - alpha_q30);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.h b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.h
new file mode 100644
index 0000000000..14ba2166d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.h
@@ -0,0 +1,211 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
+#define MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/include/module_common_types_public.h"
+#include "rtc_base/gtest_prod_util.h"
+
+//
+// The NackTracker class keeps track of the lost packets, an estimate of
+// time-to-play for each packet is also given.
+//
+// Every time a packet is pushed into NetEq, LastReceivedPacket() has to be
+// called to update the NACK list.
+//
+// Every time 10ms audio is pulled from NetEq LastDecodedPacket() should be
+// called, and time-to-play is updated at that moment.
+//
+// If packet N is received, any packet prior to N which has not arrived is
+// considered lost, and should be labeled as "missing" (the size of
+// the list might be limited and older packet eliminated from the list).
+//
+// The NackTracker class has to know about the sample rate of the packets to
+// compute time-to-play. So sample rate should be set as soon as the first
+// packet is received. If there is a change in the receive codec (sender changes
+// codec) then NackTracker should be reset. This is because NetEQ would flush
+// its buffer and re-transmission is meaning less for old packet. Therefore, in
+// that case, after reset the sampling rate has to be updated.
+//
+// Thread Safety
+// =============
+// Please note that this class in not thread safe. The class must be protected
+// if different APIs are called from different threads.
+//
+namespace webrtc {
+
+class NackTracker {
+ public:
+  // A limit for the size of the NACK list.
+  static const size_t kNackListSizeLimit = 500;  // 10 seconds for 20 ms frame
+                                                 // packets.
+  NackTracker();
+  ~NackTracker();
+
+  // Set a maximum for the size of the NACK list. If the last received packet
+  // has sequence number of N, then NACK list will not contain any element
+  // with sequence number earlier than N - `max_nack_list_size`.
+  //
+  // The largest maximum size is defined by `kNackListSizeLimit`
+  void SetMaxNackListSize(size_t max_nack_list_size);
+
+  // Set the sampling rate.
+  //
+  // If associated sampling rate of the received packets is changed, call this
+  // function to update sampling rate. Note that if there is any change in
+  // received codec then NetEq will flush its buffer and NACK has to be reset.
+  // After Reset() is called sampling rate has to be set.
+  void UpdateSampleRate(int sample_rate_hz);
+
+  // Update the sequence number and the timestamp of the last decoded RTP. This
+  // API should be called every time 10 ms audio is pulled from NetEq.
+  void UpdateLastDecodedPacket(uint16_t sequence_number, uint32_t timestamp);
+
+  // Update the sequence number and the timestamp of the last received RTP. This
+  // API should be called every time a packet pushed into ACM.
+  void UpdateLastReceivedPacket(uint16_t sequence_number, uint32_t timestamp);
+
+  // Get a list of "missing" packets which have expected time-to-play larger
+  // than the given round-trip-time (in milliseconds).
+  // Note: Late packets are not included.
+  // Calling this method multiple times may give different results, since the
+  // internal nack list may get flushed if never_nack_multiple_times_ is true.
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms);
+
+  // Reset to default values. The NACK list is cleared.
+  // `max_nack_list_size_` preserves its value.
+  void Reset();
+
+  // Returns the estimated packet loss rate in Q30, for testing only.
+  uint32_t GetPacketLossRateForTest() { return packet_loss_rate_; }
+
+ private:
+  // This test need to access the private method GetNackList().
+  FRIEND_TEST_ALL_PREFIXES(NackTrackerTest, EstimateTimestampAndTimeToPlay);
+
+  // Options that can be configured via field trial.
+  struct Config {
+    Config();
+
+    // The exponential decay factor used to estimate the packet loss rate.
+    double packet_loss_forget_factor = 0.996;
+    // How many additional ms we are willing to wait (at most) for nacked
+    // packets for each additional percentage of packet loss.
+    int ms_per_loss_percent = 20;
+    // If true, never nack packets more than once.
+    bool never_nack_multiple_times = false;
+    // Only nack if the RTT is valid.
+    bool require_valid_rtt = false;
+    // Default RTT to use unless `require_valid_rtt` is set.
+    int default_rtt_ms = 100;
+    // Do not nack if the loss rate is above this value.
+    double max_loss_rate = 1.0;
+  };
+
+  struct NackElement {
+    NackElement(int64_t initial_time_to_play_ms, uint32_t initial_timestamp)
+        : time_to_play_ms(initial_time_to_play_ms),
+          estimated_timestamp(initial_timestamp) {}
+
+    // Estimated time (ms) left for this packet to be decoded. This estimate is
+    // updated every time jitter buffer decodes a packet.
+    int64_t time_to_play_ms;
+
+    // A guess about the timestamp of the missing packet, it is used for
+    // estimation of `time_to_play_ms`. The estimate might be slightly wrong if
+    // there has been frame-size change since the last received packet and the
+    // missing packet. However, the risk of this is low, and in case of such
+    // errors, there will be a minor misestimation in time-to-play of missing
+    // packets. This will have a very minor effect on NACK performance.
+    uint32_t estimated_timestamp;
+  };
+
+  class NackListCompare {
+   public:
+    bool operator()(uint16_t sequence_number_old,
+                    uint16_t sequence_number_new) const {
+      return IsNewerSequenceNumber(sequence_number_new, sequence_number_old);
+    }
+  };
+
+  typedef std::map<uint16_t, NackElement, NackListCompare> NackList;
+
+  // This API is used only for testing to assess whether time-to-play is
+  // computed correctly.
+  NackList GetNackList() const;
+
+  // This function subtracts 10 ms of time-to-play for all packets in NACK list.
+  // This is called when 10 ms elapsed with no new RTP packet decoded.
+  void UpdateEstimatedPlayoutTimeBy10ms();
+
+  // Returns a valid number of samples per packet given the current received
+  // sequence number and timestamp or nullopt of none could be computed.
+  absl::optional<int> GetSamplesPerPacket(
+      uint16_t sequence_number_current_received_rtp,
+      uint32_t timestamp_current_received_rtp) const;
+
+  // Given the `sequence_number_current_received_rtp` of currently received RTP
+  // update the list. Packets that are older than the received packet are added
+  // to the nack list.
+  void UpdateList(uint16_t sequence_number_current_received_rtp,
+                  uint32_t timestamp_current_received_rtp);
+
+  // Packets which have sequence number older that
+  // `sequence_num_last_received_rtp_` - `max_nack_list_size_` are removed
+  // from the NACK list.
+  void LimitNackListSize();
+
+  // Estimate timestamp of a missing packet given its sequence number.
+  uint32_t EstimateTimestamp(uint16_t sequence_number, int samples_per_packet);
+
+  // Compute time-to-play given a timestamp.
+  int64_t TimeToPlay(uint32_t timestamp) const;
+
+  // Updates the estimated packet lost rate.
+  void UpdatePacketLossRate(int packets_lost);
+
+  const Config config_;
+
+  // Valid if a packet is received.
+  uint16_t sequence_num_last_received_rtp_;
+  uint32_t timestamp_last_received_rtp_;
+  bool any_rtp_received_;  // If any packet received.
+
+  // Valid if a packet is decoded.
+  uint16_t sequence_num_last_decoded_rtp_;
+  uint32_t timestamp_last_decoded_rtp_;
+  bool any_rtp_decoded_;  // If any packet decoded.
+
+  int sample_rate_khz_;  // Sample rate in kHz.
+
+  // A list of missing packets to be retransmitted. Components of the list
+  // contain the sequence number of missing packets and the estimated time that
+  // each pack is going to be played out.
+  NackList nack_list_;
+
+  // NACK list will not keep track of missing packets prior to
+  // `sequence_num_last_received_rtp_` - `max_nack_list_size_`.
+  size_t max_nack_list_size_;
+
+  // Current estimate of the packet loss rate in Q30.
+  uint32_t packet_loss_rate_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc
new file mode 100644
index 0000000000..bcc5120ff3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc
@@ -0,0 +1,565 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/nack_tracker.h"
+
+#include <stdint.h>
+
+#include <algorithm>
+#include <memory>
+
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+const int kSampleRateHz = 16000;
+const int kPacketSizeMs = 30;
+const uint32_t kTimestampIncrement = 480;  // 30 ms.
+const int64_t kShortRoundTripTimeMs = 1;
+
+bool IsNackListCorrect(const std::vector<uint16_t>& nack_list,
+                       const uint16_t* lost_sequence_numbers,
+                       size_t num_lost_packets) {
+  if (nack_list.size() != num_lost_packets)
+    return false;
+
+  if (num_lost_packets == 0)
+    return true;
+
+  for (size_t k = 0; k < nack_list.size(); ++k) {
+    int seq_num = nack_list[k];
+    bool seq_num_matched = false;
+    for (size_t n = 0; n < num_lost_packets; ++n) {
+      if (seq_num == lost_sequence_numbers[n]) {
+        seq_num_matched = true;
+        break;
+      }
+    }
+    if (!seq_num_matched)
+      return false;
+  }
+  return true;
+}
+
+}  // namespace
+
+TEST(NackTrackerTest, EmptyListWhenNoPacketLoss) {
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+
+  int seq_num = 1;
+  uint32_t timestamp = 0;
+
+  std::vector<uint16_t> nack_list;
+  for (int n = 0; n < 100; n++) {
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    seq_num++;
+    timestamp += kTimestampIncrement;
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+  }
+}
+
+TEST(NackTrackerTest, LatePacketsMovedToNackThenNackListDoesNotChange) {
+  const uint16_t kSequenceNumberLostPackets[] = {2, 3, 4, 5, 6, 7, 8, 9};
+  static const int kNumAllLostPackets = sizeof(kSequenceNumberLostPackets) /
+                                        sizeof(kSequenceNumberLostPackets[0]);
+
+  for (int k = 0; k < 2; k++) {  // Two iteration with/without wrap around.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    uint16_t sequence_num_lost_packets[kNumAllLostPackets];
+    for (int n = 0; n < kNumAllLostPackets; n++) {
+      sequence_num_lost_packets[n] =
+          kSequenceNumberLostPackets[n] +
+          k * 65531;  // Have wrap around in sequence numbers for |k == 1|.
+    }
+    uint16_t seq_num = sequence_num_lost_packets[0] - 1;
+
+    uint32_t timestamp = 0;
+    std::vector<uint16_t> nack_list;
+
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+
+    seq_num = sequence_num_lost_packets[kNumAllLostPackets - 1] + 1;
+    timestamp += kTimestampIncrement * (kNumAllLostPackets + 1);
+    int num_lost_packets = std::max(0, kNumAllLostPackets);
+
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(IsNackListCorrect(nack_list, sequence_num_lost_packets,
+                                  num_lost_packets));
+    seq_num++;
+    timestamp += kTimestampIncrement;
+    num_lost_packets++;
+
+    for (int n = 0; n < 100; ++n) {
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(IsNackListCorrect(nack_list, sequence_num_lost_packets,
+                                    kNumAllLostPackets));
+      seq_num++;
+      timestamp += kTimestampIncrement;
+    }
+  }
+}
+
+TEST(NackTrackerTest, ArrivedPacketsAreRemovedFromNackList) {
+  const uint16_t kSequenceNumberLostPackets[] = {2, 3, 4, 5, 6, 7, 8, 9};
+  static const int kNumAllLostPackets = sizeof(kSequenceNumberLostPackets) /
+                                        sizeof(kSequenceNumberLostPackets[0]);
+
+  for (int k = 0; k < 2; ++k) {  // Two iteration with/without wrap around.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    uint16_t sequence_num_lost_packets[kNumAllLostPackets];
+    for (int n = 0; n < kNumAllLostPackets; ++n) {
+      sequence_num_lost_packets[n] = kSequenceNumberLostPackets[n] +
+                                     k * 65531;  // Wrap around for |k == 1|.
+    }
+
+    uint16_t seq_num = sequence_num_lost_packets[0] - 1;
+    uint32_t timestamp = 0;
+
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+
+    size_t index_retransmitted_rtp = 0;
+    uint32_t timestamp_retransmitted_rtp = timestamp + kTimestampIncrement;
+
+    seq_num = sequence_num_lost_packets[kNumAllLostPackets - 1] + 1;
+    timestamp += kTimestampIncrement * (kNumAllLostPackets + 1);
+    size_t num_lost_packets = kNumAllLostPackets;
+    for (int n = 0; n < kNumAllLostPackets; ++n) {
+      // Number of lost packets does not change for the first
+      // |kNackThreshold + 1| packets, one is added to the list and one is
+      // removed. Thereafter, the list shrinks every iteration.
+      if (n >= 1)
+        num_lost_packets--;
+
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(IsNackListCorrect(
+          nack_list, &sequence_num_lost_packets[index_retransmitted_rtp],
+          num_lost_packets));
+      seq_num++;
+      timestamp += kTimestampIncrement;
+
+      // Retransmission of a lost RTP.
+      nack.UpdateLastReceivedPacket(
+          sequence_num_lost_packets[index_retransmitted_rtp],
+          timestamp_retransmitted_rtp);
+      index_retransmitted_rtp++;
+      timestamp_retransmitted_rtp += kTimestampIncrement;
+
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(IsNackListCorrect(
+          nack_list, &sequence_num_lost_packets[index_retransmitted_rtp],
+          num_lost_packets - 1));  // One less lost packet in the list.
+    }
+    ASSERT_TRUE(nack_list.empty());
+  }
+}
+
+// Assess if estimation of timestamps and time-to-play is correct. Introduce all
+// combinations that timestamps and sequence numbers might have wrap around.
+TEST(NackTrackerTest, EstimateTimestampAndTimeToPlay) {
+  const uint16_t kLostPackets[] = {2, 3,  4,  5,  6,  7,  8,
+                                   9, 10, 11, 12, 13, 14, 15};
+  static const int kNumAllLostPackets =
+      sizeof(kLostPackets) / sizeof(kLostPackets[0]);
+
+  for (int k = 0; k < 4; ++k) {
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    // Sequence number wrap around if `k` is 2 or 3;
+    int seq_num_offset = (k < 2) ? 0 : 65531;
+
+    // Timestamp wrap around if `k` is 1 or 3.
+    uint32_t timestamp_offset =
+        (k & 0x1) ? static_cast<uint32_t>(0xffffffff) - 6 : 0;
+
+    uint32_t timestamp_lost_packets[kNumAllLostPackets];
+    uint16_t seq_num_lost_packets[kNumAllLostPackets];
+    for (int n = 0; n < kNumAllLostPackets; ++n) {
+      timestamp_lost_packets[n] =
+          timestamp_offset + kLostPackets[n] * kTimestampIncrement;
+      seq_num_lost_packets[n] = seq_num_offset + kLostPackets[n];
+    }
+
+    // We and to push two packets before lost burst starts.
+    uint16_t seq_num = seq_num_lost_packets[0] - 2;
+    uint32_t timestamp = timestamp_lost_packets[0] - 2 * kTimestampIncrement;
+
+    const uint16_t first_seq_num = seq_num;
+    const uint32_t first_timestamp = timestamp;
+
+    // Two consecutive packets to have a correct estimate of timestamp increase.
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    seq_num++;
+    timestamp += kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    // A packet after the last one which is supposed to be lost.
+    seq_num = seq_num_lost_packets[kNumAllLostPackets - 1] + 1;
+    timestamp =
+        timestamp_lost_packets[kNumAllLostPackets - 1] + kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    NackTracker::NackList nack_list = nack.GetNackList();
+    EXPECT_EQ(static_cast<size_t>(kNumAllLostPackets), nack_list.size());
+
+    // Pretend the first packet is decoded.
+    nack.UpdateLastDecodedPacket(first_seq_num, first_timestamp);
+    nack_list = nack.GetNackList();
+
+    NackTracker::NackList::iterator it = nack_list.begin();
+    while (it != nack_list.end()) {
+      seq_num = it->first - seq_num_offset;
+      int index = seq_num - kLostPackets[0];
+      EXPECT_EQ(timestamp_lost_packets[index], it->second.estimated_timestamp);
+      EXPECT_EQ((index + 2) * kPacketSizeMs, it->second.time_to_play_ms);
+      ++it;
+    }
+
+    // Pretend 10 ms is passed, and we had pulled audio from NetEq, it still
+    // reports the same sequence number as decoded, time-to-play should be
+    // updated by 10 ms.
+    nack.UpdateLastDecodedPacket(first_seq_num, first_timestamp);
+    nack_list = nack.GetNackList();
+    it = nack_list.begin();
+    while (it != nack_list.end()) {
+      seq_num = it->first - seq_num_offset;
+      int index = seq_num - kLostPackets[0];
+      EXPECT_EQ((index + 2) * kPacketSizeMs - 10, it->second.time_to_play_ms);
+      ++it;
+    }
+  }
+}
+
+TEST(NackTrackerTest,
+     MissingPacketsPriorToLastDecodedRtpShouldNotBeInNackList) {
+  for (int m = 0; m < 2; ++m) {
+    uint16_t seq_num_offset = (m == 0) ? 0 : 65531;  // Wrap around if `m` is 1.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    // Two consecutive packets to have a correct estimate of timestamp increase.
+    uint16_t seq_num = 0;
+    nack.UpdateLastReceivedPacket(seq_num_offset + seq_num,
+                                  seq_num * kTimestampIncrement);
+    seq_num++;
+    nack.UpdateLastReceivedPacket(seq_num_offset + seq_num,
+                                  seq_num * kTimestampIncrement);
+
+    // Skip 10 packets (larger than NACK threshold).
+    const int kNumLostPackets = 10;
+    seq_num += kNumLostPackets + 1;
+    nack.UpdateLastReceivedPacket(seq_num_offset + seq_num,
+                                  seq_num * kTimestampIncrement);
+
+    const size_t kExpectedListSize = kNumLostPackets;
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_EQ(kExpectedListSize, nack_list.size());
+
+    for (int k = 0; k < 2; ++k) {
+      // Decoding of the first and the second arrived packets.
+      for (int n = 0; n < kPacketSizeMs / 10; ++n) {
+        nack.UpdateLastDecodedPacket(seq_num_offset + k,
+                                     k * kTimestampIncrement);
+        nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+        EXPECT_EQ(kExpectedListSize, nack_list.size());
+      }
+    }
+
+    // Decoding of the last received packet.
+    nack.UpdateLastDecodedPacket(seq_num + seq_num_offset,
+                                 seq_num * kTimestampIncrement);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+
+    // Make sure list of late packets is also empty. To check that, push few
+    // packets, if the late list is not empty its content will pop up in NACK
+    // list.
+    for (int n = 0; n < 10; ++n) {
+      seq_num++;
+      nack.UpdateLastReceivedPacket(seq_num_offset + seq_num,
+                                    seq_num * kTimestampIncrement);
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(nack_list.empty());
+    }
+  }
+}
+
+TEST(NackTrackerTest, Reset) {
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+
+  // Two consecutive packets to have a correct estimate of timestamp increase.
+  uint16_t seq_num = 0;
+  nack.UpdateLastReceivedPacket(seq_num, seq_num * kTimestampIncrement);
+  seq_num++;
+  nack.UpdateLastReceivedPacket(seq_num, seq_num * kTimestampIncrement);
+
+  // Skip 10 packets (larger than NACK threshold).
+  const int kNumLostPackets = 10;
+  seq_num += kNumLostPackets + 1;
+  nack.UpdateLastReceivedPacket(seq_num, seq_num * kTimestampIncrement);
+
+  const size_t kExpectedListSize = kNumLostPackets;
+  std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+  EXPECT_EQ(kExpectedListSize, nack_list.size());
+
+  nack.Reset();
+  nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+  EXPECT_TRUE(nack_list.empty());
+}
+
+TEST(NackTrackerTest, ListSizeAppliedFromBeginning) {
+  const size_t kNackListSize = 10;
+  for (int m = 0; m < 2; ++m) {
+    uint16_t seq_num_offset = (m == 0) ? 0 : 65525;  // Wrap around if `m` is 1.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+    nack.SetMaxNackListSize(kNackListSize);
+
+    uint16_t seq_num = seq_num_offset;
+    uint32_t timestamp = 0x12345678;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    // Packet lost more than NACK-list size limit.
+    uint16_t num_lost_packets = kNackListSize + 5;
+
+    seq_num += num_lost_packets + 1;
+    timestamp += (num_lost_packets + 1) * kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_EQ(kNackListSize, nack_list.size());
+  }
+}
+
+TEST(NackTrackerTest, ChangeOfListSizeAppliedAndOldElementsRemoved) {
+  const size_t kNackListSize = 10;
+  for (int m = 0; m < 2; ++m) {
+    uint16_t seq_num_offset = (m == 0) ? 0 : 65525;  // Wrap around if `m` is 1.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    uint16_t seq_num = seq_num_offset;
+    uint32_t timestamp = 0x87654321;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    // Packet lost more than NACK-list size limit.
+    uint16_t num_lost_packets = kNackListSize + 5;
+
+    std::unique_ptr<uint16_t[]> seq_num_lost(new uint16_t[num_lost_packets]);
+    for (int n = 0; n < num_lost_packets; ++n) {
+      seq_num_lost[n] = ++seq_num;
+    }
+
+    ++seq_num;
+    timestamp += (num_lost_packets + 1) * kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    size_t expected_size = num_lost_packets;
+
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_EQ(expected_size, nack_list.size());
+
+    nack.SetMaxNackListSize(kNackListSize);
+    expected_size = kNackListSize;
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(IsNackListCorrect(
+        nack_list, &seq_num_lost[num_lost_packets - kNackListSize],
+        expected_size));
+
+    // NACK list should shrink.
+    for (size_t n = 1; n < kNackListSize; ++n) {
+      ++seq_num;
+      timestamp += kTimestampIncrement;
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+      --expected_size;
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(IsNackListCorrect(
+          nack_list, &seq_num_lost[num_lost_packets - kNackListSize + n],
+          expected_size));
+    }
+
+    // After this packet, NACK list should be empty.
+    ++seq_num;
+    timestamp += kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+  }
+}
+
+TEST(NackTrackerTest, RoudTripTimeIsApplied) {
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+  // Packet lost more than NACK-list size limit.
+  uint16_t kNumLostPackets = 5;
+
+  seq_num += (1 + kNumLostPackets);
+  timestamp += (1 + kNumLostPackets) * kTimestampIncrement;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+  // Expected time-to-play are:
+  // kPacketSizeMs - 10, 2*kPacketSizeMs - 10, 3*kPacketSizeMs - 10, ...
+  //
+  // sequence number:  1,  2,  3,   4,   5
+  // time-to-play:    20, 50, 80, 110, 140
+  //
+  std::vector<uint16_t> nack_list = nack.GetNackList(100);
+  ASSERT_EQ(2u, nack_list.size());
+  EXPECT_EQ(4, nack_list[0]);
+  EXPECT_EQ(5, nack_list[1]);
+}
+
+// Set never_nack_multiple_times to true with a field trial and verify that
+// packets are not nacked multiple times.
+TEST(NackTrackerTest, DoNotNackMultipleTimes) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqNackTrackerConfig/"
+      "packet_loss_forget_factor:0.996,ms_per_loss_percent:20,"
+      "never_nack_multiple_times:true/");
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+  uint16_t kNumLostPackets = 3;
+
+  seq_num += (1 + kNumLostPackets);
+  timestamp += (1 + kNumLostPackets) * kTimestampIncrement;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+  std::vector<uint16_t> nack_list = nack.GetNackList(10);
+  ASSERT_EQ(3u, nack_list.size());
+  EXPECT_EQ(1, nack_list[0]);
+  EXPECT_EQ(2, nack_list[1]);
+  EXPECT_EQ(3, nack_list[2]);
+  // When we get the nack list again, it should be empty.
+  std::vector<uint16_t> nack_list2 = nack.GetNackList(10);
+  EXPECT_TRUE(nack_list2.empty());
+}
+
+// Test if estimated packet loss rate is correct.
+TEST(NackTrackerTest, PacketLossRateCorrect) {
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  auto add_packet = [&nack, &seq_num, &timestamp](bool received) {
+    if (received) {
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    }
+    seq_num++;
+    timestamp += kTimestampIncrement;
+  };
+  // Add some packets, but every fourth packet is lost.
+  for (int i = 0; i < 300; i++) {
+    add_packet(true);
+    add_packet(true);
+    add_packet(true);
+    add_packet(false);
+  }
+  // 1 << 28 is 0.25 in Q30. We expect the packet loss estimate to be within
+  // 0.01 of that.
+  EXPECT_NEAR(nack.GetPacketLossRateForTest(), 1 << 28, (1 << 30) / 100);
+}
+
+TEST(NackTrackerTest, DoNotNackAfterDtx) {
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+  EXPECT_TRUE(nack.GetNackList(0).empty());
+  constexpr int kDtxPeriod = 400;
+  nack.UpdateLastReceivedPacket(seq_num + 2,
+                                timestamp + kDtxPeriod * kSampleRateHz / 1000);
+  EXPECT_TRUE(nack.GetNackList(0).empty());
+}
+
+TEST(NackTrackerTest, DoNotNackIfLossRateIsTooHigh) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqNackTrackerConfig/max_loss_rate:0.4/");
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  auto add_packet = [&nack, &seq_num, &timestamp](bool received) {
+    if (received) {
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    }
+    seq_num++;
+    timestamp += kTimestampIncrement;
+  };
+  for (int i = 0; i < 500; i++) {
+    add_packet(true);
+    add_packet(false);
+  }
+  // Expect 50% loss rate which is higher that the configured maximum 40%.
+  EXPECT_NEAR(nack.GetPacketLossRateForTest(), 1 << 29, (1 << 30) / 100);
+  EXPECT_TRUE(nack.GetNackList(0).empty());
+}
+
+TEST(NackTrackerTest, OnlyNackIfRttIsValid) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqNackTrackerConfig/require_valid_rtt:true/");
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  auto add_packet = [&nack, &seq_num, &timestamp](bool received) {
+    if (received) {
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    }
+    seq_num++;
+    timestamp += kTimestampIncrement;
+  };
+  add_packet(true);
+  add_packet(false);
+  add_packet(true);
+  EXPECT_TRUE(nack.GetNackList(0).empty());
+  EXPECT_FALSE(nack.GetNackList(10).empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
new file mode 100644
index 0000000000..cf310d1efb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
@@ -0,0 +1,313 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Test to verify correct operation when using the decoder-internal PLC.
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/audio_checksum.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/encode_neteq_input.h"
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/audio_decoder_proxy_factory.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr int kSampleRateHz = 32000;
+constexpr int kRunTimeMs = 10000;
+
+// This class implements a fake decoder. The decoder will read audio from a file
+// and present as output, both for regular decoding and for PLC.
+class AudioDecoderPlc : public AudioDecoder {
+ public:
+  AudioDecoderPlc(std::unique_ptr<InputAudioFile> input, int sample_rate_hz)
+      : input_(std::move(input)), sample_rate_hz_(sample_rate_hz) {}
+
+  void Reset() override {}
+  int SampleRateHz() const override { return sample_rate_hz_; }
+  size_t Channels() const override { return 1; }
+  int DecodeInternal(const uint8_t* /*encoded*/,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override {
+    RTC_CHECK_GE(encoded_len / 2, 10 * sample_rate_hz_ / 1000);
+    RTC_CHECK_LE(encoded_len / 2, 2 * 10 * sample_rate_hz_ / 1000);
+    RTC_CHECK_EQ(sample_rate_hz, sample_rate_hz_);
+    RTC_CHECK(decoded);
+    RTC_CHECK(speech_type);
+    RTC_CHECK(input_->Read(encoded_len / 2, decoded));
+    *speech_type = kSpeech;
+    last_was_plc_ = false;
+    return encoded_len / 2;
+  }
+
+  void GeneratePlc(size_t requested_samples_per_channel,
+                   rtc::BufferT<int16_t>* concealment_audio) override {
+    // Instead of generating random data for GeneratePlc we use the same data as
+    // the input, so we can check that we produce the same result independently
+    // of the losses.
+    RTC_DCHECK_EQ(requested_samples_per_channel, 10 * sample_rate_hz_ / 1000);
+
+    // Must keep a local copy of this since DecodeInternal sets it to false.
+    const bool last_was_plc = last_was_plc_;
+
+    std::vector<int16_t> decoded(5760);
+    SpeechType speech_type;
+    int dec_len = DecodeInternal(nullptr, 2 * 10 * sample_rate_hz_ / 1000,
+                                 sample_rate_hz_, decoded.data(), &speech_type);
+    concealment_audio->AppendData(decoded.data(), dec_len);
+    concealed_samples_ += rtc::checked_cast<size_t>(dec_len);
+
+    if (!last_was_plc) {
+      ++concealment_events_;
+    }
+    last_was_plc_ = true;
+  }
+
+  size_t concealed_samples() { return concealed_samples_; }
+  size_t concealment_events() { return concealment_events_; }
+
+ private:
+  const std::unique_ptr<InputAudioFile> input_;
+  const int sample_rate_hz_;
+  size_t concealed_samples_ = 0;
+  size_t concealment_events_ = 0;
+  bool last_was_plc_ = false;
+};
+
+// An input sample generator which generates only zero-samples.
+class ZeroSampleGenerator : public EncodeNetEqInput::Generator {
+ public:
+  rtc::ArrayView<const int16_t> Generate(size_t num_samples) override {
+    vec.resize(num_samples, 0);
+    rtc::ArrayView<const int16_t> view(vec);
+    RTC_DCHECK_EQ(view.size(), num_samples);
+    return view;
+  }
+
+ private:
+  std::vector<int16_t> vec;
+};
+
+// A NetEqInput which connects to another NetEqInput, but drops a number of
+// consecutive packets on the way
+class LossyInput : public NetEqInput {
+ public:
+  LossyInput(int loss_cadence,
+             int burst_length,
+             std::unique_ptr<NetEqInput> input)
+      : loss_cadence_(loss_cadence),
+        burst_length_(burst_length),
+        input_(std::move(input)) {}
+
+  absl::optional<int64_t> NextPacketTime() const override {
+    return input_->NextPacketTime();
+  }
+
+  absl::optional<int64_t> NextOutputEventTime() const override {
+    return input_->NextOutputEventTime();
+  }
+
+  std::unique_ptr<PacketData> PopPacket() override {
+    if (loss_cadence_ != 0 && (++count_ % loss_cadence_) == 0) {
+      // Pop `burst_length_` packets to create the loss.
+      auto packet_to_return = input_->PopPacket();
+      for (int i = 0; i < burst_length_; i++) {
+        input_->PopPacket();
+      }
+      return packet_to_return;
+    }
+    return input_->PopPacket();
+  }
+
+  void AdvanceOutputEvent() override { return input_->AdvanceOutputEvent(); }
+
+  bool ended() const override { return input_->ended(); }
+
+  absl::optional<RTPHeader> NextHeader() const override {
+    return input_->NextHeader();
+  }
+
+ private:
+  const int loss_cadence_;
+  const int burst_length_;
+  int count_ = 0;
+  const std::unique_ptr<NetEqInput> input_;
+};
+
+class AudioChecksumWithOutput : public AudioChecksum {
+ public:
+  explicit AudioChecksumWithOutput(std::string* output_str)
+      : output_str_(*output_str) {}
+  ~AudioChecksumWithOutput() { output_str_ = Finish(); }
+
+ private:
+  std::string& output_str_;
+};
+
+struct TestStatistics {
+  NetEqNetworkStatistics network;
+  NetEqLifetimeStatistics lifetime;
+};
+
+TestStatistics RunTest(int loss_cadence,
+                       int burst_length,
+                       std::string* checksum) {
+  NetEq::Config config;
+  config.for_test_no_time_stretching = true;
+
+  // The input is mostly useless. It sends zero-samples to a PCM16b encoder,
+  // but the actual encoded samples will never be used by the decoder in the
+  // test. See below about the decoder.
+  auto generator = std::make_unique<ZeroSampleGenerator>();
+  constexpr int kPayloadType = 100;
+  AudioEncoderPcm16B::Config encoder_config;
+  encoder_config.sample_rate_hz = kSampleRateHz;
+  encoder_config.payload_type = kPayloadType;
+  auto encoder = std::make_unique<AudioEncoderPcm16B>(encoder_config);
+  auto input = std::make_unique<EncodeNetEqInput>(
+      std::move(generator), std::move(encoder), kRunTimeMs);
+  // Wrap the input in a loss function.
+  auto lossy_input = std::make_unique<LossyInput>(loss_cadence, burst_length,
+                                                  std::move(input));
+
+  // Setting up decoders.
+  NetEqTest::DecoderMap decoders;
+  // Using a fake decoder which simply reads the output audio from a file.
+  auto input_file = std::make_unique<InputAudioFile>(
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"));
+  AudioDecoderPlc dec(std::move(input_file), kSampleRateHz);
+  // Masquerading as a PCM16b decoder.
+  decoders.emplace(kPayloadType, SdpAudioFormat("l16", 32000, 1));
+
+  // Output is simply a checksum calculator.
+  auto output = std::make_unique<AudioChecksumWithOutput>(checksum);
+
+  // No callback objects.
+  NetEqTest::Callbacks callbacks;
+
+  NetEqTest neteq_test(
+      config, /*decoder_factory=*/
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&dec),
+      /*codecs=*/decoders, /*text_log=*/nullptr, /*neteq_factory=*/nullptr,
+      /*input=*/std::move(lossy_input), std::move(output), callbacks);
+  EXPECT_LE(kRunTimeMs, neteq_test.Run());
+
+  auto lifetime_stats = neteq_test.LifetimeStats();
+  EXPECT_EQ(dec.concealed_samples(), lifetime_stats.concealed_samples);
+  EXPECT_EQ(dec.concealment_events(), lifetime_stats.concealment_events);
+  return {neteq_test.SimulationStats(), neteq_test.LifetimeStats()};
+}
+}  // namespace
+
+// Check that some basic metrics are produced in the right direction. In
+// particular, expand_rate should only increase if there are losses present. Our
+// dummy decoder is designed such as the checksum should always be the same
+// regardless of the losses given that calls are executed in the right order.
+TEST(NetEqDecoderPlc, BasicMetrics) {
+  std::string checksum;
+
+  // Drop 1 packet every 10 packets.
+  auto stats = RunTest(10, 1, &checksum);
+
+  std::string checksum_no_loss;
+  auto stats_no_loss = RunTest(0, 0, &checksum_no_loss);
+
+  EXPECT_EQ(checksum, checksum_no_loss);
+
+  EXPECT_EQ(stats.network.preemptive_rate,
+            stats_no_loss.network.preemptive_rate);
+  EXPECT_EQ(stats.network.accelerate_rate,
+            stats_no_loss.network.accelerate_rate);
+  EXPECT_EQ(0, stats_no_loss.network.expand_rate);
+  EXPECT_GT(stats.network.expand_rate, 0);
+}
+
+// Checks that interruptions are not counted in small losses but they are
+// correctly counted in long interruptions.
+TEST(NetEqDecoderPlc, CountInterruptions) {
+  std::string checksum;
+  std::string checksum_2;
+  std::string checksum_3;
+
+  // Half of the packets lost but in short interruptions.
+  auto stats_no_interruptions = RunTest(1, 1, &checksum);
+  // One lost of 500 ms (250 packets).
+  auto stats_one_interruption = RunTest(200, 250, &checksum_2);
+  // Two losses of 250ms each (125 packets).
+  auto stats_two_interruptions = RunTest(125, 125, &checksum_3);
+
+  EXPECT_EQ(checksum, checksum_2);
+  EXPECT_EQ(checksum, checksum_3);
+  EXPECT_GT(stats_no_interruptions.network.expand_rate, 0);
+  EXPECT_EQ(stats_no_interruptions.lifetime.total_interruption_duration_ms, 0);
+  EXPECT_EQ(stats_no_interruptions.lifetime.interruption_count, 0);
+
+  EXPECT_GT(stats_one_interruption.network.expand_rate, 0);
+  EXPECT_EQ(stats_one_interruption.lifetime.total_interruption_duration_ms,
+            5000);
+  EXPECT_EQ(stats_one_interruption.lifetime.interruption_count, 1);
+
+  EXPECT_GT(stats_two_interruptions.network.expand_rate, 0);
+  EXPECT_EQ(stats_two_interruptions.lifetime.total_interruption_duration_ms,
+            5000);
+  EXPECT_EQ(stats_two_interruptions.lifetime.interruption_count, 2);
+}
+
+// Checks that small losses do not produce interruptions.
+TEST(NetEqDecoderPlc, NoInterruptionsInSmallLosses) {
+  std::string checksum_1;
+  std::string checksum_4;
+
+  auto stats_1 = RunTest(300, 1, &checksum_1);
+  auto stats_4 = RunTest(300, 4, &checksum_4);
+
+  EXPECT_EQ(checksum_1, checksum_4);
+
+  EXPECT_EQ(stats_1.lifetime.interruption_count, 0);
+  EXPECT_EQ(stats_1.lifetime.total_interruption_duration_ms, 0);
+  EXPECT_EQ(stats_1.lifetime.concealed_samples, 640u);  // 20ms of concealment.
+  EXPECT_EQ(stats_1.lifetime.concealment_events, 1u);   // in just one event.
+
+  EXPECT_EQ(stats_4.lifetime.interruption_count, 0);
+  EXPECT_EQ(stats_4.lifetime.total_interruption_duration_ms, 0);
+  EXPECT_EQ(stats_4.lifetime.concealed_samples, 2560u);  // 80ms of concealment.
+  EXPECT_EQ(stats_4.lifetime.concealment_events, 1u);    // in just one event.
+}
+
+// Checks that interruptions of different sizes report correct duration.
+TEST(NetEqDecoderPlc, InterruptionsReportCorrectSize) {
+  std::string checksum;
+
+  for (int burst_length = 5; burst_length < 10; burst_length++) {
+    auto stats = RunTest(300, burst_length, &checksum);
+    auto duration = stats.lifetime.total_interruption_duration_ms;
+    if (burst_length < 8) {
+      EXPECT_EQ(duration, 0);
+    } else {
+      EXPECT_EQ(duration, burst_length * 20);
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc
new file mode 100644
index 0000000000..6a6367d045
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -0,0 +1,2141 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/neteq_impl.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstring>
+#include <list>
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/tick_timer.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/comfort_noise.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/merge.h"
+#include "modules/audio_coding/neteq/nack_tracker.h"
+#include "modules/audio_coding/neteq/normal.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "modules/audio_coding/neteq/post_decode_vad.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/time_stretch.h"
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/sanitizer.h"
+#include "rtc_base/strings/audio_format_to_string.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+
+std::unique_ptr<NetEqController> CreateNetEqController(
+    const NetEqControllerFactory& controller_factory,
+    int base_min_delay,
+    int max_packets_in_buffer,
+    bool allow_time_stretching,
+    TickTimer* tick_timer,
+    webrtc::Clock* clock) {
+  NetEqController::Config config;
+  config.base_min_delay_ms = base_min_delay;
+  config.max_packets_in_buffer = max_packets_in_buffer;
+  config.allow_time_stretching = allow_time_stretching;
+  config.tick_timer = tick_timer;
+  config.clock = clock;
+  return controller_factory.CreateNetEqController(config);
+}
+
+}  // namespace
+
+NetEqImpl::Dependencies::Dependencies(
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+    const NetEqControllerFactory& controller_factory)
+    : clock(clock),
+      tick_timer(new TickTimer),
+      stats(new StatisticsCalculator),
+      decoder_database(
+          new DecoderDatabase(decoder_factory, config.codec_pair_id)),
+      dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),
+      dtmf_tone_generator(new DtmfToneGenerator),
+      packet_buffer(
+          new PacketBuffer(config.max_packets_in_buffer, tick_timer.get())),
+      neteq_controller(
+          CreateNetEqController(controller_factory,
+                                config.min_delay_ms,
+                                config.max_packets_in_buffer,
+                                !config.for_test_no_time_stretching,
+                                tick_timer.get(),
+                                clock)),
+      red_payload_splitter(new RedPayloadSplitter),
+      timestamp_scaler(new TimestampScaler(*decoder_database)),
+      accelerate_factory(new AccelerateFactory),
+      expand_factory(new ExpandFactory),
+      preemptive_expand_factory(new PreemptiveExpandFactory) {}
+
+NetEqImpl::Dependencies::~Dependencies() = default;
+
+NetEqImpl::NetEqImpl(const NetEq::Config& config,
+                     Dependencies&& deps,
+                     bool create_components)
+    : clock_(deps.clock),
+      tick_timer_(std::move(deps.tick_timer)),
+      decoder_database_(std::move(deps.decoder_database)),
+      dtmf_buffer_(std::move(deps.dtmf_buffer)),
+      dtmf_tone_generator_(std::move(deps.dtmf_tone_generator)),
+      packet_buffer_(std::move(deps.packet_buffer)),
+      red_payload_splitter_(std::move(deps.red_payload_splitter)),
+      timestamp_scaler_(std::move(deps.timestamp_scaler)),
+      vad_(new PostDecodeVad()),
+      expand_factory_(std::move(deps.expand_factory)),
+      accelerate_factory_(std::move(deps.accelerate_factory)),
+      preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)),
+      stats_(std::move(deps.stats)),
+      controller_(std::move(deps.neteq_controller)),
+      last_mode_(Mode::kNormal),
+      decoded_buffer_length_(kMaxFrameSize),
+      decoded_buffer_(new int16_t[decoded_buffer_length_]),
+      playout_timestamp_(0),
+      new_codec_(false),
+      timestamp_(0),
+      reset_decoder_(false),
+      first_packet_(true),
+      enable_fast_accelerate_(config.enable_fast_accelerate),
+      nack_enabled_(false),
+      enable_muted_state_(config.enable_muted_state),
+      expand_uma_logger_("WebRTC.Audio.ExpandRatePercent",
+                         10,  // Report once every 10 s.
+                         tick_timer_.get()),
+      speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent",
+                                10,  // Report once every 10 s.
+                                tick_timer_.get()),
+      no_time_stretching_(config.for_test_no_time_stretching) {
+  RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
+  int fs = config.sample_rate_hz;
+  if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
+    RTC_LOG(LS_ERROR) << "Sample rate " << fs
+                      << " Hz not supported. "
+                         "Changing to 8000 Hz.";
+    fs = 8000;
+  }
+  controller_->SetMaximumDelay(config.max_delay_ms);
+  fs_hz_ = fs;
+  fs_mult_ = fs / 8000;
+  last_output_sample_rate_hz_ = fs;
+  output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
+  controller_->SetSampleRate(fs_hz_, output_size_samples_);
+  decoder_frame_length_ = 2 * output_size_samples_;  // 20 ms.
+  if (create_components) {
+    SetSampleRateAndChannels(fs, 1);  // Default is 1 channel.
+  }
+  RTC_DCHECK(!vad_->enabled());
+  if (config.enable_post_decode_vad) {
+    vad_->Enable();
+  }
+}
+
+NetEqImpl::~NetEqImpl() = default;
+
+int NetEqImpl::InsertPacket(const RTPHeader& rtp_header,
+                            rtc::ArrayView<const uint8_t> payload) {
+  rtc::MsanCheckInitialized(payload);
+  TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket");
+  MutexLock lock(&mutex_);
+  if (InsertPacketInternal(rtp_header, payload) != 0) {
+    return kFail;
+  }
+  return kOK;
+}
+
+void NetEqImpl::InsertEmptyPacket(const RTPHeader& rtp_header) {
+  MutexLock lock(&mutex_);
+  if (nack_enabled_) {
+    nack_->UpdateLastReceivedPacket(rtp_header.sequenceNumber,
+                                    rtp_header.timestamp);
+  }
+  controller_->RegisterEmptyPacket();
+}
+
+namespace {
+void SetAudioFrameActivityAndType(bool vad_enabled,
+                                  NetEqImpl::OutputType type,
+                                  AudioFrame::VADActivity last_vad_activity,
+                                  AudioFrame* audio_frame) {
+  switch (type) {
+    case NetEqImpl::OutputType::kNormalSpeech: {
+      audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+      audio_frame->vad_activity_ = AudioFrame::kVadActive;
+      break;
+    }
+    case NetEqImpl::OutputType::kVadPassive: {
+      // This should only be reached if the VAD is enabled.
+      RTC_DCHECK(vad_enabled);
+      audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+      audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+      break;
+    }
+    case NetEqImpl::OutputType::kCNG: {
+      audio_frame->speech_type_ = AudioFrame::kCNG;
+      audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+      break;
+    }
+    case NetEqImpl::OutputType::kPLC: {
+      audio_frame->speech_type_ = AudioFrame::kPLC;
+      audio_frame->vad_activity_ = last_vad_activity;
+      break;
+    }
+    case NetEqImpl::OutputType::kPLCCNG: {
+      audio_frame->speech_type_ = AudioFrame::kPLCCNG;
+      audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+      break;
+    }
+    case NetEqImpl::OutputType::kCodecPLC: {
+      audio_frame->speech_type_ = AudioFrame::kCodecPLC;
+      audio_frame->vad_activity_ = last_vad_activity;
+      break;
+    }
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  if (!vad_enabled) {
+    // Always set kVadUnknown when receive VAD is inactive.
+    audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
+  }
+}
+}  // namespace
+
+int NetEqImpl::GetAudio(AudioFrame* audio_frame,
+                        bool* muted,
+                        int* current_sample_rate_hz,
+                        absl::optional<Operation> action_override) {
+  TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
+  MutexLock lock(&mutex_);
+  if (GetAudioInternal(audio_frame, muted, action_override) != 0) {
+    return kFail;
+  }
+  RTC_DCHECK_EQ(
+      audio_frame->sample_rate_hz_,
+      rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
+  RTC_DCHECK_EQ(*muted, audio_frame->muted());
+  SetAudioFrameActivityAndType(vad_->enabled(), LastOutputType(),
+                               last_vad_activity_, audio_frame);
+  last_vad_activity_ = audio_frame->vad_activity_;
+  last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_;
+  RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
+             last_output_sample_rate_hz_ == 16000 ||
+             last_output_sample_rate_hz_ == 32000 ||
+             last_output_sample_rate_hz_ == 48000)
+      << "Unexpected sample rate " << last_output_sample_rate_hz_;
+
+  if (current_sample_rate_hz) {
+    *current_sample_rate_hz = last_output_sample_rate_hz_;
+  }
+
+  return kOK;
+}
+
+void NetEqImpl::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+  MutexLock lock(&mutex_);
+  const std::vector<int> changed_payload_types =
+      decoder_database_->SetCodecs(codecs);
+  for (const int pt : changed_payload_types) {
+    packet_buffer_->DiscardPacketsWithPayloadType(pt, stats_.get());
+  }
+}
+
+bool NetEqImpl::RegisterPayloadType(int rtp_payload_type,
+                                    const SdpAudioFormat& audio_format) {
+  RTC_LOG(LS_VERBOSE) << "NetEqImpl::RegisterPayloadType: payload type "
+                      << rtp_payload_type << ", codec "
+                      << rtc::ToString(audio_format);
+  MutexLock lock(&mutex_);
+  return decoder_database_->RegisterPayload(rtp_payload_type, audio_format) ==
+         DecoderDatabase::kOK;
+}
+
+int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
+  MutexLock lock(&mutex_);
+  int ret = decoder_database_->Remove(rtp_payload_type);
+  if (ret == DecoderDatabase::kOK || ret == DecoderDatabase::kDecoderNotFound) {
+    packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type,
+                                                  stats_.get());
+    return kOK;
+  }
+  return kFail;
+}
+
+void NetEqImpl::RemoveAllPayloadTypes() {
+  MutexLock lock(&mutex_);
+  decoder_database_->RemoveAll();
+}
+
+bool NetEqImpl::SetMinimumDelay(int delay_ms) {
+  MutexLock lock(&mutex_);
+  if (delay_ms >= 0 && delay_ms <= 10000) {
+    RTC_DCHECK(controller_.get());
+    return controller_->SetMinimumDelay(delay_ms);
+  }
+  return false;
+}
+
+bool NetEqImpl::SetMaximumDelay(int delay_ms) {
+  MutexLock lock(&mutex_);
+  if (delay_ms >= 0 && delay_ms <= 10000) {
+    RTC_DCHECK(controller_.get());
+    return controller_->SetMaximumDelay(delay_ms);
+  }
+  return false;
+}
+
+bool NetEqImpl::SetBaseMinimumDelayMs(int delay_ms) {
+  MutexLock lock(&mutex_);
+  if (delay_ms >= 0 && delay_ms <= 10000) {
+    return controller_->SetBaseMinimumDelay(delay_ms);
+  }
+  return false;
+}
+
+int NetEqImpl::GetBaseMinimumDelayMs() const {
+  MutexLock lock(&mutex_);
+  return controller_->GetBaseMinimumDelay();
+}
+
+int NetEqImpl::TargetDelayMs() const {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(controller_.get());
+  return controller_->TargetLevelMs();
+}
+
+int NetEqImpl::FilteredCurrentDelayMs() const {
+  MutexLock lock(&mutex_);
+  // Sum up the filtered packet buffer level with the future length of the sync
+  // buffer.
+  const int delay_samples =
+      controller_->GetFilteredBufferLevel() + sync_buffer_->FutureLength();
+  // The division below will truncate. The return value is in ms.
+  return delay_samples / rtc::CheckedDivExact(fs_hz_, 1000);
+}
+
+int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(decoder_database_.get());
+  *stats = CurrentNetworkStatisticsInternal();
+  stats_->GetNetworkStatistics(decoder_frame_length_, stats);
+  return 0;
+}
+
+NetEqNetworkStatistics NetEqImpl::CurrentNetworkStatistics() const {
+  MutexLock lock(&mutex_);
+  return CurrentNetworkStatisticsInternal();
+}
+
+NetEqNetworkStatistics NetEqImpl::CurrentNetworkStatisticsInternal() const {
+  RTC_DCHECK(decoder_database_.get());
+  NetEqNetworkStatistics stats;
+  const size_t total_samples_in_buffers =
+      packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) +
+      sync_buffer_->FutureLength();
+
+  RTC_DCHECK(controller_.get());
+  stats.preferred_buffer_size_ms = controller_->TargetLevelMs();
+  stats.jitter_peaks_found = controller_->PeakFound();
+  RTC_DCHECK_GT(fs_hz_, 0);
+  stats.current_buffer_size_ms =
+      static_cast<uint16_t>(total_samples_in_buffers * 1000 / fs_hz_);
+  return stats;
+}
+
+NetEqLifetimeStatistics NetEqImpl::GetLifetimeStatistics() const {
+  MutexLock lock(&mutex_);
+  return stats_->GetLifetimeStatistics();
+}
+
+NetEqOperationsAndState NetEqImpl::GetOperationsAndState() const {
+  MutexLock lock(&mutex_);
+  auto result = stats_->GetOperationsAndState();
+  result.current_buffer_size_ms =
+      (packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) +
+       sync_buffer_->FutureLength()) *
+      1000 / fs_hz_;
+  result.current_frame_size_ms = decoder_frame_length_ * 1000 / fs_hz_;
+  result.next_packet_available = packet_buffer_->PeekNextPacket() &&
+                                 packet_buffer_->PeekNextPacket()->timestamp ==
+                                     sync_buffer_->end_timestamp();
+  return result;
+}
+
+void NetEqImpl::EnableVad() {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(vad_.get());
+  vad_->Enable();
+}
+
+void NetEqImpl::DisableVad() {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(vad_.get());
+  vad_->Disable();
+}
+
+absl::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
+  MutexLock lock(&mutex_);
+  if (first_packet_ || last_mode_ == Mode::kRfc3389Cng ||
+      last_mode_ == Mode::kCodecInternalCng) {
+    // We don't have a valid RTP timestamp until we have decoded our first
+    // RTP packet. Also, the RTP timestamp is not accurate while playing CNG,
+    // which is indicated by returning an empty value.
+    return absl::nullopt;
+  }
+  return timestamp_scaler_->ToExternal(playout_timestamp_);
+}
+
+int NetEqImpl::last_output_sample_rate_hz() const {
+  MutexLock lock(&mutex_);
+  return last_output_sample_rate_hz_;
+}
+
+absl::optional<NetEq::DecoderFormat> NetEqImpl::GetDecoderFormat(
+    int payload_type) const {
+  MutexLock lock(&mutex_);
+  const DecoderDatabase::DecoderInfo* const di =
+      decoder_database_->GetDecoderInfo(payload_type);
+  if (di) {
+    const AudioDecoder* const decoder = di->GetDecoder();
+    // TODO(kwiberg): Why the special case for RED?
+    return DecoderFormat{
+        /*sample_rate_hz=*/di->IsRed() ? 8000 : di->SampleRateHz(),
+        /*num_channels=*/
+        decoder ? rtc::dchecked_cast<int>(decoder->Channels()) : 1,
+        /*sdp_format=*/di->GetFormat()};
+  } else {
+    // Payload type not registered.
+    return absl::nullopt;
+  }
+}
+
+void NetEqImpl::FlushBuffers() {
+  MutexLock lock(&mutex_);
+  RTC_LOG(LS_VERBOSE) << "FlushBuffers";
+  packet_buffer_->Flush(stats_.get());
+  RTC_DCHECK(sync_buffer_.get());
+  RTC_DCHECK(expand_.get());
+  sync_buffer_->Flush();
+  sync_buffer_->set_next_index(sync_buffer_->next_index() -
+                               expand_->overlap_length());
+  // Set to wait for new codec.
+  first_packet_ = true;
+}
+
+void NetEqImpl::EnableNack(size_t max_nack_list_size) {
+  MutexLock lock(&mutex_);
+  if (!nack_enabled_) {
+    nack_ = std::make_unique<NackTracker>();
+    nack_enabled_ = true;
+    nack_->UpdateSampleRate(fs_hz_);
+  }
+  nack_->SetMaxNackListSize(max_nack_list_size);
+}
+
+void NetEqImpl::DisableNack() {
+  MutexLock lock(&mutex_);
+  nack_.reset();
+  nack_enabled_ = false;
+}
+
+std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const {
+  MutexLock lock(&mutex_);
+  if (!nack_enabled_) {
+    return std::vector<uint16_t>();
+  }
+  RTC_DCHECK(nack_.get());
+  return nack_->GetNackList(round_trip_time_ms);
+}
+
+int NetEqImpl::SyncBufferSizeMs() const {
+  MutexLock lock(&mutex_);
+  return rtc::dchecked_cast<int>(sync_buffer_->FutureLength() /
+                                 rtc::CheckedDivExact(fs_hz_, 1000));
+}
+
+const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
+  MutexLock lock(&mutex_);
+  return sync_buffer_.get();
+}
+
+NetEq::Operation NetEqImpl::last_operation_for_test() const {
+  MutexLock lock(&mutex_);
+  return last_operation_;
+}
+
+// Methods below this line are private.
+
+int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
+                                    rtc::ArrayView<const uint8_t> payload) {
+  if (payload.empty()) {
+    RTC_LOG_F(LS_ERROR) << "payload is empty";
+    return kInvalidPointer;
+  }
+
+  Timestamp receive_time = clock_->CurrentTime();
+  stats_->ReceivedPacket();
+
+  PacketList packet_list;
+  // Insert packet in a packet list.
+  packet_list.push_back([&rtp_header, &payload, &receive_time] {
+    // Convert to Packet.
+    Packet packet;
+    packet.payload_type = rtp_header.payloadType;
+    packet.sequence_number = rtp_header.sequenceNumber;
+    packet.timestamp = rtp_header.timestamp;
+    packet.payload.SetData(payload.data(), payload.size());
+    packet.packet_info = RtpPacketInfo(rtp_header, receive_time);
+    // Waiting time will be set upon inserting the packet in the buffer.
+    RTC_DCHECK(!packet.waiting_time);
+    return packet;
+  }());
+
+  bool update_sample_rate_and_channels = first_packet_;
+
+  if (update_sample_rate_and_channels) {
+    // Reset timestamp scaling.
+    timestamp_scaler_->Reset();
+  }
+
+  if (!decoder_database_->IsRed(rtp_header.payloadType)) {
+    // Scale timestamp to internal domain (only for some codecs).
+    timestamp_scaler_->ToInternal(&packet_list);
+  }
+
+  // Store these for later use, since the first packet may very well disappear
+  // before we need these values.
+  uint32_t main_timestamp = packet_list.front().timestamp;
+  uint8_t main_payload_type = packet_list.front().payload_type;
+  uint16_t main_sequence_number = packet_list.front().sequence_number;
+
+  // Reinitialize NetEq if it's needed (changed SSRC or first call).
+  if (update_sample_rate_and_channels) {
+    // Note: `first_packet_` will be cleared further down in this method, once
+    // the packet has been successfully inserted into the packet buffer.
+
+    // Flush the packet buffer and DTMF buffer.
+    packet_buffer_->Flush(stats_.get());
+    dtmf_buffer_->Flush();
+
+    // Update audio buffer timestamp.
+    sync_buffer_->IncreaseEndTimestamp(main_timestamp - timestamp_);
+
+    // Update codecs.
+    timestamp_ = main_timestamp;
+  }
+
+  if (nack_enabled_) {
+    RTC_DCHECK(nack_);
+    if (update_sample_rate_and_channels) {
+      nack_->Reset();
+    }
+    nack_->UpdateLastReceivedPacket(main_sequence_number, main_timestamp);
+  }
+
+  // Check for RED payload type, and separate payloads into several packets.
+  if (decoder_database_->IsRed(rtp_header.payloadType)) {
+    if (!red_payload_splitter_->SplitRed(&packet_list)) {
+      return kRedundancySplitError;
+    }
+    // Only accept a few RED payloads of the same type as the main data,
+    // DTMF events and CNG.
+    red_payload_splitter_->CheckRedPayloads(&packet_list, *decoder_database_);
+    if (packet_list.empty()) {
+      return kRedundancySplitError;
+    }
+  }
+
+  // Check payload types.
+  if (decoder_database_->CheckPayloadTypes(packet_list) ==
+      DecoderDatabase::kDecoderNotFound) {
+    return kUnknownRtpPayloadType;
+  }
+
+  RTC_DCHECK(!packet_list.empty());
+
+  // Update main_timestamp, if new packets appear in the list
+  // after RED splitting.
+  if (decoder_database_->IsRed(rtp_header.payloadType)) {
+    timestamp_scaler_->ToInternal(&packet_list);
+    main_timestamp = packet_list.front().timestamp;
+    main_payload_type = packet_list.front().payload_type;
+    main_sequence_number = packet_list.front().sequence_number;
+  }
+
+  // Process DTMF payloads. Cycle through the list of packets, and pick out any
+  // DTMF payloads found.
+  PacketList::iterator it = packet_list.begin();
+  while (it != packet_list.end()) {
+    const Packet& current_packet = (*it);
+    RTC_DCHECK(!current_packet.payload.empty());
+    if (decoder_database_->IsDtmf(current_packet.payload_type)) {
+      DtmfEvent event;
+      int ret = DtmfBuffer::ParseEvent(current_packet.timestamp,
+                                       current_packet.payload.data(),
+                                       current_packet.payload.size(), &event);
+      if (ret != DtmfBuffer::kOK) {
+        return kDtmfParsingError;
+      }
+      if (dtmf_buffer_->InsertEvent(event) != DtmfBuffer::kOK) {
+        return kDtmfInsertError;
+      }
+      it = packet_list.erase(it);
+    } else {
+      ++it;
+    }
+  }
+
+  PacketList parsed_packet_list;
+  bool is_dtx = false;
+  while (!packet_list.empty()) {
+    Packet& packet = packet_list.front();
+    const DecoderDatabase::DecoderInfo* info =
+        decoder_database_->GetDecoderInfo(packet.payload_type);
+    if (!info) {
+      RTC_LOG(LS_WARNING) << "SplitAudio unknown payload type";
+      return kUnknownRtpPayloadType;
+    }
+
+    if (info->IsComfortNoise()) {
+      // Carry comfort noise packets along.
+      parsed_packet_list.splice(parsed_packet_list.end(), packet_list,
+                                packet_list.begin());
+    } else {
+      const auto sequence_number = packet.sequence_number;
+      const auto payload_type = packet.payload_type;
+      const Packet::Priority original_priority = packet.priority;
+      const auto& packet_info = packet.packet_info;
+      auto packet_from_result = [&](AudioDecoder::ParseResult& result) {
+        Packet new_packet;
+        new_packet.sequence_number = sequence_number;
+        new_packet.payload_type = payload_type;
+        new_packet.timestamp = result.timestamp;
+        new_packet.priority.codec_level = result.priority;
+        new_packet.priority.red_level = original_priority.red_level;
+        new_packet.packet_info = packet_info;
+        new_packet.frame = std::move(result.frame);
+        return new_packet;
+      };
+
+      std::vector<AudioDecoder::ParseResult> results =
+          info->GetDecoder()->ParsePayload(std::move(packet.payload),
+                                           packet.timestamp);
+      if (results.empty()) {
+        packet_list.pop_front();
+      } else {
+        bool first = true;
+        for (auto& result : results) {
+          RTC_DCHECK(result.frame);
+          RTC_DCHECK_GE(result.priority, 0);
+          is_dtx = is_dtx || result.frame->IsDtxPacket();
+          if (first) {
+            // Re-use the node and move it to parsed_packet_list.
+            packet_list.front() = packet_from_result(result);
+            parsed_packet_list.splice(parsed_packet_list.end(), packet_list,
+                                      packet_list.begin());
+            first = false;
+          } else {
+            parsed_packet_list.push_back(packet_from_result(result));
+          }
+        }
+      }
+    }
+  }
+
+  // Calculate the number of primary (non-FEC/RED) packets.
+  const size_t number_of_primary_packets = std::count_if(
+      parsed_packet_list.begin(), parsed_packet_list.end(),
+      [](const Packet& in) { return in.priority.codec_level == 0; });
+  if (number_of_primary_packets < parsed_packet_list.size()) {
+    stats_->SecondaryPacketsReceived(parsed_packet_list.size() -
+                                     number_of_primary_packets);
+  }
+
+  // Insert packets in buffer.
+  const int target_level_ms = controller_->TargetLevelMs();
+  const int ret = packet_buffer_->InsertPacketList(
+      &parsed_packet_list, *decoder_database_, &current_rtp_payload_type_,
+      &current_cng_rtp_payload_type_, stats_.get(), decoder_frame_length_,
+      last_output_sample_rate_hz_, target_level_ms);
+  bool buffer_flush_occured = false;
+  if (ret == PacketBuffer::kFlushed) {
+    // Reset DSP timestamp etc. if packet buffer flushed.
+    new_codec_ = true;
+    update_sample_rate_and_channels = true;
+    buffer_flush_occured = true;
+  } else if (ret == PacketBuffer::kPartialFlush) {
+    // Forward sync buffer timestamp
+    timestamp_ = packet_buffer_->PeekNextPacket()->timestamp;
+    sync_buffer_->IncreaseEndTimestamp(timestamp_ -
+                                       sync_buffer_->end_timestamp());
+    buffer_flush_occured = true;
+  } else if (ret != PacketBuffer::kOK) {
+    return kOtherError;
+  }
+
+  if (first_packet_) {
+    first_packet_ = false;
+    // Update the codec on the next GetAudio call.
+    new_codec_ = true;
+  }
+
+  if (current_rtp_payload_type_) {
+    RTC_DCHECK(decoder_database_->GetDecoderInfo(*current_rtp_payload_type_))
+        << "Payload type " << static_cast<int>(*current_rtp_payload_type_)
+        << " is unknown where it shouldn't be";
+  }
+
+  if (update_sample_rate_and_channels && !packet_buffer_->Empty()) {
+    // We do not use `current_rtp_payload_type_` to |set payload_type|, but
+    // get the next RTP header from `packet_buffer_` to obtain the payload type.
+    // The reason for it is the following corner case. If NetEq receives a
+    // CNG packet with a sample rate different than the current CNG then it
+    // flushes its buffer, assuming send codec must have been changed. However,
+    // payload type of the hypothetically new send codec is not known.
+    const Packet* next_packet = packet_buffer_->PeekNextPacket();
+    RTC_DCHECK(next_packet);
+    const int payload_type = next_packet->payload_type;
+    size_t channels = 1;
+    if (!decoder_database_->IsComfortNoise(payload_type)) {
+      AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type);
+      RTC_DCHECK(decoder);  // Payloads are already checked to be valid.
+      channels = decoder->Channels();
+    }
+    const DecoderDatabase::DecoderInfo* decoder_info =
+        decoder_database_->GetDecoderInfo(payload_type);
+    RTC_DCHECK(decoder_info);
+    if (decoder_info->SampleRateHz() != fs_hz_ ||
+        channels != algorithm_buffer_->Channels()) {
+      SetSampleRateAndChannels(decoder_info->SampleRateHz(), channels);
+    }
+    if (nack_enabled_) {
+      RTC_DCHECK(nack_);
+      // Update the sample rate even if the rate is not new, because of Reset().
+      nack_->UpdateSampleRate(fs_hz_);
+    }
+  }
+
+  const DecoderDatabase::DecoderInfo* dec_info =
+      decoder_database_->GetDecoderInfo(main_payload_type);
+  RTC_DCHECK(dec_info);  // Already checked that the payload type is known.
+
+  NetEqController::PacketArrivedInfo info;
+  info.is_cng_or_dtmf = dec_info->IsComfortNoise() || dec_info->IsDtmf();
+  info.packet_length_samples =
+      number_of_primary_packets * decoder_frame_length_;
+  info.main_timestamp = main_timestamp;
+  info.main_sequence_number = main_sequence_number;
+  info.is_dtx = is_dtx;
+  info.buffer_flush = buffer_flush_occured;
+
+  const bool should_update_stats = !new_codec_;
+  auto relative_delay =
+      controller_->PacketArrived(fs_hz_, should_update_stats, info);
+  if (relative_delay) {
+    stats_->RelativePacketArrivalDelay(relative_delay.value());
+  }
+  return 0;
+}
+
+int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
+                                bool* muted,
+                                absl::optional<Operation> action_override) {
+  PacketList packet_list;
+  DtmfEvent dtmf_event;
+  Operation operation;
+  bool play_dtmf;
+  *muted = false;
+  last_decoded_packet_infos_.clear();
+  tick_timer_->Increment();
+  stats_->IncreaseCounter(output_size_samples_, fs_hz_);
+  const auto lifetime_stats = stats_->GetLifetimeStatistics();
+  expand_uma_logger_.UpdateSampleCounter(lifetime_stats.concealed_samples,
+                                         fs_hz_);
+  speech_expand_uma_logger_.UpdateSampleCounter(
+      lifetime_stats.concealed_samples -
+          lifetime_stats.silent_concealed_samples,
+      fs_hz_);
+
+  // Check for muted state.
+  if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) {
+    RTC_DCHECK_EQ(last_mode_, Mode::kExpand);
+    audio_frame->Reset();
+    RTC_DCHECK(audio_frame->muted());  // Reset() should mute the frame.
+    playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
+    audio_frame->sample_rate_hz_ = fs_hz_;
+    // Make sure the total number of samples fits in the AudioFrame.
+    if (output_size_samples_ * sync_buffer_->Channels() >
+        AudioFrame::kMaxDataSizeSamples) {
+      return kSampleUnderrun;
+    }
+    audio_frame->samples_per_channel_ = output_size_samples_;
+    audio_frame->timestamp_ =
+        first_packet_
+            ? 0
+            : timestamp_scaler_->ToExternal(playout_timestamp_) -
+                  static_cast<uint32_t>(audio_frame->samples_per_channel_);
+    audio_frame->num_channels_ = sync_buffer_->Channels();
+    stats_->ExpandedNoiseSamples(output_size_samples_, false);
+    controller_->NotifyMutedState();
+    *muted = true;
+    return 0;
+  }
+  int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
+                                 &play_dtmf, action_override);
+  if (return_value != 0) {
+    last_mode_ = Mode::kError;
+    return return_value;
+  }
+
+  AudioDecoder::SpeechType speech_type;
+  int length = 0;
+  const size_t start_num_packets = packet_list.size();
+  int decode_return_value =
+      Decode(&packet_list, &operation, &length, &speech_type);
+
+  RTC_DCHECK(vad_.get());
+  bool sid_frame_available =
+      (operation == Operation::kRfc3389Cng && !packet_list.empty());
+  vad_->Update(decoded_buffer_.get(), static_cast<size_t>(length), speech_type,
+               sid_frame_available, fs_hz_);
+
+  // This is the criterion that we did decode some data through the speech
+  // decoder, and the operation resulted in comfort noise.
+  const bool codec_internal_sid_frame =
+      (speech_type == AudioDecoder::kComfortNoise &&
+       start_num_packets > packet_list.size());
+
+  if (sid_frame_available || codec_internal_sid_frame) {
+    // Start a new stopwatch since we are decoding a new CNG packet.
+    generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch();
+  }
+
+  algorithm_buffer_->Clear();
+  switch (operation) {
+    case Operation::kNormal: {
+      DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf);
+      if (length > 0) {
+        stats_->DecodedOutputPlayed();
+      }
+      break;
+    }
+    case Operation::kMerge: {
+      DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf);
+      break;
+    }
+    case Operation::kExpand: {
+      RTC_DCHECK_EQ(return_value, 0);
+      if (!current_rtp_payload_type_ || !DoCodecPlc()) {
+        return_value = DoExpand(play_dtmf);
+      }
+      RTC_DCHECK_GE(sync_buffer_->FutureLength() - expand_->overlap_length(),
+                    output_size_samples_);
+      break;
+    }
+    case Operation::kAccelerate:
+    case Operation::kFastAccelerate: {
+      const bool fast_accelerate =
+          enable_fast_accelerate_ && (operation == Operation::kFastAccelerate);
+      return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type,
+                                  play_dtmf, fast_accelerate);
+      break;
+    }
+    case Operation::kPreemptiveExpand: {
+      return_value = DoPreemptiveExpand(decoded_buffer_.get(), length,
+                                        speech_type, play_dtmf);
+      break;
+    }
+    case Operation::kRfc3389Cng:
+    case Operation::kRfc3389CngNoPacket: {
+      return_value = DoRfc3389Cng(&packet_list, play_dtmf);
+      break;
+    }
+    case Operation::kCodecInternalCng: {
+      // This handles the case when there is no transmission and the decoder
+      // should produce internal comfort noise.
+      // TODO(hlundin): Write test for codec-internal CNG.
+      DoCodecInternalCng(decoded_buffer_.get(), length);
+      break;
+    }
+    case Operation::kDtmf: {
+      // TODO(hlundin): Write test for this.
+      return_value = DoDtmf(dtmf_event, &play_dtmf);
+      break;
+    }
+    case Operation::kUndefined: {
+      RTC_LOG(LS_ERROR) << "Invalid operation kUndefined.";
+      RTC_DCHECK_NOTREACHED();  // This should not happen.
+      last_mode_ = Mode::kError;
+      return kInvalidOperation;
+    }
+  }  // End of switch.
+  last_operation_ = operation;
+  if (return_value < 0) {
+    return return_value;
+  }
+
+  if (last_mode_ != Mode::kRfc3389Cng) {
+    comfort_noise_->Reset();
+  }
+
+  // We treat it as if all packets referenced to by `last_decoded_packet_infos_`
+  // were mashed together when creating the samples in `algorithm_buffer_`.
+  RtpPacketInfos packet_infos(last_decoded_packet_infos_);
+
+  // Copy samples from `algorithm_buffer_` to `sync_buffer_`.
+  //
+  // TODO(bugs.webrtc.org/10757):
+  //   We would in the future also like to pass `packet_infos` so that we can do
+  //   sample-perfect tracking of that information across `sync_buffer_`.
+  sync_buffer_->PushBack(*algorithm_buffer_);
+
+  // Extract data from `sync_buffer_` to `output`.
+  size_t num_output_samples_per_channel = output_size_samples_;
+  size_t num_output_samples = output_size_samples_ * sync_buffer_->Channels();
+  if (num_output_samples > AudioFrame::kMaxDataSizeSamples) {
+    RTC_LOG(LS_WARNING) << "Output array is too short. "
+                        << AudioFrame::kMaxDataSizeSamples << " < "
+                        << output_size_samples_ << " * "
+                        << sync_buffer_->Channels();
+    num_output_samples = AudioFrame::kMaxDataSizeSamples;
+    num_output_samples_per_channel =
+        AudioFrame::kMaxDataSizeSamples / sync_buffer_->Channels();
+  }
+  sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
+                                        audio_frame);
+  audio_frame->sample_rate_hz_ = fs_hz_;
+  // TODO(bugs.webrtc.org/10757):
+  //   We don't have the ability to properly track individual packets once their
+  //   audio samples have entered `sync_buffer_`. So for now, treat it as if
+  //   `packet_infos` from packets decoded by the current `GetAudioInternal()`
+  //   call were all consumed assembling the current audio frame and the current
+  //   audio frame only.
+  audio_frame->packet_infos_ = std::move(packet_infos);
+  if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
+    // The sync buffer should always contain `overlap_length` samples, but now
+    // too many samples have been extracted. Reinstall the `overlap_length`
+    // lookahead by moving the index.
+    const size_t missing_lookahead_samples =
+        expand_->overlap_length() - sync_buffer_->FutureLength();
+    RTC_DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples);
+    sync_buffer_->set_next_index(sync_buffer_->next_index() -
+                                 missing_lookahead_samples);
+  }
+  if (audio_frame->samples_per_channel_ != output_size_samples_) {
+    RTC_LOG(LS_ERROR) << "audio_frame->samples_per_channel_ ("
+                      << audio_frame->samples_per_channel_
+                      << ") != output_size_samples_ (" << output_size_samples_
+                      << ")";
+    // TODO(minyue): treatment of under-run, filling zeros
+    audio_frame->Mute();
+    return kSampleUnderrun;
+  }
+
+  // Should always have overlap samples left in the `sync_buffer_`.
+  RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
+
+  // TODO(yujo): For muted frames, this can be a copy rather than an addition.
+  if (play_dtmf) {
+    return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(),
+                               audio_frame->mutable_data());
+  }
+
+  // Update the background noise parameters if last operation wrote data
+  // straight from the decoder to the `sync_buffer_`. That is, none of the
+  // operations that modify the signal can be followed by a parameter update.
+  if ((last_mode_ == Mode::kNormal) || (last_mode_ == Mode::kAccelerateFail) ||
+      (last_mode_ == Mode::kPreemptiveExpandFail) ||
+      (last_mode_ == Mode::kRfc3389Cng) ||
+      (last_mode_ == Mode::kCodecInternalCng)) {
+    background_noise_->Update(*sync_buffer_, *vad_.get());
+  }
+
+  if (operation == Operation::kDtmf) {
+    // DTMF data was written the end of `sync_buffer_`.
+    // Update index to end of DTMF data in `sync_buffer_`.
+    sync_buffer_->set_dtmf_index(sync_buffer_->Size());
+  }
+
+  if (last_mode_ != Mode::kExpand && last_mode_ != Mode::kCodecPlc) {
+    // If last operation was not expand, calculate the `playout_timestamp_` from
+    // the `sync_buffer_`. However, do not update the `playout_timestamp_` if it
+    // would be moved "backwards".
+    uint32_t temp_timestamp =
+        sync_buffer_->end_timestamp() -
+        static_cast<uint32_t>(sync_buffer_->FutureLength());
+    if (static_cast<int32_t>(temp_timestamp - playout_timestamp_) > 0) {
+      playout_timestamp_ = temp_timestamp;
+    }
+  } else {
+    // Use dead reckoning to estimate the `playout_timestamp_`.
+    playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
+  }
+  // Set the timestamp in the audio frame to zero before the first packet has
+  // been inserted. Otherwise, subtract the frame size in samples to get the
+  // timestamp of the first sample in the frame (playout_timestamp_ is the
+  // last + 1).
+  audio_frame->timestamp_ =
+      first_packet_
+          ? 0
+          : timestamp_scaler_->ToExternal(playout_timestamp_) -
+                static_cast<uint32_t>(audio_frame->samples_per_channel_);
+
+  if (!(last_mode_ == Mode::kRfc3389Cng ||
+        last_mode_ == Mode::kCodecInternalCng || last_mode_ == Mode::kExpand ||
+        last_mode_ == Mode::kCodecPlc)) {
+    generated_noise_stopwatch_.reset();
+  }
+
+  if (decode_return_value)
+    return decode_return_value;
+  return return_value;
+}
+
+int NetEqImpl::GetDecision(Operation* operation,
+                           PacketList* packet_list,
+                           DtmfEvent* dtmf_event,
+                           bool* play_dtmf,
+                           absl::optional<Operation> action_override) {
+  // Initialize output variables.
+  *play_dtmf = false;
+  *operation = Operation::kUndefined;
+
+  RTC_DCHECK(sync_buffer_.get());
+  uint32_t end_timestamp = sync_buffer_->end_timestamp();
+  if (!new_codec_) {
+    const uint32_t five_seconds_samples = 5 * fs_hz_;
+    packet_buffer_->DiscardOldPackets(end_timestamp, five_seconds_samples,
+                                      stats_.get());
+  }
+  const Packet* packet = packet_buffer_->PeekNextPacket();
+
+  RTC_DCHECK(!generated_noise_stopwatch_ ||
+             generated_noise_stopwatch_->ElapsedTicks() >= 1);
+  uint64_t generated_noise_samples =
+      generated_noise_stopwatch_ ? (generated_noise_stopwatch_->ElapsedTicks() -
+                                    1) * output_size_samples_ +
+                                       controller_->noise_fast_forward()
+                                 : 0;
+
+  if (controller_->CngRfc3389On() || last_mode_ == Mode::kRfc3389Cng) {
+    // Because of timestamp peculiarities, we have to "manually" disallow using
+    // a CNG packet with the same timestamp as the one that was last played.
+    // This can happen when using redundancy and will cause the timing to shift.
+    while (packet && decoder_database_->IsComfortNoise(packet->payload_type) &&
+           (end_timestamp >= packet->timestamp ||
+            end_timestamp + generated_noise_samples > packet->timestamp)) {
+      // Don't use this packet, discard it.
+      if (packet_buffer_->DiscardNextPacket(stats_.get()) !=
+          PacketBuffer::kOK) {
+        RTC_DCHECK_NOTREACHED();  // Must be ok by design.
+      }
+      // Check buffer again.
+      if (!new_codec_) {
+        packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_,
+                                          stats_.get());
+      }
+      packet = packet_buffer_->PeekNextPacket();
+    }
+  }
+
+  RTC_DCHECK(expand_.get());
+  const int samples_left = static_cast<int>(sync_buffer_->FutureLength() -
+                                            expand_->overlap_length());
+  if (last_mode_ == Mode::kAccelerateSuccess ||
+      last_mode_ == Mode::kAccelerateLowEnergy ||
+      last_mode_ == Mode::kPreemptiveExpandSuccess ||
+      last_mode_ == Mode::kPreemptiveExpandLowEnergy) {
+    // Subtract (samples_left + output_size_samples_) from sampleMemory.
+    controller_->AddSampleMemory(
+        -(samples_left + rtc::dchecked_cast<int>(output_size_samples_)));
+  }
+
+  // Check if it is time to play a DTMF event.
+  if (dtmf_buffer_->GetEvent(
+          static_cast<uint32_t>(end_timestamp + generated_noise_samples),
+          dtmf_event)) {
+    *play_dtmf = true;
+  }
+
+  // Get instruction.
+  RTC_DCHECK(sync_buffer_.get());
+  RTC_DCHECK(expand_.get());
+  generated_noise_samples =
+      generated_noise_stopwatch_
+          ? generated_noise_stopwatch_->ElapsedTicks() * output_size_samples_ +
+                controller_->noise_fast_forward()
+          : 0;
+  NetEqController::NetEqStatus status;
+  status.packet_buffer_info.dtx_or_cng =
+      packet_buffer_->ContainsDtxOrCngPacket(decoder_database_.get());
+  status.packet_buffer_info.num_samples =
+      packet_buffer_->NumSamplesInBuffer(decoder_frame_length_);
+  status.packet_buffer_info.span_samples = packet_buffer_->GetSpanSamples(
+      decoder_frame_length_, last_output_sample_rate_hz_, true);
+  status.packet_buffer_info.span_samples_no_dtx =
+      packet_buffer_->GetSpanSamples(decoder_frame_length_,
+                                     last_output_sample_rate_hz_, false);
+  status.packet_buffer_info.num_packets = packet_buffer_->NumPacketsInBuffer();
+  status.target_timestamp = sync_buffer_->end_timestamp();
+  status.expand_mutefactor = expand_->MuteFactor(0);
+  status.last_packet_samples = decoder_frame_length_;
+  status.last_mode = last_mode_;
+  status.play_dtmf = *play_dtmf;
+  status.generated_noise_samples = generated_noise_samples;
+  status.sync_buffer_samples = sync_buffer_->FutureLength();
+  if (packet) {
+    status.next_packet = {
+        packet->timestamp, packet->frame && packet->frame->IsDtxPacket(),
+        decoder_database_->IsComfortNoise(packet->payload_type)};
+  }
+  *operation = controller_->GetDecision(status, &reset_decoder_);
+
+  // Disallow time stretching if this packet is DTX, because such a decision may
+  // be based on earlier buffer level estimate, as we do not update buffer level
+  // during DTX. When we have a better way to update buffer level during DTX,
+  // this can be discarded.
+  if (packet && packet->frame && packet->frame->IsDtxPacket() &&
+      (*operation == Operation::kMerge ||
+       *operation == Operation::kAccelerate ||
+       *operation == Operation::kFastAccelerate ||
+       *operation == Operation::kPreemptiveExpand)) {
+    *operation = Operation::kNormal;
+  }
+
+  if (action_override) {
+    // Use the provided action instead of the decision NetEq decided on.
+    *operation = *action_override;
+  }
+  // Check if we already have enough samples in the `sync_buffer_`. If so,
+  // change decision to normal, unless the decision was merge, accelerate, or
+  // preemptive expand.
+  if (samples_left >= rtc::dchecked_cast<int>(output_size_samples_) &&
+      *operation != Operation::kMerge && *operation != Operation::kAccelerate &&
+      *operation != Operation::kFastAccelerate &&
+      *operation != Operation::kPreemptiveExpand) {
+    *operation = Operation::kNormal;
+    return 0;
+  }
+
+  controller_->ExpandDecision(*operation);
+  if ((last_mode_ == Mode::kCodecPlc) && (*operation != Operation::kExpand)) {
+    // Getting out of the PLC expand mode, reporting interruptions.
+    // NetEq PLC reports this metrics in expand.cc
+    stats_->EndExpandEvent(fs_hz_);
+  }
+
+  // Check conditions for reset.
+  if (new_codec_ || *operation == Operation::kUndefined) {
+    // The only valid reason to get kUndefined is that new_codec_ is set.
+    RTC_DCHECK(new_codec_);
+    if (*play_dtmf && !packet) {
+      timestamp_ = dtmf_event->timestamp;
+    } else {
+      if (!packet) {
+        RTC_LOG(LS_ERROR) << "Packet missing where it shouldn't.";
+        return -1;
+      }
+      timestamp_ = packet->timestamp;
+      if (*operation == Operation::kRfc3389CngNoPacket &&
+          decoder_database_->IsComfortNoise(packet->payload_type)) {
+        // Change decision to CNG packet, since we do have a CNG packet, but it
+        // was considered too early to use. Now, use it anyway.
+        *operation = Operation::kRfc3389Cng;
+      } else if (*operation != Operation::kRfc3389Cng) {
+        *operation = Operation::kNormal;
+      }
+    }
+    // Adjust `sync_buffer_` timestamp before setting `end_timestamp` to the
+    // new value.
+    sync_buffer_->IncreaseEndTimestamp(timestamp_ - end_timestamp);
+    end_timestamp = timestamp_;
+    new_codec_ = false;
+    controller_->SoftReset();
+    stats_->ResetMcu();
+  }
+
+  size_t required_samples = output_size_samples_;
+  const size_t samples_10_ms = static_cast<size_t>(80 * fs_mult_);
+  const size_t samples_20_ms = 2 * samples_10_ms;
+  const size_t samples_30_ms = 3 * samples_10_ms;
+
+  switch (*operation) {
+    case Operation::kExpand: {
+      timestamp_ = end_timestamp;
+      return 0;
+    }
+    case Operation::kRfc3389CngNoPacket:
+    case Operation::kCodecInternalCng: {
+      return 0;
+    }
+    case Operation::kDtmf: {
+      // TODO(hlundin): Write test for this.
+      // Update timestamp.
+      timestamp_ = end_timestamp;
+      const uint64_t generated_noise_samples =
+          generated_noise_stopwatch_
+              ? generated_noise_stopwatch_->ElapsedTicks() *
+                        output_size_samples_ +
+                    controller_->noise_fast_forward()
+              : 0;
+      if (generated_noise_samples > 0 && last_mode_ != Mode::kDtmf) {
+        // Make a jump in timestamp due to the recently played comfort noise.
+        uint32_t timestamp_jump =
+            static_cast<uint32_t>(generated_noise_samples);
+        sync_buffer_->IncreaseEndTimestamp(timestamp_jump);
+        timestamp_ += timestamp_jump;
+      }
+      return 0;
+    }
+    case Operation::kAccelerate:
+    case Operation::kFastAccelerate: {
+      // In order to do an accelerate we need at least 30 ms of audio data.
+      if (samples_left >= static_cast<int>(samples_30_ms)) {
+        // Already have enough data, so we do not need to extract any more.
+        controller_->set_sample_memory(samples_left);
+        controller_->set_prev_time_scale(true);
+        return 0;
+      } else if (samples_left >= static_cast<int>(samples_10_ms) &&
+                 decoder_frame_length_ >= samples_30_ms) {
+        // Avoid decoding more data as it might overflow the playout buffer.
+        *operation = Operation::kNormal;
+        return 0;
+      } else if (samples_left < static_cast<int>(samples_20_ms) &&
+                 decoder_frame_length_ < samples_30_ms) {
+        // Build up decoded data by decoding at least 20 ms of audio data. Do
+        // not perform accelerate yet, but wait until we only need to do one
+        // decoding.
+        required_samples = 2 * output_size_samples_;
+        *operation = Operation::kNormal;
+      }
+      // If none of the above is true, we have one of two possible situations:
+      // (1) 20 ms <= samples_left < 30 ms and decoder_frame_length_ < 30 ms; or
+      // (2) samples_left < 10 ms and decoder_frame_length_ >= 30 ms.
+      // In either case, we move on with the accelerate decision, and decode one
+      // frame now.
+      break;
+    }
+    case Operation::kPreemptiveExpand: {
+      // In order to do a preemptive expand we need at least 30 ms of decoded
+      // audio data.
+      if ((samples_left >= static_cast<int>(samples_30_ms)) ||
+          (samples_left >= static_cast<int>(samples_10_ms) &&
+           decoder_frame_length_ >= samples_30_ms)) {
+        // Already have enough data, so we do not need to extract any more.
+        // Or, avoid decoding more data as it might overflow the playout buffer.
+        // Still try preemptive expand, though.
+        controller_->set_sample_memory(samples_left);
+        controller_->set_prev_time_scale(true);
+        return 0;
+      }
+      if (samples_left < static_cast<int>(samples_20_ms) &&
+          decoder_frame_length_ < samples_30_ms) {
+        // Build up decoded data by decoding at least 20 ms of audio data.
+        // Still try to perform preemptive expand.
+        required_samples = 2 * output_size_samples_;
+      }
+      // Move on with the preemptive expand decision.
+      break;
+    }
+    case Operation::kMerge: {
+      required_samples =
+          std::max(merge_->RequiredFutureSamples(), required_samples);
+      break;
+    }
+    default: {
+      // Do nothing.
+    }
+  }
+
+  // Get packets from buffer.
+  int extracted_samples = 0;
+  if (packet) {
+    sync_buffer_->IncreaseEndTimestamp(packet->timestamp - end_timestamp);
+
+    if (*operation != Operation::kRfc3389Cng) {
+      // We are about to decode and use a non-CNG packet.
+      controller_->SetCngOff();
+    }
+
+    extracted_samples = ExtractPackets(required_samples, packet_list);
+    if (extracted_samples < 0) {
+      return kPacketBufferCorruption;
+    }
+  }
+
+  if (*operation == Operation::kAccelerate ||
+      *operation == Operation::kFastAccelerate ||
+      *operation == Operation::kPreemptiveExpand) {
+    controller_->set_sample_memory(samples_left + extracted_samples);
+    controller_->set_prev_time_scale(true);
+  }
+
+  if (*operation == Operation::kAccelerate ||
+      *operation == Operation::kFastAccelerate) {
+    // Check that we have enough data (30ms) to do accelerate.
+    if (extracted_samples + samples_left < static_cast<int>(samples_30_ms)) {
+      // TODO(hlundin): Write test for this.
+      // Not enough, do normal operation instead.
+      *operation = Operation::kNormal;
+    }
+  }
+
+  timestamp_ = sync_buffer_->end_timestamp();
+  return 0;
+}
+
+int NetEqImpl::Decode(PacketList* packet_list,
+                      Operation* operation,
+                      int* decoded_length,
+                      AudioDecoder::SpeechType* speech_type) {
+  *speech_type = AudioDecoder::kSpeech;
+
+  // When packet_list is empty, we may be in kCodecInternalCng mode, and for
+  // that we use current active decoder.
+  AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
+
+  if (!packet_list->empty()) {
+    const Packet& packet = packet_list->front();
+    uint8_t payload_type = packet.payload_type;
+    if (!decoder_database_->IsComfortNoise(payload_type)) {
+      decoder = decoder_database_->GetDecoder(payload_type);
+      RTC_DCHECK(decoder);
+      if (!decoder) {
+        RTC_LOG(LS_WARNING)
+            << "Unknown payload type " << static_cast<int>(payload_type);
+        packet_list->clear();
+        return kDecoderNotFound;
+      }
+      bool decoder_changed;
+      decoder_database_->SetActiveDecoder(payload_type, &decoder_changed);
+      if (decoder_changed) {
+        // We have a new decoder. Re-init some values.
+        const DecoderDatabase::DecoderInfo* decoder_info =
+            decoder_database_->GetDecoderInfo(payload_type);
+        RTC_DCHECK(decoder_info);
+        if (!decoder_info) {
+          RTC_LOG(LS_WARNING)
+              << "Unknown payload type " << static_cast<int>(payload_type);
+          packet_list->clear();
+          return kDecoderNotFound;
+        }
+        // If sampling rate or number of channels has changed, we need to make
+        // a reset.
+        if (decoder_info->SampleRateHz() != fs_hz_ ||
+            decoder->Channels() != algorithm_buffer_->Channels()) {
+          // TODO(tlegrand): Add unittest to cover this event.
+          SetSampleRateAndChannels(decoder_info->SampleRateHz(),
+                                   decoder->Channels());
+        }
+        sync_buffer_->set_end_timestamp(timestamp_);
+        playout_timestamp_ = timestamp_;
+      }
+    }
+  }
+
+  if (reset_decoder_) {
+    // TODO(hlundin): Write test for this.
+    if (decoder)
+      decoder->Reset();
+
+    // Reset comfort noise decoder.
+    ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+    if (cng_decoder)
+      cng_decoder->Reset();
+
+    reset_decoder_ = false;
+  }
+
+  *decoded_length = 0;
+  // Update codec-internal PLC state.
+  if ((*operation == Operation::kMerge) && decoder && decoder->HasDecodePlc()) {
+    decoder->DecodePlc(1, &decoded_buffer_[*decoded_length]);
+  }
+
+  int return_value;
+  if (*operation == Operation::kCodecInternalCng) {
+    RTC_DCHECK(packet_list->empty());
+    return_value = DecodeCng(decoder, decoded_length, speech_type);
+  } else {
+    return_value = DecodeLoop(packet_list, *operation, decoder, decoded_length,
+                              speech_type);
+  }
+
+  if (*decoded_length < 0) {
+    // Error returned from the decoder.
+    *decoded_length = 0;
+    sync_buffer_->IncreaseEndTimestamp(
+        static_cast<uint32_t>(decoder_frame_length_));
+    int error_code = 0;
+    if (decoder)
+      error_code = decoder->ErrorCode();
+    if (error_code != 0) {
+      // Got some error code from the decoder.
+      return_value = kDecoderErrorCode;
+      RTC_LOG(LS_WARNING) << "Decoder returned error code: " << error_code;
+    } else {
+      // Decoder does not implement error codes. Return generic error.
+      return_value = kOtherDecoderError;
+      RTC_LOG(LS_WARNING) << "Decoder error (no error code)";
+    }
+    *operation = Operation::kExpand;  // Do expansion to get data instead.
+  }
+  if (*speech_type != AudioDecoder::kComfortNoise) {
+    // Don't increment timestamp if codec returned CNG speech type
+    // since in this case, the we will increment the CNGplayedTS counter.
+    // Increase with number of samples per channel.
+    RTC_DCHECK(*decoded_length == 0 ||
+               (decoder && decoder->Channels() == sync_buffer_->Channels()));
+    sync_buffer_->IncreaseEndTimestamp(
+        *decoded_length / static_cast<int>(sync_buffer_->Channels()));
+  }
+  return return_value;
+}
+
+int NetEqImpl::DecodeCng(AudioDecoder* decoder,
+                         int* decoded_length,
+                         AudioDecoder::SpeechType* speech_type) {
+  if (!decoder) {
+    // This happens when active decoder is not defined.
+    *decoded_length = -1;
+    return 0;
+  }
+
+  while (*decoded_length < rtc::dchecked_cast<int>(output_size_samples_)) {
+    const int length = decoder->Decode(
+        nullptr, 0, fs_hz_,
+        (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
+        &decoded_buffer_[*decoded_length], speech_type);
+    if (length > 0) {
+      *decoded_length += length;
+    } else {
+      // Error.
+      RTC_LOG(LS_WARNING) << "Failed to decode CNG";
+      *decoded_length = -1;
+      break;
+    }
+    if (*decoded_length > static_cast<int>(decoded_buffer_length_)) {
+      // Guard against overflow.
+      RTC_LOG(LS_WARNING) << "Decoded too much CNG.";
+      return kDecodedTooMuch;
+    }
+  }
+  stats_->GeneratedNoiseSamples(*decoded_length);
+  return 0;
+}
+
+int NetEqImpl::DecodeLoop(PacketList* packet_list,
+                          const Operation& operation,
+                          AudioDecoder* decoder,
+                          int* decoded_length,
+                          AudioDecoder::SpeechType* speech_type) {
+  RTC_DCHECK(last_decoded_packet_infos_.empty());
+
+  // Do decoding.
+  while (!packet_list->empty() && !decoder_database_->IsComfortNoise(
+                                      packet_list->front().payload_type)) {
+    RTC_DCHECK(decoder);  // At this point, we must have a decoder object.
+    // The number of channels in the `sync_buffer_` should be the same as the
+    // number decoder channels.
+    RTC_DCHECK_EQ(sync_buffer_->Channels(), decoder->Channels());
+    RTC_DCHECK_GE(decoded_buffer_length_, kMaxFrameSize * decoder->Channels());
+    RTC_DCHECK(operation == Operation::kNormal ||
+               operation == Operation::kAccelerate ||
+               operation == Operation::kFastAccelerate ||
+               operation == Operation::kMerge ||
+               operation == Operation::kPreemptiveExpand);
+
+    auto opt_result = packet_list->front().frame->Decode(
+        rtc::ArrayView<int16_t>(&decoded_buffer_[*decoded_length],
+                                decoded_buffer_length_ - *decoded_length));
+    last_decoded_packet_infos_.push_back(
+        std::move(packet_list->front().packet_info));
+    packet_list->pop_front();
+    if (opt_result) {
+      const auto& result = *opt_result;
+      *speech_type = result.speech_type;
+      if (result.num_decoded_samples > 0) {
+        *decoded_length += rtc::dchecked_cast<int>(result.num_decoded_samples);
+        // Update `decoder_frame_length_` with number of samples per channel.
+        decoder_frame_length_ =
+            result.num_decoded_samples / decoder->Channels();
+      }
+    } else {
+      // Error.
+      // TODO(ossu): What to put here?
+      RTC_LOG(LS_WARNING) << "Decode error";
+      *decoded_length = -1;
+      last_decoded_packet_infos_.clear();
+      packet_list->clear();
+      break;
+    }
+    if (*decoded_length > rtc::dchecked_cast<int>(decoded_buffer_length_)) {
+      // Guard against overflow.
+      RTC_LOG(LS_WARNING) << "Decoded too much.";
+      packet_list->clear();
+      return kDecodedTooMuch;
+    }
+  }  // End of decode loop.
+
+  // If the list is not empty at this point, either a decoding error terminated
+  // the while-loop, or list must hold exactly one CNG packet.
+  RTC_DCHECK(
+      packet_list->empty() || *decoded_length < 0 ||
+      (packet_list->size() == 1 &&
+       decoder_database_->IsComfortNoise(packet_list->front().payload_type)));
+  return 0;
+}
+
+void NetEqImpl::DoNormal(const int16_t* decoded_buffer,
+                         size_t decoded_length,
+                         AudioDecoder::SpeechType speech_type,
+                         bool play_dtmf) {
+  RTC_DCHECK(normal_.get());
+  normal_->Process(decoded_buffer, decoded_length, last_mode_,
+                   algorithm_buffer_.get());
+  if (decoded_length != 0) {
+    last_mode_ = Mode::kNormal;
+  }
+
+  // If last packet was decoded as an inband CNG, set mode to CNG instead.
+  if ((speech_type == AudioDecoder::kComfortNoise) ||
+      ((last_mode_ == Mode::kCodecInternalCng) && (decoded_length == 0))) {
+    // TODO(hlundin): Remove second part of || statement above.
+    last_mode_ = Mode::kCodecInternalCng;
+  }
+
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+}
+
+void NetEqImpl::DoMerge(int16_t* decoded_buffer,
+                        size_t decoded_length,
+                        AudioDecoder::SpeechType speech_type,
+                        bool play_dtmf) {
+  RTC_DCHECK(merge_.get());
+  size_t new_length =
+      merge_->Process(decoded_buffer, decoded_length, algorithm_buffer_.get());
+  // Correction can be negative.
+  int expand_length_correction =
+      rtc::dchecked_cast<int>(new_length) -
+      rtc::dchecked_cast<int>(decoded_length / algorithm_buffer_->Channels());
+
+  // Update in-call and post-call statistics.
+  if (expand_->MuteFactor(0) == 0) {
+    // Expand generates only noise.
+    stats_->ExpandedNoiseSamplesCorrection(expand_length_correction);
+  } else {
+    // Expansion generates more than only noise.
+    stats_->ExpandedVoiceSamplesCorrection(expand_length_correction);
+  }
+
+  last_mode_ = Mode::kMerge;
+  // If last packet was decoded as an inband CNG, set mode to CNG instead.
+  if (speech_type == AudioDecoder::kComfortNoise) {
+    last_mode_ = Mode::kCodecInternalCng;
+  }
+  expand_->Reset();
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+}
+
+bool NetEqImpl::DoCodecPlc() {
+  AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
+  if (!decoder) {
+    return false;
+  }
+  const size_t channels = algorithm_buffer_->Channels();
+  const size_t requested_samples_per_channel =
+      output_size_samples_ -
+      (sync_buffer_->FutureLength() - expand_->overlap_length());
+  concealment_audio_.Clear();
+  decoder->GeneratePlc(requested_samples_per_channel, &concealment_audio_);
+  if (concealment_audio_.empty()) {
+    // Nothing produced. Resort to regular expand.
+    return false;
+  }
+  RTC_CHECK_GE(concealment_audio_.size(),
+               requested_samples_per_channel * channels);
+  sync_buffer_->PushBackInterleaved(concealment_audio_);
+  RTC_DCHECK_NE(algorithm_buffer_->Channels(), 0);
+  const size_t concealed_samples_per_channel =
+      concealment_audio_.size() / channels;
+
+  // Update in-call and post-call statistics.
+  const bool is_new_concealment_event = (last_mode_ != Mode::kCodecPlc);
+  if (std::all_of(concealment_audio_.cbegin(), concealment_audio_.cend(),
+                  [](int16_t i) { return i == 0; })) {
+    // Expand operation generates only noise.
+    stats_->ExpandedNoiseSamples(concealed_samples_per_channel,
+                                 is_new_concealment_event);
+  } else {
+    // Expand operation generates more than only noise.
+    stats_->ExpandedVoiceSamples(concealed_samples_per_channel,
+                                 is_new_concealment_event);
+  }
+  last_mode_ = Mode::kCodecPlc;
+  if (!generated_noise_stopwatch_) {
+    // Start a new stopwatch since we may be covering for a lost CNG packet.
+    generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch();
+  }
+  return true;
+}
+
+int NetEqImpl::DoExpand(bool play_dtmf) {
+  while ((sync_buffer_->FutureLength() - expand_->overlap_length()) <
+         output_size_samples_) {
+    algorithm_buffer_->Clear();
+    int return_value = expand_->Process(algorithm_buffer_.get());
+    size_t length = algorithm_buffer_->Size();
+    bool is_new_concealment_event = (last_mode_ != Mode::kExpand);
+
+    // Update in-call and post-call statistics.
+    if (expand_->MuteFactor(0) == 0) {
+      // Expand operation generates only noise.
+      stats_->ExpandedNoiseSamples(length, is_new_concealment_event);
+    } else {
+      // Expand operation generates more than only noise.
+      stats_->ExpandedVoiceSamples(length, is_new_concealment_event);
+    }
+
+    last_mode_ = Mode::kExpand;
+
+    if (return_value < 0) {
+      return return_value;
+    }
+
+    sync_buffer_->PushBack(*algorithm_buffer_);
+    algorithm_buffer_->Clear();
+  }
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+
+  if (!generated_noise_stopwatch_) {
+    // Start a new stopwatch since we may be covering for a lost CNG packet.
+    generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch();
+  }
+
+  return 0;
+}
+
+int NetEqImpl::DoAccelerate(int16_t* decoded_buffer,
+                            size_t decoded_length,
+                            AudioDecoder::SpeechType speech_type,
+                            bool play_dtmf,
+                            bool fast_accelerate) {
+  const size_t required_samples =
+      static_cast<size_t>(240 * fs_mult_);  // Must have 30 ms.
+  size_t borrowed_samples_per_channel = 0;
+  size_t num_channels = algorithm_buffer_->Channels();
+  size_t decoded_length_per_channel = decoded_length / num_channels;
+  if (decoded_length_per_channel < required_samples) {
+    // Must move data from the `sync_buffer_` in order to get 30 ms.
+    borrowed_samples_per_channel =
+        static_cast<int>(required_samples - decoded_length_per_channel);
+    memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
+            decoded_buffer, sizeof(int16_t) * decoded_length);
+    sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel,
+                                         decoded_buffer);
+    decoded_length = required_samples * num_channels;
+  }
+
+  size_t samples_removed = 0;
+  Accelerate::ReturnCodes return_code =
+      accelerate_->Process(decoded_buffer, decoded_length, fast_accelerate,
+                           algorithm_buffer_.get(), &samples_removed);
+  stats_->AcceleratedSamples(samples_removed);
+  switch (return_code) {
+    case Accelerate::kSuccess:
+      last_mode_ = Mode::kAccelerateSuccess;
+      break;
+    case Accelerate::kSuccessLowEnergy:
+      last_mode_ = Mode::kAccelerateLowEnergy;
+      break;
+    case Accelerate::kNoStretch:
+      last_mode_ = Mode::kAccelerateFail;
+      break;
+    case Accelerate::kError:
+      // TODO(hlundin): Map to Modes::kError instead?
+      last_mode_ = Mode::kAccelerateFail;
+      return kAccelerateError;
+  }
+
+  if (borrowed_samples_per_channel > 0) {
+    // Copy borrowed samples back to the `sync_buffer_`.
+    size_t length = algorithm_buffer_->Size();
+    if (length < borrowed_samples_per_channel) {
+      // This destroys the beginning of the buffer, but will not cause any
+      // problems.
+      sync_buffer_->ReplaceAtIndex(
+          *algorithm_buffer_,
+          sync_buffer_->Size() - borrowed_samples_per_channel);
+      sync_buffer_->PushFrontZeros(borrowed_samples_per_channel - length);
+      algorithm_buffer_->PopFront(length);
+      RTC_DCHECK(algorithm_buffer_->Empty());
+    } else {
+      sync_buffer_->ReplaceAtIndex(
+          *algorithm_buffer_, borrowed_samples_per_channel,
+          sync_buffer_->Size() - borrowed_samples_per_channel);
+      algorithm_buffer_->PopFront(borrowed_samples_per_channel);
+    }
+  }
+
+  // If last packet was decoded as an inband CNG, set mode to CNG instead.
+  if (speech_type == AudioDecoder::kComfortNoise) {
+    last_mode_ = Mode::kCodecInternalCng;
+  }
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+  expand_->Reset();
+  return 0;
+}
+
+int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
+                                  size_t decoded_length,
+                                  AudioDecoder::SpeechType speech_type,
+                                  bool play_dtmf) {
+  const size_t required_samples =
+      static_cast<size_t>(240 * fs_mult_);  // Must have 30 ms.
+  size_t num_channels = algorithm_buffer_->Channels();
+  size_t borrowed_samples_per_channel = 0;
+  size_t old_borrowed_samples_per_channel = 0;
+  size_t decoded_length_per_channel = decoded_length / num_channels;
+  if (decoded_length_per_channel < required_samples) {
+    // Must move data from the `sync_buffer_` in order to get 30 ms.
+    borrowed_samples_per_channel =
+        required_samples - decoded_length_per_channel;
+    // Calculate how many of these were already played out.
+    old_borrowed_samples_per_channel =
+        (borrowed_samples_per_channel > sync_buffer_->FutureLength())
+            ? (borrowed_samples_per_channel - sync_buffer_->FutureLength())
+            : 0;
+    memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
+            decoded_buffer, sizeof(int16_t) * decoded_length);
+    sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel,
+                                         decoded_buffer);
+    decoded_length = required_samples * num_channels;
+  }
+
+  size_t samples_added = 0;
+  PreemptiveExpand::ReturnCodes return_code = preemptive_expand_->Process(
+      decoded_buffer, decoded_length, old_borrowed_samples_per_channel,
+      algorithm_buffer_.get(), &samples_added);
+  stats_->PreemptiveExpandedSamples(samples_added);
+  switch (return_code) {
+    case PreemptiveExpand::kSuccess:
+      last_mode_ = Mode::kPreemptiveExpandSuccess;
+      break;
+    case PreemptiveExpand::kSuccessLowEnergy:
+      last_mode_ = Mode::kPreemptiveExpandLowEnergy;
+      break;
+    case PreemptiveExpand::kNoStretch:
+      last_mode_ = Mode::kPreemptiveExpandFail;
+      break;
+    case PreemptiveExpand::kError:
+      // TODO(hlundin): Map to Modes::kError instead?
+      last_mode_ = Mode::kPreemptiveExpandFail;
+      return kPreemptiveExpandError;
+  }
+
+  if (borrowed_samples_per_channel > 0) {
+    // Copy borrowed samples back to the `sync_buffer_`.
+    sync_buffer_->ReplaceAtIndex(
+        *algorithm_buffer_, borrowed_samples_per_channel,
+        sync_buffer_->Size() - borrowed_samples_per_channel);
+    algorithm_buffer_->PopFront(borrowed_samples_per_channel);
+  }
+
+  // If last packet was decoded as an inband CNG, set mode to CNG instead.
+  if (speech_type == AudioDecoder::kComfortNoise) {
+    last_mode_ = Mode::kCodecInternalCng;
+  }
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+  expand_->Reset();
+  return 0;
+}
+
+int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
+  if (!packet_list->empty()) {
+    // Must have exactly one SID frame at this point.
+    RTC_DCHECK_EQ(packet_list->size(), 1);
+    const Packet& packet = packet_list->front();
+    if (!decoder_database_->IsComfortNoise(packet.payload_type)) {
+      RTC_LOG(LS_ERROR) << "Trying to decode non-CNG payload as CNG.";
+      return kOtherError;
+    }
+    if (comfort_noise_->UpdateParameters(packet) ==
+        ComfortNoise::kInternalError) {
+      algorithm_buffer_->Zeros(output_size_samples_);
+      return -comfort_noise_->internal_error_code();
+    }
+  }
+  int cn_return =
+      comfort_noise_->Generate(output_size_samples_, algorithm_buffer_.get());
+  expand_->Reset();
+  last_mode_ = Mode::kRfc3389Cng;
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+  if (cn_return == ComfortNoise::kInternalError) {
+    RTC_LOG(LS_WARNING) << "Comfort noise generator returned error code: "
+                        << comfort_noise_->internal_error_code();
+    return kComfortNoiseErrorCode;
+  } else if (cn_return == ComfortNoise::kUnknownPayloadType) {
+    return kUnknownRtpPayloadType;
+  }
+  return 0;
+}
+
+void NetEqImpl::DoCodecInternalCng(const int16_t* decoded_buffer,
+                                   size_t decoded_length) {
+  RTC_DCHECK(normal_.get());
+  normal_->Process(decoded_buffer, decoded_length, last_mode_,
+                   algorithm_buffer_.get());
+  last_mode_ = Mode::kCodecInternalCng;
+  expand_->Reset();
+}
+
+int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
+  // This block of the code and the block further down, handling `dtmf_switch`
+  // are commented out. Otherwise playing out-of-band DTMF would fail in VoE
+  // test, DtmfTest.ManualSuccessfullySendsOutOfBandTelephoneEvents. This is
+  // equivalent to `dtmf_switch` always be false.
+  //
+  // See http://webrtc-codereview.appspot.com/1195004/ for discussion
+  // On this issue. This change might cause some glitches at the point of
+  // switch from audio to DTMF. Issue 1545 is filed to track this.
+  //
+  //  bool dtmf_switch = false;
+  //  if ((last_mode_ != Modes::kDtmf) &&
+  //      dtmf_tone_generator_->initialized()) {
+  //    // Special case; see below.
+  //    // We must catch this before calling Generate, since `initialized` is
+  //    // modified in that call.
+  //    dtmf_switch = true;
+  //  }
+
+  int dtmf_return_value = 0;
+  if (!dtmf_tone_generator_->initialized()) {
+    // Initialize if not already done.
+    dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no,
+                                                   dtmf_event.volume);
+  }
+
+  if (dtmf_return_value == 0) {
+    // Generate DTMF signal.
+    dtmf_return_value = dtmf_tone_generator_->Generate(output_size_samples_,
+                                                       algorithm_buffer_.get());
+  }
+
+  if (dtmf_return_value < 0) {
+    algorithm_buffer_->Zeros(output_size_samples_);
+    return dtmf_return_value;
+  }
+
+  //  if (dtmf_switch) {
+  //    // This is the special case where the previous operation was DTMF
+  //    // overdub, but the current instruction is "regular" DTMF. We must make
+  //    // sure that the DTMF does not have any discontinuities. The first DTMF
+  //    // sample that we generate now must be played out immediately, therefore
+  //    // it must be copied to the speech buffer.
+  //    // TODO(hlundin): This code seems incorrect. (Legacy.) Write test and
+  //    // verify correct operation.
+  //    RTC_DCHECK_NOTREACHED();
+  //    // Must generate enough data to replace all of the `sync_buffer_`
+  //    // "future".
+  //    int required_length = sync_buffer_->FutureLength();
+  //    RTC_DCHECK(dtmf_tone_generator_->initialized());
+  //    dtmf_return_value = dtmf_tone_generator_->Generate(required_length,
+  //                                                       algorithm_buffer_);
+  //    RTC_DCHECK((size_t) required_length == algorithm_buffer_->Size());
+  //    if (dtmf_return_value < 0) {
+  //      algorithm_buffer_->Zeros(output_size_samples_);
+  //      return dtmf_return_value;
+  //    }
+  //
+  //    // Overwrite the "future" part of the speech buffer with the new DTMF
+  //    // data.
+  //    // TODO(hlundin): It seems that this overwriting has gone lost.
+  //    // Not adapted for multi-channel yet.
+  //    RTC_DCHECK(algorithm_buffer_->Channels() == 1);
+  //    if (algorithm_buffer_->Channels() != 1) {
+  //      RTC_LOG(LS_WARNING) << "DTMF not supported for more than one channel";
+  //      return kStereoNotSupported;
+  //    }
+  //    // Shuffle the remaining data to the beginning of algorithm buffer.
+  //    algorithm_buffer_->PopFront(sync_buffer_->FutureLength());
+  //  }
+
+  sync_buffer_->IncreaseEndTimestamp(
+      static_cast<uint32_t>(output_size_samples_));
+  expand_->Reset();
+  last_mode_ = Mode::kDtmf;
+
+  // Set to false because the DTMF is already in the algorithm buffer.
+  *play_dtmf = false;
+  return 0;
+}
+
+int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event,
+                           size_t num_channels,
+                           int16_t* output) const {
+  size_t out_index = 0;
+  size_t overdub_length = output_size_samples_;  // Default value.
+
+  if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) {
+    // Special operation for transition from "DTMF only" to "DTMF overdub".
+    out_index =
+        std::min(sync_buffer_->dtmf_index() - sync_buffer_->next_index(),
+                 output_size_samples_);
+    overdub_length = output_size_samples_ - out_index;
+  }
+
+  AudioMultiVector dtmf_output(num_channels);
+  int dtmf_return_value = 0;
+  if (!dtmf_tone_generator_->initialized()) {
+    dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no,
+                                                   dtmf_event.volume);
+  }
+  if (dtmf_return_value == 0) {
+    dtmf_return_value =
+        dtmf_tone_generator_->Generate(overdub_length, &dtmf_output);
+    RTC_DCHECK_EQ(overdub_length, dtmf_output.Size());
+  }
+  dtmf_output.ReadInterleaved(overdub_length, &output[out_index]);
+  return dtmf_return_value < 0 ? dtmf_return_value : 0;
+}
+
+int NetEqImpl::ExtractPackets(size_t required_samples,
+                              PacketList* packet_list) {
+  bool first_packet = true;
+  uint8_t prev_payload_type = 0;
+  uint32_t prev_timestamp = 0;
+  uint16_t prev_sequence_number = 0;
+  bool next_packet_available = false;
+
+  const Packet* next_packet = packet_buffer_->PeekNextPacket();
+  RTC_DCHECK(next_packet);
+  if (!next_packet) {
+    RTC_LOG(LS_ERROR) << "Packet buffer unexpectedly empty.";
+    return -1;
+  }
+  uint32_t first_timestamp = next_packet->timestamp;
+  size_t extracted_samples = 0;
+
+  // Packet extraction loop.
+  do {
+    timestamp_ = next_packet->timestamp;
+    absl::optional<Packet> packet = packet_buffer_->GetNextPacket();
+    // `next_packet` may be invalid after the `packet_buffer_` operation.
+    next_packet = nullptr;
+    if (!packet) {
+      RTC_LOG(LS_ERROR) << "Should always be able to extract a packet here";
+      RTC_DCHECK_NOTREACHED();  // Should always be able to extract a packet
+                                // here.
+      return -1;
+    }
+    const uint64_t waiting_time_ms = packet->waiting_time->ElapsedMs();
+    stats_->StoreWaitingTime(waiting_time_ms);
+    RTC_DCHECK(!packet->empty());
+
+    if (first_packet) {
+      first_packet = false;
+      if (nack_enabled_) {
+        RTC_DCHECK(nack_);
+        // TODO(henrik.lundin): Should we update this for all decoded packets?
+        nack_->UpdateLastDecodedPacket(packet->sequence_number,
+                                       packet->timestamp);
+      }
+      prev_sequence_number = packet->sequence_number;
+      prev_timestamp = packet->timestamp;
+      prev_payload_type = packet->payload_type;
+    }
+
+    const bool has_cng_packet =
+        decoder_database_->IsComfortNoise(packet->payload_type);
+    // Store number of extracted samples.
+    size_t packet_duration = 0;
+    if (packet->frame) {
+      packet_duration = packet->frame->Duration();
+      // TODO(ossu): Is this the correct way to track Opus FEC packets?
+      if (packet->priority.codec_level > 0) {
+        stats_->SecondaryDecodedSamples(
+            rtc::dchecked_cast<int>(packet_duration));
+      }
+    } else if (!has_cng_packet) {
+      RTC_LOG(LS_WARNING) << "Unknown payload type "
+                          << static_cast<int>(packet->payload_type);
+      RTC_DCHECK_NOTREACHED();
+    }
+
+    if (packet_duration == 0) {
+      // Decoder did not return a packet duration. Assume that the packet
+      // contains the same number of samples as the previous one.
+      packet_duration = decoder_frame_length_;
+    }
+    extracted_samples = packet->timestamp - first_timestamp + packet_duration;
+
+    RTC_DCHECK(controller_);
+    stats_->JitterBufferDelay(packet_duration, waiting_time_ms,
+                              controller_->TargetLevelMs(),
+                              controller_->UnlimitedTargetLevelMs());
+
+    packet_list->push_back(std::move(*packet));  // Store packet in list.
+    packet = absl::nullopt;  // Ensure it's never used after the move.
+
+    // Check what packet is available next.
+    next_packet = packet_buffer_->PeekNextPacket();
+    next_packet_available = false;
+    if (next_packet && prev_payload_type == next_packet->payload_type &&
+        !has_cng_packet) {
+      int16_t seq_no_diff = next_packet->sequence_number - prev_sequence_number;
+      size_t ts_diff = next_packet->timestamp - prev_timestamp;
+      if ((seq_no_diff == 1 || seq_no_diff == 0) &&
+          ts_diff <= packet_duration) {
+        // The next sequence number is available, or the next part of a packet
+        // that was split into pieces upon insertion.
+        next_packet_available = true;
+      }
+      prev_sequence_number = next_packet->sequence_number;
+      prev_timestamp = next_packet->timestamp;
+    }
+  } while (extracted_samples < required_samples && next_packet_available);
+
+  if (extracted_samples > 0) {
+    // Delete old packets only when we are going to decode something. Otherwise,
+    // we could end up in the situation where we never decode anything, since
+    // all incoming packets are considered too old but the buffer will also
+    // never be flooded and flushed.
+    packet_buffer_->DiscardAllOldPackets(timestamp_, stats_.get());
+  }
+
+  return rtc::dchecked_cast<int>(extracted_samples);
+}
+
+void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) {
+  // Delete objects and create new ones.
+  expand_.reset(expand_factory_->Create(background_noise_.get(),
+                                        sync_buffer_.get(), &random_vector_,
+                                        stats_.get(), fs_hz, channels));
+  merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get()));
+}
+
+void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
+  RTC_LOG(LS_VERBOSE) << "SetSampleRateAndChannels " << fs_hz << " "
+                      << channels;
+  // TODO(hlundin): Change to an enumerator and skip assert.
+  RTC_DCHECK(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 ||
+             fs_hz == 48000);
+  RTC_DCHECK_GT(channels, 0);
+
+  // Before changing the sample rate, end and report any ongoing expand event.
+  stats_->EndExpandEvent(fs_hz_);
+  fs_hz_ = fs_hz;
+  fs_mult_ = fs_hz / 8000;
+  output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
+  decoder_frame_length_ = 3 * output_size_samples_;  // Initialize to 30ms.
+
+  last_mode_ = Mode::kNormal;
+
+  ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+  if (cng_decoder)
+    cng_decoder->Reset();
+
+  // Reinit post-decode VAD with new sample rate.
+  RTC_DCHECK(vad_.get());  // Cannot be NULL here.
+  vad_->Init();
+
+  // Delete algorithm buffer and create a new one.
+  algorithm_buffer_.reset(new AudioMultiVector(channels));
+
+  // Delete sync buffer and create a new one.
+  sync_buffer_.reset(new SyncBuffer(channels, kSyncBufferSize * fs_mult_));
+
+  // Delete BackgroundNoise object and create a new one.
+  background_noise_.reset(new BackgroundNoise(channels));
+
+  // Reset random vector.
+  random_vector_.Reset();
+
+  UpdatePlcComponents(fs_hz, channels);
+
+  // Move index so that we create a small set of future samples (all 0).
+  sync_buffer_->set_next_index(sync_buffer_->next_index() -
+                               expand_->overlap_length());
+
+  normal_.reset(new Normal(fs_hz, decoder_database_.get(), *background_noise_,
+                           expand_.get(), stats_.get()));
+  accelerate_.reset(
+      accelerate_factory_->Create(fs_hz, channels, *background_noise_));
+  preemptive_expand_.reset(preemptive_expand_factory_->Create(
+      fs_hz, channels, *background_noise_, expand_->overlap_length()));
+
+  // Delete ComfortNoise object and create a new one.
+  comfort_noise_.reset(
+      new ComfortNoise(fs_hz, decoder_database_.get(), sync_buffer_.get()));
+
+  // Verify that `decoded_buffer_` is long enough.
+  if (decoded_buffer_length_ < kMaxFrameSize * channels) {
+    // Reallocate to larger size.
+    decoded_buffer_length_ = kMaxFrameSize * channels;
+    decoded_buffer_.reset(new int16_t[decoded_buffer_length_]);
+  }
+  RTC_CHECK(controller_) << "Unexpectedly found no NetEqController";
+  controller_->SetSampleRate(fs_hz_, output_size_samples_);
+}
+
+NetEqImpl::OutputType NetEqImpl::LastOutputType() {
+  RTC_DCHECK(vad_.get());
+  RTC_DCHECK(expand_.get());
+  if (last_mode_ == Mode::kCodecInternalCng ||
+      last_mode_ == Mode::kRfc3389Cng) {
+    return OutputType::kCNG;
+  } else if (last_mode_ == Mode::kExpand && expand_->MuteFactor(0) == 0) {
+    // Expand mode has faded down to background noise only (very long expand).
+    return OutputType::kPLCCNG;
+  } else if (last_mode_ == Mode::kExpand) {
+    return OutputType::kPLC;
+  } else if (vad_->running() && !vad_->active_speech()) {
+    return OutputType::kVadPassive;
+  } else if (last_mode_ == Mode::kCodecPlc) {
+    return OutputType::kCodecPLC;
+  } else {
+    return OutputType::kNormalSpeech;
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h
new file mode 100644
index 0000000000..6120eab5b6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -0,0 +1,404 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
+#define MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio/audio_frame.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "api/neteq/neteq_controller_factory.h"
+#include "api/neteq/tick_timer.h"
+#include "api/rtp_packet_info.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/expand_uma_logger.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class Accelerate;
+class BackgroundNoise;
+class Clock;
+class ComfortNoise;
+class DecoderDatabase;
+class DtmfBuffer;
+class DtmfToneGenerator;
+class Expand;
+class Merge;
+class NackTracker;
+class Normal;
+class PacketBuffer;
+class RedPayloadSplitter;
+class PostDecodeVad;
+class PreemptiveExpand;
+class RandomVector;
+class SyncBuffer;
+class TimestampScaler;
+struct AccelerateFactory;
+struct DtmfEvent;
+struct ExpandFactory;
+struct PreemptiveExpandFactory;
+
+class NetEqImpl : public webrtc::NetEq {
+ public:
+  enum class OutputType {
+    kNormalSpeech,
+    kPLC,
+    kCNG,
+    kPLCCNG,
+    kVadPassive,
+    kCodecPLC
+  };
+
+  enum ErrorCodes {
+    kNoError = 0,
+    kOtherError,
+    kUnknownRtpPayloadType,
+    kDecoderNotFound,
+    kInvalidPointer,
+    kAccelerateError,
+    kPreemptiveExpandError,
+    kComfortNoiseErrorCode,
+    kDecoderErrorCode,
+    kOtherDecoderError,
+    kInvalidOperation,
+    kDtmfParsingError,
+    kDtmfInsertError,
+    kSampleUnderrun,
+    kDecodedTooMuch,
+    kRedundancySplitError,
+    kPacketBufferCorruption
+  };
+
+  struct Dependencies {
+    // The constructor populates the Dependencies struct with the default
+    // implementations of the objects. They can all be replaced by the user
+    // before sending the struct to the NetEqImpl constructor. However, there
+    // are dependencies between some of the classes inside the struct, so
+    // swapping out one may make it necessary to re-create another one.
+    Dependencies(const NetEq::Config& config,
+                 Clock* clock,
+                 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+                 const NetEqControllerFactory& controller_factory);
+    ~Dependencies();
+
+    Clock* const clock;
+    std::unique_ptr<TickTimer> tick_timer;
+    std::unique_ptr<StatisticsCalculator> stats;
+    std::unique_ptr<DecoderDatabase> decoder_database;
+    std::unique_ptr<DtmfBuffer> dtmf_buffer;
+    std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
+    std::unique_ptr<PacketBuffer> packet_buffer;
+    std::unique_ptr<NetEqController> neteq_controller;
+    std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
+    std::unique_ptr<TimestampScaler> timestamp_scaler;
+    std::unique_ptr<AccelerateFactory> accelerate_factory;
+    std::unique_ptr<ExpandFactory> expand_factory;
+    std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
+  };
+
+  // Creates a new NetEqImpl object.
+  NetEqImpl(const NetEq::Config& config,
+            Dependencies&& deps,
+            bool create_components = true);
+
+  ~NetEqImpl() override;
+
+  NetEqImpl(const NetEqImpl&) = delete;
+  NetEqImpl& operator=(const NetEqImpl&) = delete;
+
+  // Inserts a new packet into NetEq. Returns 0 on success, -1 on failure.
+  int InsertPacket(const RTPHeader& rtp_header,
+                   rtc::ArrayView<const uint8_t> payload) override;
+
+  void InsertEmptyPacket(const RTPHeader& rtp_header) override;
+
+  int GetAudio(
+      AudioFrame* audio_frame,
+      bool* muted,
+      int* current_sample_rate_hz = nullptr,
+      absl::optional<Operation> action_override = absl::nullopt) override;
+
+  void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
+
+  bool RegisterPayloadType(int rtp_payload_type,
+                           const SdpAudioFormat& audio_format) override;
+
+  // Removes `rtp_payload_type` from the codec database. Returns 0 on success,
+  // -1 on failure.
+  int RemovePayloadType(uint8_t rtp_payload_type) override;
+
+  void RemoveAllPayloadTypes() override;
+
+  bool SetMinimumDelay(int delay_ms) override;
+
+  bool SetMaximumDelay(int delay_ms) override;
+
+  bool SetBaseMinimumDelayMs(int delay_ms) override;
+
+  int GetBaseMinimumDelayMs() const override;
+
+  int TargetDelayMs() const override;
+
+  int FilteredCurrentDelayMs() const override;
+
+  // Writes the current network statistics to `stats`. The statistics are reset
+  // after the call.
+  int NetworkStatistics(NetEqNetworkStatistics* stats) override;
+
+  NetEqNetworkStatistics CurrentNetworkStatistics() const override;
+
+  NetEqLifetimeStatistics GetLifetimeStatistics() const override;
+
+  NetEqOperationsAndState GetOperationsAndState() const override;
+
+  // Enables post-decode VAD. When enabled, GetAudio() will return
+  // kOutputVADPassive when the signal contains no speech.
+  void EnableVad() override;
+
+  // Disables post-decode VAD.
+  void DisableVad() override;
+
+  absl::optional<uint32_t> GetPlayoutTimestamp() const override;
+
+  int last_output_sample_rate_hz() const override;
+
+  absl::optional<DecoderFormat> GetDecoderFormat(
+      int payload_type) const override;
+
+  // Flushes both the packet buffer and the sync buffer.
+  void FlushBuffers() override;
+
+  void EnableNack(size_t max_nack_list_size) override;
+
+  void DisableNack() override;
+
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
+
+  int SyncBufferSizeMs() const override;
+
+  // This accessor method is only intended for testing purposes.
+  const SyncBuffer* sync_buffer_for_test() const;
+  Operation last_operation_for_test() const;
+
+ protected:
+  static const int kOutputSizeMs = 10;
+  static const size_t kMaxFrameSize = 5760;  // 120 ms @ 48 kHz.
+  // TODO(hlundin): Provide a better value for kSyncBufferSize.
+  // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
+  // calculating correlations of current frame against history.
+  static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
+
+  // Inserts a new packet into NetEq. This is used by the InsertPacket method
+  // above. Returns 0 on success, otherwise an error code.
+  // TODO(hlundin): Merge this with InsertPacket above?
+  int InsertPacketInternal(const RTPHeader& rtp_header,
+                           rtc::ArrayView<const uint8_t> payload)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Delivers 10 ms of audio data. The data is written to `audio_frame`.
+  // Returns 0 on success, otherwise an error code.
+  int GetAudioInternal(AudioFrame* audio_frame,
+                       bool* muted,
+                       absl::optional<Operation> action_override)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Provides a decision to the GetAudioInternal method. The decision what to
+  // do is written to `operation`. Packets to decode are written to
+  // `packet_list`, and a DTMF event to play is written to `dtmf_event`. When
+  // DTMF should be played, `play_dtmf` is set to true by the method.
+  // Returns 0 on success, otherwise an error code.
+  int GetDecision(Operation* operation,
+                  PacketList* packet_list,
+                  DtmfEvent* dtmf_event,
+                  bool* play_dtmf,
+                  absl::optional<Operation> action_override)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Decodes the speech packets in `packet_list`, and writes the results to
+  // `decoded_buffer`, which is allocated to hold `decoded_buffer_length`
+  // elements. The length of the decoded data is written to `decoded_length`.
+  // The speech type -- speech or (codec-internal) comfort noise -- is written
+  // to `speech_type`. If `packet_list` contains any SID frames for RFC 3389
+  // comfort noise, those are not decoded.
+  int Decode(PacketList* packet_list,
+             Operation* operation,
+             int* decoded_length,
+             AudioDecoder::SpeechType* speech_type)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method to Decode(). Performs codec internal CNG.
+  int DecodeCng(AudioDecoder* decoder,
+                int* decoded_length,
+                AudioDecoder::SpeechType* speech_type)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method to Decode(). Performs the actual decoding.
+  int DecodeLoop(PacketList* packet_list,
+                 const Operation& operation,
+                 AudioDecoder* decoder,
+                 int* decoded_length,
+                 AudioDecoder::SpeechType* speech_type)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the Normal class to perform the normal operation.
+  void DoNormal(const int16_t* decoded_buffer,
+                size_t decoded_length,
+                AudioDecoder::SpeechType speech_type,
+                bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the Merge class to perform the merge operation.
+  void DoMerge(int16_t* decoded_buffer,
+               size_t decoded_length,
+               AudioDecoder::SpeechType speech_type,
+               bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  bool DoCodecPlc() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the Expand class to perform the expand operation.
+  int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the Accelerate class to perform the accelerate
+  // operation.
+  int DoAccelerate(int16_t* decoded_buffer,
+                   size_t decoded_length,
+                   AudioDecoder::SpeechType speech_type,
+                   bool play_dtmf,
+                   bool fast_accelerate) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the PreemptiveExpand class to perform the
+  // preemtive expand operation.
+  int DoPreemptiveExpand(int16_t* decoded_buffer,
+                         size_t decoded_length,
+                         AudioDecoder::SpeechType speech_type,
+                         bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
+  // noise. `packet_list` can either contain one SID frame to update the
+  // noise parameters, or no payload at all, in which case the previously
+  // received parameters are used.
+  int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Calls the audio decoder to generate codec-internal comfort noise when
+  // no packet was received.
+  void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Calls the DtmfToneGenerator class to generate DTMF tones.
+  int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Overdub DTMF on top of `output`.
+  int DtmfOverdub(const DtmfEvent& dtmf_event,
+                  size_t num_channels,
+                  int16_t* output) const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Extracts packets from `packet_buffer_` to produce at least
+  // `required_samples` samples. The packets are inserted into `packet_list`.
+  // Returns the number of samples that the packets in the list will produce, or
+  // -1 in case of an error.
+  int ExtractPackets(size_t required_samples, PacketList* packet_list)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Resets various variables and objects to new values based on the sample rate
+  // `fs_hz` and `channels` number audio channels.
+  void SetSampleRateAndChannels(int fs_hz, size_t channels)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Returns the output type for the audio produced by the latest call to
+  // GetAudio().
+  OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Updates Expand and Merge.
+  virtual void UpdatePlcComponents(int fs_hz, size_t channels)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  NetEqNetworkStatistics CurrentNetworkStatisticsInternal() const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock* const clock_;
+
+  mutable Mutex mutex_;
+  const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<DecoderDatabase> decoder_database_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<TimestampScaler> timestamp_scaler_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<ExpandFactory> expand_factory_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<AccelerateFactory> accelerate_factory_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<StatisticsCalculator> stats_ RTC_GUARDED_BY(mutex_);
+
+  std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<NetEqController> controller_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<AudioMultiVector> algorithm_buffer_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<PreemptiveExpand> preemptive_expand_ RTC_GUARDED_BY(mutex_);
+  RandomVector random_vector_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(mutex_);
+  int fs_hz_ RTC_GUARDED_BY(mutex_);
+  int fs_mult_ RTC_GUARDED_BY(mutex_);
+  int last_output_sample_rate_hz_ RTC_GUARDED_BY(mutex_);
+  size_t output_size_samples_ RTC_GUARDED_BY(mutex_);
+  size_t decoder_frame_length_ RTC_GUARDED_BY(mutex_);
+  Mode last_mode_ RTC_GUARDED_BY(mutex_);
+  Operation last_operation_ RTC_GUARDED_BY(mutex_);
+  size_t decoded_buffer_length_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(mutex_);
+  uint32_t playout_timestamp_ RTC_GUARDED_BY(mutex_);
+  bool new_codec_ RTC_GUARDED_BY(mutex_);
+  uint32_t timestamp_ RTC_GUARDED_BY(mutex_);
+  bool reset_decoder_ RTC_GUARDED_BY(mutex_);
+  absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(mutex_);
+  absl::optional<uint8_t> current_cng_rtp_payload_type_ RTC_GUARDED_BY(mutex_);
+  bool first_packet_ RTC_GUARDED_BY(mutex_);
+  bool enable_fast_accelerate_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(mutex_);
+  bool nack_enabled_ RTC_GUARDED_BY(mutex_);
+  const bool enable_muted_state_ RTC_GUARDED_BY(mutex_);
+  AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(mutex_) =
+      AudioFrame::kVadPassive;
+  std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
+      RTC_GUARDED_BY(mutex_);
+  std::vector<RtpPacketInfo> last_decoded_packet_infos_ RTC_GUARDED_BY(mutex_);
+  ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(mutex_);
+  ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(mutex_);
+  bool no_time_stretching_ RTC_GUARDED_BY(mutex_);  // Only used for test.
+  rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
new file mode 100644
index 0000000000..ce2be656ef
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -0,0 +1,1871 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/neteq_impl.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/default_neteq_controller_factory.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/histogram.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/mock/mock_dtmf_buffer.h"
+#include "modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
+#include "modules/audio_coding/neteq/mock/mock_neteq_controller.h"
+#include "modules/audio_coding/neteq/mock/mock_packet_buffer.h"
+#include "modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/clock.h"
+#include "test/audio_decoder_proxy_factory.h"
+#include "test/function_audio_decoder_factory.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/mock_audio_decoder_factory.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::DoAll;
+using ::testing::ElementsAre;
+using ::testing::InSequence;
+using ::testing::Invoke;
+using ::testing::IsEmpty;
+using ::testing::IsNull;
+using ::testing::Pointee;
+using ::testing::Return;
+using ::testing::ReturnNull;
+using ::testing::SetArgPointee;
+using ::testing::SetArrayArgument;
+using ::testing::SizeIs;
+using ::testing::WithArg;
+
+namespace webrtc {
+
+// This function is called when inserting a packet list into the mock packet
+// buffer. The purpose is to delete all inserted packets properly, to avoid
+// memory leaks in the test.
+int DeletePacketsAndReturnOk(PacketList* packet_list) {
+  packet_list->clear();
+  return PacketBuffer::kOK;
+}
+
+class NetEqImplTest : public ::testing::Test {
+ protected:
+  NetEqImplTest() : clock_(0) { config_.sample_rate_hz = 8000; }
+
+  void CreateInstance(
+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+    ASSERT_TRUE(decoder_factory);
+    config_.enable_muted_state = enable_muted_state_;
+    NetEqImpl::Dependencies deps(config_, &clock_, decoder_factory,
+                                 DefaultNetEqControllerFactory());
+
+    // Get a local pointer to NetEq's TickTimer object.
+    tick_timer_ = deps.tick_timer.get();
+
+    if (use_mock_decoder_database_) {
+      std::unique_ptr<MockDecoderDatabase> mock(new MockDecoderDatabase);
+      mock_decoder_database_ = mock.get();
+      EXPECT_CALL(*mock_decoder_database_, GetActiveCngDecoder())
+          .WillOnce(ReturnNull());
+      deps.decoder_database = std::move(mock);
+    }
+    decoder_database_ = deps.decoder_database.get();
+
+    if (use_mock_dtmf_buffer_) {
+      std::unique_ptr<MockDtmfBuffer> mock(
+          new MockDtmfBuffer(config_.sample_rate_hz));
+      mock_dtmf_buffer_ = mock.get();
+      deps.dtmf_buffer = std::move(mock);
+    }
+    dtmf_buffer_ = deps.dtmf_buffer.get();
+
+    if (use_mock_dtmf_tone_generator_) {
+      std::unique_ptr<MockDtmfToneGenerator> mock(new MockDtmfToneGenerator);
+      mock_dtmf_tone_generator_ = mock.get();
+      deps.dtmf_tone_generator = std::move(mock);
+    }
+    dtmf_tone_generator_ = deps.dtmf_tone_generator.get();
+
+    if (use_mock_packet_buffer_) {
+      std::unique_ptr<MockPacketBuffer> mock(
+          new MockPacketBuffer(config_.max_packets_in_buffer, tick_timer_));
+      mock_packet_buffer_ = mock.get();
+      deps.packet_buffer = std::move(mock);
+    }
+    packet_buffer_ = deps.packet_buffer.get();
+
+    if (use_mock_neteq_controller_) {
+      std::unique_ptr<MockNetEqController> mock(new MockNetEqController());
+      mock_neteq_controller_ = mock.get();
+      deps.neteq_controller = std::move(mock);
+    } else {
+      deps.stats = std::make_unique<StatisticsCalculator>();
+      NetEqController::Config controller_config;
+      controller_config.tick_timer = tick_timer_;
+      controller_config.base_min_delay_ms = config_.min_delay_ms;
+      controller_config.allow_time_stretching = true;
+      controller_config.max_packets_in_buffer = config_.max_packets_in_buffer;
+      controller_config.clock = &clock_;
+      deps.neteq_controller =
+          std::make_unique<DecisionLogic>(std::move(controller_config));
+    }
+    neteq_controller_ = deps.neteq_controller.get();
+
+    if (use_mock_payload_splitter_) {
+      std::unique_ptr<MockRedPayloadSplitter> mock(new MockRedPayloadSplitter);
+      mock_payload_splitter_ = mock.get();
+      deps.red_payload_splitter = std::move(mock);
+    }
+    red_payload_splitter_ = deps.red_payload_splitter.get();
+
+    deps.timestamp_scaler = std::unique_ptr<TimestampScaler>(
+        new TimestampScaler(*deps.decoder_database.get()));
+
+    neteq_.reset(new NetEqImpl(config_, std::move(deps)));
+    ASSERT_TRUE(neteq_ != NULL);
+  }
+
+  void CreateInstance() { CreateInstance(CreateBuiltinAudioDecoderFactory()); }
+
+  void UseNoMocks() {
+    ASSERT_TRUE(neteq_ == NULL) << "Must call UseNoMocks before CreateInstance";
+    use_mock_decoder_database_ = false;
+    use_mock_neteq_controller_ = false;
+    use_mock_dtmf_buffer_ = false;
+    use_mock_dtmf_tone_generator_ = false;
+    use_mock_packet_buffer_ = false;
+    use_mock_payload_splitter_ = false;
+  }
+
+  virtual ~NetEqImplTest() {
+    if (use_mock_decoder_database_) {
+      EXPECT_CALL(*mock_decoder_database_, Die()).Times(1);
+    }
+    if (use_mock_neteq_controller_) {
+      EXPECT_CALL(*mock_neteq_controller_, Die()).Times(1);
+    }
+    if (use_mock_dtmf_buffer_) {
+      EXPECT_CALL(*mock_dtmf_buffer_, Die()).Times(1);
+    }
+    if (use_mock_dtmf_tone_generator_) {
+      EXPECT_CALL(*mock_dtmf_tone_generator_, Die()).Times(1);
+    }
+    if (use_mock_packet_buffer_) {
+      EXPECT_CALL(*mock_packet_buffer_, Die()).Times(1);
+    }
+  }
+
+  void TestDtmfPacket(int sample_rate_hz) {
+    const size_t kPayloadLength = 4;
+    const uint8_t kPayloadType = 110;
+    const int kSampleRateHz = 16000;
+    config_.sample_rate_hz = kSampleRateHz;
+    UseNoMocks();
+    CreateInstance();
+    // Event: 2, E bit, Volume: 17, Length: 4336.
+    uint8_t payload[kPayloadLength] = {0x02, 0x80 + 0x11, 0x10, 0xF0};
+    RTPHeader rtp_header;
+    rtp_header.payloadType = kPayloadType;
+    rtp_header.sequenceNumber = 0x1234;
+    rtp_header.timestamp = 0x12345678;
+    rtp_header.ssrc = 0x87654321;
+
+    EXPECT_TRUE(neteq_->RegisterPayloadType(
+        kPayloadType, SdpAudioFormat("telephone-event", sample_rate_hz, 1)));
+
+    // Insert first packet.
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+    // Pull audio once.
+    const size_t kMaxOutputSize =
+        static_cast<size_t>(10 * kSampleRateHz / 1000);
+    AudioFrame output;
+    bool muted;
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    ASSERT_FALSE(muted);
+    ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+    EXPECT_EQ(1u, output.num_channels_);
+    EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+    // DTMF packets are immediately consumed by `InsertPacket()` and won't be
+    // returned by `GetAudio()`.
+    EXPECT_THAT(output.packet_infos_, IsEmpty());
+
+    // Verify first 64 samples of actual output.
+    const std::vector<int16_t> kOutput(
+        {0,     0,     0,     0,     0,     0,     0,     0,     0,     0,
+         -1578, -2816, -3460, -3403, -2709, -1594, -363,  671,   1269,  1328,
+         908,   202,   -513,  -964,  -955,  -431,  504,   1617,  2602,  3164,
+         3101,  2364,  1073,  -511,  -2047, -3198, -3721, -3525, -2688, -1440,
+         -99,   1015,  1663,  1744,  1319,  588,   -171,  -680,  -747,  -315,
+         515,   1512,  2378,  2828,  2674,  1877,  568,   -986,  -2446, -3482,
+         -3864, -3516, -2534, -1163});
+    ASSERT_GE(kMaxOutputSize, kOutput.size());
+    EXPECT_TRUE(std::equal(kOutput.begin(), kOutput.end(), output.data()));
+  }
+
+  std::unique_ptr<NetEqImpl> neteq_;
+  NetEq::Config config_;
+  SimulatedClock clock_;
+  TickTimer* tick_timer_ = nullptr;
+  MockDecoderDatabase* mock_decoder_database_ = nullptr;
+  DecoderDatabase* decoder_database_ = nullptr;
+  bool use_mock_decoder_database_ = true;
+  MockNetEqController* mock_neteq_controller_ = nullptr;
+  NetEqController* neteq_controller_ = nullptr;
+  bool use_mock_neteq_controller_ = true;
+  MockDtmfBuffer* mock_dtmf_buffer_ = nullptr;
+  DtmfBuffer* dtmf_buffer_ = nullptr;
+  bool use_mock_dtmf_buffer_ = true;
+  MockDtmfToneGenerator* mock_dtmf_tone_generator_ = nullptr;
+  DtmfToneGenerator* dtmf_tone_generator_ = nullptr;
+  bool use_mock_dtmf_tone_generator_ = true;
+  MockPacketBuffer* mock_packet_buffer_ = nullptr;
+  PacketBuffer* packet_buffer_ = nullptr;
+  bool use_mock_packet_buffer_ = true;
+  MockRedPayloadSplitter* mock_payload_splitter_ = nullptr;
+  RedPayloadSplitter* red_payload_splitter_ = nullptr;
+  bool use_mock_payload_splitter_ = true;
+  bool enable_muted_state_ = false;
+};
+
+// This tests the interface class NetEq.
+// TODO(hlundin): Move to separate file?
+TEST(NetEq, CreateAndDestroy) {
+  NetEq::Config config;
+  SimulatedClock clock(0);
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  std::unique_ptr<NetEq> neteq =
+      DefaultNetEqFactory().CreateNetEq(config, decoder_factory, &clock);
+}
+
+TEST_F(NetEqImplTest, RegisterPayloadType) {
+  CreateInstance();
+  constexpr int rtp_payload_type = 0;
+  const SdpAudioFormat format("pcmu", 8000, 1);
+  EXPECT_CALL(*mock_decoder_database_,
+              RegisterPayload(rtp_payload_type, format));
+  neteq_->RegisterPayloadType(rtp_payload_type, format);
+}
+
+TEST_F(NetEqImplTest, RemovePayloadType) {
+  CreateInstance();
+  uint8_t rtp_payload_type = 0;
+  EXPECT_CALL(*mock_decoder_database_, Remove(rtp_payload_type))
+      .WillOnce(Return(DecoderDatabase::kDecoderNotFound));
+  // Check that kOK is returned when database returns kDecoderNotFound, because
+  // removing a payload type that was never registered is not an error.
+  EXPECT_EQ(NetEq::kOK, neteq_->RemovePayloadType(rtp_payload_type));
+}
+
+TEST_F(NetEqImplTest, RemoveAllPayloadTypes) {
+  CreateInstance();
+  EXPECT_CALL(*mock_decoder_database_, RemoveAll()).WillOnce(Return());
+  neteq_->RemoveAllPayloadTypes();
+}
+
+TEST_F(NetEqImplTest, InsertPacket) {
+  using ::testing::AllOf;
+  using ::testing::Field;
+  CreateInstance();
+  const size_t kPayloadLength = 100;
+  const uint8_t kPayloadType = 0;
+  const uint16_t kFirstSequenceNumber = 0x1234;
+  const uint32_t kFirstTimestamp = 0x12345678;
+  const uint32_t kSsrc = 0x87654321;
+  uint8_t payload[kPayloadLength] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = kFirstSequenceNumber;
+  rtp_header.timestamp = kFirstTimestamp;
+  rtp_header.ssrc = kSsrc;
+  Packet fake_packet;
+  fake_packet.payload_type = kPayloadType;
+  fake_packet.sequence_number = kFirstSequenceNumber;
+  fake_packet.timestamp = kFirstTimestamp;
+
+  auto mock_decoder_factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
+      .WillOnce(Invoke([&](const SdpAudioFormat& format,
+                           absl::optional<AudioCodecPairId> codec_pair_id,
+                           std::unique_ptr<AudioDecoder>* dec) {
+        EXPECT_EQ("pcmu", format.name);
+
+        std::unique_ptr<MockAudioDecoder> mock_decoder(new MockAudioDecoder);
+        EXPECT_CALL(*mock_decoder, Channels()).WillRepeatedly(Return(1));
+        EXPECT_CALL(*mock_decoder, SampleRateHz()).WillRepeatedly(Return(8000));
+        EXPECT_CALL(*mock_decoder, Die()).Times(1);  // Called when deleted.
+
+        *dec = std::move(mock_decoder);
+      }));
+  DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                    absl::nullopt, mock_decoder_factory.get());
+
+  // Expectations for decoder database.
+  EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  // Expectations for packet buffer.
+  EXPECT_CALL(*mock_packet_buffer_, Empty())
+      .WillOnce(Return(false));  // Called once after first packet is inserted.
+  EXPECT_CALL(*mock_packet_buffer_, Flush(_)).Times(1);
+  EXPECT_CALL(*mock_packet_buffer_, InsertPacketList(_, _, _, _, _, _, _, _))
+      .Times(2)
+      .WillRepeatedly(DoAll(SetArgPointee<2>(kPayloadType),
+                            WithArg<0>(Invoke(DeletePacketsAndReturnOk))));
+  // SetArgPointee<2>(kPayloadType) means that the third argument (zero-based
+  // index) is a pointer, and the variable pointed to is set to kPayloadType.
+  // Also invoke the function DeletePacketsAndReturnOk to properly delete all
+  // packets in the list (to avoid memory leaks in the test).
+  EXPECT_CALL(*mock_packet_buffer_, PeekNextPacket())
+      .Times(1)
+      .WillOnce(Return(&fake_packet));
+
+  // Expectations for DTMF buffer.
+  EXPECT_CALL(*mock_dtmf_buffer_, Flush()).Times(1);
+
+  // Expectations for delay manager.
+  {
+    // All expectations within this block must be called in this specific order.
+    InSequence sequence;  // Dummy variable.
+    // Expectations when the first packet is inserted.
+    EXPECT_CALL(
+        *mock_neteq_controller_,
+        PacketArrived(
+            /*fs_hz*/ 8000,
+            /*should_update_stats*/ _,
+            /*info*/
+            AllOf(
+                Field(&NetEqController::PacketArrivedInfo::is_cng_or_dtmf,
+                      false),
+                Field(&NetEqController::PacketArrivedInfo::main_sequence_number,
+                      kFirstSequenceNumber),
+                Field(&NetEqController::PacketArrivedInfo::main_timestamp,
+                      kFirstTimestamp))));
+    EXPECT_CALL(
+        *mock_neteq_controller_,
+        PacketArrived(
+            /*fs_hz*/ 8000,
+            /*should_update_stats*/ _,
+            /*info*/
+            AllOf(
+                Field(&NetEqController::PacketArrivedInfo::is_cng_or_dtmf,
+                      false),
+                Field(&NetEqController::PacketArrivedInfo::main_sequence_number,
+                      kFirstSequenceNumber + 1),
+                Field(&NetEqController::PacketArrivedInfo::main_timestamp,
+                      kFirstTimestamp + 160))));
+  }
+
+  // Insert first packet.
+  neteq_->InsertPacket(rtp_header, payload);
+
+  // Insert second packet.
+  rtp_header.timestamp += 160;
+  rtp_header.sequenceNumber += 1;
+  neteq_->InsertPacket(rtp_header, payload);
+}
+
+TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
+  UseNoMocks();
+  CreateInstance();
+
+  const int kPayloadLengthSamples = 80;
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;  // PCM 16-bit.
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+
+  // Insert packets. The buffer should not flush.
+  for (size_t i = 1; i <= config_.max_packets_in_buffer; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+    rtp_header.timestamp += kPayloadLengthSamples;
+    rtp_header.sequenceNumber += 1;
+    EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
+  }
+
+  // Insert one more packet and make sure the buffer got flushed. That is, it
+  // should only hold one single packet.
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
+  const Packet* test_packet = packet_buffer_->PeekNextPacket();
+  EXPECT_EQ(rtp_header.timestamp, test_packet->timestamp);
+  EXPECT_EQ(rtp_header.sequenceNumber, test_packet->sequence_number);
+}
+
+TEST_F(NetEqImplTest, TestDtmfPacketAVT) {
+  TestDtmfPacket(8000);
+}
+
+TEST_F(NetEqImplTest, TestDtmfPacketAVT16kHz) {
+  TestDtmfPacket(16000);
+}
+
+TEST_F(NetEqImplTest, TestDtmfPacketAVT32kHz) {
+  TestDtmfPacket(32000);
+}
+
+TEST_F(NetEqImplTest, TestDtmfPacketAVT48kHz) {
+  TestDtmfPacket(48000);
+}
+
+// This test verifies that timestamps propagate from the incoming packets
+// through to the sync buffer and to the playout timestamp.
+TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+  rtp_header.numCSRCs = 3;
+  rtp_header.arrOfCSRCs[0] = 43;
+  rtp_header.arrOfCSRCs[1] = 65;
+  rtp_header.arrOfCSRCs[2] = 17;
+
+  // This is a dummy decoder that produces as many output samples as the input
+  // has bytes. The output is an increasing series, starting at 1 for the first
+  // sample, and then increasing by 1 for each sample.
+  class CountingSamplesDecoder : public AudioDecoder {
+   public:
+    CountingSamplesDecoder() : next_value_(1) {}
+
+    // Produce as many samples as input bytes (`encoded_len`).
+    int DecodeInternal(const uint8_t* encoded,
+                       size_t encoded_len,
+                       int /* sample_rate_hz */,
+                       int16_t* decoded,
+                       SpeechType* speech_type) override {
+      for (size_t i = 0; i < encoded_len; ++i) {
+        decoded[i] = next_value_++;
+      }
+      *speech_type = kSpeech;
+      return rtc::checked_cast<int>(encoded_len);
+    }
+
+    void Reset() override { next_value_ = 1; }
+
+    int SampleRateHz() const override { return kSampleRateHz; }
+
+    size_t Channels() const override { return 1; }
+
+    uint16_t next_value() const { return next_value_; }
+
+   private:
+    int16_t next_value_;
+  } decoder_;
+
+  auto decoder_factory =
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&decoder_);
+
+  UseNoMocks();
+  CreateInstance(decoder_factory);
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert one packet.
+  clock_.AdvanceTimeMilliseconds(123456);
+  Timestamp expected_receive_time = clock_.CurrentTime();
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_FALSE(muted);
+  ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+  // Verify `output.packet_infos_`.
+  ASSERT_THAT(output.packet_infos_, SizeIs(1));
+  {
+    const auto& packet_info = output.packet_infos_[0];
+    EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
+    EXPECT_THAT(packet_info.csrcs(), ElementsAre(43, 65, 17));
+    EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
+    EXPECT_FALSE(packet_info.audio_level().has_value());
+    EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
+  }
+
+  // Start with a simple check that the fake decoder is behaving as expected.
+  EXPECT_EQ(kPayloadLengthSamples,
+            static_cast<size_t>(decoder_.next_value() - 1));
+
+  // The value of the last of the output samples is the same as the number of
+  // samples played from the decoded packet. Thus, this number + the RTP
+  // timestamp should match the playout timestamp.
+  // Wrap the expected value in an absl::optional to compare them as such.
+  EXPECT_EQ(
+      absl::optional<uint32_t>(rtp_header.timestamp +
+                               output.data()[output.samples_per_channel_ - 1]),
+      neteq_->GetPlayoutTimestamp());
+
+  // Check the timestamp for the last value in the sync buffer. This should
+  // be one full frame length ahead of the RTP timestamp.
+  const SyncBuffer* sync_buffer = neteq_->sync_buffer_for_test();
+  ASSERT_TRUE(sync_buffer != NULL);
+  EXPECT_EQ(rtp_header.timestamp + kPayloadLengthSamples,
+            sync_buffer->end_timestamp());
+
+  // Check that the number of samples still to play from the sync buffer add
+  // up with what was already played out.
+  EXPECT_EQ(
+      kPayloadLengthSamples - output.data()[output.samples_per_channel_ - 1],
+      sync_buffer->FutureLength());
+}
+
+TEST_F(NetEqImplTest, ReorderedPacket) {
+  UseNoMocks();
+
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+  rtp_header.extension.hasAudioLevel = true;
+  rtp_header.extension.audioLevel = 42;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, kPayloadLengthBytes))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
+  int16_t dummy_output[kPayloadLengthSamples] = {0};
+  // The below expectation will make the mock decoder write
+  // `kPayloadLengthSamples` zeros to the output array, and mark it as speech.
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
+                                           kSampleRateHz, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+                                          dummy_output + kPayloadLengthSamples),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
+                      Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert one packet.
+  clock_.AdvanceTimeMilliseconds(123456);
+  Timestamp expected_receive_time = clock_.CurrentTime();
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+  // Verify `output.packet_infos_`.
+  ASSERT_THAT(output.packet_infos_, SizeIs(1));
+  {
+    const auto& packet_info = output.packet_infos_[0];
+    EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
+    EXPECT_THAT(packet_info.csrcs(), IsEmpty());
+    EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
+    EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
+    EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
+  }
+
+  // Insert two more packets. The first one is out of order, and is already too
+  // old, the second one is the expected next packet.
+  rtp_header.sequenceNumber -= 1;
+  rtp_header.timestamp -= kPayloadLengthSamples;
+  rtp_header.extension.audioLevel = 1;
+  payload[0] = 1;
+  clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  rtp_header.sequenceNumber += 2;
+  rtp_header.timestamp += 2 * kPayloadLengthSamples;
+  rtp_header.extension.audioLevel = 2;
+  payload[0] = 2;
+  clock_.AdvanceTimeMilliseconds(2000);
+  expected_receive_time = clock_.CurrentTime();
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  // Expect only the second packet to be decoded (the one with "2" as the first
+  // payload byte).
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
+                                           kSampleRateHz, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+                                          dummy_output + kPayloadLengthSamples),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
+                      Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+
+  // Pull audio once.
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+  // Now check the packet buffer, and make sure it is empty, since the
+  // out-of-order packet should have been discarded.
+  EXPECT_TRUE(packet_buffer_->Empty());
+
+  // NetEq `packets_discarded` should capture this packet discard.
+  EXPECT_EQ(1u, neteq_->GetLifetimeStatistics().packets_discarded);
+
+  // Verify `output.packet_infos_`. Expect to only see the second packet.
+  ASSERT_THAT(output.packet_infos_, SizeIs(1));
+  {
+    const auto& packet_info = output.packet_infos_[0];
+    EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
+    EXPECT_THAT(packet_info.csrcs(), IsEmpty());
+    EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
+    EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
+    EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
+  }
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// This test verifies that NetEq can handle the situation where the first
+// incoming packet is rejected.
+TEST_F(NetEqImplTest, FirstPacketUnknown) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  // Insert one packet. Note that we have not registered any payload type, so
+  // this packet will be rejected.
+  EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_header, payload));
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
+  EXPECT_THAT(output.packet_infos_, IsEmpty());
+
+  // Register the payload type.
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+
+  // Insert 10 packets.
+  for (size_t i = 0; i < 10; ++i) {
+    rtp_header.sequenceNumber++;
+    rtp_header.timestamp += kPayloadLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+    EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
+  }
+
+  // Pull audio repeatedly and make sure we get normal output, that is not PLC.
+  for (size_t i = 0; i < 3; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
+    EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+    EXPECT_EQ(1u, output.num_channels_);
+    EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_)
+        << "NetEq did not decode the packets as expected.";
+    EXPECT_THAT(output.packet_infos_, SizeIs(1));
+  }
+}
+
+// This test verifies that audio interruption is not logged for the initial
+// PLC period before the first packet is deocoded.
+// TODO(henrik.lundin) Maybe move this test to neteq_network_stats_unittest.cc.
+// Make the test parametrized, so that we can test with different initial
+// sample rates in NetEq.
+class NetEqImplTestSampleRateParameter
+    : public NetEqImplTest,
+      public testing::WithParamInterface<int> {
+ protected:
+  NetEqImplTestSampleRateParameter()
+      : NetEqImplTest(), initial_sample_rate_hz_(GetParam()) {
+    config_.sample_rate_hz = initial_sample_rate_hz_;
+  }
+
+  const int initial_sample_rate_hz_;
+};
+
+class NetEqImplTestSdpFormatParameter
+    : public NetEqImplTest,
+      public testing::WithParamInterface<SdpAudioFormat> {
+ protected:
+  NetEqImplTestSdpFormatParameter()
+      : NetEqImplTest(), sdp_format_(GetParam()) {}
+  const SdpAudioFormat sdp_format_;
+};
+
+// This test does the following:
+// 0. Set up NetEq with initial sample rate given by test parameter, and a codec
+//    sample rate of 16000.
+// 1. Start calling GetAudio before inserting any encoded audio. The audio
+//    produced will be PLC.
+// 2. Insert a number of encoded audio packets.
+// 3. Keep calling GetAudio and verify that no audio interruption was logged.
+//    Call GetAudio until NetEq runs out of data again; PLC starts.
+// 4. Insert one more packet.
+// 5. Call GetAudio until that packet is decoded and the PLC ends.
+
+TEST_P(NetEqImplTestSampleRateParameter,
+       NoAudioInterruptionLoggedBeforeFirstDecode) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kPayloadSampleRateHz = 16000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kPayloadSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  // Register the payload type.
+  EXPECT_TRUE(neteq_->RegisterPayloadType(
+      kPayloadType, SdpAudioFormat("l16", kPayloadSampleRateHz, 1)));
+
+  // Pull audio several times. No packets have been inserted yet.
+  const size_t initial_output_size =
+      static_cast<size_t>(10 * initial_sample_rate_hz_ / 1000);  // 10 ms
+  AudioFrame output;
+  bool muted;
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    EXPECT_EQ(initial_output_size, output.samples_per_channel_);
+    EXPECT_EQ(1u, output.num_channels_);
+    EXPECT_NE(AudioFrame::kNormalSpeech, output.speech_type_);
+    EXPECT_THAT(output.packet_infos_, IsEmpty());
+  }
+
+  // Lambda for inserting packets.
+  auto insert_packet = [&]() {
+    rtp_header.sequenceNumber++;
+    rtp_header.timestamp += kPayloadLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  };
+  // Insert 10 packets.
+  for (size_t i = 0; i < 10; ++i) {
+    insert_packet();
+    EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
+  }
+
+  // Pull audio repeatedly and make sure we get normal output, that is not PLC.
+  constexpr size_t kOutputSize =
+      static_cast<size_t>(10 * kPayloadSampleRateHz / 1000);  // 10 ms
+  for (size_t i = 0; i < 3; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    EXPECT_EQ(kOutputSize, output.samples_per_channel_);
+    EXPECT_EQ(1u, output.num_channels_);
+    EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_)
+        << "NetEq did not decode the packets as expected.";
+    EXPECT_THAT(output.packet_infos_, SizeIs(1));
+  }
+
+  // Verify that no interruption was logged.
+  auto lifetime_stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(0, lifetime_stats.interruption_count);
+
+  // Keep pulling audio data until a new PLC period is started.
+  size_t count_loops = 0;
+  while (output.speech_type_ == AudioFrame::kNormalSpeech) {
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  }
+
+  // Insert a few packets to avoid postpone decoding after expand.
+  for (size_t i = 0; i < 5; ++i) {
+    insert_packet();
+  }
+
+  // Pull audio until the newly inserted packet is decoded and the PLC ends.
+  while (output.speech_type_ != AudioFrame::kNormalSpeech) {
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  }
+
+  // Verify that no interruption was logged.
+  lifetime_stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(0, lifetime_stats.interruption_count);
+}
+
+// This test does the following:
+// 0. Set up NetEq with initial sample rate given by test parameter, and a codec
+//    sample rate of 16000.
+// 1. Insert a number of encoded audio packets.
+// 2. Call GetAudio and verify that decoded audio is produced.
+// 3. Keep calling GetAudio until NetEq runs out of data; PLC starts.
+// 4. Keep calling GetAudio until PLC has been produced for at least 150 ms.
+// 5. Insert one more packet.
+// 6. Call GetAudio until that packet is decoded and the PLC ends.
+// 7. Verify that an interruption was logged.
+
+TEST_P(NetEqImplTestSampleRateParameter, AudioInterruptionLogged) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;  // Just an arbitrary number.
+  const int kPayloadSampleRateHz = 16000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kPayloadSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  // Register the payload type.
+  EXPECT_TRUE(neteq_->RegisterPayloadType(
+      kPayloadType, SdpAudioFormat("l16", kPayloadSampleRateHz, 1)));
+
+  // Lambda for inserting packets.
+  auto insert_packet = [&]() {
+    rtp_header.sequenceNumber++;
+    rtp_header.timestamp += kPayloadLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  };
+  // Insert 10 packets.
+  for (size_t i = 0; i < 10; ++i) {
+    insert_packet();
+    EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
+  }
+
+  AudioFrame output;
+  bool muted;
+  // Keep pulling audio data until a new PLC period is started.
+  size_t count_loops = 0;
+  do {
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  } while (output.speech_type_ == AudioFrame::kNormalSpeech);
+
+  // Pull audio 15 times, which produces 150 ms of output audio. This should
+  // all be produced as PLC. The total length of the gap will then be 150 ms
+  // plus an initial fraction of 10 ms at the start and the end of the PLC
+  // period. In total, less than 170 ms.
+  for (size_t i = 0; i < 15; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    EXPECT_NE(AudioFrame::kNormalSpeech, output.speech_type_);
+  }
+
+  // Insert a few packets to avoid postpone decoding after expand.
+  for (size_t i = 0; i < 5; ++i) {
+    insert_packet();
+  }
+
+  // Pull audio until the newly inserted packet is decoded and the PLC ends.
+  while (output.speech_type_ != AudioFrame::kNormalSpeech) {
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  }
+
+  // Verify that the interruption was logged.
+  auto lifetime_stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(1, lifetime_stats.interruption_count);
+  EXPECT_GT(lifetime_stats.total_interruption_duration_ms, 150);
+  EXPECT_LT(lifetime_stats.total_interruption_duration_ms, 170);
+}
+
+INSTANTIATE_TEST_SUITE_P(SampleRates,
+                         NetEqImplTestSampleRateParameter,
+                         testing::Values(8000, 16000, 32000, 48000));
+
+TEST_P(NetEqImplTestSdpFormatParameter, GetNackListScaledTimestamp) {
+  UseNoMocks();
+  CreateInstance();
+
+  neteq_->EnableNack(128);
+
+  const uint8_t kPayloadType = 17;  // Just an arbitrary number.
+  const int kPayloadSampleRateHz = sdp_format_.clockrate_hz;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kPayloadSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
+  std::vector<uint8_t> payload(kPayloadLengthBytes, 0);
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType, sdp_format_));
+
+  auto insert_packet = [&](bool lost = false) {
+    rtp_header.sequenceNumber++;
+    rtp_header.timestamp += kPayloadLengthSamples;
+    if (!lost)
+      EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  };
+
+  // Insert and decode 10 packets.
+  for (size_t i = 0; i < 10; ++i) {
+    insert_packet();
+  }
+  AudioFrame output;
+  size_t count_loops = 0;
+  do {
+    bool muted;
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  } while (output.speech_type_ == AudioFrame::kNormalSpeech);
+
+  insert_packet();
+
+  insert_packet(/*lost=*/true);
+
+  // Ensure packet gets marked as missing.
+  for (int i = 0; i < 5; ++i) {
+    insert_packet();
+  }
+
+  // Missing packet recoverable with 5ms RTT.
+  EXPECT_THAT(neteq_->GetNackList(5), Not(IsEmpty()));
+
+  // No packets should have TimeToPlay > 500ms.
+  EXPECT_THAT(neteq_->GetNackList(500), IsEmpty());
+}
+
+INSTANTIATE_TEST_SUITE_P(GetNackList,
+                         NetEqImplTestSdpFormatParameter,
+                         testing::Values(SdpAudioFormat("g722", 8000, 1),
+                                         SdpAudioFormat("opus", 48000, 2)));
+
+// This test verifies that NetEq can handle comfort noise and enters/quits codec
+// internal CNG mode properly.
+TEST_F(NetEqImplTest, CodecInternalCng) {
+  UseNoMocks();
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateKhz = 48;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(20 * kSampleRateKhz);  // 20 ms.
+  const size_t kPayloadLengthBytes = 10;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  int16_t dummy_output[kPayloadLengthSamples] = {0};
+
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateKhz * 1000));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, kPayloadLengthBytes))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
+  // Packed duration when asking the decoder for more CNG data (without a new
+  // packet).
+  EXPECT_CALL(mock_decoder, PacketDuration(nullptr, 0))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("opus", 48000, 2)));
+
+  struct Packet {
+    int sequence_number_delta;
+    int timestamp_delta;
+    AudioDecoder::SpeechType decoder_output_type;
+  };
+  std::vector<Packet> packets = {
+      {0, 0, AudioDecoder::kSpeech},
+      {1, kPayloadLengthSamples, AudioDecoder::kComfortNoise},
+      {2, 2 * kPayloadLengthSamples, AudioDecoder::kSpeech},
+      {1, kPayloadLengthSamples, AudioDecoder::kSpeech}};
+
+  for (size_t i = 0; i < packets.size(); ++i) {
+    rtp_header.sequenceNumber += packets[i].sequence_number_delta;
+    rtp_header.timestamp += packets[i].timestamp_delta;
+    payload[0] = i;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+    // Pointee(x) verifies that first byte of the payload equals x, this makes
+    // it possible to verify that the correct payload is fed to Decode().
+    EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(i), kPayloadLengthBytes,
+                                             kSampleRateKhz * 1000, _, _))
+        .WillOnce(DoAll(SetArrayArgument<3>(
+                            dummy_output, dummy_output + kPayloadLengthSamples),
+                        SetArgPointee<4>(packets[i].decoder_output_type),
+                        Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+  }
+
+  // Expect comfort noise to be returned by the decoder.
+  EXPECT_CALL(mock_decoder,
+              DecodeInternal(IsNull(), 0, kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+                                          dummy_output + kPayloadLengthSamples),
+                      SetArgPointee<4>(AudioDecoder::kComfortNoise),
+                      Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+
+  std::vector<AudioFrame::SpeechType> expected_output = {
+      AudioFrame::kNormalSpeech, AudioFrame::kCNG, AudioFrame::kNormalSpeech};
+  size_t output_index = 0;
+
+  int timeout_counter = 0;
+  while (!packet_buffer_->Empty()) {
+    ASSERT_LT(timeout_counter++, 20) << "Test timed out";
+    AudioFrame output;
+    bool muted;
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    if (output_index + 1 < expected_output.size() &&
+        output.speech_type_ == expected_output[output_index + 1]) {
+      ++output_index;
+    } else {
+      EXPECT_EQ(output.speech_type_, expected_output[output_index]);
+    }
+  }
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+TEST_F(NetEqImplTest, UnsupportedDecoder) {
+  UseNoMocks();
+  ::testing::NiceMock<MockAudioDecoder> decoder;
+
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&decoder));
+  static const size_t kNetEqMaxFrameSize = 5760;  // 120 ms @ 48 kHz.
+  static const size_t kChannels = 2;
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = 1;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  const uint8_t kFirstPayloadValue = 1;
+  const uint8_t kSecondPayloadValue = 2;
+
+  EXPECT_CALL(decoder,
+              PacketDuration(Pointee(kFirstPayloadValue), kPayloadLengthBytes))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kNetEqMaxFrameSize + 1)));
+
+  EXPECT_CALL(decoder, DecodeInternal(Pointee(kFirstPayloadValue), _, _, _, _))
+      .Times(0);
+
+  EXPECT_CALL(decoder, DecodeInternal(Pointee(kSecondPayloadValue),
+                                      kPayloadLengthBytes, kSampleRateHz, _, _))
+      .Times(1)
+      .WillOnce(DoAll(
+          SetArrayArgument<3>(dummy_output,
+                              dummy_output + kPayloadLengthSamples * kChannels),
+          SetArgPointee<4>(AudioDecoder::kSpeech),
+          Return(static_cast<int>(kPayloadLengthSamples * kChannels))));
+
+  EXPECT_CALL(decoder,
+              PacketDuration(Pointee(kSecondPayloadValue), kPayloadLengthBytes))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kNetEqMaxFrameSize)));
+
+  EXPECT_CALL(decoder, SampleRateHz()).WillRepeatedly(Return(kSampleRateHz));
+
+  EXPECT_CALL(decoder, Channels()).WillRepeatedly(Return(kChannels));
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert one packet.
+  payload[0] = kFirstPayloadValue;  // This will make Decode() fail.
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  // Insert another packet.
+  payload[0] = kSecondPayloadValue;  // This will make Decode() successful.
+  rtp_header.sequenceNumber++;
+  // The second timestamp needs to be at least 30 ms after the first to make
+  // the second packet get decoded.
+  rtp_header.timestamp += 3 * kPayloadLengthSamples;
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  AudioFrame output;
+  bool muted;
+  // First call to GetAudio will try to decode the "faulty" packet.
+  // Expect kFail return value.
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
+  // Output size and number of channels should be correct.
+  const size_t kExpectedOutputSize = 10 * (kSampleRateHz / 1000) * kChannels;
+  EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels);
+  EXPECT_EQ(kChannels, output.num_channels_);
+  EXPECT_THAT(output.packet_infos_, IsEmpty());
+
+  // Second call to GetAudio will decode the packet that is ok. No errors are
+  // expected.
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels);
+  EXPECT_EQ(kChannels, output.num_channels_);
+  EXPECT_THAT(output.packet_infos_, SizeIs(1));
+
+  // Die isn't called through NiceMock (since it's called by the
+  // MockAudioDecoder constructor), so it needs to be mocked explicitly.
+  EXPECT_CALL(decoder, Die());
+}
+
+// This test inserts packets until the buffer is flushed. After that, it asks
+// NetEq for the network statistics. The purpose of the test is to make sure
+// that even though the buffer size increment is negative (which it becomes when
+// the packet causing a flush is inserted), the packet length stored in the
+// decision logic remains valid.
+TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) {
+  UseNoMocks();
+  CreateInstance();
+
+  const size_t kPayloadLengthSamples = 80;
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;  // PCM 16-bit.
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+
+  // Insert packets until the buffer flushes.
+  for (size_t i = 0; i <= config_.max_packets_in_buffer; ++i) {
+    EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+    rtp_header.timestamp += rtc::checked_cast<uint32_t>(kPayloadLengthSamples);
+    ++rtp_header.sequenceNumber;
+  }
+  EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
+
+  // Ask for network statistics. This should not crash.
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(NetEq::kOK, neteq_->NetworkStatistics(&stats));
+}
+
+TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
+  UseNoMocks();
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
+  int16_t dummy_output[kPayloadLengthSamples] = {0};
+  // The below expectation will make the mock decoder write
+  // `kPayloadLengthSamples` - 5 zeros to the output array, and mark it as
+  // speech. That is, the decoded length is 5 samples shorter than the expected.
+  EXPECT_CALL(mock_decoder,
+              DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+      .WillOnce(
+          DoAll(SetArrayArgument<3>(dummy_output,
+                                    dummy_output + kPayloadLengthSamples - 5),
+                SetArgPointee<4>(AudioDecoder::kSpeech),
+                Return(rtc::checked_cast<int>(kPayloadLengthSamples - 5))));
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert one packet.
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+  EXPECT_THAT(output.packet_infos_, SizeIs(1));
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// This test checks the behavior of NetEq when audio decoder fails.
+TEST_F(NetEqImplTest, DecodingError) {
+  UseNoMocks();
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const int kDecoderErrorCode = -97;  // Any negative number.
+
+  // We let decoder return 5 ms each time, and therefore, 2 packets make 10 ms.
+  const size_t kFrameLengthSamples =
+      static_cast<size_t>(5 * kSampleRateHz / 1000);
+
+  const size_t kPayloadLengthBytes = 1;  // This can be arbitrary.
+
+  uint8_t payload[kPayloadLengthBytes] = {0};
+
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
+  EXPECT_CALL(mock_decoder, ErrorCode()).WillOnce(Return(kDecoderErrorCode));
+  EXPECT_CALL(mock_decoder, HasDecodePlc()).WillOnce(Return(false));
+  int16_t dummy_output[kFrameLengthSamples] = {0};
+
+  {
+    InSequence sequence;  // Dummy variable.
+    // Mock decoder works normally the first time.
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .Times(3)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kSpeech),
+                  Return(rtc::checked_cast<int>(kFrameLengthSamples))))
+        .RetiresOnSaturation();
+
+    // Then mock decoder fails. A common reason for failure can be buffer being
+    // too short
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .WillOnce(Return(-1))
+        .RetiresOnSaturation();
+
+    // Mock decoder finally returns to normal.
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .Times(2)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kSpeech),
+                  Return(rtc::checked_cast<int>(kFrameLengthSamples))));
+  }
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert packets.
+  for (int i = 0; i < 20; ++i) {
+    rtp_header.sequenceNumber += 1;
+    rtp_header.timestamp += kFrameLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  }
+
+  // Pull audio.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+  EXPECT_THAT(output.packet_infos_, SizeIs(2));  // 5 ms packets vs 10 ms output
+
+  // Pull audio again. Decoder fails.
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  // We are not expecting anything for output.speech_type_, since an error was
+  // returned.
+
+  // Pull audio again, should behave normal.
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+  EXPECT_THAT(output.packet_infos_, SizeIs(2));  // 5 ms packets vs 10 ms output
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// This test checks the behavior of NetEq when audio decoder fails during CNG.
+TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
+  UseNoMocks();
+
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const int kDecoderErrorCode = -97;  // Any negative number.
+
+  // We let decoder return 5 ms each time, and therefore, 2 packets make 10 ms.
+  const size_t kFrameLengthSamples =
+      static_cast<size_t>(5 * kSampleRateHz / 1000);
+
+  const size_t kPayloadLengthBytes = 1;  // This can be arbitrary.
+
+  uint8_t payload[kPayloadLengthBytes] = {0};
+
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
+  EXPECT_CALL(mock_decoder, ErrorCode()).WillOnce(Return(kDecoderErrorCode));
+  int16_t dummy_output[kFrameLengthSamples] = {0};
+
+  {
+    InSequence sequence;  // Dummy variable.
+    // Mock decoder works normally the first 2 times.
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .Times(2)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kComfortNoise),
+                  Return(rtc::checked_cast<int>(kFrameLengthSamples))))
+        .RetiresOnSaturation();
+
+    // Then mock decoder fails. A common reason for failure can be buffer being
+    // too short
+    EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
+        .WillOnce(Return(-1))
+        .RetiresOnSaturation();
+
+    // Mock decoder finally returns to normal.
+    EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
+        .Times(2)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kComfortNoise),
+                  Return(rtc::checked_cast<int>(kFrameLengthSamples))));
+  }
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+
+  // Insert 2 packets. This will make netEq into codec internal CNG mode.
+  for (int i = 0; i < 2; ++i) {
+    rtp_header.sequenceNumber += 1;
+    rtp_header.timestamp += kFrameLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  }
+
+  // Pull audio.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kCNG, output.speech_type_);
+
+  // Pull audio again. Decoder fails.
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  // We are not expecting anything for output.speech_type_, since an error was
+  // returned.
+
+  // Pull audio again, should resume codec CNG.
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kCNG, output.speech_type_);
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// Tests that the return value from last_output_sample_rate_hz() is equal to the
+// configured inital sample rate.
+TEST_F(NetEqImplTest, InitialLastOutputSampleRate) {
+  UseNoMocks();
+  config_.sample_rate_hz = 48000;
+  CreateInstance();
+  EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz());
+}
+
+TEST_F(NetEqImplTest, TickTimerIncrement) {
+  UseNoMocks();
+  CreateInstance();
+  ASSERT_TRUE(tick_timer_);
+  EXPECT_EQ(0u, tick_timer_->ticks());
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(1u, tick_timer_->ticks());
+}
+
+TEST_F(NetEqImplTest, SetBaseMinimumDelay) {
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+
+  EXPECT_CALL(*mock_neteq_controller_, SetBaseMinimumDelay(_))
+      .WillOnce(Return(true))
+      .WillOnce(Return(false));
+
+  const int delay_ms = 200;
+
+  EXPECT_EQ(true, neteq_->SetBaseMinimumDelayMs(delay_ms));
+  EXPECT_EQ(false, neteq_->SetBaseMinimumDelayMs(delay_ms));
+}
+
+TEST_F(NetEqImplTest, GetBaseMinimumDelayMs) {
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+
+  const int delay_ms = 200;
+
+  EXPECT_CALL(*mock_neteq_controller_, GetBaseMinimumDelay())
+      .WillOnce(Return(delay_ms));
+
+  EXPECT_EQ(delay_ms, neteq_->GetBaseMinimumDelayMs());
+}
+
+TEST_F(NetEqImplTest, TargetDelayMs) {
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+  constexpr int kTargetLevelMs = 510;
+  EXPECT_CALL(*mock_neteq_controller_, TargetLevelMs())
+      .WillOnce(Return(kTargetLevelMs));
+  EXPECT_EQ(510, neteq_->TargetDelayMs());
+}
+
+TEST_F(NetEqImplTest, InsertEmptyPacket) {
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+
+  RTPHeader rtp_header;
+  rtp_header.payloadType = 17;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(*mock_neteq_controller_, RegisterEmptyPacket());
+  neteq_->InsertEmptyPacket(rtp_header);
+}
+
+TEST_F(NetEqImplTest, NotifyControllerOfReorderedPacket) {
+  using ::testing::AllOf;
+  using ::testing::Field;
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .Times(1)
+      .WillOnce(Return(NetEq::Operation::kNormal));
+
+  const int kPayloadLengthSamples = 80;
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;  // PCM 16-bit.
+  const uint8_t kPayloadType = 17;  // Just an arbitrary number.
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+
+  // Insert second packet that was sent before the first packet.
+  rtp_header.sequenceNumber -= 1;
+  rtp_header.timestamp -= kPayloadLengthSamples;
+  EXPECT_CALL(
+      *mock_neteq_controller_,
+      PacketArrived(
+          /*fs_hz*/ 8000,
+          /*should_update_stats*/ true,
+          /*info*/
+          AllOf(
+              Field(&NetEqController::PacketArrivedInfo::packet_length_samples,
+                    kPayloadLengthSamples),
+              Field(&NetEqController::PacketArrivedInfo::main_sequence_number,
+                    rtp_header.sequenceNumber),
+              Field(&NetEqController::PacketArrivedInfo::main_timestamp,
+                    rtp_header.timestamp))));
+
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+}
+
+// When using a codec with 1000 channels, there should be no crashes.
+TEST_F(NetEqImplTest, NoCrashWith1000Channels) {
+  using ::testing::AllOf;
+  using ::testing::Field;
+  UseNoMocks();
+  use_mock_decoder_database_ = true;
+  enable_muted_state_ = true;
+  CreateInstance();
+  const size_t kPayloadLength = 100;
+  const uint8_t kPayloadType = 0;
+  const uint16_t kFirstSequenceNumber = 0x1234;
+  const uint32_t kFirstTimestamp = 0x12345678;
+  const uint32_t kSsrc = 0x87654321;
+  uint8_t payload[kPayloadLength] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = kFirstSequenceNumber;
+  rtp_header.timestamp = kFirstTimestamp;
+  rtp_header.ssrc = kSsrc;
+  Packet fake_packet;
+  fake_packet.payload_type = kPayloadType;
+  fake_packet.sequence_number = kFirstSequenceNumber;
+  fake_packet.timestamp = kFirstTimestamp;
+
+  AudioDecoder* decoder = nullptr;
+
+  auto mock_decoder_factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
+      .WillOnce(Invoke([&](const SdpAudioFormat& format,
+                           absl::optional<AudioCodecPairId> codec_pair_id,
+                           std::unique_ptr<AudioDecoder>* dec) {
+        EXPECT_EQ("pcmu", format.name);
+        *dec = std::make_unique<AudioDecoderPcmU>(1000);
+        decoder = dec->get();
+      }));
+  DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                    absl::nullopt, mock_decoder_factory.get());
+  // Expectations for decoder database.
+  EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
+      .WillRepeatedly(Return(&info));
+  EXPECT_CALL(*mock_decoder_database_, GetActiveCngDecoder())
+      .WillRepeatedly(ReturnNull());
+  EXPECT_CALL(*mock_decoder_database_, GetActiveDecoder())
+      .WillRepeatedly(Return(decoder));
+  EXPECT_CALL(*mock_decoder_database_, SetActiveDecoder(_, _))
+      .WillOnce(Invoke([](uint8_t rtp_payload_type, bool* new_decoder) {
+        *new_decoder = true;
+        return 0;
+      }));
+
+  // Insert first packet.
+  neteq_->InsertPacket(rtp_header, payload);
+
+  AudioFrame audio_frame;
+  bool muted;
+
+  // Repeat 40 times to ensure we enter muted state.
+  for (int i = 0; i < 40; i++) {
+    // GetAudio should return an error, and not crash, even in muted state.
+    EXPECT_NE(0, neteq_->GetAudio(&audio_frame, &muted));
+  }
+}
+
+class Decoder120ms : public AudioDecoder {
+ public:
+  Decoder120ms(int sample_rate_hz, SpeechType speech_type)
+      : sample_rate_hz_(sample_rate_hz),
+        next_value_(1),
+        speech_type_(speech_type) {}
+
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override {
+    EXPECT_EQ(sample_rate_hz_, sample_rate_hz);
+    size_t decoded_len =
+        rtc::CheckedDivExact(sample_rate_hz, 1000) * 120 * Channels();
+    for (size_t i = 0; i < decoded_len; ++i) {
+      decoded[i] = next_value_++;
+    }
+    *speech_type = speech_type_;
+    return rtc::checked_cast<int>(decoded_len);
+  }
+
+  void Reset() override { next_value_ = 1; }
+  int SampleRateHz() const override { return sample_rate_hz_; }
+  size_t Channels() const override { return 2; }
+
+ private:
+  int sample_rate_hz_;
+  int16_t next_value_;
+  SpeechType speech_type_;
+};
+
+class NetEqImplTest120ms : public NetEqImplTest {
+ protected:
+  NetEqImplTest120ms() : NetEqImplTest() {}
+  virtual ~NetEqImplTest120ms() {}
+
+  void CreateInstanceNoMocks() {
+    UseNoMocks();
+    CreateInstance(decoder_factory_);
+    EXPECT_TRUE(neteq_->RegisterPayloadType(
+        kPayloadType, SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}})));
+  }
+
+  void CreateInstanceWithDelayManagerMock() {
+    UseNoMocks();
+    use_mock_neteq_controller_ = true;
+    CreateInstance(decoder_factory_);
+    EXPECT_TRUE(neteq_->RegisterPayloadType(
+        kPayloadType, SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}})));
+  }
+
+  uint32_t timestamp_diff_between_packets() const {
+    return rtc::CheckedDivExact(kSamplingFreq_, 1000u) * 120;
+  }
+
+  uint32_t first_timestamp() const { return 10u; }
+
+  void GetFirstPacket() {
+    bool muted;
+    for (int i = 0; i < 12; i++) {
+      EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+      EXPECT_FALSE(muted);
+    }
+  }
+
+  void InsertPacket(uint32_t timestamp) {
+    RTPHeader rtp_header;
+    rtp_header.payloadType = kPayloadType;
+    rtp_header.sequenceNumber = sequence_number_;
+    rtp_header.timestamp = timestamp;
+    rtp_header.ssrc = 15;
+    const size_t kPayloadLengthBytes = 1;  // This can be arbitrary.
+    uint8_t payload[kPayloadLengthBytes] = {0};
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+    sequence_number_++;
+  }
+
+  void Register120msCodec(AudioDecoder::SpeechType speech_type) {
+    const uint32_t sampling_freq = kSamplingFreq_;
+    decoder_factory_ = rtc::make_ref_counted<test::FunctionAudioDecoderFactory>(
+        [sampling_freq, speech_type]() {
+          std::unique_ptr<AudioDecoder> decoder =
+              std::make_unique<Decoder120ms>(sampling_freq, speech_type);
+          RTC_CHECK_EQ(2, decoder->Channels());
+          return decoder;
+        });
+  }
+
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  AudioFrame output_;
+  const uint32_t kPayloadType = 17;
+  const uint32_t kSamplingFreq_ = 48000;
+  uint16_t sequence_number_ = 1;
+};
+
+TEST_F(NetEqImplTest120ms, CodecInternalCng) {
+  Register120msCodec(AudioDecoder::kComfortNoise);
+  CreateInstanceNoMocks();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kCodecInternalCng,
+            neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Normal) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceNoMocks();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  EXPECT_EQ(NetEq::Operation::kNormal, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Merge) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceWithDelayManagerMock();
+
+  EXPECT_CALL(*mock_neteq_controller_, CngOff()).WillRepeatedly(Return(true));
+  InsertPacket(first_timestamp());
+
+  GetFirstPacket();
+  bool muted;
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .WillOnce(Return(NetEq::Operation::kExpand));
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+
+  InsertPacket(first_timestamp() + 2 * timestamp_diff_between_packets());
+
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .WillOnce(Return(NetEq::Operation::kMerge));
+
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kMerge, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Expand) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceNoMocks();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kExpand, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, FastAccelerate) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceWithDelayManagerMock();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+  InsertPacket(first_timestamp() + timestamp_diff_between_packets());
+
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .Times(1)
+      .WillOnce(Return(NetEq::Operation::kFastAccelerate));
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kFastAccelerate,
+            neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, PreemptiveExpand) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceWithDelayManagerMock();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  InsertPacket(first_timestamp() + timestamp_diff_between_packets());
+
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .Times(1)
+      .WillOnce(Return(NetEq::Operation::kPreemptiveExpand));
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kPreemptiveExpand,
+            neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Accelerate) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceWithDelayManagerMock();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  InsertPacket(first_timestamp() + timestamp_diff_between_packets());
+
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .Times(1)
+      .WillOnce(Return(NetEq::Operation::kAccelerate));
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kAccelerate, neteq_->last_operation_for_test());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
new file mode 100644
index 0000000000..a669ad727e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -0,0 +1,345 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "absl/memory/memory.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "system_wrappers/include/clock.h"
+#include "test/audio_decoder_proxy_factory.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+std::unique_ptr<NetEq> CreateNetEq(
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+  return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
+}
+
+}  // namespace
+
+using ::testing::_;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+class MockAudioDecoder final : public AudioDecoder {
+ public:
+  static const int kPacketDuration = 960;  // 48 kHz * 20 ms
+
+  MockAudioDecoder(int sample_rate_hz, size_t num_channels)
+      : sample_rate_hz_(sample_rate_hz),
+        num_channels_(num_channels),
+        fec_enabled_(false) {}
+  ~MockAudioDecoder() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+
+  MOCK_METHOD(void, Reset, (), (override));
+
+  class MockFrame : public AudioDecoder::EncodedAudioFrame {
+   public:
+    MockFrame(size_t num_channels) : num_channels_(num_channels) {}
+
+    size_t Duration() const override { return kPacketDuration; }
+
+    absl::optional<DecodeResult> Decode(
+        rtc::ArrayView<int16_t> decoded) const override {
+      const size_t output_size =
+          sizeof(int16_t) * kPacketDuration * num_channels_;
+      if (decoded.size() >= output_size) {
+        memset(decoded.data(), 0,
+               sizeof(int16_t) * kPacketDuration * num_channels_);
+        return DecodeResult{kPacketDuration * num_channels_, kSpeech};
+      } else {
+        ADD_FAILURE() << "Expected decoded.size() to be >= output_size ("
+                      << decoded.size() << " vs. " << output_size << ")";
+        return absl::nullopt;
+      }
+    }
+
+   private:
+    const size_t num_channels_;
+  };
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override {
+    std::vector<ParseResult> results;
+    if (fec_enabled_) {
+      std::unique_ptr<MockFrame> fec_frame(new MockFrame(num_channels_));
+      results.emplace_back(timestamp - kPacketDuration, 1,
+                           std::move(fec_frame));
+    }
+
+    std::unique_ptr<MockFrame> frame(new MockFrame(num_channels_));
+    results.emplace_back(timestamp, 0, std::move(frame));
+    return results;
+  }
+
+  int PacketDuration(const uint8_t* encoded,
+                     size_t encoded_len) const override {
+    ADD_FAILURE() << "Since going through ParsePayload, PacketDuration should "
+                     "never get called.";
+    return kPacketDuration;
+  }
+
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override {
+    ADD_FAILURE() << "Since going through ParsePayload, PacketHasFec should "
+                     "never get called.";
+    return fec_enabled_;
+  }
+
+  int SampleRateHz() const override { return sample_rate_hz_; }
+
+  size_t Channels() const override { return num_channels_; }
+
+  void set_fec_enabled(bool enable_fec) { fec_enabled_ = enable_fec; }
+
+  bool fec_enabled() const { return fec_enabled_; }
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override {
+    ADD_FAILURE() << "Since going through ParsePayload, DecodeInternal should "
+                     "never get called.";
+    return -1;
+  }
+
+ private:
+  const int sample_rate_hz_;
+  const size_t num_channels_;
+  bool fec_enabled_;
+};
+
+class NetEqNetworkStatsTest {
+ public:
+  static const int kPayloadSizeByte = 30;
+  static const int kFrameSizeMs = 20;
+  static const uint8_t kPayloadType = 95;
+  static const int kOutputLengthMs = 10;
+
+  enum logic {
+    kIgnore,
+    kEqual,
+    kSmallerThan,
+    kLargerThan,
+  };
+
+  struct NetEqNetworkStatsCheck {
+    logic current_buffer_size_ms;
+    logic preferred_buffer_size_ms;
+    logic jitter_peaks_found;
+    logic packet_loss_rate;
+    logic expand_rate;
+    logic speech_expand_rate;
+    logic preemptive_rate;
+    logic accelerate_rate;
+    logic secondary_decoded_rate;
+    logic secondary_discarded_rate;
+    logic added_zero_samples;
+    NetEqNetworkStatistics stats_ref;
+  };
+
+  NetEqNetworkStatsTest(const SdpAudioFormat& format, MockAudioDecoder* decoder)
+      : decoder_(decoder),
+        decoder_factory_(
+            rtc::make_ref_counted<AudioDecoderProxyFactory>(decoder)),
+        samples_per_ms_(format.clockrate_hz / 1000),
+        frame_size_samples_(kFrameSizeMs * samples_per_ms_),
+        rtp_generator_(new RtpGenerator(samples_per_ms_)),
+        last_lost_time_(0),
+        packet_loss_interval_(0xffffffff) {
+    NetEq::Config config;
+    config.sample_rate_hz = format.clockrate_hz;
+    neteq_ = CreateNetEq(config, Clock::GetRealTimeClock(), decoder_factory_);
+    neteq_->RegisterPayloadType(kPayloadType, format);
+  }
+
+  bool Lost(uint32_t send_time) {
+    if (send_time - last_lost_time_ >= packet_loss_interval_) {
+      last_lost_time_ = send_time;
+      return true;
+    }
+    return false;
+  }
+
+  void SetPacketLossRate(double loss_rate) {
+    packet_loss_interval_ =
+        (loss_rate >= 1e-3 ? static_cast<double>(kFrameSizeMs) / loss_rate
+                           : 0xffffffff);
+  }
+
+  // `stats_ref`
+  // expects.x = -1, do not care
+  // expects.x = 0, 'x' in current stats should equal 'x' in `stats_ref`
+  // expects.x = 1, 'x' in current stats should < 'x' in `stats_ref`
+  // expects.x = 2, 'x' in current stats should > 'x' in `stats_ref`
+  void CheckNetworkStatistics(NetEqNetworkStatsCheck expects) {
+    NetEqNetworkStatistics stats;
+    neteq_->NetworkStatistics(&stats);
+
+#define CHECK_NETEQ_NETWORK_STATS(x)           \
+  switch (expects.x) {                         \
+    case kEqual:                               \
+      EXPECT_EQ(stats.x, expects.stats_ref.x); \
+      break;                                   \
+    case kSmallerThan:                         \
+      EXPECT_LT(stats.x, expects.stats_ref.x); \
+      break;                                   \
+    case kLargerThan:                          \
+      EXPECT_GT(stats.x, expects.stats_ref.x); \
+      break;                                   \
+    default:                                   \
+      break;                                   \
+  }
+
+    CHECK_NETEQ_NETWORK_STATS(current_buffer_size_ms);
+    CHECK_NETEQ_NETWORK_STATS(preferred_buffer_size_ms);
+    CHECK_NETEQ_NETWORK_STATS(jitter_peaks_found);
+    CHECK_NETEQ_NETWORK_STATS(expand_rate);
+    CHECK_NETEQ_NETWORK_STATS(speech_expand_rate);
+    CHECK_NETEQ_NETWORK_STATS(preemptive_rate);
+    CHECK_NETEQ_NETWORK_STATS(accelerate_rate);
+    CHECK_NETEQ_NETWORK_STATS(secondary_decoded_rate);
+    CHECK_NETEQ_NETWORK_STATS(secondary_discarded_rate);
+
+#undef CHECK_NETEQ_NETWORK_STATS
+  }
+
+  void RunTest(int num_loops, NetEqNetworkStatsCheck expects) {
+    uint32_t time_now;
+    uint32_t next_send_time;
+
+    // Initiate `last_lost_time_`.
+    time_now = next_send_time = last_lost_time_ = rtp_generator_->GetRtpHeader(
+        kPayloadType, frame_size_samples_, &rtp_header_);
+    for (int k = 0; k < num_loops; ++k) {
+      // Delay by one frame such that the FEC can come in.
+      while (time_now + kFrameSizeMs >= next_send_time) {
+        next_send_time = rtp_generator_->GetRtpHeader(
+            kPayloadType, frame_size_samples_, &rtp_header_);
+        if (!Lost(next_send_time)) {
+          static const uint8_t payload[kPayloadSizeByte] = {0};
+          ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header_, payload));
+        }
+      }
+      bool muted = true;
+      EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_frame_, &muted));
+      ASSERT_FALSE(muted);
+      EXPECT_EQ(decoder_->Channels(), output_frame_.num_channels_);
+      EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * samples_per_ms_),
+                output_frame_.samples_per_channel_);
+      EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz());
+
+      time_now += kOutputLengthMs;
+    }
+    CheckNetworkStatistics(expects);
+    neteq_->FlushBuffers();
+  }
+
+  void DecodeFecTest() {
+    decoder_->set_fec_enabled(false);
+    NetEqNetworkStatsCheck expects = {kIgnore,  // current_buffer_size_ms
+                                      kIgnore,  // preferred_buffer_size_ms
+                                      kIgnore,  // jitter_peaks_found
+                                      kEqual,   // packet_loss_rate
+                                      kEqual,   // expand_rate
+                                      kEqual,   // voice_expand_rate
+                                      kIgnore,  // preemptive_rate
+                                      kEqual,   // accelerate_rate
+                                      kEqual,   // decoded_fec_rate
+                                      kEqual,   // discarded_fec_rate
+                                      kEqual,   // added_zero_samples
+                                      {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}};
+    RunTest(50, expects);
+
+    // Next we introduce packet losses.
+    SetPacketLossRate(0.1);
+    expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 898;
+    RunTest(50, expects);
+
+    // Next we enable FEC.
+    decoder_->set_fec_enabled(true);
+    // If FEC fills in the lost packets, no packet loss will be counted.
+    expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 0;
+    expects.stats_ref.secondary_decoded_rate = 2006;
+    expects.stats_ref.secondary_discarded_rate = 14336;
+    RunTest(50, expects);
+  }
+
+  void NoiseExpansionTest() {
+    NetEqNetworkStatsCheck expects = {kIgnore,  // current_buffer_size_ms
+                                      kIgnore,  // preferred_buffer_size_ms
+                                      kIgnore,  // jitter_peaks_found
+                                      kEqual,   // packet_loss_rate
+                                      kEqual,   // expand_rate
+                                      kEqual,   // speech_expand_rate
+                                      kIgnore,  // preemptive_rate
+                                      kEqual,   // accelerate_rate
+                                      kEqual,   // decoded_fec_rate
+                                      kEqual,   // discard_fec_rate
+                                      kEqual,   // added_zero_samples
+                                      {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}};
+    RunTest(50, expects);
+
+    SetPacketLossRate(1);
+    expects.stats_ref.expand_rate = 16384;
+    expects.stats_ref.speech_expand_rate = 5324;
+    RunTest(10, expects);  // Lost 10 * 20ms in a row.
+  }
+
+ private:
+  MockAudioDecoder* decoder_;
+  rtc::scoped_refptr<AudioDecoderProxyFactory> decoder_factory_;
+  std::unique_ptr<NetEq> neteq_;
+
+  const int samples_per_ms_;
+  const size_t frame_size_samples_;
+  std::unique_ptr<RtpGenerator> rtp_generator_;
+  RTPHeader rtp_header_;
+  uint32_t last_lost_time_;
+  uint32_t packet_loss_interval_;
+  AudioFrame output_frame_;
+};
+
+TEST(NetEqNetworkStatsTest, DecodeFec) {
+  MockAudioDecoder decoder(48000, 1);
+  NetEqNetworkStatsTest test(SdpAudioFormat("opus", 48000, 2), &decoder);
+  test.DecodeFecTest();
+  EXPECT_CALL(decoder, Die()).Times(1);
+}
+
+TEST(NetEqNetworkStatsTest, StereoDecodeFec) {
+  MockAudioDecoder decoder(48000, 2);
+  NetEqNetworkStatsTest test(SdpAudioFormat("opus", 48000, 2), &decoder);
+  test.DecodeFecTest();
+  EXPECT_CALL(decoder, Die()).Times(1);
+}
+
+TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
+  MockAudioDecoder decoder(48000, 1);
+  NetEqNetworkStatsTest test(SdpAudioFormat("opus", 48000, 2), &decoder);
+  test.NoiseExpansionTest();
+  EXPECT_CALL(decoder, Die()).Times(1);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
new file mode 100644
index 0000000000..6fa56fd1c1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -0,0 +1,424 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Test to verify correct stereo and multi-channel operation.
+
+#include <algorithm>
+#include <list>
+#include <memory>
+#include <string>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+struct TestParameters {
+  int frame_size;
+  int sample_rate;
+  size_t num_channels;
+};
+
+// This is a parameterized test. The test parameters are supplied through a
+// TestParameters struct, which is obtained through the GetParam() method.
+//
+// The objective of the test is to create a mono input signal and a
+// multi-channel input signal, where each channel is identical to the mono
+// input channel. The two input signals are processed through their respective
+// NetEq instances. After that, the output signals are compared. The expected
+// result is that each channel in the multi-channel output is identical to the
+// mono output.
+class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
+ protected:
+  static const int kTimeStepMs = 10;
+  static const size_t kMaxBlockSize = 480;  // 10 ms @ 48 kHz.
+  static const uint8_t kPayloadTypeMono = 95;
+  static const uint8_t kPayloadTypeMulti = 96;
+
+  NetEqStereoTest()
+      : num_channels_(GetParam().num_channels),
+        sample_rate_hz_(GetParam().sample_rate),
+        samples_per_ms_(sample_rate_hz_ / 1000),
+        frame_size_ms_(GetParam().frame_size),
+        frame_size_samples_(
+            static_cast<size_t>(frame_size_ms_ * samples_per_ms_)),
+        output_size_samples_(10 * samples_per_ms_),
+        clock_(0),
+        rtp_generator_mono_(samples_per_ms_),
+        rtp_generator_(samples_per_ms_),
+        payload_size_bytes_(0),
+        multi_payload_size_bytes_(0),
+        last_send_time_(0),
+        last_arrival_time_(0) {
+    NetEq::Config config;
+    config.sample_rate_hz = sample_rate_hz_;
+    DefaultNetEqFactory neteq_factory;
+    auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+    neteq_mono_ = neteq_factory.CreateNetEq(config, decoder_factory, &clock_);
+    neteq_ = neteq_factory.CreateNetEq(config, decoder_factory, &clock_);
+    input_ = new int16_t[frame_size_samples_];
+    encoded_ = new uint8_t[2 * frame_size_samples_];
+    input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
+    encoded_multi_channel_ =
+        new uint8_t[frame_size_samples_ * 2 * num_channels_];
+  }
+
+  ~NetEqStereoTest() {
+    delete[] input_;
+    delete[] encoded_;
+    delete[] input_multi_channel_;
+    delete[] encoded_multi_channel_;
+  }
+
+  virtual void SetUp() {
+    const std::string file_name =
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+    input_file_.reset(new test::InputAudioFile(file_name));
+    RTC_CHECK_GE(num_channels_, 2);
+    ASSERT_TRUE(neteq_mono_->RegisterPayloadType(
+        kPayloadTypeMono, SdpAudioFormat("l16", sample_rate_hz_, 1)));
+    ASSERT_TRUE(neteq_->RegisterPayloadType(
+        kPayloadTypeMulti,
+        SdpAudioFormat("l16", sample_rate_hz_, num_channels_)));
+  }
+
+  virtual void TearDown() {}
+
+  int GetNewPackets() {
+    if (!input_file_->Read(frame_size_samples_, input_)) {
+      return -1;
+    }
+    payload_size_bytes_ =
+        WebRtcPcm16b_Encode(input_, frame_size_samples_, encoded_);
+    if (frame_size_samples_ * 2 != payload_size_bytes_) {
+      return -1;
+    }
+    int next_send_time = rtp_generator_mono_.GetRtpHeader(
+        kPayloadTypeMono, frame_size_samples_, &rtp_header_mono_);
+    MakeMultiChannelInput();
+    multi_payload_size_bytes_ = WebRtcPcm16b_Encode(
+        input_multi_channel_, frame_size_samples_ * num_channels_,
+        encoded_multi_channel_);
+    if (frame_size_samples_ * 2 * num_channels_ != multi_payload_size_bytes_) {
+      return -1;
+    }
+    rtp_generator_.GetRtpHeader(kPayloadTypeMulti, frame_size_samples_,
+                                &rtp_header_);
+    return next_send_time;
+  }
+
+  virtual void MakeMultiChannelInput() {
+    test::InputAudioFile::DuplicateInterleaved(
+        input_, frame_size_samples_, num_channels_, input_multi_channel_);
+  }
+
+  virtual void VerifyOutput(size_t num_samples) {
+    const int16_t* output_data = output_.data();
+    const int16_t* output_multi_channel_data = output_multi_channel_.data();
+    for (size_t i = 0; i < num_samples; ++i) {
+      for (size_t j = 0; j < num_channels_; ++j) {
+        ASSERT_EQ(output_data[i],
+                  output_multi_channel_data[i * num_channels_ + j])
+            << "Diff in sample " << i << ", channel " << j << ".";
+      }
+    }
+  }
+
+  virtual int GetArrivalTime(int send_time) {
+    int arrival_time = last_arrival_time_ + (send_time - last_send_time_);
+    last_send_time_ = send_time;
+    last_arrival_time_ = arrival_time;
+    return arrival_time;
+  }
+
+  virtual bool Lost() { return false; }
+
+  void RunTest(int num_loops) {
+    // Get next input packets (mono and multi-channel).
+    int next_send_time;
+    int next_arrival_time;
+    do {
+      next_send_time = GetNewPackets();
+      ASSERT_NE(-1, next_send_time);
+      next_arrival_time = GetArrivalTime(next_send_time);
+    } while (Lost());  // If lost, immediately read the next packet.
+
+    int time_now = 0;
+    for (int k = 0; k < num_loops; ++k) {
+      while (time_now >= next_arrival_time) {
+        // Insert packet in mono instance.
+        ASSERT_EQ(NetEq::kOK,
+                  neteq_mono_->InsertPacket(
+                      rtp_header_mono_, rtc::ArrayView<const uint8_t>(
+                                            encoded_, payload_size_bytes_)));
+        // Insert packet in multi-channel instance.
+        ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(
+                                  rtp_header_, rtc::ArrayView<const uint8_t>(
+                                                   encoded_multi_channel_,
+                                                   multi_payload_size_bytes_)));
+        // Get next input packets (mono and multi-channel).
+        do {
+          next_send_time = GetNewPackets();
+          ASSERT_NE(-1, next_send_time);
+          next_arrival_time = GetArrivalTime(next_send_time);
+        } while (Lost());  // If lost, immediately read the next packet.
+      }
+      // Get audio from mono instance.
+      bool muted;
+      EXPECT_EQ(NetEq::kOK, neteq_mono_->GetAudio(&output_, &muted));
+      ASSERT_FALSE(muted);
+      EXPECT_EQ(1u, output_.num_channels_);
+      EXPECT_EQ(output_size_samples_, output_.samples_per_channel_);
+      // Get audio from multi-channel instance.
+      ASSERT_EQ(NetEq::kOK, neteq_->GetAudio(&output_multi_channel_, &muted));
+      ASSERT_FALSE(muted);
+      EXPECT_EQ(num_channels_, output_multi_channel_.num_channels_);
+      EXPECT_EQ(output_size_samples_,
+                output_multi_channel_.samples_per_channel_);
+      rtc::StringBuilder ss;
+      ss << "Lap number " << k << ".";
+      SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+      // Compare mono and multi-channel.
+      ASSERT_NO_FATAL_FAILURE(VerifyOutput(output_size_samples_));
+
+      time_now += kTimeStepMs;
+      clock_.AdvanceTimeMilliseconds(kTimeStepMs);
+    }
+  }
+
+  const size_t num_channels_;
+  const int sample_rate_hz_;
+  const int samples_per_ms_;
+  const int frame_size_ms_;
+  const size_t frame_size_samples_;
+  const size_t output_size_samples_;
+  SimulatedClock clock_;
+  std::unique_ptr<NetEq> neteq_mono_;
+  std::unique_ptr<NetEq> neteq_;
+  test::RtpGenerator rtp_generator_mono_;
+  test::RtpGenerator rtp_generator_;
+  int16_t* input_;
+  int16_t* input_multi_channel_;
+  uint8_t* encoded_;
+  uint8_t* encoded_multi_channel_;
+  AudioFrame output_;
+  AudioFrame output_multi_channel_;
+  RTPHeader rtp_header_mono_;
+  RTPHeader rtp_header_;
+  size_t payload_size_bytes_;
+  size_t multi_payload_size_bytes_;
+  int last_send_time_;
+  int last_arrival_time_;
+  std::unique_ptr<test::InputAudioFile> input_file_;
+};
+
+class NetEqStereoTestNoJitter : public NetEqStereoTest {
+ protected:
+  NetEqStereoTestNoJitter() : NetEqStereoTest() {
+    // Start the sender 100 ms before the receiver to pre-fill the buffer.
+    // This is to avoid doing preemptive expand early in the test.
+    // TODO(hlundin): Mock the decision making instead to control the modes.
+    last_arrival_time_ = -100;
+  }
+};
+
+TEST_P(NetEqStereoTestNoJitter, RunTest) {
+  RunTest(8);
+}
+
+class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
+ protected:
+  NetEqStereoTestPositiveDrift() : NetEqStereoTest(), drift_factor(0.9) {
+    // Start the sender 100 ms before the receiver to pre-fill the buffer.
+    // This is to avoid doing preemptive expand early in the test.
+    // TODO(hlundin): Mock the decision making instead to control the modes.
+    last_arrival_time_ = -100;
+  }
+  virtual int GetArrivalTime(int send_time) {
+    int arrival_time =
+        last_arrival_time_ + drift_factor * (send_time - last_send_time_);
+    last_send_time_ = send_time;
+    last_arrival_time_ = arrival_time;
+    return arrival_time;
+  }
+
+  double drift_factor;
+};
+
+TEST_P(NetEqStereoTestPositiveDrift, RunTest) {
+  RunTest(100);
+}
+
+class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
+ protected:
+  NetEqStereoTestNegativeDrift() : NetEqStereoTestPositiveDrift() {
+    drift_factor = 1.1;
+    last_arrival_time_ = 0;
+  }
+};
+
+TEST_P(NetEqStereoTestNegativeDrift, RunTest) {
+  RunTest(100);
+}
+
+class NetEqStereoTestDelays : public NetEqStereoTest {
+ protected:
+  static const int kDelayInterval = 10;
+  static const int kDelay = 1000;
+  NetEqStereoTestDelays() : NetEqStereoTest(), frame_index_(0) {}
+
+  virtual int GetArrivalTime(int send_time) {
+    // Deliver immediately, unless we have a back-log.
+    int arrival_time = std::min(last_arrival_time_, send_time);
+    if (++frame_index_ % kDelayInterval == 0) {
+      // Delay this packet.
+      arrival_time += kDelay;
+    }
+    last_send_time_ = send_time;
+    last_arrival_time_ = arrival_time;
+    return arrival_time;
+  }
+
+  int frame_index_;
+};
+
+TEST_P(NetEqStereoTestDelays, RunTest) {
+  RunTest(1000);
+}
+
+class NetEqStereoTestLosses : public NetEqStereoTest {
+ protected:
+  static const int kLossInterval = 10;
+  NetEqStereoTestLosses() : NetEqStereoTest(), frame_index_(0) {}
+
+  virtual bool Lost() { return (++frame_index_) % kLossInterval == 0; }
+
+  // TODO(hlundin): NetEq is not giving bitexact results for these cases.
+  virtual void VerifyOutput(size_t num_samples) {
+    for (size_t i = 0; i < num_samples; ++i) {
+      const int16_t* output_data = output_.data();
+      const int16_t* output_multi_channel_data = output_multi_channel_.data();
+      auto first_channel_sample = output_multi_channel_data[i * num_channels_];
+      for (size_t j = 0; j < num_channels_; ++j) {
+        const int kErrorMargin = 200;
+        EXPECT_NEAR(output_data[i],
+                    output_multi_channel_data[i * num_channels_ + j],
+                    kErrorMargin)
+            << "Diff in sample " << i << ", channel " << j << ".";
+        EXPECT_EQ(first_channel_sample,
+                  output_multi_channel_data[i * num_channels_ + j]);
+      }
+    }
+  }
+
+  int frame_index_;
+};
+
+TEST_P(NetEqStereoTestLosses, RunTest) {
+  RunTest(100);
+}
+
+class NetEqStereoTestSingleActiveChannelPlc : public NetEqStereoTestLosses {
+ protected:
+  NetEqStereoTestSingleActiveChannelPlc() : NetEqStereoTestLosses() {}
+
+  virtual void MakeMultiChannelInput() override {
+    // Create a multi-channel input by copying the mono channel from file to the
+    // first channel, and setting the others to zero.
+    memset(input_multi_channel_, 0,
+           frame_size_samples_ * num_channels_ * sizeof(int16_t));
+    for (size_t i = 0; i < frame_size_samples_; ++i) {
+      input_multi_channel_[i * num_channels_] = input_[i];
+    }
+  }
+
+  virtual void VerifyOutput(size_t num_samples) override {
+    // Simply verify that all samples in channels other than the first are zero.
+    const int16_t* output_multi_channel_data = output_multi_channel_.data();
+    for (size_t i = 0; i < num_samples; ++i) {
+      for (size_t j = 1; j < num_channels_; ++j) {
+        EXPECT_EQ(0, output_multi_channel_data[i * num_channels_ + j])
+            << "Sample " << i << ", channel " << j << " is non-zero.";
+      }
+    }
+  }
+};
+
+TEST_P(NetEqStereoTestSingleActiveChannelPlc, RunTest) {
+  RunTest(100);
+}
+
+// Creates a list of parameter sets.
+std::list<TestParameters> GetTestParameters() {
+  std::list<TestParameters> l;
+  const int sample_rates[] = {8000, 16000, 32000};
+  const int num_rates = sizeof(sample_rates) / sizeof(sample_rates[0]);
+  // Loop through sample rates.
+  for (int rate_index = 0; rate_index < num_rates; ++rate_index) {
+    int sample_rate = sample_rates[rate_index];
+    // Loop through all frame sizes between 10 and 60 ms.
+    for (int frame_size = 10; frame_size <= 60; frame_size += 10) {
+      TestParameters p;
+      p.frame_size = frame_size;
+      p.sample_rate = sample_rate;
+      p.num_channels = 2;
+      l.push_back(p);
+      if (sample_rate == 8000) {
+        // Add a five-channel test for 8000 Hz.
+        p.num_channels = 5;
+        l.push_back(p);
+      }
+    }
+  }
+  return l;
+}
+
+// Pretty-printing the test parameters in case of an error.
+void PrintTo(const TestParameters& p, ::std::ostream* os) {
+  *os << "{frame_size = " << p.frame_size
+      << ", num_channels = " << p.num_channels
+      << ", sample_rate = " << p.sample_rate << "}";
+}
+
+// Instantiate the tests. Each test is instantiated using the function above,
+// so that all different parameter combinations are tested.
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestNoJitter,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestPositiveDrift,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestNegativeDrift,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestDelays,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestLosses,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestSingleActiveChannelPlc,
+                         ::testing::ValuesIn(GetTestParameters()));
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc
new file mode 100644
index 0000000000..fff14bf830
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -0,0 +1,1009 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/neteq/neteq.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>  // memset
+
+#include <algorithm>
+#include <memory>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/test/neteq_decoding_test.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/message_digest.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/arch.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(bool, gen_ref, false, "Generate reference files.");
+
+namespace webrtc {
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) &&         \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
+    (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
+    defined(WEBRTC_CODEC_ILBC)
+#define MAYBE_TestBitExactness TestBitExactness
+#else
+#define MAYBE_TestBitExactness DISABLED_TestBitExactness
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
+
+  const std::string output_checksum =
+      "dee7a10ab92526876a70a85bc48a4906901af3df";
+
+  const std::string network_stats_checksum =
+      "911dbf5fd97f48d25b8f0967286eb73c9d6f6158";
+
+  DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
+                   absl::GetFlag(FLAGS_gen_ref));
+}
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && defined(WEBRTC_CODEC_OPUS)
+#define MAYBE_TestOpusBitExactness TestOpusBitExactness
+#else
+#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
+
+  const std::string output_checksum =
+      "fec6827bb9ee0b21770bbbb4a3a6f8823bf537dc|"
+      "3610cc7be4b3407b9c273b1299ab7f8f47cca96b";
+
+  const std::string network_stats_checksum =
+      "3d043e47e5f4bb81d37e7bce8c44bf802965c853|"
+      "076662525572dba753b11578330bd491923f7f5e";
+
+  DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
+                   absl::GetFlag(FLAGS_gen_ref));
+}
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && defined(WEBRTC_CODEC_OPUS)
+#define MAYBE_TestOpusDtxBitExactness TestOpusDtxBitExactness
+#else
+#define MAYBE_TestOpusDtxBitExactness DISABLED_TestOpusDtxBitExactness
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp");
+
+  const std::string output_checksum =
+      "b3c4899eab5378ef5e54f2302948872149f6ad5e|"
+      "589e975ec31ea13f302457fea1425be9380ffb96";
+
+  const std::string network_stats_checksum =
+      "dc8447b9fee1a21fd5d1f4045d62b982a3fb0215";
+
+  DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
+                   absl::GetFlag(FLAGS_gen_ref));
+}
+
+// Use fax mode to avoid time-scaling. This is to simplify the testing of
+// packet waiting times in the packet buffer.
+class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
+ protected:
+  NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
+    config_.for_test_no_time_stretching = true;
+  }
+  void TestJitterBufferDelay(bool apply_packet_loss);
+};
+
+TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
+  // Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
+  size_t num_frames = 30;
+  const size_t kSamples = 10 * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  for (size_t i = 0; i < num_frames; ++i) {
+    const uint8_t payload[kPayloadBytes] = {0};
+    RTPHeader rtp_info;
+    rtp_info.sequenceNumber = rtc::checked_cast<uint16_t>(i);
+    rtp_info.timestamp = rtc::checked_cast<uint32_t>(i * kSamples);
+    rtp_info.ssrc = 0x1234;     // Just an arbitrary SSRC.
+    rtp_info.payloadType = 94;  // PCM16b WB codec.
+    rtp_info.markerBit = 0;
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+  }
+  // Pull out all data.
+  for (size_t i = 0; i < num_frames; ++i) {
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+  // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
+  // spacing (per definition), we expect the delay to increase with 10 ms for
+  // each packet. Thus, we are calculating the statistics for a series from 10
+  // to 300, in steps of 10 ms.
+  EXPECT_EQ(155, stats.mean_waiting_time_ms);
+  EXPECT_EQ(155, stats.median_waiting_time_ms);
+  EXPECT_EQ(10, stats.min_waiting_time_ms);
+  EXPECT_EQ(300, stats.max_waiting_time_ms);
+
+  // Check statistics again and make sure it's been reset.
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+  EXPECT_EQ(-1, stats.mean_waiting_time_ms);
+  EXPECT_EQ(-1, stats.median_waiting_time_ms);
+  EXPECT_EQ(-1, stats.min_waiting_time_ms);
+  EXPECT_EQ(-1, stats.max_waiting_time_ms);
+}
+
+
+TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDrift) {
+  // Apply a clock drift of -25 ms / s (sender faster than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
+  const double kNetworkFreezeTimeMs = 0.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 20;
+  const int kMaxTimeToSpeechMs = 100;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDrift) {
+  // Apply a clock drift of +25 ms / s (sender slower than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
+  const double kNetworkFreezeTimeMs = 0.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 40;
+  const int kMaxTimeToSpeechMs = 100;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDriftNetworkFreeze) {
+  // Apply a clock drift of -25 ms / s (sender faster than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
+  const double kNetworkFreezeTimeMs = 5000.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 60;
+  const int kMaxTimeToSpeechMs = 200;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreeze) {
+  // Apply a clock drift of +25 ms / s (sender slower than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
+  const double kNetworkFreezeTimeMs = 5000.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 40;
+  const int kMaxTimeToSpeechMs = 100;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreezeExtraPull) {
+  // Apply a clock drift of +25 ms / s (sender slower than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
+  const double kNetworkFreezeTimeMs = 5000.0;
+  const bool kGetAudioDuringFreezeRecovery = true;
+  const int kDelayToleranceMs = 40;
+  const int kMaxTimeToSpeechMs = 100;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithoutClockDrift) {
+  const double kDriftFactor = 1.0;  // No drift.
+  const double kNetworkFreezeTimeMs = 0.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 10;
+  const int kMaxTimeToSpeechMs = 50;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, UnknownPayloadType) {
+  const size_t kPayloadBytes = 100;
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+  PopulateRtpInfo(0, 0, &rtp_info);
+  rtp_info.payloadType = 1;  // Not registered as a decoder.
+  EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload));
+}
+
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#define MAYBE_DecoderError DecoderError
+#else
+#define MAYBE_DecoderError DISABLED_DecoderError
+#endif
+
+TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
+  const size_t kPayloadBytes = 100;
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+  PopulateRtpInfo(0, 0, &rtp_info);
+  rtp_info.payloadType = 103;  // iSAC, but the payload is invalid.
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+  // Set all of `out_data_` to 1, and verify that it was set to 0 by the call
+  // to GetAudio.
+  int16_t* out_frame_data = out_frame_.mutable_data();
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
+    out_frame_data[i] = 1;
+  }
+  bool muted;
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
+
+  // Verify that the first 160 samples are set to 0.
+  static const int kExpectedOutputLength = 160;  // 10 ms at 16 kHz sample rate.
+  const int16_t* const_out_frame_data = out_frame_.data();
+  for (int i = 0; i < kExpectedOutputLength; ++i) {
+    rtc::StringBuilder ss;
+    ss << "i = " << i;
+    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+    EXPECT_EQ(0, const_out_frame_data[i]);
+  }
+}
+
+TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
+  // Set all of `out_data_` to 1, and verify that it was set to 0 by the call
+  // to GetAudio.
+  int16_t* out_frame_data = out_frame_.mutable_data();
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
+    out_frame_data[i] = 1;
+  }
+  bool muted;
+  EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
+  // Verify that the first block of samples is set to 0.
+  static const int kExpectedOutputLength =
+      kInitSampleRateHz / 100;  // 10 ms at initial sample rate.
+  const int16_t* const_out_frame_data = out_frame_.data();
+  for (int i = 0; i < kExpectedOutputLength; ++i) {
+    rtc::StringBuilder ss;
+    ss << "i = " << i;
+    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+    EXPECT_EQ(0, const_out_frame_data[i]);
+  }
+  // Verify that the sample rate did not change from the initial configuration.
+  EXPECT_EQ(config_.sample_rate_hz, neteq_->last_output_sample_rate_hz());
+}
+
+class NetEqBgnTest : public NetEqDecodingTest {
+ protected:
+  void CheckBgn(int sampling_rate_hz) {
+    size_t expected_samples_per_channel = 0;
+    uint8_t payload_type = 0xFF;  // Invalid.
+    if (sampling_rate_hz == 8000) {
+      expected_samples_per_channel = kBlockSize8kHz;
+      payload_type = 93;  // PCM 16, 8 kHz.
+    } else if (sampling_rate_hz == 16000) {
+      expected_samples_per_channel = kBlockSize16kHz;
+      payload_type = 94;  // PCM 16, 16 kHZ.
+    } else if (sampling_rate_hz == 32000) {
+      expected_samples_per_channel = kBlockSize32kHz;
+      payload_type = 95;  // PCM 16, 32 kHz.
+    } else {
+      ASSERT_TRUE(false);  // Unsupported test case.
+    }
+
+    AudioFrame output;
+    test::AudioLoop input;
+    // We are using the same 32 kHz input file for all tests, regardless of
+    // `sampling_rate_hz`. The output may sound weird, but the test is still
+    // valid.
+    ASSERT_TRUE(input.Init(
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+        10 * sampling_rate_hz,  // Max 10 seconds loop length.
+        expected_samples_per_channel));
+
+    // Payload of 10 ms of PCM16 32 kHz.
+    uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
+    RTPHeader rtp_info;
+    PopulateRtpInfo(0, 0, &rtp_info);
+    rtp_info.payloadType = payload_type;
+
+    bool muted;
+    for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
+      auto block = input.GetNextBlock();
+      ASSERT_EQ(expected_samples_per_channel, block.size());
+      size_t enc_len_bytes =
+          WebRtcPcm16b_Encode(block.data(), block.size(), payload);
+      ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
+
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, enc_len_bytes)));
+      output.Reset();
+      ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+      ASSERT_EQ(1u, output.num_channels_);
+      ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
+      ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+      // Next packet.
+      rtp_info.timestamp +=
+          rtc::checked_cast<uint32_t>(expected_samples_per_channel);
+      rtp_info.sequenceNumber++;
+    }
+
+    output.Reset();
+
+    // Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull
+    // one frame without checking speech-type. This is the first frame pulled
+    // without inserting any packet, and might not be labeled as PLC.
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+    ASSERT_EQ(1u, output.num_channels_);
+    ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
+
+    // To be able to test the fading of background noise we need at lease to
+    // pull 611 frames.
+    const int kFadingThreshold = 611;
+
+    // Test several CNG-to-PLC packet for the expected behavior. The number 20
+    // is arbitrary, but sufficiently large to test enough number of frames.
+    const int kNumPlcToCngTestFrames = 20;
+    bool plc_to_cng = false;
+    for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) {
+      output.Reset();
+      // Set to non-zero.
+      memset(output.mutable_data(), 1, AudioFrame::kMaxDataSizeBytes);
+      ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+      ASSERT_FALSE(muted);
+      ASSERT_EQ(1u, output.num_channels_);
+      ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
+      if (output.speech_type_ == AudioFrame::kPLCCNG) {
+        plc_to_cng = true;
+        double sum_squared = 0;
+        const int16_t* output_data = output.data();
+        for (size_t k = 0;
+             k < output.num_channels_ * output.samples_per_channel_; ++k)
+          sum_squared += output_data[k] * output_data[k];
+        EXPECT_EQ(0, sum_squared);
+      } else {
+        EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
+      }
+    }
+    EXPECT_TRUE(plc_to_cng);  // Just to be sure that PLC-to-CNG has occurred.
+  }
+};
+
+TEST_F(NetEqBgnTest, RunTest) {
+  CheckBgn(8000);
+  CheckBgn(16000);
+  CheckBgn(32000);
+}
+
+TEST_F(NetEqDecodingTest, SequenceNumberWrap) {
+  // Start with a sequence number that will soon wrap.
+  std::set<uint16_t> drop_seq_numbers;  // Don't drop any packets.
+  WrapTest(0xFFFF - 10, 0, drop_seq_numbers, true, false);
+}
+
+TEST_F(NetEqDecodingTest, SequenceNumberWrapAndDrop) {
+  // Start with a sequence number that will soon wrap.
+  std::set<uint16_t> drop_seq_numbers;
+  drop_seq_numbers.insert(0xFFFF);
+  drop_seq_numbers.insert(0x0);
+  WrapTest(0xFFFF - 10, 0, drop_seq_numbers, true, false);
+}
+
+TEST_F(NetEqDecodingTest, TimestampWrap) {
+  // Start with a timestamp that will soon wrap.
+  std::set<uint16_t> drop_seq_numbers;
+  WrapTest(0, 0xFFFFFFFF - 3000, drop_seq_numbers, false, true);
+}
+
+TEST_F(NetEqDecodingTest, TimestampAndSequenceNumberWrap) {
+  // Start with a timestamp and a sequence number that will wrap at the same
+  // time.
+  std::set<uint16_t> drop_seq_numbers;
+  WrapTest(0xFFFF - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true);
+}
+
+TEST_F(NetEqDecodingTest, DiscardDuplicateCng) {
+  uint16_t seq_no = 0;
+  uint32_t timestamp = 0;
+  const int kFrameSizeMs = 10;
+  const int kSampleRateKhz = 16;
+  const int kSamples = kFrameSizeMs * kSampleRateKhz;
+  const size_t kPayloadBytes = kSamples * 2;
+
+  const int algorithmic_delay_samples =
+      std::max(algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
+  // Insert three speech packets. Three are needed to get the frame length
+  // correct.
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+  bool muted;
+  for (int i = 0; i < 3; ++i) {
+    PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+    ++seq_no;
+    timestamp += kSamples;
+
+    // Pull audio once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+  // Verify speech output.
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+
+  // Insert same CNG packet twice.
+  const int kCngPeriodMs = 100;
+  const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
+  size_t payload_len;
+  PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+  // This is the first time this CNG packet is inserted.
+  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                  payload, payload_len)));
+
+  // Pull audio once and make sure CNG is played.
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+  EXPECT_FALSE(
+      neteq_->GetPlayoutTimestamp());  // Returns empty value during CNG.
+  EXPECT_EQ(timestamp - algorithmic_delay_samples,
+            out_frame_.timestamp_ + out_frame_.samples_per_channel_);
+
+  // Insert the same CNG packet again. Note that at this point it is old, since
+  // we have already decoded the first copy of it.
+  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                  payload, payload_len)));
+
+  // Pull audio until we have played `kCngPeriodMs` of CNG. Start at 10 ms since
+  // we have already pulled out CNG once.
+  for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+    EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+    EXPECT_FALSE(
+        neteq_->GetPlayoutTimestamp());  // Returns empty value during CNG.
+    EXPECT_EQ(timestamp - algorithmic_delay_samples,
+              out_frame_.timestamp_ + out_frame_.samples_per_channel_);
+  }
+
+  ++seq_no;
+  timestamp += kCngPeriodSamples;
+  uint32_t first_speech_timestamp = timestamp;
+  // Insert speech again.
+  for (int i = 0; i < 3; ++i) {
+    PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+    ++seq_no;
+    timestamp += kSamples;
+  }
+
+  // Pull audio once and verify that the output is speech again.
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+  absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  EXPECT_EQ(first_speech_timestamp + kSamples - algorithmic_delay_samples,
+            *playout_timestamp);
+}
+
+TEST_F(NetEqDecodingTest, CngFirst) {
+  uint16_t seq_no = 0;
+  uint32_t timestamp = 0;
+  const int kFrameSizeMs = 10;
+  const int kSampleRateKhz = 16;
+  const int kSamples = kFrameSizeMs * kSampleRateKhz;
+  const int kPayloadBytes = kSamples * 2;
+  const int kCngPeriodMs = 100;
+  const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
+  size_t payload_len;
+
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+
+  PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+  ASSERT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(
+                rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len)));
+  ++seq_no;
+  timestamp += kCngPeriodSamples;
+
+  // Pull audio once and make sure CNG is played.
+  bool muted;
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+
+  // Insert some speech packets.
+  const uint32_t first_speech_timestamp = timestamp;
+  int timeout_counter = 0;
+  do {
+    ASSERT_LT(timeout_counter++, 20) << "Test timed out";
+    PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+    ++seq_no;
+    timestamp += kSamples;
+
+    // Pull audio once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  } while (!IsNewerTimestamp(out_frame_.timestamp_, first_speech_timestamp));
+  // Verify speech output.
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+}
+
+class NetEqDecodingTestWithMutedState : public NetEqDecodingTest {
+ public:
+  NetEqDecodingTestWithMutedState() : NetEqDecodingTest() {
+    config_.enable_muted_state = true;
+  }
+
+ protected:
+  static constexpr size_t kSamples = 10 * 16;
+  static constexpr size_t kPayloadBytes = kSamples * 2;
+
+  void InsertPacket(uint32_t rtp_timestamp) {
+    uint8_t payload[kPayloadBytes] = {0};
+    RTPHeader rtp_info;
+    PopulateRtpInfo(0, rtp_timestamp, &rtp_info);
+    EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+  }
+
+  void InsertCngPacket(uint32_t rtp_timestamp) {
+    uint8_t payload[kPayloadBytes] = {0};
+    RTPHeader rtp_info;
+    size_t payload_len;
+    PopulateCng(0, rtp_timestamp, &rtp_info, payload, &payload_len);
+    EXPECT_EQ(NetEq::kOK,
+              neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                 payload, payload_len)));
+  }
+
+  bool GetAudioReturnMuted() {
+    bool muted;
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    return muted;
+  }
+
+  void GetAudioUntilMuted() {
+    while (!GetAudioReturnMuted()) {
+      ASSERT_LT(counter_++, 1000) << "Test timed out";
+    }
+  }
+
+  void GetAudioUntilNormal() {
+    bool muted = false;
+    while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
+      EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+      ASSERT_LT(counter_++, 1000) << "Test timed out";
+    }
+    EXPECT_FALSE(muted);
+  }
+
+  int counter_ = 0;
+};
+
+// Verifies that NetEq goes in and out of muted state as expected.
+TEST_F(NetEqDecodingTestWithMutedState, MutedState) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+  EXPECT_TRUE(out_frame_.muted());
+
+  // Verify that output audio is not written during muted mode. Other parameters
+  // should be correct, though.
+  AudioFrame new_frame;
+  int16_t* frame_data = new_frame.mutable_data();
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
+    frame_data[i] = 17;
+  }
+  bool muted;
+  EXPECT_EQ(0, neteq_->GetAudio(&new_frame, &muted));
+  EXPECT_TRUE(muted);
+  EXPECT_TRUE(out_frame_.muted());
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
+    EXPECT_EQ(17, frame_data[i]);
+  }
+  EXPECT_EQ(out_frame_.timestamp_ + out_frame_.samples_per_channel_,
+            new_frame.timestamp_);
+  EXPECT_EQ(out_frame_.samples_per_channel_, new_frame.samples_per_channel_);
+  EXPECT_EQ(out_frame_.sample_rate_hz_, new_frame.sample_rate_hz_);
+  EXPECT_EQ(out_frame_.num_channels_, new_frame.num_channels_);
+  EXPECT_EQ(out_frame_.speech_type_, new_frame.speech_type_);
+  EXPECT_EQ(out_frame_.vad_activity_, new_frame.vad_activity_);
+
+  // Insert new data. Timestamp is corrected for the time elapsed since the last
+  // packet. Verify that normal operation resumes.
+  InsertPacket(kSamples * counter_);
+  GetAudioUntilNormal();
+  EXPECT_FALSE(out_frame_.muted());
+
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+  // NetEqNetworkStatistics::expand_rate tells the fraction of samples that were
+  // concealment samples, in Q14 (16384 = 100%) .The vast majority should be
+  // concealment samples in this test.
+  EXPECT_GT(stats.expand_rate, 14000);
+  // And, it should be greater than the speech_expand_rate.
+  EXPECT_GT(stats.expand_rate, stats.speech_expand_rate);
+}
+
+// Verifies that NetEq goes out of muted state when given a delayed packet.
+TEST_F(NetEqDecodingTestWithMutedState, MutedStateDelayedPacket) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+  // Insert new data. Timestamp is only corrected for the half of the time
+  // elapsed since the last packet. That is, the new packet is delayed. Verify
+  // that normal operation resumes.
+  InsertPacket(kSamples * counter_ / 2);
+  GetAudioUntilNormal();
+}
+
+// Verifies that NetEq goes out of muted state when given a future packet.
+TEST_F(NetEqDecodingTestWithMutedState, MutedStateFuturePacket) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+  // Insert new data. Timestamp is over-corrected for the time elapsed since the
+  // last packet. That is, the new packet is too early. Verify that normal
+  // operation resumes.
+  InsertPacket(kSamples * counter_ * 2);
+  GetAudioUntilNormal();
+}
+
+// Verifies that NetEq goes out of muted state when given an old packet.
+TEST_F(NetEqDecodingTestWithMutedState, MutedStateOldPacket) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+
+  EXPECT_NE(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+  // Insert a few packets which are older than the first packet.
+  for (int i = 0; i < 5; ++i) {
+    InsertPacket(kSamples * (i - 1000));
+  }
+  EXPECT_FALSE(GetAudioReturnMuted());
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+}
+
+// Verifies that NetEq doesn't enter muted state when CNG mode is active and the
+// packet stream is suspended for a long time.
+TEST_F(NetEqDecodingTestWithMutedState, DoNotMuteExtendedCngWithoutPackets) {
+  // Insert one CNG packet.
+  InsertCngPacket(0);
+
+  // Pull 10 seconds of audio (10 ms audio generated per lap).
+  for (int i = 0; i < 1000; ++i) {
+    bool muted;
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_FALSE(muted);
+  }
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+}
+
+// Verifies that NetEq goes back to normal after a long CNG period with the
+// packet stream suspended.
+TEST_F(NetEqDecodingTestWithMutedState, RecoverAfterExtendedCngWithoutPackets) {
+  // Insert one CNG packet.
+  InsertCngPacket(0);
+
+  // Pull 10 seconds of audio (10 ms audio generated per lap).
+  for (int i = 0; i < 1000; ++i) {
+    bool muted;
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  }
+
+  // Insert new data. Timestamp is corrected for the time elapsed since the last
+  // packet. Verify that normal operation resumes.
+  InsertPacket(kSamples * counter_);
+  GetAudioUntilNormal();
+}
+
+namespace {
+::testing::AssertionResult AudioFramesEqualExceptData(const AudioFrame& a,
+                                                      const AudioFrame& b) {
+  if (a.timestamp_ != b.timestamp_)
+    return ::testing::AssertionFailure() << "timestamp_ diff (" << a.timestamp_
+                                         << " != " << b.timestamp_ << ")";
+  if (a.sample_rate_hz_ != b.sample_rate_hz_)
+    return ::testing::AssertionFailure()
+           << "sample_rate_hz_ diff (" << a.sample_rate_hz_
+           << " != " << b.sample_rate_hz_ << ")";
+  if (a.samples_per_channel_ != b.samples_per_channel_)
+    return ::testing::AssertionFailure()
+           << "samples_per_channel_ diff (" << a.samples_per_channel_
+           << " != " << b.samples_per_channel_ << ")";
+  if (a.num_channels_ != b.num_channels_)
+    return ::testing::AssertionFailure()
+           << "num_channels_ diff (" << a.num_channels_
+           << " != " << b.num_channels_ << ")";
+  if (a.speech_type_ != b.speech_type_)
+    return ::testing::AssertionFailure()
+           << "speech_type_ diff (" << a.speech_type_
+           << " != " << b.speech_type_ << ")";
+  if (a.vad_activity_ != b.vad_activity_)
+    return ::testing::AssertionFailure()
+           << "vad_activity_ diff (" << a.vad_activity_
+           << " != " << b.vad_activity_ << ")";
+  return ::testing::AssertionSuccess();
+}
+
+::testing::AssertionResult AudioFramesEqual(const AudioFrame& a,
+                                            const AudioFrame& b) {
+  ::testing::AssertionResult res = AudioFramesEqualExceptData(a, b);
+  if (!res)
+    return res;
+  if (memcmp(a.data(), b.data(),
+             a.samples_per_channel_ * a.num_channels_ * sizeof(*a.data())) !=
+      0) {
+    return ::testing::AssertionFailure() << "data_ diff";
+  }
+  return ::testing::AssertionSuccess();
+}
+
+}  // namespace
+
+TEST_F(NetEqDecodingTestTwoInstances, CompareMutedStateOnOff) {
+  ASSERT_FALSE(config_.enable_muted_state);
+  config2_.enable_muted_state = true;
+  CreateSecondInstance();
+
+  // Insert one speech packet into both NetEqs.
+  const size_t kSamples = 10 * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+  PopulateRtpInfo(0, 0, &rtp_info);
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+  EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload));
+
+  AudioFrame out_frame1, out_frame2;
+  bool muted;
+  for (int i = 0; i < 1000; ++i) {
+    rtc::StringBuilder ss;
+    ss << "i = " << i;
+    SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
+    EXPECT_FALSE(muted);
+    EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
+    if (muted) {
+      EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
+    } else {
+      EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
+    }
+  }
+  EXPECT_TRUE(muted);
+
+  // Insert new data. Timestamp is corrected for the time elapsed since the last
+  // packet.
+  for (int i = 0; i < 5; ++i) {
+    PopulateRtpInfo(0, kSamples * 1000 + kSamples * i, &rtp_info);
+    EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+    EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload));
+  }
+
+  int counter = 0;
+  while (out_frame1.speech_type_ != AudioFrame::kNormalSpeech) {
+    ASSERT_LT(counter++, 1000) << "Test timed out";
+    rtc::StringBuilder ss;
+    ss << "counter = " << counter;
+    SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
+    EXPECT_FALSE(muted);
+    EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
+    if (muted) {
+      EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
+    } else {
+      EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
+    }
+  }
+  EXPECT_FALSE(muted);
+}
+
+TEST_F(NetEqDecodingTest, TestConcealmentEvents) {
+  const int kNumConcealmentEvents = 19;
+  const size_t kSamples = 10 * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  int seq_no = 0;
+  RTPHeader rtp_info;
+  rtp_info.ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info.payloadType = 94;  // PCM16b WB codec.
+  rtp_info.markerBit = 0;
+  const uint8_t payload[kPayloadBytes] = {0};
+  bool muted;
+
+  for (int i = 0; i < kNumConcealmentEvents; i++) {
+    // Insert some packets of 10 ms size.
+    for (int j = 0; j < 10; j++) {
+      rtp_info.sequenceNumber = seq_no++;
+      rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
+      neteq_->InsertPacket(rtp_info, payload);
+      neteq_->GetAudio(&out_frame_, &muted);
+    }
+
+    // Lose a number of packets.
+    int num_lost = 1 + i;
+    for (int j = 0; j < num_lost; j++) {
+      seq_no++;
+      neteq_->GetAudio(&out_frame_, &muted);
+    }
+  }
+
+  // Check number of concealment events.
+  NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(kNumConcealmentEvents, static_cast<int>(stats.concealment_events));
+}
+
+// Test that the jitter buffer delay stat is computed correctly.
+void NetEqDecodingTestFaxMode::TestJitterBufferDelay(bool apply_packet_loss) {
+  const int kNumPackets = 10;
+  const int kDelayInNumPackets = 2;
+  const int kPacketLenMs = 10;  // All packets are of 10 ms size.
+  const size_t kSamples = kPacketLenMs * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  RTPHeader rtp_info;
+  rtp_info.ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info.payloadType = 94;  // PCM16b WB codec.
+  rtp_info.markerBit = 0;
+  const uint8_t payload[kPayloadBytes] = {0};
+  bool muted;
+  int packets_sent = 0;
+  int packets_received = 0;
+  int expected_delay = 0;
+  int expected_target_delay = 0;
+  uint64_t expected_emitted_count = 0;
+  while (packets_received < kNumPackets) {
+    // Insert packet.
+    if (packets_sent < kNumPackets) {
+      rtp_info.sequenceNumber = packets_sent++;
+      rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
+      neteq_->InsertPacket(rtp_info, payload);
+    }
+
+    // Get packet.
+    if (packets_sent > kDelayInNumPackets) {
+      neteq_->GetAudio(&out_frame_, &muted);
+      packets_received++;
+
+      // The delay reported by the jitter buffer never exceeds
+      // the number of samples previously fetched with GetAudio
+      // (hence the min()).
+      int packets_delay = std::min(packets_received, kDelayInNumPackets + 1);
+
+      // The increase of the expected delay is the product of
+      // the current delay of the jitter buffer in ms * the
+      // number of samples that are sent for play out.
+      int current_delay_ms = packets_delay * kPacketLenMs;
+      expected_delay += current_delay_ms * kSamples;
+      expected_target_delay += neteq_->TargetDelayMs() * kSamples;
+      expected_emitted_count += kSamples;
+    }
+  }
+
+  if (apply_packet_loss) {
+    // Extra call to GetAudio to cause concealment.
+    neteq_->GetAudio(&out_frame_, &muted);
+  }
+
+  // Check jitter buffer delay.
+  NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(expected_delay,
+            rtc::checked_cast<int>(stats.jitter_buffer_delay_ms));
+  EXPECT_EQ(expected_emitted_count, stats.jitter_buffer_emitted_count);
+  EXPECT_EQ(expected_target_delay,
+            rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
+}
+
+TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithoutLoss) {
+  TestJitterBufferDelay(false);
+}
+
+TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithLoss) {
+  TestJitterBufferDelay(true);
+}
+
+TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithAcceleration) {
+  const int kPacketLenMs = 10;  // All packets are of 10 ms size.
+  const size_t kSamples = kPacketLenMs * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  RTPHeader rtp_info;
+  rtp_info.ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info.payloadType = 94;  // PCM16b WB codec.
+  rtp_info.markerBit = 0;
+  const uint8_t payload[kPayloadBytes] = {0};
+
+  int expected_target_delay = neteq_->TargetDelayMs() * kSamples;
+  neteq_->InsertPacket(rtp_info, payload);
+
+  bool muted;
+  neteq_->GetAudio(&out_frame_, &muted);
+
+  rtp_info.sequenceNumber += 1;
+  rtp_info.timestamp += kSamples;
+  neteq_->InsertPacket(rtp_info, payload);
+  rtp_info.sequenceNumber += 1;
+  rtp_info.timestamp += kSamples;
+  neteq_->InsertPacket(rtp_info, payload);
+
+  expected_target_delay += neteq_->TargetDelayMs() * 2 * kSamples;
+  // We have two packets in the buffer and kAccelerate operation will
+  // extract 20 ms of data.
+  neteq_->GetAudio(&out_frame_, &muted, nullptr, NetEq::Operation::kAccelerate);
+
+  // Check jitter buffer delay.
+  NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(10 * kSamples * 3, stats.jitter_buffer_delay_ms);
+  EXPECT_EQ(kSamples * 3, stats.jitter_buffer_emitted_count);
+  EXPECT_EQ(expected_target_delay,
+            rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
+}
+
+namespace test {
+TEST(NetEqNoTimeStretchingMode, RunTest) {
+  NetEq::Config config;
+  config.for_test_no_time_stretching = true;
+  auto codecs = NetEqTest::StandardDecoderMap();
+  NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
+      {1, kRtpExtensionAudioLevel},
+      {3, kRtpExtensionAbsoluteSendTime},
+      {5, kRtpExtensionTransportSequenceNumber},
+      {7, kRtpExtensionVideoContentType},
+      {8, kRtpExtensionVideoTiming}};
+  std::unique_ptr<NetEqInput> input(new NetEqRtpDumpInput(
+      webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"),
+      rtp_ext_map, absl::nullopt /*No SSRC filter*/));
+  std::unique_ptr<TimeLimitedNetEqInput> input_time_limit(
+      new TimeLimitedNetEqInput(std::move(input), 20000));
+  std::unique_ptr<AudioSink> output(new VoidAudioSink);
+  NetEqTest::Callbacks callbacks;
+  NetEqTest test(config, CreateBuiltinAudioDecoderFactory(), codecs,
+                 /*text_log=*/nullptr, /*neteq_factory=*/nullptr,
+                 /*input=*/std::move(input_time_limit), std::move(output),
+                 callbacks);
+  test.Run();
+  const auto stats = test.SimulationStats();
+  EXPECT_EQ(0, stats.accelerate_rate);
+  EXPECT_EQ(0, stats.preemptive_rate);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.proto b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.proto
new file mode 100644
index 0000000000..b4b4253c3d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.proto
@@ -0,0 +1,31 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.neteq_unittest;
+
+message NetEqNetworkStatistics {
+  // Next field number 18.
+  optional uint32 current_buffer_size_ms = 1;
+  optional uint32 preferred_buffer_size_ms = 2;
+  optional uint32 jitter_peaks_found = 3;
+  reserved 4;  // Was packet_loss_rate.
+  reserved 5;  // Was packet_discard_rate.
+  optional uint32 expand_rate = 6;
+  optional uint32 speech_expand_rate = 7;
+  optional uint32 preemptive_rate = 8;
+  optional uint32 accelerate_rate = 9;
+  optional uint32 secondary_decoded_rate = 10;
+  optional uint32 secondary_discarded_rate = 17;
+  optional int32 clockdrift_ppm = 11;
+  reserved 12;  // Was added_zero_samples.
+  optional int32 mean_waiting_time_ms = 13;
+  optional int32 median_waiting_time_ms = 14;
+  optional int32 min_waiting_time_ms = 15;
+  optional int32 max_waiting_time_ms = 16;
+}
+
+message RtcpStatistics {
+  optional uint32 fraction_lost = 1;
+  optional uint32 cumulative_lost = 2;
+  optional uint32 extended_max_sequence_number = 3;
+  optional uint32 jitter = 4;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/normal.cc b/third_party/libwebrtc/modules/audio_coding/neteq/normal.cc
new file mode 100644
index 0000000000..461ee7fa4a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/normal.cc
@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/normal.h"
+
+#include <string.h>  // memset, memcpy
+
+#include <algorithm>  // min
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+int Normal::Process(const int16_t* input,
+                    size_t length,
+                    NetEq::Mode last_mode,
+                    AudioMultiVector* output) {
+  if (length == 0) {
+    // Nothing to process.
+    output->Clear();
+    return static_cast<int>(length);
+  }
+
+  RTC_DCHECK(output->Empty());
+  // Output should be empty at this point.
+  if (length % output->Channels() != 0) {
+    // The length does not match the number of channels.
+    output->Clear();
+    return 0;
+  }
+  output->PushBackInterleaved(rtc::ArrayView<const int16_t>(input, length));
+
+  const int fs_mult = fs_hz_ / 8000;
+  RTC_DCHECK_GT(fs_mult, 0);
+  // fs_shift = log2(fs_mult), rounded down.
+  // Note that `fs_shift` is not "exact" for 48 kHz.
+  // TODO(hlundin): Investigate this further.
+  const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
+
+  // If last call resulted in a CodedPlc we don't need to do cross-fading but we
+  // need to report the end of the interruption once we are back to normal
+  // operation.
+  if (last_mode == NetEq::Mode::kCodecPlc) {
+    statistics_->EndExpandEvent(fs_hz_);
+  }
+
+  // Check if last RecOut call resulted in an Expand. If so, we have to take
+  // care of some cross-fading and unmuting.
+  if (last_mode == NetEq::Mode::kExpand) {
+    // Generate interpolation data using Expand.
+    // First, set Expand parameters to appropriate values.
+    expand_->SetParametersForNormalAfterExpand();
+
+    // Call Expand.
+    AudioMultiVector expanded(output->Channels());
+    expand_->Process(&expanded);
+    expand_->Reset();
+
+    size_t length_per_channel = length / output->Channels();
+    std::unique_ptr<int16_t[]> signal(new int16_t[length_per_channel]);
+    for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
+      // Set muting factor to the same as expand muting factor.
+      int16_t mute_factor = expand_->MuteFactor(channel_ix);
+
+      (*output)[channel_ix].CopyTo(length_per_channel, 0, signal.get());
+
+      // Find largest absolute value in new data.
+      int16_t decoded_max =
+          WebRtcSpl_MaxAbsValueW16(signal.get(), length_per_channel);
+      // Adjust muting factor if needed (to BGN level).
+      size_t energy_length =
+          std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
+      int scaling = 6 + fs_shift - WebRtcSpl_NormW32(decoded_max * decoded_max);
+      scaling = std::max(scaling, 0);  // `scaling` should always be >= 0.
+      int32_t energy = WebRtcSpl_DotProductWithScale(signal.get(), signal.get(),
+                                                     energy_length, scaling);
+      int32_t scaled_energy_length =
+          static_cast<int32_t>(energy_length >> scaling);
+      if (scaled_energy_length > 0) {
+        energy = energy / scaled_energy_length;
+      } else {
+        energy = 0;
+      }
+
+      int local_mute_factor = 16384;  // 1.0 in Q14.
+      if ((energy != 0) && (energy > background_noise_.Energy(channel_ix))) {
+        // Normalize new frame energy to 15 bits.
+        scaling = WebRtcSpl_NormW32(energy) - 16;
+        // We want background_noise_.energy() / energy in Q14.
+        int32_t bgn_energy = WEBRTC_SPL_SHIFT_W32(
+            background_noise_.Energy(channel_ix), scaling + 14);
+        int16_t energy_scaled =
+            static_cast<int16_t>(WEBRTC_SPL_SHIFT_W32(energy, scaling));
+        int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
+        local_mute_factor =
+            std::min(local_mute_factor, WebRtcSpl_SqrtFloor(ratio << 14));
+      }
+      mute_factor = std::max<int16_t>(mute_factor, local_mute_factor);
+      RTC_DCHECK_LE(mute_factor, 16384);
+      RTC_DCHECK_GE(mute_factor, 0);
+
+      // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14),
+      // or as fast as it takes to come back to full gain within the frame
+      // length.
+      const int back_to_fullscale_inc =
+          static_cast<int>((16384 - mute_factor) / length_per_channel);
+      const int increment = std::max(64 / fs_mult, back_to_fullscale_inc);
+      for (size_t i = 0; i < length_per_channel; i++) {
+        // Scale with mute factor.
+        RTC_DCHECK_LT(channel_ix, output->Channels());
+        RTC_DCHECK_LT(i, output->Size());
+        int32_t scaled_signal = (*output)[channel_ix][i] * mute_factor;
+        // Shift 14 with proper rounding.
+        (*output)[channel_ix][i] =
+            static_cast<int16_t>((scaled_signal + 8192) >> 14);
+        // Increase mute_factor towards 16384.
+        mute_factor =
+            static_cast<int16_t>(std::min(mute_factor + increment, 16384));
+      }
+
+      // Interpolate the expanded data into the new vector.
+      // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
+      size_t win_length = samples_per_ms_;
+      int16_t win_slope_Q14 = default_win_slope_Q14_;
+      RTC_DCHECK_LT(channel_ix, output->Channels());
+      if (win_length > output->Size()) {
+        win_length = output->Size();
+        win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
+      }
+      int16_t win_up_Q14 = 0;
+      for (size_t i = 0; i < win_length; i++) {
+        win_up_Q14 += win_slope_Q14;
+        (*output)[channel_ix][i] =
+            (win_up_Q14 * (*output)[channel_ix][i] +
+             ((1 << 14) - win_up_Q14) * expanded[channel_ix][i] + (1 << 13)) >>
+            14;
+      }
+      RTC_DCHECK_GT(win_up_Q14,
+                    (1 << 14) - 32);  // Worst case rouding is a length of 34
+    }
+  } else if (last_mode == NetEq::Mode::kRfc3389Cng) {
+    RTC_DCHECK_EQ(output->Channels(), 1);  // Not adapted for multi-channel yet.
+    static const size_t kCngLength = 48;
+    RTC_DCHECK_LE(8 * fs_mult, kCngLength);
+    int16_t cng_output[kCngLength];
+    ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+
+    if (cng_decoder) {
+      // Generate long enough for 48kHz.
+      if (!cng_decoder->Generate(cng_output, false)) {
+        // Error returned; set return vector to all zeros.
+        memset(cng_output, 0, sizeof(cng_output));
+      }
+    } else {
+      // If no CNG instance is defined, just copy from the decoded data.
+      // (This will result in interpolating the decoded with itself.)
+      (*output)[0].CopyTo(fs_mult * 8, 0, cng_output);
+    }
+    // Interpolate the CNG into the new vector.
+    // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
+    size_t win_length = samples_per_ms_;
+    int16_t win_slope_Q14 = default_win_slope_Q14_;
+    if (win_length > kCngLength) {
+      win_length = kCngLength;
+      win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
+    }
+    int16_t win_up_Q14 = 0;
+    for (size_t i = 0; i < win_length; i++) {
+      win_up_Q14 += win_slope_Q14;
+      (*output)[0][i] =
+          (win_up_Q14 * (*output)[0][i] +
+           ((1 << 14) - win_up_Q14) * cng_output[i] + (1 << 13)) >>
+          14;
+    }
+    RTC_DCHECK_GT(win_up_Q14,
+                  (1 << 14) - 32);  // Worst case rouding is a length of 34
+  }
+
+  return static_cast<int>(length);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/normal.h b/third_party/libwebrtc/modules/audio_coding/neteq/normal.h
new file mode 100644
index 0000000000..772293b605
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/normal.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
+#define MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
+
+#include <stdint.h>
+#include <string.h>  // Access to size_t.
+
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class AudioMultiVector;
+class BackgroundNoise;
+class DecoderDatabase;
+class Expand;
+
+// This class provides the "Normal" DSP operation, that is performed when
+// there is no data loss, no need to stretch the timing of the signal, and
+// no other "special circumstances" are at hand.
+class Normal {
+ public:
+  Normal(int fs_hz,
+         DecoderDatabase* decoder_database,
+         const BackgroundNoise& background_noise,
+         Expand* expand,
+         StatisticsCalculator* statistics)
+      : fs_hz_(fs_hz),
+        decoder_database_(decoder_database),
+        background_noise_(background_noise),
+        expand_(expand),
+        samples_per_ms_(rtc::CheckedDivExact(fs_hz_, 1000)),
+        default_win_slope_Q14_(
+            rtc::dchecked_cast<uint16_t>((1 << 14) / samples_per_ms_)),
+        statistics_(statistics) {}
+
+  virtual ~Normal() {}
+
+  Normal(const Normal&) = delete;
+  Normal& operator=(const Normal&) = delete;
+
+  // Performs the "Normal" operation. The decoder data is supplied in `input`,
+  // having `length` samples in total for all channels (interleaved). The
+  // result is written to `output`. The number of channels allocated in
+  // `output` defines the number of channels that will be used when
+  // de-interleaving `input`. `last_mode` contains the mode used in the previous
+  // GetAudio call (i.e., not the current one).
+  int Process(const int16_t* input,
+              size_t length,
+              NetEq::Mode last_mode,
+              AudioMultiVector* output);
+
+ private:
+  int fs_hz_;
+  DecoderDatabase* decoder_database_;
+  const BackgroundNoise& background_noise_;
+  Expand* expand_;
+  const size_t samples_per_ms_;
+  const int16_t default_win_slope_Q14_;
+  StatisticsCalculator* const statistics_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/normal_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/normal_unittest.cc
new file mode 100644
index 0000000000..4554d79576
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/normal_unittest.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for Normal class.
+
+#include "modules/audio_coding/neteq/normal.h"
+
+#include <memory>
+#include <vector>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/mock/mock_expand.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+
+namespace webrtc {
+
+namespace {
+
+int ExpandProcess120ms(AudioMultiVector* output) {
+  AudioMultiVector dummy_audio(1, 11520u);
+  dummy_audio.CopyTo(output);
+  return 0;
+}
+
+}  // namespace
+
+TEST(Normal, CreateAndDestroy) {
+  MockDecoderDatabase db;
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
+  Normal normal(fs, &db, bgn, &expand, &statistics);
+  EXPECT_CALL(db, Die());  // Called when `db` goes out of scope.
+}
+
+TEST(Normal, AvoidDivideByZero) {
+  MockDecoderDatabase db;
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
+                    channels);
+  Normal normal(fs, &db, bgn, &expand, &statistics);
+
+  int16_t input[1000] = {0};
+  AudioMultiVector output(channels);
+
+  // Zero input length.
+  EXPECT_EQ(0, normal.Process(input, 0, NetEq::Mode::kExpand, &output));
+  EXPECT_EQ(0u, output.Size());
+
+  // Try to make energy_length >> scaling = 0;
+  EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
+  EXPECT_CALL(expand, Process(_));
+  EXPECT_CALL(expand, Reset());
+  // If input_size_samples < 64, then energy_length in Normal::Process() will
+  // be equal to input_size_samples. Since the input is all zeros, decoded_max
+  // will be zero, and scaling will be >= 6. Thus, energy_length >> scaling = 0,
+  // and using this as a denominator would lead to problems.
+  int input_size_samples = 63;
+  EXPECT_EQ(input_size_samples, normal.Process(input, input_size_samples,
+                                               NetEq::Mode::kExpand, &output));
+
+  EXPECT_CALL(db, Die());      // Called when `db` goes out of scope.
+  EXPECT_CALL(expand, Die());  // Called when `expand` goes out of scope.
+}
+
+TEST(Normal, InputLengthAndChannelsDoNotMatch) {
+  MockDecoderDatabase db;
+  int fs = 8000;
+  size_t channels = 2;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(channels, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
+                    channels);
+  Normal normal(fs, &db, bgn, &expand, &statistics);
+
+  int16_t input[1000] = {0};
+  AudioMultiVector output(channels);
+
+  // Let the number of samples be one sample less than 80 samples per channel.
+  size_t input_len = 80 * channels - 1;
+  EXPECT_EQ(0, normal.Process(input, input_len, NetEq::Mode::kExpand, &output));
+  EXPECT_EQ(0u, output.Size());
+
+  EXPECT_CALL(db, Die());      // Called when `db` goes out of scope.
+  EXPECT_CALL(expand, Die());  // Called when `expand` goes out of scope.
+}
+
+TEST(Normal, LastModeExpand120msPacket) {
+  MockDecoderDatabase db;
+  const int kFs = 48000;
+  const size_t kPacketsizeBytes = 11520u;
+  const size_t kChannels = 1;
+  BackgroundNoise bgn(kChannels);
+  SyncBuffer sync_buffer(kChannels, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, kFs,
+                    kChannels);
+  Normal normal(kFs, &db, bgn, &expand, &statistics);
+
+  int16_t input[kPacketsizeBytes] = {0};
+  AudioMultiVector output(kChannels);
+
+  EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
+  EXPECT_CALL(expand, Process(_)).WillOnce(Invoke(ExpandProcess120ms));
+  EXPECT_CALL(expand, Reset());
+  EXPECT_EQ(
+      static_cast<int>(kPacketsizeBytes),
+      normal.Process(input, kPacketsizeBytes, NetEq::Mode::kExpand, &output));
+
+  EXPECT_EQ(kPacketsizeBytes, output.Size());
+
+  EXPECT_CALL(db, Die());      // Called when `db` goes out of scope.
+  EXPECT_CALL(expand, Die());  // Called when `expand` goes out of scope.
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet.cc
new file mode 100644
index 0000000000..333f161229
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet.cc
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+Packet::Packet() = default;
+Packet::Packet(Packet&& b) = default;
+
+Packet::~Packet() = default;
+
+Packet& Packet::operator=(Packet&& b) = default;
+
+Packet Packet::Clone() const {
+  RTC_CHECK(!frame);
+
+  Packet clone;
+  clone.timestamp = timestamp;
+  clone.sequence_number = sequence_number;
+  clone.payload_type = payload_type;
+  clone.payload.SetData(payload.data(), payload.size());
+  clone.priority = priority;
+  clone.packet_info = packet_info;
+
+  return clone;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet.h
new file mode 100644
index 0000000000..0c6f204edb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet.h
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_H_
+#define MODULES_AUDIO_CODING_NETEQ_PACKET_H_
+
+#include <stdint.h>
+
+#include <list>
+#include <memory>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/tick_timer.h"
+#include "api/rtp_packet_info.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Struct for holding RTP packets.
+struct Packet {
+  struct Priority {
+    Priority() : codec_level(0), red_level(0) {}
+    Priority(int codec_level, int red_level)
+        : codec_level(codec_level), red_level(red_level) {
+      CheckInvariant();
+    }
+
+    int codec_level;
+    int red_level;
+
+    // Priorities are sorted low-to-high, first on the level the codec
+    // prioritizes it, then on the level of RED packet it is; i.e. if it is a
+    // primary or secondary payload of a RED packet. For example: with Opus, an
+    // Fec packet (which the decoder prioritizes lower than a regular packet)
+    // will not be used if there is _any_ RED payload for the same
+    // timeframe. The highest priority packet will have levels {0, 0}. Negative
+    // priorities are not allowed.
+    bool operator<(const Priority& b) const {
+      CheckInvariant();
+      b.CheckInvariant();
+      if (codec_level == b.codec_level)
+        return red_level < b.red_level;
+
+      return codec_level < b.codec_level;
+    }
+    bool operator==(const Priority& b) const {
+      CheckInvariant();
+      b.CheckInvariant();
+      return codec_level == b.codec_level && red_level == b.red_level;
+    }
+    bool operator!=(const Priority& b) const { return !(*this == b); }
+    bool operator>(const Priority& b) const { return b < *this; }
+    bool operator<=(const Priority& b) const { return !(b > *this); }
+    bool operator>=(const Priority& b) const { return !(b < *this); }
+
+   private:
+    void CheckInvariant() const {
+      RTC_DCHECK_GE(codec_level, 0);
+      RTC_DCHECK_GE(red_level, 0);
+    }
+  };
+
+  uint32_t timestamp;
+  uint16_t sequence_number;
+  uint8_t payload_type;
+  // Datagram excluding RTP header and header extension.
+  rtc::Buffer payload;
+  Priority priority;
+  RtpPacketInfo packet_info;
+  std::unique_ptr<TickTimer::Stopwatch> waiting_time;
+  std::unique_ptr<AudioDecoder::EncodedAudioFrame> frame;
+
+  Packet();
+  Packet(Packet&& b);
+  ~Packet();
+
+  // Packets should generally be moved around but sometimes it's useful to make
+  // a copy, for example for testing purposes. NOTE: Will only work for
+  // un-parsed packets, i.e. `frame` must be unset. The payload will, however,
+  // be copied. `waiting_time` will also not be copied.
+  Packet Clone() const;
+
+  Packet& operator=(Packet&& b);
+
+  // Comparison operators. Establish a packet ordering based on (1) timestamp,
+  // (2) sequence number and (3) redundancy.
+  // Timestamp and sequence numbers are compared taking wrap-around into
+  // account. For two packets with the same sequence number and timestamp a
+  // primary payload is considered "smaller" than a secondary.
+  bool operator==(const Packet& rhs) const {
+    return (this->timestamp == rhs.timestamp &&
+            this->sequence_number == rhs.sequence_number &&
+            this->priority == rhs.priority);
+  }
+  bool operator!=(const Packet& rhs) const { return !operator==(rhs); }
+  bool operator<(const Packet& rhs) const {
+    if (this->timestamp == rhs.timestamp) {
+      if (this->sequence_number == rhs.sequence_number) {
+        // Timestamp and sequence numbers are identical - deem the left hand
+        // side to be "smaller" (i.e., "earlier") if it has higher priority.
+        return this->priority < rhs.priority;
+      }
+      return (static_cast<uint16_t>(rhs.sequence_number -
+                                    this->sequence_number) < 0xFFFF / 2);
+    }
+    return (static_cast<uint32_t>(rhs.timestamp - this->timestamp) <
+            0xFFFFFFFF / 2);
+  }
+  bool operator>(const Packet& rhs) const { return rhs.operator<(*this); }
+  bool operator<=(const Packet& rhs) const { return !operator>(rhs); }
+  bool operator>=(const Packet& rhs) const { return !operator<(rhs); }
+
+  bool empty() const { return !frame && payload.empty(); }
+};
+
+// A list of packets.
+typedef std::list<Packet> PacketList;
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_PACKET_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc
new file mode 100644
index 0000000000..2077383f76
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/packet_arrival_history.h"
+
+#include <algorithm>
+
+#include "api/neteq/tick_timer.h"
+
+namespace webrtc {
+
+PacketArrivalHistory::PacketArrivalHistory(int window_size_ms)
+    : window_size_ms_(window_size_ms) {}
+
+void PacketArrivalHistory::Insert(uint32_t rtp_timestamp,
+                                  int64_t arrival_time_ms) {
+  RTC_DCHECK(sample_rate_khz_ > 0);
+  int64_t unwrapped_rtp_timestamp = timestamp_unwrapper_.Unwrap(rtp_timestamp);
+  if (!newest_rtp_timestamp_ ||
+      unwrapped_rtp_timestamp > *newest_rtp_timestamp_) {
+    newest_rtp_timestamp_ = unwrapped_rtp_timestamp;
+  }
+  history_.emplace_back(unwrapped_rtp_timestamp / sample_rate_khz_,
+                        arrival_time_ms);
+  MaybeUpdateCachedArrivals(history_.back());
+  while (history_.front().rtp_timestamp_ms + window_size_ms_ <
+         unwrapped_rtp_timestamp / sample_rate_khz_) {
+    if (&history_.front() == min_packet_arrival_) {
+      min_packet_arrival_ = nullptr;
+    }
+    if (&history_.front() == max_packet_arrival_) {
+      max_packet_arrival_ = nullptr;
+    }
+    history_.pop_front();
+  }
+  if (!min_packet_arrival_ || !max_packet_arrival_) {
+    for (const PacketArrival& packet : history_) {
+      MaybeUpdateCachedArrivals(packet);
+    }
+  }
+}
+
+void PacketArrivalHistory::MaybeUpdateCachedArrivals(
+    const PacketArrival& packet_arrival) {
+  if (!min_packet_arrival_ || packet_arrival <= *min_packet_arrival_) {
+    min_packet_arrival_ = &packet_arrival;
+  }
+  if (!max_packet_arrival_ || packet_arrival >= *max_packet_arrival_) {
+    max_packet_arrival_ = &packet_arrival;
+  }
+}
+
+void PacketArrivalHistory::Reset() {
+  history_.clear();
+  min_packet_arrival_ = nullptr;
+  max_packet_arrival_ = nullptr;
+  timestamp_unwrapper_.Reset();
+  newest_rtp_timestamp_ = absl::nullopt;
+}
+
+int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp,
+                                     int64_t time_ms) const {
+  RTC_DCHECK(sample_rate_khz_ > 0);
+  int64_t unwrapped_rtp_timestamp_ms =
+      timestamp_unwrapper_.PeekUnwrap(rtp_timestamp) / sample_rate_khz_;
+  PacketArrival packet(unwrapped_rtp_timestamp_ms, time_ms);
+  return GetPacketArrivalDelayMs(packet);
+}
+
+int PacketArrivalHistory::GetMaxDelayMs() const {
+  if (!max_packet_arrival_) {
+    return 0;
+  }
+  return GetPacketArrivalDelayMs(*max_packet_arrival_);
+}
+
+bool PacketArrivalHistory::IsNewestRtpTimestamp(uint32_t rtp_timestamp) const {
+  if (!newest_rtp_timestamp_) {
+    return false;
+  }
+  int64_t unwrapped_rtp_timestamp =
+      timestamp_unwrapper_.PeekUnwrap(rtp_timestamp);
+  return unwrapped_rtp_timestamp == *newest_rtp_timestamp_;
+}
+
+int PacketArrivalHistory::GetPacketArrivalDelayMs(
+    const PacketArrival& packet_arrival) const {
+  if (!min_packet_arrival_) {
+    return 0;
+  }
+  return std::max(static_cast<int>(packet_arrival.arrival_time_ms -
+                                   min_packet_arrival_->arrival_time_ms -
+                                   (packet_arrival.rtp_timestamp_ms -
+                                    min_packet_arrival_->rtp_timestamp_ms)),
+                  0);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h
new file mode 100644
index 0000000000..cad362b469
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
+#define MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
+
+#include <cstdint>
+#include <deque>
+
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+// Stores timing information about previously received packets.
+// The history has a fixed window size beyond which old data is automatically
+// pruned.
+class PacketArrivalHistory {
+ public:
+  explicit PacketArrivalHistory(int window_size_ms);
+
+  // Insert packet with `rtp_timestamp` and `arrival_time_ms` into the history.
+  void Insert(uint32_t rtp_timestamp, int64_t arrival_time_ms);
+
+  // The delay for `rtp_timestamp` at `time_ms` is calculated as
+  // `(time_ms - p.arrival_time_ms) - (rtp_timestamp - p.rtp_timestamp)`
+  // where `p` is chosen as the packet arrival in the history that maximizes the
+  // delay.
+  int GetDelayMs(uint32_t rtp_timestamp, int64_t time_ms) const;
+
+  // Get the maximum packet arrival delay observed in the history.
+  int GetMaxDelayMs() const;
+
+  bool IsNewestRtpTimestamp(uint32_t rtp_timestamp) const;
+
+  void Reset();
+
+  void set_sample_rate(int sample_rate) {
+    sample_rate_khz_ = sample_rate / 1000;
+  }
+
+  size_t size() const { return history_.size(); }
+
+ private:
+  struct PacketArrival {
+    PacketArrival(int64_t rtp_timestamp_ms, int64_t arrival_time_ms)
+        : rtp_timestamp_ms(rtp_timestamp_ms),
+          arrival_time_ms(arrival_time_ms) {}
+    int64_t rtp_timestamp_ms;
+    int64_t arrival_time_ms;
+    bool operator<=(const PacketArrival& other) const {
+      return arrival_time_ms - rtp_timestamp_ms <=
+             other.arrival_time_ms - other.rtp_timestamp_ms;
+    }
+    bool operator>=(const PacketArrival& other) const {
+      return arrival_time_ms - rtp_timestamp_ms >=
+             other.arrival_time_ms - other.rtp_timestamp_ms;
+    }
+  };
+  std::deque<PacketArrival> history_;
+  int GetPacketArrivalDelayMs(const PacketArrival& packet_arrival) const;
+  // Updates `min_packet_arrival_` and `max_packet_arrival_`.
+  void MaybeUpdateCachedArrivals(const PacketArrival& packet);
+  const PacketArrival* min_packet_arrival_ = nullptr;
+  const PacketArrival* max_packet_arrival_ = nullptr;
+  const int window_size_ms_;
+  RtpTimestampUnwrapper timestamp_unwrapper_;
+  absl::optional<int64_t> newest_rtp_timestamp_;
+  int sample_rate_khz_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
new file mode 100644
index 0000000000..539a318fe1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/packet_arrival_history.h"
+
+#include <cstdint>
+#include <limits>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kFs = 8000;
+constexpr int kFsKhz = kFs / 1000;
+constexpr int kFrameSizeMs = 20;
+constexpr int kWindowSizeMs = 1000;
+
+class PacketArrivalHistoryTest : public testing::Test {
+ public:
+  PacketArrivalHistoryTest() : history_(kWindowSizeMs) {
+    history_.set_sample_rate(kFs);
+  }
+  void IncrementTime(int delta_ms) { time_ms_ += delta_ms; }
+  int InsertPacketAndGetDelay(int timestamp_delta_ms) {
+    uint32_t timestamp = timestamp_ + timestamp_delta_ms * kFsKhz;
+    if (timestamp_delta_ms > 0) {
+      timestamp_ = timestamp;
+    }
+    history_.Insert(timestamp, time_ms_);
+    EXPECT_EQ(history_.IsNewestRtpTimestamp(timestamp),
+              timestamp_delta_ms >= 0);
+    return history_.GetDelayMs(timestamp, time_ms_);
+  }
+
+ protected:
+  int64_t time_ms_ = 0;
+  PacketArrivalHistory history_;
+  uint32_t timestamp_ = 0x12345678;
+};
+
+TEST_F(PacketArrivalHistoryTest, RelativeArrivalDelay) {
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  IncrementTime(kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 20);
+
+  // Reordered packet.
+  EXPECT_EQ(InsertPacketAndGetDelay(-2 * kFrameSizeMs), 60);
+
+  IncrementTime(2 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 40);
+
+  // Move reference packet forward.
+  EXPECT_EQ(InsertPacketAndGetDelay(4 * kFrameSizeMs), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 20);
+
+  // Earlier packet is now more delayed due to the new reference packet.
+  EXPECT_EQ(history_.GetMaxDelayMs(), 100);
+}
+
+TEST_F(PacketArrivalHistoryTest, ReorderedPackets) {
+  // Insert first packet.
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  // Insert reordered packet.
+  EXPECT_EQ(InsertPacketAndGetDelay(-80), 80);
+
+  // Insert another reordered packet.
+  EXPECT_EQ(InsertPacketAndGetDelay(-kFrameSizeMs), 20);
+
+  // Insert the next packet in order and verify that the relative delay is
+  // estimated based on the first inserted packet.
+  IncrementTime(4 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 60);
+
+  EXPECT_EQ(history_.GetMaxDelayMs(), 80);
+}
+
+TEST_F(PacketArrivalHistoryTest, MaxHistorySize) {
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 20);
+  EXPECT_EQ(history_.GetMaxDelayMs(), 20);
+
+  // Insert next packet with a timestamp difference larger than maximum history
+  // size. This removes the previously inserted packet from the history.
+  IncrementTime(kWindowSizeMs + kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs + kWindowSizeMs), 0);
+  EXPECT_EQ(history_.GetMaxDelayMs(), 0);
+}
+
+TEST_F(PacketArrivalHistoryTest, TimestampWraparound) {
+  timestamp_ = std::numeric_limits<uint32_t>::max();
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  // Insert timestamp that will wrap around.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), kFrameSizeMs);
+
+  // Insert reordered packet before the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(-2 * kFrameSizeMs), 3 * kFrameSizeMs);
+
+  // Insert another in-order packet after the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
+
+  EXPECT_EQ(history_.GetMaxDelayMs(), 3 * kFrameSizeMs);
+}
+
+TEST_F(PacketArrivalHistoryTest, TimestampWraparoundBackwards) {
+  timestamp_ = 0;
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  // Insert timestamp that will wrap around.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), kFrameSizeMs);
+
+  // Insert reordered packet before the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(-2 * kFrameSizeMs), 3 * kFrameSizeMs);
+
+  // Insert another in-order packet after the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
+
+  EXPECT_EQ(history_.GetMaxDelayMs(), 3 * kFrameSizeMs);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc
new file mode 100644
index 0000000000..f6b5a476c9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc
@@ -0,0 +1,405 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This is the implementation of the PacketBuffer class. It is mostly based on
+// an STL list. The list is kept sorted at all times so that the next packet to
+// decode is at the beginning of the list.
+
+#include "modules/audio_coding/neteq/packet_buffer.h"
+
+#include <algorithm>
+#include <list>
+#include <memory>
+#include <type_traits>
+#include <utility>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+// Predicate used when inserting packets in the buffer list.
+// Operator() returns true when `packet` goes before `new_packet`.
+class NewTimestampIsLarger {
+ public:
+  explicit NewTimestampIsLarger(const Packet& new_packet)
+      : new_packet_(new_packet) {}
+  bool operator()(const Packet& packet) { return (new_packet_ >= packet); }
+
+ private:
+  const Packet& new_packet_;
+};
+
+// Returns true if both payload types are known to the decoder database, and
+// have the same sample rate.
+bool EqualSampleRates(uint8_t pt1,
+                      uint8_t pt2,
+                      const DecoderDatabase& decoder_database) {
+  auto* di1 = decoder_database.GetDecoderInfo(pt1);
+  auto* di2 = decoder_database.GetDecoderInfo(pt2);
+  return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz();
+}
+
+void LogPacketDiscarded(int codec_level, StatisticsCalculator* stats) {
+  RTC_CHECK(stats);
+  if (codec_level > 0) {
+    stats->SecondaryPacketsDiscarded(1);
+  } else {
+    stats->PacketsDiscarded(1);
+  }
+}
+
+absl::optional<SmartFlushingConfig> GetSmartflushingConfig() {
+  absl::optional<SmartFlushingConfig> result;
+  std::string field_trial_string =
+      field_trial::FindFullName("WebRTC-Audio-NetEqSmartFlushing");
+  result = SmartFlushingConfig();
+  bool enabled = false;
+  auto parser = StructParametersParser::Create(
+      "enabled", &enabled, "target_level_threshold_ms",
+      &result->target_level_threshold_ms, "target_level_multiplier",
+      &result->target_level_multiplier);
+  parser->Parse(field_trial_string);
+  if (!enabled) {
+    return absl::nullopt;
+  }
+  RTC_LOG(LS_INFO) << "Using smart flushing, target_level_threshold_ms: "
+                   << result->target_level_threshold_ms
+                   << ", target_level_multiplier: "
+                   << result->target_level_multiplier;
+  return result;
+}
+
+}  // namespace
+
+PacketBuffer::PacketBuffer(size_t max_number_of_packets,
+                           const TickTimer* tick_timer)
+    : smart_flushing_config_(GetSmartflushingConfig()),
+      max_number_of_packets_(max_number_of_packets),
+      tick_timer_(tick_timer) {}
+
+// Destructor. All packets in the buffer will be destroyed.
+PacketBuffer::~PacketBuffer() {
+  buffer_.clear();
+}
+
+// Flush the buffer. All packets in the buffer will be destroyed.
+void PacketBuffer::Flush(StatisticsCalculator* stats) {
+  for (auto& p : buffer_) {
+    LogPacketDiscarded(p.priority.codec_level, stats);
+  }
+  buffer_.clear();
+  stats->FlushedPacketBuffer();
+}
+
+void PacketBuffer::PartialFlush(int target_level_ms,
+                                size_t sample_rate,
+                                size_t last_decoded_length,
+                                StatisticsCalculator* stats) {
+  // Make sure that at least half the packet buffer capacity will be available
+  // after the flush. This is done to avoid getting stuck if the target level is
+  // very high.
+  int target_level_samples =
+      std::min(target_level_ms * sample_rate / 1000,
+               max_number_of_packets_ * last_decoded_length / 2);
+  // We should avoid flushing to very low levels.
+  target_level_samples = std::max(
+      target_level_samples, smart_flushing_config_->target_level_threshold_ms);
+  while (GetSpanSamples(last_decoded_length, sample_rate, true) >
+             static_cast<size_t>(target_level_samples) ||
+         buffer_.size() > max_number_of_packets_ / 2) {
+    LogPacketDiscarded(PeekNextPacket()->priority.codec_level, stats);
+    buffer_.pop_front();
+  }
+}
+
+bool PacketBuffer::Empty() const {
+  return buffer_.empty();
+}
+
+int PacketBuffer::InsertPacket(Packet&& packet,
+                               StatisticsCalculator* stats,
+                               size_t last_decoded_length,
+                               size_t sample_rate,
+                               int target_level_ms,
+                               const DecoderDatabase& decoder_database) {
+  if (packet.empty()) {
+    RTC_LOG(LS_WARNING) << "InsertPacket invalid packet";
+    return kInvalidPacket;
+  }
+
+  RTC_DCHECK_GE(packet.priority.codec_level, 0);
+  RTC_DCHECK_GE(packet.priority.red_level, 0);
+
+  int return_val = kOK;
+
+  packet.waiting_time = tick_timer_->GetNewStopwatch();
+
+  // Perform a smart flush if the buffer size exceeds a multiple of the target
+  // level.
+  const size_t span_threshold =
+      smart_flushing_config_
+          ? smart_flushing_config_->target_level_multiplier *
+                std::max(smart_flushing_config_->target_level_threshold_ms,
+                         target_level_ms) *
+                sample_rate / 1000
+          : 0;
+  const bool smart_flush =
+      smart_flushing_config_.has_value() &&
+      GetSpanSamples(last_decoded_length, sample_rate, true) >= span_threshold;
+  if (buffer_.size() >= max_number_of_packets_ || smart_flush) {
+    size_t buffer_size_before_flush = buffer_.size();
+    if (smart_flushing_config_.has_value()) {
+      // Flush down to the target level.
+      PartialFlush(target_level_ms, sample_rate, last_decoded_length, stats);
+      return_val = kPartialFlush;
+    } else {
+      // Buffer is full.
+      Flush(stats);
+      return_val = kFlushed;
+    }
+    RTC_LOG(LS_WARNING) << "Packet buffer flushed, "
+                        << (buffer_size_before_flush - buffer_.size())
+                        << " packets discarded.";
+  }
+
+  // Get an iterator pointing to the place in the buffer where the new packet
+  // should be inserted. The list is searched from the back, since the most
+  // likely case is that the new packet should be near the end of the list.
+  PacketList::reverse_iterator rit = std::find_if(
+      buffer_.rbegin(), buffer_.rend(), NewTimestampIsLarger(packet));
+
+  // The new packet is to be inserted to the right of `rit`. If it has the same
+  // timestamp as `rit`, which has a higher priority, do not insert the new
+  // packet to list.
+  if (rit != buffer_.rend() && packet.timestamp == rit->timestamp) {
+    LogPacketDiscarded(packet.priority.codec_level, stats);
+    return return_val;
+  }
+
+  // The new packet is to be inserted to the left of `it`. If it has the same
+  // timestamp as `it`, which has a lower priority, replace `it` with the new
+  // packet.
+  PacketList::iterator it = rit.base();
+  if (it != buffer_.end() && packet.timestamp == it->timestamp) {
+    LogPacketDiscarded(it->priority.codec_level, stats);
+    it = buffer_.erase(it);
+  }
+  buffer_.insert(it, std::move(packet));  // Insert the packet at that position.
+
+  return return_val;
+}
+
+int PacketBuffer::InsertPacketList(
+    PacketList* packet_list,
+    const DecoderDatabase& decoder_database,
+    absl::optional<uint8_t>* current_rtp_payload_type,
+    absl::optional<uint8_t>* current_cng_rtp_payload_type,
+    StatisticsCalculator* stats,
+    size_t last_decoded_length,
+    size_t sample_rate,
+    int target_level_ms) {
+  RTC_DCHECK(stats);
+  bool flushed = false;
+  for (auto& packet : *packet_list) {
+    if (decoder_database.IsComfortNoise(packet.payload_type)) {
+      if (*current_cng_rtp_payload_type &&
+          **current_cng_rtp_payload_type != packet.payload_type) {
+        // New CNG payload type implies new codec type.
+        *current_rtp_payload_type = absl::nullopt;
+        Flush(stats);
+        flushed = true;
+      }
+      *current_cng_rtp_payload_type = packet.payload_type;
+    } else if (!decoder_database.IsDtmf(packet.payload_type)) {
+      // This must be speech.
+      if ((*current_rtp_payload_type &&
+           **current_rtp_payload_type != packet.payload_type) ||
+          (*current_cng_rtp_payload_type &&
+           !EqualSampleRates(packet.payload_type,
+                             **current_cng_rtp_payload_type,
+                             decoder_database))) {
+        *current_cng_rtp_payload_type = absl::nullopt;
+        Flush(stats);
+        flushed = true;
+      }
+      *current_rtp_payload_type = packet.payload_type;
+    }
+    int return_val =
+        InsertPacket(std::move(packet), stats, last_decoded_length, sample_rate,
+                     target_level_ms, decoder_database);
+    if (return_val == kFlushed) {
+      // The buffer flushed, but this is not an error. We can still continue.
+      flushed = true;
+    } else if (return_val != kOK) {
+      // An error occurred. Delete remaining packets in list and return.
+      packet_list->clear();
+      return return_val;
+    }
+  }
+  packet_list->clear();
+  return flushed ? kFlushed : kOK;
+}
+
+int PacketBuffer::NextTimestamp(uint32_t* next_timestamp) const {
+  if (Empty()) {
+    return kBufferEmpty;
+  }
+  if (!next_timestamp) {
+    return kInvalidPointer;
+  }
+  *next_timestamp = buffer_.front().timestamp;
+  return kOK;
+}
+
+int PacketBuffer::NextHigherTimestamp(uint32_t timestamp,
+                                      uint32_t* next_timestamp) const {
+  if (Empty()) {
+    return kBufferEmpty;
+  }
+  if (!next_timestamp) {
+    return kInvalidPointer;
+  }
+  PacketList::const_iterator it;
+  for (it = buffer_.begin(); it != buffer_.end(); ++it) {
+    if (it->timestamp >= timestamp) {
+      // Found a packet matching the search.
+      *next_timestamp = it->timestamp;
+      return kOK;
+    }
+  }
+  return kNotFound;
+}
+
+const Packet* PacketBuffer::PeekNextPacket() const {
+  return buffer_.empty() ? nullptr : &buffer_.front();
+}
+
+absl::optional<Packet> PacketBuffer::GetNextPacket() {
+  if (Empty()) {
+    // Buffer is empty.
+    return absl::nullopt;
+  }
+
+  absl::optional<Packet> packet(std::move(buffer_.front()));
+  // Assert that the packet sanity checks in InsertPacket method works.
+  RTC_DCHECK(!packet->empty());
+  buffer_.pop_front();
+
+  return packet;
+}
+
+int PacketBuffer::DiscardNextPacket(StatisticsCalculator* stats) {
+  if (Empty()) {
+    return kBufferEmpty;
+  }
+  // Assert that the packet sanity checks in InsertPacket method works.
+  const Packet& packet = buffer_.front();
+  RTC_DCHECK(!packet.empty());
+  LogPacketDiscarded(packet.priority.codec_level, stats);
+  buffer_.pop_front();
+  return kOK;
+}
+
+void PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit,
+                                     uint32_t horizon_samples,
+                                     StatisticsCalculator* stats) {
+  buffer_.remove_if([timestamp_limit, horizon_samples, stats](const Packet& p) {
+    if (timestamp_limit == p.timestamp ||
+        !IsObsoleteTimestamp(p.timestamp, timestamp_limit, horizon_samples)) {
+      return false;
+    }
+    LogPacketDiscarded(p.priority.codec_level, stats);
+    return true;
+  });
+}
+
+void PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit,
+                                        StatisticsCalculator* stats) {
+  DiscardOldPackets(timestamp_limit, 0, stats);
+}
+
+void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type,
+                                                 StatisticsCalculator* stats) {
+  buffer_.remove_if([payload_type, stats](const Packet& p) {
+    if (p.payload_type != payload_type) {
+      return false;
+    }
+    LogPacketDiscarded(p.priority.codec_level, stats);
+    return true;
+  });
+}
+
+size_t PacketBuffer::NumPacketsInBuffer() const {
+  return buffer_.size();
+}
+
+size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
+  size_t num_samples = 0;
+  size_t last_duration = last_decoded_length;
+  for (const Packet& packet : buffer_) {
+    if (packet.frame) {
+      // TODO(hlundin): Verify that it's fine to count all packets and remove
+      // this check.
+      if (packet.priority != Packet::Priority(0, 0)) {
+        continue;
+      }
+      size_t duration = packet.frame->Duration();
+      if (duration > 0) {
+        last_duration = duration;  // Save the most up-to-date (valid) duration.
+      }
+    }
+    num_samples += last_duration;
+  }
+  return num_samples;
+}
+
+size_t PacketBuffer::GetSpanSamples(size_t last_decoded_length,
+                                    size_t sample_rate,
+                                    bool count_dtx_waiting_time) const {
+  if (buffer_.size() == 0) {
+    return 0;
+  }
+
+  size_t span = buffer_.back().timestamp - buffer_.front().timestamp;
+  if (buffer_.back().frame && buffer_.back().frame->Duration() > 0) {
+    size_t duration = buffer_.back().frame->Duration();
+    if (count_dtx_waiting_time && buffer_.back().frame->IsDtxPacket()) {
+      size_t waiting_time_samples = rtc::dchecked_cast<size_t>(
+          buffer_.back().waiting_time->ElapsedMs() * (sample_rate / 1000));
+      duration = std::max(duration, waiting_time_samples);
+    }
+    span += duration;
+  } else {
+    span += last_decoded_length;
+  }
+  return span;
+}
+
+bool PacketBuffer::ContainsDtxOrCngPacket(
+    const DecoderDatabase* decoder_database) const {
+  RTC_DCHECK(decoder_database);
+  for (const Packet& packet : buffer_) {
+    if ((packet.frame && packet.frame->IsDtxPacket()) ||
+        decoder_database->IsComfortNoise(packet.payload_type)) {
+      return true;
+    }
+  }
+  return false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h
new file mode 100644
index 0000000000..c6fb47ffbf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h
@@ -0,0 +1,181 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/include/module_common_types_public.h"  // IsNewerTimestamp
+
+namespace webrtc {
+
+class DecoderDatabase;
+class StatisticsCalculator;
+class TickTimer;
+struct SmartFlushingConfig {
+  // When calculating the flushing threshold, the maximum between the target
+  // level and this value is used.
+  int target_level_threshold_ms = 500;
+  // A smart flush is triggered when the packet buffer contains a multiple of
+  // the target level.
+  int target_level_multiplier = 3;
+};
+
+// This is the actual buffer holding the packets before decoding.
+class PacketBuffer {
+ public:
+  enum BufferReturnCodes {
+    kOK = 0,
+    kFlushed,
+    kPartialFlush,
+    kNotFound,
+    kBufferEmpty,
+    kInvalidPacket,
+    kInvalidPointer
+  };
+
+  // Constructor creates a buffer which can hold a maximum of
+  // `max_number_of_packets` packets.
+  PacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer);
+
+  // Deletes all packets in the buffer before destroying the buffer.
+  virtual ~PacketBuffer();
+
+  PacketBuffer(const PacketBuffer&) = delete;
+  PacketBuffer& operator=(const PacketBuffer&) = delete;
+
+  // Flushes the buffer and deletes all packets in it.
+  virtual void Flush(StatisticsCalculator* stats);
+
+  // Partial flush. Flush packets but leave some packets behind.
+  virtual void PartialFlush(int target_level_ms,
+                            size_t sample_rate,
+                            size_t last_decoded_length,
+                            StatisticsCalculator* stats);
+
+  // Returns true for an empty buffer.
+  virtual bool Empty() const;
+
+  // Inserts `packet` into the buffer. The buffer will take over ownership of
+  // the packet object.
+  // Returns PacketBuffer::kOK on success, PacketBuffer::kFlushed if the buffer
+  // was flushed due to overfilling.
+  virtual int InsertPacket(Packet&& packet,
+                           StatisticsCalculator* stats,
+                           size_t last_decoded_length,
+                           size_t sample_rate,
+                           int target_level_ms,
+                           const DecoderDatabase& decoder_database);
+
+  // Inserts a list of packets into the buffer. The buffer will take over
+  // ownership of the packet objects.
+  // Returns PacketBuffer::kOK if all packets were inserted successfully.
+  // If the buffer was flushed due to overfilling, only a subset of the list is
+  // inserted, and PacketBuffer::kFlushed is returned.
+  // The last three parameters are included for legacy compatibility.
+  // TODO(hlundin): Redesign to not use current_*_payload_type and
+  // decoder_database.
+  virtual int InsertPacketList(
+      PacketList* packet_list,
+      const DecoderDatabase& decoder_database,
+      absl::optional<uint8_t>* current_rtp_payload_type,
+      absl::optional<uint8_t>* current_cng_rtp_payload_type,
+      StatisticsCalculator* stats,
+      size_t last_decoded_length,
+      size_t sample_rate,
+      int target_level_ms);
+
+  // Gets the timestamp for the first packet in the buffer and writes it to the
+  // output variable `next_timestamp`.
+  // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
+  // PacketBuffer::kOK otherwise.
+  virtual int NextTimestamp(uint32_t* next_timestamp) const;
+
+  // Gets the timestamp for the first packet in the buffer with a timestamp no
+  // lower than the input limit `timestamp`. The result is written to the output
+  // variable `next_timestamp`.
+  // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
+  // PacketBuffer::kOK otherwise.
+  virtual int NextHigherTimestamp(uint32_t timestamp,
+                                  uint32_t* next_timestamp) const;
+
+  // Returns a (constant) pointer to the first packet in the buffer. Returns
+  // NULL if the buffer is empty.
+  virtual const Packet* PeekNextPacket() const;
+
+  // Extracts the first packet in the buffer and returns it.
+  // Returns an empty optional if the buffer is empty.
+  virtual absl::optional<Packet> GetNextPacket();
+
+  // Discards the first packet in the buffer. The packet is deleted.
+  // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
+  // PacketBuffer::kOK otherwise.
+  virtual int DiscardNextPacket(StatisticsCalculator* stats);
+
+  // Discards all packets that are (strictly) older than timestamp_limit,
+  // but newer than timestamp_limit - horizon_samples. Setting horizon_samples
+  // to zero implies that the horizon is set to half the timestamp range. That
+  // is, if a packet is more than 2^31 timestamps into the future compared with
+  // timestamp_limit (including wrap-around), it is considered old.
+  virtual void DiscardOldPackets(uint32_t timestamp_limit,
+                                 uint32_t horizon_samples,
+                                 StatisticsCalculator* stats);
+
+  // Discards all packets that are (strictly) older than timestamp_limit.
+  virtual void DiscardAllOldPackets(uint32_t timestamp_limit,
+                                    StatisticsCalculator* stats);
+
+  // Removes all packets with a specific payload type from the buffer.
+  virtual void DiscardPacketsWithPayloadType(uint8_t payload_type,
+                                             StatisticsCalculator* stats);
+
+  // Returns the number of packets in the buffer, including duplicates and
+  // redundant packets.
+  virtual size_t NumPacketsInBuffer() const;
+
+  // Returns the number of samples in the buffer, including samples carried in
+  // duplicate and redundant packets.
+  virtual size_t NumSamplesInBuffer(size_t last_decoded_length) const;
+
+  // Returns the total duration in samples that the packets in the buffer spans
+  // across.
+  virtual size_t GetSpanSamples(size_t last_decoded_length,
+                                size_t sample_rate,
+                                bool count_dtx_waiting_time) const;
+
+  // Returns true if the packet buffer contains any DTX or CNG packets.
+  virtual bool ContainsDtxOrCngPacket(
+      const DecoderDatabase* decoder_database) const;
+
+  // Static method returning true if `timestamp` is older than `timestamp_limit`
+  // but less than `horizon_samples` behind `timestamp_limit`. For instance,
+  // with timestamp_limit = 100 and horizon_samples = 10, a timestamp in the
+  // range (90, 100) is considered obsolete, and will yield true.
+  // Setting `horizon_samples` to 0 is the same as setting it to 2^31, i.e.,
+  // half the 32-bit timestamp range.
+  static bool IsObsoleteTimestamp(uint32_t timestamp,
+                                  uint32_t timestamp_limit,
+                                  uint32_t horizon_samples) {
+    return IsNewerTimestamp(timestamp_limit, timestamp) &&
+           (horizon_samples == 0 ||
+            IsNewerTimestamp(timestamp, timestamp_limit - horizon_samples));
+  }
+
+ private:
+  absl::optional<SmartFlushingConfig> smart_flushing_config_;
+  size_t max_number_of_packets_;
+  PacketList buffer_;
+  const TickTimer* tick_timer_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
new file mode 100644
index 0000000000..1a054daca3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -0,0 +1,989 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for PacketBuffer class.
+
+#include "modules/audio_coding/neteq/packet_buffer.h"
+
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::InSequence;
+using ::testing::MockFunction;
+using ::testing::Return;
+using ::testing::StrictMock;
+
+namespace {
+class MockEncodedAudioFrame : public webrtc::AudioDecoder::EncodedAudioFrame {
+ public:
+  MOCK_METHOD(size_t, Duration, (), (const, override));
+
+  MOCK_METHOD(bool, IsDtxPacket, (), (const, override));
+
+  MOCK_METHOD(absl::optional<DecodeResult>,
+              Decode,
+              (rtc::ArrayView<int16_t> decoded),
+              (const, override));
+};
+
+// Helper class to generate packets. Packets must be deleted by the user.
+class PacketGenerator {
+ public:
+  PacketGenerator(uint16_t seq_no, uint32_t ts, uint8_t pt, int frame_size);
+  virtual ~PacketGenerator() {}
+  void Reset(uint16_t seq_no, uint32_t ts, uint8_t pt, int frame_size);
+  webrtc::Packet NextPacket(
+      int payload_size_bytes,
+      std::unique_ptr<webrtc::AudioDecoder::EncodedAudioFrame> audio_frame);
+
+  uint16_t seq_no_;
+  uint32_t ts_;
+  uint8_t pt_;
+  int frame_size_;
+};
+
+PacketGenerator::PacketGenerator(uint16_t seq_no,
+                                 uint32_t ts,
+                                 uint8_t pt,
+                                 int frame_size) {
+  Reset(seq_no, ts, pt, frame_size);
+}
+
+void PacketGenerator::Reset(uint16_t seq_no,
+                            uint32_t ts,
+                            uint8_t pt,
+                            int frame_size) {
+  seq_no_ = seq_no;
+  ts_ = ts;
+  pt_ = pt;
+  frame_size_ = frame_size;
+}
+
+webrtc::Packet PacketGenerator::NextPacket(
+    int payload_size_bytes,
+    std::unique_ptr<webrtc::AudioDecoder::EncodedAudioFrame> audio_frame) {
+  webrtc::Packet packet;
+  packet.sequence_number = seq_no_;
+  packet.timestamp = ts_;
+  packet.payload_type = pt_;
+  packet.payload.SetSize(payload_size_bytes);
+  ++seq_no_;
+  ts_ += frame_size_;
+  packet.frame = std::move(audio_frame);
+  return packet;
+}
+
+struct PacketsToInsert {
+  uint16_t sequence_number;
+  uint32_t timestamp;
+  uint8_t payload_type;
+  bool primary;
+  // Order of this packet to appear upon extraction, after inserting a series
+  // of packets. A negative number means that it should have been discarded
+  // before extraction.
+  int extract_order;
+};
+
+}  // namespace
+
+namespace webrtc {
+
+// Start of test definitions.
+
+TEST(PacketBuffer, CreateAndDestroy) {
+  TickTimer tick_timer;
+  PacketBuffer* buffer = new PacketBuffer(10, &tick_timer);  // 10 packets.
+  EXPECT_TRUE(buffer->Empty());
+  delete buffer;
+}
+
+TEST(PacketBuffer, InsertPacket) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(17u, 4711u, 0, 10);
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  const int payload_len = 100;
+  const Packet packet = gen.NextPacket(payload_len, nullptr);
+  EXPECT_EQ(0, buffer.InsertPacket(/*packet=*/packet.Clone(),
+                                   /*stats=*/&mock_stats,
+                                   /*last_decoded_length=*/payload_len,
+                                   /*sample_rate=*/10000,
+                                   /*target_level_ms=*/60,
+                                   /*decoder_database=*/decoder_database));
+  uint32_t next_ts;
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
+  EXPECT_EQ(4711u, next_ts);
+  EXPECT_FALSE(buffer.Empty());
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
+  const Packet* next_packet = buffer.PeekNextPacket();
+  EXPECT_EQ(packet, *next_packet);  // Compare contents.
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+
+  // Do not explicitly flush buffer or delete packet to test that it is deleted
+  // with the buffer. (Tested with Valgrind or similar tool.)
+}
+
+// Test to flush buffer.
+TEST(PacketBuffer, FlushBuffer) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  const int payload_len = 10;
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  // Insert 10 small packets; should be ok.
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(
+        PacketBuffer::kOK,
+        buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                            /*stats=*/&mock_stats,
+                            /*last_decoded_length=*/payload_len,
+                            /*sample_rate=*/1000,
+                            /*target_level_ms=*/60,
+                            /*decoder_database=*/decoder_database));
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  EXPECT_FALSE(buffer.Empty());
+
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(10);
+  buffer.Flush(&mock_stats);
+  // Buffer should delete the payloads itself.
+  EXPECT_EQ(0u, buffer.NumPacketsInBuffer());
+  EXPECT_TRUE(buffer.Empty());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test to fill the buffer over the limits, and verify that it flushes.
+TEST(PacketBuffer, OverfillBuffer) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  // Insert 10 small packets; should be ok.
+  const int payload_len = 10;
+  int i;
+  for (i = 0; i < 10; ++i) {
+    EXPECT_EQ(
+        PacketBuffer::kOK,
+        buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                            /*stats=*/&mock_stats,
+                            /*last_decoded_length=*/payload_len,
+                            /*sample_rate=*/1000,
+                            /*target_level_ms=*/60,
+                            /*decoder_database=*/decoder_database));
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  uint32_t next_ts;
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
+  EXPECT_EQ(0u, next_ts);  // Expect first inserted packet to be first in line.
+
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(10);
+  const Packet packet = gen.NextPacket(payload_len, nullptr);
+  // Insert 11th packet; should flush the buffer and insert it after flushing.
+  EXPECT_EQ(PacketBuffer::kFlushed,
+            buffer.InsertPacket(/*packet=*/packet.Clone(),
+                                /*stats=*/&mock_stats,
+                                /*last_decoded_length=*/payload_len,
+                                /*sample_rate=*/1000,
+                                /*target_level_ms=*/60,
+                                /*decoder_database=*/decoder_database));
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
+  // Expect last inserted packet to be first in line.
+  EXPECT_EQ(packet.timestamp, next_ts);
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test a partial buffer flush.
+TEST(PacketBuffer, PartialFlush) {
+  // Use a field trial to configure smart flushing.
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqSmartFlushing/enabled:true,"
+      "target_level_threshold_ms:0,target_level_multiplier:2/");
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  const int payload_len = 10;
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  // Insert 10 small packets; should be ok.
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(
+        PacketBuffer::kOK,
+        buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                            /*stats=*/&mock_stats,
+                            /*last_decoded_length=*/payload_len,
+                            /*sample_rate=*/1000,
+                            /*target_level_ms=*/100,
+                            /*decoder_database=*/decoder_database));
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  EXPECT_FALSE(buffer.Empty());
+
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(7);
+  buffer.PartialFlush(/*target_level_ms=*/30,
+                      /*sample_rate=*/1000,
+                      /*last_decoded_length=*/payload_len,
+                      /*stats=*/&mock_stats);
+  // There should still be some packets left in the buffer.
+  EXPECT_EQ(3u, buffer.NumPacketsInBuffer());
+  EXPECT_FALSE(buffer.Empty());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test to fill the buffer over the limits, and verify that the smart flush
+// functionality works as expected.
+TEST(PacketBuffer, SmartFlushOverfillBuffer) {
+  // Use a field trial to configure smart flushing.
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqSmartFlushing/enabled:true,"
+      "target_level_threshold_ms:0,target_level_multiplier:2/");
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  // Insert 10 small packets; should be ok.
+  const int payload_len = 10;
+  int i;
+  for (i = 0; i < 10; ++i) {
+    EXPECT_EQ(
+        PacketBuffer::kOK,
+        buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                            /*stats=*/&mock_stats,
+                            /*last_decoded_length=*/payload_len,
+                            /*sample_rate=*/1000,
+                            /*target_level_ms=*/100,
+                            /*decoder_database=*/decoder_database));
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  uint32_t next_ts;
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
+  EXPECT_EQ(0u, next_ts);  // Expect first inserted packet to be first in line.
+
+  const Packet packet = gen.NextPacket(payload_len, nullptr);
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(6);
+  // Insert 11th packet; should cause a partial flush and insert the packet
+  // after flushing.
+  EXPECT_EQ(PacketBuffer::kPartialFlush,
+            buffer.InsertPacket(/*packet=*/packet.Clone(),
+                                /*stats=*/&mock_stats,
+                                /*last_decoded_length=*/payload_len,
+                                /*sample_rate=*/1000,
+                                /*target_level_ms=*/40,
+                                /*decoder_database=*/decoder_database));
+  EXPECT_EQ(5u, buffer.NumPacketsInBuffer());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test inserting a list of packets.
+TEST(PacketBuffer, InsertPacketList) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  PacketList list;
+  const int payload_len = 10;
+
+  // Insert 10 small packets.
+  for (int i = 0; i < 10; ++i) {
+    list.push_back(gen.NextPacket(payload_len, nullptr));
+  }
+
+  MockDecoderDatabase decoder_database;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(0))
+      .WillRepeatedly(Return(&info));
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+  EXPECT_EQ(
+      PacketBuffer::kOK,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/payload_len,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(0, current_pt);  // Current payload type changed to 0.
+  EXPECT_EQ(absl::nullopt, current_cng_pt);  // CNG payload type not changed.
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test inserting a list of packets. Last packet is of a different payload type.
+// Expecting the buffer to flush.
+// TODO(hlundin): Remove this test when legacy operation is no longer needed.
+TEST(PacketBuffer, InsertPacketListChangePayloadType) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  PacketList list;
+  const int payload_len = 10;
+
+  // Insert 10 small packets.
+  for (int i = 0; i < 10; ++i) {
+    list.push_back(gen.NextPacket(payload_len, nullptr));
+  }
+  // Insert 11th packet of another payload type (not CNG).
+  {
+    Packet packet = gen.NextPacket(payload_len, nullptr);
+    packet.payload_type = 1;
+    list.push_back(std::move(packet));
+  }
+
+  MockDecoderDatabase decoder_database;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info0(SdpAudioFormat("pcmu", 8000, 1),
+                                           absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(0))
+      .WillRepeatedly(Return(&info0));
+  const DecoderDatabase::DecoderInfo info1(SdpAudioFormat("pcma", 8000, 1),
+                                           absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(1))
+      .WillRepeatedly(Return(&info1));
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(10);
+  EXPECT_EQ(
+      PacketBuffer::kFlushed,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/payload_len,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());  // Only the last packet.
+  EXPECT_EQ(1, current_pt);  // Current payload type changed to 1.
+  EXPECT_EQ(absl::nullopt, current_cng_pt);  // CNG payload type not changed.
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+TEST(PacketBuffer, ExtractOrderRedundancy) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
+  const int kPackets = 18;
+  const int kFrameSize = 10;
+  const int kPayloadLength = 10;
+
+  PacketsToInsert packet_facts[kPackets] = {
+      {0xFFFD, 0xFFFFFFD7, 0, true, 0},   {0xFFFE, 0xFFFFFFE1, 0, true, 1},
+      {0xFFFE, 0xFFFFFFD7, 1, false, -1}, {0xFFFF, 0xFFFFFFEB, 0, true, 2},
+      {0xFFFF, 0xFFFFFFE1, 1, false, -1}, {0x0000, 0xFFFFFFF5, 0, true, 3},
+      {0x0000, 0xFFFFFFEB, 1, false, -1}, {0x0001, 0xFFFFFFFF, 0, true, 4},
+      {0x0001, 0xFFFFFFF5, 1, false, -1}, {0x0002, 0x0000000A, 0, true, 5},
+      {0x0002, 0xFFFFFFFF, 1, false, -1}, {0x0003, 0x0000000A, 1, false, -1},
+      {0x0004, 0x0000001E, 0, true, 7},   {0x0004, 0x00000014, 1, false, 6},
+      {0x0005, 0x0000001E, 0, true, -1},  {0x0005, 0x00000014, 1, false, -1},
+      {0x0006, 0x00000028, 0, true, 8},   {0x0006, 0x0000001E, 1, false, -1},
+  };
+  MockDecoderDatabase decoder_database;
+
+  const size_t kExpectPacketsInBuffer = 9;
+
+  std::vector<Packet> expect_order(kExpectPacketsInBuffer);
+
+  PacketGenerator gen(0, 0, 0, kFrameSize);
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction
+  // check ensures that exactly one call to PacketsDiscarded happens in each
+  // DiscardNextPacket call.
+  InSequence s;
+  MockFunction<void(int check_point_id)> check;
+  for (int i = 0; i < kPackets; ++i) {
+    gen.Reset(packet_facts[i].sequence_number, packet_facts[i].timestamp,
+              packet_facts[i].payload_type, kFrameSize);
+    Packet packet = gen.NextPacket(kPayloadLength, nullptr);
+    packet.priority.codec_level = packet_facts[i].primary ? 0 : 1;
+    if (packet_facts[i].extract_order < 0) {
+      if (packet.priority.codec_level > 0) {
+        EXPECT_CALL(mock_stats, SecondaryPacketsDiscarded(1));
+      } else {
+        EXPECT_CALL(mock_stats, PacketsDiscarded(1));
+      }
+    }
+    EXPECT_CALL(check, Call(i));
+    EXPECT_EQ(PacketBuffer::kOK,
+              buffer.InsertPacket(/*packet=*/packet.Clone(),
+                                  /*stats=*/&mock_stats,
+                                  /*last_decoded_length=*/kPayloadLength,
+                                  /*sample_rate=*/1000,
+                                  /*target_level_ms=*/60,
+                                  /*decoder_database=*/decoder_database));
+    if (packet_facts[i].extract_order >= 0) {
+      expect_order[packet_facts[i].extract_order] = std::move(packet);
+    }
+    check.Call(i);
+  }
+
+  EXPECT_EQ(kExpectPacketsInBuffer, buffer.NumPacketsInBuffer());
+
+  for (size_t i = 0; i < kExpectPacketsInBuffer; ++i) {
+    const absl::optional<Packet> packet = buffer.GetNextPacket();
+    EXPECT_EQ(packet, expect_order[i]);  // Compare contents.
+  }
+  EXPECT_TRUE(buffer.Empty());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+TEST(PacketBuffer, DiscardPackets) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
+  const uint16_t start_seq_no = 17;
+  const uint32_t start_ts = 4711;
+  const uint32_t ts_increment = 10;
+  PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
+  PacketList list;
+  const int payload_len = 10;
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  constexpr int kTotalPackets = 10;
+  // Insert 10 small packets.
+  for (int i = 0; i < kTotalPackets; ++i) {
+    buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                        /*stats=*/&mock_stats,
+                        /*last_decoded_length=*/payload_len,
+                        /*sample_rate=*/1000,
+                        /*target_level_ms=*/60,
+                        /*decoder_database=*/decoder_database);
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+
+  uint32_t current_ts = start_ts;
+
+  // Discard them one by one and make sure that the right packets are at the
+  // front of the buffer.
+  constexpr int kDiscardPackets = 5;
+
+  // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction
+  // check ensures that exactly one call to PacketsDiscarded happens in each
+  // DiscardNextPacket call.
+  InSequence s;
+  MockFunction<void(int check_point_id)> check;
+  for (int i = 0; i < kDiscardPackets; ++i) {
+    uint32_t ts;
+    EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&ts));
+    EXPECT_EQ(current_ts, ts);
+    EXPECT_CALL(mock_stats, PacketsDiscarded(1));
+    EXPECT_CALL(check, Call(i));
+    EXPECT_EQ(PacketBuffer::kOK, buffer.DiscardNextPacket(&mock_stats));
+    current_ts += ts_increment;
+    check.Call(i);
+  }
+
+  constexpr int kRemainingPackets = kTotalPackets - kDiscardPackets;
+  // This will discard all remaining packets but one. The oldest packet is older
+  // than the indicated horizon_samples, and will thus be left in the buffer.
+  constexpr size_t kSkipPackets = 1;
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1))
+      .Times(kRemainingPackets - kSkipPackets);
+  EXPECT_CALL(check, Call(17));  // Arbitrary id number.
+  buffer.DiscardOldPackets(start_ts + kTotalPackets * ts_increment,
+                           kRemainingPackets * ts_increment, &mock_stats);
+  check.Call(17);  // Same arbitrary id number.
+
+  EXPECT_EQ(kSkipPackets, buffer.NumPacketsInBuffer());
+  uint32_t ts;
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&ts));
+  EXPECT_EQ(current_ts, ts);
+
+  // Discard all remaining packets.
+  EXPECT_CALL(mock_stats, PacketsDiscarded(kSkipPackets));
+  buffer.DiscardAllOldPackets(start_ts + kTotalPackets * ts_increment,
+                              &mock_stats);
+
+  EXPECT_TRUE(buffer.Empty());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+TEST(PacketBuffer, Reordering) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
+  const uint16_t start_seq_no = 17;
+  const uint32_t start_ts = 4711;
+  const uint32_t ts_increment = 10;
+  PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
+  const int payload_len = 10;
+
+  // Generate 10 small packets and insert them into a PacketList. Insert every
+  // odd packet to the front, and every even packet to the back, thus creating
+  // a (rather strange) reordering.
+  PacketList list;
+  for (int i = 0; i < 10; ++i) {
+    Packet packet = gen.NextPacket(payload_len, nullptr);
+    if (i % 2) {
+      list.push_front(std::move(packet));
+    } else {
+      list.push_back(std::move(packet));
+    }
+  }
+
+  MockDecoderDatabase decoder_database;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(0))
+      .WillRepeatedly(Return(&info));
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  EXPECT_EQ(
+      PacketBuffer::kOK,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/payload_len,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+
+  // Extract them and make sure that come out in the right order.
+  uint32_t current_ts = start_ts;
+  for (int i = 0; i < 10; ++i) {
+    const absl::optional<Packet> packet = buffer.GetNextPacket();
+    ASSERT_TRUE(packet);
+    EXPECT_EQ(current_ts, packet->timestamp);
+    current_ts += ts_increment;
+  }
+  EXPECT_TRUE(buffer.Empty());
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// The test first inserts a packet with narrow-band CNG, then a packet with
+// wide-band speech. The expected behavior of the packet buffer is to detect a
+// change in sample rate, even though no speech packet has been inserted before,
+// and flush out the CNG packet.
+TEST(PacketBuffer, CngFirstThenSpeechWithNewSampleRate) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  const uint8_t kCngPt = 13;
+  const int kPayloadLen = 10;
+  const uint8_t kSpeechPt = 100;
+
+  MockDecoderDatabase decoder_database;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info_cng(SdpAudioFormat("cn", 8000, 1),
+                                              absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(kCngPt))
+      .WillRepeatedly(Return(&info_cng));
+  const DecoderDatabase::DecoderInfo info_speech(
+      SdpAudioFormat("l16", 16000, 1), absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(kSpeechPt))
+      .WillRepeatedly(Return(&info_speech));
+
+  // Insert first packet, which is narrow-band CNG.
+  PacketGenerator gen(0, 0, kCngPt, 10);
+  PacketList list;
+  list.push_back(gen.NextPacket(kPayloadLen, nullptr));
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  EXPECT_EQ(
+      PacketBuffer::kOK,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/kPayloadLen,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
+  ASSERT_TRUE(buffer.PeekNextPacket());
+  EXPECT_EQ(kCngPt, buffer.PeekNextPacket()->payload_type);
+  EXPECT_EQ(current_pt, absl::nullopt);  // Current payload type not set.
+  EXPECT_EQ(kCngPt, current_cng_pt);     // CNG payload type set.
+
+  // Insert second packet, which is wide-band speech.
+  {
+    Packet packet = gen.NextPacket(kPayloadLen, nullptr);
+    packet.payload_type = kSpeechPt;
+    list.push_back(std::move(packet));
+  }
+  // Expect the buffer to flush out the CNG packet, since it does not match the
+  // new speech sample rate.
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1));
+  EXPECT_EQ(
+      PacketBuffer::kFlushed,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/kPayloadLen,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
+  ASSERT_TRUE(buffer.PeekNextPacket());
+  EXPECT_EQ(kSpeechPt, buffer.PeekNextPacket()->payload_type);
+
+  EXPECT_EQ(kSpeechPt, current_pt);          // Current payload type set.
+  EXPECT_EQ(absl::nullopt, current_cng_pt);  // CNG payload type reset.
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+TEST(PacketBuffer, Failures) {
+  const uint16_t start_seq_no = 17;
+  const uint32_t start_ts = 4711;
+  const uint32_t ts_increment = 10;
+  int payload_len = 100;
+  PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
+  TickTimer tick_timer;
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  PacketBuffer* buffer = new PacketBuffer(100, &tick_timer);  // 100 packets.
+  {
+    Packet packet = gen.NextPacket(payload_len, nullptr);
+    packet.payload.Clear();
+    EXPECT_EQ(PacketBuffer::kInvalidPacket,
+              buffer->InsertPacket(/*packet=*/std::move(packet),
+                                   /*stats=*/&mock_stats,
+                                   /*last_decoded_length=*/payload_len,
+                                   /*sample_rate=*/1000,
+                                   /*target_level_ms=*/60,
+                                   /*decoder_database=*/decoder_database));
+  }
+  // Buffer should still be empty. Test all empty-checks.
+  uint32_t temp_ts;
+  EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer->NextTimestamp(&temp_ts));
+  EXPECT_EQ(PacketBuffer::kBufferEmpty,
+            buffer->NextHigherTimestamp(0, &temp_ts));
+  EXPECT_EQ(NULL, buffer->PeekNextPacket());
+  EXPECT_FALSE(buffer->GetNextPacket());
+
+  // Discarding packets will not invoke mock_stats.PacketDiscarded() because the
+  // packet buffer is empty.
+  EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer->DiscardNextPacket(&mock_stats));
+  buffer->DiscardAllOldPackets(0, &mock_stats);
+
+  // Insert one packet to make the buffer non-empty.
+  EXPECT_EQ(
+      PacketBuffer::kOK,
+      buffer->InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                           /*stats=*/&mock_stats,
+                           /*last_decoded_length=*/payload_len,
+                           /*sample_rate=*/1000,
+                           /*target_level_ms=*/60,
+                           /*decoder_database=*/decoder_database));
+  EXPECT_EQ(PacketBuffer::kInvalidPointer, buffer->NextTimestamp(NULL));
+  EXPECT_EQ(PacketBuffer::kInvalidPointer,
+            buffer->NextHigherTimestamp(0, NULL));
+  delete buffer;
+
+  // Insert packet list of three packets, where the second packet has an invalid
+  // payload.  Expect first packet to be inserted, and the remaining two to be
+  // discarded.
+  buffer = new PacketBuffer(100, &tick_timer);  // 100 packets.
+  PacketList list;
+  list.push_back(gen.NextPacket(payload_len, nullptr));  // Valid packet.
+  {
+    Packet packet = gen.NextPacket(payload_len, nullptr);
+    packet.payload.Clear();  // Invalid.
+    list.push_back(std::move(packet));
+  }
+  list.push_back(gen.NextPacket(payload_len, nullptr));  // Valid packet.
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(0))
+      .WillRepeatedly(Return(&info));
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+  EXPECT_EQ(
+      PacketBuffer::kInvalidPacket,
+      buffer->InsertPacketList(/*packet_list=*/&list,
+                               /*decoder_database=*/decoder_database,
+                               /*current_rtp_payload_type=*/&current_pt,
+                               /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                               /*stats=*/&mock_stats,
+                               /*last_decoded_length=*/payload_len,
+                               /*sample_rate=*/1000,
+                               /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
+  EXPECT_EQ(1u, buffer->NumPacketsInBuffer());
+  delete buffer;
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test packet comparison function.
+// The function should return true if the first packet "goes before" the second.
+TEST(PacketBuffer, ComparePackets) {
+  PacketGenerator gen(0, 0, 0, 10);
+  Packet a(gen.NextPacket(10, nullptr));  // SN = 0, TS = 0.
+  Packet b(gen.NextPacket(10, nullptr));  // SN = 1, TS = 10.
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_TRUE(a < b);
+  EXPECT_FALSE(a > b);
+  EXPECT_TRUE(a <= b);
+  EXPECT_FALSE(a >= b);
+
+  // Testing wrap-around case; 'a' is earlier but has a larger timestamp value.
+  a.timestamp = 0xFFFFFFFF - 10;
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_TRUE(a < b);
+  EXPECT_FALSE(a > b);
+  EXPECT_TRUE(a <= b);
+  EXPECT_FALSE(a >= b);
+
+  // Test equal packets.
+  EXPECT_TRUE(a == a);
+  EXPECT_FALSE(a != a);
+  EXPECT_FALSE(a < a);
+  EXPECT_FALSE(a > a);
+  EXPECT_TRUE(a <= a);
+  EXPECT_TRUE(a >= a);
+
+  // Test equal timestamps but different sequence numbers (0 and 1).
+  a.timestamp = b.timestamp;
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_TRUE(a < b);
+  EXPECT_FALSE(a > b);
+  EXPECT_TRUE(a <= b);
+  EXPECT_FALSE(a >= b);
+
+  // Test equal timestamps but different sequence numbers (32767 and 1).
+  a.sequence_number = 0xFFFF;
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_TRUE(a < b);
+  EXPECT_FALSE(a > b);
+  EXPECT_TRUE(a <= b);
+  EXPECT_FALSE(a >= b);
+
+  // Test equal timestamps and sequence numbers, but differing priorities.
+  a.sequence_number = b.sequence_number;
+  a.priority = {1, 0};
+  b.priority = {0, 0};
+  // a after b
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_FALSE(a < b);
+  EXPECT_TRUE(a > b);
+  EXPECT_FALSE(a <= b);
+  EXPECT_TRUE(a >= b);
+
+  Packet c(gen.NextPacket(0, nullptr));  // SN = 2, TS = 20.
+  Packet d(gen.NextPacket(0, nullptr));  // SN = 3, TS = 20.
+  c.timestamp = b.timestamp;
+  d.timestamp = b.timestamp;
+  c.sequence_number = b.sequence_number;
+  d.sequence_number = b.sequence_number;
+  c.priority = {1, 1};
+  d.priority = {0, 1};
+  // c after d
+  EXPECT_FALSE(c == d);
+  EXPECT_TRUE(c != d);
+  EXPECT_FALSE(c < d);
+  EXPECT_TRUE(c > d);
+  EXPECT_FALSE(c <= d);
+  EXPECT_TRUE(c >= d);
+
+  // c after a
+  EXPECT_FALSE(c == a);
+  EXPECT_TRUE(c != a);
+  EXPECT_FALSE(c < a);
+  EXPECT_TRUE(c > a);
+  EXPECT_FALSE(c <= a);
+  EXPECT_TRUE(c >= a);
+
+  // c after b
+  EXPECT_FALSE(c == b);
+  EXPECT_TRUE(c != b);
+  EXPECT_FALSE(c < b);
+  EXPECT_TRUE(c > b);
+  EXPECT_FALSE(c <= b);
+  EXPECT_TRUE(c >= b);
+
+  // a after d
+  EXPECT_FALSE(a == d);
+  EXPECT_TRUE(a != d);
+  EXPECT_FALSE(a < d);
+  EXPECT_TRUE(a > d);
+  EXPECT_FALSE(a <= d);
+  EXPECT_TRUE(a >= d);
+
+  // d after b
+  EXPECT_FALSE(d == b);
+  EXPECT_TRUE(d != b);
+  EXPECT_FALSE(d < b);
+  EXPECT_TRUE(d > b);
+  EXPECT_FALSE(d <= b);
+  EXPECT_TRUE(d >= b);
+}
+
+TEST(PacketBuffer, GetSpanSamples) {
+  constexpr size_t kFrameSizeSamples = 10;
+  constexpr int kPayloadSizeBytes = 1;  // Does not matter to this test;
+  constexpr uint32_t kStartTimeStamp = 0xFFFFFFFE;  // Close to wrap around.
+  constexpr int kSampleRateHz = 48000;
+  constexpr bool KCountDtxWaitingTime = false;
+  TickTimer tick_timer;
+  PacketBuffer buffer(3, &tick_timer);
+  PacketGenerator gen(0, kStartTimeStamp, 0, kFrameSizeSamples);
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  Packet packet_1 = gen.NextPacket(kPayloadSizeBytes, nullptr);
+
+  std::unique_ptr<MockEncodedAudioFrame> mock_audio_frame =
+      std::make_unique<MockEncodedAudioFrame>();
+  EXPECT_CALL(*mock_audio_frame, Duration())
+      .WillRepeatedly(Return(kFrameSizeSamples));
+  Packet packet_2 =
+      gen.NextPacket(kPayloadSizeBytes, std::move(mock_audio_frame));
+
+  RTC_DCHECK_GT(packet_1.timestamp,
+                packet_2.timestamp);  // Tmestamp wrapped around.
+
+  EXPECT_EQ(PacketBuffer::kOK,
+            buffer.InsertPacket(/*packet=*/std::move(packet_1),
+                                /*stats=*/&mock_stats,
+                                /*last_decoded_length=*/kFrameSizeSamples,
+                                /*sample_rate=*/1000,
+                                /*target_level_ms=*/60,
+                                /*decoder_database=*/decoder_database));
+
+  constexpr size_t kLastDecodedSizeSamples = 2;
+  // packet_1 has no access to duration, and relies last decoded duration as
+  // input.
+  EXPECT_EQ(kLastDecodedSizeSamples,
+            buffer.GetSpanSamples(kLastDecodedSizeSamples, kSampleRateHz,
+                                  KCountDtxWaitingTime));
+
+  EXPECT_EQ(PacketBuffer::kOK,
+            buffer.InsertPacket(/*packet=*/std::move(packet_2),
+                                /*stats=*/&mock_stats,
+                                /*last_decoded_length=*/kFrameSizeSamples,
+                                /*sample_rate=*/1000,
+                                /*target_level_ms=*/60,
+                                /*decoder_database=*/decoder_database));
+
+  EXPECT_EQ(kFrameSizeSamples * 2,
+            buffer.GetSpanSamples(0, kSampleRateHz, KCountDtxWaitingTime));
+
+  // packet_2 has access to duration, and ignores last decoded duration as
+  // input.
+  EXPECT_EQ(kFrameSizeSamples * 2,
+            buffer.GetSpanSamples(kLastDecodedSizeSamples, kSampleRateHz,
+                                  KCountDtxWaitingTime));
+}
+
+namespace {
+void TestIsObsoleteTimestamp(uint32_t limit_timestamp) {
+  // Check with zero horizon, which implies that the horizon is at 2^31, i.e.,
+  // half the timestamp range.
+  static const uint32_t kZeroHorizon = 0;
+  static const uint32_t k2Pow31Minus1 = 0x7FFFFFFF;
+  // Timestamp on the limit is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
+      limit_timestamp, limit_timestamp, kZeroHorizon));
+  // 1 sample behind is old.
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 1,
+                                                limit_timestamp, kZeroHorizon));
+  // 2^31 - 1 samples behind is old.
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - k2Pow31Minus1,
+                                                limit_timestamp, kZeroHorizon));
+  // 1 sample ahead is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
+      limit_timestamp + 1, limit_timestamp, kZeroHorizon));
+  // If |t1-t2|=2^31 and t1>t2, t2 is older than t1 but not the opposite.
+  uint32_t other_timestamp = limit_timestamp + (1 << 31);
+  uint32_t lowest_timestamp = std::min(limit_timestamp, other_timestamp);
+  uint32_t highest_timestamp = std::max(limit_timestamp, other_timestamp);
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(
+      lowest_timestamp, highest_timestamp, kZeroHorizon));
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
+      highest_timestamp, lowest_timestamp, kZeroHorizon));
+
+  // Fixed horizon at 10 samples.
+  static const uint32_t kHorizon = 10;
+  // Timestamp on the limit is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp,
+                                                 limit_timestamp, kHorizon));
+  // 1 sample behind is old.
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 1,
+                                                limit_timestamp, kHorizon));
+  // 9 samples behind is old.
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 9,
+                                                limit_timestamp, kHorizon));
+  // 10 samples behind is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 10,
+                                                 limit_timestamp, kHorizon));
+  // 2^31 - 1 samples behind is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
+      limit_timestamp - k2Pow31Minus1, limit_timestamp, kHorizon));
+  // 1 sample ahead is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp + 1,
+                                                 limit_timestamp, kHorizon));
+  // 2^31 samples ahead is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp + (1 << 31),
+                                                 limit_timestamp, kHorizon));
+}
+}  // namespace
+
+// Test the IsObsoleteTimestamp method with different limit timestamps.
+TEST(PacketBuffer, IsObsoleteTimestamp) {
+  TestIsObsoleteTimestamp(0);
+  TestIsObsoleteTimestamp(1);
+  TestIsObsoleteTimestamp(0xFFFFFFFF);  // -1 in uint32_t.
+  TestIsObsoleteTimestamp(0x80000000);  // 2^31.
+  TestIsObsoleteTimestamp(0x80000001);  // 2^31 + 1.
+  TestIsObsoleteTimestamp(0x7FFFFFFF);  // 2^31 - 1.
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc
new file mode 100644
index 0000000000..9999d6764b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/post_decode_vad.h"
+
+namespace webrtc {
+
+PostDecodeVad::~PostDecodeVad() {
+  if (vad_instance_)
+    WebRtcVad_Free(vad_instance_);
+}
+
+void PostDecodeVad::Enable() {
+  if (!vad_instance_) {
+    // Create the instance.
+    vad_instance_ = WebRtcVad_Create();
+    if (vad_instance_ == nullptr) {
+      // Failed to create instance.
+      Disable();
+      return;
+    }
+  }
+  Init();
+  enabled_ = true;
+}
+
+void PostDecodeVad::Disable() {
+  enabled_ = false;
+  running_ = false;
+}
+
+void PostDecodeVad::Init() {
+  running_ = false;
+  if (vad_instance_) {
+    WebRtcVad_Init(vad_instance_);
+    WebRtcVad_set_mode(vad_instance_, kVadMode);
+    running_ = true;
+  }
+}
+
+void PostDecodeVad::Update(int16_t* signal,
+                           size_t length,
+                           AudioDecoder::SpeechType speech_type,
+                           bool sid_frame,
+                           int fs_hz) {
+  if (!vad_instance_ || !enabled_) {
+    return;
+  }
+
+  if (speech_type == AudioDecoder::kComfortNoise || sid_frame ||
+      fs_hz > 16000) {
+    // TODO(hlundin): Remove restriction on fs_hz.
+    running_ = false;
+    active_speech_ = true;
+    sid_interval_counter_ = 0;
+  } else if (!running_) {
+    ++sid_interval_counter_;
+  }
+
+  if (sid_interval_counter_ >= kVadAutoEnable) {
+    Init();
+  }
+
+  if (length > 0 && running_) {
+    size_t vad_sample_index = 0;
+    active_speech_ = false;
+    // Loop through frame sizes 30, 20, and 10 ms.
+    for (int vad_frame_size_ms = 30; vad_frame_size_ms >= 10;
+         vad_frame_size_ms -= 10) {
+      size_t vad_frame_size_samples =
+          static_cast<size_t>(vad_frame_size_ms * fs_hz / 1000);
+      while (length - vad_sample_index >= vad_frame_size_samples) {
+        int vad_return =
+            WebRtcVad_Process(vad_instance_, fs_hz, &signal[vad_sample_index],
+                              vad_frame_size_samples);
+        active_speech_ |= (vad_return == 1);
+        vad_sample_index += vad_frame_size_samples;
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.h b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.h
new file mode 100644
index 0000000000..3bd91b9edb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
+#define MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "common_audio/vad/include/webrtc_vad.h"
+
+namespace webrtc {
+
+class PostDecodeVad {
+ public:
+  PostDecodeVad()
+      : enabled_(false),
+        running_(false),
+        active_speech_(true),
+        sid_interval_counter_(0),
+        vad_instance_(NULL) {}
+
+  virtual ~PostDecodeVad();
+
+  PostDecodeVad(const PostDecodeVad&) = delete;
+  PostDecodeVad& operator=(const PostDecodeVad&) = delete;
+
+  // Enables post-decode VAD.
+  void Enable();
+
+  // Disables post-decode VAD.
+  void Disable();
+
+  // Initializes post-decode VAD.
+  void Init();
+
+  // Updates post-decode VAD with the audio data in `signal` having `length`
+  // samples. The data is of type `speech_type`, at the sample rate `fs_hz`.
+  void Update(int16_t* signal,
+              size_t length,
+              AudioDecoder::SpeechType speech_type,
+              bool sid_frame,
+              int fs_hz);
+
+  // Accessors.
+  bool enabled() const { return enabled_; }
+  bool running() const { return running_; }
+  bool active_speech() const { return active_speech_; }
+
+ private:
+  static const int kVadMode = 0;  // Sets aggressiveness to "Normal".
+  // Number of Update() calls without CNG/SID before re-enabling VAD.
+  static const int kVadAutoEnable = 3000;
+
+  bool enabled_;
+  bool running_;
+  bool active_speech_;
+  int sid_interval_counter_;
+  ::VadInst* vad_instance_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc
new file mode 100644
index 0000000000..da3e4e864e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for PostDecodeVad class.
+
+#include "modules/audio_coding/neteq/post_decode_vad.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(PostDecodeVad, CreateAndDestroy) {
+  PostDecodeVad vad;
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.cc b/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.cc
new file mode 100644
index 0000000000..232170b177
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.cc
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/preemptive_expand.h"
+
+#include <algorithm>
+
+#include "api/array_view.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/time_stretch.h"
+
+namespace webrtc {
+
+PreemptiveExpand::ReturnCodes PreemptiveExpand::Process(
+    const int16_t* input,
+    size_t input_length,
+    size_t old_data_length,
+    AudioMultiVector* output,
+    size_t* length_change_samples) {
+  old_data_length_per_channel_ = old_data_length;
+  // Input length must be (almost) 30 ms.
+  // Also, the new part must be at least `overlap_samples_` elements.
+  static const size_t k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
+  if (num_channels_ == 0 ||
+      input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ ||
+      old_data_length >= input_length / num_channels_ - overlap_samples_) {
+    // Length of input data too short to do preemptive expand. Simply move all
+    // data from input to output.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kError;
+  }
+  const bool kFastMode = false;  // Fast mode is not available for PE Expand.
+  return TimeStretch::Process(input, input_length, kFastMode, output,
+                              length_change_samples);
+}
+
+void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
+                                                     int16_t* best_correlation,
+                                                     size_t* peak_index) const {
+  // When the signal does not contain any active speech, the correlation does
+  // not matter. Simply set it to zero.
+  *best_correlation = 0;
+
+  // For low energy expansion, the new data can be less than 15 ms,
+  // but we must ensure that best_correlation is not larger than the length of
+  // the new data.
+  // but we must ensure that best_correlation is not larger than the new data.
+  *peak_index = std::min(*peak_index, len - old_data_length_per_channel_);
+}
+
+PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
+    const int16_t* input,
+    size_t input_length,
+    size_t peak_index,
+    int16_t best_correlation,
+    bool active_speech,
+    bool /*fast_mode*/,
+    AudioMultiVector* output) const {
+  // Pre-calculate common multiplication with `fs_mult_`.
+  // 120 corresponds to 15 ms.
+  size_t fs_mult_120 = static_cast<size_t>(fs_mult_ * 120);
+  // Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
+  // or passive speech.
+  if (((best_correlation > kCorrelationThreshold) &&
+       (old_data_length_per_channel_ <= fs_mult_120)) ||
+      !active_speech) {
+    // Do accelerate operation by overlap add.
+
+    // Set length of the first part, not to be modified.
+    size_t unmodified_length =
+        std::max(old_data_length_per_channel_, fs_mult_120);
+    // Copy first part, including cross-fade region.
+    output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        input, (unmodified_length + peak_index) * num_channels_));
+    // Copy the last `peak_index` samples up to 15 ms to `temp_vector`.
+    AudioMultiVector temp_vector(num_channels_);
+    temp_vector.PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[(unmodified_length - peak_index) * num_channels_],
+        peak_index * num_channels_));
+    // Cross-fade `temp_vector` onto the end of `output`.
+    output->CrossFade(temp_vector, peak_index);
+    // Copy the last unmodified part, 15 ms + pitch period until the end.
+    output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[unmodified_length * num_channels_],
+        input_length - unmodified_length * num_channels_));
+
+    if (active_speech) {
+      return kSuccess;
+    } else {
+      return kSuccessLowEnergy;
+    }
+  } else {
+    // Accelerate not allowed. Simply move all data from decoded to outData.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kNoStretch;
+  }
+}
+
+PreemptiveExpand* PreemptiveExpandFactory::Create(
+    int sample_rate_hz,
+    size_t num_channels,
+    const BackgroundNoise& background_noise,
+    size_t overlap_samples) const {
+  return new PreemptiveExpand(sample_rate_hz, num_channels, background_noise,
+                              overlap_samples);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.h b/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.h
new file mode 100644
index 0000000000..6338b993fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
+#define MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/neteq/time_stretch.h"
+
+namespace webrtc {
+
+class AudioMultiVector;
+class BackgroundNoise;
+
+// This class implements the PreemptiveExpand operation. Most of the work is
+// done in the base class TimeStretch, which is shared with the Accelerate
+// operation. In the PreemptiveExpand class, the operations that are specific to
+// PreemptiveExpand are implemented.
+class PreemptiveExpand : public TimeStretch {
+ public:
+  PreemptiveExpand(int sample_rate_hz,
+                   size_t num_channels,
+                   const BackgroundNoise& background_noise,
+                   size_t overlap_samples)
+      : TimeStretch(sample_rate_hz, num_channels, background_noise),
+        old_data_length_per_channel_(0),
+        overlap_samples_(overlap_samples) {}
+
+  PreemptiveExpand(const PreemptiveExpand&) = delete;
+  PreemptiveExpand& operator=(const PreemptiveExpand&) = delete;
+
+  // This method performs the actual PreemptiveExpand operation. The samples are
+  // read from `input`, of length `input_length` elements, and are written to
+  // `output`. The number of samples added through time-stretching is
+  // is provided in the output `length_change_samples`. The method returns
+  // the outcome of the operation as an enumerator value.
+  ReturnCodes Process(const int16_t* pw16_decoded,
+                      size_t len,
+                      size_t old_data_len,
+                      AudioMultiVector* output,
+                      size_t* length_change_samples);
+
+ protected:
+  // Sets the parameters `best_correlation` and `peak_index` to suitable
+  // values when the signal contains no active speech.
+  void SetParametersForPassiveSpeech(size_t input_length,
+                                     int16_t* best_correlation,
+                                     size_t* peak_index) const override;
+
+  // Checks the criteria for performing the time-stretching operation and,
+  // if possible, performs the time-stretching.
+  ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
+                                      size_t input_length,
+                                      size_t peak_index,
+                                      int16_t best_correlation,
+                                      bool active_speech,
+                                      bool /*fast_mode*/,
+                                      AudioMultiVector* output) const override;
+
+ private:
+  size_t old_data_length_per_channel_;
+  size_t overlap_samples_;
+};
+
+struct PreemptiveExpandFactory {
+  PreemptiveExpandFactory() {}
+  virtual ~PreemptiveExpandFactory() {}
+
+  virtual PreemptiveExpand* Create(int sample_rate_hz,
+                                   size_t num_channels,
+                                   const BackgroundNoise& background_noise,
+                                   size_t overlap_samples) const;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.cc b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.cc
new file mode 100644
index 0000000000..ada175831c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/random_vector.h"
+
+namespace webrtc {
+
+const int16_t RandomVector::kRandomTable[RandomVector::kRandomTableSize] = {
+    2680,   5532,   441,    5520,   16170,  -5146,  -1024,  -8733,  3115,
+    9598,   -10380, -4959,  -1280,  -21716, 7133,   -1522,  13458,  -3902,
+    2789,   -675,   3441,   5016,   -13599, -4003,  -2739,  3922,   -7209,
+    13352,  -11617, -7241,  12905,  -2314,  5426,   10121,  -9702,  11207,
+    -13542, 1373,   816,    -5934,  -12504, 4798,   1811,   4112,   -613,
+    201,    -10367, -2960,  -2419,  3442,   4299,   -6116,  -6092,  1552,
+    -1650,  -480,   -1237,  18720,  -11858, -8303,  -8212,  865,    -2890,
+    -16968, 12052,  -5845,  -5912,  9777,   -5665,  -6294,  5426,   -4737,
+    -6335,  1652,   761,    3832,   641,    -8552,  -9084,  -5753,  8146,
+    12156,  -4915,  15086,  -1231,  -1869,  11749,  -9319,  -6403,  11407,
+    6232,   -1683,  24340,  -11166, 4017,   -10448, 3153,   -2936,  6212,
+    2891,   -866,   -404,   -4807,  -2324,  -1917,  -2388,  -6470,  -3895,
+    -10300, 5323,   -5403,  2205,   4640,   7022,   -21186, -6244,  -882,
+    -10031, -3395,  -12885, 7155,   -5339,  5079,   -2645,  -9515,  6622,
+    14651,  15852,  359,    122,    8246,   -3502,  -6696,  -3679,  -13535,
+    -1409,  -704,   -7403,  -4007,  1798,   279,    -420,   -12796, -14219,
+    1141,   3359,   11434,  7049,   -6684,  -7473,  14283,  -4115,  -9123,
+    -8969,  4152,   4117,   13792,  5742,   16168,  8661,   -1609,  -6095,
+    1881,   14380,  -5588,  6758,   -6425,  -22969, -7269,  7031,   1119,
+    -1611,  -5850,  -11281, 3559,   -8952,  -10146, -4667,  -16251, -1538,
+    2062,   -1012,  -13073, 227,    -3142,  -5265,  20,     5770,   -7559,
+    4740,   -4819,  992,    -8208,  -7130,  -4652,  6725,   7369,   -1036,
+    13144,  -1588,  -5304,  -2344,  -449,   -5705,  -8894,  5205,   -17904,
+    -11188, -1022,  4852,   10101,  -5255,  -4200,  -752,   7941,   -1543,
+    5959,   14719,  13346,  17045,  -15605, -1678,  -1600,  -9230,  68,
+    23348,  1172,   7750,   11212,  -18227, 9956,   4161,   883,    3947,
+    4341,   1014,   -4889,  -2603,  1246,   -5630,  -3596,  -870,   -1298,
+    2784,   -3317,  -6612,  -20541, 4166,   4181,   -8625,  3562,   12890,
+    4761,   3205,   -12259, -8579};
+
+void RandomVector::Reset() {
+  seed_ = 777;
+  seed_increment_ = 1;
+}
+
+void RandomVector::Generate(size_t length, int16_t* output) {
+  for (size_t i = 0; i < length; i++) {
+    seed_ += seed_increment_;
+    size_t position = seed_ & (kRandomTableSize - 1);
+    output[i] = kRandomTable[position];
+  }
+}
+
+void RandomVector::IncreaseSeedIncrement(int16_t increase_by) {
+  seed_increment_ += increase_by;
+  seed_increment_ &= kRandomTableSize - 1;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.h b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.h
new file mode 100644
index 0000000000..4a782f1116
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// This class generates pseudo-random samples.
+class RandomVector {
+ public:
+  static const size_t kRandomTableSize = 256;
+  static const int16_t kRandomTable[kRandomTableSize];
+
+  RandomVector() : seed_(777), seed_increment_(1) {}
+
+  RandomVector(const RandomVector&) = delete;
+  RandomVector& operator=(const RandomVector&) = delete;
+
+  void Reset();
+
+  void Generate(size_t length, int16_t* output);
+
+  void IncreaseSeedIncrement(int16_t increase_by);
+
+  // Accessors and mutators.
+  int16_t seed_increment() { return seed_increment_; }
+  void set_seed_increment(int16_t value) { seed_increment_ = value; }
+
+ private:
+  uint32_t seed_;
+  int16_t seed_increment_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/random_vector_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector_unittest.cc
new file mode 100644
index 0000000000..44479a6dd6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector_unittest.cc
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for RandomVector class.
+
+#include "modules/audio_coding/neteq/random_vector.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(RandomVector, CreateAndDestroy) {
+  RandomVector random_vector;
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.cc
new file mode 100644
index 0000000000..cec9f2f8a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.cc
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <list>
+#include <utility>
+#include <vector>
+
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+// The method loops through a list of packets {A, B, C, ...}. Each packet is
+// split into its corresponding RED payloads, {A1, A2, ...}, which is
+// temporarily held in the list `new_packets`.
+// When the first packet in `packet_list` has been processed, the original
+// packet is replaced by the new ones in `new_packets`, so that `packet_list`
+// becomes: {A1, A2, ..., B, C, ...}. The method then continues with B, and C,
+// until all the original packets have been replaced by their split payloads.
+bool RedPayloadSplitter::SplitRed(PacketList* packet_list) {
+  // Too many RED blocks indicates that something is wrong. Clamp it at some
+  // reasonable value.
+  const size_t kMaxRedBlocks = 32;
+  bool ret = true;
+  PacketList::iterator it = packet_list->begin();
+  while (it != packet_list->end()) {
+    const Packet& red_packet = *it;
+    RTC_DCHECK(!red_packet.payload.empty());
+    const uint8_t* payload_ptr = red_packet.payload.data();
+    size_t payload_length = red_packet.payload.size();
+
+    // Read RED headers (according to RFC 2198):
+    //
+    //    0                   1                   2                   3
+    //    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    //   |F|   block PT  |  timestamp offset         |   block length    |
+    //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // Last RED header:
+    //    0 1 2 3 4 5 6 7
+    //   +-+-+-+-+-+-+-+-+
+    //   |0|   Block PT  |
+    //   +-+-+-+-+-+-+-+-+
+
+    struct RedHeader {
+      uint8_t payload_type;
+      uint32_t timestamp;
+      size_t payload_length;
+    };
+
+    std::vector<RedHeader> new_headers;
+    bool last_block = false;
+    size_t sum_length = 0;
+    while (!last_block) {
+      if (payload_length == 0) {
+        RTC_LOG(LS_WARNING) << "SplitRed header too short";
+        return false;
+      }
+      RedHeader new_header;
+      // Check the F bit. If F == 0, this was the last block.
+      last_block = ((*payload_ptr & 0x80) == 0);
+      // Bits 1 through 7 are payload type.
+      new_header.payload_type = payload_ptr[0] & 0x7F;
+      if (last_block) {
+        // No more header data to read.
+        sum_length += kRedLastHeaderLength;  // Account for RED header size.
+        new_header.timestamp = red_packet.timestamp;
+        new_header.payload_length = red_packet.payload.size() - sum_length;
+        payload_ptr += kRedLastHeaderLength;  // Advance to first payload byte.
+        payload_length -= kRedLastHeaderLength;
+      } else {
+        if (payload_length < kRedHeaderLength) {
+          RTC_LOG(LS_WARNING) << "SplitRed header too short";
+          return false;
+        }
+        // Bits 8 through 21 are timestamp offset.
+        int timestamp_offset =
+            (payload_ptr[1] << 6) + ((payload_ptr[2] & 0xFC) >> 2);
+        new_header.timestamp = red_packet.timestamp - timestamp_offset;
+        // Bits 22 through 31 are payload length.
+        new_header.payload_length =
+            ((payload_ptr[2] & 0x03) << 8) + payload_ptr[3];
+
+        sum_length += new_header.payload_length;
+        sum_length += kRedHeaderLength;  // Account for RED header size.
+
+        payload_ptr += kRedHeaderLength;  // Advance to next RED header.
+        payload_length -= kRedHeaderLength;
+      }
+      // Store in new list of packets.
+      if (new_header.payload_length > 0) {
+        new_headers.push_back(new_header);
+      }
+    }
+
+    if (new_headers.size() <= kMaxRedBlocks) {
+      // Populate the new packets with payload data.
+      // `payload_ptr` now points at the first payload byte.
+      PacketList new_packets;  // An empty list to store the split packets in.
+      for (size_t i = 0; i != new_headers.size(); ++i) {
+        const auto& new_header = new_headers[i];
+        size_t payload_length = new_header.payload_length;
+        if (payload_ptr + payload_length >
+            red_packet.payload.data() + red_packet.payload.size()) {
+          // The block lengths in the RED headers do not match the overall
+          // packet length. Something is corrupt. Discard this and the remaining
+          // payloads from this packet.
+          RTC_LOG(LS_WARNING) << "SplitRed length mismatch";
+          ret = false;
+          break;
+        }
+
+        Packet new_packet;
+        new_packet.timestamp = new_header.timestamp;
+        new_packet.payload_type = new_header.payload_type;
+        new_packet.sequence_number = red_packet.sequence_number;
+        new_packet.priority.red_level =
+            rtc::dchecked_cast<int>((new_headers.size() - 1) - i);
+        new_packet.payload.SetData(payload_ptr, payload_length);
+        new_packet.packet_info = RtpPacketInfo(
+            /*ssrc=*/red_packet.packet_info.ssrc(),
+            /*csrcs=*/std::vector<uint32_t>(),
+            /*rtp_timestamp=*/new_packet.timestamp,
+            /*receive_time=*/red_packet.packet_info.receive_time());
+        new_packet.packet_info.set_audio_level(
+            red_packet.packet_info.audio_level());
+        new_packets.push_front(std::move(new_packet));
+        payload_ptr += payload_length;
+      }
+      // Insert new packets into original list, before the element pointed to by
+      // iterator `it`.
+      packet_list->splice(it, std::move(new_packets));
+    } else {
+      RTC_LOG(LS_WARNING) << "SplitRed too many blocks: " << new_headers.size();
+      ret = false;
+    }
+    // Remove `it` from the packet list. This operation effectively moves the
+    // iterator `it` to the next packet in the list. Thus, we do not have to
+    // increment it manually.
+    it = packet_list->erase(it);
+  }
+  return ret;
+}
+
+void RedPayloadSplitter::CheckRedPayloads(
+    PacketList* packet_list,
+    const DecoderDatabase& decoder_database) {
+  int main_payload_type = -1;
+  for (auto it = packet_list->begin(); it != packet_list->end(); /* */) {
+    uint8_t this_payload_type = it->payload_type;
+    if (decoder_database.IsRed(this_payload_type)) {
+      it = packet_list->erase(it);
+      continue;
+    }
+    if (!decoder_database.IsDtmf(this_payload_type) &&
+        !decoder_database.IsComfortNoise(this_payload_type)) {
+      if (main_payload_type == -1) {
+        // This is the first packet in the list which is non-DTMF non-CNG.
+        main_payload_type = this_payload_type;
+      } else {
+        if (this_payload_type != main_payload_type) {
+          // We do not allow redundant payloads of a different type.
+          // Remove `it` from the packet list. This operation effectively
+          // moves the iterator `it` to the next packet in the list. Thus, we
+          // do not have to increment it manually.
+          it = packet_list->erase(it);
+          continue;
+        }
+      }
+    }
+    ++it;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.h b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.h
new file mode 100644
index 0000000000..2f48e4b7d4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
+
+#include "modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+class DecoderDatabase;
+
+static const size_t kRedHeaderLength = 4;  // 4 bytes RED header.
+static const size_t kRedLastHeaderLength =
+    1;  // reduced size for last RED header.
+// This class handles splitting of RED payloads into smaller parts.
+// Codec-specific packet splitting can be performed by
+// AudioDecoder::ParsePayload.
+class RedPayloadSplitter {
+ public:
+  RedPayloadSplitter() {}
+
+  virtual ~RedPayloadSplitter() {}
+
+  RedPayloadSplitter(const RedPayloadSplitter&) = delete;
+  RedPayloadSplitter& operator=(const RedPayloadSplitter&) = delete;
+
+  // Splits each packet in `packet_list` into its separate RED payloads. Each
+  // RED payload is packetized into a Packet. The original elements in
+  // `packet_list` are properly deleted, and replaced by the new packets.
+  // Note that all packets in `packet_list` must be RED payloads, i.e., have
+  // RED headers according to RFC 2198 at the very beginning of the payload.
+  // Returns kOK or an error.
+  virtual bool SplitRed(PacketList* packet_list);
+
+  // Checks all packets in `packet_list`. Packets that are DTMF events or
+  // comfort noise payloads are kept. Except that, only one single payload type
+  // is accepted. Any packet with another payload type is discarded.
+  virtual void CheckRedPayloads(PacketList* packet_list,
+                                const DecoderDatabase& decoder_database);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
new file mode 100644
index 0000000000..a0ba5414ea
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
@@ -0,0 +1,390 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for RedPayloadSplitter class.
+
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
+
+
+#include <memory>
+#include <utility>  // pair
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
+
+using ::testing::Return;
+using ::testing::ReturnNull;
+
+namespace webrtc {
+
+static const int kRedPayloadType = 100;
+static const size_t kPayloadLength = 10;
+static const uint16_t kSequenceNumber = 0;
+static const uint32_t kBaseTimestamp = 0x12345678;
+
+// A possible Opus packet that contains FEC is the following.
+// The frame is 20 ms in duration.
+//
+// 0                   1                   2                   3
+// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |0|0|0|0|1|0|0|0|x|1|x|x|x|x|x|x|x|                             |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                             |
+// |                    Compressed frame 1 (N-2 bytes)...          :
+// :                                                               |
+// |                                                               |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+void CreateOpusFecPayload(uint8_t* payload,
+                          size_t payload_length,
+                          uint8_t payload_value) {
+  if (payload_length < 2) {
+    return;
+  }
+  payload[0] = 0x08;
+  payload[1] = 0x40;
+  memset(&payload[2], payload_value, payload_length - 2);
+}
+
+// RED headers (according to RFC 2198):
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |F|   block PT  |  timestamp offset         |   block length    |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Last RED header:
+//    0 1 2 3 4 5 6 7
+//   +-+-+-+-+-+-+-+-+
+//   |0|   Block PT  |
+//   +-+-+-+-+-+-+-+-+
+
+// Creates a RED packet, with `num_payloads` payloads, with payload types given
+// by the values in array `payload_types` (which must be of length
+// `num_payloads`). Each redundant payload is `timestamp_offset` samples
+// "behind" the the previous payload.
+Packet CreateRedPayload(size_t num_payloads,
+                        uint8_t* payload_types,
+                        int timestamp_offset,
+                        bool embed_opus_fec = false) {
+  Packet packet;
+  packet.payload_type = kRedPayloadType;
+  packet.timestamp = kBaseTimestamp;
+  packet.sequence_number = kSequenceNumber;
+  packet.payload.SetSize((kPayloadLength + 1) +
+                         (num_payloads - 1) *
+                             (kPayloadLength + kRedHeaderLength));
+  uint8_t* payload_ptr = packet.payload.data();
+  for (size_t i = 0; i < num_payloads; ++i) {
+    // Write the RED headers.
+    if (i == num_payloads - 1) {
+      // Special case for last payload.
+      *payload_ptr = payload_types[i] & 0x7F;  // F = 0;
+      ++payload_ptr;
+      break;
+    }
+    *payload_ptr = payload_types[i] & 0x7F;
+    // Not the last block; set F = 1.
+    *payload_ptr |= 0x80;
+    ++payload_ptr;
+    int this_offset =
+        rtc::checked_cast<int>((num_payloads - i - 1) * timestamp_offset);
+    *payload_ptr = this_offset >> 6;
+    ++payload_ptr;
+    RTC_DCHECK_LE(kPayloadLength, 1023);  // Max length described by 10 bits.
+    *payload_ptr = ((this_offset & 0x3F) << 2) | (kPayloadLength >> 8);
+    ++payload_ptr;
+    *payload_ptr = kPayloadLength & 0xFF;
+    ++payload_ptr;
+  }
+  for (size_t i = 0; i < num_payloads; ++i) {
+    // Write `i` to all bytes in each payload.
+    if (embed_opus_fec) {
+      CreateOpusFecPayload(payload_ptr, kPayloadLength,
+                           static_cast<uint8_t>(i));
+    } else {
+      memset(payload_ptr, static_cast<int>(i), kPayloadLength);
+    }
+    payload_ptr += kPayloadLength;
+  }
+  return packet;
+}
+
+// Create a packet with all payload bytes set to `payload_value`.
+Packet CreatePacket(uint8_t payload_type,
+                    size_t payload_length,
+                    uint8_t payload_value,
+                    bool opus_fec = false) {
+  Packet packet;
+  packet.payload_type = payload_type;
+  packet.timestamp = kBaseTimestamp;
+  packet.sequence_number = kSequenceNumber;
+  packet.payload.SetSize(payload_length);
+  if (opus_fec) {
+    CreateOpusFecPayload(packet.payload.data(), packet.payload.size(),
+                         payload_value);
+  } else {
+    memset(packet.payload.data(), payload_value, packet.payload.size());
+  }
+  return packet;
+}
+
+// Checks that `packet` has the attributes given in the remaining parameters.
+void VerifyPacket(const Packet& packet,
+                  size_t payload_length,
+                  uint8_t payload_type,
+                  uint16_t sequence_number,
+                  uint32_t timestamp,
+                  uint8_t payload_value,
+                  Packet::Priority priority) {
+  EXPECT_EQ(payload_length, packet.payload.size());
+  EXPECT_EQ(payload_type, packet.payload_type);
+  EXPECT_EQ(sequence_number, packet.sequence_number);
+  EXPECT_EQ(timestamp, packet.timestamp);
+  EXPECT_EQ(priority, packet.priority);
+  ASSERT_FALSE(packet.payload.empty());
+  for (size_t i = 0; i < packet.payload.size(); ++i) {
+    ASSERT_EQ(payload_value, packet.payload.data()[i]);
+  }
+}
+
+void VerifyPacket(const Packet& packet,
+                  size_t payload_length,
+                  uint8_t payload_type,
+                  uint16_t sequence_number,
+                  uint32_t timestamp,
+                  uint8_t payload_value,
+                  bool primary) {
+  return VerifyPacket(packet, payload_length, payload_type, sequence_number,
+                      timestamp, payload_value,
+                      Packet::Priority{0, primary ? 0 : 1});
+}
+
+// Start of test definitions.
+
+TEST(RedPayloadSplitter, CreateAndDestroy) {
+  RedPayloadSplitter* splitter = new RedPayloadSplitter;
+  delete splitter;
+}
+
+// Packet A is split into A1 and A2.
+TEST(RedPayloadSplitter, OnePacketTwoPayloads) {
+  uint8_t payload_types[] = {0, 0};
+  const int kTimestampOffset = 160;
+  PacketList packet_list;
+  packet_list.push_back(CreateRedPayload(2, payload_types, kTimestampOffset));
+  RedPayloadSplitter splitter;
+  EXPECT_TRUE(splitter.SplitRed(&packet_list));
+  ASSERT_EQ(2u, packet_list.size());
+  // Check first packet. The first in list should always be the primary payload.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[1],
+               kSequenceNumber, kBaseTimestamp, 1, true);
+  packet_list.pop_front();
+  // Check second packet.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber, kBaseTimestamp - kTimestampOffset, 0, false);
+}
+
+// Packets A and B are not split at all. Only the RED header in each packet is
+// removed.
+TEST(RedPayloadSplitter, TwoPacketsOnePayload) {
+  uint8_t payload_types[] = {0};
+  const int kTimestampOffset = 160;
+  // Create first packet, with a single RED payload.
+  PacketList packet_list;
+  packet_list.push_back(CreateRedPayload(1, payload_types, kTimestampOffset));
+  // Create second packet, with a single RED payload.
+  {
+    Packet packet = CreateRedPayload(1, payload_types, kTimestampOffset);
+    // Manually change timestamp and sequence number of second packet.
+    packet.timestamp += kTimestampOffset;
+    packet.sequence_number++;
+    packet_list.push_back(std::move(packet));
+  }
+  RedPayloadSplitter splitter;
+  EXPECT_TRUE(splitter.SplitRed(&packet_list));
+  ASSERT_EQ(2u, packet_list.size());
+  // Check first packet.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber, kBaseTimestamp, 0, true);
+  packet_list.pop_front();
+  // Check second packet.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber + 1, kBaseTimestamp + kTimestampOffset, 0, true);
+}
+
+// Packets A and B are split into packets A1, A2, A3, B1, B2, B3, with
+// attributes as follows:
+//
+//                  A1*   A2    A3    B1*   B2    B3
+// Payload type     0     1     2     0     1     2
+// Timestamp        b     b-o   b-2o  b+o   b     b-o
+// Sequence number  0     0     0     1     1     1
+//
+// b = kBaseTimestamp, o = kTimestampOffset, * = primary.
+TEST(RedPayloadSplitter, TwoPacketsThreePayloads) {
+  uint8_t payload_types[] = {2, 1, 0};  // Primary is the last one.
+  const int kTimestampOffset = 160;
+  // Create first packet, with 3 RED payloads.
+  PacketList packet_list;
+  packet_list.push_back(CreateRedPayload(3, payload_types, kTimestampOffset));
+  // Create first packet, with 3 RED payloads.
+  {
+    Packet packet = CreateRedPayload(3, payload_types, kTimestampOffset);
+    // Manually change timestamp and sequence number of second packet.
+    packet.timestamp += kTimestampOffset;
+    packet.sequence_number++;
+    packet_list.push_back(std::move(packet));
+  }
+  RedPayloadSplitter splitter;
+  EXPECT_TRUE(splitter.SplitRed(&packet_list));
+  ASSERT_EQ(6u, packet_list.size());
+  // Check first packet, A1.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[2],
+               kSequenceNumber, kBaseTimestamp, 2, {0, 0});
+  packet_list.pop_front();
+  // Check second packet, A2.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[1],
+               kSequenceNumber, kBaseTimestamp - kTimestampOffset, 1, {0, 1});
+  packet_list.pop_front();
+  // Check third packet, A3.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber, kBaseTimestamp - 2 * kTimestampOffset, 0,
+               {0, 2});
+  packet_list.pop_front();
+  // Check fourth packet, B1.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[2],
+               kSequenceNumber + 1, kBaseTimestamp + kTimestampOffset, 2,
+               {0, 0});
+  packet_list.pop_front();
+  // Check fifth packet, B2.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[1],
+               kSequenceNumber + 1, kBaseTimestamp, 1, {0, 1});
+  packet_list.pop_front();
+  // Check sixth packet, B3.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber + 1, kBaseTimestamp - kTimestampOffset, 0,
+               {0, 2});
+}
+
+// Creates a list with 4 packets with these payload types:
+// 0 = CNGnb
+// 1 = PCMu
+// 2 = DTMF (AVT)
+// 3 = iLBC
+// We expect the method CheckRedPayloads to discard the iLBC packet, since it
+// is a non-CNG, non-DTMF payload of another type than the first speech payload
+// found in the list (which is PCMu).
+TEST(RedPayloadSplitter, CheckRedPayloads) {
+  PacketList packet_list;
+  for (uint8_t i = 0; i <= 3; ++i) {
+    // Create packet with payload type `i`, payload length 10 bytes, all 0.
+    packet_list.push_back(CreatePacket(i, 10, 0));
+  }
+
+  // Use a real DecoderDatabase object here instead of a mock, since it is
+  // easier to just register the payload types and let the actual implementation
+  // do its job.
+  DecoderDatabase decoder_database(
+      rtc::make_ref_counted<MockAudioDecoderFactory>(), absl::nullopt);
+  decoder_database.RegisterPayload(0, SdpAudioFormat("cn", 8000, 1));
+  decoder_database.RegisterPayload(1, SdpAudioFormat("pcmu", 8000, 1));
+  decoder_database.RegisterPayload(2,
+                                   SdpAudioFormat("telephone-event", 8000, 1));
+  decoder_database.RegisterPayload(3, SdpAudioFormat("ilbc", 8000, 1));
+
+  RedPayloadSplitter splitter;
+  splitter.CheckRedPayloads(&packet_list, decoder_database);
+
+  ASSERT_EQ(3u, packet_list.size());  // Should have dropped the last packet.
+  // Verify packets. The loop verifies that payload types 0, 1, and 2 are in the
+  // list.
+  for (int i = 0; i <= 2; ++i) {
+    VerifyPacket(packet_list.front(), 10, i, kSequenceNumber, kBaseTimestamp, 0,
+                 true);
+    packet_list.pop_front();
+  }
+  EXPECT_TRUE(packet_list.empty());
+}
+
+// This test creates a RED packet where the payloads also have the payload type
+// for RED. That is, some kind of weird nested RED packet. This is not supported
+// and the splitter should discard all packets.
+TEST(RedPayloadSplitter, CheckRedPayloadsRecursiveRed) {
+  PacketList packet_list;
+  for (uint8_t i = 0; i <= 3; ++i) {
+    // Create packet with RED payload type, payload length 10 bytes, all 0.
+    packet_list.push_back(CreatePacket(kRedPayloadType, 10, 0));
+  }
+
+  // Use a real DecoderDatabase object here instead of a mock, since it is
+  // easier to just register the payload types and let the actual implementation
+  // do its job.
+  DecoderDatabase decoder_database(
+      rtc::make_ref_counted<MockAudioDecoderFactory>(), absl::nullopt);
+  decoder_database.RegisterPayload(kRedPayloadType,
+                                   SdpAudioFormat("red", 8000, 1));
+
+  RedPayloadSplitter splitter;
+  splitter.CheckRedPayloads(&packet_list, decoder_database);
+
+  EXPECT_TRUE(packet_list.empty());  // Should have dropped all packets.
+}
+
+// Packet A is split into A1, A2 and A3. But the length parameter is off, so
+// the last payloads should be discarded.
+TEST(RedPayloadSplitter, WrongPayloadLength) {
+  uint8_t payload_types[] = {0, 0, 0};
+  const int kTimestampOffset = 160;
+  PacketList packet_list;
+  {
+    Packet packet = CreateRedPayload(3, payload_types, kTimestampOffset);
+    // Manually tamper with the payload length of the packet.
+    // This is one byte too short for the second payload (out of three).
+    // We expect only the first payload to be returned.
+    packet.payload.SetSize(packet.payload.size() - (kPayloadLength + 1));
+    packet_list.push_back(std::move(packet));
+  }
+  RedPayloadSplitter splitter;
+  EXPECT_FALSE(splitter.SplitRed(&packet_list));
+  ASSERT_EQ(1u, packet_list.size());
+  // Check first packet.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber, kBaseTimestamp - 2 * kTimestampOffset, 0,
+               {0, 2});
+  packet_list.pop_front();
+}
+
+// Test that we reject packets too short to contain a RED header.
+TEST(RedPayloadSplitter, RejectsIncompleteHeaders) {
+  RedPayloadSplitter splitter;
+
+  uint8_t payload_types[] = {0, 0};
+  const int kTimestampOffset = 160;
+
+  PacketList packet_list;
+
+  // Truncate the packet such that the first block can not be parsed.
+  packet_list.push_back(CreateRedPayload(2, payload_types, kTimestampOffset));
+  packet_list.front().payload.SetSize(4);
+  EXPECT_FALSE(splitter.SplitRed(&packet_list));
+  EXPECT_FALSE(packet_list.empty());
+
+  // Truncate the packet such that the first block can not be parsed.
+  packet_list.front().payload.SetSize(3);
+  EXPECT_FALSE(splitter.SplitRed(&packet_list));
+  EXPECT_FALSE(packet_list.empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.cc
new file mode 100644
index 0000000000..f6e073fc88
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.cc
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/reorder_optimizer.h"
+
+#include <algorithm>
+#include <limits>
+#include <vector>
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDelayBuckets = 100;
+constexpr int kBucketSizeMs = 20;
+
+}  // namespace
+
+ReorderOptimizer::ReorderOptimizer(int forget_factor,
+                                   int ms_per_loss_percent,
+                                   absl::optional<int> start_forget_weight)
+    : histogram_(kDelayBuckets, forget_factor, start_forget_weight),
+      ms_per_loss_percent_(ms_per_loss_percent) {}
+
+void ReorderOptimizer::Update(int relative_delay_ms,
+                              bool reordered,
+                              int base_delay_ms) {
+  const int index = reordered ? relative_delay_ms / kBucketSizeMs : 0;
+  if (index < histogram_.NumBuckets()) {
+    // Maximum delay to register is 2000 ms.
+    histogram_.Add(index);
+  }
+  int bucket_index = MinimizeCostFunction(base_delay_ms);
+  optimal_delay_ms_ = (1 + bucket_index) * kBucketSizeMs;
+}
+
+void ReorderOptimizer::Reset() {
+  histogram_.Reset();
+  optimal_delay_ms_.reset();
+}
+
+int ReorderOptimizer::MinimizeCostFunction(int base_delay_ms) const {
+  const std::vector<int>& buckets = histogram_.buckets();
+
+  // Values are calculated in Q30.
+  int64_t loss_probability = 1 << 30;
+  int64_t min_cost = std::numeric_limits<int64_t>::max();
+  int min_bucket = 0;
+  for (int i = 0; i < static_cast<int>(buckets.size()); ++i) {
+    loss_probability -= buckets[i];
+    int64_t delay_ms =
+        static_cast<int64_t>(std::max(0, i * kBucketSizeMs - base_delay_ms))
+        << 30;
+    int64_t cost = delay_ms + 100 * ms_per_loss_percent_ * loss_probability;
+
+    if (cost < min_cost) {
+      min_cost = cost;
+      min_bucket = i;
+    }
+    if (loss_probability == 0) {
+      break;
+    }
+  }
+
+  return min_bucket;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.h b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.h
new file mode 100644
index 0000000000..06f6bc7e50
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_REORDER_OPTIMIZER_H_
+#define MODULES_AUDIO_CODING_NETEQ_REORDER_OPTIMIZER_H_
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/histogram.h"
+
+namespace webrtc {
+
+// Calculates an optimal delay to reduce the chance of missing reordered
+// packets. The delay/loss trade-off can be tune using the `ms_per_loss_percent`
+// parameter.
+class ReorderOptimizer {
+ public:
+  ReorderOptimizer(int forget_factor,
+                   int ms_per_loss_percent,
+                   absl::optional<int> start_forget_weight);
+
+  void Update(int relative_delay_ms, bool reordered, int base_delay_ms);
+
+  absl::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
+
+  void Reset();
+
+ private:
+  int MinimizeCostFunction(int base_delay_ms) const;
+
+  Histogram histogram_;
+  const int ms_per_loss_percent_;
+  absl::optional<int> optimal_delay_ms_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_REORDER_OPTIMIZER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer_unittest.cc
new file mode 100644
index 0000000000..aaa1062560
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer_unittest.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/reorder_optimizer.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kForgetFactor = 32745;  // 0.9993 in Q15.
+constexpr int kMsPerLossPercent = 20;
+constexpr int kStartForgetWeight = 1;
+
+}  // namespace
+
+TEST(ReorderOptimizerTest, OnlyIncreaseDelayForReorderedPackets) {
+  ReorderOptimizer reorder_optimizer(kForgetFactor, kMsPerLossPercent,
+                                     kStartForgetWeight);
+  EXPECT_FALSE(reorder_optimizer.GetOptimalDelayMs());
+
+  // Delay should not increase for in-order packets.
+  reorder_optimizer.Update(60, /*reordered=*/false, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 20);
+
+  reorder_optimizer.Update(100, /*reordered=*/false, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 20);
+
+  reorder_optimizer.Update(80, /*reordered=*/true, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 100);
+}
+
+TEST(ReorderOptimizerTest, AvoidIncreasingDelayWhenProbabilityIsLow) {
+  ReorderOptimizer reorder_optimizer(kForgetFactor, kMsPerLossPercent,
+                                     kStartForgetWeight);
+
+  reorder_optimizer.Update(40, /*reordered=*/true, 0);
+  reorder_optimizer.Update(40, /*reordered=*/true, 0);
+  reorder_optimizer.Update(40, /*reordered=*/true, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 60);
+
+  // The cost of the delay is too high relative the probability.
+  reorder_optimizer.Update(600, /*reordered=*/true, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 60);
+}
+
+TEST(ReorderOptimizerTest, BaseDelayIsSubtractedFromCost) {
+  constexpr int kBaseDelayMs = 200;
+  ReorderOptimizer reorder_optimizer(kForgetFactor, kMsPerLossPercent,
+                                     kStartForgetWeight);
+
+  reorder_optimizer.Update(40, /*reordered=*/true, kBaseDelayMs);
+  reorder_optimizer.Update(40, /*reordered=*/true, kBaseDelayMs);
+  reorder_optimizer.Update(40, /*reordered=*/true, kBaseDelayMs);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 60);
+
+  // The cost of the delay is too high relative the probability.
+  reorder_optimizer.Update(600, /*reordered=*/true, kBaseDelayMs);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 620);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.cc b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.cc
new file mode 100644
index 0000000000..52d3fa90f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -0,0 +1,394 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+
+#include <string.h>  // memset
+
+#include <algorithm>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+size_t AddIntToSizeTWithLowerCap(int a, size_t b) {
+  const size_t ret = b + a;
+  // If a + b is negative, resulting in a negative wrap, cap it to zero instead.
+  static_assert(sizeof(size_t) >= sizeof(int),
+                "int must not be wider than size_t for this to work");
+  return (a < 0 && ret > b) ? 0 : ret;
+}
+
+constexpr int kInterruptionLenMs = 150;
+}  // namespace
+
+// Allocating the static const so that it can be passed by reference to
+// RTC_DCHECK.
+const size_t StatisticsCalculator::kLenWaitingTimes;
+
+StatisticsCalculator::PeriodicUmaLogger::PeriodicUmaLogger(
+    absl::string_view uma_name,
+    int report_interval_ms,
+    int max_value)
+    : uma_name_(uma_name),
+      report_interval_ms_(report_interval_ms),
+      max_value_(max_value),
+      timer_(0) {}
+
+StatisticsCalculator::PeriodicUmaLogger::~PeriodicUmaLogger() = default;
+
+void StatisticsCalculator::PeriodicUmaLogger::AdvanceClock(int step_ms) {
+  timer_ += step_ms;
+  if (timer_ < report_interval_ms_) {
+    return;
+  }
+  LogToUma(Metric());
+  Reset();
+  timer_ -= report_interval_ms_;
+  RTC_DCHECK_GE(timer_, 0);
+}
+
+void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const {
+  RTC_HISTOGRAM_COUNTS_SPARSE(uma_name_, value, 1, max_value_, 50);
+}
+
+StatisticsCalculator::PeriodicUmaCount::PeriodicUmaCount(
+    absl::string_view uma_name,
+    int report_interval_ms,
+    int max_value)
+    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}
+
+StatisticsCalculator::PeriodicUmaCount::~PeriodicUmaCount() {
+  // Log the count for the current (incomplete) interval.
+  LogToUma(Metric());
+}
+
+void StatisticsCalculator::PeriodicUmaCount::RegisterSample() {
+  ++counter_;
+}
+
+int StatisticsCalculator::PeriodicUmaCount::Metric() const {
+  return counter_;
+}
+
+void StatisticsCalculator::PeriodicUmaCount::Reset() {
+  counter_ = 0;
+}
+
+StatisticsCalculator::PeriodicUmaAverage::PeriodicUmaAverage(
+    absl::string_view uma_name,
+    int report_interval_ms,
+    int max_value)
+    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}
+
+StatisticsCalculator::PeriodicUmaAverage::~PeriodicUmaAverage() {
+  // Log the average for the current (incomplete) interval.
+  LogToUma(Metric());
+}
+
+void StatisticsCalculator::PeriodicUmaAverage::RegisterSample(int value) {
+  sum_ += value;
+  ++counter_;
+}
+
+int StatisticsCalculator::PeriodicUmaAverage::Metric() const {
+  return counter_ == 0 ? 0 : static_cast<int>(sum_ / counter_);
+}
+
+void StatisticsCalculator::PeriodicUmaAverage::Reset() {
+  sum_ = 0.0;
+  counter_ = 0;
+}
+
+StatisticsCalculator::StatisticsCalculator()
+    : preemptive_samples_(0),
+      accelerate_samples_(0),
+      expanded_speech_samples_(0),
+      expanded_noise_samples_(0),
+      timestamps_since_last_report_(0),
+      secondary_decoded_samples_(0),
+      discarded_secondary_packets_(0),
+      delayed_packet_outage_counter_(
+          "WebRTC.Audio.DelayedPacketOutageEventsPerMinute",
+          60000,  // 60 seconds report interval.
+          100),
+      excess_buffer_delay_("WebRTC.Audio.AverageExcessBufferDelayMs",
+                           60000,  // 60 seconds report interval.
+                           1000),
+      buffer_full_counter_("WebRTC.Audio.JitterBufferFullPerMinute",
+                           60000,  // 60 seconds report interval.
+                           100) {}
+
+StatisticsCalculator::~StatisticsCalculator() = default;
+
+void StatisticsCalculator::Reset() {
+  preemptive_samples_ = 0;
+  accelerate_samples_ = 0;
+  expanded_speech_samples_ = 0;
+  expanded_noise_samples_ = 0;
+  secondary_decoded_samples_ = 0;
+  discarded_secondary_packets_ = 0;
+  waiting_times_.clear();
+}
+
+void StatisticsCalculator::ResetMcu() {
+  timestamps_since_last_report_ = 0;
+}
+
+void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples,
+                                                bool is_new_concealment_event) {
+  expanded_speech_samples_ += num_samples;
+  ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), true);
+  lifetime_stats_.concealment_events += is_new_concealment_event;
+}
+
+void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples,
+                                                bool is_new_concealment_event) {
+  expanded_noise_samples_ += num_samples;
+  ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), false);
+  lifetime_stats_.concealment_events += is_new_concealment_event;
+}
+
+void StatisticsCalculator::ExpandedVoiceSamplesCorrection(int num_samples) {
+  expanded_speech_samples_ =
+      AddIntToSizeTWithLowerCap(num_samples, expanded_speech_samples_);
+  ConcealedSamplesCorrection(num_samples, true);
+}
+
+void StatisticsCalculator::ExpandedNoiseSamplesCorrection(int num_samples) {
+  expanded_noise_samples_ =
+      AddIntToSizeTWithLowerCap(num_samples, expanded_noise_samples_);
+  ConcealedSamplesCorrection(num_samples, false);
+}
+
+void StatisticsCalculator::DecodedOutputPlayed() {
+  decoded_output_played_ = true;
+}
+
+void StatisticsCalculator::EndExpandEvent(int fs_hz) {
+  RTC_DCHECK_GE(lifetime_stats_.concealed_samples,
+                concealed_samples_at_event_end_);
+  const int event_duration_ms =
+      1000 *
+      (lifetime_stats_.concealed_samples - concealed_samples_at_event_end_) /
+      fs_hz;
+  if (event_duration_ms >= kInterruptionLenMs && decoded_output_played_) {
+    lifetime_stats_.interruption_count++;
+    lifetime_stats_.total_interruption_duration_ms += event_duration_ms;
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AudioInterruptionMs", event_duration_ms,
+                         /*min=*/150, /*max=*/5000, /*bucket_count=*/50);
+  }
+  concealed_samples_at_event_end_ = lifetime_stats_.concealed_samples;
+}
+
+void StatisticsCalculator::ConcealedSamplesCorrection(int num_samples,
+                                                      bool is_voice) {
+  if (num_samples < 0) {
+    // Store negative correction to subtract from future positive additions.
+    // See also the function comment in the header file.
+    concealed_samples_correction_ -= num_samples;
+    if (!is_voice) {
+      silent_concealed_samples_correction_ -= num_samples;
+    }
+    return;
+  }
+
+  const size_t canceled_out =
+      std::min(static_cast<size_t>(num_samples), concealed_samples_correction_);
+  concealed_samples_correction_ -= canceled_out;
+  lifetime_stats_.concealed_samples += num_samples - canceled_out;
+
+  if (!is_voice) {
+    const size_t silent_canceled_out = std::min(
+        static_cast<size_t>(num_samples), silent_concealed_samples_correction_);
+    silent_concealed_samples_correction_ -= silent_canceled_out;
+    lifetime_stats_.silent_concealed_samples +=
+        num_samples - silent_canceled_out;
+  }
+}
+
+void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
+  preemptive_samples_ += num_samples;
+  operations_and_state_.preemptive_samples += num_samples;
+  lifetime_stats_.inserted_samples_for_deceleration += num_samples;
+}
+
+void StatisticsCalculator::AcceleratedSamples(size_t num_samples) {
+  accelerate_samples_ += num_samples;
+  operations_and_state_.accelerate_samples += num_samples;
+  lifetime_stats_.removed_samples_for_acceleration += num_samples;
+}
+
+void StatisticsCalculator::GeneratedNoiseSamples(size_t num_samples) {
+  lifetime_stats_.generated_noise_samples += num_samples;
+}
+
+void StatisticsCalculator::PacketsDiscarded(size_t num_packets) {
+  lifetime_stats_.packets_discarded += num_packets;
+}
+
+void StatisticsCalculator::SecondaryPacketsDiscarded(size_t num_packets) {
+  discarded_secondary_packets_ += num_packets;
+  lifetime_stats_.fec_packets_discarded += num_packets;
+}
+
+void StatisticsCalculator::SecondaryPacketsReceived(size_t num_packets) {
+  lifetime_stats_.fec_packets_received += num_packets;
+}
+
+void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
+  const int time_step_ms =
+      rtc::CheckedDivExact(static_cast<int>(1000 * num_samples), fs_hz);
+  delayed_packet_outage_counter_.AdvanceClock(time_step_ms);
+  excess_buffer_delay_.AdvanceClock(time_step_ms);
+  buffer_full_counter_.AdvanceClock(time_step_ms);
+  timestamps_since_last_report_ += static_cast<uint32_t>(num_samples);
+  if (timestamps_since_last_report_ >
+      static_cast<uint32_t>(fs_hz * kMaxReportPeriod)) {
+    timestamps_since_last_report_ = 0;
+  }
+  lifetime_stats_.total_samples_received += num_samples;
+}
+
+void StatisticsCalculator::JitterBufferDelay(
+    size_t num_samples,
+    uint64_t waiting_time_ms,
+    uint64_t target_delay_ms,
+    uint64_t unlimited_target_delay_ms) {
+  lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * num_samples;
+  lifetime_stats_.jitter_buffer_target_delay_ms +=
+      target_delay_ms * num_samples;
+  lifetime_stats_.jitter_buffer_minimum_delay_ms +=
+      unlimited_target_delay_ms * num_samples;
+  lifetime_stats_.jitter_buffer_emitted_count += num_samples;
+}
+
+void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
+  secondary_decoded_samples_ += num_samples;
+}
+
+void StatisticsCalculator::FlushedPacketBuffer() {
+  operations_and_state_.packet_buffer_flushes++;
+  buffer_full_counter_.RegisterSample();
+}
+
+void StatisticsCalculator::ReceivedPacket() {
+  ++lifetime_stats_.jitter_buffer_packets_received;
+}
+
+void StatisticsCalculator::RelativePacketArrivalDelay(size_t delay_ms) {
+  lifetime_stats_.relative_packet_arrival_delay_ms += delay_ms;
+}
+
+void StatisticsCalculator::LogDelayedPacketOutageEvent(int num_samples,
+                                                       int fs_hz) {
+  int outage_duration_ms = num_samples / (fs_hz / 1000);
+  RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
+                       outage_duration_ms, 1 /* min */, 2000 /* max */,
+                       100 /* bucket count */);
+  delayed_packet_outage_counter_.RegisterSample();
+  lifetime_stats_.delayed_packet_outage_samples += num_samples;
+}
+
+void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
+  excess_buffer_delay_.RegisterSample(waiting_time_ms);
+  RTC_DCHECK_LE(waiting_times_.size(), kLenWaitingTimes);
+  if (waiting_times_.size() == kLenWaitingTimes) {
+    // Erase first value.
+    waiting_times_.pop_front();
+  }
+  waiting_times_.push_back(waiting_time_ms);
+  operations_and_state_.last_waiting_time_ms = waiting_time_ms;
+}
+
+void StatisticsCalculator::GetNetworkStatistics(size_t samples_per_packet,
+                                                NetEqNetworkStatistics* stats) {
+  RTC_DCHECK(stats);
+
+  stats->accelerate_rate =
+      CalculateQ14Ratio(accelerate_samples_, timestamps_since_last_report_);
+
+  stats->preemptive_rate =
+      CalculateQ14Ratio(preemptive_samples_, timestamps_since_last_report_);
+
+  stats->expand_rate =
+      CalculateQ14Ratio(expanded_speech_samples_ + expanded_noise_samples_,
+                        timestamps_since_last_report_);
+
+  stats->speech_expand_rate = CalculateQ14Ratio(expanded_speech_samples_,
+                                                timestamps_since_last_report_);
+
+  stats->secondary_decoded_rate = CalculateQ14Ratio(
+      secondary_decoded_samples_, timestamps_since_last_report_);
+
+  const size_t discarded_secondary_samples =
+      discarded_secondary_packets_ * samples_per_packet;
+  stats->secondary_discarded_rate =
+      CalculateQ14Ratio(discarded_secondary_samples,
+                        static_cast<uint32_t>(discarded_secondary_samples +
+                                              secondary_decoded_samples_));
+
+  if (waiting_times_.size() == 0) {
+    stats->mean_waiting_time_ms = -1;
+    stats->median_waiting_time_ms = -1;
+    stats->min_waiting_time_ms = -1;
+    stats->max_waiting_time_ms = -1;
+  } else {
+    std::sort(waiting_times_.begin(), waiting_times_.end());
+    // Find mid-point elements. If the size is odd, the two values
+    // `middle_left` and `middle_right` will both be the one middle element; if
+    // the size is even, they will be the the two neighboring elements at the
+    // middle of the list.
+    const int middle_left = waiting_times_[(waiting_times_.size() - 1) / 2];
+    const int middle_right = waiting_times_[waiting_times_.size() / 2];
+    // Calculate the average of the two. (Works also for odd sizes.)
+    stats->median_waiting_time_ms = (middle_left + middle_right) / 2;
+    stats->min_waiting_time_ms = waiting_times_.front();
+    stats->max_waiting_time_ms = waiting_times_.back();
+    double sum = 0;
+    for (auto time : waiting_times_) {
+      sum += time;
+    }
+    stats->mean_waiting_time_ms = static_cast<int>(sum / waiting_times_.size());
+  }
+
+  // Reset counters.
+  ResetMcu();
+  Reset();
+}
+
+NetEqLifetimeStatistics StatisticsCalculator::GetLifetimeStatistics() const {
+  return lifetime_stats_;
+}
+
+NetEqOperationsAndState StatisticsCalculator::GetOperationsAndState() const {
+  return operations_and_state_;
+}
+
+uint16_t StatisticsCalculator::CalculateQ14Ratio(size_t numerator,
+                                                 uint32_t denominator) {
+  if (numerator == 0) {
+    return 0;
+  } else if (numerator < denominator) {
+    // Ratio must be smaller than 1 in Q14.
+    RTC_DCHECK_LT((numerator << 14) / denominator, (1 << 14));
+    return static_cast<uint16_t>((numerator << 14) / denominator);
+  } else {
+    // Will not produce a ratio larger than 1, since this is probably an error.
+    return 1 << 14;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.h b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.h
new file mode 100644
index 0000000000..33a22d02dd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.h
@@ -0,0 +1,210 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
+
+#include <deque>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/neteq/neteq.h"
+
+namespace webrtc {
+
+class DelayManager;
+
+// This class handles various network statistics in NetEq.
+class StatisticsCalculator {
+ public:
+  StatisticsCalculator();
+
+  virtual ~StatisticsCalculator();
+
+  StatisticsCalculator(const StatisticsCalculator&) = delete;
+  StatisticsCalculator& operator=(const StatisticsCalculator&) = delete;
+
+  // Resets most of the counters.
+  void Reset();
+
+  // Resets the counters that are not handled by Reset().
+  void ResetMcu();
+
+  // Reports that `num_samples` samples were produced through expansion, and
+  // that the expansion produced other than just noise samples.
+  void ExpandedVoiceSamples(size_t num_samples, bool is_new_concealment_event);
+
+  // Reports that `num_samples` samples were produced through expansion, and
+  // that the expansion produced only noise samples.
+  void ExpandedNoiseSamples(size_t num_samples, bool is_new_concealment_event);
+
+  // Corrects the statistics for number of samples produced through non-noise
+  // expansion by adding `num_samples` (negative or positive) to the current
+  // value. The result is capped to zero to avoid negative values.
+  void ExpandedVoiceSamplesCorrection(int num_samples);
+
+  // Same as ExpandedVoiceSamplesCorrection but for noise samples.
+  void ExpandedNoiseSamplesCorrection(int num_samples);
+
+  void DecodedOutputPlayed();
+
+  // Mark end of expand event; triggers some stats to be reported.
+  void EndExpandEvent(int fs_hz);
+
+  // Reports that `num_samples` samples were produced through preemptive
+  // expansion.
+  void PreemptiveExpandedSamples(size_t num_samples);
+
+  // Reports that `num_samples` samples were removed through accelerate.
+  void AcceleratedSamples(size_t num_samples);
+
+  // Reports that `num_samples` comfort noise samples were generated.
+  void GeneratedNoiseSamples(size_t num_samples);
+
+  // Reports that `num_packets` packets were discarded.
+  virtual void PacketsDiscarded(size_t num_packets);
+
+  // Reports that `num_packets` secondary (FEC) packets were discarded.
+  virtual void SecondaryPacketsDiscarded(size_t num_packets);
+
+  // Reports that `num_packets` secondary (FEC) packets were received.
+  virtual void SecondaryPacketsReceived(size_t num_packets);
+
+  // Increases the report interval counter with `num_samples` at a sample rate
+  // of `fs_hz`. This is how the StatisticsCalculator gets notified that current
+  // time is increasing.
+  void IncreaseCounter(size_t num_samples, int fs_hz);
+
+  // Update jitter buffer delay counter.
+  void JitterBufferDelay(size_t num_samples,
+                         uint64_t waiting_time_ms,
+                         uint64_t target_delay_ms,
+                         uint64_t unlimited_target_delay_ms);
+
+  // Stores new packet waiting time in waiting time statistics.
+  void StoreWaitingTime(int waiting_time_ms);
+
+  // Reports that `num_samples` samples were decoded from secondary packets.
+  void SecondaryDecodedSamples(int num_samples);
+
+  // Reports that the packet buffer was flushed.
+  void FlushedPacketBuffer();
+
+  // Reports that the jitter buffer received a packet.
+  void ReceivedPacket();
+
+  // Reports that a received packet was delayed by `delay_ms` milliseconds.
+  virtual void RelativePacketArrivalDelay(size_t delay_ms);
+
+  // Logs a delayed packet outage event of `num_samples` expanded at a sample
+  // rate of `fs_hz`. A delayed packet outage event is defined as an expand
+  // period caused not by an actual packet loss, but by a delayed packet.
+  virtual void LogDelayedPacketOutageEvent(int num_samples, int fs_hz);
+
+  // Returns the current network statistics in `stats`. The number of samples
+  // per packet is `samples_per_packet`. The method does not populate
+  // `preferred_buffer_size_ms`, `jitter_peaks_found` or `clockdrift_ppm`; use
+  // the PopulateDelayManagerStats method for those.
+  void GetNetworkStatistics(size_t samples_per_packet,
+                            NetEqNetworkStatistics* stats);
+
+  // Returns a copy of this class's lifetime statistics. These statistics are
+  // never reset.
+  NetEqLifetimeStatistics GetLifetimeStatistics() const;
+
+  NetEqOperationsAndState GetOperationsAndState() const;
+
+ private:
+  static const int kMaxReportPeriod = 60;  // Seconds before auto-reset.
+  static const size_t kLenWaitingTimes = 100;
+
+  class PeriodicUmaLogger {
+   public:
+    PeriodicUmaLogger(absl::string_view uma_name,
+                      int report_interval_ms,
+                      int max_value);
+    virtual ~PeriodicUmaLogger();
+    void AdvanceClock(int step_ms);
+
+   protected:
+    void LogToUma(int value) const;
+    virtual int Metric() const = 0;
+    virtual void Reset() = 0;
+
+    const std::string uma_name_;
+    const int report_interval_ms_;
+    const int max_value_;
+    int timer_ = 0;
+  };
+
+  class PeriodicUmaCount final : public PeriodicUmaLogger {
+   public:
+    PeriodicUmaCount(absl::string_view uma_name,
+                     int report_interval_ms,
+                     int max_value);
+    ~PeriodicUmaCount() override;
+    void RegisterSample();
+
+   protected:
+    int Metric() const override;
+    void Reset() override;
+
+   private:
+    int counter_ = 0;
+  };
+
+  class PeriodicUmaAverage final : public PeriodicUmaLogger {
+   public:
+    PeriodicUmaAverage(absl::string_view uma_name,
+                       int report_interval_ms,
+                       int max_value);
+    ~PeriodicUmaAverage() override;
+    void RegisterSample(int value);
+
+   protected:
+    int Metric() const override;
+    void Reset() override;
+
+   private:
+    double sum_ = 0.0;
+    int counter_ = 0;
+  };
+
+  // Corrects the concealed samples counter in lifetime_stats_. The value of
+  // num_samples_ is added directly to the stat if the correction is positive.
+  // If the correction is negative, it is cached and will be subtracted against
+  // future additions to the counter. This is meant to be called from
+  // Expanded{Voice,Noise}Samples{Correction}.
+  void ConcealedSamplesCorrection(int num_samples, bool is_voice);
+
+  // Calculates numerator / denominator, and returns the value in Q14.
+  static uint16_t CalculateQ14Ratio(size_t numerator, uint32_t denominator);
+
+  NetEqLifetimeStatistics lifetime_stats_;
+  NetEqOperationsAndState operations_and_state_;
+  size_t concealed_samples_correction_ = 0;
+  size_t silent_concealed_samples_correction_ = 0;
+  size_t preemptive_samples_;
+  size_t accelerate_samples_;
+  size_t expanded_speech_samples_;
+  size_t expanded_noise_samples_;
+  size_t concealed_samples_at_event_end_ = 0;
+  uint32_t timestamps_since_last_report_;
+  std::deque<int> waiting_times_;
+  uint32_t secondary_decoded_samples_;
+  size_t discarded_secondary_packets_;
+  PeriodicUmaCount delayed_packet_outage_counter_;
+  PeriodicUmaAverage excess_buffer_delay_;
+  PeriodicUmaCount buffer_full_counter_;
+  bool decoded_output_played_ = false;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator_unittest.cc
new file mode 100644
index 0000000000..491cd83dc4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator_unittest.cc
@@ -0,0 +1,206 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(LifetimeStatistics, TotalSamplesReceived) {
+  StatisticsCalculator stats;
+  for (int i = 0; i < 10; ++i) {
+    stats.IncreaseCounter(480, 48000);  // 10 ms at 48 kHz.
+  }
+  EXPECT_EQ(10 * 480u, stats.GetLifetimeStatistics().total_samples_received);
+}
+
+TEST(LifetimeStatistics, SamplesConcealed) {
+  StatisticsCalculator stats;
+  stats.ExpandedVoiceSamples(100, false);
+  stats.ExpandedNoiseSamples(17, false);
+  EXPECT_EQ(100u + 17u, stats.GetLifetimeStatistics().concealed_samples);
+}
+
+// This test verifies that a negative correction of concealed_samples does not
+// result in a decrease in the stats value (because stats-consuming applications
+// would not expect the value to decrease). Instead, the correction should be
+// made to future increments to the stat.
+TEST(LifetimeStatistics, SamplesConcealedCorrection) {
+  StatisticsCalculator stats;
+  stats.ExpandedVoiceSamples(100, false);
+  EXPECT_EQ(100u, stats.GetLifetimeStatistics().concealed_samples);
+  stats.ExpandedVoiceSamplesCorrection(-10);
+  // Do not subtract directly, but keep the correction for later.
+  EXPECT_EQ(100u, stats.GetLifetimeStatistics().concealed_samples);
+  stats.ExpandedVoiceSamplesCorrection(20);
+  // The total correction is 20 - 10.
+  EXPECT_EQ(110u, stats.GetLifetimeStatistics().concealed_samples);
+
+  // Also test correction done to the next ExpandedVoiceSamples call.
+  stats.ExpandedVoiceSamplesCorrection(-17);
+  EXPECT_EQ(110u, stats.GetLifetimeStatistics().concealed_samples);
+  stats.ExpandedVoiceSamples(100, false);
+  EXPECT_EQ(110u + 100u - 17u, stats.GetLifetimeStatistics().concealed_samples);
+}
+
+// This test verifies that neither "accelerate" nor "pre-emptive expand" reults
+// in a modification to concealed_samples stats. Only PLC operations (i.e.,
+// "expand" and "merge") should affect the stat.
+TEST(LifetimeStatistics, NoUpdateOnTimeStretch) {
+  StatisticsCalculator stats;
+  stats.ExpandedVoiceSamples(100, false);
+  stats.AcceleratedSamples(4711);
+  stats.PreemptiveExpandedSamples(17);
+  stats.ExpandedVoiceSamples(100, false);
+  EXPECT_EQ(200u, stats.GetLifetimeStatistics().concealed_samples);
+}
+
+TEST(StatisticsCalculator, ExpandedSamplesCorrection) {
+  StatisticsCalculator stats;
+  NetEqNetworkStatistics stats_output;
+  constexpr int kSampleRateHz = 48000;
+  constexpr int k10MsSamples = kSampleRateHz / 100;
+  constexpr int kPacketSizeMs = 20;
+  constexpr size_t kSamplesPerPacket = kPacketSizeMs * kSampleRateHz / 1000;
+
+  // Advance time by 10 ms.
+  stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
+
+  stats.GetNetworkStatistics(kSamplesPerPacket, &stats_output);
+
+  EXPECT_EQ(0u, stats_output.expand_rate);
+  EXPECT_EQ(0u, stats_output.speech_expand_rate);
+
+  // Correct with a negative value.
+  stats.ExpandedVoiceSamplesCorrection(-100);
+  stats.ExpandedNoiseSamplesCorrection(-100);
+  stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
+  stats.GetNetworkStatistics(kSamplesPerPacket, &stats_output);
+  // Expect no change, since negative values are disallowed.
+  EXPECT_EQ(0u, stats_output.expand_rate);
+  EXPECT_EQ(0u, stats_output.speech_expand_rate);
+
+  // Correct with a positive value.
+  stats.ExpandedVoiceSamplesCorrection(50);
+  stats.ExpandedNoiseSamplesCorrection(200);
+  stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
+  stats.GetNetworkStatistics(kSamplesPerPacket, &stats_output);
+  // Calculate expected rates in Q14. Expand rate is noise + voice, while
+  // speech expand rate is only voice.
+  EXPECT_EQ(((50u + 200u) << 14) / k10MsSamples, stats_output.expand_rate);
+  EXPECT_EQ((50u << 14) / k10MsSamples, stats_output.speech_expand_rate);
+}
+
+TEST(StatisticsCalculator, RelativePacketArrivalDelay) {
+  StatisticsCalculator stats;
+
+  stats.RelativePacketArrivalDelay(50);
+  NetEqLifetimeStatistics stats_output = stats.GetLifetimeStatistics();
+  EXPECT_EQ(50u, stats_output.relative_packet_arrival_delay_ms);
+
+  stats.RelativePacketArrivalDelay(20);
+  stats_output = stats.GetLifetimeStatistics();
+  EXPECT_EQ(70u, stats_output.relative_packet_arrival_delay_ms);
+}
+
+TEST(StatisticsCalculator, ReceivedPacket) {
+  StatisticsCalculator stats;
+
+  stats.ReceivedPacket();
+  NetEqLifetimeStatistics stats_output = stats.GetLifetimeStatistics();
+  EXPECT_EQ(1u, stats_output.jitter_buffer_packets_received);
+
+  stats.ReceivedPacket();
+  stats_output = stats.GetLifetimeStatistics();
+  EXPECT_EQ(2u, stats_output.jitter_buffer_packets_received);
+}
+
+TEST(StatisticsCalculator, InterruptionCounter) {
+  constexpr int fs_khz = 48;
+  constexpr int fs_hz = fs_khz * 1000;
+  StatisticsCalculator stats;
+  stats.DecodedOutputPlayed();
+  stats.EndExpandEvent(fs_hz);
+  auto lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(0, lts.interruption_count);
+  EXPECT_EQ(0, lts.total_interruption_duration_ms);
+
+  // Add an event that is shorter than 150 ms. Should not be logged.
+  stats.ExpandedVoiceSamples(10 * fs_khz, false);   // 10 ms.
+  stats.ExpandedNoiseSamples(139 * fs_khz, false);  // 139 ms.
+  stats.EndExpandEvent(fs_hz);
+  lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(0, lts.interruption_count);
+
+  // Add an event that is longer than 150 ms. Should be logged.
+  stats.ExpandedVoiceSamples(140 * fs_khz, false);  // 140 ms.
+  stats.ExpandedNoiseSamples(11 * fs_khz, false);   // 11 ms.
+  stats.EndExpandEvent(fs_hz);
+  lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(1, lts.interruption_count);
+  EXPECT_EQ(151, lts.total_interruption_duration_ms);
+
+  // Add one more long event.
+  stats.ExpandedVoiceSamples(100 * fs_khz, false);   // 100 ms.
+  stats.ExpandedNoiseSamples(5000 * fs_khz, false);  // 5000 ms.
+  stats.EndExpandEvent(fs_hz);
+  lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(2, lts.interruption_count);
+  EXPECT_EQ(5100 + 151, lts.total_interruption_duration_ms);
+}
+
+TEST(StatisticsCalculator, InterruptionCounterDoNotLogBeforeDecoding) {
+  constexpr int fs_khz = 48;
+  constexpr int fs_hz = fs_khz * 1000;
+  StatisticsCalculator stats;
+
+  // Add an event that is longer than 150 ms. Should normally be logged, but we
+  // have not called DecodedOutputPlayed() yet, so it shouldn't this time.
+  stats.ExpandedVoiceSamples(151 * fs_khz, false);  // 151 ms.
+  stats.EndExpandEvent(fs_hz);
+  auto lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(0, lts.interruption_count);
+
+  // Call DecodedOutputPlayed(). Logging should happen after this.
+  stats.DecodedOutputPlayed();
+
+  // Add one more long event.
+  stats.ExpandedVoiceSamples(151 * fs_khz, false);  // 151 ms.
+  stats.EndExpandEvent(fs_hz);
+  lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(1, lts.interruption_count);
+}
+
+TEST(StatisticsCalculator, DiscardedPackets) {
+  StatisticsCalculator statistics_calculator;
+  EXPECT_EQ(0u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+
+  statistics_calculator.PacketsDiscarded(1);
+  EXPECT_EQ(1u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+
+  statistics_calculator.PacketsDiscarded(10);
+  EXPECT_EQ(11u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+
+  // Calling `SecondaryPacketsDiscarded` does not modify `packets_discarded`.
+  statistics_calculator.SecondaryPacketsDiscarded(1);
+  EXPECT_EQ(11u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+
+  // Calling `FlushedPacketBuffer` does not modify `packets_discarded`.
+  statistics_calculator.FlushedPacketBuffer();
+  EXPECT_EQ(11u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.cc
new file mode 100644
index 0000000000..7d7cac7157
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/sync_buffer.h"
+
+#include <algorithm>  // Access to min.
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+size_t SyncBuffer::FutureLength() const {
+  return Size() - next_index_;
+}
+
+void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
+  size_t samples_added = append_this.Size();
+  AudioMultiVector::PushBack(append_this);
+  AudioMultiVector::PopFront(samples_added);
+  if (samples_added <= next_index_) {
+    next_index_ -= samples_added;
+  } else {
+    // This means that we are pushing out future data that was never used.
+    //    RTC_DCHECK_NOTREACHED();
+    // TODO(hlundin): This assert must be disabled to support 60 ms frames.
+    // This should not happen even for 60 ms frames, but it does. Investigate
+    // why.
+    next_index_ = 0;
+  }
+  dtmf_index_ -= std::min(dtmf_index_, samples_added);
+}
+
+void SyncBuffer::PushBackInterleaved(const rtc::BufferT<int16_t>& append_this) {
+  const size_t size_before_adding = Size();
+  AudioMultiVector::PushBackInterleaved(append_this);
+  const size_t samples_added_per_channel = Size() - size_before_adding;
+  RTC_DCHECK_EQ(samples_added_per_channel * Channels(), append_this.size());
+  AudioMultiVector::PopFront(samples_added_per_channel);
+  next_index_ -= std::min(next_index_, samples_added_per_channel);
+  dtmf_index_ -= std::min(dtmf_index_, samples_added_per_channel);
+}
+
+void SyncBuffer::PushFrontZeros(size_t length) {
+  InsertZerosAtIndex(length, 0);
+}
+
+void SyncBuffer::InsertZerosAtIndex(size_t length, size_t position) {
+  position = std::min(position, Size());
+  length = std::min(length, Size() - position);
+  AudioMultiVector::PopBack(length);
+  for (size_t channel = 0; channel < Channels(); ++channel) {
+    channels_[channel]->InsertZerosAt(length, position);
+  }
+  if (next_index_ >= position) {
+    // We are moving the `next_index_` sample.
+    set_next_index(next_index_ + length);  // Overflow handled by subfunction.
+  }
+  if (dtmf_index_ > 0 && dtmf_index_ >= position) {
+    // We are moving the `dtmf_index_` sample.
+    set_dtmf_index(dtmf_index_ + length);  // Overflow handled by subfunction.
+  }
+}
+
+void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
+                                size_t length,
+                                size_t position) {
+  position = std::min(position, Size());  // Cap `position` in the valid range.
+  length = std::min(length, Size() - position);
+  AudioMultiVector::OverwriteAt(insert_this, length, position);
+}
+
+void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
+                                size_t position) {
+  ReplaceAtIndex(insert_this, insert_this.Size(), position);
+}
+
+void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
+                                         AudioFrame* output) {
+  RTC_DCHECK(output);
+  const size_t samples_to_read = std::min(FutureLength(), requested_len);
+  output->ResetWithoutMuting();
+  const size_t tot_samples_read = ReadInterleavedFromIndex(
+      next_index_, samples_to_read, output->mutable_data());
+  const size_t samples_read_per_channel = tot_samples_read / Channels();
+  next_index_ += samples_read_per_channel;
+  output->num_channels_ = Channels();
+  output->samples_per_channel_ = samples_read_per_channel;
+}
+
+void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) {
+  end_timestamp_ += increment;
+}
+
+void SyncBuffer::Flush() {
+  Zeros(Size());
+  next_index_ = Size();
+  end_timestamp_ = 0;
+  dtmf_index_ = 0;
+}
+
+void SyncBuffer::set_next_index(size_t value) {
+  // Cannot set `next_index_` larger than the size of the buffer.
+  next_index_ = std::min(value, Size());
+}
+
+void SyncBuffer::set_dtmf_index(size_t value) {
+  // Cannot set `dtmf_index_` larger than the size of the buffer.
+  dtmf_index_ = std::min(value, Size());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.h
new file mode 100644
index 0000000000..cf56c432e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.h
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class SyncBuffer : public AudioMultiVector {
+ public:
+  SyncBuffer(size_t channels, size_t length)
+      : AudioMultiVector(channels, length),
+        next_index_(length),
+        end_timestamp_(0),
+        dtmf_index_(0) {}
+
+  SyncBuffer(const SyncBuffer&) = delete;
+  SyncBuffer& operator=(const SyncBuffer&) = delete;
+
+  // Returns the number of samples yet to play out from the buffer.
+  size_t FutureLength() const;
+
+  // Adds the contents of `append_this` to the back of the SyncBuffer. Removes
+  // the same number of samples from the beginning of the SyncBuffer, to
+  // maintain a constant buffer size. The `next_index_` is updated to reflect
+  // the move of the beginning of "future" data.
+  void PushBack(const AudioMultiVector& append_this) override;
+
+  // Like PushBack, but reads the samples channel-interleaved from the input.
+  void PushBackInterleaved(const rtc::BufferT<int16_t>& append_this);
+
+  // Adds `length` zeros to the beginning of each channel. Removes
+  // the same number of samples from the end of the SyncBuffer, to
+  // maintain a constant buffer size. The `next_index_` is updated to reflect
+  // the move of the beginning of "future" data.
+  // Note that this operation may delete future samples that are waiting to
+  // be played.
+  void PushFrontZeros(size_t length);
+
+  // Inserts `length` zeros into each channel at index `position`. The size of
+  // the SyncBuffer is kept constant, which means that the last `length`
+  // elements in each channel will be purged.
+  virtual void InsertZerosAtIndex(size_t length, size_t position);
+
+  // Overwrites each channel in this SyncBuffer with values taken from
+  // `insert_this`. The values are taken from the beginning of `insert_this` and
+  // are inserted starting at `position`. `length` values are written into each
+  // channel. The size of the SyncBuffer is kept constant. That is, if `length`
+  // and `position` are selected such that the new data would extend beyond the
+  // end of the current SyncBuffer, the buffer is not extended.
+  // The `next_index_` is not updated.
+  virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
+                              size_t length,
+                              size_t position);
+
+  // Same as the above method, but where all of `insert_this` is written (with
+  // the same constraints as above, that the SyncBuffer is not extended).
+  virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
+                              size_t position);
+
+  // Reads `requested_len` samples from each channel and writes them interleaved
+  // into `output`. The `next_index_` is updated to point to the sample to read
+  // next time. The AudioFrame `output` is first reset, and the `data_`,
+  // `num_channels_`, and `samples_per_channel_` fields are updated.
+  void GetNextAudioInterleaved(size_t requested_len, AudioFrame* output);
+
+  // Adds `increment` to `end_timestamp_`.
+  void IncreaseEndTimestamp(uint32_t increment);
+
+  // Flushes the buffer. The buffer will contain only zeros after the flush, and
+  // `next_index_` will point to the end, like when the buffer was first
+  // created.
+  void Flush();
+
+  const AudioVector& Channel(size_t n) const { return *channels_[n]; }
+  AudioVector& Channel(size_t n) { return *channels_[n]; }
+
+  // Accessors and mutators.
+  size_t next_index() const { return next_index_; }
+  void set_next_index(size_t value);
+  uint32_t end_timestamp() const { return end_timestamp_; }
+  void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
+  size_t dtmf_index() const { return dtmf_index_; }
+  void set_dtmf_index(size_t value);
+
+ private:
+  size_t next_index_;
+  uint32_t end_timestamp_;  // The timestamp of the last sample in the buffer.
+  size_t dtmf_index_;       // Index to the first non-DTMF sample in the buffer.
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc
new file mode 100644
index 0000000000..bdcd92446b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc
@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/sync_buffer.h"
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(SyncBuffer, CreateAndDestroy) {
+  // Create a SyncBuffer with two channels and 10 samples each.
+  static const size_t kLen = 10;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  EXPECT_EQ(kChannels, sync_buffer.Channels());
+  EXPECT_EQ(kLen, sync_buffer.Size());
+  // When the buffer is empty, the next index to play out is at the end.
+  EXPECT_EQ(kLen, sync_buffer.next_index());
+  // Verify that all elements are zero.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kLen; ++i) {
+      EXPECT_EQ(0, sync_buffer[channel][i]);
+    }
+  }
+}
+
+TEST(SyncBuffer, SetNextIndex) {
+  // Create a SyncBuffer with two channels and 100 samples each.
+  static const size_t kLen = 100;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  sync_buffer.set_next_index(0);
+  EXPECT_EQ(0u, sync_buffer.next_index());
+  sync_buffer.set_next_index(kLen / 2);
+  EXPECT_EQ(kLen / 2, sync_buffer.next_index());
+  sync_buffer.set_next_index(kLen);
+  EXPECT_EQ(kLen, sync_buffer.next_index());
+  // Try to set larger than the buffer size; should cap at buffer size.
+  sync_buffer.set_next_index(kLen + 1);
+  EXPECT_EQ(kLen, sync_buffer.next_index());
+}
+
+TEST(SyncBuffer, PushBackAndFlush) {
+  // Create a SyncBuffer with two channels and 100 samples each.
+  static const size_t kLen = 100;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  static const size_t kNewLen = 10;
+  AudioMultiVector new_data(kChannels, kNewLen);
+  // Populate `new_data`.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen; ++i) {
+      new_data[channel][i] = rtc::checked_cast<int16_t>(i);
+    }
+  }
+  // Push back `new_data` into `sync_buffer`. This operation should pop out
+  // data from the front of `sync_buffer`, so that the size of the buffer
+  // remains the same. The `next_index_` should also move with the same length.
+  sync_buffer.PushBack(new_data);
+  ASSERT_EQ(kLen, sync_buffer.Size());
+  // Verify that `next_index_` moved accordingly.
+  EXPECT_EQ(kLen - kNewLen, sync_buffer.next_index());
+  // Verify the new contents.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen; ++i) {
+      EXPECT_EQ(new_data[channel][i],
+                sync_buffer[channel][sync_buffer.next_index() + i]);
+    }
+  }
+
+  // Now flush the buffer, and verify that it is all zeros, and that next_index
+  // points to the end.
+  sync_buffer.Flush();
+  ASSERT_EQ(kLen, sync_buffer.Size());
+  EXPECT_EQ(kLen, sync_buffer.next_index());
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kLen; ++i) {
+      EXPECT_EQ(0, sync_buffer[channel][i]);
+    }
+  }
+}
+
+TEST(SyncBuffer, PushFrontZeros) {
+  // Create a SyncBuffer with two channels and 100 samples each.
+  static const size_t kLen = 100;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  static const size_t kNewLen = 10;
+  AudioMultiVector new_data(kChannels, kNewLen);
+  // Populate `new_data`.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen; ++i) {
+      new_data[channel][i] = rtc::checked_cast<int16_t>(1000 + i);
+    }
+  }
+  sync_buffer.PushBack(new_data);
+  EXPECT_EQ(kLen, sync_buffer.Size());
+
+  // Push `kNewLen` - 1 zeros into each channel in the front of the SyncBuffer.
+  sync_buffer.PushFrontZeros(kNewLen - 1);
+  EXPECT_EQ(kLen, sync_buffer.Size());  // Size should remain the same.
+  // Verify that `next_index_` moved accordingly. Should be at the end - 1.
+  EXPECT_EQ(kLen - 1, sync_buffer.next_index());
+  // Verify the zeros.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen - 1; ++i) {
+      EXPECT_EQ(0, sync_buffer[channel][i]);
+    }
+  }
+  // Verify that the correct data is at the end of the SyncBuffer.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    EXPECT_EQ(1000, sync_buffer[channel][sync_buffer.next_index()]);
+  }
+}
+
+TEST(SyncBuffer, GetNextAudioInterleaved) {
+  // Create a SyncBuffer with two channels and 100 samples each.
+  static const size_t kLen = 100;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  static const size_t kNewLen = 10;
+  AudioMultiVector new_data(kChannels, kNewLen);
+  // Populate `new_data`.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen; ++i) {
+      new_data[channel][i] = rtc::checked_cast<int16_t>(i);
+    }
+  }
+  // Push back `new_data` into `sync_buffer`. This operation should pop out
+  // data from the front of `sync_buffer`, so that the size of the buffer
+  // remains the same. The `next_index_` should also move with the same length.
+  sync_buffer.PushBack(new_data);
+
+  // Read to interleaved output. Read in two batches, where each read operation
+  // should automatically update the `net_index_` in the SyncBuffer.
+  // Note that `samples_read` is the number of samples read from each channel.
+  // That is, the number of samples written to `output` is
+  // `samples_read` * `kChannels`.
+  AudioFrame output1;
+  sync_buffer.GetNextAudioInterleaved(kNewLen / 2, &output1);
+  EXPECT_EQ(kChannels, output1.num_channels_);
+  EXPECT_EQ(kNewLen / 2, output1.samples_per_channel_);
+
+  AudioFrame output2;
+  sync_buffer.GetNextAudioInterleaved(kNewLen / 2, &output2);
+  EXPECT_EQ(kChannels, output2.num_channels_);
+  EXPECT_EQ(kNewLen / 2, output2.samples_per_channel_);
+
+  // Verify the data.
+  const int16_t* output_ptr = output1.data();
+  for (size_t i = 0; i < kNewLen / 2; ++i) {
+    for (size_t channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(new_data[channel][i], *output_ptr);
+      ++output_ptr;
+    }
+  }
+  output_ptr = output2.data();
+  for (size_t i = kNewLen / 2; i < kNewLen; ++i) {
+    for (size_t channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(new_data[channel][i], *output_ptr);
+      ++output_ptr;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
new file mode 100644
index 0000000000..031d8a39ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
@@ -0,0 +1,201 @@
+%
+%  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+%
+%  Use of this source code is governed by a BSD-style license
+%  that can be found in the LICENSE file in the root of the source
+%  tree. An additional intellectual property rights grant can be found
+%  in the file PATENTS.  All contributing project authors may
+%  be found in the AUTHORS file in the root of the source tree.
+%
+
+function outStruct = parse_delay_file(file)
+
+fid = fopen(file, 'rb');
+if fid == -1
+    error('Cannot open file %s', file);
+end
+
+textline = fgetl(fid);
+if ~strncmp(textline, '#!NetEQ_Delay_Logging', 21)
+    error('Wrong file format');
+end
+
+ver = sscanf(textline, '#!NetEQ_Delay_Logging%d.%d');
+if ~all(ver == [2; 0])
+    error('Wrong version of delay logging function')
+end
+
+
+start_pos = ftell(fid);
+fseek(fid, -12, 'eof');
+textline = fgetl(fid);
+if ~strncmp(textline, 'End of file', 21)
+    error('File ending is not correct. Seems like the simulation ended abnormally.');
+end
+
+fseek(fid,-12-4, 'eof');
+Npackets = fread(fid, 1, 'int32');
+fseek(fid, start_pos, 'bof');
+
+rtpts = zeros(Npackets, 1);
+seqno = zeros(Npackets, 1);
+pt = zeros(Npackets, 1);
+plen = zeros(Npackets, 1);
+recin_t = nan*ones(Npackets, 1);
+decode_t = nan*ones(Npackets, 1);
+playout_delay = zeros(Npackets, 1);
+optbuf = zeros(Npackets, 1);
+
+fs_ix = 1;
+clock = 0;
+ts_ix = 1;
+ended = 0;
+late_packets = 0;
+fs_now = 8000;
+last_decode_k = 0;
+tot_expand = 0;
+tot_accelerate = 0;
+tot_preemptive = 0;
+
+while not(ended)
+    signal = fread(fid, 1, '*int32');
+    
+    switch signal
+        case 3 % NETEQ_DELAY_LOGGING_SIGNAL_CLOCK
+            clock = fread(fid, 1, '*float32');
+            
+            % keep on reading batches of M until the signal is no longer "3"
+            % read int32 + float32 in one go
+            % this is to save execution time
+            temp = [3; 0];
+            M = 120;
+            while all(temp(1,:) == 3)
+                fp = ftell(fid);
+                temp = fread(fid, [2 M], '*int32');
+            end
+            
+            % back up to last clock event
+            fseek(fid, fp - ftell(fid) + ...
+                (find(temp(1,:) ~= 3, 1 ) - 2) * 2 * 4 + 4, 'cof');
+            % read the last clock value
+            clock = fread(fid, 1, '*float32');
+            
+        case 1 % NETEQ_DELAY_LOGGING_SIGNAL_RECIN
+            temp_ts = fread(fid, 1, 'uint32');
+            
+            if late_packets > 0
+                temp_ix = ts_ix - 1;
+                while (temp_ix >= 1) && (rtpts(temp_ix) ~= temp_ts)
+                    % TODO(hlundin): use matlab vector search instead?
+                    temp_ix = temp_ix - 1;
+                end
+                
+                if temp_ix >= 1
+                    % the ts was found in the vector
+                    late_packets = late_packets - 1;
+                else
+                    temp_ix = ts_ix;
+                    ts_ix = ts_ix + 1;
+                end
+            else
+                temp_ix = ts_ix;
+                ts_ix = ts_ix + 1;
+            end
+            
+            rtpts(temp_ix) = temp_ts;
+            seqno(temp_ix) = fread(fid, 1, 'uint16');
+            pt(temp_ix) = fread(fid, 1, 'int32');
+            plen(temp_ix) = fread(fid, 1, 'int16');
+            recin_t(temp_ix) = clock;
+            
+        case 2 % NETEQ_DELAY_LOGGING_SIGNAL_FLUSH
+            % do nothing
+            
+        case 4 % NETEQ_DELAY_LOGGING_SIGNAL_EOF
+            ended = 1;
+            
+        case 5 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE
+            last_decode_ts = fread(fid, 1, 'uint32');
+            temp_delay = fread(fid, 1, 'uint16');
+            
+            k = find(rtpts(1:(ts_ix - 1))==last_decode_ts,1,'last');
+            if ~isempty(k)
+                decode_t(k) = clock;
+                playout_delay(k) = temp_delay + ...
+                    5 *  fs_now / 8000; % add overlap length
+                last_decode_k = k;
+            end
+            
+        case 6 % NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS
+            fsvec(fs_ix) = fread(fid, 1, 'uint16');
+            fschange_ts(fs_ix) = last_decode_ts;
+            fs_now = fsvec(fs_ix);
+            fs_ix = fs_ix + 1;
+            
+        case 7 % NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO
+            playout_delay(last_decode_k) = playout_delay(last_decode_k) ...
+                + fread(fid, 1, 'int32');
+            
+        case 8 % NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO
+            temp = fread(fid, 1, 'int32');
+            if last_decode_k ~= 0
+                tot_expand = tot_expand + temp / (fs_now / 1000);
+            end                
+            
+        case 9 % NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO
+            temp = fread(fid, 1, 'int32');
+            if last_decode_k ~= 0
+                tot_accelerate = tot_accelerate + temp / (fs_now / 1000);
+            end                
+
+        case 10 % NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO
+            temp = fread(fid, 1, 'int32');
+            if last_decode_k ~= 0
+                tot_preemptive = tot_preemptive + temp / (fs_now / 1000);
+            end                
+            
+        case 11 % NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF
+            optbuf(last_decode_k) = fread(fid, 1, 'int32');
+            
+        case 12 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC
+            last_decode_ts = fread(fid, 1, 'uint32');
+            k = ts_ix - 1;
+            
+            while (k >= 1) && (rtpts(k) ~= last_decode_ts)
+                % TODO(hlundin): use matlab vector search instead?
+                k = k - 1;
+            end
+            
+            if k < 1
+                % packet not received yet
+                k = ts_ix;
+                rtpts(ts_ix) = last_decode_ts;
+                late_packets = late_packets + 1;
+            end
+            
+            decode_t(k) = clock;
+            playout_delay(k) = fread(fid, 1, 'uint16') + ...
+                5 *  fs_now / 8000; % add overlap length
+            last_decode_k = k;
+             
+    end
+    
+end
+
+
+fclose(fid);
+
+outStruct = struct(...
+    'ts', rtpts, ...
+    'sn', seqno, ...
+    'pt', pt,...
+    'plen', plen,...
+    'arrival', recin_t,...
+    'decode', decode_t,...
+    'fs', fsvec(:),...
+    'fschange_ts', fschange_ts(:),...
+    'playout_delay', playout_delay,...
+    'tot_expand', tot_expand,...
+    'tot_accelerate', tot_accelerate,...
+    'tot_preemptive', tot_preemptive,...
+    'optbuf', optbuf);
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
new file mode 100644
index 0000000000..86d533fbeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
@@ -0,0 +1,197 @@
+%
+%  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+%
+%  Use of this source code is governed by a BSD-style license
+%  that can be found in the LICENSE file in the root of the source
+%  tree. An additional intellectual property rights grant can be found
+%  in the file PATENTS.  All contributing project authors may
+%  be found in the AUTHORS file in the root of the source tree.
+%
+
+function [delay_struct, delayvalues] = plot_neteq_delay(delayfile, varargin)
+
+% InfoStruct = plot_neteq_delay(delayfile)
+% InfoStruct = plot_neteq_delay(delayfile, 'skipdelay', skip_seconds)
+%
+% Henrik Lundin, 2006-11-17
+% Henrik Lundin, 2011-05-17
+%
+
+try
+    s = parse_delay_file(delayfile);
+catch
+    error(lasterr);
+end
+
+delayskip=0;
+noplot=0;
+arg_ptr=1;
+delaypoints=[];
+
+s.sn=unwrap_seqno(s.sn);
+
+while arg_ptr+1 <= nargin
+    switch lower(varargin{arg_ptr})
+    case {'skipdelay', 'delayskip'}
+        % skip a number of seconds in the beginning when calculating delays
+        delayskip = varargin{arg_ptr+1};
+        arg_ptr = arg_ptr + 2;
+    case 'noplot'
+        noplot=1;
+        arg_ptr = arg_ptr + 1;
+    case {'get_delay', 'getdelay'}
+        % return a vector of delay values for the points in the given vector
+        delaypoints = varargin{arg_ptr+1};
+        arg_ptr = arg_ptr + 2;
+    otherwise
+        warning('Unknown switch %s\n', varargin{arg_ptr});
+        arg_ptr = arg_ptr + 1;
+    end
+end
+
+% find lost frames that were covered by one-descriptor decoding
+one_desc_ix=find(isnan(s.arrival));
+for k=1:length(one_desc_ix)
+    ix=find(s.ts==max(s.ts(s.ts(one_desc_ix(k))>s.ts)));
+    s.sn(one_desc_ix(k))=s.sn(ix)+1;
+    s.pt(one_desc_ix(k))=s.pt(ix);
+    s.arrival(one_desc_ix(k))=s.arrival(ix)+s.decode(one_desc_ix(k))-s.decode(ix);
+end
+
+% remove duplicate received frames that were never decoded (RED codec)
+if length(unique(s.ts(isfinite(s.ts)))) < length(s.ts(isfinite(s.ts)))
+    ix=find(isfinite(s.decode));
+    s.sn=s.sn(ix);
+    s.ts=s.ts(ix);
+    s.arrival=s.arrival(ix);
+    s.playout_delay=s.playout_delay(ix);
+    s.pt=s.pt(ix);
+    s.optbuf=s.optbuf(ix);
+    plen=plen(ix);
+    s.decode=s.decode(ix);
+end
+
+% find non-unique sequence numbers
+[~,un_ix]=unique(s.sn);
+nonun_ix=setdiff(1:length(s.sn),un_ix);
+if ~isempty(nonun_ix)
+    warning('RTP sequence numbers are in error');
+end
+            
+% sort vectors
+[s.sn,sort_ix]=sort(s.sn);
+s.ts=s.ts(sort_ix);
+s.arrival=s.arrival(sort_ix);
+s.decode=s.decode(sort_ix);
+s.playout_delay=s.playout_delay(sort_ix);
+s.pt=s.pt(sort_ix);
+
+send_t=s.ts-s.ts(1);
+if length(s.fs)<1
+    warning('No info about sample rate found in file. Using default 8000.');
+    s.fs(1)=8000;
+    s.fschange_ts(1)=min(s.ts);
+elseif s.fschange_ts(1)>min(s.ts)
+    s.fschange_ts(1)=min(s.ts);
+end
+
+end_ix=length(send_t);
+for k=length(s.fs):-1:1
+    start_ix=find(s.ts==s.fschange_ts(k));
+    send_t(start_ix:end_ix)=send_t(start_ix:end_ix)/s.fs(k)*1000;
+    s.playout_delay(start_ix:end_ix)=s.playout_delay(start_ix:end_ix)/s.fs(k)*1000;
+    s.optbuf(start_ix:end_ix)=s.optbuf(start_ix:end_ix)/s.fs(k)*1000;
+    end_ix=start_ix-1;
+end
+
+tot_time=max(send_t)-min(send_t);
+
+seq_ix=s.sn-min(s.sn)+1;
+send_t=send_t+max(min(s.arrival-send_t),0);
+
+plot_send_t=nan*ones(max(seq_ix),1);
+plot_send_t(seq_ix)=send_t;
+plot_nw_delay=nan*ones(max(seq_ix),1);
+plot_nw_delay(seq_ix)=s.arrival-send_t;
+
+cng_ix=find(s.pt~=13); % find those packets that are not CNG/SID
+    
+if noplot==0
+    h=plot(plot_send_t/1000,plot_nw_delay);
+    set(h,'color',0.75*[1 1 1]);
+    hold on
+    if any(s.optbuf~=0)
+        peak_ix=find(s.optbuf(cng_ix)<0); % peak mode is labeled with negative values
+        no_peak_ix=find(s.optbuf(cng_ix)>0); %setdiff(1:length(cng_ix),peak_ix);
+        h1=plot(send_t(cng_ix(peak_ix))/1000,...
+            s.arrival(cng_ix(peak_ix))+abs(s.optbuf(cng_ix(peak_ix)))-send_t(cng_ix(peak_ix)),...
+            'r.');
+        h2=plot(send_t(cng_ix(no_peak_ix))/1000,...
+            s.arrival(cng_ix(no_peak_ix))+abs(s.optbuf(cng_ix(no_peak_ix)))-send_t(cng_ix(no_peak_ix)),...
+            'g.');
+        set([h1, h2],'markersize',1)
+    end
+    %h=plot(send_t(seq_ix)/1000,s.decode+s.playout_delay-send_t(seq_ix));
+    h=plot(send_t(cng_ix)/1000,s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix));
+    set(h,'linew',1.5);
+    hold off
+    ax1=axis;
+    axis tight
+    ax2=axis;
+    axis([ax2(1:3) ax1(4)])
+end
+
+
+% calculate delays and other parameters
+
+delayskip_ix = find(send_t-send_t(1)>=delayskip*1000, 1 );
+
+use_ix = intersect(cng_ix,... % use those that are not CNG/SID frames...
+    intersect(find(isfinite(s.decode)),... % ... that did arrive ...
+    (delayskip_ix:length(s.decode))')); % ... and are sent after delayskip seconds
+
+mean_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-send_t(use_ix));
+neteq_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-s.arrival(use_ix));
+
+Npack=max(s.sn(delayskip_ix:end))-min(s.sn(delayskip_ix:end))+1;
+nw_lossrate=(Npack-length(s.sn(delayskip_ix:end)))/Npack;
+neteq_lossrate=(length(s.sn(delayskip_ix:end))-length(use_ix))/Npack;
+
+delay_struct=struct('mean_delay',mean_delay,'neteq_delay',neteq_delay,...
+    'nw_lossrate',nw_lossrate,'neteq_lossrate',neteq_lossrate,...
+    'tot_expand',round(s.tot_expand),'tot_accelerate',round(s.tot_accelerate),...
+    'tot_preemptive',round(s.tot_preemptive),'tot_time',tot_time,...
+    'filename',delayfile,'units','ms','fs',unique(s.fs));
+    
+if not(isempty(delaypoints))
+    delayvalues=interp1(send_t(cng_ix),...
+        s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix),...
+        delaypoints,'nearest',NaN);
+else
+    delayvalues=[];
+end
+
+
+
+% SUBFUNCTIONS %
+
+function y=unwrap_seqno(x)
+
+jumps=find(abs((diff(x)-1))>65000);
+
+while ~isempty(jumps)
+    n=jumps(1);
+    if x(n+1)-x(n) < 0
+        % negative jump
+        x(n+1:end)=x(n+1:end)+65536;
+    else
+        % positive jump
+        x(n+1:end)=x(n+1:end)-65536;
+    end
+    
+    jumps=find(abs((diff(x(n+1:end))-1))>65000);
+end
+
+y=x;
+
+return;
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc
new file mode 100644
index 0000000000..e6c1809fb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -0,0 +1,423 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/test/neteq_decoding_test.h"
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/test/result_sink.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/testsupport/file_utils.h"
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+namespace {
+
+void LoadDecoders(webrtc::NetEq* neteq) {
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(0, SdpAudioFormat("pcmu", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(8, SdpAudioFormat("pcma", 8000, 1)));
+#ifdef WEBRTC_CODEC_ILBC
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(102, SdpAudioFormat("ilbc", 8000, 1)));
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(103, SdpAudioFormat("isac", 16000, 1)));
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(104, SdpAudioFormat("isac", 32000, 1)));
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(
+                111, SdpAudioFormat("opus", 48000, 2, {{"stereo", "0"}})));
+#endif
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(93, SdpAudioFormat("L16", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(94, SdpAudioFormat("L16", 16000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(95, SdpAudioFormat("L16", 32000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(13, SdpAudioFormat("cn", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(98, SdpAudioFormat("cn", 16000, 1)));
+}
+
+}  // namespace
+
+const int NetEqDecodingTest::kTimeStepMs;
+const size_t NetEqDecodingTest::kBlockSize8kHz;
+const size_t NetEqDecodingTest::kBlockSize16kHz;
+const size_t NetEqDecodingTest::kBlockSize32kHz;
+const int NetEqDecodingTest::kInitSampleRateHz;
+
+NetEqDecodingTest::NetEqDecodingTest()
+    : clock_(0),
+      config_(),
+      output_sample_rate_(kInitSampleRateHz),
+      algorithmic_delay_ms_(0) {
+  config_.sample_rate_hz = kInitSampleRateHz;
+}
+
+void NetEqDecodingTest::SetUp() {
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  neteq_ = DefaultNetEqFactory().CreateNetEq(config_, decoder_factory, &clock_);
+  NetEqNetworkStatistics stat;
+  ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
+  algorithmic_delay_ms_ = stat.current_buffer_size_ms;
+  ASSERT_TRUE(neteq_);
+  LoadDecoders(neteq_.get());
+}
+
+void NetEqDecodingTest::TearDown() {}
+
+void NetEqDecodingTest::OpenInputFile(absl::string_view rtp_file) {
+  rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
+}
+
+void NetEqDecodingTest::Process() {
+  // Check if time to receive.
+  while (packet_ && clock_.TimeInMilliseconds() >= packet_->time_ms()) {
+    if (packet_->payload_length_bytes() > 0) {
+#ifndef WEBRTC_CODEC_ISAC
+      // Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
+      if (packet_->header().payloadType != 104)
+#endif
+        ASSERT_EQ(
+            0, neteq_->InsertPacket(
+                   packet_->header(),
+                   rtc::ArrayView<const uint8_t>(
+                       packet_->payload(), packet_->payload_length_bytes())));
+    }
+    // Get next packet.
+    packet_ = rtp_source_->NextPacket();
+  }
+
+  // Get audio from NetEq.
+  bool muted;
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
+  ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize16kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize32kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize48kHz));
+  output_sample_rate_ = out_frame_.sample_rate_hz_;
+  EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
+
+  // Increase time.
+  clock_.AdvanceTimeMilliseconds(kTimeStepMs);
+}
+
+void NetEqDecodingTest::DecodeAndCompare(
+    absl::string_view rtp_file,
+    absl::string_view output_checksum,
+    absl::string_view network_stats_checksum,
+    bool gen_ref) {
+  OpenInputFile(rtp_file);
+
+  std::string ref_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_universal_ref.pcm" : "";
+  ResultSink output(ref_out_file);
+
+  std::string stat_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : "";
+  ResultSink network_stats(stat_out_file);
+
+  packet_ = rtp_source_->NextPacket();
+  int i = 0;
+  uint64_t last_concealed_samples = 0;
+  uint64_t last_total_samples_received = 0;
+  while (packet_) {
+    rtc::StringBuilder ss;
+    ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
+    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+    ASSERT_NO_FATAL_FAILURE(Process());
+    ASSERT_NO_FATAL_FAILURE(
+        output.AddResult(out_frame_.data(), out_frame_.samples_per_channel_));
+
+    // Query the network statistics API once per second
+    if (clock_.TimeInMilliseconds() % 1000 == 0) {
+      // Process NetworkStatistics.
+      NetEqNetworkStatistics current_network_stats;
+      ASSERT_EQ(0, neteq_->NetworkStatistics(&current_network_stats));
+      ASSERT_NO_FATAL_FAILURE(network_stats.AddResult(current_network_stats));
+
+      // Verify that liftime stats and network stats report similar loss
+      // concealment rates.
+      auto lifetime_stats = neteq_->GetLifetimeStatistics();
+      const uint64_t delta_concealed_samples =
+          lifetime_stats.concealed_samples - last_concealed_samples;
+      last_concealed_samples = lifetime_stats.concealed_samples;
+      const uint64_t delta_total_samples_received =
+          lifetime_stats.total_samples_received - last_total_samples_received;
+      last_total_samples_received = lifetime_stats.total_samples_received;
+      // The tolerance is 1% but expressed in Q14.
+      EXPECT_NEAR(
+          (delta_concealed_samples << 14) / delta_total_samples_received,
+          current_network_stats.expand_rate, (2 << 14) / 100.0);
+    }
+  }
+
+  SCOPED_TRACE("Check output audio.");
+  output.VerifyChecksum(output_checksum);
+  SCOPED_TRACE("Check network stats.");
+  network_stats.VerifyChecksum(network_stats_checksum);
+}
+
+void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
+                                        int timestamp,
+                                        RTPHeader* rtp_info) {
+  rtp_info->sequenceNumber = frame_index;
+  rtp_info->timestamp = timestamp;
+  rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info->payloadType = 94;  // PCM16b WB codec.
+  rtp_info->markerBit = false;
+}
+
+void NetEqDecodingTest::PopulateCng(int frame_index,
+                                    int timestamp,
+                                    RTPHeader* rtp_info,
+                                    uint8_t* payload,
+                                    size_t* payload_len) {
+  rtp_info->sequenceNumber = frame_index;
+  rtp_info->timestamp = timestamp;
+  rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info->payloadType = 98;  // WB CNG.
+  rtp_info->markerBit = false;
+  payload[0] = 64;   // Noise level -64 dBov, quite arbitrarily chosen.
+  *payload_len = 1;  // Only noise level, no spectral parameters.
+}
+
+void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
+                                 uint32_t start_timestamp,
+                                 const std::set<uint16_t>& drop_seq_numbers,
+                                 bool expect_seq_no_wrap,
+                                 bool expect_timestamp_wrap) {
+  uint16_t seq_no = start_seq_no;
+  uint32_t timestamp = start_timestamp;
+  const int kBlocksPerFrame = 3;  // Number of 10 ms blocks per frame.
+  const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
+  const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
+  const size_t kPayloadBytes = kSamples * sizeof(int16_t);
+  double next_input_time_ms = 0.0;
+
+  // Insert speech for 2 seconds.
+  const int kSpeechDurationMs = 2000;
+  uint16_t last_seq_no;
+  uint32_t last_timestamp;
+  bool timestamp_wrapped = false;
+  bool seq_no_wrapped = false;
+  for (double t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
+        // This sequence number was not in the set to drop. Insert it.
+        ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      }
+      NetEqNetworkStatistics network_stats;
+      ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+
+      EXPECT_LE(network_stats.preferred_buffer_size_ms, 80);
+      EXPECT_LE(network_stats.current_buffer_size_ms,
+                80 + algorithmic_delay_ms_);
+      last_seq_no = seq_no;
+      last_timestamp = timestamp;
+
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += static_cast<double>(kFrameSizeMs);
+
+      seq_no_wrapped |= seq_no < last_seq_no;
+      timestamp_wrapped |= timestamp < last_timestamp;
+    }
+    // Pull out data once.
+    AudioFrame output;
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+    ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
+    ASSERT_EQ(1u, output.num_channels_);
+
+    // Expect delay (in samples) to be less than 2 packets.
+    absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+    ASSERT_TRUE(playout_timestamp);
+    EXPECT_LE(timestamp - *playout_timestamp,
+              static_cast<uint32_t>(kSamples * 2));
+  }
+  // Make sure we have actually tested wrap-around.
+  ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
+  ASSERT_EQ(expect_timestamp_wrap, timestamp_wrapped);
+}
+
+void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
+                                              double network_freeze_ms,
+                                              bool pull_audio_during_freeze,
+                                              int delay_tolerance_ms,
+                                              int max_time_to_speech_ms) {
+  uint16_t seq_no = 0;
+  uint32_t timestamp = 0;
+  const int kFrameSizeMs = 30;
+  const size_t kSamples = kFrameSizeMs * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  double next_input_time_ms = 0.0;
+  double t_ms;
+  bool muted;
+
+  // Insert speech for 5 seconds.
+  const int kSpeechDurationMs = 5000;
+  for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+  absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  int32_t delay_before = timestamp - *playout_timestamp;
+
+  // Insert CNG for 1 minute (= 60000 ms).
+  const int kCngPeriodMs = 100;
+  const int kCngPeriodSamples = kCngPeriodMs * 16;  // Period in 16 kHz samples.
+  const int kCngDurationMs = 60000;
+  for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one CNG frame each 100 ms.
+      uint8_t payload[kPayloadBytes];
+      size_t payload_len;
+      RTPHeader rtp_info;
+      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, payload_len)));
+      ++seq_no;
+      timestamp += kCngPeriodSamples;
+      next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+
+  if (network_freeze_ms > 0) {
+    // First keep pulling audio for `network_freeze_ms` without inserting
+    // any data, then insert CNG data corresponding to `network_freeze_ms`
+    // without pulling any output audio.
+    const double loop_end_time = t_ms + network_freeze_ms;
+    for (; t_ms < loop_end_time; t_ms += 10) {
+      // Pull out data once.
+      ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+      ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+      EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+    }
+    bool pull_once = pull_audio_during_freeze;
+    // If `pull_once` is true, GetAudio will be called once half-way through
+    // the network recovery period.
+    double pull_time_ms = (t_ms + next_input_time_ms) / 2;
+    while (next_input_time_ms <= t_ms) {
+      if (pull_once && next_input_time_ms >= pull_time_ms) {
+        pull_once = false;
+        // Pull out data once.
+        ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+        ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+        EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+        t_ms += 10;
+      }
+      // Insert one CNG frame each 100 ms.
+      uint8_t payload[kPayloadBytes];
+      size_t payload_len;
+      RTPHeader rtp_info;
+      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, payload_len)));
+      ++seq_no;
+      timestamp += kCngPeriodSamples;
+      next_input_time_ms += kCngPeriodMs * drift_factor;
+    }
+  }
+
+  // Insert speech again until output type is speech.
+  double speech_restart_time_ms = t_ms;
+  while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += kFrameSizeMs * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+    // Increase clock.
+    t_ms += 10;
+  }
+
+  // Check that the speech starts again within reasonable time.
+  double time_until_speech_returns_ms = t_ms - speech_restart_time_ms;
+  EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms);
+  playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  int32_t delay_after = timestamp - *playout_timestamp;
+  // Compare delay before and after, and make sure it differs less than 20 ms.
+  EXPECT_LE(delay_after, delay_before + delay_tolerance_ms * 16);
+  EXPECT_GE(delay_after, delay_before - delay_tolerance_ms * 16);
+}
+
+void NetEqDecodingTestTwoInstances::SetUp() {
+  NetEqDecodingTest::SetUp();
+  config2_ = config_;
+}
+
+void NetEqDecodingTestTwoInstances::CreateSecondInstance() {
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  neteq2_ =
+      DefaultNetEqFactory().CreateNetEq(config2_, decoder_factory, &clock_);
+  ASSERT_TRUE(neteq2_);
+  LoadDecoders(neteq2_.get());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h
new file mode 100644
index 0000000000..456c397fdd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
+
+#include <memory>
+#include <set>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/audio_frame.h"
+#include "api/neteq/neteq.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class NetEqDecodingTest : public ::testing::Test {
+ protected:
+  // NetEQ must be polled for data once every 10 ms.
+  // Thus, none of the constants below can be changed.
+  static constexpr int kTimeStepMs = 10;
+  static constexpr size_t kBlockSize8kHz = kTimeStepMs * 8;
+  static constexpr size_t kBlockSize16kHz = kTimeStepMs * 16;
+  static constexpr size_t kBlockSize32kHz = kTimeStepMs * 32;
+  static constexpr size_t kBlockSize48kHz = kTimeStepMs * 48;
+  static constexpr int kInitSampleRateHz = 8000;
+
+  NetEqDecodingTest();
+  virtual void SetUp();
+  virtual void TearDown();
+  void OpenInputFile(absl::string_view rtp_file);
+  void Process();
+
+  void DecodeAndCompare(absl::string_view rtp_file,
+                        absl::string_view output_checksum,
+                        absl::string_view network_stats_checksum,
+                        bool gen_ref);
+
+  static void PopulateRtpInfo(int frame_index,
+                              int timestamp,
+                              RTPHeader* rtp_info);
+  static void PopulateCng(int frame_index,
+                          int timestamp,
+                          RTPHeader* rtp_info,
+                          uint8_t* payload,
+                          size_t* payload_len);
+
+  void WrapTest(uint16_t start_seq_no,
+                uint32_t start_timestamp,
+                const std::set<uint16_t>& drop_seq_numbers,
+                bool expect_seq_no_wrap,
+                bool expect_timestamp_wrap);
+
+  void LongCngWithClockDrift(double drift_factor,
+                             double network_freeze_ms,
+                             bool pull_audio_during_freeze,
+                             int delay_tolerance_ms,
+                             int max_time_to_speech_ms);
+
+  SimulatedClock clock_;
+  std::unique_ptr<NetEq> neteq_;
+  NetEq::Config config_;
+  std::unique_ptr<test::RtpFileSource> rtp_source_;
+  std::unique_ptr<test::Packet> packet_;
+  AudioFrame out_frame_;
+  int output_sample_rate_;
+  int algorithmic_delay_ms_;
+};
+
+class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
+ public:
+  NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
+
+  void SetUp() override;
+
+  void CreateSecondInstance();
+
+ protected:
+  std::unique_ptr<NetEq> neteq2_;
+  NetEq::Config config2_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
new file mode 100644
index 0000000000..1004141f16
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+}  // namespace
+
+class NetEqIlbcQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqIlbcQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("ilbc", 8000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) == 20 ||
+              absl::GetFlag(FLAGS_frame_size_ms) == 30 ||
+              absl::GetFlag(FLAGS_frame_size_ms) == 40 ||
+              absl::GetFlag(FLAGS_frame_size_ms) == 60)
+        << "Invalid frame size, should be 20, 30, 40, or 60 ms.";
+  }
+
+  void SetUp() override {
+    ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
+    AudioEncoderIlbcConfig config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    encoder_.reset(new AudioEncoderIlbcImpl(config, 102));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  std::unique_ptr<AudioEncoderIlbcImpl> encoder_;
+};
+
+TEST_F(NetEqIlbcQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
new file mode 100644
index 0000000000..5a2df24ef6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -0,0 +1,183 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+
+ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps).");
+
+ABSL_FLAG(int,
+          complexity,
+          10,
+          "Complexity: 0 ~ 10 -- defined as in Opus"
+          "specification.");
+
+ABSL_FLAG(int, maxplaybackrate, 48000, "Maximum playback rate (Hz).");
+
+ABSL_FLAG(int, application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
+
+ABSL_FLAG(int, reported_loss_rate, 10, "Reported percentile of packet loss.");
+
+ABSL_FLAG(bool, fec, false, "Enable FEC for encoding (-nofec to disable).");
+
+ABSL_FLAG(bool, dtx, false, "Enable DTX for encoding (-nodtx to disable).");
+
+ABSL_FLAG(int, sub_packets, 1, "Number of sub packets to repacketize.");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+
+static const int kOpusBlockDurationMs = 20;
+static const int kOpusSamplingKhz = 48;
+}  // namespace
+
+class NetEqOpusQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqOpusQualityTest();
+  void SetUp() override;
+  void TearDown() override;
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override;
+
+ private:
+  WebRtcOpusEncInst* opus_encoder_;
+  OpusRepacketizer* repacketizer_;
+  size_t sub_block_size_samples_;
+  int bit_rate_kbps_;
+  bool fec_;
+  bool dtx_;
+  int complexity_;
+  int maxplaybackrate_;
+  int target_loss_rate_;
+  int sub_packets_;
+  int application_;
+};
+
+NetEqOpusQualityTest::NetEqOpusQualityTest()
+    : NetEqQualityTest(kOpusBlockDurationMs * absl::GetFlag(FLAGS_sub_packets),
+                       kOpusSamplingKhz,
+                       kOpusSamplingKhz,
+                       SdpAudioFormat("opus", 48000, 2)),
+      opus_encoder_(NULL),
+      repacketizer_(NULL),
+      sub_block_size_samples_(
+          static_cast<size_t>(kOpusBlockDurationMs * kOpusSamplingKhz)),
+      bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)),
+      fec_(absl::GetFlag(FLAGS_fec)),
+      dtx_(absl::GetFlag(FLAGS_dtx)),
+      complexity_(absl::GetFlag(FLAGS_complexity)),
+      maxplaybackrate_(absl::GetFlag(FLAGS_maxplaybackrate)),
+      target_loss_rate_(absl::GetFlag(FLAGS_reported_loss_rate)),
+      sub_packets_(absl::GetFlag(FLAGS_sub_packets)) {
+  // Flag validation
+  RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 6 &&
+            absl::GetFlag(FLAGS_bit_rate_kbps) <= 510)
+      << "Invalid bit rate, should be between 6 and 510 kbps.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_complexity) >= -1 &&
+            absl::GetFlag(FLAGS_complexity) <= 10)
+      << "Invalid complexity setting, should be between 0 and 10.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_application) == 0 ||
+            absl::GetFlag(FLAGS_application) == 1)
+      << "Invalid application mode, should be 0 or 1.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_reported_loss_rate) >= 0 &&
+            absl::GetFlag(FLAGS_reported_loss_rate) <= 100)
+      << "Invalid packet loss percentile, should be between 0 and 100.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_sub_packets) >= 1 &&
+            absl::GetFlag(FLAGS_sub_packets) <= 3)
+      << "Invalid number of sub packets, should be between 1 and 3.";
+
+  // Redefine decoder type if input is stereo.
+  if (channels_ > 1) {
+    audio_format_ = SdpAudioFormat("opus", 48000, 2,
+                                   SdpAudioFormat::Parameters{{"stereo", "1"}});
+  }
+  application_ = absl::GetFlag(FLAGS_application);
+}
+
+void NetEqOpusQualityTest::SetUp() {
+  // Create encoder memory.
+  WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_, 48000);
+  ASSERT_TRUE(opus_encoder_);
+
+  // Create repacketizer.
+  repacketizer_ = opus_repacketizer_create();
+  ASSERT_TRUE(repacketizer_);
+
+  // Set bitrate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_kbps_ * 1000));
+  if (fec_) {
+    EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+  }
+  if (dtx_) {
+    EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
+  }
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity_));
+  EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, maxplaybackrate_));
+  EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, target_loss_rate_));
+  NetEqQualityTest::SetUp();
+}
+
+void NetEqOpusQualityTest::TearDown() {
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  opus_repacketizer_destroy(repacketizer_);
+  NetEqQualityTest::TearDown();
+}
+
+int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
+                                      size_t block_size_samples,
+                                      rtc::Buffer* payload,
+                                      size_t max_bytes) {
+  EXPECT_EQ(block_size_samples, sub_block_size_samples_ * sub_packets_);
+  int16_t* pointer = in_data;
+  int value;
+  opus_repacketizer_init(repacketizer_);
+  for (int idx = 0; idx < sub_packets_; idx++) {
+    payload->AppendData(max_bytes, [&](rtc::ArrayView<uint8_t> payload) {
+      value = WebRtcOpus_Encode(opus_encoder_, pointer, sub_block_size_samples_,
+                                max_bytes, payload.data());
+
+      Log() << "Encoded a frame with Opus mode "
+            << (value == 0 ? 0 : payload[0] >> 3) << std::endl;
+
+      return (value >= 0) ? static_cast<size_t>(value) : 0;
+    });
+
+    if (OPUS_OK !=
+        opus_repacketizer_cat(repacketizer_, payload->data(), value)) {
+      opus_repacketizer_init(repacketizer_);
+      // If the repacketization fails, we discard this frame.
+      return 0;
+    }
+    pointer += sub_block_size_samples_ * channels_;
+  }
+  value = opus_repacketizer_out(repacketizer_, payload->data(),
+                                static_cast<opus_int32>(max_bytes));
+  EXPECT_GE(value, 0);
+  return value;
+}
+
+TEST_F(NetEqOpusQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
new file mode 100644
index 0000000000..c3e160cb66
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 48;
+static const int kOutputSampleRateKhz = 48;
+}  // namespace
+
+class NetEqPcm16bQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqPcm16bQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("l16", 48000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
+              absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
+              (absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
+        << "Invalid frame size, should be 10, 20, ..., 60 ms.";
+  }
+
+  void SetUp() override {
+    AudioEncoderPcm16B::Config config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    config.sample_rate_hz = 48000;
+    config.num_channels = channels_;
+    encoder_.reset(new AudioEncoderPcm16B(config));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 480;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  std::unique_ptr<AudioEncoderPcm16B> encoder_;
+};
+
+TEST_F(NetEqPcm16bQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
new file mode 100644
index 0000000000..d22170c623
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+}  // namespace
+
+class NetEqPcmuQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqPcmuQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("pcmu", 8000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
+              absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
+              (absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
+        << "Invalid frame size, should be 10, 20, ..., 60 ms.";
+  }
+
+  void SetUp() override {
+    ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
+    AudioEncoderPcmU::Config config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    encoder_.reset(new AudioEncoderPcmU(config));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  std::unique_ptr<AudioEncoderPcmU> encoder_;
+};
+
+TEST_F(NetEqPcmuQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
new file mode 100644
index 0000000000..961f74ab66
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.h"
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::GetGlobalMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+// Runs a test with 10% packet losses and 10% clock drift, to exercise
+// both loss concealment and time-stretching code.
+TEST(NetEqPerformanceTest, 10_Pl_10_Drift) {
+  const int kSimulationTimeMs = 10000000;
+  const int kQuickSimulationTimeMs = 100000;
+  const int kLossPeriod = 10;  // Drop every 10th packet.
+  const double kDriftFactor = 0.1;
+  int64_t runtime = test::NetEqPerformanceTest::Run(
+      field_trial::IsEnabled("WebRTC-QuickPerfTest") ? kQuickSimulationTimeMs
+                                                     : kSimulationTimeMs,
+      kLossPeriod, kDriftFactor);
+  ASSERT_GT(runtime, 0);
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "neteq_performance", "10_pl_10_drift", runtime, Unit::kMilliseconds,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+// Runs a test with neither packet losses nor clock drift, to put
+// emphasis on the "good-weather" code path, which is presumably much
+// more lightweight.
+TEST(NetEqPerformanceTest, 0_Pl_0_Drift) {
+  const int kSimulationTimeMs = 10000000;
+  const int kQuickSimulationTimeMs = 100000;
+  const int kLossPeriod = 0;        // No losses.
+  const double kDriftFactor = 0.0;  // No clock drift.
+  int64_t runtime = test::NetEqPerformanceTest::Run(
+      field_trial::IsEnabled("WebRTC-QuickPerfTest") ? kQuickSimulationTimeMs
+                                                     : kSimulationTimeMs,
+      kLossPeriod, kDriftFactor);
+  ASSERT_GT(runtime, 0);
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "neteq_performance", "0_pl_0_drift", runtime, Unit::kMilliseconds,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
new file mode 100644
index 0000000000..a72b2009eb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <iostream>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "rtc_base/checks.h"
+
+// Define command line flags.
+ABSL_FLAG(int, runtime_ms, 10000, "Simulated runtime in ms.");
+ABSL_FLAG(int, lossrate, 10, "Packet lossrate; drop every N packets.");
+ABSL_FLAG(float, drift, 0.1f, "Clockdrift factor.");
+
+int main(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  std::string program_name = args[0];
+  std::string usage =
+      "Tool for measuring the speed of NetEq.\n"
+      "Usage: " +
+      program_name +
+      " [options]\n\n"
+      "  --runtime_ms=N         runtime in ms; default is 10000 ms\n"
+      "  --lossrate=N           drop every N packets; default is 10\n"
+      "  --drift=F              clockdrift factor between 0.0 and 1.0; "
+      "default is 0.1\n";
+  if (args.size() != 1) {
+    printf("%s", usage.c_str());
+    return 1;
+  }
+  RTC_CHECK_GT(absl::GetFlag(FLAGS_runtime_ms), 0);
+  RTC_CHECK_GE(absl::GetFlag(FLAGS_lossrate), 0);
+  RTC_CHECK(absl::GetFlag(FLAGS_drift) >= 0.0 &&
+            absl::GetFlag(FLAGS_drift) < 1.0);
+
+  int64_t result = webrtc::test::NetEqPerformanceTest::Run(
+      absl::GetFlag(FLAGS_runtime_ms), absl::GetFlag(FLAGS_lossrate),
+      absl::GetFlag(FLAGS_drift));
+  if (result <= 0) {
+    std::cout << "There was an error" << std::endl;
+    return -1;
+  }
+
+  std::cout << "Simulation done" << std::endl;
+  std::cout << "Runtime = " << result << " ms" << std::endl;
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc
new file mode 100644
index 0000000000..f5d50dc859
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/test/result_sink.h"
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/message_digest.h"
+#include "rtc_base/string_encode.h"
+#include "test/gtest.h"
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
+             webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
+  stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
+  stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
+  stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
+  stats->set_expand_rate(stats_raw.expand_rate);
+  stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
+  stats->set_preemptive_rate(stats_raw.preemptive_rate);
+  stats->set_accelerate_rate(stats_raw.accelerate_rate);
+  stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
+  stats->set_secondary_discarded_rate(stats_raw.secondary_discarded_rate);
+  stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
+  stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
+  stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
+  stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
+}
+
+void AddMessage(FILE* file,
+                rtc::MessageDigest* digest,
+                absl::string_view message) {
+  int32_t size = message.length();
+  if (file)
+    ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
+  digest->Update(&size, sizeof(size));
+
+  if (file)
+    ASSERT_EQ(static_cast<size_t>(size),
+              fwrite(message.data(), sizeof(char), size, file));
+  digest->Update(message.data(), sizeof(char) * size);
+}
+
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+
+ResultSink::ResultSink(absl::string_view output_file)
+    : output_fp_(nullptr),
+      digest_(rtc::MessageDigestFactory::Create(rtc::DIGEST_SHA_1)) {
+  if (!output_file.empty()) {
+    output_fp_ = fopen(std::string(output_file).c_str(), "wb");
+    EXPECT_TRUE(output_fp_ != NULL);
+  }
+}
+
+ResultSink::~ResultSink() {
+  if (output_fp_)
+    fclose(output_fp_);
+}
+
+void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  neteq_unittest::NetEqNetworkStatistics stats;
+  Convert(stats_raw, &stats);
+
+  std::string stats_string;
+  ASSERT_TRUE(stats.SerializeToString(&stats_string));
+  AddMessage(output_fp_, digest_.get(), stats_string);
+#else
+  FAIL() << "Writing to reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+}
+
+void ResultSink::VerifyChecksum(absl::string_view checksum) {
+  std::string buffer;
+  buffer.resize(digest_->Size());
+  digest_->Finish(buffer.data(), buffer.size());
+  const std::string result = rtc::hex_encode(buffer);
+  if (checksum.size() == result.size()) {
+    EXPECT_EQ(checksum, result);
+  } else {
+    // Check result is one the '|'-separated checksums.
+    EXPECT_NE(checksum.find(result), absl::string_view::npos)
+        << result << " should be one of these:\n"
+        << checksum;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.h b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.h
new file mode 100644
index 0000000000..c6923d7a7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_
+#define MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_
+
+#include <cstdio>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/neteq/neteq.h"
+#include "rtc_base/message_digest.h"
+
+namespace webrtc {
+
+class ResultSink {
+ public:
+  explicit ResultSink(absl::string_view output_file);
+  ~ResultSink();
+
+  template <typename T>
+  void AddResult(const T* test_results, size_t length);
+
+  void AddResult(const NetEqNetworkStatistics& stats);
+
+  void VerifyChecksum(absl::string_view ref_check_sum);
+
+ private:
+  FILE* output_fp_;
+  std::unique_ptr<rtc::MessageDigest> digest_;
+};
+
+template <typename T>
+void ResultSink::AddResult(const T* test_results, size_t length) {
+  if (output_fp_) {
+    ASSERT_EQ(length, fwrite(test_results, sizeof(T), length, output_fp_));
+  }
+  digest_->Update(test_results, sizeof(T) * length);
+}
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.cc b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.cc
new file mode 100644
index 0000000000..b89be0608d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.cc
@@ -0,0 +1,216 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/time_stretch.h"
+
+#include <algorithm>  // min, max
+#include <memory>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
+                                              size_t input_len,
+                                              bool fast_mode,
+                                              AudioMultiVector* output,
+                                              size_t* length_change_samples) {
+  // Pre-calculate common multiplication with `fs_mult_`.
+  size_t fs_mult_120 =
+      static_cast<size_t>(fs_mult_ * 120);  // Corresponds to 15 ms.
+
+  const int16_t* signal;
+  std::unique_ptr<int16_t[]> signal_array;
+  size_t signal_len;
+  if (num_channels_ == 1) {
+    signal = input;
+    signal_len = input_len;
+  } else {
+    // We want `signal` to be only the first channel of `input`, which is
+    // interleaved. Thus, we take the first sample, skip forward `num_channels`
+    // samples, and continue like that.
+    signal_len = input_len / num_channels_;
+    signal_array.reset(new int16_t[signal_len]);
+    signal = signal_array.get();
+    size_t j = kRefChannel;
+    for (size_t i = 0; i < signal_len; ++i) {
+      signal_array[i] = input[j];
+      j += num_channels_;
+    }
+  }
+
+  // Find maximum absolute value of input signal.
+  max_input_value_ = WebRtcSpl_MaxAbsValueW16(signal, signal_len);
+
+  // Downsample to 4 kHz sample rate and calculate auto-correlation.
+  DspHelper::DownsampleTo4kHz(signal, signal_len, kDownsampledLen,
+                              sample_rate_hz_, true /* compensate delay*/,
+                              downsampled_input_);
+  AutoCorrelation();
+
+  // Find the strongest correlation peak.
+  static const size_t kNumPeaks = 1;
+  size_t peak_index;
+  int16_t peak_value;
+  DspHelper::PeakDetection(auto_correlation_, kCorrelationLen, kNumPeaks,
+                           fs_mult_, &peak_index, &peak_value);
+  // Assert that `peak_index` stays within boundaries.
+  RTC_DCHECK_LE(peak_index, (2 * kCorrelationLen - 1) * fs_mult_);
+
+  // Compensate peak_index for displaced starting position. The displacement
+  // happens in AutoCorrelation(). Here, `kMinLag` is in the down-sampled 4 kHz
+  // domain, while the `peak_index` is in the original sample rate; hence, the
+  // multiplication by fs_mult_ * 2.
+  peak_index += kMinLag * fs_mult_ * 2;
+  // Assert that `peak_index` stays within boundaries.
+  RTC_DCHECK_GE(peak_index, static_cast<size_t>(20 * fs_mult_));
+  RTC_DCHECK_LE(peak_index,
+                20 * fs_mult_ + (2 * kCorrelationLen - 1) * fs_mult_);
+
+  // Calculate scaling to ensure that `peak_index` samples can be square-summed
+  // without overflowing.
+  int scaling = 31 - WebRtcSpl_NormW32(max_input_value_ * max_input_value_) -
+                WebRtcSpl_NormW32(static_cast<int32_t>(peak_index));
+  scaling = std::max(0, scaling);
+
+  // `vec1` starts at 15 ms minus one pitch period.
+  const int16_t* vec1 = &signal[fs_mult_120 - peak_index];
+  // `vec2` start at 15 ms.
+  const int16_t* vec2 = &signal[fs_mult_120];
+  // Calculate energies for `vec1` and `vec2`, assuming they both contain
+  // `peak_index` samples.
+  int32_t vec1_energy =
+      WebRtcSpl_DotProductWithScale(vec1, vec1, peak_index, scaling);
+  int32_t vec2_energy =
+      WebRtcSpl_DotProductWithScale(vec2, vec2, peak_index, scaling);
+
+  // Calculate cross-correlation between `vec1` and `vec2`.
+  int32_t cross_corr =
+      WebRtcSpl_DotProductWithScale(vec1, vec2, peak_index, scaling);
+
+  // Check if the signal seems to be active speech or not (simple VAD).
+  bool active_speech =
+      SpeechDetection(vec1_energy, vec2_energy, peak_index, scaling);
+
+  int16_t best_correlation;
+  if (!active_speech) {
+    SetParametersForPassiveSpeech(signal_len, &best_correlation, &peak_index);
+  } else {
+    // Calculate correlation:
+    // cross_corr / sqrt(vec1_energy * vec2_energy).
+
+    // Start with calculating scale values.
+    int energy1_scale = std::max(0, 16 - WebRtcSpl_NormW32(vec1_energy));
+    int energy2_scale = std::max(0, 16 - WebRtcSpl_NormW32(vec2_energy));
+
+    // Make sure total scaling is even (to simplify scale factor after sqrt).
+    if ((energy1_scale + energy2_scale) & 1) {
+      // The sum is odd.
+      energy1_scale += 1;
+    }
+
+    // Scale energies to int16_t.
+    int16_t vec1_energy_int16 =
+        static_cast<int16_t>(vec1_energy >> energy1_scale);
+    int16_t vec2_energy_int16 =
+        static_cast<int16_t>(vec2_energy >> energy2_scale);
+
+    // Calculate square-root of energy product.
+    int16_t sqrt_energy_prod =
+        WebRtcSpl_SqrtFloor(vec1_energy_int16 * vec2_energy_int16);
+
+    // Calculate cross_corr / sqrt(en1*en2) in Q14.
+    int temp_scale = 14 - (energy1_scale + energy2_scale) / 2;
+    cross_corr = WEBRTC_SPL_SHIFT_W32(cross_corr, temp_scale);
+    cross_corr = std::max(0, cross_corr);  // Don't use if negative.
+    best_correlation = WebRtcSpl_DivW32W16(cross_corr, sqrt_energy_prod);
+    // Make sure `best_correlation` is no larger than 1 in Q14.
+    best_correlation = std::min(static_cast<int16_t>(16384), best_correlation);
+  }
+
+  // Check accelerate criteria and stretch the signal.
+  ReturnCodes return_value =
+      CheckCriteriaAndStretch(input, input_len, peak_index, best_correlation,
+                              active_speech, fast_mode, output);
+  switch (return_value) {
+    case kSuccess:
+      *length_change_samples = peak_index;
+      break;
+    case kSuccessLowEnergy:
+      *length_change_samples = peak_index;
+      break;
+    case kNoStretch:
+    case kError:
+      *length_change_samples = 0;
+      break;
+  }
+  return return_value;
+}
+
+void TimeStretch::AutoCorrelation() {
+  // Calculate correlation from lag kMinLag to lag kMaxLag in 4 kHz domain.
+  int32_t auto_corr[kCorrelationLen];
+  CrossCorrelationWithAutoShift(
+      &downsampled_input_[kMaxLag], &downsampled_input_[kMaxLag - kMinLag],
+      kCorrelationLen, kMaxLag - kMinLag, -1, auto_corr);
+
+  // Normalize correlation to 14 bits and write to `auto_correlation_`.
+  int32_t max_corr = WebRtcSpl_MaxAbsValueW32(auto_corr, kCorrelationLen);
+  int scaling = std::max(0, 17 - WebRtcSpl_NormW32(max_corr));
+  WebRtcSpl_VectorBitShiftW32ToW16(auto_correlation_, kCorrelationLen,
+                                   auto_corr, scaling);
+}
+
+bool TimeStretch::SpeechDetection(int32_t vec1_energy,
+                                  int32_t vec2_energy,
+                                  size_t peak_index,
+                                  int scaling) const {
+  // Check if the signal seems to be active speech or not (simple VAD).
+  // If (vec1_energy + vec2_energy) / (2 * peak_index) <=
+  // 8 * background_noise_energy, then we say that the signal contains no
+  // active speech.
+  // Rewrite the inequality as:
+  // (vec1_energy + vec2_energy) / 16 <= peak_index * background_noise_energy.
+  // The two sides of the inequality will be denoted `left_side` and
+  // `right_side`.
+  int32_t left_side = rtc::saturated_cast<int32_t>(
+      (static_cast<int64_t>(vec1_energy) + vec2_energy) / 16);
+  int32_t right_side;
+  if (background_noise_.initialized()) {
+    right_side = background_noise_.Energy(kRefChannel);
+  } else {
+    // If noise parameters have not been estimated, use a fixed threshold.
+    right_side = 75000;
+  }
+  int right_scale = 16 - WebRtcSpl_NormW32(right_side);
+  right_scale = std::max(0, right_scale);
+  left_side = left_side >> right_scale;
+  right_side =
+      rtc::dchecked_cast<int32_t>(peak_index) * (right_side >> right_scale);
+
+  // Scale `left_side` properly before comparing with `right_side`.
+  // (`scaling` is the scale factor before energy calculation, thus the scale
+  // factor for the energy is 2 * scaling.)
+  if (WebRtcSpl_NormW32(left_side) < 2 * scaling) {
+    // Cannot scale only `left_side`, must scale `right_side` too.
+    int temp_scale = WebRtcSpl_NormW32(left_side);
+    left_side = left_side << temp_scale;
+    right_side = right_side >> (2 * scaling - temp_scale);
+  } else {
+    left_side = left_side << 2 * scaling;
+  }
+  return left_side > right_side;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.h b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.h
new file mode 100644
index 0000000000..f0ddaebeca
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.h
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
+#define MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
+
+#include <string.h>  // memset, size_t
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class BackgroundNoise;
+
+// This is the base class for Accelerate and PreemptiveExpand. This class
+// cannot be instantiated, but must be used through either of the derived
+// classes.
+class TimeStretch {
+ public:
+  enum ReturnCodes {
+    kSuccess = 0,
+    kSuccessLowEnergy = 1,
+    kNoStretch = 2,
+    kError = -1
+  };
+
+  TimeStretch(int sample_rate_hz,
+              size_t num_channels,
+              const BackgroundNoise& background_noise)
+      : sample_rate_hz_(sample_rate_hz),
+        fs_mult_(sample_rate_hz / 8000),
+        num_channels_(num_channels),
+        background_noise_(background_noise),
+        max_input_value_(0) {
+    RTC_DCHECK(sample_rate_hz_ == 8000 || sample_rate_hz_ == 16000 ||
+               sample_rate_hz_ == 32000 || sample_rate_hz_ == 48000);
+    RTC_DCHECK_GT(num_channels_, 0);
+    memset(auto_correlation_, 0, sizeof(auto_correlation_));
+  }
+
+  virtual ~TimeStretch() {}
+
+  TimeStretch(const TimeStretch&) = delete;
+  TimeStretch& operator=(const TimeStretch&) = delete;
+
+  // This method performs the processing common to both Accelerate and
+  // PreemptiveExpand.
+  ReturnCodes Process(const int16_t* input,
+                      size_t input_len,
+                      bool fast_mode,
+                      AudioMultiVector* output,
+                      size_t* length_change_samples);
+
+ protected:
+  // Sets the parameters `best_correlation` and `peak_index` to suitable
+  // values when the signal contains no active speech. This method must be
+  // implemented by the sub-classes.
+  virtual void SetParametersForPassiveSpeech(size_t input_length,
+                                             int16_t* best_correlation,
+                                             size_t* peak_index) const = 0;
+
+  // Checks the criteria for performing the time-stretching operation and,
+  // if possible, performs the time-stretching. This method must be implemented
+  // by the sub-classes.
+  virtual ReturnCodes CheckCriteriaAndStretch(
+      const int16_t* input,
+      size_t input_length,
+      size_t peak_index,
+      int16_t best_correlation,
+      bool active_speech,
+      bool fast_mode,
+      AudioMultiVector* output) const = 0;
+
+  static const size_t kCorrelationLen = 50;
+  static const size_t kLogCorrelationLen = 6;  // >= log2(kCorrelationLen).
+  static const size_t kMinLag = 10;
+  static const size_t kMaxLag = 60;
+  static const size_t kDownsampledLen = kCorrelationLen + kMaxLag;
+  static const int kCorrelationThreshold = 14746;  // 0.9 in Q14.
+  static constexpr size_t kRefChannel = 0;  // First channel is reference.
+
+  const int sample_rate_hz_;
+  const int fs_mult_;  // Sample rate multiplier = sample_rate_hz_ / 8000.
+  const size_t num_channels_;
+  const BackgroundNoise& background_noise_;
+  int16_t max_input_value_;
+  int16_t downsampled_input_[kDownsampledLen];
+  // Adding 1 to the size of `auto_correlation_` because of how it is used
+  // by the peak-detection algorithm.
+  int16_t auto_correlation_[kCorrelationLen + 1];
+
+ private:
+  // Calculates the auto-correlation of `downsampled_input_` and writes the
+  // result to `auto_correlation_`.
+  void AutoCorrelation();
+
+  // Performs a simple voice-activity detection based on the input parameters.
+  bool SpeechDetection(int32_t vec1_energy,
+                       int32_t vec2_energy,
+                       size_t peak_index,
+                       int scaling) const;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
new file mode 100644
index 0000000000..da3a98229a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for Accelerate and PreemptiveExpand classes.
+
+#include <map>
+#include <memory>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+const size_t kNumChannels = 1;
+}
+
+TEST(TimeStretch, CreateAndDestroy) {
+  const int kSampleRate = 8000;
+  const int kOverlapSamples = 5 * kSampleRate / 8000;
+  BackgroundNoise bgn(kNumChannels);
+  Accelerate accelerate(kSampleRate, kNumChannels, bgn);
+  PreemptiveExpand preemptive_expand(kSampleRate, kNumChannels, bgn,
+                                     kOverlapSamples);
+}
+
+TEST(TimeStretch, CreateUsingFactory) {
+  const int kSampleRate = 8000;
+  const int kOverlapSamples = 5 * kSampleRate / 8000;
+  BackgroundNoise bgn(kNumChannels);
+
+  AccelerateFactory accelerate_factory;
+  Accelerate* accelerate =
+      accelerate_factory.Create(kSampleRate, kNumChannels, bgn);
+  EXPECT_TRUE(accelerate != NULL);
+  delete accelerate;
+
+  PreemptiveExpandFactory preemptive_expand_factory;
+  PreemptiveExpand* preemptive_expand = preemptive_expand_factory.Create(
+      kSampleRate, kNumChannels, bgn, kOverlapSamples);
+  EXPECT_TRUE(preemptive_expand != NULL);
+  delete preemptive_expand;
+}
+
+class TimeStretchTest : public ::testing::Test {
+ protected:
+  TimeStretchTest()
+      : input_file_(new test::InputAudioFile(
+            test::ResourcePath("audio_coding/testfile32kHz", "pcm"))),
+        sample_rate_hz_(32000),
+        block_size_(30 * sample_rate_hz_ / 1000),  // 30 ms
+        audio_(new int16_t[block_size_]),
+        background_noise_(kNumChannels) {}
+
+  const int16_t* Next30Ms() {
+    RTC_CHECK(input_file_->Read(block_size_, audio_.get()));
+    return audio_.get();
+  }
+
+  // Returns the total length change (in samples) that the accelerate operation
+  // resulted in during the run.
+  size_t TestAccelerate(size_t loops, bool fast_mode) {
+    Accelerate accelerate(sample_rate_hz_, kNumChannels, background_noise_);
+    size_t total_length_change = 0;
+    for (size_t i = 0; i < loops; ++i) {
+      AudioMultiVector output(kNumChannels);
+      size_t length_change;
+      UpdateReturnStats(accelerate.Process(Next30Ms(), block_size_, fast_mode,
+                                           &output, &length_change));
+      total_length_change += length_change;
+    }
+    return total_length_change;
+  }
+
+  void UpdateReturnStats(TimeStretch::ReturnCodes ret) {
+    switch (ret) {
+      case TimeStretch::kSuccess:
+      case TimeStretch::kSuccessLowEnergy:
+      case TimeStretch::kNoStretch:
+        ++return_stats_[ret];
+        break;
+      case TimeStretch::kError:
+        FAIL() << "Process returned an error";
+    }
+  }
+
+  std::unique_ptr<test::InputAudioFile> input_file_;
+  const int sample_rate_hz_;
+  const size_t block_size_;
+  std::unique_ptr<int16_t[]> audio_;
+  std::map<TimeStretch::ReturnCodes, int> return_stats_;
+  BackgroundNoise background_noise_;
+};
+
+TEST_F(TimeStretchTest, Accelerate) {
+  // TestAccelerate returns the total length change in samples.
+  EXPECT_EQ(15268U, TestAccelerate(100, false));
+  EXPECT_EQ(9, return_stats_[TimeStretch::kSuccess]);
+  EXPECT_EQ(58, return_stats_[TimeStretch::kSuccessLowEnergy]);
+  EXPECT_EQ(33, return_stats_[TimeStretch::kNoStretch]);
+}
+
+TEST_F(TimeStretchTest, AccelerateFastMode) {
+  // TestAccelerate returns the total length change in samples.
+  EXPECT_EQ(21400U, TestAccelerate(100, true));
+  EXPECT_EQ(31, return_stats_[TimeStretch::kSuccess]);
+  EXPECT_EQ(58, return_stats_[TimeStretch::kSuccessLowEnergy]);
+  EXPECT_EQ(11, return_stats_[TimeStretch::kNoStretch]);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.cc b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.cc
new file mode 100644
index 0000000000..59177d027f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.cc
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
+
+#include "api/audio_codecs/audio_format.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void TimestampScaler::Reset() {
+  first_packet_received_ = false;
+}
+
+void TimestampScaler::ToInternal(Packet* packet) {
+  if (!packet) {
+    return;
+  }
+  packet->timestamp = ToInternal(packet->timestamp, packet->payload_type);
+}
+
+void TimestampScaler::ToInternal(PacketList* packet_list) {
+  PacketList::iterator it;
+  for (it = packet_list->begin(); it != packet_list->end(); ++it) {
+    ToInternal(&(*it));
+  }
+}
+
+uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
+                                     uint8_t rtp_payload_type) {
+  const DecoderDatabase::DecoderInfo* info =
+      decoder_database_.GetDecoderInfo(rtp_payload_type);
+  if (!info) {
+    // Payload type is unknown. Do not scale.
+    return external_timestamp;
+  }
+  if (!(info->IsComfortNoise() || info->IsDtmf())) {
+    // Do not change the timestamp scaling settings for DTMF or CNG.
+    numerator_ = info->SampleRateHz();
+    if (info->GetFormat().clockrate_hz == 0) {
+      // If the clockrate is invalid (i.e. with an old-style external codec)
+      // we cannot do any timestamp scaling.
+      denominator_ = numerator_;
+    } else {
+      denominator_ = info->GetFormat().clockrate_hz;
+    }
+  }
+  if (numerator_ != denominator_) {
+    // We have a scale factor != 1.
+    if (!first_packet_received_) {
+      external_ref_ = external_timestamp;
+      internal_ref_ = external_timestamp;
+      first_packet_received_ = true;
+    }
+    const int64_t external_diff = int64_t{external_timestamp} - external_ref_;
+    RTC_DCHECK_GT(denominator_, 0);
+    external_ref_ = external_timestamp;
+    internal_ref_ += (external_diff * numerator_) / denominator_;
+    return internal_ref_;
+  } else {
+    // No scaling.
+    return external_timestamp;
+  }
+}
+
+uint32_t TimestampScaler::ToExternal(uint32_t internal_timestamp) const {
+  if (!first_packet_received_ || (numerator_ == denominator_)) {
+    // Not initialized, or scale factor is 1.
+    return internal_timestamp;
+  } else {
+    const int64_t internal_diff = int64_t{internal_timestamp} - internal_ref_;
+    RTC_DCHECK_GT(numerator_, 0);
+    // Do not update references in this method.
+    // Switch `denominator_` and `numerator_` to convert the other way.
+    return external_ref_ + (internal_diff * denominator_) / numerator_;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.h b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.h
new file mode 100644
index 0000000000..f42ce7207a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
+
+#include "modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+// Forward declaration.
+class DecoderDatabase;
+
+// This class scales timestamps for codecs that need timestamp scaling.
+// This is done for codecs where one RTP timestamp does not correspond to
+// one sample.
+class TimestampScaler {
+ public:
+  explicit TimestampScaler(const DecoderDatabase& decoder_database)
+      : first_packet_received_(false),
+        numerator_(1),
+        denominator_(1),
+        external_ref_(0),
+        internal_ref_(0),
+        decoder_database_(decoder_database) {}
+
+  virtual ~TimestampScaler() {}
+
+  TimestampScaler(const TimestampScaler&) = delete;
+  TimestampScaler& operator=(const TimestampScaler&) = delete;
+
+  // Start over.
+  virtual void Reset();
+
+  // Scale the timestamp in `packet` from external to internal.
+  virtual void ToInternal(Packet* packet);
+
+  // Scale the timestamp for all packets in `packet_list` from external to
+  // internal.
+  virtual void ToInternal(PacketList* packet_list);
+
+  // Returns the internal equivalent of `external_timestamp`, given the
+  // RTP payload type `rtp_payload_type`.
+  virtual uint32_t ToInternal(uint32_t external_timestamp,
+                              uint8_t rtp_payload_type);
+
+  // Scales back to external timestamp. This is the inverse of ToInternal().
+  virtual uint32_t ToExternal(uint32_t internal_timestamp) const;
+
+ private:
+  bool first_packet_received_;
+  int numerator_;
+  int denominator_;
+  uint32_t external_ref_;
+  uint32_t internal_ref_;
+  const DecoderDatabase& decoder_database_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
new file mode 100644
index 0000000000..c2bb4dd95f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
@@ -0,0 +1,324 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::Return;
+using ::testing::ReturnNull;
+
+namespace webrtc {
+
+TEST(TimestampScaler, TestNoScaling) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use PCMu, because it doesn't use scaled timestamps.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 0;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  for (uint32_t timestamp = 0xFFFFFFFF - 5; timestamp != 5; ++timestamp) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(timestamp, scaler.ToInternal(timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(timestamp, scaler.ToExternal(timestamp));
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestNoScalingLargeStep) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use PCMu, because it doesn't use scaled timestamps.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 0;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  static const uint32_t kStep = 160;
+  uint32_t start_timestamp = 0;
+  // `external_timestamp` will be a large positive value.
+  start_timestamp = start_timestamp - 5 * kStep;
+  for (uint32_t timestamp = start_timestamp; timestamp != 5 * kStep;
+       timestamp += kStep) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(timestamp, scaler.ToInternal(timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(timestamp, scaler.ToExternal(timestamp));
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestG722) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  for (; external_timestamp != 5; ++external_timestamp) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += 2;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestG722LargeStep) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  static const uint32_t kStep = 320;
+  uint32_t external_timestamp = 0;
+  // `external_timestamp` will be a large positive value.
+  external_timestamp = external_timestamp - 5 * kStep;
+  uint32_t internal_timestamp = external_timestamp;
+  for (; external_timestamp != 5 * kStep; external_timestamp += kStep) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    // Internal timestamp should be incremented with twice the step.
+    internal_timestamp += 2 * kStep;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestG722WithCng) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info_g722(SdpAudioFormat("g722", 8000, 1),
+                                               absl::nullopt, factory.get());
+  const DecoderDatabase::DecoderInfo info_cng(SdpAudioFormat("cn", 16000, 1),
+                                              absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadTypeG722 = 17;
+  static const uint8_t kRtpPayloadTypeCng = 13;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeG722))
+      .WillRepeatedly(Return(&info_g722));
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeCng))
+      .WillRepeatedly(Return(&info_cng));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  bool next_is_cng = false;
+  for (; external_timestamp != 5; ++external_timestamp) {
+    // Alternate between G.722 and CNG every other packet.
+    if (next_is_cng) {
+      // Scale to internal timestamp.
+      EXPECT_EQ(internal_timestamp,
+                scaler.ToInternal(external_timestamp, kRtpPayloadTypeCng));
+      next_is_cng = false;
+    } else {
+      // Scale to internal timestamp.
+      EXPECT_EQ(internal_timestamp,
+                scaler.ToInternal(external_timestamp, kRtpPayloadTypeG722));
+      next_is_cng = true;
+    }
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += 2;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+// Make sure that the method ToInternal(Packet* packet) is wired up correctly.
+// Since it is simply calling the other ToInternal method, we are not doing
+// as many tests here.
+TEST(TimestampScaler, TestG722Packet) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  Packet packet;
+  packet.payload_type = kRtpPayloadType;
+  for (; external_timestamp != 5; ++external_timestamp) {
+    packet.timestamp = external_timestamp;
+    // Scale to internal timestamp.
+    scaler.ToInternal(&packet);
+    EXPECT_EQ(internal_timestamp, packet.timestamp);
+    internal_timestamp += 2;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+// Make sure that the method ToInternal(PacketList* packet_list) is wired up
+// correctly. Since it is simply calling the ToInternal(Packet* packet) method,
+// we are not doing as many tests here.
+TEST(TimestampScaler, TestG722PacketList) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  PacketList packet_list;
+  {
+    Packet packet1;
+    packet1.payload_type = kRtpPayloadType;
+    packet1.timestamp = external_timestamp;
+    Packet packet2;
+    packet2.payload_type = kRtpPayloadType;
+    packet2.timestamp = external_timestamp + 10;
+    packet_list.push_back(std::move(packet1));
+    packet_list.push_back(std::move(packet2));
+  }
+
+  scaler.ToInternal(&packet_list);
+  EXPECT_EQ(internal_timestamp, packet_list.front().timestamp);
+  packet_list.pop_front();
+  EXPECT_EQ(internal_timestamp + 20, packet_list.front().timestamp);
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestG722Reset) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  for (; external_timestamp != 5; ++external_timestamp) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += 2;
+  }
+  // Reset the scaler. After this, we expect the internal and external to start
+  // over at the same value again.
+  scaler.Reset();
+  internal_timestamp = external_timestamp;
+  for (; external_timestamp != 15; ++external_timestamp) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += 2;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+// TODO(minyue): This test becomes trivial since Opus does not need a timestamp
+// scaler. Therefore, this test may be removed in future. There is no harm to
+// keep it, since it can be taken as a test case for the situation of a trivial
+// timestamp scaler.
+TEST(TimestampScaler, TestOpusLargeStep) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("opus", 48000, 2),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  static const uint32_t kStep = 960;
+  uint32_t external_timestamp = 0;
+  // `external_timestamp` will be a large positive value.
+  external_timestamp = external_timestamp - 5 * kStep;
+  uint32_t internal_timestamp = external_timestamp;
+  for (; external_timestamp != 5 * kStep; external_timestamp += kStep) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += kStep;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, Failures) {
+  static const uint8_t kRtpPayloadType = 17;
+  MockDecoderDatabase db;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillOnce(ReturnNull());  // Return NULL to indicate unknown payload type.
+
+  TimestampScaler scaler(db);
+  uint32_t timestamp = 4711;  // Some number.
+  EXPECT_EQ(timestamp, scaler.ToInternal(timestamp, kRtpPayloadType));
+
+  Packet* packet = NULL;
+  scaler.ToInternal(packet);  // Should not crash. That's all we can test.
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/DEPS b/third_party/libwebrtc/modules/audio_coding/neteq/tools/DEPS
new file mode 100644
index 0000000000..4db1e1d6e5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/DEPS
@@ -0,0 +1,3 @@
+include_rules = [
+  "+logging/rtc_event_log",
+]
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/README.md b/third_party/libwebrtc/modules/audio_coding/neteq/tools/README.md
new file mode 100644
index 0000000000..e7bd95c285
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/README.md
@@ -0,0 +1,17 @@
+# NetEQ RTP Play tool
+
+## Testing of the command line arguments
+The command line tool `neteq_rtpplay` can be tested by running `neteq_rtpplay_test.sh`, which is not use on try bots, but it can be used before submitting any CLs that may break the behavior of the command line arguments of `neteq_rtpplay`.
+
+Run `neteq_rtpplay_test.sh` as follows from the `src/` folder:
+```
+src$ ./modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh  \
+  out/Default/neteq_rtpplay  \
+  resources/audio_coding/neteq_opus.rtp  \
+  resources/short_mixed_mono_48.pcm
+```
+
+You can replace the RTP and PCM files with any other compatible files.
+If you get an error using the files indicated above, try running `gclient sync`.
+
+Requirements: `awk` and `md5sum`.
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_checksum.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_checksum.h
new file mode 100644
index 0000000000..42e3a3a3a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/message_digest.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace test {
+
+class AudioChecksum : public AudioSink {
+ public:
+  AudioChecksum()
+      : checksum_(rtc::MessageDigestFactory::Create(rtc::DIGEST_MD5)),
+        checksum_result_(checksum_->Size()),
+        finished_(false) {}
+
+  AudioChecksum(const AudioChecksum&) = delete;
+  AudioChecksum& operator=(const AudioChecksum&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
+    if (finished_)
+      return false;
+
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Big-endian gives a different checksum"
+#endif
+    checksum_->Update(audio, num_samples * sizeof(*audio));
+    return true;
+  }
+
+  // Finalizes the computations, and returns the checksum.
+  std::string Finish() {
+    if (!finished_) {
+      finished_ = true;
+      checksum_->Finish(checksum_result_.data(), checksum_result_.size());
+    }
+    return rtc::hex_encode(checksum_result_);
+  }
+
+ private:
+  std::unique_ptr<rtc::MessageDigest> checksum_;
+  rtc::Buffer checksum_result_;
+  bool finished_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.cc
new file mode 100644
index 0000000000..514e6eb2ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+bool AudioLoop::Init(absl::string_view file_name,
+                     size_t max_loop_length_samples,
+                     size_t block_length_samples) {
+  FILE* fp = fopen(std::string(file_name).c_str(), "rb");
+  if (!fp)
+    return false;
+
+  audio_array_.reset(
+      new int16_t[max_loop_length_samples + block_length_samples]);
+  size_t samples_read =
+      fread(audio_array_.get(), sizeof(int16_t), max_loop_length_samples, fp);
+  fclose(fp);
+
+  // Block length must be shorter than the loop length.
+  if (block_length_samples > samples_read)
+    return false;
+
+  // Add an extra block length of samples to the end of the array, starting
+  // over again from the beginning of the array. This is done to simplify
+  // the reading process when reading over the end of the loop.
+  memcpy(&audio_array_[samples_read], audio_array_.get(),
+         block_length_samples * sizeof(int16_t));
+
+  loop_length_samples_ = samples_read;
+  block_length_samples_ = block_length_samples;
+  next_index_ = 0;
+  return true;
+}
+
+rtc::ArrayView<const int16_t> AudioLoop::GetNextBlock() {
+  // Check that the AudioLoop is initialized.
+  if (block_length_samples_ == 0)
+    return rtc::ArrayView<const int16_t>();
+
+  const int16_t* output_ptr = &audio_array_[next_index_];
+  next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
+  return rtc::ArrayView<const int16_t>(output_ptr, block_length_samples_);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.h
new file mode 100644
index 0000000000..f5f0b59011
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+namespace test {
+
+// Class serving as an infinite source of audio, realized by looping an audio
+// clip.
+class AudioLoop {
+ public:
+  AudioLoop()
+      : next_index_(0), loop_length_samples_(0), block_length_samples_(0) {}
+
+  virtual ~AudioLoop() {}
+
+  AudioLoop(const AudioLoop&) = delete;
+  AudioLoop& operator=(const AudioLoop&) = delete;
+
+  // Initializes the AudioLoop by reading from `file_name`. The loop will be no
+  // longer than `max_loop_length_samples`, if the length of the file is
+  // greater. Otherwise, the loop length is the same as the file length.
+  // The audio will be delivered in blocks of `block_length_samples`.
+  // Returns false if the initialization failed, otherwise true.
+  bool Init(absl::string_view file_name,
+            size_t max_loop_length_samples,
+            size_t block_length_samples);
+
+  // Returns a (pointer,size) pair for the next block of audio. The size is
+  // equal to the `block_length_samples` Init() argument.
+  rtc::ArrayView<const int16_t> GetNextBlock();
+
+ private:
+  size_t next_index_;
+  size_t loop_length_samples_;
+  size_t block_length_samples_;
+  std::unique_ptr<int16_t[]> audio_array_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.cc
new file mode 100644
index 0000000000..7d7af7ef9f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.cc
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+
+namespace webrtc {
+namespace test {
+
+bool AudioSinkFork::WriteArray(const int16_t* audio, size_t num_samples) {
+  return left_sink_->WriteArray(audio, num_samples) &&
+         right_sink_->WriteArray(audio, num_samples);
+}
+
+bool VoidAudioSink::WriteArray(const int16_t* audio, size_t num_samples) {
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.h
new file mode 100644
index 0000000000..53729fa920
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for an object receiving raw output audio from test
+// applications.
+class AudioSink {
+ public:
+  AudioSink() {}
+  virtual ~AudioSink() {}
+
+  AudioSink(const AudioSink&) = delete;
+  AudioSink& operator=(const AudioSink&) = delete;
+
+  // Writes `num_samples` from `audio` to the AudioSink. Returns true if
+  // successful, otherwise false.
+  virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
+
+  // Writes `audio_frame` to the AudioSink. Returns true if successful,
+  // otherwise false.
+  bool WriteAudioFrame(const AudioFrame& audio_frame) {
+    return WriteArray(audio_frame.data(), audio_frame.samples_per_channel_ *
+                                              audio_frame.num_channels_);
+  }
+};
+
+// Forks the output audio to two AudioSink objects.
+class AudioSinkFork : public AudioSink {
+ public:
+  AudioSinkFork(AudioSink* left, AudioSink* right)
+      : left_sink_(left), right_sink_(right) {}
+
+  AudioSinkFork(const AudioSinkFork&) = delete;
+  AudioSinkFork& operator=(const AudioSinkFork&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override;
+
+ private:
+  AudioSink* left_sink_;
+  AudioSink* right_sink_;
+};
+
+// An AudioSink implementation that does nothing.
+class VoidAudioSink : public AudioSink {
+ public:
+  VoidAudioSink() = default;
+
+  VoidAudioSink(const VoidAudioSink&) = delete;
+  VoidAudioSink& operator=(const VoidAudioSink&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
new file mode 100644
index 0000000000..18a910365f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
+
+#include <algorithm>
+
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+ConstantPcmPacketSource::ConstantPcmPacketSource(size_t payload_len_samples,
+                                                 int16_t sample_value,
+                                                 int sample_rate_hz,
+                                                 int payload_type)
+    : payload_len_samples_(payload_len_samples),
+      packet_len_bytes_(2 * payload_len_samples_ + kHeaderLenBytes),
+      samples_per_ms_(sample_rate_hz / 1000),
+      next_arrival_time_ms_(0.0),
+      payload_type_(payload_type),
+      seq_number_(0),
+      timestamp_(0),
+      payload_ssrc_(0xABCD1234) {
+  size_t encoded_len = WebRtcPcm16b_Encode(&sample_value, 1, encoded_sample_);
+  RTC_CHECK_EQ(2U, encoded_len);
+}
+
+std::unique_ptr<Packet> ConstantPcmPacketSource::NextPacket() {
+  RTC_CHECK_GT(packet_len_bytes_, kHeaderLenBytes);
+  rtc::CopyOnWriteBuffer packet_buffer(packet_len_bytes_);
+  uint8_t* packet_memory = packet_buffer.MutableData();
+  // Fill the payload part of the packet memory with the pre-encoded value.
+  for (unsigned i = 0; i < 2 * payload_len_samples_; ++i)
+    packet_memory[kHeaderLenBytes + i] = encoded_sample_[i % 2];
+  WriteHeader(packet_memory);
+  // `packet` assumes ownership of `packet_memory`.
+  auto packet =
+      std::make_unique<Packet>(std::move(packet_buffer), next_arrival_time_ms_);
+  next_arrival_time_ms_ += payload_len_samples_ / samples_per_ms_;
+  return packet;
+}
+
+void ConstantPcmPacketSource::WriteHeader(uint8_t* packet_memory) {
+  packet_memory[0] = 0x80;
+  packet_memory[1] = static_cast<uint8_t>(payload_type_);
+  packet_memory[2] = seq_number_ >> 8;
+  packet_memory[3] = seq_number_ & 0xFF;
+  packet_memory[4] = timestamp_ >> 24;
+  packet_memory[5] = (timestamp_ >> 16) & 0xFF;
+  packet_memory[6] = (timestamp_ >> 8) & 0xFF;
+  packet_memory[7] = timestamp_ & 0xFF;
+  packet_memory[8] = payload_ssrc_ >> 24;
+  packet_memory[9] = (payload_ssrc_ >> 16) & 0xFF;
+  packet_memory[10] = (payload_ssrc_ >> 8) & 0xFF;
+  packet_memory[11] = payload_ssrc_ & 0xFF;
+  ++seq_number_;
+  timestamp_ += static_cast<uint32_t>(payload_len_samples_);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
new file mode 100644
index 0000000000..ab4f5c2281
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
+
+#include <stdio.h>
+
+#include <string>
+
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+
+namespace webrtc {
+namespace test {
+
+// This class implements a packet source that delivers PCM16b encoded packets
+// with a constant sample value. The payload length, constant sample value,
+// sample rate, and payload type are all set in the constructor.
+class ConstantPcmPacketSource : public PacketSource {
+ public:
+  ConstantPcmPacketSource(size_t payload_len_samples,
+                          int16_t sample_value,
+                          int sample_rate_hz,
+                          int payload_type);
+
+  ConstantPcmPacketSource(const ConstantPcmPacketSource&) = delete;
+  ConstantPcmPacketSource& operator=(const ConstantPcmPacketSource&) = delete;
+
+  std::unique_ptr<Packet> NextPacket() override;
+
+ private:
+  void WriteHeader(uint8_t* packet_memory);
+
+  const size_t kHeaderLenBytes = 12;
+  const size_t payload_len_samples_;
+  const size_t packet_len_bytes_;
+  uint8_t encoded_sample_[2];
+  const int samples_per_ms_;
+  double next_arrival_time_ms_;
+  const int payload_type_;
+  uint16_t seq_number_;
+  uint32_t timestamp_;
+  const uint32_t payload_ssrc_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
new file mode 100644
index 0000000000..87b987ddb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/encode_neteq_input.h"
+
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace test {
+
+EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<Generator> generator,
+                                   std::unique_ptr<AudioEncoder> encoder,
+                                   int64_t input_duration_ms)
+    : generator_(std::move(generator)),
+      encoder_(std::move(encoder)),
+      input_duration_ms_(input_duration_ms) {
+  CreatePacket();
+}
+
+EncodeNetEqInput::~EncodeNetEqInput() = default;
+
+absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
+  RTC_DCHECK(packet_data_);
+  return static_cast<int64_t>(packet_data_->time_ms);
+}
+
+absl::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
+  return next_output_event_ms_;
+}
+
+std::unique_ptr<NetEqInput::PacketData> EncodeNetEqInput::PopPacket() {
+  RTC_DCHECK(packet_data_);
+  // Grab the packet to return...
+  std::unique_ptr<PacketData> packet_to_return = std::move(packet_data_);
+  // ... and line up the next packet for future use.
+  CreatePacket();
+
+  return packet_to_return;
+}
+
+void EncodeNetEqInput::AdvanceOutputEvent() {
+  next_output_event_ms_ += kOutputPeriodMs;
+}
+
+bool EncodeNetEqInput::ended() const {
+  return next_output_event_ms_ > input_duration_ms_;
+}
+
+absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
+  RTC_DCHECK(packet_data_);
+  return packet_data_->header;
+}
+
+void EncodeNetEqInput::CreatePacket() {
+  // Create a new PacketData object.
+  RTC_DCHECK(!packet_data_);
+  packet_data_.reset(new NetEqInput::PacketData);
+  RTC_DCHECK_EQ(packet_data_->payload.size(), 0);
+
+  // Loop until we get a packet.
+  AudioEncoder::EncodedInfo info;
+  RTC_DCHECK(!info.send_even_if_empty);
+  int num_blocks = 0;
+  while (packet_data_->payload.size() == 0 && !info.send_even_if_empty) {
+    const size_t num_samples = rtc::CheckedDivExact(
+        static_cast<int>(encoder_->SampleRateHz() * kOutputPeriodMs), 1000);
+
+    info = encoder_->Encode(rtp_timestamp_, generator_->Generate(num_samples),
+                            &packet_data_->payload);
+
+    rtp_timestamp_ += rtc::dchecked_cast<uint32_t>(
+        num_samples * encoder_->RtpTimestampRateHz() /
+        encoder_->SampleRateHz());
+    ++num_blocks;
+  }
+  packet_data_->header.timestamp = info.encoded_timestamp;
+  packet_data_->header.payloadType = info.payload_type;
+  packet_data_->header.sequenceNumber = sequence_number_++;
+  packet_data_->time_ms = next_packet_time_ms_;
+  next_packet_time_ms_ += num_blocks * kOutputPeriodMs;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
new file mode 100644
index 0000000000..caa9ac76f4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
+
+#include <memory>
+
+#include "api/audio_codecs/audio_encoder.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+
+namespace webrtc {
+namespace test {
+
+// This class provides a NetEqInput that takes audio from a generator object and
+// encodes it using a given audio encoder.
+class EncodeNetEqInput : public NetEqInput {
+ public:
+  // Generator class, to be provided to the EncodeNetEqInput constructor.
+  class Generator {
+   public:
+    virtual ~Generator() = default;
+    // Returns the next num_samples values from the signal generator.
+    virtual rtc::ArrayView<const int16_t> Generate(size_t num_samples) = 0;
+  };
+
+  // The source will end after the given input duration.
+  EncodeNetEqInput(std::unique_ptr<Generator> generator,
+                   std::unique_ptr<AudioEncoder> encoder,
+                   int64_t input_duration_ms);
+  ~EncodeNetEqInput() override;
+
+  absl::optional<int64_t> NextPacketTime() const override;
+
+  absl::optional<int64_t> NextOutputEventTime() const override;
+
+  std::unique_ptr<PacketData> PopPacket() override;
+
+  void AdvanceOutputEvent() override;
+
+  bool ended() const override;
+
+  absl::optional<RTPHeader> NextHeader() const override;
+
+ private:
+  static constexpr int64_t kOutputPeriodMs = 10;
+
+  void CreatePacket();
+
+  std::unique_ptr<Generator> generator_;
+  std::unique_ptr<AudioEncoder> encoder_;
+  std::unique_ptr<PacketData> packet_data_;
+  uint32_t rtp_timestamp_ = 0;
+  int16_t sequence_number_ = 0;
+  int64_t next_packet_time_ms_ = 0;
+  int64_t next_output_event_ms_ = 0;
+  const int64_t input_duration_ms_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
new file mode 100644
index 0000000000..6c5e5ac2e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+class FakeEncodedFrame : public AudioDecoder::EncodedAudioFrame {
+ public:
+  FakeEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload)
+      : decoder_(decoder), payload_(std::move(payload)) {}
+
+  size_t Duration() const override {
+    const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    return ret < 0 ? 0 : static_cast<size_t>(ret);
+  }
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    auto speech_type = AudioDecoder::kSpeech;
+    const int ret = decoder_->Decode(
+        payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+        decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    return ret < 0 ? absl::nullopt
+                   : absl::optional<DecodeResult>(
+                         {static_cast<size_t>(ret), speech_type});
+  }
+
+  // This is to mimic OpusFrame.
+  bool IsDtxPacket() const override {
+    uint32_t original_payload_size_bytes =
+        ByteReader<uint32_t>::ReadLittleEndian(&payload_.data()[8]);
+    return original_payload_size_bytes <= 2;
+  }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+};
+
+}  // namespace
+
+std::vector<AudioDecoder::ParseResult> FakeDecodeFromFile::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new FakeEncodedFrame(this, std::move(payload)));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
+                                       size_t encoded_len,
+                                       int sample_rate_hz,
+                                       int16_t* decoded,
+                                       SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, SampleRateHz());
+
+  const int samples_to_decode = PacketDuration(encoded, encoded_len);
+  const int total_samples_to_decode = samples_to_decode * (stereo_ ? 2 : 1);
+
+  if (encoded_len == 0) {
+    // Decoder is asked to produce codec-internal comfort noise.
+    RTC_DCHECK(!encoded);  // NetEq always sends nullptr in this case.
+    RTC_DCHECK(cng_mode_);
+    RTC_DCHECK_GT(total_samples_to_decode, 0);
+    std::fill_n(decoded, total_samples_to_decode, 0);
+    *speech_type = kComfortNoise;
+    return rtc::dchecked_cast<int>(total_samples_to_decode);
+  }
+
+  RTC_CHECK_GE(encoded_len, 12);
+  uint32_t timestamp_to_decode =
+      ByteReader<uint32_t>::ReadLittleEndian(encoded);
+
+  if (next_timestamp_from_input_ &&
+      timestamp_to_decode != *next_timestamp_from_input_) {
+    // A gap in the timestamp sequence is detected. Skip the same number of
+    // samples from the file.
+    uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_;
+    RTC_CHECK(input_->Seek(jump));
+  }
+
+  next_timestamp_from_input_ = timestamp_to_decode + samples_to_decode;
+
+  uint32_t original_payload_size_bytes =
+      ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
+  if (original_payload_size_bytes <= 2) {
+    // This is a comfort noise payload.
+    RTC_DCHECK_GT(total_samples_to_decode, 0);
+    std::fill_n(decoded, total_samples_to_decode, 0);
+    *speech_type = kComfortNoise;
+    cng_mode_ = true;
+    return rtc::dchecked_cast<int>(total_samples_to_decode);
+  }
+
+  cng_mode_ = false;
+  RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded));
+
+  if (stereo_) {
+    InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2,
+                                         decoded);
+  }
+
+  *speech_type = kSpeech;
+  last_decoded_length_ = samples_to_decode;
+  return rtc::dchecked_cast<int>(total_samples_to_decode);
+}
+
+int FakeDecodeFromFile::PacketDuration(const uint8_t* encoded,
+                                       size_t encoded_len) const {
+  const uint32_t original_payload_size_bytes =
+      encoded_len < 8 + sizeof(uint32_t)
+          ? 0
+          : ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
+  const uint32_t samples_to_decode =
+      encoded_len < 4 + sizeof(uint32_t)
+          ? 0
+          : ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
+  if (encoded_len == 0) {
+    // Decoder is asked to produce codec-internal comfort noise.
+    return rtc::CheckedDivExact(SampleRateHz(), 100);
+  }
+  bool is_dtx_payload =
+      original_payload_size_bytes <= 2 || samples_to_decode == 0;
+  bool has_error_duration =
+      samples_to_decode % rtc::CheckedDivExact(SampleRateHz(), 100) != 0;
+  if (is_dtx_payload || has_error_duration) {
+    if (last_decoded_length_ > 0) {
+      // Use length of last decoded packet.
+      return rtc::dchecked_cast<int>(last_decoded_length_);
+    } else {
+      // This is the first packet to decode, and we do not know the length of
+      // it. Set it to 10 ms.
+      return rtc::CheckedDivExact(SampleRateHz(), 100);
+    }
+  }
+  return samples_to_decode;
+}
+
+void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
+                                        size_t samples,
+                                        size_t original_payload_size_bytes,
+                                        rtc::ArrayView<uint8_t> encoded) {
+  RTC_CHECK_GE(encoded.size(), 12);
+  ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
+  ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
+                                          rtc::checked_cast<uint32_t>(samples));
+  ByteWriter<uint32_t>::WriteLittleEndian(
+      &encoded[8], rtc::checked_cast<uint32_t>(original_payload_size_bytes));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
new file mode 100644
index 0000000000..7b53653998
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+
+namespace webrtc {
+namespace test {
+// Provides an AudioDecoder implementation that delivers audio data from a file.
+// The "encoded" input should contain information about what RTP timestamp the
+// encoding represents, and how many samples the decoder should produce for that
+// encoding. A helper method PrepareEncoded is provided to prepare such
+// encodings. If packets are missing, as determined from the timestamps, the
+// file reading will skip forward to match the loss.
+class FakeDecodeFromFile : public AudioDecoder {
+ public:
+  FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input,
+                     int sample_rate_hz,
+                     bool stereo)
+      : input_(std::move(input)),
+        sample_rate_hz_(sample_rate_hz),
+        stereo_(stereo) {}
+
+  ~FakeDecodeFromFile() = default;
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+
+  void Reset() override {}
+
+  int SampleRateHz() const override { return sample_rate_hz_; }
+
+  size_t Channels() const override { return stereo_ ? 2 : 1; }
+
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+
+  // Helper method. Writes `timestamp`, `samples` and
+  // `original_payload_size_bytes` to `encoded` in a format that the
+  // FakeDecodeFromFile decoder will understand. `encoded` must be at least 12
+  // bytes long.
+  static void PrepareEncoded(uint32_t timestamp,
+                             size_t samples,
+                             size_t original_payload_size_bytes,
+                             rtc::ArrayView<uint8_t> encoded);
+
+ private:
+  std::unique_ptr<InputAudioFile> input_;
+  absl::optional<uint32_t> next_timestamp_from_input_;
+  const int sample_rate_hz_;
+  const bool stereo_;
+  size_t last_decoded_length_ = 0;
+  bool cng_mode_ = false;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc
new file mode 100644
index 0000000000..3c33aabf1c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h"
+
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+InitialPacketInserterNetEqInput::InitialPacketInserterNetEqInput(
+    std::unique_ptr<NetEqInput> source,
+    int number_of_initial_packets,
+    int sample_rate_hz)
+    : source_(std::move(source)),
+      packets_to_insert_(number_of_initial_packets),
+      sample_rate_hz_(sample_rate_hz) {}
+
+absl::optional<int64_t> InitialPacketInserterNetEqInput::NextPacketTime()
+    const {
+  return source_->NextPacketTime();
+}
+
+absl::optional<int64_t> InitialPacketInserterNetEqInput::NextOutputEventTime()
+    const {
+  return source_->NextOutputEventTime();
+}
+
+std::unique_ptr<InitialPacketInserterNetEqInput::PacketData>
+InitialPacketInserterNetEqInput::PopPacket() {
+  if (!first_packet_) {
+    first_packet_ = source_->PopPacket();
+    if (!first_packet_) {
+      // The source has no packets, so we should not insert any dummy packets.
+      packets_to_insert_ = 0;
+    }
+  }
+  if (packets_to_insert_ > 0) {
+    RTC_CHECK(first_packet_);
+    auto dummy_packet = std::unique_ptr<PacketData>(new PacketData());
+    dummy_packet->header = first_packet_->header;
+    dummy_packet->payload = rtc::Buffer(first_packet_->payload.data(),
+                                        first_packet_->payload.size());
+    dummy_packet->time_ms = first_packet_->time_ms;
+    dummy_packet->header.sequenceNumber -= packets_to_insert_;
+    // This assumes 20ms per packet.
+    dummy_packet->header.timestamp -=
+        20 * sample_rate_hz_ * packets_to_insert_ / 1000;
+    packets_to_insert_--;
+    return dummy_packet;
+  }
+  return source_->PopPacket();
+}
+
+void InitialPacketInserterNetEqInput::AdvanceOutputEvent() {
+  source_->AdvanceOutputEvent();
+}
+
+bool InitialPacketInserterNetEqInput::ended() const {
+  return source_->ended();
+}
+
+absl::optional<RTPHeader> InitialPacketInserterNetEqInput::NextHeader() const {
+  return source_->NextHeader();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h
new file mode 100644
index 0000000000..bd20a7aecf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_INITIAL_PACKET_INSERTER_NETEQ_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_INITIAL_PACKET_INSERTER_NETEQ_INPUT_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+
+namespace webrtc {
+namespace test {
+
+// Wrapper class that can insert a number of packets at the start of the
+// simulation.
+class InitialPacketInserterNetEqInput final : public NetEqInput {
+ public:
+  InitialPacketInserterNetEqInput(std::unique_ptr<NetEqInput> source,
+                                  int number_of_initial_packets,
+                                  int sample_rate_hz);
+  absl::optional<int64_t> NextPacketTime() const override;
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  void AdvanceOutputEvent() override;
+  bool ended() const override;
+  absl::optional<RTPHeader> NextHeader() const override;
+
+ private:
+  const std::unique_ptr<NetEqInput> source_;
+  int packets_to_insert_;
+  const int sample_rate_hz_;
+  std::unique_ptr<PacketData> first_packet_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_INITIAL_PACKET_INSERTER_NETEQ_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
new file mode 100644
index 0000000000..b077dbff21
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+InputAudioFile::InputAudioFile(absl::string_view file_name, bool loop_at_end)
+    : loop_at_end_(loop_at_end) {
+  fp_ = fopen(std::string(file_name).c_str(), "rb");
+  RTC_DCHECK(fp_) << file_name << " could not be opened.";
+}
+
+InputAudioFile::~InputAudioFile() {
+  RTC_DCHECK(fp_);
+  fclose(fp_);
+}
+
+bool InputAudioFile::Read(size_t samples, int16_t* destination) {
+  if (!fp_) {
+    return false;
+  }
+  size_t samples_read = fread(destination, sizeof(int16_t), samples, fp_);
+  if (samples_read < samples) {
+    if (!loop_at_end_) {
+      return false;
+    }
+    // Rewind and read the missing samples.
+    rewind(fp_);
+    size_t missing_samples = samples - samples_read;
+    if (fread(destination + samples_read, sizeof(int16_t), missing_samples,
+              fp_) < missing_samples) {
+      // Could not read enough even after rewinding the file.
+      return false;
+    }
+  }
+  return true;
+}
+
+bool InputAudioFile::Seek(int samples) {
+  if (!fp_) {
+    return false;
+  }
+  // Find file boundaries.
+  const long current_pos = ftell(fp_);
+  RTC_CHECK_NE(EOF, current_pos)
+      << "Error returned when getting file position.";
+  RTC_CHECK_EQ(0, fseek(fp_, 0, SEEK_END));  // Move to end of file.
+  const long file_size = ftell(fp_);
+  RTC_CHECK_NE(EOF, file_size) << "Error returned when getting file position.";
+  // Find new position.
+  long new_pos = current_pos + sizeof(int16_t) * samples;  // Samples to bytes.
+  if (loop_at_end_) {
+    new_pos = new_pos % file_size;  // Wrap around the end of the file.
+    if (new_pos < 0) {
+      // For negative values of new_pos, newpos % file_size will also be
+      // negative. To get the correct result it's needed to add file_size.
+      new_pos += file_size;
+    }
+  } else {
+    new_pos = new_pos > file_size ? file_size : new_pos;  // Don't loop.
+  }
+  RTC_CHECK_GE(new_pos, 0)
+      << "Trying to move to before the beginning of the file";
+  // Move to new position relative to the beginning of the file.
+  RTC_CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET));
+  return true;
+}
+
+void InputAudioFile::DuplicateInterleaved(const int16_t* source,
+                                          size_t samples,
+                                          size_t channels,
+                                          int16_t* destination) {
+  // Start from the end of `source` and `destination`, and work towards the
+  // beginning. This is to allow in-place interleaving of the same array (i.e.,
+  // `source` and `destination` are the same array).
+  for (int i = static_cast<int>(samples - 1); i >= 0; --i) {
+    for (int j = static_cast<int>(channels - 1); j >= 0; --j) {
+      destination[i * channels + j] = source[i];
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.h
new file mode 100644
index 0000000000..f538b295a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
+
+#include <stdio.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for handling a looping input audio file.
+class InputAudioFile {
+ public:
+  explicit InputAudioFile(absl::string_view file_name, bool loop_at_end = true);
+
+  virtual ~InputAudioFile();
+
+  InputAudioFile(const InputAudioFile&) = delete;
+  InputAudioFile& operator=(const InputAudioFile&) = delete;
+
+  // Reads `samples` elements from source file to `destination`. Returns true
+  // if the read was successful, otherwise false. If the file end is reached,
+  // the file is rewound and reading continues from the beginning.
+  // The output `destination` must have the capacity to hold `samples` elements.
+  virtual bool Read(size_t samples, int16_t* destination);
+
+  // Fast-forwards (`samples` > 0) or -backwards (`samples` < 0) the file by the
+  // indicated number of samples. Just like Read(), Seek() starts over at the
+  // beginning of the file if the end is reached. However, seeking backwards
+  // past the beginning of the file is not possible.
+  virtual bool Seek(int samples);
+
+  // Creates a multi-channel signal from a mono signal. Each sample is repeated
+  // `channels` times to create an interleaved multi-channel signal where all
+  // channels are identical. The output `destination` must have the capacity to
+  // hold samples * channels elements. Note that `source` and `destination` can
+  // be the same array (i.e., point to the same address).
+  static void DuplicateInterleaved(const int16_t* source,
+                                   size_t samples,
+                                   size_t channels,
+                                   int16_t* destination);
+
+ private:
+  FILE* fp_;
+  const bool loop_at_end_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
new file mode 100644
index 0000000000..52f7ea82a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for test InputAudioFile class.
+
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+TEST(TestInputAudioFile, DuplicateInterleaveSeparateSrcDst) {
+  static const size_t kSamples = 10;
+  static const size_t kChannels = 2;
+  int16_t input[kSamples];
+  for (size_t i = 0; i < kSamples; ++i) {
+    input[i] = rtc::checked_cast<int16_t>(i);
+  }
+  int16_t output[kSamples * kChannels];
+  InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, output);
+
+  // Verify output
+  int16_t* output_ptr = output;
+  for (size_t i = 0; i < kSamples; ++i) {
+    for (size_t j = 0; j < kChannels; ++j) {
+      EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++);
+    }
+  }
+}
+
+TEST(TestInputAudioFile, DuplicateInterleaveSameSrcDst) {
+  static const size_t kSamples = 10;
+  static const size_t kChannels = 5;
+  int16_t input[kSamples * kChannels];
+  for (size_t i = 0; i < kSamples; ++i) {
+    input[i] = rtc::checked_cast<int16_t>(i);
+  }
+  InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, input);
+
+  // Verify output
+  int16_t* output_ptr = input;
+  for (size_t i = 0; i < kSamples; ++i) {
+    for (size_t j = 0; j < kChannels; ++j) {
+      EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++);
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
new file mode 100644
index 0000000000..9e77457775
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
@@ -0,0 +1,307 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+
+#include <algorithm>
+#include <fstream>
+#include <ios>
+#include <iterator>
+#include <limits>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+constexpr char kArrivalDelayX[] = "arrival_delay_x";
+constexpr char kArrivalDelayY[] = "arrival_delay_y";
+constexpr char kTargetDelayX[] = "target_delay_x";
+constexpr char kTargetDelayY[] = "target_delay_y";
+constexpr char kPlayoutDelayX[] = "playout_delay_x";
+constexpr char kPlayoutDelayY[] = "playout_delay_y";
+
+// Helper function for NetEqDelayAnalyzer::CreateGraphs. Returns the
+// interpolated value of a function at the point x. Vector x_vec contains the
+// sample points, and y_vec contains the function values at these points. The
+// return value is a linear interpolation between y_vec values.
+double LinearInterpolate(double x,
+                         const std::vector<int64_t>& x_vec,
+                         const std::vector<int64_t>& y_vec) {
+  // Find first element which is larger than x.
+  auto it = std::upper_bound(x_vec.begin(), x_vec.end(), x);
+  if (it == x_vec.end()) {
+    --it;
+  }
+  const size_t upper_ix = it - x_vec.begin();
+
+  size_t lower_ix;
+  if (upper_ix == 0 || x_vec[upper_ix] <= x) {
+    lower_ix = upper_ix;
+  } else {
+    lower_ix = upper_ix - 1;
+  }
+  double y;
+  if (lower_ix == upper_ix) {
+    y = y_vec[lower_ix];
+  } else {
+    RTC_DCHECK_NE(x_vec[lower_ix], x_vec[upper_ix]);
+    y = (x - x_vec[lower_ix]) * (y_vec[upper_ix] - y_vec[lower_ix]) /
+            (x_vec[upper_ix] - x_vec[lower_ix]) +
+        y_vec[lower_ix];
+  }
+  return y;
+}
+
+void PrintDelays(const NetEqDelayAnalyzer::Delays& delays,
+                 int64_t ref_time_ms,
+                 absl::string_view var_name_x,
+                 absl::string_view var_name_y,
+                 std::ofstream& output,
+                 absl::string_view terminator = "") {
+  output << var_name_x << " = [ ";
+  for (const std::pair<int64_t, float>& delay : delays) {
+    output << (delay.first - ref_time_ms) / 1000.f << ", ";
+  }
+  output << "]" << terminator << std::endl;
+
+  output << var_name_y << " = [ ";
+  for (const std::pair<int64_t, float>& delay : delays) {
+    output << delay.second << ", ";
+  }
+  output << "]" << terminator << std::endl;
+}
+
+}  // namespace
+
+void NetEqDelayAnalyzer::AfterInsertPacket(
+    const test::NetEqInput::PacketData& packet,
+    NetEq* neteq) {
+  data_.insert(
+      std::make_pair(packet.header.timestamp, TimingData(packet.time_ms)));
+  ssrcs_.insert(packet.header.ssrc);
+  payload_types_.insert(packet.header.payloadType);
+}
+
+void NetEqDelayAnalyzer::BeforeGetAudio(NetEq* neteq) {
+  last_sync_buffer_ms_ = neteq->SyncBufferSizeMs();
+}
+
+void NetEqDelayAnalyzer::AfterGetAudio(int64_t time_now_ms,
+                                       const AudioFrame& audio_frame,
+                                       bool /*muted*/,
+                                       NetEq* neteq) {
+  get_audio_time_ms_.push_back(time_now_ms);
+  for (const RtpPacketInfo& info : audio_frame.packet_infos_) {
+    auto it = data_.find(info.rtp_timestamp());
+    if (it == data_.end()) {
+      // This is a packet that was split out from another packet. Skip it.
+      continue;
+    }
+    auto& it_timing = it->second;
+    RTC_CHECK(!it_timing.decode_get_audio_count)
+        << "Decode time already written";
+    it_timing.decode_get_audio_count = get_audio_count_;
+    RTC_CHECK(!it_timing.sync_delay_ms) << "Decode time already written";
+    it_timing.sync_delay_ms = last_sync_buffer_ms_;
+    it_timing.target_delay_ms = neteq->TargetDelayMs();
+    it_timing.current_delay_ms = neteq->FilteredCurrentDelayMs();
+  }
+  last_sample_rate_hz_ = audio_frame.sample_rate_hz_;
+  ++get_audio_count_;
+}
+
+void NetEqDelayAnalyzer::CreateGraphs(Delays* arrival_delay_ms,
+                                      Delays* corrected_arrival_delay_ms,
+                                      Delays* playout_delay_ms,
+                                      Delays* target_delay_ms) const {
+  if (get_audio_time_ms_.empty()) {
+    return;
+  }
+  // Create nominal_get_audio_time_ms, a vector starting at
+  // get_audio_time_ms_[0] and increasing by 10 for each element.
+  std::vector<int64_t> nominal_get_audio_time_ms(get_audio_time_ms_.size());
+  nominal_get_audio_time_ms[0] = get_audio_time_ms_[0];
+  std::transform(
+      nominal_get_audio_time_ms.begin(), nominal_get_audio_time_ms.end() - 1,
+      nominal_get_audio_time_ms.begin() + 1, [](int64_t& x) { return x + 10; });
+  RTC_DCHECK(
+      std::is_sorted(get_audio_time_ms_.begin(), get_audio_time_ms_.end()));
+
+  std::vector<double> rtp_timestamps_ms;
+  double offset = std::numeric_limits<double>::max();
+  RtpTimestampUnwrapper unwrapper;
+  // This loop traverses data_ and populates rtp_timestamps_ms as well as
+  // calculates the base offset.
+  for (auto& d : data_) {
+    rtp_timestamps_ms.push_back(
+        static_cast<double>(unwrapper.Unwrap(d.first)) /
+        rtc::CheckedDivExact(last_sample_rate_hz_, 1000));
+    offset =
+        std::min(offset, d.second.arrival_time_ms - rtp_timestamps_ms.back());
+  }
+
+  // This loop traverses the data again and populates the graph vectors. The
+  // reason to have two loops and traverse twice is that the offset cannot be
+  // known until the first traversal is done. Meanwhile, the final offset must
+  // be known already at the start of this second loop.
+  size_t i = 0;
+  for (const auto& data : data_) {
+    const double offset_send_time_ms = rtp_timestamps_ms[i++] + offset;
+    const auto& timing = data.second;
+    corrected_arrival_delay_ms->push_back(std::make_pair(
+        timing.arrival_time_ms,
+        LinearInterpolate(timing.arrival_time_ms, get_audio_time_ms_,
+                          nominal_get_audio_time_ms) -
+            offset_send_time_ms));
+    arrival_delay_ms->push_back(std::make_pair(
+        timing.arrival_time_ms, timing.arrival_time_ms - offset_send_time_ms));
+
+    if (timing.decode_get_audio_count) {
+      // This packet was decoded.
+      RTC_DCHECK(timing.sync_delay_ms);
+      const int64_t get_audio_time =
+          *timing.decode_get_audio_count * 10 + get_audio_time_ms_[0];
+      const float playout_ms =
+          get_audio_time + *timing.sync_delay_ms - offset_send_time_ms;
+      playout_delay_ms->push_back(std::make_pair(get_audio_time, playout_ms));
+      RTC_DCHECK(timing.target_delay_ms);
+      RTC_DCHECK(timing.current_delay_ms);
+      const float target =
+          playout_ms - *timing.current_delay_ms + *timing.target_delay_ms;
+      target_delay_ms->push_back(std::make_pair(get_audio_time, target));
+    }
+  }
+}
+
+void NetEqDelayAnalyzer::CreateMatlabScript(
+    absl::string_view script_name) const {
+  Delays arrival_delay_ms;
+  Delays corrected_arrival_delay_ms;
+  Delays playout_delay_ms;
+  Delays target_delay_ms;
+  CreateGraphs(&arrival_delay_ms, &corrected_arrival_delay_ms,
+               &playout_delay_ms, &target_delay_ms);
+
+  // Maybe better to find the actually smallest timestamp, to surely avoid
+  // x-axis starting from negative.
+  const int64_t ref_time_ms = arrival_delay_ms.front().first;
+
+  // Create an output file stream to Matlab script file.
+  std::ofstream output(std::string{script_name});
+
+  PrintDelays(corrected_arrival_delay_ms, ref_time_ms, kArrivalDelayX,
+              kArrivalDelayY, output, ";");
+
+  // PrintDelays(corrected_arrival_delay_x, kCorrectedArrivalDelayX,
+  // kCorrectedArrivalDelayY, output);
+
+  PrintDelays(playout_delay_ms, ref_time_ms, kPlayoutDelayX, kPlayoutDelayY,
+              output, ";");
+
+  PrintDelays(target_delay_ms, ref_time_ms, kTargetDelayX, kTargetDelayY,
+              output, ";");
+
+  output << "h=plot(" << kArrivalDelayX << ", " << kArrivalDelayY << ", "
+         << kTargetDelayX << ", " << kTargetDelayY << ", 'g.', "
+         << kPlayoutDelayX << ", " << kPlayoutDelayY << ");" << std::endl;
+  output << "set(h(1),'color',0.75*[1 1 1]);" << std::endl;
+  output << "set(h(2),'markersize',6);" << std::endl;
+  output << "set(h(3),'linew',1.5);" << std::endl;
+  output << "ax1=axis;" << std::endl;
+  output << "axis tight" << std::endl;
+  output << "ax2=axis;" << std::endl;
+  output << "axis([ax2(1:3) ax1(4)])" << std::endl;
+  output << "xlabel('time [s]');" << std::endl;
+  output << "ylabel('relative delay [ms]');" << std::endl;
+  if (!ssrcs_.empty()) {
+    auto ssrc_it = ssrcs_.cbegin();
+    output << "title('SSRC: 0x" << std::hex << static_cast<int64_t>(*ssrc_it++);
+    while (ssrc_it != ssrcs_.end()) {
+      output << ", 0x" << std::hex << static_cast<int64_t>(*ssrc_it++);
+    }
+    output << std::dec;
+    auto pt_it = payload_types_.cbegin();
+    output << "; Payload Types: " << *pt_it++;
+    while (pt_it != payload_types_.end()) {
+      output << ", " << *pt_it++;
+    }
+    output << "');" << std::endl;
+  }
+}
+
+void NetEqDelayAnalyzer::CreatePythonScript(
+    absl::string_view script_name) const {
+  Delays arrival_delay_ms;
+  Delays corrected_arrival_delay_ms;
+  Delays playout_delay_ms;
+  Delays target_delay_ms;
+  CreateGraphs(&arrival_delay_ms, &corrected_arrival_delay_ms,
+               &playout_delay_ms, &target_delay_ms);
+
+  // Maybe better to find the actually smallest timestamp, to surely avoid
+  // x-axis starting from negative.
+  const int64_t ref_time_ms = arrival_delay_ms.front().first;
+
+  // Create an output file stream to the python script file.
+  std::ofstream output(std::string{script_name});
+
+  // Necessary includes
+  output << "import numpy as np" << std::endl;
+  output << "import matplotlib.pyplot as plt" << std::endl;
+
+  PrintDelays(corrected_arrival_delay_ms, ref_time_ms, kArrivalDelayX,
+              kArrivalDelayY, output);
+
+  // PrintDelays(corrected_arrival_delay_x, kCorrectedArrivalDelayX,
+  // kCorrectedArrivalDelayY, output);
+
+  PrintDelays(playout_delay_ms, ref_time_ms, kPlayoutDelayX, kPlayoutDelayY,
+              output);
+
+  PrintDelays(target_delay_ms, ref_time_ms, kTargetDelayX, kTargetDelayY,
+              output);
+
+  output << "if __name__ == '__main__':" << std::endl;
+  output << "  h=plt.plot(" << kArrivalDelayX << ", " << kArrivalDelayY << ", "
+         << kTargetDelayX << ", " << kTargetDelayY << ", 'g.', "
+         << kPlayoutDelayX << ", " << kPlayoutDelayY << ")" << std::endl;
+  output << "  plt.setp(h[0],'color',[.75, .75, .75])" << std::endl;
+  output << "  plt.setp(h[1],'markersize',6)" << std::endl;
+  output << "  plt.setp(h[2],'linewidth',1.5)" << std::endl;
+  output << "  plt.axis('tight')" << std::endl;
+  output << "  plt.xlabel('time [s]')" << std::endl;
+  output << "  plt.ylabel('relative delay [ms]')" << std::endl;
+  if (!ssrcs_.empty()) {
+    auto ssrc_it = ssrcs_.cbegin();
+    output << "  plt.legend((\"arrival delay\", \"target delay\", \"playout "
+              "delay\"))"
+           << std::endl;
+    output << "  plt.title('SSRC: 0x" << std::hex
+           << static_cast<int64_t>(*ssrc_it++);
+    while (ssrc_it != ssrcs_.end()) {
+      output << ", 0x" << std::hex << static_cast<int64_t>(*ssrc_it++);
+    }
+    output << std::dec;
+    auto pt_it = payload_types_.cbegin();
+    output << "; Payload Types: " << *pt_it++;
+    while (pt_it != payload_types_.end()) {
+      output << ", " << *pt_it++;
+    }
+    output << "')" << std::endl;
+  }
+  output << "  plt.show()" << std::endl;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
new file mode 100644
index 0000000000..ffcba5843f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
+
+#include <map>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqDelayAnalyzer : public test::NetEqPostInsertPacket,
+                           public test::NetEqGetAudioCallback {
+ public:
+  void AfterInsertPacket(const test::NetEqInput::PacketData& packet,
+                         NetEq* neteq) override;
+
+  void BeforeGetAudio(NetEq* neteq) override;
+
+  void AfterGetAudio(int64_t time_now_ms,
+                     const AudioFrame& audio_frame,
+                     bool muted,
+                     NetEq* neteq) override;
+
+  using Delays = std::vector<std::pair<int64_t, float>>;
+  void CreateGraphs(Delays* arrival_delay_ms,
+                    Delays* corrected_arrival_delay_ms,
+                    Delays* playout_delay_ms,
+                    Delays* target_delay_ms) const;
+
+  // Creates a matlab script with file name script_name. When executed in
+  // Matlab, the script will generate graphs with the same timing information
+  // as provided by CreateGraphs.
+  void CreateMatlabScript(absl::string_view script_name) const;
+
+  // Creates a python script with file name `script_name`. When executed in
+  // Python, the script will generate graphs with the same timing information
+  // as provided by CreateGraphs.
+  void CreatePythonScript(absl::string_view script_name) const;
+
+ private:
+  struct TimingData {
+    explicit TimingData(int64_t at) : arrival_time_ms(at) {}
+    int64_t arrival_time_ms;
+    absl::optional<int64_t> decode_get_audio_count;
+    absl::optional<int64_t> sync_delay_ms;
+    absl::optional<int> target_delay_ms;
+    absl::optional<int> current_delay_ms;
+  };
+  std::map<uint32_t, TimingData> data_;
+  std::vector<int64_t> get_audio_time_ms_;
+  size_t get_audio_count_ = 0;
+  size_t last_sync_buffer_ms_ = 0;
+  int last_sample_rate_hz_ = 0;
+  std::set<uint32_t> ssrcs_;
+  std::set<int> payload_types_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
new file mode 100644
index 0000000000..0c1f27799a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_event_log_input.h"
+
+#include <limits>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqEventLogInput* NetEqEventLogInput::CreateFromFile(
+    absl::string_view file_name,
+    absl::optional<uint32_t> ssrc_filter) {
+  auto event_log_src =
+      RtcEventLogSource::CreateFromFile(file_name, ssrc_filter);
+  if (!event_log_src) {
+    return nullptr;
+  }
+  return new NetEqEventLogInput(std::move(event_log_src));
+}
+
+NetEqEventLogInput* NetEqEventLogInput::CreateFromString(
+    absl::string_view file_contents,
+    absl::optional<uint32_t> ssrc_filter) {
+  auto event_log_src =
+      RtcEventLogSource::CreateFromString(file_contents, ssrc_filter);
+  if (!event_log_src) {
+    return nullptr;
+  }
+  return new NetEqEventLogInput(std::move(event_log_src));
+}
+
+absl::optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
+  return next_output_event_ms_;
+}
+
+void NetEqEventLogInput::AdvanceOutputEvent() {
+  next_output_event_ms_ = source_->NextAudioOutputEventMs();
+  if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
+    next_output_event_ms_ = absl::nullopt;
+  }
+}
+
+PacketSource* NetEqEventLogInput::source() {
+  return source_.get();
+}
+
+NetEqEventLogInput::NetEqEventLogInput(
+    std::unique_ptr<RtcEventLogSource> source)
+    : source_(std::move(source)) {
+  LoadNextPacket();
+  AdvanceOutputEvent();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h
new file mode 100644
index 0000000000..c947ee1fc0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+namespace test {
+
+class RtcEventLogSource;
+
+// Implementation of NetEqPacketSourceInput to be used with an
+// RtcEventLogSource.
+class NetEqEventLogInput final : public NetEqPacketSourceInput {
+ public:
+  static NetEqEventLogInput* CreateFromFile(
+      absl::string_view file_name,
+      absl::optional<uint32_t> ssrc_filter);
+  static NetEqEventLogInput* CreateFromString(
+      absl::string_view file_contents,
+      absl::optional<uint32_t> ssrc_filter);
+
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  void AdvanceOutputEvent() override;
+
+ protected:
+  PacketSource* source() override;
+
+ private:
+  NetEqEventLogInput(std::unique_ptr<RtcEventLogSource> source);
+  std::unique_ptr<RtcEventLogSource> source_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.cc
new file mode 100644
index 0000000000..de416348f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqInput::PacketData::PacketData() = default;
+NetEqInput::PacketData::~PacketData() = default;
+
+std::string NetEqInput::PacketData::ToString() const {
+  rtc::StringBuilder ss;
+  ss << "{"
+        "time_ms: "
+     << static_cast<int64_t>(time_ms)
+     << ", "
+        "header: {"
+        "pt: "
+     << static_cast<int>(header.payloadType)
+     << ", "
+        "sn: "
+     << header.sequenceNumber
+     << ", "
+        "ts: "
+     << header.timestamp
+     << ", "
+        "ssrc: "
+     << header.ssrc
+     << "}, "
+        "payload bytes: "
+     << payload.size() << "}";
+  return ss.Release();
+}
+
+TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
+                                             int64_t duration_ms)
+    : input_(std::move(input)),
+      start_time_ms_(input_->NextEventTime()),
+      duration_ms_(duration_ms) {}
+
+TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;
+
+absl::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
+  return ended_ ? absl::nullopt : input_->NextPacketTime();
+}
+
+absl::optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
+  return ended_ ? absl::nullopt : input_->NextOutputEventTime();
+}
+
+std::unique_ptr<NetEqInput::PacketData> TimeLimitedNetEqInput::PopPacket() {
+  if (ended_) {
+    return std::unique_ptr<PacketData>();
+  }
+  auto packet = input_->PopPacket();
+  MaybeSetEnded();
+  return packet;
+}
+
+void TimeLimitedNetEqInput::AdvanceOutputEvent() {
+  if (!ended_) {
+    input_->AdvanceOutputEvent();
+    MaybeSetEnded();
+  }
+}
+
+bool TimeLimitedNetEqInput::ended() const {
+  return ended_ || input_->ended();
+}
+
+absl::optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
+  return ended_ ? absl::nullopt : input_->NextHeader();
+}
+
+void TimeLimitedNetEqInput::MaybeSetEnded() {
+  if (NextEventTime() && start_time_ms_ &&
+      *NextEventTime() - *start_time_ms_ > duration_ms_) {
+    ended_ = true;
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.h
new file mode 100644
index 0000000000..3a66264043
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.h
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+
+#include <algorithm>
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for input to the NetEqTest class.
+class NetEqInput {
+ public:
+  struct PacketData {
+    PacketData();
+    ~PacketData();
+    std::string ToString() const;
+
+    RTPHeader header;
+    rtc::Buffer payload;
+    int64_t time_ms;
+  };
+
+  virtual ~NetEqInput() = default;
+
+  // Returns at what time (in ms) NetEq::InsertPacket should be called next, or
+  // empty if the source is out of packets.
+  virtual absl::optional<int64_t> NextPacketTime() const = 0;
+
+  // Returns at what time (in ms) NetEq::GetAudio should be called next, or
+  // empty if no more output events are available.
+  virtual absl::optional<int64_t> NextOutputEventTime() const = 0;
+
+  // Returns the time (in ms) for the next event from either NextPacketTime()
+  // or NextOutputEventTime(), or empty if both are out of events.
+  absl::optional<int64_t> NextEventTime() const {
+    const auto a = NextPacketTime();
+    const auto b = NextOutputEventTime();
+    // Return the minimum of non-empty `a` and `b`, or empty if both are empty.
+    if (a) {
+      return b ? std::min(*a, *b) : a;
+    }
+    return b ? b : absl::nullopt;
+  }
+
+  // Returns the next packet to be inserted into NetEq. The packet following the
+  // returned one is pre-fetched in the NetEqInput object, such that future
+  // calls to NextPacketTime() or NextHeader() will return information from that
+  // packet.
+  virtual std::unique_ptr<PacketData> PopPacket() = 0;
+
+  // Move to the next output event. This will make NextOutputEventTime() return
+  // a new value (potentially the same if several output events share the same
+  // time).
+  virtual void AdvanceOutputEvent() = 0;
+
+  // Returns true if the source has come to an end. An implementation must
+  // eventually return true from this method, or the test will end up in an
+  // infinite loop.
+  virtual bool ended() const = 0;
+
+  // Returns the RTP header for the next packet, i.e., the packet that will be
+  // delivered next by PopPacket().
+  virtual absl::optional<RTPHeader> NextHeader() const = 0;
+};
+
+// Wrapper class to impose a time limit on a NetEqInput object, typically
+// another time limit than what the object itself provides. For example, an
+// input taken from a file can be cut shorter by wrapping it in this class.
+class TimeLimitedNetEqInput : public NetEqInput {
+ public:
+  TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
+  ~TimeLimitedNetEqInput() override;
+  absl::optional<int64_t> NextPacketTime() const override;
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  void AdvanceOutputEvent() override;
+  bool ended() const override;
+  absl::optional<RTPHeader> NextHeader() const override;
+
+ private:
+  void MaybeSetEnded();
+
+  std::unique_ptr<NetEqInput> input_;
+  const absl::optional<int64_t> start_time_ms_;
+  const int64_t duration_ms_;
+  bool ended_ = false;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
new file mode 100644
index 0000000000..55a5653238
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {}
+
+absl::optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
+  return packet_
+             ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
+             : absl::nullopt;
+}
+
+absl::optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
+  return packet_ ? absl::optional<RTPHeader>(packet_->header()) : absl::nullopt;
+}
+
+void NetEqPacketSourceInput::LoadNextPacket() {
+  packet_ = source()->NextPacket();
+}
+
+std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() {
+  if (!packet_) {
+    return std::unique_ptr<PacketData>();
+  }
+  std::unique_ptr<PacketData> packet_data(new PacketData);
+  packet_data->header = packet_->header();
+  if (packet_->payload_length_bytes() == 0 &&
+      packet_->virtual_payload_length_bytes() > 0) {
+    // This is a header-only "dummy" packet. Set the payload to all zeros, with
+    // length according to the virtual length.
+    packet_data->payload.SetSize(packet_->virtual_payload_length_bytes());
+    std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
+  } else {
+    packet_data->payload.SetData(packet_->payload(),
+                                 packet_->payload_length_bytes());
+  }
+  packet_data->time_ms = packet_->time_ms();
+
+  LoadNextPacket();
+
+  return packet_data;
+}
+
+NetEqRtpDumpInput::NetEqRtpDumpInput(absl::string_view file_name,
+                                     const RtpHeaderExtensionMap& hdr_ext_map,
+                                     absl::optional<uint32_t> ssrc_filter)
+    : source_(RtpFileSource::Create(file_name, ssrc_filter)) {
+  for (const auto& ext_pair : hdr_ext_map) {
+    source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
+  }
+  LoadNextPacket();
+}
+
+absl::optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
+  return next_output_event_ms_;
+}
+
+void NetEqRtpDumpInput::AdvanceOutputEvent() {
+  if (next_output_event_ms_) {
+    *next_output_event_ms_ += kOutputPeriodMs;
+  }
+  if (!NextPacketTime()) {
+    next_output_event_ms_ = absl::nullopt;
+  }
+}
+
+PacketSource* NetEqRtpDumpInput::source() {
+  return source_.get();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
new file mode 100644
index 0000000000..407fa491b1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+namespace test {
+
+class RtpFileSource;
+
+// An adapter class to dress up a PacketSource object as a NetEqInput.
+class NetEqPacketSourceInput : public NetEqInput {
+ public:
+  using RtpHeaderExtensionMap = std::map<int, webrtc::RTPExtensionType>;
+
+  NetEqPacketSourceInput();
+  absl::optional<int64_t> NextPacketTime() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  absl::optional<RTPHeader> NextHeader() const override;
+  bool ended() const override { return !next_output_event_ms_; }
+
+ protected:
+  virtual PacketSource* source() = 0;
+  void LoadNextPacket();
+
+  absl::optional<int64_t> next_output_event_ms_;
+
+ private:
+  std::unique_ptr<Packet> packet_;
+};
+
+// Implementation of NetEqPacketSourceInput to be used with an RtpFileSource.
+class NetEqRtpDumpInput final : public NetEqPacketSourceInput {
+ public:
+  NetEqRtpDumpInput(absl::string_view file_name,
+                    const RtpHeaderExtensionMap& hdr_ext_map,
+                    absl::optional<uint32_t> ssrc_filter);
+
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  void AdvanceOutputEvent() override;
+
+ protected:
+  PacketSource* source() override;
+
+ private:
+  static constexpr int64_t kOutputPeriodMs = 10;
+
+  std::unique_ptr<RtpFileSource> source_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
new file mode 100644
index 0000000000..ccaa87b5e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/clock.h"
+#include "test/testsupport/file_utils.h"
+
+using webrtc::NetEq;
+using webrtc::test::AudioLoop;
+using webrtc::test::RtpGenerator;
+
+namespace webrtc {
+namespace test {
+
+int64_t NetEqPerformanceTest::Run(int runtime_ms,
+                                  int lossrate,
+                                  double drift_factor) {
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const int kSampRateHz = 32000;
+  const std::string kDecoderName = "pcm16-swb32";
+  const int kPayloadType = 95;
+
+  // Initialize NetEq instance.
+  NetEq::Config config;
+  config.sample_rate_hz = kSampRateHz;
+  webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
+  auto audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
+  auto neteq =
+      DefaultNetEqFactory().CreateNetEq(config, audio_decoder_factory, clock);
+  // Register decoder in `neteq`.
+  if (!neteq->RegisterPayloadType(kPayloadType,
+                                  SdpAudioFormat("l16", kSampRateHz, 1)))
+    return -1;
+
+  // Set up AudioLoop object.
+  AudioLoop audio_loop;
+  const size_t kMaxLoopLengthSamples = kSampRateHz * 10;  // 10 second loop.
+  const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000;  // 60 ms.
+  if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                       kInputBlockSizeSamples))
+    return -1;
+
+  int32_t time_now_ms = 0;
+
+  // Get first input packet.
+  RTPHeader rtp_header;
+  RtpGenerator rtp_gen(kSampRateHz / 1000);
+  // Start with positive drift first half of simulation.
+  rtp_gen.set_drift_factor(drift_factor);
+  bool drift_flipped = false;
+  int32_t packet_input_time_ms =
+      rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
+  auto input_samples = audio_loop.GetNextBlock();
+  if (input_samples.empty())
+    exit(1);
+  uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
+  size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                           input_samples.size(), input_payload);
+  RTC_CHECK_EQ(sizeof(input_payload), payload_len);
+
+  // Main loop.
+  int64_t start_time_ms = clock->TimeInMilliseconds();
+  AudioFrame out_frame;
+  while (time_now_ms < runtime_ms) {
+    while (packet_input_time_ms <= time_now_ms) {
+      // Drop every N packets, where N = FLAG_lossrate.
+      bool lost = false;
+      if (lossrate > 0) {
+        lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0;
+      }
+      if (!lost) {
+        // Insert packet.
+        int error = neteq->InsertPacket(rtp_header, input_payload);
+        if (error != NetEq::kOK)
+          return -1;
+      }
+
+      // Get next packet.
+      packet_input_time_ms = rtp_gen.GetRtpHeader(
+          kPayloadType, kInputBlockSizeSamples, &rtp_header);
+      input_samples = audio_loop.GetNextBlock();
+      if (input_samples.empty())
+        return -1;
+      payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                        input_samples.size(), input_payload);
+      RTC_DCHECK_EQ(payload_len, kInputBlockSizeSamples * sizeof(int16_t));
+    }
+
+    // Get output audio, but don't do anything with it.
+    bool muted;
+    int error = neteq->GetAudio(&out_frame, &muted);
+    RTC_CHECK(!muted);
+    if (error != NetEq::kOK)
+      return -1;
+
+    RTC_DCHECK_EQ(out_frame.samples_per_channel_, (kSampRateHz * 10) / 1000);
+
+    static const int kOutputBlockSizeMs = 10;
+    time_now_ms += kOutputBlockSizeMs;
+    if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
+      // Apply negative drift second half of simulation.
+      rtp_gen.set_drift_factor(-drift_factor);
+      drift_flipped = true;
+    }
+  }
+  int64_t end_time_ms = clock->TimeInMilliseconds();
+  return end_time_ms - start_time_ms;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
new file mode 100644
index 0000000000..b5b4d91577
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+namespace test {
+
+class NetEqPerformanceTest {
+ public:
+  // Runs a performance test with parameters as follows:
+  //   `runtime_ms`: the simulation time, i.e., the duration of the audio data.
+  //   `lossrate`: drop one out of `lossrate` packets, e.g., one out of 10.
+  //   `drift_factor`: clock drift in [0, 1].
+  // Returns the runtime in ms.
+  static int64_t Run(int runtime_ms, int lossrate, double drift_factor);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
new file mode 100644
index 0000000000..18e6091f93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -0,0 +1,482 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+
+#include <stdio.h>
+
+#include <cmath>
+
+#include "absl/flags/flag.h"
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "modules/audio_coding/neteq/tools/output_audio_file.h"
+#include "modules/audio_coding/neteq/tools/output_wav_file.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_encode.h"
+#include "system_wrappers/include/clock.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(std::string,
+          in_filename,
+          "audio_coding/speech_mono_16kHz.pcm",
+          "Path of the input file (relative to the resources/ directory) for "
+          "input audio (specify sample rate with --input_sample_rate, "
+          "and channels with --channels).");
+
+ABSL_FLAG(int, input_sample_rate, 16000, "Sample rate of input file in Hz.");
+
+ABSL_FLAG(int, channels, 1, "Number of channels in input audio.");
+
+ABSL_FLAG(std::string,
+          out_filename,
+          "neteq_quality_test_out.pcm",
+          "Name of output audio file, which will be saved in " +
+              ::webrtc::test::OutputPath());
+
+ABSL_FLAG(
+    int,
+    runtime_ms,
+    10000,
+    "Simulated runtime (milliseconds). -1 will consume the complete file.");
+
+ABSL_FLAG(int, packet_loss_rate, 10, "Percentile of packet loss.");
+
+ABSL_FLAG(int,
+          random_loss_mode,
+          ::webrtc::test::kUniformLoss,
+          "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot "
+          "loss, 3--fixed loss.");
+
+ABSL_FLAG(int,
+          burst_length,
+          30,
+          "Burst length in milliseconds, only valid for Gilbert Elliot loss.");
+
+ABSL_FLAG(float, drift_factor, 0.0, "Time drift factor.");
+
+ABSL_FLAG(int,
+          preload_packets,
+          1,
+          "Preload the buffer with this many packets.");
+
+ABSL_FLAG(std::string,
+          loss_events,
+          "",
+          "List of loss events time and duration separated by comma: "
+          "<first_event_time> <first_event_duration>, <second_event_time> "
+          "<second_event_duration>, ...");
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+std::unique_ptr<NetEq> CreateNetEq(
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+  return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
+}
+
+const std::string& GetInFilenamePath(absl::string_view file_name) {
+  std::vector<absl::string_view> name_parts = rtc::split(file_name, '.');
+  RTC_CHECK_EQ(name_parts.size(), 2);
+  static const std::string path =
+      ::webrtc::test::ResourcePath(name_parts[0], name_parts[1]);
+  return path;
+}
+
+const std::string& GetOutFilenamePath(absl::string_view file_name) {
+  static const std::string path =
+      ::webrtc::test::OutputPath() + std::string(file_name);
+  return path;
+}
+
+}  // namespace
+
+const uint8_t kPayloadType = 95;
+const int kOutputSizeMs = 10;
+const int kInitSeed = 0x12345678;
+const int kPacketLossTimeUnitMs = 10;
+
+// Common validator for file names.
+static bool ValidateFilename(absl::string_view value, bool is_output) {
+  if (!is_output) {
+    RTC_CHECK_NE(value.substr(value.find_last_of('.') + 1), "wav")
+        << "WAV file input is not supported";
+  }
+  FILE* fid = is_output ? fopen(std::string(value).c_str(), "wb")
+                        : fopen(std::string(value).c_str(), "rb");
+  if (fid == nullptr)
+    return false;
+  fclose(fid);
+  return true;
+}
+
+// ProbTrans00Solver() is to calculate the transition probability from no-loss
+// state to itself in a modified Gilbert Elliot packet loss model. The result is
+// to achieve the target packet loss rate `loss_rate`, when a packet is not
+// lost only if all `units` drawings within the duration of the packet result in
+// no-loss.
+static double ProbTrans00Solver(int units,
+                                double loss_rate,
+                                double prob_trans_10) {
+  if (units == 1)
+    return prob_trans_10 / (1.0f - loss_rate) - prob_trans_10;
+  // 0 == prob_trans_00 ^ (units - 1) + (1 - loss_rate) / prob_trans_10 *
+  //     prob_trans_00 - (1 - loss_rate) * (1 + 1 / prob_trans_10).
+  // There is a unique solution between 0.0 and 1.0, due to the monotonicity and
+  // an opposite sign at 0.0 and 1.0.
+  // For simplicity, we reformulate the equation as
+  //     f(x) = x ^ (units - 1) + a x + b.
+  // Its derivative is
+  //     f'(x) = (units - 1) x ^ (units - 2) + a.
+  // The derivative is strictly greater than 0 when x is between 0 and 1.
+  // We use Newton's method to solve the equation, iteration is
+  //     x(k+1) = x(k) - f(x) / f'(x);
+  const double kPrecision = 0.001f;
+  const int kIterations = 100;
+  const double a = (1.0f - loss_rate) / prob_trans_10;
+  const double b = (loss_rate - 1.0f) * (1.0f + 1.0f / prob_trans_10);
+  double x = 0.0;  // Starting point;
+  double f = b;
+  double f_p;
+  int iter = 0;
+  while ((f >= kPrecision || f <= -kPrecision) && iter < kIterations) {
+    f_p = (units - 1.0f) * std::pow(x, units - 2) + a;
+    x -= f / f_p;
+    if (x > 1.0f) {
+      x = 1.0f;
+    } else if (x < 0.0f) {
+      x = 0.0f;
+    }
+    f = std::pow(x, units - 1) + a * x + b;
+    iter++;
+  }
+  return x;
+}
+
+NetEqQualityTest::NetEqQualityTest(
+    int block_duration_ms,
+    int in_sampling_khz,
+    int out_sampling_khz,
+    const SdpAudioFormat& format,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
+    : audio_format_(format),
+      channels_(absl::GetFlag(FLAGS_channels)),
+      decoded_time_ms_(0),
+      decodable_time_ms_(0),
+      drift_factor_(absl::GetFlag(FLAGS_drift_factor)),
+      packet_loss_rate_(absl::GetFlag(FLAGS_packet_loss_rate)),
+      block_duration_ms_(block_duration_ms),
+      in_sampling_khz_(in_sampling_khz),
+      out_sampling_khz_(out_sampling_khz),
+      in_size_samples_(
+          static_cast<size_t>(in_sampling_khz_ * block_duration_ms_)),
+      payload_size_bytes_(0),
+      max_payload_bytes_(0),
+      in_file_(new ResampleInputAudioFile(
+          GetInFilenamePath(absl::GetFlag(FLAGS_in_filename)),
+          absl::GetFlag(FLAGS_input_sample_rate),
+          in_sampling_khz * 1000,
+          absl::GetFlag(FLAGS_runtime_ms) > 0)),
+      rtp_generator_(
+          new RtpGenerator(in_sampling_khz_, 0, 0, decodable_time_ms_)),
+      total_payload_size_bytes_(0) {
+  // Flag validation
+  RTC_CHECK(ValidateFilename(
+      GetInFilenamePath(absl::GetFlag(FLAGS_in_filename)), false))
+      << "Invalid input filename.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_input_sample_rate) == 8000 ||
+            absl::GetFlag(FLAGS_input_sample_rate) == 16000 ||
+            absl::GetFlag(FLAGS_input_sample_rate) == 32000 ||
+            absl::GetFlag(FLAGS_input_sample_rate) == 48000)
+      << "Invalid sample rate should be 8000, 16000, 32000 or 48000 Hz.";
+
+  RTC_CHECK_EQ(absl::GetFlag(FLAGS_channels), 1)
+      << "Invalid number of channels, current support only 1.";
+
+  RTC_CHECK(ValidateFilename(
+      GetOutFilenamePath(absl::GetFlag(FLAGS_out_filename)), true))
+      << "Invalid output filename.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_packet_loss_rate) >= 0 &&
+            absl::GetFlag(FLAGS_packet_loss_rate) <= 100)
+      << "Invalid packet loss percentile, should be between 0 and 100.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_random_loss_mode) >= 0 &&
+            absl::GetFlag(FLAGS_random_loss_mode) < kLastLossMode)
+      << "Invalid random packet loss mode, should be between 0 and "
+      << kLastLossMode - 1 << ".";
+
+  RTC_CHECK_GE(absl::GetFlag(FLAGS_burst_length), kPacketLossTimeUnitMs)
+      << "Invalid burst length, should be greater than or equal to "
+      << kPacketLossTimeUnitMs << " ms.";
+
+  RTC_CHECK_GT(absl::GetFlag(FLAGS_drift_factor), -0.1)
+      << "Invalid drift factor, should be greater than -0.1.";
+
+  RTC_CHECK_GE(absl::GetFlag(FLAGS_preload_packets), 0)
+      << "Invalid number of packets to preload; must be non-negative.";
+
+  const std::string out_filename =
+      GetOutFilenamePath(absl::GetFlag(FLAGS_out_filename));
+  const std::string log_filename = out_filename + ".log";
+  log_file_.open(log_filename.c_str(), std::ofstream::out);
+  RTC_CHECK(log_file_.is_open());
+
+  if (out_filename.size() >= 4 &&
+      out_filename.substr(out_filename.size() - 4) == ".wav") {
+    // Open a wav file.
+    output_.reset(
+        new webrtc::test::OutputWavFile(out_filename, 1000 * out_sampling_khz));
+  } else {
+    // Open a pcm file.
+    output_.reset(new webrtc::test::OutputAudioFile(out_filename));
+  }
+
+  NetEq::Config config;
+  config.sample_rate_hz = out_sampling_khz_ * 1000;
+  neteq_ = CreateNetEq(config, Clock::GetRealTimeClock(), decoder_factory);
+  max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
+  in_data_.reset(new int16_t[in_size_samples_ * channels_]);
+}
+
+NetEqQualityTest::~NetEqQualityTest() {
+  log_file_.close();
+}
+
+bool NoLoss::Lost(int now_ms) {
+  return false;
+}
+
+UniformLoss::UniformLoss(double loss_rate) : loss_rate_(loss_rate) {}
+
+bool UniformLoss::Lost(int now_ms) {
+  int drop_this = rand();
+  return (drop_this < loss_rate_ * RAND_MAX);
+}
+
+GilbertElliotLoss::GilbertElliotLoss(double prob_trans_11, double prob_trans_01)
+    : prob_trans_11_(prob_trans_11),
+      prob_trans_01_(prob_trans_01),
+      lost_last_(false),
+      uniform_loss_model_(new UniformLoss(0)) {}
+
+GilbertElliotLoss::~GilbertElliotLoss() {}
+
+bool GilbertElliotLoss::Lost(int now_ms) {
+  // Simulate bursty channel (Gilbert model).
+  // (1st order) Markov chain model with memory of the previous/last
+  // packet state (lost or received).
+  if (lost_last_) {
+    // Previous packet was not received.
+    uniform_loss_model_->set_loss_rate(prob_trans_11_);
+    return lost_last_ = uniform_loss_model_->Lost(now_ms);
+  } else {
+    uniform_loss_model_->set_loss_rate(prob_trans_01_);
+    return lost_last_ = uniform_loss_model_->Lost(now_ms);
+  }
+}
+
+FixedLossModel::FixedLossModel(
+    std::set<FixedLossEvent, FixedLossEventCmp> loss_events)
+    : loss_events_(loss_events) {
+  loss_events_it_ = loss_events_.begin();
+}
+
+FixedLossModel::~FixedLossModel() {}
+
+bool FixedLossModel::Lost(int now_ms) {
+  if (loss_events_it_ != loss_events_.end() &&
+      now_ms > loss_events_it_->start_ms) {
+    if (now_ms <= loss_events_it_->start_ms + loss_events_it_->duration_ms) {
+      return true;
+    } else {
+      ++loss_events_it_;
+      return false;
+    }
+  }
+  return false;
+}
+
+void NetEqQualityTest::SetUp() {
+  ASSERT_TRUE(neteq_->RegisterPayloadType(kPayloadType, audio_format_));
+  rtp_generator_->set_drift_factor(drift_factor_);
+
+  int units = block_duration_ms_ / kPacketLossTimeUnitMs;
+  switch (absl::GetFlag(FLAGS_random_loss_mode)) {
+    case kUniformLoss: {
+      // `unit_loss_rate` is the packet loss rate for each unit time interval
+      // (kPacketLossTimeUnitMs). Since a packet loss event is generated if any
+      // of |block_duration_ms_ / kPacketLossTimeUnitMs| unit time intervals of
+      // a full packet duration is drawn with a loss, `unit_loss_rate` fulfills
+      // (1 - unit_loss_rate) ^ (block_duration_ms_ / kPacketLossTimeUnitMs) ==
+      // 1 - packet_loss_rate.
+      double unit_loss_rate =
+          (1.0 - std::pow(1.0 - 0.01 * packet_loss_rate_, 1.0 / units));
+      loss_model_.reset(new UniformLoss(unit_loss_rate));
+      break;
+    }
+    case kGilbertElliotLoss: {
+      // `FLAGS_burst_length` should be integer times of kPacketLossTimeUnitMs.
+      ASSERT_EQ(0, absl::GetFlag(FLAGS_burst_length) % kPacketLossTimeUnitMs);
+
+      // We do not allow 100 percent packet loss in Gilbert Elliot model, which
+      // makes no sense.
+      ASSERT_GT(100, packet_loss_rate_);
+
+      // To guarantee the overall packet loss rate, transition probabilities
+      // need to satisfy:
+      // pi_0 * (1 - prob_trans_01_) ^ units +
+      //     pi_1 * prob_trans_10_ ^ (units - 1) == 1 - loss_rate
+      // pi_0 = prob_trans_10 / (prob_trans_10 + prob_trans_01_)
+      //     is the stationary state probability of no-loss
+      // pi_1 = prob_trans_01_ / (prob_trans_10 + prob_trans_01_)
+      //     is the stationary state probability of loss
+      // After a derivation prob_trans_00 should satisfy:
+      // prob_trans_00 ^ (units - 1) = (loss_rate - 1) / prob_trans_10 *
+      //     prob_trans_00 + (1 - loss_rate) * (1 + 1 / prob_trans_10).
+      double loss_rate = 0.01f * packet_loss_rate_;
+      double prob_trans_10 =
+          1.0f * kPacketLossTimeUnitMs / absl::GetFlag(FLAGS_burst_length);
+      double prob_trans_00 = ProbTrans00Solver(units, loss_rate, prob_trans_10);
+      loss_model_.reset(
+          new GilbertElliotLoss(1.0f - prob_trans_10, 1.0f - prob_trans_00));
+      break;
+    }
+    case kFixedLoss: {
+      std::istringstream loss_events_stream(absl::GetFlag(FLAGS_loss_events));
+      std::string loss_event_string;
+      std::set<FixedLossEvent, FixedLossEventCmp> loss_events;
+      while (std::getline(loss_events_stream, loss_event_string, ',')) {
+        std::vector<int> loss_event_params;
+        std::istringstream loss_event_params_stream(loss_event_string);
+        std::copy(std::istream_iterator<int>(loss_event_params_stream),
+                  std::istream_iterator<int>(),
+                  std::back_inserter(loss_event_params));
+        RTC_CHECK_EQ(loss_event_params.size(), 2);
+        auto result = loss_events.insert(
+            FixedLossEvent(loss_event_params[0], loss_event_params[1]));
+        RTC_CHECK(result.second);
+      }
+      RTC_CHECK_GT(loss_events.size(), 0);
+      loss_model_.reset(new FixedLossModel(loss_events));
+      break;
+    }
+    default: {
+      loss_model_.reset(new NoLoss);
+      break;
+    }
+  }
+
+  // Make sure that the packet loss profile is same for all derived tests.
+  srand(kInitSeed);
+}
+
+std::ofstream& NetEqQualityTest::Log() {
+  return log_file_;
+}
+
+bool NetEqQualityTest::PacketLost() {
+  int cycles = block_duration_ms_ / kPacketLossTimeUnitMs;
+
+  // The loop is to make sure that codecs with different block lengths share the
+  // same packet loss profile.
+  bool lost = false;
+  for (int idx = 0; idx < cycles; idx++) {
+    if (loss_model_->Lost(decoded_time_ms_)) {
+      // The packet will be lost if any of the drawings indicates a loss, but
+      // the loop has to go on to make sure that codecs with different block
+      // lengths keep the same pace.
+      lost = true;
+    }
+  }
+  return lost;
+}
+
+int NetEqQualityTest::Transmit() {
+  int packet_input_time_ms = rtp_generator_->GetRtpHeader(
+      kPayloadType, in_size_samples_, &rtp_header_);
+  Log() << "Packet of size " << payload_size_bytes_ << " bytes, for frame at "
+        << packet_input_time_ms << " ms ";
+  if (payload_size_bytes_ > 0) {
+    if (!PacketLost()) {
+      int ret = neteq_->InsertPacket(
+          rtp_header_,
+          rtc::ArrayView<const uint8_t>(payload_.data(), payload_size_bytes_));
+      if (ret != NetEq::kOK)
+        return -1;
+      Log() << "was sent.";
+    } else {
+      Log() << "was lost.";
+    }
+  }
+  Log() << std::endl;
+  return packet_input_time_ms;
+}
+
+int NetEqQualityTest::DecodeBlock() {
+  bool muted;
+  int ret = neteq_->GetAudio(&out_frame_, &muted);
+  RTC_CHECK(!muted);
+
+  if (ret != NetEq::kOK) {
+    return -1;
+  } else {
+    RTC_DCHECK_EQ(out_frame_.num_channels_, channels_);
+    RTC_DCHECK_EQ(out_frame_.samples_per_channel_,
+                  static_cast<size_t>(kOutputSizeMs * out_sampling_khz_));
+    RTC_CHECK(output_->WriteArray(
+        out_frame_.data(),
+        out_frame_.samples_per_channel_ * out_frame_.num_channels_));
+    return static_cast<int>(out_frame_.samples_per_channel_);
+  }
+}
+
+void NetEqQualityTest::Simulate() {
+  int audio_size_samples;
+  bool end_of_input = false;
+  int runtime_ms = absl::GetFlag(FLAGS_runtime_ms) >= 0
+                       ? absl::GetFlag(FLAGS_runtime_ms)
+                       : INT_MAX;
+
+  while (!end_of_input && decoded_time_ms_ < runtime_ms) {
+    // Preload the buffer if needed.
+    while (decodable_time_ms_ -
+               absl::GetFlag(FLAGS_preload_packets) * block_duration_ms_ <
+           decoded_time_ms_) {
+      if (!in_file_->Read(in_size_samples_ * channels_, &in_data_[0])) {
+        end_of_input = true;
+        ASSERT_TRUE(end_of_input && absl::GetFlag(FLAGS_runtime_ms) < 0);
+        break;
+      }
+      payload_.Clear();
+      payload_size_bytes_ = EncodeBlock(&in_data_[0], in_size_samples_,
+                                        &payload_, max_payload_bytes_);
+      total_payload_size_bytes_ += payload_size_bytes_;
+      decodable_time_ms_ = Transmit() + block_duration_ms_;
+    }
+    audio_size_samples = DecodeBlock();
+    if (audio_size_samples > 0) {
+      decoded_time_ms_ += audio_size_samples / out_sampling_khz_;
+    }
+  }
+  Log() << "Average bit rate was "
+        << 8.0f * total_payload_size_bytes_ / absl::GetFlag(FLAGS_runtime_ms)
+        << " kbps" << std::endl;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
new file mode 100644
index 0000000000..edcb117748
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -0,0 +1,176 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
+
+#include <fstream>
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+enum LossModes {
+  kNoLoss,
+  kUniformLoss,
+  kGilbertElliotLoss,
+  kFixedLoss,
+  kLastLossMode
+};
+
+class LossModel {
+ public:
+  virtual ~LossModel() {}
+  virtual bool Lost(int now_ms) = 0;
+};
+
+class NoLoss : public LossModel {
+ public:
+  bool Lost(int now_ms) override;
+};
+
+class UniformLoss : public LossModel {
+ public:
+  UniformLoss(double loss_rate);
+  bool Lost(int now_ms) override;
+  void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
+
+ private:
+  double loss_rate_;
+};
+
+class GilbertElliotLoss : public LossModel {
+ public:
+  GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
+  ~GilbertElliotLoss() override;
+  bool Lost(int now_ms) override;
+
+ private:
+  // Prob. of losing current packet, when previous packet is lost.
+  double prob_trans_11_;
+  // Prob. of losing current packet, when previous packet is not lost.
+  double prob_trans_01_;
+  bool lost_last_;
+  std::unique_ptr<UniformLoss> uniform_loss_model_;
+};
+
+struct FixedLossEvent {
+  int start_ms;
+  int duration_ms;
+  FixedLossEvent(int start_ms, int duration_ms)
+      : start_ms(start_ms), duration_ms(duration_ms) {}
+};
+
+struct FixedLossEventCmp {
+  bool operator()(const FixedLossEvent& l_event,
+                  const FixedLossEvent& r_event) const {
+    return l_event.start_ms < r_event.start_ms;
+  }
+};
+
+class FixedLossModel : public LossModel {
+ public:
+  FixedLossModel(std::set<FixedLossEvent, FixedLossEventCmp> loss_events);
+  ~FixedLossModel() override;
+  bool Lost(int now_ms) override;
+
+ private:
+  std::set<FixedLossEvent, FixedLossEventCmp> loss_events_;
+  std::set<FixedLossEvent, FixedLossEventCmp>::iterator loss_events_it_;
+};
+
+class NetEqQualityTest : public ::testing::Test {
+ protected:
+  NetEqQualityTest(
+      int block_duration_ms,
+      int in_sampling_khz,
+      int out_sampling_khz,
+      const SdpAudioFormat& format,
+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory =
+          webrtc::CreateBuiltinAudioDecoderFactory());
+  ~NetEqQualityTest() override;
+
+  void SetUp() override;
+
+  // EncodeBlock(...) does the following:
+  // 1. encodes a block of audio, saved in `in_data` and has a length of
+  // `block_size_samples` (samples per channel),
+  // 2. save the bit stream to `payload` of `max_bytes` bytes in size,
+  // 3. returns the length of the payload (in bytes),
+  virtual int EncodeBlock(int16_t* in_data,
+                          size_t block_size_samples,
+                          rtc::Buffer* payload,
+                          size_t max_bytes) = 0;
+
+  // PacketLost(...) determines weather a packet sent at an indicated time gets
+  // lost or not.
+  bool PacketLost();
+
+  // DecodeBlock() decodes a block of audio using the payload stored in
+  // `payload_` with the length of `payload_size_bytes_` (bytes). The decoded
+  // audio is to be stored in `out_data_`.
+  int DecodeBlock();
+
+  // Transmit() uses `rtp_generator_` to generate a packet and passes it to
+  // `neteq_`.
+  int Transmit();
+
+  // Runs encoding / transmitting / decoding.
+  void Simulate();
+
+  // Write to log file. Usage Log() << ...
+  std::ofstream& Log();
+
+  SdpAudioFormat audio_format_;
+  const size_t channels_;
+
+ private:
+  int decoded_time_ms_;
+  int decodable_time_ms_;
+  double drift_factor_;
+  int packet_loss_rate_;
+  const int block_duration_ms_;
+  const int in_sampling_khz_;
+  const int out_sampling_khz_;
+
+  // Number of samples per channel in a frame.
+  const size_t in_size_samples_;
+
+  size_t payload_size_bytes_;
+  size_t max_payload_bytes_;
+
+  std::unique_ptr<InputAudioFile> in_file_;
+  std::unique_ptr<AudioSink> output_;
+  std::ofstream log_file_;
+
+  std::unique_ptr<RtpGenerator> rtp_generator_;
+  std::unique_ptr<NetEq> neteq_;
+  std::unique_ptr<LossModel> loss_model_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  rtc::Buffer payload_;
+  AudioFrame out_frame_;
+  RTPHeader rtp_header_;
+
+  size_t total_payload_size_bytes_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
new file mode 100644
index 0000000000..ffd114ae5b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
+
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqReplacementInput::NetEqReplacementInput(
+    std::unique_ptr<NetEqInput> source,
+    uint8_t replacement_payload_type,
+    const std::set<uint8_t>& comfort_noise_types,
+    const std::set<uint8_t>& forbidden_types)
+    : source_(std::move(source)),
+      replacement_payload_type_(replacement_payload_type),
+      comfort_noise_types_(comfort_noise_types),
+      forbidden_types_(forbidden_types) {
+  RTC_CHECK(source_);
+  packet_ = source_->PopPacket();
+  ReplacePacket();
+}
+
+absl::optional<int64_t> NetEqReplacementInput::NextPacketTime() const {
+  return packet_
+             ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms))
+             : absl::nullopt;
+}
+
+absl::optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const {
+  return source_->NextOutputEventTime();
+}
+
+std::unique_ptr<NetEqInput::PacketData> NetEqReplacementInput::PopPacket() {
+  std::unique_ptr<PacketData> to_return = std::move(packet_);
+  while (true) {
+    packet_ = source_->PopPacket();
+    if (!packet_)
+      break;
+    if (packet_->payload.size() > packet_->header.paddingLength) {
+      // Not padding only. Good to go. Skip this packet otherwise.
+      break;
+    }
+  }
+  ReplacePacket();
+  return to_return;
+}
+
+void NetEqReplacementInput::AdvanceOutputEvent() {
+  source_->AdvanceOutputEvent();
+}
+
+bool NetEqReplacementInput::ended() const {
+  return source_->ended();
+}
+
+absl::optional<RTPHeader> NetEqReplacementInput::NextHeader() const {
+  return source_->NextHeader();
+}
+
+void NetEqReplacementInput::ReplacePacket() {
+  if (!source_->NextPacketTime()) {
+    // End of input. Cannot do proper replacement on the very last packet, so we
+    // delete it instead.
+    packet_.reset();
+    return;
+  }
+
+  RTC_DCHECK(packet_);
+
+  RTC_CHECK_EQ(forbidden_types_.count(packet_->header.payloadType), 0)
+      << "Payload type " << static_cast<int>(packet_->header.payloadType)
+      << " is forbidden.";
+
+  // Check if this packet is comfort noise.
+  if (comfort_noise_types_.count(packet_->header.payloadType) != 0) {
+    // If CNG, simply insert a zero-energy one-byte payload.
+    uint8_t cng_payload[1] = {127};  // Max attenuation of CNG.
+    packet_->payload.SetData(cng_payload);
+    return;
+  }
+
+  absl::optional<RTPHeader> next_hdr = source_->NextHeader();
+  RTC_DCHECK(next_hdr);
+  uint8_t payload[12];
+  RTC_DCHECK_LE(last_frame_size_timestamps_, 120 * 48);
+  uint32_t input_frame_size_timestamps = last_frame_size_timestamps_;
+  const uint32_t timestamp_diff =
+      next_hdr->timestamp - packet_->header.timestamp;
+  if (next_hdr->sequenceNumber == packet_->header.sequenceNumber + 1 &&
+      timestamp_diff <= 120 * 48) {
+    // Packets are in order and the timestamp diff is less than 5760 samples.
+    // Accept the timestamp diff as a valid frame size.
+    input_frame_size_timestamps = timestamp_diff;
+    last_frame_size_timestamps_ = input_frame_size_timestamps;
+  }
+  RTC_DCHECK_LE(input_frame_size_timestamps, 120 * 48);
+  FakeDecodeFromFile::PrepareEncoded(packet_->header.timestamp,
+                                     input_frame_size_timestamps,
+                                     packet_->payload.size(), payload);
+  packet_->payload.SetData(payload);
+  packet_->header.payloadType = replacement_payload_type_;
+  return;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h
new file mode 100644
index 0000000000..9ce9b9dc63
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
+
+#include <memory>
+#include <set>
+
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+
+namespace webrtc {
+namespace test {
+
+// This class converts the packets from a NetEqInput to fake encodings to be
+// decoded by a FakeDecodeFromFile decoder.
+class NetEqReplacementInput : public NetEqInput {
+ public:
+  NetEqReplacementInput(std::unique_ptr<NetEqInput> source,
+                        uint8_t replacement_payload_type,
+                        const std::set<uint8_t>& comfort_noise_types,
+                        const std::set<uint8_t>& forbidden_types);
+
+  absl::optional<int64_t> NextPacketTime() const override;
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  void AdvanceOutputEvent() override;
+  bool ended() const override;
+  absl::optional<RTPHeader> NextHeader() const override;
+
+ private:
+  void ReplacePacket();
+
+  std::unique_ptr<NetEqInput> source_;
+  const uint8_t replacement_payload_type_;
+  const std::set<uint8_t> comfort_noise_types_;
+  const std::set<uint8_t> forbidden_types_;
+  std::unique_ptr<PacketData> packet_;         // The next packet to deliver.
+  uint32_t last_frame_size_timestamps_ = 960;  // Initial guess: 20 ms @ 48 kHz.
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
new file mode 100644
index 0000000000..b274069bd4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -0,0 +1,406 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <iostream>
+#include <string>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/field_trial.h"
+
+using TestConfig = webrtc::test::NetEqTestFactory::Config;
+
+ABSL_FLAG(bool,
+          codec_map,
+          false,
+          "Prints the mapping between RTP payload type and "
+          "codec");
+ABSL_FLAG(std::string,
+          force_fieldtrials,
+          "",
+          "Field trials control experimental feature code which can be forced. "
+          "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
+          " will assign the group Enable to field trial WebRTC-FooFeature.");
+ABSL_FLAG(int, pcmu, TestConfig::default_pcmu(), "RTP payload type for PCM-u");
+ABSL_FLAG(int, pcma, TestConfig::default_pcma(), "RTP payload type for PCM-a");
+ABSL_FLAG(int, ilbc, TestConfig::default_ilbc(), "RTP payload type for iLBC");
+ABSL_FLAG(int, isac, TestConfig::default_isac(), "RTP payload type for iSAC");
+ABSL_FLAG(int,
+          isac_swb,
+          TestConfig::default_isac_swb(),
+          "RTP payload type for iSAC-swb (32 kHz)");
+ABSL_FLAG(int, opus, TestConfig::default_opus(), "RTP payload type for Opus");
+ABSL_FLAG(int,
+          pcm16b,
+          TestConfig::default_pcm16b(),
+          "RTP payload type for PCM16b-nb (8 kHz)");
+ABSL_FLAG(int,
+          pcm16b_wb,
+          TestConfig::default_pcm16b_wb(),
+          "RTP payload type for PCM16b-wb (16 kHz)");
+ABSL_FLAG(int,
+          pcm16b_swb32,
+          TestConfig::default_pcm16b_swb32(),
+          "RTP payload type for PCM16b-swb32 (32 kHz)");
+ABSL_FLAG(int,
+          pcm16b_swb48,
+          TestConfig::default_pcm16b_swb48(),
+          "RTP payload type for PCM16b-swb48 (48 kHz)");
+ABSL_FLAG(int, g722, TestConfig::default_g722(), "RTP payload type for G.722");
+ABSL_FLAG(int,
+          avt,
+          TestConfig::default_avt(),
+          "RTP payload type for AVT/DTMF (8 kHz)");
+ABSL_FLAG(int,
+          avt_16,
+          TestConfig::default_avt_16(),
+          "RTP payload type for AVT/DTMF (16 kHz)");
+ABSL_FLAG(int,
+          avt_32,
+          TestConfig::default_avt_32(),
+          "RTP payload type for AVT/DTMF (32 kHz)");
+ABSL_FLAG(int,
+          avt_48,
+          TestConfig::default_avt_48(),
+          "RTP payload type for AVT/DTMF (48 kHz)");
+ABSL_FLAG(int,
+          red,
+          TestConfig::default_red(),
+          "RTP payload type for redundant audio (RED)");
+ABSL_FLAG(int,
+          cn_nb,
+          TestConfig::default_cn_nb(),
+          "RTP payload type for comfort noise (8 kHz)");
+ABSL_FLAG(int,
+          cn_wb,
+          TestConfig::default_cn_wb(),
+          "RTP payload type for comfort noise (16 kHz)");
+ABSL_FLAG(int,
+          cn_swb32,
+          TestConfig::default_cn_swb32(),
+          "RTP payload type for comfort noise (32 kHz)");
+ABSL_FLAG(int,
+          cn_swb48,
+          TestConfig::default_cn_swb48(),
+          "RTP payload type for comfort noise (48 kHz)");
+ABSL_FLAG(std::string,
+          replacement_audio_file,
+          "",
+          "A PCM file that will be used to populate dummy"
+          " RTP packets");
+ABSL_FLAG(std::string,
+          ssrc,
+          "",
+          "Only use packets with this SSRC (decimal or hex, the latter "
+          "starting with 0x)");
+ABSL_FLAG(int,
+          audio_level,
+          TestConfig::default_audio_level(),
+          "Extension ID for audio level (RFC 6464)");
+ABSL_FLAG(int,
+          abs_send_time,
+          TestConfig::default_abs_send_time(),
+          "Extension ID for absolute sender time");
+ABSL_FLAG(int,
+          transport_seq_no,
+          TestConfig::default_transport_seq_no(),
+          "Extension ID for transport sequence number");
+ABSL_FLAG(int,
+          video_content_type,
+          TestConfig::default_video_content_type(),
+          "Extension ID for video content type");
+ABSL_FLAG(int,
+          video_timing,
+          TestConfig::default_video_timing(),
+          "Extension ID for video timing");
+ABSL_FLAG(std::string,
+          output_files_base_name,
+          "",
+          "Custom path used as prefix for the output files - i.e., "
+          "matlab plot, python plot, text log.");
+ABSL_FLAG(bool,
+          matlabplot,
+          false,
+          "Generates a matlab script for plotting the delay profile");
+ABSL_FLAG(bool,
+          pythonplot,
+          false,
+          "Generates a python script for plotting the delay profile");
+ABSL_FLAG(bool,
+          textlog,
+          false,
+          "Generates a text log describing the simulation on a "
+          "step-by-step basis.");
+ABSL_FLAG(bool, concealment_events, false, "Prints concealment events");
+ABSL_FLAG(int,
+          max_nr_packets_in_buffer,
+          TestConfig::default_max_nr_packets_in_buffer(),
+          "Maximum allowed number of packets in the buffer");
+ABSL_FLAG(bool,
+          enable_fast_accelerate,
+          false,
+          "Enables jitter buffer fast accelerate");
+
+namespace {
+
+// Parses the input string for a valid SSRC (at the start of the string). If a
+// valid SSRC is found, it is written to the output variable `ssrc`, and true is
+// returned. Otherwise, false is returned.
+bool ParseSsrc(absl::string_view str, uint32_t* ssrc) {
+  if (str.empty())
+    return true;
+  int base = 10;
+  // Look for "0x" or "0X" at the start and change base to 16 if found.
+  if ((str.compare(0, 2, "0x") == 0) || (str.compare(0, 2, "0X") == 0))
+    base = 16;
+  errno = 0;
+  char* end_ptr;
+  std::string str_str = std::string(str);
+  unsigned long value = strtoul(str_str.c_str(), &end_ptr, base);  // NOLINT
+  if (value == ULONG_MAX && errno == ERANGE)
+    return false;  // Value out of range for unsigned long.
+  if (sizeof(unsigned long) > sizeof(uint32_t) && value > 0xFFFFFFFF)  // NOLINT
+    return false;  // Value out of range for uint32_t.
+  if (end_ptr - str_str.c_str() < static_cast<ptrdiff_t>(str.length()))
+    return false;  // Part of the string was not parsed.
+  *ssrc = static_cast<uint32_t>(value);
+  return true;
+}
+
+static bool ValidateExtensionId(int value) {
+  if (value > 0 && value <= 255)  // Value is ok.
+    return true;
+  printf("Extension ID must be between 1 and 255, not %d\n",
+         static_cast<int>(value));
+  return false;
+}
+
+// Flag validators.
+bool ValidatePayloadType(int value) {
+  if (value >= 0 && value <= 127)  // Value is ok.
+    return true;
+  printf("Payload type must be between 0 and 127, not %d\n",
+         static_cast<int>(value));
+  return false;
+}
+
+bool ValidateSsrcValue(absl::string_view str) {
+  uint32_t dummy_ssrc;
+  if (ParseSsrc(str, &dummy_ssrc))  // Value is ok.
+    return true;
+  printf("Invalid SSRC: %.*s\n", static_cast<int>(str.size()), str.data());
+  return false;
+}
+
+void PrintCodecMappingEntry(absl::string_view codec, int flag) {
+  std::cout << codec << ": " << flag << std::endl;
+}
+
+void PrintCodecMapping() {
+  PrintCodecMappingEntry("PCM-u", absl::GetFlag(FLAGS_pcmu));
+  PrintCodecMappingEntry("PCM-a", absl::GetFlag(FLAGS_pcma));
+  PrintCodecMappingEntry("iLBC", absl::GetFlag(FLAGS_ilbc));
+  PrintCodecMappingEntry("iSAC", absl::GetFlag(FLAGS_isac));
+  PrintCodecMappingEntry("iSAC-swb (32 kHz)", absl::GetFlag(FLAGS_isac_swb));
+  PrintCodecMappingEntry("Opus", absl::GetFlag(FLAGS_opus));
+  PrintCodecMappingEntry("PCM16b-nb (8 kHz)", absl::GetFlag(FLAGS_pcm16b));
+  PrintCodecMappingEntry("PCM16b-wb (16 kHz)", absl::GetFlag(FLAGS_pcm16b_wb));
+  PrintCodecMappingEntry("PCM16b-swb32 (32 kHz)",
+                         absl::GetFlag(FLAGS_pcm16b_swb32));
+  PrintCodecMappingEntry("PCM16b-swb48 (48 kHz)",
+                         absl::GetFlag(FLAGS_pcm16b_swb48));
+  PrintCodecMappingEntry("G.722", absl::GetFlag(FLAGS_g722));
+  PrintCodecMappingEntry("AVT/DTMF (8 kHz)", absl::GetFlag(FLAGS_avt));
+  PrintCodecMappingEntry("AVT/DTMF (16 kHz)", absl::GetFlag(FLAGS_avt_16));
+  PrintCodecMappingEntry("AVT/DTMF (32 kHz)", absl::GetFlag(FLAGS_avt_32));
+  PrintCodecMappingEntry("AVT/DTMF (48 kHz)", absl::GetFlag(FLAGS_avt_48));
+  PrintCodecMappingEntry("redundant audio (RED)", absl::GetFlag(FLAGS_red));
+  PrintCodecMappingEntry("comfort noise (8 kHz)", absl::GetFlag(FLAGS_cn_nb));
+  PrintCodecMappingEntry("comfort noise (16 kHz)", absl::GetFlag(FLAGS_cn_wb));
+  PrintCodecMappingEntry("comfort noise (32 kHz)",
+                         absl::GetFlag(FLAGS_cn_swb32));
+  PrintCodecMappingEntry("comfort noise (48 kHz)",
+                         absl::GetFlag(FLAGS_cn_swb48));
+}
+
+bool ValidateOutputFilesOptions(bool textlog,
+                                bool plotting,
+                                absl::string_view output_files_base_name,
+                                absl::string_view output_audio_filename) {
+  bool output_files_base_name_specified = !output_files_base_name.empty();
+  if (!textlog && !plotting && output_files_base_name_specified) {
+    std::cout << "Error: --output_files_base_name cannot be used without at "
+                 "least one of the following flags: --textlog, --matlabplot, "
+                 "--pythonplot."
+              << std::endl;
+    return false;
+  }
+  // Without `output_audio_filename`, `output_files_base_name` is required when
+  // plotting output files must be generated (in order to form a valid output
+  // file name).
+  if (output_audio_filename.empty() && plotting &&
+      !output_files_base_name_specified) {
+    std::cout << "Error: when no output audio file is specified and "
+                 "--matlabplot and/or --pythonplot are used, "
+                 "--output_files_base_name must be also used."
+              << std::endl;
+    return false;
+  }
+  return true;
+}
+
+absl::optional<std::string> CreateOptionalOutputFileName(
+    bool output_requested,
+    absl::string_view basename,
+    absl::string_view output_audio_filename,
+    absl::string_view suffix) {
+  if (!output_requested) {
+    return absl::nullopt;
+  }
+  if (!basename.empty()) {
+    // Override the automatic assignment.
+    rtc::StringBuilder sb(basename);
+    sb << suffix;
+    return sb.str();
+  }
+  if (!output_audio_filename.empty()) {
+    // Automatically assign name.
+    rtc::StringBuilder sb(output_audio_filename);
+    sb << suffix;
+    return sb.str();
+  }
+  std::cout << "Error: invalid text log file parameters.";
+  return absl::nullopt;
+}
+
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  webrtc::test::NetEqTestFactory factory;
+  std::string usage =
+      "Tool for decoding an RTP dump file using NetEq.\n"
+      "Example usage:\n"
+      "./neteq_rtpplay input.rtp [output.{pcm, wav}]\n";
+  if (absl::GetFlag(FLAGS_codec_map)) {
+    PrintCodecMapping();
+    exit(0);
+  }
+  if (args.size() != 2 &&
+      args.size() != 3) {  // The output audio file is optional.
+    // Print usage information.
+    std::cout << usage;
+    exit(0);
+  }
+  const std::string output_audio_filename((args.size() == 3) ? args[2] : "");
+  const std::string output_files_base_name(
+      absl::GetFlag(FLAGS_output_files_base_name));
+  RTC_CHECK(ValidateOutputFilesOptions(
+      absl::GetFlag(FLAGS_textlog),
+      absl::GetFlag(FLAGS_matlabplot) || absl::GetFlag(FLAGS_pythonplot),
+      output_files_base_name, output_audio_filename));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcmu)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcma)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_ilbc)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_isac)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_isac_swb)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_opus)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_wb)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_swb32)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_swb48)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_g722)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt_16)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt_32)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt_48)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_red)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_nb)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_wb)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_swb32)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_swb48)));
+  RTC_CHECK(ValidateSsrcValue(absl::GetFlag(FLAGS_ssrc)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_audio_level)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_abs_send_time)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_transport_seq_no)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_video_content_type)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_video_timing)));
+
+  // Make force_fieldtrials persistent string during entire program live as
+  // absl::GetFlag creates temporary string and c_str() will point to
+  // deallocated string.
+  const std::string force_fieldtrials = absl::GetFlag(FLAGS_force_fieldtrials);
+  webrtc::field_trial::InitFieldTrialsFromString(force_fieldtrials.c_str());
+
+  webrtc::test::NetEqTestFactory::Config config;
+  config.pcmu = absl::GetFlag(FLAGS_pcmu);
+  config.pcma = absl::GetFlag(FLAGS_pcma);
+  config.ilbc = absl::GetFlag(FLAGS_ilbc);
+  config.isac = absl::GetFlag(FLAGS_isac);
+  config.isac_swb = absl::GetFlag(FLAGS_isac_swb);
+  config.opus = absl::GetFlag(FLAGS_opus);
+  config.pcm16b = absl::GetFlag(FLAGS_pcm16b);
+  config.pcm16b_wb = absl::GetFlag(FLAGS_pcm16b_wb);
+  config.pcm16b_swb32 = absl::GetFlag(FLAGS_pcm16b_swb32);
+  config.pcm16b_swb48 = absl::GetFlag(FLAGS_pcm16b_swb48);
+  config.g722 = absl::GetFlag(FLAGS_g722);
+  config.avt = absl::GetFlag(FLAGS_avt);
+  config.avt_16 = absl::GetFlag(FLAGS_avt_16);
+  config.avt_32 = absl::GetFlag(FLAGS_avt_32);
+  config.avt_48 = absl::GetFlag(FLAGS_avt_48);
+  config.red = absl::GetFlag(FLAGS_red);
+  config.cn_nb = absl::GetFlag(FLAGS_cn_nb);
+  config.cn_wb = absl::GetFlag(FLAGS_cn_wb);
+  config.cn_swb32 = absl::GetFlag(FLAGS_cn_swb32);
+  config.cn_swb48 = absl::GetFlag(FLAGS_cn_swb48);
+  config.replacement_audio_file = absl::GetFlag(FLAGS_replacement_audio_file);
+  config.audio_level = absl::GetFlag(FLAGS_audio_level);
+  config.abs_send_time = absl::GetFlag(FLAGS_abs_send_time);
+  config.transport_seq_no = absl::GetFlag(FLAGS_transport_seq_no);
+  config.video_content_type = absl::GetFlag(FLAGS_video_content_type);
+  config.video_timing = absl::GetFlag(FLAGS_video_timing);
+  config.matlabplot = absl::GetFlag(FLAGS_matlabplot);
+  config.pythonplot = absl::GetFlag(FLAGS_pythonplot);
+  config.concealment_events = absl::GetFlag(FLAGS_concealment_events);
+  config.max_nr_packets_in_buffer =
+      absl::GetFlag(FLAGS_max_nr_packets_in_buffer);
+  config.enable_fast_accelerate = absl::GetFlag(FLAGS_enable_fast_accelerate);
+  if (!output_audio_filename.empty()) {
+    config.output_audio_filename = output_audio_filename;
+  }
+  config.textlog = absl::GetFlag(FLAGS_textlog);
+  config.textlog_filename = CreateOptionalOutputFileName(
+      absl::GetFlag(FLAGS_textlog), output_files_base_name,
+      output_audio_filename, ".text_log.txt");
+  config.plot_scripts_basename = CreateOptionalOutputFileName(
+      absl::GetFlag(FLAGS_matlabplot) || absl::GetFlag(FLAGS_pythonplot),
+      output_files_base_name, output_audio_filename, "");
+
+  // Check if an SSRC value was provided.
+  if (absl::GetFlag(FLAGS_ssrc).size() > 0) {
+    uint32_t ssrc;
+    RTC_CHECK(ParseSsrc(absl::GetFlag(FLAGS_ssrc), &ssrc))
+        << "Flag verification has failed.";
+    config.ssrc_filter = absl::make_optional(ssrc);
+  }
+
+  std::unique_ptr<webrtc::test::NetEqTest> test =
+      factory.InitializeTestFromFile(/*input_filename=*/args[1],
+                                     /*factory=*/nullptr, config);
+  RTC_CHECK(test) << "ERROR: Unable to run test";
+  test->Run();
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh
new file mode 100755
index 0000000000..0a6bf16016
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh
@@ -0,0 +1,183 @@
+#!/bin/bash
+#
+#  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+#
+#  Use of this source code is governed by a BSD-style license
+#  that can be found in the LICENSE file in the root of the source
+#  tree. An additional intellectual property rights grant can be found
+#  in the file PATENTS.  All contributing project authors may
+#  be found in the AUTHORS file in the root of the source tree.
+#
+
+# Aliases.
+BIN=$1
+TEST_RTC_EVENT_LOG=$2
+INPUT_PCM_FILE=$3
+
+# Check setup.
+if [ ! -f $BIN ]; then
+  echo "Cannot find neteq_rtpplay binary."
+  exit 99
+fi
+if [ ! -f $TEST_RTC_EVENT_LOG ]; then
+  echo "Cannot find RTC event log file."
+  exit 99
+fi
+if [ ! -f $INPUT_PCM_FILE ]; then
+  echo "Cannot find PCM file."
+  exit 99
+fi
+
+# Defines.
+
+TMP_DIR=$(mktemp -d /tmp/tmp_XXXXXXXXXX)
+PASS=0
+FAIL=1
+TEST_SUITE_RESULT=$PASS
+
+file_hash () {
+  md5sum $1 | awk '{ print $1 }'
+}
+
+test_passed () {
+  echo PASS
+}
+
+test_failed () {
+  echo "FAIL: $1"
+  TEST_SUITE_RESULT=$FAIL
+}
+
+test_file_checksums_match () {
+  if [ ! -f $1 ] || [ ! -f $2 ]; then
+    test_failed "Cannot compare hash values: file(s) not found."
+    return
+  fi
+  HASH1=$(file_hash $1)
+  HASH2=$(file_hash $2)
+  if [ "$HASH1" = "$HASH2" ]; then
+    test_passed
+  else
+    test_failed "$1 differs from $2"
+  fi
+}
+
+test_file_exists () {
+  if [ -f $1 ]; then
+    test_passed
+  else
+    test_failed "$1 does not exist"
+  fi
+}
+
+test_exit_code_0 () {
+  if [ $1 -eq 0 ]; then
+    test_passed
+  else
+    test_failed "$1 did not return 0"
+  fi
+}
+
+test_exit_code_not_0 () {
+  if [ $1 -eq 0 ]; then
+    test_failed "$1 returned 0"
+  else
+    test_passed
+  fi
+}
+
+# Generate test data.
+
+# Case 1. Pre-existing way.
+CASE1_WAV=$TMP_DIR/case1.wav
+$BIN $TEST_RTC_EVENT_LOG $CASE1_WAV  \
+    --replacement_audio_file $INPUT_PCM_FILE  \
+    --textlog --pythonplot --matlabplot  \
+    > $TMP_DIR/case1.stdout 2> /dev/null
+CASE1_RETURN_CODE=$?
+CASE1_TEXTLOG=$TMP_DIR/case1.wav.text_log.txt
+CASE1_PYPLOT=$TMP_DIR/case1_wav.py
+CASE1_MATPLOT=$TMP_DIR/case1_wav.m
+
+# Case 2. No output files.
+$BIN $TEST_RTC_EVENT_LOG --replacement_audio_file $INPUT_PCM_FILE  \
+    > $TMP_DIR/case2.stdout 2> /dev/null
+CASE2_RETURN_CODE=$?
+
+# Case 3. No output audio file.
+
+# Case 3.1 Without --output_files_base_name (won't run).
+$BIN $TEST_RTC_EVENT_LOG  \
+    --replacement_audio_file $INPUT_PCM_FILE  \
+    --textlog --pythonplot --matlabplot  \
+    &> /dev/null
+CASE3_1_RETURN_CODE=$?
+
+# Case 3.2 With --output_files_base_name (runs).
+$BIN $TEST_RTC_EVENT_LOG  \
+    --replacement_audio_file $INPUT_PCM_FILE  \
+    --output_files_base_name $TMP_DIR/case3_2  \
+    --textlog --pythonplot --matlabplot  \
+    > $TMP_DIR/case3_2.stdout 2> /dev/null
+CASE3_2_RETURN_CODE=$?
+CASE3_2_TEXTLOG=$TMP_DIR/case3_2.text_log.txt
+CASE3_2_PYPLOT=$TMP_DIR/case3_2.py
+CASE3_2_MATPLOT=$TMP_DIR/case3_2.m
+
+# Case 4. With output audio file and --output_files_base_name.
+CASE4_WAV=$TMP_DIR/case4.wav
+$BIN $TEST_RTC_EVENT_LOG $TMP_DIR/case4.wav \
+    --replacement_audio_file $INPUT_PCM_FILE  \
+    --output_files_base_name $TMP_DIR/case4  \
+    --textlog --pythonplot --matlabplot  \
+    > $TMP_DIR/case4.stdout 2> /dev/null
+CASE4_RETURN_CODE=$?
+CASE4_TEXTLOG=$TMP_DIR/case4.text_log.txt
+CASE4_PYPLOT=$TMP_DIR/case4.py
+CASE4_MATPLOT=$TMP_DIR/case4.m
+
+# Tests.
+
+echo Check exit codes
+test_exit_code_0 $CASE1_RETURN_CODE
+test_exit_code_0 $CASE2_RETURN_CODE
+test_exit_code_not_0 $CASE3_1_RETURN_CODE
+test_exit_code_0 $CASE3_2_RETURN_CODE
+test_exit_code_0 $CASE4_RETURN_CODE
+
+echo Check that the expected output files exist
+test_file_exists $CASE1_TEXTLOG
+test_file_exists $CASE3_2_TEXTLOG
+test_file_exists $CASE4_TEXTLOG
+test_file_exists $CASE1_PYPLOT
+test_file_exists $CASE3_2_PYPLOT
+test_file_exists $CASE4_PYPLOT
+test_file_exists $CASE1_MATPLOT
+test_file_exists $CASE3_2_MATPLOT
+test_file_exists $CASE4_MATPLOT
+
+echo Check that the same WAV file is produced
+test_file_checksums_match $CASE1_WAV $CASE4_WAV
+
+echo Check that the same text log is produced
+test_file_checksums_match $CASE1_TEXTLOG $CASE3_2_TEXTLOG
+test_file_checksums_match $CASE1_TEXTLOG $CASE4_TEXTLOG
+
+echo Check that the same python plot scripts is produced
+test_file_checksums_match $CASE1_PYPLOT $CASE3_2_PYPLOT
+test_file_checksums_match $CASE1_PYPLOT $CASE4_PYPLOT
+
+echo Check that the same matlab plot scripts is produced
+test_file_checksums_match $CASE1_MATPLOT $CASE3_2_MATPLOT
+test_file_checksums_match $CASE1_MATPLOT $CASE4_MATPLOT
+
+# Clean up
+rm -fr $TMP_DIR
+
+if [ $TEST_SUITE_RESULT -eq $PASS ]; then
+  echo All tests passed.
+  exit 0
+else
+  echo One or more tests failed.
+  exit 1
+fi
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc
new file mode 100644
index 0000000000..6738e494f6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc
@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
+
+#include <algorithm>
+#include <numeric>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace test {
+
+std::string NetEqStatsGetter::ConcealmentEvent::ToString() const {
+  char ss_buf[256];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  ss << "ConcealmentEvent duration_ms:" << duration_ms
+     << " event_number:" << concealment_event_number
+     << " time_from_previous_event_end_ms:" << time_from_previous_event_end_ms;
+  return ss.str();
+}
+
+NetEqStatsGetter::NetEqStatsGetter(
+    std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer)
+    : delay_analyzer_(std::move(delay_analyzer)) {}
+
+void NetEqStatsGetter::BeforeGetAudio(NetEq* neteq) {
+  if (delay_analyzer_) {
+    delay_analyzer_->BeforeGetAudio(neteq);
+  }
+}
+
+void NetEqStatsGetter::AfterGetAudio(int64_t time_now_ms,
+                                     const AudioFrame& audio_frame,
+                                     bool muted,
+                                     NetEq* neteq) {
+  // TODO(minyue): Get stats should better not be called as a call back after
+  // get audio. It is called independently from get audio in practice.
+  const auto lifetime_stat = neteq->GetLifetimeStatistics();
+  if (last_stats_query_time_ms_ == 0 ||
+      rtc::TimeDiff(time_now_ms, last_stats_query_time_ms_) >=
+          stats_query_interval_ms_) {
+    NetEqNetworkStatistics stats;
+    RTC_CHECK_EQ(neteq->NetworkStatistics(&stats), 0);
+    stats_.push_back(std::make_pair(time_now_ms, stats));
+    lifetime_stats_.push_back(std::make_pair(time_now_ms, lifetime_stat));
+    last_stats_query_time_ms_ = time_now_ms;
+  }
+
+  const auto voice_concealed_samples =
+      lifetime_stat.concealed_samples - lifetime_stat.silent_concealed_samples;
+  if (current_concealment_event_ != lifetime_stat.concealment_events &&
+      voice_concealed_samples_until_last_event_ < voice_concealed_samples) {
+    if (last_event_end_time_ms_ > 0) {
+      // Do not account for the first event to avoid start of the call
+      // skewing.
+      ConcealmentEvent concealment_event;
+      uint64_t last_event_voice_concealed_samples =
+          voice_concealed_samples - voice_concealed_samples_until_last_event_;
+      RTC_CHECK_GT(last_event_voice_concealed_samples, 0);
+      concealment_event.duration_ms = last_event_voice_concealed_samples /
+                                      (audio_frame.sample_rate_hz_ / 1000);
+      concealment_event.concealment_event_number = current_concealment_event_;
+      concealment_event.time_from_previous_event_end_ms =
+          time_now_ms - last_event_end_time_ms_;
+      concealment_events_.emplace_back(concealment_event);
+      voice_concealed_samples_until_last_event_ = voice_concealed_samples;
+    }
+    last_event_end_time_ms_ = time_now_ms;
+    voice_concealed_samples_until_last_event_ = voice_concealed_samples;
+    current_concealment_event_ = lifetime_stat.concealment_events;
+  }
+
+  if (delay_analyzer_) {
+    delay_analyzer_->AfterGetAudio(time_now_ms, audio_frame, muted, neteq);
+  }
+}
+
+double NetEqStatsGetter::AverageSpeechExpandRate() const {
+  double sum_speech_expand = std::accumulate(
+      stats_.begin(), stats_.end(), double{0.0},
+      [](double a, std::pair<int64_t, NetEqNetworkStatistics> b) {
+        return a + static_cast<double>(b.second.speech_expand_rate);
+      });
+  return sum_speech_expand / 16384.0 / stats_.size();
+}
+
+NetEqStatsGetter::Stats NetEqStatsGetter::AverageStats() const {
+  Stats sum_stats = std::accumulate(
+      stats_.begin(), stats_.end(), Stats(),
+      [](Stats a, std::pair<int64_t, NetEqNetworkStatistics> bb) {
+        const auto& b = bb.second;
+        a.current_buffer_size_ms += b.current_buffer_size_ms;
+        a.preferred_buffer_size_ms += b.preferred_buffer_size_ms;
+        a.jitter_peaks_found += b.jitter_peaks_found;
+        a.expand_rate += b.expand_rate / 16384.0;
+        a.speech_expand_rate += b.speech_expand_rate / 16384.0;
+        a.preemptive_rate += b.preemptive_rate / 16384.0;
+        a.accelerate_rate += b.accelerate_rate / 16384.0;
+        a.secondary_decoded_rate += b.secondary_decoded_rate / 16384.0;
+        a.secondary_discarded_rate += b.secondary_discarded_rate / 16384.0;
+        a.mean_waiting_time_ms += b.mean_waiting_time_ms;
+        a.median_waiting_time_ms += b.median_waiting_time_ms;
+        a.min_waiting_time_ms = std::min(
+            a.min_waiting_time_ms, static_cast<double>(b.min_waiting_time_ms));
+        a.max_waiting_time_ms = std::max(
+            a.max_waiting_time_ms, static_cast<double>(b.max_waiting_time_ms));
+        return a;
+      });
+
+  sum_stats.current_buffer_size_ms /= stats_.size();
+  sum_stats.preferred_buffer_size_ms /= stats_.size();
+  sum_stats.jitter_peaks_found /= stats_.size();
+  sum_stats.packet_loss_rate /= stats_.size();
+  sum_stats.expand_rate /= stats_.size();
+  sum_stats.speech_expand_rate /= stats_.size();
+  sum_stats.preemptive_rate /= stats_.size();
+  sum_stats.accelerate_rate /= stats_.size();
+  sum_stats.secondary_decoded_rate /= stats_.size();
+  sum_stats.secondary_discarded_rate /= stats_.size();
+  sum_stats.added_zero_samples /= stats_.size();
+  sum_stats.mean_waiting_time_ms /= stats_.size();
+  sum_stats.median_waiting_time_ms /= stats_.size();
+
+  return sum_stats;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h
new file mode 100644
index 0000000000..b1b12bb1f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqStatsGetter : public NetEqGetAudioCallback {
+ public:
+  // This struct is a replica of webrtc::NetEqNetworkStatistics, but with all
+  // values stored in double precision.
+  struct Stats {
+    double current_buffer_size_ms = 0.0;
+    double preferred_buffer_size_ms = 0.0;
+    double jitter_peaks_found = 0.0;
+    double packet_loss_rate = 0.0;
+    double expand_rate = 0.0;
+    double speech_expand_rate = 0.0;
+    double preemptive_rate = 0.0;
+    double accelerate_rate = 0.0;
+    double secondary_decoded_rate = 0.0;
+    double secondary_discarded_rate = 0.0;
+    double clockdrift_ppm = 0.0;
+    double added_zero_samples = 0.0;
+    double mean_waiting_time_ms = 0.0;
+    double median_waiting_time_ms = 0.0;
+    double min_waiting_time_ms = 0.0;
+    double max_waiting_time_ms = 0.0;
+  };
+
+  struct ConcealmentEvent {
+    uint64_t duration_ms;
+    size_t concealment_event_number;
+    int64_t time_from_previous_event_end_ms;
+    std::string ToString() const;
+  };
+
+  // Takes a pointer to another callback object, which will be invoked after
+  // this object finishes. This does not transfer ownership, and null is a
+  // valid value.
+  explicit NetEqStatsGetter(std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer);
+
+  void set_stats_query_interval_ms(int64_t stats_query_interval_ms) {
+    stats_query_interval_ms_ = stats_query_interval_ms;
+  }
+
+  void BeforeGetAudio(NetEq* neteq) override;
+
+  void AfterGetAudio(int64_t time_now_ms,
+                     const AudioFrame& audio_frame,
+                     bool muted,
+                     NetEq* neteq) override;
+
+  double AverageSpeechExpandRate() const;
+
+  NetEqDelayAnalyzer* delay_analyzer() const { return delay_analyzer_.get(); }
+
+  const std::vector<ConcealmentEvent>& concealment_events() const {
+    // Do not account for the last concealment event to avoid potential end
+    // call skewing.
+    return concealment_events_;
+  }
+
+  const std::vector<std::pair<int64_t, NetEqNetworkStatistics>>* stats() const {
+    return &stats_;
+  }
+
+  const std::vector<std::pair<int64_t, NetEqLifetimeStatistics>>*
+  lifetime_stats() const {
+    return &lifetime_stats_;
+  }
+
+  Stats AverageStats() const;
+
+ private:
+  std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer_;
+  int64_t stats_query_interval_ms_ = 1000;
+  int64_t last_stats_query_time_ms_ = 0;
+  std::vector<std::pair<int64_t, NetEqNetworkStatistics>> stats_;
+  std::vector<std::pair<int64_t, NetEqLifetimeStatistics>> lifetime_stats_;
+  size_t current_concealment_event_ = 1;
+  uint64_t voice_concealed_samples_until_last_event_ = 0;
+  std::vector<ConcealmentEvent> concealment_events_;
+  int64_t last_event_end_time_ms_ = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
new file mode 100644
index 0000000000..162a4c9300
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_stats_plotter.h"
+
+#include <inttypes.h>
+#include <stdio.h>
+
+#include <utility>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqStatsPlotter::NetEqStatsPlotter(bool make_matlab_plot,
+                                     bool make_python_plot,
+                                     bool show_concealment_events,
+                                     absl::string_view base_file_name)
+    : make_matlab_plot_(make_matlab_plot),
+      make_python_plot_(make_python_plot),
+      show_concealment_events_(show_concealment_events),
+      base_file_name_(base_file_name) {
+  std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer;
+  if (make_matlab_plot || make_python_plot) {
+    delay_analyzer.reset(new NetEqDelayAnalyzer);
+  }
+  stats_getter_.reset(new NetEqStatsGetter(std::move(delay_analyzer)));
+}
+
+void NetEqStatsPlotter::SimulationEnded(int64_t simulation_time_ms) {
+  if (make_matlab_plot_) {
+    auto matlab_script_name = base_file_name_;
+    std::replace(matlab_script_name.begin(), matlab_script_name.end(), '.',
+                 '_');
+    printf("Creating Matlab plot script %s.m\n", matlab_script_name.c_str());
+    stats_getter_->delay_analyzer()->CreateMatlabScript(matlab_script_name +
+                                                        ".m");
+  }
+  if (make_python_plot_) {
+    auto python_script_name = base_file_name_;
+    std::replace(python_script_name.begin(), python_script_name.end(), '.',
+                 '_');
+    printf("Creating Python plot script %s.py\n", python_script_name.c_str());
+    stats_getter_->delay_analyzer()->CreatePythonScript(python_script_name +
+                                                        ".py");
+  }
+
+  printf("Simulation statistics:\n");
+  printf("  output duration: %" PRId64 " ms\n", simulation_time_ms);
+  auto stats = stats_getter_->AverageStats();
+  printf("  packet_loss_rate: %f %%\n", 100.0 * stats.packet_loss_rate);
+  printf("  expand_rate: %f %%\n", 100.0 * stats.expand_rate);
+  printf("  speech_expand_rate: %f %%\n", 100.0 * stats.speech_expand_rate);
+  printf("  preemptive_rate: %f %%\n", 100.0 * stats.preemptive_rate);
+  printf("  accelerate_rate: %f %%\n", 100.0 * stats.accelerate_rate);
+  printf("  secondary_decoded_rate: %f %%\n",
+         100.0 * stats.secondary_decoded_rate);
+  printf("  secondary_discarded_rate: %f %%\n",
+         100.0 * stats.secondary_discarded_rate);
+  printf("  clockdrift_ppm: %f ppm\n", stats.clockdrift_ppm);
+  printf("  mean_waiting_time_ms: %f ms\n", stats.mean_waiting_time_ms);
+  printf("  median_waiting_time_ms: %f ms\n", stats.median_waiting_time_ms);
+  printf("  min_waiting_time_ms: %f ms\n", stats.min_waiting_time_ms);
+  printf("  max_waiting_time_ms: %f ms\n", stats.max_waiting_time_ms);
+  printf("  current_buffer_size_ms: %f ms\n", stats.current_buffer_size_ms);
+  printf("  preferred_buffer_size_ms: %f ms\n", stats.preferred_buffer_size_ms);
+  if (show_concealment_events_) {
+    printf(" concealment_events_ms:\n");
+    for (auto concealment_event : stats_getter_->concealment_events())
+      printf("%s\n", concealment_event.ToString().c_str());
+    printf(" end of concealment_events_ms\n");
+  }
+
+  const auto lifetime_stats_vector = stats_getter_->lifetime_stats();
+  if (!lifetime_stats_vector->empty()) {
+    auto lifetime_stats = lifetime_stats_vector->back().second;
+    printf("  total_samples_received: %" PRIu64 "\n",
+           lifetime_stats.total_samples_received);
+    printf("  concealed_samples: %" PRIu64 "\n",
+           lifetime_stats.concealed_samples);
+    printf("  concealment_events: %" PRIu64 "\n",
+           lifetime_stats.concealment_events);
+    printf("  delayed_packet_outage_samples: %" PRIu64 "\n",
+           lifetime_stats.delayed_packet_outage_samples);
+    printf("  num_interruptions: %d\n", lifetime_stats.interruption_count);
+    printf("  sum_interruption_length_ms: %d ms\n",
+           lifetime_stats.total_interruption_duration_ms);
+    printf("  interruption_ratio: %f\n",
+           static_cast<double>(lifetime_stats.total_interruption_duration_ms) /
+               simulation_time_ms);
+    printf("  removed_samples_for_acceleration: %" PRIu64 "\n",
+           lifetime_stats.removed_samples_for_acceleration);
+    printf("  inserted_samples_for_deceleration: %" PRIu64 "\n",
+           lifetime_stats.inserted_samples_for_deceleration);
+    printf("  generated_noise_samples: %" PRIu64 "\n",
+           lifetime_stats.generated_noise_samples);
+    printf("  packets_discarded: %" PRIu64 "\n",
+           lifetime_stats.packets_discarded);
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.h
new file mode 100644
index 0000000000..11c16da9d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqStatsPlotter : public NetEqSimulationEndedCallback {
+ public:
+  NetEqStatsPlotter(bool make_matlab_plot,
+                    bool make_python_plot,
+                    bool show_concealment_events,
+                    absl::string_view base_file_name);
+
+  void SimulationEnded(int64_t simulation_time_ms) override;
+
+  NetEqStatsGetter* stats_getter() { return stats_getter_.get(); }
+
+ private:
+  std::unique_ptr<NetEqStatsGetter> stats_getter_;
+  const bool make_matlab_plot_;
+  const bool make_python_plot_;
+  const bool show_concealment_events_;
+  const std::string base_file_name_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.cc
new file mode 100644
index 0000000000..a567efe2de
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -0,0 +1,345 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+
+#include <iomanip>
+#include <iostream>
+
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+absl::optional<NetEq::Operation> ActionToOperations(
+    absl::optional<NetEqSimulator::Action> a) {
+  if (!a) {
+    return absl::nullopt;
+  }
+  switch (*a) {
+    case NetEqSimulator::Action::kAccelerate:
+      return absl::make_optional(NetEq::Operation::kAccelerate);
+    case NetEqSimulator::Action::kExpand:
+      return absl::make_optional(NetEq::Operation::kExpand);
+    case NetEqSimulator::Action::kNormal:
+      return absl::make_optional(NetEq::Operation::kNormal);
+    case NetEqSimulator::Action::kPreemptiveExpand:
+      return absl::make_optional(NetEq::Operation::kPreemptiveExpand);
+  }
+}
+
+std::unique_ptr<NetEq> CreateNetEq(
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+  return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
+}
+
+}  // namespace
+
+void DefaultNetEqTestErrorCallback::OnInsertPacketError(
+    const NetEqInput::PacketData& packet) {
+  std::cerr << "InsertPacket returned an error." << std::endl;
+  std::cerr << "Packet data: " << packet.ToString() << std::endl;
+  RTC_FATAL();
+}
+
+void DefaultNetEqTestErrorCallback::OnGetAudioError() {
+  std::cerr << "GetAudio returned an error." << std::endl;
+  RTC_FATAL();
+}
+
+NetEqTest::NetEqTest(const NetEq::Config& config,
+                     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+                     const DecoderMap& codecs,
+                     std::unique_ptr<std::ofstream> text_log,
+                     NetEqFactory* neteq_factory,
+                     std::unique_ptr<NetEqInput> input,
+                     std::unique_ptr<AudioSink> output,
+                     Callbacks callbacks)
+    : input_(std::move(input)),
+      clock_(Timestamp::Millis(input_->NextEventTime().value_or(0))),
+      neteq_(neteq_factory
+                 ? neteq_factory->CreateNetEq(config, decoder_factory, &clock_)
+                 : CreateNetEq(config, &clock_, decoder_factory)),
+      output_(std::move(output)),
+      callbacks_(callbacks),
+      sample_rate_hz_(config.sample_rate_hz),
+      text_log_(std::move(text_log)) {
+  RTC_CHECK(!config.enable_muted_state)
+      << "The code does not handle enable_muted_state";
+  RegisterDecoders(codecs);
+}
+
+NetEqTest::~NetEqTest() = default;
+
+int64_t NetEqTest::Run() {
+  int64_t simulation_time = 0;
+  SimulationStepResult step_result;
+  do {
+    step_result = RunToNextGetAudio();
+    simulation_time += step_result.simulation_step_ms;
+  } while (!step_result.is_simulation_finished);
+  if (callbacks_.simulation_ended_callback) {
+    callbacks_.simulation_ended_callback->SimulationEnded(simulation_time);
+  }
+  return simulation_time;
+}
+
+NetEqTest::SimulationStepResult NetEqTest::RunToNextGetAudio() {
+  SimulationStepResult result;
+  const int64_t start_time_ms = *input_->NextEventTime();
+  int64_t time_now_ms = clock_.CurrentTime().ms();
+  current_state_.packet_iat_ms.clear();
+
+  while (!input_->ended()) {
+    // Advance time to next event.
+    RTC_DCHECK(input_->NextEventTime());
+    clock_.AdvanceTimeMilliseconds(*input_->NextEventTime() - time_now_ms);
+    time_now_ms = *input_->NextEventTime();
+    // Check if it is time to insert packet.
+    if (input_->NextPacketTime() && time_now_ms >= *input_->NextPacketTime()) {
+      std::unique_ptr<NetEqInput::PacketData> packet_data = input_->PopPacket();
+      RTC_CHECK(packet_data);
+      const size_t payload_data_length =
+          packet_data->payload.size() - packet_data->header.paddingLength;
+      if (payload_data_length != 0) {
+        int error = neteq_->InsertPacket(
+            packet_data->header,
+            rtc::ArrayView<const uint8_t>(packet_data->payload));
+        if (error != NetEq::kOK && callbacks_.error_callback) {
+          callbacks_.error_callback->OnInsertPacketError(*packet_data);
+        }
+        if (callbacks_.post_insert_packet) {
+          callbacks_.post_insert_packet->AfterInsertPacket(*packet_data,
+                                                           neteq_.get());
+        }
+      } else {
+        neteq_->InsertEmptyPacket(packet_data->header);
+      }
+      if (last_packet_time_ms_) {
+        current_state_.packet_iat_ms.push_back(time_now_ms -
+                                               *last_packet_time_ms_);
+      }
+      if (text_log_) {
+        const auto ops_state = neteq_->GetOperationsAndState();
+        const auto delta_wallclock =
+            last_packet_time_ms_ ? (time_now_ms - *last_packet_time_ms_) : -1;
+        const auto delta_timestamp =
+            last_packet_timestamp_
+                ? (static_cast<int64_t>(packet_data->header.timestamp) -
+                   *last_packet_timestamp_) *
+                      1000 / sample_rate_hz_
+                : -1;
+        const auto packet_size_bytes =
+            packet_data->payload.size() == 12
+                ? ByteReader<uint32_t>::ReadLittleEndian(
+                      &packet_data->payload[8])
+                : -1;
+        *text_log_ << "Packet   - wallclock: " << std::setw(5) << time_now_ms
+                   << ", delta wc: " << std::setw(4) << delta_wallclock
+                   << ", seq_no: " << packet_data->header.sequenceNumber
+                   << ", timestamp: " << std::setw(10)
+                   << packet_data->header.timestamp
+                   << ", delta ts: " << std::setw(4) << delta_timestamp
+                   << ", size: " << std::setw(5) << packet_size_bytes
+                   << ", frame size: " << std::setw(3)
+                   << ops_state.current_frame_size_ms
+                   << ", buffer size: " << std::setw(4)
+                   << ops_state.current_buffer_size_ms << std::endl;
+      }
+      last_packet_time_ms_ = absl::make_optional<int>(time_now_ms);
+      last_packet_timestamp_ =
+          absl::make_optional<uint32_t>(packet_data->header.timestamp);
+    }
+
+    // Check if it is time to get output audio.
+    if (input_->NextOutputEventTime() &&
+        time_now_ms >= *input_->NextOutputEventTime()) {
+      if (callbacks_.get_audio_callback) {
+        callbacks_.get_audio_callback->BeforeGetAudio(neteq_.get());
+      }
+      AudioFrame out_frame;
+      bool muted;
+      int error = neteq_->GetAudio(&out_frame, &muted, nullptr,
+                                   ActionToOperations(next_action_));
+      next_action_ = absl::nullopt;
+      RTC_CHECK(!muted) << "The code does not handle enable_muted_state";
+      if (error != NetEq::kOK) {
+        if (callbacks_.error_callback) {
+          callbacks_.error_callback->OnGetAudioError();
+        }
+      } else {
+        sample_rate_hz_ = out_frame.sample_rate_hz_;
+      }
+      if (callbacks_.get_audio_callback) {
+        callbacks_.get_audio_callback->AfterGetAudio(time_now_ms, out_frame,
+                                                     muted, neteq_.get());
+      }
+
+      if (output_) {
+        RTC_CHECK(output_->WriteArray(
+            out_frame.data(),
+            out_frame.samples_per_channel_ * out_frame.num_channels_));
+      }
+
+      input_->AdvanceOutputEvent();
+      result.simulation_step_ms =
+          input_->NextEventTime().value_or(time_now_ms) - start_time_ms;
+      const auto operations_state = neteq_->GetOperationsAndState();
+      current_state_.current_delay_ms = operations_state.current_buffer_size_ms;
+      current_state_.packet_size_ms = operations_state.current_frame_size_ms;
+      current_state_.next_packet_available =
+          operations_state.next_packet_available;
+      current_state_.packet_buffer_flushed =
+          operations_state.packet_buffer_flushes >
+          prev_ops_state_.packet_buffer_flushes;
+      // TODO(ivoc): Add more accurate reporting by tracking the origin of
+      // samples in the sync buffer.
+      result.action_times_ms[Action::kExpand] = 0;
+      result.action_times_ms[Action::kAccelerate] = 0;
+      result.action_times_ms[Action::kPreemptiveExpand] = 0;
+      result.action_times_ms[Action::kNormal] = 0;
+
+      if (out_frame.speech_type_ == AudioFrame::SpeechType::kPLC ||
+          out_frame.speech_type_ == AudioFrame::SpeechType::kPLCCNG) {
+        // Consider the whole frame to be the result of expansion.
+        result.action_times_ms[Action::kExpand] = 10;
+      } else if (operations_state.accelerate_samples -
+                     prev_ops_state_.accelerate_samples >
+                 0) {
+        // Consider the whole frame to be the result of acceleration.
+        result.action_times_ms[Action::kAccelerate] = 10;
+      } else if (operations_state.preemptive_samples -
+                     prev_ops_state_.preemptive_samples >
+                 0) {
+        // Consider the whole frame to be the result of preemptive expansion.
+        result.action_times_ms[Action::kPreemptiveExpand] = 10;
+      } else {
+        // Consider the whole frame to be the result of normal playout.
+        result.action_times_ms[Action::kNormal] = 10;
+      }
+      auto lifetime_stats = LifetimeStats();
+      if (text_log_) {
+        const bool plc =
+            (out_frame.speech_type_ == AudioFrame::SpeechType::kPLC) ||
+            (out_frame.speech_type_ == AudioFrame::SpeechType::kPLCCNG);
+        const bool cng = out_frame.speech_type_ == AudioFrame::SpeechType::kCNG;
+        const bool voice_concealed =
+            (lifetime_stats.concealed_samples -
+             lifetime_stats.silent_concealed_samples) >
+            (prev_lifetime_stats_.concealed_samples -
+             prev_lifetime_stats_.silent_concealed_samples);
+        *text_log_ << "GetAudio - wallclock: " << std::setw(5) << time_now_ms
+                   << ", delta wc: " << std::setw(4)
+                   << (input_->NextEventTime().value_or(time_now_ms) -
+                       start_time_ms)
+                   << ", CNG: " << cng << ", PLC: " << plc
+                   << ", voice concealed: " << voice_concealed
+                   << ", buffer size: " << std::setw(4)
+                   << current_state_.current_delay_ms << std::endl;
+        if (lifetime_stats.packets_discarded >
+            prev_lifetime_stats_.packets_discarded) {
+          *text_log_ << "Discarded "
+                     << (lifetime_stats.packets_discarded -
+                         prev_lifetime_stats_.packets_discarded)
+                     << " primary packets." << std::endl;
+        }
+        if (operations_state.packet_buffer_flushes >
+            prev_ops_state_.packet_buffer_flushes) {
+          *text_log_ << "Flushed packet buffer "
+                     << (operations_state.packet_buffer_flushes -
+                         prev_ops_state_.packet_buffer_flushes)
+                     << " times." << std::endl;
+        }
+      }
+      prev_lifetime_stats_ = lifetime_stats;
+      const bool no_more_packets_to_decode =
+          !input_->NextPacketTime() && !operations_state.next_packet_available;
+      // End the simulation if the gap is too large. This indicates an issue
+      // with the event log file.
+      const bool simulation_step_too_large = result.simulation_step_ms > 1000;
+      if (simulation_step_too_large) {
+        // If we don't reset the step time, the large gap will be included in
+        // the simulation time, which can be a large distortion.
+        result.simulation_step_ms = 10;
+      }
+      result.is_simulation_finished = simulation_step_too_large ||
+                                      no_more_packets_to_decode ||
+                                      input_->ended();
+      prev_ops_state_ = operations_state;
+      return result;
+    }
+  }
+  result.simulation_step_ms =
+      input_->NextEventTime().value_or(time_now_ms) - start_time_ms;
+  result.is_simulation_finished = true;
+  return result;
+}
+
+void NetEqTest::SetNextAction(NetEqTest::Action next_operation) {
+  next_action_ = absl::optional<Action>(next_operation);
+}
+
+NetEqTest::NetEqState NetEqTest::GetNetEqState() {
+  return current_state_;
+}
+
+NetEqNetworkStatistics NetEqTest::SimulationStats() {
+  NetEqNetworkStatistics stats;
+  RTC_CHECK_EQ(neteq_->NetworkStatistics(&stats), 0);
+  return stats;
+}
+
+NetEqLifetimeStatistics NetEqTest::LifetimeStats() const {
+  return neteq_->GetLifetimeStatistics();
+}
+
+NetEqTest::DecoderMap NetEqTest::StandardDecoderMap() {
+  DecoderMap codecs = {
+    {0, SdpAudioFormat("pcmu", 8000, 1)},
+    {8, SdpAudioFormat("pcma", 8000, 1)},
+#ifdef WEBRTC_CODEC_ILBC
+    {102, SdpAudioFormat("ilbc", 8000, 1)},
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    {111, SdpAudioFormat("opus", 48000, 2)},
+#endif
+    {93, SdpAudioFormat("l16", 8000, 1)},
+    {94, SdpAudioFormat("l16", 16000, 1)},
+    {95, SdpAudioFormat("l16", 32000, 1)},
+    {96, SdpAudioFormat("l16", 48000, 1)},
+    {9, SdpAudioFormat("g722", 8000, 1)},
+    {106, SdpAudioFormat("telephone-event", 8000, 1)},
+    {114, SdpAudioFormat("telephone-event", 16000, 1)},
+    {115, SdpAudioFormat("telephone-event", 32000, 1)},
+    {116, SdpAudioFormat("telephone-event", 48000, 1)},
+    {117, SdpAudioFormat("red", 8000, 1)},
+    {13, SdpAudioFormat("cn", 8000, 1)},
+    {98, SdpAudioFormat("cn", 16000, 1)},
+    {99, SdpAudioFormat("cn", 32000, 1)},
+    {100, SdpAudioFormat("cn", 48000, 1)}
+  };
+  return codecs;
+}
+
+void NetEqTest::RegisterDecoders(const DecoderMap& codecs) {
+  for (const auto& c : codecs) {
+    RTC_CHECK(neteq_->RegisterPayloadType(c.first, c.second))
+        << "Cannot register " << c.second.name << " to payload type "
+        << c.first;
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.h
new file mode 100644
index 0000000000..1d3eeda453
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.h
@@ -0,0 +1,129 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
+
+#include <fstream>
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/test/neteq_simulator.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqTestErrorCallback {
+ public:
+  virtual ~NetEqTestErrorCallback() = default;
+  virtual void OnInsertPacketError(const NetEqInput::PacketData& packet) {}
+  virtual void OnGetAudioError() {}
+};
+
+class DefaultNetEqTestErrorCallback : public NetEqTestErrorCallback {
+  void OnInsertPacketError(const NetEqInput::PacketData& packet) override;
+  void OnGetAudioError() override;
+};
+
+class NetEqPostInsertPacket {
+ public:
+  virtual ~NetEqPostInsertPacket() = default;
+  virtual void AfterInsertPacket(const NetEqInput::PacketData& packet,
+                                 NetEq* neteq) = 0;
+};
+
+class NetEqGetAudioCallback {
+ public:
+  virtual ~NetEqGetAudioCallback() = default;
+  virtual void BeforeGetAudio(NetEq* neteq) = 0;
+  virtual void AfterGetAudio(int64_t time_now_ms,
+                             const AudioFrame& audio_frame,
+                             bool muted,
+                             NetEq* neteq) = 0;
+};
+
+class NetEqSimulationEndedCallback {
+ public:
+  virtual ~NetEqSimulationEndedCallback() = default;
+  virtual void SimulationEnded(int64_t simulation_time_ms) = 0;
+};
+
+// Class that provides an input--output test for NetEq. The input (both packets
+// and output events) is provided by a NetEqInput object, while the output is
+// directed to an AudioSink object.
+class NetEqTest : public NetEqSimulator {
+ public:
+  using DecoderMap = std::map<int, SdpAudioFormat>;
+
+  struct Callbacks {
+    NetEqTestErrorCallback* error_callback = nullptr;
+    NetEqPostInsertPacket* post_insert_packet = nullptr;
+    NetEqGetAudioCallback* get_audio_callback = nullptr;
+    NetEqSimulationEndedCallback* simulation_ended_callback = nullptr;
+  };
+
+  // Sets up the test with given configuration, codec mappings, input, ouput,
+  // and callback objects for error reporting.
+  NetEqTest(const NetEq::Config& config,
+            rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+            const DecoderMap& codecs,
+            std::unique_ptr<std::ofstream> text_log,
+            NetEqFactory* neteq_factory,
+            std::unique_ptr<NetEqInput> input,
+            std::unique_ptr<AudioSink> output,
+            Callbacks callbacks);
+
+  ~NetEqTest() override;
+
+  // Runs the test. Returns the duration of the produced audio in ms.
+  int64_t Run() override;
+  // Runs the simulation until we hit the next GetAudio event. If the simulation
+  // is finished, is_simulation_finished will be set to true in the returned
+  // SimulationStepResult.
+  SimulationStepResult RunToNextGetAudio() override;
+
+  void SetNextAction(Action next_operation) override;
+  NetEqState GetNetEqState() override;
+
+  // Returns the statistics from NetEq.
+  NetEqNetworkStatistics SimulationStats();
+  NetEqLifetimeStatistics LifetimeStats() const;
+
+  static DecoderMap StandardDecoderMap();
+
+ private:
+  void RegisterDecoders(const DecoderMap& codecs);
+  std::unique_ptr<NetEqInput> input_;
+  SimulatedClock clock_;
+  absl::optional<Action> next_action_;
+  absl::optional<int> last_packet_time_ms_;
+  std::unique_ptr<NetEq> neteq_;
+  std::unique_ptr<AudioSink> output_;
+  Callbacks callbacks_;
+  int sample_rate_hz_;
+  NetEqState current_state_;
+  NetEqOperationsAndState prev_ops_state_;
+  NetEqLifetimeStatistics prev_lifetime_stats_;
+  absl::optional<uint32_t> last_packet_timestamp_;
+  std::unique_ptr<std::ofstream> text_log_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc
new file mode 100644
index 0000000000..6cd371406c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc
@@ -0,0 +1,342 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
+
+#include <errno.h>
+#include <limits.h>  // For ULONG_MAX returned by strtoul.
+#include <stdio.h>
+#include <stdlib.h>  // For strtoul.
+
+#include <fstream>
+#include <iostream>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+#include "modules/audio_coding/neteq/tools/neteq_event_log_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
+#include "modules/audio_coding/neteq/tools/neteq_stats_plotter.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "modules/audio_coding/neteq/tools/output_audio_file.h"
+#include "modules/audio_coding/neteq/tools/output_wav_file.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "rtc_base/checks.h"
+#include "test/function_audio_decoder_factory.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+absl::optional<int> CodecSampleRate(
+    uint8_t payload_type,
+    webrtc::test::NetEqTestFactory::Config config) {
+  if (payload_type == config.pcmu || payload_type == config.pcma ||
+      payload_type == config.ilbc || payload_type == config.pcm16b ||
+      payload_type == config.cn_nb || payload_type == config.avt)
+    return 8000;
+  if (payload_type == config.isac || payload_type == config.pcm16b_wb ||
+      payload_type == config.g722 || payload_type == config.cn_wb ||
+      payload_type == config.avt_16)
+    return 16000;
+  if (payload_type == config.isac_swb || payload_type == config.pcm16b_swb32 ||
+      payload_type == config.cn_swb32 || payload_type == config.avt_32)
+    return 32000;
+  if (payload_type == config.opus || payload_type == config.pcm16b_swb48 ||
+      payload_type == config.cn_swb48 || payload_type == config.avt_48)
+    return 48000;
+  if (payload_type == config.red)
+    return 0;
+  return absl::nullopt;
+}
+
+}  // namespace
+
+// A callback class which prints whenver the inserted packet stream changes
+// the SSRC.
+class SsrcSwitchDetector : public NetEqPostInsertPacket {
+ public:
+  // Takes a pointer to another callback object, which will be invoked after
+  // this object finishes. This does not transfer ownership, and null is a
+  // valid value.
+  explicit SsrcSwitchDetector(NetEqPostInsertPacket* other_callback)
+      : other_callback_(other_callback) {}
+
+  void AfterInsertPacket(const NetEqInput::PacketData& packet,
+                         NetEq* neteq) override {
+    if (last_ssrc_ && packet.header.ssrc != *last_ssrc_) {
+      std::cout << "Changing streams from 0x" << std::hex << *last_ssrc_
+                << " to 0x" << std::hex << packet.header.ssrc << std::dec
+                << " (payload type "
+                << static_cast<int>(packet.header.payloadType) << ")"
+                << std::endl;
+    }
+    last_ssrc_ = packet.header.ssrc;
+    if (other_callback_) {
+      other_callback_->AfterInsertPacket(packet, neteq);
+    }
+  }
+
+ private:
+  NetEqPostInsertPacket* other_callback_;
+  absl::optional<uint32_t> last_ssrc_;
+};
+
+NetEqTestFactory::NetEqTestFactory() = default;
+NetEqTestFactory::~NetEqTestFactory() = default;
+
+NetEqTestFactory::Config::Config() = default;
+NetEqTestFactory::Config::Config(const Config& other) = default;
+NetEqTestFactory::Config::~Config() = default;
+
+std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTestFromString(
+    absl::string_view input_string,
+    NetEqFactory* factory,
+    const Config& config) {
+  std::unique_ptr<NetEqInput> input(
+      NetEqEventLogInput::CreateFromString(input_string, config.ssrc_filter));
+  if (!input) {
+    std::cerr << "Error: Cannot parse input string" << std::endl;
+    return nullptr;
+  }
+  return InitializeTest(std::move(input), factory, config);
+}
+
+std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTestFromFile(
+    absl::string_view input_file_name,
+    NetEqFactory* factory,
+    const Config& config) {
+  // Gather RTP header extensions in a map.
+  NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
+      {config.audio_level, kRtpExtensionAudioLevel},
+      {config.abs_send_time, kRtpExtensionAbsoluteSendTime},
+      {config.transport_seq_no, kRtpExtensionTransportSequenceNumber},
+      {config.video_content_type, kRtpExtensionVideoContentType},
+      {config.video_timing, kRtpExtensionVideoTiming}};
+
+  std::unique_ptr<NetEqInput> input;
+  if (RtpFileSource::ValidRtpDump(input_file_name) ||
+      RtpFileSource::ValidPcap(input_file_name)) {
+    input.reset(new NetEqRtpDumpInput(input_file_name, rtp_ext_map,
+                                      config.ssrc_filter));
+  } else {
+    input.reset(NetEqEventLogInput::CreateFromFile(input_file_name,
+                                                   config.ssrc_filter));
+  }
+
+  std::cout << "Input file: " << input_file_name << std::endl;
+  if (!input) {
+    std::cerr << "Error: Cannot open input file" << std::endl;
+    return nullptr;
+  }
+  return InitializeTest(std::move(input), factory, config);
+}
+
+std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTest(
+    std::unique_ptr<NetEqInput> input,
+    NetEqFactory* factory,
+    const Config& config) {
+  if (input->ended()) {
+    std::cerr << "Error: Input is empty" << std::endl;
+    return nullptr;
+  }
+
+  if (!config.field_trial_string.empty()) {
+    field_trials_ =
+        std::make_unique<ScopedFieldTrials>(config.field_trial_string);
+  }
+
+  // Skip some initial events/packets if requested.
+  if (config.skip_get_audio_events > 0) {
+    std::cout << "Skipping " << config.skip_get_audio_events
+              << " get_audio events" << std::endl;
+    if (!input->NextPacketTime() || !input->NextOutputEventTime()) {
+      std::cerr << "No events found" << std::endl;
+      return nullptr;
+    }
+    for (int i = 0; i < config.skip_get_audio_events; i++) {
+      input->AdvanceOutputEvent();
+      if (!input->NextOutputEventTime()) {
+        std::cerr << "Not enough get_audio events found" << std::endl;
+        return nullptr;
+      }
+    }
+    while (*input->NextPacketTime() < *input->NextOutputEventTime()) {
+      input->PopPacket();
+      if (!input->NextPacketTime()) {
+        std::cerr << "Not enough incoming packets found" << std::endl;
+        return nullptr;
+      }
+    }
+  }
+
+  // Check the sample rate.
+  absl::optional<int> sample_rate_hz;
+  std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
+  while (absl::optional<RTPHeader> first_rtp_header = input->NextHeader()) {
+    RTC_DCHECK(first_rtp_header);
+    sample_rate_hz = CodecSampleRate(first_rtp_header->payloadType, config);
+    if (sample_rate_hz) {
+      std::cout << "Found valid packet with payload type "
+                << static_cast<int>(first_rtp_header->payloadType)
+                << " and SSRC 0x" << std::hex << first_rtp_header->ssrc
+                << std::dec << std::endl;
+      if (config.initial_dummy_packets > 0) {
+        std::cout << "Nr of initial dummy packets: "
+                  << config.initial_dummy_packets << std::endl;
+        input = std::make_unique<InitialPacketInserterNetEqInput>(
+            std::move(input), config.initial_dummy_packets, *sample_rate_hz);
+      }
+      break;
+    }
+    // Discard this packet and move to the next. Keep track of discarded payload
+    // types and SSRCs.
+    discarded_pt_and_ssrc.emplace(first_rtp_header->payloadType,
+                                  first_rtp_header->ssrc);
+    input->PopPacket();
+  }
+  if (!discarded_pt_and_ssrc.empty()) {
+    std::cout << "Discarded initial packets with the following payload types "
+                 "and SSRCs:"
+              << std::endl;
+    for (const auto& d : discarded_pt_and_ssrc) {
+      std::cout << "PT " << d.first << "; SSRC 0x" << std::hex
+                << static_cast<int>(d.second) << std::dec << std::endl;
+    }
+  }
+  if (!sample_rate_hz) {
+    std::cerr << "Cannot find any packets with known payload types"
+              << std::endl;
+    return nullptr;
+  }
+
+  // If an output file is requested, open it.
+  std::unique_ptr<AudioSink> output;
+  if (!config.output_audio_filename.has_value()) {
+    output = std::make_unique<VoidAudioSink>();
+    std::cout << "No output audio file" << std::endl;
+  } else if (config.output_audio_filename->size() >= 4 &&
+             config.output_audio_filename->substr(
+                 config.output_audio_filename->size() - 4) == ".wav") {
+    // Open a wav file with the known sample rate.
+    output = std::make_unique<OutputWavFile>(*config.output_audio_filename,
+                                             *sample_rate_hz);
+    std::cout << "Output WAV file: " << *config.output_audio_filename
+              << std::endl;
+  } else {
+    // Open a pcm file.
+    output = std::make_unique<OutputAudioFile>(*config.output_audio_filename);
+    std::cout << "Output PCM file: " << *config.output_audio_filename
+              << std::endl;
+  }
+
+  NetEqTest::DecoderMap codecs = NetEqTest::StandardDecoderMap();
+
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
+      CreateBuiltinAudioDecoderFactory();
+
+  // Check if a replacement audio file was provided.
+  if (config.replacement_audio_file.size() > 0) {
+    // Find largest unused payload type.
+    int replacement_pt = 127;
+    while (codecs.find(replacement_pt) != codecs.end()) {
+      --replacement_pt;
+      if (replacement_pt <= 0) {
+        std::cerr << "Error: Unable to find available replacement payload type"
+                  << std::endl;
+        return nullptr;
+      }
+    }
+
+    auto std_set_int32_to_uint8 = [](const std::set<int32_t>& a) {
+      std::set<uint8_t> b;
+      for (auto& x : a) {
+        b.insert(static_cast<uint8_t>(x));
+      }
+      return b;
+    };
+
+    std::set<uint8_t> cn_types = std_set_int32_to_uint8(
+        {config.cn_nb, config.cn_wb, config.cn_swb32, config.cn_swb48});
+    std::set<uint8_t> forbidden_types =
+        std_set_int32_to_uint8({config.g722, config.red, config.avt,
+                                config.avt_16, config.avt_32, config.avt_48});
+    input.reset(new NetEqReplacementInput(std::move(input), replacement_pt,
+                                          cn_types, forbidden_types));
+
+    // Note that capture-by-copy implies that the lambda captures the value of
+    // decoder_factory before it's reassigned on the left-hand side.
+    decoder_factory = rtc::make_ref_counted<FunctionAudioDecoderFactory>(
+        [decoder_factory, config](
+            const SdpAudioFormat& format,
+            absl::optional<AudioCodecPairId> codec_pair_id) {
+          std::unique_ptr<AudioDecoder> decoder =
+              decoder_factory->MakeAudioDecoder(format, codec_pair_id);
+          if (!decoder && format.name == "replacement") {
+            decoder = std::make_unique<FakeDecodeFromFile>(
+                std::make_unique<InputAudioFile>(config.replacement_audio_file),
+                format.clockrate_hz, format.num_channels > 1);
+          }
+          return decoder;
+        });
+
+    if (!codecs
+             .insert({replacement_pt, SdpAudioFormat("replacement", 48000, 1)})
+             .second) {
+      std::cerr << "Error: Unable to insert replacement audio codec"
+                << std::endl;
+      return nullptr;
+    }
+  }
+
+  // Create a text log output stream if needed.
+  std::unique_ptr<std::ofstream> text_log;
+  if (config.textlog && config.textlog_filename.has_value()) {
+    // Write to file.
+    text_log = std::make_unique<std::ofstream>(*config.textlog_filename);
+  } else if (config.textlog) {
+    // Print to stdout.
+    text_log = std::make_unique<std::ofstream>();
+    text_log->basic_ios<char>::rdbuf(std::cout.rdbuf());
+  }
+
+  NetEqTest::Callbacks callbacks;
+  stats_plotter_ = std::make_unique<NetEqStatsPlotter>(
+      config.matlabplot, config.pythonplot, config.concealment_events,
+      config.plot_scripts_basename.value_or(""));
+
+  ssrc_switch_detector_.reset(
+      new SsrcSwitchDetector(stats_plotter_->stats_getter()->delay_analyzer()));
+  callbacks.post_insert_packet = ssrc_switch_detector_.get();
+  callbacks.get_audio_callback = stats_plotter_->stats_getter();
+  callbacks.simulation_ended_callback = stats_plotter_.get();
+  NetEq::Config neteq_config;
+  neteq_config.sample_rate_hz = *sample_rate_hz;
+  neteq_config.max_packets_in_buffer = config.max_nr_packets_in_buffer;
+  neteq_config.enable_fast_accelerate = config.enable_fast_accelerate;
+  return std::make_unique<NetEqTest>(
+      neteq_config, decoder_factory, codecs, std::move(text_log), factory,
+      std::move(input), std::move(output), callbacks);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.h
new file mode 100644
index 0000000000..96ce0b4334
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.h
@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "test/field_trial.h"
+
+namespace webrtc {
+namespace test {
+
+class SsrcSwitchDetector;
+class NetEqStatsGetter;
+class NetEqStatsPlotter;
+
+// Note that the NetEqTestFactory needs to be alive when the NetEqTest object is
+// used for a simulation.
+class NetEqTestFactory {
+ public:
+  NetEqTestFactory();
+  ~NetEqTestFactory();
+  struct Config {
+    Config();
+    Config(const Config& other);
+    ~Config();
+    // RTP payload type for PCM-u.
+    static constexpr int default_pcmu() { return 0; }
+    int pcmu = default_pcmu();
+    // RTP payload type for PCM-a.
+    static constexpr int default_pcma() { return 8; }
+    int pcma = default_pcma();
+    // RTP payload type for iLBC.
+    static constexpr int default_ilbc() { return 102; }
+    int ilbc = default_ilbc();
+    // RTP payload type for iSAC.
+    static constexpr int default_isac() { return 103; }
+    int isac = default_isac();
+    // RTP payload type for iSAC-swb (32 kHz).
+    static constexpr int default_isac_swb() { return 104; }
+    int isac_swb = default_isac_swb();
+    // RTP payload type for Opus.
+    static constexpr int default_opus() { return 111; }
+    int opus = default_opus();
+    // RTP payload type for PCM16b-nb (8 kHz).
+    static constexpr int default_pcm16b() { return 93; }
+    int pcm16b = default_pcm16b();
+    // RTP payload type for PCM16b-wb (16 kHz).
+    static constexpr int default_pcm16b_wb() { return 94; }
+    int pcm16b_wb = default_pcm16b_wb();
+    // RTP payload type for PCM16b-swb32 (32 kHz).
+    static constexpr int default_pcm16b_swb32() { return 95; }
+    int pcm16b_swb32 = default_pcm16b_swb32();
+    // RTP payload type for PCM16b-swb48 (48 kHz).
+    static constexpr int default_pcm16b_swb48() { return 96; }
+    int pcm16b_swb48 = default_pcm16b_swb48();
+    // RTP payload type for G.722.
+    static constexpr int default_g722() { return 9; }
+    int g722 = default_g722();
+    // RTP payload type for AVT/DTMF (8 kHz).
+    static constexpr int default_avt() { return 106; }
+    int avt = default_avt();
+    // RTP payload type for AVT/DTMF (16 kHz).
+    static constexpr int default_avt_16() { return 114; }
+    int avt_16 = default_avt_16();
+    // RTP payload type for AVT/DTMF (32 kHz).
+    static constexpr int default_avt_32() { return 115; }
+    int avt_32 = default_avt_32();
+    // RTP payload type for AVT/DTMF (48 kHz).
+    static constexpr int default_avt_48() { return 116; }
+    int avt_48 = default_avt_48();
+    // RTP payload type for redundant audio (RED).
+    static constexpr int default_red() { return 117; }
+    int red = default_red();
+    // RTP payload type for comfort noise (8 kHz).
+    static constexpr int default_cn_nb() { return 13; }
+    int cn_nb = default_cn_nb();
+    // RTP payload type for comfort noise (16 kHz).
+    static constexpr int default_cn_wb() { return 98; }
+    int cn_wb = default_cn_wb();
+    // RTP payload type for comfort noise (32 kHz).
+    static constexpr int default_cn_swb32() { return 99; }
+    int cn_swb32 = default_cn_swb32();
+    // RTP payload type for comfort noise (48 kHz).
+    static constexpr int default_cn_swb48() { return 100; }
+    int cn_swb48 = default_cn_swb48();
+    // A PCM file that will be used to populate dummy RTP packets.
+    std::string replacement_audio_file;
+    // Only use packets with this SSRC.
+    absl::optional<uint32_t> ssrc_filter;
+    // Extension ID for audio level (RFC 6464).
+    static constexpr int default_audio_level() { return 1; }
+    int audio_level = default_audio_level();
+    // Extension ID for absolute sender time.
+    static constexpr int default_abs_send_time() { return 3; }
+    int abs_send_time = default_abs_send_time();
+    // Extension ID for transport sequence number.
+    static constexpr int default_transport_seq_no() { return 5; }
+    int transport_seq_no = default_transport_seq_no();
+    // Extension ID for video content type.
+    static constexpr int default_video_content_type() { return 7; }
+    int video_content_type = default_video_content_type();
+    // Extension ID for video timing.
+    static constexpr int default_video_timing() { return 8; }
+    int video_timing = default_video_timing();
+    // Generate a matlab script for plotting the delay profile.
+    bool matlabplot = false;
+    // Generates a python script for plotting the delay profile.
+    bool pythonplot = false;
+    // Prints concealment events.
+    bool concealment_events = false;
+    // Maximum allowed number of packets in the buffer.
+    static constexpr int default_max_nr_packets_in_buffer() { return 200; }
+    int max_nr_packets_in_buffer = default_max_nr_packets_in_buffer();
+    // Number of dummy packets to put in the packet buffer at the start of the
+    // simulation.
+    static constexpr int default_initial_dummy_packets() { return 0; }
+    int initial_dummy_packets = default_initial_dummy_packets();
+    // Number of getAudio events to skip at the start of the simulation.
+    static constexpr int default_skip_get_audio_events() { return 0; }
+    int skip_get_audio_events = default_skip_get_audio_events();
+    // Enables jitter buffer fast accelerate.
+    bool enable_fast_accelerate = false;
+    // Dumps events that describes the simulation on a step-by-step basis.
+    bool textlog = false;
+    // If specified and `textlog` is true, the output of `textlog` is written to
+    // the specified file name.
+    absl::optional<std::string> textlog_filename;
+    // Base name for the output script files for plotting the delay profile.
+    absl::optional<std::string> plot_scripts_basename;
+    // Path to the output audio file.
+    absl::optional<std::string> output_audio_filename;
+    // Field trials to use during the simulation.
+    std::string field_trial_string;
+  };
+
+  std::unique_ptr<NetEqTest> InitializeTestFromFile(
+      absl::string_view input_filename,
+      NetEqFactory* neteq_factory,
+      const Config& config);
+  std::unique_ptr<NetEqTest> InitializeTestFromString(
+      absl::string_view input_string,
+      NetEqFactory* neteq_factory,
+      const Config& config);
+
+ private:
+  std::unique_ptr<NetEqTest> InitializeTest(std::unique_ptr<NetEqInput> input,
+                                            NetEqFactory* neteq_factory,
+                                            const Config& config);
+  std::unique_ptr<SsrcSwitchDetector> ssrc_switch_detector_;
+  std::unique_ptr<NetEqStatsPlotter> stats_plotter_;
+  // The field trials are stored in the test factory, because neteq_test is not
+  // in a testonly target, and therefore cannot use ScopedFieldTrials.
+  std::unique_ptr<ScopedFieldTrials> field_trials_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_audio_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_audio_file.h
new file mode 100644
index 0000000000..25577fc882
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
+
+#include <stdio.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+
+namespace webrtc {
+namespace test {
+
+class OutputAudioFile : public AudioSink {
+ public:
+  // Creates an OutputAudioFile, opening a file named `file_name` for writing.
+  // The file format is 16-bit signed host-endian PCM.
+  explicit OutputAudioFile(absl::string_view file_name) {
+    out_file_ = fopen(std::string(file_name).c_str(), "wb");
+  }
+
+  virtual ~OutputAudioFile() {
+    if (out_file_)
+      fclose(out_file_);
+  }
+
+  OutputAudioFile(const OutputAudioFile&) = delete;
+  OutputAudioFile& operator=(const OutputAudioFile&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
+    RTC_DCHECK(out_file_);
+    return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
+  }
+
+ private:
+  FILE* out_file_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_wav_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_wav_file.h
new file mode 100644
index 0000000000..20eedfb554
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_wav_file.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+
+namespace webrtc {
+namespace test {
+
+class OutputWavFile : public AudioSink {
+ public:
+  // Creates an OutputWavFile, opening a file named `file_name` for writing.
+  // The output file is a PCM encoded wav file.
+  OutputWavFile(absl::string_view file_name,
+                int sample_rate_hz,
+                int num_channels = 1)
+      : wav_writer_(file_name, sample_rate_hz, num_channels) {}
+
+  OutputWavFile(const OutputWavFile&) = delete;
+  OutputWavFile& operator=(const OutputWavFile&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
+    wav_writer_.WriteSamples(audio, num_samples);
+    return true;
+  }
+
+ private:
+  WavWriter wav_writer_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.cc
new file mode 100644
index 0000000000..e540173f43
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.cc
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/packet.h"
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+namespace test {
+
+Packet::Packet(rtc::CopyOnWriteBuffer packet,
+               size_t virtual_packet_length_bytes,
+               double time_ms,
+               const RtpHeaderExtensionMap* extension_map)
+    : packet_(std::move(packet)),
+      virtual_packet_length_bytes_(virtual_packet_length_bytes),
+      time_ms_(time_ms),
+      valid_header_(ParseHeader(extension_map)) {}
+
+Packet::Packet(const RTPHeader& header,
+               size_t virtual_packet_length_bytes,
+               size_t virtual_payload_length_bytes,
+               double time_ms)
+    : header_(header),
+      virtual_packet_length_bytes_(virtual_packet_length_bytes),
+      virtual_payload_length_bytes_(virtual_payload_length_bytes),
+      time_ms_(time_ms),
+      valid_header_(true) {}
+
+Packet::~Packet() = default;
+
+bool Packet::ExtractRedHeaders(std::list<RTPHeader*>* headers) const {
+  //
+  //  0                   1                    2                   3
+  //  0 1 2 3 4 5 6 7 8 9 0 1 2 3  4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |1|   block PT  |  timestamp offset         |   block length    |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |1|    ...                                                      |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |0|   block PT  |
+  // +-+-+-+-+-+-+-+-+
+  //
+
+  const uint8_t* payload_ptr = payload();
+  const uint8_t* payload_end_ptr = payload_ptr + payload_length_bytes();
+
+  // Find all RED headers with the extension bit set to 1. That is, all headers
+  // but the last one.
+  while ((payload_ptr < payload_end_ptr) && (*payload_ptr & 0x80)) {
+    RTPHeader* header = new RTPHeader;
+    CopyToHeader(header);
+    header->payloadType = payload_ptr[0] & 0x7F;
+    uint32_t offset = (payload_ptr[1] << 6) + ((payload_ptr[2] & 0xFC) >> 2);
+    header->timestamp -= offset;
+    headers->push_front(header);
+    payload_ptr += 4;
+  }
+  // Last header.
+  RTC_DCHECK_LT(payload_ptr, payload_end_ptr);
+  if (payload_ptr >= payload_end_ptr) {
+    return false;  // Payload too short.
+  }
+  RTPHeader* header = new RTPHeader;
+  CopyToHeader(header);
+  header->payloadType = payload_ptr[0] & 0x7F;
+  headers->push_front(header);
+  return true;
+}
+
+void Packet::DeleteRedHeaders(std::list<RTPHeader*>* headers) {
+  while (!headers->empty()) {
+    delete headers->front();
+    headers->pop_front();
+  }
+}
+
+bool Packet::ParseHeader(const RtpHeaderExtensionMap* extension_map) {
+  // Use RtpPacketReceived instead of RtpPacket because former already has a
+  // converter into legacy RTPHeader.
+  webrtc::RtpPacketReceived rtp_packet(extension_map);
+
+  // Because of the special case of dummy packets that have padding marked in
+  // the RTP header, but do not have rtp payload with the padding size, handle
+  // padding manually. Regular RTP packet parser reports failure, but it is fine
+  // in this context.
+  bool padding = (packet_[0] & 0b0010'0000);
+  size_t padding_size = 0;
+  if (padding) {
+    // Clear the padding bit to prevent failure when rtp payload is omited.
+    rtc::CopyOnWriteBuffer packet(packet_);
+    packet.MutableData()[0] &= ~0b0010'0000;
+    if (!rtp_packet.Parse(std::move(packet))) {
+      return false;
+    }
+    if (rtp_packet.payload_size() > 0) {
+      padding_size = rtp_packet.data()[rtp_packet.size() - 1];
+    }
+    if (padding_size > rtp_packet.payload_size()) {
+      return false;
+    }
+  } else {
+    if (!rtp_packet.Parse(packet_)) {
+      return false;
+    }
+  }
+  rtp_payload_ = rtc::MakeArrayView(packet_.data() + rtp_packet.headers_size(),
+                                    rtp_packet.payload_size() - padding_size);
+  rtp_packet.GetHeader(&header_);
+
+  RTC_CHECK_GE(virtual_packet_length_bytes_, rtp_packet.size());
+  RTC_DCHECK_GE(virtual_packet_length_bytes_, rtp_packet.headers_size());
+  virtual_payload_length_bytes_ =
+      virtual_packet_length_bytes_ - rtp_packet.headers_size();
+  return true;
+}
+
+void Packet::CopyToHeader(RTPHeader* destination) const {
+  *destination = header_;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.h
new file mode 100644
index 0000000000..96710907df
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.h
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
+
+#include <list>
+
+#include "api/array_view.h"
+#include "api/rtp_headers.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for handling RTP packets in test applications.
+class Packet {
+ public:
+  // Creates a packet, with the packet payload (including header bytes) in
+  // `packet`. The `time_ms` is an extra time associated with this packet,
+  // typically used to denote arrival time.
+  // `virtual_packet_length_bytes` is typically used when reading RTP dump files
+  // that only contain the RTP headers, and no payload (a.k.a RTP dummy files or
+  // RTP light). The `virtual_packet_length_bytes` tells what size the packet
+  // had on wire, including the now discarded payload.
+  Packet(rtc::CopyOnWriteBuffer packet,
+         size_t virtual_packet_length_bytes,
+         double time_ms,
+         const RtpHeaderExtensionMap* extension_map = nullptr);
+
+  Packet(rtc::CopyOnWriteBuffer packet,
+         double time_ms,
+         const RtpHeaderExtensionMap* extension_map = nullptr)
+      : Packet(packet, packet.size(), time_ms, extension_map) {}
+
+  // Same as above, but creates the packet from an already parsed RTPHeader.
+  // This is typically used when reading RTP dump files that only contain the
+  // RTP headers, and no payload. The `virtual_packet_length_bytes` tells what
+  // size the packet had on wire, including the now discarded payload,
+  // The `virtual_payload_length_bytes` tells the size of the payload.
+  Packet(const RTPHeader& header,
+         size_t virtual_packet_length_bytes,
+         size_t virtual_payload_length_bytes,
+         double time_ms);
+
+  virtual ~Packet();
+
+  Packet(const Packet&) = delete;
+  Packet& operator=(const Packet&) = delete;
+
+  // Parses the first bytes of the RTP payload, interpreting them as RED headers
+  // according to RFC 2198. The headers will be inserted into `headers`. The
+  // caller of the method assumes ownership of the objects in the list, and
+  // must delete them properly.
+  bool ExtractRedHeaders(std::list<RTPHeader*>* headers) const;
+
+  // Deletes all RTPHeader objects in `headers`, but does not delete `headers`
+  // itself.
+  static void DeleteRedHeaders(std::list<RTPHeader*>* headers);
+
+  const uint8_t* payload() const { return rtp_payload_.data(); }
+
+  size_t packet_length_bytes() const { return packet_.size(); }
+
+  size_t payload_length_bytes() const { return rtp_payload_.size(); }
+
+  size_t virtual_packet_length_bytes() const {
+    return virtual_packet_length_bytes_;
+  }
+
+  size_t virtual_payload_length_bytes() const {
+    return virtual_payload_length_bytes_;
+  }
+
+  const RTPHeader& header() const { return header_; }
+
+  double time_ms() const { return time_ms_; }
+  bool valid_header() const { return valid_header_; }
+
+ private:
+  bool ParseHeader(const RtpHeaderExtensionMap* extension_map);
+  void CopyToHeader(RTPHeader* destination) const;
+
+  RTPHeader header_;
+  const rtc::CopyOnWriteBuffer packet_;
+  rtc::ArrayView<const uint8_t> rtp_payload_;  // Empty for dummy RTP packets.
+  // Virtual lengths are used when parsing RTP header files (dummy RTP files).
+  const size_t virtual_packet_length_bytes_;
+  size_t virtual_payload_length_bytes_ = 0;
+  const double time_ms_;     // Used to denote a packet's arrival time.
+  const bool valid_header_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.cc
new file mode 100644
index 0000000000..598ae6edd4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.cc
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+
+namespace webrtc {
+namespace test {
+
+PacketSource::PacketSource() = default;
+
+PacketSource::~PacketSource() = default;
+
+void PacketSource::FilterOutPayloadType(uint8_t payload_type) {
+  filter_.set(payload_type, true);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.h
new file mode 100644
index 0000000000..be1705cae1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+
+#include <bitset>
+#include <memory>
+
+#include "modules/audio_coding/neteq/tools/packet.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for an object delivering RTP packets to test applications.
+class PacketSource {
+ public:
+  PacketSource();
+  virtual ~PacketSource();
+
+  PacketSource(const PacketSource&) = delete;
+  PacketSource& operator=(const PacketSource&) = delete;
+
+  // Returns next packet. Returns nullptr if the source is depleted, or if an
+  // error occurred.
+  virtual std::unique_ptr<Packet> NextPacket() = 0;
+
+  virtual void FilterOutPayloadType(uint8_t payload_type);
+
+ protected:
+  std::bitset<128> filter_;  // Payload type is 7 bits in the RFC.
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_unittest.cc
new file mode 100644
index 0000000000..69cf56b529
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_unittest.cc
@@ -0,0 +1,226 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Unit tests for test Packet class.
+
+#include "modules/audio_coding/neteq/tools/packet.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+const int kHeaderLengthBytes = 12;
+
+void MakeRtpHeader(int payload_type,
+                   int seq_number,
+                   uint32_t timestamp,
+                   uint32_t ssrc,
+                   uint8_t* rtp_data) {
+  rtp_data[0] = 0x80;
+  rtp_data[1] = static_cast<uint8_t>(payload_type);
+  rtp_data[2] = (seq_number >> 8) & 0xFF;
+  rtp_data[3] = (seq_number)&0xFF;
+  rtp_data[4] = timestamp >> 24;
+  rtp_data[5] = (timestamp >> 16) & 0xFF;
+  rtp_data[6] = (timestamp >> 8) & 0xFF;
+  rtp_data[7] = timestamp & 0xFF;
+  rtp_data[8] = ssrc >> 24;
+  rtp_data[9] = (ssrc >> 16) & 0xFF;
+  rtp_data[10] = (ssrc >> 8) & 0xFF;
+  rtp_data[11] = ssrc & 0xFF;
+}
+}  // namespace
+
+TEST(TestPacket, RegularPacket) {
+  const size_t kPacketLengthBytes = 100;
+  rtc::CopyOnWriteBuffer packet_memory(kPacketLengthBytes);
+  const uint8_t kPayloadType = 17;
+  const uint16_t kSequenceNumber = 4711;
+  const uint32_t kTimestamp = 47114711;
+  const uint32_t kSsrc = 0x12345678;
+  MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc,
+                packet_memory.MutableData());
+  const double kPacketTime = 1.0;
+  Packet packet(std::move(packet_memory), kPacketTime);
+  ASSERT_TRUE(packet.valid_header());
+  EXPECT_EQ(kPayloadType, packet.header().payloadType);
+  EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber);
+  EXPECT_EQ(kTimestamp, packet.header().timestamp);
+  EXPECT_EQ(kSsrc, packet.header().ssrc);
+  EXPECT_EQ(0, packet.header().numCSRCs);
+  EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.payload_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes, packet.virtual_packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.virtual_payload_length_bytes());
+  EXPECT_EQ(kPacketTime, packet.time_ms());
+}
+
+TEST(TestPacket, DummyPacket) {
+  const size_t kPacketLengthBytes = kHeaderLengthBytes;  // Only RTP header.
+  const size_t kVirtualPacketLengthBytes = 100;
+  rtc::CopyOnWriteBuffer packet_memory(kPacketLengthBytes);
+  const uint8_t kPayloadType = 17;
+  const uint16_t kSequenceNumber = 4711;
+  const uint32_t kTimestamp = 47114711;
+  const uint32_t kSsrc = 0x12345678;
+  MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc,
+                packet_memory.MutableData());
+  const double kPacketTime = 1.0;
+  Packet packet(std::move(packet_memory), kVirtualPacketLengthBytes,
+                kPacketTime);
+  ASSERT_TRUE(packet.valid_header());
+  EXPECT_EQ(kPayloadType, packet.header().payloadType);
+  EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber);
+  EXPECT_EQ(kTimestamp, packet.header().timestamp);
+  EXPECT_EQ(kSsrc, packet.header().ssrc);
+  EXPECT_EQ(0, packet.header().numCSRCs);
+  EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.payload_length_bytes());
+  EXPECT_EQ(kVirtualPacketLengthBytes, packet.virtual_packet_length_bytes());
+  EXPECT_EQ(kVirtualPacketLengthBytes - kHeaderLengthBytes,
+            packet.virtual_payload_length_bytes());
+  EXPECT_EQ(kPacketTime, packet.time_ms());
+}
+
+TEST(TestPacket, DummyPaddingPacket) {
+  const size_t kPacketLengthBytes = kHeaderLengthBytes;  // Only RTP header.
+  const size_t kVirtualPacketLengthBytes = 100;
+  rtc::CopyOnWriteBuffer packet_memory(kPacketLengthBytes);
+  const uint8_t kPayloadType = 17;
+  const uint16_t kSequenceNumber = 4711;
+  const uint32_t kTimestamp = 47114711;
+  const uint32_t kSsrc = 0x12345678;
+  MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc,
+                packet_memory.MutableData());
+  packet_memory.MutableData()[0] |= 0b0010'0000;  // Set the padding bit.
+  const double kPacketTime = 1.0;
+  Packet packet(std::move(packet_memory), kVirtualPacketLengthBytes,
+                kPacketTime);
+  ASSERT_TRUE(packet.valid_header());
+  EXPECT_EQ(kPayloadType, packet.header().payloadType);
+  EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber);
+  EXPECT_EQ(kTimestamp, packet.header().timestamp);
+  EXPECT_EQ(kSsrc, packet.header().ssrc);
+  EXPECT_EQ(0, packet.header().numCSRCs);
+  EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.payload_length_bytes());
+  EXPECT_EQ(kVirtualPacketLengthBytes, packet.virtual_packet_length_bytes());
+  EXPECT_EQ(kVirtualPacketLengthBytes - kHeaderLengthBytes,
+            packet.virtual_payload_length_bytes());
+  EXPECT_EQ(kPacketTime, packet.time_ms());
+}
+
+namespace {
+// Writes one RED block header starting at `rtp_data`, according to RFC 2198.
+// returns the number of bytes written (1 or 4).
+//
+// Format if `last_payoad` is false:
+// 0                   1                    2                   3
+// 0 1 2 3 4 5 6 7 8 9 0 1 2 3  4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |1|   block PT  |  timestamp offset         |   block length    |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Format if `last_payoad` is true:
+// 0 1 2 3 4 5 6 7
+// +-+-+-+-+-+-+-+-+
+// |0|   Block PT  |
+// +-+-+-+-+-+-+-+-+
+
+int MakeRedHeader(int payload_type,
+                  uint32_t timestamp_offset,
+                  int block_length,
+                  bool last_payload,
+                  uint8_t* rtp_data) {
+  rtp_data[0] = 0x80 | (payload_type & 0x7F);  // Set the first bit to 1.
+  if (last_payload) {
+    rtp_data[0] &= 0x7F;  // Reset the first but to 0 to indicate last block.
+    return 1;
+  }
+  rtp_data[1] = timestamp_offset >> 6;
+  rtp_data[2] = (timestamp_offset & 0x3F) << 2;
+  rtp_data[2] |= block_length >> 8;
+  rtp_data[3] = block_length & 0xFF;
+  return 4;
+}
+}  // namespace
+
+TEST(TestPacket, RED) {
+  const size_t kPacketLengthBytes = 100;
+  rtc::CopyOnWriteBuffer packet_memory(kPacketLengthBytes);
+  const uint8_t kRedPayloadType = 17;
+  const uint16_t kSequenceNumber = 4711;
+  const uint32_t kTimestamp = 47114711;
+  const uint32_t kSsrc = 0x12345678;
+  MakeRtpHeader(kRedPayloadType, kSequenceNumber, kTimestamp, kSsrc,
+                packet_memory.MutableData());
+  // Create four RED headers.
+  // Payload types are just the same as the block index the offset is 100 times
+  // the block index.
+  const int kRedBlocks = 4;
+  uint8_t* payload_ptr = packet_memory.MutableData() +
+                         kHeaderLengthBytes;  // First byte after header.
+  for (int i = 0; i < kRedBlocks; ++i) {
+    int payload_type = i;
+    // Offset value is not used for the last block.
+    uint32_t timestamp_offset = 100 * i;
+    int block_length = 10 * i;
+    bool last_block = (i == kRedBlocks - 1) ? true : false;
+    payload_ptr += MakeRedHeader(payload_type, timestamp_offset, block_length,
+                                 last_block, payload_ptr);
+  }
+  const double kPacketTime = 1.0;
+  // Hand over ownership of `packet_memory` to `packet`.
+  Packet packet(packet_memory, kPacketLengthBytes, kPacketTime);
+  ASSERT_TRUE(packet.valid_header());
+  EXPECT_EQ(kRedPayloadType, packet.header().payloadType);
+  EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber);
+  EXPECT_EQ(kTimestamp, packet.header().timestamp);
+  EXPECT_EQ(kSsrc, packet.header().ssrc);
+  EXPECT_EQ(0, packet.header().numCSRCs);
+  EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.payload_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes, packet.virtual_packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.virtual_payload_length_bytes());
+  EXPECT_EQ(kPacketTime, packet.time_ms());
+  std::list<RTPHeader*> red_headers;
+  EXPECT_TRUE(packet.ExtractRedHeaders(&red_headers));
+  EXPECT_EQ(kRedBlocks, static_cast<int>(red_headers.size()));
+  int block_index = 0;
+  for (std::list<RTPHeader*>::reverse_iterator it = red_headers.rbegin();
+       it != red_headers.rend(); ++it) {
+    // Reading list from the back, since the extraction puts the main payload
+    // (which is the last one on wire) first.
+    RTPHeader* red_block = *it;
+    EXPECT_EQ(block_index, red_block->payloadType);
+    EXPECT_EQ(kSequenceNumber, red_block->sequenceNumber);
+    if (block_index == kRedBlocks - 1) {
+      // Last block has zero offset per definition.
+      EXPECT_EQ(kTimestamp, red_block->timestamp);
+    } else {
+      EXPECT_EQ(kTimestamp - 100 * block_index, red_block->timestamp);
+    }
+    EXPECT_EQ(kSsrc, red_block->ssrc);
+    EXPECT_EQ(0, red_block->numCSRCs);
+    ++block_index;
+  }
+  Packet::DeleteRedHeaders(&red_headers);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
new file mode 100644
index 0000000000..5050e1fb17
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+
+#include <memory>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+bool ResampleInputAudioFile::Read(size_t samples,
+                                  int output_rate_hz,
+                                  int16_t* destination) {
+  const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
+  RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
+      << "Frame size and sample rates don't add up to an integer.";
+  std::unique_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
+  if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
+    return false;
+  resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
+  size_t output_length = 0;
+  RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read,
+                               destination, samples, output_length),
+               0);
+  RTC_CHECK_EQ(samples, output_length);
+  return true;
+}
+
+bool ResampleInputAudioFile::Read(size_t samples, int16_t* destination) {
+  RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
+  return Read(samples, output_rate_hz_, destination);
+}
+
+void ResampleInputAudioFile::set_output_rate_hz(int rate_hz) {
+  output_rate_hz_ = rate_hz;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
new file mode 100644
index 0000000000..279fece616
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/resampler/include/resampler.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for handling a looping input audio file with resampling.
+class ResampleInputAudioFile : public InputAudioFile {
+ public:
+  ResampleInputAudioFile(absl::string_view file_name,
+                         int file_rate_hz,
+                         bool loop_at_end = true)
+      : InputAudioFile(file_name, loop_at_end),
+        file_rate_hz_(file_rate_hz),
+        output_rate_hz_(-1) {}
+  ResampleInputAudioFile(absl::string_view file_name,
+                         int file_rate_hz,
+                         int output_rate_hz,
+                         bool loop_at_end = true)
+      : InputAudioFile(file_name, loop_at_end),
+        file_rate_hz_(file_rate_hz),
+        output_rate_hz_(output_rate_hz) {}
+
+  ResampleInputAudioFile(const ResampleInputAudioFile&) = delete;
+  ResampleInputAudioFile& operator=(const ResampleInputAudioFile&) = delete;
+
+  bool Read(size_t samples, int output_rate_hz, int16_t* destination);
+  bool Read(size_t samples, int16_t* destination) override;
+  void set_output_rate_hz(int rate_hz);
+
+ private:
+  const int file_rate_hz_;
+  int output_rate_hz_;
+  Resampler resampler_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
new file mode 100644
index 0000000000..1407aab5f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
+
+#include <string.h>
+
+#include <iostream>
+#include <limits>
+#include <memory>
+#include <set>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "logging/rtc_event_log/rtc_event_processor.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+bool ShouldSkipStream(ParsedRtcEventLog::MediaType media_type,
+                      uint32_t ssrc,
+                      absl::optional<uint32_t> ssrc_filter) {
+  if (media_type != ParsedRtcEventLog::MediaType::AUDIO)
+    return true;
+  if (ssrc_filter.has_value() && ssrc != *ssrc_filter)
+    return true;
+  return false;
+}
+}  // namespace
+
+std::unique_ptr<RtcEventLogSource> RtcEventLogSource::CreateFromFile(
+    absl::string_view file_name,
+    absl::optional<uint32_t> ssrc_filter) {
+  auto source = std::unique_ptr<RtcEventLogSource>(new RtcEventLogSource());
+  ParsedRtcEventLog parsed_log;
+  auto status = parsed_log.ParseFile(file_name);
+  if (!status.ok()) {
+    std::cerr << "Failed to parse event log: " << status.message() << std::endl;
+    std::cerr << "Skipping log." << std::endl;
+    return nullptr;
+  }
+  if (!source->Initialize(parsed_log, ssrc_filter)) {
+    std::cerr << "Failed to initialize source from event log, skipping."
+              << std::endl;
+    return nullptr;
+  }
+  return source;
+}
+
+std::unique_ptr<RtcEventLogSource> RtcEventLogSource::CreateFromString(
+    absl::string_view file_contents,
+    absl::optional<uint32_t> ssrc_filter) {
+  auto source = std::unique_ptr<RtcEventLogSource>(new RtcEventLogSource());
+  ParsedRtcEventLog parsed_log;
+  auto status = parsed_log.ParseString(file_contents);
+  if (!status.ok()) {
+    std::cerr << "Failed to parse event log: " << status.message() << std::endl;
+    std::cerr << "Skipping log." << std::endl;
+    return nullptr;
+  }
+  if (!source->Initialize(parsed_log, ssrc_filter)) {
+    std::cerr << "Failed to initialize source from event log, skipping."
+              << std::endl;
+    return nullptr;
+  }
+  return source;
+}
+
+RtcEventLogSource::~RtcEventLogSource() {}
+
+std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
+  if (rtp_packet_index_ >= rtp_packets_.size())
+    return nullptr;
+
+  std::unique_ptr<Packet> packet = std::move(rtp_packets_[rtp_packet_index_++]);
+  return packet;
+}
+
+int64_t RtcEventLogSource::NextAudioOutputEventMs() {
+  if (audio_output_index_ >= audio_outputs_.size())
+    return std::numeric_limits<int64_t>::max();
+
+  int64_t output_time_ms = audio_outputs_[audio_output_index_++];
+  return output_time_ms;
+}
+
+RtcEventLogSource::RtcEventLogSource() : PacketSource() {}
+
+bool RtcEventLogSource::Initialize(const ParsedRtcEventLog& parsed_log,
+                                   absl::optional<uint32_t> ssrc_filter) {
+  const auto first_log_end_time_us =
+      parsed_log.stop_log_events().empty()
+          ? std::numeric_limits<int64_t>::max()
+          : parsed_log.stop_log_events().front().log_time_us();
+
+  std::set<uint32_t> packet_ssrcs;
+  auto handle_rtp_packet =
+      [this, first_log_end_time_us,
+       &packet_ssrcs](const webrtc::LoggedRtpPacketIncoming& incoming) {
+        if (!filter_.test(incoming.rtp.header.payloadType) &&
+            incoming.log_time_us() < first_log_end_time_us) {
+          rtp_packets_.emplace_back(std::make_unique<Packet>(
+              incoming.rtp.header, incoming.rtp.total_length,
+              incoming.rtp.total_length - incoming.rtp.header_length,
+              static_cast<double>(incoming.log_time_ms())));
+          packet_ssrcs.insert(rtp_packets_.back()->header().ssrc);
+        }
+      };
+
+  std::set<uint32_t> ignored_ssrcs;
+  auto handle_audio_playout =
+      [this, first_log_end_time_us, &packet_ssrcs,
+       &ignored_ssrcs](const webrtc::LoggedAudioPlayoutEvent& audio_playout) {
+        if (audio_playout.log_time_us() < first_log_end_time_us) {
+          if (packet_ssrcs.count(audio_playout.ssrc) > 0) {
+            audio_outputs_.emplace_back(audio_playout.log_time_ms());
+          } else {
+            ignored_ssrcs.insert(audio_playout.ssrc);
+          }
+        }
+      };
+
+  // This wouldn't be needed if we knew that there was at most one audio stream.
+  webrtc::RtcEventProcessor event_processor;
+  for (const auto& rtp_packets : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    ParsedRtcEventLog::MediaType media_type =
+        parsed_log.GetMediaType(rtp_packets.ssrc, webrtc::kIncomingPacket);
+    if (ShouldSkipStream(media_type, rtp_packets.ssrc, ssrc_filter)) {
+      continue;
+    }
+    event_processor.AddEvents(rtp_packets.incoming_packets, handle_rtp_packet);
+    // If no SSRC filter has been set, use the first SSRC only. The simulator
+    // does not work properly with interleaved packets from multiple SSRCs.
+    if (!ssrc_filter.has_value()) {
+      ssrc_filter = rtp_packets.ssrc;
+    }
+  }
+
+  for (const auto& audio_playouts : parsed_log.audio_playout_events()) {
+    if (ssrc_filter.has_value() && audio_playouts.first != *ssrc_filter)
+      continue;
+    event_processor.AddEvents(audio_playouts.second, handle_audio_playout);
+  }
+
+  // Fills in rtp_packets_ and audio_outputs_.
+  event_processor.ProcessEventsInOrder();
+
+  for (const auto& ssrc : ignored_ssrcs) {
+    std::cout << "Ignoring GetAudio events from SSRC 0x" << std::hex << ssrc
+              << " because no packets were found with a matching SSRC."
+              << std::endl;
+  }
+
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
new file mode 100644
index 0000000000..c67912a67d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class RtpHeaderParser;
+
+namespace test {
+
+class Packet;
+
+class RtcEventLogSource : public PacketSource {
+ public:
+  // Creates an RtcEventLogSource reading from `file_name`. If the file cannot
+  // be opened, or has the wrong format, NULL will be returned.
+  static std::unique_ptr<RtcEventLogSource> CreateFromFile(
+      absl::string_view file_name,
+      absl::optional<uint32_t> ssrc_filter);
+  // Same as above, but uses a string with the file contents.
+  static std::unique_ptr<RtcEventLogSource> CreateFromString(
+      absl::string_view file_contents,
+      absl::optional<uint32_t> ssrc_filter);
+
+  virtual ~RtcEventLogSource();
+
+  RtcEventLogSource(const RtcEventLogSource&) = delete;
+  RtcEventLogSource& operator=(const RtcEventLogSource&) = delete;
+
+  std::unique_ptr<Packet> NextPacket() override;
+
+  // Returns the timestamp of the next audio output event, in milliseconds. The
+  // maximum value of int64_t is returned if there are no more audio output
+  // events available.
+  int64_t NextAudioOutputEventMs();
+
+ private:
+  RtcEventLogSource();
+
+  bool Initialize(const ParsedRtcEventLog& parsed_log,
+                  absl::optional<uint32_t> ssrc_filter);
+
+  std::vector<std::unique_ptr<Packet>> rtp_packets_;
+  size_t rtp_packet_index_ = 0;
+  std::vector<int64_t> audio_outputs_;
+  size_t audio_output_index_ = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
new file mode 100644
index 0000000000..7ecf925ebb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -0,0 +1,166 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+
+ABSL_FLAG(int, red, 117, "RTP payload type for RED");
+ABSL_FLAG(int,
+          audio_level,
+          -1,
+          "Extension ID for audio level (RFC 6464); "
+          "-1 not to print audio level");
+ABSL_FLAG(int,
+          abs_send_time,
+          -1,
+          "Extension ID for absolute sender time; "
+          "-1 not to print absolute send time");
+
+int main(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  std::string usage =
+      "Tool for parsing an RTP dump file to text output.\n"
+      "Example usage:\n"
+      "./rtp_analyze input.rtp output.txt\n\n"
+      "Output is sent to stdout if no output file is given. "
+      "Note that this tool can read files with or without payloads.\n";
+  if (args.size() != 2 && args.size() != 3) {
+    printf("%s", usage.c_str());
+    return 1;
+  }
+
+  RTC_CHECK(absl::GetFlag(FLAGS_red) >= 0 &&
+            absl::GetFlag(FLAGS_red) <= 127);          // Payload type
+  RTC_CHECK(absl::GetFlag(FLAGS_audio_level) == -1 ||  // Default
+            (absl::GetFlag(FLAGS_audio_level) > 0 &&
+             absl::GetFlag(FLAGS_audio_level) <= 255));  // Extension ID
+  RTC_CHECK(absl::GetFlag(FLAGS_abs_send_time) == -1 ||  // Default
+            (absl::GetFlag(FLAGS_abs_send_time) > 0 &&
+             absl::GetFlag(FLAGS_abs_send_time) <= 255));  // Extension ID
+
+  printf("Input file: %s\n", args[1]);
+  std::unique_ptr<webrtc::test::RtpFileSource> file_source(
+      webrtc::test::RtpFileSource::Create(args[1]));
+  RTC_DCHECK(file_source.get());
+  // Set RTP extension IDs.
+  bool print_audio_level = false;
+  if (absl::GetFlag(FLAGS_audio_level) != -1) {
+    print_audio_level = true;
+    file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
+                                            absl::GetFlag(FLAGS_audio_level));
+  }
+  bool print_abs_send_time = false;
+  if (absl::GetFlag(FLAGS_abs_send_time) != -1) {
+    print_abs_send_time = true;
+    file_source->RegisterRtpHeaderExtension(
+        webrtc::kRtpExtensionAbsoluteSendTime,
+        absl::GetFlag(FLAGS_abs_send_time));
+  }
+
+  FILE* out_file;
+  if (args.size() == 3) {
+    out_file = fopen(args[2], "wt");
+    if (!out_file) {
+      printf("Cannot open output file %s\n", args[2]);
+      return -1;
+    }
+    printf("Output file: %s\n\n", args[2]);
+  } else {
+    out_file = stdout;
+  }
+
+  // Print file header.
+  fprintf(out_file, "SeqNo  TimeStamp   SendTime  Size    PT  M       SSRC");
+  if (print_audio_level) {
+    fprintf(out_file, " AuLvl (V)");
+  }
+  if (print_abs_send_time) {
+    fprintf(out_file, " AbsSendTime");
+  }
+  fprintf(out_file, "\n");
+
+  uint32_t max_abs_send_time = 0;
+  int cycles = -1;
+  std::unique_ptr<webrtc::test::Packet> packet;
+  while (true) {
+    packet = file_source->NextPacket();
+    if (!packet.get()) {
+      // End of file reached.
+      break;
+    }
+    // Write packet data to file. Use virtual_packet_length_bytes so that the
+    // correct packet sizes are printed also for RTP header-only dumps.
+    fprintf(out_file, "%5u %10u %10u %5i %5i %2i %#08X",
+            packet->header().sequenceNumber, packet->header().timestamp,
+            static_cast<unsigned int>(packet->time_ms()),
+            static_cast<int>(packet->virtual_packet_length_bytes()),
+            packet->header().payloadType, packet->header().markerBit,
+            packet->header().ssrc);
+    if (print_audio_level && packet->header().extension.hasAudioLevel) {
+      fprintf(out_file, " %5u (%1i)", packet->header().extension.audioLevel,
+              packet->header().extension.voiceActivity);
+    }
+    if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
+      if (cycles == -1) {
+        // Initialize.
+        max_abs_send_time = packet->header().extension.absoluteSendTime;
+        cycles = 0;
+      }
+      // Abs sender time is 24 bit 6.18 fixed point. Shift by 8 to normalize to
+      // 32 bits (unsigned). Calculate the difference between this packet's
+      // send time and the maximum observed. Cast to signed 32-bit to get the
+      // desired wrap-around behavior.
+      if (static_cast<int32_t>(
+              (packet->header().extension.absoluteSendTime << 8) -
+              (max_abs_send_time << 8)) >= 0) {
+        // The difference is non-negative, meaning that this packet is newer
+        // than the previously observed maximum absolute send time.
+        if (packet->header().extension.absoluteSendTime < max_abs_send_time) {
+          // Wrap detected.
+          cycles++;
+        }
+        max_abs_send_time = packet->header().extension.absoluteSendTime;
+      }
+      // Abs sender time is 24 bit 6.18 fixed point. Divide by 2^18 to convert
+      // to floating point representation.
+      double send_time_seconds =
+          static_cast<double>(packet->header().extension.absoluteSendTime) /
+              262144 +
+          64.0 * cycles;
+      fprintf(out_file, " %11f", send_time_seconds);
+    }
+    fprintf(out_file, "\n");
+
+    if (packet->header().payloadType == absl::GetFlag(FLAGS_red)) {
+      std::list<webrtc::RTPHeader*> red_headers;
+      packet->ExtractRedHeaders(&red_headers);
+      while (!red_headers.empty()) {
+        webrtc::RTPHeader* red = red_headers.front();
+        RTC_DCHECK(red);
+        fprintf(out_file, "* %5u %10u %10u %5i\n", red->sequenceNumber,
+                red->timestamp, static_cast<unsigned int>(packet->time_ms()),
+                red->payloadType);
+        red_headers.pop_front();
+        delete red;
+      }
+    }
+  }
+
+  fclose(out_file);
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_encode.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
new file mode 100644
index 0000000000..8adca927f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -0,0 +1,351 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#endif
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_FUCHSIA)
+#include <netinet/in.h>
+#endif
+
+#include <iostream>
+#include <map>
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "absl/memory/memory.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/L16/audio_encoder_L16.h"
+#include "api/audio_codecs/g711/audio_encoder_g711.h"
+#include "api/audio_codecs/g722/audio_encoder_g722.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+ABSL_FLAG(bool, list_codecs, false, "Enumerate all codecs");
+ABSL_FLAG(std::string, codec, "opus", "Codec to use");
+ABSL_FLAG(int,
+          frame_len,
+          0,
+          "Frame length in ms; 0 indicates codec default value");
+ABSL_FLAG(int, bitrate, 0, "Bitrate in kbps; 0 indicates codec default value");
+ABSL_FLAG(int,
+          payload_type,
+          -1,
+          "RTP payload type; -1 indicates codec default value");
+ABSL_FLAG(int,
+          cng_payload_type,
+          -1,
+          "RTP payload type for CNG; -1 indicates default value");
+ABSL_FLAG(int, ssrc, 0, "SSRC to write to the RTP header");
+ABSL_FLAG(bool, dtx, false, "Use DTX/CNG");
+ABSL_FLAG(int, sample_rate, 48000, "Sample rate of the input file");
+
+namespace webrtc {
+namespace test {
+namespace {
+
+// Add new codecs here, and to the map below.
+enum class CodecType {
+  kOpus,
+  kPcmU,
+  kPcmA,
+  kG722,
+  kPcm16b8,
+  kPcm16b16,
+  kPcm16b32,
+  kPcm16b48,
+  kIlbc,
+};
+
+struct CodecTypeAndInfo {
+  CodecType type;
+  int default_payload_type;
+  bool internal_dtx;
+};
+
+// List all supported codecs here. This map defines the command-line parameter
+// value (the key string) for selecting each codec, together with information
+// whether it is using internal or external DTX/CNG.
+const std::map<std::string, CodecTypeAndInfo>& CodecList() {
+  static const auto* const codec_list =
+      new std::map<std::string, CodecTypeAndInfo>{
+          {"opus", {CodecType::kOpus, 111, true}},
+          {"pcmu", {CodecType::kPcmU, 0, false}},
+          {"pcma", {CodecType::kPcmA, 8, false}},
+          {"g722", {CodecType::kG722, 9, false}},
+          {"pcm16b_8", {CodecType::kPcm16b8, 93, false}},
+          {"pcm16b_16", {CodecType::kPcm16b16, 94, false}},
+          {"pcm16b_32", {CodecType::kPcm16b32, 95, false}},
+          {"pcm16b_48", {CodecType::kPcm16b48, 96, false}},
+          {"ilbc", {CodecType::kIlbc, 102, false}}};
+  return *codec_list;
+}
+
+// This class will receive callbacks from ACM when a packet is ready, and write
+// it to the output file.
+class Packetizer : public AudioPacketizationCallback {
+ public:
+  Packetizer(FILE* out_file, uint32_t ssrc, int timestamp_rate_hz)
+      : out_file_(out_file),
+        ssrc_(ssrc),
+        timestamp_rate_hz_(timestamp_rate_hz) {}
+
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_len_bytes,
+                   int64_t absolute_capture_timestamp_ms) override {
+    if (payload_len_bytes == 0) {
+      return 0;
+    }
+
+    constexpr size_t kRtpHeaderLength = 12;
+    constexpr size_t kRtpDumpHeaderLength = 8;
+    const uint16_t length = htons(rtc::checked_cast<uint16_t>(
+        kRtpHeaderLength + kRtpDumpHeaderLength + payload_len_bytes));
+    const uint16_t plen = htons(
+        rtc::checked_cast<uint16_t>(kRtpHeaderLength + payload_len_bytes));
+    const uint32_t offset = htonl(timestamp / (timestamp_rate_hz_ / 1000));
+    RTC_CHECK_EQ(fwrite(&length, sizeof(uint16_t), 1, out_file_), 1);
+    RTC_CHECK_EQ(fwrite(&plen, sizeof(uint16_t), 1, out_file_), 1);
+    RTC_CHECK_EQ(fwrite(&offset, sizeof(uint32_t), 1, out_file_), 1);
+
+    const uint8_t rtp_header[] = {0x80,
+                                  static_cast<uint8_t>(payload_type & 0x7F),
+                                  static_cast<uint8_t>(sequence_number_ >> 8),
+                                  static_cast<uint8_t>(sequence_number_),
+                                  static_cast<uint8_t>(timestamp >> 24),
+                                  static_cast<uint8_t>(timestamp >> 16),
+                                  static_cast<uint8_t>(timestamp >> 8),
+                                  static_cast<uint8_t>(timestamp),
+                                  static_cast<uint8_t>(ssrc_ >> 24),
+                                  static_cast<uint8_t>(ssrc_ >> 16),
+                                  static_cast<uint8_t>(ssrc_ >> 8),
+                                  static_cast<uint8_t>(ssrc_)};
+    static_assert(sizeof(rtp_header) == kRtpHeaderLength, "");
+    RTC_CHECK_EQ(
+        fwrite(rtp_header, sizeof(uint8_t), kRtpHeaderLength, out_file_),
+        kRtpHeaderLength);
+    ++sequence_number_;  // Intended to wrap on overflow.
+
+    RTC_CHECK_EQ(
+        fwrite(payload_data, sizeof(uint8_t), payload_len_bytes, out_file_),
+        payload_len_bytes);
+
+    return 0;
+  }
+
+ private:
+  FILE* const out_file_;
+  const uint32_t ssrc_;
+  const int timestamp_rate_hz_;
+  uint16_t sequence_number_ = 0;
+};
+
+void SetFrameLenIfFlagIsPositive(int* config_frame_len) {
+  if (absl::GetFlag(FLAGS_frame_len) > 0) {
+    *config_frame_len = absl::GetFlag(FLAGS_frame_len);
+  }
+}
+
+template <typename T>
+typename T::Config GetCodecConfig() {
+  typename T::Config config;
+  SetFrameLenIfFlagIsPositive(&config.frame_size_ms);
+  RTC_CHECK(config.IsOk());
+  return config;
+}
+
+AudioEncoderL16::Config Pcm16bConfig(CodecType codec_type) {
+  auto config = GetCodecConfig<AudioEncoderL16>();
+  switch (codec_type) {
+    case CodecType::kPcm16b8:
+      config.sample_rate_hz = 8000;
+      return config;
+    case CodecType::kPcm16b16:
+      config.sample_rate_hz = 16000;
+      return config;
+    case CodecType::kPcm16b32:
+      config.sample_rate_hz = 32000;
+      return config;
+    case CodecType::kPcm16b48:
+      config.sample_rate_hz = 48000;
+      return config;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return config;
+  }
+}
+
+std::unique_ptr<AudioEncoder> CreateEncoder(CodecType codec_type,
+                                            int payload_type) {
+  switch (codec_type) {
+    case CodecType::kOpus: {
+      AudioEncoderOpus::Config config = GetCodecConfig<AudioEncoderOpus>();
+      if (absl::GetFlag(FLAGS_bitrate) > 0) {
+        config.bitrate_bps = absl::GetFlag(FLAGS_bitrate);
+      }
+      config.dtx_enabled = absl::GetFlag(FLAGS_dtx);
+      RTC_CHECK(config.IsOk());
+      return AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+    }
+
+    case CodecType::kPcmU:
+    case CodecType::kPcmA: {
+      AudioEncoderG711::Config config = GetCodecConfig<AudioEncoderG711>();
+      config.type = codec_type == CodecType::kPcmU
+                        ? AudioEncoderG711::Config::Type::kPcmU
+                        : AudioEncoderG711::Config::Type::kPcmA;
+      RTC_CHECK(config.IsOk());
+      return AudioEncoderG711::MakeAudioEncoder(config, payload_type);
+    }
+
+    case CodecType::kG722: {
+      return AudioEncoderG722::MakeAudioEncoder(
+          GetCodecConfig<AudioEncoderG722>(), payload_type);
+    }
+
+    case CodecType::kPcm16b8:
+    case CodecType::kPcm16b16:
+    case CodecType::kPcm16b32:
+    case CodecType::kPcm16b48: {
+      return AudioEncoderL16::MakeAudioEncoder(Pcm16bConfig(codec_type),
+                                               payload_type);
+    }
+
+    case CodecType::kIlbc: {
+      return AudioEncoderIlbc::MakeAudioEncoder(
+          GetCodecConfig<AudioEncoderIlbc>(), payload_type);
+    }
+  }
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+}
+
+AudioEncoderCngConfig GetCngConfig(int sample_rate_hz) {
+  AudioEncoderCngConfig cng_config;
+  const auto default_payload_type = [&] {
+    switch (sample_rate_hz) {
+      case 8000:
+        return 13;
+      case 16000:
+        return 98;
+      case 32000:
+        return 99;
+      case 48000:
+        return 100;
+      default:
+        RTC_DCHECK_NOTREACHED();
+    }
+    return 0;
+  };
+  cng_config.payload_type = absl::GetFlag(FLAGS_cng_payload_type) != -1
+                                ? absl::GetFlag(FLAGS_cng_payload_type)
+                                : default_payload_type();
+  return cng_config;
+}
+
+int RunRtpEncode(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  const std::string usage =
+      "Tool for generating an RTP dump file from audio input.\n"
+      "Example usage:\n"
+      "./rtp_encode input.pcm output.rtp --codec=[codec] "
+      "--frame_len=[frame_len] --bitrate=[bitrate]\n\n";
+  if (!absl::GetFlag(FLAGS_list_codecs) && args.size() != 3) {
+    printf("%s", usage.c_str());
+    return 1;
+  }
+
+  if (absl::GetFlag(FLAGS_list_codecs)) {
+    printf("The following arguments are valid --codec parameters:\n");
+    for (const auto& c : CodecList()) {
+      printf("  %s\n", c.first.c_str());
+    }
+    return 0;
+  }
+
+  const auto codec_it = CodecList().find(absl::GetFlag(FLAGS_codec));
+  if (codec_it == CodecList().end()) {
+    printf("%s is not a valid codec name.\n",
+           absl::GetFlag(FLAGS_codec).c_str());
+    printf("Use argument --list_codecs to see all valid codec names.\n");
+    return 1;
+  }
+
+  // Create the codec.
+  const int payload_type = absl::GetFlag(FLAGS_payload_type) == -1
+                               ? codec_it->second.default_payload_type
+                               : absl::GetFlag(FLAGS_payload_type);
+  std::unique_ptr<AudioEncoder> codec =
+      CreateEncoder(codec_it->second.type, payload_type);
+
+  // Create an external VAD/CNG encoder if needed.
+  if (absl::GetFlag(FLAGS_dtx) && !codec_it->second.internal_dtx) {
+    AudioEncoderCngConfig cng_config = GetCngConfig(codec->SampleRateHz());
+    RTC_DCHECK(codec);
+    cng_config.speech_encoder = std::move(codec);
+    codec = CreateComfortNoiseEncoder(std::move(cng_config));
+  }
+  RTC_DCHECK(codec);
+
+  // Set up ACM.
+  const int timestamp_rate_hz = codec->RtpTimestampRateHz();
+  AudioCodingModule::Config config;
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(config));
+  acm->SetEncoder(std::move(codec));
+
+  // Open files.
+  printf("Input file: %s\n", args[1]);
+  InputAudioFile input_file(args[1], false);  // Open input in non-looping mode.
+  FILE* out_file = fopen(args[2], "wb");
+  RTC_CHECK(out_file) << "Could not open file " << args[2] << " for writing";
+  printf("Output file: %s\n", args[2]);
+  fprintf(out_file, "#!rtpplay1.0 \n");  //,
+  // Write 3 32-bit values followed by 2 16-bit values, all set to 0. This means
+  // a total of 16 bytes.
+  const uint8_t file_header[16] = {0};
+  RTC_CHECK_EQ(fwrite(file_header, sizeof(file_header), 1, out_file), 1);
+
+  // Create and register the packetizer, which will write the packets to file.
+  Packetizer packetizer(out_file, absl::GetFlag(FLAGS_ssrc), timestamp_rate_hz);
+  RTC_DCHECK_EQ(acm->RegisterTransportCallback(&packetizer), 0);
+
+  AudioFrame audio_frame;
+  audio_frame.samples_per_channel_ =
+      absl::GetFlag(FLAGS_sample_rate) / 100;  // 10 ms
+  audio_frame.sample_rate_hz_ = absl::GetFlag(FLAGS_sample_rate);
+  audio_frame.num_channels_ = 1;
+
+  while (input_file.Read(audio_frame.samples_per_channel_,
+                         audio_frame.mutable_data())) {
+    RTC_CHECK_GE(acm->Add10MsData(audio_frame), 0);
+    audio_frame.timestamp_ += audio_frame.samples_per_channel_;
+  }
+
+  return 0;
+}
+
+}  // namespace
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::RunRtpEncode(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
new file mode 100644
index 0000000000..a43c29638c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#ifndef WIN32
+#include <netinet/in.h>
+#endif
+
+#include <memory>
+
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+#include "test/rtp_file_reader.h"
+
+namespace webrtc {
+namespace test {
+
+RtpFileSource* RtpFileSource::Create(absl::string_view file_name,
+                                     absl::optional<uint32_t> ssrc_filter) {
+  RtpFileSource* source = new RtpFileSource(ssrc_filter);
+  RTC_CHECK(source->OpenFile(file_name));
+  return source;
+}
+
+bool RtpFileSource::ValidRtpDump(absl::string_view file_name) {
+  std::unique_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
+  return !!temp_file;
+}
+
+bool RtpFileSource::ValidPcap(absl::string_view file_name) {
+  std::unique_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kPcap, file_name));
+  return !!temp_file;
+}
+
+RtpFileSource::~RtpFileSource() {}
+
+bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
+                                               uint8_t id) {
+  return rtp_header_extension_map_.RegisterByType(id, type);
+}
+
+std::unique_ptr<Packet> RtpFileSource::NextPacket() {
+  while (true) {
+    RtpPacket temp_packet;
+    if (!rtp_reader_->NextPacket(&temp_packet)) {
+      return NULL;
+    }
+    if (temp_packet.original_length == 0) {
+      // May be an RTCP packet.
+      // Read the next one.
+      continue;
+    }
+    auto packet = std::make_unique<Packet>(
+        rtc::CopyOnWriteBuffer(temp_packet.data, temp_packet.length),
+        temp_packet.original_length, temp_packet.time_ms,
+        &rtp_header_extension_map_);
+    if (!packet->valid_header()) {
+      continue;
+    }
+    if (filter_.test(packet->header().payloadType) ||
+        (ssrc_filter_ && packet->header().ssrc != *ssrc_filter_)) {
+      // This payload type should be filtered out. Continue to the next packet.
+      continue;
+    }
+    return packet;
+  }
+}
+
+RtpFileSource::RtpFileSource(absl::optional<uint32_t> ssrc_filter)
+    : PacketSource(),
+      ssrc_filter_(ssrc_filter) {}
+
+bool RtpFileSource::OpenFile(absl::string_view file_name) {
+  rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
+  if (rtp_reader_)
+    return true;
+  rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
+  if (!rtp_reader_) {
+    RTC_FATAL()
+        << "Couldn't open input file as either a rtpdump or .pcap. Note "
+        << "that .pcapng is not supported.";
+  }
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
new file mode 100644
index 0000000000..55505be630
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+namespace test {
+
+class RtpFileReader;
+
+class RtpFileSource : public PacketSource {
+ public:
+  // Creates an RtpFileSource reading from `file_name`. If the file cannot be
+  // opened, or has the wrong format, NULL will be returned.
+  static RtpFileSource* Create(
+      absl::string_view file_name,
+      absl::optional<uint32_t> ssrc_filter = absl::nullopt);
+
+  // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
+  static bool ValidRtpDump(absl::string_view file_name);
+  static bool ValidPcap(absl::string_view file_name);
+
+  ~RtpFileSource() override;
+
+  RtpFileSource(const RtpFileSource&) = delete;
+  RtpFileSource& operator=(const RtpFileSource&) = delete;
+
+  // Registers an RTP header extension and binds it to `id`.
+  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
+
+  std::unique_ptr<Packet> NextPacket() override;
+
+ private:
+  static const int kFirstLineLength = 40;
+  static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
+  static const size_t kPacketHeaderSize = 8;
+
+  explicit RtpFileSource(absl::optional<uint32_t> ssrc_filter);
+
+  bool OpenFile(absl::string_view file_name);
+
+  std::unique_ptr<RtpFileReader> rtp_reader_;
+  const absl::optional<uint32_t> ssrc_filter_;
+  RtpHeaderExtensionMap rtp_header_extension_map_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
new file mode 100644
index 0000000000..e883fc11d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+
+
+namespace webrtc {
+namespace test {
+
+uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
+                                    size_t payload_length_samples,
+                                    RTPHeader* rtp_header) {
+  RTC_DCHECK(rtp_header);
+  if (!rtp_header) {
+    return 0;
+  }
+  rtp_header->sequenceNumber = seq_number_++;
+  rtp_header->timestamp = timestamp_;
+  timestamp_ += static_cast<uint32_t>(payload_length_samples);
+  rtp_header->payloadType = payload_type;
+  rtp_header->markerBit = false;
+  rtp_header->ssrc = ssrc_;
+  rtp_header->numCSRCs = 0;
+
+  uint32_t this_send_time = next_send_time_ms_;
+  RTC_DCHECK_GT(samples_per_ms_, 0);
+  next_send_time_ms_ +=
+      ((1.0 + drift_factor_) * payload_length_samples) / samples_per_ms_;
+  return this_send_time;
+}
+
+void RtpGenerator::set_drift_factor(double factor) {
+  if (factor > -1.0) {
+    drift_factor_ = factor;
+  }
+}
+
+uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
+                                                 size_t payload_length_samples,
+                                                 RTPHeader* rtp_header) {
+  uint32_t ret = RtpGenerator::GetRtpHeader(payload_type,
+                                            payload_length_samples, rtp_header);
+  if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <=
+          jump_from_timestamp_ &&
+      timestamp_ > jump_from_timestamp_) {
+    // We just moved across the `jump_from_timestamp_` timestamp. Do the jump.
+    timestamp_ = jump_to_timestamp_;
+  }
+  return ret;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.h
new file mode 100644
index 0000000000..2e615adec5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+
+#include "api/rtp_headers.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for generating RTP headers.
+class RtpGenerator {
+ public:
+  RtpGenerator(int samples_per_ms,
+               uint16_t start_seq_number = 0,
+               uint32_t start_timestamp = 0,
+               uint32_t start_send_time_ms = 0,
+               uint32_t ssrc = 0x12345678)
+      : seq_number_(start_seq_number),
+        timestamp_(start_timestamp),
+        next_send_time_ms_(start_send_time_ms),
+        ssrc_(ssrc),
+        samples_per_ms_(samples_per_ms),
+        drift_factor_(0.0) {}
+
+  virtual ~RtpGenerator() {}
+
+  RtpGenerator(const RtpGenerator&) = delete;
+  RtpGenerator& operator=(const RtpGenerator&) = delete;
+
+  // Writes the next RTP header to `rtp_header`, which will be of type
+  // `payload_type`. Returns the send time for this packet (in ms). The value of
+  // `payload_length_samples` determines the send time for the next packet.
+  virtual uint32_t GetRtpHeader(uint8_t payload_type,
+                                size_t payload_length_samples,
+                                RTPHeader* rtp_header);
+
+  void set_drift_factor(double factor);
+
+ protected:
+  uint16_t seq_number_;
+  uint32_t timestamp_;
+  uint32_t next_send_time_ms_;
+  const uint32_t ssrc_;
+  const int samples_per_ms_;
+  double drift_factor_;
+};
+
+class TimestampJumpRtpGenerator : public RtpGenerator {
+ public:
+  TimestampJumpRtpGenerator(int samples_per_ms,
+                            uint16_t start_seq_number,
+                            uint32_t start_timestamp,
+                            uint32_t jump_from_timestamp,
+                            uint32_t jump_to_timestamp)
+      : RtpGenerator(samples_per_ms, start_seq_number, start_timestamp),
+        jump_from_timestamp_(jump_from_timestamp),
+        jump_to_timestamp_(jump_to_timestamp) {}
+
+  TimestampJumpRtpGenerator(const TimestampJumpRtpGenerator&) = delete;
+  TimestampJumpRtpGenerator& operator=(const TimestampJumpRtpGenerator&) =
+      delete;
+
+  uint32_t GetRtpHeader(uint8_t payload_type,
+                        size_t payload_length_samples,
+                        RTPHeader* rtp_header) override;
+
+ private:
+  uint32_t jump_from_timestamp_;
+  uint32_t jump_to_timestamp_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
new file mode 100644
index 0000000000..cccaa9a3bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <fstream>
+#include <iostream>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr size_t kRtpDumpHeaderLength = 8;
+
+// Returns the next packet or an empty buffer if end of file was encountered.
+rtc::Buffer ReadNextPacket(FILE* file) {
+  // Read the rtpdump header for the next packet.
+  rtc::Buffer buffer;
+  buffer.SetData(kRtpDumpHeaderLength, [&](rtc::ArrayView<uint8_t> x) {
+    return fread(x.data(), 1, x.size(), file);
+  });
+  if (buffer.size() != kRtpDumpHeaderLength) {
+    return rtc::Buffer();
+  }
+
+  // Get length field. This is the total length for this packet written to file,
+  // including the kRtpDumpHeaderLength bytes already read.
+  const uint16_t len = ByteReader<uint16_t>::ReadBigEndian(buffer.data());
+  RTC_CHECK_GE(len, kRtpDumpHeaderLength);
+
+  // Read remaining data from file directly into buffer.
+  buffer.AppendData(len - kRtpDumpHeaderLength, [&](rtc::ArrayView<uint8_t> x) {
+    return fread(x.data(), 1, x.size(), file);
+  });
+  if (buffer.size() != len) {
+    buffer.Clear();
+  }
+  return buffer;
+}
+
+struct PacketAndTime {
+  rtc::Buffer packet;
+  int time;
+};
+
+void WritePacket(const PacketAndTime& packet, FILE* file) {
+  // Write the first 4 bytes from the original packet.
+  const auto* payload_ptr = packet.packet.data();
+  RTC_CHECK_EQ(fwrite(payload_ptr, 4, 1, file), 1);
+  payload_ptr += 4;
+
+  // Convert the new time offset to network endian, and write to file.
+  uint8_t time[sizeof(uint32_t)];
+  ByteWriter<uint32_t, sizeof(uint32_t)>::WriteBigEndian(time, packet.time);
+  RTC_CHECK_EQ(fwrite(time, sizeof(uint32_t), 1, file), 1);
+  payload_ptr += 4;  // Skip the old time in the original payload.
+
+  // Write the remaining part of the payload.
+  RTC_DCHECK_EQ(payload_ptr - packet.packet.data(), kRtpDumpHeaderLength);
+  RTC_CHECK_EQ(
+      fwrite(payload_ptr, packet.packet.size() - kRtpDumpHeaderLength, 1, file),
+      1);
+}
+
+int RunRtpJitter(int argc, char* argv[]) {
+  const std::string program_name = argv[0];
+  const std::string usage =
+      "Tool for alternating the arrival times in an RTP dump file.\n"
+      "Example usage:\n" +
+      program_name + " input.rtp arrival_times_ms.txt output.rtp\n\n";
+  if (argc != 4) {
+    printf("%s", usage.c_str());
+    return 1;
+  }
+
+  printf("Input RTP file: %s\n", argv[1]);
+  FILE* in_file = fopen(argv[1], "rb");
+  RTC_CHECK(in_file) << "Could not open file " << argv[1] << " for reading";
+  printf("Timing file: %s\n", argv[2]);
+  std::ifstream timing_file(argv[2]);
+  printf("Output file: %s\n", argv[3]);
+  FILE* out_file = fopen(argv[3], "wb");
+  RTC_CHECK(out_file) << "Could not open file " << argv[2] << " for writing";
+
+  // Copy the RTP file header to the output file.
+  char header_string[30];
+  RTC_CHECK(fgets(header_string, 30, in_file));
+  fprintf(out_file, "%s", header_string);
+  uint8_t file_header[16];
+  RTC_CHECK_EQ(fread(file_header, sizeof(file_header), 1, in_file), 1);
+  RTC_CHECK_EQ(fwrite(file_header, sizeof(file_header), 1, out_file), 1);
+
+  // Read all time values from the timing file. Store in a vector.
+  std::vector<int> new_arrival_times;
+  int new_time;
+  while (timing_file >> new_time) {
+    new_arrival_times.push_back(new_time);
+  }
+
+  // Read all packets from the input RTP file, but no more than the number of
+  // new time values. Store RTP packets together with new time values.
+  auto time_it = new_arrival_times.begin();
+  std::vector<PacketAndTime> packets;
+  while (1) {
+    auto packet = ReadNextPacket(in_file);
+    if (packet.empty() || time_it == new_arrival_times.end()) {
+      break;
+    }
+    packets.push_back({std::move(packet), *time_it});
+    ++time_it;
+  }
+
+  // Sort on new time values.
+  std::sort(packets.begin(), packets.end(),
+            [](const PacketAndTime& a, const PacketAndTime& b) {
+              return a.time < b.time;
+            });
+
+  // Write packets to output file.
+  for (const auto& p : packets) {
+    WritePacket(p, out_file);
+  }
+
+  fclose(in_file);
+  fclose(out_file);
+  return 0;
+}
+
+}  // namespace
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::RunRtpJitter(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtpcat.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtpcat.cc
new file mode 100644
index 0000000000..431de553ae
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtpcat.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "rtc_base/checks.h"
+#include "test/rtp_file_reader.h"
+#include "test/rtp_file_writer.h"
+
+using webrtc::test::RtpFileReader;
+using webrtc::test::RtpFileWriter;
+
+int main(int argc, char* argv[]) {
+  if (argc < 3) {
+    printf("Concatenates multiple rtpdump files into one.\n");
+    printf("Usage: rtpcat in1.rtp int2.rtp [...] out.rtp\n");
+    exit(1);
+  }
+
+  std::unique_ptr<RtpFileWriter> output(
+      RtpFileWriter::Create(RtpFileWriter::kRtpDump, argv[argc - 1]));
+  RTC_CHECK(output.get() != NULL) << "Cannot open output file.";
+  printf("Output RTP file: %s\n", argv[argc - 1]);
+
+  for (int i = 1; i < argc - 1; i++) {
+    std::unique_ptr<RtpFileReader> input(
+        RtpFileReader::Create(RtpFileReader::kRtpDump, argv[i]));
+    RTC_CHECK(input.get() != NULL) << "Cannot open input file " << argv[i];
+    printf("Input RTP file: %s\n", argv[i]);
+
+    webrtc::test::RtpPacket packet;
+    while (input->NextPacket(&packet))
+      RTC_CHECK(output->WritePacket(&packet));
+  }
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.cc
new file mode 100644
index 0000000000..baed812327
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/underrun_optimizer.h"
+
+#include <algorithm>
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDelayBuckets = 100;
+constexpr int kBucketSizeMs = 20;
+
+}  // namespace
+
+UnderrunOptimizer::UnderrunOptimizer(const TickTimer* tick_timer,
+                                     int histogram_quantile,
+                                     int forget_factor,
+                                     absl::optional<int> start_forget_weight,
+                                     absl::optional<int> resample_interval_ms)
+    : tick_timer_(tick_timer),
+      histogram_(kDelayBuckets, forget_factor, start_forget_weight),
+      histogram_quantile_(histogram_quantile),
+      resample_interval_ms_(resample_interval_ms) {}
+
+void UnderrunOptimizer::Update(int relative_delay_ms) {
+  absl::optional<int> histogram_update;
+  if (resample_interval_ms_) {
+    if (!resample_stopwatch_) {
+      resample_stopwatch_ = tick_timer_->GetNewStopwatch();
+    }
+    if (static_cast<int>(resample_stopwatch_->ElapsedMs()) >
+        *resample_interval_ms_) {
+      histogram_update = max_delay_in_interval_ms_;
+      resample_stopwatch_ = tick_timer_->GetNewStopwatch();
+      max_delay_in_interval_ms_ = 0;
+    }
+    max_delay_in_interval_ms_ =
+        std::max(max_delay_in_interval_ms_, relative_delay_ms);
+  } else {
+    histogram_update = relative_delay_ms;
+  }
+  if (!histogram_update) {
+    return;
+  }
+
+  const int index = *histogram_update / kBucketSizeMs;
+  if (index < histogram_.NumBuckets()) {
+    // Maximum delay to register is 2000 ms.
+    histogram_.Add(index);
+  }
+  int bucket_index = histogram_.Quantile(histogram_quantile_);
+  optimal_delay_ms_ = (1 + bucket_index) * kBucketSizeMs;
+}
+
+void UnderrunOptimizer::Reset() {
+  histogram_.Reset();
+  resample_stopwatch_.reset();
+  max_delay_in_interval_ms_ = 0;
+  optimal_delay_ms_.reset();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h
new file mode 100644
index 0000000000..b37ce18795
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
+#define MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/histogram.h"
+
+namespace webrtc {
+
+// Estimates probability of buffer underrun due to late packet arrival.
+// The optimal delay is decided such that the probability of underrun is lower
+// than 1 - `histogram_quantile`.
+class UnderrunOptimizer {
+ public:
+  UnderrunOptimizer(const TickTimer* tick_timer,
+                    int histogram_quantile,
+                    int forget_factor,
+                    absl::optional<int> start_forget_weight,
+                    absl::optional<int> resample_interval_ms);
+
+  void Update(int relative_delay_ms);
+
+  absl::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
+
+  void Reset();
+
+ private:
+  const TickTimer* tick_timer_;
+  Histogram histogram_;
+  const int histogram_quantile_;  // In Q30.
+  const absl::optional<int> resample_interval_ms_;
+  std::unique_ptr<TickTimer::Stopwatch> resample_stopwatch_;
+  int max_delay_in_interval_ms_ = 0;
+  absl::optional<int> optimal_delay_ms_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer_unittest.cc
new file mode 100644
index 0000000000..a86e9cf107
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer_unittest.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/underrun_optimizer.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDefaultHistogramQuantile = 1020054733;  // 0.95 in Q30.
+constexpr int kForgetFactor = 32745;                   // 0.9993 in Q15.
+
+}  // namespace
+
+TEST(UnderrunOptimizerTest, ResamplePacketDelays) {
+  TickTimer tick_timer;
+  constexpr int kResampleIntervalMs = 500;
+  UnderrunOptimizer underrun_optimizer(&tick_timer, kDefaultHistogramQuantile,
+                                       kForgetFactor, absl::nullopt,
+                                       kResampleIntervalMs);
+
+  // The histogram should be updated once with the maximum delay observed for
+  // the following sequence of updates.
+  for (int i = 0; i < 500; i += 20) {
+    underrun_optimizer.Update(i);
+    EXPECT_FALSE(underrun_optimizer.GetOptimalDelayMs());
+  }
+  tick_timer.Increment(kResampleIntervalMs / tick_timer.ms_per_tick() + 1);
+  underrun_optimizer.Update(0);
+  EXPECT_EQ(underrun_optimizer.GetOptimalDelayMs(), 500);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build
new file mode 100644
index 0000000000..823aea6dda
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build
@@ -0,0 +1,267 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/expand.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/histogram.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/merge.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/normal.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/packet.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.cc",
+    "/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("neteq_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/pcm16b_c_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/pcm16b_c_gn/moz.build
new file mode 100644
index 0000000000..54863e95ae
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/pcm16b_c_gn/moz.build
@@ -0,0 +1,217 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("pcm16b_c_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/pcm16b_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/pcm16b_gn/moz.build
new file mode 100644
index 0000000000..afee108cb3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/pcm16b_gn/moz.build
@@ -0,0 +1,227 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("pcm16b_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/red_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/red_gn/moz.build
new file mode 100644
index 0000000000..c4fa5c8bc4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/red_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("red_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/test/Channel.cc b/third_party/libwebrtc/modules/audio_coding/test/Channel.cc
new file mode 100644
index 0000000000..35aa6cb6b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/Channel.cc
@@ -0,0 +1,274 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/test/Channel.h"
+
+#include <iostream>
+
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+int32_t Channel::SendData(AudioFrameType frameType,
+                          uint8_t payloadType,
+                          uint32_t timeStamp,
+                          const uint8_t* payloadData,
+                          size_t payloadSize,
+                          int64_t absolute_capture_timestamp_ms) {
+  RTPHeader rtp_header;
+  int32_t status;
+  size_t payloadDataSize = payloadSize;
+
+  rtp_header.markerBit = false;
+  rtp_header.ssrc = 0;
+  rtp_header.sequenceNumber =
+      (external_sequence_number_ < 0)
+          ? _seqNo++
+          : static_cast<uint16_t>(external_sequence_number_);
+  rtp_header.payloadType = payloadType;
+  rtp_header.timestamp = (external_send_timestamp_ < 0)
+                             ? timeStamp
+                             : static_cast<uint32_t>(external_send_timestamp_);
+
+  if (frameType == AudioFrameType::kEmptyFrame) {
+    // When frame is empty, we should not transmit it. The frame size of the
+    // next non-empty frame will be based on the previous frame size.
+    _useLastFrameSize = _lastFrameSizeSample > 0;
+    return 0;
+  }
+
+  memcpy(_payloadData, payloadData, payloadDataSize);
+  if (_isStereo) {
+    if (_leftChannel) {
+      _rtp_header = rtp_header;
+      _leftChannel = false;
+    } else {
+      rtp_header = _rtp_header;
+      _leftChannel = true;
+    }
+  }
+
+  _channelCritSect.Lock();
+  if (_saveBitStream) {
+    // fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
+  }
+
+  if (!_isStereo) {
+    CalcStatistics(rtp_header, payloadSize);
+  }
+  _useLastFrameSize = false;
+  _lastInTimestamp = timeStamp;
+  _totalBytes += payloadDataSize;
+  _channelCritSect.Unlock();
+
+  if (_useFECTestWithPacketLoss) {
+    _packetLoss += 1;
+    if (_packetLoss == 3) {
+      _packetLoss = 0;
+      return 0;
+    }
+  }
+
+  if (num_packets_to_drop_ > 0) {
+    num_packets_to_drop_--;
+    return 0;
+  }
+
+  status =
+      _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtp_header);
+
+  return status;
+}
+
+// TODO(turajs): rewite this method.
+void Channel::CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize) {
+  int n;
+  if ((rtp_header.payloadType != _lastPayloadType) &&
+      (_lastPayloadType != -1)) {
+    // payload-type is changed.
+    // we have to terminate the calculations on the previous payload type
+    // we ignore the last packet in that payload type just to make things
+    // easier.
+    for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+      if (_lastPayloadType == _payloadStats[n].payloadType) {
+        _payloadStats[n].newPacket = true;
+        break;
+      }
+    }
+  }
+  _lastPayloadType = rtp_header.payloadType;
+
+  bool newPayload = true;
+  ACMTestPayloadStats* currentPayloadStr = NULL;
+  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+    if (rtp_header.payloadType == _payloadStats[n].payloadType) {
+      newPayload = false;
+      currentPayloadStr = &_payloadStats[n];
+      break;
+    }
+  }
+
+  if (!newPayload) {
+    if (!currentPayloadStr->newPacket) {
+      if (!_useLastFrameSize) {
+        _lastFrameSizeSample =
+            (uint32_t)((uint32_t)rtp_header.timestamp -
+                       (uint32_t)currentPayloadStr->lastTimestamp);
+      }
+      RTC_DCHECK_GT(_lastFrameSizeSample, 0);
+      int k = 0;
+      for (; k < MAX_NUM_FRAMESIZES; ++k) {
+        if ((currentPayloadStr->frameSizeStats[k].frameSizeSample ==
+             _lastFrameSizeSample) ||
+            (currentPayloadStr->frameSizeStats[k].frameSizeSample == 0)) {
+          break;
+        }
+      }
+      if (k == MAX_NUM_FRAMESIZES) {
+        // New frame size found but no space to count statistics on it. Skip it.
+        printf("No memory to store statistics for payload %d : frame size %d\n",
+               _lastPayloadType, _lastFrameSizeSample);
+        return;
+      }
+      ACMTestFrameSizeStats* currentFrameSizeStats =
+          &(currentPayloadStr->frameSizeStats[k]);
+      currentFrameSizeStats->frameSizeSample = (int16_t)_lastFrameSizeSample;
+
+      // increment the number of encoded samples.
+      currentFrameSizeStats->totalEncodedSamples += _lastFrameSizeSample;
+      // increment the number of recveived packets
+      currentFrameSizeStats->numPackets++;
+      // increment the total number of bytes (this is based on
+      // the previous payload we don't know the frame-size of
+      // the current payload.
+      currentFrameSizeStats->totalPayloadLenByte +=
+          currentPayloadStr->lastPayloadLenByte;
+      // store the maximum payload-size (this is based on
+      // the previous payload we don't know the frame-size of
+      // the current payload.
+      if (currentFrameSizeStats->maxPayloadLen <
+          currentPayloadStr->lastPayloadLenByte) {
+        currentFrameSizeStats->maxPayloadLen =
+            currentPayloadStr->lastPayloadLenByte;
+      }
+      // store the current values for the next time
+      currentPayloadStr->lastTimestamp = rtp_header.timestamp;
+      currentPayloadStr->lastPayloadLenByte = payloadSize;
+    } else {
+      currentPayloadStr->newPacket = false;
+      currentPayloadStr->lastPayloadLenByte = payloadSize;
+      currentPayloadStr->lastTimestamp = rtp_header.timestamp;
+      currentPayloadStr->payloadType = rtp_header.payloadType;
+      memset(currentPayloadStr->frameSizeStats, 0,
+             MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
+    }
+  } else {
+    n = 0;
+    while (_payloadStats[n].payloadType != -1) {
+      n++;
+    }
+    // first packet
+    _payloadStats[n].newPacket = false;
+    _payloadStats[n].lastPayloadLenByte = payloadSize;
+    _payloadStats[n].lastTimestamp = rtp_header.timestamp;
+    _payloadStats[n].payloadType = rtp_header.payloadType;
+    memset(_payloadStats[n].frameSizeStats, 0,
+           MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
+  }
+}
+
+Channel::Channel(int16_t chID)
+    : _receiverACM(NULL),
+      _seqNo(0),
+      _bitStreamFile(NULL),
+      _saveBitStream(false),
+      _lastPayloadType(-1),
+      _isStereo(false),
+      _leftChannel(true),
+      _lastInTimestamp(0),
+      _useLastFrameSize(false),
+      _lastFrameSizeSample(0),
+      _packetLoss(0),
+      _useFECTestWithPacketLoss(false),
+      _beginTime(rtc::TimeMillis()),
+      _totalBytes(0),
+      external_send_timestamp_(-1),
+      external_sequence_number_(-1),
+      num_packets_to_drop_(0) {
+  int n;
+  int k;
+  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+    _payloadStats[n].payloadType = -1;
+    _payloadStats[n].newPacket = true;
+    for (k = 0; k < MAX_NUM_FRAMESIZES; k++) {
+      _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
+      _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
+      _payloadStats[n].frameSizeStats[k].numPackets = 0;
+      _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
+      _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
+    }
+  }
+  if (chID >= 0) {
+    _saveBitStream = true;
+    rtc::StringBuilder ss;
+    ss.AppendFormat("bitStream_%d.dat", chID);
+    _bitStreamFile = fopen(ss.str().c_str(), "wb");
+  } else {
+    _saveBitStream = false;
+  }
+}
+
+Channel::~Channel() {}
+
+void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
+  _receiverACM = acm;
+  return;
+}
+
+void Channel::ResetStats() {
+  int n;
+  int k;
+  _channelCritSect.Lock();
+  _lastPayloadType = -1;
+  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+    _payloadStats[n].payloadType = -1;
+    _payloadStats[n].newPacket = true;
+    for (k = 0; k < MAX_NUM_FRAMESIZES; k++) {
+      _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
+      _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
+      _payloadStats[n].frameSizeStats[k].numPackets = 0;
+      _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
+      _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
+    }
+  }
+  _beginTime = rtc::TimeMillis();
+  _totalBytes = 0;
+  _channelCritSect.Unlock();
+}
+
+uint32_t Channel::LastInTimestamp() {
+  uint32_t timestamp;
+  _channelCritSect.Lock();
+  timestamp = _lastInTimestamp;
+  _channelCritSect.Unlock();
+  return timestamp;
+}
+
+double Channel::BitRate() {
+  double rate;
+  uint64_t currTime = rtc::TimeMillis();
+  _channelCritSect.Lock();
+  rate = ((double)_totalBytes * 8.0) / (double)(currTime - _beginTime);
+  _channelCritSect.Unlock();
+  return rate;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/Channel.h b/third_party/libwebrtc/modules/audio_coding/test/Channel.h
new file mode 100644
index 0000000000..7a8829e1d2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/Channel.h
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+#define MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+
+#include <stdio.h>
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+#define MAX_NUM_PAYLOADS 50
+#define MAX_NUM_FRAMESIZES 6
+
+// TODO(turajs): Write constructor for this structure.
+struct ACMTestFrameSizeStats {
+  uint16_t frameSizeSample;
+  size_t maxPayloadLen;
+  uint32_t numPackets;
+  uint64_t totalPayloadLenByte;
+  uint64_t totalEncodedSamples;
+  double rateBitPerSec;
+  double usageLenSec;
+};
+
+// TODO(turajs): Write constructor for this structure.
+struct ACMTestPayloadStats {
+  bool newPacket;
+  int16_t payloadType;
+  size_t lastPayloadLenByte;
+  uint32_t lastTimestamp;
+  ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
+};
+
+class Channel : public AudioPacketizationCallback {
+ public:
+  Channel(int16_t chID = -1);
+  ~Channel() override;
+
+  int32_t SendData(AudioFrameType frameType,
+                   uint8_t payloadType,
+                   uint32_t timeStamp,
+                   const uint8_t* payloadData,
+                   size_t payloadSize,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  void RegisterReceiverACM(AudioCodingModule* acm);
+
+  void ResetStats();
+
+  void SetIsStereo(bool isStereo) { _isStereo = isStereo; }
+
+  uint32_t LastInTimestamp();
+
+  void SetFECTestWithPacketLoss(bool usePacketLoss) {
+    _useFECTestWithPacketLoss = usePacketLoss;
+  }
+
+  double BitRate();
+
+  void set_send_timestamp(uint32_t new_send_ts) {
+    external_send_timestamp_ = new_send_ts;
+  }
+
+  void set_sequence_number(uint16_t new_sequence_number) {
+    external_sequence_number_ = new_sequence_number;
+  }
+
+  void set_num_packets_to_drop(int new_num_packets_to_drop) {
+    num_packets_to_drop_ = new_num_packets_to_drop;
+  }
+
+ private:
+  void CalcStatistics(const RTPHeader& rtp_header, size_t payloadSize);
+
+  AudioCodingModule* _receiverACM;
+  uint16_t _seqNo;
+  // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
+  uint8_t _payloadData[60 * 32 * 2 * 2];
+
+  Mutex _channelCritSect;
+  FILE* _bitStreamFile;
+  bool _saveBitStream;
+  int16_t _lastPayloadType;
+  ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
+  bool _isStereo;
+  RTPHeader _rtp_header;
+  bool _leftChannel;
+  uint32_t _lastInTimestamp;
+  bool _useLastFrameSize;
+  uint32_t _lastFrameSizeSample;
+  // FEC Test variables
+  int16_t _packetLoss;
+  bool _useFECTestWithPacketLoss;
+  uint64_t _beginTime;
+  uint64_t _totalBytes;
+
+  // External timing info, defaulted to -1. Only used if they are
+  // non-negative.
+  int64_t external_send_timestamp_;
+  int32_t external_sequence_number_;
+  int num_packets_to_drop_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_CHANNEL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.cc b/third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.cc
new file mode 100644
index 0000000000..9f9c4aa74c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -0,0 +1,269 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/test/EncodeDecodeTest.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+// Buffer size for stereo 48 kHz audio.
+constexpr size_t kWebRtc10MsPcmAudio = 960;
+
+}  // namespace
+
+TestPacketization::TestPacketization(RTPStream* rtpStream, uint16_t frequency)
+    : _rtpStream(rtpStream), _frequency(frequency), _seqNo(0) {}
+
+TestPacketization::~TestPacketization() {}
+
+int32_t TestPacketization::SendData(const AudioFrameType /* frameType */,
+                                    const uint8_t payloadType,
+                                    const uint32_t timeStamp,
+                                    const uint8_t* payloadData,
+                                    const size_t payloadSize,
+                                    int64_t absolute_capture_timestamp_ms) {
+  _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
+                    _frequency);
+  return 1;
+}
+
+Sender::Sender()
+    : _acm(NULL), _pcmFile(), _audioFrame(), _packetization(NULL) {}
+
+void Sender::Setup(AudioCodingModule* acm,
+                   RTPStream* rtpStream,
+                   absl::string_view in_file_name,
+                   int in_sample_rate,
+                   int payload_type,
+                   SdpAudioFormat format) {
+  // Open input file
+  const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
+  _pcmFile.Open(file_name, in_sample_rate, "rb");
+  if (format.num_channels == 2) {
+    _pcmFile.ReadStereo(true);
+  }
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  _pcmFile.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  _pcmFile.FastForward(100);
+
+  acm->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
+      payload_type, format, absl::nullopt));
+  _packetization = new TestPacketization(rtpStream, format.clockrate_hz);
+  EXPECT_EQ(0, acm->RegisterTransportCallback(_packetization));
+
+  _acm = acm;
+}
+
+void Sender::Teardown() {
+  _pcmFile.Close();
+  delete _packetization;
+}
+
+bool Sender::Add10MsData() {
+  if (!_pcmFile.EndOfFile()) {
+    EXPECT_GT(_pcmFile.Read10MsData(_audioFrame), 0);
+    int32_t ok = _acm->Add10MsData(_audioFrame);
+    EXPECT_GE(ok, 0);
+    return ok >= 0 ? true : false;
+  }
+  return false;
+}
+
+void Sender::Run() {
+  while (true) {
+    if (!Add10MsData()) {
+      break;
+    }
+  }
+}
+
+Receiver::Receiver()
+    : _playoutLengthSmpls(kWebRtc10MsPcmAudio),
+      _payloadSizeBytes(MAX_INCOMING_PAYLOAD) {}
+
+void Receiver::Setup(AudioCodingModule* acm,
+                     RTPStream* rtpStream,
+                     absl::string_view out_file_name,
+                     size_t channels,
+                     int file_num) {
+  EXPECT_EQ(0, acm->InitializeReceiver());
+
+  if (channels == 1) {
+    acm->SetReceiveCodecs({{107, {"L16", 8000, 1}},
+                           {108, {"L16", 16000, 1}},
+                           {109, {"L16", 32000, 1}},
+                           {0, {"PCMU", 8000, 1}},
+                           {8, {"PCMA", 8000, 1}},
+                           {102, {"ILBC", 8000, 1}},
+                           {9, {"G722", 8000, 1}},
+                           {120, {"OPUS", 48000, 2}},
+                           {13, {"CN", 8000, 1}},
+                           {98, {"CN", 16000, 1}},
+                           {99, {"CN", 32000, 1}}});
+  } else {
+    ASSERT_EQ(channels, 2u);
+    acm->SetReceiveCodecs({{111, {"L16", 8000, 2}},
+                           {112, {"L16", 16000, 2}},
+                           {113, {"L16", 32000, 2}},
+                           {110, {"PCMU", 8000, 2}},
+                           {118, {"PCMA", 8000, 2}},
+                           {119, {"G722", 8000, 2}},
+                           {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
+  }
+
+  int playSampFreq;
+  std::string file_name;
+  rtc::StringBuilder file_stream;
+  file_stream << webrtc::test::OutputPath() << out_file_name << file_num
+              << ".pcm";
+  file_name = file_stream.str();
+  _rtpStream = rtpStream;
+
+  playSampFreq = 32000;
+  _pcmFile.Open(file_name, 32000, "wb+");
+
+  _realPayloadSizeBytes = 0;
+  _playoutBuffer = new int16_t[kWebRtc10MsPcmAudio];
+  _frequency = playSampFreq;
+  _acm = acm;
+  _firstTime = true;
+}
+
+void Receiver::Teardown() {
+  delete[] _playoutBuffer;
+  _pcmFile.Close();
+}
+
+bool Receiver::IncomingPacket() {
+  if (!_rtpStream->EndOfFile()) {
+    if (_firstTime) {
+      _firstTime = false;
+      _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
+                                               _payloadSizeBytes, &_nextTime);
+      if (_realPayloadSizeBytes == 0) {
+        if (_rtpStream->EndOfFile()) {
+          _firstTime = true;
+          return true;
+        } else {
+          return false;
+        }
+      }
+    }
+
+    EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
+                                      _rtpHeader));
+    _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
+                                             _payloadSizeBytes, &_nextTime);
+    if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
+      _firstTime = true;
+    }
+  }
+  return true;
+}
+
+bool Receiver::PlayoutData() {
+  AudioFrame audioFrame;
+  bool muted;
+  int32_t ok = _acm->PlayoutData10Ms(_frequency, &audioFrame, &muted);
+  if (muted) {
+    ADD_FAILURE();
+    return false;
+  }
+  EXPECT_EQ(0, ok);
+  if (ok < 0) {
+    return false;
+  }
+  if (_playoutLengthSmpls == 0) {
+    return false;
+  }
+  _pcmFile.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_ *
+                                                audioFrame.num_channels_);
+  return true;
+}
+
+void Receiver::Run() {
+  uint8_t counter500Ms = 50;
+  uint32_t clock = 0;
+
+  while (counter500Ms > 0) {
+    if (clock == 0 || clock >= _nextTime) {
+      EXPECT_TRUE(IncomingPacket());
+      if (clock == 0) {
+        clock = _nextTime;
+      }
+    }
+    if ((clock % 10) == 0) {
+      if (!PlayoutData()) {
+        clock++;
+        continue;
+      }
+    }
+    if (_rtpStream->EndOfFile()) {
+      counter500Ms--;
+    }
+    clock++;
+  }
+}
+
+EncodeDecodeTest::EncodeDecodeTest() = default;
+
+void EncodeDecodeTest::Perform() {
+  const std::map<int, SdpAudioFormat> send_codecs = {
+      {107, {"L16", 8000, 1}},   {108, {"L16", 16000, 1}},
+      {109, {"L16", 32000, 1}},  {0, {"PCMU", 8000, 1}},
+      {8, {"PCMA", 8000, 1}},
+#ifdef WEBRTC_CODEC_ILBC
+      {102, {"ILBC", 8000, 1}},
+#endif
+      {9, {"G722", 8000, 1}}};
+  int file_num = 0;
+  for (const auto& send_codec : send_codecs) {
+    RTPFile rtpFile;
+    std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
+        AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+
+    std::string fileName = webrtc::test::TempFilename(
+        webrtc::test::OutputPath(), "encode_decode_rtp");
+    rtpFile.Open(fileName.c_str(), "wb+");
+    rtpFile.WriteHeader();
+    Sender sender;
+    sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000,
+                 send_codec.first, send_codec.second);
+    sender.Run();
+    sender.Teardown();
+    rtpFile.Close();
+
+    rtpFile.Open(fileName.c_str(), "rb");
+    rtpFile.ReadHeader();
+    Receiver receiver;
+    receiver.Setup(acm.get(), &rtpFile, "encodeDecode_out", 1, file_num);
+    receiver.Run();
+    receiver.Teardown();
+    rtpFile.Close();
+
+    file_num++;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.h b/third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.h
new file mode 100644
index 0000000000..89b76440ef
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/EncodeDecodeTest.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#define MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+
+#include <stdio.h>
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/RTPFile.h"
+#include "modules/include/module_common_types.h"
+
+namespace webrtc {
+
+#define MAX_INCOMING_PAYLOAD 8096
+
+// TestPacketization callback which writes the encoded payloads to file
+class TestPacketization : public AudioPacketizationCallback {
+ public:
+  TestPacketization(RTPStream* rtpStream, uint16_t frequency);
+  ~TestPacketization();
+  int32_t SendData(AudioFrameType frameType,
+                   uint8_t payloadType,
+                   uint32_t timeStamp,
+                   const uint8_t* payloadData,
+                   size_t payloadSize,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+ private:
+  static void MakeRTPheader(uint8_t* rtpHeader,
+                            uint8_t payloadType,
+                            int16_t seqNo,
+                            uint32_t timeStamp,
+                            uint32_t ssrc);
+  RTPStream* _rtpStream;
+  int32_t _frequency;
+  int16_t _seqNo;
+};
+
+class Sender {
+ public:
+  Sender();
+  void Setup(AudioCodingModule* acm,
+             RTPStream* rtpStream,
+             absl::string_view in_file_name,
+             int in_sample_rate,
+             int payload_type,
+             SdpAudioFormat format);
+  void Teardown();
+  void Run();
+  bool Add10MsData();
+
+ protected:
+  AudioCodingModule* _acm;
+
+ private:
+  PCMFile _pcmFile;
+  AudioFrame _audioFrame;
+  TestPacketization* _packetization;
+};
+
+class Receiver {
+ public:
+  Receiver();
+  virtual ~Receiver() {}
+  void Setup(AudioCodingModule* acm,
+             RTPStream* rtpStream,
+             absl::string_view out_file_name,
+             size_t channels,
+             int file_num);
+  void Teardown();
+  void Run();
+  virtual bool IncomingPacket();
+  bool PlayoutData();
+
+ private:
+  PCMFile _pcmFile;
+  int16_t* _playoutBuffer;
+  uint16_t _playoutLengthSmpls;
+  int32_t _frequency;
+  bool _firstTime;
+
+ protected:
+  AudioCodingModule* _acm;
+  uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
+  RTPStream* _rtpStream;
+  RTPHeader _rtpHeader;
+  size_t _realPayloadSizeBytes;
+  size_t _payloadSizeBytes;
+  uint32_t _nextTime;
+};
+
+class EncodeDecodeTest {
+ public:
+  EncodeDecodeTest();
+  void Perform();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/PCMFile.cc b/third_party/libwebrtc/modules/audio_coding/test/PCMFile.cc
new file mode 100644
index 0000000000..e069a42de1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/PCMFile.cc
@@ -0,0 +1,240 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/test/PCMFile.h"
+
+#include <ctype.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+
+PCMFile::PCMFile()
+    : pcm_file_(NULL),
+      samples_10ms_(160),
+      frequency_(16000),
+      end_of_file_(false),
+      auto_rewind_(false),
+      rewinded_(false),
+      read_stereo_(false),
+      save_stereo_(false) {
+  timestamp_ =
+      (((uint32_t)rand() & 0x0000FFFF) << 16) | ((uint32_t)rand() & 0x0000FFFF);
+}
+
+PCMFile::PCMFile(uint32_t timestamp)
+    : pcm_file_(NULL),
+      samples_10ms_(160),
+      frequency_(16000),
+      end_of_file_(false),
+      auto_rewind_(false),
+      rewinded_(false),
+      read_stereo_(false),
+      save_stereo_(false) {
+  timestamp_ = timestamp;
+}
+
+PCMFile::~PCMFile() {
+  if (pcm_file_) {
+    fclose(pcm_file_);
+  }
+}
+
+int16_t PCMFile::ChooseFile(std::string* file_name,
+                            int16_t max_len,
+                            uint16_t* frequency_hz) {
+  char tmp_name[MAX_FILE_NAME_LENGTH_BYTE];
+
+  EXPECT_TRUE(fgets(tmp_name, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
+  tmp_name[MAX_FILE_NAME_LENGTH_BYTE - 1] = '\0';
+  int16_t n = 0;
+
+  // Removing trailing spaces.
+  while ((isspace(static_cast<unsigned char>(tmp_name[n])) ||
+          iscntrl(static_cast<unsigned char>(tmp_name[n]))) &&
+         (static_cast<unsigned char>(tmp_name[n]) != 0) &&
+         (n < MAX_FILE_NAME_LENGTH_BYTE)) {
+    n++;
+  }
+  if (n > 0) {
+    memmove(tmp_name, &tmp_name[n], MAX_FILE_NAME_LENGTH_BYTE - n);
+  }
+
+  // Removing trailing spaces.
+  n = (int16_t)(strlen(tmp_name) - 1);
+  if (n >= 0) {
+    while ((isspace(static_cast<unsigned char>(tmp_name[n])) ||
+            iscntrl(static_cast<unsigned char>(tmp_name[n]))) &&
+           (n >= 0)) {
+      n--;
+    }
+  }
+  if (n >= 0) {
+    tmp_name[n + 1] = '\0';
+  }
+
+  int16_t len = (int16_t)strlen(tmp_name);
+  if (len > max_len) {
+    return -1;
+  }
+  if (len > 0) {
+    std::string tmp_string(tmp_name, len + 1);
+    *file_name = tmp_string;
+  }
+  printf("Enter the sampling frequency (in Hz) of the above file [%u]: ",
+         *frequency_hz);
+  EXPECT_TRUE(fgets(tmp_name, 10, stdin) != NULL);
+  uint16_t tmp_frequency = (uint16_t)atoi(tmp_name);
+  if (tmp_frequency > 0) {
+    *frequency_hz = tmp_frequency;
+  }
+  return 0;
+}
+
+void PCMFile::Open(absl::string_view file_name,
+                   uint16_t frequency,
+                   absl::string_view mode,
+                   bool auto_rewind) {
+  if ((pcm_file_ = fopen(std::string(file_name).c_str(),
+                         std::string(mode).c_str())) == NULL) {
+    printf("Cannot open file %s.\n", std::string(file_name).c_str());
+    ADD_FAILURE() << "Unable to read file";
+  }
+  frequency_ = frequency;
+  samples_10ms_ = (uint16_t)(frequency_ / 100);
+  auto_rewind_ = auto_rewind;
+  end_of_file_ = false;
+  rewinded_ = false;
+}
+
+int32_t PCMFile::SamplingFrequency() const {
+  return frequency_;
+}
+
+uint16_t PCMFile::PayloadLength10Ms() const {
+  return samples_10ms_;
+}
+
+int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) {
+  uint16_t channels = 1;
+  if (read_stereo_) {
+    channels = 2;
+  }
+
+  int32_t payload_size =
+      (int32_t)fread(audio_frame.mutable_data(), sizeof(uint16_t),
+                     samples_10ms_ * channels, pcm_file_);
+  if (payload_size < samples_10ms_ * channels) {
+    int16_t* frame_data = audio_frame.mutable_data();
+    for (int k = payload_size; k < samples_10ms_ * channels; k++) {
+      frame_data[k] = 0;
+    }
+    if (auto_rewind_) {
+      rewind(pcm_file_);
+      rewinded_ = true;
+    } else {
+      end_of_file_ = true;
+    }
+  }
+  audio_frame.samples_per_channel_ = samples_10ms_;
+  audio_frame.sample_rate_hz_ = frequency_;
+  audio_frame.num_channels_ = channels;
+  audio_frame.timestamp_ = timestamp_;
+  timestamp_ += samples_10ms_;
+  ++blocks_read_;
+  if (num_10ms_blocks_to_read_ && blocks_read_ >= *num_10ms_blocks_to_read_)
+    end_of_file_ = true;
+  return samples_10ms_;
+}
+
+void PCMFile::Write10MsData(const AudioFrame& audio_frame) {
+  if (audio_frame.num_channels_ == 1) {
+    if (!save_stereo_) {
+      if (fwrite(audio_frame.data(), sizeof(uint16_t),
+                 audio_frame.samples_per_channel_, pcm_file_) !=
+          static_cast<size_t>(audio_frame.samples_per_channel_)) {
+        return;
+      }
+    } else {
+      const int16_t* frame_data = audio_frame.data();
+      int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
+      for (size_t k = 0; k < audio_frame.samples_per_channel_; k++) {
+        stereo_audio[k << 1] = frame_data[k];
+        stereo_audio[(k << 1) + 1] = frame_data[k];
+      }
+      if (fwrite(stereo_audio, sizeof(int16_t),
+                 2 * audio_frame.samples_per_channel_, pcm_file_) !=
+          static_cast<size_t>(2 * audio_frame.samples_per_channel_)) {
+        return;
+      }
+      delete[] stereo_audio;
+    }
+  } else {
+    if (fwrite(audio_frame.data(), sizeof(int16_t),
+               audio_frame.num_channels_ * audio_frame.samples_per_channel_,
+               pcm_file_) !=
+        static_cast<size_t>(audio_frame.num_channels_ *
+                            audio_frame.samples_per_channel_)) {
+      return;
+    }
+  }
+}
+
+void PCMFile::Write10MsData(const int16_t* playout_buffer,
+                            size_t length_smpls) {
+  if (fwrite(playout_buffer, sizeof(uint16_t), length_smpls, pcm_file_) !=
+      length_smpls) {
+    return;
+  }
+}
+
+void PCMFile::Close() {
+  fclose(pcm_file_);
+  pcm_file_ = NULL;
+  blocks_read_ = 0;
+}
+
+void PCMFile::FastForward(int num_10ms_blocks) {
+  const int channels = read_stereo_ ? 2 : 1;
+  long num_bytes_to_move =
+      num_10ms_blocks * sizeof(int16_t) * samples_10ms_ * channels;
+  int error = fseek(pcm_file_, num_bytes_to_move, SEEK_CUR);
+  RTC_DCHECK_EQ(error, 0);
+}
+
+void PCMFile::Rewind() {
+  rewind(pcm_file_);
+  end_of_file_ = false;
+  blocks_read_ = 0;
+}
+
+bool PCMFile::Rewinded() {
+  return rewinded_;
+}
+
+void PCMFile::SaveStereo(bool is_stereo) {
+  save_stereo_ = is_stereo;
+}
+
+void PCMFile::ReadStereo(bool is_stereo) {
+  read_stereo_ = is_stereo;
+}
+
+void PCMFile::SetNum10MsBlocksToRead(int value) {
+  num_10ms_blocks_to_read_ = value;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/PCMFile.h b/third_party/libwebrtc/modules/audio_coding/test/PCMFile.h
new file mode 100644
index 0000000000..5320aa63d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/PCMFile.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+#define MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+
+class PCMFile {
+ public:
+  PCMFile();
+  PCMFile(uint32_t timestamp);
+  ~PCMFile();
+
+  void Open(absl::string_view filename,
+            uint16_t frequency,
+            absl::string_view mode,
+            bool auto_rewind = false);
+
+  int32_t Read10MsData(AudioFrame& audio_frame);
+
+  void Write10MsData(const int16_t* playout_buffer, size_t length_smpls);
+  void Write10MsData(const AudioFrame& audio_frame);
+
+  uint16_t PayloadLength10Ms() const;
+  int32_t SamplingFrequency() const;
+  void Close();
+  bool EndOfFile() const { return end_of_file_; }
+  // Moves forward the specified number of 10 ms blocks. If a limit has been set
+  // with SetNum10MsBlocksToRead, fast-forwarding does not count towards this
+  // limit.
+  void FastForward(int num_10ms_blocks);
+  void Rewind();
+  static int16_t ChooseFile(std::string* file_name,
+                            int16_t max_len,
+                            uint16_t* frequency_hz);
+  bool Rewinded();
+  void SaveStereo(bool is_stereo = true);
+  void ReadStereo(bool is_stereo = true);
+  // If set, the reading will stop after the specified number of blocks have
+  // been read. When that has happened, EndOfFile() will return true. Calling
+  // Rewind() will reset the counter and start over.
+  void SetNum10MsBlocksToRead(int value);
+
+ private:
+  FILE* pcm_file_;
+  uint16_t samples_10ms_;
+  int32_t frequency_;
+  bool end_of_file_;
+  bool auto_rewind_;
+  bool rewinded_;
+  uint32_t timestamp_;
+  bool read_stereo_;
+  bool save_stereo_;
+  absl::optional<int> num_10ms_blocks_to_read_;
+  int blocks_read_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_PCMFILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.cc b/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.cc
new file mode 100644
index 0000000000..799e9c5b1f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.cc
@@ -0,0 +1,167 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/test/PacketLossTest.h"
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+ReceiverWithPacketLoss::ReceiverWithPacketLoss()
+    : loss_rate_(0),
+      burst_length_(1),
+      packet_counter_(0),
+      lost_packet_counter_(0),
+      burst_lost_counter_(burst_length_) {}
+
+void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
+                                   RTPStream* rtpStream,
+                                   absl::string_view out_file_name,
+                                   int channels,
+                                   int file_num,
+                                   int loss_rate,
+                                   int burst_length) {
+  loss_rate_ = loss_rate;
+  burst_length_ = burst_length;
+  burst_lost_counter_ = burst_length_;  // To prevent first packet gets lost.
+  rtc::StringBuilder ss;
+  ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
+  Receiver::Setup(acm, rtpStream, ss.str(), channels, file_num);
+}
+
+bool ReceiverWithPacketLoss::IncomingPacket() {
+  if (!_rtpStream->EndOfFile()) {
+    if (packet_counter_ == 0) {
+      _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
+                                               _payloadSizeBytes, &_nextTime);
+      if (_realPayloadSizeBytes == 0) {
+        if (_rtpStream->EndOfFile()) {
+          packet_counter_ = 0;
+          return true;
+        } else {
+          return false;
+        }
+      }
+    }
+
+    if (!PacketLost()) {
+      _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpHeader);
+    }
+    packet_counter_++;
+    _realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
+                                             _payloadSizeBytes, &_nextTime);
+    if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
+      packet_counter_ = 0;
+      lost_packet_counter_ = 0;
+    }
+  }
+  return true;
+}
+
+bool ReceiverWithPacketLoss::PacketLost() {
+  if (burst_lost_counter_ < burst_length_) {
+    lost_packet_counter_++;
+    burst_lost_counter_++;
+    return true;
+  }
+
+  if (lost_packet_counter_ * 100 < loss_rate_ * packet_counter_) {
+    lost_packet_counter_++;
+    burst_lost_counter_ = 1;
+    return true;
+  }
+  return false;
+}
+
+SenderWithFEC::SenderWithFEC() : expected_loss_rate_(0) {}
+
+void SenderWithFEC::Setup(AudioCodingModule* acm,
+                          RTPStream* rtpStream,
+                          absl::string_view in_file_name,
+                          int payload_type,
+                          SdpAudioFormat format,
+                          int expected_loss_rate) {
+  Sender::Setup(acm, rtpStream, in_file_name, format.clockrate_hz, payload_type,
+                format);
+  EXPECT_TRUE(SetFEC(true));
+  EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
+}
+
+bool SenderWithFEC::SetFEC(bool enable_fec) {
+  bool success = false;
+  _acm->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
+    if (*enc && (*enc)->SetFec(enable_fec)) {
+      success = true;
+    }
+  });
+  return success;
+}
+
+bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
+  if (_acm->SetPacketLossRate(expected_loss_rate) == 0) {
+    expected_loss_rate_ = expected_loss_rate;
+    return true;
+  }
+  return false;
+}
+
+PacketLossTest::PacketLossTest(int channels,
+                               int expected_loss_rate,
+                               int actual_loss_rate,
+                               int burst_length)
+    : channels_(channels),
+      in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz"
+                                   : "audio_coding/teststereo32kHz"),
+      sample_rate_hz_(32000),
+      expected_loss_rate_(expected_loss_rate),
+      actual_loss_rate_(actual_loss_rate),
+      burst_length_(burst_length) {}
+
+void PacketLossTest::Perform() {
+#ifndef WEBRTC_CODEC_OPUS
+  return;
+#else
+  RTPFile rtpFile;
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
+      AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+  SdpAudioFormat send_format = SdpAudioFormat("opus", 48000, 2);
+  if (channels_ == 2) {
+    send_format.parameters = {{"stereo", "1"}};
+  }
+
+  std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                                    "packet_loss_test");
+  rtpFile.Open(fileName.c_str(), "wb+");
+  rtpFile.WriteHeader();
+  SenderWithFEC sender;
+  sender.Setup(acm.get(), &rtpFile, in_file_name_, 120, send_format,
+               expected_loss_rate_);
+  sender.Run();
+  sender.Teardown();
+  rtpFile.Close();
+
+  rtpFile.Open(fileName.c_str(), "rb");
+  rtpFile.ReadHeader();
+  ReceiverWithPacketLoss receiver;
+  receiver.Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, 15,
+                 actual_loss_rate_, burst_length_);
+  receiver.Run();
+  receiver.Teardown();
+  rtpFile.Close();
+#endif
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.h b/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.h
new file mode 100644
index 0000000000..d841d65a1b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/PacketLossTest.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/test/EncodeDecodeTest.h"
+
+namespace webrtc {
+
+class ReceiverWithPacketLoss : public Receiver {
+ public:
+  ReceiverWithPacketLoss();
+  void Setup(AudioCodingModule* acm,
+             RTPStream* rtpStream,
+             absl::string_view out_file_name,
+             int channels,
+             int file_num,
+             int loss_rate,
+             int burst_length);
+  bool IncomingPacket() override;
+
+ protected:
+  bool PacketLost();
+  int loss_rate_;
+  int burst_length_;
+  int packet_counter_;
+  int lost_packet_counter_;
+  int burst_lost_counter_;
+};
+
+class SenderWithFEC : public Sender {
+ public:
+  SenderWithFEC();
+  void Setup(AudioCodingModule* acm,
+             RTPStream* rtpStream,
+             absl::string_view in_file_name,
+             int payload_type,
+             SdpAudioFormat format,
+             int expected_loss_rate);
+  bool SetPacketLossRate(int expected_loss_rate);
+  bool SetFEC(bool enable_fec);
+
+ protected:
+  int expected_loss_rate_;
+};
+
+class PacketLossTest {
+ public:
+  PacketLossTest(int channels,
+                 int expected_loss_rate_,
+                 int actual_loss_rate,
+                 int burst_length);
+  void Perform();
+
+ protected:
+  int channels_;
+  std::string in_file_name_;
+  int sample_rate_hz_;
+  int expected_loss_rate_;
+  int actual_loss_rate_;
+  int burst_length_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/RTPFile.cc b/third_party/libwebrtc/modules/audio_coding/test/RTPFile.cc
new file mode 100644
index 0000000000..0c2ab3c443
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/RTPFile.cc
@@ -0,0 +1,235 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "RTPFile.h"
+
+#include <stdlib.h>
+
+#include <limits>
+
+#include "absl/strings/string_view.h"
+
+#ifdef WIN32
+#include <Winsock2.h>
+#else
+#include <arpa/inet.h>
+#endif
+
+// TODO(tlegrand): Consider removing usage of gtest.
+#include "test/gtest.h"
+
+namespace webrtc {
+
+void RTPStream::ParseRTPHeader(RTPHeader* rtp_header,
+                               const uint8_t* rtpHeader) {
+  rtp_header->payloadType = rtpHeader[1];
+  rtp_header->sequenceNumber =
+      (static_cast<uint16_t>(rtpHeader[2]) << 8) | rtpHeader[3];
+  rtp_header->timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
+                          (static_cast<uint32_t>(rtpHeader[5]) << 16) |
+                          (static_cast<uint32_t>(rtpHeader[6]) << 8) |
+                          rtpHeader[7];
+  rtp_header->ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
+                     (static_cast<uint32_t>(rtpHeader[9]) << 16) |
+                     (static_cast<uint32_t>(rtpHeader[10]) << 8) |
+                     rtpHeader[11];
+}
+
+void RTPStream::MakeRTPheader(uint8_t* rtpHeader,
+                              uint8_t payloadType,
+                              int16_t seqNo,
+                              uint32_t timeStamp,
+                              uint32_t ssrc) {
+  rtpHeader[0] = 0x80;
+  rtpHeader[1] = payloadType;
+  rtpHeader[2] = (seqNo >> 8) & 0xFF;
+  rtpHeader[3] = seqNo & 0xFF;
+  rtpHeader[4] = timeStamp >> 24;
+  rtpHeader[5] = (timeStamp >> 16) & 0xFF;
+  rtpHeader[6] = (timeStamp >> 8) & 0xFF;
+  rtpHeader[7] = timeStamp & 0xFF;
+  rtpHeader[8] = ssrc >> 24;
+  rtpHeader[9] = (ssrc >> 16) & 0xFF;
+  rtpHeader[10] = (ssrc >> 8) & 0xFF;
+  rtpHeader[11] = ssrc & 0xFF;
+}
+
+RTPPacket::RTPPacket(uint8_t payloadType,
+                     uint32_t timeStamp,
+                     int16_t seqNo,
+                     const uint8_t* payloadData,
+                     size_t payloadSize,
+                     uint32_t frequency)
+    : payloadType(payloadType),
+      timeStamp(timeStamp),
+      seqNo(seqNo),
+      payloadSize(payloadSize),
+      frequency(frequency) {
+  if (payloadSize > 0) {
+    this->payloadData = new uint8_t[payloadSize];
+    memcpy(this->payloadData, payloadData, payloadSize);
+  }
+}
+
+RTPPacket::~RTPPacket() {
+  delete[] payloadData;
+}
+
+void RTPBuffer::Write(const uint8_t payloadType,
+                      const uint32_t timeStamp,
+                      const int16_t seqNo,
+                      const uint8_t* payloadData,
+                      const size_t payloadSize,
+                      uint32_t frequency) {
+  RTPPacket* packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
+                                    payloadSize, frequency);
+  MutexLock lock(&mutex_);
+  _rtpQueue.push(packet);
+}
+
+size_t RTPBuffer::Read(RTPHeader* rtp_header,
+                       uint8_t* payloadData,
+                       size_t payloadSize,
+                       uint32_t* offset) {
+  RTPPacket* packet;
+  {
+    MutexLock lock(&mutex_);
+    packet = _rtpQueue.front();
+    _rtpQueue.pop();
+  }
+  rtp_header->markerBit = 1;
+  rtp_header->payloadType = packet->payloadType;
+  rtp_header->sequenceNumber = packet->seqNo;
+  rtp_header->ssrc = 0;
+  rtp_header->timestamp = packet->timeStamp;
+  if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize) {
+    memcpy(payloadData, packet->payloadData, packet->payloadSize);
+  } else {
+    return 0;
+  }
+  *offset = (packet->timeStamp / (packet->frequency / 1000));
+
+  return packet->payloadSize;
+}
+
+bool RTPBuffer::EndOfFile() const {
+  MutexLock lock(&mutex_);
+  return _rtpQueue.empty();
+}
+
+void RTPFile::Open(absl::string_view filename, absl::string_view mode) {
+  std::string filename_str = std::string(filename);
+  if ((_rtpFile = fopen(filename_str.c_str(), std::string(mode).c_str())) ==
+      NULL) {
+    printf("Cannot write file %s.\n", filename_str.c_str());
+    ADD_FAILURE() << "Unable to write file";
+    exit(1);
+  }
+}
+
+void RTPFile::Close() {
+  if (_rtpFile != NULL) {
+    fclose(_rtpFile);
+    _rtpFile = NULL;
+  }
+}
+
+void RTPFile::WriteHeader() {
+  // Write data in a format that NetEQ and RTP Play can parse
+  fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
+  uint32_t dummy_variable = 0;
+  // should be converted to network endian format, but does not matter when 0
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
+  fflush(_rtpFile);
+}
+
+void RTPFile::ReadHeader() {
+  uint32_t start_sec, start_usec, source;
+  uint16_t port, padding;
+  char fileHeader[40];
+  EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
+  EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile));
+  start_sec = ntohl(start_sec);
+  EXPECT_EQ(1u, fread(&start_usec, 4, 1, _rtpFile));
+  start_usec = ntohl(start_usec);
+  EXPECT_EQ(1u, fread(&source, 4, 1, _rtpFile));
+  source = ntohl(source);
+  EXPECT_EQ(1u, fread(&port, 2, 1, _rtpFile));
+  port = ntohs(port);
+  EXPECT_EQ(1u, fread(&padding, 2, 1, _rtpFile));
+  padding = ntohs(padding);
+}
+
+void RTPFile::Write(const uint8_t payloadType,
+                    const uint32_t timeStamp,
+                    const int16_t seqNo,
+                    const uint8_t* payloadData,
+                    const size_t payloadSize,
+                    uint32_t frequency) {
+  /* write RTP packet to file */
+  uint8_t rtpHeader[12];
+  MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
+  ASSERT_LE(12 + payloadSize + 8, std::numeric_limits<u_short>::max());
+  uint16_t lengthBytes = htons(static_cast<u_short>(12 + payloadSize + 8));
+  uint16_t plen = htons(static_cast<u_short>(12 + payloadSize));
+  uint32_t offsetMs;
+
+  offsetMs = (timeStamp / (frequency / 1000));
+  offsetMs = htonl(offsetMs);
+  EXPECT_EQ(1u, fwrite(&lengthBytes, 2, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&plen, 2, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&offsetMs, 4, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&rtpHeader, 12, 1, _rtpFile));
+  EXPECT_EQ(payloadSize, fwrite(payloadData, 1, payloadSize, _rtpFile));
+}
+
+size_t RTPFile::Read(RTPHeader* rtp_header,
+                     uint8_t* payloadData,
+                     size_t payloadSize,
+                     uint32_t* offset) {
+  uint16_t lengthBytes;
+  uint16_t plen;
+  uint8_t rtpHeader[12];
+  size_t read_len = fread(&lengthBytes, 2, 1, _rtpFile);
+  /* Check if we have reached end of file. */
+  if ((read_len == 0) && feof(_rtpFile)) {
+    _rtpEOF = true;
+    return 0;
+  }
+  EXPECT_EQ(1u, fread(&plen, 2, 1, _rtpFile));
+  EXPECT_EQ(1u, fread(offset, 4, 1, _rtpFile));
+  lengthBytes = ntohs(lengthBytes);
+  plen = ntohs(plen);
+  *offset = ntohl(*offset);
+  EXPECT_GT(plen, 11);
+
+  EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
+  ParseRTPHeader(rtp_header, rtpHeader);
+  EXPECT_EQ(lengthBytes, plen + 8);
+
+  if (plen == 0) {
+    return 0;
+  }
+  if (lengthBytes < 20) {
+    return 0;
+  }
+  if (payloadSize < static_cast<size_t>((lengthBytes - 20))) {
+    return 0;
+  }
+  lengthBytes -= 20;
+  EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
+  return lengthBytes;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/RTPFile.h b/third_party/libwebrtc/modules/audio_coding/test/RTPFile.h
new file mode 100644
index 0000000000..b796491da9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/RTPFile.h
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+#define MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+
+#include <stdio.h>
+
+#include <queue>
+
+#include "absl/strings/string_view.h"
+#include "api/rtp_headers.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class RTPStream {
+ public:
+  virtual ~RTPStream() {}
+
+  virtual void Write(uint8_t payloadType,
+                     uint32_t timeStamp,
+                     int16_t seqNo,
+                     const uint8_t* payloadData,
+                     size_t payloadSize,
+                     uint32_t frequency) = 0;
+
+  // Returns the packet's payload size. Zero should be treated as an
+  // end-of-stream (in the case that EndOfFile() is true) or an error.
+  virtual size_t Read(RTPHeader* rtp_Header,
+                      uint8_t* payloadData,
+                      size_t payloadSize,
+                      uint32_t* offset) = 0;
+  virtual bool EndOfFile() const = 0;
+
+ protected:
+  void MakeRTPheader(uint8_t* rtpHeader,
+                     uint8_t payloadType,
+                     int16_t seqNo,
+                     uint32_t timeStamp,
+                     uint32_t ssrc);
+
+  void ParseRTPHeader(RTPHeader* rtp_header, const uint8_t* rtpHeader);
+};
+
+class RTPPacket {
+ public:
+  RTPPacket(uint8_t payloadType,
+            uint32_t timeStamp,
+            int16_t seqNo,
+            const uint8_t* payloadData,
+            size_t payloadSize,
+            uint32_t frequency);
+
+  ~RTPPacket();
+
+  uint8_t payloadType;
+  uint32_t timeStamp;
+  int16_t seqNo;
+  uint8_t* payloadData;
+  size_t payloadSize;
+  uint32_t frequency;
+};
+
+class RTPBuffer : public RTPStream {
+ public:
+  RTPBuffer() = default;
+
+  ~RTPBuffer() = default;
+
+  void Write(uint8_t payloadType,
+             uint32_t timeStamp,
+             int16_t seqNo,
+             const uint8_t* payloadData,
+             size_t payloadSize,
+             uint32_t frequency) override;
+
+  size_t Read(RTPHeader* rtp_header,
+              uint8_t* payloadData,
+              size_t payloadSize,
+              uint32_t* offset) override;
+
+  bool EndOfFile() const override;
+
+ private:
+  mutable Mutex mutex_;
+  std::queue<RTPPacket*> _rtpQueue RTC_GUARDED_BY(&mutex_);
+};
+
+class RTPFile : public RTPStream {
+ public:
+  ~RTPFile() {}
+
+  RTPFile() : _rtpFile(NULL), _rtpEOF(false) {}
+
+  void Open(absl::string_view outFilename, absl::string_view mode);
+
+  void Close();
+
+  void WriteHeader();
+
+  void ReadHeader();
+
+  void Write(uint8_t payloadType,
+             uint32_t timeStamp,
+             int16_t seqNo,
+             const uint8_t* payloadData,
+             size_t payloadSize,
+             uint32_t frequency) override;
+
+  size_t Read(RTPHeader* rtp_header,
+              uint8_t* payloadData,
+              size_t payloadSize,
+              uint32_t* offset) override;
+
+  bool EndOfFile() const override { return _rtpEOF; }
+
+ private:
+  FILE* _rtpFile;
+  bool _rtpEOF;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_RTPFILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestAllCodecs.cc b/third_party/libwebrtc/modules/audio_coding/test/TestAllCodecs.cc
new file mode 100644
index 0000000000..b44037d732
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestAllCodecs.cc
@@ -0,0 +1,412 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/test/TestAllCodecs.h"
+
+#include <cstdio>
+#include <limits>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+// Description of the test:
+// In this test we set up a one-way communication channel from a participant
+// called "a" to a participant called "b".
+// a -> channel_a_to_b -> b
+//
+// The test loops through all available mono codecs, encode at "a" sends over
+// the channel, and decodes at "b".
+
+#define CHECK_ERROR(f)                      \
+  do {                                      \
+    EXPECT_GE(f, 0) << "Error Calling API"; \
+  } while (0)
+
+namespace {
+const size_t kVariableSize = std::numeric_limits<size_t>::max();
+}
+
+namespace webrtc {
+
+// Class for simulating packet handling.
+TestPack::TestPack()
+    : receiver_acm_(NULL),
+      sequence_number_(0),
+      timestamp_diff_(0),
+      last_in_timestamp_(0),
+      total_bytes_(0),
+      payload_size_(0) {}
+
+TestPack::~TestPack() {}
+
+void TestPack::RegisterReceiverACM(AudioCodingModule* acm) {
+  receiver_acm_ = acm;
+  return;
+}
+
+int32_t TestPack::SendData(AudioFrameType frame_type,
+                           uint8_t payload_type,
+                           uint32_t timestamp,
+                           const uint8_t* payload_data,
+                           size_t payload_size,
+                           int64_t absolute_capture_timestamp_ms) {
+  RTPHeader rtp_header;
+  int32_t status;
+
+  rtp_header.markerBit = false;
+  rtp_header.ssrc = 0;
+  rtp_header.sequenceNumber = sequence_number_++;
+  rtp_header.payloadType = payload_type;
+  rtp_header.timestamp = timestamp;
+
+  if (frame_type == AudioFrameType::kEmptyFrame) {
+    // Skip this frame.
+    return 0;
+  }
+
+  // Only run mono for all test cases.
+  memcpy(payload_data_, payload_data, payload_size);
+
+  status =
+      receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_header);
+
+  payload_size_ = payload_size;
+  timestamp_diff_ = timestamp - last_in_timestamp_;
+  last_in_timestamp_ = timestamp;
+  total_bytes_ += payload_size;
+  return status;
+}
+
+size_t TestPack::payload_size() {
+  return payload_size_;
+}
+
+uint32_t TestPack::timestamp_diff() {
+  return timestamp_diff_;
+}
+
+void TestPack::reset_payload_size() {
+  payload_size_ = 0;
+}
+
+TestAllCodecs::TestAllCodecs()
+    : acm_a_(AudioCodingModule::Create(
+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+      acm_b_(AudioCodingModule::Create(
+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+      channel_a_to_b_(NULL),
+      test_count_(0),
+      packet_size_samples_(0),
+      packet_size_bytes_(0) {}
+
+TestAllCodecs::~TestAllCodecs() {
+  if (channel_a_to_b_ != NULL) {
+    delete channel_a_to_b_;
+    channel_a_to_b_ = NULL;
+  }
+}
+
+void TestAllCodecs::Perform() {
+  const std::string file_name =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  infile_a_.Open(file_name, 32000, "rb");
+
+  acm_a_->InitializeReceiver();
+  acm_b_->InitializeReceiver();
+
+  acm_b_->SetReceiveCodecs({{107, {"L16", 8000, 1}},
+                            {108, {"L16", 16000, 1}},
+                            {109, {"L16", 32000, 1}},
+                            {111, {"L16", 8000, 2}},
+                            {112, {"L16", 16000, 2}},
+                            {113, {"L16", 32000, 2}},
+                            {0, {"PCMU", 8000, 1}},
+                            {110, {"PCMU", 8000, 2}},
+                            {8, {"PCMA", 8000, 1}},
+                            {118, {"PCMA", 8000, 2}},
+                            {102, {"ILBC", 8000, 1}},
+                            {9, {"G722", 8000, 1}},
+                            {119, {"G722", 8000, 2}},
+                            {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+                            {13, {"CN", 8000, 1}},
+                            {98, {"CN", 16000, 1}},
+                            {99, {"CN", 32000, 1}}});
+
+  // Create and connect the channel
+  channel_a_to_b_ = new TestPack;
+  acm_a_->RegisterTransportCallback(channel_a_to_b_);
+  channel_a_to_b_->RegisterReceiverACM(acm_b_.get());
+
+  // All codecs are tested for all allowed sampling frequencies, rates and
+  // packet sizes.
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_g722[] = "G722";
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 480, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 640, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 800, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#ifdef WEBRTC_CODEC_ILBC
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_ilbc[] = "ILBC";
+  RegisterSendCodec('A', codec_ilbc, 8000, 13300, 240, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_ilbc, 8000, 13300, 480, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_ilbc, 8000, 15200, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_ilbc, 8000, 15200, 320, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#endif
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_l16[] = "L16";
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 80, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 240, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 320, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+
+  test_count_++;
+  OpenOutFile(test_count_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 480, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 640, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+
+  test_count_++;
+  OpenOutFile(test_count_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_pcma[] = "PCMA";
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, 0);
+  Run(channel_a_to_b_);
+
+  char codec_pcmu[] = "PCMU";
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 480, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#ifdef WEBRTC_CODEC_OPUS
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_opus[] = "OPUS";
+  RegisterSendCodec('A', codec_opus, 48000, 6000, 480, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 20000, 480 * 2, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 32000, 480 * 4, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 48000, 480, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 4, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 96000, 480 * 6, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 500000, 480 * 2, kVariableSize);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#endif
+}
+
+// Register Codec to use in the test
+//
+// Input:  side             - which ACM to use, 'A' or 'B'
+//         codec_name       - name to use when register the codec
+//         sampling_freq_hz - sampling frequency in Herz
+//         rate             - bitrate in bytes
+//         packet_size      - packet size in samples
+//         extra_byte       - if extra bytes needed compared to the bitrate
+//                            used when registering, can be an internal header
+//                            set to kVariableSize if the codec is a variable
+//                            rate codec
+void TestAllCodecs::RegisterSendCodec(char side,
+                                      char* codec_name,
+                                      int32_t sampling_freq_hz,
+                                      int rate,
+                                      int packet_size,
+                                      size_t extra_byte) {
+  // Store packet-size in samples, used to validate the received packet.
+  // If G.722, store half the size to compensate for the timestamp bug in the
+  // RFC for G.722.
+  int clockrate_hz = sampling_freq_hz;
+  size_t num_channels = 1;
+  if (absl::EqualsIgnoreCase(codec_name, "G722")) {
+    packet_size_samples_ = packet_size / 2;
+    clockrate_hz = sampling_freq_hz / 2;
+  } else if (absl::EqualsIgnoreCase(codec_name, "OPUS")) {
+    packet_size_samples_ = packet_size;
+    num_channels = 2;
+  } else {
+    packet_size_samples_ = packet_size;
+  }
+
+  // Store the expected packet size in bytes, used to validate the received
+  // packet. If variable rate codec (extra_byte == -1), set to -1.
+  if (extra_byte != kVariableSize) {
+    // Add 0.875 to always round up to a whole byte
+    packet_size_bytes_ =
+        static_cast<size_t>(static_cast<float>(packet_size * rate) /
+                                static_cast<float>(sampling_freq_hz * 8) +
+                            0.875) +
+        extra_byte;
+  } else {
+    // Packets will have a variable size.
+    packet_size_bytes_ = kVariableSize;
+  }
+
+  // Set pointer to the ACM where to register the codec.
+  AudioCodingModule* my_acm = NULL;
+  switch (side) {
+    case 'A': {
+      my_acm = acm_a_.get();
+      break;
+    }
+    case 'B': {
+      my_acm = acm_b_.get();
+      break;
+    }
+    default: {
+      break;
+    }
+  }
+  ASSERT_TRUE(my_acm != NULL);
+
+  auto factory = CreateBuiltinAudioEncoderFactory();
+  constexpr int payload_type = 17;
+  SdpAudioFormat format = {codec_name, clockrate_hz, num_channels};
+  format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
+      packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
+  my_acm->SetEncoder(
+      factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
+}
+
+void TestAllCodecs::Run(TestPack* channel) {
+  AudioFrame audio_frame;
+
+  int32_t out_freq_hz = outfile_b_.SamplingFrequency();
+  size_t receive_size;
+  uint32_t timestamp_diff;
+  channel->reset_payload_size();
+  int error_count = 0;
+  int counter = 0;
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  infile_a_.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  infile_a_.FastForward(100);
+
+  while (!infile_a_.EndOfFile()) {
+    // Add 10 msec to ACM.
+    infile_a_.Read10MsData(audio_frame);
+    CHECK_ERROR(acm_a_->Add10MsData(audio_frame));
+
+    // Verify that the received packet size matches the settings.
+    receive_size = channel->payload_size();
+    if (receive_size) {
+      if ((receive_size != packet_size_bytes_) &&
+          (packet_size_bytes_ != kVariableSize)) {
+        error_count++;
+      }
+
+      // Verify that the timestamp is updated with expected length. The counter
+      // is used to avoid problems when switching codec or frame size in the
+      // test.
+      timestamp_diff = channel->timestamp_diff();
+      if ((counter > 10) &&
+          (static_cast<int>(timestamp_diff) != packet_size_samples_) &&
+          (packet_size_samples_ > -1))
+        error_count++;
+    }
+
+    // Run received side of ACM.
+    bool muted;
+    CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
+
+    // Write output speech to file.
+    outfile_b_.Write10MsData(audio_frame.data(),
+                             audio_frame.samples_per_channel_);
+
+    // Update loop counter
+    counter++;
+  }
+
+  EXPECT_EQ(0, error_count);
+
+  if (infile_a_.EndOfFile()) {
+    infile_a_.Rewind();
+  }
+}
+
+void TestAllCodecs::OpenOutFile(int test_number) {
+  std::string filename = webrtc::test::OutputPath();
+  rtc::StringBuilder test_number_str;
+  test_number_str << test_number;
+  filename += "testallcodecs_out_";
+  filename += test_number_str.str();
+  filename += ".pcm";
+  outfile_b_.Open(filename, 32000, "wb");
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestAllCodecs.h b/third_party/libwebrtc/modules/audio_coding/test/TestAllCodecs.h
new file mode 100644
index 0000000000..0c276414e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestAllCodecs.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+#define MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+
+#include <memory>
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/PCMFile.h"
+
+namespace webrtc {
+
+class TestPack : public AudioPacketizationCallback {
+ public:
+  TestPack();
+  ~TestPack();
+
+  void RegisterReceiverACM(AudioCodingModule* acm);
+
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_size,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  size_t payload_size();
+  uint32_t timestamp_diff();
+  void reset_payload_size();
+
+ private:
+  AudioCodingModule* receiver_acm_;
+  uint16_t sequence_number_;
+  uint8_t payload_data_[60 * 32 * 2 * 2];
+  uint32_t timestamp_diff_;
+  uint32_t last_in_timestamp_;
+  uint64_t total_bytes_;
+  size_t payload_size_;
+};
+
+class TestAllCodecs {
+ public:
+  TestAllCodecs();
+  ~TestAllCodecs();
+
+  void Perform();
+
+ private:
+  // The default value of '-1' indicates that the registration is based only on
+  // codec name, and a sampling frequency matching is not required.
+  // This is useful for codecs which support several sampling frequency.
+  // Note! Only mono mode is tested in this test.
+  void RegisterSendCodec(char side,
+                         char* codec_name,
+                         int32_t sampling_freq_hz,
+                         int rate,
+                         int packet_size,
+                         size_t extra_byte);
+
+  void Run(TestPack* channel);
+  void OpenOutFile(int test_number);
+
+  std::unique_ptr<AudioCodingModule> acm_a_;
+  std::unique_ptr<AudioCodingModule> acm_b_;
+  TestPack* channel_a_to_b_;
+  PCMFile infile_a_;
+  PCMFile outfile_b_;
+  int test_count_;
+  int packet_size_samples_;
+  size_t packet_size_bytes_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestRedFec.cc b/third_party/libwebrtc/modules/audio_coding/test/TestRedFec.cc
new file mode 100644
index 0000000000..fff48b27bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestRedFec.cc
@@ -0,0 +1,200 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/test/TestRedFec.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "api/audio_codecs/L16/audio_decoder_L16.h"
+#include "api/audio_codecs/L16/audio_encoder_L16.h"
+#include "api/audio_codecs/audio_decoder_factory_template.h"
+#include "api/audio_codecs/audio_encoder_factory_template.h"
+#include "api/audio_codecs/g711/audio_decoder_g711.h"
+#include "api/audio_codecs/g711/audio_encoder_g711.h"
+#include "api/audio_codecs/g722/audio_decoder_g722.h"
+#include "api/audio_codecs/g722/audio_encoder_g722.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TestRedFec::TestRedFec()
+    : encoder_factory_(CreateAudioEncoderFactory<AudioEncoderG711,
+                                                 AudioEncoderG722,
+                                                 AudioEncoderL16,
+                                                 AudioEncoderOpus>()),
+      decoder_factory_(CreateAudioDecoderFactory<AudioDecoderG711,
+                                                 AudioDecoderG722,
+                                                 AudioDecoderL16,
+                                                 AudioDecoderOpus>()),
+      _acmA(AudioCodingModule::Create(
+          AudioCodingModule::Config(decoder_factory_))),
+      _acmB(AudioCodingModule::Create(
+          AudioCodingModule::Config(decoder_factory_))),
+      _channelA2B(NULL),
+      _testCntr(0) {}
+
+TestRedFec::~TestRedFec() {
+  if (_channelA2B != NULL) {
+    delete _channelA2B;
+    _channelA2B = NULL;
+  }
+}
+
+void TestRedFec::Perform() {
+  const std::string file_name =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  _inFileA.Open(file_name, 32000, "rb");
+
+  ASSERT_EQ(0, _acmA->InitializeReceiver());
+  ASSERT_EQ(0, _acmB->InitializeReceiver());
+
+  // Create and connect the channel
+  _channelA2B = new Channel;
+  _acmA->RegisterTransportCallback(_channelA2B);
+  _channelA2B->RegisterReceiverACM(_acmB.get());
+
+  RegisterSendCodec(_acmA, {"L16", 8000, 1}, Vad::kVadAggressive, true);
+
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+  // Switch to another 8 kHz codec; RED should remain switched on.
+  RegisterSendCodec(_acmA, {"PCMU", 8000, 1}, Vad::kVadAggressive, true);
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+  // Switch to a 16 kHz codec; RED should be switched off.
+  RegisterSendCodec(_acmA, {"G722", 8000, 1}, Vad::kVadAggressive, false);
+
+  OpenOutFile(_testCntr);
+  RegisterSendCodec(_acmA, {"G722", 8000, 1}, Vad::kVadAggressive, false);
+  Run();
+  RegisterSendCodec(_acmA, {"G722", 8000, 1}, Vad::kVadAggressive, false);
+  Run();
+  _outFileB.Close();
+
+  _channelA2B->SetFECTestWithPacketLoss(true);
+  // Following tests are under packet losses.
+
+  // Switch to a 16 kHz codec; RED should be switched off.
+  RegisterSendCodec(_acmA, {"G722", 8000, 1}, Vad::kVadAggressive, false);
+
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+  RegisterSendCodec(_acmA, {"opus", 48000, 2}, absl::nullopt, false);
+
+  // _channelA2B imposes 25% packet loss rate.
+  EXPECT_EQ(0, _acmA->SetPacketLossRate(25));
+
+  _acmA->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
+    EXPECT_EQ(true, (*enc)->SetFec(true));
+  });
+
+  OpenOutFile(_testCntr);
+  Run();
+
+  // Switch to L16 with RED.
+  RegisterSendCodec(_acmA, {"L16", 8000, 1}, absl::nullopt, true);
+  Run();
+
+  // Switch to Opus again.
+  RegisterSendCodec(_acmA, {"opus", 48000, 2}, absl::nullopt, false);
+  _acmA->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
+    EXPECT_EQ(true, (*enc)->SetFec(false));
+  });
+  Run();
+
+  _acmA->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
+    EXPECT_EQ(true, (*enc)->SetFec(true));
+  });
+  _outFileB.Close();
+}
+
+void TestRedFec::RegisterSendCodec(
+    const std::unique_ptr<AudioCodingModule>& acm,
+    const SdpAudioFormat& codec_format,
+    absl::optional<Vad::Aggressiveness> vad_mode,
+    bool use_red) {
+  constexpr int payload_type = 17, cn_payload_type = 27, red_payload_type = 37;
+  const auto& other_acm = &acm == &_acmA ? _acmB : _acmA;
+
+  auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
+                                                    absl::nullopt);
+  EXPECT_NE(encoder, nullptr);
+  std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}};
+  if (!absl::EqualsIgnoreCase(codec_format.name, "opus")) {
+    if (vad_mode.has_value()) {
+      AudioEncoderCngConfig config;
+      config.speech_encoder = std::move(encoder);
+      config.num_channels = 1;
+      config.payload_type = cn_payload_type;
+      config.vad_mode = vad_mode.value();
+      encoder = CreateComfortNoiseEncoder(std::move(config));
+      receive_codecs.emplace(std::make_pair(
+          cn_payload_type, SdpAudioFormat("CN", codec_format.clockrate_hz, 1)));
+    }
+    if (use_red) {
+      AudioEncoderCopyRed::Config config;
+      config.payload_type = red_payload_type;
+      config.speech_encoder = std::move(encoder);
+      encoder = std::make_unique<AudioEncoderCopyRed>(std::move(config),
+                                                      field_trials_);
+      receive_codecs.emplace(
+          std::make_pair(red_payload_type,
+                         SdpAudioFormat("red", codec_format.clockrate_hz, 1)));
+    }
+  }
+  acm->SetEncoder(std::move(encoder));
+  other_acm->SetReceiveCodecs(receive_codecs);
+}
+
+void TestRedFec::Run() {
+  AudioFrame audioFrame;
+  int32_t outFreqHzB = _outFileB.SamplingFrequency();
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  _inFileA.SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the file starts with silence.
+  _inFileA.FastForward(100);
+
+  while (!_inFileA.EndOfFile()) {
+    EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
+    EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
+    bool muted;
+    EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
+    _outFileB.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_);
+  }
+  _inFileA.Rewind();
+}
+
+void TestRedFec::OpenOutFile(int16_t test_number) {
+  std::string file_name;
+  rtc::StringBuilder file_stream;
+  file_stream << webrtc::test::OutputPath();
+  file_stream << "TestRedFec_outFile_";
+  file_stream << test_number << ".pcm";
+  file_name = file_stream.str();
+  _outFileB.Open(file_name, 16000, "wb");
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestRedFec.h b/third_party/libwebrtc/modules/audio_coding/test/TestRedFec.h
new file mode 100644
index 0000000000..dbadd88487
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestRedFec.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+#define MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+
+#include <memory>
+#include <string>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "common_audio/vad/include/vad.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+
+class TestRedFec {
+ public:
+  explicit TestRedFec();
+  ~TestRedFec();
+
+  void Perform();
+
+ private:
+  void RegisterSendCodec(const std::unique_ptr<AudioCodingModule>& acm,
+                         const SdpAudioFormat& codec_format,
+                         absl::optional<Vad::Aggressiveness> vad_mode,
+                         bool use_red);
+  void Run();
+  void OpenOutFile(int16_t testNumber);
+
+  test::ScopedKeyValueConfig field_trials_;
+  const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+  const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  std::unique_ptr<AudioCodingModule> _acmA;
+  std::unique_ptr<AudioCodingModule> _acmB;
+
+  Channel* _channelA2B;
+
+  PCMFile _inFileA;
+  PCMFile _outFileB;
+  int16_t _testCntr;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestStereo.cc b/third_party/libwebrtc/modules/audio_coding/test/TestStereo.cc
new file mode 100644
index 0000000000..599fafb602
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestStereo.cc
@@ -0,0 +1,599 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/test/TestStereo.h"
+
+#include <string>
+
+#include "absl/strings/match.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+// Class for simulating packet handling
+TestPackStereo::TestPackStereo()
+    : receiver_acm_(NULL),
+      seq_no_(0),
+      timestamp_diff_(0),
+      last_in_timestamp_(0),
+      total_bytes_(0),
+      payload_size_(0),
+      lost_packet_(false) {}
+
+TestPackStereo::~TestPackStereo() {}
+
+void TestPackStereo::RegisterReceiverACM(AudioCodingModule* acm) {
+  receiver_acm_ = acm;
+  return;
+}
+
+int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
+                                 const uint8_t payload_type,
+                                 const uint32_t timestamp,
+                                 const uint8_t* payload_data,
+                                 const size_t payload_size,
+                                 int64_t absolute_capture_timestamp_ms) {
+  RTPHeader rtp_header;
+  int32_t status = 0;
+
+  rtp_header.markerBit = false;
+  rtp_header.ssrc = 0;
+  rtp_header.sequenceNumber = seq_no_++;
+  rtp_header.payloadType = payload_type;
+  rtp_header.timestamp = timestamp;
+  if (frame_type == AudioFrameType::kEmptyFrame) {
+    // Skip this frame
+    return 0;
+  }
+
+  if (lost_packet_ == false) {
+    status =
+        receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_header);
+
+    if (frame_type != AudioFrameType::kAudioFrameCN) {
+      payload_size_ = static_cast<int>(payload_size);
+    } else {
+      payload_size_ = -1;
+    }
+
+    timestamp_diff_ = timestamp - last_in_timestamp_;
+    last_in_timestamp_ = timestamp;
+    total_bytes_ += payload_size;
+  }
+  return status;
+}
+
+uint16_t TestPackStereo::payload_size() {
+  return static_cast<uint16_t>(payload_size_);
+}
+
+uint32_t TestPackStereo::timestamp_diff() {
+  return timestamp_diff_;
+}
+
+void TestPackStereo::reset_payload_size() {
+  payload_size_ = 0;
+}
+
+void TestPackStereo::set_codec_mode(enum StereoMonoMode mode) {
+  codec_mode_ = mode;
+}
+
+void TestPackStereo::set_lost_packet(bool lost) {
+  lost_packet_ = lost;
+}
+
+TestStereo::TestStereo()
+    : acm_a_(AudioCodingModule::Create(
+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+      acm_b_(AudioCodingModule::Create(
+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+      channel_a2b_(NULL),
+      test_cntr_(0),
+      pack_size_samp_(0),
+      pack_size_bytes_(0),
+      counter_(0) {}
+
+TestStereo::~TestStereo() {
+  if (channel_a2b_ != NULL) {
+    delete channel_a2b_;
+    channel_a2b_ = NULL;
+  }
+}
+
+void TestStereo::Perform() {
+  uint16_t frequency_hz;
+  int audio_channels;
+  int codec_channels;
+
+  // Open both mono and stereo test files in 32 kHz.
+  const std::string file_name_stereo =
+      webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
+  const std::string file_name_mono =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  frequency_hz = 32000;
+  in_file_stereo_ = new PCMFile();
+  in_file_mono_ = new PCMFile();
+  in_file_stereo_->Open(file_name_stereo, frequency_hz, "rb");
+  in_file_stereo_->ReadStereo(true);
+  in_file_mono_->Open(file_name_mono, frequency_hz, "rb");
+  in_file_mono_->ReadStereo(false);
+
+  // Create and initialize two ACMs, one for each side of a one-to-one call.
+  ASSERT_TRUE((acm_a_.get() != NULL) && (acm_b_.get() != NULL));
+  EXPECT_EQ(0, acm_a_->InitializeReceiver());
+  EXPECT_EQ(0, acm_b_->InitializeReceiver());
+
+  acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
+                            {104, {"ISAC", 32000, 1}},
+                            {107, {"L16", 8000, 1}},
+                            {108, {"L16", 16000, 1}},
+                            {109, {"L16", 32000, 1}},
+                            {111, {"L16", 8000, 2}},
+                            {112, {"L16", 16000, 2}},
+                            {113, {"L16", 32000, 2}},
+                            {0, {"PCMU", 8000, 1}},
+                            {110, {"PCMU", 8000, 2}},
+                            {8, {"PCMA", 8000, 1}},
+                            {118, {"PCMA", 8000, 2}},
+                            {102, {"ILBC", 8000, 1}},
+                            {9, {"G722", 8000, 1}},
+                            {119, {"G722", 8000, 2}},
+                            {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
+                            {13, {"CN", 8000, 1}},
+                            {98, {"CN", 16000, 1}},
+                            {99, {"CN", 32000, 1}}});
+
+  // Create and connect the channel.
+  channel_a2b_ = new TestPackStereo;
+  EXPECT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_));
+  channel_a2b_->RegisterReceiverACM(acm_b_.get());
+
+  char codec_pcma_temp[] = "PCMA";
+  RegisterSendCodec('A', codec_pcma_temp, 8000, 64000, 80, 2);
+
+  //
+  // Test Stereo-To-Stereo for all codecs.
+  //
+  audio_channels = 2;
+  codec_channels = 2;
+
+  // All codecs are tested for all allowed sampling frequencies, rates and
+  // packet sizes.
+  channel_a2b_->set_codec_mode(kStereo);
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  char codec_g722[] = "G722";
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 320, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 480, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 640, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 800, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 960, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  channel_a2b_->set_codec_mode(kStereo);
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  char codec_l16[] = "L16";
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 160, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 240, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 320, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 320, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 480, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 640, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 640, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#ifdef PCMA_AND_PCMU
+  channel_a2b_->set_codec_mode(kStereo);
+  audio_channels = 2;
+  codec_channels = 2;
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  char codec_pcma[] = "PCMA";
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  char codec_pcmu[] = "PCMU";
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 480, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  channel_a2b_->set_codec_mode(kStereo);
+  audio_channels = 2;
+  codec_channels = 2;
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+
+  char codec_opus[] = "opus";
+  // Run Opus with 10 ms frame size.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  // Run Opus with 20 ms frame size.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 2, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  // Run Opus with 40 ms frame size.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 4, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  // Run Opus with 60 ms frame size.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 6, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  // Run Opus with 20 ms frame size and different bitrates.
+  RegisterSendCodec('A', codec_opus, 48000, 40000, 960, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_opus, 48000, 510000, 960, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+  //
+  // Test Mono-To-Stereo for all codecs.
+  //
+  audio_channels = 1;
+  codec_channels = 2;
+
+  test_cntr_++;
+  channel_a2b_->set_codec_mode(kStereo);
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  test_cntr_++;
+  channel_a2b_->set_codec_mode(kStereo);
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#ifdef PCMA_AND_PCMU
+  test_cntr_++;
+  channel_a2b_->set_codec_mode(kStereo);
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  // Keep encode and decode in stereo.
+  test_cntr_++;
+  channel_a2b_->set_codec_mode(kStereo);
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 960, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+
+  // Encode in mono, decode in stereo mode.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 960, 1);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+
+  //
+  // Test Stereo-To-Mono for all codecs.
+  //
+  audio_channels = 2;
+  codec_channels = 1;
+  channel_a2b_->set_codec_mode(kMono);
+
+  // Run stereo audio and mono codec.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#ifdef PCMA_AND_PCMU
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  // Encode and decode in mono.
+  RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels);
+  acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
+  Run(channel_a2b_, audio_channels, codec_channels);
+
+  // Encode in stereo, decode in mono.
+  RegisterSendCodec('A', codec_opus, 48000, 32000, 960, 2);
+  Run(channel_a2b_, audio_channels, codec_channels);
+
+  out_file_.Close();
+
+  // Test switching between decoding mono and stereo for Opus.
+
+  // Decode in mono.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+  // Decode in stereo.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
+  Run(channel_a2b_, audio_channels, 2);
+  out_file_.Close();
+  // Decode in mono.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+
+  // Delete the file pointers.
+  delete in_file_stereo_;
+  delete in_file_mono_;
+}
+
+// Register Codec to use in the test
+//
+// Input:   side             - which ACM to use, 'A' or 'B'
+//          codec_name       - name to use when register the codec
+//          sampling_freq_hz - sampling frequency in Herz
+//          rate             - bitrate in bytes
+//          pack_size        - packet size in samples
+//          channels         - number of channels; 1 for mono, 2 for stereo
+void TestStereo::RegisterSendCodec(char side,
+                                   char* codec_name,
+                                   int32_t sampling_freq_hz,
+                                   int rate,
+                                   int pack_size,
+                                   int channels) {
+  // Store packet size in samples, used to validate the received packet
+  pack_size_samp_ = pack_size;
+
+  // Store the expected packet size in bytes, used to validate the received
+  // packet. Add 0.875 to always round up to a whole byte.
+  pack_size_bytes_ = (uint16_t)(static_cast<float>(pack_size * rate) /
+                                    static_cast<float>(sampling_freq_hz * 8) +
+                                0.875);
+
+  // Set pointer to the ACM where to register the codec
+  AudioCodingModule* my_acm = NULL;
+  switch (side) {
+    case 'A': {
+      my_acm = acm_a_.get();
+      break;
+    }
+    case 'B': {
+      my_acm = acm_b_.get();
+      break;
+    }
+    default:
+      break;
+  }
+  ASSERT_TRUE(my_acm != NULL);
+
+  auto encoder_factory = CreateBuiltinAudioEncoderFactory();
+  const int clockrate_hz = absl::EqualsIgnoreCase(codec_name, "g722")
+                               ? sampling_freq_hz / 2
+                               : sampling_freq_hz;
+  const std::string ptime = rtc::ToString(rtc::CheckedDivExact(
+      pack_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
+  SdpAudioFormat::Parameters params = {{"ptime", ptime}};
+  RTC_CHECK(channels == 1 || channels == 2);
+  if (absl::EqualsIgnoreCase(codec_name, "opus")) {
+    if (channels == 2) {
+      params["stereo"] = "1";
+    }
+    channels = 2;
+    params["maxaveragebitrate"] = rtc::ToString(rate);
+  }
+  constexpr int payload_type = 17;
+  auto encoder = encoder_factory->MakeAudioEncoder(
+      payload_type, SdpAudioFormat(codec_name, clockrate_hz, channels, params),
+      absl::nullopt);
+  EXPECT_NE(nullptr, encoder);
+  my_acm->SetEncoder(std::move(encoder));
+
+  send_codec_name_ = codec_name;
+}
+
+void TestStereo::Run(TestPackStereo* channel,
+                     int in_channels,
+                     int out_channels,
+                     int percent_loss) {
+  AudioFrame audio_frame;
+
+  int32_t out_freq_hz_b = out_file_.SamplingFrequency();
+  uint16_t rec_size;
+  uint32_t time_stamp_diff;
+  channel->reset_payload_size();
+  int error_count = 0;
+  int variable_bytes = 0;
+  int variable_packets = 0;
+  // Set test length to 500 ms (50 blocks of 10 ms each).
+  in_file_mono_->SetNum10MsBlocksToRead(50);
+  in_file_stereo_->SetNum10MsBlocksToRead(50);
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  in_file_stereo_->FastForward(100);
+  in_file_mono_->FastForward(100);
+
+  while (true) {
+    // Simulate packet loss by setting `packet_loss_` to "true" in
+    // `percent_loss` percent of the loops.
+    if (percent_loss > 0) {
+      if (counter_ == floor((100 / percent_loss) + 0.5)) {
+        counter_ = 0;
+        channel->set_lost_packet(true);
+      } else {
+        channel->set_lost_packet(false);
+      }
+      counter_++;
+    }
+
+    // Add 10 msec to ACM
+    if (in_channels == 1) {
+      if (in_file_mono_->EndOfFile()) {
+        break;
+      }
+      in_file_mono_->Read10MsData(audio_frame);
+    } else {
+      if (in_file_stereo_->EndOfFile()) {
+        break;
+      }
+      in_file_stereo_->Read10MsData(audio_frame);
+    }
+    EXPECT_GE(acm_a_->Add10MsData(audio_frame), 0);
+
+    // Verify that the received packet size matches the settings.
+    rec_size = channel->payload_size();
+    if ((0 < rec_size) & (rec_size < 65535)) {
+      if (strcmp(send_codec_name_, "opus") == 0) {
+        // Opus is a variable rate codec, hence calculate the average packet
+        // size, and later make sure the average is in the right range.
+        variable_bytes += rec_size;
+        variable_packets++;
+      } else {
+        // For fixed rate codecs, check that packet size is correct.
+        if ((rec_size != pack_size_bytes_ * out_channels) &&
+            (pack_size_bytes_ < 65535)) {
+          error_count++;
+        }
+      }
+      // Verify that the timestamp is updated with expected length
+      time_stamp_diff = channel->timestamp_diff();
+      if ((counter_ > 10) && (time_stamp_diff != pack_size_samp_)) {
+        error_count++;
+      }
+    }
+
+    // Run receive side of ACM
+    bool muted;
+    EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
+
+    // Write output speech to file
+    out_file_.Write10MsData(
+        audio_frame.data(),
+        audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+  }
+
+  EXPECT_EQ(0, error_count);
+
+  // Check that packet size is in the right range for variable rate codecs,
+  // such as Opus.
+  if (variable_packets > 0) {
+    variable_bytes /= variable_packets;
+    EXPECT_NEAR(variable_bytes, pack_size_bytes_, 18);
+  }
+
+  if (in_file_mono_->EndOfFile()) {
+    in_file_mono_->Rewind();
+  }
+  if (in_file_stereo_->EndOfFile()) {
+    in_file_stereo_->Rewind();
+  }
+  // Reset in case we ended with a lost packet
+  channel->set_lost_packet(false);
+}
+
+void TestStereo::OpenOutFile(int16_t test_number) {
+  std::string file_name;
+  rtc::StringBuilder file_stream;
+  file_stream << webrtc::test::OutputPath() << "teststereo_out_" << test_number
+              << ".pcm";
+  file_name = file_stream.str();
+  out_file_.Open(file_name, 32000, "wb");
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestStereo.h b/third_party/libwebrtc/modules/audio_coding/test/TestStereo.h
new file mode 100644
index 0000000000..4c50a4b555
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestStereo.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+#define MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+
+#include <math.h>
+
+#include <memory>
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/PCMFile.h"
+
+#define PCMA_AND_PCMU
+
+namespace webrtc {
+
+enum StereoMonoMode { kNotSet, kMono, kStereo };
+
+class TestPackStereo : public AudioPacketizationCallback {
+ public:
+  TestPackStereo();
+  ~TestPackStereo();
+
+  void RegisterReceiverACM(AudioCodingModule* acm);
+
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_size,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  uint16_t payload_size();
+  uint32_t timestamp_diff();
+  void reset_payload_size();
+  void set_codec_mode(StereoMonoMode mode);
+  void set_lost_packet(bool lost);
+
+ private:
+  AudioCodingModule* receiver_acm_;
+  int16_t seq_no_;
+  uint32_t timestamp_diff_;
+  uint32_t last_in_timestamp_;
+  uint64_t total_bytes_;
+  int payload_size_;
+  StereoMonoMode codec_mode_;
+  // Simulate packet losses
+  bool lost_packet_;
+};
+
+class TestStereo {
+ public:
+  TestStereo();
+  ~TestStereo();
+
+  void Perform();
+
+ private:
+  // The default value of '-1' indicates that the registration is based only on
+  // codec name and a sampling frequncy matching is not required. This is useful
+  // for codecs which support several sampling frequency.
+  void RegisterSendCodec(char side,
+                         char* codec_name,
+                         int32_t samp_freq_hz,
+                         int rate,
+                         int pack_size,
+                         int channels);
+
+  void Run(TestPackStereo* channel,
+           int in_channels,
+           int out_channels,
+           int percent_loss = 0);
+  void OpenOutFile(int16_t test_number);
+
+  std::unique_ptr<AudioCodingModule> acm_a_;
+  std::unique_ptr<AudioCodingModule> acm_b_;
+
+  TestPackStereo* channel_a2b_;
+
+  PCMFile* in_file_stereo_;
+  PCMFile* in_file_mono_;
+  PCMFile out_file_;
+  int16_t test_cntr_;
+  uint16_t pack_size_samp_;
+  uint16_t pack_size_bytes_;
+  int counter_;
+  char* send_codec_name_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestVADDTX.cc b/third_party/libwebrtc/modules/audio_coding/test/TestVADDTX.cc
new file mode 100644
index 0000000000..de26cafb68
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestVADDTX.cc
@@ -0,0 +1,240 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/test/TestVADDTX.h"
+
+#include <string>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_decoder_factory_template.h"
+#include "api/audio_codecs/audio_encoder_factory_template.h"
+#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+MonitoringAudioPacketizationCallback::MonitoringAudioPacketizationCallback(
+    AudioPacketizationCallback* next)
+    : next_(next) {
+  ResetStatistics();
+}
+
+int32_t MonitoringAudioPacketizationCallback::SendData(
+    AudioFrameType frame_type,
+    uint8_t payload_type,
+    uint32_t timestamp,
+    const uint8_t* payload_data,
+    size_t payload_len_bytes,
+    int64_t absolute_capture_timestamp_ms) {
+  counter_[static_cast<int>(frame_type)]++;
+  return next_->SendData(frame_type, payload_type, timestamp, payload_data,
+                         payload_len_bytes, absolute_capture_timestamp_ms);
+}
+
+void MonitoringAudioPacketizationCallback::PrintStatistics() {
+  printf("\n");
+  printf("kEmptyFrame       %u\n",
+         counter_[static_cast<int>(AudioFrameType::kEmptyFrame)]);
+  printf("kAudioFrameSpeech %u\n",
+         counter_[static_cast<int>(AudioFrameType::kAudioFrameSpeech)]);
+  printf("kAudioFrameCN     %u\n",
+         counter_[static_cast<int>(AudioFrameType::kAudioFrameCN)]);
+  printf("\n\n");
+}
+
+void MonitoringAudioPacketizationCallback::ResetStatistics() {
+  memset(counter_, 0, sizeof(counter_));
+}
+
+void MonitoringAudioPacketizationCallback::GetStatistics(uint32_t* counter) {
+  memcpy(counter, counter_, sizeof(counter_));
+}
+
+TestVadDtx::TestVadDtx()
+    : encoder_factory_(
+          CreateAudioEncoderFactory<AudioEncoderIlbc, AudioEncoderOpus>()),
+      decoder_factory_(
+          CreateAudioDecoderFactory<AudioDecoderIlbc, AudioDecoderOpus>()),
+      acm_send_(AudioCodingModule::Create(
+          AudioCodingModule::Config(decoder_factory_))),
+      acm_receive_(AudioCodingModule::Create(
+          AudioCodingModule::Config(decoder_factory_))),
+      channel_(std::make_unique<Channel>()),
+      packetization_callback_(
+          std::make_unique<MonitoringAudioPacketizationCallback>(
+              channel_.get())) {
+  EXPECT_EQ(
+      0, acm_send_->RegisterTransportCallback(packetization_callback_.get()));
+  channel_->RegisterReceiverACM(acm_receive_.get());
+}
+
+bool TestVadDtx::RegisterCodec(const SdpAudioFormat& codec_format,
+                               absl::optional<Vad::Aggressiveness> vad_mode) {
+  constexpr int payload_type = 17, cn_payload_type = 117;
+  bool added_comfort_noise = false;
+
+  auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
+                                                    absl::nullopt);
+  if (vad_mode.has_value() &&
+      !absl::EqualsIgnoreCase(codec_format.name, "opus")) {
+    AudioEncoderCngConfig config;
+    config.speech_encoder = std::move(encoder);
+    config.num_channels = 1;
+    config.payload_type = cn_payload_type;
+    config.vad_mode = vad_mode.value();
+    encoder = CreateComfortNoiseEncoder(std::move(config));
+    added_comfort_noise = true;
+  }
+  channel_->SetIsStereo(encoder->NumChannels() > 1);
+  acm_send_->SetEncoder(std::move(encoder));
+
+  std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}};
+  acm_receive_->SetReceiveCodecs(receive_codecs);
+
+  return added_comfort_noise;
+}
+
+// Encoding a file and see if the numbers that various packets occur follow
+// the expectation.
+void TestVadDtx::Run(absl::string_view in_filename,
+                     int frequency,
+                     int channels,
+                     absl::string_view out_filename,
+                     bool append,
+                     const int* expects) {
+  packetization_callback_->ResetStatistics();
+
+  PCMFile in_file;
+  in_file.Open(in_filename, frequency, "rb");
+  in_file.ReadStereo(channels > 1);
+  // Set test length to 1000 ms (100 blocks of 10 ms each).
+  in_file.SetNum10MsBlocksToRead(100);
+  // Fast-forward both files 500 ms (50 blocks). The first second of the file is
+  // silence, but we want to keep half of that to test silence periods.
+  in_file.FastForward(50);
+
+  PCMFile out_file;
+  if (append) {
+    out_file.Open(out_filename, kOutputFreqHz, "ab");
+  } else {
+    out_file.Open(out_filename, kOutputFreqHz, "wb");
+  }
+
+  uint16_t frame_size_samples = in_file.PayloadLength10Ms();
+  AudioFrame audio_frame;
+  while (!in_file.EndOfFile()) {
+    in_file.Read10MsData(audio_frame);
+    audio_frame.timestamp_ = time_stamp_;
+    time_stamp_ += frame_size_samples;
+    EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
+    bool muted;
+    acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame, &muted);
+    ASSERT_FALSE(muted);
+    out_file.Write10MsData(audio_frame);
+  }
+
+  in_file.Close();
+  out_file.Close();
+
+#ifdef PRINT_STAT
+  packetization_callback_->PrintStatistics();
+#endif
+
+  uint32_t stats[3];
+  packetization_callback_->GetStatistics(stats);
+  packetization_callback_->ResetStatistics();
+
+  for (const auto& st : stats) {
+    int i = &st - stats;  // Calculate the current position in stats.
+    switch (expects[i]) {
+      case 0: {
+        EXPECT_EQ(0u, st) << "stats[" << i << "] error.";
+        break;
+      }
+      case 1: {
+        EXPECT_GT(st, 0u) << "stats[" << i << "] error.";
+        break;
+      }
+    }
+  }
+}
+
+// Following is the implementation of TestWebRtcVadDtx.
+TestWebRtcVadDtx::TestWebRtcVadDtx() : output_file_num_(0) {}
+
+void TestWebRtcVadDtx::Perform() {
+  RunTestCases({"ILBC", 8000, 1});
+  RunTestCases({"opus", 48000, 2});
+}
+
+// Test various configurations on VAD/DTX.
+void TestWebRtcVadDtx::RunTestCases(const SdpAudioFormat& codec_format) {
+  Test(/*new_outfile=*/true,
+       /*expect_dtx_enabled=*/RegisterCodec(codec_format, absl::nullopt));
+
+  Test(/*new_outfile=*/false,
+       /*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadAggressive));
+
+  Test(/*new_outfile=*/false,
+       /*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadLowBitrate));
+
+  Test(/*new_outfile=*/false, /*expect_dtx_enabled=*/RegisterCodec(
+           codec_format, Vad::kVadVeryAggressive));
+
+  Test(/*new_outfile=*/false,
+       /*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadNormal));
+}
+
+// Set the expectation and run the test.
+void TestWebRtcVadDtx::Test(bool new_outfile, bool expect_dtx_enabled) {
+  int expects[] = {-1, 1, expect_dtx_enabled, 0, 0};
+  if (new_outfile) {
+    output_file_num_++;
+  }
+  rtc::StringBuilder out_filename;
+  out_filename << webrtc::test::OutputPath() << "testWebRtcVadDtx_outFile_"
+               << output_file_num_ << ".pcm";
+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
+      out_filename.str(), !new_outfile, expects);
+}
+
+// Following is the implementation of TestOpusDtx.
+void TestOpusDtx::Perform() {
+  int expects[] = {0, 1, 0, 0, 0};
+
+  // Register Opus as send codec
+  std::string out_filename =
+      webrtc::test::OutputPath() + "testOpusDtx_outFile_mono.pcm";
+  RegisterCodec({"opus", 48000, 2}, absl::nullopt);
+  acm_send_->ModifyEncoder([](std::unique_ptr<AudioEncoder>* encoder_ptr) {
+    (*encoder_ptr)->SetDtx(false);
+  });
+
+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
+      out_filename, false, expects);
+
+  acm_send_->ModifyEncoder([](std::unique_ptr<AudioEncoder>* encoder_ptr) {
+    (*encoder_ptr)->SetDtx(true);
+  });
+  expects[static_cast<int>(AudioFrameType::kEmptyFrame)] = 1;
+  expects[static_cast<int>(AudioFrameType::kAudioFrameCN)] = 1;
+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
+      out_filename, true, expects);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestVADDTX.h b/third_party/libwebrtc/modules/audio_coding/test/TestVADDTX.h
new file mode 100644
index 0000000000..d81ae28beb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestVADDTX.h
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+#define MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "common_audio/vad/include/vad.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/Channel.h"
+
+namespace webrtc {
+
+// This class records the frame type, and delegates actual sending to the
+// `next_` AudioPacketizationCallback.
+class MonitoringAudioPacketizationCallback : public AudioPacketizationCallback {
+ public:
+  explicit MonitoringAudioPacketizationCallback(
+      AudioPacketizationCallback* next);
+
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_len_bytes,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  void PrintStatistics();
+  void ResetStatistics();
+  void GetStatistics(uint32_t* stats);
+
+ private:
+  // 0 - kEmptyFrame
+  // 1 - kAudioFrameSpeech
+  // 2 - kAudioFrameCN
+  uint32_t counter_[3];
+  AudioPacketizationCallback* const next_;
+};
+
+// TestVadDtx is to verify that VAD/DTX perform as they should. It runs through
+// an audio file and check if the occurrence of various packet types follows
+// expectation. TestVadDtx needs its derived class to implement the Perform()
+// to put the test together.
+class TestVadDtx {
+ public:
+  static const int kOutputFreqHz = 16000;
+
+  TestVadDtx();
+
+ protected:
+  // Returns true iff CN was added.
+  bool RegisterCodec(const SdpAudioFormat& codec_format,
+                     absl::optional<Vad::Aggressiveness> vad_mode);
+
+  // Encoding a file and see if the numbers that various packets occur follow
+  // the expectation. Saves result to a file.
+  // expects[x] means
+  // -1 : do not care,
+  // 0  : there have been no packets of type `x`,
+  // 1  : there have been packets of type `x`,
+  // with `x` indicates the following packet types
+  // 0 - kEmptyFrame
+  // 1 - kAudioFrameSpeech
+  // 2 - kAudioFrameCN
+  void Run(absl::string_view in_filename,
+           int frequency,
+           int channels,
+           absl::string_view out_filename,
+           bool append,
+           const int* expects);
+
+  const rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+  const rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  std::unique_ptr<AudioCodingModule> acm_send_;
+  std::unique_ptr<AudioCodingModule> acm_receive_;
+  std::unique_ptr<Channel> channel_;
+  std::unique_ptr<MonitoringAudioPacketizationCallback> packetization_callback_;
+  uint32_t time_stamp_ = 0x12345678;
+};
+
+// TestWebRtcVadDtx is to verify that the WebRTC VAD/DTX perform as they should.
+class TestWebRtcVadDtx final : public TestVadDtx {
+ public:
+  TestWebRtcVadDtx();
+
+  void Perform();
+
+ private:
+  void RunTestCases(const SdpAudioFormat& codec_format);
+  void Test(bool new_outfile, bool expect_dtx_enabled);
+
+  int output_file_num_;
+};
+
+// TestOpusDtx is to verify that the Opus DTX performs as it should.
+class TestOpusDtx final : public TestVadDtx {
+ public:
+  void Perform();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/Tester.cc b/third_party/libwebrtc/modules/audio_coding/test/Tester.cc
new file mode 100644
index 0000000000..7612aa43a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/Tester.cc
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <string>
+#include <vector>
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/EncodeDecodeTest.h"
+#include "modules/audio_coding/test/PacketLossTest.h"
+#include "modules/audio_coding/test/TestAllCodecs.h"
+#include "modules/audio_coding/test/TestRedFec.h"
+#include "modules/audio_coding/test/TestStereo.h"
+#include "modules/audio_coding/test/TestVADDTX.h"
+#include "modules/audio_coding/test/TwoWayCommunication.h"
+#include "modules/audio_coding/test/opus_test.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+TEST(AudioCodingModuleTest, TestAllCodecs) {
+  webrtc::TestAllCodecs().Perform();
+}
+
+#if defined(WEBRTC_ANDROID)
+TEST(AudioCodingModuleTest, DISABLED_TestEncodeDecode) {
+#else
+TEST(AudioCodingModuleTest, TestEncodeDecode) {
+#endif
+  webrtc::EncodeDecodeTest().Perform();
+}
+
+TEST(AudioCodingModuleTest, TestRedFec) {
+  webrtc::TestRedFec().Perform();
+}
+
+// Disabled on ios as flaky, see https://crbug.com/webrtc/7057
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+TEST(AudioCodingModuleTest, DISABLED_TestStereo) {
+#else
+TEST(AudioCodingModuleTest, TestStereo) {
+#endif
+  webrtc::TestStereo().Perform();
+}
+
+TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
+  webrtc::TestWebRtcVadDtx().Perform();
+}
+
+TEST(AudioCodingModuleTest, TestOpusDtx) {
+  webrtc::TestOpusDtx().Perform();
+}
+
+// Disabled on ios as flaky, see https://crbug.com/webrtc/7057
+#if defined(WEBRTC_IOS)
+TEST(AudioCodingModuleTest, DISABLED_TestOpus) {
+#else
+TEST(AudioCodingModuleTest, TestOpus) {
+#endif
+  webrtc::OpusTest().Perform();
+}
+
+TEST(AudioCodingModuleTest, TestPacketLoss) {
+  webrtc::PacketLossTest(1, 10, 10, 1).Perform();
+}
+
+TEST(AudioCodingModuleTest, TestPacketLossBurst) {
+  webrtc::PacketLossTest(1, 10, 10, 2).Perform();
+}
+
+// Disabled on ios as flake, see https://crbug.com/webrtc/7057
+#if defined(WEBRTC_IOS)
+TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereo) {
+#else
+TEST(AudioCodingModuleTest, TestPacketLossStereo) {
+#endif
+  webrtc::PacketLossTest(2, 10, 10, 1).Perform();
+}
+
+// Disabled on ios as flake, see https://crbug.com/webrtc/7057
+#if defined(WEBRTC_IOS)
+TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereoBurst) {
+#else
+TEST(AudioCodingModuleTest, TestPacketLossStereoBurst) {
+#endif
+  webrtc::PacketLossTest(2, 10, 10, 2).Perform();
+}
+
+// The full API test is too long to run automatically on bots, but can be used
+// for offline testing. User interaction is needed.
+#ifdef ACM_TEST_FULL_API
+TEST(AudioCodingModuleTest, TestAPI) {
+  webrtc::APITest().Perform();
+}
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.cc b/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.cc
new file mode 100644
index 0000000000..b42415a21a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.cc
@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "TwoWayCommunication.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+
+TwoWayCommunication::TwoWayCommunication()
+    : _acmA(AudioCodingModule::Create(
+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+      _acmRefA(AudioCodingModule::Create(
+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {
+  AudioCodingModule::Config config;
+  // The clicks will be more obvious if time-stretching is not allowed.
+  // TODO(henrik.lundin) Really?
+  config.neteq_config.for_test_no_time_stretching = true;
+  config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+  _acmB.reset(AudioCodingModule::Create(config));
+  _acmRefB.reset(AudioCodingModule::Create(config));
+}
+
+TwoWayCommunication::~TwoWayCommunication() {
+  delete _channel_A2B;
+  delete _channel_B2A;
+  delete _channelRef_A2B;
+  delete _channelRef_B2A;
+  _inFileA.Close();
+  _inFileB.Close();
+  _outFileA.Close();
+  _outFileB.Close();
+  _outFileRefA.Close();
+  _outFileRefB.Close();
+}
+
+void TwoWayCommunication::SetUpAutotest(
+    AudioEncoderFactory* const encoder_factory,
+    const SdpAudioFormat& format1,
+    const int payload_type1,
+    const SdpAudioFormat& format2,
+    const int payload_type2) {
+  //--- Set A codecs
+  _acmA->SetEncoder(
+      encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
+  _acmA->SetReceiveCodecs({{payload_type2, format2}});
+
+  //--- Set ref-A codecs
+  _acmRefA->SetEncoder(
+      encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
+  _acmRefA->SetReceiveCodecs({{payload_type2, format2}});
+
+  //--- Set B codecs
+  _acmB->SetEncoder(
+      encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
+  _acmB->SetReceiveCodecs({{payload_type1, format1}});
+
+  //--- Set ref-B codecs
+  _acmRefB->SetEncoder(
+      encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
+  _acmRefB->SetReceiveCodecs({{payload_type1, format1}});
+
+  uint16_t frequencyHz;
+
+  //--- Input A and B
+  std::string in_file_name =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  frequencyHz = 16000;
+  _inFileA.Open(in_file_name, frequencyHz, "rb");
+  _inFileB.Open(in_file_name, frequencyHz, "rb");
+
+  //--- Output A
+  std::string output_file_a = webrtc::test::OutputPath() + "outAutotestA.pcm";
+  frequencyHz = 16000;
+  _outFileA.Open(output_file_a, frequencyHz, "wb");
+  std::string output_ref_file_a =
+      webrtc::test::OutputPath() + "ref_outAutotestA.pcm";
+  _outFileRefA.Open(output_ref_file_a, frequencyHz, "wb");
+
+  //--- Output B
+  std::string output_file_b = webrtc::test::OutputPath() + "outAutotestB.pcm";
+  frequencyHz = 16000;
+  _outFileB.Open(output_file_b, frequencyHz, "wb");
+  std::string output_ref_file_b =
+      webrtc::test::OutputPath() + "ref_outAutotestB.pcm";
+  _outFileRefB.Open(output_ref_file_b, frequencyHz, "wb");
+
+  //--- Set A-to-B channel
+  _channel_A2B = new Channel;
+  _acmA->RegisterTransportCallback(_channel_A2B);
+  _channel_A2B->RegisterReceiverACM(_acmB.get());
+  //--- Do the same for the reference
+  _channelRef_A2B = new Channel;
+  _acmRefA->RegisterTransportCallback(_channelRef_A2B);
+  _channelRef_A2B->RegisterReceiverACM(_acmRefB.get());
+
+  //--- Set B-to-A channel
+  _channel_B2A = new Channel;
+  _acmB->RegisterTransportCallback(_channel_B2A);
+  _channel_B2A->RegisterReceiverACM(_acmA.get());
+  //--- Do the same for reference
+  _channelRef_B2A = new Channel;
+  _acmRefB->RegisterTransportCallback(_channelRef_B2A);
+  _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
+}
+
+void TwoWayCommunication::Perform() {
+  const SdpAudioFormat format1("ISAC", 16000, 1);
+  const SdpAudioFormat format2("L16", 8000, 1);
+  constexpr int payload_type1 = 17, payload_type2 = 18;
+
+  auto encoder_factory = CreateBuiltinAudioEncoderFactory();
+
+  SetUpAutotest(encoder_factory.get(), format1, payload_type1, format2,
+                payload_type2);
+
+  unsigned int msecPassed = 0;
+  unsigned int secPassed = 0;
+
+  int32_t outFreqHzA = _outFileA.SamplingFrequency();
+  int32_t outFreqHzB = _outFileB.SamplingFrequency();
+
+  AudioFrame audioFrame;
+
+  // In the following loop we tests that the code can handle misuse of the APIs.
+  // In the middle of a session with data flowing between two sides, called A
+  // and B, APIs will be called, and the code should continue to run, and be
+  // able to recover.
+  while (!_inFileA.EndOfFile() && !_inFileB.EndOfFile()) {
+    msecPassed += 10;
+    EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
+    EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
+    EXPECT_GE(_acmRefA->Add10MsData(audioFrame), 0);
+
+    EXPECT_GT(_inFileB.Read10MsData(audioFrame), 0);
+
+    EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
+    EXPECT_GE(_acmRefB->Add10MsData(audioFrame), 0);
+    bool muted;
+    EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
+    _outFileA.Write10MsData(audioFrame);
+    EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
+    _outFileRefA.Write10MsData(audioFrame);
+    EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
+    _outFileB.Write10MsData(audioFrame);
+    EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
+    ASSERT_FALSE(muted);
+    _outFileRefB.Write10MsData(audioFrame);
+
+    // Update time counters each time a second of data has passed.
+    if (msecPassed >= 1000) {
+      msecPassed = 0;
+      secPassed++;
+    }
+    // Re-register send codec on side B.
+    if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
+      _acmB->SetEncoder(encoder_factory->MakeAudioEncoder(
+          payload_type2, format2, absl::nullopt));
+    }
+    // Initialize receiver on side A.
+    if (((secPassed % 7) == 6) && (msecPassed == 0))
+      EXPECT_EQ(0, _acmA->InitializeReceiver());
+    // Re-register codec on side A.
+    if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
+      _acmA->SetReceiveCodecs({{payload_type2, format2}});
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.h b/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.h
new file mode 100644
index 0000000000..b7eb9e5583
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/TwoWayCommunication.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+#define MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+
+#include <memory>
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+
+namespace webrtc {
+
+class TwoWayCommunication {
+ public:
+  TwoWayCommunication();
+  ~TwoWayCommunication();
+
+  void Perform();
+
+ private:
+  void SetUpAutotest(AudioEncoderFactory* const encoder_factory,
+                     const SdpAudioFormat& format1,
+                     int payload_type1,
+                     const SdpAudioFormat& format2,
+                     int payload_type2);
+
+  std::unique_ptr<AudioCodingModule> _acmA;
+  std::unique_ptr<AudioCodingModule> _acmB;
+
+  std::unique_ptr<AudioCodingModule> _acmRefA;
+  std::unique_ptr<AudioCodingModule> _acmRefB;
+
+  Channel* _channel_A2B;
+  Channel* _channel_B2A;
+
+  Channel* _channelRef_A2B;
+  Channel* _channelRef_B2A;
+
+  PCMFile _inFileA;
+  PCMFile _inFileB;
+
+  PCMFile _outFileA;
+  PCMFile _outFileB;
+
+  PCMFile _outFileRefA;
+  PCMFile _outFileRefB;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/opus_test.cc b/third_party/libwebrtc/modules/audio_coding/test/opus_test.cc
new file mode 100644
index 0000000000..6822bc3d72
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/opus_test.cc
@@ -0,0 +1,402 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/test/opus_test.h"
+
+#include <string>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/TestStereo.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+OpusTest::OpusTest()
+    : acm_receiver_(AudioCodingModule::Create(
+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+      channel_a2b_(NULL),
+      counter_(0),
+      payload_type_(255),
+      rtp_timestamp_(0) {}
+
+OpusTest::~OpusTest() {
+  if (channel_a2b_ != NULL) {
+    delete channel_a2b_;
+    channel_a2b_ = NULL;
+  }
+  if (opus_mono_encoder_ != NULL) {
+    WebRtcOpus_EncoderFree(opus_mono_encoder_);
+    opus_mono_encoder_ = NULL;
+  }
+  if (opus_stereo_encoder_ != NULL) {
+    WebRtcOpus_EncoderFree(opus_stereo_encoder_);
+    opus_stereo_encoder_ = NULL;
+  }
+  if (opus_mono_decoder_ != NULL) {
+    WebRtcOpus_DecoderFree(opus_mono_decoder_);
+    opus_mono_decoder_ = NULL;
+  }
+  if (opus_stereo_decoder_ != NULL) {
+    WebRtcOpus_DecoderFree(opus_stereo_decoder_);
+    opus_stereo_decoder_ = NULL;
+  }
+}
+
+void OpusTest::Perform() {
+#ifndef WEBRTC_CODEC_OPUS
+  // Opus isn't defined, exit.
+  return;
+#else
+  uint16_t frequency_hz;
+  size_t audio_channels;
+  int16_t test_cntr = 0;
+
+  // Open both mono and stereo test files in 32 kHz.
+  const std::string file_name_stereo =
+      webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
+  const std::string file_name_mono =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  frequency_hz = 32000;
+  in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb");
+  in_file_stereo_.ReadStereo(true);
+  in_file_mono_.Open(file_name_mono, frequency_hz, "rb");
+  in_file_mono_.ReadStereo(false);
+
+  // Create Opus encoders for mono and stereo.
+  ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1, 0, 48000), -1);
+  ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2, 1, 48000), -1);
+
+  // Create Opus decoders for mono and stereo for stand-alone testing of Opus.
+  ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1, 48000), -1);
+  ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2, 48000), -1);
+  WebRtcOpus_DecoderInit(opus_mono_decoder_);
+  WebRtcOpus_DecoderInit(opus_stereo_decoder_);
+
+  ASSERT_TRUE(acm_receiver_.get() != NULL);
+  EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
+
+  // Register Opus stereo as receiving codec.
+  constexpr int kOpusPayloadType = 120;
+  const SdpAudioFormat kOpusFormatStereo("opus", 48000, 2, {{"stereo", "1"}});
+  payload_type_ = kOpusPayloadType;
+  acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatStereo}});
+
+  // Create and connect the channel.
+  channel_a2b_ = new TestPackStereo;
+  channel_a2b_->RegisterReceiverACM(acm_receiver_.get());
+
+  //
+  // Test Stereo.
+  //
+
+  channel_a2b_->set_codec_mode(kStereo);
+  audio_channels = 2;
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Run Opus with 2.5 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 120);
+
+  // Run Opus with 5 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 240);
+
+  // Run Opus with 10 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 480);
+
+  // Run Opus with 20 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 960);
+
+  // Run Opus with 40 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 1920);
+
+  // Run Opus with 60 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 2880);
+
+  out_file_.Close();
+  out_file_standalone_.Close();
+
+  //
+  // Test Opus stereo with packet-losses.
+  //
+
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Run Opus with 20 ms frame size, 1% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 1);
+
+  // Run Opus with 20 ms frame size, 5% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 5);
+
+  // Run Opus with 20 ms frame size, 10% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 10);
+
+  out_file_.Close();
+  out_file_standalone_.Close();
+
+  //
+  // Test Mono.
+  //
+  channel_a2b_->set_codec_mode(kMono);
+  audio_channels = 1;
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Register Opus mono as receiving codec.
+  const SdpAudioFormat kOpusFormatMono("opus", 48000, 2);
+  acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatMono}});
+
+  // Run Opus with 2.5 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 120);
+
+  // Run Opus with 5 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 240);
+
+  // Run Opus with 10 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 480);
+
+  // Run Opus with 20 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 960);
+
+  // Run Opus with 40 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 1920);
+
+  // Run Opus with 60 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 2880);
+
+  out_file_.Close();
+  out_file_standalone_.Close();
+
+  //
+  // Test Opus mono with packet-losses.
+  //
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Run Opus with 20 ms frame size, 1% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 1);
+
+  // Run Opus with 20 ms frame size, 5% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 5);
+
+  // Run Opus with 20 ms frame size, 10% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 10);
+
+  // Close the files.
+  in_file_stereo_.Close();
+  in_file_mono_.Close();
+  out_file_.Close();
+  out_file_standalone_.Close();
+#endif
+}
+
+void OpusTest::Run(TestPackStereo* channel,
+                   size_t channels,
+                   int bitrate,
+                   size_t frame_length,
+                   int percent_loss) {
+  AudioFrame audio_frame;
+  int32_t out_freq_hz_b = out_file_.SamplingFrequency();
+  const size_t kBufferSizeSamples = 480 * 12 * 2;  // 120 ms stereo audio.
+  int16_t audio[kBufferSizeSamples];
+  int16_t out_audio[kBufferSizeSamples];
+  int16_t audio_type;
+  size_t written_samples = 0;
+  size_t read_samples = 0;
+  size_t decoded_samples = 0;
+  bool first_packet = true;
+  uint32_t start_time_stamp = 0;
+
+  channel->reset_payload_size();
+  counter_ = 0;
+
+  // Set encoder rate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
+  // If we are on Android, iOS and/or ARM, use a lower complexity setting as
+  // default.
+  const int kOpusComplexity5 = 5;
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
+  EXPECT_EQ(0,
+            WebRtcOpus_SetComplexity(opus_stereo_encoder_, kOpusComplexity5));
+#endif
+
+  // Fast-forward 1 second (100 blocks) since the files start with silence.
+  in_file_stereo_.FastForward(100);
+  in_file_mono_.FastForward(100);
+
+  // Limit the runtime to 1000 blocks of 10 ms each.
+  for (size_t audio_length = 0; audio_length < 1000; audio_length += 10) {
+    bool lost_packet = false;
+
+    // Get 10 msec of audio.
+    if (channels == 1) {
+      if (in_file_mono_.EndOfFile()) {
+        break;
+      }
+      in_file_mono_.Read10MsData(audio_frame);
+    } else {
+      if (in_file_stereo_.EndOfFile()) {
+        break;
+      }
+      in_file_stereo_.Read10MsData(audio_frame);
+    }
+
+    // If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
+    EXPECT_EQ(480, resampler_.Resample10Msec(
+                       audio_frame.data(), audio_frame.sample_rate_hz_, 48000,
+                       channels, kBufferSizeSamples - written_samples,
+                       &audio[written_samples]));
+    written_samples += 480 * channels;
+
+    // Sometimes we need to loop over the audio vector to produce the right
+    // number of packets.
+    size_t loop_encode =
+        (written_samples - read_samples) / (channels * frame_length);
+
+    if (loop_encode > 0) {
+      const size_t kMaxBytes = 1000;  // Maximum number of bytes for one packet.
+      size_t bitstream_len_byte;
+      uint8_t bitstream[kMaxBytes];
+      for (size_t i = 0; i < loop_encode; i++) {
+        int bitstream_len_byte_int = WebRtcOpus_Encode(
+            (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
+            &audio[read_samples], frame_length, kMaxBytes, bitstream);
+        ASSERT_GE(bitstream_len_byte_int, 0);
+        bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
+
+        // Simulate packet loss by setting `packet_loss_` to "true" in
+        // `percent_loss` percent of the loops.
+        // TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
+        if (percent_loss > 0) {
+          if (counter_ == floor((100 / percent_loss) + 0.5)) {
+            counter_ = 0;
+            lost_packet = true;
+            channel->set_lost_packet(true);
+          } else {
+            lost_packet = false;
+            channel->set_lost_packet(false);
+          }
+          counter_++;
+        }
+
+        // Run stand-alone Opus decoder, or decode PLC.
+        if (channels == 1) {
+          if (!lost_packet) {
+            decoded_samples += WebRtcOpus_Decode(
+                opus_mono_decoder_, bitstream, bitstream_len_byte,
+                &out_audio[decoded_samples * channels], &audio_type);
+          } else {
+            // Call decoder PLC.
+            constexpr int kPlcDurationMs = 10;
+            constexpr int kPlcSamples = 48 * kPlcDurationMs;
+            size_t total_plc_samples = 0;
+            while (total_plc_samples < frame_length) {
+              int ret = WebRtcOpus_Decode(
+                  opus_mono_decoder_, NULL, 0,
+                  &out_audio[decoded_samples * channels], &audio_type);
+              EXPECT_EQ(ret, kPlcSamples);
+              decoded_samples += ret;
+              total_plc_samples += ret;
+            }
+            EXPECT_EQ(total_plc_samples, frame_length);
+          }
+        } else {
+          if (!lost_packet) {
+            decoded_samples += WebRtcOpus_Decode(
+                opus_stereo_decoder_, bitstream, bitstream_len_byte,
+                &out_audio[decoded_samples * channels], &audio_type);
+          } else {
+            // Call decoder PLC.
+            constexpr int kPlcDurationMs = 10;
+            constexpr int kPlcSamples = 48 * kPlcDurationMs;
+            size_t total_plc_samples = 0;
+            while (total_plc_samples < frame_length) {
+              int ret = WebRtcOpus_Decode(
+                  opus_stereo_decoder_, NULL, 0,
+                  &out_audio[decoded_samples * channels], &audio_type);
+              EXPECT_EQ(ret, kPlcSamples);
+              decoded_samples += ret;
+              total_plc_samples += ret;
+            }
+            EXPECT_EQ(total_plc_samples, frame_length);
+          }
+        }
+
+        // Send data to the channel. "channel" will handle the loss simulation.
+        channel->SendData(AudioFrameType::kAudioFrameSpeech, payload_type_,
+                          rtp_timestamp_, bitstream, bitstream_len_byte, 0);
+        if (first_packet) {
+          first_packet = false;
+          start_time_stamp = rtp_timestamp_;
+        }
+        rtp_timestamp_ += static_cast<uint32_t>(frame_length);
+        read_samples += frame_length * channels;
+      }
+      if (read_samples == written_samples) {
+        read_samples = 0;
+        written_samples = 0;
+      }
+    }
+
+    // Run received side of ACM.
+    bool muted;
+    ASSERT_EQ(
+        0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
+    ASSERT_FALSE(muted);
+
+    // Write output speech to file.
+    out_file_.Write10MsData(
+        audio_frame.data(),
+        audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+
+    // Write stand-alone speech to file.
+    out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
+
+    if (audio_frame.timestamp_ > start_time_stamp) {
+      // Number of channels should be the same for both stand-alone and
+      // ACM-decoding.
+      EXPECT_EQ(audio_frame.num_channels_, channels);
+    }
+
+    decoded_samples = 0;
+  }
+
+  if (in_file_mono_.EndOfFile()) {
+    in_file_mono_.Rewind();
+  }
+  if (in_file_stereo_.EndOfFile()) {
+    in_file_stereo_.Rewind();
+  }
+  // Reset in case we ended with a lost packet.
+  channel->set_lost_packet(false);
+}
+
+void OpusTest::OpenOutFile(int test_number) {
+  std::string file_name;
+  std::stringstream file_stream;
+  file_stream << webrtc::test::OutputPath() << "opustest_out_" << test_number
+              << ".pcm";
+  file_name = file_stream.str();
+  out_file_.Open(file_name, 48000, "wb");
+  file_stream.str("");
+  file_name = file_stream.str();
+  file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
+              << test_number << ".pcm";
+  file_name = file_stream.str();
+  out_file_standalone_.Open(file_name, 48000, "wb");
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/test/opus_test.h b/third_party/libwebrtc/modules/audio_coding/test/opus_test.h
new file mode 100644
index 0000000000..c69f922adb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/opus_test.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+#define MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+
+#include <math.h>
+
+#include <memory>
+
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/TestStereo.h"
+
+namespace webrtc {
+
+class OpusTest {
+ public:
+  OpusTest();
+  ~OpusTest();
+
+  void Perform();
+
+ private:
+  void Run(TestPackStereo* channel,
+           size_t channels,
+           int bitrate,
+           size_t frame_length,
+           int percent_loss = 0);
+
+  void OpenOutFile(int test_number);
+
+  std::unique_ptr<AudioCodingModule> acm_receiver_;
+  TestPackStereo* channel_a2b_;
+  PCMFile in_file_stereo_;
+  PCMFile in_file_mono_;
+  PCMFile out_file_;
+  PCMFile out_file_standalone_;
+  int counter_;
+  uint8_t payload_type_;
+  uint32_t rtp_timestamp_;
+  acm2::ACMResampler resampler_;
+  WebRtcOpusEncInst* opus_mono_encoder_;
+  WebRtcOpusEncInst* opus_stereo_encoder_;
+  WebRtcOpusDecInst* opus_mono_decoder_;
+  WebRtcOpusDecInst* opus_stereo_decoder_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/test/target_delay_unittest.cc b/third_party/libwebrtc/modules/audio_coding/test/target_delay_unittest.cc
new file mode 100644
index 0000000000..5eccdcf8eb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/test/target_delay_unittest.cc
@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+class TargetDelayTest : public ::testing::Test {
+ protected:
+  TargetDelayTest()
+      : receiver_(
+            AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())) {}
+
+  ~TargetDelayTest() {}
+
+  void SetUp() {
+    constexpr int pltype = 108;
+    std::map<int, SdpAudioFormat> receive_codecs = {
+        {pltype, {"L16", kSampleRateHz, 1}}};
+    receiver_.SetCodecs(receive_codecs);
+
+    rtp_header_.payloadType = pltype;
+    rtp_header_.timestamp = 0;
+    rtp_header_.ssrc = 0x12345678;
+    rtp_header_.markerBit = false;
+    rtp_header_.sequenceNumber = 0;
+
+    int16_t audio[kFrameSizeSamples];
+    const int kRange = 0x7FF;  // 2047, easy for masking.
+    for (size_t n = 0; n < kFrameSizeSamples; ++n)
+      audio[n] = (rand() & kRange) - kRange / 2;
+    WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_);
+  }
+
+  void OutOfRangeInput() {
+    EXPECT_EQ(-1, SetMinimumDelay(-1));
+    EXPECT_EQ(-1, SetMinimumDelay(10001));
+  }
+
+  void TargetDelayBufferMinMax() {
+    const int kTargetMinDelayMs = kNum10msPerFrame * 10;
+    ASSERT_EQ(0, SetMinimumDelay(kTargetMinDelayMs));
+    for (int m = 0; m < 30; ++m)  // Run enough iterations to fill the buffer.
+      Run(true);
+    int clean_optimal_delay = GetCurrentOptimalDelayMs();
+    EXPECT_EQ(kTargetMinDelayMs, clean_optimal_delay);
+
+    const int kTargetMaxDelayMs = 2 * (kNum10msPerFrame * 10);
+    ASSERT_EQ(0, SetMaximumDelay(kTargetMaxDelayMs));
+    for (int n = 0; n < 30; ++n)  // Run enough iterations to fill the buffer.
+      Run(false);
+
+    int capped_optimal_delay = GetCurrentOptimalDelayMs();
+    EXPECT_EQ(kTargetMaxDelayMs, capped_optimal_delay);
+  }
+
+ private:
+  static const int kSampleRateHz = 16000;
+  static const int kNum10msPerFrame = 2;
+  static const size_t kFrameSizeSamples = 320;  // 20 ms @ 16 kHz.
+  // payload-len = frame-samples * 2 bytes/sample.
+  static const int kPayloadLenBytes = 320 * 2;
+  // Inter-arrival time in number of packets in a jittery channel. One is no
+  // jitter.
+  static const int kInterarrivalJitterPacket = 2;
+
+  void Push() {
+    rtp_header_.timestamp += kFrameSizeSamples;
+    rtp_header_.sequenceNumber++;
+    ASSERT_EQ(0, receiver_.InsertPacket(rtp_header_,
+                                        rtc::ArrayView<const uint8_t>(
+                                            payload_, kFrameSizeSamples * 2)));
+  }
+
+  // Pull audio equivalent to the amount of audio in one RTP packet.
+  void Pull() {
+    AudioFrame frame;
+    bool muted;
+    for (int k = 0; k < kNum10msPerFrame; ++k) {  // Pull one frame.
+      ASSERT_EQ(0, receiver_.GetAudio(-1, &frame, &muted));
+      ASSERT_FALSE(muted);
+      // Had to use ASSERT_TRUE, ASSERT_EQ generated error.
+      ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
+      ASSERT_EQ(1u, frame.num_channels_);
+      ASSERT_TRUE(kSampleRateHz / 100 == frame.samples_per_channel_);
+    }
+  }
+
+  void Run(bool clean) {
+    for (int n = 0; n < 10; ++n) {
+      for (int m = 0; m < 5; ++m) {
+        Push();
+        Pull();
+      }
+
+      if (!clean) {
+        for (int m = 0; m < 10; ++m) {  // Long enough to trigger delay change.
+          Push();
+          for (int n = 0; n < kInterarrivalJitterPacket; ++n)
+            Pull();
+        }
+      }
+    }
+  }
+
+  int SetMinimumDelay(int delay_ms) {
+    return receiver_.SetMinimumDelay(delay_ms);
+  }
+
+  int SetMaximumDelay(int delay_ms) {
+    return receiver_.SetMaximumDelay(delay_ms);
+  }
+
+  int GetCurrentOptimalDelayMs() {
+    NetworkStatistics stats;
+    receiver_.GetNetworkStatistics(&stats);
+    return stats.preferredBufferSize;
+  }
+
+  acm2::AcmReceiver receiver_;
+  RTPHeader rtp_header_;
+  uint8_t payload_[kPayloadLenBytes];
+};
+
+// Flaky on iOS: webrtc:7057.
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_OutOfRangeInput DISABLED_OutOfRangeInput
+#else
+#define MAYBE_OutOfRangeInput OutOfRangeInput
+#endif
+TEST_F(TargetDelayTest, MAYBE_OutOfRangeInput) {
+  OutOfRangeInput();
+}
+
+// Flaky on iOS: webrtc:7057.
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_TargetDelayBufferMinMax DISABLED_TargetDelayBufferMinMax
+#else
+#define MAYBE_TargetDelayBufferMinMax TargetDelayBufferMinMax
+#endif
+TEST_F(TargetDelayTest, MAYBE_TargetDelayBufferMinMax) {
+  TargetDelayBufferMinMax();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/webrtc_cng_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/webrtc_cng_gn/moz.build
new file mode 100644
index 0000000000..4a9c24069e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/webrtc_cng_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_cng_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/webrtc_multiopus_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/webrtc_multiopus_gn/moz.build
new file mode 100644
index 0000000000..99ea47c13a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/webrtc_multiopus_gn/moz.build
@@ -0,0 +1,230 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_CODEC_ILBC"] = True
+DEFINES["WEBRTC_CODEC_OPUS"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_OPUS_SUPPORT_120MS_PTIME"] = "1"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/media/libopus/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_multiopus_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/webrtc_opus_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/webrtc_opus_gn/moz.build
new file mode 100644
index 0000000000..49fb552b0d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/webrtc_opus_gn/moz.build
@@ -0,0 +1,237 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_CODEC_ILBC"] = True
+DEFINES["WEBRTC_CODEC_OPUS"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_OPUS_SUPPORT_120MS_PTIME"] = "1"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/media/libopus/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc",
+    "/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_opus_gn")
diff --git a/third_party/libwebrtc/modules/audio_coding/webrtc_opus_wrapper_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/webrtc_opus_wrapper_gn/moz.build
new file mode 100644
index 0000000000..20e1d37306
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/webrtc_opus_wrapper_gn/moz.build
@@ -0,0 +1,229 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_CODEC_ILBC"] = True
+DEFINES["WEBRTC_CODEC_OPUS"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_OPUS_SUPPORT_120MS_PTIME"] = "1"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/media/libopus/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_opus_wrapper_gn")
diff --git a/third_party/libwebrtc/modules/audio_device/BUILD.gn b/third_party/libwebrtc/modules/audio_device/BUILD.gn
new file mode 100644
index 0000000000..61cd531edd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/BUILD.gn
@@ -0,0 +1,504 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+config("audio_device_warnings_config") {
+  if (is_win && is_clang) {
+    cflags = [
+      # Disable warnings failing when compiling with Clang on Windows.
+      # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+      "-Wno-microsoft-goto",
+    ]
+  }
+}
+
+rtc_source_set("audio_device_default") {
+  visibility = [ "*" ]
+  sources = [ "include/audio_device_default.h" ]
+  deps = [ ":audio_device_api" ]
+}
+
+rtc_source_set("audio_device") {
+if (!build_with_mozilla) { # See Bug 1820869.
+  visibility = [ "*" ]
+  public_deps = [
+    ":audio_device_api",
+
+    # Deprecated.
+    # TODO(webrtc:7452): Remove this public dep. audio_device_impl should
+    # be depended on directly if needed.
+    ":audio_device_impl",
+  ]
+}
+}
+
+rtc_source_set("audio_device_api") {
+  visibility = [ "*" ]
+  sources = [
+    "include/audio_device.h",
+    "include/audio_device_defines.h",
+  ]
+  deps = [
+    "../../api:scoped_refptr",
+    "../../api/task_queue",
+    "../../rtc_base:checks",
+    "../../rtc_base:refcount",
+    "../../rtc_base:stringutils",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_device_buffer") {
+if (!build_with_mozilla) { # See Bug 1820869.
+  sources = [
+    "audio_device_buffer.cc",
+    "audio_device_buffer.h",
+    "audio_device_config.h",
+    "fine_audio_buffer.cc",
+    "fine_audio_buffer.h",
+  ]
+  deps = [
+    ":audio_device_api",
+    "../../api:array_view",
+    "../../api:sequence_checker",
+    "../../api/task_queue",
+    "../../common_audio:common_audio_c",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rtc_task_queue",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:timestamp_aligner",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
+    "../../system_wrappers:metrics",
+  ]
+}
+}
+
+rtc_library("audio_device_generic") {
+  sources = [
+    "audio_device_generic.cc",
+    "audio_device_generic.h",
+  ]
+  deps = [
+    ":audio_device_api",
+    ":audio_device_buffer",
+    "../../rtc_base:logging",
+  ]
+}
+
+rtc_library("audio_device_name") {
+  sources = [
+    "audio_device_name.cc",
+    "audio_device_name.h",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_source_set("windows_core_audio_utility") {
+  if (is_win && !build_with_chromium) {
+    sources = [
+      "win/core_audio_utility_win.cc",
+      "win/core_audio_utility_win.h",
+    ]
+
+    deps = [
+      ":audio_device_api",
+      ":audio_device_name",
+      "../../api/units:time_delta",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:platform_thread_types",
+      "../../rtc_base:stringutils",
+      "../../rtc_base/win:windows_version",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+
+    libs = [ "oleaut32.lib" ]
+  }
+}
+
+# An ADM with a dedicated factory method which does not depend on the
+# audio_device_impl target. The goal is to use this new structure and
+# gradually phase out the old design.
+# TODO(henrika): currently only supported on Windows.
+rtc_source_set("audio_device_module_from_input_and_output") {
+  visibility = [ "*" ]
+  if (is_win && !build_with_chromium) {
+    sources = [
+      "include/audio_device_factory.cc",
+      "include/audio_device_factory.h",
+    ]
+    sources += [
+      "win/audio_device_module_win.cc",
+      "win/audio_device_module_win.h",
+      "win/core_audio_base_win.cc",
+      "win/core_audio_base_win.h",
+      "win/core_audio_input_win.cc",
+      "win/core_audio_input_win.h",
+      "win/core_audio_output_win.cc",
+      "win/core_audio_output_win.h",
+    ]
+
+    deps = [
+      ":audio_device_api",
+      ":audio_device_buffer",
+      ":windows_core_audio_utility",
+      "../../api:make_ref_counted",
+      "../../api:scoped_refptr",
+      "../../api:sequence_checker",
+      "../../api/task_queue",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:platform_thread",
+      "../../rtc_base:safe_conversions",
+      "../../rtc_base:stringutils",
+      "../../rtc_base:timeutils",
+      "../../rtc_base/win:scoped_com_initializer",
+      "../../rtc_base/win:windows_version",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings:strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+}
+
+# Contains default implementations of webrtc::AudioDeviceModule for Windows,
+# Linux, Mac, iOS and Android.
+rtc_library("audio_device_impl") {
+if (!build_with_mozilla) { # See Bug 1820869.
+  visibility = [ "*" ]
+  deps = [
+    ":audio_device_api",
+    ":audio_device_buffer",
+    ":audio_device_default",
+    ":audio_device_generic",
+    "../../api:array_view",
+    "../../api:make_ref_counted",
+    "../../api:refcountedbase",
+    "../../api:scoped_refptr",
+    "../../api:sequence_checker",
+    "../../api/task_queue",
+    "../../common_audio",
+    "../../common_audio:common_audio_c",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:platform_thread",
+    "../../rtc_base:random",
+    "../../rtc_base:rtc_event",
+    "../../rtc_base:rtc_task_queue",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:stringutils",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:arch",
+    "../../rtc_base/system:file_wrapper",
+    "../../rtc_base/task_utils:repeating_task",
+    "../../system_wrappers",
+    "../../system_wrappers:field_trial",
+    "../../system_wrappers:metrics",
+    "../utility",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings:strings",
+  ]
+  if (rtc_include_internal_audio_device && is_ios) {
+    deps += [ "../../sdk:audio_device" ]
+  }
+
+  sources = [
+    "dummy/audio_device_dummy.cc",
+    "dummy/audio_device_dummy.h",
+    "dummy/file_audio_device.cc",
+    "dummy/file_audio_device.h",
+    "include/fake_audio_device.h",
+    "include/test_audio_device.cc",
+    "include/test_audio_device.h",
+  ]
+
+  if (build_with_mozilla) {
+    sources -= [
+      "include/test_audio_device.cc",
+      "include/test_audio_device.h",
+    ]
+  }
+
+  defines = []
+  cflags = []
+  if (rtc_audio_device_plays_sinus_tone) {
+    defines += [ "AUDIO_DEVICE_PLAYS_SINUS_TONE" ]
+  }
+  if (rtc_enable_android_aaudio) {
+    defines += [ "WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO" ]
+  }
+  if (rtc_include_internal_audio_device) {
+    sources += [
+      "audio_device_data_observer.cc",
+      "audio_device_impl.cc",
+      "audio_device_impl.h",
+      "include/audio_device_data_observer.h",
+    ]
+    if (is_android) {
+      sources += [
+        "android/audio_common.h",
+        "android/audio_device_template.h",
+        "android/audio_manager.cc",
+        "android/audio_manager.h",
+        "android/audio_record_jni.cc",
+        "android/audio_record_jni.h",
+        "android/audio_track_jni.cc",
+        "android/audio_track_jni.h",
+        "android/build_info.cc",
+        "android/build_info.h",
+        "android/opensles_common.cc",
+        "android/opensles_common.h",
+        "android/opensles_player.cc",
+        "android/opensles_player.h",
+        "android/opensles_recorder.cc",
+        "android/opensles_recorder.h",
+      ]
+      libs = [
+        "log",
+        "OpenSLES",
+      ]
+      if (rtc_enable_android_aaudio) {
+        sources += [
+          "android/aaudio_player.cc",
+          "android/aaudio_player.h",
+          "android/aaudio_recorder.cc",
+          "android/aaudio_recorder.h",
+          "android/aaudio_wrapper.cc",
+          "android/aaudio_wrapper.h",
+        ]
+        libs += [ "aaudio" ]
+      }
+
+      if (build_with_mozilla) {
+        include_dirs += [
+          "/config/external/nspr",
+          "/nsprpub/lib/ds",
+          "/nsprpub/pr/include",
+        ]
+      }
+    }
+    if (rtc_use_dummy_audio_file_devices) {
+      defines += [ "WEBRTC_DUMMY_FILE_DEVICES" ]
+    } else {
+      if (is_linux || is_chromeos) {
+        sources += [
+          "linux/alsasymboltable_linux.cc",
+          "linux/alsasymboltable_linux.h",
+          "linux/audio_device_alsa_linux.cc",
+          "linux/audio_device_alsa_linux.h",
+          "linux/audio_mixer_manager_alsa_linux.cc",
+          "linux/audio_mixer_manager_alsa_linux.h",
+          "linux/latebindingsymboltable_linux.cc",
+          "linux/latebindingsymboltable_linux.h",
+        ]
+        defines += [ "WEBRTC_ENABLE_LINUX_ALSA" ]
+        libs = [ "dl" ]
+        if (rtc_use_x11) {
+          libs += [ "X11" ]
+          defines += [ "WEBRTC_USE_X11" ]
+        }
+        if (rtc_include_pulse_audio) {
+          defines += [ "WEBRTC_ENABLE_LINUX_PULSE" ]
+        }
+        sources += [
+          "linux/audio_device_pulse_linux.cc",
+          "linux/audio_device_pulse_linux.h",
+          "linux/audio_mixer_manager_pulse_linux.cc",
+          "linux/audio_mixer_manager_pulse_linux.h",
+          "linux/pulseaudiosymboltable_linux.cc",
+          "linux/pulseaudiosymboltable_linux.h",
+        ]
+      }
+      if (is_mac) {
+        sources += [
+          "mac/audio_device_mac.cc",
+          "mac/audio_device_mac.h",
+          "mac/audio_mixer_manager_mac.cc",
+          "mac/audio_mixer_manager_mac.h",
+        ]
+        deps += [
+          ":audio_device_impl_frameworks",
+          "../third_party/portaudio:mac_portaudio",
+        ]
+      }
+      if (is_win) {
+        sources += [
+          "win/audio_device_core_win.cc",
+          "win/audio_device_core_win.h",
+        ]
+        libs = [
+          # Required for the built-in WASAPI AEC.
+          "dmoguids.lib",
+          "wmcodecdspuuid.lib",
+          "amstrmid.lib",
+          "msdmo.lib",
+          "oleaut32.lib",
+        ]
+        deps += [
+          "../../rtc_base:win32",
+          "../../rtc_base/win:scoped_com_initializer",
+        ]
+      }
+      configs += [ ":audio_device_warnings_config" ]
+    }
+  } else {
+    defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ]
+  }
+
+  if (!build_with_chromium) {
+    sources += [
+      # Do not link these into Chrome since they contain static data.
+      "dummy/file_audio_device_factory.cc",
+      "dummy/file_audio_device_factory.h",
+    ]
+  }
+}
+}
+
+if (is_mac) {
+  rtc_source_set("audio_device_impl_frameworks") {
+    visibility = [ ":*" ]
+    frameworks = [
+      # Needed for CoreGraphics:
+      "ApplicationServices.framework",
+
+      "AudioToolbox.framework",
+      "CoreAudio.framework",
+
+      # Needed for CGEventSourceKeyState in audio_device_mac.cc:
+      "CoreGraphics.framework",
+    ]
+  }
+}
+
+if (!build_with_mozilla) { # See Bug 1820869.
+rtc_source_set("mock_audio_device") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "include/mock_audio_device.h",
+    "include/mock_audio_transport.h",
+    "mock_audio_device_buffer.h",
+  ]
+  deps = [
+    ":audio_device",
+    ":audio_device_buffer",
+    ":audio_device_impl",
+    "../../api:make_ref_counted",
+    "../../test:test_support",
+  ]
+}
+}
+
+# See Bug 1820869 for !build_with_mozilla.
+if (rtc_include_tests && !build_with_chromium && !build_with_mozilla) {
+  rtc_library("audio_device_unittests") {
+    testonly = true
+
+    sources = [
+      "fine_audio_buffer_unittest.cc",
+      "include/test_audio_device_unittest.cc",
+    ]
+    deps = [
+      ":audio_device",
+      ":audio_device_buffer",
+      ":audio_device_impl",
+      ":mock_audio_device",
+      "../../api:array_view",
+      "../../api:scoped_refptr",
+      "../../api:sequence_checker",
+      "../../api/task_queue",
+      "../../api/task_queue:default_task_queue_factory",
+      "../../common_audio",
+      "../../rtc_base:buffer",
+      "../../rtc_base:checks",
+      "../../rtc_base:ignore_wundef",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:race_checker",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:safe_conversions",
+      "../../rtc_base:timeutils",
+      "../../rtc_base/synchronization:mutex",
+      "../../system_wrappers",
+      "../../test:fileutils",
+      "../../test:test_support",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+    if (is_linux || is_chromeos || is_mac || is_win) {
+      sources += [ "audio_device_unittest.cc" ]
+    }
+    if (is_win) {
+      sources += [ "win/core_audio_utility_win_unittest.cc" ]
+      deps += [
+        ":audio_device_module_from_input_and_output",
+        ":windows_core_audio_utility",
+        "../../rtc_base/win:scoped_com_initializer",
+        "../../rtc_base/win:windows_version",
+      ]
+    }
+    if (is_android) {
+      sources += [
+        "android/audio_device_unittest.cc",
+        "android/audio_manager_unittest.cc",
+        "android/ensure_initialized.cc",
+        "android/ensure_initialized.h",
+      ]
+      deps += [
+        "../../sdk/android:internal_jni",
+        "../../sdk/android:libjingle_peerconnection_java",
+        "../../sdk/android:native_api_jni",
+        "../../sdk/android:native_test_jni_onload",
+        "../utility",
+      ]
+    }
+    if (!rtc_include_internal_audio_device) {
+      defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ]
+    }
+  }
+}
+
+if ((!build_with_chromium && !build_with_mozilla) && is_android) {
+  rtc_android_library("audio_device_java") {
+    sources = [
+      "android/java/src/org/webrtc/voiceengine/BuildInfo.java",
+      "android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java",
+      "android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
+      "android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
+      "android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
+      "android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
+    ]
+    deps = [
+      "../../rtc_base:base_java",
+      "//third_party/androidx:androidx_annotation_annotation_java",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_device/DEPS b/third_party/libwebrtc/modules/audio_device/DEPS
new file mode 100644
index 0000000000..9cc627d330
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/DEPS
@@ -0,0 +1,13 @@
+include_rules = [
+  "+common_audio",
+  "+system_wrappers",
+]
+
+specific_include_rules = {
+  "ensure_initialized\.cc": [
+    "+sdk/android",
+  ],
+  "audio_device_impl\.cc": [
+    "+sdk/objc",
+  ],
+}
diff --git a/third_party/libwebrtc/modules/audio_device/OWNERS b/third_party/libwebrtc/modules/audio_device/OWNERS
new file mode 100644
index 0000000000..22d03d552b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/OWNERS
@@ -0,0 +1,2 @@
+henrika@webrtc.org
+tkchin@webrtc.org
diff --git a/third_party/libwebrtc/modules/audio_device/android/aaudio_player.cc b/third_party/libwebrtc/modules/audio_device/android/aaudio_player.cc
new file mode 100644
index 0000000000..81e5bf5427
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/aaudio_player.cc
@@ -0,0 +1,216 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/aaudio_player.h"
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+AAudioPlayer::AAudioPlayer(AudioManager* audio_manager)
+    : main_thread_(TaskQueueBase::Current()),
+      aaudio_(audio_manager, AAUDIO_DIRECTION_OUTPUT, this) {
+  RTC_LOG(LS_INFO) << "ctor";
+  thread_checker_aaudio_.Detach();
+}
+
+AAudioPlayer::~AAudioPlayer() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  Terminate();
+  RTC_LOG(LS_INFO) << "#detected underruns: " << underrun_count_;
+}
+
+int AAudioPlayer::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (aaudio_.audio_parameters().channels() == 2) {
+    RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
+  }
+  return 0;
+}
+
+int AAudioPlayer::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  StopPlayout();
+  return 0;
+}
+
+int AAudioPlayer::InitPlayout() {
+  RTC_LOG(LS_INFO) << "InitPlayout";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!playing_);
+  if (!aaudio_.Init()) {
+    return -1;
+  }
+  initialized_ = true;
+  return 0;
+}
+
+bool AAudioPlayer::PlayoutIsInitialized() const {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  return initialized_;
+}
+
+int AAudioPlayer::StartPlayout() {
+  RTC_LOG(LS_INFO) << "StartPlayout";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DCHECK(!playing_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Playout can not start since InitPlayout must succeed first";
+    return 0;
+  }
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetPlayout();
+  }
+  if (!aaudio_.Start()) {
+    return -1;
+  }
+  underrun_count_ = aaudio_.xrun_count();
+  first_data_callback_ = true;
+  playing_ = true;
+  return 0;
+}
+
+int AAudioPlayer::StopPlayout() {
+  RTC_LOG(LS_INFO) << "StopPlayout";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (!initialized_ || !playing_) {
+    return 0;
+  }
+  if (!aaudio_.Stop()) {
+    RTC_LOG(LS_ERROR) << "StopPlayout failed";
+    return -1;
+  }
+  thread_checker_aaudio_.Detach();
+  initialized_ = false;
+  playing_ = false;
+  return 0;
+}
+
+bool AAudioPlayer::Playing() const {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  return playing_;
+}
+
+void AAudioPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_DLOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  audio_device_buffer_ = audioBuffer;
+  const AudioParameters audio_parameters = aaudio_.audio_parameters();
+  audio_device_buffer_->SetPlayoutSampleRate(audio_parameters.sample_rate());
+  audio_device_buffer_->SetPlayoutChannels(audio_parameters.channels());
+  RTC_CHECK(audio_device_buffer_);
+  // Create a modified audio buffer class which allows us to ask for any number
+  // of samples (and not only multiple of 10ms) to match the optimal buffer
+  // size per callback used by AAudio.
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+}
+
+int AAudioPlayer::SpeakerVolumeIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+void AAudioPlayer::OnErrorCallback(aaudio_result_t error) {
+  RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
+  // TODO(henrika): investigate if we can use a thread checker here. Initial
+  // tests shows that this callback can sometimes be called on a unique thread
+  // but according to the documentation it should be on the same thread as the
+  // data callback.
+  // RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
+  if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
+    // The stream is disconnected and any attempt to use it will return
+    // AAUDIO_ERROR_DISCONNECTED.
+    RTC_LOG(LS_WARNING) << "Output stream disconnected";
+    // AAudio documentation states: "You should not close or reopen the stream
+    // from the callback, use another thread instead". A message is therefore
+    // sent to the main thread to do the restart operation.
+    RTC_DCHECK(main_thread_);
+    main_thread_->PostTask([this] { HandleStreamDisconnected(); });
+  }
+}
+
+aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
+                                                           int32_t num_frames) {
+  RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
+  // Log device id in first data callback to ensure that a valid device is
+  // utilized.
+  if (first_data_callback_) {
+    RTC_LOG(LS_INFO) << "--- First output data callback: "
+                        "device id="
+                     << aaudio_.device_id();
+    first_data_callback_ = false;
+  }
+
+  // Check if the underrun count has increased. If it has, increase the buffer
+  // size by adding the size of a burst. It will reduce the risk of underruns
+  // at the expense of an increased latency.
+  // TODO(henrika): enable possibility to disable and/or tune the algorithm.
+  const int32_t underrun_count = aaudio_.xrun_count();
+  if (underrun_count > underrun_count_) {
+    RTC_LOG(LS_ERROR) << "Underrun detected: " << underrun_count;
+    underrun_count_ = underrun_count;
+    aaudio_.IncreaseOutputBufferSize();
+  }
+
+  // Estimate latency between writing an audio frame to the output stream and
+  // the time that same frame is played out on the output audio device.
+  latency_millis_ = aaudio_.EstimateLatencyMillis();
+  // TODO(henrika): use for development only.
+  if (aaudio_.frames_written() % (1000 * aaudio_.frames_per_burst()) == 0) {
+    RTC_DLOG(LS_INFO) << "output latency: " << latency_millis_
+                      << ", num_frames: " << num_frames;
+  }
+
+  // Read audio data from the WebRTC source using the FineAudioBuffer object
+  // and write that data into `audio_data` to be played out by AAudio.
+  // Prime output with zeros during a short initial phase to avoid distortion.
+  // TODO(henrika): do more work to figure out of if the initial forced silence
+  // period is really needed.
+  if (aaudio_.frames_written() < 50 * aaudio_.frames_per_burst()) {
+    const size_t num_bytes =
+        sizeof(int16_t) * aaudio_.samples_per_frame() * num_frames;
+    memset(audio_data, 0, num_bytes);
+  } else {
+    fine_audio_buffer_->GetPlayoutData(
+        rtc::MakeArrayView(static_cast<int16_t*>(audio_data),
+                           aaudio_.samples_per_frame() * num_frames),
+        static_cast<int>(latency_millis_ + 0.5));
+  }
+
+  // TODO(henrika): possibly add trace here to be included in systrace.
+  // See https://developer.android.com/studio/profile/systrace-commandline.html.
+  return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+void AAudioPlayer::HandleStreamDisconnected() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DLOG(LS_INFO) << "HandleStreamDisconnected";
+  if (!initialized_ || !playing_) {
+    return;
+  }
+  // Perform a restart by first closing the disconnected stream and then start
+  // a new stream; this time using the new (preferred) audio output device.
+  StopPlayout();
+  InitPlayout();
+  StartPlayout();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/aaudio_player.h b/third_party/libwebrtc/modules/audio_device/android/aaudio_player.h
new file mode 100644
index 0000000000..ea5d578092
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/aaudio_player.h
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
+
+#include <aaudio/AAudio.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_device/android/aaudio_wrapper.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+class FineAudioBuffer;
+class AudioManager;
+
+// Implements low-latency 16-bit mono PCM audio output support for Android
+// using the C based AAudio API.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will DCHECK if any method is called on an invalid thread. Audio buffers
+// are requested on a dedicated high-priority thread owned by AAudio.
+//
+// The existing design forces the user to call InitPlayout() after StopPlayout()
+// to be able to call StartPlayout() again. This is in line with how the Java-
+// based implementation works.
+//
+// An audio stream can be disconnected, e.g. when an audio device is removed.
+// This implementation will restart the audio stream using the new preferred
+// device if such an event happens.
+//
+// Also supports automatic buffer-size adjustment based on underrun detections
+// where the internal AAudio buffer can be increased when needed. It will
+// reduce the risk of underruns (~glitches) at the expense of an increased
+// latency.
+class AAudioPlayer final : public AAudioObserverInterface {
+ public:
+  explicit AAudioPlayer(AudioManager* audio_manager);
+  ~AAudioPlayer();
+
+  int Init();
+  int Terminate();
+
+  int InitPlayout();
+  bool PlayoutIsInitialized() const;
+
+  int StartPlayout();
+  int StopPlayout();
+  bool Playing() const;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+  // Not implemented in AAudio.
+  int SpeakerVolumeIsAvailable(bool& available);  // NOLINT
+  int SetSpeakerVolume(uint32_t volume) { return -1; }
+  int SpeakerVolume(uint32_t& volume) const { return -1; }        // NOLINT
+  int MaxSpeakerVolume(uint32_t& maxVolume) const { return -1; }  // NOLINT
+  int MinSpeakerVolume(uint32_t& minVolume) const { return -1; }  // NOLINT
+
+ protected:
+  // AAudioObserverInterface implementation.
+
+  // For an output stream, this function should render and write `num_frames`
+  // of data in the streams current data format to the `audio_data` buffer.
+  // Called on a real-time thread owned by AAudio.
+  aaudio_data_callback_result_t OnDataCallback(void* audio_data,
+                                               int32_t num_frames) override;
+  // AAudio calls this functions if any error occurs on a callback thread.
+  // Called on a real-time thread owned by AAudio.
+  void OnErrorCallback(aaudio_result_t error) override;
+
+ private:
+  // Closes the existing stream and starts a new stream.
+  void HandleStreamDisconnected();
+
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  SequenceChecker main_thread_checker_;
+
+  // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
+  // real-time thread owned by AAudio. Detached during construction of this
+  // object.
+  SequenceChecker thread_checker_aaudio_;
+
+  // The task queue on which this object is created on.
+  TaskQueueBase* main_thread_;
+
+  // Wraps all AAudio resources. Contains an output stream using the default
+  // output audio device. Can be accessed on both the main thread and the
+  // real-time thread owned by AAudio. See separate AAudio documentation about
+  // thread safety.
+  AAudioWrapper aaudio_;
+
+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
+  // in chunks of 10ms. It then allows for this data to be pulled in
+  // a finer or coarser granularity. I.e. interacting with this class instead
+  // of directly with the AudioDeviceBuffer one can ask for any number of
+  // audio data samples.
+  // Example: native buffer size can be 192 audio frames at 48kHz sample rate.
+  // WebRTC will provide 480 audio frames per 10ms but AAudio asks for 192
+  // in each callback (once every 4th ms). This class can then ask for 192 and
+  // the FineAudioBuffer will ask WebRTC for new data approximately only every
+  // second callback and also cache non-utilized audio.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Counts number of detected underrun events reported by AAudio.
+  int32_t underrun_count_ = 0;
+
+  // True only for the first data callback in each audio session.
+  bool first_data_callback_ = true;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and set by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_ RTC_GUARDED_BY(main_thread_checker_) =
+      nullptr;
+
+  bool initialized_ RTC_GUARDED_BY(main_thread_checker_) = false;
+  bool playing_ RTC_GUARDED_BY(main_thread_checker_) = false;
+
+  // Estimated latency between writing an audio frame to the output stream and
+  // the time that same frame is played out on the output audio device.
+  double latency_millis_ RTC_GUARDED_BY(thread_checker_aaudio_) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/aaudio_recorder.cc b/third_party/libwebrtc/modules/audio_device/android/aaudio_recorder.cc
new file mode 100644
index 0000000000..21e5dd8a74
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/aaudio_recorder.cc
@@ -0,0 +1,205 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/aaudio_recorder.h"
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+AAudioRecorder::AAudioRecorder(AudioManager* audio_manager)
+    : main_thread_(TaskQueueBase::Current()),
+      aaudio_(audio_manager, AAUDIO_DIRECTION_INPUT, this) {
+  RTC_LOG(LS_INFO) << "ctor";
+  thread_checker_aaudio_.Detach();
+}
+
+AAudioRecorder::~AAudioRecorder() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+  RTC_LOG(LS_INFO) << "detected owerflows: " << overflow_count_;
+}
+
+int AAudioRecorder::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (aaudio_.audio_parameters().channels() == 2) {
+    RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
+  }
+  return 0;
+}
+
+int AAudioRecorder::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopRecording();
+  return 0;
+}
+
+int AAudioRecorder::InitRecording() {
+  RTC_LOG(LS_INFO) << "InitRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  if (!aaudio_.Init()) {
+    return -1;
+  }
+  initialized_ = true;
+  return 0;
+}
+
+int AAudioRecorder::StartRecording() {
+  RTC_LOG(LS_INFO) << "StartRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!recording_);
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetPlayout();
+  }
+  if (!aaudio_.Start()) {
+    return -1;
+  }
+  overflow_count_ = aaudio_.xrun_count();
+  first_data_callback_ = true;
+  recording_ = true;
+  return 0;
+}
+
+int AAudioRecorder::StopRecording() {
+  RTC_LOG(LS_INFO) << "StopRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !recording_) {
+    return 0;
+  }
+  if (!aaudio_.Stop()) {
+    return -1;
+  }
+  thread_checker_aaudio_.Detach();
+  initialized_ = false;
+  recording_ = false;
+  return 0;
+}
+
+void AAudioRecorder::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_LOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const AudioParameters audio_parameters = aaudio_.audio_parameters();
+  audio_device_buffer_->SetRecordingSampleRate(audio_parameters.sample_rate());
+  audio_device_buffer_->SetRecordingChannels(audio_parameters.channels());
+  RTC_CHECK(audio_device_buffer_);
+  // Create a modified audio buffer class which allows us to deliver any number
+  // of samples (and not only multiples of 10ms which WebRTC uses) to match the
+  // native AAudio buffer size.
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+}
+
+int AAudioRecorder::EnableBuiltInAEC(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInAEC: " << enable;
+  RTC_LOG(LS_ERROR) << "Not implemented";
+  return -1;
+}
+
+int AAudioRecorder::EnableBuiltInAGC(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInAGC: " << enable;
+  RTC_LOG(LS_ERROR) << "Not implemented";
+  return -1;
+}
+
+int AAudioRecorder::EnableBuiltInNS(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInNS: " << enable;
+  RTC_LOG(LS_ERROR) << "Not implemented";
+  return -1;
+}
+
+void AAudioRecorder::OnErrorCallback(aaudio_result_t error) {
+  RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
+  // RTC_DCHECK(thread_checker_aaudio_.IsCurrent());
+  if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
+    // The stream is disconnected and any attempt to use it will return
+    // AAUDIO_ERROR_DISCONNECTED..
+    RTC_LOG(LS_WARNING) << "Input stream disconnected => restart is required";
+    // AAudio documentation states: "You should not close or reopen the stream
+    // from the callback, use another thread instead". A message is therefore
+    // sent to the main thread to do the restart operation.
+    RTC_DCHECK(main_thread_);
+    main_thread_->PostTask([this] { HandleStreamDisconnected(); });
+  }
+}
+
+// Read and process `num_frames` of data from the `audio_data` buffer.
+// TODO(henrika): possibly add trace here to be included in systrace.
+// See https://developer.android.com/studio/profile/systrace-commandline.html.
+aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
+    void* audio_data,
+    int32_t num_frames) {
+  // TODO(henrika): figure out why we sometimes hit this one.
+  // RTC_DCHECK(thread_checker_aaudio_.IsCurrent());
+  // RTC_LOG(LS_INFO) << "OnDataCallback: " << num_frames;
+  // Drain the input buffer at first callback to ensure that it does not
+  // contain any old data. Will also ensure that the lowest possible latency
+  // is obtained.
+  if (first_data_callback_) {
+    RTC_LOG(LS_INFO) << "--- First input data callback: "
+                        "device id="
+                     << aaudio_.device_id();
+    aaudio_.ClearInputStream(audio_data, num_frames);
+    first_data_callback_ = false;
+  }
+  // Check if the overflow counter has increased and if so log a warning.
+  // TODO(henrika): possible add UMA stat or capacity extension.
+  const int32_t overflow_count = aaudio_.xrun_count();
+  if (overflow_count > overflow_count_) {
+    RTC_LOG(LS_ERROR) << "Overflow detected: " << overflow_count;
+    overflow_count_ = overflow_count;
+  }
+  // Estimated time between an audio frame was recorded by the input device and
+  // it can read on the input stream.
+  latency_millis_ = aaudio_.EstimateLatencyMillis();
+  // TODO(henrika): use for development only.
+  if (aaudio_.frames_read() % (1000 * aaudio_.frames_per_burst()) == 0) {
+    RTC_DLOG(LS_INFO) << "input latency: " << latency_millis_
+                      << ", num_frames: " << num_frames;
+  }
+  // Copy recorded audio in `audio_data` to the WebRTC sink using the
+  // FineAudioBuffer object.
+  fine_audio_buffer_->DeliverRecordedData(
+      rtc::MakeArrayView(static_cast<const int16_t*>(audio_data),
+                         aaudio_.samples_per_frame() * num_frames),
+      static_cast<int>(latency_millis_ + 0.5));
+
+  return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+void AAudioRecorder::HandleStreamDisconnected() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_LOG(LS_INFO) << "HandleStreamDisconnected";
+  if (!initialized_ || !recording_) {
+    return;
+  }
+  // Perform a restart by first closing the disconnected stream and then start
+  // a new stream; this time using the new (preferred) audio input device.
+  // TODO(henrika): resolve issue where a one restart attempt leads to a long
+  // sequence of new calls to OnErrorCallback().
+  // See b/73148976 for details.
+  StopRecording();
+  InitRecording();
+  StartRecording();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/aaudio_recorder.h b/third_party/libwebrtc/modules/audio_device/android/aaudio_recorder.h
new file mode 100644
index 0000000000..6df7eed076
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/aaudio_recorder.h
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
+
+#include <aaudio/AAudio.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_device/android/aaudio_wrapper.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+class FineAudioBuffer;
+class AudioManager;
+
+// Implements low-latency 16-bit mono PCM audio input support for Android
+// using the C based AAudio API.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread. Audio buffers
+// are delivered on a dedicated high-priority thread owned by AAudio.
+//
+// The existing design forces the user to call InitRecording() after
+// StopRecording() to be able to call StartRecording() again. This is in line
+// with how the Java- based implementation works.
+//
+// TODO(henrika): add comments about device changes and adaptive buffer
+// management.
+class AAudioRecorder : public AAudioObserverInterface {
+ public:
+  explicit AAudioRecorder(AudioManager* audio_manager);
+  ~AAudioRecorder();
+
+  int Init();
+  int Terminate();
+
+  int InitRecording();
+  bool RecordingIsInitialized() const { return initialized_; }
+
+  int StartRecording();
+  int StopRecording();
+  bool Recording() const { return recording_; }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+  double latency_millis() const { return latency_millis_; }
+
+  // TODO(henrika): add support using AAudio APIs when available.
+  int EnableBuiltInAEC(bool enable);
+  int EnableBuiltInAGC(bool enable);
+  int EnableBuiltInNS(bool enable);
+
+ protected:
+  // AAudioObserverInterface implementation.
+
+  // For an input stream, this function should read `num_frames` of recorded
+  // data, in the stream's current data format, from the `audio_data` buffer.
+  // Called on a real-time thread owned by AAudio.
+  aaudio_data_callback_result_t OnDataCallback(void* audio_data,
+                                               int32_t num_frames) override;
+
+  // AAudio calls this function if any error occurs on a callback thread.
+  // Called on a real-time thread owned by AAudio.
+  void OnErrorCallback(aaudio_result_t error) override;
+
+ private:
+  // Closes the existing stream and starts a new stream.
+  void HandleStreamDisconnected();
+
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
+  // real-time thread owned by AAudio. Detached during construction of this
+  // object.
+  SequenceChecker thread_checker_aaudio_;
+
+  // The thread on which this object is created on.
+  TaskQueueBase* main_thread_;
+
+  // Wraps all AAudio resources. Contains an input stream using the default
+  // input audio device.
+  AAudioWrapper aaudio_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_ = nullptr;
+
+  bool initialized_ = false;
+  bool recording_ = false;
+
+  // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
+  // chunks of audio.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Counts number of detected overflow events reported by AAudio.
+  int32_t overflow_count_ = 0;
+
+  // Estimated time between an audio frame was recorded by the input device and
+  // it can read on the input stream.
+  double latency_millis_ = 0;
+
+  // True only for the first data callback in each audio session.
+  bool first_data_callback_ = true;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/aaudio_wrapper.cc b/third_party/libwebrtc/modules/audio_device/android/aaudio_wrapper.cc
new file mode 100644
index 0000000000..3d824b5c57
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/aaudio_wrapper.cc
@@ -0,0 +1,499 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/aaudio_wrapper.h"
+
+#include "modules/audio_device/android/audio_manager.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+
+#define LOG_ON_ERROR(op)                                                      \
+  do {                                                                        \
+    aaudio_result_t result = (op);                                            \
+    if (result != AAUDIO_OK) {                                                \
+      RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
+    }                                                                         \
+  } while (0)
+
+#define RETURN_ON_ERROR(op, ...)                                              \
+  do {                                                                        \
+    aaudio_result_t result = (op);                                            \
+    if (result != AAUDIO_OK) {                                                \
+      RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
+      return __VA_ARGS__;                                                     \
+    }                                                                         \
+  } while (0)
+
+namespace webrtc {
+
+namespace {
+
+const char* DirectionToString(aaudio_direction_t direction) {
+  switch (direction) {
+    case AAUDIO_DIRECTION_OUTPUT:
+      return "OUTPUT";
+    case AAUDIO_DIRECTION_INPUT:
+      return "INPUT";
+    default:
+      return "UNKNOWN";
+  }
+}
+
+const char* SharingModeToString(aaudio_sharing_mode_t mode) {
+  switch (mode) {
+    case AAUDIO_SHARING_MODE_EXCLUSIVE:
+      return "EXCLUSIVE";
+    case AAUDIO_SHARING_MODE_SHARED:
+      return "SHARED";
+    default:
+      return "UNKNOWN";
+  }
+}
+
+const char* PerformanceModeToString(aaudio_performance_mode_t mode) {
+  switch (mode) {
+    case AAUDIO_PERFORMANCE_MODE_NONE:
+      return "NONE";
+    case AAUDIO_PERFORMANCE_MODE_POWER_SAVING:
+      return "POWER_SAVING";
+    case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY:
+      return "LOW_LATENCY";
+    default:
+      return "UNKNOWN";
+  }
+}
+
+const char* FormatToString(int32_t id) {
+  switch (id) {
+    case AAUDIO_FORMAT_INVALID:
+      return "INVALID";
+    case AAUDIO_FORMAT_UNSPECIFIED:
+      return "UNSPECIFIED";
+    case AAUDIO_FORMAT_PCM_I16:
+      return "PCM_I16";
+    case AAUDIO_FORMAT_PCM_FLOAT:
+      return "FLOAT";
+    default:
+      return "UNKNOWN";
+  }
+}
+
+void ErrorCallback(AAudioStream* stream,
+                   void* user_data,
+                   aaudio_result_t error) {
+  RTC_DCHECK(user_data);
+  AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
+  RTC_LOG(LS_WARNING) << "ErrorCallback: "
+                      << DirectionToString(aaudio_wrapper->direction());
+  RTC_DCHECK(aaudio_wrapper->observer());
+  aaudio_wrapper->observer()->OnErrorCallback(error);
+}
+
+aaudio_data_callback_result_t DataCallback(AAudioStream* stream,
+                                           void* user_data,
+                                           void* audio_data,
+                                           int32_t num_frames) {
+  RTC_DCHECK(user_data);
+  RTC_DCHECK(audio_data);
+  AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
+  RTC_DCHECK(aaudio_wrapper->observer());
+  return aaudio_wrapper->observer()->OnDataCallback(audio_data, num_frames);
+}
+
+// Wraps the stream builder object to ensure that it is released properly when
+// the stream builder goes out of scope.
+class ScopedStreamBuilder {
+ public:
+  ScopedStreamBuilder() {
+    LOG_ON_ERROR(AAudio_createStreamBuilder(&builder_));
+    RTC_DCHECK(builder_);
+  }
+  ~ScopedStreamBuilder() {
+    if (builder_) {
+      LOG_ON_ERROR(AAudioStreamBuilder_delete(builder_));
+    }
+  }
+
+  AAudioStreamBuilder* get() const { return builder_; }
+
+ private:
+  AAudioStreamBuilder* builder_ = nullptr;
+};
+
+}  // namespace
+
+AAudioWrapper::AAudioWrapper(AudioManager* audio_manager,
+                             aaudio_direction_t direction,
+                             AAudioObserverInterface* observer)
+    : direction_(direction), observer_(observer) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_DCHECK(observer_);
+  direction_ == AAUDIO_DIRECTION_OUTPUT
+      ? audio_parameters_ = audio_manager->GetPlayoutAudioParameters()
+      : audio_parameters_ = audio_manager->GetRecordAudioParameters();
+  aaudio_thread_checker_.Detach();
+  RTC_LOG(LS_INFO) << audio_parameters_.ToString();
+}
+
+AAudioWrapper::~AAudioWrapper() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!stream_);
+}
+
+bool AAudioWrapper::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Creates a stream builder which can be used to open an audio stream.
+  ScopedStreamBuilder builder;
+  // Configures the stream builder using audio parameters given at construction.
+  SetStreamConfiguration(builder.get());
+  // Opens a stream based on options in the stream builder.
+  if (!OpenStream(builder.get())) {
+    return false;
+  }
+  // Ensures that the opened stream could activate the requested settings.
+  if (!VerifyStreamConfiguration()) {
+    return false;
+  }
+  // Optimizes the buffer scheme for lowest possible latency and creates
+  // additional buffer logic to match the 10ms buffer size used in WebRTC.
+  if (!OptimizeBuffers()) {
+    return false;
+  }
+  LogStreamState();
+  return true;
+}
+
+bool AAudioWrapper::Start() {
+  RTC_LOG(LS_INFO) << "Start";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // TODO(henrika): this state check might not be needed.
+  aaudio_stream_state_t current_state = AAudioStream_getState(stream_);
+  if (current_state != AAUDIO_STREAM_STATE_OPEN) {
+    RTC_LOG(LS_ERROR) << "Invalid state: "
+                      << AAudio_convertStreamStateToText(current_state);
+    return false;
+  }
+  // Asynchronous request for the stream to start.
+  RETURN_ON_ERROR(AAudioStream_requestStart(stream_), false);
+  LogStreamState();
+  return true;
+}
+
+bool AAudioWrapper::Stop() {
+  RTC_LOG(LS_INFO) << "Stop: " << DirectionToString(direction());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Asynchronous request for the stream to stop.
+  RETURN_ON_ERROR(AAudioStream_requestStop(stream_), false);
+  CloseStream();
+  aaudio_thread_checker_.Detach();
+  return true;
+}
+
+double AAudioWrapper::EstimateLatencyMillis() const {
+  RTC_DCHECK(stream_);
+  double latency_millis = 0.0;
+  if (direction() == AAUDIO_DIRECTION_INPUT) {
+    // For input streams. Best guess we can do is to use the current burst size
+    // as delay estimate.
+    latency_millis = static_cast<double>(frames_per_burst()) / sample_rate() *
+                     rtc::kNumMillisecsPerSec;
+  } else {
+    int64_t existing_frame_index;
+    int64_t existing_frame_presentation_time;
+    // Get the time at which a particular frame was presented to audio hardware.
+    aaudio_result_t result = AAudioStream_getTimestamp(
+        stream_, CLOCK_MONOTONIC, &existing_frame_index,
+        &existing_frame_presentation_time);
+    // Results are only valid when the stream is in AAUDIO_STREAM_STATE_STARTED.
+    if (result == AAUDIO_OK) {
+      // Get write index for next audio frame.
+      int64_t next_frame_index = frames_written();
+      // Number of frames between next frame and the existing frame.
+      int64_t frame_index_delta = next_frame_index - existing_frame_index;
+      // Assume the next frame will be written now.
+      int64_t next_frame_write_time = rtc::TimeNanos();
+      // Calculate time when next frame will be presented to the hardware taking
+      // sample rate into account.
+      int64_t frame_time_delta =
+          (frame_index_delta * rtc::kNumNanosecsPerSec) / sample_rate();
+      int64_t next_frame_presentation_time =
+          existing_frame_presentation_time + frame_time_delta;
+      // Derive a latency estimate given results above.
+      latency_millis = static_cast<double>(next_frame_presentation_time -
+                                           next_frame_write_time) /
+                       rtc::kNumNanosecsPerMillisec;
+    }
+  }
+  return latency_millis;
+}
+
+// Returns new buffer size or a negative error value if buffer size could not
+// be increased.
+bool AAudioWrapper::IncreaseOutputBufferSize() {
+  RTC_LOG(LS_INFO) << "IncreaseBufferSize";
+  RTC_DCHECK(stream_);
+  RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
+  RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_OUTPUT);
+  aaudio_result_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
+  // Try to increase size of buffer with one burst to reduce risk of underrun.
+  buffer_size += frames_per_burst();
+  // Verify that the new buffer size is not larger than max capacity.
+  // TODO(henrika): keep track of case when we reach the capacity limit.
+  const int32_t max_buffer_size = buffer_capacity_in_frames();
+  if (buffer_size > max_buffer_size) {
+    RTC_LOG(LS_ERROR) << "Required buffer size (" << buffer_size
+                      << ") is higher than max: " << max_buffer_size;
+    return false;
+  }
+  RTC_LOG(LS_INFO) << "Updating buffer size to: " << buffer_size
+                   << " (max=" << max_buffer_size << ")";
+  buffer_size = AAudioStream_setBufferSizeInFrames(stream_, buffer_size);
+  if (buffer_size < 0) {
+    RTC_LOG(LS_ERROR) << "Failed to change buffer size: "
+                      << AAudio_convertResultToText(buffer_size);
+    return false;
+  }
+  RTC_LOG(LS_INFO) << "Buffer size changed to: " << buffer_size;
+  return true;
+}
+
+void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) {
+  RTC_LOG(LS_INFO) << "ClearInputStream";
+  RTC_DCHECK(stream_);
+  RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
+  RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_INPUT);
+  aaudio_result_t cleared_frames = 0;
+  do {
+    cleared_frames = AAudioStream_read(stream_, audio_data, num_frames, 0);
+  } while (cleared_frames > 0);
+}
+
+AAudioObserverInterface* AAudioWrapper::observer() const {
+  return observer_;
+}
+
+AudioParameters AAudioWrapper::audio_parameters() const {
+  return audio_parameters_;
+}
+
+int32_t AAudioWrapper::samples_per_frame() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getSamplesPerFrame(stream_);
+}
+
+int32_t AAudioWrapper::buffer_size_in_frames() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getBufferSizeInFrames(stream_);
+}
+
+int32_t AAudioWrapper::buffer_capacity_in_frames() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getBufferCapacityInFrames(stream_);
+}
+
+int32_t AAudioWrapper::device_id() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getDeviceId(stream_);
+}
+
+int32_t AAudioWrapper::xrun_count() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getXRunCount(stream_);
+}
+
+int32_t AAudioWrapper::format() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getFormat(stream_);
+}
+
+int32_t AAudioWrapper::sample_rate() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getSampleRate(stream_);
+}
+
+int32_t AAudioWrapper::channel_count() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getChannelCount(stream_);
+}
+
+int32_t AAudioWrapper::frames_per_callback() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getFramesPerDataCallback(stream_);
+}
+
+aaudio_sharing_mode_t AAudioWrapper::sharing_mode() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getSharingMode(stream_);
+}
+
+aaudio_performance_mode_t AAudioWrapper::performance_mode() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getPerformanceMode(stream_);
+}
+
+aaudio_stream_state_t AAudioWrapper::stream_state() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getState(stream_);
+}
+
+int64_t AAudioWrapper::frames_written() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getFramesWritten(stream_);
+}
+
+int64_t AAudioWrapper::frames_read() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getFramesRead(stream_);
+}
+
+void AAudioWrapper::SetStreamConfiguration(AAudioStreamBuilder* builder) {
+  RTC_LOG(LS_INFO) << "SetStreamConfiguration";
+  RTC_DCHECK(builder);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Request usage of default primary output/input device.
+  // TODO(henrika): verify that default device follows Java APIs.
+  // https://developer.android.com/reference/android/media/AudioDeviceInfo.html.
+  AAudioStreamBuilder_setDeviceId(builder, AAUDIO_UNSPECIFIED);
+  // Use preferred sample rate given by the audio parameters.
+  AAudioStreamBuilder_setSampleRate(builder, audio_parameters().sample_rate());
+  // Use preferred channel configuration given by the audio parameters.
+  AAudioStreamBuilder_setChannelCount(builder, audio_parameters().channels());
+  // Always use 16-bit PCM audio sample format.
+  AAudioStreamBuilder_setFormat(builder, AAUDIO_FORMAT_PCM_I16);
+  // TODO(henrika): investigate effect of using AAUDIO_SHARING_MODE_EXCLUSIVE.
+  // Ask for exclusive mode since this will give us the lowest possible latency.
+  // If exclusive mode isn't available, shared mode will be used instead.
+  AAudioStreamBuilder_setSharingMode(builder, AAUDIO_SHARING_MODE_SHARED);
+  // Use the direction that was given at construction.
+  AAudioStreamBuilder_setDirection(builder, direction_);
+  // TODO(henrika): investigate performance using different performance modes.
+  AAudioStreamBuilder_setPerformanceMode(builder,
+                                         AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+  // Given that WebRTC applications require low latency, our audio stream uses
+  // an asynchronous callback function to transfer data to and from the
+  // application. AAudio executes the callback in a higher-priority thread that
+  // has better performance.
+  AAudioStreamBuilder_setDataCallback(builder, DataCallback, this);
+  // Request that AAudio calls this functions if any error occurs on a callback
+  // thread.
+  AAudioStreamBuilder_setErrorCallback(builder, ErrorCallback, this);
+}
+
+bool AAudioWrapper::OpenStream(AAudioStreamBuilder* builder) {
+  RTC_LOG(LS_INFO) << "OpenStream";
+  RTC_DCHECK(builder);
+  AAudioStream* stream = nullptr;
+  RETURN_ON_ERROR(AAudioStreamBuilder_openStream(builder, &stream), false);
+  stream_ = stream;
+  LogStreamConfiguration();
+  return true;
+}
+
+void AAudioWrapper::CloseStream() {
+  RTC_LOG(LS_INFO) << "CloseStream";
+  RTC_DCHECK(stream_);
+  LOG_ON_ERROR(AAudioStream_close(stream_));
+  stream_ = nullptr;
+}
+
+void AAudioWrapper::LogStreamConfiguration() {
+  RTC_DCHECK(stream_);
+  char ss_buf[1024];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  ss << "Stream Configuration: ";
+  ss << "sample rate=" << sample_rate() << ", channels=" << channel_count();
+  ss << ", samples per frame=" << samples_per_frame();
+  ss << ", format=" << FormatToString(format());
+  ss << ", sharing mode=" << SharingModeToString(sharing_mode());
+  ss << ", performance mode=" << PerformanceModeToString(performance_mode());
+  ss << ", direction=" << DirectionToString(direction());
+  ss << ", device id=" << AAudioStream_getDeviceId(stream_);
+  ss << ", frames per callback=" << frames_per_callback();
+  RTC_LOG(LS_INFO) << ss.str();
+}
+
+void AAudioWrapper::LogStreamState() {
+  RTC_LOG(LS_INFO) << "AAudio stream state: "
+                   << AAudio_convertStreamStateToText(stream_state());
+}
+
+bool AAudioWrapper::VerifyStreamConfiguration() {
+  RTC_LOG(LS_INFO) << "VerifyStreamConfiguration";
+  RTC_DCHECK(stream_);
+  // TODO(henrika): should we verify device ID as well?
+  if (AAudioStream_getSampleRate(stream_) != audio_parameters().sample_rate()) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested sample rate";
+    return false;
+  }
+  if (AAudioStream_getChannelCount(stream_) !=
+      static_cast<int32_t>(audio_parameters().channels())) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested channel count";
+    return false;
+  }
+  if (AAudioStream_getFormat(stream_) != AAUDIO_FORMAT_PCM_I16) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested format";
+    return false;
+  }
+  if (AAudioStream_getSharingMode(stream_) != AAUDIO_SHARING_MODE_SHARED) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested sharing mode";
+    return false;
+  }
+  if (AAudioStream_getPerformanceMode(stream_) !=
+      AAUDIO_PERFORMANCE_MODE_LOW_LATENCY) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested performance mode";
+    return false;
+  }
+  if (AAudioStream_getDirection(stream_) != direction()) {
+    RTC_LOG(LS_ERROR) << "Stream direction could not be set";
+    return false;
+  }
+  if (AAudioStream_getSamplesPerFrame(stream_) !=
+      static_cast<int32_t>(audio_parameters().channels())) {
+    RTC_LOG(LS_ERROR) << "Invalid number of samples per frame";
+    return false;
+  }
+  return true;
+}
+
+bool AAudioWrapper::OptimizeBuffers() {
+  RTC_LOG(LS_INFO) << "OptimizeBuffers";
+  RTC_DCHECK(stream_);
+  // Maximum number of frames that can be filled without blocking.
+  RTC_LOG(LS_INFO) << "max buffer capacity in frames: "
+                   << buffer_capacity_in_frames();
+  // Query the number of frames that the application should read or write at
+  // one time for optimal performance.
+  int32_t frames_per_burst = AAudioStream_getFramesPerBurst(stream_);
+  RTC_LOG(LS_INFO) << "frames per burst for optimal performance: "
+                   << frames_per_burst;
+  frames_per_burst_ = frames_per_burst;
+  if (direction() == AAUDIO_DIRECTION_INPUT) {
+    // There is no point in calling setBufferSizeInFrames() for input streams
+    // since it has no effect on the performance (latency in this case).
+    return true;
+  }
+  // Set buffer size to same as burst size to guarantee lowest possible latency.
+  // This size might change for output streams if underruns are detected and
+  // automatic buffer adjustment is enabled.
+  AAudioStream_setBufferSizeInFrames(stream_, frames_per_burst);
+  int32_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
+  if (buffer_size != frames_per_burst) {
+    RTC_LOG(LS_ERROR) << "Failed to use optimal buffer burst size";
+    return false;
+  }
+  // Maximum number of frames that can be filled without blocking.
+  RTC_LOG(LS_INFO) << "buffer burst size in frames: " << buffer_size;
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/aaudio_wrapper.h b/third_party/libwebrtc/modules/audio_device/android/aaudio_wrapper.h
new file mode 100644
index 0000000000..1f925b96d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/aaudio_wrapper.h
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
+
+#include <aaudio/AAudio.h>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioManager;
+
+// AAudio callback interface for audio transport to/from the AAudio stream.
+// The interface also contains an error callback method for notifications of
+// e.g. device changes.
+class AAudioObserverInterface {
+ public:
+  // Audio data will be passed in our out of this function dependning on the
+  // direction of the audio stream. This callback function will be called on a
+  // real-time thread owned by AAudio.
+  virtual aaudio_data_callback_result_t OnDataCallback(void* audio_data,
+                                                       int32_t num_frames) = 0;
+  // AAudio will call this functions if any error occurs on a callback thread.
+  // In response, this function could signal or launch another thread to reopen
+  // a stream on another device. Do not reopen the stream in this callback.
+  virtual void OnErrorCallback(aaudio_result_t error) = 0;
+
+ protected:
+  virtual ~AAudioObserverInterface() {}
+};
+
+// Utility class which wraps the C-based AAudio API into a more handy C++ class
+// where the underlying resources (AAudioStreamBuilder and AAudioStream) are
+// encapsulated. User must set the direction (in or out) at construction since
+// it defines the stream type and the direction of the data flow in the
+// AAudioObserverInterface.
+//
+// AAudio is a new Android C API introduced in the Android O (26) release.
+// It is designed for high-performance audio applications that require low
+// latency. Applications communicate with AAudio by reading and writing data
+// to streams.
+//
+// Each stream is attached to a single audio device, where each audio device
+// has a unique ID. The ID can be used to bind an audio stream to a specific
+// audio device but this implementation lets AAudio choose the default primary
+// device instead (device selection takes place in Java). A stream can only
+// move data in one direction. When a stream is opened, Android checks to
+// ensure that the audio device and stream direction agree.
+class AAudioWrapper {
+ public:
+  AAudioWrapper(AudioManager* audio_manager,
+                aaudio_direction_t direction,
+                AAudioObserverInterface* observer);
+  ~AAudioWrapper();
+
+  bool Init();
+  bool Start();
+  bool Stop();
+
+  // For output streams: estimates latency between writing an audio frame to
+  // the output stream and the time that same frame is played out on the output
+  // audio device.
+  // For input streams: estimates latency between reading an audio frame from
+  // the input stream and the time that same frame was recorded on the input
+  // audio device.
+  double EstimateLatencyMillis() const;
+
+  // Increases the internal buffer size for output streams by one burst size to
+  // reduce the risk of underruns. Can be used while a stream is active.
+  bool IncreaseOutputBufferSize();
+
+  // Drains the recording stream of any existing data by reading from it until
+  // it's empty. Can be used to clear out old data before starting a new audio
+  // session.
+  void ClearInputStream(void* audio_data, int32_t num_frames);
+
+  AAudioObserverInterface* observer() const;
+  AudioParameters audio_parameters() const;
+  int32_t samples_per_frame() const;
+  int32_t buffer_size_in_frames() const;
+  int32_t buffer_capacity_in_frames() const;
+  int32_t device_id() const;
+  int32_t xrun_count() const;
+  int32_t format() const;
+  int32_t sample_rate() const;
+  int32_t channel_count() const;
+  int32_t frames_per_callback() const;
+  aaudio_sharing_mode_t sharing_mode() const;
+  aaudio_performance_mode_t performance_mode() const;
+  aaudio_stream_state_t stream_state() const;
+  int64_t frames_written() const;
+  int64_t frames_read() const;
+  aaudio_direction_t direction() const { return direction_; }
+  AAudioStream* stream() const { return stream_; }
+  int32_t frames_per_burst() const { return frames_per_burst_; }
+
+ private:
+  void SetStreamConfiguration(AAudioStreamBuilder* builder);
+  bool OpenStream(AAudioStreamBuilder* builder);
+  void CloseStream();
+  void LogStreamConfiguration();
+  void LogStreamState();
+  bool VerifyStreamConfiguration();
+  bool OptimizeBuffers();
+
+  SequenceChecker thread_checker_;
+  SequenceChecker aaudio_thread_checker_;
+  AudioParameters audio_parameters_;
+  const aaudio_direction_t direction_;
+  AAudioObserverInterface* observer_ = nullptr;
+  AAudioStream* stream_ = nullptr;
+  int32_t frames_per_burst_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_common.h b/third_party/libwebrtc/modules/audio_device/android/audio_common.h
new file mode 100644
index 0000000000..81ea733aa4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_common.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
+
+namespace webrtc {
+
+const int kDefaultSampleRate = 44100;
+// Delay estimates for the two different supported modes. These values are based
+// on real-time round-trip delay estimates on a large set of devices and they
+// are lower bounds since the filter length is 128 ms, so the AEC works for
+// delays in the range [50, ~170] ms and [150, ~270] ms. Note that, in most
+// cases, the lowest delay estimate will not be utilized since devices that
+// support low-latency output audio often supports HW AEC as well.
+const int kLowLatencyModeDelayEstimateInMilliseconds = 50;
+const int kHighLatencyModeDelayEstimateInMilliseconds = 150;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_device_template.h b/third_party/libwebrtc/modules/audio_device/android/audio_device_template.h
new file mode 100644
index 0000000000..999c5878c6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_device_template.h
@@ -0,0 +1,435 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+// InputType/OutputType can be any class that implements the capturing/rendering
+// part of the AudioDeviceGeneric API.
+// Construction and destruction must be done on one and the same thread. Each
+// internal implementation of InputType and OutputType will RTC_DCHECK if that
+// is not the case. All implemented methods must also be called on the same
+// thread. See comments in each InputType/OutputType class for more info.
+// It is possible to call the two static methods (SetAndroidAudioDeviceObjects
+// and ClearAndroidAudioDeviceObjects) from a different thread but both will
+// RTC_CHECK that the calling thread is attached to a Java VM.
+
+template <class InputType, class OutputType>
+class AudioDeviceTemplate : public AudioDeviceGeneric {
+ public:
+  AudioDeviceTemplate(AudioDeviceModule::AudioLayer audio_layer,
+                      AudioManager* audio_manager)
+      : audio_layer_(audio_layer),
+        audio_manager_(audio_manager),
+        output_(audio_manager_),
+        input_(audio_manager_),
+        initialized_(false) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_CHECK(audio_manager);
+    audio_manager_->SetActiveAudioLayer(audio_layer);
+  }
+
+  virtual ~AudioDeviceTemplate() { RTC_LOG(LS_INFO) << __FUNCTION__; }
+
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    audioLayer = audio_layer_;
+    return 0;
+  }
+
+  InitStatus Init() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    RTC_DCHECK(!initialized_);
+    if (!audio_manager_->Init()) {
+      return InitStatus::OTHER_ERROR;
+    }
+    if (output_.Init() != 0) {
+      audio_manager_->Close();
+      return InitStatus::PLAYOUT_ERROR;
+    }
+    if (input_.Init() != 0) {
+      output_.Terminate();
+      audio_manager_->Close();
+      return InitStatus::RECORDING_ERROR;
+    }
+    initialized_ = true;
+    return InitStatus::OK;
+  }
+
+  int32_t Terminate() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    int32_t err = input_.Terminate();
+    err |= output_.Terminate();
+    err |= !audio_manager_->Close();
+    initialized_ = false;
+    RTC_DCHECK_EQ(err, 0);
+    return err;
+  }
+
+  bool Initialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    return initialized_;
+  }
+
+  int16_t PlayoutDevices() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 1;
+  }
+
+  int16_t RecordingDevices() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 1;
+  }
+
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t SetPlayoutDevice(uint16_t index) override {
+    // OK to use but it has no effect currently since device selection is
+    // done using Andoid APIs instead.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 0;
+  }
+
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t SetRecordingDevice(uint16_t index) override {
+    // OK to use but it has no effect currently since device selection is
+    // done using Andoid APIs instead.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 0;
+  }
+
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t PlayoutIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    available = true;
+    return 0;
+  }
+
+  int32_t InitPlayout() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.InitPlayout();
+  }
+
+  bool PlayoutIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.PlayoutIsInitialized();
+  }
+
+  int32_t RecordingIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    available = true;
+    return 0;
+  }
+
+  int32_t InitRecording() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return input_.InitRecording();
+  }
+
+  bool RecordingIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return input_.RecordingIsInitialized();
+  }
+
+  int32_t StartPlayout() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (!audio_manager_->IsCommunicationModeEnabled()) {
+      RTC_LOG(LS_WARNING)
+          << "The application should use MODE_IN_COMMUNICATION audio mode!";
+    }
+    return output_.StartPlayout();
+  }
+
+  int32_t StopPlayout() override {
+    // Avoid using audio manger (JNI/Java cost) if playout was inactive.
+    if (!Playing())
+      return 0;
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    int32_t err = output_.StopPlayout();
+    return err;
+  }
+
+  bool Playing() const override {
+    RTC_LOG(LS_INFO) << __FUNCTION__;
+    return output_.Playing();
+  }
+
+  int32_t StartRecording() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (!audio_manager_->IsCommunicationModeEnabled()) {
+      RTC_LOG(LS_WARNING)
+          << "The application should use MODE_IN_COMMUNICATION audio mode!";
+    }
+    return input_.StartRecording();
+  }
+
+  int32_t StopRecording() override {
+    // Avoid using audio manger (JNI/Java cost) if recording was inactive.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (!Recording())
+      return 0;
+    int32_t err = input_.StopRecording();
+    return err;
+  }
+
+  bool Recording() const override { return input_.Recording(); }
+
+  int32_t InitSpeaker() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 0;
+  }
+
+  bool SpeakerIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return true;
+  }
+
+  int32_t InitMicrophone() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 0;
+  }
+
+  bool MicrophoneIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return true;
+  }
+
+  int32_t SpeakerVolumeIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.SpeakerVolumeIsAvailable(available);
+  }
+
+  int32_t SetSpeakerVolume(uint32_t volume) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.SetSpeakerVolume(volume);
+  }
+
+  int32_t SpeakerVolume(uint32_t& volume) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.SpeakerVolume(volume);
+  }
+
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.MaxSpeakerVolume(maxVolume);
+  }
+
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.MinSpeakerVolume(minVolume);
+  }
+
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override {
+    available = false;
+    return -1;
+  }
+
+  int32_t SetMicrophoneVolume(uint32_t volume) override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t MicrophoneVolume(uint32_t& volume) const override {
+    RTC_CHECK_NOTREACHED();
+    return -1;
+  }
+
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t SpeakerMuteIsAvailable(bool& available) override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t SetSpeakerMute(bool enable) override { RTC_CHECK_NOTREACHED(); }
+
+  int32_t SpeakerMute(bool& enabled) const override { RTC_CHECK_NOTREACHED(); }
+
+  int32_t MicrophoneMuteIsAvailable(bool& available) override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t SetMicrophoneMute(bool enable) override { RTC_CHECK_NOTREACHED(); }
+
+  int32_t MicrophoneMute(bool& enabled) const override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  // Returns true if the audio manager has been configured to support stereo
+  // and false otherwised. Default is mono.
+  int32_t StereoPlayoutIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    available = audio_manager_->IsStereoPlayoutSupported();
+    return 0;
+  }
+
+  int32_t SetStereoPlayout(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    bool available = audio_manager_->IsStereoPlayoutSupported();
+    // Android does not support changes between mono and stero on the fly.
+    // Instead, the native audio layer is configured via the audio manager
+    // to either support mono or stereo. It is allowed to call this method
+    // if that same state is not modified.
+    return (enable == available) ? 0 : -1;
+  }
+
+  int32_t StereoPlayout(bool& enabled) const override {
+    enabled = audio_manager_->IsStereoPlayoutSupported();
+    return 0;
+  }
+
+  int32_t StereoRecordingIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    available = audio_manager_->IsStereoRecordSupported();
+    return 0;
+  }
+
+  int32_t SetStereoRecording(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    bool available = audio_manager_->IsStereoRecordSupported();
+    // Android does not support changes between mono and stero on the fly.
+    // Instead, the native audio layer is configured via the audio manager
+    // to either support mono or stereo. It is allowed to call this method
+    // if that same state is not modified.
+    return (enable == available) ? 0 : -1;
+  }
+
+  int32_t StereoRecording(bool& enabled) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    enabled = audio_manager_->IsStereoRecordSupported();
+    return 0;
+  }
+
+  int32_t PlayoutDelay(uint16_t& delay_ms) const override {
+    // Best guess we can do is to use half of the estimated total delay.
+    delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
+    RTC_DCHECK_GT(delay_ms, 0);
+    return 0;
+  }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    output_.AttachAudioBuffer(audioBuffer);
+    input_.AttachAudioBuffer(audioBuffer);
+  }
+
+  // Returns true if the device both supports built in AEC and the device
+  // is not blacklisted.
+  // Currently, if OpenSL ES is used in both directions, this method will still
+  // report the correct value and it has the correct effect. As an example:
+  // a device supports built in AEC and this method returns true. Libjingle
+  // will then disable the WebRTC based AEC and that will work for all devices
+  // (mainly Nexus) even when OpenSL ES is used for input since our current
+  // implementation will enable built-in AEC by default also for OpenSL ES.
+  // The only "bad" thing that happens today is that when Libjingle calls
+  // OpenSLESRecorder::EnableBuiltInAEC() it will not have any real effect and
+  // a "Not Implemented" log will be filed. This non-perfect state will remain
+  // until I have added full support for audio effects based on OpenSL ES APIs.
+  bool BuiltInAECIsAvailable() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return audio_manager_->IsAcousticEchoCancelerSupported();
+  }
+
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  int32_t EnableBuiltInAEC(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    RTC_CHECK(BuiltInAECIsAvailable()) << "HW AEC is not available";
+    return input_.EnableBuiltInAEC(enable);
+  }
+
+  // Returns true if the device both supports built in AGC and the device
+  // is not blacklisted.
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  // In addition, see comments for BuiltInAECIsAvailable().
+  bool BuiltInAGCIsAvailable() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return audio_manager_->IsAutomaticGainControlSupported();
+  }
+
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  int32_t EnableBuiltInAGC(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    RTC_CHECK(BuiltInAGCIsAvailable()) << "HW AGC is not available";
+    return input_.EnableBuiltInAGC(enable);
+  }
+
+  // Returns true if the device both supports built in NS and the device
+  // is not blacklisted.
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  // In addition, see comments for BuiltInAECIsAvailable().
+  bool BuiltInNSIsAvailable() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return audio_manager_->IsNoiseSuppressorSupported();
+  }
+
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  int32_t EnableBuiltInNS(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    RTC_CHECK(BuiltInNSIsAvailable()) << "HW NS is not available";
+    return input_.EnableBuiltInNS(enable);
+  }
+
+ private:
+  SequenceChecker thread_checker_;
+
+  // Local copy of the audio layer set during construction of the
+  // AudioDeviceModuleImpl instance. Read only value.
+  const AudioDeviceModule::AudioLayer audio_layer_;
+
+  // Non-owning raw pointer to AudioManager instance given to use at
+  // construction. The real object is owned by AudioDeviceModuleImpl and the
+  // life time is the same as that of the AudioDeviceModuleImpl, hence there
+  // is no risk of reading a NULL pointer at any time in this class.
+  AudioManager* const audio_manager_;
+
+  OutputType output_;
+
+  InputType input_;
+
+  bool initialized_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_device_unittest.cc b/third_party/libwebrtc/modules/audio_device/android/audio_device_unittest.cc
new file mode 100644
index 0000000000..d9d52cdcdc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_device_unittest.cc
@@ -0,0 +1,1018 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/include/audio_device.h"
+
+#include <algorithm>
+#include <limits>
+#include <list>
+#include <memory>
+#include <numeric>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/android/build_info.h"
+#include "modules/audio_device/android/ensure_initialized.h"
+#include "modules/audio_device/audio_device_impl.h"
+#include "modules/audio_device/include/mock_audio_transport.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/event.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/time_utils.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using std::cout;
+using std::endl;
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Gt;
+using ::testing::Invoke;
+using ::testing::NiceMock;
+using ::testing::NotNull;
+using ::testing::Return;
+
+// #define ENABLE_DEBUG_PRINTF
+#ifdef ENABLE_DEBUG_PRINTF
+#define PRINTD(...) fprintf(stderr, __VA_ARGS__);
+#else
+#define PRINTD(...) ((void)0)
+#endif
+#define PRINT(...) fprintf(stderr, __VA_ARGS__);
+
+namespace webrtc {
+
+// Number of callbacks (input or output) the tests waits for before we set
+// an event indicating that the test was OK.
+static const size_t kNumCallbacks = 10;
+// Max amount of time we wait for an event to be set while counting callbacks.
+static constexpr TimeDelta kTestTimeOut = TimeDelta::Seconds(10);
+// Average number of audio callbacks per second assuming 10ms packet size.
+static const size_t kNumCallbacksPerSecond = 100;
+// Play out a test file during this time (unit is in seconds).
+static const int kFilePlayTimeInSec = 5;
+static const size_t kBitsPerSample = 16;
+static const size_t kBytesPerSample = kBitsPerSample / 8;
+// Run the full-duplex test during this time (unit is in seconds).
+// Note that first `kNumIgnoreFirstCallbacks` are ignored.
+static constexpr TimeDelta kFullDuplexTime = TimeDelta::Seconds(5);
+// Wait for the callback sequence to stabilize by ignoring this amount of the
+// initial callbacks (avoids initial FIFO access).
+// Only used in the RunPlayoutAndRecordingInFullDuplex test.
+static const size_t kNumIgnoreFirstCallbacks = 50;
+// Sets the number of impulses per second in the latency test.
+static const int kImpulseFrequencyInHz = 1;
+// Length of round-trip latency measurements. Number of transmitted impulses
+// is kImpulseFrequencyInHz * kMeasureLatencyTime - 1.
+static constexpr TimeDelta kMeasureLatencyTime = TimeDelta::Seconds(11);
+// Utilized in round-trip latency measurements to avoid capturing noise samples.
+static const int kImpulseThreshold = 1000;
+static const char kTag[] = "[..........] ";
+
+enum TransportType {
+  kPlayout = 0x1,
+  kRecording = 0x2,
+};
+
+// Interface for processing the audio stream. Real implementations can e.g.
+// run audio in loopback, read audio from a file or perform latency
+// measurements.
+class AudioStreamInterface {
+ public:
+  virtual void Write(const void* source, size_t num_frames) = 0;
+  virtual void Read(void* destination, size_t num_frames) = 0;
+
+ protected:
+  virtual ~AudioStreamInterface() {}
+};
+
+// Reads audio samples from a PCM file where the file is stored in memory at
+// construction.
+class FileAudioStream : public AudioStreamInterface {
+ public:
+  FileAudioStream(size_t num_callbacks,
+                  absl::string_view file_name,
+                  int sample_rate)
+      : file_size_in_bytes_(0), sample_rate_(sample_rate), file_pos_(0) {
+    file_size_in_bytes_ = test::GetFileSize(file_name);
+    sample_rate_ = sample_rate;
+    EXPECT_GE(file_size_in_callbacks(), num_callbacks)
+        << "Size of test file is not large enough to last during the test.";
+    const size_t num_16bit_samples =
+        test::GetFileSize(file_name) / kBytesPerSample;
+    file_.reset(new int16_t[num_16bit_samples]);
+    FILE* audio_file = fopen(std::string(file_name).c_str(), "rb");
+    EXPECT_NE(audio_file, nullptr);
+    size_t num_samples_read =
+        fread(file_.get(), sizeof(int16_t), num_16bit_samples, audio_file);
+    EXPECT_EQ(num_samples_read, num_16bit_samples);
+    fclose(audio_file);
+  }
+
+  // AudioStreamInterface::Write() is not implemented.
+  void Write(const void* source, size_t num_frames) override {}
+
+  // Read samples from file stored in memory (at construction) and copy
+  // `num_frames` (<=> 10ms) to the `destination` byte buffer.
+  void Read(void* destination, size_t num_frames) override {
+    memcpy(destination, static_cast<int16_t*>(&file_[file_pos_]),
+           num_frames * sizeof(int16_t));
+    file_pos_ += num_frames;
+  }
+
+  int file_size_in_seconds() const {
+    return static_cast<int>(file_size_in_bytes_ /
+                            (kBytesPerSample * sample_rate_));
+  }
+  size_t file_size_in_callbacks() const {
+    return file_size_in_seconds() * kNumCallbacksPerSecond;
+  }
+
+ private:
+  size_t file_size_in_bytes_;
+  int sample_rate_;
+  std::unique_ptr<int16_t[]> file_;
+  size_t file_pos_;
+};
+
+// Simple first in first out (FIFO) class that wraps a list of 16-bit audio
+// buffers of fixed size and allows Write and Read operations. The idea is to
+// store recorded audio buffers (using Write) and then read (using Read) these
+// stored buffers with as short delay as possible when the audio layer needs
+// data to play out. The number of buffers in the FIFO will stabilize under
+// normal conditions since there will be a balance between Write and Read calls.
+// The container is a std::list container and access is protected with a lock
+// since both sides (playout and recording) are driven by its own thread.
+class FifoAudioStream : public AudioStreamInterface {
+ public:
+  explicit FifoAudioStream(size_t frames_per_buffer)
+      : frames_per_buffer_(frames_per_buffer),
+        bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
+        fifo_(new AudioBufferList),
+        largest_size_(0),
+        total_written_elements_(0),
+        write_count_(0) {
+    EXPECT_NE(fifo_.get(), nullptr);
+  }
+
+  ~FifoAudioStream() { Flush(); }
+
+  // Allocate new memory, copy `num_frames` samples from `source` into memory
+  // and add pointer to the memory location to end of the list.
+  // Increases the size of the FIFO by one element.
+  void Write(const void* source, size_t num_frames) override {
+    ASSERT_EQ(num_frames, frames_per_buffer_);
+    PRINTD("+");
+    if (write_count_++ < kNumIgnoreFirstCallbacks) {
+      return;
+    }
+    int16_t* memory = new int16_t[frames_per_buffer_];
+    memcpy(static_cast<int16_t*>(&memory[0]), source, bytes_per_buffer_);
+    MutexLock lock(&lock_);
+    fifo_->push_back(memory);
+    const size_t size = fifo_->size();
+    if (size > largest_size_) {
+      largest_size_ = size;
+      PRINTD("(%zu)", largest_size_);
+    }
+    total_written_elements_ += size;
+  }
+
+  // Read pointer to data buffer from front of list, copy `num_frames` of stored
+  // data into `destination` and delete the utilized memory allocation.
+  // Decreases the size of the FIFO by one element.
+  void Read(void* destination, size_t num_frames) override {
+    ASSERT_EQ(num_frames, frames_per_buffer_);
+    PRINTD("-");
+    MutexLock lock(&lock_);
+    if (fifo_->empty()) {
+      memset(destination, 0, bytes_per_buffer_);
+    } else {
+      int16_t* memory = fifo_->front();
+      fifo_->pop_front();
+      memcpy(destination, static_cast<int16_t*>(&memory[0]), bytes_per_buffer_);
+      delete memory;
+    }
+  }
+
+  size_t size() const { return fifo_->size(); }
+
+  size_t largest_size() const { return largest_size_; }
+
+  size_t average_size() const {
+    return (total_written_elements_ == 0)
+               ? 0.0
+               : 0.5 + static_cast<float>(total_written_elements_) /
+                           (write_count_ - kNumIgnoreFirstCallbacks);
+  }
+
+ private:
+  void Flush() {
+    for (auto it = fifo_->begin(); it != fifo_->end(); ++it) {
+      delete *it;
+    }
+    fifo_->clear();
+  }
+
+  using AudioBufferList = std::list<int16_t*>;
+  Mutex lock_;
+  const size_t frames_per_buffer_;
+  const size_t bytes_per_buffer_;
+  std::unique_ptr<AudioBufferList> fifo_;
+  size_t largest_size_;
+  size_t total_written_elements_;
+  size_t write_count_;
+};
+
+// Inserts periodic impulses and measures the latency between the time of
+// transmission and time of receiving the same impulse.
+// Usage requires a special hardware called Audio Loopback Dongle.
+// See http://source.android.com/devices/audio/loopback.html for details.
+class LatencyMeasuringAudioStream : public AudioStreamInterface {
+ public:
+  explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
+      : frames_per_buffer_(frames_per_buffer),
+        bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
+        play_count_(0),
+        rec_count_(0),
+        pulse_time_(0) {}
+
+  // Insert periodic impulses in first two samples of `destination`.
+  void Read(void* destination, size_t num_frames) override {
+    ASSERT_EQ(num_frames, frames_per_buffer_);
+    if (play_count_ == 0) {
+      PRINT("[");
+    }
+    play_count_++;
+    memset(destination, 0, bytes_per_buffer_);
+    if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
+      if (pulse_time_ == 0) {
+        pulse_time_ = rtc::TimeMillis();
+      }
+      PRINT(".");
+      const int16_t impulse = std::numeric_limits<int16_t>::max();
+      int16_t* ptr16 = static_cast<int16_t*>(destination);
+      for (size_t i = 0; i < 2; ++i) {
+        ptr16[i] = impulse;
+      }
+    }
+  }
+
+  // Detect received impulses in `source`, derive time between transmission and
+  // detection and add the calculated delay to list of latencies.
+  void Write(const void* source, size_t num_frames) override {
+    ASSERT_EQ(num_frames, frames_per_buffer_);
+    rec_count_++;
+    if (pulse_time_ == 0) {
+      // Avoid detection of new impulse response until a new impulse has
+      // been transmitted (sets `pulse_time_` to value larger than zero).
+      return;
+    }
+    const int16_t* ptr16 = static_cast<const int16_t*>(source);
+    std::vector<int16_t> vec(ptr16, ptr16 + num_frames);
+    // Find max value in the audio buffer.
+    int max = *std::max_element(vec.begin(), vec.end());
+    // Find index (element position in vector) of the max element.
+    int index_of_max =
+        std::distance(vec.begin(), std::find(vec.begin(), vec.end(), max));
+    if (max > kImpulseThreshold) {
+      PRINTD("(%d,%d)", max, index_of_max);
+      int64_t now_time = rtc::TimeMillis();
+      int extra_delay = IndexToMilliseconds(static_cast<double>(index_of_max));
+      PRINTD("[%d]", static_cast<int>(now_time - pulse_time_));
+      PRINTD("[%d]", extra_delay);
+      // Total latency is the difference between transmit time and detection
+      // tome plus the extra delay within the buffer in which we detected the
+      // received impulse. It is transmitted at sample 0 but can be received
+      // at sample N where N > 0. The term `extra_delay` accounts for N and it
+      // is a value between 0 and 10ms.
+      latencies_.push_back(now_time - pulse_time_ + extra_delay);
+      pulse_time_ = 0;
+    } else {
+      PRINTD("-");
+    }
+  }
+
+  size_t num_latency_values() const { return latencies_.size(); }
+
+  int min_latency() const {
+    if (latencies_.empty())
+      return 0;
+    return *std::min_element(latencies_.begin(), latencies_.end());
+  }
+
+  int max_latency() const {
+    if (latencies_.empty())
+      return 0;
+    return *std::max_element(latencies_.begin(), latencies_.end());
+  }
+
+  int average_latency() const {
+    if (latencies_.empty())
+      return 0;
+    return 0.5 + static_cast<double>(
+                     std::accumulate(latencies_.begin(), latencies_.end(), 0)) /
+                     latencies_.size();
+  }
+
+  void PrintResults() const {
+    PRINT("] ");
+    for (auto it = latencies_.begin(); it != latencies_.end(); ++it) {
+      PRINT("%d ", *it);
+    }
+    PRINT("\n");
+    PRINT("%s[min, max, avg]=[%d, %d, %d] ms\n", kTag, min_latency(),
+          max_latency(), average_latency());
+  }
+
+  int IndexToMilliseconds(double index) const {
+    return static_cast<int>(10.0 * (index / frames_per_buffer_) + 0.5);
+  }
+
+ private:
+  const size_t frames_per_buffer_;
+  const size_t bytes_per_buffer_;
+  size_t play_count_;
+  size_t rec_count_;
+  int64_t pulse_time_;
+  std::vector<int> latencies_;
+};
+
+// Mocks the AudioTransport object and proxies actions for the two callbacks
+// (RecordedDataIsAvailable and NeedMorePlayData) to different implementations
+// of AudioStreamInterface.
+class MockAudioTransportAndroid : public test::MockAudioTransport {
+ public:
+  explicit MockAudioTransportAndroid(int type)
+      : num_callbacks_(0),
+        type_(type),
+        play_count_(0),
+        rec_count_(0),
+        audio_stream_(nullptr) {}
+
+  virtual ~MockAudioTransportAndroid() {}
+
+  // Set default actions of the mock object. We are delegating to fake
+  // implementations (of AudioStreamInterface) here.
+  void HandleCallbacks(rtc::Event* test_is_done,
+                       AudioStreamInterface* audio_stream,
+                       int num_callbacks) {
+    test_is_done_ = test_is_done;
+    audio_stream_ = audio_stream;
+    num_callbacks_ = num_callbacks;
+    if (play_mode()) {
+      ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _))
+          .WillByDefault(
+              Invoke(this, &MockAudioTransportAndroid::RealNeedMorePlayData));
+    }
+    if (rec_mode()) {
+      ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _))
+          .WillByDefault(Invoke(
+              this, &MockAudioTransportAndroid::RealRecordedDataIsAvailable));
+    }
+  }
+
+  int32_t RealRecordedDataIsAvailable(const void* audioSamples,
+                                      const size_t nSamples,
+                                      const size_t nBytesPerSample,
+                                      const size_t nChannels,
+                                      const uint32_t samplesPerSec,
+                                      const uint32_t totalDelayMS,
+                                      const int32_t clockDrift,
+                                      const uint32_t currentMicLevel,
+                                      const bool keyPressed,
+                                      uint32_t& newMicLevel) {  // NOLINT
+    EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
+    rec_count_++;
+    // Process the recorded audio stream if an AudioStreamInterface
+    // implementation exists.
+    if (audio_stream_) {
+      audio_stream_->Write(audioSamples, nSamples);
+    }
+    if (ReceivedEnoughCallbacks()) {
+      test_is_done_->Set();
+    }
+    return 0;
+  }
+
+  int32_t RealNeedMorePlayData(const size_t nSamples,
+                               const size_t nBytesPerSample,
+                               const size_t nChannels,
+                               const uint32_t samplesPerSec,
+                               void* audioSamples,
+                               size_t& nSamplesOut,  // NOLINT
+                               int64_t* elapsed_time_ms,
+                               int64_t* ntp_time_ms) {
+    EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
+    play_count_++;
+    nSamplesOut = nSamples;
+    // Read (possibly processed) audio stream samples to be played out if an
+    // AudioStreamInterface implementation exists.
+    if (audio_stream_) {
+      audio_stream_->Read(audioSamples, nSamples);
+    }
+    if (ReceivedEnoughCallbacks()) {
+      test_is_done_->Set();
+    }
+    return 0;
+  }
+
+  bool ReceivedEnoughCallbacks() {
+    bool recording_done = false;
+    if (rec_mode())
+      recording_done = rec_count_ >= num_callbacks_;
+    else
+      recording_done = true;
+
+    bool playout_done = false;
+    if (play_mode())
+      playout_done = play_count_ >= num_callbacks_;
+    else
+      playout_done = true;
+
+    return recording_done && playout_done;
+  }
+
+  bool play_mode() const { return type_ & kPlayout; }
+  bool rec_mode() const { return type_ & kRecording; }
+
+ private:
+  rtc::Event* test_is_done_;
+  size_t num_callbacks_;
+  int type_;
+  size_t play_count_;
+  size_t rec_count_;
+  AudioStreamInterface* audio_stream_;
+  std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream_;
+};
+
+// AudioDeviceTest test fixture.
+class AudioDeviceTest : public ::testing::Test {
+ protected:
+  AudioDeviceTest() : task_queue_factory_(CreateDefaultTaskQueueFactory()) {
+    // One-time initialization of JVM and application context. Ensures that we
+    // can do calls between C++ and Java. Initializes both Java and OpenSL ES
+    // implementations.
+    webrtc::audiodevicemodule::EnsureInitialized();
+    // Creates an audio device using a default audio layer.
+    audio_device_ = CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
+    EXPECT_NE(audio_device_.get(), nullptr);
+    EXPECT_EQ(0, audio_device_->Init());
+    playout_parameters_ = audio_manager()->GetPlayoutAudioParameters();
+    record_parameters_ = audio_manager()->GetRecordAudioParameters();
+    build_info_.reset(new BuildInfo());
+  }
+  virtual ~AudioDeviceTest() { EXPECT_EQ(0, audio_device_->Terminate()); }
+
+  int playout_sample_rate() const { return playout_parameters_.sample_rate(); }
+  int record_sample_rate() const { return record_parameters_.sample_rate(); }
+  size_t playout_channels() const { return playout_parameters_.channels(); }
+  size_t record_channels() const { return record_parameters_.channels(); }
+  size_t playout_frames_per_10ms_buffer() const {
+    return playout_parameters_.frames_per_10ms_buffer();
+  }
+  size_t record_frames_per_10ms_buffer() const {
+    return record_parameters_.frames_per_10ms_buffer();
+  }
+
+  int total_delay_ms() const {
+    return audio_manager()->GetDelayEstimateInMilliseconds();
+  }
+
+  rtc::scoped_refptr<AudioDeviceModule> audio_device() const {
+    return audio_device_;
+  }
+
+  AudioDeviceModuleImpl* audio_device_impl() const {
+    return static_cast<AudioDeviceModuleImpl*>(audio_device_.get());
+  }
+
+  AudioManager* audio_manager() const {
+    return audio_device_impl()->GetAndroidAudioManagerForTest();
+  }
+
+  AudioManager* GetAudioManager(AudioDeviceModule* adm) const {
+    return static_cast<AudioDeviceModuleImpl*>(adm)
+        ->GetAndroidAudioManagerForTest();
+  }
+
+  AudioDeviceBuffer* audio_device_buffer() const {
+    return audio_device_impl()->GetAudioDeviceBuffer();
+  }
+
+  rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
+      AudioDeviceModule::AudioLayer audio_layer) {
+    rtc::scoped_refptr<AudioDeviceModule> module(
+        AudioDeviceModule::Create(audio_layer, task_queue_factory_.get()));
+    return module;
+  }
+
+  // Returns file name relative to the resource root given a sample rate.
+  std::string GetFileName(int sample_rate) {
+    EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100);
+    char fname[64];
+    snprintf(fname, sizeof(fname), "audio_device/audio_short%d",
+             sample_rate / 1000);
+    std::string file_name(webrtc::test::ResourcePath(fname, "pcm"));
+    EXPECT_TRUE(test::FileExists(file_name));
+#ifdef ENABLE_PRINTF
+    PRINT("file name: %s\n", file_name.c_str());
+    const size_t bytes = test::GetFileSize(file_name);
+    PRINT("file size: %zu [bytes]\n", bytes);
+    PRINT("file size: %zu [samples]\n", bytes / kBytesPerSample);
+    const int seconds =
+        static_cast<int>(bytes / (sample_rate * kBytesPerSample));
+    PRINT("file size: %d [secs]\n", seconds);
+    PRINT("file size: %zu [callbacks]\n", seconds * kNumCallbacksPerSecond);
+#endif
+    return file_name;
+  }
+
+  AudioDeviceModule::AudioLayer GetActiveAudioLayer() const {
+    AudioDeviceModule::AudioLayer audio_layer;
+    EXPECT_EQ(0, audio_device()->ActiveAudioLayer(&audio_layer));
+    return audio_layer;
+  }
+
+  int TestDelayOnAudioLayer(
+      const AudioDeviceModule::AudioLayer& layer_to_test) {
+    rtc::scoped_refptr<AudioDeviceModule> audio_device;
+    audio_device = CreateAudioDevice(layer_to_test);
+    EXPECT_NE(audio_device.get(), nullptr);
+    AudioManager* audio_manager = GetAudioManager(audio_device.get());
+    EXPECT_NE(audio_manager, nullptr);
+    return audio_manager->GetDelayEstimateInMilliseconds();
+  }
+
+  AudioDeviceModule::AudioLayer TestActiveAudioLayer(
+      const AudioDeviceModule::AudioLayer& layer_to_test) {
+    rtc::scoped_refptr<AudioDeviceModule> audio_device;
+    audio_device = CreateAudioDevice(layer_to_test);
+    EXPECT_NE(audio_device.get(), nullptr);
+    AudioDeviceModule::AudioLayer active;
+    EXPECT_EQ(0, audio_device->ActiveAudioLayer(&active));
+    return active;
+  }
+
+  bool DisableTestForThisDevice(absl::string_view model) {
+    return (build_info_->GetDeviceModel() == model);
+  }
+
+  // Volume control is currently only supported for the Java output audio layer.
+  // For OpenSL ES, the internal stream volume is always on max level and there
+  // is no need for this test to set it to max.
+  bool AudioLayerSupportsVolumeControl() const {
+    return GetActiveAudioLayer() == AudioDeviceModule::kAndroidJavaAudio;
+  }
+
+  void SetMaxPlayoutVolume() {
+    if (!AudioLayerSupportsVolumeControl())
+      return;
+    uint32_t max_volume;
+    EXPECT_EQ(0, audio_device()->MaxSpeakerVolume(&max_volume));
+    EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume));
+  }
+
+  void DisableBuiltInAECIfAvailable() {
+    if (audio_device()->BuiltInAECIsAvailable()) {
+      EXPECT_EQ(0, audio_device()->EnableBuiltInAEC(false));
+    }
+  }
+
+  void StartPlayout() {
+    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
+    EXPECT_FALSE(audio_device()->Playing());
+    EXPECT_EQ(0, audio_device()->InitPlayout());
+    EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
+    EXPECT_EQ(0, audio_device()->StartPlayout());
+    EXPECT_TRUE(audio_device()->Playing());
+  }
+
+  void StopPlayout() {
+    EXPECT_EQ(0, audio_device()->StopPlayout());
+    EXPECT_FALSE(audio_device()->Playing());
+    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
+  }
+
+  void StartRecording() {
+    EXPECT_FALSE(audio_device()->RecordingIsInitialized());
+    EXPECT_FALSE(audio_device()->Recording());
+    EXPECT_EQ(0, audio_device()->InitRecording());
+    EXPECT_TRUE(audio_device()->RecordingIsInitialized());
+    EXPECT_EQ(0, audio_device()->StartRecording());
+    EXPECT_TRUE(audio_device()->Recording());
+  }
+
+  void StopRecording() {
+    EXPECT_EQ(0, audio_device()->StopRecording());
+    EXPECT_FALSE(audio_device()->Recording());
+  }
+
+  int GetMaxSpeakerVolume() const {
+    uint32_t max_volume(0);
+    EXPECT_EQ(0, audio_device()->MaxSpeakerVolume(&max_volume));
+    return max_volume;
+  }
+
+  int GetMinSpeakerVolume() const {
+    uint32_t min_volume(0);
+    EXPECT_EQ(0, audio_device()->MinSpeakerVolume(&min_volume));
+    return min_volume;
+  }
+
+  int GetSpeakerVolume() const {
+    uint32_t volume(0);
+    EXPECT_EQ(0, audio_device()->SpeakerVolume(&volume));
+    return volume;
+  }
+
+  rtc::Event test_is_done_;
+  std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+  rtc::scoped_refptr<AudioDeviceModule> audio_device_;
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+  std::unique_ptr<BuildInfo> build_info_;
+};
+
+TEST_F(AudioDeviceTest, ConstructDestruct) {
+  // Using the test fixture to create and destruct the audio device module.
+}
+
+// We always ask for a default audio layer when the ADM is constructed. But the
+// ADM will then internally set the best suitable combination of audio layers,
+// for input and output based on if low-latency output and/or input audio in
+// combination with OpenSL ES is supported or not. This test ensures that the
+// correct selection is done.
+TEST_F(AudioDeviceTest, VerifyDefaultAudioLayer) {
+  const AudioDeviceModule::AudioLayer audio_layer = GetActiveAudioLayer();
+  bool low_latency_output = audio_manager()->IsLowLatencyPlayoutSupported();
+  bool low_latency_input = audio_manager()->IsLowLatencyRecordSupported();
+  bool aaudio = audio_manager()->IsAAudioSupported();
+  AudioDeviceModule::AudioLayer expected_audio_layer;
+  if (aaudio) {
+    expected_audio_layer = AudioDeviceModule::kAndroidAAudioAudio;
+  } else if (low_latency_output && low_latency_input) {
+    expected_audio_layer = AudioDeviceModule::kAndroidOpenSLESAudio;
+  } else if (low_latency_output && !low_latency_input) {
+    expected_audio_layer =
+        AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
+  } else {
+    expected_audio_layer = AudioDeviceModule::kAndroidJavaAudio;
+  }
+  EXPECT_EQ(expected_audio_layer, audio_layer);
+}
+
+// Verify that it is possible to explicitly create the two types of supported
+// ADMs. These two tests overrides the default selection of native audio layer
+// by ignoring if the device supports low-latency output or not.
+TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForCombinedJavaOpenSLCombo) {
+  AudioDeviceModule::AudioLayer expected_layer =
+      AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
+  AudioDeviceModule::AudioLayer active_layer =
+      TestActiveAudioLayer(expected_layer);
+  EXPECT_EQ(expected_layer, active_layer);
+}
+
+TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForJavaInBothDirections) {
+  AudioDeviceModule::AudioLayer expected_layer =
+      AudioDeviceModule::kAndroidJavaAudio;
+  AudioDeviceModule::AudioLayer active_layer =
+      TestActiveAudioLayer(expected_layer);
+  EXPECT_EQ(expected_layer, active_layer);
+}
+
+TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForOpenSLInBothDirections) {
+  AudioDeviceModule::AudioLayer expected_layer =
+      AudioDeviceModule::kAndroidOpenSLESAudio;
+  AudioDeviceModule::AudioLayer active_layer =
+      TestActiveAudioLayer(expected_layer);
+  EXPECT_EQ(expected_layer, active_layer);
+}
+
+// TODO(bugs.webrtc.org/8914)
+#if !defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
+  DISABLED_CorrectAudioLayerIsUsedForAAudioInBothDirections
+#else
+#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
+  CorrectAudioLayerIsUsedForAAudioInBothDirections
+#endif
+TEST_F(AudioDeviceTest,
+       MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections) {
+  AudioDeviceModule::AudioLayer expected_layer =
+      AudioDeviceModule::kAndroidAAudioAudio;
+  AudioDeviceModule::AudioLayer active_layer =
+      TestActiveAudioLayer(expected_layer);
+  EXPECT_EQ(expected_layer, active_layer);
+}
+
+// TODO(bugs.webrtc.org/8914)
+#if !defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+#define MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo \
+  DISABLED_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo
+#else
+#define MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo \
+  CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo
+#endif
+TEST_F(AudioDeviceTest,
+       MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo) {
+  AudioDeviceModule::AudioLayer expected_layer =
+      AudioDeviceModule::kAndroidJavaInputAndAAudioOutputAudio;
+  AudioDeviceModule::AudioLayer active_layer =
+      TestActiveAudioLayer(expected_layer);
+  EXPECT_EQ(expected_layer, active_layer);
+}
+
+// The Android ADM supports two different delay reporting modes. One for the
+// low-latency output path (in combination with OpenSL ES), and one for the
+// high-latency output path (Java backends in both directions). These two tests
+// verifies that the audio manager reports correct delay estimate given the
+// selected audio layer. Note that, this delay estimate will only be utilized
+// if the HW AEC is disabled.
+TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForHighLatencyOutputPath) {
+  EXPECT_EQ(kHighLatencyModeDelayEstimateInMilliseconds,
+            TestDelayOnAudioLayer(AudioDeviceModule::kAndroidJavaAudio));
+}
+
+TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForLowLatencyOutputPath) {
+  EXPECT_EQ(kLowLatencyModeDelayEstimateInMilliseconds,
+            TestDelayOnAudioLayer(
+                AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio));
+}
+
+// Ensure that the ADM internal audio device buffer is configured to use the
+// correct set of parameters.
+TEST_F(AudioDeviceTest, VerifyAudioDeviceBufferParameters) {
+  EXPECT_EQ(playout_parameters_.sample_rate(),
+            static_cast<int>(audio_device_buffer()->PlayoutSampleRate()));
+  EXPECT_EQ(record_parameters_.sample_rate(),
+            static_cast<int>(audio_device_buffer()->RecordingSampleRate()));
+  EXPECT_EQ(playout_parameters_.channels(),
+            audio_device_buffer()->PlayoutChannels());
+  EXPECT_EQ(record_parameters_.channels(),
+            audio_device_buffer()->RecordingChannels());
+}
+
+TEST_F(AudioDeviceTest, InitTerminate) {
+  // Initialization is part of the test fixture.
+  EXPECT_TRUE(audio_device()->Initialized());
+  EXPECT_EQ(0, audio_device()->Terminate());
+  EXPECT_FALSE(audio_device()->Initialized());
+}
+
+TEST_F(AudioDeviceTest, Devices) {
+  // Device enumeration is not supported. Verify fixed values only.
+  EXPECT_EQ(1, audio_device()->PlayoutDevices());
+  EXPECT_EQ(1, audio_device()->RecordingDevices());
+}
+
+TEST_F(AudioDeviceTest, SpeakerVolumeShouldBeAvailable) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
+  bool available;
+  EXPECT_EQ(0, audio_device()->SpeakerVolumeIsAvailable(&available));
+  EXPECT_TRUE(available);
+}
+
+TEST_F(AudioDeviceTest, MaxSpeakerVolumeIsPositive) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
+  StartPlayout();
+  EXPECT_GT(GetMaxSpeakerVolume(), 0);
+  StopPlayout();
+}
+
+TEST_F(AudioDeviceTest, MinSpeakerVolumeIsZero) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
+  EXPECT_EQ(GetMinSpeakerVolume(), 0);
+}
+
+TEST_F(AudioDeviceTest, DefaultSpeakerVolumeIsWithinMinMax) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
+  const int default_volume = GetSpeakerVolume();
+  EXPECT_GE(default_volume, GetMinSpeakerVolume());
+  EXPECT_LE(default_volume, GetMaxSpeakerVolume());
+}
+
+TEST_F(AudioDeviceTest, SetSpeakerVolumeActuallySetsVolume) {
+  // The OpenSL ES output audio path does not support volume control.
+  if (!AudioLayerSupportsVolumeControl())
+    return;
+  const int default_volume = GetSpeakerVolume();
+  const int max_volume = GetMaxSpeakerVolume();
+  EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume));
+  int new_volume = GetSpeakerVolume();
+  EXPECT_EQ(new_volume, max_volume);
+  EXPECT_EQ(0, audio_device()->SetSpeakerVolume(default_volume));
+}
+
+// Tests that playout can be initiated, started and stopped. No audio callback
+// is registered in this test.
+TEST_F(AudioDeviceTest, StartStopPlayout) {
+  StartPlayout();
+  StopPlayout();
+  StartPlayout();
+  StopPlayout();
+}
+
+// Tests that recording can be initiated, started and stopped. No audio callback
+// is registered in this test.
+TEST_F(AudioDeviceTest, StartStopRecording) {
+  StartRecording();
+  StopRecording();
+  StartRecording();
+  StopRecording();
+}
+
+// Verify that calling StopPlayout() will leave us in an uninitialized state
+// which will require a new call to InitPlayout(). This test does not call
+// StartPlayout() while being uninitialized since doing so will hit a
+// RTC_DCHECK and death tests are not supported on Android.
+TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) {
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  EXPECT_EQ(0, audio_device()->StartPlayout());
+  EXPECT_EQ(0, audio_device()->StopPlayout());
+  EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
+}
+
+// Verify that calling StopRecording() will leave us in an uninitialized state
+// which will require a new call to InitRecording(). This test does not call
+// StartRecording() while being uninitialized since doing so will hit a
+// RTC_DCHECK and death tests are not supported on Android.
+TEST_F(AudioDeviceTest, StopRecordingRequiresInitToRestart) {
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  EXPECT_EQ(0, audio_device()->StartRecording());
+  EXPECT_EQ(0, audio_device()->StopRecording());
+  EXPECT_FALSE(audio_device()->RecordingIsInitialized());
+}
+
+// Start playout and verify that the native audio layer starts asking for real
+// audio samples to play out using the NeedMorePlayData callback.
+TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
+  MockAudioTransportAndroid mock(kPlayout);
+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
+  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
+                                     kBytesPerSample, playout_channels(),
+                                     playout_sample_rate(), NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  test_is_done_.Wait(kTestTimeOut);
+  StopPlayout();
+}
+
+// Start recording and verify that the native audio layer starts feeding real
+// audio samples via the RecordedDataIsAvailable callback.
+// TODO(henrika): investigate if it is possible to perform a sanity check of
+// delay estimates as well (argument #6).
+TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
+  MockAudioTransportAndroid mock(kRecording);
+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
+  EXPECT_CALL(
+      mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
+                                    kBytesPerSample, record_channels(),
+                                    record_sample_rate(), _, 0, 0, false, _, _))
+      .Times(AtLeast(kNumCallbacks));
+
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartRecording();
+  test_is_done_.Wait(kTestTimeOut);
+  StopRecording();
+}
+
+// Start playout and recording (full-duplex audio) and verify that audio is
+// active in both directions.
+TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
+  MockAudioTransportAndroid mock(kPlayout | kRecording);
+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
+  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
+                                     kBytesPerSample, playout_channels(),
+                                     playout_sample_rate(), NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_CALL(
+      mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
+                                    kBytesPerSample, record_channels(),
+                                    record_sample_rate(), _, 0, 0, false, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  StartRecording();
+  test_is_done_.Wait(kTestTimeOut);
+  StopRecording();
+  StopPlayout();
+}
+
+// Start playout and read audio from an external PCM file when the audio layer
+// asks for data to play out. Real audio is played out in this test but it does
+// not contain any explicit verification that the audio quality is perfect.
+TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
+  // TODO(henrika): extend test when mono output is supported.
+  EXPECT_EQ(1u, playout_channels());
+  NiceMock<MockAudioTransportAndroid> mock(kPlayout);
+  const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
+  std::string file_name = GetFileName(playout_sample_rate());
+  std::unique_ptr<FileAudioStream> file_audio_stream(
+      new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
+  mock.HandleCallbacks(&test_is_done_, file_audio_stream.get(), num_callbacks);
+  // SetMaxPlayoutVolume();
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  test_is_done_.Wait(kTestTimeOut);
+  StopPlayout();
+}
+
+// Start playout and recording and store recorded data in an intermediate FIFO
+// buffer from which the playout side then reads its samples in the same order
+// as they were stored. Under ideal circumstances, a callback sequence would
+// look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-'
+// means 'packet played'. Under such conditions, the FIFO would only contain
+// one packet on average. However, under more realistic conditions, the size
+// of the FIFO will vary more due to an unbalance between the two sides.
+// This test tries to verify that the device maintains a balanced callback-
+// sequence by running in loopback for ten seconds while measuring the size
+// (max and average) of the FIFO. The size of the FIFO is increased by the
+// recording side and decreased by the playout side.
+// TODO(henrika): tune the final test parameters after running tests on several
+// different devices.
+// Disabling this test on bots since it is difficult to come up with a robust
+// test condition that all worked as intended. The main issue is that, when
+// swarming is used, an initial latency can be built up when the both sides
+// starts at different times. Hence, the test can fail even if audio works
+// as intended. Keeping the test so it can be enabled manually.
+// http://bugs.webrtc.org/7744
+TEST_F(AudioDeviceTest, DISABLED_RunPlayoutAndRecordingInFullDuplex) {
+  EXPECT_EQ(record_channels(), playout_channels());
+  EXPECT_EQ(record_sample_rate(), playout_sample_rate());
+  NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
+  std::unique_ptr<FifoAudioStream> fifo_audio_stream(
+      new FifoAudioStream(playout_frames_per_10ms_buffer()));
+  mock.HandleCallbacks(&test_is_done_, fifo_audio_stream.get(),
+                       kFullDuplexTime.seconds() * kNumCallbacksPerSecond);
+  SetMaxPlayoutVolume();
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartRecording();
+  StartPlayout();
+  test_is_done_.Wait(std::max(kTestTimeOut, kFullDuplexTime));
+  StopPlayout();
+  StopRecording();
+
+  // These thresholds are set rather high to accomodate differences in hardware
+  // in several devices, so this test can be used in swarming.
+  // See http://bugs.webrtc.org/6464
+  EXPECT_LE(fifo_audio_stream->average_size(), 60u);
+  EXPECT_LE(fifo_audio_stream->largest_size(), 70u);
+}
+
+// Measures loopback latency and reports the min, max and average values for
+// a full duplex audio session.
+// The latency is measured like so:
+// - Insert impulses periodically on the output side.
+// - Detect the impulses on the input side.
+// - Measure the time difference between the transmit time and receive time.
+// - Store time differences in a vector and calculate min, max and average.
+// This test requires a special hardware called Audio Loopback Dongle.
+// See http://source.android.com/devices/audio/loopback.html for details.
+TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
+  EXPECT_EQ(record_channels(), playout_channels());
+  EXPECT_EQ(record_sample_rate(), playout_sample_rate());
+  NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
+  std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
+      new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
+  mock.HandleCallbacks(&test_is_done_, latency_audio_stream.get(),
+                       kMeasureLatencyTime.seconds() * kNumCallbacksPerSecond);
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  SetMaxPlayoutVolume();
+  DisableBuiltInAECIfAvailable();
+  StartRecording();
+  StartPlayout();
+  test_is_done_.Wait(std::max(kTestTimeOut, kMeasureLatencyTime));
+  StopPlayout();
+  StopRecording();
+  // Verify that the correct number of transmitted impulses are detected.
+  EXPECT_EQ(latency_audio_stream->num_latency_values(),
+            static_cast<size_t>(
+                kImpulseFrequencyInHz * kMeasureLatencyTime.seconds() - 1));
+  latency_audio_stream->PrintResults();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_manager.cc b/third_party/libwebrtc/modules/audio_device/android/audio_manager.cc
new file mode 100644
index 0000000000..0b55496619
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_manager.cc
@@ -0,0 +1,318 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/audio_manager.h"
+
+#include <utility>
+
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/utility/include/helpers_android.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+
+namespace webrtc {
+
+// AudioManager::JavaAudioManager implementation
+AudioManager::JavaAudioManager::JavaAudioManager(
+    NativeRegistration* native_reg,
+    std::unique_ptr<GlobalRef> audio_manager)
+    : audio_manager_(std::move(audio_manager)),
+      init_(native_reg->GetMethodId("init", "()Z")),
+      dispose_(native_reg->GetMethodId("dispose", "()V")),
+      is_communication_mode_enabled_(
+          native_reg->GetMethodId("isCommunicationModeEnabled", "()Z")),
+      is_device_blacklisted_for_open_sles_usage_(
+          native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage",
+                                  "()Z")) {
+  RTC_LOG(LS_INFO) << "JavaAudioManager::ctor";
+}
+
+AudioManager::JavaAudioManager::~JavaAudioManager() {
+  RTC_LOG(LS_INFO) << "JavaAudioManager::~dtor";
+}
+
+bool AudioManager::JavaAudioManager::Init() {
+  return audio_manager_->CallBooleanMethod(init_);
+}
+
+void AudioManager::JavaAudioManager::Close() {
+  audio_manager_->CallVoidMethod(dispose_);
+}
+
+bool AudioManager::JavaAudioManager::IsCommunicationModeEnabled() {
+  return audio_manager_->CallBooleanMethod(is_communication_mode_enabled_);
+}
+
+bool AudioManager::JavaAudioManager::IsDeviceBlacklistedForOpenSLESUsage() {
+  return audio_manager_->CallBooleanMethod(
+      is_device_blacklisted_for_open_sles_usage_);
+}
+
+// AudioManager implementation
+AudioManager::AudioManager()
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_layer_(AudioDeviceModule::kPlatformDefaultAudio),
+      initialized_(false),
+      hardware_aec_(false),
+      hardware_agc_(false),
+      hardware_ns_(false),
+      low_latency_playout_(false),
+      low_latency_record_(false),
+      delay_estimate_in_milliseconds_(0) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheAudioParameters", "(IIIZZZZZZZIIJ)V",
+       reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioManager", native_methods,
+      arraysize(native_methods));
+  j_audio_manager_.reset(
+      new JavaAudioManager(j_native_registration_.get(),
+                           j_native_registration_->NewObject(
+                               "<init>", "(J)V", PointerTojlong(this))));
+}
+
+AudioManager::~AudioManager() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Close();
+}
+
+void AudioManager::SetActiveAudioLayer(
+    AudioDeviceModule::AudioLayer audio_layer) {
+  RTC_LOG(LS_INFO) << "SetActiveAudioLayer: " << audio_layer;
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  // Store the currently utilized audio layer.
+  audio_layer_ = audio_layer;
+  // The delay estimate can take one of two fixed values depending on if the
+  // device supports low-latency output or not. However, it is also possible
+  // that the user explicitly selects the high-latency audio path, hence we use
+  // the selected `audio_layer` here to set the delay estimate.
+  delay_estimate_in_milliseconds_ =
+      (audio_layer == AudioDeviceModule::kAndroidJavaAudio)
+          ? kHighLatencyModeDelayEstimateInMilliseconds
+          : kLowLatencyModeDelayEstimateInMilliseconds;
+  RTC_LOG(LS_INFO) << "delay_estimate_in_milliseconds: "
+                   << delay_estimate_in_milliseconds_;
+}
+
+SLObjectItf AudioManager::GetOpenSLEngine() {
+  RTC_LOG(LS_INFO) << "GetOpenSLEngine";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Only allow usage of OpenSL ES if such an audio layer has been specified.
+  if (audio_layer_ != AudioDeviceModule::kAndroidOpenSLESAudio &&
+      audio_layer_ !=
+          AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio) {
+    RTC_LOG(LS_INFO)
+        << "Unable to create OpenSL engine for the current audio layer: "
+        << audio_layer_;
+    return nullptr;
+  }
+  // OpenSL ES for Android only supports a single engine per application.
+  // If one already has been created, return existing object instead of
+  // creating a new.
+  if (engine_object_.Get() != nullptr) {
+    RTC_LOG(LS_WARNING)
+        << "The OpenSL ES engine object has already been created";
+    return engine_object_.Get();
+  }
+  // Create the engine object in thread safe mode.
+  const SLEngineOption option[] = {
+      {SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE)}};
+  SLresult result =
+      slCreateEngine(engine_object_.Receive(), 1, option, 0, NULL, NULL);
+  if (result != SL_RESULT_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "slCreateEngine() failed: "
+                      << GetSLErrorString(result);
+    engine_object_.Reset();
+    return nullptr;
+  }
+  // Realize the SL Engine in synchronous mode.
+  result = engine_object_->Realize(engine_object_.Get(), SL_BOOLEAN_FALSE);
+  if (result != SL_RESULT_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "Realize() failed: " << GetSLErrorString(result);
+    engine_object_.Reset();
+    return nullptr;
+  }
+  // Finally return the SLObjectItf interface of the engine object.
+  return engine_object_.Get();
+}
+
+bool AudioManager::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK_NE(audio_layer_, AudioDeviceModule::kPlatformDefaultAudio);
+  if (!j_audio_manager_->Init()) {
+    RTC_LOG(LS_ERROR) << "Init() failed";
+    return false;
+  }
+  initialized_ = true;
+  return true;
+}
+
+bool AudioManager::Close() {
+  RTC_LOG(LS_INFO) << "Close";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_)
+    return true;
+  j_audio_manager_->Close();
+  initialized_ = false;
+  return true;
+}
+
+bool AudioManager::IsCommunicationModeEnabled() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_manager_->IsCommunicationModeEnabled();
+}
+
+bool AudioManager::IsAcousticEchoCancelerSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return hardware_aec_;
+}
+
+bool AudioManager::IsAutomaticGainControlSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return hardware_agc_;
+}
+
+bool AudioManager::IsNoiseSuppressorSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return hardware_ns_;
+}
+
+bool AudioManager::IsLowLatencyPlayoutSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Some devices are blacklisted for usage of OpenSL ES even if they report
+  // that low-latency playout is supported. See b/21485703 for details.
+  return j_audio_manager_->IsDeviceBlacklistedForOpenSLESUsage()
+             ? false
+             : low_latency_playout_;
+}
+
+bool AudioManager::IsLowLatencyRecordSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return low_latency_record_;
+}
+
+bool AudioManager::IsProAudioSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // TODO(henrika): return the state independently of if OpenSL ES is
+  // blacklisted or not for now. We could use the same approach as in
+  // IsLowLatencyPlayoutSupported() but I can't see the need for it yet.
+  return pro_audio_;
+}
+
+// TODO(henrika): improve comments...
+bool AudioManager::IsAAudioSupported() const {
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+  return a_audio_;
+#else
+  return false;
+#endif
+}
+
+bool AudioManager::IsStereoPlayoutSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (playout_parameters_.channels() == 2);
+}
+
+bool AudioManager::IsStereoRecordSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (record_parameters_.channels() == 2);
+}
+
+int AudioManager::GetDelayEstimateInMilliseconds() const {
+  return delay_estimate_in_milliseconds_;
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env,
+                                                jobject obj,
+                                                jint sample_rate,
+                                                jint output_channels,
+                                                jint input_channels,
+                                                jboolean hardware_aec,
+                                                jboolean hardware_agc,
+                                                jboolean hardware_ns,
+                                                jboolean low_latency_output,
+                                                jboolean low_latency_input,
+                                                jboolean pro_audio,
+                                                jboolean a_audio,
+                                                jint output_buffer_size,
+                                                jint input_buffer_size,
+                                                jlong native_audio_manager) {
+  webrtc::AudioManager* this_object =
+      reinterpret_cast<webrtc::AudioManager*>(native_audio_manager);
+  this_object->OnCacheAudioParameters(
+      env, sample_rate, output_channels, input_channels, hardware_aec,
+      hardware_agc, hardware_ns, low_latency_output, low_latency_input,
+      pro_audio, a_audio, output_buffer_size, input_buffer_size);
+}
+
+void AudioManager::OnCacheAudioParameters(JNIEnv* env,
+                                          jint sample_rate,
+                                          jint output_channels,
+                                          jint input_channels,
+                                          jboolean hardware_aec,
+                                          jboolean hardware_agc,
+                                          jboolean hardware_ns,
+                                          jboolean low_latency_output,
+                                          jboolean low_latency_input,
+                                          jboolean pro_audio,
+                                          jboolean a_audio,
+                                          jint output_buffer_size,
+                                          jint input_buffer_size) {
+  RTC_LOG(LS_INFO)
+      << "OnCacheAudioParameters: "
+         "hardware_aec: "
+      << static_cast<bool>(hardware_aec)
+      << ", hardware_agc: " << static_cast<bool>(hardware_agc)
+      << ", hardware_ns: " << static_cast<bool>(hardware_ns)
+      << ", low_latency_output: " << static_cast<bool>(low_latency_output)
+      << ", low_latency_input: " << static_cast<bool>(low_latency_input)
+      << ", pro_audio: " << static_cast<bool>(pro_audio)
+      << ", a_audio: " << static_cast<bool>(a_audio)
+      << ", sample_rate: " << static_cast<int>(sample_rate)
+      << ", output_channels: " << static_cast<int>(output_channels)
+      << ", input_channels: " << static_cast<int>(input_channels)
+      << ", output_buffer_size: " << static_cast<int>(output_buffer_size)
+      << ", input_buffer_size: " << static_cast<int>(input_buffer_size);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  hardware_aec_ = hardware_aec;
+  hardware_agc_ = hardware_agc;
+  hardware_ns_ = hardware_ns;
+  low_latency_playout_ = low_latency_output;
+  low_latency_record_ = low_latency_input;
+  pro_audio_ = pro_audio;
+  a_audio_ = a_audio;
+  playout_parameters_.reset(sample_rate, static_cast<size_t>(output_channels),
+                            static_cast<size_t>(output_buffer_size));
+  record_parameters_.reset(sample_rate, static_cast<size_t>(input_channels),
+                           static_cast<size_t>(input_buffer_size));
+}
+
+const AudioParameters& AudioManager::GetPlayoutAudioParameters() {
+  RTC_CHECK(playout_parameters_.is_valid());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return playout_parameters_;
+}
+
+const AudioParameters& AudioManager::GetRecordAudioParameters() {
+  RTC_CHECK(record_parameters_.is_valid());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return record_parameters_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_manager.h b/third_party/libwebrtc/modules/audio_device/android/audio_manager.h
new file mode 100644
index 0000000000..900fc78a68
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_manager.h
@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
+
+#include <SLES/OpenSLES.h>
+#include <jni.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/opensles_common.h"
+#include "modules/audio_device/audio_device_config.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// Implements support for functions in the WebRTC audio stack for Android that
+// relies on the AudioManager in android.media. It also populates an
+// AudioParameter structure with native audio parameters detected at
+// construction. This class does not make any audio-related modifications
+// unless Init() is called. Caching audio parameters makes no changes but only
+// reads data from the Java side.
+class AudioManager {
+ public:
+  // Wraps the Java specific parts of the AudioManager into one helper class.
+  // Stores method IDs for all supported methods at construction and then
+  // allows calls like JavaAudioManager::Close() while hiding the Java/JNI
+  // parts that are associated with this call.
+  class JavaAudioManager {
+   public:
+    JavaAudioManager(NativeRegistration* native_registration,
+                     std::unique_ptr<GlobalRef> audio_manager);
+    ~JavaAudioManager();
+
+    bool Init();
+    void Close();
+    bool IsCommunicationModeEnabled();
+    bool IsDeviceBlacklistedForOpenSLESUsage();
+
+   private:
+    std::unique_ptr<GlobalRef> audio_manager_;
+    jmethodID init_;
+    jmethodID dispose_;
+    jmethodID is_communication_mode_enabled_;
+    jmethodID is_device_blacklisted_for_open_sles_usage_;
+  };
+
+  AudioManager();
+  ~AudioManager();
+
+  // Sets the currently active audio layer combination. Must be called before
+  // Init().
+  void SetActiveAudioLayer(AudioDeviceModule::AudioLayer audio_layer);
+
+  // Creates and realizes the main (global) Open SL engine object and returns
+  // a reference to it. The engine object is only created at the first call
+  // since OpenSL ES for Android only supports a single engine per application.
+  // Subsequent calls returns the already created engine. The SL engine object
+  // is destroyed when the AudioManager object is deleted. It means that the
+  // engine object will be the first OpenSL ES object to be created and last
+  // object to be destroyed.
+  // Note that NULL will be returned unless the audio layer is specified as
+  // AudioDeviceModule::kAndroidOpenSLESAudio or
+  // AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio.
+  SLObjectItf GetOpenSLEngine();
+
+  // Initializes the audio manager and stores the current audio mode.
+  bool Init();
+  // Revert any setting done by Init().
+  bool Close();
+
+  // Returns true if current audio mode is AudioManager.MODE_IN_COMMUNICATION.
+  bool IsCommunicationModeEnabled() const;
+
+  // Native audio parameters stored during construction.
+  const AudioParameters& GetPlayoutAudioParameters();
+  const AudioParameters& GetRecordAudioParameters();
+
+  // Returns true if the device supports built-in audio effects for AEC, AGC
+  // and NS. Some devices can also be blacklisted for use in combination with
+  // platform effects and these devices will return false.
+  // Can currently only be used in combination with a Java based audio backend
+  // for the recoring side (i.e. using the android.media.AudioRecord API).
+  bool IsAcousticEchoCancelerSupported() const;
+  bool IsAutomaticGainControlSupported() const;
+  bool IsNoiseSuppressorSupported() const;
+
+  // Returns true if the device supports the low-latency audio paths in
+  // combination with OpenSL ES.
+  bool IsLowLatencyPlayoutSupported() const;
+  bool IsLowLatencyRecordSupported() const;
+
+  // Returns true if the device supports (and has been configured for) stereo.
+  // Call the Java API WebRtcAudioManager.setStereoOutput/Input() with true as
+  // paramter to enable stereo. Default is mono in both directions and the
+  // setting is set once and for all when the audio manager object is created.
+  // TODO(henrika): stereo is not supported in combination with OpenSL ES.
+  bool IsStereoPlayoutSupported() const;
+  bool IsStereoRecordSupported() const;
+
+  // Returns true if the device supports pro-audio features in combination with
+  // OpenSL ES.
+  bool IsProAudioSupported() const;
+
+  // Returns true if the device supports AAudio.
+  bool IsAAudioSupported() const;
+
+  // Returns the estimated total delay of this device. Unit is in milliseconds.
+  // The vaule is set once at construction and never changes after that.
+  // Possible values are webrtc::kLowLatencyModeDelayEstimateInMilliseconds and
+  // webrtc::kHighLatencyModeDelayEstimateInMilliseconds.
+  int GetDelayEstimateInMilliseconds() const;
+
+ private:
+  // Called from Java side so we can cache the native audio parameters.
+  // This method will be called by the WebRtcAudioManager constructor, i.e.
+  // on the same thread that this object is created on.
+  static void JNICALL CacheAudioParameters(JNIEnv* env,
+                                           jobject obj,
+                                           jint sample_rate,
+                                           jint output_channels,
+                                           jint input_channels,
+                                           jboolean hardware_aec,
+                                           jboolean hardware_agc,
+                                           jboolean hardware_ns,
+                                           jboolean low_latency_output,
+                                           jboolean low_latency_input,
+                                           jboolean pro_audio,
+                                           jboolean a_audio,
+                                           jint output_buffer_size,
+                                           jint input_buffer_size,
+                                           jlong native_audio_manager);
+  void OnCacheAudioParameters(JNIEnv* env,
+                              jint sample_rate,
+                              jint output_channels,
+                              jint input_channels,
+                              jboolean hardware_aec,
+                              jboolean hardware_agc,
+                              jboolean hardware_ns,
+                              jboolean low_latency_output,
+                              jboolean low_latency_input,
+                              jboolean pro_audio,
+                              jboolean a_audio,
+                              jint output_buffer_size,
+                              jint input_buffer_size);
+
+  // Stores thread ID in the constructor.
+  // We can then use RTC_DCHECK_RUN_ON(&thread_checker_) to ensure that
+  // other methods are called from the same thread.
+  SequenceChecker thread_checker_;
+
+  // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
+  // construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  std::unique_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioManager.
+  std::unique_ptr<AudioManager::JavaAudioManager> j_audio_manager_;
+
+  // Contains the selected audio layer specified by the AudioLayer enumerator
+  // in the AudioDeviceModule class.
+  AudioDeviceModule::AudioLayer audio_layer_;
+
+  // This object is the global entry point of the OpenSL ES API.
+  // After creating the engine object, the application can obtain this object‘s
+  // SLEngineItf interface. This interface contains creation methods for all
+  // the other object types in the API. None of these interface are realized
+  // by this class. It only provides access to the global engine object.
+  webrtc::ScopedSLObjectItf engine_object_;
+
+  // Set to true by Init() and false by Close().
+  bool initialized_;
+
+  // True if device supports hardware (or built-in) AEC.
+  bool hardware_aec_;
+  // True if device supports hardware (or built-in) AGC.
+  bool hardware_agc_;
+  // True if device supports hardware (or built-in) NS.
+  bool hardware_ns_;
+
+  // True if device supports the low-latency OpenSL ES audio path for output.
+  bool low_latency_playout_;
+
+  // True if device supports the low-latency OpenSL ES audio path for input.
+  bool low_latency_record_;
+
+  // True if device supports the low-latency OpenSL ES pro-audio path.
+  bool pro_audio_;
+
+  // True if device supports the low-latency AAudio audio path.
+  bool a_audio_;
+
+  // The delay estimate can take one of two fixed values depending on if the
+  // device supports low-latency output or not.
+  int delay_estimate_in_milliseconds_;
+
+  // Contains native parameters (e.g. sample rate, channel configuration).
+  // Set at construction in OnCacheAudioParameters() which is called from
+  // Java on the same thread as this object is created on.
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_manager_unittest.cc b/third_party/libwebrtc/modules/audio_device/android/audio_manager_unittest.cc
new file mode 100644
index 0000000000..093eddd2e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_manager_unittest.cc
@@ -0,0 +1,239 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/audio_manager.h"
+
+#include <SLES/OpenSLES_Android.h>
+
+#include "modules/audio_device/android/build_info.h"
+#include "modules/audio_device/android/ensure_initialized.h"
+#include "rtc_base/arraysize.h"
+#include "test/gtest.h"
+
+#define PRINT(...) fprintf(stderr, __VA_ARGS__);
+
+namespace webrtc {
+
+static const char kTag[] = "  ";
+
+class AudioManagerTest : public ::testing::Test {
+ protected:
+  AudioManagerTest() {
+    // One-time initialization of JVM and application context. Ensures that we
+    // can do calls between C++ and Java.
+    webrtc::audiodevicemodule::EnsureInitialized();
+    audio_manager_.reset(new AudioManager());
+    SetActiveAudioLayer();
+    playout_parameters_ = audio_manager()->GetPlayoutAudioParameters();
+    record_parameters_ = audio_manager()->GetRecordAudioParameters();
+  }
+
+  AudioManager* audio_manager() const { return audio_manager_.get(); }
+
+  // A valid audio layer must always be set before calling Init(), hence we
+  // might as well make it a part of the test fixture.
+  void SetActiveAudioLayer() {
+    EXPECT_EQ(0, audio_manager()->GetDelayEstimateInMilliseconds());
+    audio_manager()->SetActiveAudioLayer(AudioDeviceModule::kAndroidJavaAudio);
+    EXPECT_NE(0, audio_manager()->GetDelayEstimateInMilliseconds());
+  }
+
+  // One way to ensure that the engine object is valid is to create an
+  // SL Engine interface since it exposes creation methods of all the OpenSL ES
+  // object types and it is only supported on the engine object. This method
+  // also verifies that the engine interface supports at least one interface.
+  // Note that, the test below is not a full test of the SLEngineItf object
+  // but only a simple sanity test to check that the global engine object is OK.
+  void ValidateSLEngine(SLObjectItf engine_object) {
+    EXPECT_NE(nullptr, engine_object);
+    // Get the SL Engine interface which is exposed by the engine object.
+    SLEngineItf engine;
+    SLresult result =
+        (*engine_object)->GetInterface(engine_object, SL_IID_ENGINE, &engine);
+    EXPECT_EQ(result, SL_RESULT_SUCCESS) << "GetInterface() on engine failed";
+    // Ensure that the SL Engine interface exposes at least one interface.
+    SLuint32 object_id = SL_OBJECTID_ENGINE;
+    SLuint32 num_supported_interfaces = 0;
+    result = (*engine)->QueryNumSupportedInterfaces(engine, object_id,
+                                                    &num_supported_interfaces);
+    EXPECT_EQ(result, SL_RESULT_SUCCESS)
+        << "QueryNumSupportedInterfaces() failed";
+    EXPECT_GE(num_supported_interfaces, 1u);
+  }
+
+  std::unique_ptr<AudioManager> audio_manager_;
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+};
+
+TEST_F(AudioManagerTest, ConstructDestruct) {}
+
+// It should not be possible to create an OpenSL engine object if Java based
+// audio is requested in both directions.
+TEST_F(AudioManagerTest, GetOpenSLEngineShouldFailForJavaAudioLayer) {
+  audio_manager()->SetActiveAudioLayer(AudioDeviceModule::kAndroidJavaAudio);
+  SLObjectItf engine_object = audio_manager()->GetOpenSLEngine();
+  EXPECT_EQ(nullptr, engine_object);
+}
+
+// It should be possible to create an OpenSL engine object if OpenSL ES based
+// audio is requested in any direction.
+TEST_F(AudioManagerTest, GetOpenSLEngineShouldSucceedForOpenSLESAudioLayer) {
+  // List of supported audio layers that uses OpenSL ES audio.
+  const AudioDeviceModule::AudioLayer opensles_audio[] = {
+      AudioDeviceModule::kAndroidOpenSLESAudio,
+      AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio};
+  // Verify that the global (singleton) OpenSL Engine can be acquired for all
+  // audio layes that uses OpenSL ES. Note that the engine is only created once.
+  for (const AudioDeviceModule::AudioLayer audio_layer : opensles_audio) {
+    audio_manager()->SetActiveAudioLayer(audio_layer);
+    SLObjectItf engine_object = audio_manager()->GetOpenSLEngine();
+    EXPECT_NE(nullptr, engine_object);
+    // Perform a simple sanity check of the created engine object.
+    ValidateSLEngine(engine_object);
+  }
+}
+
+TEST_F(AudioManagerTest, InitClose) {
+  EXPECT_TRUE(audio_manager()->Init());
+  EXPECT_TRUE(audio_manager()->Close());
+}
+
+TEST_F(AudioManagerTest, IsAcousticEchoCancelerSupported) {
+  PRINT("%sAcoustic Echo Canceler support: %s\n", kTag,
+        audio_manager()->IsAcousticEchoCancelerSupported() ? "Yes" : "No");
+}
+
+TEST_F(AudioManagerTest, IsAutomaticGainControlSupported) {
+  EXPECT_FALSE(audio_manager()->IsAutomaticGainControlSupported());
+}
+
+TEST_F(AudioManagerTest, IsNoiseSuppressorSupported) {
+  PRINT("%sNoise Suppressor support: %s\n", kTag,
+        audio_manager()->IsNoiseSuppressorSupported() ? "Yes" : "No");
+}
+
+TEST_F(AudioManagerTest, IsLowLatencyPlayoutSupported) {
+  PRINT("%sLow latency output support: %s\n", kTag,
+        audio_manager()->IsLowLatencyPlayoutSupported() ? "Yes" : "No");
+}
+
+TEST_F(AudioManagerTest, IsLowLatencyRecordSupported) {
+  PRINT("%sLow latency input support: %s\n", kTag,
+        audio_manager()->IsLowLatencyRecordSupported() ? "Yes" : "No");
+}
+
+TEST_F(AudioManagerTest, IsProAudioSupported) {
+  PRINT("%sPro audio support: %s\n", kTag,
+        audio_manager()->IsProAudioSupported() ? "Yes" : "No");
+}
+
+// Verify that playout side is configured for mono by default.
+TEST_F(AudioManagerTest, IsStereoPlayoutSupported) {
+  EXPECT_FALSE(audio_manager()->IsStereoPlayoutSupported());
+}
+
+// Verify that recording side is configured for mono by default.
+TEST_F(AudioManagerTest, IsStereoRecordSupported) {
+  EXPECT_FALSE(audio_manager()->IsStereoRecordSupported());
+}
+
+TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
+  const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
+  const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported();
+  PRINT("PLAYOUT:\n");
+  PRINT("%saudio layer: %s\n", kTag,
+        low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
+  PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate());
+  PRINT("%schannels: %zu\n", kTag, playout_parameters_.channels());
+  PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag,
+        playout_parameters_.frames_per_buffer(),
+        playout_parameters_.GetBufferSizeInMilliseconds());
+  PRINT("RECORD: \n");
+  PRINT("%saudio layer: %s\n", kTag,
+        low_latency_in ? "Low latency OpenSL" : "Java/JNI based AudioRecord");
+  PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate());
+  PRINT("%schannels: %zu\n", kTag, record_parameters_.channels());
+  PRINT("%sframes per buffer: %zu <=> %.2f ms\n", kTag,
+        record_parameters_.frames_per_buffer(),
+        record_parameters_.GetBufferSizeInMilliseconds());
+}
+
+// The audio device module only suppors the same sample rate in both directions.
+// In addition, in full-duplex low-latency mode (OpenSL ES), both input and
+// output must use the same native buffer size to allow for usage of the fast
+// audio track in Android.
+TEST_F(AudioManagerTest, VerifyAudioParameters) {
+  const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
+  const bool low_latency_in = audio_manager()->IsLowLatencyRecordSupported();
+  EXPECT_EQ(playout_parameters_.sample_rate(),
+            record_parameters_.sample_rate());
+  if (low_latency_out && low_latency_in) {
+    EXPECT_EQ(playout_parameters_.frames_per_buffer(),
+              record_parameters_.frames_per_buffer());
+  }
+}
+
+// Add device-specific information to the test for logging purposes.
+TEST_F(AudioManagerTest, ShowDeviceInfo) {
+  BuildInfo build_info;
+  PRINT("%smodel: %s\n", kTag, build_info.GetDeviceModel().c_str());
+  PRINT("%sbrand: %s\n", kTag, build_info.GetBrand().c_str());
+  PRINT("%smanufacturer: %s\n", kTag,
+        build_info.GetDeviceManufacturer().c_str());
+}
+
+// Add Android build information to the test for logging purposes.
+TEST_F(AudioManagerTest, ShowBuildInfo) {
+  BuildInfo build_info;
+  PRINT("%sbuild release: %s\n", kTag, build_info.GetBuildRelease().c_str());
+  PRINT("%sbuild id: %s\n", kTag, build_info.GetAndroidBuildId().c_str());
+  PRINT("%sbuild type: %s\n", kTag, build_info.GetBuildType().c_str());
+  PRINT("%sSDK version: %d\n", kTag, build_info.GetSdkVersion());
+}
+
+// Basic test of the AudioParameters class using default construction where
+// all members are set to zero.
+TEST_F(AudioManagerTest, AudioParametersWithDefaultConstruction) {
+  AudioParameters params;
+  EXPECT_FALSE(params.is_valid());
+  EXPECT_EQ(0, params.sample_rate());
+  EXPECT_EQ(0U, params.channels());
+  EXPECT_EQ(0U, params.frames_per_buffer());
+  EXPECT_EQ(0U, params.frames_per_10ms_buffer());
+  EXPECT_EQ(0U, params.GetBytesPerFrame());
+  EXPECT_EQ(0U, params.GetBytesPerBuffer());
+  EXPECT_EQ(0U, params.GetBytesPer10msBuffer());
+  EXPECT_EQ(0.0f, params.GetBufferSizeInMilliseconds());
+}
+
+// Basic test of the AudioParameters class using non default construction.
+TEST_F(AudioManagerTest, AudioParametersWithNonDefaultConstruction) {
+  const int kSampleRate = 48000;
+  const size_t kChannels = 1;
+  const size_t kFramesPerBuffer = 480;
+  const size_t kFramesPer10msBuffer = 480;
+  const size_t kBytesPerFrame = 2;
+  const float kBufferSizeInMs = 10.0f;
+  AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer);
+  EXPECT_TRUE(params.is_valid());
+  EXPECT_EQ(kSampleRate, params.sample_rate());
+  EXPECT_EQ(kChannels, params.channels());
+  EXPECT_EQ(kFramesPerBuffer, params.frames_per_buffer());
+  EXPECT_EQ(static_cast<size_t>(kSampleRate / 100),
+            params.frames_per_10ms_buffer());
+  EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame());
+  EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer());
+  EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer,
+            params.GetBytesPer10msBuffer());
+  EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_record_jni.cc b/third_party/libwebrtc/modules/audio_device/android/audio_record_jni.cc
new file mode 100644
index 0000000000..919eabb983
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_record_jni.cc
@@ -0,0 +1,280 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/audio_record_jni.h"
+
+#include <string>
+#include <utility>
+
+#include "modules/audio_device/android/audio_common.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+// Scoped class which logs its time of life as a UMA statistic. It generates
+// a histogram which measures the time it takes for a method/scope to execute.
+class ScopedHistogramTimer {
+ public:
+  explicit ScopedHistogramTimer(const std::string& name)
+      : histogram_name_(name), start_time_ms_(rtc::TimeMillis()) {}
+  ~ScopedHistogramTimer() {
+    const int64_t life_time_ms = rtc::TimeSince(start_time_ms_);
+    RTC_HISTOGRAM_COUNTS_1000(histogram_name_, life_time_ms);
+    RTC_LOG(LS_INFO) << histogram_name_ << ": " << life_time_ms;
+  }
+
+ private:
+  const std::string histogram_name_;
+  int64_t start_time_ms_;
+};
+}  // namespace
+
+// AudioRecordJni::JavaAudioRecord implementation.
+AudioRecordJni::JavaAudioRecord::JavaAudioRecord(
+    NativeRegistration* native_reg,
+    std::unique_ptr<GlobalRef> audio_record)
+    : audio_record_(std::move(audio_record)),
+      init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
+      start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
+      stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
+      enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
+      enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
+
+AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
+
+int AudioRecordJni::JavaAudioRecord::InitRecording(int sample_rate,
+                                                   size_t channels) {
+  return audio_record_->CallIntMethod(init_recording_,
+                                      static_cast<jint>(sample_rate),
+                                      static_cast<jint>(channels));
+}
+
+bool AudioRecordJni::JavaAudioRecord::StartRecording() {
+  return audio_record_->CallBooleanMethod(start_recording_);
+}
+
+bool AudioRecordJni::JavaAudioRecord::StopRecording() {
+  return audio_record_->CallBooleanMethod(stop_recording_);
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInAEC(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_aec_,
+                                          static_cast<jboolean>(enable));
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInNS(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_ns_,
+                                          static_cast<jboolean>(enable));
+}
+
+// AudioRecordJni implementation.
+AudioRecordJni::AudioRecordJni(AudioManager* audio_manager)
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_manager_(audio_manager),
+      audio_parameters_(audio_manager->GetRecordAudioParameters()),
+      total_delay_in_milliseconds_(0),
+      direct_buffer_address_(nullptr),
+      direct_buffer_capacity_in_bytes_(0),
+      frames_per_buffer_(0),
+      initialized_(false),
+      recording_(false),
+      audio_device_buffer_(nullptr) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_DCHECK(audio_parameters_.is_valid());
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
+       reinterpret_cast<void*>(
+           &webrtc::AudioRecordJni::CacheDirectBufferAddress)},
+      {"nativeDataIsRecorded", "(IJ)V",
+       reinterpret_cast<void*>(&webrtc::AudioRecordJni::DataIsRecorded)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioRecord", native_methods,
+      arraysize(native_methods));
+  j_audio_record_.reset(
+      new JavaAudioRecord(j_native_registration_.get(),
+                          j_native_registration_->NewObject(
+                              "<init>", "(J)V", PointerTojlong(this))));
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the Java based audio thread.
+  thread_checker_java_.Detach();
+}
+
+AudioRecordJni::~AudioRecordJni() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+}
+
+int32_t AudioRecordJni::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return 0;
+}
+
+int32_t AudioRecordJni::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopRecording();
+  return 0;
+}
+
+int32_t AudioRecordJni::InitRecording() {
+  RTC_LOG(LS_INFO) << "InitRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  ScopedHistogramTimer timer("WebRTC.Audio.InitRecordingDurationMs");
+  int frames_per_buffer = j_audio_record_->InitRecording(
+      audio_parameters_.sample_rate(), audio_parameters_.channels());
+  if (frames_per_buffer < 0) {
+    direct_buffer_address_ = nullptr;
+    RTC_LOG(LS_ERROR) << "InitRecording failed";
+    return -1;
+  }
+  frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
+  RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
+  RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
+               frames_per_buffer_ * bytes_per_frame);
+  RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
+  initialized_ = true;
+  return 0;
+}
+
+int32_t AudioRecordJni::StartRecording() {
+  RTC_LOG(LS_INFO) << "StartRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!recording_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Recording can not start since InitRecording must succeed first";
+    return 0;
+  }
+  ScopedHistogramTimer timer("WebRTC.Audio.StartRecordingDurationMs");
+  if (!j_audio_record_->StartRecording()) {
+    RTC_LOG(LS_ERROR) << "StartRecording failed";
+    return -1;
+  }
+  recording_ = true;
+  return 0;
+}
+
+int32_t AudioRecordJni::StopRecording() {
+  RTC_LOG(LS_INFO) << "StopRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !recording_) {
+    return 0;
+  }
+  if (!j_audio_record_->StopRecording()) {
+    RTC_LOG(LS_ERROR) << "StopRecording failed";
+    return -1;
+  }
+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
+  // next time StartRecording() is called since it will create a new Java
+  // thread.
+  thread_checker_java_.Detach();
+  initialized_ = false;
+  recording_ = false;
+  direct_buffer_address_ = nullptr;
+  return 0;
+}
+
+void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_LOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
+  audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
+  const size_t channels = audio_parameters_.channels();
+  RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
+  audio_device_buffer_->SetRecordingChannels(channels);
+  total_delay_in_milliseconds_ =
+      audio_manager_->GetDelayEstimateInMilliseconds();
+  RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
+  RTC_LOG(LS_INFO) << "total_delay_in_milliseconds: "
+                   << total_delay_in_milliseconds_;
+}
+
+int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInAEC(" << enable << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_record_->EnableBuiltInAEC(enable) ? 0 : -1;
+}
+
+int32_t AudioRecordJni::EnableBuiltInAGC(bool enable) {
+  // TODO(henrika): possibly remove when no longer used by any client.
+  RTC_CHECK_NOTREACHED();
+}
+
+int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInNS(" << enable << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioRecordJni::CacheDirectBufferAddress(JNIEnv* env,
+                                                      jobject obj,
+                                                      jobject byte_buffer,
+                                                      jlong nativeAudioRecord) {
+  webrtc::AudioRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioRecordJni*>(nativeAudioRecord);
+  this_object->OnCacheDirectBufferAddress(env, byte_buffer);
+}
+
+void AudioRecordJni::OnCacheDirectBufferAddress(JNIEnv* env,
+                                                jobject byte_buffer) {
+  RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!direct_buffer_address_);
+  direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
+  jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
+  RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioRecordJni::DataIsRecorded(JNIEnv* env,
+                                            jobject obj,
+                                            jint length,
+                                            jlong nativeAudioRecord) {
+  webrtc::AudioRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioRecordJni*>(nativeAudioRecord);
+  this_object->OnDataIsRecorded(length);
+}
+
+// This method is called on a high-priority thread from Java. The name of
+// the thread is 'AudioRecordThread'.
+void AudioRecordJni::OnDataIsRecorded(int length) {
+  RTC_DCHECK(thread_checker_java_.IsCurrent());
+  if (!audio_device_buffer_) {
+    RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
+    return;
+  }
+  audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
+                                          frames_per_buffer_);
+  // We provide one (combined) fixed delay estimate for the APM and use the
+  // `playDelayMs` parameter only. Components like the AEC only sees the sum
+  // of `playDelayMs` and `recDelayMs`, hence the distributions does not matter.
+  audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_, 0);
+  if (audio_device_buffer_->DeliverRecordedData() == -1) {
+    RTC_LOG(LS_INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_record_jni.h b/third_party/libwebrtc/modules/audio_device/android/audio_record_jni.h
new file mode 100644
index 0000000000..66a6a89f41
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_record_jni.h
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
+
+#include <jni.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// Implements 16-bit mono PCM audio input support for Android using the Java
+// AudioRecord interface. Most of the work is done by its Java counterpart in
+// WebRtcAudioRecord.java. This class is created and lives on a thread in
+// C++-land, but recorded audio buffers are delivered on a high-priority
+// thread managed by the Java class.
+//
+// The Java class makes use of AudioEffect features (mainly AEC) which are
+// first available in Jelly Bean. If it is instantiated running against earlier
+// SDKs, the AEC provided by the APM in WebRTC must be used and enabled
+// separately instead.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread.
+//
+// This class uses JvmThreadConnector to attach to a Java VM if needed
+// and detach when the object goes out of scope. Additional thread checking
+// guarantees that no other (possibly non attached) thread is used.
+class AudioRecordJni {
+ public:
+  // Wraps the Java specific parts of the AudioRecordJni into one helper class.
+  class JavaAudioRecord {
+   public:
+    JavaAudioRecord(NativeRegistration* native_registration,
+                    std::unique_ptr<GlobalRef> audio_track);
+    ~JavaAudioRecord();
+
+    int InitRecording(int sample_rate, size_t channels);
+    bool StartRecording();
+    bool StopRecording();
+    bool EnableBuiltInAEC(bool enable);
+    bool EnableBuiltInNS(bool enable);
+
+   private:
+    std::unique_ptr<GlobalRef> audio_record_;
+    jmethodID init_recording_;
+    jmethodID start_recording_;
+    jmethodID stop_recording_;
+    jmethodID enable_built_in_aec_;
+    jmethodID enable_built_in_ns_;
+  };
+
+  explicit AudioRecordJni(AudioManager* audio_manager);
+  ~AudioRecordJni();
+
+  int32_t Init();
+  int32_t Terminate();
+
+  int32_t InitRecording();
+  bool RecordingIsInitialized() const { return initialized_; }
+
+  int32_t StartRecording();
+  int32_t StopRecording();
+  bool Recording() const { return recording_; }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+  int32_t EnableBuiltInAEC(bool enable);
+  int32_t EnableBuiltInAGC(bool enable);
+  int32_t EnableBuiltInNS(bool enable);
+
+ private:
+  // Called from Java side so we can cache the address of the Java-manged
+  // `byte_buffer` in `direct_buffer_address_`. The size of the buffer
+  // is also stored in `direct_buffer_capacity_in_bytes_`.
+  // This method will be called by the WebRtcAudioRecord constructor, i.e.,
+  // on the same thread that this object is created on.
+  static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
+                                               jobject obj,
+                                               jobject byte_buffer,
+                                               jlong nativeAudioRecord);
+  void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
+
+  // Called periodically by the Java based WebRtcAudioRecord object when
+  // recording has started. Each call indicates that there are `length` new
+  // bytes recorded in the memory area `direct_buffer_address_` and it is
+  // now time to send these to the consumer.
+  // This method is called on a high-priority thread from Java. The name of
+  // the thread is 'AudioRecordThread'.
+  static void JNICALL DataIsRecorded(JNIEnv* env,
+                                     jobject obj,
+                                     jint length,
+                                     jlong nativeAudioRecord);
+  void OnDataIsRecorded(int length);
+
+  // Stores thread ID in constructor.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to OnDataIsRecorded() from high-priority
+  // thread in Java. Detached during construction of this object.
+  SequenceChecker thread_checker_java_;
+
+  // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
+  // construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  std::unique_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioRecordJni class.
+  std::unique_ptr<AudioRecordJni::JavaAudioRecord> j_audio_record_;
+
+  // Raw pointer to the audio manger.
+  const AudioManager* audio_manager_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Delay estimate of the total round-trip delay (input + output).
+  // Fixed value set once in AttachAudioBuffer() and it can take one out of two
+  // possible values. See audio_common.h for details.
+  int total_delay_in_milliseconds_;
+
+  // Cached copy of address to direct audio buffer owned by `j_audio_record_`.
+  void* direct_buffer_address_;
+
+  // Number of bytes in the direct audio buffer owned by `j_audio_record_`.
+  size_t direct_buffer_capacity_in_bytes_;
+
+  // Number audio frames per audio buffer. Each audio frame corresponds to
+  // one sample of PCM mono data at 16 bits per sample. Hence, each audio
+  // frame contains 2 bytes (given that the Java layer only supports mono).
+  // Example: 480 for 48000 Hz or 441 for 44100 Hz.
+  size_t frames_per_buffer_;
+
+  bool initialized_;
+
+  bool recording_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_track_jni.cc b/third_party/libwebrtc/modules/audio_device/android/audio_track_jni.cc
new file mode 100644
index 0000000000..5afa1ec252
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_track_jni.cc
@@ -0,0 +1,296 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/audio_track_jni.h"
+
+#include <utility>
+
+#include "modules/audio_device/android/audio_manager.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+// AudioTrackJni::JavaAudioTrack implementation.
+AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
+    NativeRegistration* native_reg,
+    std::unique_ptr<GlobalRef> audio_track)
+    : audio_track_(std::move(audio_track)),
+      init_playout_(native_reg->GetMethodId("initPlayout", "(IID)I")),
+      start_playout_(native_reg->GetMethodId("startPlayout", "()Z")),
+      stop_playout_(native_reg->GetMethodId("stopPlayout", "()Z")),
+      set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")),
+      get_stream_max_volume_(
+          native_reg->GetMethodId("getStreamMaxVolume", "()I")),
+      get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")),
+      get_buffer_size_in_frames_(
+          native_reg->GetMethodId("getBufferSizeInFrames", "()I")) {}
+
+AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
+
+bool AudioTrackJni::JavaAudioTrack::InitPlayout(int sample_rate, int channels) {
+  double buffer_size_factor =
+      strtod(webrtc::field_trial::FindFullName(
+                 "WebRTC-AudioDevicePlayoutBufferSizeFactor")
+                 .c_str(),
+             nullptr);
+  if (buffer_size_factor == 0)
+    buffer_size_factor = 1.0;
+  int requested_buffer_size_bytes = audio_track_->CallIntMethod(
+      init_playout_, sample_rate, channels, buffer_size_factor);
+  // Update UMA histograms for both the requested and actual buffer size.
+  if (requested_buffer_size_bytes >= 0) {
+    // To avoid division by zero, we assume the sample rate is 48k if an invalid
+    // value is found.
+    sample_rate = sample_rate <= 0 ? 48000 : sample_rate;
+    // This calculation assumes that audio is mono.
+    const int requested_buffer_size_ms =
+        (requested_buffer_size_bytes * 1000) / (2 * sample_rate);
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeRequestedAudioBufferSizeMs",
+                         requested_buffer_size_ms, 0, 1000, 100);
+    int actual_buffer_size_frames =
+        audio_track_->CallIntMethod(get_buffer_size_in_frames_);
+    if (actual_buffer_size_frames >= 0) {
+      const int actual_buffer_size_ms =
+          actual_buffer_size_frames * 1000 / sample_rate;
+      RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeAudioBufferSizeMs",
+                           actual_buffer_size_ms, 0, 1000, 100);
+    }
+    return true;
+  }
+  return false;
+}
+
+bool AudioTrackJni::JavaAudioTrack::StartPlayout() {
+  return audio_track_->CallBooleanMethod(start_playout_);
+}
+
+bool AudioTrackJni::JavaAudioTrack::StopPlayout() {
+  return audio_track_->CallBooleanMethod(stop_playout_);
+}
+
+bool AudioTrackJni::JavaAudioTrack::SetStreamVolume(int volume) {
+  return audio_track_->CallBooleanMethod(set_stream_volume_, volume);
+}
+
+int AudioTrackJni::JavaAudioTrack::GetStreamMaxVolume() {
+  return audio_track_->CallIntMethod(get_stream_max_volume_);
+}
+
+int AudioTrackJni::JavaAudioTrack::GetStreamVolume() {
+  return audio_track_->CallIntMethod(get_stream_volume_);
+}
+
+// TODO(henrika): possible extend usage of AudioManager and add it as member.
+AudioTrackJni::AudioTrackJni(AudioManager* audio_manager)
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
+      direct_buffer_address_(nullptr),
+      direct_buffer_capacity_in_bytes_(0),
+      frames_per_buffer_(0),
+      initialized_(false),
+      playing_(false),
+      audio_device_buffer_(nullptr) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_DCHECK(audio_parameters_.is_valid());
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
+       reinterpret_cast<void*>(
+           &webrtc::AudioTrackJni::CacheDirectBufferAddress)},
+      {"nativeGetPlayoutData", "(IJ)V",
+       reinterpret_cast<void*>(&webrtc::AudioTrackJni::GetPlayoutData)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioTrack", native_methods,
+      arraysize(native_methods));
+  j_audio_track_.reset(
+      new JavaAudioTrack(j_native_registration_.get(),
+                         j_native_registration_->NewObject(
+                             "<init>", "(J)V", PointerTojlong(this))));
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the Java based audio thread.
+  thread_checker_java_.Detach();
+}
+
+AudioTrackJni::~AudioTrackJni() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+}
+
+int32_t AudioTrackJni::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return 0;
+}
+
+int32_t AudioTrackJni::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopPlayout();
+  return 0;
+}
+
+int32_t AudioTrackJni::InitPlayout() {
+  RTC_LOG(LS_INFO) << "InitPlayout";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!playing_);
+  if (!j_audio_track_->InitPlayout(audio_parameters_.sample_rate(),
+                                   audio_parameters_.channels())) {
+    RTC_LOG(LS_ERROR) << "InitPlayout failed";
+    return -1;
+  }
+  initialized_ = true;
+  return 0;
+}
+
+int32_t AudioTrackJni::StartPlayout() {
+  RTC_LOG(LS_INFO) << "StartPlayout";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!playing_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Playout can not start since InitPlayout must succeed first";
+    return 0;
+  }
+  if (!j_audio_track_->StartPlayout()) {
+    RTC_LOG(LS_ERROR) << "StartPlayout failed";
+    return -1;
+  }
+  playing_ = true;
+  return 0;
+}
+
+int32_t AudioTrackJni::StopPlayout() {
+  RTC_LOG(LS_INFO) << "StopPlayout";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !playing_) {
+    return 0;
+  }
+  if (!j_audio_track_->StopPlayout()) {
+    RTC_LOG(LS_ERROR) << "StopPlayout failed";
+    return -1;
+  }
+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
+  // next time StartRecording() is called since it will create a new Java
+  // thread.
+  thread_checker_java_.Detach();
+  initialized_ = false;
+  playing_ = false;
+  direct_buffer_address_ = nullptr;
+  return 0;
+}
+
+int AudioTrackJni::SpeakerVolumeIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+int AudioTrackJni::SetSpeakerVolume(uint32_t volume) {
+  RTC_LOG(LS_INFO) << "SetSpeakerVolume(" << volume << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_track_->SetStreamVolume(volume) ? 0 : -1;
+}
+
+int AudioTrackJni::MaxSpeakerVolume(uint32_t& max_volume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  max_volume = j_audio_track_->GetStreamMaxVolume();
+  return 0;
+}
+
+int AudioTrackJni::MinSpeakerVolume(uint32_t& min_volume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  min_volume = 0;
+  return 0;
+}
+
+int AudioTrackJni::SpeakerVolume(uint32_t& volume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  volume = j_audio_track_->GetStreamVolume();
+  RTC_LOG(LS_INFO) << "SpeakerVolume: " << volume;
+  return 0;
+}
+
+// TODO(henrika): possibly add stereo support.
+void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_LOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  RTC_LOG(LS_INFO) << "SetPlayoutSampleRate(" << sample_rate_hz << ")";
+  audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
+  const size_t channels = audio_parameters_.channels();
+  RTC_LOG(LS_INFO) << "SetPlayoutChannels(" << channels << ")";
+  audio_device_buffer_->SetPlayoutChannels(channels);
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioTrackJni::CacheDirectBufferAddress(JNIEnv* env,
+                                                     jobject obj,
+                                                     jobject byte_buffer,
+                                                     jlong nativeAudioTrack) {
+  webrtc::AudioTrackJni* this_object =
+      reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack);
+  this_object->OnCacheDirectBufferAddress(env, byte_buffer);
+}
+
+void AudioTrackJni::OnCacheDirectBufferAddress(JNIEnv* env,
+                                               jobject byte_buffer) {
+  RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!direct_buffer_address_);
+  direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
+  jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
+  RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
+  frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / bytes_per_frame;
+  RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioTrackJni::GetPlayoutData(JNIEnv* env,
+                                           jobject obj,
+                                           jint length,
+                                           jlong nativeAudioTrack) {
+  webrtc::AudioTrackJni* this_object =
+      reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack);
+  this_object->OnGetPlayoutData(static_cast<size_t>(length));
+}
+
+// This method is called on a high-priority thread from Java. The name of
+// the thread is 'AudioRecordTrack'.
+void AudioTrackJni::OnGetPlayoutData(size_t length) {
+  RTC_DCHECK(thread_checker_java_.IsCurrent());
+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
+  RTC_DCHECK_EQ(frames_per_buffer_, length / bytes_per_frame);
+  if (!audio_device_buffer_) {
+    RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
+    return;
+  }
+  // Pull decoded data (in 16-bit PCM format) from jitter buffer.
+  int samples = audio_device_buffer_->RequestPlayoutData(frames_per_buffer_);
+  if (samples <= 0) {
+    RTC_LOG(LS_ERROR) << "AudioDeviceBuffer::RequestPlayoutData failed";
+    return;
+  }
+  RTC_DCHECK_EQ(samples, frames_per_buffer_);
+  // Copy decoded data into common byte buffer to ensure that it can be
+  // written to the Java based audio track.
+  samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_);
+  RTC_DCHECK_EQ(length, bytes_per_frame * samples);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/audio_track_jni.h b/third_party/libwebrtc/modules/audio_device/android/audio_track_jni.h
new file mode 100644
index 0000000000..7eb69082b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/audio_track_jni.h
@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
+
+#include <jni.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// Implements 16-bit mono PCM audio output support for Android using the Java
+// AudioTrack interface. Most of the work is done by its Java counterpart in
+// WebRtcAudioTrack.java. This class is created and lives on a thread in
+// C++-land, but decoded audio buffers are requested on a high-priority
+// thread managed by the Java class.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread.
+//
+// This class uses JvmThreadConnector to attach to a Java VM if needed
+// and detach when the object goes out of scope. Additional thread checking
+// guarantees that no other (possibly non attached) thread is used.
+class AudioTrackJni {
+ public:
+  // Wraps the Java specific parts of the AudioTrackJni into one helper class.
+  class JavaAudioTrack {
+   public:
+    JavaAudioTrack(NativeRegistration* native_registration,
+                   std::unique_ptr<GlobalRef> audio_track);
+    ~JavaAudioTrack();
+
+    bool InitPlayout(int sample_rate, int channels);
+    bool StartPlayout();
+    bool StopPlayout();
+    bool SetStreamVolume(int volume);
+    int GetStreamMaxVolume();
+    int GetStreamVolume();
+
+   private:
+    std::unique_ptr<GlobalRef> audio_track_;
+    jmethodID init_playout_;
+    jmethodID start_playout_;
+    jmethodID stop_playout_;
+    jmethodID set_stream_volume_;
+    jmethodID get_stream_max_volume_;
+    jmethodID get_stream_volume_;
+    jmethodID get_buffer_size_in_frames_;
+  };
+
+  explicit AudioTrackJni(AudioManager* audio_manager);
+  ~AudioTrackJni();
+
+  int32_t Init();
+  int32_t Terminate();
+
+  int32_t InitPlayout();
+  bool PlayoutIsInitialized() const { return initialized_; }
+
+  int32_t StartPlayout();
+  int32_t StopPlayout();
+  bool Playing() const { return playing_; }
+
+  int SpeakerVolumeIsAvailable(bool& available);
+  int SetSpeakerVolume(uint32_t volume);
+  int SpeakerVolume(uint32_t& volume) const;
+  int MaxSpeakerVolume(uint32_t& max_volume) const;
+  int MinSpeakerVolume(uint32_t& min_volume) const;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+ private:
+  // Called from Java side so we can cache the address of the Java-manged
+  // `byte_buffer` in `direct_buffer_address_`. The size of the buffer
+  // is also stored in `direct_buffer_capacity_in_bytes_`.
+  // Called on the same thread as the creating thread.
+  static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
+                                               jobject obj,
+                                               jobject byte_buffer,
+                                               jlong nativeAudioTrack);
+  void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
+
+  // Called periodically by the Java based WebRtcAudioTrack object when
+  // playout has started. Each call indicates that `length` new bytes should
+  // be written to the memory area `direct_buffer_address_` for playout.
+  // This method is called on a high-priority thread from Java. The name of
+  // the thread is 'AudioTrackThread'.
+  static void JNICALL GetPlayoutData(JNIEnv* env,
+                                     jobject obj,
+                                     jint length,
+                                     jlong nativeAudioTrack);
+  void OnGetPlayoutData(size_t length);
+
+  // Stores thread ID in constructor.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to OnGetPlayoutData() from high-priority
+  // thread in Java. Detached during construction of this object.
+  SequenceChecker thread_checker_java_;
+
+  // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
+  // construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  std::unique_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioTrackJni class.
+  std::unique_ptr<AudioTrackJni::JavaAudioTrack> j_audio_track_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Cached copy of address to direct audio buffer owned by `j_audio_track_`.
+  void* direct_buffer_address_;
+
+  // Number of bytes in the direct audio buffer owned by `j_audio_track_`.
+  size_t direct_buffer_capacity_in_bytes_;
+
+  // Number of audio frames per audio buffer. Each audio frame corresponds to
+  // one sample of PCM mono data at 16 bits per sample. Hence, each audio
+  // frame contains 2 bytes (given that the Java layer only supports mono).
+  // Example: 480 for 48000 Hz or 441 for 44100 Hz.
+  size_t frames_per_buffer_;
+
+  bool initialized_;
+
+  bool playing_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
+  // and therefore outlives this object.
+  AudioDeviceBuffer* audio_device_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/build_info.cc b/third_party/libwebrtc/modules/audio_device/android/build_info.cc
new file mode 100644
index 0000000000..916be8244e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/build_info.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/build_info.h"
+
+#include "modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+BuildInfo::BuildInfo()
+    : j_environment_(JVM::GetInstance()->environment()),
+      j_build_info_(
+          JVM::GetInstance()->GetClass("org/webrtc/voiceengine/BuildInfo")) {}
+
+std::string BuildInfo::GetStringFromJava(const char* name) {
+  jmethodID id = j_build_info_.GetStaticMethodId(name, "()Ljava/lang/String;");
+  jstring j_string =
+      static_cast<jstring>(j_build_info_.CallStaticObjectMethod(id));
+  return j_environment_->JavaToStdString(j_string);
+}
+
+std::string BuildInfo::GetDeviceModel() {
+  return GetStringFromJava("getDeviceModel");
+}
+
+std::string BuildInfo::GetBrand() {
+  return GetStringFromJava("getBrand");
+}
+
+std::string BuildInfo::GetDeviceManufacturer() {
+  return GetStringFromJava("getDeviceManufacturer");
+}
+
+std::string BuildInfo::GetAndroidBuildId() {
+  return GetStringFromJava("getAndroidBuildId");
+}
+
+std::string BuildInfo::GetBuildType() {
+  return GetStringFromJava("getBuildType");
+}
+
+std::string BuildInfo::GetBuildRelease() {
+  return GetStringFromJava("getBuildRelease");
+}
+
+SdkCode BuildInfo::GetSdkVersion() {
+  jmethodID id = j_build_info_.GetStaticMethodId("getSdkVersion", "()I");
+  jint j_version = j_build_info_.CallStaticIntMethod(id);
+  return static_cast<SdkCode>(j_version);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/build_info.h b/third_party/libwebrtc/modules/audio_device/android/build_info.h
new file mode 100644
index 0000000000..3647e56649
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/build_info.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
+
+#include <jni.h>
+
+#include <memory>
+#include <string>
+
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// This enumeration maps to the values returned by BuildInfo::GetSdkVersion(),
+// indicating the Android release associated with a given SDK version.
+// See https://developer.android.com/guide/topics/manifest/uses-sdk-element.html
+// for details.
+enum SdkCode {
+  SDK_CODE_JELLY_BEAN = 16,      // Android 4.1
+  SDK_CODE_JELLY_BEAN_MR1 = 17,  // Android 4.2
+  SDK_CODE_JELLY_BEAN_MR2 = 18,  // Android 4.3
+  SDK_CODE_KITKAT = 19,          // Android 4.4
+  SDK_CODE_WATCH = 20,           // Android 4.4W
+  SDK_CODE_LOLLIPOP = 21,        // Android 5.0
+  SDK_CODE_LOLLIPOP_MR1 = 22,    // Android 5.1
+  SDK_CODE_MARSHMALLOW = 23,     // Android 6.0
+  SDK_CODE_N = 24,
+};
+
+// Utility class used to query the Java class (org/webrtc/voiceengine/BuildInfo)
+// for device and Android build information.
+// The calling thread is attached to the JVM at construction if needed and a
+// valid Java environment object is also created.
+// All Get methods must be called on the creating thread. If not, the code will
+// hit RTC_DCHECKs when calling JNIEnvironment::JavaToStdString().
+class BuildInfo {
+ public:
+  BuildInfo();
+  ~BuildInfo() {}
+
+  // End-user-visible name for the end product (e.g. "Nexus 6").
+  std::string GetDeviceModel();
+  // Consumer-visible brand (e.g. "google").
+  std::string GetBrand();
+  // Manufacturer of the product/hardware (e.g. "motorola").
+  std::string GetDeviceManufacturer();
+  // Android build ID (e.g. LMY47D).
+  std::string GetAndroidBuildId();
+  // The type of build (e.g. "user" or "eng").
+  std::string GetBuildType();
+  // The user-visible version string (e.g. "5.1").
+  std::string GetBuildRelease();
+  // The user-visible SDK version of the framework (e.g. 21). See SdkCode enum
+  // for translation.
+  SdkCode GetSdkVersion();
+
+ private:
+  // Helper method which calls a static getter method with `name` and returns
+  // a string from Java.
+  std::string GetStringFromJava(const char* name);
+
+  // Ensures that this class can access a valid JNI interface pointer even
+  // if the creating thread was not attached to the JVM.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Provides access to the JNIEnv interface pointer and the JavaToStdString()
+  // method which is used to translate Java strings to std strings.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Holds the jclass object and provides access to CallStaticObjectMethod().
+  // Used by GetStringFromJava() during construction only.
+  JavaClass j_build_info_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/ensure_initialized.cc b/third_party/libwebrtc/modules/audio_device/android/ensure_initialized.cc
new file mode 100644
index 0000000000..59e9c8f7a6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/ensure_initialized.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/ensure_initialized.h"
+
+#include <jni.h>
+#include <pthread.h>
+#include <stddef.h>
+
+#include "modules/utility/include/jvm_android.h"
+#include "rtc_base/checks.h"
+#include "sdk/android/src/jni/jvm.h"
+
+namespace webrtc {
+namespace audiodevicemodule {
+
+static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
+
+void EnsureInitializedOnce() {
+  RTC_CHECK(::webrtc::jni::GetJVM() != nullptr);
+
+  JNIEnv* jni = ::webrtc::jni::AttachCurrentThreadIfNeeded();
+  JavaVM* jvm = NULL;
+  RTC_CHECK_EQ(0, jni->GetJavaVM(&jvm));
+
+  // Initialize the Java environment (currently only used by the audio manager).
+  webrtc::JVM::Initialize(jvm);
+}
+
+void EnsureInitialized() {
+  RTC_CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
+}
+
+}  // namespace audiodevicemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/ensure_initialized.h b/third_party/libwebrtc/modules/audio_device/android/ensure_initialized.h
new file mode 100644
index 0000000000..c1997b4acd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/ensure_initialized.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+namespace webrtc {
+namespace audiodevicemodule {
+
+void EnsureInitialized();
+
+}  // namespace audiodevicemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java
new file mode 100644
index 0000000000..aed8a06454
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.voiceengine;
+
+import android.os.Build;
+
+public final class BuildInfo {
+  public static String getDevice() {
+    return Build.DEVICE;
+  }
+
+  public static String getDeviceModel() {
+    return Build.MODEL;
+  }
+
+  public static String getProduct() {
+    return Build.PRODUCT;
+  }
+
+  public static String getBrand() {
+    return Build.BRAND;
+  }
+
+  public static String getDeviceManufacturer() {
+    return Build.MANUFACTURER;
+  }
+
+  public static String getAndroidBuildId() {
+    return Build.ID;
+  }
+
+  public static String getBuildType() {
+    return Build.TYPE;
+  }
+
+  public static String getBuildRelease() {
+    return Build.VERSION.RELEASE;
+  }
+
+  public static int getSdkVersion() {
+    return Build.VERSION.SDK_INT;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
new file mode 100644
index 0000000000..92f1c93524
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
@@ -0,0 +1,312 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.voiceengine;
+
+import android.media.audiofx.AcousticEchoCanceler;
+import android.media.audiofx.AudioEffect;
+import android.media.audiofx.AudioEffect.Descriptor;
+import android.media.audiofx.NoiseSuppressor;
+import android.os.Build;
+import androidx.annotation.Nullable;
+import java.util.List;
+import java.util.UUID;
+import org.webrtc.Logging;
+
+// This class wraps control of three different platform effects. Supported
+// effects are: AcousticEchoCanceler (AEC) and NoiseSuppressor (NS).
+// Calling enable() will active all effects that are
+// supported by the device if the corresponding `shouldEnableXXX` member is set.
+public class WebRtcAudioEffects {
+  private static final boolean DEBUG = false;
+
+  private static final String TAG = "WebRtcAudioEffects";
+
+  // UUIDs for Software Audio Effects that we want to avoid using.
+  // The implementor field will be set to "The Android Open Source Project".
+  private static final UUID AOSP_ACOUSTIC_ECHO_CANCELER =
+      UUID.fromString("bb392ec0-8d4d-11e0-a896-0002a5d5c51b");
+  private static final UUID AOSP_NOISE_SUPPRESSOR =
+      UUID.fromString("c06c8400-8e06-11e0-9cb6-0002a5d5c51b");
+
+  // Contains the available effect descriptors returned from the
+  // AudioEffect.getEffects() call. This result is cached to avoid doing the
+  // slow OS call multiple times.
+  private static @Nullable Descriptor[] cachedEffects;
+
+  // Contains the audio effect objects. Created in enable() and destroyed
+  // in release().
+  private @Nullable AcousticEchoCanceler aec;
+  private @Nullable NoiseSuppressor ns;
+
+  // Affects the final state given to the setEnabled() method on each effect.
+  // The default state is set to "disabled" but each effect can also be enabled
+  // by calling setAEC() and setNS().
+  // To enable an effect, both the shouldEnableXXX member and the static
+  // canUseXXX() must be true.
+  private boolean shouldEnableAec;
+  private boolean shouldEnableNs;
+
+  // Checks if the device implements Acoustic Echo Cancellation (AEC).
+  // Returns true if the device implements AEC, false otherwise.
+  public static boolean isAcousticEchoCancelerSupported() {
+    // Note: we're using isAcousticEchoCancelerEffectAvailable() instead of
+    // AcousticEchoCanceler.isAvailable() to avoid the expensive getEffects()
+    // OS API call.
+    return isAcousticEchoCancelerEffectAvailable();
+  }
+
+  // Checks if the device implements Noise Suppression (NS).
+  // Returns true if the device implements NS, false otherwise.
+  public static boolean isNoiseSuppressorSupported() {
+    // Note: we're using isNoiseSuppressorEffectAvailable() instead of
+    // NoiseSuppressor.isAvailable() to avoid the expensive getEffects()
+    // OS API call.
+    return isNoiseSuppressorEffectAvailable();
+  }
+
+  // Returns true if the device is blacklisted for HW AEC usage.
+  public static boolean isAcousticEchoCancelerBlacklisted() {
+    List<String> blackListedModels = WebRtcAudioUtils.getBlackListedModelsForAecUsage();
+    boolean isBlacklisted = blackListedModels.contains(Build.MODEL);
+    if (isBlacklisted) {
+      Logging.w(TAG, Build.MODEL + " is blacklisted for HW AEC usage!");
+    }
+    return isBlacklisted;
+  }
+
+  // Returns true if the device is blacklisted for HW NS usage.
+  public static boolean isNoiseSuppressorBlacklisted() {
+    List<String> blackListedModels = WebRtcAudioUtils.getBlackListedModelsForNsUsage();
+    boolean isBlacklisted = blackListedModels.contains(Build.MODEL);
+    if (isBlacklisted) {
+      Logging.w(TAG, Build.MODEL + " is blacklisted for HW NS usage!");
+    }
+    return isBlacklisted;
+  }
+
+  // Returns true if the platform AEC should be excluded based on its UUID.
+  // AudioEffect.queryEffects() can throw IllegalStateException.
+  private static boolean isAcousticEchoCancelerExcludedByUUID() {
+    for (Descriptor d : getAvailableEffects()) {
+      if (d.type.equals(AudioEffect.EFFECT_TYPE_AEC)
+          && d.uuid.equals(AOSP_ACOUSTIC_ECHO_CANCELER)) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  // Returns true if the platform NS should be excluded based on its UUID.
+  // AudioEffect.queryEffects() can throw IllegalStateException.
+  private static boolean isNoiseSuppressorExcludedByUUID() {
+    for (Descriptor d : getAvailableEffects()) {
+      if (d.type.equals(AudioEffect.EFFECT_TYPE_NS) && d.uuid.equals(AOSP_NOISE_SUPPRESSOR)) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  // Returns true if the device supports Acoustic Echo Cancellation (AEC).
+  private static boolean isAcousticEchoCancelerEffectAvailable() {
+    return isEffectTypeAvailable(AudioEffect.EFFECT_TYPE_AEC);
+  }
+
+  // Returns true if the device supports Noise Suppression (NS).
+  private static boolean isNoiseSuppressorEffectAvailable() {
+    return isEffectTypeAvailable(AudioEffect.EFFECT_TYPE_NS);
+  }
+
+  // Returns true if all conditions for supporting the HW AEC are fulfilled.
+  // It will not be possible to enable the HW AEC if this method returns false.
+  public static boolean canUseAcousticEchoCanceler() {
+    boolean canUseAcousticEchoCanceler = isAcousticEchoCancelerSupported()
+        && !WebRtcAudioUtils.useWebRtcBasedAcousticEchoCanceler()
+        && !isAcousticEchoCancelerBlacklisted() && !isAcousticEchoCancelerExcludedByUUID();
+    Logging.d(TAG, "canUseAcousticEchoCanceler: " + canUseAcousticEchoCanceler);
+    return canUseAcousticEchoCanceler;
+  }
+
+  // Returns true if all conditions for supporting the HW NS are fulfilled.
+  // It will not be possible to enable the HW NS if this method returns false.
+  public static boolean canUseNoiseSuppressor() {
+    boolean canUseNoiseSuppressor = isNoiseSuppressorSupported()
+        && !WebRtcAudioUtils.useWebRtcBasedNoiseSuppressor() && !isNoiseSuppressorBlacklisted()
+        && !isNoiseSuppressorExcludedByUUID();
+    Logging.d(TAG, "canUseNoiseSuppressor: " + canUseNoiseSuppressor);
+    return canUseNoiseSuppressor;
+  }
+
+  public static WebRtcAudioEffects create() {
+    return new WebRtcAudioEffects();
+  }
+
+  private WebRtcAudioEffects() {
+    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
+  }
+
+  // Call this method to enable or disable the platform AEC. It modifies
+  // `shouldEnableAec` which is used in enable() where the actual state
+  // of the AEC effect is modified. Returns true if HW AEC is supported and
+  // false otherwise.
+  public boolean setAEC(boolean enable) {
+    Logging.d(TAG, "setAEC(" + enable + ")");
+    if (!canUseAcousticEchoCanceler()) {
+      Logging.w(TAG, "Platform AEC is not supported");
+      shouldEnableAec = false;
+      return false;
+    }
+    if (aec != null && (enable != shouldEnableAec)) {
+      Logging.e(TAG, "Platform AEC state can't be modified while recording");
+      return false;
+    }
+    shouldEnableAec = enable;
+    return true;
+  }
+
+  // Call this method to enable or disable the platform NS. It modifies
+  // `shouldEnableNs` which is used in enable() where the actual state
+  // of the NS effect is modified. Returns true if HW NS is supported and
+  // false otherwise.
+  public boolean setNS(boolean enable) {
+    Logging.d(TAG, "setNS(" + enable + ")");
+    if (!canUseNoiseSuppressor()) {
+      Logging.w(TAG, "Platform NS is not supported");
+      shouldEnableNs = false;
+      return false;
+    }
+    if (ns != null && (enable != shouldEnableNs)) {
+      Logging.e(TAG, "Platform NS state can't be modified while recording");
+      return false;
+    }
+    shouldEnableNs = enable;
+    return true;
+  }
+
+  public void enable(int audioSession) {
+    Logging.d(TAG, "enable(audioSession=" + audioSession + ")");
+    assertTrue(aec == null);
+    assertTrue(ns == null);
+
+    if (DEBUG) {
+      // Add logging of supported effects but filter out "VoIP effects", i.e.,
+      // AEC, AEC and NS. Avoid calling AudioEffect.queryEffects() unless the
+      // DEBUG flag is set since we have seen crashes in this API.
+      for (Descriptor d : AudioEffect.queryEffects()) {
+        if (effectTypeIsVoIP(d.type)) {
+          Logging.d(TAG, "name: " + d.name + ", "
+                  + "mode: " + d.connectMode + ", "
+                  + "implementor: " + d.implementor + ", "
+                  + "UUID: " + d.uuid);
+        }
+      }
+    }
+
+    if (isAcousticEchoCancelerSupported()) {
+      // Create an AcousticEchoCanceler and attach it to the AudioRecord on
+      // the specified audio session.
+      aec = AcousticEchoCanceler.create(audioSession);
+      if (aec != null) {
+        boolean enabled = aec.getEnabled();
+        boolean enable = shouldEnableAec && canUseAcousticEchoCanceler();
+        if (aec.setEnabled(enable) != AudioEffect.SUCCESS) {
+          Logging.e(TAG, "Failed to set the AcousticEchoCanceler state");
+        }
+        Logging.d(TAG, "AcousticEchoCanceler: was " + (enabled ? "enabled" : "disabled")
+                + ", enable: " + enable + ", is now: "
+                + (aec.getEnabled() ? "enabled" : "disabled"));
+      } else {
+        Logging.e(TAG, "Failed to create the AcousticEchoCanceler instance");
+      }
+    }
+
+    if (isNoiseSuppressorSupported()) {
+      // Create an NoiseSuppressor and attach it to the AudioRecord on the
+      // specified audio session.
+      ns = NoiseSuppressor.create(audioSession);
+      if (ns != null) {
+        boolean enabled = ns.getEnabled();
+        boolean enable = shouldEnableNs && canUseNoiseSuppressor();
+        if (ns.setEnabled(enable) != AudioEffect.SUCCESS) {
+          Logging.e(TAG, "Failed to set the NoiseSuppressor state");
+        }
+        Logging.d(TAG, "NoiseSuppressor: was " + (enabled ? "enabled" : "disabled") + ", enable: "
+                + enable + ", is now: " + (ns.getEnabled() ? "enabled" : "disabled"));
+      } else {
+        Logging.e(TAG, "Failed to create the NoiseSuppressor instance");
+      }
+    }
+  }
+
+  // Releases all native audio effect resources. It is a good practice to
+  // release the effect engine when not in use as control can be returned
+  // to other applications or the native resources released.
+  public void release() {
+    Logging.d(TAG, "release");
+    if (aec != null) {
+      aec.release();
+      aec = null;
+    }
+    if (ns != null) {
+      ns.release();
+      ns = null;
+    }
+  }
+
+  // Returns true for effect types in `type` that are of "VoIP" types:
+  // Acoustic Echo Canceler (AEC) or Automatic Gain Control (AGC) or
+  // Noise Suppressor (NS). Note that, an extra check for support is needed
+  // in each comparison since some devices includes effects in the
+  // AudioEffect.Descriptor array that are actually not available on the device.
+  // As an example: Samsung Galaxy S6 includes an AGC in the descriptor but
+  // AutomaticGainControl.isAvailable() returns false.
+  private boolean effectTypeIsVoIP(UUID type) {
+    return (AudioEffect.EFFECT_TYPE_AEC.equals(type) && isAcousticEchoCancelerSupported())
+        || (AudioEffect.EFFECT_TYPE_NS.equals(type) && isNoiseSuppressorSupported());
+  }
+
+  // Helper method which throws an exception when an assertion has failed.
+  private static void assertTrue(boolean condition) {
+    if (!condition) {
+      throw new AssertionError("Expected condition to be true");
+    }
+  }
+
+  // Returns the cached copy of the audio effects array, if available, or
+  // queries the operating system for the list of effects.
+  private static @Nullable Descriptor[] getAvailableEffects() {
+    if (cachedEffects != null) {
+      return cachedEffects;
+    }
+    // The caching is best effort only - if this method is called from several
+    // threads in parallel, they may end up doing the underlying OS call
+    // multiple times. It's normally only called on one thread so there's no
+    // real need to optimize for the multiple threads case.
+    cachedEffects = AudioEffect.queryEffects();
+    return cachedEffects;
+  }
+
+  // Returns true if an effect of the specified type is available. Functionally
+  // equivalent to (NoiseSuppressor`AutomaticGainControl`...).isAvailable(), but
+  // faster as it avoids the expensive OS call to enumerate effects.
+  private static boolean isEffectTypeAvailable(UUID effectType) {
+    Descriptor[] effects = getAvailableEffects();
+    if (effects == null) {
+      return false;
+    }
+    for (Descriptor d : effects) {
+      if (d.type.equals(effectType)) {
+        return true;
+      }
+    }
+    return false;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
new file mode 100644
index 0000000000..43c416f5b1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
@@ -0,0 +1,371 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.voiceengine;
+
+import android.content.Context;
+import android.content.pm.PackageManager;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.AudioTrack;
+import android.os.Build;
+import androidx.annotation.Nullable;
+import java.util.Timer;
+import java.util.TimerTask;
+import org.webrtc.ContextUtils;
+import org.webrtc.Logging;
+
+// WebRtcAudioManager handles tasks that uses android.media.AudioManager.
+// At construction, storeAudioParameters() is called and it retrieves
+// fundamental audio parameters like native sample rate and number of channels.
+// The result is then provided to the caller by nativeCacheAudioParameters().
+// It is also possible to call init() to set up the audio environment for best
+// possible "VoIP performance". All settings done in init() are reverted by
+// dispose(). This class can also be used without calling init() if the user
+// prefers to set up the audio environment separately. However, it is
+// recommended to always use AudioManager.MODE_IN_COMMUNICATION.
+public class WebRtcAudioManager {
+  private static final boolean DEBUG = false;
+
+  private static final String TAG = "WebRtcAudioManager";
+
+  // TODO(bugs.webrtc.org/8914): disabled by default until AAudio support has
+  // been completed. Goal is to always return false on Android O MR1 and higher.
+  private static final boolean blacklistDeviceForAAudioUsage = true;
+
+  // Use mono as default for both audio directions.
+  private static boolean useStereoOutput;
+  private static boolean useStereoInput;
+
+  private static boolean blacklistDeviceForOpenSLESUsage;
+  private static boolean blacklistDeviceForOpenSLESUsageIsOverridden;
+
+  // Call this method to override the default list of blacklisted devices
+  // specified in WebRtcAudioUtils.BLACKLISTED_OPEN_SL_ES_MODELS.
+  // Allows an app to take control over which devices to exclude from using
+  // the OpenSL ES audio output path
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized void setBlacklistDeviceForOpenSLESUsage(boolean enable) {
+    blacklistDeviceForOpenSLESUsageIsOverridden = true;
+    blacklistDeviceForOpenSLESUsage = enable;
+  }
+
+  // Call these methods to override the default mono audio modes for the specified direction(s)
+  // (input and/or output).
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized void setStereoOutput(boolean enable) {
+    Logging.w(TAG, "Overriding default output behavior: setStereoOutput(" + enable + ')');
+    useStereoOutput = enable;
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized void setStereoInput(boolean enable) {
+    Logging.w(TAG, "Overriding default input behavior: setStereoInput(" + enable + ')');
+    useStereoInput = enable;
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized boolean getStereoOutput() {
+    return useStereoOutput;
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized boolean getStereoInput() {
+    return useStereoInput;
+  }
+
+  // Default audio data format is PCM 16 bit per sample.
+  // Guaranteed to be supported by all devices.
+  private static final int BITS_PER_SAMPLE = 16;
+
+  private static final int DEFAULT_FRAME_PER_BUFFER = 256;
+
+  // Private utility class that periodically checks and logs the volume level
+  // of the audio stream that is currently controlled by the volume control.
+  // A timer triggers logs once every 30 seconds and the timer's associated
+  // thread is named "WebRtcVolumeLevelLoggerThread".
+  private static class VolumeLogger {
+    private static final String THREAD_NAME = "WebRtcVolumeLevelLoggerThread";
+    private static final int TIMER_PERIOD_IN_SECONDS = 30;
+
+    private final AudioManager audioManager;
+    private @Nullable Timer timer;
+
+    public VolumeLogger(AudioManager audioManager) {
+      this.audioManager = audioManager;
+    }
+
+    public void start() {
+      timer = new Timer(THREAD_NAME);
+      timer.schedule(new LogVolumeTask(audioManager.getStreamMaxVolume(AudioManager.STREAM_RING),
+                         audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL)),
+          0, TIMER_PERIOD_IN_SECONDS * 1000);
+    }
+
+    private class LogVolumeTask extends TimerTask {
+      private final int maxRingVolume;
+      private final int maxVoiceCallVolume;
+
+      LogVolumeTask(int maxRingVolume, int maxVoiceCallVolume) {
+        this.maxRingVolume = maxRingVolume;
+        this.maxVoiceCallVolume = maxVoiceCallVolume;
+      }
+
+      @Override
+      public void run() {
+        final int mode = audioManager.getMode();
+        if (mode == AudioManager.MODE_RINGTONE) {
+          Logging.d(TAG, "STREAM_RING stream volume: "
+                  + audioManager.getStreamVolume(AudioManager.STREAM_RING) + " (max="
+                  + maxRingVolume + ")");
+        } else if (mode == AudioManager.MODE_IN_COMMUNICATION) {
+          Logging.d(TAG, "VOICE_CALL stream volume: "
+                  + audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL) + " (max="
+                  + maxVoiceCallVolume + ")");
+        }
+      }
+    }
+
+    private void stop() {
+      if (timer != null) {
+        timer.cancel();
+        timer = null;
+      }
+    }
+  }
+
+  private final long nativeAudioManager;
+  private final AudioManager audioManager;
+
+  private boolean initialized;
+  private int nativeSampleRate;
+  private int nativeChannels;
+
+  private boolean hardwareAEC;
+  private boolean hardwareAGC;
+  private boolean hardwareNS;
+  private boolean lowLatencyOutput;
+  private boolean lowLatencyInput;
+  private boolean proAudio;
+  private boolean aAudio;
+  private int sampleRate;
+  private int outputChannels;
+  private int inputChannels;
+  private int outputBufferSize;
+  private int inputBufferSize;
+
+  private final VolumeLogger volumeLogger;
+
+  WebRtcAudioManager(long nativeAudioManager) {
+    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
+    this.nativeAudioManager = nativeAudioManager;
+    audioManager =
+        (AudioManager) ContextUtils.getApplicationContext().getSystemService(Context.AUDIO_SERVICE);
+    if (DEBUG) {
+      WebRtcAudioUtils.logDeviceInfo(TAG);
+    }
+    volumeLogger = new VolumeLogger(audioManager);
+    storeAudioParameters();
+    nativeCacheAudioParameters(sampleRate, outputChannels, inputChannels, hardwareAEC, hardwareAGC,
+        hardwareNS, lowLatencyOutput, lowLatencyInput, proAudio, aAudio, outputBufferSize,
+        inputBufferSize, nativeAudioManager);
+    WebRtcAudioUtils.logAudioState(TAG);
+  }
+
+  private boolean init() {
+    Logging.d(TAG, "init" + WebRtcAudioUtils.getThreadInfo());
+    if (initialized) {
+      return true;
+    }
+    Logging.d(TAG, "audio mode is: "
+        + WebRtcAudioUtils.modeToString(audioManager.getMode()));
+    initialized = true;
+    volumeLogger.start();
+    return true;
+  }
+
+  private void dispose() {
+    Logging.d(TAG, "dispose" + WebRtcAudioUtils.getThreadInfo());
+    if (!initialized) {
+      return;
+    }
+    volumeLogger.stop();
+  }
+
+  private boolean isCommunicationModeEnabled() {
+    return (audioManager.getMode() == AudioManager.MODE_IN_COMMUNICATION);
+  }
+
+  private boolean isDeviceBlacklistedForOpenSLESUsage() {
+    boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden
+        ? blacklistDeviceForOpenSLESUsage
+        : WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
+    if (blacklisted) {
+      Logging.d(TAG, Build.MODEL + " is blacklisted for OpenSL ES usage!");
+    }
+    return blacklisted;
+  }
+
+  private void storeAudioParameters() {
+    outputChannels = getStereoOutput() ? 2 : 1;
+    inputChannels = getStereoInput() ? 2 : 1;
+    sampleRate = getNativeOutputSampleRate();
+    hardwareAEC = isAcousticEchoCancelerSupported();
+    // TODO(henrika): use of hardware AGC is no longer supported. Currently
+    // hardcoded to false. To be removed.
+    hardwareAGC = false;
+    hardwareNS = isNoiseSuppressorSupported();
+    lowLatencyOutput = isLowLatencyOutputSupported();
+    lowLatencyInput = isLowLatencyInputSupported();
+    proAudio = isProAudioSupported();
+    aAudio = isAAudioSupported();
+    outputBufferSize = lowLatencyOutput ? getLowLatencyOutputFramesPerBuffer()
+                                        : getMinOutputFrameSize(sampleRate, outputChannels);
+    inputBufferSize = lowLatencyInput ? getLowLatencyInputFramesPerBuffer()
+                                      : getMinInputFrameSize(sampleRate, inputChannels);
+  }
+
+  // Gets the current earpiece state.
+  private boolean hasEarpiece() {
+    return ContextUtils.getApplicationContext().getPackageManager().hasSystemFeature(
+        PackageManager.FEATURE_TELEPHONY);
+  }
+
+  // Returns true if low-latency audio output is supported.
+  private boolean isLowLatencyOutputSupported() {
+    return ContextUtils.getApplicationContext().getPackageManager().hasSystemFeature(
+        PackageManager.FEATURE_AUDIO_LOW_LATENCY);
+  }
+
+  // Returns true if low-latency audio input is supported.
+  // TODO(henrika): remove the hardcoded false return value when OpenSL ES
+  // input performance has been evaluated and tested more.
+  public boolean isLowLatencyInputSupported() {
+    // TODO(henrika): investigate if some sort of device list is needed here
+    // as well. The NDK doc states that: "As of API level 21, lower latency
+    // audio input is supported on select devices. To take advantage of this
+    // feature, first confirm that lower latency output is available".
+    return isLowLatencyOutputSupported();
+  }
+
+  // Returns true if the device has professional audio level of functionality
+  // and therefore supports the lowest possible round-trip latency.
+  private boolean isProAudioSupported() {
+    return Build.VERSION.SDK_INT >= 23
+        && ContextUtils.getApplicationContext().getPackageManager().hasSystemFeature(
+               PackageManager.FEATURE_AUDIO_PRO);
+  }
+
+  // AAudio is supported on Androio Oreo MR1 (API 27) and higher.
+  // TODO(bugs.webrtc.org/8914): currently disabled by default.
+  private boolean isAAudioSupported() {
+    if (blacklistDeviceForAAudioUsage) {
+      Logging.w(TAG, "AAudio support is currently disabled on all devices!");
+    }
+    return !blacklistDeviceForAAudioUsage && Build.VERSION.SDK_INT >= 27;
+  }
+
+  // Returns the native output sample rate for this device's output stream.
+  private int getNativeOutputSampleRate() {
+    // Override this if we're running on an old emulator image which only
+    // supports 8 kHz and doesn't support PROPERTY_OUTPUT_SAMPLE_RATE.
+    if (WebRtcAudioUtils.runningOnEmulator()) {
+      Logging.d(TAG, "Running emulator, overriding sample rate to 8 kHz.");
+      return 8000;
+    }
+    // Default can be overriden by WebRtcAudioUtils.setDefaultSampleRateHz().
+    // If so, use that value and return here.
+    if (WebRtcAudioUtils.isDefaultSampleRateOverridden()) {
+      Logging.d(TAG, "Default sample rate is overriden to "
+              + WebRtcAudioUtils.getDefaultSampleRateHz() + " Hz");
+      return WebRtcAudioUtils.getDefaultSampleRateHz();
+    }
+    // No overrides available. Deliver best possible estimate based on default
+    // Android AudioManager APIs.
+    final int sampleRateHz = getSampleRateForApiLevel();
+    Logging.d(TAG, "Sample rate is set to " + sampleRateHz + " Hz");
+    return sampleRateHz;
+  }
+
+  private int getSampleRateForApiLevel() {
+    String sampleRateString = audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
+    return (sampleRateString == null) ? WebRtcAudioUtils.getDefaultSampleRateHz()
+                                      : Integer.parseInt(sampleRateString);
+  }
+
+  // Returns the native output buffer size for low-latency output streams.
+  private int getLowLatencyOutputFramesPerBuffer() {
+    assertTrue(isLowLatencyOutputSupported());
+    String framesPerBuffer =
+        audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
+    return framesPerBuffer == null ? DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer);
+  }
+
+  // Returns true if the device supports an audio effect (AEC or NS).
+  // Four conditions must be fulfilled if functions are to return true:
+  // 1) the platform must support the built-in (HW) effect,
+  // 2) explicit use (override) of a WebRTC based version must not be set,
+  // 3) the device must not be blacklisted for use of the effect, and
+  // 4) the UUID of the effect must be approved (some UUIDs can be excluded).
+  private static boolean isAcousticEchoCancelerSupported() {
+    return WebRtcAudioEffects.canUseAcousticEchoCanceler();
+  }
+  private static boolean isNoiseSuppressorSupported() {
+    return WebRtcAudioEffects.canUseNoiseSuppressor();
+  }
+
+  // Returns the minimum output buffer size for Java based audio (AudioTrack).
+  // This size can also be used for OpenSL ES implementations on devices that
+  // lacks support of low-latency output.
+  private static int getMinOutputFrameSize(int sampleRateInHz, int numChannels) {
+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
+    final int channelConfig =
+        (numChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO);
+    return AudioTrack.getMinBufferSize(
+               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
+        / bytesPerFrame;
+  }
+
+  // Returns the native input buffer size for input streams.
+  private int getLowLatencyInputFramesPerBuffer() {
+    assertTrue(isLowLatencyInputSupported());
+    return getLowLatencyOutputFramesPerBuffer();
+  }
+
+  // Returns the minimum input buffer size for Java based audio (AudioRecord).
+  // This size can calso be used for OpenSL ES implementations on devices that
+  // lacks support of low-latency input.
+  private static int getMinInputFrameSize(int sampleRateInHz, int numChannels) {
+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
+    final int channelConfig =
+        (numChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
+    return AudioRecord.getMinBufferSize(
+               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
+        / bytesPerFrame;
+  }
+
+  // Helper method which throws an exception  when an assertion has failed.
+  private static void assertTrue(boolean condition) {
+    if (!condition) {
+      throw new AssertionError("Expected condition to be true");
+    }
+  }
+
+  private native void nativeCacheAudioParameters(int sampleRate, int outputChannels,
+      int inputChannels, boolean hardwareAEC, boolean hardwareAGC, boolean hardwareNS,
+      boolean lowLatencyOutput, boolean lowLatencyInput, boolean proAudio, boolean aAudio,
+      int outputBufferSize, int inputBufferSize, long nativeAudioManager);
+}
diff --git a/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
new file mode 100644
index 0000000000..8eab01cd69
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
@@ -0,0 +1,409 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.voiceengine;
+
+import android.media.AudioFormat;
+import android.media.AudioRecord;
+import android.media.MediaRecorder.AudioSource;
+import android.os.Build;
+import android.os.Process;
+import androidx.annotation.Nullable;
+import java.lang.System;
+import java.nio.ByteBuffer;
+import java.util.Arrays;
+import java.util.concurrent.TimeUnit;
+import org.webrtc.Logging;
+import org.webrtc.ThreadUtils;
+
+public class WebRtcAudioRecord {
+  private static final boolean DEBUG = false;
+
+  private static final String TAG = "WebRtcAudioRecord";
+
+  // Default audio data format is PCM 16 bit per sample.
+  // Guaranteed to be supported by all devices.
+  private static final int BITS_PER_SAMPLE = 16;
+
+  // Requested size of each recorded buffer provided to the client.
+  private static final int CALLBACK_BUFFER_SIZE_MS = 10;
+
+  // Average number of callbacks per second.
+  private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
+
+  // We ask for a native buffer size of BUFFER_SIZE_FACTOR * (minimum required
+  // buffer size). The extra space is allocated to guard against glitches under
+  // high load.
+  private static final int BUFFER_SIZE_FACTOR = 2;
+
+  // The AudioRecordJavaThread is allowed to wait for successful call to join()
+  // but the wait times out afther this amount of time.
+  private static final long AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS = 2000;
+
+  private static final int DEFAULT_AUDIO_SOURCE = getDefaultAudioSource();
+  private static int audioSource = DEFAULT_AUDIO_SOURCE;
+
+  private final long nativeAudioRecord;
+
+  private @Nullable WebRtcAudioEffects effects;
+
+  private ByteBuffer byteBuffer;
+
+  private @Nullable AudioRecord audioRecord;
+  private @Nullable AudioRecordThread audioThread;
+
+  private static volatile boolean microphoneMute;
+  private byte[] emptyBytes;
+
+  // Audio recording error handler functions.
+  public enum AudioRecordStartErrorCode {
+    AUDIO_RECORD_START_EXCEPTION,
+    AUDIO_RECORD_START_STATE_MISMATCH,
+  }
+
+  public static interface WebRtcAudioRecordErrorCallback {
+    void onWebRtcAudioRecordInitError(String errorMessage);
+    void onWebRtcAudioRecordStartError(AudioRecordStartErrorCode errorCode, String errorMessage);
+    void onWebRtcAudioRecordError(String errorMessage);
+  }
+
+  private static @Nullable WebRtcAudioRecordErrorCallback errorCallback;
+
+  public static void setErrorCallback(WebRtcAudioRecordErrorCallback errorCallback) {
+    Logging.d(TAG, "Set error callback");
+    WebRtcAudioRecord.errorCallback = errorCallback;
+  }
+
+  /**
+   * Contains audio sample information. Object is passed using {@link
+   * WebRtcAudioRecord.WebRtcAudioRecordSamplesReadyCallback}
+   */
+  public static class AudioSamples {
+    /** See {@link AudioRecord#getAudioFormat()} */
+    private final int audioFormat;
+    /** See {@link AudioRecord#getChannelCount()} */
+    private final int channelCount;
+    /** See {@link AudioRecord#getSampleRate()} */
+    private final int sampleRate;
+
+    private final byte[] data;
+
+    private AudioSamples(AudioRecord audioRecord, byte[] data) {
+      this.audioFormat = audioRecord.getAudioFormat();
+      this.channelCount = audioRecord.getChannelCount();
+      this.sampleRate = audioRecord.getSampleRate();
+      this.data = data;
+    }
+
+    public int getAudioFormat() {
+      return audioFormat;
+    }
+
+    public int getChannelCount() {
+      return channelCount;
+    }
+
+    public int getSampleRate() {
+      return sampleRate;
+    }
+
+    public byte[] getData() {
+      return data;
+    }
+  }
+
+  /** Called when new audio samples are ready. This should only be set for debug purposes */
+  public static interface WebRtcAudioRecordSamplesReadyCallback {
+    void onWebRtcAudioRecordSamplesReady(AudioSamples samples);
+  }
+
+  private static @Nullable WebRtcAudioRecordSamplesReadyCallback audioSamplesReadyCallback;
+
+  public static void setOnAudioSamplesReady(WebRtcAudioRecordSamplesReadyCallback callback) {
+    audioSamplesReadyCallback = callback;
+  }
+
+  /**
+   * Audio thread which keeps calling ByteBuffer.read() waiting for audio
+   * to be recorded. Feeds recorded data to the native counterpart as a
+   * periodic sequence of callbacks using DataIsRecorded().
+   * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
+   */
+  private class AudioRecordThread extends Thread {
+    private volatile boolean keepAlive = true;
+
+    public AudioRecordThread(String name) {
+      super(name);
+    }
+
+    // TODO(titovartem) make correct fix during webrtc:9175
+    @SuppressWarnings("ByteBufferBackingArray")
+    @Override
+    public void run() {
+      Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
+      Logging.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
+      assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING);
+
+      long lastTime = System.nanoTime();
+      while (keepAlive) {
+        int bytesRead = audioRecord.read(byteBuffer, byteBuffer.capacity());
+        if (bytesRead == byteBuffer.capacity()) {
+          if (microphoneMute) {
+            byteBuffer.clear();
+            byteBuffer.put(emptyBytes);
+          }
+          // It's possible we've been shut down during the read, and stopRecording() tried and
+          // failed to join this thread. To be a bit safer, try to avoid calling any native methods
+          // in case they've been unregistered after stopRecording() returned.
+          if (keepAlive) {
+            nativeDataIsRecorded(bytesRead, nativeAudioRecord);
+          }
+          if (audioSamplesReadyCallback != null) {
+            // Copy the entire byte buffer array.  Assume that the start of the byteBuffer is
+            // at index 0.
+            byte[] data = Arrays.copyOf(byteBuffer.array(), byteBuffer.capacity());
+            audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(
+                new AudioSamples(audioRecord, data));
+          }
+        } else {
+          String errorMessage = "AudioRecord.read failed: " + bytesRead;
+          Logging.e(TAG, errorMessage);
+          if (bytesRead == AudioRecord.ERROR_INVALID_OPERATION) {
+            keepAlive = false;
+            reportWebRtcAudioRecordError(errorMessage);
+          }
+        }
+        if (DEBUG) {
+          long nowTime = System.nanoTime();
+          long durationInMs = TimeUnit.NANOSECONDS.toMillis((nowTime - lastTime));
+          lastTime = nowTime;
+          Logging.d(TAG, "bytesRead[" + durationInMs + "] " + bytesRead);
+        }
+      }
+
+      try {
+        if (audioRecord != null) {
+          audioRecord.stop();
+        }
+      } catch (IllegalStateException e) {
+        Logging.e(TAG, "AudioRecord.stop failed: " + e.getMessage());
+      }
+    }
+
+    // Stops the inner thread loop and also calls AudioRecord.stop().
+    // Does not block the calling thread.
+    public void stopThread() {
+      Logging.d(TAG, "stopThread");
+      keepAlive = false;
+    }
+  }
+
+  WebRtcAudioRecord(long nativeAudioRecord) {
+    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
+    this.nativeAudioRecord = nativeAudioRecord;
+    if (DEBUG) {
+      WebRtcAudioUtils.logDeviceInfo(TAG);
+    }
+    effects = WebRtcAudioEffects.create();
+  }
+
+  private boolean enableBuiltInAEC(boolean enable) {
+    Logging.d(TAG, "enableBuiltInAEC(" + enable + ')');
+    if (effects == null) {
+      Logging.e(TAG, "Built-in AEC is not supported on this platform");
+      return false;
+    }
+    return effects.setAEC(enable);
+  }
+
+  private boolean enableBuiltInNS(boolean enable) {
+    Logging.d(TAG, "enableBuiltInNS(" + enable + ')');
+    if (effects == null) {
+      Logging.e(TAG, "Built-in NS is not supported on this platform");
+      return false;
+    }
+    return effects.setNS(enable);
+  }
+
+  private int initRecording(int sampleRate, int channels) {
+    Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + channels + ")");
+    if (audioRecord != null) {
+      reportWebRtcAudioRecordInitError("InitRecording called twice without StopRecording.");
+      return -1;
+    }
+    final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
+    final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
+    byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
+    Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
+    emptyBytes = new byte[byteBuffer.capacity()];
+    // Rather than passing the ByteBuffer with every callback (requiring
+    // the potentially expensive GetDirectBufferAddress) we simply have the
+    // the native class cache the address to the memory once.
+    nativeCacheDirectBufferAddress(byteBuffer, nativeAudioRecord);
+
+    // Get the minimum buffer size required for the successful creation of
+    // an AudioRecord object, in byte units.
+    // Note that this size doesn't guarantee a smooth recording under load.
+    final int channelConfig = channelCountToConfiguration(channels);
+    int minBufferSize =
+        AudioRecord.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
+    if (minBufferSize == AudioRecord.ERROR || minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
+      reportWebRtcAudioRecordInitError("AudioRecord.getMinBufferSize failed: " + minBufferSize);
+      return -1;
+    }
+    Logging.d(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize);
+
+    // Use a larger buffer size than the minimum required when creating the
+    // AudioRecord instance to ensure smooth recording under load. It has been
+    // verified that it does not increase the actual recording latency.
+    int bufferSizeInBytes = Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
+    Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
+    try {
+      audioRecord = new AudioRecord(audioSource, sampleRate, channelConfig,
+          AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes);
+    } catch (IllegalArgumentException e) {
+      reportWebRtcAudioRecordInitError("AudioRecord ctor error: " + e.getMessage());
+      releaseAudioResources();
+      return -1;
+    }
+    if (audioRecord == null || audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
+      reportWebRtcAudioRecordInitError("Failed to create a new AudioRecord instance");
+      releaseAudioResources();
+      return -1;
+    }
+    if (effects != null) {
+      effects.enable(audioRecord.getAudioSessionId());
+    }
+    logMainParameters();
+    logMainParametersExtended();
+    return framesPerBuffer;
+  }
+
+  private boolean startRecording() {
+    Logging.d(TAG, "startRecording");
+    assertTrue(audioRecord != null);
+    assertTrue(audioThread == null);
+    try {
+      audioRecord.startRecording();
+    } catch (IllegalStateException e) {
+      reportWebRtcAudioRecordStartError(AudioRecordStartErrorCode.AUDIO_RECORD_START_EXCEPTION,
+          "AudioRecord.startRecording failed: " + e.getMessage());
+      return false;
+    }
+    if (audioRecord.getRecordingState() != AudioRecord.RECORDSTATE_RECORDING) {
+      reportWebRtcAudioRecordStartError(
+          AudioRecordStartErrorCode.AUDIO_RECORD_START_STATE_MISMATCH,
+          "AudioRecord.startRecording failed - incorrect state :"
+          + audioRecord.getRecordingState());
+      return false;
+    }
+    audioThread = new AudioRecordThread("AudioRecordJavaThread");
+    audioThread.start();
+    return true;
+  }
+
+  private boolean stopRecording() {
+    Logging.d(TAG, "stopRecording");
+    assertTrue(audioThread != null);
+    audioThread.stopThread();
+    if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS)) {
+      Logging.e(TAG, "Join of AudioRecordJavaThread timed out");
+      WebRtcAudioUtils.logAudioState(TAG);
+    }
+    audioThread = null;
+    if (effects != null) {
+      effects.release();
+    }
+    releaseAudioResources();
+    return true;
+  }
+
+  private void logMainParameters() {
+    Logging.d(TAG, "AudioRecord: "
+            + "session ID: " + audioRecord.getAudioSessionId() + ", "
+            + "channels: " + audioRecord.getChannelCount() + ", "
+            + "sample rate: " + audioRecord.getSampleRate());
+  }
+
+  private void logMainParametersExtended() {
+    if (Build.VERSION.SDK_INT >= 23) {
+      Logging.d(TAG, "AudioRecord: "
+              // The frame count of the native AudioRecord buffer.
+              + "buffer size in frames: " + audioRecord.getBufferSizeInFrames());
+    }
+  }
+
+  // Helper method which throws an exception  when an assertion has failed.
+  private static void assertTrue(boolean condition) {
+    if (!condition) {
+      throw new AssertionError("Expected condition to be true");
+    }
+  }
+
+  private int channelCountToConfiguration(int channels) {
+    return (channels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
+  }
+
+  private native void nativeCacheDirectBufferAddress(ByteBuffer byteBuffer, long nativeAudioRecord);
+
+  private native void nativeDataIsRecorded(int bytes, long nativeAudioRecord);
+
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized void setAudioSource(int source) {
+    Logging.w(TAG, "Audio source is changed from: " + audioSource
+            + " to " + source);
+    audioSource = source;
+  }
+
+  private static int getDefaultAudioSource() {
+    return AudioSource.VOICE_COMMUNICATION;
+  }
+
+  // Sets all recorded samples to zero if `mute` is true, i.e., ensures that
+  // the microphone is muted.
+  public static void setMicrophoneMute(boolean mute) {
+    Logging.w(TAG, "setMicrophoneMute(" + mute + ")");
+    microphoneMute = mute;
+  }
+
+  // Releases the native AudioRecord resources.
+  private void releaseAudioResources() {
+    Logging.d(TAG, "releaseAudioResources");
+    if (audioRecord != null) {
+      audioRecord.release();
+      audioRecord = null;
+    }
+  }
+
+  private void reportWebRtcAudioRecordInitError(String errorMessage) {
+    Logging.e(TAG, "Init recording error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioRecordInitError(errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioRecordStartError(
+      AudioRecordStartErrorCode errorCode, String errorMessage) {
+    Logging.e(TAG, "Start recording error: " + errorCode + ". " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioRecordStartError(errorCode, errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioRecordError(String errorMessage) {
+    Logging.e(TAG, "Run-time recording error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioRecordError(errorMessage);
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
new file mode 100644
index 0000000000..3e1875c3d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
@@ -0,0 +1,494 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.voiceengine;
+
+import android.content.Context;
+import android.media.AudioAttributes;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioTrack;
+import android.os.Build;
+import android.os.Process;
+import androidx.annotation.Nullable;
+import java.lang.Thread;
+import java.nio.ByteBuffer;
+import org.webrtc.ContextUtils;
+import org.webrtc.Logging;
+import org.webrtc.ThreadUtils;
+
+public class WebRtcAudioTrack {
+  private static final boolean DEBUG = false;
+
+  private static final String TAG = "WebRtcAudioTrack";
+
+  // Default audio data format is PCM 16 bit per sample.
+  // Guaranteed to be supported by all devices.
+  private static final int BITS_PER_SAMPLE = 16;
+
+  // Requested size of each recorded buffer provided to the client.
+  private static final int CALLBACK_BUFFER_SIZE_MS = 10;
+
+  // Average number of callbacks per second.
+  private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
+
+  // The AudioTrackThread is allowed to wait for successful call to join()
+  // but the wait times out afther this amount of time.
+  private static final long AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS = 2000;
+
+  // By default, WebRTC creates audio tracks with a usage attribute
+  // corresponding to voice communications, such as telephony or VoIP.
+  private static final int DEFAULT_USAGE = AudioAttributes.USAGE_VOICE_COMMUNICATION;
+  private static int usageAttribute = DEFAULT_USAGE;
+
+  // This method overrides the default usage attribute and allows the user
+  // to set it to something else than AudioAttributes.USAGE_VOICE_COMMUNICATION.
+  // NOTE: calling this method will most likely break existing VoIP tuning.
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized void setAudioTrackUsageAttribute(int usage) {
+    Logging.w(TAG, "Default usage attribute is changed from: "
+        + DEFAULT_USAGE + " to " + usage);
+    usageAttribute = usage;
+  }
+
+  private final long nativeAudioTrack;
+  private final AudioManager audioManager;
+  private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker();
+
+  private ByteBuffer byteBuffer;
+
+  private @Nullable AudioTrack audioTrack;
+  private @Nullable AudioTrackThread audioThread;
+
+  // Samples to be played are replaced by zeros if `speakerMute` is set to true.
+  // Can be used to ensure that the speaker is fully muted.
+  private static volatile boolean speakerMute;
+  private byte[] emptyBytes;
+
+  // Audio playout/track error handler functions.
+  public enum AudioTrackStartErrorCode {
+    AUDIO_TRACK_START_EXCEPTION,
+    AUDIO_TRACK_START_STATE_MISMATCH,
+  }
+
+  @Deprecated
+  public static interface WebRtcAudioTrackErrorCallback {
+    void onWebRtcAudioTrackInitError(String errorMessage);
+    void onWebRtcAudioTrackStartError(String errorMessage);
+    void onWebRtcAudioTrackError(String errorMessage);
+  }
+
+  // TODO(henrika): upgrade all clients to use this new interface instead.
+  public static interface ErrorCallback {
+    void onWebRtcAudioTrackInitError(String errorMessage);
+    void onWebRtcAudioTrackStartError(AudioTrackStartErrorCode errorCode, String errorMessage);
+    void onWebRtcAudioTrackError(String errorMessage);
+  }
+
+  private static @Nullable WebRtcAudioTrackErrorCallback errorCallbackOld;
+  private static @Nullable ErrorCallback errorCallback;
+
+  @Deprecated
+  public static void setErrorCallback(WebRtcAudioTrackErrorCallback errorCallback) {
+    Logging.d(TAG, "Set error callback (deprecated");
+    WebRtcAudioTrack.errorCallbackOld = errorCallback;
+  }
+
+  public static void setErrorCallback(ErrorCallback errorCallback) {
+    Logging.d(TAG, "Set extended error callback");
+    WebRtcAudioTrack.errorCallback = errorCallback;
+  }
+
+  /**
+   * Audio thread which keeps calling AudioTrack.write() to stream audio.
+   * Data is periodically acquired from the native WebRTC layer using the
+   * nativeGetPlayoutData callback function.
+   * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
+   */
+  private class AudioTrackThread extends Thread {
+    private volatile boolean keepAlive = true;
+
+    public AudioTrackThread(String name) {
+      super(name);
+    }
+
+    @Override
+    public void run() {
+      Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
+      Logging.d(TAG, "AudioTrackThread" + WebRtcAudioUtils.getThreadInfo());
+      assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING);
+
+      // Fixed size in bytes of each 10ms block of audio data that we ask for
+      // using callbacks to the native WebRTC client.
+      final int sizeInBytes = byteBuffer.capacity();
+
+      while (keepAlive) {
+        // Get 10ms of PCM data from the native WebRTC client. Audio data is
+        // written into the common ByteBuffer using the address that was
+        // cached at construction.
+        nativeGetPlayoutData(sizeInBytes, nativeAudioTrack);
+        // Write data until all data has been written to the audio sink.
+        // Upon return, the buffer position will have been advanced to reflect
+        // the amount of data that was successfully written to the AudioTrack.
+        assertTrue(sizeInBytes <= byteBuffer.remaining());
+        if (speakerMute) {
+          byteBuffer.clear();
+          byteBuffer.put(emptyBytes);
+          byteBuffer.position(0);
+        }
+        int bytesWritten = audioTrack.write(byteBuffer, sizeInBytes, AudioTrack.WRITE_BLOCKING);
+        if (bytesWritten != sizeInBytes) {
+          Logging.e(TAG, "AudioTrack.write played invalid number of bytes: " + bytesWritten);
+          // If a write() returns a negative value, an error has occurred.
+          // Stop playing and report an error in this case.
+          if (bytesWritten < 0) {
+            keepAlive = false;
+            reportWebRtcAudioTrackError("AudioTrack.write failed: " + bytesWritten);
+          }
+        }
+        // The byte buffer must be rewinded since byteBuffer.position() is
+        // increased at each call to AudioTrack.write(). If we don't do this,
+        // next call to AudioTrack.write() will fail.
+        byteBuffer.rewind();
+
+        // TODO(henrika): it is possible to create a delay estimate here by
+        // counting number of written frames and subtracting the result from
+        // audioTrack.getPlaybackHeadPosition().
+      }
+
+      // Stops playing the audio data. Since the instance was created in
+      // MODE_STREAM mode, audio will stop playing after the last buffer that
+      // was written has been played.
+      if (audioTrack != null) {
+        Logging.d(TAG, "Calling AudioTrack.stop...");
+        try {
+          audioTrack.stop();
+          Logging.d(TAG, "AudioTrack.stop is done.");
+        } catch (IllegalStateException e) {
+          Logging.e(TAG, "AudioTrack.stop failed: " + e.getMessage());
+        }
+      }
+    }
+
+    // Stops the inner thread loop which results in calling AudioTrack.stop().
+    // Does not block the calling thread.
+    public void stopThread() {
+      Logging.d(TAG, "stopThread");
+      keepAlive = false;
+    }
+  }
+
+  WebRtcAudioTrack(long nativeAudioTrack) {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
+    this.nativeAudioTrack = nativeAudioTrack;
+    audioManager =
+        (AudioManager) ContextUtils.getApplicationContext().getSystemService(Context.AUDIO_SERVICE);
+    if (DEBUG) {
+      WebRtcAudioUtils.logDeviceInfo(TAG);
+    }
+  }
+
+  private int initPlayout(int sampleRate, int channels, double bufferSizeFactor) {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG,
+        "initPlayout(sampleRate=" + sampleRate + ", channels=" + channels
+            + ", bufferSizeFactor=" + bufferSizeFactor + ")");
+    final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
+    byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
+    Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
+    emptyBytes = new byte[byteBuffer.capacity()];
+    // Rather than passing the ByteBuffer with every callback (requiring
+    // the potentially expensive GetDirectBufferAddress) we simply have the
+    // the native class cache the address to the memory once.
+    nativeCacheDirectBufferAddress(byteBuffer, nativeAudioTrack);
+
+    // Get the minimum buffer size required for the successful creation of an
+    // AudioTrack object to be created in the MODE_STREAM mode.
+    // Note that this size doesn't guarantee a smooth playback under load.
+    final int channelConfig = channelCountToConfiguration(channels);
+    final int minBufferSizeInBytes = (int) (AudioTrack.getMinBufferSize(sampleRate, channelConfig,
+                                                AudioFormat.ENCODING_PCM_16BIT)
+        * bufferSizeFactor);
+    Logging.d(TAG, "minBufferSizeInBytes: " + minBufferSizeInBytes);
+    // For the streaming mode, data must be written to the audio sink in
+    // chunks of size (given by byteBuffer.capacity()) less than or equal
+    // to the total buffer size `minBufferSizeInBytes`. But, we have seen
+    // reports of "getMinBufferSize(): error querying hardware". Hence, it
+    // can happen that `minBufferSizeInBytes` contains an invalid value.
+    if (minBufferSizeInBytes < byteBuffer.capacity()) {
+      reportWebRtcAudioTrackInitError("AudioTrack.getMinBufferSize returns an invalid value.");
+      return -1;
+    }
+
+    // Ensure that prevision audio session was stopped correctly before trying
+    // to create a new AudioTrack.
+    if (audioTrack != null) {
+      reportWebRtcAudioTrackInitError("Conflict with existing AudioTrack.");
+      return -1;
+    }
+    try {
+      // Create an AudioTrack object and initialize its associated audio buffer.
+      // The size of this buffer determines how long an AudioTrack can play
+      // before running out of data.
+      // As we are on API level 21 or higher, it is possible to use a special AudioTrack
+      // constructor that uses AudioAttributes and AudioFormat as input. It allows us to
+      // supersede the notion of stream types for defining the behavior of audio playback,
+      // and to allow certain platforms or routing policies to use this information for more
+      // refined volume or routing decisions.
+      audioTrack = createAudioTrack(sampleRate, channelConfig, minBufferSizeInBytes);
+    } catch (IllegalArgumentException e) {
+      reportWebRtcAudioTrackInitError(e.getMessage());
+      releaseAudioResources();
+      return -1;
+    }
+
+    // It can happen that an AudioTrack is created but it was not successfully
+    // initialized upon creation. Seems to be the case e.g. when the maximum
+    // number of globally available audio tracks is exceeded.
+    if (audioTrack == null || audioTrack.getState() != AudioTrack.STATE_INITIALIZED) {
+      reportWebRtcAudioTrackInitError("Initialization of audio track failed.");
+      releaseAudioResources();
+      return -1;
+    }
+    logMainParameters();
+    logMainParametersExtended();
+    return minBufferSizeInBytes;
+  }
+
+  private boolean startPlayout() {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "startPlayout");
+    assertTrue(audioTrack != null);
+    assertTrue(audioThread == null);
+
+    // Starts playing an audio track.
+    try {
+      audioTrack.play();
+    } catch (IllegalStateException e) {
+      reportWebRtcAudioTrackStartError(AudioTrackStartErrorCode.AUDIO_TRACK_START_EXCEPTION,
+          "AudioTrack.play failed: " + e.getMessage());
+      releaseAudioResources();
+      return false;
+    }
+    if (audioTrack.getPlayState() != AudioTrack.PLAYSTATE_PLAYING) {
+      reportWebRtcAudioTrackStartError(
+          AudioTrackStartErrorCode.AUDIO_TRACK_START_STATE_MISMATCH,
+          "AudioTrack.play failed - incorrect state :"
+          + audioTrack.getPlayState());
+      releaseAudioResources();
+      return false;
+    }
+
+    // Create and start new high-priority thread which calls AudioTrack.write()
+    // and where we also call the native nativeGetPlayoutData() callback to
+    // request decoded audio from WebRTC.
+    audioThread = new AudioTrackThread("AudioTrackJavaThread");
+    audioThread.start();
+    return true;
+  }
+
+  private boolean stopPlayout() {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "stopPlayout");
+    assertTrue(audioThread != null);
+    logUnderrunCount();
+    audioThread.stopThread();
+
+    Logging.d(TAG, "Stopping the AudioTrackThread...");
+    audioThread.interrupt();
+    if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS)) {
+      Logging.e(TAG, "Join of AudioTrackThread timed out.");
+      WebRtcAudioUtils.logAudioState(TAG);
+    }
+    Logging.d(TAG, "AudioTrackThread has now been stopped.");
+    audioThread = null;
+    releaseAudioResources();
+    return true;
+  }
+
+  // Get max possible volume index for a phone call audio stream.
+  private int getStreamMaxVolume() {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "getStreamMaxVolume");
+    assertTrue(audioManager != null);
+    return audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
+  }
+
+  // Set current volume level for a phone call audio stream.
+  private boolean setStreamVolume(int volume) {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "setStreamVolume(" + volume + ")");
+    assertTrue(audioManager != null);
+    if (audioManager.isVolumeFixed()) {
+      Logging.e(TAG, "The device implements a fixed volume policy.");
+      return false;
+    }
+    audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0);
+    return true;
+  }
+
+  /** Get current volume level for a phone call audio stream. */
+  private int getStreamVolume() {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "getStreamVolume");
+    assertTrue(audioManager != null);
+    return audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
+  }
+
+  private void logMainParameters() {
+    Logging.d(TAG, "AudioTrack: "
+            + "session ID: " + audioTrack.getAudioSessionId() + ", "
+            + "channels: " + audioTrack.getChannelCount() + ", "
+            + "sample rate: " + audioTrack.getSampleRate() + ", "
+            // Gain (>=1.0) expressed as linear multiplier on sample values.
+            + "max gain: " + AudioTrack.getMaxVolume());
+  }
+
+  // Creates and AudioTrack instance using AudioAttributes and AudioFormat as input.
+  // It allows certain platforms or routing policies to use this information for more
+  // refined volume or routing decisions.
+  private static AudioTrack createAudioTrack(
+      int sampleRateInHz, int channelConfig, int bufferSizeInBytes) {
+    Logging.d(TAG, "createAudioTrack");
+    // TODO(henrika): use setPerformanceMode(int) with PERFORMANCE_MODE_LOW_LATENCY to control
+    // performance when Android O is supported. Add some logging in the mean time.
+    final int nativeOutputSampleRate =
+        AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_VOICE_CALL);
+    Logging.d(TAG, "nativeOutputSampleRate: " + nativeOutputSampleRate);
+    if (sampleRateInHz != nativeOutputSampleRate) {
+      Logging.w(TAG, "Unable to use fast mode since requested sample rate is not native");
+    }
+    if (usageAttribute != DEFAULT_USAGE) {
+      Logging.w(TAG, "A non default usage attribute is used: " + usageAttribute);
+    }
+    // Create an audio track where the audio usage is for VoIP and the content type is speech.
+    return new AudioTrack(
+        new AudioAttributes.Builder()
+            .setUsage(usageAttribute)
+            .setContentType(AudioAttributes.CONTENT_TYPE_SPEECH)
+        .build(),
+        new AudioFormat.Builder()
+          .setEncoding(AudioFormat.ENCODING_PCM_16BIT)
+          .setSampleRate(sampleRateInHz)
+          .setChannelMask(channelConfig)
+          .build(),
+        bufferSizeInBytes,
+        AudioTrack.MODE_STREAM,
+        AudioManager.AUDIO_SESSION_ID_GENERATE);
+  }
+
+  private void logBufferSizeInFrames() {
+    if (Build.VERSION.SDK_INT >= 23) {
+      Logging.d(TAG, "AudioTrack: "
+              // The effective size of the AudioTrack buffer that the app writes to.
+              + "buffer size in frames: " + audioTrack.getBufferSizeInFrames());
+    }
+  }
+
+  private int getBufferSizeInFrames() {
+    if (Build.VERSION.SDK_INT >= 23) {
+      return audioTrack.getBufferSizeInFrames();
+    }
+    return -1;
+  }
+
+  private void logBufferCapacityInFrames() {
+    if (Build.VERSION.SDK_INT >= 24) {
+      Logging.d(TAG,
+          "AudioTrack: "
+              // Maximum size of the AudioTrack buffer in frames.
+              + "buffer capacity in frames: " + audioTrack.getBufferCapacityInFrames());
+    }
+  }
+
+  private void logMainParametersExtended() {
+    logBufferSizeInFrames();
+    logBufferCapacityInFrames();
+  }
+
+  // Prints the number of underrun occurrences in the application-level write
+  // buffer since the AudioTrack was created. An underrun occurs if the app does
+  // not write audio data quickly enough, causing the buffer to underflow and a
+  // potential audio glitch.
+  // TODO(henrika): keep track of this value in the field and possibly add new
+  // UMA stat if needed.
+  private void logUnderrunCount() {
+    if (Build.VERSION.SDK_INT >= 24) {
+      Logging.d(TAG, "underrun count: " + audioTrack.getUnderrunCount());
+    }
+  }
+
+  // Helper method which throws an exception  when an assertion has failed.
+  private static void assertTrue(boolean condition) {
+    if (!condition) {
+      throw new AssertionError("Expected condition to be true");
+    }
+  }
+
+  private int channelCountToConfiguration(int channels) {
+    return (channels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO);
+  }
+
+  private native void nativeCacheDirectBufferAddress(ByteBuffer byteBuffer, long nativeAudioRecord);
+
+  private native void nativeGetPlayoutData(int bytes, long nativeAudioRecord);
+
+  // Sets all samples to be played out to zero if `mute` is true, i.e.,
+  // ensures that the speaker is muted.
+  public static void setSpeakerMute(boolean mute) {
+    Logging.w(TAG, "setSpeakerMute(" + mute + ")");
+    speakerMute = mute;
+  }
+
+  // Releases the native AudioTrack resources.
+  private void releaseAudioResources() {
+    Logging.d(TAG, "releaseAudioResources");
+    if (audioTrack != null) {
+      audioTrack.release();
+      audioTrack = null;
+    }
+  }
+
+  private void reportWebRtcAudioTrackInitError(String errorMessage) {
+    Logging.e(TAG, "Init playout error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG);
+    if (errorCallbackOld != null) {
+      errorCallbackOld.onWebRtcAudioTrackInitError(errorMessage);
+    }
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioTrackInitError(errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioTrackStartError(
+      AudioTrackStartErrorCode errorCode, String errorMessage) {
+    Logging.e(TAG, "Start playout error: "  + errorCode + ". " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG);
+    if (errorCallbackOld != null) {
+      errorCallbackOld.onWebRtcAudioTrackStartError(errorMessage);
+    }
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioTrackStartError(errorCode, errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioTrackError(String errorMessage) {
+    Logging.e(TAG, "Run-time playback error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG);
+    if (errorCallbackOld != null) {
+      errorCallbackOld.onWebRtcAudioTrackError(errorMessage);
+    }
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioTrackError(errorMessage);
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
new file mode 100644
index 0000000000..afd3d429af
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
@@ -0,0 +1,382 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.voiceengine;
+
+import static android.media.AudioManager.MODE_IN_CALL;
+import static android.media.AudioManager.MODE_IN_COMMUNICATION;
+import static android.media.AudioManager.MODE_NORMAL;
+import static android.media.AudioManager.MODE_RINGTONE;
+
+import android.annotation.SuppressLint;
+import android.content.Context;
+import android.content.pm.PackageManager;
+import android.media.AudioDeviceInfo;
+import android.media.AudioManager;
+import android.os.Build;
+import java.lang.Thread;
+import java.util.Arrays;
+import java.util.List;
+import org.webrtc.ContextUtils;
+import org.webrtc.Logging;
+
+public final class WebRtcAudioUtils {
+  private static final String TAG = "WebRtcAudioUtils";
+
+  // List of devices where we have seen issues (e.g. bad audio quality) using
+  // the low latency output mode in combination with OpenSL ES.
+  // The device name is given by Build.MODEL.
+  private static final String[] BLACKLISTED_OPEN_SL_ES_MODELS = new String[] {
+      // It is recommended to maintain a list of blacklisted models outside
+      // this package and instead call
+      // WebRtcAudioManager.setBlacklistDeviceForOpenSLESUsage(true)
+      // from the client for devices where OpenSL ES shall be disabled.
+  };
+
+  // List of devices where it has been verified that the built-in effect
+  // bad and where it makes sense to avoid using it and instead rely on the
+  // native WebRTC version instead. The device name is given by Build.MODEL.
+  private static final String[] BLACKLISTED_AEC_MODELS = new String[] {
+      // It is recommended to maintain a list of blacklisted models outside
+      // this package and instead call setWebRtcBasedAcousticEchoCanceler(true)
+      // from the client for devices where the built-in AEC shall be disabled.
+  };
+  private static final String[] BLACKLISTED_NS_MODELS = new String[] {
+    // It is recommended to maintain a list of blacklisted models outside
+    // this package and instead call setWebRtcBasedNoiseSuppressor(true)
+    // from the client for devices where the built-in NS shall be disabled.
+  };
+
+  // Use 16kHz as the default sample rate. A higher sample rate might prevent
+  // us from supporting communication mode on some older (e.g. ICS) devices.
+  private static final int DEFAULT_SAMPLE_RATE_HZ = 16000;
+  private static int defaultSampleRateHz = DEFAULT_SAMPLE_RATE_HZ;
+  // Set to true if setDefaultSampleRateHz() has been called.
+  private static boolean isDefaultSampleRateOverridden;
+
+  // By default, utilize hardware based audio effects for AEC and NS when
+  // available.
+  private static boolean useWebRtcBasedAcousticEchoCanceler;
+  private static boolean useWebRtcBasedNoiseSuppressor;
+
+  // Call these methods if any hardware based effect shall be replaced by a
+  // software based version provided by the WebRTC stack instead.
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized void setWebRtcBasedAcousticEchoCanceler(boolean enable) {
+    useWebRtcBasedAcousticEchoCanceler = enable;
+  }
+
+    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized void setWebRtcBasedNoiseSuppressor(boolean enable) {
+    useWebRtcBasedNoiseSuppressor = enable;
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized void setWebRtcBasedAutomaticGainControl(boolean enable) {
+    // TODO(henrika): deprecated; remove when no longer used by any client.
+    Logging.w(TAG, "setWebRtcBasedAutomaticGainControl() is deprecated");
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized boolean useWebRtcBasedAcousticEchoCanceler() {
+    if (useWebRtcBasedAcousticEchoCanceler) {
+      Logging.w(TAG, "Overriding default behavior; now using WebRTC AEC!");
+    }
+    return useWebRtcBasedAcousticEchoCanceler;
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized boolean useWebRtcBasedNoiseSuppressor() {
+    if (useWebRtcBasedNoiseSuppressor) {
+      Logging.w(TAG, "Overriding default behavior; now using WebRTC NS!");
+    }
+    return useWebRtcBasedNoiseSuppressor;
+  }
+
+  // TODO(henrika): deprecated; remove when no longer used by any client.
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized boolean useWebRtcBasedAutomaticGainControl() {
+    // Always return true here to avoid trying to use any built-in AGC.
+    return true;
+  }
+
+  // Returns true if the device supports an audio effect (AEC or NS).
+  // Four conditions must be fulfilled if functions are to return true:
+  // 1) the platform must support the built-in (HW) effect,
+  // 2) explicit use (override) of a WebRTC based version must not be set,
+  // 3) the device must not be blacklisted for use of the effect, and
+  // 4) the UUID of the effect must be approved (some UUIDs can be excluded).
+  public static boolean isAcousticEchoCancelerSupported() {
+    return WebRtcAudioEffects.canUseAcousticEchoCanceler();
+  }
+  public static boolean isNoiseSuppressorSupported() {
+    return WebRtcAudioEffects.canUseNoiseSuppressor();
+  }
+  // TODO(henrika): deprecated; remove when no longer used by any client.
+  public static boolean isAutomaticGainControlSupported() {
+    // Always return false here to avoid trying to use any built-in AGC.
+    return false;
+  }
+
+  // Call this method if the default handling of querying the native sample
+  // rate shall be overridden. Can be useful on some devices where the
+  // available Android APIs are known to return invalid results.
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized void setDefaultSampleRateHz(int sampleRateHz) {
+    isDefaultSampleRateOverridden = true;
+    defaultSampleRateHz = sampleRateHz;
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized boolean isDefaultSampleRateOverridden() {
+    return isDefaultSampleRateOverridden;
+  }
+
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public static synchronized int getDefaultSampleRateHz() {
+    return defaultSampleRateHz;
+  }
+
+  public static List<String> getBlackListedModelsForAecUsage() {
+    return Arrays.asList(WebRtcAudioUtils.BLACKLISTED_AEC_MODELS);
+  }
+
+  public static List<String> getBlackListedModelsForNsUsage() {
+    return Arrays.asList(WebRtcAudioUtils.BLACKLISTED_NS_MODELS);
+  }
+
+  // Helper method for building a string of thread information.
+  public static String getThreadInfo() {
+    return "@[name=" + Thread.currentThread().getName() + ", id=" + Thread.currentThread().getId()
+        + "]";
+  }
+
+  // Returns true if we're running on emulator.
+  public static boolean runningOnEmulator() {
+    return Build.HARDWARE.equals("goldfish") && Build.BRAND.startsWith("generic_");
+  }
+
+  // Returns true if the device is blacklisted for OpenSL ES usage.
+  public static boolean deviceIsBlacklistedForOpenSLESUsage() {
+    List<String> blackListedModels = Arrays.asList(BLACKLISTED_OPEN_SL_ES_MODELS);
+    return blackListedModels.contains(Build.MODEL);
+  }
+
+  // Information about the current build, taken from system properties.
+  static void logDeviceInfo(String tag) {
+    Logging.d(tag, "Android SDK: " + Build.VERSION.SDK_INT + ", "
+            + "Release: " + Build.VERSION.RELEASE + ", "
+            + "Brand: " + Build.BRAND + ", "
+            + "Device: " + Build.DEVICE + ", "
+            + "Id: " + Build.ID + ", "
+            + "Hardware: " + Build.HARDWARE + ", "
+            + "Manufacturer: " + Build.MANUFACTURER + ", "
+            + "Model: " + Build.MODEL + ", "
+            + "Product: " + Build.PRODUCT);
+  }
+
+  // Logs information about the current audio state. The idea is to call this
+  // method when errors are detected to log under what conditions the error
+  // occurred. Hopefully it will provide clues to what might be the root cause.
+  static void logAudioState(String tag) {
+    logDeviceInfo(tag);
+    final Context context = ContextUtils.getApplicationContext();
+    final AudioManager audioManager =
+        (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
+    logAudioStateBasic(tag, audioManager);
+    logAudioStateVolume(tag, audioManager);
+    logAudioDeviceInfo(tag, audioManager);
+  }
+
+  // Reports basic audio statistics.
+  private static void logAudioStateBasic(String tag, AudioManager audioManager) {
+    Logging.d(tag, "Audio State: "
+            + "audio mode: " + modeToString(audioManager.getMode()) + ", "
+            + "has mic: " + hasMicrophone() + ", "
+            + "mic muted: " + audioManager.isMicrophoneMute() + ", "
+            + "music active: " + audioManager.isMusicActive() + ", "
+            + "speakerphone: " + audioManager.isSpeakerphoneOn() + ", "
+            + "BT SCO: " + audioManager.isBluetoothScoOn());
+  }
+
+  // Adds volume information for all possible stream types.
+  private static void logAudioStateVolume(String tag, AudioManager audioManager) {
+    final int[] streams = {
+        AudioManager.STREAM_VOICE_CALL,
+        AudioManager.STREAM_MUSIC,
+        AudioManager.STREAM_RING,
+        AudioManager.STREAM_ALARM,
+        AudioManager.STREAM_NOTIFICATION,
+        AudioManager.STREAM_SYSTEM
+    };
+    Logging.d(tag, "Audio State: ");
+    // Some devices may not have volume controls and might use a fixed volume.
+    boolean fixedVolume = audioManager.isVolumeFixed();
+    Logging.d(tag, "  fixed volume=" + fixedVolume);
+    if (!fixedVolume) {
+      for (int stream : streams) {
+        StringBuilder info = new StringBuilder();
+        info.append("  " + streamTypeToString(stream) + ": ");
+        info.append("volume=").append(audioManager.getStreamVolume(stream));
+        info.append(", max=").append(audioManager.getStreamMaxVolume(stream));
+        logIsStreamMute(tag, audioManager, stream, info);
+        Logging.d(tag, info.toString());
+      }
+    }
+  }
+
+  private static void logIsStreamMute(
+      String tag, AudioManager audioManager, int stream, StringBuilder info) {
+    if (Build.VERSION.SDK_INT >= 23) {
+      info.append(", muted=").append(audioManager.isStreamMute(stream));
+    }
+  }
+
+  // Moz linting complains even though AudioManager.GET_DEVICES_ALL is
+  // listed in the docs here:
+  // https://developer.android.com/reference/android/media/AudioManager#GET_DEVICES_ALL
+  @SuppressLint("WrongConstant")
+  private static void logAudioDeviceInfo(String tag, AudioManager audioManager) {
+    if (Build.VERSION.SDK_INT < 23) {
+      return;
+    }
+    final AudioDeviceInfo[] devices =
+        audioManager.getDevices(AudioManager.GET_DEVICES_ALL);
+    if (devices.length == 0) {
+      return;
+    }
+    Logging.d(tag, "Audio Devices: ");
+    for (AudioDeviceInfo device : devices) {
+      StringBuilder info = new StringBuilder();
+      info.append("  ").append(deviceTypeToString(device.getType()));
+      info.append(device.isSource() ? "(in): " : "(out): ");
+      // An empty array indicates that the device supports arbitrary channel counts.
+      if (device.getChannelCounts().length > 0) {
+        info.append("channels=").append(Arrays.toString(device.getChannelCounts()));
+        info.append(", ");
+      }
+      if (device.getEncodings().length > 0) {
+        // Examples: ENCODING_PCM_16BIT = 2, ENCODING_PCM_FLOAT = 4.
+        info.append("encodings=").append(Arrays.toString(device.getEncodings()));
+        info.append(", ");
+      }
+      if (device.getSampleRates().length > 0) {
+        info.append("sample rates=").append(Arrays.toString(device.getSampleRates()));
+        info.append(", ");
+      }
+      info.append("id=").append(device.getId());
+      Logging.d(tag, info.toString());
+    }
+  }
+
+  // Converts media.AudioManager modes into local string representation.
+  static String modeToString(int mode) {
+    switch (mode) {
+      case MODE_IN_CALL:
+        return "MODE_IN_CALL";
+      case MODE_IN_COMMUNICATION:
+        return "MODE_IN_COMMUNICATION";
+      case MODE_NORMAL:
+        return "MODE_NORMAL";
+      case MODE_RINGTONE:
+        return "MODE_RINGTONE";
+      default:
+        return "MODE_INVALID";
+    }
+  }
+
+  private static String streamTypeToString(int stream) {
+    switch(stream) {
+      case AudioManager.STREAM_VOICE_CALL:
+        return "STREAM_VOICE_CALL";
+      case AudioManager.STREAM_MUSIC:
+        return "STREAM_MUSIC";
+      case AudioManager.STREAM_RING:
+        return "STREAM_RING";
+      case AudioManager.STREAM_ALARM:
+        return "STREAM_ALARM";
+      case AudioManager.STREAM_NOTIFICATION:
+        return "STREAM_NOTIFICATION";
+      case AudioManager.STREAM_SYSTEM:
+        return "STREAM_SYSTEM";
+      default:
+        return "STREAM_INVALID";
+    }
+  }
+
+  // Converts AudioDeviceInfo types to local string representation.
+  private static String deviceTypeToString(int type) {
+    switch (type) {
+      case AudioDeviceInfo.TYPE_UNKNOWN:
+        return "TYPE_UNKNOWN";
+      case AudioDeviceInfo.TYPE_BUILTIN_EARPIECE:
+        return "TYPE_BUILTIN_EARPIECE";
+      case AudioDeviceInfo.TYPE_BUILTIN_SPEAKER:
+        return "TYPE_BUILTIN_SPEAKER";
+      case AudioDeviceInfo.TYPE_WIRED_HEADSET:
+        return "TYPE_WIRED_HEADSET";
+      case AudioDeviceInfo.TYPE_WIRED_HEADPHONES:
+        return "TYPE_WIRED_HEADPHONES";
+      case AudioDeviceInfo.TYPE_LINE_ANALOG:
+        return "TYPE_LINE_ANALOG";
+      case AudioDeviceInfo.TYPE_LINE_DIGITAL:
+        return "TYPE_LINE_DIGITAL";
+      case AudioDeviceInfo.TYPE_BLUETOOTH_SCO:
+        return "TYPE_BLUETOOTH_SCO";
+      case AudioDeviceInfo.TYPE_BLUETOOTH_A2DP:
+        return "TYPE_BLUETOOTH_A2DP";
+      case AudioDeviceInfo.TYPE_HDMI:
+        return "TYPE_HDMI";
+      case AudioDeviceInfo.TYPE_HDMI_ARC:
+        return "TYPE_HDMI_ARC";
+      case AudioDeviceInfo.TYPE_USB_DEVICE:
+        return "TYPE_USB_DEVICE";
+      case AudioDeviceInfo.TYPE_USB_ACCESSORY:
+        return "TYPE_USB_ACCESSORY";
+      case AudioDeviceInfo.TYPE_DOCK:
+        return "TYPE_DOCK";
+      case AudioDeviceInfo.TYPE_FM:
+        return "TYPE_FM";
+      case AudioDeviceInfo.TYPE_BUILTIN_MIC:
+        return "TYPE_BUILTIN_MIC";
+      case AudioDeviceInfo.TYPE_FM_TUNER:
+        return "TYPE_FM_TUNER";
+      case AudioDeviceInfo.TYPE_TV_TUNER:
+        return "TYPE_TV_TUNER";
+      case AudioDeviceInfo.TYPE_TELEPHONY:
+        return "TYPE_TELEPHONY";
+      case AudioDeviceInfo.TYPE_AUX_LINE:
+        return "TYPE_AUX_LINE";
+      case AudioDeviceInfo.TYPE_IP:
+        return "TYPE_IP";
+      case AudioDeviceInfo.TYPE_BUS:
+        return "TYPE_BUS";
+      case AudioDeviceInfo.TYPE_USB_HEADSET:
+        return "TYPE_USB_HEADSET";
+      default:
+        return "TYPE_UNKNOWN";
+    }
+  }
+
+  // Returns true if the device can record audio via a microphone.
+  private static boolean hasMicrophone() {
+    return ContextUtils.getApplicationContext().getPackageManager().hasSystemFeature(
+        PackageManager.FEATURE_MICROPHONE);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_device/android/opensles_common.cc b/third_party/libwebrtc/modules/audio_device/android/opensles_common.cc
new file mode 100644
index 0000000000..019714dae4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/opensles_common.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/opensles_common.h"
+
+#include <SLES/OpenSLES.h>
+
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Returns a string representation given an integer SL_RESULT_XXX code.
+// The mapping can be found in <SLES/OpenSLES.h>.
+const char* GetSLErrorString(size_t code) {
+  static const char* sl_error_strings[] = {
+      "SL_RESULT_SUCCESS",                 // 0
+      "SL_RESULT_PRECONDITIONS_VIOLATED",  // 1
+      "SL_RESULT_PARAMETER_INVALID",       // 2
+      "SL_RESULT_MEMORY_FAILURE",          // 3
+      "SL_RESULT_RESOURCE_ERROR",          // 4
+      "SL_RESULT_RESOURCE_LOST",           // 5
+      "SL_RESULT_IO_ERROR",                // 6
+      "SL_RESULT_BUFFER_INSUFFICIENT",     // 7
+      "SL_RESULT_CONTENT_CORRUPTED",       // 8
+      "SL_RESULT_CONTENT_UNSUPPORTED",     // 9
+      "SL_RESULT_CONTENT_NOT_FOUND",       // 10
+      "SL_RESULT_PERMISSION_DENIED",       // 11
+      "SL_RESULT_FEATURE_UNSUPPORTED",     // 12
+      "SL_RESULT_INTERNAL_ERROR",          // 13
+      "SL_RESULT_UNKNOWN_ERROR",           // 14
+      "SL_RESULT_OPERATION_ABORTED",       // 15
+      "SL_RESULT_CONTROL_LOST",            // 16
+  };
+
+  if (code >= arraysize(sl_error_strings)) {
+    return "SL_RESULT_UNKNOWN_ERROR";
+  }
+  return sl_error_strings[code];
+}
+
+SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
+                                        int sample_rate,
+                                        size_t bits_per_sample) {
+  RTC_CHECK_EQ(bits_per_sample, SL_PCMSAMPLEFORMAT_FIXED_16);
+  SLDataFormat_PCM format;
+  format.formatType = SL_DATAFORMAT_PCM;
+  format.numChannels = static_cast<SLuint32>(channels);
+  // Note that, the unit of sample rate is actually in milliHertz and not Hertz.
+  switch (sample_rate) {
+    case 8000:
+      format.samplesPerSec = SL_SAMPLINGRATE_8;
+      break;
+    case 16000:
+      format.samplesPerSec = SL_SAMPLINGRATE_16;
+      break;
+    case 22050:
+      format.samplesPerSec = SL_SAMPLINGRATE_22_05;
+      break;
+    case 32000:
+      format.samplesPerSec = SL_SAMPLINGRATE_32;
+      break;
+    case 44100:
+      format.samplesPerSec = SL_SAMPLINGRATE_44_1;
+      break;
+    case 48000:
+      format.samplesPerSec = SL_SAMPLINGRATE_48;
+      break;
+    case 64000:
+      format.samplesPerSec = SL_SAMPLINGRATE_64;
+      break;
+    case 88200:
+      format.samplesPerSec = SL_SAMPLINGRATE_88_2;
+      break;
+    case 96000:
+      format.samplesPerSec = SL_SAMPLINGRATE_96;
+      break;
+    default:
+      RTC_CHECK(false) << "Unsupported sample rate: " << sample_rate;
+      break;
+  }
+  format.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+  format.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
+  format.endianness = SL_BYTEORDER_LITTLEENDIAN;
+  if (format.numChannels == 1) {
+    format.channelMask = SL_SPEAKER_FRONT_CENTER;
+  } else if (format.numChannels == 2) {
+    format.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+  } else {
+    RTC_CHECK(false) << "Unsupported number of channels: "
+                     << format.numChannels;
+  }
+  return format;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/opensles_common.h b/third_party/libwebrtc/modules/audio_device/android/opensles_common.h
new file mode 100644
index 0000000000..438c522072
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/opensles_common.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
+
+#include <SLES/OpenSLES.h>
+#include <stddef.h>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Returns a string representation given an integer SL_RESULT_XXX code.
+// The mapping can be found in <SLES/OpenSLES.h>.
+const char* GetSLErrorString(size_t code);
+
+// Configures an SL_DATAFORMAT_PCM structure based on native audio parameters.
+SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
+                                        int sample_rate,
+                                        size_t bits_per_sample);
+
+// Helper class for using SLObjectItf interfaces.
+template <typename SLType, typename SLDerefType>
+class ScopedSLObject {
+ public:
+  ScopedSLObject() : obj_(nullptr) {}
+
+  ~ScopedSLObject() { Reset(); }
+
+  SLType* Receive() {
+    RTC_DCHECK(!obj_);
+    return &obj_;
+  }
+
+  SLDerefType operator->() { return *obj_; }
+
+  SLType Get() const { return obj_; }
+
+  void Reset() {
+    if (obj_) {
+      (*obj_)->Destroy(obj_);
+      obj_ = nullptr;
+    }
+  }
+
+ private:
+  SLType obj_;
+};
+
+typedef ScopedSLObject<SLObjectItf, const SLObjectItf_*> ScopedSLObjectItf;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/opensles_player.cc b/third_party/libwebrtc/modules/audio_device/android/opensles_player.cc
new file mode 100644
index 0000000000..f2b3a37194
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/opensles_player.cc
@@ -0,0 +1,434 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/opensles_player.h"
+
+#include <android/log.h>
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+
+#define TAG "OpenSLESPlayer"
+#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
+#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
+#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
+
+#define RETURN_ON_ERROR(op, ...)                          \
+  do {                                                    \
+    SLresult err = (op);                                  \
+    if (err != SL_RESULT_SUCCESS) {                       \
+      ALOGE("%s failed: %s", #op, GetSLErrorString(err)); \
+      return __VA_ARGS__;                                 \
+    }                                                     \
+  } while (0)
+
+namespace webrtc {
+
+OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager)
+    : audio_manager_(audio_manager),
+      audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
+      audio_device_buffer_(nullptr),
+      initialized_(false),
+      playing_(false),
+      buffer_index_(0),
+      engine_(nullptr),
+      player_(nullptr),
+      simple_buffer_queue_(nullptr),
+      volume_(nullptr),
+      last_play_time_(0) {
+  ALOGD("ctor[tid=%d]", rtc::CurrentThreadId());
+  // Use native audio output parameters provided by the audio manager and
+  // define the PCM format structure.
+  pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
+                                       audio_parameters_.sample_rate(),
+                                       audio_parameters_.bits_per_sample());
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the internal  audio thread.
+  thread_checker_opensles_.Detach();
+}
+
+OpenSLESPlayer::~OpenSLESPlayer() {
+  ALOGD("dtor[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+  DestroyAudioPlayer();
+  DestroyMix();
+  engine_ = nullptr;
+  RTC_DCHECK(!engine_);
+  RTC_DCHECK(!output_mix_.Get());
+  RTC_DCHECK(!player_);
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_DCHECK(!volume_);
+}
+
+int OpenSLESPlayer::Init() {
+  ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (audio_parameters_.channels() == 2) {
+    ALOGW("Stereo mode is enabled");
+  }
+  return 0;
+}
+
+int OpenSLESPlayer::Terminate() {
+  ALOGD("Terminate[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopPlayout();
+  return 0;
+}
+
+int OpenSLESPlayer::InitPlayout() {
+  ALOGD("InitPlayout[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!playing_);
+  if (!ObtainEngineInterface()) {
+    ALOGE("Failed to obtain SL Engine interface");
+    return -1;
+  }
+  CreateMix();
+  initialized_ = true;
+  buffer_index_ = 0;
+  return 0;
+}
+
+int OpenSLESPlayer::StartPlayout() {
+  ALOGD("StartPlayout[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!playing_);
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetPlayout();
+  }
+  // The number of lower latency audio players is limited, hence we create the
+  // audio player in Start() and destroy it in Stop().
+  CreateAudioPlayer();
+  // Fill up audio buffers to avoid initial glitch and to ensure that playback
+  // starts when mode is later changed to SL_PLAYSTATE_PLAYING.
+  // TODO(henrika): we can save some delay by only making one call to
+  // EnqueuePlayoutData. Most likely not worth the risk of adding a glitch.
+  last_play_time_ = rtc::Time();
+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
+    EnqueuePlayoutData(true);
+  }
+  // Start streaming data by setting the play state to SL_PLAYSTATE_PLAYING.
+  // For a player object, when the object is in the SL_PLAYSTATE_PLAYING
+  // state, adding buffers will implicitly start playback.
+  RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_PLAYING), -1);
+  playing_ = (GetPlayState() == SL_PLAYSTATE_PLAYING);
+  RTC_DCHECK(playing_);
+  return 0;
+}
+
+int OpenSLESPlayer::StopPlayout() {
+  ALOGD("StopPlayout[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !playing_) {
+    return 0;
+  }
+  // Stop playing by setting the play state to SL_PLAYSTATE_STOPPED.
+  RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_STOPPED), -1);
+  // Clear the buffer queue to flush out any remaining data.
+  RETURN_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_), -1);
+#if RTC_DCHECK_IS_ON
+  // Verify that the buffer queue is in fact cleared as it should.
+  SLAndroidSimpleBufferQueueState buffer_queue_state;
+  (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &buffer_queue_state);
+  RTC_DCHECK_EQ(0, buffer_queue_state.count);
+  RTC_DCHECK_EQ(0, buffer_queue_state.index);
+#endif
+  // The number of lower latency audio players is limited, hence we create the
+  // audio player in Start() and destroy it in Stop().
+  DestroyAudioPlayer();
+  thread_checker_opensles_.Detach();
+  initialized_ = false;
+  playing_ = false;
+  return 0;
+}
+
+int OpenSLESPlayer::SpeakerVolumeIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int OpenSLESPlayer::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int OpenSLESPlayer::MinSpeakerVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int OpenSLESPlayer::SetSpeakerVolume(uint32_t volume) {
+  return -1;
+}
+
+int OpenSLESPlayer::SpeakerVolume(uint32_t& volume) const {
+  return -1;
+}
+
+void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  ALOGD("AttachAudioBuffer");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
+  audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetPlayoutChannels(%zu)", channels);
+  audio_device_buffer_->SetPlayoutChannels(channels);
+  RTC_CHECK(audio_device_buffer_);
+  AllocateDataBuffers();
+}
+
+void OpenSLESPlayer::AllocateDataBuffers() {
+  ALOGD("AllocateDataBuffers");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_CHECK(audio_device_buffer_);
+  // Create a modified audio buffer class which allows us to ask for any number
+  // of samples (and not only multiple of 10ms) to match the native OpenSL ES
+  // buffer size. The native buffer size corresponds to the
+  // PROPERTY_OUTPUT_FRAMES_PER_BUFFER property which is the number of audio
+  // frames that the HAL (Hardware Abstraction Layer) buffer can hold. It is
+  // recommended to construct audio buffers so that they contain an exact
+  // multiple of this number. If so, callbacks will occur at regular intervals,
+  // which reduces jitter.
+  const size_t buffer_size_in_samples =
+      audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
+  ALOGD("native buffer size: %zu", buffer_size_in_samples);
+  ALOGD("native buffer size in ms: %.2f",
+        audio_parameters_.GetBufferSizeInMilliseconds());
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+  // Allocated memory for audio buffers.
+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
+    audio_buffers_[i].reset(new SLint16[buffer_size_in_samples]);
+  }
+}
+
+bool OpenSLESPlayer::ObtainEngineInterface() {
+  ALOGD("ObtainEngineInterface");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (engine_)
+    return true;
+  // Get access to (or create if not already existing) the global OpenSL Engine
+  // object.
+  SLObjectItf engine_object = audio_manager_->GetOpenSLEngine();
+  if (engine_object == nullptr) {
+    ALOGE("Failed to access the global OpenSL engine");
+    return false;
+  }
+  // Get the SL Engine Interface which is implicit.
+  RETURN_ON_ERROR(
+      (*engine_object)->GetInterface(engine_object, SL_IID_ENGINE, &engine_),
+      false);
+  return true;
+}
+
+bool OpenSLESPlayer::CreateMix() {
+  ALOGD("CreateMix");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(engine_);
+  if (output_mix_.Get())
+    return true;
+
+  // Create the ouput mix on the engine object. No interfaces will be used.
+  RETURN_ON_ERROR((*engine_)->CreateOutputMix(engine_, output_mix_.Receive(), 0,
+                                              nullptr, nullptr),
+                  false);
+  RETURN_ON_ERROR(output_mix_->Realize(output_mix_.Get(), SL_BOOLEAN_FALSE),
+                  false);
+  return true;
+}
+
+void OpenSLESPlayer::DestroyMix() {
+  ALOGD("DestroyMix");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!output_mix_.Get())
+    return;
+  output_mix_.Reset();
+}
+
+bool OpenSLESPlayer::CreateAudioPlayer() {
+  ALOGD("CreateAudioPlayer");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(output_mix_.Get());
+  if (player_object_.Get())
+    return true;
+  RTC_DCHECK(!player_);
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_DCHECK(!volume_);
+
+  // source: Android Simple Buffer Queue Data Locator is source.
+  SLDataLocator_AndroidSimpleBufferQueue simple_buffer_queue = {
+      SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
+      static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
+  SLDataSource audio_source = {&simple_buffer_queue, &pcm_format_};
+
+  // sink: OutputMix-based data is sink.
+  SLDataLocator_OutputMix locator_output_mix = {SL_DATALOCATOR_OUTPUTMIX,
+                                                output_mix_.Get()};
+  SLDataSink audio_sink = {&locator_output_mix, nullptr};
+
+  // Define interfaces that we indend to use and realize.
+  const SLInterfaceID interface_ids[] = {SL_IID_ANDROIDCONFIGURATION,
+                                         SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
+  const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE,
+                                          SL_BOOLEAN_TRUE};
+
+  // Create the audio player on the engine interface.
+  RETURN_ON_ERROR(
+      (*engine_)->CreateAudioPlayer(
+          engine_, player_object_.Receive(), &audio_source, &audio_sink,
+          arraysize(interface_ids), interface_ids, interface_required),
+      false);
+
+  // Use the Android configuration interface to set platform-specific
+  // parameters. Should be done before player is realized.
+  SLAndroidConfigurationItf player_config;
+  RETURN_ON_ERROR(
+      player_object_->GetInterface(player_object_.Get(),
+                                   SL_IID_ANDROIDCONFIGURATION, &player_config),
+      false);
+  // Set audio player configuration to SL_ANDROID_STREAM_VOICE which
+  // corresponds to android.media.AudioManager.STREAM_VOICE_CALL.
+  SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
+  RETURN_ON_ERROR(
+      (*player_config)
+          ->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE,
+                             &stream_type, sizeof(SLint32)),
+      false);
+
+  // Realize the audio player object after configuration has been set.
+  RETURN_ON_ERROR(
+      player_object_->Realize(player_object_.Get(), SL_BOOLEAN_FALSE), false);
+
+  // Get the SLPlayItf interface on the audio player.
+  RETURN_ON_ERROR(
+      player_object_->GetInterface(player_object_.Get(), SL_IID_PLAY, &player_),
+      false);
+
+  // Get the SLAndroidSimpleBufferQueueItf interface on the audio player.
+  RETURN_ON_ERROR(
+      player_object_->GetInterface(player_object_.Get(), SL_IID_BUFFERQUEUE,
+                                   &simple_buffer_queue_),
+      false);
+
+  // Register callback method for the Android Simple Buffer Queue interface.
+  // This method will be called when the native audio layer needs audio data.
+  RETURN_ON_ERROR((*simple_buffer_queue_)
+                      ->RegisterCallback(simple_buffer_queue_,
+                                         SimpleBufferQueueCallback, this),
+                  false);
+
+  // Get the SLVolumeItf interface on the audio player.
+  RETURN_ON_ERROR(player_object_->GetInterface(player_object_.Get(),
+                                               SL_IID_VOLUME, &volume_),
+                  false);
+
+  // TODO(henrika): might not be required to set volume to max here since it
+  // seems to be default on most devices. Might be required for unit tests.
+  // RETURN_ON_ERROR((*volume_)->SetVolumeLevel(volume_, 0), false);
+
+  return true;
+}
+
+void OpenSLESPlayer::DestroyAudioPlayer() {
+  ALOGD("DestroyAudioPlayer");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!player_object_.Get())
+    return;
+  (*simple_buffer_queue_)
+      ->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
+  player_object_.Reset();
+  player_ = nullptr;
+  simple_buffer_queue_ = nullptr;
+  volume_ = nullptr;
+}
+
+// static
+void OpenSLESPlayer::SimpleBufferQueueCallback(
+    SLAndroidSimpleBufferQueueItf caller,
+    void* context) {
+  OpenSLESPlayer* stream = reinterpret_cast<OpenSLESPlayer*>(context);
+  stream->FillBufferQueue();
+}
+
+void OpenSLESPlayer::FillBufferQueue() {
+  RTC_DCHECK(thread_checker_opensles_.IsCurrent());
+  SLuint32 state = GetPlayState();
+  if (state != SL_PLAYSTATE_PLAYING) {
+    ALOGW("Buffer callback in non-playing state!");
+    return;
+  }
+  EnqueuePlayoutData(false);
+}
+
+void OpenSLESPlayer::EnqueuePlayoutData(bool silence) {
+  // Check delta time between two successive callbacks and provide a warning
+  // if it becomes very large.
+  // TODO(henrika): using 150ms as upper limit but this value is rather random.
+  const uint32_t current_time = rtc::Time();
+  const uint32_t diff = current_time - last_play_time_;
+  if (diff > 150) {
+    ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
+  }
+  last_play_time_ = current_time;
+  SLint8* audio_ptr8 =
+      reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get());
+  if (silence) {
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    // Avoid acquiring real audio data from WebRTC and fill the buffer with
+    // zeros instead. Used to prime the buffer with silence and to avoid asking
+    // for audio data from two different threads.
+    memset(audio_ptr8, 0, audio_parameters_.GetBytesPerBuffer());
+  } else {
+    RTC_DCHECK(thread_checker_opensles_.IsCurrent());
+    // Read audio data from the WebRTC source using the FineAudioBuffer object
+    // to adjust for differences in buffer size between WebRTC (10ms) and native
+    // OpenSL ES. Use hardcoded delay estimate since OpenSL ES does not support
+    // delay estimation.
+    fine_audio_buffer_->GetPlayoutData(
+        rtc::ArrayView<int16_t>(audio_buffers_[buffer_index_].get(),
+                                audio_parameters_.frames_per_buffer() *
+                                    audio_parameters_.channels()),
+        25);
+  }
+  // Enqueue the decoded audio buffer for playback.
+  SLresult err = (*simple_buffer_queue_)
+                     ->Enqueue(simple_buffer_queue_, audio_ptr8,
+                               audio_parameters_.GetBytesPerBuffer());
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("Enqueue failed: %d", err);
+  }
+  buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
+}
+
+SLuint32 OpenSLESPlayer::GetPlayState() const {
+  RTC_DCHECK(player_);
+  SLuint32 state;
+  SLresult err = (*player_)->GetPlayState(player_, &state);
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("GetPlayState failed: %d", err);
+  }
+  return state;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/opensles_player.h b/third_party/libwebrtc/modules/audio_device/android/opensles_player.h
new file mode 100644
index 0000000000..41593a448f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/opensles_player.h
@@ -0,0 +1,195 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <SLES/OpenSLES_AndroidConfiguration.h>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/android/opensles_common.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+class FineAudioBuffer;
+
+// Implements 16-bit mono PCM audio output support for Android using the
+// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread. Decoded audio
+// buffers are requested on a dedicated internal thread managed by the OpenSL
+// ES layer.
+//
+// The existing design forces the user to call InitPlayout() after Stoplayout()
+// to be able to call StartPlayout() again. This is inline with how the Java-
+// based implementation works.
+//
+// OpenSL ES is a native C API which have no Dalvik-related overhead such as
+// garbage collection pauses and it supports reduced audio output latency.
+// If the device doesn't claim this feature but supports API level 9 (Android
+// platform version 2.3) or later, then we can still use the OpenSL ES APIs but
+// the output latency may be higher.
+class OpenSLESPlayer {
+ public:
+  // Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
+  // required for lower latency. Beginning with API level 18 (Android 4.3), a
+  // buffer count of 1 is sufficient for lower latency. In addition, the buffer
+  // size and sample rate must be compatible with the device's native output
+  // configuration provided via the audio manager at construction.
+  // TODO(henrika): perhaps set this value dynamically based on OS version.
+  static const int kNumOfOpenSLESBuffers = 2;
+
+  explicit OpenSLESPlayer(AudioManager* audio_manager);
+  ~OpenSLESPlayer();
+
+  int Init();
+  int Terminate();
+
+  int InitPlayout();
+  bool PlayoutIsInitialized() const { return initialized_; }
+
+  int StartPlayout();
+  int StopPlayout();
+  bool Playing() const { return playing_; }
+
+  int SpeakerVolumeIsAvailable(bool& available);
+  int SetSpeakerVolume(uint32_t volume);
+  int SpeakerVolume(uint32_t& volume) const;
+  int MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int MinSpeakerVolume(uint32_t& minVolume) const;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+ private:
+  // These callback methods are called when data is required for playout.
+  // They are both called from an internal "OpenSL ES thread" which is not
+  // attached to the Dalvik VM.
+  static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
+                                        void* context);
+  void FillBufferQueue();
+  // Reads audio data in PCM format using the AudioDeviceBuffer.
+  // Can be called both on the main thread (during Start()) and from the
+  // internal audio thread while output streaming is active.
+  // If the `silence` flag is set, the audio is filled with zeros instead of
+  // asking the WebRTC layer for real audio data. This procedure is also known
+  // as audio priming.
+  void EnqueuePlayoutData(bool silence);
+
+  // Allocate memory for audio buffers which will be used to render audio
+  // via the SLAndroidSimpleBufferQueueItf interface.
+  void AllocateDataBuffers();
+
+  // Obtaines the SL Engine Interface from the existing global Engine object.
+  // The interface exposes creation methods of all the OpenSL ES object types.
+  // This method defines the `engine_` member variable.
+  bool ObtainEngineInterface();
+
+  // Creates/destroys the output mix object.
+  bool CreateMix();
+  void DestroyMix();
+
+  // Creates/destroys the audio player and the simple-buffer object.
+  // Also creates the volume object.
+  bool CreateAudioPlayer();
+  void DestroyAudioPlayer();
+
+  SLuint32 GetPlayState() const;
+
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
+  // non-application thread which is not attached to the Dalvik JVM.
+  // Detached during construction of this object.
+  SequenceChecker thread_checker_opensles_;
+
+  // Raw pointer to the audio manager injected at construction. Used to cache
+  // audio parameters and to access the global SL engine object needed by the
+  // ObtainEngineInterface() method. The audio manager outlives any instance of
+  // this class.
+  AudioManager* audio_manager_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_;
+
+  bool initialized_;
+  bool playing_;
+
+  // PCM-type format definition.
+  // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
+  // 32-bit float representation is needed.
+  SLDataFormat_PCM pcm_format_;
+
+  // Queue of audio buffers to be used by the player object for rendering
+  // audio.
+  std::unique_ptr<SLint16[]> audio_buffers_[kNumOfOpenSLESBuffers];
+
+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
+  // in chunks of 10ms. It then allows for this data to be pulled in
+  // a finer or coarser granularity. I.e. interacting with this class instead
+  // of directly with the AudioDeviceBuffer one can ask for any number of
+  // audio data samples.
+  // Example: native buffer size can be 192 audio frames at 48kHz sample rate.
+  // WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 192
+  // in each callback (one every 4th ms). This class can then ask for 192 and
+  // the FineAudioBuffer will ask WebRTC for new data approximately only every
+  // second callback and also cache non-utilized audio.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
+  // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
+  int buffer_index_;
+
+  // This interface exposes creation methods for all the OpenSL ES object types.
+  // It is the OpenSL ES API entry point.
+  SLEngineItf engine_;
+
+  // Output mix object to be used by the player object.
+  webrtc::ScopedSLObjectItf output_mix_;
+
+  // The audio player media object plays out audio to the speakers. It also
+  // supports volume control.
+  webrtc::ScopedSLObjectItf player_object_;
+
+  // This interface is supported on the audio player and it controls the state
+  // of the audio player.
+  SLPlayItf player_;
+
+  // The Android Simple Buffer Queue interface is supported on the audio player
+  // and it provides methods to send audio data from the source to the audio
+  // player for rendering.
+  SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
+
+  // This interface exposes controls for manipulating the object’s audio volume
+  // properties. This interface is supported on the Audio Player object.
+  SLVolumeItf volume_;
+
+  // Last time the OpenSL ES layer asked for audio data to play out.
+  uint32_t last_play_time_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/android/opensles_recorder.cc b/third_party/libwebrtc/modules/audio_device/android/opensles_recorder.cc
new file mode 100644
index 0000000000..4e0c26dbf0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/opensles_recorder.cc
@@ -0,0 +1,431 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/opensles_recorder.h"
+
+#include <android/log.h>
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+
+#define TAG "OpenSLESRecorder"
+#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
+#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
+#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
+
+#define LOG_ON_ERROR(op)                                    \
+  [](SLresult err) {                                        \
+    if (err != SL_RESULT_SUCCESS) {                         \
+      ALOGE("%s:%d %s failed: %s", __FILE__, __LINE__, #op, \
+            GetSLErrorString(err));                         \
+      return true;                                          \
+    }                                                       \
+    return false;                                           \
+  }(op)
+
+namespace webrtc {
+
+OpenSLESRecorder::OpenSLESRecorder(AudioManager* audio_manager)
+    : audio_manager_(audio_manager),
+      audio_parameters_(audio_manager->GetRecordAudioParameters()),
+      audio_device_buffer_(nullptr),
+      initialized_(false),
+      recording_(false),
+      engine_(nullptr),
+      recorder_(nullptr),
+      simple_buffer_queue_(nullptr),
+      buffer_index_(0),
+      last_rec_time_(0) {
+  ALOGD("ctor[tid=%d]", rtc::CurrentThreadId());
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the internal  audio thread.
+  thread_checker_opensles_.Detach();
+  // Use native audio output parameters provided by the audio manager and
+  // define the PCM format structure.
+  pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
+                                       audio_parameters_.sample_rate(),
+                                       audio_parameters_.bits_per_sample());
+}
+
+OpenSLESRecorder::~OpenSLESRecorder() {
+  ALOGD("dtor[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+  DestroyAudioRecorder();
+  engine_ = nullptr;
+  RTC_DCHECK(!engine_);
+  RTC_DCHECK(!recorder_);
+  RTC_DCHECK(!simple_buffer_queue_);
+}
+
+int OpenSLESRecorder::Init() {
+  ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (audio_parameters_.channels() == 2) {
+    ALOGD("Stereo mode is enabled");
+  }
+  return 0;
+}
+
+int OpenSLESRecorder::Terminate() {
+  ALOGD("Terminate[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopRecording();
+  return 0;
+}
+
+int OpenSLESRecorder::InitRecording() {
+  ALOGD("InitRecording[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  if (!ObtainEngineInterface()) {
+    ALOGE("Failed to obtain SL Engine interface");
+    return -1;
+  }
+  CreateAudioRecorder();
+  initialized_ = true;
+  buffer_index_ = 0;
+  return 0;
+}
+
+int OpenSLESRecorder::StartRecording() {
+  ALOGD("StartRecording[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!recording_);
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetRecord();
+  }
+  // Add buffers to the queue before changing state to SL_RECORDSTATE_RECORDING
+  // to ensure that recording starts as soon as the state is modified. On some
+  // devices, SLAndroidSimpleBufferQueue::Clear() used in Stop() does not flush
+  // the buffers as intended and we therefore check the number of buffers
+  // already queued first. Enqueue() can return SL_RESULT_BUFFER_INSUFFICIENT
+  // otherwise.
+  int num_buffers_in_queue = GetBufferCount();
+  for (int i = 0; i < kNumOfOpenSLESBuffers - num_buffers_in_queue; ++i) {
+    if (!EnqueueAudioBuffer()) {
+      recording_ = false;
+      return -1;
+    }
+  }
+  num_buffers_in_queue = GetBufferCount();
+  RTC_DCHECK_EQ(num_buffers_in_queue, kNumOfOpenSLESBuffers);
+  LogBufferState();
+  // Start audio recording by changing the state to SL_RECORDSTATE_RECORDING.
+  // Given that buffers are already enqueued, recording should start at once.
+  // The macro returns -1 if recording fails to start.
+  last_rec_time_ = rtc::Time();
+  if (LOG_ON_ERROR(
+          (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_RECORDING))) {
+    return -1;
+  }
+  recording_ = (GetRecordState() == SL_RECORDSTATE_RECORDING);
+  RTC_DCHECK(recording_);
+  return 0;
+}
+
+int OpenSLESRecorder::StopRecording() {
+  ALOGD("StopRecording[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !recording_) {
+    return 0;
+  }
+  // Stop recording by setting the record state to SL_RECORDSTATE_STOPPED.
+  if (LOG_ON_ERROR(
+          (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_STOPPED))) {
+    return -1;
+  }
+  // Clear the buffer queue to get rid of old data when resuming recording.
+  if (LOG_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_))) {
+    return -1;
+  }
+  thread_checker_opensles_.Detach();
+  initialized_ = false;
+  recording_ = false;
+  return 0;
+}
+
+void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
+  ALOGD("AttachAudioBuffer");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_CHECK(audio_buffer);
+  audio_device_buffer_ = audio_buffer;
+  // Ensure that the audio device buffer is informed about the native sample
+  // rate used on the recording side.
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
+  audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
+  // Ensure that the audio device buffer is informed about the number of
+  // channels preferred by the OS on the recording side.
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetRecordingChannels(%zu)", channels);
+  audio_device_buffer_->SetRecordingChannels(channels);
+  // Allocated memory for internal data buffers given existing audio parameters.
+  AllocateDataBuffers();
+}
+
+int OpenSLESRecorder::EnableBuiltInAEC(bool enable) {
+  ALOGD("EnableBuiltInAEC(%d)", enable);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  ALOGE("Not implemented");
+  return 0;
+}
+
+int OpenSLESRecorder::EnableBuiltInAGC(bool enable) {
+  ALOGD("EnableBuiltInAGC(%d)", enable);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  ALOGE("Not implemented");
+  return 0;
+}
+
+int OpenSLESRecorder::EnableBuiltInNS(bool enable) {
+  ALOGD("EnableBuiltInNS(%d)", enable);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  ALOGE("Not implemented");
+  return 0;
+}
+
+bool OpenSLESRecorder::ObtainEngineInterface() {
+  ALOGD("ObtainEngineInterface");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (engine_)
+    return true;
+  // Get access to (or create if not already existing) the global OpenSL Engine
+  // object.
+  SLObjectItf engine_object = audio_manager_->GetOpenSLEngine();
+  if (engine_object == nullptr) {
+    ALOGE("Failed to access the global OpenSL engine");
+    return false;
+  }
+  // Get the SL Engine Interface which is implicit.
+  if (LOG_ON_ERROR(
+          (*engine_object)
+              ->GetInterface(engine_object, SL_IID_ENGINE, &engine_))) {
+    return false;
+  }
+  return true;
+}
+
+bool OpenSLESRecorder::CreateAudioRecorder() {
+  ALOGD("CreateAudioRecorder");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (recorder_object_.Get())
+    return true;
+  RTC_DCHECK(!recorder_);
+  RTC_DCHECK(!simple_buffer_queue_);
+
+  // Audio source configuration.
+  SLDataLocator_IODevice mic_locator = {SL_DATALOCATOR_IODEVICE,
+                                        SL_IODEVICE_AUDIOINPUT,
+                                        SL_DEFAULTDEVICEID_AUDIOINPUT, NULL};
+  SLDataSource audio_source = {&mic_locator, NULL};
+
+  // Audio sink configuration.
+  SLDataLocator_AndroidSimpleBufferQueue buffer_queue = {
+      SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
+      static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
+  SLDataSink audio_sink = {&buffer_queue, &pcm_format_};
+
+  // Create the audio recorder object (requires the RECORD_AUDIO permission).
+  // Do not realize the recorder yet. Set the configuration first.
+  const SLInterfaceID interface_id[] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+                                        SL_IID_ANDROIDCONFIGURATION};
+  const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
+  if (LOG_ON_ERROR((*engine_)->CreateAudioRecorder(
+          engine_, recorder_object_.Receive(), &audio_source, &audio_sink,
+          arraysize(interface_id), interface_id, interface_required))) {
+    return false;
+  }
+
+  // Configure the audio recorder (before it is realized).
+  SLAndroidConfigurationItf recorder_config;
+  if (LOG_ON_ERROR((recorder_object_->GetInterface(recorder_object_.Get(),
+                                                   SL_IID_ANDROIDCONFIGURATION,
+                                                   &recorder_config)))) {
+    return false;
+  }
+
+  // Uses the default microphone tuned for audio communication.
+  // Note that, SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION leads to a fast
+  // track but also excludes usage of required effects like AEC, AGC and NS.
+  // SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION
+  SLint32 stream_type = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
+  if (LOG_ON_ERROR(((*recorder_config)
+                        ->SetConfiguration(recorder_config,
+                                           SL_ANDROID_KEY_RECORDING_PRESET,
+                                           &stream_type, sizeof(SLint32))))) {
+    return false;
+  }
+
+  // The audio recorder can now be realized (in synchronous mode).
+  if (LOG_ON_ERROR((recorder_object_->Realize(recorder_object_.Get(),
+                                              SL_BOOLEAN_FALSE)))) {
+    return false;
+  }
+
+  // Get the implicit recorder interface (SL_IID_RECORD).
+  if (LOG_ON_ERROR((recorder_object_->GetInterface(
+          recorder_object_.Get(), SL_IID_RECORD, &recorder_)))) {
+    return false;
+  }
+
+  // Get the simple buffer queue interface (SL_IID_ANDROIDSIMPLEBUFFERQUEUE).
+  // It was explicitly requested.
+  if (LOG_ON_ERROR((recorder_object_->GetInterface(
+          recorder_object_.Get(), SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+          &simple_buffer_queue_)))) {
+    return false;
+  }
+
+  // Register the input callback for the simple buffer queue.
+  // This callback will be called when receiving new data from the device.
+  if (LOG_ON_ERROR(((*simple_buffer_queue_)
+                        ->RegisterCallback(simple_buffer_queue_,
+                                           SimpleBufferQueueCallback, this)))) {
+    return false;
+  }
+  return true;
+}
+
+void OpenSLESRecorder::DestroyAudioRecorder() {
+  ALOGD("DestroyAudioRecorder");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!recorder_object_.Get())
+    return;
+  (*simple_buffer_queue_)
+      ->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
+  recorder_object_.Reset();
+  recorder_ = nullptr;
+  simple_buffer_queue_ = nullptr;
+}
+
+void OpenSLESRecorder::SimpleBufferQueueCallback(
+    SLAndroidSimpleBufferQueueItf buffer_queue,
+    void* context) {
+  OpenSLESRecorder* stream = static_cast<OpenSLESRecorder*>(context);
+  stream->ReadBufferQueue();
+}
+
+void OpenSLESRecorder::AllocateDataBuffers() {
+  ALOGD("AllocateDataBuffers");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_CHECK(audio_device_buffer_);
+  // Create a modified audio buffer class which allows us to deliver any number
+  // of samples (and not only multiple of 10ms) to match the native audio unit
+  // buffer size.
+  ALOGD("frames per native buffer: %zu", audio_parameters_.frames_per_buffer());
+  ALOGD("frames per 10ms buffer: %zu",
+        audio_parameters_.frames_per_10ms_buffer());
+  ALOGD("bytes per native buffer: %zu", audio_parameters_.GetBytesPerBuffer());
+  ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
+  RTC_DCHECK(audio_device_buffer_);
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+  // Allocate queue of audio buffers that stores recorded audio samples.
+  const int buffer_size_samples =
+      audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
+  audio_buffers_.reset(new std::unique_ptr<SLint16[]>[kNumOfOpenSLESBuffers]);
+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
+    audio_buffers_[i].reset(new SLint16[buffer_size_samples]);
+  }
+}
+
+void OpenSLESRecorder::ReadBufferQueue() {
+  RTC_DCHECK(thread_checker_opensles_.IsCurrent());
+  SLuint32 state = GetRecordState();
+  if (state != SL_RECORDSTATE_RECORDING) {
+    ALOGW("Buffer callback in non-recording state!");
+    return;
+  }
+  // Check delta time between two successive callbacks and provide a warning
+  // if it becomes very large.
+  // TODO(henrika): using 150ms as upper limit but this value is rather random.
+  const uint32_t current_time = rtc::Time();
+  const uint32_t diff = current_time - last_rec_time_;
+  if (diff > 150) {
+    ALOGW("Bad OpenSL ES record timing, dT=%u [ms]", diff);
+  }
+  last_rec_time_ = current_time;
+  // Send recorded audio data to the WebRTC sink.
+  // TODO(henrika): fix delay estimates. It is OK to use fixed values for now
+  // since there is no support to turn off built-in EC in combination with
+  // OpenSL ES anyhow. Hence, as is, the WebRTC based AEC (which would use
+  // these estimates) will never be active.
+  fine_audio_buffer_->DeliverRecordedData(
+      rtc::ArrayView<const int16_t>(
+          audio_buffers_[buffer_index_].get(),
+          audio_parameters_.frames_per_buffer() * audio_parameters_.channels()),
+      25);
+  // Enqueue the utilized audio buffer and use if for recording again.
+  EnqueueAudioBuffer();
+}
+
+bool OpenSLESRecorder::EnqueueAudioBuffer() {
+  SLresult err =
+      (*simple_buffer_queue_)
+          ->Enqueue(
+              simple_buffer_queue_,
+              reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get()),
+              audio_parameters_.GetBytesPerBuffer());
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("Enqueue failed: %s", GetSLErrorString(err));
+    return false;
+  }
+  buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
+  return true;
+}
+
+SLuint32 OpenSLESRecorder::GetRecordState() const {
+  RTC_DCHECK(recorder_);
+  SLuint32 state;
+  SLresult err = (*recorder_)->GetRecordState(recorder_, &state);
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("GetRecordState failed: %s", GetSLErrorString(err));
+  }
+  return state;
+}
+
+SLAndroidSimpleBufferQueueState OpenSLESRecorder::GetBufferQueueState() const {
+  RTC_DCHECK(simple_buffer_queue_);
+  // state.count: Number of buffers currently in the queue.
+  // state.index: Index of the currently filling buffer. This is a linear index
+  // that keeps a cumulative count of the number of buffers recorded.
+  SLAndroidSimpleBufferQueueState state;
+  SLresult err =
+      (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &state);
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("GetState failed: %s", GetSLErrorString(err));
+  }
+  return state;
+}
+
+void OpenSLESRecorder::LogBufferState() const {
+  SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
+  ALOGD("state.count:%d state.index:%d", state.count, state.index);
+}
+
+SLuint32 OpenSLESRecorder::GetBufferCount() {
+  SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
+  return state.count;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/android/opensles_recorder.h b/third_party/libwebrtc/modules/audio_device/android/opensles_recorder.h
new file mode 100644
index 0000000000..e659c3c157
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/android/opensles_recorder.h
@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_RECORDER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_RECORDER_H_
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <SLES/OpenSLES_AndroidConfiguration.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/android/opensles_common.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+class FineAudioBuffer;
+
+// Implements 16-bit mono PCM audio input support for Android using the
+// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread. Recorded audio
+// buffers are provided on a dedicated internal thread managed by the OpenSL
+// ES layer.
+//
+// The existing design forces the user to call InitRecording() after
+// StopRecording() to be able to call StartRecording() again. This is inline
+// with how the Java-based implementation works.
+//
+// As of API level 21, lower latency audio input is supported on select devices.
+// To take advantage of this feature, first confirm that lower latency output is
+// available. The capability for lower latency output is a prerequisite for the
+// lower latency input feature. Then, create an AudioRecorder with the same
+// sample rate and buffer size as would be used for output. OpenSL ES interfaces
+// for input effects preclude the lower latency path.
+// See https://developer.android.com/ndk/guides/audio/opensl-prog-notes.html
+// for more details.
+class OpenSLESRecorder {
+ public:
+  // Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
+  // required for lower latency. Beginning with API level 18 (Android 4.3), a
+  // buffer count of 1 is sufficient for lower latency. In addition, the buffer
+  // size and sample rate must be compatible with the device's native input
+  // configuration provided via the audio manager at construction.
+  // TODO(henrika): perhaps set this value dynamically based on OS version.
+  static const int kNumOfOpenSLESBuffers = 2;
+
+  explicit OpenSLESRecorder(AudioManager* audio_manager);
+  ~OpenSLESRecorder();
+
+  int Init();
+  int Terminate();
+
+  int InitRecording();
+  bool RecordingIsInitialized() const { return initialized_; }
+
+  int StartRecording();
+  int StopRecording();
+  bool Recording() const { return recording_; }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer);
+
+  // TODO(henrika): add support using OpenSL ES APIs when available.
+  int EnableBuiltInAEC(bool enable);
+  int EnableBuiltInAGC(bool enable);
+  int EnableBuiltInNS(bool enable);
+
+ private:
+  // Obtaines the SL Engine Interface from the existing global Engine object.
+  // The interface exposes creation methods of all the OpenSL ES object types.
+  // This method defines the `engine_` member variable.
+  bool ObtainEngineInterface();
+
+  // Creates/destroys the audio recorder and the simple-buffer queue object.
+  bool CreateAudioRecorder();
+  void DestroyAudioRecorder();
+
+  // Allocate memory for audio buffers which will be used to capture audio
+  // via the SLAndroidSimpleBufferQueueItf interface.
+  void AllocateDataBuffers();
+
+  // These callback methods are called when data has been written to the input
+  // buffer queue. They are both called from an internal "OpenSL ES thread"
+  // which is not attached to the Dalvik VM.
+  static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
+                                        void* context);
+  void ReadBufferQueue();
+
+  // Wraps calls to SLAndroidSimpleBufferQueueState::Enqueue() and it can be
+  // called both on the main thread (but before recording has started) and from
+  // the internal audio thread while input streaming is active. It uses
+  // `simple_buffer_queue_` but no lock is needed since the initial calls from
+  // the main thread and the native callback thread are mutually exclusive.
+  bool EnqueueAudioBuffer();
+
+  // Returns the current recorder state.
+  SLuint32 GetRecordState() const;
+
+  // Returns the current buffer queue state.
+  SLAndroidSimpleBufferQueueState GetBufferQueueState() const;
+
+  // Number of buffers currently in the queue.
+  SLuint32 GetBufferCount();
+
+  // Prints a log message of the current queue state. Can be used for debugging
+  // purposes.
+  void LogBufferState() const;
+
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
+  // non-application thread which is not attached to the Dalvik JVM.
+  // Detached during construction of this object.
+  SequenceChecker thread_checker_opensles_;
+
+  // Raw pointer to the audio manager injected at construction. Used to cache
+  // audio parameters and to access the global SL engine object needed by the
+  // ObtainEngineInterface() method. The audio manager outlives any instance of
+  // this class.
+  AudioManager* const audio_manager_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_;
+
+  // PCM-type format definition.
+  // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
+  // 32-bit float representation is needed.
+  SLDataFormat_PCM pcm_format_;
+
+  bool initialized_;
+  bool recording_;
+
+  // This interface exposes creation methods for all the OpenSL ES object types.
+  // It is the OpenSL ES API entry point.
+  SLEngineItf engine_;
+
+  // The audio recorder media object records audio to the destination specified
+  // by the data sink capturing it from the input specified by the data source.
+  webrtc::ScopedSLObjectItf recorder_object_;
+
+  // This interface is supported on the audio recorder object and it controls
+  // the state of the audio recorder.
+  SLRecordItf recorder_;
+
+  // The Android Simple Buffer Queue interface is supported on the audio
+  // recorder. For recording, an app should enqueue empty buffers. When a
+  // registered callback sends notification that the system has finished writing
+  // data to the buffer, the app can read the buffer.
+  SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
+
+  // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
+  // chunks of audio.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Queue of audio buffers to be used by the recorder object for capturing
+  // audio. They will be used in a Round-robin way and the size of each buffer
+  // is given by AudioParameters::frames_per_buffer(), i.e., it corresponds to
+  // the native OpenSL ES buffer size.
+  std::unique_ptr<std::unique_ptr<SLint16[]>[]> audio_buffers_;
+
+  // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
+  // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
+  int buffer_index_;
+
+  // Last time the OpenSL ES layer delivered recorded audio data.
+  uint32_t last_rec_time_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_RECORDER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_buffer.cc b/third_party/libwebrtc/modules/audio_device/audio_device_buffer.cc
new file mode 100644
index 0000000000..b1be445e0d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_buffer.cc
@@ -0,0 +1,518 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/audio_device_buffer.h"
+
+#include <string.h>
+
+#include <cmath>
+#include <cstddef>
+#include <cstdint>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+static const char kTimerQueueName[] = "AudioDeviceBufferTimer";
+
+// Time between two sucessive calls to LogStats().
+static const size_t kTimerIntervalInSeconds = 10;
+static const size_t kTimerIntervalInMilliseconds =
+    kTimerIntervalInSeconds * rtc::kNumMillisecsPerSec;
+// Min time required to qualify an audio session as a "call". If playout or
+// recording has been active for less than this time we will not store any
+// logs or UMA stats but instead consider the call as too short.
+static const size_t kMinValidCallTimeTimeInSeconds = 10;
+static const size_t kMinValidCallTimeTimeInMilliseconds =
+    kMinValidCallTimeTimeInSeconds * rtc::kNumMillisecsPerSec;
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+static const double k2Pi = 6.28318530717959;
+#endif
+
+AudioDeviceBuffer::AudioDeviceBuffer(TaskQueueFactory* task_queue_factory)
+    : task_queue_(task_queue_factory->CreateTaskQueue(
+          kTimerQueueName,
+          TaskQueueFactory::Priority::NORMAL)),
+      audio_transport_cb_(nullptr),
+      rec_sample_rate_(0),
+      play_sample_rate_(0),
+      rec_channels_(0),
+      play_channels_(0),
+      playing_(false),
+      recording_(false),
+      typing_status_(false),
+      play_delay_ms_(0),
+      rec_delay_ms_(0),
+      num_stat_reports_(0),
+      last_timer_task_time_(0),
+      rec_stat_count_(0),
+      play_stat_count_(0),
+      play_start_time_(0),
+      only_silence_recorded_(true),
+      log_stats_(false) {
+  RTC_LOG(LS_INFO) << "AudioDeviceBuffer::ctor";
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  phase_ = 0.0;
+  RTC_LOG(LS_WARNING) << "AUDIO_DEVICE_PLAYS_SINUS_TONE is defined!";
+#endif
+}
+
+AudioDeviceBuffer::~AudioDeviceBuffer() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DCHECK(!playing_);
+  RTC_DCHECK(!recording_);
+  RTC_LOG(LS_INFO) << "AudioDeviceBuffer::~dtor";
+}
+
+int32_t AudioDeviceBuffer::RegisterAudioCallback(
+    AudioTransport* audio_callback) {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  if (playing_ || recording_) {
+    RTC_LOG(LS_ERROR) << "Failed to set audio transport since media was active";
+    return -1;
+  }
+  audio_transport_cb_ = audio_callback;
+  return 0;
+}
+
+void AudioDeviceBuffer::StartPlayout() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  // TODO(henrika): allow for usage of DCHECK(!playing_) here instead. Today the
+  // ADM allows calling Start(), Start() by ignoring the second call but it
+  // makes more sense to only allow one call.
+  if (playing_) {
+    return;
+  }
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  // Clear members tracking playout stats and do it on the task queue.
+  task_queue_.PostTask([this] { ResetPlayStats(); });
+  // Start a periodic timer based on task queue if not already done by the
+  // recording side.
+  if (!recording_) {
+    StartPeriodicLogging();
+  }
+  const int64_t now_time = rtc::TimeMillis();
+  // Clear members that are only touched on the main (creating) thread.
+  play_start_time_ = now_time;
+  playing_ = true;
+}
+
+void AudioDeviceBuffer::StartRecording() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (recording_) {
+    return;
+  }
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  // Clear members tracking recording stats and do it on the task queue.
+  task_queue_.PostTask([this] { ResetRecStats(); });
+  // Start a periodic timer based on task queue if not already done by the
+  // playout side.
+  if (!playing_) {
+    StartPeriodicLogging();
+  }
+  // Clear members that will be touched on the main (creating) thread.
+  rec_start_time_ = rtc::TimeMillis();
+  recording_ = true;
+  // And finally a member which can be modified on the native audio thread.
+  // It is safe to do so since we know by design that the owning ADM has not
+  // yet started the native audio recording.
+  only_silence_recorded_ = true;
+}
+
+void AudioDeviceBuffer::StopPlayout() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (!playing_) {
+    return;
+  }
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  playing_ = false;
+  // Stop periodic logging if no more media is active.
+  if (!recording_) {
+    StopPeriodicLogging();
+  }
+  RTC_LOG(LS_INFO) << "total playout time: "
+                   << rtc::TimeSince(play_start_time_);
+}
+
+void AudioDeviceBuffer::StopRecording() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (!recording_) {
+    return;
+  }
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  recording_ = false;
+  // Stop periodic logging if no more media is active.
+  if (!playing_) {
+    StopPeriodicLogging();
+  }
+  // Add UMA histogram to keep track of the case when only zeros have been
+  // recorded. Measurements (max of absolute level) are taken twice per second,
+  // which means that if e.g 10 seconds of audio has been recorded, a total of
+  // 20 level estimates must all be identical to zero to trigger the histogram.
+  // `only_silence_recorded_` can only be cleared on the native audio thread
+  // that drives audio capture but we know by design that the audio has stopped
+  // when this method is called, hence there should not be aby conflicts. Also,
+  // the fact that `only_silence_recorded_` can be affected during the complete
+  // call makes chances of conflicts with potentially one last callback very
+  // small.
+  const size_t time_since_start = rtc::TimeSince(rec_start_time_);
+  if (time_since_start > kMinValidCallTimeTimeInMilliseconds) {
+    const int only_zeros = static_cast<int>(only_silence_recorded_);
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros", only_zeros);
+    RTC_LOG(LS_INFO) << "HISTOGRAM(WebRTC.Audio.RecordedOnlyZeros): "
+                     << only_zeros;
+  }
+  RTC_LOG(LS_INFO) << "total recording time: " << time_since_start;
+}
+
+int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) {
+  RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << fsHz << ")";
+  rec_sample_rate_ = fsHz;
+  return 0;
+}
+
+int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) {
+  RTC_LOG(LS_INFO) << "SetPlayoutSampleRate(" << fsHz << ")";
+  play_sample_rate_ = fsHz;
+  return 0;
+}
+
+uint32_t AudioDeviceBuffer::RecordingSampleRate() const {
+  return rec_sample_rate_;
+}
+
+uint32_t AudioDeviceBuffer::PlayoutSampleRate() const {
+  return play_sample_rate_;
+}
+
+int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
+  RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
+  rec_channels_ = channels;
+  return 0;
+}
+
+int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
+  RTC_LOG(LS_INFO) << "SetPlayoutChannels(" << channels << ")";
+  play_channels_ = channels;
+  return 0;
+}
+
+size_t AudioDeviceBuffer::RecordingChannels() const {
+  return rec_channels_;
+}
+
+size_t AudioDeviceBuffer::PlayoutChannels() const {
+  return play_channels_;
+}
+
+int32_t AudioDeviceBuffer::SetTypingStatus(bool typing_status) {
+  typing_status_ = typing_status;
+  return 0;
+}
+
+void AudioDeviceBuffer::SetVQEData(int play_delay_ms, int rec_delay_ms) {
+  play_delay_ms_ = play_delay_ms;
+  rec_delay_ms_ = rec_delay_ms;
+}
+
+int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
+                                             size_t samples_per_channel) {
+  return SetRecordedBuffer(audio_buffer, samples_per_channel, absl::nullopt);
+}
+
+int32_t AudioDeviceBuffer::SetRecordedBuffer(
+    const void* audio_buffer,
+    size_t samples_per_channel,
+    absl::optional<int64_t> capture_timestamp_ns) {
+  // Copy the complete input buffer to the local buffer.
+  const size_t old_size = rec_buffer_.size();
+  rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer),
+                      rec_channels_ * samples_per_channel);
+  // Keep track of the size of the recording buffer. Only updated when the
+  // size changes, which is a rare event.
+  if (old_size != rec_buffer_.size()) {
+    RTC_LOG(LS_INFO) << "Size of recording buffer: " << rec_buffer_.size();
+  }
+
+  if (capture_timestamp_ns) {
+    capture_timestamp_ns_ =
+        rtc::kNumNanosecsPerMicrosec *
+        timestamp_aligner_.TranslateTimestamp(
+            *capture_timestamp_ns / rtc::kNumNanosecsPerMicrosec,
+            rtc::TimeMicros());
+  }
+  // Derive a new level value twice per second and check if it is non-zero.
+  int16_t max_abs = 0;
+  RTC_DCHECK_LT(rec_stat_count_, 50);
+  if (++rec_stat_count_ >= 50) {
+    // Returns the largest absolute value in a signed 16-bit vector.
+    max_abs = WebRtcSpl_MaxAbsValueW16(rec_buffer_.data(), rec_buffer_.size());
+    rec_stat_count_ = 0;
+    // Set `only_silence_recorded_` to false as soon as at least one detection
+    // of a non-zero audio packet is found. It can only be restored to true
+    // again by restarting the call.
+    if (max_abs > 0) {
+      only_silence_recorded_ = false;
+    }
+  }
+  // Update recording stats which is used as base for periodic logging of the
+  // audio input state.
+  UpdateRecStats(max_abs, samples_per_channel);
+  return 0;
+}
+
+int32_t AudioDeviceBuffer::DeliverRecordedData() {
+  if (!audio_transport_cb_) {
+    RTC_LOG(LS_WARNING) << "Invalid audio transport";
+    return 0;
+  }
+  const size_t frames = rec_buffer_.size() / rec_channels_;
+  const size_t bytes_per_frame = rec_channels_ * sizeof(int16_t);
+  uint32_t new_mic_level_dummy = 0;
+  uint32_t total_delay_ms = play_delay_ms_ + rec_delay_ms_;
+  int32_t res = audio_transport_cb_->RecordedDataIsAvailable(
+      rec_buffer_.data(), frames, bytes_per_frame, rec_channels_,
+      rec_sample_rate_, total_delay_ms, 0, 0, typing_status_,
+      new_mic_level_dummy, capture_timestamp_ns_);
+  if (res == -1) {
+    RTC_LOG(LS_ERROR) << "RecordedDataIsAvailable() failed";
+  }
+  return 0;
+}
+
+int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) {
+  TRACE_EVENT1("webrtc", "AudioDeviceBuffer::RequestPlayoutData",
+               "samples_per_channel", samples_per_channel);
+
+  // The consumer can change the requested size on the fly and we therefore
+  // resize the buffer accordingly. Also takes place at the first call to this
+  // method.
+  const size_t total_samples = play_channels_ * samples_per_channel;
+  if (play_buffer_.size() != total_samples) {
+    play_buffer_.SetSize(total_samples);
+    RTC_LOG(LS_INFO) << "Size of playout buffer: " << play_buffer_.size();
+  }
+
+  size_t num_samples_out(0);
+  // It is currently supported to start playout without a valid audio
+  // transport object. Leads to warning and silence.
+  if (!audio_transport_cb_) {
+    RTC_LOG(LS_WARNING) << "Invalid audio transport";
+    return 0;
+  }
+
+  // Retrieve new 16-bit PCM audio data using the audio transport instance.
+  int64_t elapsed_time_ms = -1;
+  int64_t ntp_time_ms = -1;
+  const size_t bytes_per_frame = play_channels_ * sizeof(int16_t);
+  uint32_t res = audio_transport_cb_->NeedMorePlayData(
+      samples_per_channel, bytes_per_frame, play_channels_, play_sample_rate_,
+      play_buffer_.data(), num_samples_out, &elapsed_time_ms, &ntp_time_ms);
+  if (res != 0) {
+    RTC_LOG(LS_ERROR) << "NeedMorePlayData() failed";
+  }
+
+  // Derive a new level value twice per second.
+  int16_t max_abs = 0;
+  RTC_DCHECK_LT(play_stat_count_, 50);
+  if (++play_stat_count_ >= 50) {
+    // Returns the largest absolute value in a signed 16-bit vector.
+    max_abs =
+        WebRtcSpl_MaxAbsValueW16(play_buffer_.data(), play_buffer_.size());
+    play_stat_count_ = 0;
+  }
+  // Update playout stats which is used as base for periodic logging of the
+  // audio output state.
+  UpdatePlayStats(max_abs, num_samples_out / play_channels_);
+  return static_cast<int32_t>(num_samples_out / play_channels_);
+}
+
+int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
+  RTC_DCHECK_GT(play_buffer_.size(), 0);
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  const double phase_increment =
+      k2Pi * 440.0 / static_cast<double>(play_sample_rate_);
+  int16_t* destination_r = reinterpret_cast<int16_t*>(audio_buffer);
+  if (play_channels_ == 1) {
+    for (size_t i = 0; i < play_buffer_.size(); ++i) {
+      destination_r[i] = static_cast<int16_t>((sin(phase_) * (1 << 14)));
+      phase_ += phase_increment;
+    }
+  } else if (play_channels_ == 2) {
+    for (size_t i = 0; i < play_buffer_.size() / 2; ++i) {
+      destination_r[2 * i] = destination_r[2 * i + 1] =
+          static_cast<int16_t>((sin(phase_) * (1 << 14)));
+      phase_ += phase_increment;
+    }
+  }
+#else
+  memcpy(audio_buffer, play_buffer_.data(),
+         play_buffer_.size() * sizeof(int16_t));
+#endif
+  // Return samples per channel or number of frames.
+  return static_cast<int32_t>(play_buffer_.size() / play_channels_);
+}
+
+void AudioDeviceBuffer::StartPeriodicLogging() {
+  task_queue_.PostTask([this] { LogStats(AudioDeviceBuffer::LOG_START); });
+}
+
+void AudioDeviceBuffer::StopPeriodicLogging() {
+  task_queue_.PostTask([this] { LogStats(AudioDeviceBuffer::LOG_STOP); });
+}
+
+void AudioDeviceBuffer::LogStats(LogState state) {
+  RTC_DCHECK_RUN_ON(&task_queue_);
+  int64_t now_time = rtc::TimeMillis();
+
+  if (state == AudioDeviceBuffer::LOG_START) {
+    // Reset counters at start. We will not add any logging in this state but
+    // the timer will started by posting a new (delayed) task.
+    num_stat_reports_ = 0;
+    last_timer_task_time_ = now_time;
+    log_stats_ = true;
+  } else if (state == AudioDeviceBuffer::LOG_STOP) {
+    // Stop logging and posting new tasks.
+    log_stats_ = false;
+  } else if (state == AudioDeviceBuffer::LOG_ACTIVE) {
+    // Keep logging unless logging was disabled while task was posted.
+  }
+
+  // Avoid adding more logs since we are in STOP mode.
+  if (!log_stats_) {
+    return;
+  }
+
+  int64_t next_callback_time = now_time + kTimerIntervalInMilliseconds;
+  int64_t time_since_last = rtc::TimeDiff(now_time, last_timer_task_time_);
+  last_timer_task_time_ = now_time;
+
+  Stats stats;
+  {
+    MutexLock lock(&lock_);
+    stats = stats_;
+    stats_.max_rec_level = 0;
+    stats_.max_play_level = 0;
+  }
+
+  // Cache current sample rate from atomic members.
+  const uint32_t rec_sample_rate = rec_sample_rate_;
+  const uint32_t play_sample_rate = play_sample_rate_;
+
+  // Log the latest statistics but skip the first two rounds just after state
+  // was set to LOG_START to ensure that we have at least one full stable
+  // 10-second interval for sample-rate estimation. Hence, first printed log
+  // will be after ~20 seconds.
+  if (++num_stat_reports_ > 2 &&
+      static_cast<size_t>(time_since_last) > kTimerIntervalInMilliseconds / 2) {
+    uint32_t diff_samples = stats.rec_samples - last_stats_.rec_samples;
+    float rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
+    uint32_t abs_diff_rate_in_percent = 0;
+    if (rec_sample_rate > 0 && rate > 0) {
+      abs_diff_rate_in_percent = static_cast<uint32_t>(
+          0.5f +
+          ((100.0f * std::abs(rate - rec_sample_rate)) / rec_sample_rate));
+      RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.RecordSampleRateOffsetInPercent",
+                               abs_diff_rate_in_percent);
+      RTC_LOG(LS_INFO) << "[REC : " << time_since_last << "msec, "
+                       << rec_sample_rate / 1000 << "kHz] callbacks: "
+                       << stats.rec_callbacks - last_stats_.rec_callbacks
+                       << ", "
+                          "samples: "
+                       << diff_samples
+                       << ", "
+                          "rate: "
+                       << static_cast<int>(rate + 0.5)
+                       << ", "
+                          "rate diff: "
+                       << abs_diff_rate_in_percent
+                       << "%, "
+                          "level: "
+                       << stats.max_rec_level;
+    }
+
+    diff_samples = stats.play_samples - last_stats_.play_samples;
+    rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
+    abs_diff_rate_in_percent = 0;
+    if (play_sample_rate > 0 && rate > 0) {
+      abs_diff_rate_in_percent = static_cast<uint32_t>(
+          0.5f +
+          ((100.0f * std::abs(rate - play_sample_rate)) / play_sample_rate));
+      RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.PlayoutSampleRateOffsetInPercent",
+                               abs_diff_rate_in_percent);
+      RTC_LOG(LS_INFO) << "[PLAY: " << time_since_last << "msec, "
+                       << play_sample_rate / 1000 << "kHz] callbacks: "
+                       << stats.play_callbacks - last_stats_.play_callbacks
+                       << ", "
+                          "samples: "
+                       << diff_samples
+                       << ", "
+                          "rate: "
+                       << static_cast<int>(rate + 0.5)
+                       << ", "
+                          "rate diff: "
+                       << abs_diff_rate_in_percent
+                       << "%, "
+                          "level: "
+                       << stats.max_play_level;
+    }
+  }
+  last_stats_ = stats;
+
+  int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
+  RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";
+
+  // Keep posting new (delayed) tasks until state is changed to kLogStop.
+  task_queue_.PostDelayedTask(
+      [this] { AudioDeviceBuffer::LogStats(AudioDeviceBuffer::LOG_ACTIVE); },
+      TimeDelta::Millis(time_to_wait_ms));
+}
+
+void AudioDeviceBuffer::ResetRecStats() {
+  RTC_DCHECK_RUN_ON(&task_queue_);
+  last_stats_.ResetRecStats();
+  MutexLock lock(&lock_);
+  stats_.ResetRecStats();
+}
+
+void AudioDeviceBuffer::ResetPlayStats() {
+  RTC_DCHECK_RUN_ON(&task_queue_);
+  last_stats_.ResetPlayStats();
+  MutexLock lock(&lock_);
+  stats_.ResetPlayStats();
+}
+
+void AudioDeviceBuffer::UpdateRecStats(int16_t max_abs,
+                                       size_t samples_per_channel) {
+  MutexLock lock(&lock_);
+  ++stats_.rec_callbacks;
+  stats_.rec_samples += samples_per_channel;
+  if (max_abs > stats_.max_rec_level) {
+    stats_.max_rec_level = max_abs;
+  }
+}
+
+void AudioDeviceBuffer::UpdatePlayStats(int16_t max_abs,
+                                        size_t samples_per_channel) {
+  MutexLock lock(&lock_);
+  ++stats_.play_callbacks;
+  stats_.play_samples += samples_per_channel;
+  if (max_abs > stats_.max_play_level) {
+    stats_.max_play_level = max_abs;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_buffer.h b/third_party/libwebrtc/modules/audio_device/audio_device_buffer.h
new file mode 100644
index 0000000000..eb681a7a68
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_buffer.h
@@ -0,0 +1,245 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
+#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <atomic>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/timestamp_aligner.h"
+
+namespace webrtc {
+
+// Delta times between two successive playout callbacks are limited to this
+// value before added to an internal array.
+const size_t kMaxDeltaTimeInMs = 500;
+// TODO(henrika): remove when no longer used by external client.
+const size_t kMaxBufferSizeBytes = 3840;  // 10ms in stereo @ 96kHz
+
+class AudioDeviceBuffer {
+ public:
+  enum LogState {
+    LOG_START = 0,
+    LOG_STOP,
+    LOG_ACTIVE,
+  };
+
+  struct Stats {
+    void ResetRecStats() {
+      rec_callbacks = 0;
+      rec_samples = 0;
+      max_rec_level = 0;
+    }
+
+    void ResetPlayStats() {
+      play_callbacks = 0;
+      play_samples = 0;
+      max_play_level = 0;
+    }
+
+    // Total number of recording callbacks where the source provides 10ms audio
+    // data each time.
+    uint64_t rec_callbacks = 0;
+
+    // Total number of playback callbacks where the sink asks for 10ms audio
+    // data each time.
+    uint64_t play_callbacks = 0;
+
+    // Total number of recorded audio samples.
+    uint64_t rec_samples = 0;
+
+    // Total number of played audio samples.
+    uint64_t play_samples = 0;
+
+    // Contains max level (max(abs(x))) of recorded audio packets over the last
+    // 10 seconds where a new measurement is done twice per second. The level
+    // is reset to zero at each call to LogStats().
+    int16_t max_rec_level = 0;
+
+    // Contains max level of recorded audio packets over the last 10 seconds
+    // where a new measurement is done twice per second.
+    int16_t max_play_level = 0;
+  };
+
+  explicit AudioDeviceBuffer(TaskQueueFactory* task_queue_factory);
+  virtual ~AudioDeviceBuffer();
+
+  int32_t RegisterAudioCallback(AudioTransport* audio_callback);
+
+  void StartPlayout();
+  void StartRecording();
+  void StopPlayout();
+  void StopRecording();
+
+  int32_t SetRecordingSampleRate(uint32_t fsHz);
+  int32_t SetPlayoutSampleRate(uint32_t fsHz);
+  uint32_t RecordingSampleRate() const;
+  uint32_t PlayoutSampleRate() const;
+
+  int32_t SetRecordingChannels(size_t channels);
+  int32_t SetPlayoutChannels(size_t channels);
+  size_t RecordingChannels() const;
+  size_t PlayoutChannels() const;
+
+  // TODO(bugs.webrtc.org/13621) Deprecate this function
+  virtual int32_t SetRecordedBuffer(const void* audio_buffer,
+                                    size_t samples_per_channel);
+
+  virtual int32_t SetRecordedBuffer(
+      const void* audio_buffer,
+      size_t samples_per_channel,
+      absl::optional<int64_t> capture_timestamp_ns);
+  virtual void SetVQEData(int play_delay_ms, int rec_delay_ms);
+  virtual int32_t DeliverRecordedData();
+  uint32_t NewMicLevel() const;
+
+  virtual int32_t RequestPlayoutData(size_t samples_per_channel);
+  virtual int32_t GetPlayoutData(void* audio_buffer);
+
+  int32_t SetTypingStatus(bool typing_status);
+
+ private:
+  // Starts/stops periodic logging of audio stats.
+  void StartPeriodicLogging();
+  void StopPeriodicLogging();
+
+  // Called periodically on the internal thread created by the TaskQueue.
+  // Updates some stats but dooes it on the task queue to ensure that access of
+  // members is serialized hence avoiding usage of locks.
+  // state = LOG_START => members are initialized and the timer starts.
+  // state = LOG_STOP => no logs are printed and the timer stops.
+  // state = LOG_ACTIVE => logs are printed and the timer is kept alive.
+  void LogStats(LogState state);
+
+  // Updates counters in each play/record callback. These counters are later
+  // (periodically) read by LogStats() using a lock.
+  void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
+  void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);
+
+  // Clears all members tracking stats for recording and playout.
+  // These methods both run on the task queue.
+  void ResetRecStats();
+  void ResetPlayStats();
+
+  // This object lives on the main (creating) thread and most methods are
+  // called on that same thread. When audio has started some methods will be
+  // called on either a native audio thread for playout or a native thread for
+  // recording. Some members are not annotated since they are "protected by
+  // design" and adding e.g. a race checker can cause failures for very few
+  // edge cases and it is IMHO not worth the risk to use them in this class.
+  // TODO(henrika): see if it is possible to refactor and annotate all members.
+
+  // Main thread on which this object is created.
+  SequenceChecker main_thread_checker_;
+
+  Mutex lock_;
+
+  // Task queue used to invoke LogStats() periodically. Tasks are executed on a
+  // worker thread but it does not necessarily have to be the same thread for
+  // each task.
+  rtc::TaskQueue task_queue_;
+
+  // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
+  // and it must outlive this object. It is not possible to change this member
+  // while any media is active. It is possible to start media without calling
+  // RegisterAudioCallback() but that will lead to ignored audio callbacks in
+  // both directions where native audio will be active but no audio samples will
+  // be transported.
+  AudioTransport* audio_transport_cb_;
+
+  // Sample rate in Hertz. Accessed atomically.
+  std::atomic<uint32_t> rec_sample_rate_;
+  std::atomic<uint32_t> play_sample_rate_;
+
+  // Number of audio channels. Accessed atomically.
+  std::atomic<size_t> rec_channels_;
+  std::atomic<size_t> play_channels_;
+
+  // Keeps track of if playout/recording are active or not. A combination
+  // of these states are used to determine when to start and stop the timer.
+  // Only used on the creating thread and not used to control any media flow.
+  bool playing_ RTC_GUARDED_BY(main_thread_checker_);
+  bool recording_ RTC_GUARDED_BY(main_thread_checker_);
+
+  // Buffer used for audio samples to be played out. Size can be changed
+  // dynamically. The 16-bit samples are interleaved, hence the size is
+  // proportional to the number of channels.
+  rtc::BufferT<int16_t> play_buffer_;
+
+  // Byte buffer used for recorded audio samples. Size can be changed
+  // dynamically.
+  rtc::BufferT<int16_t> rec_buffer_;
+
+  // Contains true of a key-press has been detected.
+  bool typing_status_;
+
+  // Delay values used by the AEC.
+  int play_delay_ms_;
+  int rec_delay_ms_;
+
+  // Capture timestamp.
+  absl::optional<int64_t> capture_timestamp_ns_;
+
+  // Counts number of times LogStats() has been called.
+  size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
+
+  // Time stamp of last timer task (drives logging).
+  int64_t last_timer_task_time_ RTC_GUARDED_BY(task_queue_);
+
+  // Counts number of audio callbacks modulo 50 to create a signal when
+  // a new storage of audio stats shall be done.
+  int16_t rec_stat_count_;
+  int16_t play_stat_count_;
+
+  // Time stamps of when playout and recording starts.
+  int64_t play_start_time_ RTC_GUARDED_BY(main_thread_checker_);
+  int64_t rec_start_time_ RTC_GUARDED_BY(main_thread_checker_);
+
+  // Contains counters for playout and recording statistics.
+  Stats stats_ RTC_GUARDED_BY(lock_);
+
+  // Stores current stats at each timer task. Used to calculate differences
+  // between two successive timer events.
+  Stats last_stats_ RTC_GUARDED_BY(task_queue_);
+
+  // Set to true at construction and modified to false as soon as one audio-
+  // level estimate larger than zero is detected.
+  bool only_silence_recorded_;
+
+  // Set to true when logging of audio stats is enabled for the first time in
+  // StartPeriodicLogging() and set to false by StopPeriodicLogging().
+  // Setting this member to false prevents (possiby invalid) log messages from
+  // being printed in the LogStats() task.
+  bool log_stats_ RTC_GUARDED_BY(task_queue_);
+
+  // Used for converting capture timestaps (received from AudioRecordThread
+  // via AudioRecordJni::DataIsRecorded) to RTC clock.
+  rtc::TimestampAligner timestamp_aligner_;
+
+// Should *never* be defined in production builds. Only used for testing.
+// When defined, the output signal will be replaced by a sinus tone at 440Hz.
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  double phase_;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_config.h b/third_party/libwebrtc/modules/audio_device/audio_device_config.h
new file mode 100644
index 0000000000..fa51747b67
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_config.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_
+
+// Enumerators
+//
+enum { GET_MIC_VOLUME_INTERVAL_MS = 1000 };
+
+// Platform specifics
+//
+#if defined(_WIN32)
+#if (_MSC_VER >= 1400)
+#if !defined(WEBRTC_DUMMY_FILE_DEVICES)
+// Windows Core Audio is the default audio layer in Windows.
+// Only supported for VS 2005 and higher.
+#define WEBRTC_WINDOWS_CORE_AUDIO_BUILD
+#endif
+#endif
+#endif
+
+#endif  // AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_data_observer.cc b/third_party/libwebrtc/modules/audio_device/audio_device_data_observer.cc
new file mode 100644
index 0000000000..0524830327
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_data_observer.cc
@@ -0,0 +1,373 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/include/audio_device_data_observer.h"
+
+#include "api/make_ref_counted.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// A wrapper over AudioDeviceModule that registers itself as AudioTransport
+// callback and redirects the PCM data to AudioDeviceDataObserver callback.
+class ADMWrapper : public AudioDeviceModule, public AudioTransport {
+ public:
+  ADMWrapper(rtc::scoped_refptr<AudioDeviceModule> impl,
+             AudioDeviceDataObserver* legacy_observer,
+             std::unique_ptr<AudioDeviceDataObserver> observer)
+      : impl_(impl),
+        legacy_observer_(legacy_observer),
+        observer_(std::move(observer)) {
+    is_valid_ = impl_.get() != nullptr;
+  }
+  ADMWrapper(AudioLayer audio_layer,
+             TaskQueueFactory* task_queue_factory,
+             AudioDeviceDataObserver* legacy_observer,
+             std::unique_ptr<AudioDeviceDataObserver> observer)
+      : ADMWrapper(AudioDeviceModule::Create(audio_layer, task_queue_factory),
+                   legacy_observer,
+                   std::move(observer)) {}
+  ~ADMWrapper() override {
+    audio_transport_ = nullptr;
+    observer_ = nullptr;
+  }
+
+  // Make sure we have a valid ADM before returning it to user.
+  bool IsValid() { return is_valid_; }
+
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  size_t nSamples,
+                                  size_t nBytesPerSample,
+                                  size_t nChannels,
+                                  uint32_t samples_per_sec,
+                                  uint32_t total_delay_ms,
+                                  int32_t clockDrift,
+                                  uint32_t currentMicLevel,
+                                  bool keyPressed,
+                                  uint32_t& newMicLevel) override {
+    return RecordedDataIsAvailable(
+        audioSamples, nSamples, nBytesPerSample, nChannels, samples_per_sec,
+        total_delay_ms, clockDrift, currentMicLevel, keyPressed, newMicLevel,
+        /*capture_timestamp_ns=*/absl::nullopt);
+  }
+
+  // AudioTransport methods overrides.
+  int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t nSamples,
+      size_t nBytesPerSample,
+      size_t nChannels,
+      uint32_t samples_per_sec,
+      uint32_t total_delay_ms,
+      int32_t clockDrift,
+      uint32_t currentMicLevel,
+      bool keyPressed,
+      uint32_t& newMicLevel,
+      absl::optional<int64_t> capture_timestamp_ns) override {
+    int32_t res = 0;
+    // Capture PCM data of locally captured audio.
+    if (observer_) {
+      observer_->OnCaptureData(audioSamples, nSamples, nBytesPerSample,
+                               nChannels, samples_per_sec);
+    }
+
+    // Send to the actual audio transport.
+    if (audio_transport_) {
+      res = audio_transport_->RecordedDataIsAvailable(
+          audioSamples, nSamples, nBytesPerSample, nChannels, samples_per_sec,
+          total_delay_ms, clockDrift, currentMicLevel, keyPressed, newMicLevel,
+          capture_timestamp_ns);
+    }
+
+    return res;
+  }
+
+  int32_t NeedMorePlayData(const size_t nSamples,
+                           const size_t nBytesPerSample,
+                           const size_t nChannels,
+                           const uint32_t samples_per_sec,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override {
+    int32_t res = 0;
+    // Set out parameters to safe values to be sure not to return corrupted
+    // data.
+    nSamplesOut = 0;
+    *elapsed_time_ms = -1;
+    *ntp_time_ms = -1;
+    // Request data from audio transport.
+    if (audio_transport_) {
+      res = audio_transport_->NeedMorePlayData(
+          nSamples, nBytesPerSample, nChannels, samples_per_sec, audioSamples,
+          nSamplesOut, elapsed_time_ms, ntp_time_ms);
+    }
+
+    // Capture rendered data.
+    if (observer_) {
+      observer_->OnRenderData(audioSamples, nSamples, nBytesPerSample,
+                              nChannels, samples_per_sec);
+    }
+
+    return res;
+  }
+
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override {
+    RTC_DCHECK_NOTREACHED();
+  }
+
+  // Override AudioDeviceModule's RegisterAudioCallback method to remember the
+  // actual audio transport (e.g.: voice engine).
+  int32_t RegisterAudioCallback(AudioTransport* audio_callback) override {
+    // Remember the audio callback to forward PCM data
+    audio_transport_ = audio_callback;
+    return 0;
+  }
+
+  // AudioDeviceModule pass through method overrides.
+  int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override {
+    return impl_->ActiveAudioLayer(audio_layer);
+  }
+  int32_t Init() override {
+    int res = impl_->Init();
+    if (res != 0) {
+      return res;
+    }
+    // Register self as the audio transport callback for underlying ADM impl.
+    impl_->RegisterAudioCallback(this);
+    return res;
+  }
+  int32_t Terminate() override { return impl_->Terminate(); }
+  bool Initialized() const override { return impl_->Initialized(); }
+  int16_t PlayoutDevices() override { return impl_->PlayoutDevices(); }
+  int16_t RecordingDevices() override { return impl_->RecordingDevices(); }
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override {
+    return impl_->PlayoutDeviceName(index, name, guid);
+  }
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override {
+    return impl_->RecordingDeviceName(index, name, guid);
+  }
+  int32_t SetPlayoutDevice(uint16_t index) override {
+    return impl_->SetPlayoutDevice(index);
+  }
+  int32_t SetPlayoutDevice(WindowsDeviceType device) override {
+    return impl_->SetPlayoutDevice(device);
+  }
+  int32_t SetRecordingDevice(uint16_t index) override {
+    return impl_->SetRecordingDevice(index);
+  }
+  int32_t SetRecordingDevice(WindowsDeviceType device) override {
+    return impl_->SetRecordingDevice(device);
+  }
+  int32_t PlayoutIsAvailable(bool* available) override {
+    return impl_->PlayoutIsAvailable(available);
+  }
+  int32_t InitPlayout() override { return impl_->InitPlayout(); }
+  bool PlayoutIsInitialized() const override {
+    return impl_->PlayoutIsInitialized();
+  }
+  int32_t RecordingIsAvailable(bool* available) override {
+    return impl_->RecordingIsAvailable(available);
+  }
+  int32_t InitRecording() override { return impl_->InitRecording(); }
+  bool RecordingIsInitialized() const override {
+    return impl_->RecordingIsInitialized();
+  }
+  int32_t StartPlayout() override { return impl_->StartPlayout(); }
+  int32_t StopPlayout() override { return impl_->StopPlayout(); }
+  bool Playing() const override { return impl_->Playing(); }
+  int32_t StartRecording() override { return impl_->StartRecording(); }
+  int32_t StopRecording() override { return impl_->StopRecording(); }
+  bool Recording() const override { return impl_->Recording(); }
+  int32_t InitSpeaker() override { return impl_->InitSpeaker(); }
+  bool SpeakerIsInitialized() const override {
+    return impl_->SpeakerIsInitialized();
+  }
+  int32_t InitMicrophone() override { return impl_->InitMicrophone(); }
+  bool MicrophoneIsInitialized() const override {
+    return impl_->MicrophoneIsInitialized();
+  }
+  int32_t SpeakerVolumeIsAvailable(bool* available) override {
+    return impl_->SpeakerVolumeIsAvailable(available);
+  }
+  int32_t SetSpeakerVolume(uint32_t volume) override {
+    return impl_->SetSpeakerVolume(volume);
+  }
+  int32_t SpeakerVolume(uint32_t* volume) const override {
+    return impl_->SpeakerVolume(volume);
+  }
+  int32_t MaxSpeakerVolume(uint32_t* max_volume) const override {
+    return impl_->MaxSpeakerVolume(max_volume);
+  }
+  int32_t MinSpeakerVolume(uint32_t* min_volume) const override {
+    return impl_->MinSpeakerVolume(min_volume);
+  }
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override {
+    return impl_->MicrophoneVolumeIsAvailable(available);
+  }
+  int32_t SetMicrophoneVolume(uint32_t volume) override {
+    return impl_->SetMicrophoneVolume(volume);
+  }
+  int32_t MicrophoneVolume(uint32_t* volume) const override {
+    return impl_->MicrophoneVolume(volume);
+  }
+  int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override {
+    return impl_->MaxMicrophoneVolume(max_volume);
+  }
+  int32_t MinMicrophoneVolume(uint32_t* min_volume) const override {
+    return impl_->MinMicrophoneVolume(min_volume);
+  }
+  int32_t SpeakerMuteIsAvailable(bool* available) override {
+    return impl_->SpeakerMuteIsAvailable(available);
+  }
+  int32_t SetSpeakerMute(bool enable) override {
+    return impl_->SetSpeakerMute(enable);
+  }
+  int32_t SpeakerMute(bool* enabled) const override {
+    return impl_->SpeakerMute(enabled);
+  }
+  int32_t MicrophoneMuteIsAvailable(bool* available) override {
+    return impl_->MicrophoneMuteIsAvailable(available);
+  }
+  int32_t SetMicrophoneMute(bool enable) override {
+    return impl_->SetMicrophoneMute(enable);
+  }
+  int32_t MicrophoneMute(bool* enabled) const override {
+    return impl_->MicrophoneMute(enabled);
+  }
+  int32_t StereoPlayoutIsAvailable(bool* available) const override {
+    return impl_->StereoPlayoutIsAvailable(available);
+  }
+  int32_t SetStereoPlayout(bool enable) override {
+    return impl_->SetStereoPlayout(enable);
+  }
+  int32_t StereoPlayout(bool* enabled) const override {
+    return impl_->StereoPlayout(enabled);
+  }
+  int32_t StereoRecordingIsAvailable(bool* available) const override {
+    return impl_->StereoRecordingIsAvailable(available);
+  }
+  int32_t SetStereoRecording(bool enable) override {
+    return impl_->SetStereoRecording(enable);
+  }
+  int32_t StereoRecording(bool* enabled) const override {
+    return impl_->StereoRecording(enabled);
+  }
+  int32_t PlayoutDelay(uint16_t* delay_ms) const override {
+    return impl_->PlayoutDelay(delay_ms);
+  }
+  bool BuiltInAECIsAvailable() const override {
+    return impl_->BuiltInAECIsAvailable();
+  }
+  bool BuiltInAGCIsAvailable() const override {
+    return impl_->BuiltInAGCIsAvailable();
+  }
+  bool BuiltInNSIsAvailable() const override {
+    return impl_->BuiltInNSIsAvailable();
+  }
+  int32_t EnableBuiltInAEC(bool enable) override {
+    return impl_->EnableBuiltInAEC(enable);
+  }
+  int32_t EnableBuiltInAGC(bool enable) override {
+    return impl_->EnableBuiltInAGC(enable);
+  }
+  int32_t EnableBuiltInNS(bool enable) override {
+    return impl_->EnableBuiltInNS(enable);
+  }
+  int32_t GetPlayoutUnderrunCount() const override {
+    return impl_->GetPlayoutUnderrunCount();
+  }
+// Only supported on iOS.
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override {
+    return impl_->GetPlayoutAudioParameters(params);
+  }
+  int GetRecordAudioParameters(AudioParameters* params) const override {
+    return impl_->GetRecordAudioParameters(params);
+  }
+#endif  // WEBRTC_IOS
+
+ protected:
+  rtc::scoped_refptr<AudioDeviceModule> impl_;
+  AudioDeviceDataObserver* legacy_observer_ = nullptr;
+  std::unique_ptr<AudioDeviceDataObserver> observer_;
+  AudioTransport* audio_transport_ = nullptr;
+  bool is_valid_ = false;
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    rtc::scoped_refptr<AudioDeviceModule> impl,
+    std::unique_ptr<AudioDeviceDataObserver> observer) {
+  auto audio_device = rtc::make_ref_counted<ADMWrapper>(impl, observer.get(),
+                                                        std::move(observer));
+
+  if (!audio_device->IsValid()) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    rtc::scoped_refptr<AudioDeviceModule> impl,
+    AudioDeviceDataObserver* legacy_observer) {
+  auto audio_device =
+      rtc::make_ref_counted<ADMWrapper>(impl, legacy_observer, nullptr);
+
+  if (!audio_device->IsValid()) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    AudioDeviceModule::AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory,
+    std::unique_ptr<AudioDeviceDataObserver> observer) {
+  auto audio_device = rtc::make_ref_counted<ADMWrapper>(
+      audio_layer, task_queue_factory, observer.get(), std::move(observer));
+
+  if (!audio_device->IsValid()) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    AudioDeviceModule::AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory,
+    AudioDeviceDataObserver* legacy_observer) {
+  auto audio_device = rtc::make_ref_counted<ADMWrapper>(
+      audio_layer, task_queue_factory, legacy_observer, nullptr);
+
+  if (!audio_device->IsValid()) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_generic.cc b/third_party/libwebrtc/modules/audio_device/audio_device_generic.cc
new file mode 100644
index 0000000000..7b8cfd1734
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_generic.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/audio_device_generic.h"
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+bool AudioDeviceGeneric::BuiltInAECIsAvailable() const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return false;
+}
+
+int32_t AudioDeviceGeneric::EnableBuiltInAEC(bool enable) {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+bool AudioDeviceGeneric::BuiltInAGCIsAvailable() const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return false;
+}
+
+int32_t AudioDeviceGeneric::EnableBuiltInAGC(bool enable) {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+bool AudioDeviceGeneric::BuiltInNSIsAvailable() const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return false;
+}
+
+int32_t AudioDeviceGeneric::EnableBuiltInNS(bool enable) {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+int32_t AudioDeviceGeneric::GetPlayoutUnderrunCount() const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+#if defined(WEBRTC_IOS)
+int AudioDeviceGeneric::GetPlayoutAudioParameters(
+    AudioParameters* params) const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+int AudioDeviceGeneric::GetRecordAudioParameters(
+    AudioParameters* params) const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+#endif  // WEBRTC_IOS
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_generic.h b/third_party/libwebrtc/modules/audio_device/audio_device_generic.h
new file mode 100644
index 0000000000..41e24eb3b0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_generic.h
@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H_
+
+#include <stdint.h>
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioDeviceGeneric {
+ public:
+  // For use with UMA logging. Must be kept in sync with histograms.xml in
+  // Chrome, located at
+  // https://cs.chromium.org/chromium/src/tools/metrics/histograms/histograms.xml
+  enum class InitStatus {
+    OK = 0,
+    PLAYOUT_ERROR = 1,
+    RECORDING_ERROR = 2,
+    OTHER_ERROR = 3,
+    NUM_STATUSES = 4
+  };
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const = 0;
+
+  // Main initializaton and termination
+  virtual InitStatus Init() = 0;
+  virtual int32_t Terminate() = 0;
+  virtual bool Initialized() const = 0;
+
+  // Device enumeration
+  virtual int16_t PlayoutDevices() = 0;
+  virtual int16_t RecordingDevices() = 0;
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]) = 0;
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]) = 0;
+
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
+  virtual int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) = 0;
+  virtual int32_t SetRecordingDevice(uint16_t index) = 0;
+  virtual int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) = 0;
+
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool& available) = 0;
+  virtual int32_t InitPlayout() = 0;
+  virtual bool PlayoutIsInitialized() const = 0;
+  virtual int32_t RecordingIsAvailable(bool& available) = 0;
+  virtual int32_t InitRecording() = 0;
+  virtual bool RecordingIsInitialized() const = 0;
+
+  // Audio transport control
+  virtual int32_t StartPlayout() = 0;
+  virtual int32_t StopPlayout() = 0;
+  virtual bool Playing() const = 0;
+  virtual int32_t StartRecording() = 0;
+  virtual int32_t StopRecording() = 0;
+  virtual bool Recording() const = 0;
+
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker() = 0;
+  virtual bool SpeakerIsInitialized() const = 0;
+  virtual int32_t InitMicrophone() = 0;
+  virtual bool MicrophoneIsInitialized() const = 0;
+
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool& available) = 0;
+  virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
+  virtual int32_t SpeakerVolume(uint32_t& volume) const = 0;
+  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const = 0;
+  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const = 0;
+
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool& available) = 0;
+  virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
+  virtual int32_t MicrophoneVolume(uint32_t& volume) const = 0;
+  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const = 0;
+  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const = 0;
+
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool& available) = 0;
+  virtual int32_t SetSpeakerMute(bool enable) = 0;
+  virtual int32_t SpeakerMute(bool& enabled) const = 0;
+
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool& available) = 0;
+  virtual int32_t SetMicrophoneMute(bool enable) = 0;
+  virtual int32_t MicrophoneMute(bool& enabled) const = 0;
+
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool& available) = 0;
+  virtual int32_t SetStereoPlayout(bool enable) = 0;
+  virtual int32_t StereoPlayout(bool& enabled) const = 0;
+  virtual int32_t StereoRecordingIsAvailable(bool& available) = 0;
+  virtual int32_t SetStereoRecording(bool enable) = 0;
+  virtual int32_t StereoRecording(bool& enabled) const = 0;
+
+  // Delay information and control
+  virtual int32_t PlayoutDelay(uint16_t& delayMS) const = 0;
+
+  // Android only
+  virtual bool BuiltInAECIsAvailable() const;
+  virtual bool BuiltInAGCIsAvailable() const;
+  virtual bool BuiltInNSIsAvailable() const;
+
+  // Windows Core Audio and Android only.
+  virtual int32_t EnableBuiltInAEC(bool enable);
+  virtual int32_t EnableBuiltInAGC(bool enable);
+  virtual int32_t EnableBuiltInNS(bool enable);
+
+  // Play underrun count.
+  virtual int32_t GetPlayoutUnderrunCount() const;
+
+// iOS only.
+// TODO(henrika): add Android support.
+#if defined(WEBRTC_IOS)
+  virtual int GetPlayoutAudioParameters(AudioParameters* params) const;
+  virtual int GetRecordAudioParameters(AudioParameters* params) const;
+#endif  // WEBRTC_IOS
+
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
+
+  virtual ~AudioDeviceGeneric() {}
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H_
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_gn/moz.build b/third_party/libwebrtc/modules/audio_device/audio_device_gn/moz.build
new file mode 100644
index 0000000000..40ed29f258
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_device_gn")
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_impl.cc b/third_party/libwebrtc/modules/audio_device/audio_device_impl.cc
new file mode 100644
index 0000000000..092b98f2bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_impl.cc
@@ -0,0 +1,951 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/audio_device_impl.h"
+
+#include <stddef.h>
+
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+#include "modules/audio_device/audio_device_config.h"  // IWYU pragma: keep
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+#if defined(_WIN32)
+#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+#include "modules/audio_device/win/audio_device_core_win.h"
+#endif
+#elif defined(WEBRTC_ANDROID)
+#include <stdlib.h>
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+#include "modules/audio_device/android/aaudio_player.h"
+#include "modules/audio_device/android/aaudio_recorder.h"
+#endif
+#include "modules/audio_device/android/audio_device_template.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/android/audio_record_jni.h"
+#include "modules/audio_device/android/audio_track_jni.h"
+#include "modules/audio_device/android/opensles_player.h"
+#include "modules/audio_device/android/opensles_recorder.h"
+#elif defined(WEBRTC_LINUX)
+#if defined(WEBRTC_ENABLE_LINUX_ALSA)
+#include "modules/audio_device/linux/audio_device_alsa_linux.h"
+#endif
+#if defined(WEBRTC_ENABLE_LINUX_PULSE)
+#include "modules/audio_device/linux/audio_device_pulse_linux.h"
+#endif
+#elif defined(WEBRTC_IOS)
+#include "sdk/objc/native/src/audio/audio_device_ios.h"
+#elif defined(WEBRTC_MAC)
+#include "modules/audio_device/mac/audio_device_mac.h"
+#endif
+#if defined(WEBRTC_DUMMY_FILE_DEVICES)
+#include "modules/audio_device/dummy/file_audio_device.h"
+#include "modules/audio_device/dummy/file_audio_device_factory.h"
+#endif
+#include "modules/audio_device/dummy/audio_device_dummy.h"
+
+#define CHECKinitialized_() \
+  {                         \
+    if (!initialized_) {    \
+      return -1;            \
+    }                       \
+  }
+
+#define CHECKinitialized__BOOL() \
+  {                              \
+    if (!initialized_) {         \
+      return false;              \
+    }                            \
+  }
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
+    AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  return AudioDeviceModule::CreateForTest(audio_layer, task_queue_factory);
+}
+
+// static
+rtc::scoped_refptr<AudioDeviceModuleForTest> AudioDeviceModule::CreateForTest(
+    AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  // The "AudioDeviceModule::kWindowsCoreAudio2" audio layer has its own
+  // dedicated factory method which should be used instead.
+  if (audio_layer == AudioDeviceModule::kWindowsCoreAudio2) {
+    RTC_LOG(LS_ERROR) << "Use the CreateWindowsCoreAudioAudioDeviceModule() "
+                         "factory method instead for this option.";
+    return nullptr;
+  }
+
+  // Create the generic reference counted (platform independent) implementation.
+  auto audio_device = rtc::make_ref_counted<AudioDeviceModuleImpl>(
+      audio_layer, task_queue_factory);
+
+  // Ensure that the current platform is supported.
+  if (audio_device->CheckPlatform() == -1) {
+    return nullptr;
+  }
+
+  // Create the platform-dependent implementation.
+  if (audio_device->CreatePlatformSpecificObjects() == -1) {
+    return nullptr;
+  }
+
+  // Ensure that the generic audio buffer can communicate with the platform
+  // specific parts.
+  if (audio_device->AttachAudioBuffer() == -1) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+
+AudioDeviceModuleImpl::AudioDeviceModuleImpl(
+    AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory)
+    : audio_layer_(audio_layer), audio_device_buffer_(task_queue_factory) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+}
+
+int32_t AudioDeviceModuleImpl::CheckPlatform() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  // Ensure that the current platform is supported
+  PlatformType platform(kPlatformNotSupported);
+#if defined(_WIN32)
+  platform = kPlatformWin32;
+  RTC_LOG(LS_INFO) << "current platform is Win32";
+#elif defined(WEBRTC_ANDROID)
+  platform = kPlatformAndroid;
+  RTC_LOG(LS_INFO) << "current platform is Android";
+#elif defined(WEBRTC_LINUX)
+  platform = kPlatformLinux;
+  RTC_LOG(LS_INFO) << "current platform is Linux";
+#elif defined(WEBRTC_IOS)
+  platform = kPlatformIOS;
+  RTC_LOG(LS_INFO) << "current platform is IOS";
+#elif defined(WEBRTC_MAC)
+  platform = kPlatformMac;
+  RTC_LOG(LS_INFO) << "current platform is Mac";
+#endif
+  if (platform == kPlatformNotSupported) {
+    RTC_LOG(LS_ERROR)
+        << "current platform is not supported => this module will self "
+           "destruct!";
+    return -1;
+  }
+  platform_type_ = platform;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+// Dummy ADM implementations if build flags are set.
+#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
+  audio_device_.reset(new AudioDeviceDummy());
+  RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized";
+#elif defined(WEBRTC_DUMMY_FILE_DEVICES)
+  audio_device_.reset(FileAudioDeviceFactory::CreateFileAudioDevice());
+  if (audio_device_) {
+    RTC_LOG(LS_INFO) << "Will use file-playing dummy device.";
+  } else {
+    // Create a dummy device instead.
+    audio_device_.reset(new AudioDeviceDummy());
+    RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized";
+  }
+
+// Real (non-dummy) ADM implementations.
+#else
+  AudioLayer audio_layer(PlatformAudioLayer());
+// Windows ADM implementation.
+#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+  if ((audio_layer == kWindowsCoreAudio) ||
+      (audio_layer == kPlatformDefaultAudio)) {
+    RTC_LOG(LS_INFO) << "Attempting to use the Windows Core Audio APIs...";
+    if (AudioDeviceWindowsCore::CoreAudioIsSupported()) {
+      audio_device_.reset(new AudioDeviceWindowsCore());
+      RTC_LOG(LS_INFO) << "Windows Core Audio APIs will be utilized";
+    }
+  }
+#endif  // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+
+#if defined(WEBRTC_ANDROID)
+  // Create an Android audio manager.
+  audio_manager_android_.reset(new AudioManager());
+  // Select best possible combination of audio layers.
+  if (audio_layer == kPlatformDefaultAudio) {
+    if (audio_manager_android_->IsAAudioSupported()) {
+      // Use of AAudio for both playout and recording has highest priority.
+      audio_layer = kAndroidAAudioAudio;
+    } else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
+               audio_manager_android_->IsLowLatencyRecordSupported()) {
+      // Use OpenSL ES for both playout and recording.
+      audio_layer = kAndroidOpenSLESAudio;
+    } else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
+               !audio_manager_android_->IsLowLatencyRecordSupported()) {
+      // Use OpenSL ES for output on devices that only supports the
+      // low-latency output audio path.
+      audio_layer = kAndroidJavaInputAndOpenSLESOutputAudio;
+    } else {
+      // Use Java-based audio in both directions when low-latency output is
+      // not supported.
+      audio_layer = kAndroidJavaAudio;
+    }
+  }
+  AudioManager* audio_manager = audio_manager_android_.get();
+  if (audio_layer == kAndroidJavaAudio) {
+    // Java audio for both input and output audio.
+    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>(
+        audio_layer, audio_manager));
+  } else if (audio_layer == kAndroidOpenSLESAudio) {
+    // OpenSL ES based audio for both input and output audio.
+    audio_device_.reset(
+        new AudioDeviceTemplate<OpenSLESRecorder, OpenSLESPlayer>(
+            audio_layer, audio_manager));
+  } else if (audio_layer == kAndroidJavaInputAndOpenSLESOutputAudio) {
+    // Java audio for input and OpenSL ES for output audio (i.e. mixed APIs).
+    // This combination provides low-latency output audio and at the same
+    // time support for HW AEC using the AudioRecord Java API.
+    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, OpenSLESPlayer>(
+        audio_layer, audio_manager));
+  } else if (audio_layer == kAndroidAAudioAudio) {
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+    // AAudio based audio for both input and output.
+    audio_device_.reset(new AudioDeviceTemplate<AAudioRecorder, AAudioPlayer>(
+        audio_layer, audio_manager));
+#endif
+  } else if (audio_layer == kAndroidJavaInputAndAAudioOutputAudio) {
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+    // Java audio for input and AAudio for output audio (i.e. mixed APIs).
+    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AAudioPlayer>(
+        audio_layer, audio_manager));
+#endif
+  } else {
+    RTC_LOG(LS_ERROR) << "The requested audio layer is not supported";
+    audio_device_.reset(nullptr);
+  }
+// END #if defined(WEBRTC_ANDROID)
+
+// Linux ADM implementation.
+// Note that, WEBRTC_ENABLE_LINUX_ALSA is always defined by default when
+// WEBRTC_LINUX is defined. WEBRTC_ENABLE_LINUX_PULSE depends on the
+// 'rtc_include_pulse_audio' build flag.
+// TODO(bugs.webrtc.org/9127): improve support and make it more clear that
+// PulseAudio is the default selection.
+#elif defined(WEBRTC_LINUX)
+#if !defined(WEBRTC_ENABLE_LINUX_PULSE)
+  // Build flag 'rtc_include_pulse_audio' is set to false. In this mode:
+  // - kPlatformDefaultAudio => ALSA, and
+  // - kLinuxAlsaAudio => ALSA, and
+  // - kLinuxPulseAudio => Invalid selection.
+  RTC_LOG(LS_WARNING) << "PulseAudio is disabled using build flag.";
+  if ((audio_layer == kLinuxAlsaAudio) ||
+      (audio_layer == kPlatformDefaultAudio)) {
+    audio_device_.reset(new AudioDeviceLinuxALSA());
+    RTC_LOG(LS_INFO) << "Linux ALSA APIs will be utilized.";
+  }
+#else
+  // Build flag 'rtc_include_pulse_audio' is set to true (default). In this
+  // mode:
+  // - kPlatformDefaultAudio => PulseAudio, and
+  // - kLinuxPulseAudio => PulseAudio, and
+  // - kLinuxAlsaAudio => ALSA (supported but not default).
+  RTC_LOG(LS_INFO) << "PulseAudio support is enabled.";
+  if ((audio_layer == kLinuxPulseAudio) ||
+      (audio_layer == kPlatformDefaultAudio)) {
+    // Linux PulseAudio implementation is default.
+    audio_device_.reset(new AudioDeviceLinuxPulse());
+    RTC_LOG(LS_INFO) << "Linux PulseAudio APIs will be utilized";
+  } else if (audio_layer == kLinuxAlsaAudio) {
+    audio_device_.reset(new AudioDeviceLinuxALSA());
+    RTC_LOG(LS_WARNING) << "Linux ALSA APIs will be utilized.";
+  }
+#endif  // #if !defined(WEBRTC_ENABLE_LINUX_PULSE)
+#endif  // #if defined(WEBRTC_LINUX)
+
+// iOS ADM implementation.
+#if defined(WEBRTC_IOS)
+  if (audio_layer == kPlatformDefaultAudio) {
+    audio_device_.reset(
+        new ios_adm::AudioDeviceIOS(/*bypass_voice_processing=*/false));
+    RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
+  }
+// END #if defined(WEBRTC_IOS)
+
+// Mac OS X ADM implementation.
+#elif defined(WEBRTC_MAC)
+  if (audio_layer == kPlatformDefaultAudio) {
+    audio_device_.reset(new AudioDeviceMac());
+    RTC_LOG(LS_INFO) << "Mac OS X Audio APIs will be utilized.";
+  }
+#endif  // WEBRTC_MAC
+
+  // Dummy ADM implementation.
+  if (audio_layer == kDummyAudio) {
+    audio_device_.reset(new AudioDeviceDummy());
+    RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized.";
+  }
+#endif  // if defined(WEBRTC_DUMMY_AUDIO_BUILD)
+
+  if (!audio_device_) {
+    RTC_LOG(LS_ERROR)
+        << "Failed to create the platform specific ADM implementation.";
+    return -1;
+  }
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::AttachAudioBuffer() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  audio_device_->AttachAudioBuffer(&audio_device_buffer_);
+  return 0;
+}
+
+AudioDeviceModuleImpl::~AudioDeviceModuleImpl() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+}
+
+int32_t AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  AudioLayer activeAudio;
+  if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
+    return -1;
+  }
+  *audioLayer = activeAudio;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::Init() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  if (initialized_)
+    return 0;
+  RTC_CHECK(audio_device_);
+  AudioDeviceGeneric::InitStatus status = audio_device_->Init();
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.Audio.InitializationResult", static_cast<int>(status),
+      static_cast<int>(AudioDeviceGeneric::InitStatus::NUM_STATUSES));
+  if (status != AudioDeviceGeneric::InitStatus::OK) {
+    RTC_LOG(LS_ERROR) << "Audio device initialization failed.";
+    return -1;
+  }
+  initialized_ = true;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::Terminate() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  if (!initialized_)
+    return 0;
+  if (audio_device_->Terminate() == -1) {
+    return -1;
+  }
+  initialized_ = false;
+  return 0;
+}
+
+bool AudioDeviceModuleImpl::Initialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << ": " << initialized_;
+  return initialized_;
+}
+
+int32_t AudioDeviceModuleImpl::InitSpeaker() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  return audio_device_->InitSpeaker();
+}
+
+int32_t AudioDeviceModuleImpl::InitMicrophone() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  return audio_device_->InitMicrophone();
+}
+
+int32_t AudioDeviceModuleImpl::SpeakerVolumeIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetSpeakerVolume(uint32_t volume) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
+  CHECKinitialized_();
+  return audio_device_->SetSpeakerVolume(volume);
+}
+
+int32_t AudioDeviceModuleImpl::SpeakerVolume(uint32_t* volume) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  uint32_t level = 0;
+  if (audio_device_->SpeakerVolume(level) == -1) {
+    return -1;
+  }
+  *volume = level;
+  RTC_LOG(LS_INFO) << "output: " << *volume;
+  return 0;
+}
+
+bool AudioDeviceModuleImpl::SpeakerIsInitialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isInitialized = audio_device_->SpeakerIsInitialized();
+  RTC_LOG(LS_INFO) << "output: " << isInitialized;
+  return isInitialized;
+}
+
+bool AudioDeviceModuleImpl::MicrophoneIsInitialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isInitialized = audio_device_->MicrophoneIsInitialized();
+  RTC_LOG(LS_INFO) << "output: " << isInitialized;
+  return isInitialized;
+}
+
+int32_t AudioDeviceModuleImpl::MaxSpeakerVolume(uint32_t* maxVolume) const {
+  CHECKinitialized_();
+  uint32_t maxVol = 0;
+  if (audio_device_->MaxSpeakerVolume(maxVol) == -1) {
+    return -1;
+  }
+  *maxVolume = maxVol;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MinSpeakerVolume(uint32_t* minVolume) const {
+  CHECKinitialized_();
+  uint32_t minVol = 0;
+  if (audio_device_->MinSpeakerVolume(minVol) == -1) {
+    return -1;
+  }
+  *minVolume = minVol;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SpeakerMuteIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetSpeakerMute(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  return audio_device_->SetSpeakerMute(enable);
+}
+
+int32_t AudioDeviceModuleImpl::SpeakerMute(bool* enabled) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool muted = false;
+  if (audio_device_->SpeakerMute(muted) == -1) {
+    return -1;
+  }
+  *enabled = muted;
+  RTC_LOG(LS_INFO) << "output: " << muted;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MicrophoneMuteIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetMicrophoneMute(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  return (audio_device_->SetMicrophoneMute(enable));
+}
+
+int32_t AudioDeviceModuleImpl::MicrophoneMute(bool* enabled) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool muted = false;
+  if (audio_device_->MicrophoneMute(muted) == -1) {
+    return -1;
+  }
+  *enabled = muted;
+  RTC_LOG(LS_INFO) << "output: " << muted;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MicrophoneVolumeIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
+  CHECKinitialized_();
+  return (audio_device_->SetMicrophoneVolume(volume));
+}
+
+int32_t AudioDeviceModuleImpl::MicrophoneVolume(uint32_t* volume) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  uint32_t level = 0;
+  if (audio_device_->MicrophoneVolume(level) == -1) {
+    return -1;
+  }
+  *volume = level;
+  RTC_LOG(LS_INFO) << "output: " << *volume;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::StereoRecordingIsAvailable(
+    bool* available) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetStereoRecording(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  if (audio_device_->RecordingIsInitialized()) {
+    RTC_LOG(LS_ERROR)
+        << "unable to set stereo mode after recording is initialized";
+    return -1;
+  }
+  if (audio_device_->SetStereoRecording(enable) == -1) {
+    if (enable) {
+      RTC_LOG(LS_WARNING) << "failed to enable stereo recording";
+    }
+    return -1;
+  }
+  int8_t nChannels(1);
+  if (enable) {
+    nChannels = 2;
+  }
+  audio_device_buffer_.SetRecordingChannels(nChannels);
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::StereoRecording(bool* enabled) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool stereo = false;
+  if (audio_device_->StereoRecording(stereo) == -1) {
+    return -1;
+  }
+  *enabled = stereo;
+  RTC_LOG(LS_INFO) << "output: " << stereo;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::StereoPlayoutIsAvailable(bool* available) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetStereoPlayout(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  if (audio_device_->PlayoutIsInitialized()) {
+    RTC_LOG(LS_ERROR)
+        << "unable to set stereo mode while playing side is initialized";
+    return -1;
+  }
+  if (audio_device_->SetStereoPlayout(enable)) {
+    RTC_LOG(LS_WARNING) << "stereo playout is not supported";
+    return -1;
+  }
+  int8_t nChannels(1);
+  if (enable) {
+    nChannels = 2;
+  }
+  audio_device_buffer_.SetPlayoutChannels(nChannels);
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::StereoPlayout(bool* enabled) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool stereo = false;
+  if (audio_device_->StereoPlayout(stereo) == -1) {
+    return -1;
+  }
+  *enabled = stereo;
+  RTC_LOG(LS_INFO) << "output: " << stereo;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::PlayoutIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::RecordingIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MaxMicrophoneVolume(uint32_t* maxVolume) const {
+  CHECKinitialized_();
+  uint32_t maxVol(0);
+  if (audio_device_->MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
+  *maxVolume = maxVol;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MinMicrophoneVolume(uint32_t* minVolume) const {
+  CHECKinitialized_();
+  uint32_t minVol(0);
+  if (audio_device_->MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
+  *minVolume = minVol;
+  return 0;
+}
+
+int16_t AudioDeviceModuleImpl::PlayoutDevices() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
+  RTC_LOG(LS_INFO) << "output: " << nPlayoutDevices;
+  return (int16_t)(nPlayoutDevices);
+}
+
+int32_t AudioDeviceModuleImpl::SetPlayoutDevice(uint16_t index) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
+  CHECKinitialized_();
+  return audio_device_->SetPlayoutDevice(index);
+}
+
+int32_t AudioDeviceModuleImpl::SetPlayoutDevice(WindowsDeviceType device) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  return audio_device_->SetPlayoutDevice(device);
+}
+
+int32_t AudioDeviceModuleImpl::PlayoutDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
+  CHECKinitialized_();
+  if (name == NULL) {
+    return -1;
+  }
+  if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) {
+    return -1;
+  }
+  if (name != NULL) {
+    RTC_LOG(LS_INFO) << "output: name = " << name;
+  }
+  if (guid != NULL) {
+    RTC_LOG(LS_INFO) << "output: guid = " << guid;
+  }
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::RecordingDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
+  CHECKinitialized_();
+  if (name == NULL) {
+    return -1;
+  }
+  if (audio_device_->RecordingDeviceName(index, name, guid) == -1) {
+    return -1;
+  }
+  if (name != NULL) {
+    RTC_LOG(LS_INFO) << "output: name = " << name;
+  }
+  if (guid != NULL) {
+    RTC_LOG(LS_INFO) << "output: guid = " << guid;
+  }
+  return 0;
+}
+
+int16_t AudioDeviceModuleImpl::RecordingDevices() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  uint16_t nRecordingDevices = audio_device_->RecordingDevices();
+  RTC_LOG(LS_INFO) << "output: " << nRecordingDevices;
+  return (int16_t)nRecordingDevices;
+}
+
+int32_t AudioDeviceModuleImpl::SetRecordingDevice(uint16_t index) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
+  CHECKinitialized_();
+  return audio_device_->SetRecordingDevice(index);
+}
+
+int32_t AudioDeviceModuleImpl::SetRecordingDevice(WindowsDeviceType device) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  return audio_device_->SetRecordingDevice(device);
+}
+
+int32_t AudioDeviceModuleImpl::InitPlayout() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  if (PlayoutIsInitialized()) {
+    return 0;
+  }
+  int32_t result = audio_device_->InitPlayout();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+int32_t AudioDeviceModuleImpl::InitRecording() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  if (RecordingIsInitialized()) {
+    return 0;
+  }
+  int32_t result = audio_device_->InitRecording();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+bool AudioDeviceModuleImpl::PlayoutIsInitialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  return audio_device_->PlayoutIsInitialized();
+}
+
+bool AudioDeviceModuleImpl::RecordingIsInitialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  return audio_device_->RecordingIsInitialized();
+}
+
+int32_t AudioDeviceModuleImpl::StartPlayout() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  if (Playing()) {
+    return 0;
+  }
+  audio_device_buffer_.StartPlayout();
+  int32_t result = audio_device_->StartPlayout();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+int32_t AudioDeviceModuleImpl::StopPlayout() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  int32_t result = audio_device_->StopPlayout();
+  audio_device_buffer_.StopPlayout();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+bool AudioDeviceModuleImpl::Playing() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  return audio_device_->Playing();
+}
+
+int32_t AudioDeviceModuleImpl::StartRecording() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  if (Recording()) {
+    return 0;
+  }
+  audio_device_buffer_.StartRecording();
+  int32_t result = audio_device_->StartRecording();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+int32_t AudioDeviceModuleImpl::StopRecording() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  int32_t result = audio_device_->StopRecording();
+  audio_device_buffer_.StopRecording();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+bool AudioDeviceModuleImpl::Recording() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  return audio_device_->Recording();
+}
+
+int32_t AudioDeviceModuleImpl::RegisterAudioCallback(
+    AudioTransport* audioCallback) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  return audio_device_buffer_.RegisterAudioCallback(audioCallback);
+}
+
+int32_t AudioDeviceModuleImpl::PlayoutDelay(uint16_t* delayMS) const {
+  CHECKinitialized_();
+  uint16_t delay = 0;
+  if (audio_device_->PlayoutDelay(delay) == -1) {
+    RTC_LOG(LS_ERROR) << "failed to retrieve the playout delay";
+    return -1;
+  }
+  *delayMS = delay;
+  return 0;
+}
+
+bool AudioDeviceModuleImpl::BuiltInAECIsAvailable() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isAvailable = audio_device_->BuiltInAECIsAvailable();
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return isAvailable;
+}
+
+int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  int32_t ok = audio_device_->EnableBuiltInAEC(enable);
+  RTC_LOG(LS_INFO) << "output: " << ok;
+  return ok;
+}
+
+bool AudioDeviceModuleImpl::BuiltInAGCIsAvailable() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return isAvailable;
+}
+
+int32_t AudioDeviceModuleImpl::EnableBuiltInAGC(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  int32_t ok = audio_device_->EnableBuiltInAGC(enable);
+  RTC_LOG(LS_INFO) << "output: " << ok;
+  return ok;
+}
+
+bool AudioDeviceModuleImpl::BuiltInNSIsAvailable() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isAvailable = audio_device_->BuiltInNSIsAvailable();
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return isAvailable;
+}
+
+int32_t AudioDeviceModuleImpl::EnableBuiltInNS(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  int32_t ok = audio_device_->EnableBuiltInNS(enable);
+  RTC_LOG(LS_INFO) << "output: " << ok;
+  return ok;
+}
+
+int32_t AudioDeviceModuleImpl::GetPlayoutUnderrunCount() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  int32_t underrunCount = audio_device_->GetPlayoutUnderrunCount();
+  RTC_LOG(LS_INFO) << "output: " << underrunCount;
+  return underrunCount;
+}
+
+#if defined(WEBRTC_IOS)
+int AudioDeviceModuleImpl::GetPlayoutAudioParameters(
+    AudioParameters* params) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  int r = audio_device_->GetPlayoutAudioParameters(params);
+  RTC_LOG(LS_INFO) << "output: " << r;
+  return r;
+}
+
+int AudioDeviceModuleImpl::GetRecordAudioParameters(
+    AudioParameters* params) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  int r = audio_device_->GetRecordAudioParameters(params);
+  RTC_LOG(LS_INFO) << "output: " << r;
+  return r;
+}
+#endif  // WEBRTC_IOS
+
+AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  return platform_type_;
+}
+
+AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer()
+    const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  return audio_layer_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_impl.h b/third_party/libwebrtc/modules/audio_device/audio_device_impl.h
new file mode 100644
index 0000000000..45f73dcd65
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_impl.h
@@ -0,0 +1,180 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_
+#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_
+
+#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+
+class AudioDeviceGeneric;
+class AudioManager;
+
+class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {
+ public:
+  enum PlatformType {
+    kPlatformNotSupported = 0,
+    kPlatformWin32 = 1,
+    kPlatformWinCe = 2,
+    kPlatformLinux = 3,
+    kPlatformMac = 4,
+    kPlatformAndroid = 5,
+    kPlatformIOS = 6
+  };
+
+  int32_t CheckPlatform();
+  int32_t CreatePlatformSpecificObjects();
+  int32_t AttachAudioBuffer();
+
+  AudioDeviceModuleImpl(AudioLayer audio_layer,
+                        TaskQueueFactory* task_queue_factory);
+  ~AudioDeviceModuleImpl() override;
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
+
+  // Full-duplex transportation of PCM audio
+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
+
+  // Main initializaton and termination
+  int32_t Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool* available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool* available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool* available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t* volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t* volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool* available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool* enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool* available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool* enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool* available) const override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool* enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool* available) const override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool* enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t* delayMS) const override;
+
+  bool BuiltInAECIsAvailable() const override;
+  int32_t EnableBuiltInAEC(bool enable) override;
+  bool BuiltInAGCIsAvailable() const override;
+  int32_t EnableBuiltInAGC(bool enable) override;
+  bool BuiltInNSIsAvailable() const override;
+  int32_t EnableBuiltInNS(bool enable) override;
+
+  // Play underrun count.
+  int32_t GetPlayoutUnderrunCount() const override;
+
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
+  int GetRecordAudioParameters(AudioParameters* params) const override;
+#endif  // WEBRTC_IOS
+
+#if defined(WEBRTC_ANDROID)
+  // Only use this acccessor for test purposes on Android.
+  AudioManager* GetAndroidAudioManagerForTest() {
+    return audio_manager_android_.get();
+  }
+#endif
+  AudioDeviceBuffer* GetAudioDeviceBuffer() { return &audio_device_buffer_; }
+
+  int RestartPlayoutInternally() override { return -1; }
+  int RestartRecordingInternally() override { return -1; }
+  int SetPlayoutSampleRate(uint32_t sample_rate) override { return -1; }
+  int SetRecordingSampleRate(uint32_t sample_rate) override { return -1; }
+
+ private:
+  PlatformType Platform() const;
+  AudioLayer PlatformAudioLayer() const;
+
+  AudioLayer audio_layer_;
+  PlatformType platform_type_ = kPlatformNotSupported;
+  bool initialized_ = false;
+#if defined(WEBRTC_ANDROID)
+  // Should be declared first to ensure that it outlives other resources.
+  std::unique_ptr<AudioManager> audio_manager_android_;
+#endif
+  AudioDeviceBuffer audio_device_buffer_;
+  std::unique_ptr<AudioDeviceGeneric> audio_device_;
+};
+
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+
+#endif  // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_name.cc b/third_party/libwebrtc/modules/audio_device/audio_device_name.cc
new file mode 100644
index 0000000000..5318496768
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_name.cc
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/audio_device_name.h"
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+const char AudioDeviceName::kDefaultDeviceId[] = "default";
+
+AudioDeviceName::AudioDeviceName(absl::string_view device_name,
+                                 absl::string_view unique_id)
+    : device_name(device_name), unique_id(unique_id) {}
+
+bool AudioDeviceName::IsValid() {
+  return !device_name.empty() && !unique_id.empty();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_name.h b/third_party/libwebrtc/modules/audio_device/audio_device_name.h
new file mode 100644
index 0000000000..db37852e9a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_name.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_NAME_H_
+#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_NAME_H_
+
+#include <deque>
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+struct AudioDeviceName {
+  // Represents a default device. Note that, on Windows there are two different
+  // types of default devices (Default and Default Communication). They can
+  // either be two different physical devices or be two different roles for one
+  // single device. Hence, this id must be combined with a "role parameter" on
+  // Windows to uniquely identify a default device.
+  static const char kDefaultDeviceId[];
+
+  AudioDeviceName() = default;
+  AudioDeviceName(absl::string_view device_name, absl::string_view unique_id);
+
+  ~AudioDeviceName() = default;
+
+  // Support copy and move.
+  AudioDeviceName(const AudioDeviceName& other) = default;
+  AudioDeviceName(AudioDeviceName&&) = default;
+  AudioDeviceName& operator=(const AudioDeviceName&) = default;
+  AudioDeviceName& operator=(AudioDeviceName&&) = default;
+
+  bool IsValid();
+
+  std::string device_name;  // Friendly name of the device.
+  std::string unique_id;    // Unique identifier for the device.
+};
+
+typedef std::deque<AudioDeviceName> AudioDeviceNames;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_NAME_H_
diff --git a/third_party/libwebrtc/modules/audio_device/audio_device_unittest.cc b/third_party/libwebrtc/modules/audio_device/audio_device_unittest.cc
new file mode 100644
index 0000000000..0a3a88c2e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/audio_device_unittest.cc
@@ -0,0 +1,1241 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/include/audio_device.h"
+
+#include <algorithm>
+#include <cstring>
+#include <list>
+#include <memory>
+#include <numeric>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/audio_device_impl.h"
+#include "modules/audio_device/include/mock_audio_transport.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/time_utils.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#ifdef WEBRTC_WIN
+#include "modules/audio_device/include/audio_device_factory.h"
+#include "modules/audio_device/win/core_audio_utility_win.h"
+#include "rtc_base/win/scoped_com_initializer.h"
+#endif  // WEBRTC_WIN
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Ge;
+using ::testing::Invoke;
+using ::testing::Mock;
+using ::testing::NiceMock;
+using ::testing::NotNull;
+
+namespace webrtc {
+namespace {
+
+// Using a #define for AUDIO_DEVICE since we will call *different* versions of
+// the ADM functions, depending on the ID type.
+#if defined(WEBRTC_WIN)
+#define AUDIO_DEVICE_ID (AudioDeviceModule::WindowsDeviceType::kDefaultDevice)
+#else
+#define AUDIO_DEVICE_ID (0u)
+#endif  // defined(WEBRTC_WIN)
+
+// #define ENABLE_DEBUG_PRINTF
+#ifdef ENABLE_DEBUG_PRINTF
+#define PRINTD(...) fprintf(stderr, __VA_ARGS__);
+#else
+#define PRINTD(...) ((void)0)
+#endif
+#define PRINT(...) fprintf(stderr, __VA_ARGS__);
+
+// Don't run these tests if audio-related requirements are not met.
+#define SKIP_TEST_IF_NOT(requirements_satisfied)         \
+  do {                                                   \
+    if (!requirements_satisfied) {                       \
+      GTEST_SKIP() << "Skipped. No audio device found."; \
+    }                                                    \
+  } while (false)
+
+// Number of callbacks (input or output) the tests waits for before we set
+// an event indicating that the test was OK.
+static constexpr size_t kNumCallbacks = 10;
+// Max amount of time we wait for an event to be set while counting callbacks.
+static constexpr TimeDelta kTestTimeOut = TimeDelta::Seconds(10);
+// Average number of audio callbacks per second assuming 10ms packet size.
+static constexpr size_t kNumCallbacksPerSecond = 100;
+// Run the full-duplex test during this time (unit is in seconds).
+static constexpr TimeDelta kFullDuplexTime = TimeDelta::Seconds(5);
+// Length of round-trip latency measurements. Number of deteced impulses
+// shall be kImpulseFrequencyInHz * kMeasureLatencyTime - 1 since the
+// last transmitted pulse is not used.
+static constexpr TimeDelta kMeasureLatencyTime = TimeDelta::Seconds(10);
+// Sets the number of impulses per second in the latency test.
+static constexpr size_t kImpulseFrequencyInHz = 1;
+// Utilized in round-trip latency measurements to avoid capturing noise samples.
+static constexpr int kImpulseThreshold = 1000;
+
+enum class TransportType {
+  kInvalid,
+  kPlay,
+  kRecord,
+  kPlayAndRecord,
+};
+
+// Interface for processing the audio stream. Real implementations can e.g.
+// run audio in loopback, read audio from a file or perform latency
+// measurements.
+class AudioStream {
+ public:
+  virtual void Write(rtc::ArrayView<const int16_t> source) = 0;
+  virtual void Read(rtc::ArrayView<int16_t> destination) = 0;
+
+  virtual ~AudioStream() = default;
+};
+
+// Converts index corresponding to position within a 10ms buffer into a
+// delay value in milliseconds.
+// Example: index=240, frames_per_10ms_buffer=480 => 5ms as output.
+int IndexToMilliseconds(size_t index, size_t frames_per_10ms_buffer) {
+  return rtc::checked_cast<int>(
+      10.0 * (static_cast<double>(index) / frames_per_10ms_buffer) + 0.5);
+}
+
+}  // namespace
+
+// Simple first in first out (FIFO) class that wraps a list of 16-bit audio
+// buffers of fixed size and allows Write and Read operations. The idea is to
+// store recorded audio buffers (using Write) and then read (using Read) these
+// stored buffers with as short delay as possible when the audio layer needs
+// data to play out. The number of buffers in the FIFO will stabilize under
+// normal conditions since there will be a balance between Write and Read calls.
+// The container is a std::list container and access is protected with a lock
+// since both sides (playout and recording) are driven by its own thread.
+// Note that, we know by design that the size of the audio buffer will not
+// change over time and that both sides will in most cases use the same size.
+class FifoAudioStream : public AudioStream {
+ public:
+  void Write(rtc::ArrayView<const int16_t> source) override {
+    RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+    const size_t size = [&] {
+      MutexLock lock(&lock_);
+      fifo_.push_back(Buffer16(source.data(), source.size()));
+      return fifo_.size();
+    }();
+    if (size > max_size_) {
+      max_size_ = size;
+    }
+    // Add marker once per second to signal that audio is active.
+    if (write_count_++ % 100 == 0) {
+      PRINTD(".");
+    }
+    written_elements_ += size;
+  }
+
+  void Read(rtc::ArrayView<int16_t> destination) override {
+    MutexLock lock(&lock_);
+    if (fifo_.empty()) {
+      std::fill(destination.begin(), destination.end(), 0);
+    } else {
+      const Buffer16& buffer = fifo_.front();
+      if (buffer.size() == destination.size()) {
+        // Default case where input and output uses same sample rate and
+        // channel configuration. No conversion is needed.
+        std::copy(buffer.begin(), buffer.end(), destination.begin());
+      } else if (destination.size() == 2 * buffer.size()) {
+        // Recorded input signal in `buffer` is in mono. Do channel upmix to
+        // match stereo output (1 -> 2).
+        for (size_t i = 0; i < buffer.size(); ++i) {
+          destination[2 * i] = buffer[i];
+          destination[2 * i + 1] = buffer[i];
+        }
+      } else if (buffer.size() == 2 * destination.size()) {
+        // Recorded input signal in `buffer` is in stereo. Do channel downmix
+        // to match mono output (2 -> 1).
+        for (size_t i = 0; i < destination.size(); ++i) {
+          destination[i] =
+              (static_cast<int32_t>(buffer[2 * i]) + buffer[2 * i + 1]) / 2;
+        }
+      } else {
+        RTC_DCHECK_NOTREACHED() << "Required conversion is not support";
+      }
+      fifo_.pop_front();
+    }
+  }
+
+  size_t size() const {
+    MutexLock lock(&lock_);
+    return fifo_.size();
+  }
+
+  size_t max_size() const {
+    RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+    return max_size_;
+  }
+
+  size_t average_size() const {
+    RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+    return 0.5 + static_cast<float>(written_elements_ / write_count_);
+  }
+
+  using Buffer16 = rtc::BufferT<int16_t>;
+
+  mutable Mutex lock_;
+  rtc::RaceChecker race_checker_;
+
+  std::list<Buffer16> fifo_ RTC_GUARDED_BY(lock_);
+  size_t write_count_ RTC_GUARDED_BY(race_checker_) = 0;
+  size_t max_size_ RTC_GUARDED_BY(race_checker_) = 0;
+  size_t written_elements_ RTC_GUARDED_BY(race_checker_) = 0;
+};
+
+// Inserts periodic impulses and measures the latency between the time of
+// transmission and time of receiving the same impulse.
+class LatencyAudioStream : public AudioStream {
+ public:
+  LatencyAudioStream() {
+    // Delay thread checkers from being initialized until first callback from
+    // respective thread.
+    read_thread_checker_.Detach();
+    write_thread_checker_.Detach();
+  }
+
+  // Insert periodic impulses in first two samples of `destination`.
+  void Read(rtc::ArrayView<int16_t> destination) override {
+    RTC_DCHECK_RUN_ON(&read_thread_checker_);
+    if (read_count_ == 0) {
+      PRINT("[");
+    }
+    read_count_++;
+    std::fill(destination.begin(), destination.end(), 0);
+    if (read_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
+      PRINT(".");
+      {
+        MutexLock lock(&lock_);
+        if (!pulse_time_) {
+          pulse_time_ = rtc::TimeMillis();
+        }
+      }
+      constexpr int16_t impulse = std::numeric_limits<int16_t>::max();
+      std::fill_n(destination.begin(), 2, impulse);
+    }
+  }
+
+  // Detect received impulses in `source`, derive time between transmission and
+  // detection and add the calculated delay to list of latencies.
+  void Write(rtc::ArrayView<const int16_t> source) override {
+    RTC_DCHECK_RUN_ON(&write_thread_checker_);
+    RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+    MutexLock lock(&lock_);
+    write_count_++;
+    if (!pulse_time_) {
+      // Avoid detection of new impulse response until a new impulse has
+      // been transmitted (sets `pulse_time_` to value larger than zero).
+      return;
+    }
+    // Find index (element position in vector) of the max element.
+    const size_t index_of_max =
+        std::max_element(source.begin(), source.end()) - source.begin();
+    // Derive time between transmitted pulse and received pulse if the level
+    // is high enough (removes noise).
+    const size_t max = source[index_of_max];
+    if (max > kImpulseThreshold) {
+      PRINTD("(%zu, %zu)", max, index_of_max);
+      int64_t now_time = rtc::TimeMillis();
+      int extra_delay = IndexToMilliseconds(index_of_max, source.size());
+      PRINTD("[%d]", rtc::checked_cast<int>(now_time - pulse_time_));
+      PRINTD("[%d]", extra_delay);
+      // Total latency is the difference between transmit time and detection
+      // tome plus the extra delay within the buffer in which we detected the
+      // received impulse. It is transmitted at sample 0 but can be received
+      // at sample N where N > 0. The term `extra_delay` accounts for N and it
+      // is a value between 0 and 10ms.
+      latencies_.push_back(now_time - *pulse_time_ + extra_delay);
+      pulse_time_.reset();
+    } else {
+      PRINTD("-");
+    }
+  }
+
+  size_t num_latency_values() const {
+    RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+    return latencies_.size();
+  }
+
+  int min_latency() const {
+    RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+    if (latencies_.empty())
+      return 0;
+    return *std::min_element(latencies_.begin(), latencies_.end());
+  }
+
+  int max_latency() const {
+    RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+    if (latencies_.empty())
+      return 0;
+    return *std::max_element(latencies_.begin(), latencies_.end());
+  }
+
+  int average_latency() const {
+    RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+    if (latencies_.empty())
+      return 0;
+    return 0.5 + static_cast<double>(
+                     std::accumulate(latencies_.begin(), latencies_.end(), 0)) /
+                     latencies_.size();
+  }
+
+  void PrintResults() const {
+    RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+    PRINT("] ");
+    for (auto it = latencies_.begin(); it != latencies_.end(); ++it) {
+      PRINTD("%d ", *it);
+    }
+    PRINT("\n");
+    PRINT("[..........] [min, max, avg]=[%d, %d, %d] ms\n", min_latency(),
+          max_latency(), average_latency());
+  }
+
+  Mutex lock_;
+  rtc::RaceChecker race_checker_;
+  SequenceChecker read_thread_checker_;
+  SequenceChecker write_thread_checker_;
+
+  absl::optional<int64_t> pulse_time_ RTC_GUARDED_BY(lock_);
+  std::vector<int> latencies_ RTC_GUARDED_BY(race_checker_);
+  size_t read_count_ RTC_GUARDED_BY(read_thread_checker_) = 0;
+  size_t write_count_ RTC_GUARDED_BY(write_thread_checker_) = 0;
+};
+
+// Mocks the AudioTransport object and proxies actions for the two callbacks
+// (RecordedDataIsAvailable and NeedMorePlayData) to different implementations
+// of AudioStreamInterface.
+class MockAudioTransport : public test::MockAudioTransport {
+ public:
+  explicit MockAudioTransport(TransportType type) : type_(type) {}
+  ~MockAudioTransport() {}
+
+  // Set default actions of the mock object. We are delegating to fake
+  // implementation where the number of callbacks is counted and an event
+  // is set after a certain number of callbacks. Audio parameters are also
+  // checked.
+  void HandleCallbacks(rtc::Event* event,
+                       AudioStream* audio_stream,
+                       int num_callbacks) {
+    event_ = event;
+    audio_stream_ = audio_stream;
+    num_callbacks_ = num_callbacks;
+    if (play_mode()) {
+      ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _))
+          .WillByDefault(
+              Invoke(this, &MockAudioTransport::RealNeedMorePlayData));
+    }
+    if (rec_mode()) {
+      ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _))
+          .WillByDefault(
+              Invoke(this, &MockAudioTransport::RealRecordedDataIsAvailable));
+    }
+  }
+
+  // Special constructor used in manual tests where the user wants to run audio
+  // until e.g. a keyboard key is pressed. The event flag is set to nullptr by
+  // default since it is up to the user to stop the test. See e.g.
+  // DISABLED_RunPlayoutAndRecordingInFullDuplexAndWaitForEnterKey().
+  void HandleCallbacks(AudioStream* audio_stream) {
+    HandleCallbacks(nullptr, audio_stream, 0);
+  }
+
+  int32_t RealRecordedDataIsAvailable(const void* audio_buffer,
+                                      const size_t samples_per_channel,
+                                      const size_t bytes_per_frame,
+                                      const size_t channels,
+                                      const uint32_t sample_rate,
+                                      const uint32_t total_delay_ms,
+                                      const int32_t clock_drift,
+                                      const uint32_t current_mic_level,
+                                      const bool typing_status,
+                                      uint32_t& new_mic_level) {
+    EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
+    // Store audio parameters once in the first callback. For all other
+    // callbacks, verify that the provided audio parameters are maintained and
+    // that each callback corresponds to 10ms for any given sample rate.
+    if (!record_parameters_.is_complete()) {
+      record_parameters_.reset(sample_rate, channels, samples_per_channel);
+    } else {
+      EXPECT_EQ(samples_per_channel, record_parameters_.frames_per_buffer());
+      EXPECT_EQ(bytes_per_frame, record_parameters_.GetBytesPerFrame());
+      EXPECT_EQ(channels, record_parameters_.channels());
+      EXPECT_EQ(static_cast<int>(sample_rate),
+                record_parameters_.sample_rate());
+      EXPECT_EQ(samples_per_channel,
+                record_parameters_.frames_per_10ms_buffer());
+    }
+    {
+      MutexLock lock(&lock_);
+      rec_count_++;
+    }
+    // Write audio data to audio stream object if one has been injected.
+    if (audio_stream_) {
+      audio_stream_->Write(
+          rtc::MakeArrayView(static_cast<const int16_t*>(audio_buffer),
+                             samples_per_channel * channels));
+    }
+    // Signal the event after given amount of callbacks.
+    if (event_ && ReceivedEnoughCallbacks()) {
+      event_->Set();
+    }
+    return 0;
+  }
+
+  int32_t RealNeedMorePlayData(const size_t samples_per_channel,
+                               const size_t bytes_per_frame,
+                               const size_t channels,
+                               const uint32_t sample_rate,
+                               void* audio_buffer,
+                               size_t& samples_out,
+                               int64_t* elapsed_time_ms,
+                               int64_t* ntp_time_ms) {
+    EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
+    // Store audio parameters once in the first callback. For all other
+    // callbacks, verify that the provided audio parameters are maintained and
+    // that each callback corresponds to 10ms for any given sample rate.
+    if (!playout_parameters_.is_complete()) {
+      playout_parameters_.reset(sample_rate, channels, samples_per_channel);
+    } else {
+      EXPECT_EQ(samples_per_channel, playout_parameters_.frames_per_buffer());
+      EXPECT_EQ(bytes_per_frame, playout_parameters_.GetBytesPerFrame());
+      EXPECT_EQ(channels, playout_parameters_.channels());
+      EXPECT_EQ(static_cast<int>(sample_rate),
+                playout_parameters_.sample_rate());
+      EXPECT_EQ(samples_per_channel,
+                playout_parameters_.frames_per_10ms_buffer());
+    }
+    {
+      MutexLock lock(&lock_);
+      play_count_++;
+    }
+    samples_out = samples_per_channel * channels;
+    // Read audio data from audio stream object if one has been injected.
+    if (audio_stream_) {
+      audio_stream_->Read(rtc::MakeArrayView(
+          static_cast<int16_t*>(audio_buffer), samples_per_channel * channels));
+    } else {
+      // Fill the audio buffer with zeros to avoid disturbing audio.
+      const size_t num_bytes = samples_per_channel * bytes_per_frame;
+      std::memset(audio_buffer, 0, num_bytes);
+    }
+    // Signal the event after given amount of callbacks.
+    if (event_ && ReceivedEnoughCallbacks()) {
+      event_->Set();
+    }
+    return 0;
+  }
+
+  bool ReceivedEnoughCallbacks() {
+    bool recording_done = false;
+    if (rec_mode()) {
+      MutexLock lock(&lock_);
+      recording_done = rec_count_ >= num_callbacks_;
+    } else {
+      recording_done = true;
+    }
+    bool playout_done = false;
+    if (play_mode()) {
+      MutexLock lock(&lock_);
+      playout_done = play_count_ >= num_callbacks_;
+    } else {
+      playout_done = true;
+    }
+    return recording_done && playout_done;
+  }
+
+  bool play_mode() const {
+    return type_ == TransportType::kPlay ||
+           type_ == TransportType::kPlayAndRecord;
+  }
+
+  bool rec_mode() const {
+    return type_ == TransportType::kRecord ||
+           type_ == TransportType::kPlayAndRecord;
+  }
+
+  void ResetCallbackCounters() {
+    MutexLock lock(&lock_);
+    if (play_mode()) {
+      play_count_ = 0;
+    }
+    if (rec_mode()) {
+      rec_count_ = 0;
+    }
+  }
+
+ private:
+  Mutex lock_;
+  TransportType type_ = TransportType::kInvalid;
+  rtc::Event* event_ = nullptr;
+  AudioStream* audio_stream_ = nullptr;
+  size_t num_callbacks_ = 0;
+  size_t play_count_ RTC_GUARDED_BY(lock_) = 0;
+  size_t rec_count_ RTC_GUARDED_BY(lock_) = 0;
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+};
+
+// AudioDeviceTest test fixture.
+
+// bugs.webrtc.org/9808
+// Both the tests and the code under test are very old, unstaffed and not
+// a part of webRTC stack.
+// Here sanitizers make the tests hang, without providing usefull report.
+// So we are just disabling them, without intention to re-enable them.
+#if defined(ADDRESS_SANITIZER) || defined(MEMORY_SANITIZER) || \
+    defined(THREAD_SANITIZER) || defined(UNDEFINED_SANITIZER)
+#define MAYBE_AudioDeviceTest DISABLED_AudioDeviceTest
+#else
+#define MAYBE_AudioDeviceTest AudioDeviceTest
+#endif
+
+class MAYBE_AudioDeviceTest
+    : public ::testing::TestWithParam<webrtc::AudioDeviceModule::AudioLayer> {
+ protected:
+  MAYBE_AudioDeviceTest()
+      : audio_layer_(GetParam()),
+        task_queue_factory_(CreateDefaultTaskQueueFactory()) {
+    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+    // Add extra logging fields here if needed for debugging.
+    rtc::LogMessage::LogTimestamps();
+    rtc::LogMessage::LogThreads();
+    audio_device_ = CreateAudioDevice();
+    EXPECT_NE(audio_device_.get(), nullptr);
+    AudioDeviceModule::AudioLayer audio_layer;
+    int got_platform_audio_layer =
+        audio_device_->ActiveAudioLayer(&audio_layer);
+    // First, ensure that a valid audio layer can be activated.
+    if (got_platform_audio_layer != 0) {
+      requirements_satisfied_ = false;
+    }
+    // Next, verify that the ADM can be initialized.
+    if (requirements_satisfied_) {
+      requirements_satisfied_ = (audio_device_->Init() == 0);
+    }
+    // Finally, ensure that at least one valid device exists in each direction.
+    if (requirements_satisfied_) {
+      const int16_t num_playout_devices = audio_device_->PlayoutDevices();
+      const int16_t num_record_devices = audio_device_->RecordingDevices();
+      requirements_satisfied_ =
+          num_playout_devices > 0 && num_record_devices > 0;
+    }
+    if (requirements_satisfied_) {
+      EXPECT_EQ(0, audio_device_->SetPlayoutDevice(AUDIO_DEVICE_ID));
+      EXPECT_EQ(0, audio_device_->InitSpeaker());
+      EXPECT_EQ(0, audio_device_->StereoPlayoutIsAvailable(&stereo_playout_));
+      EXPECT_EQ(0, audio_device_->SetStereoPlayout(stereo_playout_));
+      EXPECT_EQ(0, audio_device_->SetRecordingDevice(AUDIO_DEVICE_ID));
+      EXPECT_EQ(0, audio_device_->InitMicrophone());
+      // Avoid asking for input stereo support and always record in mono
+      // since asking can cause issues in combination with remote desktop.
+      // See https://bugs.chromium.org/p/webrtc/issues/detail?id=7397 for
+      // details.
+      EXPECT_EQ(0, audio_device_->SetStereoRecording(false));
+    }
+  }
+
+  // This is needed by all tests using MockAudioTransport,
+  // since there is no way to unregister it.
+  // Without Terminate(), audio_device would still accesses
+  // the destructed mock via "webrtc_audio_module_rec_thread".
+  // An alternative would be for the mock to outlive audio_device.
+  void PreTearDown() { EXPECT_EQ(0, audio_device_->Terminate()); }
+
+  virtual ~MAYBE_AudioDeviceTest() {
+    if (audio_device_) {
+      EXPECT_EQ(0, audio_device_->Terminate());
+    }
+  }
+
+  bool requirements_satisfied() const { return requirements_satisfied_; }
+  rtc::Event* event() { return &event_; }
+  AudioDeviceModule::AudioLayer audio_layer() const { return audio_layer_; }
+
+  // AudioDeviceModuleForTest extends the default ADM interface with some extra
+  // test methods. Intended for usage in tests only and requires a unique
+  // factory method. See CreateAudioDevice() for details.
+  const rtc::scoped_refptr<AudioDeviceModuleForTest>& audio_device() const {
+    return audio_device_;
+  }
+
+  rtc::scoped_refptr<AudioDeviceModuleForTest> CreateAudioDevice() {
+    // Use the default factory for kPlatformDefaultAudio and a special factory
+    // CreateWindowsCoreAudioAudioDeviceModuleForTest() for kWindowsCoreAudio2.
+    // The value of `audio_layer_` is set at construction by GetParam() and two
+    // different layers are tested on Windows only.
+    if (audio_layer_ == AudioDeviceModule::kPlatformDefaultAudio) {
+      return AudioDeviceModule::CreateForTest(audio_layer_,
+                                              task_queue_factory_.get());
+    } else if (audio_layer_ == AudioDeviceModule::kWindowsCoreAudio2) {
+#ifdef WEBRTC_WIN
+      // We must initialize the COM library on a thread before we calling any of
+      // the library functions. All COM functions in the ADM will return
+      // CO_E_NOTINITIALIZED otherwise.
+      com_initializer_ =
+          std::make_unique<ScopedCOMInitializer>(ScopedCOMInitializer::kMTA);
+      EXPECT_TRUE(com_initializer_->Succeeded());
+      EXPECT_TRUE(webrtc_win::core_audio_utility::IsSupported());
+      EXPECT_TRUE(webrtc_win::core_audio_utility::IsMMCSSSupported());
+      return CreateWindowsCoreAudioAudioDeviceModuleForTest(
+          task_queue_factory_.get(), true);
+#else
+      return nullptr;
+#endif
+    } else {
+      return nullptr;
+    }
+  }
+
+  void StartPlayout() {
+    EXPECT_FALSE(audio_device()->Playing());
+    EXPECT_EQ(0, audio_device()->InitPlayout());
+    EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
+    EXPECT_EQ(0, audio_device()->StartPlayout());
+    EXPECT_TRUE(audio_device()->Playing());
+  }
+
+  void StopPlayout() {
+    EXPECT_EQ(0, audio_device()->StopPlayout());
+    EXPECT_FALSE(audio_device()->Playing());
+    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
+  }
+
+  void StartRecording() {
+    EXPECT_FALSE(audio_device()->Recording());
+    EXPECT_EQ(0, audio_device()->InitRecording());
+    EXPECT_TRUE(audio_device()->RecordingIsInitialized());
+    EXPECT_EQ(0, audio_device()->StartRecording());
+    EXPECT_TRUE(audio_device()->Recording());
+  }
+
+  void StopRecording() {
+    EXPECT_EQ(0, audio_device()->StopRecording());
+    EXPECT_FALSE(audio_device()->Recording());
+    EXPECT_FALSE(audio_device()->RecordingIsInitialized());
+  }
+
+  bool NewWindowsAudioDeviceModuleIsUsed() {
+#ifdef WEBRTC_WIN
+    AudioDeviceModule::AudioLayer audio_layer;
+    EXPECT_EQ(0, audio_device()->ActiveAudioLayer(&audio_layer));
+    if (audio_layer == AudioDeviceModule::kWindowsCoreAudio2) {
+      // Default device is always added as first element in the list and the
+      // default communication device as the second element. Hence, the list
+      // contains two extra elements in this case.
+      return true;
+    }
+#endif
+    return false;
+  }
+
+ private:
+#ifdef WEBRTC_WIN
+  // Windows Core Audio based ADM needs to run on a COM initialized thread.
+  std::unique_ptr<ScopedCOMInitializer> com_initializer_;
+#endif
+  AudioDeviceModule::AudioLayer audio_layer_;
+  std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+  bool requirements_satisfied_ = true;
+  rtc::Event event_;
+  rtc::scoped_refptr<AudioDeviceModuleForTest> audio_device_;
+  bool stereo_playout_ = false;
+};
+
+// Instead of using the test fixture, verify that the different factory methods
+// work as intended.
+TEST(MAYBE_AudioDeviceTestWin, ConstructDestructWithFactory) {
+  std::unique_ptr<TaskQueueFactory> task_queue_factory =
+      CreateDefaultTaskQueueFactory();
+  rtc::scoped_refptr<AudioDeviceModule> audio_device;
+  // The default factory should work for all platforms when a default ADM is
+  // requested.
+  audio_device = AudioDeviceModule::Create(
+      AudioDeviceModule::kPlatformDefaultAudio, task_queue_factory.get());
+  EXPECT_TRUE(audio_device);
+  audio_device = nullptr;
+#ifdef WEBRTC_WIN
+  // For Windows, the old factory method creates an ADM where the platform-
+  // specific parts are implemented by an AudioDeviceGeneric object. Verify
+  // that the old factory can't be used in combination with the latest audio
+  // layer AudioDeviceModule::kWindowsCoreAudio2.
+  audio_device = AudioDeviceModule::Create(
+      AudioDeviceModule::kWindowsCoreAudio2, task_queue_factory.get());
+  EXPECT_FALSE(audio_device);
+  audio_device = nullptr;
+  // Instead, ensure that the new dedicated factory method called
+  // CreateWindowsCoreAudioAudioDeviceModule() can be used on Windows and that
+  // it sets the audio layer to kWindowsCoreAudio2 implicitly. Note that, the
+  // new ADM for Windows must be created on a COM thread.
+  ScopedCOMInitializer com_initializer(ScopedCOMInitializer::kMTA);
+  EXPECT_TRUE(com_initializer.Succeeded());
+  audio_device =
+      CreateWindowsCoreAudioAudioDeviceModule(task_queue_factory.get());
+  EXPECT_TRUE(audio_device);
+  AudioDeviceModule::AudioLayer audio_layer;
+  EXPECT_EQ(0, audio_device->ActiveAudioLayer(&audio_layer));
+  EXPECT_EQ(audio_layer, AudioDeviceModule::kWindowsCoreAudio2);
+#endif
+}
+
+// Uses the test fixture to create, initialize and destruct the ADM.
+TEST_P(MAYBE_AudioDeviceTest, ConstructDestructDefault) {}
+
+TEST_P(MAYBE_AudioDeviceTest, InitTerminate) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  // Initialization is part of the test fixture.
+  EXPECT_TRUE(audio_device()->Initialized());
+  EXPECT_EQ(0, audio_device()->Terminate());
+  EXPECT_FALSE(audio_device()->Initialized());
+}
+
+// Enumerate all available and active output devices.
+TEST_P(MAYBE_AudioDeviceTest, PlayoutDeviceNames) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  char device_name[kAdmMaxDeviceNameSize];
+  char unique_id[kAdmMaxGuidSize];
+  int num_devices = audio_device()->PlayoutDevices();
+  if (NewWindowsAudioDeviceModuleIsUsed()) {
+    num_devices += 2;
+  }
+  EXPECT_GT(num_devices, 0);
+  for (int i = 0; i < num_devices; ++i) {
+    EXPECT_EQ(0, audio_device()->PlayoutDeviceName(i, device_name, unique_id));
+  }
+  EXPECT_EQ(-1, audio_device()->PlayoutDeviceName(num_devices, device_name,
+                                                  unique_id));
+}
+
+// Enumerate all available and active input devices.
+TEST_P(MAYBE_AudioDeviceTest, RecordingDeviceNames) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  char device_name[kAdmMaxDeviceNameSize];
+  char unique_id[kAdmMaxGuidSize];
+  int num_devices = audio_device()->RecordingDevices();
+  if (NewWindowsAudioDeviceModuleIsUsed()) {
+    num_devices += 2;
+  }
+  EXPECT_GT(num_devices, 0);
+  for (int i = 0; i < num_devices; ++i) {
+    EXPECT_EQ(0,
+              audio_device()->RecordingDeviceName(i, device_name, unique_id));
+  }
+  EXPECT_EQ(-1, audio_device()->RecordingDeviceName(num_devices, device_name,
+                                                    unique_id));
+}
+
+// Counts number of active output devices and ensure that all can be selected.
+TEST_P(MAYBE_AudioDeviceTest, SetPlayoutDevice) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  int num_devices = audio_device()->PlayoutDevices();
+  if (NewWindowsAudioDeviceModuleIsUsed()) {
+    num_devices += 2;
+  }
+  EXPECT_GT(num_devices, 0);
+  // Verify that all available playout devices can be set (not enabled yet).
+  for (int i = 0; i < num_devices; ++i) {
+    EXPECT_EQ(0, audio_device()->SetPlayoutDevice(i));
+  }
+  EXPECT_EQ(-1, audio_device()->SetPlayoutDevice(num_devices));
+#ifdef WEBRTC_WIN
+  // On Windows, verify the alternative method where the user can select device
+  // by role.
+  EXPECT_EQ(
+      0, audio_device()->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice));
+  EXPECT_EQ(0, audio_device()->SetPlayoutDevice(
+                   AudioDeviceModule::kDefaultCommunicationDevice));
+#endif
+}
+
+// Counts number of active input devices and ensure that all can be selected.
+TEST_P(MAYBE_AudioDeviceTest, SetRecordingDevice) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  int num_devices = audio_device()->RecordingDevices();
+  if (NewWindowsAudioDeviceModuleIsUsed()) {
+    num_devices += 2;
+  }
+  EXPECT_GT(num_devices, 0);
+  // Verify that all available recording devices can be set (not enabled yet).
+  for (int i = 0; i < num_devices; ++i) {
+    EXPECT_EQ(0, audio_device()->SetRecordingDevice(i));
+  }
+  EXPECT_EQ(-1, audio_device()->SetRecordingDevice(num_devices));
+#ifdef WEBRTC_WIN
+  // On Windows, verify the alternative method where the user can select device
+  // by role.
+  EXPECT_EQ(
+      0, audio_device()->SetRecordingDevice(AudioDeviceModule::kDefaultDevice));
+  EXPECT_EQ(0, audio_device()->SetRecordingDevice(
+                   AudioDeviceModule::kDefaultCommunicationDevice));
+#endif
+}
+
+// Tests Start/Stop playout without any registered audio callback.
+TEST_P(MAYBE_AudioDeviceTest, StartStopPlayout) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartPlayout();
+  StopPlayout();
+}
+
+// Tests Start/Stop recording without any registered audio callback.
+TEST_P(MAYBE_AudioDeviceTest, StartStopRecording) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartRecording();
+  StopRecording();
+}
+
+// Tests Start/Stop playout for all available input devices to ensure that
+// the selected device can be created and used as intended.
+TEST_P(MAYBE_AudioDeviceTest, StartStopPlayoutWithRealDevice) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  int num_devices = audio_device()->PlayoutDevices();
+  if (NewWindowsAudioDeviceModuleIsUsed()) {
+    num_devices += 2;
+  }
+  EXPECT_GT(num_devices, 0);
+  // Verify that all available playout devices can be set and used.
+  for (int i = 0; i < num_devices; ++i) {
+    EXPECT_EQ(0, audio_device()->SetPlayoutDevice(i));
+    StartPlayout();
+    StopPlayout();
+  }
+#ifdef WEBRTC_WIN
+  AudioDeviceModule::WindowsDeviceType device_role[] = {
+      AudioDeviceModule::kDefaultDevice,
+      AudioDeviceModule::kDefaultCommunicationDevice};
+  for (size_t i = 0; i < arraysize(device_role); ++i) {
+    EXPECT_EQ(0, audio_device()->SetPlayoutDevice(device_role[i]));
+    StartPlayout();
+    StopPlayout();
+  }
+#endif
+}
+
+// Tests Start/Stop recording for all available input devices to ensure that
+// the selected device can be created and used as intended.
+TEST_P(MAYBE_AudioDeviceTest, StartStopRecordingWithRealDevice) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  int num_devices = audio_device()->RecordingDevices();
+  if (NewWindowsAudioDeviceModuleIsUsed()) {
+    num_devices += 2;
+  }
+  EXPECT_GT(num_devices, 0);
+  // Verify that all available recording devices can be set and used.
+  for (int i = 0; i < num_devices; ++i) {
+    EXPECT_EQ(0, audio_device()->SetRecordingDevice(i));
+    StartRecording();
+    StopRecording();
+  }
+#ifdef WEBRTC_WIN
+  AudioDeviceModule::WindowsDeviceType device_role[] = {
+      AudioDeviceModule::kDefaultDevice,
+      AudioDeviceModule::kDefaultCommunicationDevice};
+  for (size_t i = 0; i < arraysize(device_role); ++i) {
+    EXPECT_EQ(0, audio_device()->SetRecordingDevice(device_role[i]));
+    StartRecording();
+    StopRecording();
+  }
+#endif
+}
+
+// Tests Init/Stop/Init recording without any registered audio callback.
+// See https://bugs.chromium.org/p/webrtc/issues/detail?id=8041 for details
+// on why this test is useful.
+TEST_P(MAYBE_AudioDeviceTest, InitStopInitRecording) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  EXPECT_TRUE(audio_device()->RecordingIsInitialized());
+  StopRecording();
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  StopRecording();
+}
+
+// Verify that additional attempts to initialize or start recording while
+// already being active works. Additional calls should just be ignored.
+TEST_P(MAYBE_AudioDeviceTest, StartInitRecording) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartRecording();
+  // An additional attempt to initialize at this stage should be ignored.
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  // Same for additional request to start recording while already active.
+  EXPECT_EQ(0, audio_device()->StartRecording());
+  StopRecording();
+}
+
+// Verify that additional attempts to initialize or start playou while
+// already being active works. Additional calls should just be ignored.
+TEST_P(MAYBE_AudioDeviceTest, StartInitPlayout) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartPlayout();
+  // An additional attempt to initialize at this stage should be ignored.
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  // Same for additional request to start playout while already active.
+  EXPECT_EQ(0, audio_device()->StartPlayout());
+  StopPlayout();
+}
+
+// Tests Init/Stop/Init recording while playout is active.
+TEST_P(MAYBE_AudioDeviceTest, InitStopInitRecordingWhilePlaying) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartPlayout();
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  EXPECT_TRUE(audio_device()->RecordingIsInitialized());
+  StopRecording();
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  StopRecording();
+  StopPlayout();
+}
+
+// Tests Init/Stop/Init playout without any registered audio callback.
+TEST_P(MAYBE_AudioDeviceTest, InitStopInitPlayout) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
+  StopPlayout();
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  StopPlayout();
+}
+
+// Tests Init/Stop/Init playout while recording is active.
+TEST_P(MAYBE_AudioDeviceTest, InitStopInitPlayoutWhileRecording) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartRecording();
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
+  StopPlayout();
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  StopPlayout();
+  StopRecording();
+}
+
+// TODO(henrika): restart without intermediate destruction is currently only
+// supported on Windows.
+#ifdef WEBRTC_WIN
+// Tests Start/Stop playout followed by a second session (emulates a restart
+// triggered by a user using public APIs).
+TEST_P(MAYBE_AudioDeviceTest, StartStopPlayoutWithExternalRestart) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartPlayout();
+  StopPlayout();
+  // Restart playout without destroying the ADM in between. Ensures that we
+  // support: Init(), Start(), Stop(), Init(), Start(), Stop().
+  StartPlayout();
+  StopPlayout();
+}
+
+// Tests Start/Stop recording followed by a second session (emulates a restart
+// triggered by a user using public APIs).
+TEST_P(MAYBE_AudioDeviceTest, StartStopRecordingWithExternalRestart) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartRecording();
+  StopRecording();
+  // Restart recording without destroying the ADM in between.  Ensures that we
+  // support: Init(), Start(), Stop(), Init(), Start(), Stop().
+  StartRecording();
+  StopRecording();
+}
+
+// Tests Start/Stop playout followed by a second session (emulates a restart
+// triggered by an internal callback e.g. corresponding to a device switch).
+// Note that, internal restart is only supported in combination with the latest
+// Windows ADM.
+TEST_P(MAYBE_AudioDeviceTest, StartStopPlayoutWithInternalRestart) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  if (audio_layer() != AudioDeviceModule::kWindowsCoreAudio2) {
+    return;
+  }
+  MockAudioTransport mock(TransportType::kPlay);
+  mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
+  EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  event()->Wait(kTestTimeOut);
+  EXPECT_TRUE(audio_device()->Playing());
+  // Restart playout but without stopping the internal audio thread.
+  // This procedure uses a non-public test API and it emulates what happens
+  // inside the ADM when e.g. a device is removed.
+  EXPECT_EQ(0, audio_device()->RestartPlayoutInternally());
+
+  // Run basic tests of public APIs while a restart attempt is active.
+  // These calls should now be very thin and not trigger any new actions.
+  EXPECT_EQ(-1, audio_device()->StopPlayout());
+  EXPECT_TRUE(audio_device()->Playing());
+  EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  EXPECT_EQ(0, audio_device()->StartPlayout());
+
+  // Wait until audio has restarted and a new sequence of audio callbacks
+  // becomes active.
+  // TODO(henrika): is it possible to verify that the internal state transition
+  // is Stop->Init->Start?
+  ASSERT_TRUE(Mock::VerifyAndClearExpectations(&mock));
+  mock.ResetCallbackCounters();
+  EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  event()->Wait(kTestTimeOut);
+  EXPECT_TRUE(audio_device()->Playing());
+  // Stop playout and the audio thread after successful internal restart.
+  StopPlayout();
+  PreTearDown();
+}
+
+// Tests Start/Stop recording followed by a second session (emulates a restart
+// triggered by an internal callback e.g. corresponding to a device switch).
+// Note that, internal restart is only supported in combination with the latest
+// Windows ADM.
+TEST_P(MAYBE_AudioDeviceTest, StartStopRecordingWithInternalRestart) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  if (audio_layer() != AudioDeviceModule::kWindowsCoreAudio2) {
+    return;
+  }
+  MockAudioTransport mock(TransportType::kRecord);
+  mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
+  EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
+                                            false, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartRecording();
+  event()->Wait(kTestTimeOut);
+  EXPECT_TRUE(audio_device()->Recording());
+  // Restart recording but without stopping the internal audio thread.
+  // This procedure uses a non-public test API and it emulates what happens
+  // inside the ADM when e.g. a device is removed.
+  EXPECT_EQ(0, audio_device()->RestartRecordingInternally());
+
+  // Run basic tests of public APIs while a restart attempt is active.
+  // These calls should now be very thin and not trigger any new actions.
+  EXPECT_EQ(-1, audio_device()->StopRecording());
+  EXPECT_TRUE(audio_device()->Recording());
+  EXPECT_TRUE(audio_device()->RecordingIsInitialized());
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  EXPECT_EQ(0, audio_device()->StartRecording());
+
+  // Wait until audio has restarted and a new sequence of audio callbacks
+  // becomes active.
+  // TODO(henrika): is it possible to verify that the internal state transition
+  // is Stop->Init->Start?
+  ASSERT_TRUE(Mock::VerifyAndClearExpectations(&mock));
+  mock.ResetCallbackCounters();
+  EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
+                                            false, _))
+      .Times(AtLeast(kNumCallbacks));
+  event()->Wait(kTestTimeOut);
+  EXPECT_TRUE(audio_device()->Recording());
+  // Stop recording and the audio thread after successful internal restart.
+  StopRecording();
+  PreTearDown();
+}
+#endif  // #ifdef WEBRTC_WIN
+
+// Start playout and verify that the native audio layer starts asking for real
+// audio samples to play out using the NeedMorePlayData() callback.
+// Note that we can't add expectations on audio parameters in EXPECT_CALL
+// since parameter are not provided in the each callback. We therefore test and
+// verify the parameters in the fake audio transport implementation instead.
+TEST_P(MAYBE_AudioDeviceTest, StartPlayoutVerifyCallbacks) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  MockAudioTransport mock(TransportType::kPlay);
+  mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
+  EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  event()->Wait(kTestTimeOut);
+  StopPlayout();
+  PreTearDown();
+}
+
+// Don't run these tests in combination with sanitizers.
+// They are already flaky *without* sanitizers.
+// Sanitizers seem to increase flakiness (which brings noise),
+// without reporting anything.
+// TODO(webrtc:10867): Re-enable when flakiness fixed.
+#if defined(ADDRESS_SANITIZER) || defined(MEMORY_SANITIZER) || \
+    defined(THREAD_SANITIZER)
+#define MAYBE_StartRecordingVerifyCallbacks \
+  DISABLED_StartRecordingVerifyCallbacks
+#define MAYBE_StartPlayoutAndRecordingVerifyCallbacks \
+  DISABLED_StartPlayoutAndRecordingVerifyCallbacks
+#else
+#define MAYBE_StartRecordingVerifyCallbacks StartRecordingVerifyCallbacks
+#define MAYBE_StartPlayoutAndRecordingVerifyCallbacks \
+  StartPlayoutAndRecordingVerifyCallbacks
+#endif
+
+// Start recording and verify that the native audio layer starts providing real
+// audio samples using the RecordedDataIsAvailable() callback.
+TEST_P(MAYBE_AudioDeviceTest, MAYBE_StartRecordingVerifyCallbacks) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  MockAudioTransport mock(TransportType::kRecord);
+  mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
+  EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
+                                            false, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartRecording();
+  event()->Wait(kTestTimeOut);
+  StopRecording();
+  PreTearDown();
+}
+
+// Start playout and recording (full-duplex audio) and verify that audio is
+// active in both directions.
+TEST_P(MAYBE_AudioDeviceTest, MAYBE_StartPlayoutAndRecordingVerifyCallbacks) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  MockAudioTransport mock(TransportType::kPlayAndRecord);
+  mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
+  EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(), _, _, _, _, Ge(0u), 0, _,
+                                            false, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+  StartRecording();
+  event()->Wait(kTestTimeOut);
+  StopRecording();
+  StopPlayout();
+  PreTearDown();
+}
+
+// Start playout and recording and store recorded data in an intermediate FIFO
+// buffer from which the playout side then reads its samples in the same order
+// as they were stored. Under ideal circumstances, a callback sequence would
+// look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-'
+// means 'packet played'. Under such conditions, the FIFO would contain max 1,
+// with an average somewhere in (0,1) depending on how long the packets are
+// buffered. However, under more realistic conditions, the size
+// of the FIFO will vary more due to an unbalance between the two sides.
+// This test tries to verify that the device maintains a balanced callback-
+// sequence by running in loopback for a few seconds while measuring the size
+// (max and average) of the FIFO. The size of the FIFO is increased by the
+// recording side and decreased by the playout side.
+TEST_P(MAYBE_AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  NiceMock<MockAudioTransport> mock(TransportType::kPlayAndRecord);
+  FifoAudioStream audio_stream;
+  mock.HandleCallbacks(event(), &audio_stream,
+                       kFullDuplexTime.seconds() * kNumCallbacksPerSecond);
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  // Run both sides using the same channel configuration to avoid conversions
+  // between mono/stereo while running in full duplex mode. Also, some devices
+  // (mainly on Windows) do not support mono.
+  EXPECT_EQ(0, audio_device()->SetStereoPlayout(true));
+  EXPECT_EQ(0, audio_device()->SetStereoRecording(true));
+  // Mute speakers to prevent howling.
+  EXPECT_EQ(0, audio_device()->SetSpeakerVolume(0));
+  StartPlayout();
+  StartRecording();
+  event()->Wait(std::max(kTestTimeOut, kFullDuplexTime));
+  StopRecording();
+  StopPlayout();
+  PreTearDown();
+}
+
+// Runs audio in full duplex until user hits Enter. Intended as a manual test
+// to ensure that the audio quality is good and that real device switches works
+// as intended.
+TEST_P(MAYBE_AudioDeviceTest,
+       DISABLED_RunPlayoutAndRecordingInFullDuplexAndWaitForEnterKey) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  if (audio_layer() != AudioDeviceModule::kWindowsCoreAudio2) {
+    return;
+  }
+  NiceMock<MockAudioTransport> mock(TransportType::kPlayAndRecord);
+  FifoAudioStream audio_stream;
+  mock.HandleCallbacks(&audio_stream);
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  EXPECT_EQ(0, audio_device()->SetStereoPlayout(true));
+  EXPECT_EQ(0, audio_device()->SetStereoRecording(true));
+  // Ensure that the sample rate for both directions are identical so that we
+  // always can listen to our own voice. Will lead to rate conversion (and
+  // higher latency) if the native sample rate is not 48kHz.
+  EXPECT_EQ(0, audio_device()->SetPlayoutSampleRate(48000));
+  EXPECT_EQ(0, audio_device()->SetRecordingSampleRate(48000));
+  StartPlayout();
+  StartRecording();
+  do {
+    PRINT("Loopback audio is active at 48kHz. Press Enter to stop.\n");
+  } while (getchar() != '\n');
+  StopRecording();
+  StopPlayout();
+  PreTearDown();
+}
+
+// Measures loopback latency and reports the min, max and average values for
+// a full duplex audio session.
+// The latency is measured like so:
+// - Insert impulses periodically on the output side.
+// - Detect the impulses on the input side.
+// - Measure the time difference between the transmit time and receive time.
+// - Store time differences in a vector and calculate min, max and average.
+// This test needs the '--gtest_also_run_disabled_tests' flag to run and also
+// some sort of audio feedback loop. E.g. a headset where the mic is placed
+// close to the speaker to ensure highest possible echo. It is also recommended
+// to run the test at highest possible output volume.
+TEST_P(MAYBE_AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  NiceMock<MockAudioTransport> mock(TransportType::kPlayAndRecord);
+  LatencyAudioStream audio_stream;
+  mock.HandleCallbacks(event(), &audio_stream,
+                       kMeasureLatencyTime.seconds() * kNumCallbacksPerSecond);
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  EXPECT_EQ(0, audio_device()->SetStereoPlayout(true));
+  EXPECT_EQ(0, audio_device()->SetStereoRecording(true));
+  StartPlayout();
+  StartRecording();
+  event()->Wait(std::max(kTestTimeOut, kMeasureLatencyTime));
+  StopRecording();
+  StopPlayout();
+  // Avoid concurrent access to audio_stream.
+  PreTearDown();
+  // Verify that a sufficient number of transmitted impulses are detected.
+  EXPECT_GE(audio_stream.num_latency_values(),
+            static_cast<size_t>(
+                kImpulseFrequencyInHz * kMeasureLatencyTime.seconds() - 2));
+  // Print out min, max and average delay values for debugging purposes.
+  audio_stream.PrintResults();
+}
+
+#ifdef WEBRTC_WIN
+// Test two different audio layers (or rather two different Core Audio
+// implementations) for Windows.
+INSTANTIATE_TEST_SUITE_P(
+    AudioLayerWin,
+    MAYBE_AudioDeviceTest,
+    ::testing::Values(AudioDeviceModule::kPlatformDefaultAudio,
+                      AudioDeviceModule::kWindowsCoreAudio2));
+#else
+// For all platforms but Windows, only test the default audio layer.
+INSTANTIATE_TEST_SUITE_P(
+    AudioLayer,
+    MAYBE_AudioDeviceTest,
+    ::testing::Values(AudioDeviceModule::kPlatformDefaultAudio));
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/dummy/audio_device_dummy.cc b/third_party/libwebrtc/modules/audio_device/dummy/audio_device_dummy.cc
new file mode 100644
index 0000000000..b8fd837038
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/dummy/audio_device_dummy.cc
@@ -0,0 +1,226 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/dummy/audio_device_dummy.h"
+
+namespace webrtc {
+
+int32_t AudioDeviceDummy::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  return -1;
+}
+
+AudioDeviceGeneric::InitStatus AudioDeviceDummy::Init() {
+  return InitStatus::OK;
+}
+
+int32_t AudioDeviceDummy::Terminate() {
+  return 0;
+}
+
+bool AudioDeviceDummy::Initialized() const {
+  return true;
+}
+
+int16_t AudioDeviceDummy::PlayoutDevices() {
+  return -1;
+}
+
+int16_t AudioDeviceDummy::RecordingDevices() {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::PlayoutDeviceName(uint16_t index,
+                                            char name[kAdmMaxDeviceNameSize],
+                                            char guid[kAdmMaxGuidSize]) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::RecordingDeviceName(uint16_t index,
+                                              char name[kAdmMaxDeviceNameSize],
+                                              char guid[kAdmMaxGuidSize]) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetPlayoutDevice(uint16_t index) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetRecordingDevice(uint16_t index) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::PlayoutIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::InitPlayout() {
+  return -1;
+}
+
+bool AudioDeviceDummy::PlayoutIsInitialized() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::RecordingIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::InitRecording() {
+  return -1;
+}
+
+bool AudioDeviceDummy::RecordingIsInitialized() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::StartPlayout() {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StopPlayout() {
+  return 0;
+}
+
+bool AudioDeviceDummy::Playing() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::StartRecording() {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StopRecording() {
+  return 0;
+}
+
+bool AudioDeviceDummy::Recording() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::InitSpeaker() {
+  return -1;
+}
+
+bool AudioDeviceDummy::SpeakerIsInitialized() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::InitMicrophone() {
+  return -1;
+}
+
+bool AudioDeviceDummy::MicrophoneIsInitialized() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::SpeakerVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetSpeakerVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SpeakerVolume(uint32_t& volume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MinSpeakerVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MicrophoneVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetMicrophoneVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MicrophoneVolume(uint32_t& volume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MinMicrophoneVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SpeakerMuteIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetSpeakerMute(bool enable) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SpeakerMute(bool& enabled) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MicrophoneMuteIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetMicrophoneMute(bool enable) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MicrophoneMute(bool& enabled) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StereoPlayoutIsAvailable(bool& available) {
+  return -1;
+}
+int32_t AudioDeviceDummy::SetStereoPlayout(bool enable) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StereoPlayout(bool& enabled) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StereoRecordingIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetStereoRecording(bool enable) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StereoRecording(bool& enabled) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::PlayoutDelay(uint16_t& delayMS) const {
+  return -1;
+}
+
+void AudioDeviceDummy::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/dummy/audio_device_dummy.h b/third_party/libwebrtc/modules/audio_device/dummy/audio_device_dummy.h
new file mode 100644
index 0000000000..2a2541098e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/dummy/audio_device_dummy.h
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H_
+
+#include <stdint.h>
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioDeviceDummy : public AudioDeviceGeneric {
+ public:
+  AudioDeviceDummy() {}
+  virtual ~AudioDeviceDummy() {}
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H_
diff --git a/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device.cc b/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device.cc
new file mode 100644
index 0000000000..8c10ae4186
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device.cc
@@ -0,0 +1,508 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/dummy/file_audio_device.h"
+
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/sleep.h"
+
+namespace webrtc {
+
+const int kRecordingFixedSampleRate = 48000;
+const size_t kRecordingNumChannels = 2;
+const int kPlayoutFixedSampleRate = 48000;
+const size_t kPlayoutNumChannels = 2;
+const size_t kPlayoutBufferSize =
+    kPlayoutFixedSampleRate / 100 * kPlayoutNumChannels * 2;
+const size_t kRecordingBufferSize =
+    kRecordingFixedSampleRate / 100 * kRecordingNumChannels * 2;
+
+FileAudioDevice::FileAudioDevice(absl::string_view inputFilename,
+                                 absl::string_view outputFilename)
+    : _ptrAudioBuffer(NULL),
+      _recordingBuffer(NULL),
+      _playoutBuffer(NULL),
+      _recordingFramesLeft(0),
+      _playoutFramesLeft(0),
+      _recordingBufferSizeIn10MS(0),
+      _recordingFramesIn10MS(0),
+      _playoutFramesIn10MS(0),
+      _playing(false),
+      _recording(false),
+      _lastCallPlayoutMillis(0),
+      _lastCallRecordMillis(0),
+      _outputFilename(outputFilename),
+      _inputFilename(inputFilename) {}
+
+FileAudioDevice::~FileAudioDevice() {}
+
+int32_t FileAudioDevice::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  return -1;
+}
+
+AudioDeviceGeneric::InitStatus FileAudioDevice::Init() {
+  return InitStatus::OK;
+}
+
+int32_t FileAudioDevice::Terminate() {
+  return 0;
+}
+
+bool FileAudioDevice::Initialized() const {
+  return true;
+}
+
+int16_t FileAudioDevice::PlayoutDevices() {
+  return 1;
+}
+
+int16_t FileAudioDevice::RecordingDevices() {
+  return 1;
+}
+
+int32_t FileAudioDevice::PlayoutDeviceName(uint16_t index,
+                                           char name[kAdmMaxDeviceNameSize],
+                                           char guid[kAdmMaxGuidSize]) {
+  const char* kName = "dummy_device";
+  const char* kGuid = "dummy_device_unique_id";
+  if (index < 1) {
+    memset(name, 0, kAdmMaxDeviceNameSize);
+    memset(guid, 0, kAdmMaxGuidSize);
+    memcpy(name, kName, strlen(kName));
+    memcpy(guid, kGuid, strlen(guid));
+    return 0;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::RecordingDeviceName(uint16_t index,
+                                             char name[kAdmMaxDeviceNameSize],
+                                             char guid[kAdmMaxGuidSize]) {
+  const char* kName = "dummy_device";
+  const char* kGuid = "dummy_device_unique_id";
+  if (index < 1) {
+    memset(name, 0, kAdmMaxDeviceNameSize);
+    memset(guid, 0, kAdmMaxGuidSize);
+    memcpy(name, kName, strlen(kName));
+    memcpy(guid, kGuid, strlen(guid));
+    return 0;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::SetPlayoutDevice(uint16_t index) {
+  if (index == 0) {
+    _playout_index = index;
+    return 0;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetRecordingDevice(uint16_t index) {
+  if (index == 0) {
+    _record_index = index;
+    return _record_index;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t FileAudioDevice::PlayoutIsAvailable(bool& available) {
+  if (_playout_index == 0) {
+    available = true;
+    return _playout_index;
+  }
+  available = false;
+  return -1;
+}
+
+int32_t FileAudioDevice::InitPlayout() {
+  MutexLock lock(&mutex_);
+
+  if (_playing) {
+    return -1;
+  }
+
+  _playoutFramesIn10MS = static_cast<size_t>(kPlayoutFixedSampleRate / 100);
+
+  if (_ptrAudioBuffer) {
+    // Update webrtc audio buffer with the selected parameters
+    _ptrAudioBuffer->SetPlayoutSampleRate(kPlayoutFixedSampleRate);
+    _ptrAudioBuffer->SetPlayoutChannels(kPlayoutNumChannels);
+  }
+  return 0;
+}
+
+bool FileAudioDevice::PlayoutIsInitialized() const {
+  return _playoutFramesIn10MS != 0;
+}
+
+int32_t FileAudioDevice::RecordingIsAvailable(bool& available) {
+  if (_record_index == 0) {
+    available = true;
+    return _record_index;
+  }
+  available = false;
+  return -1;
+}
+
+int32_t FileAudioDevice::InitRecording() {
+  MutexLock lock(&mutex_);
+
+  if (_recording) {
+    return -1;
+  }
+
+  _recordingFramesIn10MS = static_cast<size_t>(kRecordingFixedSampleRate / 100);
+
+  if (_ptrAudioBuffer) {
+    _ptrAudioBuffer->SetRecordingSampleRate(kRecordingFixedSampleRate);
+    _ptrAudioBuffer->SetRecordingChannels(kRecordingNumChannels);
+  }
+  return 0;
+}
+
+bool FileAudioDevice::RecordingIsInitialized() const {
+  return _recordingFramesIn10MS != 0;
+}
+
+int32_t FileAudioDevice::StartPlayout() {
+  if (_playing) {
+    return 0;
+  }
+
+  _playing = true;
+  _playoutFramesLeft = 0;
+
+  if (!_playoutBuffer) {
+    _playoutBuffer = new int8_t[kPlayoutBufferSize];
+  }
+  if (!_playoutBuffer) {
+    _playing = false;
+    return -1;
+  }
+
+  // PLAYOUT
+  if (!_outputFilename.empty()) {
+    _outputFile = FileWrapper::OpenWriteOnly(_outputFilename);
+    if (!_outputFile.is_open()) {
+      RTC_LOG(LS_ERROR) << "Failed to open playout file: " << _outputFilename;
+      _playing = false;
+      delete[] _playoutBuffer;
+      _playoutBuffer = NULL;
+      return -1;
+    }
+  }
+
+  _ptrThreadPlay = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (PlayThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_play_thread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  RTC_LOG(LS_INFO) << "Started playout capture to output file: "
+                   << _outputFilename;
+  return 0;
+}
+
+int32_t FileAudioDevice::StopPlayout() {
+  {
+    MutexLock lock(&mutex_);
+    _playing = false;
+  }
+
+  // stop playout thread first
+  if (!_ptrThreadPlay.empty())
+    _ptrThreadPlay.Finalize();
+
+  MutexLock lock(&mutex_);
+
+  _playoutFramesLeft = 0;
+  delete[] _playoutBuffer;
+  _playoutBuffer = NULL;
+  _outputFile.Close();
+
+  RTC_LOG(LS_INFO) << "Stopped playout capture to output file: "
+                   << _outputFilename;
+  return 0;
+}
+
+bool FileAudioDevice::Playing() const {
+  return _playing;
+}
+
+int32_t FileAudioDevice::StartRecording() {
+  _recording = true;
+
+  // Make sure we only create the buffer once.
+  _recordingBufferSizeIn10MS =
+      _recordingFramesIn10MS * kRecordingNumChannels * 2;
+  if (!_recordingBuffer) {
+    _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
+  }
+
+  if (!_inputFilename.empty()) {
+    _inputFile = FileWrapper::OpenReadOnly(_inputFilename);
+    if (!_inputFile.is_open()) {
+      RTC_LOG(LS_ERROR) << "Failed to open audio input file: "
+                        << _inputFilename;
+      _recording = false;
+      delete[] _recordingBuffer;
+      _recordingBuffer = NULL;
+      return -1;
+    }
+  }
+
+  _ptrThreadRec = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (RecThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_capture_thread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  RTC_LOG(LS_INFO) << "Started recording from input file: " << _inputFilename;
+
+  return 0;
+}
+
+int32_t FileAudioDevice::StopRecording() {
+  {
+    MutexLock lock(&mutex_);
+    _recording = false;
+  }
+
+  if (!_ptrThreadRec.empty())
+    _ptrThreadRec.Finalize();
+
+  MutexLock lock(&mutex_);
+  _recordingFramesLeft = 0;
+  if (_recordingBuffer) {
+    delete[] _recordingBuffer;
+    _recordingBuffer = NULL;
+  }
+  _inputFile.Close();
+
+  RTC_LOG(LS_INFO) << "Stopped recording from input file: " << _inputFilename;
+  return 0;
+}
+
+bool FileAudioDevice::Recording() const {
+  return _recording;
+}
+
+int32_t FileAudioDevice::InitSpeaker() {
+  return -1;
+}
+
+bool FileAudioDevice::SpeakerIsInitialized() const {
+  return false;
+}
+
+int32_t FileAudioDevice::InitMicrophone() {
+  return 0;
+}
+
+bool FileAudioDevice::MicrophoneIsInitialized() const {
+  return true;
+}
+
+int32_t FileAudioDevice::SpeakerVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetSpeakerVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SpeakerVolume(uint32_t& volume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MinSpeakerVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetMicrophoneVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneVolume(uint32_t& volume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MinMicrophoneVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::SpeakerMuteIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetSpeakerMute(bool enable) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SpeakerMute(bool& enabled) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneMuteIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetMicrophoneMute(bool enable) {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneMute(bool& enabled) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::StereoPlayoutIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+int32_t FileAudioDevice::SetStereoPlayout(bool enable) {
+  return 0;
+}
+
+int32_t FileAudioDevice::StereoPlayout(bool& enabled) const {
+  enabled = true;
+  return 0;
+}
+
+int32_t FileAudioDevice::StereoRecordingIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+int32_t FileAudioDevice::SetStereoRecording(bool enable) {
+  return 0;
+}
+
+int32_t FileAudioDevice::StereoRecording(bool& enabled) const {
+  enabled = true;
+  return 0;
+}
+
+int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const {
+  return 0;
+}
+
+void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  MutexLock lock(&mutex_);
+
+  _ptrAudioBuffer = audioBuffer;
+
+  // Inform the AudioBuffer about default settings for this implementation.
+  // Set all values to zero here since the actual settings will be done by
+  // InitPlayout and InitRecording later.
+  _ptrAudioBuffer->SetRecordingSampleRate(0);
+  _ptrAudioBuffer->SetPlayoutSampleRate(0);
+  _ptrAudioBuffer->SetRecordingChannels(0);
+  _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+bool FileAudioDevice::PlayThreadProcess() {
+  if (!_playing) {
+    return false;
+  }
+  int64_t currentTime = rtc::TimeMillis();
+  mutex_.Lock();
+
+  if (_lastCallPlayoutMillis == 0 ||
+      currentTime - _lastCallPlayoutMillis >= 10) {
+    mutex_.Unlock();
+    _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
+    mutex_.Lock();
+
+    _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
+    RTC_DCHECK_EQ(_playoutFramesIn10MS, _playoutFramesLeft);
+    if (_outputFile.is_open()) {
+      _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
+    }
+    _lastCallPlayoutMillis = currentTime;
+  }
+  _playoutFramesLeft = 0;
+  mutex_.Unlock();
+
+  int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+  if (deltaTimeMillis < 10) {
+    SleepMs(10 - deltaTimeMillis);
+  }
+
+  return true;
+}
+
+bool FileAudioDevice::RecThreadProcess() {
+  if (!_recording) {
+    return false;
+  }
+
+  int64_t currentTime = rtc::TimeMillis();
+  mutex_.Lock();
+
+  if (_lastCallRecordMillis == 0 || currentTime - _lastCallRecordMillis >= 10) {
+    if (_inputFile.is_open()) {
+      if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) {
+        _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+                                           _recordingFramesIn10MS);
+      } else {
+        _inputFile.Rewind();
+      }
+      _lastCallRecordMillis = currentTime;
+      mutex_.Unlock();
+      _ptrAudioBuffer->DeliverRecordedData();
+      mutex_.Lock();
+    }
+  }
+
+  mutex_.Unlock();
+
+  int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+  if (deltaTimeMillis < 10) {
+    SleepMs(10 - deltaTimeMillis);
+  }
+
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device.h b/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device.h
new file mode 100644
index 0000000000..27979933f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device.h
@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
+#define AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
+
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+// This is a fake audio device which plays audio from a file as its microphone
+// and plays out into a file.
+class FileAudioDevice : public AudioDeviceGeneric {
+ public:
+  // Constructs a file audio device with `id`. It will read audio from
+  // `inputFilename` and record output audio to `outputFilename`.
+  //
+  // The input file should be a readable 48k stereo raw file, and the output
+  // file should point to a writable location. The output format will also be
+  // 48k stereo raw audio.
+  FileAudioDevice(absl::string_view inputFilename,
+                  absl::string_view outputFilename);
+  virtual ~FileAudioDevice();
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+
+ private:
+  static void RecThreadFunc(void*);
+  static void PlayThreadFunc(void*);
+  bool RecThreadProcess();
+  bool PlayThreadProcess();
+
+  int32_t _playout_index;
+  int32_t _record_index;
+  AudioDeviceBuffer* _ptrAudioBuffer;
+  int8_t* _recordingBuffer;  // In bytes.
+  int8_t* _playoutBuffer;    // In bytes.
+  uint32_t _recordingFramesLeft;
+  uint32_t _playoutFramesLeft;
+  Mutex mutex_;
+
+  size_t _recordingBufferSizeIn10MS;
+  size_t _recordingFramesIn10MS;
+  size_t _playoutFramesIn10MS;
+
+  rtc::PlatformThread _ptrThreadRec;
+  rtc::PlatformThread _ptrThreadPlay;
+
+  bool _playing;
+  bool _recording;
+  int64_t _lastCallPlayoutMillis;
+  int64_t _lastCallRecordMillis;
+
+  FileWrapper _outputFile;
+  FileWrapper _inputFile;
+  std::string _outputFilename;
+  std::string _inputFilename;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
diff --git a/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device_factory.cc b/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device_factory.cc
new file mode 100644
index 0000000000..8c41111478
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device_factory.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/dummy/file_audio_device_factory.h"
+
+#include <stdio.h>
+
+#include <cstdlib>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_device/dummy/file_audio_device.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+
+namespace webrtc {
+
+bool FileAudioDeviceFactory::_isConfigured = false;
+char FileAudioDeviceFactory::_inputAudioFilename[MAX_FILENAME_LEN] = "";
+char FileAudioDeviceFactory::_outputAudioFilename[MAX_FILENAME_LEN] = "";
+
+FileAudioDevice* FileAudioDeviceFactory::CreateFileAudioDevice() {
+  // Bail out here if the files haven't been set explicitly.
+  // audio_device_impl.cc should then fall back to dummy audio.
+  if (!_isConfigured) {
+    RTC_LOG(LS_WARNING)
+        << "WebRTC configured with WEBRTC_DUMMY_FILE_DEVICES but "
+           "no device files supplied. Will fall back to dummy "
+           "audio.";
+
+    return nullptr;
+  }
+  return new FileAudioDevice(_inputAudioFilename, _outputAudioFilename);
+}
+
+void FileAudioDeviceFactory::SetFilenamesToUse(
+    absl::string_view inputAudioFilename,
+    absl::string_view outputAudioFilename) {
+#ifdef WEBRTC_DUMMY_FILE_DEVICES
+  RTC_DCHECK_LT(inputAudioFilename.size(), MAX_FILENAME_LEN);
+  RTC_DCHECK_LT(outputAudioFilename.size(), MAX_FILENAME_LEN);
+
+  // Copy the strings since we don't know the lifetime of the input pointers.
+  rtc::strcpyn(_inputAudioFilename, MAX_FILENAME_LEN, inputAudioFilename);
+  rtc::strcpyn(_outputAudioFilename, MAX_FILENAME_LEN, outputAudioFilename);
+  _isConfigured = true;
+#else
+  // Sanity: must be compiled with the right define to run this.
+  printf(
+      "Trying to use dummy file devices, but is not compiled "
+      "with WEBRTC_DUMMY_FILE_DEVICES. Bailing out.\n");
+  std::exit(1);
+#endif
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device_factory.h b/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device_factory.h
new file mode 100644
index 0000000000..18f9388f21
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/dummy/file_audio_device_factory.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H_
+#define AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H_
+
+#include <stdint.h>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+class FileAudioDevice;
+
+// This class is used by audio_device_impl.cc when WebRTC is compiled with
+// WEBRTC_DUMMY_FILE_DEVICES. The application must include this file and set the
+// filenames to use before the audio device module is initialized. This is
+// intended for test tools which use the audio device module.
+class FileAudioDeviceFactory {
+ public:
+  static FileAudioDevice* CreateFileAudioDevice();
+
+  // The input file must be a readable 48k stereo raw file. The output
+  // file must be writable. The strings will be copied.
+  static void SetFilenamesToUse(absl::string_view inputAudioFilename,
+                                absl::string_view outputAudioFilename);
+
+ private:
+  enum : uint32_t { MAX_FILENAME_LEN = 512 };
+  static bool _isConfigured;
+  static char _inputAudioFilename[MAX_FILENAME_LEN];
+  static char _outputAudioFilename[MAX_FILENAME_LEN];
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc
new file mode 100644
index 0000000000..86240da196
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/fine_audio_buffer.h"
+
+#include <cstdint>
+#include <cstring>
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer)
+    : audio_device_buffer_(audio_device_buffer),
+      playout_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
+          audio_device_buffer->PlayoutSampleRate() * 10 / 1000)),
+      record_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
+          audio_device_buffer->RecordingSampleRate() * 10 / 1000)),
+      playout_channels_(audio_device_buffer->PlayoutChannels()),
+      record_channels_(audio_device_buffer->RecordingChannels()) {
+  RTC_DCHECK(audio_device_buffer_);
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  if (IsReadyForPlayout()) {
+    RTC_DLOG(LS_INFO) << "playout_samples_per_channel_10ms: "
+                      << playout_samples_per_channel_10ms_;
+    RTC_DLOG(LS_INFO) << "playout_channels: " << playout_channels_;
+  }
+  if (IsReadyForRecord()) {
+    RTC_DLOG(LS_INFO) << "record_samples_per_channel_10ms: "
+                      << record_samples_per_channel_10ms_;
+    RTC_DLOG(LS_INFO) << "record_channels: " << record_channels_;
+  }
+}
+
+FineAudioBuffer::~FineAudioBuffer() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+}
+
+void FineAudioBuffer::ResetPlayout() {
+  playout_buffer_.Clear();
+}
+
+void FineAudioBuffer::ResetRecord() {
+  record_buffer_.Clear();
+}
+
+bool FineAudioBuffer::IsReadyForPlayout() const {
+  return playout_samples_per_channel_10ms_ > 0 && playout_channels_ > 0;
+}
+
+bool FineAudioBuffer::IsReadyForRecord() const {
+  return record_samples_per_channel_10ms_ > 0 && record_channels_ > 0;
+}
+
+void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
+                                     int playout_delay_ms) {
+  RTC_DCHECK(IsReadyForPlayout());
+  // Ask WebRTC for new data in chunks of 10ms until we have enough to
+  // fulfill the request. It is possible that the buffer already contains
+  // enough samples from the last round.
+  while (playout_buffer_.size() < audio_buffer.size()) {
+    // Get 10ms decoded audio from WebRTC. The ADB knows about number of
+    // channels; hence we can ask for number of samples per channel here.
+    if (audio_device_buffer_->RequestPlayoutData(
+            playout_samples_per_channel_10ms_) ==
+        static_cast<int32_t>(playout_samples_per_channel_10ms_)) {
+      // Append 10ms to the end of the local buffer taking number of channels
+      // into account.
+      const size_t num_elements_10ms =
+          playout_channels_ * playout_samples_per_channel_10ms_;
+      const size_t written_elements = playout_buffer_.AppendData(
+          num_elements_10ms, [&](rtc::ArrayView<int16_t> buf) {
+            const size_t samples_per_channel_10ms =
+                audio_device_buffer_->GetPlayoutData(buf.data());
+            return playout_channels_ * samples_per_channel_10ms;
+          });
+      RTC_DCHECK_EQ(num_elements_10ms, written_elements);
+    } else {
+      // Provide silence if AudioDeviceBuffer::RequestPlayoutData() fails.
+      // Can e.g. happen when an AudioTransport has not been registered.
+      const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
+      std::memset(audio_buffer.data(), 0, num_bytes);
+      return;
+    }
+  }
+
+  // Provide the requested number of bytes to the consumer.
+  const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
+  memcpy(audio_buffer.data(), playout_buffer_.data(), num_bytes);
+  // Move remaining samples to start of buffer to prepare for next round.
+  memmove(playout_buffer_.data(), playout_buffer_.data() + audio_buffer.size(),
+          (playout_buffer_.size() - audio_buffer.size()) * sizeof(int16_t));
+  playout_buffer_.SetSize(playout_buffer_.size() - audio_buffer.size());
+  // Cache playout latency for usage in DeliverRecordedData();
+  playout_delay_ms_ = playout_delay_ms;
+}
+
+void FineAudioBuffer::DeliverRecordedData(
+    rtc::ArrayView<const int16_t> audio_buffer,
+    int record_delay_ms) {
+  RTC_DCHECK(IsReadyForRecord());
+  // Always append new data and grow the buffer when needed.
+  record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
+  // Consume samples from buffer in chunks of 10ms until there is not
+  // enough data left. The number of remaining samples in the cache is given by
+  // the new size of the internal `record_buffer_`.
+  const size_t num_elements_10ms =
+      record_channels_ * record_samples_per_channel_10ms_;
+  while (record_buffer_.size() >= num_elements_10ms) {
+    audio_device_buffer_->SetRecordedBuffer(record_buffer_.data(),
+                                            record_samples_per_channel_10ms_);
+    audio_device_buffer_->SetVQEData(playout_delay_ms_, record_delay_ms);
+    audio_device_buffer_->DeliverRecordedData();
+    memmove(record_buffer_.data(), record_buffer_.data() + num_elements_10ms,
+            (record_buffer_.size() - num_elements_10ms) * sizeof(int16_t));
+    record_buffer_.SetSize(record_buffer_.size() - num_elements_10ms);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h
new file mode 100644
index 0000000000..a6c3042bb2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
+#define MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
+
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+
+// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with 16-bit PCM
+// audio samples corresponding to 10ms of data. It then allows for this data
+// to be pulled in a finer or coarser granularity. I.e. interacting with this
+// class instead of directly with the AudioDeviceBuffer one can ask for any
+// number of audio data samples. This class also ensures that audio data can be
+// delivered to the ADB in 10ms chunks when the size of the provided audio
+// buffers differs from 10ms.
+// As an example: calling DeliverRecordedData() with 5ms buffers will deliver
+// accumulated 10ms worth of data to the ADB every second call.
+class FineAudioBuffer {
+ public:
+  // `device_buffer` is a buffer that provides 10ms of audio data.
+  FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer);
+  ~FineAudioBuffer();
+
+  // Clears buffers and counters dealing with playout and/or recording.
+  void ResetPlayout();
+  void ResetRecord();
+
+  // Utility methods which returns true if valid parameters are acquired at
+  // constructions.
+  bool IsReadyForPlayout() const;
+  bool IsReadyForRecord() const;
+
+  // Copies audio samples into `audio_buffer` where number of requested
+  // elements is specified by `audio_buffer.size()`. The producer will always
+  // fill up the audio buffer and if no audio exists, the buffer will contain
+  // silence instead. The provided delay estimate in `playout_delay_ms` should
+  // contain an estimate of the latency between when an audio frame is read from
+  // WebRTC and when it is played out on the speaker.
+  void GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
+                      int playout_delay_ms);
+
+  // Consumes the audio data in `audio_buffer` and sends it to the WebRTC layer
+  // in chunks of 10ms. The sum of the provided delay estimate in
+  // `record_delay_ms` and the latest `playout_delay_ms` in GetPlayoutData()
+  // are given to the AEC in the audio processing module.
+  // They can be fixed values on most platforms and they are ignored if an
+  // external (hardware/built-in) AEC is used.
+  // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
+  // 5ms of data and sends a total of 10ms to WebRTC and clears the internal
+  // cache. Call #3 restarts the scheme above.
+  void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
+                           int record_delay_ms);
+
+ private:
+  // Device buffer that works with 10ms chunks of data both for playout and
+  // for recording. I.e., the WebRTC side will always be asked for audio to be
+  // played out in 10ms chunks and recorded audio will be sent to WebRTC in
+  // 10ms chunks as well. This raw pointer is owned by the constructor of this
+  // class and the owner must ensure that the pointer is valid during the life-
+  // time of this object.
+  AudioDeviceBuffer* const audio_device_buffer_;
+  // Number of audio samples per channel per 10ms. Set once at construction
+  // based on parameters in `audio_device_buffer`.
+  const size_t playout_samples_per_channel_10ms_;
+  const size_t record_samples_per_channel_10ms_;
+  // Number of audio channels. Set once at construction based on parameters in
+  // `audio_device_buffer`.
+  const size_t playout_channels_;
+  const size_t record_channels_;
+  // Storage for output samples from which a consumer can read audio buffers
+  // in any size using GetPlayoutData().
+  rtc::BufferT<int16_t> playout_buffer_;
+  // Storage for input samples that are about to be delivered to the WebRTC
+  // ADB or remains from the last successful delivery of a 10ms audio buffer.
+  rtc::BufferT<int16_t> record_buffer_;
+  // Contains latest delay estimate given to GetPlayoutData().
+  int playout_delay_ms_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc
new file mode 100644
index 0000000000..36ea85f7dd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc
@@ -0,0 +1,158 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/fine_audio_buffer.h"
+
+#include <limits.h>
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "modules/audio_device/mock_audio_device_buffer.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::InSequence;
+using ::testing::Return;
+
+namespace webrtc {
+
+const int kSampleRate = 44100;
+const int kChannels = 2;
+const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
+
+// The fake audio data is 0,1,..SCHAR_MAX-1,0,1,... This is to make it easy
+// to detect errors. This function verifies that the buffers contain such data.
+// E.g. if there are two buffers of size 3, buffer 1 would contain 0,1,2 and
+// buffer 2 would contain 3,4,5. Note that SCHAR_MAX is 127 so wrap-around
+// will happen.
+// `buffer` is the audio buffer to verify.
+bool VerifyBuffer(const int16_t* buffer, int buffer_number, int size) {
+  int start_value = (buffer_number * size) % SCHAR_MAX;
+  for (int i = 0; i < size; ++i) {
+    if (buffer[i] != (i + start_value) % SCHAR_MAX) {
+      return false;
+    }
+  }
+  return true;
+}
+
+// This function replaces the real AudioDeviceBuffer::GetPlayoutData when it's
+// called (which is done implicitly when calling GetBufferData). It writes the
+// sequence 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a
+// buffer of different size than the one VerifyBuffer verifies.
+// `iteration` is the number of calls made to UpdateBuffer prior to this call.
+// `samples_per_10_ms` is the number of samples that should be written to the
+// buffer (`arg0`).
+ACTION_P2(UpdateBuffer, iteration, samples_per_10_ms) {
+  int16_t* buffer = static_cast<int16_t*>(arg0);
+  int start_value = (iteration * samples_per_10_ms) % SCHAR_MAX;
+  for (int i = 0; i < samples_per_10_ms; ++i) {
+    buffer[i] = (i + start_value) % SCHAR_MAX;
+  }
+  // Should return samples per channel.
+  return samples_per_10_ms / kChannels;
+}
+
+// Writes a periodic ramp pattern to the supplied `buffer`. See UpdateBuffer()
+// for details.
+void UpdateInputBuffer(int16_t* buffer, int iteration, int size) {
+  int start_value = (iteration * size) % SCHAR_MAX;
+  for (int i = 0; i < size; ++i) {
+    buffer[i] = (i + start_value) % SCHAR_MAX;
+  }
+}
+
+// Action macro which verifies that the recorded 10ms chunk of audio data
+// (in `arg0`) contains the correct reference values even if they have been
+// supplied using a buffer size that is smaller or larger than 10ms.
+// See VerifyBuffer() for details.
+ACTION_P2(VerifyInputBuffer, iteration, samples_per_10_ms) {
+  const int16_t* buffer = static_cast<const int16_t*>(arg0);
+  int start_value = (iteration * samples_per_10_ms) % SCHAR_MAX;
+  for (int i = 0; i < samples_per_10_ms; ++i) {
+    EXPECT_EQ(buffer[i], (i + start_value) % SCHAR_MAX);
+  }
+  return 0;
+}
+
+void RunFineBufferTest(int frame_size_in_samples) {
+  const int kFrameSizeSamples = frame_size_in_samples;
+  const int kNumberOfFrames = 5;
+  // Ceiling of integer division: 1 + ((x - 1) / y)
+  const int kNumberOfUpdateBufferCalls =
+      1 + ((kNumberOfFrames * frame_size_in_samples - 1) / kSamplesPer10Ms);
+
+  auto task_queue_factory = CreateDefaultTaskQueueFactory();
+  MockAudioDeviceBuffer audio_device_buffer(task_queue_factory.get());
+  audio_device_buffer.SetPlayoutSampleRate(kSampleRate);
+  audio_device_buffer.SetPlayoutChannels(kChannels);
+  audio_device_buffer.SetRecordingSampleRate(kSampleRate);
+  audio_device_buffer.SetRecordingChannels(kChannels);
+
+  EXPECT_CALL(audio_device_buffer, RequestPlayoutData(_))
+      .WillRepeatedly(Return(kSamplesPer10Ms));
+  {
+    InSequence s;
+    for (int i = 0; i < kNumberOfUpdateBufferCalls; ++i) {
+      EXPECT_CALL(audio_device_buffer, GetPlayoutData(_))
+          .WillOnce(UpdateBuffer(i, kChannels * kSamplesPer10Ms))
+          .RetiresOnSaturation();
+    }
+  }
+  {
+    InSequence s;
+    for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
+      EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms))
+          .WillOnce(VerifyInputBuffer(j, kChannels * kSamplesPer10Ms))
+          .RetiresOnSaturation();
+    }
+  }
+  EXPECT_CALL(audio_device_buffer, SetVQEData(_, _))
+      .Times(kNumberOfUpdateBufferCalls - 1);
+  EXPECT_CALL(audio_device_buffer, DeliverRecordedData())
+      .Times(kNumberOfUpdateBufferCalls - 1)
+      .WillRepeatedly(Return(0));
+
+  FineAudioBuffer fine_buffer(&audio_device_buffer);
+  std::unique_ptr<int16_t[]> out_buffer(
+      new int16_t[kChannels * kFrameSizeSamples]);
+  std::unique_ptr<int16_t[]> in_buffer(
+      new int16_t[kChannels * kFrameSizeSamples]);
+
+  for (int i = 0; i < kNumberOfFrames; ++i) {
+    fine_buffer.GetPlayoutData(
+        rtc::ArrayView<int16_t>(out_buffer.get(),
+                                kChannels * kFrameSizeSamples),
+        0);
+    EXPECT_TRUE(
+        VerifyBuffer(out_buffer.get(), i, kChannels * kFrameSizeSamples));
+    UpdateInputBuffer(in_buffer.get(), i, kChannels * kFrameSizeSamples);
+    fine_buffer.DeliverRecordedData(
+        rtc::ArrayView<const int16_t>(in_buffer.get(),
+                                      kChannels * kFrameSizeSamples),
+        0);
+  }
+}
+
+TEST(FineBufferTest, BufferLessThan10ms) {
+  const int kFrameSizeSamples = kSamplesPer10Ms - 50;
+  RunFineBufferTest(kFrameSizeSamples);
+}
+
+TEST(FineBufferTest, GreaterThan10ms) {
+  const int kFrameSizeSamples = kSamplesPer10Ms + 50;
+  RunFineBufferTest(kFrameSizeSamples);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/g3doc/audio_device_module.md b/third_party/libwebrtc/modules/audio_device/g3doc/audio_device_module.md
new file mode 100644
index 0000000000..e325faacad
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/g3doc/audio_device_module.md
@@ -0,0 +1,171 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'henrika' reviewed: '2021-04-12'} *-->
+
+# Audio Device Module (ADM)
+
+## Overview
+
+The ADM is responsible for driving input (microphone) and output (speaker) audio
+in WebRTC and the API is defined in [audio_device.h][19].
+
+Main functions of the ADM are:
+
+*   Initialization and termination of native audio libraries.
+*   Registration of an [AudioTransport object][16] which handles audio callbacks
+    for audio in both directions.
+*   Device enumeration and selection (only for Linux, Windows and Mac OSX).
+*   Start/Stop physical audio streams:
+    *   Recording audio from the selected microphone, and
+    *   playing out audio on the selected speaker.
+*   Level control of the active audio streams.
+*   Control of built-in audio effects (Audio Echo Cancelation (AEC), Audio Gain
+    Control (AGC) and Noise Suppression (NS)) for Android and iOS.
+
+ADM implementations reside at two different locations in the WebRTC repository:
+`/modules/audio_device/` and `/sdk/`. The latest implementations for [iOS][20]
+and [Android][21] can be found under `/sdk/`. `/modules/audio_device/` contains
+older versions for mobile platforms and also implementations for desktop
+platforms such as [Linux][22], [Windows][23] and [Mac OSX][24]. This document is
+focusing on the parts in `/modules/audio_device/` but implementation specific
+details such as threading models are omitted to keep the descriptions as simple
+as possible.
+
+By default, the ADM in WebRTC is created in [`WebRtcVoiceEngine::Init`][1] but
+an external implementation can also be injected using
+[`rtc::CreatePeerConnectionFactory`][25]. An example of where an external ADM is
+injected can be found in [PeerConnectionInterfaceTest][26] where a so-called
+[fake ADM][29] is utilized to avoid hardware dependency in a gtest. Clients can
+also inject their own ADMs in situations where functionality is needed that is
+not provided by the default implementations.
+
+## Background
+
+This section contains a historical background of the ADM API.
+
+The ADM interface is old and has undergone many changes over the years. It used
+to be much more granular but it still contains more than 50 methods and is
+implemented on several different hardware platforms.
+
+Some APIs are not implemented on all platforms, and functionality can be spread
+out differently between the methods.
+
+The most up-to-date implementations of the ADM interface are for [iOS][27] and
+for [Android][28].
+
+Desktop version are not updated to comply with the latest
+[C++ style guide](https://chromium.googlesource.com/chromium/src/+/main/styleguide/c++/c++.md)
+and more work is also needed to improve the performance and stability of these
+versions.
+
+## WebRtcVoiceEngine
+
+[`WebRtcVoiceEngine`][2] does not utilize all methods of the ADM but it still
+serves as the best example of its architecture and how to use it. For a more
+detailed view of all methods in the ADM interface, see [ADM unit tests][3].
+
+Assuming that an external ADM implementation is not injected, a default - or
+internal - ADM is created in [`WebRtcVoiceEngine::Init`][1] using
+[`AudioDeviceModule::Create`][4].
+
+Basic initialization is done using a utility method called
+[`adm_helpers::Init`][5] which calls fundamental ADM APIs like:
+
+*   [`AudiDeviceModule::Init`][6] - initializes the native audio parts required
+    for each platform.
+*   [`AudiDeviceModule::SetPlayoutDevice`][7] - specifies which speaker to use
+    for playing out audio using an `index` retrieved by the corresponding
+    enumeration method [`AudiDeviceModule::PlayoutDeviceName`][8].
+*   [`AudiDeviceModule::SetRecordingDevice`][9] - specifies which microphone to
+    use for recording audio using an `index` retrieved by the corresponding
+    enumeration method which is [`AudiDeviceModule::RecordingDeviceName`][10].
+*   [`AudiDeviceModule::InitSpeaker`][11] - sets up the parts of the ADM needed
+    to use the selected output device.
+*   [`AudiDeviceModule::InitMicrophone`][12] - sets up the parts of the ADM
+    needed to use the selected input device.
+*   [`AudiDeviceModule::SetStereoPlayout`][13] - enables playout in stereo if
+    the selected audio device supports it.
+*   [`AudiDeviceModule::SetStereoRecording`][14] - enables recording in stereo
+    if the selected audio device supports it.
+
+[`WebRtcVoiceEngine::Init`][1] also calls
+[`AudiDeviceModule::RegisterAudioTransport`][15] to register an existing
+[AudioTransport][16] implementation which handles audio callbacks in both
+directions and therefore serves as the bridge between the native ADM and the
+upper WebRTC layers.
+
+Recorded audio samples are delivered from the ADM to the `WebRtcVoiceEngine`
+(who owns the `AudioTransport` object) via
+[`AudioTransport::RecordedDataIsAvailable`][17]:
+
+```
+int32_t RecordedDataIsAvailable(const void* audioSamples, size_t nSamples, size_t nBytesPerSample,
+                                size_t nChannels, uint32_t samplesPerSec, uint32_t totalDelayMS,
+                                int32_t clockDrift, uint32_t currentMicLevel, bool keyPressed,
+                                uint32_t& newMicLevel)
+```
+
+Decoded audio samples ready to be played out are are delivered by the
+`WebRtcVoiceEngine` to the ADM, via [`AudioTransport::NeedMorePlayoutData`][18]:
+
+```
+int32_t NeedMorePlayData(size_t nSamples, size_t nBytesPerSample, size_t nChannels, int32_t samplesPerSec,
+                         void* audioSamples, size_t& nSamplesOut,
+                         int64_t* elapsed_time_ms, int64_t* ntp_time_ms)
+```
+
+Audio samples are 16-bit [linear PCM](https://wiki.multimedia.cx/index.php/PCM)
+using regular interleaving of channels within each sample.
+
+`WebRtcVoiceEngine` also owns an [`AudioState`][30] member and this class is
+used has helper to start and stop audio to and from the ADM. To initialize and
+start recording, it calls:
+
+*   [`AudiDeviceModule::InitRecording`][31]
+*   [`AudiDeviceModule::StartRecording`][32]
+
+and to initialize and start playout:
+
+*   [`AudiDeviceModule::InitPlayout`][33]
+*   [`AudiDeviceModule::StartPlayout`][34]
+
+Finally, the corresponding stop methods [`AudiDeviceModule::StopRecording`][35]
+and [`AudiDeviceModule::StopPlayout`][36] are called followed by
+[`AudiDeviceModule::Terminate`][37].
+
+[1]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/media/engine/webrtc_voice_engine.cc;l=314;drc=f7b1b95f11c74cb5369fdd528b73c70a50f2e206
+[2]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/media/engine/webrtc_voice_engine.h;l=48;drc=d15a575ec3528c252419149d35977e55269d8a41
+[3]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/audio_device_unittest.cc;l=1;drc=d15a575ec3528c252419149d35977e55269d8a41
+[4]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=46;drc=eb8c4ca608486add9800f6bfb7a8ba3cf23e738e
+[5]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/media/engine/adm_helpers.h;drc=2222a80e79ae1ef5cb9510ec51d3868be75f47a2
+[6]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=62;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[7]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=77;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[8]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=69;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[9]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=79;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[10]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=72;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[11]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=99;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[12]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=101;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[13]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=130;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[14]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=133;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[15]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=59;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[16]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device_defines.h;l=34;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[17]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device_defines.h;l=36;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[18]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device_defines.h;l=48;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[19]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;drc=eb8c4ca608486add9800f6bfb7a8ba3cf23e738es
+[20]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/objc/native/api/audio_device_module.h;drc=76443eafa9375374d9f1d23da2b913f2acac6ac2
+[21]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/src/jni/audio_device/audio_device_module.h;drc=bbeb10925eb106eeed6143ccf571bc438ec22ce1
+[22]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/linux/;drc=d15a575ec3528c252419149d35977e55269d8a41
+[23]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/win/;drc=d15a575ec3528c252419149d35977e55269d8a41
+[24]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/mac/;drc=3b68aa346a5d3483c3448852d19d91723846825c
+[25]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/create_peerconnection_factory.h;l=45;drc=09ceed2165137c4bea4e02e8d3db31970d0bf273
+[26]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/peer_connection_interface_unittest.cc;l=692;drc=2efb8a5ec61b1b87475d046c03d20244f53b14b6
+[27]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/objc/native/api/audio_device_module.h;drc=76443eafa9375374d9f1d23da2b913f2acac6ac2
+[28]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/sdk/android/src/jni/audio_device/audio_device_module.h;drc=bbeb10925eb106eeed6143ccf571bc438ec22ce1
+[29]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/test/fake_audio_capture_module.h;l=42;drc=d15a575ec3528c252419149d35977e55269d8a41
+[30]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/audio/audio_state.h;drc=d15a575ec3528c252419149d35977e55269d8a41
+[31]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=87;drc=eb8c4ca608486add9800f6bfb7a8ba3cf23e738e
+[32]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=94;drc=eb8c4ca608486add9800f6bfb7a8ba3cf23e738e
+[33]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=84;drc=eb8c4ca608486add9800f6bfb7a8ba3cf23e738e
+[34]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=91;drc=eb8c4ca608486add9800f6bfb7a8ba3cf23e738e
+[35]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=95;drc=eb8c4ca608486add9800f6bfb7a8ba3cf23e738e
+[36]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=92;drc=eb8c4ca608486add9800f6bfb7a8ba3cf23e738e
+[37]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_device/include/audio_device.h;l=63;drc=eb8c4ca608486add9800f6bfb7a8ba3cf23e738e
diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device.h b/third_party/libwebrtc/modules/audio_device/include/audio_device.h
new file mode 100644
index 0000000000..936ee6cb04
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/audio_device.h
@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+class AudioDeviceModuleForTest;
+
+class AudioDeviceModule : public rtc::RefCountInterface {
+ public:
+  enum AudioLayer {
+    kPlatformDefaultAudio = 0,
+    kWindowsCoreAudio,
+    kWindowsCoreAudio2,
+    kLinuxAlsaAudio,
+    kLinuxPulseAudio,
+    kAndroidJavaAudio,
+    kAndroidOpenSLESAudio,
+    kAndroidJavaInputAndOpenSLESOutputAudio,
+    kAndroidAAudioAudio,
+    kAndroidJavaInputAndAAudioOutputAudio,
+    kDummyAudio,
+  };
+
+  enum WindowsDeviceType {
+    kDefaultCommunicationDevice = -1,
+    kDefaultDevice = -2
+  };
+
+  struct Stats {
+    // The fields below correspond to similarly-named fields in the WebRTC stats
+    // spec. https://w3c.github.io/webrtc-stats/#playoutstats-dict*
+    double synthesized_samples_duration_s = 0;
+    uint64_t synthesized_samples_events = 0;
+    double total_samples_duration_s = 0;
+    double total_playout_delay_s = 0;
+    uint64_t total_samples_count = 0;
+  };
+
+ public:
+  // Creates a default ADM for usage in production code.
+  static rtc::scoped_refptr<AudioDeviceModule> Create(
+      AudioLayer audio_layer,
+      TaskQueueFactory* task_queue_factory);
+  // Creates an ADM with support for extra test methods. Don't use this factory
+  // in production code.
+  static rtc::scoped_refptr<AudioDeviceModuleForTest> CreateForTest(
+      AudioLayer audio_layer,
+      TaskQueueFactory* task_queue_factory);
+
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
+
+  // Full-duplex transportation of PCM audio
+  virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
+
+  // Main initialization and termination
+  virtual int32_t Init() = 0;
+  virtual int32_t Terminate() = 0;
+  virtual bool Initialized() const = 0;
+
+  // Device enumeration
+  virtual int16_t PlayoutDevices() = 0;
+  virtual int16_t RecordingDevices() = 0;
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]) = 0;
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]) = 0;
+
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
+  virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
+  virtual int32_t SetRecordingDevice(uint16_t index) = 0;
+  virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
+
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool* available) = 0;
+  virtual int32_t InitPlayout() = 0;
+  virtual bool PlayoutIsInitialized() const = 0;
+  virtual int32_t RecordingIsAvailable(bool* available) = 0;
+  virtual int32_t InitRecording() = 0;
+  virtual bool RecordingIsInitialized() const = 0;
+
+  // Audio transport control
+  virtual int32_t StartPlayout() = 0;
+  virtual int32_t StopPlayout() = 0;
+  virtual bool Playing() const = 0;
+  virtual int32_t StartRecording() = 0;
+  virtual int32_t StopRecording() = 0;
+  virtual bool Recording() const = 0;
+
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker() = 0;
+  virtual bool SpeakerIsInitialized() const = 0;
+  virtual int32_t InitMicrophone() = 0;
+  virtual bool MicrophoneIsInitialized() const = 0;
+
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
+  virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
+  virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
+  virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
+  virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
+
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
+  virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
+  virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
+  virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
+  virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
+
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
+  virtual int32_t SetSpeakerMute(bool enable) = 0;
+  virtual int32_t SpeakerMute(bool* enabled) const = 0;
+
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
+  virtual int32_t SetMicrophoneMute(bool enable) = 0;
+  virtual int32_t MicrophoneMute(bool* enabled) const = 0;
+
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
+  virtual int32_t SetStereoPlayout(bool enable) = 0;
+  virtual int32_t StereoPlayout(bool* enabled) const = 0;
+  virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
+  virtual int32_t SetStereoRecording(bool enable) = 0;
+  virtual int32_t StereoRecording(bool* enabled) const = 0;
+
+  // Playout delay
+  virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
+
+  // Only supported on Android.
+  virtual bool BuiltInAECIsAvailable() const = 0;
+  virtual bool BuiltInAGCIsAvailable() const = 0;
+  virtual bool BuiltInNSIsAvailable() const = 0;
+
+  // Enables the built-in audio effects. Only supported on Android.
+  virtual int32_t EnableBuiltInAEC(bool enable) = 0;
+  virtual int32_t EnableBuiltInAGC(bool enable) = 0;
+  virtual int32_t EnableBuiltInNS(bool enable) = 0;
+
+  // Play underrun count. Only supported on Android.
+  // TODO(alexnarest): Make it abstract after upstream projects support it.
+  virtual int32_t GetPlayoutUnderrunCount() const { return -1; }
+
+  // Used to generate RTC stats. If not implemented, RTCAudioPlayoutStats will
+  // not be present in the stats.
+  virtual absl::optional<Stats> GetStats() const { return absl::nullopt; }
+
+// Only supported on iOS.
+#if defined(WEBRTC_IOS)
+  virtual int GetPlayoutAudioParameters(AudioParameters* params) const = 0;
+  virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
+#endif  // WEBRTC_IOS
+
+ protected:
+  ~AudioDeviceModule() override {}
+};
+
+// Extends the default ADM interface with some extra test methods.
+// Intended for usage in tests only and requires a unique factory method.
+class AudioDeviceModuleForTest : public AudioDeviceModule {
+ public:
+  // Triggers internal restart sequences of audio streaming. Can be used by
+  // tests to emulate events corresponding to e.g. removal of an active audio
+  // device or other actions which causes the stream to be disconnected.
+  virtual int RestartPlayoutInternally() = 0;
+  virtual int RestartRecordingInternally() = 0;
+
+  virtual int SetPlayoutSampleRate(uint32_t sample_rate) = 0;
+  virtual int SetRecordingSampleRate(uint32_t sample_rate) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device_data_observer.h b/third_party/libwebrtc/modules/audio_device/include/audio_device_data_observer.h
new file mode 100644
index 0000000000..36dc45f19e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/audio_device_data_observer.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DATA_OBSERVER_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DATA_OBSERVER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+
+// This interface will capture the raw PCM data of both the local captured as
+// well as the mixed/rendered remote audio.
+class AudioDeviceDataObserver {
+ public:
+  virtual void OnCaptureData(const void* audio_samples,
+                             size_t num_samples,
+                             size_t bytes_per_sample,
+                             size_t num_channels,
+                             uint32_t samples_per_sec) = 0;
+
+  virtual void OnRenderData(const void* audio_samples,
+                            size_t num_samples,
+                            size_t bytes_per_sample,
+                            size_t num_channels,
+                            uint32_t samples_per_sec) = 0;
+
+  AudioDeviceDataObserver() = default;
+  virtual ~AudioDeviceDataObserver() = default;
+};
+
+// Creates an ADMWrapper around an ADM instance that registers
+// the provided AudioDeviceDataObserver.
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    rtc::scoped_refptr<AudioDeviceModule> impl,
+    std::unique_ptr<AudioDeviceDataObserver> observer);
+
+// Creates an ADMWrapper around an ADM instance that registers
+// the provided AudioDeviceDataObserver.
+ABSL_DEPRECATED("")
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    rtc::scoped_refptr<AudioDeviceModule> impl,
+    AudioDeviceDataObserver* observer);
+
+// Creates an ADM instance with AudioDeviceDataObserver registered.
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    AudioDeviceModule::AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory,
+    std::unique_ptr<AudioDeviceDataObserver> observer);
+
+// Creates an ADM instance with AudioDeviceDataObserver registered.
+ABSL_DEPRECATED("")
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    AudioDeviceModule::AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory,
+    AudioDeviceDataObserver* observer);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DATA_OBSERVER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device_default.h b/third_party/libwebrtc/modules/audio_device/include/audio_device_default.h
new file mode 100644
index 0000000000..3779d6fb3b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/audio_device_default.h
@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
+
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+namespace webrtc_impl {
+
+// AudioDeviceModuleDefault template adds default implementation for all
+// AudioDeviceModule methods to the class, which inherits from
+// AudioDeviceModuleDefault<T>.
+template <typename T>
+class AudioDeviceModuleDefault : public T {
+ public:
+  AudioDeviceModuleDefault() {}
+  virtual ~AudioDeviceModuleDefault() {}
+
+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
+    return 0;
+  }
+  int32_t Init() override { return 0; }
+  int32_t InitSpeaker() override { return 0; }
+  int32_t SetPlayoutDevice(uint16_t index) override { return 0; }
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    return 0;
+  }
+  int32_t SetStereoPlayout(bool enable) override { return 0; }
+  int32_t StopPlayout() override { return 0; }
+  int32_t InitMicrophone() override { return 0; }
+  int32_t SetRecordingDevice(uint16_t index) override { return 0; }
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    return 0;
+  }
+  int32_t SetStereoRecording(bool enable) override { return 0; }
+  int32_t StopRecording() override { return 0; }
+
+  int32_t Terminate() override { return 0; }
+
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer* audioLayer) const override {
+    return 0;
+  }
+  bool Initialized() const override { return true; }
+  int16_t PlayoutDevices() override { return 0; }
+  int16_t RecordingDevices() override { return 0; }
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override {
+    return 0;
+  }
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override {
+    return 0;
+  }
+  int32_t PlayoutIsAvailable(bool* available) override { return 0; }
+  int32_t InitPlayout() override { return 0; }
+  bool PlayoutIsInitialized() const override { return true; }
+  int32_t RecordingIsAvailable(bool* available) override { return 0; }
+  int32_t InitRecording() override { return 0; }
+  bool RecordingIsInitialized() const override { return true; }
+  int32_t StartPlayout() override { return 0; }
+  bool Playing() const override { return false; }
+  int32_t StartRecording() override { return 0; }
+  bool Recording() const override { return false; }
+  bool SpeakerIsInitialized() const override { return true; }
+  bool MicrophoneIsInitialized() const override { return true; }
+  int32_t SpeakerVolumeIsAvailable(bool* available) override { return 0; }
+  int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
+  int32_t SpeakerVolume(uint32_t* volume) const override { return 0; }
+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override { return 0; }
+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override { return 0; }
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override { return 0; }
+  int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
+  int32_t MicrophoneVolume(uint32_t* volume) const override { return 0; }
+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override { return 0; }
+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override { return 0; }
+  int32_t SpeakerMuteIsAvailable(bool* available) override { return 0; }
+  int32_t SetSpeakerMute(bool enable) override { return 0; }
+  int32_t SpeakerMute(bool* enabled) const override { return 0; }
+  int32_t MicrophoneMuteIsAvailable(bool* available) override { return 0; }
+  int32_t SetMicrophoneMute(bool enable) override { return 0; }
+  int32_t MicrophoneMute(bool* enabled) const override { return 0; }
+  int32_t StereoPlayoutIsAvailable(bool* available) const override {
+    *available = false;
+    return 0;
+  }
+  int32_t StereoPlayout(bool* enabled) const override { return 0; }
+  int32_t StereoRecordingIsAvailable(bool* available) const override {
+    *available = false;
+    return 0;
+  }
+  int32_t StereoRecording(bool* enabled) const override { return 0; }
+  int32_t PlayoutDelay(uint16_t* delayMS) const override {
+    *delayMS = 0;
+    return 0;
+  }
+  bool BuiltInAECIsAvailable() const override { return false; }
+  int32_t EnableBuiltInAEC(bool enable) override { return -1; }
+  bool BuiltInAGCIsAvailable() const override { return false; }
+  int32_t EnableBuiltInAGC(bool enable) override { return -1; }
+  bool BuiltInNSIsAvailable() const override { return false; }
+  int32_t EnableBuiltInNS(bool enable) override { return -1; }
+
+  int32_t GetPlayoutUnderrunCount() const override { return -1; }
+
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override {
+    return -1;
+  }
+  int GetRecordAudioParameters(AudioParameters* params) const override {
+    return -1;
+  }
+#endif  // WEBRTC_IOS
+};
+
+}  // namespace webrtc_impl
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device_defines.h b/third_party/libwebrtc/modules/audio_device/include/audio_device_defines.h
new file mode 100644
index 0000000000..d677d41f69
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/audio_device_defines.h
@@ -0,0 +1,177 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+
+#include <stddef.h>
+
+#include <string>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+static const int kAdmMaxDeviceNameSize = 128;
+static const int kAdmMaxFileNameSize = 512;
+static const int kAdmMaxGuidSize = 128;
+
+static const int kAdmMinPlayoutBufferSizeMs = 10;
+static const int kAdmMaxPlayoutBufferSizeMs = 250;
+
+// ----------------------------------------------------------------------------
+//  AudioTransport
+// ----------------------------------------------------------------------------
+
+class AudioTransport {
+ public:
+  // TODO(bugs.webrtc.org/13620) Deprecate this function
+  virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                          size_t nSamples,
+                                          size_t nBytesPerSample,
+                                          size_t nChannels,
+                                          uint32_t samplesPerSec,
+                                          uint32_t totalDelayMS,
+                                          int32_t clockDrift,
+                                          uint32_t currentMicLevel,
+                                          bool keyPressed,
+                                          uint32_t& newMicLevel) = 0;  // NOLINT
+
+  virtual int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t nSamples,
+      size_t nBytesPerSample,
+      size_t nChannels,
+      uint32_t samplesPerSec,
+      uint32_t totalDelayMS,
+      int32_t clockDrift,
+      uint32_t currentMicLevel,
+      bool keyPressed,
+      uint32_t& newMicLevel,
+      absl::optional<int64_t> estimatedCaptureTimeNS) {  // NOLINT
+    // TODO(webrtc:13620) Make the default behaver of the new API to behave as
+    // the old API. This can be pure virtual if all uses of the old API is
+    // removed.
+    return RecordedDataIsAvailable(
+        audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
+        totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
+  }
+
+  // Implementation has to setup safe values for all specified out parameters.
+  virtual int32_t NeedMorePlayData(size_t nSamples,
+                                   size_t nBytesPerSample,
+                                   size_t nChannels,
+                                   uint32_t samplesPerSec,
+                                   void* audioSamples,
+                                   size_t& nSamplesOut,  // NOLINT
+                                   int64_t* elapsed_time_ms,
+                                   int64_t* ntp_time_ms) = 0;  // NOLINT
+
+  // Method to pull mixed render audio data from all active VoE channels.
+  // The data will not be passed as reference for audio processing internally.
+  virtual void PullRenderData(int bits_per_sample,
+                              int sample_rate,
+                              size_t number_of_channels,
+                              size_t number_of_frames,
+                              void* audio_data,
+                              int64_t* elapsed_time_ms,
+                              int64_t* ntp_time_ms) = 0;
+
+ protected:
+  virtual ~AudioTransport() {}
+};
+
+// Helper class for storage of fundamental audio parameters such as sample rate,
+// number of channels, native buffer size etc.
+// Note that one audio frame can contain more than one channel sample and each
+// sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
+// stereo contains 2 * (16/8) = 4 bytes of data.
+class AudioParameters {
+ public:
+  // This implementation does only support 16-bit PCM samples.
+  static const size_t kBitsPerSample = 16;
+  AudioParameters()
+      : sample_rate_(0),
+        channels_(0),
+        frames_per_buffer_(0),
+        frames_per_10ms_buffer_(0) {}
+  AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
+      : sample_rate_(sample_rate),
+        channels_(channels),
+        frames_per_buffer_(frames_per_buffer),
+        frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
+  void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
+    sample_rate_ = sample_rate;
+    channels_ = channels;
+    frames_per_buffer_ = frames_per_buffer;
+    frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
+  }
+  size_t bits_per_sample() const { return kBitsPerSample; }
+  void reset(int sample_rate, size_t channels, double buffer_duration) {
+    reset(sample_rate, channels,
+          static_cast<size_t>(sample_rate * buffer_duration + 0.5));
+  }
+  void reset(int sample_rate, size_t channels) {
+    reset(sample_rate, channels, static_cast<size_t>(0));
+  }
+  int sample_rate() const { return sample_rate_; }
+  size_t channels() const { return channels_; }
+  size_t frames_per_buffer() const { return frames_per_buffer_; }
+  size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
+  size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
+  size_t GetBytesPerBuffer() const {
+    return frames_per_buffer_ * GetBytesPerFrame();
+  }
+  // The WebRTC audio device buffer (ADB) only requires that the sample rate
+  // and number of channels are configured. Hence, to be "valid", only these
+  // two attributes must be set.
+  bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); }
+  // Most platforms also require that a native buffer size is defined.
+  // An audio parameter instance is considered to be "complete" if it is both
+  // "valid" (can be used by the ADB) and also has a native frame size.
+  bool is_complete() const { return (is_valid() && (frames_per_buffer_ > 0)); }
+  size_t GetBytesPer10msBuffer() const {
+    return frames_per_10ms_buffer_ * GetBytesPerFrame();
+  }
+  double GetBufferSizeInMilliseconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return frames_per_buffer_ / (sample_rate_ / 1000.0);
+  }
+  double GetBufferSizeInSeconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return static_cast<double>(frames_per_buffer_) / (sample_rate_);
+  }
+  std::string ToString() const {
+    char ss_buf[1024];
+    rtc::SimpleStringBuilder ss(ss_buf);
+    ss << "AudioParameters: ";
+    ss << "sample_rate=" << sample_rate() << ", channels=" << channels();
+    ss << ", frames_per_buffer=" << frames_per_buffer();
+    ss << ", frames_per_10ms_buffer=" << frames_per_10ms_buffer();
+    ss << ", bytes_per_frame=" << GetBytesPerFrame();
+    ss << ", bytes_per_buffer=" << GetBytesPerBuffer();
+    ss << ", bytes_per_10ms_buffer=" << GetBytesPer10msBuffer();
+    ss << ", size_in_ms=" << GetBufferSizeInMilliseconds();
+    return ss.str();
+  }
+
+ private:
+  int sample_rate_;
+  size_t channels_;
+  size_t frames_per_buffer_;
+  size_t frames_per_10ms_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device_factory.cc b/third_party/libwebrtc/modules/audio_device/include/audio_device_factory.cc
new file mode 100644
index 0000000000..130e096e6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/audio_device_factory.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/include/audio_device_factory.h"
+
+#include <memory>
+
+#if defined(WEBRTC_WIN)
+#include "modules/audio_device/win/audio_device_module_win.h"
+#include "modules/audio_device/win/core_audio_input_win.h"
+#include "modules/audio_device/win/core_audio_output_win.h"
+#include "modules/audio_device/win/core_audio_utility_win.h"
+#endif
+
+#include "api/task_queue/task_queue_factory.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDeviceModule> CreateWindowsCoreAudioAudioDeviceModule(
+    TaskQueueFactory* task_queue_factory,
+    bool automatic_restart) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  return CreateWindowsCoreAudioAudioDeviceModuleForTest(task_queue_factory,
+                                                        automatic_restart);
+}
+
+rtc::scoped_refptr<AudioDeviceModuleForTest>
+CreateWindowsCoreAudioAudioDeviceModuleForTest(
+    TaskQueueFactory* task_queue_factory,
+    bool automatic_restart) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  // Returns NULL if Core Audio is not supported or if COM has not been
+  // initialized correctly using ScopedCOMInitializer.
+  if (!webrtc_win::core_audio_utility::IsSupported()) {
+    RTC_LOG(LS_ERROR)
+        << "Unable to create ADM since Core Audio is not supported";
+    return nullptr;
+  }
+  return CreateWindowsCoreAudioAudioDeviceModuleFromInputAndOutput(
+      std::make_unique<webrtc_win::CoreAudioInput>(automatic_restart),
+      std::make_unique<webrtc_win::CoreAudioOutput>(automatic_restart),
+      task_queue_factory);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/include/audio_device_factory.h b/third_party/libwebrtc/modules/audio_device/include/audio_device_factory.h
new file mode 100644
index 0000000000..edd7686b8e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/audio_device_factory.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_FACTORY_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_FACTORY_H_
+
+#include <memory>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+
+// Creates an AudioDeviceModule (ADM) for Windows based on the Core Audio API.
+// The creating thread must be a COM thread; otherwise nullptr will be returned.
+// By default `automatic_restart` is set to true and it results in support for
+// automatic restart of audio if e.g. the existing device is removed. If set to
+// false, no attempt to restart audio is performed under these conditions.
+//
+// Example (assuming webrtc namespace):
+//
+//  public:
+//   rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice() {
+//     task_queue_factory_ = CreateDefaultTaskQueueFactory();
+//     // Tell COM that this thread shall live in the MTA.
+//     com_initializer_ = std::make_unique<ScopedCOMInitializer>(
+//         ScopedCOMInitializer::kMTA);
+//     if (!com_initializer_->Succeeded()) {
+//       return nullptr;
+//     }
+//     // Create the ADM with support for automatic restart if devices are
+//     // unplugged.
+//     return CreateWindowsCoreAudioAudioDeviceModule(
+//         task_queue_factory_.get());
+//   }
+//
+//   private:
+//    std::unique_ptr<ScopedCOMInitializer> com_initializer_;
+//    std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+//
+rtc::scoped_refptr<AudioDeviceModule> CreateWindowsCoreAudioAudioDeviceModule(
+    TaskQueueFactory* task_queue_factory,
+    bool automatic_restart = true);
+
+rtc::scoped_refptr<AudioDeviceModuleForTest>
+CreateWindowsCoreAudioAudioDeviceModuleForTest(
+    TaskQueueFactory* task_queue_factory,
+    bool automatic_restart = true);
+
+}  // namespace webrtc
+
+#endif  //  MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_device/include/fake_audio_device.h b/third_party/libwebrtc/modules/audio_device/include/fake_audio_device.h
new file mode 100644
index 0000000000..2322ce0263
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/fake_audio_device.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_
+
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_default.h"
+
+namespace webrtc {
+
+class FakeAudioDeviceModule
+    : public webrtc_impl::AudioDeviceModuleDefault<AudioDeviceModule> {
+ public:
+  // TODO(bugs.webrtc.org/12701): Fix all users of this class to managed
+  // references using scoped_refptr. Current code doesn't always use refcounting
+  // for this class.
+  void AddRef() const override {}
+  rtc::RefCountReleaseStatus Release() const override {
+    return rtc::RefCountReleaseStatus::kDroppedLastRef;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_
diff --git a/third_party/libwebrtc/modules/audio_device/include/mock_audio_device.h b/third_party/libwebrtc/modules/audio_device/include/mock_audio_device.h
new file mode 100644
index 0000000000..73fbdd547d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/mock_audio_device.h
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_DEVICE_H_
+
+#include <string>
+
+#include "api/make_ref_counted.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockAudioDeviceModule : public AudioDeviceModule {
+ public:
+  static rtc::scoped_refptr<MockAudioDeviceModule> CreateNice() {
+    return rtc::make_ref_counted<::testing::NiceMock<MockAudioDeviceModule>>();
+  }
+  static rtc::scoped_refptr<MockAudioDeviceModule> CreateStrict() {
+    return rtc::make_ref_counted<
+        ::testing::StrictMock<MockAudioDeviceModule>>();
+  }
+
+  // AudioDeviceModule.
+  MOCK_METHOD(int32_t,
+              ActiveAudioLayer,
+              (AudioLayer * audioLayer),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              RegisterAudioCallback,
+              (AudioTransport * audioCallback),
+              (override));
+  MOCK_METHOD(int32_t, Init, (), (override));
+  MOCK_METHOD(int32_t, Terminate, (), (override));
+  MOCK_METHOD(bool, Initialized, (), (const, override));
+  MOCK_METHOD(int16_t, PlayoutDevices, (), (override));
+  MOCK_METHOD(int16_t, RecordingDevices, (), (override));
+  MOCK_METHOD(int32_t,
+              PlayoutDeviceName,
+              (uint16_t index,
+               char name[kAdmMaxDeviceNameSize],
+               char guid[kAdmMaxGuidSize]),
+              (override));
+  MOCK_METHOD(int32_t,
+              RecordingDeviceName,
+              (uint16_t index,
+               char name[kAdmMaxDeviceNameSize],
+               char guid[kAdmMaxGuidSize]),
+              (override));
+  MOCK_METHOD(int32_t, SetPlayoutDevice, (uint16_t index), (override));
+  MOCK_METHOD(int32_t,
+              SetPlayoutDevice,
+              (WindowsDeviceType device),
+              (override));
+  MOCK_METHOD(int32_t, SetRecordingDevice, (uint16_t index), (override));
+  MOCK_METHOD(int32_t,
+              SetRecordingDevice,
+              (WindowsDeviceType device),
+              (override));
+  MOCK_METHOD(int32_t, PlayoutIsAvailable, (bool* available), (override));
+  MOCK_METHOD(int32_t, InitPlayout, (), (override));
+  MOCK_METHOD(bool, PlayoutIsInitialized, (), (const, override));
+  MOCK_METHOD(int32_t, RecordingIsAvailable, (bool* available), (override));
+  MOCK_METHOD(int32_t, InitRecording, (), (override));
+  MOCK_METHOD(bool, RecordingIsInitialized, (), (const, override));
+  MOCK_METHOD(int32_t, StartPlayout, (), (override));
+  MOCK_METHOD(int32_t, StopPlayout, (), (override));
+  MOCK_METHOD(bool, Playing, (), (const, override));
+  MOCK_METHOD(int32_t, StartRecording, (), (override));
+  MOCK_METHOD(int32_t, StopRecording, (), (override));
+  MOCK_METHOD(bool, Recording, (), (const, override));
+  MOCK_METHOD(int32_t, InitSpeaker, (), (override));
+  MOCK_METHOD(bool, SpeakerIsInitialized, (), (const, override));
+  MOCK_METHOD(int32_t, InitMicrophone, (), (override));
+  MOCK_METHOD(bool, MicrophoneIsInitialized, (), (const, override));
+  MOCK_METHOD(int32_t, SpeakerVolumeIsAvailable, (bool* available), (override));
+  MOCK_METHOD(int32_t, SetSpeakerVolume, (uint32_t volume), (override));
+  MOCK_METHOD(int32_t, SpeakerVolume, (uint32_t * volume), (const, override));
+  MOCK_METHOD(int32_t,
+              MaxSpeakerVolume,
+              (uint32_t * maxVolume),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              MinSpeakerVolume,
+              (uint32_t * minVolume),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              MicrophoneVolumeIsAvailable,
+              (bool* available),
+              (override));
+  MOCK_METHOD(int32_t, SetMicrophoneVolume, (uint32_t volume), (override));
+  MOCK_METHOD(int32_t,
+              MicrophoneVolume,
+              (uint32_t * volume),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              MaxMicrophoneVolume,
+              (uint32_t * maxVolume),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              MinMicrophoneVolume,
+              (uint32_t * minVolume),
+              (const, override));
+  MOCK_METHOD(int32_t, SpeakerMuteIsAvailable, (bool* available), (override));
+  MOCK_METHOD(int32_t, SetSpeakerMute, (bool enable), (override));
+  MOCK_METHOD(int32_t, SpeakerMute, (bool* enabled), (const, override));
+  MOCK_METHOD(int32_t,
+              MicrophoneMuteIsAvailable,
+              (bool* available),
+              (override));
+  MOCK_METHOD(int32_t, SetMicrophoneMute, (bool enable), (override));
+  MOCK_METHOD(int32_t, MicrophoneMute, (bool* enabled), (const, override));
+  MOCK_METHOD(int32_t,
+              StereoPlayoutIsAvailable,
+              (bool* available),
+              (const, override));
+  MOCK_METHOD(int32_t, SetStereoPlayout, (bool enable), (override));
+  MOCK_METHOD(int32_t, StereoPlayout, (bool* enabled), (const, override));
+  MOCK_METHOD(int32_t,
+              StereoRecordingIsAvailable,
+              (bool* available),
+              (const, override));
+  MOCK_METHOD(int32_t, SetStereoRecording, (bool enable), (override));
+  MOCK_METHOD(int32_t, StereoRecording, (bool* enabled), (const, override));
+  MOCK_METHOD(int32_t, PlayoutDelay, (uint16_t * delayMS), (const, override));
+  MOCK_METHOD(bool, BuiltInAECIsAvailable, (), (const, override));
+  MOCK_METHOD(bool, BuiltInAGCIsAvailable, (), (const, override));
+  MOCK_METHOD(bool, BuiltInNSIsAvailable, (), (const, override));
+  MOCK_METHOD(int32_t, EnableBuiltInAEC, (bool enable), (override));
+  MOCK_METHOD(int32_t, EnableBuiltInAGC, (bool enable), (override));
+  MOCK_METHOD(int32_t, EnableBuiltInNS, (bool enable), (override));
+  MOCK_METHOD(int32_t, GetPlayoutUnderrunCount, (), (const, override));
+#if defined(WEBRTC_IOS)
+  MOCK_METHOD(int,
+              GetPlayoutAudioParameters,
+              (AudioParameters * params),
+              (const, override));
+  MOCK_METHOD(int,
+              GetRecordAudioParameters,
+              (AudioParameters * params),
+              (const, override));
+#endif  // WEBRTC_IOS
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_DEVICE_H_
diff --git a/third_party/libwebrtc/modules/audio_device/include/mock_audio_transport.h b/third_party/libwebrtc/modules/audio_device/include/mock_audio_transport.h
new file mode 100644
index 0000000000..b886967319
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/mock_audio_transport.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
+
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockAudioTransport : public AudioTransport {
+ public:
+  MockAudioTransport() {}
+  ~MockAudioTransport() {}
+
+  MOCK_METHOD(int32_t,
+              RecordedDataIsAvailable,
+              (const void* audioSamples,
+               size_t nSamples,
+               size_t nBytesPerSample,
+               size_t nChannels,
+               uint32_t samplesPerSec,
+               uint32_t totalDelayMS,
+               int32_t clockDrift,
+               uint32_t currentMicLevel,
+               bool keyPressed,
+               uint32_t& newMicLevel),
+              (override));
+
+  MOCK_METHOD(int32_t,
+              RecordedDataIsAvailable,
+              (const void* audioSamples,
+               size_t nSamples,
+               size_t nBytesPerSample,
+               size_t nChannels,
+               uint32_t samplesPerSec,
+               uint32_t totalDelayMS,
+               int32_t clockDrift,
+               uint32_t currentMicLevel,
+               bool keyPressed,
+               uint32_t& newMicLevel,
+               absl::optional<int64_t> estimated_capture_time_ns),
+              (override));
+
+  MOCK_METHOD(int32_t,
+              NeedMorePlayData,
+              (size_t nSamples,
+               size_t nBytesPerSample,
+               size_t nChannels,
+               uint32_t samplesPerSec,
+               void* audioSamples,
+               size_t& nSamplesOut,
+               int64_t* elapsed_time_ms,
+               int64_t* ntp_time_ms),
+              (override));
+
+  MOCK_METHOD(void,
+              PullRenderData,
+              (int bits_per_sample,
+               int sample_rate,
+               size_t number_of_channels,
+               size_t number_of_frames,
+               void* audio_data,
+               int64_t* elapsed_time_ms,
+               int64_t* ntp_time_ms),
+              (override));
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
diff --git a/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc b/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc
new file mode 100644
index 0000000000..2189646eff
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc
@@ -0,0 +1,497 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_device/include/test_audio_device.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstdlib>
+#include <memory>
+#include <string>
+#include <type_traits>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/make_ref_counted.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_device/include/audio_device_default.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/random.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kFrameLengthUs = 10000;
+constexpr int kFramesPerSecond = rtc::kNumMicrosecsPerSec / kFrameLengthUs;
+
+// TestAudioDeviceModule implements an AudioDevice module that can act both as a
+// capturer and a renderer. It will use 10ms audio frames.
+class TestAudioDeviceModuleImpl
+    : public webrtc_impl::AudioDeviceModuleDefault<TestAudioDeviceModule> {
+ public:
+  // Creates a new TestAudioDeviceModule. When capturing or playing, 10 ms audio
+  // frames will be processed every 10ms / `speed`.
+  // `capturer` is an object that produces audio data. Can be nullptr if this
+  // device is never used for recording.
+  // `renderer` is an object that receives audio data that would have been
+  // played out. Can be nullptr if this device is never used for playing.
+  // Use one of the Create... functions to get these instances.
+  TestAudioDeviceModuleImpl(TaskQueueFactory* task_queue_factory,
+                            std::unique_ptr<Capturer> capturer,
+                            std::unique_ptr<Renderer> renderer,
+                            float speed = 1)
+      : task_queue_factory_(task_queue_factory),
+        capturer_(std::move(capturer)),
+        renderer_(std::move(renderer)),
+        process_interval_us_(kFrameLengthUs / speed),
+        audio_callback_(nullptr),
+        rendering_(false),
+        capturing_(false) {
+    auto good_sample_rate = [](int sr) {
+      return sr == 8000 || sr == 16000 || sr == 32000 || sr == 44100 ||
+             sr == 48000;
+    };
+
+    if (renderer_) {
+      const int sample_rate = renderer_->SamplingFrequency();
+      playout_buffer_.resize(
+          SamplesPerFrame(sample_rate) * renderer_->NumChannels(), 0);
+      RTC_CHECK(good_sample_rate(sample_rate));
+    }
+    if (capturer_) {
+      RTC_CHECK(good_sample_rate(capturer_->SamplingFrequency()));
+    }
+  }
+
+  ~TestAudioDeviceModuleImpl() override {
+    StopPlayout();
+    StopRecording();
+  }
+
+  int32_t Init() override {
+    task_queue_ =
+        std::make_unique<rtc::TaskQueue>(task_queue_factory_->CreateTaskQueue(
+            "TestAudioDeviceModuleImpl", TaskQueueFactory::Priority::NORMAL));
+
+    RepeatingTaskHandle::Start(task_queue_->Get(), [this]() {
+      ProcessAudio();
+      return TimeDelta::Micros(process_interval_us_);
+    });
+    return 0;
+  }
+
+  int32_t RegisterAudioCallback(AudioTransport* callback) override {
+    MutexLock lock(&lock_);
+    RTC_DCHECK(callback || audio_callback_);
+    audio_callback_ = callback;
+    return 0;
+  }
+
+  int32_t StartPlayout() override {
+    MutexLock lock(&lock_);
+    RTC_CHECK(renderer_);
+    rendering_ = true;
+    return 0;
+  }
+
+  int32_t StopPlayout() override {
+    MutexLock lock(&lock_);
+    rendering_ = false;
+    return 0;
+  }
+
+  int32_t StartRecording() override {
+    MutexLock lock(&lock_);
+    RTC_CHECK(capturer_);
+    capturing_ = true;
+    return 0;
+  }
+
+  int32_t StopRecording() override {
+    MutexLock lock(&lock_);
+    capturing_ = false;
+    return 0;
+  }
+
+  bool Playing() const override {
+    MutexLock lock(&lock_);
+    return rendering_;
+  }
+
+  bool Recording() const override {
+    MutexLock lock(&lock_);
+    return capturing_;
+  }
+
+  // Blocks forever until the Recorder stops producing data.
+  void WaitForRecordingEnd() override {
+    done_capturing_.Wait(rtc::Event::kForever);
+  }
+
+ private:
+  void ProcessAudio() {
+    MutexLock lock(&lock_);
+    if (capturing_) {
+      // Capture 10ms of audio. 2 bytes per sample.
+      const bool keep_capturing = capturer_->Capture(&recording_buffer_);
+      uint32_t new_mic_level = 0;
+      if (recording_buffer_.size() > 0) {
+        audio_callback_->RecordedDataIsAvailable(
+            recording_buffer_.data(),
+            recording_buffer_.size() / capturer_->NumChannels(),
+            2 * capturer_->NumChannels(), capturer_->NumChannels(),
+            capturer_->SamplingFrequency(), 0, 0, 0, false, new_mic_level);
+      }
+      if (!keep_capturing) {
+        capturing_ = false;
+        done_capturing_.Set();
+      }
+    }
+    if (rendering_) {
+      size_t samples_out = 0;
+      int64_t elapsed_time_ms = -1;
+      int64_t ntp_time_ms = -1;
+      const int sampling_frequency = renderer_->SamplingFrequency();
+      audio_callback_->NeedMorePlayData(
+          SamplesPerFrame(sampling_frequency), 2 * renderer_->NumChannels(),
+          renderer_->NumChannels(), sampling_frequency, playout_buffer_.data(),
+          samples_out, &elapsed_time_ms, &ntp_time_ms);
+      const bool keep_rendering = renderer_->Render(
+          rtc::ArrayView<const int16_t>(playout_buffer_.data(), samples_out));
+      if (!keep_rendering) {
+        rendering_ = false;
+        done_rendering_.Set();
+      }
+    }
+  }
+  TaskQueueFactory* const task_queue_factory_;
+  const std::unique_ptr<Capturer> capturer_ RTC_GUARDED_BY(lock_);
+  const std::unique_ptr<Renderer> renderer_ RTC_GUARDED_BY(lock_);
+  const int64_t process_interval_us_;
+
+  mutable Mutex lock_;
+  AudioTransport* audio_callback_ RTC_GUARDED_BY(lock_);
+  bool rendering_ RTC_GUARDED_BY(lock_);
+  bool capturing_ RTC_GUARDED_BY(lock_);
+  rtc::Event done_rendering_;
+  rtc::Event done_capturing_;
+
+  std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
+  rtc::BufferT<int16_t> recording_buffer_ RTC_GUARDED_BY(lock_);
+  std::unique_ptr<rtc::TaskQueue> task_queue_;
+};
+
+// A fake capturer that generates pulses with random samples between
+// -max_amplitude and +max_amplitude.
+class PulsedNoiseCapturerImpl final
+    : public TestAudioDeviceModule::PulsedNoiseCapturer {
+ public:
+  // Assuming 10ms audio packets.
+  PulsedNoiseCapturerImpl(int16_t max_amplitude,
+                          int sampling_frequency_in_hz,
+                          int num_channels)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        fill_with_zero_(false),
+        random_generator_(1),
+        max_amplitude_(max_amplitude),
+        num_channels_(num_channels) {
+    RTC_DCHECK_GT(max_amplitude, 0);
+  }
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Capture(rtc::BufferT<int16_t>* buffer) override {
+    fill_with_zero_ = !fill_with_zero_;
+    int16_t max_amplitude;
+    {
+      MutexLock lock(&lock_);
+      max_amplitude = max_amplitude_;
+    }
+    buffer->SetData(
+        TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) *
+            num_channels_,
+        [&](rtc::ArrayView<int16_t> data) {
+          if (fill_with_zero_) {
+            std::fill(data.begin(), data.end(), 0);
+          } else {
+            std::generate(data.begin(), data.end(), [&]() {
+              return random_generator_.Rand(-max_amplitude, max_amplitude);
+            });
+          }
+          return data.size();
+        });
+    return true;
+  }
+
+  void SetMaxAmplitude(int16_t amplitude) override {
+    MutexLock lock(&lock_);
+    max_amplitude_ = amplitude;
+  }
+
+ private:
+  int sampling_frequency_in_hz_;
+  bool fill_with_zero_;
+  Random random_generator_;
+  Mutex lock_;
+  int16_t max_amplitude_ RTC_GUARDED_BY(lock_);
+  const int num_channels_;
+};
+
+class WavFileReader final : public TestAudioDeviceModule::Capturer {
+ public:
+  WavFileReader(absl::string_view filename,
+                int sampling_frequency_in_hz,
+                int num_channels,
+                bool repeat)
+      : WavFileReader(std::make_unique<WavReader>(filename),
+                      sampling_frequency_in_hz,
+                      num_channels,
+                      repeat) {}
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Capture(rtc::BufferT<int16_t>* buffer) override {
+    buffer->SetData(
+        TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) *
+            num_channels_,
+        [&](rtc::ArrayView<int16_t> data) {
+          size_t read = wav_reader_->ReadSamples(data.size(), data.data());
+          if (read < data.size() && repeat_) {
+            do {
+              wav_reader_->Reset();
+              size_t delta = wav_reader_->ReadSamples(
+                  data.size() - read, data.subview(read).data());
+              RTC_CHECK_GT(delta, 0) << "No new data read from file";
+              read += delta;
+            } while (read < data.size());
+          }
+          return read;
+        });
+    return buffer->size() > 0;
+  }
+
+ private:
+  WavFileReader(std::unique_ptr<WavReader> wav_reader,
+                int sampling_frequency_in_hz,
+                int num_channels,
+                bool repeat)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        num_channels_(num_channels),
+        wav_reader_(std::move(wav_reader)),
+        repeat_(repeat) {
+    RTC_CHECK_EQ(wav_reader_->sample_rate(), sampling_frequency_in_hz);
+    RTC_CHECK_EQ(wav_reader_->num_channels(), num_channels);
+  }
+
+  const int sampling_frequency_in_hz_;
+  const int num_channels_;
+  std::unique_ptr<WavReader> wav_reader_;
+  const bool repeat_;
+};
+
+class WavFileWriter final : public TestAudioDeviceModule::Renderer {
+ public:
+  WavFileWriter(absl::string_view filename,
+                int sampling_frequency_in_hz,
+                int num_channels)
+      : WavFileWriter(std::make_unique<WavWriter>(filename,
+                                                  sampling_frequency_in_hz,
+                                                  num_channels),
+                      sampling_frequency_in_hz,
+                      num_channels) {}
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Render(rtc::ArrayView<const int16_t> data) override {
+    wav_writer_->WriteSamples(data.data(), data.size());
+    return true;
+  }
+
+ private:
+  WavFileWriter(std::unique_ptr<WavWriter> wav_writer,
+                int sampling_frequency_in_hz,
+                int num_channels)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        wav_writer_(std::move(wav_writer)),
+        num_channels_(num_channels) {}
+
+  int sampling_frequency_in_hz_;
+  std::unique_ptr<WavWriter> wav_writer_;
+  const int num_channels_;
+};
+
+class BoundedWavFileWriter : public TestAudioDeviceModule::Renderer {
+ public:
+  BoundedWavFileWriter(absl::string_view filename,
+                       int sampling_frequency_in_hz,
+                       int num_channels)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        wav_writer_(filename, sampling_frequency_in_hz, num_channels),
+        num_channels_(num_channels),
+        silent_audio_(
+            TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz) *
+                num_channels,
+            0),
+        started_writing_(false),
+        trailing_zeros_(0) {}
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Render(rtc::ArrayView<const int16_t> data) override {
+    const int16_t kAmplitudeThreshold = 5;
+
+    const int16_t* begin = data.begin();
+    const int16_t* end = data.end();
+    if (!started_writing_) {
+      // Cut off silence at the beginning.
+      while (begin < end) {
+        if (std::abs(*begin) > kAmplitudeThreshold) {
+          started_writing_ = true;
+          break;
+        }
+        ++begin;
+      }
+    }
+    if (started_writing_) {
+      // Cut off silence at the end.
+      while (begin < end) {
+        if (*(end - 1) != 0) {
+          break;
+        }
+        --end;
+      }
+      if (begin < end) {
+        // If it turns out that the silence was not final, need to write all the
+        // skipped zeros and continue writing audio.
+        while (trailing_zeros_ > 0) {
+          const size_t zeros_to_write =
+              std::min(trailing_zeros_, silent_audio_.size());
+          wav_writer_.WriteSamples(silent_audio_.data(), zeros_to_write);
+          trailing_zeros_ -= zeros_to_write;
+        }
+        wav_writer_.WriteSamples(begin, end - begin);
+      }
+      // Save the number of zeros we skipped in case this needs to be restored.
+      trailing_zeros_ += data.end() - end;
+    }
+    return true;
+  }
+
+ private:
+  int sampling_frequency_in_hz_;
+  WavWriter wav_writer_;
+  const int num_channels_;
+  std::vector<int16_t> silent_audio_;
+  bool started_writing_;
+  size_t trailing_zeros_;
+};
+
+class DiscardRenderer final : public TestAudioDeviceModule::Renderer {
+ public:
+  explicit DiscardRenderer(int sampling_frequency_in_hz, int num_channels)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        num_channels_(num_channels) {}
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Render(rtc::ArrayView<const int16_t> data) override { return true; }
+
+ private:
+  int sampling_frequency_in_hz_;
+  const int num_channels_;
+};
+
+}  // namespace
+
+size_t TestAudioDeviceModule::SamplesPerFrame(int sampling_frequency_in_hz) {
+  return rtc::CheckedDivExact(sampling_frequency_in_hz, kFramesPerSecond);
+}
+
+rtc::scoped_refptr<TestAudioDeviceModule> TestAudioDeviceModule::Create(
+    TaskQueueFactory* task_queue_factory,
+    std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+    std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
+    float speed) {
+  return rtc::make_ref_counted<TestAudioDeviceModuleImpl>(
+      task_queue_factory, std::move(capturer), std::move(renderer), speed);
+}
+
+std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer>
+TestAudioDeviceModule::CreatePulsedNoiseCapturer(int16_t max_amplitude,
+                                                 int sampling_frequency_in_hz,
+                                                 int num_channels) {
+  return std::make_unique<PulsedNoiseCapturerImpl>(
+      max_amplitude, sampling_frequency_in_hz, num_channels);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer>
+TestAudioDeviceModule::CreateDiscardRenderer(int sampling_frequency_in_hz,
+                                             int num_channels) {
+  return std::make_unique<DiscardRenderer>(sampling_frequency_in_hz,
+                                           num_channels);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Capturer>
+TestAudioDeviceModule::CreateWavFileReader(absl::string_view filename,
+                                           int sampling_frequency_in_hz,
+                                           int num_channels) {
+  return std::make_unique<WavFileReader>(filename, sampling_frequency_in_hz,
+                                         num_channels, false);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Capturer>
+TestAudioDeviceModule::CreateWavFileReader(absl::string_view filename,
+                                           bool repeat) {
+  WavReader reader(filename);
+  int sampling_frequency_in_hz = reader.sample_rate();
+  int num_channels = rtc::checked_cast<int>(reader.num_channels());
+  return std::make_unique<WavFileReader>(filename, sampling_frequency_in_hz,
+                                         num_channels, repeat);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer>
+TestAudioDeviceModule::CreateWavFileWriter(absl::string_view filename,
+                                           int sampling_frequency_in_hz,
+                                           int num_channels) {
+  return std::make_unique<WavFileWriter>(filename, sampling_frequency_in_hz,
+                                         num_channels);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer>
+TestAudioDeviceModule::CreateBoundedWavFileWriter(absl::string_view filename,
+                                                  int sampling_frequency_in_hz,
+                                                  int num_channels) {
+  return std::make_unique<BoundedWavFileWriter>(
+      filename, sampling_frequency_in_hz, num_channels);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/include/test_audio_device.h b/third_party/libwebrtc/modules/audio_device/include/test_audio_device.h
new file mode 100644
index 0000000000..8413479291
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/test_audio_device.h
@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+// TestAudioDeviceModule implements an AudioDevice module that can act both as a
+// capturer and a renderer. It will use 10ms audio frames.
+class TestAudioDeviceModule : public AudioDeviceModule {
+ public:
+  // Returns the number of samples that Capturers and Renderers with this
+  // sampling frequency will work with every time Capture or Render is called.
+  static size_t SamplesPerFrame(int sampling_frequency_in_hz);
+
+  class Capturer {
+   public:
+    virtual ~Capturer() {}
+    // Returns the sampling frequency in Hz of the audio data that this
+    // capturer produces.
+    virtual int SamplingFrequency() const = 0;
+    // Returns the number of channels of captured audio data.
+    virtual int NumChannels() const = 0;
+    // Replaces the contents of `buffer` with 10ms of captured audio data
+    // (see TestAudioDeviceModule::SamplesPerFrame). Returns true if the
+    // capturer can keep producing data, or false when the capture finishes.
+    virtual bool Capture(rtc::BufferT<int16_t>* buffer) = 0;
+  };
+
+  class Renderer {
+   public:
+    virtual ~Renderer() {}
+    // Returns the sampling frequency in Hz of the audio data that this
+    // renderer receives.
+    virtual int SamplingFrequency() const = 0;
+    // Returns the number of channels of audio data to be required.
+    virtual int NumChannels() const = 0;
+    // Renders the passed audio data and returns true if the renderer wants
+    // to keep receiving data, or false otherwise.
+    virtual bool Render(rtc::ArrayView<const int16_t> data) = 0;
+  };
+
+  // A fake capturer that generates pulses with random samples between
+  // -max_amplitude and +max_amplitude.
+  class PulsedNoiseCapturer : public Capturer {
+   public:
+    ~PulsedNoiseCapturer() override {}
+
+    virtual void SetMaxAmplitude(int16_t amplitude) = 0;
+  };
+
+  ~TestAudioDeviceModule() override {}
+
+  // Creates a new TestAudioDeviceModule. When capturing or playing, 10 ms audio
+  // frames will be processed every 10ms / `speed`.
+  // `capturer` is an object that produces audio data. Can be nullptr if this
+  // device is never used for recording.
+  // `renderer` is an object that receives audio data that would have been
+  // played out. Can be nullptr if this device is never used for playing.
+  // Use one of the Create... functions to get these instances.
+  static rtc::scoped_refptr<TestAudioDeviceModule> Create(
+      TaskQueueFactory* task_queue_factory,
+      std::unique_ptr<Capturer> capturer,
+      std::unique_ptr<Renderer> renderer,
+      float speed = 1);
+
+  // Returns a Capturer instance that generates a signal of `num_channels`
+  // channels where every second frame is zero and every second frame is evenly
+  // distributed random noise with max amplitude `max_amplitude`.
+  static std::unique_ptr<PulsedNoiseCapturer> CreatePulsedNoiseCapturer(
+      int16_t max_amplitude,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Renderer instance that does nothing with the audio data.
+  static std::unique_ptr<Renderer> CreateDiscardRenderer(
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // WavReader and WavWriter creation based on file name.
+
+  // Returns a Capturer instance that gets its data from a file. The sample rate
+  // and channels will be checked against the Wav file.
+  static std::unique_ptr<Capturer> CreateWavFileReader(
+      absl::string_view filename,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Capturer instance that gets its data from a file.
+  // Automatically detects sample rate and num of channels.
+  // `repeat` - if true, the file will be replayed from the start when we reach
+  // the end of file.
+  static std::unique_ptr<Capturer> CreateWavFileReader(
+      absl::string_view filename,
+      bool repeat = false);
+
+  // Returns a Renderer instance that writes its data to a file.
+  static std::unique_ptr<Renderer> CreateWavFileWriter(
+      absl::string_view filename,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Renderer instance that writes its data to a WAV file, cutting
+  // off silence at the beginning (not necessarily perfect silence, see
+  // kAmplitudeThreshold) and at the end (only actual 0 samples in this case).
+  static std::unique_ptr<Renderer> CreateBoundedWavFileWriter(
+      absl::string_view filename,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  int32_t Init() override = 0;
+  int32_t RegisterAudioCallback(AudioTransport* callback) override = 0;
+
+  int32_t StartPlayout() override = 0;
+  int32_t StopPlayout() override = 0;
+  int32_t StartRecording() override = 0;
+  int32_t StopRecording() override = 0;
+
+  bool Playing() const override = 0;
+  bool Recording() const override = 0;
+
+  // Blocks forever until the Recorder stops producing data.
+  virtual void WaitForRecordingEnd() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
diff --git a/third_party/libwebrtc/modules/audio_device/include/test_audio_device_unittest.cc b/third_party/libwebrtc/modules/audio_device/include/test_audio_device_unittest.cc
new file mode 100644
index 0000000000..2975b11325
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/include/test_audio_device_unittest.cc
@@ -0,0 +1,192 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/include/test_audio_device.h"
+
+#include <algorithm>
+#include <array>
+
+#include "api/array_view.h"
+#include "common_audio/wav_file.h"
+#include "common_audio/wav_header.h"
+#include "rtc_base/logging.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+void RunTest(const std::vector<int16_t>& input_samples,
+             const std::vector<int16_t>& expected_samples,
+             size_t samples_per_frame) {
+  const ::testing::TestInfo* const test_info =
+      ::testing::UnitTest::GetInstance()->current_test_info();
+
+  const std::string output_filename =
+      test::OutputPath() + "BoundedWavFileWriterTest_" + test_info->name() +
+      "_" + std::to_string(std::rand()) + ".wav";
+
+  static const size_t kSamplesPerFrame = 8;
+  static const int kSampleRate = kSamplesPerFrame * 100;
+  EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
+            kSamplesPerFrame);
+
+  // Test through file name API.
+  {
+    std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
+        TestAudioDeviceModule::CreateBoundedWavFileWriter(output_filename, 800);
+
+    for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
+      EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
+          &input_samples[i],
+          std::min(kSamplesPerFrame, input_samples.size() - i))));
+    }
+  }
+
+  {
+    WavReader reader(output_filename);
+    std::vector<int16_t> read_samples(expected_samples.size());
+    EXPECT_EQ(expected_samples.size(),
+              reader.ReadSamples(read_samples.size(), read_samples.data()));
+    EXPECT_EQ(expected_samples, read_samples);
+
+    EXPECT_EQ(0u, reader.ReadSamples(read_samples.size(), read_samples.data()));
+  }
+
+  remove(output_filename.c_str());
+}
+}  // namespace
+
+TEST(BoundedWavFileWriterTest, NoSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      75,   1234,  243,    -1231, -22222, 0,    3,      88,
+      1222, -1213, -13222, -7,    -3525,  5787, -25247, 8};
+  static const std::vector<int16_t> kExpectedSamples = kInputSamples;
+  RunTest(kInputSamples, kExpectedSamples, 8);
+}
+
+TEST(BoundedWavFileWriterTest, SomeStartSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      0, 0, 0, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 10,
+                                                     kInputSamples.end());
+  RunTest(kInputSamples, kExpectedSamples, 8);
+}
+
+TEST(BoundedWavFileWriterTest, NegativeStartSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      0, -4, -6, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 2,
+                                                     kInputSamples.end());
+  RunTest(kInputSamples, kExpectedSamples, 8);
+}
+
+TEST(BoundedWavFileWriterTest, SomeEndSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
+                                                     kInputSamples.end() - 9);
+  RunTest(kInputSamples, kExpectedSamples, 8);
+}
+
+TEST(BoundedWavFileWriterTest, DoubleEndSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      75, 1234,  243,    -1231, -22222, 0,    0, 0,
+      0,  -1213, -13222, -7,    -3525,  5787, 0, 0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
+                                                     kInputSamples.end() - 2);
+  RunTest(kInputSamples, kExpectedSamples, 8);
+}
+
+TEST(BoundedWavFileWriterTest, DoubleSilence) {
+  static const std::vector<int16_t> kInputSamples = {0,     -1213, -13222, -7,
+                                                     -3525, 5787,  0,      0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 1,
+                                                     kInputSamples.end() - 2);
+  RunTest(kInputSamples, kExpectedSamples, 8);
+}
+
+TEST(BoundedWavFileWriterTest, EndSilenceCutoff) {
+  static const std::vector<int16_t> kInputSamples = {
+      75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
+                                                     kInputSamples.end() - 4);
+  RunTest(kInputSamples, kExpectedSamples, 8);
+}
+
+TEST(WavFileReaderTest, RepeatedTrueWithSingleFrameFileReadTwice) {
+  static const std::vector<int16_t> kInputSamples = {75,     1234, 243, -1231,
+                                                     -22222, 0,    3,   88};
+  static const rtc::BufferT<int16_t> kExpectedSamples(kInputSamples.data(),
+                                                      kInputSamples.size());
+
+  const std::string output_filename = test::OutputPath() +
+                                      "WavFileReaderTest_RepeatedTrue_" +
+                                      std::to_string(std::rand()) + ".wav";
+
+  static const size_t kSamplesPerFrame = 8;
+  static const int kSampleRate = kSamplesPerFrame * 100;
+  EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
+            kSamplesPerFrame);
+
+  // Create wav file to read.
+  {
+    std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
+        TestAudioDeviceModule::CreateWavFileWriter(output_filename, 800);
+
+    for (size_t i = 0; i < kInputSamples.size(); i += kSamplesPerFrame) {
+      EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
+          &kInputSamples[i],
+          std::min(kSamplesPerFrame, kInputSamples.size() - i))));
+    }
+  }
+
+  {
+    std::unique_ptr<TestAudioDeviceModule::Capturer> reader =
+        TestAudioDeviceModule::CreateWavFileReader(output_filename, true);
+    rtc::BufferT<int16_t> buffer(kExpectedSamples.size());
+    EXPECT_TRUE(reader->Capture(&buffer));
+    EXPECT_EQ(kExpectedSamples, buffer);
+    EXPECT_TRUE(reader->Capture(&buffer));
+    EXPECT_EQ(kExpectedSamples, buffer);
+  }
+
+  remove(output_filename.c_str());
+}
+
+TEST(PulsedNoiseCapturerTest, SetMaxAmplitude) {
+  const int16_t kAmplitude = 50;
+  std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
+      TestAudioDeviceModule::CreatePulsedNoiseCapturer(
+          kAmplitude, /*sampling_frequency_in_hz=*/8000);
+  rtc::BufferT<int16_t> recording_buffer;
+
+  // Verify that the capturer doesn't create entries louder than than
+  // kAmplitude. Since the pulse generator alternates between writing
+  // zeroes and actual entries, we need to do the capturing twice.
+  capturer->Capture(&recording_buffer);
+  capturer->Capture(&recording_buffer);
+  int16_t max_sample =
+      *std::max_element(recording_buffer.begin(), recording_buffer.end());
+  EXPECT_LE(max_sample, kAmplitude);
+
+  // Increase the amplitude and verify that the samples can now be louder
+  // than the previous max.
+  capturer->SetMaxAmplitude(kAmplitude * 2);
+  capturer->Capture(&recording_buffer);
+  capturer->Capture(&recording_buffer);
+  max_sample =
+      *std::max_element(recording_buffer.begin(), recording_buffer.end());
+  EXPECT_GT(max_sample, kAmplitude);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.cc
new file mode 100644
index 0000000000..5dfb91d6f4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.cc
@@ -0,0 +1,40 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "modules/audio_device/linux/alsasymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_alsa {
+
+LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(AlsaSymbolTable, "libasound.so.2")
+#define X(sym) LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(AlsaSymbolTable, sym)
+ALSA_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DEFINE_END(AlsaSymbolTable)
+
+}  // namespace adm_linux_alsa
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.h b/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.h
new file mode 100644
index 0000000000..c9970b02bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/alsasymboltable_linux.h
@@ -0,0 +1,148 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H_
+#define AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H_
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_alsa {
+
+// The ALSA symbols we need, as an X-Macro list.
+// This list must contain precisely every libasound function that is used in
+// alsasoundsystem.cc.
+#define ALSA_SYMBOLS_LIST                      \
+  X(snd_device_name_free_hint)                 \
+  X(snd_device_name_get_hint)                  \
+  X(snd_device_name_hint)                      \
+  X(snd_pcm_avail_update)                      \
+  X(snd_pcm_close)                             \
+  X(snd_pcm_delay)                             \
+  X(snd_pcm_drop)                              \
+  X(snd_pcm_open)                              \
+  X(snd_pcm_prepare)                           \
+  X(snd_pcm_readi)                             \
+  X(snd_pcm_recover)                           \
+  X(snd_pcm_resume)                            \
+  X(snd_pcm_reset)                             \
+  X(snd_pcm_state)                             \
+  X(snd_pcm_set_params)                        \
+  X(snd_pcm_get_params)                        \
+  X(snd_pcm_start)                             \
+  X(snd_pcm_stream)                            \
+  X(snd_pcm_frames_to_bytes)                   \
+  X(snd_pcm_bytes_to_frames)                   \
+  X(snd_pcm_wait)                              \
+  X(snd_pcm_writei)                            \
+  X(snd_pcm_info_get_class)                    \
+  X(snd_pcm_info_get_subdevices_avail)         \
+  X(snd_pcm_info_get_subdevice_name)           \
+  X(snd_pcm_info_set_subdevice)                \
+  X(snd_pcm_info_get_id)                       \
+  X(snd_pcm_info_set_device)                   \
+  X(snd_pcm_info_set_stream)                   \
+  X(snd_pcm_info_get_name)                     \
+  X(snd_pcm_info_get_subdevices_count)         \
+  X(snd_pcm_info_sizeof)                       \
+  X(snd_pcm_hw_params)                         \
+  X(snd_pcm_hw_params_malloc)                  \
+  X(snd_pcm_hw_params_free)                    \
+  X(snd_pcm_hw_params_any)                     \
+  X(snd_pcm_hw_params_set_access)              \
+  X(snd_pcm_hw_params_set_format)              \
+  X(snd_pcm_hw_params_set_channels)            \
+  X(snd_pcm_hw_params_set_rate_near)           \
+  X(snd_pcm_hw_params_set_buffer_size_near)    \
+  X(snd_card_next)                             \
+  X(snd_card_get_name)                         \
+  X(snd_config_update)                         \
+  X(snd_config_copy)                           \
+  X(snd_config_get_id)                         \
+  X(snd_ctl_open)                              \
+  X(snd_ctl_close)                             \
+  X(snd_ctl_card_info)                         \
+  X(snd_ctl_card_info_sizeof)                  \
+  X(snd_ctl_card_info_get_id)                  \
+  X(snd_ctl_card_info_get_name)                \
+  X(snd_ctl_pcm_next_device)                   \
+  X(snd_ctl_pcm_info)                          \
+  X(snd_mixer_load)                            \
+  X(snd_mixer_free)                            \
+  X(snd_mixer_detach)                          \
+  X(snd_mixer_close)                           \
+  X(snd_mixer_open)                            \
+  X(snd_mixer_attach)                          \
+  X(snd_mixer_first_elem)                      \
+  X(snd_mixer_elem_next)                       \
+  X(snd_mixer_selem_get_name)                  \
+  X(snd_mixer_selem_is_active)                 \
+  X(snd_mixer_selem_register)                  \
+  X(snd_mixer_selem_set_playback_volume_all)   \
+  X(snd_mixer_selem_get_playback_volume)       \
+  X(snd_mixer_selem_has_playback_volume)       \
+  X(snd_mixer_selem_get_playback_volume_range) \
+  X(snd_mixer_selem_has_playback_switch)       \
+  X(snd_mixer_selem_get_playback_switch)       \
+  X(snd_mixer_selem_set_playback_switch_all)   \
+  X(snd_mixer_selem_has_capture_switch)        \
+  X(snd_mixer_selem_get_capture_switch)        \
+  X(snd_mixer_selem_set_capture_switch_all)    \
+  X(snd_mixer_selem_has_capture_volume)        \
+  X(snd_mixer_selem_set_capture_volume_all)    \
+  X(snd_mixer_selem_get_capture_volume)        \
+  X(snd_mixer_selem_get_capture_volume_range)  \
+  X(snd_dlopen)                                \
+  X(snd_dlclose)                               \
+  X(snd_config)                                \
+  X(snd_config_search)                         \
+  X(snd_config_get_string)                     \
+  X(snd_config_search_definition)              \
+  X(snd_config_get_type)                       \
+  X(snd_config_delete)                         \
+  X(snd_config_iterator_entry)                 \
+  X(snd_config_iterator_first)                 \
+  X(snd_config_iterator_next)                  \
+  X(snd_config_iterator_end)                   \
+  X(snd_config_delete_compound_members)        \
+  X(snd_config_get_integer)                    \
+  X(snd_config_get_bool)                       \
+  X(snd_dlsym)                                 \
+  X(snd_strerror)                              \
+  X(snd_lib_error)                             \
+  X(snd_lib_error_set_handler)
+
+LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(AlsaSymbolTable)
+#define X(sym) LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(AlsaSymbolTable, sym)
+ALSA_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DECLARE_END(AlsaSymbolTable)
+
+}  // namespace adm_linux_alsa
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
new file mode 100644
index 0000000000..50cf3beb6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -0,0 +1,1637 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/linux/audio_device_alsa_linux.h"
+
+
+#include "modules/audio_device/audio_device_config.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/sleep.h"
+
+WebRTCAlsaSymbolTable* GetAlsaSymbolTable() {
+  static WebRTCAlsaSymbolTable* alsa_symbol_table = new WebRTCAlsaSymbolTable();
+  return alsa_symbol_table;
+}
+
+// Accesses ALSA functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libasound, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                                            \
+  LATESYM_GET(webrtc::adm_linux_alsa::AlsaSymbolTable, GetAlsaSymbolTable(), \
+              sym)
+
+// Redefine these here to be able to do late-binding
+#undef snd_ctl_card_info_alloca
+#define snd_ctl_card_info_alloca(ptr)                  \
+  do {                                                 \
+    *ptr = (snd_ctl_card_info_t*)__builtin_alloca(     \
+        LATE(snd_ctl_card_info_sizeof)());             \
+    memset(*ptr, 0, LATE(snd_ctl_card_info_sizeof)()); \
+  } while (0)
+
+#undef snd_pcm_info_alloca
+#define snd_pcm_info_alloca(pInfo)                                           \
+  do {                                                                       \
+    *pInfo = (snd_pcm_info_t*)__builtin_alloca(LATE(snd_pcm_info_sizeof)()); \
+    memset(*pInfo, 0, LATE(snd_pcm_info_sizeof)());                          \
+  } while (0)
+
+// snd_lib_error_handler_t
+void WebrtcAlsaErrorHandler(const char* file,
+                            int line,
+                            const char* function,
+                            int err,
+                            const char* fmt,
+                            ...) {}
+
+namespace webrtc {
+static const unsigned int ALSA_PLAYOUT_FREQ = 48000;
+static const unsigned int ALSA_PLAYOUT_CH = 2;
+static const unsigned int ALSA_PLAYOUT_LATENCY = 40 * 1000;  // in us
+static const unsigned int ALSA_CAPTURE_FREQ = 48000;
+static const unsigned int ALSA_CAPTURE_CH = 2;
+static const unsigned int ALSA_CAPTURE_LATENCY = 40 * 1000;  // in us
+static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5;     // in ms
+
+#define FUNC_GET_NUM_OF_DEVICE 0
+#define FUNC_GET_DEVICE_NAME 1
+#define FUNC_GET_DEVICE_NAME_FOR_AN_ENUM 2
+
+AudioDeviceLinuxALSA::AudioDeviceLinuxALSA()
+    : _ptrAudioBuffer(NULL),
+      _inputDeviceIndex(0),
+      _outputDeviceIndex(0),
+      _inputDeviceIsSpecified(false),
+      _outputDeviceIsSpecified(false),
+      _handleRecord(NULL),
+      _handlePlayout(NULL),
+      _recordingBuffersizeInFrame(0),
+      _recordingPeriodSizeInFrame(0),
+      _playoutBufferSizeInFrame(0),
+      _playoutPeriodSizeInFrame(0),
+      _recordingBufferSizeIn10MS(0),
+      _playoutBufferSizeIn10MS(0),
+      _recordingFramesIn10MS(0),
+      _playoutFramesIn10MS(0),
+      _recordingFreq(ALSA_CAPTURE_FREQ),
+      _playoutFreq(ALSA_PLAYOUT_FREQ),
+      _recChannels(ALSA_CAPTURE_CH),
+      _playChannels(ALSA_PLAYOUT_CH),
+      _recordingBuffer(NULL),
+      _playoutBuffer(NULL),
+      _recordingFramesLeft(0),
+      _playoutFramesLeft(0),
+      _initialized(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _recordingDelay(0),
+      _playoutDelay(0) {
+  memset(_oldKeyState, 0, sizeof(_oldKeyState));
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+}
+
+// ----------------------------------------------------------------------------
+//  AudioDeviceLinuxALSA - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+
+  Terminate();
+
+  // Clean up the recording buffer and playout buffer.
+  if (_recordingBuffer) {
+    delete[] _recordingBuffer;
+    _recordingBuffer = NULL;
+  }
+  if (_playoutBuffer) {
+    delete[] _playoutBuffer;
+    _playoutBuffer = NULL;
+  }
+}
+
+void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  MutexLock lock(&mutex_);
+
+  _ptrAudioBuffer = audioBuffer;
+
+  // Inform the AudioBuffer about default settings for this implementation.
+  // Set all values to zero here since the actual settings will be done by
+  // InitPlayout and InitRecording later.
+  _ptrAudioBuffer->SetRecordingSampleRate(0);
+  _ptrAudioBuffer->SetPlayoutSampleRate(0);
+  _ptrAudioBuffer->SetRecordingChannels(0);
+  _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+int32_t AudioDeviceLinuxALSA::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kLinuxAlsaAudio;
+  return 0;
+}
+
+AudioDeviceGeneric::InitStatus AudioDeviceLinuxALSA::Init() {
+  MutexLock lock(&mutex_);
+
+  // Load libasound
+  if (!GetAlsaSymbolTable()->Load()) {
+    // Alsa is not installed on this system
+    RTC_LOG(LS_ERROR) << "failed to load symbol table";
+    return InitStatus::OTHER_ERROR;
+  }
+
+  if (_initialized) {
+    return InitStatus::OK;
+  }
+#if defined(WEBRTC_USE_X11)
+  // Get X display handle for typing detection
+  _XDisplay = XOpenDisplay(NULL);
+  if (!_XDisplay) {
+    RTC_LOG(LS_WARNING)
+        << "failed to open X display, typing detection will not work";
+  }
+#endif
+
+  _initialized = true;
+
+  return InitStatus::OK;
+}
+
+int32_t AudioDeviceLinuxALSA::Terminate() {
+  if (!_initialized) {
+    return 0;
+  }
+
+  MutexLock lock(&mutex_);
+
+  _mixerManager.Close();
+
+  // RECORDING
+  mutex_.Unlock();
+  _ptrThreadRec.Finalize();
+
+  // PLAYOUT
+  _ptrThreadPlay.Finalize();
+  mutex_.Lock();
+
+#if defined(WEBRTC_USE_X11)
+  if (_XDisplay) {
+    XCloseDisplay(_XDisplay);
+    _XDisplay = NULL;
+  }
+#endif
+  _initialized = false;
+  _outputDeviceIsSpecified = false;
+  _inputDeviceIsSpecified = false;
+
+  return 0;
+}
+
+bool AudioDeviceLinuxALSA::Initialized() const {
+  return (_initialized);
+}
+
+int32_t AudioDeviceLinuxALSA::InitSpeaker() {
+  MutexLock lock(&mutex_);
+  return InitSpeakerLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::InitSpeakerLocked() {
+  if (_playing) {
+    return -1;
+  }
+
+  char devName[kAdmMaxDeviceNameSize] = {0};
+  GetDevicesInfo(2, true, _outputDeviceIndex, devName, kAdmMaxDeviceNameSize);
+  return _mixerManager.OpenSpeaker(devName);
+}
+
+int32_t AudioDeviceLinuxALSA::InitMicrophone() {
+  MutexLock lock(&mutex_);
+  return InitMicrophoneLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::InitMicrophoneLocked() {
+  if (_recording) {
+    return -1;
+  }
+
+  char devName[kAdmMaxDeviceNameSize] = {0};
+  GetDevicesInfo(2, false, _inputDeviceIndex, devName, kAdmMaxDeviceNameSize);
+  return _mixerManager.OpenMicrophone(devName);
+}
+
+bool AudioDeviceLinuxALSA::SpeakerIsInitialized() const {
+  return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceLinuxALSA::MicrophoneIsInitialized() const {
+  return (_mixerManager.MicrophoneIsInitialized());
+}
+
+int32_t AudioDeviceLinuxALSA::SpeakerVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know that a volume control
+  // exists
+  available = true;
+
+  // Close the initialized output mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetSpeakerVolume(uint32_t volume) {
+  return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+int32_t AudioDeviceLinuxALSA::SpeakerVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.SpeakerVolume(level) == -1) {
+    return -1;
+  }
+
+  volume = level;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MinSpeakerVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinSpeakerVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SpeakerMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected speaker has a mute control
+  _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+  available = isAvailable;
+
+  // Close the initialized output mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetSpeakerMute(bool enable) {
+  return (_mixerManager.SetSpeakerMute(enable));
+}
+
+int32_t AudioDeviceLinuxALSA::SpeakerMute(bool& enabled) const {
+  bool muted(0);
+
+  if (_mixerManager.SpeakerMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MicrophoneMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone has a mute control
+  //
+  _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetMicrophoneMute(bool enable) {
+  return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+// ----------------------------------------------------------------------------
+//  MicrophoneMute
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceLinuxALSA::MicrophoneMute(bool& enabled) const {
+  bool muted(0);
+
+  if (_mixerManager.MicrophoneMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+
+  // If we already have initialized in stereo it's obviously available
+  if (_recIsInitialized && (2 == _recChannels)) {
+    available = true;
+    return 0;
+  }
+
+  // Save rec states and the number of rec channels
+  bool recIsInitialized = _recIsInitialized;
+  bool recording = _recording;
+  int recChannels = _recChannels;
+
+  available = false;
+
+  // Stop/uninitialize recording if initialized (and possibly started)
+  if (_recIsInitialized) {
+    StopRecordingLocked();
+  }
+
+  // Try init in stereo;
+  _recChannels = 2;
+  if (InitRecordingLocked() == 0) {
+    available = true;
+  }
+
+  // Stop/uninitialize recording
+  StopRecordingLocked();
+
+  // Recover previous states
+  _recChannels = recChannels;
+  if (recIsInitialized) {
+    InitRecordingLocked();
+  }
+  if (recording) {
+    StartRecording();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetStereoRecording(bool enable) {
+  if (enable)
+    _recChannels = 2;
+  else
+    _recChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const {
+  if (_recChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+
+  // If we already have initialized in stereo it's obviously available
+  if (_playIsInitialized && (2 == _playChannels)) {
+    available = true;
+    return 0;
+  }
+
+  // Save rec states and the number of rec channels
+  bool playIsInitialized = _playIsInitialized;
+  bool playing = _playing;
+  int playChannels = _playChannels;
+
+  available = false;
+
+  // Stop/uninitialize recording if initialized (and possibly started)
+  if (_playIsInitialized) {
+    StopPlayoutLocked();
+  }
+
+  // Try init in stereo;
+  _playChannels = 2;
+  if (InitPlayoutLocked() == 0) {
+    available = true;
+  }
+
+  // Stop/uninitialize recording
+  StopPlayoutLocked();
+
+  // Recover previous states
+  _playChannels = playChannels;
+  if (playIsInitialized) {
+    InitPlayoutLocked();
+  }
+  if (playing) {
+    StartPlayout();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetStereoPlayout(bool enable) {
+  if (enable)
+    _playChannels = 2;
+  else
+    _playChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StereoPlayout(bool& enabled) const {
+  if (_playChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MicrophoneVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a volume control
+  // exists
+  available = true;
+
+  // Close the initialized input mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetMicrophoneVolume(uint32_t volume) {
+  return (_mixerManager.SetMicrophoneVolume(volume));
+}
+
+int32_t AudioDeviceLinuxALSA::MicrophoneVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.MicrophoneVolume(level) == -1) {
+    RTC_LOG(LS_WARNING) << "failed to retrive current microphone level";
+    return -1;
+  }
+
+  volume = level;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::MinMicrophoneVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+
+  return 0;
+}
+
+int16_t AudioDeviceLinuxALSA::PlayoutDevices() {
+  return (int16_t)GetDevicesInfo(0, true);
+}
+
+int32_t AudioDeviceLinuxALSA::SetPlayoutDevice(uint16_t index) {
+  if (_playIsInitialized) {
+    return -1;
+  }
+
+  uint32_t nDevices = GetDevicesInfo(0, true);
+  RTC_LOG(LS_VERBOSE) << "number of available audio output devices is "
+                      << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _outputDeviceIndex = index;
+  _outputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceLinuxALSA::PlayoutDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(PlayoutDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDevicesInfo(1, true, index, name, kAdmMaxDeviceNameSize);
+}
+
+int32_t AudioDeviceLinuxALSA::RecordingDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(RecordingDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDevicesInfo(1, false, index, name, kAdmMaxDeviceNameSize);
+}
+
+int16_t AudioDeviceLinuxALSA::RecordingDevices() {
+  return (int16_t)GetDevicesInfo(0, false);
+}
+
+int32_t AudioDeviceLinuxALSA::SetRecordingDevice(uint16_t index) {
+  if (_recIsInitialized) {
+    return -1;
+  }
+
+  uint32_t nDevices = GetDevicesInfo(0, false);
+  RTC_LOG(LS_VERBOSE) << "number of availiable audio input devices is "
+                      << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _inputDeviceIndex = index;
+  _inputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetRecordingDevice II (II)
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceLinuxALSA::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceLinuxALSA::PlayoutIsAvailable(bool& available) {
+  available = false;
+
+  // Try to initialize the playout side with mono
+  // Assumes that user set num channels after calling this function
+  _playChannels = 1;
+  int32_t res = InitPlayout();
+
+  // Cancel effect of initialization
+  StopPlayout();
+
+  if (res != -1) {
+    available = true;
+  } else {
+    // It may be possible to play out in stereo
+    res = StereoPlayoutIsAvailable(available);
+    if (available) {
+      // Then set channels to 2 so InitPlayout doesn't fail
+      _playChannels = 2;
+    }
+  }
+
+  return res;
+}
+
+int32_t AudioDeviceLinuxALSA::RecordingIsAvailable(bool& available) {
+  available = false;
+
+  // Try to initialize the recording side with mono
+  // Assumes that user set num channels after calling this function
+  _recChannels = 1;
+  int32_t res = InitRecording();
+
+  // Cancel effect of initialization
+  StopRecording();
+
+  if (res != -1) {
+    available = true;
+  } else {
+    // It may be possible to record in stereo
+    res = StereoRecordingIsAvailable(available);
+    if (available) {
+      // Then set channels to 2 so InitPlayout doesn't fail
+      _recChannels = 2;
+    }
+  }
+
+  return res;
+}
+
+int32_t AudioDeviceLinuxALSA::InitPlayout() {
+  MutexLock lock(&mutex_);
+  return InitPlayoutLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::InitPlayoutLocked() {
+  int errVal = 0;
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_playIsInitialized) {
+    return 0;
+  }
+  // Initialize the speaker (devices might have been added or removed)
+  if (InitSpeakerLocked() == -1) {
+    RTC_LOG(LS_WARNING) << "InitSpeaker() failed";
+  }
+
+  // Start by closing any existing wave-output devices
+  //
+  if (_handlePlayout != NULL) {
+    LATE(snd_pcm_close)(_handlePlayout);
+    _handlePlayout = NULL;
+    _playIsInitialized = false;
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error closing current playout sound device, error: "
+                        << LATE(snd_strerror)(errVal);
+    }
+  }
+
+  // Open PCM device for playout
+  char deviceName[kAdmMaxDeviceNameSize] = {0};
+  GetDevicesInfo(2, true, _outputDeviceIndex, deviceName,
+                 kAdmMaxDeviceNameSize);
+
+  RTC_LOG(LS_VERBOSE) << "InitPlayout open (" << deviceName << ")";
+
+  errVal = LATE(snd_pcm_open)(&_handlePlayout, deviceName,
+                              SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+
+  if (errVal == -EBUSY)  // Device busy - try some more!
+  {
+    for (int i = 0; i < 5; i++) {
+      SleepMs(1000);
+      errVal = LATE(snd_pcm_open)(&_handlePlayout, deviceName,
+                                  SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+      if (errVal == 0) {
+        break;
+      }
+    }
+  }
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "unable to open playback device: "
+                      << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
+    _handlePlayout = NULL;
+    return -1;
+  }
+
+  _playoutFramesIn10MS = _playoutFreq / 100;
+  if ((errVal = LATE(snd_pcm_set_params)(
+           _handlePlayout,
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+           SND_PCM_FORMAT_S16_BE,
+#else
+           SND_PCM_FORMAT_S16_LE,                             // format
+#endif
+           SND_PCM_ACCESS_RW_INTERLEAVED,  // access
+           _playChannels,                  // channels
+           _playoutFreq,                   // rate
+           1,                              // soft_resample
+           ALSA_PLAYOUT_LATENCY  // 40*1000 //latency required overall latency
+                                 // in us
+           )) < 0) {             /* 0.5sec */
+    _playoutFramesIn10MS = 0;
+    RTC_LOG(LS_ERROR) << "unable to set playback device: "
+                      << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
+    ErrorRecovery(errVal, _handlePlayout);
+    errVal = LATE(snd_pcm_close)(_handlePlayout);
+    _handlePlayout = NULL;
+    return -1;
+  }
+
+  errVal = LATE(snd_pcm_get_params)(_handlePlayout, &_playoutBufferSizeInFrame,
+                                    &_playoutPeriodSizeInFrame);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_pcm_get_params: " << LATE(snd_strerror)(errVal)
+                      << " (" << errVal << ")";
+    _playoutBufferSizeInFrame = 0;
+    _playoutPeriodSizeInFrame = 0;
+  } else {
+    RTC_LOG(LS_VERBOSE) << "playout snd_pcm_get_params buffer_size:"
+                        << _playoutBufferSizeInFrame
+                        << " period_size :" << _playoutPeriodSizeInFrame;
+  }
+
+  if (_ptrAudioBuffer) {
+    // Update webrtc audio buffer with the selected parameters
+    _ptrAudioBuffer->SetPlayoutSampleRate(_playoutFreq);
+    _ptrAudioBuffer->SetPlayoutChannels(_playChannels);
+  }
+
+  // Set play buffer size
+  _playoutBufferSizeIn10MS =
+      LATE(snd_pcm_frames_to_bytes)(_handlePlayout, _playoutFramesIn10MS);
+
+  // Init varaibles used for play
+
+  if (_handlePlayout != NULL) {
+    _playIsInitialized = true;
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+int32_t AudioDeviceLinuxALSA::InitRecording() {
+  MutexLock lock(&mutex_);
+  return InitRecordingLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::InitRecordingLocked() {
+  int errVal = 0;
+
+  if (_recording) {
+    return -1;
+  }
+
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_recIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the microphone (devices might have been added or removed)
+  if (InitMicrophoneLocked() == -1) {
+    RTC_LOG(LS_WARNING) << "InitMicrophone() failed";
+  }
+
+  // Start by closing any existing pcm-input devices
+  //
+  if (_handleRecord != NULL) {
+    int errVal = LATE(snd_pcm_close)(_handleRecord);
+    _handleRecord = NULL;
+    _recIsInitialized = false;
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR)
+          << "Error closing current recording sound device, error: "
+          << LATE(snd_strerror)(errVal);
+    }
+  }
+
+  // Open PCM device for recording
+  // The corresponding settings for playout are made after the record settings
+  char deviceName[kAdmMaxDeviceNameSize] = {0};
+  GetDevicesInfo(2, false, _inputDeviceIndex, deviceName,
+                 kAdmMaxDeviceNameSize);
+
+  RTC_LOG(LS_VERBOSE) << "InitRecording open (" << deviceName << ")";
+  errVal = LATE(snd_pcm_open)(&_handleRecord, deviceName,
+                              SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
+
+  // Available modes: 0 = blocking, SND_PCM_NONBLOCK, SND_PCM_ASYNC
+  if (errVal == -EBUSY)  // Device busy - try some more!
+  {
+    for (int i = 0; i < 5; i++) {
+      SleepMs(1000);
+      errVal = LATE(snd_pcm_open)(&_handleRecord, deviceName,
+                                  SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
+      if (errVal == 0) {
+        break;
+      }
+    }
+  }
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "unable to open record device: "
+                      << LATE(snd_strerror)(errVal);
+    _handleRecord = NULL;
+    return -1;
+  }
+
+  _recordingFramesIn10MS = _recordingFreq / 100;
+  if ((errVal =
+           LATE(snd_pcm_set_params)(_handleRecord,
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+                                    SND_PCM_FORMAT_S16_BE,  // format
+#else
+                                    SND_PCM_FORMAT_S16_LE,    // format
+#endif
+                                    SND_PCM_ACCESS_RW_INTERLEAVED,  // access
+                                    _recChannels,                   // channels
+                                    _recordingFreq,                 // rate
+                                    1,                    // soft_resample
+                                    ALSA_CAPTURE_LATENCY  // latency in us
+                                    )) < 0) {
+    // Fall back to another mode then.
+    if (_recChannels == 1)
+      _recChannels = 2;
+    else
+      _recChannels = 1;
+
+    if ((errVal =
+             LATE(snd_pcm_set_params)(_handleRecord,
+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
+                                      SND_PCM_FORMAT_S16_BE,  // format
+#else
+                                      SND_PCM_FORMAT_S16_LE,  // format
+#endif
+                                      SND_PCM_ACCESS_RW_INTERLEAVED,  // access
+                                      _recChannels,         // channels
+                                      _recordingFreq,       // rate
+                                      1,                    // soft_resample
+                                      ALSA_CAPTURE_LATENCY  // latency in us
+                                      )) < 0) {
+      _recordingFramesIn10MS = 0;
+      RTC_LOG(LS_ERROR) << "unable to set record settings: "
+                        << LATE(snd_strerror)(errVal) << " (" << errVal << ")";
+      ErrorRecovery(errVal, _handleRecord);
+      errVal = LATE(snd_pcm_close)(_handleRecord);
+      _handleRecord = NULL;
+      return -1;
+    }
+  }
+
+  errVal = LATE(snd_pcm_get_params)(_handleRecord, &_recordingBuffersizeInFrame,
+                                    &_recordingPeriodSizeInFrame);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_pcm_get_params " << LATE(snd_strerror)(errVal)
+                      << " (" << errVal << ")";
+    _recordingBuffersizeInFrame = 0;
+    _recordingPeriodSizeInFrame = 0;
+  } else {
+    RTC_LOG(LS_VERBOSE) << "capture snd_pcm_get_params, buffer_size:"
+                        << _recordingBuffersizeInFrame
+                        << ", period_size:" << _recordingPeriodSizeInFrame;
+  }
+
+  if (_ptrAudioBuffer) {
+    // Update webrtc audio buffer with the selected parameters
+    _ptrAudioBuffer->SetRecordingSampleRate(_recordingFreq);
+    _ptrAudioBuffer->SetRecordingChannels(_recChannels);
+  }
+
+  // Set rec buffer size and create buffer
+  _recordingBufferSizeIn10MS =
+      LATE(snd_pcm_frames_to_bytes)(_handleRecord, _recordingFramesIn10MS);
+
+  if (_handleRecord != NULL) {
+    // Mark recording side as initialized
+    _recIsInitialized = true;
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+int32_t AudioDeviceLinuxALSA::StartRecording() {
+  if (!_recIsInitialized) {
+    return -1;
+  }
+
+  if (_recording) {
+    return 0;
+  }
+
+  _recording = true;
+
+  int errVal = 0;
+  _recordingFramesLeft = _recordingFramesIn10MS;
+
+  // Make sure we only create the buffer once.
+  if (!_recordingBuffer)
+    _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
+  if (!_recordingBuffer) {
+    RTC_LOG(LS_ERROR) << "failed to alloc recording buffer";
+    _recording = false;
+    return -1;
+  }
+  // RECORDING
+  _ptrThreadRec = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (RecThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_capture_thread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  errVal = LATE(snd_pcm_prepare)(_handleRecord);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "capture snd_pcm_prepare failed ("
+                      << LATE(snd_strerror)(errVal) << ")\n";
+    // just log error
+    // if snd_pcm_open fails will return -1
+  }
+
+  errVal = LATE(snd_pcm_start)(_handleRecord);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "capture snd_pcm_start err: "
+                      << LATE(snd_strerror)(errVal);
+    errVal = LATE(snd_pcm_start)(_handleRecord);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "capture snd_pcm_start 2nd try err: "
+                        << LATE(snd_strerror)(errVal);
+      StopRecording();
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StopRecording() {
+    MutexLock lock(&mutex_);
+    return StopRecordingLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::StopRecordingLocked() {
+  if (!_recIsInitialized) {
+    return 0;
+  }
+
+  if (_handleRecord == NULL) {
+    return -1;
+  }
+
+  // Make sure we don't start recording (it's asynchronous).
+  _recIsInitialized = false;
+  _recording = false;
+
+  _ptrThreadRec.Finalize();
+
+  _recordingFramesLeft = 0;
+  if (_recordingBuffer) {
+    delete[] _recordingBuffer;
+    _recordingBuffer = NULL;
+  }
+
+  // Stop and close pcm recording device.
+  int errVal = LATE(snd_pcm_drop)(_handleRecord);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error stop recording: " << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  errVal = LATE(snd_pcm_close)(_handleRecord);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error closing record sound device, error: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  // Check if we have muted and unmute if so.
+  bool muteEnabled = false;
+  MicrophoneMute(muteEnabled);
+  if (muteEnabled) {
+    SetMicrophoneMute(false);
+  }
+
+  // set the pcm input handle to NULL
+  _handleRecord = NULL;
+  return 0;
+}
+
+bool AudioDeviceLinuxALSA::RecordingIsInitialized() const {
+  return (_recIsInitialized);
+}
+
+bool AudioDeviceLinuxALSA::Recording() const {
+  return (_recording);
+}
+
+bool AudioDeviceLinuxALSA::PlayoutIsInitialized() const {
+  return (_playIsInitialized);
+}
+
+int32_t AudioDeviceLinuxALSA::StartPlayout() {
+  if (!_playIsInitialized) {
+    return -1;
+  }
+
+  if (_playing) {
+    return 0;
+  }
+
+  _playing = true;
+
+  _playoutFramesLeft = 0;
+  if (!_playoutBuffer)
+    _playoutBuffer = new int8_t[_playoutBufferSizeIn10MS];
+  if (!_playoutBuffer) {
+    RTC_LOG(LS_ERROR) << "failed to alloc playout buf";
+    _playing = false;
+    return -1;
+  }
+
+  // PLAYOUT
+  _ptrThreadPlay = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (PlayThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_play_thread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "playout snd_pcm_prepare failed ("
+                      << LATE(snd_strerror)(errVal) << ")\n";
+    // just log error
+    // if snd_pcm_open fails will return -1
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::StopPlayout() {
+    MutexLock lock(&mutex_);
+    return StopPlayoutLocked();
+}
+
+int32_t AudioDeviceLinuxALSA::StopPlayoutLocked() {
+  if (!_playIsInitialized) {
+    return 0;
+  }
+
+  if (_handlePlayout == NULL) {
+    return -1;
+  }
+
+  _playing = false;
+
+  // stop playout thread first
+  _ptrThreadPlay.Finalize();
+
+  _playoutFramesLeft = 0;
+  delete[] _playoutBuffer;
+  _playoutBuffer = NULL;
+
+  // stop and close pcm playout device
+  int errVal = LATE(snd_pcm_drop)(_handlePlayout);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error stop playing: " << LATE(snd_strerror)(errVal);
+  }
+
+  errVal = LATE(snd_pcm_close)(_handlePlayout);
+  if (errVal < 0)
+    RTC_LOG(LS_ERROR) << "Error closing playout sound device, error: "
+                      << LATE(snd_strerror)(errVal);
+
+  // set the pcm input handle to NULL
+  _playIsInitialized = false;
+  _handlePlayout = NULL;
+  RTC_LOG(LS_VERBOSE) << "handle_playout is now set to NULL";
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::PlayoutDelay(uint16_t& delayMS) const {
+  delayMS = (uint16_t)_playoutDelay * 1000 / _playoutFreq;
+  return 0;
+}
+
+bool AudioDeviceLinuxALSA::Playing() const {
+  return (_playing);
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+int32_t AudioDeviceLinuxALSA::GetDevicesInfo(const int32_t function,
+                                             const bool playback,
+                                             const int32_t enumDeviceNo,
+                                             char* enumDeviceName,
+                                             const int32_t ednLen) const {
+  // Device enumeration based on libjingle implementation
+  // by Tristan Schmelcher at Google Inc.
+
+  const char* type = playback ? "Output" : "Input";
+  // dmix and dsnoop are only for playback and capture, respectively, but ALSA
+  // stupidly includes them in both lists.
+  const char* ignorePrefix = playback ? "dsnoop:" : "dmix:";
+  // (ALSA lists many more "devices" of questionable interest, but we show them
+  // just in case the weird devices may actually be desirable for some
+  // users/systems.)
+
+  int err;
+  int enumCount(0);
+  bool keepSearching(true);
+
+  // From Chromium issue 95797
+  // Loop through the sound cards to get Alsa device hints.
+  // Don't use snd_device_name_hint(-1,..) since there is a access violation
+  // inside this ALSA API with libasound.so.2.0.0.
+  int card = -1;
+  while (!(LATE(snd_card_next)(&card)) && (card >= 0) && keepSearching) {
+    void** hints;
+    err = LATE(snd_device_name_hint)(card, "pcm", &hints);
+    if (err != 0) {
+      RTC_LOG(LS_ERROR) << "GetDevicesInfo - device name hint error: "
+                        << LATE(snd_strerror)(err);
+      return -1;
+    }
+
+    enumCount++;  // default is 0
+    if ((function == FUNC_GET_DEVICE_NAME ||
+         function == FUNC_GET_DEVICE_NAME_FOR_AN_ENUM) &&
+        enumDeviceNo == 0) {
+      strcpy(enumDeviceName, "default");
+
+      err = LATE(snd_device_name_free_hint)(hints);
+      if (err != 0) {
+        RTC_LOG(LS_ERROR) << "GetDevicesInfo - device name free hint error: "
+                          << LATE(snd_strerror)(err);
+      }
+
+      return 0;
+    }
+
+    for (void** list = hints; *list != NULL; ++list) {
+      char* actualType = LATE(snd_device_name_get_hint)(*list, "IOID");
+      if (actualType) {  // NULL means it's both.
+        bool wrongType = (strcmp(actualType, type) != 0);
+        free(actualType);
+        if (wrongType) {
+          // Wrong type of device (i.e., input vs. output).
+          continue;
+        }
+      }
+
+      char* name = LATE(snd_device_name_get_hint)(*list, "NAME");
+      if (!name) {
+        RTC_LOG(LS_ERROR) << "Device has no name";
+        // Skip it.
+        continue;
+      }
+
+      // Now check if we actually want to show this device.
+      if (strcmp(name, "default") != 0 && strcmp(name, "null") != 0 &&
+          strcmp(name, "pulse") != 0 &&
+          strncmp(name, ignorePrefix, strlen(ignorePrefix)) != 0) {
+        // Yes, we do.
+        char* desc = LATE(snd_device_name_get_hint)(*list, "DESC");
+        if (!desc) {
+          // Virtual devices don't necessarily have descriptions.
+          // Use their names instead.
+          desc = name;
+        }
+
+        if (FUNC_GET_NUM_OF_DEVICE == function) {
+          RTC_LOG(LS_VERBOSE) << "Enum device " << enumCount << " - " << name;
+        }
+        if ((FUNC_GET_DEVICE_NAME == function) && (enumDeviceNo == enumCount)) {
+          // We have found the enum device, copy the name to buffer.
+          strncpy(enumDeviceName, desc, ednLen);
+          enumDeviceName[ednLen - 1] = '\0';
+          keepSearching = false;
+          // Replace '\n' with '-'.
+          char* pret = strchr(enumDeviceName, '\n' /*0xa*/);  // LF
+          if (pret)
+            *pret = '-';
+        }
+        if ((FUNC_GET_DEVICE_NAME_FOR_AN_ENUM == function) &&
+            (enumDeviceNo == enumCount)) {
+          // We have found the enum device, copy the name to buffer.
+          strncpy(enumDeviceName, name, ednLen);
+          enumDeviceName[ednLen - 1] = '\0';
+          keepSearching = false;
+        }
+
+        if (keepSearching)
+          ++enumCount;
+
+        if (desc != name)
+          free(desc);
+      }
+
+      free(name);
+
+      if (!keepSearching)
+        break;
+    }
+
+    err = LATE(snd_device_name_free_hint)(hints);
+    if (err != 0) {
+      RTC_LOG(LS_ERROR) << "GetDevicesInfo - device name free hint error: "
+                        << LATE(snd_strerror)(err);
+      // Continue and return true anyway, since we did get the whole list.
+    }
+  }
+
+  if (FUNC_GET_NUM_OF_DEVICE == function) {
+    if (enumCount == 1)  // only default?
+      enumCount = 0;
+    return enumCount;  // Normal return point for function 0
+  }
+
+  if (keepSearching) {
+    // If we get here for function 1 and 2, we didn't find the specified
+    // enum device.
+    RTC_LOG(LS_ERROR)
+        << "GetDevicesInfo - Could not find device name or numbers";
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const {
+  if (_handleRecord == NULL) {
+    RTC_LOG(LS_ERROR) << "input state has been modified during unlocked period";
+    return -1;
+  }
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::OutputSanityCheckAfterUnlockedPeriod() const {
+  if (_handlePlayout == NULL) {
+    RTC_LOG(LS_ERROR)
+        << "output state has been modified during unlocked period";
+    return -1;
+  }
+  return 0;
+}
+
+int32_t AudioDeviceLinuxALSA::ErrorRecovery(int32_t error,
+                                            snd_pcm_t* deviceHandle) {
+  int st = LATE(snd_pcm_state)(deviceHandle);
+  RTC_LOG(LS_VERBOSE) << "Trying to recover from "
+                      << ((LATE(snd_pcm_stream)(deviceHandle) ==
+                           SND_PCM_STREAM_CAPTURE)
+                              ? "capture"
+                              : "playout")
+                      << " error: " << LATE(snd_strerror)(error) << " ("
+                      << error << ") (state " << st << ")";
+
+  // It is recommended to use snd_pcm_recover for all errors. If that function
+  // cannot handle the error, the input error code will be returned, otherwise
+  // 0 is returned. From snd_pcm_recover API doc: "This functions handles
+  // -EINTR (4) (interrupted system call), -EPIPE (32) (playout overrun or
+  // capture underrun) and -ESTRPIPE (86) (stream is suspended) error codes
+  // trying to prepare given stream for next I/O."
+
+  /** Open */
+  //    SND_PCM_STATE_OPEN = 0,
+  /** Setup installed */
+  //    SND_PCM_STATE_SETUP,
+  /** Ready to start */
+  //    SND_PCM_STATE_PREPARED,
+  /** Running */
+  //    SND_PCM_STATE_RUNNING,
+  /** Stopped: underrun (playback) or overrun (capture) detected */
+  //    SND_PCM_STATE_XRUN,= 4
+  /** Draining: running (playback) or stopped (capture) */
+  //    SND_PCM_STATE_DRAINING,
+  /** Paused */
+  //    SND_PCM_STATE_PAUSED,
+  /** Hardware is suspended */
+  //    SND_PCM_STATE_SUSPENDED,
+  //  ** Hardware is disconnected */
+  //    SND_PCM_STATE_DISCONNECTED,
+  //    SND_PCM_STATE_LAST = SND_PCM_STATE_DISCONNECTED
+
+  // snd_pcm_recover isn't available in older alsa, e.g. on the FC4 machine
+  // in Sthlm lab.
+
+  int res = LATE(snd_pcm_recover)(deviceHandle, error, 1);
+  if (0 == res) {
+    RTC_LOG(LS_VERBOSE) << "Recovery - snd_pcm_recover OK";
+
+    if ((error == -EPIPE || error == -ESTRPIPE) &&  // Buf underrun/overrun.
+        _recording &&
+        LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE) {
+      // For capture streams we also have to repeat the explicit start()
+      // to get data flowing again.
+      int err = LATE(snd_pcm_start)(deviceHandle);
+      if (err != 0) {
+        RTC_LOG(LS_ERROR) << "Recovery - snd_pcm_start error: " << err;
+        return -1;
+      }
+    }
+
+    if ((error == -EPIPE || error == -ESTRPIPE) &&  // Buf underrun/overrun.
+        _playing &&
+        LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_PLAYBACK) {
+      // For capture streams we also have to repeat the explicit start() to get
+      // data flowing again.
+      int err = LATE(snd_pcm_start)(deviceHandle);
+      if (err != 0) {
+        RTC_LOG(LS_ERROR) << "Recovery - snd_pcm_start error: "
+                          << LATE(snd_strerror)(err);
+        return -1;
+      }
+    }
+
+    return -EPIPE == error ? 1 : 0;
+  } else {
+    RTC_LOG(LS_ERROR) << "Unrecoverable alsa stream error: " << res;
+  }
+
+  return res;
+}
+
+// ============================================================================
+//                                  Thread Methods
+// ============================================================================
+
+bool AudioDeviceLinuxALSA::PlayThreadProcess() {
+  if (!_playing)
+    return false;
+
+  int err;
+  snd_pcm_sframes_t frames;
+  snd_pcm_sframes_t avail_frames;
+
+  Lock();
+  // return a positive number of frames ready otherwise a negative error code
+  avail_frames = LATE(snd_pcm_avail_update)(_handlePlayout);
+  if (avail_frames < 0) {
+    RTC_LOG(LS_ERROR) << "playout snd_pcm_avail_update error: "
+                      << LATE(snd_strerror)(avail_frames);
+    ErrorRecovery(avail_frames, _handlePlayout);
+    UnLock();
+    return true;
+  } else if (avail_frames == 0) {
+    UnLock();
+
+    // maximum tixe in milliseconds to wait, a negative value means infinity
+    err = LATE(snd_pcm_wait)(_handlePlayout, 2);
+    if (err == 0) {  // timeout occured
+      RTC_LOG(LS_VERBOSE) << "playout snd_pcm_wait timeout";
+    }
+
+    return true;
+  }
+
+  if (_playoutFramesLeft <= 0) {
+    UnLock();
+    _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
+    Lock();
+
+    _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
+    RTC_DCHECK_EQ(_playoutFramesLeft, _playoutFramesIn10MS);
+  }
+
+  if (static_cast<uint32_t>(avail_frames) > _playoutFramesLeft)
+    avail_frames = _playoutFramesLeft;
+
+  int size = LATE(snd_pcm_frames_to_bytes)(_handlePlayout, _playoutFramesLeft);
+  frames = LATE(snd_pcm_writei)(
+      _handlePlayout, &_playoutBuffer[_playoutBufferSizeIn10MS - size],
+      avail_frames);
+
+  if (frames < 0) {
+    RTC_LOG(LS_VERBOSE) << "playout snd_pcm_writei error: "
+                        << LATE(snd_strerror)(frames);
+    _playoutFramesLeft = 0;
+    ErrorRecovery(frames, _handlePlayout);
+    UnLock();
+    return true;
+  } else {
+    RTC_DCHECK_EQ(frames, avail_frames);
+    _playoutFramesLeft -= frames;
+  }
+
+  UnLock();
+  return true;
+}
+
+bool AudioDeviceLinuxALSA::RecThreadProcess() {
+  if (!_recording)
+    return false;
+
+  int err;
+  snd_pcm_sframes_t frames;
+  snd_pcm_sframes_t avail_frames;
+  int8_t buffer[_recordingBufferSizeIn10MS];
+
+  Lock();
+
+  // return a positive number of frames ready otherwise a negative error code
+  avail_frames = LATE(snd_pcm_avail_update)(_handleRecord);
+  if (avail_frames < 0) {
+    RTC_LOG(LS_ERROR) << "capture snd_pcm_avail_update error: "
+                      << LATE(snd_strerror)(avail_frames);
+    ErrorRecovery(avail_frames, _handleRecord);
+    UnLock();
+    return true;
+  } else if (avail_frames == 0) {  // no frame is available now
+    UnLock();
+
+    // maximum time in milliseconds to wait, a negative value means infinity
+    err = LATE(snd_pcm_wait)(_handleRecord, ALSA_CAPTURE_WAIT_TIMEOUT);
+    if (err == 0)  // timeout occured
+      RTC_LOG(LS_VERBOSE) << "capture snd_pcm_wait timeout";
+
+    return true;
+  }
+
+  if (static_cast<uint32_t>(avail_frames) > _recordingFramesLeft)
+    avail_frames = _recordingFramesLeft;
+
+  frames = LATE(snd_pcm_readi)(_handleRecord, buffer,
+                               avail_frames);  // frames to be written
+  if (frames < 0) {
+    RTC_LOG(LS_ERROR) << "capture snd_pcm_readi error: "
+                      << LATE(snd_strerror)(frames);
+    ErrorRecovery(frames, _handleRecord);
+    UnLock();
+    return true;
+  } else if (frames > 0) {
+    RTC_DCHECK_EQ(frames, avail_frames);
+
+    int left_size =
+        LATE(snd_pcm_frames_to_bytes)(_handleRecord, _recordingFramesLeft);
+    int size = LATE(snd_pcm_frames_to_bytes)(_handleRecord, frames);
+
+    memcpy(&_recordingBuffer[_recordingBufferSizeIn10MS - left_size], buffer,
+           size);
+    _recordingFramesLeft -= frames;
+
+    if (!_recordingFramesLeft) {  // buf is full
+      _recordingFramesLeft = _recordingFramesIn10MS;
+
+      // store the recorded buffer (no action will be taken if the
+      // #recorded samples is not a full buffer)
+      _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+                                         _recordingFramesIn10MS);
+
+      // calculate delay
+      _playoutDelay = 0;
+      _recordingDelay = 0;
+      if (_handlePlayout) {
+        err = LATE(snd_pcm_delay)(_handlePlayout,
+                                  &_playoutDelay);  // returned delay in frames
+        if (err < 0) {
+          // TODO(xians): Shall we call ErrorRecovery() here?
+          _playoutDelay = 0;
+          RTC_LOG(LS_ERROR)
+              << "playout snd_pcm_delay: " << LATE(snd_strerror)(err);
+        }
+      }
+
+      err = LATE(snd_pcm_delay)(_handleRecord,
+                                &_recordingDelay);  // returned delay in frames
+      if (err < 0) {
+        // TODO(xians): Shall we call ErrorRecovery() here?
+        _recordingDelay = 0;
+        RTC_LOG(LS_ERROR) << "capture snd_pcm_delay: "
+                          << LATE(snd_strerror)(err);
+      }
+
+      // TODO(xians): Shall we add 10ms buffer delay to the record delay?
+      _ptrAudioBuffer->SetVQEData(_playoutDelay * 1000 / _playoutFreq,
+                                  _recordingDelay * 1000 / _recordingFreq);
+
+      _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
+      // Deliver recorded samples at specified sample rate, mic level etc.
+      // to the observer using callback.
+      UnLock();
+      _ptrAudioBuffer->DeliverRecordedData();
+      Lock();
+    }
+  }
+
+  UnLock();
+  return true;
+}
+
+bool AudioDeviceLinuxALSA::KeyPressed() const {
+#if defined(WEBRTC_USE_X11)
+  char szKey[32];
+  unsigned int i = 0;
+  char state = 0;
+
+  if (!_XDisplay)
+    return false;
+
+  // Check key map status
+  XQueryKeymap(_XDisplay, szKey);
+
+  // A bit change in keymap means a key is pressed
+  for (i = 0; i < sizeof(szKey); i++)
+    state |= (szKey[i] ^ _oldKeyState[i]) & szKey[i];
+
+  // Save old state
+  memcpy((char*)_oldKeyState, (char*)szKey, sizeof(_oldKeyState));
+  return (state != 0);
+#else
+  return false;
+#endif
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.h
new file mode 100644
index 0000000000..23e21d3ce9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_device_alsa_linux.h
@@ -0,0 +1,208 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H_
+
+#include <memory>
+
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+
+#if defined(WEBRTC_USE_X11)
+#include <X11/Xlib.h>
+#endif
+#include <alsa/asoundlib.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+
+typedef webrtc::adm_linux_alsa::AlsaSymbolTable WebRTCAlsaSymbolTable;
+WebRTCAlsaSymbolTable* GetAlsaSymbolTable();
+
+namespace webrtc {
+
+class AudioDeviceLinuxALSA : public AudioDeviceGeneric {
+ public:
+  AudioDeviceLinuxALSA();
+  virtual ~AudioDeviceLinuxALSA();
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+
+ private:
+  int32_t InitRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t StopRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t StopPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t GetDevicesInfo(int32_t function,
+                         bool playback,
+                         int32_t enumDeviceNo = 0,
+                         char* enumDeviceName = NULL,
+                         int32_t ednLen = 0) const;
+  int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
+
+  bool KeyPressed() const;
+
+  void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }
+  void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }
+
+  inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
+  inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
+
+  static void RecThreadFunc(void*);
+  static void PlayThreadFunc(void*);
+  bool RecThreadProcess();
+  bool PlayThreadProcess();
+
+  AudioDeviceBuffer* _ptrAudioBuffer;
+
+  Mutex mutex_;
+
+  rtc::PlatformThread _ptrThreadRec;
+  rtc::PlatformThread _ptrThreadPlay;
+
+  AudioMixerManagerLinuxALSA _mixerManager;
+
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+  bool _inputDeviceIsSpecified;
+  bool _outputDeviceIsSpecified;
+
+  snd_pcm_t* _handleRecord;
+  snd_pcm_t* _handlePlayout;
+
+  snd_pcm_uframes_t _recordingBuffersizeInFrame;
+  snd_pcm_uframes_t _recordingPeriodSizeInFrame;
+  snd_pcm_uframes_t _playoutBufferSizeInFrame;
+  snd_pcm_uframes_t _playoutPeriodSizeInFrame;
+
+  ssize_t _recordingBufferSizeIn10MS;
+  ssize_t _playoutBufferSizeIn10MS;
+  uint32_t _recordingFramesIn10MS;
+  uint32_t _playoutFramesIn10MS;
+
+  uint32_t _recordingFreq;
+  uint32_t _playoutFreq;
+  uint8_t _recChannels;
+  uint8_t _playChannels;
+
+  int8_t* _recordingBuffer;  // in byte
+  int8_t* _playoutBuffer;    // in byte
+  uint32_t _recordingFramesLeft;
+  uint32_t _playoutFramesLeft;
+
+  bool _initialized;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+
+  snd_pcm_sframes_t _recordingDelay;
+  snd_pcm_sframes_t _playoutDelay;
+
+  char _oldKeyState[32];
+#if defined(WEBRTC_USE_X11)
+  Display* _XDisplay;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_ALSA_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
new file mode 100644
index 0000000000..90cd58c497
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
@@ -0,0 +1,2286 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/linux/audio_device_pulse_linux.h"
+
+#include <string.h>
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+
+WebRTCPulseSymbolTable* GetPulseSymbolTable() {
+  static WebRTCPulseSymbolTable* pulse_symbol_table =
+      new WebRTCPulseSymbolTable();
+  return pulse_symbol_table;
+}
+
+// Accesses Pulse functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libpulse, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                             \
+  LATESYM_GET(webrtc::adm_linux_pulse::PulseAudioSymbolTable, \
+              GetPulseSymbolTable(), sym)
+
+namespace webrtc {
+
+AudioDeviceLinuxPulse::AudioDeviceLinuxPulse()
+    : _ptrAudioBuffer(NULL),
+      _inputDeviceIndex(0),
+      _outputDeviceIndex(0),
+      _inputDeviceIsSpecified(false),
+      _outputDeviceIsSpecified(false),
+      sample_rate_hz_(0),
+      _recChannels(1),
+      _playChannels(1),
+      _initialized(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _startRec(false),
+      _startPlay(false),
+      update_speaker_volume_at_startup_(false),
+      quit_(false),
+      _sndCardPlayDelay(0),
+      _writeErrors(0),
+      _deviceIndex(-1),
+      _numPlayDevices(0),
+      _numRecDevices(0),
+      _playDeviceName(NULL),
+      _recDeviceName(NULL),
+      _playDisplayDeviceName(NULL),
+      _recDisplayDeviceName(NULL),
+      _playBuffer(NULL),
+      _playbackBufferSize(0),
+      _playbackBufferUnused(0),
+      _tempBufferSpace(0),
+      _recBuffer(NULL),
+      _recordBufferSize(0),
+      _recordBufferUsed(0),
+      _tempSampleData(NULL),
+      _tempSampleDataSize(0),
+      _configuredLatencyPlay(0),
+      _configuredLatencyRec(0),
+      _paDeviceIndex(-1),
+      _paStateChanged(false),
+      _paMainloop(NULL),
+      _paMainloopApi(NULL),
+      _paContext(NULL),
+      _recStream(NULL),
+      _playStream(NULL),
+      _recStreamFlags(0),
+      _playStreamFlags(0) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+
+  memset(_paServerVersion, 0, sizeof(_paServerVersion));
+  memset(&_playBufferAttr, 0, sizeof(_playBufferAttr));
+  memset(&_recBufferAttr, 0, sizeof(_recBufferAttr));
+  memset(_oldKeyState, 0, sizeof(_oldKeyState));
+}
+
+AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+
+  if (_recBuffer) {
+    delete[] _recBuffer;
+    _recBuffer = NULL;
+  }
+  if (_playBuffer) {
+    delete[] _playBuffer;
+    _playBuffer = NULL;
+  }
+  if (_playDeviceName) {
+    delete[] _playDeviceName;
+    _playDeviceName = NULL;
+  }
+  if (_recDeviceName) {
+    delete[] _recDeviceName;
+    _recDeviceName = NULL;
+  }
+}
+
+void AudioDeviceLinuxPulse::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  _ptrAudioBuffer = audioBuffer;
+
+  // Inform the AudioBuffer about default settings for this implementation.
+  // Set all values to zero here since the actual settings will be done by
+  // InitPlayout and InitRecording later.
+  _ptrAudioBuffer->SetRecordingSampleRate(0);
+  _ptrAudioBuffer->SetPlayoutSampleRate(0);
+  _ptrAudioBuffer->SetRecordingChannels(0);
+  _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+// ----------------------------------------------------------------------------
+//  ActiveAudioLayer
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceLinuxPulse::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kLinuxPulseAudio;
+  return 0;
+}
+
+AudioDeviceGeneric::InitStatus AudioDeviceLinuxPulse::Init() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_initialized) {
+    return InitStatus::OK;
+  }
+
+  // Initialize PulseAudio
+  if (InitPulseAudio() < 0) {
+    RTC_LOG(LS_ERROR) << "failed to initialize PulseAudio";
+    if (TerminatePulseAudio() < 0) {
+      RTC_LOG(LS_ERROR) << "failed to terminate PulseAudio";
+    }
+    return InitStatus::OTHER_ERROR;
+  }
+
+#if defined(WEBRTC_USE_X11)
+  // Get X display handle for typing detection
+  _XDisplay = XOpenDisplay(NULL);
+  if (!_XDisplay) {
+    RTC_LOG(LS_WARNING)
+        << "failed to open X display, typing detection will not work";
+  }
+#endif
+
+  // RECORDING
+  const auto attributes =
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
+  _ptrThreadRec = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (RecThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_rec_thread", attributes);
+
+  // PLAYOUT
+  _ptrThreadPlay = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (PlayThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_play_thread", attributes);
+  _initialized = true;
+
+  return InitStatus::OK;
+}
+
+int32_t AudioDeviceLinuxPulse::Terminate() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!_initialized) {
+    return 0;
+  }
+  {
+    MutexLock lock(&mutex_);
+    quit_ = true;
+  }
+  _mixerManager.Close();
+
+  // RECORDING
+  _timeEventRec.Set();
+  _ptrThreadRec.Finalize();
+
+  // PLAYOUT
+  _timeEventPlay.Set();
+  _ptrThreadPlay.Finalize();
+
+  // Terminate PulseAudio
+  if (TerminatePulseAudio() < 0) {
+    RTC_LOG(LS_ERROR) << "failed to terminate PulseAudio";
+    return -1;
+  }
+
+#if defined(WEBRTC_USE_X11)
+  if (_XDisplay) {
+    XCloseDisplay(_XDisplay);
+    _XDisplay = NULL;
+  }
+#endif
+
+  _initialized = false;
+  _outputDeviceIsSpecified = false;
+  _inputDeviceIsSpecified = false;
+
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::Initialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_initialized);
+}
+
+int32_t AudioDeviceLinuxPulse::InitSpeaker() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
+
+  // check if default device
+  if (_outputDeviceIndex == 0) {
+    uint16_t deviceIndex = 0;
+    GetDefaultDeviceInfo(false, NULL, deviceIndex);
+    _paDeviceIndex = deviceIndex;
+  } else {
+    // get the PA device index from
+    // the callback
+    _deviceIndex = _outputDeviceIndex;
+
+    // get playout devices
+    PlayoutDevices();
+  }
+
+  // the callback has now set the _paDeviceIndex to
+  // the PulseAudio index of the device
+  if (_mixerManager.OpenSpeaker(_paDeviceIndex) == -1) {
+    return -1;
+  }
+
+  // clear _deviceIndex
+  _deviceIndex = -1;
+  _paDeviceIndex = -1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::InitMicrophone() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_recording) {
+    return -1;
+  }
+
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
+
+  // Check if default device
+  if (_inputDeviceIndex == 0) {
+    uint16_t deviceIndex = 0;
+    GetDefaultDeviceInfo(true, NULL, deviceIndex);
+    _paDeviceIndex = deviceIndex;
+  } else {
+    // Get the PA device index from
+    // the callback
+    _deviceIndex = _inputDeviceIndex;
+
+    // get recording devices
+    RecordingDevices();
+  }
+
+  // The callback has now set the _paDeviceIndex to
+  // the PulseAudio index of the device
+  if (_mixerManager.OpenMicrophone(_paDeviceIndex) == -1) {
+    return -1;
+  }
+
+  // Clear _deviceIndex
+  _deviceIndex = -1;
+  _paDeviceIndex = -1;
+
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::SpeakerIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceLinuxPulse::MicrophoneIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_mixerManager.MicrophoneIsInitialized());
+}
+
+int32_t AudioDeviceLinuxPulse::SpeakerVolumeIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know volume control exists.
+  available = true;
+
+  // Close the initialized output mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetSpeakerVolume(uint32_t volume) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!_playing) {
+    // Only update the volume if it's been set while we weren't playing.
+    update_speaker_volume_at_startup_ = true;
+  }
+  return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+int32_t AudioDeviceLinuxPulse::SpeakerVolume(uint32_t& volume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  uint32_t level(0);
+
+  if (_mixerManager.SpeakerVolume(level) == -1) {
+    return -1;
+  }
+
+  volume = level;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MinSpeakerVolume(uint32_t& minVolume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinSpeakerVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SpeakerMuteIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected speaker has a mute control
+  _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+  available = isAvailable;
+
+  // Close the initialized output mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetSpeakerMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_mixerManager.SetSpeakerMute(enable));
+}
+
+int32_t AudioDeviceLinuxPulse::SpeakerMute(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool muted(0);
+  if (_mixerManager.SpeakerMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MicrophoneMuteIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no
+    // volume control, hence it is safe to state that there is no
+    // boost control already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone has a mute control
+  //
+  _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetMicrophoneMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+int32_t AudioDeviceLinuxPulse::MicrophoneMute(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool muted(0);
+  if (_mixerManager.MicrophoneMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StereoRecordingIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_recChannels == 2 && _recording) {
+    available = true;
+    return 0;
+  }
+
+  available = false;
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+  int error = 0;
+
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // Cannot open the specified device
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone can record stereo.
+  bool isAvailable(false);
+  error = _mixerManager.StereoRecordingIsAvailable(isAvailable);
+  if (!error)
+    available = isAvailable;
+
+  // Close the initialized input mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return error;
+}
+
+int32_t AudioDeviceLinuxPulse::SetStereoRecording(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (enable)
+    _recChannels = 2;
+  else
+    _recChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StereoRecording(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_recChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StereoPlayoutIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_playChannels == 2 && _playing) {
+    available = true;
+    return 0;
+  }
+
+  available = false;
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+  int error = 0;
+
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // Cannot open the specified device.
+    return -1;
+  }
+
+  // Check if the selected speaker can play stereo.
+  bool isAvailable(false);
+  error = _mixerManager.StereoPlayoutIsAvailable(isAvailable);
+  if (!error)
+    available = isAvailable;
+
+  // Close the initialized input mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return error;
+}
+
+int32_t AudioDeviceLinuxPulse::SetStereoPlayout(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (enable)
+    _playChannels = 2;
+  else
+    _playChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_playChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MicrophoneVolumeIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no
+    // volume control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a volume control
+  // exists.
+  available = true;
+
+  // Close the initialized input mixer
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetMicrophoneVolume(uint32_t volume) {
+  return (_mixerManager.SetMicrophoneVolume(volume));
+}
+
+int32_t AudioDeviceLinuxPulse::MicrophoneVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.MicrophoneVolume(level) == -1) {
+    RTC_LOG(LS_WARNING) << "failed to retrieve current microphone level";
+    return -1;
+  }
+
+  volume = level;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::MinMicrophoneVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+
+  return 0;
+}
+
+int16_t AudioDeviceLinuxPulse::PlayoutDevices() {
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+  _numPlayDevices = 1;  // init to 1 to account for "default"
+
+  // get the whole list of devices and update _numPlayDevices
+  paOperation =
+      LATE(pa_context_get_sink_info_list)(_paContext, PaSinkInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  return _numPlayDevices;
+}
+
+int32_t AudioDeviceLinuxPulse::SetPlayoutDevice(uint16_t index) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_playIsInitialized) {
+    return -1;
+  }
+
+  const uint16_t nDevices = PlayoutDevices();
+
+  RTC_LOG(LS_VERBOSE) << "number of availiable output devices is " << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _outputDeviceIndex = index;
+  _outputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceLinuxPulse::PlayoutDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  const uint16_t nDevices = PlayoutDevices();
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  // Check if default device
+  if (index == 0) {
+    uint16_t deviceIndex = 0;
+    return GetDefaultDeviceInfo(false, name, deviceIndex);
+  }
+
+  // Tell the callback that we want
+  // The name for this device
+  _playDisplayDeviceName = name;
+  _deviceIndex = index;
+
+  // get playout devices
+  PlayoutDevices();
+
+  // clear device name and index
+  _playDisplayDeviceName = NULL;
+  _deviceIndex = -1;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::RecordingDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  const uint16_t nDevices(RecordingDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  // Check if default device
+  if (index == 0) {
+    uint16_t deviceIndex = 0;
+    return GetDefaultDeviceInfo(true, name, deviceIndex);
+  }
+
+  // Tell the callback that we want
+  // the name for this device
+  _recDisplayDeviceName = name;
+  _deviceIndex = index;
+
+  // Get recording devices
+  RecordingDevices();
+
+  // Clear device name and index
+  _recDisplayDeviceName = NULL;
+  _deviceIndex = -1;
+
+  return 0;
+}
+
+int16_t AudioDeviceLinuxPulse::RecordingDevices() {
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+  _numRecDevices = 1;  // Init to 1 to account for "default"
+
+  // Get the whole list of devices and update _numRecDevices
+  paOperation = LATE(pa_context_get_source_info_list)(
+      _paContext, PaSourceInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  return _numRecDevices;
+}
+
+int32_t AudioDeviceLinuxPulse::SetRecordingDevice(uint16_t index) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_recIsInitialized) {
+    return -1;
+  }
+
+  const uint16_t nDevices(RecordingDevices());
+
+  RTC_LOG(LS_VERBOSE) << "number of availiable input devices is " << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _inputDeviceIndex = index;
+  _inputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceLinuxPulse::PlayoutIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  available = false;
+
+  // Try to initialize the playout side
+  int32_t res = InitPlayout();
+
+  // Cancel effect of initialization
+  StopPlayout();
+
+  if (res != -1) {
+    available = true;
+  }
+
+  return res;
+}
+
+int32_t AudioDeviceLinuxPulse::RecordingIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  available = false;
+
+  // Try to initialize the playout side
+  int32_t res = InitRecording();
+
+  // Cancel effect of initialization
+  StopRecording();
+
+  if (res != -1) {
+    available = true;
+  }
+
+  return res;
+}
+
+int32_t AudioDeviceLinuxPulse::InitPlayout() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_playIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the speaker (devices might have been added or removed)
+  if (InitSpeaker() == -1) {
+    RTC_LOG(LS_WARNING) << "InitSpeaker() failed";
+  }
+
+  // Set the play sample specification
+  pa_sample_spec playSampleSpec;
+  playSampleSpec.channels = _playChannels;
+  playSampleSpec.format = PA_SAMPLE_S16LE;
+  playSampleSpec.rate = sample_rate_hz_;
+
+  // Create a new play stream
+  {
+    MutexLock lock(&mutex_);
+    _playStream =
+        LATE(pa_stream_new)(_paContext, "playStream", &playSampleSpec, NULL);
+  }
+
+  if (!_playStream) {
+    RTC_LOG(LS_ERROR) << "failed to create play stream, err="
+                      << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  // Provide the playStream to the mixer
+  _mixerManager.SetPlayStream(_playStream);
+
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters
+    _ptrAudioBuffer->SetPlayoutSampleRate(sample_rate_hz_);
+    _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
+  }
+
+  RTC_LOG(LS_VERBOSE) << "stream state "
+                      << LATE(pa_stream_get_state)(_playStream);
+
+  // Set stream flags
+  _playStreamFlags = (pa_stream_flags_t)(PA_STREAM_AUTO_TIMING_UPDATE |
+                                         PA_STREAM_INTERPOLATE_TIMING);
+
+  if (_configuredLatencyPlay != WEBRTC_PA_NO_LATENCY_REQUIREMENTS) {
+    // If configuring a specific latency then we want to specify
+    // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
+    // automatically to reach that target latency. However, that flag
+    // doesn't exist in Ubuntu 8.04 and many people still use that,
+    // so we have to check the protocol version of libpulse.
+    if (LATE(pa_context_get_protocol_version)(_paContext) >=
+        WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) {
+      _playStreamFlags |= PA_STREAM_ADJUST_LATENCY;
+    }
+
+    const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream);
+    if (!spec) {
+      RTC_LOG(LS_ERROR) << "pa_stream_get_sample_spec()";
+      return -1;
+    }
+
+    size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
+    uint32_t latency = bytesPerSec * WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS /
+                       WEBRTC_PA_MSECS_PER_SEC;
+
+    // Set the play buffer attributes
+    _playBufferAttr.maxlength = latency;  // num bytes stored in the buffer
+    _playBufferAttr.tlength = latency;    // target fill level of play buffer
+    // minimum free num bytes before server request more data
+    _playBufferAttr.minreq = latency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR;
+    // prebuffer tlength before starting playout
+    _playBufferAttr.prebuf = _playBufferAttr.tlength - _playBufferAttr.minreq;
+
+    _configuredLatencyPlay = latency;
+  }
+
+  // num samples in bytes * num channels
+  _playbackBufferSize = sample_rate_hz_ / 100 * 2 * _playChannels;
+  _playbackBufferUnused = _playbackBufferSize;
+  _playBuffer = new int8_t[_playbackBufferSize];
+
+  // Enable underflow callback
+  LATE(pa_stream_set_underflow_callback)
+  (_playStream, PaStreamUnderflowCallback, this);
+
+  // Set the state callback function for the stream
+  LATE(pa_stream_set_state_callback)(_playStream, PaStreamStateCallback, this);
+
+  // Mark playout side as initialized
+  {
+    MutexLock lock(&mutex_);
+    _playIsInitialized = true;
+    _sndCardPlayDelay = 0;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::InitRecording() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (_recording) {
+    return -1;
+  }
+
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_recIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the microphone (devices might have been added or removed)
+  if (InitMicrophone() == -1) {
+    RTC_LOG(LS_WARNING) << "InitMicrophone() failed";
+  }
+
+  // Set the rec sample specification
+  pa_sample_spec recSampleSpec;
+  recSampleSpec.channels = _recChannels;
+  recSampleSpec.format = PA_SAMPLE_S16LE;
+  recSampleSpec.rate = sample_rate_hz_;
+
+  // Create a new rec stream
+  _recStream =
+      LATE(pa_stream_new)(_paContext, "recStream", &recSampleSpec, NULL);
+  if (!_recStream) {
+    RTC_LOG(LS_ERROR) << "failed to create rec stream, err="
+                      << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  // Provide the recStream to the mixer
+  _mixerManager.SetRecStream(_recStream);
+
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters
+    _ptrAudioBuffer->SetRecordingSampleRate(sample_rate_hz_);
+    _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+  }
+
+  if (_configuredLatencyRec != WEBRTC_PA_NO_LATENCY_REQUIREMENTS) {
+    _recStreamFlags = (pa_stream_flags_t)(PA_STREAM_AUTO_TIMING_UPDATE |
+                                          PA_STREAM_INTERPOLATE_TIMING);
+
+    // If configuring a specific latency then we want to specify
+    // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
+    // automatically to reach that target latency. However, that flag
+    // doesn't exist in Ubuntu 8.04 and many people still use that,
+    //  so we have to check the protocol version of libpulse.
+    if (LATE(pa_context_get_protocol_version)(_paContext) >=
+        WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) {
+      _recStreamFlags |= PA_STREAM_ADJUST_LATENCY;
+    }
+
+    const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_recStream);
+    if (!spec) {
+      RTC_LOG(LS_ERROR) << "pa_stream_get_sample_spec(rec)";
+      return -1;
+    }
+
+    size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
+    uint32_t latency = bytesPerSec * WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS /
+                       WEBRTC_PA_MSECS_PER_SEC;
+
+    // Set the rec buffer attributes
+    // Note: fragsize specifies a maximum transfer size, not a minimum, so
+    // it is not possible to force a high latency setting, only a low one.
+    _recBufferAttr.fragsize = latency;  // size of fragment
+    _recBufferAttr.maxlength =
+        latency + bytesPerSec * WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS /
+                      WEBRTC_PA_MSECS_PER_SEC;
+
+    _configuredLatencyRec = latency;
+  }
+
+  _recordBufferSize = sample_rate_hz_ / 100 * 2 * _recChannels;
+  _recordBufferUsed = 0;
+  _recBuffer = new int8_t[_recordBufferSize];
+
+  // Enable overflow callback
+  LATE(pa_stream_set_overflow_callback)
+  (_recStream, PaStreamOverflowCallback, this);
+
+  // Set the state callback function for the stream
+  LATE(pa_stream_set_state_callback)(_recStream, PaStreamStateCallback, this);
+
+  // Mark recording side as initialized
+  _recIsInitialized = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StartRecording() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!_recIsInitialized) {
+    return -1;
+  }
+
+  if (_recording) {
+    return 0;
+  }
+
+  // Set state to ensure that the recording starts from the audio thread.
+  _startRec = true;
+
+  // The audio thread will signal when recording has started.
+  _timeEventRec.Set();
+  if (!_recStartEvent.Wait(TimeDelta::Seconds(10))) {
+    {
+      MutexLock lock(&mutex_);
+      _startRec = false;
+    }
+    StopRecording();
+    RTC_LOG(LS_ERROR) << "failed to activate recording";
+    return -1;
+  }
+
+  {
+    MutexLock lock(&mutex_);
+    if (_recording) {
+      // The recording state is set by the audio thread after recording
+      // has started.
+    } else {
+      RTC_LOG(LS_ERROR) << "failed to activate recording";
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StopRecording() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  MutexLock lock(&mutex_);
+
+  if (!_recIsInitialized) {
+    return 0;
+  }
+
+  if (_recStream == NULL) {
+    return -1;
+  }
+
+  _recIsInitialized = false;
+  _recording = false;
+
+  RTC_LOG(LS_VERBOSE) << "stopping recording";
+
+  // Stop Recording
+  PaLock();
+
+  DisableReadCallback();
+  LATE(pa_stream_set_overflow_callback)(_recStream, NULL, NULL);
+
+  // Unset this here so that we don't get a TERMINATED callback
+  LATE(pa_stream_set_state_callback)(_recStream, NULL, NULL);
+
+  if (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_UNCONNECTED) {
+    // Disconnect the stream
+    if (LATE(pa_stream_disconnect)(_recStream) != PA_OK) {
+      RTC_LOG(LS_ERROR) << "failed to disconnect rec stream, err="
+                        << LATE(pa_context_errno)(_paContext);
+      PaUnLock();
+      return -1;
+    }
+
+    RTC_LOG(LS_VERBOSE) << "disconnected recording";
+  }
+
+  LATE(pa_stream_unref)(_recStream);
+  _recStream = NULL;
+
+  PaUnLock();
+
+  // Provide the recStream to the mixer
+  _mixerManager.SetRecStream(_recStream);
+
+  if (_recBuffer) {
+    delete[] _recBuffer;
+    _recBuffer = NULL;
+  }
+
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::RecordingIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_recIsInitialized);
+}
+
+bool AudioDeviceLinuxPulse::Recording() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_recording);
+}
+
+bool AudioDeviceLinuxPulse::PlayoutIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_playIsInitialized);
+}
+
+int32_t AudioDeviceLinuxPulse::StartPlayout() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (!_playIsInitialized) {
+    return -1;
+  }
+
+  if (_playing) {
+    return 0;
+  }
+
+  // Set state to ensure that playout starts from the audio thread.
+  {
+    MutexLock lock(&mutex_);
+    _startPlay = true;
+  }
+
+  // Both `_startPlay` and `_playing` needs protction since they are also
+  // accessed on the playout thread.
+
+  // The audio thread will signal when playout has started.
+  _timeEventPlay.Set();
+  if (!_playStartEvent.Wait(TimeDelta::Seconds(10))) {
+    {
+      MutexLock lock(&mutex_);
+      _startPlay = false;
+    }
+    StopPlayout();
+    RTC_LOG(LS_ERROR) << "failed to activate playout";
+    return -1;
+  }
+
+  {
+    MutexLock lock(&mutex_);
+    if (_playing) {
+      // The playing state is set by the audio thread after playout
+      // has started.
+    } else {
+      RTC_LOG(LS_ERROR) << "failed to activate playing";
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::StopPlayout() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  MutexLock lock(&mutex_);
+
+  if (!_playIsInitialized) {
+    return 0;
+  }
+
+  if (_playStream == NULL) {
+    return -1;
+  }
+
+  _playIsInitialized = false;
+  _playing = false;
+  _sndCardPlayDelay = 0;
+
+  RTC_LOG(LS_VERBOSE) << "stopping playback";
+
+  // Stop Playout
+  PaLock();
+
+  DisableWriteCallback();
+  LATE(pa_stream_set_underflow_callback)(_playStream, NULL, NULL);
+
+  // Unset this here so that we don't get a TERMINATED callback
+  LATE(pa_stream_set_state_callback)(_playStream, NULL, NULL);
+
+  if (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_UNCONNECTED) {
+    // Disconnect the stream
+    if (LATE(pa_stream_disconnect)(_playStream) != PA_OK) {
+      RTC_LOG(LS_ERROR) << "failed to disconnect play stream, err="
+                        << LATE(pa_context_errno)(_paContext);
+      PaUnLock();
+      return -1;
+    }
+
+    RTC_LOG(LS_VERBOSE) << "disconnected playback";
+  }
+
+  LATE(pa_stream_unref)(_playStream);
+  _playStream = NULL;
+
+  PaUnLock();
+
+  // Provide the playStream to the mixer
+  _mixerManager.SetPlayStream(_playStream);
+
+  if (_playBuffer) {
+    delete[] _playBuffer;
+    _playBuffer = NULL;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::PlayoutDelay(uint16_t& delayMS) const {
+  MutexLock lock(&mutex_);
+  delayMS = (uint16_t)_sndCardPlayDelay;
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::Playing() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (_playing);
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+void AudioDeviceLinuxPulse::PaContextStateCallback(pa_context* c, void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaContextStateCallbackHandler(c);
+}
+
+// ----------------------------------------------------------------------------
+//  PaSinkInfoCallback
+// ----------------------------------------------------------------------------
+
+void AudioDeviceLinuxPulse::PaSinkInfoCallback(pa_context* /*c*/,
+                                               const pa_sink_info* i,
+                                               int eol,
+                                               void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaSinkInfoCallbackHandler(i, eol);
+}
+
+void AudioDeviceLinuxPulse::PaSourceInfoCallback(pa_context* /*c*/,
+                                                 const pa_source_info* i,
+                                                 int eol,
+                                                 void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaSourceInfoCallbackHandler(i,
+                                                                          eol);
+}
+
+void AudioDeviceLinuxPulse::PaServerInfoCallback(pa_context* /*c*/,
+                                                 const pa_server_info* i,
+                                                 void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaServerInfoCallbackHandler(i);
+}
+
+void AudioDeviceLinuxPulse::PaStreamStateCallback(pa_stream* p, void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamStateCallbackHandler(p);
+}
+
+void AudioDeviceLinuxPulse::PaContextStateCallbackHandler(pa_context* c) {
+  RTC_LOG(LS_VERBOSE) << "context state cb";
+
+  pa_context_state_t state = LATE(pa_context_get_state)(c);
+  switch (state) {
+    case PA_CONTEXT_UNCONNECTED:
+      RTC_LOG(LS_VERBOSE) << "unconnected";
+      break;
+    case PA_CONTEXT_CONNECTING:
+    case PA_CONTEXT_AUTHORIZING:
+    case PA_CONTEXT_SETTING_NAME:
+      RTC_LOG(LS_VERBOSE) << "no state";
+      break;
+    case PA_CONTEXT_FAILED:
+    case PA_CONTEXT_TERMINATED:
+      RTC_LOG(LS_VERBOSE) << "failed";
+      _paStateChanged = true;
+      LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+      break;
+    case PA_CONTEXT_READY:
+      RTC_LOG(LS_VERBOSE) << "ready";
+      _paStateChanged = true;
+      LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+      break;
+  }
+}
+
+void AudioDeviceLinuxPulse::PaSinkInfoCallbackHandler(const pa_sink_info* i,
+                                                      int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  if (_numPlayDevices == _deviceIndex) {
+    // Convert the device index to the one of the sink
+    _paDeviceIndex = i->index;
+
+    if (_playDeviceName) {
+      // Copy the sink name
+      strncpy(_playDeviceName, i->name, kAdmMaxDeviceNameSize);
+      _playDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+    }
+    if (_playDisplayDeviceName) {
+      // Copy the sink display name
+      strncpy(_playDisplayDeviceName, i->description, kAdmMaxDeviceNameSize);
+      _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+    }
+  }
+
+  _numPlayDevices++;
+}
+
+void AudioDeviceLinuxPulse::PaSourceInfoCallbackHandler(const pa_source_info* i,
+                                                        int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  // We don't want to list output devices
+  if (i->monitor_of_sink == PA_INVALID_INDEX) {
+    if (_numRecDevices == _deviceIndex) {
+      // Convert the device index to the one of the source
+      _paDeviceIndex = i->index;
+
+      if (_recDeviceName) {
+        // copy the source name
+        strncpy(_recDeviceName, i->name, kAdmMaxDeviceNameSize);
+        _recDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+      }
+      if (_recDisplayDeviceName) {
+        // Copy the source display name
+        strncpy(_recDisplayDeviceName, i->description, kAdmMaxDeviceNameSize);
+        _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+      }
+    }
+
+    _numRecDevices++;
+  }
+}
+
+void AudioDeviceLinuxPulse::PaServerInfoCallbackHandler(
+    const pa_server_info* i) {
+  // Use PA native sampling rate
+  sample_rate_hz_ = i->sample_spec.rate;
+
+  // Copy the PA server version
+  strncpy(_paServerVersion, i->server_version, 31);
+  _paServerVersion[31] = '\0';
+
+  if (_recDisplayDeviceName) {
+    // Copy the source name
+    strncpy(_recDisplayDeviceName, i->default_source_name,
+            kAdmMaxDeviceNameSize);
+    _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+  }
+
+  if (_playDisplayDeviceName) {
+    // Copy the sink name
+    strncpy(_playDisplayDeviceName, i->default_sink_name,
+            kAdmMaxDeviceNameSize);
+    _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
+  }
+
+  LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+}
+
+void AudioDeviceLinuxPulse::PaStreamStateCallbackHandler(pa_stream* p) {
+  RTC_LOG(LS_VERBOSE) << "stream state cb";
+
+  pa_stream_state_t state = LATE(pa_stream_get_state)(p);
+  switch (state) {
+    case PA_STREAM_UNCONNECTED:
+      RTC_LOG(LS_VERBOSE) << "unconnected";
+      break;
+    case PA_STREAM_CREATING:
+      RTC_LOG(LS_VERBOSE) << "creating";
+      break;
+    case PA_STREAM_FAILED:
+    case PA_STREAM_TERMINATED:
+      RTC_LOG(LS_VERBOSE) << "failed";
+      break;
+    case PA_STREAM_READY:
+      RTC_LOG(LS_VERBOSE) << "ready";
+      break;
+  }
+
+  LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+}
+
+int32_t AudioDeviceLinuxPulse::CheckPulseAudioVersion() {
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+
+  // get the server info and update deviceName
+  paOperation =
+      LATE(pa_context_get_server_info)(_paContext, PaServerInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  RTC_LOG(LS_VERBOSE) << "checking PulseAudio version: " << _paServerVersion;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::InitSamplingFrequency() {
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+
+  // Get the server info and update sample_rate_hz_
+  paOperation =
+      LATE(pa_context_get_server_info)(_paContext, PaServerInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice,
+                                                    char* name,
+                                                    uint16_t& index) {
+  char tmpName[kAdmMaxDeviceNameSize] = {0};
+  // subtract length of "default: "
+  uint16_t nameLen = kAdmMaxDeviceNameSize - 9;
+  char* pName = NULL;
+
+  if (name) {
+    // Add "default: "
+    strcpy(name, "default: ");
+    pName = &name[9];
+  }
+
+  // Tell the callback that we want
+  // the name for this device
+  if (recDevice) {
+    _recDisplayDeviceName = tmpName;
+  } else {
+    _playDisplayDeviceName = tmpName;
+  }
+
+  // Set members
+  _paDeviceIndex = -1;
+  _deviceIndex = 0;
+  _numPlayDevices = 0;
+  _numRecDevices = 0;
+
+  PaLock();
+
+  pa_operation* paOperation = NULL;
+
+  // Get the server info and update deviceName
+  paOperation =
+      LATE(pa_context_get_server_info)(_paContext, PaServerInfoCallback, this);
+
+  WaitForOperationCompletion(paOperation);
+
+  // Get the device index
+  if (recDevice) {
+    paOperation = LATE(pa_context_get_source_info_by_name)(
+        _paContext, (char*)tmpName, PaSourceInfoCallback, this);
+  } else {
+    paOperation = LATE(pa_context_get_sink_info_by_name)(
+        _paContext, (char*)tmpName, PaSinkInfoCallback, this);
+  }
+
+  WaitForOperationCompletion(paOperation);
+
+  PaUnLock();
+
+  // Set the index
+  index = _paDeviceIndex;
+
+  if (name) {
+    // Copy to name string
+    strncpy(pName, tmpName, nameLen);
+  }
+
+  // Clear members
+  _playDisplayDeviceName = NULL;
+  _recDisplayDeviceName = NULL;
+  _paDeviceIndex = -1;
+  _deviceIndex = -1;
+  _numPlayDevices = 0;
+  _numRecDevices = 0;
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::InitPulseAudio() {
+  int retVal = 0;
+
+  // Load libpulse
+  if (!GetPulseSymbolTable()->Load()) {
+    // Most likely the Pulse library and sound server are not installed on
+    // this system
+    RTC_LOG(LS_ERROR) << "failed to load symbol table";
+    return -1;
+  }
+
+  // Create a mainloop API and connection to the default server
+  // the mainloop is the internal asynchronous API event loop
+  if (_paMainloop) {
+    RTC_LOG(LS_ERROR) << "PA mainloop has already existed";
+    return -1;
+  }
+  _paMainloop = LATE(pa_threaded_mainloop_new)();
+  if (!_paMainloop) {
+    RTC_LOG(LS_ERROR) << "could not create mainloop";
+    return -1;
+  }
+
+  // Start the threaded main loop
+  retVal = LATE(pa_threaded_mainloop_start)(_paMainloop);
+  if (retVal != PA_OK) {
+    RTC_LOG(LS_ERROR) << "failed to start main loop, error=" << retVal;
+    return -1;
+  }
+
+  RTC_LOG(LS_VERBOSE) << "mainloop running!";
+
+  PaLock();
+
+  _paMainloopApi = LATE(pa_threaded_mainloop_get_api)(_paMainloop);
+  if (!_paMainloopApi) {
+    RTC_LOG(LS_ERROR) << "could not create mainloop API";
+    PaUnLock();
+    return -1;
+  }
+
+  // Create a new PulseAudio context
+  if (_paContext) {
+    RTC_LOG(LS_ERROR) << "PA context has already existed";
+    PaUnLock();
+    return -1;
+  }
+  _paContext = LATE(pa_context_new)(_paMainloopApi, "WEBRTC VoiceEngine");
+
+  if (!_paContext) {
+    RTC_LOG(LS_ERROR) << "could not create context";
+    PaUnLock();
+    return -1;
+  }
+
+  // Set state callback function
+  LATE(pa_context_set_state_callback)(_paContext, PaContextStateCallback, this);
+
+  // Connect the context to a server (default)
+  _paStateChanged = false;
+  retVal =
+      LATE(pa_context_connect)(_paContext, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL);
+
+  if (retVal != PA_OK) {
+    RTC_LOG(LS_ERROR) << "failed to connect context, error=" << retVal;
+    PaUnLock();
+    return -1;
+  }
+
+  // Wait for state change
+  while (!_paStateChanged) {
+    LATE(pa_threaded_mainloop_wait)(_paMainloop);
+  }
+
+  // Now check to see what final state we reached.
+  pa_context_state_t state = LATE(pa_context_get_state)(_paContext);
+
+  if (state != PA_CONTEXT_READY) {
+    if (state == PA_CONTEXT_FAILED) {
+      RTC_LOG(LS_ERROR) << "failed to connect to PulseAudio sound server";
+    } else if (state == PA_CONTEXT_TERMINATED) {
+      RTC_LOG(LS_ERROR) << "PulseAudio connection terminated early";
+    } else {
+      // Shouldn't happen, because we only signal on one of those three
+      // states
+      RTC_LOG(LS_ERROR) << "unknown problem connecting to PulseAudio";
+    }
+    PaUnLock();
+    return -1;
+  }
+
+  PaUnLock();
+
+  // Give the objects to the mixer manager
+  _mixerManager.SetPulseAudioObjects(_paMainloop, _paContext);
+
+  // Check the version
+  if (CheckPulseAudioVersion() < 0) {
+    RTC_LOG(LS_ERROR) << "PulseAudio version " << _paServerVersion
+                      << " not supported";
+    return -1;
+  }
+
+  // Initialize sampling frequency
+  if (InitSamplingFrequency() < 0 || sample_rate_hz_ == 0) {
+    RTC_LOG(LS_ERROR) << "failed to initialize sampling frequency, set to "
+                      << sample_rate_hz_ << " Hz";
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::TerminatePulseAudio() {
+  // Do nothing if the instance doesn't exist
+  // likely GetPulseSymbolTable.Load() fails
+  if (!_paMainloop) {
+    return 0;
+  }
+
+  PaLock();
+
+  // Disconnect the context
+  if (_paContext) {
+    LATE(pa_context_disconnect)(_paContext);
+  }
+
+  // Unreference the context
+  if (_paContext) {
+    LATE(pa_context_unref)(_paContext);
+  }
+
+  PaUnLock();
+  _paContext = NULL;
+
+  // Stop the threaded main loop
+  if (_paMainloop) {
+    LATE(pa_threaded_mainloop_stop)(_paMainloop);
+  }
+
+  // Free the mainloop
+  if (_paMainloop) {
+    LATE(pa_threaded_mainloop_free)(_paMainloop);
+  }
+
+  _paMainloop = NULL;
+
+  RTC_LOG(LS_VERBOSE) << "PulseAudio terminated";
+
+  return 0;
+}
+
+void AudioDeviceLinuxPulse::PaLock() {
+  LATE(pa_threaded_mainloop_lock)(_paMainloop);
+}
+
+void AudioDeviceLinuxPulse::PaUnLock() {
+  LATE(pa_threaded_mainloop_unlock)(_paMainloop);
+}
+
+void AudioDeviceLinuxPulse::WaitForOperationCompletion(
+    pa_operation* paOperation) const {
+  if (!paOperation) {
+    RTC_LOG(LS_ERROR) << "paOperation NULL in WaitForOperationCompletion";
+    return;
+  }
+
+  while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING) {
+    LATE(pa_threaded_mainloop_wait)(_paMainloop);
+  }
+
+  LATE(pa_operation_unref)(paOperation);
+}
+
+// ============================================================================
+//                                  Thread Methods
+// ============================================================================
+
+void AudioDeviceLinuxPulse::EnableWriteCallback() {
+  if (LATE(pa_stream_get_state)(_playStream) == PA_STREAM_READY) {
+    // May already have available space. Must check.
+    _tempBufferSpace = LATE(pa_stream_writable_size)(_playStream);
+    if (_tempBufferSpace > 0) {
+      // Yup, there is already space available, so if we register a
+      // write callback then it will not receive any event. So dispatch
+      // one ourself instead.
+      _timeEventPlay.Set();
+      return;
+    }
+  }
+
+  LATE(pa_stream_set_write_callback)(_playStream, &PaStreamWriteCallback, this);
+}
+
+void AudioDeviceLinuxPulse::DisableWriteCallback() {
+  LATE(pa_stream_set_write_callback)(_playStream, NULL, NULL);
+}
+
+void AudioDeviceLinuxPulse::PaStreamWriteCallback(pa_stream* /*unused*/,
+                                                  size_t buffer_space,
+                                                  void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamWriteCallbackHandler(
+      buffer_space);
+}
+
+void AudioDeviceLinuxPulse::PaStreamWriteCallbackHandler(size_t bufferSpace) {
+  _tempBufferSpace = bufferSpace;
+
+  // Since we write the data asynchronously on a different thread, we have
+  // to temporarily disable the write callback or else Pulse will call it
+  // continuously until we write the data. We re-enable it below.
+  DisableWriteCallback();
+  _timeEventPlay.Set();
+}
+
+void AudioDeviceLinuxPulse::PaStreamUnderflowCallback(pa_stream* /*unused*/,
+                                                      void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)
+      ->PaStreamUnderflowCallbackHandler();
+}
+
+void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler() {
+  RTC_LOG(LS_WARNING) << "Playout underflow";
+
+  if (_configuredLatencyPlay == WEBRTC_PA_NO_LATENCY_REQUIREMENTS) {
+    // We didn't configure a pa_buffer_attr before, so switching to
+    // one now would be questionable.
+    return;
+  }
+
+  // Otherwise reconfigure the stream with a higher target latency.
+
+  const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream);
+  if (!spec) {
+    RTC_LOG(LS_ERROR) << "pa_stream_get_sample_spec()";
+    return;
+  }
+
+  size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
+  uint32_t newLatency =
+      _configuredLatencyPlay + bytesPerSec *
+                                   WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS /
+                                   WEBRTC_PA_MSECS_PER_SEC;
+
+  // Set the play buffer attributes
+  _playBufferAttr.maxlength = newLatency;
+  _playBufferAttr.tlength = newLatency;
+  _playBufferAttr.minreq = newLatency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR;
+  _playBufferAttr.prebuf = _playBufferAttr.tlength - _playBufferAttr.minreq;
+
+  pa_operation* op = LATE(pa_stream_set_buffer_attr)(
+      _playStream, &_playBufferAttr, NULL, NULL);
+  if (!op) {
+    RTC_LOG(LS_ERROR) << "pa_stream_set_buffer_attr()";
+    return;
+  }
+
+  // Don't need to wait for this to complete.
+  LATE(pa_operation_unref)(op);
+
+  // Save the new latency in case we underflow again.
+  _configuredLatencyPlay = newLatency;
+}
+
+void AudioDeviceLinuxPulse::EnableReadCallback() {
+  LATE(pa_stream_set_read_callback)(_recStream, &PaStreamReadCallback, this);
+}
+
+void AudioDeviceLinuxPulse::DisableReadCallback() {
+  LATE(pa_stream_set_read_callback)(_recStream, NULL, NULL);
+}
+
+void AudioDeviceLinuxPulse::PaStreamReadCallback(pa_stream* /*unused1*/,
+                                                 size_t /*unused2*/,
+                                                 void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamReadCallbackHandler();
+}
+
+void AudioDeviceLinuxPulse::PaStreamReadCallbackHandler() {
+  // We get the data pointer and size now in order to save one Lock/Unlock
+  // in the worker thread.
+  if (LATE(pa_stream_peek)(_recStream, &_tempSampleData,
+                           &_tempSampleDataSize) != 0) {
+    RTC_LOG(LS_ERROR) << "Can't read data!";
+    return;
+  }
+
+  // Since we consume the data asynchronously on a different thread, we have
+  // to temporarily disable the read callback or else Pulse will call it
+  // continuously until we consume the data. We re-enable it below.
+  DisableReadCallback();
+  _timeEventRec.Set();
+}
+
+void AudioDeviceLinuxPulse::PaStreamOverflowCallback(pa_stream* /*unused*/,
+                                                     void* pThis) {
+  static_cast<AudioDeviceLinuxPulse*>(pThis)->PaStreamOverflowCallbackHandler();
+}
+
+void AudioDeviceLinuxPulse::PaStreamOverflowCallbackHandler() {
+  RTC_LOG(LS_WARNING) << "Recording overflow";
+}
+
+int32_t AudioDeviceLinuxPulse::LatencyUsecs(pa_stream* stream) {
+  if (!WEBRTC_PA_REPORT_LATENCY) {
+    return 0;
+  }
+
+  if (!stream) {
+    return 0;
+  }
+
+  pa_usec_t latency;
+  int negative;
+  if (LATE(pa_stream_get_latency)(stream, &latency, &negative) != 0) {
+    RTC_LOG(LS_ERROR) << "Can't query latency";
+    // We'd rather continue playout/capture with an incorrect delay than
+    // stop it altogether, so return a valid value.
+    return 0;
+  }
+
+  if (negative) {
+    RTC_LOG(LS_VERBOSE)
+        << "warning: pa_stream_get_latency reported negative delay";
+
+    // The delay can be negative for monitoring streams if the captured
+    // samples haven't been played yet. In such a case, "latency"
+    // contains the magnitude, so we must negate it to get the real value.
+    int32_t tmpLatency = (int32_t)-latency;
+    if (tmpLatency < 0) {
+      // Make sure that we don't use a negative delay.
+      tmpLatency = 0;
+    }
+
+    return tmpLatency;
+  } else {
+    return (int32_t)latency;
+  }
+}
+
+int32_t AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData,
+                                                size_t bufferSize)
+    RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_) {
+  size_t size = bufferSize;
+  uint32_t numRecSamples = _recordBufferSize / (2 * _recChannels);
+
+  // Account for the peeked data and the used data.
+  uint32_t recDelay =
+      (uint32_t)((LatencyUsecs(_recStream) / 1000) +
+                 10 * ((size + _recordBufferUsed) / _recordBufferSize));
+
+  if (_playStream) {
+    // Get the playout delay.
+    _sndCardPlayDelay = (uint32_t)(LatencyUsecs(_playStream) / 1000);
+  }
+
+  if (_recordBufferUsed > 0) {
+    // Have to copy to the buffer until it is full.
+    size_t copy = _recordBufferSize - _recordBufferUsed;
+    if (size < copy) {
+      copy = size;
+    }
+
+    memcpy(&_recBuffer[_recordBufferUsed], bufferData, copy);
+    _recordBufferUsed += copy;
+    bufferData = static_cast<const char*>(bufferData) + copy;
+    size -= copy;
+
+    if (_recordBufferUsed != _recordBufferSize) {
+      // Not enough data yet to pass to VoE.
+      return 0;
+    }
+
+    // Provide data to VoiceEngine.
+    if (ProcessRecordedData(_recBuffer, numRecSamples, recDelay) == -1) {
+      // We have stopped recording.
+      return -1;
+    }
+
+    _recordBufferUsed = 0;
+  }
+
+  // Now process full 10ms sample sets directly from the input.
+  while (size >= _recordBufferSize) {
+    // Provide data to VoiceEngine.
+    if (ProcessRecordedData(static_cast<int8_t*>(const_cast<void*>(bufferData)),
+                            numRecSamples, recDelay) == -1) {
+      // We have stopped recording.
+      return -1;
+    }
+
+    bufferData = static_cast<const char*>(bufferData) + _recordBufferSize;
+    size -= _recordBufferSize;
+
+    // We have consumed 10ms of data.
+    recDelay -= 10;
+  }
+
+  // Now save any leftovers for later.
+  if (size > 0) {
+    memcpy(_recBuffer, bufferData, size);
+    _recordBufferUsed = size;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceLinuxPulse::ProcessRecordedData(int8_t* bufferData,
+                                                   uint32_t bufferSizeInSamples,
+                                                   uint32_t recDelay)
+    RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_) {
+  _ptrAudioBuffer->SetRecordedBuffer(bufferData, bufferSizeInSamples);
+
+  // TODO(andrew): this is a temporary hack, to avoid non-causal far- and
+  // near-end signals at the AEC for PulseAudio. I think the system delay is
+  // being correctly calculated here, but for legacy reasons we add +10 ms
+  // to the value in the AEC. The real fix will be part of a larger
+  // investigation into managing system delay in the AEC.
+  if (recDelay > 10)
+    recDelay -= 10;
+  else
+    recDelay = 0;
+  _ptrAudioBuffer->SetVQEData(_sndCardPlayDelay, recDelay);
+  _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+  // Deliver recorded samples at specified sample rate,
+  // mic level etc. to the observer using callback.
+  UnLock();
+  _ptrAudioBuffer->DeliverRecordedData();
+  Lock();
+
+  // We have been unlocked - check the flag again.
+  if (!_recording) {
+    return -1;
+  }
+
+  return 0;
+}
+
+bool AudioDeviceLinuxPulse::PlayThreadProcess() {
+  if (!_timeEventPlay.Wait(TimeDelta::Seconds(1))) {
+    return true;
+  }
+
+  MutexLock lock(&mutex_);
+
+  if (quit_) {
+    return false;
+  }
+
+  if (_startPlay) {
+    RTC_LOG(LS_VERBOSE) << "_startPlay true, performing initial actions";
+
+    _startPlay = false;
+    _playDeviceName = NULL;
+
+    // Set if not default device
+    if (_outputDeviceIndex > 0) {
+      // Get the playout device name
+      _playDeviceName = new char[kAdmMaxDeviceNameSize];
+      _deviceIndex = _outputDeviceIndex;
+      PlayoutDevices();
+    }
+
+    // Start muted only supported on 0.9.11 and up
+    if (LATE(pa_context_get_protocol_version)(_paContext) >=
+        WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) {
+      // Get the currently saved speaker mute status
+      // and set the initial mute status accordingly
+      bool enabled(false);
+      _mixerManager.SpeakerMute(enabled);
+      if (enabled) {
+        _playStreamFlags |= PA_STREAM_START_MUTED;
+      }
+    }
+
+    // Get the currently saved speaker volume
+    uint32_t volume = 0;
+    if (update_speaker_volume_at_startup_)
+      _mixerManager.SpeakerVolume(volume);
+
+    PaLock();
+
+    // NULL gives PA the choice of startup volume.
+    pa_cvolume* ptr_cvolume = NULL;
+    if (update_speaker_volume_at_startup_) {
+      pa_cvolume cVolumes;
+      ptr_cvolume = &cVolumes;
+
+      // Set the same volume for all channels
+      const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_playStream);
+      LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
+      update_speaker_volume_at_startup_ = false;
+    }
+
+    // Connect the stream to a sink
+    if (LATE(pa_stream_connect_playback)(
+            _playStream, _playDeviceName, &_playBufferAttr,
+            (pa_stream_flags_t)_playStreamFlags, ptr_cvolume, NULL) != PA_OK) {
+      RTC_LOG(LS_ERROR) << "failed to connect play stream, err="
+                        << LATE(pa_context_errno)(_paContext);
+    }
+
+    RTC_LOG(LS_VERBOSE) << "play stream connected";
+
+    // Wait for state change
+    while (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_READY) {
+      LATE(pa_threaded_mainloop_wait)(_paMainloop);
+    }
+
+    RTC_LOG(LS_VERBOSE) << "play stream ready";
+
+    // We can now handle write callbacks
+    EnableWriteCallback();
+
+    PaUnLock();
+
+    // Clear device name
+    if (_playDeviceName) {
+      delete[] _playDeviceName;
+      _playDeviceName = NULL;
+    }
+
+    _playing = true;
+    _playStartEvent.Set();
+
+    return true;
+  }
+
+  if (_playing) {
+    if (!_recording) {
+      // Update the playout delay
+      _sndCardPlayDelay = (uint32_t)(LatencyUsecs(_playStream) / 1000);
+    }
+
+    if (_playbackBufferUnused < _playbackBufferSize) {
+      size_t write = _playbackBufferSize - _playbackBufferUnused;
+      if (_tempBufferSpace < write) {
+        write = _tempBufferSpace;
+      }
+
+      PaLock();
+      if (LATE(pa_stream_write)(
+              _playStream, (void*)&_playBuffer[_playbackBufferUnused], write,
+              NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) {
+        _writeErrors++;
+        if (_writeErrors > 10) {
+          RTC_LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors
+                            << ", error=" << LATE(pa_context_errno)(_paContext);
+          _writeErrors = 0;
+        }
+      }
+      PaUnLock();
+
+      _playbackBufferUnused += write;
+      _tempBufferSpace -= write;
+    }
+
+    uint32_t numPlaySamples = _playbackBufferSize / (2 * _playChannels);
+    // Might have been reduced to zero by the above.
+    if (_tempBufferSpace > 0) {
+      // Ask for new PCM data to be played out using the
+      // AudioDeviceBuffer ensure that this callback is executed
+      // without taking the audio-thread lock.
+      UnLock();
+      RTC_LOG(LS_VERBOSE) << "requesting data";
+      uint32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(numPlaySamples);
+      Lock();
+
+      // We have been unlocked - check the flag again.
+      if (!_playing) {
+        return true;
+      }
+
+      nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer);
+      if (nSamples != numPlaySamples) {
+        RTC_LOG(LS_ERROR) << "invalid number of output samples(" << nSamples
+                          << ")";
+      }
+
+      size_t write = _playbackBufferSize;
+      if (_tempBufferSpace < write) {
+        write = _tempBufferSpace;
+      }
+
+      RTC_LOG(LS_VERBOSE) << "will write";
+      PaLock();
+      if (LATE(pa_stream_write)(_playStream, (void*)&_playBuffer[0], write,
+                                NULL, (int64_t)0, PA_SEEK_RELATIVE) != PA_OK) {
+        _writeErrors++;
+        if (_writeErrors > 10) {
+          RTC_LOG(LS_ERROR) << "Playout error: _writeErrors=" << _writeErrors
+                            << ", error=" << LATE(pa_context_errno)(_paContext);
+          _writeErrors = 0;
+        }
+      }
+      PaUnLock();
+
+      _playbackBufferUnused = write;
+    }
+
+    _tempBufferSpace = 0;
+    PaLock();
+    EnableWriteCallback();
+    PaUnLock();
+
+  }  // _playing
+
+  return true;
+}
+
+bool AudioDeviceLinuxPulse::RecThreadProcess() {
+  if (!_timeEventRec.Wait(TimeDelta::Seconds(1))) {
+    return true;
+  }
+
+  MutexLock lock(&mutex_);
+  if (quit_) {
+    return false;
+  }
+  if (_startRec) {
+    RTC_LOG(LS_VERBOSE) << "_startRec true, performing initial actions";
+
+    _recDeviceName = NULL;
+
+    // Set if not default device
+    if (_inputDeviceIndex > 0) {
+      // Get the recording device name
+      _recDeviceName = new char[kAdmMaxDeviceNameSize];
+      _deviceIndex = _inputDeviceIndex;
+      RecordingDevices();
+    }
+
+    PaLock();
+
+    RTC_LOG(LS_VERBOSE) << "connecting stream";
+
+    // Connect the stream to a source
+    if (LATE(pa_stream_connect_record)(
+            _recStream, _recDeviceName, &_recBufferAttr,
+            (pa_stream_flags_t)_recStreamFlags) != PA_OK) {
+      RTC_LOG(LS_ERROR) << "failed to connect rec stream, err="
+                        << LATE(pa_context_errno)(_paContext);
+    }
+
+    RTC_LOG(LS_VERBOSE) << "connected";
+
+    // Wait for state change
+    while (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_READY) {
+      LATE(pa_threaded_mainloop_wait)(_paMainloop);
+    }
+
+    RTC_LOG(LS_VERBOSE) << "done";
+
+    // We can now handle read callbacks
+    EnableReadCallback();
+
+    PaUnLock();
+
+    // Clear device name
+    if (_recDeviceName) {
+      delete[] _recDeviceName;
+      _recDeviceName = NULL;
+    }
+
+    _startRec = false;
+    _recording = true;
+    _recStartEvent.Set();
+
+    return true;
+  }
+
+  if (_recording) {
+    // Read data and provide it to VoiceEngine
+    if (ReadRecordedData(_tempSampleData, _tempSampleDataSize) == -1) {
+      return true;
+    }
+
+    _tempSampleData = NULL;
+    _tempSampleDataSize = 0;
+
+    PaLock();
+    while (true) {
+      // Ack the last thing we read
+      if (LATE(pa_stream_drop)(_recStream) != 0) {
+        RTC_LOG(LS_WARNING)
+            << "failed to drop, err=" << LATE(pa_context_errno)(_paContext);
+      }
+
+      if (LATE(pa_stream_readable_size)(_recStream) <= 0) {
+        // Then that was all the data
+        break;
+      }
+
+      // Else more data.
+      const void* sampleData;
+      size_t sampleDataSize;
+
+      if (LATE(pa_stream_peek)(_recStream, &sampleData, &sampleDataSize) != 0) {
+        RTC_LOG(LS_ERROR) << "RECORD_ERROR, error = "
+                          << LATE(pa_context_errno)(_paContext);
+        break;
+      }
+
+      // Drop lock for sigslot dispatch, which could take a while.
+      PaUnLock();
+      // Read data and provide it to VoiceEngine
+      if (ReadRecordedData(sampleData, sampleDataSize) == -1) {
+        return true;
+      }
+      PaLock();
+
+      // Return to top of loop for the ack and the check for more data.
+    }
+
+    EnableReadCallback();
+    PaUnLock();
+
+  }  // _recording
+
+  return true;
+}
+
+bool AudioDeviceLinuxPulse::KeyPressed() const {
+#if defined(WEBRTC_USE_X11)
+  char szKey[32];
+  unsigned int i = 0;
+  char state = 0;
+
+  if (!_XDisplay)
+    return false;
+
+  // Check key map status
+  XQueryKeymap(_XDisplay, szKey);
+
+  // A bit change in keymap means a key is pressed
+  for (i = 0; i < sizeof(szKey); i++)
+    state |= (szKey[i] ^ _oldKeyState[i]) & szKey[i];
+
+  // Save old state
+  memcpy((char*)_oldKeyState, (char*)szKey, sizeof(_oldKeyState));
+  return (state != 0);
+#else
+  return false;
+#endif
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.h
new file mode 100644
index 0000000000..0cf89ef011
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.h
@@ -0,0 +1,349 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
+#include "rtc_base/event.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+#if defined(WEBRTC_USE_X11)
+#include <X11/Xlib.h>
+#endif
+
+#include <pulse/pulseaudio.h>
+#include <stddef.h>
+#include <stdint.h>
+
+// We define this flag if it's missing from our headers, because we want to be
+// able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
+// if run against a recent version of the library.
+#ifndef PA_STREAM_ADJUST_LATENCY
+#define PA_STREAM_ADJUST_LATENCY 0x2000U
+#endif
+#ifndef PA_STREAM_START_MUTED
+#define PA_STREAM_START_MUTED 0x1000U
+#endif
+
+// Set this constant to 0 to disable latency reading
+const uint32_t WEBRTC_PA_REPORT_LATENCY = 1;
+
+// Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]
+
+// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
+const uint32_t WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;
+
+// Some timing constants for optimal operation. See
+// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
+// for a good explanation of some of the factors that go into this.
+
+// Playback.
+
+// For playback, there is a round-trip delay to fill the server-side playback
+// buffer, so setting too low of a latency is a buffer underflow risk. We will
+// automatically increase the latency if a buffer underflow does occur, but we
+// also enforce a sane minimum at start-up time. Anything lower would be
+// virtually guaranteed to underflow at least once, so there's no point in
+// allowing lower latencies.
+const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;
+
+// Every time a playback stream underflows, we will reconfigure it with target
+// latency that is greater by this amount.
+const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;
+
+// We also need to configure a suitable request size. Too small and we'd burn
+// CPU from the overhead of transfering small amounts of data at once. Too large
+// and the amount of data remaining in the buffer right before refilling it
+// would be a buffer underflow risk. We set it to half of the buffer size.
+const uint32_t WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;
+
+// Capture.
+
+// For capture, low latency is not a buffer overflow risk, but it makes us burn
+// CPU from the overhead of transfering small amounts of data at once, so we set
+// a recommended value that we use for the kLowLatency constant (but if the user
+// explicitly requests something lower then we will honour it).
+// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
+const uint32_t WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;
+
+// There is a round-trip delay to ack the data to the server, so the
+// server-side buffer needs extra space to prevent buffer overflow. 20ms is
+// sufficient, but there is no penalty to making it bigger, so we make it huge.
+// (750ms is libpulse's default value for the _total_ buffer size in the
+// kNoLatencyRequirements case.)
+const uint32_t WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;
+
+const uint32_t WEBRTC_PA_MSECS_PER_SEC = 1000;
+
+// Init _configuredLatencyRec/Play to this value to disable latency requirements
+const int32_t WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
+
+// Set this const to 1 to account for peeked and used data in latency
+// calculation
+const uint32_t WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
+
+typedef webrtc::adm_linux_pulse::PulseAudioSymbolTable WebRTCPulseSymbolTable;
+WebRTCPulseSymbolTable* GetPulseSymbolTable();
+
+namespace webrtc {
+
+class AudioDeviceLinuxPulse : public AudioDeviceGeneric {
+ public:
+  AudioDeviceLinuxPulse();
+  virtual ~AudioDeviceLinuxPulse();
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() override;
+  int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Playing() const override;
+  int32_t StartRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+
+ private:
+  void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }
+  void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }
+  void WaitForOperationCompletion(pa_operation* paOperation) const;
+  void WaitForSuccess(pa_operation* paOperation) const;
+
+  bool KeyPressed() const;
+
+  static void PaContextStateCallback(pa_context* c, void* pThis);
+  static void PaSinkInfoCallback(pa_context* c,
+                                 const pa_sink_info* i,
+                                 int eol,
+                                 void* pThis);
+  static void PaSourceInfoCallback(pa_context* c,
+                                   const pa_source_info* i,
+                                   int eol,
+                                   void* pThis);
+  static void PaServerInfoCallback(pa_context* c,
+                                   const pa_server_info* i,
+                                   void* pThis);
+  static void PaStreamStateCallback(pa_stream* p, void* pThis);
+  void PaContextStateCallbackHandler(pa_context* c);
+  void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
+  void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
+  void PaServerInfoCallbackHandler(const pa_server_info* i);
+  void PaStreamStateCallbackHandler(pa_stream* p);
+
+  void EnableWriteCallback();
+  void DisableWriteCallback();
+  static void PaStreamWriteCallback(pa_stream* unused,
+                                    size_t buffer_space,
+                                    void* pThis);
+  void PaStreamWriteCallbackHandler(size_t buffer_space);
+  static void PaStreamUnderflowCallback(pa_stream* unused, void* pThis);
+  void PaStreamUnderflowCallbackHandler();
+  void EnableReadCallback();
+  void DisableReadCallback();
+  static void PaStreamReadCallback(pa_stream* unused1,
+                                   size_t unused2,
+                                   void* pThis);
+  void PaStreamReadCallbackHandler();
+  static void PaStreamOverflowCallback(pa_stream* unused, void* pThis);
+  void PaStreamOverflowCallbackHandler();
+  int32_t LatencyUsecs(pa_stream* stream);
+  int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
+  int32_t ProcessRecordedData(int8_t* bufferData,
+                              uint32_t bufferSizeInSamples,
+                              uint32_t recDelay);
+
+  int32_t CheckPulseAudioVersion();
+  int32_t InitSamplingFrequency();
+  int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
+  int32_t InitPulseAudio();
+  int32_t TerminatePulseAudio();
+
+  void PaLock();
+  void PaUnLock();
+
+  static void RecThreadFunc(void*);
+  static void PlayThreadFunc(void*);
+  bool RecThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);
+  bool PlayThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);
+
+  AudioDeviceBuffer* _ptrAudioBuffer;
+
+  mutable Mutex mutex_;
+  rtc::Event _timeEventRec;
+  rtc::Event _timeEventPlay;
+  rtc::Event _recStartEvent;
+  rtc::Event _playStartEvent;
+
+  rtc::PlatformThread _ptrThreadPlay;
+  rtc::PlatformThread _ptrThreadRec;
+
+  AudioMixerManagerLinuxPulse _mixerManager;
+
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+  bool _inputDeviceIsSpecified;
+  bool _outputDeviceIsSpecified;
+
+  int sample_rate_hz_;
+  uint8_t _recChannels;
+  uint8_t _playChannels;
+
+  // Stores thread ID in constructor.
+  // We can then use RTC_DCHECK_RUN_ON(&worker_thread_checker_) to ensure that
+  // other methods are called from the same thread.
+  // Currently only does RTC_DCHECK(thread_checker_.IsCurrent()).
+  SequenceChecker thread_checker_;
+
+  bool _initialized;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+  bool _startRec;
+  bool _startPlay;
+  bool update_speaker_volume_at_startup_;
+  bool quit_ RTC_GUARDED_BY(&mutex_);
+
+  uint32_t _sndCardPlayDelay RTC_GUARDED_BY(&mutex_);
+
+  int32_t _writeErrors;
+
+  uint16_t _deviceIndex;
+  int16_t _numPlayDevices;
+  int16_t _numRecDevices;
+  char* _playDeviceName;
+  char* _recDeviceName;
+  char* _playDisplayDeviceName;
+  char* _recDisplayDeviceName;
+  char _paServerVersion[32];
+
+  int8_t* _playBuffer;
+  size_t _playbackBufferSize;
+  size_t _playbackBufferUnused;
+  size_t _tempBufferSpace;
+  int8_t* _recBuffer;
+  size_t _recordBufferSize;
+  size_t _recordBufferUsed;
+  const void* _tempSampleData;
+  size_t _tempSampleDataSize;
+  int32_t _configuredLatencyPlay;
+  int32_t _configuredLatencyRec;
+
+  // PulseAudio
+  uint16_t _paDeviceIndex;
+  bool _paStateChanged;
+
+  pa_threaded_mainloop* _paMainloop;
+  pa_mainloop_api* _paMainloopApi;
+  pa_context* _paContext;
+
+  pa_stream* _recStream;
+  pa_stream* _playStream;
+  uint32_t _recStreamFlags;
+  uint32_t _playStreamFlags;
+  pa_buffer_attr _playBufferAttr;
+  pa_buffer_attr _recBufferAttr;
+
+  char _oldKeyState[32];
+#if defined(WEBRTC_USE_X11)
+  Display* _XDisplay;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
new file mode 100644
index 0000000000..e7e7033173
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
@@ -0,0 +1,979 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
+
+#include "modules/audio_device/linux/audio_device_alsa_linux.h"
+#include "rtc_base/logging.h"
+
+// Accesses ALSA functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libasound, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                                            \
+  LATESYM_GET(webrtc::adm_linux_alsa::AlsaSymbolTable, GetAlsaSymbolTable(), \
+              sym)
+
+namespace webrtc {
+
+AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA()
+    : _outputMixerHandle(NULL),
+      _inputMixerHandle(NULL),
+      _outputMixerElement(NULL),
+      _inputMixerElement(NULL) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+
+  memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
+  memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
+}
+
+AudioMixerManagerLinuxALSA::~AudioMixerManagerLinuxALSA() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+  Close();
+}
+
+// ============================================================================
+//                                    PUBLIC METHODS
+// ============================================================================
+
+int32_t AudioMixerManagerLinuxALSA::Close() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  MutexLock lock(&mutex_);
+
+  CloseSpeakerLocked();
+  CloseMicrophoneLocked();
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseSpeaker() {
+  MutexLock lock(&mutex_);
+  return CloseSpeakerLocked();
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseSpeakerLocked() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  int errVal = 0;
+
+  if (_outputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing playout mixer";
+    LATE(snd_mixer_free)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+    _outputMixerHandle = NULL;
+    _outputMixerElement = NULL;
+  }
+  memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseMicrophone() {
+  MutexLock lock(&mutex_);
+  return CloseMicrophoneLocked();
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseMicrophoneLocked() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  int errVal = 0;
+
+  if (_inputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    LATE(snd_mixer_free)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer 2";
+
+    errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer 3";
+
+    errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer 4";
+    _inputMixerHandle = NULL;
+    _inputMixerElement = NULL;
+  }
+  memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenSpeaker(name="
+                      << deviceName << ")";
+
+  MutexLock lock(&mutex_);
+
+  int errVal = 0;
+
+  // Close any existing output mixer handle
+  //
+  if (_outputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing playout mixer";
+
+    LATE(snd_mixer_free)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+  }
+  _outputMixerHandle = NULL;
+  _outputMixerElement = NULL;
+
+  errVal = LATE(snd_mixer_open)(&_outputMixerHandle, 0);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_open(&_outputMixerHandle, 0) - error";
+    return -1;
+  }
+
+  char controlName[kAdmMaxDeviceNameSize] = {0};
+  GetControlName(controlName, deviceName);
+
+  RTC_LOG(LS_VERBOSE) << "snd_mixer_attach(_outputMixerHandle, " << controlName
+                      << ")";
+
+  errVal = LATE(snd_mixer_attach)(_outputMixerHandle, controlName);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_attach(_outputMixerHandle, " << controlName
+                      << ") error: " << LATE(snd_strerror)(errVal);
+    _outputMixerHandle = NULL;
+    return -1;
+  }
+  strcpy(_outputMixerStr, controlName);
+
+  errVal = LATE(snd_mixer_selem_register)(_outputMixerHandle, NULL, NULL);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR)
+        << "snd_mixer_selem_register(_outputMixerHandle, NULL, NULL), "
+           "error: "
+        << LATE(snd_strerror)(errVal);
+    _outputMixerHandle = NULL;
+    return -1;
+  }
+
+  // Load and find the proper mixer element
+  if (LoadSpeakerMixerElement() < 0) {
+    return -1;
+  }
+
+  if (_outputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "the output mixer device is now open ("
+                        << _outputMixerHandle << ")";
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenMicrophone(name="
+                      << deviceName << ")";
+
+  MutexLock lock(&mutex_);
+
+  int errVal = 0;
+
+  // Close any existing input mixer handle
+  //
+  if (_inputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    LATE(snd_mixer_free)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+  }
+  _inputMixerHandle = NULL;
+  _inputMixerElement = NULL;
+
+  errVal = LATE(snd_mixer_open)(&_inputMixerHandle, 0);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_open(&_inputMixerHandle, 0) - error";
+    return -1;
+  }
+
+  char controlName[kAdmMaxDeviceNameSize] = {0};
+  GetControlName(controlName, deviceName);
+
+  RTC_LOG(LS_VERBOSE) << "snd_mixer_attach(_inputMixerHandle, " << controlName
+                      << ")";
+
+  errVal = LATE(snd_mixer_attach)(_inputMixerHandle, controlName);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_attach(_inputMixerHandle, " << controlName
+                      << ") error: " << LATE(snd_strerror)(errVal);
+
+    _inputMixerHandle = NULL;
+    return -1;
+  }
+  strcpy(_inputMixerStr, controlName);
+
+  errVal = LATE(snd_mixer_selem_register)(_inputMixerHandle, NULL, NULL);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR)
+        << "snd_mixer_selem_register(_inputMixerHandle, NULL, NULL), "
+           "error: "
+        << LATE(snd_strerror)(errVal);
+
+    _inputMixerHandle = NULL;
+    return -1;
+  }
+  // Load and find the proper mixer element
+  if (LoadMicMixerElement() < 0) {
+    return -1;
+  }
+
+  if (_inputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "the input mixer device is now open ("
+                        << _inputMixerHandle << ")";
+  }
+
+  return 0;
+}
+
+bool AudioMixerManagerLinuxALSA::SpeakerIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_outputMixerHandle != NULL);
+}
+
+bool AudioMixerManagerLinuxALSA::MicrophoneIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_inputMixerHandle != NULL);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerVolume(volume="
+                      << volume << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  int errVal = LATE(snd_mixer_selem_set_playback_volume_all)(
+      _outputMixerElement, volume);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error changing master volume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerVolume(uint32_t& volume) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  long int vol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_playback_volume)(
+      _outputMixerElement, (snd_mixer_selem_channel_id_t)0, &vol);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting outputvolume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SpeakerVolume() => vol="
+                      << vol;
+
+  volume = static_cast<uint32_t>(vol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MaxSpeakerVolume(
+    uint32_t& maxVolume) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avilable output mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_playback_volume_range)(
+      _outputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  maxVolume = static_cast<uint32_t>(maxVol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MinSpeakerVolume(
+    uint32_t& minVolume) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_playback_volume_range)(
+      _outputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  minVolume = static_cast<uint32_t>(minVol);
+
+  return 0;
+}
+
+// TL: Have done testnig with these but they don't seem reliable and
+// they were therefore not added
+/*
+ // ----------------------------------------------------------------------------
+ //    SetMaxSpeakerVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMaxSpeakerVolume(
+     uint32_t maxVolume)
+ {
+
+ if (_outputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_playback_volume_range(
+ _outputMixerElement, &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ RTC_LOG(LS_WARNING) << "Error getting playback volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ maxVol = maxVolume;
+ errVal = snd_mixer_selem_set_playback_volume_range(
+ _outputMixerElement, minVol, maxVol);
+ RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting playback volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+
+ // ----------------------------------------------------------------------------
+ //    SetMinSpeakerVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMinSpeakerVolume(
+     uint32_t minVolume)
+ {
+
+ if (_outputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_playback_volume_range(
+ _outputMixerElement, &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ RTC_LOG(LS_WARNING) << "Error getting playback volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ minVol = minVolume;
+ errVal = snd_mixer_selem_set_playback_volume_range(
+ _outputMixerElement, minVol, maxVol);
+ RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting playback volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+ */
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerVolumeIsAvailable(bool& available) {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_playback_volume)(_outputMixerElement);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerMuteIsAvailable(bool& available) {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerMute(enable="
+                      << enable << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  // Ensure that the selected speaker destination has a valid mute control.
+  bool available(false);
+  SpeakerMuteIsAvailable(available);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the speaker";
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  int errVal = LATE(snd_mixer_selem_set_playback_switch_all)(
+      _outputMixerElement, !enable);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error setting playback switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerMute(bool& enabled) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer exists";
+    return -1;
+  }
+
+  // Ensure that the selected speaker destination has a valid mute control.
+  bool available =
+      LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the speaker";
+    return -1;
+  }
+
+  int value(false);
+
+  // Retrieve one boolean control value for a specified mute-control
+  //
+  int errVal = LATE(snd_mixer_selem_get_playback_switch)(
+      _outputMixerElement, (snd_mixer_selem_channel_id_t)0, &value);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting playback switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  enabled = (bool)!value;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneMuteIsAvailable(bool& available) {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetMicrophoneMute(enable="
+                      << enable << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  // Ensure that the selected microphone destination has a valid mute control.
+  bool available(false);
+  MicrophoneMuteIsAvailable(available);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the microphone";
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  int errVal =
+      LATE(snd_mixer_selem_set_capture_switch_all)(_inputMixerElement, !enable);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error setting capture switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneMute(bool& enabled) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer exists";
+    return -1;
+  }
+
+  // Ensure that the selected microphone destination has a valid mute control.
+  bool available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the microphone";
+    return -1;
+  }
+
+  int value(false);
+
+  // Retrieve one boolean control value for a specified mute-control
+  //
+  int errVal = LATE(snd_mixer_selem_get_capture_switch)(
+      _inputMixerElement, (snd_mixer_selem_channel_id_t)0, &value);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting capture switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  enabled = (bool)!value;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneVolumeIsAvailable(
+    bool& available) {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxALSA::SetMicrophoneVolume(volume=" << volume
+      << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  int errVal =
+      LATE(snd_mixer_selem_set_capture_volume_all)(_inputMixerElement, volume);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error changing microphone volume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+// TL: Have done testnig with these but they don't seem reliable and
+// they were therefore not added
+/*
+ // ----------------------------------------------------------------------------
+ //    SetMaxMicrophoneVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMaxMicrophoneVolume(
+     uint32_t maxVolume)
+ {
+
+ if (_inputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_capture_volume_range(_inputMixerElement,
+  &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ RTC_LOG(LS_WARNING) << "Error getting capture volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ maxVol = (long int)maxVolume;
+ printf("min %d max %d", minVol, maxVol);
+ errVal = snd_mixer_selem_set_capture_volume_range(_inputMixerElement, minVol,
+ maxVol); RTC_LOG(LS_VERBOSE) << "Capture hardware volume range, min: " <<
+ minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting capture volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+
+ // ----------------------------------------------------------------------------
+ //    SetMinMicrophoneVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMinMicrophoneVolume(
+ uint32_t minVolume)
+ {
+
+ if (_inputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_capture_volume_range(
+ _inputMixerElement, &minVol, &maxVol);
+ if (maxVol <= minVol)
+ {
+ //maxVol = 255;
+ RTC_LOG(LS_WARNING) << "Error getting capture volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ printf("min %d max %d", minVol, maxVol);
+ minVol = (long int)minVolume;
+ errVal = snd_mixer_selem_set_capture_volume_range(
+ _inputMixerElement, minVol, maxVol);
+ RTC_LOG(LS_VERBOSE) << "Capture hardware volume range, min: " << minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting capture volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+ */
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneVolume(uint32_t& volume) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  long int vol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_capture_volume)(
+      _inputMixerElement, (snd_mixer_selem_channel_id_t)0, &vol);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting inputvolume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxALSA::MicrophoneVolume() => vol=" << vol;
+
+  volume = static_cast<uint32_t>(vol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MaxMicrophoneVolume(
+    uint32_t& maxVolume) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  // check if we have mic volume at all
+  if (!LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement)) {
+    RTC_LOG(LS_ERROR) << "No microphone volume available";
+    return -1;
+  }
+
+  int errVal = LATE(snd_mixer_selem_get_capture_volume_range)(
+      _inputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting microphone volume range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  maxVolume = static_cast<uint32_t>(maxVol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MinMicrophoneVolume(
+    uint32_t& minVolume) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_capture_volume_range)(
+      _inputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting microphone volume range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  minVolume = static_cast<uint32_t>(minVol);
+
+  return 0;
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+int32_t AudioMixerManagerLinuxALSA::LoadMicMixerElement() const {
+  int errVal = LATE(snd_mixer_load)(_inputMixerHandle);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_load(_inputMixerHandle), error: "
+                      << LATE(snd_strerror)(errVal);
+    _inputMixerHandle = NULL;
+    return -1;
+  }
+
+  snd_mixer_elem_t* elem = NULL;
+  snd_mixer_elem_t* micElem = NULL;
+  unsigned mixerIdx = 0;
+  const char* selemName = NULL;
+
+  // Find and store handles to the right mixer elements
+  for (elem = LATE(snd_mixer_first_elem)(_inputMixerHandle); elem;
+       elem = LATE(snd_mixer_elem_next)(elem), mixerIdx++) {
+    if (LATE(snd_mixer_selem_is_active)(elem)) {
+      selemName = LATE(snd_mixer_selem_get_name)(elem);
+      if (strcmp(selemName, "Capture") == 0)  // "Capture", "Mic"
+      {
+        _inputMixerElement = elem;
+        RTC_LOG(LS_VERBOSE) << "Capture element set";
+      } else if (strcmp(selemName, "Mic") == 0) {
+        micElem = elem;
+        RTC_LOG(LS_VERBOSE) << "Mic element found";
+      }
+    }
+
+    if (_inputMixerElement) {
+      // Use the first Capture element that is found
+      // The second one may not work
+      break;
+    }
+  }
+
+  if (_inputMixerElement == NULL) {
+    // We didn't find a Capture handle, use Mic.
+    if (micElem != NULL) {
+      _inputMixerElement = micElem;
+      RTC_LOG(LS_VERBOSE) << "Using Mic as capture volume.";
+    } else {
+      _inputMixerElement = NULL;
+      RTC_LOG(LS_ERROR) << "Could not find capture volume on the mixer.";
+
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::LoadSpeakerMixerElement() const {
+  int errVal = LATE(snd_mixer_load)(_outputMixerHandle);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_load(_outputMixerHandle), error: "
+                      << LATE(snd_strerror)(errVal);
+    _outputMixerHandle = NULL;
+    return -1;
+  }
+
+  snd_mixer_elem_t* elem = NULL;
+  snd_mixer_elem_t* masterElem = NULL;
+  snd_mixer_elem_t* speakerElem = NULL;
+  unsigned mixerIdx = 0;
+  const char* selemName = NULL;
+
+  // Find and store handles to the right mixer elements
+  for (elem = LATE(snd_mixer_first_elem)(_outputMixerHandle); elem;
+       elem = LATE(snd_mixer_elem_next)(elem), mixerIdx++) {
+    if (LATE(snd_mixer_selem_is_active)(elem)) {
+      selemName = LATE(snd_mixer_selem_get_name)(elem);
+      RTC_LOG(LS_VERBOSE) << "snd_mixer_selem_get_name " << mixerIdx << ": "
+                          << selemName << " =" << elem;
+
+      // "Master", "PCM", "Wave", "Master Mono", "PC Speaker", "PCM", "Wave"
+      if (strcmp(selemName, "PCM") == 0) {
+        _outputMixerElement = elem;
+        RTC_LOG(LS_VERBOSE) << "PCM element set";
+      } else if (strcmp(selemName, "Master") == 0) {
+        masterElem = elem;
+        RTC_LOG(LS_VERBOSE) << "Master element found";
+      } else if (strcmp(selemName, "Speaker") == 0) {
+        speakerElem = elem;
+        RTC_LOG(LS_VERBOSE) << "Speaker element found";
+      }
+    }
+
+    if (_outputMixerElement) {
+      // We have found the element we want
+      break;
+    }
+  }
+
+  // If we didn't find a PCM Handle, use Master or Speaker
+  if (_outputMixerElement == NULL) {
+    if (masterElem != NULL) {
+      _outputMixerElement = masterElem;
+      RTC_LOG(LS_VERBOSE) << "Using Master as output volume.";
+    } else if (speakerElem != NULL) {
+      _outputMixerElement = speakerElem;
+      RTC_LOG(LS_VERBOSE) << "Using Speaker as output volume.";
+    } else {
+      _outputMixerElement = NULL;
+      RTC_LOG(LS_ERROR) << "Could not find output volume in the mixer.";
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+void AudioMixerManagerLinuxALSA::GetControlName(char* controlName,
+                                                char* deviceName) const {
+  // Example
+  // deviceName: "front:CARD=Intel,DEV=0"
+  // controlName: "hw:CARD=Intel"
+  char* pos1 = strchr(deviceName, ':');
+  char* pos2 = strchr(deviceName, ',');
+  if (!pos2) {
+    // Can also be default:CARD=Intel
+    pos2 = &deviceName[strlen(deviceName)];
+  }
+  if (pos1 && pos2) {
+    strcpy(controlName, "hw");
+    int nChar = (int)(pos2 - pos1);
+    strncpy(&controlName[2], pos1, nChar);
+    controlName[2 + nChar] = '\0';
+  } else {
+    strcpy(controlName, deviceName);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
new file mode 100644
index 0000000000..d98287822d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
+
+#include <alsa/asoundlib.h>
+
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/linux/alsasymboltable_linux.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class AudioMixerManagerLinuxALSA {
+ public:
+  int32_t OpenSpeaker(char* deviceName) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t OpenMicrophone(char* deviceName) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SetSpeakerVolume(uint32_t volume) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SpeakerVolume(uint32_t& volume) const;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  int32_t SpeakerVolumeIsAvailable(bool& available);
+  int32_t SpeakerMuteIsAvailable(bool& available);
+  int32_t SetSpeakerMute(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SpeakerMute(bool& enabled) const;
+  int32_t MicrophoneMuteIsAvailable(bool& available);
+  int32_t SetMicrophoneMute(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t MicrophoneMute(bool& enabled) const;
+  int32_t MicrophoneVolumeIsAvailable(bool& available);
+  int32_t SetMicrophoneVolume(uint32_t volume) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t MicrophoneVolume(uint32_t& volume) const;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  int32_t Close() RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t CloseSpeaker() RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t CloseMicrophone() RTC_LOCKS_EXCLUDED(mutex_);
+  bool SpeakerIsInitialized() const;
+  bool MicrophoneIsInitialized() const;
+
+ public:
+  AudioMixerManagerLinuxALSA();
+  ~AudioMixerManagerLinuxALSA();
+
+ private:
+  int32_t CloseSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t CloseMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t LoadMicMixerElement() const;
+  int32_t LoadSpeakerMixerElement() const;
+  void GetControlName(char* controlName, char* deviceName) const;
+
+ private:
+  Mutex mutex_;
+  mutable snd_mixer_t* _outputMixerHandle;
+  char _outputMixerStr[kAdmMaxDeviceNameSize];
+  mutable snd_mixer_t* _inputMixerHandle;
+  char _inputMixerStr[kAdmMaxDeviceNameSize];
+  mutable snd_mixer_elem_t* _outputMixerElement;
+  mutable snd_mixer_elem_t* _inputMixerElement;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
new file mode 100644
index 0000000000..91beee3c87
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
@@ -0,0 +1,844 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
+
+#include <stddef.h>
+
+#include "modules/audio_device/linux/audio_device_pulse_linux.h"
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+// Accesses Pulse functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libpulse, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                             \
+  LATESYM_GET(webrtc::adm_linux_pulse::PulseAudioSymbolTable, \
+              GetPulseSymbolTable(), sym)
+
+namespace webrtc {
+
+class AutoPulseLock {
+ public:
+  explicit AutoPulseLock(pa_threaded_mainloop* pa_mainloop)
+      : pa_mainloop_(pa_mainloop) {
+    LATE(pa_threaded_mainloop_lock)(pa_mainloop_);
+  }
+
+  ~AutoPulseLock() { LATE(pa_threaded_mainloop_unlock)(pa_mainloop_); }
+
+ private:
+  pa_threaded_mainloop* const pa_mainloop_;
+};
+
+AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse()
+    : _paOutputDeviceIndex(-1),
+      _paInputDeviceIndex(-1),
+      _paPlayStream(NULL),
+      _paRecStream(NULL),
+      _paMainloop(NULL),
+      _paContext(NULL),
+      _paVolume(0),
+      _paMute(0),
+      _paVolSteps(0),
+      _paSpeakerMute(false),
+      _paSpeakerVolume(PA_VOLUME_NORM),
+      _paChannels(0),
+      _paObjectsSet(false) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+}
+
+AudioMixerManagerLinuxPulse::~AudioMixerManagerLinuxPulse() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+
+  Close();
+}
+
+// ===========================================================================
+//                                    PUBLIC METHODS
+// ===========================================================================
+
+int32_t AudioMixerManagerLinuxPulse::SetPulseAudioObjects(
+    pa_threaded_mainloop* mainloop,
+    pa_context* context) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  if (!mainloop || !context) {
+    RTC_LOG(LS_ERROR) << "could not set PulseAudio objects for mixer";
+    return -1;
+  }
+
+  _paMainloop = mainloop;
+  _paContext = context;
+  _paObjectsSet = true;
+
+  RTC_LOG(LS_VERBOSE) << "the PulseAudio objects for the mixer has been set";
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::Close() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  CloseSpeaker();
+  CloseMicrophone();
+
+  _paMainloop = NULL;
+  _paContext = NULL;
+  _paObjectsSet = false;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::CloseSpeaker() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  // Reset the index to -1
+  _paOutputDeviceIndex = -1;
+  _paPlayStream = NULL;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::CloseMicrophone() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  // Reset the index to -1
+  _paInputDeviceIndex = -1;
+  _paRecStream = NULL;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetPlayStream(pa_stream* playStream) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SetPlayStream(playStream)";
+
+  _paPlayStream = playStream;
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetRecStream(pa_stream* recStream) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetRecStream(recStream)";
+
+  _paRecStream = recStream;
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::OpenSpeaker(uint16_t deviceIndex) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex="
+                      << deviceIndex << ")";
+
+  // No point in opening the speaker
+  // if PA objects have not been set
+  if (!_paObjectsSet) {
+    RTC_LOG(LS_ERROR) << "PulseAudio objects has not been set";
+    return -1;
+  }
+
+  // Set the index for the PulseAudio
+  // output device to control
+  _paOutputDeviceIndex = deviceIndex;
+
+  RTC_LOG(LS_VERBOSE) << "the output mixer device is now open";
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::OpenMicrophone(uint16_t deviceIndex) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex="
+      << deviceIndex << ")";
+
+  // No point in opening the microphone
+  // if PA objects have not been set
+  if (!_paObjectsSet) {
+    RTC_LOG(LS_ERROR) << "PulseAudio objects have not been set";
+    return -1;
+  }
+
+  // Set the index for the PulseAudio
+  // input device to control
+  _paInputDeviceIndex = deviceIndex;
+
+  RTC_LOG(LS_VERBOSE) << "the input mixer device is now open";
+
+  return 0;
+}
+
+bool AudioMixerManagerLinuxPulse::SpeakerIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_paOutputDeviceIndex != -1);
+}
+
+bool AudioMixerManagerLinuxPulse::MicrophoneIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_paInputDeviceIndex != -1);
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(uint32_t volume) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerVolume(volume="
+                      << volume << ")";
+
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  bool setFailed(false);
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only really set the volume if we have a connected stream
+    AutoPulseLock auto_lock(_paMainloop);
+
+    // Get the number of channels from the sample specification
+    const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_paPlayStream);
+    if (!spec) {
+      RTC_LOG(LS_ERROR) << "could not get sample specification";
+      return -1;
+    }
+
+    // Set the same volume for all channels
+    pa_cvolume cVolumes;
+    LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
+
+    pa_operation* paOperation = NULL;
+    paOperation = LATE(pa_context_set_sink_input_volume)(
+        _paContext, LATE(pa_stream_get_index)(_paPlayStream), &cVolumes,
+        PaSetVolumeCallback, NULL);
+    if (!paOperation) {
+      setFailed = true;
+    }
+
+    // Don't need to wait for the completion
+    LATE(pa_operation_unref)(paOperation);
+  } else {
+    // We have not created a stream or it's not connected to the sink
+    // Save the volume to be set at connection
+    _paSpeakerVolume = volume;
+  }
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not set speaker volume, error="
+                        << LATE(pa_context_errno)(_paContext);
+
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerVolume(uint32_t& volume) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only get the volume if we have a connected stream
+    if (!GetSinkInputInfo())
+      return -1;
+
+    AutoPulseLock auto_lock(_paMainloop);
+    volume = static_cast<uint32_t>(_paVolume);
+  } else {
+    AutoPulseLock auto_lock(_paMainloop);
+    volume = _paSpeakerVolume;
+  }
+
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SpeakerVolume() => vol="
+                      << volume;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MaxSpeakerVolume(
+    uint32_t& maxVolume) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  // PA_VOLUME_NORM corresponds to 100% (0db)
+  // but PA allows up to 150 db amplification
+  maxVolume = static_cast<uint32_t>(PA_VOLUME_NORM);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MinSpeakerVolume(
+    uint32_t& minVolume) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  minVolume = static_cast<uint32_t>(PA_VOLUME_MUTED);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerVolumeIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerMuteIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerMute(enable="
+                      << enable << ")";
+
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  bool setFailed(false);
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only really mute if we have a connected stream
+    AutoPulseLock auto_lock(_paMainloop);
+
+    pa_operation* paOperation = NULL;
+    paOperation = LATE(pa_context_set_sink_input_mute)(
+        _paContext, LATE(pa_stream_get_index)(_paPlayStream), (int)enable,
+        PaSetVolumeCallback, NULL);
+    if (!paOperation) {
+      setFailed = true;
+    }
+
+    // Don't need to wait for the completion
+    LATE(pa_operation_unref)(paOperation);
+  } else {
+    // We have not created a stream or it's not connected to the sink
+    // Save the mute status to be set at connection
+    _paSpeakerMute = enable;
+  }
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not mute speaker, error="
+                        << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerMute(bool& enabled) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only get the mute status if we have a connected stream
+    if (!GetSinkInputInfo())
+      return -1;
+
+    enabled = static_cast<bool>(_paMute);
+  } else {
+    enabled = _paSpeakerMute;
+  }
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SpeakerMute() => enabled=" << enabled;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::StereoPlayoutIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paOutputDeviceIndex;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+
+    // Get the actual stream device index if we have a connected stream
+    // The device used by the stream can be changed
+    // during the call
+    if (_paPlayStream &&
+        (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+      deviceIndex = LATE(pa_stream_get_device_index)(_paPlayStream);
+    }
+  }
+
+  if (!GetSinkInfoByIndex(deviceIndex))
+    return -1;
+
+  available = static_cast<bool>(_paChannels == 2);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(
+    bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  AutoPulseLock auto_lock(_paMainloop);
+
+  // Get the actual stream device index if we have a connected stream
+  // The device used by the stream can be changed
+  // during the call
+  if (_paRecStream &&
+      (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+    deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+  }
+
+  pa_operation* paOperation = NULL;
+
+  // Get info for this source
+  // We want to know if the actual device can record in stereo
+  paOperation = LATE(pa_context_get_source_info_by_index)(
+      _paContext, deviceIndex, PaSourceInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+
+  available = static_cast<bool>(_paChannels == 2);
+
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
+         " => available="
+      << available;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneMuteIsAvailable(
+    bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SetMicrophoneMute(enable=" << enable
+      << ")";
+
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  bool setFailed(false);
+  pa_operation* paOperation = NULL;
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  AutoPulseLock auto_lock(_paMainloop);
+
+  // Get the actual stream device index if we have a connected stream
+  // The device used by the stream can be changed
+  // during the call
+  if (_paRecStream &&
+      (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+    deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+  }
+
+  // Set mute switch for the source
+  paOperation = LATE(pa_context_set_source_mute_by_index)(
+      _paContext, deviceIndex, enable, PaSetVolumeCallback, NULL);
+
+  if (!paOperation) {
+    setFailed = true;
+  }
+
+  // Don't need to wait for this to complete.
+  LATE(pa_operation_unref)(paOperation);
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not mute microphone, error="
+                        << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+    // Get the actual stream device index if we have a connected stream
+    // The device used by the stream can be changed
+    // during the call
+    if (_paRecStream &&
+        (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+      deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+    }
+  }
+
+  if (!GetSourceInfoByIndex(deviceIndex))
+    return -1;
+
+  enabled = static_cast<bool>(_paMute);
+
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::MicrophoneMute() => enabled=" << enabled;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneVolumeIsAvailable(
+    bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SetMicrophoneVolume(volume=" << volume
+      << ")";
+
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // Unlike output streams, input streams have no concept of a stream
+  // volume, only a device volume. So we have to change the volume of the
+  // device itself.
+
+  // The device may have a different number of channels than the stream and
+  // their mapping may be different, so we don't want to use the channel
+  // count from our sample spec. We could use PA_CHANNELS_MAX to cover our
+  // bases, and the server allows that even if the device's channel count
+  // is lower, but some buggy PA clients don't like that (the pavucontrol
+  // on Hardy dies in an assert if the channel count is different). So
+  // instead we look up the actual number of channels that the device has.
+  AutoPulseLock auto_lock(_paMainloop);
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  // Get the actual stream device index if we have a connected stream
+  // The device used by the stream can be changed
+  // during the call
+  if (_paRecStream &&
+      (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+    deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+  }
+
+  bool setFailed(false);
+  pa_operation* paOperation = NULL;
+
+  // Get the number of channels for this source
+  paOperation = LATE(pa_context_get_source_info_by_index)(
+      _paContext, deviceIndex, PaSourceInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+
+  uint8_t channels = _paChannels;
+  pa_cvolume cVolumes;
+  LATE(pa_cvolume_set)(&cVolumes, channels, volume);
+
+  // Set the volume for the source
+  paOperation = LATE(pa_context_set_source_volume_by_index)(
+      _paContext, deviceIndex, &cVolumes, PaSetVolumeCallback, NULL);
+
+  if (!paOperation) {
+    setFailed = true;
+  }
+
+  // Don't need to wait for this to complete.
+  LATE(pa_operation_unref)(paOperation);
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not set microphone volume, error="
+                        << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneVolume(uint32_t& volume) const {
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+    // Get the actual stream device index if we have a connected stream.
+    // The device used by the stream can be changed during the call.
+    if (_paRecStream &&
+        (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+      deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+    }
+  }
+
+  if (!GetSourceInfoByIndex(deviceIndex))
+    return -1;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+    volume = static_cast<uint32_t>(_paVolume);
+  }
+
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::MicrophoneVolume() => vol=" << volume;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MaxMicrophoneVolume(
+    uint32_t& maxVolume) const {
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // PA_VOLUME_NORM corresponds to 100% (0db)
+  // PA allows up to 150 db amplification (PA_VOLUME_MAX)
+  // but that doesn't work well for all sound cards
+  maxVolume = static_cast<uint32_t>(PA_VOLUME_NORM);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MinMicrophoneVolume(
+    uint32_t& minVolume) const {
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  minVolume = static_cast<uint32_t>(PA_VOLUME_MUTED);
+
+  return 0;
+}
+
+// ===========================================================================
+//                                 Private Methods
+// ===========================================================================
+
+void AudioMixerManagerLinuxPulse::PaSinkInfoCallback(pa_context* /*c*/,
+                                                     const pa_sink_info* i,
+                                                     int eol,
+                                                     void* pThis) {
+  static_cast<AudioMixerManagerLinuxPulse*>(pThis)->PaSinkInfoCallbackHandler(
+      i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallback(
+    pa_context* /*c*/,
+    const pa_sink_input_info* i,
+    int eol,
+    void* pThis) {
+  static_cast<AudioMixerManagerLinuxPulse*>(pThis)
+      ->PaSinkInputInfoCallbackHandler(i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSourceInfoCallback(pa_context* /*c*/,
+                                                       const pa_source_info* i,
+                                                       int eol,
+                                                       void* pThis) {
+  static_cast<AudioMixerManagerLinuxPulse*>(pThis)->PaSourceInfoCallbackHandler(
+      i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSetVolumeCallback(pa_context* c,
+                                                      int success,
+                                                      void* /*pThis*/) {
+  if (!success) {
+    RTC_LOG(LS_ERROR) << "failed to set volume";
+  }
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInfoCallbackHandler(
+    const pa_sink_info* i,
+    int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  _paChannels = i->channel_map.channels;   // Get number of channels
+  pa_volume_t paVolume = PA_VOLUME_MUTED;  // Minimum possible value.
+  for (int j = 0; j < _paChannels; ++j) {
+    if (paVolume < i->volume.values[j]) {
+      paVolume = i->volume.values[j];
+    }
+  }
+  _paVolume = paVolume;  // get the max volume for any channel
+  _paMute = i->mute;     // get mute status
+
+  // supported since PA 0.9.15
+  //_paVolSteps = i->n_volume_steps; // get the number of volume steps
+  // default value is PA_VOLUME_NORM+1
+  _paVolSteps = PA_VOLUME_NORM + 1;
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallbackHandler(
+    const pa_sink_input_info* i,
+    int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  _paChannels = i->channel_map.channels;   // Get number of channels
+  pa_volume_t paVolume = PA_VOLUME_MUTED;  // Minimum possible value.
+  for (int j = 0; j < _paChannels; ++j) {
+    if (paVolume < i->volume.values[j]) {
+      paVolume = i->volume.values[j];
+    }
+  }
+  _paVolume = paVolume;  // Get the max volume for any channel
+  _paMute = i->mute;     // Get mute status
+}
+
+void AudioMixerManagerLinuxPulse::PaSourceInfoCallbackHandler(
+    const pa_source_info* i,
+    int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  _paChannels = i->channel_map.channels;   // Get number of channels
+  pa_volume_t paVolume = PA_VOLUME_MUTED;  // Minimum possible value.
+  for (int j = 0; j < _paChannels; ++j) {
+    if (paVolume < i->volume.values[j]) {
+      paVolume = i->volume.values[j];
+    }
+  }
+  _paVolume = paVolume;  // Get the max volume for any channel
+  _paMute = i->mute;     // Get mute status
+
+  // supported since PA 0.9.15
+  //_paVolSteps = i->n_volume_steps; // Get the number of volume steps
+  // default value is PA_VOLUME_NORM+1
+  _paVolSteps = PA_VOLUME_NORM + 1;
+}
+
+void AudioMixerManagerLinuxPulse::WaitForOperationCompletion(
+    pa_operation* paOperation) const {
+  while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING) {
+    LATE(pa_threaded_mainloop_wait)(_paMainloop);
+  }
+
+  LATE(pa_operation_unref)(paOperation);
+}
+
+bool AudioMixerManagerLinuxPulse::GetSinkInputInfo() const {
+  pa_operation* paOperation = NULL;
+
+  AutoPulseLock auto_lock(_paMainloop);
+  // Get info for this stream (sink input).
+  paOperation = LATE(pa_context_get_sink_input_info)(
+      _paContext, LATE(pa_stream_get_index)(_paPlayStream),
+      PaSinkInputInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+  return true;
+}
+
+bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(int device_index) const {
+  pa_operation* paOperation = NULL;
+
+  AutoPulseLock auto_lock(_paMainloop);
+  paOperation = LATE(pa_context_get_sink_info_by_index)(
+      _paContext, device_index, PaSinkInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+  return true;
+}
+
+bool AudioMixerManagerLinuxPulse::GetSourceInfoByIndex(int device_index) const {
+  pa_operation* paOperation = NULL;
+
+  AutoPulseLock auto_lock(_paMainloop);
+  paOperation = LATE(pa_context_get_source_info_by_index)(
+      _paContext, device_index, PaSourceInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
new file mode 100644
index 0000000000..546440c4a6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
+
+#include <pulse/pulseaudio.h>
+#include <stdint.h>
+
+#include "api/sequence_checker.h"
+
+#ifndef UINT32_MAX
+#define UINT32_MAX ((uint32_t)-1)
+#endif
+
+namespace webrtc {
+
+class AudioMixerManagerLinuxPulse {
+ public:
+  int32_t SetPlayStream(pa_stream* playStream);
+  int32_t SetRecStream(pa_stream* recStream);
+  int32_t OpenSpeaker(uint16_t deviceIndex);
+  int32_t OpenMicrophone(uint16_t deviceIndex);
+  int32_t SetSpeakerVolume(uint32_t volume);
+  int32_t SpeakerVolume(uint32_t& volume) const;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  int32_t SpeakerVolumeIsAvailable(bool& available);
+  int32_t SpeakerMuteIsAvailable(bool& available);
+  int32_t SetSpeakerMute(bool enable);
+  int32_t StereoPlayoutIsAvailable(bool& available);
+  int32_t StereoRecordingIsAvailable(bool& available);
+  int32_t SpeakerMute(bool& enabled) const;
+  int32_t MicrophoneMuteIsAvailable(bool& available);
+  int32_t SetMicrophoneMute(bool enable);
+  int32_t MicrophoneMute(bool& enabled) const;
+  int32_t MicrophoneVolumeIsAvailable(bool& available);
+  int32_t SetMicrophoneVolume(uint32_t volume);
+  int32_t MicrophoneVolume(uint32_t& volume) const;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  int32_t SetPulseAudioObjects(pa_threaded_mainloop* mainloop,
+                               pa_context* context);
+  int32_t Close();
+  int32_t CloseSpeaker();
+  int32_t CloseMicrophone();
+  bool SpeakerIsInitialized() const;
+  bool MicrophoneIsInitialized() const;
+
+ public:
+  AudioMixerManagerLinuxPulse();
+  ~AudioMixerManagerLinuxPulse();
+
+ private:
+  static void PaSinkInfoCallback(pa_context* c,
+                                 const pa_sink_info* i,
+                                 int eol,
+                                 void* pThis);
+  static void PaSinkInputInfoCallback(pa_context* c,
+                                      const pa_sink_input_info* i,
+                                      int eol,
+                                      void* pThis);
+  static void PaSourceInfoCallback(pa_context* c,
+                                   const pa_source_info* i,
+                                   int eol,
+                                   void* pThis);
+  static void PaSetVolumeCallback(pa_context* /*c*/,
+                                  int success,
+                                  void* /*pThis*/);
+  void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
+  void PaSinkInputInfoCallbackHandler(const pa_sink_input_info* i, int eol);
+  void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
+
+  void WaitForOperationCompletion(pa_operation* paOperation) const;
+
+  bool GetSinkInputInfo() const;
+  bool GetSinkInfoByIndex(int device_index) const;
+  bool GetSourceInfoByIndex(int device_index) const;
+
+ private:
+  int16_t _paOutputDeviceIndex;
+  int16_t _paInputDeviceIndex;
+
+  pa_stream* _paPlayStream;
+  pa_stream* _paRecStream;
+
+  pa_threaded_mainloop* _paMainloop;
+  pa_context* _paContext;
+
+  mutable uint32_t _paVolume;
+  mutable uint32_t _paMute;
+  mutable uint32_t _paVolSteps;
+  bool _paSpeakerMute;
+  mutable uint32_t _paSpeakerVolume;
+  mutable uint8_t _paChannels;
+  bool _paObjectsSet;
+
+  // Stores thread ID in constructor.
+  // We can then use RTC_DCHECK_RUN_ON(&worker_thread_checker_) to ensure that
+  // other methods are called from the same thread.
+  // Currently only does RTC_DCHECK(thread_checker_.IsCurrent()).
+  SequenceChecker thread_checker_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc
new file mode 100644
index 0000000000..751edafd8b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2010 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/logging.h"
+
+#ifdef WEBRTC_LINUX
+#include <dlfcn.h>
+#endif
+
+namespace webrtc {
+namespace adm_linux {
+
+inline static const char* GetDllError() {
+#ifdef WEBRTC_LINUX
+  char* err = dlerror();
+  if (err) {
+    return err;
+  } else {
+    return "No error";
+  }
+#else
+#error Not implemented
+#endif
+}
+
+DllHandle InternalLoadDll(absl::string_view dll_name) {
+#ifdef WEBRTC_LINUX
+  DllHandle handle = dlopen(std::string(dll_name).c_str(), RTLD_NOW);
+#else
+#error Not implemented
+#endif
+  if (handle == kInvalidDllHandle) {
+    RTC_LOG(LS_WARNING) << "Can't load " << dll_name << " : " << GetDllError();
+  }
+  return handle;
+}
+
+void InternalUnloadDll(DllHandle handle) {
+#ifdef WEBRTC_LINUX
+// TODO(pbos): Remove this dlclose() exclusion when leaks and suppressions from
+// here are gone (or AddressSanitizer can display them properly).
+//
+// Skip dlclose() on AddressSanitizer as leaks including this module in the
+// stack trace gets displayed as <unknown module> instead of the actual library
+// -> it can not be suppressed.
+// https://code.google.com/p/address-sanitizer/issues/detail?id=89
+#if !defined(ADDRESS_SANITIZER)
+  if (dlclose(handle) != 0) {
+    RTC_LOG(LS_ERROR) << GetDllError();
+  }
+#endif  // !defined(ADDRESS_SANITIZER)
+#else
+#error Not implemented
+#endif
+}
+
+static bool LoadSymbol(DllHandle handle,
+                       absl::string_view symbol_name,
+                       void** symbol) {
+#ifdef WEBRTC_LINUX
+  *symbol = dlsym(handle, std::string(symbol_name).c_str());
+  char* err = dlerror();
+  if (err) {
+    RTC_LOG(LS_ERROR) << "Error loading symbol " << symbol_name << " : " << err;
+    return false;
+  } else if (!*symbol) {
+    RTC_LOG(LS_ERROR) << "Symbol " << symbol_name << " is NULL";
+    return false;
+  }
+  return true;
+#else
+#error Not implemented
+#endif
+}
+
+// This routine MUST assign SOME value for every symbol, even if that value is
+// NULL, or else some symbols may be left with uninitialized data that the
+// caller may later interpret as a valid address.
+bool InternalLoadSymbols(DllHandle handle,
+                         int num_symbols,
+                         const char* const symbol_names[],
+                         void* symbols[]) {
+#ifdef WEBRTC_LINUX
+  // Clear any old errors.
+  dlerror();
+#endif
+  for (int i = 0; i < num_symbols; ++i) {
+    if (!LoadSymbol(handle, symbol_names[i], &symbols[i])) {
+      return false;
+    }
+  }
+  return true;
+}
+
+}  // namespace adm_linux
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.h b/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
new file mode 100644
index 0000000000..00f3c5a449
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2010 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H_
+#define AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H_
+
+#include <stddef.h>  // for NULL
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+// This file provides macros for creating "symbol table" classes to simplify the
+// dynamic loading of symbols from DLLs. Currently the implementation only
+// supports Linux and pure C symbols.
+// See talk/sound/pulseaudiosymboltable.(h|cc) for an example.
+
+namespace webrtc {
+namespace adm_linux {
+
+#ifdef WEBRTC_LINUX
+typedef void* DllHandle;
+
+const DllHandle kInvalidDllHandle = NULL;
+#else
+#error Not implemented
+#endif
+
+// These are helpers for use only by the class below.
+DllHandle InternalLoadDll(absl::string_view);
+
+void InternalUnloadDll(DllHandle handle);
+
+bool InternalLoadSymbols(DllHandle handle,
+                         int num_symbols,
+                         const char* const symbol_names[],
+                         void* symbols[]);
+
+template <int SYMBOL_TABLE_SIZE,
+          const char kDllName[],
+          const char* const kSymbolNames[]>
+class LateBindingSymbolTable {
+ public:
+  LateBindingSymbolTable()
+      : handle_(kInvalidDllHandle), undefined_symbols_(false) {
+    memset(symbols_, 0, sizeof(symbols_));
+  }
+
+  ~LateBindingSymbolTable() { Unload(); }
+
+  LateBindingSymbolTable(const LateBindingSymbolTable&) = delete;
+  LateBindingSymbolTable& operator=(LateBindingSymbolTable&) = delete;
+
+  static int NumSymbols() { return SYMBOL_TABLE_SIZE; }
+
+  // We do not use this, but we offer it for theoretical convenience.
+  static const char* GetSymbolName(int index) {
+    RTC_DCHECK_LT(index, NumSymbols());
+    return kSymbolNames[index];
+  }
+
+  bool IsLoaded() const { return handle_ != kInvalidDllHandle; }
+
+  // Loads the DLL and the symbol table. Returns true iff the DLL and symbol
+  // table loaded successfully.
+  bool Load() {
+    if (IsLoaded()) {
+      return true;
+    }
+    if (undefined_symbols_) {
+      // We do not attempt to load again because repeated attempts are not
+      // likely to succeed and DLL loading is costly.
+      return false;
+    }
+    handle_ = InternalLoadDll(kDllName);
+    if (!IsLoaded()) {
+      return false;
+    }
+    if (!InternalLoadSymbols(handle_, NumSymbols(), kSymbolNames, symbols_)) {
+      undefined_symbols_ = true;
+      Unload();
+      return false;
+    }
+    return true;
+  }
+
+  void Unload() {
+    if (!IsLoaded()) {
+      return;
+    }
+    InternalUnloadDll(handle_);
+    handle_ = kInvalidDllHandle;
+    memset(symbols_, 0, sizeof(symbols_));
+  }
+
+  // Retrieves the given symbol. NOTE: Recommended to use LATESYM_GET below
+  // instead of this.
+  void* GetSymbol(int index) const {
+    RTC_DCHECK(IsLoaded());
+    RTC_DCHECK_LT(index, NumSymbols());
+    return symbols_[index];
+  }
+
+ private:
+  DllHandle handle_;
+  bool undefined_symbols_;
+  void* symbols_[SYMBOL_TABLE_SIZE];
+};
+
+// This macro must be invoked in a header to declare a symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(ClassName) enum {
+// This macro must be invoked in the header declaration once for each symbol
+// (recommended to use an X-Macro to avoid duplication).
+// This macro defines an enum with names built from the symbols, which
+// essentially creates a hash table in the compiler from symbol names to their
+// indices in the symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(ClassName, sym) \
+  ClassName##_SYMBOL_TABLE_INDEX_##sym,
+
+// This macro completes the header declaration.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_END(ClassName)       \
+  ClassName##_SYMBOL_TABLE_SIZE                                \
+  }                                                            \
+  ;                                                            \
+                                                               \
+  extern const char ClassName##_kDllName[];                    \
+  extern const char* const                                     \
+      ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE]; \
+                                                               \
+  typedef ::webrtc::adm_linux::LateBindingSymbolTable<         \
+      ClassName##_SYMBOL_TABLE_SIZE, ClassName##_kDllName,     \
+      ClassName##_kSymbolNames>                                \
+      ClassName;
+
+// This macro must be invoked in a .cc file to define a previously-declared
+// symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(ClassName, dllName) \
+  const char ClassName##_kDllName[] = dllName;                     \
+  const char* const ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE] = {
+// This macro must be invoked in the .cc definition once for each symbol
+// (recommended to use an X-Macro to avoid duplication).
+// This would have to use the mangled name if we were to ever support C++
+// symbols.
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(ClassName, sym) #sym,
+
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_END(ClassName) \
+  }                                                     \
+  ;
+
+// Index of a given symbol in the given symbol table class.
+#define LATESYM_INDEXOF(ClassName, sym) (ClassName##_SYMBOL_TABLE_INDEX_##sym)
+
+// Returns a reference to the given late-binded symbol, with the correct type.
+#define LATESYM_GET(ClassName, inst, sym) \
+  (*reinterpret_cast<__typeof__(&sym)>(   \
+      (inst)->GetSymbol(LATESYM_INDEXOF(ClassName, sym))))
+
+}  // namespace adm_linux
+}  // namespace webrtc
+
+#endif  // ADM_LATEBINDINGSYMBOLTABLE_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc b/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc
new file mode 100644
index 0000000000..e0759e6ca3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc
@@ -0,0 +1,41 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_pulse {
+
+LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so.0")
+#define X(sym) \
+  LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(PulseAudioSymbolTable, sym)
+PULSE_AUDIO_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DEFINE_END(PulseAudioSymbolTable)
+
+}  // namespace adm_linux_pulse
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h b/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h
new file mode 100644
index 0000000000..2f6a9510d8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h
@@ -0,0 +1,106 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H_
+#define AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H_
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_pulse {
+
+// The PulseAudio symbols we need, as an X-Macro list.
+// This list must contain precisely every libpulse function that is used in
+// the ADM LINUX PULSE Device and Mixer classes
+#define PULSE_AUDIO_SYMBOLS_LIST           \
+  X(pa_bytes_per_second)                   \
+  X(pa_context_connect)                    \
+  X(pa_context_disconnect)                 \
+  X(pa_context_errno)                      \
+  X(pa_context_get_protocol_version)       \
+  X(pa_context_get_server_info)            \
+  X(pa_context_get_sink_info_list)         \
+  X(pa_context_get_sink_info_by_index)     \
+  X(pa_context_get_sink_info_by_name)      \
+  X(pa_context_get_sink_input_info)        \
+  X(pa_context_get_source_info_by_index)   \
+  X(pa_context_get_source_info_by_name)    \
+  X(pa_context_get_source_info_list)       \
+  X(pa_context_get_state)                  \
+  X(pa_context_new)                        \
+  X(pa_context_set_sink_input_volume)      \
+  X(pa_context_set_sink_input_mute)        \
+  X(pa_context_set_source_volume_by_index) \
+  X(pa_context_set_source_mute_by_index)   \
+  X(pa_context_set_state_callback)         \
+  X(pa_context_unref)                      \
+  X(pa_cvolume_set)                        \
+  X(pa_operation_get_state)                \
+  X(pa_operation_unref)                    \
+  X(pa_stream_connect_playback)            \
+  X(pa_stream_connect_record)              \
+  X(pa_stream_disconnect)                  \
+  X(pa_stream_drop)                        \
+  X(pa_stream_get_device_index)            \
+  X(pa_stream_get_index)                   \
+  X(pa_stream_get_latency)                 \
+  X(pa_stream_get_sample_spec)             \
+  X(pa_stream_get_state)                   \
+  X(pa_stream_new)                         \
+  X(pa_stream_peek)                        \
+  X(pa_stream_readable_size)               \
+  X(pa_stream_set_buffer_attr)             \
+  X(pa_stream_set_overflow_callback)       \
+  X(pa_stream_set_read_callback)           \
+  X(pa_stream_set_state_callback)          \
+  X(pa_stream_set_underflow_callback)      \
+  X(pa_stream_set_write_callback)          \
+  X(pa_stream_unref)                       \
+  X(pa_stream_writable_size)               \
+  X(pa_stream_write)                       \
+  X(pa_strerror)                           \
+  X(pa_threaded_mainloop_free)             \
+  X(pa_threaded_mainloop_get_api)          \
+  X(pa_threaded_mainloop_lock)             \
+  X(pa_threaded_mainloop_new)              \
+  X(pa_threaded_mainloop_signal)           \
+  X(pa_threaded_mainloop_start)            \
+  X(pa_threaded_mainloop_stop)             \
+  X(pa_threaded_mainloop_unlock)           \
+  X(pa_threaded_mainloop_wait)
+
+LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(PulseAudioSymbolTable)
+#define X(sym) \
+  LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(PulseAudioSymbolTable, sym)
+PULSE_AUDIO_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DECLARE_END(PulseAudioSymbolTable)
+
+}  // namespace adm_linux_pulse
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H_
diff --git a/third_party/libwebrtc/modules/audio_device/mac/audio_device_mac.cc b/third_party/libwebrtc/modules/audio_device/mac/audio_device_mac.cc
new file mode 100644
index 0000000000..527f76a371
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/mac/audio_device_mac.cc
@@ -0,0 +1,2500 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/mac/audio_device_mac.h"
+
+#include <ApplicationServices/ApplicationServices.h>
+#include <mach/mach.h>         // mach_task_self()
+#include <sys/sysctl.h>        // sysctlbyname()
+
+#include <memory>
+
+#include "modules/audio_device/audio_device_config.h"
+#include "modules/third_party/portaudio/pa_ringbuffer.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+
+#define WEBRTC_CA_RETURN_ON_ERR(expr)                                \
+  do {                                                               \
+    err = expr;                                                      \
+    if (err != noErr) {                                              \
+      logCAMsg(rtc::LS_ERROR, "Error in " #expr, (const char*)&err); \
+      return -1;                                                     \
+    }                                                                \
+  } while (0)
+
+#define WEBRTC_CA_LOG_ERR(expr)                                      \
+  do {                                                               \
+    err = expr;                                                      \
+    if (err != noErr) {                                              \
+      logCAMsg(rtc::LS_ERROR, "Error in " #expr, (const char*)&err); \
+    }                                                                \
+  } while (0)
+
+#define WEBRTC_CA_LOG_WARN(expr)                                       \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(rtc::LS_WARNING, "Error in " #expr, (const char*)&err); \
+    }                                                                  \
+  } while (0)
+
+enum { MaxNumberDevices = 64 };
+
+// CoreAudio errors are best interpreted as four character strings.
+void AudioDeviceMac::logCAMsg(const rtc::LoggingSeverity sev,
+                              const char* msg,
+                              const char* err) {
+  RTC_DCHECK(msg != NULL);
+  RTC_DCHECK(err != NULL);
+
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
+  switch (sev) {
+    case rtc::LS_ERROR:
+      RTC_LOG(LS_ERROR) << msg << ": " << err[0] << err[1] << err[2] << err[3];
+      break;
+    case rtc::LS_WARNING:
+      RTC_LOG(LS_WARNING) << msg << ": " << err[0] << err[1] << err[2]
+                          << err[3];
+      break;
+    case rtc::LS_VERBOSE:
+      RTC_LOG(LS_VERBOSE) << msg << ": " << err[0] << err[1] << err[2]
+                          << err[3];
+      break;
+    default:
+      break;
+  }
+#else
+  // We need to flip the characters in this case.
+  switch (sev) {
+    case rtc::LS_ERROR:
+      RTC_LOG(LS_ERROR) << msg << ": " << err[3] << err[2] << err[1] << err[0];
+      break;
+    case rtc::LS_WARNING:
+      RTC_LOG(LS_WARNING) << msg << ": " << err[3] << err[2] << err[1]
+                          << err[0];
+      break;
+    case rtc::LS_VERBOSE:
+      RTC_LOG(LS_VERBOSE) << msg << ": " << err[3] << err[2] << err[1]
+                          << err[0];
+      break;
+    default:
+      break;
+  }
+#endif
+}
+
+AudioDeviceMac::AudioDeviceMac()
+    : _ptrAudioBuffer(NULL),
+      _mixerManager(),
+      _inputDeviceIndex(0),
+      _outputDeviceIndex(0),
+      _inputDeviceID(kAudioObjectUnknown),
+      _outputDeviceID(kAudioObjectUnknown),
+      _inputDeviceIsSpecified(false),
+      _outputDeviceIsSpecified(false),
+      _recChannels(N_REC_CHANNELS),
+      _playChannels(N_PLAY_CHANNELS),
+      _captureBufData(NULL),
+      _renderBufData(NULL),
+      _initialized(false),
+      _isShutDown(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _renderDeviceIsAlive(1),
+      _captureDeviceIsAlive(1),
+      _twoDevices(true),
+      _doStop(false),
+      _doStopRec(false),
+      _macBookPro(false),
+      _macBookProPanRight(false),
+      _captureLatencyUs(0),
+      _renderLatencyUs(0),
+      _captureDelayUs(0),
+      _renderDelayUs(0),
+      _renderDelayOffsetSamples(0),
+      _paCaptureBuffer(NULL),
+      _paRenderBuffer(NULL),
+      _captureBufSizeSamples(0),
+      _renderBufSizeSamples(0),
+      prev_key_state_() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+
+  memset(_renderConvertData, 0, sizeof(_renderConvertData));
+  memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+}
+
+AudioDeviceMac::~AudioDeviceMac() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+
+  if (!_isShutDown) {
+    Terminate();
+  }
+
+  RTC_DCHECK(capture_worker_thread_.empty());
+  RTC_DCHECK(render_worker_thread_.empty());
+
+  if (_paRenderBuffer) {
+    delete _paRenderBuffer;
+    _paRenderBuffer = NULL;
+  }
+
+  if (_paCaptureBuffer) {
+    delete _paCaptureBuffer;
+    _paCaptureBuffer = NULL;
+  }
+
+  if (_renderBufData) {
+    delete[] _renderBufData;
+    _renderBufData = NULL;
+  }
+
+  if (_captureBufData) {
+    delete[] _captureBufData;
+    _captureBufData = NULL;
+  }
+
+  kern_return_t kernErr = KERN_SUCCESS;
+  kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "semaphore_destroy() error: " << kernErr;
+  }
+
+  kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "semaphore_destroy() error: " << kernErr;
+  }
+}
+
+// ============================================================================
+//                                     API
+// ============================================================================
+
+void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  MutexLock lock(&mutex_);
+
+  _ptrAudioBuffer = audioBuffer;
+
+  // inform the AudioBuffer about default settings for this implementation
+  _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+  _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+  _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
+  _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
+}
+
+int32_t AudioDeviceMac::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+  return 0;
+}
+
+AudioDeviceGeneric::InitStatus AudioDeviceMac::Init() {
+  MutexLock lock(&mutex_);
+
+  if (_initialized) {
+    return InitStatus::OK;
+  }
+
+  OSStatus err = noErr;
+
+  _isShutDown = false;
+
+  // PortAudio ring buffers require an elementCount which is a power of two.
+  if (_renderBufData == NULL) {
+    UInt32 powerOfTwo = 1;
+    while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES) {
+      powerOfTwo <<= 1;
+    }
+    _renderBufSizeSamples = powerOfTwo;
+    _renderBufData = new SInt16[_renderBufSizeSamples];
+  }
+
+  if (_paRenderBuffer == NULL) {
+    _paRenderBuffer = new PaUtilRingBuffer;
+    ring_buffer_size_t bufSize = -1;
+    bufSize = PaUtil_InitializeRingBuffer(
+        _paRenderBuffer, sizeof(SInt16), _renderBufSizeSamples, _renderBufData);
+    if (bufSize == -1) {
+      RTC_LOG(LS_ERROR) << "PaUtil_InitializeRingBuffer() error";
+      return InitStatus::PLAYOUT_ERROR;
+    }
+  }
+
+  if (_captureBufData == NULL) {
+    UInt32 powerOfTwo = 1;
+    while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES) {
+      powerOfTwo <<= 1;
+    }
+    _captureBufSizeSamples = powerOfTwo;
+    _captureBufData = new Float32[_captureBufSizeSamples];
+  }
+
+  if (_paCaptureBuffer == NULL) {
+    _paCaptureBuffer = new PaUtilRingBuffer;
+    ring_buffer_size_t bufSize = -1;
+    bufSize =
+        PaUtil_InitializeRingBuffer(_paCaptureBuffer, sizeof(Float32),
+                                    _captureBufSizeSamples, _captureBufData);
+    if (bufSize == -1) {
+      RTC_LOG(LS_ERROR) << "PaUtil_InitializeRingBuffer() error";
+      return InitStatus::RECORDING_ERROR;
+    }
+  }
+
+  kern_return_t kernErr = KERN_SUCCESS;
+  kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
+                             SYNC_POLICY_FIFO, 0);
+  if (kernErr != KERN_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "semaphore_create() error: " << kernErr;
+    return InitStatus::OTHER_ERROR;
+  }
+
+  kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
+                             SYNC_POLICY_FIFO, 0);
+  if (kernErr != KERN_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "semaphore_create() error: " << kernErr;
+    return InitStatus::OTHER_ERROR;
+  }
+
+  // Setting RunLoop to NULL here instructs HAL to manage its own thread for
+  // notifications. This was the default behaviour on OS X 10.5 and earlier,
+  // but now must be explicitly specified. HAL would otherwise try to use the
+  // main thread to issue notifications.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyRunLoop, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  CFRunLoopRef runLoop = NULL;
+  UInt32 size = sizeof(CFRunLoopRef);
+  int aoerr = AudioObjectSetPropertyData(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, size, &runLoop);
+  if (aoerr != noErr) {
+    RTC_LOG(LS_ERROR) << "Error in AudioObjectSetPropertyData: "
+                      << (const char*)&aoerr;
+    return InitStatus::OTHER_ERROR;
+  }
+
+  // Listen for any device changes.
+  propertyAddress.mSelector = kAudioHardwarePropertyDevices;
+  WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(
+      kAudioObjectSystemObject, &propertyAddress, &objectListenerProc, this));
+
+  // Determine if this is a MacBook Pro
+  _macBookPro = false;
+  _macBookProPanRight = false;
+  char buf[128];
+  size_t length = sizeof(buf);
+  memset(buf, 0, length);
+
+  int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
+  if (intErr != 0) {
+    RTC_LOG(LS_ERROR) << "Error in sysctlbyname(): " << err;
+  } else {
+    RTC_LOG(LS_VERBOSE) << "Hardware model: " << buf;
+    if (strncmp(buf, "MacBookPro", 10) == 0) {
+      _macBookPro = true;
+    }
+  }
+
+  _initialized = true;
+
+  return InitStatus::OK;
+}
+
+int32_t AudioDeviceMac::Terminate() {
+  if (!_initialized) {
+    return 0;
+  }
+
+  if (_recording) {
+    RTC_LOG(LS_ERROR) << "Recording must be stopped";
+    return -1;
+  }
+
+  if (_playing) {
+    RTC_LOG(LS_ERROR) << "Playback must be stopped";
+    return -1;
+  }
+
+  MutexLock lock(&mutex_);
+  _mixerManager.Close();
+
+  OSStatus err = noErr;
+  int retVal = 0;
+
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      kAudioObjectSystemObject, &propertyAddress, &objectListenerProc, this));
+
+  err = AudioHardwareUnload();
+  if (err != noErr) {
+    logCAMsg(rtc::LS_ERROR, "Error in AudioHardwareUnload()",
+             (const char*)&err);
+    retVal = -1;
+  }
+
+  _isShutDown = true;
+  _initialized = false;
+  _outputDeviceIsSpecified = false;
+  _inputDeviceIsSpecified = false;
+
+  return retVal;
+}
+
+bool AudioDeviceMac::Initialized() const {
+  return (_initialized);
+}
+
+int32_t AudioDeviceMac::SpeakerIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+  return SpeakerIsAvailableLocked(available);
+}
+
+int32_t AudioDeviceMac::SpeakerIsAvailableLocked(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeakerLocked() == -1) {
+    available = false;
+    return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know that a valid speaker
+  // exists.
+  available = true;
+
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitSpeaker() {
+  MutexLock lock(&mutex_);
+  return InitSpeakerLocked();
+}
+
+int32_t AudioDeviceMac::InitSpeakerLocked() {
+  if (_playing) {
+    return -1;
+  }
+
+  if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1) {
+    return -1;
+  }
+
+  if (_inputDeviceID == _outputDeviceID) {
+    _twoDevices = false;
+  } else {
+    _twoDevices = true;
+  }
+
+  if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1) {
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+  return MicrophoneIsAvailableLocked(available);
+}
+
+int32_t AudioDeviceMac::MicrophoneIsAvailableLocked(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitMicrophoneLocked() == -1) {
+    available = false;
+    return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a valid microphone
+  // exists.
+  available = true;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitMicrophone() {
+  MutexLock lock(&mutex_);
+  return InitMicrophoneLocked();
+}
+
+int32_t AudioDeviceMac::InitMicrophoneLocked() {
+  if (_recording) {
+    return -1;
+  }
+
+  if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1) {
+    return -1;
+  }
+
+  if (_inputDeviceID == _outputDeviceID) {
+    _twoDevices = false;
+  } else {
+    _twoDevices = true;
+  }
+
+  if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1) {
+    return -1;
+  }
+
+  return 0;
+}
+
+bool AudioDeviceMac::SpeakerIsInitialized() const {
+  return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceMac::MicrophoneIsInitialized() const {
+  return (_mixerManager.MicrophoneIsInitialized());
+}
+
+int32_t AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know that a volume control exists
+  //
+  available = true;
+
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetSpeakerVolume(uint32_t volume) {
+  return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+int32_t AudioDeviceMac::SpeakerVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.SpeakerVolume(level) == -1) {
+    return -1;
+  }
+
+  volume = level;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MinSpeakerVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinSpeakerVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+  return 0;
+}
+
+int32_t AudioDeviceMac::SpeakerMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected speaker has a mute control
+  //
+  _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+  available = isAvailable;
+
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetSpeakerMute(bool enable) {
+  return (_mixerManager.SetSpeakerMute(enable));
+}
+
+int32_t AudioDeviceMac::SpeakerMute(bool& enabled) const {
+  bool muted(0);
+
+  if (_mixerManager.SpeakerMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control, hence it is safe to state that there is no boost control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone has a mute control
+  //
+  _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetMicrophoneMute(bool enable) {
+  return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+int32_t AudioDeviceMac::MicrophoneMute(bool& enabled) const {
+  bool muted(0);
+
+  if (_mixerManager.MicrophoneMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceMac::StereoRecordingIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // Cannot open the specified device
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone can record stereo
+  //
+  _mixerManager.StereoRecordingIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetStereoRecording(bool enable) {
+  if (enable)
+    _recChannels = 2;
+  else
+    _recChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StereoRecording(bool& enabled) const {
+  if (_recChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StereoPlayoutIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // Cannot open the specified device
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone can record stereo
+  //
+  _mixerManager.StereoPlayoutIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetStereoPlayout(bool enable) {
+  if (enable)
+    _playChannels = 2;
+  else
+    _playChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StereoPlayout(bool& enabled) const {
+  if (_playChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a volume control
+  // exists
+  //
+  available = true;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetMicrophoneVolume(uint32_t volume) {
+  return (_mixerManager.SetMicrophoneVolume(volume));
+}
+
+int32_t AudioDeviceMac::MicrophoneVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.MicrophoneVolume(level) == -1) {
+    RTC_LOG(LS_WARNING) << "failed to retrieve current microphone level";
+    return -1;
+  }
+
+  volume = level;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MinMicrophoneVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+  return 0;
+}
+
+int16_t AudioDeviceMac::PlayoutDevices() {
+  AudioDeviceID playDevices[MaxNumberDevices];
+  return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
+                          MaxNumberDevices);
+}
+
+int32_t AudioDeviceMac::SetPlayoutDevice(uint16_t index) {
+  MutexLock lock(&mutex_);
+
+  if (_playIsInitialized) {
+    return -1;
+  }
+
+  AudioDeviceID playDevices[MaxNumberDevices];
+  uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
+                                       playDevices, MaxNumberDevices);
+  RTC_LOG(LS_VERBOSE) << "number of available waveform-audio output devices is "
+                      << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _outputDeviceIndex = index;
+  _outputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceMac::PlayoutDeviceName(uint16_t index,
+                                          char name[kAdmMaxDeviceNameSize],
+                                          char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(PlayoutDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDeviceName(kAudioDevicePropertyScopeOutput, index,
+                       rtc::ArrayView<char>(name, kAdmMaxDeviceNameSize));
+}
+
+int32_t AudioDeviceMac::RecordingDeviceName(uint16_t index,
+                                            char name[kAdmMaxDeviceNameSize],
+                                            char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(RecordingDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDeviceName(kAudioDevicePropertyScopeInput, index,
+                       rtc::ArrayView<char>(name, kAdmMaxDeviceNameSize));
+}
+
+int16_t AudioDeviceMac::RecordingDevices() {
+  AudioDeviceID recDevices[MaxNumberDevices];
+  return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
+                          MaxNumberDevices);
+}
+
+int32_t AudioDeviceMac::SetRecordingDevice(uint16_t index) {
+  if (_recIsInitialized) {
+    return -1;
+  }
+
+  AudioDeviceID recDevices[MaxNumberDevices];
+  uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
+                                       recDevices, MaxNumberDevices);
+  RTC_LOG(LS_VERBOSE) << "number of available waveform-audio input devices is "
+                      << nDevices;
+
+  if (index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  _inputDeviceIndex = index;
+  _inputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  RTC_LOG(LS_ERROR) << "WindowsDeviceType not supported";
+  return -1;
+}
+
+int32_t AudioDeviceMac::PlayoutIsAvailable(bool& available) {
+  available = true;
+
+  // Try to initialize the playout side
+  if (InitPlayout() == -1) {
+    available = false;
+  }
+
+  // We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
+  // We must actually start playout here in order to have the IOProc
+  // deleted by calling StopPlayout().
+  if (StartPlayout() == -1) {
+    available = false;
+  }
+
+  // Cancel effect of initialization
+  if (StopPlayout() == -1) {
+    available = false;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::RecordingIsAvailable(bool& available) {
+  available = true;
+
+  // Try to initialize the recording side
+  if (InitRecording() == -1) {
+    available = false;
+  }
+
+  // We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
+  // We must actually start recording here in order to have the IOProc
+  // deleted by calling StopRecording().
+  if (StartRecording() == -1) {
+    available = false;
+  }
+
+  // Cancel effect of initialization
+  if (StopRecording() == -1) {
+    available = false;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitPlayout() {
+  RTC_LOG(LS_INFO) << "InitPlayout";
+  MutexLock lock(&mutex_);
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_playIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the speaker (devices might have been added or removed)
+  if (InitSpeakerLocked() == -1) {
+    RTC_LOG(LS_WARNING) << "InitSpeaker() failed";
+  }
+
+  if (!MicrophoneIsInitialized()) {
+    // Make this call to check if we are using
+    // one or two devices (_twoDevices)
+    bool available = false;
+    if (MicrophoneIsAvailableLocked(available) == -1) {
+      RTC_LOG(LS_WARNING) << "MicrophoneIsAvailable() failed";
+    }
+  }
+
+  PaUtil_FlushRingBuffer(_paRenderBuffer);
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  _renderDelayOffsetSamples = 0;
+  _renderDelayUs = 0;
+  _renderLatencyUs = 0;
+  _renderDeviceIsAlive = 1;
+  _doStop = false;
+
+  // The internal microphone of a MacBook Pro is located under the left speaker
+  // grille. When the internal speakers are in use, we want to fully stereo
+  // pan to the right.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyDataSource, kAudioDevicePropertyScopeOutput, 0};
+  if (_macBookPro) {
+    _macBookProPanRight = false;
+    Boolean hasProperty =
+        AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+    if (hasProperty) {
+      UInt32 dataSource = 0;
+      size = sizeof(dataSource);
+      WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, &size, &dataSource));
+
+      if (dataSource == 'ispk') {
+        _macBookProPanRight = true;
+        RTC_LOG(LS_VERBOSE)
+            << "MacBook Pro using internal speakers; stereo panning right";
+      } else {
+        RTC_LOG(LS_VERBOSE) << "MacBook Pro not using internal speakers";
+      }
+
+      // Add a listener to determine if the status changes.
+      WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+          _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+    }
+  }
+
+  // Get current stream description
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
+  size = sizeof(_outStreamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &_outStreamFormat));
+
+  if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(rtc::LS_ERROR, "Unacceptable output stream format -> mFormatID",
+             (const char*)&_outStreamFormat.mFormatID);
+    return -1;
+  }
+
+  if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    RTC_LOG(LS_ERROR)
+        << "Too many channels on output device (mChannelsPerFrame = "
+        << _outStreamFormat.mChannelsPerFrame << ")";
+    return -1;
+  }
+
+  if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) {
+    RTC_LOG(LS_ERROR) << "Non-interleaved audio data is not supported."
+                         "AudioHardware streams should not have this format.";
+    return -1;
+  }
+
+  RTC_LOG(LS_VERBOSE) << "Ouput stream format:";
+  RTC_LOG(LS_VERBOSE) << "mSampleRate = " << _outStreamFormat.mSampleRate
+                      << ", mChannelsPerFrame = "
+                      << _outStreamFormat.mChannelsPerFrame;
+  RTC_LOG(LS_VERBOSE) << "mBytesPerPacket = "
+                      << _outStreamFormat.mBytesPerPacket
+                      << ", mFramesPerPacket = "
+                      << _outStreamFormat.mFramesPerPacket;
+  RTC_LOG(LS_VERBOSE) << "mBytesPerFrame = " << _outStreamFormat.mBytesPerFrame
+                      << ", mBitsPerChannel = "
+                      << _outStreamFormat.mBitsPerChannel;
+  RTC_LOG(LS_VERBOSE) << "mFormatFlags = " << _outStreamFormat.mFormatFlags;
+  logCAMsg(rtc::LS_VERBOSE, "mFormatID",
+           (const char*)&_outStreamFormat.mFormatID);
+
+  // Our preferred format to work with.
+  if (_outStreamFormat.mChannelsPerFrame < 2) {
+    // Disable stereo playout when we only have one channel on the device.
+    _playChannels = 1;
+    RTC_LOG(LS_VERBOSE) << "Stereo playout unavailable on this device";
+  }
+  WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
+
+  // Listen for format changes.
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  // Listen for processor overloads.
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  if (_twoDevices || !_recIsInitialized) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _outputDeviceID, deviceIOProc, this, &_deviceIOProcID));
+  }
+
+  _playIsInitialized = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitRecording() {
+  RTC_LOG(LS_INFO) << "InitRecording";
+  MutexLock lock(&mutex_);
+
+  if (_recording) {
+    return -1;
+  }
+
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_recIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the microphone (devices might have been added or removed)
+  if (InitMicrophoneLocked() == -1) {
+    RTC_LOG(LS_WARNING) << "InitMicrophone() failed";
+  }
+
+  if (!SpeakerIsInitialized()) {
+    // Make this call to check if we are using
+    // one or two devices (_twoDevices)
+    bool available = false;
+    if (SpeakerIsAvailableLocked(available) == -1) {
+      RTC_LOG(LS_WARNING) << "SpeakerIsAvailable() failed";
+    }
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+
+  PaUtil_FlushRingBuffer(_paCaptureBuffer);
+
+  _captureDelayUs = 0;
+  _captureLatencyUs = 0;
+  _captureDeviceIsAlive = 1;
+  _doStopRec = false;
+
+  // Get current stream description
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeInput, 0};
+  memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
+  size = sizeof(_inStreamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &_inStreamFormat));
+
+  if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(rtc::LS_ERROR, "Unacceptable input stream format -> mFormatID",
+             (const char*)&_inStreamFormat.mFormatID);
+    return -1;
+  }
+
+  if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    RTC_LOG(LS_ERROR)
+        << "Too many channels on input device (mChannelsPerFrame = "
+        << _inStreamFormat.mChannelsPerFrame << ")";
+    return -1;
+  }
+
+  const int io_block_size_samples = _inStreamFormat.mChannelsPerFrame *
+                                    _inStreamFormat.mSampleRate / 100 *
+                                    N_BLOCKS_IO;
+  if (io_block_size_samples > _captureBufSizeSamples) {
+    RTC_LOG(LS_ERROR) << "Input IO block size (" << io_block_size_samples
+                      << ") is larger than ring buffer ("
+                      << _captureBufSizeSamples << ")";
+    return -1;
+  }
+
+  RTC_LOG(LS_VERBOSE) << "Input stream format:";
+  RTC_LOG(LS_VERBOSE) << "mSampleRate = " << _inStreamFormat.mSampleRate
+                      << ", mChannelsPerFrame = "
+                      << _inStreamFormat.mChannelsPerFrame;
+  RTC_LOG(LS_VERBOSE) << "mBytesPerPacket = " << _inStreamFormat.mBytesPerPacket
+                      << ", mFramesPerPacket = "
+                      << _inStreamFormat.mFramesPerPacket;
+  RTC_LOG(LS_VERBOSE) << "mBytesPerFrame = " << _inStreamFormat.mBytesPerFrame
+                      << ", mBitsPerChannel = "
+                      << _inStreamFormat.mBitsPerChannel;
+  RTC_LOG(LS_VERBOSE) << "mFormatFlags = " << _inStreamFormat.mFormatFlags;
+  logCAMsg(rtc::LS_VERBOSE, "mFormatID",
+           (const char*)&_inStreamFormat.mFormatID);
+
+  // Our preferred format to work with
+  if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2)) {
+    _inDesiredFormat.mChannelsPerFrame = 2;
+  } else {
+    // Disable stereo recording when we only have one channel on the device.
+    _inDesiredFormat.mChannelsPerFrame = 1;
+    _recChannels = 1;
+    RTC_LOG(LS_VERBOSE) << "Stereo recording unavailable on this device";
+  }
+
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters
+    _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+    _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+  }
+
+  _inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
+  _inDesiredFormat.mBytesPerPacket =
+      _inDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _inDesiredFormat.mFramesPerPacket = 1;
+  _inDesiredFormat.mBytesPerFrame =
+      _inDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
+
+  _inDesiredFormat.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
+  _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+  _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+
+  WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
+                                            &_captureConverter));
+
+  // First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
+  // TODO(xians): investigate this block.
+  UInt32 bufByteCount =
+      (UInt32)((_inStreamFormat.mSampleRate / 1000.0) * 10.0 * N_BLOCKS_IO *
+               _inStreamFormat.mChannelsPerFrame * sizeof(Float32));
+  if (_inStreamFormat.mFramesPerPacket != 0) {
+    if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0) {
+      bufByteCount =
+          ((UInt32)(bufByteCount / _inStreamFormat.mFramesPerPacket) + 1) *
+          _inStreamFormat.mFramesPerPacket;
+    }
+  }
+
+  // Ensure the buffer size is within the acceptable range provided by the
+  // device.
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+  AudioValueRange range;
+  size = sizeof(range);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &range));
+  if (range.mMinimum > bufByteCount) {
+    bufByteCount = range.mMinimum;
+  } else if (range.mMaximum < bufByteCount) {
+    bufByteCount = range.mMaximum;
+  }
+
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+  size = sizeof(bufByteCount);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, size, &bufByteCount));
+
+  // Get capture device latency
+  propertyAddress.mSelector = kAudioDevicePropertyLatency;
+  UInt32 latency = 0;
+  size = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _captureLatencyUs = (UInt32)((1.0e6 * latency) / _inStreamFormat.mSampleRate);
+
+  // Get capture stream latency
+  propertyAddress.mSelector = kAudioDevicePropertyStreams;
+  AudioStreamID stream = 0;
+  size = sizeof(AudioStreamID);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &stream));
+  propertyAddress.mSelector = kAudioStreamPropertyLatency;
+  size = sizeof(UInt32);
+  latency = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _captureLatencyUs +=
+      (UInt32)((1.0e6 * latency) / _inStreamFormat.mSampleRate);
+
+  // Listen for format changes
+  // TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  // Listen for processor overloads
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  if (_twoDevices) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _inputDeviceID, inDeviceIOProc, this, &_inDeviceIOProcID));
+  } else if (!_playIsInitialized) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _inputDeviceID, deviceIOProc, this, &_deviceIOProcID));
+  }
+
+  // Mark recording side as initialized
+  _recIsInitialized = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StartRecording() {
+  RTC_LOG(LS_INFO) << "StartRecording";
+  MutexLock lock(&mutex_);
+
+  if (!_recIsInitialized) {
+    return -1;
+  }
+
+  if (_recording) {
+    return 0;
+  }
+
+  if (!_initialized) {
+    RTC_LOG(LS_ERROR) << "Recording worker thread has not been started";
+    return -1;
+  }
+
+  RTC_DCHECK(capture_worker_thread_.empty());
+  capture_worker_thread_ = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (CaptureWorkerThread()) {
+        }
+      },
+      "CaptureWorkerThread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  OSStatus err = noErr;
+  if (_twoDevices) {
+    WEBRTC_CA_RETURN_ON_ERR(
+        AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
+  } else if (!_playing) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
+  }
+
+  _recording = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StopRecording() {
+  RTC_LOG(LS_INFO) << "StopRecording";
+  MutexLock lock(&mutex_);
+
+  if (!_recIsInitialized) {
+    return 0;
+  }
+
+  OSStatus err = noErr;
+  int32_t captureDeviceIsAlive = _captureDeviceIsAlive;
+  if (_twoDevices && captureDeviceIsAlive == 1) {
+    // Recording side uses its own dedicated device and IOProc.
+    if (_recording) {
+      _recording = false;
+      _doStopRec = true;  // Signal to io proc to stop audio device
+      mutex_.Unlock();    // Cannot be under lock, risk of deadlock
+      if (!_stopEventRec.Wait(TimeDelta::Seconds(2))) {
+        MutexLock lockScoped(&mutex_);
+        RTC_LOG(LS_WARNING) << "Timed out stopping the capture IOProc."
+                               "We may have failed to detect a device removal.";
+        WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+      }
+      mutex_.Lock();
+      _doStopRec = false;
+      RTC_LOG(LS_INFO) << "Recording stopped (input device)";
+    } else if (_recIsInitialized) {
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+      RTC_LOG(LS_INFO) << "Recording uninitialized (input device)";
+    }
+  } else {
+    // We signal a stop for a shared device even when rendering has
+    // not yet ended. This is to ensure the IOProc will return early as
+    // intended (by checking `_recording`) before accessing
+    // resources we free below (e.g. the capture converter).
+    //
+    // In the case of a shared devcie, the IOProc will verify
+    // rendering has ended before stopping itself.
+    if (_recording && captureDeviceIsAlive == 1) {
+      _recording = false;
+      _doStop = true;     // Signal to io proc to stop audio device
+      mutex_.Unlock();    // Cannot be under lock, risk of deadlock
+      if (!_stopEvent.Wait(TimeDelta::Seconds(2))) {
+        MutexLock lockScoped(&mutex_);
+        RTC_LOG(LS_WARNING) << "Timed out stopping the shared IOProc."
+                               "We may have failed to detect a device removal.";
+        // We assume rendering on a shared device has stopped as well if
+        // the IOProc times out.
+        WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+      }
+      mutex_.Lock();
+      _doStop = false;
+      RTC_LOG(LS_INFO) << "Recording stopped (shared device)";
+    } else if (_recIsInitialized && !_playing && !_playIsInitialized) {
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+      RTC_LOG(LS_INFO) << "Recording uninitialized (shared device)";
+    }
+  }
+
+  // Setting this signal will allow the worker thread to be stopped.
+  _captureDeviceIsAlive = 0;
+
+  if (!capture_worker_thread_.empty()) {
+    mutex_.Unlock();
+    capture_worker_thread_.Finalize();
+    mutex_.Lock();
+  }
+
+  WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
+
+  // Remove listeners.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeInput, 0};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  _recIsInitialized = false;
+  _recording = false;
+
+  return 0;
+}
+
+bool AudioDeviceMac::RecordingIsInitialized() const {
+  return (_recIsInitialized);
+}
+
+bool AudioDeviceMac::Recording() const {
+  return (_recording);
+}
+
+bool AudioDeviceMac::PlayoutIsInitialized() const {
+  return (_playIsInitialized);
+}
+
+int32_t AudioDeviceMac::StartPlayout() {
+  RTC_LOG(LS_INFO) << "StartPlayout";
+  MutexLock lock(&mutex_);
+
+  if (!_playIsInitialized) {
+    return -1;
+  }
+
+  if (_playing) {
+    return 0;
+  }
+
+  RTC_DCHECK(render_worker_thread_.empty());
+  render_worker_thread_ = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (RenderWorkerThread()) {
+        }
+      },
+      "RenderWorkerThread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  if (_twoDevices || !_recording) {
+    OSStatus err = noErr;
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
+  }
+  _playing = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StopPlayout() {
+  RTC_LOG(LS_INFO) << "StopPlayout";
+  MutexLock lock(&mutex_);
+
+  if (!_playIsInitialized) {
+    return 0;
+  }
+
+  OSStatus err = noErr;
+  int32_t renderDeviceIsAlive = _renderDeviceIsAlive;
+  if (_playing && renderDeviceIsAlive == 1) {
+    // We signal a stop for a shared device even when capturing has not
+    // yet ended. This is to ensure the IOProc will return early as
+    // intended (by checking `_playing`) before accessing resources we
+    // free below (e.g. the render converter).
+    //
+    // In the case of a shared device, the IOProc will verify capturing
+    // has ended before stopping itself.
+    _playing = false;
+    _doStop = true;     // Signal to io proc to stop audio device
+    mutex_.Unlock();    // Cannot be under lock, risk of deadlock
+    if (!_stopEvent.Wait(TimeDelta::Seconds(2))) {
+      MutexLock lockScoped(&mutex_);
+      RTC_LOG(LS_WARNING) << "Timed out stopping the render IOProc."
+                             "We may have failed to detect a device removal.";
+
+      // We assume capturing on a shared device has stopped as well if the
+      // IOProc times out.
+      WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+    }
+    mutex_.Lock();
+    _doStop = false;
+    RTC_LOG(LS_INFO) << "Playout stopped";
+  } else if (_twoDevices && _playIsInitialized) {
+    WEBRTC_CA_LOG_WARN(
+        AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+    RTC_LOG(LS_INFO) << "Playout uninitialized (output device)";
+  } else if (!_twoDevices && _playIsInitialized && !_recIsInitialized) {
+    WEBRTC_CA_LOG_WARN(
+        AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+    RTC_LOG(LS_INFO) << "Playout uninitialized (shared device)";
+  }
+
+  // Setting this signal will allow the worker thread to be stopped.
+  _renderDeviceIsAlive = 0;
+  if (!render_worker_thread_.empty()) {
+    mutex_.Unlock();
+    render_worker_thread_.Finalize();
+    mutex_.Lock();
+  }
+
+  WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
+
+  // Remove listeners.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput, 0};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  if (_macBookPro) {
+    Boolean hasProperty =
+        AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+    if (hasProperty) {
+      propertyAddress.mSelector = kAudioDevicePropertyDataSource;
+      WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+          _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+    }
+  }
+
+  _playIsInitialized = false;
+  _playing = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::PlayoutDelay(uint16_t& delayMS) const {
+  int32_t renderDelayUs = _renderDelayUs;
+  delayMS =
+      static_cast<uint16_t>(1e-3 * (renderDelayUs + _renderLatencyUs) + 0.5);
+  return 0;
+}
+
+bool AudioDeviceMac::Playing() const {
+  return (_playing);
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
+                                         AudioDeviceID scopedDeviceIds[],
+                                         const uint32_t deviceListLength) {
+  OSStatus err = noErr;
+
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  UInt32 size = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size));
+  if (size == 0) {
+    RTC_LOG(LS_WARNING) << "No devices";
+    return 0;
+  }
+
+  UInt32 numberDevices = size / sizeof(AudioDeviceID);
+  const auto deviceIds = std::make_unique<AudioDeviceID[]>(numberDevices);
+  AudioBufferList* bufferList = NULL;
+  UInt32 numberScopedDevices = 0;
+
+  // First check if there is a default device and list it
+  UInt32 hardwareProperty = 0;
+  if (scope == kAudioDevicePropertyScopeOutput) {
+    hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+  } else {
+    hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
+  }
+
+  AudioObjectPropertyAddress propertyAddressDefault = {
+      hardwareProperty, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+
+  AudioDeviceID usedID;
+  UInt32 uintSize = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
+                                                     &propertyAddressDefault, 0,
+                                                     NULL, &uintSize, &usedID));
+  if (usedID != kAudioDeviceUnknown) {
+    scopedDeviceIds[numberScopedDevices] = usedID;
+    numberScopedDevices++;
+  } else {
+    RTC_LOG(LS_WARNING) << "GetNumberDevices(): Default device unknown";
+  }
+
+  // Then list the rest of the devices
+  bool listOK = true;
+
+  WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
+                                               &propertyAddress, 0, NULL, &size,
+                                               deviceIds.get()));
+  if (err != noErr) {
+    listOK = false;
+  } else {
+    propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
+    propertyAddress.mScope = scope;
+    propertyAddress.mElement = 0;
+    for (UInt32 i = 0; i < numberDevices; i++) {
+      // Check for input channels
+      WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(
+          deviceIds[i], &propertyAddress, 0, NULL, &size));
+      if (err == kAudioHardwareBadDeviceError) {
+        // This device doesn't actually exist; continue iterating.
+        continue;
+      } else if (err != noErr) {
+        listOK = false;
+        break;
+      }
+
+      bufferList = (AudioBufferList*)malloc(size);
+      WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
+          deviceIds[i], &propertyAddress, 0, NULL, &size, bufferList));
+      if (err != noErr) {
+        listOK = false;
+        break;
+      }
+
+      if (bufferList->mNumberBuffers > 0) {
+        if (numberScopedDevices >= deviceListLength) {
+          RTC_LOG(LS_ERROR) << "Device list is not long enough";
+          listOK = false;
+          break;
+        }
+
+        scopedDeviceIds[numberScopedDevices] = deviceIds[i];
+        numberScopedDevices++;
+      }
+
+      free(bufferList);
+      bufferList = NULL;
+    }  // for
+  }
+
+  if (!listOK) {
+    if (bufferList) {
+      free(bufferList);
+      bufferList = NULL;
+    }
+    return -1;
+  }
+
+  return numberScopedDevices;
+}
+
+int32_t AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
+                                      const uint16_t index,
+                                      rtc::ArrayView<char> name) {
+  OSStatus err = noErr;
+  AudioDeviceID deviceIds[MaxNumberDevices];
+
+  int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
+  if (numberDevices < 0) {
+    return -1;
+  } else if (numberDevices == 0) {
+    RTC_LOG(LS_ERROR) << "No devices";
+    return -1;
+  }
+
+  // If the number is below the number of devices, assume it's "WEBRTC ID"
+  // otherwise assume it's a CoreAudio ID
+  AudioDeviceID usedID;
+
+  // Check if there is a default device
+  bool isDefaultDevice = false;
+  if (index == 0) {
+    UInt32 hardwareProperty = 0;
+    if (scope == kAudioDevicePropertyScopeOutput) {
+      hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+    } else {
+      hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
+    }
+    AudioObjectPropertyAddress propertyAddress = {
+        hardwareProperty, kAudioObjectPropertyScopeGlobal,
+        kAudioObjectPropertyElementMaster};
+    UInt32 size = sizeof(UInt32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, &usedID));
+    if (usedID == kAudioDeviceUnknown) {
+      RTC_LOG(LS_WARNING) << "GetDeviceName(): Default device unknown";
+    } else {
+      isDefaultDevice = true;
+    }
+  }
+
+  AudioObjectPropertyAddress propertyAddress = {kAudioDevicePropertyDeviceName,
+                                                scope, 0};
+
+  if (isDefaultDevice) {
+    std::array<char, kAdmMaxDeviceNameSize> devName;
+    UInt32 len = devName.size();
+
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        usedID, &propertyAddress, 0, NULL, &len, devName.data()));
+
+    rtc::SimpleStringBuilder ss(name);
+    ss.AppendFormat("default (%s)", devName.data());
+  } else {
+    if (index < numberDevices) {
+      usedID = deviceIds[index];
+    } else {
+      usedID = index;
+    }
+    UInt32 len = name.size();
+
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        usedID, &propertyAddress, 0, NULL, &len, name.data()));
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitDevice(const uint16_t userDeviceIndex,
+                                   AudioDeviceID& deviceId,
+                                   const bool isInput) {
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  AudioObjectPropertyScope deviceScope;
+  AudioObjectPropertySelector defaultDeviceSelector;
+  AudioDeviceID deviceIds[MaxNumberDevices];
+
+  if (isInput) {
+    deviceScope = kAudioDevicePropertyScopeInput;
+    defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
+  } else {
+    deviceScope = kAudioDevicePropertyScopeOutput;
+    defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
+  }
+
+  AudioObjectPropertyAddress propertyAddress = {
+      defaultDeviceSelector, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+
+  // Get the actual device IDs
+  int numberDevices =
+      GetNumberDevices(deviceScope, deviceIds, MaxNumberDevices);
+  if (numberDevices < 0) {
+    return -1;
+  } else if (numberDevices == 0) {
+    RTC_LOG(LS_ERROR) << "InitDevice(): No devices";
+    return -1;
+  }
+
+  bool isDefaultDevice = false;
+  deviceId = kAudioDeviceUnknown;
+  if (userDeviceIndex == 0) {
+    // Try to use default system device
+    size = sizeof(AudioDeviceID);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, &deviceId));
+    if (deviceId == kAudioDeviceUnknown) {
+      RTC_LOG(LS_WARNING) << "No default device exists";
+    } else {
+      isDefaultDevice = true;
+    }
+  }
+
+  if (!isDefaultDevice) {
+    deviceId = deviceIds[userDeviceIndex];
+  }
+
+  // Obtain device name and manufacturer for logging.
+  // Also use this as a test to ensure a user-set device ID is valid.
+  char devName[128];
+  char devManf[128];
+  memset(devName, 0, sizeof(devName));
+  memset(devManf, 0, sizeof(devManf));
+
+  propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
+  propertyAddress.mScope = deviceScope;
+  propertyAddress.mElement = 0;
+  size = sizeof(devName);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId, &propertyAddress,
+                                                     0, NULL, &size, devName));
+
+  propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
+  size = sizeof(devManf);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId, &propertyAddress,
+                                                     0, NULL, &size, devManf));
+
+  if (isInput) {
+    RTC_LOG(LS_INFO) << "Input device: " << devManf << " " << devName;
+  } else {
+    RTC_LOG(LS_INFO) << "Output device: " << devManf << " " << devName;
+  }
+
+  return 0;
+}
+
+OSStatus AudioDeviceMac::SetDesiredPlayoutFormat() {
+  // Our preferred format to work with.
+  _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
+  _outDesiredFormat.mChannelsPerFrame = _playChannels;
+
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters.
+    _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+    _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
+  }
+
+  _renderDelayOffsetSamples =
+      _renderBufSizeSamples - N_BUFFERS_OUT * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES *
+                                  _outDesiredFormat.mChannelsPerFrame;
+
+  _outDesiredFormat.mBytesPerPacket =
+      _outDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  // In uncompressed audio, a packet is one frame.
+  _outDesiredFormat.mFramesPerPacket = 1;
+  _outDesiredFormat.mBytesPerFrame =
+      _outDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
+
+  _outDesiredFormat.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
+  _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+  _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+
+  OSStatus err = noErr;
+  WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(
+      &_outDesiredFormat, &_outStreamFormat, &_renderConverter));
+
+  // Try to set buffer size to desired value set to 20ms.
+  const uint16_t kPlayBufDelayFixed = 20;
+  UInt32 bufByteCount = static_cast<UInt32>(
+      (_outStreamFormat.mSampleRate / 1000.0) * kPlayBufDelayFixed *
+      _outStreamFormat.mChannelsPerFrame * sizeof(Float32));
+  if (_outStreamFormat.mFramesPerPacket != 0) {
+    if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0) {
+      bufByteCount = (static_cast<UInt32>(bufByteCount /
+                                          _outStreamFormat.mFramesPerPacket) +
+                      1) *
+                     _outStreamFormat.mFramesPerPacket;
+    }
+  }
+
+  // Ensure the buffer size is within the range provided by the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyDataSource, kAudioDevicePropertyScopeOutput, 0};
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+  AudioValueRange range;
+  UInt32 size = sizeof(range);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &range));
+  if (range.mMinimum > bufByteCount) {
+    bufByteCount = range.mMinimum;
+  } else if (range.mMaximum < bufByteCount) {
+    bufByteCount = range.mMaximum;
+  }
+
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+  size = sizeof(bufByteCount);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, size, &bufByteCount));
+
+  // Get render device latency.
+  propertyAddress.mSelector = kAudioDevicePropertyLatency;
+  UInt32 latency = 0;
+  size = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _renderLatencyUs =
+      static_cast<uint32_t>((1.0e6 * latency) / _outStreamFormat.mSampleRate);
+
+  // Get render stream latency.
+  propertyAddress.mSelector = kAudioDevicePropertyStreams;
+  AudioStreamID stream = 0;
+  size = sizeof(AudioStreamID);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &stream));
+  propertyAddress.mSelector = kAudioStreamPropertyLatency;
+  size = sizeof(UInt32);
+  latency = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _renderLatencyUs +=
+      static_cast<uint32_t>((1.0e6 * latency) / _outStreamFormat.mSampleRate);
+
+  RTC_LOG(LS_VERBOSE) << "initial playout status: _renderDelayOffsetSamples="
+                      << _renderDelayOffsetSamples
+                      << ", _renderDelayUs=" << _renderDelayUs
+                      << ", _renderLatencyUs=" << _renderLatencyUs;
+  return 0;
+}
+
+OSStatus AudioDeviceMac::objectListenerProc(
+    AudioObjectID objectId,
+    UInt32 numberAddresses,
+    const AudioObjectPropertyAddress addresses[],
+    void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
+
+  ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
+
+  // AudioObjectPropertyListenerProc functions are supposed to return 0
+  return 0;
+}
+
+OSStatus AudioDeviceMac::implObjectListenerProc(
+    const AudioObjectID objectId,
+    const UInt32 numberAddresses,
+    const AudioObjectPropertyAddress addresses[]) {
+  RTC_LOG(LS_VERBOSE) << "AudioDeviceMac::implObjectListenerProc()";
+
+  for (UInt32 i = 0; i < numberAddresses; i++) {
+    if (addresses[i].mSelector == kAudioHardwarePropertyDevices) {
+      HandleDeviceChange();
+    } else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat) {
+      HandleStreamFormatChange(objectId, addresses[i]);
+    } else if (addresses[i].mSelector == kAudioDevicePropertyDataSource) {
+      HandleDataSourceChange(objectId, addresses[i]);
+    } else if (addresses[i].mSelector == kAudioDeviceProcessorOverload) {
+      HandleProcessorOverload(addresses[i]);
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::HandleDeviceChange() {
+  OSStatus err = noErr;
+
+  RTC_LOG(LS_VERBOSE) << "kAudioHardwarePropertyDevices";
+
+  // A device has changed. Check if our registered devices have been removed.
+  // Ensure the devices have been initialized, meaning the IDs are valid.
+  if (MicrophoneIsInitialized()) {
+    AudioObjectPropertyAddress propertyAddress = {
+        kAudioDevicePropertyDeviceIsAlive, kAudioDevicePropertyScopeInput, 0};
+    UInt32 deviceIsAlive = 1;
+    UInt32 size = sizeof(UInt32);
+    err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0, NULL,
+                                     &size, &deviceIsAlive);
+
+    if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0) {
+      RTC_LOG(LS_WARNING) << "Capture device is not alive (probably removed)";
+      _captureDeviceIsAlive = 0;
+      _mixerManager.CloseMicrophone();
+    } else if (err != noErr) {
+      logCAMsg(rtc::LS_ERROR, "Error in AudioDeviceGetProperty()",
+               (const char*)&err);
+      return -1;
+    }
+  }
+
+  if (SpeakerIsInitialized()) {
+    AudioObjectPropertyAddress propertyAddress = {
+        kAudioDevicePropertyDeviceIsAlive, kAudioDevicePropertyScopeOutput, 0};
+    UInt32 deviceIsAlive = 1;
+    UInt32 size = sizeof(UInt32);
+    err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0, NULL,
+                                     &size, &deviceIsAlive);
+
+    if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0) {
+      RTC_LOG(LS_WARNING) << "Render device is not alive (probably removed)";
+      _renderDeviceIsAlive = 0;
+      _mixerManager.CloseSpeaker();
+    } else if (err != noErr) {
+      logCAMsg(rtc::LS_ERROR, "Error in AudioDeviceGetProperty()",
+               (const char*)&err);
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::HandleStreamFormatChange(
+    const AudioObjectID objectId,
+    const AudioObjectPropertyAddress propertyAddress) {
+  OSStatus err = noErr;
+
+  RTC_LOG(LS_VERBOSE) << "Stream format changed";
+
+  if (objectId != _inputDeviceID && objectId != _outputDeviceID) {
+    return 0;
+  }
+
+  // Get the new device format
+  AudioStreamBasicDescription streamFormat;
+  UInt32 size = sizeof(streamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      objectId, &propertyAddress, 0, NULL, &size, &streamFormat));
+
+  if (streamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(rtc::LS_ERROR, "Unacceptable input stream format -> mFormatID",
+             (const char*)&streamFormat.mFormatID);
+    return -1;
+  }
+
+  if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    RTC_LOG(LS_ERROR) << "Too many channels on device (mChannelsPerFrame = "
+                      << streamFormat.mChannelsPerFrame << ")";
+    return -1;
+  }
+
+  if (_ptrAudioBuffer && streamFormat.mChannelsPerFrame != _recChannels) {
+    RTC_LOG(LS_ERROR) << "Changing channels not supported (mChannelsPerFrame = "
+                      << streamFormat.mChannelsPerFrame << ")";
+    return -1;
+  }
+
+  RTC_LOG(LS_VERBOSE) << "Stream format:";
+  RTC_LOG(LS_VERBOSE) << "mSampleRate = " << streamFormat.mSampleRate
+                      << ", mChannelsPerFrame = "
+                      << streamFormat.mChannelsPerFrame;
+  RTC_LOG(LS_VERBOSE) << "mBytesPerPacket = " << streamFormat.mBytesPerPacket
+                      << ", mFramesPerPacket = "
+                      << streamFormat.mFramesPerPacket;
+  RTC_LOG(LS_VERBOSE) << "mBytesPerFrame = " << streamFormat.mBytesPerFrame
+                      << ", mBitsPerChannel = " << streamFormat.mBitsPerChannel;
+  RTC_LOG(LS_VERBOSE) << "mFormatFlags = " << streamFormat.mFormatFlags;
+  logCAMsg(rtc::LS_VERBOSE, "mFormatID", (const char*)&streamFormat.mFormatID);
+
+  if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) {
+    const int io_block_size_samples = streamFormat.mChannelsPerFrame *
+                                      streamFormat.mSampleRate / 100 *
+                                      N_BLOCKS_IO;
+    if (io_block_size_samples > _captureBufSizeSamples) {
+      RTC_LOG(LS_ERROR) << "Input IO block size (" << io_block_size_samples
+                        << ") is larger than ring buffer ("
+                        << _captureBufSizeSamples << ")";
+      return -1;
+    }
+
+    memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
+
+    if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2)) {
+      _inDesiredFormat.mChannelsPerFrame = 2;
+    } else {
+      // Disable stereo recording when we only have one channel on the device.
+      _inDesiredFormat.mChannelsPerFrame = 1;
+      _recChannels = 1;
+      RTC_LOG(LS_VERBOSE) << "Stereo recording unavailable on this device";
+    }
+
+    // Recreate the converter with the new format
+    // TODO(xians): make this thread safe
+    WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
+
+    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
+                                              &_captureConverter));
+  } else {
+    memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
+
+    // Our preferred format to work with
+    if (_outStreamFormat.mChannelsPerFrame < 2) {
+      _playChannels = 1;
+      RTC_LOG(LS_VERBOSE) << "Stereo playout unavailable on this device";
+    }
+    WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
+  }
+  return 0;
+}
+
+int32_t AudioDeviceMac::HandleDataSourceChange(
+    const AudioObjectID objectId,
+    const AudioObjectPropertyAddress propertyAddress) {
+  OSStatus err = noErr;
+
+  if (_macBookPro &&
+      propertyAddress.mScope == kAudioDevicePropertyScopeOutput) {
+    RTC_LOG(LS_VERBOSE) << "Data source changed";
+
+    _macBookProPanRight = false;
+    UInt32 dataSource = 0;
+    UInt32 size = sizeof(UInt32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        objectId, &propertyAddress, 0, NULL, &size, &dataSource));
+    if (dataSource == 'ispk') {
+      _macBookProPanRight = true;
+      RTC_LOG(LS_VERBOSE)
+          << "MacBook Pro using internal speakers; stereo panning right";
+    } else {
+      RTC_LOG(LS_VERBOSE) << "MacBook Pro not using internal speakers";
+    }
+  }
+
+  return 0;
+}
+int32_t AudioDeviceMac::HandleProcessorOverload(
+    const AudioObjectPropertyAddress propertyAddress) {
+  // TODO(xians): we probably want to notify the user in some way of the
+  // overload. However, the Windows interpretations of these errors seem to
+  // be more severe than what ProcessorOverload is thrown for.
+  //
+  // We don't log the notification, as it's sent from the HAL's IO thread. We
+  // don't want to slow it down even further.
+  if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) {
+    // RTC_LOG(LS_WARNING) << "Capture processor // overload";
+    //_callback->ProblemIsReported(
+    // SndCardStreamObserver::ERecordingProblem);
+  } else {
+    // RTC_LOG(LS_WARNING) << "Render processor overload";
+    //_callback->ProblemIsReported(
+    // SndCardStreamObserver::EPlaybackProblem);
+  }
+
+  return 0;
+}
+
+// ============================================================================
+//                                  Thread Methods
+// ============================================================================
+
+OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID,
+                                      const AudioTimeStamp*,
+                                      const AudioBufferList* inputData,
+                                      const AudioTimeStamp* inputTime,
+                                      AudioBufferList* outputData,
+                                      const AudioTimeStamp* outputTime,
+                                      void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
+
+  ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
+
+  // AudioDeviceIOProc functions are supposed to return 0
+  return 0;
+}
+
+OSStatus AudioDeviceMac::outConverterProc(AudioConverterRef,
+                                          UInt32* numberDataPackets,
+                                          AudioBufferList* data,
+                                          AudioStreamPacketDescription**,
+                                          void* userData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)userData;
+  RTC_DCHECK(ptrThis != NULL);
+
+  return ptrThis->implOutConverterProc(numberDataPackets, data);
+}
+
+OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID,
+                                        const AudioTimeStamp*,
+                                        const AudioBufferList* inputData,
+                                        const AudioTimeStamp* inputTime,
+                                        AudioBufferList*,
+                                        const AudioTimeStamp*,
+                                        void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
+
+  ptrThis->implInDeviceIOProc(inputData, inputTime);
+
+  // AudioDeviceIOProc functions are supposed to return 0
+  return 0;
+}
+
+OSStatus AudioDeviceMac::inConverterProc(
+    AudioConverterRef,
+    UInt32* numberDataPackets,
+    AudioBufferList* data,
+    AudioStreamPacketDescription** /*dataPacketDescription*/,
+    void* userData) {
+  AudioDeviceMac* ptrThis = static_cast<AudioDeviceMac*>(userData);
+  RTC_DCHECK(ptrThis != NULL);
+
+  return ptrThis->implInConverterProc(numberDataPackets, data);
+}
+
+OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList* inputData,
+                                          const AudioTimeStamp* inputTime,
+                                          AudioBufferList* outputData,
+                                          const AudioTimeStamp* outputTime) {
+  OSStatus err = noErr;
+  UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
+  UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+
+  if (!_twoDevices && _recording) {
+    implInDeviceIOProc(inputData, inputTime);
+  }
+
+  // Check if we should close down audio device
+  // Double-checked locking optimization to remove locking overhead
+  if (_doStop) {
+    MutexLock lock(&mutex_);
+    if (_doStop) {
+      if (_twoDevices || (!_recording && !_playing)) {
+        // In the case of a shared device, the single driving ioProc
+        // is stopped here
+        WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+        if (err == noErr) {
+          RTC_LOG(LS_VERBOSE) << "Playout or shared device stopped";
+        }
+      }
+
+      _doStop = false;
+      _stopEvent.Set();
+      return 0;
+    }
+  }
+
+  if (!_playing) {
+    // This can be the case when a shared device is capturing but not
+    // rendering. We allow the checks above before returning to avoid a
+    // timeout when capturing is stopped.
+    return 0;
+  }
+
+  RTC_DCHECK(_outStreamFormat.mBytesPerFrame != 0);
+  UInt32 size =
+      outputData->mBuffers->mDataByteSize / _outStreamFormat.mBytesPerFrame;
+
+  // TODO(xians): signal an error somehow?
+  err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
+                                        this, &size, outputData, NULL);
+  if (err != noErr) {
+    if (err == 1) {
+      // This is our own error.
+      RTC_LOG(LS_ERROR) << "Error in AudioConverterFillComplexBuffer()";
+      return 1;
+    } else {
+      logCAMsg(rtc::LS_ERROR, "Error in AudioConverterFillComplexBuffer()",
+               (const char*)&err);
+      return 1;
+    }
+  }
+
+  ring_buffer_size_t bufSizeSamples =
+      PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
+
+  int32_t renderDelayUs =
+      static_cast<int32_t>(1e-3 * (outputTimeNs - nowNs) + 0.5);
+  renderDelayUs += static_cast<int32_t>(
+      (1.0e6 * bufSizeSamples) / _outDesiredFormat.mChannelsPerFrame /
+          _outDesiredFormat.mSampleRate +
+      0.5);
+
+  _renderDelayUs = renderDelayUs;
+
+  return 0;
+}
+
+OSStatus AudioDeviceMac::implOutConverterProc(UInt32* numberDataPackets,
+                                              AudioBufferList* data) {
+  RTC_DCHECK(data->mNumberBuffers == 1);
+  ring_buffer_size_t numSamples =
+      *numberDataPackets * _outDesiredFormat.mChannelsPerFrame;
+
+  data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
+  // Always give the converter as much as it wants, zero padding as required.
+  data->mBuffers->mDataByteSize =
+      *numberDataPackets * _outDesiredFormat.mBytesPerPacket;
+  data->mBuffers->mData = _renderConvertData;
+  memset(_renderConvertData, 0, sizeof(_renderConvertData));
+
+  PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
+
+  kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "semaphore_signal_all() error: " << kernErr;
+    return 1;
+  }
+
+  return 0;
+}
+
+OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList* inputData,
+                                            const AudioTimeStamp* inputTime) {
+  OSStatus err = noErr;
+  UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
+  UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+
+  // Check if we should close down audio device
+  // Double-checked locking optimization to remove locking overhead
+  if (_doStopRec) {
+    MutexLock lock(&mutex_);
+    if (_doStopRec) {
+      // This will be signalled only when a shared device is not in use.
+      WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+      if (err == noErr) {
+        RTC_LOG(LS_VERBOSE) << "Recording device stopped";
+      }
+
+      _doStopRec = false;
+      _stopEventRec.Set();
+      return 0;
+    }
+  }
+
+  if (!_recording) {
+    // Allow above checks to avoid a timeout on stopping capture.
+    return 0;
+  }
+
+  ring_buffer_size_t bufSizeSamples =
+      PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
+
+  int32_t captureDelayUs =
+      static_cast<int32_t>(1e-3 * (nowNs - inputTimeNs) + 0.5);
+  captureDelayUs += static_cast<int32_t>((1.0e6 * bufSizeSamples) /
+                                             _inStreamFormat.mChannelsPerFrame /
+                                             _inStreamFormat.mSampleRate +
+                                         0.5);
+
+  _captureDelayUs = captureDelayUs;
+
+  RTC_DCHECK(inputData->mNumberBuffers == 1);
+  ring_buffer_size_t numSamples = inputData->mBuffers->mDataByteSize *
+                                  _inStreamFormat.mChannelsPerFrame /
+                                  _inStreamFormat.mBytesPerPacket;
+  PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
+                         numSamples);
+
+  kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "semaphore_signal_all() error: " << kernErr;
+  }
+
+  return err;
+}
+
+OSStatus AudioDeviceMac::implInConverterProc(UInt32* numberDataPackets,
+                                             AudioBufferList* data) {
+  RTC_DCHECK(data->mNumberBuffers == 1);
+  ring_buffer_size_t numSamples =
+      *numberDataPackets * _inStreamFormat.mChannelsPerFrame;
+
+  while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples) {
+    mach_timespec_t timeout;
+    timeout.tv_sec = 0;
+    timeout.tv_nsec = TIMER_PERIOD_MS;
+
+    kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
+    if (kernErr == KERN_OPERATION_TIMED_OUT) {
+      int32_t signal = _captureDeviceIsAlive;
+      if (signal == 0) {
+        // The capture device is no longer alive; stop the worker thread.
+        *numberDataPackets = 0;
+        return 1;
+      }
+    } else if (kernErr != KERN_SUCCESS) {
+      RTC_LOG(LS_ERROR) << "semaphore_wait() error: " << kernErr;
+    }
+  }
+
+  // Pass the read pointer directly to the converter to avoid a memcpy.
+  void* dummyPtr;
+  ring_buffer_size_t dummySize;
+  PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
+                                  &data->mBuffers->mData, &numSamples,
+                                  &dummyPtr, &dummySize);
+  PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
+
+  data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
+  *numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
+  data->mBuffers->mDataByteSize =
+      *numberDataPackets * _inStreamFormat.mBytesPerPacket;
+
+  return 0;
+}
+
+bool AudioDeviceMac::RenderWorkerThread() {
+  ring_buffer_size_t numSamples =
+      ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
+  while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer) -
+             _renderDelayOffsetSamples <
+         numSamples) {
+    mach_timespec_t timeout;
+    timeout.tv_sec = 0;
+    timeout.tv_nsec = TIMER_PERIOD_MS;
+
+    kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
+    if (kernErr == KERN_OPERATION_TIMED_OUT) {
+      int32_t signal = _renderDeviceIsAlive;
+      if (signal == 0) {
+        // The render device is no longer alive; stop the worker thread.
+        return false;
+      }
+    } else if (kernErr != KERN_SUCCESS) {
+      RTC_LOG(LS_ERROR) << "semaphore_timedwait() error: " << kernErr;
+    }
+  }
+
+  int8_t playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
+
+  if (!_ptrAudioBuffer) {
+    RTC_LOG(LS_ERROR) << "capture AudioBuffer is invalid";
+    return false;
+  }
+
+  // Ask for new PCM data to be played out using the AudioDeviceBuffer.
+  uint32_t nSamples =
+      _ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
+
+  nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+  if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES) {
+    RTC_LOG(LS_ERROR) << "invalid number of output samples(" << nSamples << ")";
+  }
+
+  uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
+
+  SInt16* pPlayBuffer = (SInt16*)&playBuffer;
+  if (_macBookProPanRight && (_playChannels == 2)) {
+    // Mix entirely into the right channel and zero the left channel.
+    SInt32 sampleInt32 = 0;
+    for (uint32_t sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx += 2) {
+      sampleInt32 = pPlayBuffer[sampleIdx];
+      sampleInt32 += pPlayBuffer[sampleIdx + 1];
+      sampleInt32 /= 2;
+
+      if (sampleInt32 > 32767) {
+        sampleInt32 = 32767;
+      } else if (sampleInt32 < -32768) {
+        sampleInt32 = -32768;
+      }
+
+      pPlayBuffer[sampleIdx] = 0;
+      pPlayBuffer[sampleIdx + 1] = static_cast<SInt16>(sampleInt32);
+    }
+  }
+
+  PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
+
+  return true;
+}
+
+bool AudioDeviceMac::CaptureWorkerThread() {
+  OSStatus err = noErr;
+  UInt32 noRecSamples =
+      ENGINE_REC_BUF_SIZE_IN_SAMPLES * _inDesiredFormat.mChannelsPerFrame;
+  SInt16 recordBuffer[noRecSamples];
+  UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
+
+  AudioBufferList engineBuffer;
+  engineBuffer.mNumberBuffers = 1;  // Interleaved channels.
+  engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
+  engineBuffer.mBuffers->mDataByteSize =
+      _inDesiredFormat.mBytesPerPacket * noRecSamples;
+  engineBuffer.mBuffers->mData = recordBuffer;
+
+  err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
+                                        this, &size, &engineBuffer, NULL);
+  if (err != noErr) {
+    if (err == 1) {
+      // This is our own error.
+      return false;
+    } else {
+      logCAMsg(rtc::LS_ERROR, "Error in AudioConverterFillComplexBuffer()",
+               (const char*)&err);
+      return false;
+    }
+  }
+
+  // TODO(xians): what if the returned size is incorrect?
+  if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES) {
+    int32_t msecOnPlaySide;
+    int32_t msecOnRecordSide;
+
+    int32_t captureDelayUs = _captureDelayUs;
+    int32_t renderDelayUs = _renderDelayUs;
+
+    msecOnPlaySide =
+        static_cast<int32_t>(1e-3 * (renderDelayUs + _renderLatencyUs) + 0.5);
+    msecOnRecordSide =
+        static_cast<int32_t>(1e-3 * (captureDelayUs + _captureLatencyUs) + 0.5);
+
+    if (!_ptrAudioBuffer) {
+      RTC_LOG(LS_ERROR) << "capture AudioBuffer is invalid";
+      return false;
+    }
+
+    // store the recorded buffer (no action will be taken if the
+    // #recorded samples is not a full buffer)
+    _ptrAudioBuffer->SetRecordedBuffer((int8_t*)&recordBuffer, (uint32_t)size);
+    _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide);
+    _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
+    // deliver recorded samples at specified sample rate, mic level etc.
+    // to the observer using callback
+    _ptrAudioBuffer->DeliverRecordedData();
+  }
+
+  return true;
+}
+
+bool AudioDeviceMac::KeyPressed() {
+  bool key_down = false;
+  // Loop through all Mac virtual key constant values.
+  for (unsigned int key_index = 0; key_index < arraysize(prev_key_state_);
+       ++key_index) {
+    bool keyState =
+        CGEventSourceKeyState(kCGEventSourceStateHIDSystemState, key_index);
+    // A false -> true change in keymap means a key is pressed.
+    key_down |= (keyState && !prev_key_state_[key_index]);
+    // Save current state.
+    prev_key_state_[key_index] = keyState;
+  }
+  return key_down;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/mac/audio_device_mac.h b/third_party/libwebrtc/modules/audio_device/mac/audio_device_mac.h
new file mode 100644
index 0000000000..bb06395d03
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/mac/audio_device_mac.h
@@ -0,0 +1,350 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_MAC_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_MAC_H_
+
+#include <AudioToolbox/AudioConverter.h>
+#include <CoreAudio/CoreAudio.h>
+#include <mach/semaphore.h>
+
+#include <atomic>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/mac/audio_mixer_manager_mac.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+struct PaUtilRingBuffer;
+
+namespace webrtc {
+
+const uint32_t N_REC_SAMPLES_PER_SEC = 48000;
+const uint32_t N_PLAY_SAMPLES_PER_SEC = 48000;
+
+const uint32_t N_REC_CHANNELS = 1;   // default is mono recording
+const uint32_t N_PLAY_CHANNELS = 2;  // default is stereo playout
+const uint32_t N_DEVICE_CHANNELS = 64;
+
+const int kBufferSizeMs = 10;
+
+const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES =
+    N_REC_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
+const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES =
+    N_PLAY_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
+
+const int N_BLOCKS_IO = 2;
+const int N_BUFFERS_IN = 2;   // Must be at least N_BLOCKS_IO.
+const int N_BUFFERS_OUT = 3;  // Must be at least N_BLOCKS_IO.
+
+const uint32_t TIMER_PERIOD_MS = 2 * 10 * N_BLOCKS_IO * 1000000;
+
+const uint32_t REC_BUF_SIZE_IN_SAMPLES =
+    ENGINE_REC_BUF_SIZE_IN_SAMPLES * N_DEVICE_CHANNELS * N_BUFFERS_IN;
+const uint32_t PLAY_BUF_SIZE_IN_SAMPLES =
+    ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT;
+
+const int kGetMicVolumeIntervalMs = 1000;
+
+class AudioDeviceMac : public AudioDeviceGeneric {
+ public:
+  AudioDeviceMac();
+  ~AudioDeviceMac();
+
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const;
+
+  // Main initializaton and termination
+  virtual InitStatus Init() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool Initialized() const;
+
+  // Device enumeration
+  virtual int16_t PlayoutDevices();
+  virtual int16_t RecordingDevices();
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]);
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]);
+
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index) RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
+  virtual int32_t SetRecordingDevice(uint16_t index);
+  virtual int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device);
+
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool& available);
+  virtual int32_t InitPlayout() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool PlayoutIsInitialized() const;
+  virtual int32_t RecordingIsAvailable(bool& available);
+  virtual int32_t InitRecording() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool RecordingIsInitialized() const;
+
+  // Audio transport control
+  virtual int32_t StartPlayout() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool Playing() const;
+  virtual int32_t StartRecording() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool Recording() const;
+
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool SpeakerIsInitialized() const;
+  virtual int32_t InitMicrophone() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool MicrophoneIsInitialized() const;
+
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetSpeakerVolume(uint32_t volume);
+  virtual int32_t SpeakerVolume(uint32_t& volume) const;
+  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetMicrophoneVolume(uint32_t volume);
+  virtual int32_t MicrophoneVolume(uint32_t& volume) const;
+  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetMicrophoneMute(bool enable);
+  virtual int32_t MicrophoneMute(bool& enabled) const;
+
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetSpeakerMute(bool enable);
+  virtual int32_t SpeakerMute(bool& enabled) const;
+
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetStereoPlayout(bool enable);
+  virtual int32_t StereoPlayout(bool& enabled) const;
+  virtual int32_t StereoRecordingIsAvailable(bool& available);
+  virtual int32_t SetStereoRecording(bool enable);
+  virtual int32_t StereoRecording(bool& enabled) const;
+
+  // Delay information and control
+  virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
+
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+      RTC_LOCKS_EXCLUDED(mutex_);
+
+ private:
+  int32_t InitSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  virtual int32_t MicrophoneIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t MicrophoneIsAvailableLocked(bool& available)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  virtual int32_t SpeakerIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SpeakerIsAvailableLocked(bool& available)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  static void AtomicSet32(int32_t* theValue, int32_t newValue);
+  static int32_t AtomicGet32(int32_t* theValue);
+
+  static void logCAMsg(rtc::LoggingSeverity sev,
+                       const char* msg,
+                       const char* err);
+
+  int32_t GetNumberDevices(AudioObjectPropertyScope scope,
+                           AudioDeviceID scopedDeviceIds[],
+                           uint32_t deviceListLength);
+
+  int32_t GetDeviceName(AudioObjectPropertyScope scope,
+                        uint16_t index,
+                        rtc::ArrayView<char> name);
+
+  int32_t InitDevice(uint16_t userDeviceIndex,
+                     AudioDeviceID& deviceId,
+                     bool isInput);
+
+  // Always work with our preferred playout format inside VoE.
+  // Then convert the output to the OS setting using an AudioConverter.
+  OSStatus SetDesiredPlayoutFormat();
+
+  static OSStatus objectListenerProc(
+      AudioObjectID objectId,
+      UInt32 numberAddresses,
+      const AudioObjectPropertyAddress addresses[],
+      void* clientData);
+
+  OSStatus implObjectListenerProc(AudioObjectID objectId,
+                                  UInt32 numberAddresses,
+                                  const AudioObjectPropertyAddress addresses[]);
+
+  int32_t HandleDeviceChange();
+
+  int32_t HandleStreamFormatChange(AudioObjectID objectId,
+                                   AudioObjectPropertyAddress propertyAddress);
+
+  int32_t HandleDataSourceChange(AudioObjectID objectId,
+                                 AudioObjectPropertyAddress propertyAddress);
+
+  int32_t HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
+
+  static OSStatus deviceIOProc(AudioDeviceID device,
+                               const AudioTimeStamp* now,
+                               const AudioBufferList* inputData,
+                               const AudioTimeStamp* inputTime,
+                               AudioBufferList* outputData,
+                               const AudioTimeStamp* outputTime,
+                               void* clientData);
+
+  static OSStatus outConverterProc(
+      AudioConverterRef audioConverter,
+      UInt32* numberDataPackets,
+      AudioBufferList* data,
+      AudioStreamPacketDescription** dataPacketDescription,
+      void* userData);
+
+  static OSStatus inDeviceIOProc(AudioDeviceID device,
+                                 const AudioTimeStamp* now,
+                                 const AudioBufferList* inputData,
+                                 const AudioTimeStamp* inputTime,
+                                 AudioBufferList* outputData,
+                                 const AudioTimeStamp* outputTime,
+                                 void* clientData);
+
+  static OSStatus inConverterProc(
+      AudioConverterRef audioConverter,
+      UInt32* numberDataPackets,
+      AudioBufferList* data,
+      AudioStreamPacketDescription** dataPacketDescription,
+      void* inUserData);
+
+  OSStatus implDeviceIOProc(const AudioBufferList* inputData,
+                            const AudioTimeStamp* inputTime,
+                            AudioBufferList* outputData,
+                            const AudioTimeStamp* outputTime)
+      RTC_LOCKS_EXCLUDED(mutex_);
+
+  OSStatus implOutConverterProc(UInt32* numberDataPackets,
+                                AudioBufferList* data);
+
+  OSStatus implInDeviceIOProc(const AudioBufferList* inputData,
+                              const AudioTimeStamp* inputTime)
+      RTC_LOCKS_EXCLUDED(mutex_);
+
+  OSStatus implInConverterProc(UInt32* numberDataPackets,
+                               AudioBufferList* data);
+
+  static void RunCapture(void*);
+  static void RunRender(void*);
+  bool CaptureWorkerThread();
+  bool RenderWorkerThread();
+
+  bool KeyPressed();
+
+  AudioDeviceBuffer* _ptrAudioBuffer;
+
+  Mutex mutex_;
+
+  rtc::Event _stopEventRec;
+  rtc::Event _stopEvent;
+
+  // Only valid/running between calls to StartRecording and StopRecording.
+  rtc::PlatformThread capture_worker_thread_;
+
+  // Only valid/running between calls to StartPlayout and StopPlayout.
+  rtc::PlatformThread render_worker_thread_;
+
+  AudioMixerManagerMac _mixerManager;
+
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+  AudioDeviceID _inputDeviceID;
+  AudioDeviceID _outputDeviceID;
+#if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050
+  AudioDeviceIOProcID _inDeviceIOProcID;
+  AudioDeviceIOProcID _deviceIOProcID;
+#endif
+  bool _inputDeviceIsSpecified;
+  bool _outputDeviceIsSpecified;
+
+  uint8_t _recChannels;
+  uint8_t _playChannels;
+
+  Float32* _captureBufData;
+  SInt16* _renderBufData;
+
+  SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
+
+  bool _initialized;
+  bool _isShutDown;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+
+  // Atomically set varaibles
+  std::atomic<int32_t> _renderDeviceIsAlive;
+  std::atomic<int32_t> _captureDeviceIsAlive;
+
+  bool _twoDevices;
+  bool _doStop;  // For play if not shared device or play+rec if shared device
+  bool _doStopRec;  // For rec if not shared device
+  bool _macBookPro;
+  bool _macBookProPanRight;
+
+  AudioConverterRef _captureConverter;
+  AudioConverterRef _renderConverter;
+
+  AudioStreamBasicDescription _outStreamFormat;
+  AudioStreamBasicDescription _outDesiredFormat;
+  AudioStreamBasicDescription _inStreamFormat;
+  AudioStreamBasicDescription _inDesiredFormat;
+
+  uint32_t _captureLatencyUs;
+  uint32_t _renderLatencyUs;
+
+  // Atomically set variables
+  mutable std::atomic<int32_t> _captureDelayUs;
+  mutable std::atomic<int32_t> _renderDelayUs;
+
+  int32_t _renderDelayOffsetSamples;
+
+  PaUtilRingBuffer* _paCaptureBuffer;
+  PaUtilRingBuffer* _paRenderBuffer;
+
+  semaphore_t _renderSemaphore;
+  semaphore_t _captureSemaphore;
+
+  int _captureBufSizeSamples;
+  int _renderBufSizeSamples;
+
+  // Typing detection
+  // 0x5c is key "9", after that comes function keys.
+  bool prev_key_state_[0x5d];
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_
diff --git a/third_party/libwebrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc b/third_party/libwebrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
new file mode 100644
index 0000000000..942e7db3b3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
@@ -0,0 +1,924 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/mac/audio_mixer_manager_mac.h"
+
+#include <unistd.h>  // getpid()
+
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+
+#define WEBRTC_CA_RETURN_ON_ERR(expr)                                \
+  do {                                                               \
+    err = expr;                                                      \
+    if (err != noErr) {                                              \
+      logCAMsg(rtc::LS_ERROR, "Error in " #expr, (const char*)&err); \
+      return -1;                                                     \
+    }                                                                \
+  } while (0)
+
+#define WEBRTC_CA_LOG_ERR(expr)                                      \
+  do {                                                               \
+    err = expr;                                                      \
+    if (err != noErr) {                                              \
+      logCAMsg(rtc::LS_ERROR, "Error in " #expr, (const char*)&err); \
+    }                                                                \
+  } while (0)
+
+#define WEBRTC_CA_LOG_WARN(expr)                                       \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(rtc::LS_WARNING, "Error in " #expr, (const char*)&err); \
+    }                                                                  \
+  } while (0)
+
+AudioMixerManagerMac::AudioMixerManagerMac()
+    : _inputDeviceID(kAudioObjectUnknown),
+      _outputDeviceID(kAudioObjectUnknown),
+      _noInputChannels(0),
+      _noOutputChannels(0) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+}
+
+AudioMixerManagerMac::~AudioMixerManagerMac() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+  Close();
+}
+
+// ============================================================================
+//	                                PUBLIC METHODS
+// ============================================================================
+
+int32_t AudioMixerManagerMac::Close() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  MutexLock lock(&mutex_);
+
+  CloseSpeakerLocked();
+  CloseMicrophoneLocked();
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::CloseSpeaker() {
+  MutexLock lock(&mutex_);
+  return CloseSpeakerLocked();
+}
+
+int32_t AudioMixerManagerMac::CloseSpeakerLocked() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  _outputDeviceID = kAudioObjectUnknown;
+  _noOutputChannels = 0;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::CloseMicrophone() {
+  MutexLock lock(&mutex_);
+  return CloseMicrophoneLocked();
+}
+
+int32_t AudioMixerManagerMac::CloseMicrophoneLocked() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  _inputDeviceID = kAudioObjectUnknown;
+  _noInputChannels = 0;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::OpenSpeaker(AudioDeviceID deviceID) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::OpenSpeaker(id=" << deviceID
+                      << ")";
+
+  MutexLock lock(&mutex_);
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  pid_t hogPid = -1;
+
+  _outputDeviceID = deviceID;
+
+  // Check which process, if any, has hogged the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyHogMode, kAudioDevicePropertyScopeOutput, 0};
+
+  // First, does it have the property? Aggregate devices don't.
+  if (AudioObjectHasProperty(_outputDeviceID, &propertyAddress)) {
+    size = sizeof(hogPid);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, &size, &hogPid));
+
+    if (hogPid == -1) {
+      RTC_LOG(LS_VERBOSE) << "No process has hogged the output device";
+    }
+    // getpid() is apparently "always successful"
+    else if (hogPid == getpid()) {
+      RTC_LOG(LS_VERBOSE) << "Our process has hogged the output device";
+    } else {
+      RTC_LOG(LS_WARNING) << "Another process (pid = "
+                          << static_cast<int>(hogPid)
+                          << ") has hogged the output device";
+
+      return -1;
+    }
+  }
+
+  // get number of channels from stream format
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+
+  // Get the stream format, to be able to read the number of channels.
+  AudioStreamBasicDescription streamFormat;
+  size = sizeof(AudioStreamBasicDescription);
+  memset(&streamFormat, 0, size);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &streamFormat));
+
+  _noOutputChannels = streamFormat.mChannelsPerFrame;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::OpenMicrophone(AudioDeviceID deviceID) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::OpenMicrophone(id=" << deviceID
+                      << ")";
+
+  MutexLock lock(&mutex_);
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  pid_t hogPid = -1;
+
+  _inputDeviceID = deviceID;
+
+  // Check which process, if any, has hogged the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyHogMode, kAudioDevicePropertyScopeInput, 0};
+  size = sizeof(hogPid);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &hogPid));
+  if (hogPid == -1) {
+    RTC_LOG(LS_VERBOSE) << "No process has hogged the input device";
+  }
+  // getpid() is apparently "always successful"
+  else if (hogPid == getpid()) {
+    RTC_LOG(LS_VERBOSE) << "Our process has hogged the input device";
+  } else {
+    RTC_LOG(LS_WARNING) << "Another process (pid = " << static_cast<int>(hogPid)
+                        << ") has hogged the input device";
+
+    return -1;
+  }
+
+  // get number of channels from stream format
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+
+  // Get the stream format, to be able to read the number of channels.
+  AudioStreamBasicDescription streamFormat;
+  size = sizeof(AudioStreamBasicDescription);
+  memset(&streamFormat, 0, size);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &streamFormat));
+
+  _noInputChannels = streamFormat.mChannelsPerFrame;
+
+  return 0;
+}
+
+bool AudioMixerManagerMac::SpeakerIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_outputDeviceID != kAudioObjectUnknown);
+}
+
+bool AudioMixerManagerMac::MicrophoneIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_inputDeviceID != kAudioObjectUnknown);
+}
+
+int32_t AudioMixerManagerMac::SetSpeakerVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::SetSpeakerVolume(volume="
+                      << volume << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  bool success = false;
+
+  // volume range is 0.0 - 1.0, convert from 0 -255
+  const Float32 vol = (Float32)(volume / 255.0);
+
+  RTC_DCHECK(vol <= 1.0 && vol >= 0.0);
+
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(vol);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, size, &vol));
+
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err == noErr && isSettable) {
+      size = sizeof(vol);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, size, &vol));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    RTC_LOG(LS_WARNING) << "Unable to set a volume on any output channel";
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SpeakerVolume(uint32_t& volume) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  Float32 channelVol = 0;
+  Float32 vol = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(vol);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, &size, &vol));
+
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(vol * 255 + 0.5);
+  } else {
+    // Otherwise get the average volume across channels.
+    vol = 0;
+    for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+      channelVol = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelVol);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _outputDeviceID, &propertyAddress, 0, NULL, &size, &channelVol));
+
+        vol += channelVol;
+        channels++;
+      }
+    }
+
+    if (channels == 0) {
+      RTC_LOG(LS_WARNING) << "Unable to get a volume on any channel";
+      return -1;
+    }
+
+    RTC_DCHECK_GT(channels, 0);
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(255 * vol / channels + 0.5);
+  }
+
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::SpeakerVolume() => vol=" << vol;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  maxVolume = 255;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MinSpeakerVolume(uint32_t& minVolume) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  minVolume = 0;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SpeakerVolumeIsAvailable(bool& available) {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    available = true;
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err != noErr || !isSettable) {
+      available = false;
+      RTC_LOG(LS_WARNING) << "Volume cannot be set for output channel " << i
+                          << ", err=" << err;
+      return -1;
+    }
+  }
+
+  available = true;
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SpeakerMuteIsAvailable(bool& available) {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+
+  // Does the capture device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    available = true;
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err != noErr || !isSettable) {
+      available = false;
+      RTC_LOG(LS_WARNING) << "Mute cannot be set for output channel " << i
+                          << ", err=" << err;
+      return -1;
+    }
+  }
+
+  available = true;
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SetSpeakerMute(bool enable) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::SetSpeakerMute(enable="
+                      << enable << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  UInt32 mute = enable ? 1 : 0;
+  bool success = false;
+
+  // Does the render device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeOutput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(mute);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, size, &mute));
+
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err == noErr && isSettable) {
+      size = sizeof(mute);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, size, &mute));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    RTC_LOG(LS_WARNING) << "Unable to set mute on any input channel";
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SpeakerMute(bool& enabled) const {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  UInt32 channelMuted = 0;
+  UInt32 muted = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeOutput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(muted);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _outputDeviceID, &propertyAddress, 0, NULL, &size, &muted));
+
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  } else {
+    // Otherwise check if all channels are muted.
+    for (UInt32 i = 1; i <= _noOutputChannels; i++) {
+      muted = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelMuted);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _outputDeviceID, &propertyAddress, 0, NULL, &size, &channelMuted));
+
+        muted = (muted && channelMuted);
+        channels++;
+      }
+    }
+
+    if (channels == 0) {
+      RTC_LOG(LS_WARNING) << "Unable to get mute for any channel";
+      return -1;
+    }
+
+    RTC_DCHECK_GT(channels, 0);
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  }
+
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::SpeakerMute() => enabled="
+                      << enabled;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::StereoPlayoutIsAvailable(bool& available) {
+  if (_outputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  available = (_noOutputChannels == 2);
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::StereoRecordingIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  available = (_noInputChannels == 2);
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneMuteIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+
+  // Does the capture device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    available = true;
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err != noErr || !isSettable) {
+      available = false;
+      RTC_LOG(LS_WARNING) << "Mute cannot be set for output channel " << i
+                          << ", err=" << err;
+      return -1;
+    }
+  }
+
+  available = true;
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SetMicrophoneMute(bool enable) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::SetMicrophoneMute(enable="
+                      << enable << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  UInt32 mute = enable ? 1 : 0;
+  bool success = false;
+
+  // Does the capture device have a master mute control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(mute);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, size, &mute));
+
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err == noErr && isSettable) {
+      size = sizeof(mute);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _inputDeviceID, &propertyAddress, 0, NULL, size, &mute));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    RTC_LOG(LS_WARNING) << "Unable to set mute on any input channel";
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneMute(bool& enabled) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  UInt32 channelMuted = 0;
+  UInt32 muted = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyMute, kAudioDevicePropertyScopeInput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(muted);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, &size, &muted));
+
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  } else {
+    // Otherwise check if all channels are muted.
+    for (UInt32 i = 1; i <= _noInputChannels; i++) {
+      muted = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelMuted);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _inputDeviceID, &propertyAddress, 0, NULL, &size, &channelMuted));
+
+        muted = (muted && channelMuted);
+        channels++;
+      }
+    }
+
+    if (channels == 0) {
+      RTC_LOG(LS_WARNING) << "Unable to get mute for any channel";
+      return -1;
+    }
+
+    RTC_DCHECK_GT(channels, 0);
+    // 1 means muted
+    enabled = static_cast<bool>(muted);
+  }
+
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::MicrophoneMute() => enabled="
+                      << enabled;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneVolumeIsAvailable(bool& available) {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    available = true;
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err != noErr || !isSettable) {
+      available = false;
+      RTC_LOG(LS_WARNING) << "Volume cannot be set for input channel " << i
+                          << ", err=" << err;
+      return -1;
+    }
+  }
+
+  available = true;
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::SetMicrophoneVolume(volume="
+                      << volume << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  bool success = false;
+
+  // volume range is 0.0 - 1.0, convert from 0 - 255
+  const Float32 vol = (Float32)(volume / 255.0);
+
+  RTC_DCHECK(vol <= 1.0 && vol >= 0.0);
+
+  // Does the capture device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeInput, 0};
+  Boolean isSettable = false;
+  err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                      &isSettable);
+  if (err == noErr && isSettable) {
+    size = sizeof(vol);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, size, &vol));
+
+    return 0;
+  }
+
+  // Otherwise try to set each channel.
+  for (UInt32 i = 1; i <= _noInputChannels; i++) {
+    propertyAddress.mElement = i;
+    isSettable = false;
+    err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
+                                        &isSettable);
+    if (err == noErr && isSettable) {
+      size = sizeof(vol);
+      WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+          _inputDeviceID, &propertyAddress, 0, NULL, size, &vol));
+    }
+    success = true;
+  }
+
+  if (!success) {
+    RTC_LOG(LS_WARNING) << "Unable to set a level on any input channel";
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MicrophoneVolume(uint32_t& volume) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  unsigned int channels = 0;
+  Float32 channelVol = 0;
+  Float32 volFloat32 = 0;
+
+  // Does the device have a master volume control?
+  // If so, use it exclusively.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeInput, 0};
+  Boolean hasProperty =
+      AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+  if (hasProperty) {
+    size = sizeof(volFloat32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        _inputDeviceID, &propertyAddress, 0, NULL, &size, &volFloat32));
+
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(volFloat32 * 255 + 0.5);
+  } else {
+    // Otherwise get the average volume across channels.
+    volFloat32 = 0;
+    for (UInt32 i = 1; i <= _noInputChannels; i++) {
+      channelVol = 0;
+      propertyAddress.mElement = i;
+      hasProperty = AudioObjectHasProperty(_inputDeviceID, &propertyAddress);
+      if (hasProperty) {
+        size = sizeof(channelVol);
+        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+            _inputDeviceID, &propertyAddress, 0, NULL, &size, &channelVol));
+
+        volFloat32 += channelVol;
+        channels++;
+      }
+    }
+
+    if (channels == 0) {
+      RTC_LOG(LS_WARNING) << "Unable to get a level on any channel";
+      return -1;
+    }
+
+    RTC_DCHECK_GT(channels, 0);
+    // vol 0.0 to 1.0 -> convert to 0 - 255
+    volume = static_cast<uint32_t>(255 * volFloat32 / channels + 0.5);
+  }
+
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerMac::MicrophoneVolume() => vol="
+                      << volume;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 255
+  maxVolume = 255;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerMac::MinMicrophoneVolume(uint32_t& minVolume) const {
+  if (_inputDeviceID == kAudioObjectUnknown) {
+    RTC_LOG(LS_WARNING) << "device ID has not been set";
+    return -1;
+  }
+
+  // volume range is 0.0 to 1.0
+  // we convert that to 0 - 10
+  minVolume = 0;
+
+  return 0;
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+// CoreAudio errors are best interpreted as four character strings.
+void AudioMixerManagerMac::logCAMsg(const rtc::LoggingSeverity sev,
+                                    const char* msg,
+                                    const char* err) {
+  RTC_DCHECK(msg != NULL);
+  RTC_DCHECK(err != NULL);
+  RTC_DCHECK(sev == rtc::LS_ERROR || sev == rtc::LS_WARNING);
+
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
+  switch (sev) {
+    case rtc::LS_ERROR:
+      RTC_LOG(LS_ERROR) << msg << ": " << err[0] << err[1] << err[2] << err[3];
+      break;
+    case rtc::LS_WARNING:
+      RTC_LOG(LS_WARNING) << msg << ": " << err[0] << err[1] << err[2]
+                          << err[3];
+      break;
+    default:
+      break;
+  }
+#else
+  // We need to flip the characters in this case.
+  switch (sev) {
+    case rtc::LS_ERROR:
+      RTC_LOG(LS_ERROR) << msg << ": " << err[3] << err[2] << err[1] << err[0];
+      break;
+    case rtc::LS_WARNING:
+      RTC_LOG(LS_WARNING) << msg << ": " << err[3] << err[2] << err[1]
+                          << err[0];
+      break;
+    default:
+      break;
+  }
+#endif
+}
+
+}  // namespace webrtc
+// EOF
diff --git a/third_party/libwebrtc/modules/audio_device/mac/audio_mixer_manager_mac.h b/third_party/libwebrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
new file mode 100644
index 0000000000..0ccab4879b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H_
+#define AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H_
+
+#include <CoreAudio/CoreAudio.h>
+
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class AudioMixerManagerMac {
+ public:
+  int32_t OpenSpeaker(AudioDeviceID deviceID) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t OpenMicrophone(AudioDeviceID deviceID) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SetSpeakerVolume(uint32_t volume) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SpeakerVolume(uint32_t& volume) const;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  int32_t SpeakerVolumeIsAvailable(bool& available);
+  int32_t SpeakerMuteIsAvailable(bool& available);
+  int32_t SetSpeakerMute(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SpeakerMute(bool& enabled) const;
+  int32_t StereoPlayoutIsAvailable(bool& available);
+  int32_t StereoRecordingIsAvailable(bool& available);
+  int32_t MicrophoneMuteIsAvailable(bool& available);
+  int32_t SetMicrophoneMute(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t MicrophoneMute(bool& enabled) const;
+  int32_t MicrophoneVolumeIsAvailable(bool& available);
+  int32_t SetMicrophoneVolume(uint32_t volume) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t MicrophoneVolume(uint32_t& volume) const;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  int32_t Close() RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t CloseSpeaker() RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t CloseMicrophone() RTC_LOCKS_EXCLUDED(mutex_);
+  bool SpeakerIsInitialized() const;
+  bool MicrophoneIsInitialized() const;
+
+ public:
+  AudioMixerManagerMac();
+  ~AudioMixerManagerMac();
+
+ private:
+  int32_t CloseSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t CloseMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  static void logCAMsg(rtc::LoggingSeverity sev,
+                       const char* msg,
+                       const char* err);
+
+ private:
+  Mutex mutex_;
+
+  AudioDeviceID _inputDeviceID;
+  AudioDeviceID _outputDeviceID;
+
+  uint16_t _noInputChannels;
+  uint16_t _noOutputChannels;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_MIXER_MAC_H
diff --git a/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h b/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h
new file mode 100644
index 0000000000..b0f54c20ff
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
+#define MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioDeviceBuffer : public AudioDeviceBuffer {
+ public:
+  using AudioDeviceBuffer::AudioDeviceBuffer;
+  virtual ~MockAudioDeviceBuffer() {}
+  MOCK_METHOD(int32_t, RequestPlayoutData, (size_t nSamples), (override));
+  MOCK_METHOD(int32_t, GetPlayoutData, (void* audioBuffer), (override));
+  MOCK_METHOD(int32_t,
+              SetRecordedBuffer,
+              (const void* audioBuffer, size_t nSamples),
+              (override));
+  MOCK_METHOD(void, SetVQEData, (int playDelayMS, int recDelayMS), (override));
+  MOCK_METHOD(int32_t, DeliverRecordedData, (), (override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_device/win/audio_device_core_win.cc b/third_party/libwebrtc/modules/audio_device/win/audio_device_core_win.cc
new file mode 100644
index 0000000000..1e3a94edf6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/audio_device_core_win.cc
@@ -0,0 +1,4178 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#pragma warning(disable : 4995)  // name was marked as #pragma deprecated
+
+#if (_MSC_VER >= 1310) && (_MSC_VER < 1400)
+// Reports the major and minor versions of the compiler.
+// For example, 1310 for Microsoft Visual C++ .NET 2003. 1310 represents version
+// 13 and a 1.0 point release. The Visual C++ 2005 compiler version is 1400.
+// Type cl /? at the command line to see the major and minor versions of your
+// compiler along with the build number.
+#pragma message(">> INFO: Windows Core Audio is not supported in VS 2003")
+#endif
+
+#include "modules/audio_device/audio_device_config.h"
+
+#ifdef WEBRTC_WINDOWS_CORE_AUDIO_BUILD
+
+// clang-format off
+// To get Windows includes in the right order, this must come before the Windows
+// includes below.
+#include "modules/audio_device/win/audio_device_core_win.h"
+// clang-format on
+
+#include <string.h>
+
+#include <comdef.h>
+#include <dmo.h>
+#include <functiondiscoverykeys_devpkey.h>
+#include <mmsystem.h>
+#include <strsafe.h>
+#include <uuids.h>
+#include <windows.h>
+
+#include <iomanip>
+
+#include "api/make_ref_counted.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/sleep.h"
+
+// Macro that calls a COM method returning HRESULT value.
+#define EXIT_ON_ERROR(hres) \
+  do {                      \
+    if (FAILED(hres))       \
+      goto Exit;            \
+  } while (0)
+
+// Macro that continues to a COM error.
+#define CONTINUE_ON_ERROR(hres) \
+  do {                          \
+    if (FAILED(hres))           \
+      goto Next;                \
+  } while (0)
+
+// Macro that releases a COM object if not NULL.
+#define SAFE_RELEASE(p) \
+  do {                  \
+    if ((p)) {          \
+      (p)->Release();   \
+      (p) = NULL;       \
+    }                   \
+  } while (0)
+
+#define ROUND(x) ((x) >= 0 ? (int)((x) + 0.5) : (int)((x)-0.5))
+
+// REFERENCE_TIME time units per millisecond
+#define REFTIMES_PER_MILLISEC 10000
+
+typedef struct tagTHREADNAME_INFO {
+  DWORD dwType;      // must be 0x1000
+  LPCSTR szName;     // pointer to name (in user addr space)
+  DWORD dwThreadID;  // thread ID (-1=caller thread)
+  DWORD dwFlags;     // reserved for future use, must be zero
+} THREADNAME_INFO;
+
+namespace webrtc {
+namespace {
+
+enum { COM_THREADING_MODEL = COINIT_MULTITHREADED };
+
+enum { kAecCaptureStreamIndex = 0, kAecRenderStreamIndex = 1 };
+
+// An implementation of IMediaBuffer, as required for
+// IMediaObject::ProcessOutput(). After consuming data provided by
+// ProcessOutput(), call SetLength() to update the buffer availability.
+//
+// Example implementation:
+// http://msdn.microsoft.com/en-us/library/dd376684(v=vs.85).aspx
+class MediaBufferImpl final : public IMediaBuffer {
+ public:
+  explicit MediaBufferImpl(DWORD maxLength)
+      : _data(new BYTE[maxLength]),
+        _length(0),
+        _maxLength(maxLength),
+        _refCount(0) {}
+
+  // IMediaBuffer methods.
+  STDMETHOD(GetBufferAndLength(BYTE** ppBuffer, DWORD* pcbLength)) {
+    if (!ppBuffer || !pcbLength) {
+      return E_POINTER;
+    }
+
+    *ppBuffer = _data;
+    *pcbLength = _length;
+
+    return S_OK;
+  }
+
+  STDMETHOD(GetMaxLength(DWORD* pcbMaxLength)) {
+    if (!pcbMaxLength) {
+      return E_POINTER;
+    }
+
+    *pcbMaxLength = _maxLength;
+    return S_OK;
+  }
+
+  STDMETHOD(SetLength(DWORD cbLength)) {
+    if (cbLength > _maxLength) {
+      return E_INVALIDARG;
+    }
+
+    _length = cbLength;
+    return S_OK;
+  }
+
+  // IUnknown methods.
+  STDMETHOD_(ULONG, AddRef()) { return InterlockedIncrement(&_refCount); }
+
+  STDMETHOD(QueryInterface(REFIID riid, void** ppv)) {
+    if (!ppv) {
+      return E_POINTER;
+    } else if (riid != IID_IMediaBuffer && riid != IID_IUnknown) {
+      return E_NOINTERFACE;
+    }
+
+    *ppv = static_cast<IMediaBuffer*>(this);
+    AddRef();
+    return S_OK;
+  }
+
+  STDMETHOD_(ULONG, Release()) {
+    LONG refCount = InterlockedDecrement(&_refCount);
+    if (refCount == 0) {
+      delete this;
+    }
+
+    return refCount;
+  }
+
+ private:
+  ~MediaBufferImpl() { delete[] _data; }
+
+  BYTE* _data;
+  DWORD _length;
+  const DWORD _maxLength;
+  LONG _refCount;
+};
+}  // namespace
+
+// ============================================================================
+//                              Static Methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+//  CoreAudioIsSupported
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::CoreAudioIsSupported() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  bool MMDeviceIsAvailable(false);
+  bool coreAudioIsSupported(false);
+
+  HRESULT hr(S_OK);
+  wchar_t buf[MAXERRORLENGTH];
+  wchar_t errorText[MAXERRORLENGTH];
+
+  // 1) Check if Windows version is Vista SP1 or later.
+  //
+  // CoreAudio is only available on Vista SP1 and later.
+  //
+  OSVERSIONINFOEX osvi;
+  DWORDLONG dwlConditionMask = 0;
+  int op = VER_LESS_EQUAL;
+
+  // Initialize the OSVERSIONINFOEX structure.
+  ZeroMemory(&osvi, sizeof(OSVERSIONINFOEX));
+  osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
+  osvi.dwMajorVersion = 6;
+  osvi.dwMinorVersion = 0;
+  osvi.wServicePackMajor = 0;
+  osvi.wServicePackMinor = 0;
+  osvi.wProductType = VER_NT_WORKSTATION;
+
+  // Initialize the condition mask.
+  VER_SET_CONDITION(dwlConditionMask, VER_MAJORVERSION, op);
+  VER_SET_CONDITION(dwlConditionMask, VER_MINORVERSION, op);
+  VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMAJOR, op);
+  VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMINOR, op);
+  VER_SET_CONDITION(dwlConditionMask, VER_PRODUCT_TYPE, VER_EQUAL);
+
+  DWORD dwTypeMask = VER_MAJORVERSION | VER_MINORVERSION |
+                     VER_SERVICEPACKMAJOR | VER_SERVICEPACKMINOR |
+                     VER_PRODUCT_TYPE;
+
+  // Perform the test.
+  BOOL isVistaRTMorXP = VerifyVersionInfo(&osvi, dwTypeMask, dwlConditionMask);
+  if (isVistaRTMorXP != 0) {
+    RTC_LOG(LS_VERBOSE)
+        << "*** Windows Core Audio is only supported on Vista SP1 or later";
+    return false;
+  }
+
+  // 2) Initializes the COM library for use by the calling thread.
+
+  // The COM init wrapper sets the thread's concurrency model to MTA,
+  // and creates a new apartment for the thread if one is required. The
+  // wrapper also ensures that each call to CoInitializeEx is balanced
+  // by a corresponding call to CoUninitialize.
+  //
+  ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+  if (!comInit.Succeeded()) {
+    // Things will work even if an STA thread is calling this method but we
+    // want to ensure that MTA is used and therefore return false here.
+    return false;
+  }
+
+  // 3) Check if the MMDevice API is available.
+  //
+  // The Windows Multimedia Device (MMDevice) API enables audio clients to
+  // discover audio endpoint devices, determine their capabilities, and create
+  // driver instances for those devices.
+  // Header file Mmdeviceapi.h defines the interfaces in the MMDevice API.
+  // The MMDevice API consists of several interfaces. The first of these is the
+  // IMMDeviceEnumerator interface. To access the interfaces in the MMDevice
+  // API, a client obtains a reference to the IMMDeviceEnumerator interface of a
+  // device-enumerator object by calling the CoCreateInstance function.
+  //
+  // Through the IMMDeviceEnumerator interface, the client can obtain references
+  // to the other interfaces in the MMDevice API. The MMDevice API implements
+  // the following interfaces:
+  //
+  // IMMDevice            Represents an audio device.
+  // IMMDeviceCollection  Represents a collection of audio devices.
+  // IMMDeviceEnumerator  Provides methods for enumerating audio devices.
+  // IMMEndpoint          Represents an audio endpoint device.
+  //
+  IMMDeviceEnumerator* pIMMD(NULL);
+  const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
+  const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
+
+  hr = CoCreateInstance(
+      CLSID_MMDeviceEnumerator,  // GUID value of MMDeviceEnumerator coclass
+      NULL, CLSCTX_ALL,
+      IID_IMMDeviceEnumerator,  // GUID value of the IMMDeviceEnumerator
+                                // interface
+      (void**)&pIMMD);
+
+  if (FAILED(hr)) {
+    RTC_LOG(LS_ERROR) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+                         " Failed to create the required COM object (hr="
+                      << hr << ")";
+    RTC_LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+                           " CoCreateInstance(MMDeviceEnumerator) failed (hr="
+                        << hr << ")";
+
+    const DWORD dwFlags =
+        FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_IGNORE_INSERTS;
+    const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
+
+    // Gets the system's human readable message string for this HRESULT.
+    // All error message in English by default.
+    DWORD messageLength = ::FormatMessageW(dwFlags, 0, hr, dwLangID, errorText,
+                                           MAXERRORLENGTH, NULL);
+
+    RTC_DCHECK_LE(messageLength, MAXERRORLENGTH);
+
+    // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
+    for (; messageLength && ::isspace(errorText[messageLength - 1]);
+         --messageLength) {
+      errorText[messageLength - 1] = '\0';
+    }
+
+    StringCchPrintfW(buf, MAXERRORLENGTH, L"Error details: ");
+    StringCchCatW(buf, MAXERRORLENGTH, errorText);
+    RTC_LOG(LS_VERBOSE) << buf;
+  } else {
+    MMDeviceIsAvailable = true;
+    RTC_LOG(LS_VERBOSE)
+        << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
+           " CoCreateInstance(MMDeviceEnumerator) succeeded (hr="
+        << hr << ")";
+    SAFE_RELEASE(pIMMD);
+  }
+
+  // 4) Verify that we can create and initialize our Core Audio class.
+  //
+  if (MMDeviceIsAvailable) {
+    coreAudioIsSupported = false;
+
+    AudioDeviceWindowsCore* p = new (std::nothrow) AudioDeviceWindowsCore();
+    if (p == NULL) {
+      return false;
+    }
+
+    int ok(0);
+
+    if (p->Init() != InitStatus::OK) {
+      ok |= -1;
+    }
+
+    ok |= p->Terminate();
+
+    if (ok == 0) {
+      coreAudioIsSupported = true;
+    }
+
+    delete p;
+  }
+
+  if (coreAudioIsSupported) {
+    RTC_LOG(LS_VERBOSE) << "*** Windows Core Audio is supported ***";
+  } else {
+    RTC_LOG(LS_VERBOSE) << "*** Windows Core Audio is NOT supported";
+  }
+
+  return (coreAudioIsSupported);
+}
+
+// ============================================================================
+//                            Construction & Destruction
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+//  AudioDeviceWindowsCore() - ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceWindowsCore::AudioDeviceWindowsCore()
+    : _avrtLibrary(nullptr),
+      _winSupportAvrt(false),
+      _comInit(ScopedCOMInitializer::kMTA),
+      _ptrAudioBuffer(nullptr),
+      _ptrEnumerator(nullptr),
+      _ptrRenderCollection(nullptr),
+      _ptrCaptureCollection(nullptr),
+      _ptrDeviceOut(nullptr),
+      _ptrDeviceIn(nullptr),
+      _ptrClientOut(nullptr),
+      _ptrClientIn(nullptr),
+      _ptrRenderClient(nullptr),
+      _ptrCaptureClient(nullptr),
+      _ptrCaptureVolume(nullptr),
+      _ptrRenderSimpleVolume(nullptr),
+      _dmo(nullptr),
+      _mediaBuffer(nullptr),
+      _builtInAecEnabled(false),
+      _hRenderSamplesReadyEvent(nullptr),
+      _hPlayThread(nullptr),
+      _hRenderStartedEvent(nullptr),
+      _hShutdownRenderEvent(nullptr),
+      _hCaptureSamplesReadyEvent(nullptr),
+      _hRecThread(nullptr),
+      _hCaptureStartedEvent(nullptr),
+      _hShutdownCaptureEvent(nullptr),
+      _hMmTask(nullptr),
+      _playAudioFrameSize(0),
+      _playSampleRate(0),
+      _playBlockSize(0),
+      _playChannels(2),
+      _sndCardPlayDelay(0),
+      _writtenSamples(0),
+      _readSamples(0),
+      _recAudioFrameSize(0),
+      _recSampleRate(0),
+      _recBlockSize(0),
+      _recChannels(2),
+      _initialized(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _speakerIsInitialized(false),
+      _microphoneIsInitialized(false),
+      _usingInputDeviceIndex(false),
+      _usingOutputDeviceIndex(false),
+      _inputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
+      _outputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
+      _inputDeviceIndex(0),
+      _outputDeviceIndex(0) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+  RTC_DCHECK(_comInit.Succeeded());
+
+  // Try to load the Avrt DLL
+  if (!_avrtLibrary) {
+    // Get handle to the Avrt DLL module.
+    _avrtLibrary = LoadLibrary(TEXT("Avrt.dll"));
+    if (_avrtLibrary) {
+      // Handle is valid (should only happen if OS larger than vista & win7).
+      // Try to get the function addresses.
+      RTC_LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
+                             " The Avrt DLL module is now loaded";
+
+      _PAvRevertMmThreadCharacteristics =
+          (PAvRevertMmThreadCharacteristics)GetProcAddress(
+              _avrtLibrary, "AvRevertMmThreadCharacteristics");
+      _PAvSetMmThreadCharacteristicsA =
+          (PAvSetMmThreadCharacteristicsA)GetProcAddress(
+              _avrtLibrary, "AvSetMmThreadCharacteristicsA");
+      _PAvSetMmThreadPriority = (PAvSetMmThreadPriority)GetProcAddress(
+          _avrtLibrary, "AvSetMmThreadPriority");
+
+      if (_PAvRevertMmThreadCharacteristics &&
+          _PAvSetMmThreadCharacteristicsA && _PAvSetMmThreadPriority) {
+        RTC_LOG(LS_VERBOSE)
+            << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
+               " AvRevertMmThreadCharacteristics() is OK";
+        RTC_LOG(LS_VERBOSE)
+            << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
+               " AvSetMmThreadCharacteristicsA() is OK";
+        RTC_LOG(LS_VERBOSE)
+            << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
+               " AvSetMmThreadPriority() is OK";
+        _winSupportAvrt = true;
+      }
+    }
+  }
+
+  // Create our samples ready events - we want auto reset events that start in
+  // the not-signaled state. The state of an auto-reset event object remains
+  // signaled until a single waiting thread is released, at which time the
+  // system automatically sets the state to nonsignaled. If no threads are
+  // waiting, the event object's state remains signaled. (Except for
+  // _hShutdownCaptureEvent, which is used to shutdown multiple threads).
+  _hRenderSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+  _hCaptureSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+  _hShutdownRenderEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+  _hShutdownCaptureEvent = CreateEvent(NULL, TRUE, FALSE, NULL);
+  _hRenderStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+  _hCaptureStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
+
+  _perfCounterFreq.QuadPart = 1;
+  _perfCounterFactor = 0.0;
+
+  // list of number of channels to use on recording side
+  _recChannelsPrioList[0] = 2;  // stereo is prio 1
+  _recChannelsPrioList[1] = 1;  // mono is prio 2
+  _recChannelsPrioList[2] = 4;  // quad is prio 3
+
+  // list of number of channels to use on playout side
+  _playChannelsPrioList[0] = 2;  // stereo is prio 1
+  _playChannelsPrioList[1] = 1;  // mono is prio 2
+
+  HRESULT hr;
+
+  // We know that this API will work since it has already been verified in
+  // CoreAudioIsSupported, hence no need to check for errors here as well.
+
+  // Retrive the IMMDeviceEnumerator API (should load the MMDevAPI.dll)
+  // TODO(henrika): we should probably move this allocation to Init() instead
+  // and deallocate in Terminate() to make the implementation more symmetric.
+  CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL,
+                   __uuidof(IMMDeviceEnumerator),
+                   reinterpret_cast<void**>(&_ptrEnumerator));
+  RTC_DCHECK(_ptrEnumerator);
+
+  // DMO initialization for built-in WASAPI AEC.
+  {
+    IMediaObject* ptrDMO = NULL;
+    hr = CoCreateInstance(CLSID_CWMAudioAEC, NULL, CLSCTX_INPROC_SERVER,
+                          IID_IMediaObject, reinterpret_cast<void**>(&ptrDMO));
+    if (FAILED(hr) || ptrDMO == NULL) {
+      // Since we check that _dmo is non-NULL in EnableBuiltInAEC(), the
+      // feature is prevented from being enabled.
+      _builtInAecEnabled = false;
+      _TraceCOMError(hr);
+    }
+    _dmo = ptrDMO;
+    SAFE_RELEASE(ptrDMO);
+  }
+}
+
+// ----------------------------------------------------------------------------
+//  AudioDeviceWindowsCore() - dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceWindowsCore::~AudioDeviceWindowsCore() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+
+  Terminate();
+
+  // The IMMDeviceEnumerator is created during construction. Must release
+  // it here and not in Terminate() since we don't recreate it in Init().
+  SAFE_RELEASE(_ptrEnumerator);
+
+  _ptrAudioBuffer = NULL;
+
+  if (NULL != _hRenderSamplesReadyEvent) {
+    CloseHandle(_hRenderSamplesReadyEvent);
+    _hRenderSamplesReadyEvent = NULL;
+  }
+
+  if (NULL != _hCaptureSamplesReadyEvent) {
+    CloseHandle(_hCaptureSamplesReadyEvent);
+    _hCaptureSamplesReadyEvent = NULL;
+  }
+
+  if (NULL != _hRenderStartedEvent) {
+    CloseHandle(_hRenderStartedEvent);
+    _hRenderStartedEvent = NULL;
+  }
+
+  if (NULL != _hCaptureStartedEvent) {
+    CloseHandle(_hCaptureStartedEvent);
+    _hCaptureStartedEvent = NULL;
+  }
+
+  if (NULL != _hShutdownRenderEvent) {
+    CloseHandle(_hShutdownRenderEvent);
+    _hShutdownRenderEvent = NULL;
+  }
+
+  if (NULL != _hShutdownCaptureEvent) {
+    CloseHandle(_hShutdownCaptureEvent);
+    _hShutdownCaptureEvent = NULL;
+  }
+
+  if (_avrtLibrary) {
+    BOOL freeOK = FreeLibrary(_avrtLibrary);
+    if (!freeOK) {
+      RTC_LOG(LS_WARNING)
+          << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
+             " failed to free the loaded Avrt DLL module correctly";
+    } else {
+      RTC_LOG(LS_WARNING) << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
+                             " the Avrt DLL module is now unloaded";
+    }
+  }
+}
+
+// ============================================================================
+//                                     API
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+//  AttachAudioBuffer
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  _ptrAudioBuffer = audioBuffer;
+
+  // Inform the AudioBuffer about default settings for this implementation.
+  // Set all values to zero here since the actual settings will be done by
+  // InitPlayout and InitRecording later.
+  _ptrAudioBuffer->SetRecordingSampleRate(0);
+  _ptrAudioBuffer->SetPlayoutSampleRate(0);
+  _ptrAudioBuffer->SetRecordingChannels(0);
+  _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+// ----------------------------------------------------------------------------
+//  ActiveAudioLayer
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kWindowsCoreAudio;
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  Init
+// ----------------------------------------------------------------------------
+
+AudioDeviceGeneric::InitStatus AudioDeviceWindowsCore::Init() {
+  MutexLock lock(&mutex_);
+
+  if (_initialized) {
+    return InitStatus::OK;
+  }
+
+  // Enumerate all audio rendering and capturing endpoint devices.
+  // Note that, some of these will not be able to select by the user.
+  // The complete collection is for internal use only.
+  _EnumerateEndpointDevicesAll(eRender);
+  _EnumerateEndpointDevicesAll(eCapture);
+
+  _initialized = true;
+
+  return InitStatus::OK;
+}
+
+// ----------------------------------------------------------------------------
+//  Terminate
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::Terminate() {
+  MutexLock lock(&mutex_);
+
+  if (!_initialized) {
+    return 0;
+  }
+
+  _initialized = false;
+  _speakerIsInitialized = false;
+  _microphoneIsInitialized = false;
+  _playing = false;
+  _recording = false;
+
+  SAFE_RELEASE(_ptrRenderCollection);
+  SAFE_RELEASE(_ptrCaptureCollection);
+  SAFE_RELEASE(_ptrDeviceOut);
+  SAFE_RELEASE(_ptrDeviceIn);
+  SAFE_RELEASE(_ptrClientOut);
+  SAFE_RELEASE(_ptrClientIn);
+  SAFE_RELEASE(_ptrRenderClient);
+  SAFE_RELEASE(_ptrCaptureClient);
+  SAFE_RELEASE(_ptrCaptureVolume);
+  SAFE_RELEASE(_ptrRenderSimpleVolume);
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  Initialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::Initialized() const {
+  return (_initialized);
+}
+
+// ----------------------------------------------------------------------------
+//  InitSpeaker
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::InitSpeaker() {
+  MutexLock lock(&mutex_);
+  return InitSpeakerLocked();
+}
+
+int32_t AudioDeviceWindowsCore::InitSpeakerLocked() {
+  if (_playing) {
+    return -1;
+  }
+
+  if (_ptrDeviceOut == NULL) {
+    return -1;
+  }
+
+  if (_usingOutputDeviceIndex) {
+    int16_t nDevices = PlayoutDevicesLocked();
+    if (_outputDeviceIndex > (nDevices - 1)) {
+      RTC_LOG(LS_ERROR) << "current device selection is invalid => unable to"
+                           " initialize";
+      return -1;
+    }
+  }
+
+  int32_t ret(0);
+
+  SAFE_RELEASE(_ptrDeviceOut);
+  if (_usingOutputDeviceIndex) {
+    // Refresh the selected rendering endpoint device using current index
+    ret = _GetListDevice(eRender, _outputDeviceIndex, &_ptrDeviceOut);
+  } else {
+    ERole role;
+    (_outputDevice == AudioDeviceModule::kDefaultDevice)
+        ? role = eConsole
+        : role = eCommunications;
+    // Refresh the selected rendering endpoint device using role
+    ret = _GetDefaultDevice(eRender, role, &_ptrDeviceOut);
+  }
+
+  if (ret != 0 || (_ptrDeviceOut == NULL)) {
+    RTC_LOG(LS_ERROR) << "failed to initialize the rendering enpoint device";
+    SAFE_RELEASE(_ptrDeviceOut);
+    return -1;
+  }
+
+  IAudioSessionManager* pManager = NULL;
+  ret = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager), CLSCTX_ALL,
+                                NULL, (void**)&pManager);
+  if (ret != 0 || pManager == NULL) {
+    RTC_LOG(LS_ERROR) << "failed to initialize the render manager";
+    SAFE_RELEASE(pManager);
+    return -1;
+  }
+
+  SAFE_RELEASE(_ptrRenderSimpleVolume);
+  ret = pManager->GetSimpleAudioVolume(NULL, FALSE, &_ptrRenderSimpleVolume);
+  if (ret != 0 || _ptrRenderSimpleVolume == NULL) {
+    RTC_LOG(LS_ERROR) << "failed to initialize the render simple volume";
+    SAFE_RELEASE(pManager);
+    SAFE_RELEASE(_ptrRenderSimpleVolume);
+    return -1;
+  }
+  SAFE_RELEASE(pManager);
+
+  _speakerIsInitialized = true;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  InitMicrophone
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::InitMicrophone() {
+  MutexLock lock(&mutex_);
+  return InitMicrophoneLocked();
+}
+
+int32_t AudioDeviceWindowsCore::InitMicrophoneLocked() {
+  if (_recording) {
+    return -1;
+  }
+
+  if (_ptrDeviceIn == NULL) {
+    return -1;
+  }
+
+  if (_usingInputDeviceIndex) {
+    int16_t nDevices = RecordingDevicesLocked();
+    if (_inputDeviceIndex > (nDevices - 1)) {
+      RTC_LOG(LS_ERROR) << "current device selection is invalid => unable to"
+                           " initialize";
+      return -1;
+    }
+  }
+
+  int32_t ret(0);
+
+  SAFE_RELEASE(_ptrDeviceIn);
+  if (_usingInputDeviceIndex) {
+    // Refresh the selected capture endpoint device using current index
+    ret = _GetListDevice(eCapture, _inputDeviceIndex, &_ptrDeviceIn);
+  } else {
+    ERole role;
+    (_inputDevice == AudioDeviceModule::kDefaultDevice)
+        ? role = eConsole
+        : role = eCommunications;
+    // Refresh the selected capture endpoint device using role
+    ret = _GetDefaultDevice(eCapture, role, &_ptrDeviceIn);
+  }
+
+  if (ret != 0 || (_ptrDeviceIn == NULL)) {
+    RTC_LOG(LS_ERROR) << "failed to initialize the capturing enpoint device";
+    SAFE_RELEASE(_ptrDeviceIn);
+    return -1;
+  }
+
+  SAFE_RELEASE(_ptrCaptureVolume);
+  ret = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                               reinterpret_cast<void**>(&_ptrCaptureVolume));
+  if (ret != 0 || _ptrCaptureVolume == NULL) {
+    RTC_LOG(LS_ERROR) << "failed to initialize the capture volume";
+    SAFE_RELEASE(_ptrCaptureVolume);
+    return -1;
+  }
+
+  _microphoneIsInitialized = true;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SpeakerIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::SpeakerIsInitialized() const {
+  return (_speakerIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+//  MicrophoneIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::MicrophoneIsInitialized() const {
+  return (_microphoneIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+//  SpeakerVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SpeakerVolumeIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+
+  if (_ptrDeviceOut == NULL) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+  IAudioSessionManager* pManager = NULL;
+  ISimpleAudioVolume* pVolume = NULL;
+
+  hr = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager), CLSCTX_ALL, NULL,
+                               (void**)&pManager);
+  EXIT_ON_ERROR(hr);
+
+  hr = pManager->GetSimpleAudioVolume(NULL, FALSE, &pVolume);
+  EXIT_ON_ERROR(hr);
+
+  float volume(0.0f);
+  hr = pVolume->GetMasterVolume(&volume);
+  if (FAILED(hr)) {
+    available = false;
+  }
+  available = true;
+
+  SAFE_RELEASE(pManager);
+  SAFE_RELEASE(pVolume);
+
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  SAFE_RELEASE(pManager);
+  SAFE_RELEASE(pVolume);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  SetSpeakerVolume
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetSpeakerVolume(uint32_t volume) {
+  {
+    MutexLock lock(&mutex_);
+
+    if (!_speakerIsInitialized) {
+      return -1;
+    }
+
+    if (_ptrDeviceOut == NULL) {
+      return -1;
+    }
+  }
+
+  if (volume < (uint32_t)MIN_CORE_SPEAKER_VOLUME ||
+      volume > (uint32_t)MAX_CORE_SPEAKER_VOLUME) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+
+  // scale input volume to valid range (0.0 to 1.0)
+  const float fLevel = (float)volume / MAX_CORE_SPEAKER_VOLUME;
+  volume_mutex_.Lock();
+  hr = _ptrRenderSimpleVolume->SetMasterVolume(fLevel, NULL);
+  volume_mutex_.Unlock();
+  EXIT_ON_ERROR(hr);
+
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  SpeakerVolume
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SpeakerVolume(uint32_t& volume) const {
+  {
+    MutexLock lock(&mutex_);
+
+    if (!_speakerIsInitialized) {
+      return -1;
+    }
+
+    if (_ptrDeviceOut == NULL) {
+      return -1;
+    }
+  }
+
+  HRESULT hr = S_OK;
+  float fLevel(0.0f);
+
+  volume_mutex_.Lock();
+  hr = _ptrRenderSimpleVolume->GetMasterVolume(&fLevel);
+  volume_mutex_.Unlock();
+  EXIT_ON_ERROR(hr);
+
+  // scale input volume range [0.0,1.0] to valid output range
+  volume = static_cast<uint32_t>(fLevel * MAX_CORE_SPEAKER_VOLUME);
+
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  MaxSpeakerVolume
+//
+//  The internal range for Core Audio is 0.0 to 1.0, where 0.0 indicates
+//  silence and 1.0 indicates full volume (no attenuation).
+//  We add our (webrtc-internal) own max level to match the Wave API and
+//  how it is used today in VoE.
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  if (!_speakerIsInitialized) {
+    return -1;
+  }
+
+  maxVolume = static_cast<uint32_t>(MAX_CORE_SPEAKER_VOLUME);
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  MinSpeakerVolume
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::MinSpeakerVolume(uint32_t& minVolume) const {
+  if (!_speakerIsInitialized) {
+    return -1;
+  }
+
+  minVolume = static_cast<uint32_t>(MIN_CORE_SPEAKER_VOLUME);
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SpeakerMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SpeakerMuteIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+
+  if (_ptrDeviceOut == NULL) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+  IAudioEndpointVolume* pVolume = NULL;
+
+  // Query the speaker system mute state.
+  hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                               reinterpret_cast<void**>(&pVolume));
+  EXIT_ON_ERROR(hr);
+
+  BOOL mute;
+  hr = pVolume->GetMute(&mute);
+  if (FAILED(hr))
+    available = false;
+  else
+    available = true;
+
+  SAFE_RELEASE(pVolume);
+
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  SAFE_RELEASE(pVolume);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  SetSpeakerMute
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetSpeakerMute(bool enable) {
+  MutexLock lock(&mutex_);
+
+  if (!_speakerIsInitialized) {
+    return -1;
+  }
+
+  if (_ptrDeviceOut == NULL) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+  IAudioEndpointVolume* pVolume = NULL;
+
+  // Set the speaker system mute state.
+  hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                               reinterpret_cast<void**>(&pVolume));
+  EXIT_ON_ERROR(hr);
+
+  const BOOL mute(enable);
+  hr = pVolume->SetMute(mute, NULL);
+  EXIT_ON_ERROR(hr);
+
+  SAFE_RELEASE(pVolume);
+
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  SAFE_RELEASE(pVolume);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  SpeakerMute
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SpeakerMute(bool& enabled) const {
+  if (!_speakerIsInitialized) {
+    return -1;
+  }
+
+  if (_ptrDeviceOut == NULL) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+  IAudioEndpointVolume* pVolume = NULL;
+
+  // Query the speaker system mute state.
+  hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                               reinterpret_cast<void**>(&pVolume));
+  EXIT_ON_ERROR(hr);
+
+  BOOL mute;
+  hr = pVolume->GetMute(&mute);
+  EXIT_ON_ERROR(hr);
+
+  enabled = (mute == TRUE) ? true : false;
+
+  SAFE_RELEASE(pVolume);
+
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  SAFE_RELEASE(pVolume);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  MicrophoneMuteIsAvailable
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::MicrophoneMuteIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+
+  if (_ptrDeviceIn == NULL) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+  IAudioEndpointVolume* pVolume = NULL;
+
+  // Query the microphone system mute state.
+  hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                              reinterpret_cast<void**>(&pVolume));
+  EXIT_ON_ERROR(hr);
+
+  BOOL mute;
+  hr = pVolume->GetMute(&mute);
+  if (FAILED(hr))
+    available = false;
+  else
+    available = true;
+
+  SAFE_RELEASE(pVolume);
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  SAFE_RELEASE(pVolume);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  SetMicrophoneMute
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetMicrophoneMute(bool enable) {
+  if (!_microphoneIsInitialized) {
+    return -1;
+  }
+
+  if (_ptrDeviceIn == NULL) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+  IAudioEndpointVolume* pVolume = NULL;
+
+  // Set the microphone system mute state.
+  hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                              reinterpret_cast<void**>(&pVolume));
+  EXIT_ON_ERROR(hr);
+
+  const BOOL mute(enable);
+  hr = pVolume->SetMute(mute, NULL);
+  EXIT_ON_ERROR(hr);
+
+  SAFE_RELEASE(pVolume);
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  SAFE_RELEASE(pVolume);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  MicrophoneMute
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::MicrophoneMute(bool& enabled) const {
+  if (!_microphoneIsInitialized) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+  IAudioEndpointVolume* pVolume = NULL;
+
+  // Query the microphone system mute state.
+  hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                              reinterpret_cast<void**>(&pVolume));
+  EXIT_ON_ERROR(hr);
+
+  BOOL mute;
+  hr = pVolume->GetMute(&mute);
+  EXIT_ON_ERROR(hr);
+
+  enabled = (mute == TRUE) ? true : false;
+
+  SAFE_RELEASE(pVolume);
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  SAFE_RELEASE(pVolume);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  StereoRecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::StereoRecordingIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetStereoRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetStereoRecording(bool enable) {
+  MutexLock lock(&mutex_);
+
+  if (enable) {
+    _recChannelsPrioList[0] = 2;  // try stereo first
+    _recChannelsPrioList[1] = 1;
+    _recChannels = 2;
+  } else {
+    _recChannelsPrioList[0] = 1;  // try mono first
+    _recChannelsPrioList[1] = 2;
+    _recChannels = 1;
+  }
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  StereoRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::StereoRecording(bool& enabled) const {
+  if (_recChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  StereoPlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::StereoPlayoutIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetStereoPlayout
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetStereoPlayout(bool enable) {
+  MutexLock lock(&mutex_);
+
+  if (enable) {
+    _playChannelsPrioList[0] = 2;  // try stereo first
+    _playChannelsPrioList[1] = 1;
+    _playChannels = 2;
+  } else {
+    _playChannelsPrioList[0] = 1;  // try mono first
+    _playChannelsPrioList[1] = 2;
+    _playChannels = 1;
+  }
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  StereoPlayout
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::StereoPlayout(bool& enabled) const {
+  if (_playChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  MicrophoneVolumeIsAvailable
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::MicrophoneVolumeIsAvailable(bool& available) {
+  MutexLock lock(&mutex_);
+
+  if (_ptrDeviceIn == NULL) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+  IAudioEndpointVolume* pVolume = NULL;
+
+  hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                              reinterpret_cast<void**>(&pVolume));
+  EXIT_ON_ERROR(hr);
+
+  float volume(0.0f);
+  hr = pVolume->GetMasterVolumeLevelScalar(&volume);
+  if (FAILED(hr)) {
+    available = false;
+  }
+  available = true;
+
+  SAFE_RELEASE(pVolume);
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  SAFE_RELEASE(pVolume);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  SetMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::SetMicrophoneVolume(volume="
+                      << volume << ")";
+
+  {
+    MutexLock lock(&mutex_);
+
+    if (!_microphoneIsInitialized) {
+      return -1;
+    }
+
+    if (_ptrDeviceIn == NULL) {
+      return -1;
+    }
+  }
+
+  if (volume < static_cast<uint32_t>(MIN_CORE_MICROPHONE_VOLUME) ||
+      volume > static_cast<uint32_t>(MAX_CORE_MICROPHONE_VOLUME)) {
+    return -1;
+  }
+
+  HRESULT hr = S_OK;
+  // scale input volume to valid range (0.0 to 1.0)
+  const float fLevel = static_cast<float>(volume) / MAX_CORE_MICROPHONE_VOLUME;
+  volume_mutex_.Lock();
+  _ptrCaptureVolume->SetMasterVolumeLevelScalar(fLevel, NULL);
+  volume_mutex_.Unlock();
+  EXIT_ON_ERROR(hr);
+
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  MicrophoneVolume
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::MicrophoneVolume(uint32_t& volume) const {
+  {
+    MutexLock lock(&mutex_);
+
+    if (!_microphoneIsInitialized) {
+      return -1;
+    }
+
+    if (_ptrDeviceIn == NULL) {
+      return -1;
+    }
+  }
+
+  HRESULT hr = S_OK;
+  float fLevel(0.0f);
+  volume = 0;
+  volume_mutex_.Lock();
+  hr = _ptrCaptureVolume->GetMasterVolumeLevelScalar(&fLevel);
+  volume_mutex_.Unlock();
+  EXIT_ON_ERROR(hr);
+
+  // scale input volume range [0.0,1.0] to valid output range
+  volume = static_cast<uint32_t>(fLevel * MAX_CORE_MICROPHONE_VOLUME);
+
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  MaxMicrophoneVolume
+//
+//  The internal range for Core Audio is 0.0 to 1.0, where 0.0 indicates
+//  silence and 1.0 indicates full volume (no attenuation).
+//  We add our (webrtc-internal) own max level to match the Wave API and
+//  how it is used today in VoE.
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  if (!_microphoneIsInitialized) {
+    return -1;
+  }
+
+  maxVolume = static_cast<uint32_t>(MAX_CORE_MICROPHONE_VOLUME);
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  MinMicrophoneVolume
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::MinMicrophoneVolume(uint32_t& minVolume) const {
+  if (!_microphoneIsInitialized) {
+    return -1;
+  }
+
+  minVolume = static_cast<uint32_t>(MIN_CORE_MICROPHONE_VOLUME);
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  PlayoutDevices
+// ----------------------------------------------------------------------------
+int16_t AudioDeviceWindowsCore::PlayoutDevices() {
+  MutexLock lock(&mutex_);
+  return PlayoutDevicesLocked();
+}
+
+int16_t AudioDeviceWindowsCore::PlayoutDevicesLocked() {
+  if (_RefreshDeviceList(eRender) != -1) {
+    return (_DeviceListCount(eRender));
+  }
+
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  SetPlayoutDevice I (II)
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetPlayoutDevice(uint16_t index) {
+  if (_playIsInitialized) {
+    return -1;
+  }
+
+  // Get current number of available rendering endpoint devices and refresh the
+  // rendering collection.
+  UINT nDevices = PlayoutDevices();
+
+  if (index < 0 || index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  MutexLock lock(&mutex_);
+
+  HRESULT hr(S_OK);
+
+  RTC_DCHECK(_ptrRenderCollection);
+
+  //  Select an endpoint rendering device given the specified index
+  SAFE_RELEASE(_ptrDeviceOut);
+  hr = _ptrRenderCollection->Item(index, &_ptrDeviceOut);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(_ptrDeviceOut);
+    return -1;
+  }
+
+  WCHAR szDeviceName[MAX_PATH];
+  const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
+
+  // Get the endpoint device's friendly-name
+  if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0) {
+    RTC_LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
+  }
+
+  _usingOutputDeviceIndex = true;
+  _outputDeviceIndex = index;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetPlayoutDevice II (II)
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  if (_playIsInitialized) {
+    return -1;
+  }
+
+  ERole role(eCommunications);
+
+  if (device == AudioDeviceModule::kDefaultDevice) {
+    role = eConsole;
+  } else if (device == AudioDeviceModule::kDefaultCommunicationDevice) {
+    role = eCommunications;
+  }
+
+  MutexLock lock(&mutex_);
+
+  // Refresh the list of rendering endpoint devices
+  _RefreshDeviceList(eRender);
+
+  HRESULT hr(S_OK);
+
+  RTC_DCHECK(_ptrEnumerator);
+
+  //  Select an endpoint rendering device given the specified role
+  SAFE_RELEASE(_ptrDeviceOut);
+  hr = _ptrEnumerator->GetDefaultAudioEndpoint(eRender, role, &_ptrDeviceOut);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(_ptrDeviceOut);
+    return -1;
+  }
+
+  WCHAR szDeviceName[MAX_PATH];
+  const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
+
+  // Get the endpoint device's friendly-name
+  if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0) {
+    RTC_LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
+  }
+
+  _usingOutputDeviceIndex = false;
+  _outputDevice = device;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  PlayoutDeviceName
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::PlayoutDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  bool defaultCommunicationDevice(false);
+  const int16_t nDevices(PlayoutDevices());  // also updates the list of devices
+
+  // Special fix for the case when the user selects '-1' as index (<=> Default
+  // Communication Device)
+  if (index == (uint16_t)(-1)) {
+    defaultCommunicationDevice = true;
+    index = 0;
+    RTC_LOG(LS_VERBOSE) << "Default Communication endpoint device will be used";
+  }
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  MutexLock lock(&mutex_);
+
+  int32_t ret(-1);
+  WCHAR szDeviceName[MAX_PATH];
+  const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
+
+  // Get the endpoint device's friendly-name
+  if (defaultCommunicationDevice) {
+    ret = _GetDefaultDeviceName(eRender, eCommunications, szDeviceName,
+                                bufferLen);
+  } else {
+    ret = _GetListDeviceName(eRender, index, szDeviceName, bufferLen);
+  }
+
+  if (ret == 0) {
+    // Convert the endpoint device's friendly-name to UTF-8
+    if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name,
+                            kAdmMaxDeviceNameSize, NULL, NULL) == 0) {
+      RTC_LOG(LS_ERROR)
+          << "WideCharToMultiByte(CP_UTF8) failed with error code "
+          << GetLastError();
+    }
+  }
+
+  // Get the endpoint ID string (uniquely identifies the device among all audio
+  // endpoint devices)
+  if (defaultCommunicationDevice) {
+    ret =
+        _GetDefaultDeviceID(eRender, eCommunications, szDeviceName, bufferLen);
+  } else {
+    ret = _GetListDeviceID(eRender, index, szDeviceName, bufferLen);
+  }
+
+  if (guid != NULL && ret == 0) {
+    // Convert the endpoint device's ID string to UTF-8
+    if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize,
+                            NULL, NULL) == 0) {
+      RTC_LOG(LS_ERROR)
+          << "WideCharToMultiByte(CP_UTF8) failed with error code "
+          << GetLastError();
+    }
+  }
+
+  return ret;
+}
+
+// ----------------------------------------------------------------------------
+//  RecordingDeviceName
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::RecordingDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  bool defaultCommunicationDevice(false);
+  const int16_t nDevices(
+      RecordingDevices());  // also updates the list of devices
+
+  // Special fix for the case when the user selects '-1' as index (<=> Default
+  // Communication Device)
+  if (index == (uint16_t)(-1)) {
+    defaultCommunicationDevice = true;
+    index = 0;
+    RTC_LOG(LS_VERBOSE) << "Default Communication endpoint device will be used";
+  }
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  MutexLock lock(&mutex_);
+
+  int32_t ret(-1);
+  WCHAR szDeviceName[MAX_PATH];
+  const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
+
+  // Get the endpoint device's friendly-name
+  if (defaultCommunicationDevice) {
+    ret = _GetDefaultDeviceName(eCapture, eCommunications, szDeviceName,
+                                bufferLen);
+  } else {
+    ret = _GetListDeviceName(eCapture, index, szDeviceName, bufferLen);
+  }
+
+  if (ret == 0) {
+    // Convert the endpoint device's friendly-name to UTF-8
+    if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name,
+                            kAdmMaxDeviceNameSize, NULL, NULL) == 0) {
+      RTC_LOG(LS_ERROR)
+          << "WideCharToMultiByte(CP_UTF8) failed with error code "
+          << GetLastError();
+    }
+  }
+
+  // Get the endpoint ID string (uniquely identifies the device among all audio
+  // endpoint devices)
+  if (defaultCommunicationDevice) {
+    ret =
+        _GetDefaultDeviceID(eCapture, eCommunications, szDeviceName, bufferLen);
+  } else {
+    ret = _GetListDeviceID(eCapture, index, szDeviceName, bufferLen);
+  }
+
+  if (guid != NULL && ret == 0) {
+    // Convert the endpoint device's ID string to UTF-8
+    if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize,
+                            NULL, NULL) == 0) {
+      RTC_LOG(LS_ERROR)
+          << "WideCharToMultiByte(CP_UTF8) failed with error code "
+          << GetLastError();
+    }
+  }
+
+  return ret;
+}
+
+// ----------------------------------------------------------------------------
+//  RecordingDevices
+// ----------------------------------------------------------------------------
+
+int16_t AudioDeviceWindowsCore::RecordingDevices() {
+  MutexLock lock(&mutex_);
+  return RecordingDevicesLocked();
+}
+
+int16_t AudioDeviceWindowsCore::RecordingDevicesLocked() {
+  if (_RefreshDeviceList(eCapture) != -1) {
+    return (_DeviceListCount(eCapture));
+  }
+
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  SetRecordingDevice I (II)
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetRecordingDevice(uint16_t index) {
+  if (_recIsInitialized) {
+    return -1;
+  }
+
+  // Get current number of available capture endpoint devices and refresh the
+  // capture collection.
+  UINT nDevices = RecordingDevices();
+
+  if (index < 0 || index > (nDevices - 1)) {
+    RTC_LOG(LS_ERROR) << "device index is out of range [0," << (nDevices - 1)
+                      << "]";
+    return -1;
+  }
+
+  MutexLock lock(&mutex_);
+
+  HRESULT hr(S_OK);
+
+  RTC_DCHECK(_ptrCaptureCollection);
+
+  // Select an endpoint capture device given the specified index
+  SAFE_RELEASE(_ptrDeviceIn);
+  hr = _ptrCaptureCollection->Item(index, &_ptrDeviceIn);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(_ptrDeviceIn);
+    return -1;
+  }
+
+  WCHAR szDeviceName[MAX_PATH];
+  const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
+
+  // Get the endpoint device's friendly-name
+  if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0) {
+    RTC_LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
+  }
+
+  _usingInputDeviceIndex = true;
+  _inputDeviceIndex = index;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetRecordingDevice II (II)
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  if (_recIsInitialized) {
+    return -1;
+  }
+
+  ERole role(eCommunications);
+
+  if (device == AudioDeviceModule::kDefaultDevice) {
+    role = eConsole;
+  } else if (device == AudioDeviceModule::kDefaultCommunicationDevice) {
+    role = eCommunications;
+  }
+
+  MutexLock lock(&mutex_);
+
+  // Refresh the list of capture endpoint devices
+  _RefreshDeviceList(eCapture);
+
+  HRESULT hr(S_OK);
+
+  RTC_DCHECK(_ptrEnumerator);
+
+  //  Select an endpoint capture device given the specified role
+  SAFE_RELEASE(_ptrDeviceIn);
+  hr = _ptrEnumerator->GetDefaultAudioEndpoint(eCapture, role, &_ptrDeviceIn);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(_ptrDeviceIn);
+    return -1;
+  }
+
+  WCHAR szDeviceName[MAX_PATH];
+  const int bufferLen = sizeof(szDeviceName) / sizeof(szDeviceName)[0];
+
+  // Get the endpoint device's friendly-name
+  if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0) {
+    RTC_LOG(LS_VERBOSE) << "friendly name: \"" << szDeviceName << "\"";
+  }
+
+  _usingInputDeviceIndex = false;
+  _inputDevice = device;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  PlayoutIsAvailable
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::PlayoutIsAvailable(bool& available) {
+  available = false;
+
+  // Try to initialize the playout side
+  int32_t res = InitPlayout();
+
+  // Cancel effect of initialization
+  StopPlayout();
+
+  if (res != -1) {
+    available = true;
+  }
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  RecordingIsAvailable
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::RecordingIsAvailable(bool& available) {
+  available = false;
+
+  // Try to initialize the recording side
+  int32_t res = InitRecording();
+
+  // Cancel effect of initialization
+  StopRecording();
+
+  if (res != -1) {
+    available = true;
+  }
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  InitPlayout
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::InitPlayout() {
+  MutexLock lock(&mutex_);
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (_playIsInitialized) {
+    return 0;
+  }
+
+  if (_ptrDeviceOut == NULL) {
+    return -1;
+  }
+
+  // Initialize the speaker (devices might have been added or removed)
+  if (InitSpeakerLocked() == -1) {
+    RTC_LOG(LS_WARNING) << "InitSpeaker() failed";
+  }
+
+  // Ensure that the updated rendering endpoint device is valid
+  if (_ptrDeviceOut == NULL) {
+    return -1;
+  }
+
+  if (_builtInAecEnabled && _recIsInitialized) {
+    // Ensure the correct render device is configured in case
+    // InitRecording() was called before InitPlayout().
+    if (SetDMOProperties() == -1) {
+      return -1;
+    }
+  }
+
+  HRESULT hr = S_OK;
+  WAVEFORMATEX* pWfxOut = NULL;
+  WAVEFORMATEX Wfx = WAVEFORMATEX();
+  WAVEFORMATEX* pWfxClosestMatch = NULL;
+
+  // Create COM object with IAudioClient interface.
+  SAFE_RELEASE(_ptrClientOut);
+  hr = _ptrDeviceOut->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL,
+                               (void**)&_ptrClientOut);
+  EXIT_ON_ERROR(hr);
+
+  // Retrieve the stream format that the audio engine uses for its internal
+  // processing (mixing) of shared-mode streams.
+  hr = _ptrClientOut->GetMixFormat(&pWfxOut);
+  if (SUCCEEDED(hr)) {
+    RTC_LOG(LS_VERBOSE) << "Audio Engine's current rendering mix format:";
+    // format type
+    RTC_LOG(LS_VERBOSE) << "wFormatTag     : 0x"
+                        << rtc::ToHex(pWfxOut->wFormatTag) << " ("
+                        << pWfxOut->wFormatTag << ")";
+    // number of channels (i.e. mono, stereo...)
+    RTC_LOG(LS_VERBOSE) << "nChannels      : " << pWfxOut->nChannels;
+    // sample rate
+    RTC_LOG(LS_VERBOSE) << "nSamplesPerSec : " << pWfxOut->nSamplesPerSec;
+    // for buffer estimation
+    RTC_LOG(LS_VERBOSE) << "nAvgBytesPerSec: " << pWfxOut->nAvgBytesPerSec;
+    // block size of data
+    RTC_LOG(LS_VERBOSE) << "nBlockAlign    : " << pWfxOut->nBlockAlign;
+    // number of bits per sample of mono data
+    RTC_LOG(LS_VERBOSE) << "wBitsPerSample : " << pWfxOut->wBitsPerSample;
+    RTC_LOG(LS_VERBOSE) << "cbSize         : " << pWfxOut->cbSize;
+  }
+
+  // Set wave format
+  Wfx.wFormatTag = WAVE_FORMAT_PCM;
+  Wfx.wBitsPerSample = 16;
+  Wfx.cbSize = 0;
+
+  const int freqs[] = {48000, 44100, 16000, 96000, 32000, 8000};
+  hr = S_FALSE;
+
+  // Iterate over frequencies and channels, in order of priority
+  for (unsigned int freq = 0; freq < sizeof(freqs) / sizeof(freqs[0]); freq++) {
+    for (unsigned int chan = 0; chan < sizeof(_playChannelsPrioList) /
+                                           sizeof(_playChannelsPrioList[0]);
+         chan++) {
+      Wfx.nChannels = _playChannelsPrioList[chan];
+      Wfx.nSamplesPerSec = freqs[freq];
+      Wfx.nBlockAlign = Wfx.nChannels * Wfx.wBitsPerSample / 8;
+      Wfx.nAvgBytesPerSec = Wfx.nSamplesPerSec * Wfx.nBlockAlign;
+      // If the method succeeds and the audio endpoint device supports the
+      // specified stream format, it returns S_OK. If the method succeeds and
+      // provides a closest match to the specified format, it returns S_FALSE.
+      hr = _ptrClientOut->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, &Wfx,
+                                            &pWfxClosestMatch);
+      if (hr == S_OK) {
+        break;
+      } else {
+        if (pWfxClosestMatch) {
+          RTC_LOG(LS_INFO) << "nChannels=" << Wfx.nChannels
+                           << ", nSamplesPerSec=" << Wfx.nSamplesPerSec
+                           << " is not supported. Closest match: "
+                              "nChannels="
+                           << pWfxClosestMatch->nChannels << ", nSamplesPerSec="
+                           << pWfxClosestMatch->nSamplesPerSec;
+          CoTaskMemFree(pWfxClosestMatch);
+          pWfxClosestMatch = NULL;
+        } else {
+          RTC_LOG(LS_INFO) << "nChannels=" << Wfx.nChannels
+                           << ", nSamplesPerSec=" << Wfx.nSamplesPerSec
+                           << " is not supported. No closest match.";
+        }
+      }
+    }
+    if (hr == S_OK)
+      break;
+  }
+
+  // TODO(andrew): what happens in the event of failure in the above loop?
+  //   Is _ptrClientOut->Initialize expected to fail?
+  //   Same in InitRecording().
+  if (hr == S_OK) {
+    _playAudioFrameSize = Wfx.nBlockAlign;
+    // Block size is the number of samples each channel in 10ms.
+    _playBlockSize = Wfx.nSamplesPerSec / 100;
+    _playSampleRate = Wfx.nSamplesPerSec;
+    _devicePlaySampleRate =
+        Wfx.nSamplesPerSec;  // The device itself continues to run at 44.1 kHz.
+    _devicePlayBlockSize = Wfx.nSamplesPerSec / 100;
+    _playChannels = Wfx.nChannels;
+
+    RTC_LOG(LS_VERBOSE) << "VoE selected this rendering format:";
+    RTC_LOG(LS_VERBOSE) << "wFormatTag         : 0x"
+                        << rtc::ToHex(Wfx.wFormatTag) << " (" << Wfx.wFormatTag
+                        << ")";
+    RTC_LOG(LS_VERBOSE) << "nChannels          : " << Wfx.nChannels;
+    RTC_LOG(LS_VERBOSE) << "nSamplesPerSec     : " << Wfx.nSamplesPerSec;
+    RTC_LOG(LS_VERBOSE) << "nAvgBytesPerSec    : " << Wfx.nAvgBytesPerSec;
+    RTC_LOG(LS_VERBOSE) << "nBlockAlign        : " << Wfx.nBlockAlign;
+    RTC_LOG(LS_VERBOSE) << "wBitsPerSample     : " << Wfx.wBitsPerSample;
+    RTC_LOG(LS_VERBOSE) << "cbSize             : " << Wfx.cbSize;
+    RTC_LOG(LS_VERBOSE) << "Additional settings:";
+    RTC_LOG(LS_VERBOSE) << "_playAudioFrameSize: " << _playAudioFrameSize;
+    RTC_LOG(LS_VERBOSE) << "_playBlockSize     : " << _playBlockSize;
+    RTC_LOG(LS_VERBOSE) << "_playChannels      : " << _playChannels;
+  }
+
+  // Create a rendering stream.
+  //
+  // ****************************************************************************
+  // For a shared-mode stream that uses event-driven buffering, the caller must
+  // set both hnsPeriodicity and hnsBufferDuration to 0. The Initialize method
+  // determines how large a buffer to allocate based on the scheduling period
+  // of the audio engine. Although the client's buffer processing thread is
+  // event driven, the basic buffer management process, as described previously,
+  // is unaltered.
+  // Each time the thread awakens, it should call
+  // IAudioClient::GetCurrentPadding to determine how much data to write to a
+  // rendering buffer or read from a capture buffer. In contrast to the two
+  // buffers that the Initialize method allocates for an exclusive-mode stream
+  // that uses event-driven buffering, a shared-mode stream requires a single
+  // buffer.
+  // ****************************************************************************
+  //
+  REFERENCE_TIME hnsBufferDuration =
+      0;  // ask for minimum buffer size (default)
+  if (_devicePlaySampleRate == 44100) {
+    // Ask for a larger buffer size (30ms) when using 44.1kHz as render rate.
+    // There seems to be a larger risk of underruns for 44.1 compared
+    // with the default rate (48kHz). When using default, we set the requested
+    // buffer duration to 0, which sets the buffer to the minimum size
+    // required by the engine thread. The actual buffer size can then be
+    // read by GetBufferSize() and it is 20ms on most machines.
+    hnsBufferDuration = 30 * 10000;
+  }
+  hr = _ptrClientOut->Initialize(
+      AUDCLNT_SHAREMODE_SHARED,  // share Audio Engine with other applications
+      AUDCLNT_STREAMFLAGS_EVENTCALLBACK,  // processing of the audio buffer by
+                                          // the client will be event driven
+      hnsBufferDuration,  // requested buffer capacity as a time value (in
+                          // 100-nanosecond units)
+      0,                  // periodicity
+      &Wfx,               // selected wave format
+      NULL);              // session GUID
+
+  if (FAILED(hr)) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::Initialize() failed:";
+  }
+  EXIT_ON_ERROR(hr);
+
+  if (_ptrAudioBuffer) {
+    // Update the audio buffer with the selected parameters
+    _ptrAudioBuffer->SetPlayoutSampleRate(_playSampleRate);
+    _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
+  } else {
+    // We can enter this state during CoreAudioIsSupported() when no
+    // AudioDeviceImplementation has been created, hence the AudioDeviceBuffer
+    // does not exist. It is OK to end up here since we don't initiate any media
+    // in CoreAudioIsSupported().
+    RTC_LOG(LS_VERBOSE)
+        << "AudioDeviceBuffer must be attached before streaming can start";
+  }
+
+  // Get the actual size of the shared (endpoint buffer).
+  // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
+  UINT bufferFrameCount(0);
+  hr = _ptrClientOut->GetBufferSize(&bufferFrameCount);
+  if (SUCCEEDED(hr)) {
+    RTC_LOG(LS_VERBOSE) << "IAudioClient::GetBufferSize() => "
+                        << bufferFrameCount << " (<=> "
+                        << bufferFrameCount * _playAudioFrameSize << " bytes)";
+  }
+
+  // Set the event handle that the system signals when an audio buffer is ready
+  // to be processed by the client.
+  hr = _ptrClientOut->SetEventHandle(_hRenderSamplesReadyEvent);
+  EXIT_ON_ERROR(hr);
+
+  // Get an IAudioRenderClient interface.
+  SAFE_RELEASE(_ptrRenderClient);
+  hr = _ptrClientOut->GetService(__uuidof(IAudioRenderClient),
+                                 (void**)&_ptrRenderClient);
+  EXIT_ON_ERROR(hr);
+
+  // Mark playout side as initialized
+  _playIsInitialized = true;
+
+  CoTaskMemFree(pWfxOut);
+  CoTaskMemFree(pWfxClosestMatch);
+
+  RTC_LOG(LS_VERBOSE) << "render side is now initialized";
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  CoTaskMemFree(pWfxOut);
+  CoTaskMemFree(pWfxClosestMatch);
+  SAFE_RELEASE(_ptrClientOut);
+  SAFE_RELEASE(_ptrRenderClient);
+  return -1;
+}
+
+// Capture initialization when the built-in AEC DirectX Media Object (DMO) is
+// used. Called from InitRecording(), most of which is skipped over. The DMO
+// handles device initialization itself.
+// Reference: http://msdn.microsoft.com/en-us/library/ff819492(v=vs.85).aspx
+int32_t AudioDeviceWindowsCore::InitRecordingDMO() {
+  RTC_DCHECK(_builtInAecEnabled);
+  RTC_DCHECK(_dmo);
+
+  if (SetDMOProperties() == -1) {
+    return -1;
+  }
+
+  DMO_MEDIA_TYPE mt = {};
+  HRESULT hr = MoInitMediaType(&mt, sizeof(WAVEFORMATEX));
+  if (FAILED(hr)) {
+    MoFreeMediaType(&mt);
+    _TraceCOMError(hr);
+    return -1;
+  }
+  mt.majortype = MEDIATYPE_Audio;
+  mt.subtype = MEDIASUBTYPE_PCM;
+  mt.formattype = FORMAT_WaveFormatEx;
+
+  // Supported formats
+  // nChannels: 1 (in AEC-only mode)
+  // nSamplesPerSec: 8000, 11025, 16000, 22050
+  // wBitsPerSample: 16
+  WAVEFORMATEX* ptrWav = reinterpret_cast<WAVEFORMATEX*>(mt.pbFormat);
+  ptrWav->wFormatTag = WAVE_FORMAT_PCM;
+  ptrWav->nChannels = 1;
+  // 16000 is the highest we can support with our resampler.
+  ptrWav->nSamplesPerSec = 16000;
+  ptrWav->nAvgBytesPerSec = 32000;
+  ptrWav->nBlockAlign = 2;
+  ptrWav->wBitsPerSample = 16;
+  ptrWav->cbSize = 0;
+
+  // Set the VoE format equal to the AEC output format.
+  _recAudioFrameSize = ptrWav->nBlockAlign;
+  _recSampleRate = ptrWav->nSamplesPerSec;
+  _recBlockSize = ptrWav->nSamplesPerSec / 100;
+  _recChannels = ptrWav->nChannels;
+
+  // Set the DMO output format parameters.
+  hr = _dmo->SetOutputType(kAecCaptureStreamIndex, &mt, 0);
+  MoFreeMediaType(&mt);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    return -1;
+  }
+
+  if (_ptrAudioBuffer) {
+    _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
+    _ptrAudioBuffer->SetRecordingChannels(_recChannels);
+  } else {
+    // Refer to InitRecording() for comments.
+    RTC_LOG(LS_VERBOSE)
+        << "AudioDeviceBuffer must be attached before streaming can start";
+  }
+
+  _mediaBuffer = rtc::make_ref_counted<MediaBufferImpl>(_recBlockSize *
+                                                        _recAudioFrameSize);
+
+  // Optional, but if called, must be after media types are set.
+  hr = _dmo->AllocateStreamingResources();
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    return -1;
+  }
+
+  _recIsInitialized = true;
+  RTC_LOG(LS_VERBOSE) << "Capture side is now initialized";
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  InitRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::InitRecording() {
+  MutexLock lock(&mutex_);
+
+  if (_recording) {
+    return -1;
+  }
+
+  if (_recIsInitialized) {
+    return 0;
+  }
+
+  if (QueryPerformanceFrequency(&_perfCounterFreq) == 0) {
+    return -1;
+  }
+  _perfCounterFactor = 10000000.0 / (double)_perfCounterFreq.QuadPart;
+
+  if (_ptrDeviceIn == NULL) {
+    return -1;
+  }
+
+  // Initialize the microphone (devices might have been added or removed)
+  if (InitMicrophoneLocked() == -1) {
+    RTC_LOG(LS_WARNING) << "InitMicrophone() failed";
+  }
+
+  // Ensure that the updated capturing endpoint device is valid
+  if (_ptrDeviceIn == NULL) {
+    return -1;
+  }
+
+  if (_builtInAecEnabled) {
+    // The DMO will configure the capture device.
+    return InitRecordingDMO();
+  }
+
+  HRESULT hr = S_OK;
+  WAVEFORMATEX* pWfxIn = NULL;
+  WAVEFORMATEXTENSIBLE Wfx = WAVEFORMATEXTENSIBLE();
+  WAVEFORMATEX* pWfxClosestMatch = NULL;
+
+  // Create COM object with IAudioClient interface.
+  SAFE_RELEASE(_ptrClientIn);
+  hr = _ptrDeviceIn->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL,
+                              (void**)&_ptrClientIn);
+  EXIT_ON_ERROR(hr);
+
+  // Retrieve the stream format that the audio engine uses for its internal
+  // processing (mixing) of shared-mode streams.
+  hr = _ptrClientIn->GetMixFormat(&pWfxIn);
+  if (SUCCEEDED(hr)) {
+    RTC_LOG(LS_VERBOSE) << "Audio Engine's current capturing mix format:";
+    // format type
+    RTC_LOG(LS_VERBOSE) << "wFormatTag     : 0x"
+                        << rtc::ToHex(pWfxIn->wFormatTag) << " ("
+                        << pWfxIn->wFormatTag << ")";
+    // number of channels (i.e. mono, stereo...)
+    RTC_LOG(LS_VERBOSE) << "nChannels      : " << pWfxIn->nChannels;
+    // sample rate
+    RTC_LOG(LS_VERBOSE) << "nSamplesPerSec : " << pWfxIn->nSamplesPerSec;
+    // for buffer estimation
+    RTC_LOG(LS_VERBOSE) << "nAvgBytesPerSec: " << pWfxIn->nAvgBytesPerSec;
+    // block size of data
+    RTC_LOG(LS_VERBOSE) << "nBlockAlign    : " << pWfxIn->nBlockAlign;
+    // number of bits per sample of mono data
+    RTC_LOG(LS_VERBOSE) << "wBitsPerSample : " << pWfxIn->wBitsPerSample;
+    RTC_LOG(LS_VERBOSE) << "cbSize         : " << pWfxIn->cbSize;
+  }
+
+  // Set wave format
+  Wfx.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+  Wfx.Format.wBitsPerSample = 16;
+  Wfx.Format.cbSize = 22;
+  Wfx.dwChannelMask = 0;
+  Wfx.Samples.wValidBitsPerSample = Wfx.Format.wBitsPerSample;
+  Wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+
+  const int freqs[6] = {48000, 44100, 16000, 96000, 32000, 8000};
+  hr = S_FALSE;
+
+  // Iterate over frequencies and channels, in order of priority
+  for (unsigned int freq = 0; freq < sizeof(freqs) / sizeof(freqs[0]); freq++) {
+    for (unsigned int chan = 0;
+         chan < sizeof(_recChannelsPrioList) / sizeof(_recChannelsPrioList[0]);
+         chan++) {
+      Wfx.Format.nChannels = _recChannelsPrioList[chan];
+      Wfx.Format.nSamplesPerSec = freqs[freq];
+      Wfx.Format.nBlockAlign =
+          Wfx.Format.nChannels * Wfx.Format.wBitsPerSample / 8;
+      Wfx.Format.nAvgBytesPerSec =
+          Wfx.Format.nSamplesPerSec * Wfx.Format.nBlockAlign;
+      // If the method succeeds and the audio endpoint device supports the
+      // specified stream format, it returns S_OK. If the method succeeds and
+      // provides a closest match to the specified format, it returns S_FALSE.
+      hr = _ptrClientIn->IsFormatSupported(
+          AUDCLNT_SHAREMODE_SHARED, (WAVEFORMATEX*)&Wfx, &pWfxClosestMatch);
+      if (hr == S_OK) {
+        break;
+      } else {
+        if (pWfxClosestMatch) {
+          RTC_LOG(LS_INFO) << "nChannels=" << Wfx.Format.nChannels
+                           << ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec
+                           << " is not supported. Closest match: "
+                              "nChannels="
+                           << pWfxClosestMatch->nChannels << ", nSamplesPerSec="
+                           << pWfxClosestMatch->nSamplesPerSec;
+          CoTaskMemFree(pWfxClosestMatch);
+          pWfxClosestMatch = NULL;
+        } else {
+          RTC_LOG(LS_INFO) << "nChannels=" << Wfx.Format.nChannels
+                           << ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec
+                           << " is not supported. No closest match.";
+        }
+      }
+    }
+    if (hr == S_OK)
+      break;
+  }
+
+  if (hr == S_OK) {
+    _recAudioFrameSize = Wfx.Format.nBlockAlign;
+    _recSampleRate = Wfx.Format.nSamplesPerSec;
+    _recBlockSize = Wfx.Format.nSamplesPerSec / 100;
+    _recChannels = Wfx.Format.nChannels;
+
+    RTC_LOG(LS_VERBOSE) << "VoE selected this capturing format:";
+    RTC_LOG(LS_VERBOSE) << "wFormatTag        : 0x"
+                        << rtc::ToHex(Wfx.Format.wFormatTag) << " ("
+                        << Wfx.Format.wFormatTag << ")";
+    RTC_LOG(LS_VERBOSE) << "nChannels         : " << Wfx.Format.nChannels;
+    RTC_LOG(LS_VERBOSE) << "nSamplesPerSec    : " << Wfx.Format.nSamplesPerSec;
+    RTC_LOG(LS_VERBOSE) << "nAvgBytesPerSec   : " << Wfx.Format.nAvgBytesPerSec;
+    RTC_LOG(LS_VERBOSE) << "nBlockAlign       : " << Wfx.Format.nBlockAlign;
+    RTC_LOG(LS_VERBOSE) << "wBitsPerSample    : " << Wfx.Format.wBitsPerSample;
+    RTC_LOG(LS_VERBOSE) << "cbSize            : " << Wfx.Format.cbSize;
+    RTC_LOG(LS_VERBOSE) << "Additional settings:";
+    RTC_LOG(LS_VERBOSE) << "_recAudioFrameSize: " << _recAudioFrameSize;
+    RTC_LOG(LS_VERBOSE) << "_recBlockSize     : " << _recBlockSize;
+    RTC_LOG(LS_VERBOSE) << "_recChannels      : " << _recChannels;
+  }
+
+  // Create a capturing stream.
+  hr = _ptrClientIn->Initialize(
+      AUDCLNT_SHAREMODE_SHARED,  // share Audio Engine with other applications
+      AUDCLNT_STREAMFLAGS_EVENTCALLBACK |  // processing of the audio buffer by
+                                           // the client will be event driven
+          AUDCLNT_STREAMFLAGS_NOPERSIST,   // volume and mute settings for an
+                                           // audio session will not persist
+                                           // across system restarts
+      0,                    // required for event-driven shared mode
+      0,                    // periodicity
+      (WAVEFORMATEX*)&Wfx,  // selected wave format
+      NULL);                // session GUID
+
+  if (hr != S_OK) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::Initialize() failed:";
+  }
+  EXIT_ON_ERROR(hr);
+
+  if (_ptrAudioBuffer) {
+    // Update the audio buffer with the selected parameters
+    _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
+    _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+  } else {
+    // We can enter this state during CoreAudioIsSupported() when no
+    // AudioDeviceImplementation has been created, hence the AudioDeviceBuffer
+    // does not exist. It is OK to end up here since we don't initiate any media
+    // in CoreAudioIsSupported().
+    RTC_LOG(LS_VERBOSE)
+        << "AudioDeviceBuffer must be attached before streaming can start";
+  }
+
+  // Get the actual size of the shared (endpoint buffer).
+  // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
+  UINT bufferFrameCount(0);
+  hr = _ptrClientIn->GetBufferSize(&bufferFrameCount);
+  if (SUCCEEDED(hr)) {
+    RTC_LOG(LS_VERBOSE) << "IAudioClient::GetBufferSize() => "
+                        << bufferFrameCount << " (<=> "
+                        << bufferFrameCount * _recAudioFrameSize << " bytes)";
+  }
+
+  // Set the event handle that the system signals when an audio buffer is ready
+  // to be processed by the client.
+  hr = _ptrClientIn->SetEventHandle(_hCaptureSamplesReadyEvent);
+  EXIT_ON_ERROR(hr);
+
+  // Get an IAudioCaptureClient interface.
+  SAFE_RELEASE(_ptrCaptureClient);
+  hr = _ptrClientIn->GetService(__uuidof(IAudioCaptureClient),
+                                (void**)&_ptrCaptureClient);
+  EXIT_ON_ERROR(hr);
+
+  // Mark capture side as initialized
+  _recIsInitialized = true;
+
+  CoTaskMemFree(pWfxIn);
+  CoTaskMemFree(pWfxClosestMatch);
+
+  RTC_LOG(LS_VERBOSE) << "capture side is now initialized";
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  CoTaskMemFree(pWfxIn);
+  CoTaskMemFree(pWfxClosestMatch);
+  SAFE_RELEASE(_ptrClientIn);
+  SAFE_RELEASE(_ptrCaptureClient);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  StartRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::StartRecording() {
+  if (!_recIsInitialized) {
+    return -1;
+  }
+
+  if (_hRecThread != NULL) {
+    return 0;
+  }
+
+  if (_recording) {
+    return 0;
+  }
+
+  {
+    MutexLock lockScoped(&mutex_);
+
+    // Create thread which will drive the capturing
+    LPTHREAD_START_ROUTINE lpStartAddress = WSAPICaptureThread;
+    if (_builtInAecEnabled) {
+      // Redirect to the DMO polling method.
+      lpStartAddress = WSAPICaptureThreadPollDMO;
+
+      if (!_playing) {
+        // The DMO won't provide us captured output data unless we
+        // give it render data to process.
+        RTC_LOG(LS_ERROR)
+            << "Playout must be started before recording when using"
+               " the built-in AEC";
+        return -1;
+      }
+    }
+
+    RTC_DCHECK(_hRecThread == NULL);
+    _hRecThread = CreateThread(NULL, 0, lpStartAddress, this, 0, NULL);
+    if (_hRecThread == NULL) {
+      RTC_LOG(LS_ERROR) << "failed to create the recording thread";
+      return -1;
+    }
+
+    // Set thread priority to highest possible
+    SetThreadPriority(_hRecThread, THREAD_PRIORITY_TIME_CRITICAL);
+  }  // critScoped
+
+  DWORD ret = WaitForSingleObject(_hCaptureStartedEvent, 1000);
+  if (ret != WAIT_OBJECT_0) {
+    RTC_LOG(LS_VERBOSE) << "capturing did not start up properly";
+    return -1;
+  }
+  RTC_LOG(LS_VERBOSE) << "capture audio stream has now started...";
+
+  _recording = true;
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  StopRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::StopRecording() {
+  int32_t err = 0;
+
+  if (!_recIsInitialized) {
+    return 0;
+  }
+
+  _Lock();
+
+  if (_hRecThread == NULL) {
+    RTC_LOG(LS_VERBOSE)
+        << "no capturing stream is active => close down WASAPI only";
+    SAFE_RELEASE(_ptrClientIn);
+    SAFE_RELEASE(_ptrCaptureClient);
+    _recIsInitialized = false;
+    _recording = false;
+    _UnLock();
+    return 0;
+  }
+
+  // Stop the driving thread...
+  RTC_LOG(LS_VERBOSE) << "closing down the webrtc_core_audio_capture_thread...";
+  // Manual-reset event; it will remain signalled to stop all capture threads.
+  SetEvent(_hShutdownCaptureEvent);
+
+  _UnLock();
+  DWORD ret = WaitForSingleObject(_hRecThread, 2000);
+  if (ret != WAIT_OBJECT_0) {
+    RTC_LOG(LS_ERROR)
+        << "failed to close down webrtc_core_audio_capture_thread";
+    err = -1;
+  } else {
+    RTC_LOG(LS_VERBOSE) << "webrtc_core_audio_capture_thread is now closed";
+  }
+  _Lock();
+
+  ResetEvent(_hShutdownCaptureEvent);  // Must be manually reset.
+  // Ensure that the thread has released these interfaces properly.
+  RTC_DCHECK(err == -1 || _ptrClientIn == NULL);
+  RTC_DCHECK(err == -1 || _ptrCaptureClient == NULL);
+
+  _recIsInitialized = false;
+  _recording = false;
+
+  // These will create thread leaks in the result of an error,
+  // but we can at least resume the call.
+  CloseHandle(_hRecThread);
+  _hRecThread = NULL;
+
+  if (_builtInAecEnabled) {
+    RTC_DCHECK(_dmo);
+    // This is necessary. Otherwise the DMO can generate garbage render
+    // audio even after rendering has stopped.
+    HRESULT hr = _dmo->FreeStreamingResources();
+    if (FAILED(hr)) {
+      _TraceCOMError(hr);
+      err = -1;
+    }
+  }
+
+  _UnLock();
+
+  return err;
+}
+
+// ----------------------------------------------------------------------------
+//  RecordingIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::RecordingIsInitialized() const {
+  return (_recIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+//  Recording
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::Recording() const {
+  return (_recording);
+}
+
+// ----------------------------------------------------------------------------
+//  PlayoutIsInitialized
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::PlayoutIsInitialized() const {
+  return (_playIsInitialized);
+}
+
+// ----------------------------------------------------------------------------
+//  StartPlayout
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::StartPlayout() {
+  if (!_playIsInitialized) {
+    return -1;
+  }
+
+  if (_hPlayThread != NULL) {
+    return 0;
+  }
+
+  if (_playing) {
+    return 0;
+  }
+
+  {
+    MutexLock lockScoped(&mutex_);
+
+    // Create thread which will drive the rendering.
+    RTC_DCHECK(_hPlayThread == NULL);
+    _hPlayThread = CreateThread(NULL, 0, WSAPIRenderThread, this, 0, NULL);
+    if (_hPlayThread == NULL) {
+      RTC_LOG(LS_ERROR) << "failed to create the playout thread";
+      return -1;
+    }
+
+    // Set thread priority to highest possible.
+    SetThreadPriority(_hPlayThread, THREAD_PRIORITY_TIME_CRITICAL);
+  }  // critScoped
+
+  DWORD ret = WaitForSingleObject(_hRenderStartedEvent, 1000);
+  if (ret != WAIT_OBJECT_0) {
+    RTC_LOG(LS_VERBOSE) << "rendering did not start up properly";
+    return -1;
+  }
+
+  _playing = true;
+  RTC_LOG(LS_VERBOSE) << "rendering audio stream has now started...";
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  StopPlayout
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::StopPlayout() {
+  if (!_playIsInitialized) {
+    return 0;
+  }
+
+  {
+    MutexLock lockScoped(&mutex_);
+
+    if (_hPlayThread == NULL) {
+      RTC_LOG(LS_VERBOSE)
+          << "no rendering stream is active => close down WASAPI only";
+      SAFE_RELEASE(_ptrClientOut);
+      SAFE_RELEASE(_ptrRenderClient);
+      _playIsInitialized = false;
+      _playing = false;
+      return 0;
+    }
+
+    // stop the driving thread...
+    RTC_LOG(LS_VERBOSE)
+        << "closing down the webrtc_core_audio_render_thread...";
+    SetEvent(_hShutdownRenderEvent);
+  }  // critScoped
+
+  DWORD ret = WaitForSingleObject(_hPlayThread, 2000);
+  if (ret != WAIT_OBJECT_0) {
+    // the thread did not stop as it should
+    RTC_LOG(LS_ERROR) << "failed to close down webrtc_core_audio_render_thread";
+    CloseHandle(_hPlayThread);
+    _hPlayThread = NULL;
+    _playIsInitialized = false;
+    _playing = false;
+    return -1;
+  }
+
+  {
+    MutexLock lockScoped(&mutex_);
+    RTC_LOG(LS_VERBOSE) << "webrtc_core_audio_render_thread is now closed";
+
+    // to reset this event manually at each time we finish with it,
+    // in case that the render thread has exited before StopPlayout(),
+    // this event might be caught by the new render thread within same VoE
+    // instance.
+    ResetEvent(_hShutdownRenderEvent);
+
+    SAFE_RELEASE(_ptrClientOut);
+    SAFE_RELEASE(_ptrRenderClient);
+
+    _playIsInitialized = false;
+    _playing = false;
+
+    CloseHandle(_hPlayThread);
+    _hPlayThread = NULL;
+
+    if (_builtInAecEnabled && _recording) {
+      // The DMO won't provide us captured output data unless we
+      // give it render data to process.
+      //
+      // We still permit the playout to shutdown, and trace a warning.
+      // Otherwise, VoE can get into a state which will never permit
+      // playout to stop properly.
+      RTC_LOG(LS_WARNING)
+          << "Recording should be stopped before playout when using the"
+             " built-in AEC";
+    }
+
+    // Reset the playout delay value.
+    _sndCardPlayDelay = 0;
+  }  // critScoped
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  PlayoutDelay
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const {
+  MutexLock lockScoped(&mutex_);
+  delayMS = static_cast<uint16_t>(_sndCardPlayDelay);
+  return 0;
+}
+
+bool AudioDeviceWindowsCore::BuiltInAECIsAvailable() const {
+  return _dmo != nullptr;
+}
+
+// ----------------------------------------------------------------------------
+//  Playing
+// ----------------------------------------------------------------------------
+
+bool AudioDeviceWindowsCore::Playing() const {
+  return (_playing);
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+// ----------------------------------------------------------------------------
+//  [static] WSAPIRenderThread
+// ----------------------------------------------------------------------------
+
+DWORD WINAPI AudioDeviceWindowsCore::WSAPIRenderThread(LPVOID context) {
+  return reinterpret_cast<AudioDeviceWindowsCore*>(context)->DoRenderThread();
+}
+
+// ----------------------------------------------------------------------------
+//  [static] WSAPICaptureThread
+// ----------------------------------------------------------------------------
+
+DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThread(LPVOID context) {
+  return reinterpret_cast<AudioDeviceWindowsCore*>(context)->DoCaptureThread();
+}
+
+DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThreadPollDMO(LPVOID context) {
+  return reinterpret_cast<AudioDeviceWindowsCore*>(context)
+      ->DoCaptureThreadPollDMO();
+}
+
+// ----------------------------------------------------------------------------
+//  DoRenderThread
+// ----------------------------------------------------------------------------
+
+DWORD AudioDeviceWindowsCore::DoRenderThread() {
+  bool keepPlaying = true;
+  HANDLE waitArray[2] = {_hShutdownRenderEvent, _hRenderSamplesReadyEvent};
+  HRESULT hr = S_OK;
+  HANDLE hMmTask = NULL;
+
+  // Initialize COM as MTA in this thread.
+  ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+  if (!comInit.Succeeded()) {
+    RTC_LOG(LS_ERROR) << "failed to initialize COM in render thread";
+    return 1;
+  }
+
+  rtc::SetCurrentThreadName("webrtc_core_audio_render_thread");
+
+  // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread
+  // priority.
+  //
+  if (_winSupportAvrt) {
+    DWORD taskIndex(0);
+    hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
+    if (hMmTask) {
+      if (FALSE == _PAvSetMmThreadPriority(hMmTask, AVRT_PRIORITY_CRITICAL)) {
+        RTC_LOG(LS_WARNING) << "failed to boost play-thread using MMCSS";
+      }
+      RTC_LOG(LS_VERBOSE)
+          << "render thread is now registered with MMCSS (taskIndex="
+          << taskIndex << ")";
+    } else {
+      RTC_LOG(LS_WARNING) << "failed to enable MMCSS on render thread (err="
+                          << GetLastError() << ")";
+      _TraceCOMError(GetLastError());
+    }
+  }
+
+  _Lock();
+
+  IAudioClock* clock = NULL;
+
+  // Get size of rendering buffer (length is expressed as the number of audio
+  // frames the buffer can hold). This value is fixed during the rendering
+  // session.
+  //
+  UINT32 bufferLength = 0;
+  hr = _ptrClientOut->GetBufferSize(&bufferLength);
+  EXIT_ON_ERROR(hr);
+  RTC_LOG(LS_VERBOSE) << "[REND] size of buffer       : " << bufferLength;
+
+  // Get maximum latency for the current stream (will not change for the
+  // lifetime  of the IAudioClient object).
+  //
+  REFERENCE_TIME latency;
+  _ptrClientOut->GetStreamLatency(&latency);
+  RTC_LOG(LS_VERBOSE) << "[REND] max stream latency   : " << (DWORD)latency
+                      << " (" << (double)(latency / 10000.0) << " ms)";
+
+  // Get the length of the periodic interval separating successive processing
+  // passes by the audio engine on the data in the endpoint buffer.
+  //
+  // The period between processing passes by the audio engine is fixed for a
+  // particular audio endpoint device and represents the smallest processing
+  // quantum for the audio engine. This period plus the stream latency between
+  // the buffer and endpoint device represents the minimum possible latency that
+  // an audio application can achieve. Typical value: 100000 <=> 0.01 sec =
+  // 10ms.
+  //
+  REFERENCE_TIME devPeriod = 0;
+  REFERENCE_TIME devPeriodMin = 0;
+  _ptrClientOut->GetDevicePeriod(&devPeriod, &devPeriodMin);
+  RTC_LOG(LS_VERBOSE) << "[REND] device period        : " << (DWORD)devPeriod
+                      << " (" << (double)(devPeriod / 10000.0) << " ms)";
+
+  // Derive initial rendering delay.
+  // Example: 10*(960/480) + 15 = 20 + 15 = 35ms
+  //
+  int playout_delay = 10 * (bufferLength / _playBlockSize) +
+                      (int)((latency + devPeriod) / 10000);
+  _sndCardPlayDelay = playout_delay;
+  _writtenSamples = 0;
+  RTC_LOG(LS_VERBOSE) << "[REND] initial delay        : " << playout_delay;
+
+  double endpointBufferSizeMS =
+      10.0 * ((double)bufferLength / (double)_devicePlayBlockSize);
+  RTC_LOG(LS_VERBOSE) << "[REND] endpointBufferSizeMS : "
+                      << endpointBufferSizeMS;
+
+  // Before starting the stream, fill the rendering buffer with silence.
+  //
+  BYTE* pData = NULL;
+  hr = _ptrRenderClient->GetBuffer(bufferLength, &pData);
+  EXIT_ON_ERROR(hr);
+
+  hr =
+      _ptrRenderClient->ReleaseBuffer(bufferLength, AUDCLNT_BUFFERFLAGS_SILENT);
+  EXIT_ON_ERROR(hr);
+
+  _writtenSamples += bufferLength;
+
+  hr = _ptrClientOut->GetService(__uuidof(IAudioClock), (void**)&clock);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_WARNING)
+        << "failed to get IAudioClock interface from the IAudioClient";
+  }
+
+  // Start up the rendering audio stream.
+  hr = _ptrClientOut->Start();
+  EXIT_ON_ERROR(hr);
+
+  _UnLock();
+
+  // Set event which will ensure that the calling thread modifies the playing
+  // state to true.
+  //
+  SetEvent(_hRenderStartedEvent);
+
+  // >> ------------------ THREAD LOOP ------------------
+
+  while (keepPlaying) {
+    // Wait for a render notification event or a shutdown event
+    DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
+    switch (waitResult) {
+      case WAIT_OBJECT_0 + 0:  // _hShutdownRenderEvent
+        keepPlaying = false;
+        break;
+      case WAIT_OBJECT_0 + 1:  // _hRenderSamplesReadyEvent
+        break;
+      case WAIT_TIMEOUT:  // timeout notification
+        RTC_LOG(LS_WARNING) << "render event timed out after 0.5 seconds";
+        goto Exit;
+      default:  // unexpected error
+        RTC_LOG(LS_WARNING) << "unknown wait termination on render side";
+        goto Exit;
+    }
+
+    while (keepPlaying) {
+      _Lock();
+
+      // Sanity check to ensure that essential states are not modified
+      // during the unlocked period.
+      if (_ptrRenderClient == NULL || _ptrClientOut == NULL) {
+        _UnLock();
+        RTC_LOG(LS_ERROR)
+            << "output state has been modified during unlocked period";
+        goto Exit;
+      }
+
+      // Get the number of frames of padding (queued up to play) in the endpoint
+      // buffer.
+      UINT32 padding = 0;
+      hr = _ptrClientOut->GetCurrentPadding(&padding);
+      EXIT_ON_ERROR(hr);
+
+      // Derive the amount of available space in the output buffer
+      uint32_t framesAvailable = bufferLength - padding;
+
+      // Do we have 10 ms available in the render buffer?
+      if (framesAvailable < _playBlockSize) {
+        // Not enough space in render buffer to store next render packet.
+        _UnLock();
+        break;
+      }
+
+      // Write n*10ms buffers to the render buffer
+      const uint32_t n10msBuffers = (framesAvailable / _playBlockSize);
+      for (uint32_t n = 0; n < n10msBuffers; n++) {
+        // Get pointer (i.e., grab the buffer) to next space in the shared
+        // render buffer.
+        hr = _ptrRenderClient->GetBuffer(_playBlockSize, &pData);
+        EXIT_ON_ERROR(hr);
+
+        if (_ptrAudioBuffer) {
+          // Request data to be played out (#bytes =
+          // _playBlockSize*_audioFrameSize)
+          _UnLock();
+          int32_t nSamples =
+              _ptrAudioBuffer->RequestPlayoutData(_playBlockSize);
+          _Lock();
+
+          if (nSamples == -1) {
+            _UnLock();
+            RTC_LOG(LS_ERROR) << "failed to read data from render client";
+            goto Exit;
+          }
+
+          // Sanity check to ensure that essential states are not modified
+          // during the unlocked period
+          if (_ptrRenderClient == NULL || _ptrClientOut == NULL) {
+            _UnLock();
+            RTC_LOG(LS_ERROR)
+                << "output state has been modified during unlocked"
+                   " period";
+            goto Exit;
+          }
+          if (nSamples != static_cast<int32_t>(_playBlockSize)) {
+            RTC_LOG(LS_WARNING)
+                << "nSamples(" << nSamples << ") != _playBlockSize"
+                << _playBlockSize << ")";
+          }
+
+          // Get the actual (stored) data
+          nSamples = _ptrAudioBuffer->GetPlayoutData((int8_t*)pData);
+        }
+
+        DWORD dwFlags(0);
+        hr = _ptrRenderClient->ReleaseBuffer(_playBlockSize, dwFlags);
+        // See http://msdn.microsoft.com/en-us/library/dd316605(VS.85).aspx
+        // for more details regarding AUDCLNT_E_DEVICE_INVALIDATED.
+        EXIT_ON_ERROR(hr);
+
+        _writtenSamples += _playBlockSize;
+      }
+
+      // Check the current delay on the playout side.
+      if (clock) {
+        UINT64 pos = 0;
+        UINT64 freq = 1;
+        clock->GetPosition(&pos, NULL);
+        clock->GetFrequency(&freq);
+        playout_delay = ROUND((double(_writtenSamples) / _devicePlaySampleRate -
+                               double(pos) / freq) *
+                              1000.0);
+        _sndCardPlayDelay = playout_delay;
+      }
+
+      _UnLock();
+    }
+  }
+
+  // ------------------ THREAD LOOP ------------------ <<
+
+  SleepMs(static_cast<DWORD>(endpointBufferSizeMS + 0.5));
+  hr = _ptrClientOut->Stop();
+
+Exit:
+  SAFE_RELEASE(clock);
+
+  if (FAILED(hr)) {
+    _ptrClientOut->Stop();
+    _UnLock();
+    _TraceCOMError(hr);
+  }
+
+  if (_winSupportAvrt) {
+    if (NULL != hMmTask) {
+      _PAvRevertMmThreadCharacteristics(hMmTask);
+    }
+  }
+
+  _Lock();
+
+  if (keepPlaying) {
+    if (_ptrClientOut != NULL) {
+      hr = _ptrClientOut->Stop();
+      if (FAILED(hr)) {
+        _TraceCOMError(hr);
+      }
+      hr = _ptrClientOut->Reset();
+      if (FAILED(hr)) {
+        _TraceCOMError(hr);
+      }
+    }
+    RTC_LOG(LS_ERROR)
+        << "Playout error: rendering thread has ended pre-maturely";
+  } else {
+    RTC_LOG(LS_VERBOSE) << "_Rendering thread is now terminated properly";
+  }
+
+  _UnLock();
+
+  return (DWORD)hr;
+}
+
+DWORD AudioDeviceWindowsCore::InitCaptureThreadPriority() {
+  _hMmTask = NULL;
+
+  rtc::SetCurrentThreadName("webrtc_core_audio_capture_thread");
+
+  // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread
+  // priority.
+  if (_winSupportAvrt) {
+    DWORD taskIndex(0);
+    _hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
+    if (_hMmTask) {
+      if (!_PAvSetMmThreadPriority(_hMmTask, AVRT_PRIORITY_CRITICAL)) {
+        RTC_LOG(LS_WARNING) << "failed to boost rec-thread using MMCSS";
+      }
+      RTC_LOG(LS_VERBOSE)
+          << "capture thread is now registered with MMCSS (taskIndex="
+          << taskIndex << ")";
+    } else {
+      RTC_LOG(LS_WARNING) << "failed to enable MMCSS on capture thread (err="
+                          << GetLastError() << ")";
+      _TraceCOMError(GetLastError());
+    }
+  }
+
+  return S_OK;
+}
+
+void AudioDeviceWindowsCore::RevertCaptureThreadPriority() {
+  if (_winSupportAvrt) {
+    if (NULL != _hMmTask) {
+      _PAvRevertMmThreadCharacteristics(_hMmTask);
+    }
+  }
+
+  _hMmTask = NULL;
+}
+
+DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO() {
+  RTC_DCHECK(_mediaBuffer);
+  bool keepRecording = true;
+
+  // Initialize COM as MTA in this thread.
+  ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+  if (!comInit.Succeeded()) {
+    RTC_LOG(LS_ERROR) << "failed to initialize COM in polling DMO thread";
+    return 1;
+  }
+
+  HRESULT hr = InitCaptureThreadPriority();
+  if (FAILED(hr)) {
+    return hr;
+  }
+
+  // Set event which will ensure that the calling thread modifies the
+  // recording state to true.
+  SetEvent(_hCaptureStartedEvent);
+
+  // >> ---------------------------- THREAD LOOP ----------------------------
+  while (keepRecording) {
+    // Poll the DMO every 5 ms.
+    // (The same interval used in the Wave implementation.)
+    DWORD waitResult = WaitForSingleObject(_hShutdownCaptureEvent, 5);
+    switch (waitResult) {
+      case WAIT_OBJECT_0:  // _hShutdownCaptureEvent
+        keepRecording = false;
+        break;
+      case WAIT_TIMEOUT:  // timeout notification
+        break;
+      default:  // unexpected error
+        RTC_LOG(LS_WARNING) << "Unknown wait termination on capture side";
+        hr = -1;  // To signal an error callback.
+        keepRecording = false;
+        break;
+    }
+
+    while (keepRecording) {
+      MutexLock lockScoped(&mutex_);
+
+      DWORD dwStatus = 0;
+      {
+        DMO_OUTPUT_DATA_BUFFER dmoBuffer = {0};
+        dmoBuffer.pBuffer = _mediaBuffer.get();
+        dmoBuffer.pBuffer->AddRef();
+
+        // Poll the DMO for AEC processed capture data. The DMO will
+        // copy available data to `dmoBuffer`, and should only return
+        // 10 ms frames. The value of `dwStatus` should be ignored.
+        hr = _dmo->ProcessOutput(0, 1, &dmoBuffer, &dwStatus);
+        SAFE_RELEASE(dmoBuffer.pBuffer);
+        dwStatus = dmoBuffer.dwStatus;
+      }
+      if (FAILED(hr)) {
+        _TraceCOMError(hr);
+        keepRecording = false;
+        RTC_DCHECK_NOTREACHED();
+        break;
+      }
+
+      ULONG bytesProduced = 0;
+      BYTE* data;
+      // Get a pointer to the data buffer. This should be valid until
+      // the next call to ProcessOutput.
+      hr = _mediaBuffer->GetBufferAndLength(&data, &bytesProduced);
+      if (FAILED(hr)) {
+        _TraceCOMError(hr);
+        keepRecording = false;
+        RTC_DCHECK_NOTREACHED();
+        break;
+      }
+
+      if (bytesProduced > 0) {
+        const int kSamplesProduced = bytesProduced / _recAudioFrameSize;
+        // TODO(andrew): verify that this is always satisfied. It might
+        // be that ProcessOutput will try to return more than 10 ms if
+        // we fail to call it frequently enough.
+        RTC_DCHECK_EQ(kSamplesProduced, static_cast<int>(_recBlockSize));
+        RTC_DCHECK_EQ(sizeof(BYTE), sizeof(int8_t));
+        _ptrAudioBuffer->SetRecordedBuffer(reinterpret_cast<int8_t*>(data),
+                                           kSamplesProduced);
+        _ptrAudioBuffer->SetVQEData(0, 0);
+
+        _UnLock();  // Release lock while making the callback.
+        _ptrAudioBuffer->DeliverRecordedData();
+        _Lock();
+      }
+
+      // Reset length to indicate buffer availability.
+      hr = _mediaBuffer->SetLength(0);
+      if (FAILED(hr)) {
+        _TraceCOMError(hr);
+        keepRecording = false;
+        RTC_DCHECK_NOTREACHED();
+        break;
+      }
+
+      if (!(dwStatus & DMO_OUTPUT_DATA_BUFFERF_INCOMPLETE)) {
+        // The DMO cannot currently produce more data. This is the
+        // normal case; otherwise it means the DMO had more than 10 ms
+        // of data available and ProcessOutput should be called again.
+        break;
+      }
+    }
+  }
+  // ---------------------------- THREAD LOOP ---------------------------- <<
+
+  RevertCaptureThreadPriority();
+
+  if (FAILED(hr)) {
+    RTC_LOG(LS_ERROR)
+        << "Recording error: capturing thread has ended prematurely";
+  } else {
+    RTC_LOG(LS_VERBOSE) << "Capturing thread is now terminated properly";
+  }
+
+  return hr;
+}
+
+// ----------------------------------------------------------------------------
+//  DoCaptureThread
+// ----------------------------------------------------------------------------
+
+DWORD AudioDeviceWindowsCore::DoCaptureThread() {
+  bool keepRecording = true;
+  HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hCaptureSamplesReadyEvent};
+  HRESULT hr = S_OK;
+
+  LARGE_INTEGER t1;
+
+  BYTE* syncBuffer = NULL;
+  UINT32 syncBufIndex = 0;
+
+  _readSamples = 0;
+
+  // Initialize COM as MTA in this thread.
+  ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
+  if (!comInit.Succeeded()) {
+    RTC_LOG(LS_ERROR) << "failed to initialize COM in capture thread";
+    return 1;
+  }
+
+  hr = InitCaptureThreadPriority();
+  if (FAILED(hr)) {
+    return hr;
+  }
+
+  _Lock();
+
+  // Get size of capturing buffer (length is expressed as the number of audio
+  // frames the buffer can hold). This value is fixed during the capturing
+  // session.
+  //
+  UINT32 bufferLength = 0;
+  if (_ptrClientIn == NULL) {
+    RTC_LOG(LS_ERROR)
+        << "input state has been modified before capture loop starts.";
+    return 1;
+  }
+  hr = _ptrClientIn->GetBufferSize(&bufferLength);
+  EXIT_ON_ERROR(hr);
+  RTC_LOG(LS_VERBOSE) << "[CAPT] size of buffer       : " << bufferLength;
+
+  // Allocate memory for sync buffer.
+  // It is used for compensation between native 44.1 and internal 44.0 and
+  // for cases when the capture buffer is larger than 10ms.
+  //
+  const UINT32 syncBufferSize = 2 * (bufferLength * _recAudioFrameSize);
+  syncBuffer = new BYTE[syncBufferSize];
+  if (syncBuffer == NULL) {
+    return (DWORD)E_POINTER;
+  }
+  RTC_LOG(LS_VERBOSE) << "[CAPT] size of sync buffer  : " << syncBufferSize
+                      << " [bytes]";
+
+  // Get maximum latency for the current stream (will not change for the
+  // lifetime of the IAudioClient object).
+  //
+  REFERENCE_TIME latency;
+  _ptrClientIn->GetStreamLatency(&latency);
+  RTC_LOG(LS_VERBOSE) << "[CAPT] max stream latency   : " << (DWORD)latency
+                      << " (" << (double)(latency / 10000.0) << " ms)";
+
+  // Get the length of the periodic interval separating successive processing
+  // passes by the audio engine on the data in the endpoint buffer.
+  //
+  REFERENCE_TIME devPeriod = 0;
+  REFERENCE_TIME devPeriodMin = 0;
+  _ptrClientIn->GetDevicePeriod(&devPeriod, &devPeriodMin);
+  RTC_LOG(LS_VERBOSE) << "[CAPT] device period        : " << (DWORD)devPeriod
+                      << " (" << (double)(devPeriod / 10000.0) << " ms)";
+
+  double extraDelayMS = (double)((latency + devPeriod) / 10000.0);
+  RTC_LOG(LS_VERBOSE) << "[CAPT] extraDelayMS         : " << extraDelayMS;
+
+  double endpointBufferSizeMS =
+      10.0 * ((double)bufferLength / (double)_recBlockSize);
+  RTC_LOG(LS_VERBOSE) << "[CAPT] endpointBufferSizeMS : "
+                      << endpointBufferSizeMS;
+
+  // Start up the capturing stream.
+  //
+  hr = _ptrClientIn->Start();
+  EXIT_ON_ERROR(hr);
+
+  _UnLock();
+
+  // Set event which will ensure that the calling thread modifies the recording
+  // state to true.
+  //
+  SetEvent(_hCaptureStartedEvent);
+
+  // >> ---------------------------- THREAD LOOP ----------------------------
+
+  while (keepRecording) {
+    // Wait for a capture notification event or a shutdown event
+    DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
+    switch (waitResult) {
+      case WAIT_OBJECT_0 + 0:  // _hShutdownCaptureEvent
+        keepRecording = false;
+        break;
+      case WAIT_OBJECT_0 + 1:  // _hCaptureSamplesReadyEvent
+        break;
+      case WAIT_TIMEOUT:  // timeout notification
+        RTC_LOG(LS_WARNING) << "capture event timed out after 0.5 seconds";
+        goto Exit;
+      default:  // unexpected error
+        RTC_LOG(LS_WARNING) << "unknown wait termination on capture side";
+        goto Exit;
+    }
+
+    while (keepRecording) {
+      BYTE* pData = 0;
+      UINT32 framesAvailable = 0;
+      DWORD flags = 0;
+      UINT64 recTime = 0;
+      UINT64 recPos = 0;
+
+      _Lock();
+
+      // Sanity check to ensure that essential states are not modified
+      // during the unlocked period.
+      if (_ptrCaptureClient == NULL || _ptrClientIn == NULL) {
+        _UnLock();
+        RTC_LOG(LS_ERROR)
+            << "input state has been modified during unlocked period";
+        goto Exit;
+      }
+
+      //  Find out how much capture data is available
+      //
+      hr = _ptrCaptureClient->GetBuffer(
+          &pData,            // packet which is ready to be read by used
+          &framesAvailable,  // #frames in the captured packet (can be zero)
+          &flags,            // support flags (check)
+          &recPos,    // device position of first audio frame in data packet
+          &recTime);  // value of performance counter at the time of recording
+                      // the first audio frame
+
+      if (SUCCEEDED(hr)) {
+        if (AUDCLNT_S_BUFFER_EMPTY == hr) {
+          // Buffer was empty => start waiting for a new capture notification
+          // event
+          _UnLock();
+          break;
+        }
+
+        if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
+          // Treat all of the data in the packet as silence and ignore the
+          // actual data values.
+          RTC_LOG(LS_WARNING) << "AUDCLNT_BUFFERFLAGS_SILENT";
+          pData = NULL;
+        }
+
+        RTC_DCHECK_NE(framesAvailable, 0);
+
+        if (pData) {
+          CopyMemory(&syncBuffer[syncBufIndex * _recAudioFrameSize], pData,
+                     framesAvailable * _recAudioFrameSize);
+        } else {
+          ZeroMemory(&syncBuffer[syncBufIndex * _recAudioFrameSize],
+                     framesAvailable * _recAudioFrameSize);
+        }
+        RTC_DCHECK_GE(syncBufferSize, (syncBufIndex * _recAudioFrameSize) +
+                                          framesAvailable * _recAudioFrameSize);
+
+        // Release the capture buffer
+        //
+        hr = _ptrCaptureClient->ReleaseBuffer(framesAvailable);
+        EXIT_ON_ERROR(hr);
+
+        _readSamples += framesAvailable;
+        syncBufIndex += framesAvailable;
+
+        QueryPerformanceCounter(&t1);
+
+        // Get the current recording and playout delay.
+        uint32_t sndCardRecDelay = (uint32_t)(
+            ((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime) / 10000) +
+            (10 * syncBufIndex) / _recBlockSize - 10);
+        uint32_t sndCardPlayDelay = static_cast<uint32_t>(_sndCardPlayDelay);
+
+        while (syncBufIndex >= _recBlockSize) {
+          if (_ptrAudioBuffer) {
+            _ptrAudioBuffer->SetRecordedBuffer((const int8_t*)syncBuffer,
+                                               _recBlockSize);
+            _ptrAudioBuffer->SetVQEData(sndCardPlayDelay, sndCardRecDelay);
+
+            _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
+            _UnLock();  // release lock while making the callback
+            _ptrAudioBuffer->DeliverRecordedData();
+            _Lock();  // restore the lock
+
+            // Sanity check to ensure that essential states are not modified
+            // during the unlocked period
+            if (_ptrCaptureClient == NULL || _ptrClientIn == NULL) {
+              _UnLock();
+              RTC_LOG(LS_ERROR) << "input state has been modified during"
+                                   " unlocked period";
+              goto Exit;
+            }
+          }
+
+          // store remaining data which was not able to deliver as 10ms segment
+          MoveMemory(&syncBuffer[0],
+                     &syncBuffer[_recBlockSize * _recAudioFrameSize],
+                     (syncBufIndex - _recBlockSize) * _recAudioFrameSize);
+          syncBufIndex -= _recBlockSize;
+          sndCardRecDelay -= 10;
+        }
+      } else {
+        // If GetBuffer returns AUDCLNT_E_BUFFER_ERROR, the thread consuming the
+        // audio samples must wait for the next processing pass. The client
+        // might benefit from keeping a count of the failed GetBuffer calls. If
+        // GetBuffer returns this error repeatedly, the client can start a new
+        // processing loop after shutting down the current client by calling
+        // IAudioClient::Stop, IAudioClient::Reset, and releasing the audio
+        // client.
+        RTC_LOG(LS_ERROR) << "IAudioCaptureClient::GetBuffer returned"
+                             " AUDCLNT_E_BUFFER_ERROR, hr = 0x"
+                          << rtc::ToHex(hr);
+        goto Exit;
+      }
+
+      _UnLock();
+    }
+  }
+
+  // ---------------------------- THREAD LOOP ---------------------------- <<
+
+  if (_ptrClientIn) {
+    hr = _ptrClientIn->Stop();
+  }
+
+Exit:
+  if (FAILED(hr)) {
+    _ptrClientIn->Stop();
+    _UnLock();
+    _TraceCOMError(hr);
+  }
+
+  RevertCaptureThreadPriority();
+
+  _Lock();
+
+  if (keepRecording) {
+    if (_ptrClientIn != NULL) {
+      hr = _ptrClientIn->Stop();
+      if (FAILED(hr)) {
+        _TraceCOMError(hr);
+      }
+      hr = _ptrClientIn->Reset();
+      if (FAILED(hr)) {
+        _TraceCOMError(hr);
+      }
+    }
+
+    RTC_LOG(LS_ERROR)
+        << "Recording error: capturing thread has ended pre-maturely";
+  } else {
+    RTC_LOG(LS_VERBOSE) << "_Capturing thread is now terminated properly";
+  }
+
+  SAFE_RELEASE(_ptrClientIn);
+  SAFE_RELEASE(_ptrCaptureClient);
+
+  _UnLock();
+
+  if (syncBuffer) {
+    delete[] syncBuffer;
+  }
+
+  return (DWORD)hr;
+}
+
+int32_t AudioDeviceWindowsCore::EnableBuiltInAEC(bool enable) {
+  if (_recIsInitialized) {
+    RTC_LOG(LS_ERROR)
+        << "Attempt to set Windows AEC with recording already initialized";
+    return -1;
+  }
+
+  if (_dmo == NULL) {
+    RTC_LOG(LS_ERROR)
+        << "Built-in AEC DMO was not initialized properly at create time";
+    return -1;
+  }
+
+  _builtInAecEnabled = enable;
+  return 0;
+}
+
+void AudioDeviceWindowsCore::_Lock() RTC_NO_THREAD_SAFETY_ANALYSIS {
+  mutex_.Lock();
+}
+
+void AudioDeviceWindowsCore::_UnLock() RTC_NO_THREAD_SAFETY_ANALYSIS {
+  mutex_.Unlock();
+}
+
+int AudioDeviceWindowsCore::SetDMOProperties() {
+  HRESULT hr = S_OK;
+  RTC_DCHECK(_dmo);
+
+  rtc::scoped_refptr<IPropertyStore> ps;
+  {
+    IPropertyStore* ptrPS = NULL;
+    hr = _dmo->QueryInterface(IID_IPropertyStore,
+                              reinterpret_cast<void**>(&ptrPS));
+    if (FAILED(hr) || ptrPS == NULL) {
+      _TraceCOMError(hr);
+      return -1;
+    }
+    ps = ptrPS;
+    SAFE_RELEASE(ptrPS);
+  }
+
+  // Set the AEC system mode.
+  // SINGLE_CHANNEL_AEC - AEC processing only.
+  if (SetVtI4Property(ps.get(), MFPKEY_WMAAECMA_SYSTEM_MODE,
+                      SINGLE_CHANNEL_AEC)) {
+    return -1;
+  }
+
+  // Set the AEC source mode.
+  // VARIANT_TRUE - Source mode (we poll the AEC for captured data).
+  if (SetBoolProperty(ps.get(), MFPKEY_WMAAECMA_DMO_SOURCE_MODE,
+                      VARIANT_TRUE) == -1) {
+    return -1;
+  }
+
+  // Enable the feature mode.
+  // This lets us override all the default processing settings below.
+  if (SetBoolProperty(ps.get(), MFPKEY_WMAAECMA_FEATURE_MODE, VARIANT_TRUE) ==
+      -1) {
+    return -1;
+  }
+
+  // Disable analog AGC (default enabled).
+  if (SetBoolProperty(ps.get(), MFPKEY_WMAAECMA_MIC_GAIN_BOUNDER,
+                      VARIANT_FALSE) == -1) {
+    return -1;
+  }
+
+  // Disable noise suppression (default enabled).
+  // 0 - Disabled, 1 - Enabled
+  if (SetVtI4Property(ps.get(), MFPKEY_WMAAECMA_FEATR_NS, 0) == -1) {
+    return -1;
+  }
+
+  // Relevant parameters to leave at default settings:
+  // MFPKEY_WMAAECMA_FEATR_AGC - Digital AGC (disabled).
+  // MFPKEY_WMAAECMA_FEATR_CENTER_CLIP - AEC center clipping (enabled).
+  // MFPKEY_WMAAECMA_FEATR_ECHO_LENGTH - Filter length (256 ms).
+  // TODO(andrew): investigate decresing the length to 128 ms.
+  // MFPKEY_WMAAECMA_FEATR_FRAME_SIZE - Frame size (0).
+  //   0 is automatic; defaults to 160 samples (or 10 ms frames at the
+  //   selected 16 kHz) as long as mic array processing is disabled.
+  // MFPKEY_WMAAECMA_FEATR_NOISE_FILL - Comfort noise (enabled).
+  // MFPKEY_WMAAECMA_FEATR_VAD - VAD (disabled).
+
+  // Set the devices selected by VoE. If using a default device, we need to
+  // search for the device index.
+  int inDevIndex = _inputDeviceIndex;
+  int outDevIndex = _outputDeviceIndex;
+  if (!_usingInputDeviceIndex) {
+    ERole role = eCommunications;
+    if (_inputDevice == AudioDeviceModule::kDefaultDevice) {
+      role = eConsole;
+    }
+
+    if (_GetDefaultDeviceIndex(eCapture, role, &inDevIndex) == -1) {
+      return -1;
+    }
+  }
+
+  if (!_usingOutputDeviceIndex) {
+    ERole role = eCommunications;
+    if (_outputDevice == AudioDeviceModule::kDefaultDevice) {
+      role = eConsole;
+    }
+
+    if (_GetDefaultDeviceIndex(eRender, role, &outDevIndex) == -1) {
+      return -1;
+    }
+  }
+
+  DWORD devIndex = static_cast<uint32_t>(outDevIndex << 16) +
+                   static_cast<uint32_t>(0x0000ffff & inDevIndex);
+  RTC_LOG(LS_VERBOSE) << "Capture device index: " << inDevIndex
+                      << ", render device index: " << outDevIndex;
+  if (SetVtI4Property(ps.get(), MFPKEY_WMAAECMA_DEVICE_INDEXES, devIndex) ==
+      -1) {
+    return -1;
+  }
+
+  return 0;
+}
+
+int AudioDeviceWindowsCore::SetBoolProperty(IPropertyStore* ptrPS,
+                                            REFPROPERTYKEY key,
+                                            VARIANT_BOOL value) {
+  PROPVARIANT pv;
+  PropVariantInit(&pv);
+  pv.vt = VT_BOOL;
+  pv.boolVal = value;
+  HRESULT hr = ptrPS->SetValue(key, pv);
+  PropVariantClear(&pv);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    return -1;
+  }
+  return 0;
+}
+
+int AudioDeviceWindowsCore::SetVtI4Property(IPropertyStore* ptrPS,
+                                            REFPROPERTYKEY key,
+                                            LONG value) {
+  PROPVARIANT pv;
+  PropVariantInit(&pv);
+  pv.vt = VT_I4;
+  pv.lVal = value;
+  HRESULT hr = ptrPS->SetValue(key, pv);
+  PropVariantClear(&pv);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    return -1;
+  }
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  _RefreshDeviceList
+//
+//  Creates a new list of endpoint rendering or capture devices after
+//  deleting any previously created (and possibly out-of-date) list of
+//  such devices.
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_RefreshDeviceList(EDataFlow dir) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  HRESULT hr = S_OK;
+  IMMDeviceCollection* pCollection = NULL;
+
+  RTC_DCHECK(dir == eRender || dir == eCapture);
+  RTC_DCHECK(_ptrEnumerator);
+
+  // Create a fresh list of devices using the specified direction
+  hr = _ptrEnumerator->EnumAudioEndpoints(dir, DEVICE_STATE_ACTIVE,
+                                          &pCollection);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(pCollection);
+    return -1;
+  }
+
+  if (dir == eRender) {
+    SAFE_RELEASE(_ptrRenderCollection);
+    _ptrRenderCollection = pCollection;
+  } else {
+    SAFE_RELEASE(_ptrCaptureCollection);
+    _ptrCaptureCollection = pCollection;
+  }
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  _DeviceListCount
+//
+//  Gets a count of the endpoint rendering or capture devices in the
+//  current list of such devices.
+// ----------------------------------------------------------------------------
+
+int16_t AudioDeviceWindowsCore::_DeviceListCount(EDataFlow dir) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  HRESULT hr = S_OK;
+  UINT count = 0;
+
+  RTC_DCHECK(eRender == dir || eCapture == dir);
+
+  if (eRender == dir && NULL != _ptrRenderCollection) {
+    hr = _ptrRenderCollection->GetCount(&count);
+  } else if (NULL != _ptrCaptureCollection) {
+    hr = _ptrCaptureCollection->GetCount(&count);
+  }
+
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    return -1;
+  }
+
+  return static_cast<int16_t>(count);
+}
+
+// ----------------------------------------------------------------------------
+//  _GetListDeviceName
+//
+//  Gets the friendly name of an endpoint rendering or capture device
+//  from the current list of such devices. The caller uses an index
+//  into the list to identify the device.
+//
+//  Uses: _ptrRenderCollection or _ptrCaptureCollection which is updated
+//  in _RefreshDeviceList().
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_GetListDeviceName(EDataFlow dir,
+                                                   int index,
+                                                   LPWSTR szBuffer,
+                                                   int bufferLen) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  HRESULT hr = S_OK;
+  IMMDevice* pDevice = NULL;
+
+  RTC_DCHECK(dir == eRender || dir == eCapture);
+
+  if (eRender == dir && NULL != _ptrRenderCollection) {
+    hr = _ptrRenderCollection->Item(index, &pDevice);
+  } else if (NULL != _ptrCaptureCollection) {
+    hr = _ptrCaptureCollection->Item(index, &pDevice);
+  }
+
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(pDevice);
+    return -1;
+  }
+
+  int32_t res = _GetDeviceName(pDevice, szBuffer, bufferLen);
+  SAFE_RELEASE(pDevice);
+  return res;
+}
+
+// ----------------------------------------------------------------------------
+//  _GetDefaultDeviceName
+//
+//  Gets the friendly name of an endpoint rendering or capture device
+//  given a specified device role.
+//
+//  Uses: _ptrEnumerator
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_GetDefaultDeviceName(EDataFlow dir,
+                                                      ERole role,
+                                                      LPWSTR szBuffer,
+                                                      int bufferLen) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  HRESULT hr = S_OK;
+  IMMDevice* pDevice = NULL;
+
+  RTC_DCHECK(dir == eRender || dir == eCapture);
+  RTC_DCHECK(role == eConsole || role == eCommunications);
+  RTC_DCHECK(_ptrEnumerator);
+
+  hr = _ptrEnumerator->GetDefaultAudioEndpoint(dir, role, &pDevice);
+
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(pDevice);
+    return -1;
+  }
+
+  int32_t res = _GetDeviceName(pDevice, szBuffer, bufferLen);
+  SAFE_RELEASE(pDevice);
+  return res;
+}
+
+// ----------------------------------------------------------------------------
+//  _GetListDeviceID
+//
+//  Gets the unique ID string of an endpoint rendering or capture device
+//  from the current list of such devices. The caller uses an index
+//  into the list to identify the device.
+//
+//  Uses: _ptrRenderCollection or _ptrCaptureCollection which is updated
+//  in _RefreshDeviceList().
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_GetListDeviceID(EDataFlow dir,
+                                                 int index,
+                                                 LPWSTR szBuffer,
+                                                 int bufferLen) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  HRESULT hr = S_OK;
+  IMMDevice* pDevice = NULL;
+
+  RTC_DCHECK(dir == eRender || dir == eCapture);
+
+  if (eRender == dir && NULL != _ptrRenderCollection) {
+    hr = _ptrRenderCollection->Item(index, &pDevice);
+  } else if (NULL != _ptrCaptureCollection) {
+    hr = _ptrCaptureCollection->Item(index, &pDevice);
+  }
+
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(pDevice);
+    return -1;
+  }
+
+  int32_t res = _GetDeviceID(pDevice, szBuffer, bufferLen);
+  SAFE_RELEASE(pDevice);
+  return res;
+}
+
+// ----------------------------------------------------------------------------
+//  _GetDefaultDeviceID
+//
+//  Gets the uniqe device ID of an endpoint rendering or capture device
+//  given a specified device role.
+//
+//  Uses: _ptrEnumerator
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_GetDefaultDeviceID(EDataFlow dir,
+                                                    ERole role,
+                                                    LPWSTR szBuffer,
+                                                    int bufferLen) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  HRESULT hr = S_OK;
+  IMMDevice* pDevice = NULL;
+
+  RTC_DCHECK(dir == eRender || dir == eCapture);
+  RTC_DCHECK(role == eConsole || role == eCommunications);
+  RTC_DCHECK(_ptrEnumerator);
+
+  hr = _ptrEnumerator->GetDefaultAudioEndpoint(dir, role, &pDevice);
+
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(pDevice);
+    return -1;
+  }
+
+  int32_t res = _GetDeviceID(pDevice, szBuffer, bufferLen);
+  SAFE_RELEASE(pDevice);
+  return res;
+}
+
+int32_t AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
+                                                       ERole role,
+                                                       int* index) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  HRESULT hr = S_OK;
+  WCHAR szDefaultDeviceID[MAX_PATH] = {0};
+  WCHAR szDeviceID[MAX_PATH] = {0};
+
+  const size_t kDeviceIDLength = sizeof(szDeviceID) / sizeof(szDeviceID[0]);
+  RTC_DCHECK_EQ(kDeviceIDLength,
+                sizeof(szDefaultDeviceID) / sizeof(szDefaultDeviceID[0]));
+
+  if (_GetDefaultDeviceID(dir, role, szDefaultDeviceID, kDeviceIDLength) ==
+      -1) {
+    return -1;
+  }
+
+  IMMDeviceCollection* collection = _ptrCaptureCollection;
+  if (dir == eRender) {
+    collection = _ptrRenderCollection;
+  }
+
+  if (!collection) {
+    RTC_LOG(LS_ERROR) << "Device collection not valid";
+    return -1;
+  }
+
+  UINT count = 0;
+  hr = collection->GetCount(&count);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    return -1;
+  }
+
+  *index = -1;
+  for (UINT i = 0; i < count; i++) {
+    memset(szDeviceID, 0, sizeof(szDeviceID));
+    rtc::scoped_refptr<IMMDevice> device;
+    {
+      IMMDevice* ptrDevice = NULL;
+      hr = collection->Item(i, &ptrDevice);
+      if (FAILED(hr) || ptrDevice == NULL) {
+        _TraceCOMError(hr);
+        return -1;
+      }
+      device = ptrDevice;
+      SAFE_RELEASE(ptrDevice);
+    }
+
+    if (_GetDeviceID(device.get(), szDeviceID, kDeviceIDLength) == -1) {
+      return -1;
+    }
+
+    if (wcsncmp(szDefaultDeviceID, szDeviceID, kDeviceIDLength) == 0) {
+      // Found a match.
+      *index = i;
+      break;
+    }
+  }
+
+  if (*index == -1) {
+    RTC_LOG(LS_ERROR) << "Unable to find collection index for default device";
+    return -1;
+  }
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  _GetDeviceName
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
+                                               LPWSTR pszBuffer,
+                                               int bufferLen) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  static const WCHAR szDefault[] = L"<Device not available>";
+
+  HRESULT hr = E_FAIL;
+  IPropertyStore* pProps = NULL;
+  PROPVARIANT varName;
+
+  RTC_DCHECK(pszBuffer);
+  RTC_DCHECK_GT(bufferLen, 0);
+
+  if (pDevice != NULL) {
+    hr = pDevice->OpenPropertyStore(STGM_READ, &pProps);
+    if (FAILED(hr)) {
+      RTC_LOG(LS_ERROR) << "IMMDevice::OpenPropertyStore failed, hr = 0x"
+                        << rtc::ToHex(hr);
+    }
+  }
+
+  // Initialize container for property value.
+  PropVariantInit(&varName);
+
+  if (SUCCEEDED(hr)) {
+    // Get the endpoint device's friendly-name property.
+    hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
+    if (FAILED(hr)) {
+      RTC_LOG(LS_ERROR) << "IPropertyStore::GetValue failed, hr = 0x"
+                        << rtc::ToHex(hr);
+    }
+  }
+
+  if ((SUCCEEDED(hr)) && (VT_EMPTY == varName.vt)) {
+    hr = E_FAIL;
+    RTC_LOG(LS_ERROR) << "IPropertyStore::GetValue returned no value,"
+                         " hr = 0x"
+                      << rtc::ToHex(hr);
+  }
+
+  if ((SUCCEEDED(hr)) && (VT_LPWSTR != varName.vt)) {
+    // The returned value is not a wide null terminated string.
+    hr = E_UNEXPECTED;
+    RTC_LOG(LS_ERROR) << "IPropertyStore::GetValue returned unexpected"
+                         " type, hr = 0x"
+                      << rtc::ToHex(hr);
+  }
+
+  if (SUCCEEDED(hr) && (varName.pwszVal != NULL)) {
+    // Copy the valid device name to the provided ouput buffer.
+    wcsncpy_s(pszBuffer, bufferLen, varName.pwszVal, _TRUNCATE);
+  } else {
+    // Failed to find the device name.
+    wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
+  }
+
+  PropVariantClear(&varName);
+  SAFE_RELEASE(pProps);
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  _GetDeviceID
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_GetDeviceID(IMMDevice* pDevice,
+                                             LPWSTR pszBuffer,
+                                             int bufferLen) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  static const WCHAR szDefault[] = L"<Device not available>";
+
+  HRESULT hr = E_FAIL;
+  LPWSTR pwszID = NULL;
+
+  RTC_DCHECK(pszBuffer);
+  RTC_DCHECK_GT(bufferLen, 0);
+
+  if (pDevice != NULL) {
+    hr = pDevice->GetId(&pwszID);
+  }
+
+  if (hr == S_OK) {
+    // Found the device ID.
+    wcsncpy_s(pszBuffer, bufferLen, pwszID, _TRUNCATE);
+  } else {
+    // Failed to find the device ID.
+    wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
+  }
+
+  CoTaskMemFree(pwszID);
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  _GetDefaultDevice
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_GetDefaultDevice(EDataFlow dir,
+                                                  ERole role,
+                                                  IMMDevice** ppDevice) {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  HRESULT hr(S_OK);
+
+  RTC_DCHECK(_ptrEnumerator);
+
+  hr = _ptrEnumerator->GetDefaultAudioEndpoint(dir, role, ppDevice);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    return -1;
+  }
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  _GetListDevice
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_GetListDevice(EDataFlow dir,
+                                               int index,
+                                               IMMDevice** ppDevice) {
+  HRESULT hr(S_OK);
+
+  RTC_DCHECK(_ptrEnumerator);
+
+  IMMDeviceCollection* pCollection = NULL;
+
+  hr = _ptrEnumerator->EnumAudioEndpoints(
+      dir,
+      DEVICE_STATE_ACTIVE,  // only active endpoints are OK
+      &pCollection);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(pCollection);
+    return -1;
+  }
+
+  hr = pCollection->Item(index, ppDevice);
+  if (FAILED(hr)) {
+    _TraceCOMError(hr);
+    SAFE_RELEASE(pCollection);
+    return -1;
+  }
+
+  SAFE_RELEASE(pCollection);
+
+  return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  _EnumerateEndpointDevicesAll
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(
+    EDataFlow dataFlow) const {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  RTC_DCHECK(_ptrEnumerator);
+
+  HRESULT hr = S_OK;
+  IMMDeviceCollection* pCollection = NULL;
+  IMMDevice* pEndpoint = NULL;
+  IPropertyStore* pProps = NULL;
+  IAudioEndpointVolume* pEndpointVolume = NULL;
+  LPWSTR pwszID = NULL;
+
+  // Generate a collection of audio endpoint devices in the system.
+  // Get states for *all* endpoint devices.
+  // Output: IMMDeviceCollection interface.
+  hr = _ptrEnumerator->EnumAudioEndpoints(
+      dataFlow,  // data-flow direction (input parameter)
+      DEVICE_STATE_ACTIVE | DEVICE_STATE_DISABLED | DEVICE_STATE_UNPLUGGED,
+      &pCollection);  // release interface when done
+
+  EXIT_ON_ERROR(hr);
+
+  // use the IMMDeviceCollection interface...
+
+  UINT count = 0;
+
+  // Retrieve a count of the devices in the device collection.
+  hr = pCollection->GetCount(&count);
+  EXIT_ON_ERROR(hr);
+  if (dataFlow == eRender)
+    RTC_LOG(LS_VERBOSE) << "#rendering endpoint devices (counting all): "
+                        << count;
+  else if (dataFlow == eCapture)
+    RTC_LOG(LS_VERBOSE) << "#capturing endpoint devices (counting all): "
+                        << count;
+
+  if (count == 0) {
+    return 0;
+  }
+
+  // Each loop prints the name of an endpoint device.
+  for (ULONG i = 0; i < count; i++) {
+    RTC_LOG(LS_VERBOSE) << "Endpoint " << i << ":";
+
+    // Get pointer to endpoint number i.
+    // Output: IMMDevice interface.
+    hr = pCollection->Item(i, &pEndpoint);
+    CONTINUE_ON_ERROR(hr);
+
+    // use the IMMDevice interface of the specified endpoint device...
+
+    // Get the endpoint ID string (uniquely identifies the device among all
+    // audio endpoint devices)
+    hr = pEndpoint->GetId(&pwszID);
+    CONTINUE_ON_ERROR(hr);
+    RTC_LOG(LS_VERBOSE) << "ID string    : " << pwszID;
+
+    // Retrieve an interface to the device's property store.
+    // Output: IPropertyStore interface.
+    hr = pEndpoint->OpenPropertyStore(STGM_READ, &pProps);
+    CONTINUE_ON_ERROR(hr);
+
+    // use the IPropertyStore interface...
+
+    PROPVARIANT varName;
+    // Initialize container for property value.
+    PropVariantInit(&varName);
+
+    // Get the endpoint's friendly-name property.
+    // Example: "Speakers (Realtek High Definition Audio)"
+    hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
+    CONTINUE_ON_ERROR(hr);
+    RTC_LOG(LS_VERBOSE) << "friendly name: \"" << varName.pwszVal << "\"";
+
+    // Get the endpoint's current device state
+    DWORD dwState;
+    hr = pEndpoint->GetState(&dwState);
+    CONTINUE_ON_ERROR(hr);
+    if (dwState & DEVICE_STATE_ACTIVE)
+      RTC_LOG(LS_VERBOSE) << "state (0x" << rtc::ToHex(dwState)
+                          << ")  : *ACTIVE*";
+    if (dwState & DEVICE_STATE_DISABLED)
+      RTC_LOG(LS_VERBOSE) << "state (0x" << rtc::ToHex(dwState)
+                          << ")  : DISABLED";
+    if (dwState & DEVICE_STATE_NOTPRESENT)
+      RTC_LOG(LS_VERBOSE) << "state (0x" << rtc::ToHex(dwState)
+                          << ")  : NOTPRESENT";
+    if (dwState & DEVICE_STATE_UNPLUGGED)
+      RTC_LOG(LS_VERBOSE) << "state (0x" << rtc::ToHex(dwState)
+                          << ")  : UNPLUGGED";
+
+    // Check the hardware volume capabilities.
+    DWORD dwHwSupportMask = 0;
+    hr = pEndpoint->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL,
+                             (void**)&pEndpointVolume);
+    CONTINUE_ON_ERROR(hr);
+    hr = pEndpointVolume->QueryHardwareSupport(&dwHwSupportMask);
+    CONTINUE_ON_ERROR(hr);
+    if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
+      // The audio endpoint device supports a hardware volume control
+      RTC_LOG(LS_VERBOSE) << "hwmask (0x" << rtc::ToHex(dwHwSupportMask)
+                          << ") : HARDWARE_SUPPORT_VOLUME";
+    if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_MUTE)
+      // The audio endpoint device supports a hardware mute control
+      RTC_LOG(LS_VERBOSE) << "hwmask (0x" << rtc::ToHex(dwHwSupportMask)
+                          << ") : HARDWARE_SUPPORT_MUTE";
+    if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_METER)
+      // The audio endpoint device supports a hardware peak meter
+      RTC_LOG(LS_VERBOSE) << "hwmask (0x" << rtc::ToHex(dwHwSupportMask)
+                          << ") : HARDWARE_SUPPORT_METER";
+
+    // Check the channel count (#channels in the audio stream that enters or
+    // leaves the audio endpoint device)
+    UINT nChannelCount(0);
+    hr = pEndpointVolume->GetChannelCount(&nChannelCount);
+    CONTINUE_ON_ERROR(hr);
+    RTC_LOG(LS_VERBOSE) << "#channels    : " << nChannelCount;
+
+    if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME) {
+      // Get the volume range.
+      float fLevelMinDB(0.0);
+      float fLevelMaxDB(0.0);
+      float fVolumeIncrementDB(0.0);
+      hr = pEndpointVolume->GetVolumeRange(&fLevelMinDB, &fLevelMaxDB,
+                                           &fVolumeIncrementDB);
+      CONTINUE_ON_ERROR(hr);
+      RTC_LOG(LS_VERBOSE) << "volume range : " << fLevelMinDB << " (min), "
+                          << fLevelMaxDB << " (max), " << fVolumeIncrementDB
+                          << " (inc) [dB]";
+
+      // The volume range from vmin = fLevelMinDB to vmax = fLevelMaxDB is
+      // divided into n uniform intervals of size vinc = fVolumeIncrementDB,
+      // where n = (vmax ?vmin) / vinc. The values vmin, vmax, and vinc are
+      // measured in decibels. The client can set the volume level to one of n +
+      // 1 discrete values in the range from vmin to vmax.
+      int n = (int)((fLevelMaxDB - fLevelMinDB) / fVolumeIncrementDB);
+      RTC_LOG(LS_VERBOSE) << "#intervals   : " << n;
+
+      // Get information about the current step in the volume range.
+      // This method represents the volume level of the audio stream that enters
+      // or leaves the audio endpoint device as an index or "step" in a range of
+      // discrete volume levels. Output value nStepCount is the number of steps
+      // in the range. Output value nStep is the step index of the current
+      // volume level. If the number of steps is n = nStepCount, then step index
+      // nStep can assume values from 0 (minimum volume) to n ?1 (maximum
+      // volume).
+      UINT nStep(0);
+      UINT nStepCount(0);
+      hr = pEndpointVolume->GetVolumeStepInfo(&nStep, &nStepCount);
+      CONTINUE_ON_ERROR(hr);
+      RTC_LOG(LS_VERBOSE) << "volume steps : " << nStep << " (nStep), "
+                          << nStepCount << " (nStepCount)";
+    }
+  Next:
+    if (FAILED(hr)) {
+      RTC_LOG(LS_VERBOSE) << "Error when logging device information";
+    }
+    CoTaskMemFree(pwszID);
+    pwszID = NULL;
+    PropVariantClear(&varName);
+    SAFE_RELEASE(pProps);
+    SAFE_RELEASE(pEndpoint);
+    SAFE_RELEASE(pEndpointVolume);
+  }
+  SAFE_RELEASE(pCollection);
+  return 0;
+
+Exit:
+  _TraceCOMError(hr);
+  CoTaskMemFree(pwszID);
+  pwszID = NULL;
+  SAFE_RELEASE(pCollection);
+  SAFE_RELEASE(pEndpoint);
+  SAFE_RELEASE(pEndpointVolume);
+  SAFE_RELEASE(pProps);
+  return -1;
+}
+
+// ----------------------------------------------------------------------------
+//  _TraceCOMError
+// ----------------------------------------------------------------------------
+
+void AudioDeviceWindowsCore::_TraceCOMError(HRESULT hr) const {
+  wchar_t buf[MAXERRORLENGTH];
+  wchar_t errorText[MAXERRORLENGTH];
+
+  const DWORD dwFlags =
+      FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_IGNORE_INSERTS;
+  const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
+
+  // Gets the system's human readable message string for this HRESULT.
+  // All error message in English by default.
+  DWORD messageLength = ::FormatMessageW(dwFlags, 0, hr, dwLangID, errorText,
+                                         MAXERRORLENGTH, NULL);
+
+  RTC_DCHECK_LE(messageLength, MAXERRORLENGTH);
+
+  // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
+  for (; messageLength && ::isspace(errorText[messageLength - 1]);
+       --messageLength) {
+    errorText[messageLength - 1] = '\0';
+  }
+
+  RTC_LOG(LS_ERROR) << "Core Audio method failed (hr=" << hr << ")";
+  StringCchPrintfW(buf, MAXERRORLENGTH, L"Error details: ");
+  StringCchCatW(buf, MAXERRORLENGTH, errorText);
+  RTC_LOG(LS_ERROR) << rtc::ToUtf8(buf);
+}
+
+bool AudioDeviceWindowsCore::KeyPressed() const {
+  int key_down = 0;
+  for (int key = VK_SPACE; key < VK_NUMLOCK; key++) {
+    short res = GetAsyncKeyState(key);
+    key_down |= res & 0x1;  // Get the LSB
+  }
+  return (key_down > 0);
+}
+}  // namespace webrtc
+
+#endif  // WEBRTC_WINDOWS_CORE_AUDIO_BUILD
diff --git a/third_party/libwebrtc/modules/audio_device/win/audio_device_core_win.h b/third_party/libwebrtc/modules/audio_device/win/audio_device_core_win.h
new file mode 100644
index 0000000000..7e7ef21157
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/audio_device_core_win.h
@@ -0,0 +1,300 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_WIN_AUDIO_DEVICE_CORE_WIN_H_
+#define MODULES_AUDIO_DEVICE_WIN_AUDIO_DEVICE_CORE_WIN_H_
+
+#if (_MSC_VER >= 1400)  // only include for VS 2005 and higher
+
+#include "rtc_base/win32.h"
+
+#include "modules/audio_device/audio_device_generic.h"
+
+#include <wmcodecdsp.h>   // CLSID_CWMAudioAEC
+                          // (must be before audioclient.h)
+#include <audioclient.h>  // WASAPI
+#include <audiopolicy.h>
+#include <avrt.h>  // Avrt
+#include <endpointvolume.h>
+#include <mediaobj.h>     // IMediaObject
+#include <mmdeviceapi.h>  // MMDevice
+
+#include "api/scoped_refptr.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/win/scoped_com_initializer.h"
+
+// Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority
+#pragma comment(lib, "avrt.lib")
+// AVRT function pointers
+typedef BOOL(WINAPI* PAvRevertMmThreadCharacteristics)(HANDLE);
+typedef HANDLE(WINAPI* PAvSetMmThreadCharacteristicsA)(LPCSTR, LPDWORD);
+typedef BOOL(WINAPI* PAvSetMmThreadPriority)(HANDLE, AVRT_PRIORITY);
+
+namespace webrtc {
+
+const float MAX_CORE_SPEAKER_VOLUME = 255.0f;
+const float MIN_CORE_SPEAKER_VOLUME = 0.0f;
+const float MAX_CORE_MICROPHONE_VOLUME = 255.0f;
+const float MIN_CORE_MICROPHONE_VOLUME = 0.0f;
+const uint16_t CORE_SPEAKER_VOLUME_STEP_SIZE = 1;
+const uint16_t CORE_MICROPHONE_VOLUME_STEP_SIZE = 1;
+
+class AudioDeviceWindowsCore : public AudioDeviceGeneric {
+ public:
+  AudioDeviceWindowsCore();
+  ~AudioDeviceWindowsCore();
+
+  static bool CoreAudioIsSupported();
+
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const;
+
+  // Main initializaton and termination
+  virtual InitStatus Init() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool Initialized() const;
+
+  // Device enumeration
+  virtual int16_t PlayoutDevices() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int16_t RecordingDevices() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize])
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize])
+      RTC_LOCKS_EXCLUDED(mutex_);
+
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index) RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
+  virtual int32_t SetRecordingDevice(uint16_t index) RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) RTC_LOCKS_EXCLUDED(mutex_);
+
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool& available);
+  virtual int32_t InitPlayout() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool PlayoutIsInitialized() const;
+  virtual int32_t RecordingIsAvailable(bool& available);
+  virtual int32_t InitRecording() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool RecordingIsInitialized() const;
+
+  // Audio transport control
+  virtual int32_t StartPlayout() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool Playing() const;
+  virtual int32_t StartRecording() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t StopRecording();
+  virtual bool Recording() const;
+
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool SpeakerIsInitialized() const;
+  virtual int32_t InitMicrophone() RTC_LOCKS_EXCLUDED(mutex_);
+  virtual bool MicrophoneIsInitialized() const;
+
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetSpeakerVolume(uint32_t volume) RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SpeakerVolume(uint32_t& volume) const
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetMicrophoneVolume(uint32_t volume)
+      RTC_LOCKS_EXCLUDED(mutex_, volume_mutex_);
+  virtual int32_t MicrophoneVolume(uint32_t& volume) const
+      RTC_LOCKS_EXCLUDED(mutex_, volume_mutex_);
+  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetSpeakerMute(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SpeakerMute(bool& enabled) const;
+
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t SetMicrophoneMute(bool enable);
+  virtual int32_t MicrophoneMute(bool& enabled) const;
+
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool& available);
+  virtual int32_t SetStereoPlayout(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t StereoPlayout(bool& enabled) const;
+  virtual int32_t StereoRecordingIsAvailable(bool& available);
+  virtual int32_t SetStereoRecording(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  virtual int32_t StereoRecording(bool& enabled) const
+      RTC_LOCKS_EXCLUDED(mutex_);
+
+  // Delay information and control
+  virtual int32_t PlayoutDelay(uint16_t& delayMS) const
+      RTC_LOCKS_EXCLUDED(mutex_);
+
+  virtual bool BuiltInAECIsAvailable() const;
+
+  virtual int32_t EnableBuiltInAEC(bool enable);
+
+ public:
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+ private:
+  bool KeyPressed() const;
+
+ private:  // avrt function pointers
+  PAvRevertMmThreadCharacteristics _PAvRevertMmThreadCharacteristics;
+  PAvSetMmThreadCharacteristicsA _PAvSetMmThreadCharacteristicsA;
+  PAvSetMmThreadPriority _PAvSetMmThreadPriority;
+  HMODULE _avrtLibrary;
+  bool _winSupportAvrt;
+
+ private:  // thread functions
+  int32_t InitSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int16_t PlayoutDevicesLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int16_t RecordingDevicesLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  DWORD InitCaptureThreadPriority();
+  void RevertCaptureThreadPriority();
+  static DWORD WINAPI WSAPICaptureThread(LPVOID context);
+  DWORD DoCaptureThread();
+
+  static DWORD WINAPI WSAPICaptureThreadPollDMO(LPVOID context);
+  DWORD DoCaptureThreadPollDMO() RTC_LOCKS_EXCLUDED(mutex_);
+
+  static DWORD WINAPI WSAPIRenderThread(LPVOID context);
+  DWORD DoRenderThread();
+
+  void _Lock();
+  void _UnLock();
+
+  int SetDMOProperties();
+
+  int SetBoolProperty(IPropertyStore* ptrPS,
+                      REFPROPERTYKEY key,
+                      VARIANT_BOOL value);
+
+  int SetVtI4Property(IPropertyStore* ptrPS, REFPROPERTYKEY key, LONG value);
+
+  int32_t _EnumerateEndpointDevicesAll(EDataFlow dataFlow) const;
+  void _TraceCOMError(HRESULT hr) const;
+
+  int32_t _RefreshDeviceList(EDataFlow dir);
+  int16_t _DeviceListCount(EDataFlow dir);
+  int32_t _GetDefaultDeviceName(EDataFlow dir,
+                                ERole role,
+                                LPWSTR szBuffer,
+                                int bufferLen);
+  int32_t _GetListDeviceName(EDataFlow dir,
+                             int index,
+                             LPWSTR szBuffer,
+                             int bufferLen);
+  int32_t _GetDeviceName(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
+  int32_t _GetListDeviceID(EDataFlow dir,
+                           int index,
+                           LPWSTR szBuffer,
+                           int bufferLen);
+  int32_t _GetDefaultDeviceID(EDataFlow dir,
+                              ERole role,
+                              LPWSTR szBuffer,
+                              int bufferLen);
+  int32_t _GetDefaultDeviceIndex(EDataFlow dir, ERole role, int* index);
+  int32_t _GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
+  int32_t _GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice);
+  int32_t _GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice);
+
+  int32_t InitRecordingDMO();
+
+  ScopedCOMInitializer _comInit;
+  AudioDeviceBuffer* _ptrAudioBuffer;
+  mutable Mutex mutex_;
+  mutable Mutex volume_mutex_ RTC_ACQUIRED_AFTER(mutex_);
+
+  IMMDeviceEnumerator* _ptrEnumerator;
+  IMMDeviceCollection* _ptrRenderCollection;
+  IMMDeviceCollection* _ptrCaptureCollection;
+  IMMDevice* _ptrDeviceOut;
+  IMMDevice* _ptrDeviceIn;
+
+  IAudioClient* _ptrClientOut;
+  IAudioClient* _ptrClientIn;
+  IAudioRenderClient* _ptrRenderClient;
+  IAudioCaptureClient* _ptrCaptureClient;
+  IAudioEndpointVolume* _ptrCaptureVolume;
+  ISimpleAudioVolume* _ptrRenderSimpleVolume;
+
+  // DirectX Media Object (DMO) for the built-in AEC.
+  rtc::scoped_refptr<IMediaObject> _dmo;
+  rtc::scoped_refptr<IMediaBuffer> _mediaBuffer;
+  bool _builtInAecEnabled;
+
+  HANDLE _hRenderSamplesReadyEvent;
+  HANDLE _hPlayThread;
+  HANDLE _hRenderStartedEvent;
+  HANDLE _hShutdownRenderEvent;
+
+  HANDLE _hCaptureSamplesReadyEvent;
+  HANDLE _hRecThread;
+  HANDLE _hCaptureStartedEvent;
+  HANDLE _hShutdownCaptureEvent;
+
+  HANDLE _hMmTask;
+
+  UINT _playAudioFrameSize;
+  uint32_t _playSampleRate;
+  uint32_t _devicePlaySampleRate;
+  uint32_t _playBlockSize;
+  uint32_t _devicePlayBlockSize;
+  uint32_t _playChannels;
+  uint32_t _sndCardPlayDelay;
+  UINT64 _writtenSamples;
+  UINT64 _readSamples;
+
+  UINT _recAudioFrameSize;
+  uint32_t _recSampleRate;
+  uint32_t _recBlockSize;
+  uint32_t _recChannels;
+
+  uint16_t _recChannelsPrioList[3];
+  uint16_t _playChannelsPrioList[2];
+
+  LARGE_INTEGER _perfCounterFreq;
+  double _perfCounterFactor;
+
+ private:
+  bool _initialized;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+  bool _speakerIsInitialized;
+  bool _microphoneIsInitialized;
+
+  bool _usingInputDeviceIndex;
+  bool _usingOutputDeviceIndex;
+  AudioDeviceModule::WindowsDeviceType _inputDevice;
+  AudioDeviceModule::WindowsDeviceType _outputDevice;
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+};
+
+#endif  // #if (_MSC_VER >= 1400)
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_WIN_AUDIO_DEVICE_CORE_WIN_H_
diff --git a/third_party/libwebrtc/modules/audio_device/win/audio_device_module_win.cc b/third_party/libwebrtc/modules/audio_device/win/audio_device_module_win.cc
new file mode 100644
index 0000000000..a36c40735e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/audio_device_module_win.cc
@@ -0,0 +1,522 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/win/audio_device_module_win.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/make_ref_counted.h"
+#include "api/sequence_checker.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+
+namespace webrtc {
+namespace webrtc_win {
+namespace {
+
+#define RETURN_IF_OUTPUT_RESTARTS(...) \
+  do {                                 \
+    if (output_->Restarting()) {       \
+      return __VA_ARGS__;              \
+    }                                  \
+  } while (0)
+
+#define RETURN_IF_INPUT_RESTARTS(...) \
+  do {                                \
+    if (input_->Restarting()) {       \
+      return __VA_ARGS__;             \
+    }                                 \
+  } while (0)
+
+#define RETURN_IF_OUTPUT_IS_INITIALIZED(...) \
+  do {                                       \
+    if (output_->PlayoutIsInitialized()) {   \
+      return __VA_ARGS__;                    \
+    }                                        \
+  } while (0)
+
+#define RETURN_IF_INPUT_IS_INITIALIZED(...) \
+  do {                                      \
+    if (input_->RecordingIsInitialized()) { \
+      return __VA_ARGS__;                   \
+    }                                       \
+  } while (0)
+
+#define RETURN_IF_OUTPUT_IS_ACTIVE(...) \
+  do {                                  \
+    if (output_->Playing()) {           \
+      return __VA_ARGS__;               \
+    }                                   \
+  } while (0)
+
+#define RETURN_IF_INPUT_IS_ACTIVE(...) \
+  do {                                 \
+    if (input_->Recording()) {         \
+      return __VA_ARGS__;              \
+    }                                  \
+  } while (0)
+
+// This class combines a generic instance of an AudioInput and a generic
+// instance of an AudioOutput to create an AudioDeviceModule. This is mostly
+// done by delegating to the audio input/output with some glue code. This class
+// also directly implements some of the AudioDeviceModule methods with dummy
+// implementations.
+//
+// An instance must be created, destroyed and used on one and the same thread,
+// i.e., all public methods must also be called on the same thread. A thread
+// checker will RTC_DCHECK if any method is called on an invalid thread.
+// TODO(henrika): is thread checking needed in AudioInput and AudioOutput?
+class WindowsAudioDeviceModule : public AudioDeviceModuleForTest {
+ public:
+  enum class InitStatus {
+    OK = 0,
+    PLAYOUT_ERROR = 1,
+    RECORDING_ERROR = 2,
+    OTHER_ERROR = 3,
+    NUM_STATUSES = 4
+  };
+
+  WindowsAudioDeviceModule(std::unique_ptr<AudioInput> audio_input,
+                           std::unique_ptr<AudioOutput> audio_output,
+                           TaskQueueFactory* task_queue_factory)
+      : input_(std::move(audio_input)),
+        output_(std::move(audio_output)),
+        task_queue_factory_(task_queue_factory) {
+    RTC_CHECK(input_);
+    RTC_CHECK(output_);
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+  }
+
+  ~WindowsAudioDeviceModule() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    Terminate();
+  }
+
+  WindowsAudioDeviceModule(const WindowsAudioDeviceModule&) = delete;
+  WindowsAudioDeviceModule& operator=(const WindowsAudioDeviceModule&) = delete;
+
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer* audioLayer) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    // TODO(henrika): it might be possible to remove this unique signature.
+    *audioLayer = AudioDeviceModule::kWindowsCoreAudio2;
+    return 0;
+  }
+
+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK(audio_device_buffer_);
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return audio_device_buffer_->RegisterAudioCallback(audioCallback);
+  }
+
+  int32_t Init() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(0);
+    RETURN_IF_INPUT_RESTARTS(0);
+    if (initialized_) {
+      return 0;
+    }
+    audio_device_buffer_ =
+        std::make_unique<AudioDeviceBuffer>(task_queue_factory_);
+    AttachAudioBuffer();
+    InitStatus status;
+    if (output_->Init() != 0) {
+      status = InitStatus::PLAYOUT_ERROR;
+    } else if (input_->Init() != 0) {
+      output_->Terminate();
+      status = InitStatus::RECORDING_ERROR;
+    } else {
+      initialized_ = true;
+      status = InitStatus::OK;
+    }
+    if (status != InitStatus::OK) {
+      RTC_LOG(LS_ERROR) << "Audio device initialization failed";
+      return -1;
+    }
+    return 0;
+  }
+
+  int32_t Terminate() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(0);
+    RETURN_IF_INPUT_RESTARTS(0);
+    if (!initialized_)
+      return 0;
+    int32_t err = input_->Terminate();
+    err |= output_->Terminate();
+    initialized_ = false;
+    RTC_DCHECK_EQ(err, 0);
+    return err;
+  }
+
+  bool Initialized() const override {
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return initialized_;
+  }
+
+  int16_t PlayoutDevices() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(0);
+    return output_->NumDevices();
+  }
+
+  int16_t RecordingDevices() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_INPUT_RESTARTS(0);
+    return input_->NumDevices();
+  }
+
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(0);
+    std::string name_str, guid_str;
+    int ret = -1;
+    if (guid != nullptr) {
+      ret = output_->DeviceName(index, &name_str, &guid_str);
+      rtc::strcpyn(guid, kAdmMaxGuidSize, guid_str.c_str());
+    } else {
+      ret = output_->DeviceName(index, &name_str, nullptr);
+    }
+    rtc::strcpyn(name, kAdmMaxDeviceNameSize, name_str.c_str());
+    return ret;
+  }
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_INPUT_RESTARTS(0);
+    std::string name_str, guid_str;
+    int ret = -1;
+    if (guid != nullptr) {
+      ret = input_->DeviceName(index, &name_str, &guid_str);
+      rtc::strcpyn(guid, kAdmMaxGuidSize, guid_str.c_str());
+    } else {
+      ret = input_->DeviceName(index, &name_str, nullptr);
+    }
+    rtc::strcpyn(name, kAdmMaxDeviceNameSize, name_str.c_str());
+    return ret;
+  }
+
+  int32_t SetPlayoutDevice(uint16_t index) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(0);
+    return output_->SetDevice(index);
+  }
+
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(0);
+    return output_->SetDevice(device);
+  }
+  int32_t SetRecordingDevice(uint16_t index) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return input_->SetDevice(index);
+  }
+
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return input_->SetDevice(device);
+  }
+
+  int32_t PlayoutIsAvailable(bool* available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    *available = true;
+    return 0;
+  }
+
+  int32_t InitPlayout() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(0);
+    RETURN_IF_OUTPUT_IS_INITIALIZED(0);
+    return output_->InitPlayout();
+  }
+
+  bool PlayoutIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(true);
+    return output_->PlayoutIsInitialized();
+  }
+
+  int32_t RecordingIsAvailable(bool* available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    *available = true;
+    return 0;
+  }
+
+  int32_t InitRecording() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_INPUT_RESTARTS(0);
+    RETURN_IF_INPUT_IS_INITIALIZED(0);
+    return input_->InitRecording();
+  }
+
+  bool RecordingIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_INPUT_RESTARTS(true);
+    return input_->RecordingIsInitialized();
+  }
+
+  int32_t StartPlayout() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(0);
+    RETURN_IF_OUTPUT_IS_ACTIVE(0);
+    return output_->StartPlayout();
+  }
+
+  int32_t StopPlayout() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(-1);
+    return output_->StopPlayout();
+  }
+
+  bool Playing() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(true);
+    return output_->Playing();
+  }
+
+  int32_t StartRecording() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_INPUT_RESTARTS(0);
+    RETURN_IF_INPUT_IS_ACTIVE(0);
+    return input_->StartRecording();
+  }
+
+  int32_t StopRecording() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_INPUT_RESTARTS(-1);
+    return input_->StopRecording();
+  }
+
+  bool Recording() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RETURN_IF_INPUT_RESTARTS(true);
+    return input_->Recording();
+  }
+
+  int32_t InitSpeaker() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RTC_DLOG(LS_WARNING) << "This method has no effect";
+    return initialized_ ? 0 : -1;
+  }
+
+  bool SpeakerIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RTC_DLOG(LS_WARNING) << "This method has no effect";
+    return initialized_;
+  }
+
+  int32_t InitMicrophone() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RTC_DLOG(LS_WARNING) << "This method has no effect";
+    return initialized_ ? 0 : -1;
+  }
+
+  bool MicrophoneIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RTC_DLOG(LS_WARNING) << "This method has no effect";
+    return initialized_;
+  }
+
+  int32_t SpeakerVolumeIsAvailable(bool* available) override {
+    // TODO(henrika): improve support.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    *available = false;
+    return 0;
+  }
+
+  int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
+  int32_t SpeakerVolume(uint32_t* volume) const override { return 0; }
+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override { return 0; }
+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override { return 0; }
+
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override {
+    // TODO(henrika): improve support.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    *available = false;
+    return 0;
+  }
+
+  int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
+  int32_t MicrophoneVolume(uint32_t* volume) const override { return 0; }
+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override { return 0; }
+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override { return 0; }
+
+  int32_t SpeakerMuteIsAvailable(bool* available) override { return 0; }
+  int32_t SetSpeakerMute(bool enable) override { return 0; }
+  int32_t SpeakerMute(bool* enabled) const override { return 0; }
+
+  int32_t MicrophoneMuteIsAvailable(bool* available) override { return 0; }
+  int32_t SetMicrophoneMute(bool enable) override { return 0; }
+  int32_t MicrophoneMute(bool* enabled) const override { return 0; }
+
+  int32_t StereoPlayoutIsAvailable(bool* available) const override {
+    // TODO(henrika): improve support.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    *available = true;
+    return 0;
+  }
+
+  int32_t SetStereoPlayout(bool enable) override {
+    // TODO(henrika): improve support.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return 0;
+  }
+
+  int32_t StereoPlayout(bool* enabled) const override {
+    // TODO(henrika): improve support.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    *enabled = true;
+    return 0;
+  }
+
+  int32_t StereoRecordingIsAvailable(bool* available) const override {
+    // TODO(henrika): improve support.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    *available = true;
+    return 0;
+  }
+
+  int32_t SetStereoRecording(bool enable) override {
+    // TODO(henrika): improve support.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return 0;
+  }
+
+  int32_t StereoRecording(bool* enabled) const override {
+    // TODO(henrika): improve support.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    *enabled = true;
+    return 0;
+  }
+
+  int32_t PlayoutDelay(uint16_t* delayMS) const override { return 0; }
+
+  bool BuiltInAECIsAvailable() const override { return false; }
+  bool BuiltInAGCIsAvailable() const override { return false; }
+  bool BuiltInNSIsAvailable() const override { return false; }
+
+  int32_t EnableBuiltInAEC(bool enable) override { return 0; }
+  int32_t EnableBuiltInAGC(bool enable) override { return 0; }
+  int32_t EnableBuiltInNS(bool enable) override { return 0; }
+
+  int32_t AttachAudioBuffer() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    output_->AttachAudioBuffer(audio_device_buffer_.get());
+    input_->AttachAudioBuffer(audio_device_buffer_.get());
+    return 0;
+  }
+
+  int RestartPlayoutInternally() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    RETURN_IF_OUTPUT_RESTARTS(0);
+    return output_->RestartPlayout();
+  }
+
+  int RestartRecordingInternally() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return input_->RestartRecording();
+  }
+
+  int SetPlayoutSampleRate(uint32_t sample_rate) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return output_->SetSampleRate(sample_rate);
+  }
+
+  int SetRecordingSampleRate(uint32_t sample_rate) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK_RUN_ON(&thread_checker_);
+    return input_->SetSampleRate(sample_rate);
+  }
+
+ private:
+  // Ensures that the class is used on the same thread as it is constructed
+  // and destroyed on.
+  SequenceChecker thread_checker_;
+
+  // Implements the AudioInput interface and deals with audio capturing parts.
+  const std::unique_ptr<AudioInput> input_;
+
+  // Implements the AudioOutput interface and deals with audio rendering parts.
+  const std::unique_ptr<AudioOutput> output_;
+
+  TaskQueueFactory* const task_queue_factory_;
+
+  // The AudioDeviceBuffer (ADB) instance is needed for sending/receiving audio
+  // to/from the WebRTC layer. Created and owned by this object. Used by
+  // both `input_` and `output_` but they use orthogonal parts of the ADB.
+  std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
+
+  // Set to true after a successful call to Init(). Cleared by Terminate().
+  bool initialized_ = false;
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioDeviceModuleForTest>
+CreateWindowsCoreAudioAudioDeviceModuleFromInputAndOutput(
+    std::unique_ptr<AudioInput> audio_input,
+    std::unique_ptr<AudioOutput> audio_output,
+    TaskQueueFactory* task_queue_factory) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  return rtc::make_ref_counted<WindowsAudioDeviceModule>(
+      std::move(audio_input), std::move(audio_output), task_queue_factory);
+}
+
+}  // namespace webrtc_win
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/win/audio_device_module_win.h b/third_party/libwebrtc/modules/audio_device/win/audio_device_module_win.h
new file mode 100644
index 0000000000..1ed0b25620
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/audio_device_module_win.h
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_WIN_AUDIO_DEVICE_MODULE_WIN_H_
+#define MODULES_AUDIO_DEVICE_WIN_AUDIO_DEVICE_MODULE_WIN_H_
+
+#include <memory>
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+
+namespace webrtc_win {
+
+// This interface represents the main input-related parts of the complete
+// AudioDeviceModule interface.
+class AudioInput {
+ public:
+  virtual ~AudioInput() {}
+
+  virtual int Init() = 0;
+  virtual int Terminate() = 0;
+  virtual int NumDevices() const = 0;
+  virtual int SetDevice(int index) = 0;
+  virtual int SetDevice(AudioDeviceModule::WindowsDeviceType device) = 0;
+  virtual int DeviceName(int index, std::string* name, std::string* guid) = 0;
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) = 0;
+  virtual bool RecordingIsInitialized() const = 0;
+  virtual int InitRecording() = 0;
+  virtual int StartRecording() = 0;
+  virtual int StopRecording() = 0;
+  virtual bool Recording() = 0;
+  virtual int VolumeIsAvailable(bool* available) = 0;
+  virtual int RestartRecording() = 0;
+  virtual bool Restarting() const = 0;
+  virtual int SetSampleRate(uint32_t sample_rate) = 0;
+};
+
+// This interface represents the main output-related parts of the complete
+// AudioDeviceModule interface.
+class AudioOutput {
+ public:
+  virtual ~AudioOutput() {}
+
+  virtual int Init() = 0;
+  virtual int Terminate() = 0;
+  virtual int NumDevices() const = 0;
+  virtual int SetDevice(int index) = 0;
+  virtual int SetDevice(AudioDeviceModule::WindowsDeviceType device) = 0;
+  virtual int DeviceName(int index, std::string* name, std::string* guid) = 0;
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) = 0;
+  virtual bool PlayoutIsInitialized() const = 0;
+  virtual int InitPlayout() = 0;
+  virtual int StartPlayout() = 0;
+  virtual int StopPlayout() = 0;
+  virtual bool Playing() = 0;
+  virtual int VolumeIsAvailable(bool* available) = 0;
+  virtual int RestartPlayout() = 0;
+  virtual bool Restarting() const = 0;
+  virtual int SetSampleRate(uint32_t sample_rate) = 0;
+};
+
+// Combines an AudioInput and an AudioOutput implementation to build an
+// AudioDeviceModule. Hides most parts of the full ADM interface.
+rtc::scoped_refptr<AudioDeviceModuleForTest>
+CreateWindowsCoreAudioAudioDeviceModuleFromInputAndOutput(
+    std::unique_ptr<AudioInput> audio_input,
+    std::unique_ptr<AudioOutput> audio_output,
+    TaskQueueFactory* task_queue_factory);
+
+}  // namespace webrtc_win
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_WIN_AUDIO_DEVICE_MODULE_WIN_H_
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_base_win.cc b/third_party/libwebrtc/modules/audio_device/win/core_audio_base_win.cc
new file mode 100644
index 0000000000..dc8526b625
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_base_win.cc
@@ -0,0 +1,948 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/win/core_audio_base_win.h"
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/win/scoped_com_initializer.h"
+#include "rtc_base/win/windows_version.h"
+
+using Microsoft::WRL::ComPtr;
+
+namespace webrtc {
+namespace webrtc_win {
+namespace {
+
+// Even if the device supports low latency and even if IAudioClient3 can be
+// used (requires Win10 or higher), we currently disable any attempts to
+// initialize the client for low-latency.
+// TODO(henrika): more research is needed before we can enable low-latency.
+const bool kEnableLowLatencyIfSupported = false;
+
+// Each unit of reference time is 100 nanoseconds, hence `kReftimesPerSec`
+// corresponds to one second.
+// TODO(henrika): possibly add usage in Init().
+// const REFERENCE_TIME kReferenceTimesPerSecond = 10000000;
+
+enum DefaultDeviceType {
+  kUndefined = -1,
+  kDefault = 0,
+  kDefaultCommunications = 1,
+  kDefaultDeviceTypeMaxCount = kDefaultCommunications + 1,
+};
+
+const char* DirectionToString(CoreAudioBase::Direction direction) {
+  switch (direction) {
+    case CoreAudioBase::Direction::kOutput:
+      return "Output";
+    case CoreAudioBase::Direction::kInput:
+      return "Input";
+    default:
+      return "Unkown";
+  }
+}
+
+const char* RoleToString(const ERole role) {
+  switch (role) {
+    case eConsole:
+      return "Console";
+    case eMultimedia:
+      return "Multimedia";
+    case eCommunications:
+      return "Communications";
+    default:
+      return "Unsupported";
+  }
+}
+
+std::string IndexToString(int index) {
+  std::string ss = std::to_string(index);
+  switch (index) {
+    case kDefault:
+      ss += " (Default)";
+      break;
+    case kDefaultCommunications:
+      ss += " (Communications)";
+      break;
+    default:
+      break;
+  }
+  return ss;
+}
+
+const char* SessionStateToString(AudioSessionState state) {
+  switch (state) {
+    case AudioSessionStateActive:
+      return "Active";
+    case AudioSessionStateInactive:
+      return "Inactive";
+    case AudioSessionStateExpired:
+      return "Expired";
+    default:
+      return "Invalid";
+  }
+}
+
+const char* SessionDisconnectReasonToString(
+    AudioSessionDisconnectReason reason) {
+  switch (reason) {
+    case DisconnectReasonDeviceRemoval:
+      return "DeviceRemoval";
+    case DisconnectReasonServerShutdown:
+      return "ServerShutdown";
+    case DisconnectReasonFormatChanged:
+      return "FormatChanged";
+    case DisconnectReasonSessionLogoff:
+      return "SessionLogoff";
+    case DisconnectReasonSessionDisconnected:
+      return "Disconnected";
+    case DisconnectReasonExclusiveModeOverride:
+      return "ExclusiveModeOverride";
+    default:
+      return "Invalid";
+  }
+}
+
+// Returns true if the selected audio device supports low latency, i.e, if it
+// is possible to initialize the engine using periods less than the default
+// period (10ms).
+bool IsLowLatencySupported(IAudioClient3* client3,
+                           const WAVEFORMATEXTENSIBLE* format,
+                           uint32_t* min_period_in_frames) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  // Get the range of periodicities supported by the engine for the specified
+  // stream format.
+  uint32_t default_period = 0;
+  uint32_t fundamental_period = 0;
+  uint32_t min_period = 0;
+  uint32_t max_period = 0;
+  if (FAILED(core_audio_utility::GetSharedModeEnginePeriod(
+          client3, format, &default_period, &fundamental_period, &min_period,
+          &max_period))) {
+    return false;
+  }
+
+  // Low latency is supported if the shortest allowed period is less than the
+  // default engine period.
+  // TODO(henrika): verify that this assumption is correct.
+  const bool low_latency = min_period < default_period;
+  RTC_LOG(LS_INFO) << "low_latency: " << low_latency;
+  *min_period_in_frames = low_latency ? min_period : 0;
+  return low_latency;
+}
+
+}  // namespace
+
+CoreAudioBase::CoreAudioBase(Direction direction,
+                             bool automatic_restart,
+                             OnDataCallback data_callback,
+                             OnErrorCallback error_callback)
+    : format_(),
+      direction_(direction),
+      automatic_restart_(automatic_restart),
+      on_data_callback_(data_callback),
+      on_error_callback_(error_callback),
+      device_index_(kUndefined),
+      is_restarting_(false) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction)
+                    << "]";
+  RTC_DLOG(LS_INFO) << "Automatic restart: " << automatic_restart;
+  RTC_DLOG(LS_INFO) << "Windows version: " << rtc::rtc_win::GetVersion();
+
+  // Create the event which the audio engine will signal each time a buffer
+  // becomes ready to be processed by the client.
+  audio_samples_event_.Set(CreateEvent(nullptr, false, false, nullptr));
+  RTC_DCHECK(audio_samples_event_.IsValid());
+
+  // Event to be set in Stop() when rendering/capturing shall stop.
+  stop_event_.Set(CreateEvent(nullptr, false, false, nullptr));
+  RTC_DCHECK(stop_event_.IsValid());
+
+  // Event to be set when it has been detected that an active device has been
+  // invalidated or the stream format has changed.
+  restart_event_.Set(CreateEvent(nullptr, false, false, nullptr));
+  RTC_DCHECK(restart_event_.IsValid());
+}
+
+CoreAudioBase::~CoreAudioBase() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_EQ(ref_count_, 1);
+}
+
+EDataFlow CoreAudioBase::GetDataFlow() const {
+  return direction_ == CoreAudioBase::Direction::kOutput ? eRender : eCapture;
+}
+
+bool CoreAudioBase::IsRestarting() const {
+  return is_restarting_;
+}
+
+int64_t CoreAudioBase::TimeSinceStart() const {
+  return rtc::TimeSince(start_time_);
+}
+
+int CoreAudioBase::NumberOfActiveDevices() const {
+  return core_audio_utility::NumberOfActiveDevices(GetDataFlow());
+}
+
+int CoreAudioBase::NumberOfEnumeratedDevices() const {
+  const int num_active = NumberOfActiveDevices();
+  return num_active > 0 ? num_active + kDefaultDeviceTypeMaxCount : 0;
+}
+
+void CoreAudioBase::ReleaseCOMObjects() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  // ComPtr::Reset() sets the ComPtr to nullptr releasing any previous
+  // reference.
+  if (audio_client_) {
+    audio_client_.Reset();
+  }
+  if (audio_clock_.Get()) {
+    audio_clock_.Reset();
+  }
+  if (audio_session_control_.Get()) {
+    audio_session_control_.Reset();
+  }
+}
+
+bool CoreAudioBase::IsDefaultDevice(int index) const {
+  return index == kDefault;
+}
+
+bool CoreAudioBase::IsDefaultCommunicationsDevice(int index) const {
+  return index == kDefaultCommunications;
+}
+
+bool CoreAudioBase::IsDefaultDeviceId(absl::string_view device_id) const {
+  // Returns true if `device_id` corresponds to the id of the default
+  // device. Note that, if only one device is available (or if the user has not
+  // explicitly set a default device), `device_id` will also math
+  // IsDefaultCommunicationsDeviceId().
+  return (IsInput() &&
+          (device_id == core_audio_utility::GetDefaultInputDeviceID())) ||
+         (IsOutput() &&
+          (device_id == core_audio_utility::GetDefaultOutputDeviceID()));
+}
+
+bool CoreAudioBase::IsDefaultCommunicationsDeviceId(
+    absl::string_view device_id) const {
+  // Returns true if `device_id` corresponds to the id of the default
+  // communication device. Note that, if only one device is available (or if
+  // the user has not explicitly set a communication device), `device_id` will
+  // also math IsDefaultDeviceId().
+  return (IsInput() &&
+          (device_id ==
+           core_audio_utility::GetCommunicationsInputDeviceID())) ||
+         (IsOutput() &&
+          (device_id == core_audio_utility::GetCommunicationsOutputDeviceID()));
+}
+
+bool CoreAudioBase::IsInput() const {
+  return direction_ == CoreAudioBase::Direction::kInput;
+}
+
+bool CoreAudioBase::IsOutput() const {
+  return direction_ == CoreAudioBase::Direction::kOutput;
+}
+
+std::string CoreAudioBase::GetDeviceID(int index) const {
+  if (index >= NumberOfEnumeratedDevices()) {
+    RTC_LOG(LS_ERROR) << "Invalid device index";
+    return std::string();
+  }
+
+  std::string device_id;
+  if (IsDefaultDevice(index)) {
+    device_id = IsInput() ? core_audio_utility::GetDefaultInputDeviceID()
+                          : core_audio_utility::GetDefaultOutputDeviceID();
+  } else if (IsDefaultCommunicationsDevice(index)) {
+    device_id = IsInput()
+                    ? core_audio_utility::GetCommunicationsInputDeviceID()
+                    : core_audio_utility::GetCommunicationsOutputDeviceID();
+  } else {
+    AudioDeviceNames device_names;
+    bool ok = IsInput()
+                  ? core_audio_utility::GetInputDeviceNames(&device_names)
+                  : core_audio_utility::GetOutputDeviceNames(&device_names);
+    if (ok) {
+      device_id = device_names[index].unique_id;
+    }
+  }
+  return device_id;
+}
+
+int CoreAudioBase::SetDevice(int index) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+                    << "]: index=" << IndexToString(index);
+  if (initialized_) {
+    return -1;
+  }
+
+  std::string device_id = GetDeviceID(index);
+  RTC_DLOG(LS_INFO) << "index=" << IndexToString(index)
+                    << " => device_id: " << device_id;
+  device_index_ = index;
+  device_id_ = device_id;
+
+  return device_id_.empty() ? -1 : 0;
+}
+
+int CoreAudioBase::DeviceName(int index,
+                              std::string* name,
+                              std::string* guid) const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+                    << "]: index=" << IndexToString(index);
+  if (index > NumberOfEnumeratedDevices() - 1) {
+    RTC_LOG(LS_ERROR) << "Invalid device index";
+    return -1;
+  }
+
+  AudioDeviceNames device_names;
+  bool ok = IsInput() ? core_audio_utility::GetInputDeviceNames(&device_names)
+                      : core_audio_utility::GetOutputDeviceNames(&device_names);
+  // Validate the index one extra time in-case the size of the generated list
+  // did not match NumberOfEnumeratedDevices().
+  if (!ok || static_cast<int>(device_names.size()) <= index) {
+    RTC_LOG(LS_ERROR) << "Failed to get the device name";
+    return -1;
+  }
+
+  *name = device_names[index].device_name;
+  RTC_DLOG(LS_INFO) << "name: " << *name;
+  if (guid != nullptr) {
+    *guid = device_names[index].unique_id;
+    RTC_DLOG(LS_INFO) << "guid: " << *guid;
+  }
+  return 0;
+}
+
+bool CoreAudioBase::Init() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+                    << "]";
+  RTC_DCHECK_GE(device_index_, 0);
+  RTC_DCHECK(!device_id_.empty());
+  RTC_DCHECK(audio_device_buffer_);
+  RTC_DCHECK(!audio_client_);
+  RTC_DCHECK(!audio_session_control_.Get());
+
+  // Use an existing combination of `device_index_` and `device_id_` to set
+  // parameters which are required to create an audio client. It is up to the
+  // parent class to set `device_index_` and `device_id_`.
+  std::string device_id = AudioDeviceName::kDefaultDeviceId;
+  ERole role = ERole();
+  if (IsDefaultDevice(device_index_)) {
+    role = eConsole;
+  } else if (IsDefaultCommunicationsDevice(device_index_)) {
+    role = eCommunications;
+  } else {
+    device_id = device_id_;
+  }
+  RTC_LOG(LS_INFO) << "Unique device identifier: device_id=" << device_id
+                   << ", role=" << RoleToString(role);
+
+  // Create an IAudioClient interface which enables us to create and initialize
+  // an audio stream between an audio application and the audio engine.
+  ComPtr<IAudioClient> audio_client;
+  if (core_audio_utility::GetAudioClientVersion() == 3) {
+    RTC_DLOG(LS_INFO) << "Using IAudioClient3";
+    audio_client =
+        core_audio_utility::CreateClient3(device_id, GetDataFlow(), role);
+  } else if (core_audio_utility::GetAudioClientVersion() == 2) {
+    RTC_DLOG(LS_INFO) << "Using IAudioClient2";
+    audio_client =
+        core_audio_utility::CreateClient2(device_id, GetDataFlow(), role);
+  } else {
+    RTC_DLOG(LS_INFO) << "Using IAudioClient";
+    audio_client =
+        core_audio_utility::CreateClient(device_id, GetDataFlow(), role);
+  }
+  if (!audio_client) {
+    return false;
+  }
+
+  // Set extra client properties before initialization if the audio client
+  // supports it.
+  // TODO(henrika): evaluate effect(s) of making these changes. Also, perhaps
+  // these types of settings belongs to the client and not the utility parts.
+  if (core_audio_utility::GetAudioClientVersion() >= 2) {
+    if (FAILED(core_audio_utility::SetClientProperties(
+            static_cast<IAudioClient2*>(audio_client.Get())))) {
+      return false;
+    }
+  }
+
+  // Retrieve preferred audio input or output parameters for the given client
+  // and the specified client properties. Override the preferred rate if sample
+  // rate has been defined by the user. Rate conversion will be performed by
+  // the audio engine to match the client if needed.
+  AudioParameters params;
+  HRESULT res = sample_rate_ ? core_audio_utility::GetPreferredAudioParameters(
+                                   audio_client.Get(), &params, *sample_rate_)
+                             : core_audio_utility::GetPreferredAudioParameters(
+                                   audio_client.Get(), &params);
+  if (FAILED(res)) {
+    return false;
+  }
+
+  // Define the output WAVEFORMATEXTENSIBLE format in `format_`.
+  WAVEFORMATEX* format = &format_.Format;
+  format->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+  // Check the preferred channel configuration and request implicit channel
+  // upmixing (audio engine extends from 2 to N channels internally) if the
+  // preferred number of channels is larger than two; i.e., initialize the
+  // stream in stereo even if the preferred configuration is multi-channel.
+  if (params.channels() <= 2) {
+    format->nChannels = rtc::dchecked_cast<WORD>(params.channels());
+  } else {
+    // TODO(henrika): ensure that this approach works on different multi-channel
+    // devices. Verified on:
+    // - Corsair VOID PRO Surround USB Adapter (supports 7.1)
+    RTC_LOG(LS_WARNING)
+        << "Using channel upmixing in WASAPI audio engine (2 => "
+        << params.channels() << ")";
+    format->nChannels = 2;
+  }
+  format->nSamplesPerSec = params.sample_rate();
+  format->wBitsPerSample = rtc::dchecked_cast<WORD>(params.bits_per_sample());
+  format->nBlockAlign = (format->wBitsPerSample / 8) * format->nChannels;
+  format->nAvgBytesPerSec = format->nSamplesPerSec * format->nBlockAlign;
+  format->cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
+  // Add the parts which are unique for the WAVE_FORMAT_EXTENSIBLE structure.
+  format_.Samples.wValidBitsPerSample =
+      rtc::dchecked_cast<WORD>(params.bits_per_sample());
+  format_.dwChannelMask =
+      format->nChannels == 1 ? KSAUDIO_SPEAKER_MONO : KSAUDIO_SPEAKER_STEREO;
+  format_.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+  RTC_DLOG(LS_INFO) << core_audio_utility::WaveFormatToString(&format_);
+
+  // Verify that the format is supported but exclude the test if the default
+  // sample rate has been overridden. If so, the WASAPI audio engine will do
+  // any necessary conversions between the client format we have given it and
+  // the playback mix format or recording split format.
+  if (!sample_rate_) {
+    if (!core_audio_utility::IsFormatSupported(
+            audio_client.Get(), AUDCLNT_SHAREMODE_SHARED, &format_)) {
+      return false;
+    }
+  }
+
+  // Check if low-latency is supported and use special initialization if it is.
+  // Low-latency initialization requires these things:
+  // - IAudioClient3 (>= Win10)
+  // - HDAudio driver
+  // - kEnableLowLatencyIfSupported changed from false (default) to true.
+  // TODO(henrika): IsLowLatencySupported() returns AUDCLNT_E_UNSUPPORTED_FORMAT
+  // when `sample_rate_.has_value()` returns true if rate conversion is
+  // actually required (i.e., client asks for other than the default rate).
+  bool low_latency_support = false;
+  uint32_t min_period_in_frames = 0;
+  if (kEnableLowLatencyIfSupported &&
+      core_audio_utility::GetAudioClientVersion() >= 3) {
+    low_latency_support =
+        IsLowLatencySupported(static_cast<IAudioClient3*>(audio_client.Get()),
+                              &format_, &min_period_in_frames);
+  }
+
+  if (low_latency_support) {
+    RTC_DCHECK_GE(core_audio_utility::GetAudioClientVersion(), 3);
+    // Use IAudioClient3::InitializeSharedAudioStream() API to initialize a
+    // low-latency event-driven client. Request the smallest possible
+    // periodicity.
+    // TODO(henrika): evaluate this scheme in terms of CPU etc.
+    if (FAILED(core_audio_utility::SharedModeInitializeLowLatency(
+            static_cast<IAudioClient3*>(audio_client.Get()), &format_,
+            audio_samples_event_, min_period_in_frames,
+            sample_rate_.has_value(), &endpoint_buffer_size_frames_))) {
+      return false;
+    }
+  } else {
+    // Initialize the audio stream between the client and the device in shared
+    // mode using event-driven buffer handling. Also, using 0 as requested
+    // buffer size results in a default (minimum) endpoint buffer size.
+    // TODO(henrika): possibly increase `requested_buffer_size` to add
+    // robustness.
+    const REFERENCE_TIME requested_buffer_size = 0;
+    if (FAILED(core_audio_utility::SharedModeInitialize(
+            audio_client.Get(), &format_, audio_samples_event_,
+            requested_buffer_size, sample_rate_.has_value(),
+            &endpoint_buffer_size_frames_))) {
+      return false;
+    }
+  }
+
+  // Check device period and the preferred buffer size and log a warning if
+  // WebRTC's buffer size is not an even divisor of the preferred buffer size
+  // in Core Audio.
+  // TODO(henrika): sort out if a non-perfect match really is an issue.
+  // TODO(henrika): compare with IAudioClient3::GetSharedModeEnginePeriod().
+  REFERENCE_TIME device_period;
+  if (FAILED(core_audio_utility::GetDevicePeriod(
+          audio_client.Get(), AUDCLNT_SHAREMODE_SHARED, &device_period))) {
+    return false;
+  }
+  const double device_period_in_seconds =
+      static_cast<double>(
+          core_audio_utility::ReferenceTimeToTimeDelta(device_period).ms()) /
+      1000.0L;
+  const int preferred_frames_per_buffer =
+      static_cast<int>(params.sample_rate() * device_period_in_seconds + 0.5);
+  RTC_DLOG(LS_INFO) << "preferred_frames_per_buffer: "
+                    << preferred_frames_per_buffer;
+  if (preferred_frames_per_buffer % params.frames_per_buffer()) {
+    RTC_LOG(LS_WARNING) << "Buffer size of " << params.frames_per_buffer()
+                        << " is not an even divisor of "
+                        << preferred_frames_per_buffer;
+  }
+
+  // Create an AudioSessionControl interface given the initialized client.
+  // The IAudioControl interface enables a client to configure the control
+  // parameters for an audio session and to monitor events in the session.
+  ComPtr<IAudioSessionControl> audio_session_control =
+      core_audio_utility::CreateAudioSessionControl(audio_client.Get());
+  if (!audio_session_control.Get()) {
+    return false;
+  }
+
+  // The Sndvol program displays volume and mute controls for sessions that
+  // are in the active and inactive states.
+  AudioSessionState state;
+  if (FAILED(audio_session_control->GetState(&state))) {
+    return false;
+  }
+  RTC_DLOG(LS_INFO) << "audio session state: " << SessionStateToString(state);
+  RTC_DCHECK_EQ(state, AudioSessionStateInactive);
+
+  // Register the client to receive notifications of session events, including
+  // changes in the stream state.
+  if (FAILED(audio_session_control->RegisterAudioSessionNotification(this))) {
+    return false;
+  }
+
+  // Store valid COM interfaces.
+  audio_client_ = audio_client;
+  audio_session_control_ = audio_session_control;
+
+  return true;
+}
+
+bool CoreAudioBase::Start() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+                    << "]";
+  if (IsRestarting()) {
+    // Audio thread should be alive during internal restart since the restart
+    // callback is triggered on that thread and it also makes the restart
+    // sequence less complex.
+    RTC_DCHECK(!audio_thread_.empty());
+  }
+
+  // Start an audio thread but only if one does not already exist (which is the
+  // case during restart).
+  if (audio_thread_.empty()) {
+    const absl::string_view name =
+        IsInput() ? "wasapi_capture_thread" : "wasapi_render_thread";
+    audio_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] { ThreadRun(); }, name,
+        rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+    RTC_DLOG(LS_INFO) << "Started thread with name: " << name
+                      << " and handle: " << *audio_thread_.GetHandle();
+  }
+
+  // Start streaming data between the endpoint buffer and the audio engine.
+  _com_error error = audio_client_->Start();
+  if (FAILED(error.Error())) {
+    StopThread();
+    RTC_LOG(LS_ERROR) << "IAudioClient::Start failed: "
+                      << core_audio_utility::ErrorToString(error);
+    return false;
+  }
+
+  start_time_ = rtc::TimeMillis();
+  num_data_callbacks_ = 0;
+
+  return true;
+}
+
+bool CoreAudioBase::Stop() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+                    << "]";
+  RTC_DLOG(LS_INFO) << "total activity time: " << TimeSinceStart();
+
+  // Stop audio streaming.
+  _com_error error = audio_client_->Stop();
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::Stop failed: "
+                      << core_audio_utility::ErrorToString(error);
+  }
+  // Stop and destroy the audio thread but only when a restart attempt is not
+  // ongoing.
+  if (!IsRestarting()) {
+    StopThread();
+  }
+
+  // Flush all pending data and reset the audio clock stream position to 0.
+  error = audio_client_->Reset();
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::Reset failed: "
+                      << core_audio_utility::ErrorToString(error);
+  }
+
+  if (IsOutput()) {
+    // Extra safety check to ensure that the buffers are cleared.
+    // If the buffers are not cleared correctly, the next call to Start()
+    // would fail with AUDCLNT_E_BUFFER_ERROR at
+    // IAudioRenderClient::GetBuffer().
+    UINT32 num_queued_frames = 0;
+    audio_client_->GetCurrentPadding(&num_queued_frames);
+    RTC_DCHECK_EQ(0u, num_queued_frames);
+  }
+
+  // Delete the previous registration by the client to receive notifications
+  // about audio session events.
+  RTC_DLOG(LS_INFO) << "audio session state: "
+                    << SessionStateToString(GetAudioSessionState());
+  error = audio_session_control_->UnregisterAudioSessionNotification(this);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR)
+        << "IAudioSessionControl::UnregisterAudioSessionNotification failed: "
+        << core_audio_utility::ErrorToString(error);
+  }
+
+  // To ensure that the restart process is as simple as possible, the audio
+  // thread is not destroyed during restart attempts triggered by internal
+  // error callbacks.
+  if (!IsRestarting()) {
+    thread_checker_audio_.Detach();
+  }
+
+  // Release all allocated COM interfaces to allow for a restart without
+  // intermediate destruction.
+  ReleaseCOMObjects();
+
+  return true;
+}
+
+bool CoreAudioBase::IsVolumeControlAvailable(bool* available) const {
+  // A valid IAudioClient is required to access the ISimpleAudioVolume interface
+  // properly. It is possible to use IAudioSessionManager::GetSimpleAudioVolume
+  // as well but we use the audio client here to ensure that the initialized
+  // audio session is visible under group box labeled "Applications" in
+  // Sndvol.exe.
+  if (!audio_client_) {
+    return false;
+  }
+
+  // Try to create an ISimpleAudioVolume instance.
+  ComPtr<ISimpleAudioVolume> audio_volume =
+      core_audio_utility::CreateSimpleAudioVolume(audio_client_.Get());
+  if (!audio_volume.Get()) {
+    RTC_DLOG(LS_ERROR) << "Volume control is not supported";
+    return false;
+  }
+
+  // Try to use the valid volume control.
+  float volume = 0.0;
+  _com_error error = audio_volume->GetMasterVolume(&volume);
+  if (error.Error() != S_OK) {
+    RTC_LOG(LS_ERROR) << "ISimpleAudioVolume::GetMasterVolume failed: "
+                      << core_audio_utility::ErrorToString(error);
+    *available = false;
+  }
+  RTC_DLOG(LS_INFO) << "master volume for output audio session: " << volume;
+
+  *available = true;
+  return false;
+}
+
+// Internal test method which can be used in tests to emulate a restart signal.
+// It simply sets the same event which is normally triggered by session and
+// device notifications. Hence, the emulated restart sequence covers most parts
+// of a real sequence expect the actual device switch.
+bool CoreAudioBase::Restart() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+                    << "]";
+  if (!automatic_restart()) {
+    return false;
+  }
+  is_restarting_ = true;
+  SetEvent(restart_event_.Get());
+  return true;
+}
+
+void CoreAudioBase::StopThread() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK(!IsRestarting());
+  if (!audio_thread_.empty()) {
+    RTC_DLOG(LS_INFO) << "Sets stop_event...";
+    SetEvent(stop_event_.Get());
+    RTC_DLOG(LS_INFO) << "PlatformThread::Finalize...";
+    audio_thread_.Finalize();
+
+    // Ensure that we don't quit the main thread loop immediately next
+    // time Start() is called.
+    ResetEvent(stop_event_.Get());
+    ResetEvent(restart_event_.Get());
+  }
+}
+
+bool CoreAudioBase::HandleRestartEvent() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+                    << "]";
+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
+  RTC_DCHECK(!audio_thread_.empty());
+  RTC_DCHECK(IsRestarting());
+  // Let each client (input and/or output) take care of its own restart
+  // sequence since each side might need unique actions.
+  // TODO(henrika): revisit and investigate if one common base implementation
+  // is possible
+  bool restart_ok = on_error_callback_(ErrorType::kStreamDisconnected);
+  is_restarting_ = false;
+  return restart_ok;
+}
+
+bool CoreAudioBase::SwitchDeviceIfNeeded() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
+                    << "]";
+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
+  RTC_DCHECK(IsRestarting());
+
+  RTC_DLOG(LS_INFO) << "device_index=" << device_index_
+                    << " => device_id: " << device_id_;
+
+  // Ensure that at least one device exists and can be utilized. The most
+  // probable cause for ending up here is that a device has been removed.
+  if (core_audio_utility::NumberOfActiveDevices(IsInput() ? eCapture
+                                                          : eRender) < 1) {
+    RTC_DLOG(LS_ERROR) << "All devices are disabled or removed";
+    return false;
+  }
+
+  // Get the unique device ID for the index which is currently used. It seems
+  // safe to assume that if the ID is the same as the existing device ID, then
+  // the device configuration is the same as before.
+  std::string device_id = GetDeviceID(device_index_);
+  if (device_id != device_id_) {
+    RTC_LOG(LS_WARNING)
+        << "Device configuration has changed => changing device selection...";
+    // TODO(henrika): depending on the current state and how we got here, we
+    // must select a new device here.
+    if (SetDevice(kDefault) == -1) {
+      RTC_LOG(LS_WARNING) << "Failed to set new audio device";
+      return false;
+    }
+  } else {
+    RTC_LOG(LS_INFO)
+        << "Device configuration has not changed => keeping selected device";
+  }
+  return true;
+}
+
+AudioSessionState CoreAudioBase::GetAudioSessionState() const {
+  AudioSessionState state = AudioSessionStateInactive;
+  RTC_DCHECK(audio_session_control_.Get());
+  _com_error error = audio_session_control_->GetState(&state);
+  if (FAILED(error.Error())) {
+    RTC_DLOG(LS_ERROR) << "IAudioSessionControl::GetState failed: "
+                       << core_audio_utility::ErrorToString(error);
+  }
+  return state;
+}
+
+// TODO(henrika): only used for debugging purposes currently.
+ULONG CoreAudioBase::AddRef() {
+  ULONG new_ref = InterlockedIncrement(&ref_count_);
+  // RTC_DLOG(LS_INFO) << "__AddRef => " << new_ref;
+  return new_ref;
+}
+
+// TODO(henrika): does not call delete this.
+ULONG CoreAudioBase::Release() {
+  ULONG new_ref = InterlockedDecrement(&ref_count_);
+  // RTC_DLOG(LS_INFO) << "__Release => " << new_ref;
+  return new_ref;
+}
+
+// TODO(henrika): can probably be replaced by "return S_OK" only.
+HRESULT CoreAudioBase::QueryInterface(REFIID iid, void** object) {
+  if (object == nullptr) {
+    return E_POINTER;
+  }
+  if (iid == IID_IUnknown || iid == __uuidof(IAudioSessionEvents)) {
+    *object = static_cast<IAudioSessionEvents*>(this);
+    return S_OK;
+  }
+  *object = nullptr;
+  return E_NOINTERFACE;
+}
+
+// IAudioSessionEvents::OnStateChanged.
+HRESULT CoreAudioBase::OnStateChanged(AudioSessionState new_state) {
+  RTC_DLOG(LS_INFO) << "___" << __FUNCTION__ << "["
+                    << DirectionToString(direction())
+                    << "] new_state: " << SessionStateToString(new_state);
+  return S_OK;
+}
+
+// When a session is disconnected because of a device removal or format change
+// event, we want to inform the audio thread about the lost audio session and
+// trigger an attempt to restart audio using a new (default) device.
+// This method is called on separate threads owned by the session manager and
+// it can happen that the same type of callback is called more than once for the
+// same event.
+HRESULT CoreAudioBase::OnSessionDisconnected(
+    AudioSessionDisconnectReason disconnect_reason) {
+  RTC_DLOG(LS_INFO) << "___" << __FUNCTION__ << "["
+                    << DirectionToString(direction()) << "] reason: "
+                    << SessionDisconnectReasonToString(disconnect_reason);
+  // Ignore changes in the audio session (don't try to restart) if the user
+  // has explicitly asked for this type of ADM during construction.
+  if (!automatic_restart()) {
+    RTC_DLOG(LS_WARNING) << "___Automatic restart is disabled";
+    return S_OK;
+  }
+
+  if (IsRestarting()) {
+    RTC_DLOG(LS_WARNING) << "___Ignoring since restart is already active";
+    return S_OK;
+  }
+
+  // By default, automatic restart is enabled and the restart event will be set
+  // below if the device was removed or the format was changed.
+  if (disconnect_reason == DisconnectReasonDeviceRemoval ||
+      disconnect_reason == DisconnectReasonFormatChanged) {
+    is_restarting_ = true;
+    SetEvent(restart_event_.Get());
+  }
+  return S_OK;
+}
+
+// IAudioSessionEvents::OnDisplayNameChanged
+HRESULT CoreAudioBase::OnDisplayNameChanged(LPCWSTR new_display_name,
+                                            LPCGUID event_context) {
+  return S_OK;
+}
+
+// IAudioSessionEvents::OnIconPathChanged
+HRESULT CoreAudioBase::OnIconPathChanged(LPCWSTR new_icon_path,
+                                         LPCGUID event_context) {
+  return S_OK;
+}
+
+// IAudioSessionEvents::OnSimpleVolumeChanged
+HRESULT CoreAudioBase::OnSimpleVolumeChanged(float new_simple_volume,
+                                             BOOL new_mute,
+                                             LPCGUID event_context) {
+  return S_OK;
+}
+
+// IAudioSessionEvents::OnChannelVolumeChanged
+HRESULT CoreAudioBase::OnChannelVolumeChanged(DWORD channel_count,
+                                              float new_channel_volumes[],
+                                              DWORD changed_channel,
+                                              LPCGUID event_context) {
+  return S_OK;
+}
+
+// IAudioSessionEvents::OnGroupingParamChanged
+HRESULT CoreAudioBase::OnGroupingParamChanged(LPCGUID new_grouping_param,
+                                              LPCGUID event_context) {
+  return S_OK;
+}
+
+void CoreAudioBase::ThreadRun() {
+  if (!core_audio_utility::IsMMCSSSupported()) {
+    RTC_LOG(LS_ERROR) << "MMCSS is not supported";
+    return;
+  }
+  RTC_DLOG(LS_INFO) << "[" << DirectionToString(direction())
+                    << "] ThreadRun starts...";
+  // TODO(henrika): difference between "Pro Audio" and "Audio"?
+  ScopedMMCSSRegistration mmcss_registration(L"Pro Audio");
+  ScopedCOMInitializer com_initializer(ScopedCOMInitializer::kMTA);
+  RTC_DCHECK(mmcss_registration.Succeeded());
+  RTC_DCHECK(com_initializer.Succeeded());
+  RTC_DCHECK(stop_event_.IsValid());
+  RTC_DCHECK(audio_samples_event_.IsValid());
+
+  bool streaming = true;
+  bool error = false;
+  HANDLE wait_array[] = {stop_event_.Get(), restart_event_.Get(),
+                         audio_samples_event_.Get()};
+
+  // The device frequency is the frequency generated by the hardware clock in
+  // the audio device. The GetFrequency() method reports a constant frequency.
+  UINT64 device_frequency = 0;
+  _com_error result(S_FALSE);
+  if (audio_clock_) {
+    RTC_DCHECK(IsOutput());
+    result = audio_clock_->GetFrequency(&device_frequency);
+    if (FAILED(result.Error())) {
+      RTC_LOG(LS_ERROR) << "IAudioClock::GetFrequency failed: "
+                        << core_audio_utility::ErrorToString(result);
+    }
+  }
+
+  // Keep streaming audio until the stop event or the stream-switch event
+  // is signaled. An error event can also break the main thread loop.
+  while (streaming && !error) {
+    // Wait for a close-down event, stream-switch event or a new render event.
+    DWORD wait_result = WaitForMultipleObjects(arraysize(wait_array),
+                                               wait_array, false, INFINITE);
+    switch (wait_result) {
+      case WAIT_OBJECT_0 + 0:
+        // `stop_event_` has been set.
+        streaming = false;
+        break;
+      case WAIT_OBJECT_0 + 1:
+        // `restart_event_` has been set.
+        error = !HandleRestartEvent();
+        break;
+      case WAIT_OBJECT_0 + 2:
+        // `audio_samples_event_` has been set.
+        error = !on_data_callback_(device_frequency);
+        break;
+      default:
+        error = true;
+        break;
+    }
+  }
+
+  if (streaming && error) {
+    RTC_LOG(LS_ERROR) << "[" << DirectionToString(direction())
+                      << "] WASAPI streaming failed.";
+    // Stop audio streaming since something has gone wrong in our main thread
+    // loop. Note that, we are still in a "started" state, hence a Stop() call
+    // is required to join the thread properly.
+    result = audio_client_->Stop();
+    if (FAILED(result.Error())) {
+      RTC_LOG(LS_ERROR) << "IAudioClient::Stop failed: "
+                        << core_audio_utility::ErrorToString(result);
+    }
+
+    // TODO(henrika): notify clients that something has gone wrong and that
+    // this stream should be destroyed instead of reused in the future.
+  }
+
+  RTC_DLOG(LS_INFO) << "[" << DirectionToString(direction())
+                    << "] ...ThreadRun stops";
+}
+
+}  // namespace webrtc_win
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_base_win.h b/third_party/libwebrtc/modules/audio_device/win/core_audio_base_win.h
new file mode 100644
index 0000000000..6c1357e059
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_base_win.h
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_BASE_WIN_H_
+#define MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_BASE_WIN_H_
+
+#include <atomic>
+#include <functional>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "modules/audio_device/win/core_audio_utility_win.h"
+#include "rtc_base/platform_thread.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+class FineAudioBuffer;
+
+namespace webrtc_win {
+
+// Serves as base class for CoreAudioInput and CoreAudioOutput and supports
+// device handling and audio streaming where the direction (input or output)
+// is set at constructions by the parent.
+// The IAudioSessionEvents interface provides notifications of session-related
+// events such as changes in the volume level, display name, and session state.
+// This class does not use the default ref-counting memory management method
+// provided by IUnknown: calling CoreAudioBase::Release() will not delete the
+// object. The client will receive notification from the session manager on
+// a separate thread owned and controlled by the manager.
+// TODO(henrika): investigate if CoreAudioBase should implement
+// IMMNotificationClient as well (might improve support for device changes).
+class CoreAudioBase : public IAudioSessionEvents {
+ public:
+  enum class Direction {
+    kInput,
+    kOutput,
+  };
+
+  // TODO(henrika): add more error types.
+  enum class ErrorType {
+    kStreamDisconnected,
+  };
+
+  template <typename T>
+  auto as_integer(T const value) -> typename std::underlying_type<T>::type {
+    return static_cast<typename std::underlying_type<T>::type>(value);
+  }
+
+  // Callback definition for notifications of new audio data. For input clients,
+  // it means that "new audio data has now been captured", and for output
+  // clients, "the output layer now needs new audio data".
+  typedef std::function<bool(uint64_t device_frequency)> OnDataCallback;
+
+  // Callback definition for notifications of run-time error messages. It can
+  // be called e.g. when an active audio device is removed and an audio stream
+  // is disconnected (`error` is then set to kStreamDisconnected). Both input
+  // and output clients implements OnErrorCallback() and will trigger an
+  // internal restart sequence for kStreamDisconnected.
+  // This method is currently always called on the audio thread.
+  // TODO(henrika): add support for more error types.
+  typedef std::function<bool(ErrorType error)> OnErrorCallback;
+
+  void ThreadRun();
+
+  CoreAudioBase(const CoreAudioBase&) = delete;
+  CoreAudioBase& operator=(const CoreAudioBase&) = delete;
+
+ protected:
+  explicit CoreAudioBase(Direction direction,
+                         bool automatic_restart,
+                         OnDataCallback data_callback,
+                         OnErrorCallback error_callback);
+  ~CoreAudioBase();
+
+  std::string GetDeviceID(int index) const;
+  int SetDevice(int index);
+  int DeviceName(int index, std::string* name, std::string* guid) const;
+
+  // Checks if the current device ID is no longer in use (e.g. due to a
+  // disconnected stream), and if so, switches device to the default audio
+  // device. Called on the audio thread during restart attempts.
+  bool SwitchDeviceIfNeeded();
+
+  bool Init();
+  bool Start();
+  bool Stop();
+  bool IsVolumeControlAvailable(bool* available) const;
+  bool Restart();
+
+  Direction direction() const { return direction_; }
+  bool automatic_restart() const { return automatic_restart_; }
+
+  // Releases all allocated COM resources in the base class.
+  void ReleaseCOMObjects();
+
+  // Returns number of active devices given the specified `direction_` set
+  // by the parent (input or output).
+  int NumberOfActiveDevices() const;
+
+  // Returns total number of enumerated audio devices which is the sum of all
+  // active devices plus two extra (one default and one default
+  // communications). The value in `direction_` determines if capture or
+  // render devices are counted.
+  int NumberOfEnumeratedDevices() const;
+
+  bool IsInput() const;
+  bool IsOutput() const;
+  bool IsDefaultDevice(int index) const;
+  bool IsDefaultCommunicationsDevice(int index) const;
+  bool IsDefaultDeviceId(absl::string_view device_id) const;
+  bool IsDefaultCommunicationsDeviceId(absl::string_view device_id) const;
+  EDataFlow GetDataFlow() const;
+  bool IsRestarting() const;
+  int64_t TimeSinceStart() const;
+
+  // TODO(henrika): is the existing thread checker in WindowsAudioDeviceModule
+  // sufficient? As is, we have one top-level protection and then a second
+  // level here. In addition, calls to Init(), Start() and Stop() are not
+  // included to allow for support of internal restart (where these methods are
+  // called on the audio thread).
+  SequenceChecker thread_checker_;
+  SequenceChecker thread_checker_audio_;
+  AudioDeviceBuffer* audio_device_buffer_ = nullptr;
+  bool initialized_ = false;
+  WAVEFORMATEXTENSIBLE format_ = {};
+  uint32_t endpoint_buffer_size_frames_ = 0;
+  Microsoft::WRL::ComPtr<IAudioClock> audio_clock_;
+  Microsoft::WRL::ComPtr<IAudioClient> audio_client_;
+  bool is_active_ = false;
+  int64_t num_data_callbacks_ = 0;
+  int latency_ms_ = 0;
+  absl::optional<uint32_t> sample_rate_;
+
+ private:
+  const Direction direction_;
+  const bool automatic_restart_;
+  const OnDataCallback on_data_callback_;
+  const OnErrorCallback on_error_callback_;
+  ScopedHandle audio_samples_event_;
+  ScopedHandle stop_event_;
+  ScopedHandle restart_event_;
+  int64_t start_time_ = 0;
+  std::string device_id_;
+  int device_index_ = -1;
+  // Used by the IAudioSessionEvents implementations. Currently only utilized
+  // for debugging purposes.
+  LONG ref_count_ = 1;
+  // Set when restart process starts and cleared when restart stops
+  // successfully. Accessed atomically.
+  std::atomic<bool> is_restarting_;
+  rtc::PlatformThread audio_thread_;
+  Microsoft::WRL::ComPtr<IAudioSessionControl> audio_session_control_;
+
+  void StopThread();
+  AudioSessionState GetAudioSessionState() const;
+
+  // Called on the audio thread when a restart event has been set.
+  // It will then trigger calls to the installed error callbacks with error
+  // type set to kStreamDisconnected.
+  bool HandleRestartEvent();
+
+  // IUnknown (required by IAudioSessionEvents and IMMNotificationClient).
+  ULONG __stdcall AddRef() override;
+  ULONG __stdcall Release() override;
+  HRESULT __stdcall QueryInterface(REFIID iid, void** object) override;
+
+  // IAudioSessionEvents implementation.
+  // These methods are called on separate threads owned by the session manager.
+  // More than one thread can be involved depending on the type of callback
+  // and audio session.
+  HRESULT __stdcall OnStateChanged(AudioSessionState new_state) override;
+  HRESULT __stdcall OnSessionDisconnected(
+      AudioSessionDisconnectReason disconnect_reason) override;
+  HRESULT __stdcall OnDisplayNameChanged(LPCWSTR new_display_name,
+                                         LPCGUID event_context) override;
+  HRESULT __stdcall OnIconPathChanged(LPCWSTR new_icon_path,
+                                      LPCGUID event_context) override;
+  HRESULT __stdcall OnSimpleVolumeChanged(float new_simple_volume,
+                                          BOOL new_mute,
+                                          LPCGUID event_context) override;
+  HRESULT __stdcall OnChannelVolumeChanged(DWORD channel_count,
+                                           float new_channel_volumes[],
+                                           DWORD changed_channel,
+                                           LPCGUID event_context) override;
+  HRESULT __stdcall OnGroupingParamChanged(LPCGUID new_grouping_param,
+                                           LPCGUID event_context) override;
+};
+
+}  // namespace webrtc_win
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_BASE_WIN_H_
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_input_win.cc b/third_party/libwebrtc/modules/audio_device/win/core_audio_input_win.cc
new file mode 100644
index 0000000000..17790dafc4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_input_win.cc
@@ -0,0 +1,453 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/win/core_audio_input_win.h"
+
+#include <memory>
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+using Microsoft::WRL::ComPtr;
+
+namespace webrtc {
+namespace webrtc_win {
+
+enum AudioDeviceMessageType : uint32_t {
+  kMessageInputStreamDisconnected,
+};
+
+CoreAudioInput::CoreAudioInput(bool automatic_restart)
+    : CoreAudioBase(
+          CoreAudioBase::Direction::kInput,
+          automatic_restart,
+          [this](uint64_t freq) { return OnDataCallback(freq); },
+          [this](ErrorType err) { return OnErrorCallback(err); }) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  thread_checker_audio_.Detach();
+}
+
+CoreAudioInput::~CoreAudioInput() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+}
+
+int CoreAudioInput::Init() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return 0;
+}
+
+int CoreAudioInput::Terminate() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  StopRecording();
+  return 0;
+}
+
+int CoreAudioInput::NumDevices() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return core_audio_utility::NumberOfActiveDevices(eCapture);
+}
+
+int CoreAudioInput::SetDevice(int index) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
+  RTC_DCHECK_GE(index, 0);
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return CoreAudioBase::SetDevice(index);
+}
+
+int CoreAudioInput::SetDevice(AudioDeviceModule::WindowsDeviceType device) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": "
+                    << ((device == AudioDeviceModule::kDefaultDevice)
+                            ? "Default"
+                            : "DefaultCommunication");
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return SetDevice((device == AudioDeviceModule::kDefaultDevice) ? 0 : 1);
+}
+
+int CoreAudioInput::DeviceName(int index,
+                               std::string* name,
+                               std::string* guid) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK(name);
+  return CoreAudioBase::DeviceName(index, name, guid);
+}
+
+void CoreAudioInput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  audio_device_buffer_ = audio_buffer;
+}
+
+bool CoreAudioInput::RecordingIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << initialized_;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return initialized_;
+}
+
+int CoreAudioInput::InitRecording() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!Recording());
+  RTC_DCHECK(!audio_capture_client_);
+
+  // Creates an IAudioClient instance and stores the valid interface pointer in
+  // `audio_client3_`, `audio_client2_`, or `audio_client_` depending on
+  // platform support. The base class will use optimal input parameters and do
+  // an event driven shared mode initialization. The utilized format will be
+  // stored in `format_` and can be used for configuration and allocation of
+  // audio buffers.
+  if (!CoreAudioBase::Init()) {
+    return -1;
+  }
+  RTC_DCHECK(audio_client_);
+
+  // Configure the recording side of the audio device buffer using `format_`
+  // after a trivial sanity check of the format structure.
+  RTC_DCHECK(audio_device_buffer_);
+  WAVEFORMATEX* format = &format_.Format;
+  RTC_DCHECK_EQ(format->wFormatTag, WAVE_FORMAT_EXTENSIBLE);
+  audio_device_buffer_->SetRecordingSampleRate(format->nSamplesPerSec);
+  audio_device_buffer_->SetRecordingChannels(format->nChannels);
+
+  // Create a modified audio buffer class which allows us to supply any number
+  // of samples (and not only multiple of 10ms) to match the optimal buffer
+  // size per callback used by Core Audio.
+  // TODO(henrika): can we share one FineAudioBuffer with the output side?
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+
+  // Create an IAudioCaptureClient for an initialized IAudioClient.
+  // The IAudioCaptureClient interface enables a client to read input data from
+  // a capture endpoint buffer.
+  ComPtr<IAudioCaptureClient> audio_capture_client =
+      core_audio_utility::CreateCaptureClient(audio_client_.Get());
+  if (!audio_capture_client) {
+    return -1;
+  }
+
+  // Query performance frequency.
+  LARGE_INTEGER ticks_per_sec = {};
+  qpc_to_100ns_.reset();
+  if (::QueryPerformanceFrequency(&ticks_per_sec)) {
+    double qpc_ticks_per_second =
+        rtc::dchecked_cast<double>(ticks_per_sec.QuadPart);
+    qpc_to_100ns_ = 10000000.0 / qpc_ticks_per_second;
+  }
+
+  // Store valid COM interfaces.
+  audio_capture_client_ = audio_capture_client;
+
+  initialized_ = true;
+  return 0;
+}
+
+int CoreAudioInput::StartRecording() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK(!Recording());
+  RTC_DCHECK(fine_audio_buffer_);
+  RTC_DCHECK(audio_device_buffer_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Recording can not start since InitRecording must succeed first";
+    return 0;
+  }
+
+  fine_audio_buffer_->ResetRecord();
+  if (!IsRestarting()) {
+    audio_device_buffer_->StartRecording();
+  }
+
+  if (!Start()) {
+    return -1;
+  }
+
+  is_active_ = true;
+  return 0;
+}
+
+int CoreAudioInput::StopRecording() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  if (!initialized_) {
+    return 0;
+  }
+
+  // Release resources allocated in InitRecording() and then return if this
+  // method is called without any active input audio.
+  if (!Recording()) {
+    RTC_DLOG(LS_WARNING) << "No input stream is active";
+    ReleaseCOMObjects();
+    initialized_ = false;
+    return 0;
+  }
+
+  if (!Stop()) {
+    RTC_LOG(LS_ERROR) << "StopRecording failed";
+    return -1;
+  }
+
+  if (!IsRestarting()) {
+    RTC_DCHECK(audio_device_buffer_);
+    audio_device_buffer_->StopRecording();
+  }
+
+  // Release all allocated resources to allow for a restart without
+  // intermediate destruction.
+  ReleaseCOMObjects();
+  qpc_to_100ns_.reset();
+
+  initialized_ = false;
+  is_active_ = false;
+  return 0;
+}
+
+bool CoreAudioInput::Recording() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << is_active_;
+  return is_active_;
+}
+
+// TODO(henrika): finalize support of audio session volume control. As is, we
+// are not compatible with the old ADM implementation since it allows accessing
+// the volume control with any active audio output stream.
+int CoreAudioInput::VolumeIsAvailable(bool* available) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return IsVolumeControlAvailable(available) ? 0 : -1;
+}
+
+// Triggers the restart sequence. Only used for testing purposes to emulate
+// a real event where e.g. an active input device is removed.
+int CoreAudioInput::RestartRecording() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (!Recording()) {
+    return 0;
+  }
+
+  if (!Restart()) {
+    RTC_LOG(LS_ERROR) << "RestartRecording failed";
+    return -1;
+  }
+  return 0;
+}
+
+bool CoreAudioInput::Restarting() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return IsRestarting();
+}
+
+int CoreAudioInput::SetSampleRate(uint32_t sample_rate) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  sample_rate_ = sample_rate;
+  return 0;
+}
+
+void CoreAudioInput::ReleaseCOMObjects() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  CoreAudioBase::ReleaseCOMObjects();
+  if (audio_capture_client_.Get()) {
+    audio_capture_client_.Reset();
+  }
+}
+
+bool CoreAudioInput::OnDataCallback(uint64_t device_frequency) {
+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
+
+  if (!initialized_ || !is_active_) {
+    // This is concurrent examination of state across multiple threads so will
+    // be somewhat error prone, but we should still be defensive and not use
+    // audio_capture_client_ if we know it's not there.
+    return false;
+  }
+  if (num_data_callbacks_ == 0) {
+    RTC_LOG(LS_INFO) << "--- Input audio stream is alive ---";
+  }
+  UINT32 num_frames_in_next_packet = 0;
+  _com_error error =
+      audio_capture_client_->GetNextPacketSize(&num_frames_in_next_packet);
+  if (error.Error() == AUDCLNT_E_DEVICE_INVALIDATED) {
+    // Avoid breaking the thread loop implicitly by returning false and return
+    // true instead for AUDCLNT_E_DEVICE_INVALIDATED even it is a valid error
+    // message. We will use notifications about device changes instead to stop
+    // data callbacks and attempt to restart streaming .
+    RTC_DLOG(LS_ERROR) << "AUDCLNT_E_DEVICE_INVALIDATED";
+    return true;
+  }
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioCaptureClient::GetNextPacketSize failed: "
+                      << core_audio_utility::ErrorToString(error);
+    return false;
+  }
+
+  // Drain the WASAPI capture buffer fully if audio has been recorded.
+  while (num_frames_in_next_packet > 0) {
+    uint8_t* audio_data;
+    UINT32 num_frames_to_read = 0;
+    DWORD flags = 0;
+    UINT64 device_position_frames = 0;
+    UINT64 capture_time_100ns = 0;
+    error = audio_capture_client_->GetBuffer(&audio_data, &num_frames_to_read,
+                                             &flags, &device_position_frames,
+                                             &capture_time_100ns);
+    if (error.Error() == AUDCLNT_S_BUFFER_EMPTY) {
+      // The call succeeded but no capture data is available to be read.
+      // Return and start waiting for new capture event
+      RTC_DCHECK_EQ(num_frames_to_read, 0u);
+      return true;
+    }
+    if (FAILED(error.Error())) {
+      RTC_LOG(LS_ERROR) << "IAudioCaptureClient::GetBuffer failed: "
+                        << core_audio_utility::ErrorToString(error);
+      return false;
+    }
+
+    // Update input delay estimate but only about once per second to save
+    // resources. The estimate is usually stable.
+    if (num_data_callbacks_ % 100 == 0) {
+      absl::optional<int> opt_record_delay_ms;
+      // TODO(henrika): note that FineAudioBuffer adds latency as well.
+      opt_record_delay_ms = EstimateLatencyMillis(capture_time_100ns);
+      if (opt_record_delay_ms) {
+        latency_ms_ = *opt_record_delay_ms;
+      } else {
+        RTC_DLOG(LS_WARNING) << "Input latency is set to fixed value";
+        latency_ms_ = 20;
+      }
+    }
+    if (num_data_callbacks_ % 500 == 0) {
+      RTC_DLOG(LS_INFO) << "latency: " << latency_ms_;
+    }
+
+    // The data in the packet is not correlated with the previous packet's
+    // device position; possibly due to a stream state transition or timing
+    // glitch. The behavior of the AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY flag
+    // is undefined on the application's first call to GetBuffer after Start.
+    if (device_position_frames != 0 &&
+        flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) {
+      RTC_DLOG(LS_WARNING) << "AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY";
+    }
+    // The time at which the device's stream position was recorded is uncertain.
+    // Thus, the client might be unable to accurately set a time stamp for the
+    // current data packet.
+    if (flags & AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR) {
+      RTC_DLOG(LS_WARNING) << "AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR";
+    }
+
+    // Treat all of the data in the packet as silence and ignore the actual
+    // data values when AUDCLNT_BUFFERFLAGS_SILENT is set.
+    if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
+      rtc::ExplicitZeroMemory(audio_data,
+                              format_.Format.nBlockAlign * num_frames_to_read);
+      RTC_DLOG(LS_WARNING) << "Captured audio is replaced by silence";
+    } else {
+      // Copy recorded audio in `audio_data` to the WebRTC sink using the
+      // FineAudioBuffer object.
+      fine_audio_buffer_->DeliverRecordedData(
+          rtc::MakeArrayView(reinterpret_cast<const int16_t*>(audio_data),
+                             format_.Format.nChannels * num_frames_to_read),
+
+          latency_ms_);
+    }
+
+    error = audio_capture_client_->ReleaseBuffer(num_frames_to_read);
+    if (FAILED(error.Error())) {
+      RTC_LOG(LS_ERROR) << "IAudioCaptureClient::ReleaseBuffer failed: "
+                        << core_audio_utility::ErrorToString(error);
+      return false;
+    }
+
+    error =
+        audio_capture_client_->GetNextPacketSize(&num_frames_in_next_packet);
+    if (FAILED(error.Error())) {
+      RTC_LOG(LS_ERROR) << "IAudioCaptureClient::GetNextPacketSize failed: "
+                        << core_audio_utility::ErrorToString(error);
+      return false;
+    }
+  }
+  ++num_data_callbacks_;
+  return true;
+}
+
+bool CoreAudioInput::OnErrorCallback(ErrorType error) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << as_integer(error);
+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
+  if (error == CoreAudioBase::ErrorType::kStreamDisconnected) {
+    HandleStreamDisconnected();
+  } else {
+    RTC_DLOG(LS_WARNING) << "Unsupported error type";
+  }
+  return true;
+}
+
+absl::optional<int> CoreAudioInput::EstimateLatencyMillis(
+    uint64_t capture_time_100ns) {
+  if (!qpc_to_100ns_) {
+    return absl::nullopt;
+  }
+  // Input parameter `capture_time_100ns` contains the performance counter at
+  // the time that the audio endpoint device recorded the device position of
+  // the first audio frame in the data packet converted into 100ns units.
+  // We derive a delay estimate by:
+  // - sampling the current performance counter (qpc_now_raw),
+  // - converting it into 100ns time units (now_time_100ns), and
+  // - subtracting `capture_time_100ns` from now_time_100ns.
+  LARGE_INTEGER perf_counter_now = {};
+  if (!::QueryPerformanceCounter(&perf_counter_now)) {
+    return absl::nullopt;
+  }
+  uint64_t qpc_now_raw = perf_counter_now.QuadPart;
+  uint64_t now_time_100ns = qpc_now_raw * (*qpc_to_100ns_);
+  webrtc::TimeDelta delay_us = webrtc::TimeDelta::Micros(
+      0.1 * (now_time_100ns - capture_time_100ns) + 0.5);
+  return delay_us.ms();
+}
+
+// Called from OnErrorCallback() when error type is kStreamDisconnected.
+// Note that this method is called on the audio thread and the internal restart
+// sequence is also executed on that same thread. The audio thread is therefore
+// not stopped during restart. Such a scheme also makes the restart process less
+// complex.
+// Note that, none of the called methods are thread checked since they can also
+// be called on the main thread. Thread checkers are instead added on one layer
+// above (in audio_device_module.cc) which ensures that the public API is thread
+// safe.
+// TODO(henrika): add more details.
+bool CoreAudioInput::HandleStreamDisconnected() {
+  RTC_DLOG(LS_INFO) << "<<<--- " << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
+  RTC_DCHECK(automatic_restart());
+
+  if (StopRecording() != 0) {
+    return false;
+  }
+
+  if (!SwitchDeviceIfNeeded()) {
+    return false;
+  }
+
+  if (InitRecording() != 0) {
+    return false;
+  }
+  if (StartRecording() != 0) {
+    return false;
+  }
+
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " --->>>";
+  return true;
+}
+
+}  // namespace webrtc_win
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_input_win.h b/third_party/libwebrtc/modules/audio_device/win/core_audio_input_win.h
new file mode 100644
index 0000000000..be290f9f4e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_input_win.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_INPUT_WIN_H_
+#define MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_INPUT_WIN_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "modules/audio_device/win/audio_device_module_win.h"
+#include "modules/audio_device/win/core_audio_base_win.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+class FineAudioBuffer;
+
+namespace webrtc_win {
+
+// Windows specific AudioInput implementation using a CoreAudioBase class where
+// an input direction is set at construction. Supports capture device handling
+// and streaming of captured audio to a WebRTC client.
+class CoreAudioInput final : public CoreAudioBase, public AudioInput {
+ public:
+  CoreAudioInput(bool automatic_restart);
+  ~CoreAudioInput() override;
+
+  // AudioInput implementation.
+  int Init() override;
+  int Terminate() override;
+  int NumDevices() const override;
+  int SetDevice(int index) override;
+  int SetDevice(AudioDeviceModule::WindowsDeviceType device) override;
+  int DeviceName(int index, std::string* name, std::string* guid) override;
+  void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
+  bool RecordingIsInitialized() const override;
+  int InitRecording() override;
+  int StartRecording() override;
+  int StopRecording() override;
+  bool Recording() override;
+  int VolumeIsAvailable(bool* available) override;
+  int RestartRecording() override;
+  bool Restarting() const override;
+  int SetSampleRate(uint32_t sample_rate) override;
+
+  CoreAudioInput(const CoreAudioInput&) = delete;
+  CoreAudioInput& operator=(const CoreAudioInput&) = delete;
+
+ private:
+  void ReleaseCOMObjects();
+  bool OnDataCallback(uint64_t device_frequency);
+  bool OnErrorCallback(ErrorType error);
+  absl::optional<int> EstimateLatencyMillis(uint64_t capture_time_100ns);
+  bool HandleStreamDisconnected();
+
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+  Microsoft::WRL::ComPtr<IAudioCaptureClient> audio_capture_client_;
+  absl::optional<double> qpc_to_100ns_;
+};
+
+}  // namespace webrtc_win
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_INPUT_WIN_H_
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_output_win.cc b/third_party/libwebrtc/modules/audio_device/win/core_audio_output_win.cc
new file mode 100644
index 0000000000..c92fedf0e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_output_win.cc
@@ -0,0 +1,422 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/win/core_audio_output_win.h"
+
+#include <memory>
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+
+using Microsoft::WRL::ComPtr;
+
+namespace webrtc {
+namespace webrtc_win {
+
+CoreAudioOutput::CoreAudioOutput(bool automatic_restart)
+    : CoreAudioBase(
+          CoreAudioBase::Direction::kOutput,
+          automatic_restart,
+          [this](uint64_t freq) { return OnDataCallback(freq); },
+          [this](ErrorType err) { return OnErrorCallback(err); }) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  thread_checker_audio_.Detach();
+}
+
+CoreAudioOutput::~CoreAudioOutput() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  Terminate();
+}
+
+int CoreAudioOutput::Init() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return 0;
+}
+
+int CoreAudioOutput::Terminate() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  StopPlayout();
+  return 0;
+}
+
+int CoreAudioOutput::NumDevices() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return core_audio_utility::NumberOfActiveDevices(eRender);
+}
+
+int CoreAudioOutput::SetDevice(int index) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
+  RTC_DCHECK_GE(index, 0);
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return CoreAudioBase::SetDevice(index);
+}
+
+int CoreAudioOutput::SetDevice(AudioDeviceModule::WindowsDeviceType device) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": "
+                    << ((device == AudioDeviceModule::kDefaultDevice)
+                            ? "Default"
+                            : "DefaultCommunication");
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return SetDevice((device == AudioDeviceModule::kDefaultDevice) ? 0 : 1);
+}
+
+int CoreAudioOutput::DeviceName(int index,
+                                std::string* name,
+                                std::string* guid) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << index;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK(name);
+  return CoreAudioBase::DeviceName(index, name, guid);
+}
+
+void CoreAudioOutput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  audio_device_buffer_ = audio_buffer;
+}
+
+bool CoreAudioOutput::PlayoutIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return initialized_;
+}
+
+int CoreAudioOutput::InitPlayout() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << IsRestarting();
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!Playing());
+  RTC_DCHECK(!audio_render_client_);
+
+  // Creates an IAudioClient instance and stores the valid interface pointer in
+  // `audio_client3_`, `audio_client2_`, or `audio_client_` depending on
+  // platform support. The base class will use optimal output parameters and do
+  // an event driven shared mode initialization. The utilized format will be
+  // stored in `format_` and can be used for configuration and allocation of
+  // audio buffers.
+  if (!CoreAudioBase::Init()) {
+    return -1;
+  }
+  RTC_DCHECK(audio_client_);
+
+  // Configure the playout side of the audio device buffer using `format_`
+  // after a trivial sanity check of the format structure.
+  RTC_DCHECK(audio_device_buffer_);
+  WAVEFORMATEX* format = &format_.Format;
+  RTC_DCHECK_EQ(format->wFormatTag, WAVE_FORMAT_EXTENSIBLE);
+  audio_device_buffer_->SetPlayoutSampleRate(format->nSamplesPerSec);
+  audio_device_buffer_->SetPlayoutChannels(format->nChannels);
+
+  // Create a modified audio buffer class which allows us to ask for any number
+  // of samples (and not only multiple of 10ms) to match the optimal
+  // buffer size per callback used by Core Audio.
+  // TODO(henrika): can we share one FineAudioBuffer with the input side?
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+
+  // Create an IAudioRenderClient for an initialized IAudioClient.
+  // The IAudioRenderClient interface enables us to write output data to
+  // a rendering endpoint buffer.
+  ComPtr<IAudioRenderClient> audio_render_client =
+      core_audio_utility::CreateRenderClient(audio_client_.Get());
+  if (!audio_render_client.Get()) {
+    return -1;
+  }
+
+  ComPtr<IAudioClock> audio_clock =
+      core_audio_utility::CreateAudioClock(audio_client_.Get());
+  if (!audio_clock.Get()) {
+    return -1;
+  }
+
+  // Store valid COM interfaces.
+  audio_render_client_ = audio_render_client;
+  audio_clock_ = audio_clock;
+
+  initialized_ = true;
+  return 0;
+}
+
+int CoreAudioOutput::StartPlayout() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << IsRestarting();
+  RTC_DCHECK(!Playing());
+  RTC_DCHECK(fine_audio_buffer_);
+  RTC_DCHECK(audio_device_buffer_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Playout can not start since InitPlayout must succeed first";
+  }
+
+  fine_audio_buffer_->ResetPlayout();
+  if (!IsRestarting()) {
+    audio_device_buffer_->StartPlayout();
+  }
+
+  if (!core_audio_utility::FillRenderEndpointBufferWithSilence(
+          audio_client_.Get(), audio_render_client_.Get())) {
+    RTC_LOG(LS_WARNING) << "Failed to prepare output endpoint with silence";
+  }
+
+  num_frames_written_ = endpoint_buffer_size_frames_;
+
+  if (!Start()) {
+    return -1;
+  }
+
+  is_active_ = true;
+  return 0;
+}
+
+int CoreAudioOutput::StopPlayout() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << IsRestarting();
+  if (!initialized_) {
+    return 0;
+  }
+
+  // Release resources allocated in InitPlayout() and then return if this
+  // method is called without any active output audio.
+  if (!Playing()) {
+    RTC_DLOG(LS_WARNING) << "No output stream is active";
+    ReleaseCOMObjects();
+    initialized_ = false;
+    return 0;
+  }
+
+  if (!Stop()) {
+    RTC_LOG(LS_ERROR) << "StopPlayout failed";
+    return -1;
+  }
+
+  if (!IsRestarting()) {
+    RTC_DCHECK(audio_device_buffer_);
+    audio_device_buffer_->StopPlayout();
+  }
+
+  // Release all allocated resources to allow for a restart without
+  // intermediate destruction.
+  ReleaseCOMObjects();
+
+  initialized_ = false;
+  is_active_ = false;
+  return 0;
+}
+
+bool CoreAudioOutput::Playing() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << is_active_;
+  return is_active_;
+}
+
+// TODO(henrika): finalize support of audio session volume control. As is, we
+// are not compatible with the old ADM implementation since it allows accessing
+// the volume control with any active audio output stream.
+int CoreAudioOutput::VolumeIsAvailable(bool* available) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return IsVolumeControlAvailable(available) ? 0 : -1;
+}
+
+// Triggers the restart sequence. Only used for testing purposes to emulate
+// a real event where e.g. an active output device is removed.
+int CoreAudioOutput::RestartPlayout() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (!Playing()) {
+    return 0;
+  }
+  if (!Restart()) {
+    RTC_LOG(LS_ERROR) << "RestartPlayout failed";
+    return -1;
+  }
+  return 0;
+}
+
+bool CoreAudioOutput::Restarting() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return IsRestarting();
+}
+
+int CoreAudioOutput::SetSampleRate(uint32_t sample_rate) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  sample_rate_ = sample_rate;
+  return 0;
+}
+
+void CoreAudioOutput::ReleaseCOMObjects() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  CoreAudioBase::ReleaseCOMObjects();
+  if (audio_render_client_.Get()) {
+    audio_render_client_.Reset();
+  }
+}
+
+bool CoreAudioOutput::OnErrorCallback(ErrorType error) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << as_integer(error);
+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
+  if (!initialized_ || !Playing()) {
+    return true;
+  }
+
+  if (error == CoreAudioBase::ErrorType::kStreamDisconnected) {
+    HandleStreamDisconnected();
+  } else {
+    RTC_DLOG(LS_WARNING) << "Unsupported error type";
+  }
+  return true;
+}
+
+bool CoreAudioOutput::OnDataCallback(uint64_t device_frequency) {
+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
+  if (num_data_callbacks_ == 0) {
+    RTC_LOG(LS_INFO) << "--- Output audio stream is alive ---";
+  }
+  // Get the padding value which indicates the amount of valid unread data that
+  // the endpoint buffer currently contains.
+  UINT32 num_unread_frames = 0;
+  _com_error error = audio_client_->GetCurrentPadding(&num_unread_frames);
+  if (error.Error() == AUDCLNT_E_DEVICE_INVALIDATED) {
+    // Avoid breaking the thread loop implicitly by returning false and return
+    // true instead for AUDCLNT_E_DEVICE_INVALIDATED even it is a valid error
+    // message. We will use notifications about device changes instead to stop
+    // data callbacks and attempt to restart streaming .
+    RTC_DLOG(LS_ERROR) << "AUDCLNT_E_DEVICE_INVALIDATED";
+    return true;
+  }
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::GetCurrentPadding failed: "
+                      << core_audio_utility::ErrorToString(error);
+    return false;
+  }
+
+  // Contains how much new data we can write to the buffer without the risk of
+  // overwriting previously written data that the audio engine has not yet read
+  // from the buffer. I.e., it is the maximum buffer size we can request when
+  // calling IAudioRenderClient::GetBuffer().
+  UINT32 num_requested_frames =
+      endpoint_buffer_size_frames_ - num_unread_frames;
+  if (num_requested_frames == 0) {
+    RTC_DLOG(LS_WARNING)
+        << "Audio thread is signaled but no new audio samples are needed";
+    return true;
+  }
+
+  // Request all available space in the rendering endpoint buffer into which the
+  // client can later write an audio packet.
+  uint8_t* audio_data;
+  error = audio_render_client_->GetBuffer(num_requested_frames, &audio_data);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioRenderClient::GetBuffer failed: "
+                      << core_audio_utility::ErrorToString(error);
+    return false;
+  }
+
+  // Update output delay estimate but only about once per second to save
+  // resources. The estimate is usually stable.
+  if (num_data_callbacks_ % 100 == 0) {
+    // TODO(henrika): note that FineAudioBuffer adds latency as well.
+    latency_ms_ = EstimateOutputLatencyMillis(device_frequency);
+    if (num_data_callbacks_ % 500 == 0) {
+      RTC_DLOG(LS_INFO) << "latency: " << latency_ms_;
+    }
+  }
+
+  // Get audio data from WebRTC and write it to the allocated buffer in
+  // `audio_data`. The playout latency is not updated for each callback.
+  fine_audio_buffer_->GetPlayoutData(
+      rtc::MakeArrayView(reinterpret_cast<int16_t*>(audio_data),
+                         num_requested_frames * format_.Format.nChannels),
+      latency_ms_);
+
+  // Release the buffer space acquired in IAudioRenderClient::GetBuffer.
+  error = audio_render_client_->ReleaseBuffer(num_requested_frames, 0);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioRenderClient::ReleaseBuffer failed: "
+                      << core_audio_utility::ErrorToString(error);
+    return false;
+  }
+
+  num_frames_written_ += num_requested_frames;
+  ++num_data_callbacks_;
+
+  return true;
+}
+
+// TODO(henrika): IAudioClock2::GetDevicePosition could perhaps be used here
+// instead. Tried it once, but it crashed for capture devices.
+int CoreAudioOutput::EstimateOutputLatencyMillis(uint64_t device_frequency) {
+  UINT64 position = 0;
+  UINT64 qpc_position = 0;
+  int delay_ms = 0;
+  // Get the device position through output parameter `position`. This is the
+  // stream position of the sample that is currently playing through the
+  // speakers.
+  _com_error error = audio_clock_->GetPosition(&position, &qpc_position);
+  if (error.Error() == S_OK) {
+    // Number of frames already played out through the speaker.
+    const uint64_t num_played_out_frames =
+        format_.Format.nSamplesPerSec * position / device_frequency;
+
+    // Number of frames that have been written to the buffer but not yet
+    // played out corresponding to the estimated latency measured in number
+    // of audio frames.
+    const uint64_t delay_frames = num_frames_written_ - num_played_out_frames;
+
+    // Convert latency in number of frames into milliseconds.
+    webrtc::TimeDelta delay =
+        webrtc::TimeDelta::Micros(delay_frames * rtc::kNumMicrosecsPerSec /
+                                  format_.Format.nSamplesPerSec);
+    delay_ms = delay.ms();
+  }
+  return delay_ms;
+}
+
+// Called from OnErrorCallback() when error type is kStreamDisconnected.
+// Note that this method is called on the audio thread and the internal restart
+// sequence is also executed on that same thread. The audio thread is therefore
+// not stopped during restart. Such a scheme also makes the restart process less
+// complex.
+// Note that, none of the called methods are thread checked since they can also
+// be called on the main thread. Thread checkers are instead added on one layer
+// above (in audio_device_module.cc) which ensures that the public API is thread
+// safe.
+// TODO(henrika): add more details.
+bool CoreAudioOutput::HandleStreamDisconnected() {
+  RTC_DLOG(LS_INFO) << "<<<--- " << __FUNCTION__;
+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
+  RTC_DCHECK(automatic_restart());
+
+  if (StopPlayout() != 0) {
+    return false;
+  }
+
+  if (!SwitchDeviceIfNeeded()) {
+    return false;
+  }
+
+  if (InitPlayout() != 0) {
+    return false;
+  }
+  if (StartPlayout() != 0) {
+    return false;
+  }
+
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " --->>>";
+  return true;
+}
+
+}  // namespace webrtc_win
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_output_win.h b/third_party/libwebrtc/modules/audio_device/win/core_audio_output_win.h
new file mode 100644
index 0000000000..5a547498a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_output_win.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_OUTPUT_WIN_H_
+#define MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_OUTPUT_WIN_H_
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_device/win/audio_device_module_win.h"
+#include "modules/audio_device/win/core_audio_base_win.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+class FineAudioBuffer;
+
+namespace webrtc_win {
+
+// Windows specific AudioOutput implementation using a CoreAudioBase class where
+// an output direction is set at construction. Supports render device handling
+// and streaming of decoded audio from a WebRTC client to the native audio
+// layer.
+class CoreAudioOutput final : public CoreAudioBase, public AudioOutput {
+ public:
+  CoreAudioOutput(bool automatic_restart);
+  ~CoreAudioOutput() override;
+
+  // AudioOutput implementation.
+  int Init() override;
+  int Terminate() override;
+  int NumDevices() const override;
+  int SetDevice(int index) override;
+  int SetDevice(AudioDeviceModule::WindowsDeviceType device) override;
+  int DeviceName(int index, std::string* name, std::string* guid) override;
+  void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
+  bool PlayoutIsInitialized() const override;
+  int InitPlayout() override;
+  int StartPlayout() override;
+  int StopPlayout() override;
+  bool Playing() override;
+  int VolumeIsAvailable(bool* available) override;
+  int RestartPlayout() override;
+  bool Restarting() const override;
+  int SetSampleRate(uint32_t sample_rate) override;
+
+  CoreAudioOutput(const CoreAudioOutput&) = delete;
+  CoreAudioOutput& operator=(const CoreAudioOutput&) = delete;
+
+ private:
+  void ReleaseCOMObjects();
+  bool OnDataCallback(uint64_t device_frequency);
+  bool OnErrorCallback(ErrorType error);
+  int EstimateOutputLatencyMillis(uint64_t device_frequency);
+  bool HandleStreamDisconnected();
+
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+  Microsoft::WRL::ComPtr<IAudioRenderClient> audio_render_client_;
+  uint64_t num_frames_written_ = 0;
+};
+
+}  // namespace webrtc_win
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_OUTPUT_WIN_H_
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_utility_win.cc b/third_party/libwebrtc/modules/audio_device/win/core_audio_utility_win.cc
new file mode 100644
index 0000000000..e4e2864db5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_utility_win.cc
@@ -0,0 +1,1529 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/win/core_audio_utility_win.h"
+
+#include <functiondiscoverykeys_devpkey.h>
+#include <stdio.h>
+#include <tchar.h>
+
+#include <iomanip>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread_types.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/win/windows_version.h"
+
+using Microsoft::WRL::ComPtr;
+using webrtc::AudioDeviceName;
+using webrtc::AudioParameters;
+
+namespace webrtc {
+namespace webrtc_win {
+namespace {
+
+using core_audio_utility::ErrorToString;
+
+// Converts from channel mask to list of included channels.
+// Each audio data format contains channels for one or more of the positions
+// listed below. The number of channels simply equals the number of nonzero
+// flag bits in the `channel_mask`. The relative positions of the channels
+// within each block of audio data always follow the same relative ordering
+// as the flag bits in the table below. For example, if `channel_mask` contains
+// the value 0x00000033, the format defines four audio channels that are
+// assigned for playback to the front-left, front-right, back-left,
+// and back-right speakers, respectively. The channel data should be interleaved
+// in that order within each block.
+std::string ChannelMaskToString(DWORD channel_mask) {
+  std::string ss;
+  int n = 0;
+  if (channel_mask & SPEAKER_FRONT_LEFT) {
+    ss += "FRONT_LEFT | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_FRONT_RIGHT) {
+    ss += "FRONT_RIGHT | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_FRONT_CENTER) {
+    ss += "FRONT_CENTER | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_LOW_FREQUENCY) {
+    ss += "LOW_FREQUENCY | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_BACK_LEFT) {
+    ss += "BACK_LEFT | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_BACK_RIGHT) {
+    ss += "BACK_RIGHT | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_FRONT_LEFT_OF_CENTER) {
+    ss += "FRONT_LEFT_OF_CENTER | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_FRONT_RIGHT_OF_CENTER) {
+    ss += "RIGHT_OF_CENTER | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_BACK_CENTER) {
+    ss += "BACK_CENTER | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_SIDE_LEFT) {
+    ss += "SIDE_LEFT | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_SIDE_RIGHT) {
+    ss += "SIDE_RIGHT | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_TOP_CENTER) {
+    ss += "TOP_CENTER | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_TOP_FRONT_LEFT) {
+    ss += "TOP_FRONT_LEFT | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_TOP_FRONT_CENTER) {
+    ss += "TOP_FRONT_CENTER | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_TOP_FRONT_RIGHT) {
+    ss += "TOP_FRONT_RIGHT | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_TOP_BACK_LEFT) {
+    ss += "TOP_BACK_LEFT | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_TOP_BACK_CENTER) {
+    ss += "TOP_BACK_CENTER | ";
+    ++n;
+  }
+  if (channel_mask & SPEAKER_TOP_BACK_RIGHT) {
+    ss += "TOP_BACK_RIGHT | ";
+    ++n;
+  }
+
+  if (!ss.empty()) {
+    // Delete last appended " | " substring.
+    ss.erase(ss.end() - 3, ss.end());
+  }
+  ss += " (";
+  ss += std::to_string(n);
+  ss += ")";
+  return ss;
+}
+
+#if !defined(KSAUDIO_SPEAKER_1POINT1)
+// These values are only defined in ksmedia.h after a certain version, to build
+// cleanly for older windows versions this just defines the ones that are
+// missing.
+#define KSAUDIO_SPEAKER_1POINT1 (SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY)
+#define KSAUDIO_SPEAKER_2POINT1 \
+  (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY)
+#define KSAUDIO_SPEAKER_3POINT0 \
+  (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER)
+#define KSAUDIO_SPEAKER_3POINT1                                      \
+  (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
+   SPEAKER_LOW_FREQUENCY)
+#define KSAUDIO_SPEAKER_5POINT0                                      \
+  (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
+   SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
+#define KSAUDIO_SPEAKER_7POINT0                                      \
+  (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
+   SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT |      \
+   SPEAKER_SIDE_RIGHT)
+#endif
+
+#if !defined(AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY)
+#define AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY 0x08000000
+#define AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM 0x80000000
+#endif
+
+// Converts the most common format tags defined in mmreg.h into string
+// equivalents. Mainly intended for log messages.
+const char* WaveFormatTagToString(WORD format_tag) {
+  switch (format_tag) {
+    case WAVE_FORMAT_UNKNOWN:
+      return "WAVE_FORMAT_UNKNOWN";
+    case WAVE_FORMAT_PCM:
+      return "WAVE_FORMAT_PCM";
+    case WAVE_FORMAT_IEEE_FLOAT:
+      return "WAVE_FORMAT_IEEE_FLOAT";
+    case WAVE_FORMAT_EXTENSIBLE:
+      return "WAVE_FORMAT_EXTENSIBLE";
+    default:
+      return "UNKNOWN";
+  }
+}
+
+const char* RoleToString(const ERole role) {
+  switch (role) {
+    case eConsole:
+      return "Console";
+    case eMultimedia:
+      return "Multimedia";
+    case eCommunications:
+      return "Communications";
+    default:
+      return "Unsupported";
+  }
+}
+
+const char* FlowToString(const EDataFlow flow) {
+  switch (flow) {
+    case eRender:
+      return "Render";
+    case eCapture:
+      return "Capture";
+    case eAll:
+      return "Render or Capture";
+    default:
+      return "Unsupported";
+  }
+}
+
+bool LoadAudiosesDll() {
+  static const wchar_t* const kAudiosesDLL =
+      L"%WINDIR%\\system32\\audioses.dll";
+  wchar_t path[MAX_PATH] = {0};
+  ExpandEnvironmentStringsW(kAudiosesDLL, path, arraysize(path));
+  RTC_DLOG(LS_INFO) << rtc::ToUtf8(path);
+  return (LoadLibraryExW(path, nullptr, LOAD_WITH_ALTERED_SEARCH_PATH) !=
+          nullptr);
+}
+
+bool LoadAvrtDll() {
+  static const wchar_t* const kAvrtDLL = L"%WINDIR%\\system32\\Avrt.dll";
+  wchar_t path[MAX_PATH] = {0};
+  ExpandEnvironmentStringsW(kAvrtDLL, path, arraysize(path));
+  RTC_DLOG(LS_INFO) << rtc::ToUtf8(path);
+  return (LoadLibraryExW(path, nullptr, LOAD_WITH_ALTERED_SEARCH_PATH) !=
+          nullptr);
+}
+
+ComPtr<IMMDeviceEnumerator> CreateDeviceEnumeratorInternal(
+    bool allow_reinitialize) {
+  ComPtr<IMMDeviceEnumerator> device_enumerator;
+  _com_error error =
+      ::CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr, CLSCTX_ALL,
+                         IID_PPV_ARGS(&device_enumerator));
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "CoCreateInstance failed: " << ErrorToString(error);
+  }
+
+  if (error.Error() == CO_E_NOTINITIALIZED && allow_reinitialize) {
+    RTC_LOG(LS_ERROR) << "CoCreateInstance failed with CO_E_NOTINITIALIZED";
+    // We have seen crashes which indicates that this method can in fact
+    // fail with CO_E_NOTINITIALIZED in combination with certain 3rd party
+    // modules. Calling CoInitializeEx() is an attempt to resolve the reported
+    // issues. See http://crbug.com/378465 for details.
+    error = CoInitializeEx(nullptr, COINIT_MULTITHREADED);
+    if (FAILED(error.Error())) {
+      error = ::CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr,
+                                 CLSCTX_ALL, IID_PPV_ARGS(&device_enumerator));
+      if (FAILED(error.Error())) {
+        RTC_LOG(LS_ERROR) << "CoCreateInstance failed: "
+                          << ErrorToString(error);
+      }
+    }
+  }
+  return device_enumerator;
+}
+
+bool IsSupportedInternal() {
+  // The Core Audio APIs are implemented in the user-mode system components
+  // Audioses.dll and Mmdevapi.dll. Dependency Walker shows that it is
+  // enough to verify possibility to load the Audioses DLL since it depends
+  // on Mmdevapi.dll. See http://crbug.com/166397 why this extra step is
+  // required to guarantee Core Audio support.
+  if (!LoadAudiosesDll())
+    return false;
+
+  // Being able to load the Audioses.dll does not seem to be sufficient for
+  // all devices to guarantee Core Audio support. To be 100%, we also verify
+  // that it is possible to a create the IMMDeviceEnumerator interface. If
+  // this works as well we should be home free.
+  ComPtr<IMMDeviceEnumerator> device_enumerator =
+      CreateDeviceEnumeratorInternal(false);
+  if (!device_enumerator) {
+    RTC_LOG(LS_ERROR)
+        << "Failed to create Core Audio device enumerator on thread with ID "
+        << rtc::CurrentThreadId();
+    return false;
+  }
+
+  return true;
+}
+
+bool IsDeviceActive(IMMDevice* device) {
+  DWORD state = DEVICE_STATE_DISABLED;
+  return SUCCEEDED(device->GetState(&state)) && (state & DEVICE_STATE_ACTIVE);
+}
+
+// Retrieve an audio device specified by `device_id` or a default device
+// specified by data-flow direction and role if `device_id` is default.
+ComPtr<IMMDevice> CreateDeviceInternal(absl::string_view device_id,
+                                       EDataFlow data_flow,
+                                       ERole role) {
+  RTC_DLOG(LS_INFO) << "CreateDeviceInternal: "
+                       "id="
+                    << device_id << ", flow=" << FlowToString(data_flow)
+                    << ", role=" << RoleToString(role);
+  ComPtr<IMMDevice> audio_endpoint_device;
+
+  // Create the IMMDeviceEnumerator interface.
+  ComPtr<IMMDeviceEnumerator> device_enum(CreateDeviceEnumeratorInternal(true));
+  if (!device_enum.Get())
+    return audio_endpoint_device;
+
+  _com_error error(S_FALSE);
+  if (device_id == AudioDeviceName::kDefaultDeviceId) {
+    // Get the default audio endpoint for the specified data-flow direction and
+    // role. Note that, if only a single rendering or capture device is
+    // available, the system always assigns all three rendering or capture roles
+    // to that device. If the method fails to find a rendering or capture device
+    // for the specified role, this means that no rendering or capture device is
+    // available at all. If no device is available, the method sets the output
+    // pointer to NULL and returns ERROR_NOT_FOUND.
+    error = device_enum->GetDefaultAudioEndpoint(
+        data_flow, role, audio_endpoint_device.GetAddressOf());
+    if (FAILED(error.Error())) {
+      RTC_LOG(LS_ERROR)
+          << "IMMDeviceEnumerator::GetDefaultAudioEndpoint failed: "
+          << ErrorToString(error);
+    }
+  } else {
+    // Ask for an audio endpoint device that is identified by an endpoint ID
+    // string.
+    error = device_enum->GetDevice(rtc::ToUtf16(device_id).c_str(),
+                                   audio_endpoint_device.GetAddressOf());
+    if (FAILED(error.Error())) {
+      RTC_LOG(LS_ERROR) << "IMMDeviceEnumerator::GetDevice failed: "
+                        << ErrorToString(error);
+    }
+  }
+
+  // Verify that the audio endpoint device is active, i.e., that the audio
+  // adapter that connects to the endpoint device is present and enabled.
+  if (SUCCEEDED(error.Error()) && audio_endpoint_device.Get() &&
+      !IsDeviceActive(audio_endpoint_device.Get())) {
+    RTC_LOG(LS_WARNING) << "Selected endpoint device is not active";
+    audio_endpoint_device.Reset();
+  }
+
+  return audio_endpoint_device;
+}
+
+std::string GetDeviceIdInternal(IMMDevice* device) {
+  // Retrieve unique name of endpoint device.
+  // Example: "{0.0.1.00000000}.{8db6020f-18e3-4f25-b6f5-7726c9122574}".
+  LPWSTR device_id;
+  if (SUCCEEDED(device->GetId(&device_id))) {
+    std::string device_id_utf8 = rtc::ToUtf8(device_id, wcslen(device_id));
+    CoTaskMemFree(device_id);
+    return device_id_utf8;
+  } else {
+    return std::string();
+  }
+}
+
+std::string GetDeviceFriendlyNameInternal(IMMDevice* device) {
+  // Retrieve user-friendly name of endpoint device.
+  // Example: "Microphone (Realtek High Definition Audio)".
+  ComPtr<IPropertyStore> properties;
+  HRESULT hr = device->OpenPropertyStore(STGM_READ, properties.GetAddressOf());
+  if (FAILED(hr))
+    return std::string();
+
+  ScopedPropVariant friendly_name_pv;
+  hr = properties->GetValue(PKEY_Device_FriendlyName,
+                            friendly_name_pv.Receive());
+  if (FAILED(hr))
+    return std::string();
+
+  if (friendly_name_pv.get().vt == VT_LPWSTR &&
+      friendly_name_pv.get().pwszVal) {
+    return rtc::ToUtf8(friendly_name_pv.get().pwszVal,
+                       wcslen(friendly_name_pv.get().pwszVal));
+  } else {
+    return std::string();
+  }
+}
+
+ComPtr<IAudioSessionManager2> CreateSessionManager2Internal(
+    IMMDevice* audio_device) {
+  if (!audio_device)
+    return ComPtr<IAudioSessionManager2>();
+
+  ComPtr<IAudioSessionManager2> audio_session_manager;
+  _com_error error =
+      audio_device->Activate(__uuidof(IAudioSessionManager2), CLSCTX_ALL,
+                             nullptr, &audio_session_manager);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IMMDevice::Activate(IAudioSessionManager2) failed: "
+                      << ErrorToString(error);
+  }
+  return audio_session_manager;
+}
+
+ComPtr<IAudioSessionEnumerator> CreateSessionEnumeratorInternal(
+    IMMDevice* audio_device) {
+  if (!audio_device) {
+    return ComPtr<IAudioSessionEnumerator>();
+  }
+
+  ComPtr<IAudioSessionEnumerator> audio_session_enumerator;
+  ComPtr<IAudioSessionManager2> audio_session_manager =
+      CreateSessionManager2Internal(audio_device);
+  if (!audio_session_manager.Get()) {
+    return audio_session_enumerator;
+  }
+  _com_error error =
+      audio_session_manager->GetSessionEnumerator(&audio_session_enumerator);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR)
+        << "IAudioSessionEnumerator::IAudioSessionEnumerator failed: "
+        << ErrorToString(error);
+    return ComPtr<IAudioSessionEnumerator>();
+  }
+  return audio_session_enumerator;
+}
+
+// Creates and activates an IAudioClient COM object given the selected
+// endpoint device.
+ComPtr<IAudioClient> CreateClientInternal(IMMDevice* audio_device) {
+  if (!audio_device)
+    return ComPtr<IAudioClient>();
+
+  ComPtr<IAudioClient> audio_client;
+  _com_error error = audio_device->Activate(__uuidof(IAudioClient), CLSCTX_ALL,
+                                            nullptr, &audio_client);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IMMDevice::Activate(IAudioClient) failed: "
+                      << ErrorToString(error);
+  }
+  return audio_client;
+}
+
+ComPtr<IAudioClient2> CreateClient2Internal(IMMDevice* audio_device) {
+  if (!audio_device)
+    return ComPtr<IAudioClient2>();
+
+  ComPtr<IAudioClient2> audio_client;
+  _com_error error = audio_device->Activate(__uuidof(IAudioClient2), CLSCTX_ALL,
+                                            nullptr, &audio_client);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IMMDevice::Activate(IAudioClient2) failed: "
+                      << ErrorToString(error);
+  }
+  return audio_client;
+}
+
+ComPtr<IAudioClient3> CreateClient3Internal(IMMDevice* audio_device) {
+  if (!audio_device)
+    return ComPtr<IAudioClient3>();
+
+  ComPtr<IAudioClient3> audio_client;
+  _com_error error = audio_device->Activate(__uuidof(IAudioClient3), CLSCTX_ALL,
+                                            nullptr, &audio_client);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IMMDevice::Activate(IAudioClient3) failed: "
+                      << ErrorToString(error);
+  }
+  return audio_client;
+}
+
+ComPtr<IMMDeviceCollection> CreateCollectionInternal(EDataFlow data_flow) {
+  ComPtr<IMMDeviceEnumerator> device_enumerator(
+      CreateDeviceEnumeratorInternal(true));
+  if (!device_enumerator) {
+    return ComPtr<IMMDeviceCollection>();
+  }
+
+  // Generate a collection of active (present and not disabled) audio endpoint
+  // devices for the specified data-flow direction.
+  // This method will succeed even if all devices are disabled.
+  ComPtr<IMMDeviceCollection> collection;
+  _com_error error = device_enumerator->EnumAudioEndpoints(
+      data_flow, DEVICE_STATE_ACTIVE, collection.GetAddressOf());
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IMMDeviceCollection::EnumAudioEndpoints failed: "
+                      << ErrorToString(error);
+  }
+  return collection;
+}
+
+bool GetDeviceNamesInternal(EDataFlow data_flow,
+                            webrtc::AudioDeviceNames* device_names) {
+  RTC_DLOG(LS_INFO) << "GetDeviceNamesInternal: flow="
+                    << FlowToString(data_flow);
+
+  // Generate a collection of active audio endpoint devices for the specified
+  // direction.
+  ComPtr<IMMDeviceCollection> collection = CreateCollectionInternal(data_flow);
+  if (!collection.Get()) {
+    RTC_LOG(LS_ERROR) << "Failed to create a collection of active devices";
+    return false;
+  }
+
+  // Retrieve the number of active (present, not disabled and plugged in) audio
+  // devices for the specified direction.
+  UINT number_of_active_devices = 0;
+  _com_error error = collection->GetCount(&number_of_active_devices);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IMMDeviceCollection::GetCount failed: "
+                      << ErrorToString(error);
+    return false;
+  }
+
+  if (number_of_active_devices == 0) {
+    RTC_DLOG(LS_WARNING) << "Found no active devices";
+    return false;
+  }
+
+  // Loop over all active devices and add friendly name and unique id to the
+  // `device_names` queue. For now, devices are added at indexes 0, 1, ..., N-1
+  // but they will be moved to 2,3,..., N+1 at the next stage when default and
+  // default communication devices are added at index 0 and 1.
+  ComPtr<IMMDevice> audio_device;
+  for (UINT i = 0; i < number_of_active_devices; ++i) {
+    // Retrieve a pointer to the specified item in the device collection.
+    error = collection->Item(i, audio_device.GetAddressOf());
+    if (FAILED(error.Error())) {
+      // Skip this item and try to get the next item instead; will result in an
+      // incomplete list of devices.
+      RTC_LOG(LS_WARNING) << "IMMDeviceCollection::Item failed: "
+                          << ErrorToString(error);
+      continue;
+    }
+    if (!audio_device.Get()) {
+      RTC_LOG(LS_WARNING) << "Invalid audio device";
+      continue;
+    }
+
+    // Retrieve the complete device name for the given audio device endpoint.
+    AudioDeviceName device_name(
+        GetDeviceFriendlyNameInternal(audio_device.Get()),
+        GetDeviceIdInternal(audio_device.Get()));
+    // Add combination of user-friendly and unique name to the output list.
+    device_names->push_back(device_name);
+  }
+
+  // Log a warning of the list of device is not complete but let's keep on
+  // trying to add default and default communications device at the front.
+  if (device_names->size() != number_of_active_devices) {
+    RTC_DLOG(LS_WARNING)
+        << "List of device names does not contain all active devices";
+  }
+
+  // Avoid adding default and default communication devices if no active device
+  // could be added to the queue. We might as well break here and return false
+  // since no active devices were identified.
+  if (device_names->empty()) {
+    RTC_DLOG(LS_ERROR) << "List of active devices is empty";
+    return false;
+  }
+
+  // Prepend the queue with two more elements: one for the default device and
+  // one for the default communication device (can correspond to the same unique
+  // id if only one active device exists). The first element (index 0) is the
+  // default device and the second element (index 1) is the default
+  // communication device.
+  ERole role[] = {eCommunications, eConsole};
+  ComPtr<IMMDevice> default_device;
+  AudioDeviceName default_device_name;
+  for (size_t i = 0; i < arraysize(role); ++i) {
+    default_device = CreateDeviceInternal(AudioDeviceName::kDefaultDeviceId,
+                                          data_flow, role[i]);
+    if (!default_device.Get()) {
+      // Add empty strings to device name if the device could not be created.
+      RTC_DLOG(LS_WARNING) << "Failed to add device with role: "
+                           << RoleToString(role[i]);
+      default_device_name.device_name = std::string();
+      default_device_name.unique_id = std::string();
+    } else {
+      // Populate the device name with friendly name and unique id.
+      std::string device_name;
+      device_name += (role[i] == eConsole ? "Default - " : "Communication - ");
+      device_name += GetDeviceFriendlyNameInternal(default_device.Get());
+      std::string unique_id = GetDeviceIdInternal(default_device.Get());
+      default_device_name.device_name = std::move(device_name);
+      default_device_name.unique_id = std::move(unique_id);
+    }
+
+    // Add combination of user-friendly and unique name to the output queue.
+    // The last element (<=> eConsole) will be at the front of the queue, hence
+    // at index 0. Empty strings will be added for cases where no default
+    // devices were found.
+    device_names->push_front(default_device_name);
+  }
+
+  // Example of log output when only one device is active. Note that the queue
+  // contains two extra elements at index 0 (Default) and 1 (Communication) to
+  // allow selection of device by role instead of id. All elements corresponds
+  // the same unique id.
+  // [0] friendly name: Default - Headset Microphone (2- Arctis 7 Chat)
+  // [0] unique id    : {0.0.1.00000000}.{ff9eed76-196e-467a-b295-26986e69451c}
+  // [1] friendly name: Communication - Headset Microphone (2- Arctis 7 Chat)
+  // [1] unique id    : {0.0.1.00000000}.{ff9eed76-196e-467a-b295-26986e69451c}
+  // [2] friendly name: Headset Microphone (2- Arctis 7 Chat)
+  // [2] unique id    : {0.0.1.00000000}.{ff9eed76-196e-467a-b295-26986e69451c}
+  for (size_t i = 0; i < device_names->size(); ++i) {
+    RTC_DLOG(LS_INFO) << "[" << i
+                      << "] friendly name: " << (*device_names)[i].device_name;
+    RTC_DLOG(LS_INFO) << "[" << i
+                      << "] unique id    : " << (*device_names)[i].unique_id;
+  }
+
+  return true;
+}
+
+HRESULT GetPreferredAudioParametersInternal(IAudioClient* client,
+                                            AudioParameters* params,
+                                            int fixed_sample_rate) {
+  WAVEFORMATPCMEX mix_format;
+  HRESULT hr = core_audio_utility::GetSharedModeMixFormat(client, &mix_format);
+  if (FAILED(hr))
+    return hr;
+
+  REFERENCE_TIME default_period = 0;
+  hr = core_audio_utility::GetDevicePeriod(client, AUDCLNT_SHAREMODE_SHARED,
+                                           &default_period);
+  if (FAILED(hr))
+    return hr;
+
+  int sample_rate = mix_format.Format.nSamplesPerSec;
+  // Override default sample rate if `fixed_sample_rate` is set and different
+  // from the default rate.
+  if (fixed_sample_rate > 0 && fixed_sample_rate != sample_rate) {
+    RTC_DLOG(LS_INFO) << "Using fixed sample rate instead of the preferred: "
+                      << sample_rate << " is replaced by " << fixed_sample_rate;
+    sample_rate = fixed_sample_rate;
+  }
+  // TODO(henrika): utilize full mix_format.Format.wBitsPerSample.
+  // const size_t bits_per_sample = AudioParameters::kBitsPerSample;
+  // TODO(henrika): improve channel layout support.
+  const size_t channels = mix_format.Format.nChannels;
+
+  // Use the native device period to derive the smallest possible buffer size
+  // in shared mode.
+  double device_period_in_seconds =
+      static_cast<double>(
+          core_audio_utility::ReferenceTimeToTimeDelta(default_period).ms()) /
+      1000.0L;
+  const size_t frames_per_buffer =
+      static_cast<size_t>(sample_rate * device_period_in_seconds + 0.5);
+
+  AudioParameters audio_params(sample_rate, channels, frames_per_buffer);
+  *params = audio_params;
+  RTC_DLOG(LS_INFO) << audio_params.ToString();
+
+  return hr;
+}
+
+}  // namespace
+
+namespace core_audio_utility {
+
+// core_audio_utility::WaveFormatWrapper implementation.
+WAVEFORMATEXTENSIBLE* WaveFormatWrapper::GetExtensible() const {
+  RTC_CHECK(IsExtensible());
+  return reinterpret_cast<WAVEFORMATEXTENSIBLE*>(ptr_);
+}
+
+bool WaveFormatWrapper::IsExtensible() const {
+  return ptr_->wFormatTag == WAVE_FORMAT_EXTENSIBLE && ptr_->cbSize >= 22;
+}
+
+bool WaveFormatWrapper::IsPcm() const {
+  return IsExtensible() ? GetExtensible()->SubFormat == KSDATAFORMAT_SUBTYPE_PCM
+                        : ptr_->wFormatTag == WAVE_FORMAT_PCM;
+}
+
+bool WaveFormatWrapper::IsFloat() const {
+  return IsExtensible()
+             ? GetExtensible()->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT
+             : ptr_->wFormatTag == WAVE_FORMAT_IEEE_FLOAT;
+}
+
+size_t WaveFormatWrapper::size() const {
+  return sizeof(*ptr_) + ptr_->cbSize;
+}
+
+bool IsSupported() {
+  RTC_DLOG(LS_INFO) << "IsSupported";
+  static bool g_is_supported = IsSupportedInternal();
+  return g_is_supported;
+}
+
+bool IsMMCSSSupported() {
+  RTC_DLOG(LS_INFO) << "IsMMCSSSupported";
+  return LoadAvrtDll();
+}
+
+int NumberOfActiveDevices(EDataFlow data_flow) {
+  // Generate a collection of active audio endpoint devices for the specified
+  // data-flow direction.
+  ComPtr<IMMDeviceCollection> collection = CreateCollectionInternal(data_flow);
+  if (!collection.Get()) {
+    return 0;
+  }
+
+  // Retrieve the number of active audio devices for the specified direction.
+  UINT number_of_active_devices = 0;
+  collection->GetCount(&number_of_active_devices);
+  std::string str;
+  if (data_flow == eCapture) {
+    str = "Number of capture devices: ";
+  } else if (data_flow == eRender) {
+    str = "Number of render devices: ";
+  } else if (data_flow == eAll) {
+    str = "Total number of devices: ";
+  }
+  RTC_DLOG(LS_INFO) << str << number_of_active_devices;
+  return static_cast<int>(number_of_active_devices);
+}
+
+uint32_t GetAudioClientVersion() {
+  uint32_t version = 1;
+  if (rtc::rtc_win::GetVersion() >= rtc::rtc_win::VERSION_WIN10) {
+    version = 3;
+  } else if (rtc::rtc_win::GetVersion() >= rtc::rtc_win::VERSION_WIN8) {
+    version = 2;
+  }
+  return version;
+}
+
+ComPtr<IMMDeviceEnumerator> CreateDeviceEnumerator() {
+  RTC_DLOG(LS_INFO) << "CreateDeviceEnumerator";
+  return CreateDeviceEnumeratorInternal(true);
+}
+
+std::string GetDefaultInputDeviceID() {
+  RTC_DLOG(LS_INFO) << "GetDefaultInputDeviceID";
+  ComPtr<IMMDevice> device(
+      CreateDevice(AudioDeviceName::kDefaultDeviceId, eCapture, eConsole));
+  return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
+}
+
+std::string GetDefaultOutputDeviceID() {
+  RTC_DLOG(LS_INFO) << "GetDefaultOutputDeviceID";
+  ComPtr<IMMDevice> device(
+      CreateDevice(AudioDeviceName::kDefaultDeviceId, eRender, eConsole));
+  return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
+}
+
+std::string GetCommunicationsInputDeviceID() {
+  RTC_DLOG(LS_INFO) << "GetCommunicationsInputDeviceID";
+  ComPtr<IMMDevice> device(CreateDevice(AudioDeviceName::kDefaultDeviceId,
+                                        eCapture, eCommunications));
+  return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
+}
+
+std::string GetCommunicationsOutputDeviceID() {
+  RTC_DLOG(LS_INFO) << "GetCommunicationsOutputDeviceID";
+  ComPtr<IMMDevice> device(CreateDevice(AudioDeviceName::kDefaultDeviceId,
+                                        eRender, eCommunications));
+  return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
+}
+
+ComPtr<IMMDevice> CreateDevice(absl::string_view device_id,
+                               EDataFlow data_flow,
+                               ERole role) {
+  RTC_DLOG(LS_INFO) << "CreateDevice";
+  return CreateDeviceInternal(device_id, data_flow, role);
+}
+
+AudioDeviceName GetDeviceName(IMMDevice* device) {
+  RTC_DLOG(LS_INFO) << "GetDeviceName";
+  RTC_DCHECK(device);
+  AudioDeviceName device_name(GetDeviceFriendlyNameInternal(device),
+                              GetDeviceIdInternal(device));
+  RTC_DLOG(LS_INFO) << "friendly name: " << device_name.device_name;
+  RTC_DLOG(LS_INFO) << "unique id    : " << device_name.unique_id;
+  return device_name;
+}
+
+std::string GetFriendlyName(absl::string_view device_id,
+                            EDataFlow data_flow,
+                            ERole role) {
+  RTC_DLOG(LS_INFO) << "GetFriendlyName";
+  ComPtr<IMMDevice> audio_device = CreateDevice(device_id, data_flow, role);
+  if (!audio_device.Get())
+    return std::string();
+
+  AudioDeviceName device_name = GetDeviceName(audio_device.Get());
+  return device_name.device_name;
+}
+
+EDataFlow GetDataFlow(IMMDevice* device) {
+  RTC_DLOG(LS_INFO) << "GetDataFlow";
+  RTC_DCHECK(device);
+  ComPtr<IMMEndpoint> endpoint;
+  _com_error error = device->QueryInterface(endpoint.GetAddressOf());
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IMMDevice::QueryInterface failed: "
+                      << ErrorToString(error);
+    return eAll;
+  }
+
+  EDataFlow data_flow;
+  error = endpoint->GetDataFlow(&data_flow);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IMMEndpoint::GetDataFlow failed: "
+                      << ErrorToString(error);
+    return eAll;
+  }
+  return data_flow;
+}
+
+bool GetInputDeviceNames(webrtc::AudioDeviceNames* device_names) {
+  RTC_DLOG(LS_INFO) << "GetInputDeviceNames";
+  RTC_DCHECK(device_names);
+  RTC_DCHECK(device_names->empty());
+  return GetDeviceNamesInternal(eCapture, device_names);
+}
+
+bool GetOutputDeviceNames(webrtc::AudioDeviceNames* device_names) {
+  RTC_DLOG(LS_INFO) << "GetOutputDeviceNames";
+  RTC_DCHECK(device_names);
+  RTC_DCHECK(device_names->empty());
+  return GetDeviceNamesInternal(eRender, device_names);
+}
+
+ComPtr<IAudioSessionManager2> CreateSessionManager2(IMMDevice* device) {
+  RTC_DLOG(LS_INFO) << "CreateSessionManager2";
+  return CreateSessionManager2Internal(device);
+}
+
+Microsoft::WRL::ComPtr<IAudioSessionEnumerator> CreateSessionEnumerator(
+    IMMDevice* device) {
+  RTC_DLOG(LS_INFO) << "CreateSessionEnumerator";
+  return CreateSessionEnumeratorInternal(device);
+}
+
+int NumberOfActiveSessions(IMMDevice* device) {
+  RTC_DLOG(LS_INFO) << "NumberOfActiveSessions";
+  ComPtr<IAudioSessionEnumerator> session_enumerator =
+      CreateSessionEnumerator(device);
+
+  // Iterate over all audio sessions for the given device.
+  int session_count = 0;
+  _com_error error = session_enumerator->GetCount(&session_count);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioSessionEnumerator::GetCount failed: "
+                      << ErrorToString(error);
+    return 0;
+  }
+  RTC_DLOG(LS_INFO) << "Total number of audio sessions: " << session_count;
+
+  int num_active = 0;
+  for (int session = 0; session < session_count; session++) {
+    // Acquire the session control interface.
+    ComPtr<IAudioSessionControl> session_control;
+    error = session_enumerator->GetSession(session, &session_control);
+    if (FAILED(error.Error())) {
+      RTC_LOG(LS_ERROR) << "IAudioSessionEnumerator::GetSession failed: "
+                        << ErrorToString(error);
+      return 0;
+    }
+
+    // Log the display name of the audio session for debugging purposes.
+    LPWSTR display_name;
+    if (SUCCEEDED(session_control->GetDisplayName(&display_name))) {
+      RTC_DLOG(LS_INFO) << "display name: "
+                        << rtc::ToUtf8(display_name, wcslen(display_name));
+      CoTaskMemFree(display_name);
+    }
+
+    // Get the current state and check if the state is active or not.
+    AudioSessionState state;
+    error = session_control->GetState(&state);
+    if (FAILED(error.Error())) {
+      RTC_LOG(LS_ERROR) << "IAudioSessionControl::GetState failed: "
+                        << ErrorToString(error);
+      return 0;
+    }
+    if (state == AudioSessionStateActive) {
+      ++num_active;
+    }
+  }
+
+  RTC_DLOG(LS_INFO) << "Number of active audio sessions: " << num_active;
+  return num_active;
+}
+
+ComPtr<IAudioClient> CreateClient(absl::string_view device_id,
+                                  EDataFlow data_flow,
+                                  ERole role) {
+  RTC_DLOG(LS_INFO) << "CreateClient";
+  ComPtr<IMMDevice> device(CreateDevice(device_id, data_flow, role));
+  return CreateClientInternal(device.Get());
+}
+
+ComPtr<IAudioClient2> CreateClient2(absl::string_view device_id,
+                                    EDataFlow data_flow,
+                                    ERole role) {
+  RTC_DLOG(LS_INFO) << "CreateClient2";
+  ComPtr<IMMDevice> device(CreateDevice(device_id, data_flow, role));
+  return CreateClient2Internal(device.Get());
+}
+
+ComPtr<IAudioClient3> CreateClient3(absl::string_view device_id,
+                                    EDataFlow data_flow,
+                                    ERole role) {
+  RTC_DLOG(LS_INFO) << "CreateClient3";
+  ComPtr<IMMDevice> device(CreateDevice(device_id, data_flow, role));
+  return CreateClient3Internal(device.Get());
+}
+
+HRESULT SetClientProperties(IAudioClient2* client) {
+  RTC_DLOG(LS_INFO) << "SetClientProperties";
+  RTC_DCHECK(client);
+  if (GetAudioClientVersion() < 2) {
+    RTC_LOG(LS_WARNING) << "Requires IAudioClient2 or higher";
+    return AUDCLNT_E_UNSUPPORTED_FORMAT;
+  }
+  AudioClientProperties props = {0};
+  props.cbSize = sizeof(AudioClientProperties);
+  // Real-time VoIP communication.
+  // TODO(henrika): other categories?
+  props.eCategory = AudioCategory_Communications;
+  // Hardware-offloaded audio processing allows the main audio processing tasks
+  // to be performed outside the computer's main CPU. Check support and log the
+  // result but hard-code `bIsOffload` to FALSE for now.
+  // TODO(henrika): evaluate hardware-offloading. Might complicate usage of
+  // IAudioClient::GetMixFormat().
+  BOOL supports_offload = FALSE;
+  _com_error error =
+      client->IsOffloadCapable(props.eCategory, &supports_offload);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient2::IsOffloadCapable failed: "
+                      << ErrorToString(error);
+  }
+  RTC_DLOG(LS_INFO) << "supports_offload: " << supports_offload;
+  props.bIsOffload = false;
+#if (NTDDI_VERSION < NTDDI_WINBLUE)
+  RTC_DLOG(LS_INFO) << "options: Not supported in this build";
+#else
+  // TODO(henrika): pros and cons compared with AUDCLNT_STREAMOPTIONS_NONE?
+  props.Options |= AUDCLNT_STREAMOPTIONS_NONE;
+  // Requires System.Devices.AudioDevice.RawProcessingSupported.
+  // The application can choose to *always ignore* the OEM AEC/AGC by setting
+  // the AUDCLNT_STREAMOPTIONS_RAW flag in the call to SetClientProperties.
+  // This flag will preserve the user experience aspect of Communications
+  // streams, but will not insert any OEM provided communications specific
+  // processing in the audio signal path.
+  // props.Options |= AUDCLNT_STREAMOPTIONS_RAW;
+
+  // If it is important to avoid resampling in the audio engine, set this flag.
+  // AUDCLNT_STREAMOPTIONS_MATCH_FORMAT (or anything in IAudioClient3) is not
+  // an appropriate interface to use for communications scenarios.
+  // This interface is mainly meant for pro audio scenarios.
+  // props.Options |= AUDCLNT_STREAMOPTIONS_MATCH_FORMAT;
+  RTC_DLOG(LS_INFO) << "options: 0x" << rtc::ToHex(props.Options);
+#endif
+  error = client->SetClientProperties(&props);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient2::SetClientProperties failed: "
+                      << ErrorToString(error);
+  }
+  return error.Error();
+}
+
+HRESULT GetBufferSizeLimits(IAudioClient2* client,
+                            const WAVEFORMATEXTENSIBLE* format,
+                            REFERENCE_TIME* min_buffer_duration,
+                            REFERENCE_TIME* max_buffer_duration) {
+  RTC_DLOG(LS_INFO) << "GetBufferSizeLimits";
+  RTC_DCHECK(client);
+  if (GetAudioClientVersion() < 2) {
+    RTC_LOG(LS_WARNING) << "Requires IAudioClient2 or higher";
+    return AUDCLNT_E_UNSUPPORTED_FORMAT;
+  }
+  REFERENCE_TIME min_duration = 0;
+  REFERENCE_TIME max_duration = 0;
+  _com_error error =
+      client->GetBufferSizeLimits(reinterpret_cast<const WAVEFORMATEX*>(format),
+                                  TRUE, &min_duration, &max_duration);
+  if (error.Error() == AUDCLNT_E_OFFLOAD_MODE_ONLY) {
+    // This API seems to be supported in off-load mode only but it is not
+    // documented as a valid error code. Making a special note about it here.
+    RTC_LOG(LS_ERROR) << "IAudioClient2::GetBufferSizeLimits failed: "
+                         "AUDCLNT_E_OFFLOAD_MODE_ONLY";
+  } else if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient2::GetBufferSizeLimits failed: "
+                      << ErrorToString(error);
+  } else {
+    *min_buffer_duration = min_duration;
+    *max_buffer_duration = max_duration;
+    RTC_DLOG(LS_INFO) << "min_buffer_duration: " << min_buffer_duration;
+    RTC_DLOG(LS_INFO) << "max_buffer_duration: " << max_buffer_duration;
+  }
+  return error.Error();
+}
+
+HRESULT GetSharedModeMixFormat(IAudioClient* client,
+                               WAVEFORMATEXTENSIBLE* format) {
+  RTC_DLOG(LS_INFO) << "GetSharedModeMixFormat";
+  RTC_DCHECK(client);
+
+  // The GetMixFormat method retrieves the stream format that the audio engine
+  // uses for its internal processing of shared-mode streams. The method
+  // allocates the storage for the structure and this memory will be released
+  // when `mix_format` goes out of scope. The GetMixFormat method retrieves a
+  // format descriptor that is in the form of a WAVEFORMATEXTENSIBLE structure
+  // instead of a standalone WAVEFORMATEX structure. The method outputs a
+  // pointer to the WAVEFORMATEX structure that is embedded at the start of
+  // this WAVEFORMATEXTENSIBLE structure.
+  // Note that, crbug/803056 indicates that some devices can return a format
+  // where only the WAVEFORMATEX parts is initialized and we must be able to
+  // account for that.
+  ScopedCoMem<WAVEFORMATEXTENSIBLE> mix_format;
+  _com_error error =
+      client->GetMixFormat(reinterpret_cast<WAVEFORMATEX**>(&mix_format));
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::GetMixFormat failed: "
+                      << ErrorToString(error);
+    return error.Error();
+  }
+
+  // Use a wave format wrapper to make things simpler.
+  WaveFormatWrapper wrapped_format(mix_format.Get());
+
+  // Verify that the reported format can be mixed by the audio engine in
+  // shared mode.
+  if (!wrapped_format.IsPcm() && !wrapped_format.IsFloat()) {
+    RTC_DLOG(LS_ERROR)
+        << "Only pure PCM or float audio streams can be mixed in shared mode";
+    return AUDCLNT_E_UNSUPPORTED_FORMAT;
+  }
+
+  // Log a warning for the rare case where `mix_format` only contains a
+  // stand-alone WAVEFORMATEX structure but don't return.
+  if (!wrapped_format.IsExtensible()) {
+    RTC_DLOG(LS_WARNING)
+        << "The returned format contains no extended information. "
+           "The size is "
+        << wrapped_format.size() << " bytes.";
+  }
+
+  // Copy the correct number of bytes into |*format| taking into account if
+  // the returned structure is correctly extended or not.
+  RTC_CHECK_LE(wrapped_format.size(), sizeof(WAVEFORMATEXTENSIBLE));
+  memcpy(format, wrapped_format.get(), wrapped_format.size());
+  RTC_DLOG(LS_INFO) << WaveFormatToString(format);
+
+  return error.Error();
+}
+
+bool IsFormatSupported(IAudioClient* client,
+                       AUDCLNT_SHAREMODE share_mode,
+                       const WAVEFORMATEXTENSIBLE* format) {
+  RTC_DLOG(LS_INFO) << "IsFormatSupported";
+  RTC_DCHECK(client);
+  ScopedCoMem<WAVEFORMATEX> closest_match;
+  // This method provides a way for a client to determine, before calling
+  // IAudioClient::Initialize, whether the audio engine supports a particular
+  // stream format or not. In shared mode, the audio engine always supports
+  // the mix format (see GetSharedModeMixFormat).
+  // TODO(henrika): verify support for exclusive mode as well?
+  _com_error error = client->IsFormatSupported(
+      share_mode, reinterpret_cast<const WAVEFORMATEX*>(format),
+      &closest_match);
+  RTC_LOG(LS_INFO) << WaveFormatToString(
+      const_cast<WAVEFORMATEXTENSIBLE*>(format));
+  if ((error.Error() == S_OK) && (closest_match == nullptr)) {
+    RTC_DLOG(LS_INFO)
+        << "The audio endpoint device supports the specified stream format";
+  } else if ((error.Error() == S_FALSE) && (closest_match != nullptr)) {
+    // Call succeeded with a closest match to the specified format. This log can
+    // only be triggered for shared mode.
+    RTC_LOG(LS_WARNING)
+        << "Exact format is not supported, but a closest match exists";
+    RTC_LOG(LS_INFO) << WaveFormatToString(closest_match.Get());
+  } else if ((error.Error() == AUDCLNT_E_UNSUPPORTED_FORMAT) &&
+             (closest_match == nullptr)) {
+    // The audio engine does not support the caller-specified format or any
+    // similar format.
+    RTC_DLOG(LS_INFO) << "The audio endpoint device does not support the "
+                         "specified stream format";
+  } else {
+    RTC_LOG(LS_ERROR) << "IAudioClient::IsFormatSupported failed: "
+                      << ErrorToString(error);
+  }
+
+  return (error.Error() == S_OK);
+}
+
+HRESULT GetDevicePeriod(IAudioClient* client,
+                        AUDCLNT_SHAREMODE share_mode,
+                        REFERENCE_TIME* device_period) {
+  RTC_DLOG(LS_INFO) << "GetDevicePeriod";
+  RTC_DCHECK(client);
+  // The `default_period` parameter specifies the default scheduling period
+  // for a shared-mode stream. The `minimum_period` parameter specifies the
+  // minimum scheduling period for an exclusive-mode stream.
+  // The time is expressed in 100-nanosecond units.
+  REFERENCE_TIME default_period = 0;
+  REFERENCE_TIME minimum_period = 0;
+  _com_error error = client->GetDevicePeriod(&default_period, &minimum_period);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::GetDevicePeriod failed: "
+                      << ErrorToString(error);
+    return error.Error();
+  }
+
+  *device_period = (share_mode == AUDCLNT_SHAREMODE_SHARED) ? default_period
+                                                            : minimum_period;
+  RTC_LOG(LS_INFO) << "device_period: "
+                   << ReferenceTimeToTimeDelta(*device_period).ms() << " [ms]";
+  RTC_LOG(LS_INFO) << "minimum_period: "
+                   << ReferenceTimeToTimeDelta(minimum_period).ms() << " [ms]";
+  return error.Error();
+}
+
+HRESULT GetSharedModeEnginePeriod(IAudioClient3* client3,
+                                  const WAVEFORMATEXTENSIBLE* format,
+                                  uint32_t* default_period_in_frames,
+                                  uint32_t* fundamental_period_in_frames,
+                                  uint32_t* min_period_in_frames,
+                                  uint32_t* max_period_in_frames) {
+  RTC_DLOG(LS_INFO) << "GetSharedModeEnginePeriod";
+  RTC_DCHECK(client3);
+
+  UINT32 default_period = 0;
+  UINT32 fundamental_period = 0;
+  UINT32 min_period = 0;
+  UINT32 max_period = 0;
+  _com_error error = client3->GetSharedModeEnginePeriod(
+      reinterpret_cast<const WAVEFORMATEX*>(format), &default_period,
+      &fundamental_period, &min_period, &max_period);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient3::GetSharedModeEnginePeriod failed: "
+                      << ErrorToString(error);
+    return error.Error();
+  }
+
+  WAVEFORMATEX format_ex = format->Format;
+  const WORD sample_rate = format_ex.nSamplesPerSec;
+  RTC_LOG(LS_INFO) << "default_period_in_frames: " << default_period << " ("
+                   << FramesToMilliseconds(default_period, sample_rate)
+                   << " ms)";
+  RTC_LOG(LS_INFO) << "fundamental_period_in_frames: " << fundamental_period
+                   << " ("
+                   << FramesToMilliseconds(fundamental_period, sample_rate)
+                   << " ms)";
+  RTC_LOG(LS_INFO) << "min_period_in_frames: " << min_period << " ("
+                   << FramesToMilliseconds(min_period, sample_rate) << " ms)";
+  RTC_LOG(LS_INFO) << "max_period_in_frames: " << max_period << " ("
+                   << FramesToMilliseconds(max_period, sample_rate) << " ms)";
+  *default_period_in_frames = default_period;
+  *fundamental_period_in_frames = fundamental_period;
+  *min_period_in_frames = min_period;
+  *max_period_in_frames = max_period;
+  return error.Error();
+}
+
+HRESULT GetPreferredAudioParameters(IAudioClient* client,
+                                    AudioParameters* params) {
+  RTC_DLOG(LS_INFO) << "GetPreferredAudioParameters";
+  RTC_DCHECK(client);
+  return GetPreferredAudioParametersInternal(client, params, -1);
+}
+
+HRESULT GetPreferredAudioParameters(IAudioClient* client,
+                                    webrtc::AudioParameters* params,
+                                    uint32_t sample_rate) {
+  RTC_DLOG(LS_INFO) << "GetPreferredAudioParameters: " << sample_rate;
+  RTC_DCHECK(client);
+  return GetPreferredAudioParametersInternal(client, params, sample_rate);
+}
+
+HRESULT SharedModeInitialize(IAudioClient* client,
+                             const WAVEFORMATEXTENSIBLE* format,
+                             HANDLE event_handle,
+                             REFERENCE_TIME buffer_duration,
+                             bool auto_convert_pcm,
+                             uint32_t* endpoint_buffer_size) {
+  RTC_DLOG(LS_INFO) << "SharedModeInitialize: buffer_duration="
+                    << buffer_duration
+                    << ", auto_convert_pcm=" << auto_convert_pcm;
+  RTC_DCHECK(client);
+  RTC_DCHECK_GE(buffer_duration, 0);
+  if (buffer_duration != 0) {
+    RTC_DLOG(LS_WARNING) << "Non-default buffer size is used";
+  }
+  if (auto_convert_pcm) {
+    RTC_DLOG(LS_WARNING) << "Sample rate converter can be utilized";
+  }
+  // The AUDCLNT_STREAMFLAGS_NOPERSIST flag disables persistence of the volume
+  // and mute settings for a session that contains rendering streams.
+  // By default, the volume level and muting state for a rendering session are
+  // persistent across system restarts. The volume level and muting state for a
+  // capture session are never persistent.
+  DWORD stream_flags = AUDCLNT_STREAMFLAGS_NOPERSIST;
+
+  // Enable event-driven streaming if a valid event handle is provided.
+  // After the stream starts, the audio engine will signal the event handle
+  // to notify the client each time a buffer becomes ready to process.
+  // Event-driven buffering is supported for both rendering and capturing.
+  // Both shared-mode and exclusive-mode streams can use event-driven buffering.
+  bool use_event =
+      (event_handle != nullptr && event_handle != INVALID_HANDLE_VALUE);
+  if (use_event) {
+    stream_flags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
+    RTC_DLOG(LS_INFO) << "The stream is initialized to be event driven";
+  }
+
+  // Check if sample-rate conversion is requested.
+  if (auto_convert_pcm) {
+    // Add channel matrixer (not utilized here) and rate converter to convert
+    // from our (the client's) format to the audio engine mix format.
+    // Currently only supported for testing, i.e., not possible to enable using
+    // public APIs.
+    RTC_DLOG(LS_INFO) << "The stream is initialized to support rate conversion";
+    stream_flags |= AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM;
+    stream_flags |= AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
+  }
+  RTC_DLOG(LS_INFO) << "stream_flags: 0x" << rtc::ToHex(stream_flags);
+
+  // Initialize the shared mode client for minimal delay if `buffer_duration`
+  // is 0 or possibly a higher delay (more robust) if `buffer_duration` is
+  // larger than 0. The actual size is given by IAudioClient::GetBufferSize().
+  _com_error error = client->Initialize(
+      AUDCLNT_SHAREMODE_SHARED, stream_flags, buffer_duration, 0,
+      reinterpret_cast<const WAVEFORMATEX*>(format), nullptr);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::Initialize failed: "
+                      << ErrorToString(error);
+    return error.Error();
+  }
+
+  // If a stream is initialized to be event driven and in shared mode, the
+  // associated application must also obtain a handle by making a call to
+  // IAudioClient::SetEventHandle.
+  if (use_event) {
+    error = client->SetEventHandle(event_handle);
+    if (FAILED(error.Error())) {
+      RTC_LOG(LS_ERROR) << "IAudioClient::SetEventHandle failed: "
+                        << ErrorToString(error);
+      return error.Error();
+    }
+  }
+
+  UINT32 buffer_size_in_frames = 0;
+  // Retrieves the size (maximum capacity) of the endpoint buffer. The size is
+  // expressed as the number of audio frames the buffer can hold.
+  // For rendering clients, the buffer length determines the maximum amount of
+  // rendering data that the application can write to the endpoint buffer
+  // during a single processing pass. For capture clients, the buffer length
+  // determines the maximum amount of capture data that the audio engine can
+  // read from the endpoint buffer during a single processing pass.
+  error = client->GetBufferSize(&buffer_size_in_frames);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::GetBufferSize failed: "
+                      << ErrorToString(error);
+    return error.Error();
+  }
+
+  *endpoint_buffer_size = buffer_size_in_frames;
+  RTC_DLOG(LS_INFO) << "endpoint buffer size: " << buffer_size_in_frames
+                    << " [audio frames]";
+  const double size_in_ms = static_cast<double>(buffer_size_in_frames) /
+                            (format->Format.nSamplesPerSec / 1000.0);
+  RTC_DLOG(LS_INFO) << "endpoint buffer size: "
+                    << static_cast<int>(size_in_ms + 0.5) << " [ms]";
+  RTC_DLOG(LS_INFO) << "bytes per audio frame: " << format->Format.nBlockAlign;
+  RTC_DLOG(LS_INFO) << "endpoint buffer size: "
+                    << buffer_size_in_frames * format->Format.nChannels *
+                           (format->Format.wBitsPerSample / 8)
+                    << " [bytes]";
+
+  // TODO(henrika): utilize when delay measurements are added.
+  REFERENCE_TIME latency = 0;
+  error = client->GetStreamLatency(&latency);
+  RTC_DLOG(LS_INFO) << "stream latency: "
+                    << ReferenceTimeToTimeDelta(latency).ms() << " [ms]";
+  return error.Error();
+}
+
+HRESULT SharedModeInitializeLowLatency(IAudioClient3* client,
+                                       const WAVEFORMATEXTENSIBLE* format,
+                                       HANDLE event_handle,
+                                       uint32_t period_in_frames,
+                                       bool auto_convert_pcm,
+                                       uint32_t* endpoint_buffer_size) {
+  RTC_DLOG(LS_INFO) << "SharedModeInitializeLowLatency: period_in_frames="
+                    << period_in_frames
+                    << ", auto_convert_pcm=" << auto_convert_pcm;
+  RTC_DCHECK(client);
+  RTC_DCHECK_GT(period_in_frames, 0);
+  if (auto_convert_pcm) {
+    RTC_DLOG(LS_WARNING) << "Sample rate converter is enabled";
+  }
+
+  // Define stream flags.
+  DWORD stream_flags = AUDCLNT_STREAMFLAGS_NOPERSIST;
+  bool use_event =
+      (event_handle != nullptr && event_handle != INVALID_HANDLE_VALUE);
+  if (use_event) {
+    stream_flags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
+    RTC_DLOG(LS_INFO) << "The stream is initialized to be event driven";
+  }
+  if (auto_convert_pcm) {
+    stream_flags |= AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM;
+    stream_flags |= AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
+  }
+  RTC_DLOG(LS_INFO) << "stream_flags: 0x" << rtc::ToHex(stream_flags);
+
+  // Initialize the shared mode client for lowest possible latency.
+  // It is assumed that GetSharedModeEnginePeriod() has been used to query the
+  // smallest possible engine period and that it is given by `period_in_frames`.
+  _com_error error = client->InitializeSharedAudioStream(
+      stream_flags, period_in_frames,
+      reinterpret_cast<const WAVEFORMATEX*>(format), nullptr);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient3::InitializeSharedAudioStream failed: "
+                      << ErrorToString(error);
+    return error.Error();
+  }
+
+  // Set the event handle.
+  if (use_event) {
+    error = client->SetEventHandle(event_handle);
+    if (FAILED(error.Error())) {
+      RTC_LOG(LS_ERROR) << "IAudioClient::SetEventHandle failed: "
+                        << ErrorToString(error);
+      return error.Error();
+    }
+  }
+
+  UINT32 buffer_size_in_frames = 0;
+  // Retrieve the size (maximum capacity) of the endpoint buffer.
+  error = client->GetBufferSize(&buffer_size_in_frames);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::GetBufferSize failed: "
+                      << ErrorToString(error);
+    return error.Error();
+  }
+
+  *endpoint_buffer_size = buffer_size_in_frames;
+  RTC_DLOG(LS_INFO) << "endpoint buffer size: " << buffer_size_in_frames
+                    << " [audio frames]";
+  const double size_in_ms = static_cast<double>(buffer_size_in_frames) /
+                            (format->Format.nSamplesPerSec / 1000.0);
+  RTC_DLOG(LS_INFO) << "endpoint buffer size: "
+                    << static_cast<int>(size_in_ms + 0.5) << " [ms]";
+  RTC_DLOG(LS_INFO) << "bytes per audio frame: " << format->Format.nBlockAlign;
+  RTC_DLOG(LS_INFO) << "endpoint buffer size: "
+                    << buffer_size_in_frames * format->Format.nChannels *
+                           (format->Format.wBitsPerSample / 8)
+                    << " [bytes]";
+
+  // TODO(henrika): utilize when delay measurements are added.
+  REFERENCE_TIME latency = 0;
+  error = client->GetStreamLatency(&latency);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_WARNING) << "IAudioClient::GetStreamLatency failed: "
+                        << ErrorToString(error);
+  } else {
+    RTC_DLOG(LS_INFO) << "stream latency: "
+                      << ReferenceTimeToTimeDelta(latency).ms() << " [ms]";
+  }
+  return error.Error();
+}
+
+ComPtr<IAudioRenderClient> CreateRenderClient(IAudioClient* client) {
+  RTC_DLOG(LS_INFO) << "CreateRenderClient";
+  RTC_DCHECK(client);
+  // Get access to the IAudioRenderClient interface. This interface
+  // enables us to write output data to a rendering endpoint buffer.
+  ComPtr<IAudioRenderClient> audio_render_client;
+  _com_error error = client->GetService(IID_PPV_ARGS(&audio_render_client));
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR)
+        << "IAudioClient::GetService(IID_IAudioRenderClient) failed: "
+        << ErrorToString(error);
+    return ComPtr<IAudioRenderClient>();
+  }
+  return audio_render_client;
+}
+
+ComPtr<IAudioCaptureClient> CreateCaptureClient(IAudioClient* client) {
+  RTC_DLOG(LS_INFO) << "CreateCaptureClient";
+  RTC_DCHECK(client);
+  // Get access to the IAudioCaptureClient interface. This interface
+  // enables us to read input data from a capturing endpoint buffer.
+  ComPtr<IAudioCaptureClient> audio_capture_client;
+  _com_error error = client->GetService(IID_PPV_ARGS(&audio_capture_client));
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR)
+        << "IAudioClient::GetService(IID_IAudioCaptureClient) failed: "
+        << ErrorToString(error);
+    return ComPtr<IAudioCaptureClient>();
+  }
+  return audio_capture_client;
+}
+
+ComPtr<IAudioClock> CreateAudioClock(IAudioClient* client) {
+  RTC_DLOG(LS_INFO) << "CreateAudioClock";
+  RTC_DCHECK(client);
+  // Get access to the IAudioClock interface. This interface enables us to
+  // monitor a stream's data rate and the current position in the stream.
+  ComPtr<IAudioClock> audio_clock;
+  _com_error error = client->GetService(IID_PPV_ARGS(&audio_clock));
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::GetService(IID_IAudioClock) failed: "
+                      << ErrorToString(error);
+    return ComPtr<IAudioClock>();
+  }
+  return audio_clock;
+}
+
+ComPtr<IAudioSessionControl> CreateAudioSessionControl(IAudioClient* client) {
+  RTC_DLOG(LS_INFO) << "CreateAudioSessionControl";
+  RTC_DCHECK(client);
+  ComPtr<IAudioSessionControl> audio_session_control;
+  _com_error error = client->GetService(IID_PPV_ARGS(&audio_session_control));
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::GetService(IID_IAudioControl) failed: "
+                      << ErrorToString(error);
+    return ComPtr<IAudioSessionControl>();
+  }
+  return audio_session_control;
+}
+
+ComPtr<ISimpleAudioVolume> CreateSimpleAudioVolume(IAudioClient* client) {
+  RTC_DLOG(LS_INFO) << "CreateSimpleAudioVolume";
+  RTC_DCHECK(client);
+  // Get access to the ISimpleAudioVolume interface. This interface enables a
+  // client to control the master volume level of an audio session.
+  ComPtr<ISimpleAudioVolume> simple_audio_volume;
+  _com_error error = client->GetService(IID_PPV_ARGS(&simple_audio_volume));
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR)
+        << "IAudioClient::GetService(IID_ISimpleAudioVolume) failed: "
+        << ErrorToString(error);
+    return ComPtr<ISimpleAudioVolume>();
+  }
+  return simple_audio_volume;
+}
+
+bool FillRenderEndpointBufferWithSilence(IAudioClient* client,
+                                         IAudioRenderClient* render_client) {
+  RTC_DLOG(LS_INFO) << "FillRenderEndpointBufferWithSilence";
+  RTC_DCHECK(client);
+  RTC_DCHECK(render_client);
+  UINT32 endpoint_buffer_size = 0;
+  _com_error error = client->GetBufferSize(&endpoint_buffer_size);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::GetBufferSize failed: "
+                      << ErrorToString(error);
+    return false;
+  }
+
+  UINT32 num_queued_frames = 0;
+  // Get number of audio frames that are queued up to play in the endpoint
+  // buffer.
+  error = client->GetCurrentPadding(&num_queued_frames);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioClient::GetCurrentPadding failed: "
+                      << ErrorToString(error);
+    return false;
+  }
+  RTC_DLOG(LS_INFO) << "num_queued_frames: " << num_queued_frames;
+
+  BYTE* data = nullptr;
+  int num_frames_to_fill = endpoint_buffer_size - num_queued_frames;
+  RTC_DLOG(LS_INFO) << "num_frames_to_fill: " << num_frames_to_fill;
+  error = render_client->GetBuffer(num_frames_to_fill, &data);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioRenderClient::GetBuffer failed: "
+                      << ErrorToString(error);
+    return false;
+  }
+
+  // Using the AUDCLNT_BUFFERFLAGS_SILENT flag eliminates the need to
+  // explicitly write silence data to the rendering buffer.
+  error = render_client->ReleaseBuffer(num_frames_to_fill,
+                                       AUDCLNT_BUFFERFLAGS_SILENT);
+  if (FAILED(error.Error())) {
+    RTC_LOG(LS_ERROR) << "IAudioRenderClient::ReleaseBuffer failed: "
+                      << ErrorToString(error);
+    return false;
+  }
+
+  return true;
+}
+
+std::string WaveFormatToString(const WaveFormatWrapper format) {
+  char ss_buf[1024];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  // Start with the WAVEFORMATEX part (which always exists).
+  ss.AppendFormat("wFormatTag: %s (0x%X)",
+                  WaveFormatTagToString(format->wFormatTag),
+                  format->wFormatTag);
+  ss.AppendFormat(", nChannels: %d", format->nChannels);
+  ss.AppendFormat(", nSamplesPerSec: %d", format->nSamplesPerSec);
+  ss.AppendFormat(", nAvgBytesPerSec: %d", format->nAvgBytesPerSec);
+  ss.AppendFormat(", nBlockAlign: %d", format->nBlockAlign);
+  ss.AppendFormat(", wBitsPerSample: %d", format->wBitsPerSample);
+  ss.AppendFormat(", cbSize: %d", format->cbSize);
+  if (!format.IsExtensible())
+    return ss.str();
+
+  // Append the WAVEFORMATEXTENSIBLE part (which we know exists).
+  ss.AppendFormat(
+      " [+] wValidBitsPerSample: %d, dwChannelMask: %s",
+      format.GetExtensible()->Samples.wValidBitsPerSample,
+      ChannelMaskToString(format.GetExtensible()->dwChannelMask).c_str());
+  if (format.IsPcm()) {
+    ss.AppendFormat("%s", ", SubFormat: KSDATAFORMAT_SUBTYPE_PCM");
+  } else if (format.IsFloat()) {
+    ss.AppendFormat("%s", ", SubFormat: KSDATAFORMAT_SUBTYPE_IEEE_FLOAT");
+  } else {
+    ss.AppendFormat("%s", ", SubFormat: NOT_SUPPORTED");
+  }
+  return ss.str();
+}
+
+webrtc::TimeDelta ReferenceTimeToTimeDelta(REFERENCE_TIME time) {
+  // Each unit of reference time is 100 nanoseconds <=> 0.1 microsecond.
+  return webrtc::TimeDelta::Micros(0.1 * time + 0.5);
+}
+
+double FramesToMilliseconds(uint32_t num_frames, uint16_t sample_rate) {
+  // Convert the current period in frames into milliseconds.
+  return static_cast<double>(num_frames) / (sample_rate / 1000.0);
+}
+
+std::string ErrorToString(const _com_error& error) {
+  char ss_buf[1024];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  ss.AppendFormat("(HRESULT: 0x%08X)", error.Error());
+  return ss.str();
+}
+
+}  // namespace core_audio_utility
+}  // namespace webrtc_win
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_utility_win.h b/third_party/libwebrtc/modules/audio_device/win/core_audio_utility_win.h
new file mode 100644
index 0000000000..454e60bf31
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_utility_win.h
@@ -0,0 +1,560 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_UTILITY_WIN_H_
+#define MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_UTILITY_WIN_H_
+
+#include <audioclient.h>
+#include <audiopolicy.h>
+#include <avrt.h>
+#include <comdef.h>
+#include <mmdeviceapi.h>
+#include <objbase.h>
+#include <propidl.h>
+#include <wrl/client.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_device/audio_device_name.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+
+#pragma comment(lib, "Avrt.lib")
+
+namespace webrtc {
+namespace webrtc_win {
+
+// Utility class which registers a thread with MMCSS in the constructor and
+// deregisters MMCSS in the destructor. The task name is given by `task_name`.
+// The Multimedia Class Scheduler service (MMCSS) enables multimedia
+// applications to ensure that their time-sensitive processing receives
+// prioritized access to CPU resources without denying CPU resources to
+// lower-priority applications.
+class ScopedMMCSSRegistration {
+ public:
+  const char* PriorityClassToString(DWORD priority_class) {
+    switch (priority_class) {
+      case ABOVE_NORMAL_PRIORITY_CLASS:
+        return "ABOVE_NORMAL";
+      case BELOW_NORMAL_PRIORITY_CLASS:
+        return "BELOW_NORMAL";
+      case HIGH_PRIORITY_CLASS:
+        return "HIGH";
+      case IDLE_PRIORITY_CLASS:
+        return "IDLE";
+      case NORMAL_PRIORITY_CLASS:
+        return "NORMAL";
+      case REALTIME_PRIORITY_CLASS:
+        return "REALTIME";
+      default:
+        return "INVALID";
+    }
+  }
+
+  const char* PriorityToString(int priority) {
+    switch (priority) {
+      case THREAD_PRIORITY_ABOVE_NORMAL:
+        return "ABOVE_NORMAL";
+      case THREAD_PRIORITY_BELOW_NORMAL:
+        return "BELOW_NORMAL";
+      case THREAD_PRIORITY_HIGHEST:
+        return "HIGHEST";
+      case THREAD_PRIORITY_IDLE:
+        return "IDLE";
+      case THREAD_PRIORITY_LOWEST:
+        return "LOWEST";
+      case THREAD_PRIORITY_NORMAL:
+        return "NORMAL";
+      case THREAD_PRIORITY_TIME_CRITICAL:
+        return "TIME_CRITICAL";
+      default:
+        // Can happen in combination with REALTIME_PRIORITY_CLASS.
+        return "INVALID";
+    }
+  }
+
+  explicit ScopedMMCSSRegistration(const wchar_t* task_name) {
+    RTC_DLOG(LS_INFO) << "ScopedMMCSSRegistration: " << rtc::ToUtf8(task_name);
+    // Register the calling thread with MMCSS for the supplied `task_name`.
+    DWORD mmcss_task_index = 0;
+    mmcss_handle_ = AvSetMmThreadCharacteristicsW(task_name, &mmcss_task_index);
+    if (mmcss_handle_ == nullptr) {
+      RTC_LOG(LS_ERROR) << "Failed to enable MMCSS on this thread: "
+                        << GetLastError();
+    } else {
+      const DWORD priority_class = GetPriorityClass(GetCurrentProcess());
+      const int priority = GetThreadPriority(GetCurrentThread());
+      RTC_DLOG(LS_INFO) << "priority class: "
+                        << PriorityClassToString(priority_class) << "("
+                        << priority_class << ")";
+      RTC_DLOG(LS_INFO) << "priority: " << PriorityToString(priority) << "("
+                        << priority << ")";
+    }
+  }
+
+  ~ScopedMMCSSRegistration() {
+    if (Succeeded()) {
+      // Deregister with MMCSS.
+      RTC_DLOG(LS_INFO) << "~ScopedMMCSSRegistration";
+      AvRevertMmThreadCharacteristics(mmcss_handle_);
+    }
+  }
+
+  ScopedMMCSSRegistration(const ScopedMMCSSRegistration&) = delete;
+  ScopedMMCSSRegistration& operator=(const ScopedMMCSSRegistration&) = delete;
+
+  bool Succeeded() const { return mmcss_handle_ != nullptr; }
+
+ private:
+  HANDLE mmcss_handle_ = nullptr;
+};
+
+// A PROPVARIANT that is automatically initialized and cleared upon respective
+// construction and destruction of this class.
+class ScopedPropVariant {
+ public:
+  ScopedPropVariant() { PropVariantInit(&pv_); }
+
+  ~ScopedPropVariant() { Reset(); }
+
+  ScopedPropVariant(const ScopedPropVariant&) = delete;
+  ScopedPropVariant& operator=(const ScopedPropVariant&) = delete;
+  bool operator==(const ScopedPropVariant&) const = delete;
+  bool operator!=(const ScopedPropVariant&) const = delete;
+
+  // Returns a pointer to the underlying PROPVARIANT for use as an out param in
+  // a function call.
+  PROPVARIANT* Receive() {
+    RTC_DCHECK_EQ(pv_.vt, VT_EMPTY);
+    return &pv_;
+  }
+
+  // Clears the instance to prepare it for re-use (e.g., via Receive).
+  void Reset() {
+    if (pv_.vt != VT_EMPTY) {
+      HRESULT result = PropVariantClear(&pv_);
+      RTC_DCHECK_EQ(result, S_OK);
+    }
+  }
+
+  const PROPVARIANT& get() const { return pv_; }
+  const PROPVARIANT* ptr() const { return &pv_; }
+
+ private:
+  PROPVARIANT pv_;
+};
+
+// Simple scoped memory releaser class for COM allocated memory.
+template <typename T>
+class ScopedCoMem {
+ public:
+  ScopedCoMem() : mem_ptr_(nullptr) {}
+
+  ~ScopedCoMem() { Reset(nullptr); }
+
+  ScopedCoMem(const ScopedCoMem&) = delete;
+  ScopedCoMem& operator=(const ScopedCoMem&) = delete;
+
+  T** operator&() {                   // NOLINT
+    RTC_DCHECK(mem_ptr_ == nullptr);  // To catch memory leaks.
+    return &mem_ptr_;
+  }
+
+  operator T*() { return mem_ptr_; }
+
+  T* operator->() {
+    RTC_DCHECK(mem_ptr_ != nullptr);
+    return mem_ptr_;
+  }
+
+  const T* operator->() const {
+    RTC_DCHECK(mem_ptr_ != nullptr);
+    return mem_ptr_;
+  }
+
+  explicit operator bool() const { return mem_ptr_; }
+
+  friend bool operator==(const ScopedCoMem& lhs, std::nullptr_t) {
+    return lhs.Get() == nullptr;
+  }
+
+  friend bool operator==(std::nullptr_t, const ScopedCoMem& rhs) {
+    return rhs.Get() == nullptr;
+  }
+
+  friend bool operator!=(const ScopedCoMem& lhs, std::nullptr_t) {
+    return lhs.Get() != nullptr;
+  }
+
+  friend bool operator!=(std::nullptr_t, const ScopedCoMem& rhs) {
+    return rhs.Get() != nullptr;
+  }
+
+  void Reset(T* ptr) {
+    if (mem_ptr_)
+      CoTaskMemFree(mem_ptr_);
+    mem_ptr_ = ptr;
+  }
+
+  T* Get() const { return mem_ptr_; }
+
+ private:
+  T* mem_ptr_;
+};
+
+// A HANDLE that is automatically initialized and closed upon respective
+// construction and destruction of this class.
+class ScopedHandle {
+ public:
+  ScopedHandle() : handle_(nullptr) {}
+  explicit ScopedHandle(HANDLE h) : handle_(nullptr) { Set(h); }
+
+  ~ScopedHandle() { Close(); }
+
+  ScopedHandle& operator=(const ScopedHandle&) = delete;
+  bool operator==(const ScopedHandle&) const = delete;
+  bool operator!=(const ScopedHandle&) const = delete;
+
+  // Use this instead of comparing to INVALID_HANDLE_VALUE.
+  bool IsValid() const { return handle_ != nullptr; }
+
+  void Set(HANDLE new_handle) {
+    Close();
+    // Windows is inconsistent about invalid handles.
+    // See https://blogs.msdn.microsoft.com/oldnewthing/20040302-00/?p=40443
+    // for details.
+    if (new_handle != INVALID_HANDLE_VALUE) {
+      handle_ = new_handle;
+    }
+  }
+
+  HANDLE Get() const { return handle_; }
+
+  operator HANDLE() const { return handle_; }
+
+  void Close() {
+    if (handle_) {
+      if (!::CloseHandle(handle_)) {
+        RTC_DCHECK_NOTREACHED();
+      }
+      handle_ = nullptr;
+    }
+  }
+
+ private:
+  HANDLE handle_;
+};
+
+// Utility methods for the Core Audio API on Windows.
+// Always ensure that Core Audio is supported before using these methods.
+// Use webrtc_win::core_audio_utility::IsSupported() for this purpose.
+// Also, all methods must be called on a valid COM thread. This can be done
+// by using the ScopedCOMInitializer helper class.
+// These methods are based on media::CoreAudioUtil in Chrome.
+namespace core_audio_utility {
+
+// Helper class which automates casting between WAVEFORMATEX and
+// WAVEFORMATEXTENSIBLE raw pointers using implicit constructors and
+// operator overloading. Note that, no memory is allocated by this utility
+// structure. It only serves as a handle (or a wrapper) of the structure
+// provided to it at construction.
+class WaveFormatWrapper {
+ public:
+  WaveFormatWrapper(WAVEFORMATEXTENSIBLE* p)
+      : ptr_(reinterpret_cast<WAVEFORMATEX*>(p)) {}
+  WaveFormatWrapper(WAVEFORMATEX* p) : ptr_(p) {}
+  ~WaveFormatWrapper() = default;
+
+  operator WAVEFORMATEX*() const { return ptr_; }
+  WAVEFORMATEX* operator->() const { return ptr_; }
+  WAVEFORMATEX* get() const { return ptr_; }
+  WAVEFORMATEXTENSIBLE* GetExtensible() const;
+
+  bool IsExtensible() const;
+  bool IsPcm() const;
+  bool IsFloat() const;
+  size_t size() const;
+
+ private:
+  WAVEFORMATEX* ptr_;
+};
+
+// Returns true if Windows Core Audio is supported.
+// Always verify that this method returns true before using any of the
+// other methods in this class.
+bool IsSupported();
+
+// Returns true if Multimedia Class Scheduler service (MMCSS) is supported.
+// The MMCSS enables multimedia applications to ensure that their time-sensitive
+// processing receives prioritized access to CPU resources without denying CPU
+// resources to lower-priority applications.
+bool IsMMCSSSupported();
+
+// The MMDevice API lets clients discover the audio endpoint devices in the
+// system and determine which devices are suitable for the application to use.
+// Header file Mmdeviceapi.h defines the interfaces in the MMDevice API.
+
+// Number of active audio devices in the specified data flow direction.
+// Set `data_flow` to eAll to retrieve the total number of active audio
+// devices.
+int NumberOfActiveDevices(EDataFlow data_flow);
+
+// Returns 1, 2, or 3 depending on what version of IAudioClient the platform
+// supports.
+// Example: IAudioClient2 is supported on Windows 8 and higher => 2 is returned.
+uint32_t GetAudioClientVersion();
+
+// Creates an IMMDeviceEnumerator interface which provides methods for
+// enumerating audio endpoint devices.
+// TODO(henrika): IMMDeviceEnumerator::RegisterEndpointNotificationCallback.
+Microsoft::WRL::ComPtr<IMMDeviceEnumerator> CreateDeviceEnumerator();
+
+// These functions return the unique device id of the default or
+// communications input/output device, or an empty string if no such device
+// exists or if the device has been disabled.
+std::string GetDefaultInputDeviceID();
+std::string GetDefaultOutputDeviceID();
+std::string GetCommunicationsInputDeviceID();
+std::string GetCommunicationsOutputDeviceID();
+
+// Creates an IMMDevice interface corresponding to the unique device id in
+// `device_id`, or by data-flow direction and role if `device_id` is set to
+// AudioDeviceName::kDefaultDeviceId.
+Microsoft::WRL::ComPtr<IMMDevice> CreateDevice(absl::string_view device_id,
+                                               EDataFlow data_flow,
+                                               ERole role);
+
+// Returns the unique ID and user-friendly name of a given endpoint device.
+// Example: "{0.0.1.00000000}.{8db6020f-18e3-4f25-b6f5-7726c9122574}", and
+//          "Microphone (Realtek High Definition Audio)".
+webrtc::AudioDeviceName GetDeviceName(IMMDevice* device);
+
+// Gets the user-friendly name of the endpoint device which is represented
+// by a unique id in `device_id`, or by data-flow direction and role if
+// `device_id` is set to AudioDeviceName::kDefaultDeviceId.
+std::string GetFriendlyName(absl::string_view device_id,
+                            EDataFlow data_flow,
+                            ERole role);
+
+// Query if the audio device is a rendering device or a capture device.
+EDataFlow GetDataFlow(IMMDevice* device);
+
+// Enumerates all input devices and adds the names (friendly name and unique
+// device id) to the list in `device_names`.
+bool GetInputDeviceNames(webrtc::AudioDeviceNames* device_names);
+
+// Enumerates all output devices and adds the names (friendly name and unique
+// device id) to the list in `device_names`.
+bool GetOutputDeviceNames(webrtc::AudioDeviceNames* device_names);
+
+// The Windows Audio Session API (WASAPI) enables client applications to
+// manage the flow of audio data between the application and an audio endpoint
+// device. Header files Audioclient.h and Audiopolicy.h define the WASAPI
+// interfaces.
+
+// Creates an IAudioSessionManager2 interface for the specified `device`.
+// This interface provides access to e.g. the IAudioSessionEnumerator
+Microsoft::WRL::ComPtr<IAudioSessionManager2> CreateSessionManager2(
+    IMMDevice* device);
+
+// Creates an IAudioSessionEnumerator interface for the specified `device`.
+// The client can use the interface to enumerate audio sessions on the audio
+// device
+Microsoft::WRL::ComPtr<IAudioSessionEnumerator> CreateSessionEnumerator(
+    IMMDevice* device);
+
+// Number of active audio sessions for the given `device`. Expired or inactive
+// sessions are not included.
+int NumberOfActiveSessions(IMMDevice* device);
+
+// Creates an IAudioClient instance for a specific device or the default
+// device specified by data-flow direction and role.
+Microsoft::WRL::ComPtr<IAudioClient> CreateClient(absl::string_view device_id,
+                                                  EDataFlow data_flow,
+                                                  ERole role);
+Microsoft::WRL::ComPtr<IAudioClient2> CreateClient2(absl::string_view device_id,
+                                                    EDataFlow data_flow,
+                                                    ERole role);
+Microsoft::WRL::ComPtr<IAudioClient3> CreateClient3(absl::string_view device_id,
+                                                    EDataFlow data_flow,
+                                                    ERole role);
+
+// Sets the AudioCategory_Communications category. Should be called before
+// GetSharedModeMixFormat() and IsFormatSupported(). The `client` argument must
+// be an IAudioClient2 or IAudioClient3 interface pointer, hence only supported
+// on Windows 8 and above.
+// TODO(henrika): evaluate effect (if any).
+HRESULT SetClientProperties(IAudioClient2* client);
+
+// Returns the buffer size limits of the hardware audio engine in
+// 100-nanosecond units given a specified `format`. Does not require prior
+// audio stream initialization. The `client` argument must be an IAudioClient2
+// or IAudioClient3 interface pointer, hence only supported on Windows 8 and
+// above.
+// TODO(henrika): always fails with AUDCLNT_E_OFFLOAD_MODE_ONLY.
+HRESULT GetBufferSizeLimits(IAudioClient2* client,
+                            const WAVEFORMATEXTENSIBLE* format,
+                            REFERENCE_TIME* min_buffer_duration,
+                            REFERENCE_TIME* max_buffer_duration);
+
+// Get the mix format that the audio engine uses internally for processing
+// of shared-mode streams. The client can call this method before calling
+// IAudioClient::Initialize. When creating a shared-mode stream for an audio
+// endpoint device, the Initialize method always accepts the stream format
+// obtained by this method.
+HRESULT GetSharedModeMixFormat(IAudioClient* client,
+                               WAVEFORMATEXTENSIBLE* format);
+
+// Returns true if the specified `client` supports the format in `format`
+// for the given `share_mode` (shared or exclusive). The client can call this
+// method before calling IAudioClient::Initialize.
+bool IsFormatSupported(IAudioClient* client,
+                       AUDCLNT_SHAREMODE share_mode,
+                       const WAVEFORMATEXTENSIBLE* format);
+
+// For a shared-mode stream, the audio engine periodically processes the
+// data in the endpoint buffer at the period obtained in `device_period`.
+// For an exclusive mode stream, `device_period` corresponds to the minimum
+// time interval between successive processing by the endpoint device.
+// This period plus the stream latency between the buffer and endpoint device
+// represents the minimum possible latency that an audio application can
+// achieve. The time in `device_period` is expressed in 100-nanosecond units.
+HRESULT GetDevicePeriod(IAudioClient* client,
+                        AUDCLNT_SHAREMODE share_mode,
+                        REFERENCE_TIME* device_period);
+
+// Returns the range of periodicities supported by the engine for the specified
+// stream `format`. The periodicity of the engine is the rate at which the
+// engine wakes an event-driven audio client to transfer audio data to or from
+// the engine. Can be used for low-latency support on some devices.
+// The `client` argument must be an IAudioClient3 interface pointer, hence only
+// supported on Windows 10 and above.
+HRESULT GetSharedModeEnginePeriod(IAudioClient3* client3,
+                                  const WAVEFORMATEXTENSIBLE* format,
+                                  uint32_t* default_period_in_frames,
+                                  uint32_t* fundamental_period_in_frames,
+                                  uint32_t* min_period_in_frames,
+                                  uint32_t* max_period_in_frames);
+
+// Get the preferred audio parameters for the given `client` corresponding to
+// the stream format that the audio engine uses for its internal processing of
+// shared-mode streams. The acquired values should only be utilized for shared
+// mode streamed since there are no preferred settings for an exclusive mode
+// stream.
+HRESULT GetPreferredAudioParameters(IAudioClient* client,
+                                    webrtc::AudioParameters* params);
+// As above but override the preferred sample rate and use `sample_rate`
+// instead. Intended mainly for testing purposes and in combination with rate
+// conversion.
+HRESULT GetPreferredAudioParameters(IAudioClient* client,
+                                    webrtc::AudioParameters* params,
+                                    uint32_t sample_rate);
+
+// After activating an IAudioClient interface on an audio endpoint device,
+// the client must initialize it once, and only once, to initialize the audio
+// stream between the client and the device. In shared mode, the client
+// connects indirectly through the audio engine which does the mixing.
+// If a valid event is provided in `event_handle`, the client will be
+// initialized for event-driven buffer handling. If `event_handle` is set to
+// nullptr, event-driven buffer handling is not utilized. To achieve the
+// minimum stream latency between the client application and audio endpoint
+// device, set `buffer_duration` to 0. A client has the option of requesting a
+// buffer size that is larger than what is strictly necessary to make timing
+// glitches rare or nonexistent. Increasing the buffer size does not necessarily
+// increase the stream latency. Each unit of reference time is 100 nanoseconds.
+// The `auto_convert_pcm` parameter can be used for testing purposes to ensure
+// that the sample rate of the client side does not have to match the audio
+// engine mix format. If `auto_convert_pcm` is set to true, a rate converter
+// will be inserted to convert between the sample rate in `format` and the
+// preferred rate given by GetPreferredAudioParameters().
+// The output parameter `endpoint_buffer_size` contains the size of the
+// endpoint buffer and it is expressed as the number of audio frames the
+// buffer can hold.
+HRESULT SharedModeInitialize(IAudioClient* client,
+                             const WAVEFORMATEXTENSIBLE* format,
+                             HANDLE event_handle,
+                             REFERENCE_TIME buffer_duration,
+                             bool auto_convert_pcm,
+                             uint32_t* endpoint_buffer_size);
+
+// Works as SharedModeInitialize() but adds support for using smaller engine
+// periods than the default period.
+// The `client` argument must be an IAudioClient3 interface pointer, hence only
+// supported on Windows 10 and above.
+// TODO(henrika): can probably be merged into SharedModeInitialize() to avoid
+// duplicating code. Keeping as separate method for now until decided if we
+// need low-latency support.
+HRESULT SharedModeInitializeLowLatency(IAudioClient3* client,
+                                       const WAVEFORMATEXTENSIBLE* format,
+                                       HANDLE event_handle,
+                                       uint32_t period_in_frames,
+                                       bool auto_convert_pcm,
+                                       uint32_t* endpoint_buffer_size);
+
+// Creates an IAudioRenderClient client for an existing IAudioClient given by
+// `client`. The IAudioRenderClient interface enables a client to write
+// output data to a rendering endpoint buffer. The methods in this interface
+// manage the movement of data packets that contain audio-rendering data.
+Microsoft::WRL::ComPtr<IAudioRenderClient> CreateRenderClient(
+    IAudioClient* client);
+
+// Creates an IAudioCaptureClient client for an existing IAudioClient given by
+// `client`. The IAudioCaptureClient interface enables a client to read
+// input data from a capture endpoint buffer. The methods in this interface
+// manage the movement of data packets that contain capture data.
+Microsoft::WRL::ComPtr<IAudioCaptureClient> CreateCaptureClient(
+    IAudioClient* client);
+
+// Creates an IAudioClock interface for an existing IAudioClient given by
+// `client`. The IAudioClock interface enables a client to monitor a stream's
+// data rate and the current position in the stream.
+Microsoft::WRL::ComPtr<IAudioClock> CreateAudioClock(IAudioClient* client);
+
+// Creates an AudioSessionControl interface for an existing IAudioClient given
+// by `client`. The IAudioControl interface enables a client to configure the
+// control parameters for an audio session and to monitor events in the session.
+Microsoft::WRL::ComPtr<IAudioSessionControl> CreateAudioSessionControl(
+    IAudioClient* client);
+
+// Creates an ISimpleAudioVolume interface for an existing IAudioClient given by
+// `client`. This interface enables a client to control the master volume level
+// of an active audio session.
+Microsoft::WRL::ComPtr<ISimpleAudioVolume> CreateSimpleAudioVolume(
+    IAudioClient* client);
+
+// Fills up the endpoint rendering buffer with silence for an existing
+// IAudioClient given by `client` and a corresponding IAudioRenderClient
+// given by `render_client`.
+bool FillRenderEndpointBufferWithSilence(IAudioClient* client,
+                                         IAudioRenderClient* render_client);
+
+// Prints/logs all fields of the format structure in `format`.
+// Also supports extended versions (WAVEFORMATEXTENSIBLE).
+std::string WaveFormatToString(WaveFormatWrapper format);
+
+// Converts Windows internal REFERENCE_TIME (100 nanosecond units) into
+// generic webrtc::TimeDelta which then can be converted to any time unit.
+webrtc::TimeDelta ReferenceTimeToTimeDelta(REFERENCE_TIME time);
+
+// Converts size expressed in number of audio frames, `num_frames`, into
+// milliseconds given a specified `sample_rate`.
+double FramesToMilliseconds(uint32_t num_frames, uint16_t sample_rate);
+
+// Converts a COM error into a human-readable string.
+std::string ErrorToString(const _com_error& error);
+
+}  // namespace core_audio_utility
+}  // namespace webrtc_win
+}  // namespace webrtc
+
+#endif  //  MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_UTILITY_WIN_H_
diff --git a/third_party/libwebrtc/modules/audio_device/win/core_audio_utility_win_unittest.cc b/third_party/libwebrtc/modules/audio_device/win/core_audio_utility_win_unittest.cc
new file mode 100644
index 0000000000..277f54eb35
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_device/win/core_audio_utility_win_unittest.cc
@@ -0,0 +1,876 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/win/core_audio_utility_win.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/win/scoped_com_initializer.h"
+#include "rtc_base/win/windows_version.h"
+#include "test/gtest.h"
+
+using Microsoft::WRL::ComPtr;
+using webrtc::AudioDeviceName;
+
+namespace webrtc {
+namespace webrtc_win {
+namespace {
+
+#define ABORT_TEST_IF_NOT(requirements_satisfied)                        \
+  do {                                                                   \
+    bool fail = false;                                                   \
+    if (ShouldAbortTest(requirements_satisfied, #requirements_satisfied, \
+                        &fail)) {                                        \
+      if (fail)                                                          \
+        FAIL();                                                          \
+      else                                                               \
+        return;                                                          \
+    }                                                                    \
+  } while (false)
+
+bool ShouldAbortTest(bool requirements_satisfied,
+                     const char* requirements_expression,
+                     bool* should_fail) {
+  if (!requirements_satisfied) {
+    RTC_LOG(LS_ERROR) << "Requirement(s) not satisfied ("
+                      << requirements_expression << ")";
+    // TODO(henrika): improve hard-coded condition to determine if test should
+    // fail or be ignored. Could use e.g. a command-line argument here to
+    // determine if the test should fail or not.
+    *should_fail = false;
+    return true;
+  }
+  *should_fail = false;
+  return false;
+}
+
+}  // namespace
+
+// CoreAudioUtilityWinTest test fixture.
+class CoreAudioUtilityWinTest : public ::testing::Test {
+ protected:
+  CoreAudioUtilityWinTest() : com_init_(ScopedCOMInitializer::kMTA) {
+    // We must initialize the COM library on a thread before we calling any of
+    // the library functions. All COM functions will return CO_E_NOTINITIALIZED
+    // otherwise.
+    EXPECT_TRUE(com_init_.Succeeded());
+
+    // Configure logging.
+    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+    rtc::LogMessage::LogTimestamps();
+    rtc::LogMessage::LogThreads();
+  }
+
+  virtual ~CoreAudioUtilityWinTest() {}
+
+  bool DevicesAvailable() {
+    return core_audio_utility::IsSupported() &&
+           core_audio_utility::NumberOfActiveDevices(eCapture) > 0 &&
+           core_audio_utility::NumberOfActiveDevices(eRender) > 0;
+  }
+
+ private:
+  ScopedCOMInitializer com_init_;
+};
+
+TEST_F(CoreAudioUtilityWinTest, WaveFormatWrapper) {
+  // Use default constructor for WAVEFORMATEX and verify its size.
+  WAVEFORMATEX format = {};
+  core_audio_utility::WaveFormatWrapper wave_format(&format);
+  EXPECT_FALSE(wave_format.IsExtensible());
+  EXPECT_EQ(wave_format.size(), sizeof(WAVEFORMATEX));
+  EXPECT_EQ(wave_format->cbSize, 0);
+
+  // Ensure that the stand-alone WAVEFORMATEX structure has a valid format tag
+  // and that all accessors work.
+  format.wFormatTag = WAVE_FORMAT_PCM;
+  EXPECT_FALSE(wave_format.IsExtensible());
+  EXPECT_EQ(wave_format.size(), sizeof(WAVEFORMATEX));
+  EXPECT_EQ(wave_format.get()->wFormatTag, WAVE_FORMAT_PCM);
+  EXPECT_EQ(wave_format->wFormatTag, WAVE_FORMAT_PCM);
+
+  // Next, ensure that the size is valid. Stand-alone is not extended.
+  EXPECT_EQ(wave_format.size(), sizeof(WAVEFORMATEX));
+
+  // Verify format types for the stand-alone version.
+  EXPECT_TRUE(wave_format.IsPcm());
+  EXPECT_FALSE(wave_format.IsFloat());
+  format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
+  EXPECT_TRUE(wave_format.IsFloat());
+}
+
+TEST_F(CoreAudioUtilityWinTest, WaveFormatWrapperExtended) {
+  // Use default constructor for WAVEFORMATEXTENSIBLE and verify that it
+  // results in same size as for WAVEFORMATEX even if the size of `format_ex`
+  // equals the size of WAVEFORMATEXTENSIBLE.
+  WAVEFORMATEXTENSIBLE format_ex = {};
+  core_audio_utility::WaveFormatWrapper wave_format_ex(&format_ex);
+  EXPECT_FALSE(wave_format_ex.IsExtensible());
+  EXPECT_EQ(wave_format_ex.size(), sizeof(WAVEFORMATEX));
+  EXPECT_EQ(wave_format_ex->cbSize, 0);
+
+  // Ensure that the extended structure has a valid format tag and that all
+  // accessors work.
+  format_ex.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+  EXPECT_FALSE(wave_format_ex.IsExtensible());
+  EXPECT_EQ(wave_format_ex.size(), sizeof(WAVEFORMATEX));
+  EXPECT_EQ(wave_format_ex->wFormatTag, WAVE_FORMAT_EXTENSIBLE);
+  EXPECT_EQ(wave_format_ex.get()->wFormatTag, WAVE_FORMAT_EXTENSIBLE);
+
+  // Next, ensure that the size is valid (sum of stand-alone and extended).
+  // Now the structure qualifies as extended.
+  format_ex.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
+  EXPECT_TRUE(wave_format_ex.IsExtensible());
+  EXPECT_EQ(wave_format_ex.size(), sizeof(WAVEFORMATEXTENSIBLE));
+  EXPECT_TRUE(wave_format_ex.GetExtensible());
+  EXPECT_EQ(wave_format_ex.GetExtensible()->Format.wFormatTag,
+            WAVE_FORMAT_EXTENSIBLE);
+
+  // Verify format types for the extended version.
+  EXPECT_FALSE(wave_format_ex.IsPcm());
+  format_ex.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+  EXPECT_TRUE(wave_format_ex.IsPcm());
+  EXPECT_FALSE(wave_format_ex.IsFloat());
+  format_ex.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
+  EXPECT_TRUE(wave_format_ex.IsFloat());
+}
+
+TEST_F(CoreAudioUtilityWinTest, NumberOfActiveDevices) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+  int render_devices = core_audio_utility::NumberOfActiveDevices(eRender);
+  EXPECT_GT(render_devices, 0);
+  int capture_devices = core_audio_utility::NumberOfActiveDevices(eCapture);
+  EXPECT_GT(capture_devices, 0);
+  int total_devices = core_audio_utility::NumberOfActiveDevices(eAll);
+  EXPECT_EQ(total_devices, render_devices + capture_devices);
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetAudioClientVersion) {
+  uint32_t client_version = core_audio_utility::GetAudioClientVersion();
+  EXPECT_GE(client_version, 1u);
+  EXPECT_LE(client_version, 3u);
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateDeviceEnumerator) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+  ComPtr<IMMDeviceEnumerator> enumerator =
+      core_audio_utility::CreateDeviceEnumerator();
+  EXPECT_TRUE(enumerator.Get());
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetDefaultInputDeviceID) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+  std::string default_device_id = core_audio_utility::GetDefaultInputDeviceID();
+  EXPECT_FALSE(default_device_id.empty());
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetDefaultOutputDeviceID) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+  std::string default_device_id =
+      core_audio_utility::GetDefaultOutputDeviceID();
+  EXPECT_FALSE(default_device_id.empty());
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetCommunicationsInputDeviceID) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+  std::string default_device_id =
+      core_audio_utility::GetCommunicationsInputDeviceID();
+  EXPECT_FALSE(default_device_id.empty());
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetCommunicationsOutputDeviceID) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+  std::string default_device_id =
+      core_audio_utility::GetCommunicationsOutputDeviceID();
+  EXPECT_FALSE(default_device_id.empty());
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateDefaultDevice) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  struct {
+    EDataFlow flow;
+    ERole role;
+  } data[] = {{eRender, eConsole},         {eRender, eCommunications},
+              {eRender, eMultimedia},      {eCapture, eConsole},
+              {eCapture, eCommunications}, {eCapture, eMultimedia}};
+
+  // Create default devices for all flow/role combinations above.
+  ComPtr<IMMDevice> audio_device;
+  for (size_t i = 0; i < arraysize(data); ++i) {
+    audio_device = core_audio_utility::CreateDevice(
+        AudioDeviceName::kDefaultDeviceId, data[i].flow, data[i].role);
+    EXPECT_TRUE(audio_device.Get());
+    EXPECT_EQ(data[i].flow,
+              core_audio_utility::GetDataFlow(audio_device.Get()));
+  }
+
+  // Only eRender and eCapture are allowed as flow parameter.
+  audio_device = core_audio_utility::CreateDevice(
+      AudioDeviceName::kDefaultDeviceId, eAll, eConsole);
+  EXPECT_FALSE(audio_device.Get());
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateDevice) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  // Get name and ID of default device used for playback.
+  ComPtr<IMMDevice> default_render_device = core_audio_utility::CreateDevice(
+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
+  AudioDeviceName default_render_name =
+      core_audio_utility::GetDeviceName(default_render_device.Get());
+  EXPECT_TRUE(default_render_name.IsValid());
+
+  // Use the unique ID as input to CreateDevice() and create a corresponding
+  // IMMDevice. The data-flow direction and role parameters are ignored for
+  // this scenario.
+  ComPtr<IMMDevice> audio_device = core_audio_utility::CreateDevice(
+      default_render_name.unique_id, EDataFlow(), ERole());
+  EXPECT_TRUE(audio_device.Get());
+
+  // Verify that the two IMMDevice interfaces represents the same endpoint
+  // by comparing their unique IDs.
+  AudioDeviceName device_name =
+      core_audio_utility::GetDeviceName(audio_device.Get());
+  EXPECT_EQ(default_render_name.unique_id, device_name.unique_id);
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetDefaultDeviceName) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  struct {
+    EDataFlow flow;
+    ERole role;
+  } data[] = {{eRender, eConsole},
+              {eRender, eCommunications},
+              {eCapture, eConsole},
+              {eCapture, eCommunications}};
+
+  // Get name and ID of default devices for all flow/role combinations above.
+  ComPtr<IMMDevice> audio_device;
+  AudioDeviceName device_name;
+  for (size_t i = 0; i < arraysize(data); ++i) {
+    audio_device = core_audio_utility::CreateDevice(
+        AudioDeviceName::kDefaultDeviceId, data[i].flow, data[i].role);
+    device_name = core_audio_utility::GetDeviceName(audio_device.Get());
+    EXPECT_TRUE(device_name.IsValid());
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetFriendlyName) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  // Get name and ID of default device used for recording.
+  ComPtr<IMMDevice> audio_device = core_audio_utility::CreateDevice(
+      AudioDeviceName::kDefaultDeviceId, eCapture, eConsole);
+  AudioDeviceName device_name =
+      core_audio_utility::GetDeviceName(audio_device.Get());
+  EXPECT_TRUE(device_name.IsValid());
+
+  // Use unique ID as input to GetFriendlyName() and compare the result
+  // with the already obtained friendly name for the default capture device.
+  std::string friendly_name = core_audio_utility::GetFriendlyName(
+      device_name.unique_id, eCapture, eConsole);
+  EXPECT_EQ(friendly_name, device_name.device_name);
+
+  // Same test as above but for playback.
+  audio_device = core_audio_utility::CreateDevice(
+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
+  device_name = core_audio_utility::GetDeviceName(audio_device.Get());
+  friendly_name = core_audio_utility::GetFriendlyName(device_name.unique_id,
+                                                      eRender, eConsole);
+  EXPECT_EQ(friendly_name, device_name.device_name);
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetInputDeviceNames) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  webrtc::AudioDeviceNames device_names;
+  EXPECT_TRUE(core_audio_utility::GetInputDeviceNames(&device_names));
+  // Number of elements in the list should be two more than the number of
+  // active devices since we always add default and default communication
+  // devices on index 0 and 1.
+  EXPECT_EQ(static_cast<int>(device_names.size()),
+            2 + core_audio_utility::NumberOfActiveDevices(eCapture));
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetOutputDeviceNames) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  webrtc::AudioDeviceNames device_names;
+  EXPECT_TRUE(core_audio_utility::GetOutputDeviceNames(&device_names));
+  // Number of elements in the list should be two more than the number of
+  // active devices since we always add default and default communication
+  // devices on index 0 and 1.
+  EXPECT_EQ(static_cast<int>(device_names.size()),
+            2 + core_audio_utility::NumberOfActiveDevices(eRender));
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateSessionManager2) {
+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
+                    rtc::rtc_win::GetVersion() >= rtc::rtc_win::VERSION_WIN7);
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  // Obtain reference to an IAudioSessionManager2 interface for a default audio
+  // endpoint device specified by two different data flows and the `eConsole`
+  // role.
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    ComPtr<IMMDevice> device(core_audio_utility::CreateDevice(
+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole));
+    EXPECT_TRUE(device.Get());
+    ComPtr<IAudioSessionManager2> session_manager =
+        core_audio_utility::CreateSessionManager2(device.Get());
+    EXPECT_TRUE(session_manager.Get());
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateSessionEnumerator) {
+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
+                    rtc::rtc_win::GetVersion() >= rtc::rtc_win::VERSION_WIN7);
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  // Obtain reference to an IAudioSessionEnumerator interface for a default
+  // audio endpoint device specified by two different data flows and the
+  // `eConsole` role.
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    ComPtr<IMMDevice> device(core_audio_utility::CreateDevice(
+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole));
+    EXPECT_TRUE(device.Get());
+    ComPtr<IAudioSessionEnumerator> session_enumerator =
+        core_audio_utility::CreateSessionEnumerator(device.Get());
+    EXPECT_TRUE(session_enumerator.Get());
+
+    // Perform a sanity test of the interface by asking for the total number
+    // of audio sessions that are open on the audio device. Note that, we do
+    // not check if the session is active or not.
+    int session_count = 0;
+    EXPECT_TRUE(SUCCEEDED(session_enumerator->GetCount(&session_count)));
+    EXPECT_GE(session_count, 0);
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, NumberOfActiveSessions) {
+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
+                    rtc::rtc_win::GetVersion() >= rtc::rtc_win::VERSION_WIN7);
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  // Count number of active audio session for a default audio endpoint device
+  // specified by two different data flows and the `eConsole` role.
+  // Ensure that the number of active audio sessions is less than or equal to
+  // the total number of audio sessions on that same device.
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    // Create an audio endpoint device.
+    ComPtr<IMMDevice> device(core_audio_utility::CreateDevice(
+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole));
+    EXPECT_TRUE(device.Get());
+
+    // Ask for total number of audio sessions on the created device.
+    ComPtr<IAudioSessionEnumerator> session_enumerator =
+        core_audio_utility::CreateSessionEnumerator(device.Get());
+    EXPECT_TRUE(session_enumerator.Get());
+    int total_session_count = 0;
+    EXPECT_TRUE(SUCCEEDED(session_enumerator->GetCount(&total_session_count)));
+    EXPECT_GE(total_session_count, 0);
+
+    // Use NumberOfActiveSessions and get number of active audio sessions.
+    int active_session_count =
+        core_audio_utility::NumberOfActiveSessions(device.Get());
+    EXPECT_LE(active_session_count, total_session_count);
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateClient) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  // Obtain reference to an IAudioClient interface for a default audio endpoint
+  // device specified by two different data flows and the `eConsole` role.
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    ComPtr<IAudioClient> client = core_audio_utility::CreateClient(
+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole);
+    EXPECT_TRUE(client.Get());
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateClient2) {
+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
+                    core_audio_utility::GetAudioClientVersion() >= 2);
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  // Obtain reference to an IAudioClient2 interface for a default audio endpoint
+  // device specified by two different data flows and the `eConsole` role.
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    ComPtr<IAudioClient2> client2 = core_audio_utility::CreateClient2(
+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole);
+    EXPECT_TRUE(client2.Get());
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateClient3) {
+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
+                    core_audio_utility::GetAudioClientVersion() >= 3);
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  // Obtain reference to an IAudioClient3 interface for a default audio endpoint
+  // device specified by two different data flows and the `eConsole` role.
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    ComPtr<IAudioClient3> client3 = core_audio_utility::CreateClient3(
+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole);
+    EXPECT_TRUE(client3.Get());
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, SetClientProperties) {
+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
+                    core_audio_utility::GetAudioClientVersion() >= 2);
+
+  ComPtr<IAudioClient2> client2 = core_audio_utility::CreateClient2(
+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
+  EXPECT_TRUE(client2.Get());
+  EXPECT_TRUE(
+      SUCCEEDED(core_audio_utility::SetClientProperties(client2.Get())));
+
+  ComPtr<IAudioClient3> client3 = core_audio_utility::CreateClient3(
+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
+  EXPECT_TRUE(client3.Get());
+  EXPECT_TRUE(
+      SUCCEEDED(core_audio_utility::SetClientProperties(client3.Get())));
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetSharedModeEnginePeriod) {
+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
+                    core_audio_utility::GetAudioClientVersion() >= 3);
+
+  ComPtr<IAudioClient3> client3 = core_audio_utility::CreateClient3(
+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
+  EXPECT_TRUE(client3.Get());
+
+  WAVEFORMATPCMEX format;
+  EXPECT_TRUE(SUCCEEDED(
+      core_audio_utility::GetSharedModeMixFormat(client3.Get(), &format)));
+
+  uint32_t default_period = 0;
+  uint32_t fundamental_period = 0;
+  uint32_t min_period = 0;
+  uint32_t max_period = 0;
+  EXPECT_TRUE(SUCCEEDED(core_audio_utility::GetSharedModeEnginePeriod(
+      client3.Get(), &format, &default_period, &fundamental_period, &min_period,
+      &max_period)));
+}
+
+// TODO(henrika): figure out why usage of this API always reports
+// AUDCLNT_E_OFFLOAD_MODE_ONLY.
+TEST_F(CoreAudioUtilityWinTest, DISABLED_GetBufferSizeLimits) {
+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
+                    core_audio_utility::GetAudioClientVersion() >= 2);
+
+  ComPtr<IAudioClient2> client2 = core_audio_utility::CreateClient2(
+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
+  EXPECT_TRUE(client2.Get());
+
+  WAVEFORMATPCMEX format;
+  EXPECT_TRUE(SUCCEEDED(
+      core_audio_utility::GetSharedModeMixFormat(client2.Get(), &format)));
+
+  REFERENCE_TIME min_buffer_duration = 0;
+  REFERENCE_TIME max_buffer_duration = 0;
+  EXPECT_TRUE(SUCCEEDED(core_audio_utility::GetBufferSizeLimits(
+      client2.Get(), &format, &min_buffer_duration, &max_buffer_duration)));
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetSharedModeMixFormat) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  ComPtr<IAudioClient> client = core_audio_utility::CreateClient(
+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
+  EXPECT_TRUE(client.Get());
+
+  // Perform a simple sanity test of the acquired format structure.
+  WAVEFORMATEXTENSIBLE format;
+  EXPECT_TRUE(SUCCEEDED(
+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
+  core_audio_utility::WaveFormatWrapper wformat(&format);
+  EXPECT_GE(wformat->nChannels, 1);
+  EXPECT_GE(wformat->nSamplesPerSec, 8000u);
+  EXPECT_GE(wformat->wBitsPerSample, 16);
+  if (wformat.IsExtensible()) {
+    EXPECT_EQ(wformat->wFormatTag, WAVE_FORMAT_EXTENSIBLE);
+    EXPECT_GE(wformat->cbSize, 22);
+    EXPECT_GE(wformat.GetExtensible()->Samples.wValidBitsPerSample, 16);
+  } else {
+    EXPECT_EQ(wformat->cbSize, 0);
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, IsFormatSupported) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  // Create a default render client.
+  ComPtr<IAudioClient> client = core_audio_utility::CreateClient(
+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
+  EXPECT_TRUE(client.Get());
+
+  // Get the default, shared mode, mixing format.
+  WAVEFORMATEXTENSIBLE format;
+  EXPECT_TRUE(SUCCEEDED(
+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
+
+  // In shared mode, the audio engine always supports the mix format.
+  EXPECT_TRUE(core_audio_utility::IsFormatSupported(
+      client.Get(), AUDCLNT_SHAREMODE_SHARED, &format));
+
+  // Use an invalid format and verify that it is not supported.
+  format.Format.nSamplesPerSec += 1;
+  EXPECT_FALSE(core_audio_utility::IsFormatSupported(
+      client.Get(), AUDCLNT_SHAREMODE_SHARED, &format));
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetDevicePeriod) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  // Verify that the device periods are valid for the default render and
+  // capture devices.
+  ComPtr<IAudioClient> client;
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    REFERENCE_TIME shared_time_period = 0;
+    REFERENCE_TIME exclusive_time_period = 0;
+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                              data_flow[i], eConsole);
+    EXPECT_TRUE(client.Get());
+    EXPECT_TRUE(SUCCEEDED(core_audio_utility::GetDevicePeriod(
+        client.Get(), AUDCLNT_SHAREMODE_SHARED, &shared_time_period)));
+    EXPECT_GT(shared_time_period, 0);
+    EXPECT_TRUE(SUCCEEDED(core_audio_utility::GetDevicePeriod(
+        client.Get(), AUDCLNT_SHAREMODE_EXCLUSIVE, &exclusive_time_period)));
+    EXPECT_GT(exclusive_time_period, 0);
+    EXPECT_LE(exclusive_time_period, shared_time_period);
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, GetPreferredAudioParameters) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  struct {
+    EDataFlow flow;
+    ERole role;
+  } data[] = {{eRender, eConsole},
+              {eRender, eCommunications},
+              {eCapture, eConsole},
+              {eCapture, eCommunications}};
+
+  // Verify that the preferred audio parameters are OK for all flow/role
+  // combinations above.
+  ComPtr<IAudioClient> client;
+  webrtc::AudioParameters params;
+  for (size_t i = 0; i < arraysize(data); ++i) {
+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                              data[i].flow, data[i].role);
+    EXPECT_TRUE(client.Get());
+    EXPECT_TRUE(SUCCEEDED(core_audio_utility::GetPreferredAudioParameters(
+        client.Get(), &params)));
+    EXPECT_TRUE(params.is_valid());
+    EXPECT_TRUE(params.is_complete());
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, SharedModeInitialize) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  ComPtr<IAudioClient> client;
+  client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                            eRender, eConsole);
+  EXPECT_TRUE(client.Get());
+
+  WAVEFORMATPCMEX format;
+  EXPECT_TRUE(SUCCEEDED(
+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
+
+  // Perform a shared-mode initialization without event-driven buffer handling.
+  uint32_t endpoint_buffer_size = 0;
+  HRESULT hr = core_audio_utility::SharedModeInitialize(
+      client.Get(), &format, nullptr, 0, false, &endpoint_buffer_size);
+  EXPECT_TRUE(SUCCEEDED(hr));
+  EXPECT_GT(endpoint_buffer_size, 0u);
+
+  // It is only possible to create a client once.
+  hr = core_audio_utility::SharedModeInitialize(
+      client.Get(), &format, nullptr, 0, false, &endpoint_buffer_size);
+  EXPECT_FALSE(SUCCEEDED(hr));
+  EXPECT_EQ(hr, AUDCLNT_E_ALREADY_INITIALIZED);
+
+  // Verify that it is possible to reinitialize the client after releasing it
+  // and then creating a new client.
+  client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                            eRender, eConsole);
+  EXPECT_TRUE(client.Get());
+  hr = core_audio_utility::SharedModeInitialize(
+      client.Get(), &format, nullptr, 0, false, &endpoint_buffer_size);
+  EXPECT_TRUE(SUCCEEDED(hr));
+  EXPECT_GT(endpoint_buffer_size, 0u);
+
+  // Use a non-supported format and verify that initialization fails.
+  // A simple way to emulate an invalid format is to use the shared-mode
+  // mixing format and modify the preferred sample rate.
+  client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                            eRender, eConsole);
+  EXPECT_TRUE(client.Get());
+  format.Format.nSamplesPerSec = format.Format.nSamplesPerSec + 1;
+  EXPECT_FALSE(core_audio_utility::IsFormatSupported(
+      client.Get(), AUDCLNT_SHAREMODE_SHARED, &format));
+  hr = core_audio_utility::SharedModeInitialize(
+      client.Get(), &format, nullptr, 0, false, &endpoint_buffer_size);
+  EXPECT_TRUE(FAILED(hr));
+  EXPECT_EQ(hr, E_INVALIDARG);
+
+  // Finally, perform a shared-mode initialization using event-driven buffer
+  // handling. The event handle will be signaled when an audio buffer is ready
+  // to be processed by the client (not verified here). The event handle should
+  // be in the non-signaled state.
+  ScopedHandle event_handle(::CreateEvent(nullptr, TRUE, FALSE, nullptr));
+  client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                            eRender, eConsole);
+  EXPECT_TRUE(client.Get());
+  EXPECT_TRUE(SUCCEEDED(
+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
+  EXPECT_TRUE(core_audio_utility::IsFormatSupported(
+      client.Get(), AUDCLNT_SHAREMODE_SHARED, &format));
+  hr = core_audio_utility::SharedModeInitialize(
+      client.Get(), &format, event_handle, 0, false, &endpoint_buffer_size);
+  EXPECT_TRUE(SUCCEEDED(hr));
+  EXPECT_GT(endpoint_buffer_size, 0u);
+
+  // TODO(henrika): possibly add test for signature which overrides the default
+  // sample rate.
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateRenderAndCaptureClients) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  WAVEFORMATPCMEX format;
+  uint32_t endpoint_buffer_size = 0;
+
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    ComPtr<IAudioClient> client;
+    ComPtr<IAudioRenderClient> render_client;
+    ComPtr<IAudioCaptureClient> capture_client;
+
+    // Create a default client for the given data-flow direction.
+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                              data_flow[i], eConsole);
+    EXPECT_TRUE(client.Get());
+    EXPECT_TRUE(SUCCEEDED(
+        core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
+    if (data_flow[i] == eRender) {
+      // It is not possible to create a render client using an unitialized
+      // client interface.
+      render_client = core_audio_utility::CreateRenderClient(client.Get());
+      EXPECT_FALSE(render_client.Get());
+
+      // Do a proper initialization and verify that it works this time.
+      core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
+                                               0, false, &endpoint_buffer_size);
+      render_client = core_audio_utility::CreateRenderClient(client.Get());
+      EXPECT_TRUE(render_client.Get());
+      EXPECT_GT(endpoint_buffer_size, 0u);
+    } else if (data_flow[i] == eCapture) {
+      // It is not possible to create a capture client using an unitialized
+      // client interface.
+      capture_client = core_audio_utility::CreateCaptureClient(client.Get());
+      EXPECT_FALSE(capture_client.Get());
+
+      // Do a proper initialization and verify that it works this time.
+      core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
+                                               0, false, &endpoint_buffer_size);
+      capture_client = core_audio_utility::CreateCaptureClient(client.Get());
+      EXPECT_TRUE(capture_client.Get());
+      EXPECT_GT(endpoint_buffer_size, 0u);
+    }
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateAudioClock) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  WAVEFORMATPCMEX format;
+  uint32_t endpoint_buffer_size = 0;
+
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    ComPtr<IAudioClient> client;
+    ComPtr<IAudioClock> audio_clock;
+
+    // Create a default client for the given data-flow direction.
+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                              data_flow[i], eConsole);
+    EXPECT_TRUE(client.Get());
+    EXPECT_TRUE(SUCCEEDED(
+        core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
+
+    // It is not possible to create an audio clock using an unitialized client
+    // interface.
+    audio_clock = core_audio_utility::CreateAudioClock(client.Get());
+    EXPECT_FALSE(audio_clock.Get());
+
+    // Do a proper initialization and verify that it works this time.
+    core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr, 0,
+                                             false, &endpoint_buffer_size);
+    audio_clock = core_audio_utility::CreateAudioClock(client.Get());
+    EXPECT_TRUE(audio_clock.Get());
+    EXPECT_GT(endpoint_buffer_size, 0u);
+
+    // Use the audio clock and verify that querying the device frequency works.
+    UINT64 frequency = 0;
+    EXPECT_TRUE(SUCCEEDED(audio_clock->GetFrequency(&frequency)));
+    EXPECT_GT(frequency, 0u);
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateAudioSessionControl) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  WAVEFORMATPCMEX format;
+  uint32_t endpoint_buffer_size = 0;
+
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    ComPtr<IAudioClient> client;
+    ComPtr<IAudioSessionControl> audio_session_control;
+
+    // Create a default client for the given data-flow direction.
+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                              data_flow[i], eConsole);
+    EXPECT_TRUE(client.Get());
+    EXPECT_TRUE(SUCCEEDED(
+        core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
+
+    // It is not possible to create an audio session control using an
+    // unitialized client interface.
+    audio_session_control =
+        core_audio_utility::CreateAudioSessionControl(client.Get());
+    EXPECT_FALSE(audio_session_control.Get());
+
+    // Do a proper initialization and verify that it works this time.
+    core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr, 0,
+                                             false, &endpoint_buffer_size);
+    audio_session_control =
+        core_audio_utility::CreateAudioSessionControl(client.Get());
+    EXPECT_TRUE(audio_session_control.Get());
+    EXPECT_GT(endpoint_buffer_size, 0u);
+
+    // Use the audio session control and verify that the session state can be
+    // queried. When a client opens a session by assigning the first stream to
+    // the session (by calling the IAudioClient::Initialize method), the initial
+    // session state is inactive. The session state changes from inactive to
+    // active when a stream in the session begins running (because the client
+    // has called the IAudioClient::Start method).
+    AudioSessionState state;
+    EXPECT_TRUE(SUCCEEDED(audio_session_control->GetState(&state)));
+    EXPECT_EQ(state, AudioSessionStateInactive);
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, CreateSimpleAudioVolume) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  EDataFlow data_flow[] = {eRender, eCapture};
+
+  WAVEFORMATPCMEX format;
+  uint32_t endpoint_buffer_size = 0;
+
+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
+    ComPtr<IAudioClient> client;
+    ComPtr<ISimpleAudioVolume> simple_audio_volume;
+
+    // Create a default client for the given data-flow direction.
+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
+                                              data_flow[i], eConsole);
+    EXPECT_TRUE(client.Get());
+    EXPECT_TRUE(SUCCEEDED(
+        core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
+
+    // It is not possible to create an audio volume using an uninitialized
+    // client interface.
+    simple_audio_volume =
+        core_audio_utility::CreateSimpleAudioVolume(client.Get());
+    EXPECT_FALSE(simple_audio_volume.Get());
+
+    // Do a proper initialization and verify that it works this time.
+    core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr, 0,
+                                             false, &endpoint_buffer_size);
+    simple_audio_volume =
+        core_audio_utility::CreateSimpleAudioVolume(client.Get());
+    EXPECT_TRUE(simple_audio_volume.Get());
+    EXPECT_GT(endpoint_buffer_size, 0u);
+
+    // Use the audio volume interface and validate that it works. The volume
+    // level should be value in the range 0.0 to 1.0 at first call.
+    float volume = 0.0;
+    EXPECT_TRUE(SUCCEEDED(simple_audio_volume->GetMasterVolume(&volume)));
+    EXPECT_GE(volume, 0.0);
+    EXPECT_LE(volume, 1.0);
+
+    // Next, set a new volume and verify that the setter does its job.
+    const float target_volume = 0.5;
+    EXPECT_TRUE(SUCCEEDED(
+        simple_audio_volume->SetMasterVolume(target_volume, nullptr)));
+    EXPECT_TRUE(SUCCEEDED(simple_audio_volume->GetMasterVolume(&volume)));
+    EXPECT_EQ(volume, target_volume);
+  }
+}
+
+TEST_F(CoreAudioUtilityWinTest, FillRenderEndpointBufferWithSilence) {
+  ABORT_TEST_IF_NOT(DevicesAvailable());
+
+  // Create default clients using the default mixing format for shared mode.
+  ComPtr<IAudioClient> client(core_audio_utility::CreateClient(
+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole));
+  EXPECT_TRUE(client.Get());
+
+  WAVEFORMATPCMEX format;
+  uint32_t endpoint_buffer_size = 0;
+  EXPECT_TRUE(SUCCEEDED(
+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
+  core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr, 0,
+                                           false, &endpoint_buffer_size);
+  EXPECT_GT(endpoint_buffer_size, 0u);
+
+  ComPtr<IAudioRenderClient> render_client(
+      core_audio_utility::CreateRenderClient(client.Get()));
+  EXPECT_TRUE(render_client.Get());
+
+  // The endpoint audio buffer should not be filled up by default after being
+  // created.
+  UINT32 num_queued_frames = 0;
+  client->GetCurrentPadding(&num_queued_frames);
+  EXPECT_EQ(num_queued_frames, 0u);
+
+  // Fill it up with zeros and verify that the buffer is full.
+  // It is not possible to verify that the actual data consists of zeros
+  // since we can't access data that has already been sent to the endpoint
+  // buffer.
+  EXPECT_TRUE(core_audio_utility::FillRenderEndpointBufferWithSilence(
+      client.Get(), render_client.Get()));
+  client->GetCurrentPadding(&num_queued_frames);
+  EXPECT_EQ(num_queued_frames, endpoint_buffer_size);
+}
+
+}  // namespace webrtc_win
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/BUILD.gn b/third_party/libwebrtc/modules/audio_mixer/BUILD.gn
new file mode 100644
index 0000000000..fe20f3d6c7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/BUILD.gn
@@ -0,0 +1,144 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+group("audio_mixer") {
+  deps = [
+    ":audio_frame_manipulator",
+    ":audio_mixer_impl",
+  ]
+}
+
+rtc_library("audio_mixer_impl") {
+  visibility = [ "*" ]
+  sources = [
+    "audio_mixer_impl.cc",
+    "audio_mixer_impl.h",
+    "default_output_rate_calculator.cc",
+    "default_output_rate_calculator.h",
+    "frame_combiner.cc",
+    "frame_combiner.h",
+    "output_rate_calculator.h",
+  ]
+
+  public = [
+    "audio_mixer_impl.h",
+    "default_output_rate_calculator.h",  # For creating a mixer with limiter
+                                         # disabled.
+    "frame_combiner.h",
+  ]
+
+  configs += [ "../audio_processing:apm_debug_dump" ]
+
+  deps = [
+    ":audio_frame_manipulator",
+    "../../api:array_view",
+    "../../api:rtp_packet_info",
+    "../../api:scoped_refptr",
+    "../../api/audio:audio_frame_api",
+    "../../api/audio:audio_mixer_api",
+    "../../audio/utility:audio_frame_operations",
+    "../../common_audio",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:race_checker",
+    "../../rtc_base:refcount",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
+    "../../system_wrappers:metrics",
+    "../audio_processing:api",
+    "../audio_processing:apm_logging",
+    "../audio_processing:audio_frame_view",
+    "../audio_processing/agc2:fixed_digital",
+  ]
+}
+
+rtc_library("audio_frame_manipulator") {
+  visibility = [
+    ":*",
+    "../../modules:*",
+  ]
+
+  sources = [
+    "audio_frame_manipulator.cc",
+    "audio_frame_manipulator.h",
+  ]
+
+  deps = [
+    "../../api/audio:audio_frame_api",
+    "../../audio/utility:audio_frame_operations",
+    "../../rtc_base:checks",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("audio_mixer_test_utils") {
+    testonly = true
+
+    sources = [
+      "gain_change_calculator.cc",
+      "gain_change_calculator.h",
+      "sine_wave_generator.cc",
+      "sine_wave_generator.h",
+    ]
+
+    deps = [
+      ":audio_frame_manipulator",
+      ":audio_mixer_impl",
+      "../../api:array_view",
+      "../../api/audio:audio_frame_api",
+      "../../rtc_base:checks",
+      "../../rtc_base:safe_conversions",
+    ]
+  }
+
+  rtc_library("audio_mixer_unittests") {
+    testonly = true
+
+    sources = [
+      "audio_frame_manipulator_unittest.cc",
+      "audio_mixer_impl_unittest.cc",
+      "frame_combiner_unittest.cc",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+    deps = [
+      ":audio_frame_manipulator",
+      ":audio_mixer_impl",
+      ":audio_mixer_test_utils",
+      "../../api:array_view",
+      "../../api:rtp_packet_info",
+      "../../api/audio:audio_mixer_api",
+      "../../api/units:timestamp",
+      "../../audio/utility:audio_frame_operations",
+      "../../rtc_base:checks",
+      "../../rtc_base:stringutils",
+      "../../rtc_base:task_queue_for_test",
+      "../../test:test_support",
+    ]
+  }
+
+  if (!build_with_chromium) {
+    rtc_executable("audio_mixer_test") {
+      testonly = true
+      sources = [ "audio_mixer_test.cc" ]
+
+      deps = [
+        ":audio_mixer_impl",
+        "../../api/audio:audio_mixer_api",
+        "../../common_audio",
+        "../../rtc_base:stringutils",
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/flags:parse",
+      ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_mixer/DEPS b/third_party/libwebrtc/modules/audio_mixer/DEPS
new file mode 100644
index 0000000000..46f29bccf8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/DEPS
@@ -0,0 +1,13 @@
+include_rules = [
+  "+audio/utility/audio_frame_operations.h",
+  "+audio/utility/channel_mixer.h",
+  "+call",
+  "+common_audio",
+  "+modules/audio_coding",
+  "+modules/audio_device",
+  "+modules/audio_processing",
+  "+modules/pacing",
+  "+modules/rtp_rtcp",
+  "+modules/utility",
+  "+system_wrappers",
+]
diff --git a/third_party/libwebrtc/modules/audio_mixer/OWNERS b/third_party/libwebrtc/modules/audio_mixer/OWNERS
new file mode 100644
index 0000000000..5edc304ab3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/OWNERS
@@ -0,0 +1,2 @@
+alessiob@webrtc.org
+henrik.lundin@webrtc.org
diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator.cc b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator.cc
new file mode 100644
index 0000000000..3100271cfb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator.cc
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_mixer/audio_frame_manipulator.h"
+
+#include "audio/utility/audio_frame_operations.h"
+#include "audio/utility/channel_mixer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
+  if (audio_frame.muted()) {
+    return 0;
+  }
+
+  uint32_t energy = 0;
+  const int16_t* frame_data = audio_frame.data();
+  for (size_t position = 0;
+       position < audio_frame.samples_per_channel_ * audio_frame.num_channels_;
+       position++) {
+    // TODO(aleloi): This can overflow. Convert to floats.
+    energy += frame_data[position] * frame_data[position];
+  }
+  return energy;
+}
+
+void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
+  RTC_DCHECK(audio_frame);
+  RTC_DCHECK_GE(start_gain, 0.0f);
+  RTC_DCHECK_GE(target_gain, 0.0f);
+  if (start_gain == target_gain || audio_frame->muted()) {
+    return;
+  }
+
+  size_t samples = audio_frame->samples_per_channel_;
+  RTC_DCHECK_LT(0, samples);
+  float increment = (target_gain - start_gain) / samples;
+  float gain = start_gain;
+  int16_t* frame_data = audio_frame->mutable_data();
+  for (size_t i = 0; i < samples; ++i) {
+    // If the audio is interleaved of several channels, we want to
+    // apply the same gain change to the ith sample of every channel.
+    for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
+      frame_data[audio_frame->num_channels_ * i + ch] *= gain;
+    }
+    gain += increment;
+  }
+}
+
+void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
+  RTC_DCHECK_GE(target_number_of_channels, 1);
+  // TODO(bugs.webrtc.org/10783): take channel layout into account as well.
+  if (frame->num_channels() == target_number_of_channels) {
+    return;
+  }
+
+  // Use legacy components for the most simple cases (mono <-> stereo) to ensure
+  // that native WebRTC clients are not affected when support for multi-channel
+  // audio is added to Chrome.
+  // TODO(bugs.webrtc.org/10783): utilize channel mixer for mono/stereo as well.
+  if (target_number_of_channels < 3 && frame->num_channels() < 3) {
+    if (frame->num_channels() > target_number_of_channels) {
+      AudioFrameOperations::DownmixChannels(target_number_of_channels, frame);
+    } else {
+      AudioFrameOperations::UpmixChannels(target_number_of_channels, frame);
+    }
+  } else {
+    // Use generic channel mixer when the number of channels for input our
+    // output is larger than two. E.g. stereo -> 5.1 channel up-mixing.
+    // TODO(bugs.webrtc.org/10783): ensure that actual channel layouts are used
+    // instead of guessing based on number of channels.
+    const ChannelLayout output_layout(
+        GuessChannelLayout(target_number_of_channels));
+    ChannelMixer mixer(GuessChannelLayout(frame->num_channels()),
+                       output_layout);
+    mixer.Transform(frame);
+    RTC_DCHECK_EQ(frame->channel_layout(), output_layout);
+  }
+  RTC_DCHECK_EQ(frame->num_channels(), target_number_of_channels)
+      << "Wrong number of channels, " << frame->num_channels() << " vs "
+      << target_number_of_channels;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator.h b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator.h
new file mode 100644
index 0000000000..ab3633d266
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
+#define MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+
+// Updates the audioFrame's energy (based on its samples).
+uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame);
+
+// Ramps up or down the provided audio frame. Ramp(0, 1, frame) will
+// linearly increase the samples in the frame from 0 to full volume.
+void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame);
+
+// Downmixes or upmixes a frame between stereo and mono.
+void RemixFrame(size_t target_number_of_channels, AudioFrame* frame);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_gn/moz.build b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_gn/moz.build
new file mode 100644
index 0000000000..c6a45f9a37
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_frame_manipulator_gn")
diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_unittest.cc b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_unittest.cc
new file mode 100644
index 0000000000..cfb3f2c230
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/audio_frame_manipulator_unittest.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_mixer/audio_frame_manipulator.h"
+
+#include <algorithm>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+void FillFrameWithConstants(size_t samples_per_channel,
+                            size_t number_of_channels,
+                            int16_t value,
+                            AudioFrame* frame) {
+  frame->num_channels_ = number_of_channels;
+  frame->samples_per_channel_ = samples_per_channel;
+  int16_t* frame_data = frame->mutable_data();
+  std::fill(frame_data, frame_data + samples_per_channel * number_of_channels,
+            value);
+}
+}  // namespace
+
+TEST(AudioFrameManipulator, CompareForwardRampWithExpectedResultStereo) {
+  constexpr int kSamplesPerChannel = 5;
+  constexpr int kNumberOfChannels = 2;
+
+  // Create a frame with values 5, 5, 5, ... and channels & samples as above.
+  AudioFrame frame;
+  FillFrameWithConstants(kSamplesPerChannel, kNumberOfChannels, 5, &frame);
+
+  Ramp(0.0f, 1.0f, &frame);
+
+  const int total_samples = kSamplesPerChannel * kNumberOfChannels;
+  const int16_t expected_result[total_samples] = {0, 0, 1, 1, 2, 2, 3, 3, 4, 4};
+  const int16_t* frame_data = frame.data();
+  EXPECT_TRUE(
+      std::equal(frame_data, frame_data + total_samples, expected_result));
+}
+
+TEST(AudioFrameManipulator, CompareBackwardRampWithExpectedResultMono) {
+  constexpr int kSamplesPerChannel = 5;
+  constexpr int kNumberOfChannels = 1;
+
+  // Create a frame with values 5, 5, 5, ... and channels & samples as above.
+  AudioFrame frame;
+  FillFrameWithConstants(kSamplesPerChannel, kNumberOfChannels, 5, &frame);
+
+  Ramp(1.0f, 0.0f, &frame);
+
+  const int total_samples = kSamplesPerChannel * kNumberOfChannels;
+  const int16_t expected_result[total_samples] = {5, 4, 3, 2, 1};
+  const int16_t* frame_data = frame.data();
+  EXPECT_TRUE(
+      std::equal(frame_data, frame_data + total_samples, expected_result));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl.cc b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl.cc
new file mode 100644
index 0000000000..0c203a1d9f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl.cc
@@ -0,0 +1,266 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_mixer/audio_mixer_impl.h"
+
+#include <stdint.h>
+
+#include <algorithm>
+#include <iterator>
+#include <type_traits>
+#include <utility>
+
+#include "modules/audio_mixer/audio_frame_manipulator.h"
+#include "modules/audio_mixer/default_output_rate_calculator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+struct AudioMixerImpl::SourceStatus {
+  SourceStatus(Source* audio_source, bool is_mixed, float gain)
+      : audio_source(audio_source), is_mixed(is_mixed), gain(gain) {}
+  Source* audio_source = nullptr;
+  bool is_mixed = false;
+  float gain = 0.0f;
+
+  // A frame that will be passed to audio_source->GetAudioFrameWithInfo.
+  AudioFrame audio_frame;
+};
+
+namespace {
+
+struct SourceFrame {
+  SourceFrame() = default;
+
+  SourceFrame(AudioMixerImpl::SourceStatus* source_status,
+              AudioFrame* audio_frame,
+              bool muted)
+      : source_status(source_status), audio_frame(audio_frame), muted(muted) {
+    RTC_DCHECK(source_status);
+    RTC_DCHECK(audio_frame);
+    if (!muted) {
+      energy = AudioMixerCalculateEnergy(*audio_frame);
+    }
+  }
+
+  SourceFrame(AudioMixerImpl::SourceStatus* source_status,
+              AudioFrame* audio_frame,
+              bool muted,
+              uint32_t energy)
+      : source_status(source_status),
+        audio_frame(audio_frame),
+        muted(muted),
+        energy(energy) {
+    RTC_DCHECK(source_status);
+    RTC_DCHECK(audio_frame);
+  }
+
+  AudioMixerImpl::SourceStatus* source_status = nullptr;
+  AudioFrame* audio_frame = nullptr;
+  bool muted = true;
+  uint32_t energy = 0;
+};
+
+// ShouldMixBefore(a, b) is used to select mixer sources.
+// Returns true if `a` is preferred over `b` as a source to be mixed.
+bool ShouldMixBefore(const SourceFrame& a, const SourceFrame& b) {
+  if (a.muted != b.muted) {
+    return b.muted;
+  }
+
+  const auto a_activity = a.audio_frame->vad_activity_;
+  const auto b_activity = b.audio_frame->vad_activity_;
+
+  if (a_activity != b_activity) {
+    return a_activity == AudioFrame::kVadActive;
+  }
+
+  return a.energy > b.energy;
+}
+
+void RampAndUpdateGain(
+    rtc::ArrayView<const SourceFrame> mixed_sources_and_frames) {
+  for (const auto& source_frame : mixed_sources_and_frames) {
+    float target_gain = source_frame.source_status->is_mixed ? 1.0f : 0.0f;
+    Ramp(source_frame.source_status->gain, target_gain,
+         source_frame.audio_frame);
+    source_frame.source_status->gain = target_gain;
+  }
+}
+
+std::vector<std::unique_ptr<AudioMixerImpl::SourceStatus>>::const_iterator
+FindSourceInList(
+    AudioMixerImpl::Source const* audio_source,
+    std::vector<std::unique_ptr<AudioMixerImpl::SourceStatus>> const*
+        audio_source_list) {
+  return std::find_if(
+      audio_source_list->begin(), audio_source_list->end(),
+      [audio_source](const std::unique_ptr<AudioMixerImpl::SourceStatus>& p) {
+        return p->audio_source == audio_source;
+      });
+}
+}  // namespace
+
+struct AudioMixerImpl::HelperContainers {
+  void resize(size_t size) {
+    audio_to_mix.resize(size);
+    audio_source_mixing_data_list.resize(size);
+    ramp_list.resize(size);
+    preferred_rates.resize(size);
+  }
+
+  std::vector<AudioFrame*> audio_to_mix;
+  std::vector<SourceFrame> audio_source_mixing_data_list;
+  std::vector<SourceFrame> ramp_list;
+  std::vector<int> preferred_rates;
+};
+
+AudioMixerImpl::AudioMixerImpl(
+    std::unique_ptr<OutputRateCalculator> output_rate_calculator,
+    bool use_limiter,
+    int max_sources_to_mix)
+    : max_sources_to_mix_(max_sources_to_mix),
+      output_rate_calculator_(std::move(output_rate_calculator)),
+      audio_source_list_(),
+      helper_containers_(std::make_unique<HelperContainers>()),
+      frame_combiner_(use_limiter) {
+  RTC_CHECK_GE(max_sources_to_mix, 1) << "At least one source must be mixed";
+  audio_source_list_.reserve(max_sources_to_mix);
+  helper_containers_->resize(max_sources_to_mix);
+}
+
+AudioMixerImpl::~AudioMixerImpl() {}
+
+rtc::scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create(
+    int max_sources_to_mix) {
+  return Create(std::unique_ptr<DefaultOutputRateCalculator>(
+                    new DefaultOutputRateCalculator()),
+                /*use_limiter=*/true, max_sources_to_mix);
+}
+
+rtc::scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create(
+    std::unique_ptr<OutputRateCalculator> output_rate_calculator,
+    bool use_limiter,
+    int max_sources_to_mix) {
+  return rtc::make_ref_counted<AudioMixerImpl>(
+      std::move(output_rate_calculator), use_limiter, max_sources_to_mix);
+}
+
+void AudioMixerImpl::Mix(size_t number_of_channels,
+                         AudioFrame* audio_frame_for_mixing) {
+  TRACE_EVENT0("webrtc", "AudioMixerImpl::Mix");
+  RTC_DCHECK(number_of_channels >= 1);
+  MutexLock lock(&mutex_);
+
+  size_t number_of_streams = audio_source_list_.size();
+
+  std::transform(audio_source_list_.begin(), audio_source_list_.end(),
+                 helper_containers_->preferred_rates.begin(),
+                 [&](std::unique_ptr<SourceStatus>& a) {
+                   return a->audio_source->PreferredSampleRate();
+                 });
+
+  int output_frequency = output_rate_calculator_->CalculateOutputRateFromRange(
+      rtc::ArrayView<const int>(helper_containers_->preferred_rates.data(),
+                                number_of_streams));
+
+  frame_combiner_.Combine(GetAudioFromSources(output_frequency),
+                          number_of_channels, output_frequency,
+                          number_of_streams, audio_frame_for_mixing);
+}
+
+bool AudioMixerImpl::AddSource(Source* audio_source) {
+  RTC_DCHECK(audio_source);
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(FindSourceInList(audio_source, &audio_source_list_) ==
+             audio_source_list_.end())
+      << "Source already added to mixer";
+  audio_source_list_.emplace_back(new SourceStatus(audio_source, false, 0));
+  helper_containers_->resize(audio_source_list_.size());
+  return true;
+}
+
+void AudioMixerImpl::RemoveSource(Source* audio_source) {
+  RTC_DCHECK(audio_source);
+  MutexLock lock(&mutex_);
+  const auto iter = FindSourceInList(audio_source, &audio_source_list_);
+  RTC_DCHECK(iter != audio_source_list_.end()) << "Source not present in mixer";
+  audio_source_list_.erase(iter);
+}
+
+rtc::ArrayView<AudioFrame* const> AudioMixerImpl::GetAudioFromSources(
+    int output_frequency) {
+  // Get audio from the audio sources and put it in the SourceFrame vector.
+  int audio_source_mixing_data_count = 0;
+  for (auto& source_and_status : audio_source_list_) {
+    const auto audio_frame_info =
+        source_and_status->audio_source->GetAudioFrameWithInfo(
+            output_frequency, &source_and_status->audio_frame);
+
+    if (audio_frame_info == Source::AudioFrameInfo::kError) {
+      RTC_LOG_F(LS_WARNING) << "failed to GetAudioFrameWithInfo() from source";
+      continue;
+    }
+    helper_containers_
+        ->audio_source_mixing_data_list[audio_source_mixing_data_count++] =
+        SourceFrame(source_and_status.get(), &source_and_status->audio_frame,
+                    audio_frame_info == Source::AudioFrameInfo::kMuted);
+  }
+  rtc::ArrayView<SourceFrame> audio_source_mixing_data_view(
+      helper_containers_->audio_source_mixing_data_list.data(),
+      audio_source_mixing_data_count);
+
+  // Sort frames by sorting function.
+  std::sort(audio_source_mixing_data_view.begin(),
+            audio_source_mixing_data_view.end(), ShouldMixBefore);
+
+  int max_audio_frame_counter = max_sources_to_mix_;
+  int ramp_list_lengh = 0;
+  int audio_to_mix_count = 0;
+  // Go through list in order and put unmuted frames in result list.
+  for (const auto& p : audio_source_mixing_data_view) {
+    // Filter muted.
+    if (p.muted) {
+      p.source_status->is_mixed = false;
+      continue;
+    }
+
+    // Add frame to result vector for mixing.
+    bool is_mixed = false;
+    if (max_audio_frame_counter > 0) {
+      --max_audio_frame_counter;
+      helper_containers_->audio_to_mix[audio_to_mix_count++] = p.audio_frame;
+      helper_containers_->ramp_list[ramp_list_lengh++] =
+          SourceFrame(p.source_status, p.audio_frame, false, -1);
+      is_mixed = true;
+    }
+    p.source_status->is_mixed = is_mixed;
+  }
+  RampAndUpdateGain(rtc::ArrayView<SourceFrame>(
+      helper_containers_->ramp_list.data(), ramp_list_lengh));
+  return rtc::ArrayView<AudioFrame* const>(
+      helper_containers_->audio_to_mix.data(), audio_to_mix_count);
+}
+
+bool AudioMixerImpl::GetAudioSourceMixabilityStatusForTest(
+    AudioMixerImpl::Source* audio_source) const {
+  MutexLock lock(&mutex_);
+
+  const auto iter = FindSourceInList(audio_source, &audio_source_list_);
+  if (iter != audio_source_list_.end()) {
+    return (*iter)->is_mixed;
+  }
+
+  RTC_LOG(LS_ERROR) << "Audio source unknown";
+  return false;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl.h b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl.h
new file mode 100644
index 0000000000..76b1131777
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
+#define MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio/audio_mixer.h"
+#include "api/scoped_refptr.h"
+#include "modules/audio_mixer/frame_combiner.h"
+#include "modules/audio_mixer/output_rate_calculator.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioMixerImpl : public AudioMixer {
+ public:
+  struct SourceStatus;
+
+  // AudioProcessing only accepts 10 ms frames.
+  static const int kFrameDurationInMs = 10;
+
+  static const int kDefaultNumberOfMixedAudioSources = 3;
+
+  static rtc::scoped_refptr<AudioMixerImpl> Create(
+      int max_sources_to_mix = kDefaultNumberOfMixedAudioSources);
+
+  static rtc::scoped_refptr<AudioMixerImpl> Create(
+      std::unique_ptr<OutputRateCalculator> output_rate_calculator,
+      bool use_limiter,
+      int max_sources_to_mix = kDefaultNumberOfMixedAudioSources);
+
+  ~AudioMixerImpl() override;
+
+  AudioMixerImpl(const AudioMixerImpl&) = delete;
+  AudioMixerImpl& operator=(const AudioMixerImpl&) = delete;
+
+  // AudioMixer functions
+  bool AddSource(Source* audio_source) override;
+  void RemoveSource(Source* audio_source) override;
+
+  void Mix(size_t number_of_channels,
+           AudioFrame* audio_frame_for_mixing) override
+      RTC_LOCKS_EXCLUDED(mutex_);
+
+  // Returns true if the source was mixed last round. Returns
+  // false and logs an error if the source was never added to the
+  // mixer.
+  bool GetAudioSourceMixabilityStatusForTest(Source* audio_source) const;
+
+ protected:
+  AudioMixerImpl(std::unique_ptr<OutputRateCalculator> output_rate_calculator,
+                 bool use_limiter,
+                 int max_sources_to_mix);
+
+ private:
+  struct HelperContainers;
+
+  // Compute what audio sources to mix from audio_source_list_. Ramp
+  // in and out. Update mixed status. Mixes up to
+  // kMaximumAmountOfMixedAudioSources audio sources.
+  rtc::ArrayView<AudioFrame* const> GetAudioFromSources(int output_frequency)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // The critical section lock guards audio source insertion and
+  // removal, which can be done from any thread. The race checker
+  // checks that mixing is done sequentially.
+  mutable Mutex mutex_;
+
+  const int max_sources_to_mix_;
+
+  std::unique_ptr<OutputRateCalculator> output_rate_calculator_;
+
+  // List of all audio sources.
+  std::vector<std::unique_ptr<SourceStatus>> audio_source_list_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<HelperContainers> helper_containers_
+      RTC_GUARDED_BY(mutex_);
+
+  // Component that handles actual adding of audio frames.
+  FrameCombiner frame_combiner_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_MIXER_AUDIO_MIXER_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_gn/moz.build b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_gn/moz.build
new file mode 100644
index 0000000000..32a47e6b06
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_gn/moz.build
@@ -0,0 +1,235 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl.cc",
+    "/third_party/libwebrtc/modules/audio_mixer/default_output_rate_calculator.cc",
+    "/third_party/libwebrtc/modules/audio_mixer/frame_combiner.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_mixer_impl_gn")
diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
new file mode 100644
index 0000000000..e4b12a7000
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
@@ -0,0 +1,790 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_mixer/audio_mixer_impl.h"
+
+#include <string.h>
+
+#include <cstdint>
+#include <limits>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio/audio_mixer.h"
+#include "api/rtp_packet_info.h"
+#include "api/rtp_packet_infos.h"
+#include "api/units/timestamp.h"
+#include "modules/audio_mixer/default_output_rate_calculator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::Exactly;
+using ::testing::Invoke;
+using ::testing::Return;
+using ::testing::UnorderedElementsAre;
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDefaultSampleRateHz = 48000;
+
+// Utility function that resets the frame member variables with
+// sensible defaults.
+void ResetFrame(AudioFrame* frame) {
+  frame->sample_rate_hz_ = kDefaultSampleRateHz;
+  frame->num_channels_ = 1;
+
+  // Frame duration 10ms.
+  frame->samples_per_channel_ = kDefaultSampleRateHz / 100;
+  frame->vad_activity_ = AudioFrame::kVadActive;
+  frame->speech_type_ = AudioFrame::kNormalSpeech;
+}
+
+std::string ProduceDebugText(int sample_rate_hz,
+                             int number_of_channels,
+                             int number_of_sources) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz << " ";
+  ss << "Number of channels: " << number_of_channels << " ";
+  ss << "Number of sources: " << number_of_sources;
+  return ss.Release();
+}
+
+AudioFrame frame_for_mixing;
+
+}  // namespace
+
+class MockMixerAudioSource : public ::testing::NiceMock<AudioMixer::Source> {
+ public:
+  MockMixerAudioSource()
+      : fake_audio_frame_info_(AudioMixer::Source::AudioFrameInfo::kNormal) {
+    ON_CALL(*this, GetAudioFrameWithInfo(_, _))
+        .WillByDefault(
+            Invoke(this, &MockMixerAudioSource::FakeAudioFrameWithInfo));
+    ON_CALL(*this, PreferredSampleRate())
+        .WillByDefault(Return(kDefaultSampleRateHz));
+  }
+
+  MOCK_METHOD(AudioFrameInfo,
+              GetAudioFrameWithInfo,
+              (int sample_rate_hz, AudioFrame* audio_frame),
+              (override));
+
+  MOCK_METHOD(int, PreferredSampleRate, (), (const, override));
+  MOCK_METHOD(int, Ssrc, (), (const, override));
+
+  AudioFrame* fake_frame() { return &fake_frame_; }
+  AudioFrameInfo fake_info() { return fake_audio_frame_info_; }
+  void set_fake_info(const AudioFrameInfo audio_frame_info) {
+    fake_audio_frame_info_ = audio_frame_info;
+  }
+
+  void set_packet_infos(const RtpPacketInfos& packet_infos) {
+    packet_infos_ = packet_infos;
+  }
+
+ private:
+  AudioFrameInfo FakeAudioFrameWithInfo(int sample_rate_hz,
+                                        AudioFrame* audio_frame) {
+    audio_frame->CopyFrom(fake_frame_);
+    audio_frame->sample_rate_hz_ = sample_rate_hz;
+    audio_frame->samples_per_channel_ =
+        rtc::CheckedDivExact(sample_rate_hz, 100);
+    audio_frame->packet_infos_ = packet_infos_;
+    return fake_info();
+  }
+
+  AudioFrame fake_frame_;
+  AudioFrameInfo fake_audio_frame_info_;
+  RtpPacketInfos packet_infos_;
+};
+
+class CustomRateCalculator : public OutputRateCalculator {
+ public:
+  explicit CustomRateCalculator(int rate) : rate_(rate) {}
+  int CalculateOutputRateFromRange(
+      rtc::ArrayView<const int> preferred_rates) override {
+    return rate_;
+  }
+
+ private:
+  const int rate_;
+};
+
+// Creates participants from `frames` and `frame_info` and adds them
+// to the mixer. Compares mixed status with `expected_status`
+void MixAndCompare(
+    const std::vector<AudioFrame>& frames,
+    const std::vector<AudioMixer::Source::AudioFrameInfo>& frame_info,
+    const std::vector<bool>& expected_status) {
+  const size_t num_audio_sources = frames.size();
+  RTC_DCHECK(frames.size() == frame_info.size());
+  RTC_DCHECK(frame_info.size() == expected_status.size());
+
+  const auto mixer = AudioMixerImpl::Create();
+  std::vector<MockMixerAudioSource> participants(num_audio_sources);
+
+  for (size_t i = 0; i < num_audio_sources; ++i) {
+    participants[i].fake_frame()->CopyFrom(frames[i]);
+    participants[i].set_fake_info(frame_info[i]);
+  }
+
+  for (size_t i = 0; i < num_audio_sources; ++i) {
+    EXPECT_TRUE(mixer->AddSource(&participants[i]));
+    EXPECT_CALL(participants[i], GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
+        .Times(Exactly(1));
+  }
+
+  mixer->Mix(1, &frame_for_mixing);
+
+  for (size_t i = 0; i < num_audio_sources; ++i) {
+    EXPECT_EQ(expected_status[i],
+              mixer->GetAudioSourceMixabilityStatusForTest(&participants[i]))
+        << "Mixed status of AudioSource #" << i << " wrong.";
+  }
+}
+
+void MixMonoAtGivenNativeRate(int native_sample_rate,
+                              AudioFrame* mix_frame,
+                              rtc::scoped_refptr<AudioMixer> mixer,
+                              MockMixerAudioSource* audio_source) {
+  ON_CALL(*audio_source, PreferredSampleRate())
+      .WillByDefault(Return(native_sample_rate));
+  audio_source->fake_frame()->sample_rate_hz_ = native_sample_rate;
+  audio_source->fake_frame()->samples_per_channel_ = native_sample_rate / 100;
+
+  mixer->Mix(1, mix_frame);
+}
+
+TEST(AudioMixer, LargestEnergyVadActiveMixed) {
+  constexpr int kAudioSources =
+      AudioMixerImpl::kDefaultNumberOfMixedAudioSources + 3;
+
+  const auto mixer = AudioMixerImpl::Create();
+
+  MockMixerAudioSource participants[kAudioSources];
+
+  for (int i = 0; i < kAudioSources; ++i) {
+    ResetFrame(participants[i].fake_frame());
+
+    // We set the 80-th sample value since the first 80 samples may be
+    // modified by a ramped-in window.
+    participants[i].fake_frame()->mutable_data()[80] = i;
+
+    EXPECT_TRUE(mixer->AddSource(&participants[i]));
+    EXPECT_CALL(participants[i], GetAudioFrameWithInfo(_, _)).Times(Exactly(1));
+  }
+
+  // Last participant gives audio frame with passive VAD, although it has the
+  // largest energy.
+  participants[kAudioSources - 1].fake_frame()->vad_activity_ =
+      AudioFrame::kVadPassive;
+
+  AudioFrame audio_frame;
+  mixer->Mix(1,  // number of channels
+             &audio_frame);
+
+  for (int i = 0; i < kAudioSources; ++i) {
+    bool is_mixed =
+        mixer->GetAudioSourceMixabilityStatusForTest(&participants[i]);
+    if (i == kAudioSources - 1 ||
+        i < kAudioSources - 1 -
+                AudioMixerImpl::kDefaultNumberOfMixedAudioSources) {
+      EXPECT_FALSE(is_mixed)
+          << "Mixing status of AudioSource #" << i << " wrong.";
+    } else {
+      EXPECT_TRUE(is_mixed)
+          << "Mixing status of AudioSource #" << i << " wrong.";
+    }
+  }
+}
+
+TEST(AudioMixer, FrameNotModifiedForSingleParticipant) {
+  const auto mixer = AudioMixerImpl::Create();
+
+  MockMixerAudioSource participant;
+
+  ResetFrame(participant.fake_frame());
+  const size_t n_samples = participant.fake_frame()->samples_per_channel_;
+
+  // Modify the frame so that it's not zero.
+  int16_t* fake_frame_data = participant.fake_frame()->mutable_data();
+  for (size_t j = 0; j < n_samples; ++j) {
+    fake_frame_data[j] = static_cast<int16_t>(j);
+  }
+
+  EXPECT_TRUE(mixer->AddSource(&participant));
+  EXPECT_CALL(participant, GetAudioFrameWithInfo(_, _)).Times(Exactly(2));
+
+  AudioFrame audio_frame;
+  // Two mix iteration to compare after the ramp-up step.
+  for (int i = 0; i < 2; ++i) {
+    mixer->Mix(1,  // number of channels
+               &audio_frame);
+  }
+
+  EXPECT_EQ(0, memcmp(participant.fake_frame()->data(), audio_frame.data(),
+                      n_samples));
+}
+
+TEST(AudioMixer, SourceAtNativeRateShouldNeverResample) {
+  const auto mixer = AudioMixerImpl::Create();
+
+  MockMixerAudioSource audio_source;
+  ResetFrame(audio_source.fake_frame());
+
+  mixer->AddSource(&audio_source);
+
+  for (auto frequency : {8000, 16000, 32000, 48000}) {
+    EXPECT_CALL(audio_source, GetAudioFrameWithInfo(frequency, _))
+        .Times(Exactly(1));
+
+    MixMonoAtGivenNativeRate(frequency, &frame_for_mixing, mixer,
+                             &audio_source);
+  }
+}
+
+TEST(AudioMixer, MixerShouldMixAtNativeSourceRate) {
+  const auto mixer = AudioMixerImpl::Create();
+
+  MockMixerAudioSource audio_source;
+  ResetFrame(audio_source.fake_frame());
+
+  mixer->AddSource(&audio_source);
+
+  for (auto frequency : {8000, 16000, 32000, 48000}) {
+    MixMonoAtGivenNativeRate(frequency, &frame_for_mixing, mixer,
+                             &audio_source);
+
+    EXPECT_EQ(frequency, frame_for_mixing.sample_rate_hz_);
+  }
+}
+
+TEST(AudioMixer, MixerShouldAlwaysMixAtNativeRate) {
+  const auto mixer = AudioMixerImpl::Create();
+
+  MockMixerAudioSource participant;
+  ResetFrame(participant.fake_frame());
+  mixer->AddSource(&participant);
+
+  const int needed_frequency = 44100;
+  ON_CALL(participant, PreferredSampleRate())
+      .WillByDefault(Return(needed_frequency));
+
+  // We expect mixing frequency to be native and >= needed_frequency.
+  const int expected_mix_frequency = 48000;
+  EXPECT_CALL(participant, GetAudioFrameWithInfo(expected_mix_frequency, _))
+      .Times(Exactly(1));
+  participant.fake_frame()->sample_rate_hz_ = expected_mix_frequency;
+  participant.fake_frame()->samples_per_channel_ = expected_mix_frequency / 100;
+
+  mixer->Mix(1, &frame_for_mixing);
+
+  EXPECT_EQ(48000, frame_for_mixing.sample_rate_hz_);
+}
+
+// Check that the mixing rate is always >= participants preferred rate.
+TEST(AudioMixer, ShouldNotCauseQualityLossForMultipleSources) {
+  const auto mixer = AudioMixerImpl::Create();
+
+  std::vector<MockMixerAudioSource> audio_sources(2);
+  const std::vector<int> source_sample_rates = {8000, 16000};
+  for (int i = 0; i < 2; ++i) {
+    auto& source = audio_sources[i];
+    ResetFrame(source.fake_frame());
+    mixer->AddSource(&source);
+    const auto sample_rate = source_sample_rates[i];
+    EXPECT_CALL(source, PreferredSampleRate()).WillOnce(Return(sample_rate));
+
+    EXPECT_CALL(source, GetAudioFrameWithInfo(::testing::Ge(sample_rate), _));
+  }
+  mixer->Mix(1, &frame_for_mixing);
+}
+
+TEST(AudioMixer, ParticipantNumberOfChannels) {
+  const auto mixer = AudioMixerImpl::Create();
+
+  MockMixerAudioSource participant;
+  ResetFrame(participant.fake_frame());
+
+  EXPECT_TRUE(mixer->AddSource(&participant));
+  for (size_t number_of_channels : {1, 2}) {
+    EXPECT_CALL(participant, GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
+        .Times(Exactly(1));
+    mixer->Mix(number_of_channels, &frame_for_mixing);
+    EXPECT_EQ(number_of_channels, frame_for_mixing.num_channels_);
+  }
+}
+
+// Maximal amount of participants are mixed one iteration, then
+// another participant with higher energy is added.
+TEST(AudioMixer, RampedOutSourcesShouldNotBeMarkedMixed) {
+  constexpr int kAudioSources =
+      AudioMixerImpl::kDefaultNumberOfMixedAudioSources + 1;
+
+  const auto mixer = AudioMixerImpl::Create();
+  MockMixerAudioSource participants[kAudioSources];
+
+  for (int i = 0; i < kAudioSources; ++i) {
+    ResetFrame(participants[i].fake_frame());
+    // Set the participant audio energy to increase with the index
+    // `i`.
+    participants[i].fake_frame()->mutable_data()[0] = 100 * i;
+  }
+
+  // Add all participants but the loudest for mixing.
+  for (int i = 0; i < kAudioSources - 1; ++i) {
+    EXPECT_TRUE(mixer->AddSource(&participants[i]));
+    EXPECT_CALL(participants[i], GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
+        .Times(Exactly(1));
+  }
+
+  // First mixer iteration
+  mixer->Mix(1, &frame_for_mixing);
+
+  // All participants but the loudest should have been mixed.
+  for (int i = 0; i < kAudioSources - 1; ++i) {
+    EXPECT_TRUE(mixer->GetAudioSourceMixabilityStatusForTest(&participants[i]))
+        << "Mixed status of AudioSource #" << i << " wrong.";
+  }
+
+  // Add new participant with higher energy.
+  EXPECT_TRUE(mixer->AddSource(&participants[kAudioSources - 1]));
+  for (int i = 0; i < kAudioSources; ++i) {
+    EXPECT_CALL(participants[i], GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
+        .Times(Exactly(1));
+  }
+
+  mixer->Mix(1, &frame_for_mixing);
+
+  // The most quiet participant should not have been mixed.
+  EXPECT_FALSE(mixer->GetAudioSourceMixabilityStatusForTest(&participants[0]))
+      << "Mixed status of AudioSource #0 wrong.";
+
+  // The loudest participants should have been mixed.
+  for (int i = 1; i < kAudioSources; ++i) {
+    EXPECT_EQ(true,
+              mixer->GetAudioSourceMixabilityStatusForTest(&participants[i]))
+        << "Mixed status of AudioSource #" << i << " wrong.";
+  }
+}
+
+// This test checks that the initialization and participant addition
+// can be done on a different thread.
+TEST(AudioMixer, ConstructFromOtherThread) {
+  TaskQueueForTest init_queue("init");
+  rtc::scoped_refptr<AudioMixer> mixer;
+  init_queue.SendTask([&mixer]() { mixer = AudioMixerImpl::Create(); });
+
+  MockMixerAudioSource participant;
+  EXPECT_CALL(participant, PreferredSampleRate())
+      .WillRepeatedly(Return(kDefaultSampleRateHz));
+
+  ResetFrame(participant.fake_frame());
+
+  TaskQueueForTest participant_queue("participant");
+  participant_queue.SendTask(
+      [&mixer, &participant]() { mixer->AddSource(&participant); });
+
+  EXPECT_CALL(participant, GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
+      .Times(Exactly(1));
+
+  // Do one mixer iteration
+  mixer->Mix(1, &frame_for_mixing);
+}
+
+TEST(AudioMixer, MutedShouldMixAfterUnmuted) {
+  constexpr int kAudioSources =
+      AudioMixerImpl::kDefaultNumberOfMixedAudioSources + 1;
+
+  std::vector<AudioFrame> frames(kAudioSources);
+  for (auto& frame : frames) {
+    ResetFrame(&frame);
+  }
+
+  std::vector<AudioMixer::Source::AudioFrameInfo> frame_info(
+      kAudioSources, AudioMixer::Source::AudioFrameInfo::kNormal);
+  frame_info[0] = AudioMixer::Source::AudioFrameInfo::kMuted;
+  std::vector<bool> expected_status(kAudioSources, true);
+  expected_status[0] = false;
+
+  MixAndCompare(frames, frame_info, expected_status);
+}
+
+TEST(AudioMixer, PassiveShouldMixAfterNormal) {
+  constexpr int kAudioSources =
+      AudioMixerImpl::kDefaultNumberOfMixedAudioSources + 1;
+
+  std::vector<AudioFrame> frames(kAudioSources);
+  for (auto& frame : frames) {
+    ResetFrame(&frame);
+  }
+
+  std::vector<AudioMixer::Source::AudioFrameInfo> frame_info(
+      kAudioSources, AudioMixer::Source::AudioFrameInfo::kNormal);
+  frames[0].vad_activity_ = AudioFrame::kVadPassive;
+  std::vector<bool> expected_status(kAudioSources, true);
+  expected_status[0] = false;
+
+  MixAndCompare(frames, frame_info, expected_status);
+}
+
+TEST(AudioMixer, ActiveShouldMixBeforeLoud) {
+  constexpr int kAudioSources =
+      AudioMixerImpl::kDefaultNumberOfMixedAudioSources + 1;
+
+  std::vector<AudioFrame> frames(kAudioSources);
+  for (auto& frame : frames) {
+    ResetFrame(&frame);
+  }
+
+  std::vector<AudioMixer::Source::AudioFrameInfo> frame_info(
+      kAudioSources, AudioMixer::Source::AudioFrameInfo::kNormal);
+  frames[0].vad_activity_ = AudioFrame::kVadPassive;
+  int16_t* frame_data = frames[0].mutable_data();
+  std::fill(frame_data, frame_data + kDefaultSampleRateHz / 100,
+            std::numeric_limits<int16_t>::max());
+  std::vector<bool> expected_status(kAudioSources, true);
+  expected_status[0] = false;
+
+  MixAndCompare(frames, frame_info, expected_status);
+}
+
+TEST(AudioMixer, ShouldMixUpToSpecifiedNumberOfSourcesToMix) {
+  constexpr int kAudioSources = 5;
+  constexpr int kSourcesToMix = 2;
+
+  std::vector<AudioFrame> frames(kAudioSources);
+  for (auto& frame : frames) {
+    ResetFrame(&frame);
+  }
+
+  std::vector<AudioMixer::Source::AudioFrameInfo> frame_info(
+      kAudioSources, AudioMixer::Source::AudioFrameInfo::kNormal);
+  // Set up to kSourceToMix sources with kVadActive so that they're mixed.
+  const std::vector<AudioFrame::VADActivity> kVadActivities = {
+      AudioFrame::kVadUnknown, AudioFrame::kVadPassive, AudioFrame::kVadPassive,
+      AudioFrame::kVadActive, AudioFrame::kVadActive};
+  // Populate VAD and frame for all sources.
+  for (int i = 0; i < kAudioSources; i++) {
+    frames[i].vad_activity_ = kVadActivities[i];
+  }
+
+  std::vector<MockMixerAudioSource> participants(kAudioSources);
+  for (int i = 0; i < kAudioSources; ++i) {
+    participants[i].fake_frame()->CopyFrom(frames[i]);
+    participants[i].set_fake_info(frame_info[i]);
+  }
+
+  const auto mixer = AudioMixerImpl::Create(kSourcesToMix);
+  for (int i = 0; i < kAudioSources; ++i) {
+    EXPECT_TRUE(mixer->AddSource(&participants[i]));
+    EXPECT_CALL(participants[i], GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
+        .Times(Exactly(1));
+  }
+
+  mixer->Mix(1, &frame_for_mixing);
+
+  std::vector<bool> expected_status = {false, false, false, true, true};
+  for (int i = 0; i < kAudioSources; ++i) {
+    EXPECT_EQ(expected_status[i],
+              mixer->GetAudioSourceMixabilityStatusForTest(&participants[i]))
+        << "Wrong mix status for source #" << i << " is wrong";
+  }
+}
+
+TEST(AudioMixer, UnmutedShouldMixBeforeLoud) {
+  constexpr int kAudioSources =
+      AudioMixerImpl::kDefaultNumberOfMixedAudioSources + 1;
+
+  std::vector<AudioFrame> frames(kAudioSources);
+  for (auto& frame : frames) {
+    ResetFrame(&frame);
+  }
+
+  std::vector<AudioMixer::Source::AudioFrameInfo> frame_info(
+      kAudioSources, AudioMixer::Source::AudioFrameInfo::kNormal);
+  frame_info[0] = AudioMixer::Source::AudioFrameInfo::kMuted;
+  int16_t* frame_data = frames[0].mutable_data();
+  std::fill(frame_data, frame_data + kDefaultSampleRateHz / 100,
+            std::numeric_limits<int16_t>::max());
+  std::vector<bool> expected_status(kAudioSources, true);
+  expected_status[0] = false;
+
+  MixAndCompare(frames, frame_info, expected_status);
+}
+
+TEST(AudioMixer, MixingRateShouldBeDecidedByRateCalculator) {
+  constexpr int kOutputRate = 22000;
+  const auto mixer =
+      AudioMixerImpl::Create(std::unique_ptr<OutputRateCalculator>(
+                                 new CustomRateCalculator(kOutputRate)),
+                             true);
+  MockMixerAudioSource audio_source;
+  mixer->AddSource(&audio_source);
+  ResetFrame(audio_source.fake_frame());
+
+  EXPECT_CALL(audio_source, GetAudioFrameWithInfo(kOutputRate, _))
+      .Times(Exactly(1));
+
+  mixer->Mix(1, &frame_for_mixing);
+}
+
+TEST(AudioMixer, ZeroSourceRateShouldBeDecidedByRateCalculator) {
+  constexpr int kOutputRate = 8000;
+  const auto mixer =
+      AudioMixerImpl::Create(std::unique_ptr<OutputRateCalculator>(
+                                 new CustomRateCalculator(kOutputRate)),
+                             true);
+
+  mixer->Mix(1, &frame_for_mixing);
+
+  EXPECT_EQ(kOutputRate, frame_for_mixing.sample_rate_hz_);
+}
+
+TEST(AudioMixer, NoLimiterBasicApiCalls) {
+  const auto mixer = AudioMixerImpl::Create(
+      std::unique_ptr<OutputRateCalculator>(new DefaultOutputRateCalculator()),
+      false);
+  mixer->Mix(1, &frame_for_mixing);
+}
+
+TEST(AudioMixer, AnyRateIsPossibleWithNoLimiter) {
+  // No APM limiter means no AudioProcessing::NativeRate restriction
+  // on mixing rate. The rate has to be divisible by 100 since we use
+  // 10 ms frames, though.
+  for (const auto rate : {8000, 20000, 24000, 32000, 44100}) {
+    for (const size_t number_of_channels : {1, 2}) {
+      for (const auto number_of_sources : {0, 1, 2, 3, 4}) {
+        SCOPED_TRACE(
+            ProduceDebugText(rate, number_of_sources, number_of_sources));
+        const auto mixer =
+            AudioMixerImpl::Create(std::unique_ptr<OutputRateCalculator>(
+                                       new CustomRateCalculator(rate)),
+                                   false);
+
+        std::vector<MockMixerAudioSource> sources(number_of_sources);
+        for (auto& source : sources) {
+          ResetFrame(source.fake_frame());
+          mixer->AddSource(&source);
+        }
+
+        mixer->Mix(number_of_channels, &frame_for_mixing);
+        EXPECT_EQ(rate, frame_for_mixing.sample_rate_hz_);
+        EXPECT_EQ(number_of_channels, frame_for_mixing.num_channels_);
+      }
+    }
+  }
+}
+
+TEST(AudioMixer, MultipleChannelsOneParticipant) {
+  // Set up a participant with a 6-channel frame, and make sure a 6-channel
+  // frame with the right sample values comes out from the mixer. There are 2
+  // Mix calls because of ramp-up.
+  constexpr size_t kNumberOfChannels = 6;
+  MockMixerAudioSource source;
+  ResetFrame(source.fake_frame());
+  const auto mixer = AudioMixerImpl::Create();
+  mixer->AddSource(&source);
+  mixer->Mix(1, &frame_for_mixing);
+  auto* frame = source.fake_frame();
+  frame->num_channels_ = kNumberOfChannels;
+  std::fill(frame->mutable_data(),
+            frame->mutable_data() + AudioFrame::kMaxDataSizeSamples, 0);
+  for (size_t i = 0; i < kNumberOfChannels; ++i) {
+    frame->mutable_data()[100 * frame->num_channels_ + i] = 1000 * i;
+  }
+
+  mixer->Mix(kNumberOfChannels, &frame_for_mixing);
+
+  EXPECT_EQ(frame_for_mixing.num_channels_, kNumberOfChannels);
+  for (size_t i = 0; i < kNumberOfChannels; ++i) {
+    EXPECT_EQ(frame_for_mixing.data()[100 * frame_for_mixing.num_channels_ + i],
+              static_cast<int16_t>(1000 * i));
+  }
+}
+
+TEST(AudioMixer, MultipleChannelsManyParticipants) {
+  // Sets up 2 participants. One has a 6-channel frame. Make sure a 6-channel
+  // frame with the right sample values comes out from the mixer. There are 2
+  // Mix calls because of ramp-up.
+  constexpr size_t kNumberOfChannels = 6;
+  MockMixerAudioSource source;
+  const auto mixer = AudioMixerImpl::Create();
+  mixer->AddSource(&source);
+  ResetFrame(source.fake_frame());
+  mixer->Mix(1, &frame_for_mixing);
+  auto* frame = source.fake_frame();
+  frame->num_channels_ = kNumberOfChannels;
+  std::fill(frame->mutable_data(),
+            frame->mutable_data() + AudioFrame::kMaxDataSizeSamples, 0);
+  for (size_t i = 0; i < kNumberOfChannels; ++i) {
+    frame->mutable_data()[100 * frame->num_channels_ + i] = 1000 * i;
+  }
+  MockMixerAudioSource other_source;
+  ResetFrame(other_source.fake_frame());
+  mixer->AddSource(&other_source);
+
+  mixer->Mix(kNumberOfChannels, &frame_for_mixing);
+
+  EXPECT_EQ(frame_for_mixing.num_channels_, kNumberOfChannels);
+  for (size_t i = 0; i < kNumberOfChannels; ++i) {
+    EXPECT_EQ(frame_for_mixing.data()[100 * frame_for_mixing.num_channels_ + i],
+              static_cast<int16_t>(1000 * i));
+  }
+}
+
+TEST(AudioMixer, ShouldIncludeRtpPacketInfoFromAllMixedSources) {
+  const uint32_t kSsrc0 = 10;
+  const uint32_t kSsrc1 = 11;
+  const uint32_t kSsrc2 = 12;
+  const uint32_t kCsrc0 = 20;
+  const uint32_t kCsrc1 = 21;
+  const uint32_t kCsrc2 = 22;
+  const uint32_t kCsrc3 = 23;
+  const int kAudioLevel0 = 10;
+  const int kAudioLevel1 = 40;
+  const absl::optional<uint32_t> kAudioLevel2 = absl::nullopt;
+  const uint32_t kRtpTimestamp0 = 300;
+  const uint32_t kRtpTimestamp1 = 400;
+  const Timestamp kReceiveTime0 = Timestamp::Millis(10);
+  const Timestamp kReceiveTime1 = Timestamp::Millis(20);
+
+  RtpPacketInfo p0(kSsrc0, {kCsrc0, kCsrc1}, kRtpTimestamp0, kReceiveTime0);
+  p0.set_audio_level(kAudioLevel0);
+  RtpPacketInfo p1(kSsrc1, {kCsrc2}, kRtpTimestamp1, kReceiveTime1);
+  p1.set_audio_level(kAudioLevel1);
+  RtpPacketInfo p2(kSsrc2, {kCsrc3}, kRtpTimestamp1, kReceiveTime1);
+  p2.set_audio_level(kAudioLevel2);
+
+  const auto mixer = AudioMixerImpl::Create();
+
+  MockMixerAudioSource source;
+  source.set_packet_infos(RtpPacketInfos({p0}));
+  mixer->AddSource(&source);
+  ResetFrame(source.fake_frame());
+  mixer->Mix(1, &frame_for_mixing);
+
+  MockMixerAudioSource other_source;
+  other_source.set_packet_infos(RtpPacketInfos({p1, p2}));
+  ResetFrame(other_source.fake_frame());
+  mixer->AddSource(&other_source);
+
+  mixer->Mix(/*number_of_channels=*/1, &frame_for_mixing);
+
+  EXPECT_THAT(frame_for_mixing.packet_infos_, UnorderedElementsAre(p0, p1, p2));
+}
+
+TEST(AudioMixer, MixerShouldIncludeRtpPacketInfoFromMixedSourcesOnly) {
+  const uint32_t kSsrc0 = 10;
+  const uint32_t kSsrc1 = 11;
+  const uint32_t kSsrc2 = 21;
+  const uint32_t kCsrc0 = 30;
+  const uint32_t kCsrc1 = 31;
+  const uint32_t kCsrc2 = 32;
+  const uint32_t kCsrc3 = 33;
+  const int kAudioLevel0 = 10;
+  const absl::optional<uint32_t> kAudioLevelMissing = absl::nullopt;
+  const uint32_t kRtpTimestamp0 = 300;
+  const uint32_t kRtpTimestamp1 = 400;
+  const Timestamp kReceiveTime0 = Timestamp::Millis(10);
+  const Timestamp kReceiveTime1 = Timestamp::Millis(20);
+
+  RtpPacketInfo p0(kSsrc0, {kCsrc0, kCsrc1}, kRtpTimestamp0, kReceiveTime0);
+  p0.set_audio_level(kAudioLevel0);
+  RtpPacketInfo p1(kSsrc1, {kCsrc2}, kRtpTimestamp1, kReceiveTime1);
+  p1.set_audio_level(kAudioLevelMissing);
+  RtpPacketInfo p2(kSsrc2, {kCsrc3}, kRtpTimestamp1, kReceiveTime1);
+  p2.set_audio_level(kAudioLevelMissing);
+
+  const auto mixer = AudioMixerImpl::Create(/*max_sources_to_mix=*/2);
+
+  MockMixerAudioSource source1;
+  source1.set_packet_infos(RtpPacketInfos({p0}));
+  mixer->AddSource(&source1);
+  ResetFrame(source1.fake_frame());
+  mixer->Mix(1, &frame_for_mixing);
+
+  MockMixerAudioSource source2;
+  source2.set_packet_infos(RtpPacketInfos({p1}));
+  ResetFrame(source2.fake_frame());
+  mixer->AddSource(&source2);
+
+  // The mixer prioritizes kVadActive over kVadPassive.
+  // We limit the number of sources to mix to 2 and set the third source's VAD
+  // activity to kVadPassive so that it will not be added to the mix.
+  MockMixerAudioSource source3;
+  source3.set_packet_infos(RtpPacketInfos({p2}));
+  ResetFrame(source3.fake_frame());
+  source3.fake_frame()->vad_activity_ = AudioFrame::kVadPassive;
+  mixer->AddSource(&source3);
+
+  mixer->Mix(/*number_of_channels=*/1, &frame_for_mixing);
+
+  EXPECT_THAT(frame_for_mixing.packet_infos_, UnorderedElementsAre(p0, p1));
+}
+
+class HighOutputRateCalculator : public OutputRateCalculator {
+ public:
+  static const int kDefaultFrequency = 76000;
+  int CalculateOutputRateFromRange(
+      rtc::ArrayView<const int> preferred_sample_rates) override {
+    return kDefaultFrequency;
+  }
+  ~HighOutputRateCalculator() override {}
+};
+const int HighOutputRateCalculator::kDefaultFrequency;
+
+TEST(AudioMixerDeathTest, MultipleChannelsAndHighRate) {
+  constexpr size_t kSamplesPerChannel =
+      HighOutputRateCalculator::kDefaultFrequency / 100;
+  // As many channels as an AudioFrame can fit:
+  constexpr size_t kNumberOfChannels =
+      AudioFrame::kMaxDataSizeSamples / kSamplesPerChannel;
+  MockMixerAudioSource source;
+  const auto mixer = AudioMixerImpl::Create(
+      std::make_unique<HighOutputRateCalculator>(), true);
+  mixer->AddSource(&source);
+  ResetFrame(source.fake_frame());
+  mixer->Mix(1, &frame_for_mixing);
+  auto* frame = source.fake_frame();
+  frame->num_channels_ = kNumberOfChannels;
+  frame->sample_rate_hz_ = HighOutputRateCalculator::kDefaultFrequency;
+  frame->samples_per_channel_ = kSamplesPerChannel;
+
+  std::fill(frame->mutable_data(),
+            frame->mutable_data() + AudioFrame::kMaxDataSizeSamples, 0);
+  MockMixerAudioSource other_source;
+  ResetFrame(other_source.fake_frame());
+  auto* other_frame = other_source.fake_frame();
+  other_frame->num_channels_ = kNumberOfChannels;
+  other_frame->sample_rate_hz_ = HighOutputRateCalculator::kDefaultFrequency;
+  other_frame->samples_per_channel_ = kSamplesPerChannel;
+  mixer->AddSource(&other_source);
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+  EXPECT_DEATH(mixer->Mix(kNumberOfChannels, &frame_for_mixing), "");
+#elif !RTC_DCHECK_IS_ON
+  mixer->Mix(kNumberOfChannels, &frame_for_mixing);
+  EXPECT_EQ(frame_for_mixing.num_channels_, kNumberOfChannels);
+  EXPECT_EQ(frame_for_mixing.sample_rate_hz_,
+            HighOutputRateCalculator::kDefaultFrequency);
+#endif
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/audio_mixer_test.cc b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_test.cc
new file mode 100644
index 0000000000..3ee28a7937
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/audio_mixer_test.cc
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio/audio_mixer.h"
+
+#include <cstring>
+#include <iostream>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_mixer/default_output_rate_calculator.h"
+#include "rtc_base/strings/string_builder.h"
+
+ABSL_FLAG(int,
+          sampling_rate,
+          16000,
+          "Rate at which to mix (all input streams must have this rate)");
+
+ABSL_FLAG(bool,
+          stereo,
+          false,
+          "Enable stereo (interleaved). Inputs need not be as this parameter.");
+
+ABSL_FLAG(bool, limiter, true, "Enable limiter.");
+ABSL_FLAG(std::string,
+          output_file,
+          "mixed_file.wav",
+          "File in which to store the mixed result.");
+ABSL_FLAG(std::string, input_file_1, "", "First input. Default none.");
+ABSL_FLAG(std::string, input_file_2, "", "Second input. Default none.");
+ABSL_FLAG(std::string, input_file_3, "", "Third input. Default none.");
+ABSL_FLAG(std::string, input_file_4, "", "Fourth input. Default none.");
+
+namespace webrtc {
+namespace test {
+
+class FilePlayingSource : public AudioMixer::Source {
+ public:
+  explicit FilePlayingSource(absl::string_view filename)
+      : wav_reader_(new WavReader(filename)),
+        sample_rate_hz_(wav_reader_->sample_rate()),
+        samples_per_channel_(sample_rate_hz_ / 100),
+        number_of_channels_(wav_reader_->num_channels()) {}
+
+  AudioFrameInfo GetAudioFrameWithInfo(int target_rate_hz,
+                                       AudioFrame* frame) override {
+    frame->samples_per_channel_ = samples_per_channel_;
+    frame->num_channels_ = number_of_channels_;
+    frame->sample_rate_hz_ = target_rate_hz;
+
+    RTC_CHECK_EQ(target_rate_hz, sample_rate_hz_);
+
+    const size_t num_to_read = number_of_channels_ * samples_per_channel_;
+    const size_t num_read =
+        wav_reader_->ReadSamples(num_to_read, frame->mutable_data());
+
+    file_has_ended_ = num_to_read != num_read;
+    if (file_has_ended_) {
+      frame->Mute();
+    }
+    return file_has_ended_ ? AudioFrameInfo::kMuted : AudioFrameInfo::kNormal;
+  }
+
+  int Ssrc() const override { return 0; }
+
+  int PreferredSampleRate() const override { return sample_rate_hz_; }
+
+  bool FileHasEnded() const { return file_has_ended_; }
+
+  std::string ToString() const {
+    rtc::StringBuilder ss;
+    ss << "{rate: " << sample_rate_hz_ << ", channels: " << number_of_channels_
+       << ", samples_tot: " << wav_reader_->num_samples() << "}";
+    return ss.Release();
+  }
+
+ private:
+  std::unique_ptr<WavReader> wav_reader_;
+  int sample_rate_hz_;
+  int samples_per_channel_;
+  int number_of_channels_;
+  bool file_has_ended_ = false;
+};
+}  // namespace test
+}  // namespace webrtc
+
+namespace {
+
+const std::vector<std::string> parse_input_files() {
+  std::vector<std::string> result;
+  for (auto& x :
+       {absl::GetFlag(FLAGS_input_file_1), absl::GetFlag(FLAGS_input_file_2),
+        absl::GetFlag(FLAGS_input_file_3), absl::GetFlag(FLAGS_input_file_4)}) {
+    if (!x.empty()) {
+      result.push_back(x);
+    }
+  }
+  return result;
+}
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  absl::ParseCommandLine(argc, argv);
+
+  rtc::scoped_refptr<webrtc::AudioMixerImpl> mixer(
+      webrtc::AudioMixerImpl::Create(
+          std::unique_ptr<webrtc::OutputRateCalculator>(
+              new webrtc::DefaultOutputRateCalculator()),
+          absl::GetFlag(FLAGS_limiter)));
+
+  const std::vector<std::string> input_files = parse_input_files();
+  std::vector<webrtc::test::FilePlayingSource> sources;
+  const int num_channels = absl::GetFlag(FLAGS_stereo) ? 2 : 1;
+  sources.reserve(input_files.size());
+  for (const auto& input_file : input_files) {
+    sources.emplace_back(input_file);
+  }
+
+  for (auto& source : sources) {
+    auto error = mixer->AddSource(&source);
+    RTC_CHECK(error);
+  }
+
+  if (sources.empty()) {
+    std::cout << "Need at least one source!\n";
+    return 1;
+  }
+
+  const size_t sample_rate = sources[0].PreferredSampleRate();
+  for (const auto& source : sources) {
+    RTC_CHECK_EQ(sample_rate, source.PreferredSampleRate());
+  }
+
+  // Print stats.
+  std::cout << "Limiting is: " << (absl::GetFlag(FLAGS_limiter) ? "on" : "off")
+            << "\n"
+               "Channels: "
+            << num_channels
+            << "\n"
+               "Rate: "
+            << sample_rate
+            << "\n"
+               "Number of input streams: "
+            << input_files.size() << "\n";
+  for (const auto& source : sources) {
+    std::cout << "\t" << source.ToString() << "\n";
+  }
+  std::cout << "Now mixing\n...\n";
+
+  webrtc::WavWriter wav_writer(absl::GetFlag(FLAGS_output_file), sample_rate,
+                               num_channels);
+
+  webrtc::AudioFrame frame;
+
+  bool all_streams_finished = false;
+  while (!all_streams_finished) {
+    mixer->Mix(num_channels, &frame);
+    RTC_CHECK_EQ(sample_rate / 100, frame.samples_per_channel_);
+    RTC_CHECK_EQ(sample_rate, frame.sample_rate_hz_);
+    RTC_CHECK_EQ(num_channels, frame.num_channels_);
+    wav_writer.WriteSamples(frame.data(),
+                            num_channels * frame.samples_per_channel_);
+
+    all_streams_finished =
+        std::all_of(sources.begin(), sources.end(),
+                    [](const webrtc::test::FilePlayingSource& source) {
+                      return source.FileHasEnded();
+                    });
+  }
+
+  std::cout << "Done!\n" << std::endl;
+}
diff --git a/third_party/libwebrtc/modules/audio_mixer/default_output_rate_calculator.cc b/third_party/libwebrtc/modules/audio_mixer/default_output_rate_calculator.cc
new file mode 100644
index 0000000000..5f24b653a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/default_output_rate_calculator.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_mixer/default_output_rate_calculator.h"
+
+#include <algorithm>
+#include <iterator>
+
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+int DefaultOutputRateCalculator::CalculateOutputRateFromRange(
+    rtc::ArrayView<const int> preferred_sample_rates) {
+  if (preferred_sample_rates.empty()) {
+    return DefaultOutputRateCalculator::kDefaultFrequency;
+  }
+  using NativeRate = AudioProcessing::NativeRate;
+  const int maximal_frequency = *std::max_element(
+      preferred_sample_rates.cbegin(), preferred_sample_rates.cend());
+
+  RTC_DCHECK_LE(NativeRate::kSampleRate8kHz, maximal_frequency);
+  RTC_DCHECK_GE(NativeRate::kSampleRate48kHz, maximal_frequency);
+
+  static constexpr NativeRate native_rates[] = {
+      NativeRate::kSampleRate8kHz, NativeRate::kSampleRate16kHz,
+      NativeRate::kSampleRate32kHz, NativeRate::kSampleRate48kHz};
+  const auto* rounded_up_index = std::lower_bound(
+      std::begin(native_rates), std::end(native_rates), maximal_frequency);
+  RTC_DCHECK(rounded_up_index != std::end(native_rates));
+  return *rounded_up_index;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/default_output_rate_calculator.h b/third_party/libwebrtc/modules/audio_mixer/default_output_rate_calculator.h
new file mode 100644
index 0000000000..02a3b5c37b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/default_output_rate_calculator.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_MIXER_DEFAULT_OUTPUT_RATE_CALCULATOR_H_
+#define MODULES_AUDIO_MIXER_DEFAULT_OUTPUT_RATE_CALCULATOR_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_mixer/output_rate_calculator.h"
+
+namespace webrtc {
+
+class DefaultOutputRateCalculator : public OutputRateCalculator {
+ public:
+  static const int kDefaultFrequency = 48000;
+
+  // Produces the least native rate greater or equal to the preferred
+  // sample rates. A native rate is one in
+  // AudioProcessing::NativeRate. If `preferred_sample_rates` is
+  // empty, returns `kDefaultFrequency`.
+  int CalculateOutputRateFromRange(
+      rtc::ArrayView<const int> preferred_sample_rates) override;
+  ~DefaultOutputRateCalculator() override {}
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_MIXER_DEFAULT_OUTPUT_RATE_CALCULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_mixer/frame_combiner.cc b/third_party/libwebrtc/modules/audio_mixer/frame_combiner.cc
new file mode 100644
index 0000000000..96c62f6b0d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/frame_combiner.cc
@@ -0,0 +1,213 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_mixer/frame_combiner.h"
+
+#include <algorithm>
+#include <array>
+#include <cstdint>
+#include <iterator>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/rtp_packet_info.h"
+#include "api/rtp_packet_infos.h"
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_mixer/audio_frame_manipulator.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+
+using MixingBuffer =
+    std::array<std::array<float, FrameCombiner::kMaximumChannelSize>,
+               FrameCombiner::kMaximumNumberOfChannels>;
+
+void SetAudioFrameFields(rtc::ArrayView<const AudioFrame* const> mix_list,
+                         size_t number_of_channels,
+                         int sample_rate,
+                         size_t number_of_streams,
+                         AudioFrame* audio_frame_for_mixing) {
+  const size_t samples_per_channel = static_cast<size_t>(
+      (sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000);
+
+  // TODO(minyue): Issue bugs.webrtc.org/3390.
+  // Audio frame timestamp. The 'timestamp_' field is set to dummy
+  // value '0', because it is only supported in the one channel case and
+  // is then updated in the helper functions.
+  audio_frame_for_mixing->UpdateFrame(
+      0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined,
+      AudioFrame::kVadUnknown, number_of_channels);
+
+  if (mix_list.empty()) {
+    audio_frame_for_mixing->elapsed_time_ms_ = -1;
+  } else {
+    audio_frame_for_mixing->timestamp_ = mix_list[0]->timestamp_;
+    audio_frame_for_mixing->elapsed_time_ms_ = mix_list[0]->elapsed_time_ms_;
+    audio_frame_for_mixing->ntp_time_ms_ = mix_list[0]->ntp_time_ms_;
+    std::vector<RtpPacketInfo> packet_infos;
+    for (const auto& frame : mix_list) {
+      audio_frame_for_mixing->timestamp_ =
+          std::min(audio_frame_for_mixing->timestamp_, frame->timestamp_);
+      audio_frame_for_mixing->ntp_time_ms_ =
+          std::min(audio_frame_for_mixing->ntp_time_ms_, frame->ntp_time_ms_);
+      audio_frame_for_mixing->elapsed_time_ms_ = std::max(
+          audio_frame_for_mixing->elapsed_time_ms_, frame->elapsed_time_ms_);
+      packet_infos.insert(packet_infos.end(), frame->packet_infos_.begin(),
+                          frame->packet_infos_.end());
+    }
+    audio_frame_for_mixing->packet_infos_ =
+        RtpPacketInfos(std::move(packet_infos));
+  }
+}
+
+void MixFewFramesWithNoLimiter(rtc::ArrayView<const AudioFrame* const> mix_list,
+                               AudioFrame* audio_frame_for_mixing) {
+  if (mix_list.empty()) {
+    audio_frame_for_mixing->Mute();
+    return;
+  }
+  RTC_DCHECK_LE(mix_list.size(), 1);
+  std::copy(mix_list[0]->data(),
+            mix_list[0]->data() +
+                mix_list[0]->num_channels_ * mix_list[0]->samples_per_channel_,
+            audio_frame_for_mixing->mutable_data());
+}
+
+void MixToFloatFrame(rtc::ArrayView<const AudioFrame* const> mix_list,
+                     size_t samples_per_channel,
+                     size_t number_of_channels,
+                     MixingBuffer* mixing_buffer) {
+  RTC_DCHECK_LE(samples_per_channel, FrameCombiner::kMaximumChannelSize);
+  RTC_DCHECK_LE(number_of_channels, FrameCombiner::kMaximumNumberOfChannels);
+  // Clear the mixing buffer.
+  *mixing_buffer = {};
+
+  // Convert to FloatS16 and mix.
+  for (size_t i = 0; i < mix_list.size(); ++i) {
+    const AudioFrame* const frame = mix_list[i];
+    const int16_t* const frame_data = frame->data();
+    for (size_t j = 0; j < std::min(number_of_channels,
+                                    FrameCombiner::kMaximumNumberOfChannels);
+         ++j) {
+      for (size_t k = 0; k < std::min(samples_per_channel,
+                                      FrameCombiner::kMaximumChannelSize);
+           ++k) {
+        (*mixing_buffer)[j][k] += frame_data[number_of_channels * k + j];
+      }
+    }
+  }
+}
+
+void RunLimiter(AudioFrameView<float> mixing_buffer_view, Limiter* limiter) {
+  const size_t sample_rate = mixing_buffer_view.samples_per_channel() * 1000 /
+                             AudioMixerImpl::kFrameDurationInMs;
+  // TODO(alessiob): Avoid calling SetSampleRate every time.
+  limiter->SetSampleRate(sample_rate);
+  limiter->Process(mixing_buffer_view);
+}
+
+// Both interleaves and rounds.
+void InterleaveToAudioFrame(AudioFrameView<const float> mixing_buffer_view,
+                            AudioFrame* audio_frame_for_mixing) {
+  const size_t number_of_channels = mixing_buffer_view.num_channels();
+  const size_t samples_per_channel = mixing_buffer_view.samples_per_channel();
+  int16_t* const mixing_data = audio_frame_for_mixing->mutable_data();
+  // Put data in the result frame.
+  for (size_t i = 0; i < number_of_channels; ++i) {
+    for (size_t j = 0; j < samples_per_channel; ++j) {
+      mixing_data[number_of_channels * j + i] =
+          FloatS16ToS16(mixing_buffer_view.channel(i)[j]);
+    }
+  }
+}
+}  // namespace
+
+constexpr size_t FrameCombiner::kMaximumNumberOfChannels;
+constexpr size_t FrameCombiner::kMaximumChannelSize;
+
+FrameCombiner::FrameCombiner(bool use_limiter)
+    : data_dumper_(new ApmDataDumper(0)),
+      mixing_buffer_(
+          std::make_unique<std::array<std::array<float, kMaximumChannelSize>,
+                                      kMaximumNumberOfChannels>>()),
+      limiter_(static_cast<size_t>(48000), data_dumper_.get(), "AudioMixer"),
+      use_limiter_(use_limiter) {
+  static_assert(kMaximumChannelSize * kMaximumNumberOfChannels <=
+                    AudioFrame::kMaxDataSizeSamples,
+                "");
+}
+
+FrameCombiner::~FrameCombiner() = default;
+
+void FrameCombiner::Combine(rtc::ArrayView<AudioFrame* const> mix_list,
+                            size_t number_of_channels,
+                            int sample_rate,
+                            size_t number_of_streams,
+                            AudioFrame* audio_frame_for_mixing) {
+  RTC_DCHECK(audio_frame_for_mixing);
+
+  SetAudioFrameFields(mix_list, number_of_channels, sample_rate,
+                      number_of_streams, audio_frame_for_mixing);
+
+  const size_t samples_per_channel = static_cast<size_t>(
+      (sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000);
+
+  for (const auto* frame : mix_list) {
+    RTC_DCHECK_EQ(samples_per_channel, frame->samples_per_channel_);
+    RTC_DCHECK_EQ(sample_rate, frame->sample_rate_hz_);
+  }
+
+  // The 'num_channels_' field of frames in 'mix_list' could be
+  // different from 'number_of_channels'.
+  for (auto* frame : mix_list) {
+    RemixFrame(number_of_channels, frame);
+  }
+
+  if (number_of_streams <= 1) {
+    MixFewFramesWithNoLimiter(mix_list, audio_frame_for_mixing);
+    return;
+  }
+
+  MixToFloatFrame(mix_list, samples_per_channel, number_of_channels,
+                  mixing_buffer_.get());
+
+  const size_t output_number_of_channels =
+      std::min(number_of_channels, kMaximumNumberOfChannels);
+  const size_t output_samples_per_channel =
+      std::min(samples_per_channel, kMaximumChannelSize);
+
+  // Put float data in an AudioFrameView.
+  std::array<float*, kMaximumNumberOfChannels> channel_pointers{};
+  for (size_t i = 0; i < output_number_of_channels; ++i) {
+    channel_pointers[i] = &(*mixing_buffer_.get())[i][0];
+  }
+  AudioFrameView<float> mixing_buffer_view(&channel_pointers[0],
+                                           output_number_of_channels,
+                                           output_samples_per_channel);
+
+  if (use_limiter_) {
+    RunLimiter(mixing_buffer_view, &limiter_);
+  }
+
+  InterleaveToAudioFrame(mixing_buffer_view, audio_frame_for_mixing);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/frame_combiner.h b/third_party/libwebrtc/modules/audio_mixer/frame_combiner.h
new file mode 100644
index 0000000000..6185b29f8a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/frame_combiner.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
+#define MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/audio_frame.h"
+#include "modules/audio_processing/agc2/limiter.h"
+
+namespace webrtc {
+class ApmDataDumper;
+
+class FrameCombiner {
+ public:
+  explicit FrameCombiner(bool use_limiter);
+  ~FrameCombiner();
+
+  // Combine several frames into one. Assumes sample_rate,
+  // samples_per_channel of the input frames match the parameters. The
+  // parameters 'number_of_channels' and 'sample_rate' are needed
+  // because 'mix_list' can be empty. The parameter
+  // 'number_of_streams' is used for determining whether to pass the
+  // data through a limiter.
+  void Combine(rtc::ArrayView<AudioFrame* const> mix_list,
+               size_t number_of_channels,
+               int sample_rate,
+               size_t number_of_streams,
+               AudioFrame* audio_frame_for_mixing);
+
+  // Stereo, 48 kHz, 10 ms.
+  static constexpr size_t kMaximumNumberOfChannels = 8;
+  static constexpr size_t kMaximumChannelSize = 48 * 10;
+
+  using MixingBuffer = std::array<std::array<float, kMaximumChannelSize>,
+                                  kMaximumNumberOfChannels>;
+
+ private:
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  std::unique_ptr<MixingBuffer> mixing_buffer_;
+  Limiter limiter_;
+  const bool use_limiter_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_MIXER_FRAME_COMBINER_H_
diff --git a/third_party/libwebrtc/modules/audio_mixer/frame_combiner_unittest.cc b/third_party/libwebrtc/modules/audio_mixer/frame_combiner_unittest.cc
new file mode 100644
index 0000000000..6c64d0852a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/frame_combiner_unittest.cc
@@ -0,0 +1,337 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_mixer/frame_combiner.h"
+
+#include <cstdint>
+#include <initializer_list>
+#include <numeric>
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/rtp_packet_info.h"
+#include "api/rtp_packet_infos.h"
+#include "api/units/timestamp.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "modules/audio_mixer/gain_change_calculator.h"
+#include "modules/audio_mixer/sine_wave_generator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+using ::testing::UnorderedElementsAreArray;
+
+struct FrameCombinerConfig {
+  bool use_limiter;
+  int sample_rate_hz;
+  int number_of_channels;
+  float wave_frequency;
+};
+
+std::string ProduceDebugText(int sample_rate_hz,
+                             int number_of_channels,
+                             int number_of_sources) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz << " ,";
+  ss << "number of channels: " << number_of_channels << " ,";
+  ss << "number of sources: " << number_of_sources;
+  return ss.Release();
+}
+
+std::string ProduceDebugText(const FrameCombinerConfig& config) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << config.sample_rate_hz << " ,";
+  ss << "number of channels: " << config.number_of_channels << " ,";
+  ss << "limiter active: " << (config.use_limiter ? "on" : "off") << " ,";
+  ss << "wave frequency: " << config.wave_frequency << " ,";
+  return ss.Release();
+}
+
+AudioFrame frame1;
+AudioFrame frame2;
+
+void SetUpFrames(int sample_rate_hz, int number_of_channels) {
+  RtpPacketInfo packet_info1(/*ssrc=*/1001, /*csrcs=*/{},
+                             /*rtp_timestamp=*/1000,
+                             /*receive_time=*/Timestamp::Millis(1));
+  RtpPacketInfo packet_info2(/*ssrc=*/4004, /*csrcs=*/{},
+                             /*rtp_timestamp=*/1234,
+                             /*receive_time=*/Timestamp::Millis(2));
+  RtpPacketInfo packet_info3(/*ssrc=*/7007, /*csrcs=*/{},
+                             /*rtp_timestamp=*/1333,
+                             /*receive_time=*/Timestamp::Millis(2));
+
+  frame1.packet_infos_ = RtpPacketInfos({packet_info1});
+  frame2.packet_infos_ = RtpPacketInfos({packet_info2, packet_info3});
+
+  for (auto* frame : {&frame1, &frame2}) {
+    frame->UpdateFrame(0, nullptr, rtc::CheckedDivExact(sample_rate_hz, 100),
+                       sample_rate_hz, AudioFrame::kNormalSpeech,
+                       AudioFrame::kVadActive, number_of_channels);
+  }
+}
+}  // namespace
+
+// The limiter requires sample rate divisible by 2000.
+TEST(FrameCombiner, BasicApiCallsLimiter) {
+  FrameCombiner combiner(true);
+  for (const int rate : {8000, 18000, 34000, 48000}) {
+    for (const int number_of_channels : {1, 2, 4, 8}) {
+      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+      SetUpFrames(rate, number_of_channels);
+
+      for (const int number_of_frames : {0, 1, 2}) {
+        SCOPED_TRACE(
+            ProduceDebugText(rate, number_of_channels, number_of_frames));
+        const std::vector<AudioFrame*> frames_to_combine(
+            all_frames.begin(), all_frames.begin() + number_of_frames);
+        AudioFrame audio_frame_for_mixing;
+        combiner.Combine(frames_to_combine, number_of_channels, rate,
+                         frames_to_combine.size(), &audio_frame_for_mixing);
+      }
+    }
+  }
+}
+
+// The RtpPacketInfos field of the mixed packet should contain the union of the
+// RtpPacketInfos from the frames that were actually mixed.
+TEST(FrameCombiner, ContainsAllRtpPacketInfos) {
+  static constexpr int kSampleRateHz = 48000;
+  static constexpr int kNumChannels = 1;
+  FrameCombiner combiner(true);
+  const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+  SetUpFrames(kSampleRateHz, kNumChannels);
+
+  for (const int number_of_frames : {0, 1, 2}) {
+    SCOPED_TRACE(
+        ProduceDebugText(kSampleRateHz, kNumChannels, number_of_frames));
+    const std::vector<AudioFrame*> frames_to_combine(
+        all_frames.begin(), all_frames.begin() + number_of_frames);
+
+    std::vector<RtpPacketInfo> packet_infos;
+    for (const auto& frame : frames_to_combine) {
+      packet_infos.insert(packet_infos.end(), frame->packet_infos_.begin(),
+                          frame->packet_infos_.end());
+    }
+
+    AudioFrame audio_frame_for_mixing;
+    combiner.Combine(frames_to_combine, kNumChannels, kSampleRateHz,
+                     frames_to_combine.size(), &audio_frame_for_mixing);
+    EXPECT_THAT(audio_frame_for_mixing.packet_infos_,
+                UnorderedElementsAreArray(packet_infos));
+  }
+}
+
+// There are DCHECKs in place to check for invalid parameters.
+TEST(FrameCombinerDeathTest, DebugBuildCrashesWithManyChannels) {
+  FrameCombiner combiner(true);
+  for (const int rate : {8000, 18000, 34000, 48000}) {
+    for (const int number_of_channels : {10, 20, 21}) {
+      if (static_cast<size_t>(rate / 100 * number_of_channels) >
+          AudioFrame::kMaxDataSizeSamples) {
+        continue;
+      }
+      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+      SetUpFrames(rate, number_of_channels);
+
+      const int number_of_frames = 2;
+      SCOPED_TRACE(
+          ProduceDebugText(rate, number_of_channels, number_of_frames));
+      const std::vector<AudioFrame*> frames_to_combine(
+          all_frames.begin(), all_frames.begin() + number_of_frames);
+      AudioFrame audio_frame_for_mixing;
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+      EXPECT_DEATH(
+          combiner.Combine(frames_to_combine, number_of_channels, rate,
+                           frames_to_combine.size(), &audio_frame_for_mixing),
+          "");
+#elif !RTC_DCHECK_IS_ON
+      combiner.Combine(frames_to_combine, number_of_channels, rate,
+                       frames_to_combine.size(), &audio_frame_for_mixing);
+#endif
+    }
+  }
+}
+
+TEST(FrameCombinerDeathTest, DebugBuildCrashesWithHighRate) {
+  FrameCombiner combiner(true);
+  for (const int rate : {50000, 96000, 128000, 196000}) {
+    for (const int number_of_channels : {1, 2, 3}) {
+      if (static_cast<size_t>(rate / 100 * number_of_channels) >
+          AudioFrame::kMaxDataSizeSamples) {
+        continue;
+      }
+      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+      SetUpFrames(rate, number_of_channels);
+
+      const int number_of_frames = 2;
+      SCOPED_TRACE(
+          ProduceDebugText(rate, number_of_channels, number_of_frames));
+      const std::vector<AudioFrame*> frames_to_combine(
+          all_frames.begin(), all_frames.begin() + number_of_frames);
+      AudioFrame audio_frame_for_mixing;
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+      EXPECT_DEATH(
+          combiner.Combine(frames_to_combine, number_of_channels, rate,
+                           frames_to_combine.size(), &audio_frame_for_mixing),
+          "");
+#elif !RTC_DCHECK_IS_ON
+      combiner.Combine(frames_to_combine, number_of_channels, rate,
+                       frames_to_combine.size(), &audio_frame_for_mixing);
+#endif
+    }
+  }
+}
+
+// With no limiter, the rate has to be divisible by 100 since we use
+// 10 ms frames.
+TEST(FrameCombiner, BasicApiCallsNoLimiter) {
+  FrameCombiner combiner(false);
+  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
+    for (const int number_of_channels : {1, 2, 4, 8}) {
+      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+      SetUpFrames(rate, number_of_channels);
+
+      for (const int number_of_frames : {0, 1, 2}) {
+        SCOPED_TRACE(
+            ProduceDebugText(rate, number_of_channels, number_of_frames));
+        const std::vector<AudioFrame*> frames_to_combine(
+            all_frames.begin(), all_frames.begin() + number_of_frames);
+        AudioFrame audio_frame_for_mixing;
+        combiner.Combine(frames_to_combine, number_of_channels, rate,
+                         frames_to_combine.size(), &audio_frame_for_mixing);
+      }
+    }
+  }
+}
+
+TEST(FrameCombiner, CombiningZeroFramesShouldProduceSilence) {
+  FrameCombiner combiner(false);
+  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
+    for (const int number_of_channels : {1, 2}) {
+      SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 0));
+
+      AudioFrame audio_frame_for_mixing;
+
+      const std::vector<AudioFrame*> frames_to_combine;
+      combiner.Combine(frames_to_combine, number_of_channels, rate,
+                       frames_to_combine.size(), &audio_frame_for_mixing);
+      const int16_t* audio_frame_for_mixing_data =
+          audio_frame_for_mixing.data();
+      const std::vector<int16_t> mixed_data(
+          audio_frame_for_mixing_data,
+          audio_frame_for_mixing_data + number_of_channels * rate / 100);
+
+      const std::vector<int16_t> expected(number_of_channels * rate / 100, 0);
+      EXPECT_EQ(mixed_data, expected);
+      EXPECT_THAT(audio_frame_for_mixing.packet_infos_, IsEmpty());
+    }
+  }
+}
+
+TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
+  FrameCombiner combiner(false);
+  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
+    for (const int number_of_channels : {1, 2, 4, 8, 10}) {
+      SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1));
+
+      AudioFrame audio_frame_for_mixing;
+
+      SetUpFrames(rate, number_of_channels);
+      int16_t* frame1_data = frame1.mutable_data();
+      std::iota(frame1_data, frame1_data + number_of_channels * rate / 100, 0);
+      const std::vector<AudioFrame*> frames_to_combine = {&frame1};
+      combiner.Combine(frames_to_combine, number_of_channels, rate,
+                       frames_to_combine.size(), &audio_frame_for_mixing);
+
+      const int16_t* audio_frame_for_mixing_data =
+          audio_frame_for_mixing.data();
+      const std::vector<int16_t> mixed_data(
+          audio_frame_for_mixing_data,
+          audio_frame_for_mixing_data + number_of_channels * rate / 100);
+
+      std::vector<int16_t> expected(number_of_channels * rate / 100);
+      std::iota(expected.begin(), expected.end(), 0);
+      EXPECT_EQ(mixed_data, expected);
+      EXPECT_THAT(audio_frame_for_mixing.packet_infos_,
+                  ElementsAreArray(frame1.packet_infos_));
+    }
+  }
+}
+
+// Send a sine wave through the FrameCombiner, and check that the
+// difference between input and output varies smoothly. Also check
+// that it is inside reasonable bounds. This is to catch issues like
+// chromium:695993 and chromium:816875.
+TEST(FrameCombiner, GainCurveIsSmoothForAlternatingNumberOfStreams) {
+  // Rates are divisible by 2000 when limiter is active.
+  std::vector<FrameCombinerConfig> configs = {
+      {false, 30100, 2, 50.f},  {false, 16500, 1, 3200.f},
+      {true, 8000, 1, 3200.f},  {true, 16000, 1, 50.f},
+      {true, 18000, 8, 3200.f}, {true, 10000, 2, 50.f},
+  };
+
+  for (const auto& config : configs) {
+    SCOPED_TRACE(ProduceDebugText(config));
+
+    FrameCombiner combiner(config.use_limiter);
+
+    constexpr int16_t wave_amplitude = 30000;
+    SineWaveGenerator wave_generator(config.wave_frequency, wave_amplitude);
+
+    GainChangeCalculator change_calculator;
+    float cumulative_change = 0.f;
+
+    constexpr size_t iterations = 100;
+
+    for (size_t i = 0; i < iterations; ++i) {
+      SetUpFrames(config.sample_rate_hz, config.number_of_channels);
+      wave_generator.GenerateNextFrame(&frame1);
+      AudioFrameOperations::Mute(&frame2);
+
+      std::vector<AudioFrame*> frames_to_combine = {&frame1};
+      if (i % 2 == 0) {
+        frames_to_combine.push_back(&frame2);
+      }
+      const size_t number_of_samples =
+          frame1.samples_per_channel_ * config.number_of_channels;
+
+      // Ensures limiter is on if 'use_limiter'.
+      constexpr size_t number_of_streams = 2;
+      AudioFrame audio_frame_for_mixing;
+      combiner.Combine(frames_to_combine, config.number_of_channels,
+                       config.sample_rate_hz, number_of_streams,
+                       &audio_frame_for_mixing);
+      cumulative_change += change_calculator.CalculateGainChange(
+          rtc::ArrayView<const int16_t>(frame1.data(), number_of_samples),
+          rtc::ArrayView<const int16_t>(audio_frame_for_mixing.data(),
+                                        number_of_samples));
+    }
+
+    // Check that the gain doesn't vary too much.
+    EXPECT_LT(cumulative_change, 10);
+
+    // Check that the latest gain is within reasonable bounds. It
+    // should be slightly less that 1.
+    EXPECT_LT(0.9f, change_calculator.LatestGain());
+    EXPECT_LT(change_calculator.LatestGain(), 1.01f);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/g3doc/index.md b/third_party/libwebrtc/modules/audio_mixer/g3doc/index.md
new file mode 100644
index 0000000000..6b48378fcb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/g3doc/index.md
@@ -0,0 +1,54 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'alessiob' reviewed: '2021-04-21'} *-->
+
+# The WebRTC Audio Mixer Module
+
+The WebRTC audio mixer module is responsible for mixing multiple incoming audio
+streams (sources) into a single audio stream (mix). It works with 10 ms frames,
+it supports sample rates up to 48 kHz and up to 8 audio channels. The API is
+defined in
+[`api/audio/audio_mixer.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/audio/audio_mixer.h)
+and it includes the definition of
+[`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h),
+which describes an incoming audio stream, and the definition of
+[`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h),
+which operates on a collection of
+[`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
+objects to produce a mix.
+
+## AudioMixer::Source
+
+A source has different characteristic (e.g., sample rate, number of channels,
+muted state) and it is identified by an SSRC[^1].
+[`AudioMixer::Source::GetAudioFrameWithInfo()`](https://source.chromium.org/search?q=symbol:AudioMixer::Source::GetAudioFrameWithInfo%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
+is used to retrieve the next 10 ms chunk of audio to be mixed.
+
+[^1]: A synchronization source (SSRC) is the source of a stream of RTP packets,
+    identified by a 32-bit numeric SSRC identifier carried in the RTP header
+    so as not to be dependent upon the network address (see
+    [RFC 3550](https://tools.ietf.org/html/rfc3550#section-3)).
+
+## AudioMixer
+
+The interface allows to add and remove sources and the
+[`AudioMixer::Mix()`](https://source.chromium.org/search?q=symbol:AudioMixer::Mix%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
+method allows to generates a mix with the desired number of channels.
+
+## WebRTC implementation
+
+The interface is implemented in different parts of WebRTC:
+
+*   [`AudioMixer::Source`](https://source.chromium.org/search?q=symbol:AudioMixer::Source%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h):
+    [`audio/audio_receive_stream.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/audio/audio_receive_stream.h)
+*   [`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h):
+    [`modules/audio_mixer/audio_mixer_impl.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_mixer/audio_mixer_impl.h)
+
+[`AudioMixer`](https://source.chromium.org/search?q=symbol:AudioMixer%20file:third_party%2Fwebrtc%2Fapi%2Faudio%2Faudio_mixer.h)
+is thread-safe. The output sample rate of the generated mix is automatically
+assigned depending on the sample rate of the sources; whereas the number of
+output channels is defined by the caller[^2]. Samples from the non-muted sources
+are summed up and then a limiter is used to apply soft-clipping when needed.
+
+[^2]: [`audio/utility/channel_mixer.h`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/audio/utility/channel_mixer.h)
+    is used to mix channels in the non-trivial cases - i.e., if the number of
+    channels for a source or the mix is greater than 3.
diff --git a/third_party/libwebrtc/modules/audio_mixer/gain_change_calculator.cc b/third_party/libwebrtc/modules/audio_mixer/gain_change_calculator.cc
new file mode 100644
index 0000000000..dbd0945239
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/gain_change_calculator.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_mixer/gain_change_calculator.h"
+
+#include <math.h>
+
+#include <cstdlib>
+#include <vector>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int16_t kReliabilityThreshold = 100;
+}  // namespace
+
+float GainChangeCalculator::CalculateGainChange(
+    rtc::ArrayView<const int16_t> in,
+    rtc::ArrayView<const int16_t> out) {
+  RTC_DCHECK_EQ(in.size(), out.size());
+
+  std::vector<float> gain(in.size());
+  CalculateGain(in, out, gain);
+  return CalculateDifferences(gain);
+}
+
+float GainChangeCalculator::LatestGain() const {
+  return last_reliable_gain_;
+}
+
+void GainChangeCalculator::CalculateGain(rtc::ArrayView<const int16_t> in,
+                                         rtc::ArrayView<const int16_t> out,
+                                         rtc::ArrayView<float> gain) {
+  RTC_DCHECK_EQ(in.size(), out.size());
+  RTC_DCHECK_EQ(in.size(), gain.size());
+
+  for (size_t i = 0; i < in.size(); ++i) {
+    if (std::abs(in[i]) >= kReliabilityThreshold) {
+      last_reliable_gain_ = out[i] / static_cast<float>(in[i]);
+    }
+    gain[i] = last_reliable_gain_;
+  }
+}
+
+float GainChangeCalculator::CalculateDifferences(
+    rtc::ArrayView<const float> values) {
+  float res = 0;
+  for (float f : values) {
+    res += fabs(f - last_value_);
+    last_value_ = f;
+  }
+  return res;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/gain_change_calculator.h b/third_party/libwebrtc/modules/audio_mixer/gain_change_calculator.h
new file mode 100644
index 0000000000..3dde9be61e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/gain_change_calculator.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_MIXER_GAIN_CHANGE_CALCULATOR_H_
+#define MODULES_AUDIO_MIXER_GAIN_CHANGE_CALCULATOR_H_
+
+#include <stdint.h>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+class GainChangeCalculator {
+ public:
+  // The 'out' signal is assumed to be produced from 'in' by applying
+  // a smoothly varying gain. This method computes variations of the
+  // gain and handles special cases when the samples are small.
+  float CalculateGainChange(rtc::ArrayView<const int16_t> in,
+                            rtc::ArrayView<const int16_t> out);
+
+  float LatestGain() const;
+
+ private:
+  void CalculateGain(rtc::ArrayView<const int16_t> in,
+                     rtc::ArrayView<const int16_t> out,
+                     rtc::ArrayView<float> gain);
+
+  float CalculateDifferences(rtc::ArrayView<const float> values);
+  float last_value_ = 0.f;
+  float last_reliable_gain_ = 1.0f;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_MIXER_GAIN_CHANGE_CALCULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_mixer/output_rate_calculator.h b/third_party/libwebrtc/modules/audio_mixer/output_rate_calculator.h
new file mode 100644
index 0000000000..46b65a8b57
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/output_rate_calculator.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_MIXER_OUTPUT_RATE_CALCULATOR_H_
+#define MODULES_AUDIO_MIXER_OUTPUT_RATE_CALCULATOR_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Decides the sample rate of a mixing iteration given the preferred
+// sample rates of the sources.
+class OutputRateCalculator {
+ public:
+  virtual int CalculateOutputRateFromRange(
+      rtc::ArrayView<const int> preferred_sample_rates) = 0;
+
+  virtual ~OutputRateCalculator() {}
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_MIXER_OUTPUT_RATE_CALCULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_mixer/sine_wave_generator.cc b/third_party/libwebrtc/modules/audio_mixer/sine_wave_generator.cc
new file mode 100644
index 0000000000..591fe14e8c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/sine_wave_generator.cc
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_mixer/sine_wave_generator.h"
+
+#include <math.h>
+#include <stddef.h>
+
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+constexpr float kPi = 3.14159265f;
+}  // namespace
+
+void SineWaveGenerator::GenerateNextFrame(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_; ++i) {
+    for (size_t ch = 0; ch < frame->num_channels_; ++ch) {
+      frame_data[frame->num_channels_ * i + ch] =
+          rtc::saturated_cast<int16_t>(amplitude_ * sinf(phase_));
+    }
+    phase_ += wave_frequency_hz_ * 2 * kPi / frame->sample_rate_hz_;
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_mixer/sine_wave_generator.h b/third_party/libwebrtc/modules/audio_mixer/sine_wave_generator.h
new file mode 100644
index 0000000000..ec0fcd24bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/sine_wave_generator.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_MIXER_SINE_WAVE_GENERATOR_H_
+#define MODULES_AUDIO_MIXER_SINE_WAVE_GENERATOR_H_
+
+#include <stdint.h>
+
+#include "api/audio/audio_frame.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class SineWaveGenerator {
+ public:
+  SineWaveGenerator(float wave_frequency_hz, int16_t amplitude)
+      : wave_frequency_hz_(wave_frequency_hz), amplitude_(amplitude) {
+    RTC_DCHECK_GT(wave_frequency_hz, 0);
+  }
+
+  // Produces appropriate output based on frame->num_channels_,
+  // frame->sample_rate_hz_.
+  void GenerateNextFrame(AudioFrame* frame);
+
+ private:
+  float phase_ = 0.f;
+  const float wave_frequency_hz_;
+  const int16_t amplitude_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_MIXER_SINE_WAVE_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/BUILD.gn
new file mode 100644
index 0000000000..64e83a006b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/BUILD.gn
@@ -0,0 +1,677 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+if (rtc_enable_protobuf) {
+  import("//third_party/protobuf/proto_library.gni")
+}
+
+config("apm_debug_dump") {
+  if (apm_debug_dump) {
+    defines = [ "WEBRTC_APM_DEBUG_DUMP=1" ]
+  } else {
+    defines = [ "WEBRTC_APM_DEBUG_DUMP=0" ]
+  }
+}
+
+rtc_library("api") {
+  visibility = [ "*" ]
+  sources = [
+    "include/audio_processing.cc",
+    "include/audio_processing.h",
+  ]
+  deps = [
+    ":audio_frame_view",
+    ":audio_processing_statistics",
+    "../../api:array_view",
+    "../../api:scoped_refptr",
+    "../../api/audio:aec3_config",
+    "../../api/audio:audio_frame_api",
+    "../../api/audio:echo_control",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:refcount",
+    "../../rtc_base:stringutils",
+    "../../rtc_base/system:arch",
+    "../../rtc_base/system:file_wrapper",
+    "../../rtc_base/system:rtc_export",
+    "agc:gain_control_interface",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_frame_proxies") {
+  visibility = [ "*" ]
+  sources = [
+    "include/audio_frame_proxies.cc",
+    "include/audio_frame_proxies.h",
+  ]
+  deps = [
+    ":api",
+    ":audio_frame_view",
+    "../../api/audio:audio_frame_api",
+  ]
+}
+
+rtc_library("audio_buffer") {
+  visibility = [ "*" ]
+
+  configs += [ ":apm_debug_dump" ]
+
+  sources = [
+    "audio_buffer.cc",
+    "audio_buffer.h",
+    "splitting_filter.cc",
+    "splitting_filter.h",
+    "three_band_filter_bank.cc",
+    "three_band_filter_bank.h",
+  ]
+
+  defines = []
+
+  deps = [
+    ":api",
+    "../../api:array_view",
+    "../../common_audio",
+    "../../common_audio:common_audio_c",
+    "../../rtc_base:checks",
+  ]
+}
+
+rtc_library("high_pass_filter") {
+  visibility = [ "*" ]
+
+  sources = [
+    "high_pass_filter.cc",
+    "high_pass_filter.h",
+  ]
+
+  defines = []
+
+  deps = [
+    ":audio_buffer",
+    "../../api:array_view",
+    "../../rtc_base:checks",
+    "utility:cascaded_biquad_filter",
+  ]
+}
+
+rtc_source_set("aec_dump_interface") {
+  visibility = [ "*" ]
+  sources = [
+    "include/aec_dump.cc",
+    "include/aec_dump.h",
+  ]
+
+  deps = [
+    ":api",
+    ":audio_frame_view",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("gain_controller2") {
+  configs += [ ":apm_debug_dump" ]
+  sources = [
+    "gain_controller2.cc",
+    "gain_controller2.h",
+  ]
+  defines = []
+  deps = [
+    ":aec_dump_interface",
+    ":api",
+    ":apm_logging",
+    ":audio_buffer",
+    ":audio_frame_view",
+    "../../common_audio",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:stringutils",
+    "../../system_wrappers:field_trial",
+    "agc2:adaptive_digital_gain_controller",
+    "agc2:common",
+    "agc2:cpu_features",
+    "agc2:fixed_digital",
+    "agc2:gain_applier",
+    "agc2:input_volume_controller",
+    "agc2:noise_level_estimator",
+    "agc2:saturation_protector",
+    "agc2:speech_level_estimator",
+    "agc2:vad_wrapper",
+  ]
+}
+
+rtc_library("audio_processing") {
+  visibility = [ "*" ]
+  configs += [ ":apm_debug_dump" ]
+  sources = [
+    "audio_processing_builder_impl.cc",
+    "audio_processing_impl.cc",
+    "audio_processing_impl.h",
+    "echo_control_mobile_impl.cc",
+    "echo_control_mobile_impl.h",
+    "gain_control_impl.cc",
+    "gain_control_impl.h",
+    "render_queue_item_verifier.h",
+  ]
+
+  defines = []
+  deps = [
+    ":aec_dump_interface",
+    ":api",
+    ":apm_logging",
+    ":audio_buffer",
+    ":audio_frame_proxies",
+    ":audio_frame_view",
+    ":audio_processing_statistics",
+    ":gain_controller2",
+    ":high_pass_filter",
+    ":optionally_built_submodule_creators",
+    ":rms_level",
+    "../../api:array_view",
+    "../../api:function_view",
+    "../../api:make_ref_counted",
+    "../../api/audio:aec3_config",
+    "../../api/audio:audio_frame_api",
+    "../../api/audio:echo_control",
+    "../../audio/utility:audio_frame_operations",
+    "../../common_audio:common_audio_c",
+    "../../common_audio/third_party/ooura:fft_size_256",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:gtest_prod",
+    "../../rtc_base:ignore_wundef",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base:sanitizer",
+    "../../rtc_base:swap_queue",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:rtc_export",
+    "../../system_wrappers",
+    "../../system_wrappers:denormal_disabler",
+    "../../system_wrappers:field_trial",
+    "../../system_wrappers:metrics",
+    "aec3",
+    "aec_dump:aec_dump",
+    "aecm:aecm_core",
+    "agc",
+    "agc:gain_control_interface",
+    "agc:legacy_agc",
+    "agc2:input_volume_stats_reporter",
+    "capture_levels_adjuster",
+    "ns",
+    "transient:transient_suppressor_api",
+    "vad",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+
+  deps += [
+    "../../common_audio",
+    "../../common_audio:fir_filter",
+    "../../common_audio:fir_filter_factory",
+    "../../system_wrappers",
+  ]
+
+  if (rtc_enable_protobuf) {
+    deps += [ "aec_dump:aec_dump_impl" ]
+  } else {
+    deps += [ "aec_dump:null_aec_dump_factory" ]
+  }
+}
+
+rtc_library("residual_echo_detector") {
+  poisonous = [ "default_echo_detector" ]
+  configs += [ ":apm_debug_dump" ]
+  sources = [
+    "echo_detector/circular_buffer.cc",
+    "echo_detector/circular_buffer.h",
+    "echo_detector/mean_variance_estimator.cc",
+    "echo_detector/mean_variance_estimator.h",
+    "echo_detector/moving_max.cc",
+    "echo_detector/moving_max.h",
+    "echo_detector/normalized_covariance_estimator.cc",
+    "echo_detector/normalized_covariance_estimator.h",
+    "residual_echo_detector.cc",
+    "residual_echo_detector.h",
+  ]
+  deps = [
+    ":api",
+    ":apm_logging",
+    "../../api:array_view",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../system_wrappers:metrics",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("optionally_built_submodule_creators") {
+  sources = [
+    "optionally_built_submodule_creators.cc",
+    "optionally_built_submodule_creators.h",
+  ]
+  deps = [
+    "transient:transient_suppressor_api",
+    "transient:transient_suppressor_impl",
+  ]
+}
+
+rtc_source_set("rms_level") {
+  visibility = [ "*" ]
+  sources = [
+    "rms_level.cc",
+    "rms_level.h",
+  ]
+  deps = [
+    "../../api:array_view",
+    "../../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_processing_statistics") {
+  visibility = [ "*" ]
+  sources = [
+    "include/audio_processing_statistics.cc",
+    "include/audio_processing_statistics.h",
+  ]
+  deps = [ "../../rtc_base/system:rtc_export" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_source_set("audio_frame_view") {
+  sources = [ "include/audio_frame_view.h" ]
+  deps = [ "../../api:array_view" ]
+}
+
+if (rtc_enable_protobuf) {
+  proto_library("audioproc_debug_proto") {
+    sources = [ "debug.proto" ]
+
+    proto_out_dir = "modules/audio_processing"
+  }
+}
+
+rtc_library("apm_logging") {
+  configs += [ ":apm_debug_dump" ]
+  sources = [
+    "logging/apm_data_dumper.cc",
+    "logging/apm_data_dumper.h",
+  ]
+  deps = [
+    "../../api:array_view",
+    "../../common_audio",
+    "../../rtc_base:checks",
+    "../../rtc_base:stringutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  defines = []
+}
+
+if (rtc_include_tests) {
+  rtc_source_set("mocks") {
+    testonly = true
+    sources = [ "include/mock_audio_processing.h" ]
+    deps = [
+      ":aec_dump_interface",
+      ":api",
+      ":audio_buffer",
+      ":audio_processing",
+      ":audio_processing_statistics",
+      "../../test:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  if (!build_with_chromium) {
+    group("audio_processing_tests") {
+      testonly = true
+      deps = [
+        ":audioproc_test_utils",
+        "transient:click_annotate",
+        "transient:transient_suppression_test",
+      ]
+
+      if (rtc_enable_protobuf) {
+        deps += [
+          ":audioproc_unittest_proto",
+          "aec_dump:aec_dump_unittests",
+          "test/conversational_speech",
+          "test/py_quality_assessment",
+        ]
+      }
+    }
+
+    rtc_library("audio_processing_unittests") {
+      testonly = true
+
+      configs += [ ":apm_debug_dump" ]
+      sources = [
+        "audio_buffer_unittest.cc",
+        "audio_frame_view_unittest.cc",
+        "echo_control_mobile_unittest.cc",
+        "gain_controller2_unittest.cc",
+        "splitting_filter_unittest.cc",
+        "test/fake_recording_device_unittest.cc",
+      ]
+
+      deps = [
+        ":analog_mic_simulation",
+        ":api",
+        ":apm_logging",
+        ":audio_buffer",
+        ":audio_frame_view",
+        ":audio_processing",
+        ":audioproc_test_utils",
+        ":gain_controller2",
+        ":high_pass_filter",
+        ":mocks",
+        "../../api:array_view",
+        "../../api:make_ref_counted",
+        "../../api:scoped_refptr",
+        "../../api/audio:aec3_config",
+        "../../api/audio:aec3_factory",
+        "../../api/audio:echo_detector_creator",
+        "../../common_audio",
+        "../../common_audio:common_audio_c",
+        "../../rtc_base:checks",
+        "../../rtc_base:gtest_prod",
+        "../../rtc_base:ignore_wundef",
+        "../../rtc_base:macromagic",
+        "../../rtc_base:platform_thread",
+        "../../rtc_base:protobuf_utils",
+        "../../rtc_base:random",
+        "../../rtc_base:rtc_base_tests_utils",
+        "../../rtc_base:rtc_event",
+        "../../rtc_base:safe_conversions",
+        "../../rtc_base:safe_minmax",
+        "../../rtc_base:stringutils",
+        "../../rtc_base:swap_queue",
+        "../../rtc_base:task_queue_for_test",
+        "../../rtc_base:threading",
+        "../../rtc_base/synchronization:mutex",
+        "../../rtc_base/system:arch",
+        "../../rtc_base/system:file_wrapper",
+        "../../system_wrappers",
+        "../../system_wrappers:denormal_disabler",
+        "../../test:field_trial",
+        "../../test:fileutils",
+        "../../test:rtc_expect_death",
+        "../../test:test_support",
+        "../audio_coding:neteq_input_audio_tools",
+        "aec_dump:mock_aec_dump_unittests",
+        "agc:agc_unittests",
+        "agc2:adaptive_digital_gain_controller_unittest",
+        "agc2:biquad_filter_unittests",
+        "agc2:fixed_digital_unittests",
+        "agc2:gain_applier_unittest",
+        "agc2:input_volume_controller_unittests",
+        "agc2:input_volume_stats_reporter_unittests",
+        "agc2:noise_estimator_unittests",
+        "agc2:saturation_protector_unittest",
+        "agc2:speech_level_estimator_unittest",
+        "agc2:test_utils",
+        "agc2:vad_wrapper_unittests",
+        "agc2/rnn_vad:unittests",
+        "capture_levels_adjuster",
+        "capture_levels_adjuster:capture_levels_adjuster_unittests",
+        "test/conversational_speech:unittest",
+        "transient:transient_suppression_unittests",
+        "utility:legacy_delay_estimator_unittest",
+        "utility:pffft_wrapper_unittest",
+        "vad:vad_unittests",
+        "//testing/gtest",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/strings",
+        "//third_party/abseil-cpp/absl/types:optional",
+      ]
+
+      defines = []
+
+      if (rtc_prefer_fixed_point) {
+        defines += [ "WEBRTC_AUDIOPROC_FIXED_PROFILE" ]
+      } else {
+        defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ]
+      }
+
+      if (rtc_enable_protobuf) {
+        defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
+        deps += [
+          ":audioproc_debug_proto",
+          ":audioproc_protobuf_utils",
+          ":audioproc_test_utils",
+          ":audioproc_unittest_proto",
+          ":optionally_built_submodule_creators",
+          ":residual_echo_detector",
+          ":rms_level",
+          ":runtime_settings_protobuf_utils",
+          "../../api/audio:audio_frame_api",
+          "../../api/audio:echo_control",
+          "../../rtc_base:rtc_base_tests_utils",
+          "../../rtc_base:rtc_task_queue",
+          "aec_dump",
+          "aec_dump:aec_dump_unittests",
+        ]
+        absl_deps += [ "//third_party/abseil-cpp/absl/flags:flag" ]
+        sources += [
+          "audio_processing_impl_locking_unittest.cc",
+          "audio_processing_impl_unittest.cc",
+          "audio_processing_unittest.cc",
+          "echo_control_mobile_bit_exact_unittest.cc",
+          "echo_detector/circular_buffer_unittest.cc",
+          "echo_detector/mean_variance_estimator_unittest.cc",
+          "echo_detector/moving_max_unittest.cc",
+          "echo_detector/normalized_covariance_estimator_unittest.cc",
+          "gain_control_unittest.cc",
+          "high_pass_filter_unittest.cc",
+          "residual_echo_detector_unittest.cc",
+          "rms_level_unittest.cc",
+          "test/debug_dump_replayer.cc",
+          "test/debug_dump_replayer.h",
+          "test/debug_dump_test.cc",
+          "test/echo_canceller_test_tools.cc",
+          "test/echo_canceller_test_tools.h",
+          "test/echo_canceller_test_tools_unittest.cc",
+          "test/echo_control_mock.h",
+          "test/test_utils.h",
+        ]
+      }
+    }
+  }
+
+  rtc_library("audio_processing_perf_tests") {
+    testonly = true
+    configs += [ ":apm_debug_dump" ]
+
+    sources = [ "audio_processing_performance_unittest.cc" ]
+    deps = [
+      ":audio_processing",
+      ":audioproc_test_utils",
+      "../../api:array_view",
+      "../../api/numerics",
+      "../../api/test/metrics:global_metrics_logger_and_exporter",
+      "../../api/test/metrics:metric",
+      "../../rtc_base:platform_thread",
+      "../../rtc_base:protobuf_utils",
+      "../../rtc_base:random",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:safe_conversions",
+      "../../system_wrappers",
+      "../../test:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  rtc_library("analog_mic_simulation") {
+    sources = [
+      "test/fake_recording_device.cc",
+      "test/fake_recording_device.h",
+    ]
+    deps = [
+      "../../api:array_view",
+      "../../api/audio:audio_frame_api",
+      "../../common_audio",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:safe_conversions",
+      "../../rtc_base:safe_minmax",
+      "agc2:gain_map",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
+  if (rtc_enable_protobuf && !build_with_chromium) {
+    rtc_library("audioproc_f_impl") {
+      testonly = true
+      configs += [ ":apm_debug_dump" ]
+      sources = [
+        "test/aec_dump_based_simulator.cc",
+        "test/aec_dump_based_simulator.h",
+        "test/api_call_statistics.cc",
+        "test/api_call_statistics.h",
+        "test/audio_processing_simulator.cc",
+        "test/audio_processing_simulator.h",
+        "test/audioproc_float_impl.cc",
+        "test/audioproc_float_impl.h",
+        "test/wav_based_simulator.cc",
+        "test/wav_based_simulator.h",
+      ]
+
+      deps = [
+        ":analog_mic_simulation",
+        ":api",
+        ":apm_logging",
+        ":audio_processing",
+        ":audioproc_debug_proto",
+        ":audioproc_protobuf_utils",
+        ":audioproc_test_utils",
+        ":runtime_settings_protobuf_utils",
+        "../../api/audio:aec3_config_json",
+        "../../api/audio:aec3_factory",
+        "../../api/audio:echo_detector_creator",
+        "../../common_audio",
+        "../../rtc_base:checks",
+        "../../rtc_base:ignore_wundef",
+        "../../rtc_base:logging",
+        "../../rtc_base:protobuf_utils",
+        "../../rtc_base:rtc_json",
+        "../../rtc_base:safe_conversions",
+        "../../rtc_base:stringutils",
+        "../../rtc_base:task_queue_for_test",
+        "../../rtc_base:timeutils",
+        "../../rtc_base/system:file_wrapper",
+        "../../system_wrappers",
+        "../../system_wrappers:field_trial",
+        "../../test:test_support",
+        "aec_dump",
+        "aec_dump:aec_dump_impl",
+        "//testing/gtest",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/flags:parse",
+        "//third_party/abseil-cpp/absl/strings",
+        "//third_party/abseil-cpp/absl/types:optional",
+      ]
+    }  # audioproc_f_impl
+  }
+
+  if (rtc_enable_protobuf) {
+    proto_library("audioproc_unittest_proto") {
+      sources = [ "test/unittest.proto" ]
+      proto_out_dir = "modules/audio_processing/test"
+    }
+
+    rtc_library("audioproc_protobuf_utils") {
+      sources = [
+        "test/protobuf_utils.cc",
+        "test/protobuf_utils.h",
+      ]
+
+      deps = [
+        ":audioproc_debug_proto",
+        "../../rtc_base:checks",
+        "../../rtc_base:ignore_wundef",
+        "../../rtc_base:protobuf_utils",
+        "../../rtc_base/system:arch",
+      ]
+    }
+
+    rtc_library("runtime_settings_protobuf_utils") {
+      testonly = true
+      sources = [
+        "test/runtime_setting_util.cc",
+        "test/runtime_setting_util.h",
+      ]
+
+      deps = [
+        ":api",
+        ":audioproc_debug_proto",
+        ":audioproc_protobuf_utils",
+        "../../rtc_base:checks",
+      ]
+    }
+  }
+}
+
+rtc_library("audioproc_test_utils") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "test/audio_buffer_tools.cc",
+    "test/audio_buffer_tools.h",
+    "test/audio_processing_builder_for_testing.cc",
+    "test/audio_processing_builder_for_testing.h",
+    "test/bitexactness_tools.cc",
+    "test/bitexactness_tools.h",
+    "test/performance_timer.cc",
+    "test/performance_timer.h",
+    "test/simulator_buffers.cc",
+    "test/simulator_buffers.h",
+    "test/test_utils.cc",
+    "test/test_utils.h",
+  ]
+
+  configs += [ ":apm_debug_dump" ]
+
+  deps = [
+    ":api",
+    ":audio_buffer",
+    ":audio_processing",
+    "../../api:array_view",
+    "../../api/audio:audio_frame_api",
+    "../../common_audio",
+    "../../rtc_base:checks",
+    "../../rtc_base:random",
+    "../../rtc_base/system:arch",
+    "../../system_wrappers",
+    "../../test:fileutils",
+    "../../test:test_support",
+    "../audio_coding:neteq_input_audio_tools",
+    "//testing/gtest",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/DEPS b/third_party/libwebrtc/modules/audio_processing/DEPS
new file mode 100644
index 0000000000..79fd071785
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/DEPS
@@ -0,0 +1,14 @@
+include_rules = [
+  "+audio/utility/audio_frame_operations.h",
+  "+common_audio",
+  "+system_wrappers",
+]
+
+specific_include_rules = {
+  ".*test\.cc": [
+    "+rtc_tools",
+    # Android platform build has different paths.
+    "+gtest",
+    "+external/webrtc",
+  ],
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/OWNERS b/third_party/libwebrtc/modules/audio_processing/OWNERS
new file mode 100644
index 0000000000..f5dc59ea35
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/OWNERS
@@ -0,0 +1,8 @@
+alessiob@webrtc.org
+devicentepena@webrtc.org
+gustaf@webrtc.org
+henrik.lundin@webrtc.org
+ivoc@webrtc.org
+lionelk@webrtc.org
+peah@webrtc.org
+saza@webrtc.org
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/aec3/BUILD.gn
new file mode 100644
index 0000000000..c29b893b7d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/BUILD.gn
@@ -0,0 +1,384 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("aec3") {
+  visibility = [ "*" ]
+  configs += [ "..:apm_debug_dump" ]
+  sources = [
+    "adaptive_fir_filter.cc",
+    "adaptive_fir_filter_erl.cc",
+    "aec3_common.cc",
+    "aec3_fft.cc",
+    "aec_state.cc",
+    "aec_state.h",
+    "alignment_mixer.cc",
+    "alignment_mixer.h",
+    "api_call_jitter_metrics.cc",
+    "api_call_jitter_metrics.h",
+    "block.h",
+    "block_buffer.cc",
+    "block_delay_buffer.cc",
+    "block_delay_buffer.h",
+    "block_framer.cc",
+    "block_framer.h",
+    "block_processor.cc",
+    "block_processor.h",
+    "block_processor_metrics.cc",
+    "block_processor_metrics.h",
+    "clockdrift_detector.cc",
+    "clockdrift_detector.h",
+    "coarse_filter_update_gain.cc",
+    "coarse_filter_update_gain.h",
+    "comfort_noise_generator.cc",
+    "comfort_noise_generator.h",
+    "config_selector.cc",
+    "config_selector.h",
+    "decimator.cc",
+    "decimator.h",
+    "delay_estimate.h",
+    "dominant_nearend_detector.cc",
+    "dominant_nearend_detector.h",
+    "downsampled_render_buffer.cc",
+    "downsampled_render_buffer.h",
+    "echo_audibility.cc",
+    "echo_audibility.h",
+    "echo_canceller3.cc",
+    "echo_canceller3.h",
+    "echo_path_delay_estimator.cc",
+    "echo_path_delay_estimator.h",
+    "echo_path_variability.cc",
+    "echo_path_variability.h",
+    "echo_remover.cc",
+    "echo_remover.h",
+    "echo_remover_metrics.cc",
+    "echo_remover_metrics.h",
+    "erl_estimator.cc",
+    "erl_estimator.h",
+    "erle_estimator.cc",
+    "erle_estimator.h",
+    "fft_buffer.cc",
+    "filter_analyzer.cc",
+    "filter_analyzer.h",
+    "frame_blocker.cc",
+    "frame_blocker.h",
+    "fullband_erle_estimator.cc",
+    "fullband_erle_estimator.h",
+    "matched_filter.cc",
+    "matched_filter_lag_aggregator.cc",
+    "matched_filter_lag_aggregator.h",
+    "moving_average.cc",
+    "moving_average.h",
+    "multi_channel_content_detector.cc",
+    "multi_channel_content_detector.h",
+    "nearend_detector.h",
+    "refined_filter_update_gain.cc",
+    "refined_filter_update_gain.h",
+    "render_buffer.cc",
+    "render_delay_buffer.cc",
+    "render_delay_buffer.h",
+    "render_delay_controller.cc",
+    "render_delay_controller.h",
+    "render_delay_controller_metrics.cc",
+    "render_delay_controller_metrics.h",
+    "render_signal_analyzer.cc",
+    "render_signal_analyzer.h",
+    "residual_echo_estimator.cc",
+    "residual_echo_estimator.h",
+    "reverb_decay_estimator.cc",
+    "reverb_decay_estimator.h",
+    "reverb_frequency_response.cc",
+    "reverb_frequency_response.h",
+    "reverb_model.cc",
+    "reverb_model.h",
+    "reverb_model_estimator.cc",
+    "reverb_model_estimator.h",
+    "signal_dependent_erle_estimator.cc",
+    "signal_dependent_erle_estimator.h",
+    "spectrum_buffer.cc",
+    "stationarity_estimator.cc",
+    "stationarity_estimator.h",
+    "subband_erle_estimator.cc",
+    "subband_erle_estimator.h",
+    "subband_nearend_detector.cc",
+    "subband_nearend_detector.h",
+    "subtractor.cc",
+    "subtractor.h",
+    "subtractor_output.cc",
+    "subtractor_output.h",
+    "subtractor_output_analyzer.cc",
+    "subtractor_output_analyzer.h",
+    "suppression_filter.cc",
+    "suppression_filter.h",
+    "suppression_gain.cc",
+    "suppression_gain.h",
+    "transparent_mode.cc",
+    "transparent_mode.h",
+  ]
+
+  defines = []
+  if (rtc_build_with_neon && target_cpu != "arm64") {
+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+    cflags = [ "-mfpu=neon" ]
+  }
+
+  deps = [
+    ":adaptive_fir_filter",
+    ":adaptive_fir_filter_erl",
+    ":aec3_common",
+    ":aec3_fft",
+    ":fft_data",
+    ":matched_filter",
+    ":render_buffer",
+    ":vector_math",
+    "..:apm_logging",
+    "..:audio_buffer",
+    "..:high_pass_filter",
+    "../../../api:array_view",
+    "../../../api/audio:aec3_config",
+    "../../../api/audio:echo_control",
+    "../../../common_audio:common_audio_c",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:macromagic",
+    "../../../rtc_base:race_checker",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:swap_queue",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../rtc_base/system:arch",
+    "../../../system_wrappers",
+    "../../../system_wrappers:field_trial",
+    "../../../system_wrappers:metrics",
+    "../utility:cascaded_biquad_filter",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+
+  if (target_cpu == "x86" || target_cpu == "x64") {
+    deps += [ ":aec3_avx2" ]
+  }
+}
+
+rtc_source_set("aec3_common") {
+  sources = [ "aec3_common.h" ]
+}
+
+rtc_source_set("aec3_fft") {
+  sources = [ "aec3_fft.h" ]
+  deps = [
+    ":aec3_common",
+    ":fft_data",
+    "../../../api:array_view",
+    "../../../common_audio/third_party/ooura:fft_size_128",
+    "../../../rtc_base:checks",
+    "../../../rtc_base/system:arch",
+  ]
+}
+
+rtc_source_set("render_buffer") {
+  sources = [
+    "block.h",
+    "block_buffer.h",
+    "fft_buffer.h",
+    "render_buffer.h",
+    "spectrum_buffer.h",
+  ]
+  deps = [
+    ":aec3_common",
+    ":fft_data",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base/system:arch",
+  ]
+}
+
+rtc_source_set("adaptive_fir_filter") {
+  sources = [ "adaptive_fir_filter.h" ]
+  deps = [
+    ":aec3_common",
+    ":aec3_fft",
+    ":fft_data",
+    ":render_buffer",
+    "..:apm_logging",
+    "../../../api:array_view",
+    "../../../rtc_base/system:arch",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_source_set("adaptive_fir_filter_erl") {
+  sources = [ "adaptive_fir_filter_erl.h" ]
+  deps = [
+    ":aec3_common",
+    "../../../api:array_view",
+    "../../../rtc_base/system:arch",
+  ]
+}
+
+rtc_source_set("matched_filter") {
+  sources = [ "matched_filter.h" ]
+  deps = [
+    ":aec3_common",
+    "../../../api:array_view",
+    "../../../rtc_base:gtest_prod",
+    "../../../rtc_base/system:arch",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_source_set("vector_math") {
+  sources = [ "vector_math.h" ]
+  deps = [
+    ":aec3_common",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base/system:arch",
+  ]
+}
+
+rtc_source_set("fft_data") {
+  sources = [ "fft_data.h" ]
+  deps = [
+    ":aec3_common",
+    "../../../api:array_view",
+    "../../../rtc_base/system:arch",
+  ]
+}
+
+if (target_cpu == "x86" || target_cpu == "x64") {
+  rtc_library("aec3_avx2") {
+    configs += [ "..:apm_debug_dump" ]
+    sources = [
+      "adaptive_fir_filter_avx2.cc",
+      "adaptive_fir_filter_erl_avx2.cc",
+      "fft_data_avx2.cc",
+      "matched_filter_avx2.cc",
+      "vector_math_avx2.cc",
+    ]
+
+    cflags = [
+      "-mavx",
+      "-mavx2",
+      "-mfma",
+    ]
+
+    deps = [
+      ":adaptive_fir_filter",
+      ":adaptive_fir_filter_erl",
+      ":fft_data",
+      ":matched_filter",
+      ":vector_math",
+      "../../../api:array_view",
+      "../../../rtc_base:checks",
+    ]
+  }
+}
+
+if (rtc_include_tests) {
+  rtc_library("aec3_unittests") {
+    testonly = true
+
+    configs += [ "..:apm_debug_dump" ]
+    sources = [
+      "mock/mock_block_processor.cc",
+      "mock/mock_block_processor.h",
+      "mock/mock_echo_remover.cc",
+      "mock/mock_echo_remover.h",
+      "mock/mock_render_delay_buffer.cc",
+      "mock/mock_render_delay_buffer.h",
+      "mock/mock_render_delay_controller.cc",
+      "mock/mock_render_delay_controller.h",
+    ]
+
+    deps = [
+      ":adaptive_fir_filter",
+      ":adaptive_fir_filter_erl",
+      ":aec3",
+      ":aec3_common",
+      ":aec3_fft",
+      ":fft_data",
+      ":matched_filter",
+      ":render_buffer",
+      ":vector_math",
+      "..:apm_logging",
+      "..:audio_buffer",
+      "..:audio_processing",
+      "..:high_pass_filter",
+      "../../../api:array_view",
+      "../../../api/audio:aec3_config",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:macromagic",
+      "../../../rtc_base:random",
+      "../../../rtc_base:safe_minmax",
+      "../../../rtc_base:stringutils",
+      "../../../rtc_base/system:arch",
+      "../../../system_wrappers",
+      "../../../system_wrappers:metrics",
+      "../../../test:field_trial",
+      "../../../test:test_support",
+      "../utility:cascaded_biquad_filter",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+
+    defines = []
+
+    if (rtc_enable_protobuf) {
+      sources += [
+        "adaptive_fir_filter_erl_unittest.cc",
+        "adaptive_fir_filter_unittest.cc",
+        "aec3_fft_unittest.cc",
+        "aec_state_unittest.cc",
+        "alignment_mixer_unittest.cc",
+        "api_call_jitter_metrics_unittest.cc",
+        "block_delay_buffer_unittest.cc",
+        "block_framer_unittest.cc",
+        "block_processor_metrics_unittest.cc",
+        "block_processor_unittest.cc",
+        "clockdrift_detector_unittest.cc",
+        "coarse_filter_update_gain_unittest.cc",
+        "comfort_noise_generator_unittest.cc",
+        "config_selector_unittest.cc",
+        "decimator_unittest.cc",
+        "echo_canceller3_unittest.cc",
+        "echo_path_delay_estimator_unittest.cc",
+        "echo_path_variability_unittest.cc",
+        "echo_remover_metrics_unittest.cc",
+        "echo_remover_unittest.cc",
+        "erl_estimator_unittest.cc",
+        "erle_estimator_unittest.cc",
+        "fft_data_unittest.cc",
+        "filter_analyzer_unittest.cc",
+        "frame_blocker_unittest.cc",
+        "matched_filter_lag_aggregator_unittest.cc",
+        "matched_filter_unittest.cc",
+        "moving_average_unittest.cc",
+        "multi_channel_content_detector_unittest.cc",
+        "refined_filter_update_gain_unittest.cc",
+        "render_buffer_unittest.cc",
+        "render_delay_buffer_unittest.cc",
+        "render_delay_controller_metrics_unittest.cc",
+        "render_delay_controller_unittest.cc",
+        "render_signal_analyzer_unittest.cc",
+        "residual_echo_estimator_unittest.cc",
+        "reverb_model_estimator_unittest.cc",
+        "signal_dependent_erle_estimator_unittest.cc",
+        "subtractor_unittest.cc",
+        "suppression_filter_unittest.cc",
+        "suppression_gain_unittest.cc",
+        "vector_math_unittest.cc",
+      ]
+    }
+
+    if (!build_with_chromium) {
+      deps += [ "..:audio_processing_unittests" ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
new file mode 100644
index 0000000000..917aa951ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
@@ -0,0 +1,744 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
+
+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_HAS_NEON)
+#include <arm_neon.h>
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+#include <math.h>
+
+#include <algorithm>
+#include <functional>
+
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace aec3 {
+
+// Computes and stores the frequency response of the filter.
+void ComputeFrequencyResponse(
+    size_t num_partitions,
+    const std::vector<std::vector<FftData>>& H,
+    std::vector<std::array<float, kFftLengthBy2Plus1>>* H2) {
+  for (auto& H2_ch : *H2) {
+    H2_ch.fill(0.f);
+  }
+
+  const size_t num_render_channels = H[0].size();
+  RTC_DCHECK_EQ(H.size(), H2->capacity());
+  for (size_t p = 0; p < num_partitions; ++p) {
+    RTC_DCHECK_EQ(kFftLengthBy2Plus1, (*H2)[p].size());
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      for (size_t j = 0; j < kFftLengthBy2Plus1; ++j) {
+        float tmp =
+            H[p][ch].re[j] * H[p][ch].re[j] + H[p][ch].im[j] * H[p][ch].im[j];
+        (*H2)[p][j] = std::max((*H2)[p][j], tmp);
+      }
+    }
+  }
+}
+
+#if defined(WEBRTC_HAS_NEON)
+// Computes and stores the frequency response of the filter.
+void ComputeFrequencyResponse_Neon(
+    size_t num_partitions,
+    const std::vector<std::vector<FftData>>& H,
+    std::vector<std::array<float, kFftLengthBy2Plus1>>* H2) {
+  for (auto& H2_ch : *H2) {
+    H2_ch.fill(0.f);
+  }
+
+  const size_t num_render_channels = H[0].size();
+  RTC_DCHECK_EQ(H.size(), H2->capacity());
+  for (size_t p = 0; p < num_partitions; ++p) {
+    RTC_DCHECK_EQ(kFftLengthBy2Plus1, (*H2)[p].size());
+    auto& H2_p = (*H2)[p];
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      const FftData& H_p_ch = H[p][ch];
+      for (size_t j = 0; j < kFftLengthBy2; j += 4) {
+        const float32x4_t re = vld1q_f32(&H_p_ch.re[j]);
+        const float32x4_t im = vld1q_f32(&H_p_ch.im[j]);
+        float32x4_t H2_new = vmulq_f32(re, re);
+        H2_new = vmlaq_f32(H2_new, im, im);
+        float32x4_t H2_p_j = vld1q_f32(&H2_p[j]);
+        H2_p_j = vmaxq_f32(H2_p_j, H2_new);
+        vst1q_f32(&H2_p[j], H2_p_j);
+      }
+      float H2_new = H_p_ch.re[kFftLengthBy2] * H_p_ch.re[kFftLengthBy2] +
+                     H_p_ch.im[kFftLengthBy2] * H_p_ch.im[kFftLengthBy2];
+      H2_p[kFftLengthBy2] = std::max(H2_p[kFftLengthBy2], H2_new);
+    }
+  }
+}
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Computes and stores the frequency response of the filter.
+void ComputeFrequencyResponse_Sse2(
+    size_t num_partitions,
+    const std::vector<std::vector<FftData>>& H,
+    std::vector<std::array<float, kFftLengthBy2Plus1>>* H2) {
+  for (auto& H2_ch : *H2) {
+    H2_ch.fill(0.f);
+  }
+
+  const size_t num_render_channels = H[0].size();
+  RTC_DCHECK_EQ(H.size(), H2->capacity());
+  // constexpr __mmmask8 kMaxMask = static_cast<__mmmask8>(256u);
+  for (size_t p = 0; p < num_partitions; ++p) {
+    RTC_DCHECK_EQ(kFftLengthBy2Plus1, (*H2)[p].size());
+    auto& H2_p = (*H2)[p];
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      const FftData& H_p_ch = H[p][ch];
+      for (size_t j = 0; j < kFftLengthBy2; j += 4) {
+        const __m128 re = _mm_loadu_ps(&H_p_ch.re[j]);
+        const __m128 re2 = _mm_mul_ps(re, re);
+        const __m128 im = _mm_loadu_ps(&H_p_ch.im[j]);
+        const __m128 im2 = _mm_mul_ps(im, im);
+        const __m128 H2_new = _mm_add_ps(re2, im2);
+        __m128 H2_k_j = _mm_loadu_ps(&H2_p[j]);
+        H2_k_j = _mm_max_ps(H2_k_j, H2_new);
+        _mm_storeu_ps(&H2_p[j], H2_k_j);
+      }
+      float H2_new = H_p_ch.re[kFftLengthBy2] * H_p_ch.re[kFftLengthBy2] +
+                     H_p_ch.im[kFftLengthBy2] * H_p_ch.im[kFftLengthBy2];
+      H2_p[kFftLengthBy2] = std::max(H2_p[kFftLengthBy2], H2_new);
+    }
+  }
+}
+#endif
+
+// Adapts the filter partitions as H(t+1)=H(t)+G(t)*conj(X(t)).
+void AdaptPartitions(const RenderBuffer& render_buffer,
+                     const FftData& G,
+                     size_t num_partitions,
+                     std::vector<std::vector<FftData>>* H) {
+  rtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+      render_buffer.GetFftBuffer();
+  size_t index = render_buffer.Position();
+  const size_t num_render_channels = render_buffer_data[index].size();
+  for (size_t p = 0; p < num_partitions; ++p) {
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      const FftData& X_p_ch = render_buffer_data[index][ch];
+      FftData& H_p_ch = (*H)[p][ch];
+      for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+        H_p_ch.re[k] += X_p_ch.re[k] * G.re[k] + X_p_ch.im[k] * G.im[k];
+        H_p_ch.im[k] += X_p_ch.re[k] * G.im[k] - X_p_ch.im[k] * G.re[k];
+      }
+    }
+    index = index < (render_buffer_data.size() - 1) ? index + 1 : 0;
+  }
+}
+
+#if defined(WEBRTC_HAS_NEON)
+// Adapts the filter partitions. (Neon variant)
+void AdaptPartitions_Neon(const RenderBuffer& render_buffer,
+                          const FftData& G,
+                          size_t num_partitions,
+                          std::vector<std::vector<FftData>>* H) {
+  rtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+      render_buffer.GetFftBuffer();
+  const size_t num_render_channels = render_buffer_data[0].size();
+  const size_t lim1 = std::min(
+      render_buffer_data.size() - render_buffer.Position(), num_partitions);
+  const size_t lim2 = num_partitions;
+  constexpr size_t kNumFourBinBands = kFftLengthBy2 / 4;
+
+  size_t X_partition = render_buffer.Position();
+  size_t limit = lim1;
+  size_t p = 0;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        FftData& H_p_ch = (*H)[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+        for (size_t k = 0, n = 0; n < kNumFourBinBands; ++n, k += 4) {
+          const float32x4_t G_re = vld1q_f32(&G.re[k]);
+          const float32x4_t G_im = vld1q_f32(&G.im[k]);
+          const float32x4_t X_re = vld1q_f32(&X.re[k]);
+          const float32x4_t X_im = vld1q_f32(&X.im[k]);
+          const float32x4_t H_re = vld1q_f32(&H_p_ch.re[k]);
+          const float32x4_t H_im = vld1q_f32(&H_p_ch.im[k]);
+          const float32x4_t a = vmulq_f32(X_re, G_re);
+          const float32x4_t e = vmlaq_f32(a, X_im, G_im);
+          const float32x4_t c = vmulq_f32(X_re, G_im);
+          const float32x4_t f = vmlsq_f32(c, X_im, G_re);
+          const float32x4_t g = vaddq_f32(H_re, e);
+          const float32x4_t h = vaddq_f32(H_im, f);
+          vst1q_f32(&H_p_ch.re[k], g);
+          vst1q_f32(&H_p_ch.im[k], h);
+        }
+      }
+    }
+
+    X_partition = 0;
+    limit = lim2;
+  } while (p < lim2);
+
+  X_partition = render_buffer.Position();
+  limit = lim1;
+  p = 0;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        FftData& H_p_ch = (*H)[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+
+        H_p_ch.re[kFftLengthBy2] += X.re[kFftLengthBy2] * G.re[kFftLengthBy2] +
+                                    X.im[kFftLengthBy2] * G.im[kFftLengthBy2];
+        H_p_ch.im[kFftLengthBy2] += X.re[kFftLengthBy2] * G.im[kFftLengthBy2] -
+                                    X.im[kFftLengthBy2] * G.re[kFftLengthBy2];
+      }
+    }
+    X_partition = 0;
+    limit = lim2;
+  } while (p < lim2);
+}
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Adapts the filter partitions. (SSE2 variant)
+void AdaptPartitions_Sse2(const RenderBuffer& render_buffer,
+                          const FftData& G,
+                          size_t num_partitions,
+                          std::vector<std::vector<FftData>>* H) {
+  rtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+      render_buffer.GetFftBuffer();
+  const size_t num_render_channels = render_buffer_data[0].size();
+  const size_t lim1 = std::min(
+      render_buffer_data.size() - render_buffer.Position(), num_partitions);
+  const size_t lim2 = num_partitions;
+  constexpr size_t kNumFourBinBands = kFftLengthBy2 / 4;
+
+  size_t X_partition = render_buffer.Position();
+  size_t limit = lim1;
+  size_t p = 0;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        FftData& H_p_ch = (*H)[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+
+        for (size_t k = 0, n = 0; n < kNumFourBinBands; ++n, k += 4) {
+          const __m128 G_re = _mm_loadu_ps(&G.re[k]);
+          const __m128 G_im = _mm_loadu_ps(&G.im[k]);
+          const __m128 X_re = _mm_loadu_ps(&X.re[k]);
+          const __m128 X_im = _mm_loadu_ps(&X.im[k]);
+          const __m128 H_re = _mm_loadu_ps(&H_p_ch.re[k]);
+          const __m128 H_im = _mm_loadu_ps(&H_p_ch.im[k]);
+          const __m128 a = _mm_mul_ps(X_re, G_re);
+          const __m128 b = _mm_mul_ps(X_im, G_im);
+          const __m128 c = _mm_mul_ps(X_re, G_im);
+          const __m128 d = _mm_mul_ps(X_im, G_re);
+          const __m128 e = _mm_add_ps(a, b);
+          const __m128 f = _mm_sub_ps(c, d);
+          const __m128 g = _mm_add_ps(H_re, e);
+          const __m128 h = _mm_add_ps(H_im, f);
+          _mm_storeu_ps(&H_p_ch.re[k], g);
+          _mm_storeu_ps(&H_p_ch.im[k], h);
+        }
+      }
+    }
+    X_partition = 0;
+    limit = lim2;
+  } while (p < lim2);
+
+  X_partition = render_buffer.Position();
+  limit = lim1;
+  p = 0;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        FftData& H_p_ch = (*H)[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+
+        H_p_ch.re[kFftLengthBy2] += X.re[kFftLengthBy2] * G.re[kFftLengthBy2] +
+                                    X.im[kFftLengthBy2] * G.im[kFftLengthBy2];
+        H_p_ch.im[kFftLengthBy2] += X.re[kFftLengthBy2] * G.im[kFftLengthBy2] -
+                                    X.im[kFftLengthBy2] * G.re[kFftLengthBy2];
+      }
+    }
+
+    X_partition = 0;
+    limit = lim2;
+  } while (p < lim2);
+}
+#endif
+
+// Produces the filter output.
+void ApplyFilter(const RenderBuffer& render_buffer,
+                 size_t num_partitions,
+                 const std::vector<std::vector<FftData>>& H,
+                 FftData* S) {
+  S->re.fill(0.f);
+  S->im.fill(0.f);
+
+  rtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+      render_buffer.GetFftBuffer();
+  size_t index = render_buffer.Position();
+  const size_t num_render_channels = render_buffer_data[index].size();
+  for (size_t p = 0; p < num_partitions; ++p) {
+    RTC_DCHECK_EQ(num_render_channels, H[p].size());
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      const FftData& X_p_ch = render_buffer_data[index][ch];
+      const FftData& H_p_ch = H[p][ch];
+      for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+        S->re[k] += X_p_ch.re[k] * H_p_ch.re[k] - X_p_ch.im[k] * H_p_ch.im[k];
+        S->im[k] += X_p_ch.re[k] * H_p_ch.im[k] + X_p_ch.im[k] * H_p_ch.re[k];
+      }
+    }
+    index = index < (render_buffer_data.size() - 1) ? index + 1 : 0;
+  }
+}
+
+#if defined(WEBRTC_HAS_NEON)
+// Produces the filter output (Neon variant).
+void ApplyFilter_Neon(const RenderBuffer& render_buffer,
+                      size_t num_partitions,
+                      const std::vector<std::vector<FftData>>& H,
+                      FftData* S) {
+  // const RenderBuffer& render_buffer,
+  //                     rtc::ArrayView<const FftData> H,
+  //                     FftData* S) {
+  RTC_DCHECK_GE(H.size(), H.size() - 1);
+  S->Clear();
+
+  rtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+      render_buffer.GetFftBuffer();
+  const size_t num_render_channels = render_buffer_data[0].size();
+  const size_t lim1 = std::min(
+      render_buffer_data.size() - render_buffer.Position(), num_partitions);
+  const size_t lim2 = num_partitions;
+  constexpr size_t kNumFourBinBands = kFftLengthBy2 / 4;
+
+  size_t X_partition = render_buffer.Position();
+  size_t p = 0;
+  size_t limit = lim1;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        const FftData& H_p_ch = H[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+        for (size_t k = 0, n = 0; n < kNumFourBinBands; ++n, k += 4) {
+          const float32x4_t X_re = vld1q_f32(&X.re[k]);
+          const float32x4_t X_im = vld1q_f32(&X.im[k]);
+          const float32x4_t H_re = vld1q_f32(&H_p_ch.re[k]);
+          const float32x4_t H_im = vld1q_f32(&H_p_ch.im[k]);
+          const float32x4_t S_re = vld1q_f32(&S->re[k]);
+          const float32x4_t S_im = vld1q_f32(&S->im[k]);
+          const float32x4_t a = vmulq_f32(X_re, H_re);
+          const float32x4_t e = vmlsq_f32(a, X_im, H_im);
+          const float32x4_t c = vmulq_f32(X_re, H_im);
+          const float32x4_t f = vmlaq_f32(c, X_im, H_re);
+          const float32x4_t g = vaddq_f32(S_re, e);
+          const float32x4_t h = vaddq_f32(S_im, f);
+          vst1q_f32(&S->re[k], g);
+          vst1q_f32(&S->im[k], h);
+        }
+      }
+    }
+    limit = lim2;
+    X_partition = 0;
+  } while (p < lim2);
+
+  X_partition = render_buffer.Position();
+  p = 0;
+  limit = lim1;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        const FftData& H_p_ch = H[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+        S->re[kFftLengthBy2] += X.re[kFftLengthBy2] * H_p_ch.re[kFftLengthBy2] -
+                                X.im[kFftLengthBy2] * H_p_ch.im[kFftLengthBy2];
+        S->im[kFftLengthBy2] += X.re[kFftLengthBy2] * H_p_ch.im[kFftLengthBy2] +
+                                X.im[kFftLengthBy2] * H_p_ch.re[kFftLengthBy2];
+      }
+    }
+    limit = lim2;
+    X_partition = 0;
+  } while (p < lim2);
+}
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Produces the filter output (SSE2 variant).
+void ApplyFilter_Sse2(const RenderBuffer& render_buffer,
+                      size_t num_partitions,
+                      const std::vector<std::vector<FftData>>& H,
+                      FftData* S) {
+  // const RenderBuffer& render_buffer,
+  //                     rtc::ArrayView<const FftData> H,
+  //                     FftData* S) {
+  RTC_DCHECK_GE(H.size(), H.size() - 1);
+  S->re.fill(0.f);
+  S->im.fill(0.f);
+
+  rtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+      render_buffer.GetFftBuffer();
+  const size_t num_render_channels = render_buffer_data[0].size();
+  const size_t lim1 = std::min(
+      render_buffer_data.size() - render_buffer.Position(), num_partitions);
+  const size_t lim2 = num_partitions;
+  constexpr size_t kNumFourBinBands = kFftLengthBy2 / 4;
+
+  size_t X_partition = render_buffer.Position();
+  size_t p = 0;
+  size_t limit = lim1;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        const FftData& H_p_ch = H[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+        for (size_t k = 0, n = 0; n < kNumFourBinBands; ++n, k += 4) {
+          const __m128 X_re = _mm_loadu_ps(&X.re[k]);
+          const __m128 X_im = _mm_loadu_ps(&X.im[k]);
+          const __m128 H_re = _mm_loadu_ps(&H_p_ch.re[k]);
+          const __m128 H_im = _mm_loadu_ps(&H_p_ch.im[k]);
+          const __m128 S_re = _mm_loadu_ps(&S->re[k]);
+          const __m128 S_im = _mm_loadu_ps(&S->im[k]);
+          const __m128 a = _mm_mul_ps(X_re, H_re);
+          const __m128 b = _mm_mul_ps(X_im, H_im);
+          const __m128 c = _mm_mul_ps(X_re, H_im);
+          const __m128 d = _mm_mul_ps(X_im, H_re);
+          const __m128 e = _mm_sub_ps(a, b);
+          const __m128 f = _mm_add_ps(c, d);
+          const __m128 g = _mm_add_ps(S_re, e);
+          const __m128 h = _mm_add_ps(S_im, f);
+          _mm_storeu_ps(&S->re[k], g);
+          _mm_storeu_ps(&S->im[k], h);
+        }
+      }
+    }
+    limit = lim2;
+    X_partition = 0;
+  } while (p < lim2);
+
+  X_partition = render_buffer.Position();
+  p = 0;
+  limit = lim1;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        const FftData& H_p_ch = H[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+        S->re[kFftLengthBy2] += X.re[kFftLengthBy2] * H_p_ch.re[kFftLengthBy2] -
+                                X.im[kFftLengthBy2] * H_p_ch.im[kFftLengthBy2];
+        S->im[kFftLengthBy2] += X.re[kFftLengthBy2] * H_p_ch.im[kFftLengthBy2] +
+                                X.im[kFftLengthBy2] * H_p_ch.re[kFftLengthBy2];
+      }
+    }
+    limit = lim2;
+    X_partition = 0;
+  } while (p < lim2);
+}
+#endif
+
+}  // namespace aec3
+
+namespace {
+
+// Ensures that the newly added filter partitions after a size increase are set
+// to zero.
+void ZeroFilter(size_t old_size,
+                size_t new_size,
+                std::vector<std::vector<FftData>>* H) {
+  RTC_DCHECK_GE(H->size(), old_size);
+  RTC_DCHECK_GE(H->size(), new_size);
+
+  for (size_t p = old_size; p < new_size; ++p) {
+    RTC_DCHECK_EQ((*H)[p].size(), (*H)[0].size());
+    for (size_t ch = 0; ch < (*H)[0].size(); ++ch) {
+      (*H)[p][ch].Clear();
+    }
+  }
+}
+
+}  // namespace
+
+AdaptiveFirFilter::AdaptiveFirFilter(size_t max_size_partitions,
+                                     size_t initial_size_partitions,
+                                     size_t size_change_duration_blocks,
+                                     size_t num_render_channels,
+                                     Aec3Optimization optimization,
+                                     ApmDataDumper* data_dumper)
+    : data_dumper_(data_dumper),
+      fft_(),
+      optimization_(optimization),
+      num_render_channels_(num_render_channels),
+      max_size_partitions_(max_size_partitions),
+      size_change_duration_blocks_(
+          static_cast<int>(size_change_duration_blocks)),
+      current_size_partitions_(initial_size_partitions),
+      target_size_partitions_(initial_size_partitions),
+      old_target_size_partitions_(initial_size_partitions),
+      H_(max_size_partitions_, std::vector<FftData>(num_render_channels_)) {
+  RTC_DCHECK(data_dumper_);
+  RTC_DCHECK_GE(max_size_partitions, initial_size_partitions);
+
+  RTC_DCHECK_LT(0, size_change_duration_blocks_);
+  one_by_size_change_duration_blocks_ = 1.f / size_change_duration_blocks_;
+
+  ZeroFilter(0, max_size_partitions_, &H_);
+
+  SetSizePartitions(current_size_partitions_, true);
+}
+
+AdaptiveFirFilter::~AdaptiveFirFilter() = default;
+
+void AdaptiveFirFilter::HandleEchoPathChange() {
+  // TODO(peah): Check the value and purpose of the code below.
+  ZeroFilter(current_size_partitions_, max_size_partitions_, &H_);
+}
+
+void AdaptiveFirFilter::SetSizePartitions(size_t size, bool immediate_effect) {
+  RTC_DCHECK_EQ(max_size_partitions_, H_.capacity());
+  RTC_DCHECK_LE(size, max_size_partitions_);
+
+  target_size_partitions_ = std::min(max_size_partitions_, size);
+  if (immediate_effect) {
+    size_t old_size_partitions_ = current_size_partitions_;
+    current_size_partitions_ = old_target_size_partitions_ =
+        target_size_partitions_;
+    ZeroFilter(old_size_partitions_, current_size_partitions_, &H_);
+
+    partition_to_constrain_ =
+        std::min(partition_to_constrain_, current_size_partitions_ - 1);
+    size_change_counter_ = 0;
+  } else {
+    size_change_counter_ = size_change_duration_blocks_;
+  }
+}
+
+void AdaptiveFirFilter::UpdateSize() {
+  RTC_DCHECK_GE(size_change_duration_blocks_, size_change_counter_);
+  size_t old_size_partitions_ = current_size_partitions_;
+  if (size_change_counter_ > 0) {
+    --size_change_counter_;
+
+    auto average = [](float from, float to, float from_weight) {
+      return from * from_weight + to * (1.f - from_weight);
+    };
+
+    float change_factor =
+        size_change_counter_ * one_by_size_change_duration_blocks_;
+
+    current_size_partitions_ = average(old_target_size_partitions_,
+                                       target_size_partitions_, change_factor);
+
+    partition_to_constrain_ =
+        std::min(partition_to_constrain_, current_size_partitions_ - 1);
+  } else {
+    current_size_partitions_ = old_target_size_partitions_ =
+        target_size_partitions_;
+  }
+  ZeroFilter(old_size_partitions_, current_size_partitions_, &H_);
+  RTC_DCHECK_LE(0, size_change_counter_);
+}
+
+void AdaptiveFirFilter::Filter(const RenderBuffer& render_buffer,
+                               FftData* S) const {
+  RTC_DCHECK(S);
+  switch (optimization_) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+    case Aec3Optimization::kSse2:
+      aec3::ApplyFilter_Sse2(render_buffer, current_size_partitions_, H_, S);
+      break;
+    case Aec3Optimization::kAvx2:
+      aec3::ApplyFilter_Avx2(render_buffer, current_size_partitions_, H_, S);
+      break;
+#endif
+#if defined(WEBRTC_HAS_NEON)
+    case Aec3Optimization::kNeon:
+      aec3::ApplyFilter_Neon(render_buffer, current_size_partitions_, H_, S);
+      break;
+#endif
+    default:
+      aec3::ApplyFilter(render_buffer, current_size_partitions_, H_, S);
+  }
+}
+
+void AdaptiveFirFilter::Adapt(const RenderBuffer& render_buffer,
+                              const FftData& G) {
+  // Adapt the filter and update the filter size.
+  AdaptAndUpdateSize(render_buffer, G);
+
+  // Constrain the filter partitions in a cyclic manner.
+  Constrain();
+}
+
+void AdaptiveFirFilter::Adapt(const RenderBuffer& render_buffer,
+                              const FftData& G,
+                              std::vector<float>* impulse_response) {
+  // Adapt the filter and update the filter size.
+  AdaptAndUpdateSize(render_buffer, G);
+
+  // Constrain the filter partitions in a cyclic manner.
+  ConstrainAndUpdateImpulseResponse(impulse_response);
+}
+
+void AdaptiveFirFilter::ComputeFrequencyResponse(
+    std::vector<std::array<float, kFftLengthBy2Plus1>>* H2) const {
+  RTC_DCHECK_GE(max_size_partitions_, H2->capacity());
+
+  H2->resize(current_size_partitions_);
+
+  switch (optimization_) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+    case Aec3Optimization::kSse2:
+      aec3::ComputeFrequencyResponse_Sse2(current_size_partitions_, H_, H2);
+      break;
+    case Aec3Optimization::kAvx2:
+      aec3::ComputeFrequencyResponse_Avx2(current_size_partitions_, H_, H2);
+      break;
+#endif
+#if defined(WEBRTC_HAS_NEON)
+    case Aec3Optimization::kNeon:
+      aec3::ComputeFrequencyResponse_Neon(current_size_partitions_, H_, H2);
+      break;
+#endif
+    default:
+      aec3::ComputeFrequencyResponse(current_size_partitions_, H_, H2);
+  }
+}
+
+void AdaptiveFirFilter::AdaptAndUpdateSize(const RenderBuffer& render_buffer,
+                                           const FftData& G) {
+  // Update the filter size if needed.
+  UpdateSize();
+
+  // Adapt the filter.
+  switch (optimization_) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+    case Aec3Optimization::kSse2:
+      aec3::AdaptPartitions_Sse2(render_buffer, G, current_size_partitions_,
+                                 &H_);
+      break;
+    case Aec3Optimization::kAvx2:
+      aec3::AdaptPartitions_Avx2(render_buffer, G, current_size_partitions_,
+                                 &H_);
+      break;
+#endif
+#if defined(WEBRTC_HAS_NEON)
+    case Aec3Optimization::kNeon:
+      aec3::AdaptPartitions_Neon(render_buffer, G, current_size_partitions_,
+                                 &H_);
+      break;
+#endif
+    default:
+      aec3::AdaptPartitions(render_buffer, G, current_size_partitions_, &H_);
+  }
+}
+
+// Constrains the partition of the frequency domain filter to be limited in
+// time via setting the relevant time-domain coefficients to zero and updates
+// the corresponding values in an externally stored impulse response estimate.
+void AdaptiveFirFilter::ConstrainAndUpdateImpulseResponse(
+    std::vector<float>* impulse_response) {
+  RTC_DCHECK_EQ(GetTimeDomainLength(max_size_partitions_),
+                impulse_response->capacity());
+  impulse_response->resize(GetTimeDomainLength(current_size_partitions_));
+  std::array<float, kFftLength> h;
+  impulse_response->resize(GetTimeDomainLength(current_size_partitions_));
+  std::fill(
+      impulse_response->begin() + partition_to_constrain_ * kFftLengthBy2,
+      impulse_response->begin() + (partition_to_constrain_ + 1) * kFftLengthBy2,
+      0.f);
+
+  for (size_t ch = 0; ch < num_render_channels_; ++ch) {
+    fft_.Ifft(H_[partition_to_constrain_][ch], &h);
+
+    static constexpr float kScale = 1.0f / kFftLengthBy2;
+    std::for_each(h.begin(), h.begin() + kFftLengthBy2,
+                  [](float& a) { a *= kScale; });
+    std::fill(h.begin() + kFftLengthBy2, h.end(), 0.f);
+
+    if (ch == 0) {
+      std::copy(
+          h.begin(), h.begin() + kFftLengthBy2,
+          impulse_response->begin() + partition_to_constrain_ * kFftLengthBy2);
+    } else {
+      for (size_t k = 0, j = partition_to_constrain_ * kFftLengthBy2;
+           k < kFftLengthBy2; ++k, ++j) {
+        if (fabsf((*impulse_response)[j]) < fabsf(h[k])) {
+          (*impulse_response)[j] = h[k];
+        }
+      }
+    }
+
+    fft_.Fft(&h, &H_[partition_to_constrain_][ch]);
+  }
+
+  partition_to_constrain_ =
+      partition_to_constrain_ < (current_size_partitions_ - 1)
+          ? partition_to_constrain_ + 1
+          : 0;
+}
+
+// Constrains the a partiton of the frequency domain filter to be limited in
+// time via setting the relevant time-domain coefficients to zero.
+void AdaptiveFirFilter::Constrain() {
+  std::array<float, kFftLength> h;
+  for (size_t ch = 0; ch < num_render_channels_; ++ch) {
+    fft_.Ifft(H_[partition_to_constrain_][ch], &h);
+
+    static constexpr float kScale = 1.0f / kFftLengthBy2;
+    std::for_each(h.begin(), h.begin() + kFftLengthBy2,
+                  [](float& a) { a *= kScale; });
+    std::fill(h.begin() + kFftLengthBy2, h.end(), 0.f);
+
+    fft_.Fft(&h, &H_[partition_to_constrain_][ch]);
+  }
+
+  partition_to_constrain_ =
+      partition_to_constrain_ < (current_size_partitions_ - 1)
+          ? partition_to_constrain_ + 1
+          : 0;
+}
+
+void AdaptiveFirFilter::ScaleFilter(float factor) {
+  for (auto& H_p : H_) {
+    for (auto& H_p_ch : H_p) {
+      for (auto& re : H_p_ch.re) {
+        re *= factor;
+      }
+      for (auto& im : H_p_ch.im) {
+        im *= factor;
+      }
+    }
+  }
+}
+
+// Set the filter coefficients.
+void AdaptiveFirFilter::SetFilter(size_t num_partitions,
+                                  const std::vector<std::vector<FftData>>& H) {
+  const size_t min_num_partitions =
+      std::min(current_size_partitions_, num_partitions);
+  for (size_t p = 0; p < min_num_partitions; ++p) {
+    RTC_DCHECK_EQ(H_[p].size(), H[p].size());
+    RTC_DCHECK_EQ(num_render_channels_, H_[p].size());
+
+    for (size_t ch = 0; ch < num_render_channels_; ++ch) {
+      std::copy(H[p][ch].re.begin(), H[p][ch].re.end(), H_[p][ch].re.begin());
+      std::copy(H[p][ch].im.begin(), H[p][ch].im.end(), H_[p][ch].im.begin());
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter.h b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter.h
new file mode 100644
index 0000000000..34c06f4367
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter.h
@@ -0,0 +1,192 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ADAPTIVE_FIR_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ADAPTIVE_FIR_FILTER_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace aec3 {
+// Computes and stores the frequency response of the filter.
+void ComputeFrequencyResponse(
+    size_t num_partitions,
+    const std::vector<std::vector<FftData>>& H,
+    std::vector<std::array<float, kFftLengthBy2Plus1>>* H2);
+#if defined(WEBRTC_HAS_NEON)
+void ComputeFrequencyResponse_Neon(
+    size_t num_partitions,
+    const std::vector<std::vector<FftData>>& H,
+    std::vector<std::array<float, kFftLengthBy2Plus1>>* H2);
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+void ComputeFrequencyResponse_Sse2(
+    size_t num_partitions,
+    const std::vector<std::vector<FftData>>& H,
+    std::vector<std::array<float, kFftLengthBy2Plus1>>* H2);
+
+void ComputeFrequencyResponse_Avx2(
+    size_t num_partitions,
+    const std::vector<std::vector<FftData>>& H,
+    std::vector<std::array<float, kFftLengthBy2Plus1>>* H2);
+#endif
+
+// Adapts the filter partitions.
+void AdaptPartitions(const RenderBuffer& render_buffer,
+                     const FftData& G,
+                     size_t num_partitions,
+                     std::vector<std::vector<FftData>>* H);
+#if defined(WEBRTC_HAS_NEON)
+void AdaptPartitions_Neon(const RenderBuffer& render_buffer,
+                          const FftData& G,
+                          size_t num_partitions,
+                          std::vector<std::vector<FftData>>* H);
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+void AdaptPartitions_Sse2(const RenderBuffer& render_buffer,
+                          const FftData& G,
+                          size_t num_partitions,
+                          std::vector<std::vector<FftData>>* H);
+
+void AdaptPartitions_Avx2(const RenderBuffer& render_buffer,
+                          const FftData& G,
+                          size_t num_partitions,
+                          std::vector<std::vector<FftData>>* H);
+#endif
+
+// Produces the filter output.
+void ApplyFilter(const RenderBuffer& render_buffer,
+                 size_t num_partitions,
+                 const std::vector<std::vector<FftData>>& H,
+                 FftData* S);
+#if defined(WEBRTC_HAS_NEON)
+void ApplyFilter_Neon(const RenderBuffer& render_buffer,
+                      size_t num_partitions,
+                      const std::vector<std::vector<FftData>>& H,
+                      FftData* S);
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+void ApplyFilter_Sse2(const RenderBuffer& render_buffer,
+                      size_t num_partitions,
+                      const std::vector<std::vector<FftData>>& H,
+                      FftData* S);
+
+void ApplyFilter_Avx2(const RenderBuffer& render_buffer,
+                      size_t num_partitions,
+                      const std::vector<std::vector<FftData>>& H,
+                      FftData* S);
+#endif
+
+}  // namespace aec3
+
+// Provides a frequency domain adaptive filter functionality.
+class AdaptiveFirFilter {
+ public:
+  AdaptiveFirFilter(size_t max_size_partitions,
+                    size_t initial_size_partitions,
+                    size_t size_change_duration_blocks,
+                    size_t num_render_channels,
+                    Aec3Optimization optimization,
+                    ApmDataDumper* data_dumper);
+
+  ~AdaptiveFirFilter();
+
+  AdaptiveFirFilter(const AdaptiveFirFilter&) = delete;
+  AdaptiveFirFilter& operator=(const AdaptiveFirFilter&) = delete;
+
+  // Produces the output of the filter.
+  void Filter(const RenderBuffer& render_buffer, FftData* S) const;
+
+  // Adapts the filter and updates an externally stored impulse response
+  // estimate.
+  void Adapt(const RenderBuffer& render_buffer,
+             const FftData& G,
+             std::vector<float>* impulse_response);
+
+  // Adapts the filter.
+  void Adapt(const RenderBuffer& render_buffer, const FftData& G);
+
+  // Receives reports that known echo path changes have occured and adjusts
+  // the filter adaptation accordingly.
+  void HandleEchoPathChange();
+
+  // Returns the filter size.
+  size_t SizePartitions() const { return current_size_partitions_; }
+
+  // Sets the filter size.
+  void SetSizePartitions(size_t size, bool immediate_effect);
+
+  // Computes the frequency responses for the filter partitions.
+  void ComputeFrequencyResponse(
+      std::vector<std::array<float, kFftLengthBy2Plus1>>* H2) const;
+
+  // Returns the maximum number of partitions for the filter.
+  size_t max_filter_size_partitions() const { return max_size_partitions_; }
+
+  void DumpFilter(absl::string_view name_frequency_domain) {
+    for (size_t p = 0; p < max_size_partitions_; ++p) {
+      data_dumper_->DumpRaw(name_frequency_domain, H_[p][0].re);
+      data_dumper_->DumpRaw(name_frequency_domain, H_[p][0].im);
+    }
+  }
+
+  // Scale the filter impulse response and spectrum by a factor.
+  void ScaleFilter(float factor);
+
+  // Set the filter coefficients.
+  void SetFilter(size_t num_partitions,
+                 const std::vector<std::vector<FftData>>& H);
+
+  // Gets the filter coefficients.
+  const std::vector<std::vector<FftData>>& GetFilter() const { return H_; }
+
+ private:
+  // Adapts the filter and updates the filter size.
+  void AdaptAndUpdateSize(const RenderBuffer& render_buffer, const FftData& G);
+
+  // Constrain the filter partitions in a cyclic manner.
+  void Constrain();
+  // Constrains the filter in a cyclic manner and updates the corresponding
+  // values in the supplied impulse response.
+  void ConstrainAndUpdateImpulseResponse(std::vector<float>* impulse_response);
+
+  // Gradually Updates the current filter size towards the target size.
+  void UpdateSize();
+
+  ApmDataDumper* const data_dumper_;
+  const Aec3Fft fft_;
+  const Aec3Optimization optimization_;
+  const size_t num_render_channels_;
+  const size_t max_size_partitions_;
+  const int size_change_duration_blocks_;
+  float one_by_size_change_duration_blocks_;
+  size_t current_size_partitions_;
+  size_t target_size_partitions_;
+  size_t old_target_size_partitions_;
+  int size_change_counter_ = 0;
+  std::vector<std::vector<FftData>> H_;
+  size_t partition_to_constrain_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ADAPTIVE_FIR_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
new file mode 100644
index 0000000000..44d4514275
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
@@ -0,0 +1,188 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
+
+#include "common_audio/intrin.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace aec3 {
+
+// Computes and stores the frequency response of the filter.
+void ComputeFrequencyResponse_Avx2(
+    size_t num_partitions,
+    const std::vector<std::vector<FftData>>& H,
+    std::vector<std::array<float, kFftLengthBy2Plus1>>* H2) {
+  for (auto& H2_ch : *H2) {
+    H2_ch.fill(0.f);
+  }
+
+  const size_t num_render_channels = H[0].size();
+  RTC_DCHECK_EQ(H.size(), H2->capacity());
+  for (size_t p = 0; p < num_partitions; ++p) {
+    RTC_DCHECK_EQ(kFftLengthBy2Plus1, (*H2)[p].size());
+    auto& H2_p = (*H2)[p];
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      const FftData& H_p_ch = H[p][ch];
+      for (size_t j = 0; j < kFftLengthBy2; j += 8) {
+        __m256 re = _mm256_loadu_ps(&H_p_ch.re[j]);
+        __m256 re2 = _mm256_mul_ps(re, re);
+        __m256 im = _mm256_loadu_ps(&H_p_ch.im[j]);
+        re2 = _mm256_fmadd_ps(im, im, re2);
+        __m256 H2_k_j = _mm256_loadu_ps(&H2_p[j]);
+        H2_k_j = _mm256_max_ps(H2_k_j, re2);
+        _mm256_storeu_ps(&H2_p[j], H2_k_j);
+      }
+      float H2_new = H_p_ch.re[kFftLengthBy2] * H_p_ch.re[kFftLengthBy2] +
+                     H_p_ch.im[kFftLengthBy2] * H_p_ch.im[kFftLengthBy2];
+      H2_p[kFftLengthBy2] = std::max(H2_p[kFftLengthBy2], H2_new);
+    }
+  }
+}
+
+// Adapts the filter partitions.
+void AdaptPartitions_Avx2(const RenderBuffer& render_buffer,
+                          const FftData& G,
+                          size_t num_partitions,
+                          std::vector<std::vector<FftData>>* H) {
+  rtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+      render_buffer.GetFftBuffer();
+  const size_t num_render_channels = render_buffer_data[0].size();
+  const size_t lim1 = std::min(
+      render_buffer_data.size() - render_buffer.Position(), num_partitions);
+  const size_t lim2 = num_partitions;
+  constexpr size_t kNumEightBinBands = kFftLengthBy2 / 8;
+
+  size_t X_partition = render_buffer.Position();
+  size_t limit = lim1;
+  size_t p = 0;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        FftData& H_p_ch = (*H)[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+
+        for (size_t k = 0, n = 0; n < kNumEightBinBands; ++n, k += 8) {
+          const __m256 G_re = _mm256_loadu_ps(&G.re[k]);
+          const __m256 G_im = _mm256_loadu_ps(&G.im[k]);
+          const __m256 X_re = _mm256_loadu_ps(&X.re[k]);
+          const __m256 X_im = _mm256_loadu_ps(&X.im[k]);
+          const __m256 H_re = _mm256_loadu_ps(&H_p_ch.re[k]);
+          const __m256 H_im = _mm256_loadu_ps(&H_p_ch.im[k]);
+          const __m256 a = _mm256_mul_ps(X_re, G_re);
+          const __m256 b = _mm256_mul_ps(X_im, G_im);
+          const __m256 c = _mm256_mul_ps(X_re, G_im);
+          const __m256 d = _mm256_mul_ps(X_im, G_re);
+          const __m256 e = _mm256_add_ps(a, b);
+          const __m256 f = _mm256_sub_ps(c, d);
+          const __m256 g = _mm256_add_ps(H_re, e);
+          const __m256 h = _mm256_add_ps(H_im, f);
+          _mm256_storeu_ps(&H_p_ch.re[k], g);
+          _mm256_storeu_ps(&H_p_ch.im[k], h);
+        }
+      }
+    }
+    X_partition = 0;
+    limit = lim2;
+  } while (p < lim2);
+
+  X_partition = render_buffer.Position();
+  limit = lim1;
+  p = 0;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        FftData& H_p_ch = (*H)[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+
+        H_p_ch.re[kFftLengthBy2] += X.re[kFftLengthBy2] * G.re[kFftLengthBy2] +
+                                    X.im[kFftLengthBy2] * G.im[kFftLengthBy2];
+        H_p_ch.im[kFftLengthBy2] += X.re[kFftLengthBy2] * G.im[kFftLengthBy2] -
+                                    X.im[kFftLengthBy2] * G.re[kFftLengthBy2];
+      }
+    }
+
+    X_partition = 0;
+    limit = lim2;
+  } while (p < lim2);
+}
+
+// Produces the filter output (AVX2 variant).
+void ApplyFilter_Avx2(const RenderBuffer& render_buffer,
+                      size_t num_partitions,
+                      const std::vector<std::vector<FftData>>& H,
+                      FftData* S) {
+  RTC_DCHECK_GE(H.size(), H.size() - 1);
+  S->re.fill(0.f);
+  S->im.fill(0.f);
+
+  rtc::ArrayView<const std::vector<FftData>> render_buffer_data =
+      render_buffer.GetFftBuffer();
+  const size_t num_render_channels = render_buffer_data[0].size();
+  const size_t lim1 = std::min(
+      render_buffer_data.size() - render_buffer.Position(), num_partitions);
+  const size_t lim2 = num_partitions;
+  constexpr size_t kNumEightBinBands = kFftLengthBy2 / 8;
+
+  size_t X_partition = render_buffer.Position();
+  size_t p = 0;
+  size_t limit = lim1;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        const FftData& H_p_ch = H[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+        for (size_t k = 0, n = 0; n < kNumEightBinBands; ++n, k += 8) {
+          const __m256 X_re = _mm256_loadu_ps(&X.re[k]);
+          const __m256 X_im = _mm256_loadu_ps(&X.im[k]);
+          const __m256 H_re = _mm256_loadu_ps(&H_p_ch.re[k]);
+          const __m256 H_im = _mm256_loadu_ps(&H_p_ch.im[k]);
+          const __m256 S_re = _mm256_loadu_ps(&S->re[k]);
+          const __m256 S_im = _mm256_loadu_ps(&S->im[k]);
+          const __m256 a = _mm256_mul_ps(X_re, H_re);
+          const __m256 b = _mm256_mul_ps(X_im, H_im);
+          const __m256 c = _mm256_mul_ps(X_re, H_im);
+          const __m256 d = _mm256_mul_ps(X_im, H_re);
+          const __m256 e = _mm256_sub_ps(a, b);
+          const __m256 f = _mm256_add_ps(c, d);
+          const __m256 g = _mm256_add_ps(S_re, e);
+          const __m256 h = _mm256_add_ps(S_im, f);
+          _mm256_storeu_ps(&S->re[k], g);
+          _mm256_storeu_ps(&S->im[k], h);
+        }
+      }
+    }
+    limit = lim2;
+    X_partition = 0;
+  } while (p < lim2);
+
+  X_partition = render_buffer.Position();
+  p = 0;
+  limit = lim1;
+  do {
+    for (; p < limit; ++p, ++X_partition) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        const FftData& H_p_ch = H[p][ch];
+        const FftData& X = render_buffer_data[X_partition][ch];
+        S->re[kFftLengthBy2] += X.re[kFftLengthBy2] * H_p_ch.re[kFftLengthBy2] -
+                                X.im[kFftLengthBy2] * H_p_ch.im[kFftLengthBy2];
+        S->im[kFftLengthBy2] += X.re[kFftLengthBy2] * H_p_ch.im[kFftLengthBy2] +
+                                X.im[kFftLengthBy2] * H_p_ch.re[kFftLengthBy2];
+      }
+    }
+    limit = lim2;
+    X_partition = 0;
+  } while (p < lim2);
+}
+
+}  // namespace aec3
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
new file mode 100644
index 0000000000..45b8813979
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
+
+#include <algorithm>
+#include <functional>
+
+#if defined(WEBRTC_HAS_NEON)
+#include <arm_neon.h>
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+
+namespace webrtc {
+
+namespace aec3 {
+
+// Computes and stores the echo return loss estimate of the filter, which is the
+// sum of the partition frequency responses.
+void ErlComputer(const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+                 rtc::ArrayView<float> erl) {
+  std::fill(erl.begin(), erl.end(), 0.f);
+  for (auto& H2_j : H2) {
+    std::transform(H2_j.begin(), H2_j.end(), erl.begin(), erl.begin(),
+                   std::plus<float>());
+  }
+}
+
+#if defined(WEBRTC_HAS_NEON)
+// Computes and stores the echo return loss estimate of the filter, which is the
+// sum of the partition frequency responses.
+void ErlComputer_NEON(
+    const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+    rtc::ArrayView<float> erl) {
+  std::fill(erl.begin(), erl.end(), 0.f);
+  for (auto& H2_j : H2) {
+    for (size_t k = 0; k < kFftLengthBy2; k += 4) {
+      const float32x4_t H2_j_k = vld1q_f32(&H2_j[k]);
+      float32x4_t erl_k = vld1q_f32(&erl[k]);
+      erl_k = vaddq_f32(erl_k, H2_j_k);
+      vst1q_f32(&erl[k], erl_k);
+    }
+    erl[kFftLengthBy2] += H2_j[kFftLengthBy2];
+  }
+}
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Computes and stores the echo return loss estimate of the filter, which is the
+// sum of the partition frequency responses.
+void ErlComputer_SSE2(
+    const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+    rtc::ArrayView<float> erl) {
+  std::fill(erl.begin(), erl.end(), 0.f);
+  for (auto& H2_j : H2) {
+    for (size_t k = 0; k < kFftLengthBy2; k += 4) {
+      const __m128 H2_j_k = _mm_loadu_ps(&H2_j[k]);
+      __m128 erl_k = _mm_loadu_ps(&erl[k]);
+      erl_k = _mm_add_ps(erl_k, H2_j_k);
+      _mm_storeu_ps(&erl[k], erl_k);
+    }
+    erl[kFftLengthBy2] += H2_j[kFftLengthBy2];
+  }
+}
+#endif
+
+}  // namespace aec3
+
+void ComputeErl(const Aec3Optimization& optimization,
+                const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+                rtc::ArrayView<float> erl) {
+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, erl.size());
+  // Update the frequency response and echo return loss for the filter.
+  switch (optimization) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+    case Aec3Optimization::kSse2:
+      aec3::ErlComputer_SSE2(H2, erl);
+      break;
+    case Aec3Optimization::kAvx2:
+      aec3::ErlComputer_AVX2(H2, erl);
+      break;
+#endif
+#if defined(WEBRTC_HAS_NEON)
+    case Aec3Optimization::kNeon:
+      aec3::ErlComputer_NEON(H2, erl);
+      break;
+#endif
+    default:
+      aec3::ErlComputer(H2, erl);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.h b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.h
new file mode 100644
index 0000000000..4ac13b1bc3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ADAPTIVE_FIR_FILTER_ERL_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ADAPTIVE_FIR_FILTER_ERL_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace aec3 {
+
+// Computes and stores the echo return loss estimate of the filter, which is the
+// sum of the partition frequency responses.
+void ErlComputer(const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+                 rtc::ArrayView<float> erl);
+#if defined(WEBRTC_HAS_NEON)
+void ErlComputer_NEON(
+    const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+    rtc::ArrayView<float> erl);
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+void ErlComputer_SSE2(
+    const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+    rtc::ArrayView<float> erl);
+
+void ErlComputer_AVX2(
+    const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+    rtc::ArrayView<float> erl);
+#endif
+
+}  // namespace aec3
+
+// Computes the echo return loss based on a frequency response.
+void ComputeErl(const Aec3Optimization& optimization,
+                const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+                rtc::ArrayView<float> erl);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ADAPTIVE_FIR_FILTER_ERL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc
new file mode 100644
index 0000000000..5fe7514db1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
+
+#include <immintrin.h>
+
+namespace webrtc {
+
+namespace aec3 {
+
+// Computes and stores the echo return loss estimate of the filter, which is the
+// sum of the partition frequency responses.
+void ErlComputer_AVX2(
+    const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
+    rtc::ArrayView<float> erl) {
+  std::fill(erl.begin(), erl.end(), 0.f);
+  for (auto& H2_j : H2) {
+    for (size_t k = 0; k < kFftLengthBy2; k += 8) {
+      const __m256 H2_j_k = _mm256_loadu_ps(&H2_j[k]);
+      __m256 erl_k = _mm256_loadu_ps(&erl[k]);
+      erl_k = _mm256_add_ps(erl_k, H2_j_k);
+      _mm256_storeu_ps(&erl[k], erl_k);
+    }
+    erl[kFftLengthBy2] += H2_j[kFftLengthBy2];
+  }
+}
+
+}  // namespace aec3
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_gn/moz.build
new file mode 100644
index 0000000000..60ecc93ab9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("adaptive_fir_filter_erl_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_unittest.cc
new file mode 100644
index 0000000000..d2af70a9f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_unittest.cc
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
+
+#include <array>
+#include <vector>
+
+#include "rtc_base/system/arch.h"
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace aec3 {
+
+#if defined(WEBRTC_HAS_NEON)
+// Verifies that the optimized method for echo return loss computation is
+// bitexact to the reference counterpart.
+TEST(AdaptiveFirFilter, UpdateErlNeonOptimization) {
+  const size_t kNumPartitions = 12;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> H2(kNumPartitions);
+  std::array<float, kFftLengthBy2Plus1> erl;
+  std::array<float, kFftLengthBy2Plus1> erl_NEON;
+
+  for (size_t j = 0; j < H2.size(); ++j) {
+    for (size_t k = 0; k < H2[j].size(); ++k) {
+      H2[j][k] = k + j / 3.f;
+    }
+  }
+
+  ErlComputer(H2, erl);
+  ErlComputer_NEON(H2, erl_NEON);
+
+  for (size_t j = 0; j < erl.size(); ++j) {
+    EXPECT_FLOAT_EQ(erl[j], erl_NEON[j]);
+  }
+}
+
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Verifies that the optimized method for echo return loss computation is
+// bitexact to the reference counterpart.
+TEST(AdaptiveFirFilter, UpdateErlSse2Optimization) {
+  bool use_sse2 = (GetCPUInfo(kSSE2) != 0);
+  if (use_sse2) {
+    const size_t kNumPartitions = 12;
+    std::vector<std::array<float, kFftLengthBy2Plus1>> H2(kNumPartitions);
+    std::array<float, kFftLengthBy2Plus1> erl;
+    std::array<float, kFftLengthBy2Plus1> erl_SSE2;
+
+    for (size_t j = 0; j < H2.size(); ++j) {
+      for (size_t k = 0; k < H2[j].size(); ++k) {
+        H2[j][k] = k + j / 3.f;
+      }
+    }
+
+    ErlComputer(H2, erl);
+    ErlComputer_SSE2(H2, erl_SSE2);
+
+    for (size_t j = 0; j < erl.size(); ++j) {
+      EXPECT_FLOAT_EQ(erl[j], erl_SSE2[j]);
+    }
+  }
+}
+
+// Verifies that the optimized method for echo return loss computation is
+// bitexact to the reference counterpart.
+TEST(AdaptiveFirFilter, UpdateErlAvx2Optimization) {
+  bool use_avx2 = (GetCPUInfo(kAVX2) != 0);
+  if (use_avx2) {
+    const size_t kNumPartitions = 12;
+    std::vector<std::array<float, kFftLengthBy2Plus1>> H2(kNumPartitions);
+    std::array<float, kFftLengthBy2Plus1> erl;
+    std::array<float, kFftLengthBy2Plus1> erl_AVX2;
+
+    for (size_t j = 0; j < H2.size(); ++j) {
+      for (size_t k = 0; k < H2[j].size(); ++k) {
+        H2[j][k] = k + j / 3.f;
+      }
+    }
+
+    ErlComputer(H2, erl);
+    ErlComputer_AVX2(H2, erl_AVX2);
+
+    for (size_t j = 0; j < erl.size(); ++j) {
+      EXPECT_FLOAT_EQ(erl[j], erl_AVX2[j]);
+    }
+  }
+}
+
+#endif
+
+}  // namespace aec3
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_gn/moz.build
new file mode 100644
index 0000000000..fd78a43560
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_gn/moz.build
@@ -0,0 +1,216 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("adaptive_fir_filter_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
new file mode 100644
index 0000000000..a13764c109
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
@@ -0,0 +1,594 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
+
+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
+#include <math.h>
+
+#include <algorithm>
+#include <numeric>
+#include <string>
+
+#include "rtc_base/system/arch.h"
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/coarse_filter_update_gain.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "modules/audio_processing/utility/cascaded_biquad_filter.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace aec3 {
+namespace {
+
+std::string ProduceDebugText(size_t num_render_channels, size_t delay) {
+  rtc::StringBuilder ss;
+  ss << "delay: " << delay << ", ";
+  ss << "num_render_channels:" << num_render_channels;
+  return ss.Release();
+}
+
+}  // namespace
+
+class AdaptiveFirFilterOneTwoFourEightRenderChannels
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<size_t> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         AdaptiveFirFilterOneTwoFourEightRenderChannels,
+                         ::testing::Values(1, 2, 4, 8));
+
+#if defined(WEBRTC_HAS_NEON)
+// Verifies that the optimized methods for filter adaptation are similar to
+// their reference counterparts.
+TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels,
+       FilterAdaptationNeonOptimizations) {
+  const size_t num_render_channels = GetParam();
+  for (size_t num_partitions : {2, 5, 12, 30, 50}) {
+    constexpr int kSampleRateHz = 48000;
+    constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+    std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+        RenderDelayBuffer::Create(EchoCanceller3Config(), kSampleRateHz,
+                                  num_render_channels));
+    Random random_generator(42U);
+    Block x(kNumBands, num_render_channels);
+    FftData S_C;
+    FftData S_Neon;
+    FftData G;
+    Aec3Fft fft;
+    std::vector<std::vector<FftData>> H_C(
+        num_partitions, std::vector<FftData>(num_render_channels));
+    std::vector<std::vector<FftData>> H_Neon(
+        num_partitions, std::vector<FftData>(num_render_channels));
+    for (size_t p = 0; p < num_partitions; ++p) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        H_C[p][ch].Clear();
+        H_Neon[p][ch].Clear();
+      }
+    }
+
+    for (int k = 0; k < 30; ++k) {
+      for (int band = 0; band < x.NumBands(); ++band) {
+        for (int ch = 0; ch < x.NumChannels(); ++ch) {
+          RandomizeSampleVector(&random_generator, x.View(band, ch));
+        }
+      }
+      render_delay_buffer->Insert(x);
+      if (k == 0) {
+        render_delay_buffer->Reset();
+      }
+      render_delay_buffer->PrepareCaptureProcessing();
+    }
+    auto* const render_buffer = render_delay_buffer->GetRenderBuffer();
+
+    for (size_t j = 0; j < G.re.size(); ++j) {
+      G.re[j] = j / 10001.f;
+    }
+    for (size_t j = 1; j < G.im.size() - 1; ++j) {
+      G.im[j] = j / 20001.f;
+    }
+    G.im[0] = 0.f;
+    G.im[G.im.size() - 1] = 0.f;
+
+    AdaptPartitions_Neon(*render_buffer, G, num_partitions, &H_Neon);
+    AdaptPartitions(*render_buffer, G, num_partitions, &H_C);
+    AdaptPartitions_Neon(*render_buffer, G, num_partitions, &H_Neon);
+    AdaptPartitions(*render_buffer, G, num_partitions, &H_C);
+
+    for (size_t p = 0; p < num_partitions; ++p) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        for (size_t j = 0; j < H_C[p][ch].re.size(); ++j) {
+          EXPECT_FLOAT_EQ(H_C[p][ch].re[j], H_Neon[p][ch].re[j]);
+          EXPECT_FLOAT_EQ(H_C[p][ch].im[j], H_Neon[p][ch].im[j]);
+        }
+      }
+    }
+
+    ApplyFilter_Neon(*render_buffer, num_partitions, H_Neon, &S_Neon);
+    ApplyFilter(*render_buffer, num_partitions, H_C, &S_C);
+    for (size_t j = 0; j < S_C.re.size(); ++j) {
+      EXPECT_NEAR(S_C.re[j], S_Neon.re[j], fabs(S_C.re[j] * 0.00001f));
+      EXPECT_NEAR(S_C.im[j], S_Neon.im[j], fabs(S_C.re[j] * 0.00001f));
+    }
+  }
+}
+
+// Verifies that the optimized method for frequency response computation is
+// bitexact to the reference counterpart.
+TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels,
+       ComputeFrequencyResponseNeonOptimization) {
+  const size_t num_render_channels = GetParam();
+  for (size_t num_partitions : {2, 5, 12, 30, 50}) {
+    std::vector<std::vector<FftData>> H(
+        num_partitions, std::vector<FftData>(num_render_channels));
+    std::vector<std::array<float, kFftLengthBy2Plus1>> H2(num_partitions);
+    std::vector<std::array<float, kFftLengthBy2Plus1>> H2_Neon(num_partitions);
+
+    for (size_t p = 0; p < num_partitions; ++p) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        for (size_t k = 0; k < H[p][ch].re.size(); ++k) {
+          H[p][ch].re[k] = k + p / 3.f + ch;
+          H[p][ch].im[k] = p + k / 7.f - ch;
+        }
+      }
+    }
+
+    ComputeFrequencyResponse(num_partitions, H, &H2);
+    ComputeFrequencyResponse_Neon(num_partitions, H, &H2_Neon);
+
+    for (size_t p = 0; p < num_partitions; ++p) {
+      for (size_t k = 0; k < H2[p].size(); ++k) {
+        EXPECT_FLOAT_EQ(H2[p][k], H2_Neon[p][k]);
+      }
+    }
+  }
+}
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Verifies that the optimized methods for filter adaptation are bitexact to
+// their reference counterparts.
+TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels,
+       FilterAdaptationSse2Optimizations) {
+  const size_t num_render_channels = GetParam();
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  bool use_sse2 = (GetCPUInfo(kSSE2) != 0);
+  if (use_sse2) {
+    for (size_t num_partitions : {2, 5, 12, 30, 50}) {
+      std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+          RenderDelayBuffer::Create(EchoCanceller3Config(), kSampleRateHz,
+                                    num_render_channels));
+      Random random_generator(42U);
+      Block x(kNumBands, num_render_channels);
+      FftData S_C;
+      FftData S_Sse2;
+      FftData G;
+      Aec3Fft fft;
+      std::vector<std::vector<FftData>> H_C(
+          num_partitions, std::vector<FftData>(num_render_channels));
+      std::vector<std::vector<FftData>> H_Sse2(
+          num_partitions, std::vector<FftData>(num_render_channels));
+      for (size_t p = 0; p < num_partitions; ++p) {
+        for (size_t ch = 0; ch < num_render_channels; ++ch) {
+          H_C[p][ch].Clear();
+          H_Sse2[p][ch].Clear();
+        }
+      }
+
+      for (size_t k = 0; k < 500; ++k) {
+        for (int band = 0; band < x.NumBands(); ++band) {
+          for (int ch = 0; ch < x.NumChannels(); ++ch) {
+            RandomizeSampleVector(&random_generator, x.View(band, ch));
+          }
+        }
+        render_delay_buffer->Insert(x);
+        if (k == 0) {
+          render_delay_buffer->Reset();
+        }
+        render_delay_buffer->PrepareCaptureProcessing();
+        auto* const render_buffer = render_delay_buffer->GetRenderBuffer();
+
+        ApplyFilter_Sse2(*render_buffer, num_partitions, H_Sse2, &S_Sse2);
+        ApplyFilter(*render_buffer, num_partitions, H_C, &S_C);
+        for (size_t j = 0; j < S_C.re.size(); ++j) {
+          EXPECT_FLOAT_EQ(S_C.re[j], S_Sse2.re[j]);
+          EXPECT_FLOAT_EQ(S_C.im[j], S_Sse2.im[j]);
+        }
+
+        std::for_each(G.re.begin(), G.re.end(),
+                      [&](float& a) { a = random_generator.Rand<float>(); });
+        std::for_each(G.im.begin(), G.im.end(),
+                      [&](float& a) { a = random_generator.Rand<float>(); });
+
+        AdaptPartitions_Sse2(*render_buffer, G, num_partitions, &H_Sse2);
+        AdaptPartitions(*render_buffer, G, num_partitions, &H_C);
+
+        for (size_t p = 0; p < num_partitions; ++p) {
+          for (size_t ch = 0; ch < num_render_channels; ++ch) {
+            for (size_t j = 0; j < H_C[p][ch].re.size(); ++j) {
+              EXPECT_FLOAT_EQ(H_C[p][ch].re[j], H_Sse2[p][ch].re[j]);
+              EXPECT_FLOAT_EQ(H_C[p][ch].im[j], H_Sse2[p][ch].im[j]);
+            }
+          }
+        }
+      }
+    }
+  }
+}
+
+// Verifies that the optimized methods for filter adaptation are bitexact to
+// their reference counterparts.
+TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels,
+       FilterAdaptationAvx2Optimizations) {
+  const size_t num_render_channels = GetParam();
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  bool use_avx2 = (GetCPUInfo(kAVX2) != 0);
+  if (use_avx2) {
+    for (size_t num_partitions : {2, 5, 12, 30, 50}) {
+      std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+          RenderDelayBuffer::Create(EchoCanceller3Config(), kSampleRateHz,
+                                    num_render_channels));
+      Random random_generator(42U);
+      Block x(kNumBands, num_render_channels);
+      FftData S_C;
+      FftData S_Avx2;
+      FftData G;
+      Aec3Fft fft;
+      std::vector<std::vector<FftData>> H_C(
+          num_partitions, std::vector<FftData>(num_render_channels));
+      std::vector<std::vector<FftData>> H_Avx2(
+          num_partitions, std::vector<FftData>(num_render_channels));
+      for (size_t p = 0; p < num_partitions; ++p) {
+        for (size_t ch = 0; ch < num_render_channels; ++ch) {
+          H_C[p][ch].Clear();
+          H_Avx2[p][ch].Clear();
+        }
+      }
+
+      for (size_t k = 0; k < 500; ++k) {
+        for (int band = 0; band < x.NumBands(); ++band) {
+          for (int ch = 0; ch < x.NumChannels(); ++ch) {
+            RandomizeSampleVector(&random_generator, x.View(band, ch));
+          }
+        }
+        render_delay_buffer->Insert(x);
+        if (k == 0) {
+          render_delay_buffer->Reset();
+        }
+        render_delay_buffer->PrepareCaptureProcessing();
+        auto* const render_buffer = render_delay_buffer->GetRenderBuffer();
+
+        ApplyFilter_Avx2(*render_buffer, num_partitions, H_Avx2, &S_Avx2);
+        ApplyFilter(*render_buffer, num_partitions, H_C, &S_C);
+        for (size_t j = 0; j < S_C.re.size(); ++j) {
+          EXPECT_FLOAT_EQ(S_C.re[j], S_Avx2.re[j]);
+          EXPECT_FLOAT_EQ(S_C.im[j], S_Avx2.im[j]);
+        }
+
+        std::for_each(G.re.begin(), G.re.end(),
+                      [&](float& a) { a = random_generator.Rand<float>(); });
+        std::for_each(G.im.begin(), G.im.end(),
+                      [&](float& a) { a = random_generator.Rand<float>(); });
+
+        AdaptPartitions_Avx2(*render_buffer, G, num_partitions, &H_Avx2);
+        AdaptPartitions(*render_buffer, G, num_partitions, &H_C);
+
+        for (size_t p = 0; p < num_partitions; ++p) {
+          for (size_t ch = 0; ch < num_render_channels; ++ch) {
+            for (size_t j = 0; j < H_C[p][ch].re.size(); ++j) {
+              EXPECT_FLOAT_EQ(H_C[p][ch].re[j], H_Avx2[p][ch].re[j]);
+              EXPECT_FLOAT_EQ(H_C[p][ch].im[j], H_Avx2[p][ch].im[j]);
+            }
+          }
+        }
+      }
+    }
+  }
+}
+
+// Verifies that the optimized method for frequency response computation is
+// bitexact to the reference counterpart.
+TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels,
+       ComputeFrequencyResponseSse2Optimization) {
+  const size_t num_render_channels = GetParam();
+  bool use_sse2 = (GetCPUInfo(kSSE2) != 0);
+  if (use_sse2) {
+    for (size_t num_partitions : {2, 5, 12, 30, 50}) {
+      std::vector<std::vector<FftData>> H(
+          num_partitions, std::vector<FftData>(num_render_channels));
+      std::vector<std::array<float, kFftLengthBy2Plus1>> H2(num_partitions);
+      std::vector<std::array<float, kFftLengthBy2Plus1>> H2_Sse2(
+          num_partitions);
+
+      for (size_t p = 0; p < num_partitions; ++p) {
+        for (size_t ch = 0; ch < num_render_channels; ++ch) {
+          for (size_t k = 0; k < H[p][ch].re.size(); ++k) {
+            H[p][ch].re[k] = k + p / 3.f + ch;
+            H[p][ch].im[k] = p + k / 7.f - ch;
+          }
+        }
+      }
+
+      ComputeFrequencyResponse(num_partitions, H, &H2);
+      ComputeFrequencyResponse_Sse2(num_partitions, H, &H2_Sse2);
+
+      for (size_t p = 0; p < num_partitions; ++p) {
+        for (size_t k = 0; k < H2[p].size(); ++k) {
+          EXPECT_FLOAT_EQ(H2[p][k], H2_Sse2[p][k]);
+        }
+      }
+    }
+  }
+}
+
+// Verifies that the optimized method for frequency response computation is
+// bitexact to the reference counterpart.
+TEST_P(AdaptiveFirFilterOneTwoFourEightRenderChannels,
+       ComputeFrequencyResponseAvx2Optimization) {
+  const size_t num_render_channels = GetParam();
+  bool use_avx2 = (GetCPUInfo(kAVX2) != 0);
+  if (use_avx2) {
+    for (size_t num_partitions : {2, 5, 12, 30, 50}) {
+      std::vector<std::vector<FftData>> H(
+          num_partitions, std::vector<FftData>(num_render_channels));
+      std::vector<std::array<float, kFftLengthBy2Plus1>> H2(num_partitions);
+      std::vector<std::array<float, kFftLengthBy2Plus1>> H2_Avx2(
+          num_partitions);
+
+      for (size_t p = 0; p < num_partitions; ++p) {
+        for (size_t ch = 0; ch < num_render_channels; ++ch) {
+          for (size_t k = 0; k < H[p][ch].re.size(); ++k) {
+            H[p][ch].re[k] = k + p / 3.f + ch;
+            H[p][ch].im[k] = p + k / 7.f - ch;
+          }
+        }
+      }
+
+      ComputeFrequencyResponse(num_partitions, H, &H2);
+      ComputeFrequencyResponse_Avx2(num_partitions, H, &H2_Avx2);
+
+      for (size_t p = 0; p < num_partitions; ++p) {
+        for (size_t k = 0; k < H2[p].size(); ++k) {
+          EXPECT_FLOAT_EQ(H2[p][k], H2_Avx2[p][k]);
+        }
+      }
+    }
+  }
+}
+
+#endif
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Verifies that the check for non-null data dumper works.
+TEST(AdaptiveFirFilterDeathTest, NullDataDumper) {
+  EXPECT_DEATH(AdaptiveFirFilter(9, 9, 250, 1, DetectOptimization(), nullptr),
+               "");
+}
+
+// Verifies that the check for non-null filter output works.
+TEST(AdaptiveFirFilterDeathTest, NullFilterOutput) {
+  ApmDataDumper data_dumper(42);
+  AdaptiveFirFilter filter(9, 9, 250, 1, DetectOptimization(), &data_dumper);
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(EchoCanceller3Config(), 48000, 1));
+  EXPECT_DEATH(filter.Filter(*render_delay_buffer->GetRenderBuffer(), nullptr),
+               "");
+}
+
+#endif
+
+// Verifies that the filter statistics can be accessed when filter statistics
+// are turned on.
+TEST(AdaptiveFirFilterTest, FilterStatisticsAccess) {
+  ApmDataDumper data_dumper(42);
+  Aec3Optimization optimization = DetectOptimization();
+  AdaptiveFirFilter filter(9, 9, 250, 1, optimization, &data_dumper);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> H2(
+      filter.max_filter_size_partitions(),
+      std::array<float, kFftLengthBy2Plus1>());
+  for (auto& H2_k : H2) {
+    H2_k.fill(0.f);
+  }
+
+  std::array<float, kFftLengthBy2Plus1> erl;
+  ComputeErl(optimization, H2, erl);
+  filter.ComputeFrequencyResponse(&H2);
+}
+
+// Verifies that the filter size if correctly repported.
+TEST(AdaptiveFirFilterTest, FilterSize) {
+  ApmDataDumper data_dumper(42);
+  for (size_t filter_size = 1; filter_size < 5; ++filter_size) {
+    AdaptiveFirFilter filter(filter_size, filter_size, 250, 1,
+                             DetectOptimization(), &data_dumper);
+    EXPECT_EQ(filter_size, filter.SizePartitions());
+  }
+}
+
+class AdaptiveFirFilterMultiChannel
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         AdaptiveFirFilterMultiChannel,
+                         ::testing::Combine(::testing::Values(1, 4),
+                                            ::testing::Values(1, 8)));
+
+// Verifies that the filter is being able to properly filter a signal and to
+// adapt its coefficients.
+TEST_P(AdaptiveFirFilterMultiChannel, FilterAndAdapt) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  constexpr size_t kNumBlocksToProcessPerRenderChannel = 1000;
+
+  ApmDataDumper data_dumper(42);
+  EchoCanceller3Config config;
+
+  if (num_render_channels == 33) {
+    config.filter.refined = {13, 0.00005f, 0.0005f, 0.0001f, 2.f, 20075344.f};
+    config.filter.coarse = {13, 0.1f, 20075344.f};
+    config.filter.refined_initial = {12, 0.005f, 0.5f, 0.001f, 2.f, 20075344.f};
+    config.filter.coarse_initial = {12, 0.7f, 20075344.f};
+  }
+
+  AdaptiveFirFilter filter(
+      config.filter.refined.length_blocks, config.filter.refined.length_blocks,
+      config.filter.config_change_duration_blocks, num_render_channels,
+      DetectOptimization(), &data_dumper);
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> H2(
+      num_capture_channels, std::vector<std::array<float, kFftLengthBy2Plus1>>(
+                                filter.max_filter_size_partitions(),
+                                std::array<float, kFftLengthBy2Plus1>()));
+  std::vector<std::vector<float>> h(
+      num_capture_channels,
+      std::vector<float>(
+          GetTimeDomainLength(filter.max_filter_size_partitions()), 0.f));
+  Aec3Fft fft;
+  config.delay.default_delay = 1;
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, num_render_channels));
+  CoarseFilterUpdateGain gain(config.filter.coarse,
+                              config.filter.config_change_duration_blocks);
+  Random random_generator(42U);
+  Block x(kNumBands, num_render_channels);
+  std::vector<float> n(kBlockSize, 0.f);
+  std::vector<float> y(kBlockSize, 0.f);
+  AecState aec_state(EchoCanceller3Config{}, num_capture_channels);
+  RenderSignalAnalyzer render_signal_analyzer(config);
+  absl::optional<DelayEstimate> delay_estimate;
+  std::vector<float> e(kBlockSize, 0.f);
+  std::array<float, kFftLength> s_scratch;
+  std::vector<SubtractorOutput> output(num_capture_channels);
+  FftData S;
+  FftData G;
+  FftData E;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(num_capture_channels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2_refined(
+      num_capture_channels);
+  std::array<float, kFftLengthBy2Plus1> E2_coarse;
+  // [B,A] = butter(2,100/8000,'high')
+  constexpr CascadedBiQuadFilter::BiQuadCoefficients
+      kHighPassFilterCoefficients = {{0.97261f, -1.94523f, 0.97261f},
+                                     {-1.94448f, 0.94598f}};
+  for (auto& Y2_ch : Y2) {
+    Y2_ch.fill(0.f);
+  }
+  for (auto& E2_refined_ch : E2_refined) {
+    E2_refined_ch.fill(0.f);
+  }
+  E2_coarse.fill(0.f);
+  for (auto& subtractor_output : output) {
+    subtractor_output.Reset();
+  }
+
+  constexpr float kScale = 1.0f / kFftLengthBy2;
+
+  for (size_t delay_samples : {0, 64, 150, 200, 301}) {
+    std::vector<DelayBuffer<float>> delay_buffer(
+        num_render_channels, DelayBuffer<float>(delay_samples));
+    std::vector<std::unique_ptr<CascadedBiQuadFilter>> x_hp_filter(
+        num_render_channels);
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      x_hp_filter[ch] = std::make_unique<CascadedBiQuadFilter>(
+          kHighPassFilterCoefficients, 1);
+    }
+    CascadedBiQuadFilter y_hp_filter(kHighPassFilterCoefficients, 1);
+
+    SCOPED_TRACE(ProduceDebugText(num_render_channels, delay_samples));
+    const size_t num_blocks_to_process =
+        kNumBlocksToProcessPerRenderChannel * num_render_channels;
+    for (size_t j = 0; j < num_blocks_to_process; ++j) {
+      std::fill(y.begin(), y.end(), 0.f);
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        RandomizeSampleVector(&random_generator, x.View(/*band=*/0, ch));
+        std::array<float, kBlockSize> y_channel;
+        delay_buffer[ch].Delay(x.View(/*band=*/0, ch), y_channel);
+        for (size_t k = 0; k < y.size(); ++k) {
+          y[k] += y_channel[k] / num_render_channels;
+        }
+      }
+
+      RandomizeSampleVector(&random_generator, n);
+      const float noise_scaling = 1.f / 100.f / num_render_channels;
+      for (size_t k = 0; k < y.size(); ++k) {
+        y[k] += n[k] * noise_scaling;
+      }
+
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        x_hp_filter[ch]->Process(x.View(/*band=*/0, ch));
+      }
+      y_hp_filter.Process(y);
+
+      render_delay_buffer->Insert(x);
+      if (j == 0) {
+        render_delay_buffer->Reset();
+      }
+      render_delay_buffer->PrepareCaptureProcessing();
+      auto* const render_buffer = render_delay_buffer->GetRenderBuffer();
+
+      render_signal_analyzer.Update(*render_buffer,
+                                    aec_state.MinDirectPathFilterDelay());
+
+      filter.Filter(*render_buffer, &S);
+      fft.Ifft(S, &s_scratch);
+      std::transform(y.begin(), y.end(), s_scratch.begin() + kFftLengthBy2,
+                     e.begin(),
+                     [&](float a, float b) { return a - b * kScale; });
+      std::for_each(e.begin(), e.end(),
+                    [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
+      fft.ZeroPaddedFft(e, Aec3Fft::Window::kRectangular, &E);
+      for (auto& o : output) {
+        for (size_t k = 0; k < kBlockSize; ++k) {
+          o.s_refined[k] = kScale * s_scratch[k + kFftLengthBy2];
+        }
+      }
+
+      std::array<float, kFftLengthBy2Plus1> render_power;
+      render_buffer->SpectralSum(filter.SizePartitions(), &render_power);
+      gain.Compute(render_power, render_signal_analyzer, E,
+                   filter.SizePartitions(), false, &G);
+      filter.Adapt(*render_buffer, G, &h[0]);
+      aec_state.HandleEchoPathChange(EchoPathVariability(
+          false, EchoPathVariability::DelayAdjustment::kNone, false));
+
+      filter.ComputeFrequencyResponse(&H2[0]);
+      aec_state.Update(delay_estimate, H2, h, *render_buffer, E2_refined, Y2,
+                       output);
+    }
+    // Verify that the filter is able to perform well.
+    EXPECT_LT(1000 * std::inner_product(e.begin(), e.end(), e.begin(), 0.f),
+              std::inner_product(y.begin(), y.end(), y.begin(), 0.f));
+  }
+}
+
+}  // namespace aec3
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_avx2_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_avx2_gn/moz.build
new file mode 100644
index 0000000000..6f67bd6fad
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_avx2_gn/moz.build
@@ -0,0 +1,190 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+CXXFLAGS += [
+    "-mavx2",
+    "-mfma"
+]
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_AVX2"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_avx2.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_avx2.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_avx2.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+Library("aec3_avx2_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common.cc b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common.cc
new file mode 100644
index 0000000000..3ba10d5baf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+#include <stdint.h>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+
+namespace webrtc {
+
+Aec3Optimization DetectOptimization() {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+  if (GetCPUInfo(kAVX2) != 0) {
+    return Aec3Optimization::kAvx2;
+  } else if (GetCPUInfo(kSSE2) != 0) {
+    return Aec3Optimization::kSse2;
+  }
+#endif
+
+#if defined(WEBRTC_HAS_NEON)
+  return Aec3Optimization::kNeon;
+#else
+  return Aec3Optimization::kNone;
+#endif
+}
+
+float FastApproxLog2f(const float in) {
+  RTC_DCHECK_GT(in, .0f);
+  // Read and interpret float as uint32_t and then cast to float.
+  // This is done to extract the exponent (bits 30 - 23).
+  // "Right shift" of the exponent is then performed by multiplying
+  // with the constant (1/2^23). Finally, we subtract a constant to
+  // remove the bias (https://en.wikipedia.org/wiki/Exponent_bias).
+  union {
+    float dummy;
+    uint32_t a;
+  } x = {in};
+  float out = x.a;
+  out *= 1.1920929e-7f;  // 1/2^23
+  out -= 126.942695f;    // Remove bias.
+  return out;
+}
+
+float Log2TodB(const float in_log2) {
+  return 3.0102999566398121 * in_log2;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common.h b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common.h
new file mode 100644
index 0000000000..32b564f14b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+
+#ifdef _MSC_VER /* visual c++ */
+#define ALIGN16_BEG __declspec(align(16))
+#define ALIGN16_END
+#else /* gcc or icc */
+#define ALIGN16_BEG
+#define ALIGN16_END __attribute__((aligned(16)))
+#endif
+
+enum class Aec3Optimization { kNone, kSse2, kAvx2, kNeon };
+
+constexpr int kNumBlocksPerSecond = 250;
+
+constexpr int kMetricsReportingIntervalBlocks = 10 * kNumBlocksPerSecond;
+constexpr int kMetricsComputationBlocks = 3;
+constexpr int kMetricsCollectionBlocks =
+    kMetricsReportingIntervalBlocks - kMetricsComputationBlocks;
+
+constexpr size_t kFftLengthBy2 = 64;
+constexpr size_t kFftLengthBy2Plus1 = kFftLengthBy2 + 1;
+constexpr size_t kFftLengthBy2Minus1 = kFftLengthBy2 - 1;
+constexpr size_t kFftLength = 2 * kFftLengthBy2;
+constexpr size_t kFftLengthBy2Log2 = 6;
+
+constexpr int kRenderTransferQueueSizeFrames = 100;
+
+constexpr size_t kMaxNumBands = 3;
+constexpr size_t kFrameSize = 160;
+constexpr size_t kSubFrameLength = kFrameSize / 2;
+
+constexpr size_t kBlockSize = kFftLengthBy2;
+constexpr size_t kBlockSizeLog2 = kFftLengthBy2Log2;
+
+constexpr size_t kExtendedBlockSize = 2 * kFftLengthBy2;
+constexpr size_t kMatchedFilterWindowSizeSubBlocks = 32;
+constexpr size_t kMatchedFilterAlignmentShiftSizeSubBlocks =
+    kMatchedFilterWindowSizeSubBlocks * 3 / 4;
+
+// TODO(peah): Integrate this with how it is done inside audio_processing_impl.
+constexpr size_t NumBandsForRate(int sample_rate_hz) {
+  return static_cast<size_t>(sample_rate_hz / 16000);
+}
+
+constexpr bool ValidFullBandRate(int sample_rate_hz) {
+  return sample_rate_hz == 16000 || sample_rate_hz == 32000 ||
+         sample_rate_hz == 48000;
+}
+
+constexpr int GetTimeDomainLength(int filter_length_blocks) {
+  return filter_length_blocks * kFftLengthBy2;
+}
+
+constexpr size_t GetDownSampledBufferSize(size_t down_sampling_factor,
+                                          size_t num_matched_filters) {
+  return kBlockSize / down_sampling_factor *
+         (kMatchedFilterAlignmentShiftSizeSubBlocks * num_matched_filters +
+          kMatchedFilterWindowSizeSubBlocks + 1);
+}
+
+constexpr size_t GetRenderDelayBufferSize(size_t down_sampling_factor,
+                                          size_t num_matched_filters,
+                                          size_t filter_length_blocks) {
+  return GetDownSampledBufferSize(down_sampling_factor, num_matched_filters) /
+             (kBlockSize / down_sampling_factor) +
+         filter_length_blocks + 1;
+}
+
+// Detects what kind of optimizations to use for the code.
+Aec3Optimization DetectOptimization();
+
+// Computes the log2 of the input in a fast an approximate manner.
+float FastApproxLog2f(float in);
+
+// Returns dB from a power quantity expressed in log2.
+float Log2TodB(float in_log2);
+
+static_assert(1 << kBlockSizeLog2 == kBlockSize,
+              "Proper number of shifts for blocksize");
+
+static_assert(1 << kFftLengthBy2Log2 == kFftLengthBy2,
+              "Proper number of shifts for the fft length");
+
+static_assert(1 == NumBandsForRate(16000), "Number of bands for 16 kHz");
+static_assert(2 == NumBandsForRate(32000), "Number of bands for 32 kHz");
+static_assert(3 == NumBandsForRate(48000), "Number of bands for 48 kHz");
+
+static_assert(ValidFullBandRate(16000),
+              "Test that 16 kHz is a valid sample rate");
+static_assert(ValidFullBandRate(32000),
+              "Test that 32 kHz is a valid sample rate");
+static_assert(ValidFullBandRate(48000),
+              "Test that 48 kHz is a valid sample rate");
+static_assert(!ValidFullBandRate(8001),
+              "Test that 8001 Hz is not a valid sample rate");
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common_gn/moz.build
new file mode 100644
index 0000000000..b0952a7d0c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("aec3_common_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft.cc b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft.cc
new file mode 100644
index 0000000000..9cc8016f0b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft.cc
@@ -0,0 +1,144 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/aec3_fft.h"
+
+#include <algorithm>
+#include <functional>
+#include <iterator>
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+
+namespace webrtc {
+
+namespace {
+
+const float kHanning64[kFftLengthBy2] = {
+    0.f,         0.00248461f, 0.00991376f, 0.0222136f,  0.03926189f,
+    0.06088921f, 0.08688061f, 0.11697778f, 0.15088159f, 0.1882551f,
+    0.22872687f, 0.27189467f, 0.31732949f, 0.36457977f, 0.41317591f,
+    0.46263495f, 0.51246535f, 0.56217185f, 0.61126047f, 0.65924333f,
+    0.70564355f, 0.75f,       0.79187184f, 0.83084292f, 0.86652594f,
+    0.89856625f, 0.92664544f, 0.95048443f, 0.96984631f, 0.98453864f,
+    0.99441541f, 0.99937846f, 0.99937846f, 0.99441541f, 0.98453864f,
+    0.96984631f, 0.95048443f, 0.92664544f, 0.89856625f, 0.86652594f,
+    0.83084292f, 0.79187184f, 0.75f,       0.70564355f, 0.65924333f,
+    0.61126047f, 0.56217185f, 0.51246535f, 0.46263495f, 0.41317591f,
+    0.36457977f, 0.31732949f, 0.27189467f, 0.22872687f, 0.1882551f,
+    0.15088159f, 0.11697778f, 0.08688061f, 0.06088921f, 0.03926189f,
+    0.0222136f,  0.00991376f, 0.00248461f, 0.f};
+
+// Hanning window from Matlab command win = sqrt(hanning(128)).
+const float kSqrtHanning128[kFftLength] = {
+    0.00000000000000f, 0.02454122852291f, 0.04906767432742f, 0.07356456359967f,
+    0.09801714032956f, 0.12241067519922f, 0.14673047445536f, 0.17096188876030f,
+    0.19509032201613f, 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
+    0.29028467725446f, 0.31368174039889f, 0.33688985339222f, 0.35989503653499f,
+    0.38268343236509f, 0.40524131400499f, 0.42755509343028f, 0.44961132965461f,
+    0.47139673682600f, 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
+    0.55557023301960f, 0.57580819141785f, 0.59569930449243f, 0.61523159058063f,
+    0.63439328416365f, 0.65317284295378f, 0.67155895484702f, 0.68954054473707f,
+    0.70710678118655f, 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
+    0.77301045336274f, 0.78834642762661f, 0.80320753148064f, 0.81758481315158f,
+    0.83146961230255f, 0.84485356524971f, 0.85772861000027f, 0.87008699110871f,
+    0.88192126434835f, 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
+    0.92387953251129f, 0.93299279883474f, 0.94154406518302f, 0.94952818059304f,
+    0.95694033573221f, 0.96377606579544f, 0.97003125319454f, 0.97570213003853f,
+    0.98078528040323f, 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
+    0.99518472667220f, 0.99729045667869f, 0.99879545620517f, 0.99969881869620f,
+    1.00000000000000f, 0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
+    0.99518472667220f, 0.99247953459871f, 0.98917650996478f, 0.98527764238894f,
+    0.98078528040323f, 0.97570213003853f, 0.97003125319454f, 0.96377606579544f,
+    0.95694033573221f, 0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
+    0.92387953251129f, 0.91420975570353f, 0.90398929312344f, 0.89322430119552f,
+    0.88192126434835f, 0.87008699110871f, 0.85772861000027f, 0.84485356524971f,
+    0.83146961230255f, 0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
+    0.77301045336274f, 0.75720884650648f, 0.74095112535496f, 0.72424708295147f,
+    0.70710678118655f, 0.68954054473707f, 0.67155895484702f, 0.65317284295378f,
+    0.63439328416365f, 0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
+    0.55557023301960f, 0.53499761988710f, 0.51410274419322f, 0.49289819222978f,
+    0.47139673682600f, 0.44961132965461f, 0.42755509343028f, 0.40524131400499f,
+    0.38268343236509f, 0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
+    0.29028467725446f, 0.26671275747490f, 0.24298017990326f, 0.21910124015687f,
+    0.19509032201613f, 0.17096188876030f, 0.14673047445536f, 0.12241067519922f,
+    0.09801714032956f, 0.07356456359967f, 0.04906767432742f, 0.02454122852291f};
+
+bool IsSse2Available() {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+  return GetCPUInfo(kSSE2) != 0;
+#else
+  return false;
+#endif
+}
+
+}  // namespace
+
+Aec3Fft::Aec3Fft() : ooura_fft_(IsSse2Available()) {}
+
+// TODO(peah): Change x to be std::array once the rest of the code allows this.
+void Aec3Fft::ZeroPaddedFft(rtc::ArrayView<const float> x,
+                            Window window,
+                            FftData* X) const {
+  RTC_DCHECK(X);
+  RTC_DCHECK_EQ(kFftLengthBy2, x.size());
+  std::array<float, kFftLength> fft;
+  std::fill(fft.begin(), fft.begin() + kFftLengthBy2, 0.f);
+  switch (window) {
+    case Window::kRectangular:
+      std::copy(x.begin(), x.end(), fft.begin() + kFftLengthBy2);
+      break;
+    case Window::kHanning:
+      std::transform(x.begin(), x.end(), std::begin(kHanning64),
+                     fft.begin() + kFftLengthBy2,
+                     [](float a, float b) { return a * b; });
+      break;
+    case Window::kSqrtHanning:
+      RTC_DCHECK_NOTREACHED();
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+
+  Fft(&fft, X);
+}
+
+void Aec3Fft::PaddedFft(rtc::ArrayView<const float> x,
+                        rtc::ArrayView<const float> x_old,
+                        Window window,
+                        FftData* X) const {
+  RTC_DCHECK(X);
+  RTC_DCHECK_EQ(kFftLengthBy2, x.size());
+  RTC_DCHECK_EQ(kFftLengthBy2, x_old.size());
+  std::array<float, kFftLength> fft;
+
+  switch (window) {
+    case Window::kRectangular:
+      std::copy(x_old.begin(), x_old.end(), fft.begin());
+      std::copy(x.begin(), x.end(), fft.begin() + x_old.size());
+      break;
+    case Window::kHanning:
+      RTC_DCHECK_NOTREACHED();
+      break;
+    case Window::kSqrtHanning:
+      std::transform(x_old.begin(), x_old.end(), std::begin(kSqrtHanning128),
+                     fft.begin(), std::multiplies<float>());
+      std::transform(x.begin(), x.end(),
+                     std::begin(kSqrtHanning128) + x_old.size(),
+                     fft.begin() + x_old.size(), std::multiplies<float>());
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+
+  Fft(&fft, X);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft.h b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft.h
new file mode 100644
index 0000000000..c68de53963
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_AEC3_FFT_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_AEC3_FFT_H_
+
+#include <array>
+
+#include "api/array_view.h"
+#include "common_audio/third_party/ooura/fft_size_128/ooura_fft.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Wrapper class that provides 128 point real valued FFT functionality with the
+// FftData type.
+class Aec3Fft {
+ public:
+  enum class Window { kRectangular, kHanning, kSqrtHanning };
+
+  Aec3Fft();
+
+  Aec3Fft(const Aec3Fft&) = delete;
+  Aec3Fft& operator=(const Aec3Fft&) = delete;
+
+  // Computes the FFT. Note that both the input and output are modified.
+  void Fft(std::array<float, kFftLength>* x, FftData* X) const {
+    RTC_DCHECK(x);
+    RTC_DCHECK(X);
+    ooura_fft_.Fft(x->data());
+    X->CopyFromPackedArray(*x);
+  }
+  // Computes the inverse Fft.
+  void Ifft(const FftData& X, std::array<float, kFftLength>* x) const {
+    RTC_DCHECK(x);
+    X.CopyToPackedArray(x);
+    ooura_fft_.InverseFft(x->data());
+  }
+
+  // Windows the input using a Hanning window, and then adds padding of
+  // kFftLengthBy2 initial zeros before computing the Fft.
+  void ZeroPaddedFft(rtc::ArrayView<const float> x,
+                     Window window,
+                     FftData* X) const;
+
+  // Concatenates the kFftLengthBy2 values long x and x_old before computing the
+  // Fft. After that, x is copied to x_old.
+  void PaddedFft(rtc::ArrayView<const float> x,
+                 rtc::ArrayView<const float> x_old,
+                 FftData* X) const {
+    PaddedFft(x, x_old, Window::kRectangular, X);
+  }
+
+  // Padded Fft using a time-domain window.
+  void PaddedFft(rtc::ArrayView<const float> x,
+                 rtc::ArrayView<const float> x_old,
+                 Window window,
+                 FftData* X) const;
+
+ private:
+  const OouraFft ooura_fft_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_AEC3_FFT_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_gn/moz.build
new file mode 100644
index 0000000000..97bbc43539
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_gn/moz.build
@@ -0,0 +1,216 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("aec3_fft_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_unittest.cc
new file mode 100644
index 0000000000..e60ef5b713
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft_unittest.cc
@@ -0,0 +1,213 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/aec3_fft.h"
+
+#include <algorithm>
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies that the check for non-null input in Fft works.
+TEST(Aec3FftDeathTest, NullFftInput) {
+  Aec3Fft fft;
+  FftData X;
+  EXPECT_DEATH(fft.Fft(nullptr, &X), "");
+}
+
+// Verifies that the check for non-null input in Fft works.
+TEST(Aec3FftDeathTest, NullFftOutput) {
+  Aec3Fft fft;
+  std::array<float, kFftLength> x;
+  EXPECT_DEATH(fft.Fft(&x, nullptr), "");
+}
+
+// Verifies that the check for non-null output in Ifft works.
+TEST(Aec3FftDeathTest, NullIfftOutput) {
+  Aec3Fft fft;
+  FftData X;
+  EXPECT_DEATH(fft.Ifft(X, nullptr), "");
+}
+
+// Verifies that the check for non-null output in ZeroPaddedFft works.
+TEST(Aec3FftDeathTest, NullZeroPaddedFftOutput) {
+  Aec3Fft fft;
+  std::array<float, kFftLengthBy2> x;
+  EXPECT_DEATH(fft.ZeroPaddedFft(x, Aec3Fft::Window::kRectangular, nullptr),
+               "");
+}
+
+// Verifies that the check for input length in ZeroPaddedFft works.
+TEST(Aec3FftDeathTest, ZeroPaddedFftWrongInputLength) {
+  Aec3Fft fft;
+  FftData X;
+  std::array<float, kFftLengthBy2 - 1> x;
+  EXPECT_DEATH(fft.ZeroPaddedFft(x, Aec3Fft::Window::kRectangular, &X), "");
+}
+
+// Verifies that the check for non-null output in PaddedFft works.
+TEST(Aec3FftDeathTest, NullPaddedFftOutput) {
+  Aec3Fft fft;
+  std::array<float, kFftLengthBy2> x;
+  std::array<float, kFftLengthBy2> x_old;
+  EXPECT_DEATH(fft.PaddedFft(x, x_old, nullptr), "");
+}
+
+// Verifies that the check for input length in PaddedFft works.
+TEST(Aec3FftDeathTest, PaddedFftWrongInputLength) {
+  Aec3Fft fft;
+  FftData X;
+  std::array<float, kFftLengthBy2 - 1> x;
+  std::array<float, kFftLengthBy2> x_old;
+  EXPECT_DEATH(fft.PaddedFft(x, x_old, &X), "");
+}
+
+// Verifies that the check for length in the old value in PaddedFft works.
+TEST(Aec3FftDeathTest, PaddedFftWrongOldValuesLength) {
+  Aec3Fft fft;
+  FftData X;
+  std::array<float, kFftLengthBy2> x;
+  std::array<float, kFftLengthBy2 - 1> x_old;
+  EXPECT_DEATH(fft.PaddedFft(x, x_old, &X), "");
+}
+
+#endif
+
+// Verifies that Fft works as intended.
+TEST(Aec3Fft, Fft) {
+  Aec3Fft fft;
+  FftData X;
+  std::array<float, kFftLength> x;
+  x.fill(0.f);
+  fft.Fft(&x, &X);
+  EXPECT_THAT(X.re, ::testing::Each(0.f));
+  EXPECT_THAT(X.im, ::testing::Each(0.f));
+
+  x.fill(0.f);
+  x[0] = 1.f;
+  fft.Fft(&x, &X);
+  EXPECT_THAT(X.re, ::testing::Each(1.f));
+  EXPECT_THAT(X.im, ::testing::Each(0.f));
+
+  x.fill(1.f);
+  fft.Fft(&x, &X);
+  EXPECT_EQ(128.f, X.re[0]);
+  std::for_each(X.re.begin() + 1, X.re.end(),
+                [](float a) { EXPECT_EQ(0.f, a); });
+  EXPECT_THAT(X.im, ::testing::Each(0.f));
+}
+
+// Verifies that InverseFft works as intended.
+TEST(Aec3Fft, Ifft) {
+  Aec3Fft fft;
+  FftData X;
+  std::array<float, kFftLength> x;
+
+  X.re.fill(0.f);
+  X.im.fill(0.f);
+  fft.Ifft(X, &x);
+  EXPECT_THAT(x, ::testing::Each(0.f));
+
+  X.re.fill(1.f);
+  X.im.fill(0.f);
+  fft.Ifft(X, &x);
+  EXPECT_EQ(64.f, x[0]);
+  std::for_each(x.begin() + 1, x.end(), [](float a) { EXPECT_EQ(0.f, a); });
+
+  X.re.fill(0.f);
+  X.re[0] = 128;
+  X.im.fill(0.f);
+  fft.Ifft(X, &x);
+  EXPECT_THAT(x, ::testing::Each(64.f));
+}
+
+// Verifies that InverseFft and Fft work as intended.
+TEST(Aec3Fft, FftAndIfft) {
+  Aec3Fft fft;
+  FftData X;
+  std::array<float, kFftLength> x;
+  std::array<float, kFftLength> x_ref;
+
+  int v = 0;
+  for (int k = 0; k < 20; ++k) {
+    for (size_t j = 0; j < x.size(); ++j) {
+      x[j] = v++;
+      x_ref[j] = x[j] * 64.f;
+    }
+    fft.Fft(&x, &X);
+    fft.Ifft(X, &x);
+    for (size_t j = 0; j < x.size(); ++j) {
+      EXPECT_NEAR(x_ref[j], x[j], 0.001f);
+    }
+  }
+}
+
+// Verifies that ZeroPaddedFft work as intended.
+TEST(Aec3Fft, ZeroPaddedFft) {
+  Aec3Fft fft;
+  FftData X;
+  std::array<float, kFftLengthBy2> x_in;
+  std::array<float, kFftLength> x_ref;
+  std::array<float, kFftLength> x_out;
+
+  int v = 0;
+  x_ref.fill(0.f);
+  for (int k = 0; k < 20; ++k) {
+    for (size_t j = 0; j < x_in.size(); ++j) {
+      x_in[j] = v++;
+      x_ref[j + kFftLengthBy2] = x_in[j] * 64.f;
+    }
+    fft.ZeroPaddedFft(x_in, Aec3Fft::Window::kRectangular, &X);
+    fft.Ifft(X, &x_out);
+    for (size_t j = 0; j < x_out.size(); ++j) {
+      EXPECT_NEAR(x_ref[j], x_out[j], 0.1f);
+    }
+  }
+}
+
+// Verifies that ZeroPaddedFft work as intended.
+TEST(Aec3Fft, PaddedFft) {
+  Aec3Fft fft;
+  FftData X;
+  std::array<float, kFftLengthBy2> x_in;
+  std::array<float, kFftLength> x_out;
+  std::array<float, kFftLengthBy2> x_old;
+  std::array<float, kFftLengthBy2> x_old_ref;
+  std::array<float, kFftLength> x_ref;
+
+  int v = 0;
+  x_old.fill(0.f);
+  for (int k = 0; k < 20; ++k) {
+    for (size_t j = 0; j < x_in.size(); ++j) {
+      x_in[j] = v++;
+    }
+
+    std::copy(x_old.begin(), x_old.end(), x_ref.begin());
+    std::copy(x_in.begin(), x_in.end(), x_ref.begin() + kFftLengthBy2);
+    std::copy(x_in.begin(), x_in.end(), x_old_ref.begin());
+    std::for_each(x_ref.begin(), x_ref.end(), [](float& a) { a *= 64.f; });
+
+    fft.PaddedFft(x_in, x_old, &X);
+    std::copy(x_in.begin(), x_in.end(), x_old.begin());
+    fft.Ifft(X, &x_out);
+
+    for (size_t j = 0; j < x_out.size(); ++j) {
+      EXPECT_NEAR(x_ref[j], x_out[j], 0.1f);
+    }
+
+    EXPECT_EQ(x_old_ref, x_old);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec3_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_gn/moz.build
new file mode 100644
index 0000000000..6646d41ff3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec3_gn/moz.build
@@ -0,0 +1,289 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/aec3_common.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/aec3_fft.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/aec_state.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/block_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/block_framer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/block_processor.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/config_selector.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/decimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/dominant_nearend_detector.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/echo_audibility.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/fft_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/moving_average.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/reverb_frequency_response.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/spectrum_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/stationarity_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/subband_erle_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/subband_nearend_detector.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/subtractor.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aec3/transparent_mode.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("aec3_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec_state.cc b/third_party/libwebrtc/modules/audio_processing/aec3/aec_state.cc
new file mode 100644
index 0000000000..81fd91fab9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec_state.cc
@@ -0,0 +1,481 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/aec_state.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <numeric>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+bool DeactivateInitialStateResetAtEchoPathChange() {
+  return field_trial::IsEnabled(
+      "WebRTC-Aec3DeactivateInitialStateResetKillSwitch");
+}
+
+bool FullResetAtEchoPathChange() {
+  return !field_trial::IsEnabled("WebRTC-Aec3AecStateFullResetKillSwitch");
+}
+
+bool SubtractorAnalyzerResetAtEchoPathChange() {
+  return !field_trial::IsEnabled(
+      "WebRTC-Aec3AecStateSubtractorAnalyzerResetKillSwitch");
+}
+
+void ComputeAvgRenderReverb(
+    const SpectrumBuffer& spectrum_buffer,
+    int delay_blocks,
+    float reverb_decay,
+    ReverbModel* reverb_model,
+    rtc::ArrayView<float, kFftLengthBy2Plus1> reverb_power_spectrum) {
+  RTC_DCHECK(reverb_model);
+  const size_t num_render_channels = spectrum_buffer.buffer[0].size();
+  int idx_at_delay =
+      spectrum_buffer.OffsetIndex(spectrum_buffer.read, delay_blocks);
+  int idx_past = spectrum_buffer.IncIndex(idx_at_delay);
+
+  std::array<float, kFftLengthBy2Plus1> X2_data;
+  rtc::ArrayView<const float> X2;
+  if (num_render_channels > 1) {
+    auto average_channels =
+        [](size_t num_render_channels,
+           rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+               spectrum_band_0,
+           rtc::ArrayView<float, kFftLengthBy2Plus1> render_power) {
+          std::fill(render_power.begin(), render_power.end(), 0.f);
+          for (size_t ch = 0; ch < num_render_channels; ++ch) {
+            for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+              render_power[k] += spectrum_band_0[ch][k];
+            }
+          }
+          const float normalizer = 1.f / num_render_channels;
+          for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+            render_power[k] *= normalizer;
+          }
+        };
+    average_channels(num_render_channels, spectrum_buffer.buffer[idx_past],
+                     X2_data);
+    reverb_model->UpdateReverbNoFreqShaping(
+        X2_data, /*power_spectrum_scaling=*/1.0f, reverb_decay);
+
+    average_channels(num_render_channels, spectrum_buffer.buffer[idx_at_delay],
+                     X2_data);
+    X2 = X2_data;
+  } else {
+    reverb_model->UpdateReverbNoFreqShaping(
+        spectrum_buffer.buffer[idx_past][/*channel=*/0],
+        /*power_spectrum_scaling=*/1.0f, reverb_decay);
+
+    X2 = spectrum_buffer.buffer[idx_at_delay][/*channel=*/0];
+  }
+
+  rtc::ArrayView<const float, kFftLengthBy2Plus1> reverb_power =
+      reverb_model->reverb();
+  for (size_t k = 0; k < X2.size(); ++k) {
+    reverb_power_spectrum[k] = X2[k] + reverb_power[k];
+  }
+}
+
+}  // namespace
+
+std::atomic<int> AecState::instance_count_(0);
+
+void AecState::GetResidualEchoScaling(
+    rtc::ArrayView<float> residual_scaling) const {
+  bool filter_has_had_time_to_converge;
+  if (config_.filter.conservative_initial_phase) {
+    filter_has_had_time_to_converge =
+        strong_not_saturated_render_blocks_ >= 1.5f * kNumBlocksPerSecond;
+  } else {
+    filter_has_had_time_to_converge =
+        strong_not_saturated_render_blocks_ >= 0.8f * kNumBlocksPerSecond;
+  }
+  echo_audibility_.GetResidualEchoScaling(filter_has_had_time_to_converge,
+                                          residual_scaling);
+}
+
+AecState::AecState(const EchoCanceller3Config& config,
+                   size_t num_capture_channels)
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      config_(config),
+      num_capture_channels_(num_capture_channels),
+      deactivate_initial_state_reset_at_echo_path_change_(
+          DeactivateInitialStateResetAtEchoPathChange()),
+      full_reset_at_echo_path_change_(FullResetAtEchoPathChange()),
+      subtractor_analyzer_reset_at_echo_path_change_(
+          SubtractorAnalyzerResetAtEchoPathChange()),
+      initial_state_(config_),
+      delay_state_(config_, num_capture_channels_),
+      transparent_state_(TransparentMode::Create(config_)),
+      filter_quality_state_(config_, num_capture_channels_),
+      erl_estimator_(2 * kNumBlocksPerSecond),
+      erle_estimator_(2 * kNumBlocksPerSecond, config_, num_capture_channels_),
+      filter_analyzer_(config_, num_capture_channels_),
+      echo_audibility_(
+          config_.echo_audibility.use_stationarity_properties_at_init),
+      reverb_model_estimator_(config_, num_capture_channels_),
+      subtractor_output_analyzer_(num_capture_channels_) {}
+
+AecState::~AecState() = default;
+
+void AecState::HandleEchoPathChange(
+    const EchoPathVariability& echo_path_variability) {
+  const auto full_reset = [&]() {
+    filter_analyzer_.Reset();
+    capture_signal_saturation_ = false;
+    strong_not_saturated_render_blocks_ = 0;
+    blocks_with_active_render_ = 0;
+    if (!deactivate_initial_state_reset_at_echo_path_change_) {
+      initial_state_.Reset();
+    }
+    if (transparent_state_) {
+      transparent_state_->Reset();
+    }
+    erle_estimator_.Reset(true);
+    erl_estimator_.Reset();
+    filter_quality_state_.Reset();
+  };
+
+  // TODO(peah): Refine the reset scheme according to the type of gain and
+  // delay adjustment.
+
+  if (full_reset_at_echo_path_change_ &&
+      echo_path_variability.delay_change !=
+          EchoPathVariability::DelayAdjustment::kNone) {
+    full_reset();
+  } else if (echo_path_variability.gain_change) {
+    erle_estimator_.Reset(false);
+  }
+  if (subtractor_analyzer_reset_at_echo_path_change_) {
+    subtractor_output_analyzer_.HandleEchoPathChange();
+  }
+}
+
+void AecState::Update(
+    const absl::optional<DelayEstimate>& external_delay,
+    rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+        adaptive_filter_frequency_responses,
+    rtc::ArrayView<const std::vector<float>> adaptive_filter_impulse_responses,
+    const RenderBuffer& render_buffer,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2_refined,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    rtc::ArrayView<const SubtractorOutput> subtractor_output) {
+  RTC_DCHECK_EQ(num_capture_channels_, Y2.size());
+  RTC_DCHECK_EQ(num_capture_channels_, subtractor_output.size());
+  RTC_DCHECK_EQ(num_capture_channels_,
+                adaptive_filter_frequency_responses.size());
+  RTC_DCHECK_EQ(num_capture_channels_,
+                adaptive_filter_impulse_responses.size());
+
+  // Analyze the filter outputs and filters.
+  bool any_filter_converged;
+  bool any_coarse_filter_converged;
+  bool all_filters_diverged;
+  subtractor_output_analyzer_.Update(subtractor_output, &any_filter_converged,
+                                     &any_coarse_filter_converged,
+                                     &all_filters_diverged);
+
+  bool any_filter_consistent;
+  float max_echo_path_gain;
+  filter_analyzer_.Update(adaptive_filter_impulse_responses, render_buffer,
+                          &any_filter_consistent, &max_echo_path_gain);
+
+  // Estimate the direct path delay of the filter.
+  if (config_.filter.use_linear_filter) {
+    delay_state_.Update(filter_analyzer_.FilterDelaysBlocks(), external_delay,
+                        strong_not_saturated_render_blocks_);
+  }
+
+  const Block& aligned_render_block =
+      render_buffer.GetBlock(-delay_state_.MinDirectPathFilterDelay());
+
+  // Update render counters.
+  bool active_render = false;
+  for (int ch = 0; ch < aligned_render_block.NumChannels(); ++ch) {
+    const float render_energy =
+        std::inner_product(aligned_render_block.begin(/*block=*/0, ch),
+                           aligned_render_block.end(/*block=*/0, ch),
+                           aligned_render_block.begin(/*block=*/0, ch), 0.f);
+    if (render_energy > (config_.render_levels.active_render_limit *
+                         config_.render_levels.active_render_limit) *
+                            kFftLengthBy2) {
+      active_render = true;
+      break;
+    }
+  }
+  blocks_with_active_render_ += active_render ? 1 : 0;
+  strong_not_saturated_render_blocks_ +=
+      active_render && !SaturatedCapture() ? 1 : 0;
+
+  std::array<float, kFftLengthBy2Plus1> avg_render_spectrum_with_reverb;
+
+  ComputeAvgRenderReverb(render_buffer.GetSpectrumBuffer(),
+                         delay_state_.MinDirectPathFilterDelay(),
+                         ReverbDecay(/*mild=*/false), &avg_render_reverb_,
+                         avg_render_spectrum_with_reverb);
+
+  if (config_.echo_audibility.use_stationarity_properties) {
+    // Update the echo audibility evaluator.
+    echo_audibility_.Update(render_buffer, avg_render_reverb_.reverb(),
+                            delay_state_.MinDirectPathFilterDelay(),
+                            delay_state_.ExternalDelayReported());
+  }
+
+  // Update the ERL and ERLE measures.
+  if (initial_state_.TransitionTriggered()) {
+    erle_estimator_.Reset(false);
+  }
+
+  erle_estimator_.Update(render_buffer, adaptive_filter_frequency_responses,
+                         avg_render_spectrum_with_reverb, Y2, E2_refined,
+                         subtractor_output_analyzer_.ConvergedFilters());
+
+  erl_estimator_.Update(
+      subtractor_output_analyzer_.ConvergedFilters(),
+      render_buffer.Spectrum(delay_state_.MinDirectPathFilterDelay()), Y2);
+
+  // Detect and flag echo saturation.
+  if (config_.ep_strength.echo_can_saturate) {
+    saturation_detector_.Update(aligned_render_block, SaturatedCapture(),
+                                UsableLinearEstimate(), subtractor_output,
+                                max_echo_path_gain);
+  } else {
+    RTC_DCHECK(!saturation_detector_.SaturatedEcho());
+  }
+
+  // Update the decision on whether to use the initial state parameter set.
+  initial_state_.Update(active_render, SaturatedCapture());
+
+  // Detect whether the transparent mode should be activated.
+  if (transparent_state_) {
+    transparent_state_->Update(
+        delay_state_.MinDirectPathFilterDelay(), any_filter_consistent,
+        any_filter_converged, any_coarse_filter_converged, all_filters_diverged,
+        active_render, SaturatedCapture());
+  }
+
+  // Analyze the quality of the filter.
+  filter_quality_state_.Update(active_render, TransparentModeActive(),
+                               SaturatedCapture(), external_delay,
+                               any_filter_converged);
+
+  // Update the reverb estimate.
+  const bool stationary_block =
+      config_.echo_audibility.use_stationarity_properties &&
+      echo_audibility_.IsBlockStationary();
+
+  reverb_model_estimator_.Update(
+      filter_analyzer_.GetAdjustedFilters(),
+      adaptive_filter_frequency_responses,
+      erle_estimator_.GetInstLinearQualityEstimates(),
+      delay_state_.DirectPathFilterDelays(),
+      filter_quality_state_.UsableLinearFilterOutputs(), stationary_block);
+
+  erle_estimator_.Dump(data_dumper_);
+  reverb_model_estimator_.Dump(data_dumper_.get());
+  data_dumper_->DumpRaw("aec3_active_render", active_render);
+  data_dumper_->DumpRaw("aec3_erl", Erl());
+  data_dumper_->DumpRaw("aec3_erl_time_domain", ErlTimeDomain());
+  data_dumper_->DumpRaw("aec3_erle", Erle(/*onset_compensated=*/false)[0]);
+  data_dumper_->DumpRaw("aec3_erle_onset_compensated",
+                        Erle(/*onset_compensated=*/true)[0]);
+  data_dumper_->DumpRaw("aec3_usable_linear_estimate", UsableLinearEstimate());
+  data_dumper_->DumpRaw("aec3_transparent_mode", TransparentModeActive());
+  data_dumper_->DumpRaw("aec3_filter_delay",
+                        filter_analyzer_.MinFilterDelayBlocks());
+
+  data_dumper_->DumpRaw("aec3_any_filter_consistent", any_filter_consistent);
+  data_dumper_->DumpRaw("aec3_initial_state",
+                        initial_state_.InitialStateActive());
+  data_dumper_->DumpRaw("aec3_capture_saturation", SaturatedCapture());
+  data_dumper_->DumpRaw("aec3_echo_saturation", SaturatedEcho());
+  data_dumper_->DumpRaw("aec3_any_filter_converged", any_filter_converged);
+  data_dumper_->DumpRaw("aec3_any_coarse_filter_converged",
+                        any_coarse_filter_converged);
+  data_dumper_->DumpRaw("aec3_all_filters_diverged", all_filters_diverged);
+
+  data_dumper_->DumpRaw("aec3_external_delay_avaliable",
+                        external_delay ? 1 : 0);
+  data_dumper_->DumpRaw("aec3_filter_tail_freq_resp_est",
+                        GetReverbFrequencyResponse());
+  data_dumper_->DumpRaw("aec3_subtractor_y2", subtractor_output[0].y2);
+  data_dumper_->DumpRaw("aec3_subtractor_e2_coarse",
+                        subtractor_output[0].e2_coarse);
+  data_dumper_->DumpRaw("aec3_subtractor_e2_refined",
+                        subtractor_output[0].e2_refined);
+}
+
+AecState::InitialState::InitialState(const EchoCanceller3Config& config)
+    : conservative_initial_phase_(config.filter.conservative_initial_phase),
+      initial_state_seconds_(config.filter.initial_state_seconds) {
+  Reset();
+}
+void AecState::InitialState::InitialState::Reset() {
+  initial_state_ = true;
+  strong_not_saturated_render_blocks_ = 0;
+}
+void AecState::InitialState::InitialState::Update(bool active_render,
+                                                  bool saturated_capture) {
+  strong_not_saturated_render_blocks_ +=
+      active_render && !saturated_capture ? 1 : 0;
+
+  // Flag whether the initial state is still active.
+  bool prev_initial_state = initial_state_;
+  if (conservative_initial_phase_) {
+    initial_state_ =
+        strong_not_saturated_render_blocks_ < 5 * kNumBlocksPerSecond;
+  } else {
+    initial_state_ = strong_not_saturated_render_blocks_ <
+                     initial_state_seconds_ * kNumBlocksPerSecond;
+  }
+
+  // Flag whether the transition from the initial state has started.
+  transition_triggered_ = !initial_state_ && prev_initial_state;
+}
+
+AecState::FilterDelay::FilterDelay(const EchoCanceller3Config& config,
+                                   size_t num_capture_channels)
+    : delay_headroom_blocks_(config.delay.delay_headroom_samples / kBlockSize),
+      filter_delays_blocks_(num_capture_channels, delay_headroom_blocks_),
+      min_filter_delay_(delay_headroom_blocks_) {}
+
+void AecState::FilterDelay::Update(
+    rtc::ArrayView<const int> analyzer_filter_delay_estimates_blocks,
+    const absl::optional<DelayEstimate>& external_delay,
+    size_t blocks_with_proper_filter_adaptation) {
+  // Update the delay based on the external delay.
+  if (external_delay &&
+      (!external_delay_ || external_delay_->delay != external_delay->delay)) {
+    external_delay_ = external_delay;
+    external_delay_reported_ = true;
+  }
+
+  // Override the estimated delay if it is not certain that the filter has had
+  // time to converge.
+  const bool delay_estimator_may_not_have_converged =
+      blocks_with_proper_filter_adaptation < 2 * kNumBlocksPerSecond;
+  if (delay_estimator_may_not_have_converged && external_delay_) {
+    const int delay_guess = delay_headroom_blocks_;
+    std::fill(filter_delays_blocks_.begin(), filter_delays_blocks_.end(),
+              delay_guess);
+  } else {
+    RTC_DCHECK_EQ(filter_delays_blocks_.size(),
+                  analyzer_filter_delay_estimates_blocks.size());
+    std::copy(analyzer_filter_delay_estimates_blocks.begin(),
+              analyzer_filter_delay_estimates_blocks.end(),
+              filter_delays_blocks_.begin());
+  }
+
+  min_filter_delay_ = *std::min_element(filter_delays_blocks_.begin(),
+                                        filter_delays_blocks_.end());
+}
+
+AecState::FilteringQualityAnalyzer::FilteringQualityAnalyzer(
+    const EchoCanceller3Config& config,
+    size_t num_capture_channels)
+    : use_linear_filter_(config.filter.use_linear_filter),
+      usable_linear_filter_estimates_(num_capture_channels, false) {}
+
+void AecState::FilteringQualityAnalyzer::Reset() {
+  std::fill(usable_linear_filter_estimates_.begin(),
+            usable_linear_filter_estimates_.end(), false);
+  overall_usable_linear_estimates_ = false;
+  filter_update_blocks_since_reset_ = 0;
+}
+
+void AecState::FilteringQualityAnalyzer::Update(
+    bool active_render,
+    bool transparent_mode,
+    bool saturated_capture,
+    const absl::optional<DelayEstimate>& external_delay,
+    bool any_filter_converged) {
+  // Update blocks counter.
+  const bool filter_update = active_render && !saturated_capture;
+  filter_update_blocks_since_reset_ += filter_update ? 1 : 0;
+  filter_update_blocks_since_start_ += filter_update ? 1 : 0;
+
+  // Store convergence flag when observed.
+  convergence_seen_ = convergence_seen_ || any_filter_converged;
+
+  // Verify requirements for achieving a decent filter. The requirements for
+  // filter adaptation at call startup are more restrictive than after an
+  // in-call reset.
+  const bool sufficient_data_to_converge_at_startup =
+      filter_update_blocks_since_start_ > kNumBlocksPerSecond * 0.4f;
+  const bool sufficient_data_to_converge_at_reset =
+      sufficient_data_to_converge_at_startup &&
+      filter_update_blocks_since_reset_ > kNumBlocksPerSecond * 0.2f;
+
+  // The linear filter can only be used if it has had time to converge.
+  overall_usable_linear_estimates_ = sufficient_data_to_converge_at_startup &&
+                                     sufficient_data_to_converge_at_reset;
+
+  // The linear filter can only be used if an external delay or convergence have
+  // been identified
+  overall_usable_linear_estimates_ =
+      overall_usable_linear_estimates_ && (external_delay || convergence_seen_);
+
+  // If transparent mode is on, deactivate usign the linear filter.
+  overall_usable_linear_estimates_ =
+      overall_usable_linear_estimates_ && !transparent_mode;
+
+  if (use_linear_filter_) {
+    std::fill(usable_linear_filter_estimates_.begin(),
+              usable_linear_filter_estimates_.end(),
+              overall_usable_linear_estimates_);
+  }
+}
+
+void AecState::SaturationDetector::Update(
+    const Block& x,
+    bool saturated_capture,
+    bool usable_linear_estimate,
+    rtc::ArrayView<const SubtractorOutput> subtractor_output,
+    float echo_path_gain) {
+  saturated_echo_ = false;
+  if (!saturated_capture) {
+    return;
+  }
+
+  if (usable_linear_estimate) {
+    constexpr float kSaturationThreshold = 20000.f;
+    for (size_t ch = 0; ch < subtractor_output.size(); ++ch) {
+      saturated_echo_ =
+          saturated_echo_ ||
+          (subtractor_output[ch].s_refined_max_abs > kSaturationThreshold ||
+           subtractor_output[ch].s_coarse_max_abs > kSaturationThreshold);
+    }
+  } else {
+    float max_sample = 0.f;
+    for (int ch = 0; ch < x.NumChannels(); ++ch) {
+      rtc::ArrayView<const float, kBlockSize> x_ch = x.View(/*band=*/0, ch);
+      for (float sample : x_ch) {
+        max_sample = std::max(max_sample, fabsf(sample));
+      }
+    }
+
+    const float kMargin = 10.f;
+    float peak_echo_amplitude = max_sample * echo_path_gain * kMargin;
+    saturated_echo_ = saturated_echo_ || peak_echo_amplitude > 32000;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec_state.h b/third_party/libwebrtc/modules/audio_processing/aec3/aec_state.h
new file mode 100644
index 0000000000..a39325c8b8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec_state.h
@@ -0,0 +1,300 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/delay_estimate.h"
+#include "modules/audio_processing/aec3/echo_audibility.h"
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+#include "modules/audio_processing/aec3/erl_estimator.h"
+#include "modules/audio_processing/aec3/erle_estimator.h"
+#include "modules/audio_processing/aec3/filter_analyzer.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/reverb_model_estimator.h"
+#include "modules/audio_processing/aec3/subtractor_output.h"
+#include "modules/audio_processing/aec3/subtractor_output_analyzer.h"
+#include "modules/audio_processing/aec3/transparent_mode.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+// Handles the state and the conditions for the echo removal functionality.
+class AecState {
+ public:
+  AecState(const EchoCanceller3Config& config, size_t num_capture_channels);
+  ~AecState();
+
+  // Returns whether the echo subtractor can be used to determine the residual
+  // echo.
+  bool UsableLinearEstimate() const {
+    return filter_quality_state_.LinearFilterUsable() &&
+           config_.filter.use_linear_filter;
+  }
+
+  // Returns whether the echo subtractor output should be used as output.
+  bool UseLinearFilterOutput() const {
+    return filter_quality_state_.LinearFilterUsable() &&
+           config_.filter.use_linear_filter;
+  }
+
+  // Returns whether the render signal is currently active.
+  bool ActiveRender() const { return blocks_with_active_render_ > 200; }
+
+  // Returns the appropriate scaling of the residual echo to match the
+  // audibility.
+  void GetResidualEchoScaling(rtc::ArrayView<float> residual_scaling) const;
+
+  // Returns whether the stationary properties of the signals are used in the
+  // aec.
+  bool UseStationarityProperties() const {
+    return config_.echo_audibility.use_stationarity_properties;
+  }
+
+  // Returns the ERLE.
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Erle(
+      bool onset_compensated) const {
+    return erle_estimator_.Erle(onset_compensated);
+  }
+
+  // Returns the non-capped ERLE.
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleUnbounded()
+      const {
+    return erle_estimator_.ErleUnbounded();
+  }
+
+  // Returns the fullband ERLE estimate in log2 units.
+  float FullBandErleLog2() const { return erle_estimator_.FullbandErleLog2(); }
+
+  // Returns the ERL.
+  const std::array<float, kFftLengthBy2Plus1>& Erl() const {
+    return erl_estimator_.Erl();
+  }
+
+  // Returns the time-domain ERL.
+  float ErlTimeDomain() const { return erl_estimator_.ErlTimeDomain(); }
+
+  // Returns the delay estimate based on the linear filter.
+  int MinDirectPathFilterDelay() const {
+    return delay_state_.MinDirectPathFilterDelay();
+  }
+
+  // Returns whether the capture signal is saturated.
+  bool SaturatedCapture() const { return capture_signal_saturation_; }
+
+  // Returns whether the echo signal is saturated.
+  bool SaturatedEcho() const { return saturation_detector_.SaturatedEcho(); }
+
+  // Updates the capture signal saturation.
+  void UpdateCaptureSaturation(bool capture_signal_saturation) {
+    capture_signal_saturation_ = capture_signal_saturation;
+  }
+
+  // Returns whether the transparent mode is active
+  bool TransparentModeActive() const {
+    return transparent_state_ && transparent_state_->Active();
+  }
+
+  // Takes appropriate action at an echo path change.
+  void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
+
+  // Returns the decay factor for the echo reverberation. The parameter `mild`
+  // indicates which exponential decay to return. The default one or a milder
+  // one that can be used during nearend regions.
+  float ReverbDecay(bool mild) const {
+    return reverb_model_estimator_.ReverbDecay(mild);
+  }
+
+  // Return the frequency response of the reverberant echo.
+  rtc::ArrayView<const float> GetReverbFrequencyResponse() const {
+    return reverb_model_estimator_.GetReverbFrequencyResponse();
+  }
+
+  // Returns whether the transition for going out of the initial stated has
+  // been triggered.
+  bool TransitionTriggered() const {
+    return initial_state_.TransitionTriggered();
+  }
+
+  // Updates the aec state.
+  // TODO(bugs.webrtc.org/10913): Compute multi-channel ERL.
+  void Update(
+      const absl::optional<DelayEstimate>& external_delay,
+      rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+          adaptive_filter_frequency_responses,
+      rtc::ArrayView<const std::vector<float>>
+          adaptive_filter_impulse_responses,
+      const RenderBuffer& render_buffer,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2_refined,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      rtc::ArrayView<const SubtractorOutput> subtractor_output);
+
+  // Returns filter length in blocks.
+  int FilterLengthBlocks() const {
+    // All filters have the same length, so arbitrarily return channel 0 length.
+    return filter_analyzer_.FilterLengthBlocks();
+  }
+
+ private:
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  const EchoCanceller3Config config_;
+  const size_t num_capture_channels_;
+  const bool deactivate_initial_state_reset_at_echo_path_change_;
+  const bool full_reset_at_echo_path_change_;
+  const bool subtractor_analyzer_reset_at_echo_path_change_;
+
+  // Class for controlling the transition from the intial state, which in turn
+  // controls when the filter parameters for the initial state should be used.
+  class InitialState {
+   public:
+    explicit InitialState(const EchoCanceller3Config& config);
+    // Resets the state to again begin in the initial state.
+    void Reset();
+
+    // Updates the state based on new data.
+    void Update(bool active_render, bool saturated_capture);
+
+    // Returns whether the initial state is active or not.
+    bool InitialStateActive() const { return initial_state_; }
+
+    // Returns that the transition from the initial state has was started.
+    bool TransitionTriggered() const { return transition_triggered_; }
+
+   private:
+    const bool conservative_initial_phase_;
+    const float initial_state_seconds_;
+    bool transition_triggered_ = false;
+    bool initial_state_ = true;
+    size_t strong_not_saturated_render_blocks_ = 0;
+  } initial_state_;
+
+  // Class for choosing the direct-path delay relative to the beginning of the
+  // filter, as well as any other data related to the delay used within
+  // AecState.
+  class FilterDelay {
+   public:
+    FilterDelay(const EchoCanceller3Config& config,
+                size_t num_capture_channels);
+
+    // Returns whether an external delay has been reported to the AecState (from
+    // the delay estimator).
+    bool ExternalDelayReported() const { return external_delay_reported_; }
+
+    // Returns the delay in blocks relative to the beginning of the filter that
+    // corresponds to the direct path of the echo.
+    rtc::ArrayView<const int> DirectPathFilterDelays() const {
+      return filter_delays_blocks_;
+    }
+
+    // Returns the minimum delay among the direct path delays relative to the
+    // beginning of the filter
+    int MinDirectPathFilterDelay() const { return min_filter_delay_; }
+
+    // Updates the delay estimates based on new data.
+    void Update(
+        rtc::ArrayView<const int> analyzer_filter_delay_estimates_blocks,
+        const absl::optional<DelayEstimate>& external_delay,
+        size_t blocks_with_proper_filter_adaptation);
+
+   private:
+    const int delay_headroom_blocks_;
+    bool external_delay_reported_ = false;
+    std::vector<int> filter_delays_blocks_;
+    int min_filter_delay_;
+    absl::optional<DelayEstimate> external_delay_;
+  } delay_state_;
+
+  // Classifier for toggling transparent mode when there is no echo.
+  std::unique_ptr<TransparentMode> transparent_state_;
+
+  // Class for analyzing how well the linear filter is, and can be expected to,
+  // perform on the current signals. The purpose of this is for using to
+  // select the echo suppression functionality as well as the input to the echo
+  // suppressor.
+  class FilteringQualityAnalyzer {
+   public:
+    FilteringQualityAnalyzer(const EchoCanceller3Config& config,
+                             size_t num_capture_channels);
+
+    // Returns whether the linear filter can be used for the echo
+    // canceller output.
+    bool LinearFilterUsable() const { return overall_usable_linear_estimates_; }
+
+    // Returns whether an individual filter output can be used for the echo
+    // canceller output.
+    const std::vector<bool>& UsableLinearFilterOutputs() const {
+      return usable_linear_filter_estimates_;
+    }
+
+    // Resets the state of the analyzer.
+    void Reset();
+
+    // Updates the analysis based on new data.
+    void Update(bool active_render,
+                bool transparent_mode,
+                bool saturated_capture,
+                const absl::optional<DelayEstimate>& external_delay,
+                bool any_filter_converged);
+
+   private:
+    const bool use_linear_filter_;
+    bool overall_usable_linear_estimates_ = false;
+    size_t filter_update_blocks_since_reset_ = 0;
+    size_t filter_update_blocks_since_start_ = 0;
+    bool convergence_seen_ = false;
+    std::vector<bool> usable_linear_filter_estimates_;
+  } filter_quality_state_;
+
+  // Class for detecting whether the echo is to be considered to be
+  // saturated.
+  class SaturationDetector {
+   public:
+    // Returns whether the echo is to be considered saturated.
+    bool SaturatedEcho() const { return saturated_echo_; }
+
+    // Updates the detection decision based on new data.
+    void Update(const Block& x,
+                bool saturated_capture,
+                bool usable_linear_estimate,
+                rtc::ArrayView<const SubtractorOutput> subtractor_output,
+                float echo_path_gain);
+
+   private:
+    bool saturated_echo_ = false;
+  } saturation_detector_;
+
+  ErlEstimator erl_estimator_;
+  ErleEstimator erle_estimator_;
+  size_t strong_not_saturated_render_blocks_ = 0;
+  size_t blocks_with_active_render_ = 0;
+  bool capture_signal_saturation_ = false;
+  FilterAnalyzer filter_analyzer_;
+  EchoAudibility echo_audibility_;
+  ReverbModelEstimator reverb_model_estimator_;
+  ReverbModel avg_render_reverb_;
+  SubtractorOutputAnalyzer subtractor_output_analyzer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/aec_state_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/aec_state_unittest.cc
new file mode 100644
index 0000000000..6662c8fb1a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/aec_state_unittest.cc
@@ -0,0 +1,297 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/aec_state.h"
+
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+void RunNormalUsageTest(size_t num_render_channels,
+                        size_t num_capture_channels) {
+  // TODO(bugs.webrtc.org/10913): Test with different content in different
+  // channels.
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  ApmDataDumper data_dumper(42);
+  EchoCanceller3Config config;
+  AecState state(config, num_capture_channels);
+  absl::optional<DelayEstimate> delay_estimate =
+      DelayEstimate(DelayEstimate::Quality::kRefined, 10);
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, num_render_channels));
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2_refined(
+      num_capture_channels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(num_capture_channels);
+  Block x(kNumBands, num_render_channels);
+  EchoPathVariability echo_path_variability(
+      false, EchoPathVariability::DelayAdjustment::kNone, false);
+  std::vector<std::array<float, kBlockSize>> y(num_capture_channels);
+  std::vector<SubtractorOutput> subtractor_output(num_capture_channels);
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    subtractor_output[ch].Reset();
+    subtractor_output[ch].s_refined.fill(100.f);
+    subtractor_output[ch].e_refined.fill(100.f);
+    y[ch].fill(1000.f);
+    E2_refined[ch].fill(0.f);
+    Y2[ch].fill(0.f);
+  }
+  Aec3Fft fft;
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      converged_filter_frequency_response(
+          num_capture_channels,
+          std::vector<std::array<float, kFftLengthBy2Plus1>>(10));
+  for (auto& v_ch : converged_filter_frequency_response) {
+    for (auto& v : v_ch) {
+      v.fill(0.01f);
+    }
+  }
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      diverged_filter_frequency_response = converged_filter_frequency_response;
+  converged_filter_frequency_response[0][2].fill(100.f);
+  converged_filter_frequency_response[0][2][0] = 1.f;
+  std::vector<std::vector<float>> impulse_response(
+      num_capture_channels,
+      std::vector<float>(
+          GetTimeDomainLength(config.filter.refined.length_blocks), 0.f));
+
+  // Verify that linear AEC usability is true when the filter is converged
+  for (size_t band = 0; band < kNumBands; ++band) {
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      std::fill(x.begin(band, ch), x.end(band, ch), 101.f);
+    }
+  }
+  for (int k = 0; k < 3000; ++k) {
+    render_delay_buffer->Insert(x);
+    for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+      subtractor_output[ch].ComputeMetrics(y[ch]);
+    }
+    state.Update(delay_estimate, converged_filter_frequency_response,
+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
+                 E2_refined, Y2, subtractor_output);
+  }
+  EXPECT_TRUE(state.UsableLinearEstimate());
+
+  // Verify that linear AEC usability becomes false after an echo path
+  // change is reported
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    subtractor_output[ch].ComputeMetrics(y[ch]);
+  }
+  state.HandleEchoPathChange(EchoPathVariability(
+      false, EchoPathVariability::DelayAdjustment::kNewDetectedDelay, false));
+  state.Update(delay_estimate, converged_filter_frequency_response,
+               impulse_response, *render_delay_buffer->GetRenderBuffer(),
+               E2_refined, Y2, subtractor_output);
+  EXPECT_FALSE(state.UsableLinearEstimate());
+
+  // Verify that the active render detection works as intended.
+  for (size_t ch = 0; ch < num_render_channels; ++ch) {
+    std::fill(x.begin(0, ch), x.end(0, ch), 101.f);
+  }
+  render_delay_buffer->Insert(x);
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    subtractor_output[ch].ComputeMetrics(y[ch]);
+  }
+  state.HandleEchoPathChange(EchoPathVariability(
+      true, EchoPathVariability::DelayAdjustment::kNewDetectedDelay, false));
+  state.Update(delay_estimate, converged_filter_frequency_response,
+               impulse_response, *render_delay_buffer->GetRenderBuffer(),
+               E2_refined, Y2, subtractor_output);
+  EXPECT_FALSE(state.ActiveRender());
+
+  for (int k = 0; k < 1000; ++k) {
+    render_delay_buffer->Insert(x);
+    for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+      subtractor_output[ch].ComputeMetrics(y[ch]);
+    }
+    state.Update(delay_estimate, converged_filter_frequency_response,
+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
+                 E2_refined, Y2, subtractor_output);
+  }
+  EXPECT_TRUE(state.ActiveRender());
+
+  // Verify that the ERL is properly estimated
+  for (int band = 0; band < x.NumBands(); ++band) {
+    for (int channel = 0; channel < x.NumChannels(); ++channel) {
+      std::fill(x.begin(band, channel), x.end(band, channel), 0.0f);
+    }
+  }
+
+  for (size_t ch = 0; ch < num_render_channels; ++ch) {
+    x.View(/*band=*/0, ch)[0] = 5000.f;
+  }
+  for (size_t k = 0;
+       k < render_delay_buffer->GetRenderBuffer()->GetFftBuffer().size(); ++k) {
+    render_delay_buffer->Insert(x);
+    if (k == 0) {
+      render_delay_buffer->Reset();
+    }
+    render_delay_buffer->PrepareCaptureProcessing();
+  }
+
+  for (auto& Y2_ch : Y2) {
+    Y2_ch.fill(10.f * 10000.f * 10000.f);
+  }
+  for (size_t k = 0; k < 1000; ++k) {
+    for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+      subtractor_output[ch].ComputeMetrics(y[ch]);
+    }
+    state.Update(delay_estimate, converged_filter_frequency_response,
+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
+                 E2_refined, Y2, subtractor_output);
+  }
+
+  ASSERT_TRUE(state.UsableLinearEstimate());
+  const std::array<float, kFftLengthBy2Plus1>& erl = state.Erl();
+  EXPECT_EQ(erl[0], erl[1]);
+  for (size_t k = 1; k < erl.size() - 1; ++k) {
+    EXPECT_NEAR(k % 2 == 0 ? 10.f : 1000.f, erl[k], 0.1);
+  }
+  EXPECT_EQ(erl[erl.size() - 2], erl[erl.size() - 1]);
+
+  // Verify that the ERLE is properly estimated
+  for (auto& E2_refined_ch : E2_refined) {
+    E2_refined_ch.fill(1.f * 10000.f * 10000.f);
+  }
+  for (auto& Y2_ch : Y2) {
+    Y2_ch.fill(10.f * E2_refined[0][0]);
+  }
+  for (size_t k = 0; k < 1000; ++k) {
+    for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+      subtractor_output[ch].ComputeMetrics(y[ch]);
+    }
+    state.Update(delay_estimate, converged_filter_frequency_response,
+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
+                 E2_refined, Y2, subtractor_output);
+  }
+  ASSERT_TRUE(state.UsableLinearEstimate());
+  {
+    // Note that the render spectrum is built so it does not have energy in
+    // the odd bands but just in the even bands.
+    const auto& erle = state.Erle(/*onset_compensated=*/true)[0];
+    EXPECT_EQ(erle[0], erle[1]);
+    constexpr size_t kLowFrequencyLimit = 32;
+    for (size_t k = 2; k < kLowFrequencyLimit; k = k + 2) {
+      EXPECT_NEAR(4.f, erle[k], 0.1);
+    }
+    for (size_t k = kLowFrequencyLimit; k < erle.size() - 1; k = k + 2) {
+      EXPECT_NEAR(1.5f, erle[k], 0.1);
+    }
+    EXPECT_EQ(erle[erle.size() - 2], erle[erle.size() - 1]);
+  }
+  for (auto& E2_refined_ch : E2_refined) {
+    E2_refined_ch.fill(1.f * 10000.f * 10000.f);
+  }
+  for (auto& Y2_ch : Y2) {
+    Y2_ch.fill(5.f * E2_refined[0][0]);
+  }
+  for (size_t k = 0; k < 1000; ++k) {
+    for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+      subtractor_output[ch].ComputeMetrics(y[ch]);
+    }
+    state.Update(delay_estimate, converged_filter_frequency_response,
+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
+                 E2_refined, Y2, subtractor_output);
+  }
+
+  ASSERT_TRUE(state.UsableLinearEstimate());
+  {
+    const auto& erle = state.Erle(/*onset_compensated=*/true)[0];
+    EXPECT_EQ(erle[0], erle[1]);
+    constexpr size_t kLowFrequencyLimit = 32;
+    for (size_t k = 1; k < kLowFrequencyLimit; ++k) {
+      EXPECT_NEAR(k % 2 == 0 ? 4.f : 1.f, erle[k], 0.1);
+    }
+    for (size_t k = kLowFrequencyLimit; k < erle.size() - 1; ++k) {
+      EXPECT_NEAR(k % 2 == 0 ? 1.5f : 1.f, erle[k], 0.1);
+    }
+    EXPECT_EQ(erle[erle.size() - 2], erle[erle.size() - 1]);
+  }
+}
+
+}  // namespace
+
+class AecStateMultiChannel
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         AecStateMultiChannel,
+                         ::testing::Combine(::testing::Values(1, 2, 8),
+                                            ::testing::Values(1, 2, 8)));
+
+// Verify the general functionality of AecState
+TEST_P(AecStateMultiChannel, NormalUsage) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+  RunNormalUsageTest(num_render_channels, num_capture_channels);
+}
+
+// Verifies the delay for a converged filter is correctly identified.
+TEST(AecState, ConvergedFilterDelay) {
+  constexpr int kFilterLengthBlocks = 10;
+  constexpr size_t kNumCaptureChannels = 1;
+  EchoCanceller3Config config;
+  AecState state(config, kNumCaptureChannels);
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, 48000, 1));
+  absl::optional<DelayEstimate> delay_estimate;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2_refined(
+      kNumCaptureChannels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(kNumCaptureChannels);
+  std::array<float, kBlockSize> x;
+  EchoPathVariability echo_path_variability(
+      false, EchoPathVariability::DelayAdjustment::kNone, false);
+  std::vector<SubtractorOutput> subtractor_output(kNumCaptureChannels);
+  for (auto& output : subtractor_output) {
+    output.Reset();
+    output.s_refined.fill(100.f);
+  }
+  std::array<float, kBlockSize> y;
+  x.fill(0.f);
+  y.fill(0.f);
+
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      frequency_response(kNumCaptureChannels,
+                         std::vector<std::array<float, kFftLengthBy2Plus1>>(
+                             kFilterLengthBlocks));
+  for (auto& v_ch : frequency_response) {
+    for (auto& v : v_ch) {
+      v.fill(0.01f);
+    }
+  }
+
+  std::vector<std::vector<float>> impulse_response(
+      kNumCaptureChannels,
+      std::vector<float>(
+          GetTimeDomainLength(config.filter.refined.length_blocks), 0.f));
+
+  // Verify that the filter delay for a converged filter is properly
+  // identified.
+  for (int k = 0; k < kFilterLengthBlocks; ++k) {
+    for (auto& ir : impulse_response) {
+      std::fill(ir.begin(), ir.end(), 0.f);
+      ir[k * kBlockSize + 1] = 1.f;
+    }
+
+    state.HandleEchoPathChange(echo_path_variability);
+    subtractor_output[0].ComputeMetrics(y);
+    state.Update(delay_estimate, frequency_response, impulse_response,
+                 *render_delay_buffer->GetRenderBuffer(), E2_refined, Y2,
+                 subtractor_output);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer.cc
new file mode 100644
index 0000000000..7f076dea8e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer.cc
@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/alignment_mixer.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+AlignmentMixer::MixingVariant ChooseMixingVariant(bool downmix,
+                                                  bool adaptive_selection,
+                                                  int num_channels) {
+  RTC_DCHECK(!(adaptive_selection && downmix));
+  RTC_DCHECK_LT(0, num_channels);
+
+  if (num_channels == 1) {
+    return AlignmentMixer::MixingVariant::kFixed;
+  }
+  if (downmix) {
+    return AlignmentMixer::MixingVariant::kDownmix;
+  }
+  if (adaptive_selection) {
+    return AlignmentMixer::MixingVariant::kAdaptive;
+  }
+  return AlignmentMixer::MixingVariant::kFixed;
+}
+
+}  // namespace
+
+AlignmentMixer::AlignmentMixer(
+    size_t num_channels,
+    const EchoCanceller3Config::Delay::AlignmentMixing& config)
+    : AlignmentMixer(num_channels,
+                     config.downmix,
+                     config.adaptive_selection,
+                     config.activity_power_threshold,
+                     config.prefer_first_two_channels) {}
+
+AlignmentMixer::AlignmentMixer(size_t num_channels,
+                               bool downmix,
+                               bool adaptive_selection,
+                               float activity_power_threshold,
+                               bool prefer_first_two_channels)
+    : num_channels_(num_channels),
+      one_by_num_channels_(1.f / num_channels_),
+      excitation_energy_threshold_(kBlockSize * activity_power_threshold),
+      prefer_first_two_channels_(prefer_first_two_channels),
+      selection_variant_(
+          ChooseMixingVariant(downmix, adaptive_selection, num_channels_)) {
+  if (selection_variant_ == MixingVariant::kAdaptive) {
+    std::fill(strong_block_counters_.begin(), strong_block_counters_.end(), 0);
+    cumulative_energies_.resize(num_channels_);
+    std::fill(cumulative_energies_.begin(), cumulative_energies_.end(), 0.f);
+  }
+}
+
+void AlignmentMixer::ProduceOutput(const Block& x,
+                                   rtc::ArrayView<float, kBlockSize> y) {
+  RTC_DCHECK_EQ(x.NumChannels(), num_channels_);
+
+  if (selection_variant_ == MixingVariant::kDownmix) {
+    Downmix(x, y);
+    return;
+  }
+
+  int ch = selection_variant_ == MixingVariant::kFixed ? 0 : SelectChannel(x);
+
+  RTC_DCHECK_GT(x.NumChannels(), ch);
+  std::copy(x.begin(/*band=*/0, ch), x.end(/*band=*/0, ch), y.begin());
+}
+
+void AlignmentMixer::Downmix(const Block& x,
+                             rtc::ArrayView<float, kBlockSize> y) const {
+  RTC_DCHECK_EQ(x.NumChannels(), num_channels_);
+  RTC_DCHECK_GE(num_channels_, 2);
+  std::memcpy(&y[0], x.View(/*band=*/0, /*channel=*/0).data(),
+              kBlockSize * sizeof(y[0]));
+  for (size_t ch = 1; ch < num_channels_; ++ch) {
+    const auto x_ch = x.View(/*band=*/0, ch);
+    for (size_t i = 0; i < kBlockSize; ++i) {
+      y[i] += x_ch[i];
+    }
+  }
+
+  for (size_t i = 0; i < kBlockSize; ++i) {
+    y[i] *= one_by_num_channels_;
+  }
+}
+
+int AlignmentMixer::SelectChannel(const Block& x) {
+  RTC_DCHECK_EQ(x.NumChannels(), num_channels_);
+  RTC_DCHECK_GE(num_channels_, 2);
+  RTC_DCHECK_EQ(cumulative_energies_.size(), num_channels_);
+
+  constexpr size_t kBlocksToChooseLeftOrRight =
+      static_cast<size_t>(0.5f * kNumBlocksPerSecond);
+  const bool good_signal_in_left_or_right =
+      prefer_first_two_channels_ &&
+      (strong_block_counters_[0] > kBlocksToChooseLeftOrRight ||
+       strong_block_counters_[1] > kBlocksToChooseLeftOrRight);
+
+  const int num_ch_to_analyze =
+      good_signal_in_left_or_right ? 2 : num_channels_;
+
+  constexpr int kNumBlocksBeforeEnergySmoothing = 60 * kNumBlocksPerSecond;
+  ++block_counter_;
+
+  for (int ch = 0; ch < num_ch_to_analyze; ++ch) {
+    float x2_sum = 0.f;
+    rtc::ArrayView<const float, kBlockSize> x_ch = x.View(/*band=*/0, ch);
+    for (size_t i = 0; i < kBlockSize; ++i) {
+      x2_sum += x_ch[i] * x_ch[i];
+    }
+
+    if (ch < 2 && x2_sum > excitation_energy_threshold_) {
+      ++strong_block_counters_[ch];
+    }
+
+    if (block_counter_ <= kNumBlocksBeforeEnergySmoothing) {
+      cumulative_energies_[ch] += x2_sum;
+    } else {
+      constexpr float kSmoothing = 1.f / (10 * kNumBlocksPerSecond);
+      cumulative_energies_[ch] +=
+          kSmoothing * (x2_sum - cumulative_energies_[ch]);
+    }
+  }
+
+  // Normalize the energies to allow the energy computations to from now be
+  // based on smoothing.
+  if (block_counter_ == kNumBlocksBeforeEnergySmoothing) {
+    constexpr float kOneByNumBlocksBeforeEnergySmoothing =
+        1.f / kNumBlocksBeforeEnergySmoothing;
+    for (int ch = 0; ch < num_ch_to_analyze; ++ch) {
+      cumulative_energies_[ch] *= kOneByNumBlocksBeforeEnergySmoothing;
+    }
+  }
+
+  int strongest_ch = 0;
+  for (int ch = 0; ch < num_ch_to_analyze; ++ch) {
+    if (cumulative_energies_[ch] > cumulative_energies_[strongest_ch]) {
+      strongest_ch = ch;
+    }
+  }
+
+  if ((good_signal_in_left_or_right && selected_channel_ > 1) ||
+      cumulative_energies_[strongest_ch] >
+          2.f * cumulative_energies_[selected_channel_]) {
+    selected_channel_ = strongest_ch;
+  }
+
+  return selected_channel_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer.h b/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer.h
new file mode 100644
index 0000000000..b3ed04755c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ALIGNMENT_MIXER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ALIGNMENT_MIXER_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/block.h"
+
+namespace webrtc {
+
+// Performs channel conversion to mono for the purpose of providing a decent
+// mono input for the delay estimation. This is achieved by analyzing all
+// incoming channels and produce one single channel output.
+class AlignmentMixer {
+ public:
+  AlignmentMixer(size_t num_channels,
+                 const EchoCanceller3Config::Delay::AlignmentMixing& config);
+
+  AlignmentMixer(size_t num_channels,
+                 bool downmix,
+                 bool adaptive_selection,
+                 float excitation_limit,
+                 bool prefer_first_two_channels);
+
+  void ProduceOutput(const Block& x, rtc::ArrayView<float, kBlockSize> y);
+
+  enum class MixingVariant { kDownmix, kAdaptive, kFixed };
+
+ private:
+  const size_t num_channels_;
+  const float one_by_num_channels_;
+  const float excitation_energy_threshold_;
+  const bool prefer_first_two_channels_;
+  const MixingVariant selection_variant_;
+  std::array<size_t, 2> strong_block_counters_;
+  std::vector<float> cumulative_energies_;
+  int selected_channel_ = 0;
+  size_t block_counter_ = 0;
+
+  void Downmix(const Block& x, rtc::ArrayView<float, kBlockSize> y) const;
+  int SelectChannel(const Block& x);
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ALIGNMENT_MIXER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer_unittest.cc
new file mode 100644
index 0000000000..eaf6dcb235
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer_unittest.cc
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/alignment_mixer.h"
+
+#include <string>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::AllOf;
+using ::testing::Each;
+
+namespace webrtc {
+namespace {
+std::string ProduceDebugText(bool initial_silence,
+                             bool huge_activity_threshold,
+                             bool prefer_first_two_channels,
+                             int num_channels,
+                             int strongest_ch) {
+  rtc::StringBuilder ss;
+  ss << ", Initial silence: " << initial_silence;
+  ss << ", Huge activity threshold: " << huge_activity_threshold;
+  ss << ", Prefer first two channels: " << prefer_first_two_channels;
+  ss << ", Number of channels: " << num_channels;
+  ss << ", Strongest channel: " << strongest_ch;
+  return ss.Release();
+}
+
+}  // namespace
+
+TEST(AlignmentMixer, GeneralAdaptiveMode) {
+  constexpr int kChannelOffset = 100;
+  constexpr int kMaxChannelsToTest = 8;
+  constexpr float kStrongestSignalScaling =
+      kMaxChannelsToTest * kChannelOffset * 100;
+
+  for (bool initial_silence : {false, true}) {
+    for (bool huge_activity_threshold : {false, true}) {
+      for (bool prefer_first_two_channels : {false, true}) {
+        for (int num_channels = 2; num_channels < 8; ++num_channels) {
+          for (int strongest_ch = 0; strongest_ch < num_channels;
+               ++strongest_ch) {
+            SCOPED_TRACE(ProduceDebugText(
+                initial_silence, huge_activity_threshold,
+                prefer_first_two_channels, num_channels, strongest_ch));
+            const float excitation_limit =
+                huge_activity_threshold ? 1000000000.f : 0.001f;
+            AlignmentMixer am(num_channels, /*downmix*/ false,
+                              /*adaptive_selection*/ true, excitation_limit,
+                              prefer_first_two_channels);
+
+            Block x(
+                /*num_bands=*/1, num_channels);
+            if (initial_silence) {
+              std::array<float, kBlockSize> y;
+              for (int frame = 0; frame < 10 * kNumBlocksPerSecond; ++frame) {
+                am.ProduceOutput(x, y);
+              }
+            }
+
+            for (int frame = 0; frame < 2 * kNumBlocksPerSecond; ++frame) {
+              const auto channel_value = [&](int frame_index,
+                                             int channel_index) {
+                return static_cast<float>(frame_index +
+                                          channel_index * kChannelOffset);
+              };
+
+              for (int ch = 0; ch < num_channels; ++ch) {
+                float scaling =
+                    ch == strongest_ch ? kStrongestSignalScaling : 1.f;
+                auto x_ch = x.View(/*band=*/0, ch);
+                std::fill(x_ch.begin(), x_ch.end(),
+                          channel_value(frame, ch) * scaling);
+              }
+
+              std::array<float, kBlockSize> y;
+              y.fill(-1.f);
+              am.ProduceOutput(x, y);
+
+              if (frame > 1 * kNumBlocksPerSecond) {
+                if (!prefer_first_two_channels || huge_activity_threshold) {
+                  EXPECT_THAT(y,
+                              AllOf(Each(x.View(/*band=*/0, strongest_ch)[0])));
+                } else {
+                  bool left_or_right_chosen;
+                  for (int ch = 0; ch < 2; ++ch) {
+                    left_or_right_chosen = true;
+                    const auto x_ch = x.View(/*band=*/0, ch);
+                    for (size_t k = 0; k < kBlockSize; ++k) {
+                      if (y[k] != x_ch[k]) {
+                        left_or_right_chosen = false;
+                        break;
+                      }
+                    }
+                    if (left_or_right_chosen) {
+                      break;
+                    }
+                  }
+                  EXPECT_TRUE(left_or_right_chosen);
+                }
+              }
+            }
+          }
+        }
+      }
+    }
+  }
+}
+
+TEST(AlignmentMixer, DownmixMode) {
+  for (int num_channels = 1; num_channels < 8; ++num_channels) {
+    AlignmentMixer am(num_channels, /*downmix*/ true,
+                      /*adaptive_selection*/ false, /*excitation_limit*/ 1.f,
+                      /*prefer_first_two_channels*/ false);
+
+    Block x(/*num_bands=*/1, num_channels);
+    const auto channel_value = [](int frame_index, int channel_index) {
+      return static_cast<float>(frame_index + channel_index);
+    };
+    for (int frame = 0; frame < 10; ++frame) {
+      for (int ch = 0; ch < num_channels; ++ch) {
+        auto x_ch = x.View(/*band=*/0, ch);
+        std::fill(x_ch.begin(), x_ch.end(), channel_value(frame, ch));
+      }
+
+      std::array<float, kBlockSize> y;
+      y.fill(-1.f);
+      am.ProduceOutput(x, y);
+
+      float expected_mixed_value = 0.f;
+      for (int ch = 0; ch < num_channels; ++ch) {
+        expected_mixed_value += channel_value(frame, ch);
+      }
+      expected_mixed_value *= 1.f / num_channels;
+
+      EXPECT_THAT(y, AllOf(Each(expected_mixed_value)));
+    }
+  }
+}
+
+TEST(AlignmentMixer, FixedMode) {
+  for (int num_channels = 1; num_channels < 8; ++num_channels) {
+    AlignmentMixer am(num_channels, /*downmix*/ false,
+                      /*adaptive_selection*/ false, /*excitation_limit*/ 1.f,
+                      /*prefer_first_two_channels*/ false);
+
+    Block x(/*num_band=*/1, num_channels);
+    const auto channel_value = [](int frame_index, int channel_index) {
+      return static_cast<float>(frame_index + channel_index);
+    };
+    for (int frame = 0; frame < 10; ++frame) {
+      for (int ch = 0; ch < num_channels; ++ch) {
+        auto x_ch = x.View(/*band=*/0, ch);
+        std::fill(x_ch.begin(), x_ch.end(), channel_value(frame, ch));
+      }
+
+      std::array<float, kBlockSize> y;
+      y.fill(-1.f);
+      am.ProduceOutput(x, y);
+      EXPECT_THAT(y, AllOf(Each(x.View(/*band=*/0, /*channel=*/0)[0])));
+    }
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+TEST(AlignmentMixerDeathTest, ZeroNumChannels) {
+  EXPECT_DEATH(
+      AlignmentMixer(/*num_channels*/ 0, /*downmix*/ false,
+                     /*adaptive_selection*/ false, /*excitation_limit*/ 1.f,
+                     /*prefer_first_two_channels*/ false);
+      , "");
+}
+
+TEST(AlignmentMixerDeathTest, IncorrectVariant) {
+  EXPECT_DEATH(
+      AlignmentMixer(/*num_channels*/ 1, /*downmix*/ true,
+                     /*adaptive_selection*/ true, /*excitation_limit*/ 1.f,
+                     /*prefer_first_two_channels*/ false);
+      , "");
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.cc b/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.cc
new file mode 100644
index 0000000000..45f56a5dce
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.cc
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+
+bool TimeToReportMetrics(int frames_since_last_report) {
+  constexpr int kNumFramesPerSecond = 100;
+  constexpr int kReportingIntervalFrames = 10 * kNumFramesPerSecond;
+  return frames_since_last_report == kReportingIntervalFrames;
+}
+
+}  // namespace
+
+ApiCallJitterMetrics::Jitter::Jitter()
+    : max_(0), min_(std::numeric_limits<int>::max()) {}
+
+void ApiCallJitterMetrics::Jitter::Update(int num_api_calls_in_a_row) {
+  min_ = std::min(min_, num_api_calls_in_a_row);
+  max_ = std::max(max_, num_api_calls_in_a_row);
+}
+
+void ApiCallJitterMetrics::Jitter::Reset() {
+  min_ = std::numeric_limits<int>::max();
+  max_ = 0;
+}
+
+void ApiCallJitterMetrics::Reset() {
+  render_jitter_.Reset();
+  capture_jitter_.Reset();
+  num_api_calls_in_a_row_ = 0;
+  frames_since_last_report_ = 0;
+  last_call_was_render_ = false;
+  proper_call_observed_ = false;
+}
+
+void ApiCallJitterMetrics::ReportRenderCall() {
+  if (!last_call_was_render_) {
+    // If the previous call was a capture and a proper call has been observed
+    // (containing both render and capture data), storing the last number of
+    // capture calls into the metrics.
+    if (proper_call_observed_) {
+      capture_jitter_.Update(num_api_calls_in_a_row_);
+    }
+
+    // Reset the call counter to start counting render calls.
+    num_api_calls_in_a_row_ = 0;
+  }
+  ++num_api_calls_in_a_row_;
+  last_call_was_render_ = true;
+}
+
+void ApiCallJitterMetrics::ReportCaptureCall() {
+  if (last_call_was_render_) {
+    // If the previous call was a render and a proper call has been observed
+    // (containing both render and capture data), storing the last number of
+    // render calls into the metrics.
+    if (proper_call_observed_) {
+      render_jitter_.Update(num_api_calls_in_a_row_);
+    }
+    // Reset the call counter to start counting capture calls.
+    num_api_calls_in_a_row_ = 0;
+
+    // If this statement is reached, at least one render and one capture call
+    // have been observed.
+    proper_call_observed_ = true;
+  }
+  ++num_api_calls_in_a_row_;
+  last_call_was_render_ = false;
+
+  // Only report and update jitter metrics for when a proper call, containing
+  // both render and capture data, has been observed.
+  if (proper_call_observed_ &&
+      TimeToReportMetrics(++frames_since_last_report_)) {
+    // Report jitter, where the base basic unit is frames.
+    constexpr int kMaxJitterToReport = 50;
+
+    // Report max and min jitter for render and capture, in units of 20 ms.
+    RTC_HISTOGRAM_COUNTS_LINEAR(
+        "WebRTC.Audio.EchoCanceller.MaxRenderJitter",
+        std::min(kMaxJitterToReport, render_jitter().max()), 1,
+        kMaxJitterToReport, kMaxJitterToReport);
+    RTC_HISTOGRAM_COUNTS_LINEAR(
+        "WebRTC.Audio.EchoCanceller.MinRenderJitter",
+        std::min(kMaxJitterToReport, render_jitter().min()), 1,
+        kMaxJitterToReport, kMaxJitterToReport);
+
+    RTC_HISTOGRAM_COUNTS_LINEAR(
+        "WebRTC.Audio.EchoCanceller.MaxCaptureJitter",
+        std::min(kMaxJitterToReport, capture_jitter().max()), 1,
+        kMaxJitterToReport, kMaxJitterToReport);
+    RTC_HISTOGRAM_COUNTS_LINEAR(
+        "WebRTC.Audio.EchoCanceller.MinCaptureJitter",
+        std::min(kMaxJitterToReport, capture_jitter().min()), 1,
+        kMaxJitterToReport, kMaxJitterToReport);
+
+    frames_since_last_report_ = 0;
+    Reset();
+  }
+}
+
+bool ApiCallJitterMetrics::WillReportMetricsAtNextCapture() const {
+  return TimeToReportMetrics(frames_since_last_report_ + 1);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h b/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h
new file mode 100644
index 0000000000..dd1fa82e93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_
+
+namespace webrtc {
+
+// Stores data for reporting metrics on the API call jitter.
+class ApiCallJitterMetrics {
+ public:
+  class Jitter {
+   public:
+    Jitter();
+    void Update(int num_api_calls_in_a_row);
+    void Reset();
+
+    int min() const { return min_; }
+    int max() const { return max_; }
+
+   private:
+    int max_;
+    int min_;
+  };
+
+  ApiCallJitterMetrics() { Reset(); }
+
+  // Update metrics for render API call.
+  void ReportRenderCall();
+
+  // Update and periodically report metrics for capture API call.
+  void ReportCaptureCall();
+
+  // Methods used only for testing.
+  const Jitter& render_jitter() const { return render_jitter_; }
+  const Jitter& capture_jitter() const { return capture_jitter_; }
+  bool WillReportMetricsAtNextCapture() const;
+
+ private:
+  void Reset();
+
+  Jitter render_jitter_;
+  Jitter capture_jitter_;
+
+  int num_api_calls_in_a_row_ = 0;
+  int frames_since_last_report_ = 0;
+  bool last_call_was_render_ = false;
+  bool proper_call_observed_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc
new file mode 100644
index 0000000000..b902487152
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Verify constant jitter.
+TEST(ApiCallJitterMetrics, ConstantJitter) {
+  for (int jitter = 1; jitter < 20; ++jitter) {
+    ApiCallJitterMetrics metrics;
+    for (size_t k = 0; k < 30 * kNumBlocksPerSecond; ++k) {
+      for (int j = 0; j < jitter; ++j) {
+        metrics.ReportRenderCall();
+      }
+
+      for (int j = 0; j < jitter; ++j) {
+        metrics.ReportCaptureCall();
+
+        if (metrics.WillReportMetricsAtNextCapture()) {
+          EXPECT_EQ(jitter, metrics.render_jitter().min());
+          EXPECT_EQ(jitter, metrics.render_jitter().max());
+          EXPECT_EQ(jitter, metrics.capture_jitter().min());
+          EXPECT_EQ(jitter, metrics.capture_jitter().max());
+        }
+      }
+    }
+  }
+}
+
+// Verify peaky jitter for the render.
+TEST(ApiCallJitterMetrics, JitterPeakRender) {
+  constexpr int kMinJitter = 2;
+  constexpr int kJitterPeak = 10;
+  constexpr int kPeakInterval = 100;
+
+  ApiCallJitterMetrics metrics;
+  int render_surplus = 0;
+
+  for (size_t k = 0; k < 30 * kNumBlocksPerSecond; ++k) {
+    const int num_render_calls =
+        k % kPeakInterval == 0 ? kJitterPeak : kMinJitter;
+    for (int j = 0; j < num_render_calls; ++j) {
+      metrics.ReportRenderCall();
+      ++render_surplus;
+    }
+
+    ASSERT_LE(kMinJitter, render_surplus);
+    const int num_capture_calls =
+        render_surplus == kMinJitter ? kMinJitter : kMinJitter + 1;
+    for (int j = 0; j < num_capture_calls; ++j) {
+      metrics.ReportCaptureCall();
+
+      if (metrics.WillReportMetricsAtNextCapture()) {
+        EXPECT_EQ(kMinJitter, metrics.render_jitter().min());
+        EXPECT_EQ(kJitterPeak, metrics.render_jitter().max());
+        EXPECT_EQ(kMinJitter, metrics.capture_jitter().min());
+        EXPECT_EQ(kMinJitter + 1, metrics.capture_jitter().max());
+      }
+      --render_surplus;
+    }
+  }
+}
+
+// Verify peaky jitter for the capture.
+TEST(ApiCallJitterMetrics, JitterPeakCapture) {
+  constexpr int kMinJitter = 2;
+  constexpr int kJitterPeak = 10;
+  constexpr int kPeakInterval = 100;
+
+  ApiCallJitterMetrics metrics;
+  int capture_surplus = kMinJitter;
+
+  for (size_t k = 0; k < 30 * kNumBlocksPerSecond; ++k) {
+    ASSERT_LE(kMinJitter, capture_surplus);
+    const int num_render_calls =
+        capture_surplus == kMinJitter ? kMinJitter : kMinJitter + 1;
+    for (int j = 0; j < num_render_calls; ++j) {
+      metrics.ReportRenderCall();
+      --capture_surplus;
+    }
+
+    const int num_capture_calls =
+        k % kPeakInterval == 0 ? kJitterPeak : kMinJitter;
+    for (int j = 0; j < num_capture_calls; ++j) {
+      metrics.ReportCaptureCall();
+
+      if (metrics.WillReportMetricsAtNextCapture()) {
+        EXPECT_EQ(kMinJitter, metrics.render_jitter().min());
+        EXPECT_EQ(kMinJitter + 1, metrics.render_jitter().max());
+        EXPECT_EQ(kMinJitter, metrics.capture_jitter().min());
+        EXPECT_EQ(kJitterPeak, metrics.capture_jitter().max());
+      }
+      ++capture_surplus;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block.h b/third_party/libwebrtc/modules/audio_processing/aec3/block.h
new file mode 100644
index 0000000000..c1fc70722d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_H_
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+
+// Contains one or more channels of 4 milliseconds of audio data.
+// The audio is split in one or more frequency bands, each with a sampling
+// rate of 16 kHz.
+class Block {
+ public:
+  Block(int num_bands, int num_channels, float default_value = 0.0f)
+      : num_bands_(num_bands),
+        num_channels_(num_channels),
+        data_(num_bands * num_channels * kBlockSize, default_value) {}
+
+  // Returns the number of bands.
+  int NumBands() const { return num_bands_; }
+
+  // Returns the number of channels.
+  int NumChannels() const { return num_channels_; }
+
+  // Modifies the number of channels and sets all samples to zero.
+  void SetNumChannels(int num_channels) {
+    num_channels_ = num_channels;
+    data_.resize(num_bands_ * num_channels_ * kBlockSize);
+    std::fill(data_.begin(), data_.end(), 0.0f);
+  }
+
+  // Iterators for accessing the data.
+  auto begin(int band, int channel) {
+    return data_.begin() + GetIndex(band, channel);
+  }
+
+  auto begin(int band, int channel) const {
+    return data_.begin() + GetIndex(band, channel);
+  }
+
+  auto end(int band, int channel) { return begin(band, channel) + kBlockSize; }
+
+  auto end(int band, int channel) const {
+    return begin(band, channel) + kBlockSize;
+  }
+
+  // Access data via ArrayView.
+  rtc::ArrayView<float, kBlockSize> View(int band, int channel) {
+    return rtc::ArrayView<float, kBlockSize>(&data_[GetIndex(band, channel)],
+                                             kBlockSize);
+  }
+
+  rtc::ArrayView<const float, kBlockSize> View(int band, int channel) const {
+    return rtc::ArrayView<const float, kBlockSize>(
+        &data_[GetIndex(band, channel)], kBlockSize);
+  }
+
+  // Lets two Blocks swap audio data.
+  void Swap(Block& b) {
+    std::swap(num_bands_, b.num_bands_);
+    std::swap(num_channels_, b.num_channels_);
+    data_.swap(b.data_);
+  }
+
+ private:
+  // Returns the index of the first sample of the requested |band| and
+  // |channel|.
+  int GetIndex(int band, int channel) const {
+    return (band * num_channels_ + channel) * kBlockSize;
+  }
+
+  int num_bands_;
+  int num_channels_;
+  std::vector<float> data_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_buffer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_buffer.cc
new file mode 100644
index 0000000000..289c3f0d10
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_buffer.cc
@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/block_buffer.h"
+
+#include <algorithm>
+
+namespace webrtc {
+
+BlockBuffer::BlockBuffer(size_t size, size_t num_bands, size_t num_channels)
+    : size(static_cast<int>(size)),
+      buffer(size, Block(num_bands, num_channels)) {}
+
+BlockBuffer::~BlockBuffer() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/block_buffer.h
new file mode 100644
index 0000000000..3489d51646
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_buffer.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "modules/audio_processing/aec3/block.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Struct for bundling a circular buffer of two dimensional vector objects
+// together with the read and write indices.
+struct BlockBuffer {
+  BlockBuffer(size_t size, size_t num_bands, size_t num_channels);
+  ~BlockBuffer();
+
+  int IncIndex(int index) const {
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return index < size - 1 ? index + 1 : 0;
+  }
+
+  int DecIndex(int index) const {
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return index > 0 ? index - 1 : size - 1;
+  }
+
+  int OffsetIndex(int index, int offset) const {
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    RTC_DCHECK_GE(size, offset);
+    return (size + index + offset) % size;
+  }
+
+  void UpdateWriteIndex(int offset) { write = OffsetIndex(write, offset); }
+  void IncWriteIndex() { write = IncIndex(write); }
+  void DecWriteIndex() { write = DecIndex(write); }
+  void UpdateReadIndex(int offset) { read = OffsetIndex(read, offset); }
+  void IncReadIndex() { read = IncIndex(read); }
+  void DecReadIndex() { read = DecIndex(read); }
+
+  const int size;
+  std::vector<Block> buffer;
+  int write = 0;
+  int read = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer.cc
new file mode 100644
index 0000000000..059bbafcdb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer.cc
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/block_delay_buffer.h"
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+BlockDelayBuffer::BlockDelayBuffer(size_t num_channels,
+                                   size_t num_bands,
+                                   size_t frame_length,
+                                   size_t delay_samples)
+    : frame_length_(frame_length),
+      delay_(delay_samples),
+      buf_(num_channels,
+           std::vector<std::vector<float>>(num_bands,
+                                           std::vector<float>(delay_, 0.f))) {}
+
+BlockDelayBuffer::~BlockDelayBuffer() = default;
+
+void BlockDelayBuffer::DelaySignal(AudioBuffer* frame) {
+  RTC_DCHECK_EQ(buf_.size(), frame->num_channels());
+  if (delay_ == 0) {
+    return;
+  }
+
+  const size_t num_bands = buf_[0].size();
+  const size_t num_channels = buf_.size();
+
+  const size_t i_start = last_insert_;
+  size_t i = 0;
+  for (size_t ch = 0; ch < num_channels; ++ch) {
+    RTC_DCHECK_EQ(buf_[ch].size(), frame->num_bands());
+    RTC_DCHECK_EQ(buf_[ch].size(), num_bands);
+    rtc::ArrayView<float* const> frame_ch(frame->split_bands(ch), num_bands);
+    const size_t delay = delay_;
+
+    for (size_t band = 0; band < num_bands; ++band) {
+      RTC_DCHECK_EQ(delay_, buf_[ch][band].size());
+      i = i_start;
+
+      // Offloading these pointers and class variables to local variables allows
+      // the compiler to optimize the below loop when compiling with
+      // '-fno-strict-aliasing'.
+      float* buf_ch_band = buf_[ch][band].data();
+      float* frame_ch_band = frame_ch[band];
+
+      for (size_t k = 0, frame_length = frame_length_; k < frame_length; ++k) {
+        const float tmp = buf_ch_band[i];
+        buf_ch_band[i] = frame_ch_band[k];
+        frame_ch_band[k] = tmp;
+
+        i = i < delay - 1 ? i + 1 : 0;
+      }
+    }
+  }
+
+  last_insert_ = i;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer.h
new file mode 100644
index 0000000000..711a790bfe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "modules/audio_processing/audio_buffer.h"
+
+namespace webrtc {
+
+// Class for applying a fixed delay to the samples in a signal partitioned using
+// the audiobuffer band-splitting scheme.
+class BlockDelayBuffer {
+ public:
+  BlockDelayBuffer(size_t num_channels,
+                   size_t num_bands,
+                   size_t frame_length,
+                   size_t delay_samples);
+  ~BlockDelayBuffer();
+
+  // Delays the samples by the specified delay.
+  void DelaySignal(AudioBuffer* frame);
+
+ private:
+  const size_t frame_length_;
+  const size_t delay_;
+  std::vector<std::vector<std::vector<float>>> buf_;
+  size_t last_insert_ = 0;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc
new file mode 100644
index 0000000000..011ab49651
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_delay_buffer_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/block_delay_buffer.h"
+
+#include <string>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+float SampleValue(size_t sample_index) {
+  return sample_index % 32768;
+}
+
+// Populates the frame with linearly increasing sample values for each band.
+void PopulateInputFrame(size_t frame_length,
+                        size_t num_bands,
+                        size_t first_sample_index,
+                        float* const* frame) {
+  for (size_t k = 0; k < num_bands; ++k) {
+    for (size_t i = 0; i < frame_length; ++i) {
+      frame[k][i] = SampleValue(first_sample_index + i);
+    }
+  }
+}
+
+std::string ProduceDebugText(int sample_rate_hz, size_t delay) {
+  char log_stream_buffer[8 * 1024];
+  rtc::SimpleStringBuilder ss(log_stream_buffer);
+  ss << "Sample rate: " << sample_rate_hz;
+  ss << ", Delay: " << delay;
+  return ss.str();
+}
+
+}  // namespace
+
+class BlockDelayBufferTest
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, int, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(
+    ParameterCombinations,
+    BlockDelayBufferTest,
+    ::testing::Combine(::testing::Values(0, 1, 27, 160, 4321, 7021),
+                       ::testing::Values(16000, 32000, 48000),
+                       ::testing::Values(1, 2, 4)));
+
+// Verifies that the correct signal delay is achived.
+TEST_P(BlockDelayBufferTest, CorrectDelayApplied) {
+  const size_t delay = std::get<0>(GetParam());
+  const int rate = std::get<1>(GetParam());
+  const size_t num_channels = std::get<2>(GetParam());
+
+  SCOPED_TRACE(ProduceDebugText(rate, delay));
+  size_t num_bands = NumBandsForRate(rate);
+  size_t subband_frame_length = 160;
+
+  BlockDelayBuffer delay_buffer(num_channels, num_bands, subband_frame_length,
+                                delay);
+
+  static constexpr size_t kNumFramesToProcess = 20;
+  for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
+       ++frame_index) {
+    AudioBuffer audio_buffer(rate, num_channels, rate, num_channels, rate,
+                             num_channels);
+    if (rate > 16000) {
+      audio_buffer.SplitIntoFrequencyBands();
+    }
+    size_t first_sample_index = frame_index * subband_frame_length;
+    for (size_t ch = 0; ch < num_channels; ++ch) {
+      PopulateInputFrame(subband_frame_length, num_bands, first_sample_index,
+                         &audio_buffer.split_bands(ch)[0]);
+    }
+    delay_buffer.DelaySignal(&audio_buffer);
+
+    for (size_t ch = 0; ch < num_channels; ++ch) {
+      for (size_t band = 0; band < num_bands; ++band) {
+        size_t sample_index = first_sample_index;
+        for (size_t i = 0; i < subband_frame_length; ++i, ++sample_index) {
+          if (sample_index < delay) {
+            EXPECT_EQ(0.f, audio_buffer.split_bands(ch)[band][i]);
+          } else {
+            EXPECT_EQ(SampleValue(sample_index - delay),
+                      audio_buffer.split_bands(ch)[band][i]);
+          }
+        }
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_framer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_framer.cc
new file mode 100644
index 0000000000..4243ddeba0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_framer.cc
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/block_framer.h"
+
+#include <algorithm>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+BlockFramer::BlockFramer(size_t num_bands, size_t num_channels)
+    : num_bands_(num_bands),
+      num_channels_(num_channels),
+      buffer_(num_bands_,
+              std::vector<std::vector<float>>(
+                  num_channels,
+                  std::vector<float>(kBlockSize, 0.f))) {
+  RTC_DCHECK_LT(0, num_bands);
+  RTC_DCHECK_LT(0, num_channels);
+}
+
+BlockFramer::~BlockFramer() = default;
+
+// All the constants are chosen so that the buffer is either empty or has enough
+// samples for InsertBlockAndExtractSubFrame to produce a frame. In order to
+// achieve this, the InsertBlockAndExtractSubFrame and InsertBlock methods need
+// to be called in the correct order.
+void BlockFramer::InsertBlock(const Block& block) {
+  RTC_DCHECK_EQ(num_bands_, block.NumBands());
+  RTC_DCHECK_EQ(num_channels_, block.NumChannels());
+  for (size_t band = 0; band < num_bands_; ++band) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      RTC_DCHECK_EQ(0, buffer_[band][channel].size());
+
+      buffer_[band][channel].insert(buffer_[band][channel].begin(),
+                                    block.begin(band, channel),
+                                    block.end(band, channel));
+    }
+  }
+}
+
+void BlockFramer::InsertBlockAndExtractSubFrame(
+    const Block& block,
+    std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame) {
+  RTC_DCHECK(sub_frame);
+  RTC_DCHECK_EQ(num_bands_, block.NumBands());
+  RTC_DCHECK_EQ(num_channels_, block.NumChannels());
+  RTC_DCHECK_EQ(num_bands_, sub_frame->size());
+  for (size_t band = 0; band < num_bands_; ++band) {
+    RTC_DCHECK_EQ(num_channels_, (*sub_frame)[0].size());
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      RTC_DCHECK_LE(kSubFrameLength,
+                    buffer_[band][channel].size() + kBlockSize);
+      RTC_DCHECK_GE(kBlockSize, buffer_[band][channel].size());
+      RTC_DCHECK_EQ(kSubFrameLength, (*sub_frame)[band][channel].size());
+
+      const int samples_to_frame =
+          kSubFrameLength - buffer_[band][channel].size();
+      std::copy(buffer_[band][channel].begin(), buffer_[band][channel].end(),
+                (*sub_frame)[band][channel].begin());
+      std::copy(
+          block.begin(band, channel),
+          block.begin(band, channel) + samples_to_frame,
+          (*sub_frame)[band][channel].begin() + buffer_[band][channel].size());
+      buffer_[band][channel].clear();
+      buffer_[band][channel].insert(
+          buffer_[band][channel].begin(),
+          block.begin(band, channel) + samples_to_frame,
+          block.end(band, channel));
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_framer.h b/third_party/libwebrtc/modules/audio_processing/aec3/block_framer.h
new file mode 100644
index 0000000000..e2cdd5a17c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_framer.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/block.h"
+
+namespace webrtc {
+
+// Class for producing frames consisting of 2 subframes of 80 samples each
+// from 64 sample blocks. The class is designed to work together with the
+// FrameBlocker class which performs the reverse conversion. Used together with
+// that, this class produces output frames are the same rate as frames are
+// received by the FrameBlocker class. Note that the internal buffers will
+// overrun if any other rate of packets insertion is used.
+class BlockFramer {
+ public:
+  BlockFramer(size_t num_bands, size_t num_channels);
+  ~BlockFramer();
+  BlockFramer(const BlockFramer&) = delete;
+  BlockFramer& operator=(const BlockFramer&) = delete;
+
+  // Adds a 64 sample block into the data that will form the next output frame.
+  void InsertBlock(const Block& block);
+  // Adds a 64 sample block and extracts an 80 sample subframe.
+  void InsertBlockAndExtractSubFrame(
+      const Block& block,
+      std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame);
+
+ private:
+  const size_t num_bands_;
+  const size_t num_channels_;
+  std::vector<std::vector<std::vector<float>>> buffer_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_FRAMER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_framer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_framer_unittest.cc
new file mode 100644
index 0000000000..9439623f72
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_framer_unittest.cc
@@ -0,0 +1,337 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/block_framer.h"
+
+#include <string>
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+void SetupSubFrameView(
+    std::vector<std::vector<std::vector<float>>>* sub_frame,
+    std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
+  for (size_t band = 0; band < sub_frame_view->size(); ++band) {
+    for (size_t channel = 0; channel < (*sub_frame_view)[band].size();
+         ++channel) {
+      (*sub_frame_view)[band][channel] =
+          rtc::ArrayView<float>((*sub_frame)[band][channel].data(),
+                                (*sub_frame)[band][channel].size());
+    }
+  }
+}
+
+float ComputeSampleValue(size_t chunk_counter,
+                         size_t chunk_size,
+                         size_t band,
+                         size_t channel,
+                         size_t sample_index,
+                         int offset) {
+  float value = static_cast<int>(100 + chunk_counter * chunk_size +
+                                 sample_index + channel) +
+                offset;
+  return 5000 * band + value;
+}
+
+bool VerifySubFrame(
+    size_t sub_frame_counter,
+    int offset,
+    const std::vector<std::vector<rtc::ArrayView<float>>>& sub_frame_view) {
+  for (size_t band = 0; band < sub_frame_view.size(); ++band) {
+    for (size_t channel = 0; channel < sub_frame_view[band].size(); ++channel) {
+      for (size_t sample = 0; sample < sub_frame_view[band][channel].size();
+           ++sample) {
+        const float reference_value = ComputeSampleValue(
+            sub_frame_counter, kSubFrameLength, band, channel, sample, offset);
+        if (reference_value != sub_frame_view[band][channel][sample]) {
+          return false;
+        }
+      }
+    }
+  }
+  return true;
+}
+
+void FillBlock(size_t block_counter, Block* block) {
+  for (int band = 0; band < block->NumBands(); ++band) {
+    for (int channel = 0; channel < block->NumChannels(); ++channel) {
+      auto b = block->View(band, channel);
+      for (size_t sample = 0; sample < kBlockSize; ++sample) {
+        b[sample] = ComputeSampleValue(block_counter, kBlockSize, band, channel,
+                                       sample, 0);
+      }
+    }
+  }
+}
+
+// Verifies that the BlockFramer is able to produce the expected frame content.
+void RunFramerTest(int sample_rate_hz, size_t num_channels) {
+  constexpr size_t kNumSubFramesToProcess = 10;
+  const size_t num_bands = NumBandsForRate(sample_rate_hz);
+
+  Block block(num_bands, num_channels);
+  std::vector<std::vector<std::vector<float>>> output_sub_frame(
+      num_bands, std::vector<std::vector<float>>(
+                     num_channels, std::vector<float>(kSubFrameLength, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> output_sub_frame_view(
+      num_bands, std::vector<rtc::ArrayView<float>>(num_channels));
+  SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
+  BlockFramer framer(num_bands, num_channels);
+
+  size_t block_index = 0;
+  for (size_t sub_frame_index = 0; sub_frame_index < kNumSubFramesToProcess;
+       ++sub_frame_index) {
+    FillBlock(block_index++, &block);
+    framer.InsertBlockAndExtractSubFrame(block, &output_sub_frame_view);
+    if (sub_frame_index > 1) {
+      EXPECT_TRUE(VerifySubFrame(sub_frame_index, -64, output_sub_frame_view));
+    }
+
+    if ((sub_frame_index + 1) % 4 == 0) {
+      FillBlock(block_index++, &block);
+      framer.InsertBlock(block);
+    }
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Verifies that the BlockFramer crashes if the InsertBlockAndExtractSubFrame
+// method is called for inputs with the wrong number of bands or band lengths.
+void RunWronglySizedInsertAndExtractParametersTest(
+    int sample_rate_hz,
+    size_t correct_num_channels,
+    size_t num_block_bands,
+    size_t num_block_channels,
+    size_t num_sub_frame_bands,
+    size_t num_sub_frame_channels,
+    size_t sub_frame_length) {
+  const size_t correct_num_bands = NumBandsForRate(sample_rate_hz);
+
+  Block block(num_block_bands, num_block_channels);
+  std::vector<std::vector<std::vector<float>>> output_sub_frame(
+      num_sub_frame_bands,
+      std::vector<std::vector<float>>(
+          num_sub_frame_channels, std::vector<float>(sub_frame_length, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> output_sub_frame_view(
+      output_sub_frame.size(),
+      std::vector<rtc::ArrayView<float>>(num_sub_frame_channels));
+  SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
+  BlockFramer framer(correct_num_bands, correct_num_channels);
+  EXPECT_DEATH(
+      framer.InsertBlockAndExtractSubFrame(block, &output_sub_frame_view), "");
+}
+
+// Verifies that the BlockFramer crashes if the InsertBlock method is called for
+// inputs with the wrong number of bands or band lengths.
+void RunWronglySizedInsertParameterTest(int sample_rate_hz,
+                                        size_t correct_num_channels,
+                                        size_t num_block_bands,
+                                        size_t num_block_channels) {
+  const size_t correct_num_bands = NumBandsForRate(sample_rate_hz);
+
+  Block correct_block(correct_num_bands, correct_num_channels);
+  Block wrong_block(num_block_bands, num_block_channels);
+  std::vector<std::vector<std::vector<float>>> output_sub_frame(
+      correct_num_bands,
+      std::vector<std::vector<float>>(
+          correct_num_channels, std::vector<float>(kSubFrameLength, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> output_sub_frame_view(
+      output_sub_frame.size(),
+      std::vector<rtc::ArrayView<float>>(correct_num_channels));
+  SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
+  BlockFramer framer(correct_num_bands, correct_num_channels);
+  framer.InsertBlockAndExtractSubFrame(correct_block, &output_sub_frame_view);
+  framer.InsertBlockAndExtractSubFrame(correct_block, &output_sub_frame_view);
+  framer.InsertBlockAndExtractSubFrame(correct_block, &output_sub_frame_view);
+  framer.InsertBlockAndExtractSubFrame(correct_block, &output_sub_frame_view);
+
+  EXPECT_DEATH(framer.InsertBlock(wrong_block), "");
+}
+
+// Verifies that the BlockFramer crashes if the InsertBlock method is called
+// after a wrong number of previous InsertBlockAndExtractSubFrame method calls
+// have been made.
+
+void RunWronglyInsertOrderTest(int sample_rate_hz,
+                               size_t num_channels,
+                               size_t num_preceeding_api_calls) {
+  const size_t correct_num_bands = NumBandsForRate(sample_rate_hz);
+
+  Block block(correct_num_bands, num_channels);
+  std::vector<std::vector<std::vector<float>>> output_sub_frame(
+      correct_num_bands,
+      std::vector<std::vector<float>>(
+          num_channels, std::vector<float>(kSubFrameLength, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> output_sub_frame_view(
+      output_sub_frame.size(),
+      std::vector<rtc::ArrayView<float>>(num_channels));
+  SetupSubFrameView(&output_sub_frame, &output_sub_frame_view);
+  BlockFramer framer(correct_num_bands, num_channels);
+  for (size_t k = 0; k < num_preceeding_api_calls; ++k) {
+    framer.InsertBlockAndExtractSubFrame(block, &output_sub_frame_view);
+  }
+
+  EXPECT_DEATH(framer.InsertBlock(block), "");
+}
+#endif
+
+std::string ProduceDebugText(int sample_rate_hz, size_t num_channels) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz;
+  ss << ", number of channels: " << num_channels;
+  return ss.Release();
+}
+
+}  // namespace
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(BlockFramerDeathTest,
+     WrongNumberOfBandsInBlockForInsertBlockAndExtractSubFrame) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto correct_num_channels : {1, 2, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_bands = (correct_num_bands % 3) + 1;
+      RunWronglySizedInsertAndExtractParametersTest(
+          rate, correct_num_channels, wrong_num_bands, correct_num_channels,
+          correct_num_bands, correct_num_channels, kSubFrameLength);
+    }
+  }
+}
+
+TEST(BlockFramerDeathTest,
+     WrongNumberOfChannelsInBlockForInsertBlockAndExtractSubFrame) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto correct_num_channels : {1, 2, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_channels = correct_num_channels + 1;
+      RunWronglySizedInsertAndExtractParametersTest(
+          rate, correct_num_channels, correct_num_bands, wrong_num_channels,
+          correct_num_bands, correct_num_channels, kSubFrameLength);
+    }
+  }
+}
+
+TEST(BlockFramerDeathTest,
+     WrongNumberOfBandsInSubFrameForInsertBlockAndExtractSubFrame) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto correct_num_channels : {1, 2, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_bands = (correct_num_bands % 3) + 1;
+      RunWronglySizedInsertAndExtractParametersTest(
+          rate, correct_num_channels, correct_num_bands, correct_num_channels,
+          wrong_num_bands, correct_num_channels, kSubFrameLength);
+    }
+  }
+}
+
+TEST(BlockFramerDeathTest,
+     WrongNumberOfChannelsInSubFrameForInsertBlockAndExtractSubFrame) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto correct_num_channels : {1, 2, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_channels = correct_num_channels + 1;
+      RunWronglySizedInsertAndExtractParametersTest(
+          rate, correct_num_channels, correct_num_bands, correct_num_channels,
+          correct_num_bands, wrong_num_channels, kSubFrameLength);
+    }
+  }
+}
+
+TEST(BlockFramerDeathTest,
+     WrongNumberOfSamplesInSubFrameForInsertBlockAndExtractSubFrame) {
+  const size_t correct_num_channels = 1;
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+    const size_t correct_num_bands = NumBandsForRate(rate);
+    RunWronglySizedInsertAndExtractParametersTest(
+        rate, correct_num_channels, correct_num_bands, correct_num_channels,
+        correct_num_bands, correct_num_channels, kSubFrameLength - 1);
+  }
+}
+
+TEST(BlockFramerDeathTest, WrongNumberOfBandsInBlockForInsertBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto correct_num_channels : {1, 2, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_bands = (correct_num_bands % 3) + 1;
+      RunWronglySizedInsertParameterTest(rate, correct_num_channels,
+                                         wrong_num_bands, correct_num_channels);
+    }
+  }
+}
+
+TEST(BlockFramerDeathTest, WrongNumberOfChannelsInBlockForInsertBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto correct_num_channels : {1, 2, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_channels = correct_num_channels + 1;
+      RunWronglySizedInsertParameterTest(rate, correct_num_channels,
+                                         correct_num_bands, wrong_num_channels);
+    }
+  }
+}
+
+TEST(BlockFramerDeathTest, WrongNumberOfPreceedingApiCallsForInsertBlock) {
+  for (size_t num_channels : {1, 2, 8}) {
+    for (auto rate : {16000, 32000, 48000}) {
+      for (size_t num_calls = 0; num_calls < 4; ++num_calls) {
+        rtc::StringBuilder ss;
+        ss << "Sample rate: " << rate;
+        ss << ", Num channels: " << num_channels;
+        ss << ", Num preceeding InsertBlockAndExtractSubFrame calls: "
+           << num_calls;
+
+        SCOPED_TRACE(ss.str());
+        RunWronglyInsertOrderTest(rate, num_channels, num_calls);
+      }
+    }
+  }
+}
+
+// Verifies that the verification for 0 number of channels works.
+TEST(BlockFramerDeathTest, ZeroNumberOfChannelsParameter) {
+  EXPECT_DEATH(BlockFramer(16000, 0), "");
+}
+
+// Verifies that the verification for 0 number of bands works.
+TEST(BlockFramerDeathTest, ZeroNumberOfBandsParameter) {
+  EXPECT_DEATH(BlockFramer(0, 1), "");
+}
+
+// Verifies that the verification for null sub_frame pointer works.
+TEST(BlockFramerDeathTest, NullSubFrameParameter) {
+  EXPECT_DEATH(
+      BlockFramer(1, 1).InsertBlockAndExtractSubFrame(Block(1, 1), nullptr),
+      "");
+}
+
+#endif
+
+TEST(BlockFramer, FrameBitexactness) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, num_channels));
+      RunFramerTest(rate, num_channels);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_processor.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor.cc
new file mode 100644
index 0000000000..63e3d9cc7c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor.cc
@@ -0,0 +1,290 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/block_processor.h"
+
+#include <stddef.h>
+
+#include <atomic>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "api/audio/echo_control.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/block_processor_metrics.h"
+#include "modules/audio_processing/aec3/delay_estimate.h"
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+#include "modules/audio_processing/aec3/echo_remover.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/aec3/render_delay_controller.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+enum class BlockProcessorApiCall { kCapture, kRender };
+
+class BlockProcessorImpl final : public BlockProcessor {
+ public:
+  BlockProcessorImpl(const EchoCanceller3Config& config,
+                     int sample_rate_hz,
+                     size_t num_render_channels,
+                     size_t num_capture_channels,
+                     std::unique_ptr<RenderDelayBuffer> render_buffer,
+                     std::unique_ptr<RenderDelayController> delay_controller,
+                     std::unique_ptr<EchoRemover> echo_remover);
+
+  BlockProcessorImpl() = delete;
+
+  ~BlockProcessorImpl() override;
+
+  void ProcessCapture(bool echo_path_gain_change,
+                      bool capture_signal_saturation,
+                      Block* linear_output,
+                      Block* capture_block) override;
+
+  void BufferRender(const Block& block) override;
+
+  void UpdateEchoLeakageStatus(bool leakage_detected) override;
+
+  void GetMetrics(EchoControl::Metrics* metrics) const override;
+
+  void SetAudioBufferDelay(int delay_ms) override;
+  void SetCaptureOutputUsage(bool capture_output_used) override;
+
+ private:
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  const EchoCanceller3Config config_;
+  bool capture_properly_started_ = false;
+  bool render_properly_started_ = false;
+  const size_t sample_rate_hz_;
+  std::unique_ptr<RenderDelayBuffer> render_buffer_;
+  std::unique_ptr<RenderDelayController> delay_controller_;
+  std::unique_ptr<EchoRemover> echo_remover_;
+  BlockProcessorMetrics metrics_;
+  RenderDelayBuffer::BufferingEvent render_event_;
+  size_t capture_call_counter_ = 0;
+  absl::optional<DelayEstimate> estimated_delay_;
+};
+
+std::atomic<int> BlockProcessorImpl::instance_count_(0);
+
+BlockProcessorImpl::BlockProcessorImpl(
+    const EchoCanceller3Config& config,
+    int sample_rate_hz,
+    size_t num_render_channels,
+    size_t num_capture_channels,
+    std::unique_ptr<RenderDelayBuffer> render_buffer,
+    std::unique_ptr<RenderDelayController> delay_controller,
+    std::unique_ptr<EchoRemover> echo_remover)
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      config_(config),
+      sample_rate_hz_(sample_rate_hz),
+      render_buffer_(std::move(render_buffer)),
+      delay_controller_(std::move(delay_controller)),
+      echo_remover_(std::move(echo_remover)),
+      render_event_(RenderDelayBuffer::BufferingEvent::kNone) {
+  RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
+}
+
+BlockProcessorImpl::~BlockProcessorImpl() = default;
+
+void BlockProcessorImpl::ProcessCapture(bool echo_path_gain_change,
+                                        bool capture_signal_saturation,
+                                        Block* linear_output,
+                                        Block* capture_block) {
+  RTC_DCHECK(capture_block);
+  RTC_DCHECK_EQ(NumBandsForRate(sample_rate_hz_), capture_block->NumBands());
+
+  capture_call_counter_++;
+
+  data_dumper_->DumpRaw("aec3_processblock_call_order",
+                        static_cast<int>(BlockProcessorApiCall::kCapture));
+  data_dumper_->DumpWav("aec3_processblock_capture_input",
+                        capture_block->View(/*band=*/0, /*channel=*/0), 16000,
+                        1);
+
+  if (render_properly_started_) {
+    if (!capture_properly_started_) {
+      capture_properly_started_ = true;
+      render_buffer_->Reset();
+      if (delay_controller_)
+        delay_controller_->Reset(true);
+    }
+  } else {
+    // If no render data has yet arrived, do not process the capture signal.
+    render_buffer_->HandleSkippedCaptureProcessing();
+    return;
+  }
+
+  EchoPathVariability echo_path_variability(
+      echo_path_gain_change, EchoPathVariability::DelayAdjustment::kNone,
+      false);
+
+  if (render_event_ == RenderDelayBuffer::BufferingEvent::kRenderOverrun &&
+      render_properly_started_) {
+    echo_path_variability.delay_change =
+        EchoPathVariability::DelayAdjustment::kBufferFlush;
+    if (delay_controller_)
+      delay_controller_->Reset(true);
+    RTC_LOG(LS_WARNING) << "Reset due to render buffer overrun at block  "
+                        << capture_call_counter_;
+  }
+  render_event_ = RenderDelayBuffer::BufferingEvent::kNone;
+
+  // Update the render buffers with any newly arrived render blocks and prepare
+  // the render buffers for reading the render data corresponding to the current
+  // capture block.
+  RenderDelayBuffer::BufferingEvent buffer_event =
+      render_buffer_->PrepareCaptureProcessing();
+  // Reset the delay controller at render buffer underrun.
+  if (buffer_event == RenderDelayBuffer::BufferingEvent::kRenderUnderrun) {
+    if (delay_controller_)
+      delay_controller_->Reset(false);
+  }
+
+  data_dumper_->DumpWav("aec3_processblock_capture_input2",
+                        capture_block->View(/*band=*/0, /*channel=*/0), 16000,
+                        1);
+
+  bool has_delay_estimator = !config_.delay.use_external_delay_estimator;
+  if (has_delay_estimator) {
+    RTC_DCHECK(delay_controller_);
+    // Compute and apply the render delay required to achieve proper signal
+    // alignment.
+    estimated_delay_ = delay_controller_->GetDelay(
+        render_buffer_->GetDownsampledRenderBuffer(), render_buffer_->Delay(),
+        *capture_block);
+
+    if (estimated_delay_) {
+      bool delay_change =
+          render_buffer_->AlignFromDelay(estimated_delay_->delay);
+      if (delay_change) {
+        rtc::LoggingSeverity log_level =
+            config_.delay.log_warning_on_delay_changes ? rtc::LS_WARNING
+                                                       : rtc::LS_INFO;
+        RTC_LOG_V(log_level) << "Delay changed to " << estimated_delay_->delay
+                             << " at block " << capture_call_counter_;
+        echo_path_variability.delay_change =
+            EchoPathVariability::DelayAdjustment::kNewDetectedDelay;
+      }
+    }
+
+    echo_path_variability.clock_drift = delay_controller_->HasClockdrift();
+
+  } else {
+    render_buffer_->AlignFromExternalDelay();
+  }
+
+  // Remove the echo from the capture signal.
+  if (has_delay_estimator || render_buffer_->HasReceivedBufferDelay()) {
+    echo_remover_->ProcessCapture(
+        echo_path_variability, capture_signal_saturation, estimated_delay_,
+        render_buffer_->GetRenderBuffer(), linear_output, capture_block);
+  }
+
+  // Update the metrics.
+  metrics_.UpdateCapture(false);
+}
+
+void BlockProcessorImpl::BufferRender(const Block& block) {
+  RTC_DCHECK_EQ(NumBandsForRate(sample_rate_hz_), block.NumBands());
+  data_dumper_->DumpRaw("aec3_processblock_call_order",
+                        static_cast<int>(BlockProcessorApiCall::kRender));
+  data_dumper_->DumpWav("aec3_processblock_render_input",
+                        block.View(/*band=*/0, /*channel=*/0), 16000, 1);
+
+  render_event_ = render_buffer_->Insert(block);
+
+  metrics_.UpdateRender(render_event_ !=
+                        RenderDelayBuffer::BufferingEvent::kNone);
+
+  render_properly_started_ = true;
+  if (delay_controller_)
+    delay_controller_->LogRenderCall();
+}
+
+void BlockProcessorImpl::UpdateEchoLeakageStatus(bool leakage_detected) {
+  echo_remover_->UpdateEchoLeakageStatus(leakage_detected);
+}
+
+void BlockProcessorImpl::GetMetrics(EchoControl::Metrics* metrics) const {
+  echo_remover_->GetMetrics(metrics);
+  constexpr int block_size_ms = 4;
+  absl::optional<size_t> delay = render_buffer_->Delay();
+  metrics->delay_ms = delay ? static_cast<int>(*delay) * block_size_ms : 0;
+}
+
+void BlockProcessorImpl::SetAudioBufferDelay(int delay_ms) {
+  render_buffer_->SetAudioBufferDelay(delay_ms);
+}
+
+void BlockProcessorImpl::SetCaptureOutputUsage(bool capture_output_used) {
+  echo_remover_->SetCaptureOutputUsage(capture_output_used);
+}
+
+}  // namespace
+
+BlockProcessor* BlockProcessor::Create(const EchoCanceller3Config& config,
+                                       int sample_rate_hz,
+                                       size_t num_render_channels,
+                                       size_t num_capture_channels) {
+  std::unique_ptr<RenderDelayBuffer> render_buffer(
+      RenderDelayBuffer::Create(config, sample_rate_hz, num_render_channels));
+  std::unique_ptr<RenderDelayController> delay_controller;
+  if (!config.delay.use_external_delay_estimator) {
+    delay_controller.reset(RenderDelayController::Create(config, sample_rate_hz,
+                                                         num_capture_channels));
+  }
+  std::unique_ptr<EchoRemover> echo_remover(EchoRemover::Create(
+      config, sample_rate_hz, num_render_channels, num_capture_channels));
+  return Create(config, sample_rate_hz, num_render_channels,
+                num_capture_channels, std::move(render_buffer),
+                std::move(delay_controller), std::move(echo_remover));
+}
+
+BlockProcessor* BlockProcessor::Create(
+    const EchoCanceller3Config& config,
+    int sample_rate_hz,
+    size_t num_render_channels,
+    size_t num_capture_channels,
+    std::unique_ptr<RenderDelayBuffer> render_buffer) {
+  std::unique_ptr<RenderDelayController> delay_controller;
+  if (!config.delay.use_external_delay_estimator) {
+    delay_controller.reset(RenderDelayController::Create(config, sample_rate_hz,
+                                                         num_capture_channels));
+  }
+  std::unique_ptr<EchoRemover> echo_remover(EchoRemover::Create(
+      config, sample_rate_hz, num_render_channels, num_capture_channels));
+  return Create(config, sample_rate_hz, num_render_channels,
+                num_capture_channels, std::move(render_buffer),
+                std::move(delay_controller), std::move(echo_remover));
+}
+
+BlockProcessor* BlockProcessor::Create(
+    const EchoCanceller3Config& config,
+    int sample_rate_hz,
+    size_t num_render_channels,
+    size_t num_capture_channels,
+    std::unique_ptr<RenderDelayBuffer> render_buffer,
+    std::unique_ptr<RenderDelayController> delay_controller,
+    std::unique_ptr<EchoRemover> echo_remover) {
+  return new BlockProcessorImpl(config, sample_rate_hz, num_render_channels,
+                                num_capture_channels, std::move(render_buffer),
+                                std::move(delay_controller),
+                                std::move(echo_remover));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_processor.h b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor.h
new file mode 100644
index 0000000000..01a83ae5f7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "api/audio/echo_control.h"
+#include "modules/audio_processing/aec3/block.h"
+#include "modules/audio_processing/aec3/echo_remover.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/aec3/render_delay_controller.h"
+
+namespace webrtc {
+
+// Class for performing echo cancellation on 64 sample blocks of audio data.
+class BlockProcessor {
+ public:
+  static BlockProcessor* Create(const EchoCanceller3Config& config,
+                                int sample_rate_hz,
+                                size_t num_render_channels,
+                                size_t num_capture_channels);
+  // Only used for testing purposes.
+  static BlockProcessor* Create(
+      const EchoCanceller3Config& config,
+      int sample_rate_hz,
+      size_t num_render_channels,
+      size_t num_capture_channels,
+      std::unique_ptr<RenderDelayBuffer> render_buffer);
+  static BlockProcessor* Create(
+      const EchoCanceller3Config& config,
+      int sample_rate_hz,
+      size_t num_render_channels,
+      size_t num_capture_channels,
+      std::unique_ptr<RenderDelayBuffer> render_buffer,
+      std::unique_ptr<RenderDelayController> delay_controller,
+      std::unique_ptr<EchoRemover> echo_remover);
+
+  virtual ~BlockProcessor() = default;
+
+  // Get current metrics.
+  virtual void GetMetrics(EchoControl::Metrics* metrics) const = 0;
+
+  // Provides an optional external estimate of the audio buffer delay.
+  virtual void SetAudioBufferDelay(int delay_ms) = 0;
+
+  // Processes a block of capture data.
+  virtual void ProcessCapture(bool echo_path_gain_change,
+                              bool capture_signal_saturation,
+                              Block* linear_output,
+                              Block* capture_block) = 0;
+
+  // Buffers a block of render data supplied by a FrameBlocker object.
+  virtual void BufferRender(const Block& render_block) = 0;
+
+  // Reports whether echo leakage has been detected in the echo canceller
+  // output.
+  virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0;
+
+  // Specifies whether the capture output will be used. The purpose of this is
+  // to allow the block processor to deactivate some of the processing when the
+  // resulting output is anyway not used, for instance when the endpoint is
+  // muted.
+  virtual void SetCaptureOutputUsage(bool capture_output_used) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics.cc
new file mode 100644
index 0000000000..deac1fcd22
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics.cc
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/block_processor_metrics.h"
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+enum class RenderUnderrunCategory {
+  kNone,
+  kFew,
+  kSeveral,
+  kMany,
+  kConstant,
+  kNumCategories
+};
+
+enum class RenderOverrunCategory {
+  kNone,
+  kFew,
+  kSeveral,
+  kMany,
+  kConstant,
+  kNumCategories
+};
+
+}  // namespace
+
+void BlockProcessorMetrics::UpdateCapture(bool underrun) {
+  ++capture_block_counter_;
+  if (underrun) {
+    ++render_buffer_underruns_;
+  }
+
+  if (capture_block_counter_ == kMetricsReportingIntervalBlocks) {
+    metrics_reported_ = true;
+
+    RenderUnderrunCategory underrun_category;
+    if (render_buffer_underruns_ == 0) {
+      underrun_category = RenderUnderrunCategory::kNone;
+    } else if (render_buffer_underruns_ > (capture_block_counter_ >> 1)) {
+      underrun_category = RenderUnderrunCategory::kConstant;
+    } else if (render_buffer_underruns_ > 100) {
+      underrun_category = RenderUnderrunCategory::kMany;
+    } else if (render_buffer_underruns_ > 10) {
+      underrun_category = RenderUnderrunCategory::kSeveral;
+    } else {
+      underrun_category = RenderUnderrunCategory::kFew;
+    }
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.EchoCanceller.RenderUnderruns",
+        static_cast<int>(underrun_category),
+        static_cast<int>(RenderUnderrunCategory::kNumCategories));
+
+    RenderOverrunCategory overrun_category;
+    if (render_buffer_overruns_ == 0) {
+      overrun_category = RenderOverrunCategory::kNone;
+    } else if (render_buffer_overruns_ > (buffer_render_calls_ >> 1)) {
+      overrun_category = RenderOverrunCategory::kConstant;
+    } else if (render_buffer_overruns_ > 100) {
+      overrun_category = RenderOverrunCategory::kMany;
+    } else if (render_buffer_overruns_ > 10) {
+      overrun_category = RenderOverrunCategory::kSeveral;
+    } else {
+      overrun_category = RenderOverrunCategory::kFew;
+    }
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.EchoCanceller.RenderOverruns",
+        static_cast<int>(overrun_category),
+        static_cast<int>(RenderOverrunCategory::kNumCategories));
+
+    ResetMetrics();
+    capture_block_counter_ = 0;
+  } else {
+    metrics_reported_ = false;
+  }
+}
+
+void BlockProcessorMetrics::UpdateRender(bool overrun) {
+  ++buffer_render_calls_;
+  if (overrun) {
+    ++render_buffer_overruns_;
+  }
+}
+
+void BlockProcessorMetrics::ResetMetrics() {
+  render_buffer_underruns_ = 0;
+  render_buffer_overruns_ = 0;
+  buffer_render_calls_ = 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics.h b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics.h
new file mode 100644
index 0000000000..a70d0dac5b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_METRICS_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_METRICS_H_
+
+namespace webrtc {
+
+// Handles the reporting of metrics for the block_processor.
+class BlockProcessorMetrics {
+ public:
+  BlockProcessorMetrics() = default;
+
+  BlockProcessorMetrics(const BlockProcessorMetrics&) = delete;
+  BlockProcessorMetrics& operator=(const BlockProcessorMetrics&) = delete;
+
+  // Updates the metric with new capture data.
+  void UpdateCapture(bool underrun);
+
+  // Updates the metric with new render data.
+  void UpdateRender(bool overrun);
+
+  // Returns true if the metrics have just been reported, otherwise false.
+  bool MetricsReported() { return metrics_reported_; }
+
+ private:
+  // Resets the metrics.
+  void ResetMetrics();
+
+  int capture_block_counter_ = 0;
+  bool metrics_reported_ = false;
+  int render_buffer_underruns_ = 0;
+  int render_buffer_overruns_ = 0;
+  int buffer_render_calls_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_METRICS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
new file mode 100644
index 0000000000..3e23c2499d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/block_processor_metrics.h"
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Verify the general functionality of BlockProcessorMetrics.
+TEST(BlockProcessorMetrics, NormalUsage) {
+  BlockProcessorMetrics metrics;
+
+  for (int j = 0; j < 3; ++j) {
+    for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
+      metrics.UpdateRender(false);
+      metrics.UpdateRender(false);
+      metrics.UpdateCapture(false);
+      EXPECT_FALSE(metrics.MetricsReported());
+    }
+    metrics.UpdateCapture(false);
+    EXPECT_TRUE(metrics.MetricsReported());
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_unittest.cc
new file mode 100644
index 0000000000..aba5c4186d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/block_processor_unittest.cc
@@ -0,0 +1,341 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/block_processor.h"
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/mock/mock_echo_remover.h"
+#include "modules/audio_processing/aec3/mock/mock_render_delay_buffer.h"
+#include "modules/audio_processing/aec3/mock/mock_render_delay_controller.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::StrictMock;
+
+// Verifies that the basic BlockProcessor functionality works and that the API
+// methods are callable.
+void RunBasicSetupAndApiCallTest(int sample_rate_hz, int num_iterations) {
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 1;
+
+  std::unique_ptr<BlockProcessor> block_processor(
+      BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz,
+                             kNumRenderChannels, kNumCaptureChannels));
+  Block block(NumBandsForRate(sample_rate_hz), kNumRenderChannels, 1000.f);
+  for (int k = 0; k < num_iterations; ++k) {
+    block_processor->BufferRender(block);
+    block_processor->ProcessCapture(false, false, nullptr, &block);
+    block_processor->UpdateEchoLeakageStatus(false);
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+void RunRenderBlockSizeVerificationTest(int sample_rate_hz) {
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 1;
+
+  std::unique_ptr<BlockProcessor> block_processor(
+      BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz,
+                             kNumRenderChannels, kNumCaptureChannels));
+  Block block(NumBandsForRate(sample_rate_hz), kNumRenderChannels);
+
+  EXPECT_DEATH(block_processor->BufferRender(block), "");
+}
+
+void RunRenderNumBandsVerificationTest(int sample_rate_hz) {
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 1;
+
+  const size_t wrong_num_bands = NumBandsForRate(sample_rate_hz) < 3
+                                     ? NumBandsForRate(sample_rate_hz) + 1
+                                     : 1;
+  std::unique_ptr<BlockProcessor> block_processor(
+      BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz,
+                             kNumRenderChannels, kNumCaptureChannels));
+  Block block(wrong_num_bands, kNumRenderChannels);
+
+  EXPECT_DEATH(block_processor->BufferRender(block), "");
+}
+
+void RunCaptureNumBandsVerificationTest(int sample_rate_hz) {
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 1;
+
+  const size_t wrong_num_bands = NumBandsForRate(sample_rate_hz) < 3
+                                     ? NumBandsForRate(sample_rate_hz) + 1
+                                     : 1;
+  std::unique_ptr<BlockProcessor> block_processor(
+      BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz,
+                             kNumRenderChannels, kNumCaptureChannels));
+  Block block(wrong_num_bands, kNumRenderChannels);
+
+  EXPECT_DEATH(block_processor->ProcessCapture(false, false, nullptr, &block),
+               "");
+}
+#endif
+
+std::string ProduceDebugText(int sample_rate_hz) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz;
+  return ss.Release();
+}
+
+void FillSampleVector(int call_counter,
+                      int delay,
+                      rtc::ArrayView<float> samples) {
+  for (size_t i = 0; i < samples.size(); ++i) {
+    samples[i] = (call_counter - delay) * 10000.0f + i;
+  }
+}
+
+}  // namespace
+
+// Verifies that the delay controller functionality is properly integrated with
+// the render delay buffer inside block processor.
+// TODO(peah): Activate the unittest once the required code has been landed.
+TEST(BlockProcessor, DISABLED_DelayControllerIntegration) {
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 1;
+  constexpr size_t kNumBlocks = 310;
+  constexpr size_t kDelayInSamples = 640;
+  constexpr size_t kDelayHeadroom = 1;
+  constexpr size_t kDelayInBlocks =
+      kDelayInSamples / kBlockSize - kDelayHeadroom;
+  Random random_generator(42U);
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    std::unique_ptr<testing::StrictMock<webrtc::test::MockRenderDelayBuffer>>
+        render_delay_buffer_mock(
+            new StrictMock<webrtc::test::MockRenderDelayBuffer>(rate, 1));
+    EXPECT_CALL(*render_delay_buffer_mock, Insert(_))
+        .Times(kNumBlocks)
+        .WillRepeatedly(Return(RenderDelayBuffer::BufferingEvent::kNone));
+    EXPECT_CALL(*render_delay_buffer_mock, AlignFromDelay(kDelayInBlocks))
+        .Times(AtLeast(1));
+    EXPECT_CALL(*render_delay_buffer_mock, MaxDelay()).WillOnce(Return(30));
+    EXPECT_CALL(*render_delay_buffer_mock, Delay())
+        .Times(kNumBlocks + 1)
+        .WillRepeatedly(Return(0));
+    std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
+        EchoCanceller3Config(), rate, kNumRenderChannels, kNumCaptureChannels,
+        std::move(render_delay_buffer_mock)));
+
+    Block render_block(NumBandsForRate(rate), kNumRenderChannels);
+    Block capture_block(NumBandsForRate(rate), kNumCaptureChannels);
+    DelayBuffer<float> signal_delay_buffer(kDelayInSamples);
+    for (size_t k = 0; k < kNumBlocks; ++k) {
+      RandomizeSampleVector(&random_generator,
+                            render_block.View(/*band=*/0, /*capture=*/0));
+      signal_delay_buffer.Delay(render_block.View(/*band=*/0, /*capture=*/0),
+                                capture_block.View(/*band=*/0, /*capture=*/0));
+      block_processor->BufferRender(render_block);
+      block_processor->ProcessCapture(false, false, nullptr, &capture_block);
+    }
+  }
+}
+
+// Verifies that BlockProcessor submodules are called in a proper manner.
+TEST(BlockProcessor, DISABLED_SubmoduleIntegration) {
+  constexpr size_t kNumBlocks = 310;
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 1;
+
+  Random random_generator(42U);
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    std::unique_ptr<testing::StrictMock<webrtc::test::MockRenderDelayBuffer>>
+        render_delay_buffer_mock(
+            new StrictMock<webrtc::test::MockRenderDelayBuffer>(rate, 1));
+    std::unique_ptr<
+        ::testing::StrictMock<webrtc::test::MockRenderDelayController>>
+        render_delay_controller_mock(
+            new StrictMock<webrtc::test::MockRenderDelayController>());
+    std::unique_ptr<testing::StrictMock<webrtc::test::MockEchoRemover>>
+        echo_remover_mock(new StrictMock<webrtc::test::MockEchoRemover>());
+
+    EXPECT_CALL(*render_delay_buffer_mock, Insert(_))
+        .Times(kNumBlocks - 1)
+        .WillRepeatedly(Return(RenderDelayBuffer::BufferingEvent::kNone));
+    EXPECT_CALL(*render_delay_buffer_mock, PrepareCaptureProcessing())
+        .Times(kNumBlocks);
+    EXPECT_CALL(*render_delay_buffer_mock, AlignFromDelay(9)).Times(AtLeast(1));
+    EXPECT_CALL(*render_delay_buffer_mock, Delay())
+        .Times(kNumBlocks)
+        .WillRepeatedly(Return(0));
+    EXPECT_CALL(*render_delay_controller_mock, GetDelay(_, _, _))
+        .Times(kNumBlocks);
+    EXPECT_CALL(*echo_remover_mock, ProcessCapture(_, _, _, _, _, _))
+        .Times(kNumBlocks);
+    EXPECT_CALL(*echo_remover_mock, UpdateEchoLeakageStatus(_))
+        .Times(kNumBlocks);
+
+    std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
+        EchoCanceller3Config(), rate, kNumRenderChannels, kNumCaptureChannels,
+        std::move(render_delay_buffer_mock),
+        std::move(render_delay_controller_mock), std::move(echo_remover_mock)));
+
+    Block render_block(NumBandsForRate(rate), kNumRenderChannels);
+    Block capture_block(NumBandsForRate(rate), kNumCaptureChannels);
+    DelayBuffer<float> signal_delay_buffer(640);
+    for (size_t k = 0; k < kNumBlocks; ++k) {
+      RandomizeSampleVector(&random_generator,
+                            render_block.View(/*band=*/0, /*capture=*/0));
+      signal_delay_buffer.Delay(render_block.View(/*band=*/0, /*capture=*/0),
+                                capture_block.View(/*band=*/0, /*capture=*/0));
+      block_processor->BufferRender(render_block);
+      block_processor->ProcessCapture(false, false, nullptr, &capture_block);
+      block_processor->UpdateEchoLeakageStatus(false);
+    }
+  }
+}
+
+TEST(BlockProcessor, BasicSetupAndApiCalls) {
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    RunBasicSetupAndApiCallTest(rate, 1);
+  }
+}
+
+TEST(BlockProcessor, TestLongerCall) {
+  RunBasicSetupAndApiCallTest(16000, 20 * kNumBlocksPerSecond);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// TODO(gustaf): Re-enable the test once the issue with memory leaks during
+// DEATH tests on test bots has been fixed.
+TEST(BlockProcessorDeathTest, DISABLED_VerifyRenderBlockSizeCheck) {
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    RunRenderBlockSizeVerificationTest(rate);
+  }
+}
+
+TEST(BlockProcessorDeathTest, VerifyRenderNumBandsCheck) {
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    RunRenderNumBandsVerificationTest(rate);
+  }
+}
+
+// TODO(peah): Verify the check for correct number of bands in the capture
+// signal.
+TEST(BlockProcessorDeathTest, VerifyCaptureNumBandsCheck) {
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    RunCaptureNumBandsVerificationTest(rate);
+  }
+}
+
+// Verifiers that the verification for null ProcessCapture input works.
+TEST(BlockProcessorDeathTest, NullProcessCaptureParameter) {
+  EXPECT_DEATH(std::unique_ptr<BlockProcessor>(
+                   BlockProcessor::Create(EchoCanceller3Config(), 16000, 1, 1))
+                   ->ProcessCapture(false, false, nullptr, nullptr),
+               "");
+}
+
+// Verifies the check for correct sample rate.
+// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
+// tests on test bots has been fixed.
+TEST(BlockProcessor, DISABLED_WrongSampleRate) {
+  EXPECT_DEATH(std::unique_ptr<BlockProcessor>(
+                   BlockProcessor::Create(EchoCanceller3Config(), 8001, 1, 1)),
+               "");
+}
+
+#endif
+
+// Verifies that external delay estimator delays are applied correctly when a
+// call begins with a sequence of capture blocks.
+TEST(BlockProcessor, ExternalDelayAppliedCorrectlyWithInitialCaptureCalls) {
+  constexpr int kNumRenderChannels = 1;
+  constexpr int kNumCaptureChannels = 1;
+  constexpr int kSampleRateHz = 16000;
+
+  EchoCanceller3Config config;
+  config.delay.use_external_delay_estimator = true;
+
+  std::unique_ptr<RenderDelayBuffer> delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, kNumRenderChannels));
+
+  std::unique_ptr<testing::NiceMock<webrtc::test::MockEchoRemover>>
+      echo_remover_mock(new NiceMock<webrtc::test::MockEchoRemover>());
+  webrtc::test::MockEchoRemover* echo_remover_mock_pointer =
+      echo_remover_mock.get();
+
+  std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
+      config, kSampleRateHz, kNumRenderChannels, kNumCaptureChannels,
+      std::move(delay_buffer), /*delay_controller=*/nullptr,
+      std::move(echo_remover_mock)));
+
+  Block render_block(NumBandsForRate(kSampleRateHz), kNumRenderChannels);
+  Block capture_block(NumBandsForRate(kSampleRateHz), kNumCaptureChannels);
+
+  // Process...
+  // - 10 capture calls, where no render data is available,
+  // - 10 render calls, populating the buffer,
+  // - 2 capture calls, verifying that the delay was applied correctly.
+  constexpr int kDelayInBlocks = 5;
+  constexpr int kDelayInMs = 20;
+  block_processor->SetAudioBufferDelay(kDelayInMs);
+
+  int capture_call_counter = 0;
+  int render_call_counter = 0;
+  for (size_t k = 0; k < 10; ++k) {
+    FillSampleVector(++capture_call_counter, kDelayInBlocks,
+                     capture_block.View(/*band=*/0, /*capture=*/0));
+    block_processor->ProcessCapture(false, false, nullptr, &capture_block);
+  }
+  for (size_t k = 0; k < 10; ++k) {
+    FillSampleVector(++render_call_counter, 0,
+                     render_block.View(/*band=*/0, /*capture=*/0));
+    block_processor->BufferRender(render_block);
+  }
+
+  EXPECT_CALL(*echo_remover_mock_pointer, ProcessCapture)
+      .WillRepeatedly(
+          [](EchoPathVariability /*echo_path_variability*/,
+             bool /*capture_signal_saturation*/,
+             const absl::optional<DelayEstimate>& /*external_delay*/,
+             RenderBuffer* render_buffer, Block* /*linear_output*/,
+             Block* capture) {
+            const auto& render = render_buffer->GetBlock(0);
+            const auto render_view = render.View(/*band=*/0, /*channel=*/0);
+            const auto capture_view = capture->View(/*band=*/0, /*channel=*/0);
+            for (size_t i = 0; i < kBlockSize; ++i) {
+              EXPECT_FLOAT_EQ(render_view[i], capture_view[i]);
+            }
+          });
+
+  FillSampleVector(++capture_call_counter, kDelayInBlocks,
+                   capture_block.View(/*band=*/0, /*capture=*/0));
+  block_processor->ProcessCapture(false, false, nullptr, &capture_block);
+
+  FillSampleVector(++capture_call_counter, kDelayInBlocks,
+                   capture_block.View(/*band=*/0, /*capture=*/0));
+  block_processor->ProcessCapture(false, false, nullptr, &capture_block);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector.cc b/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector.cc
new file mode 100644
index 0000000000..2c49b795c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/clockdrift_detector.h"
+
+namespace webrtc {
+
+ClockdriftDetector::ClockdriftDetector()
+    : level_(Level::kNone), stability_counter_(0) {
+  delay_history_.fill(0);
+}
+
+ClockdriftDetector::~ClockdriftDetector() = default;
+
+void ClockdriftDetector::Update(int delay_estimate) {
+  if (delay_estimate == delay_history_[0]) {
+    // Reset clockdrift level if delay estimate is stable for 7500 blocks (30
+    // seconds).
+    if (++stability_counter_ > 7500)
+      level_ = Level::kNone;
+    return;
+  }
+
+  stability_counter_ = 0;
+  const int d1 = delay_history_[0] - delay_estimate;
+  const int d2 = delay_history_[1] - delay_estimate;
+  const int d3 = delay_history_[2] - delay_estimate;
+
+  // Patterns recognized as positive clockdrift:
+  // [x-3], x-2, x-1, x.
+  // [x-3], x-1, x-2, x.
+  const bool probable_drift_up =
+      (d1 == -1 && d2 == -2) || (d1 == -2 && d2 == -1);
+  const bool drift_up = probable_drift_up && d3 == -3;
+
+  // Patterns recognized as negative clockdrift:
+  // [x+3], x+2, x+1, x.
+  // [x+3], x+1, x+2, x.
+  const bool probable_drift_down = (d1 == 1 && d2 == 2) || (d1 == 2 && d2 == 1);
+  const bool drift_down = probable_drift_down && d3 == 3;
+
+  // Set clockdrift level.
+  if (drift_up || drift_down) {
+    level_ = Level::kVerified;
+  } else if ((probable_drift_up || probable_drift_down) &&
+             level_ == Level::kNone) {
+    level_ = Level::kProbable;
+  }
+
+  // Shift delay history one step.
+  delay_history_[2] = delay_history_[1];
+  delay_history_[1] = delay_history_[0];
+  delay_history_[0] = delay_estimate;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector.h b/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector.h
new file mode 100644
index 0000000000..2ba90bb889
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_CLOCKDRIFT_DETECTOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_CLOCKDRIFT_DETECTOR_H_
+
+#include <stddef.h>
+
+#include <array>
+
+namespace webrtc {
+
+class ApmDataDumper;
+struct DownsampledRenderBuffer;
+struct EchoCanceller3Config;
+
+// Detects clockdrift by analyzing the estimated delay.
+class ClockdriftDetector {
+ public:
+  enum class Level { kNone, kProbable, kVerified, kNumCategories };
+  ClockdriftDetector();
+  ~ClockdriftDetector();
+  void Update(int delay_estimate);
+  Level ClockdriftLevel() const { return level_; }
+
+ private:
+  std::array<int, 3> delay_history_;
+  Level level_;
+  size_t stability_counter_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_CLOCKDRIFT_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector_unittest.cc
new file mode 100644
index 0000000000..0f98b01d3a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/clockdrift_detector_unittest.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/clockdrift_detector.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+TEST(ClockdriftDetector, ClockdriftDetector) {
+  ClockdriftDetector c;
+  // No clockdrift at start.
+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kNone);
+
+  // Monotonically increasing delay.
+  for (int i = 0; i < 100; i++)
+    c.Update(1000);
+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kNone);
+  for (int i = 0; i < 100; i++)
+    c.Update(1001);
+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kNone);
+  for (int i = 0; i < 100; i++)
+    c.Update(1002);
+  // Probable clockdrift.
+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kProbable);
+  for (int i = 0; i < 100; i++)
+    c.Update(1003);
+  // Verified clockdrift.
+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kVerified);
+
+  // Stable delay.
+  for (int i = 0; i < 10000; i++)
+    c.Update(1003);
+  // No clockdrift.
+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kNone);
+
+  // Decreasing delay.
+  for (int i = 0; i < 100; i++)
+    c.Update(1001);
+  for (int i = 0; i < 100; i++)
+    c.Update(999);
+  // Probable clockdrift.
+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kProbable);
+  for (int i = 0; i < 100; i++)
+    c.Update(1000);
+  for (int i = 0; i < 100; i++)
+    c.Update(998);
+  // Verified clockdrift.
+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kVerified);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain.cc b/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain.cc
new file mode 100644
index 0000000000..f4fb74d20d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/coarse_filter_update_gain.h"
+
+#include <algorithm>
+#include <functional>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+CoarseFilterUpdateGain::CoarseFilterUpdateGain(
+    const EchoCanceller3Config::Filter::CoarseConfiguration& config,
+    size_t config_change_duration_blocks)
+    : config_change_duration_blocks_(
+          static_cast<int>(config_change_duration_blocks)) {
+  SetConfig(config, true);
+  RTC_DCHECK_LT(0, config_change_duration_blocks_);
+  one_by_config_change_duration_blocks_ = 1.f / config_change_duration_blocks_;
+}
+
+void CoarseFilterUpdateGain::HandleEchoPathChange() {
+  poor_signal_excitation_counter_ = 0;
+  call_counter_ = 0;
+}
+
+void CoarseFilterUpdateGain::Compute(
+    const std::array<float, kFftLengthBy2Plus1>& render_power,
+    const RenderSignalAnalyzer& render_signal_analyzer,
+    const FftData& E_coarse,
+    size_t size_partitions,
+    bool saturated_capture_signal,
+    FftData* G) {
+  RTC_DCHECK(G);
+  ++call_counter_;
+
+  UpdateCurrentConfig();
+
+  if (render_signal_analyzer.PoorSignalExcitation()) {
+    poor_signal_excitation_counter_ = 0;
+  }
+
+  // Do not update the filter if the render is not sufficiently excited.
+  if (++poor_signal_excitation_counter_ < size_partitions ||
+      saturated_capture_signal || call_counter_ <= size_partitions) {
+    G->re.fill(0.f);
+    G->im.fill(0.f);
+    return;
+  }
+
+  // Compute mu.
+  std::array<float, kFftLengthBy2Plus1> mu;
+  const auto& X2 = render_power;
+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+    if (X2[k] > current_config_.noise_gate) {
+      mu[k] = current_config_.rate / X2[k];
+    } else {
+      mu[k] = 0.f;
+    }
+  }
+
+  // Avoid updating the filter close to narrow bands in the render signals.
+  render_signal_analyzer.MaskRegionsAroundNarrowBands(&mu);
+
+  // G = mu * E * X2.
+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+    G->re[k] = mu[k] * E_coarse.re[k];
+    G->im[k] = mu[k] * E_coarse.im[k];
+  }
+}
+
+void CoarseFilterUpdateGain::UpdateCurrentConfig() {
+  RTC_DCHECK_GE(config_change_duration_blocks_, config_change_counter_);
+  if (config_change_counter_ > 0) {
+    if (--config_change_counter_ > 0) {
+      auto average = [](float from, float to, float from_weight) {
+        return from * from_weight + to * (1.f - from_weight);
+      };
+
+      float change_factor =
+          config_change_counter_ * one_by_config_change_duration_blocks_;
+
+      current_config_.rate =
+          average(old_target_config_.rate, target_config_.rate, change_factor);
+      current_config_.noise_gate =
+          average(old_target_config_.noise_gate, target_config_.noise_gate,
+                  change_factor);
+    } else {
+      current_config_ = old_target_config_ = target_config_;
+    }
+  }
+  RTC_DCHECK_LE(0, config_change_counter_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain.h b/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain.h
new file mode 100644
index 0000000000..a1a1399b2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_COARSE_FILTER_UPDATE_GAIN_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_COARSE_FILTER_UPDATE_GAIN_H_
+
+#include <stddef.h>
+
+#include <array>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+
+namespace webrtc {
+
+// Provides functionality for computing the fixed gain for the coarse filter.
+class CoarseFilterUpdateGain {
+ public:
+  explicit CoarseFilterUpdateGain(
+      const EchoCanceller3Config::Filter::CoarseConfiguration& config,
+      size_t config_change_duration_blocks);
+
+  // Takes action in the case of a known echo path change.
+  void HandleEchoPathChange();
+
+  // Computes the gain.
+  void Compute(const std::array<float, kFftLengthBy2Plus1>& render_power,
+               const RenderSignalAnalyzer& render_signal_analyzer,
+               const FftData& E_coarse,
+               size_t size_partitions,
+               bool saturated_capture_signal,
+               FftData* G);
+
+  // Sets a new config.
+  void SetConfig(
+      const EchoCanceller3Config::Filter::CoarseConfiguration& config,
+      bool immediate_effect) {
+    if (immediate_effect) {
+      old_target_config_ = current_config_ = target_config_ = config;
+      config_change_counter_ = 0;
+    } else {
+      old_target_config_ = current_config_;
+      target_config_ = config;
+      config_change_counter_ = config_change_duration_blocks_;
+    }
+  }
+
+ private:
+  EchoCanceller3Config::Filter::CoarseConfiguration current_config_;
+  EchoCanceller3Config::Filter::CoarseConfiguration target_config_;
+  EchoCanceller3Config::Filter::CoarseConfiguration old_target_config_;
+  const int config_change_duration_blocks_;
+  float one_by_config_change_duration_blocks_;
+  // TODO(peah): Check whether this counter should instead be initialized to a
+  // large value.
+  size_t poor_signal_excitation_counter_ = 0;
+  size_t call_counter_ = 0;
+  int config_change_counter_ = 0;
+
+  void UpdateCurrentConfig();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_COARSE_FILTER_UPDATE_GAIN_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc
new file mode 100644
index 0000000000..55b79bb812
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/coarse_filter_update_gain_unittest.cc
@@ -0,0 +1,268 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/coarse_filter_update_gain.h"
+
+#include <algorithm>
+#include <memory>
+#include <numeric>
+#include <string>
+#include <vector>
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+// Method for performing the simulations needed to test the refined filter
+// update gain functionality.
+void RunFilterUpdateTest(int num_blocks_to_process,
+                         size_t delay_samples,
+                         size_t num_render_channels,
+                         int filter_length_blocks,
+                         const std::vector<int>& blocks_with_saturation,
+                         std::array<float, kBlockSize>* e_last_block,
+                         std::array<float, kBlockSize>* y_last_block,
+                         FftData* G_last_block) {
+  ApmDataDumper data_dumper(42);
+  EchoCanceller3Config config;
+  config.filter.refined.length_blocks = filter_length_blocks;
+  AdaptiveFirFilter refined_filter(
+      config.filter.refined.length_blocks, config.filter.refined.length_blocks,
+      config.filter.config_change_duration_blocks, num_render_channels,
+      DetectOptimization(), &data_dumper);
+  AdaptiveFirFilter coarse_filter(
+      config.filter.coarse.length_blocks, config.filter.coarse.length_blocks,
+      config.filter.config_change_duration_blocks, num_render_channels,
+      DetectOptimization(), &data_dumper);
+  Aec3Fft fft;
+
+  constexpr int kSampleRateHz = 48000;
+  config.delay.default_delay = 1;
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, num_render_channels));
+
+  CoarseFilterUpdateGain coarse_gain(
+      config.filter.coarse, config.filter.config_change_duration_blocks);
+  Random random_generator(42U);
+  Block x(NumBandsForRate(kSampleRateHz), num_render_channels);
+  std::array<float, kBlockSize> y;
+  RenderSignalAnalyzer render_signal_analyzer(config);
+  std::array<float, kFftLength> s;
+  FftData S;
+  FftData G;
+  FftData E_coarse;
+  std::array<float, kBlockSize> e_coarse;
+
+  constexpr float kScale = 1.0f / kFftLengthBy2;
+
+  DelayBuffer<float> delay_buffer(delay_samples);
+  for (int k = 0; k < num_blocks_to_process; ++k) {
+    // Handle saturation.
+    bool saturation =
+        std::find(blocks_with_saturation.begin(), blocks_with_saturation.end(),
+                  k) != blocks_with_saturation.end();
+
+    // Create the render signal.
+    for (int band = 0; band < x.NumBands(); ++band) {
+      for (int channel = 0; channel < x.NumChannels(); ++channel) {
+        RandomizeSampleVector(&random_generator, x.View(band, channel));
+      }
+    }
+    delay_buffer.Delay(x.View(/*band=*/0, /*channel*/ 0), y);
+
+    render_delay_buffer->Insert(x);
+    if (k == 0) {
+      render_delay_buffer->Reset();
+    }
+    render_delay_buffer->PrepareCaptureProcessing();
+
+    render_signal_analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
+                                  delay_samples / kBlockSize);
+
+    coarse_filter.Filter(*render_delay_buffer->GetRenderBuffer(), &S);
+    fft.Ifft(S, &s);
+    std::transform(y.begin(), y.end(), s.begin() + kFftLengthBy2,
+                   e_coarse.begin(),
+                   [&](float a, float b) { return a - b * kScale; });
+    std::for_each(e_coarse.begin(), e_coarse.end(),
+                  [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
+    fft.ZeroPaddedFft(e_coarse, Aec3Fft::Window::kRectangular, &E_coarse);
+
+    std::array<float, kFftLengthBy2Plus1> render_power;
+    render_delay_buffer->GetRenderBuffer()->SpectralSum(
+        coarse_filter.SizePartitions(), &render_power);
+    coarse_gain.Compute(render_power, render_signal_analyzer, E_coarse,
+                        coarse_filter.SizePartitions(), saturation, &G);
+    coarse_filter.Adapt(*render_delay_buffer->GetRenderBuffer(), G);
+  }
+
+  std::copy(e_coarse.begin(), e_coarse.end(), e_last_block->begin());
+  std::copy(y.begin(), y.end(), y_last_block->begin());
+  std::copy(G.re.begin(), G.re.end(), G_last_block->re.begin());
+  std::copy(G.im.begin(), G.im.end(), G_last_block->im.begin());
+}
+
+std::string ProduceDebugText(int filter_length_blocks) {
+  rtc::StringBuilder ss;
+  ss << "Length: " << filter_length_blocks;
+  return ss.Release();
+}
+
+std::string ProduceDebugText(size_t delay, int filter_length_blocks) {
+  rtc::StringBuilder ss;
+  ss << "Delay: " << delay << ", ";
+  ss << ProduceDebugText(filter_length_blocks);
+  return ss.Release();
+}
+
+}  // namespace
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies that the check for non-null output gain parameter works.
+TEST(CoarseFilterUpdateGainDeathTest, NullDataOutputGain) {
+  ApmDataDumper data_dumper(42);
+  FftBuffer fft_buffer(1, 1);
+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
+  FftData E;
+  const EchoCanceller3Config::Filter::CoarseConfiguration& config = {
+      12, 0.5f, 220075344.f};
+  CoarseFilterUpdateGain gain(config, 250);
+  std::array<float, kFftLengthBy2Plus1> render_power;
+  render_power.fill(0.f);
+  EXPECT_DEATH(gain.Compute(render_power, analyzer, E, 1, false, nullptr), "");
+}
+
+#endif
+
+class CoarseFilterUpdateGainOneTwoEightRenderChannels
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<size_t> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         CoarseFilterUpdateGainOneTwoEightRenderChannels,
+                         ::testing::Values(1, 2, 8));
+
+// Verifies that the gain formed causes the filter using it to converge.
+TEST_P(CoarseFilterUpdateGainOneTwoEightRenderChannels,
+       GainCausesFilterToConverge) {
+  const size_t num_render_channels = GetParam();
+  std::vector<int> blocks_with_echo_path_changes;
+  std::vector<int> blocks_with_saturation;
+
+  for (size_t filter_length_blocks : {12, 20, 30}) {
+    for (size_t delay_samples : {0, 64, 150, 200, 301}) {
+      SCOPED_TRACE(ProduceDebugText(delay_samples, filter_length_blocks));
+
+      std::array<float, kBlockSize> e;
+      std::array<float, kBlockSize> y;
+      FftData G;
+
+      RunFilterUpdateTest(5000, delay_samples, num_render_channels,
+                          filter_length_blocks, blocks_with_saturation, &e, &y,
+                          &G);
+
+      // Verify that the refined filter is able to perform well.
+      // Use different criteria to take overmodelling into account.
+      if (filter_length_blocks == 12) {
+        EXPECT_LT(1000 * std::inner_product(e.begin(), e.end(), e.begin(), 0.f),
+                  std::inner_product(y.begin(), y.end(), y.begin(), 0.f));
+      } else {
+        EXPECT_LT(std::inner_product(e.begin(), e.end(), e.begin(), 0.f),
+                  std::inner_product(y.begin(), y.end(), y.begin(), 0.f));
+      }
+    }
+  }
+}
+
+// Verifies that the gain is zero when there is saturation.
+TEST_P(CoarseFilterUpdateGainOneTwoEightRenderChannels, SaturationBehavior) {
+  const size_t num_render_channels = GetParam();
+  std::vector<int> blocks_with_echo_path_changes;
+  std::vector<int> blocks_with_saturation;
+  for (int k = 99; k < 200; ++k) {
+    blocks_with_saturation.push_back(k);
+  }
+  for (size_t filter_length_blocks : {12, 20, 30}) {
+    SCOPED_TRACE(ProduceDebugText(filter_length_blocks));
+
+    std::array<float, kBlockSize> e;
+    std::array<float, kBlockSize> y;
+    FftData G_a;
+    FftData G_a_ref;
+    G_a_ref.re.fill(0.f);
+    G_a_ref.im.fill(0.f);
+
+    RunFilterUpdateTest(100, 65, num_render_channels, filter_length_blocks,
+                        blocks_with_saturation, &e, &y, &G_a);
+
+    EXPECT_EQ(G_a_ref.re, G_a.re);
+    EXPECT_EQ(G_a_ref.im, G_a.im);
+  }
+}
+
+class CoarseFilterUpdateGainOneTwoFourRenderChannels
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<size_t> {};
+
+INSTANTIATE_TEST_SUITE_P(
+    MultiChannel,
+    CoarseFilterUpdateGainOneTwoFourRenderChannels,
+    ::testing::Values(1, 2, 4),
+    [](const ::testing::TestParamInfo<
+        CoarseFilterUpdateGainOneTwoFourRenderChannels::ParamType>& info) {
+      return (rtc::StringBuilder() << "Render" << info.param).str();
+    });
+
+// Verifies that the magnitude of the gain on average decreases for a
+// persistently exciting signal.
+TEST_P(CoarseFilterUpdateGainOneTwoFourRenderChannels, DecreasingGain) {
+  const size_t num_render_channels = GetParam();
+  for (size_t filter_length_blocks : {12, 20, 30}) {
+    SCOPED_TRACE(ProduceDebugText(filter_length_blocks));
+    std::vector<int> blocks_with_echo_path_changes;
+    std::vector<int> blocks_with_saturation;
+
+    std::array<float, kBlockSize> e;
+    std::array<float, kBlockSize> y;
+    FftData G_a;
+    FftData G_b;
+    FftData G_c;
+    std::array<float, kFftLengthBy2Plus1> G_a_power;
+    std::array<float, kFftLengthBy2Plus1> G_b_power;
+    std::array<float, kFftLengthBy2Plus1> G_c_power;
+
+    RunFilterUpdateTest(100, 65, num_render_channels, filter_length_blocks,
+                        blocks_with_saturation, &e, &y, &G_a);
+    RunFilterUpdateTest(200, 65, num_render_channels, filter_length_blocks,
+                        blocks_with_saturation, &e, &y, &G_b);
+    RunFilterUpdateTest(300, 65, num_render_channels, filter_length_blocks,
+                        blocks_with_saturation, &e, &y, &G_c);
+
+    G_a.Spectrum(Aec3Optimization::kNone, G_a_power);
+    G_b.Spectrum(Aec3Optimization::kNone, G_b_power);
+    G_c.Spectrum(Aec3Optimization::kNone, G_c_power);
+
+    EXPECT_GT(std::accumulate(G_a_power.begin(), G_a_power.end(), 0.),
+              std::accumulate(G_b_power.begin(), G_b_power.end(), 0.));
+
+    EXPECT_GT(std::accumulate(G_b_power.begin(), G_b_power.end(), 0.),
+              std::accumulate(G_c_power.begin(), G_c_power.end(), 0.));
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator.cc
new file mode 100644
index 0000000000..de5227c089
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator.cc
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/comfort_noise_generator.h"
+
+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <cstdint>
+#include <functional>
+#include <numeric>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_processing/aec3/vector_math.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Computes the noise floor value that matches a WGN input of noise_floor_dbfs.
+float GetNoiseFloorFactor(float noise_floor_dbfs) {
+  // kdBfsNormalization = 20.f*log10(32768.f).
+  constexpr float kdBfsNormalization = 90.30899869919436f;
+  return 64.f * powf(10.f, (kdBfsNormalization + noise_floor_dbfs) * 0.1f);
+}
+
+// Table of sqrt(2) * sin(2*pi*i/32).
+constexpr float kSqrt2Sin[32] = {
+    +0.0000000f, +0.2758994f, +0.5411961f, +0.7856950f, +1.0000000f,
+    +1.1758756f, +1.3065630f, +1.3870398f, +1.4142136f, +1.3870398f,
+    +1.3065630f, +1.1758756f, +1.0000000f, +0.7856950f, +0.5411961f,
+    +0.2758994f, +0.0000000f, -0.2758994f, -0.5411961f, -0.7856950f,
+    -1.0000000f, -1.1758756f, -1.3065630f, -1.3870398f, -1.4142136f,
+    -1.3870398f, -1.3065630f, -1.1758756f, -1.0000000f, -0.7856950f,
+    -0.5411961f, -0.2758994f};
+
+void GenerateComfortNoise(Aec3Optimization optimization,
+                          const std::array<float, kFftLengthBy2Plus1>& N2,
+                          uint32_t* seed,
+                          FftData* lower_band_noise,
+                          FftData* upper_band_noise) {
+  FftData* N_low = lower_band_noise;
+  FftData* N_high = upper_band_noise;
+
+  // Compute square root spectrum.
+  std::array<float, kFftLengthBy2Plus1> N;
+  std::copy(N2.begin(), N2.end(), N.begin());
+  aec3::VectorMath(optimization).Sqrt(N);
+
+  // Compute the noise level for the upper bands.
+  constexpr float kOneByNumBands = 1.f / (kFftLengthBy2Plus1 / 2 + 1);
+  constexpr int kFftLengthBy2Plus1By2 = kFftLengthBy2Plus1 / 2;
+  const float high_band_noise_level =
+      std::accumulate(N.begin() + kFftLengthBy2Plus1By2, N.end(), 0.f) *
+      kOneByNumBands;
+
+  // The analysis and synthesis windowing cause loss of power when
+  // cross-fading the noise where frames are completely uncorrelated
+  // (generated with random phase), hence the factor sqrt(2).
+  // This is not the case for the speech signal where the input is overlapping
+  // (strong correlation).
+  N_low->re[0] = N_low->re[kFftLengthBy2] = N_high->re[0] =
+      N_high->re[kFftLengthBy2] = 0.f;
+  for (size_t k = 1; k < kFftLengthBy2; k++) {
+    constexpr int kIndexMask = 32 - 1;
+    // Generate a random 31-bit integer.
+    seed[0] = (seed[0] * 69069 + 1) & (0x80000000 - 1);
+    // Convert to a 5-bit index.
+    int i = seed[0] >> 26;
+
+    // y = sqrt(2) * sin(a)
+    const float x = kSqrt2Sin[i];
+    // x = sqrt(2) * cos(a) = sqrt(2) * sin(a + pi/2)
+    const float y = kSqrt2Sin[(i + 8) & kIndexMask];
+
+    // Form low-frequency noise via spectral shaping.
+    N_low->re[k] = N[k] * x;
+    N_low->im[k] = N[k] * y;
+
+    // Form the high-frequency noise via simple levelling.
+    N_high->re[k] = high_band_noise_level * x;
+    N_high->im[k] = high_band_noise_level * y;
+  }
+}
+
+}  // namespace
+
+ComfortNoiseGenerator::ComfortNoiseGenerator(const EchoCanceller3Config& config,
+                                             Aec3Optimization optimization,
+                                             size_t num_capture_channels)
+    : optimization_(optimization),
+      seed_(42),
+      num_capture_channels_(num_capture_channels),
+      noise_floor_(GetNoiseFloorFactor(config.comfort_noise.noise_floor_dbfs)),
+      N2_initial_(
+          std::make_unique<std::vector<std::array<float, kFftLengthBy2Plus1>>>(
+              num_capture_channels_)),
+      Y2_smoothed_(num_capture_channels_),
+      N2_(num_capture_channels_) {
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    (*N2_initial_)[ch].fill(0.f);
+    Y2_smoothed_[ch].fill(0.f);
+    N2_[ch].fill(1.0e6f);
+  }
+}
+
+ComfortNoiseGenerator::~ComfortNoiseGenerator() = default;
+
+void ComfortNoiseGenerator::Compute(
+    bool saturated_capture,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        capture_spectrum,
+    rtc::ArrayView<FftData> lower_band_noise,
+    rtc::ArrayView<FftData> upper_band_noise) {
+  const auto& Y2 = capture_spectrum;
+
+  if (!saturated_capture) {
+    // Smooth Y2.
+    for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+      std::transform(Y2_smoothed_[ch].begin(), Y2_smoothed_[ch].end(),
+                     Y2[ch].begin(), Y2_smoothed_[ch].begin(),
+                     [](float a, float b) { return a + 0.1f * (b - a); });
+    }
+
+    if (N2_counter_ > 50) {
+      // Update N2 from Y2_smoothed.
+      for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+        std::transform(N2_[ch].begin(), N2_[ch].end(), Y2_smoothed_[ch].begin(),
+                       N2_[ch].begin(), [](float a, float b) {
+                         return b < a ? (0.9f * b + 0.1f * a) * 1.0002f
+                                      : a * 1.0002f;
+                       });
+      }
+    }
+
+    if (N2_initial_) {
+      if (++N2_counter_ == 1000) {
+        N2_initial_.reset();
+      } else {
+        // Compute the N2_initial from N2.
+        for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+          std::transform(N2_[ch].begin(), N2_[ch].end(),
+                         (*N2_initial_)[ch].begin(), (*N2_initial_)[ch].begin(),
+                         [](float a, float b) {
+                           return a > b ? b + 0.001f * (a - b) : a;
+                         });
+        }
+      }
+    }
+
+    for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+      for (auto& n : N2_[ch]) {
+        n = std::max(n, noise_floor_);
+      }
+      if (N2_initial_) {
+        for (auto& n : (*N2_initial_)[ch]) {
+          n = std::max(n, noise_floor_);
+        }
+      }
+    }
+  }
+
+  // Choose N2 estimate to use.
+  const auto& N2 = N2_initial_ ? (*N2_initial_) : N2_;
+
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    GenerateComfortNoise(optimization_, N2[ch], &seed_, &lower_band_noise[ch],
+                         &upper_band_noise[ch]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator.h b/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator.h
new file mode 100644
index 0000000000..2785b765c5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_COMFORT_NOISE_GENERATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_COMFORT_NOISE_GENERATOR_H_
+
+#include <stdint.h>
+
+#include <array>
+#include <memory>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace aec3 {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+
+void EstimateComfortNoise_SSE2(const std::array<float, kFftLengthBy2Plus1>& N2,
+                               uint32_t* seed,
+                               FftData* lower_band_noise,
+                               FftData* upper_band_noise);
+#endif
+void EstimateComfortNoise(const std::array<float, kFftLengthBy2Plus1>& N2,
+                          uint32_t* seed,
+                          FftData* lower_band_noise,
+                          FftData* upper_band_noise);
+
+}  // namespace aec3
+
+// Generates the comfort noise.
+class ComfortNoiseGenerator {
+ public:
+  ComfortNoiseGenerator(const EchoCanceller3Config& config,
+                        Aec3Optimization optimization,
+                        size_t num_capture_channels);
+  ComfortNoiseGenerator() = delete;
+  ~ComfortNoiseGenerator();
+  ComfortNoiseGenerator(const ComfortNoiseGenerator&) = delete;
+
+  // Computes the comfort noise.
+  void Compute(bool saturated_capture,
+               rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+                   capture_spectrum,
+               rtc::ArrayView<FftData> lower_band_noise,
+               rtc::ArrayView<FftData> upper_band_noise);
+
+  // Returns the estimate of the background noise spectrum.
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> NoiseSpectrum()
+      const {
+    return N2_;
+  }
+
+ private:
+  const Aec3Optimization optimization_;
+  uint32_t seed_;
+  const size_t num_capture_channels_;
+  const float noise_floor_;
+  std::unique_ptr<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      N2_initial_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2_smoothed_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> N2_;
+  int N2_counter_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_COMFORT_NOISE_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
new file mode 100644
index 0000000000..a9da17559a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/comfort_noise_generator.h"
+
+#include <algorithm>
+#include <numeric>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "rtc_base/random.h"
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace aec3 {
+namespace {
+
+float Power(const FftData& N) {
+  std::array<float, kFftLengthBy2Plus1> N2;
+  N.Spectrum(Aec3Optimization::kNone, N2);
+  return std::accumulate(N2.begin(), N2.end(), 0.f) / N2.size();
+}
+
+}  // namespace
+
+TEST(ComfortNoiseGenerator, CorrectLevel) {
+  constexpr size_t kNumChannels = 5;
+  EchoCanceller3Config config;
+  ComfortNoiseGenerator cng(config, DetectOptimization(), kNumChannels);
+  AecState aec_state(config, kNumChannels);
+
+  std::vector<std::array<float, kFftLengthBy2Plus1>> N2(kNumChannels);
+  std::vector<FftData> n_lower(kNumChannels);
+  std::vector<FftData> n_upper(kNumChannels);
+
+  for (size_t ch = 0; ch < kNumChannels; ++ch) {
+    N2[ch].fill(1000.f * 1000.f / (ch + 1));
+    n_lower[ch].re.fill(0.f);
+    n_lower[ch].im.fill(0.f);
+    n_upper[ch].re.fill(0.f);
+    n_upper[ch].im.fill(0.f);
+  }
+
+  // Ensure instantaneous updata to nonzero noise.
+  cng.Compute(false, N2, n_lower, n_upper);
+
+  for (size_t ch = 0; ch < kNumChannels; ++ch) {
+    EXPECT_LT(0.f, Power(n_lower[ch]));
+    EXPECT_LT(0.f, Power(n_upper[ch]));
+  }
+
+  for (int k = 0; k < 10000; ++k) {
+    cng.Compute(false, N2, n_lower, n_upper);
+  }
+
+  for (size_t ch = 0; ch < kNumChannels; ++ch) {
+    EXPECT_NEAR(2.f * N2[ch][0], Power(n_lower[ch]), N2[ch][0] / 10.f);
+    EXPECT_NEAR(2.f * N2[ch][0], Power(n_upper[ch]), N2[ch][0] / 10.f);
+  }
+}
+
+}  // namespace aec3
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/config_selector.cc b/third_party/libwebrtc/modules/audio_processing/aec3/config_selector.cc
new file mode 100644
index 0000000000..c55344da79
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/config_selector.cc
@@ -0,0 +1,71 @@
+
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/config_selector.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+// Validates that the mono and the multichannel configs have compatible fields.
+bool CompatibleConfigs(const EchoCanceller3Config& mono_config,
+                       const EchoCanceller3Config& multichannel_config) {
+  if (mono_config.delay.fixed_capture_delay_samples !=
+      multichannel_config.delay.fixed_capture_delay_samples) {
+    return false;
+  }
+  if (mono_config.filter.export_linear_aec_output !=
+      multichannel_config.filter.export_linear_aec_output) {
+    return false;
+  }
+  if (mono_config.filter.high_pass_filter_echo_reference !=
+      multichannel_config.filter.high_pass_filter_echo_reference) {
+    return false;
+  }
+  if (mono_config.multi_channel.detect_stereo_content !=
+      multichannel_config.multi_channel.detect_stereo_content) {
+    return false;
+  }
+  if (mono_config.multi_channel.stereo_detection_timeout_threshold_seconds !=
+      multichannel_config.multi_channel
+          .stereo_detection_timeout_threshold_seconds) {
+    return false;
+  }
+  return true;
+}
+
+}  // namespace
+
+ConfigSelector::ConfigSelector(
+    const EchoCanceller3Config& config,
+    const absl::optional<EchoCanceller3Config>& multichannel_config,
+    int num_render_input_channels)
+    : config_(config), multichannel_config_(multichannel_config) {
+  if (multichannel_config_.has_value()) {
+    RTC_DCHECK(CompatibleConfigs(config_, *multichannel_config_));
+  }
+
+  Update(!config_.multi_channel.detect_stereo_content &&
+         num_render_input_channels > 1);
+
+  RTC_DCHECK(active_config_);
+}
+
+void ConfigSelector::Update(bool multichannel_content) {
+  if (multichannel_content && multichannel_config_.has_value()) {
+    active_config_ = &(*multichannel_config_);
+  } else {
+    active_config_ = &config_;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/config_selector.h b/third_party/libwebrtc/modules/audio_processing/aec3/config_selector.h
new file mode 100644
index 0000000000..3b3f94e5ac
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/config_selector.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_CONFIG_SELECTOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_CONFIG_SELECTOR_H_
+
+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+
+namespace webrtc {
+
+// Selects the config to use.
+class ConfigSelector {
+ public:
+  ConfigSelector(
+      const EchoCanceller3Config& config,
+      const absl::optional<EchoCanceller3Config>& multichannel_config,
+      int num_render_input_channels);
+
+  // Updates the config selection based on the detection of multichannel
+  // content.
+  void Update(bool multichannel_content);
+
+  const EchoCanceller3Config& active_config() const { return *active_config_; }
+
+ private:
+  const EchoCanceller3Config config_;
+  const absl::optional<EchoCanceller3Config> multichannel_config_;
+  const EchoCanceller3Config* active_config_ = nullptr;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_CONFIG_SELECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/config_selector_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/config_selector_unittest.cc
new file mode 100644
index 0000000000..1826bfcace
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/config_selector_unittest.cc
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/config_selector.h"
+
+#include <tuple>
+
+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class ConfigSelectorChannelsAndContentDetection
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<int, bool>> {};
+
+INSTANTIATE_TEST_SUITE_P(ConfigSelectorMultiParameters,
+                         ConfigSelectorChannelsAndContentDetection,
+                         ::testing::Combine(::testing::Values(1, 2, 8),
+                                            ::testing::Values(false, true)));
+
+class ConfigSelectorChannels : public ::testing::Test,
+                               public ::testing::WithParamInterface<int> {};
+
+INSTANTIATE_TEST_SUITE_P(ConfigSelectorMultiParameters,
+                         ConfigSelectorChannels,
+                         ::testing::Values(1, 2, 8));
+
+TEST_P(ConfigSelectorChannelsAndContentDetection,
+       MonoConfigIsSelectedWhenNoMultiChannelConfigPresent) {
+  const auto [num_channels, detect_stereo_content] = GetParam();
+  EchoCanceller3Config config;
+  config.multi_channel.detect_stereo_content = detect_stereo_content;
+  absl::optional<EchoCanceller3Config> multichannel_config;
+
+  config.delay.default_delay = config.delay.default_delay + 1;
+  const size_t custom_delay_value_in_config = config.delay.default_delay;
+
+  ConfigSelector cs(config, multichannel_config,
+                    /*num_render_input_channels=*/num_channels);
+  EXPECT_EQ(cs.active_config().delay.default_delay,
+            custom_delay_value_in_config);
+
+  cs.Update(/*multichannel_content=*/false);
+  EXPECT_EQ(cs.active_config().delay.default_delay,
+            custom_delay_value_in_config);
+
+  cs.Update(/*multichannel_content=*/true);
+  EXPECT_EQ(cs.active_config().delay.default_delay,
+            custom_delay_value_in_config);
+}
+
+TEST_P(ConfigSelectorChannelsAndContentDetection,
+       CorrectInitialConfigIsSelected) {
+  const auto [num_channels, detect_stereo_content] = GetParam();
+  EchoCanceller3Config config;
+  config.multi_channel.detect_stereo_content = detect_stereo_content;
+  absl::optional<EchoCanceller3Config> multichannel_config = config;
+
+  config.delay.default_delay += 1;
+  const size_t custom_delay_value_in_config = config.delay.default_delay;
+  multichannel_config->delay.default_delay += 2;
+  const size_t custom_delay_value_in_multichannel_config =
+      multichannel_config->delay.default_delay;
+
+  ConfigSelector cs(config, multichannel_config,
+                    /*num_render_input_channels=*/num_channels);
+
+  if (num_channels == 1 || detect_stereo_content) {
+    EXPECT_EQ(cs.active_config().delay.default_delay,
+              custom_delay_value_in_config);
+  } else {
+    EXPECT_EQ(cs.active_config().delay.default_delay,
+              custom_delay_value_in_multichannel_config);
+  }
+}
+
+TEST_P(ConfigSelectorChannels, CorrectConfigUpdateBehavior) {
+  const int num_channels = GetParam();
+  EchoCanceller3Config config;
+  config.multi_channel.detect_stereo_content = true;
+  absl::optional<EchoCanceller3Config> multichannel_config = config;
+
+  config.delay.default_delay += 1;
+  const size_t custom_delay_value_in_config = config.delay.default_delay;
+  multichannel_config->delay.default_delay += 2;
+  const size_t custom_delay_value_in_multichannel_config =
+      multichannel_config->delay.default_delay;
+
+  ConfigSelector cs(config, multichannel_config,
+                    /*num_render_input_channels=*/num_channels);
+
+  cs.Update(/*multichannel_content=*/false);
+  EXPECT_EQ(cs.active_config().delay.default_delay,
+            custom_delay_value_in_config);
+
+  if (num_channels == 1) {
+    cs.Update(/*multichannel_content=*/false);
+    EXPECT_EQ(cs.active_config().delay.default_delay,
+              custom_delay_value_in_config);
+  } else {
+    cs.Update(/*multichannel_content=*/true);
+    EXPECT_EQ(cs.active_config().delay.default_delay,
+              custom_delay_value_in_multichannel_config);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/decimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/decimator.cc
new file mode 100644
index 0000000000..bd03237ca0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/decimator.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/decimator.h"
+
+#include <array>
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+// signal.butter(2, 3400/8000.0, 'lowpass', analog=False)
+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetLowPassFilterDS2() {
+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{
+      {{-1.f, 0.f}, {0.13833231f, 0.40743176f}, 0.22711796393486466f},
+      {{-1.f, 0.f}, {0.13833231f, 0.40743176f}, 0.22711796393486466f},
+      {{-1.f, 0.f}, {0.13833231f, 0.40743176f}, 0.22711796393486466f}};
+}
+
+// signal.ellip(6, 1, 40, 1800/8000, btype='lowpass', analog=False)
+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetLowPassFilterDS4() {
+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{
+      {{-0.08873842f, 0.99605496f}, {0.75916227f, 0.23841065f}, 0.26250696827f},
+      {{0.62273832f, 0.78243018f}, {0.74892112f, 0.5410152f}, 0.26250696827f},
+      {{0.71107693f, 0.70311421f}, {0.74895534f, 0.63924616f}, 0.26250696827f}};
+}
+
+// signal.cheby1(1, 6, [1000/8000, 2000/8000], btype='bandpass', analog=False)
+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetBandPassFilterDS8() {
+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{
+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true},
+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true},
+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true},
+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true},
+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true}};
+}
+
+// signal.butter(2, 1000/8000.0, 'highpass', analog=False)
+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetHighPassFilter() {
+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{
+      {{1.f, 0.f}, {0.72712179f, 0.21296904f}, 0.7570763753338849f}};
+}
+
+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetPassThroughFilter() {
+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{};
+}
+}  // namespace
+
+Decimator::Decimator(size_t down_sampling_factor)
+    : down_sampling_factor_(down_sampling_factor),
+      anti_aliasing_filter_(down_sampling_factor_ == 4
+                                ? GetLowPassFilterDS4()
+                                : (down_sampling_factor_ == 8
+                                       ? GetBandPassFilterDS8()
+                                       : GetLowPassFilterDS2())),
+      noise_reduction_filter_(down_sampling_factor_ == 8
+                                  ? GetPassThroughFilter()
+                                  : GetHighPassFilter()) {
+  RTC_DCHECK(down_sampling_factor_ == 2 || down_sampling_factor_ == 4 ||
+             down_sampling_factor_ == 8);
+}
+
+void Decimator::Decimate(rtc::ArrayView<const float> in,
+                         rtc::ArrayView<float> out) {
+  RTC_DCHECK_EQ(kBlockSize, in.size());
+  RTC_DCHECK_EQ(kBlockSize / down_sampling_factor_, out.size());
+  std::array<float, kBlockSize> x;
+
+  // Limit the frequency content of the signal to avoid aliasing.
+  anti_aliasing_filter_.Process(in, x);
+
+  // Reduce the impact of near-end noise.
+  noise_reduction_filter_.Process(x);
+
+  // Downsample the signal.
+  for (size_t j = 0, k = 0; j < out.size(); ++j, k += down_sampling_factor_) {
+    RTC_DCHECK_GT(kBlockSize, k);
+    out[j] = x[k];
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/decimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/decimator.h
new file mode 100644
index 0000000000..dbff3d9fff
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/decimator.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/utility/cascaded_biquad_filter.h"
+
+namespace webrtc {
+
+// Provides functionality for decimating a signal.
+class Decimator {
+ public:
+  explicit Decimator(size_t down_sampling_factor);
+
+  Decimator(const Decimator&) = delete;
+  Decimator& operator=(const Decimator&) = delete;
+
+  // Downsamples the signal.
+  void Decimate(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);
+
+ private:
+  const size_t down_sampling_factor_;
+  CascadedBiQuadFilter anti_aliasing_filter_;
+  CascadedBiQuadFilter noise_reduction_filter_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/decimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/decimator_unittest.cc
new file mode 100644
index 0000000000..e6f5ea0403
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/decimator_unittest.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/decimator.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <cstring>
+#include <numeric>
+#include <string>
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+std::string ProduceDebugText(int sample_rate_hz) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz;
+  return ss.Release();
+}
+
+constexpr size_t kDownSamplingFactors[] = {2, 4, 8};
+constexpr float kPi = 3.141592f;
+constexpr size_t kNumStartupBlocks = 50;
+constexpr size_t kNumBlocks = 1000;
+
+void ProduceDecimatedSinusoidalOutputPower(int sample_rate_hz,
+                                           size_t down_sampling_factor,
+                                           float sinusoidal_frequency_hz,
+                                           float* input_power,
+                                           float* output_power) {
+  float input[kBlockSize * kNumBlocks];
+  const size_t sub_block_size = kBlockSize / down_sampling_factor;
+
+  // Produce a sinusoid of the specified frequency.
+  for (size_t k = 0; k < kBlockSize * kNumBlocks; ++k) {
+    input[k] = 32767.f * std::sin(2.f * kPi * sinusoidal_frequency_hz * k /
+                                  sample_rate_hz);
+  }
+
+  Decimator decimator(down_sampling_factor);
+  std::vector<float> output(sub_block_size * kNumBlocks);
+
+  for (size_t k = 0; k < kNumBlocks; ++k) {
+    std::vector<float> sub_block(sub_block_size);
+    decimator.Decimate(
+        rtc::ArrayView<const float>(&input[k * kBlockSize], kBlockSize),
+        sub_block);
+
+    std::copy(sub_block.begin(), sub_block.end(),
+              output.begin() + k * sub_block_size);
+  }
+
+  ASSERT_GT(kNumBlocks, kNumStartupBlocks);
+  rtc::ArrayView<const float> input_to_evaluate(
+      &input[kNumStartupBlocks * kBlockSize],
+      (kNumBlocks - kNumStartupBlocks) * kBlockSize);
+  rtc::ArrayView<const float> output_to_evaluate(
+      &output[kNumStartupBlocks * sub_block_size],
+      (kNumBlocks - kNumStartupBlocks) * sub_block_size);
+  *input_power =
+      std::inner_product(input_to_evaluate.begin(), input_to_evaluate.end(),
+                         input_to_evaluate.begin(), 0.f) /
+      input_to_evaluate.size();
+  *output_power =
+      std::inner_product(output_to_evaluate.begin(), output_to_evaluate.end(),
+                         output_to_evaluate.begin(), 0.f) /
+      output_to_evaluate.size();
+}
+
+}  // namespace
+
+// Verifies that there is little aliasing from upper frequencies in the
+// downsampling.
+TEST(Decimator, NoLeakageFromUpperFrequencies) {
+  float input_power;
+  float output_power;
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto down_sampling_factor : kDownSamplingFactors) {
+      ProduceDebugText(rate);
+      ProduceDecimatedSinusoidalOutputPower(rate, down_sampling_factor,
+                                            3.f / 8.f * rate, &input_power,
+                                            &output_power);
+      EXPECT_GT(0.0001f * input_power, output_power);
+    }
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Verifies the check for the input size.
+TEST(DecimatorDeathTest, WrongInputSize) {
+  Decimator decimator(4);
+  std::vector<float> x(kBlockSize - 1, 0.f);
+  std::array<float, kBlockSize / 4> x_downsampled;
+  EXPECT_DEATH(decimator.Decimate(x, x_downsampled), "");
+}
+
+// Verifies the check for non-null output parameter.
+TEST(DecimatorDeathTest, NullOutput) {
+  Decimator decimator(4);
+  std::vector<float> x(kBlockSize, 0.f);
+  EXPECT_DEATH(decimator.Decimate(x, nullptr), "");
+}
+
+// Verifies the check for the output size.
+TEST(DecimatorDeathTest, WrongOutputSize) {
+  Decimator decimator(4);
+  std::vector<float> x(kBlockSize, 0.f);
+  std::array<float, kBlockSize / 4 - 1> x_downsampled;
+  EXPECT_DEATH(decimator.Decimate(x, x_downsampled), "");
+}
+
+// Verifies the check for the correct downsampling factor.
+TEST(DecimatorDeathTest, CorrectDownSamplingFactor) {
+  EXPECT_DEATH(Decimator(3), "");
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/delay_estimate.h b/third_party/libwebrtc/modules/audio_processing/aec3/delay_estimate.h
new file mode 100644
index 0000000000..7838a0c255
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/delay_estimate.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_DELAY_ESTIMATE_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_DELAY_ESTIMATE_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+
+// Stores delay_estimates.
+struct DelayEstimate {
+  enum class Quality { kCoarse, kRefined };
+
+  DelayEstimate(Quality quality, size_t delay)
+      : quality(quality), delay(delay) {}
+
+  Quality quality;
+  size_t delay;
+  size_t blocks_since_last_change = 0;
+  size_t blocks_since_last_update = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_DELAY_ESTIMATE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/dominant_nearend_detector.cc b/third_party/libwebrtc/modules/audio_processing/aec3/dominant_nearend_detector.cc
new file mode 100644
index 0000000000..40073cf615
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/dominant_nearend_detector.cc
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/dominant_nearend_detector.h"
+
+#include <numeric>
+
+namespace webrtc {
+DominantNearendDetector::DominantNearendDetector(
+    const EchoCanceller3Config::Suppressor::DominantNearendDetection& config,
+    size_t num_capture_channels)
+    : enr_threshold_(config.enr_threshold),
+      enr_exit_threshold_(config.enr_exit_threshold),
+      snr_threshold_(config.snr_threshold),
+      hold_duration_(config.hold_duration),
+      trigger_threshold_(config.trigger_threshold),
+      use_during_initial_phase_(config.use_during_initial_phase),
+      num_capture_channels_(num_capture_channels),
+      trigger_counters_(num_capture_channels_),
+      hold_counters_(num_capture_channels_) {}
+
+void DominantNearendDetector::Update(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        nearend_spectrum,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        residual_echo_spectrum,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        comfort_noise_spectrum,
+    bool initial_state) {
+  nearend_state_ = false;
+
+  auto low_frequency_energy = [](rtc::ArrayView<const float> spectrum) {
+    RTC_DCHECK_LE(16, spectrum.size());
+    return std::accumulate(spectrum.begin() + 1, spectrum.begin() + 16, 0.f);
+  };
+
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    const float ne_sum = low_frequency_energy(nearend_spectrum[ch]);
+    const float echo_sum = low_frequency_energy(residual_echo_spectrum[ch]);
+    const float noise_sum = low_frequency_energy(comfort_noise_spectrum[ch]);
+
+    // Detect strong active nearend if the nearend is sufficiently stronger than
+    // the echo and the nearend noise.
+    if ((!initial_state || use_during_initial_phase_) &&
+        echo_sum < enr_threshold_ * ne_sum &&
+        ne_sum > snr_threshold_ * noise_sum) {
+      if (++trigger_counters_[ch] >= trigger_threshold_) {
+        // After a period of strong active nearend activity, flag nearend mode.
+        hold_counters_[ch] = hold_duration_;
+        trigger_counters_[ch] = trigger_threshold_;
+      }
+    } else {
+      // Forget previously detected strong active nearend activity.
+      trigger_counters_[ch] = std::max(0, trigger_counters_[ch] - 1);
+    }
+
+    // Exit nearend-state early at strong echo.
+    if (echo_sum > enr_exit_threshold_ * ne_sum &&
+        echo_sum > snr_threshold_ * noise_sum) {
+      hold_counters_[ch] = 0;
+    }
+
+    // Remain in any nearend mode for a certain duration.
+    hold_counters_[ch] = std::max(0, hold_counters_[ch] - 1);
+    nearend_state_ = nearend_state_ || hold_counters_[ch] > 0;
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/dominant_nearend_detector.h b/third_party/libwebrtc/modules/audio_processing/aec3/dominant_nearend_detector.h
new file mode 100644
index 0000000000..046d1488d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/dominant_nearend_detector.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_DOMINANT_NEAREND_DETECTOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_DOMINANT_NEAREND_DETECTOR_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/nearend_detector.h"
+
+namespace webrtc {
+// Class for selecting whether the suppressor is in the nearend or echo state.
+class DominantNearendDetector : public NearendDetector {
+ public:
+  DominantNearendDetector(
+      const EchoCanceller3Config::Suppressor::DominantNearendDetection& config,
+      size_t num_capture_channels);
+
+  // Returns whether the current state is the nearend state.
+  bool IsNearendState() const override { return nearend_state_; }
+
+  // Updates the state selection based on latest spectral estimates.
+  void Update(rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+                  nearend_spectrum,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+                  residual_echo_spectrum,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+                  comfort_noise_spectrum,
+              bool initial_state) override;
+
+ private:
+  const float enr_threshold_;
+  const float enr_exit_threshold_;
+  const float snr_threshold_;
+  const int hold_duration_;
+  const int trigger_threshold_;
+  const bool use_during_initial_phase_;
+  const size_t num_capture_channels_;
+
+  bool nearend_state_ = false;
+  std::vector<int> trigger_counters_;
+  std::vector<int> hold_counters_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_DOMINANT_NEAREND_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc
new file mode 100644
index 0000000000..c105911aa8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+
+#include <algorithm>
+
+namespace webrtc {
+
+DownsampledRenderBuffer::DownsampledRenderBuffer(size_t downsampled_buffer_size)
+    : size(static_cast<int>(downsampled_buffer_size)),
+      buffer(downsampled_buffer_size, 0.f) {
+  std::fill(buffer.begin(), buffer.end(), 0.f);
+}
+
+DownsampledRenderBuffer::~DownsampledRenderBuffer() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.h
new file mode 100644
index 0000000000..fbdc9b4e93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Holds the circular buffer of the downsampled render data.
+struct DownsampledRenderBuffer {
+  explicit DownsampledRenderBuffer(size_t downsampled_buffer_size);
+  ~DownsampledRenderBuffer();
+
+  int IncIndex(int index) const {
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return index < size - 1 ? index + 1 : 0;
+  }
+
+  int DecIndex(int index) const {
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return index > 0 ? index - 1 : size - 1;
+  }
+
+  int OffsetIndex(int index, int offset) const {
+    RTC_DCHECK_GE(buffer.size(), offset);
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return (size + index + offset) % size;
+  }
+
+  void UpdateWriteIndex(int offset) { write = OffsetIndex(write, offset); }
+  void IncWriteIndex() { write = IncIndex(write); }
+  void DecWriteIndex() { write = DecIndex(write); }
+  void UpdateReadIndex(int offset) { read = OffsetIndex(read, offset); }
+  void IncReadIndex() { read = IncIndex(read); }
+  void DecReadIndex() { read = DecIndex(read); }
+
+  const int size;
+  std::vector<float> buffer;
+  int write = 0;
+  int read = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_audibility.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_audibility.cc
new file mode 100644
index 0000000000..142a33d5e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_audibility.cc
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/echo_audibility.h"
+
+#include <algorithm>
+#include <cmath>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/block_buffer.h"
+#include "modules/audio_processing/aec3/spectrum_buffer.h"
+#include "modules/audio_processing/aec3/stationarity_estimator.h"
+
+namespace webrtc {
+
+EchoAudibility::EchoAudibility(bool use_render_stationarity_at_init)
+    : use_render_stationarity_at_init_(use_render_stationarity_at_init) {
+  Reset();
+}
+
+EchoAudibility::~EchoAudibility() = default;
+
+void EchoAudibility::Update(const RenderBuffer& render_buffer,
+                            rtc::ArrayView<const float> average_reverb,
+                            int delay_blocks,
+                            bool external_delay_seen) {
+  UpdateRenderNoiseEstimator(render_buffer.GetSpectrumBuffer(),
+                             render_buffer.GetBlockBuffer(),
+                             external_delay_seen);
+
+  if (external_delay_seen || use_render_stationarity_at_init_) {
+    UpdateRenderStationarityFlags(render_buffer, average_reverb, delay_blocks);
+  }
+}
+
+void EchoAudibility::Reset() {
+  render_stationarity_.Reset();
+  non_zero_render_seen_ = false;
+  render_spectrum_write_prev_ = absl::nullopt;
+}
+
+void EchoAudibility::UpdateRenderStationarityFlags(
+    const RenderBuffer& render_buffer,
+    rtc::ArrayView<const float> average_reverb,
+    int min_channel_delay_blocks) {
+  const SpectrumBuffer& spectrum_buffer = render_buffer.GetSpectrumBuffer();
+  int idx_at_delay = spectrum_buffer.OffsetIndex(spectrum_buffer.read,
+                                                 min_channel_delay_blocks);
+
+  int num_lookahead = render_buffer.Headroom() - min_channel_delay_blocks + 1;
+  num_lookahead = std::max(0, num_lookahead);
+
+  render_stationarity_.UpdateStationarityFlags(spectrum_buffer, average_reverb,
+                                               idx_at_delay, num_lookahead);
+}
+
+void EchoAudibility::UpdateRenderNoiseEstimator(
+    const SpectrumBuffer& spectrum_buffer,
+    const BlockBuffer& block_buffer,
+    bool external_delay_seen) {
+  if (!render_spectrum_write_prev_) {
+    render_spectrum_write_prev_ = spectrum_buffer.write;
+    render_block_write_prev_ = block_buffer.write;
+    return;
+  }
+  int render_spectrum_write_current = spectrum_buffer.write;
+  if (!non_zero_render_seen_ && !external_delay_seen) {
+    non_zero_render_seen_ = !IsRenderTooLow(block_buffer);
+  }
+  if (non_zero_render_seen_) {
+    for (int idx = render_spectrum_write_prev_.value();
+         idx != render_spectrum_write_current;
+         idx = spectrum_buffer.DecIndex(idx)) {
+      render_stationarity_.UpdateNoiseEstimator(spectrum_buffer.buffer[idx]);
+    }
+  }
+  render_spectrum_write_prev_ = render_spectrum_write_current;
+}
+
+bool EchoAudibility::IsRenderTooLow(const BlockBuffer& block_buffer) {
+  const int num_render_channels =
+      static_cast<int>(block_buffer.buffer[0].NumChannels());
+  bool too_low = false;
+  const int render_block_write_current = block_buffer.write;
+  if (render_block_write_current == render_block_write_prev_) {
+    too_low = true;
+  } else {
+    for (int idx = render_block_write_prev_; idx != render_block_write_current;
+         idx = block_buffer.IncIndex(idx)) {
+      float max_abs_over_channels = 0.f;
+      for (int ch = 0; ch < num_render_channels; ++ch) {
+        rtc::ArrayView<const float, kBlockSize> block =
+            block_buffer.buffer[idx].View(/*band=*/0, /*channel=*/ch);
+        auto r = std::minmax_element(block.cbegin(), block.cend());
+        float max_abs_channel =
+            std::max(std::fabs(*r.first), std::fabs(*r.second));
+        max_abs_over_channels =
+            std::max(max_abs_over_channels, max_abs_channel);
+      }
+      if (max_abs_over_channels < 10.f) {
+        too_low = true;  // Discards all blocks if one of them is too low.
+        break;
+      }
+    }
+  }
+  render_block_write_prev_ = render_block_write_current;
+  return too_low;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_audibility.h b/third_party/libwebrtc/modules/audio_processing/aec3/echo_audibility.h
new file mode 100644
index 0000000000..b9d6f87d2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_audibility.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/block_buffer.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/spectrum_buffer.h"
+#include "modules/audio_processing/aec3/stationarity_estimator.h"
+
+namespace webrtc {
+
+class EchoAudibility {
+ public:
+  explicit EchoAudibility(bool use_render_stationarity_at_init);
+  ~EchoAudibility();
+
+  EchoAudibility(const EchoAudibility&) = delete;
+  EchoAudibility& operator=(const EchoAudibility&) = delete;
+
+  // Feed new render data to the echo audibility estimator.
+  void Update(const RenderBuffer& render_buffer,
+              rtc::ArrayView<const float> average_reverb,
+              int min_channel_delay_blocks,
+              bool external_delay_seen);
+  // Get the residual echo scaling.
+  void GetResidualEchoScaling(bool filter_has_had_time_to_converge,
+                              rtc::ArrayView<float> residual_scaling) const {
+    for (size_t band = 0; band < residual_scaling.size(); ++band) {
+      if (render_stationarity_.IsBandStationary(band) &&
+          (filter_has_had_time_to_converge ||
+           use_render_stationarity_at_init_)) {
+        residual_scaling[band] = 0.f;
+      } else {
+        residual_scaling[band] = 1.0f;
+      }
+    }
+  }
+
+  // Returns true if the current render block is estimated as stationary.
+  bool IsBlockStationary() const {
+    return render_stationarity_.IsBlockStationary();
+  }
+
+ private:
+  // Reset the EchoAudibility class.
+  void Reset();
+
+  // Updates the render stationarity flags for the current frame.
+  void UpdateRenderStationarityFlags(const RenderBuffer& render_buffer,
+                                     rtc::ArrayView<const float> average_reverb,
+                                     int delay_blocks);
+
+  // Updates the noise estimator with the new render data since the previous
+  // call to this method.
+  void UpdateRenderNoiseEstimator(const SpectrumBuffer& spectrum_buffer,
+                                  const BlockBuffer& block_buffer,
+                                  bool external_delay_seen);
+
+  // Returns a bool being true if the render signal contains just close to zero
+  // values.
+  bool IsRenderTooLow(const BlockBuffer& block_buffer);
+
+  absl::optional<int> render_spectrum_write_prev_;
+  int render_block_write_prev_;
+  bool non_zero_render_seen_;
+  const bool use_render_stationarity_at_init_;
+  StationarityEstimator render_stationarity_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3.cc
new file mode 100644
index 0000000000..e8e2175994
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3.cc
@@ -0,0 +1,992 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/echo_canceller3.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/high_pass_filter.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+enum class EchoCanceller3ApiCall { kCapture, kRender };
+
+bool DetectSaturation(rtc::ArrayView<const float> y) {
+  for (size_t k = 0; k < y.size(); ++k) {
+    if (y[k] >= 32700.0f || y[k] <= -32700.0f) {
+      return true;
+    }
+  }
+  return false;
+}
+
+// Retrieves a value from a field trial if it is available. If no value is
+// present, the default value is returned. If the retrieved value is beyond the
+// specified limits, the default value is returned instead.
+void RetrieveFieldTrialValue(absl::string_view trial_name,
+                             float min,
+                             float max,
+                             float* value_to_update) {
+  const std::string field_trial_str = field_trial::FindFullName(trial_name);
+
+  FieldTrialParameter<double> field_trial_param(/*key=*/"", *value_to_update);
+
+  ParseFieldTrial({&field_trial_param}, field_trial_str);
+  float field_trial_value = static_cast<float>(field_trial_param.Get());
+
+  if (field_trial_value >= min && field_trial_value <= max &&
+      field_trial_value != *value_to_update) {
+    RTC_LOG(LS_INFO) << "Key " << trial_name
+                     << " changing AEC3 parameter value from "
+                     << *value_to_update << " to " << field_trial_value;
+    *value_to_update = field_trial_value;
+  }
+}
+
+void RetrieveFieldTrialValue(absl::string_view trial_name,
+                             int min,
+                             int max,
+                             int* value_to_update) {
+  const std::string field_trial_str = field_trial::FindFullName(trial_name);
+
+  FieldTrialParameter<int> field_trial_param(/*key=*/"", *value_to_update);
+
+  ParseFieldTrial({&field_trial_param}, field_trial_str);
+  float field_trial_value = field_trial_param.Get();
+
+  if (field_trial_value >= min && field_trial_value <= max &&
+      field_trial_value != *value_to_update) {
+    RTC_LOG(LS_INFO) << "Key " << trial_name
+                     << " changing AEC3 parameter value from "
+                     << *value_to_update << " to " << field_trial_value;
+    *value_to_update = field_trial_value;
+  }
+}
+
+void FillSubFrameView(
+    AudioBuffer* frame,
+    size_t sub_frame_index,
+    std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
+  RTC_DCHECK_GE(1, sub_frame_index);
+  RTC_DCHECK_LE(0, sub_frame_index);
+  RTC_DCHECK_EQ(frame->num_bands(), sub_frame_view->size());
+  RTC_DCHECK_EQ(frame->num_channels(), (*sub_frame_view)[0].size());
+  for (size_t band = 0; band < sub_frame_view->size(); ++band) {
+    for (size_t channel = 0; channel < (*sub_frame_view)[0].size(); ++channel) {
+      (*sub_frame_view)[band][channel] = rtc::ArrayView<float>(
+          &frame->split_bands(channel)[band][sub_frame_index * kSubFrameLength],
+          kSubFrameLength);
+    }
+  }
+}
+
+void FillSubFrameView(
+    bool proper_downmix_needed,
+    std::vector<std::vector<std::vector<float>>>* frame,
+    size_t sub_frame_index,
+    std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
+  RTC_DCHECK_GE(1, sub_frame_index);
+  RTC_DCHECK_EQ(frame->size(), sub_frame_view->size());
+  const size_t frame_num_channels = (*frame)[0].size();
+  const size_t sub_frame_num_channels = (*sub_frame_view)[0].size();
+  if (frame_num_channels > sub_frame_num_channels) {
+    RTC_DCHECK_EQ(sub_frame_num_channels, 1u);
+    if (proper_downmix_needed) {
+      // When a proper downmix is needed (which is the case when proper stereo
+      // is present in the echo reference signal but the echo canceller does the
+      // processing in mono) downmix the echo reference by averaging the channel
+      // content (otherwise downmixing is done by selecting channel 0).
+      for (size_t band = 0; band < frame->size(); ++band) {
+        for (size_t ch = 1; ch < frame_num_channels; ++ch) {
+          for (size_t k = 0; k < kSubFrameLength; ++k) {
+            (*frame)[band][/*channel=*/0]
+                    [sub_frame_index * kSubFrameLength + k] +=
+                (*frame)[band][ch][sub_frame_index * kSubFrameLength + k];
+          }
+        }
+        const float one_by_num_channels = 1.0f / frame_num_channels;
+        for (size_t k = 0; k < kSubFrameLength; ++k) {
+          (*frame)[band][/*channel=*/0][sub_frame_index * kSubFrameLength +
+                                        k] *= one_by_num_channels;
+        }
+      }
+    }
+    for (size_t band = 0; band < frame->size(); ++band) {
+      (*sub_frame_view)[band][/*channel=*/0] = rtc::ArrayView<float>(
+          &(*frame)[band][/*channel=*/0][sub_frame_index * kSubFrameLength],
+          kSubFrameLength);
+    }
+  } else {
+    RTC_DCHECK_EQ(frame_num_channels, sub_frame_num_channels);
+    for (size_t band = 0; band < frame->size(); ++band) {
+      for (size_t channel = 0; channel < (*frame)[band].size(); ++channel) {
+        (*sub_frame_view)[band][channel] = rtc::ArrayView<float>(
+            &(*frame)[band][channel][sub_frame_index * kSubFrameLength],
+            kSubFrameLength);
+      }
+    }
+  }
+}
+
+void ProcessCaptureFrameContent(
+    AudioBuffer* linear_output,
+    AudioBuffer* capture,
+    bool level_change,
+    bool aec_reference_is_downmixed_stereo,
+    bool saturated_microphone_signal,
+    size_t sub_frame_index,
+    FrameBlocker* capture_blocker,
+    BlockFramer* linear_output_framer,
+    BlockFramer* output_framer,
+    BlockProcessor* block_processor,
+    Block* linear_output_block,
+    std::vector<std::vector<rtc::ArrayView<float>>>*
+        linear_output_sub_frame_view,
+    Block* capture_block,
+    std::vector<std::vector<rtc::ArrayView<float>>>* capture_sub_frame_view) {
+  FillSubFrameView(capture, sub_frame_index, capture_sub_frame_view);
+
+  if (linear_output) {
+    RTC_DCHECK(linear_output_framer);
+    RTC_DCHECK(linear_output_block);
+    RTC_DCHECK(linear_output_sub_frame_view);
+    FillSubFrameView(linear_output, sub_frame_index,
+                     linear_output_sub_frame_view);
+  }
+
+  capture_blocker->InsertSubFrameAndExtractBlock(*capture_sub_frame_view,
+                                                 capture_block);
+  block_processor->ProcessCapture(
+      /*echo_path_gain_change=*/level_change ||
+          aec_reference_is_downmixed_stereo,
+      saturated_microphone_signal, linear_output_block, capture_block);
+  output_framer->InsertBlockAndExtractSubFrame(*capture_block,
+                                               capture_sub_frame_view);
+
+  if (linear_output) {
+    RTC_DCHECK(linear_output_framer);
+    linear_output_framer->InsertBlockAndExtractSubFrame(
+        *linear_output_block, linear_output_sub_frame_view);
+  }
+}
+
+void ProcessRemainingCaptureFrameContent(bool level_change,
+                                         bool aec_reference_is_downmixed_stereo,
+                                         bool saturated_microphone_signal,
+                                         FrameBlocker* capture_blocker,
+                                         BlockFramer* linear_output_framer,
+                                         BlockFramer* output_framer,
+                                         BlockProcessor* block_processor,
+                                         Block* linear_output_block,
+                                         Block* block) {
+  if (!capture_blocker->IsBlockAvailable()) {
+    return;
+  }
+
+  capture_blocker->ExtractBlock(block);
+  block_processor->ProcessCapture(
+      /*echo_path_gain_change=*/level_change ||
+          aec_reference_is_downmixed_stereo,
+      saturated_microphone_signal, linear_output_block, block);
+  output_framer->InsertBlock(*block);
+
+  if (linear_output_framer) {
+    RTC_DCHECK(linear_output_block);
+    linear_output_framer->InsertBlock(*linear_output_block);
+  }
+}
+
+void BufferRenderFrameContent(
+    bool proper_downmix_needed,
+    std::vector<std::vector<std::vector<float>>>* render_frame,
+    size_t sub_frame_index,
+    FrameBlocker* render_blocker,
+    BlockProcessor* block_processor,
+    Block* block,
+    std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
+  FillSubFrameView(proper_downmix_needed, render_frame, sub_frame_index,
+                   sub_frame_view);
+  render_blocker->InsertSubFrameAndExtractBlock(*sub_frame_view, block);
+  block_processor->BufferRender(*block);
+}
+
+void BufferRemainingRenderFrameContent(FrameBlocker* render_blocker,
+                                       BlockProcessor* block_processor,
+                                       Block* block) {
+  if (!render_blocker->IsBlockAvailable()) {
+    return;
+  }
+  render_blocker->ExtractBlock(block);
+  block_processor->BufferRender(*block);
+}
+
+void CopyBufferIntoFrame(const AudioBuffer& buffer,
+                         size_t num_bands,
+                         size_t num_channels,
+                         std::vector<std::vector<std::vector<float>>>* frame) {
+  RTC_DCHECK_EQ(num_bands, frame->size());
+  RTC_DCHECK_EQ(num_channels, (*frame)[0].size());
+  RTC_DCHECK_EQ(AudioBuffer::kSplitBandSize, (*frame)[0][0].size());
+  for (size_t band = 0; band < num_bands; ++band) {
+    for (size_t channel = 0; channel < num_channels; ++channel) {
+      rtc::ArrayView<const float> buffer_view(
+          &buffer.split_bands_const(channel)[band][0],
+          AudioBuffer::kSplitBandSize);
+      std::copy(buffer_view.begin(), buffer_view.end(),
+                (*frame)[band][channel].begin());
+    }
+  }
+}
+
+}  // namespace
+
+// TODO(webrtc:5298): Move this to a separate file.
+EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
+  EchoCanceller3Config adjusted_cfg = config;
+
+  if (field_trial::IsEnabled("WebRTC-Aec3StereoContentDetectionKillSwitch")) {
+    adjusted_cfg.multi_channel.detect_stereo_content = false;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3AntiHowlingMinimizationKillSwitch")) {
+    adjusted_cfg.suppressor.high_bands_suppression
+        .anti_howling_activation_threshold = 25.f;
+    adjusted_cfg.suppressor.high_bands_suppression.anti_howling_gain = 0.01f;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3UseShortConfigChangeDuration")) {
+    adjusted_cfg.filter.config_change_duration_blocks = 10;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3UseZeroInitialStateDuration")) {
+    adjusted_cfg.filter.initial_state_seconds = 0.f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3UseDot1SecondsInitialStateDuration")) {
+    adjusted_cfg.filter.initial_state_seconds = .1f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3UseDot2SecondsInitialStateDuration")) {
+    adjusted_cfg.filter.initial_state_seconds = .2f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3UseDot3SecondsInitialStateDuration")) {
+    adjusted_cfg.filter.initial_state_seconds = .3f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3UseDot6SecondsInitialStateDuration")) {
+    adjusted_cfg.filter.initial_state_seconds = .6f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3UseDot9SecondsInitialStateDuration")) {
+    adjusted_cfg.filter.initial_state_seconds = .9f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3Use1Dot2SecondsInitialStateDuration")) {
+    adjusted_cfg.filter.initial_state_seconds = 1.2f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3Use1Dot6SecondsInitialStateDuration")) {
+    adjusted_cfg.filter.initial_state_seconds = 1.6f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3Use2Dot0SecondsInitialStateDuration")) {
+    adjusted_cfg.filter.initial_state_seconds = 2.0f;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3HighPassFilterEchoReference")) {
+    adjusted_cfg.filter.high_pass_filter_echo_reference = true;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3EchoSaturationDetectionKillSwitch")) {
+    adjusted_cfg.ep_strength.echo_can_saturate = false;
+  }
+
+  const std::string use_nearend_reverb_len_tunings =
+      field_trial::FindFullName("WebRTC-Aec3UseNearendReverbLen");
+  FieldTrialParameter<double> nearend_reverb_default_len(
+      "default_len", adjusted_cfg.ep_strength.default_len);
+  FieldTrialParameter<double> nearend_reverb_nearend_len(
+      "nearend_len", adjusted_cfg.ep_strength.nearend_len);
+
+  ParseFieldTrial({&nearend_reverb_default_len, &nearend_reverb_nearend_len},
+                  use_nearend_reverb_len_tunings);
+  float default_len = static_cast<float>(nearend_reverb_default_len.Get());
+  float nearend_len = static_cast<float>(nearend_reverb_nearend_len.Get());
+  if (default_len > -1 && default_len < 1 && nearend_len > -1 &&
+      nearend_len < 1) {
+    adjusted_cfg.ep_strength.default_len =
+        static_cast<float>(nearend_reverb_default_len.Get());
+    adjusted_cfg.ep_strength.nearend_len =
+        static_cast<float>(nearend_reverb_nearend_len.Get());
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3ConservativeTailFreqResponse")) {
+    adjusted_cfg.ep_strength.use_conservative_tail_frequency_response = true;
+  }
+
+  if (field_trial::IsDisabled("WebRTC-Aec3ConservativeTailFreqResponse")) {
+    adjusted_cfg.ep_strength.use_conservative_tail_frequency_response = false;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3ShortHeadroomKillSwitch")) {
+    // Two blocks headroom.
+    adjusted_cfg.delay.delay_headroom_samples = kBlockSize * 2;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3ClampInstQualityToZeroKillSwitch")) {
+    adjusted_cfg.erle.clamp_quality_estimate_to_zero = false;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3ClampInstQualityToOneKillSwitch")) {
+    adjusted_cfg.erle.clamp_quality_estimate_to_one = false;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3OnsetDetectionKillSwitch")) {
+    adjusted_cfg.erle.onset_detection = false;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceRenderDelayEstimationDownmixing")) {
+    adjusted_cfg.delay.render_alignment_mixing.downmix = true;
+    adjusted_cfg.delay.render_alignment_mixing.adaptive_selection = false;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceCaptureDelayEstimationDownmixing")) {
+    adjusted_cfg.delay.capture_alignment_mixing.downmix = true;
+    adjusted_cfg.delay.capture_alignment_mixing.adaptive_selection = false;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceCaptureDelayEstimationLeftRightPrioritization")) {
+    adjusted_cfg.delay.capture_alignment_mixing.prefer_first_two_channels =
+        true;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-"
+          "Aec3RenderDelayEstimationLeftRightPrioritizationKillSwitch")) {
+    adjusted_cfg.delay.capture_alignment_mixing.prefer_first_two_channels =
+        false;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3DelayEstimatorDetectPreEcho")) {
+    adjusted_cfg.delay.detect_pre_echo = true;
+  }
+
+  if (field_trial::IsDisabled("WebRTC-Aec3DelayEstimatorDetectPreEcho")) {
+    adjusted_cfg.delay.detect_pre_echo = false;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3SensitiveDominantNearendActivation")) {
+    adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = 0.5f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3VerySensitiveDominantNearendActivation")) {
+    adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = 0.75f;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3TransparentAntiHowlingGain")) {
+    adjusted_cfg.suppressor.high_bands_suppression.anti_howling_gain = 1.f;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceMoreTransparentNormalSuppressorTuning")) {
+    adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_transparent = 0.4f;
+    adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_suppress = 0.5f;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceMoreTransparentNearendSuppressorTuning")) {
+    adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent = 1.29f;
+    adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress = 1.3f;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceMoreTransparentNormalSuppressorHfTuning")) {
+    adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_transparent = 0.3f;
+    adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_suppress = 0.4f;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceMoreTransparentNearendSuppressorHfTuning")) {
+    adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent = 1.09f;
+    adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_suppress = 1.1f;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceRapidlyAdjustingNormalSuppressorTunings")) {
+    adjusted_cfg.suppressor.normal_tuning.max_inc_factor = 2.5f;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceRapidlyAdjustingNearendSuppressorTunings")) {
+    adjusted_cfg.suppressor.nearend_tuning.max_inc_factor = 2.5f;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceSlowlyAdjustingNormalSuppressorTunings")) {
+    adjusted_cfg.suppressor.normal_tuning.max_dec_factor_lf = .2f;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceSlowlyAdjustingNearendSuppressorTunings")) {
+    adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf = .2f;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3EnforceConservativeHfSuppression")) {
+    adjusted_cfg.suppressor.conservative_hf_suppression = true;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3EnforceStationarityProperties")) {
+    adjusted_cfg.echo_audibility.use_stationarity_properties = true;
+  }
+
+  if (field_trial::IsEnabled(
+          "WebRTC-Aec3EnforceStationarityPropertiesAtInit")) {
+    adjusted_cfg.echo_audibility.use_stationarity_properties_at_init = true;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3EnforceLowActiveRenderLimit")) {
+    adjusted_cfg.render_levels.active_render_limit = 50.f;
+  } else if (field_trial::IsEnabled(
+                 "WebRTC-Aec3EnforceVeryLowActiveRenderLimit")) {
+    adjusted_cfg.render_levels.active_render_limit = 30.f;
+  }
+
+  if (field_trial::IsEnabled("WebRTC-Aec3NonlinearModeReverbKillSwitch")) {
+    adjusted_cfg.echo_model.model_reverb_in_nonlinear_mode = false;
+  }
+
+  // Field-trial based override for the whole suppressor tuning.
+  const std::string suppressor_tuning_override_trial_name =
+      field_trial::FindFullName("WebRTC-Aec3SuppressorTuningOverride");
+
+  FieldTrialParameter<double> nearend_tuning_mask_lf_enr_transparent(
+      "nearend_tuning_mask_lf_enr_transparent",
+      adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent);
+  FieldTrialParameter<double> nearend_tuning_mask_lf_enr_suppress(
+      "nearend_tuning_mask_lf_enr_suppress",
+      adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress);
+  FieldTrialParameter<double> nearend_tuning_mask_hf_enr_transparent(
+      "nearend_tuning_mask_hf_enr_transparent",
+      adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent);
+  FieldTrialParameter<double> nearend_tuning_mask_hf_enr_suppress(
+      "nearend_tuning_mask_hf_enr_suppress",
+      adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_suppress);
+  FieldTrialParameter<double> nearend_tuning_max_inc_factor(
+      "nearend_tuning_max_inc_factor",
+      adjusted_cfg.suppressor.nearend_tuning.max_inc_factor);
+  FieldTrialParameter<double> nearend_tuning_max_dec_factor_lf(
+      "nearend_tuning_max_dec_factor_lf",
+      adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf);
+  FieldTrialParameter<double> normal_tuning_mask_lf_enr_transparent(
+      "normal_tuning_mask_lf_enr_transparent",
+      adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_transparent);
+  FieldTrialParameter<double> normal_tuning_mask_lf_enr_suppress(
+      "normal_tuning_mask_lf_enr_suppress",
+      adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_suppress);
+  FieldTrialParameter<double> normal_tuning_mask_hf_enr_transparent(
+      "normal_tuning_mask_hf_enr_transparent",
+      adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_transparent);
+  FieldTrialParameter<double> normal_tuning_mask_hf_enr_suppress(
+      "normal_tuning_mask_hf_enr_suppress",
+      adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_suppress);
+  FieldTrialParameter<double> normal_tuning_max_inc_factor(
+      "normal_tuning_max_inc_factor",
+      adjusted_cfg.suppressor.normal_tuning.max_inc_factor);
+  FieldTrialParameter<double> normal_tuning_max_dec_factor_lf(
+      "normal_tuning_max_dec_factor_lf",
+      adjusted_cfg.suppressor.normal_tuning.max_dec_factor_lf);
+  FieldTrialParameter<double> dominant_nearend_detection_enr_threshold(
+      "dominant_nearend_detection_enr_threshold",
+      adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold);
+  FieldTrialParameter<double> dominant_nearend_detection_enr_exit_threshold(
+      "dominant_nearend_detection_enr_exit_threshold",
+      adjusted_cfg.suppressor.dominant_nearend_detection.enr_exit_threshold);
+  FieldTrialParameter<double> dominant_nearend_detection_snr_threshold(
+      "dominant_nearend_detection_snr_threshold",
+      adjusted_cfg.suppressor.dominant_nearend_detection.snr_threshold);
+  FieldTrialParameter<int> dominant_nearend_detection_hold_duration(
+      "dominant_nearend_detection_hold_duration",
+      adjusted_cfg.suppressor.dominant_nearend_detection.hold_duration);
+  FieldTrialParameter<int> dominant_nearend_detection_trigger_threshold(
+      "dominant_nearend_detection_trigger_threshold",
+      adjusted_cfg.suppressor.dominant_nearend_detection.trigger_threshold);
+
+  ParseFieldTrial(
+      {&nearend_tuning_mask_lf_enr_transparent,
+       &nearend_tuning_mask_lf_enr_suppress,
+       &nearend_tuning_mask_hf_enr_transparent,
+       &nearend_tuning_mask_hf_enr_suppress, &nearend_tuning_max_inc_factor,
+       &nearend_tuning_max_dec_factor_lf,
+       &normal_tuning_mask_lf_enr_transparent,
+       &normal_tuning_mask_lf_enr_suppress,
+       &normal_tuning_mask_hf_enr_transparent,
+       &normal_tuning_mask_hf_enr_suppress, &normal_tuning_max_inc_factor,
+       &normal_tuning_max_dec_factor_lf,
+       &dominant_nearend_detection_enr_threshold,
+       &dominant_nearend_detection_enr_exit_threshold,
+       &dominant_nearend_detection_snr_threshold,
+       &dominant_nearend_detection_hold_duration,
+       &dominant_nearend_detection_trigger_threshold},
+      suppressor_tuning_override_trial_name);
+
+  adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent =
+      static_cast<float>(nearend_tuning_mask_lf_enr_transparent.Get());
+  adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress =
+      static_cast<float>(nearend_tuning_mask_lf_enr_suppress.Get());
+  adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent =
+      static_cast<float>(nearend_tuning_mask_hf_enr_transparent.Get());
+  adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_suppress =
+      static_cast<float>(nearend_tuning_mask_hf_enr_suppress.Get());
+  adjusted_cfg.suppressor.nearend_tuning.max_inc_factor =
+      static_cast<float>(nearend_tuning_max_inc_factor.Get());
+  adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf =
+      static_cast<float>(nearend_tuning_max_dec_factor_lf.Get());
+  adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_transparent =
+      static_cast<float>(normal_tuning_mask_lf_enr_transparent.Get());
+  adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_suppress =
+      static_cast<float>(normal_tuning_mask_lf_enr_suppress.Get());
+  adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_transparent =
+      static_cast<float>(normal_tuning_mask_hf_enr_transparent.Get());
+  adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_suppress =
+      static_cast<float>(normal_tuning_mask_hf_enr_suppress.Get());
+  adjusted_cfg.suppressor.normal_tuning.max_inc_factor =
+      static_cast<float>(normal_tuning_max_inc_factor.Get());
+  adjusted_cfg.suppressor.normal_tuning.max_dec_factor_lf =
+      static_cast<float>(normal_tuning_max_dec_factor_lf.Get());
+  adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold =
+      static_cast<float>(dominant_nearend_detection_enr_threshold.Get());
+  adjusted_cfg.suppressor.dominant_nearend_detection.enr_exit_threshold =
+      static_cast<float>(dominant_nearend_detection_enr_exit_threshold.Get());
+  adjusted_cfg.suppressor.dominant_nearend_detection.snr_threshold =
+      static_cast<float>(dominant_nearend_detection_snr_threshold.Get());
+  adjusted_cfg.suppressor.dominant_nearend_detection.hold_duration =
+      dominant_nearend_detection_hold_duration.Get();
+  adjusted_cfg.suppressor.dominant_nearend_detection.trigger_threshold =
+      dominant_nearend_detection_trigger_threshold.Get();
+
+  // Field trial-based overrides of individual suppressor parameters.
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNearendLfMaskTransparentOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNearendLfMaskSuppressOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNearendHfMaskTransparentOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNearendHfMaskSuppressOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_suppress);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNearendMaxIncFactorOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.nearend_tuning.max_inc_factor);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNearendMaxDecFactorLfOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf);
+
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNormalLfMaskTransparentOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_transparent);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNormalLfMaskSuppressOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_suppress);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNormalHfMaskTransparentOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_transparent);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNormalHfMaskSuppressOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_suppress);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNormalMaxIncFactorOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.normal_tuning.max_inc_factor);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorNormalMaxDecFactorLfOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.normal_tuning.max_dec_factor_lf);
+
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorDominantNearendEnrThresholdOverride", 0.f, 100.f,
+      &adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorDominantNearendEnrExitThresholdOverride", 0.f,
+      100.f,
+      &adjusted_cfg.suppressor.dominant_nearend_detection.enr_exit_threshold);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorDominantNearendSnrThresholdOverride", 0.f, 100.f,
+      &adjusted_cfg.suppressor.dominant_nearend_detection.snr_threshold);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorDominantNearendHoldDurationOverride", 0, 1000,
+      &adjusted_cfg.suppressor.dominant_nearend_detection.hold_duration);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorDominantNearendTriggerThresholdOverride", 0, 1000,
+      &adjusted_cfg.suppressor.dominant_nearend_detection.trigger_threshold);
+
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3SuppressorAntiHowlingGainOverride", 0.f, 10.f,
+      &adjusted_cfg.suppressor.high_bands_suppression.anti_howling_gain);
+
+  // Field trial-based overrides of individual delay estimator parameters.
+  RetrieveFieldTrialValue("WebRTC-Aec3DelayEstimateSmoothingOverride", 0.f, 1.f,
+                          &adjusted_cfg.delay.delay_estimate_smoothing);
+  RetrieveFieldTrialValue(
+      "WebRTC-Aec3DelayEstimateSmoothingDelayFoundOverride", 0.f, 1.f,
+      &adjusted_cfg.delay.delay_estimate_smoothing_delay_found);
+
+  return adjusted_cfg;
+}
+
+class EchoCanceller3::RenderWriter {
+ public:
+  RenderWriter(ApmDataDumper* data_dumper,
+               const EchoCanceller3Config& config,
+               SwapQueue<std::vector<std::vector<std::vector<float>>>,
+                         Aec3RenderQueueItemVerifier>* render_transfer_queue,
+               size_t num_bands,
+               size_t num_channels);
+
+  RenderWriter() = delete;
+  RenderWriter(const RenderWriter&) = delete;
+  RenderWriter& operator=(const RenderWriter&) = delete;
+
+  ~RenderWriter();
+  void Insert(const AudioBuffer& input);
+
+ private:
+  ApmDataDumper* data_dumper_;
+  const size_t num_bands_;
+  const size_t num_channels_;
+  std::unique_ptr<HighPassFilter> high_pass_filter_;
+  std::vector<std::vector<std::vector<float>>> render_queue_input_frame_;
+  SwapQueue<std::vector<std::vector<std::vector<float>>>,
+            Aec3RenderQueueItemVerifier>* render_transfer_queue_;
+};
+
+EchoCanceller3::RenderWriter::RenderWriter(
+    ApmDataDumper* data_dumper,
+    const EchoCanceller3Config& config,
+    SwapQueue<std::vector<std::vector<std::vector<float>>>,
+              Aec3RenderQueueItemVerifier>* render_transfer_queue,
+    size_t num_bands,
+    size_t num_channels)
+    : data_dumper_(data_dumper),
+      num_bands_(num_bands),
+      num_channels_(num_channels),
+      render_queue_input_frame_(
+          num_bands_,
+          std::vector<std::vector<float>>(
+              num_channels_,
+              std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))),
+      render_transfer_queue_(render_transfer_queue) {
+  RTC_DCHECK(data_dumper);
+  if (config.filter.high_pass_filter_echo_reference) {
+    high_pass_filter_ = std::make_unique<HighPassFilter>(16000, num_channels);
+  }
+}
+
+EchoCanceller3::RenderWriter::~RenderWriter() = default;
+
+void EchoCanceller3::RenderWriter::Insert(const AudioBuffer& input) {
+  RTC_DCHECK_EQ(AudioBuffer::kSplitBandSize, input.num_frames_per_band());
+  RTC_DCHECK_EQ(num_bands_, input.num_bands());
+  RTC_DCHECK_EQ(num_channels_, input.num_channels());
+
+  // TODO(bugs.webrtc.org/8759) Temporary work-around.
+  if (num_bands_ != input.num_bands())
+    return;
+
+  data_dumper_->DumpWav("aec3_render_input", AudioBuffer::kSplitBandSize,
+                        &input.split_bands_const(0)[0][0], 16000, 1);
+
+  CopyBufferIntoFrame(input, num_bands_, num_channels_,
+                      &render_queue_input_frame_);
+  if (high_pass_filter_) {
+    high_pass_filter_->Process(&render_queue_input_frame_[0]);
+  }
+
+  static_cast<void>(render_transfer_queue_->Insert(&render_queue_input_frame_));
+}
+
+std::atomic<int> EchoCanceller3::instance_count_(0);
+
+EchoCanceller3::EchoCanceller3(
+    const EchoCanceller3Config& config,
+    const absl::optional<EchoCanceller3Config>& multichannel_config,
+    int sample_rate_hz,
+    size_t num_render_channels,
+    size_t num_capture_channels)
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      config_(AdjustConfig(config)),
+      sample_rate_hz_(sample_rate_hz),
+      num_bands_(NumBandsForRate(sample_rate_hz_)),
+      num_render_input_channels_(num_render_channels),
+      num_capture_channels_(num_capture_channels),
+      config_selector_(AdjustConfig(config),
+                       multichannel_config,
+                       num_render_input_channels_),
+      multichannel_content_detector_(
+          config_selector_.active_config().multi_channel.detect_stereo_content,
+          num_render_input_channels_,
+          config_selector_.active_config()
+              .multi_channel.stereo_detection_threshold,
+          config_selector_.active_config()
+              .multi_channel.stereo_detection_timeout_threshold_seconds,
+          config_selector_.active_config()
+              .multi_channel.stereo_detection_hysteresis_seconds),
+      output_framer_(num_bands_, num_capture_channels_),
+      capture_blocker_(num_bands_, num_capture_channels_),
+      render_transfer_queue_(
+          kRenderTransferQueueSizeFrames,
+          std::vector<std::vector<std::vector<float>>>(
+              num_bands_,
+              std::vector<std::vector<float>>(
+                  num_render_input_channels_,
+                  std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))),
+          Aec3RenderQueueItemVerifier(num_bands_,
+                                      num_render_input_channels_,
+                                      AudioBuffer::kSplitBandSize)),
+      render_queue_output_frame_(
+          num_bands_,
+          std::vector<std::vector<float>>(
+              num_render_input_channels_,
+              std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))),
+      render_block_(num_bands_, num_render_input_channels_),
+      capture_block_(num_bands_, num_capture_channels_),
+      capture_sub_frame_view_(
+          num_bands_,
+          std::vector<rtc::ArrayView<float>>(num_capture_channels_)) {
+  RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
+
+  if (config_selector_.active_config().delay.fixed_capture_delay_samples > 0) {
+    block_delay_buffer_.reset(new BlockDelayBuffer(
+        num_capture_channels_, num_bands_, AudioBuffer::kSplitBandSize,
+        config_.delay.fixed_capture_delay_samples));
+  }
+
+  render_writer_.reset(new RenderWriter(
+      data_dumper_.get(), config_selector_.active_config(),
+      &render_transfer_queue_, num_bands_, num_render_input_channels_));
+
+  RTC_DCHECK_EQ(num_bands_, std::max(sample_rate_hz_, 16000) / 16000);
+  RTC_DCHECK_GE(kMaxNumBands, num_bands_);
+
+  if (config_selector_.active_config().filter.export_linear_aec_output) {
+    linear_output_framer_.reset(
+        new BlockFramer(/*num_bands=*/1, num_capture_channels_));
+    linear_output_block_ =
+        std::make_unique<Block>(/*num_bands=*/1, num_capture_channels_),
+    linear_output_sub_frame_view_ =
+        std::vector<std::vector<rtc::ArrayView<float>>>(
+            1, std::vector<rtc::ArrayView<float>>(num_capture_channels_));
+  }
+
+  Initialize();
+
+  RTC_LOG(LS_INFO) << "AEC3 created with sample rate: " << sample_rate_hz_
+                   << " Hz, num render channels: " << num_render_input_channels_
+                   << ", num capture channels: " << num_capture_channels_;
+}
+
+EchoCanceller3::~EchoCanceller3() = default;
+
+void EchoCanceller3::Initialize() {
+  RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
+
+  num_render_channels_to_aec_ =
+      multichannel_content_detector_.IsProperMultiChannelContentDetected()
+          ? num_render_input_channels_
+          : 1;
+
+  config_selector_.Update(
+      multichannel_content_detector_.IsProperMultiChannelContentDetected());
+
+  render_block_.SetNumChannels(num_render_channels_to_aec_);
+
+  render_blocker_.reset(
+      new FrameBlocker(num_bands_, num_render_channels_to_aec_));
+
+  block_processor_.reset(BlockProcessor::Create(
+      config_selector_.active_config(), sample_rate_hz_,
+      num_render_channels_to_aec_, num_capture_channels_));
+
+  render_sub_frame_view_ = std::vector<std::vector<rtc::ArrayView<float>>>(
+      num_bands_,
+      std::vector<rtc::ArrayView<float>>(num_render_channels_to_aec_));
+}
+
+void EchoCanceller3::AnalyzeRender(const AudioBuffer& render) {
+  RTC_DCHECK_RUNS_SERIALIZED(&render_race_checker_);
+
+  RTC_DCHECK_EQ(render.num_channels(), num_render_input_channels_);
+  data_dumper_->DumpRaw("aec3_call_order",
+                        static_cast<int>(EchoCanceller3ApiCall::kRender));
+
+  return render_writer_->Insert(render);
+}
+
+void EchoCanceller3::AnalyzeCapture(const AudioBuffer& capture) {
+  RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
+  data_dumper_->DumpWav("aec3_capture_analyze_input", capture.num_frames(),
+                        capture.channels_const()[0], sample_rate_hz_, 1);
+  saturated_microphone_signal_ = false;
+  for (size_t channel = 0; channel < capture.num_channels(); ++channel) {
+    saturated_microphone_signal_ |=
+        DetectSaturation(rtc::ArrayView<const float>(
+            capture.channels_const()[channel], capture.num_frames()));
+    if (saturated_microphone_signal_) {
+      break;
+    }
+  }
+}
+
+void EchoCanceller3::ProcessCapture(AudioBuffer* capture, bool level_change) {
+  ProcessCapture(capture, nullptr, level_change);
+}
+
+void EchoCanceller3::ProcessCapture(AudioBuffer* capture,
+                                    AudioBuffer* linear_output,
+                                    bool level_change) {
+  RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
+  RTC_DCHECK(capture);
+  RTC_DCHECK_EQ(num_bands_, capture->num_bands());
+  RTC_DCHECK_EQ(AudioBuffer::kSplitBandSize, capture->num_frames_per_band());
+  RTC_DCHECK_EQ(capture->num_channels(), num_capture_channels_);
+  data_dumper_->DumpRaw("aec3_call_order",
+                        static_cast<int>(EchoCanceller3ApiCall::kCapture));
+
+  if (linear_output && !linear_output_framer_) {
+    RTC_LOG(LS_ERROR) << "Trying to retrieve the linear AEC output without "
+                         "properly configuring AEC3.";
+    RTC_DCHECK_NOTREACHED();
+  }
+
+  // Report capture call in the metrics and periodically update API call
+  // metrics.
+  api_call_metrics_.ReportCaptureCall();
+
+  // Optionally delay the capture signal.
+  if (config_selector_.active_config().delay.fixed_capture_delay_samples > 0) {
+    RTC_DCHECK(block_delay_buffer_);
+    block_delay_buffer_->DelaySignal(capture);
+  }
+
+  rtc::ArrayView<float> capture_lower_band = rtc::ArrayView<float>(
+      &capture->split_bands(0)[0][0], AudioBuffer::kSplitBandSize);
+
+  data_dumper_->DumpWav("aec3_capture_input", capture_lower_band, 16000, 1);
+
+  EmptyRenderQueue();
+
+  ProcessCaptureFrameContent(
+      linear_output, capture, level_change,
+      multichannel_content_detector_.IsTemporaryMultiChannelContentDetected(),
+      saturated_microphone_signal_, 0, &capture_blocker_,
+      linear_output_framer_.get(), &output_framer_, block_processor_.get(),
+      linear_output_block_.get(), &linear_output_sub_frame_view_,
+      &capture_block_, &capture_sub_frame_view_);
+
+  ProcessCaptureFrameContent(
+      linear_output, capture, level_change,
+      multichannel_content_detector_.IsTemporaryMultiChannelContentDetected(),
+      saturated_microphone_signal_, 1, &capture_blocker_,
+      linear_output_framer_.get(), &output_framer_, block_processor_.get(),
+      linear_output_block_.get(), &linear_output_sub_frame_view_,
+      &capture_block_, &capture_sub_frame_view_);
+
+  ProcessRemainingCaptureFrameContent(
+      level_change,
+      multichannel_content_detector_.IsTemporaryMultiChannelContentDetected(),
+      saturated_microphone_signal_, &capture_blocker_,
+      linear_output_framer_.get(), &output_framer_, block_processor_.get(),
+      linear_output_block_.get(), &capture_block_);
+
+  data_dumper_->DumpWav("aec3_capture_output", AudioBuffer::kSplitBandSize,
+                        &capture->split_bands(0)[0][0], 16000, 1);
+}
+
+EchoControl::Metrics EchoCanceller3::GetMetrics() const {
+  RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
+  Metrics metrics;
+  block_processor_->GetMetrics(&metrics);
+  return metrics;
+}
+
+void EchoCanceller3::SetAudioBufferDelay(int delay_ms) {
+  RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
+  block_processor_->SetAudioBufferDelay(delay_ms);
+}
+
+void EchoCanceller3::SetCaptureOutputUsage(bool capture_output_used) {
+  RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
+  block_processor_->SetCaptureOutputUsage(capture_output_used);
+}
+
+bool EchoCanceller3::ActiveProcessing() const {
+  return true;
+}
+
+EchoCanceller3Config EchoCanceller3::CreateDefaultMultichannelConfig() {
+  EchoCanceller3Config cfg;
+  // Use shorter and more rapidly adapting coarse filter to compensate for
+  // thge increased number of total filter parameters to adapt.
+  cfg.filter.coarse.length_blocks = 11;
+  cfg.filter.coarse.rate = 0.95f;
+  cfg.filter.coarse_initial.length_blocks = 11;
+  cfg.filter.coarse_initial.rate = 0.95f;
+
+  // Use more concervative suppressor behavior for non-nearend speech.
+  cfg.suppressor.normal_tuning.max_dec_factor_lf = 0.35f;
+  cfg.suppressor.normal_tuning.max_inc_factor = 1.5f;
+  return cfg;
+}
+
+void EchoCanceller3::SetBlockProcessorForTesting(
+    std::unique_ptr<BlockProcessor> block_processor) {
+  RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
+  RTC_DCHECK(block_processor);
+  block_processor_ = std::move(block_processor);
+}
+
+void EchoCanceller3::EmptyRenderQueue() {
+  RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
+  bool frame_to_buffer =
+      render_transfer_queue_.Remove(&render_queue_output_frame_);
+  while (frame_to_buffer) {
+    // Report render call in the metrics.
+    api_call_metrics_.ReportRenderCall();
+
+    if (multichannel_content_detector_.UpdateDetection(
+            render_queue_output_frame_)) {
+      // Reinitialize the AEC when proper stereo is detected.
+      Initialize();
+    }
+
+    // Buffer frame content.
+    BufferRenderFrameContent(
+        /*proper_downmix_needed=*/multichannel_content_detector_
+            .IsTemporaryMultiChannelContentDetected(),
+        &render_queue_output_frame_, 0, render_blocker_.get(),
+        block_processor_.get(), &render_block_, &render_sub_frame_view_);
+
+    BufferRenderFrameContent(
+        /*proper_downmix_needed=*/multichannel_content_detector_
+            .IsTemporaryMultiChannelContentDetected(),
+        &render_queue_output_frame_, 1, render_blocker_.get(),
+        block_processor_.get(), &render_block_, &render_sub_frame_view_);
+
+    BufferRemainingRenderFrameContent(render_blocker_.get(),
+                                      block_processor_.get(), &render_block_);
+
+    frame_to_buffer =
+        render_transfer_queue_.Remove(&render_queue_output_frame_);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3.h b/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3.h
new file mode 100644
index 0000000000..7bf8e51a4b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3.h
@@ -0,0 +1,230 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
+
+#include <stddef.h>
+
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "api/audio/echo_control.h"
+#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
+#include "modules/audio_processing/aec3/block_delay_buffer.h"
+#include "modules/audio_processing/aec3/block_framer.h"
+#include "modules/audio_processing/aec3/block_processor.h"
+#include "modules/audio_processing/aec3/config_selector.h"
+#include "modules/audio_processing/aec3/frame_blocker.h"
+#include "modules/audio_processing/aec3/multi_channel_content_detector.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/swap_queue.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Method for adjusting config parameter dependencies.
+// Only to be used externally to AEC3 for testing purposes.
+// TODO(webrtc:5298): Move this to a separate file.
+EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config);
+
+// Functor for verifying the invariance of the frames being put into the render
+// queue.
+class Aec3RenderQueueItemVerifier {
+ public:
+  Aec3RenderQueueItemVerifier(size_t num_bands,
+                              size_t num_channels,
+                              size_t frame_length)
+      : num_bands_(num_bands),
+        num_channels_(num_channels),
+        frame_length_(frame_length) {}
+
+  bool operator()(const std::vector<std::vector<std::vector<float>>>& v) const {
+    if (v.size() != num_bands_) {
+      return false;
+    }
+    for (const auto& band : v) {
+      if (band.size() != num_channels_) {
+        return false;
+      }
+      for (const auto& channel : band) {
+        if (channel.size() != frame_length_) {
+          return false;
+        }
+      }
+    }
+    return true;
+  }
+
+ private:
+  const size_t num_bands_;
+  const size_t num_channels_;
+  const size_t frame_length_;
+};
+
+// Main class for the echo canceller3.
+// It does 4 things:
+// -Receives 10 ms frames of band-split audio.
+// -Provides the lower level echo canceller functionality with
+// blocks of 64 samples of audio data.
+// -Partially handles the jitter in the render and capture API
+// call sequence.
+//
+// The class is supposed to be used in a non-concurrent manner apart from the
+// AnalyzeRender call which can be called concurrently with the other methods.
+class EchoCanceller3 : public EchoControl {
+ public:
+  EchoCanceller3(
+      const EchoCanceller3Config& config,
+      const absl::optional<EchoCanceller3Config>& multichannel_config,
+      int sample_rate_hz,
+      size_t num_render_channels,
+      size_t num_capture_channels);
+
+  ~EchoCanceller3() override;
+
+  EchoCanceller3(const EchoCanceller3&) = delete;
+  EchoCanceller3& operator=(const EchoCanceller3&) = delete;
+
+  // Analyzes and stores an internal copy of the split-band domain render
+  // signal.
+  void AnalyzeRender(AudioBuffer* render) override { AnalyzeRender(*render); }
+  // Analyzes the full-band domain capture signal to detect signal saturation.
+  void AnalyzeCapture(AudioBuffer* capture) override {
+    AnalyzeCapture(*capture);
+  }
+  // Processes the split-band domain capture signal in order to remove any echo
+  // present in the signal.
+  void ProcessCapture(AudioBuffer* capture, bool level_change) override;
+  // As above, but also returns the linear filter output.
+  void ProcessCapture(AudioBuffer* capture,
+                      AudioBuffer* linear_output,
+                      bool level_change) override;
+  // Collect current metrics from the echo canceller.
+  Metrics GetMetrics() const override;
+  // Provides an optional external estimate of the audio buffer delay.
+  void SetAudioBufferDelay(int delay_ms) override;
+
+  // Specifies whether the capture output will be used. The purpose of this is
+  // to allow the echo controller to deactivate some of the processing when the
+  // resulting output is anyway not used, for instance when the endpoint is
+  // muted.
+  void SetCaptureOutputUsage(bool capture_output_used) override;
+
+  bool ActiveProcessing() const override;
+
+  // Signals whether an external detector has detected echo leakage from the
+  // echo canceller.
+  // Note that in the case echo leakage has been flagged, it should be unflagged
+  // once it is no longer occurring.
+  void UpdateEchoLeakageStatus(bool leakage_detected) {
+    RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
+    block_processor_->UpdateEchoLeakageStatus(leakage_detected);
+  }
+
+  // Produces a default configuration for multichannel.
+  static EchoCanceller3Config CreateDefaultMultichannelConfig();
+
+ private:
+  friend class EchoCanceller3Tester;
+  FRIEND_TEST_ALL_PREFIXES(EchoCanceller3, DetectionOfProperStereo);
+  FRIEND_TEST_ALL_PREFIXES(EchoCanceller3,
+                           DetectionOfProperStereoUsingThreshold);
+  FRIEND_TEST_ALL_PREFIXES(EchoCanceller3,
+                           DetectionOfProperStereoUsingHysteresis);
+  FRIEND_TEST_ALL_PREFIXES(EchoCanceller3,
+                           StereoContentDetectionForMonoSignals);
+
+  class RenderWriter;
+
+  // (Re-)Initializes the selected subset of the EchoCanceller3 fields, at
+  // creation as well as during reconfiguration.
+  void Initialize();
+
+  // Only for testing. Replaces the internal block processor.
+  void SetBlockProcessorForTesting(
+      std::unique_ptr<BlockProcessor> block_processor);
+
+  // Only for testing. Returns whether stereo processing is active.
+  bool StereoRenderProcessingActiveForTesting() const {
+    return multichannel_content_detector_.IsProperMultiChannelContentDetected();
+  }
+
+  // Only for testing.
+  const EchoCanceller3Config& GetActiveConfigForTesting() const {
+    return config_selector_.active_config();
+  }
+
+  // Empties the render SwapQueue.
+  void EmptyRenderQueue();
+
+  // Analyzes and stores an internal copy of the split-band domain render
+  // signal.
+  void AnalyzeRender(const AudioBuffer& render);
+  // Analyzes the full-band domain capture signal to detect signal saturation.
+  void AnalyzeCapture(const AudioBuffer& capture);
+
+  rtc::RaceChecker capture_race_checker_;
+  rtc::RaceChecker render_race_checker_;
+
+  // State that is accessed by the AnalyzeRender call.
+  std::unique_ptr<RenderWriter> render_writer_
+      RTC_GUARDED_BY(render_race_checker_);
+
+  // State that may be accessed by the capture thread.
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  const EchoCanceller3Config config_;
+  const int sample_rate_hz_;
+  const int num_bands_;
+  const size_t num_render_input_channels_;
+  size_t num_render_channels_to_aec_;
+  const size_t num_capture_channels_;
+  ConfigSelector config_selector_;
+  MultiChannelContentDetector multichannel_content_detector_;
+  std::unique_ptr<BlockFramer> linear_output_framer_
+      RTC_GUARDED_BY(capture_race_checker_);
+  BlockFramer output_framer_ RTC_GUARDED_BY(capture_race_checker_);
+  FrameBlocker capture_blocker_ RTC_GUARDED_BY(capture_race_checker_);
+  std::unique_ptr<FrameBlocker> render_blocker_
+      RTC_GUARDED_BY(capture_race_checker_);
+  SwapQueue<std::vector<std::vector<std::vector<float>>>,
+            Aec3RenderQueueItemVerifier>
+      render_transfer_queue_;
+  std::unique_ptr<BlockProcessor> block_processor_
+      RTC_GUARDED_BY(capture_race_checker_);
+  std::vector<std::vector<std::vector<float>>> render_queue_output_frame_
+      RTC_GUARDED_BY(capture_race_checker_);
+  bool saturated_microphone_signal_ RTC_GUARDED_BY(capture_race_checker_) =
+      false;
+  Block render_block_ RTC_GUARDED_BY(capture_race_checker_);
+  std::unique_ptr<Block> linear_output_block_
+      RTC_GUARDED_BY(capture_race_checker_);
+  Block capture_block_ RTC_GUARDED_BY(capture_race_checker_);
+  std::vector<std::vector<rtc::ArrayView<float>>> render_sub_frame_view_
+      RTC_GUARDED_BY(capture_race_checker_);
+  std::vector<std::vector<rtc::ArrayView<float>>> linear_output_sub_frame_view_
+      RTC_GUARDED_BY(capture_race_checker_);
+  std::vector<std::vector<rtc::ArrayView<float>>> capture_sub_frame_view_
+      RTC_GUARDED_BY(capture_race_checker_);
+  std::unique_ptr<BlockDelayBuffer> block_delay_buffer_
+      RTC_GUARDED_BY(capture_race_checker_);
+  ApiCallJitterMetrics api_call_metrics_ RTC_GUARDED_BY(capture_race_checker_);
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc
new file mode 100644
index 0000000000..ad126af4d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc
@@ -0,0 +1,1160 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/echo_canceller3.h"
+
+#include <deque>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/block_processor.h"
+#include "modules/audio_processing/aec3/frame_blocker.h"
+#include "modules/audio_processing/aec3/mock/mock_block_processor.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/high_pass_filter.h"
+#include "modules/audio_processing/utility/cascaded_biquad_filter.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::StrictMock;
+
+// Populates the frame with linearly increasing sample values for each band,
+// with a band-specific offset, in order to allow simple bitexactness
+// verification for each band.
+void PopulateInputFrame(size_t frame_length,
+                        size_t num_bands,
+                        size_t frame_index,
+                        float* const* frame,
+                        int offset) {
+  for (size_t k = 0; k < num_bands; ++k) {
+    for (size_t i = 0; i < frame_length; ++i) {
+      float value = static_cast<int>(frame_index * frame_length + i) + offset;
+      frame[k][i] = (value > 0 ? 5000 * k + value : 0);
+    }
+  }
+}
+
+// Populates the frame with linearly increasing sample values.
+void PopulateInputFrame(size_t frame_length,
+                        size_t frame_index,
+                        float* frame,
+                        int offset) {
+  for (size_t i = 0; i < frame_length; ++i) {
+    float value = static_cast<int>(frame_index * frame_length + i) + offset;
+    frame[i] = std::max(value, 0.f);
+  }
+}
+
+// Verifies the that samples in the output frame are identical to the samples
+// that were produced for the input frame, with an offset in order to compensate
+// for buffering delays.
+bool VerifyOutputFrameBitexactness(size_t frame_length,
+                                   size_t num_bands,
+                                   size_t frame_index,
+                                   const float* const* frame,
+                                   int offset) {
+  float reference_frame_data[kMaxNumBands][2 * kSubFrameLength];
+  float* reference_frame[kMaxNumBands];
+  for (size_t k = 0; k < num_bands; ++k) {
+    reference_frame[k] = &reference_frame_data[k][0];
+  }
+
+  PopulateInputFrame(frame_length, num_bands, frame_index, reference_frame,
+                     offset);
+  for (size_t k = 0; k < num_bands; ++k) {
+    for (size_t i = 0; i < frame_length; ++i) {
+      if (reference_frame[k][i] != frame[k][i]) {
+        return false;
+      }
+    }
+  }
+
+  return true;
+}
+
+bool VerifyOutputFrameBitexactness(rtc::ArrayView<const float> reference,
+                                   rtc::ArrayView<const float> frame,
+                                   int offset) {
+  for (size_t k = 0; k < frame.size(); ++k) {
+    int reference_index = static_cast<int>(k) + offset;
+    if (reference_index >= 0) {
+      if (reference[reference_index] != frame[k]) {
+        return false;
+      }
+    }
+  }
+  return true;
+}
+
+// Class for testing that the capture data is properly received by the block
+// processor and that the processor data is properly passed to the
+// EchoCanceller3 output.
+class CaptureTransportVerificationProcessor : public BlockProcessor {
+ public:
+  explicit CaptureTransportVerificationProcessor(size_t num_bands) {}
+
+  CaptureTransportVerificationProcessor() = delete;
+  CaptureTransportVerificationProcessor(
+      const CaptureTransportVerificationProcessor&) = delete;
+  CaptureTransportVerificationProcessor& operator=(
+      const CaptureTransportVerificationProcessor&) = delete;
+
+  ~CaptureTransportVerificationProcessor() override = default;
+
+  void ProcessCapture(bool level_change,
+                      bool saturated_microphone_signal,
+                      Block* linear_output,
+                      Block* capture_block) override {}
+
+  void BufferRender(const Block& block) override {}
+
+  void UpdateEchoLeakageStatus(bool leakage_detected) override {}
+
+  void GetMetrics(EchoControl::Metrics* metrics) const override {}
+
+  void SetAudioBufferDelay(int delay_ms) override {}
+
+  void SetCaptureOutputUsage(bool capture_output_used) {}
+};
+
+// Class for testing that the render data is properly received by the block
+// processor.
+class RenderTransportVerificationProcessor : public BlockProcessor {
+ public:
+  explicit RenderTransportVerificationProcessor(size_t num_bands) {}
+
+  RenderTransportVerificationProcessor() = delete;
+  RenderTransportVerificationProcessor(
+      const RenderTransportVerificationProcessor&) = delete;
+  RenderTransportVerificationProcessor& operator=(
+      const RenderTransportVerificationProcessor&) = delete;
+
+  ~RenderTransportVerificationProcessor() override = default;
+
+  void ProcessCapture(bool level_change,
+                      bool saturated_microphone_signal,
+                      Block* linear_output,
+                      Block* capture_block) override {
+    Block render_block = received_render_blocks_.front();
+    received_render_blocks_.pop_front();
+    capture_block->Swap(render_block);
+  }
+
+  void BufferRender(const Block& block) override {
+    received_render_blocks_.push_back(block);
+  }
+
+  void UpdateEchoLeakageStatus(bool leakage_detected) override {}
+
+  void GetMetrics(EchoControl::Metrics* metrics) const override {}
+
+  void SetAudioBufferDelay(int delay_ms) override {}
+
+  void SetCaptureOutputUsage(bool capture_output_used) {}
+
+ private:
+  std::deque<Block> received_render_blocks_;
+};
+
+std::string ProduceDebugText(int sample_rate_hz) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz;
+  return ss.Release();
+}
+
+std::string ProduceDebugText(int sample_rate_hz, int variant) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz << ", variant: " << variant;
+  return ss.Release();
+}
+
+void RunAecInStereo(AudioBuffer& buffer,
+                    EchoCanceller3& aec3,
+                    float channel_0_value,
+                    float channel_1_value) {
+  rtc::ArrayView<float> data_channel_0(&buffer.channels()[0][0],
+                                       buffer.num_frames());
+  std::fill(data_channel_0.begin(), data_channel_0.end(), channel_0_value);
+  rtc::ArrayView<float> data_channel_1(&buffer.channels()[1][0],
+                                       buffer.num_frames());
+  std::fill(data_channel_1.begin(), data_channel_1.end(), channel_1_value);
+  aec3.AnalyzeRender(&buffer);
+  aec3.AnalyzeCapture(&buffer);
+  aec3.ProcessCapture(&buffer, /*level_change=*/false);
+}
+
+void RunAecInSMono(AudioBuffer& buffer,
+                   EchoCanceller3& aec3,
+                   float channel_0_value) {
+  rtc::ArrayView<float> data_channel_0(&buffer.channels()[0][0],
+                                       buffer.num_frames());
+  std::fill(data_channel_0.begin(), data_channel_0.end(), channel_0_value);
+  aec3.AnalyzeRender(&buffer);
+  aec3.AnalyzeCapture(&buffer);
+  aec3.ProcessCapture(&buffer, /*level_change=*/false);
+}
+
+}  // namespace
+
+class EchoCanceller3Tester {
+ public:
+  explicit EchoCanceller3Tester(int sample_rate_hz)
+      : sample_rate_hz_(sample_rate_hz),
+        num_bands_(NumBandsForRate(sample_rate_hz_)),
+        frame_length_(160),
+        fullband_frame_length_(rtc::CheckedDivExact(sample_rate_hz_, 100)),
+        capture_buffer_(fullband_frame_length_ * 100,
+                        1,
+                        fullband_frame_length_ * 100,
+                        1,
+                        fullband_frame_length_ * 100,
+                        1),
+        render_buffer_(fullband_frame_length_ * 100,
+                       1,
+                       fullband_frame_length_ * 100,
+                       1,
+                       fullband_frame_length_ * 100,
+                       1) {}
+
+  EchoCanceller3Tester() = delete;
+  EchoCanceller3Tester(const EchoCanceller3Tester&) = delete;
+  EchoCanceller3Tester& operator=(const EchoCanceller3Tester&) = delete;
+
+  // Verifies that the capture data is properly received by the block processor
+  // and that the processor data is properly passed to the EchoCanceller3
+  // output.
+  void RunCaptureTransportVerificationTest() {
+    EchoCanceller3 aec3(EchoCanceller3Config(),
+                        /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+                        1, 1);
+    aec3.SetBlockProcessorForTesting(
+        std::make_unique<CaptureTransportVerificationProcessor>(num_bands_));
+
+    for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
+         ++frame_index) {
+      aec3.AnalyzeCapture(&capture_buffer_);
+      OptionalBandSplit();
+      PopulateInputFrame(frame_length_, num_bands_, frame_index,
+                         &capture_buffer_.split_bands(0)[0], 0);
+      PopulateInputFrame(frame_length_, frame_index,
+                         &render_buffer_.channels()[0][0], 0);
+
+      aec3.AnalyzeRender(&render_buffer_);
+      aec3.ProcessCapture(&capture_buffer_, false);
+      EXPECT_TRUE(VerifyOutputFrameBitexactness(
+          frame_length_, num_bands_, frame_index,
+          &capture_buffer_.split_bands(0)[0], -64));
+    }
+  }
+
+  // Test method for testing that the render data is properly received by the
+  // block processor.
+  void RunRenderTransportVerificationTest() {
+    EchoCanceller3 aec3(EchoCanceller3Config(),
+                        /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+                        1, 1);
+    aec3.SetBlockProcessorForTesting(
+        std::make_unique<RenderTransportVerificationProcessor>(num_bands_));
+
+    std::vector<std::vector<float>> render_input(1);
+    std::vector<float> capture_output;
+    for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
+         ++frame_index) {
+      aec3.AnalyzeCapture(&capture_buffer_);
+      OptionalBandSplit();
+      PopulateInputFrame(frame_length_, num_bands_, frame_index,
+                         &capture_buffer_.split_bands(0)[0], 100);
+      PopulateInputFrame(frame_length_, num_bands_, frame_index,
+                         &render_buffer_.split_bands(0)[0], 0);
+
+      for (size_t k = 0; k < frame_length_; ++k) {
+        render_input[0].push_back(render_buffer_.split_bands(0)[0][k]);
+      }
+      aec3.AnalyzeRender(&render_buffer_);
+      aec3.ProcessCapture(&capture_buffer_, false);
+      for (size_t k = 0; k < frame_length_; ++k) {
+        capture_output.push_back(capture_buffer_.split_bands(0)[0][k]);
+      }
+    }
+
+    EXPECT_TRUE(
+        VerifyOutputFrameBitexactness(render_input[0], capture_output, -64));
+  }
+
+  // Verifies that information about echo path changes are properly propagated
+  // to the block processor.
+  // The cases tested are:
+  // -That no set echo path change flags are received when there is no echo path
+  // change.
+  // -That set echo path change flags are received and continues to be received
+  // as long as echo path changes are flagged.
+  // -That set echo path change flags are no longer received when echo path
+  // change events stop being flagged.
+  enum class EchoPathChangeTestVariant { kNone, kOneSticky, kOneNonSticky };
+
+  void RunEchoPathChangeVerificationTest(
+      EchoPathChangeTestVariant echo_path_change_test_variant) {
+    constexpr size_t kNumFullBlocksPerFrame = 160 / kBlockSize;
+    constexpr size_t kExpectedNumBlocksToProcess =
+        (kNumFramesToProcess * 160) / kBlockSize;
+    std::unique_ptr<testing::StrictMock<webrtc::test::MockBlockProcessor>>
+        block_processor_mock(
+            new StrictMock<webrtc::test::MockBlockProcessor>());
+    EXPECT_CALL(*block_processor_mock, BufferRender(_))
+        .Times(kExpectedNumBlocksToProcess);
+    EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(_)).Times(0);
+
+    switch (echo_path_change_test_variant) {
+      case EchoPathChangeTestVariant::kNone:
+        EXPECT_CALL(*block_processor_mock, ProcessCapture(false, _, _, _))
+            .Times(kExpectedNumBlocksToProcess);
+        break;
+      case EchoPathChangeTestVariant::kOneSticky:
+        EXPECT_CALL(*block_processor_mock, ProcessCapture(true, _, _, _))
+            .Times(kExpectedNumBlocksToProcess);
+        break;
+      case EchoPathChangeTestVariant::kOneNonSticky:
+        EXPECT_CALL(*block_processor_mock, ProcessCapture(true, _, _, _))
+            .Times(kNumFullBlocksPerFrame);
+        EXPECT_CALL(*block_processor_mock, ProcessCapture(false, _, _, _))
+            .Times(kExpectedNumBlocksToProcess - kNumFullBlocksPerFrame);
+        break;
+    }
+
+    EchoCanceller3 aec3(EchoCanceller3Config(),
+                        /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+                        1, 1);
+    aec3.SetBlockProcessorForTesting(std::move(block_processor_mock));
+
+    for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
+         ++frame_index) {
+      bool echo_path_change = false;
+      switch (echo_path_change_test_variant) {
+        case EchoPathChangeTestVariant::kNone:
+          break;
+        case EchoPathChangeTestVariant::kOneSticky:
+          echo_path_change = true;
+          break;
+        case EchoPathChangeTestVariant::kOneNonSticky:
+          if (frame_index == 0) {
+            echo_path_change = true;
+          }
+          break;
+      }
+
+      aec3.AnalyzeCapture(&capture_buffer_);
+      OptionalBandSplit();
+
+      PopulateInputFrame(frame_length_, num_bands_, frame_index,
+                         &capture_buffer_.split_bands(0)[0], 0);
+      PopulateInputFrame(frame_length_, frame_index,
+                         &render_buffer_.channels()[0][0], 0);
+
+      aec3.AnalyzeRender(&render_buffer_);
+      aec3.ProcessCapture(&capture_buffer_, echo_path_change);
+    }
+  }
+
+  // Test for verifying that echo leakage information is being properly passed
+  // to the processor.
+  // The cases tested are:
+  // -That no method calls are received when they should not.
+  // -That false values are received each time they are flagged.
+  // -That true values are received each time they are flagged.
+  // -That a false value is received when flagged after a true value has been
+  // flagged.
+  enum class EchoLeakageTestVariant {
+    kNone,
+    kFalseSticky,
+    kTrueSticky,
+    kTrueNonSticky
+  };
+
+  void RunEchoLeakageVerificationTest(
+      EchoLeakageTestVariant leakage_report_variant) {
+    constexpr size_t kExpectedNumBlocksToProcess =
+        (kNumFramesToProcess * 160) / kBlockSize;
+    std::unique_ptr<testing::StrictMock<webrtc::test::MockBlockProcessor>>
+        block_processor_mock(
+            new StrictMock<webrtc::test::MockBlockProcessor>());
+    EXPECT_CALL(*block_processor_mock, BufferRender(_))
+        .Times(kExpectedNumBlocksToProcess);
+    EXPECT_CALL(*block_processor_mock, ProcessCapture(_, _, _, _))
+        .Times(kExpectedNumBlocksToProcess);
+
+    switch (leakage_report_variant) {
+      case EchoLeakageTestVariant::kNone:
+        EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(_)).Times(0);
+        break;
+      case EchoLeakageTestVariant::kFalseSticky:
+        EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(false))
+            .Times(1);
+        break;
+      case EchoLeakageTestVariant::kTrueSticky:
+        EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(true))
+            .Times(1);
+        break;
+      case EchoLeakageTestVariant::kTrueNonSticky: {
+        ::testing::InSequence s;
+        EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(true))
+            .Times(1);
+        EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(false))
+            .Times(kNumFramesToProcess - 1);
+      } break;
+    }
+
+    EchoCanceller3 aec3(EchoCanceller3Config(),
+                        /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+                        1, 1);
+    aec3.SetBlockProcessorForTesting(std::move(block_processor_mock));
+
+    for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
+         ++frame_index) {
+      switch (leakage_report_variant) {
+        case EchoLeakageTestVariant::kNone:
+          break;
+        case EchoLeakageTestVariant::kFalseSticky:
+          if (frame_index == 0) {
+            aec3.UpdateEchoLeakageStatus(false);
+          }
+          break;
+        case EchoLeakageTestVariant::kTrueSticky:
+          if (frame_index == 0) {
+            aec3.UpdateEchoLeakageStatus(true);
+          }
+          break;
+        case EchoLeakageTestVariant::kTrueNonSticky:
+          if (frame_index == 0) {
+            aec3.UpdateEchoLeakageStatus(true);
+          } else {
+            aec3.UpdateEchoLeakageStatus(false);
+          }
+          break;
+      }
+
+      aec3.AnalyzeCapture(&capture_buffer_);
+      OptionalBandSplit();
+
+      PopulateInputFrame(frame_length_, num_bands_, frame_index,
+                         &capture_buffer_.split_bands(0)[0], 0);
+      PopulateInputFrame(frame_length_, frame_index,
+                         &render_buffer_.channels()[0][0], 0);
+
+      aec3.AnalyzeRender(&render_buffer_);
+      aec3.ProcessCapture(&capture_buffer_, false);
+    }
+  }
+
+  // This verifies that saturation information is properly passed to the
+  // BlockProcessor.
+  // The cases tested are:
+  // -That no saturation event is passed to the processor if there is no
+  // saturation.
+  // -That one frame with one negative saturated sample value is reported to be
+  // saturated and that following non-saturated frames are properly reported as
+  // not being saturated.
+  // -That one frame with one positive saturated sample value is reported to be
+  // saturated and that following non-saturated frames are properly reported as
+  // not being saturated.
+  enum class SaturationTestVariant { kNone, kOneNegative, kOnePositive };
+
+  void RunCaptureSaturationVerificationTest(
+      SaturationTestVariant saturation_variant) {
+    const size_t kNumFullBlocksPerFrame = 160 / kBlockSize;
+    const size_t kExpectedNumBlocksToProcess =
+        (kNumFramesToProcess * 160) / kBlockSize;
+    std::unique_ptr<testing::StrictMock<webrtc::test::MockBlockProcessor>>
+        block_processor_mock(
+            new StrictMock<webrtc::test::MockBlockProcessor>());
+    EXPECT_CALL(*block_processor_mock, BufferRender(_))
+        .Times(kExpectedNumBlocksToProcess);
+    EXPECT_CALL(*block_processor_mock, UpdateEchoLeakageStatus(_)).Times(0);
+
+    switch (saturation_variant) {
+      case SaturationTestVariant::kNone:
+        EXPECT_CALL(*block_processor_mock, ProcessCapture(_, false, _, _))
+            .Times(kExpectedNumBlocksToProcess);
+        break;
+      case SaturationTestVariant::kOneNegative: {
+        ::testing::InSequence s;
+        EXPECT_CALL(*block_processor_mock, ProcessCapture(_, true, _, _))
+            .Times(kNumFullBlocksPerFrame);
+        EXPECT_CALL(*block_processor_mock, ProcessCapture(_, false, _, _))
+            .Times(kExpectedNumBlocksToProcess - kNumFullBlocksPerFrame);
+      } break;
+      case SaturationTestVariant::kOnePositive: {
+        ::testing::InSequence s;
+        EXPECT_CALL(*block_processor_mock, ProcessCapture(_, true, _, _))
+            .Times(kNumFullBlocksPerFrame);
+        EXPECT_CALL(*block_processor_mock, ProcessCapture(_, false, _, _))
+            .Times(kExpectedNumBlocksToProcess - kNumFullBlocksPerFrame);
+      } break;
+    }
+
+    EchoCanceller3 aec3(EchoCanceller3Config(),
+                        /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+                        1, 1);
+    aec3.SetBlockProcessorForTesting(std::move(block_processor_mock));
+    for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
+         ++frame_index) {
+      for (int k = 0; k < fullband_frame_length_; ++k) {
+        capture_buffer_.channels()[0][k] = 0.f;
+      }
+      switch (saturation_variant) {
+        case SaturationTestVariant::kNone:
+          break;
+        case SaturationTestVariant::kOneNegative:
+          if (frame_index == 0) {
+            capture_buffer_.channels()[0][10] = -32768.f;
+          }
+          break;
+        case SaturationTestVariant::kOnePositive:
+          if (frame_index == 0) {
+            capture_buffer_.channels()[0][10] = 32767.f;
+          }
+          break;
+      }
+
+      aec3.AnalyzeCapture(&capture_buffer_);
+      OptionalBandSplit();
+
+      PopulateInputFrame(frame_length_, num_bands_, frame_index,
+                         &capture_buffer_.split_bands(0)[0], 0);
+      PopulateInputFrame(frame_length_, num_bands_, frame_index,
+                         &render_buffer_.split_bands(0)[0], 0);
+
+      aec3.AnalyzeRender(&render_buffer_);
+      aec3.ProcessCapture(&capture_buffer_, false);
+    }
+  }
+
+  // This test verifies that the swapqueue is able to handle jitter in the
+  // capture and render API calls.
+  void RunRenderSwapQueueVerificationTest() {
+    const EchoCanceller3Config config;
+    EchoCanceller3 aec3(config, /*multichannel_config=*/absl::nullopt,
+                        sample_rate_hz_, 1, 1);
+    aec3.SetBlockProcessorForTesting(
+        std::make_unique<RenderTransportVerificationProcessor>(num_bands_));
+
+    std::vector<std::vector<float>> render_input(1);
+    std::vector<float> capture_output;
+
+    for (size_t frame_index = 0; frame_index < kRenderTransferQueueSizeFrames;
+         ++frame_index) {
+      if (sample_rate_hz_ > 16000) {
+        render_buffer_.SplitIntoFrequencyBands();
+      }
+      PopulateInputFrame(frame_length_, num_bands_, frame_index,
+                         &render_buffer_.split_bands(0)[0], 0);
+
+      if (sample_rate_hz_ > 16000) {
+        render_buffer_.SplitIntoFrequencyBands();
+      }
+
+      for (size_t k = 0; k < frame_length_; ++k) {
+        render_input[0].push_back(render_buffer_.split_bands(0)[0][k]);
+      }
+      aec3.AnalyzeRender(&render_buffer_);
+    }
+
+    for (size_t frame_index = 0; frame_index < kRenderTransferQueueSizeFrames;
+         ++frame_index) {
+      aec3.AnalyzeCapture(&capture_buffer_);
+      if (sample_rate_hz_ > 16000) {
+        capture_buffer_.SplitIntoFrequencyBands();
+      }
+
+      PopulateInputFrame(frame_length_, num_bands_, frame_index,
+                         &capture_buffer_.split_bands(0)[0], 0);
+
+      aec3.ProcessCapture(&capture_buffer_, false);
+      for (size_t k = 0; k < frame_length_; ++k) {
+        capture_output.push_back(capture_buffer_.split_bands(0)[0][k]);
+      }
+    }
+
+    EXPECT_TRUE(
+        VerifyOutputFrameBitexactness(render_input[0], capture_output, -64));
+  }
+
+  // This test verifies that a buffer overrun in the render swapqueue is
+  // properly reported.
+  void RunRenderPipelineSwapQueueOverrunReturnValueTest() {
+    EchoCanceller3 aec3(EchoCanceller3Config(),
+                        /*multichannel_config=*/absl::nullopt, sample_rate_hz_,
+                        1, 1);
+
+    constexpr size_t kRenderTransferQueueSize = 30;
+    for (size_t k = 0; k < 2; ++k) {
+      for (size_t frame_index = 0; frame_index < kRenderTransferQueueSize;
+           ++frame_index) {
+        if (sample_rate_hz_ > 16000) {
+          render_buffer_.SplitIntoFrequencyBands();
+        }
+        PopulateInputFrame(frame_length_, frame_index,
+                           &render_buffer_.channels()[0][0], 0);
+
+        aec3.AnalyzeRender(&render_buffer_);
+      }
+    }
+  }
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+  // Verifies the that the check for the number of bands in the AnalyzeRender
+  // input is correct by adjusting the sample rates of EchoCanceller3 and the
+  // input AudioBuffer to have a different number of bands.
+  void RunAnalyzeRenderNumBandsCheckVerification() {
+    // Set aec3_sample_rate_hz to be different from sample_rate_hz_ in such a
+    // way that the number of bands for the rates are different.
+    const int aec3_sample_rate_hz = sample_rate_hz_ == 48000 ? 32000 : 48000;
+    EchoCanceller3 aec3(EchoCanceller3Config(),
+                        /*multichannel_config=*/absl::nullopt,
+                        aec3_sample_rate_hz, 1, 1);
+    PopulateInputFrame(frame_length_, 0, &render_buffer_.channels_f()[0][0], 0);
+
+    EXPECT_DEATH(aec3.AnalyzeRender(&render_buffer_), "");
+  }
+
+  // Verifies the that the check for the number of bands in the ProcessCapture
+  // input is correct by adjusting the sample rates of EchoCanceller3 and the
+  // input AudioBuffer to have a different number of bands.
+  void RunProcessCaptureNumBandsCheckVerification() {
+    // Set aec3_sample_rate_hz to be different from sample_rate_hz_ in such a
+    // way that the number of bands for the rates are different.
+    const int aec3_sample_rate_hz = sample_rate_hz_ == 48000 ? 32000 : 48000;
+    EchoCanceller3 aec3(EchoCanceller3Config(),
+                        /*multichannel_config=*/absl::nullopt,
+                        aec3_sample_rate_hz, 1, 1);
+    PopulateInputFrame(frame_length_, num_bands_, 0,
+                       &capture_buffer_.split_bands_f(0)[0], 100);
+    EXPECT_DEATH(aec3.ProcessCapture(&capture_buffer_, false), "");
+  }
+
+#endif
+
+ private:
+  void OptionalBandSplit() {
+    if (sample_rate_hz_ > 16000) {
+      capture_buffer_.SplitIntoFrequencyBands();
+      render_buffer_.SplitIntoFrequencyBands();
+    }
+  }
+
+  static constexpr size_t kNumFramesToProcess = 20;
+  const int sample_rate_hz_;
+  const size_t num_bands_;
+  const size_t frame_length_;
+  const int fullband_frame_length_;
+  AudioBuffer capture_buffer_;
+  AudioBuffer render_buffer_;
+};
+
+TEST(EchoCanceller3Buffering, CaptureBitexactness) {
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    EchoCanceller3Tester(rate).RunCaptureTransportVerificationTest();
+  }
+}
+
+TEST(EchoCanceller3Buffering, RenderBitexactness) {
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    EchoCanceller3Tester(rate).RunRenderTransportVerificationTest();
+  }
+}
+
+TEST(EchoCanceller3Buffering, RenderSwapQueue) {
+  EchoCanceller3Tester(16000).RunRenderSwapQueueVerificationTest();
+}
+
+TEST(EchoCanceller3Buffering, RenderSwapQueueOverrunReturnValue) {
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    EchoCanceller3Tester(rate)
+        .RunRenderPipelineSwapQueueOverrunReturnValueTest();
+  }
+}
+
+TEST(EchoCanceller3Messaging, CaptureSaturation) {
+  auto variants = {EchoCanceller3Tester::SaturationTestVariant::kNone,
+                   EchoCanceller3Tester::SaturationTestVariant::kOneNegative,
+                   EchoCanceller3Tester::SaturationTestVariant::kOnePositive};
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto variant : variants) {
+      SCOPED_TRACE(ProduceDebugText(rate, static_cast<int>(variant)));
+      EchoCanceller3Tester(rate).RunCaptureSaturationVerificationTest(variant);
+    }
+  }
+}
+
+TEST(EchoCanceller3Messaging, EchoPathChange) {
+  auto variants = {
+      EchoCanceller3Tester::EchoPathChangeTestVariant::kNone,
+      EchoCanceller3Tester::EchoPathChangeTestVariant::kOneSticky,
+      EchoCanceller3Tester::EchoPathChangeTestVariant::kOneNonSticky};
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto variant : variants) {
+      SCOPED_TRACE(ProduceDebugText(rate, static_cast<int>(variant)));
+      EchoCanceller3Tester(rate).RunEchoPathChangeVerificationTest(variant);
+    }
+  }
+}
+
+TEST(EchoCanceller3Messaging, EchoLeakage) {
+  auto variants = {
+      EchoCanceller3Tester::EchoLeakageTestVariant::kNone,
+      EchoCanceller3Tester::EchoLeakageTestVariant::kFalseSticky,
+      EchoCanceller3Tester::EchoLeakageTestVariant::kTrueSticky,
+      EchoCanceller3Tester::EchoLeakageTestVariant::kTrueNonSticky};
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto variant : variants) {
+      SCOPED_TRACE(ProduceDebugText(rate, static_cast<int>(variant)));
+      EchoCanceller3Tester(rate).RunEchoLeakageVerificationTest(variant);
+    }
+  }
+}
+
+// Tests the parameter functionality for the field trial override for the
+// anti-howling gain.
+TEST(EchoCanceller3FieldTrials, Aec3SuppressorAntiHowlingGainOverride) {
+  EchoCanceller3Config default_config;
+  EchoCanceller3Config adjusted_config = AdjustConfig(default_config);
+  ASSERT_EQ(
+      default_config.suppressor.high_bands_suppression.anti_howling_gain,
+      adjusted_config.suppressor.high_bands_suppression.anti_howling_gain);
+
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Aec3SuppressorAntiHowlingGainOverride/0.02/");
+  adjusted_config = AdjustConfig(default_config);
+
+  ASSERT_NE(
+      default_config.suppressor.high_bands_suppression.anti_howling_gain,
+      adjusted_config.suppressor.high_bands_suppression.anti_howling_gain);
+  EXPECT_FLOAT_EQ(
+      0.02f,
+      adjusted_config.suppressor.high_bands_suppression.anti_howling_gain);
+}
+
+// Tests the field trial override for the enforcement of a low active render
+// limit.
+TEST(EchoCanceller3FieldTrials, Aec3EnforceLowActiveRenderLimit) {
+  EchoCanceller3Config default_config;
+  EchoCanceller3Config adjusted_config = AdjustConfig(default_config);
+  ASSERT_EQ(default_config.render_levels.active_render_limit,
+            adjusted_config.render_levels.active_render_limit);
+
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Aec3EnforceLowActiveRenderLimit/Enabled/");
+  adjusted_config = AdjustConfig(default_config);
+
+  ASSERT_NE(default_config.render_levels.active_render_limit,
+            adjusted_config.render_levels.active_render_limit);
+  EXPECT_FLOAT_EQ(50.f, adjusted_config.render_levels.active_render_limit);
+}
+
+// Testing the field trial-based override of the suppressor parameters for a
+// joint passing of all parameters.
+TEST(EchoCanceller3FieldTrials, Aec3SuppressorTuningOverrideAllParams) {
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Aec3SuppressorTuningOverride/"
+      "nearend_tuning_mask_lf_enr_transparent:0.1,nearend_tuning_mask_lf_enr_"
+      "suppress:0.2,nearend_tuning_mask_hf_enr_transparent:0.3,nearend_tuning_"
+      "mask_hf_enr_suppress:0.4,nearend_tuning_max_inc_factor:0.5,nearend_"
+      "tuning_max_dec_factor_lf:0.6,normal_tuning_mask_lf_enr_transparent:0.7,"
+      "normal_tuning_mask_lf_enr_suppress:0.8,normal_tuning_mask_hf_enr_"
+      "transparent:0.9,normal_tuning_mask_hf_enr_suppress:1.0,normal_tuning_"
+      "max_inc_factor:1.1,normal_tuning_max_dec_factor_lf:1.2,dominant_nearend_"
+      "detection_enr_threshold:1.3,dominant_nearend_detection_enr_exit_"
+      "threshold:1.4,dominant_nearend_detection_snr_threshold:1.5,dominant_"
+      "nearend_detection_hold_duration:10,dominant_nearend_detection_trigger_"
+      "threshold:11/");
+
+  EchoCanceller3Config default_config;
+  EchoCanceller3Config adjusted_config = AdjustConfig(default_config);
+
+  ASSERT_NE(adjusted_config.suppressor.nearend_tuning.mask_lf.enr_transparent,
+            default_config.suppressor.nearend_tuning.mask_lf.enr_transparent);
+  ASSERT_NE(adjusted_config.suppressor.nearend_tuning.mask_lf.enr_suppress,
+            default_config.suppressor.nearend_tuning.mask_lf.enr_suppress);
+  ASSERT_NE(adjusted_config.suppressor.nearend_tuning.mask_hf.enr_transparent,
+            default_config.suppressor.nearend_tuning.mask_hf.enr_transparent);
+  ASSERT_NE(adjusted_config.suppressor.nearend_tuning.mask_hf.enr_suppress,
+            default_config.suppressor.nearend_tuning.mask_hf.enr_suppress);
+  ASSERT_NE(adjusted_config.suppressor.nearend_tuning.max_inc_factor,
+            default_config.suppressor.nearend_tuning.max_inc_factor);
+  ASSERT_NE(adjusted_config.suppressor.nearend_tuning.max_dec_factor_lf,
+            default_config.suppressor.nearend_tuning.max_dec_factor_lf);
+  ASSERT_NE(adjusted_config.suppressor.normal_tuning.mask_lf.enr_transparent,
+            default_config.suppressor.normal_tuning.mask_lf.enr_transparent);
+  ASSERT_NE(adjusted_config.suppressor.normal_tuning.mask_lf.enr_suppress,
+            default_config.suppressor.normal_tuning.mask_lf.enr_suppress);
+  ASSERT_NE(adjusted_config.suppressor.normal_tuning.mask_hf.enr_transparent,
+            default_config.suppressor.normal_tuning.mask_hf.enr_transparent);
+  ASSERT_NE(adjusted_config.suppressor.normal_tuning.mask_hf.enr_suppress,
+            default_config.suppressor.normal_tuning.mask_hf.enr_suppress);
+  ASSERT_NE(adjusted_config.suppressor.normal_tuning.max_inc_factor,
+            default_config.suppressor.normal_tuning.max_inc_factor);
+  ASSERT_NE(adjusted_config.suppressor.normal_tuning.max_dec_factor_lf,
+            default_config.suppressor.normal_tuning.max_dec_factor_lf);
+  ASSERT_NE(adjusted_config.suppressor.dominant_nearend_detection.enr_threshold,
+            default_config.suppressor.dominant_nearend_detection.enr_threshold);
+  ASSERT_NE(
+      adjusted_config.suppressor.dominant_nearend_detection.enr_exit_threshold,
+      default_config.suppressor.dominant_nearend_detection.enr_exit_threshold);
+  ASSERT_NE(adjusted_config.suppressor.dominant_nearend_detection.snr_threshold,
+            default_config.suppressor.dominant_nearend_detection.snr_threshold);
+  ASSERT_NE(adjusted_config.suppressor.dominant_nearend_detection.hold_duration,
+            default_config.suppressor.dominant_nearend_detection.hold_duration);
+  ASSERT_NE(
+      adjusted_config.suppressor.dominant_nearend_detection.trigger_threshold,
+      default_config.suppressor.dominant_nearend_detection.trigger_threshold);
+
+  EXPECT_FLOAT_EQ(
+      adjusted_config.suppressor.nearend_tuning.mask_lf.enr_transparent, 0.1);
+  EXPECT_FLOAT_EQ(
+      adjusted_config.suppressor.nearend_tuning.mask_lf.enr_suppress, 0.2);
+  EXPECT_FLOAT_EQ(
+      adjusted_config.suppressor.nearend_tuning.mask_hf.enr_transparent, 0.3);
+  EXPECT_FLOAT_EQ(
+      adjusted_config.suppressor.nearend_tuning.mask_hf.enr_suppress, 0.4);
+  EXPECT_FLOAT_EQ(adjusted_config.suppressor.nearend_tuning.max_inc_factor,
+                  0.5);
+  EXPECT_FLOAT_EQ(adjusted_config.suppressor.nearend_tuning.max_dec_factor_lf,
+                  0.6);
+  EXPECT_FLOAT_EQ(
+      adjusted_config.suppressor.normal_tuning.mask_lf.enr_transparent, 0.7);
+  EXPECT_FLOAT_EQ(adjusted_config.suppressor.normal_tuning.mask_lf.enr_suppress,
+                  0.8);
+  EXPECT_FLOAT_EQ(
+      adjusted_config.suppressor.normal_tuning.mask_hf.enr_transparent, 0.9);
+  EXPECT_FLOAT_EQ(adjusted_config.suppressor.normal_tuning.mask_hf.enr_suppress,
+                  1.0);
+  EXPECT_FLOAT_EQ(adjusted_config.suppressor.normal_tuning.max_inc_factor, 1.1);
+  EXPECT_FLOAT_EQ(adjusted_config.suppressor.normal_tuning.max_dec_factor_lf,
+                  1.2);
+  EXPECT_FLOAT_EQ(
+      adjusted_config.suppressor.dominant_nearend_detection.enr_threshold, 1.3);
+  EXPECT_FLOAT_EQ(
+      adjusted_config.suppressor.dominant_nearend_detection.enr_exit_threshold,
+      1.4);
+  EXPECT_FLOAT_EQ(
+      adjusted_config.suppressor.dominant_nearend_detection.snr_threshold, 1.5);
+  EXPECT_EQ(adjusted_config.suppressor.dominant_nearend_detection.hold_duration,
+            10);
+  EXPECT_EQ(
+      adjusted_config.suppressor.dominant_nearend_detection.trigger_threshold,
+      11);
+}
+
+// Testing the field trial-based override of the suppressor parameters for
+// passing one parameter.
+TEST(EchoCanceller3FieldTrials, Aec3SuppressorTuningOverrideOneParam) {
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Aec3SuppressorTuningOverride/nearend_tuning_max_inc_factor:0.5/");
+
+  EchoCanceller3Config default_config;
+  EchoCanceller3Config adjusted_config = AdjustConfig(default_config);
+
+  ASSERT_EQ(adjusted_config.suppressor.nearend_tuning.mask_lf.enr_transparent,
+            default_config.suppressor.nearend_tuning.mask_lf.enr_transparent);
+  ASSERT_EQ(adjusted_config.suppressor.nearend_tuning.mask_lf.enr_suppress,
+            default_config.suppressor.nearend_tuning.mask_lf.enr_suppress);
+  ASSERT_EQ(adjusted_config.suppressor.nearend_tuning.mask_hf.enr_transparent,
+            default_config.suppressor.nearend_tuning.mask_hf.enr_transparent);
+  ASSERT_EQ(adjusted_config.suppressor.nearend_tuning.mask_hf.enr_suppress,
+            default_config.suppressor.nearend_tuning.mask_hf.enr_suppress);
+  ASSERT_EQ(adjusted_config.suppressor.nearend_tuning.max_dec_factor_lf,
+            default_config.suppressor.nearend_tuning.max_dec_factor_lf);
+  ASSERT_EQ(adjusted_config.suppressor.normal_tuning.mask_lf.enr_transparent,
+            default_config.suppressor.normal_tuning.mask_lf.enr_transparent);
+  ASSERT_EQ(adjusted_config.suppressor.normal_tuning.mask_lf.enr_suppress,
+            default_config.suppressor.normal_tuning.mask_lf.enr_suppress);
+  ASSERT_EQ(adjusted_config.suppressor.normal_tuning.mask_hf.enr_transparent,
+            default_config.suppressor.normal_tuning.mask_hf.enr_transparent);
+  ASSERT_EQ(adjusted_config.suppressor.normal_tuning.mask_hf.enr_suppress,
+            default_config.suppressor.normal_tuning.mask_hf.enr_suppress);
+  ASSERT_EQ(adjusted_config.suppressor.normal_tuning.max_inc_factor,
+            default_config.suppressor.normal_tuning.max_inc_factor);
+  ASSERT_EQ(adjusted_config.suppressor.normal_tuning.max_dec_factor_lf,
+            default_config.suppressor.normal_tuning.max_dec_factor_lf);
+  ASSERT_EQ(adjusted_config.suppressor.dominant_nearend_detection.enr_threshold,
+            default_config.suppressor.dominant_nearend_detection.enr_threshold);
+  ASSERT_EQ(
+      adjusted_config.suppressor.dominant_nearend_detection.enr_exit_threshold,
+      default_config.suppressor.dominant_nearend_detection.enr_exit_threshold);
+  ASSERT_EQ(adjusted_config.suppressor.dominant_nearend_detection.snr_threshold,
+            default_config.suppressor.dominant_nearend_detection.snr_threshold);
+  ASSERT_EQ(adjusted_config.suppressor.dominant_nearend_detection.hold_duration,
+            default_config.suppressor.dominant_nearend_detection.hold_duration);
+  ASSERT_EQ(
+      adjusted_config.suppressor.dominant_nearend_detection.trigger_threshold,
+      default_config.suppressor.dominant_nearend_detection.trigger_threshold);
+
+  ASSERT_NE(adjusted_config.suppressor.nearend_tuning.max_inc_factor,
+            default_config.suppressor.nearend_tuning.max_inc_factor);
+
+  EXPECT_FLOAT_EQ(adjusted_config.suppressor.nearend_tuning.max_inc_factor,
+                  0.5);
+}
+
+// Testing the field trial-based that override the exponential decay parameters.
+TEST(EchoCanceller3FieldTrials, Aec3UseNearendReverb) {
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Aec3UseNearendReverbLen/default_len:0.9,nearend_len:0.8/");
+  EchoCanceller3Config default_config;
+  EchoCanceller3Config adjusted_config = AdjustConfig(default_config);
+  EXPECT_FLOAT_EQ(adjusted_config.ep_strength.default_len, 0.9);
+  EXPECT_FLOAT_EQ(adjusted_config.ep_strength.nearend_len, 0.8);
+}
+
+TEST(EchoCanceller3, DetectionOfProperStereo) {
+  constexpr int kSampleRateHz = 16000;
+  constexpr int kNumChannels = 2;
+  AudioBuffer buffer(/*input_rate=*/kSampleRateHz,
+                     /*input_num_channels=*/kNumChannels,
+                     /*input_rate=*/kSampleRateHz,
+                     /*buffer_num_channels=*/kNumChannels,
+                     /*output_rate=*/kSampleRateHz,
+                     /*output_num_channels=*/kNumChannels);
+
+  constexpr size_t kNumBlocksForMonoConfig = 1;
+  constexpr size_t kNumBlocksForSurroundConfig = 2;
+  EchoCanceller3Config mono_config;
+  absl::optional<EchoCanceller3Config> multichannel_config;
+
+  mono_config.multi_channel.detect_stereo_content = true;
+  mono_config.multi_channel.stereo_detection_threshold = 0.0f;
+  mono_config.multi_channel.stereo_detection_hysteresis_seconds = 0.0f;
+  multichannel_config = mono_config;
+  mono_config.filter.coarse_initial.length_blocks = kNumBlocksForMonoConfig;
+  multichannel_config->filter.coarse_initial.length_blocks =
+      kNumBlocksForSurroundConfig;
+
+  EchoCanceller3 aec3(mono_config, multichannel_config,
+                      /*sample_rate_hz=*/kSampleRateHz,
+                      /*num_render_channels=*/kNumChannels,
+                      /*num_capture_input_channels=*/kNumChannels);
+
+  EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting());
+  EXPECT_EQ(
+      aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+      kNumBlocksForMonoConfig);
+
+  RunAecInStereo(buffer, aec3, 100.0f, 100.0f);
+  EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting());
+  EXPECT_EQ(
+      aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+      kNumBlocksForMonoConfig);
+
+  RunAecInStereo(buffer, aec3, 100.0f, 101.0f);
+  EXPECT_TRUE(aec3.StereoRenderProcessingActiveForTesting());
+  EXPECT_EQ(
+      aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+      kNumBlocksForSurroundConfig);
+}
+
+TEST(EchoCanceller3, DetectionOfProperStereoUsingThreshold) {
+  constexpr int kSampleRateHz = 16000;
+  constexpr int kNumChannels = 2;
+  AudioBuffer buffer(/*input_rate=*/kSampleRateHz,
+                     /*input_num_channels=*/kNumChannels,
+                     /*input_rate=*/kSampleRateHz,
+                     /*buffer_num_channels=*/kNumChannels,
+                     /*output_rate=*/kSampleRateHz,
+                     /*output_num_channels=*/kNumChannels);
+
+  constexpr size_t kNumBlocksForMonoConfig = 1;
+  constexpr size_t kNumBlocksForSurroundConfig = 2;
+  EchoCanceller3Config mono_config;
+  absl::optional<EchoCanceller3Config> multichannel_config;
+
+  constexpr float kStereoDetectionThreshold = 2.0f;
+  mono_config.multi_channel.detect_stereo_content = true;
+  mono_config.multi_channel.stereo_detection_threshold =
+      kStereoDetectionThreshold;
+  mono_config.multi_channel.stereo_detection_hysteresis_seconds = 0.0f;
+  multichannel_config = mono_config;
+  mono_config.filter.coarse_initial.length_blocks = kNumBlocksForMonoConfig;
+  multichannel_config->filter.coarse_initial.length_blocks =
+      kNumBlocksForSurroundConfig;
+
+  EchoCanceller3 aec3(mono_config, multichannel_config,
+                      /*sample_rate_hz=*/kSampleRateHz,
+                      /*num_render_channels=*/kNumChannels,
+                      /*num_capture_input_channels=*/kNumChannels);
+
+  EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting());
+  EXPECT_EQ(
+      aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+      kNumBlocksForMonoConfig);
+
+  RunAecInStereo(buffer, aec3, 100.0f,
+                 100.0f + kStereoDetectionThreshold - 1.0f);
+  EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting());
+  EXPECT_EQ(
+      aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+      kNumBlocksForMonoConfig);
+
+  RunAecInStereo(buffer, aec3, 100.0f,
+                 100.0f + kStereoDetectionThreshold + 10.0f);
+  EXPECT_TRUE(aec3.StereoRenderProcessingActiveForTesting());
+  EXPECT_EQ(
+      aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+      kNumBlocksForSurroundConfig);
+}
+
+TEST(EchoCanceller3, DetectionOfProperStereoUsingHysteresis) {
+  constexpr int kSampleRateHz = 16000;
+  constexpr int kNumChannels = 2;
+  AudioBuffer buffer(/*input_rate=*/kSampleRateHz,
+                     /*input_num_channels=*/kNumChannels,
+                     /*input_rate=*/kSampleRateHz,
+                     /*buffer_num_channels=*/kNumChannels,
+                     /*output_rate=*/kSampleRateHz,
+                     /*output_num_channels=*/kNumChannels);
+
+  constexpr size_t kNumBlocksForMonoConfig = 1;
+  constexpr size_t kNumBlocksForSurroundConfig = 2;
+  EchoCanceller3Config mono_config;
+  absl::optional<EchoCanceller3Config> surround_config;
+
+  mono_config.multi_channel.detect_stereo_content = true;
+  mono_config.multi_channel.stereo_detection_hysteresis_seconds = 0.5f;
+  surround_config = mono_config;
+  mono_config.filter.coarse_initial.length_blocks = kNumBlocksForMonoConfig;
+  surround_config->filter.coarse_initial.length_blocks =
+      kNumBlocksForSurroundConfig;
+
+  EchoCanceller3 aec3(mono_config, surround_config,
+                      /*sample_rate_hz=*/kSampleRateHz,
+                      /*num_render_channels=*/kNumChannels,
+                      /*num_capture_input_channels=*/kNumChannels);
+
+  EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting());
+  EXPECT_EQ(
+      aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+      kNumBlocksForMonoConfig);
+
+  RunAecInStereo(buffer, aec3, 100.0f, 100.0f);
+  EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting());
+  EXPECT_EQ(
+      aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+      kNumBlocksForMonoConfig);
+
+  constexpr int kNumFramesPerSecond = 100;
+  for (int k = 0;
+       k < static_cast<int>(
+               kNumFramesPerSecond *
+               mono_config.multi_channel.stereo_detection_hysteresis_seconds);
+       ++k) {
+    RunAecInStereo(buffer, aec3, 100.0f, 101.0f);
+    EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting());
+    EXPECT_EQ(
+        aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+        kNumBlocksForMonoConfig);
+  }
+
+  RunAecInStereo(buffer, aec3, 100.0f, 101.0f);
+  EXPECT_TRUE(aec3.StereoRenderProcessingActiveForTesting());
+  EXPECT_EQ(
+      aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+      kNumBlocksForSurroundConfig);
+}
+
+TEST(EchoCanceller3, StereoContentDetectionForMonoSignals) {
+  constexpr int kSampleRateHz = 16000;
+  constexpr int kNumChannels = 2;
+  AudioBuffer buffer(/*input_rate=*/kSampleRateHz,
+                     /*input_num_channels=*/kNumChannels,
+                     /*input_rate=*/kSampleRateHz,
+                     /*buffer_num_channels=*/kNumChannels,
+                     /*output_rate=*/kSampleRateHz,
+                     /*output_num_channels=*/kNumChannels);
+
+  constexpr size_t kNumBlocksForMonoConfig = 1;
+  constexpr size_t kNumBlocksForSurroundConfig = 2;
+  EchoCanceller3Config mono_config;
+  absl::optional<EchoCanceller3Config> multichannel_config;
+
+  for (bool detect_stereo_content : {false, true}) {
+    mono_config.multi_channel.detect_stereo_content = detect_stereo_content;
+    multichannel_config = mono_config;
+    mono_config.filter.coarse_initial.length_blocks = kNumBlocksForMonoConfig;
+    multichannel_config->filter.coarse_initial.length_blocks =
+        kNumBlocksForSurroundConfig;
+
+    AudioBuffer mono_buffer(/*input_rate=*/kSampleRateHz,
+                            /*input_num_channels=*/1,
+                            /*input_rate=*/kSampleRateHz,
+                            /*buffer_num_channels=*/1,
+                            /*output_rate=*/kSampleRateHz,
+                            /*output_num_channels=*/1);
+
+    EchoCanceller3 aec3(mono_config, multichannel_config,
+                        /*sample_rate_hz=*/kSampleRateHz,
+                        /*num_render_channels=*/1,
+                        /*num_capture_input_channels=*/1);
+
+    EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting());
+    EXPECT_EQ(
+        aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+        kNumBlocksForMonoConfig);
+
+    RunAecInSMono(mono_buffer, aec3, 100.0f);
+    EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting());
+    EXPECT_EQ(
+        aec3.GetActiveConfigForTesting().filter.coarse_initial.length_blocks,
+        kNumBlocksForMonoConfig);
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+TEST(EchoCanceller3InputCheckDeathTest, WrongCaptureNumBandsCheckVerification) {
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    EchoCanceller3Tester(rate).RunProcessCaptureNumBandsCheckVerification();
+  }
+}
+
+// Verifiers that the verification for null input to the capture processing api
+// call works.
+TEST(EchoCanceller3InputCheckDeathTest, NullCaptureProcessingParameter) {
+  EXPECT_DEATH(
+      EchoCanceller3(EchoCanceller3Config(),
+                     /*multichannel_config_=*/absl::nullopt, 16000, 1, 1)
+          .ProcessCapture(nullptr, false),
+      "");
+}
+
+// Verifies the check for correct sample rate.
+// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
+// tests on test bots has been fixed.
+TEST(EchoCanceller3InputCheckDeathTest, DISABLED_WrongSampleRate) {
+  ApmDataDumper data_dumper(0);
+  EXPECT_DEATH(
+      EchoCanceller3(EchoCanceller3Config(),
+                     /*multichannel_config_=*/absl::nullopt, 8001, 1, 1),
+      "");
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc
new file mode 100644
index 0000000000..510e4b8a8d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/echo_path_delay_estimator.h"
+
+#include <array>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+EchoPathDelayEstimator::EchoPathDelayEstimator(
+    ApmDataDumper* data_dumper,
+    const EchoCanceller3Config& config,
+    size_t num_capture_channels)
+    : data_dumper_(data_dumper),
+      down_sampling_factor_(config.delay.down_sampling_factor),
+      sub_block_size_(down_sampling_factor_ != 0
+                          ? kBlockSize / down_sampling_factor_
+                          : kBlockSize),
+      capture_mixer_(num_capture_channels,
+                     config.delay.capture_alignment_mixing),
+      capture_decimator_(down_sampling_factor_),
+      matched_filter_(
+          data_dumper_,
+          DetectOptimization(),
+          sub_block_size_,
+          kMatchedFilterWindowSizeSubBlocks,
+          config.delay.num_filters,
+          kMatchedFilterAlignmentShiftSizeSubBlocks,
+          config.delay.down_sampling_factor == 8
+              ? config.render_levels.poor_excitation_render_limit_ds8
+              : config.render_levels.poor_excitation_render_limit,
+          config.delay.delay_estimate_smoothing,
+          config.delay.delay_estimate_smoothing_delay_found,
+          config.delay.delay_candidate_detection_threshold,
+          config.delay.detect_pre_echo),
+      matched_filter_lag_aggregator_(data_dumper_,
+                                     matched_filter_.GetMaxFilterLag(),
+                                     config.delay) {
+  RTC_DCHECK(data_dumper);
+  RTC_DCHECK(down_sampling_factor_ > 0);
+}
+
+EchoPathDelayEstimator::~EchoPathDelayEstimator() = default;
+
+void EchoPathDelayEstimator::Reset(bool reset_delay_confidence) {
+  Reset(true, reset_delay_confidence);
+}
+
+absl::optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
+    const DownsampledRenderBuffer& render_buffer,
+    const Block& capture) {
+  std::array<float, kBlockSize> downsampled_capture_data;
+  rtc::ArrayView<float> downsampled_capture(downsampled_capture_data.data(),
+                                            sub_block_size_);
+
+  std::array<float, kBlockSize> downmixed_capture;
+  capture_mixer_.ProduceOutput(capture, downmixed_capture);
+  capture_decimator_.Decimate(downmixed_capture, downsampled_capture);
+  data_dumper_->DumpWav("aec3_capture_decimator_output",
+                        downsampled_capture.size(), downsampled_capture.data(),
+                        16000 / down_sampling_factor_, 1);
+  matched_filter_.Update(render_buffer, downsampled_capture,
+                         matched_filter_lag_aggregator_.ReliableDelayFound());
+
+  absl::optional<DelayEstimate> aggregated_matched_filter_lag =
+      matched_filter_lag_aggregator_.Aggregate(
+          matched_filter_.GetBestLagEstimate());
+
+  // Run clockdrift detection.
+  if (aggregated_matched_filter_lag &&
+      (*aggregated_matched_filter_lag).quality ==
+          DelayEstimate::Quality::kRefined)
+    clockdrift_detector_.Update(
+        matched_filter_lag_aggregator_.GetDelayAtHighestPeak());
+
+  // TODO(peah): Move this logging outside of this class once EchoCanceller3
+  // development is done.
+  data_dumper_->DumpRaw(
+      "aec3_echo_path_delay_estimator_delay",
+      aggregated_matched_filter_lag
+          ? static_cast<int>(aggregated_matched_filter_lag->delay *
+                             down_sampling_factor_)
+          : -1);
+
+  // Return the detected delay in samples as the aggregated matched filter lag
+  // compensated by the down sampling factor for the signal being correlated.
+  if (aggregated_matched_filter_lag) {
+    aggregated_matched_filter_lag->delay *= down_sampling_factor_;
+  }
+
+  if (old_aggregated_lag_ && aggregated_matched_filter_lag &&
+      old_aggregated_lag_->delay == aggregated_matched_filter_lag->delay) {
+    ++consistent_estimate_counter_;
+  } else {
+    consistent_estimate_counter_ = 0;
+  }
+  old_aggregated_lag_ = aggregated_matched_filter_lag;
+  constexpr size_t kNumBlocksPerSecondBy2 = kNumBlocksPerSecond / 2;
+  if (consistent_estimate_counter_ > kNumBlocksPerSecondBy2) {
+    Reset(false, false);
+  }
+
+  return aggregated_matched_filter_lag;
+}
+
+void EchoPathDelayEstimator::Reset(bool reset_lag_aggregator,
+                                   bool reset_delay_confidence) {
+  if (reset_lag_aggregator) {
+    matched_filter_lag_aggregator_.Reset(reset_delay_confidence);
+  }
+  matched_filter_.Reset(/*full_reset=*/reset_lag_aggregator);
+  old_aggregated_lag_ = absl::nullopt;
+  consistent_estimate_counter_ = 0;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
new file mode 100644
index 0000000000..b24d0a29ec
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/alignment_mixer.h"
+#include "modules/audio_processing/aec3/block.h"
+#include "modules/audio_processing/aec3/clockdrift_detector.h"
+#include "modules/audio_processing/aec3/decimator.h"
+#include "modules/audio_processing/aec3/delay_estimate.h"
+#include "modules/audio_processing/aec3/matched_filter.h"
+#include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+struct DownsampledRenderBuffer;
+struct EchoCanceller3Config;
+
+// Estimates the delay of the echo path.
+class EchoPathDelayEstimator {
+ public:
+  EchoPathDelayEstimator(ApmDataDumper* data_dumper,
+                         const EchoCanceller3Config& config,
+                         size_t num_capture_channels);
+  ~EchoPathDelayEstimator();
+
+  EchoPathDelayEstimator(const EchoPathDelayEstimator&) = delete;
+  EchoPathDelayEstimator& operator=(const EchoPathDelayEstimator&) = delete;
+
+  // Resets the estimation. If the delay confidence is reset, the reset behavior
+  // is as if the call is restarted.
+  void Reset(bool reset_delay_confidence);
+
+  // Produce a delay estimate if such is avaliable.
+  absl::optional<DelayEstimate> EstimateDelay(
+      const DownsampledRenderBuffer& render_buffer,
+      const Block& capture);
+
+  // Log delay estimator properties.
+  void LogDelayEstimationProperties(int sample_rate_hz, size_t shift) const {
+    matched_filter_.LogFilterProperties(sample_rate_hz, shift,
+                                        down_sampling_factor_);
+  }
+
+  // Returns the level of detected clockdrift.
+  ClockdriftDetector::Level Clockdrift() const {
+    return clockdrift_detector_.ClockdriftLevel();
+  }
+
+ private:
+  ApmDataDumper* const data_dumper_;
+  const size_t down_sampling_factor_;
+  const size_t sub_block_size_;
+  AlignmentMixer capture_mixer_;
+  Decimator capture_decimator_;
+  MatchedFilter matched_filter_;
+  MatchedFilterLagAggregator matched_filter_lag_aggregator_;
+  absl::optional<DelayEstimate> old_aggregated_lag_;
+  size_t consistent_estimate_counter_ = 0;
+  ClockdriftDetector clockdrift_detector_;
+
+  // Internal reset method with more granularity.
+  void Reset(bool reset_lag_aggregator, bool reset_delay_confidence);
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
new file mode 100644
index 0000000000..e2c101fb04
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/echo_path_delay_estimator.h"
+
+#include <algorithm>
+#include <string>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+std::string ProduceDebugText(size_t delay, size_t down_sampling_factor) {
+  rtc::StringBuilder ss;
+  ss << "Delay: " << delay;
+  ss << ", Down sampling factor: " << down_sampling_factor;
+  return ss.Release();
+}
+
+}  // namespace
+
+class EchoPathDelayEstimatorMultiChannel
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         EchoPathDelayEstimatorMultiChannel,
+                         ::testing::Combine(::testing::Values(1, 2, 3, 6, 8),
+                                            ::testing::Values(1, 2, 4)));
+
+// Verifies that the basic API calls work.
+TEST_P(EchoPathDelayEstimatorMultiChannel, BasicApiCalls) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, num_render_channels));
+  EchoPathDelayEstimator estimator(&data_dumper, config, num_capture_channels);
+  Block render(kNumBands, num_render_channels);
+  Block capture(/*num_bands=*/1, num_capture_channels);
+  for (size_t k = 0; k < 100; ++k) {
+    render_delay_buffer->Insert(render);
+    estimator.EstimateDelay(render_delay_buffer->GetDownsampledRenderBuffer(),
+                            capture);
+  }
+}
+
+// Verifies that the delay estimator produces correct delay for artificially
+// delayed signals.
+TEST(EchoPathDelayEstimator, DelayEstimation) {
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  Random random_generator(42U);
+  Block render(kNumBands, kNumRenderChannels);
+  Block capture(/*num_bands=*/1, kNumCaptureChannels);
+  ApmDataDumper data_dumper(0);
+  constexpr size_t kDownSamplingFactors[] = {2, 4, 8};
+  for (auto down_sampling_factor : kDownSamplingFactors) {
+    EchoCanceller3Config config;
+    config.delay.delay_headroom_samples = 0;
+    config.delay.down_sampling_factor = down_sampling_factor;
+    config.delay.num_filters = 10;
+    for (size_t delay_samples : {30, 64, 150, 200, 800, 4000}) {
+      SCOPED_TRACE(ProduceDebugText(delay_samples, down_sampling_factor));
+      std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+          RenderDelayBuffer::Create(config, kSampleRateHz, kNumRenderChannels));
+      DelayBuffer<float> signal_delay_buffer(delay_samples);
+      EchoPathDelayEstimator estimator(&data_dumper, config,
+                                       kNumCaptureChannels);
+
+      absl::optional<DelayEstimate> estimated_delay_samples;
+      for (size_t k = 0; k < (500 + (delay_samples) / kBlockSize); ++k) {
+        RandomizeSampleVector(&random_generator,
+                              render.View(/*band=*/0, /*channel=*/0));
+        signal_delay_buffer.Delay(render.View(/*band=*/0, /*channel=*/0),
+                                  capture.View(/*band=*/0, /*channel=*/0));
+        render_delay_buffer->Insert(render);
+
+        if (k == 0) {
+          render_delay_buffer->Reset();
+        }
+
+        render_delay_buffer->PrepareCaptureProcessing();
+
+        auto estimate = estimator.EstimateDelay(
+            render_delay_buffer->GetDownsampledRenderBuffer(), capture);
+
+        if (estimate) {
+          estimated_delay_samples = estimate;
+        }
+      }
+
+      if (estimated_delay_samples) {
+        // Allow estimated delay to be off by a block as internally the delay is
+        // quantized with an error up to a block.
+        size_t delay_ds = delay_samples / down_sampling_factor;
+        size_t estimated_delay_ds =
+            estimated_delay_samples->delay / down_sampling_factor;
+        EXPECT_NEAR(delay_ds, estimated_delay_ds,
+                    kBlockSize / down_sampling_factor);
+      } else {
+        ADD_FAILURE();
+      }
+    }
+  }
+}
+
+// Verifies that the delay estimator does not produce delay estimates for render
+// signals of low level.
+TEST(EchoPathDelayEstimator, NoDelayEstimatesForLowLevelRenderSignals) {
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  Random random_generator(42U);
+  EchoCanceller3Config config;
+  Block render(kNumBands, kNumRenderChannels);
+  Block capture(/*num_bands=*/1, kNumCaptureChannels);
+  ApmDataDumper data_dumper(0);
+  EchoPathDelayEstimator estimator(&data_dumper, config, kNumCaptureChannels);
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(EchoCanceller3Config(), kSampleRateHz,
+                                kNumRenderChannels));
+  for (size_t k = 0; k < 100; ++k) {
+    RandomizeSampleVector(&random_generator,
+                          render.View(/*band=*/0, /*channel=*/0));
+    for (auto& render_k : render.View(/*band=*/0, /*channel=*/0)) {
+      render_k *= 100.f / 32767.f;
+    }
+    std::copy(render.begin(/*band=*/0, /*channel=*/0),
+              render.end(/*band=*/0, /*channel=*/0),
+              capture.begin(/*band*/ 0, /*channel=*/0));
+    render_delay_buffer->Insert(render);
+    render_delay_buffer->PrepareCaptureProcessing();
+    EXPECT_FALSE(estimator.EstimateDelay(
+        render_delay_buffer->GetDownsampledRenderBuffer(), capture));
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for the render blocksize.
+// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
+// tests on test bots has been fixed.
+TEST(EchoPathDelayEstimatorDeathTest, DISABLED_WrongRenderBlockSize) {
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  EchoPathDelayEstimator estimator(&data_dumper, config, 1);
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, 48000, 1));
+  Block capture(/*num_bands=*/1, /*num_channels=*/1);
+  EXPECT_DEATH(estimator.EstimateDelay(
+                   render_delay_buffer->GetDownsampledRenderBuffer(), capture),
+               "");
+}
+
+// Verifies the check for non-null data dumper.
+TEST(EchoPathDelayEstimatorDeathTest, NullDataDumper) {
+  EXPECT_DEATH(EchoPathDelayEstimator(nullptr, EchoCanceller3Config(), 1), "");
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability.cc
new file mode 100644
index 0000000000..0ae9cff98e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability.cc
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+
+namespace webrtc {
+
+EchoPathVariability::EchoPathVariability(bool gain_change,
+                                         DelayAdjustment delay_change,
+                                         bool clock_drift)
+    : gain_change(gain_change),
+      delay_change(delay_change),
+      clock_drift(clock_drift) {}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability.h b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability.h
new file mode 100644
index 0000000000..78e4f64b2b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_VARIABILITY_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_VARIABILITY_H_
+
+namespace webrtc {
+
+struct EchoPathVariability {
+  enum class DelayAdjustment {
+    kNone,
+    kBufferFlush,
+    kNewDetectedDelay
+  };
+
+  EchoPathVariability(bool gain_change,
+                      DelayAdjustment delay_change,
+                      bool clock_drift);
+
+  bool AudioPathChanged() const {
+    return gain_change || delay_change != DelayAdjustment::kNone;
+  }
+  bool gain_change;
+  DelayAdjustment delay_change;
+  bool clock_drift;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_VARIABILITY_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability_unittest.cc
new file mode 100644
index 0000000000..0f10f95f72
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_path_variability_unittest.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(EchoPathVariability, CorrectBehavior) {
+  // Test correct passing and reporting of the gain change information.
+  EchoPathVariability v(
+      true, EchoPathVariability::DelayAdjustment::kNewDetectedDelay, false);
+  EXPECT_TRUE(v.gain_change);
+  EXPECT_TRUE(v.delay_change ==
+              EchoPathVariability::DelayAdjustment::kNewDetectedDelay);
+  EXPECT_TRUE(v.AudioPathChanged());
+  EXPECT_FALSE(v.clock_drift);
+
+  v = EchoPathVariability(true, EchoPathVariability::DelayAdjustment::kNone,
+                          false);
+  EXPECT_TRUE(v.gain_change);
+  EXPECT_TRUE(v.delay_change == EchoPathVariability::DelayAdjustment::kNone);
+  EXPECT_TRUE(v.AudioPathChanged());
+  EXPECT_FALSE(v.clock_drift);
+
+  v = EchoPathVariability(
+      false, EchoPathVariability::DelayAdjustment::kNewDetectedDelay, false);
+  EXPECT_FALSE(v.gain_change);
+  EXPECT_TRUE(v.delay_change ==
+              EchoPathVariability::DelayAdjustment::kNewDetectedDelay);
+  EXPECT_TRUE(v.AudioPathChanged());
+  EXPECT_FALSE(v.clock_drift);
+
+  v = EchoPathVariability(false, EchoPathVariability::DelayAdjustment::kNone,
+                          false);
+  EXPECT_FALSE(v.gain_change);
+  EXPECT_TRUE(v.delay_change == EchoPathVariability::DelayAdjustment::kNone);
+  EXPECT_FALSE(v.AudioPathChanged());
+  EXPECT_FALSE(v.clock_drift);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover.cc
new file mode 100644
index 0000000000..673d88af03
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover.cc
@@ -0,0 +1,521 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/echo_remover.h"
+
+#include <math.h>
+#include <stddef.h>
+
+#include <algorithm>
+#include <array>
+#include <atomic>
+#include <cmath>
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/comfort_noise_generator.h"
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+#include "modules/audio_processing/aec3/echo_remover_metrics.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+#include "modules/audio_processing/aec3/residual_echo_estimator.h"
+#include "modules/audio_processing/aec3/subtractor.h"
+#include "modules/audio_processing/aec3/subtractor_output.h"
+#include "modules/audio_processing/aec3/suppression_filter.h"
+#include "modules/audio_processing/aec3/suppression_gain.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// Maximum number of channels for which the capture channel data is stored on
+// the stack. If the number of channels are larger than this, they are stored
+// using scratch memory that is pre-allocated on the heap. The reason for this
+// partitioning is not to waste heap space for handling the more common numbers
+// of channels, while at the same time not limiting the support for higher
+// numbers of channels by enforcing the capture channel data to be stored on the
+// stack using a fixed maximum value.
+constexpr size_t kMaxNumChannelsOnStack = 2;
+
+// Chooses the number of channels to store on the heap when that is required due
+// to the number of capture channels being larger than the pre-defined number
+// of channels to store on the stack.
+size_t NumChannelsOnHeap(size_t num_capture_channels) {
+  return num_capture_channels > kMaxNumChannelsOnStack ? num_capture_channels
+                                                       : 0;
+}
+
+void LinearEchoPower(const FftData& E,
+                     const FftData& Y,
+                     std::array<float, kFftLengthBy2Plus1>* S2) {
+  for (size_t k = 0; k < E.re.size(); ++k) {
+    (*S2)[k] = (Y.re[k] - E.re[k]) * (Y.re[k] - E.re[k]) +
+               (Y.im[k] - E.im[k]) * (Y.im[k] - E.im[k]);
+  }
+}
+
+// Fades between two input signals using a fix-sized transition.
+void SignalTransition(rtc::ArrayView<const float> from,
+                      rtc::ArrayView<const float> to,
+                      rtc::ArrayView<float> out) {
+  if (from == to) {
+    RTC_DCHECK_EQ(to.size(), out.size());
+    std::copy(to.begin(), to.end(), out.begin());
+  } else {
+    constexpr size_t kTransitionSize = 30;
+    constexpr float kOneByTransitionSizePlusOne = 1.f / (kTransitionSize + 1);
+
+    RTC_DCHECK_EQ(from.size(), to.size());
+    RTC_DCHECK_EQ(from.size(), out.size());
+    RTC_DCHECK_LE(kTransitionSize, out.size());
+
+    for (size_t k = 0; k < kTransitionSize; ++k) {
+      float a = (k + 1) * kOneByTransitionSizePlusOne;
+      out[k] = a * to[k] + (1.f - a) * from[k];
+    }
+
+    std::copy(to.begin() + kTransitionSize, to.end(),
+              out.begin() + kTransitionSize);
+  }
+}
+
+// Computes a windowed (square root Hanning) padded FFT and updates the related
+// memory.
+void WindowedPaddedFft(const Aec3Fft& fft,
+                       rtc::ArrayView<const float> v,
+                       rtc::ArrayView<float> v_old,
+                       FftData* V) {
+  fft.PaddedFft(v, v_old, Aec3Fft::Window::kSqrtHanning, V);
+  std::copy(v.begin(), v.end(), v_old.begin());
+}
+
+// Class for removing the echo from the capture signal.
+class EchoRemoverImpl final : public EchoRemover {
+ public:
+  EchoRemoverImpl(const EchoCanceller3Config& config,
+                  int sample_rate_hz,
+                  size_t num_render_channels,
+                  size_t num_capture_channels);
+  ~EchoRemoverImpl() override;
+  EchoRemoverImpl(const EchoRemoverImpl&) = delete;
+  EchoRemoverImpl& operator=(const EchoRemoverImpl&) = delete;
+
+  void GetMetrics(EchoControl::Metrics* metrics) const override;
+
+  // Removes the echo from a block of samples from the capture signal. The
+  // supplied render signal is assumed to be pre-aligned with the capture
+  // signal.
+  void ProcessCapture(EchoPathVariability echo_path_variability,
+                      bool capture_signal_saturation,
+                      const absl::optional<DelayEstimate>& external_delay,
+                      RenderBuffer* render_buffer,
+                      Block* linear_output,
+                      Block* capture) override;
+
+  // Updates the status on whether echo leakage is detected in the output of the
+  // echo remover.
+  void UpdateEchoLeakageStatus(bool leakage_detected) override {
+    echo_leakage_detected_ = leakage_detected;
+  }
+
+  void SetCaptureOutputUsage(bool capture_output_used) override {
+    capture_output_used_ = capture_output_used;
+  }
+
+ private:
+  // Selects which of the coarse and refined linear filter outputs that is most
+  // appropriate to pass to the suppressor and forms the linear filter output by
+  // smoothly transition between those.
+  void FormLinearFilterOutput(const SubtractorOutput& subtractor_output,
+                              rtc::ArrayView<float> output);
+
+  static std::atomic<int> instance_count_;
+  const EchoCanceller3Config config_;
+  const Aec3Fft fft_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  const Aec3Optimization optimization_;
+  const int sample_rate_hz_;
+  const size_t num_render_channels_;
+  const size_t num_capture_channels_;
+  const bool use_coarse_filter_output_;
+  Subtractor subtractor_;
+  SuppressionGain suppression_gain_;
+  ComfortNoiseGenerator cng_;
+  SuppressionFilter suppression_filter_;
+  RenderSignalAnalyzer render_signal_analyzer_;
+  ResidualEchoEstimator residual_echo_estimator_;
+  bool echo_leakage_detected_ = false;
+  bool capture_output_used_ = true;
+  AecState aec_state_;
+  EchoRemoverMetrics metrics_;
+  std::vector<std::array<float, kFftLengthBy2>> e_old_;
+  std::vector<std::array<float, kFftLengthBy2>> y_old_;
+  size_t block_counter_ = 0;
+  int gain_change_hangover_ = 0;
+  bool refined_filter_output_last_selected_ = true;
+
+  std::vector<std::array<float, kFftLengthBy2>> e_heap_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2_heap_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2_heap_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> R2_heap_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> R2_unbounded_heap_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> S2_linear_heap_;
+  std::vector<FftData> Y_heap_;
+  std::vector<FftData> E_heap_;
+  std::vector<FftData> comfort_noise_heap_;
+  std::vector<FftData> high_band_comfort_noise_heap_;
+  std::vector<SubtractorOutput> subtractor_output_heap_;
+};
+
+std::atomic<int> EchoRemoverImpl::instance_count_(0);
+
+EchoRemoverImpl::EchoRemoverImpl(const EchoCanceller3Config& config,
+                                 int sample_rate_hz,
+                                 size_t num_render_channels,
+                                 size_t num_capture_channels)
+    : config_(config),
+      fft_(),
+      data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      optimization_(DetectOptimization()),
+      sample_rate_hz_(sample_rate_hz),
+      num_render_channels_(num_render_channels),
+      num_capture_channels_(num_capture_channels),
+      use_coarse_filter_output_(
+          config_.filter.enable_coarse_filter_output_usage),
+      subtractor_(config,
+                  num_render_channels_,
+                  num_capture_channels_,
+                  data_dumper_.get(),
+                  optimization_),
+      suppression_gain_(config_,
+                        optimization_,
+                        sample_rate_hz,
+                        num_capture_channels),
+      cng_(config_, optimization_, num_capture_channels_),
+      suppression_filter_(optimization_,
+                          sample_rate_hz_,
+                          num_capture_channels_),
+      render_signal_analyzer_(config_),
+      residual_echo_estimator_(config_, num_render_channels),
+      aec_state_(config_, num_capture_channels_),
+      e_old_(num_capture_channels_, {0.f}),
+      y_old_(num_capture_channels_, {0.f}),
+      e_heap_(NumChannelsOnHeap(num_capture_channels_), {0.f}),
+      Y2_heap_(NumChannelsOnHeap(num_capture_channels_)),
+      E2_heap_(NumChannelsOnHeap(num_capture_channels_)),
+      R2_heap_(NumChannelsOnHeap(num_capture_channels_)),
+      R2_unbounded_heap_(NumChannelsOnHeap(num_capture_channels_)),
+      S2_linear_heap_(NumChannelsOnHeap(num_capture_channels_)),
+      Y_heap_(NumChannelsOnHeap(num_capture_channels_)),
+      E_heap_(NumChannelsOnHeap(num_capture_channels_)),
+      comfort_noise_heap_(NumChannelsOnHeap(num_capture_channels_)),
+      high_band_comfort_noise_heap_(NumChannelsOnHeap(num_capture_channels_)),
+      subtractor_output_heap_(NumChannelsOnHeap(num_capture_channels_)) {
+  RTC_DCHECK(ValidFullBandRate(sample_rate_hz));
+}
+
+EchoRemoverImpl::~EchoRemoverImpl() = default;
+
+void EchoRemoverImpl::GetMetrics(EchoControl::Metrics* metrics) const {
+  // Echo return loss (ERL) is inverted to go from gain to attenuation.
+  metrics->echo_return_loss = -10.0 * std::log10(aec_state_.ErlTimeDomain());
+  metrics->echo_return_loss_enhancement =
+      Log2TodB(aec_state_.FullBandErleLog2());
+}
+
+void EchoRemoverImpl::ProcessCapture(
+    EchoPathVariability echo_path_variability,
+    bool capture_signal_saturation,
+    const absl::optional<DelayEstimate>& external_delay,
+    RenderBuffer* render_buffer,
+    Block* linear_output,
+    Block* capture) {
+  ++block_counter_;
+  const Block& x = render_buffer->GetBlock(0);
+  Block* y = capture;
+  RTC_DCHECK(render_buffer);
+  RTC_DCHECK(y);
+  RTC_DCHECK_EQ(x.NumBands(), NumBandsForRate(sample_rate_hz_));
+  RTC_DCHECK_EQ(y->NumBands(), NumBandsForRate(sample_rate_hz_));
+  RTC_DCHECK_EQ(x.NumChannels(), num_render_channels_);
+  RTC_DCHECK_EQ(y->NumChannels(), num_capture_channels_);
+
+  // Stack allocated data to use when the number of channels is low.
+  std::array<std::array<float, kFftLengthBy2>, kMaxNumChannelsOnStack> e_stack;
+  std::array<std::array<float, kFftLengthBy2Plus1>, kMaxNumChannelsOnStack>
+      Y2_stack;
+  std::array<std::array<float, kFftLengthBy2Plus1>, kMaxNumChannelsOnStack>
+      E2_stack;
+  std::array<std::array<float, kFftLengthBy2Plus1>, kMaxNumChannelsOnStack>
+      R2_stack;
+  std::array<std::array<float, kFftLengthBy2Plus1>, kMaxNumChannelsOnStack>
+      R2_unbounded_stack;
+  std::array<std::array<float, kFftLengthBy2Plus1>, kMaxNumChannelsOnStack>
+      S2_linear_stack;
+  std::array<FftData, kMaxNumChannelsOnStack> Y_stack;
+  std::array<FftData, kMaxNumChannelsOnStack> E_stack;
+  std::array<FftData, kMaxNumChannelsOnStack> comfort_noise_stack;
+  std::array<FftData, kMaxNumChannelsOnStack> high_band_comfort_noise_stack;
+  std::array<SubtractorOutput, kMaxNumChannelsOnStack> subtractor_output_stack;
+
+  rtc::ArrayView<std::array<float, kFftLengthBy2>> e(e_stack.data(),
+                                                     num_capture_channels_);
+  rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> Y2(
+      Y2_stack.data(), num_capture_channels_);
+  rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> E2(
+      E2_stack.data(), num_capture_channels_);
+  rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2(
+      R2_stack.data(), num_capture_channels_);
+  rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2_unbounded(
+      R2_unbounded_stack.data(), num_capture_channels_);
+  rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> S2_linear(
+      S2_linear_stack.data(), num_capture_channels_);
+  rtc::ArrayView<FftData> Y(Y_stack.data(), num_capture_channels_);
+  rtc::ArrayView<FftData> E(E_stack.data(), num_capture_channels_);
+  rtc::ArrayView<FftData> comfort_noise(comfort_noise_stack.data(),
+                                        num_capture_channels_);
+  rtc::ArrayView<FftData> high_band_comfort_noise(
+      high_band_comfort_noise_stack.data(), num_capture_channels_);
+  rtc::ArrayView<SubtractorOutput> subtractor_output(
+      subtractor_output_stack.data(), num_capture_channels_);
+  if (NumChannelsOnHeap(num_capture_channels_) > 0) {
+    // If the stack-allocated space is too small, use the heap for storing the
+    // microphone data.
+    e = rtc::ArrayView<std::array<float, kFftLengthBy2>>(e_heap_.data(),
+                                                         num_capture_channels_);
+    Y2 = rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+        Y2_heap_.data(), num_capture_channels_);
+    E2 = rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+        E2_heap_.data(), num_capture_channels_);
+    R2 = rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+        R2_heap_.data(), num_capture_channels_);
+    R2_unbounded = rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+        R2_unbounded_heap_.data(), num_capture_channels_);
+    S2_linear = rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>>(
+        S2_linear_heap_.data(), num_capture_channels_);
+    Y = rtc::ArrayView<FftData>(Y_heap_.data(), num_capture_channels_);
+    E = rtc::ArrayView<FftData>(E_heap_.data(), num_capture_channels_);
+    comfort_noise = rtc::ArrayView<FftData>(comfort_noise_heap_.data(),
+                                            num_capture_channels_);
+    high_band_comfort_noise = rtc::ArrayView<FftData>(
+        high_band_comfort_noise_heap_.data(), num_capture_channels_);
+    subtractor_output = rtc::ArrayView<SubtractorOutput>(
+        subtractor_output_heap_.data(), num_capture_channels_);
+  }
+
+  data_dumper_->DumpWav("aec3_echo_remover_capture_input",
+                        y->View(/*band=*/0, /*channel=*/0), 16000, 1);
+  data_dumper_->DumpWav("aec3_echo_remover_render_input",
+                        x.View(/*band=*/0, /*channel=*/0), 16000, 1);
+  data_dumper_->DumpRaw("aec3_echo_remover_capture_input",
+                        y->View(/*band=*/0, /*channel=*/0));
+  data_dumper_->DumpRaw("aec3_echo_remover_render_input",
+                        x.View(/*band=*/0, /*channel=*/0));
+
+  aec_state_.UpdateCaptureSaturation(capture_signal_saturation);
+
+  if (echo_path_variability.AudioPathChanged()) {
+    // Ensure that the gain change is only acted on once per frame.
+    if (echo_path_variability.gain_change) {
+      if (gain_change_hangover_ == 0) {
+        constexpr int kMaxBlocksPerFrame = 3;
+        gain_change_hangover_ = kMaxBlocksPerFrame;
+        rtc::LoggingSeverity log_level =
+            config_.delay.log_warning_on_delay_changes ? rtc::LS_WARNING
+                                                       : rtc::LS_VERBOSE;
+        RTC_LOG_V(log_level)
+            << "Gain change detected at block " << block_counter_;
+      } else {
+        echo_path_variability.gain_change = false;
+      }
+    }
+
+    subtractor_.HandleEchoPathChange(echo_path_variability);
+    aec_state_.HandleEchoPathChange(echo_path_variability);
+
+    if (echo_path_variability.delay_change !=
+        EchoPathVariability::DelayAdjustment::kNone) {
+      suppression_gain_.SetInitialState(true);
+    }
+  }
+  if (gain_change_hangover_ > 0) {
+    --gain_change_hangover_;
+  }
+
+  // Analyze the render signal.
+  render_signal_analyzer_.Update(*render_buffer,
+                                 aec_state_.MinDirectPathFilterDelay());
+
+  // State transition.
+  if (aec_state_.TransitionTriggered()) {
+    subtractor_.ExitInitialState();
+    suppression_gain_.SetInitialState(false);
+  }
+
+  // Perform linear echo cancellation.
+  subtractor_.Process(*render_buffer, *y, render_signal_analyzer_, aec_state_,
+                      subtractor_output);
+
+  // Compute spectra.
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    FormLinearFilterOutput(subtractor_output[ch], e[ch]);
+    WindowedPaddedFft(fft_, y->View(/*band=*/0, ch), y_old_[ch], &Y[ch]);
+    WindowedPaddedFft(fft_, e[ch], e_old_[ch], &E[ch]);
+    LinearEchoPower(E[ch], Y[ch], &S2_linear[ch]);
+    Y[ch].Spectrum(optimization_, Y2[ch]);
+    E[ch].Spectrum(optimization_, E2[ch]);
+  }
+
+  // Optionally return the linear filter output.
+  if (linear_output) {
+    RTC_DCHECK_GE(1, linear_output->NumBands());
+    RTC_DCHECK_EQ(num_capture_channels_, linear_output->NumChannels());
+    for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+      std::copy(e[ch].begin(), e[ch].end(),
+                linear_output->begin(/*band=*/0, ch));
+    }
+  }
+
+  // Update the AEC state information.
+  aec_state_.Update(external_delay, subtractor_.FilterFrequencyResponses(),
+                    subtractor_.FilterImpulseResponses(), *render_buffer, E2,
+                    Y2, subtractor_output);
+
+  // Choose the linear output.
+  const auto& Y_fft = aec_state_.UseLinearFilterOutput() ? E : Y;
+
+  data_dumper_->DumpWav("aec3_output_linear",
+                        y->View(/*band=*/0, /*channel=*/0), 16000, 1);
+  data_dumper_->DumpWav("aec3_output_linear2", kBlockSize, &e[0][0], 16000, 1);
+
+  // Estimate the comfort noise.
+  cng_.Compute(aec_state_.SaturatedCapture(), Y2, comfort_noise,
+               high_band_comfort_noise);
+
+  // Only do the below processing if the output of the audio processing module
+  // is used.
+  std::array<float, kFftLengthBy2Plus1> G;
+  if (capture_output_used_) {
+    // Estimate the residual echo power.
+    residual_echo_estimator_.Estimate(aec_state_, *render_buffer, S2_linear, Y2,
+                                      suppression_gain_.IsDominantNearend(), R2,
+                                      R2_unbounded);
+
+    // Suppressor nearend estimate.
+    if (aec_state_.UsableLinearEstimate()) {
+      // E2 is bound by Y2.
+      for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+        std::transform(E2[ch].begin(), E2[ch].end(), Y2[ch].begin(),
+                       E2[ch].begin(),
+                       [](float a, float b) { return std::min(a, b); });
+      }
+    }
+    const auto& nearend_spectrum = aec_state_.UsableLinearEstimate() ? E2 : Y2;
+
+    // Suppressor echo estimate.
+    const auto& echo_spectrum =
+        aec_state_.UsableLinearEstimate() ? S2_linear : R2;
+
+    // Determine if the suppressor should assume clock drift.
+    const bool clock_drift = config_.echo_removal_control.has_clock_drift ||
+                             echo_path_variability.clock_drift;
+
+    // Compute preferred gains.
+    float high_bands_gain;
+    suppression_gain_.GetGain(nearend_spectrum, echo_spectrum, R2, R2_unbounded,
+                              cng_.NoiseSpectrum(), render_signal_analyzer_,
+                              aec_state_, x, clock_drift, &high_bands_gain, &G);
+
+    suppression_filter_.ApplyGain(comfort_noise, high_band_comfort_noise, G,
+                                  high_bands_gain, Y_fft, y);
+
+  } else {
+    G.fill(0.f);
+  }
+
+  // Update the metrics.
+  metrics_.Update(aec_state_, cng_.NoiseSpectrum()[0], G);
+
+  // Debug outputs for the purpose of development and analysis.
+  data_dumper_->DumpWav("aec3_echo_estimate", kBlockSize,
+                        &subtractor_output[0].s_refined[0], 16000, 1);
+  data_dumper_->DumpRaw("aec3_output", y->View(/*band=*/0, /*channel=*/0));
+  data_dumper_->DumpRaw("aec3_narrow_render",
+                        render_signal_analyzer_.NarrowPeakBand() ? 1 : 0);
+  data_dumper_->DumpRaw("aec3_N2", cng_.NoiseSpectrum()[0]);
+  data_dumper_->DumpRaw("aec3_suppressor_gain", G);
+  data_dumper_->DumpWav("aec3_output", y->View(/*band=*/0, /*channel=*/0),
+                        16000, 1);
+  data_dumper_->DumpRaw("aec3_using_subtractor_output[0]",
+                        aec_state_.UseLinearFilterOutput() ? 1 : 0);
+  data_dumper_->DumpRaw("aec3_E2", E2[0]);
+  data_dumper_->DumpRaw("aec3_S2_linear", S2_linear[0]);
+  data_dumper_->DumpRaw("aec3_Y2", Y2[0]);
+  data_dumper_->DumpRaw(
+      "aec3_X2", render_buffer->Spectrum(
+                     aec_state_.MinDirectPathFilterDelay())[/*channel=*/0]);
+  data_dumper_->DumpRaw("aec3_R2", R2[0]);
+  data_dumper_->DumpRaw("aec3_filter_delay",
+                        aec_state_.MinDirectPathFilterDelay());
+  data_dumper_->DumpRaw("aec3_capture_saturation",
+                        aec_state_.SaturatedCapture() ? 1 : 0);
+}
+
+void EchoRemoverImpl::FormLinearFilterOutput(
+    const SubtractorOutput& subtractor_output,
+    rtc::ArrayView<float> output) {
+  RTC_DCHECK_EQ(subtractor_output.e_refined.size(), output.size());
+  RTC_DCHECK_EQ(subtractor_output.e_coarse.size(), output.size());
+  bool use_refined_output = true;
+  if (use_coarse_filter_output_) {
+    // As the output of the refined adaptive filter generally should be better
+    // than the coarse filter output, add a margin and threshold for when
+    // choosing the coarse filter output.
+    if (subtractor_output.e2_coarse < 0.9f * subtractor_output.e2_refined &&
+        subtractor_output.y2 > 30.f * 30.f * kBlockSize &&
+        (subtractor_output.s2_refined > 60.f * 60.f * kBlockSize ||
+         subtractor_output.s2_coarse > 60.f * 60.f * kBlockSize)) {
+      use_refined_output = false;
+    } else {
+      // If the refined filter is diverged, choose the filter output that has
+      // the lowest power.
+      if (subtractor_output.e2_coarse < subtractor_output.e2_refined &&
+          subtractor_output.y2 < subtractor_output.e2_refined) {
+        use_refined_output = false;
+      }
+    }
+  }
+
+  SignalTransition(refined_filter_output_last_selected_
+                       ? subtractor_output.e_refined
+                       : subtractor_output.e_coarse,
+                   use_refined_output ? subtractor_output.e_refined
+                                      : subtractor_output.e_coarse,
+                   output);
+  refined_filter_output_last_selected_ = use_refined_output;
+}
+
+}  // namespace
+
+EchoRemover* EchoRemover::Create(const EchoCanceller3Config& config,
+                                 int sample_rate_hz,
+                                 size_t num_render_channels,
+                                 size_t num_capture_channels) {
+  return new EchoRemoverImpl(config, sample_rate_hz, num_render_channels,
+                             num_capture_channels);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover.h b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover.h
new file mode 100644
index 0000000000..f2f4f5e64d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "api/audio/echo_control.h"
+#include "modules/audio_processing/aec3/block.h"
+#include "modules/audio_processing/aec3/delay_estimate.h"
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+
+namespace webrtc {
+
+// Class for removing the echo from the capture signal.
+class EchoRemover {
+ public:
+  static EchoRemover* Create(const EchoCanceller3Config& config,
+                             int sample_rate_hz,
+                             size_t num_render_channels,
+                             size_t num_capture_channels);
+  virtual ~EchoRemover() = default;
+
+  // Get current metrics.
+  virtual void GetMetrics(EchoControl::Metrics* metrics) const = 0;
+
+  // Removes the echo from a block of samples from the capture signal. The
+  // supplied render signal is assumed to be pre-aligned with the capture
+  // signal.
+  virtual void ProcessCapture(
+      EchoPathVariability echo_path_variability,
+      bool capture_signal_saturation,
+      const absl::optional<DelayEstimate>& external_delay,
+      RenderBuffer* render_buffer,
+      Block* linear_output,
+      Block* capture) = 0;
+
+  // Updates the status on whether echo leakage is detected in the output of the
+  // echo remover.
+  virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0;
+
+  // Specifies whether the capture output will be used. The purpose of this is
+  // to allow the echo remover to deactivate some of the processing when the
+  // resulting output is anyway not used, for instance when the endpoint is
+  // muted.
+  virtual void SetCaptureOutputUsage(bool capture_output_used) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
new file mode 100644
index 0000000000..c3fc80773a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
@@ -0,0 +1,157 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/echo_remover_metrics.h"
+
+#include <math.h>
+#include <stddef.h>
+
+#include <algorithm>
+#include <cmath>
+#include <numeric>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+EchoRemoverMetrics::DbMetric::DbMetric() : DbMetric(0.f, 0.f, 0.f) {}
+EchoRemoverMetrics::DbMetric::DbMetric(float sum_value,
+                                       float floor_value,
+                                       float ceil_value)
+    : sum_value(sum_value), floor_value(floor_value), ceil_value(ceil_value) {}
+
+void EchoRemoverMetrics::DbMetric::Update(float value) {
+  sum_value += value;
+  floor_value = std::min(floor_value, value);
+  ceil_value = std::max(ceil_value, value);
+}
+
+void EchoRemoverMetrics::DbMetric::UpdateInstant(float value) {
+  sum_value = value;
+  floor_value = std::min(floor_value, value);
+  ceil_value = std::max(ceil_value, value);
+}
+
+EchoRemoverMetrics::EchoRemoverMetrics() {
+  ResetMetrics();
+}
+
+void EchoRemoverMetrics::ResetMetrics() {
+  erl_time_domain_ = DbMetric(0.f, 10000.f, 0.000f);
+  erle_time_domain_ = DbMetric(0.f, 0.f, 1000.f);
+  saturated_capture_ = false;
+}
+
+void EchoRemoverMetrics::Update(
+    const AecState& aec_state,
+    const std::array<float, kFftLengthBy2Plus1>& comfort_noise_spectrum,
+    const std::array<float, kFftLengthBy2Plus1>& suppressor_gain) {
+  metrics_reported_ = false;
+  if (++block_counter_ <= kMetricsCollectionBlocks) {
+    erl_time_domain_.UpdateInstant(aec_state.ErlTimeDomain());
+    erle_time_domain_.UpdateInstant(aec_state.FullBandErleLog2());
+    saturated_capture_ = saturated_capture_ || aec_state.SaturatedCapture();
+  } else {
+    // Report the metrics over several frames in order to lower the impact of
+    // the logarithms involved on the computational complexity.
+    switch (block_counter_) {
+      case kMetricsCollectionBlocks + 1:
+        RTC_HISTOGRAM_BOOLEAN(
+            "WebRTC.Audio.EchoCanceller.UsableLinearEstimate",
+            static_cast<int>(aec_state.UsableLinearEstimate() ? 1 : 0));
+        RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.FilterDelay",
+                                    aec_state.MinDirectPathFilterDelay(), 0, 30,
+                                    31);
+        RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.EchoCanceller.CaptureSaturation",
+                              static_cast<int>(saturated_capture_ ? 1 : 0));
+        break;
+      case kMetricsCollectionBlocks + 2:
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.Erl.Value",
+            aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+                                                erl_time_domain_.sum_value),
+            0, 59, 30);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.Erl.Max",
+            aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+                                                erl_time_domain_.ceil_value),
+            0, 59, 30);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.Erl.Min",
+            aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+                                                erl_time_domain_.floor_value),
+            0, 59, 30);
+        break;
+      case kMetricsCollectionBlocks + 3:
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.Erle.Value",
+            aec3::TransformDbMetricForReporting(false, 0.f, 19.f, 0.f, 1.f,
+                                                erle_time_domain_.sum_value),
+            0, 19, 20);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.Erle.Max",
+            aec3::TransformDbMetricForReporting(false, 0.f, 19.f, 0.f, 1.f,
+                                                erle_time_domain_.ceil_value),
+            0, 19, 20);
+        RTC_HISTOGRAM_COUNTS_LINEAR(
+            "WebRTC.Audio.EchoCanceller.Erle.Min",
+            aec3::TransformDbMetricForReporting(false, 0.f, 19.f, 0.f, 1.f,
+                                                erle_time_domain_.floor_value),
+            0, 19, 20);
+        metrics_reported_ = true;
+        RTC_DCHECK_EQ(kMetricsReportingIntervalBlocks, block_counter_);
+        block_counter_ = 0;
+        ResetMetrics();
+        break;
+      default:
+        RTC_DCHECK_NOTREACHED();
+        break;
+    }
+  }
+}
+
+namespace aec3 {
+
+void UpdateDbMetric(const std::array<float, kFftLengthBy2Plus1>& value,
+                    std::array<EchoRemoverMetrics::DbMetric, 2>* statistic) {
+  RTC_DCHECK(statistic);
+  // Truncation is intended in the band width computation.
+  constexpr int kNumBands = 2;
+  constexpr int kBandWidth = 65 / kNumBands;
+  constexpr float kOneByBandWidth = 1.f / kBandWidth;
+  RTC_DCHECK_EQ(kNumBands, statistic->size());
+  RTC_DCHECK_EQ(65, value.size());
+  for (size_t k = 0; k < statistic->size(); ++k) {
+    float average_band =
+        std::accumulate(value.begin() + kBandWidth * k,
+                        value.begin() + kBandWidth * (k + 1), 0.f) *
+        kOneByBandWidth;
+    (*statistic)[k].Update(average_band);
+  }
+}
+
+int TransformDbMetricForReporting(bool negate,
+                                  float min_value,
+                                  float max_value,
+                                  float offset,
+                                  float scaling,
+                                  float value) {
+  float new_value = 10.f * std::log10(value * scaling + 1e-10f) + offset;
+  if (negate) {
+    new_value = -new_value;
+  }
+  return static_cast<int>(rtc::SafeClamp(new_value, min_value, max_value));
+}
+
+}  // namespace aec3
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics.h b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics.h
new file mode 100644
index 0000000000..aec8084d78
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics.h
@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_METRICS_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_METRICS_H_
+
+#include <array>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+
+namespace webrtc {
+
+// Handles the reporting of metrics for the echo remover.
+class EchoRemoverMetrics {
+ public:
+  struct DbMetric {
+    DbMetric();
+    DbMetric(float sum_value, float floor_value, float ceil_value);
+    void Update(float value);
+    void UpdateInstant(float value);
+    float sum_value;
+    float floor_value;
+    float ceil_value;
+  };
+
+  EchoRemoverMetrics();
+
+  EchoRemoverMetrics(const EchoRemoverMetrics&) = delete;
+  EchoRemoverMetrics& operator=(const EchoRemoverMetrics&) = delete;
+
+  // Updates the metric with new data.
+  void Update(
+      const AecState& aec_state,
+      const std::array<float, kFftLengthBy2Plus1>& comfort_noise_spectrum,
+      const std::array<float, kFftLengthBy2Plus1>& suppressor_gain);
+
+  // Returns true if the metrics have just been reported, otherwise false.
+  bool MetricsReported() { return metrics_reported_; }
+
+ private:
+  // Resets the metrics.
+  void ResetMetrics();
+
+  int block_counter_ = 0;
+  DbMetric erl_time_domain_;
+  DbMetric erle_time_domain_;
+  bool saturated_capture_ = false;
+  bool metrics_reported_ = false;
+};
+
+namespace aec3 {
+
+// Updates a banded metric of type DbMetric with the values in the supplied
+// array.
+void UpdateDbMetric(const std::array<float, kFftLengthBy2Plus1>& value,
+                    std::array<EchoRemoverMetrics::DbMetric, 2>* statistic);
+
+// Transforms a DbMetric from the linear domain into the logarithmic domain.
+int TransformDbMetricForReporting(bool negate,
+                                  float min_value,
+                                  float max_value,
+                                  float offset,
+                                  float scaling,
+                                  float value);
+
+}  // namespace aec3
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_METRICS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
new file mode 100644
index 0000000000..45b30a9c74
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/echo_remover_metrics.h"
+
+#include <math.h>
+
+#include <cmath>
+
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for non-null input.
+TEST(UpdateDbMetricDeathTest, NullValue) {
+  std::array<float, kFftLengthBy2Plus1> value;
+  value.fill(0.f);
+  EXPECT_DEATH(aec3::UpdateDbMetric(value, nullptr), "");
+}
+
+#endif
+
+// Verifies the updating functionality of UpdateDbMetric.
+TEST(UpdateDbMetric, Updating) {
+  std::array<float, kFftLengthBy2Plus1> value;
+  std::array<EchoRemoverMetrics::DbMetric, 2> statistic;
+  statistic.fill(EchoRemoverMetrics::DbMetric(0.f, 100.f, -100.f));
+  constexpr float kValue0 = 10.f;
+  constexpr float kValue1 = 20.f;
+  std::fill(value.begin(), value.begin() + 32, kValue0);
+  std::fill(value.begin() + 32, value.begin() + 64, kValue1);
+
+  aec3::UpdateDbMetric(value, &statistic);
+  EXPECT_FLOAT_EQ(kValue0, statistic[0].sum_value);
+  EXPECT_FLOAT_EQ(kValue0, statistic[0].ceil_value);
+  EXPECT_FLOAT_EQ(kValue0, statistic[0].floor_value);
+  EXPECT_FLOAT_EQ(kValue1, statistic[1].sum_value);
+  EXPECT_FLOAT_EQ(kValue1, statistic[1].ceil_value);
+  EXPECT_FLOAT_EQ(kValue1, statistic[1].floor_value);
+
+  aec3::UpdateDbMetric(value, &statistic);
+  EXPECT_FLOAT_EQ(2.f * kValue0, statistic[0].sum_value);
+  EXPECT_FLOAT_EQ(kValue0, statistic[0].ceil_value);
+  EXPECT_FLOAT_EQ(kValue0, statistic[0].floor_value);
+  EXPECT_FLOAT_EQ(2.f * kValue1, statistic[1].sum_value);
+  EXPECT_FLOAT_EQ(kValue1, statistic[1].ceil_value);
+  EXPECT_FLOAT_EQ(kValue1, statistic[1].floor_value);
+}
+
+// Verifies that the TransformDbMetricForReporting method produces the desired
+// output for values for dBFS.
+TEST(TransformDbMetricForReporting, DbFsScaling) {
+  std::array<float, kBlockSize> x;
+  FftData X;
+  std::array<float, kFftLengthBy2Plus1> X2;
+  Aec3Fft fft;
+  x.fill(1000.f);
+  fft.ZeroPaddedFft(x, Aec3Fft::Window::kRectangular, &X);
+  X.Spectrum(Aec3Optimization::kNone, X2);
+
+  float offset = -10.f * std::log10(32768.f * 32768.f);
+  EXPECT_NEAR(offset, -90.3f, 0.1f);
+  EXPECT_EQ(
+      static_cast<int>(30.3f),
+      aec3::TransformDbMetricForReporting(
+          true, 0.f, 90.f, offset, 1.f / (kBlockSize * kBlockSize), X2[0]));
+}
+
+// Verifies that the TransformDbMetricForReporting method is able to properly
+// limit the output.
+TEST(TransformDbMetricForReporting, Limits) {
+  EXPECT_EQ(0, aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f,
+                                                   0.001f));
+  EXPECT_EQ(10, aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f,
+                                                    100.f));
+}
+
+// Verifies that the TransformDbMetricForReporting method is able to properly
+// negate output.
+TEST(TransformDbMetricForReporting, Negate) {
+  EXPECT_EQ(10, aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f, 1.f,
+                                                    0.1f));
+  EXPECT_EQ(-10, aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f,
+                                                     1.f, 10.f));
+}
+
+// Verify the Update functionality of DbMetric.
+TEST(DbMetric, Update) {
+  EchoRemoverMetrics::DbMetric metric(0.f, 20.f, -20.f);
+  constexpr int kNumValues = 100;
+  constexpr float kValue = 10.f;
+  for (int k = 0; k < kNumValues; ++k) {
+    metric.Update(kValue);
+  }
+  EXPECT_FLOAT_EQ(kValue * kNumValues, metric.sum_value);
+  EXPECT_FLOAT_EQ(kValue, metric.ceil_value);
+  EXPECT_FLOAT_EQ(kValue, metric.floor_value);
+}
+
+// Verify the Update functionality of DbMetric.
+TEST(DbMetric, UpdateInstant) {
+  EchoRemoverMetrics::DbMetric metric(0.f, 20.f, -20.f);
+  constexpr float kMinValue = -77.f;
+  constexpr float kMaxValue = 33.f;
+  constexpr float kLastValue = (kMinValue + kMaxValue) / 2.0f;
+  for (float value = kMinValue; value <= kMaxValue; value++)
+    metric.UpdateInstant(value);
+  metric.UpdateInstant(kLastValue);
+  EXPECT_FLOAT_EQ(kLastValue, metric.sum_value);
+  EXPECT_FLOAT_EQ(kMaxValue, metric.ceil_value);
+  EXPECT_FLOAT_EQ(kMinValue, metric.floor_value);
+}
+
+// Verify the constructor functionality of DbMetric.
+TEST(DbMetric, Constructor) {
+  EchoRemoverMetrics::DbMetric metric;
+  EXPECT_FLOAT_EQ(0.f, metric.sum_value);
+  EXPECT_FLOAT_EQ(0.f, metric.ceil_value);
+  EXPECT_FLOAT_EQ(0.f, metric.floor_value);
+
+  metric = EchoRemoverMetrics::DbMetric(1.f, 2.f, 3.f);
+  EXPECT_FLOAT_EQ(1.f, metric.sum_value);
+  EXPECT_FLOAT_EQ(2.f, metric.floor_value);
+  EXPECT_FLOAT_EQ(3.f, metric.ceil_value);
+}
+
+// Verify the general functionality of EchoRemoverMetrics.
+TEST(EchoRemoverMetrics, NormalUsage) {
+  EchoRemoverMetrics metrics;
+  AecState aec_state(EchoCanceller3Config{}, 1);
+  std::array<float, kFftLengthBy2Plus1> comfort_noise_spectrum;
+  std::array<float, kFftLengthBy2Plus1> suppressor_gain;
+  comfort_noise_spectrum.fill(10.f);
+  suppressor_gain.fill(1.f);
+  for (int j = 0; j < 3; ++j) {
+    for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
+      metrics.Update(aec_state, comfort_noise_spectrum, suppressor_gain);
+      EXPECT_FALSE(metrics.MetricsReported());
+    }
+    metrics.Update(aec_state, comfort_noise_spectrum, suppressor_gain);
+    EXPECT_TRUE(metrics.MetricsReported());
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_unittest.cc
new file mode 100644
index 0000000000..66168ab08d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/echo_remover_unittest.cc
@@ -0,0 +1,210 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/echo_remover.h"
+
+#include <algorithm>
+#include <memory>
+#include <numeric>
+#include <string>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+std::string ProduceDebugText(int sample_rate_hz) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz;
+  return ss.Release();
+}
+
+std::string ProduceDebugText(int sample_rate_hz, int delay) {
+  rtc::StringBuilder ss(ProduceDebugText(sample_rate_hz));
+  ss << ", Delay: " << delay;
+  return ss.Release();
+}
+
+}  // namespace
+
+class EchoRemoverMultiChannel
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         EchoRemoverMultiChannel,
+                         ::testing::Combine(::testing::Values(1, 2, 8),
+                                            ::testing::Values(1, 2, 8)));
+
+// Verifies the basic API call sequence
+TEST_P(EchoRemoverMultiChannel, BasicApiCalls) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+  absl::optional<DelayEstimate> delay_estimate;
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    std::unique_ptr<EchoRemover> remover(
+        EchoRemover::Create(EchoCanceller3Config(), rate, num_render_channels,
+                            num_capture_channels));
+    std::unique_ptr<RenderDelayBuffer> render_buffer(RenderDelayBuffer::Create(
+        EchoCanceller3Config(), rate, num_render_channels));
+
+    Block render(NumBandsForRate(rate), num_render_channels);
+    Block capture(NumBandsForRate(rate), num_capture_channels);
+    for (size_t k = 0; k < 100; ++k) {
+      EchoPathVariability echo_path_variability(
+          k % 3 == 0 ? true : false,
+          k % 5 == 0 ? EchoPathVariability::DelayAdjustment::kNewDetectedDelay
+                     : EchoPathVariability::DelayAdjustment::kNone,
+          false);
+      render_buffer->Insert(render);
+      render_buffer->PrepareCaptureProcessing();
+
+      remover->ProcessCapture(echo_path_variability, k % 2 == 0 ? true : false,
+                              delay_estimate, render_buffer->GetRenderBuffer(),
+                              nullptr, &capture);
+    }
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for the samplerate.
+// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
+// tests on test bots has been fixed.
+TEST(EchoRemoverDeathTest, DISABLED_WrongSampleRate) {
+  EXPECT_DEATH(std::unique_ptr<EchoRemover>(
+                   EchoRemover::Create(EchoCanceller3Config(), 8001, 1, 1)),
+               "");
+}
+
+// Verifies the check for the number of capture bands.
+// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
+// tests on test bots has been fixed.c
+TEST(EchoRemoverDeathTest, DISABLED_WrongCaptureNumBands) {
+  absl::optional<DelayEstimate> delay_estimate;
+  for (auto rate : {16000, 32000, 48000}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    std::unique_ptr<EchoRemover> remover(
+        EchoRemover::Create(EchoCanceller3Config(), rate, 1, 1));
+    std::unique_ptr<RenderDelayBuffer> render_buffer(
+        RenderDelayBuffer::Create(EchoCanceller3Config(), rate, 1));
+    Block capture(NumBandsForRate(rate == 48000 ? 16000 : rate + 16000), 1);
+    EchoPathVariability echo_path_variability(
+        false, EchoPathVariability::DelayAdjustment::kNone, false);
+    EXPECT_DEATH(remover->ProcessCapture(
+                     echo_path_variability, false, delay_estimate,
+                     render_buffer->GetRenderBuffer(), nullptr, &capture),
+                 "");
+  }
+}
+
+// Verifies the check for non-null capture block.
+TEST(EchoRemoverDeathTest, NullCapture) {
+  absl::optional<DelayEstimate> delay_estimate;
+  std::unique_ptr<EchoRemover> remover(
+      EchoRemover::Create(EchoCanceller3Config(), 16000, 1, 1));
+  std::unique_ptr<RenderDelayBuffer> render_buffer(
+      RenderDelayBuffer::Create(EchoCanceller3Config(), 16000, 1));
+  EchoPathVariability echo_path_variability(
+      false, EchoPathVariability::DelayAdjustment::kNone, false);
+  EXPECT_DEATH(remover->ProcessCapture(
+                   echo_path_variability, false, delay_estimate,
+                   render_buffer->GetRenderBuffer(), nullptr, nullptr),
+               "");
+}
+
+#endif
+
+// Performs a sanity check that the echo_remover is able to properly
+// remove echoes.
+TEST(EchoRemover, BasicEchoRemoval) {
+  constexpr int kNumBlocksToProcess = 500;
+  Random random_generator(42U);
+  absl::optional<DelayEstimate> delay_estimate;
+  for (size_t num_channels : {1, 2, 4}) {
+    for (auto rate : {16000, 32000, 48000}) {
+      Block x(NumBandsForRate(rate), num_channels);
+      Block y(NumBandsForRate(rate), num_channels);
+      EchoPathVariability echo_path_variability(
+          false, EchoPathVariability::DelayAdjustment::kNone, false);
+      for (size_t delay_samples : {0, 64, 150, 200, 301}) {
+        SCOPED_TRACE(ProduceDebugText(rate, delay_samples));
+        EchoCanceller3Config config;
+        std::unique_ptr<EchoRemover> remover(
+            EchoRemover::Create(config, rate, num_channels, num_channels));
+        std::unique_ptr<RenderDelayBuffer> render_buffer(
+            RenderDelayBuffer::Create(config, rate, num_channels));
+        render_buffer->AlignFromDelay(delay_samples / kBlockSize);
+
+        std::vector<std::vector<std::unique_ptr<DelayBuffer<float>>>>
+            delay_buffers(x.NumBands());
+        for (size_t band = 0; band < delay_buffers.size(); ++band) {
+          delay_buffers[band].resize(x.NumChannels());
+        }
+
+        for (int band = 0; band < x.NumBands(); ++band) {
+          for (int channel = 0; channel < x.NumChannels(); ++channel) {
+            delay_buffers[band][channel].reset(
+                new DelayBuffer<float>(delay_samples));
+          }
+        }
+
+        float input_energy = 0.f;
+        float output_energy = 0.f;
+        for (int k = 0; k < kNumBlocksToProcess; ++k) {
+          const bool silence = k < 100 || (k % 100 >= 10);
+
+          for (int band = 0; band < x.NumBands(); ++band) {
+            for (int channel = 0; channel < x.NumChannels(); ++channel) {
+              if (silence) {
+                std::fill(x.begin(band, channel), x.end(band, channel), 0.f);
+              } else {
+                RandomizeSampleVector(&random_generator, x.View(band, channel));
+              }
+              delay_buffers[band][channel]->Delay(x.View(band, channel),
+                                                  y.View(band, channel));
+            }
+          }
+
+          if (k > kNumBlocksToProcess / 2) {
+            input_energy = std::inner_product(
+                y.begin(/*band=*/0, /*channel=*/0),
+                y.end(/*band=*/0, /*channel=*/0),
+                y.begin(/*band=*/0, /*channel=*/0), input_energy);
+          }
+
+          render_buffer->Insert(x);
+          render_buffer->PrepareCaptureProcessing();
+
+          remover->ProcessCapture(echo_path_variability, false, delay_estimate,
+                                  render_buffer->GetRenderBuffer(), nullptr,
+                                  &y);
+
+          if (k > kNumBlocksToProcess / 2) {
+            output_energy = std::inner_product(
+                y.begin(/*band=*/0, /*channel=*/0),
+                y.end(/*band=*/0, /*channel=*/0),
+                y.begin(/*band=*/0, /*channel=*/0), output_energy);
+          }
+        }
+        EXPECT_GT(input_energy, 10.f * output_energy);
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator.cc
new file mode 100644
index 0000000000..01cc33cb80
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator.cc
@@ -0,0 +1,146 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/erl_estimator.h"
+
+#include <algorithm>
+#include <numeric>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr float kMinErl = 0.01f;
+constexpr float kMaxErl = 1000.f;
+
+}  // namespace
+
+ErlEstimator::ErlEstimator(size_t startup_phase_length_blocks_)
+    : startup_phase_length_blocks__(startup_phase_length_blocks_) {
+  erl_.fill(kMaxErl);
+  hold_counters_.fill(0);
+  erl_time_domain_ = kMaxErl;
+  hold_counter_time_domain_ = 0;
+}
+
+ErlEstimator::~ErlEstimator() = default;
+
+void ErlEstimator::Reset() {
+  blocks_since_reset_ = 0;
+}
+
+void ErlEstimator::Update(
+    const std::vector<bool>& converged_filters,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> render_spectra,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        capture_spectra) {
+  const size_t num_capture_channels = converged_filters.size();
+  RTC_DCHECK_EQ(capture_spectra.size(), num_capture_channels);
+
+  // Corresponds to WGN of power -46 dBFS.
+  constexpr float kX2Min = 44015068.0f;
+
+  const auto first_converged_iter =
+      std::find(converged_filters.begin(), converged_filters.end(), true);
+  const bool any_filter_converged =
+      first_converged_iter != converged_filters.end();
+
+  if (++blocks_since_reset_ < startup_phase_length_blocks__ ||
+      !any_filter_converged) {
+    return;
+  }
+
+  // Use the maximum spectrum across capture and the maximum across render.
+  std::array<float, kFftLengthBy2Plus1> max_capture_spectrum_data;
+  std::array<float, kFftLengthBy2Plus1> max_capture_spectrum =
+      capture_spectra[/*channel=*/0];
+  if (num_capture_channels > 1) {
+    // Initialize using the first channel with a converged filter.
+    const size_t first_converged =
+        std::distance(converged_filters.begin(), first_converged_iter);
+    RTC_DCHECK_GE(first_converged, 0);
+    RTC_DCHECK_LT(first_converged, num_capture_channels);
+    max_capture_spectrum_data = capture_spectra[first_converged];
+
+    for (size_t ch = first_converged + 1; ch < num_capture_channels; ++ch) {
+      if (!converged_filters[ch]) {
+        continue;
+      }
+      for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+        max_capture_spectrum_data[k] =
+            std::max(max_capture_spectrum_data[k], capture_spectra[ch][k]);
+      }
+    }
+    max_capture_spectrum = max_capture_spectrum_data;
+  }
+
+  const size_t num_render_channels = render_spectra.size();
+  std::array<float, kFftLengthBy2Plus1> max_render_spectrum_data;
+  rtc::ArrayView<const float, kFftLengthBy2Plus1> max_render_spectrum =
+      render_spectra[/*channel=*/0];
+  if (num_render_channels > 1) {
+    std::copy(render_spectra[0].begin(), render_spectra[0].end(),
+              max_render_spectrum_data.begin());
+    for (size_t ch = 1; ch < num_render_channels; ++ch) {
+      for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+        max_render_spectrum_data[k] =
+            std::max(max_render_spectrum_data[k], render_spectra[ch][k]);
+      }
+    }
+    max_render_spectrum = max_render_spectrum_data;
+  }
+
+  const auto& X2 = max_render_spectrum;
+  const auto& Y2 = max_capture_spectrum;
+
+  // Update the estimates in a maximum statistics manner.
+  for (size_t k = 1; k < kFftLengthBy2; ++k) {
+    if (X2[k] > kX2Min) {
+      const float new_erl = Y2[k] / X2[k];
+      if (new_erl < erl_[k]) {
+        hold_counters_[k - 1] = 1000;
+        erl_[k] += 0.1f * (new_erl - erl_[k]);
+        erl_[k] = std::max(erl_[k], kMinErl);
+      }
+    }
+  }
+
+  std::for_each(hold_counters_.begin(), hold_counters_.end(),
+                [](int& a) { --a; });
+  std::transform(hold_counters_.begin(), hold_counters_.end(), erl_.begin() + 1,
+                 erl_.begin() + 1, [](int a, float b) {
+                   return a > 0 ? b : std::min(kMaxErl, 2.f * b);
+                 });
+
+  erl_[0] = erl_[1];
+  erl_[kFftLengthBy2] = erl_[kFftLengthBy2 - 1];
+
+  // Compute ERL over all frequency bins.
+  const float X2_sum = std::accumulate(X2.begin(), X2.end(), 0.0f);
+
+  if (X2_sum > kX2Min * X2.size()) {
+    const float Y2_sum = std::accumulate(Y2.begin(), Y2.end(), 0.0f);
+    const float new_erl = Y2_sum / X2_sum;
+    if (new_erl < erl_time_domain_) {
+      hold_counter_time_domain_ = 1000;
+      erl_time_domain_ += 0.1f * (new_erl - erl_time_domain_);
+      erl_time_domain_ = std::max(erl_time_domain_, kMinErl);
+    }
+  }
+
+  --hold_counter_time_domain_;
+  erl_time_domain_ = (hold_counter_time_domain_ > 0)
+                         ? erl_time_domain_
+                         : std::min(kMaxErl, 2.f * erl_time_domain_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator.h
new file mode 100644
index 0000000000..639a52c561
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ERL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ERL_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+
+// Estimates the echo return loss based on the signal spectra.
+class ErlEstimator {
+ public:
+  explicit ErlEstimator(size_t startup_phase_length_blocks_);
+  ~ErlEstimator();
+
+  ErlEstimator(const ErlEstimator&) = delete;
+  ErlEstimator& operator=(const ErlEstimator&) = delete;
+
+  // Resets the ERL estimation.
+  void Reset();
+
+  // Updates the ERL estimate.
+  void Update(const std::vector<bool>& converged_filters,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+                  render_spectra,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+                  capture_spectra);
+
+  // Returns the most recent ERL estimate.
+  const std::array<float, kFftLengthBy2Plus1>& Erl() const { return erl_; }
+  float ErlTimeDomain() const { return erl_time_domain_; }
+
+ private:
+  const size_t startup_phase_length_blocks__;
+  std::array<float, kFftLengthBy2Plus1> erl_;
+  std::array<int, kFftLengthBy2Minus1> hold_counters_;
+  float erl_time_domain_;
+  int hold_counter_time_domain_;
+  size_t blocks_since_reset_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ERL_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator_unittest.cc
new file mode 100644
index 0000000000..79e5465e3c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/erl_estimator_unittest.cc
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/erl_estimator.h"
+
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+std::string ProduceDebugText(size_t num_render_channels,
+                             size_t num_capture_channels) {
+  rtc::StringBuilder ss;
+  ss << "Render channels: " << num_render_channels;
+  ss << ", Capture channels: " << num_capture_channels;
+  return ss.Release();
+}
+
+void VerifyErl(const std::array<float, kFftLengthBy2Plus1>& erl,
+               float erl_time_domain,
+               float reference) {
+  std::for_each(erl.begin(), erl.end(),
+                [reference](float a) { EXPECT_NEAR(reference, a, 0.001); });
+  EXPECT_NEAR(reference, erl_time_domain, 0.001);
+}
+
+}  // namespace
+
+class ErlEstimatorMultiChannel
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         ErlEstimatorMultiChannel,
+                         ::testing::Combine(::testing::Values(1, 2, 8),
+                                            ::testing::Values(1, 2, 8)));
+
+// Verifies that the correct ERL estimates are achieved.
+TEST_P(ErlEstimatorMultiChannel, Estimates) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+  SCOPED_TRACE(ProduceDebugText(num_render_channels, num_capture_channels));
+  std::vector<std::array<float, kFftLengthBy2Plus1>> X2(num_render_channels);
+  for (auto& X2_ch : X2) {
+    X2_ch.fill(0.f);
+  }
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(num_capture_channels);
+  for (auto& Y2_ch : Y2) {
+    Y2_ch.fill(0.f);
+  }
+  std::vector<bool> converged_filters(num_capture_channels, false);
+  const size_t converged_idx = num_capture_channels - 1;
+  converged_filters[converged_idx] = true;
+
+  ErlEstimator estimator(0);
+
+  // Verifies that the ERL estimate is properly reduced to lower values.
+  for (auto& X2_ch : X2) {
+    X2_ch.fill(500 * 1000.f * 1000.f);
+  }
+  Y2[converged_idx].fill(10 * X2[0][0]);
+  for (size_t k = 0; k < 200; ++k) {
+    estimator.Update(converged_filters, X2, Y2);
+  }
+  VerifyErl(estimator.Erl(), estimator.ErlTimeDomain(), 10.f);
+
+  // Verifies that the ERL is not immediately increased when the ERL in the
+  // data increases.
+  Y2[converged_idx].fill(10000 * X2[0][0]);
+  for (size_t k = 0; k < 998; ++k) {
+    estimator.Update(converged_filters, X2, Y2);
+  }
+  VerifyErl(estimator.Erl(), estimator.ErlTimeDomain(), 10.f);
+
+  // Verifies that the rate of increase is 3 dB.
+  estimator.Update(converged_filters, X2, Y2);
+  VerifyErl(estimator.Erl(), estimator.ErlTimeDomain(), 20.f);
+
+  // Verifies that the maximum ERL is achieved when there are no low RLE
+  // estimates.
+  for (size_t k = 0; k < 1000; ++k) {
+    estimator.Update(converged_filters, X2, Y2);
+  }
+  VerifyErl(estimator.Erl(), estimator.ErlTimeDomain(), 1000.f);
+
+  // Verifies that the ERL estimate is is not updated for low-level signals
+  for (auto& X2_ch : X2) {
+    X2_ch.fill(1000.f * 1000.f);
+  }
+  Y2[converged_idx].fill(10 * X2[0][0]);
+  for (size_t k = 0; k < 200; ++k) {
+    estimator.Update(converged_filters, X2, Y2);
+  }
+  VerifyErl(estimator.Erl(), estimator.ErlTimeDomain(), 1000.f);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator.cc
new file mode 100644
index 0000000000..0e3d715c59
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/erle_estimator.h"
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+ErleEstimator::ErleEstimator(size_t startup_phase_length_blocks,
+                             const EchoCanceller3Config& config,
+                             size_t num_capture_channels)
+    : startup_phase_length_blocks_(startup_phase_length_blocks),
+      fullband_erle_estimator_(config.erle, num_capture_channels),
+      subband_erle_estimator_(config, num_capture_channels) {
+  if (config.erle.num_sections > 1) {
+    signal_dependent_erle_estimator_ =
+        std::make_unique<SignalDependentErleEstimator>(config,
+                                                       num_capture_channels);
+  }
+  Reset(true);
+}
+
+ErleEstimator::~ErleEstimator() = default;
+
+void ErleEstimator::Reset(bool delay_change) {
+  fullband_erle_estimator_.Reset();
+  subband_erle_estimator_.Reset();
+  if (signal_dependent_erle_estimator_) {
+    signal_dependent_erle_estimator_->Reset();
+  }
+  if (delay_change) {
+    blocks_since_reset_ = 0;
+  }
+}
+
+void ErleEstimator::Update(
+    const RenderBuffer& render_buffer,
+    rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+        filter_frequency_responses,
+    rtc::ArrayView<const float, kFftLengthBy2Plus1>
+        avg_render_spectrum_with_reverb,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> capture_spectra,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        subtractor_spectra,
+    const std::vector<bool>& converged_filters) {
+  RTC_DCHECK_EQ(subband_erle_estimator_.Erle(/*onset_compensated=*/true).size(),
+                capture_spectra.size());
+  RTC_DCHECK_EQ(subband_erle_estimator_.Erle(/*onset_compensated=*/true).size(),
+                subtractor_spectra.size());
+  const auto& X2_reverb = avg_render_spectrum_with_reverb;
+  const auto& Y2 = capture_spectra;
+  const auto& E2 = subtractor_spectra;
+
+  if (++blocks_since_reset_ < startup_phase_length_blocks_) {
+    return;
+  }
+
+  subband_erle_estimator_.Update(X2_reverb, Y2, E2, converged_filters);
+
+  if (signal_dependent_erle_estimator_) {
+    signal_dependent_erle_estimator_->Update(
+        render_buffer, filter_frequency_responses, X2_reverb, Y2, E2,
+        subband_erle_estimator_.Erle(/*onset_compensated=*/false),
+        subband_erle_estimator_.Erle(/*onset_compensated=*/true),
+        converged_filters);
+  }
+
+  fullband_erle_estimator_.Update(X2_reverb, Y2, E2, converged_filters);
+}
+
+void ErleEstimator::Dump(
+    const std::unique_ptr<ApmDataDumper>& data_dumper) const {
+  fullband_erle_estimator_.Dump(data_dumper);
+  subband_erle_estimator_.Dump(data_dumper);
+  if (signal_dependent_erle_estimator_) {
+    signal_dependent_erle_estimator_->Dump(data_dumper);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator.h
new file mode 100644
index 0000000000..55797592a9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator.h
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ERLE_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_ERLE_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/fullband_erle_estimator.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/signal_dependent_erle_estimator.h"
+#include "modules/audio_processing/aec3/subband_erle_estimator.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+// Estimates the echo return loss enhancement. One estimate is done per subband
+// and another one is done using the aggreation of energy over all the subbands.
+class ErleEstimator {
+ public:
+  ErleEstimator(size_t startup_phase_length_blocks,
+                const EchoCanceller3Config& config,
+                size_t num_capture_channels);
+  ~ErleEstimator();
+
+  // Resets the fullband ERLE estimator and the subbands ERLE estimators.
+  void Reset(bool delay_change);
+
+  // Updates the ERLE estimates.
+  void Update(
+      const RenderBuffer& render_buffer,
+      rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+          filter_frequency_responses,
+      rtc::ArrayView<const float, kFftLengthBy2Plus1>
+          avg_render_spectrum_with_reverb,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          capture_spectra,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          subtractor_spectra,
+      const std::vector<bool>& converged_filters);
+
+  // Returns the most recent subband ERLE estimates.
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Erle(
+      bool onset_compensated) const {
+    return signal_dependent_erle_estimator_
+               ? signal_dependent_erle_estimator_->Erle(onset_compensated)
+               : subband_erle_estimator_.Erle(onset_compensated);
+  }
+
+  // Returns the non-capped subband ERLE.
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleUnbounded()
+      const {
+    // Unbounded ERLE is only used with the subband erle estimator where the
+    // ERLE is often capped at low values. When the signal dependent ERLE
+    // estimator is used the capped ERLE is returned.
+    return !signal_dependent_erle_estimator_
+               ? subband_erle_estimator_.ErleUnbounded()
+               : signal_dependent_erle_estimator_->Erle(
+                     /*onset_compensated=*/false);
+  }
+
+  // Returns the subband ERLE that are estimated during onsets (only used for
+  // testing).
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleDuringOnsets()
+      const {
+    return subband_erle_estimator_.ErleDuringOnsets();
+  }
+
+  // Returns the fullband ERLE estimate.
+  float FullbandErleLog2() const {
+    return fullband_erle_estimator_.FullbandErleLog2();
+  }
+
+  // Returns an estimation of the current linear filter quality based on the
+  // current and past fullband ERLE estimates. The returned value is a float
+  // vector with content between 0 and 1 where 1 indicates that, at this current
+  // time instant, the linear filter is reaching its maximum subtraction
+  // performance.
+  rtc::ArrayView<const absl::optional<float>> GetInstLinearQualityEstimates()
+      const {
+    return fullband_erle_estimator_.GetInstLinearQualityEstimates();
+  }
+
+  void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper) const;
+
+ private:
+  const size_t startup_phase_length_blocks_;
+  FullBandErleEstimator fullband_erle_estimator_;
+  SubbandErleEstimator subband_erle_estimator_;
+  std::unique_ptr<SignalDependentErleEstimator>
+      signal_dependent_erle_estimator_;
+  size_t blocks_since_reset_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ERLE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc
new file mode 100644
index 0000000000..42be7d9c7d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc
@@ -0,0 +1,288 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/erle_estimator.h"
+
+#include <cmath>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/aec3/spectrum_buffer.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kLowFrequencyLimit = kFftLengthBy2 / 2;
+constexpr float kTrueErle = 10.f;
+constexpr float kTrueErleOnsets = 1.0f;
+constexpr float kEchoPathGain = 3.f;
+
+void VerifyErleBands(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle,
+    float reference_lf,
+    float reference_hf) {
+  for (size_t ch = 0; ch < erle.size(); ++ch) {
+    std::for_each(
+        erle[ch].begin(), erle[ch].begin() + kLowFrequencyLimit,
+        [reference_lf](float a) { EXPECT_NEAR(reference_lf, a, 0.001); });
+    std::for_each(
+        erle[ch].begin() + kLowFrequencyLimit, erle[ch].end(),
+        [reference_hf](float a) { EXPECT_NEAR(reference_hf, a, 0.001); });
+  }
+}
+
+void VerifyErle(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle,
+    float erle_time_domain,
+    float reference_lf,
+    float reference_hf) {
+  VerifyErleBands(erle, reference_lf, reference_hf);
+  EXPECT_NEAR(kTrueErle, erle_time_domain, 0.5);
+}
+
+void VerifyErleGreaterOrEqual(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle1,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle2) {
+  for (size_t ch = 0; ch < erle1.size(); ++ch) {
+    for (size_t i = 0; i < kFftLengthBy2Plus1; ++i) {
+      EXPECT_GE(erle1[ch][i], erle2[ch][i]);
+    }
+  }
+}
+
+void FormFarendTimeFrame(Block* x) {
+  const std::array<float, kBlockSize> frame = {
+      7459.88, 17209.6, 17383,   20768.9, 16816.7, 18386.3, 4492.83, 9675.85,
+      6665.52, 14808.6, 9342.3,  7483.28, 19261.7, 4145.98, 1622.18, 13475.2,
+      7166.32, 6856.61, 21937,   7263.14, 9569.07, 14919,   8413.32, 7551.89,
+      7848.65, 6011.27, 13080.6, 15865.2, 12656,   17459.6, 4263.93, 4503.03,
+      9311.79, 21095.8, 12657.9, 13906.6, 19267.2, 11338.1, 16828.9, 11501.6,
+      11405,   15031.4, 14541.6, 19765.5, 18346.3, 19350.2, 3157.47, 18095.8,
+      1743.68, 21328.2, 19727.5, 7295.16, 10332.4, 11055.5, 20107.4, 14708.4,
+      12416.2, 16434,   2454.69, 9840.8,  6867.23, 1615.75, 6059.9,  8394.19};
+  for (int band = 0; band < x->NumBands(); ++band) {
+    for (int channel = 0; channel < x->NumChannels(); ++channel) {
+      RTC_DCHECK_GE(kBlockSize, frame.size());
+      std::copy(frame.begin(), frame.end(), x->begin(band, channel));
+    }
+  }
+}
+
+void FormFarendFrame(const RenderBuffer& render_buffer,
+                     float erle,
+                     std::array<float, kFftLengthBy2Plus1>* X2,
+                     rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> E2,
+                     rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> Y2) {
+  const auto& spectrum_buffer = render_buffer.GetSpectrumBuffer();
+  const int num_render_channels = spectrum_buffer.buffer[0].size();
+  const int num_capture_channels = Y2.size();
+
+  X2->fill(0.f);
+  for (int ch = 0; ch < num_render_channels; ++ch) {
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      (*X2)[k] += spectrum_buffer.buffer[spectrum_buffer.write][ch][k] /
+                  num_render_channels;
+    }
+  }
+
+  for (int ch = 0; ch < num_capture_channels; ++ch) {
+    std::transform(X2->begin(), X2->end(), Y2[ch].begin(),
+                   [](float a) { return a * kEchoPathGain * kEchoPathGain; });
+    std::transform(Y2[ch].begin(), Y2[ch].end(), E2[ch].begin(),
+                   [erle](float a) { return a / erle; });
+  }
+}
+
+void FormNearendFrame(
+    Block* x,
+    std::array<float, kFftLengthBy2Plus1>* X2,
+    rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> E2,
+    rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> Y2) {
+  for (int band = 0; band < x->NumBands(); ++band) {
+    for (int ch = 0; ch < x->NumChannels(); ++ch) {
+      std::fill(x->begin(band, ch), x->end(band, ch), 0.f);
+    }
+  }
+
+  X2->fill(0.f);
+  for (size_t ch = 0; ch < Y2.size(); ++ch) {
+    Y2[ch].fill(500.f * 1000.f * 1000.f);
+    E2[ch].fill(Y2[ch][0]);
+  }
+}
+
+void GetFilterFreq(
+    size_t delay_headroom_samples,
+    rtc::ArrayView<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+        filter_frequency_response) {
+  const size_t delay_headroom_blocks = delay_headroom_samples / kBlockSize;
+  for (size_t ch = 0; ch < filter_frequency_response[0].size(); ++ch) {
+    for (auto& block_freq_resp : filter_frequency_response) {
+      block_freq_resp[ch].fill(0.f);
+    }
+
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      filter_frequency_response[delay_headroom_blocks][ch][k] = kEchoPathGain;
+    }
+  }
+}
+
+}  // namespace
+
+class ErleEstimatorMultiChannel
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         ErleEstimatorMultiChannel,
+                         ::testing::Combine(::testing::Values(1, 2, 4, 8),
+                                            ::testing::Values(1, 2, 8)));
+
+TEST_P(ErleEstimatorMultiChannel, VerifyErleIncreaseAndHold) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  std::array<float, kFftLengthBy2Plus1> X2;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2(num_capture_channels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(num_capture_channels);
+  std::vector<bool> converged_filters(num_capture_channels, true);
+
+  EchoCanceller3Config config;
+  config.erle.onset_detection = true;
+
+  Block x(kNumBands, num_render_channels);
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      filter_frequency_response(
+          config.filter.refined.length_blocks,
+          std::vector<std::array<float, kFftLengthBy2Plus1>>(
+              num_capture_channels));
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, num_render_channels));
+
+  GetFilterFreq(config.delay.delay_headroom_samples, filter_frequency_response);
+
+  ErleEstimator estimator(0, config, num_capture_channels);
+
+  FormFarendTimeFrame(&x);
+  render_delay_buffer->Insert(x);
+  render_delay_buffer->PrepareCaptureProcessing();
+  // Verifies that the ERLE estimate is properly increased to higher values.
+  FormFarendFrame(*render_delay_buffer->GetRenderBuffer(), kTrueErle, &X2, E2,
+                  Y2);
+  for (size_t k = 0; k < 1000; ++k) {
+    render_delay_buffer->Insert(x);
+    render_delay_buffer->PrepareCaptureProcessing();
+    estimator.Update(*render_delay_buffer->GetRenderBuffer(),
+                     filter_frequency_response, X2, Y2, E2, converged_filters);
+  }
+  VerifyErle(estimator.Erle(/*onset_compensated=*/true),
+             std::pow(2.f, estimator.FullbandErleLog2()), config.erle.max_l,
+             config.erle.max_h);
+  VerifyErleGreaterOrEqual(estimator.Erle(/*onset_compensated=*/false),
+                           estimator.Erle(/*onset_compensated=*/true));
+  VerifyErleGreaterOrEqual(estimator.ErleUnbounded(),
+                           estimator.Erle(/*onset_compensated=*/false));
+
+  FormNearendFrame(&x, &X2, E2, Y2);
+  // Verifies that the ERLE is not immediately decreased during nearend
+  // activity.
+  for (size_t k = 0; k < 50; ++k) {
+    render_delay_buffer->Insert(x);
+    render_delay_buffer->PrepareCaptureProcessing();
+    estimator.Update(*render_delay_buffer->GetRenderBuffer(),
+                     filter_frequency_response, X2, Y2, E2, converged_filters);
+  }
+  VerifyErle(estimator.Erle(/*onset_compensated=*/true),
+             std::pow(2.f, estimator.FullbandErleLog2()), config.erle.max_l,
+             config.erle.max_h);
+  VerifyErleGreaterOrEqual(estimator.Erle(/*onset_compensated=*/false),
+                           estimator.Erle(/*onset_compensated=*/true));
+  VerifyErleGreaterOrEqual(estimator.ErleUnbounded(),
+                           estimator.Erle(/*onset_compensated=*/false));
+}
+
+TEST_P(ErleEstimatorMultiChannel, VerifyErleTrackingOnOnsets) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  std::array<float, kFftLengthBy2Plus1> X2;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2(num_capture_channels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(num_capture_channels);
+  std::vector<bool> converged_filters(num_capture_channels, true);
+  EchoCanceller3Config config;
+  config.erle.onset_detection = true;
+  Block x(kNumBands, num_render_channels);
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      filter_frequency_response(
+          config.filter.refined.length_blocks,
+          std::vector<std::array<float, kFftLengthBy2Plus1>>(
+              num_capture_channels));
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, num_render_channels));
+
+  GetFilterFreq(config.delay.delay_headroom_samples, filter_frequency_response);
+
+  ErleEstimator estimator(/*startup_phase_length_blocks=*/0, config,
+                          num_capture_channels);
+
+  FormFarendTimeFrame(&x);
+  render_delay_buffer->Insert(x);
+  render_delay_buffer->PrepareCaptureProcessing();
+
+  for (size_t burst = 0; burst < 20; ++burst) {
+    FormFarendFrame(*render_delay_buffer->GetRenderBuffer(), kTrueErleOnsets,
+                    &X2, E2, Y2);
+    for (size_t k = 0; k < 10; ++k) {
+      render_delay_buffer->Insert(x);
+      render_delay_buffer->PrepareCaptureProcessing();
+      estimator.Update(*render_delay_buffer->GetRenderBuffer(),
+                       filter_frequency_response, X2, Y2, E2,
+                       converged_filters);
+    }
+    FormFarendFrame(*render_delay_buffer->GetRenderBuffer(), kTrueErle, &X2, E2,
+                    Y2);
+    for (size_t k = 0; k < 1000; ++k) {
+      render_delay_buffer->Insert(x);
+      render_delay_buffer->PrepareCaptureProcessing();
+      estimator.Update(*render_delay_buffer->GetRenderBuffer(),
+                       filter_frequency_response, X2, Y2, E2,
+                       converged_filters);
+    }
+    FormNearendFrame(&x, &X2, E2, Y2);
+    for (size_t k = 0; k < 300; ++k) {
+      render_delay_buffer->Insert(x);
+      render_delay_buffer->PrepareCaptureProcessing();
+      estimator.Update(*render_delay_buffer->GetRenderBuffer(),
+                       filter_frequency_response, X2, Y2, E2,
+                       converged_filters);
+    }
+  }
+  VerifyErleBands(estimator.ErleDuringOnsets(), config.erle.min,
+                  config.erle.min);
+  FormNearendFrame(&x, &X2, E2, Y2);
+  for (size_t k = 0; k < 1000; k++) {
+    estimator.Update(*render_delay_buffer->GetRenderBuffer(),
+                     filter_frequency_response, X2, Y2, E2, converged_filters);
+  }
+  // Verifies that during ne activity, Erle converges to the Erle for
+  // onsets.
+  VerifyErle(estimator.Erle(/*onset_compensated=*/true),
+             std::pow(2.f, estimator.FullbandErleLog2()), config.erle.min,
+             config.erle.min);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/fft_buffer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/fft_buffer.cc
new file mode 100644
index 0000000000..1ce2d31d8f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/fft_buffer.cc
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/fft_buffer.h"
+
+namespace webrtc {
+
+FftBuffer::FftBuffer(size_t size, size_t num_channels)
+    : size(static_cast<int>(size)),
+      buffer(size, std::vector<FftData>(num_channels)) {
+  for (auto& block : buffer) {
+    for (auto& channel_fft_data : block) {
+      channel_fft_data.Clear();
+    }
+  }
+}
+
+FftBuffer::~FftBuffer() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/fft_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/fft_buffer.h
new file mode 100644
index 0000000000..4187315863
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/fft_buffer.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Struct for bundling a circular buffer of FftData objects together with the
+// read and write indices.
+struct FftBuffer {
+  FftBuffer(size_t size, size_t num_channels);
+  ~FftBuffer();
+
+  int IncIndex(int index) const {
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return index < size - 1 ? index + 1 : 0;
+  }
+
+  int DecIndex(int index) const {
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return index > 0 ? index - 1 : size - 1;
+  }
+
+  int OffsetIndex(int index, int offset) const {
+    RTC_DCHECK_GE(buffer.size(), offset);
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return (size + index + offset) % size;
+  }
+
+  void UpdateWriteIndex(int offset) { write = OffsetIndex(write, offset); }
+  void IncWriteIndex() { write = IncIndex(write); }
+  void DecWriteIndex() { write = DecIndex(write); }
+  void UpdateReadIndex(int offset) { read = OffsetIndex(read, offset); }
+  void IncReadIndex() { read = IncIndex(read); }
+  void DecReadIndex() { read = DecIndex(read); }
+
+  const int size;
+  std::vector<std::vector<FftData>> buffer;
+  int write = 0;
+  int read = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/fft_data.h b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data.h
new file mode 100644
index 0000000000..9c25e784aa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data.h
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_FFT_DATA_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_FFT_DATA_H_
+
+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+#include <algorithm>
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+
+// Struct that holds imaginary data produced from 128 point real-valued FFTs.
+struct FftData {
+  // Copies the data in src.
+  void Assign(const FftData& src) {
+    std::copy(src.re.begin(), src.re.end(), re.begin());
+    std::copy(src.im.begin(), src.im.end(), im.begin());
+    im[0] = im[kFftLengthBy2] = 0;
+  }
+
+  // Clears all the imaginary.
+  void Clear() {
+    re.fill(0.f);
+    im.fill(0.f);
+  }
+
+  // Computes the power spectrum of the data.
+  void SpectrumAVX2(rtc::ArrayView<float> power_spectrum) const;
+
+  // Computes the power spectrum of the data.
+  void Spectrum(Aec3Optimization optimization,
+                rtc::ArrayView<float> power_spectrum) const {
+    RTC_DCHECK_EQ(kFftLengthBy2Plus1, power_spectrum.size());
+    switch (optimization) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+      case Aec3Optimization::kSse2: {
+        constexpr int kNumFourBinBands = kFftLengthBy2 / 4;
+        constexpr int kLimit = kNumFourBinBands * 4;
+        for (size_t k = 0; k < kLimit; k += 4) {
+          const __m128 r = _mm_loadu_ps(&re[k]);
+          const __m128 i = _mm_loadu_ps(&im[k]);
+          const __m128 ii = _mm_mul_ps(i, i);
+          const __m128 rr = _mm_mul_ps(r, r);
+          const __m128 rrii = _mm_add_ps(rr, ii);
+          _mm_storeu_ps(&power_spectrum[k], rrii);
+        }
+        power_spectrum[kFftLengthBy2] = re[kFftLengthBy2] * re[kFftLengthBy2] +
+                                        im[kFftLengthBy2] * im[kFftLengthBy2];
+      } break;
+      case Aec3Optimization::kAvx2:
+        SpectrumAVX2(power_spectrum);
+        break;
+#endif
+      default:
+        std::transform(re.begin(), re.end(), im.begin(), power_spectrum.begin(),
+                       [](float a, float b) { return a * a + b * b; });
+    }
+  }
+
+  // Copy the data from an interleaved array.
+  void CopyFromPackedArray(const std::array<float, kFftLength>& v) {
+    re[0] = v[0];
+    re[kFftLengthBy2] = v[1];
+    im[0] = im[kFftLengthBy2] = 0;
+    for (size_t k = 1, j = 2; k < kFftLengthBy2; ++k) {
+      re[k] = v[j++];
+      im[k] = v[j++];
+    }
+  }
+
+  // Copies the data into an interleaved array.
+  void CopyToPackedArray(std::array<float, kFftLength>* v) const {
+    RTC_DCHECK(v);
+    (*v)[0] = re[0];
+    (*v)[1] = re[kFftLengthBy2];
+    for (size_t k = 1, j = 2; k < kFftLengthBy2; ++k) {
+      (*v)[j++] = re[k];
+      (*v)[j++] = im[k];
+    }
+  }
+
+  std::array<float, kFftLengthBy2Plus1> re;
+  std::array<float, kFftLengthBy2Plus1> im;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_FFT_DATA_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_avx2.cc b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_avx2.cc
new file mode 100644
index 0000000000..1fe4bd69c6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_avx2.cc
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/fft_data.h"
+
+#include <immintrin.h>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Computes the power spectrum of the data.
+void FftData::SpectrumAVX2(rtc::ArrayView<float> power_spectrum) const {
+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, power_spectrum.size());
+  for (size_t k = 0; k < kFftLengthBy2; k += 8) {
+    __m256 r = _mm256_loadu_ps(&re[k]);
+    __m256 i = _mm256_loadu_ps(&im[k]);
+    __m256 ii = _mm256_mul_ps(i, i);
+    ii = _mm256_fmadd_ps(r, r, ii);
+    _mm256_storeu_ps(&power_spectrum[k], ii);
+  }
+  power_spectrum[kFftLengthBy2] = re[kFftLengthBy2] * re[kFftLengthBy2] +
+                                  im[kFftLengthBy2] * im[kFftLengthBy2];
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_gn/moz.build
new file mode 100644
index 0000000000..d77163999b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("fft_data_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_unittest.cc
new file mode 100644
index 0000000000..d76fabdbd6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/fft_data_unittest.cc
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/fft_data.h"
+
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Verifies that the optimized methods are bitexact to their reference
+// counterparts.
+TEST(FftData, TestSse2Optimizations) {
+  if (GetCPUInfo(kSSE2) != 0) {
+    FftData x;
+
+    for (size_t k = 0; k < x.re.size(); ++k) {
+      x.re[k] = k + 1;
+    }
+
+    x.im[0] = x.im[x.im.size() - 1] = 0.f;
+    for (size_t k = 1; k < x.im.size() - 1; ++k) {
+      x.im[k] = 2.f * (k + 1);
+    }
+
+    std::array<float, kFftLengthBy2Plus1> spectrum;
+    std::array<float, kFftLengthBy2Plus1> spectrum_sse2;
+    x.Spectrum(Aec3Optimization::kNone, spectrum);
+    x.Spectrum(Aec3Optimization::kSse2, spectrum_sse2);
+    EXPECT_EQ(spectrum, spectrum_sse2);
+  }
+}
+
+// Verifies that the optimized methods are bitexact to their reference
+// counterparts.
+TEST(FftData, TestAvx2Optimizations) {
+  if (GetCPUInfo(kAVX2) != 0) {
+    FftData x;
+
+    for (size_t k = 0; k < x.re.size(); ++k) {
+      x.re[k] = k + 1;
+    }
+
+    x.im[0] = x.im[x.im.size() - 1] = 0.f;
+    for (size_t k = 1; k < x.im.size() - 1; ++k) {
+      x.im[k] = 2.f * (k + 1);
+    }
+
+    std::array<float, kFftLengthBy2Plus1> spectrum;
+    std::array<float, kFftLengthBy2Plus1> spectrum_avx2;
+    x.Spectrum(Aec3Optimization::kNone, spectrum);
+    x.Spectrum(Aec3Optimization::kAvx2, spectrum_avx2);
+    EXPECT_EQ(spectrum, spectrum_avx2);
+  }
+}
+#endif
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for null output in CopyToPackedArray.
+TEST(FftDataDeathTest, NonNullCopyToPackedArrayOutput) {
+  EXPECT_DEATH(FftData().CopyToPackedArray(nullptr), "");
+}
+
+// Verifies the check for null output in Spectrum.
+TEST(FftDataDeathTest, NonNullSpectrumOutput) {
+  EXPECT_DEATH(FftData().Spectrum(Aec3Optimization::kNone, nullptr), "");
+}
+
+#endif
+
+// Verifies that the Assign method properly copies the data from the source and
+// ensures that the imaginary components for the DC and Nyquist bins are 0.
+TEST(FftData, Assign) {
+  FftData x;
+  FftData y;
+
+  x.re.fill(1.f);
+  x.im.fill(2.f);
+  y.Assign(x);
+  EXPECT_EQ(x.re, y.re);
+  EXPECT_EQ(0.f, y.im[0]);
+  EXPECT_EQ(0.f, y.im[x.im.size() - 1]);
+  for (size_t k = 1; k < x.im.size() - 1; ++k) {
+    EXPECT_EQ(x.im[k], y.im[k]);
+  }
+}
+
+// Verifies that the Clear method properly clears all the data.
+TEST(FftData, Clear) {
+  FftData x_ref;
+  FftData x;
+
+  x_ref.re.fill(0.f);
+  x_ref.im.fill(0.f);
+
+  x.re.fill(1.f);
+  x.im.fill(2.f);
+  x.Clear();
+
+  EXPECT_EQ(x_ref.re, x.re);
+  EXPECT_EQ(x_ref.im, x.im);
+}
+
+// Verifies that the spectrum is correctly computed.
+TEST(FftData, Spectrum) {
+  FftData x;
+
+  for (size_t k = 0; k < x.re.size(); ++k) {
+    x.re[k] = k + 1;
+  }
+
+  x.im[0] = x.im[x.im.size() - 1] = 0.f;
+  for (size_t k = 1; k < x.im.size() - 1; ++k) {
+    x.im[k] = 2.f * (k + 1);
+  }
+
+  std::array<float, kFftLengthBy2Plus1> spectrum;
+  x.Spectrum(Aec3Optimization::kNone, spectrum);
+
+  EXPECT_EQ(x.re[0] * x.re[0], spectrum[0]);
+  EXPECT_EQ(x.re[spectrum.size() - 1] * x.re[spectrum.size() - 1],
+            spectrum[spectrum.size() - 1]);
+  for (size_t k = 1; k < spectrum.size() - 1; ++k) {
+    EXPECT_EQ(x.re[k] * x.re[k] + x.im[k] * x.im[k], spectrum[k]);
+  }
+}
+
+// Verifies that the functionality in CopyToPackedArray works as intended.
+TEST(FftData, CopyToPackedArray) {
+  FftData x;
+  std::array<float, kFftLength> x_packed;
+
+  for (size_t k = 0; k < x.re.size(); ++k) {
+    x.re[k] = k + 1;
+  }
+
+  x.im[0] = x.im[x.im.size() - 1] = 0.f;
+  for (size_t k = 1; k < x.im.size() - 1; ++k) {
+    x.im[k] = 2.f * (k + 1);
+  }
+
+  x.CopyToPackedArray(&x_packed);
+
+  EXPECT_EQ(x.re[0], x_packed[0]);
+  EXPECT_EQ(x.re[x.re.size() - 1], x_packed[1]);
+  for (size_t k = 1; k < x_packed.size() / 2; ++k) {
+    EXPECT_EQ(x.re[k], x_packed[2 * k]);
+    EXPECT_EQ(x.im[k], x_packed[2 * k + 1]);
+  }
+}
+
+// Verifies that the functionality in CopyFromPackedArray works as intended
+// (relies on that the functionality in CopyToPackedArray has been verified in
+// the test above).
+TEST(FftData, CopyFromPackedArray) {
+  FftData x_ref;
+  FftData x;
+  std::array<float, kFftLength> x_packed;
+
+  for (size_t k = 0; k < x_ref.re.size(); ++k) {
+    x_ref.re[k] = k + 1;
+  }
+
+  x_ref.im[0] = x_ref.im[x_ref.im.size() - 1] = 0.f;
+  for (size_t k = 1; k < x_ref.im.size() - 1; ++k) {
+    x_ref.im[k] = 2.f * (k + 1);
+  }
+
+  x_ref.CopyToPackedArray(&x_packed);
+  x.CopyFromPackedArray(x_packed);
+
+  EXPECT_EQ(x_ref.re, x.re);
+  EXPECT_EQ(x_ref.im, x.im);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer.cc
new file mode 100644
index 0000000000..d8fd3aa275
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer.cc
@@ -0,0 +1,289 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/filter_analyzer.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <array>
+#include <numeric>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+size_t FindPeakIndex(rtc::ArrayView<const float> filter_time_domain,
+                     size_t peak_index_in,
+                     size_t start_sample,
+                     size_t end_sample) {
+  size_t peak_index_out = peak_index_in;
+  float max_h2 =
+      filter_time_domain[peak_index_out] * filter_time_domain[peak_index_out];
+  for (size_t k = start_sample; k <= end_sample; ++k) {
+    float tmp = filter_time_domain[k] * filter_time_domain[k];
+    if (tmp > max_h2) {
+      peak_index_out = k;
+      max_h2 = tmp;
+    }
+  }
+
+  return peak_index_out;
+}
+
+}  // namespace
+
+std::atomic<int> FilterAnalyzer::instance_count_(0);
+
+FilterAnalyzer::FilterAnalyzer(const EchoCanceller3Config& config,
+                               size_t num_capture_channels)
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      bounded_erl_(config.ep_strength.bounded_erl),
+      default_gain_(config.ep_strength.default_gain),
+      h_highpass_(num_capture_channels,
+                  std::vector<float>(
+                      GetTimeDomainLength(config.filter.refined.length_blocks),
+                      0.f)),
+      filter_analysis_states_(num_capture_channels,
+                              FilterAnalysisState(config)),
+      filter_delays_blocks_(num_capture_channels, 0) {
+  Reset();
+}
+
+FilterAnalyzer::~FilterAnalyzer() = default;
+
+void FilterAnalyzer::Reset() {
+  blocks_since_reset_ = 0;
+  ResetRegion();
+  for (auto& state : filter_analysis_states_) {
+    state.Reset(default_gain_);
+  }
+  std::fill(filter_delays_blocks_.begin(), filter_delays_blocks_.end(), 0);
+}
+
+void FilterAnalyzer::Update(
+    rtc::ArrayView<const std::vector<float>> filters_time_domain,
+    const RenderBuffer& render_buffer,
+    bool* any_filter_consistent,
+    float* max_echo_path_gain) {
+  RTC_DCHECK(any_filter_consistent);
+  RTC_DCHECK(max_echo_path_gain);
+  RTC_DCHECK_EQ(filters_time_domain.size(), filter_analysis_states_.size());
+  RTC_DCHECK_EQ(filters_time_domain.size(), h_highpass_.size());
+
+  ++blocks_since_reset_;
+  SetRegionToAnalyze(filters_time_domain[0].size());
+  AnalyzeRegion(filters_time_domain, render_buffer);
+
+  // Aggregate the results for all capture channels.
+  auto& st_ch0 = filter_analysis_states_[0];
+  *any_filter_consistent = st_ch0.consistent_estimate;
+  *max_echo_path_gain = st_ch0.gain;
+  min_filter_delay_blocks_ = filter_delays_blocks_[0];
+  for (size_t ch = 1; ch < filters_time_domain.size(); ++ch) {
+    auto& st_ch = filter_analysis_states_[ch];
+    *any_filter_consistent =
+        *any_filter_consistent || st_ch.consistent_estimate;
+    *max_echo_path_gain = std::max(*max_echo_path_gain, st_ch.gain);
+    min_filter_delay_blocks_ =
+        std::min(min_filter_delay_blocks_, filter_delays_blocks_[ch]);
+  }
+}
+
+void FilterAnalyzer::AnalyzeRegion(
+    rtc::ArrayView<const std::vector<float>> filters_time_domain,
+    const RenderBuffer& render_buffer) {
+  // Preprocess the filter to avoid issues with low-frequency components in the
+  // filter.
+  PreProcessFilters(filters_time_domain);
+  data_dumper_->DumpRaw("aec3_linear_filter_processed_td", h_highpass_[0]);
+
+  constexpr float kOneByBlockSize = 1.f / kBlockSize;
+  for (size_t ch = 0; ch < filters_time_domain.size(); ++ch) {
+    RTC_DCHECK_LT(region_.start_sample_, filters_time_domain[ch].size());
+    RTC_DCHECK_LT(region_.end_sample_, filters_time_domain[ch].size());
+
+    auto& st_ch = filter_analysis_states_[ch];
+    RTC_DCHECK_EQ(h_highpass_[ch].size(), filters_time_domain[ch].size());
+    RTC_DCHECK_GT(h_highpass_[ch].size(), 0);
+    st_ch.peak_index = std::min(st_ch.peak_index, h_highpass_[ch].size() - 1);
+
+    st_ch.peak_index =
+        FindPeakIndex(h_highpass_[ch], st_ch.peak_index, region_.start_sample_,
+                      region_.end_sample_);
+    filter_delays_blocks_[ch] = st_ch.peak_index >> kBlockSizeLog2;
+    UpdateFilterGain(h_highpass_[ch], &st_ch);
+    st_ch.filter_length_blocks =
+        filters_time_domain[ch].size() * kOneByBlockSize;
+
+    st_ch.consistent_estimate = st_ch.consistent_filter_detector.Detect(
+        h_highpass_[ch], region_,
+        render_buffer.GetBlock(-filter_delays_blocks_[ch]), st_ch.peak_index,
+        filter_delays_blocks_[ch]);
+  }
+}
+
+void FilterAnalyzer::UpdateFilterGain(
+    rtc::ArrayView<const float> filter_time_domain,
+    FilterAnalysisState* st) {
+  bool sufficient_time_to_converge =
+      blocks_since_reset_ > 5 * kNumBlocksPerSecond;
+
+  if (sufficient_time_to_converge && st->consistent_estimate) {
+    st->gain = fabsf(filter_time_domain[st->peak_index]);
+  } else {
+    // TODO(peah): Verify whether this check against a float is ok.
+    if (st->gain) {
+      st->gain = std::max(st->gain, fabsf(filter_time_domain[st->peak_index]));
+    }
+  }
+
+  if (bounded_erl_ && st->gain) {
+    st->gain = std::max(st->gain, 0.01f);
+  }
+}
+
+void FilterAnalyzer::PreProcessFilters(
+    rtc::ArrayView<const std::vector<float>> filters_time_domain) {
+  for (size_t ch = 0; ch < filters_time_domain.size(); ++ch) {
+    RTC_DCHECK_LT(region_.start_sample_, filters_time_domain[ch].size());
+    RTC_DCHECK_LT(region_.end_sample_, filters_time_domain[ch].size());
+
+    RTC_DCHECK_GE(h_highpass_[ch].capacity(), filters_time_domain[ch].size());
+    h_highpass_[ch].resize(filters_time_domain[ch].size());
+    // Minimum phase high-pass filter with cutoff frequency at about 600 Hz.
+    constexpr std::array<float, 3> h = {
+        {0.7929742f, -0.36072128f, -0.47047766f}};
+
+    std::fill(h_highpass_[ch].begin() + region_.start_sample_,
+              h_highpass_[ch].begin() + region_.end_sample_ + 1, 0.f);
+    float* h_highpass_ch = h_highpass_[ch].data();
+    const float* filters_time_domain_ch = filters_time_domain[ch].data();
+    const size_t region_end = region_.end_sample_;
+    for (size_t k = std::max(h.size() - 1, region_.start_sample_);
+         k <= region_end; ++k) {
+      float tmp = h_highpass_ch[k];
+      for (size_t j = 0; j < h.size(); ++j) {
+        tmp += filters_time_domain_ch[k - j] * h[j];
+      }
+      h_highpass_ch[k] = tmp;
+    }
+  }
+}
+
+void FilterAnalyzer::ResetRegion() {
+  region_.start_sample_ = 0;
+  region_.end_sample_ = 0;
+}
+
+void FilterAnalyzer::SetRegionToAnalyze(size_t filter_size) {
+  constexpr size_t kNumberBlocksToUpdate = 1;
+  auto& r = region_;
+  r.start_sample_ = r.end_sample_ >= filter_size - 1 ? 0 : r.end_sample_ + 1;
+  r.end_sample_ =
+      std::min(r.start_sample_ + kNumberBlocksToUpdate * kBlockSize - 1,
+               filter_size - 1);
+
+  // Check range.
+  RTC_DCHECK_LT(r.start_sample_, filter_size);
+  RTC_DCHECK_LT(r.end_sample_, filter_size);
+  RTC_DCHECK_LE(r.start_sample_, r.end_sample_);
+}
+
+FilterAnalyzer::ConsistentFilterDetector::ConsistentFilterDetector(
+    const EchoCanceller3Config& config)
+    : active_render_threshold_(config.render_levels.active_render_limit *
+                               config.render_levels.active_render_limit *
+                               kFftLengthBy2) {
+  Reset();
+}
+
+void FilterAnalyzer::ConsistentFilterDetector::Reset() {
+  significant_peak_ = false;
+  filter_floor_accum_ = 0.f;
+  filter_secondary_peak_ = 0.f;
+  filter_floor_low_limit_ = 0;
+  filter_floor_high_limit_ = 0;
+  consistent_estimate_counter_ = 0;
+  consistent_delay_reference_ = -10;
+}
+
+bool FilterAnalyzer::ConsistentFilterDetector::Detect(
+    rtc::ArrayView<const float> filter_to_analyze,
+    const FilterRegion& region,
+    const Block& x_block,
+    size_t peak_index,
+    int delay_blocks) {
+  if (region.start_sample_ == 0) {
+    filter_floor_accum_ = 0.f;
+    filter_secondary_peak_ = 0.f;
+    filter_floor_low_limit_ = peak_index < 64 ? 0 : peak_index - 64;
+    filter_floor_high_limit_ =
+        peak_index > filter_to_analyze.size() - 129 ? 0 : peak_index + 128;
+  }
+
+  float filter_floor_accum = filter_floor_accum_;
+  float filter_secondary_peak = filter_secondary_peak_;
+  for (size_t k = region.start_sample_;
+       k < std::min(region.end_sample_ + 1, filter_floor_low_limit_); ++k) {
+    float abs_h = fabsf(filter_to_analyze[k]);
+    filter_floor_accum += abs_h;
+    filter_secondary_peak = std::max(filter_secondary_peak, abs_h);
+  }
+
+  for (size_t k = std::max(filter_floor_high_limit_, region.start_sample_);
+       k <= region.end_sample_; ++k) {
+    float abs_h = fabsf(filter_to_analyze[k]);
+    filter_floor_accum += abs_h;
+    filter_secondary_peak = std::max(filter_secondary_peak, abs_h);
+  }
+  filter_floor_accum_ = filter_floor_accum;
+  filter_secondary_peak_ = filter_secondary_peak;
+
+  if (region.end_sample_ == filter_to_analyze.size() - 1) {
+    float filter_floor = filter_floor_accum_ /
+                         (filter_floor_low_limit_ + filter_to_analyze.size() -
+                          filter_floor_high_limit_);
+
+    float abs_peak = fabsf(filter_to_analyze[peak_index]);
+    significant_peak_ = abs_peak > 10.f * filter_floor &&
+                        abs_peak > 2.f * filter_secondary_peak_;
+  }
+
+  if (significant_peak_) {
+    bool active_render_block = false;
+    for (int ch = 0; ch < x_block.NumChannels(); ++ch) {
+      rtc::ArrayView<const float, kBlockSize> x_channel =
+          x_block.View(/*band=*/0, ch);
+      const float x_energy = std::inner_product(
+          x_channel.begin(), x_channel.end(), x_channel.begin(), 0.f);
+      if (x_energy > active_render_threshold_) {
+        active_render_block = true;
+        break;
+      }
+    }
+
+    if (consistent_delay_reference_ == delay_blocks) {
+      if (active_render_block) {
+        ++consistent_estimate_counter_;
+      }
+    } else {
+      consistent_estimate_counter_ = 0;
+      consistent_delay_reference_ = delay_blocks;
+    }
+  }
+  return consistent_estimate_counter_ > 1.5f * kNumBlocksPerSecond;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer.h b/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer.h
new file mode 100644
index 0000000000..9aec8b14d7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer.h
@@ -0,0 +1,150 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_FILTER_ANALYZER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_FILTER_ANALYZER_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/block.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class RenderBuffer;
+
+// Class for analyzing the properties of an adaptive filter.
+class FilterAnalyzer {
+ public:
+  FilterAnalyzer(const EchoCanceller3Config& config,
+                 size_t num_capture_channels);
+  ~FilterAnalyzer();
+
+  FilterAnalyzer(const FilterAnalyzer&) = delete;
+  FilterAnalyzer& operator=(const FilterAnalyzer&) = delete;
+
+  // Resets the analysis.
+  void Reset();
+
+  // Updates the estimates with new input data.
+  void Update(rtc::ArrayView<const std::vector<float>> filters_time_domain,
+              const RenderBuffer& render_buffer,
+              bool* any_filter_consistent,
+              float* max_echo_path_gain);
+
+  // Returns the delay in blocks for each filter.
+  rtc::ArrayView<const int> FilterDelaysBlocks() const {
+    return filter_delays_blocks_;
+  }
+
+  // Returns the minimum delay of all filters in terms of blocks.
+  int MinFilterDelayBlocks() const { return min_filter_delay_blocks_; }
+
+  // Returns the number of blocks for the current used filter.
+  int FilterLengthBlocks() const {
+    return filter_analysis_states_[0].filter_length_blocks;
+  }
+
+  // Returns the preprocessed filter.
+  rtc::ArrayView<const std::vector<float>> GetAdjustedFilters() const {
+    return h_highpass_;
+  }
+
+  // Public for testing purposes only.
+  void SetRegionToAnalyze(size_t filter_size);
+
+ private:
+  struct FilterAnalysisState;
+
+  void AnalyzeRegion(
+      rtc::ArrayView<const std::vector<float>> filters_time_domain,
+      const RenderBuffer& render_buffer);
+
+  void UpdateFilterGain(rtc::ArrayView<const float> filters_time_domain,
+                        FilterAnalysisState* st);
+  void PreProcessFilters(
+      rtc::ArrayView<const std::vector<float>> filters_time_domain);
+
+  void ResetRegion();
+
+  struct FilterRegion {
+    size_t start_sample_;
+    size_t end_sample_;
+  };
+
+  // This class checks whether the shape of the impulse response has been
+  // consistent over time.
+  class ConsistentFilterDetector {
+   public:
+    explicit ConsistentFilterDetector(const EchoCanceller3Config& config);
+    void Reset();
+    bool Detect(rtc::ArrayView<const float> filter_to_analyze,
+                const FilterRegion& region,
+                const Block& x_block,
+                size_t peak_index,
+                int delay_blocks);
+
+   private:
+    bool significant_peak_;
+    float filter_floor_accum_;
+    float filter_secondary_peak_;
+    size_t filter_floor_low_limit_;
+    size_t filter_floor_high_limit_;
+    const float active_render_threshold_;
+    size_t consistent_estimate_counter_ = 0;
+    int consistent_delay_reference_ = -10;
+  };
+
+  struct FilterAnalysisState {
+    explicit FilterAnalysisState(const EchoCanceller3Config& config)
+        : filter_length_blocks(config.filter.refined_initial.length_blocks),
+          consistent_filter_detector(config) {
+      Reset(config.ep_strength.default_gain);
+    }
+
+    void Reset(float default_gain) {
+      peak_index = 0;
+      gain = default_gain;
+      consistent_filter_detector.Reset();
+    }
+
+    float gain;
+    size_t peak_index;
+    int filter_length_blocks;
+    bool consistent_estimate = false;
+    ConsistentFilterDetector consistent_filter_detector;
+  };
+
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  const bool bounded_erl_;
+  const float default_gain_;
+  std::vector<std::vector<float>> h_highpass_;
+
+  size_t blocks_since_reset_ = 0;
+  FilterRegion region_;
+
+  std::vector<FilterAnalysisState> filter_analysis_states_;
+  std::vector<int> filter_delays_blocks_;
+
+  int min_filter_delay_blocks_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_FILTER_ANALYZER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer_unittest.cc
new file mode 100644
index 0000000000..f1e2e4c188
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/filter_analyzer_unittest.cc
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/filter_analyzer.h"
+
+#include <algorithm>
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Verifies that the filter analyzer handles filter resizes properly.
+TEST(FilterAnalyzer, FilterResize) {
+  EchoCanceller3Config c;
+  std::vector<float> filter(65, 0.f);
+  for (size_t num_capture_channels : {1, 2, 4}) {
+    FilterAnalyzer fa(c, num_capture_channels);
+    fa.SetRegionToAnalyze(filter.size());
+    fa.SetRegionToAnalyze(filter.size());
+    filter.resize(32);
+    fa.SetRegionToAnalyze(filter.size());
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker.cc b/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker.cc
new file mode 100644
index 0000000000..3039dcf7f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker.cc
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/frame_blocker.h"
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+FrameBlocker::FrameBlocker(size_t num_bands, size_t num_channels)
+    : num_bands_(num_bands),
+      num_channels_(num_channels),
+      buffer_(num_bands_, std::vector<std::vector<float>>(num_channels)) {
+  RTC_DCHECK_LT(0, num_bands);
+  RTC_DCHECK_LT(0, num_channels);
+  for (auto& band : buffer_) {
+    for (auto& channel : band) {
+      channel.reserve(kBlockSize);
+      RTC_DCHECK(channel.empty());
+    }
+  }
+}
+
+FrameBlocker::~FrameBlocker() = default;
+
+void FrameBlocker::InsertSubFrameAndExtractBlock(
+    const std::vector<std::vector<rtc::ArrayView<float>>>& sub_frame,
+    Block* block) {
+  RTC_DCHECK(block);
+  RTC_DCHECK_EQ(num_bands_, block->NumBands());
+  RTC_DCHECK_EQ(num_bands_, sub_frame.size());
+  for (size_t band = 0; band < num_bands_; ++band) {
+    RTC_DCHECK_EQ(num_channels_, block->NumChannels());
+    RTC_DCHECK_EQ(num_channels_, sub_frame[band].size());
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      RTC_DCHECK_GE(kBlockSize - 16, buffer_[band][channel].size());
+      RTC_DCHECK_EQ(kSubFrameLength, sub_frame[band][channel].size());
+      const int samples_to_block = kBlockSize - buffer_[band][channel].size();
+      std::copy(buffer_[band][channel].begin(), buffer_[band][channel].end(),
+                block->begin(band, channel));
+      std::copy(sub_frame[band][channel].begin(),
+                sub_frame[band][channel].begin() + samples_to_block,
+                block->begin(band, channel) + kBlockSize - samples_to_block);
+      buffer_[band][channel].clear();
+      buffer_[band][channel].insert(
+          buffer_[band][channel].begin(),
+          sub_frame[band][channel].begin() + samples_to_block,
+          sub_frame[band][channel].end());
+    }
+  }
+}
+
+bool FrameBlocker::IsBlockAvailable() const {
+  return kBlockSize == buffer_[0][0].size();
+}
+
+void FrameBlocker::ExtractBlock(Block* block) {
+  RTC_DCHECK(block);
+  RTC_DCHECK_EQ(num_bands_, block->NumBands());
+  RTC_DCHECK_EQ(num_channels_, block->NumChannels());
+  RTC_DCHECK(IsBlockAvailable());
+  for (size_t band = 0; band < num_bands_; ++band) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      RTC_DCHECK_EQ(kBlockSize, buffer_[band][channel].size());
+      std::copy(buffer_[band][channel].begin(), buffer_[band][channel].end(),
+                block->begin(band, channel));
+      buffer_[band][channel].clear();
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker.h b/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker.h
new file mode 100644
index 0000000000..623c812157
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_FRAME_BLOCKER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_FRAME_BLOCKER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/block.h"
+
+namespace webrtc {
+
+// Class for producing 64 sample multiband blocks from frames consisting of 2
+// subframes of 80 samples.
+class FrameBlocker {
+ public:
+  FrameBlocker(size_t num_bands, size_t num_channels);
+  ~FrameBlocker();
+  FrameBlocker(const FrameBlocker&) = delete;
+  FrameBlocker& operator=(const FrameBlocker&) = delete;
+
+  // Inserts one 80 sample multiband subframe from the multiband frame and
+  // extracts one 64 sample multiband block.
+  void InsertSubFrameAndExtractBlock(
+      const std::vector<std::vector<rtc::ArrayView<float>>>& sub_frame,
+      Block* block);
+  // Reports whether a multiband block of 64 samples is available for
+  // extraction.
+  bool IsBlockAvailable() const;
+  // Extracts a multiband block of 64 samples.
+  void ExtractBlock(Block* block);
+
+ private:
+  const size_t num_bands_;
+  const size_t num_channels_;
+  std::vector<std::vector<std::vector<float>>> buffer_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_FRAME_BLOCKER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker_unittest.cc
new file mode 100644
index 0000000000..92e393023a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/frame_blocker_unittest.cc
@@ -0,0 +1,425 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/frame_blocker.h"
+
+#include <string>
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/block_framer.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+float ComputeSampleValue(size_t chunk_counter,
+                         size_t chunk_size,
+                         size_t band,
+                         size_t channel,
+                         size_t sample_index,
+                         int offset) {
+  float value =
+      static_cast<int>(chunk_counter * chunk_size + sample_index + channel) +
+      offset;
+  return value > 0 ? 5000 * band + value : 0;
+}
+
+void FillSubFrame(size_t sub_frame_counter,
+                  int offset,
+                  std::vector<std::vector<std::vector<float>>>* sub_frame) {
+  for (size_t band = 0; band < sub_frame->size(); ++band) {
+    for (size_t channel = 0; channel < (*sub_frame)[band].size(); ++channel) {
+      for (size_t sample = 0; sample < (*sub_frame)[band][channel].size();
+           ++sample) {
+        (*sub_frame)[band][channel][sample] = ComputeSampleValue(
+            sub_frame_counter, kSubFrameLength, band, channel, sample, offset);
+      }
+    }
+  }
+}
+
+void FillSubFrameView(
+    size_t sub_frame_counter,
+    int offset,
+    std::vector<std::vector<std::vector<float>>>* sub_frame,
+    std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
+  FillSubFrame(sub_frame_counter, offset, sub_frame);
+  for (size_t band = 0; band < sub_frame_view->size(); ++band) {
+    for (size_t channel = 0; channel < (*sub_frame_view)[band].size();
+         ++channel) {
+      (*sub_frame_view)[band][channel] = rtc::ArrayView<float>(
+          &(*sub_frame)[band][channel][0], (*sub_frame)[band][channel].size());
+    }
+  }
+}
+
+bool VerifySubFrame(
+    size_t sub_frame_counter,
+    int offset,
+    const std::vector<std::vector<rtc::ArrayView<float>>>& sub_frame_view) {
+  std::vector<std::vector<std::vector<float>>> reference_sub_frame(
+      sub_frame_view.size(),
+      std::vector<std::vector<float>>(
+          sub_frame_view[0].size(),
+          std::vector<float>(sub_frame_view[0][0].size(), 0.f)));
+  FillSubFrame(sub_frame_counter, offset, &reference_sub_frame);
+  for (size_t band = 0; band < sub_frame_view.size(); ++band) {
+    for (size_t channel = 0; channel < sub_frame_view[band].size(); ++channel) {
+      for (size_t sample = 0; sample < sub_frame_view[band][channel].size();
+           ++sample) {
+        if (reference_sub_frame[band][channel][sample] !=
+            sub_frame_view[band][channel][sample]) {
+          return false;
+        }
+      }
+    }
+  }
+  return true;
+}
+
+bool VerifyBlock(size_t block_counter, int offset, const Block& block) {
+  for (int band = 0; band < block.NumBands(); ++band) {
+    for (int channel = 0; channel < block.NumChannels(); ++channel) {
+      for (size_t sample = 0; sample < kBlockSize; ++sample) {
+        auto it = block.begin(band, channel) + sample;
+        const float reference_value = ComputeSampleValue(
+            block_counter, kBlockSize, band, channel, sample, offset);
+        if (reference_value != *it) {
+          return false;
+        }
+      }
+    }
+  }
+  return true;
+}
+
+// Verifies that the FrameBlocker properly forms blocks out of the frames.
+void RunBlockerTest(int sample_rate_hz, size_t num_channels) {
+  constexpr size_t kNumSubFramesToProcess = 20;
+  const size_t num_bands = NumBandsForRate(sample_rate_hz);
+
+  Block block(num_bands, num_channels);
+  std::vector<std::vector<std::vector<float>>> input_sub_frame(
+      num_bands, std::vector<std::vector<float>>(
+                     num_channels, std::vector<float>(kSubFrameLength, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> input_sub_frame_view(
+      num_bands, std::vector<rtc::ArrayView<float>>(num_channels));
+  FrameBlocker blocker(num_bands, num_channels);
+
+  size_t block_counter = 0;
+  for (size_t sub_frame_index = 0; sub_frame_index < kNumSubFramesToProcess;
+       ++sub_frame_index) {
+    FillSubFrameView(sub_frame_index, 0, &input_sub_frame,
+                     &input_sub_frame_view);
+
+    blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &block);
+    VerifyBlock(block_counter++, 0, block);
+
+    if ((sub_frame_index + 1) % 4 == 0) {
+      EXPECT_TRUE(blocker.IsBlockAvailable());
+    } else {
+      EXPECT_FALSE(blocker.IsBlockAvailable());
+    }
+    if (blocker.IsBlockAvailable()) {
+      blocker.ExtractBlock(&block);
+      VerifyBlock(block_counter++, 0, block);
+    }
+  }
+}
+
+// Verifies that the FrameBlocker and BlockFramer work well together and produce
+// the expected output.
+void RunBlockerAndFramerTest(int sample_rate_hz, size_t num_channels) {
+  const size_t kNumSubFramesToProcess = 20;
+  const size_t num_bands = NumBandsForRate(sample_rate_hz);
+
+  Block block(num_bands, num_channels);
+  std::vector<std::vector<std::vector<float>>> input_sub_frame(
+      num_bands, std::vector<std::vector<float>>(
+                     num_channels, std::vector<float>(kSubFrameLength, 0.f)));
+  std::vector<std::vector<std::vector<float>>> output_sub_frame(
+      num_bands, std::vector<std::vector<float>>(
+                     num_channels, std::vector<float>(kSubFrameLength, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> output_sub_frame_view(
+      num_bands, std::vector<rtc::ArrayView<float>>(num_channels));
+  std::vector<std::vector<rtc::ArrayView<float>>> input_sub_frame_view(
+      num_bands, std::vector<rtc::ArrayView<float>>(num_channels));
+  FrameBlocker blocker(num_bands, num_channels);
+  BlockFramer framer(num_bands, num_channels);
+
+  for (size_t sub_frame_index = 0; sub_frame_index < kNumSubFramesToProcess;
+       ++sub_frame_index) {
+    FillSubFrameView(sub_frame_index, 0, &input_sub_frame,
+                     &input_sub_frame_view);
+    FillSubFrameView(sub_frame_index, 0, &output_sub_frame,
+                     &output_sub_frame_view);
+
+    blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &block);
+    framer.InsertBlockAndExtractSubFrame(block, &output_sub_frame_view);
+
+    if ((sub_frame_index + 1) % 4 == 0) {
+      EXPECT_TRUE(blocker.IsBlockAvailable());
+    } else {
+      EXPECT_FALSE(blocker.IsBlockAvailable());
+    }
+    if (blocker.IsBlockAvailable()) {
+      blocker.ExtractBlock(&block);
+      framer.InsertBlock(block);
+    }
+    if (sub_frame_index > 1) {
+      EXPECT_TRUE(VerifySubFrame(sub_frame_index, -64, output_sub_frame_view));
+    }
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Verifies that the FrameBlocker crashes if the InsertSubFrameAndExtractBlock
+// method is called for inputs with the wrong number of bands or band lengths.
+void RunWronglySizedInsertAndExtractParametersTest(
+    int sample_rate_hz,
+    size_t correct_num_channels,
+    size_t num_block_bands,
+    size_t num_block_channels,
+    size_t num_sub_frame_bands,
+    size_t num_sub_frame_channels,
+    size_t sub_frame_length) {
+  const size_t correct_num_bands = NumBandsForRate(sample_rate_hz);
+
+  Block block(num_block_bands, num_block_channels);
+  std::vector<std::vector<std::vector<float>>> input_sub_frame(
+      num_sub_frame_bands,
+      std::vector<std::vector<float>>(
+          num_sub_frame_channels, std::vector<float>(sub_frame_length, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> input_sub_frame_view(
+      input_sub_frame.size(),
+      std::vector<rtc::ArrayView<float>>(num_sub_frame_channels));
+  FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view);
+  FrameBlocker blocker(correct_num_bands, correct_num_channels);
+  EXPECT_DEATH(
+      blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &block), "");
+}
+
+// Verifies that the FrameBlocker crashes if the ExtractBlock method is called
+// for inputs with the wrong number of bands or band lengths.
+void RunWronglySizedExtractParameterTest(int sample_rate_hz,
+                                         size_t correct_num_channels,
+                                         size_t num_block_bands,
+                                         size_t num_block_channels) {
+  const size_t correct_num_bands = NumBandsForRate(sample_rate_hz);
+
+  Block correct_block(correct_num_bands, correct_num_channels);
+  Block wrong_block(num_block_bands, num_block_channels);
+  std::vector<std::vector<std::vector<float>>> input_sub_frame(
+      correct_num_bands,
+      std::vector<std::vector<float>>(
+          correct_num_channels, std::vector<float>(kSubFrameLength, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> input_sub_frame_view(
+      input_sub_frame.size(),
+      std::vector<rtc::ArrayView<float>>(correct_num_channels));
+  FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view);
+  FrameBlocker blocker(correct_num_bands, correct_num_channels);
+  blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &correct_block);
+  blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &correct_block);
+  blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &correct_block);
+  blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &correct_block);
+
+  EXPECT_DEATH(blocker.ExtractBlock(&wrong_block), "");
+}
+
+// Verifies that the FrameBlocker crashes if the ExtractBlock method is called
+// after a wrong number of previous InsertSubFrameAndExtractBlock method calls
+// have been made.
+void RunWrongExtractOrderTest(int sample_rate_hz,
+                              size_t num_channels,
+                              size_t num_preceeding_api_calls) {
+  const size_t num_bands = NumBandsForRate(sample_rate_hz);
+
+  Block block(num_bands, num_channels);
+  std::vector<std::vector<std::vector<float>>> input_sub_frame(
+      num_bands, std::vector<std::vector<float>>(
+                     num_channels, std::vector<float>(kSubFrameLength, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> input_sub_frame_view(
+      input_sub_frame.size(), std::vector<rtc::ArrayView<float>>(num_channels));
+  FillSubFrameView(0, 0, &input_sub_frame, &input_sub_frame_view);
+  FrameBlocker blocker(num_bands, num_channels);
+  for (size_t k = 0; k < num_preceeding_api_calls; ++k) {
+    blocker.InsertSubFrameAndExtractBlock(input_sub_frame_view, &block);
+  }
+
+  EXPECT_DEATH(blocker.ExtractBlock(&block), "");
+}
+#endif
+
+std::string ProduceDebugText(int sample_rate_hz, size_t num_channels) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz;
+  ss << ", number of channels: " << num_channels;
+  return ss.Release();
+}
+
+}  // namespace
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(FrameBlockerDeathTest,
+     WrongNumberOfBandsInBlockForInsertSubFrameAndExtractBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t correct_num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_bands = (correct_num_bands % 3) + 1;
+      RunWronglySizedInsertAndExtractParametersTest(
+          rate, correct_num_channels, wrong_num_bands, correct_num_channels,
+          correct_num_bands, correct_num_channels, kSubFrameLength);
+    }
+  }
+}
+
+TEST(FrameBlockerDeathTest,
+     WrongNumberOfChannelsInBlockForInsertSubFrameAndExtractBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t correct_num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_channels = correct_num_channels + 1;
+      RunWronglySizedInsertAndExtractParametersTest(
+          rate, correct_num_channels, correct_num_bands, wrong_num_channels,
+          correct_num_bands, correct_num_channels, kSubFrameLength);
+    }
+  }
+}
+
+TEST(FrameBlockerDeathTest,
+     WrongNumberOfBandsInSubFrameForInsertSubFrameAndExtractBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t correct_num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_bands = (correct_num_bands % 3) + 1;
+      RunWronglySizedInsertAndExtractParametersTest(
+          rate, correct_num_channels, correct_num_bands, correct_num_channels,
+          wrong_num_bands, correct_num_channels, kSubFrameLength);
+    }
+  }
+}
+
+TEST(FrameBlockerDeathTest,
+     WrongNumberOfChannelsInSubFrameForInsertSubFrameAndExtractBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t correct_num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_channels = correct_num_channels + 1;
+      RunWronglySizedInsertAndExtractParametersTest(
+          rate, correct_num_channels, correct_num_bands, wrong_num_channels,
+          correct_num_bands, wrong_num_channels, kSubFrameLength);
+    }
+  }
+}
+
+TEST(FrameBlockerDeathTest,
+     WrongNumberOfSamplesInSubFrameForInsertSubFrameAndExtractBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t correct_num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      RunWronglySizedInsertAndExtractParametersTest(
+          rate, correct_num_channels, correct_num_bands, correct_num_channels,
+          correct_num_bands, correct_num_channels, kSubFrameLength - 1);
+    }
+  }
+}
+
+TEST(FrameBlockerDeathTest, WrongNumberOfBandsInBlockForExtractBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t correct_num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_bands = (correct_num_bands % 3) + 1;
+      RunWronglySizedExtractParameterTest(
+          rate, correct_num_channels, wrong_num_bands, correct_num_channels);
+    }
+  }
+}
+
+TEST(FrameBlockerDeathTest, WrongNumberOfChannelsInBlockForExtractBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t correct_num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, correct_num_channels));
+      const size_t correct_num_bands = NumBandsForRate(rate);
+      const size_t wrong_num_channels = correct_num_channels + 1;
+      RunWronglySizedExtractParameterTest(
+          rate, correct_num_channels, correct_num_bands, wrong_num_channels);
+    }
+  }
+}
+
+TEST(FrameBlockerDeathTest, WrongNumberOfPreceedingApiCallsForExtractBlock) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t num_channels : {1, 2, 4, 8}) {
+      for (size_t num_calls = 0; num_calls < 4; ++num_calls) {
+        rtc::StringBuilder ss;
+        ss << "Sample rate: " << rate;
+        ss << "Num channels: " << num_channels;
+        ss << ", Num preceeding InsertSubFrameAndExtractBlock calls: "
+           << num_calls;
+
+        SCOPED_TRACE(ss.str());
+        RunWrongExtractOrderTest(rate, num_channels, num_calls);
+      }
+    }
+  }
+}
+
+// Verifies that the verification for 0 number of channels works.
+TEST(FrameBlockerDeathTest, ZeroNumberOfChannelsParameter) {
+  EXPECT_DEATH(FrameBlocker(16000, 0), "");
+}
+
+// Verifies that the verification for 0 number of bands works.
+TEST(FrameBlockerDeathTest, ZeroNumberOfBandsParameter) {
+  EXPECT_DEATH(FrameBlocker(0, 1), "");
+}
+
+// Verifiers that the verification for null sub_frame pointer works.
+TEST(FrameBlockerDeathTest, NullBlockParameter) {
+  std::vector<std::vector<std::vector<float>>> sub_frame(
+      1, std::vector<std::vector<float>>(
+             1, std::vector<float>(kSubFrameLength, 0.f)));
+  std::vector<std::vector<rtc::ArrayView<float>>> sub_frame_view(
+      sub_frame.size());
+  FillSubFrameView(0, 0, &sub_frame, &sub_frame_view);
+  EXPECT_DEATH(
+      FrameBlocker(1, 1).InsertSubFrameAndExtractBlock(sub_frame_view, nullptr),
+      "");
+}
+
+#endif
+
+TEST(FrameBlocker, BlockBitexactness) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, num_channels));
+      RunBlockerTest(rate, num_channels);
+    }
+  }
+}
+
+TEST(FrameBlocker, BlockerAndFramer) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t num_channels : {1, 2, 4, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate, num_channels));
+      RunBlockerAndFramerTest(rate, num_channels);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc
new file mode 100644
index 0000000000..e56674e4c9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc
@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/fullband_erle_estimator.h"
+
+#include <algorithm>
+#include <memory>
+#include <numeric>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+namespace {
+constexpr float kEpsilon = 1e-3f;
+constexpr float kX2BandEnergyThreshold = 44015068.0f;
+constexpr int kBlocksToHoldErle = 100;
+constexpr int kPointsToAccumulate = 6;
+}  // namespace
+
+FullBandErleEstimator::FullBandErleEstimator(
+    const EchoCanceller3Config::Erle& config,
+    size_t num_capture_channels)
+    : min_erle_log2_(FastApproxLog2f(config.min + kEpsilon)),
+      max_erle_lf_log2_(FastApproxLog2f(config.max_l + kEpsilon)),
+      hold_counters_instantaneous_erle_(num_capture_channels, 0),
+      erle_time_domain_log2_(num_capture_channels, min_erle_log2_),
+      instantaneous_erle_(num_capture_channels, ErleInstantaneous(config)),
+      linear_filters_qualities_(num_capture_channels) {
+  Reset();
+}
+
+FullBandErleEstimator::~FullBandErleEstimator() = default;
+
+void FullBandErleEstimator::Reset() {
+  for (auto& instantaneous_erle_ch : instantaneous_erle_) {
+    instantaneous_erle_ch.Reset();
+  }
+
+  UpdateQualityEstimates();
+  std::fill(erle_time_domain_log2_.begin(), erle_time_domain_log2_.end(),
+            min_erle_log2_);
+  std::fill(hold_counters_instantaneous_erle_.begin(),
+            hold_counters_instantaneous_erle_.end(), 0);
+}
+
+void FullBandErleEstimator::Update(
+    rtc::ArrayView<const float> X2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    const std::vector<bool>& converged_filters) {
+  for (size_t ch = 0; ch < Y2.size(); ++ch) {
+    if (converged_filters[ch]) {
+      // Computes the fullband ERLE.
+      const float X2_sum = std::accumulate(X2.begin(), X2.end(), 0.0f);
+      if (X2_sum > kX2BandEnergyThreshold * X2.size()) {
+        const float Y2_sum =
+            std::accumulate(Y2[ch].begin(), Y2[ch].end(), 0.0f);
+        const float E2_sum =
+            std::accumulate(E2[ch].begin(), E2[ch].end(), 0.0f);
+        if (instantaneous_erle_[ch].Update(Y2_sum, E2_sum)) {
+          hold_counters_instantaneous_erle_[ch] = kBlocksToHoldErle;
+          erle_time_domain_log2_[ch] +=
+              0.05f * ((instantaneous_erle_[ch].GetInstErleLog2().value()) -
+                       erle_time_domain_log2_[ch]);
+          erle_time_domain_log2_[ch] =
+              std::max(erle_time_domain_log2_[ch], min_erle_log2_);
+        }
+      }
+    }
+    --hold_counters_instantaneous_erle_[ch];
+    if (hold_counters_instantaneous_erle_[ch] == 0) {
+      instantaneous_erle_[ch].ResetAccumulators();
+    }
+  }
+
+  UpdateQualityEstimates();
+}
+
+void FullBandErleEstimator::Dump(
+    const std::unique_ptr<ApmDataDumper>& data_dumper) const {
+  data_dumper->DumpRaw("aec3_fullband_erle_log2", FullbandErleLog2());
+  instantaneous_erle_[0].Dump(data_dumper);
+}
+
+void FullBandErleEstimator::UpdateQualityEstimates() {
+  for (size_t ch = 0; ch < instantaneous_erle_.size(); ++ch) {
+    linear_filters_qualities_[ch] =
+        instantaneous_erle_[ch].GetQualityEstimate();
+  }
+}
+
+FullBandErleEstimator::ErleInstantaneous::ErleInstantaneous(
+    const EchoCanceller3Config::Erle& config)
+    : clamp_inst_quality_to_zero_(config.clamp_quality_estimate_to_zero),
+      clamp_inst_quality_to_one_(config.clamp_quality_estimate_to_one) {
+  Reset();
+}
+
+FullBandErleEstimator::ErleInstantaneous::~ErleInstantaneous() = default;
+
+bool FullBandErleEstimator::ErleInstantaneous::Update(const float Y2_sum,
+                                                      const float E2_sum) {
+  bool update_estimates = false;
+  E2_acum_ += E2_sum;
+  Y2_acum_ += Y2_sum;
+  num_points_++;
+  if (num_points_ == kPointsToAccumulate) {
+    if (E2_acum_ > 0.f) {
+      update_estimates = true;
+      erle_log2_ = FastApproxLog2f(Y2_acum_ / E2_acum_ + kEpsilon);
+    }
+    num_points_ = 0;
+    E2_acum_ = 0.f;
+    Y2_acum_ = 0.f;
+  }
+
+  if (update_estimates) {
+    UpdateMaxMin();
+    UpdateQualityEstimate();
+  }
+  return update_estimates;
+}
+
+void FullBandErleEstimator::ErleInstantaneous::Reset() {
+  ResetAccumulators();
+  max_erle_log2_ = -10.f;  // -30 dB.
+  min_erle_log2_ = 33.f;   // 100 dB.
+  inst_quality_estimate_ = 0.f;
+}
+
+void FullBandErleEstimator::ErleInstantaneous::ResetAccumulators() {
+  erle_log2_ = absl::nullopt;
+  inst_quality_estimate_ = 0.f;
+  num_points_ = 0;
+  E2_acum_ = 0.f;
+  Y2_acum_ = 0.f;
+}
+
+void FullBandErleEstimator::ErleInstantaneous::Dump(
+    const std::unique_ptr<ApmDataDumper>& data_dumper) const {
+  data_dumper->DumpRaw("aec3_fullband_erle_inst_log2",
+                       erle_log2_ ? *erle_log2_ : -10.f);
+  data_dumper->DumpRaw(
+      "aec3_erle_instantaneous_quality",
+      GetQualityEstimate() ? GetQualityEstimate().value() : 0.f);
+  data_dumper->DumpRaw("aec3_fullband_erle_max_log2", max_erle_log2_);
+  data_dumper->DumpRaw("aec3_fullband_erle_min_log2", min_erle_log2_);
+}
+
+void FullBandErleEstimator::ErleInstantaneous::UpdateMaxMin() {
+  RTC_DCHECK(erle_log2_);
+  // Adding the forgetting factors for the maximum and minimum and capping the
+  // result to the incoming value.
+  max_erle_log2_ -= 0.0004f;  // Forget factor, approx 1dB every 3 sec.
+  max_erle_log2_ = std::max(max_erle_log2_, erle_log2_.value());
+  min_erle_log2_ += 0.0004f;  // Forget factor, approx 1dB every 3 sec.
+  min_erle_log2_ = std::min(min_erle_log2_, erle_log2_.value());
+}
+
+void FullBandErleEstimator::ErleInstantaneous::UpdateQualityEstimate() {
+  const float alpha = 0.07f;
+  float quality_estimate = 0.f;
+  RTC_DCHECK(erle_log2_);
+  // TODO(peah): Currently, the estimate can become be less than 0; this should
+  // be corrected.
+  if (max_erle_log2_ > min_erle_log2_) {
+    quality_estimate = (erle_log2_.value() - min_erle_log2_) /
+                       (max_erle_log2_ - min_erle_log2_);
+  }
+  if (quality_estimate > inst_quality_estimate_) {
+    inst_quality_estimate_ = quality_estimate;
+  } else {
+    inst_quality_estimate_ +=
+        alpha * (quality_estimate - inst_quality_estimate_);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/fullband_erle_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/fullband_erle_estimator.h
new file mode 100644
index 0000000000..7a082176d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/fullband_erle_estimator.h
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_FULLBAND_ERLE_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_FULLBAND_ERLE_ESTIMATOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+// Estimates the echo return loss enhancement using the energy of all the
+// freuquency bands.
+class FullBandErleEstimator {
+ public:
+  FullBandErleEstimator(const EchoCanceller3Config::Erle& config,
+                        size_t num_capture_channels);
+  ~FullBandErleEstimator();
+  // Resets the ERLE estimator.
+  void Reset();
+
+  // Updates the ERLE estimator.
+  void Update(rtc::ArrayView<const float> X2,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+              const std::vector<bool>& converged_filters);
+
+  // Returns the fullband ERLE estimates in log2 units.
+  float FullbandErleLog2() const {
+    float min_erle = erle_time_domain_log2_[0];
+    for (size_t ch = 1; ch < erle_time_domain_log2_.size(); ++ch) {
+      min_erle = std::min(min_erle, erle_time_domain_log2_[ch]);
+    }
+    return min_erle;
+  }
+
+  // Returns an estimation of the current linear filter quality. It returns a
+  // float number between 0 and 1 mapping 1 to the highest possible quality.
+  rtc::ArrayView<const absl::optional<float>> GetInstLinearQualityEstimates()
+      const {
+    return linear_filters_qualities_;
+  }
+
+  void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper) const;
+
+ private:
+  void UpdateQualityEstimates();
+
+  class ErleInstantaneous {
+   public:
+    explicit ErleInstantaneous(const EchoCanceller3Config::Erle& config);
+    ~ErleInstantaneous();
+
+    // Updates the estimator with a new point, returns true
+    // if the instantaneous ERLE was updated due to having enough
+    // points for performing the estimate.
+    bool Update(float Y2_sum, float E2_sum);
+    // Resets the instantaneous ERLE estimator to its initial state.
+    void Reset();
+    // Resets the members related with an instantaneous estimate.
+    void ResetAccumulators();
+    // Returns the instantaneous ERLE in log2 units.
+    absl::optional<float> GetInstErleLog2() const { return erle_log2_; }
+    // Gets an indication between 0 and 1 of the performance of the linear
+    // filter for the current time instant.
+    absl::optional<float> GetQualityEstimate() const {
+      if (erle_log2_) {
+        float value = inst_quality_estimate_;
+        if (clamp_inst_quality_to_zero_) {
+          value = std::max(0.f, value);
+        }
+        if (clamp_inst_quality_to_one_) {
+          value = std::min(1.f, value);
+        }
+        return absl::optional<float>(value);
+      }
+      return absl::nullopt;
+    }
+    void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper) const;
+
+   private:
+    void UpdateMaxMin();
+    void UpdateQualityEstimate();
+    const bool clamp_inst_quality_to_zero_;
+    const bool clamp_inst_quality_to_one_;
+    absl::optional<float> erle_log2_;
+    float inst_quality_estimate_;
+    float max_erle_log2_;
+    float min_erle_log2_;
+    float Y2_acum_;
+    float E2_acum_;
+    int num_points_;
+  };
+
+  const float min_erle_log2_;
+  const float max_erle_lf_log2_;
+  std::vector<int> hold_counters_instantaneous_erle_;
+  std::vector<float> erle_time_domain_log2_;
+  std::vector<ErleInstantaneous> instantaneous_erle_;
+  std::vector<absl::optional<float>> linear_filters_qualities_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_FULLBAND_ERLE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter.cc b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter.cc
new file mode 100644
index 0000000000..af30ff1b9f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter.cc
@@ -0,0 +1,900 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/matched_filter.h"
+
+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_HAS_NEON)
+#include <arm_neon.h>
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+#include <algorithm>
+#include <cstddef>
+#include <initializer_list>
+#include <iterator>
+#include <numeric>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace {
+
+// Subsample rate used for computing the accumulated error.
+// The implementation of some core functions depends on this constant being
+// equal to 4.
+constexpr int kAccumulatedErrorSubSampleRate = 4;
+
+void UpdateAccumulatedError(
+    const rtc::ArrayView<const float> instantaneous_accumulated_error,
+    const rtc::ArrayView<float> accumulated_error,
+    float one_over_error_sum_anchor,
+    float smooth_constant_increases) {
+  for (size_t k = 0; k < instantaneous_accumulated_error.size(); ++k) {
+    float error_norm =
+        instantaneous_accumulated_error[k] * one_over_error_sum_anchor;
+    if (error_norm < accumulated_error[k]) {
+      accumulated_error[k] = error_norm;
+    } else {
+      accumulated_error[k] +=
+          smooth_constant_increases * (error_norm - accumulated_error[k]);
+    }
+  }
+}
+
+size_t ComputePreEchoLag(
+    const webrtc::MatchedFilter::PreEchoConfiguration& pre_echo_configuration,
+    const rtc::ArrayView<const float> accumulated_error,
+    size_t lag,
+    size_t alignment_shift_winner) {
+  RTC_DCHECK_GE(lag, alignment_shift_winner);
+  size_t pre_echo_lag_estimate = lag - alignment_shift_winner;
+  size_t maximum_pre_echo_lag =
+      std::min(pre_echo_lag_estimate / kAccumulatedErrorSubSampleRate,
+               accumulated_error.size());
+  switch (pre_echo_configuration.mode) {
+    case 0:
+      // Mode 0: Pre echo lag is defined as the first coefficient with an error
+      // lower than a threshold with a certain decrease slope.
+      for (size_t k = 1; k < maximum_pre_echo_lag; ++k) {
+        if (accumulated_error[k] <
+                pre_echo_configuration.threshold * accumulated_error[k - 1] &&
+            accumulated_error[k] < pre_echo_configuration.threshold) {
+          pre_echo_lag_estimate = (k + 1) * kAccumulatedErrorSubSampleRate - 1;
+          break;
+        }
+      }
+      break;
+    case 1:
+      // Mode 1: Pre echo lag is defined as the first coefficient with an error
+      // lower than a certain threshold.
+      for (size_t k = 0; k < maximum_pre_echo_lag; ++k) {
+        if (accumulated_error[k] < pre_echo_configuration.threshold) {
+          pre_echo_lag_estimate = (k + 1) * kAccumulatedErrorSubSampleRate - 1;
+          break;
+        }
+      }
+      break;
+    case 2:
+    case 3:
+      // Mode 2,3: Pre echo lag is defined as the closest coefficient to the lag
+      // with an error lower than a certain threshold.
+      for (int k = static_cast<int>(maximum_pre_echo_lag) - 1; k >= 0; --k) {
+        if (accumulated_error[k] > pre_echo_configuration.threshold) {
+          break;
+        }
+        pre_echo_lag_estimate = (k + 1) * kAccumulatedErrorSubSampleRate - 1;
+      }
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+  return pre_echo_lag_estimate + alignment_shift_winner;
+}
+
+webrtc::MatchedFilter::PreEchoConfiguration FetchPreEchoConfiguration() {
+  float threshold = 0.5f;
+  int mode = 0;
+  const std::string pre_echo_configuration_field_trial =
+      webrtc::field_trial::FindFullName("WebRTC-Aec3PreEchoConfiguration");
+  webrtc::FieldTrialParameter<double> threshold_field_trial_parameter(
+      /*key=*/"threshold", /*default_value=*/threshold);
+  webrtc::FieldTrialParameter<int> mode_field_trial_parameter(
+      /*key=*/"mode", /*default_value=*/mode);
+  webrtc::ParseFieldTrial(
+      {&threshold_field_trial_parameter, &mode_field_trial_parameter},
+      pre_echo_configuration_field_trial);
+  float threshold_read =
+      static_cast<float>(threshold_field_trial_parameter.Get());
+  int mode_read = mode_field_trial_parameter.Get();
+  if (threshold_read < 1.0f && threshold_read > 0.0f) {
+    threshold = threshold_read;
+  } else {
+    RTC_LOG(LS_ERROR)
+        << "AEC3: Pre echo configuration:  wrong input, threshold = "
+        << threshold_read << ".";
+  }
+  if (mode_read >= 0 && mode_read <= 3) {
+    mode = mode_read;
+  } else {
+    RTC_LOG(LS_ERROR) << "AEC3: Pre echo configuration:  wrong input, mode = "
+                      << mode_read << ".";
+  }
+  RTC_LOG(LS_INFO) << "AEC3: Pre echo configuration:  threshold = " << threshold
+                   << ", mode =  " << mode << ".";
+  return {.threshold = threshold, .mode = mode};
+}
+
+}  // namespace
+
+namespace webrtc {
+namespace aec3 {
+
+#if defined(WEBRTC_HAS_NEON)
+
+inline float SumAllElements(float32x4_t elements) {
+  float32x2_t sum = vpadd_f32(vget_low_f32(elements), vget_high_f32(elements));
+  sum = vpadd_f32(sum, sum);
+  return vget_lane_f32(sum, 0);
+}
+
+void MatchedFilterCoreWithAccumulatedError_NEON(
+    size_t x_start_index,
+    float x2_sum_threshold,
+    float smoothing,
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<const float> y,
+    rtc::ArrayView<float> h,
+    bool* filters_updated,
+    float* error_sum,
+    rtc::ArrayView<float> accumulated_error,
+    rtc::ArrayView<float> scratch_memory) {
+  const int h_size = static_cast<int>(h.size());
+  const int x_size = static_cast<int>(x.size());
+  RTC_DCHECK_EQ(0, h_size % 4);
+  std::fill(accumulated_error.begin(), accumulated_error.end(), 0.0f);
+  // Process for all samples in the sub-block.
+  for (size_t i = 0; i < y.size(); ++i) {
+    // Apply the matched filter as filter * x, and compute x * x.
+    RTC_DCHECK_GT(x_size, x_start_index);
+    // Compute loop chunk sizes until, and after, the wraparound of the circular
+    // buffer for x.
+    const int chunk1 =
+        std::min(h_size, static_cast<int>(x_size - x_start_index));
+    if (chunk1 != h_size) {
+      const int chunk2 = h_size - chunk1;
+      std::copy(x.begin() + x_start_index, x.end(), scratch_memory.begin());
+      std::copy(x.begin(), x.begin() + chunk2, scratch_memory.begin() + chunk1);
+    }
+    const float* x_p =
+        chunk1 != h_size ? scratch_memory.data() : &x[x_start_index];
+    const float* h_p = &h[0];
+    float* accumulated_error_p = &accumulated_error[0];
+    // Initialize values for the accumulation.
+    float32x4_t x2_sum_128 = vdupq_n_f32(0);
+    float x2_sum = 0.f;
+    float s = 0;
+    // Perform 128 bit vector operations.
+    const int limit_by_4 = h_size >> 2;
+    for (int k = limit_by_4; k > 0;
+         --k, h_p += 4, x_p += 4, accumulated_error_p++) {
+      // Load the data into 128 bit vectors.
+      const float32x4_t x_k = vld1q_f32(x_p);
+      const float32x4_t h_k = vld1q_f32(h_p);
+      // Compute and accumulate x * x.
+      x2_sum_128 = vmlaq_f32(x2_sum_128, x_k, x_k);
+      // Compute x * h
+      float32x4_t hk_xk_128 = vmulq_f32(h_k, x_k);
+      s += SumAllElements(hk_xk_128);
+      const float e = s - y[i];
+      accumulated_error_p[0] += e * e;
+    }
+    // Combine the accumulated vector and scalar values.
+    x2_sum += SumAllElements(x2_sum_128);
+    // Compute the matched filter error.
+    float e = y[i] - s;
+    const bool saturation = y[i] >= 32000.f || y[i] <= -32000.f;
+    (*error_sum) += e * e;
+    // Update the matched filter estimate in an NLMS manner.
+    if (x2_sum > x2_sum_threshold && !saturation) {
+      RTC_DCHECK_LT(0.f, x2_sum);
+      const float alpha = smoothing * e / x2_sum;
+      const float32x4_t alpha_128 = vmovq_n_f32(alpha);
+      // filter = filter + smoothing * (y - filter * x) * x / x * x.
+      float* h_p = &h[0];
+      x_p = chunk1 != h_size ? scratch_memory.data() : &x[x_start_index];
+      // Perform 128 bit vector operations.
+      const int limit_by_4 = h_size >> 2;
+      for (int k = limit_by_4; k > 0; --k, h_p += 4, x_p += 4) {
+        // Load the data into 128 bit vectors.
+        float32x4_t h_k = vld1q_f32(h_p);
+        const float32x4_t x_k = vld1q_f32(x_p);
+        // Compute h = h + alpha * x.
+        h_k = vmlaq_f32(h_k, alpha_128, x_k);
+        // Store the result.
+        vst1q_f32(h_p, h_k);
+      }
+      *filters_updated = true;
+    }
+    x_start_index = x_start_index > 0 ? x_start_index - 1 : x_size - 1;
+  }
+}
+
+void MatchedFilterCore_NEON(size_t x_start_index,
+                            float x2_sum_threshold,
+                            float smoothing,
+                            rtc::ArrayView<const float> x,
+                            rtc::ArrayView<const float> y,
+                            rtc::ArrayView<float> h,
+                            bool* filters_updated,
+                            float* error_sum,
+                            bool compute_accumulated_error,
+                            rtc::ArrayView<float> accumulated_error,
+                            rtc::ArrayView<float> scratch_memory) {
+  const int h_size = static_cast<int>(h.size());
+  const int x_size = static_cast<int>(x.size());
+  RTC_DCHECK_EQ(0, h_size % 4);
+
+  if (compute_accumulated_error) {
+    return MatchedFilterCoreWithAccumulatedError_NEON(
+        x_start_index, x2_sum_threshold, smoothing, x, y, h, filters_updated,
+        error_sum, accumulated_error, scratch_memory);
+  }
+
+  // Process for all samples in the sub-block.
+  for (size_t i = 0; i < y.size(); ++i) {
+    // Apply the matched filter as filter * x, and compute x * x.
+
+    RTC_DCHECK_GT(x_size, x_start_index);
+    const float* x_p = &x[x_start_index];
+    const float* h_p = &h[0];
+
+    // Initialize values for the accumulation.
+    float32x4_t s_128 = vdupq_n_f32(0);
+    float32x4_t x2_sum_128 = vdupq_n_f32(0);
+    float x2_sum = 0.f;
+    float s = 0;
+
+    // Compute loop chunk sizes until, and after, the wraparound of the circular
+    // buffer for x.
+    const int chunk1 =
+        std::min(h_size, static_cast<int>(x_size - x_start_index));
+
+    // Perform the loop in two chunks.
+    const int chunk2 = h_size - chunk1;
+    for (int limit : {chunk1, chunk2}) {
+      // Perform 128 bit vector operations.
+      const int limit_by_4 = limit >> 2;
+      for (int k = limit_by_4; k > 0; --k, h_p += 4, x_p += 4) {
+        // Load the data into 128 bit vectors.
+        const float32x4_t x_k = vld1q_f32(x_p);
+        const float32x4_t h_k = vld1q_f32(h_p);
+        // Compute and accumulate x * x and h * x.
+        x2_sum_128 = vmlaq_f32(x2_sum_128, x_k, x_k);
+        s_128 = vmlaq_f32(s_128, h_k, x_k);
+      }
+
+      // Perform non-vector operations for any remaining items.
+      for (int k = limit - limit_by_4 * 4; k > 0; --k, ++h_p, ++x_p) {
+        const float x_k = *x_p;
+        x2_sum += x_k * x_k;
+        s += *h_p * x_k;
+      }
+
+      x_p = &x[0];
+    }
+
+    // Combine the accumulated vector and scalar values.
+    s += SumAllElements(s_128);
+    x2_sum += SumAllElements(x2_sum_128);
+
+    // Compute the matched filter error.
+    float e = y[i] - s;
+    const bool saturation = y[i] >= 32000.f || y[i] <= -32000.f;
+    (*error_sum) += e * e;
+
+    // Update the matched filter estimate in an NLMS manner.
+    if (x2_sum > x2_sum_threshold && !saturation) {
+      RTC_DCHECK_LT(0.f, x2_sum);
+      const float alpha = smoothing * e / x2_sum;
+      const float32x4_t alpha_128 = vmovq_n_f32(alpha);
+
+      // filter = filter + smoothing * (y - filter * x) * x / x * x.
+      float* h_p = &h[0];
+      x_p = &x[x_start_index];
+
+      // Perform the loop in two chunks.
+      for (int limit : {chunk1, chunk2}) {
+        // Perform 128 bit vector operations.
+        const int limit_by_4 = limit >> 2;
+        for (int k = limit_by_4; k > 0; --k, h_p += 4, x_p += 4) {
+          // Load the data into 128 bit vectors.
+          float32x4_t h_k = vld1q_f32(h_p);
+          const float32x4_t x_k = vld1q_f32(x_p);
+          // Compute h = h + alpha * x.
+          h_k = vmlaq_f32(h_k, alpha_128, x_k);
+
+          // Store the result.
+          vst1q_f32(h_p, h_k);
+        }
+
+        // Perform non-vector operations for any remaining items.
+        for (int k = limit - limit_by_4 * 4; k > 0; --k, ++h_p, ++x_p) {
+          *h_p += alpha * *x_p;
+        }
+
+        x_p = &x[0];
+      }
+
+      *filters_updated = true;
+    }
+
+    x_start_index = x_start_index > 0 ? x_start_index - 1 : x_size - 1;
+  }
+}
+
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+
+void MatchedFilterCore_AccumulatedError_SSE2(
+    size_t x_start_index,
+    float x2_sum_threshold,
+    float smoothing,
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<const float> y,
+    rtc::ArrayView<float> h,
+    bool* filters_updated,
+    float* error_sum,
+    rtc::ArrayView<float> accumulated_error,
+    rtc::ArrayView<float> scratch_memory) {
+  const int h_size = static_cast<int>(h.size());
+  const int x_size = static_cast<int>(x.size());
+  RTC_DCHECK_EQ(0, h_size % 8);
+  std::fill(accumulated_error.begin(), accumulated_error.end(), 0.0f);
+  // Process for all samples in the sub-block.
+  for (size_t i = 0; i < y.size(); ++i) {
+    // Apply the matched filter as filter * x, and compute x * x.
+    RTC_DCHECK_GT(x_size, x_start_index);
+    const int chunk1 =
+        std::min(h_size, static_cast<int>(x_size - x_start_index));
+    if (chunk1 != h_size) {
+      const int chunk2 = h_size - chunk1;
+      std::copy(x.begin() + x_start_index, x.end(), scratch_memory.begin());
+      std::copy(x.begin(), x.begin() + chunk2, scratch_memory.begin() + chunk1);
+    }
+    const float* x_p =
+        chunk1 != h_size ? scratch_memory.data() : &x[x_start_index];
+    const float* h_p = &h[0];
+    float* a_p = &accumulated_error[0];
+    __m128 s_inst_128;
+    __m128 s_inst_128_4;
+    __m128 x2_sum_128 = _mm_set1_ps(0);
+    __m128 x2_sum_128_4 = _mm_set1_ps(0);
+    __m128 e_128;
+    float* const s_p = reinterpret_cast<float*>(&s_inst_128);
+    float* const s_4_p = reinterpret_cast<float*>(&s_inst_128_4);
+    float* const e_p = reinterpret_cast<float*>(&e_128);
+    float x2_sum = 0.0f;
+    float s_acum = 0;
+    // Perform 128 bit vector operations.
+    const int limit_by_8 = h_size >> 3;
+    for (int k = limit_by_8; k > 0; --k, h_p += 8, x_p += 8, a_p += 2) {
+      // Load the data into 128 bit vectors.
+      const __m128 x_k = _mm_loadu_ps(x_p);
+      const __m128 h_k = _mm_loadu_ps(h_p);
+      const __m128 x_k_4 = _mm_loadu_ps(x_p + 4);
+      const __m128 h_k_4 = _mm_loadu_ps(h_p + 4);
+      const __m128 xx = _mm_mul_ps(x_k, x_k);
+      const __m128 xx_4 = _mm_mul_ps(x_k_4, x_k_4);
+      // Compute and accumulate x * x and h * x.
+      x2_sum_128 = _mm_add_ps(x2_sum_128, xx);
+      x2_sum_128_4 = _mm_add_ps(x2_sum_128_4, xx_4);
+      s_inst_128 = _mm_mul_ps(h_k, x_k);
+      s_inst_128_4 = _mm_mul_ps(h_k_4, x_k_4);
+      s_acum += s_p[0] + s_p[1] + s_p[2] + s_p[3];
+      e_p[0] = s_acum - y[i];
+      s_acum += s_4_p[0] + s_4_p[1] + s_4_p[2] + s_4_p[3];
+      e_p[1] = s_acum - y[i];
+      a_p[0] += e_p[0] * e_p[0];
+      a_p[1] += e_p[1] * e_p[1];
+    }
+    // Combine the accumulated vector and scalar values.
+    x2_sum_128 = _mm_add_ps(x2_sum_128, x2_sum_128_4);
+    float* v = reinterpret_cast<float*>(&x2_sum_128);
+    x2_sum += v[0] + v[1] + v[2] + v[3];
+    // Compute the matched filter error.
+    float e = y[i] - s_acum;
+    const bool saturation = y[i] >= 32000.f || y[i] <= -32000.f;
+    (*error_sum) += e * e;
+    // Update the matched filter estimate in an NLMS manner.
+    if (x2_sum > x2_sum_threshold && !saturation) {
+      RTC_DCHECK_LT(0.f, x2_sum);
+      const float alpha = smoothing * e / x2_sum;
+      const __m128 alpha_128 = _mm_set1_ps(alpha);
+      // filter = filter + smoothing * (y - filter * x) * x / x * x.
+      float* h_p = &h[0];
+      const float* x_p =
+          chunk1 != h_size ? scratch_memory.data() : &x[x_start_index];
+      // Perform 128 bit vector operations.
+      const int limit_by_4 = h_size >> 2;
+      for (int k = limit_by_4; k > 0; --k, h_p += 4, x_p += 4) {
+        // Load the data into 128 bit vectors.
+        __m128 h_k = _mm_loadu_ps(h_p);
+        const __m128 x_k = _mm_loadu_ps(x_p);
+        // Compute h = h + alpha * x.
+        const __m128 alpha_x = _mm_mul_ps(alpha_128, x_k);
+        h_k = _mm_add_ps(h_k, alpha_x);
+        // Store the result.
+        _mm_storeu_ps(h_p, h_k);
+      }
+      *filters_updated = true;
+    }
+    x_start_index = x_start_index > 0 ? x_start_index - 1 : x_size - 1;
+  }
+}
+
+void MatchedFilterCore_SSE2(size_t x_start_index,
+                            float x2_sum_threshold,
+                            float smoothing,
+                            rtc::ArrayView<const float> x,
+                            rtc::ArrayView<const float> y,
+                            rtc::ArrayView<float> h,
+                            bool* filters_updated,
+                            float* error_sum,
+                            bool compute_accumulated_error,
+                            rtc::ArrayView<float> accumulated_error,
+                            rtc::ArrayView<float> scratch_memory) {
+  if (compute_accumulated_error) {
+    return MatchedFilterCore_AccumulatedError_SSE2(
+        x_start_index, x2_sum_threshold, smoothing, x, y, h, filters_updated,
+        error_sum, accumulated_error, scratch_memory);
+  }
+  const int h_size = static_cast<int>(h.size());
+  const int x_size = static_cast<int>(x.size());
+  RTC_DCHECK_EQ(0, h_size % 4);
+  // Process for all samples in the sub-block.
+  for (size_t i = 0; i < y.size(); ++i) {
+    // Apply the matched filter as filter * x, and compute x * x.
+    RTC_DCHECK_GT(x_size, x_start_index);
+    const float* x_p = &x[x_start_index];
+    const float* h_p = &h[0];
+    // Initialize values for the accumulation.
+    __m128 s_128 = _mm_set1_ps(0);
+    __m128 s_128_4 = _mm_set1_ps(0);
+    __m128 x2_sum_128 = _mm_set1_ps(0);
+    __m128 x2_sum_128_4 = _mm_set1_ps(0);
+    float x2_sum = 0.f;
+    float s = 0;
+    // Compute loop chunk sizes until, and after, the wraparound of the circular
+    // buffer for x.
+    const int chunk1 =
+        std::min(h_size, static_cast<int>(x_size - x_start_index));
+    // Perform the loop in two chunks.
+    const int chunk2 = h_size - chunk1;
+    for (int limit : {chunk1, chunk2}) {
+      // Perform 128 bit vector operations.
+      const int limit_by_8 = limit >> 3;
+      for (int k = limit_by_8; k > 0; --k, h_p += 8, x_p += 8) {
+        // Load the data into 128 bit vectors.
+        const __m128 x_k = _mm_loadu_ps(x_p);
+        const __m128 h_k = _mm_loadu_ps(h_p);
+        const __m128 x_k_4 = _mm_loadu_ps(x_p + 4);
+        const __m128 h_k_4 = _mm_loadu_ps(h_p + 4);
+        const __m128 xx = _mm_mul_ps(x_k, x_k);
+        const __m128 xx_4 = _mm_mul_ps(x_k_4, x_k_4);
+        // Compute and accumulate x * x and h * x.
+        x2_sum_128 = _mm_add_ps(x2_sum_128, xx);
+        x2_sum_128_4 = _mm_add_ps(x2_sum_128_4, xx_4);
+        const __m128 hx = _mm_mul_ps(h_k, x_k);
+        const __m128 hx_4 = _mm_mul_ps(h_k_4, x_k_4);
+        s_128 = _mm_add_ps(s_128, hx);
+        s_128_4 = _mm_add_ps(s_128_4, hx_4);
+      }
+      // Perform non-vector operations for any remaining items.
+      for (int k = limit - limit_by_8 * 8; k > 0; --k, ++h_p, ++x_p) {
+        const float x_k = *x_p;
+        x2_sum += x_k * x_k;
+        s += *h_p * x_k;
+      }
+      x_p = &x[0];
+    }
+    // Combine the accumulated vector and scalar values.
+    x2_sum_128 = _mm_add_ps(x2_sum_128, x2_sum_128_4);
+    float* v = reinterpret_cast<float*>(&x2_sum_128);
+    x2_sum += v[0] + v[1] + v[2] + v[3];
+    s_128 = _mm_add_ps(s_128, s_128_4);
+    v = reinterpret_cast<float*>(&s_128);
+    s += v[0] + v[1] + v[2] + v[3];
+    // Compute the matched filter error.
+    float e = y[i] - s;
+    const bool saturation = y[i] >= 32000.f || y[i] <= -32000.f;
+    (*error_sum) += e * e;
+    // Update the matched filter estimate in an NLMS manner.
+    if (x2_sum > x2_sum_threshold && !saturation) {
+      RTC_DCHECK_LT(0.f, x2_sum);
+      const float alpha = smoothing * e / x2_sum;
+      const __m128 alpha_128 = _mm_set1_ps(alpha);
+      // filter = filter + smoothing * (y - filter * x) * x / x * x.
+      float* h_p = &h[0];
+      x_p = &x[x_start_index];
+      // Perform the loop in two chunks.
+      for (int limit : {chunk1, chunk2}) {
+        // Perform 128 bit vector operations.
+        const int limit_by_4 = limit >> 2;
+        for (int k = limit_by_4; k > 0; --k, h_p += 4, x_p += 4) {
+          // Load the data into 128 bit vectors.
+          __m128 h_k = _mm_loadu_ps(h_p);
+          const __m128 x_k = _mm_loadu_ps(x_p);
+
+          // Compute h = h + alpha * x.
+          const __m128 alpha_x = _mm_mul_ps(alpha_128, x_k);
+          h_k = _mm_add_ps(h_k, alpha_x);
+          // Store the result.
+          _mm_storeu_ps(h_p, h_k);
+        }
+        // Perform non-vector operations for any remaining items.
+        for (int k = limit - limit_by_4 * 4; k > 0; --k, ++h_p, ++x_p) {
+          *h_p += alpha * *x_p;
+        }
+        x_p = &x[0];
+      }
+      *filters_updated = true;
+    }
+    x_start_index = x_start_index > 0 ? x_start_index - 1 : x_size - 1;
+  }
+}
+#endif
+
+void MatchedFilterCore(size_t x_start_index,
+                       float x2_sum_threshold,
+                       float smoothing,
+                       rtc::ArrayView<const float> x,
+                       rtc::ArrayView<const float> y,
+                       rtc::ArrayView<float> h,
+                       bool* filters_updated,
+                       float* error_sum,
+                       bool compute_accumulated_error,
+                       rtc::ArrayView<float> accumulated_error) {
+  if (compute_accumulated_error) {
+    std::fill(accumulated_error.begin(), accumulated_error.end(), 0.0f);
+  }
+
+  // Process for all samples in the sub-block.
+  for (size_t i = 0; i < y.size(); ++i) {
+    // Apply the matched filter as filter * x, and compute x * x.
+    float x2_sum = 0.f;
+    float s = 0;
+    size_t x_index = x_start_index;
+    if (compute_accumulated_error) {
+      for (size_t k = 0; k < h.size(); ++k) {
+        x2_sum += x[x_index] * x[x_index];
+        s += h[k] * x[x_index];
+        x_index = x_index < (x.size() - 1) ? x_index + 1 : 0;
+        if ((k + 1 & 0b11) == 0) {
+          int idx = k >> 2;
+          accumulated_error[idx] += (y[i] - s) * (y[i] - s);
+        }
+      }
+    } else {
+      for (size_t k = 0; k < h.size(); ++k) {
+        x2_sum += x[x_index] * x[x_index];
+        s += h[k] * x[x_index];
+        x_index = x_index < (x.size() - 1) ? x_index + 1 : 0;
+      }
+    }
+
+    // Compute the matched filter error.
+    float e = y[i] - s;
+    const bool saturation = y[i] >= 32000.f || y[i] <= -32000.f;
+    (*error_sum) += e * e;
+
+    // Update the matched filter estimate in an NLMS manner.
+    if (x2_sum > x2_sum_threshold && !saturation) {
+      RTC_DCHECK_LT(0.f, x2_sum);
+      const float alpha = smoothing * e / x2_sum;
+
+      // filter = filter + smoothing * (y - filter * x) * x / x * x.
+      size_t x_index = x_start_index;
+      for (size_t k = 0; k < h.size(); ++k) {
+        h[k] += alpha * x[x_index];
+        x_index = x_index < (x.size() - 1) ? x_index + 1 : 0;
+      }
+      *filters_updated = true;
+    }
+
+    x_start_index = x_start_index > 0 ? x_start_index - 1 : x.size() - 1;
+  }
+}
+
+size_t MaxSquarePeakIndex(rtc::ArrayView<const float> h) {
+  if (h.size() < 2) {
+    return 0;
+  }
+  float max_element1 = h[0] * h[0];
+  float max_element2 = h[1] * h[1];
+  size_t lag_estimate1 = 0;
+  size_t lag_estimate2 = 1;
+  const size_t last_index = h.size() - 1;
+  // Keeping track of even & odd max elements separately typically allows the
+  // compiler to produce more efficient code.
+  for (size_t k = 2; k < last_index; k += 2) {
+    float element1 = h[k] * h[k];
+    float element2 = h[k + 1] * h[k + 1];
+    if (element1 > max_element1) {
+      max_element1 = element1;
+      lag_estimate1 = k;
+    }
+    if (element2 > max_element2) {
+      max_element2 = element2;
+      lag_estimate2 = k + 1;
+    }
+  }
+  if (max_element2 > max_element1) {
+    max_element1 = max_element2;
+    lag_estimate1 = lag_estimate2;
+  }
+  // In case of odd h size, we have not yet checked the last element.
+  float last_element = h[last_index] * h[last_index];
+  if (last_element > max_element1) {
+    return last_index;
+  }
+  return lag_estimate1;
+}
+
+}  // namespace aec3
+
+MatchedFilter::MatchedFilter(ApmDataDumper* data_dumper,
+                             Aec3Optimization optimization,
+                             size_t sub_block_size,
+                             size_t window_size_sub_blocks,
+                             int num_matched_filters,
+                             size_t alignment_shift_sub_blocks,
+                             float excitation_limit,
+                             float smoothing_fast,
+                             float smoothing_slow,
+                             float matching_filter_threshold,
+                             bool detect_pre_echo)
+    : data_dumper_(data_dumper),
+      optimization_(optimization),
+      sub_block_size_(sub_block_size),
+      filter_intra_lag_shift_(alignment_shift_sub_blocks * sub_block_size_),
+      filters_(
+          num_matched_filters,
+          std::vector<float>(window_size_sub_blocks * sub_block_size_, 0.f)),
+      filters_offsets_(num_matched_filters, 0),
+      excitation_limit_(excitation_limit),
+      smoothing_fast_(smoothing_fast),
+      smoothing_slow_(smoothing_slow),
+      matching_filter_threshold_(matching_filter_threshold),
+      detect_pre_echo_(detect_pre_echo),
+      pre_echo_config_(FetchPreEchoConfiguration()) {
+  RTC_DCHECK(data_dumper);
+  RTC_DCHECK_LT(0, window_size_sub_blocks);
+  RTC_DCHECK((kBlockSize % sub_block_size) == 0);
+  RTC_DCHECK((sub_block_size % 4) == 0);
+  static_assert(kAccumulatedErrorSubSampleRate == 4);
+  if (detect_pre_echo_) {
+    accumulated_error_ = std::vector<std::vector<float>>(
+        num_matched_filters,
+        std::vector<float>(window_size_sub_blocks * sub_block_size_ /
+                               kAccumulatedErrorSubSampleRate,
+                           1.0f));
+
+    instantaneous_accumulated_error_ =
+        std::vector<float>(window_size_sub_blocks * sub_block_size_ /
+                               kAccumulatedErrorSubSampleRate,
+                           0.0f);
+    scratch_memory_ =
+        std::vector<float>(window_size_sub_blocks * sub_block_size_);
+  }
+}
+
+MatchedFilter::~MatchedFilter() = default;
+
+void MatchedFilter::Reset(bool full_reset) {
+  for (auto& f : filters_) {
+    std::fill(f.begin(), f.end(), 0.f);
+  }
+
+  winner_lag_ = absl::nullopt;
+  reported_lag_estimate_ = absl::nullopt;
+  if (pre_echo_config_.mode != 3 || full_reset) {
+    for (auto& e : accumulated_error_) {
+      std::fill(e.begin(), e.end(), 1.0f);
+    }
+    number_pre_echo_updates_ = 0;
+  }
+}
+
+void MatchedFilter::Update(const DownsampledRenderBuffer& render_buffer,
+                           rtc::ArrayView<const float> capture,
+                           bool use_slow_smoothing) {
+  RTC_DCHECK_EQ(sub_block_size_, capture.size());
+  auto& y = capture;
+
+  const float smoothing =
+      use_slow_smoothing ? smoothing_slow_ : smoothing_fast_;
+
+  const float x2_sum_threshold =
+      filters_[0].size() * excitation_limit_ * excitation_limit_;
+
+  // Compute anchor for the matched filter error.
+  float error_sum_anchor = 0.0f;
+  for (size_t k = 0; k < y.size(); ++k) {
+    error_sum_anchor += y[k] * y[k];
+  }
+
+  // Apply all matched filters.
+  float winner_error_sum = error_sum_anchor;
+  winner_lag_ = absl::nullopt;
+  reported_lag_estimate_ = absl::nullopt;
+  size_t alignment_shift = 0;
+  absl::optional<size_t> previous_lag_estimate;
+  const int num_filters = static_cast<int>(filters_.size());
+  int winner_index = -1;
+  for (int n = 0; n < num_filters; ++n) {
+    float error_sum = 0.f;
+    bool filters_updated = false;
+    const bool compute_pre_echo =
+        detect_pre_echo_ && n == last_detected_best_lag_filter_;
+
+    size_t x_start_index =
+        (render_buffer.read + alignment_shift + sub_block_size_ - 1) %
+        render_buffer.buffer.size();
+
+    switch (optimization_) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+      case Aec3Optimization::kSse2:
+        aec3::MatchedFilterCore_SSE2(
+            x_start_index, x2_sum_threshold, smoothing, render_buffer.buffer, y,
+            filters_[n], &filters_updated, &error_sum, compute_pre_echo,
+            instantaneous_accumulated_error_, scratch_memory_);
+        break;
+      case Aec3Optimization::kAvx2:
+        aec3::MatchedFilterCore_AVX2(
+            x_start_index, x2_sum_threshold, smoothing, render_buffer.buffer, y,
+            filters_[n], &filters_updated, &error_sum, compute_pre_echo,
+            instantaneous_accumulated_error_, scratch_memory_);
+        break;
+#endif
+#if defined(WEBRTC_HAS_NEON)
+      case Aec3Optimization::kNeon:
+        aec3::MatchedFilterCore_NEON(
+            x_start_index, x2_sum_threshold, smoothing, render_buffer.buffer, y,
+            filters_[n], &filters_updated, &error_sum, compute_pre_echo,
+            instantaneous_accumulated_error_, scratch_memory_);
+        break;
+#endif
+      default:
+        aec3::MatchedFilterCore(x_start_index, x2_sum_threshold, smoothing,
+                                render_buffer.buffer, y, filters_[n],
+                                &filters_updated, &error_sum, compute_pre_echo,
+                                instantaneous_accumulated_error_);
+    }
+
+    // Estimate the lag in the matched filter as the distance to the portion in
+    // the filter that contributes the most to the matched filter output. This
+    // is detected as the peak of the matched filter.
+    const size_t lag_estimate = aec3::MaxSquarePeakIndex(filters_[n]);
+    const bool reliable =
+        lag_estimate > 2 && lag_estimate < (filters_[n].size() - 10) &&
+        error_sum < matching_filter_threshold_ * error_sum_anchor;
+
+    // Find the best estimate
+    const size_t lag = lag_estimate + alignment_shift;
+    if (filters_updated && reliable && error_sum < winner_error_sum) {
+      winner_error_sum = error_sum;
+      winner_index = n;
+      // In case that 2 matched filters return the same winner candidate
+      // (overlap region), the one with the smaller index is chosen in order
+      // to search for pre-echoes.
+      if (previous_lag_estimate && previous_lag_estimate == lag) {
+        winner_lag_ = previous_lag_estimate;
+        winner_index = n - 1;
+      } else {
+        winner_lag_ = lag;
+      }
+    }
+    previous_lag_estimate = lag;
+    alignment_shift += filter_intra_lag_shift_;
+  }
+
+  if (winner_index != -1) {
+    RTC_DCHECK(winner_lag_.has_value());
+    reported_lag_estimate_ =
+        LagEstimate(winner_lag_.value(), /*pre_echo_lag=*/winner_lag_.value());
+    if (detect_pre_echo_ && last_detected_best_lag_filter_ == winner_index) {
+      const float energy_threshold =
+          pre_echo_config_.mode == 3 ? 1.0f : 30.0f * 30.0f * y.size();
+
+      if (error_sum_anchor > energy_threshold) {
+        const float smooth_constant_increases =
+            pre_echo_config_.mode != 3 ? 0.01f : 0.015f;
+
+        UpdateAccumulatedError(
+            instantaneous_accumulated_error_, accumulated_error_[winner_index],
+            1.0f / error_sum_anchor, smooth_constant_increases);
+        number_pre_echo_updates_++;
+      }
+      if (pre_echo_config_.mode != 3 || number_pre_echo_updates_ >= 50) {
+        reported_lag_estimate_->pre_echo_lag = ComputePreEchoLag(
+            pre_echo_config_, accumulated_error_[winner_index],
+            winner_lag_.value(),
+            winner_index * filter_intra_lag_shift_ /*alignment_shift_winner*/);
+      } else {
+        reported_lag_estimate_->pre_echo_lag = winner_lag_.value();
+      }
+    }
+    last_detected_best_lag_filter_ = winner_index;
+  }
+  if (ApmDataDumper::IsAvailable()) {
+    Dump();
+    data_dumper_->DumpRaw("error_sum_anchor", error_sum_anchor / y.size());
+    data_dumper_->DumpRaw("number_pre_echo_updates", number_pre_echo_updates_);
+    data_dumper_->DumpRaw("filter_smoothing", smoothing);
+  }
+}
+
+void MatchedFilter::LogFilterProperties(int sample_rate_hz,
+                                        size_t shift,
+                                        size_t downsampling_factor) const {
+  size_t alignment_shift = 0;
+  constexpr int kFsBy1000 = 16;
+  for (size_t k = 0; k < filters_.size(); ++k) {
+    int start = static_cast<int>(alignment_shift * downsampling_factor);
+    int end = static_cast<int>((alignment_shift + filters_[k].size()) *
+                               downsampling_factor);
+    RTC_LOG(LS_VERBOSE) << "Filter " << k << ": start: "
+                        << (start - static_cast<int>(shift)) / kFsBy1000
+                        << " ms, end: "
+                        << (end - static_cast<int>(shift)) / kFsBy1000
+                        << " ms.";
+    alignment_shift += filter_intra_lag_shift_;
+  }
+}
+
+void MatchedFilter::Dump() {
+  for (size_t n = 0; n < filters_.size(); ++n) {
+    const size_t lag_estimate = aec3::MaxSquarePeakIndex(filters_[n]);
+    std::string dumper_filter = "aec3_correlator_" + std::to_string(n) + "_h";
+    data_dumper_->DumpRaw(dumper_filter.c_str(), filters_[n]);
+    std::string dumper_lag = "aec3_correlator_lag_" + std::to_string(n);
+    data_dumper_->DumpRaw(dumper_lag.c_str(),
+                          lag_estimate + n * filter_intra_lag_shift_);
+    if (detect_pre_echo_) {
+      std::string dumper_error =
+          "aec3_correlator_error_" + std::to_string(n) + "_h";
+      data_dumper_->DumpRaw(dumper_error.c_str(), accumulated_error_[n]);
+
+      size_t pre_echo_lag =
+          ComputePreEchoLag(pre_echo_config_, accumulated_error_[n],
+                            lag_estimate + n * filter_intra_lag_shift_,
+                            n * filter_intra_lag_shift_);
+      std::string dumper_pre_lag =
+          "aec3_correlator_pre_echo_lag_" + std::to_string(n);
+      data_dumper_->DumpRaw(dumper_pre_lag.c_str(), pre_echo_lag);
+      if (static_cast<int>(n) == last_detected_best_lag_filter_) {
+        data_dumper_->DumpRaw("aec3_pre_echo_delay_winner_inst", pre_echo_lag);
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter.h b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter.h
new file mode 100644
index 0000000000..bb54fba2b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter.h
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+struct DownsampledRenderBuffer;
+
+namespace aec3 {
+
+#if defined(WEBRTC_HAS_NEON)
+
+// Filter core for the matched filter that is optimized for NEON.
+void MatchedFilterCore_NEON(size_t x_start_index,
+                            float x2_sum_threshold,
+                            float smoothing,
+                            rtc::ArrayView<const float> x,
+                            rtc::ArrayView<const float> y,
+                            rtc::ArrayView<float> h,
+                            bool* filters_updated,
+                            float* error_sum,
+                            bool compute_accumulation_error,
+                            rtc::ArrayView<float> accumulated_error,
+                            rtc::ArrayView<float> scratch_memory);
+
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+
+// Filter core for the matched filter that is optimized for SSE2.
+void MatchedFilterCore_SSE2(size_t x_start_index,
+                            float x2_sum_threshold,
+                            float smoothing,
+                            rtc::ArrayView<const float> x,
+                            rtc::ArrayView<const float> y,
+                            rtc::ArrayView<float> h,
+                            bool* filters_updated,
+                            float* error_sum,
+                            bool compute_accumulated_error,
+                            rtc::ArrayView<float> accumulated_error,
+                            rtc::ArrayView<float> scratch_memory);
+
+// Filter core for the matched filter that is optimized for AVX2.
+void MatchedFilterCore_AVX2(size_t x_start_index,
+                            float x2_sum_threshold,
+                            float smoothing,
+                            rtc::ArrayView<const float> x,
+                            rtc::ArrayView<const float> y,
+                            rtc::ArrayView<float> h,
+                            bool* filters_updated,
+                            float* error_sum,
+                            bool compute_accumulated_error,
+                            rtc::ArrayView<float> accumulated_error,
+                            rtc::ArrayView<float> scratch_memory);
+
+#endif
+
+// Filter core for the matched filter.
+void MatchedFilterCore(size_t x_start_index,
+                       float x2_sum_threshold,
+                       float smoothing,
+                       rtc::ArrayView<const float> x,
+                       rtc::ArrayView<const float> y,
+                       rtc::ArrayView<float> h,
+                       bool* filters_updated,
+                       float* error_sum,
+                       bool compute_accumulation_error,
+                       rtc::ArrayView<float> accumulated_error);
+
+// Find largest peak of squared values in array.
+size_t MaxSquarePeakIndex(rtc::ArrayView<const float> h);
+
+}  // namespace aec3
+
+// Produces recursively updated cross-correlation estimates for several signal
+// shifts where the intra-shift spacing is uniform.
+class MatchedFilter {
+ public:
+  // Stores properties for the lag estimate corresponding to a particular signal
+  // shift.
+  struct LagEstimate {
+    LagEstimate() = default;
+    LagEstimate(size_t lag, size_t pre_echo_lag)
+        : lag(lag), pre_echo_lag(pre_echo_lag) {}
+    size_t lag = 0;
+    size_t pre_echo_lag = 0;
+  };
+
+  struct PreEchoConfiguration {
+    const float threshold;
+    const int mode;
+  };
+
+  MatchedFilter(ApmDataDumper* data_dumper,
+                Aec3Optimization optimization,
+                size_t sub_block_size,
+                size_t window_size_sub_blocks,
+                int num_matched_filters,
+                size_t alignment_shift_sub_blocks,
+                float excitation_limit,
+                float smoothing_fast,
+                float smoothing_slow,
+                float matching_filter_threshold,
+                bool detect_pre_echo);
+
+  MatchedFilter() = delete;
+  MatchedFilter(const MatchedFilter&) = delete;
+  MatchedFilter& operator=(const MatchedFilter&) = delete;
+
+  ~MatchedFilter();
+
+  // Updates the correlation with the values in the capture buffer.
+  void Update(const DownsampledRenderBuffer& render_buffer,
+              rtc::ArrayView<const float> capture,
+              bool use_slow_smoothing);
+
+  // Resets the matched filter.
+  void Reset(bool full_reset);
+
+  // Returns the current lag estimates.
+  absl::optional<const MatchedFilter::LagEstimate> GetBestLagEstimate() const {
+    return reported_lag_estimate_;
+  }
+
+  // Returns the maximum filter lag.
+  size_t GetMaxFilterLag() const {
+    return filters_.size() * filter_intra_lag_shift_ + filters_[0].size();
+  }
+
+  // Log matched filter properties.
+  void LogFilterProperties(int sample_rate_hz,
+                           size_t shift,
+                           size_t downsampling_factor) const;
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(MatchedFilterFieldTrialTest,
+                           PreEchoConfigurationTest);
+  FRIEND_TEST_ALL_PREFIXES(MatchedFilterFieldTrialTest,
+                           WrongPreEchoConfigurationTest);
+
+  // Only for testing. Gets the pre echo detection configuration.
+  const PreEchoConfiguration& GetPreEchoConfiguration() const {
+    return pre_echo_config_;
+  }
+  void Dump();
+
+  ApmDataDumper* const data_dumper_;
+  const Aec3Optimization optimization_;
+  const size_t sub_block_size_;
+  const size_t filter_intra_lag_shift_;
+  std::vector<std::vector<float>> filters_;
+  std::vector<std::vector<float>> accumulated_error_;
+  std::vector<float> instantaneous_accumulated_error_;
+  std::vector<float> scratch_memory_;
+  absl::optional<MatchedFilter::LagEstimate> reported_lag_estimate_;
+  absl::optional<size_t> winner_lag_;
+  int last_detected_best_lag_filter_ = -1;
+  std::vector<size_t> filters_offsets_;
+  int number_pre_echo_updates_ = 0;
+  const float excitation_limit_;
+  const float smoothing_fast_;
+  const float smoothing_slow_;
+  const float matching_filter_threshold_;
+  const bool detect_pre_echo_;
+  const PreEchoConfiguration pre_echo_config_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_avx2.cc b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_avx2.cc
new file mode 100644
index 0000000000..8c2ffcbd1e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_avx2.cc
@@ -0,0 +1,261 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <immintrin.h>
+
+#include "modules/audio_processing/aec3/matched_filter.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace aec3 {
+
+// Let ha denote the horizontal of a, and hb the horizontal sum of b
+// returns [ha, hb, ha, hb]
+inline __m128 hsum_ab(__m256 a, __m256 b) {
+  __m256 s_256 = _mm256_hadd_ps(a, b);
+  const __m256i mask = _mm256_set_epi32(7, 6, 3, 2, 5, 4, 1, 0);
+  s_256 = _mm256_permutevar8x32_ps(s_256, mask);
+  __m128 s = _mm_hadd_ps(_mm256_extractf128_ps(s_256, 0),
+                         _mm256_extractf128_ps(s_256, 1));
+  s = _mm_hadd_ps(s, s);
+  return s;
+}
+
+void MatchedFilterCore_AccumulatedError_AVX2(
+    size_t x_start_index,
+    float x2_sum_threshold,
+    float smoothing,
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<const float> y,
+    rtc::ArrayView<float> h,
+    bool* filters_updated,
+    float* error_sum,
+    rtc::ArrayView<float> accumulated_error,
+    rtc::ArrayView<float> scratch_memory) {
+  const int h_size = static_cast<int>(h.size());
+  const int x_size = static_cast<int>(x.size());
+  RTC_DCHECK_EQ(0, h_size % 16);
+  std::fill(accumulated_error.begin(), accumulated_error.end(), 0.0f);
+
+  // Process for all samples in the sub-block.
+  for (size_t i = 0; i < y.size(); ++i) {
+    // Apply the matched filter as filter * x, and compute x * x.
+    RTC_DCHECK_GT(x_size, x_start_index);
+    const int chunk1 =
+        std::min(h_size, static_cast<int>(x_size - x_start_index));
+    if (chunk1 != h_size) {
+      const int chunk2 = h_size - chunk1;
+      std::copy(x.begin() + x_start_index, x.end(), scratch_memory.begin());
+      std::copy(x.begin(), x.begin() + chunk2, scratch_memory.begin() + chunk1);
+    }
+    const float* x_p =
+        chunk1 != h_size ? scratch_memory.data() : &x[x_start_index];
+    const float* h_p = &h[0];
+    float* a_p = &accumulated_error[0];
+    __m256 s_inst_hadd_256;
+    __m256 s_inst_256;
+    __m256 s_inst_256_8;
+    __m256 x2_sum_256 = _mm256_set1_ps(0);
+    __m256 x2_sum_256_8 = _mm256_set1_ps(0);
+    __m128 e_128;
+    float x2_sum = 0.0f;
+    float s_acum = 0;
+    const int limit_by_16 = h_size >> 4;
+    for (int k = limit_by_16; k > 0; --k, h_p += 16, x_p += 16, a_p += 4) {
+      // Load the data into 256 bit vectors.
+      __m256 x_k = _mm256_loadu_ps(x_p);
+      __m256 h_k = _mm256_loadu_ps(h_p);
+      __m256 x_k_8 = _mm256_loadu_ps(x_p + 8);
+      __m256 h_k_8 = _mm256_loadu_ps(h_p + 8);
+      // Compute and accumulate x * x and h * x.
+      x2_sum_256 = _mm256_fmadd_ps(x_k, x_k, x2_sum_256);
+      x2_sum_256_8 = _mm256_fmadd_ps(x_k_8, x_k_8, x2_sum_256_8);
+      s_inst_256 = _mm256_mul_ps(h_k, x_k);
+      s_inst_256_8 = _mm256_mul_ps(h_k_8, x_k_8);
+      s_inst_hadd_256 = _mm256_hadd_ps(s_inst_256, s_inst_256_8);
+      s_inst_hadd_256 = _mm256_hadd_ps(s_inst_hadd_256, s_inst_hadd_256);
+      s_acum += s_inst_hadd_256[0];
+      e_128[0] = s_acum - y[i];
+      s_acum += s_inst_hadd_256[4];
+      e_128[1] = s_acum - y[i];
+      s_acum += s_inst_hadd_256[1];
+      e_128[2] = s_acum - y[i];
+      s_acum += s_inst_hadd_256[5];
+      e_128[3] = s_acum - y[i];
+
+      __m128 accumulated_error = _mm_load_ps(a_p);
+      accumulated_error = _mm_fmadd_ps(e_128, e_128, accumulated_error);
+      _mm_storeu_ps(a_p, accumulated_error);
+    }
+    // Sum components together.
+    x2_sum_256 = _mm256_add_ps(x2_sum_256, x2_sum_256_8);
+    __m128 x2_sum_128 = _mm_add_ps(_mm256_extractf128_ps(x2_sum_256, 0),
+                                   _mm256_extractf128_ps(x2_sum_256, 1));
+    // Combine the accumulated vector and scalar values.
+    float* v = reinterpret_cast<float*>(&x2_sum_128);
+    x2_sum += v[0] + v[1] + v[2] + v[3];
+
+    // Compute the matched filter error.
+    float e = y[i] - s_acum;
+    const bool saturation = y[i] >= 32000.f || y[i] <= -32000.f;
+    (*error_sum) += e * e;
+
+    // Update the matched filter estimate in an NLMS manner.
+    if (x2_sum > x2_sum_threshold && !saturation) {
+      RTC_DCHECK_LT(0.f, x2_sum);
+      const float alpha = smoothing * e / x2_sum;
+      const __m256 alpha_256 = _mm256_set1_ps(alpha);
+
+      // filter = filter + smoothing * (y - filter * x) * x / x * x.
+      float* h_p = &h[0];
+      const float* x_p =
+          chunk1 != h_size ? scratch_memory.data() : &x[x_start_index];
+      // Perform 256 bit vector operations.
+      const int limit_by_8 = h_size >> 3;
+      for (int k = limit_by_8; k > 0; --k, h_p += 8, x_p += 8) {
+        // Load the data into 256 bit vectors.
+        __m256 h_k = _mm256_loadu_ps(h_p);
+        __m256 x_k = _mm256_loadu_ps(x_p);
+        // Compute h = h + alpha * x.
+        h_k = _mm256_fmadd_ps(x_k, alpha_256, h_k);
+
+        // Store the result.
+        _mm256_storeu_ps(h_p, h_k);
+      }
+      *filters_updated = true;
+    }
+
+    x_start_index = x_start_index > 0 ? x_start_index - 1 : x_size - 1;
+  }
+}
+
+void MatchedFilterCore_AVX2(size_t x_start_index,
+                            float x2_sum_threshold,
+                            float smoothing,
+                            rtc::ArrayView<const float> x,
+                            rtc::ArrayView<const float> y,
+                            rtc::ArrayView<float> h,
+                            bool* filters_updated,
+                            float* error_sum,
+                            bool compute_accumulated_error,
+                            rtc::ArrayView<float> accumulated_error,
+                            rtc::ArrayView<float> scratch_memory) {
+  if (compute_accumulated_error) {
+    return MatchedFilterCore_AccumulatedError_AVX2(
+        x_start_index, x2_sum_threshold, smoothing, x, y, h, filters_updated,
+        error_sum, accumulated_error, scratch_memory);
+  }
+  const int h_size = static_cast<int>(h.size());
+  const int x_size = static_cast<int>(x.size());
+  RTC_DCHECK_EQ(0, h_size % 8);
+
+  // Process for all samples in the sub-block.
+  for (size_t i = 0; i < y.size(); ++i) {
+    // Apply the matched filter as filter * x, and compute x * x.
+
+    RTC_DCHECK_GT(x_size, x_start_index);
+    const float* x_p = &x[x_start_index];
+    const float* h_p = &h[0];
+
+    // Initialize values for the accumulation.
+    __m256 s_256 = _mm256_set1_ps(0);
+    __m256 s_256_8 = _mm256_set1_ps(0);
+    __m256 x2_sum_256 = _mm256_set1_ps(0);
+    __m256 x2_sum_256_8 = _mm256_set1_ps(0);
+    float x2_sum = 0.f;
+    float s = 0;
+
+    // Compute loop chunk sizes until, and after, the wraparound of the circular
+    // buffer for x.
+    const int chunk1 =
+        std::min(h_size, static_cast<int>(x_size - x_start_index));
+
+    // Perform the loop in two chunks.
+    const int chunk2 = h_size - chunk1;
+    for (int limit : {chunk1, chunk2}) {
+      // Perform 256 bit vector operations.
+      const int limit_by_16 = limit >> 4;
+      for (int k = limit_by_16; k > 0; --k, h_p += 16, x_p += 16) {
+        // Load the data into 256 bit vectors.
+        __m256 x_k = _mm256_loadu_ps(x_p);
+        __m256 h_k = _mm256_loadu_ps(h_p);
+        __m256 x_k_8 = _mm256_loadu_ps(x_p + 8);
+        __m256 h_k_8 = _mm256_loadu_ps(h_p + 8);
+        // Compute and accumulate x * x and h * x.
+        x2_sum_256 = _mm256_fmadd_ps(x_k, x_k, x2_sum_256);
+        x2_sum_256_8 = _mm256_fmadd_ps(x_k_8, x_k_8, x2_sum_256_8);
+        s_256 = _mm256_fmadd_ps(h_k, x_k, s_256);
+        s_256_8 = _mm256_fmadd_ps(h_k_8, x_k_8, s_256_8);
+      }
+
+      // Perform non-vector operations for any remaining items.
+      for (int k = limit - limit_by_16 * 16; k > 0; --k, ++h_p, ++x_p) {
+        const float x_k = *x_p;
+        x2_sum += x_k * x_k;
+        s += *h_p * x_k;
+      }
+
+      x_p = &x[0];
+    }
+
+    // Sum components together.
+    x2_sum_256 = _mm256_add_ps(x2_sum_256, x2_sum_256_8);
+    s_256 = _mm256_add_ps(s_256, s_256_8);
+    __m128 sum = hsum_ab(x2_sum_256, s_256);
+    x2_sum += sum[0];
+    s += sum[1];
+
+    // Compute the matched filter error.
+    float e = y[i] - s;
+    const bool saturation = y[i] >= 32000.f || y[i] <= -32000.f;
+    (*error_sum) += e * e;
+
+    // Update the matched filter estimate in an NLMS manner.
+    if (x2_sum > x2_sum_threshold && !saturation) {
+      RTC_DCHECK_LT(0.f, x2_sum);
+      const float alpha = smoothing * e / x2_sum;
+      const __m256 alpha_256 = _mm256_set1_ps(alpha);
+
+      // filter = filter + smoothing * (y - filter * x) * x / x * x.
+      float* h_p = &h[0];
+      x_p = &x[x_start_index];
+
+      // Perform the loop in two chunks.
+      for (int limit : {chunk1, chunk2}) {
+        // Perform 256 bit vector operations.
+        const int limit_by_8 = limit >> 3;
+        for (int k = limit_by_8; k > 0; --k, h_p += 8, x_p += 8) {
+          // Load the data into 256 bit vectors.
+          __m256 h_k = _mm256_loadu_ps(h_p);
+          __m256 x_k = _mm256_loadu_ps(x_p);
+          // Compute h = h + alpha * x.
+          h_k = _mm256_fmadd_ps(x_k, alpha_256, h_k);
+
+          // Store the result.
+          _mm256_storeu_ps(h_p, h_k);
+        }
+
+        // Perform non-vector operations for any remaining items.
+        for (int k = limit - limit_by_8 * 8; k > 0; --k, ++h_p, ++x_p) {
+          *h_p += alpha * *x_p;
+        }
+
+        x_p = &x[0];
+      }
+
+      *filters_updated = true;
+    }
+
+    x_start_index = x_start_index > 0 ? x_start_index - 1 : x_size - 1;
+  }
+}
+
+}  // namespace aec3
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_gn/moz.build
new file mode 100644
index 0000000000..bae4fa2972
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("matched_filter_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
new file mode 100644
index 0000000000..bea7868a91
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
@@ -0,0 +1,166 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
+
+#include <algorithm>
+#include <iterator>
+
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+constexpr int kPreEchoHistogramDataNotUpdated = -1;
+
+int GetDownSamplingBlockSizeLog2(int down_sampling_factor) {
+  int down_sampling_factor_log2 = 0;
+  down_sampling_factor >>= 1;
+  while (down_sampling_factor > 0) {
+    down_sampling_factor_log2++;
+    down_sampling_factor >>= 1;
+  }
+  return static_cast<int>(kBlockSizeLog2) > down_sampling_factor_log2
+             ? static_cast<int>(kBlockSizeLog2) - down_sampling_factor_log2
+             : 0;
+}
+}  // namespace
+
+MatchedFilterLagAggregator::MatchedFilterLagAggregator(
+    ApmDataDumper* data_dumper,
+    size_t max_filter_lag,
+    const EchoCanceller3Config::Delay& delay_config)
+    : data_dumper_(data_dumper),
+      thresholds_(delay_config.delay_selection_thresholds),
+      headroom_(static_cast<int>(delay_config.delay_headroom_samples /
+                                 delay_config.down_sampling_factor)),
+      highest_peak_aggregator_(max_filter_lag) {
+  if (delay_config.detect_pre_echo) {
+    pre_echo_lag_aggregator_ = std::make_unique<PreEchoLagAggregator>(
+        max_filter_lag, delay_config.down_sampling_factor);
+  }
+  RTC_DCHECK(data_dumper);
+  RTC_DCHECK_LE(thresholds_.initial, thresholds_.converged);
+}
+
+MatchedFilterLagAggregator::~MatchedFilterLagAggregator() = default;
+
+void MatchedFilterLagAggregator::Reset(bool hard_reset) {
+  highest_peak_aggregator_.Reset();
+  if (pre_echo_lag_aggregator_ != nullptr) {
+    pre_echo_lag_aggregator_->Reset();
+  }
+  if (hard_reset) {
+    significant_candidate_found_ = false;
+  }
+}
+
+absl::optional<DelayEstimate> MatchedFilterLagAggregator::Aggregate(
+    const absl::optional<const MatchedFilter::LagEstimate>& lag_estimate) {
+  if (lag_estimate && pre_echo_lag_aggregator_) {
+    pre_echo_lag_aggregator_->Dump(data_dumper_);
+    pre_echo_lag_aggregator_->Aggregate(
+        std::max(0, static_cast<int>(lag_estimate->pre_echo_lag) - headroom_));
+  }
+
+  if (lag_estimate) {
+    highest_peak_aggregator_.Aggregate(
+        std::max(0, static_cast<int>(lag_estimate->lag) - headroom_));
+    rtc::ArrayView<const int> histogram = highest_peak_aggregator_.histogram();
+    int candidate = highest_peak_aggregator_.candidate();
+    significant_candidate_found_ = significant_candidate_found_ ||
+                                   histogram[candidate] > thresholds_.converged;
+    if (histogram[candidate] > thresholds_.converged ||
+        (histogram[candidate] > thresholds_.initial &&
+         !significant_candidate_found_)) {
+      DelayEstimate::Quality quality = significant_candidate_found_
+                                           ? DelayEstimate::Quality::kRefined
+                                           : DelayEstimate::Quality::kCoarse;
+      int reported_delay = pre_echo_lag_aggregator_ != nullptr
+                               ? pre_echo_lag_aggregator_->pre_echo_candidate()
+                               : candidate;
+      return DelayEstimate(quality, reported_delay);
+    }
+  }
+
+  return absl::nullopt;
+}
+
+MatchedFilterLagAggregator::HighestPeakAggregator::HighestPeakAggregator(
+    size_t max_filter_lag)
+    : histogram_(max_filter_lag + 1, 0) {
+  histogram_data_.fill(0);
+}
+
+void MatchedFilterLagAggregator::HighestPeakAggregator::Reset() {
+  std::fill(histogram_.begin(), histogram_.end(), 0);
+  histogram_data_.fill(0);
+  histogram_data_index_ = 0;
+}
+
+void MatchedFilterLagAggregator::HighestPeakAggregator::Aggregate(int lag) {
+  RTC_DCHECK_GT(histogram_.size(), histogram_data_[histogram_data_index_]);
+  RTC_DCHECK_LE(0, histogram_data_[histogram_data_index_]);
+  --histogram_[histogram_data_[histogram_data_index_]];
+  histogram_data_[histogram_data_index_] = lag;
+  RTC_DCHECK_GT(histogram_.size(), histogram_data_[histogram_data_index_]);
+  RTC_DCHECK_LE(0, histogram_data_[histogram_data_index_]);
+  ++histogram_[histogram_data_[histogram_data_index_]];
+  histogram_data_index_ = (histogram_data_index_ + 1) % histogram_data_.size();
+  candidate_ =
+      std::distance(histogram_.begin(),
+                    std::max_element(histogram_.begin(), histogram_.end()));
+}
+
+MatchedFilterLagAggregator::PreEchoLagAggregator::PreEchoLagAggregator(
+    size_t max_filter_lag,
+    size_t down_sampling_factor)
+    : block_size_log2_(GetDownSamplingBlockSizeLog2(down_sampling_factor)),
+      histogram_(
+          ((max_filter_lag + 1) * down_sampling_factor) >> kBlockSizeLog2,
+          0) {
+  Reset();
+}
+
+void MatchedFilterLagAggregator::PreEchoLagAggregator::Reset() {
+  std::fill(histogram_.begin(), histogram_.end(), 0);
+  histogram_data_.fill(kPreEchoHistogramDataNotUpdated);
+  histogram_data_index_ = 0;
+  pre_echo_candidate_ = 0;
+}
+
+void MatchedFilterLagAggregator::PreEchoLagAggregator::Aggregate(
+    int pre_echo_lag) {
+  int pre_echo_block_size = pre_echo_lag >> block_size_log2_;
+  RTC_DCHECK(pre_echo_block_size >= 0 &&
+             pre_echo_block_size < static_cast<int>(histogram_.size()));
+  pre_echo_block_size =
+      rtc::SafeClamp(pre_echo_block_size, 0, histogram_.size() - 1);
+  // Remove the oldest point from the `histogram_`, it ignores the initial
+  // points where no updates have been done to the `histogram_data_` array.
+  if (histogram_data_[histogram_data_index_] !=
+      kPreEchoHistogramDataNotUpdated) {
+    --histogram_[histogram_data_[histogram_data_index_]];
+  }
+  histogram_data_[histogram_data_index_] = pre_echo_block_size;
+  ++histogram_[histogram_data_[histogram_data_index_]];
+  histogram_data_index_ = (histogram_data_index_ + 1) % histogram_data_.size();
+  int pre_echo_candidate_block_size =
+      std::distance(histogram_.begin(),
+                    std::max_element(histogram_.begin(), histogram_.end()));
+  pre_echo_candidate_ = (pre_echo_candidate_block_size << block_size_log2_);
+}
+
+void MatchedFilterLagAggregator::PreEchoLagAggregator::Dump(
+    ApmDataDumper* const data_dumper) {
+  data_dumper->DumpRaw("aec3_pre_echo_delay_candidate", pre_echo_candidate_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
new file mode 100644
index 0000000000..c0598bf226
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
@@ -0,0 +1,97 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/delay_estimate.h"
+#include "modules/audio_processing/aec3/matched_filter.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+// Aggregates lag estimates produced by the MatchedFilter class into a single
+// reliable combined lag estimate.
+class MatchedFilterLagAggregator {
+ public:
+  MatchedFilterLagAggregator(ApmDataDumper* data_dumper,
+                             size_t max_filter_lag,
+                             const EchoCanceller3Config::Delay& delay_config);
+
+  MatchedFilterLagAggregator() = delete;
+  MatchedFilterLagAggregator(const MatchedFilterLagAggregator&) = delete;
+  MatchedFilterLagAggregator& operator=(const MatchedFilterLagAggregator&) =
+      delete;
+
+  ~MatchedFilterLagAggregator();
+
+  // Resets the aggregator.
+  void Reset(bool hard_reset);
+
+  // Aggregates the provided lag estimates.
+  absl::optional<DelayEstimate> Aggregate(
+      const absl::optional<const MatchedFilter::LagEstimate>& lag_estimate);
+
+  // Returns whether a reliable delay estimate has been found.
+  bool ReliableDelayFound() const { return significant_candidate_found_; }
+
+  // Returns the delay candidate that is computed by looking at the highest peak
+  // on the matched filters.
+  int GetDelayAtHighestPeak() const {
+    return highest_peak_aggregator_.candidate();
+  }
+
+ private:
+  class PreEchoLagAggregator {
+   public:
+    PreEchoLagAggregator(size_t max_filter_lag, size_t down_sampling_factor);
+    void Reset();
+    void Aggregate(int pre_echo_lag);
+    int pre_echo_candidate() const { return pre_echo_candidate_; }
+    void Dump(ApmDataDumper* const data_dumper);
+
+   private:
+    const int block_size_log2_;
+    std::array<int, 250> histogram_data_;
+    std::vector<int> histogram_;
+    int histogram_data_index_ = 0;
+    int pre_echo_candidate_ = 0;
+  };
+
+  class HighestPeakAggregator {
+   public:
+    explicit HighestPeakAggregator(size_t max_filter_lag);
+    void Reset();
+    void Aggregate(int lag);
+    int candidate() const { return candidate_; }
+    rtc::ArrayView<const int> histogram() const { return histogram_; }
+
+   private:
+    std::vector<int> histogram_;
+    std::array<int, 250> histogram_data_;
+    int histogram_data_index_ = 0;
+    int candidate_ = -1;
+  };
+
+  ApmDataDumper* const data_dumper_;
+  bool significant_candidate_found_ = false;
+  const EchoCanceller3Config::Delay::DelaySelectionThresholds thresholds_;
+  const int headroom_;
+  HighestPeakAggregator highest_peak_aggregator_;
+  std::unique_ptr<PreEchoLagAggregator> pre_echo_lag_aggregator_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_LAG_AGGREGATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
new file mode 100644
index 0000000000..6804102584
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
+
+#include <sstream>
+#include <string>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kNumLagsBeforeDetection = 26;
+
+}  // namespace
+
+// Verifies that varying lag estimates causes lag estimates to not be deemed
+// reliable.
+TEST(MatchedFilterLagAggregator,
+     LagEstimateInvarianceRequiredForAggregatedLag) {
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  MatchedFilterLagAggregator aggregator(&data_dumper, /*max_filter_lag=*/100,
+                                        config.delay);
+
+  absl::optional<DelayEstimate> aggregated_lag;
+  for (size_t k = 0; k < kNumLagsBeforeDetection; ++k) {
+    aggregated_lag = aggregator.Aggregate(
+        MatchedFilter::LagEstimate(/*lag=*/10, /*pre_echo_lag=*/10));
+  }
+  EXPECT_TRUE(aggregated_lag);
+
+  for (size_t k = 0; k < kNumLagsBeforeDetection * 100; ++k) {
+    aggregated_lag = aggregator.Aggregate(
+        MatchedFilter::LagEstimate(/*lag=*/k % 100, /*pre_echo_lag=*/k % 100));
+  }
+  EXPECT_FALSE(aggregated_lag);
+
+  for (size_t k = 0; k < kNumLagsBeforeDetection * 100; ++k) {
+    aggregated_lag = aggregator.Aggregate(
+        MatchedFilter::LagEstimate(/*lag=*/k % 100, /*pre_echo_lag=*/k % 100));
+    EXPECT_FALSE(aggregated_lag);
+  }
+}
+
+// Verifies that lag estimate updates are required to produce an updated lag
+// aggregate.
+TEST(MatchedFilterLagAggregator,
+     DISABLED_LagEstimateUpdatesRequiredForAggregatedLag) {
+  constexpr size_t kLag = 5;
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  MatchedFilterLagAggregator aggregator(&data_dumper, /*max_filter_lag=*/kLag,
+                                        config.delay);
+  for (size_t k = 0; k < kNumLagsBeforeDetection * 10; ++k) {
+    absl::optional<DelayEstimate> aggregated_lag = aggregator.Aggregate(
+        MatchedFilter::LagEstimate(/*lag=*/kLag, /*pre_echo_lag=*/kLag));
+    EXPECT_FALSE(aggregated_lag);
+    EXPECT_EQ(kLag, aggregated_lag->delay);
+  }
+}
+
+// Verifies that an aggregated lag is persistent if the lag estimates do not
+// change and that an aggregated lag is not produced without gaining lag
+// estimate confidence.
+TEST(MatchedFilterLagAggregator, DISABLED_PersistentAggregatedLag) {
+  constexpr size_t kLag1 = 5;
+  constexpr size_t kLag2 = 10;
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  std::vector<MatchedFilter::LagEstimate> lag_estimates(1);
+  MatchedFilterLagAggregator aggregator(&data_dumper, std::max(kLag1, kLag2),
+                                        config.delay);
+  absl::optional<DelayEstimate> aggregated_lag;
+  for (size_t k = 0; k < kNumLagsBeforeDetection; ++k) {
+    aggregated_lag = aggregator.Aggregate(
+        MatchedFilter::LagEstimate(/*lag=*/kLag1, /*pre_echo_lag=*/kLag1));
+  }
+  EXPECT_TRUE(aggregated_lag);
+  EXPECT_EQ(kLag1, aggregated_lag->delay);
+
+  for (size_t k = 0; k < kNumLagsBeforeDetection * 40; ++k) {
+    aggregated_lag = aggregator.Aggregate(
+        MatchedFilter::LagEstimate(/*lag=*/kLag2, /*pre_echo_lag=*/kLag2));
+    EXPECT_TRUE(aggregated_lag);
+    EXPECT_EQ(kLag1, aggregated_lag->delay);
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for non-null data dumper.
+TEST(MatchedFilterLagAggregatorDeathTest, NullDataDumper) {
+  EchoCanceller3Config config;
+  EXPECT_DEATH(MatchedFilterLagAggregator(nullptr, 10, config.delay), "");
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_unittest.cc
new file mode 100644
index 0000000000..0a04c7809c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/matched_filter_unittest.cc
@@ -0,0 +1,612 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/matched_filter.h"
+
+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+#include <algorithm>
+#include <string>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/decimator.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace aec3 {
+namespace {
+
+std::string ProduceDebugText(size_t delay, size_t down_sampling_factor) {
+  rtc::StringBuilder ss;
+  ss << "Delay: " << delay;
+  ss << ", Down sampling factor: " << down_sampling_factor;
+  return ss.Release();
+}
+
+constexpr size_t kNumMatchedFilters = 10;
+constexpr size_t kDownSamplingFactors[] = {2, 4, 8};
+constexpr size_t kWindowSizeSubBlocks = 32;
+constexpr size_t kAlignmentShiftSubBlocks = kWindowSizeSubBlocks * 3 / 4;
+
+}  // namespace
+
+class MatchedFilterTest : public ::testing::TestWithParam<bool> {};
+
+#if defined(WEBRTC_HAS_NEON)
+// Verifies that the optimized methods for NEON are similar to their reference
+// counterparts.
+TEST_P(MatchedFilterTest, TestNeonOptimizations) {
+  Random random_generator(42U);
+  constexpr float kSmoothing = 0.7f;
+  const bool kComputeAccumulatederror = GetParam();
+  for (auto down_sampling_factor : kDownSamplingFactors) {
+    const size_t sub_block_size = kBlockSize / down_sampling_factor;
+
+    std::vector<float> x(2000);
+    RandomizeSampleVector(&random_generator, x);
+    std::vector<float> y(sub_block_size);
+    std::vector<float> h_NEON(512);
+    std::vector<float> h(512);
+    std::vector<float> accumulated_error(512);
+    std::vector<float> accumulated_error_NEON(512);
+    std::vector<float> scratch_memory(512);
+
+    int x_index = 0;
+    for (int k = 0; k < 1000; ++k) {
+      RandomizeSampleVector(&random_generator, y);
+
+      bool filters_updated = false;
+      float error_sum = 0.f;
+      bool filters_updated_NEON = false;
+      float error_sum_NEON = 0.f;
+
+      MatchedFilterCore_NEON(x_index, h.size() * 150.f * 150.f, kSmoothing, x,
+                             y, h_NEON, &filters_updated_NEON, &error_sum_NEON,
+                             kComputeAccumulatederror, accumulated_error_NEON,
+                             scratch_memory);
+
+      MatchedFilterCore(x_index, h.size() * 150.f * 150.f, kSmoothing, x, y, h,
+                        &filters_updated, &error_sum, kComputeAccumulatederror,
+                        accumulated_error);
+
+      EXPECT_EQ(filters_updated, filters_updated_NEON);
+      EXPECT_NEAR(error_sum, error_sum_NEON, error_sum / 100000.f);
+
+      for (size_t j = 0; j < h.size(); ++j) {
+        EXPECT_NEAR(h[j], h_NEON[j], 0.00001f);
+      }
+
+      if (kComputeAccumulatederror) {
+        for (size_t j = 0; j < accumulated_error.size(); ++j) {
+          float difference =
+              std::abs(accumulated_error[j] - accumulated_error_NEON[j]);
+          float relative_difference = accumulated_error[j] > 0
+                                          ? difference / accumulated_error[j]
+                                          : difference;
+          EXPECT_NEAR(relative_difference, 0.0f, 0.02f);
+        }
+      }
+
+      x_index = (x_index + sub_block_size) % x.size();
+    }
+  }
+}
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Verifies that the optimized methods for SSE2 are bitexact to their reference
+// counterparts.
+TEST_P(MatchedFilterTest, TestSse2Optimizations) {
+  const bool kComputeAccumulatederror = GetParam();
+  bool use_sse2 = (GetCPUInfo(kSSE2) != 0);
+  if (use_sse2) {
+    Random random_generator(42U);
+    constexpr float kSmoothing = 0.7f;
+    for (auto down_sampling_factor : kDownSamplingFactors) {
+      const size_t sub_block_size = kBlockSize / down_sampling_factor;
+      std::vector<float> x(2000);
+      RandomizeSampleVector(&random_generator, x);
+      std::vector<float> y(sub_block_size);
+      std::vector<float> h_SSE2(512);
+      std::vector<float> h(512);
+      std::vector<float> accumulated_error(512 / 4);
+      std::vector<float> accumulated_error_SSE2(512 / 4);
+      std::vector<float> scratch_memory(512);
+      int x_index = 0;
+      for (int k = 0; k < 1000; ++k) {
+        RandomizeSampleVector(&random_generator, y);
+
+        bool filters_updated = false;
+        float error_sum = 0.f;
+        bool filters_updated_SSE2 = false;
+        float error_sum_SSE2 = 0.f;
+
+        MatchedFilterCore_SSE2(x_index, h.size() * 150.f * 150.f, kSmoothing, x,
+                               y, h_SSE2, &filters_updated_SSE2,
+                               &error_sum_SSE2, kComputeAccumulatederror,
+                               accumulated_error_SSE2, scratch_memory);
+
+        MatchedFilterCore(x_index, h.size() * 150.f * 150.f, kSmoothing, x, y,
+                          h, &filters_updated, &error_sum,
+                          kComputeAccumulatederror, accumulated_error);
+
+        EXPECT_EQ(filters_updated, filters_updated_SSE2);
+        EXPECT_NEAR(error_sum, error_sum_SSE2, error_sum / 100000.f);
+
+        for (size_t j = 0; j < h.size(); ++j) {
+          EXPECT_NEAR(h[j], h_SSE2[j], 0.00001f);
+        }
+
+        for (size_t j = 0; j < accumulated_error.size(); ++j) {
+          float difference =
+              std::abs(accumulated_error[j] - accumulated_error_SSE2[j]);
+          float relative_difference = accumulated_error[j] > 0
+                                          ? difference / accumulated_error[j]
+                                          : difference;
+          EXPECT_NEAR(relative_difference, 0.0f, 0.00001f);
+        }
+
+        x_index = (x_index + sub_block_size) % x.size();
+      }
+    }
+  }
+}
+
+TEST_P(MatchedFilterTest, TestAvx2Optimizations) {
+  bool use_avx2 = (GetCPUInfo(kAVX2) != 0);
+  const bool kComputeAccumulatederror = GetParam();
+  if (use_avx2) {
+    Random random_generator(42U);
+    constexpr float kSmoothing = 0.7f;
+    for (auto down_sampling_factor : kDownSamplingFactors) {
+      const size_t sub_block_size = kBlockSize / down_sampling_factor;
+      std::vector<float> x(2000);
+      RandomizeSampleVector(&random_generator, x);
+      std::vector<float> y(sub_block_size);
+      std::vector<float> h_AVX2(512);
+      std::vector<float> h(512);
+      std::vector<float> accumulated_error(512 / 4);
+      std::vector<float> accumulated_error_AVX2(512 / 4);
+      std::vector<float> scratch_memory(512);
+      int x_index = 0;
+      for (int k = 0; k < 1000; ++k) {
+        RandomizeSampleVector(&random_generator, y);
+        bool filters_updated = false;
+        float error_sum = 0.f;
+        bool filters_updated_AVX2 = false;
+        float error_sum_AVX2 = 0.f;
+        MatchedFilterCore_AVX2(x_index, h.size() * 150.f * 150.f, kSmoothing, x,
+                               y, h_AVX2, &filters_updated_AVX2,
+                               &error_sum_AVX2, kComputeAccumulatederror,
+                               accumulated_error_AVX2, scratch_memory);
+        MatchedFilterCore(x_index, h.size() * 150.f * 150.f, kSmoothing, x, y,
+                          h, &filters_updated, &error_sum,
+                          kComputeAccumulatederror, accumulated_error);
+        EXPECT_EQ(filters_updated, filters_updated_AVX2);
+        EXPECT_NEAR(error_sum, error_sum_AVX2, error_sum / 100000.f);
+        for (size_t j = 0; j < h.size(); ++j) {
+          EXPECT_NEAR(h[j], h_AVX2[j], 0.00001f);
+        }
+        for (size_t j = 0; j < accumulated_error.size(); j += 4) {
+          float difference =
+              std::abs(accumulated_error[j] - accumulated_error_AVX2[j]);
+          float relative_difference = accumulated_error[j] > 0
+                                          ? difference / accumulated_error[j]
+                                          : difference;
+          EXPECT_NEAR(relative_difference, 0.0f, 0.00001f);
+        }
+        x_index = (x_index + sub_block_size) % x.size();
+      }
+    }
+  }
+}
+
+#endif
+
+// Verifies that the (optimized) function MaxSquarePeakIndex() produces output
+// equal to the corresponding std-functions.
+TEST(MatchedFilter, MaxSquarePeakIndex) {
+  Random random_generator(42U);
+  constexpr int kMaxLength = 128;
+  constexpr int kNumIterationsPerLength = 256;
+  for (int length = 1; length < kMaxLength; ++length) {
+    std::vector<float> y(length);
+    for (int i = 0; i < kNumIterationsPerLength; ++i) {
+      RandomizeSampleVector(&random_generator, y);
+
+      size_t lag_from_function = MaxSquarePeakIndex(y);
+      size_t lag_from_std = std::distance(
+          y.begin(),
+          std::max_element(y.begin(), y.end(), [](float a, float b) -> bool {
+            return a * a < b * b;
+          }));
+      EXPECT_EQ(lag_from_function, lag_from_std);
+    }
+  }
+}
+
+// Verifies that the matched filter produces proper lag estimates for
+// artificially delayed signals.
+TEST_P(MatchedFilterTest, LagEstimation) {
+  const bool kDetectPreEcho = GetParam();
+  Random random_generator(42U);
+  constexpr size_t kNumChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  for (auto down_sampling_factor : kDownSamplingFactors) {
+    const size_t sub_block_size = kBlockSize / down_sampling_factor;
+
+    Block render(kNumBands, kNumChannels);
+    std::vector<std::vector<float>> capture(
+        1, std::vector<float>(kBlockSize, 0.f));
+    ApmDataDumper data_dumper(0);
+    for (size_t delay_samples : {5, 64, 150, 200, 800, 1000}) {
+      SCOPED_TRACE(ProduceDebugText(delay_samples, down_sampling_factor));
+      EchoCanceller3Config config;
+      config.delay.down_sampling_factor = down_sampling_factor;
+      config.delay.num_filters = kNumMatchedFilters;
+      Decimator capture_decimator(down_sampling_factor);
+      DelayBuffer<float> signal_delay_buffer(down_sampling_factor *
+                                             delay_samples);
+      MatchedFilter filter(
+          &data_dumper, DetectOptimization(), sub_block_size,
+          kWindowSizeSubBlocks, kNumMatchedFilters, kAlignmentShiftSubBlocks,
+          150, config.delay.delay_estimate_smoothing,
+          config.delay.delay_estimate_smoothing_delay_found,
+          config.delay.delay_candidate_detection_threshold, kDetectPreEcho);
+
+      std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+          RenderDelayBuffer::Create(config, kSampleRateHz, kNumChannels));
+
+      // Analyze the correlation between render and capture.
+      for (size_t k = 0; k < (600 + delay_samples / sub_block_size); ++k) {
+        for (size_t band = 0; band < kNumBands; ++band) {
+          for (size_t channel = 0; channel < kNumChannels; ++channel) {
+            RandomizeSampleVector(&random_generator,
+                                  render.View(band, channel));
+          }
+        }
+        signal_delay_buffer.Delay(render.View(/*band=*/0, /*channel=*/0),
+                                  capture[0]);
+        render_delay_buffer->Insert(render);
+
+        if (k == 0) {
+          render_delay_buffer->Reset();
+        }
+
+        render_delay_buffer->PrepareCaptureProcessing();
+        std::array<float, kBlockSize> downsampled_capture_data;
+        rtc::ArrayView<float> downsampled_capture(
+            downsampled_capture_data.data(), sub_block_size);
+        capture_decimator.Decimate(capture[0], downsampled_capture);
+        filter.Update(render_delay_buffer->GetDownsampledRenderBuffer(),
+                      downsampled_capture, /*use_slow_smoothing=*/false);
+      }
+
+      // Obtain the lag estimates.
+      auto lag_estimate = filter.GetBestLagEstimate();
+      EXPECT_TRUE(lag_estimate.has_value());
+
+      // Verify that the expected most accurate lag estimate is correct.
+      if (lag_estimate.has_value()) {
+        EXPECT_EQ(delay_samples, lag_estimate->lag);
+        EXPECT_EQ(delay_samples, lag_estimate->pre_echo_lag);
+      }
+    }
+  }
+}
+
+// Test the pre echo estimation.
+TEST_P(MatchedFilterTest, PreEchoEstimation) {
+  const bool kDetectPreEcho = GetParam();
+  Random random_generator(42U);
+  constexpr size_t kNumChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  for (auto down_sampling_factor : kDownSamplingFactors) {
+    const size_t sub_block_size = kBlockSize / down_sampling_factor;
+
+    Block render(kNumBands, kNumChannels);
+    std::vector<std::vector<float>> capture(
+        1, std::vector<float>(kBlockSize, 0.f));
+    std::vector<float> capture_with_pre_echo(kBlockSize, 0.f);
+    ApmDataDumper data_dumper(0);
+    // data_dumper.SetActivated(true);
+    size_t pre_echo_delay_samples = 20e-3 * 16000 / down_sampling_factor;
+    size_t echo_delay_samples = 50e-3 * 16000 / down_sampling_factor;
+    EchoCanceller3Config config;
+    config.delay.down_sampling_factor = down_sampling_factor;
+    config.delay.num_filters = kNumMatchedFilters;
+    Decimator capture_decimator(down_sampling_factor);
+    DelayBuffer<float> signal_echo_delay_buffer(down_sampling_factor *
+                                                echo_delay_samples);
+    DelayBuffer<float> signal_pre_echo_delay_buffer(down_sampling_factor *
+                                                    pre_echo_delay_samples);
+    MatchedFilter filter(
+        &data_dumper, DetectOptimization(), sub_block_size,
+        kWindowSizeSubBlocks, kNumMatchedFilters, kAlignmentShiftSubBlocks, 150,
+        config.delay.delay_estimate_smoothing,
+        config.delay.delay_estimate_smoothing_delay_found,
+        config.delay.delay_candidate_detection_threshold, kDetectPreEcho);
+    std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+        RenderDelayBuffer::Create(config, kSampleRateHz, kNumChannels));
+    // Analyze the correlation between render and capture.
+    for (size_t k = 0; k < (600 + echo_delay_samples / sub_block_size); ++k) {
+      for (size_t band = 0; band < kNumBands; ++band) {
+        for (size_t channel = 0; channel < kNumChannels; ++channel) {
+          RandomizeSampleVector(&random_generator, render.View(band, channel));
+        }
+      }
+      signal_echo_delay_buffer.Delay(render.View(0, 0), capture[0]);
+      signal_pre_echo_delay_buffer.Delay(render.View(0, 0),
+                                         capture_with_pre_echo);
+      for (size_t k = 0; k < capture[0].size(); ++k) {
+        constexpr float gain_pre_echo = 0.8f;
+        capture[0][k] += gain_pre_echo * capture_with_pre_echo[k];
+      }
+      render_delay_buffer->Insert(render);
+      if (k == 0) {
+        render_delay_buffer->Reset();
+      }
+      render_delay_buffer->PrepareCaptureProcessing();
+      std::array<float, kBlockSize> downsampled_capture_data;
+      rtc::ArrayView<float> downsampled_capture(downsampled_capture_data.data(),
+                                                sub_block_size);
+      capture_decimator.Decimate(capture[0], downsampled_capture);
+      filter.Update(render_delay_buffer->GetDownsampledRenderBuffer(),
+                    downsampled_capture, /*use_slow_smoothing=*/false);
+    }
+    // Obtain the lag estimates.
+    auto lag_estimate = filter.GetBestLagEstimate();
+    EXPECT_TRUE(lag_estimate.has_value());
+    // Verify that the expected most accurate lag estimate is correct.
+    if (lag_estimate.has_value()) {
+      EXPECT_EQ(echo_delay_samples, lag_estimate->lag);
+      if (kDetectPreEcho) {
+        // The pre echo delay is estimated in a subsampled domain and a larger
+        // error is allowed.
+        EXPECT_NEAR(pre_echo_delay_samples, lag_estimate->pre_echo_lag, 4);
+      } else {
+        // The pre echo delay fallback to the highest mached filter peak when
+        // its detection is disabled.
+        EXPECT_EQ(echo_delay_samples, lag_estimate->pre_echo_lag);
+      }
+    }
+  }
+}
+
+// Verifies that the matched filter does not produce reliable and accurate
+// estimates for uncorrelated render and capture signals.
+TEST_P(MatchedFilterTest, LagNotReliableForUncorrelatedRenderAndCapture) {
+  const bool kDetectPreEcho = GetParam();
+  constexpr size_t kNumChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  Random random_generator(42U);
+  for (auto down_sampling_factor : kDownSamplingFactors) {
+    EchoCanceller3Config config;
+    config.delay.down_sampling_factor = down_sampling_factor;
+    config.delay.num_filters = kNumMatchedFilters;
+    const size_t sub_block_size = kBlockSize / down_sampling_factor;
+
+    Block render(kNumBands, kNumChannels);
+    std::array<float, kBlockSize> capture_data;
+    rtc::ArrayView<float> capture(capture_data.data(), sub_block_size);
+    std::fill(capture.begin(), capture.end(), 0.f);
+    ApmDataDumper data_dumper(0);
+    std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+        RenderDelayBuffer::Create(config, kSampleRateHz, kNumChannels));
+    MatchedFilter filter(
+        &data_dumper, DetectOptimization(), sub_block_size,
+        kWindowSizeSubBlocks, kNumMatchedFilters, kAlignmentShiftSubBlocks, 150,
+        config.delay.delay_estimate_smoothing,
+        config.delay.delay_estimate_smoothing_delay_found,
+        config.delay.delay_candidate_detection_threshold, kDetectPreEcho);
+
+    // Analyze the correlation between render and capture.
+    for (size_t k = 0; k < 100; ++k) {
+      RandomizeSampleVector(&random_generator,
+                            render.View(/*band=*/0, /*channel=*/0));
+      RandomizeSampleVector(&random_generator, capture);
+      render_delay_buffer->Insert(render);
+      filter.Update(render_delay_buffer->GetDownsampledRenderBuffer(), capture,
+                    false);
+    }
+
+    // Obtain the best lag estimate and Verify that no lag estimates are
+    // reliable.
+    auto best_lag_estimates = filter.GetBestLagEstimate();
+    EXPECT_FALSE(best_lag_estimates.has_value());
+  }
+}
+
+// Verifies that the matched filter does not produce updated lag estimates for
+// render signals of low level.
+TEST_P(MatchedFilterTest, LagNotUpdatedForLowLevelRender) {
+  const bool kDetectPreEcho = GetParam();
+  Random random_generator(42U);
+  constexpr size_t kNumChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  for (auto down_sampling_factor : kDownSamplingFactors) {
+    const size_t sub_block_size = kBlockSize / down_sampling_factor;
+
+    Block render(kNumBands, kNumChannels);
+    std::vector<std::vector<float>> capture(
+        1, std::vector<float>(kBlockSize, 0.f));
+    ApmDataDumper data_dumper(0);
+    EchoCanceller3Config config;
+    MatchedFilter filter(
+        &data_dumper, DetectOptimization(), sub_block_size,
+        kWindowSizeSubBlocks, kNumMatchedFilters, kAlignmentShiftSubBlocks, 150,
+        config.delay.delay_estimate_smoothing,
+        config.delay.delay_estimate_smoothing_delay_found,
+        config.delay.delay_candidate_detection_threshold, kDetectPreEcho);
+    std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+        RenderDelayBuffer::Create(EchoCanceller3Config(), kSampleRateHz,
+                                  kNumChannels));
+    Decimator capture_decimator(down_sampling_factor);
+
+    // Analyze the correlation between render and capture.
+    for (size_t k = 0; k < 100; ++k) {
+      RandomizeSampleVector(&random_generator, render.View(0, 0));
+      for (auto& render_k : render.View(0, 0)) {
+        render_k *= 149.f / 32767.f;
+      }
+      std::copy(render.begin(0, 0), render.end(0, 0), capture[0].begin());
+      std::array<float, kBlockSize> downsampled_capture_data;
+      rtc::ArrayView<float> downsampled_capture(downsampled_capture_data.data(),
+                                                sub_block_size);
+      capture_decimator.Decimate(capture[0], downsampled_capture);
+      filter.Update(render_delay_buffer->GetDownsampledRenderBuffer(),
+                    downsampled_capture, false);
+    }
+
+    // Verify that no lag estimate has been produced.
+    auto lag_estimate = filter.GetBestLagEstimate();
+    EXPECT_FALSE(lag_estimate.has_value());
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(_, MatchedFilterTest, testing::Values(true, false));
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+class MatchedFilterDeathTest : public ::testing::TestWithParam<bool> {};
+
+// Verifies the check for non-zero windows size.
+TEST_P(MatchedFilterDeathTest, ZeroWindowSize) {
+  const bool kDetectPreEcho = GetParam();
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  EXPECT_DEATH(MatchedFilter(&data_dumper, DetectOptimization(), 16, 0, 1, 1,
+                             150, config.delay.delay_estimate_smoothing,
+                             config.delay.delay_estimate_smoothing_delay_found,
+                             config.delay.delay_candidate_detection_threshold,
+                             kDetectPreEcho),
+               "");
+}
+
+// Verifies the check for non-null data dumper.
+TEST_P(MatchedFilterDeathTest, NullDataDumper) {
+  const bool kDetectPreEcho = GetParam();
+  EchoCanceller3Config config;
+  EXPECT_DEATH(MatchedFilter(nullptr, DetectOptimization(), 16, 1, 1, 1, 150,
+                             config.delay.delay_estimate_smoothing,
+                             config.delay.delay_estimate_smoothing_delay_found,
+                             config.delay.delay_candidate_detection_threshold,
+                             kDetectPreEcho),
+               "");
+}
+
+// Verifies the check for that the sub block size is a multiple of 4.
+// TODO(peah): Activate the unittest once the required code has been landed.
+TEST_P(MatchedFilterDeathTest, DISABLED_BlockSizeMultipleOf4) {
+  const bool kDetectPreEcho = GetParam();
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  EXPECT_DEATH(MatchedFilter(&data_dumper, DetectOptimization(), 15, 1, 1, 1,
+                             150, config.delay.delay_estimate_smoothing,
+                             config.delay.delay_estimate_smoothing_delay_found,
+                             config.delay.delay_candidate_detection_threshold,
+                             kDetectPreEcho),
+               "");
+}
+
+// Verifies the check for that there is an integer number of sub blocks that add
+// up to a block size.
+// TODO(peah): Activate the unittest once the required code has been landed.
+TEST_P(MatchedFilterDeathTest, DISABLED_SubBlockSizeAddsUpToBlockSize) {
+  const bool kDetectPreEcho = GetParam();
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  EXPECT_DEATH(MatchedFilter(&data_dumper, DetectOptimization(), 12, 1, 1, 1,
+                             150, config.delay.delay_estimate_smoothing,
+                             config.delay.delay_estimate_smoothing_delay_found,
+                             config.delay.delay_candidate_detection_threshold,
+                             kDetectPreEcho),
+               "");
+}
+
+INSTANTIATE_TEST_SUITE_P(_,
+                         MatchedFilterDeathTest,
+                         testing::Values(true, false));
+
+#endif
+
+}  // namespace aec3
+
+TEST(MatchedFilterFieldTrialTest, PreEchoConfigurationTest) {
+  float threshold_in = 0.1f;
+  int mode_in = 2;
+  rtc::StringBuilder field_trial_name;
+  field_trial_name << "WebRTC-Aec3PreEchoConfiguration/threshold:"
+                   << threshold_in << ",mode:" << mode_in << "/";
+  webrtc::test::ScopedFieldTrials field_trials(field_trial_name.str());
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  MatchedFilter matched_filter(
+      &data_dumper, DetectOptimization(),
+      kBlockSize / config.delay.down_sampling_factor,
+      aec3::kWindowSizeSubBlocks, aec3::kNumMatchedFilters,
+      aec3::kAlignmentShiftSubBlocks,
+      config.render_levels.poor_excitation_render_limit,
+      config.delay.delay_estimate_smoothing,
+      config.delay.delay_estimate_smoothing_delay_found,
+      config.delay.delay_candidate_detection_threshold,
+      config.delay.detect_pre_echo);
+
+  auto& pre_echo_config = matched_filter.GetPreEchoConfiguration();
+  EXPECT_EQ(pre_echo_config.threshold, threshold_in);
+  EXPECT_EQ(pre_echo_config.mode, mode_in);
+}
+
+TEST(MatchedFilterFieldTrialTest, WrongPreEchoConfigurationTest) {
+  constexpr float kDefaultThreshold = 0.5f;
+  constexpr int kDefaultMode = 0;
+  float threshold_in = -0.1f;
+  int mode_in = 5;
+  rtc::StringBuilder field_trial_name;
+  field_trial_name << "WebRTC-Aec3PreEchoConfiguration/threshold:"
+                   << threshold_in << ",mode:" << mode_in << "/";
+  webrtc::test::ScopedFieldTrials field_trials(field_trial_name.str());
+  ApmDataDumper data_dumper(0);
+  EchoCanceller3Config config;
+  MatchedFilter matched_filter(
+      &data_dumper, DetectOptimization(),
+      kBlockSize / config.delay.down_sampling_factor,
+      aec3::kWindowSizeSubBlocks, aec3::kNumMatchedFilters,
+      aec3::kAlignmentShiftSubBlocks,
+      config.render_levels.poor_excitation_render_limit,
+      config.delay.delay_estimate_smoothing,
+      config.delay.delay_estimate_smoothing_delay_found,
+      config.delay.delay_candidate_detection_threshold,
+      config.delay.detect_pre_echo);
+
+  auto& pre_echo_config = matched_filter.GetPreEchoConfiguration();
+  EXPECT_EQ(pre_echo_config.threshold, kDefaultThreshold);
+  EXPECT_EQ(pre_echo_config.mode, kDefaultMode);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.cc b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.cc
new file mode 100644
index 0000000000..c5c33dbd68
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/mock/mock_block_processor.h"
+
+namespace webrtc {
+namespace test {
+
+MockBlockProcessor::MockBlockProcessor() = default;
+MockBlockProcessor::~MockBlockProcessor() = default;
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.h b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.h
new file mode 100644
index 0000000000..c9ae38c4aa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_
+
+#include <vector>
+
+#include "modules/audio_processing/aec3/block_processor.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockBlockProcessor : public BlockProcessor {
+ public:
+  MockBlockProcessor();
+  virtual ~MockBlockProcessor();
+
+  MOCK_METHOD(void,
+              ProcessCapture,
+              (bool level_change,
+               bool saturated_microphone_signal,
+               Block* linear_output,
+               Block* capture_block),
+              (override));
+  MOCK_METHOD(void, BufferRender, (const Block& block), (override));
+  MOCK_METHOD(void,
+              UpdateEchoLeakageStatus,
+              (bool leakage_detected),
+              (override));
+  MOCK_METHOD(void,
+              GetMetrics,
+              (EchoControl::Metrics * metrics),
+              (const, override));
+  MOCK_METHOD(void, SetAudioBufferDelay, (int delay_ms), (override));
+  MOCK_METHOD(void,
+              SetCaptureOutputUsage,
+              (bool capture_output_used),
+              (override));
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.cc b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.cc
new file mode 100644
index 0000000000..b903bf0785
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/mock/mock_echo_remover.h"
+
+namespace webrtc {
+namespace test {
+
+MockEchoRemover::MockEchoRemover() = default;
+MockEchoRemover::~MockEchoRemover() = default;
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h
new file mode 100644
index 0000000000..31f075ef0a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_ECHO_REMOVER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_ECHO_REMOVER_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+#include "modules/audio_processing/aec3/echo_remover.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockEchoRemover : public EchoRemover {
+ public:
+  MockEchoRemover();
+  virtual ~MockEchoRemover();
+
+  MOCK_METHOD(void,
+              ProcessCapture,
+              (EchoPathVariability echo_path_variability,
+               bool capture_signal_saturation,
+               const absl::optional<DelayEstimate>& delay_estimate,
+               RenderBuffer* render_buffer,
+               Block* linear_output,
+               Block* capture),
+              (override));
+  MOCK_METHOD(void,
+              UpdateEchoLeakageStatus,
+              (bool leakage_detected),
+              (override));
+  MOCK_METHOD(void,
+              GetMetrics,
+              (EchoControl::Metrics * metrics),
+              (const, override));
+  MOCK_METHOD(void,
+              SetCaptureOutputUsage,
+              (bool capture_output_used),
+              (override));
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_ECHO_REMOVER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc
new file mode 100644
index 0000000000..d4ad09b4bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/mock/mock_render_delay_buffer.h"
+
+namespace webrtc {
+namespace test {
+
+MockRenderDelayBuffer::MockRenderDelayBuffer(int sample_rate_hz,
+                                             size_t num_channels)
+    : block_buffer_(GetRenderDelayBufferSize(4, 4, 12),
+                    NumBandsForRate(sample_rate_hz),
+                    num_channels),
+      spectrum_buffer_(block_buffer_.buffer.size(), num_channels),
+      fft_buffer_(block_buffer_.buffer.size(), num_channels),
+      render_buffer_(&block_buffer_, &spectrum_buffer_, &fft_buffer_),
+      downsampled_render_buffer_(GetDownSampledBufferSize(4, 4)) {
+  ON_CALL(*this, GetRenderBuffer())
+      .WillByDefault(
+          ::testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer));
+  ON_CALL(*this, GetDownsampledRenderBuffer())
+      .WillByDefault(::testing::Invoke(
+          this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer));
+}
+
+MockRenderDelayBuffer::~MockRenderDelayBuffer() = default;
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h
new file mode 100644
index 0000000000..c17fd62caa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
+
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockRenderDelayBuffer : public RenderDelayBuffer {
+ public:
+  MockRenderDelayBuffer(int sample_rate_hz, size_t num_channels);
+  virtual ~MockRenderDelayBuffer();
+
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(RenderDelayBuffer::BufferingEvent,
+              Insert,
+              (const Block& block),
+              (override));
+  MOCK_METHOD(void, HandleSkippedCaptureProcessing, (), (override));
+  MOCK_METHOD(RenderDelayBuffer::BufferingEvent,
+              PrepareCaptureProcessing,
+              (),
+              (override));
+  MOCK_METHOD(bool, AlignFromDelay, (size_t delay), (override));
+  MOCK_METHOD(void, AlignFromExternalDelay, (), (override));
+  MOCK_METHOD(size_t, Delay, (), (const, override));
+  MOCK_METHOD(size_t, MaxDelay, (), (const, override));
+  MOCK_METHOD(RenderBuffer*, GetRenderBuffer, (), (override));
+  MOCK_METHOD(const DownsampledRenderBuffer&,
+              GetDownsampledRenderBuffer,
+              (),
+              (const, override));
+  MOCK_METHOD(void, SetAudioBufferDelay, (int delay_ms), (override));
+  MOCK_METHOD(bool, HasReceivedBufferDelay, (), (override));
+
+ private:
+  RenderBuffer* FakeGetRenderBuffer() { return &render_buffer_; }
+  const DownsampledRenderBuffer& FakeGetDownsampledRenderBuffer() const {
+    return downsampled_render_buffer_;
+  }
+  BlockBuffer block_buffer_;
+  SpectrumBuffer spectrum_buffer_;
+  FftBuffer fft_buffer_;
+  RenderBuffer render_buffer_;
+  DownsampledRenderBuffer downsampled_render_buffer_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.cc b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.cc
new file mode 100644
index 0000000000..4ae2af96bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/mock/mock_render_delay_controller.h"
+
+namespace webrtc {
+namespace test {
+
+MockRenderDelayController::MockRenderDelayController() = default;
+MockRenderDelayController::~MockRenderDelayController() = default;
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
new file mode 100644
index 0000000000..14d499dd28
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/aec3/render_delay_controller.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockRenderDelayController : public RenderDelayController {
+ public:
+  MockRenderDelayController();
+  virtual ~MockRenderDelayController();
+
+  MOCK_METHOD(void, Reset, (bool reset_delay_statistics), (override));
+  MOCK_METHOD(void, LogRenderCall, (), (override));
+  MOCK_METHOD(absl::optional<DelayEstimate>,
+              GetDelay,
+              (const DownsampledRenderBuffer& render_buffer,
+               size_t render_delay_buffer_delay,
+               const Block& capture),
+              (override));
+  MOCK_METHOD(bool, HasClockdrift, (), (const, override));
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/moving_average.cc b/third_party/libwebrtc/modules/audio_processing/aec3/moving_average.cc
new file mode 100644
index 0000000000..7a81ee89ea
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/moving_average.cc
@@ -0,0 +1,60 @@
+
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/moving_average.h"
+
+#include <algorithm>
+#include <functional>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace aec3 {
+
+MovingAverage::MovingAverage(size_t num_elem, size_t mem_len)
+    : num_elem_(num_elem),
+      mem_len_(mem_len - 1),
+      scaling_(1.0f / static_cast<float>(mem_len)),
+      memory_(num_elem * mem_len_, 0.f),
+      mem_index_(0) {
+  RTC_DCHECK(num_elem_ > 0);
+  RTC_DCHECK(mem_len > 0);
+}
+
+MovingAverage::~MovingAverage() = default;
+
+void MovingAverage::Average(rtc::ArrayView<const float> input,
+                            rtc::ArrayView<float> output) {
+  RTC_DCHECK(input.size() == num_elem_);
+  RTC_DCHECK(output.size() == num_elem_);
+
+  // Sum all contributions.
+  std::copy(input.begin(), input.end(), output.begin());
+  for (auto i = memory_.begin(); i < memory_.end(); i += num_elem_) {
+    std::transform(i, i + num_elem_, output.begin(), output.begin(),
+                   std::plus<float>());
+  }
+
+  // Divide by mem_len_.
+  for (float& o : output) {
+    o *= scaling_;
+  }
+
+  // Update memory.
+  if (mem_len_ > 0) {
+    std::copy(input.begin(), input.end(),
+              memory_.begin() + mem_index_ * num_elem_);
+    mem_index_ = (mem_index_ + 1) % mem_len_;
+  }
+}
+
+}  // namespace aec3
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/moving_average.h b/third_party/libwebrtc/modules/audio_processing/aec3/moving_average.h
new file mode 100644
index 0000000000..913d78519c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/moving_average.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+namespace aec3 {
+
+class MovingAverage {
+ public:
+  // Creates an instance of MovingAverage that accepts inputs of length num_elem
+  // and averages over mem_len inputs.
+  MovingAverage(size_t num_elem, size_t mem_len);
+  ~MovingAverage();
+
+  // Computes the average of input and mem_len-1 previous inputs and stores the
+  // result in output.
+  void Average(rtc::ArrayView<const float> input, rtc::ArrayView<float> output);
+
+ private:
+  const size_t num_elem_;
+  const size_t mem_len_;
+  const float scaling_;
+  std::vector<float> memory_;
+  size_t mem_index_;
+};
+
+}  // namespace aec3
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/moving_average_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/moving_average_unittest.cc
new file mode 100644
index 0000000000..84ba9cbc5b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/moving_average_unittest.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/moving_average.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(MovingAverage, Average) {
+  constexpr size_t num_elem = 4;
+  constexpr size_t mem_len = 3;
+  constexpr float e = 1e-6f;
+  aec3::MovingAverage ma(num_elem, mem_len);
+  std::array<float, num_elem> data1 = {1, 2, 3, 4};
+  std::array<float, num_elem> data2 = {5, 1, 9, 7};
+  std::array<float, num_elem> data3 = {3, 3, 5, 6};
+  std::array<float, num_elem> data4 = {8, 4, 2, 1};
+  std::array<float, num_elem> output;
+
+  ma.Average(data1, output);
+  EXPECT_NEAR(output[0], data1[0] / 3.0f, e);
+  EXPECT_NEAR(output[1], data1[1] / 3.0f, e);
+  EXPECT_NEAR(output[2], data1[2] / 3.0f, e);
+  EXPECT_NEAR(output[3], data1[3] / 3.0f, e);
+
+  ma.Average(data2, output);
+  EXPECT_NEAR(output[0], (data1[0] + data2[0]) / 3.0f, e);
+  EXPECT_NEAR(output[1], (data1[1] + data2[1]) / 3.0f, e);
+  EXPECT_NEAR(output[2], (data1[2] + data2[2]) / 3.0f, e);
+  EXPECT_NEAR(output[3], (data1[3] + data2[3]) / 3.0f, e);
+
+  ma.Average(data3, output);
+  EXPECT_NEAR(output[0], (data1[0] + data2[0] + data3[0]) / 3.0f, e);
+  EXPECT_NEAR(output[1], (data1[1] + data2[1] + data3[1]) / 3.0f, e);
+  EXPECT_NEAR(output[2], (data1[2] + data2[2] + data3[2]) / 3.0f, e);
+  EXPECT_NEAR(output[3], (data1[3] + data2[3] + data3[3]) / 3.0f, e);
+
+  ma.Average(data4, output);
+  EXPECT_NEAR(output[0], (data2[0] + data3[0] + data4[0]) / 3.0f, e);
+  EXPECT_NEAR(output[1], (data2[1] + data3[1] + data4[1]) / 3.0f, e);
+  EXPECT_NEAR(output[2], (data2[2] + data3[2] + data4[2]) / 3.0f, e);
+  EXPECT_NEAR(output[3], (data2[3] + data3[3] + data4[3]) / 3.0f, e);
+}
+
+TEST(MovingAverage, PassThrough) {
+  constexpr size_t num_elem = 4;
+  constexpr size_t mem_len = 1;
+  constexpr float e = 1e-6f;
+  aec3::MovingAverage ma(num_elem, mem_len);
+  std::array<float, num_elem> data1 = {1, 2, 3, 4};
+  std::array<float, num_elem> data2 = {5, 1, 9, 7};
+  std::array<float, num_elem> data3 = {3, 3, 5, 6};
+  std::array<float, num_elem> data4 = {8, 4, 2, 1};
+  std::array<float, num_elem> output;
+
+  ma.Average(data1, output);
+  EXPECT_NEAR(output[0], data1[0], e);
+  EXPECT_NEAR(output[1], data1[1], e);
+  EXPECT_NEAR(output[2], data1[2], e);
+  EXPECT_NEAR(output[3], data1[3], e);
+
+  ma.Average(data2, output);
+  EXPECT_NEAR(output[0], data2[0], e);
+  EXPECT_NEAR(output[1], data2[1], e);
+  EXPECT_NEAR(output[2], data2[2], e);
+  EXPECT_NEAR(output[3], data2[3], e);
+
+  ma.Average(data3, output);
+  EXPECT_NEAR(output[0], data3[0], e);
+  EXPECT_NEAR(output[1], data3[1], e);
+  EXPECT_NEAR(output[2], data3[2], e);
+  EXPECT_NEAR(output[3], data3[3], e);
+
+  ma.Average(data4, output);
+  EXPECT_NEAR(output[0], data4[0], e);
+  EXPECT_NEAR(output[1], data4[1], e);
+  EXPECT_NEAR(output[2], data4[2], e);
+  EXPECT_NEAR(output[3], data4[3], e);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector.cc b/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector.cc
new file mode 100644
index 0000000000..98068964d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/multi_channel_content_detector.h"
+
+#include <cmath>
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kNumFramesPerSecond = 100;
+
+// Compares the left and right channels in the render `frame` to determine
+// whether the signal is a proper stereo signal. To allow for differences
+// introduced by hardware drivers, a threshold `detection_threshold` is used for
+// the detection.
+bool HasStereoContent(const std::vector<std::vector<std::vector<float>>>& frame,
+                      float detection_threshold) {
+  if (frame[0].size() < 2) {
+    return false;
+  }
+
+  for (size_t band = 0; band < frame.size(); ++band) {
+    for (size_t k = 0; k < frame[band][0].size(); ++k) {
+      if (std::fabs(frame[band][0][k] - frame[band][1][k]) >
+          detection_threshold) {
+        return true;
+      }
+    }
+  }
+  return false;
+}
+
+// In order to avoid logging metrics for very short lifetimes that are unlikely
+// to reflect real calls and that may dilute the "real" data, logging is limited
+// to lifetimes of at leats 5 seconds.
+constexpr int kMinNumberOfFramesRequiredToLogMetrics = 500;
+
+// Continuous metrics are logged every 10 seconds.
+constexpr int kFramesPer10Seconds = 1000;
+
+}  // namespace
+
+MultiChannelContentDetector::MetricsLogger::MetricsLogger() {}
+
+MultiChannelContentDetector::MetricsLogger::~MetricsLogger() {
+  if (frame_counter_ < kMinNumberOfFramesRequiredToLogMetrics)
+    return;
+
+  RTC_HISTOGRAM_BOOLEAN(
+      "WebRTC.Audio.EchoCanceller.PersistentMultichannelContentEverDetected",
+      any_multichannel_content_detected_ ? 1 : 0);
+}
+
+void MultiChannelContentDetector::MetricsLogger::Update(
+    bool persistent_multichannel_content_detected) {
+  ++frame_counter_;
+  if (persistent_multichannel_content_detected) {
+    any_multichannel_content_detected_ = true;
+    ++persistent_multichannel_frame_counter_;
+  }
+
+  if (frame_counter_ < kMinNumberOfFramesRequiredToLogMetrics)
+    return;
+  if (frame_counter_ % kFramesPer10Seconds != 0)
+    return;
+  const bool mostly_multichannel_last_10_seconds =
+      (persistent_multichannel_frame_counter_ >= kFramesPer10Seconds / 2);
+  RTC_HISTOGRAM_BOOLEAN(
+      "WebRTC.Audio.EchoCanceller.ProcessingPersistentMultichannelContent",
+      mostly_multichannel_last_10_seconds ? 1 : 0);
+
+  persistent_multichannel_frame_counter_ = 0;
+}
+
+MultiChannelContentDetector::MultiChannelContentDetector(
+    bool detect_stereo_content,
+    int num_render_input_channels,
+    float detection_threshold,
+    int stereo_detection_timeout_threshold_seconds,
+    float stereo_detection_hysteresis_seconds)
+    : detect_stereo_content_(detect_stereo_content),
+      detection_threshold_(detection_threshold),
+      detection_timeout_threshold_frames_(
+          stereo_detection_timeout_threshold_seconds > 0
+              ? absl::make_optional(stereo_detection_timeout_threshold_seconds *
+                                    kNumFramesPerSecond)
+              : absl::nullopt),
+      stereo_detection_hysteresis_frames_(static_cast<int>(
+          stereo_detection_hysteresis_seconds * kNumFramesPerSecond)),
+      metrics_logger_((detect_stereo_content && num_render_input_channels > 1)
+                          ? std::make_unique<MetricsLogger>()
+                          : nullptr),
+      persistent_multichannel_content_detected_(
+          !detect_stereo_content && num_render_input_channels > 1) {}
+
+bool MultiChannelContentDetector::UpdateDetection(
+    const std::vector<std::vector<std::vector<float>>>& frame) {
+  if (!detect_stereo_content_) {
+    RTC_DCHECK_EQ(frame[0].size() > 1,
+                  persistent_multichannel_content_detected_);
+    return false;
+  }
+
+  const bool previous_persistent_multichannel_content_detected =
+      persistent_multichannel_content_detected_;
+  const bool stereo_detected_in_frame =
+      HasStereoContent(frame, detection_threshold_);
+
+  consecutive_frames_with_stereo_ =
+      stereo_detected_in_frame ? consecutive_frames_with_stereo_ + 1 : 0;
+  frames_since_stereo_detected_last_ =
+      stereo_detected_in_frame ? 0 : frames_since_stereo_detected_last_ + 1;
+
+  // Detect persistent multichannel content.
+  if (consecutive_frames_with_stereo_ > stereo_detection_hysteresis_frames_) {
+    persistent_multichannel_content_detected_ = true;
+  }
+  if (detection_timeout_threshold_frames_.has_value() &&
+      frames_since_stereo_detected_last_ >=
+          *detection_timeout_threshold_frames_) {
+    persistent_multichannel_content_detected_ = false;
+  }
+
+  // Detect temporary multichannel content.
+  temporary_multichannel_content_detected_ =
+      persistent_multichannel_content_detected_ ? false
+                                                : stereo_detected_in_frame;
+
+  if (metrics_logger_)
+    metrics_logger_->Update(persistent_multichannel_content_detected_);
+
+  return previous_persistent_multichannel_content_detected !=
+         persistent_multichannel_content_detected_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector.h b/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector.h
new file mode 100644
index 0000000000..1742c5fc17
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector.h
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MULTI_CHANNEL_CONTENT_DETECTOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_MULTI_CHANNEL_CONTENT_DETECTOR_H_
+
+#include <memory>
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+// Analyzes audio content to determine whether the contained audio is proper
+// multichannel, or only upmixed mono. To allow for differences introduced by
+// hardware drivers, a threshold `detection_threshold` is used for the
+// detection.
+// Logs metrics continously and upon destruction.
+class MultiChannelContentDetector {
+ public:
+  // If |stereo_detection_timeout_threshold_seconds| <= 0, no timeout is
+  // applied: Once multichannel is detected, the detector remains in that state
+  // for its lifetime.
+  MultiChannelContentDetector(bool detect_stereo_content,
+                              int num_render_input_channels,
+                              float detection_threshold,
+                              int stereo_detection_timeout_threshold_seconds,
+                              float stereo_detection_hysteresis_seconds);
+
+  // Compares the left and right channels in the render `frame` to determine
+  // whether the signal is a proper multichannel signal. Returns a bool
+  // indicating whether a change in the proper multichannel content was
+  // detected.
+  bool UpdateDetection(
+      const std::vector<std::vector<std::vector<float>>>& frame);
+
+  bool IsProperMultiChannelContentDetected() const {
+    return persistent_multichannel_content_detected_;
+  }
+
+  bool IsTemporaryMultiChannelContentDetected() const {
+    return temporary_multichannel_content_detected_;
+  }
+
+ private:
+  // Tracks and logs metrics for the amount of multichannel content detected.
+  class MetricsLogger {
+   public:
+    MetricsLogger();
+
+    // The destructor logs call summary statistics.
+    ~MetricsLogger();
+
+    // Updates and logs metrics.
+    void Update(bool persistent_multichannel_content_detected);
+
+   private:
+    int frame_counter_ = 0;
+
+    // Counts the number of frames of persistent multichannel audio observed
+    // during the current metrics collection interval.
+    int persistent_multichannel_frame_counter_ = 0;
+
+    // Indicates whether persistent multichannel content has ever been detected.
+    bool any_multichannel_content_detected_ = false;
+  };
+
+  const bool detect_stereo_content_;
+  const float detection_threshold_;
+  const absl::optional<int> detection_timeout_threshold_frames_;
+  const int stereo_detection_hysteresis_frames_;
+
+  // Collects and reports metrics on the amount of multichannel content
+  // detected. Only created if |num_render_input_channels| > 1 and
+  // |detect_stereo_content_| is true.
+  const std::unique_ptr<MetricsLogger> metrics_logger_;
+
+  bool persistent_multichannel_content_detected_;
+  bool temporary_multichannel_content_detected_ = false;
+  int64_t frames_since_stereo_detected_last_ = 0;
+  int64_t consecutive_frames_with_stereo_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MULTI_CHANNEL_CONTENT_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector_unittest.cc
new file mode 100644
index 0000000000..8d38dd0991
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/multi_channel_content_detector_unittest.cc
@@ -0,0 +1,470 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/multi_channel_content_detector.h"
+
+#include "system_wrappers/include/metrics.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(MultiChannelContentDetector, HandlingOfMono) {
+  MultiChannelContentDetector mc(
+      /*detect_stereo_content=*/true,
+      /*num_render_input_channels=*/1,
+      /*detection_threshold=*/0.0f,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      /*stereo_detection_hysteresis_seconds=*/0.0f);
+  EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+}
+
+TEST(MultiChannelContentDetector, HandlingOfMonoAndDetectionOff) {
+  MultiChannelContentDetector mc(
+      /*detect_stereo_content=*/false,
+      /*num_render_input_channels=*/1,
+      /*detection_threshold=*/0.0f,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      /*stereo_detection_hysteresis_seconds=*/0.0f);
+  EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+}
+
+TEST(MultiChannelContentDetector, HandlingOfDetectionOff) {
+  MultiChannelContentDetector mc(
+      /*detect_stereo_content=*/false,
+      /*num_render_input_channels=*/2,
+      /*detection_threshold=*/0.0f,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      /*stereo_detection_hysteresis_seconds=*/0.0f);
+  EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+
+  std::vector<std::vector<std::vector<float>>> frame(
+      1, std::vector<std::vector<float>>(2, std::vector<float>(160, 0.0f)));
+  std::fill(frame[0][0].begin(), frame[0][0].end(), 100.0f);
+  std::fill(frame[0][1].begin(), frame[0][1].end(), 101.0f);
+
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+  EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+}
+
+TEST(MultiChannelContentDetector, InitialDetectionOfStereo) {
+  MultiChannelContentDetector mc(
+      /*detect_stereo_content=*/true,
+      /*num_render_input_channels=*/2,
+      /*detection_threshold=*/0.0f,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      /*stereo_detection_hysteresis_seconds=*/0.0f);
+  EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+}
+
+TEST(MultiChannelContentDetector, DetectionWhenFakeStereo) {
+  MultiChannelContentDetector mc(
+      /*detect_stereo_content=*/true,
+      /*num_render_input_channels=*/2,
+      /*detection_threshold=*/0.0f,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      /*stereo_detection_hysteresis_seconds=*/0.0f);
+  std::vector<std::vector<std::vector<float>>> frame(
+      1, std::vector<std::vector<float>>(2, std::vector<float>(160, 0.0f)));
+  std::fill(frame[0][0].begin(), frame[0][0].end(), 100.0f);
+  std::fill(frame[0][1].begin(), frame[0][1].end(), 100.0f);
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+  EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+}
+
+TEST(MultiChannelContentDetector, DetectionWhenStereo) {
+  MultiChannelContentDetector mc(
+      /*detect_stereo_content=*/true,
+      /*num_render_input_channels=*/2,
+      /*detection_threshold=*/0.0f,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      /*stereo_detection_hysteresis_seconds=*/0.0f);
+  std::vector<std::vector<std::vector<float>>> frame(
+      1, std::vector<std::vector<float>>(2, std::vector<float>(160, 0.0f)));
+  std::fill(frame[0][0].begin(), frame[0][0].end(), 100.0f);
+  std::fill(frame[0][1].begin(), frame[0][1].end(), 101.0f);
+  EXPECT_TRUE(mc.UpdateDetection(frame));
+  EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+}
+
+TEST(MultiChannelContentDetector, DetectionWhenStereoAfterAWhile) {
+  MultiChannelContentDetector mc(
+      /*detect_stereo_content=*/true,
+      /*num_render_input_channels=*/2,
+      /*detection_threshold=*/0.0f,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      /*stereo_detection_hysteresis_seconds=*/0.0f);
+  std::vector<std::vector<std::vector<float>>> frame(
+      1, std::vector<std::vector<float>>(2, std::vector<float>(160, 0.0f)));
+
+  std::fill(frame[0][0].begin(), frame[0][0].end(), 100.0f);
+  std::fill(frame[0][1].begin(), frame[0][1].end(), 100.0f);
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+  EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+
+  std::fill(frame[0][0].begin(), frame[0][0].end(), 100.0f);
+  std::fill(frame[0][1].begin(), frame[0][1].end(), 101.0f);
+
+  EXPECT_TRUE(mc.UpdateDetection(frame));
+  EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+}
+
+TEST(MultiChannelContentDetector, DetectionWithStereoBelowThreshold) {
+  constexpr float kThreshold = 1.0f;
+  MultiChannelContentDetector mc(
+      /*detect_stereo_content=*/true,
+      /*num_render_input_channels=*/2,
+      /*detection_threshold=*/kThreshold,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      /*stereo_detection_hysteresis_seconds=*/0.0f);
+  std::vector<std::vector<std::vector<float>>> frame(
+      1, std::vector<std::vector<float>>(2, std::vector<float>(160, 0.0f)));
+  std::fill(frame[0][0].begin(), frame[0][0].end(), 100.0f);
+  std::fill(frame[0][1].begin(), frame[0][1].end(), 100.0f + kThreshold);
+
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+  EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+}
+
+TEST(MultiChannelContentDetector, DetectionWithStereoAboveThreshold) {
+  constexpr float kThreshold = 1.0f;
+  MultiChannelContentDetector mc(
+      /*detect_stereo_content=*/true,
+      /*num_render_input_channels=*/2,
+      /*detection_threshold=*/kThreshold,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      /*stereo_detection_hysteresis_seconds=*/0.0f);
+  std::vector<std::vector<std::vector<float>>> frame(
+      1, std::vector<std::vector<float>>(2, std::vector<float>(160, 0.0f)));
+  std::fill(frame[0][0].begin(), frame[0][0].end(), 100.0f);
+  std::fill(frame[0][1].begin(), frame[0][1].end(), 100.0f + kThreshold + 0.1f);
+
+  EXPECT_TRUE(mc.UpdateDetection(frame));
+  EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+
+  EXPECT_FALSE(mc.UpdateDetection(frame));
+}
+
+class MultiChannelContentDetectorTimeoutBehavior
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<bool, int>> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannelContentDetector,
+                         MultiChannelContentDetectorTimeoutBehavior,
+                         ::testing::Combine(::testing::Values(false, true),
+                                            ::testing::Values(0, 1, 10)));
+
+TEST_P(MultiChannelContentDetectorTimeoutBehavior,
+       TimeOutBehaviorForNonTrueStereo) {
+  constexpr int kNumFramesPerSecond = 100;
+  const bool detect_stereo_content = std::get<0>(GetParam());
+  const int stereo_detection_timeout_threshold_seconds =
+      std::get<1>(GetParam());
+  const int stereo_detection_timeout_threshold_frames =
+      stereo_detection_timeout_threshold_seconds * kNumFramesPerSecond;
+
+  MultiChannelContentDetector mc(detect_stereo_content,
+                                 /*num_render_input_channels=*/2,
+                                 /*detection_threshold=*/0.0f,
+                                 stereo_detection_timeout_threshold_seconds,
+                                 /*stereo_detection_hysteresis_seconds=*/0.0f);
+  std::vector<std::vector<std::vector<float>>> true_stereo_frame = {
+      {std::vector<float>(160, 100.0f), std::vector<float>(160, 101.0f)}};
+
+  std::vector<std::vector<std::vector<float>>> fake_stereo_frame = {
+      {std::vector<float>(160, 100.0f), std::vector<float>(160, 100.0f)}};
+
+  // Pass fake stereo frames and verify the content detection.
+  for (int k = 0; k < 10; ++k) {
+    EXPECT_FALSE(mc.UpdateDetection(fake_stereo_frame));
+    if (detect_stereo_content) {
+      EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+    } else {
+      EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+    }
+  }
+
+  // Pass a true stereo frame and verify that it is properly detected.
+  if (detect_stereo_content) {
+    EXPECT_TRUE(mc.UpdateDetection(true_stereo_frame));
+  } else {
+    EXPECT_FALSE(mc.UpdateDetection(true_stereo_frame));
+  }
+  EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+
+  // Pass fake stereo frames until any timeouts are about to occur.
+  for (int k = 0; k < stereo_detection_timeout_threshold_frames - 1; ++k) {
+    EXPECT_FALSE(mc.UpdateDetection(fake_stereo_frame));
+    EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+  }
+
+  // Pass a fake stereo frame and verify that any timeouts properly occur.
+  if (detect_stereo_content && stereo_detection_timeout_threshold_frames > 0) {
+    EXPECT_TRUE(mc.UpdateDetection(fake_stereo_frame));
+    EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+  } else {
+    EXPECT_FALSE(mc.UpdateDetection(fake_stereo_frame));
+    EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+  }
+
+  // Pass fake stereo frames and verify the behavior after any timeout.
+  for (int k = 0; k < 10; ++k) {
+    EXPECT_FALSE(mc.UpdateDetection(fake_stereo_frame));
+    if (detect_stereo_content &&
+        stereo_detection_timeout_threshold_frames > 0) {
+      EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+    } else {
+      EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+    }
+  }
+}
+
+class MultiChannelContentDetectorHysteresisBehavior
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<bool, float>> {};
+
+INSTANTIATE_TEST_SUITE_P(
+    MultiChannelContentDetector,
+    MultiChannelContentDetectorHysteresisBehavior,
+    ::testing::Combine(::testing::Values(false, true),
+                       ::testing::Values(0.0f, 0.1f, 0.2f)));
+
+TEST_P(MultiChannelContentDetectorHysteresisBehavior,
+       PeriodBeforeStereoDetectionIsTriggered) {
+  constexpr int kNumFramesPerSecond = 100;
+  const bool detect_stereo_content = std::get<0>(GetParam());
+  const int stereo_detection_hysteresis_seconds = std::get<1>(GetParam());
+  const int stereo_detection_hysteresis_frames =
+      stereo_detection_hysteresis_seconds * kNumFramesPerSecond;
+
+  MultiChannelContentDetector mc(
+      detect_stereo_content,
+      /*num_render_input_channels=*/2,
+      /*detection_threshold=*/0.0f,
+      /*stereo_detection_timeout_threshold_seconds=*/0,
+      stereo_detection_hysteresis_seconds);
+  std::vector<std::vector<std::vector<float>>> true_stereo_frame = {
+      {std::vector<float>(160, 100.0f), std::vector<float>(160, 101.0f)}};
+
+  std::vector<std::vector<std::vector<float>>> fake_stereo_frame = {
+      {std::vector<float>(160, 100.0f), std::vector<float>(160, 100.0f)}};
+
+  // Pass fake stereo frames and verify the content detection.
+  for (int k = 0; k < 10; ++k) {
+    EXPECT_FALSE(mc.UpdateDetection(fake_stereo_frame));
+    if (detect_stereo_content) {
+      EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+    } else {
+      EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+    }
+    EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+  }
+
+  // Pass a two true stereo frames and verify that they are properly detected.
+  ASSERT_TRUE(stereo_detection_hysteresis_frames > 2 ||
+              stereo_detection_hysteresis_frames == 0);
+  for (int k = 0; k < 2; ++k) {
+    if (detect_stereo_content) {
+      if (stereo_detection_hysteresis_seconds == 0.0f) {
+        if (k == 0) {
+          EXPECT_TRUE(mc.UpdateDetection(true_stereo_frame));
+        } else {
+          EXPECT_FALSE(mc.UpdateDetection(true_stereo_frame));
+        }
+        EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+        EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+      } else {
+        EXPECT_FALSE(mc.UpdateDetection(true_stereo_frame));
+        EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+        EXPECT_TRUE(mc.IsTemporaryMultiChannelContentDetected());
+      }
+    } else {
+      EXPECT_FALSE(mc.UpdateDetection(true_stereo_frame));
+      EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+      EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+    }
+  }
+
+  if (stereo_detection_hysteresis_seconds == 0.0f) {
+    return;
+  }
+
+  // Pass true stereo frames until any timeouts are about to occur.
+  for (int k = 0; k < stereo_detection_hysteresis_frames - 3; ++k) {
+    if (detect_stereo_content) {
+      EXPECT_FALSE(mc.UpdateDetection(true_stereo_frame));
+      EXPECT_FALSE(mc.IsProperMultiChannelContentDetected());
+      EXPECT_TRUE(mc.IsTemporaryMultiChannelContentDetected());
+    } else {
+      EXPECT_FALSE(mc.UpdateDetection(true_stereo_frame));
+      EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+      EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+    }
+  }
+
+  // Pass a true stereo frame and verify that it is properly detected.
+  if (detect_stereo_content) {
+    EXPECT_TRUE(mc.UpdateDetection(true_stereo_frame));
+    EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+    EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+  } else {
+    EXPECT_FALSE(mc.UpdateDetection(true_stereo_frame));
+    EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+    EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+  }
+
+  // Pass an additional true stereo frame and verify that it is properly
+  // detected.
+  if (detect_stereo_content) {
+    EXPECT_FALSE(mc.UpdateDetection(true_stereo_frame));
+    EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+    EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+  } else {
+    EXPECT_FALSE(mc.UpdateDetection(true_stereo_frame));
+    EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+    EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+  }
+
+  // Pass a fake stereo frame and verify that it is properly detected.
+  if (detect_stereo_content) {
+    EXPECT_FALSE(mc.UpdateDetection(fake_stereo_frame));
+    EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+    EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+  } else {
+    EXPECT_FALSE(mc.UpdateDetection(fake_stereo_frame));
+    EXPECT_TRUE(mc.IsProperMultiChannelContentDetected());
+    EXPECT_FALSE(mc.IsTemporaryMultiChannelContentDetected());
+  }
+}
+
+class MultiChannelContentDetectorMetricsDisabled
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<bool, int>> {};
+
+INSTANTIATE_TEST_SUITE_P(
+    /*no prefix*/,
+    MultiChannelContentDetectorMetricsDisabled,
+    ::testing::Values(std::tuple<bool, int>(false, 2),
+                      std::tuple<bool, int>(true, 1)));
+
+// Test that no metrics are logged when they are clearly uninteresting and would
+// dilute relevant data: when the reference audio is single channel, or when
+// dynamic detection is disabled.
+TEST_P(MultiChannelContentDetectorMetricsDisabled, ReportsNoMetrics) {
+  metrics::Reset();
+  constexpr int kNumFramesPerSecond = 100;
+  const bool detect_stereo_content = std::get<0>(GetParam());
+  const int channel_count = std::get<1>(GetParam());
+  std::vector<std::vector<std::vector<float>>> audio_frame = {
+      std::vector<std::vector<float>>(channel_count,
+                                      std::vector<float>(160, 100.0f))};
+  {
+    MultiChannelContentDetector mc(
+        /*detect_stereo_content=*/detect_stereo_content,
+        /*num_render_input_channels=*/channel_count,
+        /*detection_threshold=*/0.0f,
+        /*stereo_detection_timeout_threshold_seconds=*/1,
+        /*stereo_detection_hysteresis_seconds=*/0.0f);
+    for (int k = 0; k < 20 * kNumFramesPerSecond; ++k) {
+      mc.UpdateDetection(audio_frame);
+    }
+  }
+  EXPECT_METRIC_EQ(
+      0, metrics::NumSamples("WebRTC.Audio.EchoCanceller."
+                             "ProcessingPersistentMultichannelContent"));
+  EXPECT_METRIC_EQ(
+      0, metrics::NumSamples("WebRTC.Audio.EchoCanceller."
+                             "PersistentMultichannelContentEverDetected"));
+}
+
+// Tests that after 3 seconds, no metrics are reported.
+TEST(MultiChannelContentDetectorMetrics, ReportsNoMetricsForShortLifetime) {
+  metrics::Reset();
+  constexpr int kNumFramesPerSecond = 100;
+  constexpr int kTooFewFramesToLogMetrics = 3 * kNumFramesPerSecond;
+  std::vector<std::vector<std::vector<float>>> audio_frame = {
+      std::vector<std::vector<float>>(2, std::vector<float>(160, 100.0f))};
+  {
+    MultiChannelContentDetector mc(
+        /*detect_stereo_content=*/true,
+        /*num_render_input_channels=*/2,
+        /*detection_threshold=*/0.0f,
+        /*stereo_detection_timeout_threshold_seconds=*/1,
+        /*stereo_detection_hysteresis_seconds=*/0.0f);
+    for (int k = 0; k < kTooFewFramesToLogMetrics; ++k) {
+      mc.UpdateDetection(audio_frame);
+    }
+  }
+  EXPECT_METRIC_EQ(
+      0, metrics::NumSamples("WebRTC.Audio.EchoCanceller."
+                             "ProcessingPersistentMultichannelContent"));
+  EXPECT_METRIC_EQ(
+      0, metrics::NumSamples("WebRTC.Audio.EchoCanceller."
+                             "PersistentMultichannelContentEverDetected"));
+}
+
+// Tests that after 25 seconds, metrics are reported.
+TEST(MultiChannelContentDetectorMetrics, ReportsMetrics) {
+  metrics::Reset();
+  constexpr int kNumFramesPerSecond = 100;
+  std::vector<std::vector<std::vector<float>>> true_stereo_frame = {
+      {std::vector<float>(160, 100.0f), std::vector<float>(160, 101.0f)}};
+  std::vector<std::vector<std::vector<float>>> fake_stereo_frame = {
+      {std::vector<float>(160, 100.0f), std::vector<float>(160, 100.0f)}};
+  {
+    MultiChannelContentDetector mc(
+        /*detect_stereo_content=*/true,
+        /*num_render_input_channels=*/2,
+        /*detection_threshold=*/0.0f,
+        /*stereo_detection_timeout_threshold_seconds=*/1,
+        /*stereo_detection_hysteresis_seconds=*/0.0f);
+    for (int k = 0; k < 10 * kNumFramesPerSecond; ++k) {
+      mc.UpdateDetection(true_stereo_frame);
+    }
+    for (int k = 0; k < 15 * kNumFramesPerSecond; ++k) {
+      mc.UpdateDetection(fake_stereo_frame);
+    }
+  }
+  // After 10 seconds of true stereo and the remainder fake stereo, we expect
+  // one lifetime metric sample (multichannel detected) and two periodic samples
+  // (one multichannel, one mono).
+
+  // Check lifetime metric.
+  EXPECT_METRIC_EQ(
+      1, metrics::NumSamples("WebRTC.Audio.EchoCanceller."
+                             "PersistentMultichannelContentEverDetected"));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.Audio.EchoCanceller."
+                             "PersistentMultichannelContentEverDetected", 1));
+
+  // Check periodic metric.
+  EXPECT_METRIC_EQ(
+      2, metrics::NumSamples("WebRTC.Audio.EchoCanceller."
+                             "ProcessingPersistentMultichannelContent"));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.Audio.EchoCanceller."
+                             "ProcessingPersistentMultichannelContent", 0));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.Audio.EchoCanceller."
+                             "ProcessingPersistentMultichannelContent", 1));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/nearend_detector.h b/third_party/libwebrtc/modules/audio_processing/aec3/nearend_detector.h
new file mode 100644
index 0000000000..0d8a06b2cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/nearend_detector.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_NEAREND_DETECTOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_NEAREND_DETECTOR_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+// Class for selecting whether the suppressor is in the nearend or echo state.
+class NearendDetector {
+ public:
+  virtual ~NearendDetector() {}
+
+  // Returns whether the current state is the nearend state.
+  virtual bool IsNearendState() const = 0;
+
+  // Updates the state selection based on latest spectral estimates.
+  virtual void Update(
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          nearend_spectrum,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          residual_echo_spectrum,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          comfort_noise_spectrum,
+      bool initial_state) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_NEAREND_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain.cc b/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain.cc
new file mode 100644
index 0000000000..8e391d6fa6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain.cc
@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/refined_filter_update_gain.h"
+
+#include <algorithm>
+#include <functional>
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+#include "modules/audio_processing/aec3/subtractor_output.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kHErrorInitial = 10000.f;
+constexpr int kPoorExcitationCounterInitial = 1000;
+
+}  // namespace
+
+std::atomic<int> RefinedFilterUpdateGain::instance_count_(0);
+
+RefinedFilterUpdateGain::RefinedFilterUpdateGain(
+    const EchoCanceller3Config::Filter::RefinedConfiguration& config,
+    size_t config_change_duration_blocks)
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      config_change_duration_blocks_(
+          static_cast<int>(config_change_duration_blocks)),
+      poor_excitation_counter_(kPoorExcitationCounterInitial) {
+  SetConfig(config, true);
+  H_error_.fill(kHErrorInitial);
+  RTC_DCHECK_LT(0, config_change_duration_blocks_);
+  one_by_config_change_duration_blocks_ = 1.f / config_change_duration_blocks_;
+}
+
+RefinedFilterUpdateGain::~RefinedFilterUpdateGain() {}
+
+void RefinedFilterUpdateGain::HandleEchoPathChange(
+    const EchoPathVariability& echo_path_variability) {
+  if (echo_path_variability.gain_change) {
+    // TODO(bugs.webrtc.org/9526) Handle gain changes.
+  }
+
+  if (echo_path_variability.delay_change !=
+      EchoPathVariability::DelayAdjustment::kNone) {
+    H_error_.fill(kHErrorInitial);
+  }
+
+  if (!echo_path_variability.gain_change) {
+    poor_excitation_counter_ = kPoorExcitationCounterInitial;
+    call_counter_ = 0;
+  }
+}
+
+void RefinedFilterUpdateGain::Compute(
+    const std::array<float, kFftLengthBy2Plus1>& render_power,
+    const RenderSignalAnalyzer& render_signal_analyzer,
+    const SubtractorOutput& subtractor_output,
+    rtc::ArrayView<const float> erl,
+    size_t size_partitions,
+    bool saturated_capture_signal,
+    bool disallow_leakage_diverged,
+    FftData* gain_fft) {
+  RTC_DCHECK(gain_fft);
+  // Introducing shorter notation to improve readability.
+  const FftData& E_refined = subtractor_output.E_refined;
+  const auto& E2_refined = subtractor_output.E2_refined;
+  const auto& E2_coarse = subtractor_output.E2_coarse;
+  FftData* G = gain_fft;
+  const auto& X2 = render_power;
+
+  ++call_counter_;
+
+  UpdateCurrentConfig();
+
+  if (render_signal_analyzer.PoorSignalExcitation()) {
+    poor_excitation_counter_ = 0;
+  }
+
+  // Do not update the filter if the render is not sufficiently excited.
+  if (++poor_excitation_counter_ < size_partitions ||
+      saturated_capture_signal || call_counter_ <= size_partitions) {
+    G->re.fill(0.f);
+    G->im.fill(0.f);
+  } else {
+    // Corresponds to WGN of power -39 dBFS.
+    std::array<float, kFftLengthBy2Plus1> mu;
+    // mu = H_error / (0.5* H_error* X2 + n * E2).
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      if (X2[k] >= current_config_.noise_gate) {
+        mu[k] = H_error_[k] /
+                (0.5f * H_error_[k] * X2[k] + size_partitions * E2_refined[k]);
+      } else {
+        mu[k] = 0.f;
+      }
+    }
+
+    // Avoid updating the filter close to narrow bands in the render signals.
+    render_signal_analyzer.MaskRegionsAroundNarrowBands(&mu);
+
+    // H_error = H_error - 0.5 * mu * X2 * H_error.
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      H_error_[k] -= 0.5f * mu[k] * X2[k] * H_error_[k];
+    }
+
+    // G = mu * E.
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      G->re[k] = mu[k] * E_refined.re[k];
+      G->im[k] = mu[k] * E_refined.im[k];
+    }
+  }
+
+  // H_error = H_error + factor * erl.
+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+    if (E2_refined[k] <= E2_coarse[k] || disallow_leakage_diverged) {
+      H_error_[k] += current_config_.leakage_converged * erl[k];
+    } else {
+      H_error_[k] += current_config_.leakage_diverged * erl[k];
+    }
+
+    H_error_[k] = std::max(H_error_[k], current_config_.error_floor);
+    H_error_[k] = std::min(H_error_[k], current_config_.error_ceil);
+  }
+
+  data_dumper_->DumpRaw("aec3_refined_gain_H_error", H_error_);
+}
+
+void RefinedFilterUpdateGain::UpdateCurrentConfig() {
+  RTC_DCHECK_GE(config_change_duration_blocks_, config_change_counter_);
+  if (config_change_counter_ > 0) {
+    if (--config_change_counter_ > 0) {
+      auto average = [](float from, float to, float from_weight) {
+        return from * from_weight + to * (1.f - from_weight);
+      };
+
+      float change_factor =
+          config_change_counter_ * one_by_config_change_duration_blocks_;
+
+      current_config_.leakage_converged =
+          average(old_target_config_.leakage_converged,
+                  target_config_.leakage_converged, change_factor);
+      current_config_.leakage_diverged =
+          average(old_target_config_.leakage_diverged,
+                  target_config_.leakage_diverged, change_factor);
+      current_config_.error_floor =
+          average(old_target_config_.error_floor, target_config_.error_floor,
+                  change_factor);
+      current_config_.error_ceil =
+          average(old_target_config_.error_ceil, target_config_.error_ceil,
+                  change_factor);
+      current_config_.noise_gate =
+          average(old_target_config_.noise_gate, target_config_.noise_gate,
+                  change_factor);
+    } else {
+      current_config_ = old_target_config_ = target_config_;
+    }
+  }
+  RTC_DCHECK_LE(0, config_change_counter_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain.h b/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain.h
new file mode 100644
index 0000000000..1a68ebc296
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REFINED_FILTER_UPDATE_GAIN_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_REFINED_FILTER_UPDATE_GAIN_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <atomic>
+#include <memory>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+
+class AdaptiveFirFilter;
+class ApmDataDumper;
+struct EchoPathVariability;
+struct FftData;
+class RenderSignalAnalyzer;
+struct SubtractorOutput;
+
+// Provides functionality for  computing the adaptive gain for the refined
+// filter.
+class RefinedFilterUpdateGain {
+ public:
+  RefinedFilterUpdateGain(
+      const EchoCanceller3Config::Filter::RefinedConfiguration& config,
+      size_t config_change_duration_blocks);
+  ~RefinedFilterUpdateGain();
+
+  RefinedFilterUpdateGain(const RefinedFilterUpdateGain&) = delete;
+  RefinedFilterUpdateGain& operator=(const RefinedFilterUpdateGain&) = delete;
+
+  // Takes action in the case of a known echo path change.
+  void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
+
+  // Computes the gain.
+  void Compute(const std::array<float, kFftLengthBy2Plus1>& render_power,
+               const RenderSignalAnalyzer& render_signal_analyzer,
+               const SubtractorOutput& subtractor_output,
+               rtc::ArrayView<const float> erl,
+               size_t size_partitions,
+               bool saturated_capture_signal,
+               bool disallow_leakage_diverged,
+               FftData* gain_fft);
+
+  // Sets a new config.
+  void SetConfig(
+      const EchoCanceller3Config::Filter::RefinedConfiguration& config,
+      bool immediate_effect) {
+    if (immediate_effect) {
+      old_target_config_ = current_config_ = target_config_ = config;
+      config_change_counter_ = 0;
+    } else {
+      old_target_config_ = current_config_;
+      target_config_ = config;
+      config_change_counter_ = config_change_duration_blocks_;
+    }
+  }
+
+ private:
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  const int config_change_duration_blocks_;
+  float one_by_config_change_duration_blocks_;
+  EchoCanceller3Config::Filter::RefinedConfiguration current_config_;
+  EchoCanceller3Config::Filter::RefinedConfiguration target_config_;
+  EchoCanceller3Config::Filter::RefinedConfiguration old_target_config_;
+  std::array<float, kFftLengthBy2Plus1> H_error_;
+  size_t poor_excitation_counter_;
+  size_t call_counter_ = 0;
+  int config_change_counter_ = 0;
+
+  // Updates the current config towards the target config.
+  void UpdateCurrentConfig();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REFINED_FILTER_UPDATE_GAIN_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
new file mode 100644
index 0000000000..c77c5b53d5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
@@ -0,0 +1,392 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/refined_filter_update_gain.h"
+
+#include <algorithm>
+#include <numeric>
+#include <string>
+#include <vector>
+
+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
+#include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/coarse_filter_update_gain.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+#include "modules/audio_processing/aec3/subtractor_output.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+// Method for performing the simulations needed to test the refined filter
+// update gain functionality.
+void RunFilterUpdateTest(int num_blocks_to_process,
+                         size_t delay_samples,
+                         int filter_length_blocks,
+                         const std::vector<int>& blocks_with_echo_path_changes,
+                         const std::vector<int>& blocks_with_saturation,
+                         bool use_silent_render_in_second_half,
+                         std::array<float, kBlockSize>* e_last_block,
+                         std::array<float, kBlockSize>* y_last_block,
+                         FftData* G_last_block) {
+  ApmDataDumper data_dumper(42);
+  Aec3Optimization optimization = DetectOptimization();
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  EchoCanceller3Config config;
+  config.filter.refined.length_blocks = filter_length_blocks;
+  config.filter.coarse.length_blocks = filter_length_blocks;
+  AdaptiveFirFilter refined_filter(
+      config.filter.refined.length_blocks, config.filter.refined.length_blocks,
+      config.filter.config_change_duration_blocks, kNumRenderChannels,
+      optimization, &data_dumper);
+  AdaptiveFirFilter coarse_filter(
+      config.filter.coarse.length_blocks, config.filter.coarse.length_blocks,
+      config.filter.config_change_duration_blocks, kNumRenderChannels,
+      optimization, &data_dumper);
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> H2(
+      kNumCaptureChannels, std::vector<std::array<float, kFftLengthBy2Plus1>>(
+                               refined_filter.max_filter_size_partitions(),
+                               std::array<float, kFftLengthBy2Plus1>()));
+  for (auto& H2_ch : H2) {
+    for (auto& H2_k : H2_ch) {
+      H2_k.fill(0.f);
+    }
+  }
+  std::vector<std::vector<float>> h(
+      kNumCaptureChannels,
+      std::vector<float>(
+          GetTimeDomainLength(refined_filter.max_filter_size_partitions()),
+          0.f));
+
+  Aec3Fft fft;
+  std::array<float, kBlockSize> x_old;
+  x_old.fill(0.f);
+  CoarseFilterUpdateGain coarse_gain(
+      config.filter.coarse, config.filter.config_change_duration_blocks);
+  RefinedFilterUpdateGain refined_gain(
+      config.filter.refined, config.filter.config_change_duration_blocks);
+  Random random_generator(42U);
+  Block x(kNumBands, kNumRenderChannels);
+  std::vector<float> y(kBlockSize, 0.f);
+  config.delay.default_delay = 1;
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, kNumRenderChannels));
+  AecState aec_state(config, kNumCaptureChannels);
+  RenderSignalAnalyzer render_signal_analyzer(config);
+  absl::optional<DelayEstimate> delay_estimate;
+  std::array<float, kFftLength> s_scratch;
+  std::array<float, kBlockSize> s;
+  FftData S;
+  FftData G;
+  std::vector<SubtractorOutput> output(kNumCaptureChannels);
+  for (auto& subtractor_output : output) {
+    subtractor_output.Reset();
+  }
+  FftData& E_refined = output[0].E_refined;
+  FftData E_coarse;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(kNumCaptureChannels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2_refined(
+      kNumCaptureChannels);
+  std::array<float, kBlockSize>& e_refined = output[0].e_refined;
+  std::array<float, kBlockSize>& e_coarse = output[0].e_coarse;
+  for (auto& Y2_ch : Y2) {
+    Y2_ch.fill(0.f);
+  }
+
+  constexpr float kScale = 1.0f / kFftLengthBy2;
+
+  DelayBuffer<float> delay_buffer(delay_samples);
+  for (int k = 0; k < num_blocks_to_process; ++k) {
+    // Handle echo path changes.
+    if (std::find(blocks_with_echo_path_changes.begin(),
+                  blocks_with_echo_path_changes.end(),
+                  k) != blocks_with_echo_path_changes.end()) {
+      refined_filter.HandleEchoPathChange();
+    }
+
+    // Handle saturation.
+    const bool saturation =
+        std::find(blocks_with_saturation.begin(), blocks_with_saturation.end(),
+                  k) != blocks_with_saturation.end();
+
+    // Create the render signal.
+    if (use_silent_render_in_second_half && k > num_blocks_to_process / 2) {
+      for (int band = 0; band < x.NumBands(); ++band) {
+        for (int channel = 0; channel < x.NumChannels(); ++channel) {
+          std::fill(x.begin(band, channel), x.end(band, channel), 0.f);
+        }
+      }
+    } else {
+      for (int band = 0; band < x.NumChannels(); ++band) {
+        for (int channel = 0; channel < x.NumChannels(); ++channel) {
+          RandomizeSampleVector(&random_generator, x.View(band, channel));
+        }
+      }
+    }
+    delay_buffer.Delay(x.View(/*band=*/0, /*channel=*/0), y);
+
+    render_delay_buffer->Insert(x);
+    if (k == 0) {
+      render_delay_buffer->Reset();
+    }
+    render_delay_buffer->PrepareCaptureProcessing();
+
+    render_signal_analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
+                                  aec_state.MinDirectPathFilterDelay());
+
+    // Apply the refined filter.
+    refined_filter.Filter(*render_delay_buffer->GetRenderBuffer(), &S);
+    fft.Ifft(S, &s_scratch);
+    std::transform(y.begin(), y.end(), s_scratch.begin() + kFftLengthBy2,
+                   e_refined.begin(),
+                   [&](float a, float b) { return a - b * kScale; });
+    std::for_each(e_refined.begin(), e_refined.end(),
+                  [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
+    fft.ZeroPaddedFft(e_refined, Aec3Fft::Window::kRectangular, &E_refined);
+    for (size_t k = 0; k < kBlockSize; ++k) {
+      s[k] = kScale * s_scratch[k + kFftLengthBy2];
+    }
+
+    // Apply the coarse filter.
+    coarse_filter.Filter(*render_delay_buffer->GetRenderBuffer(), &S);
+    fft.Ifft(S, &s_scratch);
+    std::transform(y.begin(), y.end(), s_scratch.begin() + kFftLengthBy2,
+                   e_coarse.begin(),
+                   [&](float a, float b) { return a - b * kScale; });
+    std::for_each(e_coarse.begin(), e_coarse.end(),
+                  [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
+    fft.ZeroPaddedFft(e_coarse, Aec3Fft::Window::kRectangular, &E_coarse);
+
+    // Compute spectra for future use.
+    E_refined.Spectrum(Aec3Optimization::kNone, output[0].E2_refined);
+    E_coarse.Spectrum(Aec3Optimization::kNone, output[0].E2_coarse);
+
+    // Adapt the coarse filter.
+    std::array<float, kFftLengthBy2Plus1> render_power;
+    render_delay_buffer->GetRenderBuffer()->SpectralSum(
+        coarse_filter.SizePartitions(), &render_power);
+    coarse_gain.Compute(render_power, render_signal_analyzer, E_coarse,
+                        coarse_filter.SizePartitions(), saturation, &G);
+    coarse_filter.Adapt(*render_delay_buffer->GetRenderBuffer(), G);
+
+    // Adapt the refined filter
+    render_delay_buffer->GetRenderBuffer()->SpectralSum(
+        refined_filter.SizePartitions(), &render_power);
+
+    std::array<float, kFftLengthBy2Plus1> erl;
+    ComputeErl(optimization, H2[0], erl);
+    refined_gain.Compute(render_power, render_signal_analyzer, output[0], erl,
+                         refined_filter.SizePartitions(), saturation, false,
+                         &G);
+    refined_filter.Adapt(*render_delay_buffer->GetRenderBuffer(), G, &h[0]);
+
+    // Update the delay.
+    aec_state.HandleEchoPathChange(EchoPathVariability(
+        false, EchoPathVariability::DelayAdjustment::kNone, false));
+    refined_filter.ComputeFrequencyResponse(&H2[0]);
+    std::copy(output[0].E2_refined.begin(), output[0].E2_refined.end(),
+              E2_refined[0].begin());
+    aec_state.Update(delay_estimate, H2, h,
+                     *render_delay_buffer->GetRenderBuffer(), E2_refined, Y2,
+                     output);
+  }
+
+  std::copy(e_refined.begin(), e_refined.end(), e_last_block->begin());
+  std::copy(y.begin(), y.end(), y_last_block->begin());
+  std::copy(G.re.begin(), G.re.end(), G_last_block->re.begin());
+  std::copy(G.im.begin(), G.im.end(), G_last_block->im.begin());
+}
+
+std::string ProduceDebugText(int filter_length_blocks) {
+  rtc::StringBuilder ss;
+  ss << "Length: " << filter_length_blocks;
+  return ss.Release();
+}
+
+std::string ProduceDebugText(size_t delay, int filter_length_blocks) {
+  rtc::StringBuilder ss;
+  ss << "Delay: " << delay << ", ";
+  ss << ProduceDebugText(filter_length_blocks);
+  return ss.Release();
+}
+
+}  // namespace
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies that the check for non-null output gain parameter works.
+TEST(RefinedFilterUpdateGainDeathTest, NullDataOutputGain) {
+  ApmDataDumper data_dumper(42);
+  EchoCanceller3Config config;
+  RenderSignalAnalyzer analyzer(config);
+  SubtractorOutput output;
+  RefinedFilterUpdateGain gain(config.filter.refined,
+                               config.filter.config_change_duration_blocks);
+  std::array<float, kFftLengthBy2Plus1> render_power;
+  render_power.fill(0.f);
+  std::array<float, kFftLengthBy2Plus1> erl;
+  erl.fill(0.f);
+  EXPECT_DEATH(
+      gain.Compute(render_power, analyzer, output, erl,
+                   config.filter.refined.length_blocks, false, false, nullptr),
+      "");
+}
+
+#endif
+
+// Verifies that the gain formed causes the filter using it to converge.
+TEST(RefinedFilterUpdateGain, GainCausesFilterToConverge) {
+  std::vector<int> blocks_with_echo_path_changes;
+  std::vector<int> blocks_with_saturation;
+  for (size_t filter_length_blocks : {12, 20, 30}) {
+    for (size_t delay_samples : {0, 64, 150, 200, 301}) {
+      SCOPED_TRACE(ProduceDebugText(delay_samples, filter_length_blocks));
+
+      std::array<float, kBlockSize> e;
+      std::array<float, kBlockSize> y;
+      FftData G;
+
+      RunFilterUpdateTest(600, delay_samples, filter_length_blocks,
+                          blocks_with_echo_path_changes, blocks_with_saturation,
+                          false, &e, &y, &G);
+
+      // Verify that the refined filter is able to perform well.
+      // Use different criteria to take overmodelling into account.
+      if (filter_length_blocks == 12) {
+        EXPECT_LT(1000 * std::inner_product(e.begin(), e.end(), e.begin(), 0.f),
+                  std::inner_product(y.begin(), y.end(), y.begin(), 0.f));
+      } else {
+        EXPECT_LT(std::inner_product(e.begin(), e.end(), e.begin(), 0.f),
+                  std::inner_product(y.begin(), y.end(), y.begin(), 0.f));
+      }
+    }
+  }
+}
+
+// Verifies that the magnitude of the gain on average decreases for a
+// persistently exciting signal.
+TEST(RefinedFilterUpdateGain, DecreasingGain) {
+  std::vector<int> blocks_with_echo_path_changes;
+  std::vector<int> blocks_with_saturation;
+
+  std::array<float, kBlockSize> e;
+  std::array<float, kBlockSize> y;
+  FftData G_a;
+  FftData G_b;
+  FftData G_c;
+  std::array<float, kFftLengthBy2Plus1> G_a_power;
+  std::array<float, kFftLengthBy2Plus1> G_b_power;
+  std::array<float, kFftLengthBy2Plus1> G_c_power;
+
+  RunFilterUpdateTest(250, 65, 12, blocks_with_echo_path_changes,
+                      blocks_with_saturation, false, &e, &y, &G_a);
+  RunFilterUpdateTest(500, 65, 12, blocks_with_echo_path_changes,
+                      blocks_with_saturation, false, &e, &y, &G_b);
+  RunFilterUpdateTest(750, 65, 12, blocks_with_echo_path_changes,
+                      blocks_with_saturation, false, &e, &y, &G_c);
+
+  G_a.Spectrum(Aec3Optimization::kNone, G_a_power);
+  G_b.Spectrum(Aec3Optimization::kNone, G_b_power);
+  G_c.Spectrum(Aec3Optimization::kNone, G_c_power);
+
+  EXPECT_GT(std::accumulate(G_a_power.begin(), G_a_power.end(), 0.),
+            std::accumulate(G_b_power.begin(), G_b_power.end(), 0.));
+
+  EXPECT_GT(std::accumulate(G_b_power.begin(), G_b_power.end(), 0.),
+            std::accumulate(G_c_power.begin(), G_c_power.end(), 0.));
+}
+
+// Verifies that the gain is zero when there is saturation and that the internal
+// error estimates cause the gain to increase after a period of saturation.
+TEST(RefinedFilterUpdateGain, SaturationBehavior) {
+  std::vector<int> blocks_with_echo_path_changes;
+  std::vector<int> blocks_with_saturation;
+  for (int k = 99; k < 200; ++k) {
+    blocks_with_saturation.push_back(k);
+  }
+
+  for (size_t filter_length_blocks : {12, 20, 30}) {
+    SCOPED_TRACE(ProduceDebugText(filter_length_blocks));
+    std::array<float, kBlockSize> e;
+    std::array<float, kBlockSize> y;
+    FftData G_a;
+    FftData G_b;
+    FftData G_a_ref;
+    G_a_ref.re.fill(0.f);
+    G_a_ref.im.fill(0.f);
+
+    std::array<float, kFftLengthBy2Plus1> G_a_power;
+    std::array<float, kFftLengthBy2Plus1> G_b_power;
+
+    RunFilterUpdateTest(100, 65, filter_length_blocks,
+                        blocks_with_echo_path_changes, blocks_with_saturation,
+                        false, &e, &y, &G_a);
+
+    EXPECT_EQ(G_a_ref.re, G_a.re);
+    EXPECT_EQ(G_a_ref.im, G_a.im);
+
+    RunFilterUpdateTest(99, 65, filter_length_blocks,
+                        blocks_with_echo_path_changes, blocks_with_saturation,
+                        false, &e, &y, &G_a);
+    RunFilterUpdateTest(201, 65, filter_length_blocks,
+                        blocks_with_echo_path_changes, blocks_with_saturation,
+                        false, &e, &y, &G_b);
+
+    G_a.Spectrum(Aec3Optimization::kNone, G_a_power);
+    G_b.Spectrum(Aec3Optimization::kNone, G_b_power);
+
+    EXPECT_LT(std::accumulate(G_a_power.begin(), G_a_power.end(), 0.),
+              std::accumulate(G_b_power.begin(), G_b_power.end(), 0.));
+  }
+}
+
+// Verifies that the gain increases after an echo path change.
+// TODO(peah): Correct and reactivate this test.
+TEST(RefinedFilterUpdateGain, DISABLED_EchoPathChangeBehavior) {
+  for (size_t filter_length_blocks : {12, 20, 30}) {
+    SCOPED_TRACE(ProduceDebugText(filter_length_blocks));
+    std::vector<int> blocks_with_echo_path_changes;
+    std::vector<int> blocks_with_saturation;
+    blocks_with_echo_path_changes.push_back(99);
+
+    std::array<float, kBlockSize> e;
+    std::array<float, kBlockSize> y;
+    FftData G_a;
+    FftData G_b;
+    std::array<float, kFftLengthBy2Plus1> G_a_power;
+    std::array<float, kFftLengthBy2Plus1> G_b_power;
+
+    RunFilterUpdateTest(100, 65, filter_length_blocks,
+                        blocks_with_echo_path_changes, blocks_with_saturation,
+                        false, &e, &y, &G_a);
+    RunFilterUpdateTest(101, 65, filter_length_blocks,
+                        blocks_with_echo_path_changes, blocks_with_saturation,
+                        false, &e, &y, &G_b);
+
+    G_a.Spectrum(Aec3Optimization::kNone, G_a_power);
+    G_b.Spectrum(Aec3Optimization::kNone, G_b_power);
+
+    EXPECT_LT(std::accumulate(G_a_power.begin(), G_a_power.end(), 0.),
+              std::accumulate(G_b_power.begin(), G_b_power.end(), 0.));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer.cc
new file mode 100644
index 0000000000..aa511e2b6b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/render_buffer.h"
+
+#include <algorithm>
+#include <functional>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+RenderBuffer::RenderBuffer(BlockBuffer* block_buffer,
+                           SpectrumBuffer* spectrum_buffer,
+                           FftBuffer* fft_buffer)
+    : block_buffer_(block_buffer),
+      spectrum_buffer_(spectrum_buffer),
+      fft_buffer_(fft_buffer) {
+  RTC_DCHECK(block_buffer_);
+  RTC_DCHECK(spectrum_buffer_);
+  RTC_DCHECK(fft_buffer_);
+  RTC_DCHECK_EQ(block_buffer_->buffer.size(), fft_buffer_->buffer.size());
+  RTC_DCHECK_EQ(spectrum_buffer_->buffer.size(), fft_buffer_->buffer.size());
+  RTC_DCHECK_EQ(spectrum_buffer_->read, fft_buffer_->read);
+  RTC_DCHECK_EQ(spectrum_buffer_->write, fft_buffer_->write);
+}
+
+RenderBuffer::~RenderBuffer() = default;
+
+void RenderBuffer::SpectralSum(
+    size_t num_spectra,
+    std::array<float, kFftLengthBy2Plus1>* X2) const {
+  X2->fill(0.f);
+  int position = spectrum_buffer_->read;
+  for (size_t j = 0; j < num_spectra; ++j) {
+    for (const auto& channel_spectrum : spectrum_buffer_->buffer[position]) {
+      for (size_t k = 0; k < X2->size(); ++k) {
+        (*X2)[k] += channel_spectrum[k];
+      }
+    }
+    position = spectrum_buffer_->IncIndex(position);
+  }
+}
+
+void RenderBuffer::SpectralSums(
+    size_t num_spectra_shorter,
+    size_t num_spectra_longer,
+    std::array<float, kFftLengthBy2Plus1>* X2_shorter,
+    std::array<float, kFftLengthBy2Plus1>* X2_longer) const {
+  RTC_DCHECK_LE(num_spectra_shorter, num_spectra_longer);
+  X2_shorter->fill(0.f);
+  int position = spectrum_buffer_->read;
+  size_t j = 0;
+  for (; j < num_spectra_shorter; ++j) {
+    for (const auto& channel_spectrum : spectrum_buffer_->buffer[position]) {
+      for (size_t k = 0; k < X2_shorter->size(); ++k) {
+        (*X2_shorter)[k] += channel_spectrum[k];
+      }
+    }
+    position = spectrum_buffer_->IncIndex(position);
+  }
+  std::copy(X2_shorter->begin(), X2_shorter->end(), X2_longer->begin());
+  for (; j < num_spectra_longer; ++j) {
+    for (const auto& channel_spectrum : spectrum_buffer_->buffer[position]) {
+      for (size_t k = 0; k < X2_longer->size(); ++k) {
+        (*X2_longer)[k] += channel_spectrum[k];
+      }
+    }
+    position = spectrum_buffer_->IncIndex(position);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer.h
new file mode 100644
index 0000000000..8adc996087
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer.h
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/block_buffer.h"
+#include "modules/audio_processing/aec3/fft_buffer.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "modules/audio_processing/aec3/spectrum_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Provides a buffer of the render data for the echo remover.
+class RenderBuffer {
+ public:
+  RenderBuffer(BlockBuffer* block_buffer,
+               SpectrumBuffer* spectrum_buffer,
+               FftBuffer* fft_buffer);
+
+  RenderBuffer() = delete;
+  RenderBuffer(const RenderBuffer&) = delete;
+  RenderBuffer& operator=(const RenderBuffer&) = delete;
+
+  ~RenderBuffer();
+
+  // Get a block.
+  const Block& GetBlock(int buffer_offset_blocks) const {
+    int position =
+        block_buffer_->OffsetIndex(block_buffer_->read, buffer_offset_blocks);
+    return block_buffer_->buffer[position];
+  }
+
+  // Get the spectrum from one of the FFTs in the buffer.
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Spectrum(
+      int buffer_offset_ffts) const {
+    int position = spectrum_buffer_->OffsetIndex(spectrum_buffer_->read,
+                                                 buffer_offset_ffts);
+    return spectrum_buffer_->buffer[position];
+  }
+
+  // Returns the circular fft buffer.
+  rtc::ArrayView<const std::vector<FftData>> GetFftBuffer() const {
+    return fft_buffer_->buffer;
+  }
+
+  // Returns the current position in the circular buffer.
+  size_t Position() const {
+    RTC_DCHECK_EQ(spectrum_buffer_->read, fft_buffer_->read);
+    RTC_DCHECK_EQ(spectrum_buffer_->write, fft_buffer_->write);
+    return fft_buffer_->read;
+  }
+
+  // Returns the sum of the spectrums for a certain number of FFTs.
+  void SpectralSum(size_t num_spectra,
+                   std::array<float, kFftLengthBy2Plus1>* X2) const;
+
+  // Returns the sums of the spectrums for two numbers of FFTs.
+  void SpectralSums(size_t num_spectra_shorter,
+                    size_t num_spectra_longer,
+                    std::array<float, kFftLengthBy2Plus1>* X2_shorter,
+                    std::array<float, kFftLengthBy2Plus1>* X2_longer) const;
+
+  // Gets the recent activity seen in the render signal.
+  bool GetRenderActivity() const { return render_activity_; }
+
+  // Specifies the recent activity seen in the render signal.
+  void SetRenderActivity(bool activity) { render_activity_ = activity; }
+
+  // Returns the headroom between the write and the read positions in the
+  // buffer.
+  int Headroom() const {
+    // The write and read indices are decreased over time.
+    int headroom =
+        fft_buffer_->write < fft_buffer_->read
+            ? fft_buffer_->read - fft_buffer_->write
+            : fft_buffer_->size - fft_buffer_->write + fft_buffer_->read;
+
+    RTC_DCHECK_LE(0, headroom);
+    RTC_DCHECK_GE(fft_buffer_->size, headroom);
+
+    return headroom;
+  }
+
+  // Returns a reference to the spectrum buffer.
+  const SpectrumBuffer& GetSpectrumBuffer() const { return *spectrum_buffer_; }
+
+  // Returns a reference to the block buffer.
+  const BlockBuffer& GetBlockBuffer() const { return *block_buffer_; }
+
+ private:
+  const BlockBuffer* const block_buffer_;
+  const SpectrumBuffer* const spectrum_buffer_;
+  const FftBuffer* const fft_buffer_;
+  bool render_activity_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_gn/moz.build
new file mode 100644
index 0000000000..b7a10f5d7c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("render_buffer_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_unittest.cc
new file mode 100644
index 0000000000..5d9d646e76
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_buffer_unittest.cc
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/render_buffer.h"
+
+#include <algorithm>
+#include <functional>
+#include <vector>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for non-null fft buffer.
+TEST(RenderBufferDeathTest, NullExternalFftBuffer) {
+  BlockBuffer block_buffer(10, 3, 1);
+  SpectrumBuffer spectrum_buffer(10, 1);
+  EXPECT_DEATH(RenderBuffer(&block_buffer, &spectrum_buffer, nullptr), "");
+}
+
+// Verifies the check for non-null spectrum buffer.
+TEST(RenderBufferDeathTest, NullExternalSpectrumBuffer) {
+  FftBuffer fft_buffer(10, 1);
+  BlockBuffer block_buffer(10, 3, 1);
+  EXPECT_DEATH(RenderBuffer(&block_buffer, nullptr, &fft_buffer), "");
+}
+
+// Verifies the check for non-null block buffer.
+TEST(RenderBufferDeathTest, NullExternalBlockBuffer) {
+  FftBuffer fft_buffer(10, 1);
+  SpectrumBuffer spectrum_buffer(10, 1);
+  EXPECT_DEATH(RenderBuffer(nullptr, &spectrum_buffer, &fft_buffer), "");
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer.cc
new file mode 100644
index 0000000000..ec5d35507e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer.cc
@@ -0,0 +1,519 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <atomic>
+#include <cmath>
+#include <memory>
+#include <numeric>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/alignment_mixer.h"
+#include "modules/audio_processing/aec3/block_buffer.h"
+#include "modules/audio_processing/aec3/decimator.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/aec3/fft_buffer.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/spectrum_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+bool UpdateCaptureCallCounterOnSkippedBlocks() {
+  return !field_trial::IsEnabled(
+      "WebRTC-Aec3RenderBufferCallCounterUpdateKillSwitch");
+}
+
+class RenderDelayBufferImpl final : public RenderDelayBuffer {
+ public:
+  RenderDelayBufferImpl(const EchoCanceller3Config& config,
+                        int sample_rate_hz,
+                        size_t num_render_channels);
+  RenderDelayBufferImpl() = delete;
+  ~RenderDelayBufferImpl() override;
+
+  void Reset() override;
+  BufferingEvent Insert(const Block& block) override;
+  BufferingEvent PrepareCaptureProcessing() override;
+  void HandleSkippedCaptureProcessing() override;
+  bool AlignFromDelay(size_t delay) override;
+  void AlignFromExternalDelay() override;
+  size_t Delay() const override { return ComputeDelay(); }
+  size_t MaxDelay() const override {
+    return blocks_.buffer.size() - 1 - buffer_headroom_;
+  }
+  RenderBuffer* GetRenderBuffer() override { return &echo_remover_buffer_; }
+
+  const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const override {
+    return low_rate_;
+  }
+
+  int BufferLatency() const;
+  void SetAudioBufferDelay(int delay_ms) override;
+  bool HasReceivedBufferDelay() override;
+
+ private:
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  const Aec3Optimization optimization_;
+  const EchoCanceller3Config config_;
+  const bool update_capture_call_counter_on_skipped_blocks_;
+  const float render_linear_amplitude_gain_;
+  const rtc::LoggingSeverity delay_log_level_;
+  size_t down_sampling_factor_;
+  const int sub_block_size_;
+  BlockBuffer blocks_;
+  SpectrumBuffer spectra_;
+  FftBuffer ffts_;
+  absl::optional<size_t> delay_;
+  RenderBuffer echo_remover_buffer_;
+  DownsampledRenderBuffer low_rate_;
+  AlignmentMixer render_mixer_;
+  Decimator render_decimator_;
+  const Aec3Fft fft_;
+  std::vector<float> render_ds_;
+  const int buffer_headroom_;
+  bool last_call_was_render_ = false;
+  int num_api_calls_in_a_row_ = 0;
+  int max_observed_jitter_ = 1;
+  int64_t capture_call_counter_ = 0;
+  int64_t render_call_counter_ = 0;
+  bool render_activity_ = false;
+  size_t render_activity_counter_ = 0;
+  absl::optional<int> external_audio_buffer_delay_;
+  bool external_audio_buffer_delay_verified_after_reset_ = false;
+  size_t min_latency_blocks_ = 0;
+  size_t excess_render_detection_counter_ = 0;
+
+  int MapDelayToTotalDelay(size_t delay) const;
+  int ComputeDelay() const;
+  void ApplyTotalDelay(int delay);
+  void InsertBlock(const Block& block, int previous_write);
+  bool DetectActiveRender(rtc::ArrayView<const float> x) const;
+  bool DetectExcessRenderBlocks();
+  void IncrementWriteIndices();
+  void IncrementLowRateReadIndices();
+  void IncrementReadIndices();
+  bool RenderOverrun();
+  bool RenderUnderrun();
+};
+
+std::atomic<int> RenderDelayBufferImpl::instance_count_ = 0;
+
+RenderDelayBufferImpl::RenderDelayBufferImpl(const EchoCanceller3Config& config,
+                                             int sample_rate_hz,
+                                             size_t num_render_channels)
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      optimization_(DetectOptimization()),
+      config_(config),
+      update_capture_call_counter_on_skipped_blocks_(
+          UpdateCaptureCallCounterOnSkippedBlocks()),
+      render_linear_amplitude_gain_(
+          std::pow(10.0f, config_.render_levels.render_power_gain_db / 20.f)),
+      delay_log_level_(config_.delay.log_warning_on_delay_changes
+                           ? rtc::LS_WARNING
+                           : rtc::LS_VERBOSE),
+      down_sampling_factor_(config.delay.down_sampling_factor),
+      sub_block_size_(static_cast<int>(down_sampling_factor_ > 0
+                                           ? kBlockSize / down_sampling_factor_
+                                           : kBlockSize)),
+      blocks_(GetRenderDelayBufferSize(down_sampling_factor_,
+                                       config.delay.num_filters,
+                                       config.filter.refined.length_blocks),
+              NumBandsForRate(sample_rate_hz),
+              num_render_channels),
+      spectra_(blocks_.buffer.size(), num_render_channels),
+      ffts_(blocks_.buffer.size(), num_render_channels),
+      delay_(config_.delay.default_delay),
+      echo_remover_buffer_(&blocks_, &spectra_, &ffts_),
+      low_rate_(GetDownSampledBufferSize(down_sampling_factor_,
+                                         config.delay.num_filters)),
+      render_mixer_(num_render_channels, config.delay.render_alignment_mixing),
+      render_decimator_(down_sampling_factor_),
+      fft_(),
+      render_ds_(sub_block_size_, 0.f),
+      buffer_headroom_(config.filter.refined.length_blocks) {
+  RTC_DCHECK_EQ(blocks_.buffer.size(), ffts_.buffer.size());
+  RTC_DCHECK_EQ(spectra_.buffer.size(), ffts_.buffer.size());
+  for (size_t i = 0; i < blocks_.buffer.size(); ++i) {
+    RTC_DCHECK_EQ(blocks_.buffer[i].NumChannels(), ffts_.buffer[i].size());
+    RTC_DCHECK_EQ(spectra_.buffer[i].size(), ffts_.buffer[i].size());
+  }
+
+  Reset();
+}
+
+RenderDelayBufferImpl::~RenderDelayBufferImpl() = default;
+
+// Resets the buffer delays and clears the reported delays.
+void RenderDelayBufferImpl::Reset() {
+  last_call_was_render_ = false;
+  num_api_calls_in_a_row_ = 1;
+  min_latency_blocks_ = 0;
+  excess_render_detection_counter_ = 0;
+
+  // Initialize the read index to one sub-block before the write index.
+  low_rate_.read = low_rate_.OffsetIndex(low_rate_.write, sub_block_size_);
+
+  // Check for any external audio buffer delay and whether it is feasible.
+  if (external_audio_buffer_delay_) {
+    const int headroom = 2;
+    size_t audio_buffer_delay_to_set;
+    // Minimum delay is 1 (like the low-rate render buffer).
+    if (*external_audio_buffer_delay_ <= headroom) {
+      audio_buffer_delay_to_set = 1;
+    } else {
+      audio_buffer_delay_to_set = *external_audio_buffer_delay_ - headroom;
+    }
+
+    audio_buffer_delay_to_set = std::min(audio_buffer_delay_to_set, MaxDelay());
+
+    // When an external delay estimate is available, use that delay as the
+    // initial render buffer delay.
+    ApplyTotalDelay(audio_buffer_delay_to_set);
+    delay_ = ComputeDelay();
+
+    external_audio_buffer_delay_verified_after_reset_ = false;
+  } else {
+    // If an external delay estimate is not available, use that delay as the
+    // initial delay. Set the render buffer delays to the default delay.
+    ApplyTotalDelay(config_.delay.default_delay);
+
+    // Unset the delays which are set by AlignFromDelay.
+    delay_ = absl::nullopt;
+  }
+}
+
+// Inserts a new block into the render buffers.
+RenderDelayBuffer::BufferingEvent RenderDelayBufferImpl::Insert(
+    const Block& block) {
+  ++render_call_counter_;
+  if (delay_) {
+    if (!last_call_was_render_) {
+      last_call_was_render_ = true;
+      num_api_calls_in_a_row_ = 1;
+    } else {
+      if (++num_api_calls_in_a_row_ > max_observed_jitter_) {
+        max_observed_jitter_ = num_api_calls_in_a_row_;
+        RTC_LOG_V(delay_log_level_)
+            << "New max number api jitter observed at render block "
+            << render_call_counter_ << ":  " << num_api_calls_in_a_row_
+            << " blocks";
+      }
+    }
+  }
+
+  // Increase the write indices to where the new blocks should be written.
+  const int previous_write = blocks_.write;
+  IncrementWriteIndices();
+
+  // Allow overrun and do a reset when render overrun occurrs due to more render
+  // data being inserted than capture data is received.
+  BufferingEvent event =
+      RenderOverrun() ? BufferingEvent::kRenderOverrun : BufferingEvent::kNone;
+
+  // Detect and update render activity.
+  if (!render_activity_) {
+    render_activity_counter_ +=
+        DetectActiveRender(block.View(/*band=*/0, /*channel=*/0)) ? 1 : 0;
+    render_activity_ = render_activity_counter_ >= 20;
+  }
+
+  // Insert the new render block into the specified position.
+  InsertBlock(block, previous_write);
+
+  if (event != BufferingEvent::kNone) {
+    Reset();
+  }
+
+  return event;
+}
+
+void RenderDelayBufferImpl::HandleSkippedCaptureProcessing() {
+  if (update_capture_call_counter_on_skipped_blocks_) {
+    ++capture_call_counter_;
+  }
+}
+
+// Prepares the render buffers for processing another capture block.
+RenderDelayBuffer::BufferingEvent
+RenderDelayBufferImpl::PrepareCaptureProcessing() {
+  RenderDelayBuffer::BufferingEvent event = BufferingEvent::kNone;
+  ++capture_call_counter_;
+
+  if (delay_) {
+    if (last_call_was_render_) {
+      last_call_was_render_ = false;
+      num_api_calls_in_a_row_ = 1;
+    } else {
+      if (++num_api_calls_in_a_row_ > max_observed_jitter_) {
+        max_observed_jitter_ = num_api_calls_in_a_row_;
+        RTC_LOG_V(delay_log_level_)
+            << "New max number api jitter observed at capture block "
+            << capture_call_counter_ << ":  " << num_api_calls_in_a_row_
+            << " blocks";
+      }
+    }
+  }
+
+  if (DetectExcessRenderBlocks()) {
+    // Too many render blocks compared to capture blocks. Risk of delay ending
+    // up before the filter used by the delay estimator.
+    RTC_LOG_V(delay_log_level_)
+        << "Excess render blocks detected at block " << capture_call_counter_;
+    Reset();
+    event = BufferingEvent::kRenderOverrun;
+  } else if (RenderUnderrun()) {
+    // Don't increment the read indices of the low rate buffer if there is a
+    // render underrun.
+    RTC_LOG_V(delay_log_level_)
+        << "Render buffer underrun detected at block " << capture_call_counter_;
+    IncrementReadIndices();
+    // Incrementing the buffer index without increasing the low rate buffer
+    // index means that the delay is reduced by one.
+    if (delay_ && *delay_ > 0)
+      delay_ = *delay_ - 1;
+    event = BufferingEvent::kRenderUnderrun;
+  } else {
+    // Increment the read indices in the render buffers to point to the most
+    // recent block to use in the capture processing.
+    IncrementLowRateReadIndices();
+    IncrementReadIndices();
+  }
+
+  echo_remover_buffer_.SetRenderActivity(render_activity_);
+  if (render_activity_) {
+    render_activity_counter_ = 0;
+    render_activity_ = false;
+  }
+
+  return event;
+}
+
+// Sets the delay and returns a bool indicating whether the delay was changed.
+bool RenderDelayBufferImpl::AlignFromDelay(size_t delay) {
+  RTC_DCHECK(!config_.delay.use_external_delay_estimator);
+  if (!external_audio_buffer_delay_verified_after_reset_ &&
+      external_audio_buffer_delay_ && delay_) {
+    int difference = static_cast<int>(delay) - static_cast<int>(*delay_);
+    RTC_LOG_V(delay_log_level_)
+        << "Mismatch between first estimated delay after reset "
+           "and externally reported audio buffer delay: "
+        << difference << " blocks";
+    external_audio_buffer_delay_verified_after_reset_ = true;
+  }
+  if (delay_ && *delay_ == delay) {
+    return false;
+  }
+  delay_ = delay;
+
+  // Compute the total delay and limit the delay to the allowed range.
+  int total_delay = MapDelayToTotalDelay(*delay_);
+  total_delay =
+      std::min(MaxDelay(), static_cast<size_t>(std::max(total_delay, 0)));
+
+  // Apply the delay to the buffers.
+  ApplyTotalDelay(total_delay);
+  return true;
+}
+
+void RenderDelayBufferImpl::SetAudioBufferDelay(int delay_ms) {
+  if (!external_audio_buffer_delay_) {
+    RTC_LOG_V(delay_log_level_)
+        << "Receiving a first externally reported audio buffer delay of "
+        << delay_ms << " ms.";
+  }
+
+  // Convert delay from milliseconds to blocks (rounded down).
+  external_audio_buffer_delay_ = delay_ms / 4;
+}
+
+bool RenderDelayBufferImpl::HasReceivedBufferDelay() {
+  return external_audio_buffer_delay_.has_value();
+}
+
+// Maps the externally computed delay to the delay used internally.
+int RenderDelayBufferImpl::MapDelayToTotalDelay(
+    size_t external_delay_blocks) const {
+  const int latency_blocks = BufferLatency();
+  return latency_blocks + static_cast<int>(external_delay_blocks);
+}
+
+// Returns the delay (not including call jitter).
+int RenderDelayBufferImpl::ComputeDelay() const {
+  const int latency_blocks = BufferLatency();
+  int internal_delay = spectra_.read >= spectra_.write
+                           ? spectra_.read - spectra_.write
+                           : spectra_.size + spectra_.read - spectra_.write;
+
+  return internal_delay - latency_blocks;
+}
+
+// Set the read indices according to the delay.
+void RenderDelayBufferImpl::ApplyTotalDelay(int delay) {
+  RTC_LOG_V(delay_log_level_)
+      << "Applying total delay of " << delay << " blocks.";
+  blocks_.read = blocks_.OffsetIndex(blocks_.write, -delay);
+  spectra_.read = spectra_.OffsetIndex(spectra_.write, delay);
+  ffts_.read = ffts_.OffsetIndex(ffts_.write, delay);
+}
+
+void RenderDelayBufferImpl::AlignFromExternalDelay() {
+  RTC_DCHECK(config_.delay.use_external_delay_estimator);
+  if (external_audio_buffer_delay_) {
+    const int64_t delay = render_call_counter_ - capture_call_counter_ +
+                          *external_audio_buffer_delay_;
+    const int64_t delay_with_headroom =
+        delay - config_.delay.delay_headroom_samples / kBlockSize;
+    ApplyTotalDelay(delay_with_headroom);
+  }
+}
+
+// Inserts a block into the render buffers.
+void RenderDelayBufferImpl::InsertBlock(const Block& block,
+                                        int previous_write) {
+  auto& b = blocks_;
+  auto& lr = low_rate_;
+  auto& ds = render_ds_;
+  auto& f = ffts_;
+  auto& s = spectra_;
+  const size_t num_bands = b.buffer[b.write].NumBands();
+  const size_t num_render_channels = b.buffer[b.write].NumChannels();
+  RTC_DCHECK_EQ(block.NumBands(), num_bands);
+  RTC_DCHECK_EQ(block.NumChannels(), num_render_channels);
+  for (size_t band = 0; band < num_bands; ++band) {
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      std::copy(block.begin(band, ch), block.end(band, ch),
+                b.buffer[b.write].begin(band, ch));
+    }
+  }
+
+  if (render_linear_amplitude_gain_ != 1.f) {
+    for (size_t band = 0; band < num_bands; ++band) {
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        rtc::ArrayView<float, kBlockSize> b_view =
+            b.buffer[b.write].View(band, ch);
+        for (float& sample : b_view) {
+          sample *= render_linear_amplitude_gain_;
+        }
+      }
+    }
+  }
+
+  std::array<float, kBlockSize> downmixed_render;
+  render_mixer_.ProduceOutput(b.buffer[b.write], downmixed_render);
+  render_decimator_.Decimate(downmixed_render, ds);
+  data_dumper_->DumpWav("aec3_render_decimator_output", ds.size(), ds.data(),
+                        16000 / down_sampling_factor_, 1);
+  std::copy(ds.rbegin(), ds.rend(), lr.buffer.begin() + lr.write);
+  for (int channel = 0; channel < b.buffer[b.write].NumChannels(); ++channel) {
+    fft_.PaddedFft(b.buffer[b.write].View(/*band=*/0, channel),
+                   b.buffer[previous_write].View(/*band=*/0, channel),
+                   &f.buffer[f.write][channel]);
+    f.buffer[f.write][channel].Spectrum(optimization_,
+                                        s.buffer[s.write][channel]);
+  }
+}
+
+bool RenderDelayBufferImpl::DetectActiveRender(
+    rtc::ArrayView<const float> x) const {
+  const float x_energy = std::inner_product(x.begin(), x.end(), x.begin(), 0.f);
+  return x_energy > (config_.render_levels.active_render_limit *
+                     config_.render_levels.active_render_limit) *
+                        kFftLengthBy2;
+}
+
+bool RenderDelayBufferImpl::DetectExcessRenderBlocks() {
+  bool excess_render_detected = false;
+  const size_t latency_blocks = static_cast<size_t>(BufferLatency());
+  // The recently seen minimum latency in blocks. Should be close to 0.
+  min_latency_blocks_ = std::min(min_latency_blocks_, latency_blocks);
+  // After processing a configurable number of blocks the minimum latency is
+  // checked.
+  if (++excess_render_detection_counter_ >=
+      config_.buffering.excess_render_detection_interval_blocks) {
+    // If the minimum latency is not lower than the threshold there have been
+    // more render than capture frames.
+    excess_render_detected = min_latency_blocks_ >
+                             config_.buffering.max_allowed_excess_render_blocks;
+    // Reset the counter and let the minimum latency be the current latency.
+    min_latency_blocks_ = latency_blocks;
+    excess_render_detection_counter_ = 0;
+  }
+
+  data_dumper_->DumpRaw("aec3_latency_blocks", latency_blocks);
+  data_dumper_->DumpRaw("aec3_min_latency_blocks", min_latency_blocks_);
+  data_dumper_->DumpRaw("aec3_excess_render_detected", excess_render_detected);
+  return excess_render_detected;
+}
+
+// Computes the latency in the buffer (the number of unread sub-blocks).
+int RenderDelayBufferImpl::BufferLatency() const {
+  const DownsampledRenderBuffer& l = low_rate_;
+  int latency_samples = (l.buffer.size() + l.read - l.write) % l.buffer.size();
+  int latency_blocks = latency_samples / sub_block_size_;
+  return latency_blocks;
+}
+
+// Increments the write indices for the render buffers.
+void RenderDelayBufferImpl::IncrementWriteIndices() {
+  low_rate_.UpdateWriteIndex(-sub_block_size_);
+  blocks_.IncWriteIndex();
+  spectra_.DecWriteIndex();
+  ffts_.DecWriteIndex();
+}
+
+// Increments the read indices of the low rate render buffers.
+void RenderDelayBufferImpl::IncrementLowRateReadIndices() {
+  low_rate_.UpdateReadIndex(-sub_block_size_);
+}
+
+// Increments the read indices for the render buffers.
+void RenderDelayBufferImpl::IncrementReadIndices() {
+  if (blocks_.read != blocks_.write) {
+    blocks_.IncReadIndex();
+    spectra_.DecReadIndex();
+    ffts_.DecReadIndex();
+  }
+}
+
+// Checks for a render buffer overrun.
+bool RenderDelayBufferImpl::RenderOverrun() {
+  return low_rate_.read == low_rate_.write || blocks_.read == blocks_.write;
+}
+
+// Checks for a render buffer underrun.
+bool RenderDelayBufferImpl::RenderUnderrun() {
+  return low_rate_.read == low_rate_.write;
+}
+
+}  // namespace
+
+RenderDelayBuffer* RenderDelayBuffer::Create(const EchoCanceller3Config& config,
+                                             int sample_rate_hz,
+                                             size_t num_render_channels) {
+  return new RenderDelayBufferImpl(config, sample_rate_hz, num_render_channels);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer.h
new file mode 100644
index 0000000000..6dc1aefb85
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/block.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+
+namespace webrtc {
+
+// Class for buffering the incoming render blocks such that these may be
+// extracted with a specified delay.
+class RenderDelayBuffer {
+ public:
+  enum class BufferingEvent {
+    kNone,
+    kRenderUnderrun,
+    kRenderOverrun,
+    kApiCallSkew
+  };
+
+  static RenderDelayBuffer* Create(const EchoCanceller3Config& config,
+                                   int sample_rate_hz,
+                                   size_t num_render_channels);
+  virtual ~RenderDelayBuffer() = default;
+
+  // Resets the buffer alignment.
+  virtual void Reset() = 0;
+
+  // Inserts a block into the buffer.
+  virtual BufferingEvent Insert(const Block& block) = 0;
+
+  // Updates the buffers one step based on the specified buffer delay. Returns
+  // an enum indicating whether there was a special event that occurred.
+  virtual BufferingEvent PrepareCaptureProcessing() = 0;
+
+  // Called on capture blocks where PrepareCaptureProcessing is not called.
+  virtual void HandleSkippedCaptureProcessing() = 0;
+
+  // Sets the buffer delay and returns a bool indicating whether the delay
+  // changed.
+  virtual bool AlignFromDelay(size_t delay) = 0;
+
+  // Sets the buffer delay from the most recently reported external delay.
+  virtual void AlignFromExternalDelay() = 0;
+
+  // Gets the buffer delay.
+  virtual size_t Delay() const = 0;
+
+  // Gets the buffer delay.
+  virtual size_t MaxDelay() const = 0;
+
+  // Returns the render buffer for the echo remover.
+  virtual RenderBuffer* GetRenderBuffer() = 0;
+
+  // Returns the downsampled render buffer.
+  virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
+
+  // Returns the maximum non calusal offset that can occur in the delay buffer.
+  static int DelayEstimatorOffset(const EchoCanceller3Config& config);
+
+  // Provides an optional external estimate of the audio buffer delay.
+  virtual void SetAudioBufferDelay(int delay_ms) = 0;
+
+  // Returns whether an external delay estimate has been reported via
+  // SetAudioBufferDelay.
+  virtual bool HasReceivedBufferDelay() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
new file mode 100644
index 0000000000..d51e06a1ac
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+std::string ProduceDebugText(int sample_rate_hz) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz;
+  return ss.Release();
+}
+
+}  // namespace
+
+// Verifies that the buffer overflow is correctly reported.
+TEST(RenderDelayBuffer, BufferOverflow) {
+  const EchoCanceller3Config config;
+  for (auto num_channels : {1, 2, 8}) {
+    for (auto rate : {16000, 32000, 48000}) {
+      SCOPED_TRACE(ProduceDebugText(rate));
+      std::unique_ptr<RenderDelayBuffer> delay_buffer(
+          RenderDelayBuffer::Create(config, rate, num_channels));
+      Block block_to_insert(NumBandsForRate(rate), num_channels);
+      for (size_t k = 0; k < 10; ++k) {
+        EXPECT_EQ(RenderDelayBuffer::BufferingEvent::kNone,
+                  delay_buffer->Insert(block_to_insert));
+      }
+      bool overrun_occurred = false;
+      for (size_t k = 0; k < 1000; ++k) {
+        RenderDelayBuffer::BufferingEvent event =
+            delay_buffer->Insert(block_to_insert);
+        overrun_occurred =
+            overrun_occurred ||
+            RenderDelayBuffer::BufferingEvent::kRenderOverrun == event;
+      }
+
+      EXPECT_TRUE(overrun_occurred);
+    }
+  }
+}
+
+// Verifies that the check for available block works.
+TEST(RenderDelayBuffer, AvailableBlock) {
+  constexpr size_t kNumChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
+      EchoCanceller3Config(), kSampleRateHz, kNumChannels));
+  Block input_block(kNumBands, kNumChannels, 1.0f);
+  EXPECT_EQ(RenderDelayBuffer::BufferingEvent::kNone,
+            delay_buffer->Insert(input_block));
+  delay_buffer->PrepareCaptureProcessing();
+}
+
+// Verifies the AlignFromDelay method.
+TEST(RenderDelayBuffer, AlignFromDelay) {
+  EchoCanceller3Config config;
+  std::unique_ptr<RenderDelayBuffer> delay_buffer(
+      RenderDelayBuffer::Create(config, 16000, 1));
+  ASSERT_TRUE(delay_buffer->Delay());
+  delay_buffer->Reset();
+  size_t initial_internal_delay = 0;
+  for (size_t delay = initial_internal_delay;
+       delay < initial_internal_delay + 20; ++delay) {
+    ASSERT_TRUE(delay_buffer->AlignFromDelay(delay));
+    EXPECT_EQ(delay, delay_buffer->Delay());
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for feasible delay.
+// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
+// tests on test bots has been fixed.
+TEST(RenderDelayBufferDeathTest, DISABLED_WrongDelay) {
+  std::unique_ptr<RenderDelayBuffer> delay_buffer(
+      RenderDelayBuffer::Create(EchoCanceller3Config(), 48000, 1));
+  EXPECT_DEATH(delay_buffer->AlignFromDelay(21), "");
+}
+
+// Verifies the check for the number of bands in the inserted blocks.
+TEST(RenderDelayBufferDeathTest, WrongNumberOfBands) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t num_channels : {1, 2, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate));
+      std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
+          EchoCanceller3Config(), rate, num_channels));
+      Block block_to_insert(
+          NumBandsForRate(rate < 48000 ? rate + 16000 : 16000), num_channels);
+      EXPECT_DEATH(delay_buffer->Insert(block_to_insert), "");
+    }
+  }
+}
+
+// Verifies the check for the number of channels in the inserted blocks.
+TEST(RenderDelayBufferDeathTest, WrongNumberOfChannels) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (size_t num_channels : {1, 2, 8}) {
+      SCOPED_TRACE(ProduceDebugText(rate));
+      std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
+          EchoCanceller3Config(), rate, num_channels));
+      Block block_to_insert(NumBandsForRate(rate), num_channels + 1);
+      EXPECT_DEATH(delay_buffer->Insert(block_to_insert), "");
+    }
+  }
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller.cc
new file mode 100644
index 0000000000..465e77fb7c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller.cc
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec3/render_delay_controller.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <atomic>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/delay_estimate.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/aec3/echo_path_delay_estimator.h"
+#include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+class RenderDelayControllerImpl final : public RenderDelayController {
+ public:
+  RenderDelayControllerImpl(const EchoCanceller3Config& config,
+                            int sample_rate_hz,
+                            size_t num_capture_channels);
+
+  RenderDelayControllerImpl() = delete;
+  RenderDelayControllerImpl(const RenderDelayControllerImpl&) = delete;
+  RenderDelayControllerImpl& operator=(const RenderDelayControllerImpl&) =
+      delete;
+
+  ~RenderDelayControllerImpl() override;
+  void Reset(bool reset_delay_confidence) override;
+  void LogRenderCall() override;
+  absl::optional<DelayEstimate> GetDelay(
+      const DownsampledRenderBuffer& render_buffer,
+      size_t render_delay_buffer_delay,
+      const Block& capture) override;
+  bool HasClockdrift() const override;
+
+ private:
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  const int hysteresis_limit_blocks_;
+  absl::optional<DelayEstimate> delay_;
+  EchoPathDelayEstimator delay_estimator_;
+  RenderDelayControllerMetrics metrics_;
+  absl::optional<DelayEstimate> delay_samples_;
+  size_t capture_call_counter_ = 0;
+  int delay_change_counter_ = 0;
+  DelayEstimate::Quality last_delay_estimate_quality_;
+};
+
+DelayEstimate ComputeBufferDelay(
+    const absl::optional<DelayEstimate>& current_delay,
+    int hysteresis_limit_blocks,
+    DelayEstimate estimated_delay) {
+  // Compute the buffer delay increase required to achieve the desired latency.
+  size_t new_delay_blocks = estimated_delay.delay >> kBlockSizeLog2;
+  // Add hysteresis.
+  if (current_delay) {
+    size_t current_delay_blocks = current_delay->delay;
+    if (new_delay_blocks > current_delay_blocks &&
+        new_delay_blocks <= current_delay_blocks + hysteresis_limit_blocks) {
+      new_delay_blocks = current_delay_blocks;
+    }
+  }
+  DelayEstimate new_delay = estimated_delay;
+  new_delay.delay = new_delay_blocks;
+  return new_delay;
+}
+
+std::atomic<int> RenderDelayControllerImpl::instance_count_(0);
+
+RenderDelayControllerImpl::RenderDelayControllerImpl(
+    const EchoCanceller3Config& config,
+    int sample_rate_hz,
+    size_t num_capture_channels)
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      hysteresis_limit_blocks_(
+          static_cast<int>(config.delay.hysteresis_limit_blocks)),
+      delay_estimator_(data_dumper_.get(), config, num_capture_channels),
+      last_delay_estimate_quality_(DelayEstimate::Quality::kCoarse) {
+  RTC_DCHECK(ValidFullBandRate(sample_rate_hz));
+  delay_estimator_.LogDelayEstimationProperties(sample_rate_hz, 0);
+}
+
+RenderDelayControllerImpl::~RenderDelayControllerImpl() = default;
+
+void RenderDelayControllerImpl::Reset(bool reset_delay_confidence) {
+  delay_ = absl::nullopt;
+  delay_samples_ = absl::nullopt;
+  delay_estimator_.Reset(reset_delay_confidence);
+  delay_change_counter_ = 0;
+  if (reset_delay_confidence) {
+    last_delay_estimate_quality_ = DelayEstimate::Quality::kCoarse;
+  }
+}
+
+void RenderDelayControllerImpl::LogRenderCall() {}
+
+absl::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
+    const DownsampledRenderBuffer& render_buffer,
+    size_t render_delay_buffer_delay,
+    const Block& capture) {
+  ++capture_call_counter_;
+
+  auto delay_samples = delay_estimator_.EstimateDelay(render_buffer, capture);
+
+  if (delay_samples) {
+    if (!delay_samples_ || delay_samples->delay != delay_samples_->delay) {
+      delay_change_counter_ = 0;
+    }
+    if (delay_samples_) {
+      delay_samples_->blocks_since_last_change =
+          delay_samples_->delay == delay_samples->delay
+              ? delay_samples_->blocks_since_last_change + 1
+              : 0;
+      delay_samples_->blocks_since_last_update = 0;
+      delay_samples_->delay = delay_samples->delay;
+      delay_samples_->quality = delay_samples->quality;
+    } else {
+      delay_samples_ = delay_samples;
+    }
+  } else {
+    if (delay_samples_) {
+      ++delay_samples_->blocks_since_last_change;
+      ++delay_samples_->blocks_since_last_update;
+    }
+  }
+
+  if (delay_change_counter_ < 2 * kNumBlocksPerSecond) {
+    ++delay_change_counter_;
+  }
+
+  if (delay_samples_) {
+    // Compute the render delay buffer delay.
+    const bool use_hysteresis =
+        last_delay_estimate_quality_ == DelayEstimate::Quality::kRefined &&
+        delay_samples_->quality == DelayEstimate::Quality::kRefined;
+    delay_ = ComputeBufferDelay(
+        delay_, use_hysteresis ? hysteresis_limit_blocks_ : 0, *delay_samples_);
+    last_delay_estimate_quality_ = delay_samples_->quality;
+  }
+
+  metrics_.Update(
+      delay_samples_ ? absl::optional<size_t>(delay_samples_->delay)
+                     : absl::nullopt,
+      delay_ ? absl::optional<size_t>(delay_->delay) : absl::nullopt,
+      delay_estimator_.Clockdrift());
+
+  data_dumper_->DumpRaw("aec3_render_delay_controller_delay",
+                        delay_samples ? delay_samples->delay : 0);
+  data_dumper_->DumpRaw("aec3_render_delay_controller_buffer_delay",
+                        delay_ ? delay_->delay : 0);
+
+  return delay_;
+}
+
+bool RenderDelayControllerImpl::HasClockdrift() const {
+  return delay_estimator_.Clockdrift() != ClockdriftDetector::Level::kNone;
+}
+
+}  // namespace
+
+RenderDelayController* RenderDelayController::Create(
+    const EchoCanceller3Config& config,
+    int sample_rate_hz,
+    size_t num_capture_channels) {
+  return new RenderDelayControllerImpl(config, sample_rate_hz,
+                                       num_capture_channels);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller.h b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller.h
new file mode 100644
index 0000000000..4a18a11e36
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/block.h"
+#include "modules/audio_processing/aec3/delay_estimate.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+// Class for aligning the render and capture signal using a RenderDelayBuffer.
+class RenderDelayController {
+ public:
+  static RenderDelayController* Create(const EchoCanceller3Config& config,
+                                       int sample_rate_hz,
+                                       size_t num_capture_channels);
+  virtual ~RenderDelayController() = default;
+
+  // Resets the delay controller. If the delay confidence is reset, the reset
+  // behavior is as if the call is restarted.
+  virtual void Reset(bool reset_delay_confidence) = 0;
+
+  // Logs a render call.
+  virtual void LogRenderCall() = 0;
+
+  // Aligns the render buffer content with the capture signal.
+  virtual absl::optional<DelayEstimate> GetDelay(
+      const DownsampledRenderBuffer& render_buffer,
+      size_t render_delay_buffer_delay,
+      const Block& capture) = 0;
+
+  // Returns true if clockdrift has been detected.
+  virtual bool HasClockdrift() const = 0;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
new file mode 100644
index 0000000000..1e0a0f443e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
+
+#include <algorithm>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+enum class DelayReliabilityCategory {
+  kNone,
+  kPoor,
+  kMedium,
+  kGood,
+  kExcellent,
+  kNumCategories
+};
+enum class DelayChangesCategory {
+  kNone,
+  kFew,
+  kSeveral,
+  kMany,
+  kConstant,
+  kNumCategories
+};
+
+}  // namespace
+
+RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default;
+
+void RenderDelayControllerMetrics::Update(
+    absl::optional<size_t> delay_samples,
+    absl::optional<size_t> buffer_delay_blocks,
+    ClockdriftDetector::Level clockdrift) {
+  ++call_counter_;
+
+  if (!initial_update) {
+    size_t delay_blocks;
+    if (delay_samples) {
+      ++reliable_delay_estimate_counter_;
+      // Add an offset by 1 (metric is halved before reporting) to reserve 0 for
+      // absent delay.
+      delay_blocks = (*delay_samples) / kBlockSize + 2;
+    } else {
+      delay_blocks = 0;
+    }
+
+    if (delay_blocks != delay_blocks_) {
+      ++delay_change_counter_;
+      delay_blocks_ = delay_blocks;
+    }
+
+  } else if (++initial_call_counter_ == 5 * kNumBlocksPerSecond) {
+    initial_update = false;
+  }
+
+  if (call_counter_ == kMetricsReportingIntervalBlocks) {
+    int value_to_report = static_cast<int>(delay_blocks_);
+    // Divide by 2 to compress metric range.
+    value_to_report = std::min(124, value_to_report >> 1);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay",
+                                value_to_report, 0, 124, 125);
+
+    // Divide by 2 to compress metric range.
+    // Offset by 1 to reserve 0 for absent delay.
+    value_to_report = buffer_delay_blocks ? (*buffer_delay_blocks + 2) >> 1 : 0;
+    value_to_report = std::min(124, value_to_report);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay",
+                                value_to_report, 0, 124, 125);
+
+    DelayReliabilityCategory delay_reliability;
+    if (reliable_delay_estimate_counter_ == 0) {
+      delay_reliability = DelayReliabilityCategory::kNone;
+    } else if (reliable_delay_estimate_counter_ > (call_counter_ >> 1)) {
+      delay_reliability = DelayReliabilityCategory::kExcellent;
+    } else if (reliable_delay_estimate_counter_ > 100) {
+      delay_reliability = DelayReliabilityCategory::kGood;
+    } else if (reliable_delay_estimate_counter_ > 10) {
+      delay_reliability = DelayReliabilityCategory::kMedium;
+    } else {
+      delay_reliability = DelayReliabilityCategory::kPoor;
+    }
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.EchoCanceller.ReliableDelayEstimates",
+        static_cast<int>(delay_reliability),
+        static_cast<int>(DelayReliabilityCategory::kNumCategories));
+
+    DelayChangesCategory delay_changes;
+    if (delay_change_counter_ == 0) {
+      delay_changes = DelayChangesCategory::kNone;
+    } else if (delay_change_counter_ > 10) {
+      delay_changes = DelayChangesCategory::kConstant;
+    } else if (delay_change_counter_ > 5) {
+      delay_changes = DelayChangesCategory::kMany;
+    } else if (delay_change_counter_ > 2) {
+      delay_changes = DelayChangesCategory::kSeveral;
+    } else {
+      delay_changes = DelayChangesCategory::kFew;
+    }
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.EchoCanceller.DelayChanges",
+        static_cast<int>(delay_changes),
+        static_cast<int>(DelayChangesCategory::kNumCategories));
+
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.EchoCanceller.Clockdrift", static_cast<int>(clockdrift),
+        static_cast<int>(ClockdriftDetector::Level::kNumCategories));
+
+    call_counter_ = 0;
+    ResetMetrics();
+  }
+}
+
+void RenderDelayControllerMetrics::ResetMetrics() {
+  delay_change_counter_ = 0;
+  reliable_delay_estimate_counter_ = 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
new file mode 100644
index 0000000000..b81833b43f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/aec3/clockdrift_detector.h"
+
+namespace webrtc {
+
+// Handles the reporting of metrics for the render delay controller.
+class RenderDelayControllerMetrics {
+ public:
+  RenderDelayControllerMetrics();
+
+  RenderDelayControllerMetrics(const RenderDelayControllerMetrics&) = delete;
+  RenderDelayControllerMetrics& operator=(const RenderDelayControllerMetrics&) =
+      delete;
+
+  // Updates the metric with new data.
+  void Update(absl::optional<size_t> delay_samples,
+              absl::optional<size_t> buffer_delay_blocks,
+              ClockdriftDetector::Level clockdrift);
+
+ private:
+  // Resets the metrics.
+  void ResetMetrics();
+
+  size_t delay_blocks_ = 0;
+  int reliable_delay_estimate_counter_ = 0;
+  int delay_change_counter_ = 0;
+  int call_counter_ = 0;
+  int initial_call_counter_ = 0;
+  bool initial_update = true;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
new file mode 100644
index 0000000000..cf9df6b297
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "system_wrappers/include/metrics.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Verify the general functionality of RenderDelayControllerMetrics.
+TEST(RenderDelayControllerMetrics, NormalUsage) {
+  metrics::Reset();
+
+  RenderDelayControllerMetrics metrics;
+
+  int expected_num_metric_reports = 0;
+
+  for (int j = 0; j < 3; ++j) {
+    for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
+      metrics.Update(absl::nullopt, absl::nullopt,
+                     ClockdriftDetector::Level::kNone);
+    }
+    EXPECT_METRIC_EQ(
+        metrics::NumSamples("WebRTC.Audio.EchoCanceller.EchoPathDelay"),
+        expected_num_metric_reports);
+    EXPECT_METRIC_EQ(
+        metrics::NumSamples("WebRTC.Audio.EchoCanceller.BufferDelay"),
+        expected_num_metric_reports);
+    EXPECT_METRIC_EQ(metrics::NumSamples(
+                         "WebRTC.Audio.EchoCanceller.ReliableDelayEstimates"),
+                     expected_num_metric_reports);
+    EXPECT_METRIC_EQ(
+        metrics::NumSamples("WebRTC.Audio.EchoCanceller.DelayChanges"),
+        expected_num_metric_reports);
+    EXPECT_METRIC_EQ(
+        metrics::NumSamples("WebRTC.Audio.EchoCanceller.Clockdrift"),
+        expected_num_metric_reports);
+
+    // We expect metric reports every kMetricsReportingIntervalBlocks blocks.
+    ++expected_num_metric_reports;
+
+    metrics.Update(absl::nullopt, absl::nullopt,
+                   ClockdriftDetector::Level::kNone);
+    EXPECT_METRIC_EQ(
+        metrics::NumSamples("WebRTC.Audio.EchoCanceller.EchoPathDelay"),
+        expected_num_metric_reports);
+    EXPECT_METRIC_EQ(
+        metrics::NumSamples("WebRTC.Audio.EchoCanceller.BufferDelay"),
+        expected_num_metric_reports);
+    EXPECT_METRIC_EQ(metrics::NumSamples(
+                         "WebRTC.Audio.EchoCanceller.ReliableDelayEstimates"),
+                     expected_num_metric_reports);
+    EXPECT_METRIC_EQ(
+        metrics::NumSamples("WebRTC.Audio.EchoCanceller.DelayChanges"),
+        expected_num_metric_reports);
+    EXPECT_METRIC_EQ(
+        metrics::NumSamples("WebRTC.Audio.EchoCanceller.Clockdrift"),
+        expected_num_metric_reports);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc
new file mode 100644
index 0000000000..e1a54fca9e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc
@@ -0,0 +1,334 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/render_delay_controller.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/block_processor.h"
+#include "modules/audio_processing/aec3/decimator.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+std::string ProduceDebugText(int sample_rate_hz) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz;
+  return ss.Release();
+}
+
+std::string ProduceDebugText(int sample_rate_hz,
+                             size_t delay,
+                             size_t num_render_channels,
+                             size_t num_capture_channels) {
+  rtc::StringBuilder ss;
+  ss << ProduceDebugText(sample_rate_hz) << ", Delay: " << delay
+     << ", Num render channels: " << num_render_channels
+     << ", Num capture channels: " << num_capture_channels;
+  return ss.Release();
+}
+
+constexpr size_t kDownSamplingFactors[] = {2, 4, 8};
+
+}  // namespace
+
+// Verifies the output of GetDelay when there are no AnalyzeRender calls.
+// TODO(bugs.webrtc.org/11161): Re-enable tests.
+TEST(RenderDelayController, DISABLED_NoRenderSignal) {
+  for (size_t num_render_channels : {1, 2, 8}) {
+    Block block(/*num_bands=1*/ 1, /*num_channels=*/1);
+    EchoCanceller3Config config;
+    for (size_t num_matched_filters = 4; num_matched_filters <= 10;
+         num_matched_filters++) {
+      for (auto down_sampling_factor : kDownSamplingFactors) {
+        config.delay.down_sampling_factor = down_sampling_factor;
+        config.delay.num_filters = num_matched_filters;
+        for (auto rate : {16000, 32000, 48000}) {
+          SCOPED_TRACE(ProduceDebugText(rate));
+          std::unique_ptr<RenderDelayBuffer> delay_buffer(
+              RenderDelayBuffer::Create(config, rate, num_render_channels));
+          std::unique_ptr<RenderDelayController> delay_controller(
+              RenderDelayController::Create(config, rate,
+                                            /*num_capture_channels*/ 1));
+          for (size_t k = 0; k < 100; ++k) {
+            auto delay = delay_controller->GetDelay(
+                delay_buffer->GetDownsampledRenderBuffer(),
+                delay_buffer->Delay(), block);
+            EXPECT_FALSE(delay->delay);
+          }
+        }
+      }
+    }
+  }
+}
+
+// Verifies the basic API call sequence.
+// TODO(bugs.webrtc.org/11161): Re-enable tests.
+TEST(RenderDelayController, DISABLED_BasicApiCalls) {
+  for (size_t num_capture_channels : {1, 2, 4}) {
+    for (size_t num_render_channels : {1, 2, 8}) {
+      Block capture_block(/*num_bands=*/1, num_capture_channels);
+      absl::optional<DelayEstimate> delay_blocks;
+      for (size_t num_matched_filters = 4; num_matched_filters <= 10;
+           num_matched_filters++) {
+        for (auto down_sampling_factor : kDownSamplingFactors) {
+          EchoCanceller3Config config;
+          config.delay.down_sampling_factor = down_sampling_factor;
+          config.delay.num_filters = num_matched_filters;
+          config.delay.capture_alignment_mixing.downmix = false;
+          config.delay.capture_alignment_mixing.adaptive_selection = false;
+
+          for (auto rate : {16000, 32000, 48000}) {
+            Block render_block(NumBandsForRate(rate), num_render_channels);
+            std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+                RenderDelayBuffer::Create(config, rate, num_render_channels));
+            std::unique_ptr<RenderDelayController> delay_controller(
+                RenderDelayController::Create(EchoCanceller3Config(), rate,
+                                              num_capture_channels));
+            for (size_t k = 0; k < 10; ++k) {
+              render_delay_buffer->Insert(render_block);
+              render_delay_buffer->PrepareCaptureProcessing();
+
+              delay_blocks = delay_controller->GetDelay(
+                  render_delay_buffer->GetDownsampledRenderBuffer(),
+                  render_delay_buffer->Delay(), capture_block);
+            }
+            EXPECT_TRUE(delay_blocks);
+            EXPECT_FALSE(delay_blocks->delay);
+          }
+        }
+      }
+    }
+  }
+}
+
+// Verifies that the RenderDelayController is able to align the signals for
+// simple timeshifts between the signals.
+// TODO(bugs.webrtc.org/11161): Re-enable tests.
+TEST(RenderDelayController, DISABLED_Alignment) {
+  Random random_generator(42U);
+  for (size_t num_capture_channels : {1, 2, 4}) {
+    Block capture_block(/*num_bands=*/1, num_capture_channels);
+    for (size_t num_matched_filters = 4; num_matched_filters <= 10;
+         num_matched_filters++) {
+      for (auto down_sampling_factor : kDownSamplingFactors) {
+        EchoCanceller3Config config;
+        config.delay.down_sampling_factor = down_sampling_factor;
+        config.delay.num_filters = num_matched_filters;
+        config.delay.capture_alignment_mixing.downmix = false;
+        config.delay.capture_alignment_mixing.adaptive_selection = false;
+
+        for (size_t num_render_channels : {1, 2, 8}) {
+          for (auto rate : {16000, 32000, 48000}) {
+            Block render_block(NumBandsForRate(rate), num_render_channels);
+
+            for (size_t delay_samples : {15, 50, 150, 200, 800, 4000}) {
+              absl::optional<DelayEstimate> delay_blocks;
+              SCOPED_TRACE(ProduceDebugText(rate, delay_samples,
+                                            num_render_channels,
+                                            num_capture_channels));
+              std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+                  RenderDelayBuffer::Create(config, rate, num_render_channels));
+              std::unique_ptr<RenderDelayController> delay_controller(
+                  RenderDelayController::Create(config, rate,
+                                                num_capture_channels));
+              DelayBuffer<float> signal_delay_buffer(delay_samples);
+              for (size_t k = 0; k < (400 + delay_samples / kBlockSize); ++k) {
+                for (int band = 0; band < render_block.NumBands(); ++band) {
+                  for (int channel = 0; channel < render_block.NumChannels();
+                       ++channel) {
+                    RandomizeSampleVector(&random_generator,
+                                          render_block.View(band, channel));
+                  }
+                }
+                signal_delay_buffer.Delay(
+                    render_block.View(/*band=*/0, /*channel=*/0),
+                    capture_block.View(/*band=*/0, /*channel=*/0));
+                render_delay_buffer->Insert(render_block);
+                render_delay_buffer->PrepareCaptureProcessing();
+                delay_blocks = delay_controller->GetDelay(
+                    render_delay_buffer->GetDownsampledRenderBuffer(),
+                    render_delay_buffer->Delay(), capture_block);
+              }
+              ASSERT_TRUE(!!delay_blocks);
+
+              constexpr int kDelayHeadroomBlocks = 1;
+              size_t expected_delay_blocks =
+                  std::max(0, static_cast<int>(delay_samples / kBlockSize) -
+                                  kDelayHeadroomBlocks);
+
+              EXPECT_EQ(expected_delay_blocks, delay_blocks->delay);
+            }
+          }
+        }
+      }
+    }
+  }
+}
+
+// Verifies that the RenderDelayController is able to properly handle noncausal
+// delays.
+// TODO(bugs.webrtc.org/11161): Re-enable tests.
+TEST(RenderDelayController, DISABLED_NonCausalAlignment) {
+  Random random_generator(42U);
+  for (size_t num_capture_channels : {1, 2, 4}) {
+    for (size_t num_render_channels : {1, 2, 8}) {
+      for (size_t num_matched_filters = 4; num_matched_filters <= 10;
+           num_matched_filters++) {
+        for (auto down_sampling_factor : kDownSamplingFactors) {
+          EchoCanceller3Config config;
+          config.delay.down_sampling_factor = down_sampling_factor;
+          config.delay.num_filters = num_matched_filters;
+          config.delay.capture_alignment_mixing.downmix = false;
+          config.delay.capture_alignment_mixing.adaptive_selection = false;
+          for (auto rate : {16000, 32000, 48000}) {
+            Block render_block(NumBandsForRate(rate), num_render_channels);
+            Block capture_block(NumBandsForRate(rate), num_capture_channels);
+
+            for (int delay_samples : {-15, -50, -150, -200}) {
+              absl::optional<DelayEstimate> delay_blocks;
+              SCOPED_TRACE(ProduceDebugText(rate, -delay_samples,
+                                            num_render_channels,
+                                            num_capture_channels));
+              std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+                  RenderDelayBuffer::Create(config, rate, num_render_channels));
+              std::unique_ptr<RenderDelayController> delay_controller(
+                  RenderDelayController::Create(EchoCanceller3Config(), rate,
+                                                num_capture_channels));
+              DelayBuffer<float> signal_delay_buffer(-delay_samples);
+              for (int k = 0;
+                   k < (400 - delay_samples / static_cast<int>(kBlockSize));
+                   ++k) {
+                RandomizeSampleVector(
+                    &random_generator,
+                    capture_block.View(/*band=*/0, /*channel=*/0));
+                signal_delay_buffer.Delay(
+                    capture_block.View(/*band=*/0, /*channel=*/0),
+                    render_block.View(/*band=*/0, /*channel=*/0));
+                render_delay_buffer->Insert(render_block);
+                render_delay_buffer->PrepareCaptureProcessing();
+                delay_blocks = delay_controller->GetDelay(
+                    render_delay_buffer->GetDownsampledRenderBuffer(),
+                    render_delay_buffer->Delay(), capture_block);
+              }
+
+              ASSERT_FALSE(delay_blocks);
+            }
+          }
+        }
+      }
+    }
+  }
+}
+
+// Verifies that the RenderDelayController is able to align the signals for
+// simple timeshifts between the signals when there is jitter in the API calls.
+// TODO(bugs.webrtc.org/11161): Re-enable tests.
+TEST(RenderDelayController, DISABLED_AlignmentWithJitter) {
+  Random random_generator(42U);
+  for (size_t num_capture_channels : {1, 2, 4}) {
+    for (size_t num_render_channels : {1, 2, 8}) {
+      Block capture_block(
+          /*num_bands=*/1, num_capture_channels);
+      for (size_t num_matched_filters = 4; num_matched_filters <= 10;
+           num_matched_filters++) {
+        for (auto down_sampling_factor : kDownSamplingFactors) {
+          EchoCanceller3Config config;
+          config.delay.down_sampling_factor = down_sampling_factor;
+          config.delay.num_filters = num_matched_filters;
+          config.delay.capture_alignment_mixing.downmix = false;
+          config.delay.capture_alignment_mixing.adaptive_selection = false;
+
+          for (auto rate : {16000, 32000, 48000}) {
+            Block render_block(NumBandsForRate(rate), num_render_channels);
+            for (size_t delay_samples : {15, 50, 300, 800}) {
+              absl::optional<DelayEstimate> delay_blocks;
+              SCOPED_TRACE(ProduceDebugText(rate, delay_samples,
+                                            num_render_channels,
+                                            num_capture_channels));
+              std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+                  RenderDelayBuffer::Create(config, rate, num_render_channels));
+              std::unique_ptr<RenderDelayController> delay_controller(
+                  RenderDelayController::Create(config, rate,
+                                                num_capture_channels));
+              DelayBuffer<float> signal_delay_buffer(delay_samples);
+              constexpr size_t kMaxTestJitterBlocks = 26;
+              for (size_t j = 0; j < (1000 + delay_samples / kBlockSize) /
+                                             kMaxTestJitterBlocks +
+                                         1;
+                   ++j) {
+                std::vector<Block> capture_block_buffer;
+                for (size_t k = 0; k < (kMaxTestJitterBlocks - 1); ++k) {
+                  RandomizeSampleVector(
+                      &random_generator,
+                      render_block.View(/*band=*/0, /*channel=*/0));
+                  signal_delay_buffer.Delay(
+                      render_block.View(/*band=*/0, /*channel=*/0),
+                      capture_block.View(/*band=*/0, /*channel=*/0));
+                  capture_block_buffer.push_back(capture_block);
+                  render_delay_buffer->Insert(render_block);
+                }
+                for (size_t k = 0; k < (kMaxTestJitterBlocks - 1); ++k) {
+                  render_delay_buffer->PrepareCaptureProcessing();
+                  delay_blocks = delay_controller->GetDelay(
+                      render_delay_buffer->GetDownsampledRenderBuffer(),
+                      render_delay_buffer->Delay(), capture_block_buffer[k]);
+                }
+              }
+
+              constexpr int kDelayHeadroomBlocks = 1;
+              size_t expected_delay_blocks =
+                  std::max(0, static_cast<int>(delay_samples / kBlockSize) -
+                                  kDelayHeadroomBlocks);
+              if (expected_delay_blocks < 2) {
+                expected_delay_blocks = 0;
+              }
+
+              ASSERT_TRUE(delay_blocks);
+              EXPECT_EQ(expected_delay_blocks, delay_blocks->delay);
+            }
+          }
+        }
+      }
+    }
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for correct sample rate.
+// TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
+// tests on test bots has been fixed.
+TEST(RenderDelayControllerDeathTest, DISABLED_WrongSampleRate) {
+  for (auto rate : {-1, 0, 8001, 16001}) {
+    SCOPED_TRACE(ProduceDebugText(rate));
+    EchoCanceller3Config config;
+    std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+        RenderDelayBuffer::Create(config, rate, 1));
+    EXPECT_DEATH(
+        std::unique_ptr<RenderDelayController>(
+            RenderDelayController::Create(EchoCanceller3Config(), rate, 1)),
+        "");
+  }
+}
+
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.cc
new file mode 100644
index 0000000000..bfbeb0ec2e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+constexpr size_t kCounterThreshold = 5;
+
+// Identifies local bands with narrow characteristics.
+void IdentifySmallNarrowBandRegions(
+    const RenderBuffer& render_buffer,
+    const absl::optional<size_t>& delay_partitions,
+    std::array<size_t, kFftLengthBy2 - 1>* narrow_band_counters) {
+  RTC_DCHECK(narrow_band_counters);
+
+  if (!delay_partitions) {
+    narrow_band_counters->fill(0);
+    return;
+  }
+
+  std::array<size_t, kFftLengthBy2 - 1> channel_counters;
+  channel_counters.fill(0);
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> X2 =
+      render_buffer.Spectrum(*delay_partitions);
+  for (size_t ch = 0; ch < X2.size(); ++ch) {
+    for (size_t k = 1; k < kFftLengthBy2; ++k) {
+      if (X2[ch][k] > 3 * std::max(X2[ch][k - 1], X2[ch][k + 1])) {
+        ++channel_counters[k - 1];
+      }
+    }
+  }
+  for (size_t k = 1; k < kFftLengthBy2; ++k) {
+    (*narrow_band_counters)[k - 1] =
+        channel_counters[k - 1] > 0 ? (*narrow_band_counters)[k - 1] + 1 : 0;
+  }
+}
+
+// Identifies whether the signal has a single strong narrow-band component.
+void IdentifyStrongNarrowBandComponent(const RenderBuffer& render_buffer,
+                                       int strong_peak_freeze_duration,
+                                       absl::optional<int>* narrow_peak_band,
+                                       size_t* narrow_peak_counter) {
+  RTC_DCHECK(narrow_peak_band);
+  RTC_DCHECK(narrow_peak_counter);
+  if (*narrow_peak_band &&
+      ++(*narrow_peak_counter) >
+          static_cast<size_t>(strong_peak_freeze_duration)) {
+    *narrow_peak_band = absl::nullopt;
+  }
+
+  const Block& x_latest = render_buffer.GetBlock(0);
+  float max_peak_level = 0.f;
+  for (int channel = 0; channel < x_latest.NumChannels(); ++channel) {
+    rtc::ArrayView<const float, kFftLengthBy2Plus1> X2_latest =
+        render_buffer.Spectrum(0)[channel];
+
+    // Identify the spectral peak.
+    const int peak_bin =
+        static_cast<int>(std::max_element(X2_latest.begin(), X2_latest.end()) -
+                         X2_latest.begin());
+
+    // Compute the level around the peak.
+    float non_peak_power = 0.f;
+    for (int k = std::max(0, peak_bin - 14); k < peak_bin - 4; ++k) {
+      non_peak_power = std::max(X2_latest[k], non_peak_power);
+    }
+    for (int k = peak_bin + 5;
+         k < std::min(peak_bin + 15, static_cast<int>(kFftLengthBy2Plus1));
+         ++k) {
+      non_peak_power = std::max(X2_latest[k], non_peak_power);
+    }
+
+    // Assess the render signal strength.
+    auto result0 = std::minmax_element(x_latest.begin(/*band=*/0, channel),
+                                       x_latest.end(/*band=*/0, channel));
+    float max_abs = std::max(fabs(*result0.first), fabs(*result0.second));
+
+    if (x_latest.NumBands() > 1) {
+      const auto result1 =
+          std::minmax_element(x_latest.begin(/*band=*/1, channel),
+                              x_latest.end(/*band=*/1, channel));
+      max_abs =
+          std::max(max_abs, static_cast<float>(std::max(
+                                fabs(*result1.first), fabs(*result1.second))));
+    }
+
+    // Detect whether the spectral peak has as strong narrowband nature.
+    const float peak_level = X2_latest[peak_bin];
+    if (peak_bin > 0 && max_abs > 100 && peak_level > 100 * non_peak_power) {
+      // Store the strongest peak across channels.
+      if (peak_level > max_peak_level) {
+        max_peak_level = peak_level;
+        *narrow_peak_band = peak_bin;
+        *narrow_peak_counter = 0;
+      }
+    }
+  }
+}
+
+}  // namespace
+
+RenderSignalAnalyzer::RenderSignalAnalyzer(const EchoCanceller3Config& config)
+    : strong_peak_freeze_duration_(config.filter.refined.length_blocks) {
+  narrow_band_counters_.fill(0);
+}
+RenderSignalAnalyzer::~RenderSignalAnalyzer() = default;
+
+void RenderSignalAnalyzer::Update(
+    const RenderBuffer& render_buffer,
+    const absl::optional<size_t>& delay_partitions) {
+  // Identify bands of narrow nature.
+  IdentifySmallNarrowBandRegions(render_buffer, delay_partitions,
+                                 &narrow_band_counters_);
+
+  // Identify the presence of a strong narrow band.
+  IdentifyStrongNarrowBandComponent(render_buffer, strong_peak_freeze_duration_,
+                                    &narrow_peak_band_, &narrow_peak_counter_);
+}
+
+void RenderSignalAnalyzer::MaskRegionsAroundNarrowBands(
+    std::array<float, kFftLengthBy2Plus1>* v) const {
+  RTC_DCHECK(v);
+
+  // Set v to zero around narrow band signal regions.
+  if (narrow_band_counters_[0] > kCounterThreshold) {
+    (*v)[1] = (*v)[0] = 0.f;
+  }
+  for (size_t k = 2; k < kFftLengthBy2 - 1; ++k) {
+    if (narrow_band_counters_[k - 1] > kCounterThreshold) {
+      (*v)[k - 2] = (*v)[k - 1] = (*v)[k] = (*v)[k + 1] = (*v)[k + 2] = 0.f;
+    }
+  }
+  if (narrow_band_counters_[kFftLengthBy2 - 2] > kCounterThreshold) {
+    (*v)[kFftLengthBy2] = (*v)[kFftLengthBy2 - 1] = 0.f;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.h b/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.h
new file mode 100644
index 0000000000..2e4aaa4ba7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
+
+#include <algorithm>
+#include <array>
+#include <cstddef>
+
+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Provides functionality for analyzing the properties of the render signal.
+class RenderSignalAnalyzer {
+ public:
+  explicit RenderSignalAnalyzer(const EchoCanceller3Config& config);
+  ~RenderSignalAnalyzer();
+
+  RenderSignalAnalyzer(const RenderSignalAnalyzer&) = delete;
+  RenderSignalAnalyzer& operator=(const RenderSignalAnalyzer&) = delete;
+
+  // Updates the render signal analysis with the most recent render signal.
+  void Update(const RenderBuffer& render_buffer,
+              const absl::optional<size_t>& delay_partitions);
+
+  // Returns true if the render signal is poorly exciting.
+  bool PoorSignalExcitation() const {
+    RTC_DCHECK_LT(2, narrow_band_counters_.size());
+    return std::any_of(narrow_band_counters_.begin(),
+                       narrow_band_counters_.end(),
+                       [](size_t a) { return a > 10; });
+  }
+
+  // Zeros the array around regions with narrow bands signal characteristics.
+  void MaskRegionsAroundNarrowBands(
+      std::array<float, kFftLengthBy2Plus1>* v) const;
+
+  absl::optional<int> NarrowPeakBand() const { return narrow_peak_band_; }
+
+ private:
+  const int strong_peak_freeze_duration_;
+  std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_;
+  absl::optional<int> narrow_peak_band_;
+  size_t narrow_peak_counter_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
new file mode 100644
index 0000000000..16f6280cb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
@@ -0,0 +1,171 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+
+#include <math.h>
+
+#include <array>
+#include <cmath>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kPi = 3.141592f;
+
+void ProduceSinusoidInNoise(int sample_rate_hz,
+                            size_t sinusoid_channel,
+                            float sinusoidal_frequency_hz,
+                            Random* random_generator,
+                            size_t* sample_counter,
+                            Block* x) {
+  // Fill x with low-amplitude noise.
+  for (int band = 0; band < x->NumBands(); ++band) {
+    for (int channel = 0; channel < x->NumChannels(); ++channel) {
+      RandomizeSampleVector(random_generator, x->View(band, channel),
+                            /*amplitude=*/500.f);
+    }
+  }
+  // Produce a sinusoid of the specified frequency in the specified channel.
+  for (size_t k = *sample_counter, j = 0; k < (*sample_counter + kBlockSize);
+       ++k, ++j) {
+    x->View(/*band=*/0, sinusoid_channel)[j] +=
+        32000.f *
+        std::sin(2.f * kPi * sinusoidal_frequency_hz * k / sample_rate_hz);
+  }
+  *sample_counter = *sample_counter + kBlockSize;
+}
+
+void RunNarrowBandDetectionTest(size_t num_channels) {
+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
+  Random random_generator(42U);
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  Block x(kNumBands, num_channels);
+  std::array<float, kBlockSize> x_old;
+  Aec3Fft fft;
+  EchoCanceller3Config config;
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, num_channels));
+
+  std::array<float, kFftLengthBy2Plus1> mask;
+  x_old.fill(0.f);
+  constexpr int kSinusFrequencyBin = 32;
+
+  auto generate_sinusoid_test = [&](bool known_delay) {
+    size_t sample_counter = 0;
+    for (size_t k = 0; k < 100; ++k) {
+      ProduceSinusoidInNoise(16000, num_channels - 1,
+                             16000 / 2 * kSinusFrequencyBin / kFftLengthBy2,
+                             &random_generator, &sample_counter, &x);
+
+      render_delay_buffer->Insert(x);
+      if (k == 0) {
+        render_delay_buffer->Reset();
+      }
+      render_delay_buffer->PrepareCaptureProcessing();
+
+      analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
+                      known_delay ? absl::optional<size_t>(0) : absl::nullopt);
+    }
+  };
+
+  generate_sinusoid_test(true);
+  mask.fill(1.f);
+  analyzer.MaskRegionsAroundNarrowBands(&mask);
+  for (int k = 0; k < static_cast<int>(mask.size()); ++k) {
+    EXPECT_EQ(abs(k - kSinusFrequencyBin) <= 2 ? 0.f : 1.f, mask[k]);
+  }
+  EXPECT_TRUE(analyzer.PoorSignalExcitation());
+  EXPECT_TRUE(static_cast<bool>(analyzer.NarrowPeakBand()));
+  EXPECT_EQ(*analyzer.NarrowPeakBand(), 32);
+
+  // Verify that no bands are detected as narrow when the delay is unknown.
+  generate_sinusoid_test(false);
+  mask.fill(1.f);
+  analyzer.MaskRegionsAroundNarrowBands(&mask);
+  std::for_each(mask.begin(), mask.end(), [](float a) { EXPECT_EQ(1.f, a); });
+  EXPECT_FALSE(analyzer.PoorSignalExcitation());
+}
+
+std::string ProduceDebugText(size_t num_channels) {
+  rtc::StringBuilder ss;
+  ss << "number of channels: " << num_channels;
+  return ss.Release();
+}
+}  // namespace
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Verifies that the check for non-null output parameter works.
+TEST(RenderSignalAnalyzerDeathTest, NullMaskOutput) {
+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
+  EXPECT_DEATH(analyzer.MaskRegionsAroundNarrowBands(nullptr), "");
+}
+
+#endif
+
+// Verify that no narrow bands are detected in a Gaussian noise signal.
+TEST(RenderSignalAnalyzer, NoFalseDetectionOfNarrowBands) {
+  for (auto num_channels : {1, 2, 8}) {
+    SCOPED_TRACE(ProduceDebugText(num_channels));
+    RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
+    Random random_generator(42U);
+    Block x(3, num_channels);
+    std::array<float, kBlockSize> x_old;
+    std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+        RenderDelayBuffer::Create(EchoCanceller3Config(), 48000, num_channels));
+    std::array<float, kFftLengthBy2Plus1> mask;
+    x_old.fill(0.f);
+
+    for (int k = 0; k < 100; ++k) {
+      for (int band = 0; band < x.NumBands(); ++band) {
+        for (int channel = 0; channel < x.NumChannels(); ++channel) {
+          RandomizeSampleVector(&random_generator, x.View(band, channel));
+        }
+      }
+
+      render_delay_buffer->Insert(x);
+      if (k == 0) {
+        render_delay_buffer->Reset();
+      }
+      render_delay_buffer->PrepareCaptureProcessing();
+
+      analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
+                      absl::optional<size_t>(0));
+    }
+
+    mask.fill(1.f);
+    analyzer.MaskRegionsAroundNarrowBands(&mask);
+    EXPECT_TRUE(std::all_of(mask.begin(), mask.end(),
+                            [](float a) { return a == 1.f; }));
+    EXPECT_FALSE(analyzer.PoorSignalExcitation());
+    EXPECT_FALSE(static_cast<bool>(analyzer.NarrowPeakBand()));
+  }
+}
+
+// Verify that a sinusoid signal is detected as narrow bands.
+TEST(RenderSignalAnalyzer, NarrowBandDetection) {
+  for (auto num_channels : {1, 2, 8}) {
+    SCOPED_TRACE(ProduceDebugText(num_channels));
+    RunNarrowBandDetectionTest(num_channels);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator.cc
new file mode 100644
index 0000000000..640a3e3cb9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator.cc
@@ -0,0 +1,379 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/residual_echo_estimator.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/reverb_model.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kDefaultTransparentModeGain = 0.01f;
+
+float GetTransparentModeGain() {
+  return kDefaultTransparentModeGain;
+}
+
+float GetEarlyReflectionsDefaultModeGain(
+    const EchoCanceller3Config::EpStrength& config) {
+  if (field_trial::IsEnabled("WebRTC-Aec3UseLowEarlyReflectionsDefaultGain")) {
+    return 0.1f;
+  }
+  return config.default_gain;
+}
+
+float GetLateReflectionsDefaultModeGain(
+    const EchoCanceller3Config::EpStrength& config) {
+  if (field_trial::IsEnabled("WebRTC-Aec3UseLowLateReflectionsDefaultGain")) {
+    return 0.1f;
+  }
+  return config.default_gain;
+}
+
+bool UseErleOnsetCompensationInDominantNearend(
+    const EchoCanceller3Config::EpStrength& config) {
+  return config.erle_onset_compensation_in_dominant_nearend ||
+         field_trial::IsEnabled(
+             "WebRTC-Aec3UseErleOnsetCompensationInDominantNearend");
+}
+
+// Computes the indexes that will be used for computing spectral power over
+// the blocks surrounding the delay.
+void GetRenderIndexesToAnalyze(
+    const SpectrumBuffer& spectrum_buffer,
+    const EchoCanceller3Config::EchoModel& echo_model,
+    int filter_delay_blocks,
+    int* idx_start,
+    int* idx_stop) {
+  RTC_DCHECK(idx_start);
+  RTC_DCHECK(idx_stop);
+  size_t window_start;
+  size_t window_end;
+  window_start =
+      std::max(0, filter_delay_blocks -
+                      static_cast<int>(echo_model.render_pre_window_size));
+  window_end = filter_delay_blocks +
+               static_cast<int>(echo_model.render_post_window_size);
+  *idx_start = spectrum_buffer.OffsetIndex(spectrum_buffer.read, window_start);
+  *idx_stop = spectrum_buffer.OffsetIndex(spectrum_buffer.read, window_end + 1);
+}
+
+// Estimates the residual echo power based on the echo return loss enhancement
+// (ERLE) and the linear power estimate.
+void LinearEstimate(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> S2_linear,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> erle,
+    rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2) {
+  RTC_DCHECK_EQ(S2_linear.size(), erle.size());
+  RTC_DCHECK_EQ(S2_linear.size(), R2.size());
+
+  const size_t num_capture_channels = R2.size();
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      RTC_DCHECK_LT(0.f, erle[ch][k]);
+      R2[ch][k] = S2_linear[ch][k] / erle[ch][k];
+    }
+  }
+}
+
+// Estimates the residual echo power based on the estimate of the echo path
+// gain.
+void NonLinearEstimate(
+    float echo_path_gain,
+    const std::array<float, kFftLengthBy2Plus1>& X2,
+    rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2) {
+  const size_t num_capture_channels = R2.size();
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      R2[ch][k] = X2[k] * echo_path_gain;
+    }
+  }
+}
+
+// Applies a soft noise gate to the echo generating power.
+void ApplyNoiseGate(const EchoCanceller3Config::EchoModel& config,
+                    rtc::ArrayView<float, kFftLengthBy2Plus1> X2) {
+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+    if (config.noise_gate_power > X2[k]) {
+      X2[k] = std::max(0.f, X2[k] - config.noise_gate_slope *
+                                        (config.noise_gate_power - X2[k]));
+    }
+  }
+}
+
+// Estimates the echo generating signal power as gated maximal power over a
+// time window.
+void EchoGeneratingPower(size_t num_render_channels,
+                         const SpectrumBuffer& spectrum_buffer,
+                         const EchoCanceller3Config::EchoModel& echo_model,
+                         int filter_delay_blocks,
+                         rtc::ArrayView<float, kFftLengthBy2Plus1> X2) {
+  int idx_stop;
+  int idx_start;
+  GetRenderIndexesToAnalyze(spectrum_buffer, echo_model, filter_delay_blocks,
+                            &idx_start, &idx_stop);
+
+  std::fill(X2.begin(), X2.end(), 0.f);
+  if (num_render_channels == 1) {
+    for (int k = idx_start; k != idx_stop; k = spectrum_buffer.IncIndex(k)) {
+      for (size_t j = 0; j < kFftLengthBy2Plus1; ++j) {
+        X2[j] = std::max(X2[j], spectrum_buffer.buffer[k][/*channel=*/0][j]);
+      }
+    }
+  } else {
+    for (int k = idx_start; k != idx_stop; k = spectrum_buffer.IncIndex(k)) {
+      std::array<float, kFftLengthBy2Plus1> render_power;
+      render_power.fill(0.f);
+      for (size_t ch = 0; ch < num_render_channels; ++ch) {
+        const auto& channel_power = spectrum_buffer.buffer[k][ch];
+        for (size_t j = 0; j < kFftLengthBy2Plus1; ++j) {
+          render_power[j] += channel_power[j];
+        }
+      }
+      for (size_t j = 0; j < kFftLengthBy2Plus1; ++j) {
+        X2[j] = std::max(X2[j], render_power[j]);
+      }
+    }
+  }
+}
+
+}  // namespace
+
+ResidualEchoEstimator::ResidualEchoEstimator(const EchoCanceller3Config& config,
+                                             size_t num_render_channels)
+    : config_(config),
+      num_render_channels_(num_render_channels),
+      early_reflections_transparent_mode_gain_(GetTransparentModeGain()),
+      late_reflections_transparent_mode_gain_(GetTransparentModeGain()),
+      early_reflections_general_gain_(
+          GetEarlyReflectionsDefaultModeGain(config_.ep_strength)),
+      late_reflections_general_gain_(
+          GetLateReflectionsDefaultModeGain(config_.ep_strength)),
+      erle_onset_compensation_in_dominant_nearend_(
+          UseErleOnsetCompensationInDominantNearend(config_.ep_strength)) {
+  Reset();
+}
+
+ResidualEchoEstimator::~ResidualEchoEstimator() = default;
+
+void ResidualEchoEstimator::Estimate(
+    const AecState& aec_state,
+    const RenderBuffer& render_buffer,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> S2_linear,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    bool dominant_nearend,
+    rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2,
+    rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2_unbounded) {
+  RTC_DCHECK_EQ(R2.size(), Y2.size());
+  RTC_DCHECK_EQ(R2.size(), S2_linear.size());
+
+  const size_t num_capture_channels = R2.size();
+
+  // Estimate the power of the stationary noise in the render signal.
+  UpdateRenderNoisePower(render_buffer);
+
+  // Estimate the residual echo power.
+  if (aec_state.UsableLinearEstimate()) {
+    // When there is saturated echo, assume the same spectral content as is
+    // present in the microphone signal.
+    if (aec_state.SaturatedEcho()) {
+      for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+        std::copy(Y2[ch].begin(), Y2[ch].end(), R2[ch].begin());
+        std::copy(Y2[ch].begin(), Y2[ch].end(), R2_unbounded[ch].begin());
+      }
+    } else {
+      const bool onset_compensated =
+          erle_onset_compensation_in_dominant_nearend_ || !dominant_nearend;
+      LinearEstimate(S2_linear, aec_state.Erle(onset_compensated), R2);
+      LinearEstimate(S2_linear, aec_state.ErleUnbounded(), R2_unbounded);
+    }
+
+    UpdateReverb(ReverbType::kLinear, aec_state, render_buffer,
+                 dominant_nearend);
+    AddReverb(R2);
+    AddReverb(R2_unbounded);
+  } else {
+    const float echo_path_gain =
+        GetEchoPathGain(aec_state, /*gain_for_early_reflections=*/true);
+
+    // When there is saturated echo, assume the same spectral content as is
+    // present in the microphone signal.
+    if (aec_state.SaturatedEcho()) {
+      for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+        std::copy(Y2[ch].begin(), Y2[ch].end(), R2[ch].begin());
+        std::copy(Y2[ch].begin(), Y2[ch].end(), R2_unbounded[ch].begin());
+      }
+    } else {
+      // Estimate the echo generating signal power.
+      std::array<float, kFftLengthBy2Plus1> X2;
+      EchoGeneratingPower(num_render_channels_,
+                          render_buffer.GetSpectrumBuffer(), config_.echo_model,
+                          aec_state.MinDirectPathFilterDelay(), X2);
+      if (!aec_state.UseStationarityProperties()) {
+        ApplyNoiseGate(config_.echo_model, X2);
+      }
+
+      // Subtract the stationary noise power to avoid stationary noise causing
+      // excessive echo suppression.
+      for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+        X2[k] -= config_.echo_model.stationary_gate_slope * X2_noise_floor_[k];
+        X2[k] = std::max(0.f, X2[k]);
+      }
+
+      NonLinearEstimate(echo_path_gain, X2, R2);
+      NonLinearEstimate(echo_path_gain, X2, R2_unbounded);
+    }
+
+    if (config_.echo_model.model_reverb_in_nonlinear_mode &&
+        !aec_state.TransparentModeActive()) {
+      UpdateReverb(ReverbType::kNonLinear, aec_state, render_buffer,
+                   dominant_nearend);
+      AddReverb(R2);
+      AddReverb(R2_unbounded);
+    }
+  }
+
+  if (aec_state.UseStationarityProperties()) {
+    // Scale the echo according to echo audibility.
+    std::array<float, kFftLengthBy2Plus1> residual_scaling;
+    aec_state.GetResidualEchoScaling(residual_scaling);
+    for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+      for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+        R2[ch][k] *= residual_scaling[k];
+        R2_unbounded[ch][k] *= residual_scaling[k];
+      }
+    }
+  }
+}
+
+void ResidualEchoEstimator::Reset() {
+  echo_reverb_.Reset();
+  X2_noise_floor_counter_.fill(config_.echo_model.noise_floor_hold);
+  X2_noise_floor_.fill(config_.echo_model.min_noise_floor_power);
+}
+
+void ResidualEchoEstimator::UpdateRenderNoisePower(
+    const RenderBuffer& render_buffer) {
+  std::array<float, kFftLengthBy2Plus1> render_power_data;
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> X2 =
+      render_buffer.Spectrum(0);
+  rtc::ArrayView<const float, kFftLengthBy2Plus1> render_power =
+      X2[/*channel=*/0];
+  if (num_render_channels_ > 1) {
+    render_power_data.fill(0.f);
+    for (size_t ch = 0; ch < num_render_channels_; ++ch) {
+      const auto& channel_power = X2[ch];
+      for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+        render_power_data[k] += channel_power[k];
+      }
+    }
+    render_power = render_power_data;
+  }
+
+  // Estimate the stationary noise power in a minimum statistics manner.
+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+    // Decrease rapidly.
+    if (render_power[k] < X2_noise_floor_[k]) {
+      X2_noise_floor_[k] = render_power[k];
+      X2_noise_floor_counter_[k] = 0;
+    } else {
+      // Increase in a delayed, leaky manner.
+      if (X2_noise_floor_counter_[k] >=
+          static_cast<int>(config_.echo_model.noise_floor_hold)) {
+        X2_noise_floor_[k] = std::max(X2_noise_floor_[k] * 1.1f,
+                                      config_.echo_model.min_noise_floor_power);
+      } else {
+        ++X2_noise_floor_counter_[k];
+      }
+    }
+  }
+}
+
+// Updates the reverb estimation.
+void ResidualEchoEstimator::UpdateReverb(ReverbType reverb_type,
+                                         const AecState& aec_state,
+                                         const RenderBuffer& render_buffer,
+                                         bool dominant_nearend) {
+  // Choose reverb partition based on what type of echo power model is used.
+  const size_t first_reverb_partition =
+      reverb_type == ReverbType::kLinear
+          ? aec_state.FilterLengthBlocks() + 1
+          : aec_state.MinDirectPathFilterDelay() + 1;
+
+  // Compute render power for the reverb.
+  std::array<float, kFftLengthBy2Plus1> render_power_data;
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> X2 =
+      render_buffer.Spectrum(first_reverb_partition);
+  rtc::ArrayView<const float, kFftLengthBy2Plus1> render_power =
+      X2[/*channel=*/0];
+  if (num_render_channels_ > 1) {
+    render_power_data.fill(0.f);
+    for (size_t ch = 0; ch < num_render_channels_; ++ch) {
+      const auto& channel_power = X2[ch];
+      for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+        render_power_data[k] += channel_power[k];
+      }
+    }
+    render_power = render_power_data;
+  }
+
+  // Update the reverb estimate.
+  float reverb_decay = aec_state.ReverbDecay(/*mild=*/dominant_nearend);
+  if (reverb_type == ReverbType::kLinear) {
+    echo_reverb_.UpdateReverb(
+        render_power, aec_state.GetReverbFrequencyResponse(), reverb_decay);
+  } else {
+    const float echo_path_gain =
+        GetEchoPathGain(aec_state, /*gain_for_early_reflections=*/false);
+    echo_reverb_.UpdateReverbNoFreqShaping(render_power, echo_path_gain,
+                                           reverb_decay);
+  }
+}
+// Adds the estimated power of the reverb to the residual echo power.
+void ResidualEchoEstimator::AddReverb(
+    rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2) const {
+  const size_t num_capture_channels = R2.size();
+
+  // Add the reverb power.
+  rtc::ArrayView<const float, kFftLengthBy2Plus1> reverb_power =
+      echo_reverb_.reverb();
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      R2[ch][k] += reverb_power[k];
+    }
+  }
+}
+
+// Chooses the echo path gain to use.
+float ResidualEchoEstimator::GetEchoPathGain(
+    const AecState& aec_state,
+    bool gain_for_early_reflections) const {
+  float gain_amplitude;
+  if (aec_state.TransparentModeActive()) {
+    gain_amplitude = gain_for_early_reflections
+                         ? early_reflections_transparent_mode_gain_
+                         : late_reflections_transparent_mode_gain_;
+  } else {
+    gain_amplitude = gain_for_early_reflections
+                         ? early_reflections_general_gain_
+                         : late_reflections_general_gain_;
+  }
+  return gain_amplitude * gain_amplitude;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator.h
new file mode 100644
index 0000000000..c468764002
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_RESIDUAL_ECHO_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_RESIDUAL_ECHO_ESTIMATOR_H_
+
+#include <array>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/reverb_model.h"
+#include "modules/audio_processing/aec3/spectrum_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class ResidualEchoEstimator {
+ public:
+  ResidualEchoEstimator(const EchoCanceller3Config& config,
+                        size_t num_render_channels);
+  ~ResidualEchoEstimator();
+
+  ResidualEchoEstimator(const ResidualEchoEstimator&) = delete;
+  ResidualEchoEstimator& operator=(const ResidualEchoEstimator&) = delete;
+
+  void Estimate(
+      const AecState& aec_state,
+      const RenderBuffer& render_buffer,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> S2_linear,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      bool dominant_nearend,
+      rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2,
+      rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2_unbounded);
+
+ private:
+  enum class ReverbType { kLinear, kNonLinear };
+
+  // Resets the state.
+  void Reset();
+
+  // Updates estimate for the power of the stationary noise component in the
+  // render signal.
+  void UpdateRenderNoisePower(const RenderBuffer& render_buffer);
+
+  // Updates the reverb estimation.
+  void UpdateReverb(ReverbType reverb_type,
+                    const AecState& aec_state,
+                    const RenderBuffer& render_buffer,
+                    bool dominant_nearend);
+
+  // Adds the estimated unmodelled echo power to the residual echo power
+  // estimate.
+  void AddReverb(
+      rtc::ArrayView<std::array<float, kFftLengthBy2Plus1>> R2) const;
+
+  // Gets the echo path gain to apply.
+  float GetEchoPathGain(const AecState& aec_state,
+                        bool gain_for_early_reflections) const;
+
+  const EchoCanceller3Config config_;
+  const size_t num_render_channels_;
+  const float early_reflections_transparent_mode_gain_;
+  const float late_reflections_transparent_mode_gain_;
+  const float early_reflections_general_gain_;
+  const float late_reflections_general_gain_;
+  const bool erle_onset_compensation_in_dominant_nearend_;
+  std::array<float, kFftLengthBy2Plus1> X2_noise_floor_;
+  std::array<int, kFftLengthBy2Plus1> X2_noise_floor_counter_;
+  ReverbModel echo_reverb_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RESIDUAL_ECHO_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
new file mode 100644
index 0000000000..9a7bf0a89c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
@@ -0,0 +1,199 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/residual_echo_estimator.h"
+
+#include <numeric>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kSampleRateHz = 48000;
+constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+constexpr float kEpsilon = 1e-4f;
+}  // namespace
+
+class ResidualEchoEstimatorTest {
+ public:
+  ResidualEchoEstimatorTest(size_t num_render_channels,
+                            size_t num_capture_channels,
+                            const EchoCanceller3Config& config)
+      : num_render_channels_(num_render_channels),
+        num_capture_channels_(num_capture_channels),
+        config_(config),
+        estimator_(config_, num_render_channels_),
+        aec_state_(config_, num_capture_channels_),
+        render_delay_buffer_(RenderDelayBuffer::Create(config_,
+                                                       kSampleRateHz,
+                                                       num_render_channels_)),
+        E2_refined_(num_capture_channels_),
+        S2_linear_(num_capture_channels_),
+        Y2_(num_capture_channels_),
+        R2_(num_capture_channels_),
+        R2_unbounded_(num_capture_channels_),
+        x_(kNumBands, num_render_channels_),
+        H2_(num_capture_channels_,
+            std::vector<std::array<float, kFftLengthBy2Plus1>>(10)),
+        h_(num_capture_channels_,
+           std::vector<float>(
+               GetTimeDomainLength(config_.filter.refined.length_blocks),
+               0.0f)),
+        random_generator_(42U),
+        output_(num_capture_channels_) {
+    for (auto& H2_ch : H2_) {
+      for (auto& H2_k : H2_ch) {
+        H2_k.fill(0.01f);
+      }
+      H2_ch[2].fill(10.f);
+      H2_ch[2][0] = 0.1f;
+    }
+
+    for (auto& subtractor_output : output_) {
+      subtractor_output.Reset();
+      subtractor_output.s_refined.fill(100.f);
+    }
+    y_.fill(0.f);
+
+    constexpr float kLevel = 10.f;
+    for (auto& E2_refined_ch : E2_refined_) {
+      E2_refined_ch.fill(kLevel);
+    }
+    S2_linear_[0].fill(kLevel);
+    for (auto& Y2_ch : Y2_) {
+      Y2_ch.fill(kLevel);
+    }
+  }
+
+  void RunOneFrame(bool dominant_nearend) {
+    RandomizeSampleVector(&random_generator_,
+                          x_.View(/*band=*/0, /*channel=*/0));
+    render_delay_buffer_->Insert(x_);
+    if (first_frame_) {
+      render_delay_buffer_->Reset();
+      first_frame_ = false;
+    }
+    render_delay_buffer_->PrepareCaptureProcessing();
+
+    aec_state_.Update(delay_estimate_, H2_, h_,
+                      *render_delay_buffer_->GetRenderBuffer(), E2_refined_,
+                      Y2_, output_);
+
+    estimator_.Estimate(aec_state_, *render_delay_buffer_->GetRenderBuffer(),
+                        S2_linear_, Y2_, dominant_nearend, R2_, R2_unbounded_);
+  }
+
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> R2() const {
+    return R2_;
+  }
+
+ private:
+  const size_t num_render_channels_;
+  const size_t num_capture_channels_;
+  const EchoCanceller3Config& config_;
+  ResidualEchoEstimator estimator_;
+  AecState aec_state_;
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2_refined_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> S2_linear_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> R2_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> R2_unbounded_;
+  Block x_;
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> H2_;
+  std::vector<std::vector<float>> h_;
+  Random random_generator_;
+  std::vector<SubtractorOutput> output_;
+  std::array<float, kBlockSize> y_;
+  absl::optional<DelayEstimate> delay_estimate_;
+  bool first_frame_ = true;
+};
+
+class ResidualEchoEstimatorMultiChannel
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         ResidualEchoEstimatorMultiChannel,
+                         ::testing::Combine(::testing::Values(1, 2, 4),
+                                            ::testing::Values(1, 2, 4)));
+
+TEST_P(ResidualEchoEstimatorMultiChannel, BasicTest) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+
+  EchoCanceller3Config config;
+  ResidualEchoEstimatorTest residual_echo_estimator_test(
+      num_render_channels, num_capture_channels, config);
+  for (int k = 0; k < 1993; ++k) {
+    residual_echo_estimator_test.RunOneFrame(/*dominant_nearend=*/false);
+  }
+}
+
+TEST(ResidualEchoEstimatorMultiChannel, ReverbTest) {
+  const size_t num_render_channels = 1;
+  const size_t num_capture_channels = 1;
+  const size_t nFrames = 100;
+
+  EchoCanceller3Config reference_config;
+  reference_config.ep_strength.default_len = 0.95f;
+  reference_config.ep_strength.nearend_len = 0.95f;
+  EchoCanceller3Config config_use_nearend_len = reference_config;
+  config_use_nearend_len.ep_strength.default_len = 0.95f;
+  config_use_nearend_len.ep_strength.nearend_len = 0.83f;
+
+  ResidualEchoEstimatorTest reference_residual_echo_estimator_test(
+      num_render_channels, num_capture_channels, reference_config);
+  ResidualEchoEstimatorTest use_nearend_len_residual_echo_estimator_test(
+      num_render_channels, num_capture_channels, config_use_nearend_len);
+
+  std::vector<float> acum_energy_reference_R2(num_capture_channels, 0.0f);
+  std::vector<float> acum_energy_R2(num_capture_channels, 0.0f);
+  for (size_t frame = 0; frame < nFrames; ++frame) {
+    bool dominant_nearend = frame <= nFrames / 2 ? false : true;
+    reference_residual_echo_estimator_test.RunOneFrame(dominant_nearend);
+    use_nearend_len_residual_echo_estimator_test.RunOneFrame(dominant_nearend);
+    const auto& reference_R2 = reference_residual_echo_estimator_test.R2();
+    const auto& R2 = use_nearend_len_residual_echo_estimator_test.R2();
+    ASSERT_EQ(reference_R2.size(), R2.size());
+    for (size_t ch = 0; ch < reference_R2.size(); ++ch) {
+      float energy_reference_R2 = std::accumulate(
+          reference_R2[ch].cbegin(), reference_R2[ch].cend(), 0.0f);
+      float energy_R2 = std::accumulate(R2[ch].cbegin(), R2[ch].cend(), 0.0f);
+      if (dominant_nearend) {
+        EXPECT_GE(energy_reference_R2, energy_R2);
+      } else {
+        EXPECT_NEAR(energy_reference_R2, energy_R2, kEpsilon);
+      }
+      acum_energy_reference_R2[ch] += energy_reference_R2;
+      acum_energy_R2[ch] += energy_R2;
+    }
+    if (frame == nFrames / 2 || frame == nFrames - 1) {
+      for (size_t ch = 0; ch < acum_energy_reference_R2.size(); ch++) {
+        if (dominant_nearend) {
+          EXPECT_GT(acum_energy_reference_R2[ch], acum_energy_R2[ch]);
+        } else {
+          EXPECT_NEAR(acum_energy_reference_R2[ch], acum_energy_R2[ch],
+                      kEpsilon);
+        }
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc
new file mode 100644
index 0000000000..2daf376911
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc
@@ -0,0 +1,410 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/reverb_decay_estimator.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <cmath>
+#include <numeric>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kEarlyReverbMinSizeBlocks = 3;
+constexpr int kBlocksPerSection = 6;
+// Linear regression approach assumes symmetric index around 0.
+constexpr float kEarlyReverbFirstPointAtLinearRegressors =
+    -0.5f * kBlocksPerSection * kFftLengthBy2 + 0.5f;
+
+// Averages the values in a block of size kFftLengthBy2;
+float BlockAverage(rtc::ArrayView<const float> v, size_t block_index) {
+  constexpr float kOneByFftLengthBy2 = 1.f / kFftLengthBy2;
+  const int i = block_index * kFftLengthBy2;
+  RTC_DCHECK_GE(v.size(), i + kFftLengthBy2);
+  const float sum =
+      std::accumulate(v.begin() + i, v.begin() + i + kFftLengthBy2, 0.f);
+  return sum * kOneByFftLengthBy2;
+}
+
+// Analyzes the gain in a block.
+void AnalyzeBlockGain(const std::array<float, kFftLengthBy2>& h2,
+                      float floor_gain,
+                      float* previous_gain,
+                      bool* block_adapting,
+                      bool* decaying_gain) {
+  float gain = std::max(BlockAverage(h2, 0), 1e-32f);
+  *block_adapting =
+      *previous_gain > 1.1f * gain || *previous_gain < 0.9f * gain;
+  *decaying_gain = gain > floor_gain;
+  *previous_gain = gain;
+}
+
+// Arithmetic sum of $2 \sum_{i=0.5}^{(N-1)/2}i^2$ calculated directly.
+constexpr float SymmetricArithmetricSum(int N) {
+  return N * (N * N - 1.0f) * (1.f / 12.f);
+}
+
+// Returns the peak energy of an impulse response.
+float BlockEnergyPeak(rtc::ArrayView<const float> h, int peak_block) {
+  RTC_DCHECK_LE((peak_block + 1) * kFftLengthBy2, h.size());
+  RTC_DCHECK_GE(peak_block, 0);
+  float peak_value =
+      *std::max_element(h.begin() + peak_block * kFftLengthBy2,
+                        h.begin() + (peak_block + 1) * kFftLengthBy2,
+                        [](float a, float b) { return a * a < b * b; });
+  return peak_value * peak_value;
+}
+
+// Returns the average energy of an impulse response block.
+float BlockEnergyAverage(rtc::ArrayView<const float> h, int block_index) {
+  RTC_DCHECK_LE((block_index + 1) * kFftLengthBy2, h.size());
+  RTC_DCHECK_GE(block_index, 0);
+  constexpr float kOneByFftLengthBy2 = 1.f / kFftLengthBy2;
+  const auto sum_of_squares = [](float a, float b) { return a + b * b; };
+  return std::accumulate(h.begin() + block_index * kFftLengthBy2,
+                         h.begin() + (block_index + 1) * kFftLengthBy2, 0.f,
+                         sum_of_squares) *
+         kOneByFftLengthBy2;
+}
+
+}  // namespace
+
+ReverbDecayEstimator::ReverbDecayEstimator(const EchoCanceller3Config& config)
+    : filter_length_blocks_(config.filter.refined.length_blocks),
+      filter_length_coefficients_(GetTimeDomainLength(filter_length_blocks_)),
+      use_adaptive_echo_decay_(config.ep_strength.default_len < 0.f),
+      early_reverb_estimator_(config.filter.refined.length_blocks -
+                              kEarlyReverbMinSizeBlocks),
+      late_reverb_start_(kEarlyReverbMinSizeBlocks),
+      late_reverb_end_(kEarlyReverbMinSizeBlocks),
+      previous_gains_(config.filter.refined.length_blocks, 0.f),
+      decay_(std::fabs(config.ep_strength.default_len)),
+      mild_decay_(std::fabs(config.ep_strength.nearend_len)) {
+  RTC_DCHECK_GT(config.filter.refined.length_blocks,
+                static_cast<size_t>(kEarlyReverbMinSizeBlocks));
+}
+
+ReverbDecayEstimator::~ReverbDecayEstimator() = default;
+
+void ReverbDecayEstimator::Update(rtc::ArrayView<const float> filter,
+                                  const absl::optional<float>& filter_quality,
+                                  int filter_delay_blocks,
+                                  bool usable_linear_filter,
+                                  bool stationary_signal) {
+  const int filter_size = static_cast<int>(filter.size());
+
+  if (stationary_signal) {
+    return;
+  }
+
+  bool estimation_feasible =
+      filter_delay_blocks <=
+      filter_length_blocks_ - kEarlyReverbMinSizeBlocks - 1;
+  estimation_feasible =
+      estimation_feasible && filter_size == filter_length_coefficients_;
+  estimation_feasible = estimation_feasible && filter_delay_blocks > 0;
+  estimation_feasible = estimation_feasible && usable_linear_filter;
+
+  if (!estimation_feasible) {
+    ResetDecayEstimation();
+    return;
+  }
+
+  if (!use_adaptive_echo_decay_) {
+    return;
+  }
+
+  const float new_smoothing = filter_quality ? *filter_quality * 0.2f : 0.f;
+  smoothing_constant_ = std::max(new_smoothing, smoothing_constant_);
+  if (smoothing_constant_ == 0.f) {
+    return;
+  }
+
+  if (block_to_analyze_ < filter_length_blocks_) {
+    // Analyze the filter and accumulate data for reverb estimation.
+    AnalyzeFilter(filter);
+    ++block_to_analyze_;
+  } else {
+    // When the filter is fully analyzed, estimate the reverb decay and reset
+    // the block_to_analyze_ counter.
+    EstimateDecay(filter, filter_delay_blocks);
+  }
+}
+
+void ReverbDecayEstimator::ResetDecayEstimation() {
+  early_reverb_estimator_.Reset();
+  late_reverb_decay_estimator_.Reset(0);
+  block_to_analyze_ = 0;
+  estimation_region_candidate_size_ = 0;
+  estimation_region_identified_ = false;
+  smoothing_constant_ = 0.f;
+  late_reverb_start_ = 0;
+  late_reverb_end_ = 0;
+}
+
+void ReverbDecayEstimator::EstimateDecay(rtc::ArrayView<const float> filter,
+                                         int peak_block) {
+  auto& h = filter;
+  RTC_DCHECK_EQ(0, h.size() % kFftLengthBy2);
+
+  // Reset the block analysis counter.
+  block_to_analyze_ =
+      std::min(peak_block + kEarlyReverbMinSizeBlocks, filter_length_blocks_);
+
+  // To estimate the reverb decay, the energy of the first filter section must
+  // be substantially larger than the last. Also, the first filter section
+  // energy must not deviate too much from the max peak.
+  const float first_reverb_gain = BlockEnergyAverage(h, block_to_analyze_);
+  const size_t h_size_blocks = h.size() >> kFftLengthBy2Log2;
+  tail_gain_ = BlockEnergyAverage(h, h_size_blocks - 1);
+  float peak_energy = BlockEnergyPeak(h, peak_block);
+  const bool sufficient_reverb_decay = first_reverb_gain > 4.f * tail_gain_;
+  const bool valid_filter =
+      first_reverb_gain > 2.f * tail_gain_ && peak_energy < 100.f;
+
+  // Estimate the size of the regions with early and late reflections.
+  const int size_early_reverb = early_reverb_estimator_.Estimate();
+  const int size_late_reverb =
+      std::max(estimation_region_candidate_size_ - size_early_reverb, 0);
+
+  // Only update the reverb decay estimate if the size of the identified late
+  // reverb is sufficiently large.
+  if (size_late_reverb >= 5) {
+    if (valid_filter && late_reverb_decay_estimator_.EstimateAvailable()) {
+      float decay = std::pow(
+          2.0f, late_reverb_decay_estimator_.Estimate() * kFftLengthBy2);
+      constexpr float kMaxDecay = 0.95f;  // ~1 sec min RT60.
+      constexpr float kMinDecay = 0.02f;  // ~15 ms max RT60.
+      decay = std::max(.97f * decay_, decay);
+      decay = std::min(decay, kMaxDecay);
+      decay = std::max(decay, kMinDecay);
+      decay_ += smoothing_constant_ * (decay - decay_);
+    }
+
+    // Update length of decay. Must have enough data (number of sections) in
+    // order to estimate decay rate.
+    late_reverb_decay_estimator_.Reset(size_late_reverb * kFftLengthBy2);
+    late_reverb_start_ =
+        peak_block + kEarlyReverbMinSizeBlocks + size_early_reverb;
+    late_reverb_end_ =
+        block_to_analyze_ + estimation_region_candidate_size_ - 1;
+  } else {
+    late_reverb_decay_estimator_.Reset(0);
+    late_reverb_start_ = 0;
+    late_reverb_end_ = 0;
+  }
+
+  // Reset variables for the identification of the region for reverb decay
+  // estimation.
+  estimation_region_identified_ = !(valid_filter && sufficient_reverb_decay);
+  estimation_region_candidate_size_ = 0;
+
+  // Stop estimation of the decay until another good filter is received.
+  smoothing_constant_ = 0.f;
+
+  // Reset early reflections detector.
+  early_reverb_estimator_.Reset();
+}
+
+void ReverbDecayEstimator::AnalyzeFilter(rtc::ArrayView<const float> filter) {
+  auto h = rtc::ArrayView<const float>(
+      filter.begin() + block_to_analyze_ * kFftLengthBy2, kFftLengthBy2);
+
+  // Compute squared filter coeffiecients for the block to analyze_;
+  std::array<float, kFftLengthBy2> h2;
+  std::transform(h.begin(), h.end(), h2.begin(), [](float a) { return a * a; });
+
+  // Map out the region for estimating the reverb decay.
+  bool adapting;
+  bool above_noise_floor;
+  AnalyzeBlockGain(h2, tail_gain_, &previous_gains_[block_to_analyze_],
+                   &adapting, &above_noise_floor);
+
+  // Count consecutive number of "good" filter sections, where "good" means:
+  // 1) energy is above noise floor.
+  // 2) energy of current section has not changed too much from last check.
+  estimation_region_identified_ =
+      estimation_region_identified_ || adapting || !above_noise_floor;
+  if (!estimation_region_identified_) {
+    ++estimation_region_candidate_size_;
+  }
+
+  // Accumulate data for reverb decay estimation and for the estimation of early
+  // reflections.
+  if (block_to_analyze_ <= late_reverb_end_) {
+    if (block_to_analyze_ >= late_reverb_start_) {
+      for (float h2_k : h2) {
+        float h2_log2 = FastApproxLog2f(h2_k + 1e-10);
+        late_reverb_decay_estimator_.Accumulate(h2_log2);
+        early_reverb_estimator_.Accumulate(h2_log2, smoothing_constant_);
+      }
+    } else {
+      for (float h2_k : h2) {
+        float h2_log2 = FastApproxLog2f(h2_k + 1e-10);
+        early_reverb_estimator_.Accumulate(h2_log2, smoothing_constant_);
+      }
+    }
+  }
+}
+
+void ReverbDecayEstimator::Dump(ApmDataDumper* data_dumper) const {
+  data_dumper->DumpRaw("aec3_reverb_decay", decay_);
+  data_dumper->DumpRaw("aec3_reverb_tail_energy", tail_gain_);
+  data_dumper->DumpRaw("aec3_reverb_alpha", smoothing_constant_);
+  data_dumper->DumpRaw("aec3_num_reverb_decay_blocks",
+                       late_reverb_end_ - late_reverb_start_);
+  data_dumper->DumpRaw("aec3_late_reverb_start", late_reverb_start_);
+  data_dumper->DumpRaw("aec3_late_reverb_end", late_reverb_end_);
+  early_reverb_estimator_.Dump(data_dumper);
+}
+
+void ReverbDecayEstimator::LateReverbLinearRegressor::Reset(
+    int num_data_points) {
+  RTC_DCHECK_LE(0, num_data_points);
+  RTC_DCHECK_EQ(0, num_data_points % 2);
+  const int N = num_data_points;
+  nz_ = 0.f;
+  // Arithmetic sum of $2 \sum_{i=0.5}^{(N-1)/2}i^2$ calculated directly.
+  nn_ = SymmetricArithmetricSum(N);
+  // The linear regression approach assumes symmetric index around 0.
+  count_ = N > 0 ? -N * 0.5f + 0.5f : 0.f;
+  N_ = N;
+  n_ = 0;
+}
+
+void ReverbDecayEstimator::LateReverbLinearRegressor::Accumulate(float z) {
+  nz_ += count_ * z;
+  ++count_;
+  ++n_;
+}
+
+float ReverbDecayEstimator::LateReverbLinearRegressor::Estimate() {
+  RTC_DCHECK(EstimateAvailable());
+  if (nn_ == 0.f) {
+    RTC_DCHECK_NOTREACHED();
+    return 0.f;
+  }
+  return nz_ / nn_;
+}
+
+ReverbDecayEstimator::EarlyReverbLengthEstimator::EarlyReverbLengthEstimator(
+    int max_blocks)
+    : numerators_smooth_(max_blocks - kBlocksPerSection, 0.f),
+      numerators_(numerators_smooth_.size(), 0.f),
+      coefficients_counter_(0) {
+  RTC_DCHECK_LE(0, max_blocks);
+}
+
+ReverbDecayEstimator::EarlyReverbLengthEstimator::
+    ~EarlyReverbLengthEstimator() = default;
+
+void ReverbDecayEstimator::EarlyReverbLengthEstimator::Reset() {
+  coefficients_counter_ = 0;
+  std::fill(numerators_.begin(), numerators_.end(), 0.f);
+  block_counter_ = 0;
+}
+
+void ReverbDecayEstimator::EarlyReverbLengthEstimator::Accumulate(
+    float value,
+    float smoothing) {
+  // Each section is composed by kBlocksPerSection blocks and each section
+  // overlaps with the next one in (kBlocksPerSection - 1) blocks. For example,
+  // the first section covers the blocks [0:5], the second covers the blocks
+  // [1:6] and so on. As a result, for each value, kBlocksPerSection sections
+  // need to be updated.
+  int first_section_index = std::max(block_counter_ - kBlocksPerSection + 1, 0);
+  int last_section_index =
+      std::min(block_counter_, static_cast<int>(numerators_.size() - 1));
+  float x_value = static_cast<float>(coefficients_counter_) +
+                  kEarlyReverbFirstPointAtLinearRegressors;
+  const float value_to_inc = kFftLengthBy2 * value;
+  float value_to_add =
+      x_value * value + (block_counter_ - last_section_index) * value_to_inc;
+  for (int section = last_section_index; section >= first_section_index;
+       --section, value_to_add += value_to_inc) {
+    numerators_[section] += value_to_add;
+  }
+
+  // Check if this update was the last coefficient of the current block. In that
+  // case, check if we are at the end of one of the sections and update the
+  // numerator of the linear regressor that is computed in such section.
+  if (++coefficients_counter_ == kFftLengthBy2) {
+    if (block_counter_ >= (kBlocksPerSection - 1)) {
+      size_t section = block_counter_ - (kBlocksPerSection - 1);
+      RTC_DCHECK_GT(numerators_.size(), section);
+      RTC_DCHECK_GT(numerators_smooth_.size(), section);
+      numerators_smooth_[section] +=
+          smoothing * (numerators_[section] - numerators_smooth_[section]);
+      n_sections_ = section + 1;
+    }
+    ++block_counter_;
+    coefficients_counter_ = 0;
+  }
+}
+
+// Estimates the size in blocks of the early reverb. The estimation is done by
+// comparing the tilt that is estimated in each section. As an optimization
+// detail and due to the fact that all the linear regressors that are computed
+// shared the same denominator, the comparison of the tilts is done by a
+// comparison of the numerator of the linear regressors.
+int ReverbDecayEstimator::EarlyReverbLengthEstimator::Estimate() {
+  constexpr float N = kBlocksPerSection * kFftLengthBy2;
+  constexpr float nn = SymmetricArithmetricSum(N);
+  // numerator_11 refers to the quantity that the linear regressor needs in the
+  // numerator for getting a decay equal to 1.1 (which is not a decay).
+  // log2(1.1) * nn / kFftLengthBy2.
+  constexpr float numerator_11 = 0.13750352374993502f * nn / kFftLengthBy2;
+  // log2(0.8) *  nn / kFftLengthBy2.
+  constexpr float numerator_08 = -0.32192809488736229f * nn / kFftLengthBy2;
+  constexpr int kNumSectionsToAnalyze = 9;
+
+  if (n_sections_ < kNumSectionsToAnalyze) {
+    return 0;
+  }
+
+  // Estimation of the blocks that correspond to early reverberations. The
+  // estimation is done by analyzing the impulse response. The portions of the
+  // impulse response whose energy is not decreasing over its coefficients are
+  // considered to be part of the early reverberations. Furthermore, the blocks
+  // where the energy is decreasing faster than what it does at the end of the
+  // impulse response are also considered to be part of the early
+  // reverberations. The estimation is limited to the first
+  // kNumSectionsToAnalyze sections.
+
+  RTC_DCHECK_LE(n_sections_, numerators_smooth_.size());
+  const float min_numerator_tail =
+      *std::min_element(numerators_smooth_.begin() + kNumSectionsToAnalyze,
+                        numerators_smooth_.begin() + n_sections_);
+  int early_reverb_size_minus_1 = 0;
+  for (int k = 0; k < kNumSectionsToAnalyze; ++k) {
+    if ((numerators_smooth_[k] > numerator_11) ||
+        (numerators_smooth_[k] < numerator_08 &&
+         numerators_smooth_[k] < 0.9f * min_numerator_tail)) {
+      early_reverb_size_minus_1 = k;
+    }
+  }
+
+  return early_reverb_size_minus_1 == 0 ? 0 : early_reverb_size_minus_1 + 1;
+}
+
+void ReverbDecayEstimator::EarlyReverbLengthEstimator::Dump(
+    ApmDataDumper* data_dumper) const {
+  data_dumper->DumpRaw("aec3_er_acum_numerator", numerators_smooth_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/reverb_decay_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_decay_estimator.h
new file mode 100644
index 0000000000..fee54210e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_decay_estimator.h
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_
+
+#include <array>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"  // kMaxAdaptiveFilter...
+
+namespace webrtc {
+
+class ApmDataDumper;
+struct EchoCanceller3Config;
+
+// Class for estimating the decay of the late reverb.
+class ReverbDecayEstimator {
+ public:
+  explicit ReverbDecayEstimator(const EchoCanceller3Config& config);
+  ~ReverbDecayEstimator();
+  // Updates the decay estimate.
+  void Update(rtc::ArrayView<const float> filter,
+              const absl::optional<float>& filter_quality,
+              int filter_delay_blocks,
+              bool usable_linear_filter,
+              bool stationary_signal);
+  // Returns the decay for the exponential model. The parameter `mild` indicates
+  // which exponential decay to return, the default one or a milder one.
+  float Decay(bool mild) const {
+    if (use_adaptive_echo_decay_) {
+      return decay_;
+    } else {
+      return mild ? mild_decay_ : decay_;
+    }
+  }
+  // Dumps debug data.
+  void Dump(ApmDataDumper* data_dumper) const;
+
+ private:
+  void EstimateDecay(rtc::ArrayView<const float> filter, int peak_block);
+  void AnalyzeFilter(rtc::ArrayView<const float> filter);
+
+  void ResetDecayEstimation();
+
+  // Class for estimating the decay of the late reverb from the linear filter.
+  class LateReverbLinearRegressor {
+   public:
+    // Resets the estimator to receive a specified number of data points.
+    void Reset(int num_data_points);
+    // Accumulates estimation data.
+    void Accumulate(float z);
+    // Estimates the decay.
+    float Estimate();
+    // Returns whether an estimate is available.
+    bool EstimateAvailable() const { return n_ == N_ && N_ != 0; }
+
+   public:
+    float nz_ = 0.f;
+    float nn_ = 0.f;
+    float count_ = 0.f;
+    int N_ = 0;
+    int n_ = 0;
+  };
+
+  // Class for identifying the length of the early reverb from the linear
+  // filter. For identifying the early reverberations, the impulse response is
+  // divided in sections and the tilt of each section is computed by a linear
+  // regressor.
+  class EarlyReverbLengthEstimator {
+   public:
+    explicit EarlyReverbLengthEstimator(int max_blocks);
+    ~EarlyReverbLengthEstimator();
+
+    // Resets the estimator.
+    void Reset();
+    // Accumulates estimation data.
+    void Accumulate(float value, float smoothing);
+    // Estimates the size in blocks of the early reverb.
+    int Estimate();
+    // Dumps debug data.
+    void Dump(ApmDataDumper* data_dumper) const;
+
+   private:
+    std::vector<float> numerators_smooth_;
+    std::vector<float> numerators_;
+    int coefficients_counter_;
+    int block_counter_ = 0;
+    int n_sections_ = 0;
+  };
+
+  const int filter_length_blocks_;
+  const int filter_length_coefficients_;
+  const bool use_adaptive_echo_decay_;
+  LateReverbLinearRegressor late_reverb_decay_estimator_;
+  EarlyReverbLengthEstimator early_reverb_estimator_;
+  int late_reverb_start_;
+  int late_reverb_end_;
+  int block_to_analyze_ = 0;
+  int estimation_region_candidate_size_ = 0;
+  bool estimation_region_identified_ = false;
+  std::vector<float> previous_gains_;
+  float decay_;
+  float mild_decay_;
+  float tail_gain_ = 0.f;
+  float smoothing_constant_ = 0.f;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/reverb_frequency_response.cc b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_frequency_response.cc
new file mode 100644
index 0000000000..6e7282a1fc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_frequency_response.cc
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/reverb_frequency_response.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <array>
+#include <numeric>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Computes the ratio of the energies between the direct path and the tail. The
+// energy is computed in the power spectrum domain discarding the DC
+// contributions.
+float AverageDecayWithinFilter(
+    rtc::ArrayView<const float> freq_resp_direct_path,
+    rtc::ArrayView<const float> freq_resp_tail) {
+  // Skipping the DC for the ratio computation
+  constexpr size_t kSkipBins = 1;
+  RTC_CHECK_EQ(freq_resp_direct_path.size(), freq_resp_tail.size());
+
+  float direct_path_energy =
+      std::accumulate(freq_resp_direct_path.begin() + kSkipBins,
+                      freq_resp_direct_path.end(), 0.f);
+
+  if (direct_path_energy == 0.f) {
+    return 0.f;
+  }
+
+  float tail_energy = std::accumulate(freq_resp_tail.begin() + kSkipBins,
+                                      freq_resp_tail.end(), 0.f);
+  return tail_energy / direct_path_energy;
+}
+
+}  // namespace
+
+ReverbFrequencyResponse::ReverbFrequencyResponse(
+    bool use_conservative_tail_frequency_response)
+    : use_conservative_tail_frequency_response_(
+          use_conservative_tail_frequency_response) {
+  tail_response_.fill(0.0f);
+}
+
+ReverbFrequencyResponse::~ReverbFrequencyResponse() = default;
+
+void ReverbFrequencyResponse::Update(
+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
+        frequency_response,
+    int filter_delay_blocks,
+    const absl::optional<float>& linear_filter_quality,
+    bool stationary_block) {
+  if (stationary_block || !linear_filter_quality) {
+    return;
+  }
+
+  Update(frequency_response, filter_delay_blocks, *linear_filter_quality);
+}
+
+void ReverbFrequencyResponse::Update(
+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
+        frequency_response,
+    int filter_delay_blocks,
+    float linear_filter_quality) {
+  rtc::ArrayView<const float> freq_resp_tail(
+      frequency_response[frequency_response.size() - 1]);
+
+  rtc::ArrayView<const float> freq_resp_direct_path(
+      frequency_response[filter_delay_blocks]);
+
+  float average_decay =
+      AverageDecayWithinFilter(freq_resp_direct_path, freq_resp_tail);
+
+  const float smoothing = 0.2f * linear_filter_quality;
+  average_decay_ += smoothing * (average_decay - average_decay_);
+
+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+    tail_response_[k] = freq_resp_direct_path[k] * average_decay_;
+  }
+
+  if (use_conservative_tail_frequency_response_) {
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      tail_response_[k] = std::max(freq_resp_tail[k], tail_response_[k]);
+    }
+  }
+
+  for (size_t k = 1; k < kFftLengthBy2; ++k) {
+    const float avg_neighbour =
+        0.5f * (tail_response_[k - 1] + tail_response_[k + 1]);
+    tail_response_[k] = std::max(tail_response_[k], avg_neighbour);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/reverb_frequency_response.h b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_frequency_response.h
new file mode 100644
index 0000000000..69b16b54d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_frequency_response.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
+
+#include <array>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+
+// Class for updating the frequency response for the reverb.
+class ReverbFrequencyResponse {
+ public:
+  explicit ReverbFrequencyResponse(
+      bool use_conservative_tail_frequency_response);
+  ~ReverbFrequencyResponse();
+
+  // Updates the frequency response estimate of the reverb.
+  void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
+                  frequency_response,
+              int filter_delay_blocks,
+              const absl::optional<float>& linear_filter_quality,
+              bool stationary_block);
+
+  // Returns the estimated frequency response for the reverb.
+  rtc::ArrayView<const float> FrequencyResponse() const {
+    return tail_response_;
+  }
+
+ private:
+  void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
+                  frequency_response,
+              int filter_delay_blocks,
+              float linear_filter_quality);
+
+  const bool use_conservative_tail_frequency_response_;
+  float average_decay_ = 0.f;
+  std::array<float, kFftLengthBy2Plus1> tail_response_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model.cc b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model.cc
new file mode 100644
index 0000000000..e4f3507d31
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/reverb_model.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <functional>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+ReverbModel::ReverbModel() {
+  Reset();
+}
+
+ReverbModel::~ReverbModel() = default;
+
+void ReverbModel::Reset() {
+  reverb_.fill(0.);
+}
+
+void ReverbModel::UpdateReverbNoFreqShaping(
+    rtc::ArrayView<const float> power_spectrum,
+    float power_spectrum_scaling,
+    float reverb_decay) {
+  if (reverb_decay > 0) {
+    // Update the estimate of the reverberant power.
+    for (size_t k = 0; k < power_spectrum.size(); ++k) {
+      reverb_[k] = (reverb_[k] + power_spectrum[k] * power_spectrum_scaling) *
+                   reverb_decay;
+    }
+  }
+}
+
+void ReverbModel::UpdateReverb(
+    rtc::ArrayView<const float> power_spectrum,
+    rtc::ArrayView<const float> power_spectrum_scaling,
+    float reverb_decay) {
+  if (reverb_decay > 0) {
+    // Update the estimate of the reverberant power.
+    for (size_t k = 0; k < power_spectrum.size(); ++k) {
+      reverb_[k] =
+          (reverb_[k] + power_spectrum[k] * power_spectrum_scaling[k]) *
+          reverb_decay;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model.h b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model.h
new file mode 100644
index 0000000000..5ba54853da
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_H_
+
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+
+// The ReverbModel class describes an exponential reverberant model
+// that can be applied over power spectrums.
+class ReverbModel {
+ public:
+  ReverbModel();
+  ~ReverbModel();
+
+  // Resets the state.
+  void Reset();
+
+  // Returns the reverb.
+  rtc::ArrayView<const float, kFftLengthBy2Plus1> reverb() const {
+    return reverb_;
+  }
+
+  // The methods UpdateReverbNoFreqShaping and UpdateReverb update the
+  // estimate of the reverberation contribution to an input/output power
+  // spectrum. Before applying the exponential reverberant model, the input
+  // power spectrum is pre-scaled. Use the method UpdateReverb when a different
+  // scaling should be applied per frequency and UpdateReverb_no_freq_shape if
+  // the same scaling should be used for all the frequencies.
+  void UpdateReverbNoFreqShaping(rtc::ArrayView<const float> power_spectrum,
+                                 float power_spectrum_scaling,
+                                 float reverb_decay);
+
+  // Update the reverb based on new data.
+  void UpdateReverb(rtc::ArrayView<const float> power_spectrum,
+                    rtc::ArrayView<const float> power_spectrum_scaling,
+                    float reverb_decay);
+
+ private:
+
+  std::array<float, kFftLengthBy2Plus1> reverb_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator.cc
new file mode 100644
index 0000000000..5cd7a7870d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/reverb_model_estimator.h"
+
+namespace webrtc {
+
+ReverbModelEstimator::ReverbModelEstimator(const EchoCanceller3Config& config,
+                                           size_t num_capture_channels)
+    : reverb_decay_estimators_(num_capture_channels),
+      reverb_frequency_responses_(
+          num_capture_channels,
+          ReverbFrequencyResponse(
+              config.ep_strength.use_conservative_tail_frequency_response)) {
+  for (size_t ch = 0; ch < reverb_decay_estimators_.size(); ++ch) {
+    reverb_decay_estimators_[ch] =
+        std::make_unique<ReverbDecayEstimator>(config);
+  }
+}
+
+ReverbModelEstimator::~ReverbModelEstimator() = default;
+
+void ReverbModelEstimator::Update(
+    rtc::ArrayView<const std::vector<float>> impulse_responses,
+    rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+        frequency_responses,
+    rtc::ArrayView<const absl::optional<float>> linear_filter_qualities,
+    rtc::ArrayView<const int> filter_delays_blocks,
+    const std::vector<bool>& usable_linear_estimates,
+    bool stationary_block) {
+  const size_t num_capture_channels = reverb_decay_estimators_.size();
+  RTC_DCHECK_EQ(num_capture_channels, impulse_responses.size());
+  RTC_DCHECK_EQ(num_capture_channels, frequency_responses.size());
+  RTC_DCHECK_EQ(num_capture_channels, usable_linear_estimates.size());
+
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    // Estimate the frequency response for the reverb.
+    reverb_frequency_responses_[ch].Update(
+        frequency_responses[ch], filter_delays_blocks[ch],
+        linear_filter_qualities[ch], stationary_block);
+
+    // Estimate the reverb decay,
+    reverb_decay_estimators_[ch]->Update(
+        impulse_responses[ch], linear_filter_qualities[ch],
+        filter_delays_blocks[ch], usable_linear_estimates[ch],
+        stationary_block);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator.h
new file mode 100644
index 0000000000..63bade977f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_ESTIMATOR_H_
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"  // kFftLengthBy2Plus1
+#include "modules/audio_processing/aec3/reverb_decay_estimator.h"
+#include "modules/audio_processing/aec3/reverb_frequency_response.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+// Class for estimating the model parameters for the reverberant echo.
+class ReverbModelEstimator {
+ public:
+  ReverbModelEstimator(const EchoCanceller3Config& config,
+                       size_t num_capture_channels);
+  ~ReverbModelEstimator();
+
+  // Updates the estimates based on new data.
+  void Update(
+      rtc::ArrayView<const std::vector<float>> impulse_responses,
+      rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+          frequency_responses,
+      rtc::ArrayView<const absl::optional<float>> linear_filter_qualities,
+      rtc::ArrayView<const int> filter_delays_blocks,
+      const std::vector<bool>& usable_linear_estimates,
+      bool stationary_block);
+
+  // Returns the exponential decay of the reverberant echo. The parameter `mild`
+  // indicates which exponential decay to return, the default one or a milder
+  // one.
+  // TODO(peah): Correct to properly support multiple channels.
+  float ReverbDecay(bool mild) const {
+    return reverb_decay_estimators_[0]->Decay(mild);
+  }
+
+  // Return the frequency response of the reverberant echo.
+  // TODO(peah): Correct to properly support multiple channels.
+  rtc::ArrayView<const float> GetReverbFrequencyResponse() const {
+    return reverb_frequency_responses_[0].FrequencyResponse();
+  }
+
+  // Dumps debug data.
+  void Dump(ApmDataDumper* data_dumper) const {
+    reverb_decay_estimators_[0]->Dump(data_dumper);
+  }
+
+ private:
+  std::vector<std::unique_ptr<ReverbDecayEstimator>> reverb_decay_estimators_;
+  std::vector<ReverbFrequencyResponse> reverb_frequency_responses_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
new file mode 100644
index 0000000000..fb7dcef37f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
@@ -0,0 +1,157 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/reverb_model_estimator.h"
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <numeric>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+EchoCanceller3Config CreateConfigForTest(float default_decay) {
+  EchoCanceller3Config cfg;
+  cfg.ep_strength.default_len = default_decay;
+  cfg.filter.refined.length_blocks = 40;
+  return cfg;
+}
+
+constexpr int kFilterDelayBlocks = 2;
+
+}  // namespace
+
+class ReverbModelEstimatorTest {
+ public:
+  ReverbModelEstimatorTest(float default_decay, size_t num_capture_channels)
+      : aec3_config_(CreateConfigForTest(default_decay)),
+        estimated_decay_(default_decay),
+        h_(num_capture_channels,
+           std::vector<float>(
+               aec3_config_.filter.refined.length_blocks * kBlockSize,
+               0.f)),
+        H2_(num_capture_channels,
+            std::vector<std::array<float, kFftLengthBy2Plus1>>(
+                aec3_config_.filter.refined.length_blocks)),
+        quality_linear_(num_capture_channels, 1.0f) {
+    CreateImpulseResponseWithDecay();
+  }
+  void RunEstimator();
+  float GetDecay(bool mild) {
+    return mild ? mild_estimated_decay_ : estimated_decay_;
+  }
+  float GetTrueDecay() { return kTruePowerDecay; }
+  float GetPowerTailDb() { return 10.f * std::log10(estimated_power_tail_); }
+  float GetTruePowerTailDb() { return 10.f * std::log10(true_power_tail_); }
+
+ private:
+  void CreateImpulseResponseWithDecay();
+  static constexpr bool kStationaryBlock = false;
+  static constexpr float kTruePowerDecay = 0.5f;
+  const EchoCanceller3Config aec3_config_;
+  float estimated_decay_;
+  float mild_estimated_decay_;
+  float estimated_power_tail_ = 0.f;
+  float true_power_tail_ = 0.f;
+  std::vector<std::vector<float>> h_;
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> H2_;
+  std::vector<absl::optional<float>> quality_linear_;
+};
+
+void ReverbModelEstimatorTest::CreateImpulseResponseWithDecay() {
+  const Aec3Fft fft;
+  for (const auto& h_k : h_) {
+    RTC_DCHECK_EQ(h_k.size(),
+                  aec3_config_.filter.refined.length_blocks * kBlockSize);
+  }
+  for (const auto& H2_k : H2_) {
+    RTC_DCHECK_EQ(H2_k.size(), aec3_config_.filter.refined.length_blocks);
+  }
+  RTC_DCHECK_EQ(kFilterDelayBlocks, 2);
+
+  float decay_sample = std::sqrt(powf(kTruePowerDecay, 1.f / kBlockSize));
+  const size_t filter_delay_coefficients = kFilterDelayBlocks * kBlockSize;
+  for (auto& h_i : h_) {
+    std::fill(h_i.begin(), h_i.end(), 0.f);
+    h_i[filter_delay_coefficients] = 1.f;
+    for (size_t k = filter_delay_coefficients + 1; k < h_i.size(); ++k) {
+      h_i[k] = h_i[k - 1] * decay_sample;
+    }
+  }
+
+  for (size_t ch = 0; ch < H2_.size(); ++ch) {
+    for (size_t j = 0, k = 0; j < H2_[ch].size(); ++j, k += kBlockSize) {
+      std::array<float, kFftLength> fft_data;
+      fft_data.fill(0.f);
+      std::copy(h_[ch].begin() + k, h_[ch].begin() + k + kBlockSize,
+                fft_data.begin());
+      FftData H_j;
+      fft.Fft(&fft_data, &H_j);
+      H_j.Spectrum(Aec3Optimization::kNone, H2_[ch][j]);
+    }
+  }
+  rtc::ArrayView<float> H2_tail(H2_[0][H2_[0].size() - 1]);
+  true_power_tail_ = std::accumulate(H2_tail.begin(), H2_tail.end(), 0.f);
+}
+void ReverbModelEstimatorTest::RunEstimator() {
+  const size_t num_capture_channels = H2_.size();
+  constexpr bool kUsableLinearEstimate = true;
+  ReverbModelEstimator estimator(aec3_config_, num_capture_channels);
+  std::vector<bool> usable_linear_estimates(num_capture_channels,
+                                            kUsableLinearEstimate);
+  std::vector<int> filter_delay_blocks(num_capture_channels,
+                                       kFilterDelayBlocks);
+  for (size_t k = 0; k < 3000; ++k) {
+    estimator.Update(h_, H2_, quality_linear_, filter_delay_blocks,
+                     usable_linear_estimates, kStationaryBlock);
+  }
+  estimated_decay_ = estimator.ReverbDecay(/*mild=*/false);
+  mild_estimated_decay_ = estimator.ReverbDecay(/*mild=*/true);
+  auto freq_resp_tail = estimator.GetReverbFrequencyResponse();
+  estimated_power_tail_ =
+      std::accumulate(freq_resp_tail.begin(), freq_resp_tail.end(), 0.f);
+}
+
+TEST(ReverbModelEstimatorTests, NotChangingDecay) {
+  constexpr float kDefaultDecay = 0.9f;
+  for (size_t num_capture_channels : {1, 2, 4, 8}) {
+    ReverbModelEstimatorTest test(kDefaultDecay, num_capture_channels);
+    test.RunEstimator();
+    EXPECT_EQ(test.GetDecay(/*mild=*/false), kDefaultDecay);
+    EXPECT_EQ(test.GetDecay(/*mild=*/true),
+              EchoCanceller3Config().ep_strength.nearend_len);
+    EXPECT_NEAR(test.GetPowerTailDb(), test.GetTruePowerTailDb(), 5.f);
+  }
+}
+
+TEST(ReverbModelEstimatorTests, ChangingDecay) {
+  constexpr float kDefaultDecay = -0.9f;
+  for (size_t num_capture_channels : {1, 2, 4, 8}) {
+    ReverbModelEstimatorTest test(kDefaultDecay, num_capture_channels);
+    test.RunEstimator();
+    EXPECT_NEAR(test.GetDecay(/*mild=*/false), test.GetTrueDecay(), 0.1f);
+    EXPECT_NEAR(test.GetDecay(/*mild=*/true), test.GetTrueDecay(), 0.1f);
+    EXPECT_NEAR(test.GetPowerTailDb(), test.GetTruePowerTailDb(), 5.f);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
new file mode 100644
index 0000000000..a5e77092a6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
@@ -0,0 +1,416 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/signal_dependent_erle_estimator.h"
+
+#include <algorithm>
+#include <functional>
+#include <numeric>
+
+#include "modules/audio_processing/aec3/spectrum_buffer.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr std::array<size_t, SignalDependentErleEstimator::kSubbands + 1>
+    kBandBoundaries = {1, 8, 16, 24, 32, 48, kFftLengthBy2Plus1};
+
+std::array<size_t, kFftLengthBy2Plus1> FormSubbandMap() {
+  std::array<size_t, kFftLengthBy2Plus1> map_band_to_subband;
+  size_t subband = 1;
+  for (size_t k = 0; k < map_band_to_subband.size(); ++k) {
+    RTC_DCHECK_LT(subband, kBandBoundaries.size());
+    if (k >= kBandBoundaries[subband]) {
+      subband++;
+      RTC_DCHECK_LT(k, kBandBoundaries[subband]);
+    }
+    map_band_to_subband[k] = subband - 1;
+  }
+  return map_band_to_subband;
+}
+
+// Defines the size in blocks of the sections that are used for dividing the
+// linear filter. The sections are split in a non-linear manner so that lower
+// sections that typically represent the direct path have a larger resolution
+// than the higher sections which typically represent more reverberant acoustic
+// paths.
+std::vector<size_t> DefineFilterSectionSizes(size_t delay_headroom_blocks,
+                                             size_t num_blocks,
+                                             size_t num_sections) {
+  size_t filter_length_blocks = num_blocks - delay_headroom_blocks;
+  std::vector<size_t> section_sizes(num_sections);
+  size_t remaining_blocks = filter_length_blocks;
+  size_t remaining_sections = num_sections;
+  size_t estimator_size = 2;
+  size_t idx = 0;
+  while (remaining_sections > 1 &&
+         remaining_blocks > estimator_size * remaining_sections) {
+    RTC_DCHECK_LT(idx, section_sizes.size());
+    section_sizes[idx] = estimator_size;
+    remaining_blocks -= estimator_size;
+    remaining_sections--;
+    estimator_size *= 2;
+    idx++;
+  }
+
+  size_t last_groups_size = remaining_blocks / remaining_sections;
+  for (; idx < num_sections; idx++) {
+    section_sizes[idx] = last_groups_size;
+  }
+  section_sizes[num_sections - 1] +=
+      remaining_blocks - last_groups_size * remaining_sections;
+  return section_sizes;
+}
+
+// Forms the limits in blocks for each filter section. Those sections
+// are used for analyzing the echo estimates and investigating which
+// linear filter sections contribute most to the echo estimate energy.
+std::vector<size_t> SetSectionsBoundaries(size_t delay_headroom_blocks,
+                                          size_t num_blocks,
+                                          size_t num_sections) {
+  std::vector<size_t> estimator_boundaries_blocks(num_sections + 1);
+  if (estimator_boundaries_blocks.size() == 2) {
+    estimator_boundaries_blocks[0] = 0;
+    estimator_boundaries_blocks[1] = num_blocks;
+    return estimator_boundaries_blocks;
+  }
+  RTC_DCHECK_GT(estimator_boundaries_blocks.size(), 2);
+  const std::vector<size_t> section_sizes =
+      DefineFilterSectionSizes(delay_headroom_blocks, num_blocks,
+                               estimator_boundaries_blocks.size() - 1);
+
+  size_t idx = 0;
+  size_t current_size_block = 0;
+  RTC_DCHECK_EQ(section_sizes.size() + 1, estimator_boundaries_blocks.size());
+  estimator_boundaries_blocks[0] = delay_headroom_blocks;
+  for (size_t k = delay_headroom_blocks; k < num_blocks; ++k) {
+    current_size_block++;
+    if (current_size_block >= section_sizes[idx]) {
+      idx = idx + 1;
+      if (idx == section_sizes.size()) {
+        break;
+      }
+      estimator_boundaries_blocks[idx] = k + 1;
+      current_size_block = 0;
+    }
+  }
+  estimator_boundaries_blocks[section_sizes.size()] = num_blocks;
+  return estimator_boundaries_blocks;
+}
+
+std::array<float, SignalDependentErleEstimator::kSubbands>
+SetMaxErleSubbands(float max_erle_l, float max_erle_h, size_t limit_subband_l) {
+  std::array<float, SignalDependentErleEstimator::kSubbands> max_erle;
+  std::fill(max_erle.begin(), max_erle.begin() + limit_subband_l, max_erle_l);
+  std::fill(max_erle.begin() + limit_subband_l, max_erle.end(), max_erle_h);
+  return max_erle;
+}
+
+}  // namespace
+
+SignalDependentErleEstimator::SignalDependentErleEstimator(
+    const EchoCanceller3Config& config,
+    size_t num_capture_channels)
+    : min_erle_(config.erle.min),
+      num_sections_(config.erle.num_sections),
+      num_blocks_(config.filter.refined.length_blocks),
+      delay_headroom_blocks_(config.delay.delay_headroom_samples / kBlockSize),
+      band_to_subband_(FormSubbandMap()),
+      max_erle_(SetMaxErleSubbands(config.erle.max_l,
+                                   config.erle.max_h,
+                                   band_to_subband_[kFftLengthBy2 / 2])),
+      section_boundaries_blocks_(SetSectionsBoundaries(delay_headroom_blocks_,
+                                                       num_blocks_,
+                                                       num_sections_)),
+      use_onset_detection_(config.erle.onset_detection),
+      erle_(num_capture_channels),
+      erle_onset_compensated_(num_capture_channels),
+      S2_section_accum_(
+          num_capture_channels,
+          std::vector<std::array<float, kFftLengthBy2Plus1>>(num_sections_)),
+      erle_estimators_(
+          num_capture_channels,
+          std::vector<std::array<float, kSubbands>>(num_sections_)),
+      erle_ref_(num_capture_channels),
+      correction_factors_(
+          num_capture_channels,
+          std::vector<std::array<float, kSubbands>>(num_sections_)),
+      num_updates_(num_capture_channels),
+      n_active_sections_(num_capture_channels) {
+  RTC_DCHECK_LE(num_sections_, num_blocks_);
+  RTC_DCHECK_GE(num_sections_, 1);
+  Reset();
+}
+
+SignalDependentErleEstimator::~SignalDependentErleEstimator() = default;
+
+void SignalDependentErleEstimator::Reset() {
+  for (size_t ch = 0; ch < erle_.size(); ++ch) {
+    erle_[ch].fill(min_erle_);
+    erle_onset_compensated_[ch].fill(min_erle_);
+    for (auto& erle_estimator : erle_estimators_[ch]) {
+      erle_estimator.fill(min_erle_);
+    }
+    erle_ref_[ch].fill(min_erle_);
+    for (auto& factor : correction_factors_[ch]) {
+      factor.fill(1.0f);
+    }
+    num_updates_[ch].fill(0);
+    n_active_sections_[ch].fill(0);
+  }
+}
+
+// Updates the Erle estimate by analyzing the current input signals. It takes
+// the render buffer and the filter frequency response in order to do an
+// estimation of the number of sections of the linear filter that are needed
+// for getting the majority of the energy in the echo estimate. Based on that
+// number of sections, it updates the erle estimation by introducing a
+// correction factor to the erle that is given as an input to this method.
+void SignalDependentErleEstimator::Update(
+    const RenderBuffer& render_buffer,
+    rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+        filter_frequency_responses,
+    rtc::ArrayView<const float, kFftLengthBy2Plus1> X2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> average_erle,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        average_erle_onset_compensated,
+    const std::vector<bool>& converged_filters) {
+  RTC_DCHECK_GT(num_sections_, 1);
+
+  // Gets the number of filter sections that are needed for achieving 90 %
+  // of the power spectrum energy of the echo estimate.
+  ComputeNumberOfActiveFilterSections(render_buffer,
+                                      filter_frequency_responses);
+
+  // Updates the correction factors that is used for correcting the erle and
+  // adapt it to the particular characteristics of the input signal.
+  UpdateCorrectionFactors(X2, Y2, E2, converged_filters);
+
+  // Applies the correction factor to the input erle for getting a more refined
+  // erle estimation for the current input signal.
+  for (size_t ch = 0; ch < erle_.size(); ++ch) {
+    for (size_t k = 0; k < kFftLengthBy2; ++k) {
+      RTC_DCHECK_GT(correction_factors_[ch].size(), n_active_sections_[ch][k]);
+      float correction_factor =
+          correction_factors_[ch][n_active_sections_[ch][k]]
+                             [band_to_subband_[k]];
+      erle_[ch][k] = rtc::SafeClamp(average_erle[ch][k] * correction_factor,
+                                    min_erle_, max_erle_[band_to_subband_[k]]);
+      if (use_onset_detection_) {
+        erle_onset_compensated_[ch][k] = rtc::SafeClamp(
+            average_erle_onset_compensated[ch][k] * correction_factor,
+            min_erle_, max_erle_[band_to_subband_[k]]);
+      }
+    }
+  }
+}
+
+void SignalDependentErleEstimator::Dump(
+    const std::unique_ptr<ApmDataDumper>& data_dumper) const {
+  for (auto& erle : erle_estimators_[0]) {
+    data_dumper->DumpRaw("aec3_all_erle", erle);
+  }
+  data_dumper->DumpRaw("aec3_ref_erle", erle_ref_[0]);
+  for (auto& factor : correction_factors_[0]) {
+    data_dumper->DumpRaw("aec3_erle_correction_factor", factor);
+  }
+}
+
+// Estimates for each band the smallest number of sections in the filter that
+// together constitute 90% of the estimated echo energy.
+void SignalDependentErleEstimator::ComputeNumberOfActiveFilterSections(
+    const RenderBuffer& render_buffer,
+    rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+        filter_frequency_responses) {
+  RTC_DCHECK_GT(num_sections_, 1);
+  // Computes an approximation of the power spectrum if the filter would have
+  // been limited to a certain number of filter sections.
+  ComputeEchoEstimatePerFilterSection(render_buffer,
+                                      filter_frequency_responses);
+  // For each band, computes the number of filter sections that are needed for
+  // achieving the 90 % energy in the echo estimate.
+  ComputeActiveFilterSections();
+}
+
+void SignalDependentErleEstimator::UpdateCorrectionFactors(
+    rtc::ArrayView<const float, kFftLengthBy2Plus1> X2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    const std::vector<bool>& converged_filters) {
+  for (size_t ch = 0; ch < converged_filters.size(); ++ch) {
+    if (converged_filters[ch]) {
+      constexpr float kX2BandEnergyThreshold = 44015068.0f;
+      constexpr float kSmthConstantDecreases = 0.1f;
+      constexpr float kSmthConstantIncreases = kSmthConstantDecreases / 2.f;
+      auto subband_powers = [](rtc::ArrayView<const float> power_spectrum,
+                               rtc::ArrayView<float> power_spectrum_subbands) {
+        for (size_t subband = 0; subband < kSubbands; ++subband) {
+          RTC_DCHECK_LE(kBandBoundaries[subband + 1], power_spectrum.size());
+          power_spectrum_subbands[subband] = std::accumulate(
+              power_spectrum.begin() + kBandBoundaries[subband],
+              power_spectrum.begin() + kBandBoundaries[subband + 1], 0.f);
+        }
+      };
+
+      std::array<float, kSubbands> X2_subbands, E2_subbands, Y2_subbands;
+      subband_powers(X2, X2_subbands);
+      subband_powers(E2[ch], E2_subbands);
+      subband_powers(Y2[ch], Y2_subbands);
+      std::array<size_t, kSubbands> idx_subbands;
+      for (size_t subband = 0; subband < kSubbands; ++subband) {
+        // When aggregating the number of active sections in the filter for
+        // different bands we choose to take the minimum of all of them. As an
+        // example, if for one of the bands it is the direct path its refined
+        // contributor to the final echo estimate, we consider the direct path
+        // is as well the refined contributor for the subband that contains that
+        // particular band. That aggregate number of sections will be later used
+        // as the identifier of the erle estimator that needs to be updated.
+        RTC_DCHECK_LE(kBandBoundaries[subband + 1],
+                      n_active_sections_[ch].size());
+        idx_subbands[subband] = *std::min_element(
+            n_active_sections_[ch].begin() + kBandBoundaries[subband],
+            n_active_sections_[ch].begin() + kBandBoundaries[subband + 1]);
+      }
+
+      std::array<float, kSubbands> new_erle;
+      std::array<bool, kSubbands> is_erle_updated;
+      is_erle_updated.fill(false);
+      new_erle.fill(0.f);
+      for (size_t subband = 0; subband < kSubbands; ++subband) {
+        if (X2_subbands[subband] > kX2BandEnergyThreshold &&
+            E2_subbands[subband] > 0) {
+          new_erle[subband] = Y2_subbands[subband] / E2_subbands[subband];
+          RTC_DCHECK_GT(new_erle[subband], 0);
+          is_erle_updated[subband] = true;
+          ++num_updates_[ch][subband];
+        }
+      }
+
+      for (size_t subband = 0; subband < kSubbands; ++subband) {
+        const size_t idx = idx_subbands[subband];
+        RTC_DCHECK_LT(idx, erle_estimators_[ch].size());
+        float alpha = new_erle[subband] > erle_estimators_[ch][idx][subband]
+                          ? kSmthConstantIncreases
+                          : kSmthConstantDecreases;
+        alpha = static_cast<float>(is_erle_updated[subband]) * alpha;
+        erle_estimators_[ch][idx][subband] +=
+            alpha * (new_erle[subband] - erle_estimators_[ch][idx][subband]);
+        erle_estimators_[ch][idx][subband] = rtc::SafeClamp(
+            erle_estimators_[ch][idx][subband], min_erle_, max_erle_[subband]);
+      }
+
+      for (size_t subband = 0; subband < kSubbands; ++subband) {
+        float alpha = new_erle[subband] > erle_ref_[ch][subband]
+                          ? kSmthConstantIncreases
+                          : kSmthConstantDecreases;
+        alpha = static_cast<float>(is_erle_updated[subband]) * alpha;
+        erle_ref_[ch][subband] +=
+            alpha * (new_erle[subband] - erle_ref_[ch][subband]);
+        erle_ref_[ch][subband] = rtc::SafeClamp(erle_ref_[ch][subband],
+                                                min_erle_, max_erle_[subband]);
+      }
+
+      for (size_t subband = 0; subband < kSubbands; ++subband) {
+        constexpr int kNumUpdateThr = 50;
+        if (is_erle_updated[subband] &&
+            num_updates_[ch][subband] > kNumUpdateThr) {
+          const size_t idx = idx_subbands[subband];
+          RTC_DCHECK_GT(erle_ref_[ch][subband], 0.f);
+          // Computes the ratio between the erle that is updated using all the
+          // points and the erle that is updated only on signals that share the
+          // same number of active filter sections.
+          float new_correction_factor =
+              erle_estimators_[ch][idx][subband] / erle_ref_[ch][subband];
+
+          correction_factors_[ch][idx][subband] +=
+              0.1f *
+              (new_correction_factor - correction_factors_[ch][idx][subband]);
+        }
+      }
+    }
+  }
+}
+
+void SignalDependentErleEstimator::ComputeEchoEstimatePerFilterSection(
+    const RenderBuffer& render_buffer,
+    rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+        filter_frequency_responses) {
+  const SpectrumBuffer& spectrum_render_buffer =
+      render_buffer.GetSpectrumBuffer();
+  const size_t num_render_channels = spectrum_render_buffer.buffer[0].size();
+  const size_t num_capture_channels = S2_section_accum_.size();
+  const float one_by_num_render_channels = 1.f / num_render_channels;
+
+  RTC_DCHECK_EQ(S2_section_accum_.size(), filter_frequency_responses.size());
+
+  for (size_t capture_ch = 0; capture_ch < num_capture_channels; ++capture_ch) {
+    RTC_DCHECK_EQ(S2_section_accum_[capture_ch].size() + 1,
+                  section_boundaries_blocks_.size());
+    size_t idx_render = render_buffer.Position();
+    idx_render = spectrum_render_buffer.OffsetIndex(
+        idx_render, section_boundaries_blocks_[0]);
+
+    for (size_t section = 0; section < num_sections_; ++section) {
+      std::array<float, kFftLengthBy2Plus1> X2_section;
+      std::array<float, kFftLengthBy2Plus1> H2_section;
+      X2_section.fill(0.f);
+      H2_section.fill(0.f);
+      const size_t block_limit =
+          std::min(section_boundaries_blocks_[section + 1],
+                   filter_frequency_responses[capture_ch].size());
+      for (size_t block = section_boundaries_blocks_[section];
+           block < block_limit; ++block) {
+        for (size_t render_ch = 0;
+             render_ch < spectrum_render_buffer.buffer[idx_render].size();
+             ++render_ch) {
+          for (size_t k = 0; k < X2_section.size(); ++k) {
+            X2_section[k] +=
+                spectrum_render_buffer.buffer[idx_render][render_ch][k] *
+                one_by_num_render_channels;
+          }
+        }
+        std::transform(H2_section.begin(), H2_section.end(),
+                       filter_frequency_responses[capture_ch][block].begin(),
+                       H2_section.begin(), std::plus<float>());
+        idx_render = spectrum_render_buffer.IncIndex(idx_render);
+      }
+
+      std::transform(X2_section.begin(), X2_section.end(), H2_section.begin(),
+                     S2_section_accum_[capture_ch][section].begin(),
+                     std::multiplies<float>());
+    }
+
+    for (size_t section = 1; section < num_sections_; ++section) {
+      std::transform(S2_section_accum_[capture_ch][section - 1].begin(),
+                     S2_section_accum_[capture_ch][section - 1].end(),
+                     S2_section_accum_[capture_ch][section].begin(),
+                     S2_section_accum_[capture_ch][section].begin(),
+                     std::plus<float>());
+    }
+  }
+}
+
+void SignalDependentErleEstimator::ComputeActiveFilterSections() {
+  for (size_t ch = 0; ch < n_active_sections_.size(); ++ch) {
+    std::fill(n_active_sections_[ch].begin(), n_active_sections_[ch].end(), 0);
+    for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+      size_t section = num_sections_;
+      float target = 0.9f * S2_section_accum_[ch][num_sections_ - 1][k];
+      while (section > 0 && S2_section_accum_[ch][section - 1][k] >= target) {
+        n_active_sections_[ch][k] = --section;
+      }
+    }
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.h
new file mode 100644
index 0000000000..6847c1ab13
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.h
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SIGNAL_DEPENDENT_ERLE_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SIGNAL_DEPENDENT_ERLE_ESTIMATOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+// This class estimates the dependency of the Erle to the input signal. By
+// looking at the input signal, an estimation on whether the current echo
+// estimate is due to the direct path or to a more reverberant one is performed.
+// Once that estimation is done, it is possible to refine the average Erle that
+// this class receive as an input.
+class SignalDependentErleEstimator {
+ public:
+  SignalDependentErleEstimator(const EchoCanceller3Config& config,
+                               size_t num_capture_channels);
+
+  ~SignalDependentErleEstimator();
+
+  void Reset();
+
+  // Returns the Erle per frequency subband.
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Erle(
+      bool onset_compensated) const {
+    return onset_compensated && use_onset_detection_ ? erle_onset_compensated_
+                                                     : erle_;
+  }
+
+  // Updates the Erle estimate. The Erle that is passed as an input is required
+  // to be an estimation of the average Erle achieved by the linear filter.
+  void Update(
+      const RenderBuffer& render_buffer,
+      rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+          filter_frequency_response,
+      rtc::ArrayView<const float, kFftLengthBy2Plus1> X2,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> average_erle,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          average_erle_onset_compensated,
+      const std::vector<bool>& converged_filters);
+
+  void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper) const;
+
+  static constexpr size_t kSubbands = 6;
+
+ private:
+  void ComputeNumberOfActiveFilterSections(
+      const RenderBuffer& render_buffer,
+      rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+          filter_frequency_responses);
+
+  void UpdateCorrectionFactors(
+      rtc::ArrayView<const float, kFftLengthBy2Plus1> X2,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+      const std::vector<bool>& converged_filters);
+
+  void ComputeEchoEstimatePerFilterSection(
+      const RenderBuffer& render_buffer,
+      rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+          filter_frequency_responses);
+
+  void ComputeActiveFilterSections();
+
+  const float min_erle_;
+  const size_t num_sections_;
+  const size_t num_blocks_;
+  const size_t delay_headroom_blocks_;
+  const std::array<size_t, kFftLengthBy2Plus1> band_to_subband_;
+  const std::array<float, kSubbands> max_erle_;
+  const std::vector<size_t> section_boundaries_blocks_;
+  const bool use_onset_detection_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> erle_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> erle_onset_compensated_;
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      S2_section_accum_;
+  std::vector<std::vector<std::array<float, kSubbands>>> erle_estimators_;
+  std::vector<std::array<float, kSubbands>> erle_ref_;
+  std::vector<std::vector<std::array<float, kSubbands>>> correction_factors_;
+  std::vector<std::array<int, kSubbands>> num_updates_;
+  std::vector<std::array<size_t, kFftLengthBy2Plus1>> n_active_sections_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SIGNAL_DEPENDENT_ERLE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc
new file mode 100644
index 0000000000..67927a6c68
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc
@@ -0,0 +1,208 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/signal_dependent_erle_estimator.h"
+
+#include <algorithm>
+#include <iostream>
+#include <string>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+void GetActiveFrame(Block* x) {
+  const std::array<float, kBlockSize> frame = {
+      7459.88, 17209.6, 17383,   20768.9, 16816.7, 18386.3, 4492.83, 9675.85,
+      6665.52, 14808.6, 9342.3,  7483.28, 19261.7, 4145.98, 1622.18, 13475.2,
+      7166.32, 6856.61, 21937,   7263.14, 9569.07, 14919,   8413.32, 7551.89,
+      7848.65, 6011.27, 13080.6, 15865.2, 12656,   17459.6, 4263.93, 4503.03,
+      9311.79, 21095.8, 12657.9, 13906.6, 19267.2, 11338.1, 16828.9, 11501.6,
+      11405,   15031.4, 14541.6, 19765.5, 18346.3, 19350.2, 3157.47, 18095.8,
+      1743.68, 21328.2, 19727.5, 7295.16, 10332.4, 11055.5, 20107.4, 14708.4,
+      12416.2, 16434,   2454.69, 9840.8,  6867.23, 1615.75, 6059.9,  8394.19};
+  for (int band = 0; band < x->NumBands(); ++band) {
+    for (int channel = 0; channel < x->NumChannels(); ++channel) {
+      RTC_DCHECK_GE(kBlockSize, frame.size());
+      std::copy(frame.begin(), frame.end(), x->begin(band, channel));
+    }
+  }
+}
+
+class TestInputs {
+ public:
+  TestInputs(const EchoCanceller3Config& cfg,
+             size_t num_render_channels,
+             size_t num_capture_channels);
+  ~TestInputs();
+  const RenderBuffer& GetRenderBuffer() { return *render_buffer_; }
+  rtc::ArrayView<const float, kFftLengthBy2Plus1> GetX2() { return X2_; }
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> GetY2() const {
+    return Y2_;
+  }
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> GetE2() const {
+    return E2_;
+  }
+  rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+  GetH2() const {
+    return H2_;
+  }
+  const std::vector<bool>& GetConvergedFilters() const {
+    return converged_filters_;
+  }
+  void Update();
+
+ private:
+  void UpdateCurrentPowerSpectra();
+  int n_ = 0;
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer_;
+  RenderBuffer* render_buffer_;
+  std::array<float, kFftLengthBy2Plus1> X2_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2_;
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> H2_;
+  Block x_;
+  std::vector<bool> converged_filters_;
+};
+
+TestInputs::TestInputs(const EchoCanceller3Config& cfg,
+                       size_t num_render_channels,
+                       size_t num_capture_channels)
+    : render_delay_buffer_(
+          RenderDelayBuffer::Create(cfg, 16000, num_render_channels)),
+      Y2_(num_capture_channels),
+      E2_(num_capture_channels),
+      H2_(num_capture_channels,
+          std::vector<std::array<float, kFftLengthBy2Plus1>>(
+              cfg.filter.refined.length_blocks)),
+      x_(1, num_render_channels),
+      converged_filters_(num_capture_channels, true) {
+  render_delay_buffer_->AlignFromDelay(4);
+  render_buffer_ = render_delay_buffer_->GetRenderBuffer();
+  for (auto& H2_ch : H2_) {
+    for (auto& H2_p : H2_ch) {
+      H2_p.fill(0.f);
+    }
+  }
+  for (auto& H2_p : H2_[0]) {
+    H2_p.fill(1.f);
+  }
+}
+
+TestInputs::~TestInputs() = default;
+
+void TestInputs::Update() {
+  if (n_ % 2 == 0) {
+    std::fill(x_.begin(/*band=*/0, /*channel=*/0),
+              x_.end(/*band=*/0, /*channel=*/0), 0.f);
+  } else {
+    GetActiveFrame(&x_);
+  }
+
+  render_delay_buffer_->Insert(x_);
+  render_delay_buffer_->PrepareCaptureProcessing();
+  UpdateCurrentPowerSpectra();
+  ++n_;
+}
+
+void TestInputs::UpdateCurrentPowerSpectra() {
+  const SpectrumBuffer& spectrum_render_buffer =
+      render_buffer_->GetSpectrumBuffer();
+  size_t idx = render_buffer_->Position();
+  size_t prev_idx = spectrum_render_buffer.OffsetIndex(idx, 1);
+  auto& X2 = spectrum_render_buffer.buffer[idx][/*channel=*/0];
+  auto& X2_prev = spectrum_render_buffer.buffer[prev_idx][/*channel=*/0];
+  std::copy(X2.begin(), X2.end(), X2_.begin());
+  for (size_t ch = 0; ch < Y2_.size(); ++ch) {
+    RTC_DCHECK_EQ(X2.size(), Y2_[ch].size());
+    for (size_t k = 0; k < X2.size(); ++k) {
+      E2_[ch][k] = 0.01f * X2_prev[k];
+      Y2_[ch][k] = X2[k] + E2_[ch][k];
+    }
+  }
+}
+
+}  // namespace
+
+class SignalDependentErleEstimatorMultiChannel
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         SignalDependentErleEstimatorMultiChannel,
+                         ::testing::Combine(::testing::Values(1, 2, 4),
+                                            ::testing::Values(1, 2, 4)));
+
+TEST_P(SignalDependentErleEstimatorMultiChannel, SweepSettings) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+  EchoCanceller3Config cfg;
+  size_t max_length_blocks = 50;
+  for (size_t blocks = 1; blocks < max_length_blocks; blocks = blocks + 10) {
+    for (size_t delay_headroom = 0; delay_headroom < 5; ++delay_headroom) {
+      for (size_t num_sections = 2; num_sections < max_length_blocks;
+           ++num_sections) {
+        cfg.filter.refined.length_blocks = blocks;
+        cfg.filter.refined_initial.length_blocks =
+            std::min(cfg.filter.refined_initial.length_blocks, blocks);
+        cfg.delay.delay_headroom_samples = delay_headroom * kBlockSize;
+        cfg.erle.num_sections = num_sections;
+        if (EchoCanceller3Config::Validate(&cfg)) {
+          SignalDependentErleEstimator s(cfg, num_capture_channels);
+          std::vector<std::array<float, kFftLengthBy2Plus1>> average_erle(
+              num_capture_channels);
+          for (auto& e : average_erle) {
+            e.fill(cfg.erle.max_l);
+          }
+          TestInputs inputs(cfg, num_render_channels, num_capture_channels);
+          for (size_t n = 0; n < 10; ++n) {
+            inputs.Update();
+            s.Update(inputs.GetRenderBuffer(), inputs.GetH2(), inputs.GetX2(),
+                     inputs.GetY2(), inputs.GetE2(), average_erle, average_erle,
+                     inputs.GetConvergedFilters());
+          }
+        }
+      }
+    }
+  }
+}
+
+TEST_P(SignalDependentErleEstimatorMultiChannel, LongerRun) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+  EchoCanceller3Config cfg;
+  cfg.filter.refined.length_blocks = 2;
+  cfg.filter.refined_initial.length_blocks = 1;
+  cfg.delay.delay_headroom_samples = 0;
+  cfg.delay.hysteresis_limit_blocks = 0;
+  cfg.erle.num_sections = 2;
+  EXPECT_EQ(EchoCanceller3Config::Validate(&cfg), true);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> average_erle(
+      num_capture_channels);
+  for (auto& e : average_erle) {
+    e.fill(cfg.erle.max_l);
+  }
+  SignalDependentErleEstimator s(cfg, num_capture_channels);
+  TestInputs inputs(cfg, num_render_channels, num_capture_channels);
+  for (size_t n = 0; n < 200; ++n) {
+    inputs.Update();
+    s.Update(inputs.GetRenderBuffer(), inputs.GetH2(), inputs.GetX2(),
+             inputs.GetY2(), inputs.GetE2(), average_erle, average_erle,
+             inputs.GetConvergedFilters());
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/spectrum_buffer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/spectrum_buffer.cc
new file mode 100644
index 0000000000..fe32ece09c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/spectrum_buffer.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/spectrum_buffer.h"
+
+#include <algorithm>
+
+namespace webrtc {
+
+SpectrumBuffer::SpectrumBuffer(size_t size, size_t num_channels)
+    : size(static_cast<int>(size)),
+      buffer(size,
+             std::vector<std::array<float, kFftLengthBy2Plus1>>(num_channels)) {
+  for (auto& channel : buffer) {
+    for (auto& c : channel) {
+      std::fill(c.begin(), c.end(), 0.f);
+    }
+  }
+}
+
+SpectrumBuffer::~SpectrumBuffer() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/spectrum_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/spectrum_buffer.h
new file mode 100644
index 0000000000..51e1317f55
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/spectrum_buffer.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SPECTRUM_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SPECTRUM_BUFFER_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Struct for bundling a circular buffer of one dimensional vector objects
+// together with the read and write indices.
+struct SpectrumBuffer {
+  SpectrumBuffer(size_t size, size_t num_channels);
+  ~SpectrumBuffer();
+
+  int IncIndex(int index) const {
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return index < size - 1 ? index + 1 : 0;
+  }
+
+  int DecIndex(int index) const {
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    return index > 0 ? index - 1 : size - 1;
+  }
+
+  int OffsetIndex(int index, int offset) const {
+    RTC_DCHECK_GE(size, offset);
+    RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
+    RTC_DCHECK_GE(size + index + offset, 0);
+    return (size + index + offset) % size;
+  }
+
+  void UpdateWriteIndex(int offset) { write = OffsetIndex(write, offset); }
+  void IncWriteIndex() { write = IncIndex(write); }
+  void DecWriteIndex() { write = DecIndex(write); }
+  void UpdateReadIndex(int offset) { read = OffsetIndex(read, offset); }
+  void IncReadIndex() { read = IncIndex(read); }
+  void DecReadIndex() { read = DecIndex(read); }
+
+  const int size;
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> buffer;
+  int write = 0;
+  int read = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SPECTRUM_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/stationarity_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/stationarity_estimator.cc
new file mode 100644
index 0000000000..4d364041b3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/stationarity_estimator.cc
@@ -0,0 +1,241 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/stationarity_estimator.h"
+
+#include <algorithm>
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/spectrum_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+namespace {
+constexpr float kMinNoisePower = 10.f;
+constexpr int kHangoverBlocks = kNumBlocksPerSecond / 20;
+constexpr int kNBlocksAverageInitPhase = 20;
+constexpr int kNBlocksInitialPhase = kNumBlocksPerSecond * 2.;
+}  // namespace
+
+StationarityEstimator::StationarityEstimator()
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)) {
+  Reset();
+}
+
+StationarityEstimator::~StationarityEstimator() = default;
+
+void StationarityEstimator::Reset() {
+  noise_.Reset();
+  hangovers_.fill(0);
+  stationarity_flags_.fill(false);
+}
+
+// Update just the noise estimator. Usefull until the delay is known
+void StationarityEstimator::UpdateNoiseEstimator(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> spectrum) {
+  noise_.Update(spectrum);
+  data_dumper_->DumpRaw("aec3_stationarity_noise_spectrum", noise_.Spectrum());
+  data_dumper_->DumpRaw("aec3_stationarity_is_block_stationary",
+                        IsBlockStationary());
+}
+
+void StationarityEstimator::UpdateStationarityFlags(
+    const SpectrumBuffer& spectrum_buffer,
+    rtc::ArrayView<const float> render_reverb_contribution_spectrum,
+    int idx_current,
+    int num_lookahead) {
+  std::array<int, kWindowLength> indexes;
+  int num_lookahead_bounded = std::min(num_lookahead, kWindowLength - 1);
+  int idx = idx_current;
+
+  if (num_lookahead_bounded < kWindowLength - 1) {
+    int num_lookback = (kWindowLength - 1) - num_lookahead_bounded;
+    idx = spectrum_buffer.OffsetIndex(idx_current, num_lookback);
+  }
+  // For estimating the stationarity properties of the current frame, the
+  // power for each band is accumulated for several consecutive spectra in the
+  // method EstimateBandStationarity.
+  // In order to avoid getting the indexes of the spectra for every band with
+  // its associated overhead, those indexes are stored in an array and then use
+  // when the estimation is done.
+  indexes[0] = idx;
+  for (size_t k = 1; k < indexes.size(); ++k) {
+    indexes[k] = spectrum_buffer.DecIndex(indexes[k - 1]);
+  }
+  RTC_DCHECK_EQ(
+      spectrum_buffer.DecIndex(indexes[kWindowLength - 1]),
+      spectrum_buffer.OffsetIndex(idx_current, -(num_lookahead_bounded + 1)));
+
+  for (size_t k = 0; k < stationarity_flags_.size(); ++k) {
+    stationarity_flags_[k] = EstimateBandStationarity(
+        spectrum_buffer, render_reverb_contribution_spectrum, indexes, k);
+  }
+  UpdateHangover();
+  SmoothStationaryPerFreq();
+}
+
+bool StationarityEstimator::IsBlockStationary() const {
+  float acum_stationarity = 0.f;
+  RTC_DCHECK_EQ(stationarity_flags_.size(), kFftLengthBy2Plus1);
+  for (size_t band = 0; band < stationarity_flags_.size(); ++band) {
+    bool st = IsBandStationary(band);
+    acum_stationarity += static_cast<float>(st);
+  }
+  return ((acum_stationarity * (1.f / kFftLengthBy2Plus1)) > 0.75f);
+}
+
+bool StationarityEstimator::EstimateBandStationarity(
+    const SpectrumBuffer& spectrum_buffer,
+    rtc::ArrayView<const float> average_reverb,
+    const std::array<int, kWindowLength>& indexes,
+    size_t band) const {
+  constexpr float kThrStationarity = 10.f;
+  float acum_power = 0.f;
+  const int num_render_channels =
+      static_cast<int>(spectrum_buffer.buffer[0].size());
+  const float one_by_num_channels = 1.f / num_render_channels;
+  for (auto idx : indexes) {
+    for (int ch = 0; ch < num_render_channels; ++ch) {
+      acum_power += spectrum_buffer.buffer[idx][ch][band] * one_by_num_channels;
+    }
+  }
+  acum_power += average_reverb[band];
+  float noise = kWindowLength * GetStationarityPowerBand(band);
+  RTC_CHECK_LT(0.f, noise);
+  bool stationary = acum_power < kThrStationarity * noise;
+  data_dumper_->DumpRaw("aec3_stationarity_long_ratio", acum_power / noise);
+  return stationary;
+}
+
+bool StationarityEstimator::AreAllBandsStationary() {
+  for (auto b : stationarity_flags_) {
+    if (!b)
+      return false;
+  }
+  return true;
+}
+
+void StationarityEstimator::UpdateHangover() {
+  bool reduce_hangover = AreAllBandsStationary();
+  for (size_t k = 0; k < stationarity_flags_.size(); ++k) {
+    if (!stationarity_flags_[k]) {
+      hangovers_[k] = kHangoverBlocks;
+    } else if (reduce_hangover) {
+      hangovers_[k] = std::max(hangovers_[k] - 1, 0);
+    }
+  }
+}
+
+void StationarityEstimator::SmoothStationaryPerFreq() {
+  std::array<bool, kFftLengthBy2Plus1> all_ahead_stationary_smooth;
+  for (size_t k = 1; k < kFftLengthBy2Plus1 - 1; ++k) {
+    all_ahead_stationary_smooth[k] = stationarity_flags_[k - 1] &&
+                                     stationarity_flags_[k] &&
+                                     stationarity_flags_[k + 1];
+  }
+
+  all_ahead_stationary_smooth[0] = all_ahead_stationary_smooth[1];
+  all_ahead_stationary_smooth[kFftLengthBy2Plus1 - 1] =
+      all_ahead_stationary_smooth[kFftLengthBy2Plus1 - 2];
+
+  stationarity_flags_ = all_ahead_stationary_smooth;
+}
+
+std::atomic<int> StationarityEstimator::instance_count_(0);
+
+StationarityEstimator::NoiseSpectrum::NoiseSpectrum() {
+  Reset();
+}
+
+StationarityEstimator::NoiseSpectrum::~NoiseSpectrum() = default;
+
+void StationarityEstimator::NoiseSpectrum::Reset() {
+  block_counter_ = 0;
+  noise_spectrum_.fill(kMinNoisePower);
+}
+
+void StationarityEstimator::NoiseSpectrum::Update(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> spectrum) {
+  RTC_DCHECK_LE(1, spectrum[0].size());
+  const int num_render_channels = static_cast<int>(spectrum.size());
+
+  std::array<float, kFftLengthBy2Plus1> avg_spectrum_data;
+  rtc::ArrayView<const float> avg_spectrum;
+  if (num_render_channels == 1) {
+    avg_spectrum = spectrum[0];
+  } else {
+    // For multiple channels, average the channel spectra before passing to the
+    // noise spectrum estimator.
+    avg_spectrum = avg_spectrum_data;
+    std::copy(spectrum[0].begin(), spectrum[0].end(),
+              avg_spectrum_data.begin());
+    for (int ch = 1; ch < num_render_channels; ++ch) {
+      for (size_t k = 1; k < kFftLengthBy2Plus1; ++k) {
+        avg_spectrum_data[k] += spectrum[ch][k];
+      }
+    }
+
+    const float one_by_num_channels = 1.f / num_render_channels;
+    for (size_t k = 1; k < kFftLengthBy2Plus1; ++k) {
+      avg_spectrum_data[k] *= one_by_num_channels;
+    }
+  }
+
+  ++block_counter_;
+  float alpha = GetAlpha();
+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
+    if (block_counter_ <= kNBlocksAverageInitPhase) {
+      noise_spectrum_[k] += (1.f / kNBlocksAverageInitPhase) * avg_spectrum[k];
+    } else {
+      noise_spectrum_[k] =
+          UpdateBandBySmoothing(avg_spectrum[k], noise_spectrum_[k], alpha);
+    }
+  }
+}
+
+float StationarityEstimator::NoiseSpectrum::GetAlpha() const {
+  constexpr float kAlpha = 0.004f;
+  constexpr float kAlphaInit = 0.04f;
+  constexpr float kTiltAlpha = (kAlphaInit - kAlpha) / kNBlocksInitialPhase;
+
+  if (block_counter_ > (kNBlocksInitialPhase + kNBlocksAverageInitPhase)) {
+    return kAlpha;
+  } else {
+    return kAlphaInit -
+           kTiltAlpha * (block_counter_ - kNBlocksAverageInitPhase);
+  }
+}
+
+float StationarityEstimator::NoiseSpectrum::UpdateBandBySmoothing(
+    float power_band,
+    float power_band_noise,
+    float alpha) const {
+  float power_band_noise_updated = power_band_noise;
+  if (power_band_noise < power_band) {
+    RTC_DCHECK_GT(power_band, 0.f);
+    float alpha_inc = alpha * (power_band_noise / power_band);
+    if (block_counter_ > kNBlocksInitialPhase) {
+      if (10.f * power_band_noise < power_band) {
+        alpha_inc *= 0.1f;
+      }
+    }
+    power_band_noise_updated += alpha_inc * (power_band - power_band_noise);
+  } else {
+    power_band_noise_updated += alpha * (power_band - power_band_noise);
+    power_band_noise_updated =
+        std::max(power_band_noise_updated, kMinNoisePower);
+  }
+  return power_band_noise_updated;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/stationarity_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/stationarity_estimator.h
new file mode 100644
index 0000000000..8bcd3b789e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/stationarity_estimator.h
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_STATIONARITY_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_STATIONARITY_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <atomic>
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"  // kFftLengthBy2Plus1...
+#include "modules/audio_processing/aec3/reverb_model.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+struct SpectrumBuffer;
+
+class StationarityEstimator {
+ public:
+  StationarityEstimator();
+  ~StationarityEstimator();
+
+  // Reset the stationarity estimator.
+  void Reset();
+
+  // Update just the noise estimator. Usefull until the delay is known
+  void UpdateNoiseEstimator(
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> spectrum);
+
+  // Update the flag indicating whether this current frame is stationary. For
+  // getting a more robust estimation, it looks at future and/or past frames.
+  void UpdateStationarityFlags(
+      const SpectrumBuffer& spectrum_buffer,
+      rtc::ArrayView<const float> render_reverb_contribution_spectrum,
+      int idx_current,
+      int num_lookahead);
+
+  // Returns true if the current band is stationary.
+  bool IsBandStationary(size_t band) const {
+    return stationarity_flags_[band] && (hangovers_[band] == 0);
+  }
+
+  // Returns true if the current block is estimated as stationary.
+  bool IsBlockStationary() const;
+
+ private:
+  static constexpr int kWindowLength = 13;
+  // Returns the power of the stationary noise spectrum at a band.
+  float GetStationarityPowerBand(size_t k) const { return noise_.Power(k); }
+
+  // Get an estimation of the stationarity for the current band by looking
+  // at the past/present/future available data.
+  bool EstimateBandStationarity(const SpectrumBuffer& spectrum_buffer,
+                                rtc::ArrayView<const float> average_reverb,
+                                const std::array<int, kWindowLength>& indexes,
+                                size_t band) const;
+
+  // True if all bands at the current point are stationary.
+  bool AreAllBandsStationary();
+
+  // Update the hangover depending on the stationary status of the current
+  // frame.
+  void UpdateHangover();
+
+  // Smooth the stationarity detection by looking at neighbouring frequency
+  // bands.
+  void SmoothStationaryPerFreq();
+
+  class NoiseSpectrum {
+   public:
+    NoiseSpectrum();
+    ~NoiseSpectrum();
+
+    // Reset the noise power spectrum estimate state.
+    void Reset();
+
+    // Update the noise power spectrum with a new frame.
+    void Update(
+        rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> spectrum);
+
+    // Get the noise estimation power spectrum.
+    rtc::ArrayView<const float> Spectrum() const { return noise_spectrum_; }
+
+    // Get the noise power spectrum at a certain band.
+    float Power(size_t band) const {
+      RTC_DCHECK_LT(band, noise_spectrum_.size());
+      return noise_spectrum_[band];
+    }
+
+   private:
+    // Get the update coefficient to be used for the current frame.
+    float GetAlpha() const;
+
+    // Update the noise power spectrum at a certain band with a new frame.
+    float UpdateBandBySmoothing(float power_band,
+                                float power_band_noise,
+                                float alpha) const;
+    std::array<float, kFftLengthBy2Plus1> noise_spectrum_;
+    size_t block_counter_;
+  };
+
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  NoiseSpectrum noise_;
+  std::array<int, kFftLengthBy2Plus1> hangovers_;
+  std::array<bool, kFftLengthBy2Plus1> stationarity_flags_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_STATIONARITY_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subband_erle_estimator.cc b/third_party/libwebrtc/modules/audio_processing/aec3/subband_erle_estimator.cc
new file mode 100644
index 0000000000..dc7f92fd99
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subband_erle_estimator.cc
@@ -0,0 +1,251 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/subband_erle_estimator.h"
+
+#include <algorithm>
+#include <functional>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr float kX2BandEnergyThreshold = 44015068.0f;
+constexpr int kBlocksToHoldErle = 100;
+constexpr int kBlocksForOnsetDetection = kBlocksToHoldErle + 150;
+constexpr int kPointsToAccumulate = 6;
+
+std::array<float, kFftLengthBy2Plus1> SetMaxErleBands(float max_erle_l,
+                                                      float max_erle_h) {
+  std::array<float, kFftLengthBy2Plus1> max_erle;
+  std::fill(max_erle.begin(), max_erle.begin() + kFftLengthBy2 / 2, max_erle_l);
+  std::fill(max_erle.begin() + kFftLengthBy2 / 2, max_erle.end(), max_erle_h);
+  return max_erle;
+}
+
+bool EnableMinErleDuringOnsets() {
+  return !field_trial::IsEnabled("WebRTC-Aec3MinErleDuringOnsetsKillSwitch");
+}
+
+}  // namespace
+
+SubbandErleEstimator::SubbandErleEstimator(const EchoCanceller3Config& config,
+                                           size_t num_capture_channels)
+    : use_onset_detection_(config.erle.onset_detection),
+      min_erle_(config.erle.min),
+      max_erle_(SetMaxErleBands(config.erle.max_l, config.erle.max_h)),
+      use_min_erle_during_onsets_(EnableMinErleDuringOnsets()),
+      accum_spectra_(num_capture_channels),
+      erle_(num_capture_channels),
+      erle_onset_compensated_(num_capture_channels),
+      erle_unbounded_(num_capture_channels),
+      erle_during_onsets_(num_capture_channels),
+      coming_onset_(num_capture_channels),
+      hold_counters_(num_capture_channels) {
+  Reset();
+}
+
+SubbandErleEstimator::~SubbandErleEstimator() = default;
+
+void SubbandErleEstimator::Reset() {
+  const size_t num_capture_channels = erle_.size();
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    erle_[ch].fill(min_erle_);
+    erle_onset_compensated_[ch].fill(min_erle_);
+    erle_unbounded_[ch].fill(min_erle_);
+    erle_during_onsets_[ch].fill(min_erle_);
+    coming_onset_[ch].fill(true);
+    hold_counters_[ch].fill(0);
+  }
+  ResetAccumulatedSpectra();
+}
+
+void SubbandErleEstimator::Update(
+    rtc::ArrayView<const float, kFftLengthBy2Plus1> X2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    const std::vector<bool>& converged_filters) {
+  UpdateAccumulatedSpectra(X2, Y2, E2, converged_filters);
+  UpdateBands(converged_filters);
+
+  if (use_onset_detection_) {
+    DecreaseErlePerBandForLowRenderSignals();
+  }
+
+  const size_t num_capture_channels = erle_.size();
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    auto& erle = erle_[ch];
+    erle[0] = erle[1];
+    erle[kFftLengthBy2] = erle[kFftLengthBy2 - 1];
+
+    auto& erle_oc = erle_onset_compensated_[ch];
+    erle_oc[0] = erle_oc[1];
+    erle_oc[kFftLengthBy2] = erle_oc[kFftLengthBy2 - 1];
+
+    auto& erle_u = erle_unbounded_[ch];
+    erle_u[0] = erle_u[1];
+    erle_u[kFftLengthBy2] = erle_u[kFftLengthBy2 - 1];
+  }
+}
+
+void SubbandErleEstimator::Dump(
+    const std::unique_ptr<ApmDataDumper>& data_dumper) const {
+  data_dumper->DumpRaw("aec3_erle_onset", ErleDuringOnsets()[0]);
+}
+
+void SubbandErleEstimator::UpdateBands(
+    const std::vector<bool>& converged_filters) {
+  const int num_capture_channels = static_cast<int>(accum_spectra_.Y2.size());
+  for (int ch = 0; ch < num_capture_channels; ++ch) {
+    // Note that the use of the converged_filter flag already imposed
+    // a minimum of the erle that can be estimated as that flag would
+    // be false if the filter is performing poorly.
+    if (!converged_filters[ch]) {
+      continue;
+    }
+
+    if (accum_spectra_.num_points[ch] != kPointsToAccumulate) {
+      continue;
+    }
+
+    std::array<float, kFftLengthBy2> new_erle;
+    std::array<bool, kFftLengthBy2> is_erle_updated;
+    is_erle_updated.fill(false);
+
+    for (size_t k = 1; k < kFftLengthBy2; ++k) {
+      if (accum_spectra_.E2[ch][k] > 0.f) {
+        new_erle[k] = accum_spectra_.Y2[ch][k] / accum_spectra_.E2[ch][k];
+        is_erle_updated[k] = true;
+      }
+    }
+
+    if (use_onset_detection_) {
+      for (size_t k = 1; k < kFftLengthBy2; ++k) {
+        if (is_erle_updated[k] && !accum_spectra_.low_render_energy[ch][k]) {
+          if (coming_onset_[ch][k]) {
+            coming_onset_[ch][k] = false;
+            if (!use_min_erle_during_onsets_) {
+              float alpha =
+                  new_erle[k] < erle_during_onsets_[ch][k] ? 0.3f : 0.15f;
+              erle_during_onsets_[ch][k] = rtc::SafeClamp(
+                  erle_during_onsets_[ch][k] +
+                      alpha * (new_erle[k] - erle_during_onsets_[ch][k]),
+                  min_erle_, max_erle_[k]);
+            }
+          }
+          hold_counters_[ch][k] = kBlocksForOnsetDetection;
+        }
+      }
+    }
+
+    auto update_erle_band = [](float& erle, float new_erle,
+                               bool low_render_energy, float min_erle,
+                               float max_erle) {
+      float alpha = 0.05f;
+      if (new_erle < erle) {
+        alpha = low_render_energy ? 0.f : 0.1f;
+      }
+      erle =
+          rtc::SafeClamp(erle + alpha * (new_erle - erle), min_erle, max_erle);
+    };
+
+    for (size_t k = 1; k < kFftLengthBy2; ++k) {
+      if (is_erle_updated[k]) {
+        const bool low_render_energy = accum_spectra_.low_render_energy[ch][k];
+        update_erle_band(erle_[ch][k], new_erle[k], low_render_energy,
+                         min_erle_, max_erle_[k]);
+        if (use_onset_detection_) {
+          update_erle_band(erle_onset_compensated_[ch][k], new_erle[k],
+                           low_render_energy, min_erle_, max_erle_[k]);
+        }
+
+        // Virtually unbounded ERLE.
+        constexpr float kUnboundedErleMax = 100000.0f;
+        update_erle_band(erle_unbounded_[ch][k], new_erle[k], low_render_energy,
+                         min_erle_, kUnboundedErleMax);
+      }
+    }
+  }
+}
+
+void SubbandErleEstimator::DecreaseErlePerBandForLowRenderSignals() {
+  const int num_capture_channels = static_cast<int>(accum_spectra_.Y2.size());
+  for (int ch = 0; ch < num_capture_channels; ++ch) {
+    for (size_t k = 1; k < kFftLengthBy2; ++k) {
+      --hold_counters_[ch][k];
+      if (hold_counters_[ch][k] <=
+          (kBlocksForOnsetDetection - kBlocksToHoldErle)) {
+        if (erle_onset_compensated_[ch][k] > erle_during_onsets_[ch][k]) {
+          erle_onset_compensated_[ch][k] =
+              std::max(erle_during_onsets_[ch][k],
+                       0.97f * erle_onset_compensated_[ch][k]);
+          RTC_DCHECK_LE(min_erle_, erle_onset_compensated_[ch][k]);
+        }
+        if (hold_counters_[ch][k] <= 0) {
+          coming_onset_[ch][k] = true;
+          hold_counters_[ch][k] = 0;
+        }
+      }
+    }
+  }
+}
+
+void SubbandErleEstimator::ResetAccumulatedSpectra() {
+  for (size_t ch = 0; ch < erle_during_onsets_.size(); ++ch) {
+    accum_spectra_.Y2[ch].fill(0.f);
+    accum_spectra_.E2[ch].fill(0.f);
+    accum_spectra_.num_points[ch] = 0;
+    accum_spectra_.low_render_energy[ch].fill(false);
+  }
+}
+
+void SubbandErleEstimator::UpdateAccumulatedSpectra(
+    rtc::ArrayView<const float, kFftLengthBy2Plus1> X2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+    const std::vector<bool>& converged_filters) {
+  auto& st = accum_spectra_;
+  RTC_DCHECK_EQ(st.E2.size(), E2.size());
+  RTC_DCHECK_EQ(st.E2.size(), E2.size());
+  const int num_capture_channels = static_cast<int>(Y2.size());
+  for (int ch = 0; ch < num_capture_channels; ++ch) {
+    // Note that the use of the converged_filter flag already imposed
+    // a minimum of the erle that can be estimated as that flag would
+    // be false if the filter is performing poorly.
+    if (!converged_filters[ch]) {
+      continue;
+    }
+
+    if (st.num_points[ch] == kPointsToAccumulate) {
+      st.num_points[ch] = 0;
+      st.Y2[ch].fill(0.f);
+      st.E2[ch].fill(0.f);
+      st.low_render_energy[ch].fill(false);
+    }
+
+    std::transform(Y2[ch].begin(), Y2[ch].end(), st.Y2[ch].begin(),
+                   st.Y2[ch].begin(), std::plus<float>());
+    std::transform(E2[ch].begin(), E2[ch].end(), st.E2[ch].begin(),
+                   st.E2[ch].begin(), std::plus<float>());
+
+    for (size_t k = 0; k < X2.size(); ++k) {
+      st.low_render_energy[ch][k] =
+          st.low_render_energy[ch][k] || X2[k] < kX2BandEnergyThreshold;
+    }
+
+    ++st.num_points[ch];
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subband_erle_estimator.h b/third_party/libwebrtc/modules/audio_processing/aec3/subband_erle_estimator.h
new file mode 100644
index 0000000000..8bf9c4d645
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subband_erle_estimator.h
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBBAND_ERLE_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SUBBAND_ERLE_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+// Estimates the echo return loss enhancement for each frequency subband.
+class SubbandErleEstimator {
+ public:
+  SubbandErleEstimator(const EchoCanceller3Config& config,
+                       size_t num_capture_channels);
+  ~SubbandErleEstimator();
+
+  // Resets the ERLE estimator.
+  void Reset();
+
+  // Updates the ERLE estimate.
+  void Update(rtc::ArrayView<const float, kFftLengthBy2Plus1> X2,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+              const std::vector<bool>& converged_filters);
+
+  // Returns the ERLE estimate.
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Erle(
+      bool onset_compensated) const {
+    return onset_compensated && use_onset_detection_ ? erle_onset_compensated_
+                                                     : erle_;
+  }
+
+  // Returns the non-capped ERLE estimate.
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleUnbounded()
+      const {
+    return erle_unbounded_;
+  }
+
+  // Returns the ERLE estimate at onsets (only used for testing).
+  rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> ErleDuringOnsets()
+      const {
+    return erle_during_onsets_;
+  }
+
+  void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper) const;
+
+ private:
+  struct AccumulatedSpectra {
+    explicit AccumulatedSpectra(size_t num_capture_channels)
+        : Y2(num_capture_channels),
+          E2(num_capture_channels),
+          low_render_energy(num_capture_channels),
+          num_points(num_capture_channels) {}
+    std::vector<std::array<float, kFftLengthBy2Plus1>> Y2;
+    std::vector<std::array<float, kFftLengthBy2Plus1>> E2;
+    std::vector<std::array<bool, kFftLengthBy2Plus1>> low_render_energy;
+    std::vector<int> num_points;
+  };
+
+  void UpdateAccumulatedSpectra(
+      rtc::ArrayView<const float, kFftLengthBy2Plus1> X2,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> Y2,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> E2,
+      const std::vector<bool>& converged_filters);
+
+  void ResetAccumulatedSpectra();
+
+  void UpdateBands(const std::vector<bool>& converged_filters);
+  void DecreaseErlePerBandForLowRenderSignals();
+
+  const bool use_onset_detection_;
+  const float min_erle_;
+  const std::array<float, kFftLengthBy2Plus1> max_erle_;
+  const bool use_min_erle_during_onsets_;
+  AccumulatedSpectra accum_spectra_;
+  // ERLE without special handling of render onsets.
+  std::vector<std::array<float, kFftLengthBy2Plus1>> erle_;
+  // ERLE lowered during render onsets.
+  std::vector<std::array<float, kFftLengthBy2Plus1>> erle_onset_compensated_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> erle_unbounded_;
+  // Estimation of ERLE during render onsets.
+  std::vector<std::array<float, kFftLengthBy2Plus1>> erle_during_onsets_;
+  std::vector<std::array<bool, kFftLengthBy2Plus1>> coming_onset_;
+  std::vector<std::array<int, kFftLengthBy2Plus1>> hold_counters_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUBBAND_ERLE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subband_nearend_detector.cc b/third_party/libwebrtc/modules/audio_processing/aec3/subband_nearend_detector.cc
new file mode 100644
index 0000000000..2aa400c3af
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subband_nearend_detector.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/subband_nearend_detector.h"
+
+#include <numeric>
+
+namespace webrtc {
+SubbandNearendDetector::SubbandNearendDetector(
+    const EchoCanceller3Config::Suppressor::SubbandNearendDetection& config,
+    size_t num_capture_channels)
+    : config_(config),
+      num_capture_channels_(num_capture_channels),
+      nearend_smoothers_(num_capture_channels_,
+                         aec3::MovingAverage(kFftLengthBy2Plus1,
+                                             config_.nearend_average_blocks)),
+      one_over_subband_length1_(
+          1.f / (config_.subband1.high - config_.subband1.low + 1)),
+      one_over_subband_length2_(
+          1.f / (config_.subband2.high - config_.subband2.low + 1)) {}
+
+void SubbandNearendDetector::Update(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        nearend_spectrum,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        residual_echo_spectrum,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        comfort_noise_spectrum,
+    bool initial_state) {
+  nearend_state_ = false;
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    const std::array<float, kFftLengthBy2Plus1>& noise =
+        comfort_noise_spectrum[ch];
+    std::array<float, kFftLengthBy2Plus1> nearend;
+    nearend_smoothers_[ch].Average(nearend_spectrum[ch], nearend);
+
+    // Noise power of the first region.
+    float noise_power =
+        std::accumulate(noise.begin() + config_.subband1.low,
+                        noise.begin() + config_.subband1.high + 1, 0.f) *
+        one_over_subband_length1_;
+
+    // Nearend power of the first region.
+    float nearend_power_subband1 =
+        std::accumulate(nearend.begin() + config_.subband1.low,
+                        nearend.begin() + config_.subband1.high + 1, 0.f) *
+        one_over_subband_length1_;
+
+    // Nearend power of the second region.
+    float nearend_power_subband2 =
+        std::accumulate(nearend.begin() + config_.subband2.low,
+                        nearend.begin() + config_.subband2.high + 1, 0.f) *
+        one_over_subband_length2_;
+
+    // One channel is sufficient to trigger nearend state.
+    nearend_state_ =
+        nearend_state_ ||
+        (nearend_power_subband1 <
+             config_.nearend_threshold * nearend_power_subband2 &&
+         (nearend_power_subband1 > config_.snr_threshold * noise_power));
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subband_nearend_detector.h b/third_party/libwebrtc/modules/audio_processing/aec3/subband_nearend_detector.h
new file mode 100644
index 0000000000..8357edb65f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subband_nearend_detector.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBBAND_NEAREND_DETECTOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SUBBAND_NEAREND_DETECTOR_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/moving_average.h"
+#include "modules/audio_processing/aec3/nearend_detector.h"
+
+namespace webrtc {
+// Class for selecting whether the suppressor is in the nearend or echo state.
+class SubbandNearendDetector : public NearendDetector {
+ public:
+  SubbandNearendDetector(
+      const EchoCanceller3Config::Suppressor::SubbandNearendDetection& config,
+      size_t num_capture_channels);
+
+  // Returns whether the current state is the nearend state.
+  bool IsNearendState() const override { return nearend_state_; }
+
+  // Updates the state selection based on latest spectral estimates.
+  void Update(rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+                  nearend_spectrum,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+                  residual_echo_spectrum,
+              rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+                  comfort_noise_spectrum,
+              bool initial_state) override;
+
+ private:
+  const EchoCanceller3Config::Suppressor::SubbandNearendDetection config_;
+  const size_t num_capture_channels_;
+  std::vector<aec3::MovingAverage> nearend_smoothers_;
+  const float one_over_subband_length1_;
+  const float one_over_subband_length2_;
+  bool nearend_state_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUBBAND_NEAREND_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subtractor.cc b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor.cc
new file mode 100644
index 0000000000..aa36bb272a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor.cc
@@ -0,0 +1,364 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/subtractor.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+bool UseCoarseFilterResetHangover() {
+  return !field_trial::IsEnabled(
+      "WebRTC-Aec3CoarseFilterResetHangoverKillSwitch");
+}
+
+void PredictionError(const Aec3Fft& fft,
+                     const FftData& S,
+                     rtc::ArrayView<const float> y,
+                     std::array<float, kBlockSize>* e,
+                     std::array<float, kBlockSize>* s) {
+  std::array<float, kFftLength> tmp;
+  fft.Ifft(S, &tmp);
+  constexpr float kScale = 1.0f / kFftLengthBy2;
+  std::transform(y.begin(), y.end(), tmp.begin() + kFftLengthBy2, e->begin(),
+                 [&](float a, float b) { return a - b * kScale; });
+
+  if (s) {
+    for (size_t k = 0; k < s->size(); ++k) {
+      (*s)[k] = kScale * tmp[k + kFftLengthBy2];
+    }
+  }
+}
+
+void ScaleFilterOutput(rtc::ArrayView<const float> y,
+                       float factor,
+                       rtc::ArrayView<float> e,
+                       rtc::ArrayView<float> s) {
+  RTC_DCHECK_EQ(y.size(), e.size());
+  RTC_DCHECK_EQ(y.size(), s.size());
+  for (size_t k = 0; k < y.size(); ++k) {
+    s[k] *= factor;
+    e[k] = y[k] - s[k];
+  }
+}
+
+}  // namespace
+
+Subtractor::Subtractor(const EchoCanceller3Config& config,
+                       size_t num_render_channels,
+                       size_t num_capture_channels,
+                       ApmDataDumper* data_dumper,
+                       Aec3Optimization optimization)
+    : fft_(),
+      data_dumper_(data_dumper),
+      optimization_(optimization),
+      config_(config),
+      num_capture_channels_(num_capture_channels),
+      use_coarse_filter_reset_hangover_(UseCoarseFilterResetHangover()),
+      refined_filters_(num_capture_channels_),
+      coarse_filter_(num_capture_channels_),
+      refined_gains_(num_capture_channels_),
+      coarse_gains_(num_capture_channels_),
+      filter_misadjustment_estimators_(num_capture_channels_),
+      poor_coarse_filter_counters_(num_capture_channels_, 0),
+      coarse_filter_reset_hangover_(num_capture_channels_, 0),
+      refined_frequency_responses_(
+          num_capture_channels_,
+          std::vector<std::array<float, kFftLengthBy2Plus1>>(
+              std::max(config_.filter.refined_initial.length_blocks,
+                       config_.filter.refined.length_blocks),
+              std::array<float, kFftLengthBy2Plus1>())),
+      refined_impulse_responses_(
+          num_capture_channels_,
+          std::vector<float>(GetTimeDomainLength(std::max(
+                                 config_.filter.refined_initial.length_blocks,
+                                 config_.filter.refined.length_blocks)),
+                             0.f)),
+      coarse_impulse_responses_(0) {
+  // Set up the storing of coarse impulse responses if data dumping is
+  // available.
+  if (ApmDataDumper::IsAvailable()) {
+    coarse_impulse_responses_.resize(num_capture_channels_);
+    const size_t filter_size = GetTimeDomainLength(
+        std::max(config_.filter.coarse_initial.length_blocks,
+                 config_.filter.coarse.length_blocks));
+    for (std::vector<float>& impulse_response : coarse_impulse_responses_) {
+      impulse_response.resize(filter_size, 0.f);
+    }
+  }
+
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    refined_filters_[ch] = std::make_unique<AdaptiveFirFilter>(
+        config_.filter.refined.length_blocks,
+        config_.filter.refined_initial.length_blocks,
+        config.filter.config_change_duration_blocks, num_render_channels,
+        optimization, data_dumper_);
+
+    coarse_filter_[ch] = std::make_unique<AdaptiveFirFilter>(
+        config_.filter.coarse.length_blocks,
+        config_.filter.coarse_initial.length_blocks,
+        config.filter.config_change_duration_blocks, num_render_channels,
+        optimization, data_dumper_);
+    refined_gains_[ch] = std::make_unique<RefinedFilterUpdateGain>(
+        config_.filter.refined_initial,
+        config_.filter.config_change_duration_blocks);
+    coarse_gains_[ch] = std::make_unique<CoarseFilterUpdateGain>(
+        config_.filter.coarse_initial,
+        config.filter.config_change_duration_blocks);
+  }
+
+  RTC_DCHECK(data_dumper_);
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    for (auto& H2_k : refined_frequency_responses_[ch]) {
+      H2_k.fill(0.f);
+    }
+  }
+}
+
+Subtractor::~Subtractor() = default;
+
+void Subtractor::HandleEchoPathChange(
+    const EchoPathVariability& echo_path_variability) {
+  const auto full_reset = [&]() {
+    for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+      refined_filters_[ch]->HandleEchoPathChange();
+      coarse_filter_[ch]->HandleEchoPathChange();
+      refined_gains_[ch]->HandleEchoPathChange(echo_path_variability);
+      coarse_gains_[ch]->HandleEchoPathChange();
+      refined_gains_[ch]->SetConfig(config_.filter.refined_initial, true);
+      coarse_gains_[ch]->SetConfig(config_.filter.coarse_initial, true);
+      refined_filters_[ch]->SetSizePartitions(
+          config_.filter.refined_initial.length_blocks, true);
+      coarse_filter_[ch]->SetSizePartitions(
+          config_.filter.coarse_initial.length_blocks, true);
+    }
+  };
+
+  if (echo_path_variability.delay_change !=
+      EchoPathVariability::DelayAdjustment::kNone) {
+    full_reset();
+  }
+
+  if (echo_path_variability.gain_change) {
+    for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+      refined_gains_[ch]->HandleEchoPathChange(echo_path_variability);
+    }
+  }
+}
+
+void Subtractor::ExitInitialState() {
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    refined_gains_[ch]->SetConfig(config_.filter.refined, false);
+    coarse_gains_[ch]->SetConfig(config_.filter.coarse, false);
+    refined_filters_[ch]->SetSizePartitions(
+        config_.filter.refined.length_blocks, false);
+    coarse_filter_[ch]->SetSizePartitions(config_.filter.coarse.length_blocks,
+                                          false);
+  }
+}
+
+void Subtractor::Process(const RenderBuffer& render_buffer,
+                         const Block& capture,
+                         const RenderSignalAnalyzer& render_signal_analyzer,
+                         const AecState& aec_state,
+                         rtc::ArrayView<SubtractorOutput> outputs) {
+  RTC_DCHECK_EQ(num_capture_channels_, capture.NumChannels());
+
+  // Compute the render powers.
+  const bool same_filter_sizes = refined_filters_[0]->SizePartitions() ==
+                                 coarse_filter_[0]->SizePartitions();
+  std::array<float, kFftLengthBy2Plus1> X2_refined;
+  std::array<float, kFftLengthBy2Plus1> X2_coarse_data;
+  auto& X2_coarse = same_filter_sizes ? X2_refined : X2_coarse_data;
+  if (same_filter_sizes) {
+    render_buffer.SpectralSum(refined_filters_[0]->SizePartitions(),
+                              &X2_refined);
+  } else if (refined_filters_[0]->SizePartitions() >
+             coarse_filter_[0]->SizePartitions()) {
+    render_buffer.SpectralSums(coarse_filter_[0]->SizePartitions(),
+                               refined_filters_[0]->SizePartitions(),
+                               &X2_coarse, &X2_refined);
+  } else {
+    render_buffer.SpectralSums(refined_filters_[0]->SizePartitions(),
+                               coarse_filter_[0]->SizePartitions(), &X2_refined,
+                               &X2_coarse);
+  }
+
+  // Process all capture channels
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    SubtractorOutput& output = outputs[ch];
+    rtc::ArrayView<const float> y = capture.View(/*band=*/0, ch);
+    FftData& E_refined = output.E_refined;
+    FftData E_coarse;
+    std::array<float, kBlockSize>& e_refined = output.e_refined;
+    std::array<float, kBlockSize>& e_coarse = output.e_coarse;
+
+    FftData S;
+    FftData& G = S;
+
+    // Form the outputs of the refined and coarse filters.
+    refined_filters_[ch]->Filter(render_buffer, &S);
+    PredictionError(fft_, S, y, &e_refined, &output.s_refined);
+
+    coarse_filter_[ch]->Filter(render_buffer, &S);
+    PredictionError(fft_, S, y, &e_coarse, &output.s_coarse);
+
+    // Compute the signal powers in the subtractor output.
+    output.ComputeMetrics(y);
+
+    // Adjust the filter if needed.
+    bool refined_filters_adjusted = false;
+    filter_misadjustment_estimators_[ch].Update(output);
+    if (filter_misadjustment_estimators_[ch].IsAdjustmentNeeded()) {
+      float scale = filter_misadjustment_estimators_[ch].GetMisadjustment();
+      refined_filters_[ch]->ScaleFilter(scale);
+      for (auto& h_k : refined_impulse_responses_[ch]) {
+        h_k *= scale;
+      }
+      ScaleFilterOutput(y, scale, e_refined, output.s_refined);
+      filter_misadjustment_estimators_[ch].Reset();
+      refined_filters_adjusted = true;
+    }
+
+    // Compute the FFts of the refined and coarse filter outputs.
+    fft_.ZeroPaddedFft(e_refined, Aec3Fft::Window::kHanning, &E_refined);
+    fft_.ZeroPaddedFft(e_coarse, Aec3Fft::Window::kHanning, &E_coarse);
+
+    // Compute spectra for future use.
+    E_coarse.Spectrum(optimization_, output.E2_coarse);
+    E_refined.Spectrum(optimization_, output.E2_refined);
+
+    // Update the refined filter.
+    if (!refined_filters_adjusted) {
+      // Do not allow the performance of the coarse filter to affect the
+      // adaptation speed of the refined filter just after the coarse filter has
+      // been reset.
+      const bool disallow_leakage_diverged =
+          coarse_filter_reset_hangover_[ch] > 0 &&
+          use_coarse_filter_reset_hangover_;
+
+      std::array<float, kFftLengthBy2Plus1> erl;
+      ComputeErl(optimization_, refined_frequency_responses_[ch], erl);
+      refined_gains_[ch]->Compute(X2_refined, render_signal_analyzer, output,
+                                  erl, refined_filters_[ch]->SizePartitions(),
+                                  aec_state.SaturatedCapture(),
+                                  disallow_leakage_diverged, &G);
+    } else {
+      G.re.fill(0.f);
+      G.im.fill(0.f);
+    }
+    refined_filters_[ch]->Adapt(render_buffer, G,
+                                &refined_impulse_responses_[ch]);
+    refined_filters_[ch]->ComputeFrequencyResponse(
+        &refined_frequency_responses_[ch]);
+
+    if (ch == 0) {
+      data_dumper_->DumpRaw("aec3_subtractor_G_refined", G.re);
+      data_dumper_->DumpRaw("aec3_subtractor_G_refined", G.im);
+    }
+
+    // Update the coarse filter.
+    poor_coarse_filter_counters_[ch] =
+        output.e2_refined < output.e2_coarse
+            ? poor_coarse_filter_counters_[ch] + 1
+            : 0;
+    if (poor_coarse_filter_counters_[ch] < 5) {
+      coarse_gains_[ch]->Compute(X2_coarse, render_signal_analyzer, E_coarse,
+                                 coarse_filter_[ch]->SizePartitions(),
+                                 aec_state.SaturatedCapture(), &G);
+      coarse_filter_reset_hangover_[ch] =
+          std::max(coarse_filter_reset_hangover_[ch] - 1, 0);
+    } else {
+      poor_coarse_filter_counters_[ch] = 0;
+      coarse_filter_[ch]->SetFilter(refined_filters_[ch]->SizePartitions(),
+                                    refined_filters_[ch]->GetFilter());
+      coarse_gains_[ch]->Compute(X2_coarse, render_signal_analyzer, E_refined,
+                                 coarse_filter_[ch]->SizePartitions(),
+                                 aec_state.SaturatedCapture(), &G);
+      coarse_filter_reset_hangover_[ch] =
+          config_.filter.coarse_reset_hangover_blocks;
+    }
+
+    if (ApmDataDumper::IsAvailable()) {
+      RTC_DCHECK_LT(ch, coarse_impulse_responses_.size());
+      coarse_filter_[ch]->Adapt(render_buffer, G,
+                                &coarse_impulse_responses_[ch]);
+    } else {
+      coarse_filter_[ch]->Adapt(render_buffer, G);
+    }
+
+    if (ch == 0) {
+      data_dumper_->DumpRaw("aec3_subtractor_G_coarse", G.re);
+      data_dumper_->DumpRaw("aec3_subtractor_G_coarse", G.im);
+      filter_misadjustment_estimators_[ch].Dump(data_dumper_);
+      DumpFilters();
+    }
+
+    std::for_each(e_refined.begin(), e_refined.end(),
+                  [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
+
+    if (ch == 0) {
+      data_dumper_->DumpWav("aec3_refined_filters_output", kBlockSize,
+                            &e_refined[0], 16000, 1);
+      data_dumper_->DumpWav("aec3_coarse_filter_output", kBlockSize,
+                            &e_coarse[0], 16000, 1);
+    }
+  }
+}
+
+void Subtractor::FilterMisadjustmentEstimator::Update(
+    const SubtractorOutput& output) {
+  e2_acum_ += output.e2_refined;
+  y2_acum_ += output.y2;
+  if (++n_blocks_acum_ == n_blocks_) {
+    if (y2_acum_ > n_blocks_ * 200.f * 200.f * kBlockSize) {
+      float update = (e2_acum_ / y2_acum_);
+      if (e2_acum_ > n_blocks_ * 7500.f * 7500.f * kBlockSize) {
+        // Duration equal to blockSizeMs * n_blocks_ * 4.
+        overhang_ = 4;
+      } else {
+        overhang_ = std::max(overhang_ - 1, 0);
+      }
+
+      if ((update < inv_misadjustment_) || (overhang_ > 0)) {
+        inv_misadjustment_ += 0.1f * (update - inv_misadjustment_);
+      }
+    }
+    e2_acum_ = 0.f;
+    y2_acum_ = 0.f;
+    n_blocks_acum_ = 0;
+  }
+}
+
+void Subtractor::FilterMisadjustmentEstimator::Reset() {
+  e2_acum_ = 0.f;
+  y2_acum_ = 0.f;
+  n_blocks_acum_ = 0;
+  inv_misadjustment_ = 0.f;
+  overhang_ = 0.f;
+}
+
+void Subtractor::FilterMisadjustmentEstimator::Dump(
+    ApmDataDumper* data_dumper) const {
+  data_dumper->DumpRaw("aec3_inv_misadjustment_factor", inv_misadjustment_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subtractor.h b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor.h
new file mode 100644
index 0000000000..86159a3442
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor.h
@@ -0,0 +1,150 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
+
+#include <math.h>
+#include <stddef.h>
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/block.h"
+#include "modules/audio_processing/aec3/coarse_filter_update_gain.h"
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+#include "modules/audio_processing/aec3/refined_filter_update_gain.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+#include "modules/audio_processing/aec3/subtractor_output.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Proves linear echo cancellation functionality
+class Subtractor {
+ public:
+  Subtractor(const EchoCanceller3Config& config,
+             size_t num_render_channels,
+             size_t num_capture_channels,
+             ApmDataDumper* data_dumper,
+             Aec3Optimization optimization);
+  ~Subtractor();
+  Subtractor(const Subtractor&) = delete;
+  Subtractor& operator=(const Subtractor&) = delete;
+
+  // Performs the echo subtraction.
+  void Process(const RenderBuffer& render_buffer,
+               const Block& capture,
+               const RenderSignalAnalyzer& render_signal_analyzer,
+               const AecState& aec_state,
+               rtc::ArrayView<SubtractorOutput> outputs);
+
+  void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
+
+  // Exits the initial state.
+  void ExitInitialState();
+
+  // Returns the block-wise frequency responses for the refined adaptive
+  // filters.
+  const std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>&
+  FilterFrequencyResponses() const {
+    return refined_frequency_responses_;
+  }
+
+  // Returns the estimates of the impulse responses for the refined adaptive
+  // filters.
+  const std::vector<std::vector<float>>& FilterImpulseResponses() const {
+    return refined_impulse_responses_;
+  }
+
+  void DumpFilters() {
+    data_dumper_->DumpRaw(
+        "aec3_subtractor_h_refined",
+        rtc::ArrayView<const float>(
+            refined_impulse_responses_[0].data(),
+            GetTimeDomainLength(
+                refined_filters_[0]->max_filter_size_partitions())));
+    if (ApmDataDumper::IsAvailable()) {
+      RTC_DCHECK_GT(coarse_impulse_responses_.size(), 0);
+      data_dumper_->DumpRaw(
+          "aec3_subtractor_h_coarse",
+          rtc::ArrayView<const float>(
+              coarse_impulse_responses_[0].data(),
+              GetTimeDomainLength(
+                  coarse_filter_[0]->max_filter_size_partitions())));
+    }
+
+    refined_filters_[0]->DumpFilter("aec3_subtractor_H_refined");
+    coarse_filter_[0]->DumpFilter("aec3_subtractor_H_coarse");
+  }
+
+ private:
+  class FilterMisadjustmentEstimator {
+   public:
+    FilterMisadjustmentEstimator() = default;
+    ~FilterMisadjustmentEstimator() = default;
+    // Update the misadjustment estimator.
+    void Update(const SubtractorOutput& output);
+    // GetMisadjustment() Returns a recommended scale for the filter so the
+    // prediction error energy gets closer to the energy that is seen at the
+    // microphone input.
+    float GetMisadjustment() const {
+      RTC_DCHECK_GT(inv_misadjustment_, 0.0f);
+      // It is not aiming to adjust all the estimated mismatch. Instead,
+      // it adjusts half of that estimated mismatch.
+      return 2.f / sqrtf(inv_misadjustment_);
+    }
+    // Returns true if the prediciton error energy is significantly larger
+    // than the microphone signal energy and, therefore, an adjustment is
+    // recommended.
+    bool IsAdjustmentNeeded() const { return inv_misadjustment_ > 10.f; }
+    void Reset();
+    void Dump(ApmDataDumper* data_dumper) const;
+
+   private:
+    const int n_blocks_ = 4;
+    int n_blocks_acum_ = 0;
+    float e2_acum_ = 0.f;
+    float y2_acum_ = 0.f;
+    float inv_misadjustment_ = 0.f;
+    int overhang_ = 0.f;
+  };
+
+  const Aec3Fft fft_;
+  ApmDataDumper* data_dumper_;
+  const Aec3Optimization optimization_;
+  const EchoCanceller3Config config_;
+  const size_t num_capture_channels_;
+  const bool use_coarse_filter_reset_hangover_;
+
+  std::vector<std::unique_ptr<AdaptiveFirFilter>> refined_filters_;
+  std::vector<std::unique_ptr<AdaptiveFirFilter>> coarse_filter_;
+  std::vector<std::unique_ptr<RefinedFilterUpdateGain>> refined_gains_;
+  std::vector<std::unique_ptr<CoarseFilterUpdateGain>> coarse_gains_;
+  std::vector<FilterMisadjustmentEstimator> filter_misadjustment_estimators_;
+  std::vector<size_t> poor_coarse_filter_counters_;
+  std::vector<int> coarse_filter_reset_hangover_;
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      refined_frequency_responses_;
+  std::vector<std::vector<float>> refined_impulse_responses_;
+  std::vector<std::vector<float>> coarse_impulse_responses_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output.cc b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output.cc
new file mode 100644
index 0000000000..ed80101f06
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/subtractor_output.h"
+
+#include <numeric>
+
+namespace webrtc {
+
+SubtractorOutput::SubtractorOutput() = default;
+SubtractorOutput::~SubtractorOutput() = default;
+
+void SubtractorOutput::Reset() {
+  s_refined.fill(0.f);
+  s_coarse.fill(0.f);
+  e_refined.fill(0.f);
+  e_coarse.fill(0.f);
+  E_refined.re.fill(0.f);
+  E_refined.im.fill(0.f);
+  E2_refined.fill(0.f);
+  E2_coarse.fill(0.f);
+  e2_refined = 0.f;
+  e2_coarse = 0.f;
+  s2_refined = 0.f;
+  s2_coarse = 0.f;
+  y2 = 0.f;
+}
+
+void SubtractorOutput::ComputeMetrics(rtc::ArrayView<const float> y) {
+  const auto sum_of_squares = [](float a, float b) { return a + b * b; };
+  y2 = std::accumulate(y.begin(), y.end(), 0.f, sum_of_squares);
+  e2_refined =
+      std::accumulate(e_refined.begin(), e_refined.end(), 0.f, sum_of_squares);
+  e2_coarse =
+      std::accumulate(e_coarse.begin(), e_coarse.end(), 0.f, sum_of_squares);
+  s2_refined =
+      std::accumulate(s_refined.begin(), s_refined.end(), 0.f, sum_of_squares);
+  s2_coarse =
+      std::accumulate(s_coarse.begin(), s_coarse.end(), 0.f, sum_of_squares);
+
+  s_refined_max_abs = *std::max_element(s_refined.begin(), s_refined.end());
+  s_refined_max_abs =
+      std::max(s_refined_max_abs,
+               -(*std::min_element(s_refined.begin(), s_refined.end())));
+
+  s_coarse_max_abs = *std::max_element(s_coarse.begin(), s_coarse.end());
+  s_coarse_max_abs = std::max(
+      s_coarse_max_abs, -(*std::min_element(s_coarse.begin(), s_coarse.end())));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output.h b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output.h
new file mode 100644
index 0000000000..d2d12082c6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_H_
+
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+
+namespace webrtc {
+
+// Stores the values being returned from the echo subtractor for a single
+// capture channel.
+struct SubtractorOutput {
+  SubtractorOutput();
+  ~SubtractorOutput();
+
+  std::array<float, kBlockSize> s_refined;
+  std::array<float, kBlockSize> s_coarse;
+  std::array<float, kBlockSize> e_refined;
+  std::array<float, kBlockSize> e_coarse;
+  FftData E_refined;
+  std::array<float, kFftLengthBy2Plus1> E2_refined;
+  std::array<float, kFftLengthBy2Plus1> E2_coarse;
+  float s2_refined = 0.f;
+  float s2_coarse = 0.f;
+  float e2_refined = 0.f;
+  float e2_coarse = 0.f;
+  float y2 = 0.f;
+  float s_refined_max_abs = 0.f;
+  float s_coarse_max_abs = 0.f;
+
+  // Reset the struct content.
+  void Reset();
+
+  // Updates the powers of the signals.
+  void ComputeMetrics(rtc::ArrayView<const float> y);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc
new file mode 100644
index 0000000000..baf0600161
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/subtractor_output_analyzer.h"
+
+#include <algorithm>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+
+SubtractorOutputAnalyzer::SubtractorOutputAnalyzer(size_t num_capture_channels)
+    : filters_converged_(num_capture_channels, false) {}
+
+void SubtractorOutputAnalyzer::Update(
+    rtc::ArrayView<const SubtractorOutput> subtractor_output,
+    bool* any_filter_converged,
+    bool* any_coarse_filter_converged,
+    bool* all_filters_diverged) {
+  RTC_DCHECK(any_filter_converged);
+  RTC_DCHECK(all_filters_diverged);
+  RTC_DCHECK_EQ(subtractor_output.size(), filters_converged_.size());
+
+  *any_filter_converged = false;
+  *any_coarse_filter_converged = false;
+  *all_filters_diverged = true;
+
+  for (size_t ch = 0; ch < subtractor_output.size(); ++ch) {
+    const float y2 = subtractor_output[ch].y2;
+    const float e2_refined = subtractor_output[ch].e2_refined;
+    const float e2_coarse = subtractor_output[ch].e2_coarse;
+
+    constexpr float kConvergenceThreshold = 50 * 50 * kBlockSize;
+    constexpr float kConvergenceThresholdLowLevel = 20 * 20 * kBlockSize;
+    bool refined_filter_converged =
+        e2_refined < 0.5f * y2 && y2 > kConvergenceThreshold;
+    bool coarse_filter_converged_strict =
+        e2_coarse < 0.05f * y2 && y2 > kConvergenceThreshold;
+    bool coarse_filter_converged_relaxed =
+        e2_coarse < 0.2f * y2 && y2 > kConvergenceThresholdLowLevel;
+    float min_e2 = std::min(e2_refined, e2_coarse);
+    bool filter_diverged = min_e2 > 1.5f * y2 && y2 > 30.f * 30.f * kBlockSize;
+    filters_converged_[ch] =
+        refined_filter_converged || coarse_filter_converged_strict;
+
+    *any_filter_converged = *any_filter_converged || filters_converged_[ch];
+    *any_coarse_filter_converged =
+        *any_coarse_filter_converged || coarse_filter_converged_relaxed;
+    *all_filters_diverged = *all_filters_diverged && filter_diverged;
+  }
+}
+
+void SubtractorOutputAnalyzer::HandleEchoPathChange() {
+  std::fill(filters_converged_.begin(), filters_converged_.end(), false);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h
new file mode 100644
index 0000000000..32707dbb19
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
+
+#include <vector>
+
+#include "modules/audio_processing/aec3/subtractor_output.h"
+
+namespace webrtc {
+
+// Class for analyzing the properties subtractor output.
+class SubtractorOutputAnalyzer {
+ public:
+  explicit SubtractorOutputAnalyzer(size_t num_capture_channels);
+  ~SubtractorOutputAnalyzer() = default;
+
+  // Analyses the subtractor output.
+  void Update(rtc::ArrayView<const SubtractorOutput> subtractor_output,
+              bool* any_filter_converged,
+              bool* any_coarse_filter_converged,
+              bool* all_filters_diverged);
+
+  const std::vector<bool>& ConvergedFilters() const {
+    return filters_converged_;
+  }
+
+  // Handle echo path change.
+  void HandleEchoPathChange();
+
+ private:
+  std::vector<bool> filters_converged_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_unittest.cc
new file mode 100644
index 0000000000..56b9cec9f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/subtractor_unittest.cc
@@ -0,0 +1,320 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/subtractor.h"
+
+#include <algorithm>
+#include <memory>
+#include <numeric>
+#include <string>
+
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "modules/audio_processing/utility/cascaded_biquad_filter.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+std::vector<float> RunSubtractorTest(
+    size_t num_render_channels,
+    size_t num_capture_channels,
+    int num_blocks_to_process,
+    int delay_samples,
+    int refined_filter_length_blocks,
+    int coarse_filter_length_blocks,
+    bool uncorrelated_inputs,
+    const std::vector<int>& blocks_with_echo_path_changes) {
+  ApmDataDumper data_dumper(42);
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  EchoCanceller3Config config;
+  config.filter.refined.length_blocks = refined_filter_length_blocks;
+  config.filter.coarse.length_blocks = coarse_filter_length_blocks;
+
+  Subtractor subtractor(config, num_render_channels, num_capture_channels,
+                        &data_dumper, DetectOptimization());
+  absl::optional<DelayEstimate> delay_estimate;
+  Block x(kNumBands, num_render_channels);
+  Block y(/*num_bands=*/1, num_capture_channels);
+  std::array<float, kBlockSize> x_old;
+  std::vector<SubtractorOutput> output(num_capture_channels);
+  config.delay.default_delay = 1;
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, num_render_channels));
+  RenderSignalAnalyzer render_signal_analyzer(config);
+  Random random_generator(42U);
+  Aec3Fft fft;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(num_capture_channels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2_refined(
+      num_capture_channels);
+  std::array<float, kFftLengthBy2Plus1> E2_coarse;
+  AecState aec_state(config, num_capture_channels);
+  x_old.fill(0.f);
+  for (auto& Y2_ch : Y2) {
+    Y2_ch.fill(0.f);
+  }
+  for (auto& E2_refined_ch : E2_refined) {
+    E2_refined_ch.fill(0.f);
+  }
+  E2_coarse.fill(0.f);
+
+  std::vector<std::vector<std::unique_ptr<DelayBuffer<float>>>> delay_buffer(
+      num_capture_channels);
+  for (size_t capture_ch = 0; capture_ch < num_capture_channels; ++capture_ch) {
+    delay_buffer[capture_ch].resize(num_render_channels);
+    for (size_t render_ch = 0; render_ch < num_render_channels; ++render_ch) {
+      delay_buffer[capture_ch][render_ch] =
+          std::make_unique<DelayBuffer<float>>(delay_samples);
+    }
+  }
+
+  // [B,A] = butter(2,100/8000,'high')
+  constexpr CascadedBiQuadFilter::BiQuadCoefficients
+      kHighPassFilterCoefficients = {{0.97261f, -1.94523f, 0.97261f},
+                                     {-1.94448f, 0.94598f}};
+  std::vector<std::unique_ptr<CascadedBiQuadFilter>> x_hp_filter(
+      num_render_channels);
+  for (size_t ch = 0; ch < num_render_channels; ++ch) {
+    x_hp_filter[ch] =
+        std::make_unique<CascadedBiQuadFilter>(kHighPassFilterCoefficients, 1);
+  }
+  std::vector<std::unique_ptr<CascadedBiQuadFilter>> y_hp_filter(
+      num_capture_channels);
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    y_hp_filter[ch] =
+        std::make_unique<CascadedBiQuadFilter>(kHighPassFilterCoefficients, 1);
+  }
+
+  for (int k = 0; k < num_blocks_to_process; ++k) {
+    for (size_t render_ch = 0; render_ch < num_render_channels; ++render_ch) {
+      RandomizeSampleVector(&random_generator, x.View(/*band=*/0, render_ch));
+    }
+    if (uncorrelated_inputs) {
+      for (size_t capture_ch = 0; capture_ch < num_capture_channels;
+           ++capture_ch) {
+        RandomizeSampleVector(&random_generator,
+                              y.View(/*band=*/0, capture_ch));
+      }
+    } else {
+      for (size_t capture_ch = 0; capture_ch < num_capture_channels;
+           ++capture_ch) {
+        rtc::ArrayView<float> y_view = y.View(/*band=*/0, capture_ch);
+        for (size_t render_ch = 0; render_ch < num_render_channels;
+             ++render_ch) {
+          std::array<float, kBlockSize> y_channel;
+          delay_buffer[capture_ch][render_ch]->Delay(
+              x.View(/*band=*/0, render_ch), y_channel);
+          for (size_t k = 0; k < kBlockSize; ++k) {
+            y_view[k] += y_channel[k] / num_render_channels;
+          }
+        }
+      }
+    }
+    for (size_t ch = 0; ch < num_render_channels; ++ch) {
+      x_hp_filter[ch]->Process(x.View(/*band=*/0, ch));
+    }
+    for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+      y_hp_filter[ch]->Process(y.View(/*band=*/0, ch));
+    }
+
+    render_delay_buffer->Insert(x);
+    if (k == 0) {
+      render_delay_buffer->Reset();
+    }
+    render_delay_buffer->PrepareCaptureProcessing();
+    render_signal_analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
+                                  aec_state.MinDirectPathFilterDelay());
+
+    // Handle echo path changes.
+    if (std::find(blocks_with_echo_path_changes.begin(),
+                  blocks_with_echo_path_changes.end(),
+                  k) != blocks_with_echo_path_changes.end()) {
+      subtractor.HandleEchoPathChange(EchoPathVariability(
+          true, EchoPathVariability::DelayAdjustment::kNewDetectedDelay,
+          false));
+    }
+    subtractor.Process(*render_delay_buffer->GetRenderBuffer(), y,
+                       render_signal_analyzer, aec_state, output);
+
+    aec_state.HandleEchoPathChange(EchoPathVariability(
+        false, EchoPathVariability::DelayAdjustment::kNone, false));
+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponses(),
+                     subtractor.FilterImpulseResponses(),
+                     *render_delay_buffer->GetRenderBuffer(), E2_refined, Y2,
+                     output);
+  }
+
+  std::vector<float> results(num_capture_channels);
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    const float output_power = std::inner_product(
+        output[ch].e_refined.begin(), output[ch].e_refined.end(),
+        output[ch].e_refined.begin(), 0.f);
+    const float y_power =
+        std::inner_product(y.begin(/*band=*/0, ch), y.end(/*band=*/0, ch),
+                           y.begin(/*band=*/0, ch), 0.f);
+    if (y_power == 0.f) {
+      ADD_FAILURE();
+      results[ch] = -1.f;
+    }
+    results[ch] = output_power / y_power;
+  }
+  return results;
+}
+
+std::string ProduceDebugText(size_t num_render_channels,
+                             size_t num_capture_channels,
+                             size_t delay,
+                             int filter_length_blocks) {
+  rtc::StringBuilder ss;
+  ss << "delay: " << delay << ", ";
+  ss << "filter_length_blocks:" << filter_length_blocks << ", ";
+  ss << "num_render_channels:" << num_render_channels << ", ";
+  ss << "num_capture_channels:" << num_capture_channels;
+  return ss.Release();
+}
+
+}  // namespace
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies that the check for non data dumper works.
+TEST(SubtractorDeathTest, NullDataDumper) {
+  EXPECT_DEATH(
+      Subtractor(EchoCanceller3Config(), 1, 1, nullptr, DetectOptimization()),
+      "");
+}
+
+#endif
+
+// Verifies that the subtractor is able to converge on correlated data.
+TEST(Subtractor, Convergence) {
+  std::vector<int> blocks_with_echo_path_changes;
+  for (size_t filter_length_blocks : {12, 20, 30}) {
+    for (size_t delay_samples : {0, 64, 150, 200, 301}) {
+      SCOPED_TRACE(ProduceDebugText(1, 1, delay_samples, filter_length_blocks));
+      std::vector<float> echo_to_nearend_powers = RunSubtractorTest(
+          1, 1, 2500, delay_samples, filter_length_blocks, filter_length_blocks,
+          false, blocks_with_echo_path_changes);
+
+      for (float echo_to_nearend_power : echo_to_nearend_powers) {
+        EXPECT_GT(0.1f, echo_to_nearend_power);
+      }
+    }
+  }
+}
+
+// Verifies that the subtractor is able to handle the case when the refined
+// filter is longer than the coarse filter.
+TEST(Subtractor, RefinedFilterLongerThanCoarseFilter) {
+  std::vector<int> blocks_with_echo_path_changes;
+  std::vector<float> echo_to_nearend_powers = RunSubtractorTest(
+      1, 1, 400, 64, 20, 15, false, blocks_with_echo_path_changes);
+  for (float echo_to_nearend_power : echo_to_nearend_powers) {
+    EXPECT_GT(0.5f, echo_to_nearend_power);
+  }
+}
+
+// Verifies that the subtractor is able to handle the case when the coarse
+// filter is longer than the refined filter.
+TEST(Subtractor, CoarseFilterLongerThanRefinedFilter) {
+  std::vector<int> blocks_with_echo_path_changes;
+  std::vector<float> echo_to_nearend_powers = RunSubtractorTest(
+      1, 1, 400, 64, 15, 20, false, blocks_with_echo_path_changes);
+  for (float echo_to_nearend_power : echo_to_nearend_powers) {
+    EXPECT_GT(0.5f, echo_to_nearend_power);
+  }
+}
+
+// Verifies that the subtractor does not converge on uncorrelated signals.
+TEST(Subtractor, NonConvergenceOnUncorrelatedSignals) {
+  std::vector<int> blocks_with_echo_path_changes;
+  for (size_t filter_length_blocks : {12, 20, 30}) {
+    for (size_t delay_samples : {0, 64, 150, 200, 301}) {
+      SCOPED_TRACE(ProduceDebugText(1, 1, delay_samples, filter_length_blocks));
+
+      std::vector<float> echo_to_nearend_powers = RunSubtractorTest(
+          1, 1, 3000, delay_samples, filter_length_blocks, filter_length_blocks,
+          true, blocks_with_echo_path_changes);
+      for (float echo_to_nearend_power : echo_to_nearend_powers) {
+        EXPECT_NEAR(1.f, echo_to_nearend_power, 0.1);
+      }
+    }
+  }
+}
+
+class SubtractorMultiChannelUpToEightRender
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+#if defined(NDEBUG)
+INSTANTIATE_TEST_SUITE_P(NonDebugMultiChannel,
+                         SubtractorMultiChannelUpToEightRender,
+                         ::testing::Combine(::testing::Values(1, 2, 8),
+                                            ::testing::Values(1, 2, 4)));
+#else
+INSTANTIATE_TEST_SUITE_P(DebugMultiChannel,
+                         SubtractorMultiChannelUpToEightRender,
+                         ::testing::Combine(::testing::Values(1, 2),
+                                            ::testing::Values(1, 2)));
+#endif
+
+// Verifies that the subtractor is able to converge on correlated data.
+TEST_P(SubtractorMultiChannelUpToEightRender, Convergence) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+
+  std::vector<int> blocks_with_echo_path_changes;
+  size_t num_blocks_to_process = 2500 * num_render_channels;
+  std::vector<float> echo_to_nearend_powers = RunSubtractorTest(
+      num_render_channels, num_capture_channels, num_blocks_to_process, 64, 20,
+      20, false, blocks_with_echo_path_changes);
+
+  for (float echo_to_nearend_power : echo_to_nearend_powers) {
+    EXPECT_GT(0.1f, echo_to_nearend_power);
+  }
+}
+
+class SubtractorMultiChannelUpToFourRender
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<size_t, size_t>> {};
+
+#if defined(NDEBUG)
+INSTANTIATE_TEST_SUITE_P(NonDebugMultiChannel,
+                         SubtractorMultiChannelUpToFourRender,
+                         ::testing::Combine(::testing::Values(1, 2, 4),
+                                            ::testing::Values(1, 2, 4)));
+#else
+INSTANTIATE_TEST_SUITE_P(DebugMultiChannel,
+                         SubtractorMultiChannelUpToFourRender,
+                         ::testing::Combine(::testing::Values(1, 2),
+                                            ::testing::Values(1, 2)));
+#endif
+
+// Verifies that the subtractor does not converge on uncorrelated signals.
+TEST_P(SubtractorMultiChannelUpToFourRender,
+       NonConvergenceOnUncorrelatedSignals) {
+  const size_t num_render_channels = std::get<0>(GetParam());
+  const size_t num_capture_channels = std::get<1>(GetParam());
+
+  std::vector<int> blocks_with_echo_path_changes;
+  size_t num_blocks_to_process = 5000 * num_render_channels;
+  std::vector<float> echo_to_nearend_powers = RunSubtractorTest(
+      num_render_channels, num_capture_channels, num_blocks_to_process, 64, 20,
+      20, true, blocks_with_echo_path_changes);
+  for (float echo_to_nearend_power : echo_to_nearend_powers) {
+    EXPECT_LT(.8f, echo_to_nearend_power);
+    EXPECT_NEAR(1.f, echo_to_nearend_power, 0.25f);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter.cc b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter.cc
new file mode 100644
index 0000000000..83ded425d5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter.cc
@@ -0,0 +1,180 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/suppression_filter.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstring>
+#include <functional>
+#include <iterator>
+
+#include "modules/audio_processing/aec3/vector_math.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+
+// Hanning window from Matlab command win = sqrt(hanning(128)).
+const float kSqrtHanning[kFftLength] = {
+    0.00000000000000f, 0.02454122852291f, 0.04906767432742f, 0.07356456359967f,
+    0.09801714032956f, 0.12241067519922f, 0.14673047445536f, 0.17096188876030f,
+    0.19509032201613f, 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
+    0.29028467725446f, 0.31368174039889f, 0.33688985339222f, 0.35989503653499f,
+    0.38268343236509f, 0.40524131400499f, 0.42755509343028f, 0.44961132965461f,
+    0.47139673682600f, 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
+    0.55557023301960f, 0.57580819141785f, 0.59569930449243f, 0.61523159058063f,
+    0.63439328416365f, 0.65317284295378f, 0.67155895484702f, 0.68954054473707f,
+    0.70710678118655f, 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
+    0.77301045336274f, 0.78834642762661f, 0.80320753148064f, 0.81758481315158f,
+    0.83146961230255f, 0.84485356524971f, 0.85772861000027f, 0.87008699110871f,
+    0.88192126434835f, 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
+    0.92387953251129f, 0.93299279883474f, 0.94154406518302f, 0.94952818059304f,
+    0.95694033573221f, 0.96377606579544f, 0.97003125319454f, 0.97570213003853f,
+    0.98078528040323f, 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
+    0.99518472667220f, 0.99729045667869f, 0.99879545620517f, 0.99969881869620f,
+    1.00000000000000f, 0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
+    0.99518472667220f, 0.99247953459871f, 0.98917650996478f, 0.98527764238894f,
+    0.98078528040323f, 0.97570213003853f, 0.97003125319454f, 0.96377606579544f,
+    0.95694033573221f, 0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
+    0.92387953251129f, 0.91420975570353f, 0.90398929312344f, 0.89322430119552f,
+    0.88192126434835f, 0.87008699110871f, 0.85772861000027f, 0.84485356524971f,
+    0.83146961230255f, 0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
+    0.77301045336274f, 0.75720884650648f, 0.74095112535496f, 0.72424708295147f,
+    0.70710678118655f, 0.68954054473707f, 0.67155895484702f, 0.65317284295378f,
+    0.63439328416365f, 0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
+    0.55557023301960f, 0.53499761988710f, 0.51410274419322f, 0.49289819222978f,
+    0.47139673682600f, 0.44961132965461f, 0.42755509343028f, 0.40524131400499f,
+    0.38268343236509f, 0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
+    0.29028467725446f, 0.26671275747490f, 0.24298017990326f, 0.21910124015687f,
+    0.19509032201613f, 0.17096188876030f, 0.14673047445536f, 0.12241067519922f,
+    0.09801714032956f, 0.07356456359967f, 0.04906767432742f, 0.02454122852291f};
+
+}  // namespace
+
+SuppressionFilter::SuppressionFilter(Aec3Optimization optimization,
+                                     int sample_rate_hz,
+                                     size_t num_capture_channels)
+    : optimization_(optimization),
+      sample_rate_hz_(sample_rate_hz),
+      num_capture_channels_(num_capture_channels),
+      fft_(),
+      e_output_old_(NumBandsForRate(sample_rate_hz_),
+                    std::vector<std::array<float, kFftLengthBy2>>(
+                        num_capture_channels_)) {
+  RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
+  for (size_t b = 0; b < e_output_old_.size(); ++b) {
+    for (size_t ch = 0; ch < e_output_old_[b].size(); ++ch) {
+      e_output_old_[b][ch].fill(0.f);
+    }
+  }
+}
+
+SuppressionFilter::~SuppressionFilter() = default;
+
+void SuppressionFilter::ApplyGain(
+    rtc::ArrayView<const FftData> comfort_noise,
+    rtc::ArrayView<const FftData> comfort_noise_high_band,
+    const std::array<float, kFftLengthBy2Plus1>& suppression_gain,
+    float high_bands_gain,
+    rtc::ArrayView<const FftData> E_lowest_band,
+    Block* e) {
+  RTC_DCHECK(e);
+  RTC_DCHECK_EQ(e->NumBands(), NumBandsForRate(sample_rate_hz_));
+
+  // Comfort noise gain is sqrt(1-g^2), where g is the suppression gain.
+  std::array<float, kFftLengthBy2Plus1> noise_gain;
+  for (size_t i = 0; i < kFftLengthBy2Plus1; ++i) {
+    noise_gain[i] = 1.f - suppression_gain[i] * suppression_gain[i];
+  }
+  aec3::VectorMath(optimization_).Sqrt(noise_gain);
+
+  const float high_bands_noise_scaling =
+      0.4f * std::sqrt(1.f - high_bands_gain * high_bands_gain);
+
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    FftData E;
+
+    // Analysis filterbank.
+    E.Assign(E_lowest_band[ch]);
+
+    for (size_t i = 0; i < kFftLengthBy2Plus1; ++i) {
+      // Apply suppression gains.
+      float E_real = E.re[i] * suppression_gain[i];
+      float E_imag = E.im[i] * suppression_gain[i];
+
+      // Scale and add the comfort noise.
+      E.re[i] = E_real + noise_gain[i] * comfort_noise[ch].re[i];
+      E.im[i] = E_imag + noise_gain[i] * comfort_noise[ch].im[i];
+    }
+
+    // Synthesis filterbank.
+    std::array<float, kFftLength> e_extended;
+    constexpr float kIfftNormalization = 2.f / kFftLength;
+    fft_.Ifft(E, &e_extended);
+
+    auto e0 = e->View(/*band=*/0, ch);
+    float* e0_old = e_output_old_[0][ch].data();
+
+    // Window and add the first half of e_extended with the second half of
+    // e_extended from the previous block.
+    for (size_t i = 0; i < kFftLengthBy2; ++i) {
+      float e0_i = e0_old[i] * kSqrtHanning[kFftLengthBy2 + i];
+      e0_i += e_extended[i] * kSqrtHanning[i];
+      e0[i] = e0_i * kIfftNormalization;
+    }
+
+    // The second half of e_extended is stored for the succeeding frame.
+    std::copy(e_extended.begin() + kFftLengthBy2,
+              e_extended.begin() + kFftLength,
+              std::begin(e_output_old_[0][ch]));
+
+    // Apply suppression gain to upper bands.
+    for (int b = 1; b < e->NumBands(); ++b) {
+      auto e_band = e->View(b, ch);
+      for (size_t i = 0; i < kFftLengthBy2; ++i) {
+        e_band[i] *= high_bands_gain;
+      }
+    }
+
+    // Add comfort noise to band 1.
+    if (e->NumBands() > 1) {
+      E.Assign(comfort_noise_high_band[ch]);
+      std::array<float, kFftLength> time_domain_high_band_noise;
+      fft_.Ifft(E, &time_domain_high_band_noise);
+
+      auto e1 = e->View(/*band=*/1, ch);
+      const float gain = high_bands_noise_scaling * kIfftNormalization;
+      for (size_t i = 0; i < kFftLengthBy2; ++i) {
+        e1[i] += time_domain_high_band_noise[i] * gain;
+      }
+    }
+
+    // Delay upper bands to match the delay of the filter bank.
+    for (int b = 1; b < e->NumBands(); ++b) {
+      auto e_band = e->View(b, ch);
+      float* e_band_old = e_output_old_[b][ch].data();
+      for (size_t i = 0; i < kFftLengthBy2; ++i) {
+        std::swap(e_band[i], e_band_old[i]);
+      }
+    }
+
+    // Clamp output of all bands.
+    for (int b = 0; b < e->NumBands(); ++b) {
+      auto e_band = e->View(b, ch);
+      for (size_t i = 0; i < kFftLengthBy2; ++i) {
+        e_band[i] = rtc::SafeClamp(e_band[i], -32768.f, 32767.f);
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter.h b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter.h
new file mode 100644
index 0000000000..c18b2334bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_FILTER_H_
+
+#include <array>
+#include <vector>
+
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/block.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+
+namespace webrtc {
+
+class SuppressionFilter {
+ public:
+  SuppressionFilter(Aec3Optimization optimization,
+                    int sample_rate_hz,
+                    size_t num_capture_channels_);
+  ~SuppressionFilter();
+
+  SuppressionFilter(const SuppressionFilter&) = delete;
+  SuppressionFilter& operator=(const SuppressionFilter&) = delete;
+
+  void ApplyGain(rtc::ArrayView<const FftData> comfort_noise,
+                 rtc::ArrayView<const FftData> comfort_noise_high_bands,
+                 const std::array<float, kFftLengthBy2Plus1>& suppression_gain,
+                 float high_bands_gain,
+                 rtc::ArrayView<const FftData> E_lowest_band,
+                 Block* e);
+
+ private:
+  const Aec3Optimization optimization_;
+  const int sample_rate_hz_;
+  const size_t num_capture_channels_;
+  const Aec3Fft fft_;
+  std::vector<std::vector<std::array<float, kFftLengthBy2>>> e_output_old_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc
new file mode 100644
index 0000000000..464f5cfed2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc
@@ -0,0 +1,257 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/suppression_filter.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <cmath>
+#include <numeric>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kPi = 3.141592f;
+
+void ProduceSinusoid(int sample_rate_hz,
+                     float sinusoidal_frequency_hz,
+                     size_t* sample_counter,
+                     Block* x) {
+  // Produce a sinusoid of the specified frequency.
+  for (size_t k = *sample_counter, j = 0; k < (*sample_counter + kBlockSize);
+       ++k, ++j) {
+    for (int channel = 0; channel < x->NumChannels(); ++channel) {
+      x->View(/*band=*/0, channel)[j] =
+          32767.f *
+          std::sin(2.f * kPi * sinusoidal_frequency_hz * k / sample_rate_hz);
+    }
+  }
+  *sample_counter = *sample_counter + kBlockSize;
+
+  for (int band = 1; band < x->NumBands(); ++band) {
+    for (int channel = 0; channel < x->NumChannels(); ++channel) {
+      std::fill(x->begin(band, channel), x->end(band, channel), 0.f);
+    }
+  }
+}
+
+}  // namespace
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for null suppressor output.
+TEST(SuppressionFilterDeathTest, NullOutput) {
+  std::vector<FftData> cn(1);
+  std::vector<FftData> cn_high_bands(1);
+  std::vector<FftData> E(1);
+  std::array<float, kFftLengthBy2Plus1> gain;
+
+  EXPECT_DEATH(SuppressionFilter(Aec3Optimization::kNone, 16000, 1)
+                   .ApplyGain(cn, cn_high_bands, gain, 1.0f, E, nullptr),
+               "");
+}
+
+// Verifies the check for allowed sample rate.
+TEST(SuppressionFilterDeathTest, ProperSampleRate) {
+  EXPECT_DEATH(SuppressionFilter(Aec3Optimization::kNone, 16001, 1), "");
+}
+
+#endif
+
+// Verifies that no comfort noise is added when the gain is 1.
+TEST(SuppressionFilter, ComfortNoiseInUnityGain) {
+  SuppressionFilter filter(Aec3Optimization::kNone, 48000, 1);
+  std::vector<FftData> cn(1);
+  std::vector<FftData> cn_high_bands(1);
+  std::array<float, kFftLengthBy2Plus1> gain;
+  std::array<float, kFftLengthBy2> e_old_;
+  Aec3Fft fft;
+
+  e_old_.fill(0.f);
+  gain.fill(1.f);
+  cn[0].re.fill(1.f);
+  cn[0].im.fill(1.f);
+  cn_high_bands[0].re.fill(1.f);
+  cn_high_bands[0].im.fill(1.f);
+
+  Block e(3, kBlockSize);
+  Block e_ref = e;
+
+  std::vector<FftData> E(1);
+  fft.PaddedFft(e.View(/*band=*/0, /*channel=*/0), e_old_,
+                Aec3Fft::Window::kSqrtHanning, &E[0]);
+  std::copy(e.begin(/*band=*/0, /*channel=*/0),
+            e.end(/*band=*/0, /*channel=*/0), e_old_.begin());
+
+  filter.ApplyGain(cn, cn_high_bands, gain, 1.f, E, &e);
+
+  for (int band = 0; band < e.NumBands(); ++band) {
+    for (int channel = 0; channel < e.NumChannels(); ++channel) {
+      const auto e_view = e.View(band, channel);
+      const auto e_ref_view = e_ref.View(band, channel);
+      for (size_t sample = 0; sample < e_view.size(); ++sample) {
+        EXPECT_EQ(e_ref_view[sample], e_view[sample]);
+      }
+    }
+  }
+}
+
+// Verifies that the suppressor is able to suppress a signal.
+TEST(SuppressionFilter, SignalSuppression) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  constexpr size_t kNumChannels = 1;
+
+  SuppressionFilter filter(Aec3Optimization::kNone, kSampleRateHz, 1);
+  std::vector<FftData> cn(1);
+  std::vector<FftData> cn_high_bands(1);
+  std::array<float, kFftLengthBy2> e_old_;
+  Aec3Fft fft;
+  std::array<float, kFftLengthBy2Plus1> gain;
+  Block e(kNumBands, kNumChannels);
+  e_old_.fill(0.f);
+
+  gain.fill(1.f);
+  std::for_each(gain.begin() + 10, gain.end(), [](float& a) { a = 0.f; });
+
+  cn[0].re.fill(0.f);
+  cn[0].im.fill(0.f);
+  cn_high_bands[0].re.fill(0.f);
+  cn_high_bands[0].im.fill(0.f);
+
+  size_t sample_counter = 0;
+
+  float e0_input = 0.f;
+  float e0_output = 0.f;
+  for (size_t k = 0; k < 100; ++k) {
+    ProduceSinusoid(16000, 16000 * 40 / kFftLengthBy2 / 2, &sample_counter, &e);
+    e0_input = std::inner_product(e.begin(/*band=*/0, /*channel=*/0),
+                                  e.end(/*band=*/0, /*channel=*/0),
+                                  e.begin(/*band=*/0, /*channel=*/0), e0_input);
+
+    std::vector<FftData> E(1);
+    fft.PaddedFft(e.View(/*band=*/0, /*channel=*/0), e_old_,
+                  Aec3Fft::Window::kSqrtHanning, &E[0]);
+    std::copy(e.begin(/*band=*/0, /*channel=*/0),
+              e.end(/*band=*/0, /*channel=*/0), e_old_.begin());
+
+    filter.ApplyGain(cn, cn_high_bands, gain, 1.f, E, &e);
+    e0_output = std::inner_product(
+        e.begin(/*band=*/0, /*channel=*/0), e.end(/*band=*/0, /*channel=*/0),
+        e.begin(/*band=*/0, /*channel=*/0), e0_output);
+  }
+
+  EXPECT_LT(e0_output, e0_input / 1000.f);
+}
+
+// Verifies that the suppressor is able to pass through a desired signal while
+// applying suppressing for some frequencies.
+TEST(SuppressionFilter, SignalTransparency) {
+  constexpr size_t kNumChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  SuppressionFilter filter(Aec3Optimization::kNone, kSampleRateHz, 1);
+  std::vector<FftData> cn(1);
+  std::array<float, kFftLengthBy2> e_old_;
+  Aec3Fft fft;
+  std::vector<FftData> cn_high_bands(1);
+  std::array<float, kFftLengthBy2Plus1> gain;
+  Block e(kNumBands, kNumChannels);
+  e_old_.fill(0.f);
+  gain.fill(1.f);
+  std::for_each(gain.begin() + 30, gain.end(), [](float& a) { a = 0.f; });
+
+  cn[0].re.fill(0.f);
+  cn[0].im.fill(0.f);
+  cn_high_bands[0].re.fill(0.f);
+  cn_high_bands[0].im.fill(0.f);
+
+  size_t sample_counter = 0;
+
+  float e0_input = 0.f;
+  float e0_output = 0.f;
+  for (size_t k = 0; k < 100; ++k) {
+    ProduceSinusoid(16000, 16000 * 10 / kFftLengthBy2 / 2, &sample_counter, &e);
+    e0_input = std::inner_product(e.begin(/*band=*/0, /*channel=*/0),
+                                  e.end(/*band=*/0, /*channel=*/0),
+                                  e.begin(/*band=*/0, /*channel=*/0), e0_input);
+
+    std::vector<FftData> E(1);
+    fft.PaddedFft(e.View(/*band=*/0, /*channel=*/0), e_old_,
+                  Aec3Fft::Window::kSqrtHanning, &E[0]);
+    std::copy(e.begin(/*band=*/0, /*channel=*/0),
+              e.end(/*band=*/0, /*channel=*/0), e_old_.begin());
+
+    filter.ApplyGain(cn, cn_high_bands, gain, 1.f, E, &e);
+    e0_output = std::inner_product(
+        e.begin(/*band=*/0, /*channel=*/0), e.end(/*band=*/0, /*channel=*/0),
+        e.begin(/*band=*/0, /*channel=*/0), e0_output);
+  }
+
+  EXPECT_LT(0.9f * e0_input, e0_output);
+}
+
+// Verifies that the suppressor delay.
+TEST(SuppressionFilter, Delay) {
+  constexpr size_t kNumChannels = 1;
+  constexpr int kSampleRateHz = 48000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+
+  SuppressionFilter filter(Aec3Optimization::kNone, kSampleRateHz, 1);
+  std::vector<FftData> cn(1);
+  std::vector<FftData> cn_high_bands(1);
+  std::array<float, kFftLengthBy2> e_old_;
+  Aec3Fft fft;
+  std::array<float, kFftLengthBy2Plus1> gain;
+  Block e(kNumBands, kNumChannels);
+
+  gain.fill(1.f);
+
+  cn[0].re.fill(0.f);
+  cn[0].im.fill(0.f);
+  cn_high_bands[0].re.fill(0.f);
+  cn_high_bands[0].im.fill(0.f);
+
+  for (size_t k = 0; k < 100; ++k) {
+    for (size_t band = 0; band < kNumBands; ++band) {
+      for (size_t channel = 0; channel < kNumChannels; ++channel) {
+        auto e_view = e.View(band, channel);
+        for (size_t sample = 0; sample < kBlockSize; ++sample) {
+          e_view[sample] = k * kBlockSize + sample + channel;
+        }
+      }
+    }
+
+    std::vector<FftData> E(1);
+    fft.PaddedFft(e.View(/*band=*/0, /*channel=*/0), e_old_,
+                  Aec3Fft::Window::kSqrtHanning, &E[0]);
+    std::copy(e.begin(/*band=*/0, /*channel=*/0),
+              e.end(/*band=*/0, /*channel=*/0), e_old_.begin());
+
+    filter.ApplyGain(cn, cn_high_bands, gain, 1.f, E, &e);
+    if (k > 2) {
+      for (size_t band = 0; band < kNumBands; ++band) {
+        for (size_t channel = 0; channel < kNumChannels; ++channel) {
+          const auto e_view = e.View(band, channel);
+          for (size_t sample = 0; sample < kBlockSize; ++sample) {
+            EXPECT_NEAR(k * kBlockSize + sample - kBlockSize + channel,
+                        e_view[sample], 0.01);
+          }
+        }
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain.cc b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain.cc
new file mode 100644
index 0000000000..037dabaabe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain.cc
@@ -0,0 +1,465 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/suppression_gain.h"
+
+#include <math.h>
+#include <stddef.h>
+
+#include <algorithm>
+#include <numeric>
+
+#include "modules/audio_processing/aec3/dominant_nearend_detector.h"
+#include "modules/audio_processing/aec3/moving_average.h"
+#include "modules/audio_processing/aec3/subband_nearend_detector.h"
+#include "modules/audio_processing/aec3/vector_math.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+void LimitLowFrequencyGains(std::array<float, kFftLengthBy2Plus1>* gain) {
+  // Limit the low frequency gains to avoid the impact of the high-pass filter
+  // on the lower-frequency gain influencing the overall achieved gain.
+  (*gain)[0] = (*gain)[1] = std::min((*gain)[1], (*gain)[2]);
+}
+
+void LimitHighFrequencyGains(bool conservative_hf_suppression,
+                             std::array<float, kFftLengthBy2Plus1>* gain) {
+  // Limit the high frequency gains to avoid echo leakage due to an imperfect
+  // filter.
+  constexpr size_t kFirstBandToLimit = (64 * 2000) / 8000;
+  const float min_upper_gain = (*gain)[kFirstBandToLimit];
+  std::for_each(
+      gain->begin() + kFirstBandToLimit + 1, gain->end(),
+      [min_upper_gain](float& a) { a = std::min(a, min_upper_gain); });
+  (*gain)[kFftLengthBy2] = (*gain)[kFftLengthBy2Minus1];
+
+  if (conservative_hf_suppression) {
+    // Limits the gain in the frequencies for which the adaptive filter has not
+    // converged.
+    // TODO(peah): Make adaptive to take the actual filter error into account.
+    constexpr size_t kUpperAccurateBandPlus1 = 29;
+
+    constexpr float oneByBandsInSum =
+        1 / static_cast<float>(kUpperAccurateBandPlus1 - 20);
+    const float hf_gain_bound =
+        std::accumulate(gain->begin() + 20,
+                        gain->begin() + kUpperAccurateBandPlus1, 0.f) *
+        oneByBandsInSum;
+
+    std::for_each(
+        gain->begin() + kUpperAccurateBandPlus1, gain->end(),
+        [hf_gain_bound](float& a) { a = std::min(a, hf_gain_bound); });
+  }
+}
+
+// Scales the echo according to assessed audibility at the other end.
+void WeightEchoForAudibility(const EchoCanceller3Config& config,
+                             rtc::ArrayView<const float> echo,
+                             rtc::ArrayView<float> weighted_echo) {
+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, echo.size());
+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, weighted_echo.size());
+
+  auto weigh = [](float threshold, float normalizer, size_t begin, size_t end,
+                  rtc::ArrayView<const float> echo,
+                  rtc::ArrayView<float> weighted_echo) {
+    for (size_t k = begin; k < end; ++k) {
+      if (echo[k] < threshold) {
+        float tmp = (threshold - echo[k]) * normalizer;
+        weighted_echo[k] = echo[k] * std::max(0.f, 1.f - tmp * tmp);
+      } else {
+        weighted_echo[k] = echo[k];
+      }
+    }
+  };
+
+  float threshold = config.echo_audibility.floor_power *
+                    config.echo_audibility.audibility_threshold_lf;
+  float normalizer = 1.f / (threshold - config.echo_audibility.floor_power);
+  weigh(threshold, normalizer, 0, 3, echo, weighted_echo);
+
+  threshold = config.echo_audibility.floor_power *
+              config.echo_audibility.audibility_threshold_mf;
+  normalizer = 1.f / (threshold - config.echo_audibility.floor_power);
+  weigh(threshold, normalizer, 3, 7, echo, weighted_echo);
+
+  threshold = config.echo_audibility.floor_power *
+              config.echo_audibility.audibility_threshold_hf;
+  normalizer = 1.f / (threshold - config.echo_audibility.floor_power);
+  weigh(threshold, normalizer, 7, kFftLengthBy2Plus1, echo, weighted_echo);
+}
+
+}  // namespace
+
+std::atomic<int> SuppressionGain::instance_count_(0);
+
+float SuppressionGain::UpperBandsGain(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        comfort_noise_spectrum,
+    const absl::optional<int>& narrow_peak_band,
+    bool saturated_echo,
+    const Block& render,
+    const std::array<float, kFftLengthBy2Plus1>& low_band_gain) const {
+  RTC_DCHECK_LT(0, render.NumBands());
+  if (render.NumBands() == 1) {
+    return 1.f;
+  }
+  const int num_render_channels = render.NumChannels();
+
+  if (narrow_peak_band &&
+      (*narrow_peak_band > static_cast<int>(kFftLengthBy2Plus1 - 10))) {
+    return 0.001f;
+  }
+
+  constexpr size_t kLowBandGainLimit = kFftLengthBy2 / 2;
+  const float gain_below_8_khz = *std::min_element(
+      low_band_gain.begin() + kLowBandGainLimit, low_band_gain.end());
+
+  // Always attenuate the upper bands when there is saturated echo.
+  if (saturated_echo) {
+    return std::min(0.001f, gain_below_8_khz);
+  }
+
+  // Compute the upper and lower band energies.
+  const auto sum_of_squares = [](float a, float b) { return a + b * b; };
+  float low_band_energy = 0.f;
+  for (int ch = 0; ch < num_render_channels; ++ch) {
+    const float channel_energy =
+        std::accumulate(render.begin(/*band=*/0, ch),
+                        render.end(/*band=*/0, ch), 0.0f, sum_of_squares);
+    low_band_energy = std::max(low_band_energy, channel_energy);
+  }
+  float high_band_energy = 0.f;
+  for (int k = 1; k < render.NumBands(); ++k) {
+    for (int ch = 0; ch < num_render_channels; ++ch) {
+      const float energy = std::accumulate(
+          render.begin(k, ch), render.end(k, ch), 0.f, sum_of_squares);
+      high_band_energy = std::max(high_band_energy, energy);
+    }
+  }
+
+  // If there is more power in the lower frequencies than the upper frequencies,
+  // or if the power in upper frequencies is low, do not bound the gain in the
+  // upper bands.
+  float anti_howling_gain;
+  const float activation_threshold =
+      kBlockSize * config_.suppressor.high_bands_suppression
+                       .anti_howling_activation_threshold;
+  if (high_band_energy < std::max(low_band_energy, activation_threshold)) {
+    anti_howling_gain = 1.f;
+  } else {
+    // In all other cases, bound the gain for upper frequencies.
+    RTC_DCHECK_LE(low_band_energy, high_band_energy);
+    RTC_DCHECK_NE(0.f, high_band_energy);
+    anti_howling_gain =
+        config_.suppressor.high_bands_suppression.anti_howling_gain *
+        sqrtf(low_band_energy / high_band_energy);
+  }
+
+  float gain_bound = 1.f;
+  if (!dominant_nearend_detector_->IsNearendState()) {
+    // Bound the upper gain during significant echo activity.
+    const auto& cfg = config_.suppressor.high_bands_suppression;
+    auto low_frequency_energy = [](rtc::ArrayView<const float> spectrum) {
+      RTC_DCHECK_LE(16, spectrum.size());
+      return std::accumulate(spectrum.begin() + 1, spectrum.begin() + 16, 0.f);
+    };
+    for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+      const float echo_sum = low_frequency_energy(echo_spectrum[ch]);
+      const float noise_sum = low_frequency_energy(comfort_noise_spectrum[ch]);
+      if (echo_sum > cfg.enr_threshold * noise_sum) {
+        gain_bound = cfg.max_gain_during_echo;
+        break;
+      }
+    }
+  }
+
+  // Choose the gain as the minimum of the lower and upper gains.
+  return std::min(std::min(gain_below_8_khz, anti_howling_gain), gain_bound);
+}
+
+// Computes the gain to reduce the echo to a non audible level.
+void SuppressionGain::GainToNoAudibleEcho(
+    const std::array<float, kFftLengthBy2Plus1>& nearend,
+    const std::array<float, kFftLengthBy2Plus1>& echo,
+    const std::array<float, kFftLengthBy2Plus1>& masker,
+    std::array<float, kFftLengthBy2Plus1>* gain) const {
+  const auto& p = dominant_nearend_detector_->IsNearendState() ? nearend_params_
+                                                               : normal_params_;
+  for (size_t k = 0; k < gain->size(); ++k) {
+    float enr = echo[k] / (nearend[k] + 1.f);  // Echo-to-nearend ratio.
+    float emr = echo[k] / (masker[k] + 1.f);   // Echo-to-masker (noise) ratio.
+    float g = 1.0f;
+    if (enr > p.enr_transparent_[k] && emr > p.emr_transparent_[k]) {
+      g = (p.enr_suppress_[k] - enr) /
+          (p.enr_suppress_[k] - p.enr_transparent_[k]);
+      g = std::max(g, p.emr_transparent_[k] / emr);
+    }
+    (*gain)[k] = g;
+  }
+}
+
+// Compute the minimum gain as the attenuating gain to put the signal just
+// above the zero sample values.
+void SuppressionGain::GetMinGain(
+    rtc::ArrayView<const float> weighted_residual_echo,
+    rtc::ArrayView<const float> last_nearend,
+    rtc::ArrayView<const float> last_echo,
+    bool low_noise_render,
+    bool saturated_echo,
+    rtc::ArrayView<float> min_gain) const {
+  if (!saturated_echo) {
+    const float min_echo_power =
+        low_noise_render ? config_.echo_audibility.low_render_limit
+                         : config_.echo_audibility.normal_render_limit;
+
+    for (size_t k = 0; k < min_gain.size(); ++k) {
+      min_gain[k] = weighted_residual_echo[k] > 0.f
+                        ? min_echo_power / weighted_residual_echo[k]
+                        : 1.f;
+      min_gain[k] = std::min(min_gain[k], 1.f);
+    }
+
+    if (!initial_state_ ||
+        config_.suppressor.lf_smoothing_during_initial_phase) {
+      const float& dec = dominant_nearend_detector_->IsNearendState()
+                             ? nearend_params_.max_dec_factor_lf
+                             : normal_params_.max_dec_factor_lf;
+
+      for (int k = 0; k <= config_.suppressor.last_lf_smoothing_band; ++k) {
+        // Make sure the gains of the low frequencies do not decrease too
+        // quickly after strong nearend.
+        if (last_nearend[k] > last_echo[k] ||
+            k <= config_.suppressor.last_permanent_lf_smoothing_band) {
+          min_gain[k] = std::max(min_gain[k], last_gain_[k] * dec);
+          min_gain[k] = std::min(min_gain[k], 1.f);
+        }
+      }
+    }
+  } else {
+    std::fill(min_gain.begin(), min_gain.end(), 0.f);
+  }
+}
+
+// Compute the maximum gain by limiting the gain increase from the previous
+// gain.
+void SuppressionGain::GetMaxGain(rtc::ArrayView<float> max_gain) const {
+  const auto& inc = dominant_nearend_detector_->IsNearendState()
+                        ? nearend_params_.max_inc_factor
+                        : normal_params_.max_inc_factor;
+  const auto& floor = config_.suppressor.floor_first_increase;
+  for (size_t k = 0; k < max_gain.size(); ++k) {
+    max_gain[k] = std::min(std::max(last_gain_[k] * inc, floor), 1.f);
+  }
+}
+
+void SuppressionGain::LowerBandGain(
+    bool low_noise_render,
+    const AecState& aec_state,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        suppressor_input,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> residual_echo,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> comfort_noise,
+    bool clock_drift,
+    std::array<float, kFftLengthBy2Plus1>* gain) {
+  gain->fill(1.f);
+  const bool saturated_echo = aec_state.SaturatedEcho();
+  std::array<float, kFftLengthBy2Plus1> max_gain;
+  GetMaxGain(max_gain);
+
+  for (size_t ch = 0; ch < num_capture_channels_; ++ch) {
+    std::array<float, kFftLengthBy2Plus1> G;
+    std::array<float, kFftLengthBy2Plus1> nearend;
+    nearend_smoothers_[ch].Average(suppressor_input[ch], nearend);
+
+    // Weight echo power in terms of audibility.
+    std::array<float, kFftLengthBy2Plus1> weighted_residual_echo;
+    WeightEchoForAudibility(config_, residual_echo[ch], weighted_residual_echo);
+
+    std::array<float, kFftLengthBy2Plus1> min_gain;
+    GetMinGain(weighted_residual_echo, last_nearend_[ch], last_echo_[ch],
+               low_noise_render, saturated_echo, min_gain);
+
+    GainToNoAudibleEcho(nearend, weighted_residual_echo, comfort_noise[0], &G);
+
+    // Clamp gains.
+    for (size_t k = 0; k < gain->size(); ++k) {
+      G[k] = std::max(std::min(G[k], max_gain[k]), min_gain[k]);
+      (*gain)[k] = std::min((*gain)[k], G[k]);
+    }
+
+    // Store data required for the gain computation of the next block.
+    std::copy(nearend.begin(), nearend.end(), last_nearend_[ch].begin());
+    std::copy(weighted_residual_echo.begin(), weighted_residual_echo.end(),
+              last_echo_[ch].begin());
+  }
+
+  LimitLowFrequencyGains(gain);
+  // Use conservative high-frequency gains during clock-drift or when not in
+  // dominant nearend.
+  if (!dominant_nearend_detector_->IsNearendState() || clock_drift ||
+      config_.suppressor.conservative_hf_suppression) {
+    LimitHighFrequencyGains(config_.suppressor.conservative_hf_suppression,
+                            gain);
+  }
+
+  // Store computed gains.
+  std::copy(gain->begin(), gain->end(), last_gain_.begin());
+
+  // Transform gains to amplitude domain.
+  aec3::VectorMath(optimization_).Sqrt(*gain);
+}
+
+SuppressionGain::SuppressionGain(const EchoCanceller3Config& config,
+                                 Aec3Optimization optimization,
+                                 int sample_rate_hz,
+                                 size_t num_capture_channels)
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      optimization_(optimization),
+      config_(config),
+      num_capture_channels_(num_capture_channels),
+      state_change_duration_blocks_(
+          static_cast<int>(config_.filter.config_change_duration_blocks)),
+      last_nearend_(num_capture_channels_, {0}),
+      last_echo_(num_capture_channels_, {0}),
+      nearend_smoothers_(
+          num_capture_channels_,
+          aec3::MovingAverage(kFftLengthBy2Plus1,
+                              config.suppressor.nearend_average_blocks)),
+      nearend_params_(config_.suppressor.last_lf_band,
+                      config_.suppressor.first_hf_band,
+                      config_.suppressor.nearend_tuning),
+      normal_params_(config_.suppressor.last_lf_band,
+                     config_.suppressor.first_hf_band,
+                     config_.suppressor.normal_tuning),
+      use_unbounded_echo_spectrum_(config.suppressor.dominant_nearend_detection
+                                       .use_unbounded_echo_spectrum) {
+  RTC_DCHECK_LT(0, state_change_duration_blocks_);
+  last_gain_.fill(1.f);
+  if (config_.suppressor.use_subband_nearend_detection) {
+    dominant_nearend_detector_ = std::make_unique<SubbandNearendDetector>(
+        config_.suppressor.subband_nearend_detection, num_capture_channels_);
+  } else {
+    dominant_nearend_detector_ = std::make_unique<DominantNearendDetector>(
+        config_.suppressor.dominant_nearend_detection, num_capture_channels_);
+  }
+  RTC_DCHECK(dominant_nearend_detector_);
+}
+
+SuppressionGain::~SuppressionGain() = default;
+
+void SuppressionGain::GetGain(
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        nearend_spectrum,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        residual_echo_spectrum,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        residual_echo_spectrum_unbounded,
+    rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+        comfort_noise_spectrum,
+    const RenderSignalAnalyzer& render_signal_analyzer,
+    const AecState& aec_state,
+    const Block& render,
+    bool clock_drift,
+    float* high_bands_gain,
+    std::array<float, kFftLengthBy2Plus1>* low_band_gain) {
+  RTC_DCHECK(high_bands_gain);
+  RTC_DCHECK(low_band_gain);
+
+  // Choose residual echo spectrum for dominant nearend detection.
+  const auto echo = use_unbounded_echo_spectrum_
+                        ? residual_echo_spectrum_unbounded
+                        : residual_echo_spectrum;
+
+  // Update the nearend state selection.
+  dominant_nearend_detector_->Update(nearend_spectrum, echo,
+                                     comfort_noise_spectrum, initial_state_);
+
+  // Compute gain for the lower band.
+  bool low_noise_render = low_render_detector_.Detect(render);
+  LowerBandGain(low_noise_render, aec_state, nearend_spectrum,
+                residual_echo_spectrum, comfort_noise_spectrum, clock_drift,
+                low_band_gain);
+
+  // Compute the gain for the upper bands.
+  const absl::optional<int> narrow_peak_band =
+      render_signal_analyzer.NarrowPeakBand();
+
+  *high_bands_gain =
+      UpperBandsGain(echo_spectrum, comfort_noise_spectrum, narrow_peak_band,
+                     aec_state.SaturatedEcho(), render, *low_band_gain);
+
+  data_dumper_->DumpRaw("aec3_dominant_nearend",
+                        dominant_nearend_detector_->IsNearendState());
+}
+
+void SuppressionGain::SetInitialState(bool state) {
+  initial_state_ = state;
+  if (state) {
+    initial_state_change_counter_ = state_change_duration_blocks_;
+  } else {
+    initial_state_change_counter_ = 0;
+  }
+}
+
+// Detects when the render signal can be considered to have low power and
+// consist of stationary noise.
+bool SuppressionGain::LowNoiseRenderDetector::Detect(const Block& render) {
+  float x2_sum = 0.f;
+  float x2_max = 0.f;
+  for (int ch = 0; ch < render.NumChannels(); ++ch) {
+    for (float x_k : render.View(/*band=*/0, ch)) {
+      const float x2 = x_k * x_k;
+      x2_sum += x2;
+      x2_max = std::max(x2_max, x2);
+    }
+  }
+  x2_sum = x2_sum / render.NumChannels();
+
+  constexpr float kThreshold = 50.f * 50.f * 64.f;
+  const bool low_noise_render =
+      average_power_ < kThreshold && x2_max < 3 * average_power_;
+  average_power_ = average_power_ * 0.9f + x2_sum * 0.1f;
+  return low_noise_render;
+}
+
+SuppressionGain::GainParameters::GainParameters(
+    int last_lf_band,
+    int first_hf_band,
+    const EchoCanceller3Config::Suppressor::Tuning& tuning)
+    : max_inc_factor(tuning.max_inc_factor),
+      max_dec_factor_lf(tuning.max_dec_factor_lf) {
+  // Compute per-band masking thresholds.
+  RTC_DCHECK_LT(last_lf_band, first_hf_band);
+  auto& lf = tuning.mask_lf;
+  auto& hf = tuning.mask_hf;
+  RTC_DCHECK_LT(lf.enr_transparent, lf.enr_suppress);
+  RTC_DCHECK_LT(hf.enr_transparent, hf.enr_suppress);
+  for (int k = 0; k < static_cast<int>(kFftLengthBy2Plus1); k++) {
+    float a;
+    if (k <= last_lf_band) {
+      a = 0.f;
+    } else if (k < first_hf_band) {
+      a = (k - last_lf_band) / static_cast<float>(first_hf_band - last_lf_band);
+    } else {
+      a = 1.f;
+    }
+    enr_transparent_[k] = (1 - a) * lf.enr_transparent + a * hf.enr_transparent;
+    enr_suppress_[k] = (1 - a) * lf.enr_suppress + a * hf.enr_suppress;
+    emr_transparent_[k] = (1 - a) * lf.emr_transparent + a * hf.emr_transparent;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain.h b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain.h
new file mode 100644
index 0000000000..c19ddd7e30
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain.h
@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_H_
+
+#include <array>
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "modules/audio_processing/aec3/moving_average.h"
+#include "modules/audio_processing/aec3/nearend_detector.h"
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+class SuppressionGain {
+ public:
+  SuppressionGain(const EchoCanceller3Config& config,
+                  Aec3Optimization optimization,
+                  int sample_rate_hz,
+                  size_t num_capture_channels);
+  ~SuppressionGain();
+
+  SuppressionGain(const SuppressionGain&) = delete;
+  SuppressionGain& operator=(const SuppressionGain&) = delete;
+
+  void GetGain(
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          nearend_spectrum,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          residual_echo_spectrum,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          residual_echo_spectrum_unbounded,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          comfort_noise_spectrum,
+      const RenderSignalAnalyzer& render_signal_analyzer,
+      const AecState& aec_state,
+      const Block& render,
+      bool clock_drift,
+      float* high_bands_gain,
+      std::array<float, kFftLengthBy2Plus1>* low_band_gain);
+
+  bool IsDominantNearend() {
+    return dominant_nearend_detector_->IsNearendState();
+  }
+
+  // Toggles the usage of the initial state.
+  void SetInitialState(bool state);
+
+ private:
+  // Computes the gain to apply for the bands beyond the first band.
+  float UpperBandsGain(
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> echo_spectrum,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          comfort_noise_spectrum,
+      const absl::optional<int>& narrow_peak_band,
+      bool saturated_echo,
+      const Block& render,
+      const std::array<float, kFftLengthBy2Plus1>& low_band_gain) const;
+
+  void GainToNoAudibleEcho(const std::array<float, kFftLengthBy2Plus1>& nearend,
+                           const std::array<float, kFftLengthBy2Plus1>& echo,
+                           const std::array<float, kFftLengthBy2Plus1>& masker,
+                           std::array<float, kFftLengthBy2Plus1>* gain) const;
+
+  void LowerBandGain(
+      bool stationary_with_low_power,
+      const AecState& aec_state,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>>
+          suppressor_input,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> residual_echo,
+      rtc::ArrayView<const std::array<float, kFftLengthBy2Plus1>> comfort_noise,
+      bool clock_drift,
+      std::array<float, kFftLengthBy2Plus1>* gain);
+
+  void GetMinGain(rtc::ArrayView<const float> weighted_residual_echo,
+                  rtc::ArrayView<const float> last_nearend,
+                  rtc::ArrayView<const float> last_echo,
+                  bool low_noise_render,
+                  bool saturated_echo,
+                  rtc::ArrayView<float> min_gain) const;
+
+  void GetMaxGain(rtc::ArrayView<float> max_gain) const;
+
+  class LowNoiseRenderDetector {
+   public:
+    bool Detect(const Block& render);
+
+   private:
+    float average_power_ = 32768.f * 32768.f;
+  };
+
+  struct GainParameters {
+    explicit GainParameters(
+        int last_lf_band,
+        int first_hf_band,
+        const EchoCanceller3Config::Suppressor::Tuning& tuning);
+    const float max_inc_factor;
+    const float max_dec_factor_lf;
+    std::array<float, kFftLengthBy2Plus1> enr_transparent_;
+    std::array<float, kFftLengthBy2Plus1> enr_suppress_;
+    std::array<float, kFftLengthBy2Plus1> emr_transparent_;
+  };
+
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  const Aec3Optimization optimization_;
+  const EchoCanceller3Config config_;
+  const size_t num_capture_channels_;
+  const int state_change_duration_blocks_;
+  std::array<float, kFftLengthBy2Plus1> last_gain_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> last_nearend_;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> last_echo_;
+  LowNoiseRenderDetector low_render_detector_;
+  bool initial_state_ = true;
+  int initial_state_change_counter_ = 0;
+  std::vector<aec3::MovingAverage> nearend_smoothers_;
+  const GainParameters nearend_params_;
+  const GainParameters normal_params_;
+  // Determines if the dominant nearend detector uses the unbounded residual
+  // echo spectrum.
+  const bool use_unbounded_echo_spectrum_;
+  std::unique_ptr<NearendDetector> dominant_nearend_detector_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc
new file mode 100644
index 0000000000..02de706c77
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc
@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/suppression_gain.h"
+
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/aec3/subtractor.h"
+#include "modules/audio_processing/aec3/subtractor_output.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace aec3 {
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies that the check for non-null output gains works.
+TEST(SuppressionGainDeathTest, NullOutputGains) {
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2(1, {0.0f});
+  std::vector<std::array<float, kFftLengthBy2Plus1>> R2(1, {0.0f});
+  std::vector<std::array<float, kFftLengthBy2Plus1>> R2_unbounded(1, {0.0f});
+  std::vector<std::array<float, kFftLengthBy2Plus1>> S2(1);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> N2(1, {0.0f});
+  for (auto& S2_k : S2) {
+    S2_k.fill(0.1f);
+  }
+  FftData E;
+  FftData Y;
+  E.re.fill(0.0f);
+  E.im.fill(0.0f);
+  Y.re.fill(0.0f);
+  Y.im.fill(0.0f);
+
+  float high_bands_gain;
+  AecState aec_state(EchoCanceller3Config{}, 1);
+  EXPECT_DEATH(
+      SuppressionGain(EchoCanceller3Config{}, DetectOptimization(), 16000, 1)
+          .GetGain(E2, S2, R2, R2_unbounded, N2,
+                   RenderSignalAnalyzer((EchoCanceller3Config{})), aec_state,
+                   Block(3, 1), false, &high_bands_gain, nullptr),
+      "");
+}
+
+#endif
+
+// Does a sanity check that the gains are correctly computed.
+TEST(SuppressionGain, BasicGainComputation) {
+  constexpr size_t kNumRenderChannels = 1;
+  constexpr size_t kNumCaptureChannels = 2;
+  constexpr int kSampleRateHz = 16000;
+  constexpr size_t kNumBands = NumBandsForRate(kSampleRateHz);
+  SuppressionGain suppression_gain(EchoCanceller3Config(), DetectOptimization(),
+                                   kSampleRateHz, kNumCaptureChannels);
+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
+  float high_bands_gain;
+  std::vector<std::array<float, kFftLengthBy2Plus1>> E2(kNumCaptureChannels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> S2(kNumCaptureChannels,
+                                                        {0.0f});
+  std::vector<std::array<float, kFftLengthBy2Plus1>> Y2(kNumCaptureChannels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> R2(kNumCaptureChannels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> R2_unbounded(
+      kNumCaptureChannels);
+  std::vector<std::array<float, kFftLengthBy2Plus1>> N2(kNumCaptureChannels);
+  std::array<float, kFftLengthBy2Plus1> g;
+  std::vector<SubtractorOutput> output(kNumCaptureChannels);
+  Block x(kNumBands, kNumRenderChannels);
+  EchoCanceller3Config config;
+  AecState aec_state(config, kNumCaptureChannels);
+  ApmDataDumper data_dumper(42);
+  Subtractor subtractor(config, kNumRenderChannels, kNumCaptureChannels,
+                        &data_dumper, DetectOptimization());
+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
+      RenderDelayBuffer::Create(config, kSampleRateHz, kNumRenderChannels));
+  absl::optional<DelayEstimate> delay_estimate;
+
+  // Ensure that a strong noise is detected to mask any echoes.
+  for (size_t ch = 0; ch < kNumCaptureChannels; ++ch) {
+    E2[ch].fill(10.f);
+    Y2[ch].fill(10.f);
+    R2[ch].fill(0.1f);
+    R2_unbounded[ch].fill(0.1f);
+    N2[ch].fill(100.0f);
+  }
+  for (auto& subtractor_output : output) {
+    subtractor_output.Reset();
+  }
+
+  // Ensure that the gain is no longer forced to zero.
+  for (int k = 0; k <= kNumBlocksPerSecond / 5 + 1; ++k) {
+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponses(),
+                     subtractor.FilterImpulseResponses(),
+                     *render_delay_buffer->GetRenderBuffer(), E2, Y2, output);
+  }
+
+  for (int k = 0; k < 100; ++k) {
+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponses(),
+                     subtractor.FilterImpulseResponses(),
+                     *render_delay_buffer->GetRenderBuffer(), E2, Y2, output);
+    suppression_gain.GetGain(E2, S2, R2, R2_unbounded, N2, analyzer, aec_state,
+                             x, false, &high_bands_gain, &g);
+  }
+  std::for_each(g.begin(), g.end(),
+                [](float a) { EXPECT_NEAR(1.0f, a, 0.001f); });
+
+  // Ensure that a strong nearend is detected to mask any echoes.
+  for (size_t ch = 0; ch < kNumCaptureChannels; ++ch) {
+    E2[ch].fill(100.f);
+    Y2[ch].fill(100.f);
+    R2[ch].fill(0.1f);
+    R2_unbounded[ch].fill(0.1f);
+    S2[ch].fill(0.1f);
+    N2[ch].fill(0.f);
+  }
+
+  for (int k = 0; k < 100; ++k) {
+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponses(),
+                     subtractor.FilterImpulseResponses(),
+                     *render_delay_buffer->GetRenderBuffer(), E2, Y2, output);
+    suppression_gain.GetGain(E2, S2, R2, R2_unbounded, N2, analyzer, aec_state,
+                             x, false, &high_bands_gain, &g);
+  }
+  std::for_each(g.begin(), g.end(),
+                [](float a) { EXPECT_NEAR(1.0f, a, 0.001f); });
+
+  // Add a strong echo to one of the channels and ensure that it is suppressed.
+  E2[1].fill(1000000000.0f);
+  R2[1].fill(10000000000000.0f);
+  R2_unbounded[1].fill(10000000000000.0f);
+
+  for (int k = 0; k < 10; ++k) {
+    suppression_gain.GetGain(E2, S2, R2, R2_unbounded, N2, analyzer, aec_state,
+                             x, false, &high_bands_gain, &g);
+  }
+  std::for_each(g.begin(), g.end(),
+                [](float a) { EXPECT_NEAR(0.0f, a, 0.001f); });
+}
+
+}  // namespace aec3
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/transparent_mode.cc b/third_party/libwebrtc/modules/audio_processing/aec3/transparent_mode.cc
new file mode 100644
index 0000000000..489f53f4f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/transparent_mode.cc
@@ -0,0 +1,243 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/transparent_mode.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kBlocksSinceConvergencedFilterInit = 10000;
+constexpr size_t kBlocksSinceConsistentEstimateInit = 10000;
+
+bool DeactivateTransparentMode() {
+  return field_trial::IsEnabled("WebRTC-Aec3TransparentModeKillSwitch");
+}
+
+bool ActivateTransparentModeHmm() {
+  return field_trial::IsEnabled("WebRTC-Aec3TransparentModeHmm");
+}
+
+}  // namespace
+
+// Classifier that toggles transparent mode which reduces echo suppression when
+// headsets are used.
+class TransparentModeImpl : public TransparentMode {
+ public:
+  bool Active() const override { return transparency_activated_; }
+
+  void Reset() override {
+    // Determines if transparent mode is used.
+    transparency_activated_ = false;
+
+    // The estimated probability of being transparent mode.
+    prob_transparent_state_ = 0.f;
+  }
+
+  void Update(int filter_delay_blocks,
+              bool any_filter_consistent,
+              bool any_filter_converged,
+              bool any_coarse_filter_converged,
+              bool all_filters_diverged,
+              bool active_render,
+              bool saturated_capture) override {
+    // The classifier is implemented as a Hidden Markov Model (HMM) with two
+    // hidden states: "normal" and "transparent". The estimated probabilities of
+    // the two states are updated by observing filter convergence during active
+    // render. The filters are less likely to be reported as converged when
+    // there is no echo present in the microphone signal.
+
+    // The constants have been obtained by observing active_render and
+    // any_coarse_filter_converged under varying call scenarios. They
+    // have further been hand tuned to prefer normal state during uncertain
+    // regions (to avoid echo leaks).
+
+    // The model is only updated during active render.
+    if (!active_render)
+      return;
+
+    // Probability of switching from one state to the other.
+    constexpr float kSwitch = 0.000001f;
+
+    // Probability of observing converged filters in states "normal" and
+    // "transparent" during active render.
+    constexpr float kConvergedNormal = 0.01f;
+    constexpr float kConvergedTransparent = 0.001f;
+
+    // Probability of transitioning to transparent state from normal state and
+    // transparent state respectively.
+    constexpr float kA[2] = {kSwitch, 1.f - kSwitch};
+
+    // Probability of the two observations (converged filter or not converged
+    // filter) in normal state and transparent state respectively.
+    constexpr float kB[2][2] = {
+        {1.f - kConvergedNormal, kConvergedNormal},
+        {1.f - kConvergedTransparent, kConvergedTransparent}};
+
+    // Probability of the two states before the update.
+    const float prob_transparent = prob_transparent_state_;
+    const float prob_normal = 1.f - prob_transparent;
+
+    // Probability of transitioning to transparent state.
+    const float prob_transition_transparent =
+        prob_normal * kA[0] + prob_transparent * kA[1];
+    const float prob_transition_normal = 1.f - prob_transition_transparent;
+
+    // Observed output.
+    const int out = static_cast<int>(any_coarse_filter_converged);
+
+    // Joint probabilites of the observed output and respective states.
+    const float prob_joint_normal = prob_transition_normal * kB[0][out];
+    const float prob_joint_transparent =
+        prob_transition_transparent * kB[1][out];
+
+    // Conditional probability of transparent state and the observed output.
+    RTC_DCHECK_GT(prob_joint_normal + prob_joint_transparent, 0.f);
+    prob_transparent_state_ =
+        prob_joint_transparent / (prob_joint_normal + prob_joint_transparent);
+
+    // Transparent mode is only activated when its state probability is high.
+    // Dead zone between activation/deactivation thresholds to avoid switching
+    // back and forth.
+    if (prob_transparent_state_ > 0.95f) {
+      transparency_activated_ = true;
+    } else if (prob_transparent_state_ < 0.5f) {
+      transparency_activated_ = false;
+    }
+  }
+
+ private:
+  bool transparency_activated_ = false;
+  float prob_transparent_state_ = 0.f;
+};
+
+// Legacy classifier for toggling transparent mode.
+class LegacyTransparentModeImpl : public TransparentMode {
+ public:
+  explicit LegacyTransparentModeImpl(const EchoCanceller3Config& config)
+      : linear_and_stable_echo_path_(
+            config.echo_removal_control.linear_and_stable_echo_path),
+        active_blocks_since_sane_filter_(kBlocksSinceConsistentEstimateInit),
+        non_converged_sequence_size_(kBlocksSinceConvergencedFilterInit) {}
+
+  bool Active() const override { return transparency_activated_; }
+
+  void Reset() override {
+    non_converged_sequence_size_ = kBlocksSinceConvergencedFilterInit;
+    diverged_sequence_size_ = 0;
+    strong_not_saturated_render_blocks_ = 0;
+    if (linear_and_stable_echo_path_) {
+      recent_convergence_during_activity_ = false;
+    }
+  }
+
+  void Update(int filter_delay_blocks,
+              bool any_filter_consistent,
+              bool any_filter_converged,
+              bool any_coarse_filter_converged,
+              bool all_filters_diverged,
+              bool active_render,
+              bool saturated_capture) override {
+    ++capture_block_counter_;
+    strong_not_saturated_render_blocks_ +=
+        active_render && !saturated_capture ? 1 : 0;
+
+    if (any_filter_consistent && filter_delay_blocks < 5) {
+      sane_filter_observed_ = true;
+      active_blocks_since_sane_filter_ = 0;
+    } else if (active_render) {
+      ++active_blocks_since_sane_filter_;
+    }
+
+    bool sane_filter_recently_seen;
+    if (!sane_filter_observed_) {
+      sane_filter_recently_seen =
+          capture_block_counter_ <= 5 * kNumBlocksPerSecond;
+    } else {
+      sane_filter_recently_seen =
+          active_blocks_since_sane_filter_ <= 30 * kNumBlocksPerSecond;
+    }
+
+    if (any_filter_converged) {
+      recent_convergence_during_activity_ = true;
+      active_non_converged_sequence_size_ = 0;
+      non_converged_sequence_size_ = 0;
+      ++num_converged_blocks_;
+    } else {
+      if (++non_converged_sequence_size_ > 20 * kNumBlocksPerSecond) {
+        num_converged_blocks_ = 0;
+      }
+
+      if (active_render &&
+          ++active_non_converged_sequence_size_ > 60 * kNumBlocksPerSecond) {
+        recent_convergence_during_activity_ = false;
+      }
+    }
+
+    if (!all_filters_diverged) {
+      diverged_sequence_size_ = 0;
+    } else if (++diverged_sequence_size_ >= 60) {
+      // TODO(peah): Change these lines to ensure proper triggering of usable
+      // filter.
+      non_converged_sequence_size_ = kBlocksSinceConvergencedFilterInit;
+    }
+
+    if (active_non_converged_sequence_size_ > 60 * kNumBlocksPerSecond) {
+      finite_erl_recently_detected_ = false;
+    }
+    if (num_converged_blocks_ > 50) {
+      finite_erl_recently_detected_ = true;
+    }
+
+    if (finite_erl_recently_detected_) {
+      transparency_activated_ = false;
+    } else if (sane_filter_recently_seen &&
+               recent_convergence_during_activity_) {
+      transparency_activated_ = false;
+    } else {
+      const bool filter_should_have_converged =
+          strong_not_saturated_render_blocks_ > 6 * kNumBlocksPerSecond;
+      transparency_activated_ = filter_should_have_converged;
+    }
+  }
+
+ private:
+  const bool linear_and_stable_echo_path_;
+  size_t capture_block_counter_ = 0;
+  bool transparency_activated_ = false;
+  size_t active_blocks_since_sane_filter_;
+  bool sane_filter_observed_ = false;
+  bool finite_erl_recently_detected_ = false;
+  size_t non_converged_sequence_size_;
+  size_t diverged_sequence_size_ = 0;
+  size_t active_non_converged_sequence_size_ = 0;
+  size_t num_converged_blocks_ = 0;
+  bool recent_convergence_during_activity_ = false;
+  size_t strong_not_saturated_render_blocks_ = 0;
+};
+
+std::unique_ptr<TransparentMode> TransparentMode::Create(
+    const EchoCanceller3Config& config) {
+  if (config.ep_strength.bounded_erl || DeactivateTransparentMode()) {
+    RTC_LOG(LS_INFO) << "AEC3 Transparent Mode: Disabled";
+    return nullptr;
+  }
+  if (ActivateTransparentModeHmm()) {
+    RTC_LOG(LS_INFO) << "AEC3 Transparent Mode: HMM";
+    return std::make_unique<TransparentModeImpl>();
+  }
+  RTC_LOG(LS_INFO) << "AEC3 Transparent Mode: Legacy";
+  return std::make_unique<LegacyTransparentModeImpl>(config);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/transparent_mode.h b/third_party/libwebrtc/modules/audio_processing/aec3/transparent_mode.h
new file mode 100644
index 0000000000..bc5dd0391b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/transparent_mode.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_TRANSPARENT_MODE_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_TRANSPARENT_MODE_H_
+
+#include <memory>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+
+namespace webrtc {
+
+// Class for detecting and toggling the transparent mode which causes the
+// suppressor to apply less suppression.
+class TransparentMode {
+ public:
+  static std::unique_ptr<TransparentMode> Create(
+      const EchoCanceller3Config& config);
+
+  virtual ~TransparentMode() {}
+
+  // Returns whether the transparent mode should be active.
+  virtual bool Active() const = 0;
+
+  // Resets the state of the detector.
+  virtual void Reset() = 0;
+
+  // Updates the detection decision based on new data.
+  virtual void Update(int filter_delay_blocks,
+                      bool any_filter_consistent,
+                      bool any_filter_converged,
+                      bool any_coarse_filter_converged,
+                      bool all_filters_diverged,
+                      bool active_render,
+                      bool saturated_capture) = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_TRANSPARENT_MODE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/vector_math.h b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math.h
new file mode 100644
index 0000000000..e4d1381ae1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math.h
@@ -0,0 +1,229 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC3_VECTOR_MATH_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_VECTOR_MATH_H_
+
+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_HAS_NEON)
+#include <arm_neon.h>
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+#include <math.h>
+
+#include <algorithm>
+#include <array>
+#include <functional>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace aec3 {
+
+// Provides optimizations for mathematical operations based on vectors.
+class VectorMath {
+ public:
+  explicit VectorMath(Aec3Optimization optimization)
+      : optimization_(optimization) {}
+
+  // Elementwise square root.
+  void SqrtAVX2(rtc::ArrayView<float> x);
+  void Sqrt(rtc::ArrayView<float> x) {
+    switch (optimization_) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+      case Aec3Optimization::kSse2: {
+        const int x_size = static_cast<int>(x.size());
+        const int vector_limit = x_size >> 2;
+
+        int j = 0;
+        for (; j < vector_limit * 4; j += 4) {
+          __m128 g = _mm_loadu_ps(&x[j]);
+          g = _mm_sqrt_ps(g);
+          _mm_storeu_ps(&x[j], g);
+        }
+
+        for (; j < x_size; ++j) {
+          x[j] = sqrtf(x[j]);
+        }
+      } break;
+      case Aec3Optimization::kAvx2:
+        SqrtAVX2(x);
+        break;
+#endif
+#if defined(WEBRTC_HAS_NEON)
+      case Aec3Optimization::kNeon: {
+        const int x_size = static_cast<int>(x.size());
+        const int vector_limit = x_size >> 2;
+
+        int j = 0;
+        for (; j < vector_limit * 4; j += 4) {
+          float32x4_t g = vld1q_f32(&x[j]);
+#if !defined(WEBRTC_ARCH_ARM64)
+          float32x4_t y = vrsqrteq_f32(g);
+
+          // Code to handle sqrt(0).
+          // If the input to sqrtf() is zero, a zero will be returned.
+          // If the input to vrsqrteq_f32() is zero, positive infinity is
+          // returned.
+          const uint32x4_t vec_p_inf = vdupq_n_u32(0x7F800000);
+          // check for divide by zero
+          const uint32x4_t div_by_zero =
+              vceqq_u32(vec_p_inf, vreinterpretq_u32_f32(y));
+          // zero out the positive infinity results
+          y = vreinterpretq_f32_u32(
+              vandq_u32(vmvnq_u32(div_by_zero), vreinterpretq_u32_f32(y)));
+          // from arm documentation
+          // The Newton-Raphson iteration:
+          //     y[n+1] = y[n] * (3 - d * (y[n] * y[n])) / 2)
+          // converges to (1/√d) if y0 is the result of VRSQRTE applied to d.
+          //
+          // Note: The precision did not improve after 2 iterations.
+          for (int i = 0; i < 2; i++) {
+            y = vmulq_f32(vrsqrtsq_f32(vmulq_f32(y, y), g), y);
+          }
+          // sqrt(g) = g * 1/sqrt(g)
+          g = vmulq_f32(g, y);
+#else
+          g = vsqrtq_f32(g);
+#endif
+          vst1q_f32(&x[j], g);
+        }
+
+        for (; j < x_size; ++j) {
+          x[j] = sqrtf(x[j]);
+        }
+      }
+#endif
+      break;
+      default:
+        std::for_each(x.begin(), x.end(), [](float& a) { a = sqrtf(a); });
+    }
+  }
+
+  // Elementwise vector multiplication z = x * y.
+  void MultiplyAVX2(rtc::ArrayView<const float> x,
+                    rtc::ArrayView<const float> y,
+                    rtc::ArrayView<float> z);
+  void Multiply(rtc::ArrayView<const float> x,
+                rtc::ArrayView<const float> y,
+                rtc::ArrayView<float> z) {
+    RTC_DCHECK_EQ(z.size(), x.size());
+    RTC_DCHECK_EQ(z.size(), y.size());
+    switch (optimization_) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+      case Aec3Optimization::kSse2: {
+        const int x_size = static_cast<int>(x.size());
+        const int vector_limit = x_size >> 2;
+
+        int j = 0;
+        for (; j < vector_limit * 4; j += 4) {
+          const __m128 x_j = _mm_loadu_ps(&x[j]);
+          const __m128 y_j = _mm_loadu_ps(&y[j]);
+          const __m128 z_j = _mm_mul_ps(x_j, y_j);
+          _mm_storeu_ps(&z[j], z_j);
+        }
+
+        for (; j < x_size; ++j) {
+          z[j] = x[j] * y[j];
+        }
+      } break;
+      case Aec3Optimization::kAvx2:
+        MultiplyAVX2(x, y, z);
+        break;
+#endif
+#if defined(WEBRTC_HAS_NEON)
+      case Aec3Optimization::kNeon: {
+        const int x_size = static_cast<int>(x.size());
+        const int vector_limit = x_size >> 2;
+
+        int j = 0;
+        for (; j < vector_limit * 4; j += 4) {
+          const float32x4_t x_j = vld1q_f32(&x[j]);
+          const float32x4_t y_j = vld1q_f32(&y[j]);
+          const float32x4_t z_j = vmulq_f32(x_j, y_j);
+          vst1q_f32(&z[j], z_j);
+        }
+
+        for (; j < x_size; ++j) {
+          z[j] = x[j] * y[j];
+        }
+      } break;
+#endif
+      default:
+        std::transform(x.begin(), x.end(), y.begin(), z.begin(),
+                       std::multiplies<float>());
+    }
+  }
+
+  // Elementwise vector accumulation z += x.
+  void AccumulateAVX2(rtc::ArrayView<const float> x, rtc::ArrayView<float> z);
+  void Accumulate(rtc::ArrayView<const float> x, rtc::ArrayView<float> z) {
+    RTC_DCHECK_EQ(z.size(), x.size());
+    switch (optimization_) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+      case Aec3Optimization::kSse2: {
+        const int x_size = static_cast<int>(x.size());
+        const int vector_limit = x_size >> 2;
+
+        int j = 0;
+        for (; j < vector_limit * 4; j += 4) {
+          const __m128 x_j = _mm_loadu_ps(&x[j]);
+          __m128 z_j = _mm_loadu_ps(&z[j]);
+          z_j = _mm_add_ps(x_j, z_j);
+          _mm_storeu_ps(&z[j], z_j);
+        }
+
+        for (; j < x_size; ++j) {
+          z[j] += x[j];
+        }
+      } break;
+      case Aec3Optimization::kAvx2:
+        AccumulateAVX2(x, z);
+        break;
+#endif
+#if defined(WEBRTC_HAS_NEON)
+      case Aec3Optimization::kNeon: {
+        const int x_size = static_cast<int>(x.size());
+        const int vector_limit = x_size >> 2;
+
+        int j = 0;
+        for (; j < vector_limit * 4; j += 4) {
+          const float32x4_t x_j = vld1q_f32(&x[j]);
+          float32x4_t z_j = vld1q_f32(&z[j]);
+          z_j = vaddq_f32(z_j, x_j);
+          vst1q_f32(&z[j], z_j);
+        }
+
+        for (; j < x_size; ++j) {
+          z[j] += x[j];
+        }
+      } break;
+#endif
+      default:
+        std::transform(x.begin(), x.end(), z.begin(), z.begin(),
+                       std::plus<float>());
+    }
+  }
+
+ private:
+  Aec3Optimization optimization_;
+};
+
+}  // namespace aec3
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_VECTOR_MATH_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_avx2.cc b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_avx2.cc
new file mode 100644
index 0000000000..0b5f3c142e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_avx2.cc
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/vector_math.h"
+
+#include <immintrin.h>
+#include <math.h>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace aec3 {
+
+// Elementwise square root.
+void VectorMath::SqrtAVX2(rtc::ArrayView<float> x) {
+  const int x_size = static_cast<int>(x.size());
+  const int vector_limit = x_size >> 3;
+
+  int j = 0;
+  for (; j < vector_limit * 8; j += 8) {
+    __m256 g = _mm256_loadu_ps(&x[j]);
+    g = _mm256_sqrt_ps(g);
+    _mm256_storeu_ps(&x[j], g);
+  }
+
+  for (; j < x_size; ++j) {
+    x[j] = sqrtf(x[j]);
+  }
+}
+
+// Elementwise vector multiplication z = x * y.
+void VectorMath::MultiplyAVX2(rtc::ArrayView<const float> x,
+                              rtc::ArrayView<const float> y,
+                              rtc::ArrayView<float> z) {
+  RTC_DCHECK_EQ(z.size(), x.size());
+  RTC_DCHECK_EQ(z.size(), y.size());
+  const int x_size = static_cast<int>(x.size());
+  const int vector_limit = x_size >> 3;
+
+  int j = 0;
+  for (; j < vector_limit * 8; j += 8) {
+    const __m256 x_j = _mm256_loadu_ps(&x[j]);
+    const __m256 y_j = _mm256_loadu_ps(&y[j]);
+    const __m256 z_j = _mm256_mul_ps(x_j, y_j);
+    _mm256_storeu_ps(&z[j], z_j);
+  }
+
+  for (; j < x_size; ++j) {
+    z[j] = x[j] * y[j];
+  }
+}
+
+// Elementwise vector accumulation z += x.
+void VectorMath::AccumulateAVX2(rtc::ArrayView<const float> x,
+                                rtc::ArrayView<float> z) {
+  RTC_DCHECK_EQ(z.size(), x.size());
+  const int x_size = static_cast<int>(x.size());
+  const int vector_limit = x_size >> 3;
+
+  int j = 0;
+  for (; j < vector_limit * 8; j += 8) {
+    const __m256 x_j = _mm256_loadu_ps(&x[j]);
+    __m256 z_j = _mm256_loadu_ps(&z[j]);
+    z_j = _mm256_add_ps(x_j, z_j);
+    _mm256_storeu_ps(&z[j], z_j);
+  }
+
+  for (; j < x_size; ++j) {
+    z[j] += x[j];
+  }
+}
+
+}  // namespace aec3
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_gn/moz.build
new file mode 100644
index 0000000000..89ee0b6a81
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("vector_math_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_unittest.cc
new file mode 100644
index 0000000000..a9c37e33cf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec3/vector_math_unittest.cc
@@ -0,0 +1,209 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec3/vector_math.h"
+
+#include <math.h>
+
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+#if defined(WEBRTC_HAS_NEON)
+
+TEST(VectorMath, Sqrt) {
+  std::array<float, kFftLengthBy2Plus1> x;
+  std::array<float, kFftLengthBy2Plus1> z;
+  std::array<float, kFftLengthBy2Plus1> z_neon;
+
+  for (size_t k = 0; k < x.size(); ++k) {
+    x[k] = (2.f / 3.f) * k;
+  }
+
+  std::copy(x.begin(), x.end(), z.begin());
+  aec3::VectorMath(Aec3Optimization::kNone).Sqrt(z);
+  std::copy(x.begin(), x.end(), z_neon.begin());
+  aec3::VectorMath(Aec3Optimization::kNeon).Sqrt(z_neon);
+  for (size_t k = 0; k < z.size(); ++k) {
+    EXPECT_NEAR(z[k], z_neon[k], 0.0001f);
+    EXPECT_NEAR(sqrtf(x[k]), z_neon[k], 0.0001f);
+  }
+}
+
+TEST(VectorMath, Multiply) {
+  std::array<float, kFftLengthBy2Plus1> x;
+  std::array<float, kFftLengthBy2Plus1> y;
+  std::array<float, kFftLengthBy2Plus1> z;
+  std::array<float, kFftLengthBy2Plus1> z_neon;
+
+  for (size_t k = 0; k < x.size(); ++k) {
+    x[k] = k;
+    y[k] = (2.f / 3.f) * k;
+  }
+
+  aec3::VectorMath(Aec3Optimization::kNone).Multiply(x, y, z);
+  aec3::VectorMath(Aec3Optimization::kNeon).Multiply(x, y, z_neon);
+  for (size_t k = 0; k < z.size(); ++k) {
+    EXPECT_FLOAT_EQ(z[k], z_neon[k]);
+    EXPECT_FLOAT_EQ(x[k] * y[k], z_neon[k]);
+  }
+}
+
+TEST(VectorMath, Accumulate) {
+  std::array<float, kFftLengthBy2Plus1> x;
+  std::array<float, kFftLengthBy2Plus1> z;
+  std::array<float, kFftLengthBy2Plus1> z_neon;
+
+  for (size_t k = 0; k < x.size(); ++k) {
+    x[k] = k;
+    z[k] = z_neon[k] = 2.f * k;
+  }
+
+  aec3::VectorMath(Aec3Optimization::kNone).Accumulate(x, z);
+  aec3::VectorMath(Aec3Optimization::kNeon).Accumulate(x, z_neon);
+  for (size_t k = 0; k < z.size(); ++k) {
+    EXPECT_FLOAT_EQ(z[k], z_neon[k]);
+    EXPECT_FLOAT_EQ(x[k] + 2.f * x[k], z_neon[k]);
+  }
+}
+#endif
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+
+TEST(VectorMath, Sse2Sqrt) {
+  if (GetCPUInfo(kSSE2) != 0) {
+    std::array<float, kFftLengthBy2Plus1> x;
+    std::array<float, kFftLengthBy2Plus1> z;
+    std::array<float, kFftLengthBy2Plus1> z_sse2;
+
+    for (size_t k = 0; k < x.size(); ++k) {
+      x[k] = (2.f / 3.f) * k;
+    }
+
+    std::copy(x.begin(), x.end(), z.begin());
+    aec3::VectorMath(Aec3Optimization::kNone).Sqrt(z);
+    std::copy(x.begin(), x.end(), z_sse2.begin());
+    aec3::VectorMath(Aec3Optimization::kSse2).Sqrt(z_sse2);
+    EXPECT_EQ(z, z_sse2);
+    for (size_t k = 0; k < z.size(); ++k) {
+      EXPECT_FLOAT_EQ(z[k], z_sse2[k]);
+      EXPECT_FLOAT_EQ(sqrtf(x[k]), z_sse2[k]);
+    }
+  }
+}
+
+TEST(VectorMath, Avx2Sqrt) {
+  if (GetCPUInfo(kAVX2) != 0) {
+    std::array<float, kFftLengthBy2Plus1> x;
+    std::array<float, kFftLengthBy2Plus1> z;
+    std::array<float, kFftLengthBy2Plus1> z_avx2;
+
+    for (size_t k = 0; k < x.size(); ++k) {
+      x[k] = (2.f / 3.f) * k;
+    }
+
+    std::copy(x.begin(), x.end(), z.begin());
+    aec3::VectorMath(Aec3Optimization::kNone).Sqrt(z);
+    std::copy(x.begin(), x.end(), z_avx2.begin());
+    aec3::VectorMath(Aec3Optimization::kAvx2).Sqrt(z_avx2);
+    EXPECT_EQ(z, z_avx2);
+    for (size_t k = 0; k < z.size(); ++k) {
+      EXPECT_FLOAT_EQ(z[k], z_avx2[k]);
+      EXPECT_FLOAT_EQ(sqrtf(x[k]), z_avx2[k]);
+    }
+  }
+}
+
+TEST(VectorMath, Sse2Multiply) {
+  if (GetCPUInfo(kSSE2) != 0) {
+    std::array<float, kFftLengthBy2Plus1> x;
+    std::array<float, kFftLengthBy2Plus1> y;
+    std::array<float, kFftLengthBy2Plus1> z;
+    std::array<float, kFftLengthBy2Plus1> z_sse2;
+
+    for (size_t k = 0; k < x.size(); ++k) {
+      x[k] = k;
+      y[k] = (2.f / 3.f) * k;
+    }
+
+    aec3::VectorMath(Aec3Optimization::kNone).Multiply(x, y, z);
+    aec3::VectorMath(Aec3Optimization::kSse2).Multiply(x, y, z_sse2);
+    for (size_t k = 0; k < z.size(); ++k) {
+      EXPECT_FLOAT_EQ(z[k], z_sse2[k]);
+      EXPECT_FLOAT_EQ(x[k] * y[k], z_sse2[k]);
+    }
+  }
+}
+
+TEST(VectorMath, Avx2Multiply) {
+  if (GetCPUInfo(kAVX2) != 0) {
+    std::array<float, kFftLengthBy2Plus1> x;
+    std::array<float, kFftLengthBy2Plus1> y;
+    std::array<float, kFftLengthBy2Plus1> z;
+    std::array<float, kFftLengthBy2Plus1> z_avx2;
+
+    for (size_t k = 0; k < x.size(); ++k) {
+      x[k] = k;
+      y[k] = (2.f / 3.f) * k;
+    }
+
+    aec3::VectorMath(Aec3Optimization::kNone).Multiply(x, y, z);
+    aec3::VectorMath(Aec3Optimization::kAvx2).Multiply(x, y, z_avx2);
+    for (size_t k = 0; k < z.size(); ++k) {
+      EXPECT_FLOAT_EQ(z[k], z_avx2[k]);
+      EXPECT_FLOAT_EQ(x[k] * y[k], z_avx2[k]);
+    }
+  }
+}
+
+TEST(VectorMath, Sse2Accumulate) {
+  if (GetCPUInfo(kSSE2) != 0) {
+    std::array<float, kFftLengthBy2Plus1> x;
+    std::array<float, kFftLengthBy2Plus1> z;
+    std::array<float, kFftLengthBy2Plus1> z_sse2;
+
+    for (size_t k = 0; k < x.size(); ++k) {
+      x[k] = k;
+      z[k] = z_sse2[k] = 2.f * k;
+    }
+
+    aec3::VectorMath(Aec3Optimization::kNone).Accumulate(x, z);
+    aec3::VectorMath(Aec3Optimization::kSse2).Accumulate(x, z_sse2);
+    for (size_t k = 0; k < z.size(); ++k) {
+      EXPECT_FLOAT_EQ(z[k], z_sse2[k]);
+      EXPECT_FLOAT_EQ(x[k] + 2.f * x[k], z_sse2[k]);
+    }
+  }
+}
+
+TEST(VectorMath, Avx2Accumulate) {
+  if (GetCPUInfo(kAVX2) != 0) {
+    std::array<float, kFftLengthBy2Plus1> x;
+    std::array<float, kFftLengthBy2Plus1> z;
+    std::array<float, kFftLengthBy2Plus1> z_avx2;
+
+    for (size_t k = 0; k < x.size(); ++k) {
+      x[k] = k;
+      z[k] = z_avx2[k] = 2.f * k;
+    }
+
+    aec3::VectorMath(Aec3Optimization::kNone).Accumulate(x, z);
+    aec3::VectorMath(Aec3Optimization::kAvx2).Accumulate(x, z_avx2);
+    for (size_t k = 0; k < z.size(); ++k) {
+      EXPECT_FLOAT_EQ(z[k], z_avx2[k]);
+      EXPECT_FLOAT_EQ(x[k] + 2.f * x[k], z_avx2[k]);
+    }
+  }
+}
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn
new file mode 100644
index 0000000000..38d8776258
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn
@@ -0,0 +1,112 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")  # This contains def of 'rtc_enable_protobuf'
+
+rtc_source_set("aec_dump") {
+  visibility = [ "*" ]
+  sources = [ "aec_dump_factory.h" ]
+
+  deps = [
+    "..:aec_dump_interface",
+    "../../../rtc_base/system:file_wrapper",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("mock_aec_dump") {
+    testonly = true
+    sources = [
+      "mock_aec_dump.cc",
+      "mock_aec_dump.h",
+    ]
+
+    deps = [
+      "..:aec_dump_interface",
+      "..:audioproc_test_utils",
+      "../",
+      "../../../test:test_support",
+    ]
+  }
+
+  rtc_library("mock_aec_dump_unittests") {
+    testonly = true
+    configs += [ "..:apm_debug_dump" ]
+    sources = [ "aec_dump_integration_test.cc" ]
+
+    deps = [
+      ":mock_aec_dump",
+      "..:api",
+      "..:audioproc_test_utils",
+      "../",
+      "//testing/gtest",
+    ]
+  }
+}
+
+if (rtc_enable_protobuf) {
+  rtc_library("aec_dump_impl") {
+    sources = [
+      "aec_dump_impl.cc",
+      "aec_dump_impl.h",
+      "capture_stream_info.cc",
+      "capture_stream_info.h",
+    ]
+
+    deps = [
+      ":aec_dump",
+      "..:aec_dump_interface",
+      "../../../api/audio:audio_frame_api",
+      "../../../api/task_queue",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:ignore_wundef",
+      "../../../rtc_base:logging",
+      "../../../rtc_base:macromagic",
+      "../../../rtc_base:protobuf_utils",
+      "../../../rtc_base:race_checker",
+      "../../../rtc_base:rtc_event",
+      "../../../rtc_base:rtc_task_queue",
+      "../../../rtc_base/system:file_wrapper",
+      "../../../system_wrappers",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+
+    deps += [ "../:audioproc_debug_proto" ]
+  }
+
+  if (rtc_include_tests) {
+    rtc_library("aec_dump_unittests") {
+      testonly = true
+      defines = []
+      deps = [
+        ":aec_dump",
+        ":aec_dump_impl",
+        "..:audioproc_debug_proto",
+        "../",
+        "../../../rtc_base:task_queue_for_test",
+        "../../../test:fileutils",
+        "../../../test:test_support",
+        "//testing/gtest",
+      ]
+      sources = [ "aec_dump_unittest.cc" ]
+    }
+  }
+}
+
+rtc_library("null_aec_dump_factory") {
+  assert_no_deps = [ ":aec_dump_impl" ]
+  sources = [ "null_aec_dump_factory.cc" ]
+
+  deps = [
+    ":aec_dump",
+    "..:aec_dump_interface",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
new file mode 100644
index 0000000000..20718c3d7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace rtc {
+class TaskQueue;
+}  // namespace rtc
+
+namespace webrtc {
+
+class RTC_EXPORT AecDumpFactory {
+ public:
+  // The `worker_queue` may not be null and must outlive the created
+  // AecDump instance. `max_log_size_bytes == -1` means the log size
+  // will be unlimited. `handle` may not be null. The AecDump takes
+  // responsibility for `handle` and closes it in the destructor. A
+  // non-null return value indicates that the file has been
+  // sucessfully opened.
+  static std::unique_ptr<AecDump> Create(webrtc::FileWrapper file,
+                                         int64_t max_log_size_bytes,
+                                         rtc::TaskQueue* worker_queue);
+  static std::unique_ptr<AecDump> Create(absl::string_view file_name,
+                                         int64_t max_log_size_bytes,
+                                         rtc::TaskQueue* worker_queue);
+  static std::unique_ptr<AecDump> Create(FILE* handle,
+                                         int64_t max_log_size_bytes,
+                                         rtc::TaskQueue* worker_queue);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_gn/moz.build
new file mode 100644
index 0000000000..201cd58360
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_gn/moz.build
@@ -0,0 +1,209 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("aec_dump_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
new file mode 100644
index 0000000000..94c24048e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
@@ -0,0 +1,281 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec_dump/aec_dump_impl.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/task_queue.h"
+
+namespace webrtc {
+
+namespace {
+void CopyFromConfigToEvent(const webrtc::InternalAPMConfig& config,
+                           webrtc::audioproc::Config* pb_cfg) {
+  pb_cfg->set_aec_enabled(config.aec_enabled);
+  pb_cfg->set_aec_delay_agnostic_enabled(config.aec_delay_agnostic_enabled);
+  pb_cfg->set_aec_drift_compensation_enabled(
+      config.aec_drift_compensation_enabled);
+  pb_cfg->set_aec_extended_filter_enabled(config.aec_extended_filter_enabled);
+  pb_cfg->set_aec_suppression_level(config.aec_suppression_level);
+
+  pb_cfg->set_aecm_enabled(config.aecm_enabled);
+  pb_cfg->set_aecm_comfort_noise_enabled(config.aecm_comfort_noise_enabled);
+  pb_cfg->set_aecm_routing_mode(config.aecm_routing_mode);
+
+  pb_cfg->set_agc_enabled(config.agc_enabled);
+  pb_cfg->set_agc_mode(config.agc_mode);
+  pb_cfg->set_agc_limiter_enabled(config.agc_limiter_enabled);
+  pb_cfg->set_noise_robust_agc_enabled(config.noise_robust_agc_enabled);
+
+  pb_cfg->set_hpf_enabled(config.hpf_enabled);
+
+  pb_cfg->set_ns_enabled(config.ns_enabled);
+  pb_cfg->set_ns_level(config.ns_level);
+
+  pb_cfg->set_transient_suppression_enabled(
+      config.transient_suppression_enabled);
+
+  pb_cfg->set_pre_amplifier_enabled(config.pre_amplifier_enabled);
+  pb_cfg->set_pre_amplifier_fixed_gain_factor(
+      config.pre_amplifier_fixed_gain_factor);
+
+  pb_cfg->set_experiments_description(config.experiments_description);
+}
+
+}  // namespace
+
+AecDumpImpl::AecDumpImpl(FileWrapper debug_file,
+                         int64_t max_log_size_bytes,
+                         rtc::TaskQueue* worker_queue)
+    : debug_file_(std::move(debug_file)),
+      num_bytes_left_for_log_(max_log_size_bytes),
+      worker_queue_(worker_queue) {}
+
+AecDumpImpl::~AecDumpImpl() {
+  // Block until all tasks have finished running.
+  rtc::Event thread_sync_event;
+  worker_queue_->PostTask([&thread_sync_event] { thread_sync_event.Set(); });
+  // Wait until the event has been signaled with .Set(). By then all
+  // pending tasks will have finished.
+  thread_sync_event.Wait(rtc::Event::kForever);
+}
+
+void AecDumpImpl::WriteInitMessage(const ProcessingConfig& api_format,
+                                   int64_t time_now_ms) {
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::INIT);
+  audioproc::Init* msg = event->mutable_init();
+
+  msg->set_sample_rate(api_format.input_stream().sample_rate_hz());
+  msg->set_output_sample_rate(api_format.output_stream().sample_rate_hz());
+  msg->set_reverse_sample_rate(
+      api_format.reverse_input_stream().sample_rate_hz());
+  msg->set_reverse_output_sample_rate(
+      api_format.reverse_output_stream().sample_rate_hz());
+
+  msg->set_num_input_channels(
+      static_cast<int32_t>(api_format.input_stream().num_channels()));
+  msg->set_num_output_channels(
+      static_cast<int32_t>(api_format.output_stream().num_channels()));
+  msg->set_num_reverse_channels(
+      static_cast<int32_t>(api_format.reverse_input_stream().num_channels()));
+  msg->set_num_reverse_output_channels(
+      api_format.reverse_output_stream().num_channels());
+  msg->set_timestamp_ms(time_now_ms);
+
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::AddCaptureStreamInput(
+    const AudioFrameView<const float>& src) {
+  capture_stream_info_.AddInput(src);
+}
+
+void AecDumpImpl::AddCaptureStreamOutput(
+    const AudioFrameView<const float>& src) {
+  capture_stream_info_.AddOutput(src);
+}
+
+void AecDumpImpl::AddCaptureStreamInput(const int16_t* const data,
+                                        int num_channels,
+                                        int samples_per_channel) {
+  capture_stream_info_.AddInput(data, num_channels, samples_per_channel);
+}
+
+void AecDumpImpl::AddCaptureStreamOutput(const int16_t* const data,
+                                         int num_channels,
+                                         int samples_per_channel) {
+  capture_stream_info_.AddOutput(data, num_channels, samples_per_channel);
+}
+
+void AecDumpImpl::AddAudioProcessingState(const AudioProcessingState& state) {
+  capture_stream_info_.AddAudioProcessingState(state);
+}
+
+void AecDumpImpl::WriteCaptureStreamMessage() {
+  PostWriteToFileTask(capture_stream_info_.FetchEvent());
+}
+
+void AecDumpImpl::WriteRenderStreamMessage(const int16_t* const data,
+                                           int num_channels,
+                                           int samples_per_channel) {
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::REVERSE_STREAM);
+  audioproc::ReverseStream* msg = event->mutable_reverse_stream();
+  const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
+  msg->set_data(data, data_size);
+
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::WriteRenderStreamMessage(
+    const AudioFrameView<const float>& src) {
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::REVERSE_STREAM);
+
+  audioproc::ReverseStream* msg = event->mutable_reverse_stream();
+
+  for (int i = 0; i < src.num_channels(); ++i) {
+    const auto& channel_view = src.channel(i);
+    msg->add_channel(channel_view.begin(), sizeof(float) * channel_view.size());
+  }
+
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::WriteConfig(const InternalAPMConfig& config) {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::CONFIG);
+  CopyFromConfigToEvent(config, event->mutable_config());
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::WriteRuntimeSetting(
+    const AudioProcessing::RuntimeSetting& runtime_setting) {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::RUNTIME_SETTING);
+  audioproc::RuntimeSetting* setting = event->mutable_runtime_setting();
+  switch (runtime_setting.type()) {
+    case AudioProcessing::RuntimeSetting::Type::kCapturePreGain: {
+      float x;
+      runtime_setting.GetFloat(&x);
+      setting->set_capture_pre_gain(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kCapturePostGain: {
+      float x;
+      runtime_setting.GetFloat(&x);
+      setting->set_capture_post_gain(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::
+        kCustomRenderProcessingRuntimeSetting: {
+      float x;
+      runtime_setting.GetFloat(&x);
+      setting->set_custom_render_processing_setting(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kCaptureCompressionGain:
+      // Runtime AGC1 compression gain is ignored.
+      // TODO(http://bugs.webrtc.org/10432): Store compression gain in aecdumps.
+      break;
+    case AudioProcessing::RuntimeSetting::Type::kCaptureFixedPostGain: {
+      float x;
+      runtime_setting.GetFloat(&x);
+      setting->set_capture_fixed_post_gain(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kCaptureOutputUsed: {
+      bool x;
+      runtime_setting.GetBool(&x);
+      setting->set_capture_output_used(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kPlayoutVolumeChange: {
+      int x;
+      runtime_setting.GetInt(&x);
+      setting->set_playout_volume_change(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kPlayoutAudioDeviceChange: {
+      AudioProcessing::RuntimeSetting::PlayoutAudioDeviceInfo src;
+      runtime_setting.GetPlayoutAudioDeviceInfo(&src);
+      auto* dst = setting->mutable_playout_audio_device_change();
+      dst->set_id(src.id);
+      dst->set_max_volume(src.max_volume);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kNotSpecified:
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::PostWriteToFileTask(std::unique_ptr<audioproc::Event> event) {
+  RTC_DCHECK(event);
+  worker_queue_->PostTask([event = std::move(event), this] {
+    std::string event_string = event->SerializeAsString();
+    const size_t event_byte_size = event_string.size();
+
+    if (num_bytes_left_for_log_ >= 0) {
+      const int64_t next_message_size = sizeof(int32_t) + event_byte_size;
+      if (num_bytes_left_for_log_ < next_message_size) {
+        // Ensure that no further events are written, even if they're smaller
+        // than the current event.
+        num_bytes_left_for_log_ = 0;
+        return;
+      }
+      num_bytes_left_for_log_ -= next_message_size;
+    }
+
+    // Write message preceded by its size.
+    if (!debug_file_.Write(&event_byte_size, sizeof(int32_t))) {
+      RTC_DCHECK_NOTREACHED();
+    }
+    if (!debug_file_.Write(event_string.data(), event_string.size())) {
+      RTC_DCHECK_NOTREACHED();
+    }
+  });
+}
+
+std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
+                                                int64_t max_log_size_bytes,
+                                                rtc::TaskQueue* worker_queue) {
+  RTC_DCHECK(worker_queue);
+  if (!file.is_open())
+    return nullptr;
+
+  return std::make_unique<AecDumpImpl>(std::move(file), max_log_size_bytes,
+                                       worker_queue);
+}
+
+std::unique_ptr<AecDump> AecDumpFactory::Create(absl::string_view file_name,
+                                                int64_t max_log_size_bytes,
+                                                rtc::TaskQueue* worker_queue) {
+  return Create(FileWrapper::OpenWriteOnly(file_name), max_log_size_bytes,
+                worker_queue);
+}
+
+std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
+                                                int64_t max_log_size_bytes,
+                                                rtc::TaskQueue* worker_queue) {
+  return Create(FileWrapper(handle), max_log_size_bytes, worker_queue);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
new file mode 100644
index 0000000000..fac3712b7a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "modules/audio_processing/aec_dump/capture_stream_info.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread_annotations.h"
+
+// Files generated at build-time by the protobuf compiler.
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "modules/audio_processing/debug.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+
+// Task-queue based implementation of AecDump. It is thread safe by
+// relying on locks in TaskQueue.
+class AecDumpImpl : public AecDump {
+ public:
+  // `max_log_size_bytes` - maximum number of bytes to write to the debug file,
+  // `max_log_size_bytes == -1` means the log size will be unlimited.
+  AecDumpImpl(FileWrapper debug_file,
+              int64_t max_log_size_bytes,
+              rtc::TaskQueue* worker_queue);
+  AecDumpImpl(const AecDumpImpl&) = delete;
+  AecDumpImpl& operator=(const AecDumpImpl&) = delete;
+  ~AecDumpImpl() override;
+
+  void WriteInitMessage(const ProcessingConfig& api_format,
+                        int64_t time_now_ms) override;
+  void AddCaptureStreamInput(const AudioFrameView<const float>& src) override;
+  void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override;
+  void AddCaptureStreamInput(const int16_t* const data,
+                             int num_channels,
+                             int samples_per_channel) override;
+  void AddCaptureStreamOutput(const int16_t* const data,
+                              int num_channels,
+                              int samples_per_channel) override;
+  void AddAudioProcessingState(const AudioProcessingState& state) override;
+  void WriteCaptureStreamMessage() override;
+
+  void WriteRenderStreamMessage(const int16_t* const data,
+                                int num_channels,
+                                int samples_per_channel) override;
+  void WriteRenderStreamMessage(
+      const AudioFrameView<const float>& src) override;
+
+  void WriteConfig(const InternalAPMConfig& config) override;
+
+  void WriteRuntimeSetting(
+      const AudioProcessing::RuntimeSetting& runtime_setting) override;
+
+ private:
+  void PostWriteToFileTask(std::unique_ptr<audioproc::Event> event);
+
+  FileWrapper debug_file_;
+  int64_t num_bytes_left_for_log_ = 0;
+  rtc::RaceChecker race_checker_;
+  rtc::TaskQueue* worker_queue_;
+  CaptureStreamInfo capture_stream_info_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
new file mode 100644
index 0000000000..503135d87f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <array>
+#include <memory>
+#include <utility>
+
+#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
+#include "modules/audio_processing/audio_processing_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Exactly;
+using ::testing::StrictMock;
+
+namespace {
+rtc::scoped_refptr<webrtc::AudioProcessing> CreateAudioProcessing() {
+  rtc::scoped_refptr<webrtc::AudioProcessing> apm(
+      webrtc::AudioProcessingBuilderForTesting().Create());
+  RTC_DCHECK(apm);
+  return apm;
+}
+
+std::unique_ptr<webrtc::test::MockAecDump> CreateMockAecDump() {
+  auto mock_aec_dump =
+      std::make_unique<testing::StrictMock<webrtc::test::MockAecDump>>();
+  EXPECT_CALL(*mock_aec_dump.get(), WriteConfig(_)).Times(AtLeast(1));
+  EXPECT_CALL(*mock_aec_dump.get(), WriteInitMessage(_, _)).Times(AtLeast(1));
+  return std::unique_ptr<webrtc::test::MockAecDump>(std::move(mock_aec_dump));
+}
+
+}  // namespace
+
+TEST(AecDumpIntegration, ConfigurationAndInitShouldBeLogged) {
+  auto apm = CreateAudioProcessing();
+
+  apm->AttachAecDump(CreateMockAecDump());
+}
+
+TEST(AecDumpIntegration,
+     RenderStreamShouldBeLoggedOnceEveryProcessReverseStream) {
+  auto apm = CreateAudioProcessing();
+  auto mock_aec_dump = CreateMockAecDump();
+  constexpr int kNumChannels = 1;
+  constexpr int kNumSampleRateHz = 16000;
+  constexpr int kNumSamplesPerChannel = kNumSampleRateHz / 100;
+  std::array<int16_t, kNumSamplesPerChannel * kNumChannels> frame;
+  frame.fill(0.f);
+  webrtc::StreamConfig stream_config(kNumSampleRateHz, kNumChannels);
+
+  EXPECT_CALL(*mock_aec_dump.get(), WriteRenderStreamMessage(_, _, _))
+      .Times(Exactly(1));
+
+  apm->AttachAecDump(std::move(mock_aec_dump));
+  apm->ProcessReverseStream(frame.data(), stream_config, stream_config,
+                            frame.data());
+}
+
+TEST(AecDumpIntegration, CaptureStreamShouldBeLoggedOnceEveryProcessStream) {
+  auto apm = CreateAudioProcessing();
+  auto mock_aec_dump = CreateMockAecDump();
+  constexpr int kNumChannels = 1;
+  constexpr int kNumSampleRateHz = 16000;
+  constexpr int kNumSamplesPerChannel = kNumSampleRateHz / 100;
+  std::array<int16_t, kNumSamplesPerChannel * kNumChannels> frame;
+  frame.fill(0.f);
+
+  webrtc::StreamConfig stream_config(kNumSampleRateHz, kNumChannels);
+
+  EXPECT_CALL(*mock_aec_dump.get(), AddCaptureStreamInput(_, _, _))
+      .Times(AtLeast(1));
+
+  EXPECT_CALL(*mock_aec_dump.get(), AddCaptureStreamOutput(_, _, _))
+      .Times(Exactly(1));
+
+  EXPECT_CALL(*mock_aec_dump.get(), AddAudioProcessingState(_))
+      .Times(Exactly(1));
+
+  EXPECT_CALL(*mock_aec_dump.get(), WriteCaptureStreamMessage())
+      .Times(Exactly(1));
+
+  apm->AttachAecDump(std::move(mock_aec_dump));
+  apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
new file mode 100644
index 0000000000..62f896fe14
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <array>
+#include <utility>
+
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TEST(AecDumper, APICallsDoNotCrash) {
+  // Note order of initialization: Task queue has to be initialized
+  // before AecDump.
+  webrtc::TaskQueueForTest file_writer_queue("file_writer_queue");
+
+  const std::string filename =
+      webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump");
+
+  {
+    std::unique_ptr<webrtc::AecDump> aec_dump =
+        webrtc::AecDumpFactory::Create(filename, -1, &file_writer_queue);
+
+    constexpr int kNumChannels = 1;
+    constexpr int kNumSamplesPerChannel = 160;
+    std::array<int16_t, kNumSamplesPerChannel * kNumChannels> frame;
+    frame.fill(0.f);
+    aec_dump->WriteRenderStreamMessage(frame.data(), kNumChannels,
+                                       kNumSamplesPerChannel);
+
+    aec_dump->AddCaptureStreamInput(frame.data(), kNumChannels,
+                                    kNumSamplesPerChannel);
+    aec_dump->AddCaptureStreamOutput(frame.data(), kNumChannels,
+                                     kNumSamplesPerChannel);
+
+    aec_dump->WriteCaptureStreamMessage();
+
+    webrtc::InternalAPMConfig apm_config;
+    aec_dump->WriteConfig(apm_config);
+
+    webrtc::ProcessingConfig api_format;
+    constexpr int64_t kTimeNowMs = 123456789ll;
+    aec_dump->WriteInitMessage(api_format, kTimeNowMs);
+  }
+  // Remove file after the AecDump d-tor has finished.
+  ASSERT_EQ(0, remove(filename.c_str()));
+}
+
+TEST(AecDumper, WriteToFile) {
+  webrtc::TaskQueueForTest file_writer_queue("file_writer_queue");
+
+  const std::string filename =
+      webrtc::test::TempFilename(webrtc::test::OutputPath(), "aec_dump");
+
+  {
+    std::unique_ptr<webrtc::AecDump> aec_dump =
+        webrtc::AecDumpFactory::Create(filename, -1, &file_writer_queue);
+
+    constexpr int kNumChannels = 1;
+    constexpr int kNumSamplesPerChannel = 160;
+    std::array<int16_t, kNumSamplesPerChannel * kNumChannels> frame;
+    frame.fill(0.f);
+
+    aec_dump->WriteRenderStreamMessage(frame.data(), kNumChannels,
+                                       kNumSamplesPerChannel);
+  }
+
+  // Verify the file has been written after the AecDump d-tor has
+  // finished.
+  FILE* fid = fopen(filename.c_str(), "r");
+  ASSERT_TRUE(fid != NULL);
+
+  // Clean it up.
+  ASSERT_EQ(0, fclose(fid));
+  ASSERT_EQ(0, remove(filename.c_str()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.cc
new file mode 100644
index 0000000000..7d82a39729
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec_dump/capture_stream_info.h"
+
+namespace webrtc {
+
+void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
+  auto* stream = event_->mutable_stream();
+
+  for (int i = 0; i < src.num_channels(); ++i) {
+    const auto& channel_view = src.channel(i);
+    stream->add_input_channel(channel_view.begin(),
+                              sizeof(float) * channel_view.size());
+  }
+}
+
+void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
+  auto* stream = event_->mutable_stream();
+
+  for (int i = 0; i < src.num_channels(); ++i) {
+    const auto& channel_view = src.channel(i);
+    stream->add_output_channel(channel_view.begin(),
+                               sizeof(float) * channel_view.size());
+  }
+}
+
+void CaptureStreamInfo::AddInput(const int16_t* const data,
+                                 int num_channels,
+                                 int samples_per_channel) {
+  auto* stream = event_->mutable_stream();
+  const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
+  stream->set_input_data(data, data_size);
+}
+
+void CaptureStreamInfo::AddOutput(const int16_t* const data,
+                                  int num_channels,
+                                  int samples_per_channel) {
+  auto* stream = event_->mutable_stream();
+  const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
+  stream->set_output_data(data, data_size);
+}
+
+void CaptureStreamInfo::AddAudioProcessingState(
+    const AecDump::AudioProcessingState& state) {
+  auto* stream = event_->mutable_stream();
+  stream->set_delay(state.delay);
+  stream->set_drift(state.drift);
+  if (state.applied_input_volume.has_value()) {
+    stream->set_applied_input_volume(*state.applied_input_volume);
+  }
+  stream->set_keypress(state.keypress);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h
new file mode 100644
index 0000000000..0819bbcb23
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
+
+#include <memory>
+#include <utility>
+
+#include "modules/audio_processing/include/aec_dump.h"
+#include "rtc_base/ignore_wundef.h"
+
+// Files generated at build-time by the protobuf compiler.
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "modules/audio_processing/debug.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+
+class CaptureStreamInfo {
+ public:
+  CaptureStreamInfo() { CreateNewEvent(); }
+  CaptureStreamInfo(const CaptureStreamInfo&) = delete;
+  CaptureStreamInfo& operator=(const CaptureStreamInfo&) = delete;
+  ~CaptureStreamInfo() = default;
+
+  void AddInput(const AudioFrameView<const float>& src);
+  void AddOutput(const AudioFrameView<const float>& src);
+
+  void AddInput(const int16_t* const data,
+                int num_channels,
+                int samples_per_channel);
+  void AddOutput(const int16_t* const data,
+                 int num_channels,
+                 int samples_per_channel);
+
+  void AddAudioProcessingState(const AecDump::AudioProcessingState& state);
+
+  std::unique_ptr<audioproc::Event> FetchEvent() {
+    std::unique_ptr<audioproc::Event> result = std::move(event_);
+    CreateNewEvent();
+    return result;
+  }
+
+ private:
+  void CreateNewEvent() {
+    event_ = std::make_unique<audioproc::Event>();
+    event_->set_type(audioproc::Event::STREAM);
+  }
+  std::unique_ptr<audioproc::Event> event_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.cc
new file mode 100644
index 0000000000..fe35d81db9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.cc
@@ -0,0 +1,19 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
+
+namespace webrtc {
+
+namespace test {
+
+MockAecDump::MockAecDump() = default;
+MockAecDump::~MockAecDump() = default;
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.h
new file mode 100644
index 0000000000..b396739de4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
+
+#include <memory>
+
+#include "modules/audio_processing/include/aec_dump.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+namespace test {
+
+class MockAecDump : public AecDump {
+ public:
+  MockAecDump();
+  virtual ~MockAecDump();
+
+  MOCK_METHOD(void,
+              WriteInitMessage,
+              (const ProcessingConfig& api_format, int64_t time_now_ms),
+              (override));
+
+  MOCK_METHOD(void,
+              AddCaptureStreamInput,
+              (const AudioFrameView<const float>& src),
+              (override));
+  MOCK_METHOD(void,
+              AddCaptureStreamOutput,
+              (const AudioFrameView<const float>& src),
+              (override));
+  MOCK_METHOD(void,
+              AddCaptureStreamInput,
+              (const int16_t* const data,
+               int num_channels,
+               int samples_per_channel),
+              (override));
+  MOCK_METHOD(void,
+              AddCaptureStreamOutput,
+              (const int16_t* const data,
+               int num_channels,
+               int samples_per_channel),
+              (override));
+  MOCK_METHOD(void,
+              AddAudioProcessingState,
+              (const AudioProcessingState& state),
+              (override));
+  MOCK_METHOD(void, WriteCaptureStreamMessage, (), (override));
+
+  MOCK_METHOD(void,
+              WriteRenderStreamMessage,
+              (const int16_t* const data,
+               int num_channels,
+               int samples_per_channel),
+              (override));
+  MOCK_METHOD(void,
+              WriteRenderStreamMessage,
+              (const AudioFrameView<const float>& src),
+              (override));
+
+  MOCK_METHOD(void, WriteConfig, (const InternalAPMConfig& config), (override));
+
+  MOCK_METHOD(void,
+              WriteRuntimeSetting,
+              (const AudioProcessing::RuntimeSetting& config),
+              (override));
+};
+
+}  // namespace test
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
new file mode 100644
index 0000000000..9bd9745069
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "modules/audio_processing/include/aec_dump.h"
+
+namespace webrtc {
+
+std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
+                                                int64_t max_log_size_bytes,
+                                                rtc::TaskQueue* worker_queue) {
+  return nullptr;
+}
+
+std::unique_ptr<AecDump> AecDumpFactory::Create(absl::string_view file_name,
+                                                int64_t max_log_size_bytes,
+                                                rtc::TaskQueue* worker_queue) {
+  return nullptr;
+}
+
+std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
+                                                int64_t max_log_size_bytes,
+                                                rtc::TaskQueue* worker_queue) {
+  return nullptr;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory_gn/moz.build
new file mode 100644
index 0000000000..974b70b087
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("null_aec_dump_factory_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump_interface_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aec_dump_interface_gn/moz.build
new file mode 100644
index 0000000000..1c47bbd5cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump_interface_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/include/aec_dump.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("aec_dump_interface_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/aecm/BUILD.gn
new file mode 100644
index 0000000000..a77f04aba5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/BUILD.gn
@@ -0,0 +1,44 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("aecm_core") {
+  sources = [
+    "aecm_core.cc",
+    "aecm_core.h",
+    "aecm_defines.h",
+    "echo_control_mobile.cc",
+    "echo_control_mobile.h",
+  ]
+  deps = [
+    "../../../common_audio:common_audio_c",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:sanitizer",
+    "../../../system_wrappers",
+    "../utility:legacy_delay_estimator",
+  ]
+  cflags = []
+
+  if (rtc_build_with_neon) {
+    sources += [ "aecm_core_neon.cc" ]
+
+    if (target_cpu != "arm64") {
+      # Enable compilation for the NEON instruction set.
+      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+      cflags += [ "-mfpu=neon" ]
+    }
+  }
+
+  if (target_cpu == "mipsel") {
+    sources += [ "aecm_core_mips.cc" ]
+  } else {
+    sources += [ "aecm_core_c.cc" ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core.cc b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core.cc
new file mode 100644
index 0000000000..fbc3239732
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core.cc
@@ -0,0 +1,1125 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aecm/aecm_core.h"
+
+#include <stddef.h>
+#include <stdlib.h>
+#include <string.h>
+
+extern "C" {
+#include "common_audio/ring_buffer.h"
+#include "common_audio/signal_processing/include/real_fft.h"
+}
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_processing/aecm/echo_control_mobile.h"
+#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+#ifdef AEC_DEBUG
+FILE* dfile;
+FILE* testfile;
+#endif
+
+// Initialization table for echo channel in 8 kHz
+static const int16_t kChannelStored8kHz[PART_LEN1] = {
+    2040, 1815, 1590, 1498, 1405, 1395, 1385, 1418, 1451, 1506, 1562,
+    1644, 1726, 1804, 1882, 1918, 1953, 1982, 2010, 2025, 2040, 2034,
+    2027, 2021, 2014, 1997, 1980, 1925, 1869, 1800, 1732, 1683, 1635,
+    1604, 1572, 1545, 1517, 1481, 1444, 1405, 1367, 1331, 1294, 1270,
+    1245, 1239, 1233, 1247, 1260, 1282, 1303, 1338, 1373, 1407, 1441,
+    1470, 1499, 1524, 1549, 1565, 1582, 1601, 1621, 1649, 1676};
+
+// Initialization table for echo channel in 16 kHz
+static const int16_t kChannelStored16kHz[PART_LEN1] = {
+    2040, 1590, 1405, 1385, 1451, 1562, 1726, 1882, 1953, 2010, 2040,
+    2027, 2014, 1980, 1869, 1732, 1635, 1572, 1517, 1444, 1367, 1294,
+    1245, 1233, 1260, 1303, 1373, 1441, 1499, 1549, 1582, 1621, 1676,
+    1741, 1802, 1861, 1921, 1983, 2040, 2102, 2170, 2265, 2375, 2515,
+    2651, 2781, 2922, 3075, 3253, 3471, 3738, 3976, 4151, 4258, 4308,
+    4288, 4270, 4253, 4237, 4179, 4086, 3947, 3757, 3484, 3153};
+
+}  // namespace
+
+const int16_t WebRtcAecm_kCosTable[] = {
+    8192,  8190,  8187,  8180,  8172,  8160,  8147,  8130,  8112,  8091,  8067,
+    8041,  8012,  7982,  7948,  7912,  7874,  7834,  7791,  7745,  7697,  7647,
+    7595,  7540,  7483,  7424,  7362,  7299,  7233,  7164,  7094,  7021,  6947,
+    6870,  6791,  6710,  6627,  6542,  6455,  6366,  6275,  6182,  6087,  5991,
+    5892,  5792,  5690,  5586,  5481,  5374,  5265,  5155,  5043,  4930,  4815,
+    4698,  4580,  4461,  4341,  4219,  4096,  3971,  3845,  3719,  3591,  3462,
+    3331,  3200,  3068,  2935,  2801,  2667,  2531,  2395,  2258,  2120,  1981,
+    1842,  1703,  1563,  1422,  1281,  1140,  998,   856,   713,   571,   428,
+    285,   142,   0,     -142,  -285,  -428,  -571,  -713,  -856,  -998,  -1140,
+    -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395, -2531, -2667,
+    -2801, -2935, -3068, -3200, -3331, -3462, -3591, -3719, -3845, -3971, -4095,
+    -4219, -4341, -4461, -4580, -4698, -4815, -4930, -5043, -5155, -5265, -5374,
+    -5481, -5586, -5690, -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455,
+    -6542, -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233, -7299,
+    -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745, -7791, -7834, -7874,
+    -7912, -7948, -7982, -8012, -8041, -8067, -8091, -8112, -8130, -8147, -8160,
+    -8172, -8180, -8187, -8190, -8191, -8190, -8187, -8180, -8172, -8160, -8147,
+    -8130, -8112, -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
+    -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362, -7299, -7233,
+    -7164, -7094, -7021, -6947, -6870, -6791, -6710, -6627, -6542, -6455, -6366,
+    -6275, -6182, -6087, -5991, -5892, -5792, -5690, -5586, -5481, -5374, -5265,
+    -5155, -5043, -4930, -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971,
+    -3845, -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667, -2531,
+    -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422, -1281, -1140, -998,
+    -856,  -713,  -571,  -428,  -285,  -142,  0,     142,   285,   428,   571,
+    713,   856,   998,   1140,  1281,  1422,  1563,  1703,  1842,  1981,  2120,
+    2258,  2395,  2531,  2667,  2801,  2935,  3068,  3200,  3331,  3462,  3591,
+    3719,  3845,  3971,  4095,  4219,  4341,  4461,  4580,  4698,  4815,  4930,
+    5043,  5155,  5265,  5374,  5481,  5586,  5690,  5792,  5892,  5991,  6087,
+    6182,  6275,  6366,  6455,  6542,  6627,  6710,  6791,  6870,  6947,  7021,
+    7094,  7164,  7233,  7299,  7362,  7424,  7483,  7540,  7595,  7647,  7697,
+    7745,  7791,  7834,  7874,  7912,  7948,  7982,  8012,  8041,  8067,  8091,
+    8112,  8130,  8147,  8160,  8172,  8180,  8187,  8190};
+
+const int16_t WebRtcAecm_kSinTable[] = {
+    0,     142,   285,   428,   571,   713,   856,   998,   1140,  1281,  1422,
+    1563,  1703,  1842,  1981,  2120,  2258,  2395,  2531,  2667,  2801,  2935,
+    3068,  3200,  3331,  3462,  3591,  3719,  3845,  3971,  4095,  4219,  4341,
+    4461,  4580,  4698,  4815,  4930,  5043,  5155,  5265,  5374,  5481,  5586,
+    5690,  5792,  5892,  5991,  6087,  6182,  6275,  6366,  6455,  6542,  6627,
+    6710,  6791,  6870,  6947,  7021,  7094,  7164,  7233,  7299,  7362,  7424,
+    7483,  7540,  7595,  7647,  7697,  7745,  7791,  7834,  7874,  7912,  7948,
+    7982,  8012,  8041,  8067,  8091,  8112,  8130,  8147,  8160,  8172,  8180,
+    8187,  8190,  8191,  8190,  8187,  8180,  8172,  8160,  8147,  8130,  8112,
+    8091,  8067,  8041,  8012,  7982,  7948,  7912,  7874,  7834,  7791,  7745,
+    7697,  7647,  7595,  7540,  7483,  7424,  7362,  7299,  7233,  7164,  7094,
+    7021,  6947,  6870,  6791,  6710,  6627,  6542,  6455,  6366,  6275,  6182,
+    6087,  5991,  5892,  5792,  5690,  5586,  5481,  5374,  5265,  5155,  5043,
+    4930,  4815,  4698,  4580,  4461,  4341,  4219,  4096,  3971,  3845,  3719,
+    3591,  3462,  3331,  3200,  3068,  2935,  2801,  2667,  2531,  2395,  2258,
+    2120,  1981,  1842,  1703,  1563,  1422,  1281,  1140,  998,   856,   713,
+    571,   428,   285,   142,   0,     -142,  -285,  -428,  -571,  -713,  -856,
+    -998,  -1140, -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
+    -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591, -3719, -3845,
+    -3971, -4095, -4219, -4341, -4461, -4580, -4698, -4815, -4930, -5043, -5155,
+    -5265, -5374, -5481, -5586, -5690, -5792, -5892, -5991, -6087, -6182, -6275,
+    -6366, -6455, -6542, -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164,
+    -7233, -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745, -7791,
+    -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067, -8091, -8112, -8130,
+    -8147, -8160, -8172, -8180, -8187, -8190, -8191, -8190, -8187, -8180, -8172,
+    -8160, -8147, -8130, -8112, -8091, -8067, -8041, -8012, -7982, -7948, -7912,
+    -7874, -7834, -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
+    -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710, -6627, -6542,
+    -6455, -6366, -6275, -6182, -6087, -5991, -5892, -5792, -5690, -5586, -5481,
+    -5374, -5265, -5155, -5043, -4930, -4815, -4698, -4580, -4461, -4341, -4219,
+    -4096, -3971, -3845, -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801,
+    -2667, -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422, -1281,
+    -1140, -998,  -856,  -713,  -571,  -428,  -285,  -142};
+
+
+// Moves the pointer to the next entry and inserts `far_spectrum` and
+// corresponding Q-domain in its buffer.
+//
+// Inputs:
+//      - self          : Pointer to the delay estimation instance
+//      - far_spectrum  : Pointer to the far end spectrum
+//      - far_q         : Q-domain of far end spectrum
+//
+void WebRtcAecm_UpdateFarHistory(AecmCore* self,
+                                 uint16_t* far_spectrum,
+                                 int far_q) {
+  // Get new buffer position
+  self->far_history_pos++;
+  if (self->far_history_pos >= MAX_DELAY) {
+    self->far_history_pos = 0;
+  }
+  // Update Q-domain buffer
+  self->far_q_domains[self->far_history_pos] = far_q;
+  // Update far end spectrum buffer
+  memcpy(&(self->far_history[self->far_history_pos * PART_LEN1]), far_spectrum,
+         sizeof(uint16_t) * PART_LEN1);
+}
+
+// Returns a pointer to the far end spectrum aligned to current near end
+// spectrum. The function WebRtc_DelayEstimatorProcessFix(...) should have been
+// called before AlignedFarend(...). Otherwise, you get the pointer to the
+// previous frame. The memory is only valid until the next call of
+// WebRtc_DelayEstimatorProcessFix(...).
+//
+// Inputs:
+//      - self              : Pointer to the AECM instance.
+//      - delay             : Current delay estimate.
+//
+// Output:
+//      - far_q             : The Q-domain of the aligned far end spectrum
+//
+// Return value:
+//      - far_spectrum      : Pointer to the aligned far end spectrum
+//                            NULL - Error
+//
+const uint16_t* WebRtcAecm_AlignedFarend(AecmCore* self,
+                                         int* far_q,
+                                         int delay) {
+  int buffer_position = 0;
+  RTC_DCHECK(self);
+  buffer_position = self->far_history_pos - delay;
+
+  // Check buffer position
+  if (buffer_position < 0) {
+    buffer_position += MAX_DELAY;
+  }
+  // Get Q-domain
+  *far_q = self->far_q_domains[buffer_position];
+  // Return far end spectrum
+  return &(self->far_history[buffer_position * PART_LEN1]);
+}
+
+// Declare function pointers.
+CalcLinearEnergies WebRtcAecm_CalcLinearEnergies;
+StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
+ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
+
+AecmCore* WebRtcAecm_CreateCore() {
+  // Allocate zero-filled memory.
+  AecmCore* aecm = static_cast<AecmCore*>(calloc(1, sizeof(AecmCore)));
+
+  aecm->farFrameBuf =
+      WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
+  if (!aecm->farFrameBuf) {
+    WebRtcAecm_FreeCore(aecm);
+    return NULL;
+  }
+
+  aecm->nearNoisyFrameBuf =
+      WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
+  if (!aecm->nearNoisyFrameBuf) {
+    WebRtcAecm_FreeCore(aecm);
+    return NULL;
+  }
+
+  aecm->nearCleanFrameBuf =
+      WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
+  if (!aecm->nearCleanFrameBuf) {
+    WebRtcAecm_FreeCore(aecm);
+    return NULL;
+  }
+
+  aecm->outFrameBuf =
+      WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
+  if (!aecm->outFrameBuf) {
+    WebRtcAecm_FreeCore(aecm);
+    return NULL;
+  }
+
+  aecm->delay_estimator_farend =
+      WebRtc_CreateDelayEstimatorFarend(PART_LEN1, MAX_DELAY);
+  if (aecm->delay_estimator_farend == NULL) {
+    WebRtcAecm_FreeCore(aecm);
+    return NULL;
+  }
+  aecm->delay_estimator =
+      WebRtc_CreateDelayEstimator(aecm->delay_estimator_farend, 0);
+  if (aecm->delay_estimator == NULL) {
+    WebRtcAecm_FreeCore(aecm);
+    return NULL;
+  }
+  // TODO(bjornv): Explicitly disable robust delay validation until no
+  // performance regression has been established.  Then remove the line.
+  WebRtc_enable_robust_validation(aecm->delay_estimator, 0);
+
+  aecm->real_fft = WebRtcSpl_CreateRealFFT(PART_LEN_SHIFT);
+  if (aecm->real_fft == NULL) {
+    WebRtcAecm_FreeCore(aecm);
+    return NULL;
+  }
+
+  // Init some aecm pointers. 16 and 32 byte alignment is only necessary
+  // for Neon code currently.
+  aecm->xBuf = (int16_t*)(((uintptr_t)aecm->xBuf_buf + 31) & ~31);
+  aecm->dBufClean = (int16_t*)(((uintptr_t)aecm->dBufClean_buf + 31) & ~31);
+  aecm->dBufNoisy = (int16_t*)(((uintptr_t)aecm->dBufNoisy_buf + 31) & ~31);
+  aecm->outBuf = (int16_t*)(((uintptr_t)aecm->outBuf_buf + 15) & ~15);
+  aecm->channelStored =
+      (int16_t*)(((uintptr_t)aecm->channelStored_buf + 15) & ~15);
+  aecm->channelAdapt16 =
+      (int16_t*)(((uintptr_t)aecm->channelAdapt16_buf + 15) & ~15);
+  aecm->channelAdapt32 =
+      (int32_t*)(((uintptr_t)aecm->channelAdapt32_buf + 31) & ~31);
+
+  return aecm;
+}
+
+void WebRtcAecm_InitEchoPathCore(AecmCore* aecm, const int16_t* echo_path) {
+  int i = 0;
+
+  // Reset the stored channel
+  memcpy(aecm->channelStored, echo_path, sizeof(int16_t) * PART_LEN1);
+  // Reset the adapted channels
+  memcpy(aecm->channelAdapt16, echo_path, sizeof(int16_t) * PART_LEN1);
+  for (i = 0; i < PART_LEN1; i++) {
+    aecm->channelAdapt32[i] = (int32_t)aecm->channelAdapt16[i] << 16;
+  }
+
+  // Reset channel storing variables
+  aecm->mseAdaptOld = 1000;
+  aecm->mseStoredOld = 1000;
+  aecm->mseThreshold = WEBRTC_SPL_WORD32_MAX;
+  aecm->mseChannelCount = 0;
+}
+
+static void CalcLinearEnergiesC(AecmCore* aecm,
+                                const uint16_t* far_spectrum,
+                                int32_t* echo_est,
+                                uint32_t* far_energy,
+                                uint32_t* echo_energy_adapt,
+                                uint32_t* echo_energy_stored) {
+  int i;
+
+  // Get energy for the delayed far end signal and estimated
+  // echo using both stored and adapted channels.
+  for (i = 0; i < PART_LEN1; i++) {
+    echo_est[i] =
+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+    (*far_energy) += (uint32_t)(far_spectrum[i]);
+    *echo_energy_adapt += aecm->channelAdapt16[i] * far_spectrum[i];
+    (*echo_energy_stored) += (uint32_t)echo_est[i];
+  }
+}
+
+static void StoreAdaptiveChannelC(AecmCore* aecm,
+                                  const uint16_t* far_spectrum,
+                                  int32_t* echo_est) {
+  int i;
+
+  // During startup we store the channel every block.
+  memcpy(aecm->channelStored, aecm->channelAdapt16,
+         sizeof(int16_t) * PART_LEN1);
+  // Recalculate echo estimate
+  for (i = 0; i < PART_LEN; i += 4) {
+    echo_est[i] =
+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+    echo_est[i + 1] =
+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 1], far_spectrum[i + 1]);
+    echo_est[i + 2] =
+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 2], far_spectrum[i + 2]);
+    echo_est[i + 3] =
+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 3], far_spectrum[i + 3]);
+  }
+  echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+}
+
+static void ResetAdaptiveChannelC(AecmCore* aecm) {
+  int i;
+
+  // The stored channel has a significantly lower MSE than the adaptive one for
+  // two consecutive calculations. Reset the adaptive channel.
+  memcpy(aecm->channelAdapt16, aecm->channelStored,
+         sizeof(int16_t) * PART_LEN1);
+  // Restore the W32 channel
+  for (i = 0; i < PART_LEN; i += 4) {
+    aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
+    aecm->channelAdapt32[i + 1] = (int32_t)aecm->channelStored[i + 1] << 16;
+    aecm->channelAdapt32[i + 2] = (int32_t)aecm->channelStored[i + 2] << 16;
+    aecm->channelAdapt32[i + 3] = (int32_t)aecm->channelStored[i + 3] << 16;
+  }
+  aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
+}
+
+// Initialize function pointers for ARM Neon platform.
+#if defined(WEBRTC_HAS_NEON)
+static void WebRtcAecm_InitNeon(void) {
+  WebRtcAecm_StoreAdaptiveChannel = WebRtcAecm_StoreAdaptiveChannelNeon;
+  WebRtcAecm_ResetAdaptiveChannel = WebRtcAecm_ResetAdaptiveChannelNeon;
+  WebRtcAecm_CalcLinearEnergies = WebRtcAecm_CalcLinearEnergiesNeon;
+}
+#endif
+
+// Initialize function pointers for MIPS platform.
+#if defined(MIPS32_LE)
+static void WebRtcAecm_InitMips(void) {
+#if defined(MIPS_DSP_R1_LE)
+  WebRtcAecm_StoreAdaptiveChannel = WebRtcAecm_StoreAdaptiveChannel_mips;
+  WebRtcAecm_ResetAdaptiveChannel = WebRtcAecm_ResetAdaptiveChannel_mips;
+#endif
+  WebRtcAecm_CalcLinearEnergies = WebRtcAecm_CalcLinearEnergies_mips;
+}
+#endif
+
+// WebRtcAecm_InitCore(...)
+//
+// This function initializes the AECM instant created with
+// WebRtcAecm_CreateCore(...) Input:
+//      - aecm            : Pointer to the Echo Suppression instance
+//      - samplingFreq   : Sampling Frequency
+//
+// Output:
+//      - aecm            : Initialized instance
+//
+// Return value         :  0 - Ok
+//                        -1 - Error
+//
+int WebRtcAecm_InitCore(AecmCore* const aecm, int samplingFreq) {
+  int i = 0;
+  int32_t tmp32 = PART_LEN1 * PART_LEN1;
+  int16_t tmp16 = PART_LEN1;
+
+  if (samplingFreq != 8000 && samplingFreq != 16000) {
+    samplingFreq = 8000;
+    return -1;
+  }
+  // sanity check of sampling frequency
+  aecm->mult = (int16_t)samplingFreq / 8000;
+
+  aecm->farBufWritePos = 0;
+  aecm->farBufReadPos = 0;
+  aecm->knownDelay = 0;
+  aecm->lastKnownDelay = 0;
+
+  WebRtc_InitBuffer(aecm->farFrameBuf);
+  WebRtc_InitBuffer(aecm->nearNoisyFrameBuf);
+  WebRtc_InitBuffer(aecm->nearCleanFrameBuf);
+  WebRtc_InitBuffer(aecm->outFrameBuf);
+
+  memset(aecm->xBuf_buf, 0, sizeof(aecm->xBuf_buf));
+  memset(aecm->dBufClean_buf, 0, sizeof(aecm->dBufClean_buf));
+  memset(aecm->dBufNoisy_buf, 0, sizeof(aecm->dBufNoisy_buf));
+  memset(aecm->outBuf_buf, 0, sizeof(aecm->outBuf_buf));
+
+  aecm->seed = 666;
+  aecm->totCount = 0;
+
+  if (WebRtc_InitDelayEstimatorFarend(aecm->delay_estimator_farend) != 0) {
+    return -1;
+  }
+  if (WebRtc_InitDelayEstimator(aecm->delay_estimator) != 0) {
+    return -1;
+  }
+  // Set far end histories to zero
+  memset(aecm->far_history, 0, sizeof(uint16_t) * PART_LEN1 * MAX_DELAY);
+  memset(aecm->far_q_domains, 0, sizeof(int) * MAX_DELAY);
+  aecm->far_history_pos = MAX_DELAY;
+
+  aecm->nlpFlag = 1;
+  aecm->fixedDelay = -1;
+
+  aecm->dfaCleanQDomain = 0;
+  aecm->dfaCleanQDomainOld = 0;
+  aecm->dfaNoisyQDomain = 0;
+  aecm->dfaNoisyQDomainOld = 0;
+
+  memset(aecm->nearLogEnergy, 0, sizeof(aecm->nearLogEnergy));
+  aecm->farLogEnergy = 0;
+  memset(aecm->echoAdaptLogEnergy, 0, sizeof(aecm->echoAdaptLogEnergy));
+  memset(aecm->echoStoredLogEnergy, 0, sizeof(aecm->echoStoredLogEnergy));
+
+  // Initialize the echo channels with a stored shape.
+  if (samplingFreq == 8000) {
+    WebRtcAecm_InitEchoPathCore(aecm, kChannelStored8kHz);
+  } else {
+    WebRtcAecm_InitEchoPathCore(aecm, kChannelStored16kHz);
+  }
+
+  memset(aecm->echoFilt, 0, sizeof(aecm->echoFilt));
+  memset(aecm->nearFilt, 0, sizeof(aecm->nearFilt));
+  aecm->noiseEstCtr = 0;
+
+  aecm->cngMode = AecmTrue;
+
+  memset(aecm->noiseEstTooLowCtr, 0, sizeof(aecm->noiseEstTooLowCtr));
+  memset(aecm->noiseEstTooHighCtr, 0, sizeof(aecm->noiseEstTooHighCtr));
+  // Shape the initial noise level to an approximate pink noise.
+  for (i = 0; i < (PART_LEN1 >> 1) - 1; i++) {
+    aecm->noiseEst[i] = (tmp32 << 8);
+    tmp16--;
+    tmp32 -= (int32_t)((tmp16 << 1) + 1);
+  }
+  for (; i < PART_LEN1; i++) {
+    aecm->noiseEst[i] = (tmp32 << 8);
+  }
+
+  aecm->farEnergyMin = WEBRTC_SPL_WORD16_MAX;
+  aecm->farEnergyMax = WEBRTC_SPL_WORD16_MIN;
+  aecm->farEnergyMaxMin = 0;
+  aecm->farEnergyVAD = FAR_ENERGY_MIN;  // This prevents false speech detection
+                                        // at the beginning.
+  aecm->farEnergyMSE = 0;
+  aecm->currentVADValue = 0;
+  aecm->vadUpdateCount = 0;
+  aecm->firstVAD = 1;
+
+  aecm->startupState = 0;
+  aecm->supGain = SUPGAIN_DEFAULT;
+  aecm->supGainOld = SUPGAIN_DEFAULT;
+
+  aecm->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
+  aecm->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
+  aecm->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
+  aecm->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
+
+  // Assert a preprocessor definition at compile-time. It's an assumption
+  // used in assembly code, so check the assembly files before any change.
+  static_assert(PART_LEN % 16 == 0, "PART_LEN is not a multiple of 16");
+
+  // Initialize function pointers.
+  WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesC;
+  WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelC;
+  WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelC;
+
+#if defined(WEBRTC_HAS_NEON)
+  WebRtcAecm_InitNeon();
+#endif
+
+#if defined(MIPS32_LE)
+  WebRtcAecm_InitMips();
+#endif
+  return 0;
+}
+
+// TODO(bjornv): This function is currently not used. Add support for these
+// parameters from a higher level
+int WebRtcAecm_Control(AecmCore* aecm, int delay, int nlpFlag) {
+  aecm->nlpFlag = nlpFlag;
+  aecm->fixedDelay = delay;
+
+  return 0;
+}
+
+void WebRtcAecm_FreeCore(AecmCore* aecm) {
+  if (aecm == NULL) {
+    return;
+  }
+
+  WebRtc_FreeBuffer(aecm->farFrameBuf);
+  WebRtc_FreeBuffer(aecm->nearNoisyFrameBuf);
+  WebRtc_FreeBuffer(aecm->nearCleanFrameBuf);
+  WebRtc_FreeBuffer(aecm->outFrameBuf);
+
+  WebRtc_FreeDelayEstimator(aecm->delay_estimator);
+  WebRtc_FreeDelayEstimatorFarend(aecm->delay_estimator_farend);
+  WebRtcSpl_FreeRealFFT(aecm->real_fft);
+
+  free(aecm);
+}
+
+int WebRtcAecm_ProcessFrame(AecmCore* aecm,
+                            const int16_t* farend,
+                            const int16_t* nearendNoisy,
+                            const int16_t* nearendClean,
+                            int16_t* out) {
+  int16_t outBlock_buf[PART_LEN + 8];  // Align buffer to 8-byte boundary.
+  int16_t* outBlock = (int16_t*)(((uintptr_t)outBlock_buf + 15) & ~15);
+
+  int16_t farFrame[FRAME_LEN];
+  const int16_t* out_ptr = NULL;
+  int size = 0;
+
+  // Buffer the current frame.
+  // Fetch an older one corresponding to the delay.
+  WebRtcAecm_BufferFarFrame(aecm, farend, FRAME_LEN);
+  WebRtcAecm_FetchFarFrame(aecm, farFrame, FRAME_LEN, aecm->knownDelay);
+
+  // Buffer the synchronized far and near frames,
+  // to pass the smaller blocks individually.
+  WebRtc_WriteBuffer(aecm->farFrameBuf, farFrame, FRAME_LEN);
+  WebRtc_WriteBuffer(aecm->nearNoisyFrameBuf, nearendNoisy, FRAME_LEN);
+  if (nearendClean != NULL) {
+    WebRtc_WriteBuffer(aecm->nearCleanFrameBuf, nearendClean, FRAME_LEN);
+  }
+
+  // Process as many blocks as possible.
+  while (WebRtc_available_read(aecm->farFrameBuf) >= PART_LEN) {
+    int16_t far_block[PART_LEN];
+    const int16_t* far_block_ptr = NULL;
+    int16_t near_noisy_block[PART_LEN];
+    const int16_t* near_noisy_block_ptr = NULL;
+
+    WebRtc_ReadBuffer(aecm->farFrameBuf, (void**)&far_block_ptr, far_block,
+                      PART_LEN);
+    WebRtc_ReadBuffer(aecm->nearNoisyFrameBuf, (void**)&near_noisy_block_ptr,
+                      near_noisy_block, PART_LEN);
+    if (nearendClean != NULL) {
+      int16_t near_clean_block[PART_LEN];
+      const int16_t* near_clean_block_ptr = NULL;
+
+      WebRtc_ReadBuffer(aecm->nearCleanFrameBuf, (void**)&near_clean_block_ptr,
+                        near_clean_block, PART_LEN);
+      if (WebRtcAecm_ProcessBlock(aecm, far_block_ptr, near_noisy_block_ptr,
+                                  near_clean_block_ptr, outBlock) == -1) {
+        return -1;
+      }
+    } else {
+      if (WebRtcAecm_ProcessBlock(aecm, far_block_ptr, near_noisy_block_ptr,
+                                  NULL, outBlock) == -1) {
+        return -1;
+      }
+    }
+
+    WebRtc_WriteBuffer(aecm->outFrameBuf, outBlock, PART_LEN);
+  }
+
+  // Stuff the out buffer if we have less than a frame to output.
+  // This should only happen for the first frame.
+  size = (int)WebRtc_available_read(aecm->outFrameBuf);
+  if (size < FRAME_LEN) {
+    WebRtc_MoveReadPtr(aecm->outFrameBuf, size - FRAME_LEN);
+  }
+
+  // Obtain an output frame.
+  WebRtc_ReadBuffer(aecm->outFrameBuf, (void**)&out_ptr, out, FRAME_LEN);
+  if (out_ptr != out) {
+    // ReadBuffer() hasn't copied to `out` in this case.
+    memcpy(out, out_ptr, FRAME_LEN * sizeof(int16_t));
+  }
+
+  return 0;
+}
+
+// WebRtcAecm_AsymFilt(...)
+//
+// Performs asymmetric filtering.
+//
+// Inputs:
+//      - filtOld       : Previous filtered value.
+//      - inVal         : New input value.
+//      - stepSizePos   : Step size when we have a positive contribution.
+//      - stepSizeNeg   : Step size when we have a negative contribution.
+//
+// Output:
+//
+// Return: - Filtered value.
+//
+int16_t WebRtcAecm_AsymFilt(const int16_t filtOld,
+                            const int16_t inVal,
+                            const int16_t stepSizePos,
+                            const int16_t stepSizeNeg) {
+  int16_t retVal;
+
+  if ((filtOld == WEBRTC_SPL_WORD16_MAX) | (filtOld == WEBRTC_SPL_WORD16_MIN)) {
+    return inVal;
+  }
+  retVal = filtOld;
+  if (filtOld > inVal) {
+    retVal -= (filtOld - inVal) >> stepSizeNeg;
+  } else {
+    retVal += (inVal - filtOld) >> stepSizePos;
+  }
+
+  return retVal;
+}
+
+// ExtractFractionPart(a, zeros)
+//
+// returns the fraction part of `a`, with `zeros` number of leading zeros, as an
+// int16_t scaled to Q8. There is no sanity check of `a` in the sense that the
+// number of zeros match.
+static int16_t ExtractFractionPart(uint32_t a, int zeros) {
+  return (int16_t)(((a << zeros) & 0x7FFFFFFF) >> 23);
+}
+
+// Calculates and returns the log of `energy` in Q8. The input `energy` is
+// supposed to be in Q(`q_domain`).
+static int16_t LogOfEnergyInQ8(uint32_t energy, int q_domain) {
+  static const int16_t kLogLowValue = PART_LEN_SHIFT << 7;
+  int16_t log_energy_q8 = kLogLowValue;
+  if (energy > 0) {
+    int zeros = WebRtcSpl_NormU32(energy);
+    int16_t frac = ExtractFractionPart(energy, zeros);
+    // log2 of `energy` in Q8.
+    log_energy_q8 += ((31 - zeros) << 8) + frac - (q_domain << 8);
+  }
+  return log_energy_q8;
+}
+
+// WebRtcAecm_CalcEnergies(...)
+//
+// This function calculates the log of energies for nearend, farend and
+// estimated echoes. There is also an update of energy decision levels, i.e.
+// internal VAD.
+//
+//
+// @param  aecm         [i/o]   Handle of the AECM instance.
+// @param  far_spectrum [in]    Pointer to farend spectrum.
+// @param  far_q        [in]    Q-domain of farend spectrum.
+// @param  nearEner     [in]    Near end energy for current block in
+//                              Q(aecm->dfaQDomain).
+// @param  echoEst      [out]   Estimated echo in Q(xfa_q+RESOLUTION_CHANNEL16).
+//
+void WebRtcAecm_CalcEnergies(AecmCore* aecm,
+                             const uint16_t* far_spectrum,
+                             const int16_t far_q,
+                             const uint32_t nearEner,
+                             int32_t* echoEst) {
+  // Local variables
+  uint32_t tmpAdapt = 0;
+  uint32_t tmpStored = 0;
+  uint32_t tmpFar = 0;
+
+  int i;
+
+  int16_t tmp16;
+  int16_t increase_max_shifts = 4;
+  int16_t decrease_max_shifts = 11;
+  int16_t increase_min_shifts = 11;
+  int16_t decrease_min_shifts = 3;
+
+  // Get log of near end energy and store in buffer
+
+  // Shift buffer
+  memmove(aecm->nearLogEnergy + 1, aecm->nearLogEnergy,
+          sizeof(int16_t) * (MAX_BUF_LEN - 1));
+
+  // Logarithm of integrated magnitude spectrum (nearEner)
+  aecm->nearLogEnergy[0] = LogOfEnergyInQ8(nearEner, aecm->dfaNoisyQDomain);
+
+  WebRtcAecm_CalcLinearEnergies(aecm, far_spectrum, echoEst, &tmpFar, &tmpAdapt,
+                                &tmpStored);
+
+  // Shift buffers
+  memmove(aecm->echoAdaptLogEnergy + 1, aecm->echoAdaptLogEnergy,
+          sizeof(int16_t) * (MAX_BUF_LEN - 1));
+  memmove(aecm->echoStoredLogEnergy + 1, aecm->echoStoredLogEnergy,
+          sizeof(int16_t) * (MAX_BUF_LEN - 1));
+
+  // Logarithm of delayed far end energy
+  aecm->farLogEnergy = LogOfEnergyInQ8(tmpFar, far_q);
+
+  // Logarithm of estimated echo energy through adapted channel
+  aecm->echoAdaptLogEnergy[0] =
+      LogOfEnergyInQ8(tmpAdapt, RESOLUTION_CHANNEL16 + far_q);
+
+  // Logarithm of estimated echo energy through stored channel
+  aecm->echoStoredLogEnergy[0] =
+      LogOfEnergyInQ8(tmpStored, RESOLUTION_CHANNEL16 + far_q);
+
+  // Update farend energy levels (min, max, vad, mse)
+  if (aecm->farLogEnergy > FAR_ENERGY_MIN) {
+    if (aecm->startupState == 0) {
+      increase_max_shifts = 2;
+      decrease_min_shifts = 2;
+      increase_min_shifts = 8;
+    }
+
+    aecm->farEnergyMin =
+        WebRtcAecm_AsymFilt(aecm->farEnergyMin, aecm->farLogEnergy,
+                            increase_min_shifts, decrease_min_shifts);
+    aecm->farEnergyMax =
+        WebRtcAecm_AsymFilt(aecm->farEnergyMax, aecm->farLogEnergy,
+                            increase_max_shifts, decrease_max_shifts);
+    aecm->farEnergyMaxMin = (aecm->farEnergyMax - aecm->farEnergyMin);
+
+    // Dynamic VAD region size
+    tmp16 = 2560 - aecm->farEnergyMin;
+    if (tmp16 > 0) {
+      tmp16 = (int16_t)((tmp16 * FAR_ENERGY_VAD_REGION) >> 9);
+    } else {
+      tmp16 = 0;
+    }
+    tmp16 += FAR_ENERGY_VAD_REGION;
+
+    if ((aecm->startupState == 0) | (aecm->vadUpdateCount > 1024)) {
+      // In startup phase or VAD update halted
+      aecm->farEnergyVAD = aecm->farEnergyMin + tmp16;
+    } else {
+      if (aecm->farEnergyVAD > aecm->farLogEnergy) {
+        aecm->farEnergyVAD +=
+            (aecm->farLogEnergy + tmp16 - aecm->farEnergyVAD) >> 6;
+        aecm->vadUpdateCount = 0;
+      } else {
+        aecm->vadUpdateCount++;
+      }
+    }
+    // Put MSE threshold higher than VAD
+    aecm->farEnergyMSE = aecm->farEnergyVAD + (1 << 8);
+  }
+
+  // Update VAD variables
+  if (aecm->farLogEnergy > aecm->farEnergyVAD) {
+    if ((aecm->startupState == 0) | (aecm->farEnergyMaxMin > FAR_ENERGY_DIFF)) {
+      // We are in startup or have significant dynamics in input speech level
+      aecm->currentVADValue = 1;
+    }
+  } else {
+    aecm->currentVADValue = 0;
+  }
+  if ((aecm->currentVADValue) && (aecm->firstVAD)) {
+    aecm->firstVAD = 0;
+    if (aecm->echoAdaptLogEnergy[0] > aecm->nearLogEnergy[0]) {
+      // The estimated echo has higher energy than the near end signal.
+      // This means that the initialization was too aggressive. Scale
+      // down by a factor 8
+      for (i = 0; i < PART_LEN1; i++) {
+        aecm->channelAdapt16[i] >>= 3;
+      }
+      // Compensate the adapted echo energy level accordingly.
+      aecm->echoAdaptLogEnergy[0] -= (3 << 8);
+      aecm->firstVAD = 1;
+    }
+  }
+}
+
+// WebRtcAecm_CalcStepSize(...)
+//
+// This function calculates the step size used in channel estimation
+//
+//
+// @param  aecm  [in]    Handle of the AECM instance.
+// @param  mu    [out]   (Return value) Stepsize in log2(), i.e. number of
+// shifts.
+//
+//
+int16_t WebRtcAecm_CalcStepSize(AecmCore* const aecm) {
+  int32_t tmp32;
+  int16_t tmp16;
+  int16_t mu = MU_MAX;
+
+  // Here we calculate the step size mu used in the
+  // following NLMS based Channel estimation algorithm
+  if (!aecm->currentVADValue) {
+    // Far end energy level too low, no channel update
+    mu = 0;
+  } else if (aecm->startupState > 0) {
+    if (aecm->farEnergyMin >= aecm->farEnergyMax) {
+      mu = MU_MIN;
+    } else {
+      tmp16 = (aecm->farLogEnergy - aecm->farEnergyMin);
+      tmp32 = tmp16 * MU_DIFF;
+      tmp32 = WebRtcSpl_DivW32W16(tmp32, aecm->farEnergyMaxMin);
+      mu = MU_MIN - 1 - (int16_t)(tmp32);
+      // The -1 is an alternative to rounding. This way we get a larger
+      // stepsize, so we in some sense compensate for truncation in NLMS
+    }
+    if (mu < MU_MAX) {
+      mu = MU_MAX;  // Equivalent with maximum step size of 2^-MU_MAX
+    }
+  }
+
+  return mu;
+}
+
+// WebRtcAecm_UpdateChannel(...)
+//
+// This function performs channel estimation. NLMS and decision on channel
+// storage.
+//
+//
+// @param  aecm         [i/o]   Handle of the AECM instance.
+// @param  far_spectrum [in]    Absolute value of the farend signal in Q(far_q)
+// @param  far_q        [in]    Q-domain of the farend signal
+// @param  dfa          [in]    Absolute value of the nearend signal
+// (Q[aecm->dfaQDomain])
+// @param  mu           [in]    NLMS step size.
+// @param  echoEst      [i/o]   Estimated echo in Q(far_q+RESOLUTION_CHANNEL16).
+//
+void WebRtcAecm_UpdateChannel(AecmCore* aecm,
+                              const uint16_t* far_spectrum,
+                              const int16_t far_q,
+                              const uint16_t* const dfa,
+                              const int16_t mu,
+                              int32_t* echoEst) {
+  uint32_t tmpU32no1, tmpU32no2;
+  int32_t tmp32no1, tmp32no2;
+  int32_t mseStored;
+  int32_t mseAdapt;
+
+  int i;
+
+  int16_t zerosFar, zerosNum, zerosCh, zerosDfa;
+  int16_t shiftChFar, shiftNum, shift2ResChan;
+  int16_t tmp16no1;
+  int16_t xfaQ, dfaQ;
+
+  // This is the channel estimation algorithm. It is base on NLMS but has a
+  // variable step length, which was calculated above.
+  if (mu) {
+    for (i = 0; i < PART_LEN1; i++) {
+      // Determine norm of channel and farend to make sure we don't get overflow
+      // in multiplication
+      zerosCh = WebRtcSpl_NormU32(aecm->channelAdapt32[i]);
+      zerosFar = WebRtcSpl_NormU32((uint32_t)far_spectrum[i]);
+      if (zerosCh + zerosFar > 31) {
+        // Multiplication is safe
+        tmpU32no1 =
+            WEBRTC_SPL_UMUL_32_16(aecm->channelAdapt32[i], far_spectrum[i]);
+        shiftChFar = 0;
+      } else {
+        // We need to shift down before multiplication
+        shiftChFar = 32 - zerosCh - zerosFar;
+        // If zerosCh == zerosFar == 0, shiftChFar is 32. A
+        // right shift of 32 is undefined. To avoid that, we
+        // do this check.
+        tmpU32no1 =
+            rtc::dchecked_cast<uint32_t>(
+                shiftChFar >= 32 ? 0 : aecm->channelAdapt32[i] >> shiftChFar) *
+            far_spectrum[i];
+      }
+      // Determine Q-domain of numerator
+      zerosNum = WebRtcSpl_NormU32(tmpU32no1);
+      if (dfa[i]) {
+        zerosDfa = WebRtcSpl_NormU32((uint32_t)dfa[i]);
+      } else {
+        zerosDfa = 32;
+      }
+      tmp16no1 = zerosDfa - 2 + aecm->dfaNoisyQDomain - RESOLUTION_CHANNEL32 -
+                 far_q + shiftChFar;
+      if (zerosNum > tmp16no1 + 1) {
+        xfaQ = tmp16no1;
+        dfaQ = zerosDfa - 2;
+      } else {
+        xfaQ = zerosNum - 2;
+        dfaQ = RESOLUTION_CHANNEL32 + far_q - aecm->dfaNoisyQDomain -
+               shiftChFar + xfaQ;
+      }
+      // Add in the same Q-domain
+      tmpU32no1 = WEBRTC_SPL_SHIFT_W32(tmpU32no1, xfaQ);
+      tmpU32no2 = WEBRTC_SPL_SHIFT_W32((uint32_t)dfa[i], dfaQ);
+      tmp32no1 = (int32_t)tmpU32no2 - (int32_t)tmpU32no1;
+      zerosNum = WebRtcSpl_NormW32(tmp32no1);
+      if ((tmp32no1) && (far_spectrum[i] > (CHANNEL_VAD << far_q))) {
+        //
+        // Update is needed
+        //
+        // This is what we would like to compute
+        //
+        // tmp32no1 = dfa[i] - (aecm->channelAdapt[i] * far_spectrum[i])
+        // tmp32norm = (i + 1)
+        // aecm->channelAdapt[i] += (2^mu) * tmp32no1
+        //                        / (tmp32norm * far_spectrum[i])
+        //
+
+        // Make sure we don't get overflow in multiplication.
+        if (zerosNum + zerosFar > 31) {
+          if (tmp32no1 > 0) {
+            tmp32no2 =
+                (int32_t)WEBRTC_SPL_UMUL_32_16(tmp32no1, far_spectrum[i]);
+          } else {
+            tmp32no2 =
+                -(int32_t)WEBRTC_SPL_UMUL_32_16(-tmp32no1, far_spectrum[i]);
+          }
+          shiftNum = 0;
+        } else {
+          shiftNum = 32 - (zerosNum + zerosFar);
+          if (tmp32no1 > 0) {
+            tmp32no2 = (tmp32no1 >> shiftNum) * far_spectrum[i];
+          } else {
+            tmp32no2 = -((-tmp32no1 >> shiftNum) * far_spectrum[i]);
+          }
+        }
+        // Normalize with respect to frequency bin
+        tmp32no2 = WebRtcSpl_DivW32W16(tmp32no2, i + 1);
+        // Make sure we are in the right Q-domain
+        shift2ResChan =
+            shiftNum + shiftChFar - xfaQ - mu - ((30 - zerosFar) << 1);
+        if (WebRtcSpl_NormW32(tmp32no2) < shift2ResChan) {
+          tmp32no2 = WEBRTC_SPL_WORD32_MAX;
+        } else {
+          tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, shift2ResChan);
+        }
+        aecm->channelAdapt32[i] =
+            WebRtcSpl_AddSatW32(aecm->channelAdapt32[i], tmp32no2);
+        if (aecm->channelAdapt32[i] < 0) {
+          // We can never have negative channel gain
+          aecm->channelAdapt32[i] = 0;
+        }
+        aecm->channelAdapt16[i] = (int16_t)(aecm->channelAdapt32[i] >> 16);
+      }
+    }
+  }
+  // END: Adaptive channel update
+
+  // Determine if we should store or restore the channel
+  if ((aecm->startupState == 0) & (aecm->currentVADValue)) {
+    // During startup we store the channel every block,
+    // and we recalculate echo estimate
+    WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
+  } else {
+    if (aecm->farLogEnergy < aecm->farEnergyMSE) {
+      aecm->mseChannelCount = 0;
+    } else {
+      aecm->mseChannelCount++;
+    }
+    // Enough data for validation. Store channel if we can.
+    if (aecm->mseChannelCount >= (MIN_MSE_COUNT + 10)) {
+      // We have enough data.
+      // Calculate MSE of "Adapt" and "Stored" versions.
+      // It is actually not MSE, but average absolute error.
+      mseStored = 0;
+      mseAdapt = 0;
+      for (i = 0; i < MIN_MSE_COUNT; i++) {
+        tmp32no1 = ((int32_t)aecm->echoStoredLogEnergy[i] -
+                    (int32_t)aecm->nearLogEnergy[i]);
+        tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
+        mseStored += tmp32no2;
+
+        tmp32no1 = ((int32_t)aecm->echoAdaptLogEnergy[i] -
+                    (int32_t)aecm->nearLogEnergy[i]);
+        tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
+        mseAdapt += tmp32no2;
+      }
+      if (((mseStored << MSE_RESOLUTION) < (MIN_MSE_DIFF * mseAdapt)) &
+          ((aecm->mseStoredOld << MSE_RESOLUTION) <
+           (MIN_MSE_DIFF * aecm->mseAdaptOld))) {
+        // The stored channel has a significantly lower MSE than the adaptive
+        // one for two consecutive calculations. Reset the adaptive channel.
+        WebRtcAecm_ResetAdaptiveChannel(aecm);
+      } else if (((MIN_MSE_DIFF * mseStored) > (mseAdapt << MSE_RESOLUTION)) &
+                 (mseAdapt < aecm->mseThreshold) &
+                 (aecm->mseAdaptOld < aecm->mseThreshold)) {
+        // The adaptive channel has a significantly lower MSE than the stored
+        // one. The MSE for the adaptive channel has also been low for two
+        // consecutive calculations. Store the adaptive channel.
+        WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
+
+        // Update threshold
+        if (aecm->mseThreshold == WEBRTC_SPL_WORD32_MAX) {
+          aecm->mseThreshold = (mseAdapt + aecm->mseAdaptOld);
+        } else {
+          int scaled_threshold = aecm->mseThreshold * 5 / 8;
+          aecm->mseThreshold += ((mseAdapt - scaled_threshold) * 205) >> 8;
+        }
+      }
+
+      // Reset counter
+      aecm->mseChannelCount = 0;
+
+      // Store the MSE values.
+      aecm->mseStoredOld = mseStored;
+      aecm->mseAdaptOld = mseAdapt;
+    }
+  }
+  // END: Determine if we should store or reset channel estimate.
+}
+
+// CalcSuppressionGain(...)
+//
+// This function calculates the suppression gain that is used in the Wiener
+// filter.
+//
+//
+// @param  aecm     [i/n]   Handle of the AECM instance.
+// @param  supGain  [out]   (Return value) Suppression gain with which to scale
+// the noise
+//                          level (Q14).
+//
+//
+int16_t WebRtcAecm_CalcSuppressionGain(AecmCore* const aecm) {
+  int32_t tmp32no1;
+
+  int16_t supGain = SUPGAIN_DEFAULT;
+  int16_t tmp16no1;
+  int16_t dE = 0;
+
+  // Determine suppression gain used in the Wiener filter. The gain is based on
+  // a mix of far end energy and echo estimation error. Adjust for the far end
+  // signal level. A low signal level indicates no far end signal, hence we set
+  // the suppression gain to 0
+  if (!aecm->currentVADValue) {
+    supGain = 0;
+  } else {
+    // Adjust for possible double talk. If we have large variations in
+    // estimation error we likely have double talk (or poor channel).
+    tmp16no1 = (aecm->nearLogEnergy[0] - aecm->echoStoredLogEnergy[0] -
+                ENERGY_DEV_OFFSET);
+    dE = WEBRTC_SPL_ABS_W16(tmp16no1);
+
+    if (dE < ENERGY_DEV_TOL) {
+      // Likely no double talk. The better estimation, the more we can suppress
+      // signal. Update counters
+      if (dE < SUPGAIN_EPC_DT) {
+        tmp32no1 = aecm->supGainErrParamDiffAB * dE;
+        tmp32no1 += (SUPGAIN_EPC_DT >> 1);
+        tmp16no1 = (int16_t)WebRtcSpl_DivW32W16(tmp32no1, SUPGAIN_EPC_DT);
+        supGain = aecm->supGainErrParamA - tmp16no1;
+      } else {
+        tmp32no1 = aecm->supGainErrParamDiffBD * (ENERGY_DEV_TOL - dE);
+        tmp32no1 += ((ENERGY_DEV_TOL - SUPGAIN_EPC_DT) >> 1);
+        tmp16no1 = (int16_t)WebRtcSpl_DivW32W16(
+            tmp32no1, (ENERGY_DEV_TOL - SUPGAIN_EPC_DT));
+        supGain = aecm->supGainErrParamD + tmp16no1;
+      }
+    } else {
+      // Likely in double talk. Use default value
+      supGain = aecm->supGainErrParamD;
+    }
+  }
+
+  if (supGain > aecm->supGainOld) {
+    tmp16no1 = supGain;
+  } else {
+    tmp16no1 = aecm->supGainOld;
+  }
+  aecm->supGainOld = supGain;
+  if (tmp16no1 < aecm->supGain) {
+    aecm->supGain += (int16_t)((tmp16no1 - aecm->supGain) >> 4);
+  } else {
+    aecm->supGain += (int16_t)((tmp16no1 - aecm->supGain) >> 4);
+  }
+
+  // END: Update suppression gain
+
+  return aecm->supGain;
+}
+
+void WebRtcAecm_BufferFarFrame(AecmCore* const aecm,
+                               const int16_t* const farend,
+                               const int farLen) {
+  int writeLen = farLen, writePos = 0;
+
+  // Check if the write position must be wrapped
+  while (aecm->farBufWritePos + writeLen > FAR_BUF_LEN) {
+    // Write to remaining buffer space before wrapping
+    writeLen = FAR_BUF_LEN - aecm->farBufWritePos;
+    memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
+           sizeof(int16_t) * writeLen);
+    aecm->farBufWritePos = 0;
+    writePos = writeLen;
+    writeLen = farLen - writeLen;
+  }
+
+  memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
+         sizeof(int16_t) * writeLen);
+  aecm->farBufWritePos += writeLen;
+}
+
+void WebRtcAecm_FetchFarFrame(AecmCore* const aecm,
+                              int16_t* const farend,
+                              const int farLen,
+                              const int knownDelay) {
+  int readLen = farLen;
+  int readPos = 0;
+  int delayChange = knownDelay - aecm->lastKnownDelay;
+
+  aecm->farBufReadPos -= delayChange;
+
+  // Check if delay forces a read position wrap
+  while (aecm->farBufReadPos < 0) {
+    aecm->farBufReadPos += FAR_BUF_LEN;
+  }
+  while (aecm->farBufReadPos > FAR_BUF_LEN - 1) {
+    aecm->farBufReadPos -= FAR_BUF_LEN;
+  }
+
+  aecm->lastKnownDelay = knownDelay;
+
+  // Check if read position must be wrapped
+  while (aecm->farBufReadPos + readLen > FAR_BUF_LEN) {
+    // Read from remaining buffer space before wrapping
+    readLen = FAR_BUF_LEN - aecm->farBufReadPos;
+    memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
+           sizeof(int16_t) * readLen);
+    aecm->farBufReadPos = 0;
+    readPos = readLen;
+    readLen = farLen - readLen;
+  }
+  memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
+         sizeof(int16_t) * readLen);
+  aecm->farBufReadPos += readLen;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core.h b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core.h
new file mode 100644
index 0000000000..3de49315c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core.h
@@ -0,0 +1,441 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Performs echo control (suppression) with fft routines in fixed-point.
+
+#ifndef MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
+#define MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
+
+extern "C" {
+#include "common_audio/ring_buffer.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+}
+#include "modules/audio_processing/aecm/aecm_defines.h"
+
+struct RealFFT;
+
+namespace webrtc {
+
+#ifdef _MSC_VER  // visual c++
+#define ALIGN8_BEG __declspec(align(8))
+#define ALIGN8_END
+#else  // gcc or icc
+#define ALIGN8_BEG
+#define ALIGN8_END __attribute__((aligned(8)))
+#endif
+
+typedef struct {
+  int16_t real;
+  int16_t imag;
+} ComplexInt16;
+
+typedef struct {
+  int farBufWritePos;
+  int farBufReadPos;
+  int knownDelay;
+  int lastKnownDelay;
+  int firstVAD;  // Parameter to control poorly initialized channels
+
+  RingBuffer* farFrameBuf;
+  RingBuffer* nearNoisyFrameBuf;
+  RingBuffer* nearCleanFrameBuf;
+  RingBuffer* outFrameBuf;
+
+  int16_t farBuf[FAR_BUF_LEN];
+
+  int16_t mult;
+  uint32_t seed;
+
+  // Delay estimation variables
+  void* delay_estimator_farend;
+  void* delay_estimator;
+  uint16_t currentDelay;
+  // Far end history variables
+  // TODO(bjornv): Replace `far_history` with ring_buffer.
+  uint16_t far_history[PART_LEN1 * MAX_DELAY];
+  int far_history_pos;
+  int far_q_domains[MAX_DELAY];
+
+  int16_t nlpFlag;
+  int16_t fixedDelay;
+
+  uint32_t totCount;
+
+  int16_t dfaCleanQDomain;
+  int16_t dfaCleanQDomainOld;
+  int16_t dfaNoisyQDomain;
+  int16_t dfaNoisyQDomainOld;
+
+  int16_t nearLogEnergy[MAX_BUF_LEN];
+  int16_t farLogEnergy;
+  int16_t echoAdaptLogEnergy[MAX_BUF_LEN];
+  int16_t echoStoredLogEnergy[MAX_BUF_LEN];
+
+  // The extra 16 or 32 bytes in the following buffers are for alignment based
+  // Neon code.
+  // It's designed this way since the current GCC compiler can't align a
+  // buffer in 16 or 32 byte boundaries properly.
+  int16_t channelStored_buf[PART_LEN1 + 8];
+  int16_t channelAdapt16_buf[PART_LEN1 + 8];
+  int32_t channelAdapt32_buf[PART_LEN1 + 8];
+  int16_t xBuf_buf[PART_LEN2 + 16];       // farend
+  int16_t dBufClean_buf[PART_LEN2 + 16];  // nearend
+  int16_t dBufNoisy_buf[PART_LEN2 + 16];  // nearend
+  int16_t outBuf_buf[PART_LEN + 8];
+
+  // Pointers to the above buffers
+  int16_t* channelStored;
+  int16_t* channelAdapt16;
+  int32_t* channelAdapt32;
+  int16_t* xBuf;
+  int16_t* dBufClean;
+  int16_t* dBufNoisy;
+  int16_t* outBuf;
+
+  int32_t echoFilt[PART_LEN1];
+  int16_t nearFilt[PART_LEN1];
+  int32_t noiseEst[PART_LEN1];
+  int noiseEstTooLowCtr[PART_LEN1];
+  int noiseEstTooHighCtr[PART_LEN1];
+  int16_t noiseEstCtr;
+  int16_t cngMode;
+
+  int32_t mseAdaptOld;
+  int32_t mseStoredOld;
+  int32_t mseThreshold;
+
+  int16_t farEnergyMin;
+  int16_t farEnergyMax;
+  int16_t farEnergyMaxMin;
+  int16_t farEnergyVAD;
+  int16_t farEnergyMSE;
+  int currentVADValue;
+  int16_t vadUpdateCount;
+
+  int16_t startupState;
+  int16_t mseChannelCount;
+  int16_t supGain;
+  int16_t supGainOld;
+
+  int16_t supGainErrParamA;
+  int16_t supGainErrParamD;
+  int16_t supGainErrParamDiffAB;
+  int16_t supGainErrParamDiffBD;
+
+  struct RealFFT* real_fft;
+
+#ifdef AEC_DEBUG
+  FILE* farFile;
+  FILE* nearFile;
+  FILE* outFile;
+#endif
+} AecmCore;
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_CreateCore()
+//
+// Allocates the memory needed by the AECM. The memory needs to be
+// initialized separately using the WebRtcAecm_InitCore() function.
+// Returns a pointer to the instance and a nullptr at failure.
+AecmCore* WebRtcAecm_CreateCore();
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_InitCore(...)
+//
+// This function initializes the AECM instant created with
+// WebRtcAecm_CreateCore()
+// Input:
+//      - aecm          : Pointer to the AECM instance
+//      - samplingFreq  : Sampling Frequency
+//
+// Output:
+//      - aecm          : Initialized instance
+//
+// Return value         :  0 - Ok
+//                        -1 - Error
+//
+int WebRtcAecm_InitCore(AecmCore* const aecm, int samplingFreq);
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_FreeCore(...)
+//
+// This function releases the memory allocated by WebRtcAecm_CreateCore()
+// Input:
+//      - aecm          : Pointer to the AECM instance
+//
+void WebRtcAecm_FreeCore(AecmCore* aecm);
+
+int WebRtcAecm_Control(AecmCore* aecm, int delay, int nlpFlag);
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_InitEchoPathCore(...)
+//
+// This function resets the echo channel adaptation with the specified channel.
+// Input:
+//      - aecm          : Pointer to the AECM instance
+//      - echo_path     : Pointer to the data that should initialize the echo
+//                        path
+//
+// Output:
+//      - aecm          : Initialized instance
+//
+void WebRtcAecm_InitEchoPathCore(AecmCore* aecm, const int16_t* echo_path);
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_ProcessFrame(...)
+//
+// This function processes frames and sends blocks to
+// WebRtcAecm_ProcessBlock(...)
+//
+// Inputs:
+//      - aecm          : Pointer to the AECM instance
+//      - farend        : In buffer containing one frame of echo signal
+//      - nearendNoisy  : In buffer containing one frame of nearend+echo signal
+//                        without NS
+//      - nearendClean  : In buffer containing one frame of nearend+echo signal
+//                        with NS
+//
+// Output:
+//      - out           : Out buffer, one frame of nearend signal          :
+//
+//
+int WebRtcAecm_ProcessFrame(AecmCore* aecm,
+                            const int16_t* farend,
+                            const int16_t* nearendNoisy,
+                            const int16_t* nearendClean,
+                            int16_t* out);
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_ProcessBlock(...)
+//
+// This function is called for every block within one frame
+// This function is called by WebRtcAecm_ProcessFrame(...)
+//
+// Inputs:
+//      - aecm          : Pointer to the AECM instance
+//      - farend        : In buffer containing one block of echo signal
+//      - nearendNoisy  : In buffer containing one frame of nearend+echo signal
+//                        without NS
+//      - nearendClean  : In buffer containing one frame of nearend+echo signal
+//                        with NS
+//
+// Output:
+//      - out           : Out buffer, one block of nearend signal          :
+//
+//
+int WebRtcAecm_ProcessBlock(AecmCore* aecm,
+                            const int16_t* farend,
+                            const int16_t* nearendNoisy,
+                            const int16_t* noisyClean,
+                            int16_t* out);
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_BufferFarFrame()
+//
+// Inserts a frame of data into farend buffer.
+//
+// Inputs:
+//      - aecm          : Pointer to the AECM instance
+//      - farend        : In buffer containing one frame of farend signal
+//      - farLen        : Length of frame
+//
+void WebRtcAecm_BufferFarFrame(AecmCore* const aecm,
+                               const int16_t* const farend,
+                               int farLen);
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_FetchFarFrame()
+//
+// Read the farend buffer to account for known delay
+//
+// Inputs:
+//      - aecm          : Pointer to the AECM instance
+//      - farend        : In buffer containing one frame of farend signal
+//      - farLen        : Length of frame
+//      - knownDelay    : known delay
+//
+void WebRtcAecm_FetchFarFrame(AecmCore* const aecm,
+                              int16_t* const farend,
+                              int farLen,
+                              int knownDelay);
+
+// All the functions below are intended to be private
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_UpdateFarHistory()
+//
+// Moves the pointer to the next entry and inserts `far_spectrum` and
+// corresponding Q-domain in its buffer.
+//
+// Inputs:
+//      - self          : Pointer to the delay estimation instance
+//      - far_spectrum  : Pointer to the far end spectrum
+//      - far_q         : Q-domain of far end spectrum
+//
+void WebRtcAecm_UpdateFarHistory(AecmCore* self,
+                                 uint16_t* far_spectrum,
+                                 int far_q);
+
+////////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_AlignedFarend()
+//
+// Returns a pointer to the far end spectrum aligned to current near end
+// spectrum. The function WebRtc_DelayEstimatorProcessFix(...) should have been
+// called before AlignedFarend(...). Otherwise, you get the pointer to the
+// previous frame. The memory is only valid until the next call of
+// WebRtc_DelayEstimatorProcessFix(...).
+//
+// Inputs:
+//      - self              : Pointer to the AECM instance.
+//      - delay             : Current delay estimate.
+//
+// Output:
+//      - far_q             : The Q-domain of the aligned far end spectrum
+//
+// Return value:
+//      - far_spectrum      : Pointer to the aligned far end spectrum
+//                            NULL - Error
+//
+const uint16_t* WebRtcAecm_AlignedFarend(AecmCore* self, int* far_q, int delay);
+
+///////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_CalcSuppressionGain()
+//
+// This function calculates the suppression gain that is used in the
+// Wiener filter.
+//
+// Inputs:
+//      - aecm              : Pointer to the AECM instance.
+//
+// Return value:
+//      - supGain           : Suppression gain with which to scale the noise
+//                            level (Q14).
+//
+int16_t WebRtcAecm_CalcSuppressionGain(AecmCore* const aecm);
+
+///////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_CalcEnergies()
+//
+// This function calculates the log of energies for nearend, farend and
+// estimated echoes. There is also an update of energy decision levels,
+// i.e. internal VAD.
+//
+// Inputs:
+//      - aecm              : Pointer to the AECM instance.
+//      - far_spectrum      : Pointer to farend spectrum.
+//      - far_q             : Q-domain of farend spectrum.
+//      - nearEner          : Near end energy for current block in
+//                            Q(aecm->dfaQDomain).
+//
+// Output:
+//     - echoEst            : Estimated echo in Q(xfa_q+RESOLUTION_CHANNEL16).
+//
+void WebRtcAecm_CalcEnergies(AecmCore* aecm,
+                             const uint16_t* far_spectrum,
+                             int16_t far_q,
+                             uint32_t nearEner,
+                             int32_t* echoEst);
+
+///////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_CalcStepSize()
+//
+// This function calculates the step size used in channel estimation
+//
+// Inputs:
+//      - aecm              : Pointer to the AECM instance.
+//
+// Return value:
+//      - mu                : Stepsize in log2(), i.e. number of shifts.
+//
+int16_t WebRtcAecm_CalcStepSize(AecmCore* const aecm);
+
+///////////////////////////////////////////////////////////////////////////////
+// WebRtcAecm_UpdateChannel(...)
+//
+// This function performs channel estimation.
+// NLMS and decision on channel storage.
+//
+// Inputs:
+//      - aecm              : Pointer to the AECM instance.
+//      - far_spectrum      : Absolute value of the farend signal in Q(far_q)
+//      - far_q             : Q-domain of the farend signal
+//      - dfa               : Absolute value of the nearend signal
+//                            (Q[aecm->dfaQDomain])
+//      - mu                : NLMS step size.
+// Input/Output:
+//      - echoEst           : Estimated echo in Q(far_q+RESOLUTION_CHANNEL16).
+//
+void WebRtcAecm_UpdateChannel(AecmCore* aecm,
+                              const uint16_t* far_spectrum,
+                              int16_t far_q,
+                              const uint16_t* const dfa,
+                              int16_t mu,
+                              int32_t* echoEst);
+
+extern const int16_t WebRtcAecm_kCosTable[];
+extern const int16_t WebRtcAecm_kSinTable[];
+
+///////////////////////////////////////////////////////////////////////////////
+// Some function pointers, for internal functions shared by ARM NEON and
+// generic C code.
+//
+typedef void (*CalcLinearEnergies)(AecmCore* aecm,
+                                   const uint16_t* far_spectrum,
+                                   int32_t* echoEst,
+                                   uint32_t* far_energy,
+                                   uint32_t* echo_energy_adapt,
+                                   uint32_t* echo_energy_stored);
+extern CalcLinearEnergies WebRtcAecm_CalcLinearEnergies;
+
+typedef void (*StoreAdaptiveChannel)(AecmCore* aecm,
+                                     const uint16_t* far_spectrum,
+                                     int32_t* echo_est);
+extern StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
+
+typedef void (*ResetAdaptiveChannel)(AecmCore* aecm);
+extern ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
+
+// For the above function pointers, functions for generic platforms are declared
+// and defined as static in file aecm_core.c, while those for ARM Neon platforms
+// are declared below and defined in file aecm_core_neon.c.
+#if defined(WEBRTC_HAS_NEON)
+void WebRtcAecm_CalcLinearEnergiesNeon(AecmCore* aecm,
+                                       const uint16_t* far_spectrum,
+                                       int32_t* echo_est,
+                                       uint32_t* far_energy,
+                                       uint32_t* echo_energy_adapt,
+                                       uint32_t* echo_energy_stored);
+
+void WebRtcAecm_StoreAdaptiveChannelNeon(AecmCore* aecm,
+                                         const uint16_t* far_spectrum,
+                                         int32_t* echo_est);
+
+void WebRtcAecm_ResetAdaptiveChannelNeon(AecmCore* aecm);
+#endif
+
+#if defined(MIPS32_LE)
+void WebRtcAecm_CalcLinearEnergies_mips(AecmCore* aecm,
+                                        const uint16_t* far_spectrum,
+                                        int32_t* echo_est,
+                                        uint32_t* far_energy,
+                                        uint32_t* echo_energy_adapt,
+                                        uint32_t* echo_energy_stored);
+#if defined(MIPS_DSP_R1_LE)
+void WebRtcAecm_StoreAdaptiveChannel_mips(AecmCore* aecm,
+                                          const uint16_t* far_spectrum,
+                                          int32_t* echo_est);
+
+void WebRtcAecm_ResetAdaptiveChannel_mips(AecmCore* aecm);
+#endif
+#endif
+
+}  // namespace webrtc
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc
new file mode 100644
index 0000000000..d363dd2cfd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc
@@ -0,0 +1,671 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdlib.h>
+
+#include "modules/audio_processing/aecm/aecm_core.h"
+
+extern "C" {
+#include "common_audio/ring_buffer.h"
+#include "common_audio/signal_processing/include/real_fft.h"
+}
+#include "modules/audio_processing/aecm/echo_control_mobile.h"
+#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
+extern "C" {
+#include "system_wrappers/include/cpu_features_wrapper.h"
+}
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/sanitizer.h"
+
+namespace webrtc {
+
+namespace {
+
+// Square root of Hanning window in Q14.
+static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
+    0,     399,   798,   1196,  1594,  1990,  2386,  2780,  3172,  3562,  3951,
+    4337,  4720,  5101,  5478,  5853,  6224,  6591,  6954,  7313,  7668,  8019,
+    8364,  8705,  9040,  9370,  9695,  10013, 10326, 10633, 10933, 11227, 11514,
+    11795, 12068, 12335, 12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189,
+    14384, 14571, 14749, 14918, 15079, 15231, 15373, 15506, 15631, 15746, 15851,
+    15947, 16034, 16111, 16179, 16237, 16286, 16325, 16354, 16373, 16384};
+
+#ifdef AECM_WITH_ABS_APPROX
+// Q15 alpha = 0.99439986968132  const Factor for magnitude approximation
+static const uint16_t kAlpha1 = 32584;
+// Q15 beta = 0.12967166976970   const Factor for magnitude approximation
+static const uint16_t kBeta1 = 4249;
+// Q15 alpha = 0.94234827210087  const Factor for magnitude approximation
+static const uint16_t kAlpha2 = 30879;
+// Q15 beta = 0.33787806009150   const Factor for magnitude approximation
+static const uint16_t kBeta2 = 11072;
+// Q15 alpha = 0.82247698684306  const Factor for magnitude approximation
+static const uint16_t kAlpha3 = 26951;
+// Q15 beta = 0.57762063060713   const Factor for magnitude approximation
+static const uint16_t kBeta3 = 18927;
+#endif
+
+static const int16_t kNoiseEstQDomain = 15;
+static const int16_t kNoiseEstIncCount = 5;
+
+static void ComfortNoise(AecmCore* aecm,
+                         const uint16_t* dfa,
+                         ComplexInt16* out,
+                         const int16_t* lambda) {
+  int16_t i;
+  int16_t tmp16;
+  int32_t tmp32;
+
+  int16_t randW16[PART_LEN];
+  int16_t uReal[PART_LEN1];
+  int16_t uImag[PART_LEN1];
+  int32_t outLShift32;
+  int16_t noiseRShift16[PART_LEN1];
+
+  int16_t shiftFromNearToNoise = kNoiseEstQDomain - aecm->dfaCleanQDomain;
+  int16_t minTrackShift;
+
+  RTC_DCHECK_GE(shiftFromNearToNoise, 0);
+  RTC_DCHECK_LT(shiftFromNearToNoise, 16);
+
+  if (aecm->noiseEstCtr < 100) {
+    // Track the minimum more quickly initially.
+    aecm->noiseEstCtr++;
+    minTrackShift = 6;
+  } else {
+    minTrackShift = 9;
+  }
+
+  // Estimate noise power.
+  for (i = 0; i < PART_LEN1; i++) {
+    // Shift to the noise domain.
+    tmp32 = (int32_t)dfa[i];
+    outLShift32 = tmp32 << shiftFromNearToNoise;
+
+    if (outLShift32 < aecm->noiseEst[i]) {
+      // Reset "too low" counter
+      aecm->noiseEstTooLowCtr[i] = 0;
+      // Track the minimum.
+      if (aecm->noiseEst[i] < (1 << minTrackShift)) {
+        // For small values, decrease noiseEst[i] every
+        // `kNoiseEstIncCount` block. The regular approach below can not
+        // go further down due to truncation.
+        aecm->noiseEstTooHighCtr[i]++;
+        if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount) {
+          aecm->noiseEst[i]--;
+          aecm->noiseEstTooHighCtr[i] = 0;  // Reset the counter
+        }
+      } else {
+        aecm->noiseEst[i] -=
+            ((aecm->noiseEst[i] - outLShift32) >> minTrackShift);
+      }
+    } else {
+      // Reset "too high" counter
+      aecm->noiseEstTooHighCtr[i] = 0;
+      // Ramp slowly upwards until we hit the minimum again.
+      if ((aecm->noiseEst[i] >> 19) > 0) {
+        // Avoid overflow.
+        // Multiplication with 2049 will cause wrap around. Scale
+        // down first and then multiply
+        aecm->noiseEst[i] >>= 11;
+        aecm->noiseEst[i] *= 2049;
+      } else if ((aecm->noiseEst[i] >> 11) > 0) {
+        // Large enough for relative increase
+        aecm->noiseEst[i] *= 2049;
+        aecm->noiseEst[i] >>= 11;
+      } else {
+        // Make incremental increases based on size every
+        // `kNoiseEstIncCount` block
+        aecm->noiseEstTooLowCtr[i]++;
+        if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount) {
+          aecm->noiseEst[i] += (aecm->noiseEst[i] >> 9) + 1;
+          aecm->noiseEstTooLowCtr[i] = 0;  // Reset counter
+        }
+      }
+    }
+  }
+
+  for (i = 0; i < PART_LEN1; i++) {
+    tmp32 = aecm->noiseEst[i] >> shiftFromNearToNoise;
+    if (tmp32 > 32767) {
+      tmp32 = 32767;
+      aecm->noiseEst[i] = tmp32 << shiftFromNearToNoise;
+    }
+    noiseRShift16[i] = (int16_t)tmp32;
+
+    tmp16 = ONE_Q14 - lambda[i];
+    noiseRShift16[i] = (int16_t)((tmp16 * noiseRShift16[i]) >> 14);
+  }
+
+  // Generate a uniform random array on [0 2^15-1].
+  WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
+
+  // Generate noise according to estimated energy.
+  uReal[0] = 0;  // Reject LF noise.
+  uImag[0] = 0;
+  for (i = 1; i < PART_LEN1; i++) {
+    // Get a random index for the cos and sin tables over [0 359].
+    tmp16 = (int16_t)((359 * randW16[i - 1]) >> 15);
+
+    // Tables are in Q13.
+    uReal[i] =
+        (int16_t)((noiseRShift16[i] * WebRtcAecm_kCosTable[tmp16]) >> 13);
+    uImag[i] =
+        (int16_t)((-noiseRShift16[i] * WebRtcAecm_kSinTable[tmp16]) >> 13);
+  }
+  uImag[PART_LEN] = 0;
+
+  for (i = 0; i < PART_LEN1; i++) {
+    out[i].real = WebRtcSpl_AddSatW16(out[i].real, uReal[i]);
+    out[i].imag = WebRtcSpl_AddSatW16(out[i].imag, uImag[i]);
+  }
+}
+
+static void WindowAndFFT(AecmCore* aecm,
+                         int16_t* fft,
+                         const int16_t* time_signal,
+                         ComplexInt16* freq_signal,
+                         int time_signal_scaling) {
+  int i = 0;
+
+  // FFT of signal
+  for (i = 0; i < PART_LEN; i++) {
+    // Window time domain signal and insert into real part of
+    // transformation array `fft`
+    int16_t scaled_time_signal = time_signal[i] * (1 << time_signal_scaling);
+    fft[i] = (int16_t)((scaled_time_signal * WebRtcAecm_kSqrtHanning[i]) >> 14);
+    scaled_time_signal = time_signal[i + PART_LEN] * (1 << time_signal_scaling);
+    fft[PART_LEN + i] = (int16_t)(
+        (scaled_time_signal * WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14);
+  }
+
+  // Do forward FFT, then take only the first PART_LEN complex samples,
+  // and change signs of the imaginary parts.
+  WebRtcSpl_RealForwardFFT(aecm->real_fft, fft, (int16_t*)freq_signal);
+  for (i = 0; i < PART_LEN; i++) {
+    freq_signal[i].imag = -freq_signal[i].imag;
+  }
+}
+
+static void InverseFFTAndWindow(AecmCore* aecm,
+                                int16_t* fft,
+                                ComplexInt16* efw,
+                                int16_t* output,
+                                const int16_t* nearendClean) {
+  int i, j, outCFFT;
+  int32_t tmp32no1;
+  // Reuse `efw` for the inverse FFT output after transferring
+  // the contents to `fft`.
+  int16_t* ifft_out = (int16_t*)efw;
+
+  // Synthesis
+  for (i = 1, j = 2; i < PART_LEN; i += 1, j += 2) {
+    fft[j] = efw[i].real;
+    fft[j + 1] = -efw[i].imag;
+  }
+  fft[0] = efw[0].real;
+  fft[1] = -efw[0].imag;
+
+  fft[PART_LEN2] = efw[PART_LEN].real;
+  fft[PART_LEN2 + 1] = -efw[PART_LEN].imag;
+
+  // Inverse FFT. Keep outCFFT to scale the samples in the next block.
+  outCFFT = WebRtcSpl_RealInverseFFT(aecm->real_fft, fft, ifft_out);
+  for (i = 0; i < PART_LEN; i++) {
+    ifft_out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
+        ifft_out[i], WebRtcAecm_kSqrtHanning[i], 14);
+    tmp32no1 = WEBRTC_SPL_SHIFT_W32((int32_t)ifft_out[i],
+                                    outCFFT - aecm->dfaCleanQDomain);
+    output[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
+                                        tmp32no1 + aecm->outBuf[i],
+                                        WEBRTC_SPL_WORD16_MIN);
+
+    tmp32no1 =
+        (ifft_out[PART_LEN + i] * WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14;
+    tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, outCFFT - aecm->dfaCleanQDomain);
+    aecm->outBuf[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, tmp32no1,
+                                              WEBRTC_SPL_WORD16_MIN);
+  }
+
+  // Copy the current block to the old position
+  // (aecm->outBuf is shifted elsewhere)
+  memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(int16_t) * PART_LEN);
+  memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN,
+         sizeof(int16_t) * PART_LEN);
+  if (nearendClean != NULL) {
+    memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN,
+           sizeof(int16_t) * PART_LEN);
+  }
+}
+
+// Transforms a time domain signal into the frequency domain, outputting the
+// complex valued signal, absolute value and sum of absolute values.
+//
+// time_signal          [in]    Pointer to time domain signal
+// freq_signal_real     [out]   Pointer to real part of frequency domain array
+// freq_signal_imag     [out]   Pointer to imaginary part of frequency domain
+//                              array
+// freq_signal_abs      [out]   Pointer to absolute value of frequency domain
+//                              array
+// freq_signal_sum_abs  [out]   Pointer to the sum of all absolute values in
+//                              the frequency domain array
+// return value                 The Q-domain of current frequency values
+//
+static int TimeToFrequencyDomain(AecmCore* aecm,
+                                 const int16_t* time_signal,
+                                 ComplexInt16* freq_signal,
+                                 uint16_t* freq_signal_abs,
+                                 uint32_t* freq_signal_sum_abs) {
+  int i = 0;
+  int time_signal_scaling = 0;
+
+  int32_t tmp32no1 = 0;
+  int32_t tmp32no2 = 0;
+
+  // In fft_buf, +16 for 32-byte alignment.
+  int16_t fft_buf[PART_LEN4 + 16];
+  int16_t* fft = (int16_t*)(((uintptr_t)fft_buf + 31) & ~31);
+
+  int16_t tmp16no1;
+#ifndef WEBRTC_ARCH_ARM_V7
+  int16_t tmp16no2;
+#endif
+#ifdef AECM_WITH_ABS_APPROX
+  int16_t max_value = 0;
+  int16_t min_value = 0;
+  uint16_t alpha = 0;
+  uint16_t beta = 0;
+#endif
+
+#ifdef AECM_DYNAMIC_Q
+  tmp16no1 = WebRtcSpl_MaxAbsValueW16(time_signal, PART_LEN2);
+  time_signal_scaling = WebRtcSpl_NormW16(tmp16no1);
+#endif
+
+  WindowAndFFT(aecm, fft, time_signal, freq_signal, time_signal_scaling);
+
+  // Extract imaginary and real part, calculate the magnitude for
+  // all frequency bins
+  freq_signal[0].imag = 0;
+  freq_signal[PART_LEN].imag = 0;
+  freq_signal_abs[0] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[0].real);
+  freq_signal_abs[PART_LEN] =
+      (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[PART_LEN].real);
+  (*freq_signal_sum_abs) =
+      (uint32_t)(freq_signal_abs[0]) + (uint32_t)(freq_signal_abs[PART_LEN]);
+
+  for (i = 1; i < PART_LEN; i++) {
+    if (freq_signal[i].real == 0) {
+      freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
+    } else if (freq_signal[i].imag == 0) {
+      freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].real);
+    } else {
+      // Approximation for magnitude of complex fft output
+      // magn = sqrt(real^2 + imag^2)
+      // magn ~= alpha * max(`imag`,`real`) + beta * min(`imag`,`real`)
+      //
+      // The parameters alpha and beta are stored in Q15
+
+#ifdef AECM_WITH_ABS_APPROX
+      tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
+      tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
+
+      if (tmp16no1 > tmp16no2) {
+        max_value = tmp16no1;
+        min_value = tmp16no2;
+      } else {
+        max_value = tmp16no2;
+        min_value = tmp16no1;
+      }
+
+      // Magnitude in Q(-6)
+      if ((max_value >> 2) > min_value) {
+        alpha = kAlpha1;
+        beta = kBeta1;
+      } else if ((max_value >> 1) > min_value) {
+        alpha = kAlpha2;
+        beta = kBeta2;
+      } else {
+        alpha = kAlpha3;
+        beta = kBeta3;
+      }
+      tmp16no1 = (int16_t)((max_value * alpha) >> 15);
+      tmp16no2 = (int16_t)((min_value * beta) >> 15);
+      freq_signal_abs[i] = (uint16_t)tmp16no1 + (uint16_t)tmp16no2;
+#else
+#ifdef WEBRTC_ARCH_ARM_V7
+      __asm __volatile(
+          "smulbb %[tmp32no1], %[real], %[real]\n\t"
+          "smlabb %[tmp32no2], %[imag], %[imag], %[tmp32no1]\n\t"
+          : [tmp32no1] "+&r"(tmp32no1), [tmp32no2] "=r"(tmp32no2)
+          : [real] "r"(freq_signal[i].real), [imag] "r"(freq_signal[i].imag));
+#else
+      tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
+      tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
+      tmp32no1 = tmp16no1 * tmp16no1;
+      tmp32no2 = tmp16no2 * tmp16no2;
+      tmp32no2 = WebRtcSpl_AddSatW32(tmp32no1, tmp32no2);
+#endif  // WEBRTC_ARCH_ARM_V7
+      tmp32no1 = WebRtcSpl_SqrtFloor(tmp32no2);
+
+      freq_signal_abs[i] = (uint16_t)tmp32no1;
+#endif  // AECM_WITH_ABS_APPROX
+    }
+    (*freq_signal_sum_abs) += (uint32_t)freq_signal_abs[i];
+  }
+
+  return time_signal_scaling;
+}
+
+}  // namespace
+
+int RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/8200
+    WebRtcAecm_ProcessBlock(AecmCore* aecm,
+                            const int16_t* farend,
+                            const int16_t* nearendNoisy,
+                            const int16_t* nearendClean,
+                            int16_t* output) {
+  int i;
+
+  uint32_t xfaSum;
+  uint32_t dfaNoisySum;
+  uint32_t dfaCleanSum;
+  uint32_t echoEst32Gained;
+  uint32_t tmpU32;
+
+  int32_t tmp32no1;
+
+  uint16_t xfa[PART_LEN1];
+  uint16_t dfaNoisy[PART_LEN1];
+  uint16_t dfaClean[PART_LEN1];
+  uint16_t* ptrDfaClean = dfaClean;
+  const uint16_t* far_spectrum_ptr = NULL;
+
+  // 32 byte aligned buffers (with +8 or +16).
+  // TODO(kma): define fft with ComplexInt16.
+  int16_t fft_buf[PART_LEN4 + 2 + 16];  // +2 to make a loop safe.
+  int32_t echoEst32_buf[PART_LEN1 + 8];
+  int32_t dfw_buf[PART_LEN2 + 8];
+  int32_t efw_buf[PART_LEN2 + 8];
+
+  int16_t* fft = (int16_t*)(((uintptr_t)fft_buf + 31) & ~31);
+  int32_t* echoEst32 = (int32_t*)(((uintptr_t)echoEst32_buf + 31) & ~31);
+  ComplexInt16* dfw = (ComplexInt16*)(((uintptr_t)dfw_buf + 31) & ~31);
+  ComplexInt16* efw = (ComplexInt16*)(((uintptr_t)efw_buf + 31) & ~31);
+
+  int16_t hnl[PART_LEN1];
+  int16_t numPosCoef = 0;
+  int16_t nlpGain = ONE_Q14;
+  int delay;
+  int16_t tmp16no1;
+  int16_t tmp16no2;
+  int16_t mu;
+  int16_t supGain;
+  int16_t zeros32, zeros16;
+  int16_t zerosDBufNoisy, zerosDBufClean, zerosXBuf;
+  int far_q;
+  int16_t resolutionDiff, qDomainDiff, dfa_clean_q_domain_diff;
+
+  const int kMinPrefBand = 4;
+  const int kMaxPrefBand = 24;
+  int32_t avgHnl32 = 0;
+
+  // Determine startup state. There are three states:
+  // (0) the first CONV_LEN blocks
+  // (1) another CONV_LEN blocks
+  // (2) the rest
+
+  if (aecm->startupState < 2) {
+    aecm->startupState =
+        (aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
+  }
+  // END: Determine startup state
+
+  // Buffer near and far end signals
+  memcpy(aecm->xBuf + PART_LEN, farend, sizeof(int16_t) * PART_LEN);
+  memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(int16_t) * PART_LEN);
+  if (nearendClean != NULL) {
+    memcpy(aecm->dBufClean + PART_LEN, nearendClean,
+           sizeof(int16_t) * PART_LEN);
+  }
+
+  // Transform far end signal from time domain to frequency domain.
+  far_q = TimeToFrequencyDomain(aecm, aecm->xBuf, dfw, xfa, &xfaSum);
+
+  // Transform noisy near end signal from time domain to frequency domain.
+  zerosDBufNoisy =
+      TimeToFrequencyDomain(aecm, aecm->dBufNoisy, dfw, dfaNoisy, &dfaNoisySum);
+  aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
+  aecm->dfaNoisyQDomain = (int16_t)zerosDBufNoisy;
+
+  if (nearendClean == NULL) {
+    ptrDfaClean = dfaNoisy;
+    aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld;
+    aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain;
+    dfaCleanSum = dfaNoisySum;
+  } else {
+    // Transform clean near end signal from time domain to frequency domain.
+    zerosDBufClean = TimeToFrequencyDomain(aecm, aecm->dBufClean, dfw, dfaClean,
+                                           &dfaCleanSum);
+    aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
+    aecm->dfaCleanQDomain = (int16_t)zerosDBufClean;
+  }
+
+  // Get the delay
+  // Save far-end history and estimate delay
+  WebRtcAecm_UpdateFarHistory(aecm, xfa, far_q);
+  if (WebRtc_AddFarSpectrumFix(aecm->delay_estimator_farend, xfa, PART_LEN1,
+                               far_q) == -1) {
+    return -1;
+  }
+  delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator, dfaNoisy,
+                                          PART_LEN1, zerosDBufNoisy);
+  if (delay == -1) {
+    return -1;
+  } else if (delay == -2) {
+    // If the delay is unknown, we assume zero.
+    // NOTE: this will have to be adjusted if we ever add lookahead.
+    delay = 0;
+  }
+
+  if (aecm->fixedDelay >= 0) {
+    // Use fixed delay
+    delay = aecm->fixedDelay;
+  }
+
+  // Get aligned far end spectrum
+  far_spectrum_ptr = WebRtcAecm_AlignedFarend(aecm, &far_q, delay);
+  zerosXBuf = (int16_t)far_q;
+  if (far_spectrum_ptr == NULL) {
+    return -1;
+  }
+
+  // Calculate log(energy) and update energy threshold levels
+  WebRtcAecm_CalcEnergies(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisySum,
+                          echoEst32);
+
+  // Calculate stepsize
+  mu = WebRtcAecm_CalcStepSize(aecm);
+
+  // Update counters
+  aecm->totCount++;
+
+  // This is the channel estimation algorithm.
+  // It is base on NLMS but has a variable step length,
+  // which was calculated above.
+  WebRtcAecm_UpdateChannel(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisy, mu,
+                           echoEst32);
+  supGain = WebRtcAecm_CalcSuppressionGain(aecm);
+
+  // Calculate Wiener filter hnl[]
+  for (i = 0; i < PART_LEN1; i++) {
+    // Far end signal through channel estimate in Q8
+    // How much can we shift right to preserve resolution
+    tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
+    aecm->echoFilt[i] +=
+        rtc::dchecked_cast<int32_t>((int64_t{tmp32no1} * 50) >> 8);
+
+    zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
+    zeros16 = WebRtcSpl_NormW16(supGain) + 1;
+    if (zeros32 + zeros16 > 16) {
+      // Multiplication is safe
+      // Result in
+      // Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+
+      //   aecm->xfaQDomainBuf[diff])
+      echoEst32Gained =
+          WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i], (uint16_t)supGain);
+      resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
+      resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
+    } else {
+      tmp16no1 = 17 - zeros32 - zeros16;
+      resolutionDiff =
+          14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
+      resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
+      if (zeros32 > tmp16no1) {
+        echoEst32Gained = WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i],
+                                                supGain >> tmp16no1);
+      } else {
+        // Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
+        echoEst32Gained = (aecm->echoFilt[i] >> tmp16no1) * supGain;
+      }
+    }
+
+    zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
+    RTC_DCHECK_GE(zeros16, 0);  // `zeros16` is a norm, hence non-negative.
+    dfa_clean_q_domain_diff = aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld;
+    if (zeros16 < dfa_clean_q_domain_diff && aecm->nearFilt[i]) {
+      tmp16no1 = aecm->nearFilt[i] * (1 << zeros16);
+      qDomainDiff = zeros16 - dfa_clean_q_domain_diff;
+      tmp16no2 = ptrDfaClean[i] >> -qDomainDiff;
+    } else {
+      tmp16no1 = dfa_clean_q_domain_diff < 0
+                     ? aecm->nearFilt[i] >> -dfa_clean_q_domain_diff
+                     : aecm->nearFilt[i] * (1 << dfa_clean_q_domain_diff);
+      qDomainDiff = 0;
+      tmp16no2 = ptrDfaClean[i];
+    }
+    tmp32no1 = (int32_t)(tmp16no2 - tmp16no1);
+    tmp16no2 = (int16_t)(tmp32no1 >> 4);
+    tmp16no2 += tmp16no1;
+    zeros16 = WebRtcSpl_NormW16(tmp16no2);
+    if ((tmp16no2) & (-qDomainDiff > zeros16)) {
+      aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
+    } else {
+      aecm->nearFilt[i] = qDomainDiff < 0 ? tmp16no2 * (1 << -qDomainDiff)
+                                          : tmp16no2 >> qDomainDiff;
+    }
+
+    // Wiener filter coefficients, resulting hnl in Q14
+    if (echoEst32Gained == 0) {
+      hnl[i] = ONE_Q14;
+    } else if (aecm->nearFilt[i] == 0) {
+      hnl[i] = 0;
+    } else {
+      // Multiply the suppression gain
+      // Rounding
+      echoEst32Gained += (uint32_t)(aecm->nearFilt[i] >> 1);
+      tmpU32 =
+          WebRtcSpl_DivU32U16(echoEst32Gained, (uint16_t)aecm->nearFilt[i]);
+
+      // Current resolution is
+      // Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN- max(0,17-zeros16- zeros32))
+      // Make sure we are in Q14
+      tmp32no1 = (int32_t)WEBRTC_SPL_SHIFT_W32(tmpU32, resolutionDiff);
+      if (tmp32no1 > ONE_Q14) {
+        hnl[i] = 0;
+      } else if (tmp32no1 < 0) {
+        hnl[i] = ONE_Q14;
+      } else {
+        // 1-echoEst/dfa
+        hnl[i] = ONE_Q14 - (int16_t)tmp32no1;
+        if (hnl[i] < 0) {
+          hnl[i] = 0;
+        }
+      }
+    }
+    if (hnl[i]) {
+      numPosCoef++;
+    }
+  }
+  // Only in wideband. Prevent the gain in upper band from being larger than
+  // in lower band.
+  if (aecm->mult == 2) {
+    // TODO(bjornv): Investigate if the scaling of hnl[i] below can cause
+    //               speech distortion in double-talk.
+    for (i = 0; i < PART_LEN1; i++) {
+      hnl[i] = (int16_t)((hnl[i] * hnl[i]) >> 14);
+    }
+
+    for (i = kMinPrefBand; i <= kMaxPrefBand; i++) {
+      avgHnl32 += (int32_t)hnl[i];
+    }
+    RTC_DCHECK_GT(kMaxPrefBand - kMinPrefBand + 1, 0);
+    avgHnl32 /= (kMaxPrefBand - kMinPrefBand + 1);
+
+    for (i = kMaxPrefBand; i < PART_LEN1; i++) {
+      if (hnl[i] > (int16_t)avgHnl32) {
+        hnl[i] = (int16_t)avgHnl32;
+      }
+    }
+  }
+
+  // Calculate NLP gain, result is in Q14
+  if (aecm->nlpFlag) {
+    for (i = 0; i < PART_LEN1; i++) {
+      // Truncate values close to zero and one.
+      if (hnl[i] > NLP_COMP_HIGH) {
+        hnl[i] = ONE_Q14;
+      } else if (hnl[i] < NLP_COMP_LOW) {
+        hnl[i] = 0;
+      }
+
+      // Remove outliers
+      if (numPosCoef < 3) {
+        nlpGain = 0;
+      } else {
+        nlpGain = ONE_Q14;
+      }
+
+      // NLP
+      if ((hnl[i] == ONE_Q14) && (nlpGain == ONE_Q14)) {
+        hnl[i] = ONE_Q14;
+      } else {
+        hnl[i] = (int16_t)((hnl[i] * nlpGain) >> 14);
+      }
+
+      // multiply with Wiener coefficients
+      efw[i].real = (int16_t)(
+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14));
+      efw[i].imag = (int16_t)(
+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14));
+    }
+  } else {
+    // multiply with Wiener coefficients
+    for (i = 0; i < PART_LEN1; i++) {
+      efw[i].real = (int16_t)(
+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14));
+      efw[i].imag = (int16_t)(
+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14));
+    }
+  }
+
+  if (aecm->cngMode == AecmTrue) {
+    ComfortNoise(aecm, ptrDfaClean, efw, hnl);
+  }
+
+  InverseFFTAndWindow(aecm, fft, efw, output, nearendClean);
+
+  return 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_gn/moz.build
new file mode 100644
index 0000000000..f0e41cd6bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_gn/moz.build
@@ -0,0 +1,293 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core.cc",
+    "/third_party/libwebrtc/modules/audio_processing/aecm/echo_control_mobile.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_neon.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_neon.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_mips.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "ppc64":
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "riscv64":
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc"
+    ]
+
+Library("aecm_core_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_mips.cc b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_mips.cc
new file mode 100644
index 0000000000..828aa6d2fb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_mips.cc
@@ -0,0 +1,1656 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aecm/aecm_core.h"
+#include "modules/audio_processing/aecm/echo_control_mobile.h"
+#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
+    0,     399,   798,   1196,  1594,  1990,  2386,  2780,  3172,  3562,  3951,
+    4337,  4720,  5101,  5478,  5853,  6224,  6591,  6954,  7313,  7668,  8019,
+    8364,  8705,  9040,  9370,  9695,  10013, 10326, 10633, 10933, 11227, 11514,
+    11795, 12068, 12335, 12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189,
+    14384, 14571, 14749, 14918, 15079, 15231, 15373, 15506, 15631, 15746, 15851,
+    15947, 16034, 16111, 16179, 16237, 16286, 16325, 16354, 16373, 16384};
+
+static const int16_t kNoiseEstQDomain = 15;
+static const int16_t kNoiseEstIncCount = 5;
+
+static int16_t coefTable[] = {
+    0,   4,   256, 260, 128, 132, 384, 388, 64,  68,  320, 324, 192, 196, 448,
+    452, 32,  36,  288, 292, 160, 164, 416, 420, 96,  100, 352, 356, 224, 228,
+    480, 484, 16,  20,  272, 276, 144, 148, 400, 404, 80,  84,  336, 340, 208,
+    212, 464, 468, 48,  52,  304, 308, 176, 180, 432, 436, 112, 116, 368, 372,
+    240, 244, 496, 500, 8,   12,  264, 268, 136, 140, 392, 396, 72,  76,  328,
+    332, 200, 204, 456, 460, 40,  44,  296, 300, 168, 172, 424, 428, 104, 108,
+    360, 364, 232, 236, 488, 492, 24,  28,  280, 284, 152, 156, 408, 412, 88,
+    92,  344, 348, 216, 220, 472, 476, 56,  60,  312, 316, 184, 188, 440, 444,
+    120, 124, 376, 380, 248, 252, 504, 508};
+
+static int16_t coefTable_ifft[] = {
+    0,   512, 256, 508, 128, 252, 384, 380, 64,  124, 320, 444, 192, 188, 448,
+    316, 32,  60,  288, 476, 160, 220, 416, 348, 96,  92,  352, 412, 224, 156,
+    480, 284, 16,  28,  272, 492, 144, 236, 400, 364, 80,  108, 336, 428, 208,
+    172, 464, 300, 48,  44,  304, 460, 176, 204, 432, 332, 112, 76,  368, 396,
+    240, 140, 496, 268, 8,   12,  264, 500, 136, 244, 392, 372, 72,  116, 328,
+    436, 200, 180, 456, 308, 40,  52,  296, 468, 168, 212, 424, 340, 104, 84,
+    360, 404, 232, 148, 488, 276, 24,  20,  280, 484, 152, 228, 408, 356, 88,
+    100, 344, 420, 216, 164, 472, 292, 56,  36,  312, 452, 184, 196, 440, 324,
+    120, 68,  376, 388, 248, 132, 504, 260};
+
+}  // namespace
+
+static void ComfortNoise(AecmCore* aecm,
+                         const uint16_t* dfa,
+                         ComplexInt16* out,
+                         const int16_t* lambda);
+
+static void WindowAndFFT(AecmCore* aecm,
+                         int16_t* fft,
+                         const int16_t* time_signal,
+                         ComplexInt16* freq_signal,
+                         int time_signal_scaling) {
+  int i, j;
+  int32_t tmp1, tmp2, tmp3, tmp4;
+  int16_t* pfrfi;
+  ComplexInt16* pfreq_signal;
+  int16_t f_coef, s_coef;
+  int32_t load_ptr, store_ptr1, store_ptr2, shift, shift1;
+  int32_t hann, hann1, coefs;
+
+  memset(fft, 0, sizeof(int16_t) * PART_LEN4);
+
+  // FFT of signal
+  __asm __volatile(
+      ".set        push                                                    \n\t"
+      ".set        noreorder                                               \n\t"
+      "addiu       %[shift],          %[time_signal_scaling], -14          \n\t"
+      "addiu       %[i],              $zero,                  64           \n\t"
+      "addiu       %[load_ptr],       %[time_signal],         0            \n\t"
+      "addiu       %[hann],           %[hanning],             0            \n\t"
+      "addiu       %[hann1],          %[hanning],             128          \n\t"
+      "addiu       %[coefs],          %[coefTable],           0            \n\t"
+      "bltz        %[shift],          2f                                   \n\t"
+      " negu       %[shift1],         %[shift]                             \n\t"
+      "1:                                                                   "
+      "\n\t"
+      "lh          %[tmp1],           0(%[load_ptr])                       \n\t"
+      "lh          %[tmp2],           0(%[hann])                           \n\t"
+      "lh          %[tmp3],           128(%[load_ptr])                     \n\t"
+      "lh          %[tmp4],           0(%[hann1])                          \n\t"
+      "addiu       %[i],              %[i],                   -1           \n\t"
+      "mul         %[tmp1],           %[tmp1],                %[tmp2]      \n\t"
+      "mul         %[tmp3],           %[tmp3],                %[tmp4]      \n\t"
+      "lh          %[f_coef],         0(%[coefs])                          \n\t"
+      "lh          %[s_coef],         2(%[coefs])                          \n\t"
+      "addiu       %[load_ptr],       %[load_ptr],            2            \n\t"
+      "addiu       %[hann],           %[hann],                2            \n\t"
+      "addiu       %[hann1],          %[hann1],               -2           \n\t"
+      "addu        %[store_ptr1],     %[fft],                 %[f_coef]    \n\t"
+      "addu        %[store_ptr2],     %[fft],                 %[s_coef]    \n\t"
+      "sllv        %[tmp1],           %[tmp1],                %[shift]     \n\t"
+      "sllv        %[tmp3],           %[tmp3],                %[shift]     \n\t"
+      "sh          %[tmp1],           0(%[store_ptr1])                     \n\t"
+      "sh          %[tmp3],           0(%[store_ptr2])                     \n\t"
+      "bgtz        %[i],              1b                                   \n\t"
+      " addiu      %[coefs],          %[coefs],               4            \n\t"
+      "b           3f                                                      \n\t"
+      " nop                                                                \n\t"
+      "2:                                                                   "
+      "\n\t"
+      "lh          %[tmp1],           0(%[load_ptr])                       \n\t"
+      "lh          %[tmp2],           0(%[hann])                           \n\t"
+      "lh          %[tmp3],           128(%[load_ptr])                     \n\t"
+      "lh          %[tmp4],           0(%[hann1])                          \n\t"
+      "addiu       %[i],              %[i],                   -1           \n\t"
+      "mul         %[tmp1],           %[tmp1],                %[tmp2]      \n\t"
+      "mul         %[tmp3],           %[tmp3],                %[tmp4]      \n\t"
+      "lh          %[f_coef],         0(%[coefs])                          \n\t"
+      "lh          %[s_coef],         2(%[coefs])                          \n\t"
+      "addiu       %[load_ptr],       %[load_ptr],            2            \n\t"
+      "addiu       %[hann],           %[hann],                2            \n\t"
+      "addiu       %[hann1],          %[hann1],               -2           \n\t"
+      "addu        %[store_ptr1],     %[fft],                 %[f_coef]    \n\t"
+      "addu        %[store_ptr2],     %[fft],                 %[s_coef]    \n\t"
+      "srav        %[tmp1],           %[tmp1],                %[shift1]    \n\t"
+      "srav        %[tmp3],           %[tmp3],                %[shift1]    \n\t"
+      "sh          %[tmp1],           0(%[store_ptr1])                     \n\t"
+      "sh          %[tmp3],           0(%[store_ptr2])                     \n\t"
+      "bgtz        %[i],              2b                                   \n\t"
+      " addiu      %[coefs],          %[coefs],               4            \n\t"
+      "3:                                                                   "
+      "\n\t"
+      ".set        pop                                                     \n\t"
+      : [load_ptr] "=&r"(load_ptr), [shift] "=&r"(shift), [hann] "=&r"(hann),
+        [hann1] "=&r"(hann1), [shift1] "=&r"(shift1), [coefs] "=&r"(coefs),
+        [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [tmp3] "=&r"(tmp3),
+        [tmp4] "=&r"(tmp4), [i] "=&r"(i), [f_coef] "=&r"(f_coef),
+        [s_coef] "=&r"(s_coef), [store_ptr1] "=&r"(store_ptr1),
+        [store_ptr2] "=&r"(store_ptr2)
+      : [time_signal] "r"(time_signal), [coefTable] "r"(coefTable),
+        [time_signal_scaling] "r"(time_signal_scaling),
+        [hanning] "r"(WebRtcAecm_kSqrtHanning), [fft] "r"(fft)
+      : "memory", "hi", "lo");
+
+  WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
+  pfrfi = fft;
+  pfreq_signal = freq_signal;
+
+  __asm __volatile(
+      ".set        push                                                     "
+      "\n\t"
+      ".set        noreorder                                                "
+      "\n\t"
+      "addiu       %[j],              $zero,                 128            "
+      "\n\t"
+      "1:                                                                    "
+      "\n\t"
+      "lh          %[tmp1],           0(%[pfrfi])                           "
+      "\n\t"
+      "lh          %[tmp2],           2(%[pfrfi])                           "
+      "\n\t"
+      "lh          %[tmp3],           4(%[pfrfi])                           "
+      "\n\t"
+      "lh          %[tmp4],           6(%[pfrfi])                           "
+      "\n\t"
+      "subu        %[tmp2],           $zero,                 %[tmp2]        "
+      "\n\t"
+      "sh          %[tmp1],           0(%[pfreq_signal])                    "
+      "\n\t"
+      "sh          %[tmp2],           2(%[pfreq_signal])                    "
+      "\n\t"
+      "subu        %[tmp4],           $zero,                 %[tmp4]        "
+      "\n\t"
+      "sh          %[tmp3],           4(%[pfreq_signal])                    "
+      "\n\t"
+      "sh          %[tmp4],           6(%[pfreq_signal])                    "
+      "\n\t"
+      "lh          %[tmp1],           8(%[pfrfi])                           "
+      "\n\t"
+      "lh          %[tmp2],           10(%[pfrfi])                          "
+      "\n\t"
+      "lh          %[tmp3],           12(%[pfrfi])                          "
+      "\n\t"
+      "lh          %[tmp4],           14(%[pfrfi])                          "
+      "\n\t"
+      "addiu       %[j],              %[j],                  -8             "
+      "\n\t"
+      "subu        %[tmp2],           $zero,                 %[tmp2]        "
+      "\n\t"
+      "sh          %[tmp1],           8(%[pfreq_signal])                    "
+      "\n\t"
+      "sh          %[tmp2],           10(%[pfreq_signal])                   "
+      "\n\t"
+      "subu        %[tmp4],           $zero,                 %[tmp4]        "
+      "\n\t"
+      "sh          %[tmp3],           12(%[pfreq_signal])                   "
+      "\n\t"
+      "sh          %[tmp4],           14(%[pfreq_signal])                   "
+      "\n\t"
+      "addiu       %[pfreq_signal],   %[pfreq_signal],       16             "
+      "\n\t"
+      "bgtz        %[j],              1b                                    "
+      "\n\t"
+      " addiu      %[pfrfi],          %[pfrfi],              16             "
+      "\n\t"
+      ".set        pop                                                      "
+      "\n\t"
+      : [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [tmp3] "=&r"(tmp3),
+        [j] "=&r"(j), [pfrfi] "+r"(pfrfi), [pfreq_signal] "+r"(pfreq_signal),
+        [tmp4] "=&r"(tmp4)
+      :
+      : "memory");
+}
+
+static void InverseFFTAndWindow(AecmCore* aecm,
+                                int16_t* fft,
+                                ComplexInt16* efw,
+                                int16_t* output,
+                                const int16_t* nearendClean) {
+  int i, outCFFT;
+  int32_t tmp1, tmp2, tmp3, tmp4, tmp_re, tmp_im;
+  int16_t* pcoefTable_ifft = coefTable_ifft;
+  int16_t* pfft = fft;
+  int16_t* ppfft = fft;
+  ComplexInt16* pefw = efw;
+  int32_t out_aecm;
+  int16_t* paecm_buf = aecm->outBuf;
+  const int16_t* p_kSqrtHanning = WebRtcAecm_kSqrtHanning;
+  const int16_t* pp_kSqrtHanning = &WebRtcAecm_kSqrtHanning[PART_LEN];
+  int16_t* output1 = output;
+
+  __asm __volatile(
+      ".set      push                                                        "
+      "\n\t"
+      ".set      noreorder                                                   "
+      "\n\t"
+      "addiu     %[i],                $zero,                   64            "
+      "\n\t"
+      "1:                                                                     "
+      "\n\t"
+      "lh        %[tmp1],             0(%[pcoefTable_ifft])                  "
+      "\n\t"
+      "lh        %[tmp2],             2(%[pcoefTable_ifft])                  "
+      "\n\t"
+      "lh        %[tmp_re],           0(%[pefw])                             "
+      "\n\t"
+      "lh        %[tmp_im],           2(%[pefw])                             "
+      "\n\t"
+      "addu      %[pfft],             %[fft],                  %[tmp2]       "
+      "\n\t"
+      "sh        %[tmp_re],           0(%[pfft])                             "
+      "\n\t"
+      "sh        %[tmp_im],           2(%[pfft])                             "
+      "\n\t"
+      "addu      %[pfft],             %[fft],                  %[tmp1]       "
+      "\n\t"
+      "sh        %[tmp_re],           0(%[pfft])                             "
+      "\n\t"
+      "subu      %[tmp_im],           $zero,                   %[tmp_im]     "
+      "\n\t"
+      "sh        %[tmp_im],           2(%[pfft])                             "
+      "\n\t"
+      "lh        %[tmp1],             4(%[pcoefTable_ifft])                  "
+      "\n\t"
+      "lh        %[tmp2],             6(%[pcoefTable_ifft])                  "
+      "\n\t"
+      "lh        %[tmp_re],           4(%[pefw])                             "
+      "\n\t"
+      "lh        %[tmp_im],           6(%[pefw])                             "
+      "\n\t"
+      "addu      %[pfft],             %[fft],                  %[tmp2]       "
+      "\n\t"
+      "sh        %[tmp_re],           0(%[pfft])                             "
+      "\n\t"
+      "sh        %[tmp_im],           2(%[pfft])                             "
+      "\n\t"
+      "addu      %[pfft],             %[fft],                  %[tmp1]       "
+      "\n\t"
+      "sh        %[tmp_re],           0(%[pfft])                             "
+      "\n\t"
+      "subu      %[tmp_im],           $zero,                   %[tmp_im]     "
+      "\n\t"
+      "sh        %[tmp_im],           2(%[pfft])                             "
+      "\n\t"
+      "lh        %[tmp1],             8(%[pcoefTable_ifft])                  "
+      "\n\t"
+      "lh        %[tmp2],             10(%[pcoefTable_ifft])                 "
+      "\n\t"
+      "lh        %[tmp_re],           8(%[pefw])                             "
+      "\n\t"
+      "lh        %[tmp_im],           10(%[pefw])                            "
+      "\n\t"
+      "addu      %[pfft],             %[fft],                  %[tmp2]       "
+      "\n\t"
+      "sh        %[tmp_re],           0(%[pfft])                             "
+      "\n\t"
+      "sh        %[tmp_im],           2(%[pfft])                             "
+      "\n\t"
+      "addu      %[pfft],             %[fft],                  %[tmp1]       "
+      "\n\t"
+      "sh        %[tmp_re],           0(%[pfft])                             "
+      "\n\t"
+      "subu      %[tmp_im],           $zero,                   %[tmp_im]     "
+      "\n\t"
+      "sh        %[tmp_im],           2(%[pfft])                             "
+      "\n\t"
+      "lh        %[tmp1],             12(%[pcoefTable_ifft])                 "
+      "\n\t"
+      "lh        %[tmp2],             14(%[pcoefTable_ifft])                 "
+      "\n\t"
+      "lh        %[tmp_re],           12(%[pefw])                            "
+      "\n\t"
+      "lh        %[tmp_im],           14(%[pefw])                            "
+      "\n\t"
+      "addu      %[pfft],             %[fft],                  %[tmp2]       "
+      "\n\t"
+      "sh        %[tmp_re],           0(%[pfft])                             "
+      "\n\t"
+      "sh        %[tmp_im],           2(%[pfft])                             "
+      "\n\t"
+      "addu      %[pfft],             %[fft],                  %[tmp1]       "
+      "\n\t"
+      "sh        %[tmp_re],           0(%[pfft])                             "
+      "\n\t"
+      "subu      %[tmp_im],           $zero,                   %[tmp_im]     "
+      "\n\t"
+      "sh        %[tmp_im],           2(%[pfft])                             "
+      "\n\t"
+      "addiu     %[pcoefTable_ifft],  %[pcoefTable_ifft],      16            "
+      "\n\t"
+      "addiu     %[i],                %[i],                    -4            "
+      "\n\t"
+      "bgtz      %[i],                1b                                     "
+      "\n\t"
+      " addiu    %[pefw],             %[pefw],                 16            "
+      "\n\t"
+      ".set      pop                                                         "
+      "\n\t"
+      : [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [pfft] "+r"(pfft), [i] "=&r"(i),
+        [tmp_re] "=&r"(tmp_re), [tmp_im] "=&r"(tmp_im), [pefw] "+r"(pefw),
+        [pcoefTable_ifft] "+r"(pcoefTable_ifft), [fft] "+r"(fft)
+      :
+      : "memory");
+
+  fft[2] = efw[PART_LEN].real;
+  fft[3] = -efw[PART_LEN].imag;
+
+  outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
+  pfft = fft;
+
+  __asm __volatile(
+      ".set       push                                               \n\t"
+      ".set       noreorder                                          \n\t"
+      "addiu      %[i],            $zero,               128          \n\t"
+      "1:                                                             \n\t"
+      "lh         %[tmp1],         0(%[ppfft])                       \n\t"
+      "lh         %[tmp2],         4(%[ppfft])                       \n\t"
+      "lh         %[tmp3],         8(%[ppfft])                       \n\t"
+      "lh         %[tmp4],         12(%[ppfft])                      \n\t"
+      "addiu      %[i],            %[i],                -4           \n\t"
+      "sh         %[tmp1],         0(%[pfft])                        \n\t"
+      "sh         %[tmp2],         2(%[pfft])                        \n\t"
+      "sh         %[tmp3],         4(%[pfft])                        \n\t"
+      "sh         %[tmp4],         6(%[pfft])                        \n\t"
+      "addiu      %[ppfft],        %[ppfft],            16           \n\t"
+      "bgtz       %[i],            1b                                \n\t"
+      " addiu     %[pfft],         %[pfft],             8            \n\t"
+      ".set       pop                                                \n\t"
+      : [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [pfft] "+r"(pfft), [i] "=&r"(i),
+        [tmp3] "=&r"(tmp3), [tmp4] "=&r"(tmp4), [ppfft] "+r"(ppfft)
+      :
+      : "memory");
+
+  pfft = fft;
+  out_aecm = (int32_t)(outCFFT - aecm->dfaCleanQDomain);
+
+  __asm __volatile(
+      ".set       push                                                       "
+      "\n\t"
+      ".set       noreorder                                                  "
+      "\n\t"
+      "addiu      %[i],                $zero,                  64            "
+      "\n\t"
+      "11:                                                                    "
+      "\n\t"
+      "lh         %[tmp1],             0(%[pfft])                            "
+      "\n\t"
+      "lh         %[tmp2],             0(%[p_kSqrtHanning])                  "
+      "\n\t"
+      "addiu      %[i],                %[i],                   -2            "
+      "\n\t"
+      "mul        %[tmp1],             %[tmp1],                %[tmp2]       "
+      "\n\t"
+      "lh         %[tmp3],             2(%[pfft])                            "
+      "\n\t"
+      "lh         %[tmp4],             2(%[p_kSqrtHanning])                  "
+      "\n\t"
+      "mul        %[tmp3],             %[tmp3],                %[tmp4]       "
+      "\n\t"
+      "addiu      %[tmp1],             %[tmp1],                8192          "
+      "\n\t"
+      "sra        %[tmp1],             %[tmp1],                14            "
+      "\n\t"
+      "addiu      %[tmp3],             %[tmp3],                8192          "
+      "\n\t"
+      "sra        %[tmp3],             %[tmp3],                14            "
+      "\n\t"
+      "bgez       %[out_aecm],         1f                                    "
+      "\n\t"
+      " negu      %[tmp2],             %[out_aecm]                           "
+      "\n\t"
+      "srav       %[tmp1],             %[tmp1],                %[tmp2]       "
+      "\n\t"
+      "b          2f                                                         "
+      "\n\t"
+      " srav      %[tmp3],             %[tmp3],                %[tmp2]       "
+      "\n\t"
+      "1:                                                                     "
+      "\n\t"
+      "sllv       %[tmp1],             %[tmp1],                %[out_aecm]   "
+      "\n\t"
+      "sllv       %[tmp3],             %[tmp3],                %[out_aecm]   "
+      "\n\t"
+      "2:                                                                     "
+      "\n\t"
+      "lh         %[tmp4],             0(%[paecm_buf])                       "
+      "\n\t"
+      "lh         %[tmp2],             2(%[paecm_buf])                       "
+      "\n\t"
+      "addu       %[tmp3],             %[tmp3],                %[tmp2]       "
+      "\n\t"
+      "addu       %[tmp1],             %[tmp1],                %[tmp4]       "
+      "\n\t"
+#if defined(MIPS_DSP_R1_LE)
+      "shll_s.w   %[tmp1],             %[tmp1],                16            "
+      "\n\t"
+      "sra        %[tmp1],             %[tmp1],                16            "
+      "\n\t"
+      "shll_s.w   %[tmp3],             %[tmp3],                16            "
+      "\n\t"
+      "sra        %[tmp3],             %[tmp3],                16            "
+      "\n\t"
+#else   // #if defined(MIPS_DSP_R1_LE)
+      "sra        %[tmp4],             %[tmp1],                31            "
+      "\n\t"
+      "sra        %[tmp2],             %[tmp1],                15            "
+      "\n\t"
+      "beq        %[tmp4],             %[tmp2],                3f            "
+      "\n\t"
+      " ori       %[tmp2],             $zero,                  0x7fff        "
+      "\n\t"
+      "xor        %[tmp1],             %[tmp2],                %[tmp4]       "
+      "\n\t"
+      "3:                                                                     "
+      "\n\t"
+      "sra        %[tmp2],             %[tmp3],                31            "
+      "\n\t"
+      "sra        %[tmp4],             %[tmp3],                15            "
+      "\n\t"
+      "beq        %[tmp2],             %[tmp4],                4f            "
+      "\n\t"
+      " ori       %[tmp4],             $zero,                  0x7fff        "
+      "\n\t"
+      "xor        %[tmp3],             %[tmp4],                %[tmp2]       "
+      "\n\t"
+      "4:                                                                     "
+      "\n\t"
+#endif  // #if defined(MIPS_DSP_R1_LE)
+      "sh         %[tmp1],             0(%[pfft])                            "
+      "\n\t"
+      "sh         %[tmp1],             0(%[output1])                         "
+      "\n\t"
+      "sh         %[tmp3],             2(%[pfft])                            "
+      "\n\t"
+      "sh         %[tmp3],             2(%[output1])                         "
+      "\n\t"
+      "lh         %[tmp1],             128(%[pfft])                          "
+      "\n\t"
+      "lh         %[tmp2],             0(%[pp_kSqrtHanning])                 "
+      "\n\t"
+      "mul        %[tmp1],             %[tmp1],                %[tmp2]       "
+      "\n\t"
+      "lh         %[tmp3],             130(%[pfft])                          "
+      "\n\t"
+      "lh         %[tmp4],             -2(%[pp_kSqrtHanning])                "
+      "\n\t"
+      "mul        %[tmp3],             %[tmp3],                %[tmp4]       "
+      "\n\t"
+      "sra        %[tmp1],             %[tmp1],                14            "
+      "\n\t"
+      "sra        %[tmp3],             %[tmp3],                14            "
+      "\n\t"
+      "bgez       %[out_aecm],         5f                                    "
+      "\n\t"
+      " negu      %[tmp2],             %[out_aecm]                           "
+      "\n\t"
+      "srav       %[tmp3],             %[tmp3],                %[tmp2]       "
+      "\n\t"
+      "b          6f                                                         "
+      "\n\t"
+      " srav      %[tmp1],             %[tmp1],                %[tmp2]       "
+      "\n\t"
+      "5:                                                                     "
+      "\n\t"
+      "sllv       %[tmp1],             %[tmp1],                %[out_aecm]   "
+      "\n\t"
+      "sllv       %[tmp3],             %[tmp3],                %[out_aecm]   "
+      "\n\t"
+      "6:                                                                     "
+      "\n\t"
+#if defined(MIPS_DSP_R1_LE)
+      "shll_s.w   %[tmp1],             %[tmp1],                16            "
+      "\n\t"
+      "sra        %[tmp1],             %[tmp1],                16            "
+      "\n\t"
+      "shll_s.w   %[tmp3],             %[tmp3],                16            "
+      "\n\t"
+      "sra        %[tmp3],             %[tmp3],                16            "
+      "\n\t"
+#else   // #if defined(MIPS_DSP_R1_LE)
+      "sra        %[tmp4],             %[tmp1],                31            "
+      "\n\t"
+      "sra        %[tmp2],             %[tmp1],                15            "
+      "\n\t"
+      "beq        %[tmp4],             %[tmp2],                7f            "
+      "\n\t"
+      " ori       %[tmp2],             $zero,                  0x7fff        "
+      "\n\t"
+      "xor        %[tmp1],             %[tmp2],                %[tmp4]       "
+      "\n\t"
+      "7:                                                                     "
+      "\n\t"
+      "sra        %[tmp2],             %[tmp3],                31            "
+      "\n\t"
+      "sra        %[tmp4],             %[tmp3],                15            "
+      "\n\t"
+      "beq        %[tmp2],             %[tmp4],                8f            "
+      "\n\t"
+      " ori       %[tmp4],             $zero,                  0x7fff        "
+      "\n\t"
+      "xor        %[tmp3],             %[tmp4],                %[tmp2]       "
+      "\n\t"
+      "8:                                                                     "
+      "\n\t"
+#endif  // #if defined(MIPS_DSP_R1_LE)
+      "sh         %[tmp1],             0(%[paecm_buf])                       "
+      "\n\t"
+      "sh         %[tmp3],             2(%[paecm_buf])                       "
+      "\n\t"
+      "addiu      %[output1],          %[output1],             4             "
+      "\n\t"
+      "addiu      %[paecm_buf],        %[paecm_buf],           4             "
+      "\n\t"
+      "addiu      %[pfft],             %[pfft],                4             "
+      "\n\t"
+      "addiu      %[p_kSqrtHanning],   %[p_kSqrtHanning],      4             "
+      "\n\t"
+      "bgtz       %[i],                11b                                   "
+      "\n\t"
+      " addiu     %[pp_kSqrtHanning],  %[pp_kSqrtHanning],     -4            "
+      "\n\t"
+      ".set       pop                                                        "
+      "\n\t"
+      : [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [pfft] "+r"(pfft),
+        [output1] "+r"(output1), [tmp3] "=&r"(tmp3), [tmp4] "=&r"(tmp4),
+        [paecm_buf] "+r"(paecm_buf), [i] "=&r"(i),
+        [pp_kSqrtHanning] "+r"(pp_kSqrtHanning),
+        [p_kSqrtHanning] "+r"(p_kSqrtHanning)
+      : [out_aecm] "r"(out_aecm),
+        [WebRtcAecm_kSqrtHanning] "r"(WebRtcAecm_kSqrtHanning)
+      : "hi", "lo", "memory");
+
+  // Copy the current block to the old position
+  // (aecm->outBuf is shifted elsewhere)
+  memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(int16_t) * PART_LEN);
+  memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN,
+         sizeof(int16_t) * PART_LEN);
+  if (nearendClean != NULL) {
+    memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN,
+           sizeof(int16_t) * PART_LEN);
+  }
+}
+
+void WebRtcAecm_CalcLinearEnergies_mips(AecmCore* aecm,
+                                        const uint16_t* far_spectrum,
+                                        int32_t* echo_est,
+                                        uint32_t* far_energy,
+                                        uint32_t* echo_energy_adapt,
+                                        uint32_t* echo_energy_stored) {
+  int i;
+  uint32_t par1 = (*far_energy);
+  uint32_t par2 = (*echo_energy_adapt);
+  uint32_t par3 = (*echo_energy_stored);
+  int16_t* ch_stored_p = &(aecm->channelStored[0]);
+  int16_t* ch_adapt_p = &(aecm->channelAdapt16[0]);
+  uint16_t* spectrum_p = (uint16_t*)(&(far_spectrum[0]));
+  int32_t* echo_p = &(echo_est[0]);
+  int32_t temp0, stored0, echo0, adept0, spectrum0;
+  int32_t stored1, adept1, spectrum1, echo1, temp1;
+
+  // Get energy for the delayed far end signal and estimated
+  // echo using both stored and adapted channels.
+  for (i = 0; i < PART_LEN; i += 4) {
+    __asm __volatile(
+        ".set           push                                            \n\t"
+        ".set           noreorder                                       \n\t"
+        "lh             %[stored0],     0(%[ch_stored_p])               \n\t"
+        "lhu            %[adept0],      0(%[ch_adapt_p])                \n\t"
+        "lhu            %[spectrum0],   0(%[spectrum_p])                \n\t"
+        "lh             %[stored1],     2(%[ch_stored_p])               \n\t"
+        "lhu            %[adept1],      2(%[ch_adapt_p])                \n\t"
+        "lhu            %[spectrum1],   2(%[spectrum_p])                \n\t"
+        "mul            %[echo0],       %[stored0],     %[spectrum0]    \n\t"
+        "mul            %[temp0],       %[adept0],      %[spectrum0]    \n\t"
+        "mul            %[echo1],       %[stored1],     %[spectrum1]    \n\t"
+        "mul            %[temp1],       %[adept1],      %[spectrum1]    \n\t"
+        "addu           %[par1],        %[par1],        %[spectrum0]    \n\t"
+        "addu           %[par1],        %[par1],        %[spectrum1]    \n\t"
+        "addiu          %[echo_p],      %[echo_p],      16              \n\t"
+        "addu           %[par3],        %[par3],        %[echo0]        \n\t"
+        "addu           %[par2],        %[par2],        %[temp0]        \n\t"
+        "addu           %[par3],        %[par3],        %[echo1]        \n\t"
+        "addu           %[par2],        %[par2],        %[temp1]        \n\t"
+        "usw            %[echo0],       -16(%[echo_p])                  \n\t"
+        "usw            %[echo1],       -12(%[echo_p])                  \n\t"
+        "lh             %[stored0],     4(%[ch_stored_p])               \n\t"
+        "lhu            %[adept0],      4(%[ch_adapt_p])                \n\t"
+        "lhu            %[spectrum0],   4(%[spectrum_p])                \n\t"
+        "lh             %[stored1],     6(%[ch_stored_p])               \n\t"
+        "lhu            %[adept1],      6(%[ch_adapt_p])                \n\t"
+        "lhu            %[spectrum1],   6(%[spectrum_p])                \n\t"
+        "mul            %[echo0],       %[stored0],     %[spectrum0]    \n\t"
+        "mul            %[temp0],       %[adept0],      %[spectrum0]    \n\t"
+        "mul            %[echo1],       %[stored1],     %[spectrum1]    \n\t"
+        "mul            %[temp1],       %[adept1],      %[spectrum1]    \n\t"
+        "addu           %[par1],        %[par1],        %[spectrum0]    \n\t"
+        "addu           %[par1],        %[par1],        %[spectrum1]    \n\t"
+        "addiu          %[ch_stored_p], %[ch_stored_p], 8               \n\t"
+        "addiu          %[ch_adapt_p],  %[ch_adapt_p],  8               \n\t"
+        "addiu          %[spectrum_p],  %[spectrum_p],  8               \n\t"
+        "addu           %[par3],        %[par3],        %[echo0]        \n\t"
+        "addu           %[par2],        %[par2],        %[temp0]        \n\t"
+        "addu           %[par3],        %[par3],        %[echo1]        \n\t"
+        "addu           %[par2],        %[par2],        %[temp1]        \n\t"
+        "usw            %[echo0],       -8(%[echo_p])                   \n\t"
+        "usw            %[echo1],       -4(%[echo_p])                   \n\t"
+        ".set           pop                                             \n\t"
+        : [temp0] "=&r"(temp0), [stored0] "=&r"(stored0),
+          [adept0] "=&r"(adept0), [spectrum0] "=&r"(spectrum0),
+          [echo0] "=&r"(echo0), [echo_p] "+r"(echo_p), [par3] "+r"(par3),
+          [par1] "+r"(par1), [par2] "+r"(par2), [stored1] "=&r"(stored1),
+          [adept1] "=&r"(adept1), [echo1] "=&r"(echo1),
+          [spectrum1] "=&r"(spectrum1), [temp1] "=&r"(temp1),
+          [ch_stored_p] "+r"(ch_stored_p), [ch_adapt_p] "+r"(ch_adapt_p),
+          [spectrum_p] "+r"(spectrum_p)
+        :
+        : "hi", "lo", "memory");
+  }
+
+  echo_est[PART_LEN] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[PART_LEN],
+                                             far_spectrum[PART_LEN]);
+  par1 += (uint32_t)(far_spectrum[PART_LEN]);
+  par2 += aecm->channelAdapt16[PART_LEN] * far_spectrum[PART_LEN];
+  par3 += (uint32_t)echo_est[PART_LEN];
+
+  (*far_energy) = par1;
+  (*echo_energy_adapt) = par2;
+  (*echo_energy_stored) = par3;
+}
+
+#if defined(MIPS_DSP_R1_LE)
+void WebRtcAecm_StoreAdaptiveChannel_mips(AecmCore* aecm,
+                                          const uint16_t* far_spectrum,
+                                          int32_t* echo_est) {
+  int i;
+  int16_t* temp1;
+  uint16_t* temp8;
+  int32_t temp0, temp2, temp3, temp4, temp5, temp6;
+  int32_t* temp7 = &(echo_est[0]);
+  temp1 = &(aecm->channelStored[0]);
+  temp8 = (uint16_t*)(&far_spectrum[0]);
+
+  // During startup we store the channel every block.
+  memcpy(aecm->channelStored, aecm->channelAdapt16,
+         sizeof(int16_t) * PART_LEN1);
+  // Recalculate echo estimate
+  for (i = 0; i < PART_LEN; i += 4) {
+    __asm __volatile(
+        "ulw            %[temp0],   0(%[temp8])               \n\t"
+        "ulw            %[temp2],   0(%[temp1])               \n\t"
+        "ulw            %[temp4],   4(%[temp8])               \n\t"
+        "ulw            %[temp5],   4(%[temp1])               \n\t"
+        "muleq_s.w.phl  %[temp3],   %[temp2],     %[temp0]    \n\t"
+        "muleq_s.w.phr  %[temp0],   %[temp2],     %[temp0]    \n\t"
+        "muleq_s.w.phl  %[temp6],   %[temp5],     %[temp4]    \n\t"
+        "muleq_s.w.phr  %[temp4],   %[temp5],     %[temp4]    \n\t"
+        "addiu          %[temp7],   %[temp7],     16          \n\t"
+        "addiu          %[temp1],   %[temp1],     8           \n\t"
+        "addiu          %[temp8],   %[temp8],     8           \n\t"
+        "sra            %[temp3],   %[temp3],     1           \n\t"
+        "sra            %[temp0],   %[temp0],     1           \n\t"
+        "sra            %[temp6],   %[temp6],     1           \n\t"
+        "sra            %[temp4],   %[temp4],     1           \n\t"
+        "usw            %[temp3],   -12(%[temp7])             \n\t"
+        "usw            %[temp0],   -16(%[temp7])             \n\t"
+        "usw            %[temp6],   -4(%[temp7])              \n\t"
+        "usw            %[temp4],   -8(%[temp7])              \n\t"
+        : [temp0] "=&r"(temp0), [temp2] "=&r"(temp2), [temp3] "=&r"(temp3),
+          [temp4] "=&r"(temp4), [temp5] "=&r"(temp5), [temp6] "=&r"(temp6),
+          [temp1] "+r"(temp1), [temp8] "+r"(temp8), [temp7] "+r"(temp7)
+        :
+        : "hi", "lo", "memory");
+  }
+  echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
+}
+
+void WebRtcAecm_ResetAdaptiveChannel_mips(AecmCore* aecm) {
+  int i;
+  int32_t* temp3;
+  int16_t* temp0;
+  int32_t temp1, temp2, temp4, temp5;
+
+  temp0 = &(aecm->channelStored[0]);
+  temp3 = &(aecm->channelAdapt32[0]);
+
+  // The stored channel has a significantly lower MSE than the adaptive one for
+  // two consecutive calculations. Reset the adaptive channel.
+  memcpy(aecm->channelAdapt16, aecm->channelStored,
+         sizeof(int16_t) * PART_LEN1);
+
+  // Restore the W32 channel
+  for (i = 0; i < PART_LEN; i += 4) {
+    __asm __volatile(
+        "ulw            %[temp1], 0(%[temp0])           \n\t"
+        "ulw            %[temp4], 4(%[temp0])           \n\t"
+        "preceq.w.phl   %[temp2], %[temp1]              \n\t"
+        "preceq.w.phr   %[temp1], %[temp1]              \n\t"
+        "preceq.w.phl   %[temp5], %[temp4]              \n\t"
+        "preceq.w.phr   %[temp4], %[temp4]              \n\t"
+        "addiu          %[temp0], %[temp0], 8           \n\t"
+        "usw            %[temp2], 4(%[temp3])           \n\t"
+        "usw            %[temp1], 0(%[temp3])           \n\t"
+        "usw            %[temp5], 12(%[temp3])          \n\t"
+        "usw            %[temp4], 8(%[temp3])           \n\t"
+        "addiu          %[temp3], %[temp3], 16          \n\t"
+        : [temp1] "=&r"(temp1), [temp2] "=&r"(temp2), [temp4] "=&r"(temp4),
+          [temp5] "=&r"(temp5), [temp3] "+r"(temp3), [temp0] "+r"(temp0)
+        :
+        : "memory");
+  }
+
+  aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
+}
+#endif  // #if defined(MIPS_DSP_R1_LE)
+
+// Transforms a time domain signal into the frequency domain, outputting the
+// complex valued signal, absolute value and sum of absolute values.
+//
+// time_signal          [in]    Pointer to time domain signal
+// freq_signal_real     [out]   Pointer to real part of frequency domain array
+// freq_signal_imag     [out]   Pointer to imaginary part of frequency domain
+//                              array
+// freq_signal_abs      [out]   Pointer to absolute value of frequency domain
+//                              array
+// freq_signal_sum_abs  [out]   Pointer to the sum of all absolute values in
+//                              the frequency domain array
+// return value                 The Q-domain of current frequency values
+//
+static int TimeToFrequencyDomain(AecmCore* aecm,
+                                 const int16_t* time_signal,
+                                 ComplexInt16* freq_signal,
+                                 uint16_t* freq_signal_abs,
+                                 uint32_t* freq_signal_sum_abs) {
+  int i = 0;
+  int time_signal_scaling = 0;
+
+  // In fft_buf, +16 for 32-byte alignment.
+  int16_t fft_buf[PART_LEN4 + 16];
+  int16_t* fft = (int16_t*)(((uintptr_t)fft_buf + 31) & ~31);
+
+  int16_t tmp16no1;
+#if !defined(MIPS_DSP_R2_LE)
+  int32_t tmp32no1;
+  int32_t tmp32no2;
+  int16_t tmp16no2;
+#else
+  int32_t tmp32no10, tmp32no11, tmp32no12, tmp32no13;
+  int32_t tmp32no20, tmp32no21, tmp32no22, tmp32no23;
+  int16_t* freqp;
+  uint16_t* freqabsp;
+  uint32_t freqt0, freqt1, freqt2, freqt3;
+  uint32_t freqs;
+#endif
+
+#ifdef AECM_DYNAMIC_Q
+  tmp16no1 = WebRtcSpl_MaxAbsValueW16(time_signal, PART_LEN2);
+  time_signal_scaling = WebRtcSpl_NormW16(tmp16no1);
+#endif
+
+  WindowAndFFT(aecm, fft, time_signal, freq_signal, time_signal_scaling);
+
+  // Extract imaginary and real part,
+  // calculate the magnitude for all frequency bins
+  freq_signal[0].imag = 0;
+  freq_signal[PART_LEN].imag = 0;
+  freq_signal[PART_LEN].real = fft[PART_LEN2];
+  freq_signal_abs[0] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[0].real);
+  freq_signal_abs[PART_LEN] =
+      (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[PART_LEN].real);
+  (*freq_signal_sum_abs) =
+      (uint32_t)(freq_signal_abs[0]) + (uint32_t)(freq_signal_abs[PART_LEN]);
+
+#if !defined(MIPS_DSP_R2_LE)
+  for (i = 1; i < PART_LEN; i++) {
+    if (freq_signal[i].real == 0) {
+      freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
+    } else if (freq_signal[i].imag == 0) {
+      freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].real);
+    } else {
+      // Approximation for magnitude of complex fft output
+      // magn = sqrt(real^2 + imag^2)
+      // magn ~= alpha * max(`imag`,`real`) + beta * min(`imag`,`real`)
+      //
+      // The parameters alpha and beta are stored in Q15
+      tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
+      tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
+      tmp32no1 = tmp16no1 * tmp16no1;
+      tmp32no2 = tmp16no2 * tmp16no2;
+      tmp32no2 = WebRtcSpl_AddSatW32(tmp32no1, tmp32no2);
+      tmp32no1 = WebRtcSpl_SqrtFloor(tmp32no2);
+
+      freq_signal_abs[i] = (uint16_t)tmp32no1;
+    }
+    (*freq_signal_sum_abs) += (uint32_t)freq_signal_abs[i];
+  }
+#else  // #if !defined(MIPS_DSP_R2_LE)
+  freqs =
+      (uint32_t)(freq_signal_abs[0]) + (uint32_t)(freq_signal_abs[PART_LEN]);
+  freqp = &(freq_signal[1].real);
+
+  __asm __volatile(
+      "lw             %[freqt0],      0(%[freqp])             \n\t"
+      "lw             %[freqt1],      4(%[freqp])             \n\t"
+      "lw             %[freqt2],      8(%[freqp])             \n\t"
+      "mult           $ac0,           $zero,      $zero       \n\t"
+      "mult           $ac1,           $zero,      $zero       \n\t"
+      "mult           $ac2,           $zero,      $zero       \n\t"
+      "dpaq_s.w.ph    $ac0,           %[freqt0],  %[freqt0]   \n\t"
+      "dpaq_s.w.ph    $ac1,           %[freqt1],  %[freqt1]   \n\t"
+      "dpaq_s.w.ph    $ac2,           %[freqt2],  %[freqt2]   \n\t"
+      "addiu          %[freqp],       %[freqp],   12          \n\t"
+      "extr.w         %[tmp32no20],   $ac0,       1           \n\t"
+      "extr.w         %[tmp32no21],   $ac1,       1           \n\t"
+      "extr.w         %[tmp32no22],   $ac2,       1           \n\t"
+      : [freqt0] "=&r"(freqt0), [freqt1] "=&r"(freqt1), [freqt2] "=&r"(freqt2),
+        [freqp] "+r"(freqp), [tmp32no20] "=r"(tmp32no20),
+        [tmp32no21] "=r"(tmp32no21), [tmp32no22] "=r"(tmp32no22)
+      :
+      : "memory", "hi", "lo", "$ac1hi", "$ac1lo", "$ac2hi", "$ac2lo");
+
+  tmp32no10 = WebRtcSpl_SqrtFloor(tmp32no20);
+  tmp32no11 = WebRtcSpl_SqrtFloor(tmp32no21);
+  tmp32no12 = WebRtcSpl_SqrtFloor(tmp32no22);
+  freq_signal_abs[1] = (uint16_t)tmp32no10;
+  freq_signal_abs[2] = (uint16_t)tmp32no11;
+  freq_signal_abs[3] = (uint16_t)tmp32no12;
+  freqs += (uint32_t)tmp32no10;
+  freqs += (uint32_t)tmp32no11;
+  freqs += (uint32_t)tmp32no12;
+  freqabsp = &(freq_signal_abs[4]);
+  for (i = 4; i < PART_LEN; i += 4) {
+    __asm __volatile(
+        "ulw            %[freqt0],      0(%[freqp])                 \n\t"
+        "ulw            %[freqt1],      4(%[freqp])                 \n\t"
+        "ulw            %[freqt2],      8(%[freqp])                 \n\t"
+        "ulw            %[freqt3],      12(%[freqp])                \n\t"
+        "mult           $ac0,           $zero,          $zero       \n\t"
+        "mult           $ac1,           $zero,          $zero       \n\t"
+        "mult           $ac2,           $zero,          $zero       \n\t"
+        "mult           $ac3,           $zero,          $zero       \n\t"
+        "dpaq_s.w.ph    $ac0,           %[freqt0],      %[freqt0]   \n\t"
+        "dpaq_s.w.ph    $ac1,           %[freqt1],      %[freqt1]   \n\t"
+        "dpaq_s.w.ph    $ac2,           %[freqt2],      %[freqt2]   \n\t"
+        "dpaq_s.w.ph    $ac3,           %[freqt3],      %[freqt3]   \n\t"
+        "addiu          %[freqp],       %[freqp],       16          \n\t"
+        "addiu          %[freqabsp],    %[freqabsp],    8           \n\t"
+        "extr.w         %[tmp32no20],   $ac0,           1           \n\t"
+        "extr.w         %[tmp32no21],   $ac1,           1           \n\t"
+        "extr.w         %[tmp32no22],   $ac2,           1           \n\t"
+        "extr.w         %[tmp32no23],   $ac3,           1           \n\t"
+        : [freqt0] "=&r"(freqt0), [freqt1] "=&r"(freqt1),
+          [freqt2] "=&r"(freqt2), [freqt3] "=&r"(freqt3),
+          [tmp32no20] "=r"(tmp32no20), [tmp32no21] "=r"(tmp32no21),
+          [tmp32no22] "=r"(tmp32no22), [tmp32no23] "=r"(tmp32no23),
+          [freqabsp] "+r"(freqabsp), [freqp] "+r"(freqp)
+        :
+        : "memory", "hi", "lo", "$ac1hi", "$ac1lo", "$ac2hi", "$ac2lo",
+          "$ac3hi", "$ac3lo");
+
+    tmp32no10 = WebRtcSpl_SqrtFloor(tmp32no20);
+    tmp32no11 = WebRtcSpl_SqrtFloor(tmp32no21);
+    tmp32no12 = WebRtcSpl_SqrtFloor(tmp32no22);
+    tmp32no13 = WebRtcSpl_SqrtFloor(tmp32no23);
+
+    __asm __volatile(
+        "sh             %[tmp32no10],   -8(%[freqabsp])                 \n\t"
+        "sh             %[tmp32no11],   -6(%[freqabsp])                 \n\t"
+        "sh             %[tmp32no12],   -4(%[freqabsp])                 \n\t"
+        "sh             %[tmp32no13],   -2(%[freqabsp])                 \n\t"
+        "addu           %[freqs],       %[freqs],       %[tmp32no10]    \n\t"
+        "addu           %[freqs],       %[freqs],       %[tmp32no11]    \n\t"
+        "addu           %[freqs],       %[freqs],       %[tmp32no12]    \n\t"
+        "addu           %[freqs],       %[freqs],       %[tmp32no13]    \n\t"
+        : [freqs] "+r"(freqs)
+        : [tmp32no10] "r"(tmp32no10), [tmp32no11] "r"(tmp32no11),
+          [tmp32no12] "r"(tmp32no12), [tmp32no13] "r"(tmp32no13),
+          [freqabsp] "r"(freqabsp)
+        : "memory");
+  }
+
+  (*freq_signal_sum_abs) = freqs;
+#endif
+
+  return time_signal_scaling;
+}
+
+int WebRtcAecm_ProcessBlock(AecmCore* aecm,
+                            const int16_t* farend,
+                            const int16_t* nearendNoisy,
+                            const int16_t* nearendClean,
+                            int16_t* output) {
+  int i;
+  uint32_t xfaSum;
+  uint32_t dfaNoisySum;
+  uint32_t dfaCleanSum;
+  uint32_t echoEst32Gained;
+  uint32_t tmpU32;
+  int32_t tmp32no1;
+
+  uint16_t xfa[PART_LEN1];
+  uint16_t dfaNoisy[PART_LEN1];
+  uint16_t dfaClean[PART_LEN1];
+  uint16_t* ptrDfaClean = dfaClean;
+  const uint16_t* far_spectrum_ptr = NULL;
+
+  // 32 byte aligned buffers (with +8 or +16).
+  int16_t fft_buf[PART_LEN4 + 2 + 16];  // +2 to make a loop safe.
+  int32_t echoEst32_buf[PART_LEN1 + 8];
+  int32_t dfw_buf[PART_LEN2 + 8];
+  int32_t efw_buf[PART_LEN2 + 8];
+
+  int16_t* fft = (int16_t*)(((uint32_t)fft_buf + 31) & ~31);
+  int32_t* echoEst32 = (int32_t*)(((uint32_t)echoEst32_buf + 31) & ~31);
+  ComplexInt16* dfw = (ComplexInt16*)(((uint32_t)dfw_buf + 31) & ~31);
+  ComplexInt16* efw = (ComplexInt16*)(((uint32_t)efw_buf + 31) & ~31);
+
+  int16_t hnl[PART_LEN1];
+  int16_t numPosCoef = 0;
+  int delay;
+  int16_t tmp16no1;
+  int16_t tmp16no2;
+  int16_t mu;
+  int16_t supGain;
+  int16_t zeros32, zeros16;
+  int16_t zerosDBufNoisy, zerosDBufClean, zerosXBuf;
+  int far_q;
+  int16_t resolutionDiff, qDomainDiff, dfa_clean_q_domain_diff;
+
+  const int kMinPrefBand = 4;
+  const int kMaxPrefBand = 24;
+  int32_t avgHnl32 = 0;
+
+  int32_t temp1, temp2, temp3, temp4, temp5, temp6, temp7, temp8;
+  int16_t* ptr;
+  int16_t* ptr1;
+  int16_t* er_ptr;
+  int16_t* dr_ptr;
+
+  ptr = &hnl[0];
+  ptr1 = &hnl[0];
+  er_ptr = &efw[0].real;
+  dr_ptr = &dfw[0].real;
+
+  // Determine startup state. There are three states:
+  // (0) the first CONV_LEN blocks
+  // (1) another CONV_LEN blocks
+  // (2) the rest
+
+  if (aecm->startupState < 2) {
+    aecm->startupState =
+        (aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
+  }
+  // END: Determine startup state
+
+  // Buffer near and far end signals
+  memcpy(aecm->xBuf + PART_LEN, farend, sizeof(int16_t) * PART_LEN);
+  memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(int16_t) * PART_LEN);
+  if (nearendClean != NULL) {
+    memcpy(aecm->dBufClean + PART_LEN, nearendClean,
+           sizeof(int16_t) * PART_LEN);
+  }
+
+  // Transform far end signal from time domain to frequency domain.
+  far_q = TimeToFrequencyDomain(aecm, aecm->xBuf, dfw, xfa, &xfaSum);
+
+  // Transform noisy near end signal from time domain to frequency domain.
+  zerosDBufNoisy =
+      TimeToFrequencyDomain(aecm, aecm->dBufNoisy, dfw, dfaNoisy, &dfaNoisySum);
+  aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
+  aecm->dfaNoisyQDomain = (int16_t)zerosDBufNoisy;
+
+  if (nearendClean == NULL) {
+    ptrDfaClean = dfaNoisy;
+    aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld;
+    aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain;
+    dfaCleanSum = dfaNoisySum;
+  } else {
+    // Transform clean near end signal from time domain to frequency domain.
+    zerosDBufClean = TimeToFrequencyDomain(aecm, aecm->dBufClean, dfw, dfaClean,
+                                           &dfaCleanSum);
+    aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
+    aecm->dfaCleanQDomain = (int16_t)zerosDBufClean;
+  }
+
+  // Get the delay
+  // Save far-end history and estimate delay
+  WebRtcAecm_UpdateFarHistory(aecm, xfa, far_q);
+
+  if (WebRtc_AddFarSpectrumFix(aecm->delay_estimator_farend, xfa, PART_LEN1,
+                               far_q) == -1) {
+    return -1;
+  }
+  delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator, dfaNoisy,
+                                          PART_LEN1, zerosDBufNoisy);
+  if (delay == -1) {
+    return -1;
+  } else if (delay == -2) {
+    // If the delay is unknown, we assume zero.
+    // NOTE: this will have to be adjusted if we ever add lookahead.
+    delay = 0;
+  }
+
+  if (aecm->fixedDelay >= 0) {
+    // Use fixed delay
+    delay = aecm->fixedDelay;
+  }
+
+  // Get aligned far end spectrum
+  far_spectrum_ptr = WebRtcAecm_AlignedFarend(aecm, &far_q, delay);
+  zerosXBuf = (int16_t)far_q;
+
+  if (far_spectrum_ptr == NULL) {
+    return -1;
+  }
+
+  // Calculate log(energy) and update energy threshold levels
+  WebRtcAecm_CalcEnergies(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisySum,
+                          echoEst32);
+  // Calculate stepsize
+  mu = WebRtcAecm_CalcStepSize(aecm);
+
+  // Update counters
+  aecm->totCount++;
+
+  // This is the channel estimation algorithm.
+  // It is base on NLMS but has a variable step length,
+  // which was calculated above.
+  WebRtcAecm_UpdateChannel(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisy, mu,
+                           echoEst32);
+
+  supGain = WebRtcAecm_CalcSuppressionGain(aecm);
+
+  // Calculate Wiener filter hnl[]
+  for (i = 0; i < PART_LEN1; i++) {
+    // Far end signal through channel estimate in Q8
+    // How much can we shift right to preserve resolution
+    tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
+    aecm->echoFilt[i] +=
+        rtc::dchecked_cast<int32_t>((int64_t{tmp32no1} * 50) >> 8);
+
+    zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
+    zeros16 = WebRtcSpl_NormW16(supGain) + 1;
+    if (zeros32 + zeros16 > 16) {
+      // Multiplication is safe
+      // Result in
+      // Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+aecm->xfaQDomainBuf[diff])
+      echoEst32Gained =
+          WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i], (uint16_t)supGain);
+      resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
+      resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
+    } else {
+      tmp16no1 = 17 - zeros32 - zeros16;
+      resolutionDiff =
+          14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
+      resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
+      if (zeros32 > tmp16no1) {
+        echoEst32Gained = WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i],
+                                                supGain >> tmp16no1);
+      } else {
+        // Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
+        echoEst32Gained = (aecm->echoFilt[i] >> tmp16no1) * supGain;
+      }
+    }
+
+    zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
+    RTC_DCHECK_GE(zeros16, 0);  // `zeros16` is a norm, hence non-negative.
+    dfa_clean_q_domain_diff = aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld;
+    if (zeros16 < dfa_clean_q_domain_diff && aecm->nearFilt[i]) {
+      tmp16no1 = aecm->nearFilt[i] << zeros16;
+      qDomainDiff = zeros16 - dfa_clean_q_domain_diff;
+      tmp16no2 = ptrDfaClean[i] >> -qDomainDiff;
+    } else {
+      tmp16no1 = dfa_clean_q_domain_diff < 0
+                     ? aecm->nearFilt[i] >> -dfa_clean_q_domain_diff
+                     : aecm->nearFilt[i] << dfa_clean_q_domain_diff;
+      qDomainDiff = 0;
+      tmp16no2 = ptrDfaClean[i];
+    }
+
+    tmp32no1 = (int32_t)(tmp16no2 - tmp16no1);
+    tmp16no2 = (int16_t)(tmp32no1 >> 4);
+    tmp16no2 += tmp16no1;
+    zeros16 = WebRtcSpl_NormW16(tmp16no2);
+    if ((tmp16no2) & (-qDomainDiff > zeros16)) {
+      aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
+    } else {
+      aecm->nearFilt[i] =
+          qDomainDiff < 0 ? tmp16no2 << -qDomainDiff : tmp16no2 >> qDomainDiff;
+    }
+
+    // Wiener filter coefficients, resulting hnl in Q14
+    if (echoEst32Gained == 0) {
+      hnl[i] = ONE_Q14;
+      numPosCoef++;
+    } else if (aecm->nearFilt[i] == 0) {
+      hnl[i] = 0;
+    } else {
+      // Multiply the suppression gain
+      // Rounding
+      echoEst32Gained += (uint32_t)(aecm->nearFilt[i] >> 1);
+      tmpU32 =
+          WebRtcSpl_DivU32U16(echoEst32Gained, (uint16_t)aecm->nearFilt[i]);
+
+      // Current resolution is
+      // Q-(RESOLUTION_CHANNEL + RESOLUTION_SUPGAIN
+      //    - max(0, 17 - zeros16 - zeros32))
+      // Make sure we are in Q14
+      tmp32no1 = (int32_t)WEBRTC_SPL_SHIFT_W32(tmpU32, resolutionDiff);
+      if (tmp32no1 > ONE_Q14) {
+        hnl[i] = 0;
+      } else if (tmp32no1 < 0) {
+        hnl[i] = ONE_Q14;
+        numPosCoef++;
+      } else {
+        // 1-echoEst/dfa
+        hnl[i] = ONE_Q14 - (int16_t)tmp32no1;
+        if (hnl[i] <= 0) {
+          hnl[i] = 0;
+        } else {
+          numPosCoef++;
+        }
+      }
+    }
+  }
+
+  // Only in wideband. Prevent the gain in upper band from being larger than
+  // in lower band.
+  if (aecm->mult == 2) {
+    // TODO(bjornv): Investigate if the scaling of hnl[i] below can cause
+    //               speech distortion in double-talk.
+    for (i = 0; i < (PART_LEN1 >> 3); i++) {
+      __asm __volatile(
+          "lh         %[temp1],       0(%[ptr1])                  \n\t"
+          "lh         %[temp2],       2(%[ptr1])                  \n\t"
+          "lh         %[temp3],       4(%[ptr1])                  \n\t"
+          "lh         %[temp4],       6(%[ptr1])                  \n\t"
+          "lh         %[temp5],       8(%[ptr1])                  \n\t"
+          "lh         %[temp6],       10(%[ptr1])                 \n\t"
+          "lh         %[temp7],       12(%[ptr1])                 \n\t"
+          "lh         %[temp8],       14(%[ptr1])                 \n\t"
+          "mul        %[temp1],       %[temp1],       %[temp1]    \n\t"
+          "mul        %[temp2],       %[temp2],       %[temp2]    \n\t"
+          "mul        %[temp3],       %[temp3],       %[temp3]    \n\t"
+          "mul        %[temp4],       %[temp4],       %[temp4]    \n\t"
+          "mul        %[temp5],       %[temp5],       %[temp5]    \n\t"
+          "mul        %[temp6],       %[temp6],       %[temp6]    \n\t"
+          "mul        %[temp7],       %[temp7],       %[temp7]    \n\t"
+          "mul        %[temp8],       %[temp8],       %[temp8]    \n\t"
+          "sra        %[temp1],       %[temp1],       14          \n\t"
+          "sra        %[temp2],       %[temp2],       14          \n\t"
+          "sra        %[temp3],       %[temp3],       14          \n\t"
+          "sra        %[temp4],       %[temp4],       14          \n\t"
+          "sra        %[temp5],       %[temp5],       14          \n\t"
+          "sra        %[temp6],       %[temp6],       14          \n\t"
+          "sra        %[temp7],       %[temp7],       14          \n\t"
+          "sra        %[temp8],       %[temp8],       14          \n\t"
+          "sh         %[temp1],       0(%[ptr1])                  \n\t"
+          "sh         %[temp2],       2(%[ptr1])                  \n\t"
+          "sh         %[temp3],       4(%[ptr1])                  \n\t"
+          "sh         %[temp4],       6(%[ptr1])                  \n\t"
+          "sh         %[temp5],       8(%[ptr1])                  \n\t"
+          "sh         %[temp6],       10(%[ptr1])                 \n\t"
+          "sh         %[temp7],       12(%[ptr1])                 \n\t"
+          "sh         %[temp8],       14(%[ptr1])                 \n\t"
+          "addiu      %[ptr1],        %[ptr1],        16          \n\t"
+          : [temp1] "=&r"(temp1), [temp2] "=&r"(temp2), [temp3] "=&r"(temp3),
+            [temp4] "=&r"(temp4), [temp5] "=&r"(temp5), [temp6] "=&r"(temp6),
+            [temp7] "=&r"(temp7), [temp8] "=&r"(temp8), [ptr1] "+r"(ptr1)
+          :
+          : "memory", "hi", "lo");
+    }
+    for (i = 0; i < (PART_LEN1 & 7); i++) {
+      __asm __volatile(
+          "lh         %[temp1],       0(%[ptr1])                  \n\t"
+          "mul        %[temp1],       %[temp1],       %[temp1]    \n\t"
+          "sra        %[temp1],       %[temp1],       14          \n\t"
+          "sh         %[temp1],       0(%[ptr1])                  \n\t"
+          "addiu      %[ptr1],        %[ptr1],        2           \n\t"
+          : [temp1] "=&r"(temp1), [ptr1] "+r"(ptr1)
+          :
+          : "memory", "hi", "lo");
+    }
+
+    for (i = kMinPrefBand; i <= kMaxPrefBand; i++) {
+      avgHnl32 += (int32_t)hnl[i];
+    }
+
+    RTC_DCHECK_GT(kMaxPrefBand - kMinPrefBand + 1, 0);
+    avgHnl32 /= (kMaxPrefBand - kMinPrefBand + 1);
+
+    for (i = kMaxPrefBand; i < PART_LEN1; i++) {
+      if (hnl[i] > (int16_t)avgHnl32) {
+        hnl[i] = (int16_t)avgHnl32;
+      }
+    }
+  }
+
+  // Calculate NLP gain, result is in Q14
+  if (aecm->nlpFlag) {
+    if (numPosCoef < 3) {
+      for (i = 0; i < PART_LEN1; i++) {
+        efw[i].real = 0;
+        efw[i].imag = 0;
+        hnl[i] = 0;
+      }
+    } else {
+      for (i = 0; i < PART_LEN1; i++) {
+#if defined(MIPS_DSP_R1_LE)
+        __asm __volatile(
+            ".set       push                                        \n\t"
+            ".set       noreorder                                   \n\t"
+            "lh         %[temp1],       0(%[ptr])                   \n\t"
+            "lh         %[temp2],       0(%[dr_ptr])                \n\t"
+            "slti       %[temp4],       %[temp1],       0x4001      \n\t"
+            "beqz       %[temp4],       3f                          \n\t"
+            " lh        %[temp3],       2(%[dr_ptr])                \n\t"
+            "slti       %[temp5],       %[temp1],       3277        \n\t"
+            "bnez       %[temp5],       2f                          \n\t"
+            " addiu     %[dr_ptr],      %[dr_ptr],      4           \n\t"
+            "mul        %[temp2],       %[temp2],       %[temp1]    \n\t"
+            "mul        %[temp3],       %[temp3],       %[temp1]    \n\t"
+            "shra_r.w   %[temp2],       %[temp2],       14          \n\t"
+            "shra_r.w   %[temp3],       %[temp3],       14          \n\t"
+            "b          4f                                          \n\t"
+            " nop                                                   \n\t"
+            "2:                                                      \n\t"
+            "addu       %[temp1],       $zero,          $zero       \n\t"
+            "addu       %[temp2],       $zero,          $zero       \n\t"
+            "addu       %[temp3],       $zero,          $zero       \n\t"
+            "b          1f                                          \n\t"
+            " nop                                                   \n\t"
+            "3:                                                      \n\t"
+            "addiu      %[temp1],       $0,             0x4000      \n\t"
+            "1:                                                      \n\t"
+            "sh         %[temp1],       0(%[ptr])                   \n\t"
+            "4:                                                      \n\t"
+            "sh         %[temp2],       0(%[er_ptr])                \n\t"
+            "sh         %[temp3],       2(%[er_ptr])                \n\t"
+            "addiu      %[ptr],         %[ptr],         2           \n\t"
+            "addiu      %[er_ptr],      %[er_ptr],      4           \n\t"
+            ".set       pop                                         \n\t"
+            : [temp1] "=&r"(temp1), [temp2] "=&r"(temp2), [temp3] "=&r"(temp3),
+              [temp4] "=&r"(temp4), [temp5] "=&r"(temp5), [ptr] "+r"(ptr),
+              [er_ptr] "+r"(er_ptr), [dr_ptr] "+r"(dr_ptr)
+            :
+            : "memory", "hi", "lo");
+#else
+        __asm __volatile(
+            ".set       push                                        \n\t"
+            ".set       noreorder                                   \n\t"
+            "lh         %[temp1],       0(%[ptr])                   \n\t"
+            "lh         %[temp2],       0(%[dr_ptr])                \n\t"
+            "slti       %[temp4],       %[temp1],       0x4001      \n\t"
+            "beqz       %[temp4],       3f                          \n\t"
+            " lh        %[temp3],       2(%[dr_ptr])                \n\t"
+            "slti       %[temp5],       %[temp1],       3277        \n\t"
+            "bnez       %[temp5],       2f                          \n\t"
+            " addiu     %[dr_ptr],      %[dr_ptr],      4           \n\t"
+            "mul        %[temp2],       %[temp2],       %[temp1]    \n\t"
+            "mul        %[temp3],       %[temp3],       %[temp1]    \n\t"
+            "addiu      %[temp2],       %[temp2],       0x2000      \n\t"
+            "addiu      %[temp3],       %[temp3],       0x2000      \n\t"
+            "sra        %[temp2],       %[temp2],       14          \n\t"
+            "sra        %[temp3],       %[temp3],       14          \n\t"
+            "b          4f                                          \n\t"
+            " nop                                                   \n\t"
+            "2:                                                      \n\t"
+            "addu       %[temp1],       $zero,          $zero       \n\t"
+            "addu       %[temp2],       $zero,          $zero       \n\t"
+            "addu       %[temp3],       $zero,          $zero       \n\t"
+            "b          1f                                          \n\t"
+            " nop                                                   \n\t"
+            "3:                                                      \n\t"
+            "addiu      %[temp1],       $0,             0x4000      \n\t"
+            "1:                                                      \n\t"
+            "sh         %[temp1],       0(%[ptr])                   \n\t"
+            "4:                                                      \n\t"
+            "sh         %[temp2],       0(%[er_ptr])                \n\t"
+            "sh         %[temp3],       2(%[er_ptr])                \n\t"
+            "addiu      %[ptr],         %[ptr],         2           \n\t"
+            "addiu      %[er_ptr],      %[er_ptr],      4           \n\t"
+            ".set       pop                                         \n\t"
+            : [temp1] "=&r"(temp1), [temp2] "=&r"(temp2), [temp3] "=&r"(temp3),
+              [temp4] "=&r"(temp4), [temp5] "=&r"(temp5), [ptr] "+r"(ptr),
+              [er_ptr] "+r"(er_ptr), [dr_ptr] "+r"(dr_ptr)
+            :
+            : "memory", "hi", "lo");
+#endif
+      }
+    }
+  } else {
+    // multiply with Wiener coefficients
+    for (i = 0; i < PART_LEN1; i++) {
+      efw[i].real = (int16_t)(
+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14));
+      efw[i].imag = (int16_t)(
+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14));
+    }
+  }
+
+  if (aecm->cngMode == AecmTrue) {
+    ComfortNoise(aecm, ptrDfaClean, efw, hnl);
+  }
+
+  InverseFFTAndWindow(aecm, fft, efw, output, nearendClean);
+
+  return 0;
+}
+
+// Generate comfort noise and add to output signal.
+static void ComfortNoise(AecmCore* aecm,
+                         const uint16_t* dfa,
+                         ComplexInt16* out,
+                         const int16_t* lambda) {
+  int16_t i;
+  int16_t tmp16, tmp161, tmp162, tmp163, nrsh1, nrsh2;
+  int32_t tmp32, tmp321, tnoise, tnoise1;
+  int32_t tmp322, tmp323, *tmp1;
+  int16_t* dfap;
+  int16_t* lambdap;
+  const int32_t c2049 = 2049;
+  const int32_t c359 = 359;
+  const int32_t c114 = ONE_Q14;
+
+  int16_t randW16[PART_LEN];
+  int16_t uReal[PART_LEN1];
+  int16_t uImag[PART_LEN1];
+  int32_t outLShift32;
+
+  int16_t shiftFromNearToNoise = kNoiseEstQDomain - aecm->dfaCleanQDomain;
+  int16_t minTrackShift = 9;
+
+  RTC_DCHECK_GE(shiftFromNearToNoise, 0);
+  RTC_DCHECK_LT(shiftFromNearToNoise, 16);
+
+  if (aecm->noiseEstCtr < 100) {
+    // Track the minimum more quickly initially.
+    aecm->noiseEstCtr++;
+    minTrackShift = 6;
+  }
+
+  // Generate a uniform random array on [0 2^15-1].
+  WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
+  int16_t* randW16p = (int16_t*)randW16;
+#if defined(MIPS_DSP_R1_LE)
+  int16_t* kCosTablep = (int16_t*)WebRtcAecm_kCosTable;
+  int16_t* kSinTablep = (int16_t*)WebRtcAecm_kSinTable;
+#endif  // #if defined(MIPS_DSP_R1_LE)
+  tmp1 = (int32_t*)aecm->noiseEst + 1;
+  dfap = (int16_t*)dfa + 1;
+  lambdap = (int16_t*)lambda + 1;
+  // Estimate noise power.
+  for (i = 1; i < PART_LEN1; i += 2) {
+    // Shift to the noise domain.
+    __asm __volatile(
+        "lh     %[tmp32],       0(%[dfap])                              \n\t"
+        "lw     %[tnoise],      0(%[tmp1])                              \n\t"
+        "sllv   %[outLShift32], %[tmp32],   %[shiftFromNearToNoise]     \n\t"
+        : [tmp32] "=&r"(tmp32), [outLShift32] "=r"(outLShift32),
+          [tnoise] "=&r"(tnoise)
+        : [tmp1] "r"(tmp1), [dfap] "r"(dfap),
+          [shiftFromNearToNoise] "r"(shiftFromNearToNoise)
+        : "memory");
+
+    if (outLShift32 < tnoise) {
+      // Reset "too low" counter
+      aecm->noiseEstTooLowCtr[i] = 0;
+      // Track the minimum.
+      if (tnoise < (1 << minTrackShift)) {
+        // For small values, decrease noiseEst[i] every
+        // `kNoiseEstIncCount` block. The regular approach below can not
+        // go further down due to truncation.
+        aecm->noiseEstTooHighCtr[i]++;
+        if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount) {
+          tnoise--;
+          aecm->noiseEstTooHighCtr[i] = 0;  // Reset the counter
+        }
+      } else {
+        __asm __volatile(
+            "subu   %[tmp32],       %[tnoise],      %[outLShift32]      \n\t"
+            "srav   %[tmp32],       %[tmp32],       %[minTrackShift]    \n\t"
+            "subu   %[tnoise],      %[tnoise],      %[tmp32]            \n\t"
+            : [tmp32] "=&r"(tmp32), [tnoise] "+r"(tnoise)
+            :
+            [outLShift32] "r"(outLShift32), [minTrackShift] "r"(minTrackShift));
+      }
+    } else {
+      // Reset "too high" counter
+      aecm->noiseEstTooHighCtr[i] = 0;
+      // Ramp slowly upwards until we hit the minimum again.
+      if ((tnoise >> 19) <= 0) {
+        if ((tnoise >> 11) > 0) {
+          // Large enough for relative increase
+          __asm __volatile(
+              "mul    %[tnoise],  %[tnoise],  %[c2049]    \n\t"
+              "sra    %[tnoise],  %[tnoise],  11          \n\t"
+              : [tnoise] "+r"(tnoise)
+              : [c2049] "r"(c2049)
+              : "hi", "lo");
+        } else {
+          // Make incremental increases based on size every
+          // `kNoiseEstIncCount` block
+          aecm->noiseEstTooLowCtr[i]++;
+          if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount) {
+            __asm __volatile(
+                "sra    %[tmp32],   %[tnoise],  9           \n\t"
+                "addi   %[tnoise],  %[tnoise],  1           \n\t"
+                "addu   %[tnoise],  %[tnoise],  %[tmp32]    \n\t"
+                : [tnoise] "+r"(tnoise), [tmp32] "=&r"(tmp32)
+                :);
+            aecm->noiseEstTooLowCtr[i] = 0;  // Reset counter
+          }
+        }
+      } else {
+        // Avoid overflow.
+        // Multiplication with 2049 will cause wrap around. Scale
+        // down first and then multiply
+        __asm __volatile(
+            "sra    %[tnoise],  %[tnoise],  11          \n\t"
+            "mul    %[tnoise],  %[tnoise],  %[c2049]    \n\t"
+            : [tnoise] "+r"(tnoise)
+            : [c2049] "r"(c2049)
+            : "hi", "lo");
+      }
+    }
+
+    // Shift to the noise domain.
+    __asm __volatile(
+        "lh     %[tmp32],       2(%[dfap])                              \n\t"
+        "lw     %[tnoise1],     4(%[tmp1])                              \n\t"
+        "addiu  %[dfap],        %[dfap],    4                           \n\t"
+        "sllv   %[outLShift32], %[tmp32],   %[shiftFromNearToNoise]     \n\t"
+        : [tmp32] "=&r"(tmp32), [dfap] "+r"(dfap),
+          [outLShift32] "=r"(outLShift32), [tnoise1] "=&r"(tnoise1)
+        : [tmp1] "r"(tmp1), [shiftFromNearToNoise] "r"(shiftFromNearToNoise)
+        : "memory");
+
+    if (outLShift32 < tnoise1) {
+      // Reset "too low" counter
+      aecm->noiseEstTooLowCtr[i + 1] = 0;
+      // Track the minimum.
+      if (tnoise1 < (1 << minTrackShift)) {
+        // For small values, decrease noiseEst[i] every
+        // `kNoiseEstIncCount` block. The regular approach below can not
+        // go further down due to truncation.
+        aecm->noiseEstTooHighCtr[i + 1]++;
+        if (aecm->noiseEstTooHighCtr[i + 1] >= kNoiseEstIncCount) {
+          tnoise1--;
+          aecm->noiseEstTooHighCtr[i + 1] = 0;  // Reset the counter
+        }
+      } else {
+        __asm __volatile(
+            "subu   %[tmp32],       %[tnoise1],     %[outLShift32]      \n\t"
+            "srav   %[tmp32],       %[tmp32],       %[minTrackShift]    \n\t"
+            "subu   %[tnoise1],     %[tnoise1],     %[tmp32]            \n\t"
+            : [tmp32] "=&r"(tmp32), [tnoise1] "+r"(tnoise1)
+            :
+            [outLShift32] "r"(outLShift32), [minTrackShift] "r"(minTrackShift));
+      }
+    } else {
+      // Reset "too high" counter
+      aecm->noiseEstTooHighCtr[i + 1] = 0;
+      // Ramp slowly upwards until we hit the minimum again.
+      if ((tnoise1 >> 19) <= 0) {
+        if ((tnoise1 >> 11) > 0) {
+          // Large enough for relative increase
+          __asm __volatile(
+              "mul    %[tnoise1], %[tnoise1], %[c2049]   \n\t"
+              "sra    %[tnoise1], %[tnoise1], 11         \n\t"
+              : [tnoise1] "+r"(tnoise1)
+              : [c2049] "r"(c2049)
+              : "hi", "lo");
+        } else {
+          // Make incremental increases based on size every
+          // `kNoiseEstIncCount` block
+          aecm->noiseEstTooLowCtr[i + 1]++;
+          if (aecm->noiseEstTooLowCtr[i + 1] >= kNoiseEstIncCount) {
+            __asm __volatile(
+                "sra    %[tmp32],   %[tnoise1], 9           \n\t"
+                "addi   %[tnoise1], %[tnoise1], 1           \n\t"
+                "addu   %[tnoise1], %[tnoise1], %[tmp32]    \n\t"
+                : [tnoise1] "+r"(tnoise1), [tmp32] "=&r"(tmp32)
+                :);
+            aecm->noiseEstTooLowCtr[i + 1] = 0;  // Reset counter
+          }
+        }
+      } else {
+        // Avoid overflow.
+        // Multiplication with 2049 will cause wrap around. Scale
+        // down first and then multiply
+        __asm __volatile(
+            "sra    %[tnoise1], %[tnoise1], 11          \n\t"
+            "mul    %[tnoise1], %[tnoise1], %[c2049]    \n\t"
+            : [tnoise1] "+r"(tnoise1)
+            : [c2049] "r"(c2049)
+            : "hi", "lo");
+      }
+    }
+
+    __asm __volatile(
+        "lh     %[tmp16],   0(%[lambdap])                           \n\t"
+        "lh     %[tmp161],  2(%[lambdap])                           \n\t"
+        "sw     %[tnoise],  0(%[tmp1])                              \n\t"
+        "sw     %[tnoise1], 4(%[tmp1])                              \n\t"
+        "subu   %[tmp16],   %[c114],        %[tmp16]                \n\t"
+        "subu   %[tmp161],  %[c114],        %[tmp161]               \n\t"
+        "srav   %[tmp32],   %[tnoise],      %[shiftFromNearToNoise] \n\t"
+        "srav   %[tmp321],  %[tnoise1],     %[shiftFromNearToNoise] \n\t"
+        "addiu  %[lambdap], %[lambdap],     4                       \n\t"
+        "addiu  %[tmp1],    %[tmp1],        8                       \n\t"
+        : [tmp16] "=&r"(tmp16), [tmp161] "=&r"(tmp161), [tmp1] "+r"(tmp1),
+          [tmp32] "=&r"(tmp32), [tmp321] "=&r"(tmp321), [lambdap] "+r"(lambdap)
+        : [tnoise] "r"(tnoise), [tnoise1] "r"(tnoise1), [c114] "r"(c114),
+          [shiftFromNearToNoise] "r"(shiftFromNearToNoise)
+        : "memory");
+
+    if (tmp32 > 32767) {
+      tmp32 = 32767;
+      aecm->noiseEst[i] = tmp32 << shiftFromNearToNoise;
+    }
+    if (tmp321 > 32767) {
+      tmp321 = 32767;
+      aecm->noiseEst[i + 1] = tmp321 << shiftFromNearToNoise;
+    }
+
+    __asm __volatile(
+        "mul    %[tmp32],   %[tmp32],       %[tmp16]                \n\t"
+        "mul    %[tmp321],  %[tmp321],      %[tmp161]               \n\t"
+        "sra    %[nrsh1],   %[tmp32],       14                      \n\t"
+        "sra    %[nrsh2],   %[tmp321],      14                      \n\t"
+        : [nrsh1] "=&r"(nrsh1), [nrsh2] "=r"(nrsh2)
+        : [tmp16] "r"(tmp16), [tmp161] "r"(tmp161), [tmp32] "r"(tmp32),
+          [tmp321] "r"(tmp321)
+        : "memory", "hi", "lo");
+
+    __asm __volatile(
+        "lh     %[tmp32],       0(%[randW16p])              \n\t"
+        "lh     %[tmp321],      2(%[randW16p])              \n\t"
+        "addiu  %[randW16p],    %[randW16p],    4           \n\t"
+        "mul    %[tmp32],       %[tmp32],       %[c359]     \n\t"
+        "mul    %[tmp321],      %[tmp321],      %[c359]     \n\t"
+        "sra    %[tmp16],       %[tmp32],       15          \n\t"
+        "sra    %[tmp161],      %[tmp321],      15          \n\t"
+        : [randW16p] "+r"(randW16p), [tmp32] "=&r"(tmp32), [tmp16] "=r"(tmp16),
+          [tmp161] "=r"(tmp161), [tmp321] "=&r"(tmp321)
+        : [c359] "r"(c359)
+        : "memory", "hi", "lo");
+
+#if !defined(MIPS_DSP_R1_LE)
+    tmp32 = WebRtcAecm_kCosTable[tmp16];
+    tmp321 = WebRtcAecm_kSinTable[tmp16];
+    tmp322 = WebRtcAecm_kCosTable[tmp161];
+    tmp323 = WebRtcAecm_kSinTable[tmp161];
+#else
+    __asm __volatile(
+        "sll    %[tmp16],       %[tmp16],                   1           \n\t"
+        "sll    %[tmp161],      %[tmp161],                  1           \n\t"
+        "lhx    %[tmp32],       %[tmp16](%[kCosTablep])                 \n\t"
+        "lhx    %[tmp321],      %[tmp16](%[kSinTablep])                 \n\t"
+        "lhx    %[tmp322],      %[tmp161](%[kCosTablep])                \n\t"
+        "lhx    %[tmp323],      %[tmp161](%[kSinTablep])                \n\t"
+        : [tmp32] "=&r"(tmp32), [tmp321] "=&r"(tmp321), [tmp322] "=&r"(tmp322),
+          [tmp323] "=&r"(tmp323)
+        : [kCosTablep] "r"(kCosTablep), [tmp16] "r"(tmp16),
+          [tmp161] "r"(tmp161), [kSinTablep] "r"(kSinTablep)
+        : "memory");
+#endif
+    __asm __volatile(
+        "mul    %[tmp32],       %[tmp32],                   %[nrsh1]    \n\t"
+        "negu   %[tmp162],      %[nrsh1]                                \n\t"
+        "mul    %[tmp322],      %[tmp322],                  %[nrsh2]    \n\t"
+        "negu   %[tmp163],      %[nrsh2]                                \n\t"
+        "sra    %[tmp32],       %[tmp32],                   13          \n\t"
+        "mul    %[tmp321],      %[tmp321],                  %[tmp162]   \n\t"
+        "sra    %[tmp322],      %[tmp322],                  13          \n\t"
+        "mul    %[tmp323],      %[tmp323],                  %[tmp163]   \n\t"
+        "sra    %[tmp321],      %[tmp321],                  13          \n\t"
+        "sra    %[tmp323],      %[tmp323],                  13          \n\t"
+        : [tmp32] "+r"(tmp32), [tmp321] "+r"(tmp321), [tmp162] "=&r"(tmp162),
+          [tmp322] "+r"(tmp322), [tmp323] "+r"(tmp323), [tmp163] "=&r"(tmp163)
+        : [nrsh1] "r"(nrsh1), [nrsh2] "r"(nrsh2)
+        : "hi", "lo");
+    // Tables are in Q13.
+    uReal[i] = (int16_t)tmp32;
+    uImag[i] = (int16_t)tmp321;
+    uReal[i + 1] = (int16_t)tmp322;
+    uImag[i + 1] = (int16_t)tmp323;
+  }
+
+  int32_t tt, sgn;
+  tt = out[0].real;
+  sgn = ((int)tt) >> 31;
+  out[0].real = sgn == (int16_t)(tt >> 15) ? (int16_t)tt : (16384 ^ sgn);
+  tt = out[0].imag;
+  sgn = ((int)tt) >> 31;
+  out[0].imag = sgn == (int16_t)(tt >> 15) ? (int16_t)tt : (16384 ^ sgn);
+  for (i = 1; i < PART_LEN; i++) {
+    tt = out[i].real + uReal[i];
+    sgn = ((int)tt) >> 31;
+    out[i].real = sgn == (int16_t)(tt >> 15) ? (int16_t)tt : (16384 ^ sgn);
+    tt = out[i].imag + uImag[i];
+    sgn = ((int)tt) >> 31;
+    out[i].imag = sgn == (int16_t)(tt >> 15) ? (int16_t)tt : (16384 ^ sgn);
+  }
+  tt = out[PART_LEN].real + uReal[PART_LEN];
+  sgn = ((int)tt) >> 31;
+  out[PART_LEN].real = sgn == (int16_t)(tt >> 15) ? (int16_t)tt : (16384 ^ sgn);
+  tt = out[PART_LEN].imag;
+  sgn = ((int)tt) >> 31;
+  out[PART_LEN].imag = sgn == (int16_t)(tt >> 15) ? (int16_t)tt : (16384 ^ sgn);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_neon.cc b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_neon.cc
new file mode 100644
index 0000000000..584110d3af
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_neon.cc
@@ -0,0 +1,206 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <arm_neon.h>
+
+#include "common_audio/signal_processing/include/real_fft.h"
+#include "modules/audio_processing/aecm/aecm_core.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// TODO(kma): Re-write the corresponding assembly file, the offset
+// generating script and makefile, to replace these C functions.
+
+static inline void AddLanes(uint32_t* ptr, uint32x4_t v) {
+#if defined(WEBRTC_ARCH_ARM64)
+  *(ptr) = vaddvq_u32(v);
+#else
+  uint32x2_t tmp_v;
+  tmp_v = vadd_u32(vget_low_u32(v), vget_high_u32(v));
+  tmp_v = vpadd_u32(tmp_v, tmp_v);
+  *(ptr) = vget_lane_u32(tmp_v, 0);
+#endif
+}
+
+}  // namespace
+
+void WebRtcAecm_CalcLinearEnergiesNeon(AecmCore* aecm,
+                                       const uint16_t* far_spectrum,
+                                       int32_t* echo_est,
+                                       uint32_t* far_energy,
+                                       uint32_t* echo_energy_adapt,
+                                       uint32_t* echo_energy_stored) {
+  int16_t* start_stored_p = aecm->channelStored;
+  int16_t* start_adapt_p = aecm->channelAdapt16;
+  int32_t* echo_est_p = echo_est;
+  const int16_t* end_stored_p = aecm->channelStored + PART_LEN;
+  const uint16_t* far_spectrum_p = far_spectrum;
+  int16x8_t store_v, adapt_v;
+  uint16x8_t spectrum_v;
+  uint32x4_t echo_est_v_low, echo_est_v_high;
+  uint32x4_t far_energy_v, echo_stored_v, echo_adapt_v;
+
+  far_energy_v = vdupq_n_u32(0);
+  echo_adapt_v = vdupq_n_u32(0);
+  echo_stored_v = vdupq_n_u32(0);
+
+  // Get energy for the delayed far end signal and estimated
+  // echo using both stored and adapted channels.
+  // The C code:
+  //  for (i = 0; i < PART_LEN1; i++) {
+  //      echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
+  //                                         far_spectrum[i]);
+  //      (*far_energy) += (uint32_t)(far_spectrum[i]);
+  //      *echo_energy_adapt += aecm->channelAdapt16[i] * far_spectrum[i];
+  //      (*echo_energy_stored) += (uint32_t)echo_est[i];
+  //  }
+  while (start_stored_p < end_stored_p) {
+    spectrum_v = vld1q_u16(far_spectrum_p);
+    adapt_v = vld1q_s16(start_adapt_p);
+    store_v = vld1q_s16(start_stored_p);
+
+    far_energy_v = vaddw_u16(far_energy_v, vget_low_u16(spectrum_v));
+    far_energy_v = vaddw_u16(far_energy_v, vget_high_u16(spectrum_v));
+
+    echo_est_v_low = vmull_u16(vreinterpret_u16_s16(vget_low_s16(store_v)),
+                               vget_low_u16(spectrum_v));
+    echo_est_v_high = vmull_u16(vreinterpret_u16_s16(vget_high_s16(store_v)),
+                                vget_high_u16(spectrum_v));
+    vst1q_s32(echo_est_p, vreinterpretq_s32_u32(echo_est_v_low));
+    vst1q_s32(echo_est_p + 4, vreinterpretq_s32_u32(echo_est_v_high));
+
+    echo_stored_v = vaddq_u32(echo_est_v_low, echo_stored_v);
+    echo_stored_v = vaddq_u32(echo_est_v_high, echo_stored_v);
+
+    echo_adapt_v =
+        vmlal_u16(echo_adapt_v, vreinterpret_u16_s16(vget_low_s16(adapt_v)),
+                  vget_low_u16(spectrum_v));
+    echo_adapt_v =
+        vmlal_u16(echo_adapt_v, vreinterpret_u16_s16(vget_high_s16(adapt_v)),
+                  vget_high_u16(spectrum_v));
+
+    start_stored_p += 8;
+    start_adapt_p += 8;
+    far_spectrum_p += 8;
+    echo_est_p += 8;
+  }
+
+  AddLanes(far_energy, far_energy_v);
+  AddLanes(echo_energy_stored, echo_stored_v);
+  AddLanes(echo_energy_adapt, echo_adapt_v);
+
+  echo_est[PART_LEN] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[PART_LEN],
+                                             far_spectrum[PART_LEN]);
+  *echo_energy_stored += (uint32_t)echo_est[PART_LEN];
+  *far_energy += (uint32_t)far_spectrum[PART_LEN];
+  *echo_energy_adapt += aecm->channelAdapt16[PART_LEN] * far_spectrum[PART_LEN];
+}
+
+void WebRtcAecm_StoreAdaptiveChannelNeon(AecmCore* aecm,
+                                         const uint16_t* far_spectrum,
+                                         int32_t* echo_est) {
+  RTC_DCHECK_EQ(0, (uintptr_t)echo_est % 32);
+  RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelStored % 16);
+  RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelAdapt16 % 16);
+
+  // This is C code of following optimized code.
+  // During startup we store the channel every block.
+  //  memcpy(aecm->channelStored,
+  //         aecm->channelAdapt16,
+  //         sizeof(int16_t) * PART_LEN1);
+  // Recalculate echo estimate
+  //  for (i = 0; i < PART_LEN; i += 4) {
+  //    echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
+  //                                        far_spectrum[i]);
+  //    echo_est[i + 1] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 1],
+  //                                            far_spectrum[i + 1]);
+  //    echo_est[i + 2] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 2],
+  //                                            far_spectrum[i + 2]);
+  //    echo_est[i + 3] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 3],
+  //                                            far_spectrum[i + 3]);
+  //  }
+  //  echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
+  //                                     far_spectrum[i]);
+  const uint16_t* far_spectrum_p = far_spectrum;
+  int16_t* start_adapt_p = aecm->channelAdapt16;
+  int16_t* start_stored_p = aecm->channelStored;
+  const int16_t* end_stored_p = aecm->channelStored + PART_LEN;
+  int32_t* echo_est_p = echo_est;
+
+  uint16x8_t far_spectrum_v;
+  int16x8_t adapt_v;
+  uint32x4_t echo_est_v_low, echo_est_v_high;
+
+  while (start_stored_p < end_stored_p) {
+    far_spectrum_v = vld1q_u16(far_spectrum_p);
+    adapt_v = vld1q_s16(start_adapt_p);
+
+    vst1q_s16(start_stored_p, adapt_v);
+
+    echo_est_v_low = vmull_u16(vget_low_u16(far_spectrum_v),
+                               vget_low_u16(vreinterpretq_u16_s16(adapt_v)));
+    echo_est_v_high = vmull_u16(vget_high_u16(far_spectrum_v),
+                                vget_high_u16(vreinterpretq_u16_s16(adapt_v)));
+
+    vst1q_s32(echo_est_p, vreinterpretq_s32_u32(echo_est_v_low));
+    vst1q_s32(echo_est_p + 4, vreinterpretq_s32_u32(echo_est_v_high));
+
+    far_spectrum_p += 8;
+    start_adapt_p += 8;
+    start_stored_p += 8;
+    echo_est_p += 8;
+  }
+  aecm->channelStored[PART_LEN] = aecm->channelAdapt16[PART_LEN];
+  echo_est[PART_LEN] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[PART_LEN],
+                                             far_spectrum[PART_LEN]);
+}
+
+void WebRtcAecm_ResetAdaptiveChannelNeon(AecmCore* aecm) {
+  RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelStored % 16);
+  RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelAdapt16 % 16);
+  RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelAdapt32 % 32);
+
+  // The C code of following optimized code.
+  // for (i = 0; i < PART_LEN1; i++) {
+  //   aecm->channelAdapt16[i] = aecm->channelStored[i];
+  //   aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
+  //              (int32_t)aecm->channelStored[i], 16);
+  // }
+
+  int16_t* start_stored_p = aecm->channelStored;
+  int16_t* start_adapt16_p = aecm->channelAdapt16;
+  int32_t* start_adapt32_p = aecm->channelAdapt32;
+  const int16_t* end_stored_p = start_stored_p + PART_LEN;
+
+  int16x8_t stored_v;
+  int32x4_t adapt32_v_low, adapt32_v_high;
+
+  while (start_stored_p < end_stored_p) {
+    stored_v = vld1q_s16(start_stored_p);
+    vst1q_s16(start_adapt16_p, stored_v);
+
+    adapt32_v_low = vshll_n_s16(vget_low_s16(stored_v), 16);
+    adapt32_v_high = vshll_n_s16(vget_high_s16(stored_v), 16);
+
+    vst1q_s32(start_adapt32_p, adapt32_v_low);
+    vst1q_s32(start_adapt32_p + 4, adapt32_v_high);
+
+    start_stored_p += 8;
+    start_adapt16_p += 8;
+    start_adapt32_p += 8;
+  }
+  aecm->channelAdapt16[PART_LEN] = aecm->channelStored[PART_LEN];
+  aecm->channelAdapt32[PART_LEN] = (int32_t)aecm->channelStored[PART_LEN] << 16;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/aecm_defines.h b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_defines.h
new file mode 100644
index 0000000000..5805549e2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/aecm_defines.h
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AECM_AECM_DEFINES_H_
+#define MODULES_AUDIO_PROCESSING_AECM_AECM_DEFINES_H_
+
+#define AECM_DYNAMIC_Q /* Turn on/off dynamic Q-domain. */
+
+/* Algorithm parameters */
+#define FRAME_LEN 80 /* Total frame length, 10 ms. */
+
+#define PART_LEN 64      /* Length of partition. */
+#define PART_LEN_SHIFT 7 /* Length of (PART_LEN * 2) in base 2. */
+
+#define PART_LEN1 (PART_LEN + 1)  /* Unique fft coefficients. */
+#define PART_LEN2 (PART_LEN << 1) /* Length of partition * 2. */
+#define PART_LEN4 (PART_LEN << 2) /* Length of partition * 4. */
+#define FAR_BUF_LEN PART_LEN4     /* Length of buffers. */
+#define MAX_DELAY 100
+
+/* Counter parameters */
+#define CONV_LEN 512              /* Convergence length used at startup. */
+#define CONV_LEN2 (CONV_LEN << 1) /* Used at startup. */
+
+/* Energy parameters */
+#define MAX_BUF_LEN 64            /* History length of energy signals. */
+#define FAR_ENERGY_MIN 1025       /* Lowest Far energy level: At least 2 */
+                                  /* in energy. */
+#define FAR_ENERGY_DIFF 929       /* Allowed difference between max */
+                                  /* and min. */
+#define ENERGY_DEV_OFFSET 0       /* The energy error offset in Q8. */
+#define ENERGY_DEV_TOL 400        /* The energy estimation tolerance (Q8). */
+#define FAR_ENERGY_VAD_REGION 230 /* Far VAD tolerance region. */
+
+/* Stepsize parameters */
+#define MU_MIN 10 /* Min stepsize 2^-MU_MIN (far end energy */
+                  /* dependent). */
+#define MU_MAX 1  /* Max stepsize 2^-MU_MAX (far end energy */
+                  /* dependent). */
+#define MU_DIFF 9 /* MU_MIN - MU_MAX */
+
+/* Channel parameters */
+#define MIN_MSE_COUNT 20 /* Min number of consecutive blocks with enough */
+                         /* far end energy to compare channel estimates. */
+#define MIN_MSE_DIFF 29  /* The ratio between adapted and stored channel to */
+                         /* accept a new storage (0.8 in Q-MSE_RESOLUTION). */
+#define MSE_RESOLUTION 5 /* MSE parameter resolution. */
+#define RESOLUTION_CHANNEL16 12 /* W16 Channel in Q-RESOLUTION_CHANNEL16. */
+#define RESOLUTION_CHANNEL32 28 /* W32 Channel in Q-RESOLUTION_CHANNEL. */
+#define CHANNEL_VAD 16          /* Minimum energy in frequency band */
+                                /* to update channel. */
+
+/* Suppression gain parameters: SUPGAIN parameters in Q-(RESOLUTION_SUPGAIN). */
+#define RESOLUTION_SUPGAIN 8 /* Channel in Q-(RESOLUTION_SUPGAIN). */
+#define SUPGAIN_DEFAULT (1 << RESOLUTION_SUPGAIN) /* Default. */
+#define SUPGAIN_ERROR_PARAM_A 3072            /* Estimation error parameter */
+                                              /* (Maximum gain) (8 in Q8). */
+#define SUPGAIN_ERROR_PARAM_B 1536            /* Estimation error parameter */
+                                              /* (Gain before going down). */
+#define SUPGAIN_ERROR_PARAM_D SUPGAIN_DEFAULT /* Estimation error parameter */
+/* (Should be the same as Default) (1 in Q8). */
+#define SUPGAIN_EPC_DT 200 /* SUPGAIN_ERROR_PARAM_C * ENERGY_DEV_TOL */
+
+/* Defines for "check delay estimation" */
+#define CORR_WIDTH 31 /* Number of samples to correlate over. */
+#define CORR_MAX 16   /* Maximum correlation offset. */
+#define CORR_MAX_BUF 63
+#define CORR_DEV 4
+#define CORR_MAX_LEVEL 20
+#define CORR_MAX_LOW 4
+#define CORR_BUF_LEN (CORR_MAX << 1) + 1
+/* Note that CORR_WIDTH + 2*CORR_MAX <= MAX_BUF_LEN. */
+
+#define ONE_Q14 (1 << 14)
+
+/* NLP defines */
+#define NLP_COMP_LOW 3277     /* 0.2 in Q14 */
+#define NLP_COMP_HIGH ONE_Q14 /* 1 in Q14 */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/echo_control_mobile.cc b/third_party/libwebrtc/modules/audio_processing/aecm/echo_control_mobile.cc
new file mode 100644
index 0000000000..14522c0f1d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/echo_control_mobile.cc
@@ -0,0 +1,599 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aecm/echo_control_mobile.h"
+
+#ifdef AEC_DEBUG
+#include <stdio.h>
+#endif
+#include <stdlib.h>
+#include <string.h>
+
+extern "C" {
+#include "common_audio/ring_buffer.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_processing/aecm/aecm_defines.h"
+}
+#include "modules/audio_processing/aecm/aecm_core.h"
+
+namespace webrtc {
+
+namespace {
+
+#define BUF_SIZE_FRAMES 50  // buffer size (frames)
+// Maximum length of resampled signal. Must be an integer multiple of frames
+// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
+// The factor of 2 handles wb, and the + 1 is as a safety margin
+#define MAX_RESAMP_LEN (5 * FRAME_LEN)
+
+static const size_t kBufSizeSamp =
+    BUF_SIZE_FRAMES * FRAME_LEN;  // buffer size (samples)
+static const int kSampMsNb = 8;   // samples per ms in nb
+// Target suppression levels for nlp modes
+// log{0.001, 0.00001, 0.00000001}
+static const int kInitCheck = 42;
+
+typedef struct {
+  int sampFreq;
+  int scSampFreq;
+  short bufSizeStart;
+  int knownDelay;
+
+  // Stores the last frame added to the farend buffer
+  short farendOld[2][FRAME_LEN];
+  short initFlag;  // indicates if AEC has been initialized
+
+  // Variables used for averaging far end buffer size
+  short counter;
+  short sum;
+  short firstVal;
+  short checkBufSizeCtr;
+
+  // Variables used for delay shifts
+  short msInSndCardBuf;
+  short filtDelay;
+  int timeForDelayChange;
+  int ECstartup;
+  int checkBuffSize;
+  int delayChange;
+  short lastDelayDiff;
+
+  int16_t echoMode;
+
+#ifdef AEC_DEBUG
+  FILE* bufFile;
+  FILE* delayFile;
+  FILE* preCompFile;
+  FILE* postCompFile;
+#endif  // AEC_DEBUG
+  // Structures
+  RingBuffer* farendBuf;
+
+  AecmCore* aecmCore;
+} AecMobile;
+
+}  // namespace
+
+// Estimates delay to set the position of the farend buffer read pointer
+// (controlled by knownDelay)
+static int WebRtcAecm_EstBufDelay(AecMobile* aecm, short msInSndCardBuf);
+
+// Stuffs the farend buffer if the estimated delay is too large
+static int WebRtcAecm_DelayComp(AecMobile* aecm);
+
+void* WebRtcAecm_Create() {
+  // Allocate zero-filled memory.
+  AecMobile* aecm = static_cast<AecMobile*>(calloc(1, sizeof(AecMobile)));
+
+  aecm->aecmCore = WebRtcAecm_CreateCore();
+  if (!aecm->aecmCore) {
+    WebRtcAecm_Free(aecm);
+    return NULL;
+  }
+
+  aecm->farendBuf = WebRtc_CreateBuffer(kBufSizeSamp, sizeof(int16_t));
+  if (!aecm->farendBuf) {
+    WebRtcAecm_Free(aecm);
+    return NULL;
+  }
+
+#ifdef AEC_DEBUG
+  aecm->aecmCore->farFile = fopen("aecFar.pcm", "wb");
+  aecm->aecmCore->nearFile = fopen("aecNear.pcm", "wb");
+  aecm->aecmCore->outFile = fopen("aecOut.pcm", "wb");
+  // aecm->aecmCore->outLpFile = fopen("aecOutLp.pcm","wb");
+
+  aecm->bufFile = fopen("aecBuf.dat", "wb");
+  aecm->delayFile = fopen("aecDelay.dat", "wb");
+  aecm->preCompFile = fopen("preComp.pcm", "wb");
+  aecm->postCompFile = fopen("postComp.pcm", "wb");
+#endif  // AEC_DEBUG
+  return aecm;
+}
+
+void WebRtcAecm_Free(void* aecmInst) {
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+
+  if (aecm == NULL) {
+    return;
+  }
+
+#ifdef AEC_DEBUG
+  fclose(aecm->aecmCore->farFile);
+  fclose(aecm->aecmCore->nearFile);
+  fclose(aecm->aecmCore->outFile);
+  // fclose(aecm->aecmCore->outLpFile);
+
+  fclose(aecm->bufFile);
+  fclose(aecm->delayFile);
+  fclose(aecm->preCompFile);
+  fclose(aecm->postCompFile);
+#endif  // AEC_DEBUG
+  WebRtcAecm_FreeCore(aecm->aecmCore);
+  WebRtc_FreeBuffer(aecm->farendBuf);
+  free(aecm);
+}
+
+int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq) {
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+  AecmConfig aecConfig;
+
+  if (aecm == NULL) {
+    return -1;
+  }
+
+  if (sampFreq != 8000 && sampFreq != 16000) {
+    return AECM_BAD_PARAMETER_ERROR;
+  }
+  aecm->sampFreq = sampFreq;
+
+  // Initialize AECM core
+  if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1) {
+    return AECM_UNSPECIFIED_ERROR;
+  }
+
+  // Initialize farend buffer
+  WebRtc_InitBuffer(aecm->farendBuf);
+
+  aecm->initFlag = kInitCheck;  // indicates that initialization has been done
+
+  aecm->delayChange = 1;
+
+  aecm->sum = 0;
+  aecm->counter = 0;
+  aecm->checkBuffSize = 1;
+  aecm->firstVal = 0;
+
+  aecm->ECstartup = 1;
+  aecm->bufSizeStart = 0;
+  aecm->checkBufSizeCtr = 0;
+  aecm->filtDelay = 0;
+  aecm->timeForDelayChange = 0;
+  aecm->knownDelay = 0;
+  aecm->lastDelayDiff = 0;
+
+  memset(&aecm->farendOld, 0, sizeof(aecm->farendOld));
+
+  // Default settings.
+  aecConfig.cngMode = AecmTrue;
+  aecConfig.echoMode = 3;
+
+  if (WebRtcAecm_set_config(aecm, aecConfig) == -1) {
+    return AECM_UNSPECIFIED_ERROR;
+  }
+
+  return 0;
+}
+
+// Returns any error that is caused when buffering the
+// farend signal.
+int32_t WebRtcAecm_GetBufferFarendError(void* aecmInst,
+                                        const int16_t* farend,
+                                        size_t nrOfSamples) {
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+
+  if (aecm == NULL)
+    return -1;
+
+  if (farend == NULL)
+    return AECM_NULL_POINTER_ERROR;
+
+  if (aecm->initFlag != kInitCheck)
+    return AECM_UNINITIALIZED_ERROR;
+
+  if (nrOfSamples != 80 && nrOfSamples != 160)
+    return AECM_BAD_PARAMETER_ERROR;
+
+  return 0;
+}
+
+int32_t WebRtcAecm_BufferFarend(void* aecmInst,
+                                const int16_t* farend,
+                                size_t nrOfSamples) {
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+
+  const int32_t err =
+      WebRtcAecm_GetBufferFarendError(aecmInst, farend, nrOfSamples);
+
+  if (err != 0)
+    return err;
+
+  // TODO(unknown): Is this really a good idea?
+  if (!aecm->ECstartup) {
+    WebRtcAecm_DelayComp(aecm);
+  }
+
+  WebRtc_WriteBuffer(aecm->farendBuf, farend, nrOfSamples);
+
+  return 0;
+}
+
+int32_t WebRtcAecm_Process(void* aecmInst,
+                           const int16_t* nearendNoisy,
+                           const int16_t* nearendClean,
+                           int16_t* out,
+                           size_t nrOfSamples,
+                           int16_t msInSndCardBuf) {
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+  int32_t retVal = 0;
+  size_t i;
+  short nmbrOfFilledBuffers;
+  size_t nBlocks10ms;
+  size_t nFrames;
+#ifdef AEC_DEBUG
+  short msInAECBuf;
+#endif
+
+  if (aecm == NULL) {
+    return -1;
+  }
+
+  if (nearendNoisy == NULL) {
+    return AECM_NULL_POINTER_ERROR;
+  }
+
+  if (out == NULL) {
+    return AECM_NULL_POINTER_ERROR;
+  }
+
+  if (aecm->initFlag != kInitCheck) {
+    return AECM_UNINITIALIZED_ERROR;
+  }
+
+  if (nrOfSamples != 80 && nrOfSamples != 160) {
+    return AECM_BAD_PARAMETER_ERROR;
+  }
+
+  if (msInSndCardBuf < 0) {
+    msInSndCardBuf = 0;
+    retVal = AECM_BAD_PARAMETER_WARNING;
+  } else if (msInSndCardBuf > 500) {
+    msInSndCardBuf = 500;
+    retVal = AECM_BAD_PARAMETER_WARNING;
+  }
+  msInSndCardBuf += 10;
+  aecm->msInSndCardBuf = msInSndCardBuf;
+
+  nFrames = nrOfSamples / FRAME_LEN;
+  nBlocks10ms = nFrames / aecm->aecmCore->mult;
+
+  if (aecm->ECstartup) {
+    if (nearendClean == NULL) {
+      if (out != nearendNoisy) {
+        memcpy(out, nearendNoisy, sizeof(short) * nrOfSamples);
+      }
+    } else if (out != nearendClean) {
+      memcpy(out, nearendClean, sizeof(short) * nrOfSamples);
+    }
+
+    nmbrOfFilledBuffers =
+        (short)WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
+    // The AECM is in the start up mode
+    // AECM is disabled until the soundcard buffer and farend buffers are OK
+
+    // Mechanism to ensure that the soundcard buffer is reasonably stable.
+    if (aecm->checkBuffSize) {
+      aecm->checkBufSizeCtr++;
+      // Before we fill up the far end buffer we require the amount of data on
+      // the sound card to be stable (+/-8 ms) compared to the first value. This
+      // comparison is made during the following 4 consecutive frames. If it
+      // seems to be stable then we start to fill up the far end buffer.
+
+      if (aecm->counter == 0) {
+        aecm->firstVal = aecm->msInSndCardBuf;
+        aecm->sum = 0;
+      }
+
+      if (abs(aecm->firstVal - aecm->msInSndCardBuf) <
+          WEBRTC_SPL_MAX(0.2 * aecm->msInSndCardBuf, kSampMsNb)) {
+        aecm->sum += aecm->msInSndCardBuf;
+        aecm->counter++;
+      } else {
+        aecm->counter = 0;
+      }
+
+      if (aecm->counter * nBlocks10ms >= 6) {
+        // The farend buffer size is determined in blocks of 80 samples
+        // Use 75% of the average value of the soundcard buffer
+        aecm->bufSizeStart = WEBRTC_SPL_MIN(
+            (3 * aecm->sum * aecm->aecmCore->mult) / (aecm->counter * 40),
+            BUF_SIZE_FRAMES);
+        // buffersize has now been determined
+        aecm->checkBuffSize = 0;
+      }
+
+      if (aecm->checkBufSizeCtr * nBlocks10ms > 50) {
+        // for really bad sound cards, don't disable echocanceller for more than
+        // 0.5 sec
+        aecm->bufSizeStart = WEBRTC_SPL_MIN(
+            (3 * aecm->msInSndCardBuf * aecm->aecmCore->mult) / 40,
+            BUF_SIZE_FRAMES);
+        aecm->checkBuffSize = 0;
+      }
+    }
+
+    // if checkBuffSize changed in the if-statement above
+    if (!aecm->checkBuffSize) {
+      // soundcard buffer is now reasonably stable
+      // When the far end buffer is filled with approximately the same amount of
+      // data as the amount on the sound card we end the start up phase and
+      // start to cancel echoes.
+
+      if (nmbrOfFilledBuffers == aecm->bufSizeStart) {
+        aecm->ECstartup = 0;  // Enable the AECM
+      } else if (nmbrOfFilledBuffers > aecm->bufSizeStart) {
+        WebRtc_MoveReadPtr(aecm->farendBuf,
+                           (int)WebRtc_available_read(aecm->farendBuf) -
+                               (int)aecm->bufSizeStart * FRAME_LEN);
+        aecm->ECstartup = 0;
+      }
+    }
+
+  } else {
+    // AECM is enabled
+
+    // Note only 1 block supported for nb and 2 blocks for wb
+    for (i = 0; i < nFrames; i++) {
+      int16_t farend[FRAME_LEN];
+      const int16_t* farend_ptr = NULL;
+
+      nmbrOfFilledBuffers =
+          (short)WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
+
+      // Check that there is data in the far end buffer
+      if (nmbrOfFilledBuffers > 0) {
+        // Get the next 80 samples from the farend buffer
+        WebRtc_ReadBuffer(aecm->farendBuf, (void**)&farend_ptr, farend,
+                          FRAME_LEN);
+
+        // Always store the last frame for use when we run out of data
+        memcpy(&(aecm->farendOld[i][0]), farend_ptr, FRAME_LEN * sizeof(short));
+      } else {
+        // We have no data so we use the last played frame
+        memcpy(farend, &(aecm->farendOld[i][0]), FRAME_LEN * sizeof(short));
+        farend_ptr = farend;
+      }
+
+      // Call buffer delay estimator when all data is extracted,
+      // i,e. i = 0 for NB and i = 1 for WB
+      if ((i == 0 && aecm->sampFreq == 8000) ||
+          (i == 1 && aecm->sampFreq == 16000)) {
+        WebRtcAecm_EstBufDelay(aecm, aecm->msInSndCardBuf);
+      }
+
+      // Call the AECM
+      /*WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearend[FRAME_LEN * i],
+       &out[FRAME_LEN * i], aecm->knownDelay);*/
+      if (WebRtcAecm_ProcessFrame(
+              aecm->aecmCore, farend_ptr, &nearendNoisy[FRAME_LEN * i],
+              (nearendClean ? &nearendClean[FRAME_LEN * i] : NULL),
+              &out[FRAME_LEN * i]) == -1)
+        return -1;
+    }
+  }
+
+#ifdef AEC_DEBUG
+  msInAECBuf = (short)WebRtc_available_read(aecm->farendBuf) /
+               (kSampMsNb * aecm->aecmCore->mult);
+  fwrite(&msInAECBuf, 2, 1, aecm->bufFile);
+  fwrite(&(aecm->knownDelay), sizeof(aecm->knownDelay), 1, aecm->delayFile);
+#endif
+
+  return retVal;
+}
+
+int32_t WebRtcAecm_set_config(void* aecmInst, AecmConfig config) {
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+
+  if (aecm == NULL) {
+    return -1;
+  }
+
+  if (aecm->initFlag != kInitCheck) {
+    return AECM_UNINITIALIZED_ERROR;
+  }
+
+  if (config.cngMode != AecmFalse && config.cngMode != AecmTrue) {
+    return AECM_BAD_PARAMETER_ERROR;
+  }
+  aecm->aecmCore->cngMode = config.cngMode;
+
+  if (config.echoMode < 0 || config.echoMode > 4) {
+    return AECM_BAD_PARAMETER_ERROR;
+  }
+  aecm->echoMode = config.echoMode;
+
+  if (aecm->echoMode == 0) {
+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 3;
+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 3;
+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 3;
+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 3;
+    aecm->aecmCore->supGainErrParamDiffAB =
+        (SUPGAIN_ERROR_PARAM_A >> 3) - (SUPGAIN_ERROR_PARAM_B >> 3);
+    aecm->aecmCore->supGainErrParamDiffBD =
+        (SUPGAIN_ERROR_PARAM_B >> 3) - (SUPGAIN_ERROR_PARAM_D >> 3);
+  } else if (aecm->echoMode == 1) {
+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 2;
+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 2;
+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 2;
+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 2;
+    aecm->aecmCore->supGainErrParamDiffAB =
+        (SUPGAIN_ERROR_PARAM_A >> 2) - (SUPGAIN_ERROR_PARAM_B >> 2);
+    aecm->aecmCore->supGainErrParamDiffBD =
+        (SUPGAIN_ERROR_PARAM_B >> 2) - (SUPGAIN_ERROR_PARAM_D >> 2);
+  } else if (aecm->echoMode == 2) {
+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 1;
+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 1;
+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 1;
+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 1;
+    aecm->aecmCore->supGainErrParamDiffAB =
+        (SUPGAIN_ERROR_PARAM_A >> 1) - (SUPGAIN_ERROR_PARAM_B >> 1);
+    aecm->aecmCore->supGainErrParamDiffBD =
+        (SUPGAIN_ERROR_PARAM_B >> 1) - (SUPGAIN_ERROR_PARAM_D >> 1);
+  } else if (aecm->echoMode == 3) {
+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT;
+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT;
+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
+    aecm->aecmCore->supGainErrParamDiffAB =
+        SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
+    aecm->aecmCore->supGainErrParamDiffBD =
+        SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
+  } else if (aecm->echoMode == 4) {
+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT << 1;
+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT << 1;
+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A << 1;
+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D << 1;
+    aecm->aecmCore->supGainErrParamDiffAB =
+        (SUPGAIN_ERROR_PARAM_A << 1) - (SUPGAIN_ERROR_PARAM_B << 1);
+    aecm->aecmCore->supGainErrParamDiffBD =
+        (SUPGAIN_ERROR_PARAM_B << 1) - (SUPGAIN_ERROR_PARAM_D << 1);
+  }
+
+  return 0;
+}
+
+int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
+                                const void* echo_path,
+                                size_t size_bytes) {
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+  const int16_t* echo_path_ptr = static_cast<const int16_t*>(echo_path);
+
+  if (aecmInst == NULL) {
+    return -1;
+  }
+  if (echo_path == NULL) {
+    return AECM_NULL_POINTER_ERROR;
+  }
+  if (size_bytes != WebRtcAecm_echo_path_size_bytes()) {
+    // Input channel size does not match the size of AECM
+    return AECM_BAD_PARAMETER_ERROR;
+  }
+  if (aecm->initFlag != kInitCheck) {
+    return AECM_UNINITIALIZED_ERROR;
+  }
+
+  WebRtcAecm_InitEchoPathCore(aecm->aecmCore, echo_path_ptr);
+
+  return 0;
+}
+
+int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
+                               void* echo_path,
+                               size_t size_bytes) {
+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
+  int16_t* echo_path_ptr = static_cast<int16_t*>(echo_path);
+
+  if (aecmInst == NULL) {
+    return -1;
+  }
+  if (echo_path == NULL) {
+    return AECM_NULL_POINTER_ERROR;
+  }
+  if (size_bytes != WebRtcAecm_echo_path_size_bytes()) {
+    // Input channel size does not match the size of AECM
+    return AECM_BAD_PARAMETER_ERROR;
+  }
+  if (aecm->initFlag != kInitCheck) {
+    return AECM_UNINITIALIZED_ERROR;
+  }
+
+  memcpy(echo_path_ptr, aecm->aecmCore->channelStored, size_bytes);
+  return 0;
+}
+
+size_t WebRtcAecm_echo_path_size_bytes() {
+  return (PART_LEN1 * sizeof(int16_t));
+}
+
+static int WebRtcAecm_EstBufDelay(AecMobile* aecm, short msInSndCardBuf) {
+  short delayNew, nSampSndCard;
+  short nSampFar = (short)WebRtc_available_read(aecm->farendBuf);
+  short diff;
+
+  nSampSndCard = msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
+
+  delayNew = nSampSndCard - nSampFar;
+
+  if (delayNew < FRAME_LEN) {
+    WebRtc_MoveReadPtr(aecm->farendBuf, FRAME_LEN);
+    delayNew += FRAME_LEN;
+  }
+
+  aecm->filtDelay =
+      WEBRTC_SPL_MAX(0, (8 * aecm->filtDelay + 2 * delayNew) / 10);
+
+  diff = aecm->filtDelay - aecm->knownDelay;
+  if (diff > 224) {
+    if (aecm->lastDelayDiff < 96) {
+      aecm->timeForDelayChange = 0;
+    } else {
+      aecm->timeForDelayChange++;
+    }
+  } else if (diff < 96 && aecm->knownDelay > 0) {
+    if (aecm->lastDelayDiff > 224) {
+      aecm->timeForDelayChange = 0;
+    } else {
+      aecm->timeForDelayChange++;
+    }
+  } else {
+    aecm->timeForDelayChange = 0;
+  }
+  aecm->lastDelayDiff = diff;
+
+  if (aecm->timeForDelayChange > 25) {
+    aecm->knownDelay = WEBRTC_SPL_MAX((int)aecm->filtDelay - 160, 0);
+  }
+  return 0;
+}
+
+static int WebRtcAecm_DelayComp(AecMobile* aecm) {
+  int nSampFar = (int)WebRtc_available_read(aecm->farendBuf);
+  int nSampSndCard, delayNew, nSampAdd;
+  const int maxStuffSamp = 10 * FRAME_LEN;
+
+  nSampSndCard = aecm->msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
+  delayNew = nSampSndCard - nSampFar;
+
+  if (delayNew > FAR_BUF_LEN - FRAME_LEN * aecm->aecmCore->mult) {
+    // The difference of the buffer sizes is larger than the maximum
+    // allowed known delay. Compensate by stuffing the buffer.
+    nSampAdd =
+        (int)(WEBRTC_SPL_MAX(((nSampSndCard >> 1) - nSampFar), FRAME_LEN));
+    nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
+
+    WebRtc_MoveReadPtr(aecm->farendBuf, -nSampAdd);
+    aecm->delayChange = 1;  // the delay needs to be updated
+  }
+
+  return 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aecm/echo_control_mobile.h b/third_party/libwebrtc/modules/audio_processing/aecm/echo_control_mobile.h
new file mode 100644
index 0000000000..ee780524de
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aecm/echo_control_mobile.h
@@ -0,0 +1,209 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
+#define MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+enum { AecmFalse = 0, AecmTrue };
+
+// Errors
+#define AECM_UNSPECIFIED_ERROR 12000
+#define AECM_UNSUPPORTED_FUNCTION_ERROR 12001
+#define AECM_UNINITIALIZED_ERROR 12002
+#define AECM_NULL_POINTER_ERROR 12003
+#define AECM_BAD_PARAMETER_ERROR 12004
+
+// Warnings
+#define AECM_BAD_PARAMETER_WARNING 12100
+
+typedef struct {
+  int16_t cngMode;   // AECM_FALSE, AECM_TRUE (default)
+  int16_t echoMode;  // 0, 1, 2, 3 (default), 4
+} AecmConfig;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * Allocates the memory needed by the AECM. The memory needs to be
+ * initialized separately using the WebRtcAecm_Init() function.
+ * Returns a pointer to the instance and a nullptr at failure.
+ */
+void* WebRtcAecm_Create();
+
+/*
+ * This function releases the memory allocated by WebRtcAecm_Create()
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void*    aecmInst            Pointer to the AECM instance
+ */
+void WebRtcAecm_Free(void* aecmInst);
+
+/*
+ * Initializes an AECM instance.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void*          aecmInst      Pointer to the AECM instance
+ * int32_t        sampFreq      Sampling frequency of data
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * int32_t        return        0: OK
+ *                              1200-12004,12100: error/warning
+ */
+int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq);
+
+/*
+ * Inserts an 80 or 160 sample block of data into the farend buffer.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void*          aecmInst      Pointer to the AECM instance
+ * int16_t*       farend        In buffer containing one frame of
+ *                              farend signal
+ * int16_t        nrOfSamples   Number of samples in farend buffer
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * int32_t        return        0: OK
+ *                              1200-12004,12100: error/warning
+ */
+int32_t WebRtcAecm_BufferFarend(void* aecmInst,
+                                const int16_t* farend,
+                                size_t nrOfSamples);
+
+/*
+ * Reports any errors that would arise when buffering a farend buffer.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void*          aecmInst      Pointer to the AECM instance
+ * int16_t*       farend        In buffer containing one frame of
+ *                              farend signal
+ * int16_t        nrOfSamples   Number of samples in farend buffer
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * int32_t        return        0: OK
+ *                              1200-12004,12100: error/warning
+ */
+int32_t WebRtcAecm_GetBufferFarendError(void* aecmInst,
+                                        const int16_t* farend,
+                                        size_t nrOfSamples);
+
+/*
+ * Runs the AECM on an 80 or 160 sample blocks of data.
+ *
+ * Inputs                        Description
+ * -------------------------------------------------------------------
+ * void*          aecmInst       Pointer to the AECM instance
+ * int16_t*       nearendNoisy   In buffer containing one frame of
+ *                               reference nearend+echo signal. If
+ *                               noise reduction is active, provide
+ *                               the noisy signal here.
+ * int16_t*       nearendClean   In buffer containing one frame of
+ *                               nearend+echo signal. If noise
+ *                               reduction is active, provide the
+ *                               clean signal here. Otherwise pass a
+ *                               NULL pointer.
+ * int16_t        nrOfSamples    Number of samples in nearend buffer
+ * int16_t        msInSndCardBuf Delay estimate for sound card and
+ *                               system buffers
+ *
+ * Outputs                       Description
+ * -------------------------------------------------------------------
+ * int16_t*       out            Out buffer, one frame of processed nearend
+ * int32_t        return         0: OK
+ *                               1200-12004,12100: error/warning
+ */
+int32_t WebRtcAecm_Process(void* aecmInst,
+                           const int16_t* nearendNoisy,
+                           const int16_t* nearendClean,
+                           int16_t* out,
+                           size_t nrOfSamples,
+                           int16_t msInSndCardBuf);
+
+/*
+ * This function enables the user to set certain parameters on-the-fly
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void*          aecmInst      Pointer to the AECM instance
+ * AecmConfig     config        Config instance that contains all
+ *                              properties to be set
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * int32_t        return        0: OK
+ *                              1200-12004,12100: error/warning
+ */
+int32_t WebRtcAecm_set_config(void* aecmInst, AecmConfig config);
+
+/*
+ * This function enables the user to set the echo path on-the-fly.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void*        aecmInst        Pointer to the AECM instance
+ * void*        echo_path       Pointer to the echo path to be set
+ * size_t       size_bytes      Size in bytes of the echo path
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * int32_t      return          0: OK
+ *                              1200-12004,12100: error/warning
+ */
+int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
+                                const void* echo_path,
+                                size_t size_bytes);
+
+/*
+ * This function enables the user to get the currently used echo path
+ * on-the-fly
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void*        aecmInst        Pointer to the AECM instance
+ * void*        echo_path       Pointer to echo path
+ * size_t       size_bytes      Size in bytes of the echo path
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * int32_t      return          0: OK
+ *                              1200-12004,12100: error/warning
+ */
+int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
+                               void* echo_path,
+                               size_t size_bytes);
+
+/*
+ * This function enables the user to get the echo path size in bytes
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * size_t       return          Size in bytes
+ */
+size_t WebRtcAecm_echo_path_size_bytes();
+
+#ifdef __cplusplus
+}
+#endif
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/agc/BUILD.gn
new file mode 100644
index 0000000000..75bef1450f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/BUILD.gn
@@ -0,0 +1,126 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("gain_control_interface") {
+  sources = [ "gain_control.h" ]
+}
+
+rtc_library("agc") {
+  sources = [
+    "agc_manager_direct.cc",
+    "agc_manager_direct.h",
+  ]
+  configs += [ "..:apm_debug_dump" ]
+  deps = [
+    ":gain_control_interface",
+    ":level_estimation",
+    "..:api",
+    "..:apm_logging",
+    "..:audio_buffer",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../common_audio",
+    "../../../common_audio:common_audio_c",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:gtest_prod",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:safe_minmax",
+    "../../../system_wrappers:field_trial",
+    "../../../system_wrappers:metrics",
+    "../agc2:clipping_predictor",
+    "../agc2:gain_map",
+    "../agc2:input_volume_stats_reporter",
+    "../vad",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("level_estimation") {
+  sources = [
+    "agc.cc",
+    "agc.h",
+    "loudness_histogram.cc",
+    "loudness_histogram.h",
+    "utility.cc",
+    "utility.h",
+  ]
+  deps = [
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../vad",
+  ]
+}
+
+rtc_library("legacy_agc") {
+  visibility = [
+    ":*",
+    "..:*",
+  ]  # Only targets in this file and in
+     # audio_processing can depend on
+     # this.
+
+  sources = [
+    "legacy/analog_agc.cc",
+    "legacy/analog_agc.h",
+    "legacy/digital_agc.cc",
+    "legacy/digital_agc.h",
+    "legacy/gain_control.h",
+  ]
+
+  deps = [
+    "../../../common_audio",
+    "../../../common_audio:common_audio_c",
+    "../../../common_audio/third_party/ooura:fft_size_256",
+    "../../../rtc_base:checks",
+    "../../../system_wrappers",
+  ]
+
+  if (rtc_build_with_neon) {
+    if (target_cpu != "arm64") {
+      # Enable compilation for the NEON instruction set.
+      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+      cflags = [ "-mfpu=neon" ]
+    }
+  }
+}
+
+if (rtc_include_tests) {
+  rtc_library("agc_unittests") {
+    testonly = true
+    sources = [
+      "agc_manager_direct_unittest.cc",
+      "loudness_histogram_unittest.cc",
+      "mock_agc.h",
+    ]
+    configs += [ "..:apm_debug_dump" ]
+
+    deps = [
+      ":agc",
+      ":gain_control_interface",
+      ":level_estimation",
+      "..:mocks",
+      "../../../api:array_view",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:random",
+      "../../../rtc_base:safe_conversions",
+      "../../../rtc_base:safe_minmax",
+      "../../../rtc_base:stringutils",
+      "../../../system_wrappers:metrics",
+      "../../../test:field_trial",
+      "../../../test:fileutils",
+      "../../../test:test_support",
+      "//testing/gtest",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/agc.cc b/third_party/libwebrtc/modules/audio_processing/agc/agc.cc
new file mode 100644
index 0000000000..a018ff9f93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/agc.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc/agc.h"
+
+#include <cmath>
+#include <cstdlib>
+#include <vector>
+
+#include "modules/audio_processing/agc/loudness_histogram.h"
+#include "modules/audio_processing/agc/utility.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kDefaultLevelDbfs = -18;
+constexpr int kNumAnalysisFrames = 100;
+constexpr double kActivityThreshold = 0.3;
+constexpr int kNum10msFramesInOneSecond = 100;
+constexpr int kMaxSampleRateHz = 384000;
+
+}  // namespace
+
+Agc::Agc()
+    : target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
+      target_level_dbfs_(kDefaultLevelDbfs),
+      histogram_(LoudnessHistogram::Create(kNumAnalysisFrames)),
+      inactive_histogram_(LoudnessHistogram::Create()) {}
+
+Agc::~Agc() = default;
+
+void Agc::Process(rtc::ArrayView<const int16_t> audio) {
+  const int sample_rate_hz = audio.size() * kNum10msFramesInOneSecond;
+  RTC_DCHECK_LE(sample_rate_hz, kMaxSampleRateHz);
+  vad_.ProcessChunk(audio.data(), audio.size(), sample_rate_hz);
+  const std::vector<double>& rms = vad_.chunkwise_rms();
+  const std::vector<double>& probabilities =
+      vad_.chunkwise_voice_probabilities();
+  RTC_DCHECK_EQ(rms.size(), probabilities.size());
+  for (size_t i = 0; i < rms.size(); ++i) {
+    histogram_->Update(rms[i], probabilities[i]);
+  }
+}
+
+bool Agc::GetRmsErrorDb(int* error) {
+  if (!error) {
+    RTC_DCHECK_NOTREACHED();
+    return false;
+  }
+
+  if (histogram_->num_updates() < kNumAnalysisFrames) {
+    // We haven't yet received enough frames.
+    return false;
+  }
+
+  if (histogram_->AudioContent() < kNumAnalysisFrames * kActivityThreshold) {
+    // We are likely in an inactive segment.
+    return false;
+  }
+
+  double loudness = Linear2Loudness(histogram_->CurrentRms());
+  *error = std::floor(Loudness2Db(target_level_loudness_ - loudness) + 0.5);
+  histogram_->Reset();
+  return true;
+}
+
+void Agc::Reset() {
+  histogram_->Reset();
+}
+
+int Agc::set_target_level_dbfs(int level) {
+  // TODO(turajs): just some arbitrary sanity check. We can come up with better
+  // limits. The upper limit should be chosen such that the risk of clipping is
+  // low. The lower limit should not result in a too quiet signal.
+  if (level >= 0 || level <= -100)
+    return -1;
+  target_level_dbfs_ = level;
+  target_level_loudness_ = Dbfs2Loudness(level);
+  return 0;
+}
+
+int Agc::target_level_dbfs() const {
+  return target_level_dbfs_;
+}
+
+float Agc::voice_probability() const {
+  return vad_.last_voice_probability();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/agc.h b/third_party/libwebrtc/modules/audio_processing/agc/agc.h
new file mode 100644
index 0000000000..da42808225
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/agc.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_AGC_H_
+#define MODULES_AUDIO_PROCESSING_AGC_AGC_H_
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/vad/voice_activity_detector.h"
+
+namespace webrtc {
+
+class LoudnessHistogram;
+
+class Agc {
+ public:
+  Agc();
+  virtual ~Agc();
+
+  // `audio` must be mono; in a multi-channel stream, provide the first (usually
+  // left) channel.
+  virtual void Process(rtc::ArrayView<const int16_t> audio);
+
+  // Retrieves the difference between the target RMS level and the current
+  // signal RMS level in dB. Returns true if an update is available and false
+  // otherwise, in which case `error` should be ignored and no action taken.
+  virtual bool GetRmsErrorDb(int* error);
+  virtual void Reset();
+
+  virtual int set_target_level_dbfs(int level);
+  virtual int target_level_dbfs() const;
+  virtual float voice_probability() const;
+
+ private:
+  double target_level_loudness_;
+  int target_level_dbfs_;
+  std::unique_ptr<LoudnessHistogram> histogram_;
+  std::unique_ptr<LoudnessHistogram> inactive_histogram_;
+  VoiceActivityDetector vad_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_AGC_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/agc_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc/agc_gn/moz.build
new file mode 100644
index 0000000000..45e6cad306
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/agc_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("agc_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct.cc b/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct.cc
new file mode 100644
index 0000000000..b8ad4a8bb9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct.cc
@@ -0,0 +1,713 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc/agc_manager_direct.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "api/array_view.h"
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc/gain_control.h"
+#include "modules/audio_processing/agc2/gain_map_internal.h"
+#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+// Amount of error we tolerate in the microphone level (presumably due to OS
+// quantization) before we assume the user has manually adjusted the microphone.
+constexpr int kLevelQuantizationSlack = 25;
+
+constexpr int kDefaultCompressionGain = 7;
+constexpr int kMaxCompressionGain = 12;
+constexpr int kMinCompressionGain = 2;
+// Controls the rate of compression changes towards the target.
+constexpr float kCompressionGainStep = 0.05f;
+
+constexpr int kMaxMicLevel = 255;
+static_assert(kGainMapSize > kMaxMicLevel, "gain map too small");
+constexpr int kMinMicLevel = 12;
+
+// Prevent very large microphone level changes.
+constexpr int kMaxResidualGainChange = 15;
+
+// Maximum additional gain allowed to compensate for microphone level
+// restrictions from clipping events.
+constexpr int kSurplusCompressionGain = 6;
+
+// Target speech level (dBFs) and speech probability threshold used to compute
+// the RMS error override in `GetSpeechLevelErrorDb()`. These are only used for
+// computing the error override and they are not passed to `agc_`.
+// TODO(webrtc:7494): Move these to a config and pass in the ctor.
+constexpr float kOverrideTargetSpeechLevelDbfs = -18.0f;
+constexpr float kOverrideSpeechProbabilitySilenceThreshold = 0.5f;
+// The minimum number of frames between `UpdateGain()` calls.
+// TODO(webrtc:7494): Move this to a config and pass in the ctor with
+// kOverrideWaitFrames = 100. Default value zero needed for the unit tests.
+constexpr int kOverrideWaitFrames = 0;
+
+using AnalogAgcConfig =
+    AudioProcessing::Config::GainController1::AnalogGainController;
+
+// If the "WebRTC-Audio-2ndAgcMinMicLevelExperiment" field trial is specified,
+// parses it and returns a value between 0 and 255 depending on the field-trial
+// string. Returns an unspecified value if the field trial is not specified, if
+// disabled or if it cannot be parsed. Example:
+// 'WebRTC-Audio-2ndAgcMinMicLevelExperiment/Enabled-80' => returns 80.
+absl::optional<int> GetMinMicLevelOverride() {
+  constexpr char kMinMicLevelFieldTrial[] =
+      "WebRTC-Audio-2ndAgcMinMicLevelExperiment";
+  if (!webrtc::field_trial::IsEnabled(kMinMicLevelFieldTrial)) {
+    return absl::nullopt;
+  }
+  const auto field_trial_string =
+      webrtc::field_trial::FindFullName(kMinMicLevelFieldTrial);
+  int min_mic_level = -1;
+  sscanf(field_trial_string.c_str(), "Enabled-%d", &min_mic_level);
+  if (min_mic_level >= 0 && min_mic_level <= 255) {
+    return min_mic_level;
+  } else {
+    RTC_LOG(LS_WARNING) << "[agc] Invalid parameter for "
+                        << kMinMicLevelFieldTrial << ", ignored.";
+    return absl::nullopt;
+  }
+}
+
+int LevelFromGainError(int gain_error, int level, int min_mic_level) {
+  RTC_DCHECK_GE(level, 0);
+  RTC_DCHECK_LE(level, kMaxMicLevel);
+  if (gain_error == 0) {
+    return level;
+  }
+
+  int new_level = level;
+  if (gain_error > 0) {
+    while (kGainMap[new_level] - kGainMap[level] < gain_error &&
+           new_level < kMaxMicLevel) {
+      ++new_level;
+    }
+  } else {
+    while (kGainMap[new_level] - kGainMap[level] > gain_error &&
+           new_level > min_mic_level) {
+      --new_level;
+    }
+  }
+  return new_level;
+}
+
+// Returns the proportion of samples in the buffer which are at full-scale
+// (and presumably clipped).
+float ComputeClippedRatio(const float* const* audio,
+                          size_t num_channels,
+                          size_t samples_per_channel) {
+  RTC_DCHECK_GT(samples_per_channel, 0);
+  int num_clipped = 0;
+  for (size_t ch = 0; ch < num_channels; ++ch) {
+    int num_clipped_in_ch = 0;
+    for (size_t i = 0; i < samples_per_channel; ++i) {
+      RTC_DCHECK(audio[ch]);
+      if (audio[ch][i] >= 32767.0f || audio[ch][i] <= -32768.0f) {
+        ++num_clipped_in_ch;
+      }
+    }
+    num_clipped = std::max(num_clipped, num_clipped_in_ch);
+  }
+  return static_cast<float>(num_clipped) / (samples_per_channel);
+}
+
+void LogClippingMetrics(int clipping_rate) {
+  RTC_LOG(LS_INFO) << "Input clipping rate: " << clipping_rate << "%";
+  RTC_HISTOGRAM_COUNTS_LINEAR(/*name=*/"WebRTC.Audio.Agc.InputClippingRate",
+                              /*sample=*/clipping_rate, /*min=*/0, /*max=*/100,
+                              /*bucket_count=*/50);
+}
+
+// Computes the speech level error in dB. `speech_level_dbfs` is required to be
+// in the range [-90.0f, 30.0f] and `speech_probability` in the range
+// [0.0f, 1.0f].
+int GetSpeechLevelErrorDb(float speech_level_dbfs, float speech_probability) {
+  constexpr float kMinSpeechLevelDbfs = -90.0f;
+  constexpr float kMaxSpeechLevelDbfs = 30.0f;
+  RTC_DCHECK_GE(speech_level_dbfs, kMinSpeechLevelDbfs);
+  RTC_DCHECK_LE(speech_level_dbfs, kMaxSpeechLevelDbfs);
+  RTC_DCHECK_GE(speech_probability, 0.0f);
+  RTC_DCHECK_LE(speech_probability, 1.0f);
+
+  if (speech_probability < kOverrideSpeechProbabilitySilenceThreshold) {
+    return 0;
+  }
+
+  const float speech_level = rtc::SafeClamp<float>(
+      speech_level_dbfs, kMinSpeechLevelDbfs, kMaxSpeechLevelDbfs);
+
+  return std::round(kOverrideTargetSpeechLevelDbfs - speech_level);
+}
+
+}  // namespace
+
+MonoAgc::MonoAgc(ApmDataDumper* data_dumper,
+                 int clipped_level_min,
+                 bool disable_digital_adaptive,
+                 int min_mic_level)
+    : min_mic_level_(min_mic_level),
+      disable_digital_adaptive_(disable_digital_adaptive),
+      agc_(std::make_unique<Agc>()),
+      max_level_(kMaxMicLevel),
+      max_compression_gain_(kMaxCompressionGain),
+      target_compression_(kDefaultCompressionGain),
+      compression_(target_compression_),
+      compression_accumulator_(compression_),
+      clipped_level_min_(clipped_level_min) {}
+
+MonoAgc::~MonoAgc() = default;
+
+void MonoAgc::Initialize() {
+  max_level_ = kMaxMicLevel;
+  max_compression_gain_ = kMaxCompressionGain;
+  target_compression_ = disable_digital_adaptive_ ? 0 : kDefaultCompressionGain;
+  compression_ = disable_digital_adaptive_ ? 0 : target_compression_;
+  compression_accumulator_ = compression_;
+  capture_output_used_ = true;
+  check_volume_on_next_process_ = true;
+  frames_since_update_gain_ = 0;
+  is_first_frame_ = true;
+}
+
+void MonoAgc::Process(rtc::ArrayView<const int16_t> audio,
+                      absl::optional<int> rms_error_override) {
+  new_compression_to_set_ = absl::nullopt;
+
+  if (check_volume_on_next_process_) {
+    check_volume_on_next_process_ = false;
+    // We have to wait until the first process call to check the volume,
+    // because Chromium doesn't guarantee it to be valid any earlier.
+    CheckVolumeAndReset();
+  }
+
+  agc_->Process(audio);
+
+  // Always check if `agc_` has a new error available. If yes, `agc_` gets
+  // reset.
+  // TODO(webrtc:7494) Replace the `agc_` call `GetRmsErrorDb()` with `Reset()`
+  // if an error override is used.
+  int rms_error = 0;
+  bool update_gain = agc_->GetRmsErrorDb(&rms_error);
+  if (rms_error_override.has_value()) {
+    if (is_first_frame_ || frames_since_update_gain_ < kOverrideWaitFrames) {
+      update_gain = false;
+    } else {
+      rms_error = *rms_error_override;
+      update_gain = true;
+    }
+  }
+
+  if (update_gain) {
+    UpdateGain(rms_error);
+  }
+
+  if (!disable_digital_adaptive_) {
+    UpdateCompressor();
+  }
+
+  is_first_frame_ = false;
+  if (frames_since_update_gain_ < kOverrideWaitFrames) {
+    ++frames_since_update_gain_;
+  }
+}
+
+void MonoAgc::HandleClipping(int clipped_level_step) {
+  RTC_DCHECK_GT(clipped_level_step, 0);
+  // Always decrease the maximum level, even if the current level is below
+  // threshold.
+  SetMaxLevel(std::max(clipped_level_min_, max_level_ - clipped_level_step));
+  if (log_to_histograms_) {
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.AgcClippingAdjustmentAllowed",
+                          level_ - clipped_level_step >= clipped_level_min_);
+  }
+  if (level_ > clipped_level_min_) {
+    // Don't try to adjust the level if we're already below the limit. As
+    // a consequence, if the user has brought the level above the limit, we
+    // will still not react until the postproc updates the level.
+    SetLevel(std::max(clipped_level_min_, level_ - clipped_level_step));
+    // Reset the AGCs for all channels since the level has changed.
+    agc_->Reset();
+    frames_since_update_gain_ = 0;
+    is_first_frame_ = false;
+  }
+}
+
+void MonoAgc::SetLevel(int new_level) {
+  int voe_level = recommended_input_volume_;
+  if (voe_level == 0) {
+    RTC_DLOG(LS_INFO)
+        << "[agc] VolumeCallbacks returned level=0, taking no action.";
+    return;
+  }
+  if (voe_level < 0 || voe_level > kMaxMicLevel) {
+    RTC_LOG(LS_ERROR) << "VolumeCallbacks returned an invalid level="
+                      << voe_level;
+    return;
+  }
+
+  // Detect manual input volume adjustments by checking if the current level
+  // `voe_level` is outside of the `[level_ - kLevelQuantizationSlack, level_ +
+  // kLevelQuantizationSlack]` range where `level_` is the last input volume
+  // known by this gain controller.
+  if (voe_level > level_ + kLevelQuantizationSlack ||
+      voe_level < level_ - kLevelQuantizationSlack) {
+    RTC_DLOG(LS_INFO) << "[agc] Mic volume was manually adjusted. Updating "
+                         "stored level from "
+                      << level_ << " to " << voe_level;
+    level_ = voe_level;
+    // Always allow the user to increase the volume.
+    if (level_ > max_level_) {
+      SetMaxLevel(level_);
+    }
+    // Take no action in this case, since we can't be sure when the volume
+    // was manually adjusted. The compressor will still provide some of the
+    // desired gain change.
+    agc_->Reset();
+    frames_since_update_gain_ = 0;
+    is_first_frame_ = false;
+    return;
+  }
+
+  new_level = std::min(new_level, max_level_);
+  if (new_level == level_) {
+    return;
+  }
+
+  recommended_input_volume_ = new_level;
+  RTC_DLOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", level_=" << level_
+                    << ", new_level=" << new_level;
+  level_ = new_level;
+}
+
+void MonoAgc::SetMaxLevel(int level) {
+  RTC_DCHECK_GE(level, clipped_level_min_);
+  max_level_ = level;
+  // Scale the `kSurplusCompressionGain` linearly across the restricted
+  // level range.
+  max_compression_gain_ =
+      kMaxCompressionGain + std::floor((1.f * kMaxMicLevel - max_level_) /
+                                           (kMaxMicLevel - clipped_level_min_) *
+                                           kSurplusCompressionGain +
+                                       0.5f);
+  RTC_DLOG(LS_INFO) << "[agc] max_level_=" << max_level_
+                    << ", max_compression_gain_=" << max_compression_gain_;
+}
+
+void MonoAgc::HandleCaptureOutputUsedChange(bool capture_output_used) {
+  if (capture_output_used_ == capture_output_used) {
+    return;
+  }
+  capture_output_used_ = capture_output_used;
+
+  if (capture_output_used) {
+    // When we start using the output, we should reset things to be safe.
+    check_volume_on_next_process_ = true;
+  }
+}
+
+int MonoAgc::CheckVolumeAndReset() {
+  int level = recommended_input_volume_;
+  // Reasons for taking action at startup:
+  // 1) A person starting a call is expected to be heard.
+  // 2) Independent of interpretation of `level` == 0 we should raise it so the
+  // AGC can do its job properly.
+  if (level == 0 && !startup_) {
+    RTC_DLOG(LS_INFO)
+        << "[agc] VolumeCallbacks returned level=0, taking no action.";
+    return 0;
+  }
+  if (level < 0 || level > kMaxMicLevel) {
+    RTC_LOG(LS_ERROR) << "[agc] VolumeCallbacks returned an invalid level="
+                      << level;
+    return -1;
+  }
+  RTC_DLOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
+
+  if (level < min_mic_level_) {
+    level = min_mic_level_;
+    RTC_DLOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
+    recommended_input_volume_ = level;
+  }
+  agc_->Reset();
+  level_ = level;
+  startup_ = false;
+  frames_since_update_gain_ = 0;
+  is_first_frame_ = true;
+  return 0;
+}
+
+// Distributes the required gain change between the digital compression stage
+// and volume slider. We use the compressor first, providing a slack region
+// around the current slider position to reduce movement.
+//
+// If the slider needs to be moved, we check first if the user has adjusted
+// it, in which case we take no action and cache the updated level.
+void MonoAgc::UpdateGain(int rms_error_db) {
+  int rms_error = rms_error_db;
+
+  // Always reset the counter regardless of whether the gain is changed
+  // or not. This matches with the bahvior of `agc_` where the histogram is
+  // reset every time an RMS error is successfully read.
+  frames_since_update_gain_ = 0;
+
+  // The compressor will always add at least kMinCompressionGain. In effect,
+  // this adjusts our target gain upward by the same amount and rms_error
+  // needs to reflect that.
+  rms_error += kMinCompressionGain;
+
+  // Handle as much error as possible with the compressor first.
+  int raw_compression =
+      rtc::SafeClamp(rms_error, kMinCompressionGain, max_compression_gain_);
+
+  // Deemphasize the compression gain error. Move halfway between the current
+  // target and the newly received target. This serves to soften perceptible
+  // intra-talkspurt adjustments, at the cost of some adaptation speed.
+  if ((raw_compression == max_compression_gain_ &&
+       target_compression_ == max_compression_gain_ - 1) ||
+      (raw_compression == kMinCompressionGain &&
+       target_compression_ == kMinCompressionGain + 1)) {
+    // Special case to allow the target to reach the endpoints of the
+    // compression range. The deemphasis would otherwise halt it at 1 dB shy.
+    target_compression_ = raw_compression;
+  } else {
+    target_compression_ =
+        (raw_compression - target_compression_) / 2 + target_compression_;
+  }
+
+  // Residual error will be handled by adjusting the volume slider. Use the
+  // raw rather than deemphasized compression here as we would otherwise
+  // shrink the amount of slack the compressor provides.
+  const int residual_gain =
+      rtc::SafeClamp(rms_error - raw_compression, -kMaxResidualGainChange,
+                     kMaxResidualGainChange);
+  RTC_DLOG(LS_INFO) << "[agc] rms_error=" << rms_error
+                    << ", target_compression=" << target_compression_
+                    << ", residual_gain=" << residual_gain;
+  if (residual_gain == 0)
+    return;
+
+  int old_level = level_;
+  SetLevel(LevelFromGainError(residual_gain, level_, min_mic_level_));
+  if (old_level != level_) {
+    // Reset the AGC since the level has changed.
+    agc_->Reset();
+  }
+}
+
+void MonoAgc::UpdateCompressor() {
+  if (compression_ == target_compression_) {
+    return;
+  }
+
+  // Adapt the compression gain slowly towards the target, in order to avoid
+  // highly perceptible changes.
+  if (target_compression_ > compression_) {
+    compression_accumulator_ += kCompressionGainStep;
+  } else {
+    compression_accumulator_ -= kCompressionGainStep;
+  }
+
+  // The compressor accepts integer gains in dB. Adjust the gain when
+  // we've come within half a stepsize of the nearest integer.  (We don't
+  // check for equality due to potential floating point imprecision).
+  int new_compression = compression_;
+  int nearest_neighbor = std::floor(compression_accumulator_ + 0.5);
+  if (std::fabs(compression_accumulator_ - nearest_neighbor) <
+      kCompressionGainStep / 2) {
+    new_compression = nearest_neighbor;
+  }
+
+  // Set the new compression gain.
+  if (new_compression != compression_) {
+    compression_ = new_compression;
+    compression_accumulator_ = new_compression;
+    new_compression_to_set_ = compression_;
+  }
+}
+
+std::atomic<int> AgcManagerDirect::instance_counter_(0);
+
+AgcManagerDirect::AgcManagerDirect(
+    const AudioProcessing::Config::GainController1::AnalogGainController&
+        analog_config,
+    Agc* agc)
+    : AgcManagerDirect(/*num_capture_channels=*/1, analog_config) {
+  RTC_DCHECK(channel_agcs_[0]);
+  RTC_DCHECK(agc);
+  channel_agcs_[0]->set_agc(agc);
+}
+
+AgcManagerDirect::AgcManagerDirect(int num_capture_channels,
+                                   const AnalogAgcConfig& analog_config)
+    : analog_controller_enabled_(analog_config.enabled),
+      min_mic_level_override_(GetMinMicLevelOverride()),
+      data_dumper_(new ApmDataDumper(instance_counter_.fetch_add(1) + 1)),
+      num_capture_channels_(num_capture_channels),
+      disable_digital_adaptive_(!analog_config.enable_digital_adaptive),
+      frames_since_clipped_(analog_config.clipped_wait_frames),
+      capture_output_used_(true),
+      clipped_level_step_(analog_config.clipped_level_step),
+      clipped_ratio_threshold_(analog_config.clipped_ratio_threshold),
+      clipped_wait_frames_(analog_config.clipped_wait_frames),
+      channel_agcs_(num_capture_channels),
+      new_compressions_to_set_(num_capture_channels),
+      clipping_predictor_(
+          CreateClippingPredictor(num_capture_channels,
+                                  analog_config.clipping_predictor)),
+      use_clipping_predictor_step_(
+          !!clipping_predictor_ &&
+          analog_config.clipping_predictor.use_predicted_step),
+      clipping_rate_log_(0.0f),
+      clipping_rate_log_counter_(0) {
+  RTC_LOG(LS_INFO) << "[agc] analog controller enabled: "
+                   << (analog_controller_enabled_ ? "yes" : "no");
+  const int min_mic_level = min_mic_level_override_.value_or(kMinMicLevel);
+  RTC_LOG(LS_INFO) << "[agc] Min mic level: " << min_mic_level
+                   << " (overridden: "
+                   << (min_mic_level_override_.has_value() ? "yes" : "no")
+                   << ")";
+  for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
+    ApmDataDumper* data_dumper_ch = ch == 0 ? data_dumper_.get() : nullptr;
+
+    channel_agcs_[ch] = std::make_unique<MonoAgc>(
+        data_dumper_ch, analog_config.clipped_level_min,
+        disable_digital_adaptive_, min_mic_level);
+  }
+  RTC_DCHECK(!channel_agcs_.empty());
+  RTC_DCHECK_GT(clipped_level_step_, 0);
+  RTC_DCHECK_LE(clipped_level_step_, 255);
+  RTC_DCHECK_GT(clipped_ratio_threshold_, 0.0f);
+  RTC_DCHECK_LT(clipped_ratio_threshold_, 1.0f);
+  RTC_DCHECK_GT(clipped_wait_frames_, 0);
+  channel_agcs_[0]->ActivateLogging();
+}
+
+AgcManagerDirect::~AgcManagerDirect() {}
+
+void AgcManagerDirect::Initialize() {
+  RTC_DLOG(LS_INFO) << "AgcManagerDirect::Initialize";
+  data_dumper_->InitiateNewSetOfRecordings();
+  for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
+    channel_agcs_[ch]->Initialize();
+  }
+  capture_output_used_ = true;
+
+  AggregateChannelLevels();
+  clipping_rate_log_ = 0.0f;
+  clipping_rate_log_counter_ = 0;
+}
+
+void AgcManagerDirect::SetupDigitalGainControl(
+    GainControl& gain_control) const {
+  if (gain_control.set_mode(GainControl::kFixedDigital) != 0) {
+    RTC_LOG(LS_ERROR) << "set_mode(GainControl::kFixedDigital) failed.";
+  }
+  const int target_level_dbfs = disable_digital_adaptive_ ? 0 : 2;
+  if (gain_control.set_target_level_dbfs(target_level_dbfs) != 0) {
+    RTC_LOG(LS_ERROR) << "set_target_level_dbfs() failed.";
+  }
+  const int compression_gain_db =
+      disable_digital_adaptive_ ? 0 : kDefaultCompressionGain;
+  if (gain_control.set_compression_gain_db(compression_gain_db) != 0) {
+    RTC_LOG(LS_ERROR) << "set_compression_gain_db() failed.";
+  }
+  const bool enable_limiter = !disable_digital_adaptive_;
+  if (gain_control.enable_limiter(enable_limiter) != 0) {
+    RTC_LOG(LS_ERROR) << "enable_limiter() failed.";
+  }
+}
+
+void AgcManagerDirect::AnalyzePreProcess(const AudioBuffer& audio_buffer) {
+  const float* const* audio = audio_buffer.channels_const();
+  size_t samples_per_channel = audio_buffer.num_frames();
+  RTC_DCHECK(audio);
+
+  AggregateChannelLevels();
+  if (!capture_output_used_) {
+    return;
+  }
+
+  if (!!clipping_predictor_) {
+    AudioFrameView<const float> frame = AudioFrameView<const float>(
+        audio, num_capture_channels_, static_cast<int>(samples_per_channel));
+    clipping_predictor_->Analyze(frame);
+  }
+
+  // Check for clipped samples, as the AGC has difficulty detecting pitch
+  // under clipping distortion. We do this in the preprocessing phase in order
+  // to catch clipped echo as well.
+  //
+  // If we find a sufficiently clipped frame, drop the current microphone level
+  // and enforce a new maximum level, dropped the same amount from the current
+  // maximum. This harsh treatment is an effort to avoid repeated clipped echo
+  // events. As compensation for this restriction, the maximum compression
+  // gain is increased, through SetMaxLevel().
+  float clipped_ratio =
+      ComputeClippedRatio(audio, num_capture_channels_, samples_per_channel);
+  clipping_rate_log_ = std::max(clipped_ratio, clipping_rate_log_);
+  clipping_rate_log_counter_++;
+  constexpr int kNumFramesIn30Seconds = 3000;
+  if (clipping_rate_log_counter_ == kNumFramesIn30Seconds) {
+    LogClippingMetrics(std::round(100.0f * clipping_rate_log_));
+    clipping_rate_log_ = 0.0f;
+    clipping_rate_log_counter_ = 0;
+  }
+
+  if (frames_since_clipped_ < clipped_wait_frames_) {
+    ++frames_since_clipped_;
+    return;
+  }
+
+  const bool clipping_detected = clipped_ratio > clipped_ratio_threshold_;
+  bool clipping_predicted = false;
+  int predicted_step = 0;
+  if (!!clipping_predictor_) {
+    for (int channel = 0; channel < num_capture_channels_; ++channel) {
+      const auto step = clipping_predictor_->EstimateClippedLevelStep(
+          channel, recommended_input_volume_, clipped_level_step_,
+          channel_agcs_[channel]->min_mic_level(), kMaxMicLevel);
+      if (step.has_value()) {
+        predicted_step = std::max(predicted_step, step.value());
+        clipping_predicted = true;
+      }
+    }
+  }
+  if (clipping_detected) {
+    RTC_DLOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
+                      << clipped_ratio;
+  }
+  int step = clipped_level_step_;
+  if (clipping_predicted) {
+    predicted_step = std::max(predicted_step, clipped_level_step_);
+    RTC_DLOG(LS_INFO) << "[agc] Clipping predicted. step=" << predicted_step;
+    if (use_clipping_predictor_step_) {
+      step = predicted_step;
+    }
+  }
+  if (clipping_detected ||
+      (clipping_predicted && use_clipping_predictor_step_)) {
+    for (auto& state_ch : channel_agcs_) {
+      state_ch->HandleClipping(step);
+    }
+    frames_since_clipped_ = 0;
+    if (!!clipping_predictor_) {
+      clipping_predictor_->Reset();
+    }
+  }
+  AggregateChannelLevels();
+}
+
+void AgcManagerDirect::Process(const AudioBuffer& audio_buffer) {
+  Process(audio_buffer, /*speech_probability=*/absl::nullopt,
+          /*speech_level_dbfs=*/absl::nullopt);
+}
+
+void AgcManagerDirect::Process(const AudioBuffer& audio_buffer,
+                               absl::optional<float> speech_probability,
+                               absl::optional<float> speech_level_dbfs) {
+  AggregateChannelLevels();
+  const int volume_after_clipping_handling = recommended_input_volume_;
+
+  if (!capture_output_used_) {
+    return;
+  }
+
+  const size_t num_frames_per_band = audio_buffer.num_frames_per_band();
+  absl::optional<int> rms_error_override = absl::nullopt;
+  if (speech_probability.has_value() && speech_level_dbfs.has_value()) {
+    rms_error_override =
+        GetSpeechLevelErrorDb(*speech_level_dbfs, *speech_probability);
+  }
+  for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
+    std::array<int16_t, AudioBuffer::kMaxSampleRate / 100> audio_data;
+    int16_t* audio_use = audio_data.data();
+    FloatS16ToS16(audio_buffer.split_bands_const_f(ch)[0], num_frames_per_band,
+                  audio_use);
+    channel_agcs_[ch]->Process({audio_use, num_frames_per_band},
+                               rms_error_override);
+    new_compressions_to_set_[ch] = channel_agcs_[ch]->new_compression();
+  }
+
+  AggregateChannelLevels();
+  if (volume_after_clipping_handling != recommended_input_volume_) {
+    // The recommended input volume was adjusted in order to match the target
+    // level.
+    UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(
+        recommended_input_volume_);
+  }
+}
+
+absl::optional<int> AgcManagerDirect::GetDigitalComressionGain() {
+  return new_compressions_to_set_[channel_controlling_gain_];
+}
+
+void AgcManagerDirect::HandleCaptureOutputUsedChange(bool capture_output_used) {
+  for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
+    channel_agcs_[ch]->HandleCaptureOutputUsedChange(capture_output_used);
+  }
+  capture_output_used_ = capture_output_used;
+}
+
+float AgcManagerDirect::voice_probability() const {
+  float max_prob = 0.f;
+  for (const auto& state_ch : channel_agcs_) {
+    max_prob = std::max(max_prob, state_ch->voice_probability());
+  }
+
+  return max_prob;
+}
+
+void AgcManagerDirect::set_stream_analog_level(int level) {
+  if (!analog_controller_enabled_) {
+    recommended_input_volume_ = level;
+  }
+
+  for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
+    channel_agcs_[ch]->set_stream_analog_level(level);
+  }
+
+  AggregateChannelLevels();
+}
+
+void AgcManagerDirect::AggregateChannelLevels() {
+  int new_recommended_input_volume =
+      channel_agcs_[0]->recommended_analog_level();
+  channel_controlling_gain_ = 0;
+  for (size_t ch = 1; ch < channel_agcs_.size(); ++ch) {
+    int level = channel_agcs_[ch]->recommended_analog_level();
+    if (level < new_recommended_input_volume) {
+      new_recommended_input_volume = level;
+      channel_controlling_gain_ = static_cast<int>(ch);
+    }
+  }
+
+  if (min_mic_level_override_.has_value() && new_recommended_input_volume > 0) {
+    new_recommended_input_volume =
+        std::max(new_recommended_input_volume, *min_mic_level_override_);
+  }
+
+  if (analog_controller_enabled_) {
+    recommended_input_volume_ = new_recommended_input_volume;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct.h b/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct.h
new file mode 100644
index 0000000000..adb2f5a63f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct.h
@@ -0,0 +1,278 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
+#define MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
+
+#include <atomic>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc/agc.h"
+#include "modules/audio_processing/agc2/clipping_predictor.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gtest_prod_util.h"
+
+namespace webrtc {
+
+class MonoAgc;
+class GainControl;
+
+// Adaptive Gain Controller (AGC) that controls the input volume and a digital
+// gain. The input volume controller recommends what volume to use, handles
+// volume changes and clipping. In particular, it handles changes triggered by
+// the user (e.g., volume set to zero by a HW mute button). The digital
+// controller chooses and applies the digital compression gain.
+// This class is not thread-safe.
+// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
+// convention.
+class AgcManagerDirect final {
+ public:
+  // Ctor. `num_capture_channels` specifies the number of channels for the audio
+  // passed to `AnalyzePreProcess()` and `Process()`. Clamps
+  // `analog_config.startup_min_level` in the [12, 255] range.
+  AgcManagerDirect(
+      int num_capture_channels,
+      const AudioProcessing::Config::GainController1::AnalogGainController&
+          analog_config);
+
+  ~AgcManagerDirect();
+  AgcManagerDirect(const AgcManagerDirect&) = delete;
+  AgcManagerDirect& operator=(const AgcManagerDirect&) = delete;
+
+  void Initialize();
+
+  // Configures `gain_control` to work as a fixed digital controller so that the
+  // adaptive part is only handled by this gain controller. Must be called if
+  // `gain_control` is also used to avoid the side-effects of running two AGCs.
+  void SetupDigitalGainControl(GainControl& gain_control) const;
+
+  // Sets the applied input volume.
+  void set_stream_analog_level(int level);
+
+  // TODO(bugs.webrtc.org/7494): Add argument for the applied input volume and
+  // remove `set_stream_analog_level()`.
+  // Analyzes `audio` before `Process()` is called so that the analysis can be
+  // performed before external digital processing operations take place (e.g.,
+  // echo cancellation). The analysis consists of input clipping detection and
+  // prediction (if enabled). Must be called after `set_stream_analog_level()`.
+  void AnalyzePreProcess(const AudioBuffer& audio_buffer);
+
+  // Processes `audio_buffer`. Chooses a digital compression gain and the new
+  // input volume to recommend. Must be called after `AnalyzePreProcess()`. If
+  // `speech_probability` (range [0.0f, 1.0f]) and `speech_level_dbfs` (range
+  // [-90.f, 30.0f]) are given, uses them to override the estimated RMS error.
+  // TODO(webrtc:7494): This signature is needed for testing purposes, unify
+  // the signatures when the clean-up is done.
+  void Process(const AudioBuffer& audio_buffer,
+               absl::optional<float> speech_probability,
+               absl::optional<float> speech_level_dbfs);
+
+  // Processes `audio_buffer`. Chooses a digital compression gain and the new
+  // input volume to recommend. Must be called after `AnalyzePreProcess()`.
+  void Process(const AudioBuffer& audio_buffer);
+
+  // TODO(bugs.webrtc.org/7494): Return recommended input volume and remove
+  // `recommended_analog_level()`.
+  // Returns the recommended input volume. If the input volume contoller is
+  // disabled, returns the input volume set via the latest
+  // `set_stream_analog_level()` call. Must be called after
+  // `AnalyzePreProcess()` and `Process()`.
+  int recommended_analog_level() const { return recommended_input_volume_; }
+
+  // Call when the capture stream output has been flagged to be used/not-used.
+  // If unused, the manager  disregards all incoming audio.
+  void HandleCaptureOutputUsedChange(bool capture_output_used);
+
+  float voice_probability() const;
+
+  int num_channels() const { return num_capture_channels_; }
+
+  // If available, returns the latest digital compression gain that has been
+  // chosen.
+  absl::optional<int> GetDigitalComressionGain();
+
+  // Returns true if clipping prediction is enabled.
+  bool clipping_predictor_enabled() const { return !!clipping_predictor_; }
+
+  // Returns true if clipping prediction is used to adjust the input volume.
+  bool use_clipping_predictor_step() const {
+    return use_clipping_predictor_step_;
+  }
+
+ private:
+  friend class AgcManagerDirectTestHelper;
+
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest, DisableDigitalDisablesDigital);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
+                           AgcMinMicLevelExperimentDefault);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
+                           AgcMinMicLevelExperimentDisabled);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
+                           AgcMinMicLevelExperimentOutOfRangeAbove);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
+                           AgcMinMicLevelExperimentOutOfRangeBelow);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
+                           AgcMinMicLevelExperimentEnabled50);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectTest,
+                           AgcMinMicLevelExperimentEnabledAboveStartupLevel);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
+                           ClippingParametersVerified);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
+                           DisableClippingPredictorDoesNotLowerVolume);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
+                           UsedClippingPredictionsProduceLowerAnalogLevels);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
+                           UnusedClippingPredictionsProduceEqualAnalogLevels);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
+                           EmptyRmsErrorOverrideHasNoEffect);
+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectParametrizedTest,
+                           NonEmptyRmsErrorOverrideHasEffect);
+
+  // Ctor that creates a single channel AGC and by injecting `agc`.
+  // `agc` will be owned by this class; hence, do not delete it.
+  AgcManagerDirect(
+      const AudioProcessing::Config::GainController1::AnalogGainController&
+          analog_config,
+      Agc* agc);
+
+  void AggregateChannelLevels();
+
+  const bool analog_controller_enabled_;
+
+  const absl::optional<int> min_mic_level_override_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  static std::atomic<int> instance_counter_;
+  const int num_capture_channels_;
+  const bool disable_digital_adaptive_;
+
+  int frames_since_clipped_;
+
+  // TODO(bugs.webrtc.org/7494): Create a separate member for the applied input
+  // volume.
+  // TODO(bugs.webrtc.org/7494): Once
+  // `AudioProcessingImpl::recommended_stream_analog_level()` becomes a trivial
+  // getter, leave uninitialized.
+  // Recommended input volume. After `set_stream_analog_level()` is called it
+  // holds the observed input volume. Possibly updated by `AnalyzePreProcess()`
+  // and `Process()`; after these calls, holds the recommended input volume.
+  int recommended_input_volume_ = 0;
+
+  bool capture_output_used_;
+  int channel_controlling_gain_ = 0;
+
+  const int clipped_level_step_;
+  const float clipped_ratio_threshold_;
+  const int clipped_wait_frames_;
+
+  std::vector<std::unique_ptr<MonoAgc>> channel_agcs_;
+  std::vector<absl::optional<int>> new_compressions_to_set_;
+
+  const std::unique_ptr<ClippingPredictor> clipping_predictor_;
+  const bool use_clipping_predictor_step_;
+  float clipping_rate_log_;
+  int clipping_rate_log_counter_;
+};
+
+// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
+// convention.
+class MonoAgc {
+ public:
+  MonoAgc(ApmDataDumper* data_dumper,
+          int clipped_level_min,
+          bool disable_digital_adaptive,
+          int min_mic_level);
+  ~MonoAgc();
+  MonoAgc(const MonoAgc&) = delete;
+  MonoAgc& operator=(const MonoAgc&) = delete;
+
+  void Initialize();
+  void HandleCaptureOutputUsedChange(bool capture_output_used);
+
+  // Sets the current input volume.
+  void set_stream_analog_level(int level) { recommended_input_volume_ = level; }
+
+  // Lowers the recommended input volume in response to clipping based on the
+  // suggested reduction `clipped_level_step`. Must be called after
+  // `set_stream_analog_level()`.
+  void HandleClipping(int clipped_level_step);
+
+  // Analyzes `audio`, requests the RMS error from AGC, updates the recommended
+  // input volume based on the estimated speech level and, if enabled, updates
+  // the (digital) compression gain to be applied by `agc_`. Must be called
+  // after `HandleClipping()`. If `rms_error_override` has a value, RMS error
+  // from AGC is overridden by it.
+  void Process(rtc::ArrayView<const int16_t> audio,
+               absl::optional<int> rms_error_override);
+
+  // Returns the recommended input volume. Must be called after `Process()`.
+  int recommended_analog_level() const { return recommended_input_volume_; }
+
+  float voice_probability() const { return agc_->voice_probability(); }
+  void ActivateLogging() { log_to_histograms_ = true; }
+  absl::optional<int> new_compression() const {
+    return new_compression_to_set_;
+  }
+
+  // Only used for testing.
+  void set_agc(Agc* agc) { agc_.reset(agc); }
+  int min_mic_level() const { return min_mic_level_; }
+
+ private:
+  // Sets a new input volume, after first checking that it hasn't been updated
+  // by the user, in which case no action is taken.
+  void SetLevel(int new_level);
+
+  // Set the maximum input volume the AGC is allowed to apply. Also updates the
+  // maximum compression gain to compensate. The volume must be at least
+  // `kClippedLevelMin`.
+  void SetMaxLevel(int level);
+
+  int CheckVolumeAndReset();
+  void UpdateGain(int rms_error_db);
+  void UpdateCompressor();
+
+  const int min_mic_level_;
+  const bool disable_digital_adaptive_;
+  std::unique_ptr<Agc> agc_;
+  int level_ = 0;
+  int max_level_;
+  int max_compression_gain_;
+  int target_compression_;
+  int compression_;
+  float compression_accumulator_;
+  bool capture_output_used_ = true;
+  bool check_volume_on_next_process_ = true;
+  bool startup_ = true;
+
+  // TODO(bugs.webrtc.org/7494): Create a separate member for the applied
+  // input volume.
+  // Recommended input volume. After `set_stream_analog_level()` is
+  // called, it holds the observed applied input volume. Possibly updated by
+  // `HandleClipping()` and `Process()`; after these calls, holds the
+  // recommended input volume.
+  int recommended_input_volume_ = 0;
+
+  absl::optional<int> new_compression_to_set_;
+  bool log_to_histograms_ = false;
+  const int clipped_level_min_;
+
+  // Frames since the last `UpdateGain()` call.
+  int frames_since_update_gain_ = 0;
+  // Set to true for the first frame after startup and reset, otherwise false.
+  bool is_first_frame_ = true;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc
new file mode 100644
index 0000000000..70ac0b5b34
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc
@@ -0,0 +1,2184 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc/agc_manager_direct.h"
+
+#include <fstream>
+#include <limits>
+#include <tuple>
+#include <vector>
+
+#include "modules/audio_processing/agc/gain_control.h"
+#include "modules/audio_processing/agc/mock_agc.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::DoAll;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+namespace webrtc {
+namespace {
+
+constexpr int kSampleRateHz = 32000;
+constexpr int kNumChannels = 1;
+constexpr int kInitialInputVolume = 128;
+constexpr int kClippedMin = 165;  // Arbitrary, but different from the default.
+constexpr float kAboveClippedThreshold = 0.2f;
+constexpr int kMinMicLevel = 12;
+constexpr int kClippedLevelStep = 15;
+constexpr float kClippedRatioThreshold = 0.1f;
+constexpr int kClippedWaitFrames = 300;
+constexpr float kLowSpeechProbability = 0.1f;
+constexpr float kHighSpeechProbability = 0.7f;
+constexpr float kSpeechLevelDbfs = -25.0f;
+
+constexpr float kMinSample = std::numeric_limits<int16_t>::min();
+constexpr float kMaxSample = std::numeric_limits<int16_t>::max();
+
+using AnalogAgcConfig =
+    AudioProcessing::Config::GainController1::AnalogGainController;
+using ClippingPredictorConfig = AudioProcessing::Config::GainController1::
+    AnalogGainController::ClippingPredictor;
+constexpr AnalogAgcConfig kDefaultAnalogConfig{};
+
+class MockGainControl : public GainControl {
+ public:
+  virtual ~MockGainControl() {}
+  MOCK_METHOD(int, set_stream_analog_level, (int level), (override));
+  MOCK_METHOD(int, stream_analog_level, (), (const, override));
+  MOCK_METHOD(int, set_mode, (Mode mode), (override));
+  MOCK_METHOD(Mode, mode, (), (const, override));
+  MOCK_METHOD(int, set_target_level_dbfs, (int level), (override));
+  MOCK_METHOD(int, target_level_dbfs, (), (const, override));
+  MOCK_METHOD(int, set_compression_gain_db, (int gain), (override));
+  MOCK_METHOD(int, compression_gain_db, (), (const, override));
+  MOCK_METHOD(int, enable_limiter, (bool enable), (override));
+  MOCK_METHOD(bool, is_limiter_enabled, (), (const, override));
+  MOCK_METHOD(int,
+              set_analog_level_limits,
+              (int minimum, int maximum),
+              (override));
+  MOCK_METHOD(int, analog_level_minimum, (), (const, override));
+  MOCK_METHOD(int, analog_level_maximum, (), (const, override));
+  MOCK_METHOD(bool, stream_is_saturated, (), (const, override));
+};
+
+// TODO(bugs.webrtc.org/12874): Remove and use designated initializers once
+// fixed.
+std::unique_ptr<AgcManagerDirect> CreateAgcManagerDirect(
+    int startup_min_volume,
+    int clipped_level_step,
+    float clipped_ratio_threshold,
+    int clipped_wait_frames,
+    const ClippingPredictorConfig& clipping_predictor_config =
+        kDefaultAnalogConfig.clipping_predictor) {
+  AnalogAgcConfig config;
+  config.startup_min_volume = startup_min_volume;
+  config.clipped_level_min = kClippedMin;
+  config.enable_digital_adaptive = false;
+  config.clipped_level_step = clipped_level_step;
+  config.clipped_ratio_threshold = clipped_ratio_threshold;
+  config.clipped_wait_frames = clipped_wait_frames;
+  config.clipping_predictor = clipping_predictor_config;
+  return std::make_unique<AgcManagerDirect>(/*num_capture_channels=*/1, config);
+}
+
+// Deprecated.
+// TODO(bugs.webrtc.org/7494): Delete this helper, use
+// `AgcManagerDirectTestHelper::CallAgcSequence()` instead.
+// Calls `AnalyzePreProcess()` on `manager` `num_calls` times. `peak_ratio` is a
+// value in [0, 1] which determines the amplitude of the samples (1 maps to full
+// scale). The first half of the calls is made on frames which are half filled
+// with zeros in order to simulate a signal with different crest factors.
+void CallPreProcessAudioBuffer(int num_calls,
+                               float peak_ratio,
+                               AgcManagerDirect& manager) {
+  RTC_DCHECK_LE(peak_ratio, 1.0f);
+  AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
+                           kNumChannels, kSampleRateHz, kNumChannels);
+  const int num_channels = audio_buffer.num_channels();
+  const int num_frames = audio_buffer.num_frames();
+
+  // Make half of the calls with half zeroed frames.
+  for (int ch = 0; ch < num_channels; ++ch) {
+    // 50% of the samples in one frame are zero.
+    for (int i = 0; i < num_frames; i += 2) {
+      audio_buffer.channels()[ch][i] = peak_ratio * 32767.0f;
+      audio_buffer.channels()[ch][i + 1] = 0.0f;
+    }
+  }
+  for (int n = 0; n < num_calls / 2; ++n) {
+    manager.AnalyzePreProcess(audio_buffer);
+  }
+
+  // Make the remaining half of the calls with frames whose samples are all set.
+  for (int ch = 0; ch < num_channels; ++ch) {
+    for (int i = 0; i < num_frames; ++i) {
+      audio_buffer.channels()[ch][i] = peak_ratio * 32767.0f;
+    }
+  }
+  for (int n = 0; n < num_calls - num_calls / 2; ++n) {
+    manager.AnalyzePreProcess(audio_buffer);
+  }
+}
+
+constexpr char kMinMicLevelFieldTrial[] =
+    "WebRTC-Audio-2ndAgcMinMicLevelExperiment";
+
+std::string GetAgcMinMicLevelExperimentFieldTrial(const std::string& value) {
+  char field_trial_buffer[64];
+  rtc::SimpleStringBuilder builder(field_trial_buffer);
+  builder << kMinMicLevelFieldTrial << "/" << value << "/";
+  return builder.str();
+}
+
+std::string GetAgcMinMicLevelExperimentFieldTrialEnabled(
+    int enabled_value,
+    const std::string& suffix = "") {
+  RTC_DCHECK_GE(enabled_value, 0);
+  RTC_DCHECK_LE(enabled_value, 255);
+  char field_trial_buffer[64];
+  rtc::SimpleStringBuilder builder(field_trial_buffer);
+  builder << kMinMicLevelFieldTrial << "/Enabled-" << enabled_value << suffix
+          << "/";
+  return builder.str();
+}
+
+std::string GetAgcMinMicLevelExperimentFieldTrial(
+    absl::optional<int> min_mic_level) {
+  if (min_mic_level.has_value()) {
+    return GetAgcMinMicLevelExperimentFieldTrialEnabled(*min_mic_level);
+  }
+  return GetAgcMinMicLevelExperimentFieldTrial("Disabled");
+}
+
+// (Over)writes `samples_value` for the samples in `audio_buffer`.
+// When `clipped_ratio`, a value in [0, 1], is greater than 0, the corresponding
+// fraction of the frame is set to a full scale value to simulate clipping.
+void WriteAudioBufferSamples(float samples_value,
+                             float clipped_ratio,
+                             AudioBuffer& audio_buffer) {
+  RTC_DCHECK_GE(samples_value, kMinSample);
+  RTC_DCHECK_LE(samples_value, kMaxSample);
+  RTC_DCHECK_GE(clipped_ratio, 0.0f);
+  RTC_DCHECK_LE(clipped_ratio, 1.0f);
+  int num_channels = audio_buffer.num_channels();
+  int num_samples = audio_buffer.num_frames();
+  int num_clipping_samples = clipped_ratio * num_samples;
+  for (int ch = 0; ch < num_channels; ++ch) {
+    int i = 0;
+    for (; i < num_clipping_samples; ++i) {
+      audio_buffer.channels()[ch][i] = 32767.0f;
+    }
+    for (; i < num_samples; ++i) {
+      audio_buffer.channels()[ch][i] = samples_value;
+    }
+  }
+}
+
+// Deprecated.
+// TODO(bugs.webrtc.org/7494): Delete this helper, use
+// `AgcManagerDirectTestHelper::CallAgcSequence()` instead.
+void CallPreProcessAndProcess(int num_calls,
+                              const AudioBuffer& audio_buffer,
+                              absl::optional<float> speech_probability_override,
+                              absl::optional<float> speech_level_override,
+                              AgcManagerDirect& manager) {
+  for (int n = 0; n < num_calls; ++n) {
+    manager.AnalyzePreProcess(audio_buffer);
+    manager.Process(audio_buffer, speech_probability_override,
+                    speech_level_override);
+  }
+}
+
+// Reads a given number of 10 ms chunks from a PCM file and feeds them to
+// `AgcManagerDirect`.
+class SpeechSamplesReader {
+ private:
+  // Recording properties.
+  static constexpr int kPcmSampleRateHz = 16000;
+  static constexpr int kPcmNumChannels = 1;
+  static constexpr int kPcmBytesPerSamples = sizeof(int16_t);
+
+ public:
+  SpeechSamplesReader()
+      : is_(test::ResourcePath("audio_processing/agc/agc_audio", "pcm"),
+            std::ios::binary | std::ios::ate),
+        audio_buffer_(kPcmSampleRateHz,
+                      kPcmNumChannels,
+                      kPcmSampleRateHz,
+                      kPcmNumChannels,
+                      kPcmSampleRateHz,
+                      kPcmNumChannels),
+        buffer_(audio_buffer_.num_frames()),
+        buffer_num_bytes_(buffer_.size() * kPcmBytesPerSamples) {
+    RTC_CHECK(is_);
+  }
+
+  // Reads `num_frames` 10 ms frames from the beginning of the PCM file, applies
+  // `gain_db` and feeds the frames into `agc` by calling `AnalyzePreProcess()`
+  // and `Process()` for each frame. Reads the number of 10 ms frames available
+  // in the PCM file if `num_frames` is too large - i.e., does not loop.
+  void Feed(int num_frames, int gain_db, AgcManagerDirect& agc) {
+    float gain = std::pow(10.0f, gain_db / 20.0f);  // From dB to linear gain.
+    is_.seekg(0, is_.beg);  // Start from the beginning of the PCM file.
+
+    // Read and feed frames.
+    for (int i = 0; i < num_frames; ++i) {
+      is_.read(reinterpret_cast<char*>(buffer_.data()), buffer_num_bytes_);
+      if (is_.gcount() < buffer_num_bytes_) {
+        // EOF reached. Stop.
+        break;
+      }
+      // Apply gain and copy samples into `audio_buffer_`.
+      std::transform(buffer_.begin(), buffer_.end(),
+                     audio_buffer_.channels()[0], [gain](int16_t v) -> float {
+                       return rtc::SafeClamp(static_cast<float>(v) * gain,
+                                             kMinSample, kMaxSample);
+                     });
+
+      agc.AnalyzePreProcess(audio_buffer_);
+      agc.Process(audio_buffer_);
+    }
+  }
+
+  // Reads `num_frames` 10 ms frames from the beginning of the PCM file, applies
+  // `gain_db` and feeds the frames into `agc` by calling `AnalyzePreProcess()`
+  // and `Process()` for each frame. Reads the number of 10 ms frames available
+  // in the PCM file if `num_frames` is too large - i.e., does not loop.
+  // `speech_probability_override` and `speech_level_override` are passed to
+  // `Process()` where they are used to override the `agc` RMS error if they
+  // have a value.
+  void Feed(int num_frames,
+            int gain_db,
+            absl::optional<float> speech_probability_override,
+            absl::optional<float> speech_level_override,
+            AgcManagerDirect& agc) {
+    float gain = std::pow(10.0f, gain_db / 20.0f);  // From dB to linear gain.
+    is_.seekg(0, is_.beg);  // Start from the beginning of the PCM file.
+
+    // Read and feed frames.
+    for (int i = 0; i < num_frames; ++i) {
+      is_.read(reinterpret_cast<char*>(buffer_.data()), buffer_num_bytes_);
+      if (is_.gcount() < buffer_num_bytes_) {
+        // EOF reached. Stop.
+        break;
+      }
+      // Apply gain and copy samples into `audio_buffer_`.
+      std::transform(buffer_.begin(), buffer_.end(),
+                     audio_buffer_.channels()[0], [gain](int16_t v) -> float {
+                       return rtc::SafeClamp(static_cast<float>(v) * gain,
+                                             kMinSample, kMaxSample);
+                     });
+
+      agc.AnalyzePreProcess(audio_buffer_);
+      agc.Process(audio_buffer_, speech_probability_override,
+                  speech_level_override);
+    }
+  }
+
+ private:
+  std::ifstream is_;
+  AudioBuffer audio_buffer_;
+  std::vector<int16_t> buffer_;
+  const std::streamsize buffer_num_bytes_;
+};
+
+}  // namespace
+
+// TODO(bugs.webrtc.org/12874): Use constexpr struct with designated
+// initializers once fixed.
+constexpr AnalogAgcConfig GetAnalogAgcTestConfig() {
+  AnalogAgcConfig config;
+  config.enabled = true;
+  config.startup_min_volume = kInitialInputVolume;
+  config.clipped_level_min = kClippedMin;
+  config.enable_digital_adaptive = true;
+  config.clipped_level_step = kClippedLevelStep;
+  config.clipped_ratio_threshold = kClippedRatioThreshold;
+  config.clipped_wait_frames = kClippedWaitFrames;
+  config.clipping_predictor = kDefaultAnalogConfig.clipping_predictor;
+  return config;
+}
+
+constexpr AnalogAgcConfig GetDisabledAnalogAgcConfig() {
+  AnalogAgcConfig config = GetAnalogAgcTestConfig();
+  config.enabled = false;
+  return config;
+}
+
+// Helper class that provides an `AgcManagerDirect` instance with an injected
+// `Agc` mock, an `AudioBuffer` instance and `CallAgcSequence()`, a helper
+// method that runs the `AgcManagerDirect` instance on the `AudioBuffer` one by
+// sticking to the API contract.
+class AgcManagerDirectTestHelper {
+ public:
+  // Ctor. Initializes `audio_buffer` with zeros.
+  AgcManagerDirectTestHelper()
+      : audio_buffer(kSampleRateHz,
+                     kNumChannels,
+                     kSampleRateHz,
+                     kNumChannels,
+                     kSampleRateHz,
+                     kNumChannels),
+        mock_agc(new ::testing::NiceMock<MockAgc>()),
+        manager(GetAnalogAgcTestConfig(), mock_agc) {
+    manager.Initialize();
+    manager.SetupDigitalGainControl(mock_gain_control);
+    WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.0f,
+                            audio_buffer);
+  }
+
+  // Calls the sequence of `AgcManagerDirect` methods according to the API
+  // contract, namely:
+  // - Sets the applied input volume;
+  // - Uses `audio_buffer` to call `AnalyzePreProcess()` and `Process()`;
+  // - Sets the digital compression gain, if specified, on the injected
+  // `mock_agc`. Returns the recommended input volume. The RMS error from
+  // AGC is replaced by an override value if `speech_probability_override`
+  // and `speech_level_override` have a value.
+  int CallAgcSequence(int applied_input_volume,
+                      absl::optional<float> speech_probability_override,
+                      absl::optional<float> speech_level_override) {
+    manager.set_stream_analog_level(applied_input_volume);
+    manager.AnalyzePreProcess(audio_buffer);
+    manager.Process(audio_buffer, speech_probability_override,
+                    speech_level_override);
+    absl::optional<int> digital_gain = manager.GetDigitalComressionGain();
+    if (digital_gain) {
+      mock_gain_control.set_compression_gain_db(*digital_gain);
+    }
+    return manager.recommended_analog_level();
+  }
+
+  // Deprecated.
+  // TODO(bugs.webrtc.org/7494): Let the caller write `audio_buffer` and use
+  // `CallAgcSequence()`. The RMS error from AGC is replaced by an override
+  // value if `speech_probability_override` and `speech_level_override` have
+  // a value.
+  void CallProcess(int num_calls,
+                   absl::optional<float> speech_probability_override,
+                   absl::optional<float> speech_level_override) {
+    for (int i = 0; i < num_calls; ++i) {
+      EXPECT_CALL(*mock_agc, Process(_)).WillOnce(Return());
+      manager.Process(audio_buffer, speech_probability_override,
+                      speech_level_override);
+      absl::optional<int> new_digital_gain = manager.GetDigitalComressionGain();
+      if (new_digital_gain) {
+        mock_gain_control.set_compression_gain_db(*new_digital_gain);
+      }
+    }
+  }
+
+  // Deprecated.
+  // TODO(bugs.webrtc.org/7494): Let the caller write `audio_buffer` and use
+  // `CallAgcSequence()`.
+  void CallPreProc(int num_calls, float clipped_ratio) {
+    RTC_DCHECK_GE(clipped_ratio, 0.0f);
+    RTC_DCHECK_LE(clipped_ratio, 1.0f);
+    WriteAudioBufferSamples(/*samples_value=*/0.0f, clipped_ratio,
+                            audio_buffer);
+    for (int i = 0; i < num_calls; ++i) {
+      manager.AnalyzePreProcess(audio_buffer);
+    }
+  }
+
+  // Deprecated.
+  // TODO(bugs.webrtc.org/7494): Let the caller write `audio_buffer` and use
+  // `CallAgcSequence()`.
+  void CallPreProcForChangingAudio(int num_calls, float peak_ratio) {
+    RTC_DCHECK_GE(peak_ratio, 0.0f);
+    RTC_DCHECK_LE(peak_ratio, 1.0f);
+    const float samples_value = peak_ratio * 32767.0f;
+
+    // Make half of the calls on a frame where the samples alternate
+    // `sample_values` and zeros.
+    WriteAudioBufferSamples(samples_value, /*clipped_ratio=*/0.0f,
+                            audio_buffer);
+    for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
+      for (size_t k = 1; k < audio_buffer.num_frames(); k += 2) {
+        audio_buffer.channels()[ch][k] = 0.0f;
+      }
+    }
+    for (int i = 0; i < num_calls / 2; ++i) {
+      manager.AnalyzePreProcess(audio_buffer);
+    }
+
+    // Make half of thecalls on a frame where all the samples equal
+    // `sample_values`.
+    WriteAudioBufferSamples(samples_value, /*clipped_ratio=*/0.0f,
+                            audio_buffer);
+    for (int i = 0; i < num_calls - num_calls / 2; ++i) {
+      manager.AnalyzePreProcess(audio_buffer);
+    }
+  }
+
+  AudioBuffer audio_buffer;
+  MockAgc* mock_agc;
+  AgcManagerDirect manager;
+  MockGainControl mock_gain_control;
+};
+
+class AgcManagerDirectParametrizedTest
+    : public ::testing::TestWithParam<std::tuple<absl::optional<int>, bool>> {
+ protected:
+  AgcManagerDirectParametrizedTest()
+      : field_trials_(
+            GetAgcMinMicLevelExperimentFieldTrial(std::get<0>(GetParam()))) {}
+
+  bool IsMinMicLevelOverridden() const {
+    return std::get<0>(GetParam()).has_value();
+  }
+  int GetMinMicLevel() const {
+    return std::get<0>(GetParam()).value_or(kMinMicLevel);
+  }
+
+  bool IsRmsErrorOverridden() const { return std::get<1>(GetParam()); }
+  absl::optional<float> GetOverrideOrEmpty(float value) const {
+    return IsRmsErrorOverridden() ? absl::optional<float>(value)
+                                  : absl::nullopt;
+  }
+
+ private:
+  test::ScopedFieldTrials field_trials_;
+};
+
+INSTANTIATE_TEST_SUITE_P(
+    ,
+    AgcManagerDirectParametrizedTest,
+    ::testing::Combine(testing::Values(absl::nullopt, 12, 20),
+                       testing::Bool()));
+
+// Checks that when the analog controller is disabled, no downward adaptation
+// takes place.
+// TODO(webrtc:7494): Revisit the test after moving the number of override wait
+// frames to AMP config. The test passes but internally the gain update timing
+// differs.
+TEST_P(AgcManagerDirectParametrizedTest,
+       DisabledAnalogAgcDoesNotAdaptDownwards) {
+  AgcManagerDirect manager_no_analog_agc(kNumChannels,
+                                         GetDisabledAnalogAgcConfig());
+  manager_no_analog_agc.Initialize();
+  AgcManagerDirect manager_with_analog_agc(kNumChannels,
+                                           GetAnalogAgcTestConfig());
+  manager_with_analog_agc.Initialize();
+
+  AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
+                           kNumChannels, kSampleRateHz, kNumChannels);
+
+  constexpr int kAnalogLevel = 250;
+  static_assert(kAnalogLevel > kInitialInputVolume, "Increase `kAnalogLevel`.");
+  manager_no_analog_agc.set_stream_analog_level(kAnalogLevel);
+  manager_with_analog_agc.set_stream_analog_level(kAnalogLevel);
+
+  // Make a first call with input that doesn't clip in order to let the
+  // controller read the input volume. That is needed because clipping input
+  // causes the controller to stay in idle state for
+  // `AnalogAgcConfig::clipped_wait_frames` frames.
+  WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipping_ratio=*/0.0f,
+                          audio_buffer);
+  manager_no_analog_agc.AnalyzePreProcess(audio_buffer);
+  manager_with_analog_agc.AnalyzePreProcess(audio_buffer);
+  manager_no_analog_agc.Process(audio_buffer,
+                                GetOverrideOrEmpty(kHighSpeechProbability),
+                                GetOverrideOrEmpty(-18.0f));
+  manager_with_analog_agc.Process(audio_buffer,
+                                  GetOverrideOrEmpty(kHighSpeechProbability),
+                                  GetOverrideOrEmpty(-18.0f));
+
+  // Feed clipping input to trigger a downward adapation of the analog level.
+  WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipping_ratio=*/0.2f,
+                          audio_buffer);
+  manager_no_analog_agc.AnalyzePreProcess(audio_buffer);
+  manager_with_analog_agc.AnalyzePreProcess(audio_buffer);
+  manager_no_analog_agc.Process(audio_buffer,
+                                GetOverrideOrEmpty(kHighSpeechProbability),
+                                GetOverrideOrEmpty(-10.0f));
+  manager_with_analog_agc.Process(audio_buffer,
+                                  GetOverrideOrEmpty(kHighSpeechProbability),
+                                  GetOverrideOrEmpty(-10.0f));
+
+  // Check that no adaptation occurs when the analog controller is disabled
+  // and make sure that the test triggers a downward adaptation otherwise.
+  EXPECT_EQ(manager_no_analog_agc.recommended_analog_level(), kAnalogLevel);
+  ASSERT_LT(manager_with_analog_agc.recommended_analog_level(), kAnalogLevel);
+}
+
+// Checks that when the analog controller is disabled, no upward adaptation
+// takes place.
+// TODO(webrtc:7494): Revisit the test after moving the number of override wait
+// frames to APM config. The test passes but internally the gain update timing
+// differs.
+TEST_P(AgcManagerDirectParametrizedTest, DisabledAnalogAgcDoesNotAdaptUpwards) {
+  AgcManagerDirect manager_no_analog_agc(kNumChannels,
+                                         GetDisabledAnalogAgcConfig());
+  manager_no_analog_agc.Initialize();
+  AgcManagerDirect manager_with_analog_agc(kNumChannels,
+                                           GetAnalogAgcTestConfig());
+  manager_with_analog_agc.Initialize();
+
+  constexpr int kAnalogLevel = kInitialInputVolume;
+  manager_no_analog_agc.set_stream_analog_level(kAnalogLevel);
+  manager_with_analog_agc.set_stream_analog_level(kAnalogLevel);
+
+  // Feed speech with low energy to trigger an upward adapation of the analog
+  // level.
+  constexpr int kNumFrames = 125;
+  constexpr int kGainDb = -20;
+  SpeechSamplesReader reader;
+  reader.Feed(kNumFrames, kGainDb, GetOverrideOrEmpty(kHighSpeechProbability),
+              GetOverrideOrEmpty(-42.0f), manager_no_analog_agc);
+  reader.Feed(kNumFrames, kGainDb, GetOverrideOrEmpty(kHighSpeechProbability),
+              GetOverrideOrEmpty(-42.0f), manager_with_analog_agc);
+
+  // Check that no adaptation occurs when the analog controller is disabled
+  // and make sure that the test triggers an upward adaptation otherwise.
+  EXPECT_EQ(manager_no_analog_agc.recommended_analog_level(), kAnalogLevel);
+  ASSERT_GT(manager_with_analog_agc.recommended_analog_level(), kAnalogLevel);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       StartupMinVolumeConfigurationIsRespected) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+  EXPECT_EQ(kInitialInputVolume, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, MicVolumeResponseToRmsError) {
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  // Compressor default; no residual error.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-23.0f));
+
+  // Inside the compressor's window; no change of volume.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-28.0f));
+
+  // Above the compressor's window; volume should be increased.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-29.0f));
+  EXPECT_EQ(130, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-38.0f));
+  EXPECT_EQ(168, helper.manager.recommended_analog_level());
+
+  // Inside the compressor's window; no change of volume.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-23.0f));
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-18.0f));
+
+  // Below the compressor's window; volume should be decreased.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-17.0f));
+  EXPECT_EQ(167, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-17.0f));
+  EXPECT_EQ(163, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-9), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-9.0f));
+  EXPECT_EQ(129, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, MicVolumeIsLimited) {
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  // Maximum upwards change is limited.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(183, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(243, helper.manager.recommended_analog_level());
+
+  // Won't go higher than the maximum.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(255, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-17.0f));
+  EXPECT_EQ(254, helper.manager.recommended_analog_level());
+
+  // Maximum downwards change is limited.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(22.0f));
+  EXPECT_EQ(194, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(22.0f));
+  EXPECT_EQ(137, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(22.0f));
+  EXPECT_EQ(88, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(22.0f));
+  EXPECT_EQ(54, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(22.0f));
+  EXPECT_EQ(33, helper.manager.recommended_analog_level());
+
+  // Won't go lower than the minimum.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(22.0f));
+  EXPECT_EQ(std::max(18, GetMinMicLevel()),
+            helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(22.0f));
+  EXPECT_EQ(std::max(12, GetMinMicLevel()),
+            helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, CompressorStepsTowardsTarget) {
+  constexpr absl::optional<float> kNoOverride = absl::nullopt;
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  // Compressor default; no call to set_compression_gain_db.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-23.0f));
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  // The mock `GetRmsErrorDb()` returns false; mimic this by passing
+  // absl::nullopt as an override.
+  helper.CallProcess(/*num_calls=*/19, kNoOverride, kNoOverride);
+
+  // Moves slowly upwards.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-27.0f));
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/18, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(8))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/19, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(9))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+
+  // Moves slowly downward, then reverses before reaching the original target.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-23.0f));
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/18, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(8))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
+      .WillRepeatedly(Return(false));
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-27.0f));
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/18, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(9))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, CompressorErrorIsDeemphasized) {
+  constexpr absl::optional<float> kNoOverride = absl::nullopt;
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillRepeatedly(Return(false));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-28.0f));
+  // The mock `GetRmsErrorDb()` returns false; mimic this by passing
+  // absl::nullopt as an override.
+  helper.CallProcess(/*num_calls=*/18, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(8))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(9))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillRepeatedly(Return(false));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-18.0f));
+  helper.CallProcess(/*num_calls=*/18, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(8))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(7))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(6))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(_)).Times(0);
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, CompressorReachesMaximum) {
+  constexpr absl::optional<float> kNoOverride = absl::nullopt;
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
+      .WillRepeatedly(Return(false));
+  helper.CallProcess(/*num_calls=*/4, speech_probability_override,
+                     GetOverrideOrEmpty(-28.0f));
+  // The mock `GetRmsErrorDb()` returns false; mimic this by passing
+  // absl::nullopt as an override.
+  helper.CallProcess(/*num_calls=*/15, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(8))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(9))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(10))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(11))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(12))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, CompressorReachesMinimum) {
+  constexpr absl::optional<float> kNoOverride = absl::nullopt;
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
+      .WillRepeatedly(Return(false));
+  helper.CallProcess(/*num_calls=*/4, speech_probability_override,
+                     GetOverrideOrEmpty(-18.0f));
+  // The mock `GetRmsErrorDb()` returns false; mimic this by passing
+  // absl::nullopt as an override.
+  helper.CallProcess(/*num_calls=*/15, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(6))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(5))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(4))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(3))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(2))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, NoActionWhileMuted) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  helper.manager.HandleCaptureOutputUsedChange(false);
+  helper.manager.Process(helper.audio_buffer,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  absl::optional<int> new_digital_gain =
+      helper.manager.GetDigitalComressionGain();
+  if (new_digital_gain) {
+    helper.mock_gain_control.set_compression_gain_db(*new_digital_gain);
+  }
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, UnmutingChecksVolumeWithoutRaising) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  helper.manager.HandleCaptureOutputUsedChange(false);
+  helper.manager.HandleCaptureOutputUsedChange(true);
+
+  constexpr int kInputVolume = 127;
+  helper.manager.set_stream_analog_level(kInputVolume);
+  EXPECT_CALL(*helper.mock_agc, Reset());
+
+  // SetMicVolume should not be called.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_)).WillOnce(Return(false));
+  helper.CallProcess(/*num_calls=*/1,
+                     GetOverrideOrEmpty(kHighSpeechProbability),
+                     GetOverrideOrEmpty(kSpeechLevelDbfs));
+  EXPECT_EQ(127, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, UnmutingRaisesTooLowVolume) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  helper.manager.HandleCaptureOutputUsedChange(false);
+  helper.manager.HandleCaptureOutputUsedChange(true);
+
+  constexpr int kInputVolume = 11;
+  helper.manager.set_stream_analog_level(kInputVolume);
+  EXPECT_CALL(*helper.mock_agc, Reset());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_)).WillOnce(Return(false));
+  helper.CallProcess(/*num_calls=*/1,
+                     GetOverrideOrEmpty(kHighSpeechProbability),
+                     GetOverrideOrEmpty(kSpeechLevelDbfs));
+  EXPECT_EQ(GetMinMicLevel(), helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       ManualLevelChangeResultsInNoSetMicCall) {
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  // Change outside of compressor's range, which would normally trigger a call
+  // to `SetMicVolume()`.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
+
+  // When the analog volume changes, the gain controller is reset.
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+
+  // GetMicVolume returns a value outside of the quantization slack, indicating
+  // a manual volume change.
+  ASSERT_NE(helper.manager.recommended_analog_level(), 154);
+  helper.manager.set_stream_analog_level(154);
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-29.0f));
+  EXPECT_EQ(154, helper.manager.recommended_analog_level());
+
+  // Do the same thing, except downwards now.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  helper.manager.set_stream_analog_level(100);
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-17.0f));
+  EXPECT_EQ(100, helper.manager.recommended_analog_level());
+
+  // And finally verify the AGC continues working without a manual change.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-17.0f));
+  EXPECT_EQ(99, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       RecoveryAfterManualLevelChangeFromMax) {
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  // Force the mic up to max volume. Takes a few steps due to the residual
+  // gain limitation.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(183, helper.manager.recommended_analog_level());
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(243, helper.manager.recommended_analog_level());
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(255, helper.manager.recommended_analog_level());
+
+  // Manual change does not result in SetMicVolume call.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  helper.manager.set_stream_analog_level(50);
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-17.0f));
+  EXPECT_EQ(50, helper.manager.recommended_analog_level());
+
+  // Continues working as usual afterwards.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-38.0f));
+
+  EXPECT_EQ(69, helper.manager.recommended_analog_level());
+}
+
+// Checks that, when the min mic level override is not specified, AGC ramps up
+// towards the minimum mic level after the mic level is manually set below the
+// minimum gain to enforce.
+TEST_P(AgcManagerDirectParametrizedTest,
+       RecoveryAfterManualLevelChangeBelowMinWithoutMiMicLevelnOverride) {
+  if (IsMinMicLevelOverridden()) {
+    GTEST_SKIP() << "Skipped. Min mic level overridden.";
+  }
+
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  // Manual change below min, but strictly positive, otherwise AGC won't take
+  // any action.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  helper.manager.set_stream_analog_level(1);
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-17.0f));
+  EXPECT_EQ(1, helper.manager.recommended_analog_level());
+
+  // Continues working as usual afterwards.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-29.0f));
+  EXPECT_EQ(2, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(11, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-38.0f));
+  EXPECT_EQ(18, helper.manager.recommended_analog_level());
+}
+
+// Checks that, when the min mic level override is specified, AGC immediately
+// applies the minimum mic level after the mic level is manually set below the
+// minimum gain to enforce.
+TEST_P(AgcManagerDirectParametrizedTest,
+       RecoveryAfterManualLevelChangeBelowMin) {
+  if (!IsMinMicLevelOverridden()) {
+    GTEST_SKIP() << "Skipped. Min mic level not overridden.";
+  }
+
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume, speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  // Manual change below min, but strictly positive, otherwise
+  // AGC won't take any action.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
+  helper.manager.set_stream_analog_level(1);
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-17.0f));
+  EXPECT_EQ(GetMinMicLevel(), helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, NoClippingHasNoImpact) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  helper.CallPreProc(/*num_calls=*/100, /*clipped_ratio=*/0);
+  EXPECT_EQ(128, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, ClippingUnderThresholdHasNoImpact) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/0.099);
+  EXPECT_EQ(128, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, ClippingLowersVolume) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/255,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/0.2);
+  EXPECT_EQ(240, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, WaitingPeriodBetweenClippingChecks) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/255,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(240, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(0);
+  helper.CallPreProc(/*num_calls=*/300,
+                     /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(240, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(225, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, ClippingLoweringIsLimited) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/180,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(kClippedMin, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(0);
+  helper.CallPreProc(/*num_calls=*/1000,
+                     /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(kClippedMin, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       ClippingMaxIsRespectedWhenEqualToLevel) {
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/255,
+                         speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(240, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
+  helper.CallProcess(/*num_calls=*/10, speech_probability_override,
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(240, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       ClippingMaxIsRespectedWhenHigherThanLevel) {
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/200,
+                         speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(185, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(40), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-58.0f));
+  EXPECT_EQ(240, helper.manager.recommended_analog_level());
+  helper.CallProcess(/*num_calls=*/10, speech_probability_override,
+                     GetOverrideOrEmpty(-58.0f));
+  EXPECT_EQ(240, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       MaxCompressionIsIncreasedAfterClipping) {
+  constexpr absl::optional<float> kNoOverride = absl::nullopt;
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/210,
+                         speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(/*num_calls=*/1, kAboveClippedThreshold);
+  EXPECT_EQ(195, helper.manager.recommended_analog_level());
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
+      .WillRepeatedly(Return(false));
+  helper.CallProcess(/*num_calls=*/5, speech_probability_override,
+                     GetOverrideOrEmpty(-29.0f));
+  // The mock `GetRmsErrorDb()` returns false; mimic this by passing
+  // absl::nullopt as an override.
+  helper.CallProcess(/*num_calls=*/14, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(8))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(9))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(10))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(11))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(12))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(13))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+
+  // Continue clipping until we hit the maximum surplus compression.
+  helper.CallPreProc(/*num_calls=*/300,
+                     /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(180, helper.manager.recommended_analog_level());
+
+  helper.CallPreProc(/*num_calls=*/300,
+                     /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(1, kAboveClippedThreshold);
+  EXPECT_EQ(kClippedMin, helper.manager.recommended_analog_level());
+
+  // Current level is now at the minimum, but the maximum allowed level still
+  // has more to decrease.
+  helper.CallPreProc(/*num_calls=*/300,
+                     /*clipped_ratio=*/kAboveClippedThreshold);
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+
+  helper.CallPreProc(/*num_calls=*/300,
+                     /*clipped_ratio=*/kAboveClippedThreshold);
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+
+  helper.CallPreProc(/*num_calls=*/300,
+                     /*clipped_ratio=*/kAboveClippedThreshold);
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
+      .WillRepeatedly(Return(false));
+  helper.CallProcess(/*num_calls=*/4, speech_probability_override,
+                     GetOverrideOrEmpty(-34.0f));
+  helper.CallProcess(/*num_calls=*/15, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(14))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(15))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(16))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(17))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/20, kNoOverride, kNoOverride);
+  EXPECT_CALL(helper.mock_gain_control, set_compression_gain_db(18))
+      .WillOnce(Return(0));
+  helper.CallProcess(/*num_calls=*/1, kNoOverride, kNoOverride);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, UserCanRaiseVolumeAfterClipping) {
+  const auto speech_probability_override =
+      GetOverrideOrEmpty(kHighSpeechProbability);
+
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/225,
+                         speech_probability_override,
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(210, helper.manager.recommended_analog_level());
+
+  // High enough error to trigger a volume check.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(14), Return(true)));
+  // User changed the volume.
+  helper.manager.set_stream_analog_level(250);
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(AtLeast(1));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-32.0f));
+  EXPECT_EQ(250, helper.manager.recommended_analog_level());
+
+  // Move down...
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(-10), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-8.0f));
+  EXPECT_EQ(210, helper.manager.recommended_analog_level());
+  // And back up to the new max established by the user.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(40), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-58.0f));
+  EXPECT_EQ(250, helper.manager.recommended_analog_level());
+  // Will not move above new maximum.
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
+  helper.CallProcess(/*num_calls=*/1, speech_probability_override,
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(250, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, ClippingDoesNotPullLowVolumeBackUp) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/80,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, Reset()).Times(0);
+  int initial_volume = helper.manager.recommended_analog_level();
+  helper.CallPreProc(/*num_calls=*/1, /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(initial_volume, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, TakesNoActionOnZeroMicVolume) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(kInitialInputVolume,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_CALL(*helper.mock_agc, GetRmsErrorDb(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
+  helper.manager.set_stream_analog_level(0);
+  helper.CallProcess(/*num_calls=*/10,
+                     GetOverrideOrEmpty(kHighSpeechProbability),
+                     GetOverrideOrEmpty(-48.0f));
+  EXPECT_EQ(0, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, ClippingDetectionLowersVolume) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/255,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_EQ(255, helper.manager.recommended_analog_level());
+  helper.CallPreProcForChangingAudio(/*num_calls=*/100, /*peak_ratio=*/0.99f);
+  EXPECT_EQ(255, helper.manager.recommended_analog_level());
+  helper.CallPreProcForChangingAudio(/*num_calls=*/100, /*peak_ratio=*/1.0f);
+  EXPECT_EQ(240, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       DisabledClippingPredictorDoesNotLowerVolume) {
+  AgcManagerDirectTestHelper helper;
+  helper.CallAgcSequence(/*applied_input_volume=*/255,
+                         GetOverrideOrEmpty(kHighSpeechProbability),
+                         GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_FALSE(helper.manager.clipping_predictor_enabled());
+  EXPECT_EQ(255, helper.manager.recommended_analog_level());
+  helper.CallPreProcForChangingAudio(/*num_calls=*/100, /*peak_ratio=*/0.99f);
+  EXPECT_EQ(255, helper.manager.recommended_analog_level());
+  helper.CallPreProcForChangingAudio(/*num_calls=*/100, /*peak_ratio=*/0.99f);
+  EXPECT_EQ(255, helper.manager.recommended_analog_level());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, DisableDigitalDisablesDigital) {
+  if (IsRmsErrorOverridden()) {
+    GTEST_SKIP() << "Skipped. RMS error override does not affect the test.";
+  }
+
+  auto agc = std::unique_ptr<Agc>(new ::testing::NiceMock<MockAgc>());
+  MockGainControl mock_gain_control;
+  EXPECT_CALL(mock_gain_control, set_mode(GainControl::kFixedDigital));
+  EXPECT_CALL(mock_gain_control, set_target_level_dbfs(0));
+  EXPECT_CALL(mock_gain_control, set_compression_gain_db(0));
+  EXPECT_CALL(mock_gain_control, enable_limiter(false));
+
+  AnalogAgcConfig config;
+  config.enable_digital_adaptive = false;
+  auto manager = std::make_unique<AgcManagerDirect>(kNumChannels, config);
+  manager->Initialize();
+  manager->SetupDigitalGainControl(mock_gain_control);
+}
+
+TEST(AgcManagerDirectTest, AgcMinMicLevelExperimentDefault) {
+  std::unique_ptr<AgcManagerDirect> manager =
+      CreateAgcManagerDirect(kInitialInputVolume, kClippedLevelStep,
+                             kClippedRatioThreshold, kClippedWaitFrames);
+  EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
+}
+
+TEST(AgcManagerDirectTest, AgcMinMicLevelExperimentDisabled) {
+  for (const std::string& field_trial_suffix : {"", "_20220210"}) {
+    test::ScopedFieldTrials field_trial(
+        GetAgcMinMicLevelExperimentFieldTrial("Disabled" + field_trial_suffix));
+    std::unique_ptr<AgcManagerDirect> manager =
+        CreateAgcManagerDirect(kInitialInputVolume, kClippedLevelStep,
+                               kClippedRatioThreshold, kClippedWaitFrames);
+    EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
+  }
+}
+
+// Checks that a field-trial parameter outside of the valid range [0,255] is
+// ignored.
+TEST(AgcManagerDirectTest, AgcMinMicLevelExperimentOutOfRangeAbove) {
+  test::ScopedFieldTrials field_trial(
+      GetAgcMinMicLevelExperimentFieldTrial("Enabled-256"));
+  std::unique_ptr<AgcManagerDirect> manager =
+      CreateAgcManagerDirect(kInitialInputVolume, kClippedLevelStep,
+                             kClippedRatioThreshold, kClippedWaitFrames);
+  EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
+}
+
+// Checks that a field-trial parameter outside of the valid range [0,255] is
+// ignored.
+TEST(AgcManagerDirectTest, AgcMinMicLevelExperimentOutOfRangeBelow) {
+  test::ScopedFieldTrials field_trial(
+      GetAgcMinMicLevelExperimentFieldTrial("Enabled--1"));
+  std::unique_ptr<AgcManagerDirect> manager =
+      CreateAgcManagerDirect(kInitialInputVolume, kClippedLevelStep,
+                             kClippedRatioThreshold, kClippedWaitFrames);
+  EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
+}
+
+// Verifies that a valid experiment changes the minimum microphone level. The
+// start volume is larger than the min level and should therefore not be
+// changed.
+TEST(AgcManagerDirectTest, AgcMinMicLevelExperimentEnabled50) {
+  constexpr int kMinMicLevelOverride = 50;
+  for (const std::string& field_trial_suffix : {"", "_20220210"}) {
+    SCOPED_TRACE(field_trial_suffix);
+    test::ScopedFieldTrials field_trial(
+        GetAgcMinMicLevelExperimentFieldTrialEnabled(kMinMicLevelOverride,
+                                                     field_trial_suffix));
+    std::unique_ptr<AgcManagerDirect> manager =
+        CreateAgcManagerDirect(kInitialInputVolume, kClippedLevelStep,
+                               kClippedRatioThreshold, kClippedWaitFrames);
+    EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), kMinMicLevelOverride);
+  }
+}
+
+// Checks that, when the "WebRTC-Audio-AgcMinMicLevelExperiment" field trial is
+// specified with a valid value, the mic level never gets lowered beyond the
+// override value in the presence of clipping.
+TEST(AgcManagerDirectTest, AgcMinMicLevelExperimentCheckMinLevelWithClipping) {
+  constexpr int kMinMicLevelOverride = 250;
+
+  // Create and initialize two AGCs by specifying and leaving unspecified the
+  // relevant field trial.
+  const auto factory = []() {
+    std::unique_ptr<AgcManagerDirect> manager =
+        CreateAgcManagerDirect(kInitialInputVolume, kClippedLevelStep,
+                               kClippedRatioThreshold, kClippedWaitFrames);
+    manager->Initialize();
+    manager->set_stream_analog_level(kInitialInputVolume);
+    return manager;
+  };
+  std::unique_ptr<AgcManagerDirect> manager = factory();
+  std::unique_ptr<AgcManagerDirect> manager_with_override;
+  {
+    test::ScopedFieldTrials field_trial(
+        GetAgcMinMicLevelExperimentFieldTrialEnabled(kMinMicLevelOverride));
+    manager_with_override = factory();
+  }
+
+  // Create a test input signal which containts 80% of clipped samples.
+  AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz,
+                           1);
+  WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
+                          audio_buffer);
+
+  // Simulate 4 seconds of clipping; it is expected to trigger a downward
+  // adjustment of the analog gain.
+  CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
+                           /*speech_probability_override=*/absl::nullopt,
+                           /*speech_level_override=*/absl::nullopt, *manager);
+  CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
+                           /*speech_probability_override=*/absl::nullopt,
+                           /*speech_level_override=*/absl::nullopt,
+                           *manager_with_override);
+
+  // Make sure that an adaptation occurred.
+  ASSERT_GT(manager->recommended_analog_level(), 0);
+
+  // Check that the test signal triggers a larger downward adaptation for
+  // `manager`, which is allowed to reach a lower gain.
+  EXPECT_GT(manager_with_override->recommended_analog_level(),
+            manager->recommended_analog_level());
+  // Check that the gain selected by `manager_with_override` equals the minimum
+  // value overridden via field trial.
+  EXPECT_EQ(manager_with_override->recommended_analog_level(),
+            kMinMicLevelOverride);
+}
+
+// Checks that, when the "WebRTC-Audio-AgcMinMicLevelExperiment" field trial is
+// specified with a valid value, the mic level never gets lowered beyond the
+// override value in the presence of clipping when RMS error override is used.
+// TODO(webrtc:7494): Revisit the test after moving the number of override wait
+// frames to APM config. The test passes but internally the gain update timing
+// differs.
+TEST(AgcManagerDirectTest,
+     AgcMinMicLevelExperimentCheckMinLevelWithClippingWithRmsErrorOverride) {
+  constexpr int kMinMicLevelOverride = 250;
+
+  // Create and initialize two AGCs by specifying and leaving unspecified the
+  // relevant field trial.
+  const auto factory = []() {
+    std::unique_ptr<AgcManagerDirect> manager =
+        CreateAgcManagerDirect(kInitialInputVolume, kClippedLevelStep,
+                               kClippedRatioThreshold, kClippedWaitFrames);
+    manager->Initialize();
+    manager->set_stream_analog_level(kInitialInputVolume);
+    return manager;
+  };
+  std::unique_ptr<AgcManagerDirect> manager = factory();
+  std::unique_ptr<AgcManagerDirect> manager_with_override;
+  {
+    test::ScopedFieldTrials field_trial(
+        GetAgcMinMicLevelExperimentFieldTrialEnabled(kMinMicLevelOverride));
+    manager_with_override = factory();
+  }
+
+  // Create a test input signal which containts 80% of clipped samples.
+  AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz,
+                           1);
+  WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
+                          audio_buffer);
+
+  // Simulate 4 seconds of clipping; it is expected to trigger a downward
+  // adjustment of the analog gain.
+  CallPreProcessAndProcess(
+      /*num_calls=*/400, audio_buffer,
+      /*speech_probability_override=*/0.7f,
+      /*speech_probability_level=*/-18.0f, *manager);
+  CallPreProcessAndProcess(
+      /*num_calls=*/400, audio_buffer,
+      /*speech_probability_override=*/absl::optional<float>(0.7f),
+      /*speech_probability_level=*/absl::optional<float>(-18.0f),
+      *manager_with_override);
+
+  // Make sure that an adaptation occurred.
+  ASSERT_GT(manager->recommended_analog_level(), 0);
+
+  // Check that the test signal triggers a larger downward adaptation for
+  // `manager`, which is allowed to reach a lower gain.
+  EXPECT_GT(manager_with_override->recommended_analog_level(),
+            manager->recommended_analog_level());
+  // Check that the gain selected by `manager_with_override` equals the minimum
+  // value overridden via field trial.
+  EXPECT_EQ(manager_with_override->recommended_analog_level(),
+            kMinMicLevelOverride);
+}
+
+// Checks that, when the "WebRTC-Audio-AgcMinMicLevelExperiment" field trial is
+// specified with a value lower than the `clipped_level_min`, the behavior of
+// the analog gain controller is the same as that obtained when the field trial
+// is not specified.
+TEST(AgcManagerDirectTest,
+     AgcMinMicLevelExperimentCompareMicLevelWithClipping) {
+  // Create and initialize two AGCs by specifying and leaving unspecified the
+  // relevant field trial.
+  const auto factory = []() {
+    // Use a large clipped level step to more quickly decrease the analog gain
+    // with clipping.
+    AnalogAgcConfig config = kDefaultAnalogConfig;
+    config.startup_min_volume = kInitialInputVolume;
+    config.enable_digital_adaptive = false;
+    config.clipped_level_step = 64;
+    config.clipped_ratio_threshold = kClippedRatioThreshold;
+    config.clipped_wait_frames = kClippedWaitFrames;
+    auto controller =
+        std::make_unique<AgcManagerDirect>(/*num_capture_channels=*/1, config);
+    controller->Initialize();
+    controller->set_stream_analog_level(kInitialInputVolume);
+    return controller;
+  };
+  std::unique_ptr<AgcManagerDirect> manager = factory();
+  std::unique_ptr<AgcManagerDirect> manager_with_override;
+  {
+    constexpr int kMinMicLevelOverride = 20;
+    static_assert(
+        kDefaultAnalogConfig.clipped_level_min >= kMinMicLevelOverride,
+        "Use a lower override value.");
+    test::ScopedFieldTrials field_trial(
+        GetAgcMinMicLevelExperimentFieldTrialEnabled(kMinMicLevelOverride));
+    manager_with_override = factory();
+  }
+
+  // Create a test input signal which containts 80% of clipped samples.
+  AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz,
+                           1);
+  WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
+                          audio_buffer);
+
+  // Simulate 4 seconds of clipping; it is expected to trigger a downward
+  // adjustment of the analog gain.
+  CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
+                           /*speech_probability_override=*/absl::nullopt,
+                           /*speech_level_override=*/absl::nullopt, *manager);
+  CallPreProcessAndProcess(/*num_calls=*/400, audio_buffer,
+                           /*speech_probability_override=*/absl::nullopt,
+                           /*speech_level_override=*/absl::nullopt,
+                           *manager_with_override);
+
+  // Make sure that an adaptation occurred.
+  ASSERT_GT(manager->recommended_analog_level(), 0);
+
+  // Check that the selected analog gain is the same for both controllers and
+  // that it equals the minimum level reached when clipping is handled. That is
+  // expected because the minimum microphone level override is less than the
+  // minimum level used when clipping is detected.
+  EXPECT_EQ(manager->recommended_analog_level(),
+            manager_with_override->recommended_analog_level());
+  EXPECT_EQ(manager_with_override->recommended_analog_level(),
+            kDefaultAnalogConfig.clipped_level_min);
+}
+
+// Checks that, when the "WebRTC-Audio-AgcMinMicLevelExperiment" field trial is
+// specified with a value lower than the `clipped_level_min`, the behavior of
+// the analog gain controller is the same as that obtained when the field trial
+// is not specified.
+// TODO(webrtc:7494): Revisit the test after moving the number of override wait
+// frames to APM config. The test passes but internally the gain update timing
+// differs.
+TEST(AgcManagerDirectTest,
+     AgcMinMicLevelExperimentCompareMicLevelWithClippingWithRmsErrorOverride) {
+  // Create and initialize two AGCs by specifying and leaving unspecified the
+  // relevant field trial.
+  const auto factory = []() {
+    // Use a large clipped level step to more quickly decrease the analog gain
+    // with clipping.
+    AnalogAgcConfig config = kDefaultAnalogConfig;
+    config.startup_min_volume = kInitialInputVolume;
+    config.enable_digital_adaptive = false;
+    config.clipped_level_step = 64;
+    config.clipped_ratio_threshold = kClippedRatioThreshold;
+    config.clipped_wait_frames = kClippedWaitFrames;
+    auto controller =
+        std::make_unique<AgcManagerDirect>(/*num_capture_channels=*/1, config);
+    controller->Initialize();
+    controller->set_stream_analog_level(kInitialInputVolume);
+    return controller;
+  };
+  std::unique_ptr<AgcManagerDirect> manager = factory();
+  std::unique_ptr<AgcManagerDirect> manager_with_override;
+  {
+    constexpr int kMinMicLevelOverride = 20;
+    static_assert(
+        kDefaultAnalogConfig.clipped_level_min >= kMinMicLevelOverride,
+        "Use a lower override value.");
+    test::ScopedFieldTrials field_trial(
+        GetAgcMinMicLevelExperimentFieldTrialEnabled(kMinMicLevelOverride));
+    manager_with_override = factory();
+  }
+
+  // Create a test input signal which containts 80% of clipped samples.
+  AudioBuffer audio_buffer(kSampleRateHz, 1, kSampleRateHz, 1, kSampleRateHz,
+                           1);
+  WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
+                          audio_buffer);
+
+  CallPreProcessAndProcess(
+      /*num_calls=*/400, audio_buffer,
+      /*speech_probability_override=*/absl::optional<float>(0.7f),
+      /*speech_level_override=*/absl::optional<float>(-18.0f), *manager);
+  CallPreProcessAndProcess(
+      /*num_calls=*/400, audio_buffer,
+      /*speech_probability_override=*/absl::optional<float>(0.7f),
+      /*speech_level_override=*/absl::optional<float>(-18.0f),
+      *manager_with_override);
+
+  // Make sure that an adaptation occurred.
+  ASSERT_GT(manager->recommended_analog_level(), 0);
+
+  // Check that the selected analog gain is the same for both controllers and
+  // that it equals the minimum level reached when clipping is handled. That is
+  // expected because the minimum microphone level override is less than the
+  // minimum level used when clipping is detected.
+  EXPECT_EQ(manager->recommended_analog_level(),
+            manager_with_override->recommended_analog_level());
+  EXPECT_EQ(manager_with_override->recommended_analog_level(),
+            kDefaultAnalogConfig.clipped_level_min);
+}
+
+// TODO(bugs.webrtc.org/12774): Test the bahavior of `clipped_level_step`.
+// TODO(bugs.webrtc.org/12774): Test the bahavior of `clipped_ratio_threshold`.
+// TODO(bugs.webrtc.org/12774): Test the bahavior of `clipped_wait_frames`.
+// Verifies that configurable clipping parameters are initialized as intended.
+TEST_P(AgcManagerDirectParametrizedTest, ClippingParametersVerified) {
+  if (IsRmsErrorOverridden()) {
+    GTEST_SKIP() << "Skipped. RMS error override does not affect the test.";
+  }
+
+  std::unique_ptr<AgcManagerDirect> manager =
+      CreateAgcManagerDirect(kInitialInputVolume, kClippedLevelStep,
+                             kClippedRatioThreshold, kClippedWaitFrames);
+  manager->Initialize();
+  EXPECT_EQ(manager->clipped_level_step_, kClippedLevelStep);
+  EXPECT_EQ(manager->clipped_ratio_threshold_, kClippedRatioThreshold);
+  EXPECT_EQ(manager->clipped_wait_frames_, kClippedWaitFrames);
+  std::unique_ptr<AgcManagerDirect> manager_custom =
+      CreateAgcManagerDirect(kInitialInputVolume,
+                             /*clipped_level_step=*/10,
+                             /*clipped_ratio_threshold=*/0.2f,
+                             /*clipped_wait_frames=*/50);
+  manager_custom->Initialize();
+  EXPECT_EQ(manager_custom->clipped_level_step_, 10);
+  EXPECT_EQ(manager_custom->clipped_ratio_threshold_, 0.2f);
+  EXPECT_EQ(manager_custom->clipped_wait_frames_, 50);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       DisableClippingPredictorDisablesClippingPredictor) {
+  if (IsRmsErrorOverridden()) {
+    GTEST_SKIP() << "Skipped. RMS error override does not affect the test.";
+  }
+
+  // TODO(bugs.webrtc.org/12874): Use designated initializers once fixed.
+  ClippingPredictorConfig config;
+  config.enabled = false;
+
+  std::unique_ptr<AgcManagerDirect> manager = CreateAgcManagerDirect(
+      kInitialInputVolume, kClippedLevelStep, kClippedRatioThreshold,
+      kClippedWaitFrames, config);
+  manager->Initialize();
+  EXPECT_FALSE(manager->clipping_predictor_enabled());
+  EXPECT_FALSE(manager->use_clipping_predictor_step());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest, ClippingPredictorDisabledByDefault) {
+  if (IsRmsErrorOverridden()) {
+    GTEST_SKIP() << "Skipped. RMS error override does not affect the test.";
+  }
+
+  constexpr ClippingPredictorConfig kDefaultConfig;
+  EXPECT_FALSE(kDefaultConfig.enabled);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       EnableClippingPredictorEnablesClippingPredictor) {
+  if (IsRmsErrorOverridden()) {
+    GTEST_SKIP() << "Skipped. RMS error override does not affect the test.";
+  }
+
+  // TODO(bugs.webrtc.org/12874): Use designated initializers once fixed.
+  ClippingPredictorConfig config;
+  config.enabled = true;
+  config.use_predicted_step = true;
+
+  std::unique_ptr<AgcManagerDirect> manager = CreateAgcManagerDirect(
+      kInitialInputVolume, kClippedLevelStep, kClippedRatioThreshold,
+      kClippedWaitFrames, config);
+  manager->Initialize();
+  EXPECT_TRUE(manager->clipping_predictor_enabled());
+  EXPECT_TRUE(manager->use_clipping_predictor_step());
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       DisableClippingPredictorDoesNotLowerVolume) {
+  AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
+                           kNumChannels, kSampleRateHz, kNumChannels);
+
+  AnalogAgcConfig config = GetAnalogAgcTestConfig();
+  config.clipping_predictor.enabled = false;
+  AgcManagerDirect manager(config, new ::testing::NiceMock<MockAgc>());
+  manager.Initialize();
+  manager.set_stream_analog_level(/*level=*/255);
+  EXPECT_FALSE(manager.clipping_predictor_enabled());
+  EXPECT_FALSE(manager.use_clipping_predictor_step());
+  EXPECT_EQ(manager.recommended_analog_level(), 255);
+  manager.Process(audio_buffer, GetOverrideOrEmpty(kHighSpeechProbability),
+                  GetOverrideOrEmpty(kSpeechLevelDbfs));
+  CallPreProcessAudioBuffer(/*num_calls=*/10, /*peak_ratio=*/0.99f, manager);
+  EXPECT_EQ(manager.recommended_analog_level(), 255);
+  CallPreProcessAudioBuffer(/*num_calls=*/300, /*peak_ratio=*/0.99f, manager);
+  EXPECT_EQ(manager.recommended_analog_level(), 255);
+  CallPreProcessAudioBuffer(/*num_calls=*/10, /*peak_ratio=*/0.99f, manager);
+  EXPECT_EQ(manager.recommended_analog_level(), 255);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       UsedClippingPredictionsProduceLowerAnalogLevels) {
+  AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
+                           kNumChannels, kSampleRateHz, kNumChannels);
+
+  AnalogAgcConfig config_with_prediction = GetAnalogAgcTestConfig();
+  config_with_prediction.clipping_predictor.enabled = true;
+  config_with_prediction.clipping_predictor.use_predicted_step = true;
+  AnalogAgcConfig config_without_prediction = GetAnalogAgcTestConfig();
+  config_without_prediction.clipping_predictor.enabled = false;
+  AgcManagerDirect manager_with_prediction(config_with_prediction,
+                                           new ::testing::NiceMock<MockAgc>());
+  AgcManagerDirect manager_without_prediction(
+      config_without_prediction, new ::testing::NiceMock<MockAgc>());
+
+  manager_with_prediction.Initialize();
+  manager_without_prediction.Initialize();
+
+  constexpr int kInitialLevel = 255;
+  constexpr float kClippingPeakRatio = 1.0f;
+  constexpr float kCloseToClippingPeakRatio = 0.99f;
+  constexpr float kZeroPeakRatio = 0.0f;
+  manager_with_prediction.set_stream_analog_level(kInitialLevel);
+  manager_without_prediction.set_stream_analog_level(kInitialLevel);
+  manager_with_prediction.Process(audio_buffer,
+                                  GetOverrideOrEmpty(kHighSpeechProbability),
+                                  GetOverrideOrEmpty(kSpeechLevelDbfs));
+  manager_without_prediction.Process(audio_buffer,
+                                     GetOverrideOrEmpty(kHighSpeechProbability),
+                                     GetOverrideOrEmpty(kSpeechLevelDbfs));
+  EXPECT_TRUE(manager_with_prediction.clipping_predictor_enabled());
+  EXPECT_FALSE(manager_without_prediction.clipping_predictor_enabled());
+  EXPECT_TRUE(manager_with_prediction.use_clipping_predictor_step());
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(), kInitialLevel);
+  EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
+            kInitialLevel);
+
+  // Expect a change in the analog level when the prediction step is used.
+  CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            kInitialLevel - kClippedLevelStep);
+  EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
+            kInitialLevel);
+
+  // Expect no change during waiting.
+  CallPreProcessAudioBuffer(kClippedWaitFrames, kCloseToClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(kClippedWaitFrames, kCloseToClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            kInitialLevel - kClippedLevelStep);
+  EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
+            kInitialLevel);
+
+  // Expect a change when the prediction step is used.
+  CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            kInitialLevel - 2 * kClippedLevelStep);
+  EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
+            kInitialLevel);
+
+  // Expect no change when clipping is not detected or predicted.
+  CallPreProcessAudioBuffer(2 * kClippedWaitFrames, kZeroPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(2 * kClippedWaitFrames, kZeroPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            kInitialLevel - 2 * kClippedLevelStep);
+  EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
+            kInitialLevel);
+
+  // Expect a change for clipping frames.
+  CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            kInitialLevel - 3 * kClippedLevelStep);
+  EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
+            kInitialLevel - kClippedLevelStep);
+
+  // Expect no change during waiting.
+  CallPreProcessAudioBuffer(kClippedWaitFrames, kClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(kClippedWaitFrames, kClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            kInitialLevel - 3 * kClippedLevelStep);
+  EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
+            kInitialLevel - kClippedLevelStep);
+
+  // Expect a change for clipping frames.
+  CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            kInitialLevel - 4 * kClippedLevelStep);
+  EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
+            kInitialLevel - 2 * kClippedLevelStep);
+}
+
+TEST_P(AgcManagerDirectParametrizedTest,
+       UnusedClippingPredictionsProduceEqualAnalogLevels) {
+  AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
+                           kNumChannels, kSampleRateHz, kNumChannels);
+
+  AnalogAgcConfig config_with_prediction = GetAnalogAgcTestConfig();
+  config_with_prediction.clipping_predictor.enabled = true;
+  config_with_prediction.clipping_predictor.use_predicted_step = false;
+  AnalogAgcConfig config_without_prediction = GetAnalogAgcTestConfig();
+  config_without_prediction.clipping_predictor.enabled = false;
+  AgcManagerDirect manager_with_prediction(config_with_prediction,
+                                           new ::testing::NiceMock<MockAgc>());
+  AgcManagerDirect manager_without_prediction(
+      config_without_prediction, new ::testing::NiceMock<MockAgc>());
+
+  constexpr int kInitialLevel = 255;
+  constexpr float kClippingPeakRatio = 1.0f;
+  constexpr float kCloseToClippingPeakRatio = 0.99f;
+  constexpr float kZeroPeakRatio = 0.0f;
+  manager_with_prediction.Initialize();
+  manager_without_prediction.Initialize();
+  manager_with_prediction.set_stream_analog_level(kInitialLevel);
+  manager_without_prediction.set_stream_analog_level(kInitialLevel);
+  manager_with_prediction.Process(audio_buffer,
+                                  GetOverrideOrEmpty(kHighSpeechProbability),
+                                  GetOverrideOrEmpty(kSpeechLevelDbfs));
+  manager_without_prediction.Process(audio_buffer,
+                                     GetOverrideOrEmpty(kHighSpeechProbability),
+                                     GetOverrideOrEmpty(kSpeechLevelDbfs));
+
+  EXPECT_TRUE(manager_with_prediction.clipping_predictor_enabled());
+  EXPECT_FALSE(manager_without_prediction.clipping_predictor_enabled());
+  EXPECT_FALSE(manager_with_prediction.use_clipping_predictor_step());
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(), kInitialLevel);
+  EXPECT_EQ(manager_without_prediction.recommended_analog_level(),
+            kInitialLevel);
+
+  // Expect no change in the analog level for non-clipping frames.
+  CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            manager_without_prediction.recommended_analog_level());
+
+  // Expect no change for non-clipping frames.
+  CallPreProcessAudioBuffer(kClippedWaitFrames, kCloseToClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(kClippedWaitFrames, kCloseToClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            manager_without_prediction.recommended_analog_level());
+
+  // Expect no change for non-clipping frames.
+  CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(/*num_calls=*/10, kCloseToClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            manager_without_prediction.recommended_analog_level());
+
+  // Expect no change when clipping is not detected or predicted.
+  CallPreProcessAudioBuffer(2 * kClippedWaitFrames, kZeroPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(2 * kClippedWaitFrames, kZeroPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            manager_without_prediction.recommended_analog_level());
+
+  // Expect a change for clipping frames.
+  CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            manager_without_prediction.recommended_analog_level());
+
+  // Expect no change during waiting.
+  CallPreProcessAudioBuffer(kClippedWaitFrames, kClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(kClippedWaitFrames, kClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            manager_without_prediction.recommended_analog_level());
+
+  // Expect a change for clipping frames.
+  CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
+                            manager_with_prediction);
+  CallPreProcessAudioBuffer(/*num_calls=*/1, kClippingPeakRatio,
+                            manager_without_prediction);
+  EXPECT_EQ(manager_with_prediction.recommended_analog_level(),
+            manager_without_prediction.recommended_analog_level());
+}
+
+// Checks that passing an empty speech level and probability overrides to
+// `Process()` has the same effect as passing no overrides.
+TEST_P(AgcManagerDirectParametrizedTest, EmptyRmsErrorOverrideHasNoEffect) {
+  AgcManagerDirect manager_1(kNumChannels, GetAnalogAgcTestConfig());
+  AgcManagerDirect manager_2(kNumChannels, GetAnalogAgcTestConfig());
+  manager_1.Initialize();
+  manager_2.Initialize();
+
+  constexpr int kAnalogLevel = 50;
+  manager_1.set_stream_analog_level(kAnalogLevel);
+  manager_2.set_stream_analog_level(kAnalogLevel);
+
+  // Feed speech with low energy to trigger an upward adapation of the analog
+  // level.
+  constexpr int kNumFrames = 125;
+  constexpr int kGainDb = -20;
+  SpeechSamplesReader reader;
+
+  // Check the initial input volume.
+  ASSERT_EQ(manager_1.recommended_analog_level(), kAnalogLevel);
+  ASSERT_EQ(manager_2.recommended_analog_level(), kAnalogLevel);
+
+  reader.Feed(kNumFrames, kGainDb, absl::nullopt, absl::nullopt, manager_1);
+  reader.Feed(kNumFrames, kGainDb, manager_2);
+
+  // Check that the states are the same and adaptation occurs.
+  EXPECT_EQ(manager_1.recommended_analog_level(),
+            manager_2.recommended_analog_level());
+  ASSERT_GT(manager_1.recommended_analog_level(), kAnalogLevel);
+  EXPECT_EQ(manager_1.voice_probability(), manager_2.voice_probability());
+  EXPECT_EQ(manager_1.frames_since_clipped_, manager_2.frames_since_clipped_);
+
+  // Check that the states of the channel AGCs are the same.
+  EXPECT_EQ(manager_1.num_channels(), manager_2.num_channels());
+  for (int i = 0; i < manager_1.num_channels(); ++i) {
+    EXPECT_EQ(manager_1.channel_agcs_[i]->recommended_analog_level(),
+              manager_2.channel_agcs_[i]->recommended_analog_level());
+    EXPECT_EQ(manager_1.channel_agcs_[i]->voice_probability(),
+              manager_2.channel_agcs_[i]->voice_probability());
+  }
+}
+
+// Checks that passing a non-empty speech level and probability overrides to
+// `Process()` has an effect.
+TEST_P(AgcManagerDirectParametrizedTest, NonEmptyRmsErrorOverrideHasEffect) {
+  AgcManagerDirect manager_1(kNumChannels, GetAnalogAgcTestConfig());
+  AgcManagerDirect manager_2(kNumChannels, GetAnalogAgcTestConfig());
+  manager_1.Initialize();
+  manager_2.Initialize();
+
+  constexpr int kInputVolume = 128;
+  manager_1.set_stream_analog_level(kInputVolume);
+  manager_2.set_stream_analog_level(kInputVolume);
+
+  // Feed speech with low energy to trigger an upward adapation of the input
+  // volume.
+  constexpr int kNumFrames = 125;
+  constexpr int kGainDb = -20;
+  SpeechSamplesReader reader;
+
+  // Make sure that the feeding samples triggers an adaptation when no override
+  // is specified.
+  reader.Feed(kNumFrames, kGainDb, manager_1);
+  ASSERT_GT(manager_1.recommended_analog_level(), kInputVolume);
+
+  // Expect that feeding samples triggers an adaptation when the speech
+  // probability and speech level overrides are specified.
+  reader.Feed(kNumFrames, kGainDb,
+              /*speech_probability_override=*/kHighSpeechProbability,
+              /*speech_level_override=*/-45.0f, manager_2);
+  EXPECT_GT(manager_2.recommended_analog_level(), kInputVolume);
+
+  // The voice probability override does not affect the `voice_probability()`
+  // getter.
+  EXPECT_EQ(manager_1.voice_probability(), manager_2.voice_probability());
+}
+
+class AgcManagerDirectChannelSampleRateTest
+    : public ::testing::TestWithParam<std::tuple<int, int>> {
+ protected:
+  int GetNumChannels() const { return std::get<0>(GetParam()); }
+  int GetSampleRateHz() const { return std::get<1>(GetParam()); }
+};
+
+TEST_P(AgcManagerDirectChannelSampleRateTest, CheckIsAlive) {
+  const int num_channels = GetNumChannels();
+  const int sample_rate_hz = GetSampleRateHz();
+
+  constexpr AnalogAgcConfig kConfig{.enabled = true,
+                                    .clipping_predictor{.enabled = true}};
+  AgcManagerDirect manager(num_channels, kConfig);
+  manager.Initialize();
+  AudioBuffer buffer(sample_rate_hz, num_channels, sample_rate_hz, num_channels,
+                     sample_rate_hz, num_channels);
+
+  constexpr int kStartupVolume = 100;
+  int applied_initial_volume = kStartupVolume;
+
+  // Trigger a downward adaptation with clipping.
+  WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.5f,
+                          buffer);
+  const int initial_volume1 = applied_initial_volume;
+  for (int i = 0; i < 400; ++i) {
+    manager.set_stream_analog_level(applied_initial_volume);
+    manager.AnalyzePreProcess(buffer);
+    manager.Process(buffer, kLowSpeechProbability,
+                    /*speech_level_dbfs=*/-20.0f);
+    applied_initial_volume = manager.recommended_analog_level();
+  }
+  ASSERT_LT(manager.recommended_analog_level(), initial_volume1);
+
+  // Fill in audio that does not clip.
+  WriteAudioBufferSamples(/*samples_value=*/1234.5f, /*clipped_ratio=*/0.0f,
+                          buffer);
+
+  // Trigger an upward adaptation.
+  const int initial_volume2 = manager.recommended_analog_level();
+  for (int i = 0; i < kConfig.clipped_wait_frames; ++i) {
+    manager.set_stream_analog_level(applied_initial_volume);
+    manager.AnalyzePreProcess(buffer);
+    manager.Process(buffer, kHighSpeechProbability,
+                    /*speech_level_dbfs=*/-65.0f);
+    applied_initial_volume = manager.recommended_analog_level();
+  }
+  EXPECT_GT(manager.recommended_analog_level(), initial_volume2);
+
+  // Trigger a downward adaptation.
+  const int initial_volume = manager.recommended_analog_level();
+  for (int i = 0; i < 100; ++i) {
+    manager.set_stream_analog_level(applied_initial_volume);
+    manager.AnalyzePreProcess(buffer);
+    manager.Process(buffer, kHighSpeechProbability,
+                    /*speech_level_dbfs=*/-5.0f);
+    applied_initial_volume = manager.recommended_analog_level();
+  }
+  EXPECT_LT(manager.recommended_analog_level(), initial_volume);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    ,
+    AgcManagerDirectChannelSampleRateTest,
+    ::testing::Combine(::testing::Values(1, 2, 3, 6),
+                       ::testing::Values(8000, 16000, 32000, 48000)));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/gain_control.h b/third_party/libwebrtc/modules/audio_processing/agc/gain_control.h
new file mode 100644
index 0000000000..389b2114af
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/gain_control.h
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_GAIN_CONTROL_H_
+#define MODULES_AUDIO_PROCESSING_AGC_GAIN_CONTROL_H_
+
+namespace webrtc {
+
+// The automatic gain control (AGC) component brings the signal to an
+// appropriate range. This is done by applying a digital gain directly and, in
+// the analog mode, prescribing an analog gain to be applied at the audio HAL.
+//
+// Recommended to be enabled on the client-side.
+class GainControl {
+ public:
+  // When an analog mode is set, this must be called prior to `ProcessStream()`
+  // to pass the current analog level from the audio HAL. Must be within the
+  // range provided to `set_analog_level_limits()`.
+  virtual int set_stream_analog_level(int level) = 0;
+
+  // When an analog mode is set, this should be called after `ProcessStream()`
+  // to obtain the recommended new analog level for the audio HAL. It is the
+  // users responsibility to apply this level.
+  virtual int stream_analog_level() const = 0;
+
+  enum Mode {
+    // Adaptive mode intended for use if an analog volume control is available
+    // on the capture device. It will require the user to provide coupling
+    // between the OS mixer controls and AGC through the `stream_analog_level()`
+    // functions.
+    //
+    // It consists of an analog gain prescription for the audio device and a
+    // digital compression stage.
+    kAdaptiveAnalog,
+
+    // Adaptive mode intended for situations in which an analog volume control
+    // is unavailable. It operates in a similar fashion to the adaptive analog
+    // mode, but with scaling instead applied in the digital domain. As with
+    // the analog mode, it additionally uses a digital compression stage.
+    kAdaptiveDigital,
+
+    // Fixed mode which enables only the digital compression stage also used by
+    // the two adaptive modes.
+    //
+    // It is distinguished from the adaptive modes by considering only a
+    // short time-window of the input signal. It applies a fixed gain through
+    // most of the input level range, and compresses (gradually reduces gain
+    // with increasing level) the input signal at higher levels. This mode is
+    // preferred on embedded devices where the capture signal level is
+    // predictable, so that a known gain can be applied.
+    kFixedDigital
+  };
+
+  virtual int set_mode(Mode mode) = 0;
+  virtual Mode mode() const = 0;
+
+  // Sets the target peak `level` (or envelope) of the AGC in dBFs (decibels
+  // from digital full-scale). The convention is to use positive values. For
+  // instance, passing in a value of 3 corresponds to -3 dBFs, or a target
+  // level 3 dB below full-scale. Limited to [0, 31].
+  //
+  // TODO(ajm): use a negative value here instead, if/when VoE will similarly
+  //            update its interface.
+  virtual int set_target_level_dbfs(int level) = 0;
+  virtual int target_level_dbfs() const = 0;
+
+  // Sets the maximum `gain` the digital compression stage may apply, in dB. A
+  // higher number corresponds to greater compression, while a value of 0 will
+  // leave the signal uncompressed. Limited to [0, 90].
+  virtual int set_compression_gain_db(int gain) = 0;
+  virtual int compression_gain_db() const = 0;
+
+  // When enabled, the compression stage will hard limit the signal to the
+  // target level. Otherwise, the signal will be compressed but not limited
+  // above the target level.
+  virtual int enable_limiter(bool enable) = 0;
+  virtual bool is_limiter_enabled() const = 0;
+
+  // Sets the `minimum` and `maximum` analog levels of the audio capture device.
+  // Must be set if and only if an analog mode is used. Limited to [0, 65535].
+  virtual int set_analog_level_limits(int minimum, int maximum) = 0;
+  virtual int analog_level_minimum() const = 0;
+  virtual int analog_level_maximum() const = 0;
+
+  // Returns true if the AGC has detected a saturation event (period where the
+  // signal reaches digital full-scale) in the current frame and the analog
+  // level cannot be reduced.
+  //
+  // This could be used as an indicator to reduce or disable analog mic gain at
+  // the audio HAL.
+  virtual bool stream_is_saturated() const = 0;
+
+ protected:
+  virtual ~GainControl() {}
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_GAIN_CONTROL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/gain_control_interface_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc/gain_control_interface_gn/moz.build
new file mode 100644
index 0000000000..c6ab9b3160
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/gain_control_interface_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("gain_control_interface_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/legacy/analog_agc.cc b/third_party/libwebrtc/modules/audio_processing/agc/legacy/analog_agc.cc
new file mode 100644
index 0000000000..e40a3f1629
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/legacy/analog_agc.cc
@@ -0,0 +1,1238 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ *
+ * Using a feedback system, determines an appropriate analog volume level
+ * given an input signal and current volume level. Targets a conservative
+ * signal level and is intended for use with a digital AGC to apply
+ * additional gain.
+ *
+ */
+
+#include "modules/audio_processing/agc/legacy/analog_agc.h"
+
+#include <stdlib.h>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Errors
+#define AGC_UNSPECIFIED_ERROR 18000
+#define AGC_UNINITIALIZED_ERROR 18002
+#define AGC_NULL_POINTER_ERROR 18003
+#define AGC_BAD_PARAMETER_ERROR 18004
+
+/* The slope of in Q13*/
+static const int16_t kSlope1[8] = {21793, 12517, 7189, 4129,
+                                   2372,  1362,  472,  78};
+
+/* The offset in Q14 */
+static const int16_t kOffset1[8] = {25395, 23911, 22206, 20737,
+                                    19612, 18805, 17951, 17367};
+
+/* The slope of in Q13*/
+static const int16_t kSlope2[8] = {2063, 1731, 1452, 1218, 1021, 857, 597, 337};
+
+/* The offset in Q14 */
+static const int16_t kOffset2[8] = {18432, 18379, 18290, 18177,
+                                    18052, 17920, 17670, 17286};
+
+static const int16_t kMuteGuardTimeMs = 8000;
+static const int16_t kInitCheck = 42;
+static const size_t kNumSubframes = 10;
+
+/* Default settings if config is not used */
+#define AGC_DEFAULT_TARGET_LEVEL 3
+#define AGC_DEFAULT_COMP_GAIN 9
+/* This is the target level for the analog part in ENV scale. To convert to RMS
+ * scale you
+ * have to add OFFSET_ENV_TO_RMS.
+ */
+#define ANALOG_TARGET_LEVEL 11
+#define ANALOG_TARGET_LEVEL_2 5  // ANALOG_TARGET_LEVEL / 2
+/* Offset between RMS scale (analog part) and ENV scale (digital part). This
+ * value actually
+ * varies with the FIXED_ANALOG_TARGET_LEVEL, hence we should in the future
+ * replace it with
+ * a table.
+ */
+#define OFFSET_ENV_TO_RMS 9
+/* The reference input level at which the digital part gives an output of
+ * targetLevelDbfs
+ * (desired level) if we have no compression gain. This level should be set high
+ * enough not
+ * to compress the peaks due to the dynamics.
+ */
+#define DIGITAL_REF_AT_0_COMP_GAIN 4
+/* Speed of reference level decrease.
+ */
+#define DIFF_REF_TO_ANALOG 5
+
+/* Size of analog gain table */
+#define GAIN_TBL_LEN 32
+/* Matlab code:
+ * fprintf(1, '\t%i, %i, %i, %i,\n', round(10.^(linspace(0,10,32)/20) * 2^12));
+ */
+/* Q12 */
+static const uint16_t kGainTableAnalog[GAIN_TBL_LEN] = {
+    4096, 4251, 4412, 4579,  4752,  4932,  5118,  5312,  5513,  5722, 5938,
+    6163, 6396, 6638, 6889,  7150,  7420,  7701,  7992,  8295,  8609, 8934,
+    9273, 9623, 9987, 10365, 10758, 11165, 11587, 12025, 12480, 12953};
+
+/* Gain/Suppression tables for virtual Mic (in Q10) */
+static const uint16_t kGainTableVirtualMic[128] = {
+    1052,  1081,  1110,  1141,  1172,  1204,  1237,  1271,  1305,  1341,  1378,
+    1416,  1454,  1494,  1535,  1577,  1620,  1664,  1710,  1757,  1805,  1854,
+    1905,  1957,  2010,  2065,  2122,  2180,  2239,  2301,  2364,  2428,  2495,
+    2563,  2633,  2705,  2779,  2855,  2933,  3013,  3096,  3180,  3267,  3357,
+    3449,  3543,  3640,  3739,  3842,  3947,  4055,  4166,  4280,  4397,  4517,
+    4640,  4767,  4898,  5032,  5169,  5311,  5456,  5605,  5758,  5916,  6078,
+    6244,  6415,  6590,  6770,  6956,  7146,  7341,  7542,  7748,  7960,  8178,
+    8402,  8631,  8867,  9110,  9359,  9615,  9878,  10148, 10426, 10711, 11004,
+    11305, 11614, 11932, 12258, 12593, 12938, 13292, 13655, 14029, 14412, 14807,
+    15212, 15628, 16055, 16494, 16945, 17409, 17885, 18374, 18877, 19393, 19923,
+    20468, 21028, 21603, 22194, 22801, 23425, 24065, 24724, 25400, 26095, 26808,
+    27541, 28295, 29069, 29864, 30681, 31520, 32382};
+static const uint16_t kSuppressionTableVirtualMic[128] = {
+    1024, 1006, 988, 970, 952, 935, 918, 902, 886, 870, 854, 839, 824, 809, 794,
+    780,  766,  752, 739, 726, 713, 700, 687, 675, 663, 651, 639, 628, 616, 605,
+    594,  584,  573, 563, 553, 543, 533, 524, 514, 505, 496, 487, 478, 470, 461,
+    453,  445,  437, 429, 421, 414, 406, 399, 392, 385, 378, 371, 364, 358, 351,
+    345,  339,  333, 327, 321, 315, 309, 304, 298, 293, 288, 283, 278, 273, 268,
+    263,  258,  254, 249, 244, 240, 236, 232, 227, 223, 219, 215, 211, 208, 204,
+    200,  197,  193, 190, 186, 183, 180, 176, 173, 170, 167, 164, 161, 158, 155,
+    153,  150,  147, 145, 142, 139, 137, 134, 132, 130, 127, 125, 123, 121, 118,
+    116,  114,  112, 110, 108, 106, 104, 102};
+
+/* Table for target energy levels. Values in Q(-7)
+ * Matlab code
+ * targetLevelTable = fprintf('%d,\t%d,\t%d,\t%d,\n',
+ * round((32767*10.^(-(0:63)'/20)).^2*16/2^7) */
+
+static const int32_t kTargetLevelTable[64] = {
+    134209536, 106606424, 84680493, 67264106, 53429779, 42440782, 33711911,
+    26778323,  21270778,  16895980, 13420954, 10660642, 8468049,  6726411,
+    5342978,   4244078,   3371191,  2677832,  2127078,  1689598,  1342095,
+    1066064,   846805,    672641,   534298,   424408,   337119,   267783,
+    212708,    168960,    134210,   106606,   84680,    67264,    53430,
+    42441,     33712,     26778,    21271,    16896,    13421,    10661,
+    8468,      6726,      5343,     4244,     3371,     2678,     2127,
+    1690,      1342,      1066,     847,      673,      534,      424,
+    337,       268,       213,      169,      134,      107,      85,
+    67};
+
+}  // namespace
+
+int WebRtcAgc_AddMic(void* state,
+                     int16_t* const* in_mic,
+                     size_t num_bands,
+                     size_t samples) {
+  int32_t nrg, max_nrg, sample, tmp32;
+  int32_t* ptr;
+  uint16_t targetGainIdx, gain;
+  size_t i;
+  int16_t n, L, tmp16, tmp_speech[16];
+  LegacyAgc* stt;
+  stt = reinterpret_cast<LegacyAgc*>(state);
+
+  if (stt->fs == 8000) {
+    L = 8;
+    if (samples != 80) {
+      return -1;
+    }
+  } else {
+    L = 16;
+    if (samples != 160) {
+      return -1;
+    }
+  }
+
+  /* apply slowly varying digital gain */
+  if (stt->micVol > stt->maxAnalog) {
+    /* `maxLevel` is strictly >= `micVol`, so this condition should be
+     * satisfied here, ensuring there is no divide-by-zero. */
+    RTC_DCHECK_GT(stt->maxLevel, stt->maxAnalog);
+
+    /* Q1 */
+    tmp16 = (int16_t)(stt->micVol - stt->maxAnalog);
+    tmp32 = (GAIN_TBL_LEN - 1) * tmp16;
+    tmp16 = (int16_t)(stt->maxLevel - stt->maxAnalog);
+    targetGainIdx = tmp32 / tmp16;
+    RTC_DCHECK_LT(targetGainIdx, GAIN_TBL_LEN);
+
+    /* Increment through the table towards the target gain.
+     * If micVol drops below maxAnalog, we allow the gain
+     * to be dropped immediately. */
+    if (stt->gainTableIdx < targetGainIdx) {
+      stt->gainTableIdx++;
+    } else if (stt->gainTableIdx > targetGainIdx) {
+      stt->gainTableIdx--;
+    }
+
+    /* Q12 */
+    gain = kGainTableAnalog[stt->gainTableIdx];
+
+    for (i = 0; i < samples; i++) {
+      size_t j;
+      for (j = 0; j < num_bands; ++j) {
+        sample = (in_mic[j][i] * gain) >> 12;
+        if (sample > 32767) {
+          in_mic[j][i] = 32767;
+        } else if (sample < -32768) {
+          in_mic[j][i] = -32768;
+        } else {
+          in_mic[j][i] = (int16_t)sample;
+        }
+      }
+    }
+  } else {
+    stt->gainTableIdx = 0;
+  }
+
+  /* compute envelope */
+  if (stt->inQueue > 0) {
+    ptr = stt->env[1];
+  } else {
+    ptr = stt->env[0];
+  }
+
+  for (i = 0; i < kNumSubframes; i++) {
+    /* iterate over samples */
+    max_nrg = 0;
+    for (n = 0; n < L; n++) {
+      nrg = in_mic[0][i * L + n] * in_mic[0][i * L + n];
+      if (nrg > max_nrg) {
+        max_nrg = nrg;
+      }
+    }
+    ptr[i] = max_nrg;
+  }
+
+  /* compute energy */
+  if (stt->inQueue > 0) {
+    ptr = stt->Rxx16w32_array[1];
+  } else {
+    ptr = stt->Rxx16w32_array[0];
+  }
+
+  for (i = 0; i < kNumSubframes / 2; i++) {
+    if (stt->fs == 16000) {
+      WebRtcSpl_DownsampleBy2(&in_mic[0][i * 32], 32, tmp_speech,
+                              stt->filterState);
+    } else {
+      memcpy(tmp_speech, &in_mic[0][i * 16], 16 * sizeof(int16_t));
+    }
+    /* Compute energy in blocks of 16 samples */
+    ptr[i] = WebRtcSpl_DotProductWithScale(tmp_speech, tmp_speech, 16, 4);
+  }
+
+  /* update queue information */
+  if (stt->inQueue == 0) {
+    stt->inQueue = 1;
+  } else {
+    stt->inQueue = 2;
+  }
+
+  /* call VAD (use low band only) */
+  WebRtcAgc_ProcessVad(&stt->vadMic, in_mic[0], samples);
+
+  return 0;
+}
+
+int WebRtcAgc_AddFarend(void* state, const int16_t* in_far, size_t samples) {
+  LegacyAgc* stt = reinterpret_cast<LegacyAgc*>(state);
+
+  int err = WebRtcAgc_GetAddFarendError(state, samples);
+
+  if (err != 0)
+    return err;
+
+  return WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, in_far, samples);
+}
+
+int WebRtcAgc_GetAddFarendError(void* state, size_t samples) {
+  LegacyAgc* stt;
+  stt = reinterpret_cast<LegacyAgc*>(state);
+
+  if (stt == NULL)
+    return -1;
+
+  if (stt->fs == 8000) {
+    if (samples != 80)
+      return -1;
+  } else if (stt->fs == 16000 || stt->fs == 32000 || stt->fs == 48000) {
+    if (samples != 160)
+      return -1;
+  } else {
+    return -1;
+  }
+
+  return 0;
+}
+
+int WebRtcAgc_VirtualMic(void* agcInst,
+                         int16_t* const* in_near,
+                         size_t num_bands,
+                         size_t samples,
+                         int32_t micLevelIn,
+                         int32_t* micLevelOut) {
+  int32_t tmpFlt, micLevelTmp, gainIdx;
+  uint16_t gain;
+  size_t ii, j;
+  LegacyAgc* stt;
+
+  uint32_t nrg;
+  size_t sampleCntr;
+  uint32_t frameNrg = 0;
+  uint32_t frameNrgLimit = 5500;
+  int16_t numZeroCrossing = 0;
+  const int16_t kZeroCrossingLowLim = 15;
+  const int16_t kZeroCrossingHighLim = 20;
+
+  stt = reinterpret_cast<LegacyAgc*>(agcInst);
+
+  /*
+   *  Before applying gain decide if this is a low-level signal.
+   *  The idea is that digital AGC will not adapt to low-level
+   *  signals.
+   */
+  if (stt->fs != 8000) {
+    frameNrgLimit = frameNrgLimit << 1;
+  }
+
+  frameNrg = (uint32_t)(in_near[0][0] * in_near[0][0]);
+  for (sampleCntr = 1; sampleCntr < samples; sampleCntr++) {
+    // increment frame energy if it is less than the limit
+    // the correct value of the energy is not important
+    if (frameNrg < frameNrgLimit) {
+      nrg = (uint32_t)(in_near[0][sampleCntr] * in_near[0][sampleCntr]);
+      frameNrg += nrg;
+    }
+
+    // Count the zero crossings
+    numZeroCrossing +=
+        ((in_near[0][sampleCntr] ^ in_near[0][sampleCntr - 1]) < 0);
+  }
+
+  if ((frameNrg < 500) || (numZeroCrossing <= 5)) {
+    stt->lowLevelSignal = 1;
+  } else if (numZeroCrossing <= kZeroCrossingLowLim) {
+    stt->lowLevelSignal = 0;
+  } else if (frameNrg <= frameNrgLimit) {
+    stt->lowLevelSignal = 1;
+  } else if (numZeroCrossing >= kZeroCrossingHighLim) {
+    stt->lowLevelSignal = 1;
+  } else {
+    stt->lowLevelSignal = 0;
+  }
+
+  micLevelTmp = micLevelIn << stt->scale;
+  /* Set desired level */
+  gainIdx = stt->micVol;
+  if (stt->micVol > stt->maxAnalog) {
+    gainIdx = stt->maxAnalog;
+  }
+  if (micLevelTmp != stt->micRef) {
+    /* Something has happened with the physical level, restart. */
+    stt->micRef = micLevelTmp;
+    stt->micVol = 127;
+    *micLevelOut = 127;
+    stt->micGainIdx = 127;
+    gainIdx = 127;
+  }
+  /* Pre-process the signal to emulate the microphone level. */
+  /* Take one step at a time in the gain table. */
+  if (gainIdx > 127) {
+    gain = kGainTableVirtualMic[gainIdx - 128];
+  } else {
+    gain = kSuppressionTableVirtualMic[127 - gainIdx];
+  }
+  for (ii = 0; ii < samples; ii++) {
+    tmpFlt = (in_near[0][ii] * gain) >> 10;
+    if (tmpFlt > 32767) {
+      tmpFlt = 32767;
+      gainIdx--;
+      if (gainIdx >= 127) {
+        gain = kGainTableVirtualMic[gainIdx - 127];
+      } else {
+        gain = kSuppressionTableVirtualMic[127 - gainIdx];
+      }
+    }
+    if (tmpFlt < -32768) {
+      tmpFlt = -32768;
+      gainIdx--;
+      if (gainIdx >= 127) {
+        gain = kGainTableVirtualMic[gainIdx - 127];
+      } else {
+        gain = kSuppressionTableVirtualMic[127 - gainIdx];
+      }
+    }
+    in_near[0][ii] = (int16_t)tmpFlt;
+    for (j = 1; j < num_bands; ++j) {
+      tmpFlt = (in_near[j][ii] * gain) >> 10;
+      if (tmpFlt > 32767) {
+        tmpFlt = 32767;
+      }
+      if (tmpFlt < -32768) {
+        tmpFlt = -32768;
+      }
+      in_near[j][ii] = (int16_t)tmpFlt;
+    }
+  }
+  /* Set the level we (finally) used */
+  stt->micGainIdx = gainIdx;
+  //    *micLevelOut = stt->micGainIdx;
+  *micLevelOut = stt->micGainIdx >> stt->scale;
+  /* Add to Mic as if it was the output from a true microphone */
+  if (WebRtcAgc_AddMic(agcInst, in_near, num_bands, samples) != 0) {
+    return -1;
+  }
+  return 0;
+}
+
+void WebRtcAgc_UpdateAgcThresholds(LegacyAgc* stt) {
+  int16_t tmp16;
+
+  /* Set analog target level in envelope dBOv scale */
+  tmp16 = (DIFF_REF_TO_ANALOG * stt->compressionGaindB) + ANALOG_TARGET_LEVEL_2;
+  tmp16 = WebRtcSpl_DivW32W16ResW16((int32_t)tmp16, ANALOG_TARGET_LEVEL);
+  stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN + tmp16;
+  if (stt->analogTarget < DIGITAL_REF_AT_0_COMP_GAIN) {
+    stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN;
+  }
+  if (stt->agcMode == kAgcModeFixedDigital) {
+    /* Adjust for different parameter interpretation in FixedDigital mode */
+    stt->analogTarget = stt->compressionGaindB;
+  }
+  /* Since the offset between RMS and ENV is not constant, we should make this
+   * into a
+   * table, but for now, we'll stick with a constant, tuned for the chosen
+   * analog
+   * target level.
+   */
+  stt->targetIdx = ANALOG_TARGET_LEVEL + OFFSET_ENV_TO_RMS;
+  /* Analog adaptation limits */
+  /* analogTargetLevel = round((32767*10^(-targetIdx/20))^2*16/2^7) */
+  stt->analogTargetLevel =
+      kRxxBufferLen * kTargetLevelTable[stt->targetIdx]; /* ex. -20 dBov */
+  stt->startUpperLimit =
+      kRxxBufferLen * kTargetLevelTable[stt->targetIdx - 1]; /* -19 dBov */
+  stt->startLowerLimit =
+      kRxxBufferLen * kTargetLevelTable[stt->targetIdx + 1]; /* -21 dBov */
+  stt->upperPrimaryLimit =
+      kRxxBufferLen * kTargetLevelTable[stt->targetIdx - 2]; /* -18 dBov */
+  stt->lowerPrimaryLimit =
+      kRxxBufferLen * kTargetLevelTable[stt->targetIdx + 2]; /* -22 dBov */
+  stt->upperSecondaryLimit =
+      kRxxBufferLen * kTargetLevelTable[stt->targetIdx - 5]; /* -15 dBov */
+  stt->lowerSecondaryLimit =
+      kRxxBufferLen * kTargetLevelTable[stt->targetIdx + 5]; /* -25 dBov */
+  stt->upperLimit = stt->startUpperLimit;
+  stt->lowerLimit = stt->startLowerLimit;
+}
+
+void WebRtcAgc_SaturationCtrl(LegacyAgc* stt,
+                              uint8_t* saturated,
+                              int32_t* env) {
+  int16_t i, tmpW16;
+
+  /* Check if the signal is saturated */
+  for (i = 0; i < 10; i++) {
+    tmpW16 = (int16_t)(env[i] >> 20);
+    if (tmpW16 > 875) {
+      stt->envSum += tmpW16;
+    }
+  }
+
+  if (stt->envSum > 25000) {
+    *saturated = 1;
+    stt->envSum = 0;
+  }
+
+  /* stt->envSum *= 0.99; */
+  stt->envSum = (int16_t)((stt->envSum * 32440) >> 15);
+}
+
+void WebRtcAgc_ZeroCtrl(LegacyAgc* stt, int32_t* inMicLevel, int32_t* env) {
+  int16_t i;
+  int64_t tmp = 0;
+  int32_t midVal;
+
+  /* Is the input signal zero? */
+  for (i = 0; i < 10; i++) {
+    tmp += env[i];
+  }
+
+  /* Each block is allowed to have a few non-zero
+   * samples.
+   */
+  if (tmp < 500) {
+    stt->msZero += 10;
+  } else {
+    stt->msZero = 0;
+  }
+
+  if (stt->muteGuardMs > 0) {
+    stt->muteGuardMs -= 10;
+  }
+
+  if (stt->msZero > 500) {
+    stt->msZero = 0;
+
+    /* Increase microphone level only if it's less than 50% */
+    midVal = (stt->maxAnalog + stt->minLevel + 1) / 2;
+    if (*inMicLevel < midVal) {
+      /* *inMicLevel *= 1.1; */
+      *inMicLevel = (1126 * *inMicLevel) >> 10;
+      /* Reduces risk of a muted mic repeatedly triggering excessive levels due
+       * to zero signal detection. */
+      *inMicLevel = WEBRTC_SPL_MIN(*inMicLevel, stt->zeroCtrlMax);
+      stt->micVol = *inMicLevel;
+    }
+
+    stt->activeSpeech = 0;
+    stt->Rxx16_LPw32Max = 0;
+
+    /* The AGC has a tendency (due to problems with the VAD parameters), to
+     * vastly increase the volume after a muting event. This timer prevents
+     * upwards adaptation for a short period. */
+    stt->muteGuardMs = kMuteGuardTimeMs;
+  }
+}
+
+void WebRtcAgc_SpeakerInactiveCtrl(LegacyAgc* stt) {
+  /* Check if the near end speaker is inactive.
+   * If that is the case the VAD threshold is
+   * increased since the VAD speech model gets
+   * more sensitive to any sound after a long
+   * silence.
+   */
+
+  int32_t tmp32;
+  int16_t vadThresh;
+
+  if (stt->vadMic.stdLongTerm < 2500) {
+    stt->vadThreshold = 1500;
+  } else {
+    vadThresh = kNormalVadThreshold;
+    if (stt->vadMic.stdLongTerm < 4500) {
+      /* Scale between min and max threshold */
+      vadThresh += (4500 - stt->vadMic.stdLongTerm) / 2;
+    }
+
+    /* stt->vadThreshold = (31 * stt->vadThreshold + vadThresh) / 32; */
+    tmp32 = vadThresh + 31 * stt->vadThreshold;
+    stt->vadThreshold = (int16_t)(tmp32 >> 5);
+  }
+}
+
+void WebRtcAgc_ExpCurve(int16_t volume, int16_t* index) {
+  // volume in Q14
+  // index in [0-7]
+  /* 8 different curves */
+  if (volume > 5243) {
+    if (volume > 7864) {
+      if (volume > 12124) {
+        *index = 7;
+      } else {
+        *index = 6;
+      }
+    } else {
+      if (volume > 6554) {
+        *index = 5;
+      } else {
+        *index = 4;
+      }
+    }
+  } else {
+    if (volume > 2621) {
+      if (volume > 3932) {
+        *index = 3;
+      } else {
+        *index = 2;
+      }
+    } else {
+      if (volume > 1311) {
+        *index = 1;
+      } else {
+        *index = 0;
+      }
+    }
+  }
+}
+
+int32_t WebRtcAgc_ProcessAnalog(void* state,
+                                int32_t inMicLevel,
+                                int32_t* outMicLevel,
+                                int16_t vadLogRatio,
+                                int16_t echo,
+                                uint8_t* saturationWarning) {
+  uint32_t tmpU32;
+  int32_t Rxx16w32, tmp32;
+  int32_t inMicLevelTmp, lastMicVol;
+  int16_t i;
+  uint8_t saturated = 0;
+  LegacyAgc* stt;
+
+  stt = reinterpret_cast<LegacyAgc*>(state);
+  inMicLevelTmp = inMicLevel << stt->scale;
+
+  if (inMicLevelTmp > stt->maxAnalog) {
+    return -1;
+  } else if (inMicLevelTmp < stt->minLevel) {
+    return -1;
+  }
+
+  if (stt->firstCall == 0) {
+    int32_t tmpVol;
+    stt->firstCall = 1;
+    tmp32 = ((stt->maxLevel - stt->minLevel) * 51) >> 9;
+    tmpVol = (stt->minLevel + tmp32);
+
+    /* If the mic level is very low at start, increase it! */
+    if ((inMicLevelTmp < tmpVol) && (stt->agcMode == kAgcModeAdaptiveAnalog)) {
+      inMicLevelTmp = tmpVol;
+    }
+    stt->micVol = inMicLevelTmp;
+  }
+
+  /* Set the mic level to the previous output value if there is digital input
+   * gain */
+  if ((inMicLevelTmp == stt->maxAnalog) && (stt->micVol > stt->maxAnalog)) {
+    inMicLevelTmp = stt->micVol;
+  }
+
+  /* If the mic level was manually changed to a very low value raise it! */
+  if ((inMicLevelTmp != stt->micVol) && (inMicLevelTmp < stt->minOutput)) {
+    tmp32 = ((stt->maxLevel - stt->minLevel) * 51) >> 9;
+    inMicLevelTmp = (stt->minLevel + tmp32);
+    stt->micVol = inMicLevelTmp;
+  }
+
+  if (inMicLevelTmp != stt->micVol) {
+    if (inMicLevel == stt->lastInMicLevel) {
+      // We requested a volume adjustment, but it didn't occur. This is
+      // probably due to a coarse quantization of the volume slider.
+      // Restore the requested value to prevent getting stuck.
+      inMicLevelTmp = stt->micVol;
+    } else {
+      // As long as the value changed, update to match.
+      stt->micVol = inMicLevelTmp;
+    }
+  }
+
+  if (inMicLevelTmp > stt->maxLevel) {
+    // Always allow the user to raise the volume above the maxLevel.
+    stt->maxLevel = inMicLevelTmp;
+  }
+
+  // Store last value here, after we've taken care of manual updates etc.
+  stt->lastInMicLevel = inMicLevel;
+  lastMicVol = stt->micVol;
+
+  /* Checks if the signal is saturated. Also a check if individual samples
+   * are larger than 12000 is done. If they are the counter for increasing
+   * the volume level is set to -100ms
+   */
+  WebRtcAgc_SaturationCtrl(stt, &saturated, stt->env[0]);
+
+  /* The AGC is always allowed to lower the level if the signal is saturated */
+  if (saturated == 1) {
+    /* Lower the recording level
+     * Rxx160_LP is adjusted down because it is so slow it could
+     * cause the AGC to make wrong decisions. */
+    /* stt->Rxx160_LPw32 *= 0.875; */
+    stt->Rxx160_LPw32 = (stt->Rxx160_LPw32 / 8) * 7;
+
+    stt->zeroCtrlMax = stt->micVol;
+
+    /* stt->micVol *= 0.903; */
+    tmp32 = inMicLevelTmp - stt->minLevel;
+    tmpU32 = WEBRTC_SPL_UMUL(29591, (uint32_t)(tmp32));
+    stt->micVol = (tmpU32 >> 15) + stt->minLevel;
+    if (stt->micVol > lastMicVol - 2) {
+      stt->micVol = lastMicVol - 2;
+    }
+    inMicLevelTmp = stt->micVol;
+
+    if (stt->micVol < stt->minOutput) {
+      *saturationWarning = 1;
+    }
+
+    /* Reset counter for decrease of volume level to avoid
+     * decreasing too much. The saturation control can still
+     * lower the level if needed. */
+    stt->msTooHigh = -100;
+
+    /* Enable the control mechanism to ensure that our measure,
+     * Rxx160_LP, is in the correct range. This must be done since
+     * the measure is very slow. */
+    stt->activeSpeech = 0;
+    stt->Rxx16_LPw32Max = 0;
+
+    /* Reset to initial values */
+    stt->msecSpeechInnerChange = kMsecSpeechInner;
+    stt->msecSpeechOuterChange = kMsecSpeechOuter;
+    stt->changeToSlowMode = 0;
+
+    stt->muteGuardMs = 0;
+
+    stt->upperLimit = stt->startUpperLimit;
+    stt->lowerLimit = stt->startLowerLimit;
+  }
+
+  /* Check if the input speech is zero. If so the mic volume
+   * is increased. On some computers the input is zero up as high
+   * level as 17% */
+  WebRtcAgc_ZeroCtrl(stt, &inMicLevelTmp, stt->env[0]);
+
+  /* Check if the near end speaker is inactive.
+   * If that is the case the VAD threshold is
+   * increased since the VAD speech model gets
+   * more sensitive to any sound after a long
+   * silence.
+   */
+  WebRtcAgc_SpeakerInactiveCtrl(stt);
+
+  for (i = 0; i < 5; i++) {
+    /* Computed on blocks of 16 samples */
+
+    Rxx16w32 = stt->Rxx16w32_array[0][i];
+
+    /* Rxx160w32 in Q(-7) */
+    tmp32 = (Rxx16w32 - stt->Rxx16_vectorw32[stt->Rxx16pos]) >> 3;
+    stt->Rxx160w32 = stt->Rxx160w32 + tmp32;
+    stt->Rxx16_vectorw32[stt->Rxx16pos] = Rxx16w32;
+
+    /* Circular buffer */
+    stt->Rxx16pos++;
+    if (stt->Rxx16pos == kRxxBufferLen) {
+      stt->Rxx16pos = 0;
+    }
+
+    /* Rxx16_LPw32 in Q(-4) */
+    tmp32 = (Rxx16w32 - stt->Rxx16_LPw32) >> kAlphaShortTerm;
+    stt->Rxx16_LPw32 = (stt->Rxx16_LPw32) + tmp32;
+
+    if (vadLogRatio > stt->vadThreshold) {
+      /* Speech detected! */
+
+      /* Check if Rxx160_LP is in the correct range. If
+       * it is too high/low then we set it to the maximum of
+       * Rxx16_LPw32 during the first 200ms of speech.
+       */
+      if (stt->activeSpeech < 250) {
+        stt->activeSpeech += 2;
+
+        if (stt->Rxx16_LPw32 > stt->Rxx16_LPw32Max) {
+          stt->Rxx16_LPw32Max = stt->Rxx16_LPw32;
+        }
+      } else if (stt->activeSpeech == 250) {
+        stt->activeSpeech += 2;
+        tmp32 = stt->Rxx16_LPw32Max >> 3;
+        stt->Rxx160_LPw32 = tmp32 * kRxxBufferLen;
+      }
+
+      tmp32 = (stt->Rxx160w32 - stt->Rxx160_LPw32) >> kAlphaLongTerm;
+      stt->Rxx160_LPw32 = stt->Rxx160_LPw32 + tmp32;
+
+      if (stt->Rxx160_LPw32 > stt->upperSecondaryLimit) {
+        stt->msTooHigh += 2;
+        stt->msTooLow = 0;
+        stt->changeToSlowMode = 0;
+
+        if (stt->msTooHigh > stt->msecSpeechOuterChange) {
+          stt->msTooHigh = 0;
+
+          /* Lower the recording level */
+          /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
+          tmp32 = stt->Rxx160_LPw32 >> 6;
+          stt->Rxx160_LPw32 = tmp32 * 53;
+
+          /* Reduce the max gain to avoid excessive oscillation
+           * (but never drop below the maximum analog level).
+           */
+          stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
+          stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
+
+          stt->zeroCtrlMax = stt->micVol;
+
+          /* 0.95 in Q15 */
+          tmp32 = inMicLevelTmp - stt->minLevel;
+          tmpU32 = WEBRTC_SPL_UMUL(31130, (uint32_t)(tmp32));
+          stt->micVol = (tmpU32 >> 15) + stt->minLevel;
+          if (stt->micVol > lastMicVol - 1) {
+            stt->micVol = lastMicVol - 1;
+          }
+          inMicLevelTmp = stt->micVol;
+
+          /* Enable the control mechanism to ensure that our measure,
+           * Rxx160_LP, is in the correct range.
+           */
+          stt->activeSpeech = 0;
+          stt->Rxx16_LPw32Max = 0;
+        }
+      } else if (stt->Rxx160_LPw32 > stt->upperLimit) {
+        stt->msTooHigh += 2;
+        stt->msTooLow = 0;
+        stt->changeToSlowMode = 0;
+
+        if (stt->msTooHigh > stt->msecSpeechInnerChange) {
+          /* Lower the recording level */
+          stt->msTooHigh = 0;
+          /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
+          stt->Rxx160_LPw32 = (stt->Rxx160_LPw32 / 64) * 53;
+
+          /* Reduce the max gain to avoid excessive oscillation
+           * (but never drop below the maximum analog level).
+           */
+          stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
+          stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
+
+          stt->zeroCtrlMax = stt->micVol;
+
+          /* 0.965 in Q15 */
+          tmp32 = inMicLevelTmp - stt->minLevel;
+          tmpU32 =
+              WEBRTC_SPL_UMUL(31621, (uint32_t)(inMicLevelTmp - stt->minLevel));
+          stt->micVol = (tmpU32 >> 15) + stt->minLevel;
+          if (stt->micVol > lastMicVol - 1) {
+            stt->micVol = lastMicVol - 1;
+          }
+          inMicLevelTmp = stt->micVol;
+        }
+      } else if (stt->Rxx160_LPw32 < stt->lowerSecondaryLimit) {
+        stt->msTooHigh = 0;
+        stt->changeToSlowMode = 0;
+        stt->msTooLow += 2;
+
+        if (stt->msTooLow > stt->msecSpeechOuterChange) {
+          /* Raise the recording level */
+          int16_t index, weightFIX;
+          int16_t volNormFIX = 16384;  // =1 in Q14.
+
+          stt->msTooLow = 0;
+
+          /* Normalize the volume level */
+          tmp32 = (inMicLevelTmp - stt->minLevel) << 14;
+          if (stt->maxInit != stt->minLevel) {
+            volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
+          }
+
+          /* Find correct curve */
+          WebRtcAgc_ExpCurve(volNormFIX, &index);
+
+          /* Compute weighting factor for the volume increase, 32^(-2*X)/2+1.05
+           */
+          weightFIX =
+              kOffset1[index] - (int16_t)((kSlope1[index] * volNormFIX) >> 13);
+
+          /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
+          stt->Rxx160_LPw32 = (stt->Rxx160_LPw32 / 64) * 67;
+
+          tmp32 = inMicLevelTmp - stt->minLevel;
+          tmpU32 =
+              ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel));
+          stt->micVol = (tmpU32 >> 14) + stt->minLevel;
+          if (stt->micVol < lastMicVol + 2) {
+            stt->micVol = lastMicVol + 2;
+          }
+
+          inMicLevelTmp = stt->micVol;
+        }
+      } else if (stt->Rxx160_LPw32 < stt->lowerLimit) {
+        stt->msTooHigh = 0;
+        stt->changeToSlowMode = 0;
+        stt->msTooLow += 2;
+
+        if (stt->msTooLow > stt->msecSpeechInnerChange) {
+          /* Raise the recording level */
+          int16_t index, weightFIX;
+          int16_t volNormFIX = 16384;  // =1 in Q14.
+
+          stt->msTooLow = 0;
+
+          /* Normalize the volume level */
+          tmp32 = (inMicLevelTmp - stt->minLevel) << 14;
+          if (stt->maxInit != stt->minLevel) {
+            volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
+          }
+
+          /* Find correct curve */
+          WebRtcAgc_ExpCurve(volNormFIX, &index);
+
+          /* Compute weighting factor for the volume increase, (3.^(-2.*X))/8+1
+           */
+          weightFIX =
+              kOffset2[index] - (int16_t)((kSlope2[index] * volNormFIX) >> 13);
+
+          /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
+          stt->Rxx160_LPw32 = (stt->Rxx160_LPw32 / 64) * 67;
+
+          tmp32 = inMicLevelTmp - stt->minLevel;
+          tmpU32 =
+              ((uint32_t)weightFIX * (uint32_t)(inMicLevelTmp - stt->minLevel));
+          stt->micVol = (tmpU32 >> 14) + stt->minLevel;
+          if (stt->micVol < lastMicVol + 1) {
+            stt->micVol = lastMicVol + 1;
+          }
+
+          inMicLevelTmp = stt->micVol;
+        }
+      } else {
+        /* The signal is inside the desired range which is:
+         * lowerLimit < Rxx160_LP/640 < upperLimit
+         */
+        if (stt->changeToSlowMode > 4000) {
+          stt->msecSpeechInnerChange = 1000;
+          stt->msecSpeechOuterChange = 500;
+          stt->upperLimit = stt->upperPrimaryLimit;
+          stt->lowerLimit = stt->lowerPrimaryLimit;
+        } else {
+          stt->changeToSlowMode += 2;  // in milliseconds
+        }
+        stt->msTooLow = 0;
+        stt->msTooHigh = 0;
+
+        stt->micVol = inMicLevelTmp;
+      }
+    }
+  }
+
+  /* Ensure gain is not increased in presence of echo or after a mute event
+   * (but allow the zeroCtrl() increase on the frame of a mute detection).
+   */
+  if (echo == 1 ||
+      (stt->muteGuardMs > 0 && stt->muteGuardMs < kMuteGuardTimeMs)) {
+    if (stt->micVol > lastMicVol) {
+      stt->micVol = lastMicVol;
+    }
+  }
+
+  /* limit the gain */
+  if (stt->micVol > stt->maxLevel) {
+    stt->micVol = stt->maxLevel;
+  } else if (stt->micVol < stt->minOutput) {
+    stt->micVol = stt->minOutput;
+  }
+
+  *outMicLevel = WEBRTC_SPL_MIN(stt->micVol, stt->maxAnalog) >> stt->scale;
+
+  return 0;
+}
+
+int WebRtcAgc_Analyze(void* agcInst,
+                      const int16_t* const* in_near,
+                      size_t num_bands,
+                      size_t samples,
+                      int32_t inMicLevel,
+                      int32_t* outMicLevel,
+                      int16_t echo,
+                      uint8_t* saturationWarning,
+                      int32_t gains[11]) {
+  LegacyAgc* stt = reinterpret_cast<LegacyAgc*>(agcInst);
+
+  if (stt == NULL) {
+    return -1;
+  }
+
+  if (stt->fs == 8000) {
+    if (samples != 80) {
+      return -1;
+    }
+  } else if (stt->fs == 16000 || stt->fs == 32000 || stt->fs == 48000) {
+    if (samples != 160) {
+      return -1;
+    }
+  } else {
+    return -1;
+  }
+
+  *saturationWarning = 0;
+  // TODO(minyue): PUT IN RANGE CHECKING FOR INPUT LEVELS
+  *outMicLevel = inMicLevel;
+
+  int32_t error =
+      WebRtcAgc_ComputeDigitalGains(&stt->digitalAgc, in_near, num_bands,
+                                    stt->fs, stt->lowLevelSignal, gains);
+  if (error == -1) {
+    return -1;
+  }
+
+  if (stt->agcMode < kAgcModeFixedDigital &&
+      (stt->lowLevelSignal == 0 || stt->agcMode != kAgcModeAdaptiveDigital)) {
+    if (WebRtcAgc_ProcessAnalog(agcInst, inMicLevel, outMicLevel,
+                                stt->vadMic.logRatio, echo,
+                                saturationWarning) == -1) {
+      return -1;
+    }
+  }
+
+  /* update queue */
+  if (stt->inQueue > 1) {
+    memcpy(stt->env[0], stt->env[1], 10 * sizeof(int32_t));
+    memcpy(stt->Rxx16w32_array[0], stt->Rxx16w32_array[1], 5 * sizeof(int32_t));
+  }
+
+  if (stt->inQueue > 0) {
+    stt->inQueue--;
+  }
+
+  return 0;
+}
+
+int WebRtcAgc_Process(const void* agcInst,
+                      const int32_t gains[11],
+                      const int16_t* const* in_near,
+                      size_t num_bands,
+                      int16_t* const* out) {
+  const LegacyAgc* stt = (const LegacyAgc*)agcInst;
+  return WebRtcAgc_ApplyDigitalGains(gains, num_bands, stt->fs, in_near, out);
+}
+
+int WebRtcAgc_set_config(void* agcInst, WebRtcAgcConfig agcConfig) {
+  LegacyAgc* stt;
+  stt = reinterpret_cast<LegacyAgc*>(agcInst);
+
+  if (stt == NULL) {
+    return -1;
+  }
+
+  if (stt->initFlag != kInitCheck) {
+    stt->lastError = AGC_UNINITIALIZED_ERROR;
+    return -1;
+  }
+
+  if (agcConfig.limiterEnable != kAgcFalse &&
+      agcConfig.limiterEnable != kAgcTrue) {
+    stt->lastError = AGC_BAD_PARAMETER_ERROR;
+    return -1;
+  }
+  stt->limiterEnable = agcConfig.limiterEnable;
+  stt->compressionGaindB = agcConfig.compressionGaindB;
+  if ((agcConfig.targetLevelDbfs < 0) || (agcConfig.targetLevelDbfs > 31)) {
+    stt->lastError = AGC_BAD_PARAMETER_ERROR;
+    return -1;
+  }
+  stt->targetLevelDbfs = agcConfig.targetLevelDbfs;
+
+  if (stt->agcMode == kAgcModeFixedDigital) {
+    /* Adjust for different parameter interpretation in FixedDigital mode */
+    stt->compressionGaindB += agcConfig.targetLevelDbfs;
+  }
+
+  /* Update threshold levels for analog adaptation */
+  WebRtcAgc_UpdateAgcThresholds(stt);
+
+  /* Recalculate gain table */
+  if (WebRtcAgc_CalculateGainTable(
+          &(stt->digitalAgc.gainTable[0]), stt->compressionGaindB,
+          stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget) == -1) {
+    return -1;
+  }
+  /* Store the config in a WebRtcAgcConfig */
+  stt->usedConfig.compressionGaindB = agcConfig.compressionGaindB;
+  stt->usedConfig.limiterEnable = agcConfig.limiterEnable;
+  stt->usedConfig.targetLevelDbfs = agcConfig.targetLevelDbfs;
+
+  return 0;
+}
+
+int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config) {
+  LegacyAgc* stt;
+  stt = reinterpret_cast<LegacyAgc*>(agcInst);
+
+  if (stt == NULL) {
+    return -1;
+  }
+
+  if (config == NULL) {
+    stt->lastError = AGC_NULL_POINTER_ERROR;
+    return -1;
+  }
+
+  if (stt->initFlag != kInitCheck) {
+    stt->lastError = AGC_UNINITIALIZED_ERROR;
+    return -1;
+  }
+
+  config->limiterEnable = stt->usedConfig.limiterEnable;
+  config->targetLevelDbfs = stt->usedConfig.targetLevelDbfs;
+  config->compressionGaindB = stt->usedConfig.compressionGaindB;
+
+  return 0;
+}
+
+void* WebRtcAgc_Create() {
+  LegacyAgc* stt = static_cast<LegacyAgc*>(malloc(sizeof(LegacyAgc)));
+
+  stt->initFlag = 0;
+  stt->lastError = 0;
+
+  return stt;
+}
+
+void WebRtcAgc_Free(void* state) {
+  LegacyAgc* stt;
+
+  stt = reinterpret_cast<LegacyAgc*>(state);
+  free(stt);
+}
+
+/* minLevel     - Minimum volume level
+ * maxLevel     - Maximum volume level
+ */
+int WebRtcAgc_Init(void* agcInst,
+                   int32_t minLevel,
+                   int32_t maxLevel,
+                   int16_t agcMode,
+                   uint32_t fs) {
+  int32_t max_add, tmp32;
+  int16_t i;
+  int tmpNorm;
+  LegacyAgc* stt;
+
+  /* typecast state pointer */
+  stt = reinterpret_cast<LegacyAgc*>(agcInst);
+
+  if (WebRtcAgc_InitDigital(&stt->digitalAgc, agcMode) != 0) {
+    stt->lastError = AGC_UNINITIALIZED_ERROR;
+    return -1;
+  }
+
+  /* Analog AGC variables */
+  stt->envSum = 0;
+
+  /* mode     = 0 - Only saturation protection
+   *            1 - Analog Automatic Gain Control [-targetLevelDbfs (default -3
+   * dBOv)]
+   *            2 - Digital Automatic Gain Control [-targetLevelDbfs (default -3
+   * dBOv)]
+   *            3 - Fixed Digital Gain [compressionGaindB (default 8 dB)]
+   */
+  if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital) {
+    return -1;
+  }
+  stt->agcMode = agcMode;
+  stt->fs = fs;
+
+  /* initialize input VAD */
+  WebRtcAgc_InitVad(&stt->vadMic);
+
+  /* If the volume range is smaller than 0-256 then
+   * the levels are shifted up to Q8-domain */
+  tmpNorm = WebRtcSpl_NormU32((uint32_t)maxLevel);
+  stt->scale = tmpNorm - 23;
+  if (stt->scale < 0) {
+    stt->scale = 0;
+  }
+  // TODO(bjornv): Investigate if we really need to scale up a small range now
+  // when we have
+  // a guard against zero-increments. For now, we do not support scale up (scale
+  // = 0).
+  stt->scale = 0;
+  maxLevel <<= stt->scale;
+  minLevel <<= stt->scale;
+
+  /* Make minLevel and maxLevel static in AdaptiveDigital */
+  if (stt->agcMode == kAgcModeAdaptiveDigital) {
+    minLevel = 0;
+    maxLevel = 255;
+    stt->scale = 0;
+  }
+  /* The maximum supplemental volume range is based on a vague idea
+   * of how much lower the gain will be than the real analog gain. */
+  max_add = (maxLevel - minLevel) / 4;
+
+  /* Minimum/maximum volume level that can be set */
+  stt->minLevel = minLevel;
+  stt->maxAnalog = maxLevel;
+  stt->maxLevel = maxLevel + max_add;
+  stt->maxInit = stt->maxLevel;
+
+  stt->zeroCtrlMax = stt->maxAnalog;
+  stt->lastInMicLevel = 0;
+
+  /* Initialize micVol parameter */
+  stt->micVol = stt->maxAnalog;
+  if (stt->agcMode == kAgcModeAdaptiveDigital) {
+    stt->micVol = 127; /* Mid-point of mic level */
+  }
+  stt->micRef = stt->micVol;
+  stt->micGainIdx = 127;
+
+  /* Minimum output volume is 4% higher than the available lowest volume level
+   */
+  tmp32 = ((stt->maxLevel - stt->minLevel) * 10) >> 8;
+  stt->minOutput = (stt->minLevel + tmp32);
+
+  stt->msTooLow = 0;
+  stt->msTooHigh = 0;
+  stt->changeToSlowMode = 0;
+  stt->firstCall = 0;
+  stt->msZero = 0;
+  stt->muteGuardMs = 0;
+  stt->gainTableIdx = 0;
+
+  stt->msecSpeechInnerChange = kMsecSpeechInner;
+  stt->msecSpeechOuterChange = kMsecSpeechOuter;
+
+  stt->activeSpeech = 0;
+  stt->Rxx16_LPw32Max = 0;
+
+  stt->vadThreshold = kNormalVadThreshold;
+  stt->inActive = 0;
+
+  for (i = 0; i < kRxxBufferLen; i++) {
+    stt->Rxx16_vectorw32[i] = (int32_t)1000; /* -54dBm0 */
+  }
+  stt->Rxx160w32 = 125 * kRxxBufferLen; /* (stt->Rxx16_vectorw32[0]>>3) = 125 */
+
+  stt->Rxx16pos = 0;
+  stt->Rxx16_LPw32 = (int32_t)16284; /* Q(-4) */
+
+  for (i = 0; i < 5; i++) {
+    stt->Rxx16w32_array[0][i] = 0;
+  }
+  for (i = 0; i < 10; i++) {
+    stt->env[0][i] = 0;
+    stt->env[1][i] = 0;
+  }
+  stt->inQueue = 0;
+
+  WebRtcSpl_MemSetW32(stt->filterState, 0, 8);
+
+  stt->initFlag = kInitCheck;
+  // Default config settings.
+  stt->defaultConfig.limiterEnable = kAgcTrue;
+  stt->defaultConfig.targetLevelDbfs = AGC_DEFAULT_TARGET_LEVEL;
+  stt->defaultConfig.compressionGaindB = AGC_DEFAULT_COMP_GAIN;
+
+  if (WebRtcAgc_set_config(stt, stt->defaultConfig) == -1) {
+    stt->lastError = AGC_UNSPECIFIED_ERROR;
+    return -1;
+  }
+  stt->Rxx160_LPw32 = stt->analogTargetLevel;  // Initialize rms value
+
+  stt->lowLevelSignal = 0;
+
+  /* Only positive values are allowed that are not too large */
+  if ((minLevel >= maxLevel) || (maxLevel & 0xFC000000)) {
+    return -1;
+  } else {
+    return 0;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/legacy/analog_agc.h b/third_party/libwebrtc/modules/audio_processing/agc/legacy/analog_agc.h
new file mode 100644
index 0000000000..22cd924a93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/legacy/analog_agc.h
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
+#define MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
+
+
+#include "modules/audio_processing/agc/legacy/digital_agc.h"
+#include "modules/audio_processing/agc/legacy/gain_control.h"
+
+namespace webrtc {
+
+/* Analog Automatic Gain Control variables:
+ * Constant declarations (inner limits inside which no changes are done)
+ * In the beginning the range is narrower to widen as soon as the measure
+ * 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0
+ * and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal
+ * go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm
+ * The limits are created by running the AGC with a file having the desired
+ * signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined
+ * by out=10*log10(in/260537279.7); Set the target level to the average level
+ * of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in
+ * Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) )
+ */
+constexpr int16_t kRxxBufferLen = 10;
+
+static const int16_t kMsecSpeechInner = 520;
+static const int16_t kMsecSpeechOuter = 340;
+
+static const int16_t kNormalVadThreshold = 400;
+
+static const int16_t kAlphaShortTerm = 6;  // 1 >> 6 = 0.0156
+static const int16_t kAlphaLongTerm = 10;  // 1 >> 10 = 0.000977
+
+typedef struct {
+  // Configurable parameters/variables
+  uint32_t fs;                // Sampling frequency
+  int16_t compressionGaindB;  // Fixed gain level in dB
+  int16_t targetLevelDbfs;    // Target level in -dBfs of envelope (default -3)
+  int16_t agcMode;            // Hard coded mode (adaptAna/adaptDig/fixedDig)
+  uint8_t limiterEnable;      // Enabling limiter (on/off (default off))
+  WebRtcAgcConfig defaultConfig;
+  WebRtcAgcConfig usedConfig;
+
+  // General variables
+  int16_t initFlag;
+  int16_t lastError;
+
+  // Target level parameters
+  // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7)
+  int32_t analogTargetLevel;    // = kRxxBufferLen * 846805;       -22 dBfs
+  int32_t startUpperLimit;      // = kRxxBufferLen * 1066064;      -21 dBfs
+  int32_t startLowerLimit;      // = kRxxBufferLen * 672641;       -23 dBfs
+  int32_t upperPrimaryLimit;    // = kRxxBufferLen * 1342095;      -20 dBfs
+  int32_t lowerPrimaryLimit;    // = kRxxBufferLen * 534298;       -24 dBfs
+  int32_t upperSecondaryLimit;  // = kRxxBufferLen * 2677832;      -17 dBfs
+  int32_t lowerSecondaryLimit;  // = kRxxBufferLen * 267783;       -27 dBfs
+  uint16_t targetIdx;           // Table index for corresponding target level
+  int16_t analogTarget;  // Digital reference level in ENV scale
+
+  // Analog AGC specific variables
+  int32_t filterState[8];  // For downsampling wb to nb
+  int32_t upperLimit;      // Upper limit for mic energy
+  int32_t lowerLimit;      // Lower limit for mic energy
+  int32_t Rxx160w32;       // Average energy for one frame
+  int32_t Rxx16_LPw32;     // Low pass filtered subframe energies
+  int32_t Rxx160_LPw32;    // Low pass filtered frame energies
+  int32_t Rxx16_LPw32Max;  // Keeps track of largest energy subframe
+  int32_t Rxx16_vectorw32[kRxxBufferLen];  // Array with subframe energies
+  int32_t Rxx16w32_array[2][5];  // Energy values of microphone signal
+  int32_t env[2][10];            // Envelope values of subframes
+
+  int16_t Rxx16pos;          // Current position in the Rxx16_vectorw32
+  int16_t envSum;            // Filtered scaled envelope in subframes
+  int16_t vadThreshold;      // Threshold for VAD decision
+  int16_t inActive;          // Inactive time in milliseconds
+  int16_t msTooLow;          // Milliseconds of speech at a too low level
+  int16_t msTooHigh;         // Milliseconds of speech at a too high level
+  int16_t changeToSlowMode;  // Change to slow mode after some time at target
+  int16_t firstCall;         // First call to the process-function
+  int16_t msZero;            // Milliseconds of zero input
+  int16_t msecSpeechOuterChange;  // Min ms of speech between volume changes
+  int16_t msecSpeechInnerChange;  // Min ms of speech between volume changes
+  int16_t activeSpeech;           // Milliseconds of active speech
+  int16_t muteGuardMs;            // Counter to prevent mute action
+  int16_t inQueue;                // 10 ms batch indicator
+
+  // Microphone level variables
+  int32_t micRef;         // Remember ref. mic level for virtual mic
+  uint16_t gainTableIdx;  // Current position in virtual gain table
+  int32_t micGainIdx;     // Gain index of mic level to increase slowly
+  int32_t micVol;         // Remember volume between frames
+  int32_t maxLevel;       // Max possible vol level, incl dig gain
+  int32_t maxAnalog;      // Maximum possible analog volume level
+  int32_t maxInit;        // Initial value of "max"
+  int32_t minLevel;       // Minimum possible volume level
+  int32_t minOutput;      // Minimum output volume level
+  int32_t zeroCtrlMax;    // Remember max gain => don't amp low input
+  int32_t lastInMicLevel;
+
+  int16_t scale;  // Scale factor for internal volume levels
+  // Structs for VAD and digital_agc
+  AgcVad vadMic;
+  DigitalAgc digitalAgc;
+
+  int16_t lowLevelSignal;
+} LegacyAgc;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.cc b/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.cc
new file mode 100644
index 0000000000..4cd86acba8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.cc
@@ -0,0 +1,704 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc/legacy/digital_agc.h"
+
+#include <string.h>
+
+#include "modules/audio_processing/agc/legacy/gain_control.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// To generate the gaintable, copy&paste the following lines to a Matlab window:
+// MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1;
+// zeros = 0:31; lvl = 2.^(1-zeros);
+// A = -10*log10(lvl) * (CompRatio - 1) / CompRatio;
+// B = MaxGain - MinGain;
+// gains = round(2^16*10.^(0.05 * (MinGain + B * (
+// log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) /
+// log(1/(1+exp(Knee*B))))));
+// fprintf(1, '\t%i, %i, %i, %i,\n', gains);
+// % Matlab code for plotting the gain and input/output level characteristic
+// (copy/paste the following 3 lines):
+// in = 10*log10(lvl); out = 20*log10(gains/65536);
+// subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input
+// (dB)'); ylabel('Gain (dB)');
+// subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on;
+// xlabel('Input (dB)'); ylabel('Output (dB)');
+// zoom on;
+
+// Generator table for y=log2(1+e^x) in Q8.
+enum { kGenFuncTableSize = 128 };
+static const uint16_t kGenFuncTable[kGenFuncTableSize] = {
+    256,   485,   786,   1126,  1484,  1849,  2217,  2586,  2955,  3324,  3693,
+    4063,  4432,  4801,  5171,  5540,  5909,  6279,  6648,  7017,  7387,  7756,
+    8125,  8495,  8864,  9233,  9603,  9972,  10341, 10711, 11080, 11449, 11819,
+    12188, 12557, 12927, 13296, 13665, 14035, 14404, 14773, 15143, 15512, 15881,
+    16251, 16620, 16989, 17359, 17728, 18097, 18466, 18836, 19205, 19574, 19944,
+    20313, 20682, 21052, 21421, 21790, 22160, 22529, 22898, 23268, 23637, 24006,
+    24376, 24745, 25114, 25484, 25853, 26222, 26592, 26961, 27330, 27700, 28069,
+    28438, 28808, 29177, 29546, 29916, 30285, 30654, 31024, 31393, 31762, 32132,
+    32501, 32870, 33240, 33609, 33978, 34348, 34717, 35086, 35456, 35825, 36194,
+    36564, 36933, 37302, 37672, 38041, 38410, 38780, 39149, 39518, 39888, 40257,
+    40626, 40996, 41365, 41734, 42104, 42473, 42842, 43212, 43581, 43950, 44320,
+    44689, 45058, 45428, 45797, 46166, 46536, 46905};
+
+static const int16_t kAvgDecayTime = 250;  // frames; < 3000
+
+// the 32 most significant bits of A(19) * B(26) >> 13
+#define AGC_MUL32(A, B) (((B) >> 13) * (A) + (((0x00001FFF & (B)) * (A)) >> 13))
+// C + the 32 most significant bits of A * B
+#define AGC_SCALEDIFF32(A, B, C) \
+  ((C) + ((B) >> 16) * (A) + (((0x0000FFFF & (B)) * (A)) >> 16))
+
+}  // namespace
+
+int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable,       // Q16
+                                     int16_t digCompGaindB,    // Q0
+                                     int16_t targetLevelDbfs,  // Q0
+                                     uint8_t limiterEnable,
+                                     int16_t analogTarget) {  // Q0
+  // This function generates the compressor gain table used in the fixed digital
+  // part.
+  uint32_t tmpU32no1, tmpU32no2, absInLevel, logApprox;
+  int32_t inLevel, limiterLvl;
+  int32_t tmp32, tmp32no1, tmp32no2, numFIX, den, y32;
+  const uint16_t kLog10 = 54426;    // log2(10)     in Q14
+  const uint16_t kLog10_2 = 49321;  // 10*log10(2)  in Q14
+  const uint16_t kLogE_1 = 23637;   // log2(e)      in Q14
+  uint16_t constMaxGain;
+  uint16_t tmpU16, intPart, fracPart;
+  const int16_t kCompRatio = 3;
+  int16_t limiterOffset = 0;  // Limiter offset
+  int16_t limiterIdx, limiterLvlX;
+  int16_t constLinApprox, maxGain, diffGain;
+  int16_t i, tmp16, tmp16no1;
+  int zeros, zerosScale;
+
+  // Constants
+  //    kLogE_1 = 23637; // log2(e)      in Q14
+  //    kLog10 = 54426; // log2(10)     in Q14
+  //    kLog10_2 = 49321; // 10*log10(2)  in Q14
+
+  // Calculate maximum digital gain and zero gain level
+  tmp32no1 = (digCompGaindB - analogTarget) * (kCompRatio - 1);
+  tmp16no1 = analogTarget - targetLevelDbfs;
+  tmp16no1 +=
+      WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
+  maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs));
+  tmp32no1 = maxGain * kCompRatio;
+  if ((digCompGaindB <= analogTarget) && (limiterEnable)) {
+    limiterOffset = 0;
+  }
+
+  // Calculate the difference between maximum gain and gain at 0dB0v
+  tmp32no1 = digCompGaindB * (kCompRatio - 1);
+  diffGain =
+      WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
+  if (diffGain < 0 || diffGain >= kGenFuncTableSize) {
+    RTC_DCHECK(0);
+    return -1;
+  }
+
+  // Calculate the limiter level and index:
+  //  limiterLvlX = analogTarget - limiterOffset
+  //  limiterLvl  = targetLevelDbfs + limiterOffset/compRatio
+  limiterLvlX = analogTarget - limiterOffset;
+  limiterIdx = 2 + WebRtcSpl_DivW32W16ResW16((int32_t)limiterLvlX * (1 << 13),
+                                             kLog10_2 / 2);
+  tmp16no1 =
+      WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio);
+  limiterLvl = targetLevelDbfs + tmp16no1;
+
+  // Calculate (through table lookup):
+  //  constMaxGain = log2(1+2^(log2(e)*diffGain)); (in Q8)
+  constMaxGain = kGenFuncTable[diffGain];  // in Q8
+
+  // Calculate a parameter used to approximate the fractional part of 2^x with a
+  // piecewise linear function in Q14:
+  //  constLinApprox = round(3/2*(4*(3-2*sqrt(2))/(log(2)^2)-0.5)*2^14);
+  constLinApprox = 22817;  // in Q14
+
+  // Calculate a denominator used in the exponential part to convert from dB to
+  // linear scale:
+  //  den = 20*constMaxGain (in Q8)
+  den = WEBRTC_SPL_MUL_16_U16(20, constMaxGain);  // in Q8
+
+  for (i = 0; i < 32; i++) {
+    // Calculate scaled input level (compressor):
+    //  inLevel =
+    //  fix((-constLog10_2*(compRatio-1)*(1-i)+fix(compRatio/2))/compRatio)
+    tmp16 = (int16_t)((kCompRatio - 1) * (i - 1));       // Q0
+    tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1;  // Q14
+    inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio);    // Q14
+
+    // Calculate diffGain-inLevel, to map using the genFuncTable
+    inLevel = (int32_t)diffGain * (1 << 14) - inLevel;  // Q14
+
+    // Make calculations on abs(inLevel) and compensate for the sign afterwards.
+    absInLevel = (uint32_t)WEBRTC_SPL_ABS_W32(inLevel);  // Q14
+
+    // LUT with interpolation
+    intPart = (uint16_t)(absInLevel >> 14);
+    fracPart =
+        (uint16_t)(absInLevel & 0x00003FFF);  // extract the fractional part
+    tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart];  // Q8
+    tmpU32no1 = tmpU16 * fracPart;                                 // Q22
+    tmpU32no1 += (uint32_t)kGenFuncTable[intPart] << 14;           // Q22
+    logApprox = tmpU32no1 >> 8;                                    // Q14
+    // Compensate for negative exponent using the relation:
+    //  log2(1 + 2^-x) = log2(1 + 2^x) - x
+    if (inLevel < 0) {
+      zeros = WebRtcSpl_NormU32(absInLevel);
+      zerosScale = 0;
+      if (zeros < 15) {
+        // Not enough space for multiplication
+        tmpU32no2 = absInLevel >> (15 - zeros);                 // Q(zeros-1)
+        tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1);  // Q(zeros+13)
+        if (zeros < 9) {
+          zerosScale = 9 - zeros;
+          tmpU32no1 >>= zerosScale;  // Q(zeros+13)
+        } else {
+          tmpU32no2 >>= zeros - 9;  // Q22
+        }
+      } else {
+        tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1);  // Q28
+        tmpU32no2 >>= 6;                                         // Q22
+      }
+      logApprox = 0;
+      if (tmpU32no2 < tmpU32no1) {
+        logApprox = (tmpU32no1 - tmpU32no2) >> (8 - zerosScale);  // Q14
+      }
+    }
+    numFIX = (maxGain * constMaxGain) * (1 << 6);  // Q14
+    numFIX -= (int32_t)logApprox * diffGain;       // Q14
+
+    // Calculate ratio
+    // Shift `numFIX` as much as possible.
+    // Ensure we avoid wrap-around in `den` as well.
+    if (numFIX > (den >> 8) || -numFIX > (den >> 8)) {  // `den` is Q8.
+      zeros = WebRtcSpl_NormW32(numFIX);
+    } else {
+      zeros = WebRtcSpl_NormW32(den) + 8;
+    }
+    numFIX *= 1 << zeros;  // Q(14+zeros)
+
+    // Shift den so we end up in Qy1
+    tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 9);  // Q(zeros - 1)
+    y32 = numFIX / tmp32no1;                          // in Q15
+    // This is to do rounding in Q14.
+    y32 = y32 >= 0 ? (y32 + 1) >> 1 : -((-y32 + 1) >> 1);
+
+    if (limiterEnable && (i < limiterIdx)) {
+      tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2);  // Q14
+      tmp32 -= limiterLvl * (1 << 14);                 // Q14
+      y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20);
+    }
+    if (y32 > 39000) {
+      tmp32 = (y32 >> 1) * kLog10 + 4096;  // in Q27
+      tmp32 >>= 13;                        // In Q14.
+    } else {
+      tmp32 = y32 * kLog10 + 8192;  // in Q28
+      tmp32 >>= 14;                 // In Q14.
+    }
+    tmp32 += 16 << 14;  // in Q14 (Make sure final output is in Q16)
+
+    // Calculate power
+    if (tmp32 > 0) {
+      intPart = (int16_t)(tmp32 >> 14);
+      fracPart = (uint16_t)(tmp32 & 0x00003FFF);  // in Q14
+      if ((fracPart >> 13) != 0) {
+        tmp16 = (2 << 14) - constLinApprox;
+        tmp32no2 = (1 << 14) - fracPart;
+        tmp32no2 *= tmp16;
+        tmp32no2 >>= 13;
+        tmp32no2 = (1 << 14) - tmp32no2;
+      } else {
+        tmp16 = constLinApprox - (1 << 14);
+        tmp32no2 = (fracPart * tmp16) >> 13;
+      }
+      fracPart = (uint16_t)tmp32no2;
+      gainTable[i] =
+          (1 << intPart) + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14);
+    } else {
+      gainTable[i] = 0;
+    }
+  }
+
+  return 0;
+}
+
+int32_t WebRtcAgc_InitDigital(DigitalAgc* stt, int16_t agcMode) {
+  if (agcMode == kAgcModeFixedDigital) {
+    // start at minimum to find correct gain faster
+    stt->capacitorSlow = 0;
+  } else {
+    // start out with 0 dB gain
+    stt->capacitorSlow = 134217728;  // (int32_t)(0.125f * 32768.0f * 32768.0f);
+  }
+  stt->capacitorFast = 0;
+  stt->gain = 65536;
+  stt->gatePrevious = 0;
+  stt->agcMode = agcMode;
+
+  // initialize VADs
+  WebRtcAgc_InitVad(&stt->vadNearend);
+  WebRtcAgc_InitVad(&stt->vadFarend);
+
+  return 0;
+}
+
+int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* stt,
+                                     const int16_t* in_far,
+                                     size_t nrSamples) {
+  RTC_DCHECK(stt);
+  // VAD for far end
+  WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
+
+  return 0;
+}
+
+// Gains is an 11 element long array (one value per ms, incl start & end).
+int32_t WebRtcAgc_ComputeDigitalGains(DigitalAgc* stt,
+                                      const int16_t* const* in_near,
+                                      size_t num_bands,
+                                      uint32_t FS,
+                                      int16_t lowlevelSignal,
+                                      int32_t gains[11]) {
+  int32_t tmp32;
+  int32_t env[10];
+  int32_t max_nrg;
+  int32_t cur_level;
+  int32_t gain32;
+  int16_t logratio;
+  int16_t lower_thr, upper_thr;
+  int16_t zeros = 0, zeros_fast, frac = 0;
+  int16_t decay;
+  int16_t gate, gain_adj;
+  int16_t k;
+  size_t n, L;
+
+  // determine number of samples per ms
+  if (FS == 8000) {
+    L = 8;
+  } else if (FS == 16000 || FS == 32000 || FS == 48000) {
+    L = 16;
+  } else {
+    return -1;
+  }
+
+  // VAD for near end
+  logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, in_near[0], L * 10);
+
+  // Account for far end VAD
+  if (stt->vadFarend.counter > 10) {
+    tmp32 = 3 * logratio;
+    logratio = (int16_t)((tmp32 - stt->vadFarend.logRatio) >> 2);
+  }
+
+  // Determine decay factor depending on VAD
+  //  upper_thr = 1.0f;
+  //  lower_thr = 0.25f;
+  upper_thr = 1024;  // Q10
+  lower_thr = 0;     // Q10
+  if (logratio > upper_thr) {
+    // decay = -2^17 / DecayTime;  ->  -65
+    decay = -65;
+  } else if (logratio < lower_thr) {
+    decay = 0;
+  } else {
+    // decay = (int16_t)(((lower_thr - logratio)
+    //       * (2^27/(DecayTime*(upper_thr-lower_thr)))) >> 10);
+    // SUBSTITUTED: 2^27/(DecayTime*(upper_thr-lower_thr))  ->  65
+    tmp32 = (lower_thr - logratio) * 65;
+    decay = (int16_t)(tmp32 >> 10);
+  }
+
+  // adjust decay factor for long silence (detected as low standard deviation)
+  // This is only done in the adaptive modes
+  if (stt->agcMode != kAgcModeFixedDigital) {
+    if (stt->vadNearend.stdLongTerm < 4000) {
+      decay = 0;
+    } else if (stt->vadNearend.stdLongTerm < 8096) {
+      // decay = (int16_t)(((stt->vadNearend.stdLongTerm - 4000) * decay) >>
+      // 12);
+      tmp32 = (stt->vadNearend.stdLongTerm - 4000) * decay;
+      decay = (int16_t)(tmp32 >> 12);
+    }
+
+    if (lowlevelSignal != 0) {
+      decay = 0;
+    }
+  }
+  // Find max amplitude per sub frame
+  // iterate over sub frames
+  for (k = 0; k < 10; k++) {
+    // iterate over samples
+    max_nrg = 0;
+    for (n = 0; n < L; n++) {
+      int32_t nrg = in_near[0][k * L + n] * in_near[0][k * L + n];
+      if (nrg > max_nrg) {
+        max_nrg = nrg;
+      }
+    }
+    env[k] = max_nrg;
+  }
+
+  // Calculate gain per sub frame
+  gains[0] = stt->gain;
+  for (k = 0; k < 10; k++) {
+    // Fast envelope follower
+    //  decay time = -131000 / -1000 = 131 (ms)
+    stt->capacitorFast =
+        AGC_SCALEDIFF32(-1000, stt->capacitorFast, stt->capacitorFast);
+    if (env[k] > stt->capacitorFast) {
+      stt->capacitorFast = env[k];
+    }
+    // Slow envelope follower
+    if (env[k] > stt->capacitorSlow) {
+      // increase capacitorSlow
+      stt->capacitorSlow = AGC_SCALEDIFF32(500, (env[k] - stt->capacitorSlow),
+                                           stt->capacitorSlow);
+    } else {
+      // decrease capacitorSlow
+      stt->capacitorSlow =
+          AGC_SCALEDIFF32(decay, stt->capacitorSlow, stt->capacitorSlow);
+    }
+
+    // use maximum of both capacitors as current level
+    if (stt->capacitorFast > stt->capacitorSlow) {
+      cur_level = stt->capacitorFast;
+    } else {
+      cur_level = stt->capacitorSlow;
+    }
+    // Translate signal level into gain, using a piecewise linear approximation
+    // find number of leading zeros
+    zeros = WebRtcSpl_NormU32((uint32_t)cur_level);
+    if (cur_level == 0) {
+      zeros = 31;
+    }
+    tmp32 = ((uint32_t)cur_level << zeros) & 0x7FFFFFFF;
+    frac = (int16_t)(tmp32 >> 19);  // Q12.
+    // Interpolate between gainTable[zeros] and gainTable[zeros-1].
+    tmp32 =
+        ((stt->gainTable[zeros - 1] - stt->gainTable[zeros]) * (int64_t)frac) >>
+        12;
+    gains[k + 1] = stt->gainTable[zeros] + tmp32;
+  }
+
+  // Gate processing (lower gain during absence of speech)
+  zeros = (zeros << 9) - (frac >> 3);
+  // find number of leading zeros
+  zeros_fast = WebRtcSpl_NormU32((uint32_t)stt->capacitorFast);
+  if (stt->capacitorFast == 0) {
+    zeros_fast = 31;
+  }
+  tmp32 = ((uint32_t)stt->capacitorFast << zeros_fast) & 0x7FFFFFFF;
+  zeros_fast <<= 9;
+  zeros_fast -= (int16_t)(tmp32 >> 22);
+
+  gate = 1000 + zeros_fast - zeros - stt->vadNearend.stdShortTerm;
+
+  if (gate < 0) {
+    stt->gatePrevious = 0;
+  } else {
+    tmp32 = stt->gatePrevious * 7;
+    gate = (int16_t)((gate + tmp32) >> 3);
+    stt->gatePrevious = gate;
+  }
+  // gate < 0     -> no gate
+  // gate > 2500  -> max gate
+  if (gate > 0) {
+    if (gate < 2500) {
+      gain_adj = (2500 - gate) >> 5;
+    } else {
+      gain_adj = 0;
+    }
+    for (k = 0; k < 10; k++) {
+      if ((gains[k + 1] - stt->gainTable[0]) > 8388608) {
+        // To prevent wraparound
+        tmp32 = (gains[k + 1] - stt->gainTable[0]) >> 8;
+        tmp32 *= 178 + gain_adj;
+      } else {
+        tmp32 = (gains[k + 1] - stt->gainTable[0]) * (178 + gain_adj);
+        tmp32 >>= 8;
+      }
+      gains[k + 1] = stt->gainTable[0] + tmp32;
+    }
+  }
+
+  // Limit gain to avoid overload distortion
+  for (k = 0; k < 10; k++) {
+    // Find a shift of gains[k + 1] such that it can be squared without
+    // overflow, but at least by 10 bits.
+    zeros = 10;
+    if (gains[k + 1] > 47452159) {
+      zeros = 16 - WebRtcSpl_NormW32(gains[k + 1]);
+    }
+    gain32 = (gains[k + 1] >> zeros) + 1;
+    gain32 *= gain32;
+    // check for overflow
+    while (AGC_MUL32((env[k] >> 12) + 1, gain32) >
+           WEBRTC_SPL_SHIFT_W32((int32_t)32767, 2 * (1 - zeros + 10))) {
+      // multiply by 253/256 ==> -0.1 dB
+      if (gains[k + 1] > 8388607) {
+        // Prevent wrap around
+        gains[k + 1] = (gains[k + 1] / 256) * 253;
+      } else {
+        gains[k + 1] = (gains[k + 1] * 253) / 256;
+      }
+      gain32 = (gains[k + 1] >> zeros) + 1;
+      gain32 *= gain32;
+    }
+  }
+  // gain reductions should be done 1 ms earlier than gain increases
+  for (k = 1; k < 10; k++) {
+    if (gains[k] > gains[k + 1]) {
+      gains[k] = gains[k + 1];
+    }
+  }
+  // save start gain for next frame
+  stt->gain = gains[10];
+
+  return 0;
+}
+
+int32_t WebRtcAgc_ApplyDigitalGains(const int32_t gains[11],
+                                    size_t num_bands,
+                                    uint32_t FS,
+                                    const int16_t* const* in_near,
+                                    int16_t* const* out) {
+  // Apply gain
+  // handle first sub frame separately
+  size_t L;
+  int16_t L2;  // samples/subframe
+
+  // determine number of samples per ms
+  if (FS == 8000) {
+    L = 8;
+    L2 = 3;
+  } else if (FS == 16000 || FS == 32000 || FS == 48000) {
+    L = 16;
+    L2 = 4;
+  } else {
+    return -1;
+  }
+
+  for (size_t i = 0; i < num_bands; ++i) {
+    if (in_near[i] != out[i]) {
+      // Only needed if they don't already point to the same place.
+      memcpy(out[i], in_near[i], 10 * L * sizeof(in_near[i][0]));
+    }
+  }
+
+  // iterate over samples
+  int32_t delta = (gains[1] - gains[0]) * (1 << (4 - L2));
+  int32_t gain32 = gains[0] * (1 << 4);
+  for (size_t n = 0; n < L; n++) {
+    for (size_t i = 0; i < num_bands; ++i) {
+      int32_t out_tmp = (int64_t)out[i][n] * ((gain32 + 127) >> 7) >> 16;
+      if (out_tmp > 4095) {
+        out[i][n] = (int16_t)32767;
+      } else if (out_tmp < -4096) {
+        out[i][n] = (int16_t)-32768;
+      } else {
+        int32_t tmp32 = ((int64_t)out[i][n] * (gain32 >> 4)) >> 16;
+        out[i][n] = (int16_t)tmp32;
+      }
+    }
+
+    gain32 += delta;
+  }
+  // iterate over subframes
+  for (int k = 1; k < 10; k++) {
+    delta = (gains[k + 1] - gains[k]) * (1 << (4 - L2));
+    gain32 = gains[k] * (1 << 4);
+    // iterate over samples
+    for (size_t n = 0; n < L; n++) {
+      for (size_t i = 0; i < num_bands; ++i) {
+        int64_t tmp64 = ((int64_t)(out[i][k * L + n])) * (gain32 >> 4);
+        tmp64 = tmp64 >> 16;
+        if (tmp64 > 32767) {
+          out[i][k * L + n] = 32767;
+        } else if (tmp64 < -32768) {
+          out[i][k * L + n] = -32768;
+        } else {
+          out[i][k * L + n] = (int16_t)(tmp64);
+        }
+      }
+      gain32 += delta;
+    }
+  }
+  return 0;
+}
+
+void WebRtcAgc_InitVad(AgcVad* state) {
+  int16_t k;
+
+  state->HPstate = 0;   // state of high pass filter
+  state->logRatio = 0;  // log( P(active) / P(inactive) )
+  // average input level (Q10)
+  state->meanLongTerm = 15 << 10;
+
+  // variance of input level (Q8)
+  state->varianceLongTerm = 500 << 8;
+
+  state->stdLongTerm = 0;  // standard deviation of input level in dB
+  // short-term average input level (Q10)
+  state->meanShortTerm = 15 << 10;
+
+  // short-term variance of input level (Q8)
+  state->varianceShortTerm = 500 << 8;
+
+  state->stdShortTerm =
+      0;               // short-term standard deviation of input level in dB
+  state->counter = 3;  // counts updates
+  for (k = 0; k < 8; k++) {
+    // downsampling filter
+    state->downState[k] = 0;
+  }
+}
+
+int16_t WebRtcAgc_ProcessVad(AgcVad* state,       // (i) VAD state
+                             const int16_t* in,   // (i) Speech signal
+                             size_t nrSamples) {  // (i) number of samples
+  uint32_t nrg;
+  int32_t out, tmp32, tmp32b;
+  uint16_t tmpU16;
+  int16_t k, subfr, tmp16;
+  int16_t buf1[8];
+  int16_t buf2[4];
+  int16_t HPstate;
+  int16_t zeros, dB;
+  int64_t tmp64;
+
+  // process in 10 sub frames of 1 ms (to save on memory)
+  nrg = 0;
+  HPstate = state->HPstate;
+  for (subfr = 0; subfr < 10; subfr++) {
+    // downsample to 4 kHz
+    if (nrSamples == 160) {
+      for (k = 0; k < 8; k++) {
+        tmp32 = (int32_t)in[2 * k] + (int32_t)in[2 * k + 1];
+        tmp32 >>= 1;
+        buf1[k] = (int16_t)tmp32;
+      }
+      in += 16;
+
+      WebRtcSpl_DownsampleBy2(buf1, 8, buf2, state->downState);
+    } else {
+      WebRtcSpl_DownsampleBy2(in, 8, buf2, state->downState);
+      in += 8;
+    }
+
+    // high pass filter and compute energy
+    for (k = 0; k < 4; k++) {
+      out = buf2[k] + HPstate;
+      tmp32 = 600 * out;
+      HPstate = (int16_t)((tmp32 >> 10) - buf2[k]);
+
+      // Add 'out * out / 2**6' to 'nrg' in a non-overflowing
+      // way. Guaranteed to work as long as 'out * out / 2**6' fits in
+      // an int32_t.
+      nrg += out * (out / (1 << 6));
+      nrg += out * (out % (1 << 6)) / (1 << 6);
+    }
+  }
+  state->HPstate = HPstate;
+
+  // find number of leading zeros
+  if (!(0xFFFF0000 & nrg)) {
+    zeros = 16;
+  } else {
+    zeros = 0;
+  }
+  if (!(0xFF000000 & (nrg << zeros))) {
+    zeros += 8;
+  }
+  if (!(0xF0000000 & (nrg << zeros))) {
+    zeros += 4;
+  }
+  if (!(0xC0000000 & (nrg << zeros))) {
+    zeros += 2;
+  }
+  if (!(0x80000000 & (nrg << zeros))) {
+    zeros += 1;
+  }
+
+  // energy level (range {-32..30}) (Q10)
+  dB = (15 - zeros) * (1 << 11);
+
+  // Update statistics
+
+  if (state->counter < kAvgDecayTime) {
+    // decay time = AvgDecTime * 10 ms
+    state->counter++;
+  }
+
+  // update short-term estimate of mean energy level (Q10)
+  tmp32 = state->meanShortTerm * 15 + dB;
+  state->meanShortTerm = (int16_t)(tmp32 >> 4);
+
+  // update short-term estimate of variance in energy level (Q8)
+  tmp32 = (dB * dB) >> 12;
+  tmp32 += state->varianceShortTerm * 15;
+  state->varianceShortTerm = tmp32 / 16;
+
+  // update short-term estimate of standard deviation in energy level (Q10)
+  tmp32 = state->meanShortTerm * state->meanShortTerm;
+  tmp32 = (state->varianceShortTerm << 12) - tmp32;
+  state->stdShortTerm = (int16_t)WebRtcSpl_Sqrt(tmp32);
+
+  // update long-term estimate of mean energy level (Q10)
+  tmp32 = state->meanLongTerm * state->counter + dB;
+  state->meanLongTerm =
+      WebRtcSpl_DivW32W16ResW16(tmp32, WebRtcSpl_AddSatW16(state->counter, 1));
+
+  // update long-term estimate of variance in energy level (Q8)
+  tmp32 = (dB * dB) >> 12;
+  tmp32 += state->varianceLongTerm * state->counter;
+  state->varianceLongTerm =
+      WebRtcSpl_DivW32W16(tmp32, WebRtcSpl_AddSatW16(state->counter, 1));
+
+  // update long-term estimate of standard deviation in energy level (Q10)
+  tmp32 = state->meanLongTerm * state->meanLongTerm;
+  tmp32 = (state->varianceLongTerm << 12) - tmp32;
+  state->stdLongTerm = (int16_t)WebRtcSpl_Sqrt(tmp32);
+
+  // update voice activity measure (Q10)
+  tmp16 = 3 << 12;
+  // TODO(bjornv): (dB - state->meanLongTerm) can overflow, e.g., in
+  // ApmTest.Process unit test. Previously the macro WEBRTC_SPL_MUL_16_16()
+  // was used, which did an intermediate cast to (int16_t), hence losing
+  // significant bits. This cause logRatio to max out positive, rather than
+  // negative. This is a bug, but has very little significance.
+  tmp32 = tmp16 * (int16_t)(dB - state->meanLongTerm);
+  tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm);
+  tmpU16 = (13 << 12);
+  tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16);
+  tmp64 = tmp32;
+  tmp64 += tmp32b >> 10;
+  tmp64 >>= 6;
+
+  // limit
+  if (tmp64 > 2048) {
+    tmp64 = 2048;
+  } else if (tmp64 < -2048) {
+    tmp64 = -2048;
+  }
+  state->logRatio = (int16_t)tmp64;
+
+  return state->logRatio;  // Q10
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.h b/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.h
new file mode 100644
index 0000000000..223c74b9bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
+#define MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+typedef struct {
+  int32_t downState[8];
+  int16_t HPstate;
+  int16_t counter;
+  int16_t logRatio;           // log( P(active) / P(inactive) ) (Q10)
+  int16_t meanLongTerm;       // Q10
+  int32_t varianceLongTerm;   // Q8
+  int16_t stdLongTerm;        // Q10
+  int16_t meanShortTerm;      // Q10
+  int32_t varianceShortTerm;  // Q8
+  int16_t stdShortTerm;       // Q10
+} AgcVad;                     // total = 54 bytes
+
+typedef struct {
+  int32_t capacitorSlow;
+  int32_t capacitorFast;
+  int32_t gain;
+  int32_t gainTable[32];
+  int16_t gatePrevious;
+  int16_t agcMode;
+  AgcVad vadNearend;
+  AgcVad vadFarend;
+} DigitalAgc;
+
+int32_t WebRtcAgc_InitDigital(DigitalAgc* digitalAgcInst, int16_t agcMode);
+
+int32_t WebRtcAgc_ComputeDigitalGains(DigitalAgc* digitalAgcInst,
+                                      const int16_t* const* inNear,
+                                      size_t num_bands,
+                                      uint32_t FS,
+                                      int16_t lowLevelSignal,
+                                      int32_t gains[11]);
+
+int32_t WebRtcAgc_ApplyDigitalGains(const int32_t gains[11],
+                                    size_t num_bands,
+                                    uint32_t FS,
+                                    const int16_t* const* in_near,
+                                    int16_t* const* out);
+
+int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
+                                     const int16_t* inFar,
+                                     size_t nrSamples);
+
+void WebRtcAgc_InitVad(AgcVad* vadInst);
+
+int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst,    // (i) VAD state
+                             const int16_t* in,  // (i) Speech signal
+                             size_t nrSamples);  // (i) number of samples
+
+int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable,         // Q16
+                                     int16_t compressionGaindB,  // Q0 (in dB)
+                                     int16_t targetLevelDbfs,    // Q0 (in dB)
+                                     uint8_t limiterEnable,
+                                     int16_t analogTarget);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/legacy/gain_control.h b/third_party/libwebrtc/modules/audio_processing/agc/legacy/gain_control.h
new file mode 100644
index 0000000000..6010a988fa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/legacy/gain_control.h
@@ -0,0 +1,256 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
+#define MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+enum {
+  kAgcModeUnchanged,
+  kAgcModeAdaptiveAnalog,
+  kAgcModeAdaptiveDigital,
+  kAgcModeFixedDigital
+};
+
+enum { kAgcFalse = 0, kAgcTrue };
+
+typedef struct {
+  int16_t targetLevelDbfs;    // default 3 (-3 dBOv)
+  int16_t compressionGaindB;  // default 9 dB
+  uint8_t limiterEnable;      // default kAgcTrue (on)
+} WebRtcAgcConfig;
+
+/*
+ * This function analyses the number of samples passed to
+ * farend and produces any error code that could arise.
+ *
+ * Input:
+ *      - agcInst           : AGC instance.
+ *      - samples           : Number of samples in input vector.
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error.
+ */
+int WebRtcAgc_GetAddFarendError(void* state, size_t samples);
+
+/*
+ * This function processes a 10 ms frame of far-end speech to determine
+ * if there is active speech. The length of the input speech vector must be
+ * given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
+ * FS=48000).
+ *
+ * Input:
+ *      - agcInst           : AGC instance.
+ *      - inFar             : Far-end input speech vector
+ *      - samples           : Number of samples in input vector
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error
+ */
+int WebRtcAgc_AddFarend(void* agcInst, const int16_t* inFar, size_t samples);
+
+/*
+ * This function processes a 10 ms frame of microphone speech to determine
+ * if there is active speech. The length of the input speech vector must be
+ * given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
+ * FS=48000). For very low input levels, the input signal is increased in level
+ * by multiplying and overwriting the samples in inMic[].
+ *
+ * This function should be called before any further processing of the
+ * near-end microphone signal.
+ *
+ * Input:
+ *      - agcInst           : AGC instance.
+ *      - inMic             : Microphone input speech vector for each band
+ *      - num_bands         : Number of bands in input vector
+ *      - samples           : Number of samples in input vector
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error
+ */
+int WebRtcAgc_AddMic(void* agcInst,
+                     int16_t* const* inMic,
+                     size_t num_bands,
+                     size_t samples);
+
+/*
+ * This function replaces the analog microphone with a virtual one.
+ * It is a digital gain applied to the input signal and is used in the
+ * agcAdaptiveDigital mode where no microphone level is adjustable. The length
+ * of the input speech vector must be given in samples (80 when FS=8000, and 160
+ * when FS=16000, FS=32000 or FS=48000).
+ *
+ * Input:
+ *      - agcInst           : AGC instance.
+ *      - inMic             : Microphone input speech vector for each band
+ *      - num_bands         : Number of bands in input vector
+ *      - samples           : Number of samples in input vector
+ *      - micLevelIn        : Input level of microphone (static)
+ *
+ * Output:
+ *      - inMic             : Microphone output after processing (L band)
+ *      - inMic_H           : Microphone output after processing (H band)
+ *      - micLevelOut       : Adjusted microphone level after processing
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error
+ */
+int WebRtcAgc_VirtualMic(void* agcInst,
+                         int16_t* const* inMic,
+                         size_t num_bands,
+                         size_t samples,
+                         int32_t micLevelIn,
+                         int32_t* micLevelOut);
+
+/*
+ * This function analyses a 10 ms frame and produces the analog and digital
+ * gains required to normalize the signal. The gain adjustments are done only
+ * during active periods of speech. The length of the speech vectors must be
+ * given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
+ * FS=48000). The echo parameter can be used to ensure the AGC will not adjust
+ * upward in the presence of echo.
+ *
+ * This function should be called after processing the near-end microphone
+ * signal, in any case after any echo cancellation.
+ *
+ * Input:
+ *      - agcInst           : AGC instance
+ *      - inNear            : Near-end input speech vector for each band
+ *      - num_bands         : Number of bands in input/output vector
+ *      - samples           : Number of samples in input/output vector
+ *      - inMicLevel        : Current microphone volume level
+ *      - echo              : Set to 0 if the signal passed to add_mic is
+ *                            almost certainly free of echo; otherwise set
+ *                            to 1. If you have no information regarding echo
+ *                            set to 0.
+ *
+ * Output:
+ *      - outMicLevel       : Adjusted microphone volume level
+ *      - saturationWarning : A returned value of 1 indicates a saturation event
+ *                            has occurred and the volume cannot be further
+ *                            reduced. Otherwise will be set to 0.
+ *      - gains             : Vector of gains to apply for digital normalization
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error
+ */
+int WebRtcAgc_Analyze(void* agcInst,
+                      const int16_t* const* inNear,
+                      size_t num_bands,
+                      size_t samples,
+                      int32_t inMicLevel,
+                      int32_t* outMicLevel,
+                      int16_t echo,
+                      uint8_t* saturationWarning,
+                      int32_t gains[11]);
+
+/*
+ * This function processes a 10 ms frame by applying precomputed digital gains.
+ *
+ * Input:
+ *      - agcInst           : AGC instance
+ *      - gains             : Vector of gains to apply for digital normalization
+ *      - in_near           : Near-end input speech vector for each band
+ *      - num_bands         : Number of bands in input/output vector
+ *
+ * Output:
+ *      - out               : Gain-adjusted near-end speech vector
+ *                          : May be the same vector as the input.
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error
+ */
+int WebRtcAgc_Process(const void* agcInst,
+                      const int32_t gains[11],
+                      const int16_t* const* in_near,
+                      size_t num_bands,
+                      int16_t* const* out);
+
+/*
+ * This function sets the config parameters (targetLevelDbfs,
+ * compressionGaindB and limiterEnable).
+ *
+ * Input:
+ *      - agcInst           : AGC instance
+ *      - config            : config struct
+ *
+ * Output:
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error
+ */
+int WebRtcAgc_set_config(void* agcInst, WebRtcAgcConfig config);
+
+/*
+ * This function returns the config parameters (targetLevelDbfs,
+ * compressionGaindB and limiterEnable).
+ *
+ * Input:
+ *      - agcInst           : AGC instance
+ *
+ * Output:
+ *      - config            : config struct
+ *
+ * Return value:
+ *                          :  0 - Normal operation.
+ *                          : -1 - Error
+ */
+int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config);
+
+/*
+ * This function creates and returns an AGC instance, which will contain the
+ * state information for one (duplex) channel.
+ */
+void* WebRtcAgc_Create(void);
+
+/*
+ * This function frees the AGC instance created at the beginning.
+ *
+ * Input:
+ *      - agcInst           : AGC instance.
+ */
+void WebRtcAgc_Free(void* agcInst);
+
+/*
+ * This function initializes an AGC instance.
+ *
+ * Input:
+ *      - agcInst           : AGC instance.
+ *      - minLevel          : Minimum possible mic level
+ *      - maxLevel          : Maximum possible mic level
+ *      - agcMode           : 0 - Unchanged
+ *                          : 1 - Adaptive Analog Automatic Gain Control -3dBOv
+ *                          : 2 - Adaptive Digital Automatic Gain Control -3dBOv
+ *                          : 3 - Fixed Digital Gain 0dB
+ *      - fs                : Sampling frequency
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+int WebRtcAgc_Init(void* agcInst,
+                   int32_t minLevel,
+                   int32_t maxLevel,
+                   int16_t agcMode,
+                   uint32_t fs);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/legacy_agc_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc/legacy_agc_gn/moz.build
new file mode 100644
index 0000000000..0188a8ac10
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/legacy_agc_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc/legacy/analog_agc.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc/legacy/digital_agc.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("legacy_agc_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/level_estimation_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc/level_estimation_gn/moz.build
new file mode 100644
index 0000000000..9db9a639e7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/level_estimation_gn/moz.build
@@ -0,0 +1,234 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc/agc.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc/utility.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("level_estimation_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram.cc b/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram.cc
new file mode 100644
index 0000000000..b0a1f53b97
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram.cc
@@ -0,0 +1,229 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc/loudness_histogram.h"
+
+#include <string.h>
+
+#include <cmath>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+static const double kHistBinCenters[] = {
+    7.59621091765857e-02, 9.02036021061016e-02, 1.07115112009343e-01,
+    1.27197217770508e-01, 1.51044347572047e-01, 1.79362373905283e-01,
+    2.12989507320644e-01, 2.52921107370304e-01, 3.00339145144454e-01,
+    3.56647189489147e-01, 4.23511952494003e-01, 5.02912623991786e-01,
+    5.97199455365749e-01, 7.09163326739184e-01, 8.42118356728544e-01,
+    1.00000000000000e+00, 1.18748153630660e+00, 1.41011239906908e+00,
+    1.67448243801153e+00, 1.98841697800836e+00, 2.36120844786349e+00,
+    2.80389143520905e+00, 3.32956930911896e+00, 3.95380207843188e+00,
+    4.69506696634852e+00, 5.57530533426190e+00, 6.62057214370769e+00,
+    7.86180718043869e+00, 9.33575086877358e+00, 1.10860317842269e+01,
+    1.31644580546776e+01, 1.56325508754123e+01, 1.85633655299256e+01,
+    2.20436538184971e+01, 2.61764319021997e+01, 3.10840295702492e+01,
+    3.69117111886792e+01, 4.38319755100383e+01, 5.20496616180135e+01,
+    6.18080121423973e+01, 7.33958732149108e+01, 8.71562442838066e+01,
+    1.03496430860848e+02, 1.22900100720889e+02, 1.45941600416277e+02,
+    1.73302955873365e+02, 2.05794060286978e+02, 2.44376646872353e+02,
+    2.90192756065437e+02, 3.44598539797631e+02, 4.09204403447902e+02,
+    4.85922673669740e+02, 5.77024203055553e+02, 6.85205587130498e+02,
+    8.13668983291589e+02, 9.66216894324125e+02, 1.14736472207740e+03,
+    1.36247442287647e+03, 1.61791322085579e+03, 1.92124207711260e+03,
+    2.28143949334655e+03, 2.70916727454970e+03, 3.21708611729384e+03,
+    3.82023036499473e+03, 4.53645302286906e+03, 5.38695420497926e+03,
+    6.39690865534207e+03, 7.59621091765857e+03, 9.02036021061016e+03,
+    1.07115112009343e+04, 1.27197217770508e+04, 1.51044347572047e+04,
+    1.79362373905283e+04, 2.12989507320644e+04, 2.52921107370304e+04,
+    3.00339145144454e+04, 3.56647189489147e+04};
+
+static const double kProbQDomain = 1024.0;
+// Loudness of -15 dB (smallest expected loudness) in log domain,
+// loudness_db = 13.5 * log10(rms);
+static const double kLogDomainMinBinCenter = -2.57752062648587;
+// Loudness step of 1 dB in log domain
+static const double kLogDomainStepSizeInverse = 5.81954605750359;
+
+static const int kTransientWidthThreshold = 7;
+static const double kLowProbabilityThreshold = 0.2;
+
+static const int kLowProbThresholdQ10 =
+    static_cast<int>(kLowProbabilityThreshold * kProbQDomain);
+
+LoudnessHistogram::LoudnessHistogram()
+    : num_updates_(0),
+      audio_content_q10_(0),
+      bin_count_q10_(),
+      activity_probability_(),
+      hist_bin_index_(),
+      buffer_index_(0),
+      buffer_is_full_(false),
+      len_circular_buffer_(0),
+      len_high_activity_(0) {
+  static_assert(
+      kHistSize == sizeof(kHistBinCenters) / sizeof(kHistBinCenters[0]),
+      "histogram bin centers incorrect size");
+}
+
+LoudnessHistogram::LoudnessHistogram(int window_size)
+    : num_updates_(0),
+      audio_content_q10_(0),
+      bin_count_q10_(),
+      activity_probability_(new int[window_size]),
+      hist_bin_index_(new int[window_size]),
+      buffer_index_(0),
+      buffer_is_full_(false),
+      len_circular_buffer_(window_size),
+      len_high_activity_(0) {}
+
+LoudnessHistogram::~LoudnessHistogram() {}
+
+void LoudnessHistogram::Update(double rms, double activity_probaility) {
+  // If circular histogram is activated then remove the oldest entry.
+  if (len_circular_buffer_ > 0)
+    RemoveOldestEntryAndUpdate();
+
+  // Find the corresponding bin.
+  int hist_index = GetBinIndex(rms);
+  // To Q10 domain.
+  int prob_q10 =
+      static_cast<int16_t>(floor(activity_probaility * kProbQDomain));
+  InsertNewestEntryAndUpdate(prob_q10, hist_index);
+}
+
+// Doing nothing if buffer is not full, yet.
+void LoudnessHistogram::RemoveOldestEntryAndUpdate() {
+  RTC_DCHECK_GT(len_circular_buffer_, 0);
+  // Do nothing if circular buffer is not full.
+  if (!buffer_is_full_)
+    return;
+
+  int oldest_prob = activity_probability_[buffer_index_];
+  int oldest_hist_index = hist_bin_index_[buffer_index_];
+  UpdateHist(-oldest_prob, oldest_hist_index);
+}
+
+void LoudnessHistogram::RemoveTransient() {
+  // Don't expect to be here if high-activity region is longer than
+  // `kTransientWidthThreshold` or there has not been any transient.
+  RTC_DCHECK_LE(len_high_activity_, kTransientWidthThreshold);
+  int index =
+      (buffer_index_ > 0) ? (buffer_index_ - 1) : len_circular_buffer_ - 1;
+  while (len_high_activity_ > 0) {
+    UpdateHist(-activity_probability_[index], hist_bin_index_[index]);
+    activity_probability_[index] = 0;
+    index = (index > 0) ? (index - 1) : (len_circular_buffer_ - 1);
+    len_high_activity_--;
+  }
+}
+
+void LoudnessHistogram::InsertNewestEntryAndUpdate(int activity_prob_q10,
+                                                   int hist_index) {
+  // Update the circular buffer if it is enabled.
+  if (len_circular_buffer_ > 0) {
+    // Removing transient.
+    if (activity_prob_q10 <= kLowProbThresholdQ10) {
+      // Lower than threshold probability, set it to zero.
+      activity_prob_q10 = 0;
+      // Check if this has been a transient.
+      if (len_high_activity_ <= kTransientWidthThreshold)
+        RemoveTransient();  // Remove this transient.
+      len_high_activity_ = 0;
+    } else if (len_high_activity_ <= kTransientWidthThreshold) {
+      len_high_activity_++;
+    }
+    // Updating the circular buffer.
+    activity_probability_[buffer_index_] = activity_prob_q10;
+    hist_bin_index_[buffer_index_] = hist_index;
+    // Increment the buffer index and check for wrap-around.
+    buffer_index_++;
+    if (buffer_index_ >= len_circular_buffer_) {
+      buffer_index_ = 0;
+      buffer_is_full_ = true;
+    }
+  }
+
+  num_updates_++;
+  if (num_updates_ < 0)
+    num_updates_--;
+
+  UpdateHist(activity_prob_q10, hist_index);
+}
+
+void LoudnessHistogram::UpdateHist(int activity_prob_q10, int hist_index) {
+  bin_count_q10_[hist_index] += activity_prob_q10;
+  audio_content_q10_ += activity_prob_q10;
+}
+
+double LoudnessHistogram::AudioContent() const {
+  return audio_content_q10_ / kProbQDomain;
+}
+
+LoudnessHistogram* LoudnessHistogram::Create() {
+  return new LoudnessHistogram;
+}
+
+LoudnessHistogram* LoudnessHistogram::Create(int window_size) {
+  if (window_size < 0)
+    return NULL;
+  return new LoudnessHistogram(window_size);
+}
+
+void LoudnessHistogram::Reset() {
+  // Reset the histogram, audio-content and number of updates.
+  memset(bin_count_q10_, 0, sizeof(bin_count_q10_));
+  audio_content_q10_ = 0;
+  num_updates_ = 0;
+  // Empty the circular buffer.
+  buffer_index_ = 0;
+  buffer_is_full_ = false;
+  len_high_activity_ = 0;
+}
+
+int LoudnessHistogram::GetBinIndex(double rms) {
+  // First exclude overload cases.
+  if (rms <= kHistBinCenters[0]) {
+    return 0;
+  } else if (rms >= kHistBinCenters[kHistSize - 1]) {
+    return kHistSize - 1;
+  } else {
+    // The quantizer is uniform in log domain. Alternatively we could do binary
+    // search in linear domain.
+    double rms_log = log(rms);
+
+    int index = static_cast<int>(
+        floor((rms_log - kLogDomainMinBinCenter) * kLogDomainStepSizeInverse));
+    // The final decision is in linear domain.
+    double b = 0.5 * (kHistBinCenters[index] + kHistBinCenters[index + 1]);
+    if (rms > b) {
+      return index + 1;
+    }
+    return index;
+  }
+}
+
+double LoudnessHistogram::CurrentRms() const {
+  double p;
+  double mean_val = 0;
+  if (audio_content_q10_ > 0) {
+    double p_total_inverse = 1. / static_cast<double>(audio_content_q10_);
+    for (int n = 0; n < kHistSize; n++) {
+      p = static_cast<double>(bin_count_q10_[n]) * p_total_inverse;
+      mean_val += p * kHistBinCenters[n];
+    }
+  } else {
+    mean_val = kHistBinCenters[0];
+  }
+  return mean_val;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram.h b/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram.h
new file mode 100644
index 0000000000..51b38714c2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram.h
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
+#define MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
+
+#include <stdint.h>
+
+#include <memory>
+
+namespace webrtc {
+
+// This class implements the histogram of loudness with circular buffers so that
+// the histogram tracks the last T seconds of the loudness.
+class LoudnessHistogram {
+ public:
+  // Create a non-sliding LoudnessHistogram.
+  static LoudnessHistogram* Create();
+
+  // Create a sliding LoudnessHistogram, i.e. the histogram represents the last
+  // `window_size` samples.
+  static LoudnessHistogram* Create(int window_size);
+  ~LoudnessHistogram();
+
+  // Insert RMS and the corresponding activity probability.
+  void Update(double rms, double activity_probability);
+
+  // Reset the histogram, forget the past.
+  void Reset();
+
+  // Current loudness, which is actually the mean of histogram in loudness
+  // domain.
+  double CurrentRms() const;
+
+  // Sum of the histogram content.
+  double AudioContent() const;
+
+  // Number of times the histogram has been updated.
+  int num_updates() const { return num_updates_; }
+
+ private:
+  LoudnessHistogram();
+  explicit LoudnessHistogram(int window);
+
+  // Find the histogram bin associated with the given `rms`.
+  int GetBinIndex(double rms);
+
+  void RemoveOldestEntryAndUpdate();
+  void InsertNewestEntryAndUpdate(int activity_prob_q10, int hist_index);
+  void UpdateHist(int activity_prob_q10, int hist_index);
+  void RemoveTransient();
+
+  // Number of histogram bins.
+  static const int kHistSize = 77;
+
+  // Number of times the histogram is updated
+  int num_updates_;
+  // Audio content, this should be equal to the sum of the components of
+  // `bin_count_q10_`.
+  int64_t audio_content_q10_;
+
+  // LoudnessHistogram of input RMS in Q10 with `kHistSize_` bins. In each
+  // 'Update(),' we increment the associated histogram-bin with the given
+  // probability. The increment is implemented in Q10 to avoid rounding errors.
+  int64_t bin_count_q10_[kHistSize];
+
+  // Circular buffer for probabilities
+  std::unique_ptr<int[]> activity_probability_;
+  // Circular buffer for histogram-indices of probabilities.
+  std::unique_ptr<int[]> hist_bin_index_;
+  // Current index of circular buffer, where the newest data will be written to,
+  // therefore, pointing to the oldest data if buffer is full.
+  int buffer_index_;
+  // Indicating if buffer is full and we had a wrap around.
+  int buffer_is_full_;
+  // Size of circular buffer.
+  int len_circular_buffer_;
+  int len_high_activity_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram_unittest.cc
new file mode 100644
index 0000000000..bbc0a7ee92
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/loudness_histogram_unittest.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Use CreateHistUnittestFile.m to generate the input file.
+
+#include "modules/audio_processing/agc/loudness_histogram.h"
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <cmath>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/agc/utility.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+struct InputOutput {
+  double rms;
+  double activity_probability;
+  double audio_content;
+  double loudness;
+};
+
+const double kRelativeErrTol = 1e-10;
+
+class LoudnessHistogramTest : public ::testing::Test {
+ protected:
+  void RunTest(bool enable_circular_buff, absl::string_view filename);
+
+ private:
+  void TestClean();
+  std::unique_ptr<LoudnessHistogram> hist_;
+};
+
+void LoudnessHistogramTest::TestClean() {
+  EXPECT_EQ(hist_->CurrentRms(), 7.59621091765857e-02);
+  EXPECT_EQ(hist_->AudioContent(), 0);
+  EXPECT_EQ(hist_->num_updates(), 0);
+}
+
+void LoudnessHistogramTest::RunTest(bool enable_circular_buff,
+                                    absl::string_view filename) {
+  FILE* in_file = fopen(std::string(filename).c_str(), "rb");
+  ASSERT_TRUE(in_file != NULL);
+  if (enable_circular_buff) {
+    int buffer_size;
+    EXPECT_EQ(fread(&buffer_size, sizeof(buffer_size), 1, in_file), 1u);
+    hist_.reset(LoudnessHistogram::Create(buffer_size));
+  } else {
+    hist_.reset(LoudnessHistogram::Create());
+  }
+  TestClean();
+
+  InputOutput io;
+  int num_updates = 0;
+  while (fread(&io, sizeof(InputOutput), 1, in_file) == 1) {
+    if (io.rms < 0) {
+      // We have to reset.
+      hist_->Reset();
+      TestClean();
+      num_updates = 0;
+      // Read the next chunk of input.
+      if (fread(&io, sizeof(InputOutput), 1, in_file) != 1)
+        break;
+    }
+    hist_->Update(io.rms, io.activity_probability);
+    num_updates++;
+    EXPECT_EQ(hist_->num_updates(), num_updates);
+    double audio_content = hist_->AudioContent();
+
+    double abs_err =
+        std::min(audio_content, io.audio_content) * kRelativeErrTol;
+
+    ASSERT_NEAR(audio_content, io.audio_content, abs_err);
+    double current_loudness = Linear2Loudness(hist_->CurrentRms());
+    abs_err =
+        std::min(fabs(current_loudness), fabs(io.loudness)) * kRelativeErrTol;
+    ASSERT_NEAR(current_loudness, io.loudness, abs_err);
+  }
+  fclose(in_file);
+}
+
+TEST_F(LoudnessHistogramTest, ActiveCircularBuffer) {
+  RunTest(true, test::ResourcePath(
+                    "audio_processing/agc/agc_with_circular_buffer", "dat")
+                    .c_str());
+}
+
+TEST_F(LoudnessHistogramTest, InactiveCircularBuffer) {
+  RunTest(false, test::ResourcePath(
+                     "audio_processing/agc/agc_no_circular_buffer", "dat")
+                     .c_str());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/mock_agc.h b/third_party/libwebrtc/modules/audio_processing/agc/mock_agc.h
new file mode 100644
index 0000000000..3080e1563c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/mock_agc.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_MOCK_AGC_H_
+#define MODULES_AUDIO_PROCESSING_AGC_MOCK_AGC_H_
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc/agc.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAgc : public Agc {
+ public:
+  virtual ~MockAgc() {}
+  MOCK_METHOD(void, Process, (rtc::ArrayView<const int16_t> audio), (override));
+  MOCK_METHOD(bool, GetRmsErrorDb, (int* error), (override));
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(int, set_target_level_dbfs, (int level), (override));
+  MOCK_METHOD(int, target_level_dbfs, (), (const, override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_MOCK_AGC_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/utility.cc b/third_party/libwebrtc/modules/audio_processing/agc/utility.cc
new file mode 100644
index 0000000000..2a87e5ce74
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/utility.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc/utility.h"
+
+#include <math.h>
+
+namespace webrtc {
+
+static const double kLog10 = 2.30258509299;
+static const double kLinear2DbScale = 20.0 / kLog10;
+static const double kLinear2LoudnessScale = 13.4 / kLog10;
+
+double Loudness2Db(double loudness) {
+  return loudness * kLinear2DbScale / kLinear2LoudnessScale;
+}
+
+double Linear2Loudness(double rms) {
+  if (rms == 0)
+    return -15;
+  return kLinear2LoudnessScale * log(rms);
+}
+
+double Db2Loudness(double db) {
+  return db * kLinear2LoudnessScale / kLinear2DbScale;
+}
+
+double Dbfs2Loudness(double dbfs) {
+  return Db2Loudness(90 + dbfs);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc/utility.h b/third_party/libwebrtc/modules/audio_processing/agc/utility.h
new file mode 100644
index 0000000000..56eec244a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc/utility.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_
+#define MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_
+
+namespace webrtc {
+
+// TODO(turajs): Add description of function.
+double Loudness2Db(double loudness);
+
+double Linear2Loudness(double rms);
+
+double Db2Loudness(double db);
+
+double Dbfs2Loudness(double dbfs);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_UTILITY_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/agc2/BUILD.gn
new file mode 100644
index 0000000000..bd59ad3dae
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/BUILD.gn
@@ -0,0 +1,511 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("speech_level_estimator") {
+  sources = [
+    "speech_level_estimator.cc",
+    "speech_level_estimator.h",
+  ]
+
+  visibility = [
+    "..:gain_controller2",
+    "./*",
+  ]
+
+  configs += [ "..:apm_debug_dump" ]
+
+  deps = [
+    ":common",
+    "..:api",
+    "..:apm_logging",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:safe_minmax",
+  ]
+}
+
+rtc_library("adaptive_digital_gain_controller") {
+  sources = [
+    "adaptive_digital_gain_controller.cc",
+    "adaptive_digital_gain_controller.h",
+  ]
+
+  visibility = [
+    "..:gain_controller2",
+    "./*",
+  ]
+
+  configs += [ "..:apm_debug_dump" ]
+
+  deps = [
+    ":common",
+    ":gain_applier",
+    "..:api",
+    "..:apm_logging",
+    "..:audio_frame_view",
+    "../../../common_audio",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:safe_minmax",
+    "../../../system_wrappers:metrics",
+  ]
+}
+
+rtc_library("saturation_protector") {
+  sources = [
+    "saturation_protector.cc",
+    "saturation_protector.h",
+    "saturation_protector_buffer.cc",
+    "saturation_protector_buffer.h",
+  ]
+
+  visibility = [
+    "..:gain_controller2",
+    "./*",
+  ]
+
+  configs += [ "..:apm_debug_dump" ]
+
+  deps = [
+    ":common",
+    "..:apm_logging",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:safe_compare",
+    "../../../rtc_base:safe_minmax",
+  ]
+
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("biquad_filter") {
+  visibility = [ "./*" ]
+  sources = [
+    "biquad_filter.cc",
+    "biquad_filter.h",
+  ]
+  deps = [
+    "../../../api:array_view",
+    "../../../rtc_base:macromagic",
+  ]
+}
+
+rtc_library("clipping_predictor") {
+  visibility = [
+    "../agc:agc",
+    "./*",
+  ]
+
+  sources = [
+    "clipping_predictor.cc",
+    "clipping_predictor.h",
+    "clipping_predictor_level_buffer.cc",
+    "clipping_predictor_level_buffer.h",
+  ]
+
+  deps = [
+    ":gain_map",
+    "..:api",
+    "..:audio_frame_view",
+    "../../../common_audio",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:safe_minmax",
+  ]
+
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_source_set("common") {
+  sources = [ "agc2_common.h" ]
+}
+
+rtc_library("fixed_digital") {
+  sources = [
+    "fixed_digital_level_estimator.cc",
+    "fixed_digital_level_estimator.h",
+    "interpolated_gain_curve.cc",
+    "interpolated_gain_curve.h",
+    "limiter.cc",
+    "limiter.h",
+  ]
+
+  visibility = [
+    "..:gain_controller2",
+    "../../audio_mixer:audio_mixer_impl",
+    "./*",
+  ]
+
+  configs += [ "..:apm_debug_dump" ]
+
+  deps = [
+    ":common",
+    "..:apm_logging",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../common_audio",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:gtest_prod",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../system_wrappers:metrics",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_library("gain_applier") {
+  sources = [
+    "gain_applier.cc",
+    "gain_applier.h",
+  ]
+
+  visibility = [
+    "..:gain_controller2",
+    "./*",
+  ]
+
+  deps = [
+    ":common",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../rtc_base:safe_minmax",
+  ]
+}
+
+rtc_source_set("gain_map") {
+  visibility = [
+    "..:analog_mic_simulation",
+    "../agc:agc",
+    "./*",
+  ]
+
+  sources = [ "gain_map_internal.h" ]
+}
+
+rtc_library("input_volume_controller") {
+  sources = [
+    "input_volume_controller.cc",
+    "input_volume_controller.h",
+    "speech_probability_buffer.cc",
+    "speech_probability_buffer.h",
+  ]
+
+  visibility = [
+    "..:gain_controller2",
+    "./*",
+  ]
+
+  configs += [ "..:apm_debug_dump" ]
+
+  deps = [
+    ":clipping_predictor",
+    ":gain_map",
+    ":input_volume_stats_reporter",
+    "..:api",
+    "..:audio_buffer",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:gtest_prod",
+    "../../../rtc_base:gtest_prod",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:safe_minmax",
+    "../../../system_wrappers:field_trial",
+    "../../../system_wrappers:metrics",
+  ]
+
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("noise_level_estimator") {
+  sources = [
+    "noise_level_estimator.cc",
+    "noise_level_estimator.h",
+  ]
+  deps = [
+    ":biquad_filter",
+    "..:apm_logging",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../system_wrappers",
+  ]
+
+  visibility = [
+    "..:gain_controller2",
+    "./*",
+  ]
+
+  configs += [ "..:apm_debug_dump" ]
+}
+
+rtc_library("vad_wrapper") {
+  sources = [
+    "vad_wrapper.cc",
+    "vad_wrapper.h",
+  ]
+
+  visibility = [
+    "..:gain_controller2",
+    "./*",
+  ]
+
+  defines = []
+  if (rtc_build_with_neon && current_cpu != "arm64") {
+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+    cflags = [ "-mfpu=neon" ]
+  }
+
+  deps = [
+    ":common",
+    ":cpu_features",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../common_audio",
+    "../../../rtc_base:checks",
+    "rnn_vad",
+    "rnn_vad:rnn_vad_common",
+  ]
+}
+
+rtc_library("cpu_features") {
+  sources = [
+    "cpu_features.cc",
+    "cpu_features.h",
+  ]
+
+  visibility = [
+    "..:gain_controller2",
+    "./*",
+  ]
+
+  deps = [
+    "../../../rtc_base:stringutils",
+    "../../../rtc_base/system:arch",
+    "../../../system_wrappers",
+  ]
+}
+
+rtc_library("speech_level_estimator_unittest") {
+  testonly = true
+  configs += [ "..:apm_debug_dump" ]
+
+  sources = [ "speech_level_estimator_unittest.cc" ]
+  deps = [
+    ":common",
+    ":speech_level_estimator",
+    "..:api",
+    "..:apm_logging",
+    "../../../rtc_base:gunit_helpers",
+    "../../../test:test_support",
+  ]
+}
+
+rtc_library("adaptive_digital_gain_controller_unittest") {
+  testonly = true
+  configs += [ "..:apm_debug_dump" ]
+
+  sources = [ "adaptive_digital_gain_controller_unittest.cc" ]
+
+  deps = [
+    ":adaptive_digital_gain_controller",
+    ":common",
+    ":test_utils",
+    "..:api",
+    "..:apm_logging",
+    "..:audio_frame_view",
+    "../../../common_audio",
+    "../../../rtc_base:gunit_helpers",
+    "../../../test:test_support",
+  ]
+}
+
+rtc_library("gain_applier_unittest") {
+  testonly = true
+  configs += [ "..:apm_debug_dump" ]
+
+  sources = [ "gain_applier_unittest.cc" ]
+  deps = [
+    ":gain_applier",
+    ":test_utils",
+    "..:audio_frame_view",
+    "../../../rtc_base:gunit_helpers",
+    "../../../test:test_support",
+  ]
+}
+
+rtc_library("saturation_protector_unittest") {
+  testonly = true
+  configs += [ "..:apm_debug_dump" ]
+
+  sources = [
+    "saturation_protector_buffer_unittest.cc",
+    "saturation_protector_unittest.cc",
+  ]
+  deps = [
+    ":common",
+    ":saturation_protector",
+    "..:apm_logging",
+    "../../../rtc_base:gunit_helpers",
+    "../../../test:test_support",
+  ]
+}
+
+rtc_library("biquad_filter_unittests") {
+  testonly = true
+  sources = [ "biquad_filter_unittest.cc" ]
+  deps = [
+    ":biquad_filter",
+    "../../../rtc_base:gunit_helpers",
+  ]
+}
+
+rtc_library("fixed_digital_unittests") {
+  testonly = true
+  configs += [ "..:apm_debug_dump" ]
+
+  sources = [
+    "agc2_testing_common_unittest.cc",
+    "compute_interpolated_gain_curve.cc",
+    "compute_interpolated_gain_curve.h",
+    "fixed_digital_level_estimator_unittest.cc",
+    "interpolated_gain_curve_unittest.cc",
+    "limiter_db_gain_curve.cc",
+    "limiter_db_gain_curve.h",
+    "limiter_db_gain_curve_unittest.cc",
+    "limiter_unittest.cc",
+  ]
+  deps = [
+    ":common",
+    ":fixed_digital",
+    ":test_utils",
+    "..:apm_logging",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../common_audio",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:gunit_helpers",
+    "../../../system_wrappers:metrics",
+  ]
+}
+
+rtc_library("input_volume_controller_unittests") {
+  testonly = true
+  sources = [
+    "clipping_predictor_level_buffer_unittest.cc",
+    "clipping_predictor_unittest.cc",
+    "input_volume_controller_unittest.cc",
+    "speech_probability_buffer_unittest.cc",
+  ]
+
+  configs += [ "..:apm_debug_dump" ]
+
+  deps = [
+    ":clipping_predictor",
+    ":gain_map",
+    ":input_volume_controller",
+    "..:api",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:random",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../system_wrappers:metrics",
+    "../../../test:field_trial",
+    "../../../test:fileutils",
+    "../../../test:test_support",
+    "//testing/gtest",
+  ]
+
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("noise_estimator_unittests") {
+  testonly = true
+  configs += [ "..:apm_debug_dump" ]
+
+  sources = [ "noise_level_estimator_unittest.cc" ]
+  deps = [
+    ":noise_level_estimator",
+    ":test_utils",
+    "..:apm_logging",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../api:function_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:gunit_helpers",
+  ]
+}
+
+rtc_library("vad_wrapper_unittests") {
+  testonly = true
+  sources = [ "vad_wrapper_unittest.cc" ]
+  deps = [
+    ":common",
+    ":vad_wrapper",
+    "..:audio_frame_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:gunit_helpers",
+    "../../../rtc_base:safe_compare",
+    "../../../test:test_support",
+  ]
+}
+
+rtc_library("test_utils") {
+  testonly = true
+  visibility = [
+    ":*",
+    "..:audio_processing_unittests",
+  ]
+  sources = [
+    "agc2_testing_common.cc",
+    "agc2_testing_common.h",
+    "vector_float_frame.cc",
+    "vector_float_frame.h",
+  ]
+  deps = [
+    "..:audio_frame_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:random",
+  ]
+}
+
+rtc_library("input_volume_stats_reporter") {
+  sources = [
+    "input_volume_stats_reporter.cc",
+    "input_volume_stats_reporter.h",
+  ]
+  deps = [
+    "../../../rtc_base:gtest_prod",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../system_wrappers:metrics",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("input_volume_stats_reporter_unittests") {
+  testonly = true
+  sources = [ "input_volume_stats_reporter_unittest.cc" ]
+  deps = [
+    ":input_volume_stats_reporter",
+    "../../../rtc_base:stringutils",
+    "../../../system_wrappers:metrics",
+    "../../../test:test_support",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
new file mode 100644
index 0000000000..e8edab602c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
@@ -0,0 +1,216 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
+
+#include <algorithm>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+
+using AdaptiveDigitalConfig =
+    AudioProcessing::Config::GainController2::AdaptiveDigital;
+
+constexpr int kHeadroomHistogramMin = 0;
+constexpr int kHeadroomHistogramMax = 50;
+constexpr int kGainDbHistogramMax = 30;
+
+// Computes the gain for `input_level_dbfs` to reach `-config.headroom_db`.
+// Clamps the gain in [0, `config.max_gain_db`]. `config.headroom_db` is a
+// safety margin to allow transient peaks to exceed the target peak level
+// without clipping.
+float ComputeGainDb(float input_level_dbfs,
+                    const AdaptiveDigitalConfig& config) {
+  // If the level is very low, apply the maximum gain.
+  if (input_level_dbfs < -(config.headroom_db + config.max_gain_db)) {
+    return config.max_gain_db;
+  }
+  // We expect to end up here most of the time: the level is below
+  // -headroom, but we can boost it to -headroom.
+  if (input_level_dbfs < -config.headroom_db) {
+    return -config.headroom_db - input_level_dbfs;
+  }
+  // The level is too high and we can't boost.
+  RTC_DCHECK_GE(input_level_dbfs, -config.headroom_db);
+  return 0.0f;
+}
+
+// Returns `target_gain_db` if applying such a gain to `input_noise_level_dbfs`
+// does not exceed `max_output_noise_level_dbfs`. Otherwise lowers and returns
+// `target_gain_db` so that the output noise level equals
+// `max_output_noise_level_dbfs`.
+float LimitGainByNoise(float target_gain_db,
+                       float input_noise_level_dbfs,
+                       float max_output_noise_level_dbfs,
+                       ApmDataDumper& apm_data_dumper) {
+  const float max_allowed_gain_db =
+      max_output_noise_level_dbfs - input_noise_level_dbfs;
+  apm_data_dumper.DumpRaw("agc2_adaptive_gain_applier_max_allowed_gain_db",
+                          max_allowed_gain_db);
+  return std::min(target_gain_db, std::max(max_allowed_gain_db, 0.0f));
+}
+
+float LimitGainByLowConfidence(float target_gain_db,
+                               float last_gain_db,
+                               float limiter_audio_level_dbfs,
+                               bool estimate_is_confident) {
+  if (estimate_is_confident ||
+      limiter_audio_level_dbfs <= kLimiterThresholdForAgcGainDbfs) {
+    return target_gain_db;
+  }
+  const float limiter_level_dbfs_before_gain =
+      limiter_audio_level_dbfs - last_gain_db;
+
+  // Compute a new gain so that `limiter_level_dbfs_before_gain` +
+  // `new_target_gain_db` is not great than `kLimiterThresholdForAgcGainDbfs`.
+  const float new_target_gain_db = std::max(
+      kLimiterThresholdForAgcGainDbfs - limiter_level_dbfs_before_gain, 0.0f);
+  return std::min(new_target_gain_db, target_gain_db);
+}
+
+// Computes how the gain should change during this frame.
+// Return the gain difference in db to 'last_gain_db'.
+float ComputeGainChangeThisFrameDb(float target_gain_db,
+                                   float last_gain_db,
+                                   bool gain_increase_allowed,
+                                   float max_gain_decrease_db,
+                                   float max_gain_increase_db) {
+  RTC_DCHECK_GT(max_gain_decrease_db, 0);
+  RTC_DCHECK_GT(max_gain_increase_db, 0);
+  float target_gain_difference_db = target_gain_db - last_gain_db;
+  if (!gain_increase_allowed) {
+    target_gain_difference_db = std::min(target_gain_difference_db, 0.0f);
+  }
+  return rtc::SafeClamp(target_gain_difference_db, -max_gain_decrease_db,
+                        max_gain_increase_db);
+}
+
+}  // namespace
+
+AdaptiveDigitalGainController::AdaptiveDigitalGainController(
+    ApmDataDumper* apm_data_dumper,
+    const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+    int adjacent_speech_frames_threshold)
+    : apm_data_dumper_(apm_data_dumper),
+      gain_applier_(
+          /*hard_clip_samples=*/false,
+          /*initial_gain_factor=*/DbToRatio(config.initial_gain_db)),
+      config_(config),
+      adjacent_speech_frames_threshold_(adjacent_speech_frames_threshold),
+      max_gain_change_db_per_10ms_(config_.max_gain_change_db_per_second *
+                                   kFrameDurationMs / 1000.0f),
+      calls_since_last_gain_log_(0),
+      frames_to_gain_increase_allowed_(adjacent_speech_frames_threshold),
+      last_gain_db_(config_.initial_gain_db) {
+  RTC_DCHECK_GT(max_gain_change_db_per_10ms_, 0.0f);
+  RTC_DCHECK_GE(frames_to_gain_increase_allowed_, 1);
+  RTC_DCHECK_GE(config_.max_output_noise_level_dbfs, -90.0f);
+  RTC_DCHECK_LE(config_.max_output_noise_level_dbfs, 0.0f);
+}
+
+void AdaptiveDigitalGainController::Process(const FrameInfo& info,
+                                            AudioFrameView<float> frame) {
+  RTC_DCHECK_GE(info.speech_level_dbfs, -150.0f);
+  RTC_DCHECK_GE(frame.num_channels(), 1);
+  RTC_DCHECK(
+      frame.samples_per_channel() == 80 || frame.samples_per_channel() == 160 ||
+      frame.samples_per_channel() == 320 || frame.samples_per_channel() == 480)
+      << "`frame` does not look like a 10 ms frame for an APM supported sample "
+         "rate";
+
+  // Compute the input level used to select the desired gain.
+  RTC_DCHECK_GT(info.headroom_db, 0.0f);
+  const float input_level_dbfs = info.speech_level_dbfs + info.headroom_db;
+
+  const float target_gain_db = LimitGainByLowConfidence(
+      LimitGainByNoise(ComputeGainDb(input_level_dbfs, config_),
+                       info.noise_rms_dbfs, config_.max_output_noise_level_dbfs,
+                       *apm_data_dumper_),
+      last_gain_db_, info.limiter_envelope_dbfs, info.speech_level_reliable);
+
+  // Forbid increasing the gain until enough adjacent speech frames are
+  // observed.
+  bool first_confident_speech_frame = false;
+  if (info.speech_probability < kVadConfidenceThreshold) {
+    frames_to_gain_increase_allowed_ = adjacent_speech_frames_threshold_;
+  } else if (frames_to_gain_increase_allowed_ > 0) {
+    frames_to_gain_increase_allowed_--;
+    first_confident_speech_frame = frames_to_gain_increase_allowed_ == 0;
+  }
+  apm_data_dumper_->DumpRaw(
+      "agc2_adaptive_gain_applier_frames_to_gain_increase_allowed",
+      frames_to_gain_increase_allowed_);
+
+  const bool gain_increase_allowed = frames_to_gain_increase_allowed_ == 0;
+
+  float max_gain_increase_db = max_gain_change_db_per_10ms_;
+  if (first_confident_speech_frame) {
+    // No gain increase happened while waiting for a long enough speech
+    // sequence. Therefore, temporarily allow a faster gain increase.
+    RTC_DCHECK(gain_increase_allowed);
+    max_gain_increase_db *= adjacent_speech_frames_threshold_;
+  }
+
+  const float gain_change_this_frame_db = ComputeGainChangeThisFrameDb(
+      target_gain_db, last_gain_db_, gain_increase_allowed,
+      /*max_gain_decrease_db=*/max_gain_change_db_per_10ms_,
+      max_gain_increase_db);
+
+  apm_data_dumper_->DumpRaw("agc2_adaptive_gain_applier_want_to_change_by_db",
+                            target_gain_db - last_gain_db_);
+  apm_data_dumper_->DumpRaw("agc2_adaptive_gain_applier_will_change_by_db",
+                            gain_change_this_frame_db);
+
+  // Optimization: avoid calling math functions if gain does not
+  // change.
+  if (gain_change_this_frame_db != 0.f) {
+    gain_applier_.SetGainFactor(
+        DbToRatio(last_gain_db_ + gain_change_this_frame_db));
+  }
+
+  gain_applier_.ApplyGain(frame);
+
+  // Remember that the gain has changed for the next iteration.
+  last_gain_db_ = last_gain_db_ + gain_change_this_frame_db;
+  apm_data_dumper_->DumpRaw("agc2_adaptive_gain_applier_applied_gain_db",
+                            last_gain_db_);
+
+  // Log every 10 seconds.
+  calls_since_last_gain_log_++;
+  if (calls_since_last_gain_log_ == 1000) {
+    calls_since_last_gain_log_ = 0;
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedSpeechLevel",
+                                -info.speech_level_dbfs, 0, 100, 101);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedNoiseLevel",
+                                -info.noise_rms_dbfs, 0, 100, 101);
+    RTC_HISTOGRAM_COUNTS_LINEAR(
+        "WebRTC.Audio.Agc2.Headroom", info.headroom_db, kHeadroomHistogramMin,
+        kHeadroomHistogramMax,
+        kHeadroomHistogramMax - kHeadroomHistogramMin + 1);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.DigitalGainApplied",
+                                last_gain_db_, 0, kGainDbHistogramMax,
+                                kGainDbHistogramMax + 1);
+    RTC_LOG(LS_INFO) << "AGC2 adaptive digital"
+                     << " | speech_dbfs: " << info.speech_level_dbfs
+                     << " | noise_dbfs: " << info.noise_rms_dbfs
+                     << " | headroom_db: " << info.headroom_db
+                     << " | gain_db: " << last_gain_db_;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller.h b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller.h
new file mode 100644
index 0000000000..01335e79db
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
+
+#include <vector>
+
+#include "modules/audio_processing/agc2/gain_applier.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+// Selects the target digital gain, decides when and how quickly to adapt to the
+// target and applies the current gain to 10 ms frames.
+class AdaptiveDigitalGainController {
+ public:
+  // Information about a frame to process.
+  struct FrameInfo {
+    float speech_probability;    // Probability of speech in the [0, 1] range.
+    float speech_level_dbfs;     // Estimated speech level (dBFS).
+    bool speech_level_reliable;  // True with reliable speech level estimation.
+    float noise_rms_dbfs;        // Estimated noise RMS level (dBFS).
+    float headroom_db;           // Headroom (dB).
+    // TODO(bugs.webrtc.org/7494): Remove `limiter_envelope_dbfs`.
+    float limiter_envelope_dbfs;  // Envelope level from the limiter (dBFS).
+  };
+
+  AdaptiveDigitalGainController(
+      ApmDataDumper* apm_data_dumper,
+      const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+      int adjacent_speech_frames_threshold);
+  AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete;
+  AdaptiveDigitalGainController& operator=(
+      const AdaptiveDigitalGainController&) = delete;
+
+  // Analyzes `info`, updates the digital gain and applies it to a 10 ms
+  // `frame`. Supports any sample rate supported by APM.
+  void Process(const FrameInfo& info, AudioFrameView<float> frame);
+
+ private:
+  ApmDataDumper* const apm_data_dumper_;
+  GainApplier gain_applier_;
+
+  const AudioProcessing::Config::GainController2::AdaptiveDigital config_;
+  const int adjacent_speech_frames_threshold_;
+  const float max_gain_change_db_per_10ms_;
+
+  int calls_since_last_gain_log_;
+  int frames_to_gain_increase_allowed_;
+  float last_gain_db_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_gn/moz.build
new file mode 100644
index 0000000000..7d16c9a9f5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("adaptive_digital_gain_controller_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc
new file mode 100644
index 0000000000..e95cbb5067
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc
@@ -0,0 +1,312 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/agc2/vector_float_frame.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kMono = 1;
+constexpr int kStereo = 2;
+constexpr int kFrameLen10ms8kHz = 80;
+constexpr int kFrameLen10ms48kHz = 480;
+
+constexpr float kMaxSpeechProbability = 1.0f;
+
+// Constants used in place of estimated noise levels.
+constexpr float kNoNoiseDbfs = kMinLevelDbfs;
+constexpr float kWithNoiseDbfs = -20.0f;
+
+// Number of additional frames to process in the tests to ensure that the tested
+// adaptation processes have converged.
+constexpr int kNumExtraFrames = 10;
+
+constexpr float GetMaxGainChangePerFrameDb(
+    float max_gain_change_db_per_second) {
+  return max_gain_change_db_per_second * kFrameDurationMs / 1000.0f;
+}
+
+using AdaptiveDigitalConfig =
+    AudioProcessing::Config::GainController2::AdaptiveDigital;
+
+constexpr AdaptiveDigitalConfig kDefaultConfig{};
+
+// Helper to create initialized `AdaptiveDigitalGainController` objects.
+struct GainApplierHelper {
+  GainApplierHelper(const AdaptiveDigitalConfig& config,
+                    int adjacent_speech_frames_threshold)
+      : apm_data_dumper(0),
+        gain_applier(std::make_unique<AdaptiveDigitalGainController>(
+            &apm_data_dumper,
+            config,
+            adjacent_speech_frames_threshold)) {}
+  ApmDataDumper apm_data_dumper;
+  std::unique_ptr<AdaptiveDigitalGainController> gain_applier;
+};
+
+// Returns a `FrameInfo` sample to simulate noiseless speech detected with
+// maximum probability and with level, headroom and limiter envelope chosen
+// so that the resulting gain equals the default initial adaptive digital gain
+// i.e., no gain adaptation is expected.
+AdaptiveDigitalGainController::FrameInfo GetFrameInfoToNotAdapt(
+    const AdaptiveDigitalConfig& config) {
+  AdaptiveDigitalGainController::FrameInfo info;
+  info.speech_probability = kMaxSpeechProbability;
+  info.speech_level_dbfs = -config.initial_gain_db - config.headroom_db;
+  info.speech_level_reliable = true;
+  info.noise_rms_dbfs = kNoNoiseDbfs;
+  info.headroom_db = config.headroom_db;
+  info.limiter_envelope_dbfs = -2.0f;
+  return info;
+}
+
+TEST(GainController2AdaptiveDigitalGainControllerTest,
+     GainApplierShouldNotCrash) {
+  GainApplierHelper helper(kDefaultConfig, kAdjacentSpeechFramesThreshold);
+  // Make one call with reasonable audio level values and settings.
+  VectorFloatFrame fake_audio(kStereo, kFrameLen10ms48kHz, 10000.0f);
+  helper.gain_applier->Process(GetFrameInfoToNotAdapt(kDefaultConfig),
+                               fake_audio.float_frame_view());
+}
+
+// Checks that the maximum allowed gain is applied.
+TEST(GainController2AdaptiveDigitalGainControllerTest, MaxGainApplied) {
+  constexpr int kNumFramesToAdapt =
+      static_cast<int>(kDefaultConfig.max_gain_db /
+                       GetMaxGainChangePerFrameDb(
+                           kDefaultConfig.max_gain_change_db_per_second)) +
+      kNumExtraFrames;
+
+  GainApplierHelper helper(kDefaultConfig, kAdjacentSpeechFramesThreshold);
+  AdaptiveDigitalGainController::FrameInfo info =
+      GetFrameInfoToNotAdapt(kDefaultConfig);
+  info.speech_level_dbfs = -60.0f;
+  float applied_gain;
+  for (int i = 0; i < kNumFramesToAdapt; ++i) {
+    VectorFloatFrame fake_audio(kMono, kFrameLen10ms8kHz, 1.0f);
+    helper.gain_applier->Process(info, fake_audio.float_frame_view());
+    applied_gain = fake_audio.float_frame_view().channel(0)[0];
+  }
+  const float applied_gain_db = 20.0f * std::log10f(applied_gain);
+  EXPECT_NEAR(applied_gain_db, kDefaultConfig.max_gain_db, 0.1f);
+}
+
+TEST(GainController2AdaptiveDigitalGainControllerTest, GainDoesNotChangeFast) {
+  GainApplierHelper helper(kDefaultConfig, kAdjacentSpeechFramesThreshold);
+
+  constexpr float initial_level_dbfs = -25.0f;
+  constexpr float kMaxGainChangeDbPerFrame =
+      GetMaxGainChangePerFrameDb(kDefaultConfig.max_gain_change_db_per_second);
+  constexpr int kNumFramesToAdapt =
+      static_cast<int>(initial_level_dbfs / kMaxGainChangeDbPerFrame) +
+      kNumExtraFrames;
+
+  const float max_change_per_frame_linear = DbToRatio(kMaxGainChangeDbPerFrame);
+
+  float last_gain_linear = 1.f;
+  for (int i = 0; i < kNumFramesToAdapt; ++i) {
+    SCOPED_TRACE(i);
+    VectorFloatFrame fake_audio(kMono, kFrameLen10ms8kHz, 1.0f);
+    AdaptiveDigitalGainController::FrameInfo info =
+        GetFrameInfoToNotAdapt(kDefaultConfig);
+    info.speech_level_dbfs = initial_level_dbfs;
+    helper.gain_applier->Process(info, fake_audio.float_frame_view());
+    float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
+    EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
+              max_change_per_frame_linear);
+    last_gain_linear = current_gain_linear;
+  }
+
+  // Check that the same is true when gain decreases as well.
+  for (int i = 0; i < kNumFramesToAdapt; ++i) {
+    SCOPED_TRACE(i);
+    VectorFloatFrame fake_audio(kMono, kFrameLen10ms8kHz, 1.0f);
+    AdaptiveDigitalGainController::FrameInfo info =
+        GetFrameInfoToNotAdapt(kDefaultConfig);
+    info.speech_level_dbfs = 0.f;
+    helper.gain_applier->Process(info, fake_audio.float_frame_view());
+    float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
+    EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
+              max_change_per_frame_linear);
+    last_gain_linear = current_gain_linear;
+  }
+}
+
+TEST(GainController2AdaptiveDigitalGainControllerTest, GainIsRampedInAFrame) {
+  GainApplierHelper helper(kDefaultConfig, kAdjacentSpeechFramesThreshold);
+
+  constexpr float initial_level_dbfs = -25.0f;
+
+  VectorFloatFrame fake_audio(kMono, kFrameLen10ms48kHz, 1.0f);
+  AdaptiveDigitalGainController::FrameInfo info =
+      GetFrameInfoToNotAdapt(kDefaultConfig);
+  info.speech_level_dbfs = initial_level_dbfs;
+  helper.gain_applier->Process(info, fake_audio.float_frame_view());
+  float maximal_difference = 0.0f;
+  float current_value = 1.0f * DbToRatio(kDefaultConfig.initial_gain_db);
+  for (const auto& x : fake_audio.float_frame_view().channel(0)) {
+    const float difference = std::abs(x - current_value);
+    maximal_difference = std::max(maximal_difference, difference);
+    current_value = x;
+  }
+
+  const float max_change_per_frame_linear = DbToRatio(
+      GetMaxGainChangePerFrameDb(kDefaultConfig.max_gain_change_db_per_second));
+  const float max_change_per_sample =
+      max_change_per_frame_linear / kFrameLen10ms48kHz;
+
+  EXPECT_LE(maximal_difference, max_change_per_sample);
+}
+
+TEST(GainController2AdaptiveDigitalGainControllerTest, NoiseLimitsGain) {
+  GainApplierHelper helper(kDefaultConfig, kAdjacentSpeechFramesThreshold);
+
+  constexpr float initial_level_dbfs = -25.0f;
+  constexpr int num_initial_frames =
+      kDefaultConfig.initial_gain_db /
+      GetMaxGainChangePerFrameDb(kDefaultConfig.max_gain_change_db_per_second);
+  constexpr int num_frames = 50;
+
+  ASSERT_GT(kWithNoiseDbfs, kDefaultConfig.max_output_noise_level_dbfs)
+      << "kWithNoiseDbfs is too low";
+
+  for (int i = 0; i < num_initial_frames + num_frames; ++i) {
+    VectorFloatFrame fake_audio(kMono, kFrameLen10ms48kHz, 1.0f);
+    AdaptiveDigitalGainController::FrameInfo info =
+        GetFrameInfoToNotAdapt(kDefaultConfig);
+    info.speech_level_dbfs = initial_level_dbfs;
+    info.noise_rms_dbfs = kWithNoiseDbfs;
+    helper.gain_applier->Process(info, fake_audio.float_frame_view());
+
+    // Wait so that the adaptive gain applier has time to lower the gain.
+    if (i > num_initial_frames) {
+      const float maximal_ratio =
+          *std::max_element(fake_audio.float_frame_view().channel(0).begin(),
+                            fake_audio.float_frame_view().channel(0).end());
+
+      EXPECT_NEAR(maximal_ratio, 1.0f, 0.001f);
+    }
+  }
+}
+
+TEST(GainController2AdaptiveDigitalGainControllerTest,
+     CanHandlePositiveSpeechLevels) {
+  GainApplierHelper helper(kDefaultConfig, kAdjacentSpeechFramesThreshold);
+
+  // Make one call with positive audio level values and settings.
+  VectorFloatFrame fake_audio(kStereo, kFrameLen10ms48kHz, 10000.0f);
+  AdaptiveDigitalGainController::FrameInfo info =
+      GetFrameInfoToNotAdapt(kDefaultConfig);
+  info.speech_level_dbfs = 5.0f;
+  helper.gain_applier->Process(info, fake_audio.float_frame_view());
+}
+
+TEST(GainController2AdaptiveDigitalGainControllerTest, AudioLevelLimitsGain) {
+  GainApplierHelper helper(kDefaultConfig, kAdjacentSpeechFramesThreshold);
+
+  constexpr float initial_level_dbfs = -25.0f;
+  constexpr int num_initial_frames =
+      kDefaultConfig.initial_gain_db /
+      GetMaxGainChangePerFrameDb(kDefaultConfig.max_gain_change_db_per_second);
+  constexpr int num_frames = 50;
+
+  ASSERT_GT(kWithNoiseDbfs, kDefaultConfig.max_output_noise_level_dbfs)
+      << "kWithNoiseDbfs is too low";
+
+  for (int i = 0; i < num_initial_frames + num_frames; ++i) {
+    VectorFloatFrame fake_audio(kMono, kFrameLen10ms48kHz, 1.0f);
+    AdaptiveDigitalGainController::FrameInfo info =
+        GetFrameInfoToNotAdapt(kDefaultConfig);
+    info.speech_level_dbfs = initial_level_dbfs;
+    info.limiter_envelope_dbfs = 1.0f;
+    info.speech_level_reliable = false;
+    helper.gain_applier->Process(info, fake_audio.float_frame_view());
+
+    // Wait so that the adaptive gain applier has time to lower the gain.
+    if (i > num_initial_frames) {
+      const float maximal_ratio =
+          *std::max_element(fake_audio.float_frame_view().channel(0).begin(),
+                            fake_audio.float_frame_view().channel(0).end());
+
+      EXPECT_NEAR(maximal_ratio, 1.0f, 0.001f);
+    }
+  }
+}
+
+class AdaptiveDigitalGainControllerParametrizedTest
+    : public ::testing::TestWithParam<int> {
+ protected:
+  int adjacent_speech_frames_threshold() const { return GetParam(); }
+};
+
+TEST_P(AdaptiveDigitalGainControllerParametrizedTest,
+       DoNotIncreaseGainWithTooFewSpeechFrames) {
+  GainApplierHelper helper(kDefaultConfig, adjacent_speech_frames_threshold());
+
+  // Lower the speech level so that the target gain will be increased.
+  AdaptiveDigitalGainController::FrameInfo info =
+      GetFrameInfoToNotAdapt(kDefaultConfig);
+  info.speech_level_dbfs -= 12.0f;
+
+  float prev_gain = 0.0f;
+  for (int i = 0; i < adjacent_speech_frames_threshold(); ++i) {
+    SCOPED_TRACE(i);
+    VectorFloatFrame audio(kMono, kFrameLen10ms48kHz, 1.0f);
+    helper.gain_applier->Process(info, audio.float_frame_view());
+    const float gain = audio.float_frame_view().channel(0)[0];
+    if (i > 0) {
+      EXPECT_EQ(prev_gain, gain);  // No gain increase applied.
+    }
+    prev_gain = gain;
+  }
+}
+
+TEST_P(AdaptiveDigitalGainControllerParametrizedTest,
+       IncreaseGainWithEnoughSpeechFrames) {
+  GainApplierHelper helper(kDefaultConfig, adjacent_speech_frames_threshold());
+
+  // Lower the speech level so that the target gain will be increased.
+  AdaptiveDigitalGainController::FrameInfo info =
+      GetFrameInfoToNotAdapt(kDefaultConfig);
+  info.speech_level_dbfs -= 12.0f;
+
+  float prev_gain = 0.0f;
+  for (int i = 0; i < adjacent_speech_frames_threshold(); ++i) {
+    SCOPED_TRACE(i);
+    VectorFloatFrame audio(kMono, kFrameLen10ms48kHz, 1.0f);
+    helper.gain_applier->Process(info, audio.float_frame_view());
+    prev_gain = audio.float_frame_view().channel(0)[0];
+  }
+
+  // Process one more speech frame.
+  VectorFloatFrame audio(kMono, kFrameLen10ms48kHz, 1.0f);
+  helper.gain_applier->Process(info, audio.float_frame_view());
+
+  // An increased gain has been applied.
+  EXPECT_GT(audio.float_frame_view().channel(0)[0], prev_gain);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    GainController2,
+    AdaptiveDigitalGainControllerParametrizedTest,
+    ::testing::Values(1, 7, 31, kAdjacentSpeechFramesThreshold));
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/agc2_common.h b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_common.h
new file mode 100644
index 0000000000..4597bcd015
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_common.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
+
+namespace webrtc {
+
+constexpr float kMinFloatS16Value = -32768.0f;
+constexpr float kMaxFloatS16Value = 32767.0f;
+constexpr float kMaxAbsFloatS16Value = 32768.0f;
+
+// Minimum audio level in dBFS scale for S16 samples.
+constexpr float kMinLevelDbfs = -90.31f;
+
+constexpr int kFrameDurationMs = 10;
+constexpr int kSubFramesInFrame = 20;
+constexpr int kMaximalNumberOfSamplesPerChannel = 480;
+
+// Adaptive digital gain applier settings.
+
+// At what limiter levels should we start decreasing the adaptive digital gain.
+constexpr float kLimiterThresholdForAgcGainDbfs = -1.0f;
+
+// Number of milliseconds to wait to periodically reset the VAD.
+constexpr int kVadResetPeriodMs = 1500;
+
+// Speech probability threshold to detect speech activity.
+constexpr float kVadConfidenceThreshold = 0.95f;
+
+// Minimum number of adjacent speech frames having a sufficiently high speech
+// probability to reliably detect speech activity.
+constexpr int kAdjacentSpeechFramesThreshold = 12;
+
+// Number of milliseconds of speech frames to observe to make the estimator
+// confident.
+constexpr float kLevelEstimatorTimeToConfidenceMs = 400;
+constexpr float kLevelEstimatorLeakFactor =
+    1.0f - 1.0f / kLevelEstimatorTimeToConfidenceMs;
+
+// Saturation Protector settings.
+constexpr float kSaturationProtectorInitialHeadroomDb = 20.0f;
+constexpr int kSaturationProtectorBufferSize = 4;
+
+// Number of interpolation points for each region of the limiter.
+// These values have been tuned to limit the interpolated gain curve error given
+// the limiter parameters and allowing a maximum error of +/- 32768^-1.
+constexpr int kInterpolatedGainCurveKneePoints = 22;
+constexpr int kInterpolatedGainCurveBeyondKneePoints = 10;
+constexpr int kInterpolatedGainCurveTotalPoints =
+    kInterpolatedGainCurveKneePoints + kInterpolatedGainCurveBeyondKneePoints;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.cc b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.cc
new file mode 100644
index 0000000000..125e551b72
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+
+#include <cmath>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+std::vector<double> LinSpace(double l, double r, int num_points) {
+  RTC_CHECK_GE(num_points, 2);
+  std::vector<double> points(num_points);
+  const double step = (r - l) / (num_points - 1.0);
+  points[0] = l;
+  for (int i = 1; i < num_points - 1; i++) {
+    points[i] = static_cast<double>(l) + i * step;
+  }
+  points[num_points - 1] = r;
+  return points;
+}
+
+WhiteNoiseGenerator::WhiteNoiseGenerator(int min_amplitude, int max_amplitude)
+    : rand_gen_(42),
+      min_amplitude_(min_amplitude),
+      max_amplitude_(max_amplitude) {
+  RTC_DCHECK_LT(min_amplitude_, max_amplitude_);
+  RTC_DCHECK_LE(kMinS16, min_amplitude_);
+  RTC_DCHECK_LE(min_amplitude_, kMaxS16);
+  RTC_DCHECK_LE(kMinS16, max_amplitude_);
+  RTC_DCHECK_LE(max_amplitude_, kMaxS16);
+}
+
+float WhiteNoiseGenerator::operator()() {
+  return static_cast<float>(rand_gen_.Rand(min_amplitude_, max_amplitude_));
+}
+
+SineGenerator::SineGenerator(float amplitude,
+                             float frequency_hz,
+                             int sample_rate_hz)
+    : amplitude_(amplitude),
+      frequency_hz_(frequency_hz),
+      sample_rate_hz_(sample_rate_hz),
+      x_radians_(0.0f) {
+  RTC_DCHECK_GT(amplitude_, 0);
+  RTC_DCHECK_LE(amplitude_, kMaxS16);
+}
+
+float SineGenerator::operator()() {
+  constexpr float kPi = 3.1415926536f;
+  x_radians_ += frequency_hz_ / sample_rate_hz_ * 2 * kPi;
+  if (x_radians_ >= 2 * kPi) {
+    x_radians_ -= 2 * kPi;
+  }
+  return amplitude_ * std::sinf(x_radians_);
+}
+
+PulseGenerator::PulseGenerator(float pulse_amplitude,
+                               float no_pulse_amplitude,
+                               float frequency_hz,
+                               int sample_rate_hz)
+    : pulse_amplitude_(pulse_amplitude),
+      no_pulse_amplitude_(no_pulse_amplitude),
+      samples_period_(
+          static_cast<int>(static_cast<float>(sample_rate_hz) / frequency_hz)),
+      sample_counter_(0) {
+  RTC_DCHECK_GE(pulse_amplitude_, kMinS16);
+  RTC_DCHECK_LE(pulse_amplitude_, kMaxS16);
+  RTC_DCHECK_GT(no_pulse_amplitude_, kMinS16);
+  RTC_DCHECK_LE(no_pulse_amplitude_, kMaxS16);
+  RTC_DCHECK_GT(sample_rate_hz, frequency_hz);
+}
+
+float PulseGenerator::operator()() {
+  sample_counter_++;
+  if (sample_counter_ >= samples_period_) {
+    sample_counter_ -= samples_period_;
+  }
+  return static_cast<float>(sample_counter_ == 0 ? pulse_amplitude_
+                                                 : no_pulse_amplitude_);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.h b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.h
new file mode 100644
index 0000000000..afed97e83b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
+
+#include <limits>
+#include <vector>
+
+#include "rtc_base/random.h"
+
+namespace webrtc {
+namespace test {
+
+constexpr float kMinS16 =
+    static_cast<float>(std::numeric_limits<int16_t>::min());
+constexpr float kMaxS16 =
+    static_cast<float>(std::numeric_limits<int16_t>::max());
+
+// Level Estimator test parameters.
+constexpr float kDecayMs = 20.0f;
+
+// Limiter parameters.
+constexpr float kLimiterMaxInputLevelDbFs = 1.f;
+constexpr float kLimiterKneeSmoothnessDb = 1.f;
+constexpr float kLimiterCompressionRatio = 5.f;
+
+// Returns evenly spaced `num_points` numbers over a specified interval [l, r].
+std::vector<double> LinSpace(double l, double r, int num_points);
+
+// Generates white noise.
+class WhiteNoiseGenerator {
+ public:
+  WhiteNoiseGenerator(int min_amplitude, int max_amplitude);
+  float operator()();
+
+ private:
+  Random rand_gen_;
+  const int min_amplitude_;
+  const int max_amplitude_;
+};
+
+// Generates a sine function.
+class SineGenerator {
+ public:
+  SineGenerator(float amplitude, float frequency_hz, int sample_rate_hz);
+  float operator()();
+
+ private:
+  const float amplitude_;
+  const float frequency_hz_;
+  const int sample_rate_hz_;
+  float x_radians_;
+};
+
+// Generates periodic pulses.
+class PulseGenerator {
+ public:
+  PulseGenerator(float pulse_amplitude,
+                 float no_pulse_amplitude,
+                 float frequency_hz,
+                 int sample_rate_hz);
+  float operator()();
+
+ private:
+  const float pulse_amplitude_;
+  const float no_pulse_amplitude_;
+  const int samples_period_;
+  int sample_counter_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common_unittest.cc
new file mode 100644
index 0000000000..79c3cc95d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/agc2_testing_common_unittest.cc
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+
+TEST(GainController2TestingCommon, LinSpace) {
+  std::vector<double> points1 = test::LinSpace(-1.0, 2.0, 4);
+  const std::vector<double> expected_points1{{-1.0, 0.0, 1.0, 2.0}};
+  EXPECT_EQ(expected_points1, points1);
+
+  std::vector<double> points2 = test::LinSpace(0.0, 1.0, 4);
+  const std::vector<double> expected_points2{{0.0, 1.0 / 3.0, 2.0 / 3.0, 1.0}};
+  EXPECT_EQ(points2, expected_points2);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter.cc b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter.cc
new file mode 100644
index 0000000000..c1b80d7320
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/biquad_filter.h"
+
+#include "rtc_base/arraysize.h"
+
+namespace webrtc {
+
+BiQuadFilter::BiQuadFilter(const Config& config)
+    : config_(config), state_({}) {}
+
+BiQuadFilter::~BiQuadFilter() = default;
+
+void BiQuadFilter::SetConfig(const Config& config) {
+  config_ = config;
+  state_ = {};
+}
+
+void BiQuadFilter::Reset() {
+  state_ = {};
+}
+
+void BiQuadFilter::Process(rtc::ArrayView<const float> x,
+                           rtc::ArrayView<float> y) {
+  RTC_DCHECK_EQ(x.size(), y.size());
+  const float config_a0 = config_.a[0];
+  const float config_a1 = config_.a[1];
+  const float config_b0 = config_.b[0];
+  const float config_b1 = config_.b[1];
+  const float config_b2 = config_.b[2];
+  float state_a0 = state_.a[0];
+  float state_a1 = state_.a[1];
+  float state_b0 = state_.b[0];
+  float state_b1 = state_.b[1];
+  for (size_t k = 0, x_size = x.size(); k < x_size; ++k) {
+    // Use a temporary variable for `x[k]` to allow in-place processing.
+    const float tmp = x[k];
+    float y_k = config_b0 * tmp + config_b1 * state_b0 + config_b2 * state_b1 -
+                config_a0 * state_a0 - config_a1 * state_a1;
+    state_b1 = state_b0;
+    state_b0 = tmp;
+    state_a1 = state_a0;
+    state_a0 = y_k;
+    y[k] = y_k;
+  }
+  state_.a[0] = state_a0;
+  state_.a[1] = state_a1;
+  state_.b[0] = state_b0;
+  state_.b[1] = state_b1;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter.h b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter.h
new file mode 100644
index 0000000000..5273ff9386
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Transposed direct form I implementation of a bi-quad filter.
+//        b[0] + b[1] • z^(-1) + b[2] • z^(-2)
+// H(z) = ------------------------------------
+//          1 + a[1] • z^(-1) + a[2] • z^(-2)
+class BiQuadFilter {
+ public:
+  // Normalized filter coefficients.
+  // Computed as `[b, a] = scipy.signal.butter(N=2, Wn, btype)`.
+  struct Config {
+    float b[3];  // b[0], b[1], b[2].
+    float a[2];  // a[1], a[2].
+  };
+
+  explicit BiQuadFilter(const Config& config);
+  BiQuadFilter(const BiQuadFilter&) = delete;
+  BiQuadFilter& operator=(const BiQuadFilter&) = delete;
+  ~BiQuadFilter();
+
+  // Sets the filter configuration and resets the internal state.
+  void SetConfig(const Config& config);
+
+  // Zeroes the filter state.
+  void Reset();
+
+  // Filters `x` and writes the output in `y`, which must have the same length
+  // of `x`. In-place processing is supported.
+  void Process(rtc::ArrayView<const float> x, rtc::ArrayView<float> y);
+
+ private:
+  Config config_;
+  struct State {
+    float b[2];
+    float a[2];
+  } state_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_gn/moz.build
new file mode 100644
index 0000000000..f396f42e57
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("biquad_filter_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_unittest.cc
new file mode 100644
index 0000000000..a53036b08e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/biquad_filter_unittest.cc
@@ -0,0 +1,175 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/biquad_filter.h"
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kFrameSize = 8;
+constexpr int kNumFrames = 4;
+using FloatArraySequence =
+    std::array<std::array<float, kFrameSize>, kNumFrames>;
+
+constexpr FloatArraySequence kBiQuadInputSeq = {
+    {{{-87.166290f, -8.029022f, 101.619583f, -0.294296f, -5.825764f, -8.890625f,
+       10.310432f, 54.845333f}},
+     {{-64.647644f, -6.883945f, 11.059189f, -95.242538f, -108.870834f,
+       11.024944f, 63.044102f, -52.709583f}},
+     {{-32.350529f, -18.108028f, -74.022339f, -8.986874f, -1.525581f,
+       103.705513f, 6.346226f, -14.319557f}},
+     {{22.645832f, -64.597153f, 55.462521f, -109.393188f, 10.117825f,
+       -40.019642f, -98.612228f, -8.330326f}}}};
+
+// Computed as `scipy.signal.butter(N=2, Wn=60/24000, btype='highpass')`.
+constexpr BiQuadFilter::Config kBiQuadConfig{
+    {0.99446179f, -1.98892358f, 0.99446179f},
+    {-1.98889291f, 0.98895425f}};
+
+// Comparing to scipy. The expected output is generated as follows:
+// zi = np.float32([0, 0])
+// for i in range(4):
+//   yn, zi = scipy.signal.lfilter(B, A, x[i], zi=zi)
+//   print(yn)
+constexpr FloatArraySequence kBiQuadOutputSeq = {
+    {{{-86.68354497f, -7.02175351f, 102.10290352f, -0.37487333f, -5.87205847f,
+       -8.85521608f, 10.33772563f, 54.51157181f}},
+     {{-64.92531604f, -6.76395978f, 11.15534507f, -94.68073341f, -107.18177856f,
+       13.24642474f, 64.84288941f, -50.97822629f}},
+     {{-30.1579652f, -15.64850899f, -71.06662821f, -5.5883229f, 1.91175353f,
+       106.5572003f, 8.57183046f, -12.06298473f}},
+     {{24.84286614f, -62.18094158f, 57.91488056f, -106.65685933f, 13.38760103f,
+       -36.60367134f, -94.44880104f, -3.59920354f}}}};
+
+// Fails for every pair from two equally sized rtc::ArrayView<float> views such
+// that their relative error is above a given threshold. If the expected value
+// of a pair is 0, `tolerance` is used to check the absolute error.
+void ExpectNearRelative(rtc::ArrayView<const float> expected,
+                        rtc::ArrayView<const float> computed,
+                        const float tolerance) {
+  // The relative error is undefined when the expected value is 0.
+  // When that happens, check the absolute error instead. `safe_den` is used
+  // below to implement such logic.
+  auto safe_den = [](float x) { return (x == 0.0f) ? 1.0f : std::fabs(x); };
+  ASSERT_EQ(expected.size(), computed.size());
+  for (size_t i = 0; i < expected.size(); ++i) {
+    const float abs_diff = std::fabs(expected[i] - computed[i]);
+    // No failure when the values are equal.
+    if (abs_diff == 0.0f) {
+      continue;
+    }
+    SCOPED_TRACE(i);
+    SCOPED_TRACE(expected[i]);
+    SCOPED_TRACE(computed[i]);
+    EXPECT_LE(abs_diff / safe_den(expected[i]), tolerance);
+  }
+}
+
+// Checks that filtering works when different containers are used both as input
+// and as output.
+TEST(BiQuadFilterTest, FilterNotInPlace) {
+  BiQuadFilter filter(kBiQuadConfig);
+  std::array<float, kFrameSize> samples;
+
+  // TODO(https://bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+
+  for (int i = 0; i < kNumFrames; ++i) {
+    SCOPED_TRACE(i);
+    filter.Process(kBiQuadInputSeq[i], samples);
+    ExpectNearRelative(kBiQuadOutputSeq[i], samples, 2e-4f);
+  }
+}
+
+// Checks that filtering works when the same container is used both as input and
+// as output.
+TEST(BiQuadFilterTest, FilterInPlace) {
+  BiQuadFilter filter(kBiQuadConfig);
+  std::array<float, kFrameSize> samples;
+
+  // TODO(https://bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+
+  for (int i = 0; i < kNumFrames; ++i) {
+    SCOPED_TRACE(i);
+    std::copy(kBiQuadInputSeq[i].begin(), kBiQuadInputSeq[i].end(),
+              samples.begin());
+    filter.Process({samples}, {samples});
+    ExpectNearRelative(kBiQuadOutputSeq[i], samples, 2e-4f);
+  }
+}
+
+// Checks that different configurations produce different outputs.
+TEST(BiQuadFilterTest, SetConfigDifferentOutput) {
+  BiQuadFilter filter(/*config=*/{{0.97803048f, -1.95606096f, 0.97803048f},
+                                  {-1.95557824f, 0.95654368f}});
+
+  std::array<float, kFrameSize> samples1;
+  for (int i = 0; i < kNumFrames; ++i) {
+    filter.Process(kBiQuadInputSeq[i], samples1);
+  }
+
+  filter.SetConfig(
+      {{0.09763107f, 0.19526215f, 0.09763107f}, {-0.94280904f, 0.33333333f}});
+  std::array<float, kFrameSize> samples2;
+  for (int i = 0; i < kNumFrames; ++i) {
+    filter.Process(kBiQuadInputSeq[i], samples2);
+  }
+
+  EXPECT_NE(samples1, samples2);
+}
+
+// Checks that when `SetConfig()` is called but the filter coefficients are the
+// same the filter state is reset.
+TEST(BiQuadFilterTest, SetConfigResetsState) {
+  BiQuadFilter filter(kBiQuadConfig);
+
+  std::array<float, kFrameSize> samples1;
+  for (int i = 0; i < kNumFrames; ++i) {
+    filter.Process(kBiQuadInputSeq[i], samples1);
+  }
+
+  filter.SetConfig(kBiQuadConfig);
+  std::array<float, kFrameSize> samples2;
+  for (int i = 0; i < kNumFrames; ++i) {
+    filter.Process(kBiQuadInputSeq[i], samples2);
+  }
+
+  EXPECT_EQ(samples1, samples2);
+}
+
+// Checks that when `Reset()` is called the filter state is reset.
+TEST(BiQuadFilterTest, Reset) {
+  BiQuadFilter filter(kBiQuadConfig);
+
+  std::array<float, kFrameSize> samples1;
+  for (int i = 0; i < kNumFrames; ++i) {
+    filter.Process(kBiQuadInputSeq[i], samples1);
+  }
+
+  filter.Reset();
+  std::array<float, kFrameSize> samples2;
+  for (int i = 0; i < kNumFrames; ++i) {
+    filter.Process(kBiQuadInputSeq[i], samples2);
+  }
+
+  EXPECT_EQ(samples1, samples2);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc
new file mode 100644
index 0000000000..fd759c63e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc
@@ -0,0 +1,384 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/clipping_predictor.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/clipping_predictor_level_buffer.h"
+#include "modules/audio_processing/agc2/gain_map_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kClippingPredictorMaxGainChange = 15;
+
+// Returns an input volume in the [`min_input_volume`, `max_input_volume`] range
+// that reduces `gain_error_db`, which is a gain error estimated when
+// `input_volume` was applied, according to a fixed gain map.
+int ComputeVolumeUpdate(int gain_error_db,
+                        int input_volume,
+                        int min_input_volume,
+                        int max_input_volume) {
+  RTC_DCHECK_GE(input_volume, 0);
+  RTC_DCHECK_LE(input_volume, max_input_volume);
+  if (gain_error_db == 0) {
+    return input_volume;
+  }
+  int new_volume = input_volume;
+  if (gain_error_db > 0) {
+    while (kGainMap[new_volume] - kGainMap[input_volume] < gain_error_db &&
+           new_volume < max_input_volume) {
+      ++new_volume;
+    }
+  } else {
+    while (kGainMap[new_volume] - kGainMap[input_volume] > gain_error_db &&
+           new_volume > min_input_volume) {
+      --new_volume;
+    }
+  }
+  return new_volume;
+}
+
+float ComputeCrestFactor(const ClippingPredictorLevelBuffer::Level& level) {
+  const float crest_factor =
+      FloatS16ToDbfs(level.max) - FloatS16ToDbfs(std::sqrt(level.average));
+  return crest_factor;
+}
+
+// Crest factor-based clipping prediction and clipped level step estimation.
+class ClippingEventPredictor : public ClippingPredictor {
+ public:
+  // ClippingEventPredictor with `num_channels` channels (limited to values
+  // higher than zero); window size `window_length` and reference window size
+  // `reference_window_length` (both referring to the number of frames in the
+  // respective sliding windows and limited to values higher than zero);
+  // reference window delay `reference_window_delay` (delay in frames, limited
+  // to values zero and higher with an additional requirement of
+  // `window_length` < `reference_window_length` + reference_window_delay`);
+  // and an estimation peak threshold `clipping_threshold` and a crest factor
+  // drop threshold `crest_factor_margin` (both in dB).
+  ClippingEventPredictor(int num_channels,
+                         int window_length,
+                         int reference_window_length,
+                         int reference_window_delay,
+                         float clipping_threshold,
+                         float crest_factor_margin)
+      : window_length_(window_length),
+        reference_window_length_(reference_window_length),
+        reference_window_delay_(reference_window_delay),
+        clipping_threshold_(clipping_threshold),
+        crest_factor_margin_(crest_factor_margin) {
+    RTC_DCHECK_GT(num_channels, 0);
+    RTC_DCHECK_GT(window_length, 0);
+    RTC_DCHECK_GT(reference_window_length, 0);
+    RTC_DCHECK_GE(reference_window_delay, 0);
+    RTC_DCHECK_GT(reference_window_length + reference_window_delay,
+                  window_length);
+    const int buffer_length = GetMinFramesProcessed();
+    RTC_DCHECK_GT(buffer_length, 0);
+    for (int i = 0; i < num_channels; ++i) {
+      ch_buffers_.push_back(
+          std::make_unique<ClippingPredictorLevelBuffer>(buffer_length));
+    }
+  }
+
+  ClippingEventPredictor(const ClippingEventPredictor&) = delete;
+  ClippingEventPredictor& operator=(const ClippingEventPredictor&) = delete;
+  ~ClippingEventPredictor() {}
+
+  void Reset() {
+    const int num_channels = ch_buffers_.size();
+    for (int i = 0; i < num_channels; ++i) {
+      ch_buffers_[i]->Reset();
+    }
+  }
+
+  // Analyzes a frame of audio and stores the framewise metrics in
+  // `ch_buffers_`.
+  void Analyze(const AudioFrameView<const float>& frame) {
+    const int num_channels = frame.num_channels();
+    RTC_DCHECK_EQ(num_channels, ch_buffers_.size());
+    const int samples_per_channel = frame.samples_per_channel();
+    RTC_DCHECK_GT(samples_per_channel, 0);
+    for (int channel = 0; channel < num_channels; ++channel) {
+      float sum_squares = 0.0f;
+      float peak = 0.0f;
+      for (const auto& sample : frame.channel(channel)) {
+        sum_squares += sample * sample;
+        peak = std::max(std::fabs(sample), peak);
+      }
+      ch_buffers_[channel]->Push(
+          {sum_squares / static_cast<float>(samples_per_channel), peak});
+    }
+  }
+
+  // Estimates the analog gain adjustment for channel `channel` using a
+  // sliding window over the frame-wise metrics in `ch_buffers_`. Returns an
+  // estimate for the clipped level step equal to `default_clipped_level_step_`
+  // if at least `GetMinFramesProcessed()` frames have been processed since the
+  // last reset and a clipping event is predicted. `level`, `min_mic_level`, and
+  // `max_mic_level` are limited to [0, 255] and `default_step` to [1, 255].
+  absl::optional<int> EstimateClippedLevelStep(int channel,
+                                               int level,
+                                               int default_step,
+                                               int min_mic_level,
+                                               int max_mic_level) const {
+    RTC_CHECK_GE(channel, 0);
+    RTC_CHECK_LT(channel, ch_buffers_.size());
+    RTC_DCHECK_GE(level, 0);
+    RTC_DCHECK_LE(level, 255);
+    RTC_DCHECK_GT(default_step, 0);
+    RTC_DCHECK_LE(default_step, 255);
+    RTC_DCHECK_GE(min_mic_level, 0);
+    RTC_DCHECK_LE(min_mic_level, 255);
+    RTC_DCHECK_GE(max_mic_level, 0);
+    RTC_DCHECK_LE(max_mic_level, 255);
+    if (level <= min_mic_level) {
+      return absl::nullopt;
+    }
+    if (PredictClippingEvent(channel)) {
+      const int new_level =
+          rtc::SafeClamp(level - default_step, min_mic_level, max_mic_level);
+      const int step = level - new_level;
+      if (step > 0) {
+        return step;
+      }
+    }
+    return absl::nullopt;
+  }
+
+ private:
+  int GetMinFramesProcessed() const {
+    return reference_window_delay_ + reference_window_length_;
+  }
+
+  // Predicts clipping events based on the processed audio frames. Returns
+  // true if a clipping event is likely.
+  bool PredictClippingEvent(int channel) const {
+    const auto metrics =
+        ch_buffers_[channel]->ComputePartialMetrics(0, window_length_);
+    if (!metrics.has_value() ||
+        !(FloatS16ToDbfs(metrics.value().max) > clipping_threshold_)) {
+      return false;
+    }
+    const auto reference_metrics = ch_buffers_[channel]->ComputePartialMetrics(
+        reference_window_delay_, reference_window_length_);
+    if (!reference_metrics.has_value()) {
+      return false;
+    }
+    const float crest_factor = ComputeCrestFactor(metrics.value());
+    const float reference_crest_factor =
+        ComputeCrestFactor(reference_metrics.value());
+    if (crest_factor < reference_crest_factor - crest_factor_margin_) {
+      return true;
+    }
+    return false;
+  }
+
+  std::vector<std::unique_ptr<ClippingPredictorLevelBuffer>> ch_buffers_;
+  const int window_length_;
+  const int reference_window_length_;
+  const int reference_window_delay_;
+  const float clipping_threshold_;
+  const float crest_factor_margin_;
+};
+
+// Performs crest factor-based clipping peak prediction.
+class ClippingPeakPredictor : public ClippingPredictor {
+ public:
+  // Ctor. ClippingPeakPredictor with `num_channels` channels (limited to values
+  // higher than zero); window size `window_length` and reference window size
+  // `reference_window_length` (both referring to the number of frames in the
+  // respective sliding windows and limited to values higher than zero);
+  // reference window delay `reference_window_delay` (delay in frames, limited
+  // to values zero and higher with an additional requirement of
+  // `window_length` < `reference_window_length` + reference_window_delay`);
+  // and a clipping prediction threshold `clipping_threshold` (in dB). Adaptive
+  // clipped level step estimation is used if `adaptive_step_estimation` is
+  // true.
+  explicit ClippingPeakPredictor(int num_channels,
+                                 int window_length,
+                                 int reference_window_length,
+                                 int reference_window_delay,
+                                 int clipping_threshold,
+                                 bool adaptive_step_estimation)
+      : window_length_(window_length),
+        reference_window_length_(reference_window_length),
+        reference_window_delay_(reference_window_delay),
+        clipping_threshold_(clipping_threshold),
+        adaptive_step_estimation_(adaptive_step_estimation) {
+    RTC_DCHECK_GT(num_channels, 0);
+    RTC_DCHECK_GT(window_length, 0);
+    RTC_DCHECK_GT(reference_window_length, 0);
+    RTC_DCHECK_GE(reference_window_delay, 0);
+    RTC_DCHECK_GT(reference_window_length + reference_window_delay,
+                  window_length);
+    const int buffer_length = GetMinFramesProcessed();
+    RTC_DCHECK_GT(buffer_length, 0);
+    for (int i = 0; i < num_channels; ++i) {
+      ch_buffers_.push_back(
+          std::make_unique<ClippingPredictorLevelBuffer>(buffer_length));
+    }
+  }
+
+  ClippingPeakPredictor(const ClippingPeakPredictor&) = delete;
+  ClippingPeakPredictor& operator=(const ClippingPeakPredictor&) = delete;
+  ~ClippingPeakPredictor() {}
+
+  void Reset() {
+    const int num_channels = ch_buffers_.size();
+    for (int i = 0; i < num_channels; ++i) {
+      ch_buffers_[i]->Reset();
+    }
+  }
+
+  // Analyzes a frame of audio and stores the framewise metrics in
+  // `ch_buffers_`.
+  void Analyze(const AudioFrameView<const float>& frame) {
+    const int num_channels = frame.num_channels();
+    RTC_DCHECK_EQ(num_channels, ch_buffers_.size());
+    const int samples_per_channel = frame.samples_per_channel();
+    RTC_DCHECK_GT(samples_per_channel, 0);
+    for (int channel = 0; channel < num_channels; ++channel) {
+      float sum_squares = 0.0f;
+      float peak = 0.0f;
+      for (const auto& sample : frame.channel(channel)) {
+        sum_squares += sample * sample;
+        peak = std::max(std::fabs(sample), peak);
+      }
+      ch_buffers_[channel]->Push(
+          {sum_squares / static_cast<float>(samples_per_channel), peak});
+    }
+  }
+
+  // Estimates the analog gain adjustment for channel `channel` using a
+  // sliding window over the frame-wise metrics in `ch_buffers_`. Returns an
+  // estimate for the clipped level step (equal to
+  // `default_clipped_level_step_` if `adaptive_estimation_` is false) if at
+  // least `GetMinFramesProcessed()` frames have been processed since the last
+  // reset and a clipping event is predicted. `level`, `min_mic_level`, and
+  // `max_mic_level` are limited to [0, 255] and `default_step` to [1, 255].
+  absl::optional<int> EstimateClippedLevelStep(int channel,
+                                               int level,
+                                               int default_step,
+                                               int min_mic_level,
+                                               int max_mic_level) const {
+    RTC_DCHECK_GE(channel, 0);
+    RTC_DCHECK_LT(channel, ch_buffers_.size());
+    RTC_DCHECK_GE(level, 0);
+    RTC_DCHECK_LE(level, 255);
+    RTC_DCHECK_GT(default_step, 0);
+    RTC_DCHECK_LE(default_step, 255);
+    RTC_DCHECK_GE(min_mic_level, 0);
+    RTC_DCHECK_LE(min_mic_level, 255);
+    RTC_DCHECK_GE(max_mic_level, 0);
+    RTC_DCHECK_LE(max_mic_level, 255);
+    if (level <= min_mic_level) {
+      return absl::nullopt;
+    }
+    absl::optional<float> estimate_db = EstimatePeakValue(channel);
+    if (estimate_db.has_value() && estimate_db.value() > clipping_threshold_) {
+      int step = 0;
+      if (!adaptive_step_estimation_) {
+        step = default_step;
+      } else {
+        const int estimated_gain_change =
+            rtc::SafeClamp(-static_cast<int>(std::ceil(estimate_db.value())),
+                           -kClippingPredictorMaxGainChange, 0);
+        step =
+            std::max(level - ComputeVolumeUpdate(estimated_gain_change, level,
+                                                 min_mic_level, max_mic_level),
+                     default_step);
+      }
+      const int new_level =
+          rtc::SafeClamp(level - step, min_mic_level, max_mic_level);
+      if (level > new_level) {
+        return level - new_level;
+      }
+    }
+    return absl::nullopt;
+  }
+
+ private:
+  int GetMinFramesProcessed() {
+    return reference_window_delay_ + reference_window_length_;
+  }
+
+  // Predicts clipping sample peaks based on the processed audio frames.
+  // Returns the estimated peak value if clipping is predicted. Otherwise
+  // returns absl::nullopt.
+  absl::optional<float> EstimatePeakValue(int channel) const {
+    const auto reference_metrics = ch_buffers_[channel]->ComputePartialMetrics(
+        reference_window_delay_, reference_window_length_);
+    if (!reference_metrics.has_value()) {
+      return absl::nullopt;
+    }
+    const auto metrics =
+        ch_buffers_[channel]->ComputePartialMetrics(0, window_length_);
+    if (!metrics.has_value() ||
+        !(FloatS16ToDbfs(metrics.value().max) > clipping_threshold_)) {
+      return absl::nullopt;
+    }
+    const float reference_crest_factor =
+        ComputeCrestFactor(reference_metrics.value());
+    const float& mean_squares = metrics.value().average;
+    const float projected_peak =
+        reference_crest_factor + FloatS16ToDbfs(std::sqrt(mean_squares));
+    return projected_peak;
+  }
+
+  std::vector<std::unique_ptr<ClippingPredictorLevelBuffer>> ch_buffers_;
+  const int window_length_;
+  const int reference_window_length_;
+  const int reference_window_delay_;
+  const int clipping_threshold_;
+  const bool adaptive_step_estimation_;
+};
+
+}  // namespace
+
+std::unique_ptr<ClippingPredictor> CreateClippingPredictor(
+    int num_channels,
+    const AudioProcessing::Config::GainController1::AnalogGainController::
+        ClippingPredictor& config) {
+  if (!config.enabled) {
+    RTC_LOG(LS_INFO) << "[AGC2] Clipping prediction disabled.";
+    return nullptr;
+  }
+  RTC_LOG(LS_INFO) << "[AGC2] Clipping prediction enabled.";
+  using ClippingPredictorMode = AudioProcessing::Config::GainController1::
+      AnalogGainController::ClippingPredictor::Mode;
+  switch (config.mode) {
+    case ClippingPredictorMode::kClippingEventPrediction:
+      return std::make_unique<ClippingEventPredictor>(
+          num_channels, config.window_length, config.reference_window_length,
+          config.reference_window_delay, config.clipping_threshold,
+          config.crest_factor_margin);
+    case ClippingPredictorMode::kAdaptiveStepClippingPeakPrediction:
+      return std::make_unique<ClippingPeakPredictor>(
+          num_channels, config.window_length, config.reference_window_length,
+          config.reference_window_delay, config.clipping_threshold,
+          /*adaptive_step_estimation=*/true);
+    case ClippingPredictorMode::kFixedStepClippingPeakPrediction:
+      return std::make_unique<ClippingPeakPredictor>(
+          num_channels, config.window_length, config.reference_window_length,
+          config.reference_window_delay, config.clipping_threshold,
+          /*adaptive_step_estimation=*/false);
+  }
+  RTC_DCHECK_NOTREACHED();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.h b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.h
new file mode 100644
index 0000000000..14612508c0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+// Frame-wise clipping prediction and clipped level step estimation. Analyzes
+// 10 ms multi-channel frames and estimates an analog mic level decrease step
+// to possibly avoid clipping when predicted. `Analyze()` and
+// `EstimateClippedLevelStep()` can be called in any order.
+class ClippingPredictor {
+ public:
+  virtual ~ClippingPredictor() = default;
+
+  virtual void Reset() = 0;
+
+  // Analyzes a 10 ms multi-channel audio frame.
+  virtual void Analyze(const AudioFrameView<const float>& frame) = 0;
+
+  // Predicts if clipping is going to occur for the specified `channel` in the
+  // near-future and, if so, it returns a recommended analog mic level decrease
+  // step. Returns absl::nullopt if clipping is not predicted.
+  // `level` is the current analog mic level, `default_step` is the amount the
+  // mic level is lowered by the analog controller with every clipping event and
+  // `min_mic_level` and `max_mic_level` is the range of allowed analog mic
+  // levels.
+  virtual absl::optional<int> EstimateClippedLevelStep(
+      int channel,
+      int level,
+      int default_step,
+      int min_mic_level,
+      int max_mic_level) const = 0;
+};
+
+// Creates a ClippingPredictor based on the provided `config`. When enabled,
+// the following must hold for `config`:
+// `window_length < reference_window_length + reference_window_delay`.
+// Returns `nullptr` if `config.enabled` is false.
+std::unique_ptr<ClippingPredictor> CreateClippingPredictor(
+    int num_channels,
+    const AudioProcessing::Config::GainController1::AnalogGainController::
+        ClippingPredictor& config);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_gn/moz.build
new file mode 100644
index 0000000000..9cddd69abe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("clipping_predictor_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer.cc b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer.cc
new file mode 100644
index 0000000000..fe4cf2a154
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer.cc
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/clipping_predictor_level_buffer.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+bool ClippingPredictorLevelBuffer::Level::operator==(const Level& level) const {
+  constexpr float kEpsilon = 1e-6f;
+  return std::fabs(average - level.average) < kEpsilon &&
+         std::fabs(max - level.max) < kEpsilon;
+}
+
+ClippingPredictorLevelBuffer::ClippingPredictorLevelBuffer(int capacity)
+    : tail_(-1), size_(0), data_(std::max(1, capacity)) {
+  if (capacity > kMaxCapacity) {
+    RTC_LOG(LS_WARNING) << "[agc]: ClippingPredictorLevelBuffer exceeds the "
+                        << "maximum allowed capacity. Capacity: " << capacity;
+  }
+  RTC_DCHECK(!data_.empty());
+}
+
+void ClippingPredictorLevelBuffer::Reset() {
+  tail_ = -1;
+  size_ = 0;
+}
+
+void ClippingPredictorLevelBuffer::Push(Level level) {
+  ++tail_;
+  if (tail_ == Capacity()) {
+    tail_ = 0;
+  }
+  if (size_ < Capacity()) {
+    size_++;
+  }
+  data_[tail_] = level;
+}
+
+// TODO(bugs.webrtc.org/12774): Optimize partial computation for long buffers.
+absl::optional<ClippingPredictorLevelBuffer::Level>
+ClippingPredictorLevelBuffer::ComputePartialMetrics(int delay,
+                                                    int num_items) const {
+  RTC_DCHECK_GE(delay, 0);
+  RTC_DCHECK_LT(delay, Capacity());
+  RTC_DCHECK_GT(num_items, 0);
+  RTC_DCHECK_LE(num_items, Capacity());
+  RTC_DCHECK_LE(delay + num_items, Capacity());
+  if (delay + num_items > Size()) {
+    return absl::nullopt;
+  }
+  float sum = 0.0f;
+  float max = 0.0f;
+  for (int i = 0; i < num_items && i < Size(); ++i) {
+    int idx = tail_ - delay - i;
+    if (idx < 0) {
+      idx += Capacity();
+    }
+    sum += data_[idx].average;
+    max = std::fmax(data_[idx].max, max);
+  }
+  return absl::optional<Level>({sum / static_cast<float>(num_items), max});
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer.h b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer.h
new file mode 100644
index 0000000000..c9032773a6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_LEVEL_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_LEVEL_BUFFER_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+// A circular buffer to store frame-wise `Level` items for clipping prediction.
+// The current implementation is not optimized for large buffer lengths.
+class ClippingPredictorLevelBuffer {
+ public:
+  struct Level {
+    float average;
+    float max;
+    bool operator==(const Level& level) const;
+  };
+
+  // Recommended maximum capacity. It is possible to create a buffer with a
+  // larger capacity, but the implementation is not optimized for large values.
+  static constexpr int kMaxCapacity = 100;
+
+  // Ctor. Sets the buffer capacity to max(1, `capacity`) and logs a warning
+  // message if the capacity is greater than `kMaxCapacity`.
+  explicit ClippingPredictorLevelBuffer(int capacity);
+  ~ClippingPredictorLevelBuffer() {}
+  ClippingPredictorLevelBuffer(const ClippingPredictorLevelBuffer&) = delete;
+  ClippingPredictorLevelBuffer& operator=(const ClippingPredictorLevelBuffer&) =
+      delete;
+
+  void Reset();
+
+  // Returns the current number of items stored in the buffer.
+  int Size() const { return size_; }
+
+  // Returns the capacity of the buffer.
+  int Capacity() const { return data_.size(); }
+
+  // Adds a `level` item into the circular buffer `data_`. Stores at most
+  // `Capacity()` items. If more items are pushed, the new item replaces the
+  // least recently pushed item.
+  void Push(Level level);
+
+  // If at least `num_items` + `delay` items have been pushed, returns the
+  // average and maximum value for the `num_items` most recently pushed items
+  // from `delay` to `delay` - `num_items` (a delay equal to zero corresponds
+  // to the most recently pushed item). The value of `delay` is limited to
+  // [0, N] and `num_items` to [1, M] where N + M is the capacity of the buffer.
+  absl::optional<Level> ComputePartialMetrics(int delay, int num_items) const;
+
+ private:
+  int tail_;
+  int size_;
+  std::vector<Level> data_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_CLIPPING_PREDICTOR_LEVEL_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer_unittest.cc
new file mode 100644
index 0000000000..7af9a436c9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_level_buffer_unittest.cc
@@ -0,0 +1,131 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/clipping_predictor_level_buffer.h"
+
+#include <algorithm>
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Eq;
+using ::testing::Optional;
+
+class ClippingPredictorLevelBufferParametrization
+    : public ::testing::TestWithParam<int> {
+ protected:
+  int capacity() const { return GetParam(); }
+};
+
+TEST_P(ClippingPredictorLevelBufferParametrization, CheckEmptyBufferSize) {
+  ClippingPredictorLevelBuffer buffer(capacity());
+  EXPECT_EQ(buffer.Capacity(), std::max(capacity(), 1));
+  EXPECT_EQ(buffer.Size(), 0);
+}
+
+TEST_P(ClippingPredictorLevelBufferParametrization, CheckHalfEmptyBufferSize) {
+  ClippingPredictorLevelBuffer buffer(capacity());
+  for (int i = 0; i < buffer.Capacity() / 2; ++i) {
+    buffer.Push({2, 4});
+  }
+  EXPECT_EQ(buffer.Capacity(), std::max(capacity(), 1));
+  EXPECT_EQ(buffer.Size(), std::max(capacity(), 1) / 2);
+}
+
+TEST_P(ClippingPredictorLevelBufferParametrization, CheckFullBufferSize) {
+  ClippingPredictorLevelBuffer buffer(capacity());
+  for (int i = 0; i < buffer.Capacity(); ++i) {
+    buffer.Push({2, 4});
+  }
+  EXPECT_EQ(buffer.Capacity(), std::max(capacity(), 1));
+  EXPECT_EQ(buffer.Size(), std::max(capacity(), 1));
+}
+
+TEST_P(ClippingPredictorLevelBufferParametrization, CheckLargeBufferSize) {
+  ClippingPredictorLevelBuffer buffer(capacity());
+  for (int i = 0; i < 2 * buffer.Capacity(); ++i) {
+    buffer.Push({2, 4});
+  }
+  EXPECT_EQ(buffer.Capacity(), std::max(capacity(), 1));
+  EXPECT_EQ(buffer.Size(), std::max(capacity(), 1));
+}
+
+TEST_P(ClippingPredictorLevelBufferParametrization, CheckSizeAfterReset) {
+  ClippingPredictorLevelBuffer buffer(capacity());
+  buffer.Push({1, 1});
+  buffer.Push({1, 1});
+  buffer.Reset();
+  EXPECT_EQ(buffer.Capacity(), std::max(capacity(), 1));
+  EXPECT_EQ(buffer.Size(), 0);
+  buffer.Push({1, 1});
+  EXPECT_EQ(buffer.Capacity(), std::max(capacity(), 1));
+  EXPECT_EQ(buffer.Size(), 1);
+}
+
+INSTANTIATE_TEST_SUITE_P(ClippingPredictorLevelBufferTest,
+                         ClippingPredictorLevelBufferParametrization,
+                         ::testing::Values(-1, 0, 1, 123));
+
+TEST(ClippingPredictorLevelBufferTest, CheckMetricsAfterFullBuffer) {
+  ClippingPredictorLevelBuffer buffer(/*capacity=*/2);
+  buffer.Push({1, 2});
+  buffer.Push({3, 6});
+  EXPECT_THAT(buffer.ComputePartialMetrics(/*delay=*/0, /*num_items=*/1),
+              Optional(Eq(ClippingPredictorLevelBuffer::Level{3, 6})));
+  EXPECT_THAT(buffer.ComputePartialMetrics(/*delay=*/1, /*num_items=*/1),
+              Optional(Eq(ClippingPredictorLevelBuffer::Level{1, 2})));
+  EXPECT_THAT(buffer.ComputePartialMetrics(/*delay=*/0, /*num_items=*/2),
+              Optional(Eq(ClippingPredictorLevelBuffer::Level{2, 6})));
+}
+
+TEST(ClippingPredictorLevelBufferTest, CheckMetricsAfterPushBeyondCapacity) {
+  ClippingPredictorLevelBuffer buffer(/*capacity=*/2);
+  buffer.Push({1, 1});
+  buffer.Push({3, 6});
+  buffer.Push({5, 10});
+  buffer.Push({7, 14});
+  buffer.Push({6, 12});
+  EXPECT_THAT(buffer.ComputePartialMetrics(/*delay=*/0, /*num_items=*/1),
+              Optional(Eq(ClippingPredictorLevelBuffer::Level{6, 12})));
+  EXPECT_THAT(buffer.ComputePartialMetrics(/*delay=*/1, /*num_items=*/1),
+              Optional(Eq(ClippingPredictorLevelBuffer::Level{7, 14})));
+  EXPECT_THAT(buffer.ComputePartialMetrics(/*delay=*/0, /*num_items=*/2),
+              Optional(Eq(ClippingPredictorLevelBuffer::Level{6.5f, 14})));
+}
+
+TEST(ClippingPredictorLevelBufferTest, CheckMetricsAfterTooFewItems) {
+  ClippingPredictorLevelBuffer buffer(/*capacity=*/4);
+  buffer.Push({1, 2});
+  buffer.Push({3, 6});
+  EXPECT_EQ(buffer.ComputePartialMetrics(/*delay=*/0, /*num_items=*/3),
+            absl::nullopt);
+  EXPECT_EQ(buffer.ComputePartialMetrics(/*delay=*/2, /*num_items=*/1),
+            absl::nullopt);
+}
+
+TEST(ClippingPredictorLevelBufferTest, CheckMetricsAfterReset) {
+  ClippingPredictorLevelBuffer buffer(/*capacity=*/2);
+  buffer.Push({1, 2});
+  buffer.Reset();
+  buffer.Push({5, 10});
+  buffer.Push({7, 14});
+  EXPECT_THAT(buffer.ComputePartialMetrics(/*delay=*/0, /*num_items=*/1),
+              Optional(Eq(ClippingPredictorLevelBuffer::Level{7, 14})));
+  EXPECT_THAT(buffer.ComputePartialMetrics(/*delay=*/0, /*num_items=*/2),
+              Optional(Eq(ClippingPredictorLevelBuffer::Level{6, 14})));
+  EXPECT_THAT(buffer.ComputePartialMetrics(/*delay=*/1, /*num_items=*/1),
+              Optional(Eq(ClippingPredictorLevelBuffer::Level{5, 10})));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_unittest.cc
new file mode 100644
index 0000000000..af73107749
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor_unittest.cc
@@ -0,0 +1,491 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/clipping_predictor.h"
+
+#include <cstdint>
+#include <limits>
+#include <tuple>
+
+#include "rtc_base/checks.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Eq;
+using ::testing::Optional;
+using ClippingPredictorConfig = AudioProcessing::Config::GainController1::
+    AnalogGainController::ClippingPredictor;
+using ClippingPredictorMode = AudioProcessing::Config::GainController1::
+    AnalogGainController::ClippingPredictor::Mode;
+
+constexpr int kSampleRateHz = 32000;
+constexpr int kNumChannels = 1;
+constexpr int kSamplesPerChannel = kSampleRateHz / 100;
+constexpr int kMaxMicLevel = 255;
+constexpr int kMinMicLevel = 12;
+constexpr int kDefaultClippedLevelStep = 15;
+constexpr float kMaxSampleS16 =
+    static_cast<float>(std::numeric_limits<int16_t>::max());
+
+// Threshold in dB corresponding to a signal with an amplitude equal to 99% of
+// the dynamic range - i.e., computed as `20*log10(0.99)`.
+constexpr float kClippingThresholdDb = -0.08729610804900176f;
+
+void CallAnalyze(int num_calls,
+                 const AudioFrameView<const float>& frame,
+                 ClippingPredictor& predictor) {
+  for (int i = 0; i < num_calls; ++i) {
+    predictor.Analyze(frame);
+  }
+}
+
+// Creates and analyzes an audio frame with a non-zero (approx. 4.15dB) crest
+// factor.
+void AnalyzeNonZeroCrestFactorAudio(int num_calls,
+                                    int num_channels,
+                                    float peak_ratio,
+                                    ClippingPredictor& predictor) {
+  RTC_DCHECK_GT(num_calls, 0);
+  RTC_DCHECK_GT(num_channels, 0);
+  RTC_DCHECK_LE(peak_ratio, 1.0f);
+  std::vector<float*> audio(num_channels);
+  std::vector<float> audio_data(num_channels * kSamplesPerChannel, 0.0f);
+  for (int channel = 0; channel < num_channels; ++channel) {
+    audio[channel] = &audio_data[channel * kSamplesPerChannel];
+    for (int sample = 0; sample < kSamplesPerChannel; sample += 10) {
+      audio[channel][sample] = 0.1f * peak_ratio * kMaxSampleS16;
+      audio[channel][sample + 1] = 0.2f * peak_ratio * kMaxSampleS16;
+      audio[channel][sample + 2] = 0.3f * peak_ratio * kMaxSampleS16;
+      audio[channel][sample + 3] = 0.4f * peak_ratio * kMaxSampleS16;
+      audio[channel][sample + 4] = 0.5f * peak_ratio * kMaxSampleS16;
+      audio[channel][sample + 5] = 0.6f * peak_ratio * kMaxSampleS16;
+      audio[channel][sample + 6] = 0.7f * peak_ratio * kMaxSampleS16;
+      audio[channel][sample + 7] = 0.8f * peak_ratio * kMaxSampleS16;
+      audio[channel][sample + 8] = 0.9f * peak_ratio * kMaxSampleS16;
+      audio[channel][sample + 9] = 1.0f * peak_ratio * kMaxSampleS16;
+    }
+  }
+  AudioFrameView<const float> frame(audio.data(), num_channels,
+                                    kSamplesPerChannel);
+  CallAnalyze(num_calls, frame, predictor);
+}
+
+void CheckChannelEstimatesWithValue(int num_channels,
+                                    int level,
+                                    int default_step,
+                                    int min_mic_level,
+                                    int max_mic_level,
+                                    const ClippingPredictor& predictor,
+                                    int expected) {
+  for (int i = 0; i < num_channels; ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_THAT(predictor.EstimateClippedLevelStep(
+                    i, level, default_step, min_mic_level, max_mic_level),
+                Optional(Eq(expected)));
+  }
+}
+
+void CheckChannelEstimatesWithoutValue(int num_channels,
+                                       int level,
+                                       int default_step,
+                                       int min_mic_level,
+                                       int max_mic_level,
+                                       const ClippingPredictor& predictor) {
+  for (int i = 0; i < num_channels; ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_EQ(predictor.EstimateClippedLevelStep(i, level, default_step,
+                                                 min_mic_level, max_mic_level),
+              absl::nullopt);
+  }
+}
+
+// Creates and analyzes an audio frame with a zero crest factor.
+void AnalyzeZeroCrestFactorAudio(int num_calls,
+                                 int num_channels,
+                                 float peak_ratio,
+                                 ClippingPredictor& predictor) {
+  RTC_DCHECK_GT(num_calls, 0);
+  RTC_DCHECK_GT(num_channels, 0);
+  RTC_DCHECK_LE(peak_ratio, 1.f);
+  std::vector<float*> audio(num_channels);
+  std::vector<float> audio_data(num_channels * kSamplesPerChannel, 0.f);
+  for (int channel = 0; channel < num_channels; ++channel) {
+    audio[channel] = &audio_data[channel * kSamplesPerChannel];
+    for (int sample = 0; sample < kSamplesPerChannel; ++sample) {
+      audio[channel][sample] = peak_ratio * kMaxSampleS16;
+    }
+  }
+  auto frame = AudioFrameView<const float>(audio.data(), num_channels,
+                                           kSamplesPerChannel);
+  CallAnalyze(num_calls, frame, predictor);
+}
+
+TEST(ClippingPeakPredictorTest, NoPredictorCreated) {
+  auto predictor =
+      CreateClippingPredictor(kNumChannels, /*config=*/{/*enabled=*/false});
+  EXPECT_FALSE(predictor);
+}
+
+TEST(ClippingPeakPredictorTest, ClippingEventPredictionCreated) {
+  // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+  auto predictor = CreateClippingPredictor(
+      kNumChannels,
+      /*config=*/{/*enabled=*/true,
+                  /*mode=*/ClippingPredictorMode::kClippingEventPrediction});
+  EXPECT_TRUE(predictor);
+}
+
+TEST(ClippingPeakPredictorTest, AdaptiveStepClippingPeakPredictionCreated) {
+  // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+  auto predictor = CreateClippingPredictor(
+      kNumChannels, /*config=*/{
+          /*enabled=*/true,
+          /*mode=*/ClippingPredictorMode::kAdaptiveStepClippingPeakPrediction});
+  EXPECT_TRUE(predictor);
+}
+
+TEST(ClippingPeakPredictorTest, FixedStepClippingPeakPredictionCreated) {
+  // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+  auto predictor = CreateClippingPredictor(
+      kNumChannels, /*config=*/{
+          /*enabled=*/true,
+          /*mode=*/ClippingPredictorMode::kFixedStepClippingPeakPrediction});
+  EXPECT_TRUE(predictor);
+}
+
+class ClippingPredictorParameterization
+    : public ::testing::TestWithParam<std::tuple<int, int, int, int>> {
+ protected:
+  int num_channels() const { return std::get<0>(GetParam()); }
+  ClippingPredictorConfig GetConfig(ClippingPredictorMode mode) const {
+    // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+    return {/*enabled=*/true,
+            /*mode=*/mode,
+            /*window_length=*/std::get<1>(GetParam()),
+            /*reference_window_length=*/std::get<2>(GetParam()),
+            /*reference_window_delay=*/std::get<3>(GetParam()),
+            /*clipping_threshold=*/-1.0f,
+            /*crest_factor_margin=*/0.5f};
+  }
+};
+
+TEST_P(ClippingPredictorParameterization,
+       CheckClippingEventPredictorEstimateAfterCrestFactorDrop) {
+  const ClippingPredictorConfig config =
+      GetConfig(ClippingPredictorMode::kClippingEventPrediction);
+  if (config.reference_window_length + config.reference_window_delay <=
+      config.window_length) {
+    return;
+  }
+  auto predictor = CreateClippingPredictor(num_channels(), config);
+  AnalyzeNonZeroCrestFactorAudio(
+      /*num_calls=*/config.reference_window_length +
+          config.reference_window_delay - config.window_length,
+      num_channels(), /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(num_channels(), /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  AnalyzeZeroCrestFactorAudio(config.window_length, num_channels(),
+                              /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithValue(
+      num_channels(), /*level=*/255, kDefaultClippedLevelStep, kMinMicLevel,
+      kMaxMicLevel, *predictor, kDefaultClippedLevelStep);
+}
+
+TEST_P(ClippingPredictorParameterization,
+       CheckClippingEventPredictorNoEstimateAfterConstantCrestFactor) {
+  const ClippingPredictorConfig config =
+      GetConfig(ClippingPredictorMode::kClippingEventPrediction);
+  if (config.reference_window_length + config.reference_window_delay <=
+      config.window_length) {
+    return;
+  }
+  auto predictor = CreateClippingPredictor(num_channels(), config);
+  AnalyzeNonZeroCrestFactorAudio(
+      /*num_calls=*/config.reference_window_length +
+          config.reference_window_delay - config.window_length,
+      num_channels(), /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(num_channels(), /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/config.window_length,
+                                 num_channels(),
+                                 /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(num_channels(), /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+}
+
+TEST_P(ClippingPredictorParameterization,
+       CheckClippingPeakPredictorEstimateAfterHighCrestFactor) {
+  const ClippingPredictorConfig config =
+      GetConfig(ClippingPredictorMode::kAdaptiveStepClippingPeakPrediction);
+  if (config.reference_window_length + config.reference_window_delay <=
+      config.window_length) {
+    return;
+  }
+  auto predictor = CreateClippingPredictor(num_channels(), config);
+  AnalyzeNonZeroCrestFactorAudio(
+      /*num_calls=*/config.reference_window_length +
+          config.reference_window_delay - config.window_length,
+      num_channels(), /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(num_channels(), /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/config.window_length,
+                                 num_channels(),
+                                 /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithValue(
+      num_channels(), /*level=*/255, kDefaultClippedLevelStep, kMinMicLevel,
+      kMaxMicLevel, *predictor, kDefaultClippedLevelStep);
+}
+
+TEST_P(ClippingPredictorParameterization,
+       CheckClippingPeakPredictorNoEstimateAfterLowCrestFactor) {
+  const ClippingPredictorConfig config =
+      GetConfig(ClippingPredictorMode::kAdaptiveStepClippingPeakPrediction);
+  if (config.reference_window_length + config.reference_window_delay <=
+      config.window_length) {
+    return;
+  }
+  auto predictor = CreateClippingPredictor(num_channels(), config);
+  AnalyzeZeroCrestFactorAudio(
+      /*num_calls=*/config.reference_window_length +
+          config.reference_window_delay - config.window_length,
+      num_channels(), /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(num_channels(), /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/config.window_length,
+                                 num_channels(),
+                                 /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(num_channels(), /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+}
+
+INSTANTIATE_TEST_SUITE_P(GainController1ClippingPredictor,
+                         ClippingPredictorParameterization,
+                         ::testing::Combine(::testing::Values(1, 5),
+                                            ::testing::Values(1, 5, 10),
+                                            ::testing::Values(1, 5),
+                                            ::testing::Values(0, 1, 5)));
+
+class ClippingEventPredictorParameterization
+    : public ::testing::TestWithParam<std::tuple<float, float>> {
+ protected:
+  ClippingPredictorConfig GetConfig() const {
+    // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+    return {/*enabled=*/true,
+            /*mode=*/ClippingPredictorMode::kClippingEventPrediction,
+            /*window_length=*/5,
+            /*reference_window_length=*/5,
+            /*reference_window_delay=*/5,
+            /*clipping_threshold=*/std::get<0>(GetParam()),
+            /*crest_factor_margin=*/std::get<1>(GetParam())};
+  }
+};
+
+TEST_P(ClippingEventPredictorParameterization,
+       CheckEstimateAfterCrestFactorDrop) {
+  const ClippingPredictorConfig config = GetConfig();
+  auto predictor = CreateClippingPredictor(kNumChannels, config);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/config.reference_window_length,
+                                 kNumChannels, /*peak_ratio=*/0.99f,
+                                 *predictor);
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  AnalyzeZeroCrestFactorAudio(config.window_length, kNumChannels,
+                              /*peak_ratio=*/0.99f, *predictor);
+  // TODO(bugs.webrtc.org/12774): Add clarifying comment.
+  // TODO(bugs.webrtc.org/12774): Remove 4.15f threshold and split tests.
+  if (config.clipping_threshold < kClippingThresholdDb &&
+      config.crest_factor_margin < 4.15f) {
+    CheckChannelEstimatesWithValue(
+        kNumChannels, /*level=*/255, kDefaultClippedLevelStep, kMinMicLevel,
+        kMaxMicLevel, *predictor, kDefaultClippedLevelStep);
+  } else {
+    CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                      kDefaultClippedLevelStep, kMinMicLevel,
+                                      kMaxMicLevel, *predictor);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(GainController1ClippingPredictor,
+                         ClippingEventPredictorParameterization,
+                         ::testing::Combine(::testing::Values(-1.0f, 0.0f),
+                                            ::testing::Values(3.0f, 4.16f)));
+
+class ClippingPredictorModeParameterization
+    : public ::testing::TestWithParam<ClippingPredictorMode> {
+ protected:
+  ClippingPredictorConfig GetConfig(float clipping_threshold_dbfs) const {
+    // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+    return {/*enabled=*/true,
+            /*mode=*/GetParam(),
+            /*window_length=*/5,
+            /*reference_window_length=*/5,
+            /*reference_window_delay=*/5,
+            /*clipping_threshold=*/clipping_threshold_dbfs,
+            /*crest_factor_margin=*/3.0f};
+  }
+};
+
+TEST_P(ClippingPredictorModeParameterization,
+       CheckEstimateAfterHighCrestFactorWithNoClippingMargin) {
+  const ClippingPredictorConfig config = GetConfig(
+      /*clipping_threshold_dbfs=*/0.0f);
+  auto predictor = CreateClippingPredictor(kNumChannels, config);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/config.reference_window_length,
+                                 kNumChannels, /*peak_ratio=*/0.99f,
+                                 *predictor);
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  AnalyzeZeroCrestFactorAudio(config.window_length, kNumChannels,
+                              /*peak_ratio=*/0.99f, *predictor);
+  // Since the clipping threshold is set to 0 dBFS, `EstimateClippedLevelStep()`
+  // is expected to return an unavailable value.
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+}
+
+TEST_P(ClippingPredictorModeParameterization,
+       CheckEstimateAfterHighCrestFactorWithClippingMargin) {
+  const ClippingPredictorConfig config =
+      GetConfig(/*clipping_threshold_dbfs=*/-1.0f);
+  auto predictor = CreateClippingPredictor(kNumChannels, config);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/config.reference_window_length,
+                                 kNumChannels,
+                                 /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  AnalyzeZeroCrestFactorAudio(config.window_length, kNumChannels,
+                              /*peak_ratio=*/0.99f, *predictor);
+  // TODO(bugs.webrtc.org/12774): Add clarifying comment.
+  const float expected_step =
+      config.mode == ClippingPredictorMode::kAdaptiveStepClippingPeakPrediction
+          ? 17
+          : kDefaultClippedLevelStep;
+  CheckChannelEstimatesWithValue(kNumChannels, /*level=*/255,
+                                 kDefaultClippedLevelStep, kMinMicLevel,
+                                 kMaxMicLevel, *predictor, expected_step);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    GainController1ClippingPredictor,
+    ClippingPredictorModeParameterization,
+    ::testing::Values(
+        ClippingPredictorMode::kAdaptiveStepClippingPeakPrediction,
+        ClippingPredictorMode::kFixedStepClippingPeakPrediction));
+
+TEST(ClippingEventPredictorTest, CheckEstimateAfterReset) {
+  // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+  constexpr ClippingPredictorConfig kConfig{
+      /*enabled=*/true,
+      /*mode=*/ClippingPredictorMode::kClippingEventPrediction,
+      /*window_length=*/5,
+      /*reference_window_length=*/5,
+      /*reference_window_delay=*/5,
+      /*clipping_threshold=*/-1.0f,
+      /*crest_factor_margin=*/3.0f};
+  auto predictor = CreateClippingPredictor(kNumChannels, kConfig);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/kConfig.reference_window_length,
+                                 kNumChannels,
+                                 /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  predictor->Reset();
+  AnalyzeZeroCrestFactorAudio(kConfig.window_length, kNumChannels,
+                              /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+}
+
+TEST(ClippingPeakPredictorTest, CheckNoEstimateAfterReset) {
+  // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+  constexpr ClippingPredictorConfig kConfig{
+      /*enabled=*/true,
+      /*mode=*/ClippingPredictorMode::kAdaptiveStepClippingPeakPrediction,
+      /*window_length=*/5,
+      /*reference_window_length=*/5,
+      /*reference_window_delay=*/5,
+      /*clipping_threshold=*/-1.0f};
+  auto predictor = CreateClippingPredictor(kNumChannels, kConfig);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/kConfig.reference_window_length,
+                                 kNumChannels,
+                                 /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  predictor->Reset();
+  AnalyzeZeroCrestFactorAudio(kConfig.window_length, kNumChannels,
+                              /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+}
+
+TEST(ClippingPeakPredictorTest, CheckAdaptiveStepEstimate) {
+  // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+  constexpr ClippingPredictorConfig kConfig{
+      /*enabled=*/true,
+      /*mode=*/ClippingPredictorMode::kAdaptiveStepClippingPeakPrediction,
+      /*window_length=*/5,
+      /*reference_window_length=*/5,
+      /*reference_window_delay=*/5,
+      /*clipping_threshold=*/-1.0f};
+  auto predictor = CreateClippingPredictor(kNumChannels, kConfig);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/kConfig.reference_window_length,
+                                 kNumChannels, /*peak_ratio=*/0.99f,
+                                 *predictor);
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  AnalyzeZeroCrestFactorAudio(kConfig.window_length, kNumChannels,
+                              /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithValue(kNumChannels, /*level=*/255,
+                                 kDefaultClippedLevelStep, kMinMicLevel,
+                                 kMaxMicLevel, *predictor, /*expected=*/17);
+}
+
+TEST(ClippingPeakPredictorTest, CheckFixedStepEstimate) {
+  // TODO(bugs.webrtc.org/12874): Use designated initializers one fixed.
+  constexpr ClippingPredictorConfig kConfig{
+      /*enabled=*/true,
+      /*mode=*/ClippingPredictorMode::kFixedStepClippingPeakPrediction,
+      /*window_length=*/5,
+      /*reference_window_length=*/5,
+      /*reference_window_delay=*/5,
+      /*clipping_threshold=*/-1.0f};
+  auto predictor = CreateClippingPredictor(kNumChannels, kConfig);
+  AnalyzeNonZeroCrestFactorAudio(/*num_calls=*/kConfig.reference_window_length,
+                                 kNumChannels, /*peak_ratio=*/0.99f,
+                                 *predictor);
+  CheckChannelEstimatesWithoutValue(kNumChannels, /*level=*/255,
+                                    kDefaultClippedLevelStep, kMinMicLevel,
+                                    kMaxMicLevel, *predictor);
+  AnalyzeZeroCrestFactorAudio(kConfig.window_length, kNumChannels,
+                              /*peak_ratio=*/0.99f, *predictor);
+  CheckChannelEstimatesWithValue(
+      kNumChannels, /*level=*/255, kDefaultClippedLevelStep, kMinMicLevel,
+      kMaxMicLevel, *predictor, kDefaultClippedLevelStep);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/common_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/common_gn/moz.build
new file mode 100644
index 0000000000..274cae23a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/common_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("common_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc b/third_party/libwebrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
new file mode 100644
index 0000000000..221b499e32
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
@@ -0,0 +1,229 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/compute_interpolated_gain_curve.h"
+
+#include <algorithm>
+#include <cmath>
+#include <queue>
+#include <tuple>
+#include <utility>
+#include <vector>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+#include "modules/audio_processing/agc2/limiter_db_gain_curve.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+std::pair<double, double> ComputeLinearApproximationParams(
+    const LimiterDbGainCurve* limiter,
+    const double x) {
+  const double m = limiter->GetGainFirstDerivativeLinear(x);
+  const double q = limiter->GetGainLinear(x) - m * x;
+  return {m, q};
+}
+
+double ComputeAreaUnderPiecewiseLinearApproximation(
+    const LimiterDbGainCurve* limiter,
+    const double x0,
+    const double x1) {
+  RTC_CHECK_LT(x0, x1);
+
+  // Linear approximation in x0 and x1.
+  double m0, q0, m1, q1;
+  std::tie(m0, q0) = ComputeLinearApproximationParams(limiter, x0);
+  std::tie(m1, q1) = ComputeLinearApproximationParams(limiter, x1);
+
+  // Intersection point between two adjacent linear pieces.
+  RTC_CHECK_NE(m1, m0);
+  const double x_split = (q0 - q1) / (m1 - m0);
+  RTC_CHECK_LT(x0, x_split);
+  RTC_CHECK_LT(x_split, x1);
+
+  auto area_under_linear_piece = [](double x_l, double x_r, double m,
+                                    double q) {
+    return x_r * (m * x_r / 2.0 + q) - x_l * (m * x_l / 2.0 + q);
+  };
+  return area_under_linear_piece(x0, x_split, m0, q0) +
+         area_under_linear_piece(x_split, x1, m1, q1);
+}
+
+// Computes the approximation error in the limiter region for a given interval.
+// The error is computed as the difference between the areas beneath the limiter
+// curve to approximate and its linear under-approximation.
+double LimiterUnderApproximationNegativeError(const LimiterDbGainCurve* limiter,
+                                              const double x0,
+                                              const double x1) {
+  const double area_limiter = limiter->GetGainIntegralLinear(x0, x1);
+  const double area_interpolated_curve =
+      ComputeAreaUnderPiecewiseLinearApproximation(limiter, x0, x1);
+  RTC_CHECK_GE(area_limiter, area_interpolated_curve);
+  return area_limiter - area_interpolated_curve;
+}
+
+// Automatically finds where to sample the beyond-knee region of a limiter using
+// a greedy optimization algorithm that iteratively decreases the approximation
+// error.
+// The solution is sub-optimal because the algorithm is greedy and the points
+// are assigned by halving intervals (starting with the whole beyond-knee region
+// as a single interval). However, even if sub-optimal, this algorithm works
+// well in practice and it is efficiently implemented using priority queues.
+std::vector<double> SampleLimiterRegion(const LimiterDbGainCurve* limiter) {
+  static_assert(kInterpolatedGainCurveBeyondKneePoints > 2, "");
+
+  struct Interval {
+    Interval() = default;  // Ctor required by std::priority_queue.
+    Interval(double l, double r, double e) : x0(l), x1(r), error(e) {
+      RTC_CHECK(x0 < x1);
+    }
+    bool operator<(const Interval& other) const { return error < other.error; }
+
+    double x0;
+    double x1;
+    double error;
+  };
+
+  std::priority_queue<Interval, std::vector<Interval>> q;
+  q.emplace(limiter->limiter_start_linear(), limiter->max_input_level_linear(),
+            LimiterUnderApproximationNegativeError(
+                limiter, limiter->limiter_start_linear(),
+                limiter->max_input_level_linear()));
+
+  // Iteratively find points by halving the interval with greatest error.
+  while (q.size() < kInterpolatedGainCurveBeyondKneePoints) {
+    // Get the interval with highest error.
+    const auto interval = q.top();
+    q.pop();
+
+    // Split `interval` and enqueue.
+    double x_split = (interval.x0 + interval.x1) / 2.0;
+    q.emplace(interval.x0, x_split,
+              LimiterUnderApproximationNegativeError(limiter, interval.x0,
+                                                     x_split));  // Left.
+    q.emplace(x_split, interval.x1,
+              LimiterUnderApproximationNegativeError(limiter, x_split,
+                                                     interval.x1));  // Right.
+  }
+
+  // Copy x1 values and sort them.
+  RTC_CHECK_EQ(q.size(), kInterpolatedGainCurveBeyondKneePoints);
+  std::vector<double> samples(kInterpolatedGainCurveBeyondKneePoints);
+  for (size_t i = 0; i < kInterpolatedGainCurveBeyondKneePoints; ++i) {
+    const auto interval = q.top();
+    q.pop();
+    samples[i] = interval.x1;
+  }
+  RTC_CHECK(q.empty());
+  std::sort(samples.begin(), samples.end());
+
+  return samples;
+}
+
+// Compute the parameters to over-approximate the knee region via linear
+// interpolation. Over-approximating is saturation-safe since the knee region is
+// convex.
+void PrecomputeKneeApproxParams(const LimiterDbGainCurve* limiter,
+                                test::InterpolatedParameters* parameters) {
+  static_assert(kInterpolatedGainCurveKneePoints > 2, "");
+  // Get `kInterpolatedGainCurveKneePoints` - 1 equally spaced points.
+  const std::vector<double> points = test::LinSpace(
+      limiter->knee_start_linear(), limiter->limiter_start_linear(),
+      kInterpolatedGainCurveKneePoints - 1);
+
+  // Set the first two points. The second is computed to help with the beginning
+  // of the knee region, which has high curvature.
+  parameters->computed_approximation_params_x[0] = points[0];
+  parameters->computed_approximation_params_x[1] =
+      (points[0] + points[1]) / 2.0;
+  // Copy the remaining points.
+  std::copy(std::begin(points) + 1, std::end(points),
+            std::begin(parameters->computed_approximation_params_x) + 2);
+
+  // Compute (m, q) pairs for each linear piece y = mx + q.
+  for (size_t i = 0; i < kInterpolatedGainCurveKneePoints - 1; ++i) {
+    const double x0 = parameters->computed_approximation_params_x[i];
+    const double x1 = parameters->computed_approximation_params_x[i + 1];
+    const double y0 = limiter->GetGainLinear(x0);
+    const double y1 = limiter->GetGainLinear(x1);
+    RTC_CHECK_NE(x1, x0);
+    parameters->computed_approximation_params_m[i] = (y1 - y0) / (x1 - x0);
+    parameters->computed_approximation_params_q[i] =
+        y0 - parameters->computed_approximation_params_m[i] * x0;
+  }
+}
+
+// Compute the parameters to under-approximate the beyond-knee region via linear
+// interpolation and greedy sampling. Under-approximating is saturation-safe
+// since the beyond-knee region is concave.
+void PrecomputeBeyondKneeApproxParams(
+    const LimiterDbGainCurve* limiter,
+    test::InterpolatedParameters* parameters) {
+  // Find points on which the linear pieces are tangent to the gain curve.
+  const auto samples = SampleLimiterRegion(limiter);
+
+  // Parametrize each linear piece.
+  double m, q;
+  std::tie(m, q) = ComputeLinearApproximationParams(
+      limiter,
+      parameters
+          ->computed_approximation_params_x[kInterpolatedGainCurveKneePoints -
+                                            1]);
+  parameters
+      ->computed_approximation_params_m[kInterpolatedGainCurveKneePoints - 1] =
+      m;
+  parameters
+      ->computed_approximation_params_q[kInterpolatedGainCurveKneePoints - 1] =
+      q;
+  for (size_t i = 0; i < samples.size(); ++i) {
+    std::tie(m, q) = ComputeLinearApproximationParams(limiter, samples[i]);
+    parameters
+        ->computed_approximation_params_m[i +
+                                          kInterpolatedGainCurveKneePoints] = m;
+    parameters
+        ->computed_approximation_params_q[i +
+                                          kInterpolatedGainCurveKneePoints] = q;
+  }
+
+  // Find the point of intersection between adjacent linear pieces. They will be
+  // used as boundaries between adjacent linear pieces.
+  for (size_t i = kInterpolatedGainCurveKneePoints;
+       i < kInterpolatedGainCurveKneePoints +
+               kInterpolatedGainCurveBeyondKneePoints;
+       ++i) {
+    RTC_CHECK_NE(parameters->computed_approximation_params_m[i],
+                 parameters->computed_approximation_params_m[i - 1]);
+    parameters->computed_approximation_params_x[i] =
+        (  // Formula: (q0 - q1) / (m1 - m0).
+            parameters->computed_approximation_params_q[i - 1] -
+            parameters->computed_approximation_params_q[i]) /
+        (parameters->computed_approximation_params_m[i] -
+         parameters->computed_approximation_params_m[i - 1]);
+  }
+}
+
+}  // namespace
+
+namespace test {
+
+InterpolatedParameters ComputeInterpolatedGainCurveApproximationParams() {
+  InterpolatedParameters parameters;
+  LimiterDbGainCurve limiter;
+  parameters.computed_approximation_params_x.fill(0.0f);
+  parameters.computed_approximation_params_m.fill(0.0f);
+  parameters.computed_approximation_params_q.fill(0.0f);
+  PrecomputeKneeApproxParams(&limiter, &parameters);
+  PrecomputeBeyondKneeApproxParams(&limiter, &parameters);
+  return parameters;
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.h b/third_party/libwebrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.h
new file mode 100644
index 0000000000..08b676f5fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_COMPUTE_INTERPOLATED_GAIN_CURVE_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_COMPUTE_INTERPOLATED_GAIN_CURVE_H_
+
+#include <array>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+
+namespace webrtc {
+
+namespace test {
+
+// Parameters for interpolated gain curve using under-approximation to
+// avoid saturation.
+//
+// The saturation gain is defined in order to let hard-clipping occur for
+// those samples having a level that falls in the saturation region. It is an
+// upper bound of the actual gain to apply - i.e., that returned by the
+// limiter.
+
+// Knee and beyond-knee regions approximation parameters.
+// The gain curve is approximated as a piece-wise linear function.
+// `approx_params_x_` are the boundaries between adjacent linear pieces,
+// `approx_params_m_` and `approx_params_q_` are the slope and the y-intercept
+// values of each piece.
+struct InterpolatedParameters {
+  std::array<float, kInterpolatedGainCurveTotalPoints>
+      computed_approximation_params_x;
+  std::array<float, kInterpolatedGainCurveTotalPoints>
+      computed_approximation_params_m;
+  std::array<float, kInterpolatedGainCurveTotalPoints>
+      computed_approximation_params_q;
+};
+
+InterpolatedParameters ComputeInterpolatedGainCurveApproximationParams();
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_COMPUTE_INTERPOLATED_GAIN_CURVE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features.cc b/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features.cc
new file mode 100644
index 0000000000..cced7614bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/cpu_features.h"
+
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+
+namespace webrtc {
+
+std::string AvailableCpuFeatures::ToString() const {
+  char buf[64];
+  rtc::SimpleStringBuilder builder(buf);
+  bool first = true;
+  if (sse2) {
+    builder << (first ? "SSE2" : "_SSE2");
+    first = false;
+  }
+  if (avx2) {
+    builder << (first ? "AVX2" : "_AVX2");
+    first = false;
+  }
+  if (neon) {
+    builder << (first ? "NEON" : "_NEON");
+    first = false;
+  }
+  if (first) {
+    return "none";
+  }
+  return builder.str();
+}
+
+// Detects available CPU features.
+AvailableCpuFeatures GetAvailableCpuFeatures() {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+  return {/*sse2=*/GetCPUInfo(kSSE2) != 0,
+          /*avx2=*/GetCPUInfo(kAVX2) != 0,
+          /*neon=*/false};
+#elif defined(WEBRTC_HAS_NEON)
+  return {/*sse2=*/false,
+          /*avx2=*/false,
+          /*neon=*/true};
+#else
+  return {/*sse2=*/false,
+          /*avx2=*/false,
+          /*neon=*/false};
+#endif
+}
+
+AvailableCpuFeatures NoAvailableCpuFeatures() {
+  return {/*sse2=*/false, /*avx2=*/false, /*neon=*/false};
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features.h b/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features.h
new file mode 100644
index 0000000000..54ddfb3055
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_CPU_FEATURES_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_CPU_FEATURES_H_
+
+#include <string>
+
+namespace webrtc {
+
+// Collection of flags indicating which CPU features are available on the
+// current platform. True means available.
+struct AvailableCpuFeatures {
+  AvailableCpuFeatures(bool sse2, bool avx2, bool neon)
+      : sse2(sse2), avx2(avx2), neon(neon) {}
+  // Intel.
+  bool sse2;
+  bool avx2;
+  // ARM.
+  bool neon;
+  std::string ToString() const;
+};
+
+// Detects what CPU features are available.
+AvailableCpuFeatures GetAvailableCpuFeatures();
+
+// Returns the CPU feature flags all set to false.
+AvailableCpuFeatures NoAvailableCpuFeatures();
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_CPU_FEATURES_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features_gn/moz.build
new file mode 100644
index 0000000000..a4572251ad
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/cpu_features.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("cpu_features_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_gn/moz.build
new file mode 100644
index 0000000000..3a54fc3171
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_gn/moz.build
@@ -0,0 +1,235 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/limiter.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("fixed_digital_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
new file mode 100644
index 0000000000..1995b24913
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kInitialFilterStateLevel = 0.0f;
+
+// Instant attack.
+constexpr float kAttackFilterConstant = 0.0f;
+
+// Limiter decay constant.
+// Computed as `10 ** (-1/20 * subframe_duration / kDecayMs)` where:
+// - `subframe_duration` is `kFrameDurationMs / kSubFramesInFrame`;
+// - `kDecayMs` is defined in agc2_testing_common.h.
+constexpr float kDecayFilterConstant = 0.9971259f;
+
+}  // namespace
+
+FixedDigitalLevelEstimator::FixedDigitalLevelEstimator(
+    int sample_rate_hz,
+    ApmDataDumper* apm_data_dumper)
+    : apm_data_dumper_(apm_data_dumper),
+      filter_state_level_(kInitialFilterStateLevel) {
+  SetSampleRate(sample_rate_hz);
+  CheckParameterCombination();
+  RTC_DCHECK(apm_data_dumper_);
+  apm_data_dumper_->DumpRaw("agc2_level_estimator_samplerate", sample_rate_hz);
+}
+
+void FixedDigitalLevelEstimator::CheckParameterCombination() {
+  RTC_DCHECK_GT(samples_in_frame_, 0);
+  RTC_DCHECK_LE(kSubFramesInFrame, samples_in_frame_);
+  RTC_DCHECK_EQ(samples_in_frame_ % kSubFramesInFrame, 0);
+  RTC_DCHECK_GT(samples_in_sub_frame_, 1);
+}
+
+std::array<float, kSubFramesInFrame> FixedDigitalLevelEstimator::ComputeLevel(
+    const AudioFrameView<const float>& float_frame) {
+  RTC_DCHECK_GT(float_frame.num_channels(), 0);
+  RTC_DCHECK_EQ(float_frame.samples_per_channel(), samples_in_frame_);
+
+  // Compute max envelope without smoothing.
+  std::array<float, kSubFramesInFrame> envelope{};
+  for (int channel_idx = 0; channel_idx < float_frame.num_channels();
+       ++channel_idx) {
+    const auto channel = float_frame.channel(channel_idx);
+    for (int sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
+      for (int sample_in_sub_frame = 0;
+           sample_in_sub_frame < samples_in_sub_frame_; ++sample_in_sub_frame) {
+        envelope[sub_frame] =
+            std::max(envelope[sub_frame],
+                     std::abs(channel[sub_frame * samples_in_sub_frame_ +
+                                      sample_in_sub_frame]));
+      }
+    }
+  }
+
+  // Make sure envelope increases happen one step earlier so that the
+  // corresponding *gain decrease* doesn't miss a sudden signal
+  // increase due to interpolation.
+  for (int sub_frame = 0; sub_frame < kSubFramesInFrame - 1; ++sub_frame) {
+    if (envelope[sub_frame] < envelope[sub_frame + 1]) {
+      envelope[sub_frame] = envelope[sub_frame + 1];
+    }
+  }
+
+  // Add attack / decay smoothing.
+  for (int sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
+    const float envelope_value = envelope[sub_frame];
+    if (envelope_value > filter_state_level_) {
+      envelope[sub_frame] = envelope_value * (1 - kAttackFilterConstant) +
+                            filter_state_level_ * kAttackFilterConstant;
+    } else {
+      envelope[sub_frame] = envelope_value * (1 - kDecayFilterConstant) +
+                            filter_state_level_ * kDecayFilterConstant;
+    }
+    filter_state_level_ = envelope[sub_frame];
+
+    // Dump data for debug.
+    RTC_DCHECK(apm_data_dumper_);
+    const auto channel = float_frame.channel(0);
+    apm_data_dumper_->DumpRaw("agc2_level_estimator_samples",
+                              samples_in_sub_frame_,
+                              &channel[sub_frame * samples_in_sub_frame_]);
+    apm_data_dumper_->DumpRaw("agc2_level_estimator_level",
+                              envelope[sub_frame]);
+  }
+
+  return envelope;
+}
+
+void FixedDigitalLevelEstimator::SetSampleRate(int sample_rate_hz) {
+  samples_in_frame_ =
+      rtc::CheckedDivExact(sample_rate_hz * kFrameDurationMs, 1000);
+  samples_in_sub_frame_ =
+      rtc::CheckedDivExact(samples_in_frame_, kSubFramesInFrame);
+  CheckParameterCombination();
+}
+
+void FixedDigitalLevelEstimator::Reset() {
+  filter_state_level_ = kInitialFilterStateLevel;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h
new file mode 100644
index 0000000000..d26b55950c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
+
+#include <array>
+#include <vector>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+// Produces a smooth signal level estimate from an input audio
+// stream. The estimate smoothing is done through exponential
+// filtering.
+class FixedDigitalLevelEstimator {
+ public:
+  // Sample rates are allowed if the number of samples in a frame
+  // (sample_rate_hz * kFrameDurationMs / 1000) is divisible by
+  // kSubFramesInSample. For kFrameDurationMs=10 and
+  // kSubFramesInSample=20, this means that sample_rate_hz has to be
+  // divisible by 2000.
+  FixedDigitalLevelEstimator(int sample_rate_hz,
+                             ApmDataDumper* apm_data_dumper);
+
+  FixedDigitalLevelEstimator(const FixedDigitalLevelEstimator&) = delete;
+  FixedDigitalLevelEstimator& operator=(const FixedDigitalLevelEstimator&) =
+      delete;
+
+  // The input is assumed to be in FloatS16 format. Scaled input will
+  // produce similarly scaled output. A frame of with kFrameDurationMs
+  // ms of audio produces a level estimates in the same scale. The
+  // level estimate contains kSubFramesInFrame values.
+  std::array<float, kSubFramesInFrame> ComputeLevel(
+      const AudioFrameView<const float>& float_frame);
+
+  // Rate may be changed at any time (but not concurrently) from the
+  // value passed to the constructor. The class is not thread safe.
+  void SetSampleRate(int sample_rate_hz);
+
+  // Resets the level estimator internal state.
+  void Reset();
+
+  float LastAudioLevel() const { return filter_state_level_; }
+
+ private:
+  void CheckParameterCombination();
+
+  ApmDataDumper* const apm_data_dumper_ = nullptr;
+  float filter_state_level_;
+  int samples_in_frame_;
+  int samples_in_sub_frame_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator_unittest.cc
new file mode 100644
index 0000000000..97b421d04c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/fixed_digital_level_estimator_unittest.cc
@@ -0,0 +1,159 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
+
+#include <limits>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+#include "modules/audio_processing/agc2/vector_float_frame.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kInputLevel = 10000.f;
+
+// Run audio at specified settings through the level estimator, and
+// verify that the output level falls within the bounds.
+void TestLevelEstimator(int sample_rate_hz,
+                        int num_channels,
+                        float input_level_linear_scale,
+                        float expected_min,
+                        float expected_max) {
+  ApmDataDumper apm_data_dumper(0);
+  FixedDigitalLevelEstimator level_estimator(sample_rate_hz, &apm_data_dumper);
+
+  const VectorFloatFrame vectors_with_float_frame(
+      num_channels, rtc::CheckedDivExact(sample_rate_hz, 100),
+      input_level_linear_scale);
+
+  for (int i = 0; i < 500; ++i) {
+    const auto level = level_estimator.ComputeLevel(
+        vectors_with_float_frame.float_frame_view());
+
+    // Give the estimator some time to ramp up.
+    if (i < 50) {
+      continue;
+    }
+
+    for (const auto& x : level) {
+      EXPECT_LE(expected_min, x);
+      EXPECT_LE(x, expected_max);
+    }
+  }
+}
+
+// Returns time it takes for the level estimator to decrease its level
+// estimate by 'level_reduction_db'.
+float TimeMsToDecreaseLevel(int sample_rate_hz,
+                            int num_channels,
+                            float input_level_db,
+                            float level_reduction_db) {
+  const float input_level = DbfsToFloatS16(input_level_db);
+  RTC_DCHECK_GT(level_reduction_db, 0);
+
+  const VectorFloatFrame vectors_with_float_frame(
+      num_channels, rtc::CheckedDivExact(sample_rate_hz, 100), input_level);
+
+  ApmDataDumper apm_data_dumper(0);
+  FixedDigitalLevelEstimator level_estimator(sample_rate_hz, &apm_data_dumper);
+
+  // Give the LevelEstimator plenty of time to ramp up and stabilize
+  float last_level = 0.f;
+  for (int i = 0; i < 500; ++i) {
+    const auto level_envelope = level_estimator.ComputeLevel(
+        vectors_with_float_frame.float_frame_view());
+    last_level = *level_envelope.rbegin();
+  }
+
+  // Set input to 0.
+  VectorFloatFrame vectors_with_zero_float_frame(
+      num_channels, rtc::CheckedDivExact(sample_rate_hz, 100), 0);
+
+  const float reduced_level_linear =
+      DbfsToFloatS16(input_level_db - level_reduction_db);
+  int sub_frames_until_level_reduction = 0;
+  while (last_level > reduced_level_linear) {
+    const auto level_envelope = level_estimator.ComputeLevel(
+        vectors_with_zero_float_frame.float_frame_view());
+    for (const auto& v : level_envelope) {
+      EXPECT_LT(v, last_level);
+      sub_frames_until_level_reduction++;
+      last_level = v;
+      if (last_level <= reduced_level_linear) {
+        break;
+      }
+    }
+  }
+  return static_cast<float>(sub_frames_until_level_reduction) *
+         kFrameDurationMs / kSubFramesInFrame;
+}
+}  // namespace
+
+TEST(GainController2FixedDigitalLevelEstimator, EstimatorShouldNotCrash) {
+  TestLevelEstimator(8000, 1, 0, std::numeric_limits<float>::lowest(),
+                     std::numeric_limits<float>::max());
+}
+
+TEST(GainController2FixedDigitalLevelEstimator,
+     EstimatorShouldEstimateConstantLevel) {
+  TestLevelEstimator(10000, 1, kInputLevel, kInputLevel * 0.99,
+                     kInputLevel * 1.01);
+}
+
+TEST(GainController2FixedDigitalLevelEstimator,
+     EstimatorShouldEstimateConstantLevelForManyChannels) {
+  constexpr size_t num_channels = 10;
+  TestLevelEstimator(20000, num_channels, kInputLevel, kInputLevel * 0.99,
+                     kInputLevel * 1.01);
+}
+
+TEST(GainController2FixedDigitalLevelEstimator, TimeToDecreaseForLowLevel) {
+  constexpr float kLevelReductionDb = 25;
+  constexpr float kInitialLowLevel = -40;
+  constexpr float kExpectedTime = kLevelReductionDb * test::kDecayMs;
+
+  const float time_to_decrease =
+      TimeMsToDecreaseLevel(22000, 1, kInitialLowLevel, kLevelReductionDb);
+
+  EXPECT_LE(kExpectedTime * 0.9, time_to_decrease);
+  EXPECT_LE(time_to_decrease, kExpectedTime * 1.1);
+}
+
+TEST(GainController2FixedDigitalLevelEstimator,
+     TimeToDecreaseForFullScaleLevel) {
+  constexpr float kLevelReductionDb = 25;
+  constexpr float kExpectedTime = kLevelReductionDb * test::kDecayMs;
+
+  const float time_to_decrease =
+      TimeMsToDecreaseLevel(26000, 1, 0, kLevelReductionDb);
+
+  EXPECT_LE(kExpectedTime * 0.9, time_to_decrease);
+  EXPECT_LE(time_to_decrease, kExpectedTime * 1.1);
+}
+
+TEST(GainController2FixedDigitalLevelEstimator,
+     TimeToDecreaseForMultipleChannels) {
+  constexpr float kLevelReductionDb = 25;
+  constexpr float kExpectedTime = kLevelReductionDb * test::kDecayMs;
+  constexpr size_t kNumChannels = 10;
+
+  const float time_to_decrease =
+      TimeMsToDecreaseLevel(28000, kNumChannels, 0, kLevelReductionDb);
+
+  EXPECT_LE(kExpectedTime * 0.9, time_to_decrease);
+  EXPECT_LE(time_to_decrease, kExpectedTime * 1.1);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc
new file mode 100644
index 0000000000..f9e276d3a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/gain_applier.h"
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+
+// Returns true when the gain factor is so close to 1 that it would
+// not affect int16 samples.
+bool GainCloseToOne(float gain_factor) {
+  return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
+         gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
+}
+
+void ClipSignal(AudioFrameView<float> signal) {
+  for (int k = 0; k < signal.num_channels(); ++k) {
+    rtc::ArrayView<float> channel_view = signal.channel(k);
+    for (auto& sample : channel_view) {
+      sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
+    }
+  }
+}
+
+void ApplyGainWithRamping(float last_gain_linear,
+                          float gain_at_end_of_frame_linear,
+                          float inverse_samples_per_channel,
+                          AudioFrameView<float> float_frame) {
+  // Do not modify the signal.
+  if (last_gain_linear == gain_at_end_of_frame_linear &&
+      GainCloseToOne(gain_at_end_of_frame_linear)) {
+    return;
+  }
+
+  // Gain is constant and different from 1.
+  if (last_gain_linear == gain_at_end_of_frame_linear) {
+    for (int k = 0; k < float_frame.num_channels(); ++k) {
+      rtc::ArrayView<float> channel_view = float_frame.channel(k);
+      for (auto& sample : channel_view) {
+        sample *= gain_at_end_of_frame_linear;
+      }
+    }
+    return;
+  }
+
+  // The gain changes. We have to change slowly to avoid discontinuities.
+  const float increment = (gain_at_end_of_frame_linear - last_gain_linear) *
+                          inverse_samples_per_channel;
+  float gain = last_gain_linear;
+  for (int i = 0; i < float_frame.samples_per_channel(); ++i) {
+    for (int ch = 0; ch < float_frame.num_channels(); ++ch) {
+      float_frame.channel(ch)[i] *= gain;
+    }
+    gain += increment;
+  }
+}
+
+}  // namespace
+
+GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
+    : hard_clip_samples_(hard_clip_samples),
+      last_gain_factor_(initial_gain_factor),
+      current_gain_factor_(initial_gain_factor) {}
+
+void GainApplier::ApplyGain(AudioFrameView<float> signal) {
+  if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) {
+    Initialize(signal.samples_per_channel());
+  }
+
+  ApplyGainWithRamping(last_gain_factor_, current_gain_factor_,
+                       inverse_samples_per_channel_, signal);
+
+  last_gain_factor_ = current_gain_factor_;
+
+  if (hard_clip_samples_) {
+    ClipSignal(signal);
+  }
+}
+
+// TODO(bugs.webrtc.org/7494): Remove once switched to gains in dB.
+void GainApplier::SetGainFactor(float gain_factor) {
+  RTC_DCHECK_GT(gain_factor, 0.f);
+  current_gain_factor_ = gain_factor;
+}
+
+void GainApplier::Initialize(int samples_per_channel) {
+  RTC_DCHECK_GT(samples_per_channel, 0);
+  samples_per_channel_ = static_cast<int>(samples_per_channel);
+  inverse_samples_per_channel_ = 1.f / samples_per_channel_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.h b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.h
new file mode 100644
index 0000000000..ba8a4a4cd2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
+
+#include <stddef.h>
+
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+class GainApplier {
+ public:
+  GainApplier(bool hard_clip_samples, float initial_gain_factor);
+
+  void ApplyGain(AudioFrameView<float> signal);
+  void SetGainFactor(float gain_factor);
+  float GetGainFactor() const { return current_gain_factor_; }
+
+ private:
+  void Initialize(int samples_per_channel);
+
+  // Whether to clip samples after gain is applied. If 'true', result
+  // will fit in FloatS16 range.
+  const bool hard_clip_samples_;
+  float last_gain_factor_;
+
+  // If this value is not equal to 'last_gain_factor', gain will be
+  // ramped from 'last_gain_factor_' to this value during the next
+  // 'ApplyGain'.
+  float current_gain_factor_;
+  int samples_per_channel_ = -1;
+  float inverse_samples_per_channel_ = -1.f;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_gn/moz.build
new file mode 100644
index 0000000000..394aa109fa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("gain_applier_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_unittest.cc
new file mode 100644
index 0000000000..3296345e62
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier_unittest.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/gain_applier.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <limits>
+
+#include "modules/audio_processing/agc2/vector_float_frame.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+TEST(AutomaticGainController2GainApplier, InitialGainIsRespected) {
+  constexpr float initial_signal_level = 123.f;
+  constexpr float gain_factor = 10.f;
+  VectorFloatFrame fake_audio(1, 1, initial_signal_level);
+  GainApplier gain_applier(true, gain_factor);
+
+  gain_applier.ApplyGain(fake_audio.float_frame_view());
+  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
+              initial_signal_level * gain_factor, 0.1f);
+}
+
+TEST(AutomaticGainController2GainApplier, ClippingIsDone) {
+  constexpr float initial_signal_level = 30000.f;
+  constexpr float gain_factor = 10.f;
+  VectorFloatFrame fake_audio(1, 1, initial_signal_level);
+  GainApplier gain_applier(true, gain_factor);
+
+  gain_applier.ApplyGain(fake_audio.float_frame_view());
+  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
+              std::numeric_limits<int16_t>::max(), 0.1f);
+}
+
+TEST(AutomaticGainController2GainApplier, ClippingIsNotDone) {
+  constexpr float initial_signal_level = 30000.f;
+  constexpr float gain_factor = 10.f;
+  VectorFloatFrame fake_audio(1, 1, initial_signal_level);
+  GainApplier gain_applier(false, gain_factor);
+
+  gain_applier.ApplyGain(fake_audio.float_frame_view());
+
+  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
+              initial_signal_level * gain_factor, 0.1f);
+}
+
+TEST(AutomaticGainController2GainApplier, RampingIsDone) {
+  constexpr float initial_signal_level = 30000.f;
+  constexpr float initial_gain_factor = 1.f;
+  constexpr float target_gain_factor = 0.5f;
+  constexpr int num_channels = 3;
+  constexpr int samples_per_channel = 4;
+  VectorFloatFrame fake_audio(num_channels, samples_per_channel,
+                              initial_signal_level);
+  GainApplier gain_applier(false, initial_gain_factor);
+
+  gain_applier.SetGainFactor(target_gain_factor);
+  gain_applier.ApplyGain(fake_audio.float_frame_view());
+
+  // The maximal gain change should be close to that in linear interpolation.
+  for (size_t channel = 0; channel < num_channels; ++channel) {
+    float max_signal_change = 0.f;
+    float last_signal_level = initial_signal_level;
+    for (const auto sample : fake_audio.float_frame_view().channel(channel)) {
+      const float current_change = fabs(last_signal_level - sample);
+      max_signal_change = std::max(max_signal_change, current_change);
+      last_signal_level = sample;
+    }
+    const float total_gain_change =
+        fabs((initial_gain_factor - target_gain_factor) * initial_signal_level);
+    EXPECT_NEAR(max_signal_change, total_gain_change / samples_per_channel,
+                0.1f);
+  }
+
+  // Next frame should have the desired level.
+  VectorFloatFrame next_fake_audio_frame(num_channels, samples_per_channel,
+                                         initial_signal_level);
+  gain_applier.ApplyGain(next_fake_audio_frame.float_frame_view());
+
+  // The last sample should have the new gain.
+  EXPECT_NEAR(next_fake_audio_frame.float_frame_view().channel(0)[0],
+              initial_signal_level * target_gain_factor, 0.1f);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_gn/moz.build
new file mode 100644
index 0000000000..5b1d68415a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("gain_map_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_internal.h b/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_internal.h
new file mode 100644
index 0000000000..7c669fc9dd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_map_internal.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_GAIN_MAP_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_MAP_INTERNAL_H_
+
+namespace webrtc {
+
+static constexpr int kGainMapSize = 256;
+// Maps input volumes, which are values in the [0, 255] range, to gains in dB.
+// The values below are generated with numpy as follows:
+// SI = 2                        # Initial slope.
+// SF = 0.25                     # Final slope.
+// D = 8/256                     # Quantization factor.
+// x = np.linspace(0, 255, 256)  # Input volumes.
+// y = (SF * x + (SI - SF) * (1 - np.exp(-D*x)) / D - 56).round()
+static const int kGainMap[kGainMapSize] = {
+    -56, -54, -52, -50, -48, -47, -45, -43, -42, -40, -38, -37, -35, -34, -33,
+    -31, -30, -29, -27, -26, -25, -24, -23, -22, -20, -19, -18, -17, -16, -15,
+    -14, -14, -13, -12, -11, -10, -9,  -8,  -8,  -7,  -6,  -5,  -5,  -4,  -3,
+    -2,  -2,  -1,  0,   0,   1,   1,   2,   3,   3,   4,   4,   5,   5,   6,
+    6,   7,   7,   8,   8,   9,   9,   10,  10,  11,  11,  12,  12,  13,  13,
+    13,  14,  14,  15,  15,  15,  16,  16,  17,  17,  17,  18,  18,  18,  19,
+    19,  19,  20,  20,  21,  21,  21,  22,  22,  22,  23,  23,  23,  24,  24,
+    24,  24,  25,  25,  25,  26,  26,  26,  27,  27,  27,  28,  28,  28,  28,
+    29,  29,  29,  30,  30,  30,  30,  31,  31,  31,  32,  32,  32,  32,  33,
+    33,  33,  33,  34,  34,  34,  35,  35,  35,  35,  36,  36,  36,  36,  37,
+    37,  37,  38,  38,  38,  38,  39,  39,  39,  39,  40,  40,  40,  40,  41,
+    41,  41,  41,  42,  42,  42,  42,  43,  43,  43,  44,  44,  44,  44,  45,
+    45,  45,  45,  46,  46,  46,  46,  47,  47,  47,  47,  48,  48,  48,  48,
+    49,  49,  49,  49,  50,  50,  50,  50,  51,  51,  51,  51,  52,  52,  52,
+    52,  53,  53,  53,  53,  54,  54,  54,  54,  55,  55,  55,  55,  56,  56,
+    56,  56,  57,  57,  57,  57,  58,  58,  58,  58,  59,  59,  59,  59,  60,
+    60,  60,  60,  61,  61,  61,  61,  62,  62,  62,  62,  63,  63,  63,  63,
+    64};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_GAIN_MAP_INTERNAL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.cc b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.cc
new file mode 100644
index 0000000000..bcc650fb3e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.cc
@@ -0,0 +1,580 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/input_volume_controller.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/gain_map_internal.h"
+#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+// Amount of error we tolerate in the microphone input volume (presumably due to
+// OS quantization) before we assume the user has manually adjusted the volume.
+constexpr int kVolumeQuantizationSlack = 25;
+
+constexpr int kMaxInputVolume = 255;
+static_assert(kGainMapSize > kMaxInputVolume, "gain map too small");
+
+// Maximum absolute RMS error.
+constexpr int KMaxAbsRmsErrorDbfs = 15;
+static_assert(KMaxAbsRmsErrorDbfs > 0, "");
+
+using Agc1ClippingPredictorConfig = AudioProcessing::Config::GainController1::
+    AnalogGainController::ClippingPredictor;
+
+// TODO(webrtc:7494): Hardcode clipping predictor parameters and remove this
+// function after no longer needed in the ctor.
+Agc1ClippingPredictorConfig CreateClippingPredictorConfig(bool enabled) {
+  Agc1ClippingPredictorConfig config;
+  config.enabled = enabled;
+
+  return config;
+}
+
+// Returns an input volume in the [`min_input_volume`, `kMaxInputVolume`] range
+// that reduces `gain_error_db`, which is a gain error estimated when
+// `input_volume` was applied, according to a fixed gain map.
+int ComputeVolumeUpdate(int gain_error_db,
+                        int input_volume,
+                        int min_input_volume) {
+  RTC_DCHECK_GE(input_volume, 0);
+  RTC_DCHECK_LE(input_volume, kMaxInputVolume);
+  if (gain_error_db == 0) {
+    return input_volume;
+  }
+
+  int new_volume = input_volume;
+  if (gain_error_db > 0) {
+    while (kGainMap[new_volume] - kGainMap[input_volume] < gain_error_db &&
+           new_volume < kMaxInputVolume) {
+      ++new_volume;
+    }
+  } else {
+    while (kGainMap[new_volume] - kGainMap[input_volume] > gain_error_db &&
+           new_volume > min_input_volume) {
+      --new_volume;
+    }
+  }
+  return new_volume;
+}
+
+// Returns the proportion of samples in the buffer which are at full-scale
+// (and presumably clipped).
+float ComputeClippedRatio(const float* const* audio,
+                          size_t num_channels,
+                          size_t samples_per_channel) {
+  RTC_DCHECK_GT(samples_per_channel, 0);
+  int num_clipped = 0;
+  for (size_t ch = 0; ch < num_channels; ++ch) {
+    int num_clipped_in_ch = 0;
+    for (size_t i = 0; i < samples_per_channel; ++i) {
+      RTC_DCHECK(audio[ch]);
+      if (audio[ch][i] >= 32767.0f || audio[ch][i] <= -32768.0f) {
+        ++num_clipped_in_ch;
+      }
+    }
+    num_clipped = std::max(num_clipped, num_clipped_in_ch);
+  }
+  return static_cast<float>(num_clipped) / (samples_per_channel);
+}
+
+void LogClippingMetrics(int clipping_rate) {
+  RTC_LOG(LS_INFO) << "[AGC2] Input clipping rate: " << clipping_rate << "%";
+  RTC_HISTOGRAM_COUNTS_LINEAR(/*name=*/"WebRTC.Audio.Agc.InputClippingRate",
+                              /*sample=*/clipping_rate, /*min=*/0, /*max=*/100,
+                              /*bucket_count=*/50);
+}
+
+// Compares `speech_level_dbfs` to the [`target_range_min_dbfs`,
+// `target_range_max_dbfs`] range and returns the error to be compensated via
+// input volume adjustment. Returns a positive value when the level is below
+// the range, a negative value when the level is above the range, zero
+// otherwise.
+int GetSpeechLevelRmsErrorDb(float speech_level_dbfs,
+                             int target_range_min_dbfs,
+                             int target_range_max_dbfs) {
+  constexpr float kMinSpeechLevelDbfs = -90.0f;
+  constexpr float kMaxSpeechLevelDbfs = 30.0f;
+  RTC_DCHECK_GE(speech_level_dbfs, kMinSpeechLevelDbfs);
+  RTC_DCHECK_LE(speech_level_dbfs, kMaxSpeechLevelDbfs);
+  speech_level_dbfs = rtc::SafeClamp<float>(
+      speech_level_dbfs, kMinSpeechLevelDbfs, kMaxSpeechLevelDbfs);
+
+  int rms_error_db = 0;
+  if (speech_level_dbfs > target_range_max_dbfs) {
+    rms_error_db = std::round(target_range_max_dbfs - speech_level_dbfs);
+  } else if (speech_level_dbfs < target_range_min_dbfs) {
+    rms_error_db = std::round(target_range_min_dbfs - speech_level_dbfs);
+  }
+
+  return rms_error_db;
+}
+
+}  // namespace
+
+MonoInputVolumeController::MonoInputVolumeController(
+    int min_input_volume_after_clipping,
+    int min_input_volume,
+    int update_input_volume_wait_frames,
+    float speech_probability_threshold,
+    float speech_ratio_threshold)
+    : min_input_volume_(min_input_volume),
+      min_input_volume_after_clipping_(min_input_volume_after_clipping),
+      max_input_volume_(kMaxInputVolume),
+      update_input_volume_wait_frames_(
+          std::max(update_input_volume_wait_frames, 1)),
+      speech_probability_threshold_(speech_probability_threshold),
+      speech_ratio_threshold_(speech_ratio_threshold) {
+  RTC_DCHECK_GE(min_input_volume_, 0);
+  RTC_DCHECK_LE(min_input_volume_, 255);
+  RTC_DCHECK_GE(min_input_volume_after_clipping_, 0);
+  RTC_DCHECK_LE(min_input_volume_after_clipping_, 255);
+  RTC_DCHECK_GE(max_input_volume_, 0);
+  RTC_DCHECK_LE(max_input_volume_, 255);
+  RTC_DCHECK_GE(update_input_volume_wait_frames_, 0);
+  RTC_DCHECK_GE(speech_probability_threshold_, 0.0f);
+  RTC_DCHECK_LE(speech_probability_threshold_, 1.0f);
+  RTC_DCHECK_GE(speech_ratio_threshold_, 0.0f);
+  RTC_DCHECK_LE(speech_ratio_threshold_, 1.0f);
+}
+
+MonoInputVolumeController::~MonoInputVolumeController() = default;
+
+void MonoInputVolumeController::Initialize() {
+  max_input_volume_ = kMaxInputVolume;
+  capture_output_used_ = true;
+  check_volume_on_next_process_ = true;
+  frames_since_update_input_volume_ = 0;
+  speech_frames_since_update_input_volume_ = 0;
+  is_first_frame_ = true;
+}
+
+// A speeh segment is considered active if at least
+// `update_input_volume_wait_frames_` new frames have been processed since the
+// previous update and the ratio of non-silence frames (i.e., frames with a
+// `speech_probability` higher than `speech_probability_threshold_`) is at least
+// `speech_ratio_threshold_`.
+void MonoInputVolumeController::Process(absl::optional<int> rms_error_db,
+                                        float speech_probability) {
+  if (check_volume_on_next_process_) {
+    check_volume_on_next_process_ = false;
+    // We have to wait until the first process call to check the volume,
+    // because Chromium doesn't guarantee it to be valid any earlier.
+    CheckVolumeAndReset();
+  }
+
+  // Count frames with a high speech probability as speech.
+  if (speech_probability >= speech_probability_threshold_) {
+    ++speech_frames_since_update_input_volume_;
+  }
+
+  // Reset the counters and maybe update the input volume.
+  if (++frames_since_update_input_volume_ >= update_input_volume_wait_frames_) {
+    const float speech_ratio =
+        static_cast<float>(speech_frames_since_update_input_volume_) /
+        static_cast<float>(update_input_volume_wait_frames_);
+
+    // Always reset the counters regardless of whether the volume changes or
+    // not.
+    frames_since_update_input_volume_ = 0;
+    speech_frames_since_update_input_volume_ = 0;
+
+    // Update the input volume if allowed.
+    if (!is_first_frame_ && speech_ratio >= speech_ratio_threshold_ &&
+        rms_error_db.has_value()) {
+      UpdateInputVolume(*rms_error_db);
+    }
+  }
+
+  is_first_frame_ = false;
+}
+
+void MonoInputVolumeController::HandleClipping(int clipped_level_step) {
+  RTC_DCHECK_GT(clipped_level_step, 0);
+  // Always decrease the maximum input volume, even if the current input volume
+  // is below threshold.
+  SetMaxLevel(std::max(min_input_volume_after_clipping_,
+                       max_input_volume_ - clipped_level_step));
+  if (log_to_histograms_) {
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.AgcClippingAdjustmentAllowed",
+                          last_recommended_input_volume_ - clipped_level_step >=
+                              min_input_volume_after_clipping_);
+  }
+  if (last_recommended_input_volume_ > min_input_volume_after_clipping_) {
+    // Don't try to adjust the input volume if we're already below the limit. As
+    // a consequence, if the user has brought the input volume above the limit,
+    // we will still not react until the postproc updates the input volume.
+    SetInputVolume(
+        std::max(min_input_volume_after_clipping_,
+                 last_recommended_input_volume_ - clipped_level_step));
+    frames_since_update_input_volume_ = 0;
+    speech_frames_since_update_input_volume_ = 0;
+    is_first_frame_ = false;
+  }
+}
+
+void MonoInputVolumeController::SetInputVolume(int new_volume) {
+  int applied_input_volume = recommended_input_volume_;
+  if (applied_input_volume == 0) {
+    RTC_DLOG(LS_INFO)
+        << "[AGC2] The applied input volume is zero, taking no action.";
+    return;
+  }
+  if (applied_input_volume < 0 || applied_input_volume > kMaxInputVolume) {
+    RTC_LOG(LS_ERROR) << "[AGC2] Invalid value for the applied input volume: "
+                      << applied_input_volume;
+    return;
+  }
+
+  // Detect manual input volume adjustments by checking if the
+  // `applied_input_volume` is outside of the `[last_recommended_input_volume_ -
+  // kVolumeQuantizationSlack, last_recommended_input_volume_ +
+  // kVolumeQuantizationSlack]` range.
+  if (applied_input_volume >
+          last_recommended_input_volume_ + kVolumeQuantizationSlack ||
+      applied_input_volume <
+          last_recommended_input_volume_ - kVolumeQuantizationSlack) {
+    RTC_DLOG(LS_INFO)
+        << "[AGC2] The input volume was manually adjusted. Updating "
+           "stored input volume from "
+        << last_recommended_input_volume_ << " to " << applied_input_volume;
+    last_recommended_input_volume_ = applied_input_volume;
+    // Always allow the user to increase the volume.
+    if (last_recommended_input_volume_ > max_input_volume_) {
+      SetMaxLevel(last_recommended_input_volume_);
+    }
+    // Take no action in this case, since we can't be sure when the volume
+    // was manually adjusted.
+    frames_since_update_input_volume_ = 0;
+    speech_frames_since_update_input_volume_ = 0;
+    is_first_frame_ = false;
+    return;
+  }
+
+  new_volume = std::min(new_volume, max_input_volume_);
+  if (new_volume == last_recommended_input_volume_) {
+    return;
+  }
+
+  recommended_input_volume_ = new_volume;
+  RTC_DLOG(LS_INFO) << "[AGC2] Applied input volume: " << applied_input_volume
+                    << " | last recommended input volume: "
+                    << last_recommended_input_volume_
+                    << " | newly recommended input volume: " << new_volume;
+  last_recommended_input_volume_ = new_volume;
+}
+
+void MonoInputVolumeController::SetMaxLevel(int input_volume) {
+  RTC_DCHECK_GE(input_volume, min_input_volume_after_clipping_);
+  max_input_volume_ = input_volume;
+  RTC_DLOG(LS_INFO) << "[AGC2] Maximum input volume updated: "
+                    << max_input_volume_;
+}
+
+void MonoInputVolumeController::HandleCaptureOutputUsedChange(
+    bool capture_output_used) {
+  if (capture_output_used_ == capture_output_used) {
+    return;
+  }
+  capture_output_used_ = capture_output_used;
+
+  if (capture_output_used) {
+    // When we start using the output, we should reset things to be safe.
+    check_volume_on_next_process_ = true;
+  }
+}
+
+int MonoInputVolumeController::CheckVolumeAndReset() {
+  int input_volume = recommended_input_volume_;
+  // Reasons for taking action at startup:
+  // 1) A person starting a call is expected to be heard.
+  // 2) Independent of interpretation of `input_volume` == 0 we should raise it
+  // so the AGC can do its job properly.
+  if (input_volume == 0 && !startup_) {
+    RTC_DLOG(LS_INFO)
+        << "[AGC2] The applied input volume is zero, taking no action.";
+    return 0;
+  }
+  if (input_volume < 0 || input_volume > kMaxInputVolume) {
+    RTC_LOG(LS_ERROR) << "[AGC2] Invalid value for the applied input volume: "
+                      << input_volume;
+    return -1;
+  }
+  RTC_DLOG(LS_INFO) << "[AGC2] Initial input volume: " << input_volume;
+
+  if (input_volume < min_input_volume_) {
+    input_volume = min_input_volume_;
+    RTC_DLOG(LS_INFO)
+        << "[AGC2] The initial input volume is too low, raising to "
+        << input_volume;
+    recommended_input_volume_ = input_volume;
+  }
+
+  last_recommended_input_volume_ = input_volume;
+  startup_ = false;
+  frames_since_update_input_volume_ = 0;
+  speech_frames_since_update_input_volume_ = 0;
+  is_first_frame_ = true;
+
+  return 0;
+}
+
+void MonoInputVolumeController::UpdateInputVolume(int rms_error_db) {
+  RTC_DLOG(LS_INFO) << "[AGC2] RMS error: " << rms_error_db << " dB";
+  // Prevent too large microphone input volume changes by clamping the RMS
+  // error.
+  rms_error_db =
+      rtc::SafeClamp(rms_error_db, -KMaxAbsRmsErrorDbfs, KMaxAbsRmsErrorDbfs);
+  if (rms_error_db == 0) {
+    return;
+  }
+  SetInputVolume(ComputeVolumeUpdate(
+      rms_error_db, last_recommended_input_volume_, min_input_volume_));
+}
+
+InputVolumeController::InputVolumeController(int num_capture_channels,
+                                             const Config& config)
+    : num_capture_channels_(num_capture_channels),
+      min_input_volume_(config.min_input_volume),
+      capture_output_used_(true),
+      clipped_level_step_(config.clipped_level_step),
+      clipped_ratio_threshold_(config.clipped_ratio_threshold),
+      clipped_wait_frames_(config.clipped_wait_frames),
+      clipping_predictor_(CreateClippingPredictor(
+          num_capture_channels,
+          CreateClippingPredictorConfig(config.enable_clipping_predictor))),
+      use_clipping_predictor_step_(
+          !!clipping_predictor_ &&
+          CreateClippingPredictorConfig(config.enable_clipping_predictor)
+              .use_predicted_step),
+      frames_since_clipped_(config.clipped_wait_frames),
+      clipping_rate_log_counter_(0),
+      clipping_rate_log_(0.0f),
+      target_range_max_dbfs_(config.target_range_max_dbfs),
+      target_range_min_dbfs_(config.target_range_min_dbfs),
+      channel_controllers_(num_capture_channels) {
+  RTC_LOG(LS_INFO)
+      << "[AGC2] Input volume controller enabled. Minimum input volume: "
+      << min_input_volume_;
+
+  for (auto& controller : channel_controllers_) {
+    controller = std::make_unique<MonoInputVolumeController>(
+        config.clipped_level_min, min_input_volume_,
+        config.update_input_volume_wait_frames,
+        config.speech_probability_threshold, config.speech_ratio_threshold);
+  }
+
+  RTC_DCHECK(!channel_controllers_.empty());
+  RTC_DCHECK_GT(clipped_level_step_, 0);
+  RTC_DCHECK_LE(clipped_level_step_, 255);
+  RTC_DCHECK_GT(clipped_ratio_threshold_, 0.0f);
+  RTC_DCHECK_LT(clipped_ratio_threshold_, 1.0f);
+  RTC_DCHECK_GT(clipped_wait_frames_, 0);
+  channel_controllers_[0]->ActivateLogging();
+}
+
+InputVolumeController::~InputVolumeController() {}
+
+void InputVolumeController::Initialize() {
+  for (auto& controller : channel_controllers_) {
+    controller->Initialize();
+  }
+  capture_output_used_ = true;
+
+  AggregateChannelLevels();
+  clipping_rate_log_ = 0.0f;
+  clipping_rate_log_counter_ = 0;
+
+  applied_input_volume_ = absl::nullopt;
+}
+
+void InputVolumeController::AnalyzeInputAudio(int applied_input_volume,
+                                              const AudioBuffer& audio_buffer) {
+  RTC_DCHECK_GE(applied_input_volume, 0);
+  RTC_DCHECK_LE(applied_input_volume, 255);
+
+  SetAppliedInputVolume(applied_input_volume);
+
+  RTC_DCHECK_EQ(audio_buffer.num_channels(), channel_controllers_.size());
+  const float* const* audio = audio_buffer.channels_const();
+  size_t samples_per_channel = audio_buffer.num_frames();
+  RTC_DCHECK(audio);
+
+  AggregateChannelLevels();
+  if (!capture_output_used_) {
+    return;
+  }
+
+  if (!!clipping_predictor_) {
+    AudioFrameView<const float> frame = AudioFrameView<const float>(
+        audio, num_capture_channels_, static_cast<int>(samples_per_channel));
+    clipping_predictor_->Analyze(frame);
+  }
+
+  // Check for clipped samples. We do this in the preprocessing phase in order
+  // to catch clipped echo as well.
+  //
+  // If we find a sufficiently clipped frame, drop the current microphone
+  // input volume and enforce a new maximum input volume, dropped the same
+  // amount from the current maximum. This harsh treatment is an effort to avoid
+  // repeated clipped echo events.
+  float clipped_ratio =
+      ComputeClippedRatio(audio, num_capture_channels_, samples_per_channel);
+  clipping_rate_log_ = std::max(clipped_ratio, clipping_rate_log_);
+  clipping_rate_log_counter_++;
+  constexpr int kNumFramesIn30Seconds = 3000;
+  if (clipping_rate_log_counter_ == kNumFramesIn30Seconds) {
+    LogClippingMetrics(std::round(100.0f * clipping_rate_log_));
+    clipping_rate_log_ = 0.0f;
+    clipping_rate_log_counter_ = 0;
+  }
+
+  if (frames_since_clipped_ < clipped_wait_frames_) {
+    ++frames_since_clipped_;
+    return;
+  }
+
+  const bool clipping_detected = clipped_ratio > clipped_ratio_threshold_;
+  bool clipping_predicted = false;
+  int predicted_step = 0;
+  if (!!clipping_predictor_) {
+    for (int channel = 0; channel < num_capture_channels_; ++channel) {
+      const auto step = clipping_predictor_->EstimateClippedLevelStep(
+          channel, recommended_input_volume_, clipped_level_step_,
+          channel_controllers_[channel]->min_input_volume_after_clipping(),
+          kMaxInputVolume);
+      if (step.has_value()) {
+        predicted_step = std::max(predicted_step, step.value());
+        clipping_predicted = true;
+      }
+    }
+  }
+
+  if (clipping_detected) {
+    RTC_DLOG(LS_INFO) << "[AGC2] Clipping detected (ratio: " << clipped_ratio
+                      << ")";
+  }
+
+  int step = clipped_level_step_;
+  if (clipping_predicted) {
+    predicted_step = std::max(predicted_step, clipped_level_step_);
+    RTC_DLOG(LS_INFO) << "[AGC2] Clipping predicted (volume down step: "
+                      << predicted_step << ")";
+    if (use_clipping_predictor_step_) {
+      step = predicted_step;
+    }
+  }
+
+  if (clipping_detected ||
+      (clipping_predicted && use_clipping_predictor_step_)) {
+    for (auto& state_ch : channel_controllers_) {
+      state_ch->HandleClipping(step);
+    }
+    frames_since_clipped_ = 0;
+    if (!!clipping_predictor_) {
+      clipping_predictor_->Reset();
+    }
+  }
+
+  AggregateChannelLevels();
+}
+
+absl::optional<int> InputVolumeController::RecommendInputVolume(
+    float speech_probability,
+    absl::optional<float> speech_level_dbfs) {
+  // Only process if applied input volume is set.
+  if (!applied_input_volume_.has_value()) {
+    RTC_LOG(LS_ERROR) << "[AGC2] Applied input volume not set.";
+    return absl::nullopt;
+  }
+
+  AggregateChannelLevels();
+  const int volume_after_clipping_handling = recommended_input_volume_;
+
+  if (!capture_output_used_) {
+    return applied_input_volume_;
+  }
+
+  absl::optional<int> rms_error_db;
+  if (speech_level_dbfs.has_value()) {
+    // Compute the error for all frames (both speech and non-speech frames).
+    rms_error_db = GetSpeechLevelRmsErrorDb(
+        *speech_level_dbfs, target_range_min_dbfs_, target_range_max_dbfs_);
+  }
+
+  for (auto& controller : channel_controllers_) {
+    controller->Process(rms_error_db, speech_probability);
+  }
+
+  AggregateChannelLevels();
+  if (volume_after_clipping_handling != recommended_input_volume_) {
+    // The recommended input volume was adjusted in order to match the target
+    // level.
+    UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(
+        recommended_input_volume_);
+  }
+
+  applied_input_volume_ = absl::nullopt;
+  return recommended_input_volume();
+}
+
+void InputVolumeController::HandleCaptureOutputUsedChange(
+    bool capture_output_used) {
+  for (auto& controller : channel_controllers_) {
+    controller->HandleCaptureOutputUsedChange(capture_output_used);
+  }
+
+  capture_output_used_ = capture_output_used;
+}
+
+void InputVolumeController::SetAppliedInputVolume(int input_volume) {
+  applied_input_volume_ = input_volume;
+
+  for (auto& controller : channel_controllers_) {
+    controller->set_stream_analog_level(input_volume);
+  }
+
+  AggregateChannelLevels();
+}
+
+void InputVolumeController::AggregateChannelLevels() {
+  int new_recommended_input_volume =
+      channel_controllers_[0]->recommended_analog_level();
+  channel_controlling_gain_ = 0;
+  for (size_t ch = 1; ch < channel_controllers_.size(); ++ch) {
+    int input_volume = channel_controllers_[ch]->recommended_analog_level();
+    if (input_volume < new_recommended_input_volume) {
+      new_recommended_input_volume = input_volume;
+      channel_controlling_gain_ = static_cast<int>(ch);
+    }
+  }
+
+  // Enforce the minimum input volume when a recommendation is made.
+  if (applied_input_volume_.has_value() && *applied_input_volume_ > 0) {
+    new_recommended_input_volume =
+        std::max(new_recommended_input_volume, min_input_volume_);
+  }
+
+  recommended_input_volume_ = new_recommended_input_volume;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.h b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.h
new file mode 100644
index 0000000000..21405542dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.h
@@ -0,0 +1,282 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/clipping_predictor.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/gtest_prod_util.h"
+
+namespace webrtc {
+
+class MonoInputVolumeController;
+
+// The input volume controller recommends what volume to use, handles volume
+// changes and clipping detection and prediction. In particular, it handles
+// changes triggered by the user (e.g., volume set to zero by a HW mute button).
+// This class is not thread-safe.
+// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
+// convention.
+class InputVolumeController final {
+ public:
+  // Config for the constructor.
+  struct Config {
+    // Minimum input volume that can be recommended. Not enforced when the
+    // applied input volume is zero outside startup.
+    int min_input_volume = 20;
+    // Lowest input volume level that will be applied in response to clipping.
+    int clipped_level_min = 70;
+    // Amount input volume level is lowered with every clipping event. Limited
+    // to (0, 255].
+    int clipped_level_step = 15;
+    // Proportion of clipped samples required to declare a clipping event.
+    // Limited to (0.0f, 1.0f).
+    float clipped_ratio_threshold = 0.1f;
+    // Time in frames to wait after a clipping event before checking again.
+    // Limited to values higher than 0.
+    int clipped_wait_frames = 300;
+    // Enables clipping prediction functionality.
+    bool enable_clipping_predictor = false;
+    // Speech level target range (dBFS). If the speech level is in the range
+    // [`target_range_min_dbfs`, `target_range_max_dbfs`], no input volume
+    // adjustments are done based on the speech level. For speech levels below
+    // and above the range, the targets `target_range_min_dbfs` and
+    // `target_range_max_dbfs` are used, respectively.
+    int target_range_max_dbfs = -30;
+    int target_range_min_dbfs = -50;
+    // Number of wait frames between the recommended input volume updates.
+    int update_input_volume_wait_frames = 100;
+    // Speech probability threshold: speech probabilities below the threshold
+    // are considered silence. Limited to [0.0f, 1.0f].
+    float speech_probability_threshold = 0.7f;
+    // Minimum speech frame ratio for volume updates to be allowed. Limited to
+    // [0.0f, 1.0f].
+    float speech_ratio_threshold = 0.6f;
+  };
+
+  // Ctor. `num_capture_channels` specifies the number of channels for the audio
+  // passed to `AnalyzePreProcess()` and `Process()`. Clamps
+  // `config.startup_min_level` in the [12, 255] range.
+  InputVolumeController(int num_capture_channels, const Config& config);
+
+  ~InputVolumeController();
+  InputVolumeController(const InputVolumeController&) = delete;
+  InputVolumeController& operator=(const InputVolumeController&) = delete;
+
+  // TODO(webrtc:7494): Integrate initialization into ctor and remove.
+  void Initialize();
+
+  // Analyzes `audio_buffer` before `RecommendInputVolume()` is called so tha
+  // the analysis can be performed before digital processing operations take
+  // place (e.g., echo cancellation). The analysis consists of input clipping
+  // detection and prediction (if enabled).
+  void AnalyzeInputAudio(int applied_input_volume,
+                         const AudioBuffer& audio_buffer);
+
+  // Adjusts the recommended input volume upwards/downwards based on the result
+  // of `AnalyzeInputAudio()` and on `speech_level_dbfs` (if specified). Must
+  // be called after `AnalyzeInputAudio()`.  The value of `speech_probability`
+  // is expected to be in the range [0, 1] and `speech_level_dbfs` in the range
+  // [-90, 30] and both should be estimated after echo cancellation and noise
+  // suppression are applied. Returns a non-empty input volume recommendation if
+  // available. If `capture_output_used_` is true, returns the applied input
+  // volume.
+  absl::optional<int> RecommendInputVolume(
+      float speech_probability,
+      absl::optional<float> speech_level_dbfs);
+
+  // Stores whether the capture output will be used or not. Call when the
+  // capture stream output has been flagged to be used/not-used. If unused, the
+  // controller disregards all incoming audio.
+  void HandleCaptureOutputUsedChange(bool capture_output_used);
+
+  // Returns true if clipping prediction is enabled.
+  // TODO(bugs.webrtc.org/7494): Deprecate this method.
+  bool clipping_predictor_enabled() const { return !!clipping_predictor_; }
+
+  // Returns true if clipping prediction is used to adjust the input volume.
+  // TODO(bugs.webrtc.org/7494): Deprecate this method.
+  bool use_clipping_predictor_step() const {
+    return use_clipping_predictor_step_;
+  }
+
+  // Only use for testing: Use `RecommendInputVolume()` elsewhere.
+  // Returns the value of a member variable, needed for testing
+  // `AnalyzeInputAudio()`.
+  int recommended_input_volume() const { return recommended_input_volume_; }
+
+  // Only use for testing.
+  bool capture_output_used() const { return capture_output_used_; }
+
+ private:
+  friend class InputVolumeControllerTestHelper;
+
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest, MinInputVolumeDefault);
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest, MinInputVolumeDisabled);
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
+                           MinInputVolumeOutOfRangeAbove);
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
+                           MinInputVolumeOutOfRangeBelow);
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest, MinInputVolumeEnabled50);
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
+                           ClippingParametersVerified);
+
+  // Sets the applied input volume and resets the recommended input volume.
+  void SetAppliedInputVolume(int level);
+
+  void AggregateChannelLevels();
+
+  const int num_capture_channels_;
+
+  // Minimum input volume that can be recommended.
+  const int min_input_volume_;
+
+  // TODO(bugs.webrtc.org/7494): Once
+  // `AudioProcessingImpl::recommended_stream_analog_level()` becomes a trivial
+  // getter, leave uninitialized.
+  // Recommended input volume. After `SetAppliedInputVolume()` is called it
+  // holds holds the observed input volume. Possibly updated by
+  // `AnalyzePreProcess()` and `Process()`; after these calls, holds the
+  // recommended input volume.
+  int recommended_input_volume_ = 0;
+  // Applied input volume. After `SetAppliedInputVolume()` is called it holds
+  // the current applied volume.
+  absl::optional<int> applied_input_volume_;
+
+  bool capture_output_used_;
+
+  // Clipping detection and prediction.
+  const int clipped_level_step_;
+  const float clipped_ratio_threshold_;
+  const int clipped_wait_frames_;
+  const std::unique_ptr<ClippingPredictor> clipping_predictor_;
+  const bool use_clipping_predictor_step_;
+  int frames_since_clipped_;
+  int clipping_rate_log_counter_;
+  float clipping_rate_log_;
+
+  // Target range minimum and maximum. If the seech level is in the range
+  // [`target_range_min_dbfs`, `target_range_max_dbfs`], no volume adjustments
+  // take place. Instead, the digital gain controller is assumed to adapt to
+  // compensate for the speech level RMS error.
+  const int target_range_max_dbfs_;
+  const int target_range_min_dbfs_;
+
+  // Channel controllers updating the gain upwards/downwards.
+  std::vector<std::unique_ptr<MonoInputVolumeController>> channel_controllers_;
+  int channel_controlling_gain_ = 0;
+};
+
+// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
+// convention.
+class MonoInputVolumeController {
+ public:
+  MonoInputVolumeController(int min_input_volume_after_clipping,
+                            int min_input_volume,
+                            int update_input_volume_wait_frames,
+                            float speech_probability_threshold,
+                            float speech_ratio_threshold);
+  ~MonoInputVolumeController();
+  MonoInputVolumeController(const MonoInputVolumeController&) = delete;
+  MonoInputVolumeController& operator=(const MonoInputVolumeController&) =
+      delete;
+
+  void Initialize();
+  void HandleCaptureOutputUsedChange(bool capture_output_used);
+
+  // Sets the current input volume.
+  void set_stream_analog_level(int input_volume) {
+    recommended_input_volume_ = input_volume;
+  }
+
+  // Lowers the recommended input volume in response to clipping based on the
+  // suggested reduction `clipped_level_step`. Must be called after
+  // `set_stream_analog_level()`.
+  void HandleClipping(int clipped_level_step);
+
+  // TODO(bugs.webrtc.org/7494): Rename, audio not passed to the method anymore.
+  // Adjusts the recommended input volume upwards/downwards depending on the
+  // result of `HandleClipping()` and on `rms_error_dbfs`. Updates are only
+  // allowed for active speech segments and when `rms_error_dbfs` is not empty.
+  // Must be called after `HandleClipping()`.
+  void Process(absl::optional<int> rms_error_dbfs, float speech_probability);
+
+  // Returns the recommended input volume. Must be called after `Process()`.
+  int recommended_analog_level() const { return recommended_input_volume_; }
+
+  void ActivateLogging() { log_to_histograms_ = true; }
+
+  int min_input_volume_after_clipping() const {
+    return min_input_volume_after_clipping_;
+  }
+
+  // Only used for testing.
+  int min_input_volume() const { return min_input_volume_; }
+
+ private:
+  // Sets a new input volume, after first checking that it hasn't been updated
+  // by the user, in which case no action is taken.
+  void SetInputVolume(int new_volume);
+
+  // Sets the maximum input volume that the input volume controller is allowed
+  // to apply. The volume must be at least `kClippedLevelMin`.
+  void SetMaxLevel(int level);
+
+  int CheckVolumeAndReset();
+
+  // Updates the recommended input volume. If the volume slider needs to be
+  // moved, we check first if the user has adjusted it, in which case we take no
+  // action and cache the updated level.
+  void UpdateInputVolume(int rms_error_dbfs);
+
+  const int min_input_volume_;
+  const int min_input_volume_after_clipping_;
+  int max_input_volume_;
+
+  int last_recommended_input_volume_ = 0;
+
+  bool capture_output_used_ = true;
+  bool check_volume_on_next_process_ = true;
+  bool startup_ = true;
+
+  // TODO(bugs.webrtc.org/7494): Create a separate member for the applied
+  // input volume.
+  // Recommended input volume. After `set_stream_analog_level()` is
+  // called, it holds the observed applied input volume. Possibly updated by
+  // `HandleClipping()` and `Process()`; after these calls, holds the
+  // recommended input volume.
+  int recommended_input_volume_ = 0;
+
+  bool log_to_histograms_ = false;
+
+  // Counters for frames and speech frames since the last update in the
+  // recommended input volume.
+  const int update_input_volume_wait_frames_;
+  int frames_since_update_input_volume_ = 0;
+  int speech_frames_since_update_input_volume_ = 0;
+  bool is_first_frame_ = true;
+
+  // Speech probability threshold for a frame to be considered speech (instead
+  // of silence). Limited to [0.0f, 1.0f].
+  const float speech_probability_threshold_;
+  // Minimum ratio of speech frames. Limited to [0.0f, 1.0f].
+  const float speech_ratio_threshold_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_gn/moz.build
new file mode 100644
index 0000000000..582eb326f3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_gn/moz.build
@@ -0,0 +1,234 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("input_volume_controller_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_unittest.cc
new file mode 100644
index 0000000000..d1bdcf25a5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_controller_unittest.cc
@@ -0,0 +1,1857 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/input_volume_controller.h"
+
+#include <algorithm>
+#include <fstream>
+#include <limits>
+#include <string>
+#include <vector>
+
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/metrics.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::DoAll;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+namespace webrtc {
+namespace {
+
+constexpr int kSampleRateHz = 32000;
+constexpr int kNumChannels = 1;
+constexpr int kInitialInputVolume = 128;
+constexpr int kClippedMin = 165;  // Arbitrary, but different from the default.
+constexpr float kAboveClippedThreshold = 0.2f;
+constexpr int kMinMicLevel = 20;
+constexpr int kClippedLevelStep = 15;
+constexpr float kClippedRatioThreshold = 0.1f;
+constexpr int kClippedWaitFrames = 300;
+constexpr float kHighSpeechProbability = 0.7f;
+constexpr float kLowSpeechProbability = 0.1f;
+constexpr float kSpeechLevel = -25.0f;
+constexpr float kSpeechProbabilityThreshold = 0.5f;
+constexpr float kSpeechRatioThreshold = 0.8f;
+
+constexpr float kMinSample = std::numeric_limits<int16_t>::min();
+constexpr float kMaxSample = std::numeric_limits<int16_t>::max();
+
+using ClippingPredictorConfig = AudioProcessing::Config::GainController1::
+    AnalogGainController::ClippingPredictor;
+
+using InputVolumeControllerConfig = InputVolumeController::Config;
+
+constexpr ClippingPredictorConfig kDefaultClippingPredictorConfig{};
+
+std::unique_ptr<InputVolumeController> CreateInputVolumeController(
+    int clipped_level_step = kClippedLevelStep,
+    float clipped_ratio_threshold = kClippedRatioThreshold,
+    int clipped_wait_frames = kClippedWaitFrames,
+    bool enable_clipping_predictor = false,
+    int update_input_volume_wait_frames = 0) {
+  InputVolumeControllerConfig config{
+      .min_input_volume = kMinMicLevel,
+      .clipped_level_min = kClippedMin,
+      .clipped_level_step = clipped_level_step,
+      .clipped_ratio_threshold = clipped_ratio_threshold,
+      .clipped_wait_frames = clipped_wait_frames,
+      .enable_clipping_predictor = enable_clipping_predictor,
+      .target_range_max_dbfs = -18,
+      .target_range_min_dbfs = -30,
+      .update_input_volume_wait_frames = update_input_volume_wait_frames,
+      .speech_probability_threshold = kSpeechProbabilityThreshold,
+      .speech_ratio_threshold = kSpeechRatioThreshold,
+  };
+
+  return std::make_unique<InputVolumeController>(/*num_capture_channels=*/1,
+                                                 config);
+}
+
+// (Over)writes `samples_value` for the samples in `audio_buffer`.
+// When `clipped_ratio`, a value in [0, 1], is greater than 0, the corresponding
+// fraction of the frame is set to a full scale value to simulate clipping.
+void WriteAudioBufferSamples(float samples_value,
+                             float clipped_ratio,
+                             AudioBuffer& audio_buffer) {
+  RTC_DCHECK_GE(samples_value, kMinSample);
+  RTC_DCHECK_LE(samples_value, kMaxSample);
+  RTC_DCHECK_GE(clipped_ratio, 0.0f);
+  RTC_DCHECK_LE(clipped_ratio, 1.0f);
+  int num_channels = audio_buffer.num_channels();
+  int num_samples = audio_buffer.num_frames();
+  int num_clipping_samples = clipped_ratio * num_samples;
+  for (int ch = 0; ch < num_channels; ++ch) {
+    int i = 0;
+    for (; i < num_clipping_samples; ++i) {
+      audio_buffer.channels()[ch][i] = 32767.0f;
+    }
+    for (; i < num_samples; ++i) {
+      audio_buffer.channels()[ch][i] = samples_value;
+    }
+  }
+}
+
+// (Over)writes samples in `audio_buffer`. Alternates samples `samples_value`
+// and zero.
+void WriteAlternatingAudioBufferSamples(float samples_value,
+                                        AudioBuffer& audio_buffer) {
+  RTC_DCHECK_GE(samples_value, kMinSample);
+  RTC_DCHECK_LE(samples_value, kMaxSample);
+  const int num_channels = audio_buffer.num_channels();
+  const int num_frames = audio_buffer.num_frames();
+  for (int ch = 0; ch < num_channels; ++ch) {
+    for (int i = 0; i < num_frames; i += 2) {
+      audio_buffer.channels()[ch][i] = samples_value;
+      audio_buffer.channels()[ch][i + 1] = 0.0f;
+    }
+  }
+}
+
+// Reads a given number of 10 ms chunks from a PCM file and feeds them to
+// `InputVolumeController`.
+class SpeechSamplesReader {
+ private:
+  // Recording properties.
+  static constexpr int kPcmSampleRateHz = 16000;
+  static constexpr int kPcmNumChannels = 1;
+  static constexpr int kPcmBytesPerSamples = sizeof(int16_t);
+
+ public:
+  SpeechSamplesReader()
+      : is_(test::ResourcePath("audio_processing/agc/agc_audio", "pcm"),
+            std::ios::binary | std::ios::ate),
+        audio_buffer_(kPcmSampleRateHz,
+                      kPcmNumChannels,
+                      kPcmSampleRateHz,
+                      kPcmNumChannels,
+                      kPcmSampleRateHz,
+                      kPcmNumChannels),
+        buffer_(audio_buffer_.num_frames()),
+        buffer_num_bytes_(buffer_.size() * kPcmBytesPerSamples) {
+    RTC_CHECK(is_);
+  }
+
+  // Reads `num_frames` 10 ms frames from the beginning of the PCM file, applies
+  // `gain_db` and feeds the frames into `controller` by calling
+  // `AnalyzeInputAudio()` and `RecommendInputVolume()` for each frame. Reads
+  // the number of 10 ms frames available in the PCM file if `num_frames` is too
+  // large - i.e., does not loop. `speech_probability` and `speech_level_dbfs`
+  // are passed to `RecommendInputVolume()`.
+  int Feed(int num_frames,
+           int applied_input_volume,
+           int gain_db,
+           float speech_probability,
+           absl::optional<float> speech_level_dbfs,
+           InputVolumeController& controller) {
+    RTC_DCHECK(controller.capture_output_used());
+
+    float gain = std::pow(10.0f, gain_db / 20.0f);  // From dB to linear gain.
+    is_.seekg(0, is_.beg);  // Start from the beginning of the PCM file.
+
+    // Read and feed frames.
+    for (int i = 0; i < num_frames; ++i) {
+      is_.read(reinterpret_cast<char*>(buffer_.data()), buffer_num_bytes_);
+      if (is_.gcount() < buffer_num_bytes_) {
+        // EOF reached. Stop.
+        break;
+      }
+      // Apply gain and copy samples into `audio_buffer_`.
+      std::transform(buffer_.begin(), buffer_.end(),
+                     audio_buffer_.channels()[0], [gain](int16_t v) -> float {
+                       return rtc::SafeClamp(static_cast<float>(v) * gain,
+                                             kMinSample, kMaxSample);
+                     });
+      controller.AnalyzeInputAudio(applied_input_volume, audio_buffer_);
+      const auto recommended_input_volume = controller.RecommendInputVolume(
+          speech_probability, speech_level_dbfs);
+
+      // Expect no errors: Applied volume set for every frame;
+      // `RecommendInputVolume()` returns a non-empty value.
+      EXPECT_TRUE(recommended_input_volume.has_value());
+
+      applied_input_volume = *recommended_input_volume;
+    }
+    return applied_input_volume;
+  }
+
+ private:
+  std::ifstream is_;
+  AudioBuffer audio_buffer_;
+  std::vector<int16_t> buffer_;
+  const std::streamsize buffer_num_bytes_;
+};
+
+// Runs the MonoInputVolumeControl processing sequence following the API
+// contract. Returns the updated recommended input volume.
+float UpdateRecommendedInputVolume(MonoInputVolumeController& mono_controller,
+                                   int applied_input_volume,
+                                   float speech_probability,
+                                   absl::optional<float> rms_error_dbfs) {
+  mono_controller.set_stream_analog_level(applied_input_volume);
+  EXPECT_EQ(mono_controller.recommended_analog_level(), applied_input_volume);
+  mono_controller.Process(rms_error_dbfs, speech_probability);
+  return mono_controller.recommended_analog_level();
+}
+
+}  // namespace
+
+// TODO(bugs.webrtc.org/12874): Use constexpr struct with designated
+// initializers once fixed.
+constexpr InputVolumeControllerConfig GetInputVolumeControllerTestConfig() {
+  InputVolumeControllerConfig config{
+      .clipped_level_min = kClippedMin,
+      .clipped_level_step = kClippedLevelStep,
+      .clipped_ratio_threshold = kClippedRatioThreshold,
+      .clipped_wait_frames = kClippedWaitFrames,
+      .enable_clipping_predictor = kDefaultClippingPredictorConfig.enabled,
+      .target_range_max_dbfs = -18,
+      .target_range_min_dbfs = -30,
+      .update_input_volume_wait_frames = 0,
+      .speech_probability_threshold = 0.5f,
+      .speech_ratio_threshold = 1.0f,
+  };
+  return config;
+}
+
+// Helper class that provides an `InputVolumeController` instance with an
+// `AudioBuffer` instance and `CallAgcSequence()`, a helper method that runs the
+// `InputVolumeController` instance on the `AudioBuffer` one by sticking to the
+// API contract.
+class InputVolumeControllerTestHelper {
+ public:
+  // Ctor. Initializes `audio_buffer` with zeros.
+  // TODO(bugs.webrtc.org/7494): Remove the default argument.
+  InputVolumeControllerTestHelper(const InputVolumeController::Config& config =
+                                      GetInputVolumeControllerTestConfig())
+      : audio_buffer(kSampleRateHz,
+                     kNumChannels,
+                     kSampleRateHz,
+                     kNumChannels,
+                     kSampleRateHz,
+                     kNumChannels),
+        controller(/*num_capture_channels=*/1, config) {
+    controller.Initialize();
+    WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.0f,
+                            audio_buffer);
+  }
+
+  // Calls the sequence of `InputVolumeController` methods according to the API
+  // contract, namely:
+  // - Sets the applied input volume;
+  // - Uses `audio_buffer` to call `AnalyzeInputAudio()` and
+  // `RecommendInputVolume()`;
+  //  Returns the recommended input volume.
+  absl::optional<int> CallAgcSequence(int applied_input_volume,
+                                      float speech_probability,
+                                      absl::optional<float> speech_level_dbfs,
+                                      int num_calls = 1) {
+    RTC_DCHECK_GE(num_calls, 1);
+    absl::optional<int> volume = applied_input_volume;
+    for (int i = 0; i < num_calls; ++i) {
+      // Repeat the initial volume if `RecommendInputVolume()` doesn't return a
+      // value.
+      controller.AnalyzeInputAudio(volume.value_or(applied_input_volume),
+                                   audio_buffer);
+      volume = controller.RecommendInputVolume(speech_probability,
+                                               speech_level_dbfs);
+
+      // Allow deviation from the API contract: `RecommendInputVolume()` doesn't
+      // return a recommended input volume.
+      if (volume.has_value()) {
+        EXPECT_EQ(*volume, controller.recommended_input_volume());
+      }
+    }
+    return volume;
+  }
+
+  // Deprecated.
+  // TODO(bugs.webrtc.org/7494): Let the caller write `audio_buffer` and use
+  // `CallAgcSequence()`.
+  int CallRecommendInputVolume(int num_calls,
+                               int initial_volume,
+                               float speech_probability,
+                               absl::optional<float> speech_level_dbfs) {
+    RTC_DCHECK(controller.capture_output_used());
+
+    // Create non-clipping audio for `AnalyzeInputAudio()`.
+    WriteAlternatingAudioBufferSamples(0.1f * kMaxSample, audio_buffer);
+    int volume = initial_volume;
+    for (int i = 0; i < num_calls; ++i) {
+      controller.AnalyzeInputAudio(volume, audio_buffer);
+      const auto recommended_input_volume = controller.RecommendInputVolume(
+          speech_probability, speech_level_dbfs);
+
+      // Expect no errors: Applied volume set for every frame;
+      // `RecommendInputVolume()` returns a non-empty value.
+      EXPECT_TRUE(recommended_input_volume.has_value());
+
+      volume = *recommended_input_volume;
+    }
+    return volume;
+  }
+
+  // Deprecated.
+  // TODO(bugs.webrtc.org/7494): Let the caller write `audio_buffer` and use
+  // `CallAgcSequence()`.
+  void CallAnalyzeInputAudio(int num_calls, float clipped_ratio) {
+    RTC_DCHECK(controller.capture_output_used());
+
+    RTC_DCHECK_GE(clipped_ratio, 0.0f);
+    RTC_DCHECK_LE(clipped_ratio, 1.0f);
+    WriteAudioBufferSamples(/*samples_value=*/0.0f, clipped_ratio,
+                            audio_buffer);
+    for (int i = 0; i < num_calls; ++i) {
+      controller.AnalyzeInputAudio(controller.recommended_input_volume(),
+                                   audio_buffer);
+    }
+  }
+
+  AudioBuffer audio_buffer;
+  InputVolumeController controller;
+};
+
+class InputVolumeControllerChannelSampleRateTest
+    : public ::testing::TestWithParam<std::tuple<int, int>> {
+ protected:
+  int GetNumChannels() const { return std::get<0>(GetParam()); }
+  int GetSampleRateHz() const { return std::get<1>(GetParam()); }
+};
+
+TEST_P(InputVolumeControllerChannelSampleRateTest, CheckIsAlive) {
+  const int num_channels = GetNumChannels();
+  const int sample_rate_hz = GetSampleRateHz();
+
+  constexpr InputVolumeController::Config kConfig{.enable_clipping_predictor =
+                                                      true};
+  InputVolumeController controller(num_channels, kConfig);
+  controller.Initialize();
+  AudioBuffer buffer(sample_rate_hz, num_channels, sample_rate_hz, num_channels,
+                     sample_rate_hz, num_channels);
+
+  constexpr int kStartupVolume = 100;
+  int applied_initial_volume = kStartupVolume;
+
+  // Trigger a downward adaptation with clipping.
+  constexpr int kLevelWithinTargetDbfs =
+      (kConfig.target_range_min_dbfs + kConfig.target_range_max_dbfs) / 2;
+  WriteAlternatingAudioBufferSamples(/*samples_value=*/kMaxSample, buffer);
+  const int initial_volume1 = applied_initial_volume;
+  for (int i = 0; i < 400; ++i) {
+    controller.AnalyzeInputAudio(applied_initial_volume, buffer);
+    auto recommended_input_volume = controller.RecommendInputVolume(
+        kLowSpeechProbability,
+        /*speech_level_dbfs=*/kLevelWithinTargetDbfs);
+    ASSERT_TRUE(recommended_input_volume.has_value());
+    applied_initial_volume = *recommended_input_volume;
+  }
+  ASSERT_LT(controller.recommended_input_volume(), initial_volume1);
+
+  // Fill in audio that does not clip.
+  WriteAlternatingAudioBufferSamples(/*samples_value=*/1234.5f, buffer);
+
+  // Trigger an upward adaptation.
+  const int initial_volume2 = controller.recommended_input_volume();
+  for (int i = 0; i < kConfig.clipped_wait_frames; ++i) {
+    controller.AnalyzeInputAudio(applied_initial_volume, buffer);
+    auto recommended_input_volume = controller.RecommendInputVolume(
+        kHighSpeechProbability,
+        /*speech_level_dbfs=*/kConfig.target_range_min_dbfs - 5);
+    ASSERT_TRUE(recommended_input_volume.has_value());
+    applied_initial_volume = *recommended_input_volume;
+  }
+  EXPECT_GT(controller.recommended_input_volume(), initial_volume2);
+
+  // Trigger a downward adaptation.
+  const int initial_volume = controller.recommended_input_volume();
+  for (int i = 0; i < kConfig.update_input_volume_wait_frames; ++i) {
+    controller.AnalyzeInputAudio(applied_initial_volume, buffer);
+    auto recommended_input_volume = controller.RecommendInputVolume(
+        kHighSpeechProbability,
+        /*speech_level_dbfs=*/kConfig.target_range_max_dbfs + 5);
+    ASSERT_TRUE(recommended_input_volume.has_value());
+    applied_initial_volume = *recommended_input_volume;
+  }
+  EXPECT_LT(controller.recommended_input_volume(), initial_volume);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    ,
+    InputVolumeControllerChannelSampleRateTest,
+    ::testing::Combine(::testing::Values(1, 2, 3, 6),
+                       ::testing::Values(8000, 16000, 32000, 48000)));
+
+class InputVolumeControllerParametrizedTest
+    : public ::testing::TestWithParam<int> {};
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       StartupMinVolumeConfigurationRespectedWhenAppliedInputVolumeAboveMin) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = GetParam()});
+
+  EXPECT_EQ(*helper.CallAgcSequence(/*applied_input_volume=*/128,
+                                    /*speech_probability=*/0.9f,
+                                    /*speech_level_dbfs=*/-80),
+            128);
+}
+
+TEST_P(
+    InputVolumeControllerParametrizedTest,
+    StartupMinVolumeConfigurationRespectedWhenAppliedInputVolumeMaybeBelowMin) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = GetParam()});
+
+  EXPECT_GE(*helper.CallAgcSequence(/*applied_input_volume=*/10,
+                                    /*speech_probability=*/0.9f,
+                                    /*speech_level_dbfs=*/-80),
+            10);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       StartupMinVolumeRespectedWhenAppliedVolumeNonZero) {
+  const int kMinInputVolume = GetParam();
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = kMinInputVolume,
+                  .target_range_min_dbfs = -30,
+                  .update_input_volume_wait_frames = 1,
+                  .speech_probability_threshold = 0.5f,
+                  .speech_ratio_threshold = 0.5f});
+
+  // Volume change possible; speech level below the digital gain window.
+  int volume = *helper.CallAgcSequence(/*applied_input_volume=*/1,
+                                       /*speech_probability=*/0.9f,
+                                       /*speech_level_dbfs=*/-80);
+
+  EXPECT_EQ(volume, kMinInputVolume);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       MinVolumeRepeatedlyRespectedWhenAppliedVolumeNonZero) {
+  const int kMinInputVolume = GetParam();
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = kMinInputVolume,
+                  .target_range_min_dbfs = -30,
+                  .update_input_volume_wait_frames = 1,
+                  .speech_probability_threshold = 0.5f,
+                  .speech_ratio_threshold = 0.5f});
+
+  // Volume change possible; speech level below the digital gain window.
+  for (int i = 0; i < 100; ++i) {
+    const int volume = *helper.CallAgcSequence(/*applied_input_volume=*/1,
+                                               /*speech_probability=*/0.9f,
+                                               /*speech_level_dbfs=*/-80);
+    EXPECT_GE(volume, kMinInputVolume);
+  }
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       StartupMinVolumeRespectedOnceWhenAppliedVolumeZero) {
+  const int kMinInputVolume = GetParam();
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = kMinInputVolume,
+                  .target_range_min_dbfs = -30,
+                  .update_input_volume_wait_frames = 1,
+                  .speech_probability_threshold = 0.5f,
+                  .speech_ratio_threshold = 0.5f});
+
+  int volume = *helper.CallAgcSequence(/*applied_input_volume=*/0,
+                                       /*speech_probability=*/0.9f,
+                                       /*speech_level_dbfs=*/-80);
+
+  EXPECT_EQ(volume, kMinInputVolume);
+
+  // No change of volume regardless of a speech level below the digital gain
+  // window; applied volume is zero.
+  volume = *helper.CallAgcSequence(/*applied_input_volume=*/0,
+                                   /*speech_probability=*/0.9f,
+                                   /*speech_level_dbfs=*/-80);
+
+  EXPECT_EQ(volume, 0);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, MicVolumeResponseToRmsError) {
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(config);
+  int volume = *helper.CallAgcSequence(kInitialInputVolume,
+                                       kHighSpeechProbability, kSpeechLevel);
+
+  // Inside the digital gain's window; no change of volume.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -23.0f);
+
+  // Inside the digital gain's window; no change of volume.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -28.0f);
+
+  // Above the digital gain's  window; volume should be increased.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -29.0f);
+  EXPECT_EQ(volume, 128);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -38.0f);
+  EXPECT_EQ(volume, 156);
+
+  // Inside the digital gain's window; no change of volume.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -23.0f);
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -18.0f);
+
+  // Below the digial gain's window; volume should be decreased.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -17.0f);
+  EXPECT_EQ(volume, 155);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -17.0f);
+  EXPECT_EQ(volume, 151);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -9.0f);
+  EXPECT_EQ(volume, 119);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, MicVolumeIsLimited) {
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  const int min_input_volume = GetParam();
+  config.min_input_volume = min_input_volume;
+  InputVolumeControllerTestHelper helper(config);
+  int volume = *helper.CallAgcSequence(kInitialInputVolume,
+                                       kHighSpeechProbability, kSpeechLevel);
+
+  // Maximum upwards change is limited.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -48.0f);
+  EXPECT_EQ(volume, 183);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -48.0f);
+  EXPECT_EQ(volume, 243);
+
+  // Won't go higher than the maximum.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -48.0f);
+  EXPECT_EQ(volume, 255);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -17.0f);
+  EXPECT_EQ(volume, 254);
+
+  // Maximum downwards change is limited.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, 22.0f);
+  EXPECT_EQ(volume, 194);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, 22.0f);
+  EXPECT_EQ(volume, 137);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, 22.0f);
+  EXPECT_EQ(volume, 88);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, 22.0f);
+  EXPECT_EQ(volume, 54);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, 22.0f);
+  EXPECT_EQ(volume, 33);
+
+  // Won't go lower than the minimum.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, 22.0f);
+  EXPECT_EQ(volume, std::max(18, min_input_volume));
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, 22.0f);
+  EXPECT_EQ(volume, std::max(12, min_input_volume));
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, NoActionWhileMuted) {
+  InputVolumeControllerTestHelper helper_1(
+      /*config=*/{.min_input_volume = GetParam()});
+  InputVolumeControllerTestHelper helper_2(
+      /*config=*/{.min_input_volume = GetParam()});
+
+  int volume_1 = *helper_1.CallAgcSequence(/*applied_input_volume=*/255,
+                                           kHighSpeechProbability, kSpeechLevel,
+                                           /*num_calls=*/1);
+  int volume_2 = *helper_2.CallAgcSequence(/*applied_input_volume=*/255,
+                                           kHighSpeechProbability, kSpeechLevel,
+                                           /*num_calls=*/1);
+
+  EXPECT_EQ(volume_1, 255);
+  EXPECT_EQ(volume_2, 255);
+
+  helper_2.controller.HandleCaptureOutputUsedChange(false);
+
+  WriteAlternatingAudioBufferSamples(kMaxSample, helper_1.audio_buffer);
+  WriteAlternatingAudioBufferSamples(kMaxSample, helper_2.audio_buffer);
+
+  volume_1 =
+      *helper_1.CallAgcSequence(volume_1, kHighSpeechProbability, kSpeechLevel,
+                                /*num_calls=*/1);
+  volume_2 =
+      *helper_2.CallAgcSequence(volume_2, kHighSpeechProbability, kSpeechLevel,
+                                /*num_calls=*/1);
+
+  EXPECT_LT(volume_1, 255);
+  EXPECT_EQ(volume_2, 255);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       UnmutingChecksVolumeWithoutRaising) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = GetParam()});
+  helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.controller.HandleCaptureOutputUsedChange(false);
+  helper.controller.HandleCaptureOutputUsedChange(true);
+
+  constexpr int kInputVolume = 127;
+
+  // SetMicVolume should not be called.
+  EXPECT_EQ(
+      helper.CallRecommendInputVolume(/*num_calls=*/1, kInputVolume,
+                                      kHighSpeechProbability, kSpeechLevel),
+      kInputVolume);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, UnmutingRaisesTooLowVolume) {
+  const int min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = min_input_volume});
+  helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.controller.HandleCaptureOutputUsedChange(false);
+  helper.controller.HandleCaptureOutputUsedChange(true);
+
+  constexpr int kInputVolume = 11;
+
+  EXPECT_EQ(
+      helper.CallRecommendInputVolume(/*num_calls=*/1, kInputVolume,
+                                      kHighSpeechProbability, kSpeechLevel),
+      min_input_volume);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       ManualLevelChangeResultsInNoSetMicCall) {
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(config);
+  int volume = *helper.CallAgcSequence(kInitialInputVolume,
+                                       kHighSpeechProbability, kSpeechLevel);
+
+  // GetMicVolume returns a value outside of the quantization slack, indicating
+  // a manual volume change.
+  ASSERT_NE(volume, 154);
+  volume = helper.CallRecommendInputVolume(
+      /*num_calls=*/1, /*initial_volume=*/154, kHighSpeechProbability, -29.0f);
+  EXPECT_EQ(volume, 154);
+
+  // Do the same thing, except downwards now.
+  volume = helper.CallRecommendInputVolume(
+      /*num_calls=*/1, /*initial_volume=*/100, kHighSpeechProbability, -17.0f);
+  EXPECT_EQ(volume, 100);
+
+  // And finally verify the AGC continues working without a manual change.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -17.0f);
+  EXPECT_EQ(volume, 99);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       RecoveryAfterManualLevelChangeFromMax) {
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(config);
+  int volume = *helper.CallAgcSequence(kInitialInputVolume,
+                                       kHighSpeechProbability, kSpeechLevel);
+
+  // Force the mic up to max volume. Takes a few steps due to the residual
+  // gain limitation.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -48.0f);
+  EXPECT_EQ(volume, 183);
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -48.0f);
+  EXPECT_EQ(volume, 243);
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -48.0f);
+  EXPECT_EQ(volume, 255);
+
+  // Manual change does not result in SetMicVolume call.
+  volume = helper.CallRecommendInputVolume(
+      /*num_calls=*/1, /*initial_volume=*/50, kHighSpeechProbability, -17.0f);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 50);
+
+  // Continues working as usual afterwards.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -38.0f);
+
+  EXPECT_EQ(volume, 65);
+}
+
+// Checks that the minimum input volume is enforced during the upward adjustment
+// of the input volume.
+TEST_P(InputVolumeControllerParametrizedTest,
+       EnforceMinInputVolumeDuringUpwardsAdjustment) {
+  const int min_input_volume = GetParam();
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.min_input_volume = min_input_volume;
+  InputVolumeControllerTestHelper helper(config);
+  int volume = *helper.CallAgcSequence(kInitialInputVolume,
+                                       kHighSpeechProbability, kSpeechLevel);
+
+  // Manual change below min, but strictly positive, otherwise no action will be
+  // taken.
+  volume = helper.CallRecommendInputVolume(
+      /*num_calls=*/1, /*initial_volume=*/1, kHighSpeechProbability, -17.0f);
+
+  // Trigger an upward adjustment of the input volume.
+  EXPECT_EQ(volume, min_input_volume);
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -29.0f);
+  EXPECT_EQ(volume, min_input_volume);
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -30.0f);
+  EXPECT_EQ(volume, min_input_volume);
+
+  // After a number of consistently low speech level observations, the input
+  // volume is eventually raised above the minimum.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/10, volume,
+                                           kHighSpeechProbability, -38.0f);
+  EXPECT_GT(volume, min_input_volume);
+}
+
+// Checks that, when the min mic level override is specified, AGC immediately
+// applies the minimum mic level after the mic level is manually set below the
+// minimum gain to enforce.
+TEST_P(InputVolumeControllerParametrizedTest,
+       RecoveryAfterManualLevelChangeBelowMin) {
+  const int min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = min_input_volume});
+  int volume = *helper.CallAgcSequence(kInitialInputVolume,
+                                       kHighSpeechProbability, kSpeechLevel);
+
+  // Manual change below min, but strictly positive, otherwise
+  // AGC won't take any action.
+  volume = helper.CallRecommendInputVolume(
+      /*num_calls=*/1, /*initial_volume=*/1, kHighSpeechProbability, -17.0f);
+  EXPECT_EQ(volume, min_input_volume);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, NoClippingHasNoImpact) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = GetParam()});
+  helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/100, /*clipped_ratio=*/0);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 128);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       ClippingUnderThresholdHasNoImpact) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = GetParam()});
+  helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1, /*clipped_ratio=*/0.099);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 128);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, ClippingLowersVolume) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = GetParam()});
+  helper.CallAgcSequence(/*applied_input_volume=*/255, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1, /*clipped_ratio=*/0.2);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 240);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       WaitingPeriodBetweenClippingChecks) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = GetParam()});
+  helper.CallAgcSequence(/*applied_input_volume=*/255, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1,
+                               /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 240);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/300,
+                               /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 240);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1,
+                               /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 225);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, ClippingLoweringIsLimited) {
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(config);
+  helper.CallAgcSequence(/*applied_input_volume=*/180, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1,
+                               /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), kClippedMin);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1000,
+                               /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), kClippedMin);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       ClippingMaxIsRespectedWhenEqualToLevel) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = GetParam()});
+  helper.CallAgcSequence(/*applied_input_volume=*/255, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1,
+                               /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 240);
+
+  helper.CallRecommendInputVolume(/*num_calls=*/10, /*initial_volume=*/240,
+                                  kHighSpeechProbability, -48.0f);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 240);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       ClippingMaxIsRespectedWhenHigherThanLevel) {
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(config);
+  helper.CallAgcSequence(/*applied_input_volume=*/200, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1,
+                               /*clipped_ratio=*/kAboveClippedThreshold);
+  int volume = helper.controller.recommended_input_volume();
+  EXPECT_EQ(volume, 185);
+
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -58.0f);
+  EXPECT_EQ(volume, 240);
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/10, volume,
+                                           kHighSpeechProbability, -58.0f);
+  EXPECT_EQ(volume, 240);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, UserCanRaiseVolumeAfterClipping) {
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(config);
+  helper.CallAgcSequence(/*applied_input_volume=*/225, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1,
+                               /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 210);
+
+  // User changed the volume.
+  int volume = helper.CallRecommendInputVolume(
+      /*num_calls=*/1, /*initial_volume-*/ 250, kHighSpeechProbability, -32.0f);
+  EXPECT_EQ(volume, 250);
+
+  // Move down...
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -8.0f);
+  EXPECT_EQ(volume, 210);
+  // And back up to the new max established by the user.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -58.0f);
+  EXPECT_EQ(volume, 250);
+  // Will not move above new maximum.
+  volume = helper.CallRecommendInputVolume(/*num_calls=*/1, volume,
+                                           kHighSpeechProbability, -48.0f);
+  EXPECT_EQ(volume, 250);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       ClippingDoesNotPullLowVolumeBackUp) {
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(config);
+  helper.CallAgcSequence(/*applied_input_volume=*/80, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  int initial_volume = helper.controller.recommended_input_volume();
+  helper.CallAnalyzeInputAudio(/*num_calls=*/1,
+                               /*clipped_ratio=*/kAboveClippedThreshold);
+  EXPECT_EQ(helper.controller.recommended_input_volume(), initial_volume);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, TakesNoActionOnZeroMicVolume) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = GetParam()});
+  helper.CallAgcSequence(kInitialInputVolume, kHighSpeechProbability,
+                         kSpeechLevel);
+
+  EXPECT_EQ(
+      helper.CallRecommendInputVolume(/*num_calls=*/10, /*initial_volume=*/0,
+                                      kHighSpeechProbability, -48.0f),
+      0);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest, ClippingDetectionLowersVolume) {
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.min_input_volume = GetParam();
+  InputVolumeControllerTestHelper helper(config);
+  int volume = *helper.CallAgcSequence(/*applied_input_volume=*/255,
+                                       kHighSpeechProbability, kSpeechLevel,
+                                       /*num_calls=*/1);
+
+  EXPECT_EQ(volume, 255);
+
+  WriteAlternatingAudioBufferSamples(0.99f * kMaxSample, helper.audio_buffer);
+  volume = *helper.CallAgcSequence(volume, kHighSpeechProbability, kSpeechLevel,
+                                   /*num_calls=*/100);
+
+  EXPECT_EQ(volume, 255);
+
+  WriteAlternatingAudioBufferSamples(kMaxSample, helper.audio_buffer);
+  volume = *helper.CallAgcSequence(volume, kHighSpeechProbability, kSpeechLevel,
+                                   /*num_calls=*/100);
+
+  EXPECT_EQ(volume, 240);
+}
+
+// TODO(bugs.webrtc.org/12774): Test the bahavior of `clipped_level_step`.
+// TODO(bugs.webrtc.org/12774): Test the bahavior of `clipped_ratio_threshold`.
+// TODO(bugs.webrtc.org/12774): Test the bahavior of `clipped_wait_frames`.
+// Verifies that configurable clipping parameters are initialized as intended.
+TEST_P(InputVolumeControllerParametrizedTest, ClippingParametersVerified) {
+  std::unique_ptr<InputVolumeController> controller =
+      CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
+                                  kClippedWaitFrames);
+  controller->Initialize();
+  EXPECT_EQ(controller->clipped_level_step_, kClippedLevelStep);
+  EXPECT_EQ(controller->clipped_ratio_threshold_, kClippedRatioThreshold);
+  EXPECT_EQ(controller->clipped_wait_frames_, kClippedWaitFrames);
+  std::unique_ptr<InputVolumeController> controller_custom =
+      CreateInputVolumeController(/*clipped_level_step=*/10,
+                                  /*clipped_ratio_threshold=*/0.2f,
+                                  /*clipped_wait_frames=*/50);
+  controller_custom->Initialize();
+  EXPECT_EQ(controller_custom->clipped_level_step_, 10);
+  EXPECT_EQ(controller_custom->clipped_ratio_threshold_, 0.2f);
+  EXPECT_EQ(controller_custom->clipped_wait_frames_, 50);
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       DisableClippingPredictorDisablesClippingPredictor) {
+  std::unique_ptr<InputVolumeController> controller =
+      CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
+                                  kClippedWaitFrames,
+                                  /*enable_clipping_predictor=*/false);
+  controller->Initialize();
+
+  EXPECT_FALSE(controller->clipping_predictor_enabled());
+  EXPECT_FALSE(controller->use_clipping_predictor_step());
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       EnableClippingPredictorEnablesClippingPredictor) {
+  std::unique_ptr<InputVolumeController> controller =
+      CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
+                                  kClippedWaitFrames,
+                                  /*enable_clipping_predictor=*/true);
+  controller->Initialize();
+
+  EXPECT_TRUE(controller->clipping_predictor_enabled());
+  EXPECT_TRUE(controller->use_clipping_predictor_step());
+}
+
+TEST_P(InputVolumeControllerParametrizedTest,
+       DisableClippingPredictorDoesNotLowerVolume) {
+  int volume = 255;
+  InputVolumeControllerConfig config = GetInputVolumeControllerTestConfig();
+  config.enable_clipping_predictor = false;
+  auto helper = InputVolumeControllerTestHelper(config);
+  helper.controller.Initialize();
+
+  EXPECT_FALSE(helper.controller.clipping_predictor_enabled());
+  EXPECT_FALSE(helper.controller.use_clipping_predictor_step());
+
+  // Expect no change if clipping prediction is enabled.
+  for (int j = 0; j < 31; ++j) {
+    WriteAlternatingAudioBufferSamples(0.99f * kMaxSample, helper.audio_buffer);
+    volume =
+        *helper.CallAgcSequence(volume, kLowSpeechProbability, kSpeechLevel,
+                                /*num_calls=*/5);
+
+    WriteAudioBufferSamples(0.99f * kMaxSample, /*clipped_ratio=*/0.0f,
+                            helper.audio_buffer);
+    volume =
+        *helper.CallAgcSequence(volume, kLowSpeechProbability, kSpeechLevel,
+                                /*num_calls=*/5);
+
+    EXPECT_EQ(volume, 255);
+  }
+}
+
+// TODO(bugs.webrtc.org/7494): Split into several smaller tests.
+TEST_P(InputVolumeControllerParametrizedTest,
+       UsedClippingPredictionsProduceLowerAnalogLevels) {
+  constexpr int kInitialLevel = 255;
+  constexpr float kCloseToClippingPeakRatio = 0.99f;
+  int volume_1 = kInitialLevel;
+  int volume_2 = kInitialLevel;
+
+  // Create two helpers, one with clipping prediction and one without.
+  auto config_1 = GetInputVolumeControllerTestConfig();
+  auto config_2 = GetInputVolumeControllerTestConfig();
+  config_1.enable_clipping_predictor = true;
+  config_2.enable_clipping_predictor = false;
+  auto helper_1 = InputVolumeControllerTestHelper(config_1);
+  auto helper_2 = InputVolumeControllerTestHelper(config_2);
+  helper_1.controller.Initialize();
+  helper_2.controller.Initialize();
+
+  EXPECT_TRUE(helper_1.controller.clipping_predictor_enabled());
+  EXPECT_FALSE(helper_2.controller.clipping_predictor_enabled());
+  EXPECT_TRUE(helper_1.controller.use_clipping_predictor_step());
+
+  // Expect a change if clipping prediction is enabled.
+  WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                                     helper_1.audio_buffer);
+  WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                                     helper_2.audio_buffer);
+  volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                       kSpeechLevel, 5);
+  volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                       kSpeechLevel, 5);
+
+  WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                          /*clipped_ratio=*/0.0f, helper_1.audio_buffer);
+  WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                          /*clipped_ratio=*/0.0f, helper_2.audio_buffer);
+  volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                       kSpeechLevel, 5);
+  volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                       kSpeechLevel, 5);
+
+  EXPECT_EQ(volume_1, kInitialLevel - kClippedLevelStep);
+  EXPECT_EQ(volume_2, kInitialLevel);
+
+  // Expect no change during waiting.
+  for (int i = 0; i < kClippedWaitFrames / 10; ++i) {
+    WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                                       helper_1.audio_buffer);
+    WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                                       helper_2.audio_buffer);
+    volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+    volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+
+    WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                            /*clipped_ratio=*/0.0f, helper_1.audio_buffer);
+    WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                            /*clipped_ratio=*/0.0f, helper_2.audio_buffer);
+    volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+    volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+
+    EXPECT_EQ(volume_1, kInitialLevel - kClippedLevelStep);
+    EXPECT_EQ(volume_2, kInitialLevel);
+  }
+
+  // Expect a change when the prediction step is used.
+  WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                                     helper_1.audio_buffer);
+  WriteAlternatingAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                                     helper_2.audio_buffer);
+  volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                       kSpeechLevel, 5);
+  volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                       kSpeechLevel, 5);
+
+  WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                          /*clipped_ratio=*/0.0f, helper_1.audio_buffer);
+  WriteAudioBufferSamples(kCloseToClippingPeakRatio * kMaxSample,
+                          /*clipped_ratio=*/0.0f, helper_2.audio_buffer);
+  volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                       kSpeechLevel, 5);
+  volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                       kSpeechLevel, 5);
+
+  EXPECT_EQ(volume_1, kInitialLevel - 2 * kClippedLevelStep);
+  EXPECT_EQ(volume_2, kInitialLevel);
+
+  // Expect no change when clipping is not detected or predicted.
+  for (int i = 0; i < 2 * kClippedWaitFrames / 10; ++i) {
+    WriteAlternatingAudioBufferSamples(/*samples_value=*/0.0f,
+                                       helper_1.audio_buffer);
+    WriteAlternatingAudioBufferSamples(/*samples_value=*/0.0f,
+                                       helper_2.audio_buffer);
+    volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+    volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+
+    WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.0f,
+                            helper_1.audio_buffer);
+    WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.0f,
+                            helper_2.audio_buffer);
+    volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+    volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+  }
+
+  EXPECT_EQ(volume_1, kInitialLevel - 2 * kClippedLevelStep);
+  EXPECT_EQ(volume_2, kInitialLevel);
+
+  // Expect a change for clipping frames.
+  WriteAlternatingAudioBufferSamples(kMaxSample, helper_1.audio_buffer);
+  WriteAlternatingAudioBufferSamples(kMaxSample, helper_2.audio_buffer);
+  volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                       kSpeechLevel, 1);
+  volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                       kSpeechLevel, 1);
+
+  EXPECT_EQ(volume_1, kInitialLevel - 3 * kClippedLevelStep);
+  EXPECT_EQ(volume_2, kInitialLevel - kClippedLevelStep);
+
+  // Expect no change during waiting.
+  for (int i = 0; i < kClippedWaitFrames / 10; ++i) {
+    WriteAlternatingAudioBufferSamples(kMaxSample, helper_1.audio_buffer);
+    WriteAlternatingAudioBufferSamples(kMaxSample, helper_2.audio_buffer);
+    volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+    volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+
+    WriteAudioBufferSamples(kMaxSample, /*clipped_ratio=*/1.0f,
+                            helper_1.audio_buffer);
+    WriteAudioBufferSamples(kMaxSample, /*clipped_ratio=*/1.0f,
+                            helper_2.audio_buffer);
+    volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+    volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                         kSpeechLevel, 5);
+  }
+
+  EXPECT_EQ(volume_1, kInitialLevel - 3 * kClippedLevelStep);
+  EXPECT_EQ(volume_2, kInitialLevel - kClippedLevelStep);
+
+  // Expect a change for clipping frames.
+  WriteAlternatingAudioBufferSamples(kMaxSample, helper_1.audio_buffer);
+  WriteAlternatingAudioBufferSamples(kMaxSample, helper_2.audio_buffer);
+  volume_1 = *helper_1.CallAgcSequence(volume_1, kLowSpeechProbability,
+                                       kSpeechLevel, 1);
+  volume_2 = *helper_2.CallAgcSequence(volume_2, kLowSpeechProbability,
+                                       kSpeechLevel, 1);
+
+  EXPECT_EQ(volume_1, kInitialLevel - 4 * kClippedLevelStep);
+  EXPECT_EQ(volume_2, kInitialLevel - 2 * kClippedLevelStep);
+}
+
+// Checks that passing an empty speech level has no effect on the input volume.
+TEST_P(InputVolumeControllerParametrizedTest, EmptyRmsErrorHasNoEffect) {
+  InputVolumeController controller(kNumChannels,
+                                   GetInputVolumeControllerTestConfig());
+  controller.Initialize();
+
+  // Feed speech with low energy that would trigger an upward adapation of
+  // the analog level if an speech level was not low and the RMS level empty.
+  constexpr int kNumFrames = 125;
+  constexpr int kGainDb = -20;
+  SpeechSamplesReader reader;
+  int volume = reader.Feed(kNumFrames, kInitialInputVolume, kGainDb,
+                           kLowSpeechProbability, absl::nullopt, controller);
+
+  // Check that no adaptation occurs.
+  ASSERT_EQ(volume, kInitialInputVolume);
+}
+
+// Checks that the recommended input volume is not updated unless enough
+// frames have been processed after the previous update.
+TEST(InputVolumeControllerTest, UpdateInputVolumeWaitFramesIsEffective) {
+  constexpr int kInputVolume = kInitialInputVolume;
+  std::unique_ptr<InputVolumeController> controller_wait_0 =
+      CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
+                                  kClippedWaitFrames,
+                                  /*enable_clipping_predictor=*/false,
+                                  /*update_input_volume_wait_frames=*/0);
+  std::unique_ptr<InputVolumeController> controller_wait_100 =
+      CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
+                                  kClippedWaitFrames,
+                                  /*enable_clipping_predictor=*/false,
+                                  /*update_input_volume_wait_frames=*/100);
+  controller_wait_0->Initialize();
+  controller_wait_100->Initialize();
+
+  SpeechSamplesReader reader_1;
+  SpeechSamplesReader reader_2;
+  int volume_wait_0 = reader_1.Feed(
+      /*num_frames=*/99, kInputVolume, /*gain_db=*/0, kHighSpeechProbability,
+      /*speech_level_dbfs=*/-42.0f, *controller_wait_0);
+  int volume_wait_100 = reader_2.Feed(
+      /*num_frames=*/99, kInputVolume, /*gain_db=*/0, kHighSpeechProbability,
+      /*speech_level_dbfs=*/-42.0f, *controller_wait_100);
+
+  // Check that adaptation only occurs if enough frames have been processed.
+  ASSERT_GT(volume_wait_0, kInputVolume);
+  ASSERT_EQ(volume_wait_100, kInputVolume);
+
+  volume_wait_0 =
+      reader_1.Feed(/*num_frames=*/1, volume_wait_0,
+                    /*gain_db=*/0, kHighSpeechProbability,
+                    /*speech_level_dbfs=*/-42.0f, *controller_wait_0);
+  volume_wait_100 =
+      reader_2.Feed(/*num_frames=*/1, volume_wait_100,
+                    /*gain_db=*/0, kHighSpeechProbability,
+                    /*speech_level_dbfs=*/-42.0f, *controller_wait_100);
+
+  // Check that adaptation only occurs when enough frames have been processed.
+  ASSERT_GT(volume_wait_0, kInputVolume);
+  ASSERT_GT(volume_wait_100, kInputVolume);
+}
+
+INSTANTIATE_TEST_SUITE_P(,
+                         InputVolumeControllerParametrizedTest,
+                         ::testing::Values(12, 20));
+
+TEST(InputVolumeControllerTest,
+     MinInputVolumeEnforcedWithClippingWhenAboveClippedLevelMin) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = 80, .clipped_level_min = 70});
+
+  // Trigger a downward adjustment caused by clipping input. Use a low speech
+  // probability to limit the volume changes to clipping handling.
+  WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
+                          helper.audio_buffer);
+  constexpr int kNumCalls = 800;
+  helper.CallAgcSequence(/*applied_input_volume=*/100, kLowSpeechProbability,
+                         /*speech_level_dbfs=*/-18.0f, kNumCalls);
+
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 80);
+}
+
+TEST(InputVolumeControllerTest,
+     ClippedlevelMinEnforcedWithClippingWhenAboveMinInputVolume) {
+  InputVolumeControllerTestHelper helper(
+      /*config=*/{.min_input_volume = 70, .clipped_level_min = 80});
+
+  // Trigger a downward adjustment caused by clipping input. Use a low speech
+  // probability to limit the volume changes to clipping handling.
+  WriteAudioBufferSamples(/*samples_value=*/4000.0f, /*clipped_ratio=*/0.8f,
+                          helper.audio_buffer);
+  constexpr int kNumCalls = 800;
+  helper.CallAgcSequence(/*applied_input_volume=*/100, kLowSpeechProbability,
+                         /*speech_level_dbfs=*/-18.0f, kNumCalls);
+
+  EXPECT_EQ(helper.controller.recommended_input_volume(), 80);
+}
+
+TEST(InputVolumeControllerTest, SpeechRatioThresholdIsEffective) {
+  constexpr int kInputVolume = kInitialInputVolume;
+  // Create two input volume controllers with 10 frames between volume updates
+  // and the minimum speech ratio of 0.8 and speech probability threshold 0.5.
+  std::unique_ptr<InputVolumeController> controller_1 =
+      CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
+                                  kClippedWaitFrames,
+                                  /*enable_clipping_predictor=*/false,
+                                  /*update_input_volume_wait_frames=*/10);
+  std::unique_ptr<InputVolumeController> controller_2 =
+      CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
+                                  kClippedWaitFrames,
+                                  /*enable_clipping_predictor=*/false,
+                                  /*update_input_volume_wait_frames=*/10);
+  controller_1->Initialize();
+  controller_2->Initialize();
+
+  SpeechSamplesReader reader_1;
+  SpeechSamplesReader reader_2;
+
+  int volume_1 = reader_1.Feed(/*num_frames=*/1, kInputVolume, /*gain_db=*/0,
+                               /*speech_probability=*/0.7f,
+                               /*speech_level_dbfs=*/-42.0f, *controller_1);
+  int volume_2 = reader_2.Feed(/*num_frames=*/1, kInputVolume, /*gain_db=*/0,
+                               /*speech_probability=*/0.4f,
+                               /*speech_level_dbfs=*/-42.0f, *controller_2);
+
+  ASSERT_EQ(volume_1, kInputVolume);
+  ASSERT_EQ(volume_2, kInputVolume);
+
+  volume_1 = reader_1.Feed(/*num_frames=*/2, volume_1, /*gain_db=*/0,
+                           /*speech_probability=*/0.4f,
+                           /*speech_level_dbfs=*/-42.0f, *controller_1);
+  volume_2 = reader_2.Feed(/*num_frames=*/2, volume_2, /*gain_db=*/0,
+                           /*speech_probability=*/0.4f,
+                           /*speech_level_dbfs=*/-42.0f, *controller_2);
+
+  ASSERT_EQ(volume_1, kInputVolume);
+  ASSERT_EQ(volume_2, kInputVolume);
+
+  volume_1 = reader_1.Feed(
+      /*num_frames=*/7, volume_1, /*gain_db=*/0,
+      /*speech_probability=*/0.7f, /*speech_level_dbfs=*/-42.0f, *controller_1);
+  volume_2 = reader_2.Feed(
+      /*num_frames=*/7, volume_2, /*gain_db=*/0,
+      /*speech_probability=*/0.7f, /*speech_level_dbfs=*/-42.0f, *controller_2);
+
+  ASSERT_GT(volume_1, kInputVolume);
+  ASSERT_EQ(volume_2, kInputVolume);
+}
+
+TEST(InputVolumeControllerTest, SpeechProbabilityThresholdIsEffective) {
+  constexpr int kInputVolume = kInitialInputVolume;
+  // Create two input volume controllers with the exact same settings and
+  // 10 frames between volume updates.
+  std::unique_ptr<InputVolumeController> controller_1 =
+      CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
+                                  kClippedWaitFrames,
+                                  /*enable_clipping_predictor=*/false,
+                                  /*update_input_volume_wait_frames=*/10);
+  std::unique_ptr<InputVolumeController> controller_2 =
+      CreateInputVolumeController(kClippedLevelStep, kClippedRatioThreshold,
+                                  kClippedWaitFrames,
+                                  /*enable_clipping_predictor=*/false,
+                                  /*update_input_volume_wait_frames=*/10);
+  controller_1->Initialize();
+  controller_2->Initialize();
+
+  SpeechSamplesReader reader_1;
+  SpeechSamplesReader reader_2;
+
+  // Process with two sets of inputs: Use `reader_1` to process inputs
+  // that make the volume to be adjusted after enough frames have been
+  // processsed and `reader_2` to process inputs that won't make the volume
+  // to be adjusted.
+  int volume_1 = reader_1.Feed(/*num_frames=*/1, kInputVolume, /*gain_db=*/0,
+                               /*speech_probability=*/0.5f,
+                               /*speech_level_dbfs=*/-42.0f, *controller_1);
+  int volume_2 = reader_2.Feed(/*num_frames=*/1, kInputVolume, /*gain_db=*/0,
+                               /*speech_probability=*/0.49f,
+                               /*speech_level_dbfs=*/-42.0f, *controller_2);
+
+  ASSERT_EQ(volume_1, kInputVolume);
+  ASSERT_EQ(volume_2, kInputVolume);
+
+  reader_1.Feed(/*num_frames=*/2, volume_1, /*gain_db=*/0,
+                /*speech_probability=*/0.49f, /*speech_level_dbfs=*/-42.0f,
+                *controller_1);
+  reader_2.Feed(/*num_frames=*/2, volume_2, /*gain_db=*/0,
+                /*speech_probability=*/0.49f, /*speech_level_dbfs=*/-42.0f,
+                *controller_2);
+
+  ASSERT_EQ(volume_1, kInputVolume);
+  ASSERT_EQ(volume_2, kInputVolume);
+
+  volume_1 = reader_1.Feed(
+      /*num_frames=*/7, volume_1, /*gain_db=*/0,
+      /*speech_probability=*/0.5f, /*speech_level_dbfs=*/-42.0f, *controller_1);
+  volume_2 = reader_2.Feed(
+      /*num_frames=*/7, volume_2, /*gain_db=*/0,
+      /*speech_probability=*/0.5f, /*speech_level_dbfs=*/-42.0f, *controller_2);
+
+  ASSERT_GT(volume_1, kInputVolume);
+  ASSERT_EQ(volume_2, kInputVolume);
+}
+
+TEST(InputVolumeControllerTest,
+     DoNotLogRecommendedInputVolumeOnChangeToMatchTarget) {
+  metrics::Reset();
+
+  SpeechSamplesReader reader;
+  auto controller = CreateInputVolumeController();
+  controller->Initialize();
+  // Trigger a downward volume change by inputting audio that clips. Pass a
+  // speech level that falls in the target range to make sure that the
+  // adaptation is not made to match the target range.
+  constexpr int kStartupVolume = 255;
+  const int volume = reader.Feed(/*num_frames=*/14, kStartupVolume,
+                                 /*gain_db=*/50, kHighSpeechProbability,
+                                 /*speech_level_dbfs=*/-20.0f, *controller);
+  ASSERT_LT(volume, kStartupVolume);
+  EXPECT_METRIC_THAT(
+      metrics::Samples(
+          "WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget"),
+      ::testing::IsEmpty());
+}
+
+TEST(InputVolumeControllerTest,
+     LogRecommendedInputVolumeOnUpwardChangeToMatchTarget) {
+  metrics::Reset();
+
+  SpeechSamplesReader reader;
+  auto controller = CreateInputVolumeController();
+  controller->Initialize();
+  constexpr int kStartupVolume = 100;
+  // Trigger an upward volume change by inputting audio that does not clip and
+  // by passing a speech level below the target range.
+  const int volume = reader.Feed(/*num_frames=*/14, kStartupVolume,
+                                 /*gain_db=*/-6, kHighSpeechProbability,
+                                 /*speech_level_dbfs=*/-50.0f, *controller);
+  ASSERT_GT(volume, kStartupVolume);
+  EXPECT_METRIC_THAT(
+      metrics::Samples(
+          "WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget"),
+      ::testing::Not(::testing::IsEmpty()));
+}
+
+TEST(InputVolumeControllerTest,
+     LogRecommendedInputVolumeOnDownwardChangeToMatchTarget) {
+  metrics::Reset();
+
+  SpeechSamplesReader reader;
+  auto controller = CreateInputVolumeController();
+  controller->Initialize();
+  constexpr int kStartupVolume = 100;
+  // Trigger a downward volume change by inputting audio that does not clip and
+  // by passing a speech level above the target range.
+  const int volume = reader.Feed(/*num_frames=*/14, kStartupVolume,
+                                 /*gain_db=*/-6, kHighSpeechProbability,
+                                 /*speech_level_dbfs=*/-5.0f, *controller);
+  ASSERT_LT(volume, kStartupVolume);
+  EXPECT_METRIC_THAT(
+      metrics::Samples(
+          "WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget"),
+      ::testing::Not(::testing::IsEmpty()));
+}
+
+TEST(MonoInputVolumeControllerTest, CheckHandleClippingLowersVolume) {
+  constexpr int kInitialInputVolume = 100;
+  constexpr int kInputVolumeStep = 29;
+  MonoInputVolumeController mono_controller(
+      /*clipped_level_min=*/70,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/3, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller.Initialize();
+
+  UpdateRecommendedInputVolume(mono_controller, kInitialInputVolume,
+                               kLowSpeechProbability,
+                               /*rms_error_dbfs*/ -10.0f);
+
+  mono_controller.HandleClipping(kInputVolumeStep);
+
+  EXPECT_EQ(mono_controller.recommended_analog_level(),
+            kInitialInputVolume - kInputVolumeStep);
+}
+
+TEST(MonoInputVolumeControllerTest,
+     CheckProcessNegativeRmsErrorDecreasesInputVolume) {
+  constexpr int kInitialInputVolume = 100;
+  MonoInputVolumeController mono_controller(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/3, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller.Initialize();
+
+  int volume = UpdateRecommendedInputVolume(
+      mono_controller, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+  volume = UpdateRecommendedInputVolume(mono_controller, volume,
+                                        kHighSpeechProbability, -10.0f);
+  volume = UpdateRecommendedInputVolume(mono_controller, volume,
+                                        kHighSpeechProbability, -10.0f);
+
+  EXPECT_LT(volume, kInitialInputVolume);
+}
+
+TEST(MonoInputVolumeControllerTest,
+     CheckProcessPositiveRmsErrorIncreasesInputVolume) {
+  constexpr int kInitialInputVolume = 100;
+  MonoInputVolumeController mono_controller(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/3, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller.Initialize();
+
+  int volume = UpdateRecommendedInputVolume(
+      mono_controller, kInitialInputVolume, kHighSpeechProbability, 10.0f);
+  volume = UpdateRecommendedInputVolume(mono_controller, volume,
+                                        kHighSpeechProbability, 10.0f);
+  volume = UpdateRecommendedInputVolume(mono_controller, volume,
+                                        kHighSpeechProbability, 10.0f);
+
+  EXPECT_GT(volume, kInitialInputVolume);
+}
+
+TEST(MonoInputVolumeControllerTest,
+     CheckProcessNegativeRmsErrorDecreasesInputVolumeWithLimit) {
+  constexpr int kInitialInputVolume = 100;
+  MonoInputVolumeController mono_controller_1(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  MonoInputVolumeController mono_controller_2(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  MonoInputVolumeController mono_controller_3(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/2,
+      /*speech_probability_threshold=*/0.7,
+      /*speech_ratio_threshold=*/0.8);
+  mono_controller_1.Initialize();
+  mono_controller_2.Initialize();
+  mono_controller_3.Initialize();
+
+  // Process RMS errors in the range
+  // [`-kMaxResidualGainChange`, `kMaxResidualGainChange`].
+  int volume_1 = UpdateRecommendedInputVolume(
+      mono_controller_1, kInitialInputVolume, kHighSpeechProbability, -14.0f);
+  volume_1 = UpdateRecommendedInputVolume(mono_controller_1, volume_1,
+                                          kHighSpeechProbability, -14.0f);
+  // Process RMS errors outside the range
+  // [`-kMaxResidualGainChange`, `kMaxResidualGainChange`].
+  int volume_2 = UpdateRecommendedInputVolume(
+      mono_controller_2, kInitialInputVolume, kHighSpeechProbability, -15.0f);
+  int volume_3 = UpdateRecommendedInputVolume(
+      mono_controller_3, kInitialInputVolume, kHighSpeechProbability, -30.0f);
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kHighSpeechProbability, -15.0f);
+  volume_3 = UpdateRecommendedInputVolume(mono_controller_3, volume_3,
+                                          kHighSpeechProbability, -30.0f);
+
+  EXPECT_LT(volume_1, kInitialInputVolume);
+  EXPECT_LT(volume_2, volume_1);
+  EXPECT_EQ(volume_2, volume_3);
+}
+
+TEST(MonoInputVolumeControllerTest,
+     CheckProcessPositiveRmsErrorIncreasesInputVolumeWithLimit) {
+  constexpr int kInitialInputVolume = 100;
+  MonoInputVolumeController mono_controller_1(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  MonoInputVolumeController mono_controller_2(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  MonoInputVolumeController mono_controller_3(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller_1.Initialize();
+  mono_controller_2.Initialize();
+  mono_controller_3.Initialize();
+
+  // Process RMS errors in the range
+  // [`-kMaxResidualGainChange`, `kMaxResidualGainChange`].
+  int volume_1 = UpdateRecommendedInputVolume(
+      mono_controller_1, kInitialInputVolume, kHighSpeechProbability, 14.0f);
+  volume_1 = UpdateRecommendedInputVolume(mono_controller_1, volume_1,
+                                          kHighSpeechProbability, 14.0f);
+  // Process RMS errors outside the range
+  // [`-kMaxResidualGainChange`, `kMaxResidualGainChange`].
+  int volume_2 = UpdateRecommendedInputVolume(
+      mono_controller_2, kInitialInputVolume, kHighSpeechProbability, 15.0f);
+  int volume_3 = UpdateRecommendedInputVolume(
+      mono_controller_3, kInitialInputVolume, kHighSpeechProbability, 30.0f);
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kHighSpeechProbability, 15.0f);
+  volume_3 = UpdateRecommendedInputVolume(mono_controller_3, volume_3,
+                                          kHighSpeechProbability, 30.0f);
+
+  EXPECT_GT(volume_1, kInitialInputVolume);
+  EXPECT_GT(volume_2, volume_1);
+  EXPECT_EQ(volume_2, volume_3);
+}
+
+TEST(MonoInputVolumeControllerTest,
+     CheckProcessRmsErrorDecreasesInputVolumeRepeatedly) {
+  constexpr int kInitialInputVolume = 100;
+  MonoInputVolumeController mono_controller(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller.Initialize();
+
+  int volume_before = UpdateRecommendedInputVolume(
+      mono_controller, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+  volume_before = UpdateRecommendedInputVolume(mono_controller, volume_before,
+                                               kHighSpeechProbability, -10.0f);
+
+  EXPECT_LT(volume_before, kInitialInputVolume);
+
+  int volume_after = UpdateRecommendedInputVolume(
+      mono_controller, volume_before, kHighSpeechProbability, -10.0f);
+  volume_after = UpdateRecommendedInputVolume(mono_controller, volume_after,
+                                              kHighSpeechProbability, -10.0f);
+
+  EXPECT_LT(volume_after, volume_before);
+}
+
+TEST(MonoInputVolumeControllerTest,
+     CheckProcessPositiveRmsErrorIncreasesInputVolumeRepeatedly) {
+  constexpr int kInitialInputVolume = 100;
+  MonoInputVolumeController mono_controller(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/32,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller.Initialize();
+
+  int volume_before = UpdateRecommendedInputVolume(
+      mono_controller, kInitialInputVolume, kHighSpeechProbability, 10.0f);
+  volume_before = UpdateRecommendedInputVolume(mono_controller, volume_before,
+                                               kHighSpeechProbability, 10.0f);
+
+  EXPECT_GT(volume_before, kInitialInputVolume);
+
+  int volume_after = UpdateRecommendedInputVolume(
+      mono_controller, volume_before, kHighSpeechProbability, 10.0f);
+  volume_after = UpdateRecommendedInputVolume(mono_controller, volume_after,
+                                              kHighSpeechProbability, 10.0f);
+
+  EXPECT_GT(volume_after, volume_before);
+}
+
+TEST(MonoInputVolumeControllerTest, CheckClippedLevelMinIsEffective) {
+  constexpr int kInitialInputVolume = 100;
+  constexpr int kClippedLevelMin = 70;
+  MonoInputVolumeController mono_controller_1(
+      kClippedLevelMin,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  MonoInputVolumeController mono_controller_2(
+      kClippedLevelMin,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller_1.Initialize();
+  mono_controller_2.Initialize();
+
+  // Process one frame to reset the state for `HandleClipping()`.
+  EXPECT_EQ(UpdateRecommendedInputVolume(mono_controller_1, kInitialInputVolume,
+                                         kLowSpeechProbability, -10.0f),
+            kInitialInputVolume);
+  EXPECT_EQ(UpdateRecommendedInputVolume(mono_controller_2, kInitialInputVolume,
+                                         kLowSpeechProbability, -10.0f),
+            kInitialInputVolume);
+
+  mono_controller_1.HandleClipping(29);
+  mono_controller_2.HandleClipping(31);
+
+  EXPECT_EQ(mono_controller_2.recommended_analog_level(), kClippedLevelMin);
+  EXPECT_LT(mono_controller_2.recommended_analog_level(),
+            mono_controller_1.recommended_analog_level());
+}
+
+TEST(MonoInputVolumeControllerTest, CheckMinMicLevelIsEffective) {
+  constexpr int kInitialInputVolume = 100;
+  constexpr int kMinMicLevel = 64;
+  MonoInputVolumeController mono_controller_1(
+      /*clipped_level_min=*/64, kMinMicLevel,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  MonoInputVolumeController mono_controller_2(
+      /*clipped_level_min=*/64, kMinMicLevel,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller_1.Initialize();
+  mono_controller_2.Initialize();
+
+  int volume_1 = UpdateRecommendedInputVolume(
+      mono_controller_1, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+  int volume_2 = UpdateRecommendedInputVolume(
+      mono_controller_2, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+
+  EXPECT_EQ(volume_1, kInitialInputVolume);
+  EXPECT_EQ(volume_2, kInitialInputVolume);
+
+  volume_1 = UpdateRecommendedInputVolume(mono_controller_1, volume_1,
+                                          kHighSpeechProbability, -10.0f);
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kHighSpeechProbability, -30.0f);
+
+  EXPECT_LT(volume_1, kInitialInputVolume);
+  EXPECT_LT(volume_2, volume_1);
+  EXPECT_EQ(volume_2, kMinMicLevel);
+}
+
+TEST(MonoInputVolumeControllerTest,
+     CheckUpdateInputVolumeWaitFramesIsEffective) {
+  constexpr int kInitialInputVolume = 100;
+  MonoInputVolumeController mono_controller_1(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/1, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  MonoInputVolumeController mono_controller_2(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/3, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller_1.Initialize();
+  mono_controller_2.Initialize();
+
+  int volume_1 = UpdateRecommendedInputVolume(
+      mono_controller_1, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+  int volume_2 = UpdateRecommendedInputVolume(
+      mono_controller_2, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+
+  EXPECT_EQ(volume_1, kInitialInputVolume);
+  EXPECT_EQ(volume_2, kInitialInputVolume);
+
+  volume_1 = UpdateRecommendedInputVolume(mono_controller_1, volume_1,
+                                          kHighSpeechProbability, -10.0f);
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kHighSpeechProbability, -10.0f);
+
+  EXPECT_LT(volume_1, kInitialInputVolume);
+  EXPECT_EQ(volume_2, kInitialInputVolume);
+
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kHighSpeechProbability, -10.0f);
+
+  EXPECT_LT(volume_2, kInitialInputVolume);
+}
+
+TEST(MonoInputVolumeControllerTest,
+     CheckSpeechProbabilityThresholdIsEffective) {
+  constexpr int kInitialInputVolume = 100;
+  constexpr float kSpeechProbabilityThreshold = 0.8f;
+  MonoInputVolumeController mono_controller_1(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/2, kSpeechProbabilityThreshold,
+      kSpeechRatioThreshold);
+  MonoInputVolumeController mono_controller_2(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/2, kSpeechProbabilityThreshold,
+      kSpeechRatioThreshold);
+  mono_controller_1.Initialize();
+  mono_controller_2.Initialize();
+
+  int volume_1 =
+      UpdateRecommendedInputVolume(mono_controller_1, kInitialInputVolume,
+                                   kSpeechProbabilityThreshold, -10.0f);
+  int volume_2 =
+      UpdateRecommendedInputVolume(mono_controller_2, kInitialInputVolume,
+                                   kSpeechProbabilityThreshold, -10.0f);
+
+  EXPECT_EQ(volume_1, kInitialInputVolume);
+  EXPECT_EQ(volume_2, kInitialInputVolume);
+
+  volume_1 = UpdateRecommendedInputVolume(
+      mono_controller_1, volume_1, kSpeechProbabilityThreshold - 0.1f, -10.0f);
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kSpeechProbabilityThreshold, -10.0f);
+
+  EXPECT_EQ(volume_1, kInitialInputVolume);
+  EXPECT_LT(volume_2, volume_1);
+}
+
+TEST(MonoInputVolumeControllerTest, CheckSpeechRatioThresholdIsEffective) {
+  constexpr int kInitialInputVolume = 100;
+  MonoInputVolumeController mono_controller_1(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/4, kHighSpeechProbability,
+      /*speech_ratio_threshold=*/0.75f);
+  MonoInputVolumeController mono_controller_2(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/4, kHighSpeechProbability,
+      /*speech_ratio_threshold=*/0.75f);
+  mono_controller_1.Initialize();
+  mono_controller_2.Initialize();
+
+  int volume_1 = UpdateRecommendedInputVolume(
+      mono_controller_1, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+  int volume_2 = UpdateRecommendedInputVolume(
+      mono_controller_2, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+
+  volume_1 = UpdateRecommendedInputVolume(mono_controller_1, volume_1,
+                                          kHighSpeechProbability, -10.0f);
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kHighSpeechProbability, -10.0f);
+
+  volume_1 = UpdateRecommendedInputVolume(mono_controller_1, volume_1,
+                                          kLowSpeechProbability, -10.0f);
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kLowSpeechProbability, -10.0f);
+
+  EXPECT_EQ(volume_1, kInitialInputVolume);
+  EXPECT_EQ(volume_2, kInitialInputVolume);
+
+  volume_1 = UpdateRecommendedInputVolume(mono_controller_1, volume_1,
+                                          kLowSpeechProbability, -10.0f);
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kHighSpeechProbability, -10.0f);
+
+  EXPECT_EQ(volume_1, kInitialInputVolume);
+  EXPECT_LT(volume_2, volume_1);
+}
+
+TEST(MonoInputVolumeControllerTest,
+     CheckProcessEmptyRmsErrorDoesNotLowerVolume) {
+  constexpr int kInitialInputVolume = 100;
+  MonoInputVolumeController mono_controller_1(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  MonoInputVolumeController mono_controller_2(
+      /*clipped_level_min=*/64,
+      /*min_mic_level=*/84,
+      /*update_input_volume_wait_frames=*/2, kHighSpeechProbability,
+      kSpeechRatioThreshold);
+  mono_controller_1.Initialize();
+  mono_controller_2.Initialize();
+
+  int volume_1 = UpdateRecommendedInputVolume(
+      mono_controller_1, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+  int volume_2 = UpdateRecommendedInputVolume(
+      mono_controller_2, kInitialInputVolume, kHighSpeechProbability, -10.0f);
+
+  EXPECT_EQ(volume_1, kInitialInputVolume);
+  EXPECT_EQ(volume_2, kInitialInputVolume);
+
+  volume_1 = UpdateRecommendedInputVolume(
+      mono_controller_1, volume_1, kHighSpeechProbability, absl::nullopt);
+  volume_2 = UpdateRecommendedInputVolume(mono_controller_2, volume_2,
+                                          kHighSpeechProbability, -10.0f);
+
+  EXPECT_EQ(volume_1, kInitialInputVolume);
+  EXPECT_LT(volume_2, volume_1);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter.cc b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter.cc
new file mode 100644
index 0000000000..05624b1f92
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter.cc
@@ -0,0 +1,171 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
+
+#include <cmath>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+
+using InputVolumeType = InputVolumeStatsReporter::InputVolumeType;
+
+constexpr int kFramesIn60Seconds = 6000;
+constexpr int kMinInputVolume = 0;
+constexpr int kMaxInputVolume = 255;
+constexpr int kMaxUpdate = kMaxInputVolume - kMinInputVolume;
+
+int ComputeAverageUpdate(int sum_updates, int num_updates) {
+  RTC_DCHECK_GE(sum_updates, 0);
+  RTC_DCHECK_LE(sum_updates, kMaxUpdate * kFramesIn60Seconds);
+  RTC_DCHECK_GE(num_updates, 0);
+  RTC_DCHECK_LE(num_updates, kFramesIn60Seconds);
+  if (num_updates == 0) {
+    return 0;
+  }
+  return std::round(static_cast<float>(sum_updates) /
+                    static_cast<float>(num_updates));
+}
+
+constexpr absl::string_view MetricNamePrefix(
+    InputVolumeType input_volume_type) {
+  switch (input_volume_type) {
+    case InputVolumeType::kApplied:
+      return "WebRTC.Audio.Apm.AppliedInputVolume.";
+    case InputVolumeType::kRecommended:
+      return "WebRTC.Audio.Apm.RecommendedInputVolume.";
+  }
+}
+
+metrics::Histogram* CreateVolumeHistogram(InputVolumeType input_volume_type) {
+  char buffer[64];
+  rtc::SimpleStringBuilder builder(buffer);
+  builder << MetricNamePrefix(input_volume_type) << "OnChange";
+  return metrics::HistogramFactoryGetCountsLinear(/*name=*/builder.str(),
+                                                  /*min=*/1,
+                                                  /*max=*/kMaxInputVolume,
+                                                  /*bucket_count=*/50);
+}
+
+metrics::Histogram* CreateRateHistogram(InputVolumeType input_volume_type,
+                                        absl::string_view name) {
+  char buffer[64];
+  rtc::SimpleStringBuilder builder(buffer);
+  builder << MetricNamePrefix(input_volume_type) << name;
+  return metrics::HistogramFactoryGetCountsLinear(/*name=*/builder.str(),
+                                                  /*min=*/1,
+                                                  /*max=*/kFramesIn60Seconds,
+                                                  /*bucket_count=*/50);
+}
+
+metrics::Histogram* CreateAverageHistogram(InputVolumeType input_volume_type,
+                                           absl::string_view name) {
+  char buffer[64];
+  rtc::SimpleStringBuilder builder(buffer);
+  builder << MetricNamePrefix(input_volume_type) << name;
+  return metrics::HistogramFactoryGetCountsLinear(/*name=*/builder.str(),
+                                                  /*min=*/1,
+                                                  /*max=*/kMaxUpdate,
+                                                  /*bucket_count=*/50);
+}
+
+}  // namespace
+
+InputVolumeStatsReporter::InputVolumeStatsReporter(InputVolumeType type)
+    : histograms_(
+          {.on_volume_change = CreateVolumeHistogram(type),
+           .decrease_rate = CreateRateHistogram(type, "DecreaseRate"),
+           .decrease_average = CreateAverageHistogram(type, "DecreaseAverage"),
+           .increase_rate = CreateRateHistogram(type, "IncreaseRate"),
+           .increase_average = CreateAverageHistogram(type, "IncreaseAverage"),
+           .update_rate = CreateRateHistogram(type, "UpdateRate"),
+           .update_average = CreateAverageHistogram(type, "UpdateAverage")}),
+      cannot_log_stats_(!histograms_.AllPointersSet()) {
+  if (cannot_log_stats_) {
+    RTC_LOG(LS_WARNING) << "Will not log any `" << MetricNamePrefix(type)
+                        << "*` histogram stats.";
+  }
+}
+
+InputVolumeStatsReporter::~InputVolumeStatsReporter() = default;
+
+void InputVolumeStatsReporter::UpdateStatistics(int input_volume) {
+  if (cannot_log_stats_) {
+    // Since the stats cannot be logged, do not bother updating them.
+    return;
+  }
+
+  RTC_DCHECK_GE(input_volume, kMinInputVolume);
+  RTC_DCHECK_LE(input_volume, kMaxInputVolume);
+  if (previous_input_volume_.has_value() &&
+      input_volume != previous_input_volume_.value()) {
+    // Update stats when the input volume changes.
+    metrics::HistogramAdd(histograms_.on_volume_change, input_volume);
+    // Update stats that are periodically logged.
+    const int volume_change = input_volume - previous_input_volume_.value();
+    if (volume_change < 0) {
+      ++volume_update_stats_.num_decreases;
+      volume_update_stats_.sum_decreases -= volume_change;
+    } else {
+      ++volume_update_stats_.num_increases;
+      volume_update_stats_.sum_increases += volume_change;
+    }
+  }
+  // Periodically log input volume change metrics.
+  if (++log_volume_update_stats_counter_ >= kFramesIn60Seconds) {
+    LogVolumeUpdateStats();
+    volume_update_stats_ = {};
+    log_volume_update_stats_counter_ = 0;
+  }
+  previous_input_volume_ = input_volume;
+}
+
+void InputVolumeStatsReporter::LogVolumeUpdateStats() const {
+  // Decrease rate and average.
+  metrics::HistogramAdd(histograms_.decrease_rate,
+                        volume_update_stats_.num_decreases);
+  if (volume_update_stats_.num_decreases > 0) {
+    int average_decrease = ComputeAverageUpdate(
+        volume_update_stats_.sum_decreases, volume_update_stats_.num_decreases);
+    metrics::HistogramAdd(histograms_.decrease_average, average_decrease);
+  }
+  // Increase rate and average.
+  metrics::HistogramAdd(histograms_.increase_rate,
+                        volume_update_stats_.num_increases);
+  if (volume_update_stats_.num_increases > 0) {
+    int average_increase = ComputeAverageUpdate(
+        volume_update_stats_.sum_increases, volume_update_stats_.num_increases);
+    metrics::HistogramAdd(histograms_.increase_average, average_increase);
+  }
+  // Update rate and average.
+  int num_updates =
+      volume_update_stats_.num_decreases + volume_update_stats_.num_increases;
+  metrics::HistogramAdd(histograms_.update_rate, num_updates);
+  if (num_updates > 0) {
+    int average_update = ComputeAverageUpdate(
+        volume_update_stats_.sum_decreases + volume_update_stats_.sum_increases,
+        num_updates);
+    metrics::HistogramAdd(histograms_.update_average, average_update);
+  }
+}
+
+void UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(int volume) {
+  RTC_HISTOGRAM_COUNTS_LINEAR(
+      "WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget", volume,
+      1, kMaxInputVolume, 50);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter.h b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter.h
new file mode 100644
index 0000000000..31b110031c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter.h
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_STATS_REPORTER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_STATS_REPORTER_H_
+
+#include "absl/types/optional.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+// Input volume statistics calculator. Computes aggregate stats based on the
+// framewise input volume observed by `UpdateStatistics()`. Periodically logs
+// the statistics into a histogram.
+class InputVolumeStatsReporter {
+ public:
+  enum class InputVolumeType {
+    kApplied = 0,
+    kRecommended = 1,
+  };
+
+  explicit InputVolumeStatsReporter(InputVolumeType input_volume_type);
+  InputVolumeStatsReporter(const InputVolumeStatsReporter&) = delete;
+  InputVolumeStatsReporter operator=(const InputVolumeStatsReporter&) = delete;
+  ~InputVolumeStatsReporter();
+
+  // Updates the stats based on `input_volume`. Periodically logs the stats into
+  // a histogram.
+  void UpdateStatistics(int input_volume);
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeStatsReporterTest,
+                           CheckVolumeUpdateStatsForEmptyStats);
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeStatsReporterTest,
+                           CheckVolumeUpdateStatsAfterNoVolumeChange);
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeStatsReporterTest,
+                           CheckVolumeUpdateStatsAfterVolumeIncrease);
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeStatsReporterTest,
+                           CheckVolumeUpdateStatsAfterVolumeDecrease);
+  FRIEND_TEST_ALL_PREFIXES(InputVolumeStatsReporterTest,
+                           CheckVolumeUpdateStatsAfterReset);
+
+  // Stores input volume update stats to enable calculation of update rate and
+  // average update separately for volume increases and decreases.
+  struct VolumeUpdateStats {
+    int num_decreases = 0;
+    int num_increases = 0;
+    int sum_decreases = 0;
+    int sum_increases = 0;
+  } volume_update_stats_;
+
+  // Returns a copy of the stored statistics. Use only for testing.
+  VolumeUpdateStats volume_update_stats() const { return volume_update_stats_; }
+
+  // Computes aggregate stat and logs them into a histogram.
+  void LogVolumeUpdateStats() const;
+
+  // Histograms.
+  struct Histograms {
+    metrics::Histogram* const on_volume_change;
+    metrics::Histogram* const decrease_rate;
+    metrics::Histogram* const decrease_average;
+    metrics::Histogram* const increase_rate;
+    metrics::Histogram* const increase_average;
+    metrics::Histogram* const update_rate;
+    metrics::Histogram* const update_average;
+    bool AllPointersSet() const {
+      return !!on_volume_change && !!decrease_rate && !!decrease_average &&
+             !!increase_rate && !!increase_average && !!update_rate &&
+             !!update_average;
+    }
+  } histograms_;
+
+  // True if the stats cannot be logged.
+  const bool cannot_log_stats_;
+
+  int log_volume_update_stats_counter_ = 0;
+  absl::optional<int> previous_input_volume_ = absl::nullopt;
+};
+
+// Updates the histogram that keeps track of recommended input volume changes
+// required in order to match the target level in the input volume adaptation
+// process.
+void UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(int volume);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_STATS_REPORTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_gn/moz.build
new file mode 100644
index 0000000000..5a662f4079
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("input_volume_stats_reporter_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_unittest.cc
new file mode 100644
index 0000000000..e762c1fb59
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/input_volume_stats_reporter_unittest.cc
@@ -0,0 +1,246 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/metrics.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace {
+
+using InputVolumeType = InputVolumeStatsReporter::InputVolumeType;
+
+constexpr int kFramesIn60Seconds = 6000;
+
+constexpr absl::string_view kLabelPrefix = "WebRTC.Audio.Apm.";
+
+class InputVolumeStatsReporterTest
+    : public ::testing::TestWithParam<InputVolumeType> {
+ public:
+  InputVolumeStatsReporterTest() { metrics::Reset(); }
+
+ protected:
+  InputVolumeType InputVolumeType() const { return GetParam(); }
+  std::string VolumeLabel() const {
+    return (rtc::StringBuilder(kLabelPrefix) << VolumeTypeLabel() << "OnChange")
+        .str();
+  }
+  std::string DecreaseRateLabel() const {
+    return (rtc::StringBuilder(kLabelPrefix)
+            << VolumeTypeLabel() << "DecreaseRate")
+        .str();
+  }
+  std::string DecreaseAverageLabel() const {
+    return (rtc::StringBuilder(kLabelPrefix)
+            << VolumeTypeLabel() << "DecreaseAverage")
+        .str();
+  }
+  std::string IncreaseRateLabel() const {
+    return (rtc::StringBuilder(kLabelPrefix)
+            << VolumeTypeLabel() << "IncreaseRate")
+        .str();
+  }
+  std::string IncreaseAverageLabel() const {
+    return (rtc::StringBuilder(kLabelPrefix)
+            << VolumeTypeLabel() << "IncreaseAverage")
+        .str();
+  }
+  std::string UpdateRateLabel() const {
+    return (rtc::StringBuilder(kLabelPrefix)
+            << VolumeTypeLabel() << "UpdateRate")
+        .str();
+  }
+  std::string UpdateAverageLabel() const {
+    return (rtc::StringBuilder(kLabelPrefix)
+            << VolumeTypeLabel() << "UpdateAverage")
+        .str();
+  }
+
+ private:
+  absl::string_view VolumeTypeLabel() const {
+    switch (InputVolumeType()) {
+      case InputVolumeType::kApplied:
+        return "AppliedInputVolume.";
+      case InputVolumeType::kRecommended:
+        return "RecommendedInputVolume.";
+    }
+  }
+};
+
+TEST_P(InputVolumeStatsReporterTest, CheckVolumeOnChangeIsEmpty) {
+  InputVolumeStatsReporter stats_reporter(InputVolumeType());
+  stats_reporter.UpdateStatistics(10);
+  EXPECT_METRIC_THAT(metrics::Samples(VolumeLabel()), ::testing::ElementsAre());
+}
+
+TEST_P(InputVolumeStatsReporterTest, CheckRateAverageStatsEmpty) {
+  InputVolumeStatsReporter stats_reporter(InputVolumeType());
+  constexpr int kInputVolume = 10;
+  stats_reporter.UpdateStatistics(kInputVolume);
+  // Update almost until the periodic logging and reset.
+  for (int i = 0; i < kFramesIn60Seconds - 2; i += 2) {
+    stats_reporter.UpdateStatistics(kInputVolume + 2);
+    stats_reporter.UpdateStatistics(kInputVolume);
+  }
+  EXPECT_METRIC_THAT(metrics::Samples(UpdateRateLabel()),
+                     ::testing::ElementsAre());
+  EXPECT_METRIC_THAT(metrics::Samples(DecreaseRateLabel()),
+                     ::testing::ElementsAre());
+  EXPECT_METRIC_THAT(metrics::Samples(IncreaseRateLabel()),
+                     ::testing::ElementsAre());
+  EXPECT_METRIC_THAT(metrics::Samples(UpdateAverageLabel()),
+                     ::testing::ElementsAre());
+  EXPECT_METRIC_THAT(metrics::Samples(DecreaseAverageLabel()),
+                     ::testing::ElementsAre());
+  EXPECT_METRIC_THAT(metrics::Samples(IncreaseAverageLabel()),
+                     ::testing::ElementsAre());
+}
+
+TEST_P(InputVolumeStatsReporterTest, CheckSamples) {
+  InputVolumeStatsReporter stats_reporter(InputVolumeType());
+
+  constexpr int kInputVolume1 = 10;
+  stats_reporter.UpdateStatistics(kInputVolume1);
+  // Update until periodic logging.
+  constexpr int kInputVolume2 = 12;
+  for (int i = 0; i < kFramesIn60Seconds; i += 2) {
+    stats_reporter.UpdateStatistics(kInputVolume2);
+    stats_reporter.UpdateStatistics(kInputVolume1);
+  }
+  // Update until periodic logging.
+  constexpr int kInputVolume3 = 13;
+  for (int i = 0; i < kFramesIn60Seconds; i += 2) {
+    stats_reporter.UpdateStatistics(kInputVolume3);
+    stats_reporter.UpdateStatistics(kInputVolume1);
+  }
+
+  // Check volume changes stats.
+  EXPECT_METRIC_THAT(
+      metrics::Samples(VolumeLabel()),
+      ::testing::ElementsAre(
+          ::testing::Pair(kInputVolume1, kFramesIn60Seconds),
+          ::testing::Pair(kInputVolume2, kFramesIn60Seconds / 2),
+          ::testing::Pair(kInputVolume3, kFramesIn60Seconds / 2)));
+
+  // Check volume change rate stats.
+  EXPECT_METRIC_THAT(
+      metrics::Samples(UpdateRateLabel()),
+      ::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds - 1, 1),
+                             ::testing::Pair(kFramesIn60Seconds, 1)));
+  EXPECT_METRIC_THAT(
+      metrics::Samples(DecreaseRateLabel()),
+      ::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2 - 1, 1),
+                             ::testing::Pair(kFramesIn60Seconds / 2, 1)));
+  EXPECT_METRIC_THAT(
+      metrics::Samples(IncreaseRateLabel()),
+      ::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2, 2)));
+
+  // Check volume change average stats.
+  EXPECT_METRIC_THAT(
+      metrics::Samples(UpdateAverageLabel()),
+      ::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
+  EXPECT_METRIC_THAT(
+      metrics::Samples(DecreaseAverageLabel()),
+      ::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
+  EXPECT_METRIC_THAT(
+      metrics::Samples(IncreaseAverageLabel()),
+      ::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
+}
+}  // namespace
+
+TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsForEmptyStats) {
+  InputVolumeStatsReporter stats_reporter(InputVolumeType());
+  const auto& update_stats = stats_reporter.volume_update_stats();
+  EXPECT_EQ(update_stats.num_decreases, 0);
+  EXPECT_EQ(update_stats.sum_decreases, 0);
+  EXPECT_EQ(update_stats.num_increases, 0);
+  EXPECT_EQ(update_stats.sum_increases, 0);
+}
+
+TEST_P(InputVolumeStatsReporterTest,
+       CheckVolumeUpdateStatsAfterNoVolumeChange) {
+  constexpr int kInputVolume = 10;
+  InputVolumeStatsReporter stats_reporter(InputVolumeType());
+  stats_reporter.UpdateStatistics(kInputVolume);
+  stats_reporter.UpdateStatistics(kInputVolume);
+  stats_reporter.UpdateStatistics(kInputVolume);
+  const auto& update_stats = stats_reporter.volume_update_stats();
+  EXPECT_EQ(update_stats.num_decreases, 0);
+  EXPECT_EQ(update_stats.sum_decreases, 0);
+  EXPECT_EQ(update_stats.num_increases, 0);
+  EXPECT_EQ(update_stats.sum_increases, 0);
+}
+
+TEST_P(InputVolumeStatsReporterTest,
+       CheckVolumeUpdateStatsAfterVolumeIncrease) {
+  constexpr int kInputVolume = 10;
+  InputVolumeStatsReporter stats_reporter(InputVolumeType());
+  stats_reporter.UpdateStatistics(kInputVolume);
+  stats_reporter.UpdateStatistics(kInputVolume + 4);
+  stats_reporter.UpdateStatistics(kInputVolume + 5);
+  const auto& update_stats = stats_reporter.volume_update_stats();
+  EXPECT_EQ(update_stats.num_decreases, 0);
+  EXPECT_EQ(update_stats.sum_decreases, 0);
+  EXPECT_EQ(update_stats.num_increases, 2);
+  EXPECT_EQ(update_stats.sum_increases, 5);
+}
+
+TEST_P(InputVolumeStatsReporterTest,
+       CheckVolumeUpdateStatsAfterVolumeDecrease) {
+  constexpr int kInputVolume = 10;
+  InputVolumeStatsReporter stats_reporter(InputVolumeType());
+  stats_reporter.UpdateStatistics(kInputVolume);
+  stats_reporter.UpdateStatistics(kInputVolume - 4);
+  stats_reporter.UpdateStatistics(kInputVolume - 5);
+  const auto& stats_update = stats_reporter.volume_update_stats();
+  EXPECT_EQ(stats_update.num_decreases, 2);
+  EXPECT_EQ(stats_update.sum_decreases, 5);
+  EXPECT_EQ(stats_update.num_increases, 0);
+  EXPECT_EQ(stats_update.sum_increases, 0);
+}
+
+TEST_P(InputVolumeStatsReporterTest, CheckVolumeUpdateStatsAfterReset) {
+  InputVolumeStatsReporter stats_reporter(InputVolumeType());
+  constexpr int kInputVolume = 10;
+  stats_reporter.UpdateStatistics(kInputVolume);
+  // Update until the periodic reset.
+  for (int i = 0; i < kFramesIn60Seconds - 2; i += 2) {
+    stats_reporter.UpdateStatistics(kInputVolume + 2);
+    stats_reporter.UpdateStatistics(kInputVolume);
+  }
+  const auto& stats_before_reset = stats_reporter.volume_update_stats();
+  EXPECT_EQ(stats_before_reset.num_decreases, kFramesIn60Seconds / 2 - 1);
+  EXPECT_EQ(stats_before_reset.sum_decreases, kFramesIn60Seconds - 2);
+  EXPECT_EQ(stats_before_reset.num_increases, kFramesIn60Seconds / 2 - 1);
+  EXPECT_EQ(stats_before_reset.sum_increases, kFramesIn60Seconds - 2);
+  stats_reporter.UpdateStatistics(kInputVolume + 2);
+  const auto& stats_during_reset = stats_reporter.volume_update_stats();
+  EXPECT_EQ(stats_during_reset.num_decreases, 0);
+  EXPECT_EQ(stats_during_reset.sum_decreases, 0);
+  EXPECT_EQ(stats_during_reset.num_increases, 0);
+  EXPECT_EQ(stats_during_reset.sum_increases, 0);
+  stats_reporter.UpdateStatistics(kInputVolume);
+  stats_reporter.UpdateStatistics(kInputVolume + 3);
+  const auto& stats_after_reset = stats_reporter.volume_update_stats();
+  EXPECT_EQ(stats_after_reset.num_decreases, 1);
+  EXPECT_EQ(stats_after_reset.sum_decreases, 2);
+  EXPECT_EQ(stats_after_reset.num_increases, 1);
+  EXPECT_EQ(stats_after_reset.sum_increases, 3);
+}
+
+INSTANTIATE_TEST_SUITE_P(,
+                         InputVolumeStatsReporterTest,
+                         ::testing::Values(InputVolumeType::kApplied,
+                                           InputVolumeType::kRecommended));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc b/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc
new file mode 100644
index 0000000000..bb6e038514
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc
@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
+
+#include <algorithm>
+#include <iterator>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
+    InterpolatedGainCurve::approximation_params_x_;
+
+constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
+    InterpolatedGainCurve::approximation_params_m_;
+
+constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
+    InterpolatedGainCurve::approximation_params_q_;
+
+InterpolatedGainCurve::InterpolatedGainCurve(
+    ApmDataDumper* apm_data_dumper,
+    absl::string_view histogram_name_prefix)
+    : region_logger_(
+          (rtc::StringBuilder("WebRTC.Audio.")
+           << histogram_name_prefix << ".FixedDigitalGainCurveRegion.Identity")
+              .str(),
+          (rtc::StringBuilder("WebRTC.Audio.")
+           << histogram_name_prefix << ".FixedDigitalGainCurveRegion.Knee")
+              .str(),
+          (rtc::StringBuilder("WebRTC.Audio.")
+           << histogram_name_prefix << ".FixedDigitalGainCurveRegion.Limiter")
+              .str(),
+          (rtc::StringBuilder("WebRTC.Audio.")
+           << histogram_name_prefix
+           << ".FixedDigitalGainCurveRegion.Saturation")
+              .str()),
+      apm_data_dumper_(apm_data_dumper) {}
+
+InterpolatedGainCurve::~InterpolatedGainCurve() {
+  if (stats_.available) {
+    RTC_DCHECK(apm_data_dumper_);
+    apm_data_dumper_->DumpRaw("agc2_interp_gain_curve_lookups_identity",
+                              stats_.look_ups_identity_region);
+    apm_data_dumper_->DumpRaw("agc2_interp_gain_curve_lookups_knee",
+                              stats_.look_ups_knee_region);
+    apm_data_dumper_->DumpRaw("agc2_interp_gain_curve_lookups_limiter",
+                              stats_.look_ups_limiter_region);
+    apm_data_dumper_->DumpRaw("agc2_interp_gain_curve_lookups_saturation",
+                              stats_.look_ups_saturation_region);
+    region_logger_.LogRegionStats(stats_);
+  }
+}
+
+InterpolatedGainCurve::RegionLogger::RegionLogger(
+    absl::string_view identity_histogram_name,
+    absl::string_view knee_histogram_name,
+    absl::string_view limiter_histogram_name,
+    absl::string_view saturation_histogram_name)
+    : identity_histogram(
+          metrics::HistogramFactoryGetCounts(identity_histogram_name,
+                                             1,
+                                             10000,
+                                             50)),
+      knee_histogram(metrics::HistogramFactoryGetCounts(knee_histogram_name,
+                                                        1,
+                                                        10000,
+                                                        50)),
+      limiter_histogram(
+          metrics::HistogramFactoryGetCounts(limiter_histogram_name,
+                                             1,
+                                             10000,
+                                             50)),
+      saturation_histogram(
+          metrics::HistogramFactoryGetCounts(saturation_histogram_name,
+                                             1,
+                                             10000,
+                                             50)) {}
+
+InterpolatedGainCurve::RegionLogger::~RegionLogger() = default;
+
+void InterpolatedGainCurve::RegionLogger::LogRegionStats(
+    const InterpolatedGainCurve::Stats& stats) const {
+  using Region = InterpolatedGainCurve::GainCurveRegion;
+  const int duration_s =
+      stats.region_duration_frames / (1000 / kFrameDurationMs);
+
+  switch (stats.region) {
+    case Region::kIdentity: {
+      if (identity_histogram) {
+        metrics::HistogramAdd(identity_histogram, duration_s);
+      }
+      break;
+    }
+    case Region::kKnee: {
+      if (knee_histogram) {
+        metrics::HistogramAdd(knee_histogram, duration_s);
+      }
+      break;
+    }
+    case Region::kLimiter: {
+      if (limiter_histogram) {
+        metrics::HistogramAdd(limiter_histogram, duration_s);
+      }
+      break;
+    }
+    case Region::kSaturation: {
+      if (saturation_histogram) {
+        metrics::HistogramAdd(saturation_histogram, duration_s);
+      }
+      break;
+    }
+    default: {
+      RTC_DCHECK_NOTREACHED();
+    }
+  }
+}
+
+void InterpolatedGainCurve::UpdateStats(float input_level) const {
+  stats_.available = true;
+
+  GainCurveRegion region;
+
+  if (input_level < approximation_params_x_[0]) {
+    stats_.look_ups_identity_region++;
+    region = GainCurveRegion::kIdentity;
+  } else if (input_level <
+             approximation_params_x_[kInterpolatedGainCurveKneePoints - 1]) {
+    stats_.look_ups_knee_region++;
+    region = GainCurveRegion::kKnee;
+  } else if (input_level < kMaxInputLevelLinear) {
+    stats_.look_ups_limiter_region++;
+    region = GainCurveRegion::kLimiter;
+  } else {
+    stats_.look_ups_saturation_region++;
+    region = GainCurveRegion::kSaturation;
+  }
+
+  if (region == stats_.region) {
+    ++stats_.region_duration_frames;
+  } else {
+    region_logger_.LogRegionStats(stats_);
+
+    stats_.region_duration_frames = 0;
+    stats_.region = region;
+  }
+}
+
+// Looks up a gain to apply given a non-negative input level.
+// The cost of this operation depends on the region in which `input_level`
+// falls.
+// For the identity and the saturation regions the cost is O(1).
+// For the other regions, namely knee and limiter, the cost is
+// O(2 + log2(`LightkInterpolatedGainCurveTotalPoints`), plus O(1) for the
+// linear interpolation (one product and one sum).
+float InterpolatedGainCurve::LookUpGainToApply(float input_level) const {
+  UpdateStats(input_level);
+
+  if (input_level <= approximation_params_x_[0]) {
+    // Identity region.
+    return 1.0f;
+  }
+
+  if (input_level >= kMaxInputLevelLinear) {
+    // Saturating lower bound. The saturing samples exactly hit the clipping
+    // level. This method achieves has the lowest harmonic distorsion, but it
+    // may reduce the amplitude of the non-saturating samples too much.
+    return 32768.f / input_level;
+  }
+
+  // Knee and limiter regions; find the linear piece index. Spelling
+  // out the complete type was the only way to silence both the clang
+  // plugin and the windows compilers.
+  std::array<float, kInterpolatedGainCurveTotalPoints>::const_iterator it =
+      std::lower_bound(approximation_params_x_.begin(),
+                       approximation_params_x_.end(), input_level);
+  const size_t index = std::distance(approximation_params_x_.begin(), it) - 1;
+  RTC_DCHECK_LE(0, index);
+  RTC_DCHECK_LT(index, approximation_params_m_.size());
+  RTC_DCHECK_LE(approximation_params_x_[index], input_level);
+  if (index < approximation_params_m_.size() - 1) {
+    RTC_DCHECK_LE(input_level, approximation_params_x_[index + 1]);
+  }
+
+  // Piece-wise linear interploation.
+  const float gain = approximation_params_m_[index] * input_level +
+                     approximation_params_q_[index];
+  RTC_DCHECK_LE(0.f, gain);
+  return gain;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve.h b/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve.h
new file mode 100644
index 0000000000..8dd3e48f21
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve.h
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_INTERPOLATED_GAIN_CURVE_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_INTERPOLATED_GAIN_CURVE_H_
+
+#include <array>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+constexpr float kInputLevelScalingFactor = 32768.0f;
+
+// Defined as DbfsToLinear(kLimiterMaxInputLevelDbFs)
+constexpr float kMaxInputLevelLinear = static_cast<float>(36766.300710566735);
+
+// Interpolated gain curve using under-approximation to avoid saturation.
+//
+// The goal of this class is allowing fast look ups to get an accurate
+// estimates of the gain to apply given an estimated input level.
+class InterpolatedGainCurve {
+ public:
+  enum class GainCurveRegion {
+    kIdentity = 0,
+    kKnee = 1,
+    kLimiter = 2,
+    kSaturation = 3
+  };
+
+  struct Stats {
+    // Region in which the output level equals the input one.
+    size_t look_ups_identity_region = 0;
+    // Smoothing between the identity and the limiter regions.
+    size_t look_ups_knee_region = 0;
+    // Limiter region in which the output and input levels are linearly related.
+    size_t look_ups_limiter_region = 0;
+    // Region in which saturation may occur since the input level is beyond the
+    // maximum expected by the limiter.
+    size_t look_ups_saturation_region = 0;
+    // True if stats have been populated.
+    bool available = false;
+
+    // The current region, and for how many frames the level has been
+    // in that region.
+    GainCurveRegion region = GainCurveRegion::kIdentity;
+    int64_t region_duration_frames = 0;
+  };
+
+  InterpolatedGainCurve(ApmDataDumper* apm_data_dumper,
+                        absl::string_view histogram_name_prefix);
+  ~InterpolatedGainCurve();
+
+  InterpolatedGainCurve(const InterpolatedGainCurve&) = delete;
+  InterpolatedGainCurve& operator=(const InterpolatedGainCurve&) = delete;
+
+  Stats get_stats() const { return stats_; }
+
+  // Given a non-negative input level (linear scale), a scalar factor to apply
+  // to a sub-frame is returned.
+  // Levels above kLimiterMaxInputLevelDbFs will be reduced to 0 dBFS
+  // after applying this gain
+  float LookUpGainToApply(float input_level) const;
+
+ private:
+  // For comparing 'approximation_params_*_' with ones computed by
+  // ComputeInterpolatedGainCurve.
+  FRIEND_TEST_ALL_PREFIXES(GainController2InterpolatedGainCurve,
+                           CheckApproximationParams);
+
+  struct RegionLogger {
+    metrics::Histogram* identity_histogram;
+    metrics::Histogram* knee_histogram;
+    metrics::Histogram* limiter_histogram;
+    metrics::Histogram* saturation_histogram;
+
+    RegionLogger(absl::string_view identity_histogram_name,
+                 absl::string_view knee_histogram_name,
+                 absl::string_view limiter_histogram_name,
+                 absl::string_view saturation_histogram_name);
+
+    ~RegionLogger();
+
+    void LogRegionStats(const InterpolatedGainCurve::Stats& stats) const;
+  } region_logger_;
+
+  void UpdateStats(float input_level) const;
+
+  ApmDataDumper* const apm_data_dumper_;
+
+  static constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
+      approximation_params_x_ = {
+          {30057.296875,    30148.986328125, 30240.67578125,  30424.052734375,
+           30607.4296875,   30790.806640625, 30974.18359375,  31157.560546875,
+           31340.939453125, 31524.31640625,  31707.693359375, 31891.0703125,
+           32074.447265625, 32257.82421875,  32441.201171875, 32624.580078125,
+           32807.95703125,  32991.33203125,  33174.7109375,   33358.08984375,
+           33541.46484375,  33724.84375,     33819.53515625,  34009.5390625,
+           34200.05859375,  34389.81640625,  34674.48828125,  35054.375,
+           35434.86328125,  35814.81640625,  36195.16796875,  36575.03125}};
+  static constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
+      approximation_params_m_ = {
+          {-3.515235675877192989e-07, -1.050251626111275982e-06,
+           -2.085213736791047268e-06, -3.443004743530764244e-06,
+           -4.773849468620028347e-06, -6.077375928725814447e-06,
+           -7.353257842623861507e-06, -8.601219633419532329e-06,
+           -9.821013009059242904e-06, -1.101243378798244521e-05,
+           -1.217532644659513608e-05, -1.330956911260727793e-05,
+           -1.441507538402220234e-05, -1.549179251014720649e-05,
+           -1.653970684856176376e-05, -1.755882840370759368e-05,
+           -1.854918446042574942e-05, -1.951086778717581183e-05,
+           -2.044398024736437947e-05, -2.1348627342376858e-05,
+           -2.222496914328075945e-05, -2.265374678245279938e-05,
+           -2.242570917587727308e-05, -2.220122041762806475e-05,
+           -2.19802095671184361e-05,  -2.176260204578284174e-05,
+           -2.133731686626560986e-05, -2.092481918225530535e-05,
+           -2.052459603874012828e-05, -2.013615448959171772e-05,
+           -1.975903069251216948e-05, -1.939277899509761482e-05}};
+
+  static constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
+      approximation_params_q_ = {
+          {1.010565876960754395, 1.031631827354431152, 1.062929749488830566,
+           1.104239225387573242, 1.144973039627075195, 1.185109615325927734,
+           1.224629044532775879, 1.263512492179870605, 1.301741957664489746,
+           1.339300632476806641, 1.376173257827758789, 1.412345528602600098,
+           1.447803974151611328, 1.482536554336547852, 1.516532182693481445,
+           1.549780607223510742, 1.582272171974182129, 1.613999366760253906,
+           1.644955039024353027, 1.675132393836975098, 1.704526185989379883,
+           1.718986630439758301, 1.711274504661560059, 1.703639745712280273,
+           1.696081161499023438, 1.688597679138183594, 1.673851132392883301,
+           1.659391283988952637, 1.645209431648254395, 1.631297469139099121,
+           1.617647409439086914, 1.604251742362976074}};
+
+  // Stats.
+  mutable Stats stats_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_INTERPOLATED_GAIN_CURVE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
new file mode 100644
index 0000000000..7861ae997d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
+
+#include <array>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/agc2/compute_interpolated_gain_curve.h"
+#include "modules/audio_processing/agc2/limiter_db_gain_curve.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+namespace {
+
+constexpr double kLevelEpsilon = 1e-2 * kMaxAbsFloatS16Value;
+constexpr float kInterpolatedGainCurveTolerance = 1.f / 32768.f;
+ApmDataDumper apm_data_dumper(0);
+static_assert(std::is_trivially_destructible<LimiterDbGainCurve>::value, "");
+const LimiterDbGainCurve limiter;
+
+}  // namespace
+
+TEST(GainController2InterpolatedGainCurve, CreateUse) {
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  const auto levels = test::LinSpace(
+      kLevelEpsilon, DbfsToFloatS16(limiter.max_input_level_db() + 1), 500);
+  for (const auto level : levels) {
+    EXPECT_GE(igc.LookUpGainToApply(level), 0.0f);
+  }
+}
+
+TEST(GainController2InterpolatedGainCurve, CheckValidOutput) {
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  const auto levels = test::LinSpace(
+      kLevelEpsilon, limiter.max_input_level_linear() * 2.0, 500);
+  for (const auto level : levels) {
+    SCOPED_TRACE(std::to_string(level));
+    const float gain = igc.LookUpGainToApply(level);
+    EXPECT_LE(0.0f, gain);
+    EXPECT_LE(gain, 1.0f);
+  }
+}
+
+TEST(GainController2InterpolatedGainCurve, CheckMonotonicity) {
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  const auto levels = test::LinSpace(
+      kLevelEpsilon, limiter.max_input_level_linear() + kLevelEpsilon + 0.5,
+      500);
+  float prev_gain = igc.LookUpGainToApply(0.0f);
+  for (const auto level : levels) {
+    const float gain = igc.LookUpGainToApply(level);
+    EXPECT_GE(prev_gain, gain);
+    prev_gain = gain;
+  }
+}
+
+TEST(GainController2InterpolatedGainCurve, CheckApproximation) {
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  const auto levels = test::LinSpace(
+      kLevelEpsilon, limiter.max_input_level_linear() - kLevelEpsilon, 500);
+  for (const auto level : levels) {
+    SCOPED_TRACE(std::to_string(level));
+    EXPECT_LT(
+        std::fabs(limiter.GetGainLinear(level) - igc.LookUpGainToApply(level)),
+        kInterpolatedGainCurveTolerance);
+  }
+}
+
+TEST(GainController2InterpolatedGainCurve, CheckRegionBoundaries) {
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  const std::vector<double> levels{
+      {kLevelEpsilon, limiter.knee_start_linear() + kLevelEpsilon,
+       limiter.limiter_start_linear() + kLevelEpsilon,
+       limiter.max_input_level_linear() + kLevelEpsilon}};
+  for (const auto level : levels) {
+    igc.LookUpGainToApply(level);
+  }
+
+  const auto stats = igc.get_stats();
+  EXPECT_EQ(1ul, stats.look_ups_identity_region);
+  EXPECT_EQ(1ul, stats.look_ups_knee_region);
+  EXPECT_EQ(1ul, stats.look_ups_limiter_region);
+  EXPECT_EQ(1ul, stats.look_ups_saturation_region);
+}
+
+TEST(GainController2InterpolatedGainCurve, CheckIdentityRegion) {
+  constexpr size_t kNumSteps = 10;
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  const auto levels =
+      test::LinSpace(kLevelEpsilon, limiter.knee_start_linear(), kNumSteps);
+  for (const auto level : levels) {
+    SCOPED_TRACE(std::to_string(level));
+    EXPECT_EQ(1.0f, igc.LookUpGainToApply(level));
+  }
+
+  const auto stats = igc.get_stats();
+  EXPECT_EQ(kNumSteps - 1, stats.look_ups_identity_region);
+  EXPECT_EQ(1ul, stats.look_ups_knee_region);
+  EXPECT_EQ(0ul, stats.look_ups_limiter_region);
+  EXPECT_EQ(0ul, stats.look_ups_saturation_region);
+}
+
+TEST(GainController2InterpolatedGainCurve, CheckNoOverApproximationKnee) {
+  constexpr size_t kNumSteps = 10;
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  const auto levels =
+      test::LinSpace(limiter.knee_start_linear() + kLevelEpsilon,
+                     limiter.limiter_start_linear(), kNumSteps);
+  for (const auto level : levels) {
+    SCOPED_TRACE(std::to_string(level));
+    // Small tolerance added (needed because comparing a float with a double).
+    EXPECT_LE(igc.LookUpGainToApply(level),
+              limiter.GetGainLinear(level) + 1e-7);
+  }
+
+  const auto stats = igc.get_stats();
+  EXPECT_EQ(0ul, stats.look_ups_identity_region);
+  EXPECT_EQ(kNumSteps - 1, stats.look_ups_knee_region);
+  EXPECT_EQ(1ul, stats.look_ups_limiter_region);
+  EXPECT_EQ(0ul, stats.look_ups_saturation_region);
+}
+
+TEST(GainController2InterpolatedGainCurve, CheckNoOverApproximationBeyondKnee) {
+  constexpr size_t kNumSteps = 10;
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  const auto levels = test::LinSpace(
+      limiter.limiter_start_linear() + kLevelEpsilon,
+      limiter.max_input_level_linear() - kLevelEpsilon, kNumSteps);
+  for (const auto level : levels) {
+    SCOPED_TRACE(std::to_string(level));
+    // Small tolerance added (needed because comparing a float with a double).
+    EXPECT_LE(igc.LookUpGainToApply(level),
+              limiter.GetGainLinear(level) + 1e-7);
+  }
+
+  const auto stats = igc.get_stats();
+  EXPECT_EQ(0ul, stats.look_ups_identity_region);
+  EXPECT_EQ(0ul, stats.look_ups_knee_region);
+  EXPECT_EQ(kNumSteps, stats.look_ups_limiter_region);
+  EXPECT_EQ(0ul, stats.look_ups_saturation_region);
+}
+
+TEST(GainController2InterpolatedGainCurve,
+     CheckNoOverApproximationWithSaturation) {
+  constexpr size_t kNumSteps = 3;
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  const auto levels = test::LinSpace(
+      limiter.max_input_level_linear() + kLevelEpsilon,
+      limiter.max_input_level_linear() + kLevelEpsilon + 0.5, kNumSteps);
+  for (const auto level : levels) {
+    SCOPED_TRACE(std::to_string(level));
+    EXPECT_LE(igc.LookUpGainToApply(level), limiter.GetGainLinear(level));
+  }
+
+  const auto stats = igc.get_stats();
+  EXPECT_EQ(0ul, stats.look_ups_identity_region);
+  EXPECT_EQ(0ul, stats.look_ups_knee_region);
+  EXPECT_EQ(0ul, stats.look_ups_limiter_region);
+  EXPECT_EQ(kNumSteps, stats.look_ups_saturation_region);
+}
+
+TEST(GainController2InterpolatedGainCurve, CheckApproximationParams) {
+  test::InterpolatedParameters parameters =
+      test::ComputeInterpolatedGainCurveApproximationParams();
+
+  InterpolatedGainCurve igc(&apm_data_dumper, "");
+
+  for (size_t i = 0; i < kInterpolatedGainCurveTotalPoints; ++i) {
+    // The tolerance levels are chosen to account for deviations due
+    // to computing with single precision floating point numbers.
+    EXPECT_NEAR(igc.approximation_params_x_[i],
+                parameters.computed_approximation_params_x[i], 0.9f);
+    EXPECT_NEAR(igc.approximation_params_m_[i],
+                parameters.computed_approximation_params_m[i], 0.00001f);
+    EXPECT_NEAR(igc.approximation_params_q_[i],
+                parameters.computed_approximation_params_q[i], 0.001f);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/limiter.cc b/third_party/libwebrtc/modules/audio_processing/agc2/limiter.cc
new file mode 100644
index 0000000000..7a1e2202be
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/limiter.cc
@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/limiter.h"
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+
+// This constant affects the way scaling factors are interpolated for the first
+// sub-frame of a frame. Only in the case in which the first sub-frame has an
+// estimated level which is greater than the that of the previous analyzed
+// sub-frame, linear interpolation is replaced with a power function which
+// reduces the chances of over-shooting (and hence saturation), however reducing
+// the fixed gain effectiveness.
+constexpr float kAttackFirstSubframeInterpolationPower = 8.0f;
+
+void InterpolateFirstSubframe(float last_factor,
+                              float current_factor,
+                              rtc::ArrayView<float> subframe) {
+  const int n = rtc::dchecked_cast<int>(subframe.size());
+  constexpr float p = kAttackFirstSubframeInterpolationPower;
+  for (int i = 0; i < n; ++i) {
+    subframe[i] = std::pow(1.f - i / n, p) * (last_factor - current_factor) +
+                  current_factor;
+  }
+}
+
+void ComputePerSampleSubframeFactors(
+    const std::array<float, kSubFramesInFrame + 1>& scaling_factors,
+    int samples_per_channel,
+    rtc::ArrayView<float> per_sample_scaling_factors) {
+  const int num_subframes = scaling_factors.size() - 1;
+  const int subframe_size =
+      rtc::CheckedDivExact(samples_per_channel, num_subframes);
+
+  // Handle first sub-frame differently in case of attack.
+  const bool is_attack = scaling_factors[0] > scaling_factors[1];
+  if (is_attack) {
+    InterpolateFirstSubframe(
+        scaling_factors[0], scaling_factors[1],
+        rtc::ArrayView<float>(
+            per_sample_scaling_factors.subview(0, subframe_size)));
+  }
+
+  for (int i = is_attack ? 1 : 0; i < num_subframes; ++i) {
+    const int subframe_start = i * subframe_size;
+    const float scaling_start = scaling_factors[i];
+    const float scaling_end = scaling_factors[i + 1];
+    const float scaling_diff = (scaling_end - scaling_start) / subframe_size;
+    for (int j = 0; j < subframe_size; ++j) {
+      per_sample_scaling_factors[subframe_start + j] =
+          scaling_start + scaling_diff * j;
+    }
+  }
+}
+
+void ScaleSamples(rtc::ArrayView<const float> per_sample_scaling_factors,
+                  AudioFrameView<float> signal) {
+  const int samples_per_channel = signal.samples_per_channel();
+  RTC_DCHECK_EQ(samples_per_channel, per_sample_scaling_factors.size());
+  for (int i = 0; i < signal.num_channels(); ++i) {
+    rtc::ArrayView<float> channel = signal.channel(i);
+    for (int j = 0; j < samples_per_channel; ++j) {
+      channel[j] = rtc::SafeClamp(channel[j] * per_sample_scaling_factors[j],
+                                  kMinFloatS16Value, kMaxFloatS16Value);
+    }
+  }
+}
+
+void CheckLimiterSampleRate(int sample_rate_hz) {
+  // Check that per_sample_scaling_factors_ is large enough.
+  RTC_DCHECK_LE(sample_rate_hz,
+                kMaximalNumberOfSamplesPerChannel * 1000 / kFrameDurationMs);
+}
+
+}  // namespace
+
+Limiter::Limiter(int sample_rate_hz,
+                 ApmDataDumper* apm_data_dumper,
+                 absl::string_view histogram_name)
+    : interp_gain_curve_(apm_data_dumper, histogram_name),
+      level_estimator_(sample_rate_hz, apm_data_dumper),
+      apm_data_dumper_(apm_data_dumper) {
+  CheckLimiterSampleRate(sample_rate_hz);
+}
+
+Limiter::~Limiter() = default;
+
+void Limiter::Process(AudioFrameView<float> signal) {
+  const std::array<float, kSubFramesInFrame> level_estimate =
+      level_estimator_.ComputeLevel(signal);
+
+  RTC_DCHECK_EQ(level_estimate.size() + 1, scaling_factors_.size());
+  scaling_factors_[0] = last_scaling_factor_;
+  std::transform(level_estimate.begin(), level_estimate.end(),
+                 scaling_factors_.begin() + 1, [this](float x) {
+                   return interp_gain_curve_.LookUpGainToApply(x);
+                 });
+
+  const int samples_per_channel = signal.samples_per_channel();
+  RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
+
+  auto per_sample_scaling_factors = rtc::ArrayView<float>(
+      &per_sample_scaling_factors_[0], samples_per_channel);
+  ComputePerSampleSubframeFactors(scaling_factors_, samples_per_channel,
+                                  per_sample_scaling_factors);
+  ScaleSamples(per_sample_scaling_factors, signal);
+
+  last_scaling_factor_ = scaling_factors_.back();
+
+  // Dump data for debug.
+  apm_data_dumper_->DumpRaw("agc2_limiter_last_scaling_factor",
+                            last_scaling_factor_);
+  apm_data_dumper_->DumpRaw(
+      "agc2_limiter_region",
+      static_cast<int>(interp_gain_curve_.get_stats().region));
+}
+
+InterpolatedGainCurve::Stats Limiter::GetGainCurveStats() const {
+  return interp_gain_curve_.get_stats();
+}
+
+void Limiter::SetSampleRate(int sample_rate_hz) {
+  CheckLimiterSampleRate(sample_rate_hz);
+  level_estimator_.SetSampleRate(sample_rate_hz);
+}
+
+void Limiter::Reset() {
+  level_estimator_.Reset();
+}
+
+float Limiter::LastAudioLevel() const {
+  return level_estimator_.LastAudioLevel();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/limiter.h b/third_party/libwebrtc/modules/audio_processing/agc2/limiter.h
new file mode 100644
index 0000000000..d4d556349c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/limiter.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
+
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+class ApmDataDumper;
+
+class Limiter {
+ public:
+  Limiter(int sample_rate_hz,
+          ApmDataDumper* apm_data_dumper,
+          absl::string_view histogram_name_prefix);
+  Limiter(const Limiter& limiter) = delete;
+  Limiter& operator=(const Limiter& limiter) = delete;
+  ~Limiter();
+
+  // Applies limiter and hard-clipping to `signal`.
+  void Process(AudioFrameView<float> signal);
+  InterpolatedGainCurve::Stats GetGainCurveStats() const;
+
+  // Supported rates must be
+  // * supported by FixedDigitalLevelEstimator
+  // * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
+  //   so that samples_per_channel fit in the
+  //   per_sample_scaling_factors_ array.
+  void SetSampleRate(int sample_rate_hz);
+
+  // Resets the internal state.
+  void Reset();
+
+  float LastAudioLevel() const;
+
+ private:
+  const InterpolatedGainCurve interp_gain_curve_;
+  FixedDigitalLevelEstimator level_estimator_;
+  ApmDataDumper* const apm_data_dumper_ = nullptr;
+
+  // Work array containing the sub-frame scaling factors to be interpolated.
+  std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
+  std::array<float, kMaximalNumberOfSamplesPerChannel>
+      per_sample_scaling_factors_ = {};
+  float last_scaling_factor_ = 1.f;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/limiter_db_gain_curve.cc b/third_party/libwebrtc/modules/audio_processing/agc2/limiter_db_gain_curve.cc
new file mode 100644
index 0000000000..d47c0b2e17
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/limiter_db_gain_curve.cc
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/limiter_db_gain_curve.h"
+
+#include <cmath>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+double ComputeKneeStart(double max_input_level_db,
+                        double knee_smoothness_db,
+                        double compression_ratio) {
+  RTC_CHECK_LT((compression_ratio - 1.0) * knee_smoothness_db /
+                   (2.0 * compression_ratio),
+               max_input_level_db);
+  return -knee_smoothness_db / 2.0 -
+         max_input_level_db / (compression_ratio - 1.0);
+}
+
+std::array<double, 3> ComputeKneeRegionPolynomial(double knee_start_dbfs,
+                                                  double knee_smoothness_db,
+                                                  double compression_ratio) {
+  const double a = (1.0 - compression_ratio) /
+                   (2.0 * knee_smoothness_db * compression_ratio);
+  const double b = 1.0 - 2.0 * a * knee_start_dbfs;
+  const double c = a * knee_start_dbfs * knee_start_dbfs;
+  return {{a, b, c}};
+}
+
+double ComputeLimiterD1(double max_input_level_db, double compression_ratio) {
+  return (std::pow(10.0, -max_input_level_db / (20.0 * compression_ratio)) *
+          (1.0 - compression_ratio) / compression_ratio) /
+         kMaxAbsFloatS16Value;
+}
+
+constexpr double ComputeLimiterD2(double compression_ratio) {
+  return (1.0 - 2.0 * compression_ratio) / compression_ratio;
+}
+
+double ComputeLimiterI2(double max_input_level_db,
+                        double compression_ratio,
+                        double gain_curve_limiter_i1) {
+  RTC_CHECK_NE(gain_curve_limiter_i1, 0.f);
+  return std::pow(10.0, -max_input_level_db / (20.0 * compression_ratio)) /
+         gain_curve_limiter_i1 /
+         std::pow(kMaxAbsFloatS16Value, gain_curve_limiter_i1 - 1);
+}
+
+}  // namespace
+
+LimiterDbGainCurve::LimiterDbGainCurve()
+    : max_input_level_linear_(DbfsToFloatS16(max_input_level_db_)),
+      knee_start_dbfs_(ComputeKneeStart(max_input_level_db_,
+                                        knee_smoothness_db_,
+                                        compression_ratio_)),
+      knee_start_linear_(DbfsToFloatS16(knee_start_dbfs_)),
+      limiter_start_dbfs_(knee_start_dbfs_ + knee_smoothness_db_),
+      limiter_start_linear_(DbfsToFloatS16(limiter_start_dbfs_)),
+      knee_region_polynomial_(ComputeKneeRegionPolynomial(knee_start_dbfs_,
+                                                          knee_smoothness_db_,
+                                                          compression_ratio_)),
+      gain_curve_limiter_d1_(
+          ComputeLimiterD1(max_input_level_db_, compression_ratio_)),
+      gain_curve_limiter_d2_(ComputeLimiterD2(compression_ratio_)),
+      gain_curve_limiter_i1_(1.0 / compression_ratio_),
+      gain_curve_limiter_i2_(ComputeLimiterI2(max_input_level_db_,
+                                              compression_ratio_,
+                                              gain_curve_limiter_i1_)) {
+  static_assert(knee_smoothness_db_ > 0.0f, "");
+  static_assert(compression_ratio_ > 1.0f, "");
+  RTC_CHECK_GE(max_input_level_db_, knee_start_dbfs_ + knee_smoothness_db_);
+}
+
+constexpr double LimiterDbGainCurve::max_input_level_db_;
+constexpr double LimiterDbGainCurve::knee_smoothness_db_;
+constexpr double LimiterDbGainCurve::compression_ratio_;
+
+double LimiterDbGainCurve::GetOutputLevelDbfs(double input_level_dbfs) const {
+  if (input_level_dbfs < knee_start_dbfs_) {
+    return input_level_dbfs;
+  } else if (input_level_dbfs < limiter_start_dbfs_) {
+    return GetKneeRegionOutputLevelDbfs(input_level_dbfs);
+  }
+  return GetCompressorRegionOutputLevelDbfs(input_level_dbfs);
+}
+
+double LimiterDbGainCurve::GetGainLinear(double input_level_linear) const {
+  if (input_level_linear < knee_start_linear_) {
+    return 1.0;
+  }
+  return DbfsToFloatS16(
+             GetOutputLevelDbfs(FloatS16ToDbfs(input_level_linear))) /
+         input_level_linear;
+}
+
+// Computes the first derivative of GetGainLinear() in `x`.
+double LimiterDbGainCurve::GetGainFirstDerivativeLinear(double x) const {
+  // Beyond-knee region only.
+  RTC_CHECK_GE(x, limiter_start_linear_ - 1e-7 * kMaxAbsFloatS16Value);
+  return gain_curve_limiter_d1_ *
+         std::pow(x / kMaxAbsFloatS16Value, gain_curve_limiter_d2_);
+}
+
+// Computes the integral of GetGainLinear() in the range [x0, x1].
+double LimiterDbGainCurve::GetGainIntegralLinear(double x0, double x1) const {
+  RTC_CHECK_LE(x0, x1);                     // Valid interval.
+  RTC_CHECK_GE(x0, limiter_start_linear_);  // Beyond-knee region only.
+  auto limiter_integral = [this](const double& x) {
+    return gain_curve_limiter_i2_ * std::pow(x, gain_curve_limiter_i1_);
+  };
+  return limiter_integral(x1) - limiter_integral(x0);
+}
+
+double LimiterDbGainCurve::GetKneeRegionOutputLevelDbfs(
+    double input_level_dbfs) const {
+  return knee_region_polynomial_[0] * input_level_dbfs * input_level_dbfs +
+         knee_region_polynomial_[1] * input_level_dbfs +
+         knee_region_polynomial_[2];
+}
+
+double LimiterDbGainCurve::GetCompressorRegionOutputLevelDbfs(
+    double input_level_dbfs) const {
+  return (input_level_dbfs - max_input_level_db_) / compression_ratio_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/limiter_db_gain_curve.h b/third_party/libwebrtc/modules/audio_processing/agc2/limiter_db_gain_curve.h
new file mode 100644
index 0000000000..9086e26739
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/limiter_db_gain_curve.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_LIMITER_DB_GAIN_CURVE_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_DB_GAIN_CURVE_H_
+
+#include <array>
+
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+
+namespace webrtc {
+
+// A class for computing a limiter gain curve (in dB scale) given a set of
+// hard-coded parameters (namely, kLimiterDbGainCurveMaxInputLevelDbFs,
+// kLimiterDbGainCurveKneeSmoothnessDb, and
+// kLimiterDbGainCurveCompressionRatio). The generated curve consists of four
+// regions: identity (linear), knee (quadratic polynomial), compression
+// (linear), saturation (linear). The aforementioned constants are used to shape
+// the different regions.
+class LimiterDbGainCurve {
+ public:
+  LimiterDbGainCurve();
+
+  double max_input_level_db() const { return max_input_level_db_; }
+  double max_input_level_linear() const { return max_input_level_linear_; }
+  double knee_start_linear() const { return knee_start_linear_; }
+  double limiter_start_linear() const { return limiter_start_linear_; }
+
+  // These methods can be marked 'constexpr' in C++ 14.
+  double GetOutputLevelDbfs(double input_level_dbfs) const;
+  double GetGainLinear(double input_level_linear) const;
+  double GetGainFirstDerivativeLinear(double x) const;
+  double GetGainIntegralLinear(double x0, double x1) const;
+
+ private:
+  double GetKneeRegionOutputLevelDbfs(double input_level_dbfs) const;
+  double GetCompressorRegionOutputLevelDbfs(double input_level_dbfs) const;
+
+  static constexpr double max_input_level_db_ = test::kLimiterMaxInputLevelDbFs;
+  static constexpr double knee_smoothness_db_ = test::kLimiterKneeSmoothnessDb;
+  static constexpr double compression_ratio_ = test::kLimiterCompressionRatio;
+
+  const double max_input_level_linear_;
+
+  // Do not modify signal with level <= knee_start_dbfs_.
+  const double knee_start_dbfs_;
+  const double knee_start_linear_;
+
+  // The upper end of the knee region, which is between knee_start_dbfs_ and
+  // limiter_start_dbfs_.
+  const double limiter_start_dbfs_;
+  const double limiter_start_linear_;
+
+  // Coefficients {a, b, c} of the knee region polynomial
+  // ax^2 + bx + c in the DB scale.
+  const std::array<double, 3> knee_region_polynomial_;
+
+  // Parameters for the computation of the first derivative of GetGainLinear().
+  const double gain_curve_limiter_d1_;
+  const double gain_curve_limiter_d2_;
+
+  // Parameters for the computation of the integral of GetGainLinear().
+  const double gain_curve_limiter_i1_;
+  const double gain_curve_limiter_i2_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_DB_GAIN_CURVE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/limiter_db_gain_curve_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/limiter_db_gain_curve_unittest.cc
new file mode 100644
index 0000000000..049c8d568e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/limiter_db_gain_curve_unittest.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/limiter_db_gain_curve.h"
+
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+
+TEST(FixedDigitalGainController2Limiter, ConstructDestruct) {
+  LimiterDbGainCurve l;
+}
+
+TEST(FixedDigitalGainController2Limiter, GainCurveShouldBeMonotone) {
+  LimiterDbGainCurve l;
+  float last_output_level = 0.f;
+  bool has_last_output_level = false;
+  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
+    const float current_output_level = l.GetOutputLevelDbfs(level);
+    if (!has_last_output_level) {
+      last_output_level = current_output_level;
+      has_last_output_level = true;
+    }
+    EXPECT_LE(last_output_level, current_output_level);
+    last_output_level = current_output_level;
+  }
+}
+
+TEST(FixedDigitalGainController2Limiter, GainCurveShouldBeContinuous) {
+  LimiterDbGainCurve l;
+  float last_output_level = 0.f;
+  bool has_last_output_level = false;
+  constexpr float kMaxDelta = 0.5f;
+  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
+    const float current_output_level = l.GetOutputLevelDbfs(level);
+    if (!has_last_output_level) {
+      last_output_level = current_output_level;
+      has_last_output_level = true;
+    }
+    EXPECT_LE(current_output_level, last_output_level + kMaxDelta);
+    last_output_level = current_output_level;
+  }
+}
+
+TEST(FixedDigitalGainController2Limiter, OutputGainShouldBeLessThanFullScale) {
+  LimiterDbGainCurve l;
+  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
+    const float current_output_level = l.GetOutputLevelDbfs(level);
+    EXPECT_LE(current_output_level, 0.f);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/limiter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/limiter_unittest.cc
new file mode 100644
index 0000000000..e662a7fc89
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/limiter_unittest.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/limiter.h"
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+#include "modules/audio_processing/agc2/vector_float_frame.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+
+TEST(Limiter, LimiterShouldConstructAndRun) {
+  const int sample_rate_hz = 48000;
+  ApmDataDumper apm_data_dumper(0);
+
+  Limiter limiter(sample_rate_hz, &apm_data_dumper, "");
+
+  VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
+                                            kMaxAbsFloatS16Value);
+  limiter.Process(vectors_with_float_frame.float_frame_view());
+}
+
+TEST(Limiter, OutputVolumeAboveThreshold) {
+  const int sample_rate_hz = 48000;
+  const float input_level =
+      (kMaxAbsFloatS16Value + DbfsToFloatS16(test::kLimiterMaxInputLevelDbFs)) /
+      2.f;
+  ApmDataDumper apm_data_dumper(0);
+
+  Limiter limiter(sample_rate_hz, &apm_data_dumper, "");
+
+  // Give the level estimator time to adapt.
+  for (int i = 0; i < 5; ++i) {
+    VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
+                                              input_level);
+    limiter.Process(vectors_with_float_frame.float_frame_view());
+  }
+
+  VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
+                                            input_level);
+  limiter.Process(vectors_with_float_frame.float_frame_view());
+  rtc::ArrayView<const float> channel =
+      vectors_with_float_frame.float_frame_view().channel(0);
+
+  for (const auto& sample : channel) {
+    EXPECT_LT(0.9f * kMaxAbsFloatS16Value, sample);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator.cc b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator.cc
new file mode 100644
index 0000000000..691513b509
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator.cc
@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/noise_level_estimator.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <cmath>
+#include <numeric>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kFramesPerSecond = 100;
+
+float FrameEnergy(const AudioFrameView<const float>& audio) {
+  float energy = 0.0f;
+  for (int k = 0; k < audio.num_channels(); ++k) {
+    float channel_energy =
+        std::accumulate(audio.channel(k).begin(), audio.channel(k).end(), 0.0f,
+                        [](float a, float b) -> float { return a + b * b; });
+    energy = std::max(channel_energy, energy);
+  }
+  return energy;
+}
+
+float EnergyToDbfs(float signal_energy, int num_samples) {
+  RTC_DCHECK_GE(signal_energy, 0.0f);
+  const float rms_square = signal_energy / num_samples;
+  constexpr float kMinDbfs = -90.30899869919436f;
+  if (rms_square <= 1.0f) {
+    return kMinDbfs;
+  }
+  return 10.0f * std::log10(rms_square) + kMinDbfs;
+}
+
+// Updates the noise floor with instant decay and slow attack. This tuning is
+// specific for AGC2, so that (i) it can promptly increase the gain if the noise
+// floor drops (instant decay) and (ii) in case of music or fast speech, due to
+// which the noise floor can be overestimated, the gain reduction is slowed
+// down.
+float SmoothNoiseFloorEstimate(float current_estimate, float new_estimate) {
+  constexpr float kAttack = 0.5f;
+  if (current_estimate < new_estimate) {
+    // Attack phase.
+    return kAttack * new_estimate + (1.0f - kAttack) * current_estimate;
+  }
+  // Instant attack.
+  return new_estimate;
+}
+
+class NoiseFloorEstimator : public NoiseLevelEstimator {
+ public:
+  // Update the noise floor every 5 seconds.
+  static constexpr int kUpdatePeriodNumFrames = 500;
+  static_assert(kUpdatePeriodNumFrames >= 200,
+                "A too small value may cause noise level overestimation.");
+  static_assert(kUpdatePeriodNumFrames <= 1500,
+                "A too large value may make AGC2 slow at reacting to increased "
+                "noise levels.");
+
+  NoiseFloorEstimator(ApmDataDumper* data_dumper) : data_dumper_(data_dumper) {
+    RTC_DCHECK(data_dumper_);
+    // Initially assume that 48 kHz will be used. `Analyze()` will detect the
+    // used sample rate and call `Initialize()` again if needed.
+    Initialize(/*sample_rate_hz=*/48000);
+  }
+  NoiseFloorEstimator(const NoiseFloorEstimator&) = delete;
+  NoiseFloorEstimator& operator=(const NoiseFloorEstimator&) = delete;
+  ~NoiseFloorEstimator() = default;
+
+  float Analyze(const AudioFrameView<const float>& frame) override {
+    // Detect sample rate changes.
+    const int sample_rate_hz =
+        static_cast<int>(frame.samples_per_channel() * kFramesPerSecond);
+    if (sample_rate_hz != sample_rate_hz_) {
+      Initialize(sample_rate_hz);
+    }
+
+    const float frame_energy = FrameEnergy(frame);
+    if (frame_energy <= min_noise_energy_) {
+      // Ignore frames when muted or below the minimum measurable energy.
+      if (data_dumper_)
+        data_dumper_->DumpRaw("agc2_noise_floor_estimator_preliminary_level",
+                              noise_energy_);
+      return EnergyToDbfs(noise_energy_,
+                          static_cast<int>(frame.samples_per_channel()));
+    }
+
+    if (preliminary_noise_energy_set_) {
+      preliminary_noise_energy_ =
+          std::min(preliminary_noise_energy_, frame_energy);
+    } else {
+      preliminary_noise_energy_ = frame_energy;
+      preliminary_noise_energy_set_ = true;
+    }
+    if (data_dumper_)
+      data_dumper_->DumpRaw("agc2_noise_floor_estimator_preliminary_level",
+                            preliminary_noise_energy_);
+
+    if (counter_ == 0) {
+      // Full period observed.
+      first_period_ = false;
+      // Update the estimated noise floor energy with the preliminary
+      // estimation.
+      noise_energy_ = SmoothNoiseFloorEstimate(
+          /*current_estimate=*/noise_energy_,
+          /*new_estimate=*/preliminary_noise_energy_);
+      // Reset for a new observation period.
+      counter_ = kUpdatePeriodNumFrames;
+      preliminary_noise_energy_set_ = false;
+    } else if (first_period_) {
+      // While analyzing the signal during the initial period, continuously
+      // update the estimated noise energy, which is monotonic.
+      noise_energy_ = preliminary_noise_energy_;
+      counter_--;
+    } else {
+      // During the observation period it's only allowed to lower the energy.
+      noise_energy_ = std::min(noise_energy_, preliminary_noise_energy_);
+      counter_--;
+    }
+
+    float noise_rms_dbfs = EnergyToDbfs(
+        noise_energy_, static_cast<int>(frame.samples_per_channel()));
+    if (data_dumper_)
+      data_dumper_->DumpRaw("agc2_noise_rms_dbfs", noise_rms_dbfs);
+
+    return noise_rms_dbfs;
+  }
+
+ private:
+  void Initialize(int sample_rate_hz) {
+    sample_rate_hz_ = sample_rate_hz;
+    first_period_ = true;
+    preliminary_noise_energy_set_ = false;
+    // Initialize the minimum noise energy to -84 dBFS.
+    min_noise_energy_ = sample_rate_hz * 2.0f * 2.0f / kFramesPerSecond;
+    preliminary_noise_energy_ = min_noise_energy_;
+    noise_energy_ = min_noise_energy_;
+    counter_ = kUpdatePeriodNumFrames;
+  }
+
+  ApmDataDumper* const data_dumper_;
+  int sample_rate_hz_;
+  float min_noise_energy_;
+  bool first_period_;
+  bool preliminary_noise_energy_set_;
+  float preliminary_noise_energy_;
+  float noise_energy_;
+  int counter_;
+};
+
+}  // namespace
+
+std::unique_ptr<NoiseLevelEstimator> CreateNoiseFloorEstimator(
+    ApmDataDumper* data_dumper) {
+  return std::make_unique<NoiseFloorEstimator>(data_dumper);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator.h b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator.h
new file mode 100644
index 0000000000..9f3b957486
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_
+
+#include <memory>
+
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+class ApmDataDumper;
+
+// Noise level estimator interface.
+class NoiseLevelEstimator {
+ public:
+  virtual ~NoiseLevelEstimator() = default;
+  // Analyzes a 10 ms `frame`, updates the noise level estimation and returns
+  // the value for the latter in dBFS.
+  virtual float Analyze(const AudioFrameView<const float>& frame) = 0;
+};
+
+// Creates a noise level estimator based on noise floor detection.
+std::unique_ptr<NoiseLevelEstimator> CreateNoiseFloorEstimator(
+    ApmDataDumper* data_dumper);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_gn/moz.build
new file mode 100644
index 0000000000..ba000d3862
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("noise_level_estimator_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
new file mode 100644
index 0000000000..8168c5a229
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/noise_level_estimator.h"
+
+#include <array>
+#include <cmath>
+#include <functional>
+#include <limits>
+
+#include "api/function_view.h"
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+#include "modules/audio_processing/agc2/vector_float_frame.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kNumIterations = 200;
+constexpr int kFramesPerSecond = 100;
+
+// Runs the noise estimator on audio generated by 'sample_generator'
+// for kNumIterations. Returns the last noise level estimate.
+float RunEstimator(rtc::FunctionView<float()> sample_generator,
+                   NoiseLevelEstimator& estimator,
+                   int sample_rate_hz) {
+  const int samples_per_channel =
+      rtc::CheckedDivExact(sample_rate_hz, kFramesPerSecond);
+  VectorFloatFrame signal(1, samples_per_channel, 0.0f);
+  for (int i = 0; i < kNumIterations; ++i) {
+    AudioFrameView<float> frame_view = signal.float_frame_view();
+    for (int j = 0; j < samples_per_channel; ++j) {
+      frame_view.channel(0)[j] = sample_generator();
+    }
+    estimator.Analyze(frame_view);
+  }
+  return estimator.Analyze(signal.float_frame_view());
+}
+
+class NoiseEstimatorParametrization : public ::testing::TestWithParam<int> {
+ protected:
+  int sample_rate_hz() const { return GetParam(); }
+};
+
+// Checks that full scale white noise maps to about -5.5 dBFS.
+TEST_P(NoiseEstimatorParametrization, NoiseFloorEstimatorWithRandomNoise) {
+  ApmDataDumper data_dumper(0);
+  auto estimator = CreateNoiseFloorEstimator(&data_dumper);
+
+  test::WhiteNoiseGenerator gen(/*min_amplitude=*/test::kMinS16,
+                                /*max_amplitude=*/test::kMaxS16);
+  const float noise_level_dbfs =
+      RunEstimator(gen, *estimator, sample_rate_hz());
+  EXPECT_NEAR(noise_level_dbfs, -5.5f, 0.5f);
+}
+
+// Checks that a full scale sine wave maps to about -3 dBFS.
+TEST_P(NoiseEstimatorParametrization, NoiseFloorEstimatorWithSineTone) {
+  ApmDataDumper data_dumper(0);
+  auto estimator = CreateNoiseFloorEstimator(&data_dumper);
+
+  test::SineGenerator gen(/*amplitude=*/test::kMaxS16, /*frequency_hz=*/600.0f,
+                          sample_rate_hz());
+  const float noise_level_dbfs =
+      RunEstimator(gen, *estimator, sample_rate_hz());
+  EXPECT_NEAR(noise_level_dbfs, -3.0f, 0.1f);
+}
+
+// Check that sufficiently spaced periodic pulses do not raise the estimated
+// noise floor, which is determined by the amplitude of the non-pulse samples.
+TEST_P(NoiseEstimatorParametrization, NoiseFloorEstimatorWithPulseTone) {
+  ApmDataDumper data_dumper(0);
+  auto estimator = CreateNoiseFloorEstimator(&data_dumper);
+
+  constexpr float kNoPulseAmplitude = 10.0f;
+  test::PulseGenerator gen(/*pulse_amplitude=*/test::kMaxS16, kNoPulseAmplitude,
+                           /*frequency_hz=*/20.0f, sample_rate_hz());
+  const float noise_level_dbfs =
+      RunEstimator(gen, *estimator, sample_rate_hz());
+  const float expected_noise_floor_dbfs =
+      20.0f * std::log10f(kNoPulseAmplitude / test::kMaxS16);
+  EXPECT_NEAR(noise_level_dbfs, expected_noise_floor_dbfs, 0.5f);
+}
+
+INSTANTIATE_TEST_SUITE_P(GainController2NoiseEstimator,
+                         NoiseEstimatorParametrization,
+                         ::testing::Values(8000, 16000, 32000, 48000));
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn
new file mode 100644
index 0000000000..d709eb3699
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -0,0 +1,334 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../../webrtc.gni")
+
+rtc_library("rnn_vad") {
+  visibility = [ "../*" ]
+  sources = [
+    "features_extraction.cc",
+    "features_extraction.h",
+    "rnn.cc",
+    "rnn.h",
+  ]
+
+  defines = []
+  if (rtc_build_with_neon && target_cpu != "arm64") {
+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+    cflags = [ "-mfpu=neon" ]
+  }
+
+  deps = [
+    ":rnn_vad_common",
+    ":rnn_vad_layers",
+    ":rnn_vad_lp_residual",
+    ":rnn_vad_pitch",
+    ":rnn_vad_sequence_buffer",
+    ":rnn_vad_spectral_features",
+    "..:biquad_filter",
+    "..:cpu_features",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_compare",
+    "../../../../rtc_base:safe_conversions",
+    "//third_party/rnnoise:rnn_vad",
+  ]
+}
+
+rtc_library("rnn_vad_auto_correlation") {
+  sources = [
+    "auto_correlation.cc",
+    "auto_correlation.h",
+  ]
+  deps = [
+    ":rnn_vad_common",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../utility:pffft_wrapper",
+  ]
+}
+
+rtc_source_set("rnn_vad_common") {
+  # TODO(alessiob): Make this target visibility private.
+  visibility = [
+    ":*",
+    "..:vad_wrapper",
+  ]
+  sources = [ "common.h" ]
+  deps = [
+    "../../../../rtc_base/system:arch",
+    "../../../../system_wrappers",
+  ]
+}
+
+rtc_library("rnn_vad_lp_residual") {
+  sources = [
+    "lp_residual.cc",
+    "lp_residual.h",
+  ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_compare",
+  ]
+}
+
+rtc_source_set("rnn_vad_layers") {
+  sources = [
+    "rnn_fc.cc",
+    "rnn_fc.h",
+    "rnn_gru.cc",
+    "rnn_gru.h",
+  ]
+
+  defines = []
+  if (rtc_build_with_neon && current_cpu != "arm64") {
+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+    cflags = [ "-mfpu=neon" ]
+  }
+
+  deps = [
+    ":rnn_vad_common",
+    ":vector_math",
+    "..:cpu_features",
+    "../../../../api:array_view",
+    "../../../../api:function_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_conversions",
+    "//third_party/rnnoise:rnn_vad",
+  ]
+  if (current_cpu == "x86" || current_cpu == "x64") {
+    deps += [ ":vector_math_avx2" ]
+  }
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_source_set("vector_math") {
+  sources = [ "vector_math.h" ]
+  deps = [
+    "..:cpu_features",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_conversions",
+    "../../../../rtc_base/system:arch",
+  ]
+}
+
+if (current_cpu == "x86" || current_cpu == "x64") {
+  rtc_library("vector_math_avx2") {
+    sources = [ "vector_math_avx2.cc" ]
+    if (is_win && !build_with_mozilla) {
+      cflags = [ "/arch:AVX2" ]
+    } else {
+      cflags = [
+        "-mavx2",
+        "-mfma",
+      ]
+    }
+    deps = [
+      ":vector_math",
+      "../../../../api:array_view",
+      "../../../../rtc_base:checks",
+      "../../../../rtc_base:safe_conversions",
+    ]
+  }
+}
+
+rtc_library("rnn_vad_pitch") {
+  sources = [
+    "pitch_search.cc",
+    "pitch_search.h",
+    "pitch_search_internal.cc",
+    "pitch_search_internal.h",
+  ]
+
+  defines = []
+  if (rtc_build_with_neon && current_cpu != "arm64") {
+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+    cflags = [ "-mfpu=neon" ]
+  }
+
+  deps = [
+    ":rnn_vad_auto_correlation",
+    ":rnn_vad_common",
+    ":vector_math",
+    "..:cpu_features",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:gtest_prod",
+    "../../../../rtc_base:safe_compare",
+    "../../../../rtc_base:safe_conversions",
+    "../../../../rtc_base/system:arch",
+  ]
+  if (current_cpu == "x86" || current_cpu == "x64") {
+    deps += [ ":vector_math_avx2" ]
+  }
+}
+
+rtc_source_set("rnn_vad_ring_buffer") {
+  sources = [ "ring_buffer.h" ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+  ]
+}
+
+rtc_source_set("rnn_vad_sequence_buffer") {
+  sources = [ "sequence_buffer.h" ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+  ]
+}
+
+rtc_library("rnn_vad_spectral_features") {
+  sources = [
+    "spectral_features.cc",
+    "spectral_features.h",
+    "spectral_features_internal.cc",
+    "spectral_features_internal.h",
+  ]
+  deps = [
+    ":rnn_vad_common",
+    ":rnn_vad_ring_buffer",
+    ":rnn_vad_symmetric_matrix_buffer",
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_compare",
+    "../../utility:pffft_wrapper",
+  ]
+}
+
+rtc_source_set("rnn_vad_symmetric_matrix_buffer") {
+  sources = [ "symmetric_matrix_buffer.h" ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:safe_compare",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("test_utils") {
+    testonly = true
+    sources = [
+      "test_utils.cc",
+      "test_utils.h",
+    ]
+    deps = [
+      ":rnn_vad",
+      ":rnn_vad_common",
+      "../../../../api:array_view",
+      "../../../../api:scoped_refptr",
+      "../../../../rtc_base:checks",
+      "../../../../rtc_base:safe_compare",
+      "../../../../test:fileutils",
+      "../../../../test:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  unittest_resources = [
+    "../../../../resources/audio_processing/agc2/rnn_vad/band_energies.dat",
+    "../../../../resources/audio_processing/agc2/rnn_vad/pitch_buf_24k.dat",
+    "../../../../resources/audio_processing/agc2/rnn_vad/pitch_lp_res.dat",
+    "../../../../resources/audio_processing/agc2/rnn_vad/pitch_search_int.dat",
+    "../../../../resources/audio_processing/agc2/rnn_vad/samples.pcm",
+    "../../../../resources/audio_processing/agc2/rnn_vad/vad_prob.dat",
+  ]
+
+  if (is_ios) {
+    bundle_data("unittests_bundle_data") {
+      testonly = true
+      sources = unittest_resources
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+    }
+  }
+
+  rtc_library("unittests") {
+    testonly = true
+    sources = [
+      "auto_correlation_unittest.cc",
+      "features_extraction_unittest.cc",
+      "lp_residual_unittest.cc",
+      "pitch_search_internal_unittest.cc",
+      "pitch_search_unittest.cc",
+      "ring_buffer_unittest.cc",
+      "rnn_fc_unittest.cc",
+      "rnn_gru_unittest.cc",
+      "rnn_unittest.cc",
+      "rnn_vad_unittest.cc",
+      "sequence_buffer_unittest.cc",
+      "spectral_features_internal_unittest.cc",
+      "spectral_features_unittest.cc",
+      "symmetric_matrix_buffer_unittest.cc",
+      "vector_math_unittest.cc",
+    ]
+
+    defines = []
+    if (rtc_build_with_neon && current_cpu != "arm64") {
+      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+      cflags = [ "-mfpu=neon" ]
+    }
+
+    deps = [
+      ":rnn_vad",
+      ":rnn_vad_auto_correlation",
+      ":rnn_vad_common",
+      ":rnn_vad_layers",
+      ":rnn_vad_lp_residual",
+      ":rnn_vad_pitch",
+      ":rnn_vad_ring_buffer",
+      ":rnn_vad_sequence_buffer",
+      ":rnn_vad_spectral_features",
+      ":rnn_vad_symmetric_matrix_buffer",
+      ":test_utils",
+      ":vector_math",
+      "..:cpu_features",
+      "../..:audioproc_test_utils",
+      "../../../../api:array_view",
+      "../../../../common_audio/",
+      "../../../../rtc_base:checks",
+      "../../../../rtc_base:logging",
+      "../../../../rtc_base:safe_compare",
+      "../../../../rtc_base:safe_conversions",
+      "../../../../rtc_base:stringutils",
+      "../../../../rtc_base/system:arch",
+      "../../../../test:test_support",
+      "../../utility:pffft_wrapper",
+      "//third_party/rnnoise:rnn_vad",
+    ]
+    if (current_cpu == "x86" || current_cpu == "x64") {
+      deps += [ ":vector_math_avx2" ]
+    }
+    absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+    data = unittest_resources
+    if (is_ios) {
+      deps += [ ":unittests_bundle_data" ]
+    }
+  }
+
+  if (!build_with_chromium) {
+    rtc_executable("rnn_vad_tool") {
+      testonly = true
+      sources = [ "rnn_vad_tool.cc" ]
+      deps = [
+        ":rnn_vad",
+        ":rnn_vad_common",
+        "..:cpu_features",
+        "../../../../api:array_view",
+        "../../../../common_audio",
+        "../../../../rtc_base:logging",
+        "../../../../rtc_base:safe_compare",
+        "../../../../test:test_support",
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/flags:parse",
+      ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/DEPS b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/DEPS
new file mode 100644
index 0000000000..773c2d7edd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/DEPS
@@ -0,0 +1,3 @@
+include_rules = [
+  "+third_party/rnnoise",
+]
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
new file mode 100644
index 0000000000..3ddeec8dba
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/auto_correlation.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kAutoCorrelationFftOrder = 9;  // Length-512 FFT.
+static_assert(1 << kAutoCorrelationFftOrder >
+                  kNumLags12kHz + kBufSize12kHz - kMaxPitch12kHz,
+              "");
+
+}  // namespace
+
+AutoCorrelationCalculator::AutoCorrelationCalculator()
+    : fft_(1 << kAutoCorrelationFftOrder, Pffft::FftType::kReal),
+      tmp_(fft_.CreateBuffer()),
+      X_(fft_.CreateBuffer()),
+      H_(fft_.CreateBuffer()) {}
+
+AutoCorrelationCalculator::~AutoCorrelationCalculator() = default;
+
+// The auto-correlations coefficients are computed as follows:
+// |.........|...........|  <- pitch buffer
+//           [ x (fixed) ]
+// [   y_0   ]
+//         [ y_{m-1} ]
+// x and y are sub-array of equal length; x is never moved, whereas y slides.
+// The cross-correlation between y_0 and x corresponds to the auto-correlation
+// for the maximum pitch period. Hence, the first value in `auto_corr` has an
+// inverted lag equal to 0 that corresponds to a lag equal to the maximum
+// pitch period.
+void AutoCorrelationCalculator::ComputeOnPitchBuffer(
+    rtc::ArrayView<const float, kBufSize12kHz> pitch_buf,
+    rtc::ArrayView<float, kNumLags12kHz> auto_corr) {
+  RTC_DCHECK_LT(auto_corr.size(), kMaxPitch12kHz);
+  RTC_DCHECK_GT(pitch_buf.size(), kMaxPitch12kHz);
+  constexpr int kFftFrameSize = 1 << kAutoCorrelationFftOrder;
+  constexpr int kConvolutionLength = kBufSize12kHz - kMaxPitch12kHz;
+  static_assert(kConvolutionLength == kFrameSize20ms12kHz,
+                "Mismatch between pitch buffer size, frame size and maximum "
+                "pitch period.");
+  static_assert(kFftFrameSize > kNumLags12kHz + kConvolutionLength,
+                "The FFT length is not sufficiently big to avoid cyclic "
+                "convolution errors.");
+  auto tmp = tmp_->GetView();
+
+  // Compute the FFT for the reversed reference frame - i.e.,
+  // pitch_buf[-kConvolutionLength:].
+  std::reverse_copy(pitch_buf.end() - kConvolutionLength, pitch_buf.end(),
+                    tmp.begin());
+  std::fill(tmp.begin() + kConvolutionLength, tmp.end(), 0.f);
+  fft_.ForwardTransform(*tmp_, H_.get(), /*ordered=*/false);
+
+  // Compute the FFT for the sliding frames chunk. The sliding frames are
+  // defined as pitch_buf[i:i+kConvolutionLength] where i in
+  // [0, kNumLags12kHz). The chunk includes all of them, hence it is
+  // defined as pitch_buf[:kNumLags12kHz+kConvolutionLength].
+  std::copy(pitch_buf.begin(),
+            pitch_buf.begin() + kConvolutionLength + kNumLags12kHz,
+            tmp.begin());
+  std::fill(tmp.begin() + kNumLags12kHz + kConvolutionLength, tmp.end(), 0.f);
+  fft_.ForwardTransform(*tmp_, X_.get(), /*ordered=*/false);
+
+  // Convolve in the frequency domain.
+  constexpr float kScalingFactor = 1.f / static_cast<float>(kFftFrameSize);
+  std::fill(tmp.begin(), tmp.end(), 0.f);
+  fft_.FrequencyDomainConvolve(*X_, *H_, tmp_.get(), kScalingFactor);
+  fft_.BackwardTransform(*tmp_, tmp_.get(), /*ordered=*/false);
+
+  // Extract the auto-correlation coefficients.
+  std::copy(tmp.begin() + kConvolutionLength - 1,
+            tmp.begin() + kConvolutionLength + kNumLags12kHz - 1,
+            auto_corr.begin());
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
new file mode 100644
index 0000000000..1ae5054567
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_AUTO_CORRELATION_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_AUTO_CORRELATION_H_
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/utility/pffft_wrapper.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Class to compute the auto correlation on the pitch buffer for a target pitch
+// interval.
+class AutoCorrelationCalculator {
+ public:
+  AutoCorrelationCalculator();
+  AutoCorrelationCalculator(const AutoCorrelationCalculator&) = delete;
+  AutoCorrelationCalculator& operator=(const AutoCorrelationCalculator&) =
+      delete;
+  ~AutoCorrelationCalculator();
+
+  // Computes the auto-correlation coefficients for a target pitch interval.
+  // `auto_corr` indexes are inverted lags.
+  void ComputeOnPitchBuffer(
+      rtc::ArrayView<const float, kBufSize12kHz> pitch_buf,
+      rtc::ArrayView<float, kNumLags12kHz> auto_corr);
+
+ private:
+  Pffft fft_;
+  std::unique_ptr<Pffft::FloatBuffer> tmp_;
+  std::unique_ptr<Pffft::FloatBuffer> X_;
+  std::unique_ptr<Pffft::FloatBuffer> H_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_AUTO_CORRELATION_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation_unittest.cc
new file mode 100644
index 0000000000..76001ed7b7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation_unittest.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/auto_correlation.h"
+
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Checks that the auto correlation function produces output within tolerance
+// given test input data.
+TEST(RnnVadTest, PitchBufferAutoCorrelationWithinTolerance) {
+  PitchTestData test_data;
+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
+  Decimate2x(test_data.PitchBuffer24kHzView(), pitch_buf_decimated);
+  std::array<float, kNumLags12kHz> computed_output;
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    AutoCorrelationCalculator auto_corr_calculator;
+    auto_corr_calculator.ComputeOnPitchBuffer(pitch_buf_decimated,
+                                              computed_output);
+  }
+  auto auto_corr_view = test_data.AutoCorrelation12kHzView();
+  ExpectNearAbsolute({auto_corr_view.data(), auto_corr_view.size()},
+                     computed_output, 3e-3f);
+}
+
+// Checks that the auto correlation function computes the right thing for a
+// simple use case.
+TEST(RnnVadTest, CheckAutoCorrelationOnConstantPitchBuffer) {
+  // Create constant signal with no pitch.
+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
+  std::fill(pitch_buf_decimated.begin(), pitch_buf_decimated.end(), 1.f);
+  std::array<float, kNumLags12kHz> computed_output;
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    AutoCorrelationCalculator auto_corr_calculator;
+    auto_corr_calculator.ComputeOnPitchBuffer(pitch_buf_decimated,
+                                              computed_output);
+  }
+  // The expected output is a vector filled with the same expected
+  // auto-correlation value. The latter equals the length of a 20 ms frame.
+  constexpr int kFrameSize20ms12kHz = kFrameSize20ms24kHz / 2;
+  std::array<float, kNumLags12kHz> expected_output;
+  std::fill(expected_output.begin(), expected_output.end(),
+            static_cast<float>(kFrameSize20ms12kHz));
+  ExpectNearAbsolute(expected_output, computed_output, 4e-5f);
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/common.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/common.h
new file mode 100644
index 0000000000..c099373200
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/common.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+namespace rnn_vad {
+
+constexpr double kPi = 3.14159265358979323846;
+
+constexpr int kSampleRate24kHz = 24000;
+constexpr int kFrameSize10ms24kHz = kSampleRate24kHz / 100;
+constexpr int kFrameSize20ms24kHz = kFrameSize10ms24kHz * 2;
+
+// Pitch buffer.
+constexpr int kMinPitch24kHz = kSampleRate24kHz / 800;   // 0.00125 s.
+constexpr int kMaxPitch24kHz = kSampleRate24kHz / 62.5;  // 0.016 s.
+constexpr int kBufSize24kHz = kMaxPitch24kHz + kFrameSize20ms24kHz;
+static_assert((kBufSize24kHz & 1) == 0, "The buffer size must be even.");
+
+// 24 kHz analysis.
+// Define a higher minimum pitch period for the initial search. This is used to
+// avoid searching for very short periods, for which a refinement step is
+// responsible.
+constexpr int kInitialMinPitch24kHz = 3 * kMinPitch24kHz;
+static_assert(kMinPitch24kHz < kInitialMinPitch24kHz, "");
+static_assert(kInitialMinPitch24kHz < kMaxPitch24kHz, "");
+static_assert(kMaxPitch24kHz > kInitialMinPitch24kHz, "");
+// Number of (inverted) lags during the initial pitch search phase at 24 kHz.
+constexpr int kInitialNumLags24kHz = kMaxPitch24kHz - kInitialMinPitch24kHz;
+// Number of (inverted) lags during the pitch search refinement phase at 24 kHz.
+constexpr int kRefineNumLags24kHz = kMaxPitch24kHz + 1;
+static_assert(
+    kRefineNumLags24kHz > kInitialNumLags24kHz,
+    "The refinement step must search the pitch in an extended pitch range.");
+
+// 12 kHz analysis.
+constexpr int kSampleRate12kHz = 12000;
+constexpr int kFrameSize10ms12kHz = kSampleRate12kHz / 100;
+constexpr int kFrameSize20ms12kHz = kFrameSize10ms12kHz * 2;
+constexpr int kBufSize12kHz = kBufSize24kHz / 2;
+constexpr int kInitialMinPitch12kHz = kInitialMinPitch24kHz / 2;
+constexpr int kMaxPitch12kHz = kMaxPitch24kHz / 2;
+static_assert(kMaxPitch12kHz > kInitialMinPitch12kHz, "");
+// The inverted lags for the pitch interval [`kInitialMinPitch12kHz`,
+// `kMaxPitch12kHz`] are in the range [0, `kNumLags12kHz`].
+constexpr int kNumLags12kHz = kMaxPitch12kHz - kInitialMinPitch12kHz;
+
+// 48 kHz constants.
+constexpr int kMinPitch48kHz = kMinPitch24kHz * 2;
+constexpr int kMaxPitch48kHz = kMaxPitch24kHz * 2;
+
+// Spectral features.
+constexpr int kNumBands = 22;
+constexpr int kNumLowerBands = 6;
+static_assert((0 < kNumLowerBands) && (kNumLowerBands < kNumBands), "");
+constexpr int kCepstralCoeffsHistorySize = 8;
+static_assert(kCepstralCoeffsHistorySize > 2,
+              "The history size must at least be 3 to compute first and second "
+              "derivatives.");
+
+constexpr int kFeatureVectorSize = 42;
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
new file mode 100644
index 0000000000..502023428d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
+
+#include <array>
+
+#include "modules/audio_processing/agc2/rnn_vad/lp_residual.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Computed as `scipy.signal.butter(N=2, Wn=60/24000, btype='highpass')`.
+constexpr BiQuadFilter::Config kHpfConfig24k{
+    {0.99446179f, -1.98892358f, 0.99446179f},
+    {-1.98889291f, 0.98895425f}};
+
+}  // namespace
+
+FeaturesExtractor::FeaturesExtractor(const AvailableCpuFeatures& cpu_features)
+    : use_high_pass_filter_(false),
+      hpf_(kHpfConfig24k),
+      pitch_buf_24kHz_(),
+      pitch_buf_24kHz_view_(pitch_buf_24kHz_.GetBufferView()),
+      lp_residual_(kBufSize24kHz),
+      lp_residual_view_(lp_residual_.data(), kBufSize24kHz),
+      pitch_estimator_(cpu_features),
+      reference_frame_view_(pitch_buf_24kHz_.GetMostRecentValuesView()) {
+  RTC_DCHECK_EQ(kBufSize24kHz, lp_residual_.size());
+  Reset();
+}
+
+FeaturesExtractor::~FeaturesExtractor() = default;
+
+void FeaturesExtractor::Reset() {
+  pitch_buf_24kHz_.Reset();
+  spectral_features_extractor_.Reset();
+  if (use_high_pass_filter_) {
+    hpf_.Reset();
+  }
+}
+
+bool FeaturesExtractor::CheckSilenceComputeFeatures(
+    rtc::ArrayView<const float, kFrameSize10ms24kHz> samples,
+    rtc::ArrayView<float, kFeatureVectorSize> feature_vector) {
+  // Pre-processing.
+  if (use_high_pass_filter_) {
+    std::array<float, kFrameSize10ms24kHz> samples_filtered;
+    hpf_.Process(samples, samples_filtered);
+    // Feed buffer with the pre-processed version of `samples`.
+    pitch_buf_24kHz_.Push(samples_filtered);
+  } else {
+    // Feed buffer with `samples`.
+    pitch_buf_24kHz_.Push(samples);
+  }
+  // Extract the LP residual.
+  float lpc_coeffs[kNumLpcCoefficients];
+  ComputeAndPostProcessLpcCoefficients(pitch_buf_24kHz_view_, lpc_coeffs);
+  ComputeLpResidual(lpc_coeffs, pitch_buf_24kHz_view_, lp_residual_view_);
+  // Estimate pitch on the LP-residual and write the normalized pitch period
+  // into the output vector (normalization based on training data stats).
+  pitch_period_48kHz_ = pitch_estimator_.Estimate(lp_residual_view_);
+  feature_vector[kFeatureVectorSize - 2] = 0.01f * (pitch_period_48kHz_ - 300);
+  // Extract lagged frames (according to the estimated pitch period).
+  RTC_DCHECK_LE(pitch_period_48kHz_ / 2, kMaxPitch24kHz);
+  auto lagged_frame = pitch_buf_24kHz_view_.subview(
+      kMaxPitch24kHz - pitch_period_48kHz_ / 2, kFrameSize20ms24kHz);
+  // Analyze reference and lagged frames checking if silence has been detected
+  // and write the feature vector.
+  return spectral_features_extractor_.CheckSilenceComputeFeatures(
+      reference_frame_view_, {lagged_frame.data(), kFrameSize20ms24kHz},
+      {feature_vector.data() + kNumLowerBands, kNumBands - kNumLowerBands},
+      {feature_vector.data(), kNumLowerBands},
+      {feature_vector.data() + kNumBands, kNumLowerBands},
+      {feature_vector.data() + kNumBands + kNumLowerBands, kNumLowerBands},
+      {feature_vector.data() + kNumBands + 2 * kNumLowerBands, kNumLowerBands},
+      &feature_vector[kFeatureVectorSize - 1]);
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h
new file mode 100644
index 0000000000..d47a85bfb0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/biquad_filter.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
+#include "modules/audio_processing/agc2/rnn_vad/sequence_buffer.h"
+#include "modules/audio_processing/agc2/rnn_vad/spectral_features.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Feature extractor to feed the VAD RNN.
+class FeaturesExtractor {
+ public:
+  explicit FeaturesExtractor(const AvailableCpuFeatures& cpu_features);
+  FeaturesExtractor(const FeaturesExtractor&) = delete;
+  FeaturesExtractor& operator=(const FeaturesExtractor&) = delete;
+  ~FeaturesExtractor();
+  void Reset();
+  // Analyzes the samples, computes the feature vector and returns true if
+  // silence is detected (false if not). When silence is detected,
+  // `feature_vector` is partially written and therefore must not be used to
+  // feed the VAD RNN.
+  bool CheckSilenceComputeFeatures(
+      rtc::ArrayView<const float, kFrameSize10ms24kHz> samples,
+      rtc::ArrayView<float, kFeatureVectorSize> feature_vector);
+
+ private:
+  const bool use_high_pass_filter_;
+  // TODO(bugs.webrtc.org/7494): Remove HPF depending on how AGC2 is used in APM
+  // and on whether an HPF is already used as pre-processing step in APM.
+  BiQuadFilter hpf_;
+  SequenceBuffer<float, kBufSize24kHz, kFrameSize10ms24kHz, kFrameSize20ms24kHz>
+      pitch_buf_24kHz_;
+  rtc::ArrayView<const float, kBufSize24kHz> pitch_buf_24kHz_view_;
+  std::vector<float> lp_residual_;
+  rtc::ArrayView<float, kBufSize24kHz> lp_residual_view_;
+  PitchEstimator pitch_estimator_;
+  rtc::ArrayView<const float, kFrameSize20ms24kHz> reference_frame_view_;
+  SpectralFeaturesExtractor spectral_features_extractor_;
+  int pitch_period_48kHz_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
new file mode 100644
index 0000000000..96f956adfe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
+
+#include <cmath>
+#include <vector>
+
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "rtc_base/numerics/safe_conversions.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int ceil(int n, int m) {
+  return (n + m - 1) / m;
+}
+
+// Number of 10 ms frames required to fill a pitch buffer having size
+// `kBufSize24kHz`.
+constexpr int kNumTestDataFrames = ceil(kBufSize24kHz, kFrameSize10ms24kHz);
+// Number of samples for the test data.
+constexpr int kNumTestDataSize = kNumTestDataFrames * kFrameSize10ms24kHz;
+
+// Verifies that the pitch in Hz is in the detectable range.
+bool PitchIsValid(float pitch_hz) {
+  const int pitch_period = static_cast<float>(kSampleRate24kHz) / pitch_hz;
+  return kInitialMinPitch24kHz <= pitch_period &&
+         pitch_period <= kMaxPitch24kHz;
+}
+
+void CreatePureTone(float amplitude, float freq_hz, rtc::ArrayView<float> dst) {
+  for (int i = 0; rtc::SafeLt(i, dst.size()); ++i) {
+    dst[i] = amplitude * std::sin(2.f * kPi * freq_hz * i / kSampleRate24kHz);
+  }
+}
+
+// Feeds `features_extractor` with `samples` splitting it in 10 ms frames.
+// For every frame, the output is written into `feature_vector`. Returns true
+// if silence is detected in the last frame.
+bool FeedTestData(FeaturesExtractor& features_extractor,
+                  rtc::ArrayView<const float> samples,
+                  rtc::ArrayView<float, kFeatureVectorSize> feature_vector) {
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  bool is_silence = true;
+  const int num_frames = samples.size() / kFrameSize10ms24kHz;
+  for (int i = 0; i < num_frames; ++i) {
+    is_silence = features_extractor.CheckSilenceComputeFeatures(
+        {samples.data() + i * kFrameSize10ms24kHz, kFrameSize10ms24kHz},
+        feature_vector);
+  }
+  return is_silence;
+}
+
+// Extracts the features for two pure tones and verifies that the pitch field
+// values reflect the known tone frequencies.
+TEST(RnnVadTest, FeatureExtractionLowHighPitch) {
+  constexpr float amplitude = 1000.f;
+  constexpr float low_pitch_hz = 150.f;
+  constexpr float high_pitch_hz = 250.f;
+  ASSERT_TRUE(PitchIsValid(low_pitch_hz));
+  ASSERT_TRUE(PitchIsValid(high_pitch_hz));
+
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+  FeaturesExtractor features_extractor(cpu_features);
+  std::vector<float> samples(kNumTestDataSize);
+  std::vector<float> feature_vector(kFeatureVectorSize);
+  ASSERT_EQ(kFeatureVectorSize, rtc::dchecked_cast<int>(feature_vector.size()));
+  rtc::ArrayView<float, kFeatureVectorSize> feature_vector_view(
+      feature_vector.data(), kFeatureVectorSize);
+
+  // Extract the normalized scalar feature that is proportional to the estimated
+  // pitch period.
+  constexpr int pitch_feature_index = kFeatureVectorSize - 2;
+  // Low frequency tone - i.e., high period.
+  CreatePureTone(amplitude, low_pitch_hz, samples);
+  ASSERT_FALSE(FeedTestData(features_extractor, samples, feature_vector_view));
+  float high_pitch_period = feature_vector_view[pitch_feature_index];
+  // High frequency tone - i.e., low period.
+  features_extractor.Reset();
+  CreatePureTone(amplitude, high_pitch_hz, samples);
+  ASSERT_FALSE(FeedTestData(features_extractor, samples, feature_vector_view));
+  float low_pitch_period = feature_vector_view[pitch_feature_index];
+  // Check.
+  EXPECT_LT(low_pitch_period, high_pitch_period);
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
new file mode 100644
index 0000000000..484bfba459
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/lp_residual.h"
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <numeric>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Computes auto-correlation coefficients for `x` and writes them in
+// `auto_corr`. The lag values are in {0, ..., max_lag - 1}, where max_lag
+// equals the size of `auto_corr`.
+void ComputeAutoCorrelation(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float, kNumLpcCoefficients> auto_corr) {
+  constexpr int max_lag = auto_corr.size();
+  RTC_DCHECK_LT(max_lag, x.size());
+  for (int lag = 0; lag < max_lag; ++lag) {
+    auto_corr[lag] =
+        std::inner_product(x.begin(), x.end() - lag, x.begin() + lag, 0.f);
+  }
+}
+
+// Applies denoising to the auto-correlation coefficients.
+void DenoiseAutoCorrelation(
+    rtc::ArrayView<float, kNumLpcCoefficients> auto_corr) {
+  // Assume -40 dB white noise floor.
+  auto_corr[0] *= 1.0001f;
+  // Hard-coded values obtained as
+  // [np.float32((0.008*0.008*i*i)) for i in range(1,5)].
+  auto_corr[1] -= auto_corr[1] * 0.000064f;
+  auto_corr[2] -= auto_corr[2] * 0.000256f;
+  auto_corr[3] -= auto_corr[3] * 0.000576f;
+  auto_corr[4] -= auto_corr[4] * 0.001024f;
+  static_assert(kNumLpcCoefficients == 5, "Update `auto_corr`.");
+}
+
+// Computes the initial inverse filter coefficients given the auto-correlation
+// coefficients of an input frame.
+void ComputeInitialInverseFilterCoefficients(
+    rtc::ArrayView<const float, kNumLpcCoefficients> auto_corr,
+    rtc::ArrayView<float, kNumLpcCoefficients - 1> lpc_coeffs) {
+  float error = auto_corr[0];
+  for (int i = 0; i < kNumLpcCoefficients - 1; ++i) {
+    float reflection_coeff = 0.f;
+    for (int j = 0; j < i; ++j) {
+      reflection_coeff += lpc_coeffs[j] * auto_corr[i - j];
+    }
+    reflection_coeff += auto_corr[i + 1];
+
+    // Avoid division by numbers close to zero.
+    constexpr float kMinErrorMagnitude = 1e-6f;
+    if (std::fabs(error) < kMinErrorMagnitude) {
+      error = std::copysign(kMinErrorMagnitude, error);
+    }
+
+    reflection_coeff /= -error;
+    // Update LPC coefficients and total error.
+    lpc_coeffs[i] = reflection_coeff;
+    for (int j = 0; j < ((i + 1) >> 1); ++j) {
+      const float tmp1 = lpc_coeffs[j];
+      const float tmp2 = lpc_coeffs[i - 1 - j];
+      lpc_coeffs[j] = tmp1 + reflection_coeff * tmp2;
+      lpc_coeffs[i - 1 - j] = tmp2 + reflection_coeff * tmp1;
+    }
+    error -= reflection_coeff * reflection_coeff * error;
+    if (error < 0.001f * auto_corr[0]) {
+      break;
+    }
+  }
+}
+
+}  // namespace
+
+void ComputeAndPostProcessLpcCoefficients(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float, kNumLpcCoefficients> lpc_coeffs) {
+  std::array<float, kNumLpcCoefficients> auto_corr;
+  ComputeAutoCorrelation(x, auto_corr);
+  if (auto_corr[0] == 0.f) {  // Empty frame.
+    std::fill(lpc_coeffs.begin(), lpc_coeffs.end(), 0);
+    return;
+  }
+  DenoiseAutoCorrelation(auto_corr);
+  std::array<float, kNumLpcCoefficients - 1> lpc_coeffs_pre{};
+  ComputeInitialInverseFilterCoefficients(auto_corr, lpc_coeffs_pre);
+  // LPC coefficients post-processing.
+  // TODO(bugs.webrtc.org/9076): Consider removing these steps.
+  lpc_coeffs_pre[0] *= 0.9f;
+  lpc_coeffs_pre[1] *= 0.9f * 0.9f;
+  lpc_coeffs_pre[2] *= 0.9f * 0.9f * 0.9f;
+  lpc_coeffs_pre[3] *= 0.9f * 0.9f * 0.9f * 0.9f;
+  constexpr float kC = 0.8f;
+  lpc_coeffs[0] = lpc_coeffs_pre[0] + kC;
+  lpc_coeffs[1] = lpc_coeffs_pre[1] + kC * lpc_coeffs_pre[0];
+  lpc_coeffs[2] = lpc_coeffs_pre[2] + kC * lpc_coeffs_pre[1];
+  lpc_coeffs[3] = lpc_coeffs_pre[3] + kC * lpc_coeffs_pre[2];
+  lpc_coeffs[4] = kC * lpc_coeffs_pre[3];
+  static_assert(kNumLpcCoefficients == 5, "Update `lpc_coeffs(_pre)`.");
+}
+
+void ComputeLpResidual(
+    rtc::ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float> y) {
+  RTC_DCHECK_GT(x.size(), kNumLpcCoefficients);
+  RTC_DCHECK_EQ(x.size(), y.size());
+  // The code below implements the following operation:
+  // y[i] = x[i] + dot_product({x[i], ..., x[i - kNumLpcCoefficients + 1]},
+  //                           lpc_coeffs)
+  // Edge case: i < kNumLpcCoefficients.
+  y[0] = x[0];
+  for (int i = 1; i < kNumLpcCoefficients; ++i) {
+    y[i] =
+        std::inner_product(x.crend() - i, x.crend(), lpc_coeffs.cbegin(), x[i]);
+  }
+  // Regular case.
+  auto last = x.crend();
+  for (int i = kNumLpcCoefficients; rtc::SafeLt(i, y.size()); ++i, --last) {
+    y[i] = std::inner_product(last - kNumLpcCoefficients, last,
+                              lpc_coeffs.cbegin(), x[i]);
+  }
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h
new file mode 100644
index 0000000000..d04c536ec1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
+
+#include <stddef.h>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Linear predictive coding (LPC) inverse filter length.
+constexpr int kNumLpcCoefficients = 5;
+
+// Given a frame `x`, computes a post-processed version of LPC coefficients
+// tailored for pitch estimation.
+void ComputeAndPostProcessLpcCoefficients(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float, kNumLpcCoefficients> lpc_coeffs);
+
+// Computes the LP residual for the input frame `x` and the LPC coefficients
+// `lpc_coeffs`. `y` and `x` can point to the same array for in-place
+// computation.
+void ComputeLpResidual(
+    rtc::ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float> y);
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual_unittest.cc
new file mode 100644
index 0000000000..7b3a4a3f65
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual_unittest.cc
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/lp_residual.h"
+
+#include <algorithm>
+#include <array>
+#include <vector>
+
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Checks that the LP residual can be computed on an empty frame.
+TEST(RnnVadTest, LpResidualOfEmptyFrame) {
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+
+  // Input frame (empty, i.e., all samples set to 0).
+  std::array<float, kFrameSize10ms24kHz> empty_frame;
+  empty_frame.fill(0.f);
+  // Compute inverse filter coefficients.
+  std::array<float, kNumLpcCoefficients> lpc;
+  ComputeAndPostProcessLpcCoefficients(empty_frame, lpc);
+  // Compute LP residual.
+  std::array<float, kFrameSize10ms24kHz> lp_residual;
+  ComputeLpResidual(lpc, empty_frame, lp_residual);
+}
+
+// Checks that the computed LP residual is bit-exact given test input data.
+TEST(RnnVadTest, LpResidualPipelineBitExactness) {
+  // Input and expected output readers.
+  ChunksFileReader pitch_buffer_reader = CreatePitchBuffer24kHzReader();
+  ChunksFileReader lp_pitch_reader = CreateLpResidualAndPitchInfoReader();
+
+  // Buffers.
+  std::vector<float> pitch_buffer_24kHz(kBufSize24kHz);
+  std::array<float, kNumLpcCoefficients> lpc;
+  std::vector<float> computed_lp_residual(kBufSize24kHz);
+  std::vector<float> expected_lp_residual(kBufSize24kHz);
+
+  // Test length.
+  const int num_frames =
+      std::min(pitch_buffer_reader.num_chunks, 300);  // Max 3 s.
+  ASSERT_GE(lp_pitch_reader.num_chunks, num_frames);
+
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  for (int i = 0; i < num_frames; ++i) {
+    SCOPED_TRACE(i);
+    // Read input.
+    ASSERT_TRUE(pitch_buffer_reader.reader->ReadChunk(pitch_buffer_24kHz));
+    // Read expected output (ignore pitch gain and period).
+    ASSERT_TRUE(lp_pitch_reader.reader->ReadChunk(expected_lp_residual));
+    lp_pitch_reader.reader->SeekForward(2);  // Pitch period and strength.
+    // Check every 200 ms.
+    if (i % 20 == 0) {
+      ComputeAndPostProcessLpcCoefficients(pitch_buffer_24kHz, lpc);
+      ComputeLpResidual(lpc, pitch_buffer_24kHz, computed_lp_residual);
+      ExpectNearAbsolute(expected_lp_residual, computed_lp_residual, kFloatMin);
+    }
+  }
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
new file mode 100644
index 0000000000..419620fc0c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
+
+#include <array>
+#include <cstddef>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+PitchEstimator::PitchEstimator(const AvailableCpuFeatures& cpu_features)
+    : cpu_features_(cpu_features),
+      y_energy_24kHz_(kRefineNumLags24kHz, 0.f),
+      pitch_buffer_12kHz_(kBufSize12kHz),
+      auto_correlation_12kHz_(kNumLags12kHz) {}
+
+PitchEstimator::~PitchEstimator() = default;
+
+int PitchEstimator::Estimate(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer) {
+  rtc::ArrayView<float, kBufSize12kHz> pitch_buffer_12kHz_view(
+      pitch_buffer_12kHz_.data(), kBufSize12kHz);
+  RTC_DCHECK_EQ(pitch_buffer_12kHz_.size(), pitch_buffer_12kHz_view.size());
+  rtc::ArrayView<float, kNumLags12kHz> auto_correlation_12kHz_view(
+      auto_correlation_12kHz_.data(), kNumLags12kHz);
+  RTC_DCHECK_EQ(auto_correlation_12kHz_.size(),
+                auto_correlation_12kHz_view.size());
+
+  // TODO(bugs.chromium.org/10480): Use `cpu_features_` to estimate pitch.
+  // Perform the initial pitch search at 12 kHz.
+  Decimate2x(pitch_buffer, pitch_buffer_12kHz_view);
+  auto_corr_calculator_.ComputeOnPitchBuffer(pitch_buffer_12kHz_view,
+                                             auto_correlation_12kHz_view);
+  CandidatePitchPeriods pitch_periods = ComputePitchPeriod12kHz(
+      pitch_buffer_12kHz_view, auto_correlation_12kHz_view, cpu_features_);
+  // The refinement is done using the pitch buffer that contains 24 kHz samples.
+  // Therefore, adapt the inverted lags in `pitch_candidates_inv_lags` from 12
+  // to 24 kHz.
+  pitch_periods.best *= 2;
+  pitch_periods.second_best *= 2;
+
+  // Refine the initial pitch period estimation from 12 kHz to 48 kHz.
+  // Pre-compute frame energies at 24 kHz.
+  rtc::ArrayView<float, kRefineNumLags24kHz> y_energy_24kHz_view(
+      y_energy_24kHz_.data(), kRefineNumLags24kHz);
+  RTC_DCHECK_EQ(y_energy_24kHz_.size(), y_energy_24kHz_view.size());
+  ComputeSlidingFrameSquareEnergies24kHz(pitch_buffer, y_energy_24kHz_view,
+                                         cpu_features_);
+  // Estimation at 48 kHz.
+  const int pitch_lag_48kHz = ComputePitchPeriod48kHz(
+      pitch_buffer, y_energy_24kHz_view, pitch_periods, cpu_features_);
+  last_pitch_48kHz_ = ComputeExtendedPitchPeriod48kHz(
+      pitch_buffer, y_energy_24kHz_view,
+      /*initial_pitch_period_48kHz=*/kMaxPitch48kHz - pitch_lag_48kHz,
+      last_pitch_48kHz_, cpu_features_);
+  return last_pitch_48kHz_.period;
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.h
new file mode 100644
index 0000000000..42c448eb56
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/auto_correlation.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
+#include "rtc_base/gtest_prod_util.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Pitch estimator.
+class PitchEstimator {
+ public:
+  explicit PitchEstimator(const AvailableCpuFeatures& cpu_features);
+  PitchEstimator(const PitchEstimator&) = delete;
+  PitchEstimator& operator=(const PitchEstimator&) = delete;
+  ~PitchEstimator();
+  // Returns the estimated pitch period at 48 kHz.
+  int Estimate(rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer);
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(RnnVadTest, PitchSearchWithinTolerance);
+  float GetLastPitchStrengthForTesting() const {
+    return last_pitch_48kHz_.strength;
+  }
+
+  const AvailableCpuFeatures cpu_features_;
+  PitchInfo last_pitch_48kHz_{};
+  AutoCorrelationCalculator auto_corr_calculator_;
+  std::vector<float> y_energy_24kHz_;
+  std::vector<float> pitch_buffer_12kHz_;
+  std::vector<float> auto_correlation_12kHz_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
new file mode 100644
index 0000000000..e8c912518d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
@@ -0,0 +1,513 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
+
+#include <stdlib.h>
+
+#include <algorithm>
+#include <cmath>
+#include <cstddef>
+#include <numeric>
+
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+float ComputeAutoCorrelation(
+    int inverted_lag,
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    const VectorMath& vector_math) {
+  RTC_DCHECK_LT(inverted_lag, kBufSize24kHz);
+  RTC_DCHECK_LT(inverted_lag, kRefineNumLags24kHz);
+  static_assert(kMaxPitch24kHz < kBufSize24kHz, "");
+  return vector_math.DotProduct(
+      pitch_buffer.subview(/*offset=*/kMaxPitch24kHz),
+      pitch_buffer.subview(inverted_lag, kFrameSize20ms24kHz));
+}
+
+// Given an auto-correlation coefficient `curr_auto_correlation` and its
+// neighboring values `prev_auto_correlation` and `next_auto_correlation`
+// computes a pseudo-interpolation offset to be applied to the pitch period
+// associated to `curr`. The output is a lag in {-1, 0, +1}.
+// TODO(bugs.webrtc.org/9076): Consider removing this method.
+// `GetPitchPseudoInterpolationOffset()` it is relevant only if the spectral
+// analysis works at a sample rate that is twice as that of the pitch buffer;
+// In particular, it is not relevant for the estimated pitch period feature fed
+// into the RNN.
+int GetPitchPseudoInterpolationOffset(float prev_auto_correlation,
+                                      float curr_auto_correlation,
+                                      float next_auto_correlation) {
+  if ((next_auto_correlation - prev_auto_correlation) >
+      0.7f * (curr_auto_correlation - prev_auto_correlation)) {
+    return 1;  // `next_auto_correlation` is the largest auto-correlation
+               // coefficient.
+  } else if ((prev_auto_correlation - next_auto_correlation) >
+             0.7f * (curr_auto_correlation - next_auto_correlation)) {
+    return -1;  // `prev_auto_correlation` is the largest auto-correlation
+                // coefficient.
+  }
+  return 0;
+}
+
+// Refines a pitch period `lag` encoded as lag with pseudo-interpolation. The
+// output sample rate is twice as that of `lag`.
+int PitchPseudoInterpolationLagPitchBuf(
+    int lag,
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    const VectorMath& vector_math) {
+  int offset = 0;
+  // Cannot apply pseudo-interpolation at the boundaries.
+  if (lag > 0 && lag < kMaxPitch24kHz) {
+    const int inverted_lag = kMaxPitch24kHz - lag;
+    offset = GetPitchPseudoInterpolationOffset(
+        ComputeAutoCorrelation(inverted_lag + 1, pitch_buffer, vector_math),
+        ComputeAutoCorrelation(inverted_lag, pitch_buffer, vector_math),
+        ComputeAutoCorrelation(inverted_lag - 1, pitch_buffer, vector_math));
+  }
+  return 2 * lag + offset;
+}
+
+// Integer multipliers used in ComputeExtendedPitchPeriod48kHz() when
+// looking for sub-harmonics.
+// The values have been chosen to serve the following algorithm. Given the
+// initial pitch period T, we examine whether one of its harmonics is the true
+// fundamental frequency. We consider T/k with k in {2, ..., 15}. For each of
+// these harmonics, in addition to the pitch strength of itself, we choose one
+// multiple of its pitch period, n*T/k, to validate it (by averaging their pitch
+// strengths). The multiplier n is chosen so that n*T/k is used only one time
+// over all k. When for example k = 4, we should also expect a peak at 3*T/4.
+// When k = 8 instead we don't want to look at 2*T/8, since we have already
+// checked T/4 before. Instead, we look at T*3/8.
+// The array can be generate in Python as follows:
+//   from fractions import Fraction
+//   # Smallest positive integer not in X.
+//   def mex(X):
+//     for i in range(1, int(max(X)+2)):
+//       if i not in X:
+//         return i
+//   # Visited multiples of the period.
+//   S = {1}
+//   for n in range(2, 16):
+//     sn = mex({n * i for i in S} | {1})
+//     S = S | {Fraction(1, n), Fraction(sn, n)}
+//     print(sn, end=', ')
+constexpr std::array<int, 14> kSubHarmonicMultipliers = {
+    {3, 2, 3, 2, 5, 2, 3, 2, 3, 2, 5, 2, 3, 2}};
+
+struct Range {
+  int min;
+  int max;
+};
+
+// Number of analyzed pitches to the left(right) of a pitch candidate.
+constexpr int kPitchNeighborhoodRadius = 2;
+
+// Creates a pitch period interval centered in `inverted_lag` with hard-coded
+// radius. Clipping is applied so that the interval is always valid for a 24 kHz
+// pitch buffer.
+Range CreateInvertedLagRange(int inverted_lag) {
+  return {std::max(inverted_lag - kPitchNeighborhoodRadius, 0),
+          std::min(inverted_lag + kPitchNeighborhoodRadius,
+                   kInitialNumLags24kHz - 1)};
+}
+
+constexpr int kNumPitchCandidates = 2;  // Best and second best.
+// Maximum number of analyzed pitch periods.
+constexpr int kMaxPitchPeriods24kHz =
+    kNumPitchCandidates * (2 * kPitchNeighborhoodRadius + 1);
+
+// Collection of inverted lags.
+class InvertedLagsIndex {
+ public:
+  InvertedLagsIndex() : num_entries_(0) {}
+  // Adds an inverted lag to the index. Cannot add more than
+  // `kMaxPitchPeriods24kHz` values.
+  void Append(int inverted_lag) {
+    RTC_DCHECK_LT(num_entries_, kMaxPitchPeriods24kHz);
+    inverted_lags_[num_entries_++] = inverted_lag;
+  }
+  const int* data() const { return inverted_lags_.data(); }
+  int size() const { return num_entries_; }
+
+ private:
+  std::array<int, kMaxPitchPeriods24kHz> inverted_lags_;
+  int num_entries_;
+};
+
+// Computes the auto correlation coefficients for the inverted lags in the
+// closed interval `inverted_lags`. Updates `inverted_lags_index` by appending
+// the inverted lags for the computed auto correlation values.
+void ComputeAutoCorrelation(
+    Range inverted_lags,
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<float, kInitialNumLags24kHz> auto_correlation,
+    InvertedLagsIndex& inverted_lags_index,
+    const VectorMath& vector_math) {
+  // Check valid range.
+  RTC_DCHECK_LE(inverted_lags.min, inverted_lags.max);
+  // Trick to avoid zero initialization of `auto_correlation`.
+  // Needed by the pseudo-interpolation.
+  if (inverted_lags.min > 0) {
+    auto_correlation[inverted_lags.min - 1] = 0.f;
+  }
+  if (inverted_lags.max < kInitialNumLags24kHz - 1) {
+    auto_correlation[inverted_lags.max + 1] = 0.f;
+  }
+  // Check valid `inverted_lag` indexes.
+  RTC_DCHECK_GE(inverted_lags.min, 0);
+  RTC_DCHECK_LT(inverted_lags.max, kInitialNumLags24kHz);
+  for (int inverted_lag = inverted_lags.min; inverted_lag <= inverted_lags.max;
+       ++inverted_lag) {
+    auto_correlation[inverted_lag] =
+        ComputeAutoCorrelation(inverted_lag, pitch_buffer, vector_math);
+    inverted_lags_index.Append(inverted_lag);
+  }
+}
+
+// Searches the strongest pitch period at 24 kHz and returns its inverted lag at
+// 48 kHz.
+int ComputePitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const int> inverted_lags,
+    rtc::ArrayView<const float, kInitialNumLags24kHz> auto_correlation,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    const VectorMath& vector_math) {
+  static_assert(kMaxPitch24kHz > kInitialNumLags24kHz, "");
+  static_assert(kMaxPitch24kHz < kBufSize24kHz, "");
+  int best_inverted_lag = 0;     // Pitch period.
+  float best_numerator = -1.f;   // Pitch strength numerator.
+  float best_denominator = 0.f;  // Pitch strength denominator.
+  for (int inverted_lag : inverted_lags) {
+    // A pitch candidate must have positive correlation.
+    if (auto_correlation[inverted_lag] > 0.f) {
+      // Auto-correlation energy normalized by frame energy.
+      const float numerator =
+          auto_correlation[inverted_lag] * auto_correlation[inverted_lag];
+      const float denominator = y_energy[inverted_lag];
+      // Compare numerator/denominator ratios without using divisions.
+      if (numerator * best_denominator > best_numerator * denominator) {
+        best_inverted_lag = inverted_lag;
+        best_numerator = numerator;
+        best_denominator = denominator;
+      }
+    }
+  }
+  // Pseudo-interpolation to transform `best_inverted_lag` (24 kHz pitch) to a
+  // 48 kHz pitch period.
+  if (best_inverted_lag == 0 || best_inverted_lag >= kInitialNumLags24kHz - 1) {
+    // Cannot apply pseudo-interpolation at the boundaries.
+    return best_inverted_lag * 2;
+  }
+  int offset = GetPitchPseudoInterpolationOffset(
+      auto_correlation[best_inverted_lag + 1],
+      auto_correlation[best_inverted_lag],
+      auto_correlation[best_inverted_lag - 1]);
+  // TODO(bugs.webrtc.org/9076): When retraining, check if `offset` below should
+  // be subtracted since `inverted_lag` is an inverted lag but offset is a lag.
+  return 2 * best_inverted_lag + offset;
+}
+
+// Returns an alternative pitch period for `pitch_period` given a `multiplier`
+// and a `divisor` of the period.
+constexpr int GetAlternativePitchPeriod(int pitch_period,
+                                        int multiplier,
+                                        int divisor) {
+  RTC_DCHECK_GT(divisor, 0);
+  // Same as `round(multiplier * pitch_period / divisor)`.
+  return (2 * multiplier * pitch_period + divisor) / (2 * divisor);
+}
+
+// Returns true if the alternative pitch period is stronger than the initial one
+// given the last estimated pitch and the value of `period_divisor` used to
+// compute the alternative pitch period via `GetAlternativePitchPeriod()`.
+bool IsAlternativePitchStrongerThanInitial(PitchInfo last,
+                                           PitchInfo initial,
+                                           PitchInfo alternative,
+                                           int period_divisor) {
+  // Initial pitch period candidate thresholds for a sample rate of 24 kHz.
+  // Computed as [5*k*k for k in range(16)].
+  constexpr std::array<int, 14> kInitialPitchPeriodThresholds = {
+      {20, 45, 80, 125, 180, 245, 320, 405, 500, 605, 720, 845, 980, 1125}};
+  static_assert(
+      kInitialPitchPeriodThresholds.size() == kSubHarmonicMultipliers.size(),
+      "");
+  RTC_DCHECK_GE(last.period, 0);
+  RTC_DCHECK_GE(initial.period, 0);
+  RTC_DCHECK_GE(alternative.period, 0);
+  RTC_DCHECK_GE(period_divisor, 2);
+  // Compute a term that lowers the threshold when `alternative.period` is close
+  // to the last estimated period `last.period` - i.e., pitch tracking.
+  float lower_threshold_term = 0.f;
+  if (std::abs(alternative.period - last.period) <= 1) {
+    // The candidate pitch period is within 1 sample from the last one.
+    // Make the candidate at `alternative.period` very easy to be accepted.
+    lower_threshold_term = last.strength;
+  } else if (std::abs(alternative.period - last.period) == 2 &&
+             initial.period >
+                 kInitialPitchPeriodThresholds[period_divisor - 2]) {
+    // The candidate pitch period is 2 samples far from the last one and the
+    // period `initial.period` (from which `alternative.period` has been
+    // derived) is greater than a threshold. Make `alternative.period` easy to
+    // be accepted.
+    lower_threshold_term = 0.5f * last.strength;
+  }
+  // Set the threshold based on the strength of the initial estimate
+  // `initial.period`. Also reduce the chance of false positives caused by a
+  // bias towards high frequencies (originating from short-term correlations).
+  float threshold =
+      std::max(0.3f, 0.7f * initial.strength - lower_threshold_term);
+  if (alternative.period < 3 * kMinPitch24kHz) {
+    // High frequency.
+    threshold = std::max(0.4f, 0.85f * initial.strength - lower_threshold_term);
+  } else if (alternative.period < 2 * kMinPitch24kHz) {
+    // Even higher frequency.
+    threshold = std::max(0.5f, 0.9f * initial.strength - lower_threshold_term);
+  }
+  return alternative.strength > threshold;
+}
+
+}  // namespace
+
+void Decimate2x(rtc::ArrayView<const float, kBufSize24kHz> src,
+                rtc::ArrayView<float, kBufSize12kHz> dst) {
+  // TODO(bugs.webrtc.org/9076): Consider adding anti-aliasing filter.
+  static_assert(2 * kBufSize12kHz == kBufSize24kHz, "");
+  for (int i = 0; i < kBufSize12kHz; ++i) {
+    dst[i] = src[2 * i];
+  }
+}
+
+void ComputeSlidingFrameSquareEnergies24kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<float, kRefineNumLags24kHz> y_energy,
+    AvailableCpuFeatures cpu_features) {
+  VectorMath vector_math(cpu_features);
+  static_assert(kFrameSize20ms24kHz < kBufSize24kHz, "");
+  const auto frame_20ms_view = pitch_buffer.subview(0, kFrameSize20ms24kHz);
+  float yy = vector_math.DotProduct(frame_20ms_view, frame_20ms_view);
+  y_energy[0] = yy;
+  static_assert(kMaxPitch24kHz - 1 + kFrameSize20ms24kHz < kBufSize24kHz, "");
+  static_assert(kMaxPitch24kHz < kRefineNumLags24kHz, "");
+  for (int inverted_lag = 0; inverted_lag < kMaxPitch24kHz; ++inverted_lag) {
+    yy -= pitch_buffer[inverted_lag] * pitch_buffer[inverted_lag];
+    yy += pitch_buffer[inverted_lag + kFrameSize20ms24kHz] *
+          pitch_buffer[inverted_lag + kFrameSize20ms24kHz];
+    yy = std::max(1.f, yy);
+    y_energy[inverted_lag + 1] = yy;
+  }
+}
+
+CandidatePitchPeriods ComputePitchPeriod12kHz(
+    rtc::ArrayView<const float, kBufSize12kHz> pitch_buffer,
+    rtc::ArrayView<const float, kNumLags12kHz> auto_correlation,
+    AvailableCpuFeatures cpu_features) {
+  static_assert(kMaxPitch12kHz > kNumLags12kHz, "");
+  static_assert(kMaxPitch12kHz < kBufSize12kHz, "");
+
+  // Stores a pitch candidate period and strength information.
+  struct PitchCandidate {
+    // Pitch period encoded as inverted lag.
+    int period_inverted_lag = 0;
+    // Pitch strength encoded as a ratio.
+    float strength_numerator = -1.f;
+    float strength_denominator = 0.f;
+    // Compare the strength of two pitch candidates.
+    bool HasStrongerPitchThan(const PitchCandidate& b) const {
+      // Comparing the numerator/denominator ratios without using divisions.
+      return strength_numerator * b.strength_denominator >
+             b.strength_numerator * strength_denominator;
+    }
+  };
+
+  VectorMath vector_math(cpu_features);
+  static_assert(kFrameSize20ms12kHz + 1 < kBufSize12kHz, "");
+  const auto frame_view = pitch_buffer.subview(0, kFrameSize20ms12kHz + 1);
+  float denominator = 1.f + vector_math.DotProduct(frame_view, frame_view);
+  // Search best and second best pitches by looking at the scaled
+  // auto-correlation.
+  PitchCandidate best;
+  PitchCandidate second_best;
+  second_best.period_inverted_lag = 1;
+  for (int inverted_lag = 0; inverted_lag < kNumLags12kHz; ++inverted_lag) {
+    // A pitch candidate must have positive correlation.
+    if (auto_correlation[inverted_lag] > 0.f) {
+      PitchCandidate candidate{
+          inverted_lag,
+          auto_correlation[inverted_lag] * auto_correlation[inverted_lag],
+          denominator};
+      if (candidate.HasStrongerPitchThan(second_best)) {
+        if (candidate.HasStrongerPitchThan(best)) {
+          second_best = best;
+          best = candidate;
+        } else {
+          second_best = candidate;
+        }
+      }
+    }
+    // Update `squared_energy_y` for the next inverted lag.
+    const float y_old = pitch_buffer[inverted_lag];
+    const float y_new = pitch_buffer[inverted_lag + kFrameSize20ms12kHz];
+    denominator -= y_old * y_old;
+    denominator += y_new * y_new;
+    denominator = std::max(0.f, denominator);
+  }
+  return {best.period_inverted_lag, second_best.period_inverted_lag};
+}
+
+int ComputePitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    CandidatePitchPeriods pitch_candidates,
+    AvailableCpuFeatures cpu_features) {
+  // Compute the auto-correlation terms only for neighbors of the two pitch
+  // candidates (best and second best).
+  std::array<float, kInitialNumLags24kHz> auto_correlation;
+  InvertedLagsIndex inverted_lags_index;
+  // Create two inverted lag ranges so that `r1` precedes `r2`.
+  const bool swap_candidates =
+      pitch_candidates.best > pitch_candidates.second_best;
+  const Range r1 = CreateInvertedLagRange(
+      swap_candidates ? pitch_candidates.second_best : pitch_candidates.best);
+  const Range r2 = CreateInvertedLagRange(
+      swap_candidates ? pitch_candidates.best : pitch_candidates.second_best);
+  // Check valid ranges.
+  RTC_DCHECK_LE(r1.min, r1.max);
+  RTC_DCHECK_LE(r2.min, r2.max);
+  // Check `r1` precedes `r2`.
+  RTC_DCHECK_LE(r1.min, r2.min);
+  RTC_DCHECK_LE(r1.max, r2.max);
+  VectorMath vector_math(cpu_features);
+  if (r1.max + 1 >= r2.min) {
+    // Overlapping or adjacent ranges.
+    ComputeAutoCorrelation({r1.min, r2.max}, pitch_buffer, auto_correlation,
+                           inverted_lags_index, vector_math);
+  } else {
+    // Disjoint ranges.
+    ComputeAutoCorrelation(r1, pitch_buffer, auto_correlation,
+                           inverted_lags_index, vector_math);
+    ComputeAutoCorrelation(r2, pitch_buffer, auto_correlation,
+                           inverted_lags_index, vector_math);
+  }
+  return ComputePitchPeriod48kHz(pitch_buffer, inverted_lags_index,
+                                 auto_correlation, y_energy, vector_math);
+}
+
+PitchInfo ComputeExtendedPitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    int initial_pitch_period_48kHz,
+    PitchInfo last_pitch_48kHz,
+    AvailableCpuFeatures cpu_features) {
+  RTC_DCHECK_LE(kMinPitch48kHz, initial_pitch_period_48kHz);
+  RTC_DCHECK_LE(initial_pitch_period_48kHz, kMaxPitch48kHz);
+
+  // Stores information for a refined pitch candidate.
+  struct RefinedPitchCandidate {
+    int period;
+    float strength;
+    // Additional strength data used for the final pitch estimation.
+    float xy;        // Auto-correlation.
+    float y_energy;  // Energy of the sliding frame `y`.
+  };
+
+  const float x_energy = y_energy[kMaxPitch24kHz];
+  const auto pitch_strength = [x_energy](float xy, float y_energy) {
+    RTC_DCHECK_GE(x_energy * y_energy, 0.f);
+    return xy / std::sqrt(1.f + x_energy * y_energy);
+  };
+  VectorMath vector_math(cpu_features);
+
+  // Initialize the best pitch candidate with `initial_pitch_period_48kHz`.
+  RefinedPitchCandidate best_pitch;
+  best_pitch.period =
+      std::min(initial_pitch_period_48kHz / 2, kMaxPitch24kHz - 1);
+  best_pitch.xy = ComputeAutoCorrelation(kMaxPitch24kHz - best_pitch.period,
+                                         pitch_buffer, vector_math);
+  best_pitch.y_energy = y_energy[kMaxPitch24kHz - best_pitch.period];
+  best_pitch.strength = pitch_strength(best_pitch.xy, best_pitch.y_energy);
+  // Keep a copy of the initial pitch candidate.
+  const PitchInfo initial_pitch{best_pitch.period, best_pitch.strength};
+  // 24 kHz version of the last estimated pitch.
+  const PitchInfo last_pitch{last_pitch_48kHz.period / 2,
+                             last_pitch_48kHz.strength};
+
+  // Find `max_period_divisor` such that the result of
+  // `GetAlternativePitchPeriod(initial_pitch_period, 1, max_period_divisor)`
+  // equals `kMinPitch24kHz`.
+  const int max_period_divisor =
+      (2 * initial_pitch.period) / (2 * kMinPitch24kHz - 1);
+  for (int period_divisor = 2; period_divisor <= max_period_divisor;
+       ++period_divisor) {
+    PitchInfo alternative_pitch;
+    alternative_pitch.period = GetAlternativePitchPeriod(
+        initial_pitch.period, /*multiplier=*/1, period_divisor);
+    RTC_DCHECK_GE(alternative_pitch.period, kMinPitch24kHz);
+    // When looking at `alternative_pitch.period`, we also look at one of its
+    // sub-harmonics. `kSubHarmonicMultipliers` is used to know where to look.
+    // `period_divisor` == 2 is a special case since `dual_alternative_period`
+    // might be greater than the maximum pitch period.
+    int dual_alternative_period = GetAlternativePitchPeriod(
+        initial_pitch.period, kSubHarmonicMultipliers[period_divisor - 2],
+        period_divisor);
+    RTC_DCHECK_GT(dual_alternative_period, 0);
+    if (period_divisor == 2 && dual_alternative_period > kMaxPitch24kHz) {
+      dual_alternative_period = initial_pitch.period;
+    }
+    RTC_DCHECK_NE(alternative_pitch.period, dual_alternative_period)
+        << "The lower pitch period and the additional sub-harmonic must not "
+           "coincide.";
+    // Compute an auto-correlation score for the primary pitch candidate
+    // `alternative_pitch.period` by also looking at its possible sub-harmonic
+    // `dual_alternative_period`.
+    const float xy_primary_period = ComputeAutoCorrelation(
+        kMaxPitch24kHz - alternative_pitch.period, pitch_buffer, vector_math);
+    // TODO(webrtc:10480): Copy `xy_primary_period` if the secondary period is
+    // equal to the primary one.
+    const float xy_secondary_period = ComputeAutoCorrelation(
+        kMaxPitch24kHz - dual_alternative_period, pitch_buffer, vector_math);
+    const float xy = 0.5f * (xy_primary_period + xy_secondary_period);
+    const float yy =
+        0.5f * (y_energy[kMaxPitch24kHz - alternative_pitch.period] +
+                y_energy[kMaxPitch24kHz - dual_alternative_period]);
+    alternative_pitch.strength = pitch_strength(xy, yy);
+
+    // Maybe update best period.
+    if (IsAlternativePitchStrongerThanInitial(
+            last_pitch, initial_pitch, alternative_pitch, period_divisor)) {
+      best_pitch = {alternative_pitch.period, alternative_pitch.strength, xy,
+                    yy};
+    }
+  }
+
+  // Final pitch strength and period.
+  best_pitch.xy = std::max(0.f, best_pitch.xy);
+  RTC_DCHECK_LE(0.f, best_pitch.y_energy);
+  float final_pitch_strength =
+      (best_pitch.y_energy <= best_pitch.xy)
+          ? 1.f
+          : best_pitch.xy / (best_pitch.y_energy + 1.f);
+  final_pitch_strength = std::min(best_pitch.strength, final_pitch_strength);
+  int final_pitch_period_48kHz = std::max(
+      kMinPitch48kHz, PitchPseudoInterpolationLagPitchBuf(
+                          best_pitch.period, pitch_buffer, vector_math));
+
+  return {final_pitch_period_48kHz, final_pitch_strength};
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
new file mode 100644
index 0000000000..aa2dd13745
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <utility>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Performs 2x decimation without any anti-aliasing filter.
+void Decimate2x(rtc::ArrayView<const float, kBufSize24kHz> src,
+                rtc::ArrayView<float, kBufSize12kHz> dst);
+
+// Key concepts and keywords used below in this file.
+//
+// The pitch estimation relies on a pitch buffer, which is an array-like data
+// structured designed as follows:
+//
+// |....A....|.....B.....|
+//
+// The part on the left, named `A` contains the oldest samples, whereas `B`
+// contains the most recent ones. The size of `A` corresponds to the maximum
+// pitch period, that of `B` to the analysis frame size (e.g., 16 ms and 20 ms
+// respectively).
+//
+// Pitch estimation is essentially based on the analysis of two 20 ms frames
+// extracted from the pitch buffer. One frame, called `x`, is kept fixed and
+// corresponds to `B` - i.e., the most recent 20 ms. The other frame, called
+// `y`, is extracted from different parts of the buffer instead.
+//
+// The offset between `x` and `y` corresponds to a specific pitch period.
+// For instance, if `y` is positioned at the beginning of the pitch buffer, then
+// the cross-correlation between `x` and `y` can be used as an indication of the
+// strength for the maximum pitch.
+//
+// Such an offset can be encoded in two ways:
+// - As a lag, which is the index in the pitch buffer for the first item in `y`
+// - As an inverted lag, which is the number of samples from the beginning of
+//   `x` and the end of `y`
+//
+// |---->| lag
+// |....A....|.....B.....|
+//       |<--| inverted lag
+//       |.....y.....| `y` 20 ms frame
+//
+// The inverted lag has the advantage of being directly proportional to the
+// corresponding pitch period.
+
+// Computes the sum of squared samples for every sliding frame `y` in the pitch
+// buffer. The indexes of `y_energy` are inverted lags.
+void ComputeSlidingFrameSquareEnergies24kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<float, kRefineNumLags24kHz> y_energy,
+    AvailableCpuFeatures cpu_features);
+
+// Top-2 pitch period candidates. Unit: number of samples - i.e., inverted lags.
+struct CandidatePitchPeriods {
+  int best;
+  int second_best;
+};
+
+// Computes the candidate pitch periods at 12 kHz given a view on the 12 kHz
+// pitch buffer and the auto-correlation values (having inverted lags as
+// indexes).
+CandidatePitchPeriods ComputePitchPeriod12kHz(
+    rtc::ArrayView<const float, kBufSize12kHz> pitch_buffer,
+    rtc::ArrayView<const float, kNumLags12kHz> auto_correlation,
+    AvailableCpuFeatures cpu_features);
+
+// Computes the pitch period at 48 kHz given a view on the 24 kHz pitch buffer,
+// the energies for the sliding frames `y` at 24 kHz and the pitch period
+// candidates at 24 kHz (encoded as inverted lag).
+int ComputePitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    CandidatePitchPeriods pitch_candidates_24kHz,
+    AvailableCpuFeatures cpu_features);
+
+struct PitchInfo {
+  int period;
+  float strength;
+};
+
+// Computes the pitch period at 48 kHz searching in an extended pitch range
+// given a view on the 24 kHz pitch buffer, the energies for the sliding frames
+// `y` at 24 kHz, the initial 48 kHz estimation (computed by
+// `ComputePitchPeriod48kHz()`) and the last estimated pitch.
+PitchInfo ComputeExtendedPitchPeriod48kHz(
+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buffer,
+    rtc::ArrayView<const float, kRefineNumLags24kHz> y_energy,
+    int initial_pitch_period_48kHz,
+    PitchInfo last_pitch_48kHz,
+    AvailableCpuFeatures cpu_features);
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc
new file mode 100644
index 0000000000..2a6e68f157
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc
@@ -0,0 +1,217 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
+
+#include <array>
+#include <string>
+#include <tuple>
+
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "rtc_base/strings/string_builder.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kTestPitchPeriodsLow = 3 * kMinPitch48kHz / 2;
+constexpr int kTestPitchPeriodsHigh = (3 * kMinPitch48kHz + kMaxPitch48kHz) / 2;
+
+constexpr float kTestPitchStrengthLow = 0.35f;
+constexpr float kTestPitchStrengthHigh = 0.75f;
+
+template <class T>
+std::string PrintTestIndexAndCpuFeatures(
+    const ::testing::TestParamInfo<T>& info) {
+  rtc::StringBuilder builder;
+  builder << info.index << "_" << info.param.cpu_features.ToString();
+  return builder.str();
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back(NoAvailableCpuFeatures());
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.avx2) {
+    v.push_back({/*sse2=*/false, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  return v;
+}
+
+// Checks that the frame-wise sliding square energy function produces output
+// within tolerance given test input data.
+TEST(RnnVadTest, ComputeSlidingFrameSquareEnergies24kHzWithinTolerance) {
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+
+  PitchTestData test_data;
+  std::array<float, kRefineNumLags24kHz> computed_output;
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
+                                         computed_output, cpu_features);
+  auto square_energies_view = test_data.SquareEnergies24kHzView();
+  ExpectNearAbsolute({square_energies_view.data(), square_energies_view.size()},
+                     computed_output, 1e-3f);
+}
+
+// Checks that the estimated pitch period is bit-exact given test input data.
+TEST(RnnVadTest, ComputePitchPeriod12kHzBitExactness) {
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+
+  PitchTestData test_data;
+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
+  Decimate2x(test_data.PitchBuffer24kHzView(), pitch_buf_decimated);
+  CandidatePitchPeriods pitch_candidates;
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  pitch_candidates = ComputePitchPeriod12kHz(
+      pitch_buf_decimated, test_data.AutoCorrelation12kHzView(), cpu_features);
+  EXPECT_EQ(pitch_candidates.best, 140);
+  EXPECT_EQ(pitch_candidates.second_best, 142);
+}
+
+// Checks that the refined pitch period is bit-exact given test input data.
+TEST(RnnVadTest, ComputePitchPeriod48kHzBitExactness) {
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+
+  PitchTestData test_data;
+  std::vector<float> y_energy(kRefineNumLags24kHz);
+  rtc::ArrayView<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
+                                                           kRefineNumLags24kHz);
+  ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
+                                         y_energy_view, cpu_features);
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  EXPECT_EQ(
+      ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view,
+                              /*pitch_candidates=*/{280, 284}, cpu_features),
+      560);
+  EXPECT_EQ(
+      ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view,
+                              /*pitch_candidates=*/{260, 284}, cpu_features),
+      568);
+}
+
+struct PitchCandidatesParameters {
+  CandidatePitchPeriods pitch_candidates;
+  AvailableCpuFeatures cpu_features;
+};
+
+class PitchCandidatesParametrization
+    : public ::testing::TestWithParam<PitchCandidatesParameters> {};
+
+// Checks that the result of `ComputePitchPeriod48kHz()` does not depend on the
+// order of the input pitch candidates.
+TEST_P(PitchCandidatesParametrization,
+       ComputePitchPeriod48kHzOrderDoesNotMatter) {
+  const PitchCandidatesParameters params = GetParam();
+  const CandidatePitchPeriods swapped_pitch_candidates{
+      params.pitch_candidates.second_best, params.pitch_candidates.best};
+
+  PitchTestData test_data;
+  std::vector<float> y_energy(kRefineNumLags24kHz);
+  rtc::ArrayView<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
+                                                           kRefineNumLags24kHz);
+  ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
+                                         y_energy_view, params.cpu_features);
+  EXPECT_EQ(
+      ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view,
+                              params.pitch_candidates, params.cpu_features),
+      ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view,
+                              swapped_pitch_candidates, params.cpu_features));
+}
+
+std::vector<PitchCandidatesParameters> CreatePitchCandidatesParameters() {
+  std::vector<PitchCandidatesParameters> v;
+  for (AvailableCpuFeatures cpu_features : GetCpuFeaturesToTest()) {
+    v.push_back({{0, 2}, cpu_features});
+    v.push_back({{260, 284}, cpu_features});
+    v.push_back({{280, 284}, cpu_features});
+    v.push_back(
+        {{kInitialNumLags24kHz - 2, kInitialNumLags24kHz - 1}, cpu_features});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    PitchCandidatesParametrization,
+    ::testing::ValuesIn(CreatePitchCandidatesParameters()),
+    PrintTestIndexAndCpuFeatures<PitchCandidatesParameters>);
+
+struct ExtendedPitchPeriodSearchParameters {
+  int initial_pitch_period;
+  PitchInfo last_pitch;
+  PitchInfo expected_pitch;
+  AvailableCpuFeatures cpu_features;
+};
+
+class ExtendedPitchPeriodSearchParametrizaion
+    : public ::testing::TestWithParam<ExtendedPitchPeriodSearchParameters> {};
+
+// Checks that the computed pitch period is bit-exact and that the computed
+// pitch strength is within tolerance given test input data.
+TEST_P(ExtendedPitchPeriodSearchParametrizaion,
+       PeriodBitExactnessGainWithinTolerance) {
+  const ExtendedPitchPeriodSearchParameters params = GetParam();
+
+  PitchTestData test_data;
+  std::vector<float> y_energy(kRefineNumLags24kHz);
+  rtc::ArrayView<float, kRefineNumLags24kHz> y_energy_view(y_energy.data(),
+                                                           kRefineNumLags24kHz);
+  ComputeSlidingFrameSquareEnergies24kHz(test_data.PitchBuffer24kHzView(),
+                                         y_energy_view, params.cpu_features);
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+  const auto computed_output = ComputeExtendedPitchPeriod48kHz(
+      test_data.PitchBuffer24kHzView(), y_energy_view,
+      params.initial_pitch_period, params.last_pitch, params.cpu_features);
+  EXPECT_EQ(params.expected_pitch.period, computed_output.period);
+  EXPECT_NEAR(params.expected_pitch.strength, computed_output.strength, 1e-6f);
+}
+
+std::vector<ExtendedPitchPeriodSearchParameters>
+CreateExtendedPitchPeriodSearchParameters() {
+  std::vector<ExtendedPitchPeriodSearchParameters> v;
+  for (AvailableCpuFeatures cpu_features : GetCpuFeaturesToTest()) {
+    for (int last_pitch_period :
+         {kTestPitchPeriodsLow, kTestPitchPeriodsHigh}) {
+      for (float last_pitch_strength :
+           {kTestPitchStrengthLow, kTestPitchStrengthHigh}) {
+        v.push_back({kTestPitchPeriodsLow,
+                     {last_pitch_period, last_pitch_strength},
+                     {91, -0.0188608f},
+                     cpu_features});
+        v.push_back({kTestPitchPeriodsHigh,
+                     {last_pitch_period, last_pitch_strength},
+                     {475, -0.0904344f},
+                     cpu_features});
+      }
+    }
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    ExtendedPitchPeriodSearchParametrizaion,
+    ::testing::ValuesIn(CreateExtendedPitchPeriodSearchParameters()),
+    PrintTestIndexAndCpuFeatures<ExtendedPitchPeriodSearchParameters>);
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
new file mode 100644
index 0000000000..79b44b995c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Checks that the computed pitch period is bit-exact and that the computed
+// pitch gain is within tolerance given test input data.
+TEST(RnnVadTest, PitchSearchWithinTolerance) {
+  ChunksFileReader reader = CreateLpResidualAndPitchInfoReader();
+  const int num_frames = std::min(reader.num_chunks, 300);  // Max 3 s.
+  std::vector<float> lp_residual(kBufSize24kHz);
+  float expected_pitch_period, expected_pitch_strength;
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+  PitchEstimator pitch_estimator(cpu_features);
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    for (int i = 0; i < num_frames; ++i) {
+      SCOPED_TRACE(i);
+      ASSERT_TRUE(reader.reader->ReadChunk(lp_residual));
+      ASSERT_TRUE(reader.reader->ReadValue(expected_pitch_period));
+      ASSERT_TRUE(reader.reader->ReadValue(expected_pitch_strength));
+      int pitch_period =
+          pitch_estimator.Estimate({lp_residual.data(), kBufSize24kHz});
+      EXPECT_EQ(expected_pitch_period, pitch_period);
+      EXPECT_NEAR(expected_pitch_strength,
+                  pitch_estimator.GetLastPitchStrengthForTesting(), 15e-6f);
+    }
+  }
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
new file mode 100644
index 0000000000..a6f7fdd1a6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RING_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RING_BUFFER_H_
+
+#include <array>
+#include <cstring>
+#include <type_traits>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Ring buffer for N arrays of type T each one with size S.
+template <typename T, int S, int N>
+class RingBuffer {
+  static_assert(S > 0, "");
+  static_assert(N > 0, "");
+  static_assert(std::is_arithmetic<T>::value,
+                "Integral or floating point required.");
+
+ public:
+  RingBuffer() : tail_(0) {}
+  RingBuffer(const RingBuffer&) = delete;
+  RingBuffer& operator=(const RingBuffer&) = delete;
+  ~RingBuffer() = default;
+  // Set the ring buffer values to zero.
+  void Reset() { buffer_.fill(0); }
+  // Replace the least recently pushed array in the buffer with `new_values`.
+  void Push(rtc::ArrayView<const T, S> new_values) {
+    std::memcpy(buffer_.data() + S * tail_, new_values.data(), S * sizeof(T));
+    tail_ += 1;
+    if (tail_ == N)
+      tail_ = 0;
+  }
+  // Return an array view onto the array with a given delay. A view on the last
+  // and least recently push array is returned when `delay` is 0 and N - 1
+  // respectively.
+  rtc::ArrayView<const T, S> GetArrayView(int delay) const {
+    RTC_DCHECK_LE(0, delay);
+    RTC_DCHECK_LT(delay, N);
+    int offset = tail_ - 1 - delay;
+    if (offset < 0)
+      offset += N;
+    return {buffer_.data() + S * offset, S};
+  }
+
+ private:
+  int tail_;  // Index of the least recently pushed sub-array.
+  std::array<T, S * N> buffer_{};
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RING_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
new file mode 100644
index 0000000000..d11d4eac3e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Compare the elements of two given array views.
+template <typename T, std::ptrdiff_t S>
+void ExpectEq(rtc::ArrayView<const T, S> a, rtc::ArrayView<const T, S> b) {
+  for (int i = 0; i < S; ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_EQ(a[i], b[i]);
+  }
+}
+
+// Test push/read sequences.
+template <typename T, int S, int N>
+void TestRingBuffer() {
+  SCOPED_TRACE(N);
+  SCOPED_TRACE(S);
+  std::array<T, S> prev_pushed_array;
+  std::array<T, S> pushed_array;
+  rtc::ArrayView<const T, S> pushed_array_view(pushed_array.data(), S);
+
+  // Init.
+  RingBuffer<T, S, N> ring_buf;
+  ring_buf.GetArrayView(0);
+  pushed_array.fill(0);
+  ring_buf.Push(pushed_array_view);
+  ExpectEq(pushed_array_view, ring_buf.GetArrayView(0));
+
+  // Push N times and check most recent and second most recent.
+  for (T v = 1; v <= static_cast<T>(N); ++v) {
+    SCOPED_TRACE(v);
+    prev_pushed_array = pushed_array;
+    pushed_array.fill(v);
+    ring_buf.Push(pushed_array_view);
+    ExpectEq(pushed_array_view, ring_buf.GetArrayView(0));
+    if (N > 1) {
+      pushed_array.fill(v - 1);
+      ExpectEq(pushed_array_view, ring_buf.GetArrayView(1));
+    }
+  }
+
+  // Check buffer.
+  for (int delay = 2; delay < N; ++delay) {
+    SCOPED_TRACE(delay);
+    T expected_value = N - static_cast<T>(delay);
+    pushed_array.fill(expected_value);
+    ExpectEq(pushed_array_view, ring_buf.GetArrayView(delay));
+  }
+}
+
+// Check that for different delays, different views are returned.
+TEST(RnnVadTest, RingBufferArrayViews) {
+  constexpr int s = 3;
+  constexpr int n = 4;
+  RingBuffer<int, s, n> ring_buf;
+  std::array<int, s> pushed_array;
+  pushed_array.fill(1);
+  for (int k = 0; k <= n; ++k) {  // Push data n + 1 times.
+    SCOPED_TRACE(k);
+    // Check array views.
+    for (int i = 0; i < n; ++i) {
+      SCOPED_TRACE(i);
+      auto view_i = ring_buf.GetArrayView(i);
+      for (int j = i + 1; j < n; ++j) {
+        SCOPED_TRACE(j);
+        auto view_j = ring_buf.GetArrayView(j);
+        EXPECT_NE(view_i, view_j);
+      }
+    }
+    ring_buf.Push(pushed_array);
+  }
+}
+
+TEST(RnnVadTest, RingBufferUnsigned) {
+  TestRingBuffer<uint8_t, 1, 1>();
+  TestRingBuffer<uint8_t, 2, 5>();
+  TestRingBuffer<uint8_t, 5, 2>();
+  TestRingBuffer<uint8_t, 5, 5>();
+}
+
+TEST(RnnVadTest, RingBufferSigned) {
+  TestRingBuffer<int, 1, 1>();
+  TestRingBuffer<int, 2, 5>();
+  TestRingBuffer<int, 5, 2>();
+  TestRingBuffer<int, 5, 5>();
+}
+
+TEST(RnnVadTest, RingBufferFloating) {
+  TestRingBuffer<float, 1, 1>();
+  TestRingBuffer<float, 2, 5>();
+  TestRingBuffer<float, 5, 2>();
+  TestRingBuffer<float, 5, 5>();
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
new file mode 100644
index 0000000000..475bef9775
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
+
+#include "rtc_base/checks.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+using ::rnnoise::kInputLayerInputSize;
+static_assert(kFeatureVectorSize == kInputLayerInputSize, "");
+using ::rnnoise::kInputDenseBias;
+using ::rnnoise::kInputDenseWeights;
+using ::rnnoise::kInputLayerOutputSize;
+static_assert(kInputLayerOutputSize <= kFullyConnectedLayerMaxUnits, "");
+
+using ::rnnoise::kHiddenGruBias;
+using ::rnnoise::kHiddenGruRecurrentWeights;
+using ::rnnoise::kHiddenGruWeights;
+using ::rnnoise::kHiddenLayerOutputSize;
+static_assert(kHiddenLayerOutputSize <= kGruLayerMaxUnits, "");
+
+using ::rnnoise::kOutputDenseBias;
+using ::rnnoise::kOutputDenseWeights;
+using ::rnnoise::kOutputLayerOutputSize;
+static_assert(kOutputLayerOutputSize <= kFullyConnectedLayerMaxUnits, "");
+
+}  // namespace
+
+RnnVad::RnnVad(const AvailableCpuFeatures& cpu_features)
+    : input_(kInputLayerInputSize,
+             kInputLayerOutputSize,
+             kInputDenseBias,
+             kInputDenseWeights,
+             ActivationFunction::kTansigApproximated,
+             cpu_features,
+             /*layer_name=*/"FC1"),
+      hidden_(kInputLayerOutputSize,
+              kHiddenLayerOutputSize,
+              kHiddenGruBias,
+              kHiddenGruWeights,
+              kHiddenGruRecurrentWeights,
+              cpu_features,
+              /*layer_name=*/"GRU1"),
+      output_(kHiddenLayerOutputSize,
+              kOutputLayerOutputSize,
+              kOutputDenseBias,
+              kOutputDenseWeights,
+              ActivationFunction::kSigmoidApproximated,
+              // The output layer is just 24x1. The unoptimized code is faster.
+              NoAvailableCpuFeatures(),
+              /*layer_name=*/"FC2") {
+  // Input-output chaining size checks.
+  RTC_DCHECK_EQ(input_.size(), hidden_.input_size())
+      << "The input and the hidden layers sizes do not match.";
+  RTC_DCHECK_EQ(hidden_.size(), output_.input_size())
+      << "The hidden and the output layers sizes do not match.";
+}
+
+RnnVad::~RnnVad() = default;
+
+void RnnVad::Reset() {
+  hidden_.Reset();
+}
+
+float RnnVad::ComputeVadProbability(
+    rtc::ArrayView<const float, kFeatureVectorSize> feature_vector,
+    bool is_silence) {
+  if (is_silence) {
+    Reset();
+    return 0.f;
+  }
+  input_.ComputeOutput(feature_vector);
+  hidden_.ComputeOutput(input_);
+  output_.ComputeOutput(hidden_);
+  RTC_DCHECK_EQ(output_.size(), 1);
+  return output_.data()[0];
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.h
new file mode 100644
index 0000000000..3148f1b3ff
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
+
+#include <stddef.h>
+#include <sys/types.h>
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn_fc.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn_gru.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Recurrent network with hard-coded architecture and weights for voice activity
+// detection.
+class RnnVad {
+ public:
+  explicit RnnVad(const AvailableCpuFeatures& cpu_features);
+  RnnVad(const RnnVad&) = delete;
+  RnnVad& operator=(const RnnVad&) = delete;
+  ~RnnVad();
+  void Reset();
+  // Observes `feature_vector` and `is_silence`, updates the RNN and returns the
+  // current voice probability.
+  float ComputeVadProbability(
+      rtc::ArrayView<const float, kFeatureVectorSize> feature_vector,
+      bool is_silence);
+
+ private:
+  FullyConnectedLayer input_;
+  GatedRecurrentLayer hidden_;
+  FullyConnectedLayer output_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
new file mode 100644
index 0000000000..91501fb6e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <numeric>
+
+#include "modules/audio_processing/agc2/rnn_vad/rnn_fc.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "third_party/rnnoise/src/rnn_activations.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+std::vector<float> GetScaledParams(rtc::ArrayView<const int8_t> params) {
+  std::vector<float> scaled_params(params.size());
+  std::transform(params.begin(), params.end(), scaled_params.begin(),
+                 [](int8_t x) -> float {
+                   return ::rnnoise::kWeightsScale * static_cast<float>(x);
+                 });
+  return scaled_params;
+}
+
+// TODO(bugs.chromium.org/10480): Hard-code optimized layout and remove this
+// function to improve setup time.
+// Casts and scales `weights` and re-arranges the layout.
+std::vector<float> PreprocessWeights(rtc::ArrayView<const int8_t> weights,
+                                     int output_size) {
+  if (output_size == 1) {
+    return GetScaledParams(weights);
+  }
+  // Transpose, scale and cast.
+  const int input_size = rtc::CheckedDivExact(
+      rtc::dchecked_cast<int>(weights.size()), output_size);
+  std::vector<float> w(weights.size());
+  for (int o = 0; o < output_size; ++o) {
+    for (int i = 0; i < input_size; ++i) {
+      w[o * input_size + i] = rnnoise::kWeightsScale *
+                              static_cast<float>(weights[i * output_size + o]);
+    }
+  }
+  return w;
+}
+
+rtc::FunctionView<float(float)> GetActivationFunction(
+    ActivationFunction activation_function) {
+  switch (activation_function) {
+    case ActivationFunction::kTansigApproximated:
+      return ::rnnoise::TansigApproximated;
+    case ActivationFunction::kSigmoidApproximated:
+      return ::rnnoise::SigmoidApproximated;
+  }
+}
+
+}  // namespace
+
+FullyConnectedLayer::FullyConnectedLayer(
+    const int input_size,
+    const int output_size,
+    const rtc::ArrayView<const int8_t> bias,
+    const rtc::ArrayView<const int8_t> weights,
+    ActivationFunction activation_function,
+    const AvailableCpuFeatures& cpu_features,
+    absl::string_view layer_name)
+    : input_size_(input_size),
+      output_size_(output_size),
+      bias_(GetScaledParams(bias)),
+      weights_(PreprocessWeights(weights, output_size)),
+      vector_math_(cpu_features),
+      activation_function_(GetActivationFunction(activation_function)) {
+  RTC_DCHECK_LE(output_size_, kFullyConnectedLayerMaxUnits)
+      << "Insufficient FC layer over-allocation (" << layer_name << ").";
+  RTC_DCHECK_EQ(output_size_, bias_.size())
+      << "Mismatching output size and bias terms array size (" << layer_name
+      << ").";
+  RTC_DCHECK_EQ(input_size_ * output_size_, weights_.size())
+      << "Mismatching input-output size and weight coefficients array size ("
+      << layer_name << ").";
+}
+
+FullyConnectedLayer::~FullyConnectedLayer() = default;
+
+void FullyConnectedLayer::ComputeOutput(rtc::ArrayView<const float> input) {
+  RTC_DCHECK_EQ(input.size(), input_size_);
+  rtc::ArrayView<const float> weights(weights_);
+  for (int o = 0; o < output_size_; ++o) {
+    output_[o] = activation_function_(
+        bias_[o] + vector_math_.DotProduct(
+                       input, weights.subview(o * input_size_, input_size_)));
+  }
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
new file mode 100644
index 0000000000..d23957a6f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_FC_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_FC_H_
+
+#include <array>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/function_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Activation function for a neural network cell.
+enum class ActivationFunction { kTansigApproximated, kSigmoidApproximated };
+
+// Maximum number of units for an FC layer.
+constexpr int kFullyConnectedLayerMaxUnits = 24;
+
+// Fully-connected layer with a custom activation function which owns the output
+// buffer.
+class FullyConnectedLayer {
+ public:
+  // Ctor. `output_size` cannot be greater than `kFullyConnectedLayerMaxUnits`.
+  FullyConnectedLayer(int input_size,
+                      int output_size,
+                      rtc::ArrayView<const int8_t> bias,
+                      rtc::ArrayView<const int8_t> weights,
+                      ActivationFunction activation_function,
+                      const AvailableCpuFeatures& cpu_features,
+                      absl::string_view layer_name);
+  FullyConnectedLayer(const FullyConnectedLayer&) = delete;
+  FullyConnectedLayer& operator=(const FullyConnectedLayer&) = delete;
+  ~FullyConnectedLayer();
+
+  // Returns the size of the input vector.
+  int input_size() const { return input_size_; }
+  // Returns the pointer to the first element of the output buffer.
+  const float* data() const { return output_.data(); }
+  // Returns the size of the output buffer.
+  int size() const { return output_size_; }
+
+  // Computes the fully-connected layer output.
+  void ComputeOutput(rtc::ArrayView<const float> input);
+
+ private:
+  const int input_size_;
+  const int output_size_;
+  const std::vector<float> bias_;
+  const std::vector<float> weights_;
+  const VectorMath vector_math_;
+  rtc::FunctionView<float(float)> activation_function_;
+  // Over-allocated array with size equal to `output_size_`.
+  std::array<float, kFullyConnectedLayerMaxUnits> output_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_FC_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
new file mode 100644
index 0000000000..ff9bb18bc2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc_unittest.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/rnn_fc.h"
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "modules/audio_processing/test/performance_timer.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/system/arch.h"
+#include "test/gtest.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+using ::rnnoise::kInputDenseBias;
+using ::rnnoise::kInputDenseWeights;
+using ::rnnoise::kInputLayerInputSize;
+using ::rnnoise::kInputLayerOutputSize;
+
+// Fully connected layer test data.
+constexpr std::array<float, 42> kFullyConnectedInputVector = {
+    -1.00131f,   -0.627069f, -7.81097f,  7.86285f,    -2.87145f,  3.32365f,
+    -0.653161f,  0.529839f,  -0.425307f, 0.25583f,    0.235094f,  0.230527f,
+    -0.144687f,  0.182785f,  0.57102f,   0.125039f,   0.479482f,  -0.0255439f,
+    -0.0073141f, -0.147346f, -0.217106f, -0.0846906f, -8.34943f,  3.09065f,
+    1.42628f,    -0.85235f,  -0.220207f, -0.811163f,  2.09032f,   -2.01425f,
+    -0.690268f,  -0.925327f, -0.541354f, 0.58455f,    -0.606726f, -0.0372358f,
+    0.565991f,   0.435854f,  0.420812f,  0.162198f,   -2.13f,     10.0089f};
+constexpr std::array<float, 24> kFullyConnectedExpectedOutput = {
+    -0.623293f, -0.988299f, 0.999378f,  0.967168f,  0.103087f,  -0.978545f,
+    -0.856347f, 0.346675f,  1.f,        -0.717442f, -0.544176f, 0.960363f,
+    0.983443f,  0.999991f,  -0.824335f, 0.984742f,  0.990208f,  0.938179f,
+    0.875092f,  0.999846f,  0.997707f,  -0.999382f, 0.973153f,  -0.966605f};
+
+class RnnFcParametrization
+    : public ::testing::TestWithParam<AvailableCpuFeatures> {};
+
+// Checks that the output of a fully connected layer is within tolerance given
+// test input data.
+TEST_P(RnnFcParametrization, CheckFullyConnectedLayerOutput) {
+  FullyConnectedLayer fc(kInputLayerInputSize, kInputLayerOutputSize,
+                         kInputDenseBias, kInputDenseWeights,
+                         ActivationFunction::kTansigApproximated,
+                         /*cpu_features=*/GetParam(),
+                         /*layer_name=*/"FC");
+  fc.ComputeOutput(kFullyConnectedInputVector);
+  ExpectNearAbsolute(kFullyConnectedExpectedOutput, fc, 1e-5f);
+}
+
+TEST_P(RnnFcParametrization, DISABLED_BenchmarkFullyConnectedLayer) {
+  const AvailableCpuFeatures cpu_features = GetParam();
+  FullyConnectedLayer fc(kInputLayerInputSize, kInputLayerOutputSize,
+                         kInputDenseBias, kInputDenseWeights,
+                         ActivationFunction::kTansigApproximated, cpu_features,
+                         /*layer_name=*/"FC");
+
+  constexpr int kNumTests = 10000;
+  ::webrtc::test::PerformanceTimer perf_timer(kNumTests);
+  for (int k = 0; k < kNumTests; ++k) {
+    perf_timer.StartTimer();
+    fc.ComputeOutput(kFullyConnectedInputVector);
+    perf_timer.StopTimer();
+  }
+  RTC_LOG(LS_INFO) << "CPU features: " << cpu_features.ToString() << " | "
+                   << (perf_timer.GetDurationAverage() / 1000) << " +/- "
+                   << (perf_timer.GetDurationStandardDeviation() / 1000)
+                   << " ms";
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back(NoAvailableCpuFeatures());
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  if (available.avx2) {
+    v.push_back({/*sse2=*/false, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.neon) {
+    v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/true});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    RnnFcParametrization,
+    ::testing::ValuesIn(GetCpuFeaturesToTest()),
+    [](const ::testing::TestParamInfo<AvailableCpuFeatures>& info) {
+      return info.param.ToString();
+    });
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
new file mode 100644
index 0000000000..ef37410caa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc
@@ -0,0 +1,198 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/rnn_gru.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "third_party/rnnoise/src/rnn_activations.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kNumGruGates = 3;  // Update, reset, output.
+
+std::vector<float> PreprocessGruTensor(rtc::ArrayView<const int8_t> tensor_src,
+                                       int output_size) {
+  // Transpose, cast and scale.
+  // `n` is the size of the first dimension of the 3-dim tensor `weights`.
+  const int n = rtc::CheckedDivExact(rtc::dchecked_cast<int>(tensor_src.size()),
+                                     output_size * kNumGruGates);
+  const int stride_src = kNumGruGates * output_size;
+  const int stride_dst = n * output_size;
+  std::vector<float> tensor_dst(tensor_src.size());
+  for (int g = 0; g < kNumGruGates; ++g) {
+    for (int o = 0; o < output_size; ++o) {
+      for (int i = 0; i < n; ++i) {
+        tensor_dst[g * stride_dst + o * n + i] =
+            ::rnnoise::kWeightsScale *
+            static_cast<float>(
+                tensor_src[i * stride_src + g * output_size + o]);
+      }
+    }
+  }
+  return tensor_dst;
+}
+
+// Computes the output for the update or the reset gate.
+// Operation: `g = sigmoid(W^T∙i + R^T∙s + b)` where
+// - `g`: output gate vector
+// - `W`: weights matrix
+// - `i`: input vector
+// - `R`: recurrent weights matrix
+// - `s`: state gate vector
+// - `b`: bias vector
+void ComputeUpdateResetGate(int input_size,
+                            int output_size,
+                            const VectorMath& vector_math,
+                            rtc::ArrayView<const float> input,
+                            rtc::ArrayView<const float> state,
+                            rtc::ArrayView<const float> bias,
+                            rtc::ArrayView<const float> weights,
+                            rtc::ArrayView<const float> recurrent_weights,
+                            rtc::ArrayView<float> gate) {
+  RTC_DCHECK_EQ(input.size(), input_size);
+  RTC_DCHECK_EQ(state.size(), output_size);
+  RTC_DCHECK_EQ(bias.size(), output_size);
+  RTC_DCHECK_EQ(weights.size(), input_size * output_size);
+  RTC_DCHECK_EQ(recurrent_weights.size(), output_size * output_size);
+  RTC_DCHECK_GE(gate.size(), output_size);  // `gate` is over-allocated.
+  for (int o = 0; o < output_size; ++o) {
+    float x = bias[o];
+    x += vector_math.DotProduct(input,
+                                weights.subview(o * input_size, input_size));
+    x += vector_math.DotProduct(
+        state, recurrent_weights.subview(o * output_size, output_size));
+    gate[o] = ::rnnoise::SigmoidApproximated(x);
+  }
+}
+
+// Computes the output for the state gate.
+// Operation: `s' = u .* s + (1 - u) .* ReLU(W^T∙i + R^T∙(s .* r) + b)` where
+// - `s'`: output state gate vector
+// - `s`: previous state gate vector
+// - `u`: update gate vector
+// - `W`: weights matrix
+// - `i`: input vector
+// - `R`: recurrent weights matrix
+// - `r`: reset gate vector
+// - `b`: bias vector
+// - `.*` element-wise product
+void ComputeStateGate(int input_size,
+                      int output_size,
+                      const VectorMath& vector_math,
+                      rtc::ArrayView<const float> input,
+                      rtc::ArrayView<const float> update,
+                      rtc::ArrayView<const float> reset,
+                      rtc::ArrayView<const float> bias,
+                      rtc::ArrayView<const float> weights,
+                      rtc::ArrayView<const float> recurrent_weights,
+                      rtc::ArrayView<float> state) {
+  RTC_DCHECK_EQ(input.size(), input_size);
+  RTC_DCHECK_GE(update.size(), output_size);  // `update` is over-allocated.
+  RTC_DCHECK_GE(reset.size(), output_size);   // `reset` is over-allocated.
+  RTC_DCHECK_EQ(bias.size(), output_size);
+  RTC_DCHECK_EQ(weights.size(), input_size * output_size);
+  RTC_DCHECK_EQ(recurrent_weights.size(), output_size * output_size);
+  RTC_DCHECK_EQ(state.size(), output_size);
+  std::array<float, kGruLayerMaxUnits> reset_x_state;
+  for (int o = 0; o < output_size; ++o) {
+    reset_x_state[o] = state[o] * reset[o];
+  }
+  for (int o = 0; o < output_size; ++o) {
+    float x = bias[o];
+    x += vector_math.DotProduct(input,
+                                weights.subview(o * input_size, input_size));
+    x += vector_math.DotProduct(
+        {reset_x_state.data(), static_cast<size_t>(output_size)},
+        recurrent_weights.subview(o * output_size, output_size));
+    state[o] = update[o] * state[o] + (1.f - update[o]) * std::max(0.f, x);
+  }
+}
+
+}  // namespace
+
+GatedRecurrentLayer::GatedRecurrentLayer(
+    const int input_size,
+    const int output_size,
+    const rtc::ArrayView<const int8_t> bias,
+    const rtc::ArrayView<const int8_t> weights,
+    const rtc::ArrayView<const int8_t> recurrent_weights,
+    const AvailableCpuFeatures& cpu_features,
+    absl::string_view layer_name)
+    : input_size_(input_size),
+      output_size_(output_size),
+      bias_(PreprocessGruTensor(bias, output_size)),
+      weights_(PreprocessGruTensor(weights, output_size)),
+      recurrent_weights_(PreprocessGruTensor(recurrent_weights, output_size)),
+      vector_math_(cpu_features) {
+  RTC_DCHECK_LE(output_size_, kGruLayerMaxUnits)
+      << "Insufficient GRU layer over-allocation (" << layer_name << ").";
+  RTC_DCHECK_EQ(kNumGruGates * output_size_, bias_.size())
+      << "Mismatching output size and bias terms array size (" << layer_name
+      << ").";
+  RTC_DCHECK_EQ(kNumGruGates * input_size_ * output_size_, weights_.size())
+      << "Mismatching input-output size and weight coefficients array size ("
+      << layer_name << ").";
+  RTC_DCHECK_EQ(kNumGruGates * output_size_ * output_size_,
+                recurrent_weights_.size())
+      << "Mismatching input-output size and recurrent weight coefficients array"
+         " size ("
+      << layer_name << ").";
+  Reset();
+}
+
+GatedRecurrentLayer::~GatedRecurrentLayer() = default;
+
+void GatedRecurrentLayer::Reset() {
+  state_.fill(0.f);
+}
+
+void GatedRecurrentLayer::ComputeOutput(rtc::ArrayView<const float> input) {
+  RTC_DCHECK_EQ(input.size(), input_size_);
+
+  // The tensors below are organized as a sequence of flattened tensors for the
+  // `update`, `reset` and `state` gates.
+  rtc::ArrayView<const float> bias(bias_);
+  rtc::ArrayView<const float> weights(weights_);
+  rtc::ArrayView<const float> recurrent_weights(recurrent_weights_);
+  // Strides to access to the flattened tensors for a specific gate.
+  const int stride_weights = input_size_ * output_size_;
+  const int stride_recurrent_weights = output_size_ * output_size_;
+
+  rtc::ArrayView<float> state(state_.data(), output_size_);
+
+  // Update gate.
+  std::array<float, kGruLayerMaxUnits> update;
+  ComputeUpdateResetGate(
+      input_size_, output_size_, vector_math_, input, state,
+      bias.subview(0, output_size_), weights.subview(0, stride_weights),
+      recurrent_weights.subview(0, stride_recurrent_weights), update);
+  // Reset gate.
+  std::array<float, kGruLayerMaxUnits> reset;
+  ComputeUpdateResetGate(input_size_, output_size_, vector_math_, input, state,
+                         bias.subview(output_size_, output_size_),
+                         weights.subview(stride_weights, stride_weights),
+                         recurrent_weights.subview(stride_recurrent_weights,
+                                                   stride_recurrent_weights),
+                         reset);
+  // State gate.
+  ComputeStateGate(input_size_, output_size_, vector_math_, input, update,
+                   reset, bias.subview(2 * output_size_, output_size_),
+                   weights.subview(2 * stride_weights, stride_weights),
+                   recurrent_weights.subview(2 * stride_recurrent_weights,
+                                             stride_recurrent_weights),
+                   state);
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
new file mode 100644
index 0000000000..3407dfcdf1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_GRU_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_GRU_H_
+
+#include <array>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Maximum number of units for a GRU layer.
+constexpr int kGruLayerMaxUnits = 24;
+
+// Recurrent layer with gated recurrent units (GRUs) with sigmoid and ReLU as
+// activation functions for the update/reset and output gates respectively.
+class GatedRecurrentLayer {
+ public:
+  // Ctor. `output_size` cannot be greater than `kGruLayerMaxUnits`.
+  GatedRecurrentLayer(int input_size,
+                      int output_size,
+                      rtc::ArrayView<const int8_t> bias,
+                      rtc::ArrayView<const int8_t> weights,
+                      rtc::ArrayView<const int8_t> recurrent_weights,
+                      const AvailableCpuFeatures& cpu_features,
+                      absl::string_view layer_name);
+  GatedRecurrentLayer(const GatedRecurrentLayer&) = delete;
+  GatedRecurrentLayer& operator=(const GatedRecurrentLayer&) = delete;
+  ~GatedRecurrentLayer();
+
+  // Returns the size of the input vector.
+  int input_size() const { return input_size_; }
+  // Returns the pointer to the first element of the output buffer.
+  const float* data() const { return state_.data(); }
+  // Returns the size of the output buffer.
+  int size() const { return output_size_; }
+
+  // Resets the GRU state.
+  void Reset();
+  // Computes the recurrent layer output and updates the status.
+  void ComputeOutput(rtc::ArrayView<const float> input);
+
+ private:
+  const int input_size_;
+  const int output_size_;
+  const std::vector<float> bias_;
+  const std::vector<float> weights_;
+  const std::vector<float> recurrent_weights_;
+  const VectorMath vector_math_;
+  // Over-allocated array with size equal to `output_size_`.
+  std::array<float, kGruLayerMaxUnits> state_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_GRU_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
new file mode 100644
index 0000000000..88ae72803a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru_unittest.cc
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/rnn_gru.h"
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "modules/audio_processing/test/performance_timer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/gtest.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+void TestGatedRecurrentLayer(
+    GatedRecurrentLayer& gru,
+    rtc::ArrayView<const float> input_sequence,
+    rtc::ArrayView<const float> expected_output_sequence) {
+  const int input_sequence_length = rtc::CheckedDivExact(
+      rtc::dchecked_cast<int>(input_sequence.size()), gru.input_size());
+  const int output_sequence_length = rtc::CheckedDivExact(
+      rtc::dchecked_cast<int>(expected_output_sequence.size()), gru.size());
+  ASSERT_EQ(input_sequence_length, output_sequence_length)
+      << "The test data length is invalid.";
+  // Feed the GRU layer and check the output at every step.
+  gru.Reset();
+  for (int i = 0; i < input_sequence_length; ++i) {
+    SCOPED_TRACE(i);
+    gru.ComputeOutput(
+        input_sequence.subview(i * gru.input_size(), gru.input_size()));
+    const auto expected_output =
+        expected_output_sequence.subview(i * gru.size(), gru.size());
+    ExpectNearAbsolute(expected_output, gru, 3e-6f);
+  }
+}
+
+// Gated recurrent units layer test data.
+constexpr int kGruInputSize = 5;
+constexpr int kGruOutputSize = 4;
+constexpr std::array<int8_t, 12> kGruBias = {96,   -99, -81, -114, 49,  119,
+                                             -118, 68,  -76, 91,   121, 125};
+constexpr std::array<int8_t, 60> kGruWeights = {
+    // Input 0.
+    124, 9, 1, 116,        // Update.
+    -66, -21, -118, -110,  // Reset.
+    104, 75, -23, -51,     // Output.
+    // Input 1.
+    -72, -111, 47, 93,   // Update.
+    77, -98, 41, -8,     // Reset.
+    40, -23, -43, -107,  // Output.
+    // Input 2.
+    9, -73, 30, -32,      // Update.
+    -2, 64, -26, 91,      // Reset.
+    -48, -24, -28, -104,  // Output.
+    // Input 3.
+    74, -46, 116, 15,    // Update.
+    32, 52, -126, -38,   // Reset.
+    -121, 12, -16, 110,  // Output.
+    // Input 4.
+    -95, 66, -103, -35,  // Update.
+    -38, 3, -126, -61,   // Reset.
+    28, 98, -117, -43    // Output.
+};
+constexpr std::array<int8_t, 48> kGruRecurrentWeights = {
+    // Output 0.
+    -3, 87, 50, 51,     // Update.
+    -22, 27, -39, 62,   // Reset.
+    31, -83, -52, -48,  // Output.
+    // Output 1.
+    -6, 83, -19, 104,  // Update.
+    105, 48, 23, 68,   // Reset.
+    23, 40, 7, -120,   // Output.
+    // Output 2.
+    64, -62, 117, 85,     // Update.
+    51, -43, 54, -105,    // Reset.
+    120, 56, -128, -107,  // Output.
+    // Output 3.
+    39, 50, -17, -47,   // Update.
+    -117, 14, 108, 12,  // Reset.
+    -7, -72, 103, -87,  // Output.
+};
+constexpr std::array<float, 20> kGruInputSequence = {
+    0.89395463f, 0.93224651f, 0.55788344f, 0.32341808f, 0.93355054f,
+    0.13475326f, 0.97370994f, 0.14253306f, 0.93710381f, 0.76093364f,
+    0.65780413f, 0.41657975f, 0.49403164f, 0.46843281f, 0.75138855f,
+    0.24517593f, 0.47657707f, 0.57064998f, 0.435184f,   0.19319285f};
+constexpr std::array<float, 16> kGruExpectedOutputSequence = {
+    0.0239123f,  0.5773077f,  0.f,         0.f,
+    0.01282811f, 0.64330572f, 0.f,         0.04863098f,
+    0.00781069f, 0.75267816f, 0.f,         0.02579715f,
+    0.00471378f, 0.59162533f, 0.11087593f, 0.01334511f};
+
+class RnnGruParametrization
+    : public ::testing::TestWithParam<AvailableCpuFeatures> {};
+
+// Checks that the output of a GRU layer is within tolerance given test input
+// data.
+TEST_P(RnnGruParametrization, CheckGatedRecurrentLayer) {
+  GatedRecurrentLayer gru(kGruInputSize, kGruOutputSize, kGruBias, kGruWeights,
+                          kGruRecurrentWeights,
+                          /*cpu_features=*/GetParam(),
+                          /*layer_name=*/"GRU");
+  TestGatedRecurrentLayer(gru, kGruInputSequence, kGruExpectedOutputSequence);
+}
+
+TEST_P(RnnGruParametrization, DISABLED_BenchmarkGatedRecurrentLayer) {
+  // Prefetch test data.
+  std::unique_ptr<FileReader> reader = CreateGruInputReader();
+  std::vector<float> gru_input_sequence(reader->size());
+  reader->ReadChunk(gru_input_sequence);
+
+  using ::rnnoise::kHiddenGruBias;
+  using ::rnnoise::kHiddenGruRecurrentWeights;
+  using ::rnnoise::kHiddenGruWeights;
+  using ::rnnoise::kHiddenLayerOutputSize;
+  using ::rnnoise::kInputLayerOutputSize;
+
+  GatedRecurrentLayer gru(kInputLayerOutputSize, kHiddenLayerOutputSize,
+                          kHiddenGruBias, kHiddenGruWeights,
+                          kHiddenGruRecurrentWeights,
+                          /*cpu_features=*/GetParam(),
+                          /*layer_name=*/"GRU");
+
+  rtc::ArrayView<const float> input_sequence(gru_input_sequence);
+  ASSERT_EQ(input_sequence.size() % kInputLayerOutputSize,
+            static_cast<size_t>(0));
+  const int input_sequence_length =
+      input_sequence.size() / kInputLayerOutputSize;
+
+  constexpr int kNumTests = 100;
+  ::webrtc::test::PerformanceTimer perf_timer(kNumTests);
+  for (int k = 0; k < kNumTests; ++k) {
+    perf_timer.StartTimer();
+    for (int i = 0; i < input_sequence_length; ++i) {
+      gru.ComputeOutput(
+          input_sequence.subview(i * gru.input_size(), gru.input_size()));
+    }
+    perf_timer.StopTimer();
+  }
+  RTC_LOG(LS_INFO) << (perf_timer.GetDurationAverage() / 1000) << " +/- "
+                   << (perf_timer.GetDurationStandardDeviation() / 1000)
+                   << " ms";
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back(NoAvailableCpuFeatures());
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  if (available.avx2) {
+    v.push_back({/*sse2=*/false, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.neon) {
+    v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/true});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    RnnGruParametrization,
+    ::testing::ValuesIn(GetCpuFeaturesToTest()),
+    [](const ::testing::TestParamInfo<AvailableCpuFeatures>& info) {
+      return info.param.ToString();
+    });
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
new file mode 100644
index 0000000000..4c5409a14e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr std::array<float, kFeatureVectorSize> kFeatures = {
+    -1.00131f,   -0.627069f, -7.81097f,  7.86285f,    -2.87145f,  3.32365f,
+    -0.653161f,  0.529839f,  -0.425307f, 0.25583f,    0.235094f,  0.230527f,
+    -0.144687f,  0.182785f,  0.57102f,   0.125039f,   0.479482f,  -0.0255439f,
+    -0.0073141f, -0.147346f, -0.217106f, -0.0846906f, -8.34943f,  3.09065f,
+    1.42628f,    -0.85235f,  -0.220207f, -0.811163f,  2.09032f,   -2.01425f,
+    -0.690268f,  -0.925327f, -0.541354f, 0.58455f,    -0.606726f, -0.0372358f,
+    0.565991f,   0.435854f,  0.420812f,  0.162198f,   -2.13f,     10.0089f};
+
+void WarmUpRnnVad(RnnVad& rnn_vad) {
+  for (int i = 0; i < 10; ++i) {
+    rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  }
+}
+
+// Checks that the speech probability is zero with silence.
+TEST(RnnVadTest, CheckZeroProbabilityWithSilence) {
+  RnnVad rnn_vad(GetAvailableCpuFeatures());
+  WarmUpRnnVad(rnn_vad);
+  EXPECT_EQ(rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/true), 0.f);
+}
+
+// Checks that the same output is produced after reset given the same input
+// sequence.
+TEST(RnnVadTest, CheckRnnVadReset) {
+  RnnVad rnn_vad(GetAvailableCpuFeatures());
+  WarmUpRnnVad(rnn_vad);
+  float pre = rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  rnn_vad.Reset();
+  WarmUpRnnVad(rnn_vad);
+  float post = rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  EXPECT_EQ(pre, post);
+}
+
+// Checks that the same output is produced after silence is observed given the
+// same input sequence.
+TEST(RnnVadTest, CheckRnnVadSilence) {
+  RnnVad rnn_vad(GetAvailableCpuFeatures());
+  WarmUpRnnVad(rnn_vad);
+  float pre = rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/true);
+  WarmUpRnnVad(rnn_vad);
+  float post = rnn_vad.ComputeVadProbability(kFeatures, /*is_silence=*/false);
+  EXPECT_EQ(pre, post);
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build
new file mode 100644
index 0000000000..3f00e43e7c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_auto_correlation_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/auto_correlation.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_auto_correlation_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build
new file mode 100644
index 0000000000..64d3371d81
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_common_gn/moz.build
@@ -0,0 +1,216 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_common_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build
new file mode 100644
index 0000000000..82e5302312
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build
new file mode 100644
index 0000000000..7dd4619965
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_layers_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_gru.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_layers_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build
new file mode 100644
index 0000000000..1fcbbbe408
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_lp_residual_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_lp_residual_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build
new file mode 100644
index 0000000000..6a69de9e29
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_pitch_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_pitch_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build
new file mode 100644
index 0000000000..d2a4eb7261
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_ring_buffer_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_ring_buffer_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build
new file mode 100644
index 0000000000..c6913470c3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_sequence_buffer_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_sequence_buffer_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build
new file mode 100644
index 0000000000..8298c7e091
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_spectral_features_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_spectral_features_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build
new file mode 100644
index 0000000000..acb9330454
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_symmetric_matrix_buffer_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rnn_vad_symmetric_matrix_buffer_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
new file mode 100644
index 0000000000..a0e1242eb4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <array>
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+ABSL_FLAG(std::string, i, "", "Path to the input wav file");
+ABSL_FLAG(std::string, f, "", "Path to the output features file");
+ABSL_FLAG(std::string, o, "", "Path to the output VAD probabilities file");
+
+namespace webrtc {
+namespace rnn_vad {
+namespace test {
+
+int main(int argc, char* argv[]) {
+  absl::ParseCommandLine(argc, argv);
+  rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+
+  // Open wav input file and check properties.
+  const std::string input_wav_file = absl::GetFlag(FLAGS_i);
+  WavReader wav_reader(input_wav_file);
+  if (wav_reader.num_channels() != 1) {
+    RTC_LOG(LS_ERROR) << "Only mono wav files are supported";
+    return 1;
+  }
+  if (wav_reader.sample_rate() % 100 != 0) {
+    RTC_LOG(LS_ERROR) << "The sample rate rate must allow 10 ms frames.";
+    return 1;
+  }
+  RTC_LOG(LS_INFO) << "Input sample rate: " << wav_reader.sample_rate();
+
+  // Init output files.
+  const std::string output_vad_probs_file = absl::GetFlag(FLAGS_o);
+  FILE* vad_probs_file = fopen(output_vad_probs_file.c_str(), "wb");
+  FILE* features_file = nullptr;
+  const std::string output_feature_file = absl::GetFlag(FLAGS_f);
+  if (!output_feature_file.empty()) {
+    features_file = fopen(output_feature_file.c_str(), "wb");
+  }
+
+  // Initialize.
+  const int frame_size_10ms =
+      rtc::CheckedDivExact(wav_reader.sample_rate(), 100);
+  std::vector<float> samples_10ms;
+  samples_10ms.resize(frame_size_10ms);
+  std::array<float, kFrameSize10ms24kHz> samples_10ms_24kHz;
+  PushSincResampler resampler(frame_size_10ms, kFrameSize10ms24kHz);
+  const AvailableCpuFeatures cpu_features = GetAvailableCpuFeatures();
+  FeaturesExtractor features_extractor(cpu_features);
+  std::array<float, kFeatureVectorSize> feature_vector;
+  RnnVad rnn_vad(cpu_features);
+
+  // Compute VAD probabilities.
+  while (true) {
+    // Read frame at the input sample rate.
+    const size_t read_samples =
+        wav_reader.ReadSamples(frame_size_10ms, samples_10ms.data());
+    if (rtc::SafeLt(read_samples, frame_size_10ms)) {
+      break;  // EOF.
+    }
+    // Resample input.
+    resampler.Resample(samples_10ms.data(), samples_10ms.size(),
+                       samples_10ms_24kHz.data(), samples_10ms_24kHz.size());
+
+    // Extract features and feed the RNN.
+    bool is_silence = features_extractor.CheckSilenceComputeFeatures(
+        samples_10ms_24kHz, feature_vector);
+    float vad_probability =
+        rnn_vad.ComputeVadProbability(feature_vector, is_silence);
+    // Write voice probability.
+    RTC_DCHECK_GE(vad_probability, 0.f);
+    RTC_DCHECK_GE(1.f, vad_probability);
+    fwrite(&vad_probability, sizeof(float), 1, vad_probs_file);
+    // Write features.
+    if (features_file) {
+      const float float_is_silence = is_silence ? 1.f : 0.f;
+      fwrite(&float_is_silence, sizeof(float), 1, features_file);
+      if (is_silence) {
+        // Do not write uninitialized values.
+        feature_vector.fill(0.f);
+      }
+      fwrite(feature_vector.data(), sizeof(float), kFeatureVectorSize,
+             features_file);
+    }
+  }
+
+  // Close output file(s).
+  fclose(vad_probs_file);
+  RTC_LOG(LS_INFO) << "VAD probabilities written to " << output_vad_probs_file;
+  if (features_file) {
+    fclose(features_file);
+    RTC_LOG(LS_INFO) << "features written to " << output_feature_file;
+  }
+
+  return 0;
+}
+
+}  // namespace test
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::rnn_vad::test::main(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
new file mode 100644
index 0000000000..f33cd14a8a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <array>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "modules/audio_processing/test/performance_timer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/gtest.h"
+#include "third_party/rnnoise/src/rnn_activations.h"
+#include "third_party/rnnoise/src/rnn_vad_weights.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kFrameSize10ms48kHz = 480;
+
+void DumpPerfStats(int num_samples,
+                   int sample_rate,
+                   double average_us,
+                   double standard_deviation) {
+  float audio_track_length_ms =
+      1e3f * static_cast<float>(num_samples) / static_cast<float>(sample_rate);
+  float average_ms = static_cast<float>(average_us) / 1e3f;
+  float speed = audio_track_length_ms / average_ms;
+  RTC_LOG(LS_INFO) << "track duration (ms): " << audio_track_length_ms;
+  RTC_LOG(LS_INFO) << "average processing time (ms): " << average_ms << " +/- "
+                   << (standard_deviation / 1e3);
+  RTC_LOG(LS_INFO) << "speed: " << speed << "x";
+}
+
+// When the RNN VAD model is updated and the expected output changes, set the
+// constant below to true in order to write new expected output binary files.
+constexpr bool kWriteComputedOutputToFile = false;
+
+// Avoids that one forgets to set `kWriteComputedOutputToFile` back to false
+// when the expected output files are re-exported.
+TEST(RnnVadTest, CheckWriteComputedOutputIsFalse) {
+  ASSERT_FALSE(kWriteComputedOutputToFile)
+      << "Cannot land if kWriteComputedOutput is true.";
+}
+
+class RnnVadProbabilityParametrization
+    : public ::testing::TestWithParam<AvailableCpuFeatures> {};
+
+// Checks that the computed VAD probability for a test input sequence sampled at
+// 48 kHz is within tolerance.
+TEST_P(RnnVadProbabilityParametrization, RnnVadProbabilityWithinTolerance) {
+  // Init resampler, feature extractor and RNN.
+  PushSincResampler decimator(kFrameSize10ms48kHz, kFrameSize10ms24kHz);
+  const AvailableCpuFeatures cpu_features = GetParam();
+  FeaturesExtractor features_extractor(cpu_features);
+  RnnVad rnn_vad(cpu_features);
+
+  // Init input samples and expected output readers.
+  std::unique_ptr<FileReader> samples_reader = CreatePcmSamplesReader();
+  std::unique_ptr<FileReader> expected_vad_prob_reader = CreateVadProbsReader();
+
+  // Input length. The last incomplete frame is ignored.
+  const int num_frames = samples_reader->size() / kFrameSize10ms48kHz;
+
+  // Init buffers.
+  std::vector<float> samples_48k(kFrameSize10ms48kHz);
+  std::vector<float> samples_24k(kFrameSize10ms24kHz);
+  std::vector<float> feature_vector(kFeatureVectorSize);
+  std::vector<float> computed_vad_prob(num_frames);
+  std::vector<float> expected_vad_prob(num_frames);
+
+  // Read expected output.
+  ASSERT_TRUE(expected_vad_prob_reader->ReadChunk(expected_vad_prob));
+
+  // Compute VAD probabilities on the downsampled input.
+  float cumulative_error = 0.f;
+  for (int i = 0; i < num_frames; ++i) {
+    ASSERT_TRUE(samples_reader->ReadChunk(samples_48k));
+    decimator.Resample(samples_48k.data(), samples_48k.size(),
+                       samples_24k.data(), samples_24k.size());
+    bool is_silence = features_extractor.CheckSilenceComputeFeatures(
+        {samples_24k.data(), kFrameSize10ms24kHz},
+        {feature_vector.data(), kFeatureVectorSize});
+    computed_vad_prob[i] = rnn_vad.ComputeVadProbability(
+        {feature_vector.data(), kFeatureVectorSize}, is_silence);
+    EXPECT_NEAR(computed_vad_prob[i], expected_vad_prob[i], 1e-3f);
+    cumulative_error += std::abs(computed_vad_prob[i] - expected_vad_prob[i]);
+  }
+  // Check average error.
+  EXPECT_LT(cumulative_error / num_frames, 1e-4f);
+
+  if (kWriteComputedOutputToFile) {
+    FileWriter vad_prob_writer("new_vad_prob.dat");
+    vad_prob_writer.WriteChunk(computed_vad_prob);
+  }
+}
+
+// Performance test for the RNN VAD (pre-fetching and downsampling are
+// excluded). Keep disabled and only enable locally to measure performance as
+// follows:
+// - on desktop: run the this unit test adding "--logs";
+// - on android: run the this unit test adding "--logcat-output-file".
+TEST_P(RnnVadProbabilityParametrization, DISABLED_RnnVadPerformance) {
+  // PCM samples reader and buffers.
+  std::unique_ptr<FileReader> samples_reader = CreatePcmSamplesReader();
+  // The last incomplete frame is ignored.
+  const int num_frames = samples_reader->size() / kFrameSize10ms48kHz;
+  std::array<float, kFrameSize10ms48kHz> samples;
+  // Pre-fetch and decimate samples.
+  PushSincResampler decimator(kFrameSize10ms48kHz, kFrameSize10ms24kHz);
+  std::vector<float> prefetched_decimated_samples;
+  prefetched_decimated_samples.resize(num_frames * kFrameSize10ms24kHz);
+  for (int i = 0; i < num_frames; ++i) {
+    ASSERT_TRUE(samples_reader->ReadChunk(samples));
+    decimator.Resample(samples.data(), samples.size(),
+                       &prefetched_decimated_samples[i * kFrameSize10ms24kHz],
+                       kFrameSize10ms24kHz);
+  }
+  // Initialize.
+  const AvailableCpuFeatures cpu_features = GetParam();
+  FeaturesExtractor features_extractor(cpu_features);
+  std::array<float, kFeatureVectorSize> feature_vector;
+  RnnVad rnn_vad(cpu_features);
+  constexpr int number_of_tests = 100;
+  ::webrtc::test::PerformanceTimer perf_timer(number_of_tests);
+  for (int k = 0; k < number_of_tests; ++k) {
+    features_extractor.Reset();
+    rnn_vad.Reset();
+    // Process frames.
+    perf_timer.StartTimer();
+    for (int i = 0; i < num_frames; ++i) {
+      bool is_silence = features_extractor.CheckSilenceComputeFeatures(
+          {&prefetched_decimated_samples[i * kFrameSize10ms24kHz],
+           kFrameSize10ms24kHz},
+          feature_vector);
+      rnn_vad.ComputeVadProbability(feature_vector, is_silence);
+    }
+    perf_timer.StopTimer();
+  }
+  DumpPerfStats(num_frames * kFrameSize10ms24kHz, kSampleRate24kHz,
+                perf_timer.GetDurationAverage(),
+                perf_timer.GetDurationStandardDeviation());
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back(NoAvailableCpuFeatures());
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.avx2 && available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  if (available.neon) {
+    v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/true});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    RnnVadProbabilityParametrization,
+    ::testing::ValuesIn(GetCpuFeaturesToTest()),
+    [](const ::testing::TestParamInfo<AvailableCpuFeatures>& info) {
+      return info.param.ToString();
+    });
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
new file mode 100644
index 0000000000..a7402788c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
+
+#include <algorithm>
+#include <cstring>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Linear buffer implementation to (i) push fixed size chunks of sequential data
+// and (ii) view contiguous parts of the buffer. The buffer and the pushed
+// chunks have size S and N respectively. For instance, when S = 2N the first
+// half of the sequence buffer is replaced with its second half, and the new N
+// values are written at the end of the buffer.
+// The class also provides a view on the most recent M values, where 0 < M <= S
+// and by default M = N.
+template <typename T, int S, int N, int M = N>
+class SequenceBuffer {
+  static_assert(N <= S,
+                "The new chunk size cannot be larger than the sequence buffer "
+                "size.");
+  static_assert(std::is_arithmetic<T>::value,
+                "Integral or floating point required.");
+
+ public:
+  SequenceBuffer() : buffer_(S) {
+    RTC_DCHECK_EQ(S, buffer_.size());
+    Reset();
+  }
+  SequenceBuffer(const SequenceBuffer&) = delete;
+  SequenceBuffer& operator=(const SequenceBuffer&) = delete;
+  ~SequenceBuffer() = default;
+  int size() const { return S; }
+  int chunks_size() const { return N; }
+  // Sets the sequence buffer values to zero.
+  void Reset() { std::fill(buffer_.begin(), buffer_.end(), 0); }
+  // Returns a view on the whole buffer.
+  rtc::ArrayView<const T, S> GetBufferView() const {
+    return {buffer_.data(), S};
+  }
+  // Returns a view on the M most recent values of the buffer.
+  rtc::ArrayView<const T, M> GetMostRecentValuesView() const {
+    static_assert(M <= S,
+                  "The number of most recent values cannot be larger than the "
+                  "sequence buffer size.");
+    return {buffer_.data() + S - M, M};
+  }
+  // Shifts left the buffer by N items and add new N items at the end.
+  void Push(rtc::ArrayView<const T, N> new_values) {
+    // Make space for the new values.
+    if (S > N)
+      std::memmove(buffer_.data(), buffer_.data() + N, (S - N) * sizeof(T));
+    // Copy the new values at the end of the buffer.
+    std::memcpy(buffer_.data() + S - N, new_values.data(), N * sizeof(T));
+  }
+
+ private:
+  std::vector<T> buffer_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
new file mode 100644
index 0000000000..af005833c1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/sequence_buffer.h"
+
+#include <algorithm>
+#include <array>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+template <typename T, int S, int N>
+void TestSequenceBufferPushOp() {
+  SCOPED_TRACE(S);
+  SCOPED_TRACE(N);
+  SequenceBuffer<T, S, N> seq_buf;
+  auto seq_buf_view = seq_buf.GetBufferView();
+  std::array<T, N> chunk;
+
+  // Check that a chunk is fully gone after ceil(S / N) push ops.
+  chunk.fill(1);
+  seq_buf.Push(chunk);
+  chunk.fill(0);
+  constexpr int required_push_ops = (S % N) ? S / N + 1 : S / N;
+  for (int i = 0; i < required_push_ops - 1; ++i) {
+    SCOPED_TRACE(i);
+    seq_buf.Push(chunk);
+    // Still in the buffer.
+    const auto* m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
+    EXPECT_EQ(1, *m);
+  }
+  // Gone after another push.
+  seq_buf.Push(chunk);
+  const auto* m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
+  EXPECT_EQ(0, *m);
+
+  // Check that the last item moves left by N positions after a push op.
+  if (S > N) {
+    // Fill in with non-zero values.
+    for (int i = 0; i < N; ++i)
+      chunk[i] = static_cast<T>(i + 1);
+    seq_buf.Push(chunk);
+    // With the next Push(), `last` will be moved left by N positions.
+    const T last = chunk[N - 1];
+    for (int i = 0; i < N; ++i)
+      chunk[i] = static_cast<T>(last + i + 1);
+    seq_buf.Push(chunk);
+    EXPECT_EQ(last, seq_buf_view[S - N - 1]);
+  }
+}
+
+TEST(RnnVadTest, SequenceBufferGetters) {
+  constexpr int buffer_size = 8;
+  constexpr int chunk_size = 8;
+  SequenceBuffer<int, buffer_size, chunk_size> seq_buf;
+  EXPECT_EQ(buffer_size, seq_buf.size());
+  EXPECT_EQ(chunk_size, seq_buf.chunks_size());
+  // Test view.
+  auto seq_buf_view = seq_buf.GetBufferView();
+  EXPECT_EQ(0, seq_buf_view[0]);
+  EXPECT_EQ(0, seq_buf_view[seq_buf_view.size() - 1]);
+  constexpr std::array<int, chunk_size> chunk = {10, 20, 30, 40,
+                                                 50, 60, 70, 80};
+  seq_buf.Push(chunk);
+  EXPECT_EQ(10, *seq_buf_view.begin());
+  EXPECT_EQ(80, *(seq_buf_view.end() - 1));
+}
+
+TEST(RnnVadTest, SequenceBufferPushOpsUnsigned) {
+  TestSequenceBufferPushOp<uint8_t, 32, 8>();   // Chunk size: 25%.
+  TestSequenceBufferPushOp<uint8_t, 32, 16>();  // Chunk size: 50%.
+  TestSequenceBufferPushOp<uint8_t, 32, 32>();  // Chunk size: 100%.
+  TestSequenceBufferPushOp<uint8_t, 23, 7>();   // Non-integer ratio.
+}
+
+TEST(RnnVadTest, SequenceBufferPushOpsSigned) {
+  TestSequenceBufferPushOp<int, 32, 8>();   // Chunk size: 25%.
+  TestSequenceBufferPushOp<int, 32, 16>();  // Chunk size: 50%.
+  TestSequenceBufferPushOp<int, 32, 32>();  // Chunk size: 100%.
+  TestSequenceBufferPushOp<int, 23, 7>();   // Non-integer ratio.
+}
+
+TEST(RnnVadTest, SequenceBufferPushOpsFloating) {
+  TestSequenceBufferPushOp<float, 32, 8>();   // Chunk size: 25%.
+  TestSequenceBufferPushOp<float, 32, 16>();  // Chunk size: 50%.
+  TestSequenceBufferPushOp<float, 32, 32>();  // Chunk size: 100%.
+  TestSequenceBufferPushOp<float, 23, 7>();   // Non-integer ratio.
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
new file mode 100644
index 0000000000..96086babb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
@@ -0,0 +1,214 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/spectral_features.h"
+
+#include <algorithm>
+#include <cmath>
+#include <limits>
+#include <numeric>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr float kSilenceThreshold = 0.04f;
+
+// Computes the new cepstral difference stats and pushes them into the passed
+// symmetric matrix buffer.
+void UpdateCepstralDifferenceStats(
+    rtc::ArrayView<const float, kNumBands> new_cepstral_coeffs,
+    const RingBuffer<float, kNumBands, kCepstralCoeffsHistorySize>& ring_buf,
+    SymmetricMatrixBuffer<float, kCepstralCoeffsHistorySize>* sym_matrix_buf) {
+  RTC_DCHECK(sym_matrix_buf);
+  // Compute the new cepstral distance stats.
+  std::array<float, kCepstralCoeffsHistorySize - 1> distances;
+  for (int i = 0; i < kCepstralCoeffsHistorySize - 1; ++i) {
+    const int delay = i + 1;
+    auto old_cepstral_coeffs = ring_buf.GetArrayView(delay);
+    distances[i] = 0.f;
+    for (int k = 0; k < kNumBands; ++k) {
+      const float c = new_cepstral_coeffs[k] - old_cepstral_coeffs[k];
+      distances[i] += c * c;
+    }
+  }
+  // Push the new spectral distance stats into the symmetric matrix buffer.
+  sym_matrix_buf->Push(distances);
+}
+
+// Computes the first half of the Vorbis window.
+std::array<float, kFrameSize20ms24kHz / 2> ComputeScaledHalfVorbisWindow(
+    float scaling = 1.f) {
+  constexpr int kHalfSize = kFrameSize20ms24kHz / 2;
+  std::array<float, kHalfSize> half_window{};
+  for (int i = 0; i < kHalfSize; ++i) {
+    half_window[i] =
+        scaling *
+        std::sin(0.5 * kPi * std::sin(0.5 * kPi * (i + 0.5) / kHalfSize) *
+                 std::sin(0.5 * kPi * (i + 0.5) / kHalfSize));
+  }
+  return half_window;
+}
+
+// Computes the forward FFT on a 20 ms frame to which a given window function is
+// applied. The Fourier coefficient corresponding to the Nyquist frequency is
+// set to zero (it is never used and this allows to simplify the code).
+void ComputeWindowedForwardFft(
+    rtc::ArrayView<const float, kFrameSize20ms24kHz> frame,
+    const std::array<float, kFrameSize20ms24kHz / 2>& half_window,
+    Pffft::FloatBuffer* fft_input_buffer,
+    Pffft::FloatBuffer* fft_output_buffer,
+    Pffft* fft) {
+  RTC_DCHECK_EQ(frame.size(), 2 * half_window.size());
+  // Apply windowing.
+  auto in = fft_input_buffer->GetView();
+  for (int i = 0, j = kFrameSize20ms24kHz - 1;
+       rtc::SafeLt(i, half_window.size()); ++i, --j) {
+    in[i] = frame[i] * half_window[i];
+    in[j] = frame[j] * half_window[i];
+  }
+  fft->ForwardTransform(*fft_input_buffer, fft_output_buffer, /*ordered=*/true);
+  // Set the Nyquist frequency coefficient to zero.
+  auto out = fft_output_buffer->GetView();
+  out[1] = 0.f;
+}
+
+}  // namespace
+
+SpectralFeaturesExtractor::SpectralFeaturesExtractor()
+    : half_window_(ComputeScaledHalfVorbisWindow(
+          1.f / static_cast<float>(kFrameSize20ms24kHz))),
+      fft_(kFrameSize20ms24kHz, Pffft::FftType::kReal),
+      fft_buffer_(fft_.CreateBuffer()),
+      reference_frame_fft_(fft_.CreateBuffer()),
+      lagged_frame_fft_(fft_.CreateBuffer()),
+      dct_table_(ComputeDctTable()) {}
+
+SpectralFeaturesExtractor::~SpectralFeaturesExtractor() = default;
+
+void SpectralFeaturesExtractor::Reset() {
+  cepstral_coeffs_ring_buf_.Reset();
+  cepstral_diffs_buf_.Reset();
+}
+
+bool SpectralFeaturesExtractor::CheckSilenceComputeFeatures(
+    rtc::ArrayView<const float, kFrameSize20ms24kHz> reference_frame,
+    rtc::ArrayView<const float, kFrameSize20ms24kHz> lagged_frame,
+    rtc::ArrayView<float, kNumBands - kNumLowerBands> higher_bands_cepstrum,
+    rtc::ArrayView<float, kNumLowerBands> average,
+    rtc::ArrayView<float, kNumLowerBands> first_derivative,
+    rtc::ArrayView<float, kNumLowerBands> second_derivative,
+    rtc::ArrayView<float, kNumLowerBands> bands_cross_corr,
+    float* variability) {
+  // Compute the Opus band energies for the reference frame.
+  ComputeWindowedForwardFft(reference_frame, half_window_, fft_buffer_.get(),
+                            reference_frame_fft_.get(), &fft_);
+  spectral_correlator_.ComputeAutoCorrelation(
+      reference_frame_fft_->GetConstView(), reference_frame_bands_energy_);
+  // Check if the reference frame has silence.
+  const float tot_energy =
+      std::accumulate(reference_frame_bands_energy_.begin(),
+                      reference_frame_bands_energy_.end(), 0.f);
+  if (tot_energy < kSilenceThreshold) {
+    return true;
+  }
+  // Compute the Opus band energies for the lagged frame.
+  ComputeWindowedForwardFft(lagged_frame, half_window_, fft_buffer_.get(),
+                            lagged_frame_fft_.get(), &fft_);
+  spectral_correlator_.ComputeAutoCorrelation(lagged_frame_fft_->GetConstView(),
+                                              lagged_frame_bands_energy_);
+  // Log of the band energies for the reference frame.
+  std::array<float, kNumBands> log_bands_energy;
+  ComputeSmoothedLogMagnitudeSpectrum(reference_frame_bands_energy_,
+                                      log_bands_energy);
+  // Reference frame cepstrum.
+  std::array<float, kNumBands> cepstrum;
+  ComputeDct(log_bands_energy, dct_table_, cepstrum);
+  // Ad-hoc correction terms for the first two cepstral coefficients.
+  cepstrum[0] -= 12.f;
+  cepstrum[1] -= 4.f;
+  // Update the ring buffer and the cepstral difference stats.
+  cepstral_coeffs_ring_buf_.Push(cepstrum);
+  UpdateCepstralDifferenceStats(cepstrum, cepstral_coeffs_ring_buf_,
+                                &cepstral_diffs_buf_);
+  // Write the higher bands cepstral coefficients.
+  RTC_DCHECK_EQ(cepstrum.size() - kNumLowerBands, higher_bands_cepstrum.size());
+  std::copy(cepstrum.begin() + kNumLowerBands, cepstrum.end(),
+            higher_bands_cepstrum.begin());
+  // Compute and write remaining features.
+  ComputeAvgAndDerivatives(average, first_derivative, second_derivative);
+  ComputeNormalizedCepstralCorrelation(bands_cross_corr);
+  RTC_DCHECK(variability);
+  *variability = ComputeVariability();
+  return false;
+}
+
+void SpectralFeaturesExtractor::ComputeAvgAndDerivatives(
+    rtc::ArrayView<float, kNumLowerBands> average,
+    rtc::ArrayView<float, kNumLowerBands> first_derivative,
+    rtc::ArrayView<float, kNumLowerBands> second_derivative) const {
+  auto curr = cepstral_coeffs_ring_buf_.GetArrayView(0);
+  auto prev1 = cepstral_coeffs_ring_buf_.GetArrayView(1);
+  auto prev2 = cepstral_coeffs_ring_buf_.GetArrayView(2);
+  RTC_DCHECK_EQ(average.size(), first_derivative.size());
+  RTC_DCHECK_EQ(first_derivative.size(), second_derivative.size());
+  RTC_DCHECK_LE(average.size(), curr.size());
+  for (int i = 0; rtc::SafeLt(i, average.size()); ++i) {
+    // Average, kernel: [1, 1, 1].
+    average[i] = curr[i] + prev1[i] + prev2[i];
+    // First derivative, kernel: [1, 0, - 1].
+    first_derivative[i] = curr[i] - prev2[i];
+    // Second derivative, Laplacian kernel: [1, -2, 1].
+    second_derivative[i] = curr[i] - 2 * prev1[i] + prev2[i];
+  }
+}
+
+void SpectralFeaturesExtractor::ComputeNormalizedCepstralCorrelation(
+    rtc::ArrayView<float, kNumLowerBands> bands_cross_corr) {
+  spectral_correlator_.ComputeCrossCorrelation(
+      reference_frame_fft_->GetConstView(), lagged_frame_fft_->GetConstView(),
+      bands_cross_corr_);
+  // Normalize.
+  for (int i = 0; rtc::SafeLt(i, bands_cross_corr_.size()); ++i) {
+    bands_cross_corr_[i] =
+        bands_cross_corr_[i] /
+        std::sqrt(0.001f + reference_frame_bands_energy_[i] *
+                               lagged_frame_bands_energy_[i]);
+  }
+  // Cepstrum.
+  ComputeDct(bands_cross_corr_, dct_table_, bands_cross_corr);
+  // Ad-hoc correction terms for the first two cepstral coefficients.
+  bands_cross_corr[0] -= 1.3f;
+  bands_cross_corr[1] -= 0.9f;
+}
+
+float SpectralFeaturesExtractor::ComputeVariability() const {
+  // Compute cepstral variability score.
+  float variability = 0.f;
+  for (int delay1 = 0; delay1 < kCepstralCoeffsHistorySize; ++delay1) {
+    float min_dist = std::numeric_limits<float>::max();
+    for (int delay2 = 0; delay2 < kCepstralCoeffsHistorySize; ++delay2) {
+      if (delay1 == delay2)  // The distance would be 0.
+        continue;
+      min_dist =
+          std::min(min_dist, cepstral_diffs_buf_.GetValue(delay1, delay2));
+    }
+    variability += min_dist;
+  }
+  // Normalize (based on training set stats).
+  // TODO(bugs.webrtc.org/10480): Isolate normalization from feature extraction.
+  return variability / kCepstralCoeffsHistorySize - 2.1f;
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h
new file mode 100644
index 0000000000..d327ef8e01
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_H_
+
+#include <array>
+#include <cstddef>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
+#include "modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h"
+#include "modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h"
+#include "modules/audio_processing/utility/pffft_wrapper.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Class to compute spectral features.
+class SpectralFeaturesExtractor {
+ public:
+  SpectralFeaturesExtractor();
+  SpectralFeaturesExtractor(const SpectralFeaturesExtractor&) = delete;
+  SpectralFeaturesExtractor& operator=(const SpectralFeaturesExtractor&) =
+      delete;
+  ~SpectralFeaturesExtractor();
+  // Resets the internal state of the feature extractor.
+  void Reset();
+  // Analyzes a pair of reference and lagged frames from the pitch buffer,
+  // detects silence and computes features. If silence is detected, the output
+  // is neither computed nor written.
+  bool CheckSilenceComputeFeatures(
+      rtc::ArrayView<const float, kFrameSize20ms24kHz> reference_frame,
+      rtc::ArrayView<const float, kFrameSize20ms24kHz> lagged_frame,
+      rtc::ArrayView<float, kNumBands - kNumLowerBands> higher_bands_cepstrum,
+      rtc::ArrayView<float, kNumLowerBands> average,
+      rtc::ArrayView<float, kNumLowerBands> first_derivative,
+      rtc::ArrayView<float, kNumLowerBands> second_derivative,
+      rtc::ArrayView<float, kNumLowerBands> bands_cross_corr,
+      float* variability);
+
+ private:
+  void ComputeAvgAndDerivatives(
+      rtc::ArrayView<float, kNumLowerBands> average,
+      rtc::ArrayView<float, kNumLowerBands> first_derivative,
+      rtc::ArrayView<float, kNumLowerBands> second_derivative) const;
+  void ComputeNormalizedCepstralCorrelation(
+      rtc::ArrayView<float, kNumLowerBands> bands_cross_corr);
+  float ComputeVariability() const;
+
+  const std::array<float, kFrameSize20ms24kHz / 2> half_window_;
+  Pffft fft_;
+  std::unique_ptr<Pffft::FloatBuffer> fft_buffer_;
+  std::unique_ptr<Pffft::FloatBuffer> reference_frame_fft_;
+  std::unique_ptr<Pffft::FloatBuffer> lagged_frame_fft_;
+  SpectralCorrelator spectral_correlator_;
+  std::array<float, kOpusBands24kHz> reference_frame_bands_energy_;
+  std::array<float, kOpusBands24kHz> lagged_frame_bands_energy_;
+  std::array<float, kOpusBands24kHz> bands_cross_corr_;
+  const std::array<float, kNumBands * kNumBands> dct_table_;
+  RingBuffer<float, kNumBands, kCepstralCoeffsHistorySize>
+      cepstral_coeffs_ring_buf_;
+  SymmetricMatrixBuffer<float, kCepstralCoeffsHistorySize> cepstral_diffs_buf_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
new file mode 100644
index 0000000000..a10b0f7ec9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
@@ -0,0 +1,188 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstddef>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Weights for each FFT coefficient for each Opus band (Nyquist frequency
+// excluded). The size of each band is specified in
+// `kOpusScaleNumBins24kHz20ms`.
+constexpr std::array<float, kFrameSize20ms24kHz / 2> kOpusBandWeights24kHz20ms =
+    {{
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 0
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 1
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 2
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 3
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 4
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 5
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 6
+        0.f,       0.25f,      0.5f,       0.75f,  // Band 7
+        0.f,       0.125f,     0.25f,      0.375f,    0.5f,
+        0.625f,    0.75f,      0.875f,  // Band 8
+        0.f,       0.125f,     0.25f,      0.375f,    0.5f,
+        0.625f,    0.75f,      0.875f,  // Band 9
+        0.f,       0.125f,     0.25f,      0.375f,    0.5f,
+        0.625f,    0.75f,      0.875f,  // Band 10
+        0.f,       0.125f,     0.25f,      0.375f,    0.5f,
+        0.625f,    0.75f,      0.875f,  // Band 11
+        0.f,       0.0625f,    0.125f,     0.1875f,   0.25f,
+        0.3125f,   0.375f,     0.4375f,    0.5f,      0.5625f,
+        0.625f,    0.6875f,    0.75f,      0.8125f,   0.875f,
+        0.9375f,  // Band 12
+        0.f,       0.0625f,    0.125f,     0.1875f,   0.25f,
+        0.3125f,   0.375f,     0.4375f,    0.5f,      0.5625f,
+        0.625f,    0.6875f,    0.75f,      0.8125f,   0.875f,
+        0.9375f,  // Band 13
+        0.f,       0.0625f,    0.125f,     0.1875f,   0.25f,
+        0.3125f,   0.375f,     0.4375f,    0.5f,      0.5625f,
+        0.625f,    0.6875f,    0.75f,      0.8125f,   0.875f,
+        0.9375f,  // Band 14
+        0.f,       0.0416667f, 0.0833333f, 0.125f,    0.166667f,
+        0.208333f, 0.25f,      0.291667f,  0.333333f, 0.375f,
+        0.416667f, 0.458333f,  0.5f,       0.541667f, 0.583333f,
+        0.625f,    0.666667f,  0.708333f,  0.75f,     0.791667f,
+        0.833333f, 0.875f,     0.916667f,  0.958333f,  // Band 15
+        0.f,       0.0416667f, 0.0833333f, 0.125f,    0.166667f,
+        0.208333f, 0.25f,      0.291667f,  0.333333f, 0.375f,
+        0.416667f, 0.458333f,  0.5f,       0.541667f, 0.583333f,
+        0.625f,    0.666667f,  0.708333f,  0.75f,     0.791667f,
+        0.833333f, 0.875f,     0.916667f,  0.958333f,  // Band 16
+        0.f,       0.03125f,   0.0625f,    0.09375f,  0.125f,
+        0.15625f,  0.1875f,    0.21875f,   0.25f,     0.28125f,
+        0.3125f,   0.34375f,   0.375f,     0.40625f,  0.4375f,
+        0.46875f,  0.5f,       0.53125f,   0.5625f,   0.59375f,
+        0.625f,    0.65625f,   0.6875f,    0.71875f,  0.75f,
+        0.78125f,  0.8125f,    0.84375f,   0.875f,    0.90625f,
+        0.9375f,   0.96875f,  // Band 17
+        0.f,       0.0208333f, 0.0416667f, 0.0625f,   0.0833333f,
+        0.104167f, 0.125f,     0.145833f,  0.166667f, 0.1875f,
+        0.208333f, 0.229167f,  0.25f,      0.270833f, 0.291667f,
+        0.3125f,   0.333333f,  0.354167f,  0.375f,    0.395833f,
+        0.416667f, 0.4375f,    0.458333f,  0.479167f, 0.5f,
+        0.520833f, 0.541667f,  0.5625f,    0.583333f, 0.604167f,
+        0.625f,    0.645833f,  0.666667f,  0.6875f,   0.708333f,
+        0.729167f, 0.75f,      0.770833f,  0.791667f, 0.8125f,
+        0.833333f, 0.854167f,  0.875f,     0.895833f, 0.916667f,
+        0.9375f,   0.958333f,  0.979167f  // Band 18
+    }};
+
+}  // namespace
+
+SpectralCorrelator::SpectralCorrelator()
+    : weights_(kOpusBandWeights24kHz20ms.begin(),
+               kOpusBandWeights24kHz20ms.end()) {}
+
+SpectralCorrelator::~SpectralCorrelator() = default;
+
+void SpectralCorrelator::ComputeAutoCorrelation(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float, kOpusBands24kHz> auto_corr) const {
+  ComputeCrossCorrelation(x, x, auto_corr);
+}
+
+void SpectralCorrelator::ComputeCrossCorrelation(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<const float> y,
+    rtc::ArrayView<float, kOpusBands24kHz> cross_corr) const {
+  RTC_DCHECK_EQ(x.size(), kFrameSize20ms24kHz);
+  RTC_DCHECK_EQ(x.size(), y.size());
+  RTC_DCHECK_EQ(x[1], 0.f) << "The Nyquist coefficient must be zeroed.";
+  RTC_DCHECK_EQ(y[1], 0.f) << "The Nyquist coefficient must be zeroed.";
+  constexpr auto kOpusScaleNumBins24kHz20ms = GetOpusScaleNumBins24kHz20ms();
+  int k = 0;  // Next Fourier coefficient index.
+  cross_corr[0] = 0.f;
+  for (int i = 0; i < kOpusBands24kHz - 1; ++i) {
+    cross_corr[i + 1] = 0.f;
+    for (int j = 0; j < kOpusScaleNumBins24kHz20ms[i]; ++j) {  // Band size.
+      const float v = x[2 * k] * y[2 * k] + x[2 * k + 1] * y[2 * k + 1];
+      const float tmp = weights_[k] * v;
+      cross_corr[i] += v - tmp;
+      cross_corr[i + 1] += tmp;
+      k++;
+    }
+  }
+  cross_corr[0] *= 2.f;  // The first band only gets half contribution.
+  RTC_DCHECK_EQ(k, kFrameSize20ms24kHz / 2);  // Nyquist coefficient never used.
+}
+
+void ComputeSmoothedLogMagnitudeSpectrum(
+    rtc::ArrayView<const float> bands_energy,
+    rtc::ArrayView<float, kNumBands> log_bands_energy) {
+  RTC_DCHECK_LE(bands_energy.size(), kNumBands);
+  constexpr float kOneByHundred = 1e-2f;
+  constexpr float kLogOneByHundred = -2.f;
+  // Init.
+  float log_max = kLogOneByHundred;
+  float follow = kLogOneByHundred;
+  const auto smooth = [&log_max, &follow](float x) {
+    x = std::max(log_max - 7.f, std::max(follow - 1.5f, x));
+    log_max = std::max(log_max, x);
+    follow = std::max(follow - 1.5f, x);
+    return x;
+  };
+  // Smoothing over the bands for which the band energy is defined.
+  for (int i = 0; rtc::SafeLt(i, bands_energy.size()); ++i) {
+    log_bands_energy[i] = smooth(std::log10(kOneByHundred + bands_energy[i]));
+  }
+  // Smoothing over the remaining bands (zero energy).
+  for (int i = bands_energy.size(); i < kNumBands; ++i) {
+    log_bands_energy[i] = smooth(kLogOneByHundred);
+  }
+}
+
+std::array<float, kNumBands * kNumBands> ComputeDctTable() {
+  std::array<float, kNumBands * kNumBands> dct_table;
+  const double k = std::sqrt(0.5);
+  for (int i = 0; i < kNumBands; ++i) {
+    for (int j = 0; j < kNumBands; ++j)
+      dct_table[i * kNumBands + j] = std::cos((i + 0.5) * j * kPi / kNumBands);
+    dct_table[i * kNumBands] *= k;
+  }
+  return dct_table;
+}
+
+void ComputeDct(rtc::ArrayView<const float> in,
+                rtc::ArrayView<const float, kNumBands * kNumBands> dct_table,
+                rtc::ArrayView<float> out) {
+  // DCT scaling factor - i.e., sqrt(2 / kNumBands).
+  constexpr float kDctScalingFactor = 0.301511345f;
+  constexpr float kDctScalingFactorError =
+      kDctScalingFactor * kDctScalingFactor -
+      2.f / static_cast<float>(kNumBands);
+  static_assert(
+      (kDctScalingFactorError >= 0.f && kDctScalingFactorError < 1e-1f) ||
+          (kDctScalingFactorError < 0.f && kDctScalingFactorError > -1e-1f),
+      "kNumBands changed and kDctScalingFactor has not been updated.");
+  RTC_DCHECK_NE(in.data(), out.data()) << "In-place DCT is not supported.";
+  RTC_DCHECK_LE(in.size(), kNumBands);
+  RTC_DCHECK_LE(1, out.size());
+  RTC_DCHECK_LE(out.size(), in.size());
+  for (int i = 0; rtc::SafeLt(i, out.size()); ++i) {
+    out[i] = 0.f;
+    for (int j = 0; rtc::SafeLt(j, in.size()); ++j) {
+      out[i] += in[j] * dct_table[j * kNumBands + i];
+    }
+    // TODO(bugs.webrtc.org/10480): Scaling factor in the DCT table.
+    out[i] *= kDctScalingFactor;
+  }
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
new file mode 100644
index 0000000000..f4b293a567
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// At a sample rate of 24 kHz, the last 3 Opus bands are beyond the Nyquist
+// frequency. However, band #19 gets the contributions from band #18 because
+// of the symmetric triangular filter with peak response at 12 kHz.
+constexpr int kOpusBands24kHz = 20;
+static_assert(kOpusBands24kHz < kNumBands,
+              "The number of bands at 24 kHz must be less than those defined "
+              "in the Opus scale at 48 kHz.");
+
+// Number of FFT frequency bins covered by each band in the Opus scale at a
+// sample rate of 24 kHz for 20 ms frames.
+// Declared here for unit testing.
+constexpr std::array<int, kOpusBands24kHz - 1> GetOpusScaleNumBins24kHz20ms() {
+  return {4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 16, 16, 16, 24, 24, 32, 48};
+}
+
+// TODO(bugs.webrtc.org/10480): Move to a separate file.
+// Class to compute band-wise spectral features in the Opus perceptual scale
+// for 20 ms frames sampled at 24 kHz. The analysis methods apply triangular
+// filters with peak response at the each band boundary.
+class SpectralCorrelator {
+ public:
+  // Ctor.
+  SpectralCorrelator();
+  SpectralCorrelator(const SpectralCorrelator&) = delete;
+  SpectralCorrelator& operator=(const SpectralCorrelator&) = delete;
+  ~SpectralCorrelator();
+
+  // Computes the band-wise spectral auto-correlations.
+  // `x` must:
+  //  - have size equal to `kFrameSize20ms24kHz`;
+  //  - be encoded as vectors of interleaved real-complex FFT coefficients
+  //    where x[1] = y[1] = 0 (the Nyquist frequency coefficient is omitted).
+  void ComputeAutoCorrelation(
+      rtc::ArrayView<const float> x,
+      rtc::ArrayView<float, kOpusBands24kHz> auto_corr) const;
+
+  // Computes the band-wise spectral cross-correlations.
+  // `x` and `y` must:
+  //  - have size equal to `kFrameSize20ms24kHz`;
+  //  - be encoded as vectors of interleaved real-complex FFT coefficients where
+  //    x[1] = y[1] = 0 (the Nyquist frequency coefficient is omitted).
+  void ComputeCrossCorrelation(
+      rtc::ArrayView<const float> x,
+      rtc::ArrayView<const float> y,
+      rtc::ArrayView<float, kOpusBands24kHz> cross_corr) const;
+
+ private:
+  const std::vector<float> weights_;  // Weights for each Fourier coefficient.
+};
+
+// TODO(bugs.webrtc.org/10480): Move to anonymous namespace in
+// spectral_features.cc. Given a vector of Opus-bands energy coefficients,
+// computes the log magnitude spectrum applying smoothing both over time and
+// over frequency. Declared here for unit testing.
+void ComputeSmoothedLogMagnitudeSpectrum(
+    rtc::ArrayView<const float> bands_energy,
+    rtc::ArrayView<float, kNumBands> log_bands_energy);
+
+// TODO(bugs.webrtc.org/10480): Move to anonymous namespace in
+// spectral_features.cc. Creates a DCT table for arrays having size equal to
+// `kNumBands`. Declared here for unit testing.
+std::array<float, kNumBands * kNumBands> ComputeDctTable();
+
+// TODO(bugs.webrtc.org/10480): Move to anonymous namespace in
+// spectral_features.cc. Computes DCT for `in` given a pre-computed DCT table.
+// In-place computation is not allowed and `out` can be smaller than `in` in
+// order to only compute the first DCT coefficients. Declared here for unit
+// testing.
+void ComputeDct(rtc::ArrayView<const float> in,
+                rtc::ArrayView<const float, kNumBands * kNumBands> dct_table,
+                rtc::ArrayView<float> out);
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
new file mode 100644
index 0000000000..ece4eb5024
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
@@ -0,0 +1,160 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h"
+
+#include <algorithm>
+#include <array>
+#include <complex>
+#include <numeric>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "modules/audio_processing/utility/pffft_wrapper.h"
+#include "rtc_base/numerics/safe_compare.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// Generates the values for the array named `kOpusBandWeights24kHz20ms` in the
+// anonymous namespace of the .cc file, which is the array of FFT coefficient
+// weights for the Opus scale triangular filters.
+std::vector<float> ComputeTriangularFiltersWeights() {
+  constexpr auto kOpusScaleNumBins24kHz20ms = GetOpusScaleNumBins24kHz20ms();
+  const auto& v = kOpusScaleNumBins24kHz20ms;  // Alias.
+  const int num_weights = std::accumulate(kOpusScaleNumBins24kHz20ms.begin(),
+                                          kOpusScaleNumBins24kHz20ms.end(), 0);
+  std::vector<float> weights(num_weights);
+  int next_fft_coeff_index = 0;
+  for (int band = 0; rtc::SafeLt(band, v.size()); ++band) {
+    const int band_size = v[band];
+    for (int j = 0; rtc::SafeLt(j, band_size); ++j) {
+      weights[next_fft_coeff_index + j] = static_cast<float>(j) / band_size;
+    }
+    next_fft_coeff_index += band_size;
+  }
+  return weights;
+}
+
+// Checks that the values returned by GetOpusScaleNumBins24kHz20ms() match the
+// Opus scale frequency boundaries.
+TEST(RnnVadTest, TestOpusScaleBoundaries) {
+  constexpr int kBandFrequencyBoundariesHz[kNumBands - 1] = {
+      200,  400,  600,  800,  1000, 1200, 1400, 1600,  2000,  2400, 2800,
+      3200, 4000, 4800, 5600, 6800, 8000, 9600, 12000, 15600, 20000};
+  constexpr auto kOpusScaleNumBins24kHz20ms = GetOpusScaleNumBins24kHz20ms();
+  int prev = 0;
+  for (int i = 0; rtc::SafeLt(i, kOpusScaleNumBins24kHz20ms.size()); ++i) {
+    int boundary =
+        kBandFrequencyBoundariesHz[i] * kFrameSize20ms24kHz / kSampleRate24kHz;
+    EXPECT_EQ(kOpusScaleNumBins24kHz20ms[i], boundary - prev);
+    prev = boundary;
+  }
+}
+
+// Checks that the computed triangular filters weights for the Opus scale are
+// monotonic withing each Opus band. This test should only be enabled when
+// ComputeTriangularFiltersWeights() is changed and `kOpusBandWeights24kHz20ms`
+// is updated accordingly.
+TEST(RnnVadTest, DISABLED_TestOpusScaleWeights) {
+  auto weights = ComputeTriangularFiltersWeights();
+  int i = 0;
+  for (int band_size : GetOpusScaleNumBins24kHz20ms()) {
+    SCOPED_TRACE(band_size);
+    rtc::ArrayView<float> band_weights(weights.data() + i, band_size);
+    float prev = -1.f;
+    for (float weight : band_weights) {
+      EXPECT_LT(prev, weight);
+      prev = weight;
+    }
+    i += band_size;
+  }
+}
+
+// Checks that the computed band-wise auto-correlation is non-negative for a
+// simple input vector of FFT coefficients.
+TEST(RnnVadTest, SpectralCorrelatorValidOutput) {
+  // Input: vector of (1, 1j) values.
+  Pffft fft(kFrameSize20ms24kHz, Pffft::FftType::kReal);
+  auto in = fft.CreateBuffer();
+  std::array<float, kOpusBands24kHz> out;
+  auto in_view = in->GetView();
+  std::fill(in_view.begin(), in_view.end(), 1.f);
+  in_view[1] = 0.f;  // Nyquist frequency.
+  // Compute and check output.
+  SpectralCorrelator e;
+  e.ComputeAutoCorrelation(in_view, out);
+  for (int i = 0; i < kOpusBands24kHz; ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_GT(out[i], 0.f);
+  }
+}
+
+// Checks that the computed smoothed log magnitude spectrum is within tolerance
+// given hard-coded test input data.
+TEST(RnnVadTest, ComputeSmoothedLogMagnitudeSpectrumWithinTolerance) {
+  constexpr std::array<float, kNumBands> input = {
+      {86.060539245605f, 275.668334960938f, 43.406528472900f, 6.541896820068f,
+       17.964015960693f, 8.090919494629f,   1.261920094490f,  1.212702631950f,
+       1.619154453278f,  0.508935272694f,   0.346316039562f,  0.237035423517f,
+       0.172424271703f,  0.271657168865f,   0.126088857651f,  0.139967113733f,
+       0.207200810313f,  0.155893072486f,   0.091090843081f,  0.033391401172f,
+       0.013879744336f,  0.011973354965f}};
+  constexpr std::array<float, kNumBands> expected_output = {
+      {1.934854507446f,  2.440402746201f,  1.637655138969f,  0.816367030144f,
+       1.254645109177f,  0.908534288406f,  0.104459829628f,  0.087320849299f,
+       0.211962252855f,  -0.284886807203f, -0.448164641857f, -0.607240796089f,
+       -0.738917350769f, -0.550279200077f, -0.866177439690f, -0.824003994465f,
+       -0.663138568401f, -0.780171751976f, -0.995288193226f, -1.362596273422f,
+       -1.621970295906f, -1.658103585243f}};
+  std::array<float, kNumBands> computed_output;
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    ComputeSmoothedLogMagnitudeSpectrum(input, computed_output);
+    ExpectNearAbsolute(expected_output, computed_output, 1e-5f);
+  }
+}
+
+// Checks that the computed DCT is within tolerance given hard-coded test input
+// data.
+TEST(RnnVadTest, ComputeDctWithinTolerance) {
+  constexpr std::array<float, kNumBands> input = {
+      {0.232155621052f,  0.678957760334f, 0.220818966627f,  -0.077363930643f,
+       -0.559227049351f, 0.432545185089f, 0.353900641203f,  0.398993015289f,
+       0.409774333239f,  0.454977899790f, 0.300520688295f,  -0.010286616161f,
+       0.272525429726f,  0.098067551851f, 0.083649002016f,  0.046226885170f,
+       -0.033228103071f, 0.144773483276f, -0.117661058903f, -0.005628800020f,
+       -0.009547689930f, -0.045382082462f}};
+  constexpr std::array<float, kNumBands> expected_output = {
+      {0.697072803974f,  0.442710995674f,  -0.293156713247f, -0.060711503029f,
+       0.292050391436f,  0.489301353693f,  0.402255415916f,  0.134404733777f,
+       -0.086305990815f, -0.199605688453f, -0.234511867166f, -0.413774639368f,
+       -0.388507157564f, -0.032798115164f, 0.044605545700f,  0.112466648221f,
+       -0.050096966326f, 0.045971218497f,  -0.029815061018f, -0.410366982222f,
+       -0.209233760834f, -0.128037497401f}};
+  auto dct_table = ComputeDctTable();
+  std::array<float, kNumBands> computed_output;
+  {
+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+    // FloatingPointExceptionObserver fpe_observer;
+    ComputeDct(input, dct_table, computed_output);
+    ExpectNearAbsolute(expected_output, computed_output, 1e-5f);
+  }
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
new file mode 100644
index 0000000000..324d694957
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
@@ -0,0 +1,160 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/spectral_features.h"
+
+#include <algorithm>
+
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+// #include "test/fpe_observer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kTestFeatureVectorSize = kNumBands + 3 * kNumLowerBands + 1;
+
+// Writes non-zero sample values.
+void WriteTestData(rtc::ArrayView<float> samples) {
+  for (int i = 0; rtc::SafeLt(i, samples.size()); ++i) {
+    samples[i] = i % 100;
+  }
+}
+
+rtc::ArrayView<float, kNumBands - kNumLowerBands> GetHigherBandsSpectrum(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data() + kNumLowerBands, kNumBands - kNumLowerBands};
+}
+
+rtc::ArrayView<float, kNumLowerBands> GetAverage(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data(), kNumLowerBands};
+}
+
+rtc::ArrayView<float, kNumLowerBands> GetFirstDerivative(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data() + kNumBands, kNumLowerBands};
+}
+
+rtc::ArrayView<float, kNumLowerBands> GetSecondDerivative(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data() + kNumBands + kNumLowerBands, kNumLowerBands};
+}
+
+rtc::ArrayView<float, kNumLowerBands> GetCepstralCrossCorrelation(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return {feature_vector->data() + kNumBands + 2 * kNumLowerBands,
+          kNumLowerBands};
+}
+
+float* GetCepstralVariability(
+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
+  return feature_vector->data() + kNumBands + 3 * kNumLowerBands;
+}
+
+constexpr float kInitialFeatureVal = -9999.f;
+
+// Checks that silence is detected when the input signal is 0 and that the
+// feature vector is written only if the input signal is not tagged as silence.
+TEST(RnnVadTest, SpectralFeaturesWithAndWithoutSilence) {
+  // Initialize.
+  SpectralFeaturesExtractor sfe;
+  std::array<float, kFrameSize20ms24kHz> samples;
+  rtc::ArrayView<float, kFrameSize20ms24kHz> samples_view(samples);
+  bool is_silence;
+  std::array<float, kTestFeatureVectorSize> feature_vector;
+
+  // Write an initial value in the feature vector to detect changes.
+  std::fill(feature_vector.begin(), feature_vector.end(), kInitialFeatureVal);
+
+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
+  // FloatingPointExceptionObserver fpe_observer;
+
+  // With silence.
+  std::fill(samples.begin(), samples.end(), 0.f);
+  is_silence = sfe.CheckSilenceComputeFeatures(
+      samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector),
+      GetAverage(&feature_vector), GetFirstDerivative(&feature_vector),
+      GetSecondDerivative(&feature_vector),
+      GetCepstralCrossCorrelation(&feature_vector),
+      GetCepstralVariability(&feature_vector));
+  // Silence is expected, the output won't be overwritten.
+  EXPECT_TRUE(is_silence);
+  EXPECT_TRUE(std::all_of(feature_vector.begin(), feature_vector.end(),
+                          [](float x) { return x == kInitialFeatureVal; }));
+
+  // With no silence.
+  WriteTestData(samples);
+  is_silence = sfe.CheckSilenceComputeFeatures(
+      samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector),
+      GetAverage(&feature_vector), GetFirstDerivative(&feature_vector),
+      GetSecondDerivative(&feature_vector),
+      GetCepstralCrossCorrelation(&feature_vector),
+      GetCepstralVariability(&feature_vector));
+  // Silence is not expected, the output will be overwritten.
+  EXPECT_FALSE(is_silence);
+  EXPECT_FALSE(std::all_of(feature_vector.begin(), feature_vector.end(),
+                           [](float x) { return x == kInitialFeatureVal; }));
+}
+
+// Feeds a constant input signal and checks that:
+// - the cepstral coefficients average does not change;
+// - the derivatives are zero;
+// - the cepstral variability score does not change.
+TEST(RnnVadTest, CepstralFeaturesConstantAverageZeroDerivative) {
+  // Initialize.
+  SpectralFeaturesExtractor sfe;
+  std::array<float, kFrameSize20ms24kHz> samples;
+  rtc::ArrayView<float, kFrameSize20ms24kHz> samples_view(samples);
+  WriteTestData(samples);
+
+  // Fill the spectral features with test data.
+  std::array<float, kTestFeatureVectorSize> feature_vector;
+  for (int i = 0; i < kCepstralCoeffsHistorySize; ++i) {
+    sfe.CheckSilenceComputeFeatures(
+        samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector),
+        GetAverage(&feature_vector), GetFirstDerivative(&feature_vector),
+        GetSecondDerivative(&feature_vector),
+        GetCepstralCrossCorrelation(&feature_vector),
+        GetCepstralVariability(&feature_vector));
+  }
+
+  // Feed the test data one last time but using a different output vector.
+  std::array<float, kTestFeatureVectorSize> feature_vector_last;
+  sfe.CheckSilenceComputeFeatures(
+      samples_view, samples_view, GetHigherBandsSpectrum(&feature_vector_last),
+      GetAverage(&feature_vector_last),
+      GetFirstDerivative(&feature_vector_last),
+      GetSecondDerivative(&feature_vector_last),
+      GetCepstralCrossCorrelation(&feature_vector_last),
+      GetCepstralVariability(&feature_vector_last));
+
+  // Average is unchanged.
+  ExpectEqualFloatArray({feature_vector.data(), kNumLowerBands},
+                        {feature_vector_last.data(), kNumLowerBands});
+  // First and second derivatives are zero.
+  constexpr std::array<float, kNumLowerBands> zeros{};
+  ExpectEqualFloatArray(
+      {feature_vector_last.data() + kNumBands, kNumLowerBands}, zeros);
+  ExpectEqualFloatArray(
+      {feature_vector_last.data() + kNumBands + kNumLowerBands, kNumLowerBands},
+      zeros);
+  // Variability is unchanged.
+  EXPECT_FLOAT_EQ(feature_vector[kNumBands + 3 * kNumLowerBands],
+                  feature_vector_last[kNumBands + 3 * kNumLowerBands]);
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
new file mode 100644
index 0000000000..d186479551
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SYMMETRIC_MATRIX_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SYMMETRIC_MATRIX_BUFFER_H_
+
+#include <algorithm>
+#include <array>
+#include <cstring>
+#include <utility>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Data structure to buffer the results of pair-wise comparisons between items
+// stored in a ring buffer. Every time that the oldest item is replaced in the
+// ring buffer, the new one is compared to the remaining items in the ring
+// buffer. The results of such comparisons need to be buffered and automatically
+// removed when one of the two corresponding items that have been compared is
+// removed from the ring buffer. It is assumed that the comparison is symmetric
+// and that comparing an item with itself is not needed.
+template <typename T, int S>
+class SymmetricMatrixBuffer {
+  static_assert(S > 2, "");
+
+ public:
+  SymmetricMatrixBuffer() = default;
+  SymmetricMatrixBuffer(const SymmetricMatrixBuffer&) = delete;
+  SymmetricMatrixBuffer& operator=(const SymmetricMatrixBuffer&) = delete;
+  ~SymmetricMatrixBuffer() = default;
+  // Sets the buffer values to zero.
+  void Reset() {
+    static_assert(std::is_arithmetic<T>::value,
+                  "Integral or floating point required.");
+    buf_.fill(0);
+  }
+  // Pushes the results from the comparison between the most recent item and
+  // those that are still in the ring buffer. The first element in `values` must
+  // correspond to the comparison between the most recent item and the second
+  // most recent one in the ring buffer, whereas the last element in `values`
+  // must correspond to the comparison between the most recent item and the
+  // oldest one in the ring buffer.
+  void Push(rtc::ArrayView<T, S - 1> values) {
+    // Move the lower-right sub-matrix of size (S-2) x (S-2) one row up and one
+    // column left.
+    std::memmove(buf_.data(), buf_.data() + S, (buf_.size() - S) * sizeof(T));
+    // Copy new values in the last column in the right order.
+    for (int i = 0; rtc::SafeLt(i, values.size()); ++i) {
+      const int index = (S - 1 - i) * (S - 1) - 1;
+      RTC_DCHECK_GE(index, 0);
+      RTC_DCHECK_LT(index, buf_.size());
+      buf_[index] = values[i];
+    }
+  }
+  // Reads the value that corresponds to comparison of two items in the ring
+  // buffer having delay `delay1` and `delay2`. The two arguments must not be
+  // equal and both must be in {0, ..., S - 1}.
+  T GetValue(int delay1, int delay2) const {
+    int row = S - 1 - delay1;
+    int col = S - 1 - delay2;
+    RTC_DCHECK_NE(row, col) << "The diagonal cannot be accessed.";
+    if (row > col)
+      std::swap(row, col);  // Swap to access the upper-right triangular part.
+    RTC_DCHECK_LE(0, row);
+    RTC_DCHECK_LT(row, S - 1) << "Not enforcing row < col and row != col.";
+    RTC_DCHECK_LE(1, col) << "Not enforcing row < col and row != col.";
+    RTC_DCHECK_LT(col, S);
+    const int index = row * (S - 1) + (col - 1);
+    RTC_DCHECK_LE(0, index);
+    RTC_DCHECK_LT(index, buf_.size());
+    return buf_[index];
+  }
+
+ private:
+  // Encode an upper-right triangular matrix (excluding its diagonal) using a
+  // square matrix. This allows to move the data in Push() with one single
+  // operation.
+  std::array<T, (S - 1) * (S - 1)> buf_{};
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SYMMETRIC_MATRIX_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
new file mode 100644
index 0000000000..1509ca5ac1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h"
+
+#include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+template <typename T, int S>
+void CheckSymmetry(const SymmetricMatrixBuffer<T, S>* sym_matrix_buf) {
+  for (int row = 0; row < S - 1; ++row)
+    for (int col = row + 1; col < S; ++col)
+      EXPECT_EQ(sym_matrix_buf->GetValue(row, col),
+                sym_matrix_buf->GetValue(col, row));
+}
+
+using PairType = std::pair<int, int>;
+
+// Checks that the symmetric matrix buffer contains any pair with a value equal
+// to the given one.
+template <int S>
+bool CheckPairsWithValueExist(
+    const SymmetricMatrixBuffer<PairType, S>* sym_matrix_buf,
+    const int value) {
+  for (int row = 0; row < S - 1; ++row) {
+    for (int col = row + 1; col < S; ++col) {
+      auto p = sym_matrix_buf->GetValue(row, col);
+      if (p.first == value || p.second == value)
+        return true;
+    }
+  }
+  return false;
+}
+
+// Test that shows how to combine RingBuffer and SymmetricMatrixBuffer to
+// efficiently compute pair-wise scores. This test verifies that the evolution
+// of a SymmetricMatrixBuffer instance follows that of RingBuffer.
+TEST(RnnVadTest, SymmetricMatrixBufferUseCase) {
+  // Instance a ring buffer which will be fed with a series of integer values.
+  constexpr int kRingBufSize = 10;
+  RingBuffer<int, 1, kRingBufSize> ring_buf;
+  // Instance a symmetric matrix buffer for the ring buffer above. It stores
+  // pairs of integers with which this test can easily check that the evolution
+  // of RingBuffer and SymmetricMatrixBuffer match.
+  SymmetricMatrixBuffer<PairType, kRingBufSize> sym_matrix_buf;
+  for (int t = 1; t <= 100; ++t) {  // Evolution steps.
+    SCOPED_TRACE(t);
+    const int t_removed = ring_buf.GetArrayView(kRingBufSize - 1)[0];
+    ring_buf.Push({&t, 1});
+    // The head of the ring buffer is `t`.
+    ASSERT_EQ(t, ring_buf.GetArrayView(0)[0]);
+    // Create the comparisons between `t` and the older elements in the ring
+    // buffer.
+    std::array<PairType, kRingBufSize - 1> new_comparions;
+    for (int i = 0; i < kRingBufSize - 1; ++i) {
+      // Start comparing `t` to the second newest element in the ring buffer.
+      const int delay = i + 1;
+      const auto t_prev = ring_buf.GetArrayView(delay)[0];
+      ASSERT_EQ(std::max(0, t - delay), t_prev);
+      // Compare the last element `t` with `t_prev`.
+      new_comparions[i].first = t_prev;
+      new_comparions[i].second = t;
+    }
+    // Push the new comparisons in the symmetric matrix buffer.
+    sym_matrix_buf.Push({new_comparions.data(), new_comparions.size()});
+    // Tests.
+    CheckSymmetry(&sym_matrix_buf);
+    // Check that the pairs resulting from the content in the ring buffer are
+    // in the right position.
+    for (int delay1 = 0; delay1 < kRingBufSize - 1; ++delay1) {
+      for (int delay2 = delay1 + 1; delay2 < kRingBufSize; ++delay2) {
+        const auto t1 = ring_buf.GetArrayView(delay1)[0];
+        const auto t2 = ring_buf.GetArrayView(delay2)[0];
+        ASSERT_LE(t2, t1);
+        const auto p = sym_matrix_buf.GetValue(delay1, delay2);
+        EXPECT_EQ(p.first, t2);
+        EXPECT_EQ(p.second, t1);
+      }
+    }
+    // Check that every older element in the ring buffer still has a
+    // corresponding pair in the symmetric matrix buffer.
+    for (int delay = 1; delay < kRingBufSize; ++delay) {
+      const auto t_prev = ring_buf.GetArrayView(delay)[0];
+      EXPECT_TRUE(CheckPairsWithValueExist(&sym_matrix_buf, t_prev));
+    }
+    // Check that the element removed from the ring buffer has no corresponding
+    // pairs in the symmetric matrix buffer.
+    if (t > kRingBufSize - 1) {
+      EXPECT_FALSE(CheckPairsWithValueExist(&sym_matrix_buf, t_removed));
+    }
+  }
+}
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.cc
new file mode 100644
index 0000000000..857a9f2706
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.cc
@@ -0,0 +1,143 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
+
+#include <algorithm>
+#include <fstream>
+#include <memory>
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+// File reader for binary files that contain a sequence of values with
+// arithmetic type `T`. The values of type `T` that are read are cast to float.
+template <typename T>
+class FloatFileReader : public FileReader {
+ public:
+  static_assert(std::is_arithmetic<T>::value, "");
+  explicit FloatFileReader(absl::string_view filename)
+      : is_(std::string(filename), std::ios::binary | std::ios::ate),
+        size_(is_.tellg() / sizeof(T)) {
+    RTC_CHECK(is_);
+    SeekBeginning();
+  }
+  FloatFileReader(const FloatFileReader&) = delete;
+  FloatFileReader& operator=(const FloatFileReader&) = delete;
+  ~FloatFileReader() = default;
+
+  int size() const override { return size_; }
+  bool ReadChunk(rtc::ArrayView<float> dst) override {
+    const std::streamsize bytes_to_read = dst.size() * sizeof(T);
+    if (std::is_same<T, float>::value) {
+      is_.read(reinterpret_cast<char*>(dst.data()), bytes_to_read);
+    } else {
+      buffer_.resize(dst.size());
+      is_.read(reinterpret_cast<char*>(buffer_.data()), bytes_to_read);
+      std::transform(buffer_.begin(), buffer_.end(), dst.begin(),
+                     [](const T& v) -> float { return static_cast<float>(v); });
+    }
+    return is_.gcount() == bytes_to_read;
+  }
+  bool ReadValue(float& dst) override { return ReadChunk({&dst, 1}); }
+  void SeekForward(int hop) override { is_.seekg(hop * sizeof(T), is_.cur); }
+  void SeekBeginning() override { is_.seekg(0, is_.beg); }
+
+ private:
+  std::ifstream is_;
+  const int size_;
+  std::vector<T> buffer_;
+};
+
+}  // namespace
+
+using webrtc::test::ResourcePath;
+
+void ExpectEqualFloatArray(rtc::ArrayView<const float> expected,
+                           rtc::ArrayView<const float> computed) {
+  ASSERT_EQ(expected.size(), computed.size());
+  for (int i = 0; rtc::SafeLt(i, expected.size()); ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_FLOAT_EQ(expected[i], computed[i]);
+  }
+}
+
+void ExpectNearAbsolute(rtc::ArrayView<const float> expected,
+                        rtc::ArrayView<const float> computed,
+                        float tolerance) {
+  ASSERT_EQ(expected.size(), computed.size());
+  for (int i = 0; rtc::SafeLt(i, expected.size()); ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_NEAR(expected[i], computed[i], tolerance);
+  }
+}
+
+std::unique_ptr<FileReader> CreatePcmSamplesReader() {
+  return std::make_unique<FloatFileReader<int16_t>>(
+      /*filename=*/test::ResourcePath("audio_processing/agc2/rnn_vad/samples",
+                                      "pcm"));
+}
+
+ChunksFileReader CreatePitchBuffer24kHzReader() {
+  auto reader = std::make_unique<FloatFileReader<float>>(
+      /*filename=*/test::ResourcePath(
+          "audio_processing/agc2/rnn_vad/pitch_buf_24k", "dat"));
+  const int num_chunks = rtc::CheckedDivExact(reader->size(), kBufSize24kHz);
+  return {/*chunk_size=*/kBufSize24kHz, num_chunks, std::move(reader)};
+}
+
+ChunksFileReader CreateLpResidualAndPitchInfoReader() {
+  constexpr int kPitchInfoSize = 2;  // Pitch period and strength.
+  constexpr int kChunkSize = kBufSize24kHz + kPitchInfoSize;
+  auto reader = std::make_unique<FloatFileReader<float>>(
+      /*filename=*/test::ResourcePath(
+          "audio_processing/agc2/rnn_vad/pitch_lp_res", "dat"));
+  const int num_chunks = rtc::CheckedDivExact(reader->size(), kChunkSize);
+  return {kChunkSize, num_chunks, std::move(reader)};
+}
+
+std::unique_ptr<FileReader> CreateGruInputReader() {
+  return std::make_unique<FloatFileReader<float>>(
+      /*filename=*/test::ResourcePath("audio_processing/agc2/rnn_vad/gru_in",
+                                      "dat"));
+}
+
+std::unique_ptr<FileReader> CreateVadProbsReader() {
+  return std::make_unique<FloatFileReader<float>>(
+      /*filename=*/test::ResourcePath("audio_processing/agc2/rnn_vad/vad_prob",
+                                      "dat"));
+}
+
+PitchTestData::PitchTestData() {
+  FloatFileReader<float> reader(
+      /*filename=*/ResourcePath(
+          "audio_processing/agc2/rnn_vad/pitch_search_int", "dat"));
+  reader.ReadChunk(pitch_buffer_24k_);
+  reader.ReadChunk(square_energies_24k_);
+  reader.ReadChunk(auto_correlation_12k_);
+  // Reverse the order of the squared energy values.
+  // Required after the WebRTC CL 191703 which switched to forward computation.
+  std::reverse(square_energies_24k_.begin(), square_energies_24k_.end());
+}
+
+PitchTestData::~PitchTestData() = default;
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.h
new file mode 100644
index 0000000000..e64b7b7ecd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/test_utils.h
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_TEST_UTILS_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_TEST_UTILS_H_
+
+#include <array>
+#include <fstream>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+constexpr float kFloatMin = std::numeric_limits<float>::min();
+
+// Fails for every pair from two equally sized rtc::ArrayView<float> views such
+// that the values in the pair do not match.
+void ExpectEqualFloatArray(rtc::ArrayView<const float> expected,
+                           rtc::ArrayView<const float> computed);
+
+// Fails for every pair from two equally sized rtc::ArrayView<float> views such
+// that their absolute error is above a given threshold.
+void ExpectNearAbsolute(rtc::ArrayView<const float> expected,
+                        rtc::ArrayView<const float> computed,
+                        float tolerance);
+
+// File reader interface.
+class FileReader {
+ public:
+  virtual ~FileReader() = default;
+  // Number of values in the file.
+  virtual int size() const = 0;
+  // Reads `dst.size()` float values into `dst`, advances the internal file
+  // position according to the number of read bytes and returns true if the
+  // values are correctly read. If the number of remaining bytes in the file is
+  // not sufficient to read `dst.size()` float values, `dst` is partially
+  // modified and false is returned.
+  virtual bool ReadChunk(rtc::ArrayView<float> dst) = 0;
+  // Reads a single float value, advances the internal file position according
+  // to the number of read bytes and returns true if the value is correctly
+  // read. If the number of remaining bytes in the file is not sufficient to
+  // read one float, `dst` is not modified and false is returned.
+  virtual bool ReadValue(float& dst) = 0;
+  // Advances the internal file position by `hop` float values.
+  virtual void SeekForward(int hop) = 0;
+  // Resets the internal file position to BOF.
+  virtual void SeekBeginning() = 0;
+};
+
+// File reader for files that contain `num_chunks` chunks with size equal to
+// `chunk_size`.
+struct ChunksFileReader {
+  const int chunk_size;
+  const int num_chunks;
+  std::unique_ptr<FileReader> reader;
+};
+
+// Creates a reader for the PCM S16 samples file.
+std::unique_ptr<FileReader> CreatePcmSamplesReader();
+
+// Creates a reader for the 24 kHz pitch buffer test data.
+ChunksFileReader CreatePitchBuffer24kHzReader();
+
+// Creates a reader for the LP residual and pitch information test data.
+ChunksFileReader CreateLpResidualAndPitchInfoReader();
+
+// Creates a reader for the sequence of GRU input vectors.
+std::unique_ptr<FileReader> CreateGruInputReader();
+
+// Creates a reader for the VAD probabilities test data.
+std::unique_ptr<FileReader> CreateVadProbsReader();
+
+// Class to retrieve a test pitch buffer content and the expected output for the
+// analysis steps.
+class PitchTestData {
+ public:
+  PitchTestData();
+  ~PitchTestData();
+  rtc::ArrayView<const float, kBufSize24kHz> PitchBuffer24kHzView() const {
+    return pitch_buffer_24k_;
+  }
+  rtc::ArrayView<const float, kRefineNumLags24kHz> SquareEnergies24kHzView()
+      const {
+    return square_energies_24k_;
+  }
+  rtc::ArrayView<const float, kNumLags12kHz> AutoCorrelation12kHzView() const {
+    return auto_correlation_12k_;
+  }
+
+ private:
+  std::array<float, kBufSize24kHz> pitch_buffer_24k_;
+  std::array<float, kRefineNumLags24kHz> square_energies_24k_;
+  std::array<float, kNumLags12kHz> auto_correlation_12k_;
+};
+
+// Writer for binary files.
+class FileWriter {
+ public:
+  explicit FileWriter(absl::string_view file_path)
+      : os_(std::string(file_path), std::ios::binary) {}
+  FileWriter(const FileWriter&) = delete;
+  FileWriter& operator=(const FileWriter&) = delete;
+  ~FileWriter() = default;
+  void WriteChunk(rtc::ArrayView<const float> value) {
+    const std::streamsize bytes_to_write = value.size() * sizeof(float);
+    os_.write(reinterpret_cast<const char*>(value.data()), bytes_to_write);
+  }
+
+ private:
+  std::ofstream os_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_TEST_UTILS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math.h b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math.h
new file mode 100644
index 0000000000..47f681196a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_VECTOR_MATH_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_VECTOR_MATH_H_
+
+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_HAS_NEON)
+#include <arm_neon.h>
+#endif
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include <emmintrin.h>
+#endif
+
+#include <numeric>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Provides optimizations for mathematical operations having vectors as
+// operand(s).
+class VectorMath {
+ public:
+  explicit VectorMath(AvailableCpuFeatures cpu_features)
+      : cpu_features_(cpu_features) {}
+
+  // Computes the dot product between two equally sized vectors.
+  float DotProduct(rtc::ArrayView<const float> x,
+                   rtc::ArrayView<const float> y) const {
+    RTC_DCHECK_EQ(x.size(), y.size());
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+    if (cpu_features_.avx2) {
+      return DotProductAvx2(x, y);
+    } else if (cpu_features_.sse2) {
+      __m128 accumulator = _mm_setzero_ps();
+      constexpr int kBlockSizeLog2 = 2;
+      constexpr int kBlockSize = 1 << kBlockSizeLog2;
+      const int incomplete_block_index = (x.size() >> kBlockSizeLog2)
+                                         << kBlockSizeLog2;
+      for (int i = 0; i < incomplete_block_index; i += kBlockSize) {
+        RTC_DCHECK_LE(i + kBlockSize, x.size());
+        const __m128 x_i = _mm_loadu_ps(&x[i]);
+        const __m128 y_i = _mm_loadu_ps(&y[i]);
+        // Multiply-add.
+        const __m128 z_j = _mm_mul_ps(x_i, y_i);
+        accumulator = _mm_add_ps(accumulator, z_j);
+      }
+      // Reduce `accumulator` by addition.
+      __m128 high = _mm_movehl_ps(accumulator, accumulator);
+      accumulator = _mm_add_ps(accumulator, high);
+      high = _mm_shuffle_ps(accumulator, accumulator, 1);
+      accumulator = _mm_add_ps(accumulator, high);
+      float dot_product = _mm_cvtss_f32(accumulator);
+      // Add the result for the last block if incomplete.
+      for (int i = incomplete_block_index;
+           i < rtc::dchecked_cast<int>(x.size()); ++i) {
+        dot_product += x[i] * y[i];
+      }
+      return dot_product;
+    }
+#elif defined(WEBRTC_HAS_NEON) && defined(WEBRTC_ARCH_ARM64)
+    if (cpu_features_.neon) {
+      float32x4_t accumulator = vdupq_n_f32(0.f);
+      constexpr int kBlockSizeLog2 = 2;
+      constexpr int kBlockSize = 1 << kBlockSizeLog2;
+      const int incomplete_block_index = (x.size() >> kBlockSizeLog2)
+                                         << kBlockSizeLog2;
+      for (int i = 0; i < incomplete_block_index; i += kBlockSize) {
+        RTC_DCHECK_LE(i + kBlockSize, x.size());
+        const float32x4_t x_i = vld1q_f32(&x[i]);
+        const float32x4_t y_i = vld1q_f32(&y[i]);
+        accumulator = vfmaq_f32(accumulator, x_i, y_i);
+      }
+      // Reduce `accumulator` by addition.
+      const float32x2_t tmp =
+          vpadd_f32(vget_low_f32(accumulator), vget_high_f32(accumulator));
+      float dot_product = vget_lane_f32(vpadd_f32(tmp, vrev64_f32(tmp)), 0);
+      // Add the result for the last block if incomplete.
+      for (int i = incomplete_block_index;
+           i < rtc::dchecked_cast<int>(x.size()); ++i) {
+        dot_product += x[i] * y[i];
+      }
+      return dot_product;
+    }
+#endif
+    return std::inner_product(x.begin(), x.end(), y.begin(), 0.f);
+  }
+
+ private:
+  float DotProductAvx2(rtc::ArrayView<const float> x,
+                       rtc::ArrayView<const float> y) const;
+
+  const AvailableCpuFeatures cpu_features_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_VECTOR_MATH_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc
new file mode 100644
index 0000000000..e4d246d9ab
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
+
+#include <immintrin.h>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+float VectorMath::DotProductAvx2(rtc::ArrayView<const float> x,
+                                 rtc::ArrayView<const float> y) const {
+  RTC_DCHECK(cpu_features_.avx2);
+  RTC_DCHECK_EQ(x.size(), y.size());
+  __m256 accumulator = _mm256_setzero_ps();
+  constexpr int kBlockSizeLog2 = 3;
+  constexpr int kBlockSize = 1 << kBlockSizeLog2;
+  const int incomplete_block_index = (x.size() >> kBlockSizeLog2)
+                                     << kBlockSizeLog2;
+  for (int i = 0; i < incomplete_block_index; i += kBlockSize) {
+    RTC_DCHECK_LE(i + kBlockSize, x.size());
+    const __m256 x_i = _mm256_loadu_ps(&x[i]);
+    const __m256 y_i = _mm256_loadu_ps(&y[i]);
+    accumulator = _mm256_fmadd_ps(x_i, y_i, accumulator);
+  }
+  // Reduce `accumulator` by addition.
+  __m128 high = _mm256_extractf128_ps(accumulator, 1);
+  __m128 low = _mm256_extractf128_ps(accumulator, 0);
+  low = _mm_add_ps(high, low);
+  high = _mm_movehl_ps(high, low);
+  low = _mm_add_ps(high, low);
+  high = _mm_shuffle_ps(low, low, 1);
+  low = _mm_add_ss(high, low);
+  float dot_product = _mm_cvtss_f32(low);
+  // Add the result for the last block if incomplete.
+  for (int i = incomplete_block_index; i < rtc::dchecked_cast<int>(x.size());
+       ++i) {
+    dot_product += x[i] * y[i];
+  }
+  return dot_product;
+}
+
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build
new file mode 100644
index 0000000000..275c512cf6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2_gn/moz.build
@@ -0,0 +1,185 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+CXXFLAGS += [
+    "-mavx2",
+    "-mfma"
+]
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_AVX2"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+Library("vector_math_avx2_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build
new file mode 100644
index 0000000000..263ec679e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_gn/moz.build
@@ -0,0 +1,216 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("vector_math_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_unittest.cc
new file mode 100644
index 0000000000..45fd65d61e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/vector_math_unittest.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
+
+#include <vector>
+
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rnn_vad {
+namespace {
+
+constexpr int kSizeOfX = 19;
+constexpr float kX[kSizeOfX] = {
+    0.31593041f, 0.9350786f,   -0.25252445f, -0.86956251f, -0.9673632f,
+    0.54571901f, -0.72504495f, -0.79509912f, -0.25525012f, -0.73340473f,
+    0.15747377f, -0.04370565f, 0.76135145f,  -0.57239645f, 0.68616848f,
+    0.3740298f,  0.34710799f,  -0.92207423f, 0.10738454f};
+constexpr int kSizeOfXSubSpan = 16;
+static_assert(kSizeOfXSubSpan < kSizeOfX, "");
+constexpr float kEnergyOfX = 7.315563958160327f;
+constexpr float kEnergyOfXSubspan = 6.333327669592963f;
+
+class VectorMathParametrization
+    : public ::testing::TestWithParam<AvailableCpuFeatures> {};
+
+TEST_P(VectorMathParametrization, TestDotProduct) {
+  VectorMath vector_math(/*cpu_features=*/GetParam());
+  EXPECT_FLOAT_EQ(vector_math.DotProduct(kX, kX), kEnergyOfX);
+  EXPECT_FLOAT_EQ(
+      vector_math.DotProduct({kX, kSizeOfXSubSpan}, {kX, kSizeOfXSubSpan}),
+      kEnergyOfXSubspan);
+}
+
+// Finds the relevant CPU features combinations to test.
+std::vector<AvailableCpuFeatures> GetCpuFeaturesToTest() {
+  std::vector<AvailableCpuFeatures> v;
+  v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/false});
+  AvailableCpuFeatures available = GetAvailableCpuFeatures();
+  if (available.avx2) {
+    v.push_back({/*sse2=*/false, /*avx2=*/true, /*neon=*/false});
+  }
+  if (available.sse2) {
+    v.push_back({/*sse2=*/true, /*avx2=*/false, /*neon=*/false});
+  }
+  if (available.neon) {
+    v.push_back({/*sse2=*/false, /*avx2=*/false, /*neon=*/true});
+  }
+  return v;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RnnVadTest,
+    VectorMathParametrization,
+    ::testing::ValuesIn(GetCpuFeaturesToTest()),
+    [](const ::testing::TestParamInfo<AvailableCpuFeatures>& info) {
+      return info.param.ToString();
+    });
+
+}  // namespace
+}  // namespace rnn_vad
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector.cc b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector.cc
new file mode 100644
index 0000000000..961baf4cd3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector.cc
@@ -0,0 +1,183 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/saturation_protector.h"
+
+#include <memory>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/agc2/saturation_protector_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kPeakEnveloperSuperFrameLengthMs = 400;
+constexpr float kMinMarginDb = 12.0f;
+constexpr float kMaxMarginDb = 25.0f;
+constexpr float kAttack = 0.9988493699365052f;
+constexpr float kDecay = 0.9997697679981565f;
+
+// Saturation protector state. Defined outside of `SaturationProtectorImpl` to
+// implement check-point and restore ops.
+struct SaturationProtectorState {
+  bool operator==(const SaturationProtectorState& s) const {
+    return headroom_db == s.headroom_db &&
+           peak_delay_buffer == s.peak_delay_buffer &&
+           max_peaks_dbfs == s.max_peaks_dbfs &&
+           time_since_push_ms == s.time_since_push_ms;
+  }
+  inline bool operator!=(const SaturationProtectorState& s) const {
+    return !(*this == s);
+  }
+
+  float headroom_db;
+  SaturationProtectorBuffer peak_delay_buffer;
+  float max_peaks_dbfs;
+  int time_since_push_ms;  // Time since the last ring buffer push operation.
+};
+
+// Resets the saturation protector state.
+void ResetSaturationProtectorState(float initial_headroom_db,
+                                   SaturationProtectorState& state) {
+  state.headroom_db = initial_headroom_db;
+  state.peak_delay_buffer.Reset();
+  state.max_peaks_dbfs = kMinLevelDbfs;
+  state.time_since_push_ms = 0;
+}
+
+// Updates `state` by analyzing the estimated speech level `speech_level_dbfs`
+// and the peak level `peak_dbfs` for an observed frame. `state` must not be
+// modified without calling this function.
+void UpdateSaturationProtectorState(float peak_dbfs,
+                                    float speech_level_dbfs,
+                                    SaturationProtectorState& state) {
+  // Get the max peak over `kPeakEnveloperSuperFrameLengthMs` ms.
+  state.max_peaks_dbfs = std::max(state.max_peaks_dbfs, peak_dbfs);
+  state.time_since_push_ms += kFrameDurationMs;
+  if (rtc::SafeGt(state.time_since_push_ms, kPeakEnveloperSuperFrameLengthMs)) {
+    // Push `max_peaks_dbfs` back into the ring buffer.
+    state.peak_delay_buffer.PushBack(state.max_peaks_dbfs);
+    // Reset.
+    state.max_peaks_dbfs = kMinLevelDbfs;
+    state.time_since_push_ms = 0;
+  }
+
+  // Update the headroom by comparing the estimated speech level and the delayed
+  // max speech peak.
+  const float delayed_peak_dbfs =
+      state.peak_delay_buffer.Front().value_or(state.max_peaks_dbfs);
+  const float difference_db = delayed_peak_dbfs - speech_level_dbfs;
+  if (difference_db > state.headroom_db) {
+    // Attack.
+    state.headroom_db =
+        state.headroom_db * kAttack + difference_db * (1.0f - kAttack);
+  } else {
+    // Decay.
+    state.headroom_db =
+        state.headroom_db * kDecay + difference_db * (1.0f - kDecay);
+  }
+
+  state.headroom_db =
+      rtc::SafeClamp<float>(state.headroom_db, kMinMarginDb, kMaxMarginDb);
+}
+
+// Saturation protector which recommends a headroom based on the recent peaks.
+class SaturationProtectorImpl : public SaturationProtector {
+ public:
+  explicit SaturationProtectorImpl(float initial_headroom_db,
+                                   int adjacent_speech_frames_threshold,
+                                   ApmDataDumper* apm_data_dumper)
+      : apm_data_dumper_(apm_data_dumper),
+        initial_headroom_db_(initial_headroom_db),
+        adjacent_speech_frames_threshold_(adjacent_speech_frames_threshold) {
+    Reset();
+  }
+  SaturationProtectorImpl(const SaturationProtectorImpl&) = delete;
+  SaturationProtectorImpl& operator=(const SaturationProtectorImpl&) = delete;
+  ~SaturationProtectorImpl() = default;
+
+  float HeadroomDb() override { return headroom_db_; }
+
+  void Analyze(float speech_probability,
+               float peak_dbfs,
+               float speech_level_dbfs) override {
+    if (speech_probability < kVadConfidenceThreshold) {
+      // Not a speech frame.
+      if (adjacent_speech_frames_threshold_ > 1) {
+        // When two or more adjacent speech frames are required in order to
+        // update the state, we need to decide whether to discard or confirm the
+        // updates based on the speech sequence length.
+        if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) {
+          // First non-speech frame after a long enough sequence of speech
+          // frames. Update the reliable state.
+          reliable_state_ = preliminary_state_;
+        } else if (num_adjacent_speech_frames_ > 0) {
+          // First non-speech frame after a too short sequence of speech frames.
+          // Reset to the last reliable state.
+          preliminary_state_ = reliable_state_;
+        }
+      }
+      num_adjacent_speech_frames_ = 0;
+    } else {
+      // Speech frame observed.
+      num_adjacent_speech_frames_++;
+
+      // Update preliminary level estimate.
+      UpdateSaturationProtectorState(peak_dbfs, speech_level_dbfs,
+                                     preliminary_state_);
+
+      if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) {
+        // `preliminary_state_` is now reliable. Update the headroom.
+        headroom_db_ = preliminary_state_.headroom_db;
+      }
+    }
+    DumpDebugData();
+  }
+
+  void Reset() override {
+    num_adjacent_speech_frames_ = 0;
+    headroom_db_ = initial_headroom_db_;
+    ResetSaturationProtectorState(initial_headroom_db_, preliminary_state_);
+    ResetSaturationProtectorState(initial_headroom_db_, reliable_state_);
+  }
+
+ private:
+  void DumpDebugData() {
+    apm_data_dumper_->DumpRaw(
+        "agc2_saturation_protector_preliminary_max_peak_dbfs",
+        preliminary_state_.max_peaks_dbfs);
+    apm_data_dumper_->DumpRaw(
+        "agc2_saturation_protector_reliable_max_peak_dbfs",
+        reliable_state_.max_peaks_dbfs);
+  }
+
+  ApmDataDumper* const apm_data_dumper_;
+  const float initial_headroom_db_;
+  const int adjacent_speech_frames_threshold_;
+  int num_adjacent_speech_frames_;
+  float headroom_db_;
+  SaturationProtectorState preliminary_state_;
+  SaturationProtectorState reliable_state_;
+};
+
+}  // namespace
+
+std::unique_ptr<SaturationProtector> CreateSaturationProtector(
+    float initial_headroom_db,
+    int adjacent_speech_frames_threshold,
+    ApmDataDumper* apm_data_dumper) {
+  return std::make_unique<SaturationProtectorImpl>(
+      initial_headroom_db, adjacent_speech_frames_threshold, apm_data_dumper);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector.h b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector.h
new file mode 100644
index 0000000000..ef22145d5f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
+
+#include <memory>
+
+namespace webrtc {
+class ApmDataDumper;
+
+// Saturation protector. Analyzes peak levels and recommends a headroom to
+// reduce the chances of clipping.
+class SaturationProtector {
+ public:
+  virtual ~SaturationProtector() = default;
+
+  // Returns the recommended headroom in dB.
+  virtual float HeadroomDb() = 0;
+
+  // Analyzes the peak level of a 10 ms frame along with its speech probability
+  // and the current speech level estimate to update the recommended headroom.
+  virtual void Analyze(float speech_probability,
+                       float peak_dbfs,
+                       float speech_level_dbfs) = 0;
+
+  // Resets the internal state.
+  virtual void Reset() = 0;
+};
+
+// Creates a saturation protector that starts at `initial_headroom_db`.
+std::unique_ptr<SaturationProtector> CreateSaturationProtector(
+    float initial_headroom_db,
+    int adjacent_speech_frames_threshold,
+    ApmDataDumper* apm_data_dumper);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc
new file mode 100644
index 0000000000..41efdad2c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/saturation_protector_buffer.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_compare.h"
+
+namespace webrtc {
+
+SaturationProtectorBuffer::SaturationProtectorBuffer() = default;
+
+SaturationProtectorBuffer::~SaturationProtectorBuffer() = default;
+
+bool SaturationProtectorBuffer::operator==(
+    const SaturationProtectorBuffer& b) const {
+  RTC_DCHECK_LE(size_, buffer_.size());
+  RTC_DCHECK_LE(b.size_, b.buffer_.size());
+  if (size_ != b.size_) {
+    return false;
+  }
+  for (int i = 0, i0 = FrontIndex(), i1 = b.FrontIndex(); i < size_;
+       ++i, ++i0, ++i1) {
+    if (buffer_[i0 % buffer_.size()] != b.buffer_[i1 % b.buffer_.size()]) {
+      return false;
+    }
+  }
+  return true;
+}
+
+int SaturationProtectorBuffer::Capacity() const {
+  return buffer_.size();
+}
+
+int SaturationProtectorBuffer::Size() const {
+  return size_;
+}
+
+void SaturationProtectorBuffer::Reset() {
+  next_ = 0;
+  size_ = 0;
+}
+
+void SaturationProtectorBuffer::PushBack(float v) {
+  RTC_DCHECK_GE(next_, 0);
+  RTC_DCHECK_GE(size_, 0);
+  RTC_DCHECK_LT(next_, buffer_.size());
+  RTC_DCHECK_LE(size_, buffer_.size());
+  buffer_[next_++] = v;
+  if (rtc::SafeEq(next_, buffer_.size())) {
+    next_ = 0;
+  }
+  if (rtc::SafeLt(size_, buffer_.size())) {
+    size_++;
+  }
+}
+
+absl::optional<float> SaturationProtectorBuffer::Front() const {
+  if (size_ == 0) {
+    return absl::nullopt;
+  }
+  RTC_DCHECK_LT(FrontIndex(), buffer_.size());
+  return buffer_[FrontIndex()];
+}
+
+int SaturationProtectorBuffer::FrontIndex() const {
+  return rtc::SafeEq(size_, buffer_.size()) ? next_ : 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.h b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.h
new file mode 100644
index 0000000000..e17d0998c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_
+
+#include <array>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+
+namespace webrtc {
+
+// Ring buffer for the saturation protector which only supports (i) push back
+// and (ii) read oldest item.
+class SaturationProtectorBuffer {
+ public:
+  SaturationProtectorBuffer();
+  ~SaturationProtectorBuffer();
+
+  bool operator==(const SaturationProtectorBuffer& b) const;
+  inline bool operator!=(const SaturationProtectorBuffer& b) const {
+    return !(*this == b);
+  }
+
+  // Maximum number of values that the buffer can contain.
+  int Capacity() const;
+
+  // Number of values in the buffer.
+  int Size() const;
+
+  void Reset();
+
+  // Pushes back `v`. If the buffer is full, the oldest value is replaced.
+  void PushBack(float v);
+
+  // Returns the oldest item in the buffer. Returns an empty value if the
+  // buffer is empty.
+  absl::optional<float> Front() const;
+
+ private:
+  int FrontIndex() const;
+  // `buffer_` has `size_` elements (up to the size of `buffer_`) and `next_` is
+  // the position where the next new value is written in `buffer_`.
+  std::array<float, kSaturationProtectorBufferSize> buffer_;
+  int next_ = 0;
+  int size_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer_unittest.cc
new file mode 100644
index 0000000000..22187bf027
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer_unittest.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/saturation_protector_buffer.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Eq;
+using ::testing::Optional;
+
+TEST(GainController2SaturationProtectorBuffer, Init) {
+  SaturationProtectorBuffer b;
+  EXPECT_EQ(b.Size(), 0);
+  EXPECT_FALSE(b.Front().has_value());
+}
+
+TEST(GainController2SaturationProtectorBuffer, PushBack) {
+  SaturationProtectorBuffer b;
+  constexpr float kValue = 123.0f;
+  b.PushBack(kValue);
+  EXPECT_EQ(b.Size(), 1);
+  EXPECT_THAT(b.Front(), Optional(Eq(kValue)));
+}
+
+TEST(GainController2SaturationProtectorBuffer, Reset) {
+  SaturationProtectorBuffer b;
+  b.PushBack(123.0f);
+  b.Reset();
+  EXPECT_EQ(b.Size(), 0);
+  EXPECT_FALSE(b.Front().has_value());
+}
+
+// Checks that the front value does not change until the ring buffer gets full.
+TEST(GainController2SaturationProtectorBuffer, FrontUntilBufferIsFull) {
+  SaturationProtectorBuffer b;
+  constexpr float kValue = 123.0f;
+  b.PushBack(kValue);
+  for (int i = 1; i < b.Capacity(); ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_THAT(b.Front(), Optional(Eq(kValue)));
+    b.PushBack(kValue + i);
+  }
+}
+
+// Checks that when the buffer is full it behaves as a shift register.
+TEST(GainController2SaturationProtectorBuffer, FrontIsDelayed) {
+  SaturationProtectorBuffer b;
+  // Fill the buffer.
+  for (int i = 0; i < b.Capacity(); ++i) {
+    b.PushBack(i);
+  }
+  // The ring buffer should now behave as a shift register with a delay equal to
+  // its capacity.
+  for (int i = b.Capacity(); i < 2 * b.Capacity() + 1; ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_THAT(b.Front(), Optional(Eq(i - b.Capacity())));
+    b.PushBack(i);
+  }
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_gn/moz.build
new file mode 100644
index 0000000000..a1e6f309bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_gn/moz.build
@@ -0,0 +1,234 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector.cc",
+    "/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_buffer.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("saturation_protector_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_unittest.cc
new file mode 100644
index 0000000000..3b104be8cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/saturation_protector_unittest.cc
@@ -0,0 +1,140 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/saturation_protector.h"
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kInitialHeadroomDb = 20.0f;
+constexpr int kNoAdjacentSpeechFramesRequired = 1;
+constexpr float kMaxSpeechProbability = 1.0f;
+
+// Calls `Analyze(speech_probability, peak_dbfs, speech_level_dbfs)`
+// `num_iterations` times on `saturation_protector` and return the largest
+// headroom difference between two consecutive calls.
+float RunOnConstantLevel(int num_iterations,
+                         float speech_probability,
+                         float peak_dbfs,
+                         float speech_level_dbfs,
+                         SaturationProtector& saturation_protector) {
+  float last_headroom = saturation_protector.HeadroomDb();
+  float max_difference = 0.0f;
+  for (int i = 0; i < num_iterations; ++i) {
+    saturation_protector.Analyze(speech_probability, peak_dbfs,
+                                 speech_level_dbfs);
+    const float new_headroom = saturation_protector.HeadroomDb();
+    max_difference =
+        std::max(max_difference, std::fabs(new_headroom - last_headroom));
+    last_headroom = new_headroom;
+  }
+  return max_difference;
+}
+
+// Checks that the returned headroom value is correctly reset.
+TEST(GainController2SaturationProtector, Reset) {
+  ApmDataDumper apm_data_dumper(0);
+  auto saturation_protector = CreateSaturationProtector(
+      kInitialHeadroomDb, kNoAdjacentSpeechFramesRequired, &apm_data_dumper);
+  const float initial_headroom_db = saturation_protector->HeadroomDb();
+  RunOnConstantLevel(/*num_iterations=*/10, kMaxSpeechProbability,
+                     /*peak_dbfs=*/0.0f,
+                     /*speech_level_dbfs=*/-10.0f, *saturation_protector);
+  // Make sure that there are side-effects.
+  ASSERT_NE(initial_headroom_db, saturation_protector->HeadroomDb());
+  saturation_protector->Reset();
+  EXPECT_EQ(initial_headroom_db, saturation_protector->HeadroomDb());
+}
+
+// Checks that the estimate converges to the ratio between peaks and level
+// estimator values after a while.
+TEST(GainController2SaturationProtector, EstimatesCrestRatio) {
+  constexpr int kNumIterations = 2000;
+  constexpr float kPeakLevelDbfs = -20.0f;
+  constexpr float kCrestFactorDb = kInitialHeadroomDb + 1.0f;
+  constexpr float kSpeechLevelDbfs = kPeakLevelDbfs - kCrestFactorDb;
+  const float kMaxDifferenceDb =
+      0.5f * std::fabs(kInitialHeadroomDb - kCrestFactorDb);
+
+  ApmDataDumper apm_data_dumper(0);
+  auto saturation_protector = CreateSaturationProtector(
+      kInitialHeadroomDb, kNoAdjacentSpeechFramesRequired, &apm_data_dumper);
+  RunOnConstantLevel(kNumIterations, kMaxSpeechProbability, kPeakLevelDbfs,
+                     kSpeechLevelDbfs, *saturation_protector);
+  EXPECT_NEAR(saturation_protector->HeadroomDb(), kCrestFactorDb,
+              kMaxDifferenceDb);
+}
+
+// Checks that the headroom does not change too quickly.
+TEST(GainController2SaturationProtector, ChangeSlowly) {
+  constexpr int kNumIterations = 1000;
+  constexpr float kPeakLevelDbfs = -20.f;
+  constexpr float kCrestFactorDb = kInitialHeadroomDb - 5.f;
+  constexpr float kOtherCrestFactorDb = kInitialHeadroomDb;
+  constexpr float kSpeechLevelDbfs = kPeakLevelDbfs - kCrestFactorDb;
+  constexpr float kOtherSpeechLevelDbfs = kPeakLevelDbfs - kOtherCrestFactorDb;
+
+  ApmDataDumper apm_data_dumper(0);
+  auto saturation_protector = CreateSaturationProtector(
+      kInitialHeadroomDb, kNoAdjacentSpeechFramesRequired, &apm_data_dumper);
+  float max_difference_db =
+      RunOnConstantLevel(kNumIterations, kMaxSpeechProbability, kPeakLevelDbfs,
+                         kSpeechLevelDbfs, *saturation_protector);
+  max_difference_db = std::max(
+      RunOnConstantLevel(kNumIterations, kMaxSpeechProbability, kPeakLevelDbfs,
+                         kOtherSpeechLevelDbfs, *saturation_protector),
+      max_difference_db);
+  constexpr float kMaxChangeSpeedDbPerSecond = 0.5f;  // 1 db / 2 seconds.
+  EXPECT_LE(max_difference_db,
+            kMaxChangeSpeedDbPerSecond / 1000 * kFrameDurationMs);
+}
+
+class SaturationProtectorParametrization
+    : public ::testing::TestWithParam<int> {
+ protected:
+  int adjacent_speech_frames_threshold() const { return GetParam(); }
+};
+
+TEST_P(SaturationProtectorParametrization, DoNotAdaptToShortSpeechSegments) {
+  ApmDataDumper apm_data_dumper(0);
+  auto saturation_protector = CreateSaturationProtector(
+      kInitialHeadroomDb, adjacent_speech_frames_threshold(), &apm_data_dumper);
+  const float initial_headroom_db = saturation_protector->HeadroomDb();
+  RunOnConstantLevel(/*num_iterations=*/adjacent_speech_frames_threshold() - 1,
+                     kMaxSpeechProbability,
+                     /*peak_dbfs=*/0.0f,
+                     /*speech_level_dbfs=*/-10.0f, *saturation_protector);
+  // No adaptation expected.
+  EXPECT_EQ(initial_headroom_db, saturation_protector->HeadroomDb());
+}
+
+TEST_P(SaturationProtectorParametrization, AdaptToEnoughSpeechSegments) {
+  ApmDataDumper apm_data_dumper(0);
+  auto saturation_protector = CreateSaturationProtector(
+      kInitialHeadroomDb, adjacent_speech_frames_threshold(), &apm_data_dumper);
+  const float initial_headroom_db = saturation_protector->HeadroomDb();
+  RunOnConstantLevel(/*num_iterations=*/adjacent_speech_frames_threshold() + 1,
+                     kMaxSpeechProbability,
+                     /*peak_dbfs=*/0.0f,
+                     /*speech_level_dbfs=*/-10.0f, *saturation_protector);
+  // Adaptation expected.
+  EXPECT_NE(initial_headroom_db, saturation_protector->HeadroomDb());
+}
+
+INSTANTIATE_TEST_SUITE_P(GainController2,
+                         SaturationProtectorParametrization,
+                         ::testing::Values(2, 9, 17));
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.cc b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.cc
new file mode 100644
index 0000000000..7bf3252116
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.cc
@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/speech_level_estimator.h"
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+
+float ClampLevelEstimateDbfs(float level_estimate_dbfs) {
+  return rtc::SafeClamp<float>(level_estimate_dbfs, -90.0f, 30.0f);
+}
+
+// Returns the initial speech level estimate needed to apply the initial gain.
+float GetInitialSpeechLevelEstimateDbfs(
+    const AudioProcessing::Config::GainController2::AdaptiveDigital& config) {
+  return ClampLevelEstimateDbfs(-kSaturationProtectorInitialHeadroomDb -
+                                config.initial_gain_db - config.headroom_db);
+}
+
+}  // namespace
+
+bool SpeechLevelEstimator::LevelEstimatorState::operator==(
+    const SpeechLevelEstimator::LevelEstimatorState& b) const {
+  return time_to_confidence_ms == b.time_to_confidence_ms &&
+         level_dbfs.numerator == b.level_dbfs.numerator &&
+         level_dbfs.denominator == b.level_dbfs.denominator;
+}
+
+float SpeechLevelEstimator::LevelEstimatorState::Ratio::GetRatio() const {
+  RTC_DCHECK_NE(denominator, 0.f);
+  return numerator / denominator;
+}
+
+SpeechLevelEstimator::SpeechLevelEstimator(
+    ApmDataDumper* apm_data_dumper,
+    const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+    int adjacent_speech_frames_threshold)
+    : apm_data_dumper_(apm_data_dumper),
+      initial_speech_level_dbfs_(GetInitialSpeechLevelEstimateDbfs(config)),
+      adjacent_speech_frames_threshold_(adjacent_speech_frames_threshold),
+      level_dbfs_(initial_speech_level_dbfs_),
+      // TODO(bugs.webrtc.org/7494): Remove init below when AGC2 input volume
+      // controller temporal dependency removed.
+      is_confident_(false) {
+  RTC_DCHECK(apm_data_dumper_);
+  RTC_DCHECK_GE(adjacent_speech_frames_threshold_, 1);
+  Reset();
+}
+
+void SpeechLevelEstimator::Update(float rms_dbfs,
+                                  float peak_dbfs,
+                                  float speech_probability) {
+  RTC_DCHECK_GT(rms_dbfs, -150.0f);
+  RTC_DCHECK_LT(rms_dbfs, 50.0f);
+  RTC_DCHECK_GT(peak_dbfs, -150.0f);
+  RTC_DCHECK_LT(peak_dbfs, 50.0f);
+  RTC_DCHECK_GE(speech_probability, 0.0f);
+  RTC_DCHECK_LE(speech_probability, 1.0f);
+  if (speech_probability < kVadConfidenceThreshold) {
+    // Not a speech frame.
+    if (adjacent_speech_frames_threshold_ > 1) {
+      // When two or more adjacent speech frames are required in order to update
+      // the state, we need to decide whether to discard or confirm the updates
+      // based on the speech sequence length.
+      if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) {
+        // First non-speech frame after a long enough sequence of speech frames.
+        // Update the reliable state.
+        reliable_state_ = preliminary_state_;
+      } else if (num_adjacent_speech_frames_ > 0) {
+        // First non-speech frame after a too short sequence of speech frames.
+        // Reset to the last reliable state.
+        preliminary_state_ = reliable_state_;
+      }
+    }
+    num_adjacent_speech_frames_ = 0;
+  } else {
+    // Speech frame observed.
+    num_adjacent_speech_frames_++;
+
+    // Update preliminary level estimate.
+    RTC_DCHECK_GE(preliminary_state_.time_to_confidence_ms, 0);
+    const bool buffer_is_full = preliminary_state_.time_to_confidence_ms == 0;
+    if (!buffer_is_full) {
+      preliminary_state_.time_to_confidence_ms -= kFrameDurationMs;
+    }
+    // Weighted average of levels with speech probability as weight.
+    RTC_DCHECK_GT(speech_probability, 0.0f);
+    const float leak_factor = buffer_is_full ? kLevelEstimatorLeakFactor : 1.0f;
+    preliminary_state_.level_dbfs.numerator =
+        preliminary_state_.level_dbfs.numerator * leak_factor +
+        rms_dbfs * speech_probability;
+    preliminary_state_.level_dbfs.denominator =
+        preliminary_state_.level_dbfs.denominator * leak_factor +
+        speech_probability;
+
+    const float level_dbfs = preliminary_state_.level_dbfs.GetRatio();
+
+    if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) {
+      // `preliminary_state_` is now reliable. Update the last level estimation.
+      level_dbfs_ = ClampLevelEstimateDbfs(level_dbfs);
+    }
+  }
+  UpdateIsConfident();
+  DumpDebugData();
+}
+
+void SpeechLevelEstimator::UpdateIsConfident() {
+  if (adjacent_speech_frames_threshold_ == 1) {
+    // Ignore `reliable_state_` when a single frame is enough to update the
+    // level estimate (because it is not used).
+    is_confident_ = preliminary_state_.time_to_confidence_ms == 0;
+    return;
+  }
+  // Once confident, it remains confident.
+  RTC_DCHECK(reliable_state_.time_to_confidence_ms != 0 ||
+             preliminary_state_.time_to_confidence_ms == 0);
+  // During the first long enough speech sequence, `reliable_state_` must be
+  // ignored since `preliminary_state_` is used.
+  is_confident_ =
+      reliable_state_.time_to_confidence_ms == 0 ||
+      (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_ &&
+       preliminary_state_.time_to_confidence_ms == 0);
+}
+
+void SpeechLevelEstimator::Reset() {
+  ResetLevelEstimatorState(preliminary_state_);
+  ResetLevelEstimatorState(reliable_state_);
+  level_dbfs_ = initial_speech_level_dbfs_;
+  num_adjacent_speech_frames_ = 0;
+}
+
+void SpeechLevelEstimator::ResetLevelEstimatorState(
+    LevelEstimatorState& state) const {
+  state.time_to_confidence_ms = kLevelEstimatorTimeToConfidenceMs;
+  state.level_dbfs.numerator = initial_speech_level_dbfs_;
+  state.level_dbfs.denominator = 1.0f;
+}
+
+void SpeechLevelEstimator::DumpDebugData() const {
+  if (!apm_data_dumper_)
+    return;
+  apm_data_dumper_->DumpRaw("agc2_speech_level_dbfs", level_dbfs_);
+  apm_data_dumper_->DumpRaw("agc2_speech_level_is_confident", is_confident_);
+  apm_data_dumper_->DumpRaw(
+      "agc2_adaptive_level_estimator_num_adjacent_speech_frames",
+      num_adjacent_speech_frames_);
+  apm_data_dumper_->DumpRaw(
+      "agc2_adaptive_level_estimator_preliminary_level_estimate_num",
+      preliminary_state_.level_dbfs.numerator);
+  apm_data_dumper_->DumpRaw(
+      "agc2_adaptive_level_estimator_preliminary_level_estimate_den",
+      preliminary_state_.level_dbfs.denominator);
+  apm_data_dumper_->DumpRaw(
+      "agc2_adaptive_level_estimator_preliminary_time_to_confidence_ms",
+      preliminary_state_.time_to_confidence_ms);
+  apm_data_dumper_->DumpRaw(
+      "agc2_adaptive_level_estimator_reliable_time_to_confidence_ms",
+      reliable_state_.time_to_confidence_ms);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.h b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.h
new file mode 100644
index 0000000000..4d9f106ba9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include <type_traits>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+class ApmDataDumper;
+
+// Active speech level estimator based on the analysis of the following
+// framewise properties: RMS level (dBFS), peak level (dBFS), speech
+// probability.
+class SpeechLevelEstimator {
+ public:
+  SpeechLevelEstimator(
+      ApmDataDumper* apm_data_dumper,
+      const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+      int adjacent_speech_frames_threshold);
+  SpeechLevelEstimator(const SpeechLevelEstimator&) = delete;
+  SpeechLevelEstimator& operator=(const SpeechLevelEstimator&) = delete;
+
+  // Updates the level estimation.
+  void Update(float rms_dbfs, float peak_dbfs, float speech_probability);
+  // Returns the estimated speech plus noise level.
+  float level_dbfs() const { return level_dbfs_; }
+  // Returns true if the estimator is confident on its current estimate.
+  bool is_confident() const { return is_confident_; }
+
+  void Reset();
+
+ private:
+  // Part of the level estimator state used for check-pointing and restore ops.
+  struct LevelEstimatorState {
+    bool operator==(const LevelEstimatorState& s) const;
+    inline bool operator!=(const LevelEstimatorState& s) const {
+      return !(*this == s);
+    }
+    // TODO(bugs.webrtc.org/7494): Remove `time_to_confidence_ms` if redundant.
+    int time_to_confidence_ms;
+    struct Ratio {
+      float numerator;
+      float denominator;
+      float GetRatio() const;
+    } level_dbfs;
+  };
+  static_assert(std::is_trivially_copyable<LevelEstimatorState>::value, "");
+
+  void UpdateIsConfident();
+
+  void ResetLevelEstimatorState(LevelEstimatorState& state) const;
+
+  void DumpDebugData() const;
+
+  ApmDataDumper* const apm_data_dumper_;
+
+  const float initial_speech_level_dbfs_;
+  const int adjacent_speech_frames_threshold_;
+  LevelEstimatorState preliminary_state_;
+  LevelEstimatorState reliable_state_;
+  float level_dbfs_;
+  bool is_confident_;
+  int num_adjacent_speech_frames_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_LEVEL_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_gn/moz.build
new file mode 100644
index 0000000000..bb1dbc67b8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("speech_level_estimator_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_unittest.cc
new file mode 100644
index 0000000000..e1c5f85434
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator_unittest.cc
@@ -0,0 +1,207 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/speech_level_estimator.h"
+
+#include <memory>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gunit.h"
+
+namespace webrtc {
+namespace {
+
+using AdaptiveDigitalConfig =
+    AudioProcessing::Config::GainController2::AdaptiveDigital;
+
+// Number of speech frames that the level estimator must observe in order to
+// become confident about the estimated level.
+constexpr int kNumFramesToConfidence =
+    kLevelEstimatorTimeToConfidenceMs / kFrameDurationMs;
+static_assert(kNumFramesToConfidence > 0, "");
+
+constexpr float kConvergenceSpeedTestsLevelTolerance = 0.5f;
+
+// Provides the `vad_level` value `num_iterations` times to `level_estimator`.
+void RunOnConstantLevel(int num_iterations,
+                        float rms_dbfs,
+                        float peak_dbfs,
+                        float speech_probability,
+                        SpeechLevelEstimator& level_estimator) {
+  for (int i = 0; i < num_iterations; ++i) {
+    level_estimator.Update(rms_dbfs, peak_dbfs, speech_probability);
+  }
+}
+
+constexpr float kNoSpeechProbability = 0.0f;
+constexpr float kLowSpeechProbability = kVadConfidenceThreshold / 2.0f;
+constexpr float kMaxSpeechProbability = 1.0f;
+
+// Level estimator with data dumper.
+struct TestLevelEstimator {
+  explicit TestLevelEstimator(int adjacent_speech_frames_threshold)
+      : data_dumper(0),
+        estimator(std::make_unique<SpeechLevelEstimator>(
+            &data_dumper,
+            AdaptiveDigitalConfig{},
+            adjacent_speech_frames_threshold)),
+        initial_speech_level_dbfs(estimator->level_dbfs()),
+        level_rms_dbfs(initial_speech_level_dbfs / 2.0f),
+        level_peak_dbfs(initial_speech_level_dbfs / 3.0f) {
+    RTC_DCHECK_LT(level_rms_dbfs, level_peak_dbfs);
+    RTC_DCHECK_LT(initial_speech_level_dbfs, level_rms_dbfs);
+    RTC_DCHECK_GT(level_rms_dbfs - initial_speech_level_dbfs, 5.0f)
+        << "Adjust `level_rms_dbfs` so that the difference from the initial "
+           "level is wide enough for the tests";
+  }
+  ApmDataDumper data_dumper;
+  std::unique_ptr<SpeechLevelEstimator> estimator;
+  const float initial_speech_level_dbfs;
+  const float level_rms_dbfs;
+  const float level_peak_dbfs;
+};
+
+// Checks that the level estimator converges to a constant input speech level.
+TEST(GainController2SpeechLevelEstimator, LevelStabilizes) {
+  TestLevelEstimator level_estimator(/*adjacent_speech_frames_threshold=*/1);
+  RunOnConstantLevel(/*num_iterations=*/kNumFramesToConfidence,
+                     level_estimator.level_rms_dbfs,
+                     level_estimator.level_peak_dbfs, kMaxSpeechProbability,
+                     *level_estimator.estimator);
+  const float estimated_level_dbfs = level_estimator.estimator->level_dbfs();
+  RunOnConstantLevel(/*num_iterations=*/1, level_estimator.level_rms_dbfs,
+                     level_estimator.level_peak_dbfs, kMaxSpeechProbability,
+                     *level_estimator.estimator);
+  EXPECT_NEAR(level_estimator.estimator->level_dbfs(), estimated_level_dbfs,
+              0.1f);
+}
+
+// Checks that the level controller does not become confident when too few
+// speech frames are observed.
+TEST(GainController2SpeechLevelEstimator, IsNotConfident) {
+  TestLevelEstimator level_estimator(/*adjacent_speech_frames_threshold=*/1);
+  RunOnConstantLevel(/*num_iterations=*/kNumFramesToConfidence / 2,
+                     level_estimator.level_rms_dbfs,
+                     level_estimator.level_peak_dbfs, kMaxSpeechProbability,
+                     *level_estimator.estimator);
+  EXPECT_FALSE(level_estimator.estimator->is_confident());
+}
+
+// Checks that the level controller becomes confident when enough speech frames
+// are observed.
+TEST(GainController2SpeechLevelEstimator, IsConfident) {
+  TestLevelEstimator level_estimator(/*adjacent_speech_frames_threshold=*/1);
+  RunOnConstantLevel(/*num_iterations=*/kNumFramesToConfidence,
+                     level_estimator.level_rms_dbfs,
+                     level_estimator.level_peak_dbfs, kMaxSpeechProbability,
+                     *level_estimator.estimator);
+  EXPECT_TRUE(level_estimator.estimator->is_confident());
+}
+
+// Checks that the estimated level is not affected by the level of non-speech
+// frames.
+TEST(GainController2SpeechLevelEstimator, EstimatorIgnoresNonSpeechFrames) {
+  TestLevelEstimator level_estimator(/*adjacent_speech_frames_threshold=*/1);
+  // Simulate speech.
+  RunOnConstantLevel(/*num_iterations=*/kNumFramesToConfidence,
+                     level_estimator.level_rms_dbfs,
+                     level_estimator.level_peak_dbfs, kMaxSpeechProbability,
+                     *level_estimator.estimator);
+  const float estimated_level_dbfs = level_estimator.estimator->level_dbfs();
+  // Simulate full-scale non-speech.
+  RunOnConstantLevel(/*num_iterations=*/kNumFramesToConfidence,
+                     /*rms_dbfs=*/0.0f, /*peak_dbfs=*/0.0f,
+                     kNoSpeechProbability, *level_estimator.estimator);
+  // No estimated level change is expected.
+  EXPECT_FLOAT_EQ(level_estimator.estimator->level_dbfs(),
+                  estimated_level_dbfs);
+}
+
+// Checks the convergence speed of the estimator before it becomes confident.
+TEST(GainController2SpeechLevelEstimator, ConvergenceSpeedBeforeConfidence) {
+  TestLevelEstimator level_estimator(/*adjacent_speech_frames_threshold=*/1);
+  RunOnConstantLevel(/*num_iterations=*/kNumFramesToConfidence,
+                     level_estimator.level_rms_dbfs,
+                     level_estimator.level_peak_dbfs, kMaxSpeechProbability,
+                     *level_estimator.estimator);
+  EXPECT_NEAR(level_estimator.estimator->level_dbfs(),
+              level_estimator.level_rms_dbfs,
+              kConvergenceSpeedTestsLevelTolerance);
+}
+
+// Checks the convergence speed of the estimator after it becomes confident.
+TEST(GainController2SpeechLevelEstimator, ConvergenceSpeedAfterConfidence) {
+  TestLevelEstimator level_estimator(/*adjacent_speech_frames_threshold=*/1);
+  // Reach confidence using the initial level estimate.
+  RunOnConstantLevel(
+      /*num_iterations=*/kNumFramesToConfidence,
+      /*rms_dbfs=*/level_estimator.initial_speech_level_dbfs,
+      /*peak_dbfs=*/level_estimator.initial_speech_level_dbfs + 6.0f,
+      kMaxSpeechProbability, *level_estimator.estimator);
+  // No estimate change should occur, but confidence is achieved.
+  ASSERT_FLOAT_EQ(level_estimator.estimator->level_dbfs(),
+                  level_estimator.initial_speech_level_dbfs);
+  ASSERT_TRUE(level_estimator.estimator->is_confident());
+  // After confidence.
+  constexpr float kConvergenceTimeAfterConfidenceNumFrames = 600;  // 6 seconds.
+  static_assert(
+      kConvergenceTimeAfterConfidenceNumFrames > kNumFramesToConfidence, "");
+  RunOnConstantLevel(
+      /*num_iterations=*/kConvergenceTimeAfterConfidenceNumFrames,
+      level_estimator.level_rms_dbfs, level_estimator.level_peak_dbfs,
+      kMaxSpeechProbability, *level_estimator.estimator);
+  EXPECT_NEAR(level_estimator.estimator->level_dbfs(),
+              level_estimator.level_rms_dbfs,
+              kConvergenceSpeedTestsLevelTolerance);
+}
+
+class SpeechLevelEstimatorParametrization
+    : public ::testing::TestWithParam<int> {
+ protected:
+  int adjacent_speech_frames_threshold() const { return GetParam(); }
+};
+
+TEST_P(SpeechLevelEstimatorParametrization, DoNotAdaptToShortSpeechSegments) {
+  TestLevelEstimator level_estimator(adjacent_speech_frames_threshold());
+  const float initial_level = level_estimator.estimator->level_dbfs();
+  ASSERT_LT(initial_level, level_estimator.level_peak_dbfs);
+  for (int i = 0; i < adjacent_speech_frames_threshold() - 1; ++i) {
+    SCOPED_TRACE(i);
+    level_estimator.estimator->Update(level_estimator.level_rms_dbfs,
+                                      level_estimator.level_peak_dbfs,
+                                      kMaxSpeechProbability);
+    EXPECT_EQ(initial_level, level_estimator.estimator->level_dbfs());
+  }
+  level_estimator.estimator->Update(level_estimator.level_rms_dbfs,
+                                    level_estimator.level_peak_dbfs,
+                                    kLowSpeechProbability);
+  EXPECT_EQ(initial_level, level_estimator.estimator->level_dbfs());
+}
+
+TEST_P(SpeechLevelEstimatorParametrization, AdaptToEnoughSpeechSegments) {
+  TestLevelEstimator level_estimator(adjacent_speech_frames_threshold());
+  const float initial_level = level_estimator.estimator->level_dbfs();
+  ASSERT_LT(initial_level, level_estimator.level_peak_dbfs);
+  for (int i = 0; i < adjacent_speech_frames_threshold(); ++i) {
+    level_estimator.estimator->Update(level_estimator.level_rms_dbfs,
+                                      level_estimator.level_peak_dbfs,
+                                      kMaxSpeechProbability);
+  }
+  EXPECT_LT(initial_level, level_estimator.estimator->level_dbfs());
+}
+
+INSTANTIATE_TEST_SUITE_P(GainController2,
+                         SpeechLevelEstimatorParametrization,
+                         ::testing::Values(1, 9, 17));
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer.cc b/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer.cc
new file mode 100644
index 0000000000..7746f6c000
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/speech_probability_buffer.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kActivityThreshold = 0.9f;
+constexpr int kNumAnalysisFrames = 100;
+// We use 12 in AGC2 adaptive digital, but with a slightly different logic.
+constexpr int kTransientWidthThreshold = 7;
+
+}  // namespace
+
+SpeechProbabilityBuffer::SpeechProbabilityBuffer(
+    float low_probability_threshold)
+    : low_probability_threshold_(low_probability_threshold),
+      probabilities_(kNumAnalysisFrames) {
+  RTC_DCHECK_GE(low_probability_threshold, 0.0f);
+  RTC_DCHECK_LE(low_probability_threshold, 1.0f);
+  RTC_DCHECK(!probabilities_.empty());
+}
+
+void SpeechProbabilityBuffer::Update(float probability) {
+  // Remove the oldest entry if the circular buffer is full.
+  if (buffer_is_full_) {
+    const float oldest_probability = probabilities_[buffer_index_];
+    sum_probabilities_ -= oldest_probability;
+  }
+
+  // Check for transients.
+  if (probability <= low_probability_threshold_) {
+    // Set a probability lower than the threshold to zero.
+    probability = 0.0f;
+
+    // Check if this has been a transient.
+    if (num_high_probability_observations_ <= kTransientWidthThreshold) {
+      RemoveTransient();
+    }
+    num_high_probability_observations_ = 0;
+  } else if (num_high_probability_observations_ <= kTransientWidthThreshold) {
+    ++num_high_probability_observations_;
+  }
+
+  // Update the circular buffer and the current sum.
+  probabilities_[buffer_index_] = probability;
+  sum_probabilities_ += probability;
+
+  // Increment the buffer index and check for wrap-around.
+  if (++buffer_index_ >= kNumAnalysisFrames) {
+    buffer_index_ = 0;
+    buffer_is_full_ = true;
+  }
+}
+
+void SpeechProbabilityBuffer::RemoveTransient() {
+  // Don't expect to be here if high-activity region is longer than
+  // `kTransientWidthThreshold` or there has not been any transient.
+  RTC_DCHECK_LE(num_high_probability_observations_, kTransientWidthThreshold);
+
+  // Replace previously added probabilities with zero.
+  int index =
+      (buffer_index_ > 0) ? (buffer_index_ - 1) : (kNumAnalysisFrames - 1);
+
+  while (num_high_probability_observations_-- > 0) {
+    sum_probabilities_ -= probabilities_[index];
+    probabilities_[index] = 0.0f;
+
+    // Update the circular buffer index.
+    index = (index > 0) ? (index - 1) : (kNumAnalysisFrames - 1);
+  }
+}
+
+bool SpeechProbabilityBuffer::IsActiveSegment() const {
+  if (!buffer_is_full_) {
+    return false;
+  }
+  if (sum_probabilities_ < kActivityThreshold * kNumAnalysisFrames) {
+    return false;
+  }
+  return true;
+}
+
+void SpeechProbabilityBuffer::Reset() {
+  sum_probabilities_ = 0.0f;
+
+  // Empty the circular buffer.
+  buffer_index_ = 0;
+  buffer_is_full_ = false;
+  num_high_probability_observations_ = 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer.h b/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer.h
new file mode 100644
index 0000000000..3056a3eeab
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_SPEECH_PROBABILITY_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_SPEECH_PROBABILITY_BUFFER_H_
+
+#include <vector>
+
+#include "rtc_base/gtest_prod_util.h"
+
+namespace webrtc {
+
+// This class implements a circular buffer that stores speech probabilities
+// for a speech segment and estimates speech activity for that segment.
+class SpeechProbabilityBuffer {
+ public:
+  // Ctor. The value of `low_probability_threshold` is required to be on the
+  // range [0.0f, 1.0f].
+  explicit SpeechProbabilityBuffer(float low_probability_threshold);
+  ~SpeechProbabilityBuffer() {}
+  SpeechProbabilityBuffer(const SpeechProbabilityBuffer&) = delete;
+  SpeechProbabilityBuffer& operator=(const SpeechProbabilityBuffer&) = delete;
+
+  // Adds `probability` in the buffer and computes an updatds sum of the buffer
+  // probabilities. Value of `probability` is required to be on the range
+  // [0.0f, 1.0f].
+  void Update(float probability);
+
+  // Resets the histogram, forgets the past.
+  void Reset();
+
+  // Returns true if the segment is active (a long enough segment with an
+  // average speech probability above `low_probability_threshold`).
+  bool IsActiveSegment() const;
+
+ private:
+  void RemoveTransient();
+
+  // Use only for testing.
+  float GetSumProbabilities() const { return sum_probabilities_; }
+
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest,
+                           CheckSumAfterInitialization);
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest, CheckSumAfterUpdate);
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest, CheckSumAfterReset);
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest,
+                           CheckSumAfterTransientNotRemoved);
+  FRIEND_TEST_ALL_PREFIXES(SpeechProbabilityBufferTest,
+                           CheckSumAfterTransientRemoved);
+
+  const float low_probability_threshold_;
+
+  // Sum of probabilities stored in `probabilities_`. Must be updated if
+  // `probabilities_` is updated.
+  float sum_probabilities_ = 0.0f;
+
+  // Circular buffer for probabilities.
+  std::vector<float> probabilities_;
+
+  // Current index of the circular buffer, where the newest data will be written
+  // to, therefore, pointing to the oldest data if buffer is full.
+  int buffer_index_ = 0;
+
+  // Indicates if the buffer is full and adding a new value removes the oldest
+  // value.
+  int buffer_is_full_ = false;
+
+  int num_high_probability_observations_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_PROBABILITY_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer_unittest.cc
new file mode 100644
index 0000000000..89cc209d9d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/speech_probability_buffer_unittest.cc
@@ -0,0 +1,346 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/speech_probability_buffer.h"
+
+#include <algorithm>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kAbsError = 0.001f;
+constexpr float kActivityThreshold = 0.9f;
+constexpr float kLowProbabilityThreshold = 0.2f;
+constexpr int kNumAnalysisFrames = 100;
+
+}  // namespace
+
+TEST(SpeechProbabilityBufferTest, CheckSumAfterInitialization) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  EXPECT_EQ(buffer.GetSumProbabilities(), 0.0f);
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSumAfterUpdate) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  buffer.Update(0.7f);
+
+  EXPECT_NEAR(buffer.GetSumProbabilities(), 0.7f, kAbsError);
+
+  buffer.Update(0.6f);
+
+  EXPECT_NEAR(buffer.GetSumProbabilities(), 1.3f, kAbsError);
+
+  for (int i = 0; i < kNumAnalysisFrames - 1; ++i) {
+    buffer.Update(1.0f);
+  }
+
+  EXPECT_NEAR(buffer.GetSumProbabilities(), 99.6f, kAbsError);
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSumAfterReset) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  buffer.Update(0.7f);
+  buffer.Update(0.6f);
+  buffer.Update(0.3f);
+
+  EXPECT_GT(buffer.GetSumProbabilities(), 0.0f);
+
+  buffer.Reset();
+
+  EXPECT_EQ(buffer.GetSumProbabilities(), 0.0f);
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSumAfterTransientNotRemoved) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+
+  buffer.Update(0.0f);
+
+  EXPECT_NEAR(buffer.GetSumProbabilities(), 9.0f, kAbsError);
+
+  buffer.Update(0.0f);
+
+  EXPECT_NEAR(buffer.GetSumProbabilities(), 9.0f, kAbsError);
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSumAfterTransientRemoved) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  buffer.Update(0.0f);
+  buffer.Update(0.0f);
+  buffer.Update(0.0f);
+  buffer.Update(0.0f);
+  buffer.Update(0.0f);
+  buffer.Update(0.0f);
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+  buffer.Update(1.0f);
+
+  EXPECT_NEAR(buffer.GetSumProbabilities(), 3.0f, kAbsError);
+
+  buffer.Update(0.0f);
+
+  EXPECT_NEAR(buffer.GetSumProbabilities(), 0.0f, kAbsError);
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSegmentIsNotActiveAfterNoUpdates) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSegmentIsActiveChangesFromFalseToTrue) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  // Add low probabilities until the buffer is full. That's not enough
+  // to make `IsActiveSegment()` to return true.
+  for (int i = 0; i < kNumAnalysisFrames; ++i) {
+    buffer.Update(0.0f);
+  }
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  // Add high probabilities until `IsActiveSegment()` returns true.
+  for (int i = 0; i < kActivityThreshold * kNumAnalysisFrames - 1; ++i) {
+    buffer.Update(1.0f);
+  }
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(1.0f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSegmentIsActiveChangesFromTrueToFalse) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  // Add high probabilities until the buffer is full. That's enough to
+  // make `IsActiveSegment()` to return true.
+  for (int i = 0; i < kNumAnalysisFrames; ++i) {
+    buffer.Update(1.0f);
+  }
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+
+  // Add low probabilities until `IsActiveSegment()` returns false.
+  for (int i = 0; i < (1.0f - kActivityThreshold) * kNumAnalysisFrames - 1;
+       ++i) {
+    buffer.Update(0.0f);
+  }
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+
+  buffer.Update(1.0f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest,
+     CheckSegmentIsActiveAfterUpdatesWithHighProbabilities) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  for (int i = 0; i < kNumAnalysisFrames - 1; ++i) {
+    buffer.Update(1.0f);
+  }
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(1.0f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest,
+     CheckSegmentIsNotActiveAfterUpdatesWithLowProbabilities) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  for (int i = 0; i < kNumAnalysisFrames - 1; ++i) {
+    buffer.Update(0.3f);
+  }
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(0.3f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSegmentIsActiveAfterBufferIsFull) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  for (int i = 0; i < kNumAnalysisFrames - 1; ++i) {
+    buffer.Update(1.0f);
+  }
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(1.0f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+
+  buffer.Update(1.0f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSegmentIsNotActiveAfterBufferIsFull) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  for (int i = 0; i < kNumAnalysisFrames - 1; ++i) {
+    buffer.Update(0.29f);
+  }
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(0.29f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(0.29f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSegmentIsNotActiveAfterReset) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  for (int i = 0; i < kNumAnalysisFrames; ++i) {
+    buffer.Update(1.0f);
+  }
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+
+  buffer.Reset();
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest,
+     CheckSegmentIsNotActiveAfterTransientRemovedAfterFewUpdates) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  buffer.Update(0.4f);
+  buffer.Update(0.4f);
+  buffer.Update(0.0f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest,
+     CheckSegmentIsActiveAfterTransientNotRemoved) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  for (int i = 0; i < kNumAnalysisFrames; ++i) {
+    buffer.Update(1.0f);
+  }
+
+  buffer.Update(0.7f);
+  buffer.Update(0.8f);
+  buffer.Update(0.9f);
+  buffer.Update(1.0f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+
+  buffer.Update(0.0f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+
+  buffer.Update(0.7f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest,
+     CheckSegmentIsNotActiveAfterTransientNotRemoved) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  for (int i = 0; i < kNumAnalysisFrames; ++i) {
+    buffer.Update(0.1f);
+  }
+
+  buffer.Update(0.7f);
+  buffer.Update(0.8f);
+  buffer.Update(0.9f);
+  buffer.Update(1.0f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(0.0f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(0.7f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest,
+     CheckSegmentIsNotActiveAfterTransientRemoved) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  for (int i = 0; i < kNumAnalysisFrames; ++i) {
+    buffer.Update(0.1f);
+  }
+
+  buffer.Update(0.7f);
+  buffer.Update(0.8f);
+  buffer.Update(0.9f);
+  buffer.Update(1.0f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(0.0f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+
+  buffer.Update(0.7f);
+
+  EXPECT_FALSE(buffer.IsActiveSegment());
+}
+
+TEST(SpeechProbabilityBufferTest, CheckSegmentIsActiveAfterTransientRemoved) {
+  SpeechProbabilityBuffer buffer(kLowProbabilityThreshold);
+
+  for (int i = 0; i < kNumAnalysisFrames; ++i) {
+    buffer.Update(1.0f);
+  }
+
+  buffer.Update(0.7f);
+  buffer.Update(0.8f);
+  buffer.Update(0.9f);
+  buffer.Update(1.0f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+
+  buffer.Update(0.0f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+
+  buffer.Update(0.7f);
+
+  EXPECT_TRUE(buffer.IsActiveSegment());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper.cc b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper.cc
new file mode 100644
index 0000000000..af6325dea7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper.cc
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/vad_wrapper.h"
+
+#include <array>
+#include <utility>
+
+#include "api/array_view.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/agc2/rnn_vad/common.h"
+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kNumFramesPerSecond = 100;
+
+class MonoVadImpl : public VoiceActivityDetectorWrapper::MonoVad {
+ public:
+  explicit MonoVadImpl(const AvailableCpuFeatures& cpu_features)
+      : features_extractor_(cpu_features), rnn_vad_(cpu_features) {}
+  MonoVadImpl(const MonoVadImpl&) = delete;
+  MonoVadImpl& operator=(const MonoVadImpl&) = delete;
+  ~MonoVadImpl() = default;
+
+  int SampleRateHz() const override { return rnn_vad::kSampleRate24kHz; }
+  void Reset() override { rnn_vad_.Reset(); }
+  float Analyze(rtc::ArrayView<const float> frame) override {
+    RTC_DCHECK_EQ(frame.size(), rnn_vad::kFrameSize10ms24kHz);
+    std::array<float, rnn_vad::kFeatureVectorSize> feature_vector;
+    const bool is_silence = features_extractor_.CheckSilenceComputeFeatures(
+        /*samples=*/{frame.data(), rnn_vad::kFrameSize10ms24kHz},
+        feature_vector);
+    return rnn_vad_.ComputeVadProbability(feature_vector, is_silence);
+  }
+
+ private:
+  rnn_vad::FeaturesExtractor features_extractor_;
+  rnn_vad::RnnVad rnn_vad_;
+};
+
+}  // namespace
+
+VoiceActivityDetectorWrapper::VoiceActivityDetectorWrapper(
+    const AvailableCpuFeatures& cpu_features,
+    int sample_rate_hz)
+    : VoiceActivityDetectorWrapper(kVadResetPeriodMs,
+                                   cpu_features,
+                                   sample_rate_hz) {}
+
+VoiceActivityDetectorWrapper::VoiceActivityDetectorWrapper(
+    int vad_reset_period_ms,
+    const AvailableCpuFeatures& cpu_features,
+    int sample_rate_hz)
+    : VoiceActivityDetectorWrapper(vad_reset_period_ms,
+                                   std::make_unique<MonoVadImpl>(cpu_features),
+                                   sample_rate_hz) {}
+
+VoiceActivityDetectorWrapper::VoiceActivityDetectorWrapper(
+    int vad_reset_period_ms,
+    std::unique_ptr<MonoVad> vad,
+    int sample_rate_hz)
+    : vad_reset_period_frames_(
+          rtc::CheckedDivExact(vad_reset_period_ms, kFrameDurationMs)),
+      time_to_vad_reset_(vad_reset_period_frames_),
+      vad_(std::move(vad)) {
+  RTC_DCHECK(vad_);
+  RTC_DCHECK_GT(vad_reset_period_frames_, 1);
+  resampled_buffer_.resize(
+      rtc::CheckedDivExact(vad_->SampleRateHz(), kNumFramesPerSecond));
+  Initialize(sample_rate_hz);
+}
+
+VoiceActivityDetectorWrapper::~VoiceActivityDetectorWrapper() = default;
+
+void VoiceActivityDetectorWrapper::Initialize(int sample_rate_hz) {
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  frame_size_ = rtc::CheckedDivExact(sample_rate_hz, kNumFramesPerSecond);
+  int status =
+      resampler_.InitializeIfNeeded(sample_rate_hz, vad_->SampleRateHz(),
+                                    /*num_channels=*/1);
+  constexpr int kStatusOk = 0;
+  RTC_DCHECK_EQ(status, kStatusOk);
+  vad_->Reset();
+}
+
+float VoiceActivityDetectorWrapper::Analyze(AudioFrameView<const float> frame) {
+  // Periodically reset the VAD.
+  time_to_vad_reset_--;
+  if (time_to_vad_reset_ <= 0) {
+    vad_->Reset();
+    time_to_vad_reset_ = vad_reset_period_frames_;
+  }
+  // Resample the first channel of `frame`.
+  RTC_DCHECK_EQ(frame.samples_per_channel(), frame_size_);
+  resampler_.Resample(frame.channel(0).data(), frame_size_,
+                      resampled_buffer_.data(), resampled_buffer_.size());
+
+  return vad_->Analyze(resampled_buffer_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper.h b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper.h
new file mode 100644
index 0000000000..459c471630
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_VAD_WRAPPER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_VAD_WRAPPER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+
+// Wraps a single-channel Voice Activity Detector (VAD) which is used to analyze
+// the first channel of the input audio frames. Takes care of resampling the
+// input frames to match the sample rate of the wrapped VAD and periodically
+// resets the VAD.
+class VoiceActivityDetectorWrapper {
+ public:
+  // Single channel VAD interface.
+  class MonoVad {
+   public:
+    virtual ~MonoVad() = default;
+    // Returns the sample rate (Hz) required for the input frames analyzed by
+    // `ComputeProbability`.
+    virtual int SampleRateHz() const = 0;
+    // Resets the internal state.
+    virtual void Reset() = 0;
+    // Analyzes an audio frame and returns the speech probability.
+    virtual float Analyze(rtc::ArrayView<const float> frame) = 0;
+  };
+
+  // Ctor. Uses `cpu_features` to instantiate the default VAD.
+  VoiceActivityDetectorWrapper(const AvailableCpuFeatures& cpu_features,
+                               int sample_rate_hz);
+
+  // Ctor. `vad_reset_period_ms` indicates the period in milliseconds to call
+  // `MonoVad::Reset()`; it must be equal to or greater than the duration of two
+  // frames. Uses `cpu_features` to instantiate the default VAD.
+  VoiceActivityDetectorWrapper(int vad_reset_period_ms,
+                               const AvailableCpuFeatures& cpu_features,
+                               int sample_rate_hz);
+  // Ctor. Uses a custom `vad`.
+  VoiceActivityDetectorWrapper(int vad_reset_period_ms,
+                               std::unique_ptr<MonoVad> vad,
+                               int sample_rate_hz);
+
+  VoiceActivityDetectorWrapper(const VoiceActivityDetectorWrapper&) = delete;
+  VoiceActivityDetectorWrapper& operator=(const VoiceActivityDetectorWrapper&) =
+      delete;
+  ~VoiceActivityDetectorWrapper();
+
+  // Initializes the VAD wrapper.
+  void Initialize(int sample_rate_hz);
+
+  // Analyzes the first channel of `frame` and returns the speech probability.
+  // `frame` must be a 10 ms frame with the sample rate specified in the last
+  // `Initialize()` call.
+  float Analyze(AudioFrameView<const float> frame);
+
+ private:
+  const int vad_reset_period_frames_;
+  int frame_size_;
+  int time_to_vad_reset_;
+  PushResampler<float> resampler_;
+  std::unique_ptr<MonoVad> vad_;
+  std::vector<float> resampled_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_VAD_WRAPPER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_gn/moz.build
new file mode 100644
index 0000000000..dfa2765108
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("vad_wrapper_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_unittest.cc
new file mode 100644
index 0000000000..91efdb566e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/vad_wrapper_unittest.cc
@@ -0,0 +1,181 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/vad_wrapper.h"
+
+#include <limits>
+#include <memory>
+#include <tuple>
+#include <utility>
+#include <vector>
+
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::AnyNumber;
+using ::testing::Return;
+using ::testing::ReturnRoundRobin;
+using ::testing::Truly;
+
+constexpr int kNumFramesPerSecond = 100;
+
+constexpr int kNoVadPeriodicReset =
+    kFrameDurationMs * (std::numeric_limits<int>::max() / kFrameDurationMs);
+
+constexpr int kSampleRate8kHz = 8000;
+
+class MockVad : public VoiceActivityDetectorWrapper::MonoVad {
+ public:
+  MOCK_METHOD(int, SampleRateHz, (), (const, override));
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(float, Analyze, (rtc::ArrayView<const float> frame), (override));
+};
+
+// Checks that the ctor and `Initialize()` read the sample rate of the wrapped
+// VAD.
+TEST(GainController2VoiceActivityDetectorWrapper, CtorAndInitReadSampleRate) {
+  auto vad = std::make_unique<MockVad>();
+  EXPECT_CALL(*vad, SampleRateHz)
+      .Times(2)
+      .WillRepeatedly(Return(kSampleRate8kHz));
+  EXPECT_CALL(*vad, Reset).Times(AnyNumber());
+  auto vad_wrapper = std::make_unique<VoiceActivityDetectorWrapper>(
+      kNoVadPeriodicReset, std::move(vad), kSampleRate8kHz);
+}
+
+// Creates a `VoiceActivityDetectorWrapper` injecting a mock VAD that
+// repeatedly returns the next value from `speech_probabilities` and that
+// restarts from the beginning when after the last element is returned.
+std::unique_ptr<VoiceActivityDetectorWrapper> CreateMockVadWrapper(
+    int vad_reset_period_ms,
+    int sample_rate_hz,
+    const std::vector<float>& speech_probabilities,
+    int expected_vad_reset_calls) {
+  auto vad = std::make_unique<MockVad>();
+  EXPECT_CALL(*vad, SampleRateHz)
+      .Times(AnyNumber())
+      .WillRepeatedly(Return(sample_rate_hz));
+  if (expected_vad_reset_calls >= 0) {
+    EXPECT_CALL(*vad, Reset).Times(expected_vad_reset_calls);
+  }
+  EXPECT_CALL(*vad, Analyze)
+      .Times(AnyNumber())
+      .WillRepeatedly(ReturnRoundRobin(speech_probabilities));
+  return std::make_unique<VoiceActivityDetectorWrapper>(
+      vad_reset_period_ms, std::move(vad), kSampleRate8kHz);
+}
+
+// 10 ms mono frame.
+struct FrameWithView {
+  // Ctor. Initializes the frame samples with `value`.
+  explicit FrameWithView(int sample_rate_hz)
+      : samples(rtc::CheckedDivExact(sample_rate_hz, kNumFramesPerSecond),
+                0.0f),
+        channel0(samples.data()),
+        view(&channel0, /*num_channels=*/1, samples.size()) {}
+  std::vector<float> samples;
+  const float* const channel0;
+  const AudioFrameView<const float> view;
+};
+
+// Checks that the expected speech probabilities are returned.
+TEST(GainController2VoiceActivityDetectorWrapper, CheckSpeechProbabilities) {
+  const std::vector<float> speech_probabilities{0.709f, 0.484f, 0.882f, 0.167f,
+                                                0.44f,  0.525f, 0.858f, 0.314f,
+                                                0.653f, 0.965f, 0.413f, 0.0f};
+  auto vad_wrapper = CreateMockVadWrapper(kNoVadPeriodicReset, kSampleRate8kHz,
+                                          speech_probabilities,
+                                          /*expected_vad_reset_calls=*/1);
+  FrameWithView frame(kSampleRate8kHz);
+  for (int i = 0; rtc::SafeLt(i, speech_probabilities.size()); ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_EQ(speech_probabilities[i], vad_wrapper->Analyze(frame.view));
+  }
+}
+
+// Checks that the VAD is not periodically reset.
+TEST(GainController2VoiceActivityDetectorWrapper, VadNoPeriodicReset) {
+  constexpr int kNumFrames = 19;
+  auto vad_wrapper = CreateMockVadWrapper(kNoVadPeriodicReset, kSampleRate8kHz,
+                                          /*speech_probabilities=*/{1.0f},
+                                          /*expected_vad_reset_calls=*/1);
+  FrameWithView frame(kSampleRate8kHz);
+  for (int i = 0; i < kNumFrames; ++i) {
+    vad_wrapper->Analyze(frame.view);
+  }
+}
+
+class VadPeriodResetParametrization
+    : public ::testing::TestWithParam<std::tuple<int, int>> {
+ protected:
+  int num_frames() const { return std::get<0>(GetParam()); }
+  int vad_reset_period_frames() const { return std::get<1>(GetParam()); }
+};
+
+// Checks that the VAD is periodically reset with the expected period.
+TEST_P(VadPeriodResetParametrization, VadPeriodicReset) {
+  auto vad_wrapper = CreateMockVadWrapper(
+      /*vad_reset_period_ms=*/vad_reset_period_frames() * kFrameDurationMs,
+      kSampleRate8kHz,
+      /*speech_probabilities=*/{1.0f},
+      /*expected_vad_reset_calls=*/1 +
+          num_frames() / vad_reset_period_frames());
+  FrameWithView frame(kSampleRate8kHz);
+  for (int i = 0; i < num_frames(); ++i) {
+    vad_wrapper->Analyze(frame.view);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(GainController2VoiceActivityDetectorWrapper,
+                         VadPeriodResetParametrization,
+                         ::testing::Combine(::testing::Values(1, 19, 123),
+                                            ::testing::Values(2, 5, 20, 53)));
+
+class VadResamplingParametrization
+    : public ::testing::TestWithParam<std::tuple<int, int>> {
+ protected:
+  int input_sample_rate_hz() const { return std::get<0>(GetParam()); }
+  int vad_sample_rate_hz() const { return std::get<1>(GetParam()); }
+};
+
+// Checks that regardless of the input audio sample rate, the wrapped VAD
+// analyzes frames having the expected size, that is according to its internal
+// sample rate.
+TEST_P(VadResamplingParametrization, CheckResampledFrameSize) {
+  auto vad = std::make_unique<MockVad>();
+  EXPECT_CALL(*vad, SampleRateHz)
+      .Times(AnyNumber())
+      .WillRepeatedly(Return(vad_sample_rate_hz()));
+  EXPECT_CALL(*vad, Reset).Times(1);
+  EXPECT_CALL(*vad, Analyze(Truly([this](rtc::ArrayView<const float> frame) {
+    return rtc::SafeEq(frame.size(), rtc::CheckedDivExact(vad_sample_rate_hz(),
+                                                          kNumFramesPerSecond));
+  }))).Times(1);
+  auto vad_wrapper = std::make_unique<VoiceActivityDetectorWrapper>(
+      kNoVadPeriodicReset, std::move(vad), input_sample_rate_hz());
+  FrameWithView frame(input_sample_rate_hz());
+  vad_wrapper->Analyze(frame.view);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    GainController2VoiceActivityDetectorWrapper,
+    VadResamplingParametrization,
+    ::testing::Combine(::testing::Values(8000, 16000, 44100, 48000),
+                       ::testing::Values(6000, 8000, 12000, 16000, 24000)));
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/vector_float_frame.cc b/third_party/libwebrtc/modules/audio_processing/agc2/vector_float_frame.cc
new file mode 100644
index 0000000000..a70d815196
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/vector_float_frame.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/agc2/vector_float_frame.h"
+
+namespace webrtc {
+
+namespace {
+
+std::vector<float*> ConstructChannelPointers(
+    std::vector<std::vector<float>>* x) {
+  std::vector<float*> channel_ptrs;
+  for (auto& v : *x) {
+    channel_ptrs.push_back(v.data());
+  }
+  return channel_ptrs;
+}
+}  // namespace
+
+VectorFloatFrame::VectorFloatFrame(int num_channels,
+                                   int samples_per_channel,
+                                   float start_value)
+    : channels_(num_channels,
+                std::vector<float>(samples_per_channel, start_value)),
+      channel_ptrs_(ConstructChannelPointers(&channels_)),
+      float_frame_view_(channel_ptrs_.data(),
+                        channels_.size(),
+                        samples_per_channel) {}
+
+VectorFloatFrame::~VectorFloatFrame() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/vector_float_frame.h b/third_party/libwebrtc/modules/audio_processing/agc2/vector_float_frame.h
new file mode 100644
index 0000000000..b521f346f9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/agc2/vector_float_frame.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_VECTOR_FLOAT_FRAME_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_VECTOR_FLOAT_FRAME_H_
+
+#include <vector>
+
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+
+// A construct consisting of a multi-channel audio frame, and a FloatFrame view
+// of it.
+class VectorFloatFrame {
+ public:
+  VectorFloatFrame(int num_channels,
+                   int samples_per_channel,
+                   float start_value);
+  const AudioFrameView<float>& float_frame_view() { return float_frame_view_; }
+  AudioFrameView<const float> float_frame_view() const {
+    return float_frame_view_;
+  }
+
+  ~VectorFloatFrame();
+
+ private:
+  std::vector<std::vector<float>> channels_;
+  std::vector<float*> channel_ptrs_;
+  AudioFrameView<float> float_frame_view_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_VECTOR_FLOAT_FRAME_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/api_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/api_gn/moz.build
new file mode 100644
index 0000000000..7f1c48ac39
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/api_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/include/audio_processing.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("api_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/apm_logging_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/apm_logging_gn/moz.build
new file mode 100644
index 0000000000..b6d96414b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/apm_logging_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("apm_logging_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_buffer.cc b/third_party/libwebrtc/modules/audio_processing/audio_buffer.cc
new file mode 100644
index 0000000000..3dbe1fe072
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_buffer.cc
@@ -0,0 +1,396 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/audio_buffer.h"
+
+#include <string.h>
+
+#include <cstdint>
+
+#include "common_audio/channel_buffer.h"
+#include "common_audio/include/audio_util.h"
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "modules/audio_processing/splitting_filter.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kSamplesPer32kHzChannel = 320;
+constexpr size_t kSamplesPer48kHzChannel = 480;
+constexpr size_t kMaxSamplesPerChannel = AudioBuffer::kMaxSampleRate / 100;
+
+size_t NumBandsFromFramesPerChannel(size_t num_frames) {
+  if (num_frames == kSamplesPer32kHzChannel) {
+    return 2;
+  }
+  if (num_frames == kSamplesPer48kHzChannel) {
+    return 3;
+  }
+  return 1;
+}
+
+}  // namespace
+
+AudioBuffer::AudioBuffer(size_t input_rate,
+                         size_t input_num_channels,
+                         size_t buffer_rate,
+                         size_t buffer_num_channels,
+                         size_t output_rate,
+                         size_t output_num_channels)
+    : input_num_frames_(static_cast<int>(input_rate) / 100),
+      input_num_channels_(input_num_channels),
+      buffer_num_frames_(static_cast<int>(buffer_rate) / 100),
+      buffer_num_channels_(buffer_num_channels),
+      output_num_frames_(static_cast<int>(output_rate) / 100),
+      output_num_channels_(0),
+      num_channels_(buffer_num_channels),
+      num_bands_(NumBandsFromFramesPerChannel(buffer_num_frames_)),
+      num_split_frames_(rtc::CheckedDivExact(buffer_num_frames_, num_bands_)),
+      data_(
+          new ChannelBuffer<float>(buffer_num_frames_, buffer_num_channels_)) {
+  RTC_DCHECK_GT(input_num_frames_, 0);
+  RTC_DCHECK_GT(buffer_num_frames_, 0);
+  RTC_DCHECK_GT(output_num_frames_, 0);
+  RTC_DCHECK_GT(input_num_channels_, 0);
+  RTC_DCHECK_GT(buffer_num_channels_, 0);
+  RTC_DCHECK_LE(buffer_num_channels_, input_num_channels_);
+
+  const bool input_resampling_needed = input_num_frames_ != buffer_num_frames_;
+  const bool output_resampling_needed =
+      output_num_frames_ != buffer_num_frames_;
+  if (input_resampling_needed) {
+    for (size_t i = 0; i < buffer_num_channels_; ++i) {
+      input_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
+          new PushSincResampler(input_num_frames_, buffer_num_frames_)));
+    }
+  }
+
+  if (output_resampling_needed) {
+    for (size_t i = 0; i < buffer_num_channels_; ++i) {
+      output_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
+          new PushSincResampler(buffer_num_frames_, output_num_frames_)));
+    }
+  }
+
+  if (num_bands_ > 1) {
+    split_data_.reset(new ChannelBuffer<float>(
+        buffer_num_frames_, buffer_num_channels_, num_bands_));
+    splitting_filter_.reset(new SplittingFilter(
+        buffer_num_channels_, num_bands_, buffer_num_frames_));
+  }
+}
+
+AudioBuffer::~AudioBuffer() {}
+
+void AudioBuffer::set_downmixing_to_specific_channel(size_t channel) {
+  downmix_by_averaging_ = false;
+  RTC_DCHECK_GT(input_num_channels_, channel);
+  channel_for_downmixing_ = std::min(channel, input_num_channels_ - 1);
+}
+
+void AudioBuffer::set_downmixing_by_averaging() {
+  downmix_by_averaging_ = true;
+}
+
+void AudioBuffer::CopyFrom(const float* const* stacked_data,
+                           const StreamConfig& stream_config) {
+  RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_);
+  RTC_DCHECK_EQ(stream_config.num_channels(), input_num_channels_);
+  RestoreNumChannels();
+  const bool downmix_needed = input_num_channels_ > 1 && num_channels_ == 1;
+
+  const bool resampling_needed = input_num_frames_ != buffer_num_frames_;
+
+  if (downmix_needed) {
+    RTC_DCHECK_GE(kMaxSamplesPerChannel, input_num_frames_);
+
+    std::array<float, kMaxSamplesPerChannel> downmix;
+    if (downmix_by_averaging_) {
+      const float kOneByNumChannels = 1.f / input_num_channels_;
+      for (size_t i = 0; i < input_num_frames_; ++i) {
+        float value = stacked_data[0][i];
+        for (size_t j = 1; j < input_num_channels_; ++j) {
+          value += stacked_data[j][i];
+        }
+        downmix[i] = value * kOneByNumChannels;
+      }
+    }
+    const float* downmixed_data = downmix_by_averaging_
+                                      ? downmix.data()
+                                      : stacked_data[channel_for_downmixing_];
+
+    if (resampling_needed) {
+      input_resamplers_[0]->Resample(downmixed_data, input_num_frames_,
+                                     data_->channels()[0], buffer_num_frames_);
+    }
+    const float* data_to_convert =
+        resampling_needed ? data_->channels()[0] : downmixed_data;
+    FloatToFloatS16(data_to_convert, buffer_num_frames_, data_->channels()[0]);
+  } else {
+    if (resampling_needed) {
+      for (size_t i = 0; i < num_channels_; ++i) {
+        input_resamplers_[i]->Resample(stacked_data[i], input_num_frames_,
+                                       data_->channels()[i],
+                                       buffer_num_frames_);
+        FloatToFloatS16(data_->channels()[i], buffer_num_frames_,
+                        data_->channels()[i]);
+      }
+    } else {
+      for (size_t i = 0; i < num_channels_; ++i) {
+        FloatToFloatS16(stacked_data[i], buffer_num_frames_,
+                        data_->channels()[i]);
+      }
+    }
+  }
+}
+
+void AudioBuffer::CopyTo(const StreamConfig& stream_config,
+                         float* const* stacked_data) {
+  RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_);
+
+  const bool resampling_needed = output_num_frames_ != buffer_num_frames_;
+  if (resampling_needed) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      FloatS16ToFloat(data_->channels()[i], buffer_num_frames_,
+                      data_->channels()[i]);
+      output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_,
+                                      stacked_data[i], output_num_frames_);
+    }
+  } else {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      FloatS16ToFloat(data_->channels()[i], buffer_num_frames_,
+                      stacked_data[i]);
+    }
+  }
+
+  for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
+    memcpy(stacked_data[i], stacked_data[0],
+           output_num_frames_ * sizeof(**stacked_data));
+  }
+}
+
+void AudioBuffer::CopyTo(AudioBuffer* buffer) const {
+  RTC_DCHECK_EQ(buffer->num_frames(), output_num_frames_);
+
+  const bool resampling_needed = output_num_frames_ != buffer_num_frames_;
+  if (resampling_needed) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_,
+                                      buffer->channels()[i],
+                                      buffer->num_frames());
+    }
+  } else {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      memcpy(buffer->channels()[i], data_->channels()[i],
+             buffer_num_frames_ * sizeof(**buffer->channels()));
+    }
+  }
+
+  for (size_t i = num_channels_; i < buffer->num_channels(); ++i) {
+    memcpy(buffer->channels()[i], buffer->channels()[0],
+           output_num_frames_ * sizeof(**buffer->channels()));
+  }
+}
+
+void AudioBuffer::RestoreNumChannels() {
+  num_channels_ = buffer_num_channels_;
+  data_->set_num_channels(buffer_num_channels_);
+  if (split_data_.get()) {
+    split_data_->set_num_channels(buffer_num_channels_);
+  }
+}
+
+void AudioBuffer::set_num_channels(size_t num_channels) {
+  RTC_DCHECK_GE(buffer_num_channels_, num_channels);
+  num_channels_ = num_channels;
+  data_->set_num_channels(num_channels);
+  if (split_data_.get()) {
+    split_data_->set_num_channels(num_channels);
+  }
+}
+
+// The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
+void AudioBuffer::CopyFrom(const int16_t* const interleaved_data,
+                           const StreamConfig& stream_config) {
+  RTC_DCHECK_EQ(stream_config.num_channels(), input_num_channels_);
+  RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_);
+  RestoreNumChannels();
+
+  const bool resampling_required = input_num_frames_ != buffer_num_frames_;
+
+  const int16_t* interleaved = interleaved_data;
+  if (num_channels_ == 1) {
+    if (input_num_channels_ == 1) {
+      if (resampling_required) {
+        std::array<float, kMaxSamplesPerChannel> float_buffer;
+        S16ToFloatS16(interleaved, input_num_frames_, float_buffer.data());
+        input_resamplers_[0]->Resample(float_buffer.data(), input_num_frames_,
+                                       data_->channels()[0],
+                                       buffer_num_frames_);
+      } else {
+        S16ToFloatS16(interleaved, input_num_frames_, data_->channels()[0]);
+      }
+    } else {
+      std::array<float, kMaxSamplesPerChannel> float_buffer;
+      float* downmixed_data =
+          resampling_required ? float_buffer.data() : data_->channels()[0];
+      if (downmix_by_averaging_) {
+        for (size_t j = 0, k = 0; j < input_num_frames_; ++j) {
+          int32_t sum = 0;
+          for (size_t i = 0; i < input_num_channels_; ++i, ++k) {
+            sum += interleaved[k];
+          }
+          downmixed_data[j] = sum / static_cast<int16_t>(input_num_channels_);
+        }
+      } else {
+        for (size_t j = 0, k = channel_for_downmixing_; j < input_num_frames_;
+             ++j, k += input_num_channels_) {
+          downmixed_data[j] = interleaved[k];
+        }
+      }
+
+      if (resampling_required) {
+        input_resamplers_[0]->Resample(downmixed_data, input_num_frames_,
+                                       data_->channels()[0],
+                                       buffer_num_frames_);
+      }
+    }
+  } else {
+    auto deinterleave_channel = [](size_t channel, size_t num_channels,
+                                   size_t samples_per_channel, const int16_t* x,
+                                   float* y) {
+      for (size_t j = 0, k = channel; j < samples_per_channel;
+           ++j, k += num_channels) {
+        y[j] = x[k];
+      }
+    };
+
+    if (resampling_required) {
+      std::array<float, kMaxSamplesPerChannel> float_buffer;
+      for (size_t i = 0; i < num_channels_; ++i) {
+        deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
+                             float_buffer.data());
+        input_resamplers_[i]->Resample(float_buffer.data(), input_num_frames_,
+                                       data_->channels()[i],
+                                       buffer_num_frames_);
+      }
+    } else {
+      for (size_t i = 0; i < num_channels_; ++i) {
+        deinterleave_channel(i, num_channels_, input_num_frames_, interleaved,
+                             data_->channels()[i]);
+      }
+    }
+  }
+}
+
+void AudioBuffer::CopyTo(const StreamConfig& stream_config,
+                         int16_t* const interleaved_data) {
+  const size_t config_num_channels = stream_config.num_channels();
+
+  RTC_DCHECK(config_num_channels == num_channels_ || num_channels_ == 1);
+  RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_);
+
+  const bool resampling_required = buffer_num_frames_ != output_num_frames_;
+
+  int16_t* interleaved = interleaved_data;
+  if (num_channels_ == 1) {
+    std::array<float, kMaxSamplesPerChannel> float_buffer;
+
+    if (resampling_required) {
+      output_resamplers_[0]->Resample(data_->channels()[0], buffer_num_frames_,
+                                      float_buffer.data(), output_num_frames_);
+    }
+    const float* deinterleaved =
+        resampling_required ? float_buffer.data() : data_->channels()[0];
+
+    if (config_num_channels == 1) {
+      for (size_t j = 0; j < output_num_frames_; ++j) {
+        interleaved[j] = FloatS16ToS16(deinterleaved[j]);
+      }
+    } else {
+      for (size_t i = 0, k = 0; i < output_num_frames_; ++i) {
+        float tmp = FloatS16ToS16(deinterleaved[i]);
+        for (size_t j = 0; j < config_num_channels; ++j, ++k) {
+          interleaved[k] = tmp;
+        }
+      }
+    }
+  } else {
+    auto interleave_channel = [](size_t channel, size_t num_channels,
+                                 size_t samples_per_channel, const float* x,
+                                 int16_t* y) {
+      for (size_t k = 0, j = channel; k < samples_per_channel;
+           ++k, j += num_channels) {
+        y[j] = FloatS16ToS16(x[k]);
+      }
+    };
+
+    if (resampling_required) {
+      for (size_t i = 0; i < num_channels_; ++i) {
+        std::array<float, kMaxSamplesPerChannel> float_buffer;
+        output_resamplers_[i]->Resample(data_->channels()[i],
+                                        buffer_num_frames_, float_buffer.data(),
+                                        output_num_frames_);
+        interleave_channel(i, config_num_channels, output_num_frames_,
+                           float_buffer.data(), interleaved);
+      }
+    } else {
+      for (size_t i = 0; i < num_channels_; ++i) {
+        interleave_channel(i, config_num_channels, output_num_frames_,
+                           data_->channels()[i], interleaved);
+      }
+    }
+
+    for (size_t i = num_channels_; i < config_num_channels; ++i) {
+      for (size_t j = 0, k = i, n = num_channels_; j < output_num_frames_;
+           ++j, k += config_num_channels, n += config_num_channels) {
+        interleaved[k] = interleaved[n];
+      }
+    }
+  }
+}
+
+void AudioBuffer::SplitIntoFrequencyBands() {
+  splitting_filter_->Analysis(data_.get(), split_data_.get());
+}
+
+void AudioBuffer::MergeFrequencyBands() {
+  splitting_filter_->Synthesis(split_data_.get(), data_.get());
+}
+
+void AudioBuffer::ExportSplitChannelData(
+    size_t channel,
+    int16_t* const* split_band_data) const {
+  for (size_t k = 0; k < num_bands(); ++k) {
+    const float* band_data = split_bands_const(channel)[k];
+
+    RTC_DCHECK(split_band_data[k]);
+    RTC_DCHECK(band_data);
+    for (size_t i = 0; i < num_frames_per_band(); ++i) {
+      split_band_data[k][i] = FloatS16ToS16(band_data[i]);
+    }
+  }
+}
+
+void AudioBuffer::ImportSplitChannelData(
+    size_t channel,
+    const int16_t* const* split_band_data) {
+  for (size_t k = 0; k < num_bands(); ++k) {
+    float* band_data = split_bands(channel)[k];
+    RTC_DCHECK(split_band_data[k]);
+    RTC_DCHECK(band_data);
+    for (size_t i = 0; i < num_frames_per_band(); ++i) {
+      band_data[i] = split_band_data[k][i];
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_buffer.h b/third_party/libwebrtc/modules/audio_processing/audio_buffer.h
new file mode 100644
index 0000000000..b9ea3000a2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_buffer.h
@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "common_audio/channel_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+class PushSincResampler;
+class SplittingFilter;
+
+enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
+
+// Stores any audio data in a way that allows the audio processing module to
+// operate on it in a controlled manner.
+class AudioBuffer {
+ public:
+  static const int kSplitBandSize = 160;
+  static const int kMaxSampleRate = 384000;
+  AudioBuffer(size_t input_rate,
+              size_t input_num_channels,
+              size_t buffer_rate,
+              size_t buffer_num_channels,
+              size_t output_rate,
+              size_t output_num_channels);
+
+  virtual ~AudioBuffer();
+
+  AudioBuffer(const AudioBuffer&) = delete;
+  AudioBuffer& operator=(const AudioBuffer&) = delete;
+
+  // Specify that downmixing should be done by selecting a single channel.
+  void set_downmixing_to_specific_channel(size_t channel);
+
+  // Specify that downmixing should be done by averaging all channels,.
+  void set_downmixing_by_averaging();
+
+  // Set the number of channels in the buffer. The specified number of channels
+  // cannot be larger than the specified buffer_num_channels. The number is also
+  // reset at each call to CopyFrom or InterleaveFrom.
+  void set_num_channels(size_t num_channels);
+
+  size_t num_channels() const { return num_channels_; }
+  size_t num_frames() const { return buffer_num_frames_; }
+  size_t num_frames_per_band() const { return num_split_frames_; }
+  size_t num_bands() const { return num_bands_; }
+
+  // Returns pointer arrays to the full-band channels.
+  // Usage:
+  // channels()[channel][sample].
+  // Where:
+  // 0 <= channel < `buffer_num_channels_`
+  // 0 <= sample < `buffer_num_frames_`
+  float* const* channels() { return data_->channels(); }
+  const float* const* channels_const() const { return data_->channels(); }
+
+  // Returns pointer arrays to the bands for a specific channel.
+  // Usage:
+  // split_bands(channel)[band][sample].
+  // Where:
+  // 0 <= channel < `buffer_num_channels_`
+  // 0 <= band < `num_bands_`
+  // 0 <= sample < `num_split_frames_`
+  const float* const* split_bands_const(size_t channel) const {
+    return split_data_.get() ? split_data_->bands(channel)
+                             : data_->bands(channel);
+  }
+  float* const* split_bands(size_t channel) {
+    return split_data_.get() ? split_data_->bands(channel)
+                             : data_->bands(channel);
+  }
+
+  // Returns a pointer array to the channels for a specific band.
+  // Usage:
+  // split_channels(band)[channel][sample].
+  // Where:
+  // 0 <= band < `num_bands_`
+  // 0 <= channel < `buffer_num_channels_`
+  // 0 <= sample < `num_split_frames_`
+  const float* const* split_channels_const(Band band) const {
+    if (split_data_.get()) {
+      return split_data_->channels(band);
+    } else {
+      return band == kBand0To8kHz ? data_->channels() : nullptr;
+    }
+  }
+
+  // Copies data into the buffer.
+  void CopyFrom(const int16_t* const interleaved_data,
+                const StreamConfig& stream_config);
+  void CopyFrom(const float* const* stacked_data,
+                const StreamConfig& stream_config);
+
+  // Copies data from the buffer.
+  void CopyTo(const StreamConfig& stream_config,
+              int16_t* const interleaved_data);
+  void CopyTo(const StreamConfig& stream_config, float* const* stacked_data);
+  void CopyTo(AudioBuffer* buffer) const;
+
+  // Splits the buffer data into frequency bands.
+  void SplitIntoFrequencyBands();
+
+  // Recombines the frequency bands into a full-band signal.
+  void MergeFrequencyBands();
+
+  // Copies the split bands data into the integer two-dimensional array.
+  void ExportSplitChannelData(size_t channel,
+                              int16_t* const* split_band_data) const;
+
+  // Copies the data in the integer two-dimensional array into the split_bands
+  // data.
+  void ImportSplitChannelData(size_t channel,
+                              const int16_t* const* split_band_data);
+
+  static const size_t kMaxSplitFrameLength = 160;
+  static const size_t kMaxNumBands = 3;
+
+  // Deprecated methods, will be removed soon.
+  float* const* channels_f() { return channels(); }
+  const float* const* channels_const_f() const { return channels_const(); }
+  const float* const* split_bands_const_f(size_t channel) const {
+    return split_bands_const(channel);
+  }
+  float* const* split_bands_f(size_t channel) { return split_bands(channel); }
+  const float* const* split_channels_const_f(Band band) const {
+    return split_channels_const(band);
+  }
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(AudioBufferTest,
+                           SetNumChannelsSetsChannelBuffersNumChannels);
+  void RestoreNumChannels();
+
+  const size_t input_num_frames_;
+  const size_t input_num_channels_;
+  const size_t buffer_num_frames_;
+  const size_t buffer_num_channels_;
+  const size_t output_num_frames_;
+  const size_t output_num_channels_;
+
+  size_t num_channels_;
+  size_t num_bands_;
+  size_t num_split_frames_;
+
+  std::unique_ptr<ChannelBuffer<float>> data_;
+  std::unique_ptr<ChannelBuffer<float>> split_data_;
+  std::unique_ptr<SplittingFilter> splitting_filter_;
+  std::vector<std::unique_ptr<PushSincResampler>> input_resamplers_;
+  std::vector<std::unique_ptr<PushSincResampler>> output_resamplers_;
+  bool downmix_by_averaging_ = true;
+  size_t channel_for_downmixing_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_buffer_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_buffer_gn/moz.build
new file mode 100644
index 0000000000..2291d2981d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_buffer_gn/moz.build
@@ -0,0 +1,235 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/audio_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/splitting_filter.cc",
+    "/third_party/libwebrtc/modules/audio_processing/three_band_filter_bank.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_buffer_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_buffer_unittest.cc
new file mode 100644
index 0000000000..f3b2ddc689
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_buffer_unittest.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/audio_buffer.h"
+
+#include <cmath>
+
+#include "test/gtest.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kSampleRateHz = 48000u;
+const size_t kStereo = 2u;
+const size_t kMono = 1u;
+
+void ExpectNumChannels(const AudioBuffer& ab, size_t num_channels) {
+  EXPECT_EQ(ab.num_channels(), num_channels);
+}
+
+}  // namespace
+
+TEST(AudioBufferTest, SetNumChannelsSetsChannelBuffersNumChannels) {
+  AudioBuffer ab(kSampleRateHz, kStereo, kSampleRateHz, kStereo, kSampleRateHz,
+                 kStereo);
+  ExpectNumChannels(ab, kStereo);
+  ab.set_num_channels(1);
+  ExpectNumChannels(ab, kMono);
+  ab.RestoreNumChannels();
+  ExpectNumChannels(ab, kStereo);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(AudioBufferDeathTest, SetNumChannelsDeathTest) {
+  AudioBuffer ab(kSampleRateHz, kMono, kSampleRateHz, kMono, kSampleRateHz,
+                 kMono);
+  RTC_EXPECT_DEATH(ab.set_num_channels(kStereo), "num_channels");
+}
+#endif
+
+TEST(AudioBufferTest, CopyWithoutResampling) {
+  AudioBuffer ab1(32000, 2, 32000, 2, 32000, 2);
+  AudioBuffer ab2(32000, 2, 32000, 2, 32000, 2);
+  // Fill first buffer.
+  for (size_t ch = 0; ch < ab1.num_channels(); ++ch) {
+    for (size_t i = 0; i < ab1.num_frames(); ++i) {
+      ab1.channels()[ch][i] = i + ch;
+    }
+  }
+  // Copy to second buffer.
+  ab1.CopyTo(&ab2);
+  // Verify content of second buffer.
+  for (size_t ch = 0; ch < ab2.num_channels(); ++ch) {
+    for (size_t i = 0; i < ab2.num_frames(); ++i) {
+      EXPECT_EQ(ab2.channels()[ch][i], i + ch);
+    }
+  }
+}
+
+TEST(AudioBufferTest, CopyWithResampling) {
+  AudioBuffer ab1(32000, 2, 32000, 2, 48000, 2);
+  AudioBuffer ab2(48000, 2, 48000, 2, 48000, 2);
+  float energy_ab1 = 0.f;
+  float energy_ab2 = 0.f;
+  const float pi = std::acos(-1.f);
+  // Put a sine and compute energy of first buffer.
+  for (size_t ch = 0; ch < ab1.num_channels(); ++ch) {
+    for (size_t i = 0; i < ab1.num_frames(); ++i) {
+      ab1.channels()[ch][i] = std::sin(2 * pi * 100.f / 32000.f * i);
+      energy_ab1 += ab1.channels()[ch][i] * ab1.channels()[ch][i];
+    }
+  }
+  // Copy to second buffer.
+  ab1.CopyTo(&ab2);
+  // Compute energy of second buffer.
+  for (size_t ch = 0; ch < ab2.num_channels(); ++ch) {
+    for (size_t i = 0; i < ab2.num_frames(); ++i) {
+      energy_ab2 += ab2.channels()[ch][i] * ab2.channels()[ch][i];
+    }
+  }
+  // Verify that energies match.
+  EXPECT_NEAR(energy_ab1, energy_ab2 * 32000.f / 48000.f, .01f * energy_ab1);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_frame_proxies_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_frame_proxies_gn/moz.build
new file mode 100644
index 0000000000..683d86e743
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_frame_proxies_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_frame_proxies_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_frame_view_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_frame_view_gn/moz.build
new file mode 100644
index 0000000000..9b90bca379
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_frame_view_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_frame_view_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc
new file mode 100644
index 0000000000..fd25bc3b0b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_frame_view_unittest.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+TEST(AudioFrameTest, ConstructFromAudioBuffer) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 2;
+  constexpr float kFloatConstant = 1272.f;
+  constexpr float kIntConstant = 17252;
+  const webrtc::StreamConfig stream_config(kSampleRateHz, kNumChannels);
+  webrtc::AudioBuffer buffer(
+      stream_config.sample_rate_hz(), stream_config.num_channels(),
+      stream_config.sample_rate_hz(), stream_config.num_channels(),
+      stream_config.sample_rate_hz(), stream_config.num_channels());
+
+  AudioFrameView<float> non_const_view(buffer.channels(), buffer.num_channels(),
+                                       buffer.num_frames());
+  // Modification is allowed.
+  non_const_view.channel(0)[0] = kFloatConstant;
+  EXPECT_EQ(buffer.channels()[0][0], kFloatConstant);
+
+  AudioFrameView<const float> const_view(
+      buffer.channels(), buffer.num_channels(), buffer.num_frames());
+  // Modification is not allowed.
+  // const_view.channel(0)[0] = kFloatConstant;
+
+  // Assignment is allowed.
+  AudioFrameView<const float> other_const_view = non_const_view;
+  static_cast<void>(other_const_view);
+
+  // But not the other way. The following will fail:
+  // non_const_view = other_const_view;
+
+  AudioFrameView<float> non_const_float_view(
+      buffer.channels(), buffer.num_channels(), buffer.num_frames());
+  non_const_float_view.channel(0)[0] = kIntConstant;
+  EXPECT_EQ(buffer.channels()[0][0], kIntConstant);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_builder_impl.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_builder_impl.cc
new file mode 100644
index 0000000000..a246448c26
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_builder_impl.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "api/make_ref_counted.h"
+#include "modules/audio_processing/audio_processing_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+AudioProcessingBuilder::AudioProcessingBuilder() = default;
+AudioProcessingBuilder::~AudioProcessingBuilder() = default;
+
+rtc::scoped_refptr<AudioProcessing> AudioProcessingBuilder::Create() {
+#ifdef WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE
+  // Return a null pointer when the APM is excluded from the build.
+  return nullptr;
+#else  // WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE
+  return rtc::make_ref_counted<AudioProcessingImpl>(
+      config_, std::move(capture_post_processing_),
+      std::move(render_pre_processing_), std::move(echo_control_factory_),
+      std::move(echo_detector_), std::move(capture_analyzer_));
+#endif
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_processing_gn/moz.build
new file mode 100644
index 0000000000..ab0ca7113f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_gn/moz.build
@@ -0,0 +1,239 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_impl.cc",
+    "/third_party/libwebrtc/modules/audio_processing/gain_control_impl.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/audio_processing_builder_impl.cc",
+    "/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_processing_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
new file mode 100644
index 0000000000..c304453388
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
@@ -0,0 +1,2649 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/audio_processing_impl.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstring>
+#include <memory>
+#include <string>
+#include <type_traits>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/audio_frame.h"
+#include "common_audio/audio_converter.h"
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/optionally_built_submodule_creators.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/denormal_disabler.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+
+#define RETURN_ON_ERR(expr) \
+  do {                      \
+    int err = (expr);       \
+    if (err != kNoError) {  \
+      return err;           \
+    }                       \
+  } while (0)
+
+namespace webrtc {
+
+namespace {
+
+bool SampleRateSupportsMultiBand(int sample_rate_hz) {
+  return sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
+         sample_rate_hz == AudioProcessing::kSampleRate48kHz;
+}
+
+// Checks whether the high-pass filter should be done in the full-band.
+bool EnforceSplitBandHpf() {
+  return field_trial::IsEnabled("WebRTC-FullBandHpfKillSwitch");
+}
+
+// Checks whether AEC3 should be allowed to decide what the default
+// configuration should be based on the render and capture channel configuration
+// at hand.
+bool UseSetupSpecificDefaultAec3Congfig() {
+  return !field_trial::IsEnabled(
+      "WebRTC-Aec3SetupSpecificDefaultConfigDefaultsKillSwitch");
+}
+
+// Identify the native processing rate that best handles a sample rate.
+int SuitableProcessRate(int minimum_rate,
+                        int max_splitting_rate,
+                        bool band_splitting_required) {
+  const int uppermost_native_rate =
+      band_splitting_required ? max_splitting_rate : 48000;
+  for (auto rate : {16000, 32000, 48000}) {
+    if (rate >= uppermost_native_rate) {
+      return uppermost_native_rate;
+    }
+    if (rate >= minimum_rate) {
+      return rate;
+    }
+  }
+  RTC_DCHECK_NOTREACHED();
+  return uppermost_native_rate;
+}
+
+GainControl::Mode Agc1ConfigModeToInterfaceMode(
+    AudioProcessing::Config::GainController1::Mode mode) {
+  using Agc1Config = AudioProcessing::Config::GainController1;
+  switch (mode) {
+    case Agc1Config::kAdaptiveAnalog:
+      return GainControl::kAdaptiveAnalog;
+    case Agc1Config::kAdaptiveDigital:
+      return GainControl::kAdaptiveDigital;
+    case Agc1Config::kFixedDigital:
+      return GainControl::kFixedDigital;
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+bool MinimizeProcessingForUnusedOutput() {
+  return !field_trial::IsEnabled("WebRTC-MutedStateKillSwitch");
+}
+
+// Maximum lengths that frame of samples being passed from the render side to
+// the capture side can have (does not apply to AEC3).
+static const size_t kMaxAllowedValuesOfSamplesPerBand = 160;
+static const size_t kMaxAllowedValuesOfSamplesPerFrame = 480;
+
+// Maximum number of frames to buffer in the render queue.
+// TODO(peah): Decrease this once we properly handle hugely unbalanced
+// reverse and forward call numbers.
+static const size_t kMaxNumFramesToBuffer = 100;
+
+void PackRenderAudioBufferForEchoDetector(const AudioBuffer& audio,
+                                          std::vector<float>& packed_buffer) {
+  packed_buffer.clear();
+  packed_buffer.insert(packed_buffer.end(), audio.channels_const()[0],
+                       audio.channels_const()[0] + audio.num_frames());
+}
+
+// Options for gracefully handling processing errors.
+enum class FormatErrorOutputOption {
+  kOutputExactCopyOfInput,
+  kOutputBroadcastCopyOfFirstInputChannel,
+  kOutputSilence,
+  kDoNothing
+};
+
+enum class AudioFormatValidity {
+  // Format is supported by APM.
+  kValidAndSupported,
+  // Format has a reasonable interpretation but is not supported.
+  kValidButUnsupportedSampleRate,
+  // The remaining enums values signal that the audio does not have a reasonable
+  // interpretation and cannot be used.
+  kInvalidSampleRate,
+  kInvalidChannelCount
+};
+
+AudioFormatValidity ValidateAudioFormat(const StreamConfig& config) {
+  if (config.sample_rate_hz() < 0)
+    return AudioFormatValidity::kInvalidSampleRate;
+  if (config.num_channels() == 0)
+    return AudioFormatValidity::kInvalidChannelCount;
+
+  // Format has a reasonable interpretation, but may still be unsupported.
+  if (config.sample_rate_hz() < 8000 ||
+      config.sample_rate_hz() > AudioBuffer::kMaxSampleRate)
+    return AudioFormatValidity::kValidButUnsupportedSampleRate;
+
+  // Format is fully supported.
+  return AudioFormatValidity::kValidAndSupported;
+}
+
+int AudioFormatValidityToErrorCode(AudioFormatValidity validity) {
+  switch (validity) {
+    case AudioFormatValidity::kValidAndSupported:
+      return AudioProcessing::kNoError;
+    case AudioFormatValidity::kValidButUnsupportedSampleRate:  // fall-through
+    case AudioFormatValidity::kInvalidSampleRate:
+      return AudioProcessing::kBadSampleRateError;
+    case AudioFormatValidity::kInvalidChannelCount:
+      return AudioProcessing::kBadNumberChannelsError;
+  }
+  RTC_DCHECK(false);
+}
+
+// Returns an AudioProcessing::Error together with the best possible option for
+// output audio content.
+std::pair<int, FormatErrorOutputOption> ChooseErrorOutputOption(
+    const StreamConfig& input_config,
+    const StreamConfig& output_config) {
+  AudioFormatValidity input_validity = ValidateAudioFormat(input_config);
+  AudioFormatValidity output_validity = ValidateAudioFormat(output_config);
+
+  if (input_validity == AudioFormatValidity::kValidAndSupported &&
+      output_validity == AudioFormatValidity::kValidAndSupported &&
+      (output_config.num_channels() == 1 ||
+       output_config.num_channels() == input_config.num_channels())) {
+    return {AudioProcessing::kNoError, FormatErrorOutputOption::kDoNothing};
+  }
+
+  int error_code = AudioFormatValidityToErrorCode(input_validity);
+  if (error_code == AudioProcessing::kNoError) {
+    error_code = AudioFormatValidityToErrorCode(output_validity);
+  }
+  if (error_code == AudioProcessing::kNoError) {
+    // The individual formats are valid but there is some error - must be
+    // channel mismatch.
+    error_code = AudioProcessing::kBadNumberChannelsError;
+  }
+
+  FormatErrorOutputOption output_option;
+  if (output_validity != AudioFormatValidity::kValidAndSupported &&
+      output_validity != AudioFormatValidity::kValidButUnsupportedSampleRate) {
+    // The output format is uninterpretable: cannot do anything.
+    output_option = FormatErrorOutputOption::kDoNothing;
+  } else if (input_validity != AudioFormatValidity::kValidAndSupported &&
+             input_validity !=
+                 AudioFormatValidity::kValidButUnsupportedSampleRate) {
+    // The input format is uninterpretable: cannot use it, must output silence.
+    output_option = FormatErrorOutputOption::kOutputSilence;
+  } else if (input_config.sample_rate_hz() != output_config.sample_rate_hz()) {
+    // Sample rates do not match: Cannot copy input into output, output silence.
+    // Note: If the sample rates are in a supported range, we could resample.
+    // However, that would significantly increase complexity of this error
+    // handling code.
+    output_option = FormatErrorOutputOption::kOutputSilence;
+  } else if (input_config.num_channels() != output_config.num_channels()) {
+    // Channel counts do not match: We cannot easily map input channels to
+    // output channels.
+    output_option =
+        FormatErrorOutputOption::kOutputBroadcastCopyOfFirstInputChannel;
+  } else {
+    // The formats match exactly.
+    RTC_DCHECK(input_config == output_config);
+    output_option = FormatErrorOutputOption::kOutputExactCopyOfInput;
+  }
+  return std::make_pair(error_code, output_option);
+}
+
+// Checks if the audio format is supported. If not, the output is populated in a
+// best-effort manner and an APM error code is returned.
+int HandleUnsupportedAudioFormats(const int16_t* const src,
+                                  const StreamConfig& input_config,
+                                  const StreamConfig& output_config,
+                                  int16_t* const dest) {
+  RTC_DCHECK(src);
+  RTC_DCHECK(dest);
+
+  auto [error_code, output_option] =
+      ChooseErrorOutputOption(input_config, output_config);
+  if (error_code == AudioProcessing::kNoError)
+    return AudioProcessing::kNoError;
+
+  const size_t num_output_channels = output_config.num_channels();
+  switch (output_option) {
+    case FormatErrorOutputOption::kOutputSilence:
+      memset(dest, 0, output_config.num_samples() * sizeof(int16_t));
+      break;
+    case FormatErrorOutputOption::kOutputBroadcastCopyOfFirstInputChannel:
+      for (size_t i = 0; i < output_config.num_frames(); ++i) {
+        int16_t sample = src[input_config.num_channels() * i];
+        for (size_t ch = 0; ch < num_output_channels; ++ch) {
+          dest[ch + num_output_channels * i] = sample;
+        }
+      }
+      break;
+    case FormatErrorOutputOption::kOutputExactCopyOfInput:
+      memcpy(dest, src, output_config.num_samples() * sizeof(int16_t));
+      break;
+    case FormatErrorOutputOption::kDoNothing:
+      break;
+  }
+  return error_code;
+}
+
+// Checks if the audio format is supported. If not, the output is populated in a
+// best-effort manner and an APM error code is returned.
+int HandleUnsupportedAudioFormats(const float* const* src,
+                                  const StreamConfig& input_config,
+                                  const StreamConfig& output_config,
+                                  float* const* dest) {
+  RTC_DCHECK(src);
+  RTC_DCHECK(dest);
+  for (size_t i = 0; i < input_config.num_channels(); ++i) {
+    RTC_DCHECK(src[i]);
+  }
+  for (size_t i = 0; i < output_config.num_channels(); ++i) {
+    RTC_DCHECK(dest[i]);
+  }
+
+  auto [error_code, output_option] =
+      ChooseErrorOutputOption(input_config, output_config);
+  if (error_code == AudioProcessing::kNoError)
+    return AudioProcessing::kNoError;
+
+  const size_t num_output_channels = output_config.num_channels();
+  switch (output_option) {
+    case FormatErrorOutputOption::kOutputSilence:
+      for (size_t ch = 0; ch < num_output_channels; ++ch) {
+        memset(dest[ch], 0, output_config.num_frames() * sizeof(float));
+      }
+      break;
+    case FormatErrorOutputOption::kOutputBroadcastCopyOfFirstInputChannel:
+      for (size_t ch = 0; ch < num_output_channels; ++ch) {
+        memcpy(dest[ch], src[0], output_config.num_frames() * sizeof(float));
+      }
+      break;
+    case FormatErrorOutputOption::kOutputExactCopyOfInput:
+      for (size_t ch = 0; ch < num_output_channels; ++ch) {
+        memcpy(dest[ch], src[ch], output_config.num_frames() * sizeof(float));
+      }
+      break;
+    case FormatErrorOutputOption::kDoNothing:
+      break;
+  }
+  return error_code;
+}
+
+using DownmixMethod = AudioProcessing::Config::Pipeline::DownmixMethod;
+
+void SetDownmixMethod(AudioBuffer& buffer, DownmixMethod method) {
+  switch (method) {
+    case DownmixMethod::kAverageChannels:
+      buffer.set_downmixing_by_averaging();
+      break;
+    case DownmixMethod::kUseFirstChannel:
+      buffer.set_downmixing_to_specific_channel(/*channel=*/0);
+      break;
+  }
+}
+
+constexpr int kUnspecifiedDataDumpInputVolume = -100;
+
+}  // namespace
+
+// Throughout webrtc, it's assumed that success is represented by zero.
+static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
+
+absl::optional<AudioProcessingImpl::GainController2ExperimentParams>
+AudioProcessingImpl::GetGainController2ExperimentParams() {
+  constexpr char kFieldTrialName[] = "WebRTC-Audio-GainController2";
+
+  if (!field_trial::IsEnabled(kFieldTrialName)) {
+    return absl::nullopt;
+  }
+
+  FieldTrialFlag enabled("Enabled", false);
+
+  // Whether the gain control should switch to AGC2. Enabled by default.
+  FieldTrialParameter<bool> switch_to_agc2("switch_to_agc2", true);
+
+  // AGC2 input volume controller configuration.
+  constexpr InputVolumeController::Config kDefaultInputVolumeControllerConfig;
+  FieldTrialConstrained<int> min_input_volume(
+      "min_input_volume", kDefaultInputVolumeControllerConfig.min_input_volume,
+      0, 255);
+  FieldTrialConstrained<int> clipped_level_min(
+      "clipped_level_min",
+      kDefaultInputVolumeControllerConfig.clipped_level_min, 0, 255);
+  FieldTrialConstrained<int> clipped_level_step(
+      "clipped_level_step",
+      kDefaultInputVolumeControllerConfig.clipped_level_step, 0, 255);
+  FieldTrialConstrained<double> clipped_ratio_threshold(
+      "clipped_ratio_threshold",
+      kDefaultInputVolumeControllerConfig.clipped_ratio_threshold, 0, 1);
+  FieldTrialConstrained<int> clipped_wait_frames(
+      "clipped_wait_frames",
+      kDefaultInputVolumeControllerConfig.clipped_wait_frames, 0,
+      absl::nullopt);
+  FieldTrialParameter<bool> enable_clipping_predictor(
+      "enable_clipping_predictor",
+      kDefaultInputVolumeControllerConfig.enable_clipping_predictor);
+  FieldTrialConstrained<int> target_range_max_dbfs(
+      "target_range_max_dbfs",
+      kDefaultInputVolumeControllerConfig.target_range_max_dbfs, -90, 30);
+  FieldTrialConstrained<int> target_range_min_dbfs(
+      "target_range_min_dbfs",
+      kDefaultInputVolumeControllerConfig.target_range_min_dbfs, -90, 30);
+  FieldTrialConstrained<int> update_input_volume_wait_frames(
+      "update_input_volume_wait_frames",
+      kDefaultInputVolumeControllerConfig.update_input_volume_wait_frames, 0,
+      absl::nullopt);
+  FieldTrialConstrained<double> speech_probability_threshold(
+      "speech_probability_threshold",
+      kDefaultInputVolumeControllerConfig.speech_probability_threshold, 0, 1);
+  FieldTrialConstrained<double> speech_ratio_threshold(
+      "speech_ratio_threshold",
+      kDefaultInputVolumeControllerConfig.speech_ratio_threshold, 0, 1);
+
+  // AGC2 adaptive digital controller configuration.
+  constexpr AudioProcessing::Config::GainController2::AdaptiveDigital
+      kDefaultAdaptiveDigitalConfig;
+  FieldTrialConstrained<double> headroom_db(
+      "headroom_db", kDefaultAdaptiveDigitalConfig.headroom_db, 0,
+      absl::nullopt);
+  FieldTrialConstrained<double> max_gain_db(
+      "max_gain_db", kDefaultAdaptiveDigitalConfig.max_gain_db, 0,
+      absl::nullopt);
+  FieldTrialConstrained<double> initial_gain_db(
+      "initial_gain_db", kDefaultAdaptiveDigitalConfig.initial_gain_db, 0,
+      absl::nullopt);
+  FieldTrialConstrained<double> max_gain_change_db_per_second(
+      "max_gain_change_db_per_second",
+      kDefaultAdaptiveDigitalConfig.max_gain_change_db_per_second, 0,
+      absl::nullopt);
+  FieldTrialConstrained<double> max_output_noise_level_dbfs(
+      "max_output_noise_level_dbfs",
+      kDefaultAdaptiveDigitalConfig.max_output_noise_level_dbfs, absl::nullopt,
+      0);
+
+  // Transient suppressor.
+  FieldTrialParameter<bool> disallow_transient_suppressor_usage(
+      "disallow_transient_suppressor_usage", false);
+
+  // Field-trial based override for the input volume controller and adaptive
+  // digital configs.
+  ParseFieldTrial(
+      {&enabled, &switch_to_agc2, &min_input_volume, &clipped_level_min,
+       &clipped_level_step, &clipped_ratio_threshold, &clipped_wait_frames,
+       &enable_clipping_predictor, &target_range_max_dbfs,
+       &target_range_min_dbfs, &update_input_volume_wait_frames,
+       &speech_probability_threshold, &speech_ratio_threshold, &headroom_db,
+       &max_gain_db, &initial_gain_db, &max_gain_change_db_per_second,
+       &max_output_noise_level_dbfs, &disallow_transient_suppressor_usage},
+      field_trial::FindFullName(kFieldTrialName));
+  // Checked already by `IsEnabled()` before parsing, therefore always true.
+  RTC_DCHECK(enabled);
+
+  const bool do_not_change_agc_config = !switch_to_agc2.Get();
+  if (do_not_change_agc_config && !disallow_transient_suppressor_usage.Get()) {
+    // Return an unspecifed value since, in this case, both the AGC2 and TS
+    // configurations won't be adjusted.
+    return absl::nullopt;
+  }
+  using Params = AudioProcessingImpl::GainController2ExperimentParams;
+  if (do_not_change_agc_config) {
+    // Return a value that leaves the AGC2 config unchanged and that always
+    // disables TS.
+    return Params{.agc2_config = absl::nullopt,
+                  .disallow_transient_suppressor_usage = true};
+  }
+  // Return a value that switches all the gain control to AGC2.
+  return Params{
+      .agc2_config =
+          Params::Agc2Config{
+              .input_volume_controller =
+                  {
+                      .min_input_volume = min_input_volume.Get(),
+                      .clipped_level_min = clipped_level_min.Get(),
+                      .clipped_level_step = clipped_level_step.Get(),
+                      .clipped_ratio_threshold =
+                          static_cast<float>(clipped_ratio_threshold.Get()),
+                      .clipped_wait_frames = clipped_wait_frames.Get(),
+                      .enable_clipping_predictor =
+                          enable_clipping_predictor.Get(),
+                      .target_range_max_dbfs = target_range_max_dbfs.Get(),
+                      .target_range_min_dbfs = target_range_min_dbfs.Get(),
+                      .update_input_volume_wait_frames =
+                          update_input_volume_wait_frames.Get(),
+                      .speech_probability_threshold = static_cast<float>(
+                          speech_probability_threshold.Get()),
+                      .speech_ratio_threshold =
+                          static_cast<float>(speech_ratio_threshold.Get()),
+                  },
+              .adaptive_digital_controller =
+                  {
+                      .enabled = false,
+                      .headroom_db = static_cast<float>(headroom_db.Get()),
+                      .max_gain_db = static_cast<float>(max_gain_db.Get()),
+                      .initial_gain_db =
+                          static_cast<float>(initial_gain_db.Get()),
+                      .max_gain_change_db_per_second = static_cast<float>(
+                          max_gain_change_db_per_second.Get()),
+                      .max_output_noise_level_dbfs =
+                          static_cast<float>(max_output_noise_level_dbfs.Get()),
+                  }},
+      .disallow_transient_suppressor_usage =
+          disallow_transient_suppressor_usage.Get()};
+}
+
+AudioProcessing::Config AudioProcessingImpl::AdjustConfig(
+    const AudioProcessing::Config& config,
+    const absl::optional<AudioProcessingImpl::GainController2ExperimentParams>&
+        experiment_params) {
+  if (!experiment_params.has_value() ||
+      (!experiment_params->agc2_config.has_value() &&
+       !experiment_params->disallow_transient_suppressor_usage)) {
+    // When the experiment parameters are unspecified or when the AGC and TS
+    // configuration are not overridden, return the unmodified configuration.
+    return config;
+  }
+
+  AudioProcessing::Config adjusted_config = config;
+
+  // Override the transient suppressor configuration.
+  if (experiment_params->disallow_transient_suppressor_usage) {
+    adjusted_config.transient_suppression.enabled = false;
+  }
+
+  // Override the auto gain control configuration if the AGC1 analog gain
+  // controller is active and `experiment_params->agc2_config` is specified.
+  const bool agc1_analog_enabled =
+      config.gain_controller1.enabled &&
+      (config.gain_controller1.mode ==
+           AudioProcessing::Config::GainController1::kAdaptiveAnalog ||
+       config.gain_controller1.analog_gain_controller.enabled);
+  if (agc1_analog_enabled && experiment_params->agc2_config.has_value()) {
+    // Check that the unadjusted AGC config meets the preconditions.
+    const bool hybrid_agc_config_detected =
+        config.gain_controller1.enabled &&
+        config.gain_controller1.analog_gain_controller.enabled &&
+        !config.gain_controller1.analog_gain_controller
+             .enable_digital_adaptive &&
+        config.gain_controller2.enabled &&
+        config.gain_controller2.adaptive_digital.enabled;
+    const bool full_agc1_config_detected =
+        config.gain_controller1.enabled &&
+        config.gain_controller1.analog_gain_controller.enabled &&
+        config.gain_controller1.analog_gain_controller
+            .enable_digital_adaptive &&
+        !config.gain_controller2.enabled;
+    const bool one_and_only_one_input_volume_controller =
+        hybrid_agc_config_detected != full_agc1_config_detected;
+    const bool agc2_input_volume_controller_enabled =
+        config.gain_controller2.enabled &&
+        config.gain_controller2.input_volume_controller.enabled;
+    if (!one_and_only_one_input_volume_controller ||
+        agc2_input_volume_controller_enabled) {
+      RTC_LOG(LS_ERROR) << "Cannot adjust AGC config (precondition failed)";
+      if (!one_and_only_one_input_volume_controller)
+        RTC_LOG(LS_ERROR)
+            << "One and only one input volume controller must be enabled.";
+      if (agc2_input_volume_controller_enabled)
+        RTC_LOG(LS_ERROR)
+            << "The AGC2 input volume controller must be disabled.";
+    } else {
+      adjusted_config.gain_controller1.enabled = false;
+      adjusted_config.gain_controller1.analog_gain_controller.enabled = false;
+
+      adjusted_config.gain_controller2.enabled = true;
+      adjusted_config.gain_controller2.input_volume_controller.enabled = true;
+      adjusted_config.gain_controller2.adaptive_digital =
+          experiment_params->agc2_config->adaptive_digital_controller;
+      adjusted_config.gain_controller2.adaptive_digital.enabled = true;
+    }
+  }
+
+  return adjusted_config;
+}
+
+bool AudioProcessingImpl::UseApmVadSubModule(
+    const AudioProcessing::Config& config,
+    const absl::optional<GainController2ExperimentParams>& experiment_params) {
+  // The VAD as an APM sub-module is needed only in one case, that is when TS
+  // and AGC2 are both enabled and when the AGC2 experiment is running and its
+  // parameters require to fully switch the gain control to AGC2.
+  return config.transient_suppression.enabled &&
+         config.gain_controller2.enabled &&
+         (config.gain_controller2.input_volume_controller.enabled ||
+          config.gain_controller2.adaptive_digital.enabled) &&
+         experiment_params.has_value() &&
+         experiment_params->agc2_config.has_value();
+}
+
+AudioProcessingImpl::SubmoduleStates::SubmoduleStates(
+    bool capture_post_processor_enabled,
+    bool render_pre_processor_enabled,
+    bool capture_analyzer_enabled)
+    : capture_post_processor_enabled_(capture_post_processor_enabled),
+      render_pre_processor_enabled_(render_pre_processor_enabled),
+      capture_analyzer_enabled_(capture_analyzer_enabled) {}
+
+bool AudioProcessingImpl::SubmoduleStates::Update(
+    bool high_pass_filter_enabled,
+    bool mobile_echo_controller_enabled,
+    bool noise_suppressor_enabled,
+    bool adaptive_gain_controller_enabled,
+    bool gain_controller2_enabled,
+    bool voice_activity_detector_enabled,
+    bool gain_adjustment_enabled,
+    bool echo_controller_enabled,
+    bool transient_suppressor_enabled) {
+  bool changed = false;
+  changed |= (high_pass_filter_enabled != high_pass_filter_enabled_);
+  changed |=
+      (mobile_echo_controller_enabled != mobile_echo_controller_enabled_);
+  changed |= (noise_suppressor_enabled != noise_suppressor_enabled_);
+  changed |=
+      (adaptive_gain_controller_enabled != adaptive_gain_controller_enabled_);
+  changed |= (gain_controller2_enabled != gain_controller2_enabled_);
+  changed |=
+      (voice_activity_detector_enabled != voice_activity_detector_enabled_);
+  changed |= (gain_adjustment_enabled != gain_adjustment_enabled_);
+  changed |= (echo_controller_enabled != echo_controller_enabled_);
+  changed |= (transient_suppressor_enabled != transient_suppressor_enabled_);
+  if (changed) {
+    high_pass_filter_enabled_ = high_pass_filter_enabled;
+    mobile_echo_controller_enabled_ = mobile_echo_controller_enabled;
+    noise_suppressor_enabled_ = noise_suppressor_enabled;
+    adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled;
+    gain_controller2_enabled_ = gain_controller2_enabled;
+    voice_activity_detector_enabled_ = voice_activity_detector_enabled;
+    gain_adjustment_enabled_ = gain_adjustment_enabled;
+    echo_controller_enabled_ = echo_controller_enabled;
+    transient_suppressor_enabled_ = transient_suppressor_enabled;
+  }
+
+  changed |= first_update_;
+  first_update_ = false;
+  return changed;
+}
+
+bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandSubModulesActive()
+    const {
+  return CaptureMultiBandProcessingPresent();
+}
+
+bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandProcessingPresent()
+    const {
+  // If echo controller is present, assume it performs active processing.
+  return CaptureMultiBandProcessingActive(/*ec_processing_active=*/true);
+}
+
+bool AudioProcessingImpl::SubmoduleStates::CaptureMultiBandProcessingActive(
+    bool ec_processing_active) const {
+  return high_pass_filter_enabled_ || mobile_echo_controller_enabled_ ||
+         noise_suppressor_enabled_ || adaptive_gain_controller_enabled_ ||
+         (echo_controller_enabled_ && ec_processing_active);
+}
+
+bool AudioProcessingImpl::SubmoduleStates::CaptureFullBandProcessingActive()
+    const {
+  return gain_controller2_enabled_ || capture_post_processor_enabled_ ||
+         gain_adjustment_enabled_;
+}
+
+bool AudioProcessingImpl::SubmoduleStates::CaptureAnalyzerActive() const {
+  return capture_analyzer_enabled_;
+}
+
+bool AudioProcessingImpl::SubmoduleStates::RenderMultiBandSubModulesActive()
+    const {
+  return RenderMultiBandProcessingActive() || mobile_echo_controller_enabled_ ||
+         adaptive_gain_controller_enabled_ || echo_controller_enabled_;
+}
+
+bool AudioProcessingImpl::SubmoduleStates::RenderFullBandProcessingActive()
+    const {
+  return render_pre_processor_enabled_;
+}
+
+bool AudioProcessingImpl::SubmoduleStates::RenderMultiBandProcessingActive()
+    const {
+  return false;
+}
+
+bool AudioProcessingImpl::SubmoduleStates::HighPassFilteringRequired() const {
+  return high_pass_filter_enabled_ || mobile_echo_controller_enabled_ ||
+         noise_suppressor_enabled_;
+}
+
+AudioProcessingImpl::AudioProcessingImpl()
+    : AudioProcessingImpl(/*config=*/{},
+                          /*capture_post_processor=*/nullptr,
+                          /*render_pre_processor=*/nullptr,
+                          /*echo_control_factory=*/nullptr,
+                          /*echo_detector=*/nullptr,
+                          /*capture_analyzer=*/nullptr) {}
+
+std::atomic<int> AudioProcessingImpl::instance_count_(0);
+
+AudioProcessingImpl::AudioProcessingImpl(
+    const AudioProcessing::Config& config,
+    std::unique_ptr<CustomProcessing> capture_post_processor,
+    std::unique_ptr<CustomProcessing> render_pre_processor,
+    std::unique_ptr<EchoControlFactory> echo_control_factory,
+    rtc::scoped_refptr<EchoDetector> echo_detector,
+    std::unique_ptr<CustomAudioAnalyzer> capture_analyzer)
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      use_setup_specific_default_aec3_config_(
+          UseSetupSpecificDefaultAec3Congfig()),
+      gain_controller2_experiment_params_(GetGainController2ExperimentParams()),
+      transient_suppressor_vad_mode_(TransientSuppressor::VadMode::kDefault),
+      capture_runtime_settings_(RuntimeSettingQueueSize()),
+      render_runtime_settings_(RuntimeSettingQueueSize()),
+      capture_runtime_settings_enqueuer_(&capture_runtime_settings_),
+      render_runtime_settings_enqueuer_(&render_runtime_settings_),
+      echo_control_factory_(std::move(echo_control_factory)),
+      config_(AdjustConfig(config, gain_controller2_experiment_params_)),
+      submodule_states_(!!capture_post_processor,
+                        !!render_pre_processor,
+                        !!capture_analyzer),
+      submodules_(std::move(capture_post_processor),
+                  std::move(render_pre_processor),
+                  std::move(echo_detector),
+                  std::move(capture_analyzer)),
+      constants_(!field_trial::IsEnabled(
+                     "WebRTC-ApmExperimentalMultiChannelRenderKillSwitch"),
+                 !field_trial::IsEnabled(
+                     "WebRTC-ApmExperimentalMultiChannelCaptureKillSwitch"),
+                 EnforceSplitBandHpf(),
+                 MinimizeProcessingForUnusedOutput(),
+                 field_trial::IsEnabled("WebRTC-TransientSuppressorForcedOff")),
+      capture_(),
+      capture_nonlocked_(),
+      applied_input_volume_stats_reporter_(
+          InputVolumeStatsReporter::InputVolumeType::kApplied),
+      recommended_input_volume_stats_reporter_(
+          InputVolumeStatsReporter::InputVolumeType::kRecommended) {
+  RTC_LOG(LS_INFO) << "Injected APM submodules:"
+                      "\nEcho control factory: "
+                   << !!echo_control_factory_
+                   << "\nEcho detector: " << !!submodules_.echo_detector
+                   << "\nCapture analyzer: " << !!submodules_.capture_analyzer
+                   << "\nCapture post processor: "
+                   << !!submodules_.capture_post_processor
+                   << "\nRender pre processor: "
+                   << !!submodules_.render_pre_processor;
+  if (!DenormalDisabler::IsSupported()) {
+    RTC_LOG(LS_INFO) << "Denormal disabler unsupported";
+  }
+
+  RTC_LOG(LS_INFO) << "AudioProcessing: " << config_.ToString();
+
+  // Mark Echo Controller enabled if a factory is injected.
+  capture_nonlocked_.echo_controller_enabled =
+      static_cast<bool>(echo_control_factory_);
+
+  Initialize();
+}
+
+AudioProcessingImpl::~AudioProcessingImpl() = default;
+
+int AudioProcessingImpl::Initialize() {
+  // Run in a single-threaded manner during initialization.
+  MutexLock lock_render(&mutex_render_);
+  MutexLock lock_capture(&mutex_capture_);
+  InitializeLocked();
+  return kNoError;
+}
+
+int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
+  // Run in a single-threaded manner during initialization.
+  MutexLock lock_render(&mutex_render_);
+  MutexLock lock_capture(&mutex_capture_);
+  InitializeLocked(processing_config);
+  return kNoError;
+}
+
+void AudioProcessingImpl::MaybeInitializeRender(
+    const StreamConfig& input_config,
+    const StreamConfig& output_config) {
+  ProcessingConfig processing_config = formats_.api_format;
+  processing_config.reverse_input_stream() = input_config;
+  processing_config.reverse_output_stream() = output_config;
+
+  if (processing_config == formats_.api_format) {
+    return;
+  }
+
+  MutexLock lock_capture(&mutex_capture_);
+  InitializeLocked(processing_config);
+}
+
+void AudioProcessingImpl::InitializeLocked() {
+  UpdateActiveSubmoduleStates();
+
+  const int render_audiobuffer_sample_rate_hz =
+      formats_.api_format.reverse_output_stream().num_frames() == 0
+          ? formats_.render_processing_format.sample_rate_hz()
+          : formats_.api_format.reverse_output_stream().sample_rate_hz();
+  if (formats_.api_format.reverse_input_stream().num_channels() > 0) {
+    render_.render_audio.reset(new AudioBuffer(
+        formats_.api_format.reverse_input_stream().sample_rate_hz(),
+        formats_.api_format.reverse_input_stream().num_channels(),
+        formats_.render_processing_format.sample_rate_hz(),
+        formats_.render_processing_format.num_channels(),
+        render_audiobuffer_sample_rate_hz,
+        formats_.render_processing_format.num_channels()));
+    if (formats_.api_format.reverse_input_stream() !=
+        formats_.api_format.reverse_output_stream()) {
+      render_.render_converter = AudioConverter::Create(
+          formats_.api_format.reverse_input_stream().num_channels(),
+          formats_.api_format.reverse_input_stream().num_frames(),
+          formats_.api_format.reverse_output_stream().num_channels(),
+          formats_.api_format.reverse_output_stream().num_frames());
+    } else {
+      render_.render_converter.reset(nullptr);
+    }
+  } else {
+    render_.render_audio.reset(nullptr);
+    render_.render_converter.reset(nullptr);
+  }
+
+  capture_.capture_audio.reset(new AudioBuffer(
+      formats_.api_format.input_stream().sample_rate_hz(),
+      formats_.api_format.input_stream().num_channels(),
+      capture_nonlocked_.capture_processing_format.sample_rate_hz(),
+      formats_.api_format.output_stream().num_channels(),
+      formats_.api_format.output_stream().sample_rate_hz(),
+      formats_.api_format.output_stream().num_channels()));
+  SetDownmixMethod(*capture_.capture_audio,
+                   config_.pipeline.capture_downmix_method);
+
+  if (capture_nonlocked_.capture_processing_format.sample_rate_hz() <
+          formats_.api_format.output_stream().sample_rate_hz() &&
+      formats_.api_format.output_stream().sample_rate_hz() == 48000) {
+    capture_.capture_fullband_audio.reset(
+        new AudioBuffer(formats_.api_format.input_stream().sample_rate_hz(),
+                        formats_.api_format.input_stream().num_channels(),
+                        formats_.api_format.output_stream().sample_rate_hz(),
+                        formats_.api_format.output_stream().num_channels(),
+                        formats_.api_format.output_stream().sample_rate_hz(),
+                        formats_.api_format.output_stream().num_channels()));
+    SetDownmixMethod(*capture_.capture_fullband_audio,
+                     config_.pipeline.capture_downmix_method);
+  } else {
+    capture_.capture_fullband_audio.reset();
+  }
+
+  AllocateRenderQueue();
+
+  InitializeGainController1();
+  InitializeTransientSuppressor();
+  InitializeHighPassFilter(true);
+  InitializeResidualEchoDetector();
+  InitializeEchoController();
+  InitializeGainController2();
+  InitializeVoiceActivityDetector();
+  InitializeNoiseSuppressor();
+  InitializeAnalyzer();
+  InitializePostProcessor();
+  InitializePreProcessor();
+  InitializeCaptureLevelsAdjuster();
+
+  if (aec_dump_) {
+    aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis());
+  }
+}
+
+void AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
+  UpdateActiveSubmoduleStates();
+
+  formats_.api_format = config;
+
+  // Choose maximum rate to use for the split filtering.
+  RTC_DCHECK(config_.pipeline.maximum_internal_processing_rate == 48000 ||
+             config_.pipeline.maximum_internal_processing_rate == 32000);
+  int max_splitting_rate = 48000;
+  if (config_.pipeline.maximum_internal_processing_rate == 32000) {
+    max_splitting_rate = config_.pipeline.maximum_internal_processing_rate;
+  }
+
+  int capture_processing_rate = SuitableProcessRate(
+      std::min(formats_.api_format.input_stream().sample_rate_hz(),
+               formats_.api_format.output_stream().sample_rate_hz()),
+      max_splitting_rate,
+      submodule_states_.CaptureMultiBandSubModulesActive() ||
+          submodule_states_.RenderMultiBandSubModulesActive());
+  RTC_DCHECK_NE(8000, capture_processing_rate);
+
+  capture_nonlocked_.capture_processing_format =
+      StreamConfig(capture_processing_rate);
+
+  int render_processing_rate;
+  if (!capture_nonlocked_.echo_controller_enabled) {
+    render_processing_rate = SuitableProcessRate(
+        std::min(formats_.api_format.reverse_input_stream().sample_rate_hz(),
+                 formats_.api_format.reverse_output_stream().sample_rate_hz()),
+        max_splitting_rate,
+        submodule_states_.CaptureMultiBandSubModulesActive() ||
+            submodule_states_.RenderMultiBandSubModulesActive());
+  } else {
+    render_processing_rate = capture_processing_rate;
+  }
+
+  // If the forward sample rate is 8 kHz, the render stream is also processed
+  // at this rate.
+  if (capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
+      kSampleRate8kHz) {
+    render_processing_rate = kSampleRate8kHz;
+  } else {
+    render_processing_rate =
+        std::max(render_processing_rate, static_cast<int>(kSampleRate16kHz));
+  }
+
+  RTC_DCHECK_NE(8000, render_processing_rate);
+
+  if (submodule_states_.RenderMultiBandSubModulesActive()) {
+    // By default, downmix the render stream to mono for analysis. This has been
+    // demonstrated to work well for AEC in most practical scenarios.
+    const bool multi_channel_render = config_.pipeline.multi_channel_render &&
+                                      constants_.multi_channel_render_support;
+    int render_processing_num_channels =
+        multi_channel_render
+            ? formats_.api_format.reverse_input_stream().num_channels()
+            : 1;
+    formats_.render_processing_format =
+        StreamConfig(render_processing_rate, render_processing_num_channels);
+  } else {
+    formats_.render_processing_format = StreamConfig(
+        formats_.api_format.reverse_input_stream().sample_rate_hz(),
+        formats_.api_format.reverse_input_stream().num_channels());
+  }
+
+  if (capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
+          kSampleRate32kHz ||
+      capture_nonlocked_.capture_processing_format.sample_rate_hz() ==
+          kSampleRate48kHz) {
+    capture_nonlocked_.split_rate = kSampleRate16kHz;
+  } else {
+    capture_nonlocked_.split_rate =
+        capture_nonlocked_.capture_processing_format.sample_rate_hz();
+  }
+
+  InitializeLocked();
+}
+
+void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
+  // Run in a single-threaded manner when applying the settings.
+  MutexLock lock_render(&mutex_render_);
+  MutexLock lock_capture(&mutex_capture_);
+
+  const auto adjusted_config =
+      AdjustConfig(config, gain_controller2_experiment_params_);
+  RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: "
+                   << adjusted_config.ToString();
+
+  const bool pipeline_config_changed =
+      config_.pipeline.multi_channel_render !=
+          adjusted_config.pipeline.multi_channel_render ||
+      config_.pipeline.multi_channel_capture !=
+          adjusted_config.pipeline.multi_channel_capture ||
+      config_.pipeline.maximum_internal_processing_rate !=
+          adjusted_config.pipeline.maximum_internal_processing_rate ||
+      config_.pipeline.capture_downmix_method !=
+          adjusted_config.pipeline.capture_downmix_method;
+
+  const bool aec_config_changed =
+      config_.echo_canceller.enabled !=
+          adjusted_config.echo_canceller.enabled ||
+      config_.echo_canceller.mobile_mode !=
+          adjusted_config.echo_canceller.mobile_mode;
+
+  const bool agc1_config_changed =
+      config_.gain_controller1 != adjusted_config.gain_controller1;
+
+  const bool agc2_config_changed =
+      config_.gain_controller2 != adjusted_config.gain_controller2;
+
+  const bool ns_config_changed =
+      config_.noise_suppression.enabled !=
+          adjusted_config.noise_suppression.enabled ||
+      config_.noise_suppression.level !=
+          adjusted_config.noise_suppression.level;
+
+  const bool ts_config_changed = config_.transient_suppression.enabled !=
+                                 adjusted_config.transient_suppression.enabled;
+
+  const bool pre_amplifier_config_changed =
+      config_.pre_amplifier.enabled != adjusted_config.pre_amplifier.enabled ||
+      config_.pre_amplifier.fixed_gain_factor !=
+          adjusted_config.pre_amplifier.fixed_gain_factor;
+
+  const bool gain_adjustment_config_changed =
+      config_.capture_level_adjustment !=
+      adjusted_config.capture_level_adjustment;
+
+  config_ = adjusted_config;
+
+  if (aec_config_changed) {
+    InitializeEchoController();
+  }
+
+  if (ns_config_changed) {
+    InitializeNoiseSuppressor();
+  }
+
+  if (ts_config_changed) {
+    InitializeTransientSuppressor();
+  }
+
+  InitializeHighPassFilter(false);
+
+  if (agc1_config_changed) {
+    InitializeGainController1();
+  }
+
+  const bool config_ok = GainController2::Validate(config_.gain_controller2);
+  if (!config_ok) {
+    RTC_LOG(LS_ERROR)
+        << "Invalid Gain Controller 2 config; using the default config.";
+    config_.gain_controller2 = AudioProcessing::Config::GainController2();
+  }
+
+  if (agc2_config_changed || ts_config_changed) {
+    // AGC2 also depends on TS because of the possible dependency on the APM VAD
+    // sub-module.
+    InitializeGainController2();
+    InitializeVoiceActivityDetector();
+  }
+
+  if (pre_amplifier_config_changed || gain_adjustment_config_changed) {
+    InitializeCaptureLevelsAdjuster();
+  }
+
+  // Reinitialization must happen after all submodule configuration to avoid
+  // additional reinitializations on the next capture / render processing call.
+  if (pipeline_config_changed) {
+    InitializeLocked(formats_.api_format);
+  }
+}
+
+void AudioProcessingImpl::OverrideSubmoduleCreationForTesting(
+    const ApmSubmoduleCreationOverrides& overrides) {
+  MutexLock lock(&mutex_capture_);
+  submodule_creation_overrides_ = overrides;
+}
+
+int AudioProcessingImpl::proc_sample_rate_hz() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.capture_processing_format.sample_rate_hz();
+}
+
+int AudioProcessingImpl::proc_fullband_sample_rate_hz() const {
+  return capture_.capture_fullband_audio
+             ? capture_.capture_fullband_audio->num_frames() * 100
+             : capture_nonlocked_.capture_processing_format.sample_rate_hz();
+}
+
+int AudioProcessingImpl::proc_split_sample_rate_hz() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.split_rate;
+}
+
+size_t AudioProcessingImpl::num_reverse_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return formats_.render_processing_format.num_channels();
+}
+
+size_t AudioProcessingImpl::num_input_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return formats_.api_format.input_stream().num_channels();
+}
+
+size_t AudioProcessingImpl::num_proc_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  const bool multi_channel_capture = config_.pipeline.multi_channel_capture &&
+                                     constants_.multi_channel_capture_support;
+  if (capture_nonlocked_.echo_controller_enabled && !multi_channel_capture) {
+    return 1;
+  }
+  return num_output_channels();
+}
+
+size_t AudioProcessingImpl::num_output_channels() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return formats_.api_format.output_stream().num_channels();
+}
+
+void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
+  MutexLock lock(&mutex_capture_);
+  HandleCaptureOutputUsedSetting(!muted);
+}
+
+void AudioProcessingImpl::HandleCaptureOutputUsedSetting(
+    bool capture_output_used) {
+  capture_.capture_output_used =
+      capture_output_used || !constants_.minimize_processing_for_unused_output;
+
+  if (submodules_.agc_manager.get()) {
+    submodules_.agc_manager->HandleCaptureOutputUsedChange(
+        capture_.capture_output_used);
+  }
+  if (submodules_.echo_controller) {
+    submodules_.echo_controller->SetCaptureOutputUsage(
+        capture_.capture_output_used);
+  }
+  if (submodules_.noise_suppressor) {
+    submodules_.noise_suppressor->SetCaptureOutputUsage(
+        capture_.capture_output_used);
+  }
+  if (submodules_.gain_controller2) {
+    submodules_.gain_controller2->SetCaptureOutputUsed(
+        capture_.capture_output_used);
+  }
+}
+
+void AudioProcessingImpl::SetRuntimeSetting(RuntimeSetting setting) {
+  PostRuntimeSetting(setting);
+}
+
+bool AudioProcessingImpl::PostRuntimeSetting(RuntimeSetting setting) {
+  switch (setting.type()) {
+    case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
+    case RuntimeSetting::Type::kPlayoutAudioDeviceChange:
+      return render_runtime_settings_enqueuer_.Enqueue(setting);
+    case RuntimeSetting::Type::kCapturePreGain:
+    case RuntimeSetting::Type::kCapturePostGain:
+    case RuntimeSetting::Type::kCaptureCompressionGain:
+    case RuntimeSetting::Type::kCaptureFixedPostGain:
+    case RuntimeSetting::Type::kCaptureOutputUsed:
+      return capture_runtime_settings_enqueuer_.Enqueue(setting);
+    case RuntimeSetting::Type::kPlayoutVolumeChange: {
+      bool enqueueing_successful;
+      enqueueing_successful =
+          capture_runtime_settings_enqueuer_.Enqueue(setting);
+      enqueueing_successful =
+          render_runtime_settings_enqueuer_.Enqueue(setting) &&
+          enqueueing_successful;
+      return enqueueing_successful;
+    }
+    case RuntimeSetting::Type::kNotSpecified:
+      RTC_DCHECK_NOTREACHED();
+      return true;
+  }
+  // The language allows the enum to have a non-enumerator
+  // value. Check that this doesn't happen.
+  RTC_DCHECK_NOTREACHED();
+  return true;
+}
+
+AudioProcessingImpl::RuntimeSettingEnqueuer::RuntimeSettingEnqueuer(
+    SwapQueue<RuntimeSetting>* runtime_settings)
+    : runtime_settings_(*runtime_settings) {
+  RTC_DCHECK(runtime_settings);
+}
+
+AudioProcessingImpl::RuntimeSettingEnqueuer::~RuntimeSettingEnqueuer() =
+    default;
+
+bool AudioProcessingImpl::RuntimeSettingEnqueuer::Enqueue(
+    RuntimeSetting setting) {
+  const bool successful_insert = runtime_settings_.Insert(&setting);
+
+  if (!successful_insert) {
+    RTC_LOG(LS_ERROR) << "Cannot enqueue a new runtime setting.";
+  }
+  return successful_insert;
+}
+
+void AudioProcessingImpl::MaybeInitializeCapture(
+    const StreamConfig& input_config,
+    const StreamConfig& output_config) {
+  ProcessingConfig processing_config;
+  bool reinitialization_required = false;
+  {
+    // Acquire the capture lock in order to access api_format. The lock is
+    // released immediately, as we may need to acquire the render lock as part
+    // of the conditional reinitialization.
+    MutexLock lock_capture(&mutex_capture_);
+    processing_config = formats_.api_format;
+    reinitialization_required = UpdateActiveSubmoduleStates();
+  }
+
+  if (processing_config.input_stream() != input_config) {
+    reinitialization_required = true;
+  }
+
+  if (processing_config.output_stream() != output_config) {
+    reinitialization_required = true;
+  }
+
+  if (reinitialization_required) {
+    MutexLock lock_render(&mutex_render_);
+    MutexLock lock_capture(&mutex_capture_);
+    // Reread the API format since the render format may have changed.
+    processing_config = formats_.api_format;
+    processing_config.input_stream() = input_config;
+    processing_config.output_stream() = output_config;
+    InitializeLocked(processing_config);
+  }
+}
+
+int AudioProcessingImpl::ProcessStream(const float* const* src,
+                                       const StreamConfig& input_config,
+                                       const StreamConfig& output_config,
+                                       float* const* dest) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_StreamConfig");
+  DenormalDisabler denormal_disabler;
+  RETURN_ON_ERR(
+      HandleUnsupportedAudioFormats(src, input_config, output_config, dest));
+  MaybeInitializeCapture(input_config, output_config);
+
+  MutexLock lock_capture(&mutex_capture_);
+
+  if (aec_dump_) {
+    RecordUnprocessedCaptureStream(src);
+  }
+
+  capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream());
+  if (capture_.capture_fullband_audio) {
+    capture_.capture_fullband_audio->CopyFrom(
+        src, formats_.api_format.input_stream());
+  }
+  RETURN_ON_ERR(ProcessCaptureStreamLocked());
+  if (capture_.capture_fullband_audio) {
+    capture_.capture_fullband_audio->CopyTo(formats_.api_format.output_stream(),
+                                            dest);
+  } else {
+    capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
+  }
+
+  if (aec_dump_) {
+    RecordProcessedCaptureStream(dest);
+  }
+  return kNoError;
+}
+
+void AudioProcessingImpl::HandleCaptureRuntimeSettings() {
+  RuntimeSetting setting;
+  int num_settings_processed = 0;
+  while (capture_runtime_settings_.Remove(&setting)) {
+    if (aec_dump_) {
+      aec_dump_->WriteRuntimeSetting(setting);
+    }
+    switch (setting.type()) {
+      case RuntimeSetting::Type::kCapturePreGain:
+        if (config_.pre_amplifier.enabled ||
+            config_.capture_level_adjustment.enabled) {
+          float value;
+          setting.GetFloat(&value);
+          // If the pre-amplifier is used, apply the new gain to the
+          // pre-amplifier regardless if the capture level adjustment is
+          // activated. This approach allows both functionalities to coexist
+          // until they have been properly merged.
+          if (config_.pre_amplifier.enabled) {
+            config_.pre_amplifier.fixed_gain_factor = value;
+          } else {
+            config_.capture_level_adjustment.pre_gain_factor = value;
+          }
+
+          // Use both the pre-amplifier and the capture level adjustment gains
+          // as pre-gains.
+          float gain = 1.f;
+          if (config_.pre_amplifier.enabled) {
+            gain *= config_.pre_amplifier.fixed_gain_factor;
+          }
+          if (config_.capture_level_adjustment.enabled) {
+            gain *= config_.capture_level_adjustment.pre_gain_factor;
+          }
+
+          submodules_.capture_levels_adjuster->SetPreGain(gain);
+        }
+        // TODO(bugs.chromium.org/9138): Log setting handling by Aec Dump.
+        break;
+      case RuntimeSetting::Type::kCapturePostGain:
+        if (config_.capture_level_adjustment.enabled) {
+          float value;
+          setting.GetFloat(&value);
+          config_.capture_level_adjustment.post_gain_factor = value;
+          submodules_.capture_levels_adjuster->SetPostGain(
+              config_.capture_level_adjustment.post_gain_factor);
+        }
+        // TODO(bugs.chromium.org/9138): Log setting handling by Aec Dump.
+        break;
+      case RuntimeSetting::Type::kCaptureCompressionGain: {
+        if (!submodules_.agc_manager &&
+            !(submodules_.gain_controller2 &&
+              config_.gain_controller2.input_volume_controller.enabled)) {
+          float value;
+          setting.GetFloat(&value);
+          int int_value = static_cast<int>(value + .5f);
+          config_.gain_controller1.compression_gain_db = int_value;
+          if (submodules_.gain_control) {
+            int error =
+                submodules_.gain_control->set_compression_gain_db(int_value);
+            RTC_DCHECK_EQ(kNoError, error);
+          }
+        }
+        break;
+      }
+      case RuntimeSetting::Type::kCaptureFixedPostGain: {
+        if (submodules_.gain_controller2) {
+          float value;
+          setting.GetFloat(&value);
+          config_.gain_controller2.fixed_digital.gain_db = value;
+          submodules_.gain_controller2->SetFixedGainDb(value);
+        }
+        break;
+      }
+      case RuntimeSetting::Type::kPlayoutVolumeChange: {
+        int value;
+        setting.GetInt(&value);
+        capture_.playout_volume = value;
+        break;
+      }
+      case RuntimeSetting::Type::kPlayoutAudioDeviceChange:
+        RTC_DCHECK_NOTREACHED();
+        break;
+      case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
+        RTC_DCHECK_NOTREACHED();
+        break;
+      case RuntimeSetting::Type::kNotSpecified:
+        RTC_DCHECK_NOTREACHED();
+        break;
+      case RuntimeSetting::Type::kCaptureOutputUsed:
+        bool value;
+        setting.GetBool(&value);
+        HandleCaptureOutputUsedSetting(value);
+        break;
+    }
+    ++num_settings_processed;
+  }
+
+  if (num_settings_processed >= RuntimeSettingQueueSize()) {
+    // Handle overrun of the runtime settings queue, which likely will has
+    // caused settings to be discarded.
+    HandleOverrunInCaptureRuntimeSettingsQueue();
+  }
+}
+
+void AudioProcessingImpl::HandleOverrunInCaptureRuntimeSettingsQueue() {
+  // Fall back to a safe state for the case when a setting for capture output
+  // usage setting has been missed.
+  HandleCaptureOutputUsedSetting(/*capture_output_used=*/true);
+}
+
+void AudioProcessingImpl::HandleRenderRuntimeSettings() {
+  RuntimeSetting setting;
+  while (render_runtime_settings_.Remove(&setting)) {
+    if (aec_dump_) {
+      aec_dump_->WriteRuntimeSetting(setting);
+    }
+    switch (setting.type()) {
+      case RuntimeSetting::Type::kPlayoutAudioDeviceChange:  // fall-through
+      case RuntimeSetting::Type::kPlayoutVolumeChange:       // fall-through
+      case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
+        if (submodules_.render_pre_processor) {
+          submodules_.render_pre_processor->SetRuntimeSetting(setting);
+        }
+        break;
+      case RuntimeSetting::Type::kCapturePreGain:          // fall-through
+      case RuntimeSetting::Type::kCapturePostGain:         // fall-through
+      case RuntimeSetting::Type::kCaptureCompressionGain:  // fall-through
+      case RuntimeSetting::Type::kCaptureFixedPostGain:    // fall-through
+      case RuntimeSetting::Type::kCaptureOutputUsed:       // fall-through
+      case RuntimeSetting::Type::kNotSpecified:
+        RTC_DCHECK_NOTREACHED();
+        break;
+    }
+  }
+}
+
+void AudioProcessingImpl::QueueBandedRenderAudio(AudioBuffer* audio) {
+  RTC_DCHECK_GE(160, audio->num_frames_per_band());
+
+  if (submodules_.echo_control_mobile) {
+    EchoControlMobileImpl::PackRenderAudioBuffer(audio, num_output_channels(),
+                                                 num_reverse_channels(),
+                                                 &aecm_render_queue_buffer_);
+    RTC_DCHECK(aecm_render_signal_queue_);
+    // Insert the samples into the queue.
+    if (!aecm_render_signal_queue_->Insert(&aecm_render_queue_buffer_)) {
+      // The data queue is full and needs to be emptied.
+      EmptyQueuedRenderAudio();
+
+      // Retry the insert (should always work).
+      bool result =
+          aecm_render_signal_queue_->Insert(&aecm_render_queue_buffer_);
+      RTC_DCHECK(result);
+    }
+  }
+
+  if (!submodules_.agc_manager && submodules_.gain_control) {
+    GainControlImpl::PackRenderAudioBuffer(*audio, &agc_render_queue_buffer_);
+    // Insert the samples into the queue.
+    if (!agc_render_signal_queue_->Insert(&agc_render_queue_buffer_)) {
+      // The data queue is full and needs to be emptied.
+      EmptyQueuedRenderAudio();
+
+      // Retry the insert (should always work).
+      bool result = agc_render_signal_queue_->Insert(&agc_render_queue_buffer_);
+      RTC_DCHECK(result);
+    }
+  }
+}
+
+void AudioProcessingImpl::QueueNonbandedRenderAudio(AudioBuffer* audio) {
+  if (submodules_.echo_detector) {
+    PackRenderAudioBufferForEchoDetector(*audio, red_render_queue_buffer_);
+    RTC_DCHECK(red_render_signal_queue_);
+    // Insert the samples into the queue.
+    if (!red_render_signal_queue_->Insert(&red_render_queue_buffer_)) {
+      // The data queue is full and needs to be emptied.
+      EmptyQueuedRenderAudio();
+
+      // Retry the insert (should always work).
+      bool result = red_render_signal_queue_->Insert(&red_render_queue_buffer_);
+      RTC_DCHECK(result);
+    }
+  }
+}
+
+void AudioProcessingImpl::AllocateRenderQueue() {
+  const size_t new_agc_render_queue_element_max_size =
+      std::max(static_cast<size_t>(1), kMaxAllowedValuesOfSamplesPerBand);
+
+  const size_t new_red_render_queue_element_max_size =
+      std::max(static_cast<size_t>(1), kMaxAllowedValuesOfSamplesPerFrame);
+
+  // Reallocate the queues if the queue item sizes are too small to fit the
+  // data to put in the queues.
+
+  if (agc_render_queue_element_max_size_ <
+      new_agc_render_queue_element_max_size) {
+    agc_render_queue_element_max_size_ = new_agc_render_queue_element_max_size;
+
+    std::vector<int16_t> template_queue_element(
+        agc_render_queue_element_max_size_);
+
+    agc_render_signal_queue_.reset(
+        new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<int16_t>(
+                agc_render_queue_element_max_size_)));
+
+    agc_render_queue_buffer_.resize(agc_render_queue_element_max_size_);
+    agc_capture_queue_buffer_.resize(agc_render_queue_element_max_size_);
+  } else {
+    agc_render_signal_queue_->Clear();
+  }
+
+  if (submodules_.echo_detector) {
+    if (red_render_queue_element_max_size_ <
+        new_red_render_queue_element_max_size) {
+      red_render_queue_element_max_size_ =
+          new_red_render_queue_element_max_size;
+
+      std::vector<float> template_queue_element(
+          red_render_queue_element_max_size_);
+
+      red_render_signal_queue_.reset(
+          new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
+              kMaxNumFramesToBuffer, template_queue_element,
+              RenderQueueItemVerifier<float>(
+                  red_render_queue_element_max_size_)));
+
+      red_render_queue_buffer_.resize(red_render_queue_element_max_size_);
+      red_capture_queue_buffer_.resize(red_render_queue_element_max_size_);
+    } else {
+      red_render_signal_queue_->Clear();
+    }
+  }
+}
+
+void AudioProcessingImpl::EmptyQueuedRenderAudio() {
+  MutexLock lock_capture(&mutex_capture_);
+  EmptyQueuedRenderAudioLocked();
+}
+
+void AudioProcessingImpl::EmptyQueuedRenderAudioLocked() {
+  if (submodules_.echo_control_mobile) {
+    RTC_DCHECK(aecm_render_signal_queue_);
+    while (aecm_render_signal_queue_->Remove(&aecm_capture_queue_buffer_)) {
+      submodules_.echo_control_mobile->ProcessRenderAudio(
+          aecm_capture_queue_buffer_);
+    }
+  }
+
+  if (submodules_.gain_control) {
+    while (agc_render_signal_queue_->Remove(&agc_capture_queue_buffer_)) {
+      submodules_.gain_control->ProcessRenderAudio(agc_capture_queue_buffer_);
+    }
+  }
+
+  if (submodules_.echo_detector) {
+    while (red_render_signal_queue_->Remove(&red_capture_queue_buffer_)) {
+      submodules_.echo_detector->AnalyzeRenderAudio(red_capture_queue_buffer_);
+    }
+  }
+}
+
+int AudioProcessingImpl::ProcessStream(const int16_t* const src,
+                                       const StreamConfig& input_config,
+                                       const StreamConfig& output_config,
+                                       int16_t* const dest) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
+
+  RETURN_ON_ERR(
+      HandleUnsupportedAudioFormats(src, input_config, output_config, dest));
+  MaybeInitializeCapture(input_config, output_config);
+
+  MutexLock lock_capture(&mutex_capture_);
+  DenormalDisabler denormal_disabler;
+
+  if (aec_dump_) {
+    RecordUnprocessedCaptureStream(src, input_config);
+  }
+
+  capture_.capture_audio->CopyFrom(src, input_config);
+  if (capture_.capture_fullband_audio) {
+    capture_.capture_fullband_audio->CopyFrom(src, input_config);
+  }
+  RETURN_ON_ERR(ProcessCaptureStreamLocked());
+  if (submodule_states_.CaptureMultiBandProcessingPresent() ||
+      submodule_states_.CaptureFullBandProcessingActive()) {
+    if (capture_.capture_fullband_audio) {
+      capture_.capture_fullband_audio->CopyTo(output_config, dest);
+    } else {
+      capture_.capture_audio->CopyTo(output_config, dest);
+    }
+  }
+
+  if (aec_dump_) {
+    RecordProcessedCaptureStream(dest, output_config);
+  }
+  return kNoError;
+}
+
+int AudioProcessingImpl::ProcessCaptureStreamLocked() {
+  EmptyQueuedRenderAudioLocked();
+  HandleCaptureRuntimeSettings();
+  DenormalDisabler denormal_disabler;
+
+  // Ensure that not both the AEC and AECM are active at the same time.
+  // TODO(peah): Simplify once the public API Enable functions for these
+  // are moved to APM.
+  RTC_DCHECK_LE(
+      !!submodules_.echo_controller + !!submodules_.echo_control_mobile, 1);
+
+  data_dumper_->DumpRaw(
+      "applied_input_volume",
+      capture_.applied_input_volume.value_or(kUnspecifiedDataDumpInputVolume));
+
+  AudioBuffer* capture_buffer = capture_.capture_audio.get();  // For brevity.
+  AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get();
+
+  if (submodules_.high_pass_filter &&
+      config_.high_pass_filter.apply_in_full_band &&
+      !constants_.enforce_split_band_hpf) {
+    submodules_.high_pass_filter->Process(capture_buffer,
+                                          /*use_split_band_data=*/false);
+  }
+
+  if (submodules_.capture_levels_adjuster) {
+    if (config_.capture_level_adjustment.analog_mic_gain_emulation.enabled) {
+      // When the input volume is emulated, retrieve the volume applied to the
+      // input audio and notify that to APM so that the volume is passed to the
+      // active AGC.
+      set_stream_analog_level_locked(
+          submodules_.capture_levels_adjuster->GetAnalogMicGainLevel());
+    }
+    submodules_.capture_levels_adjuster->ApplyPreLevelAdjustment(
+        *capture_buffer);
+  }
+
+  capture_input_rms_.Analyze(rtc::ArrayView<const float>(
+      capture_buffer->channels_const()[0],
+      capture_nonlocked_.capture_processing_format.num_frames()));
+  const bool log_rms = ++capture_rms_interval_counter_ >= 1000;
+  if (log_rms) {
+    capture_rms_interval_counter_ = 0;
+    RmsLevel::Levels levels = capture_input_rms_.AverageAndPeak();
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelAverageRms",
+                                levels.average, 1, RmsLevel::kMinLevelDb, 64);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelPeakRms",
+                                levels.peak, 1, RmsLevel::kMinLevelDb, 64);
+  }
+
+  if (capture_.applied_input_volume.has_value()) {
+    applied_input_volume_stats_reporter_.UpdateStatistics(
+        *capture_.applied_input_volume);
+  }
+
+  if (submodules_.echo_controller) {
+    // Determine if the echo path gain has changed by checking all the gains
+    // applied before AEC.
+    capture_.echo_path_gain_change = capture_.applied_input_volume_changed;
+
+    // Detect and flag any change in the capture level adjustment pre-gain.
+    if (submodules_.capture_levels_adjuster) {
+      float pre_adjustment_gain =
+          submodules_.capture_levels_adjuster->GetPreAdjustmentGain();
+      capture_.echo_path_gain_change =
+          capture_.echo_path_gain_change ||
+          (capture_.prev_pre_adjustment_gain != pre_adjustment_gain &&
+           capture_.prev_pre_adjustment_gain >= 0.0f);
+      capture_.prev_pre_adjustment_gain = pre_adjustment_gain;
+    }
+
+    // Detect volume change.
+    capture_.echo_path_gain_change =
+        capture_.echo_path_gain_change ||
+        (capture_.prev_playout_volume != capture_.playout_volume &&
+         capture_.prev_playout_volume >= 0);
+    capture_.prev_playout_volume = capture_.playout_volume;
+
+    submodules_.echo_controller->AnalyzeCapture(capture_buffer);
+  }
+
+  if (submodules_.agc_manager) {
+    submodules_.agc_manager->AnalyzePreProcess(*capture_buffer);
+  }
+
+  if (submodules_.gain_controller2 &&
+      config_.gain_controller2.input_volume_controller.enabled) {
+    // Expect the volume to be available if the input controller is enabled.
+    RTC_DCHECK(capture_.applied_input_volume.has_value());
+    if (capture_.applied_input_volume.has_value()) {
+      submodules_.gain_controller2->Analyze(*capture_.applied_input_volume,
+                                            *capture_buffer);
+    }
+  }
+
+  if (submodule_states_.CaptureMultiBandSubModulesActive() &&
+      SampleRateSupportsMultiBand(
+          capture_nonlocked_.capture_processing_format.sample_rate_hz())) {
+    capture_buffer->SplitIntoFrequencyBands();
+  }
+
+  const bool multi_channel_capture = config_.pipeline.multi_channel_capture &&
+                                     constants_.multi_channel_capture_support;
+  if (submodules_.echo_controller && !multi_channel_capture) {
+    // Force down-mixing of the number of channels after the detection of
+    // capture signal saturation.
+    // TODO(peah): Look into ensuring that this kind of tampering with the
+    // AudioBuffer functionality should not be needed.
+    capture_buffer->set_num_channels(1);
+  }
+
+  if (submodules_.high_pass_filter &&
+      (!config_.high_pass_filter.apply_in_full_band ||
+       constants_.enforce_split_band_hpf)) {
+    submodules_.high_pass_filter->Process(capture_buffer,
+                                          /*use_split_band_data=*/true);
+  }
+
+  if (submodules_.gain_control) {
+    RETURN_ON_ERR(
+        submodules_.gain_control->AnalyzeCaptureAudio(*capture_buffer));
+  }
+
+  if ((!config_.noise_suppression.analyze_linear_aec_output_when_available ||
+       !linear_aec_buffer || submodules_.echo_control_mobile) &&
+      submodules_.noise_suppressor) {
+    submodules_.noise_suppressor->Analyze(*capture_buffer);
+  }
+
+  if (submodules_.echo_control_mobile) {
+    // Ensure that the stream delay was set before the call to the
+    // AECM ProcessCaptureAudio function.
+    if (!capture_.was_stream_delay_set) {
+      return AudioProcessing::kStreamParameterNotSetError;
+    }
+
+    if (submodules_.noise_suppressor) {
+      submodules_.noise_suppressor->Process(capture_buffer);
+    }
+
+    RETURN_ON_ERR(submodules_.echo_control_mobile->ProcessCaptureAudio(
+        capture_buffer, stream_delay_ms()));
+  } else {
+    if (submodules_.echo_controller) {
+      data_dumper_->DumpRaw("stream_delay", stream_delay_ms());
+
+      if (capture_.was_stream_delay_set) {
+        submodules_.echo_controller->SetAudioBufferDelay(stream_delay_ms());
+      }
+
+      submodules_.echo_controller->ProcessCapture(
+          capture_buffer, linear_aec_buffer, capture_.echo_path_gain_change);
+    }
+
+    if (config_.noise_suppression.analyze_linear_aec_output_when_available &&
+        linear_aec_buffer && submodules_.noise_suppressor) {
+      submodules_.noise_suppressor->Analyze(*linear_aec_buffer);
+    }
+
+    if (submodules_.noise_suppressor) {
+      submodules_.noise_suppressor->Process(capture_buffer);
+    }
+  }
+
+  if (submodules_.agc_manager) {
+    submodules_.agc_manager->Process(*capture_buffer);
+
+    absl::optional<int> new_digital_gain =
+        submodules_.agc_manager->GetDigitalComressionGain();
+    if (new_digital_gain && submodules_.gain_control) {
+      submodules_.gain_control->set_compression_gain_db(*new_digital_gain);
+    }
+  }
+
+  if (submodules_.gain_control) {
+    // TODO(peah): Add reporting from AEC3 whether there is echo.
+    RETURN_ON_ERR(submodules_.gain_control->ProcessCaptureAudio(
+        capture_buffer, /*stream_has_echo*/ false));
+  }
+
+  if (submodule_states_.CaptureMultiBandProcessingPresent() &&
+      SampleRateSupportsMultiBand(
+          capture_nonlocked_.capture_processing_format.sample_rate_hz())) {
+    capture_buffer->MergeFrequencyBands();
+  }
+
+  if (capture_.capture_output_used) {
+    if (capture_.capture_fullband_audio) {
+      const auto& ec = submodules_.echo_controller;
+      bool ec_active = ec ? ec->ActiveProcessing() : false;
+      // Only update the fullband buffer if the multiband processing has changed
+      // the signal. Keep the original signal otherwise.
+      if (submodule_states_.CaptureMultiBandProcessingActive(ec_active)) {
+        capture_buffer->CopyTo(capture_.capture_fullband_audio.get());
+      }
+      capture_buffer = capture_.capture_fullband_audio.get();
+    }
+
+    if (submodules_.echo_detector) {
+      submodules_.echo_detector->AnalyzeCaptureAudio(
+          rtc::ArrayView<const float>(capture_buffer->channels()[0],
+                                      capture_buffer->num_frames()));
+    }
+
+    absl::optional<float> voice_probability;
+    if (!!submodules_.voice_activity_detector) {
+      voice_probability = submodules_.voice_activity_detector->Analyze(
+          AudioFrameView<const float>(capture_buffer->channels(),
+                                      capture_buffer->num_channels(),
+                                      capture_buffer->num_frames()));
+    }
+
+    if (submodules_.transient_suppressor) {
+      float transient_suppressor_voice_probability = 1.0f;
+      switch (transient_suppressor_vad_mode_) {
+        case TransientSuppressor::VadMode::kDefault:
+          if (submodules_.agc_manager) {
+            transient_suppressor_voice_probability =
+                submodules_.agc_manager->voice_probability();
+          }
+          break;
+        case TransientSuppressor::VadMode::kRnnVad:
+          RTC_DCHECK(voice_probability.has_value());
+          transient_suppressor_voice_probability = *voice_probability;
+          break;
+        case TransientSuppressor::VadMode::kNoVad:
+          // The transient suppressor will ignore `voice_probability`.
+          break;
+      }
+      float delayed_voice_probability =
+          submodules_.transient_suppressor->Suppress(
+              capture_buffer->channels()[0], capture_buffer->num_frames(),
+              capture_buffer->num_channels(),
+              capture_buffer->split_bands_const(0)[kBand0To8kHz],
+              capture_buffer->num_frames_per_band(),
+              /*reference_data=*/nullptr, /*reference_length=*/0,
+              transient_suppressor_voice_probability, capture_.key_pressed);
+      if (voice_probability.has_value()) {
+        *voice_probability = delayed_voice_probability;
+      }
+    }
+
+    // Experimental APM sub-module that analyzes `capture_buffer`.
+    if (submodules_.capture_analyzer) {
+      submodules_.capture_analyzer->Analyze(capture_buffer);
+    }
+
+    if (submodules_.gain_controller2) {
+      // TODO(bugs.webrtc.org/7494): Let AGC2 detect applied input volume
+      // changes.
+      submodules_.gain_controller2->Process(
+          voice_probability, capture_.applied_input_volume_changed,
+          capture_buffer);
+    }
+
+    if (submodules_.capture_post_processor) {
+      submodules_.capture_post_processor->Process(capture_buffer);
+    }
+
+    capture_output_rms_.Analyze(rtc::ArrayView<const float>(
+        capture_buffer->channels_const()[0],
+        capture_nonlocked_.capture_processing_format.num_frames()));
+    if (log_rms) {
+      RmsLevel::Levels levels = capture_output_rms_.AverageAndPeak();
+      RTC_HISTOGRAM_COUNTS_LINEAR(
+          "WebRTC.Audio.ApmCaptureOutputLevelAverageRms", levels.average, 1,
+          RmsLevel::kMinLevelDb, 64);
+      RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureOutputLevelPeakRms",
+                                  levels.peak, 1, RmsLevel::kMinLevelDb, 64);
+    }
+
+    // Compute echo-detector stats.
+    if (submodules_.echo_detector) {
+      auto ed_metrics = submodules_.echo_detector->GetMetrics();
+      capture_.stats.residual_echo_likelihood = ed_metrics.echo_likelihood;
+      capture_.stats.residual_echo_likelihood_recent_max =
+          ed_metrics.echo_likelihood_recent_max;
+    }
+  }
+
+  // Compute echo-controller stats.
+  if (submodules_.echo_controller) {
+    auto ec_metrics = submodules_.echo_controller->GetMetrics();
+    capture_.stats.echo_return_loss = ec_metrics.echo_return_loss;
+    capture_.stats.echo_return_loss_enhancement =
+        ec_metrics.echo_return_loss_enhancement;
+    capture_.stats.delay_ms = ec_metrics.delay_ms;
+  }
+
+  // Pass stats for reporting.
+  stats_reporter_.UpdateStatistics(capture_.stats);
+
+  UpdateRecommendedInputVolumeLocked();
+  if (capture_.recommended_input_volume.has_value()) {
+    recommended_input_volume_stats_reporter_.UpdateStatistics(
+        *capture_.recommended_input_volume);
+  }
+
+  if (submodules_.capture_levels_adjuster) {
+    submodules_.capture_levels_adjuster->ApplyPostLevelAdjustment(
+        *capture_buffer);
+
+    if (config_.capture_level_adjustment.analog_mic_gain_emulation.enabled) {
+      // If the input volume emulation is used, retrieve the recommended input
+      // volume and set that to emulate the input volume on the next processed
+      // audio frame.
+      RTC_DCHECK(capture_.recommended_input_volume.has_value());
+      submodules_.capture_levels_adjuster->SetAnalogMicGainLevel(
+          *capture_.recommended_input_volume);
+    }
+  }
+
+  // Temporarily set the output to zero after the stream has been unmuted
+  // (capture output is again used). The purpose of this is to avoid clicks and
+  // artefacts in the audio that results when the processing again is
+  // reactivated after unmuting.
+  if (!capture_.capture_output_used_last_frame &&
+      capture_.capture_output_used) {
+    for (size_t ch = 0; ch < capture_buffer->num_channels(); ++ch) {
+      rtc::ArrayView<float> channel_view(capture_buffer->channels()[ch],
+                                         capture_buffer->num_frames());
+      std::fill(channel_view.begin(), channel_view.end(), 0.f);
+    }
+  }
+  capture_.capture_output_used_last_frame = capture_.capture_output_used;
+
+  capture_.was_stream_delay_set = false;
+
+  data_dumper_->DumpRaw("recommended_input_volume",
+                        capture_.recommended_input_volume.value_or(
+                            kUnspecifiedDataDumpInputVolume));
+
+  return kNoError;
+}
+
+int AudioProcessingImpl::AnalyzeReverseStream(
+    const float* const* data,
+    const StreamConfig& reverse_config) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_StreamConfig");
+  MutexLock lock(&mutex_render_);
+  DenormalDisabler denormal_disabler;
+  RTC_DCHECK(data);
+  for (size_t i = 0; i < reverse_config.num_channels(); ++i) {
+    RTC_DCHECK(data[i]);
+  }
+  RETURN_ON_ERR(
+      AudioFormatValidityToErrorCode(ValidateAudioFormat(reverse_config)));
+
+  MaybeInitializeRender(reverse_config, reverse_config);
+  return AnalyzeReverseStreamLocked(data, reverse_config, reverse_config);
+}
+
+int AudioProcessingImpl::ProcessReverseStream(const float* const* src,
+                                              const StreamConfig& input_config,
+                                              const StreamConfig& output_config,
+                                              float* const* dest) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_StreamConfig");
+  MutexLock lock(&mutex_render_);
+  DenormalDisabler denormal_disabler;
+  RETURN_ON_ERR(
+      HandleUnsupportedAudioFormats(src, input_config, output_config, dest));
+
+  MaybeInitializeRender(input_config, output_config);
+
+  RETURN_ON_ERR(AnalyzeReverseStreamLocked(src, input_config, output_config));
+
+  if (submodule_states_.RenderMultiBandProcessingActive() ||
+      submodule_states_.RenderFullBandProcessingActive()) {
+    render_.render_audio->CopyTo(formats_.api_format.reverse_output_stream(),
+                                 dest);
+  } else if (formats_.api_format.reverse_input_stream() !=
+             formats_.api_format.reverse_output_stream()) {
+    render_.render_converter->Convert(src, input_config.num_samples(), dest,
+                                      output_config.num_samples());
+  } else {
+    CopyAudioIfNeeded(src, input_config.num_frames(),
+                      input_config.num_channels(), dest);
+  }
+
+  return kNoError;
+}
+
+int AudioProcessingImpl::AnalyzeReverseStreamLocked(
+    const float* const* src,
+    const StreamConfig& input_config,
+    const StreamConfig& output_config) {
+  if (aec_dump_) {
+    const size_t channel_size =
+        formats_.api_format.reverse_input_stream().num_frames();
+    const size_t num_channels =
+        formats_.api_format.reverse_input_stream().num_channels();
+    aec_dump_->WriteRenderStreamMessage(
+        AudioFrameView<const float>(src, num_channels, channel_size));
+  }
+  render_.render_audio->CopyFrom(src,
+                                 formats_.api_format.reverse_input_stream());
+  return ProcessRenderStreamLocked();
+}
+
+int AudioProcessingImpl::ProcessReverseStream(const int16_t* const src,
+                                              const StreamConfig& input_config,
+                                              const StreamConfig& output_config,
+                                              int16_t* const dest) {
+  TRACE_EVENT0("webrtc", "AudioProcessing::ProcessReverseStream_AudioFrame");
+
+  MutexLock lock(&mutex_render_);
+  DenormalDisabler denormal_disabler;
+
+  RETURN_ON_ERR(
+      HandleUnsupportedAudioFormats(src, input_config, output_config, dest));
+  MaybeInitializeRender(input_config, output_config);
+
+  if (aec_dump_) {
+    aec_dump_->WriteRenderStreamMessage(src, input_config.num_frames(),
+                                        input_config.num_channels());
+  }
+
+  render_.render_audio->CopyFrom(src, input_config);
+  RETURN_ON_ERR(ProcessRenderStreamLocked());
+  if (submodule_states_.RenderMultiBandProcessingActive() ||
+      submodule_states_.RenderFullBandProcessingActive()) {
+    render_.render_audio->CopyTo(output_config, dest);
+  }
+  return kNoError;
+}
+
+int AudioProcessingImpl::ProcessRenderStreamLocked() {
+  AudioBuffer* render_buffer = render_.render_audio.get();  // For brevity.
+
+  HandleRenderRuntimeSettings();
+  DenormalDisabler denormal_disabler;
+
+  if (submodules_.render_pre_processor) {
+    submodules_.render_pre_processor->Process(render_buffer);
+  }
+
+  QueueNonbandedRenderAudio(render_buffer);
+
+  if (submodule_states_.RenderMultiBandSubModulesActive() &&
+      SampleRateSupportsMultiBand(
+          formats_.render_processing_format.sample_rate_hz())) {
+    render_buffer->SplitIntoFrequencyBands();
+  }
+
+  if (submodule_states_.RenderMultiBandSubModulesActive()) {
+    QueueBandedRenderAudio(render_buffer);
+  }
+
+  // TODO(peah): Perform the queuing inside QueueRenderAudiuo().
+  if (submodules_.echo_controller) {
+    submodules_.echo_controller->AnalyzeRender(render_buffer);
+  }
+
+  if (submodule_states_.RenderMultiBandProcessingActive() &&
+      SampleRateSupportsMultiBand(
+          formats_.render_processing_format.sample_rate_hz())) {
+    render_buffer->MergeFrequencyBands();
+  }
+
+  return kNoError;
+}
+
+int AudioProcessingImpl::set_stream_delay_ms(int delay) {
+  MutexLock lock(&mutex_capture_);
+  Error retval = kNoError;
+  capture_.was_stream_delay_set = true;
+
+  if (delay < 0) {
+    delay = 0;
+    retval = kBadStreamParameterWarning;
+  }
+
+  // TODO(ajm): the max is rather arbitrarily chosen; investigate.
+  if (delay > 500) {
+    delay = 500;
+    retval = kBadStreamParameterWarning;
+  }
+
+  capture_nonlocked_.stream_delay_ms = delay;
+  return retval;
+}
+
+bool AudioProcessingImpl::GetLinearAecOutput(
+    rtc::ArrayView<std::array<float, 160>> linear_output) const {
+  MutexLock lock(&mutex_capture_);
+  AudioBuffer* linear_aec_buffer = capture_.linear_aec_output.get();
+
+  RTC_DCHECK(linear_aec_buffer);
+  if (linear_aec_buffer) {
+    RTC_DCHECK_EQ(1, linear_aec_buffer->num_bands());
+    RTC_DCHECK_EQ(linear_output.size(), linear_aec_buffer->num_channels());
+
+    for (size_t ch = 0; ch < linear_aec_buffer->num_channels(); ++ch) {
+      RTC_DCHECK_EQ(linear_output[ch].size(), linear_aec_buffer->num_frames());
+      rtc::ArrayView<const float> channel_view =
+          rtc::ArrayView<const float>(linear_aec_buffer->channels_const()[ch],
+                                      linear_aec_buffer->num_frames());
+      FloatS16ToFloat(channel_view.data(), channel_view.size(),
+                      linear_output[ch].data());
+    }
+    return true;
+  }
+  RTC_LOG(LS_ERROR) << "No linear AEC output available";
+  RTC_DCHECK_NOTREACHED();
+  return false;
+}
+
+int AudioProcessingImpl::stream_delay_ms() const {
+  // Used as callback from submodules, hence locking is not allowed.
+  return capture_nonlocked_.stream_delay_ms;
+}
+
+void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
+  MutexLock lock(&mutex_capture_);
+  capture_.key_pressed = key_pressed;
+}
+
+void AudioProcessingImpl::set_stream_analog_level(int level) {
+  MutexLock lock_capture(&mutex_capture_);
+  set_stream_analog_level_locked(level);
+}
+
+void AudioProcessingImpl::set_stream_analog_level_locked(int level) {
+  capture_.applied_input_volume_changed =
+      capture_.applied_input_volume.has_value() &&
+      *capture_.applied_input_volume != level;
+  capture_.applied_input_volume = level;
+
+  // Invalidate any previously recommended input volume which will be updated by
+  // `ProcessStream()`.
+  capture_.recommended_input_volume = absl::nullopt;
+
+  if (submodules_.agc_manager) {
+    submodules_.agc_manager->set_stream_analog_level(level);
+    return;
+  }
+
+  if (submodules_.gain_control) {
+    int error = submodules_.gain_control->set_stream_analog_level(level);
+    RTC_DCHECK_EQ(kNoError, error);
+    return;
+  }
+}
+
+int AudioProcessingImpl::recommended_stream_analog_level() const {
+  MutexLock lock_capture(&mutex_capture_);
+  if (!capture_.applied_input_volume.has_value()) {
+    RTC_LOG(LS_ERROR) << "set_stream_analog_level has not been called";
+  }
+  // Input volume to recommend when `set_stream_analog_level()` is not called.
+  constexpr int kFallBackInputVolume = 255;
+  // When APM has no input volume to recommend, return the latest applied input
+  // volume that has been observed in order to possibly produce no input volume
+  // change. If no applied input volume has been observed, return a fall-back
+  // value.
+  return capture_.recommended_input_volume.value_or(
+      capture_.applied_input_volume.value_or(kFallBackInputVolume));
+}
+
+void AudioProcessingImpl::UpdateRecommendedInputVolumeLocked() {
+  if (!capture_.applied_input_volume.has_value()) {
+    // When `set_stream_analog_level()` is not called, no input level can be
+    // recommended.
+    capture_.recommended_input_volume = absl::nullopt;
+    return;
+  }
+
+  if (submodules_.agc_manager) {
+    capture_.recommended_input_volume =
+        submodules_.agc_manager->recommended_analog_level();
+    return;
+  }
+
+  if (submodules_.gain_control) {
+    capture_.recommended_input_volume =
+        submodules_.gain_control->stream_analog_level();
+    return;
+  }
+
+  if (submodules_.gain_controller2 &&
+      config_.gain_controller2.input_volume_controller.enabled) {
+    capture_.recommended_input_volume =
+        submodules_.gain_controller2->recommended_input_volume();
+    return;
+  }
+
+  capture_.recommended_input_volume = capture_.applied_input_volume;
+}
+
+bool AudioProcessingImpl::CreateAndAttachAecDump(absl::string_view file_name,
+                                                 int64_t max_log_size_bytes,
+                                                 rtc::TaskQueue* worker_queue) {
+  std::unique_ptr<AecDump> aec_dump =
+      AecDumpFactory::Create(file_name, max_log_size_bytes, worker_queue);
+  if (!aec_dump) {
+    return false;
+  }
+
+  AttachAecDump(std::move(aec_dump));
+  return true;
+}
+
+bool AudioProcessingImpl::CreateAndAttachAecDump(FILE* handle,
+                                                 int64_t max_log_size_bytes,
+                                                 rtc::TaskQueue* worker_queue) {
+  std::unique_ptr<AecDump> aec_dump =
+      AecDumpFactory::Create(handle, max_log_size_bytes, worker_queue);
+  if (!aec_dump) {
+    return false;
+  }
+
+  AttachAecDump(std::move(aec_dump));
+  return true;
+}
+
+void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) {
+  RTC_DCHECK(aec_dump);
+  MutexLock lock_render(&mutex_render_);
+  MutexLock lock_capture(&mutex_capture_);
+
+  // The previously attached AecDump will be destroyed with the
+  // 'aec_dump' parameter, which is after locks are released.
+  aec_dump_.swap(aec_dump);
+  WriteAecDumpConfigMessage(true);
+  aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis());
+}
+
+void AudioProcessingImpl::DetachAecDump() {
+  // The d-tor of a task-queue based AecDump blocks until all pending
+  // tasks are done. This construction avoids blocking while holding
+  // the render and capture locks.
+  std::unique_ptr<AecDump> aec_dump = nullptr;
+  {
+    MutexLock lock_render(&mutex_render_);
+    MutexLock lock_capture(&mutex_capture_);
+    aec_dump = std::move(aec_dump_);
+  }
+}
+
+AudioProcessing::Config AudioProcessingImpl::GetConfig() const {
+  MutexLock lock_render(&mutex_render_);
+  MutexLock lock_capture(&mutex_capture_);
+  return config_;
+}
+
+bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
+  return submodule_states_.Update(
+      config_.high_pass_filter.enabled, !!submodules_.echo_control_mobile,
+      !!submodules_.noise_suppressor, !!submodules_.gain_control,
+      !!submodules_.gain_controller2, !!submodules_.voice_activity_detector,
+      config_.pre_amplifier.enabled || config_.capture_level_adjustment.enabled,
+      capture_nonlocked_.echo_controller_enabled,
+      !!submodules_.transient_suppressor);
+}
+
+void AudioProcessingImpl::InitializeTransientSuppressor() {
+  // Choose the VAD mode for TS and detect a VAD mode change.
+  const TransientSuppressor::VadMode previous_vad_mode =
+      transient_suppressor_vad_mode_;
+  transient_suppressor_vad_mode_ = TransientSuppressor::VadMode::kDefault;
+  if (UseApmVadSubModule(config_, gain_controller2_experiment_params_)) {
+    transient_suppressor_vad_mode_ = TransientSuppressor::VadMode::kRnnVad;
+  }
+  const bool vad_mode_changed =
+      previous_vad_mode != transient_suppressor_vad_mode_;
+
+  if (config_.transient_suppression.enabled &&
+      !constants_.transient_suppressor_forced_off) {
+    // Attempt to create a transient suppressor, if one is not already created.
+    if (!submodules_.transient_suppressor || vad_mode_changed) {
+      submodules_.transient_suppressor = CreateTransientSuppressor(
+          submodule_creation_overrides_, transient_suppressor_vad_mode_,
+          proc_fullband_sample_rate_hz(), capture_nonlocked_.split_rate,
+          num_proc_channels());
+      if (!submodules_.transient_suppressor) {
+        RTC_LOG(LS_WARNING)
+            << "No transient suppressor created (probably disabled)";
+      }
+    } else {
+      submodules_.transient_suppressor->Initialize(
+          proc_fullband_sample_rate_hz(), capture_nonlocked_.split_rate,
+          num_proc_channels());
+    }
+  } else {
+    submodules_.transient_suppressor.reset();
+  }
+}
+
+void AudioProcessingImpl::InitializeHighPassFilter(bool forced_reset) {
+  bool high_pass_filter_needed_by_aec =
+      config_.echo_canceller.enabled &&
+      config_.echo_canceller.enforce_high_pass_filtering &&
+      !config_.echo_canceller.mobile_mode;
+  if (submodule_states_.HighPassFilteringRequired() ||
+      high_pass_filter_needed_by_aec) {
+    bool use_full_band = config_.high_pass_filter.apply_in_full_band &&
+                         !constants_.enforce_split_band_hpf;
+    int rate = use_full_band ? proc_fullband_sample_rate_hz()
+                             : proc_split_sample_rate_hz();
+    size_t num_channels =
+        use_full_band ? num_output_channels() : num_proc_channels();
+
+    if (!submodules_.high_pass_filter ||
+        rate != submodules_.high_pass_filter->sample_rate_hz() ||
+        forced_reset ||
+        num_channels != submodules_.high_pass_filter->num_channels()) {
+      submodules_.high_pass_filter.reset(
+          new HighPassFilter(rate, num_channels));
+    }
+  } else {
+    submodules_.high_pass_filter.reset();
+  }
+}
+
+void AudioProcessingImpl::InitializeEchoController() {
+  bool use_echo_controller =
+      echo_control_factory_ ||
+      (config_.echo_canceller.enabled && !config_.echo_canceller.mobile_mode);
+
+  if (use_echo_controller) {
+    // Create and activate the echo controller.
+    if (echo_control_factory_) {
+      submodules_.echo_controller = echo_control_factory_->Create(
+          proc_sample_rate_hz(), num_reverse_channels(), num_proc_channels());
+      RTC_DCHECK(submodules_.echo_controller);
+    } else {
+      EchoCanceller3Config config;
+      absl::optional<EchoCanceller3Config> multichannel_config;
+      if (use_setup_specific_default_aec3_config_) {
+        multichannel_config = EchoCanceller3::CreateDefaultMultichannelConfig();
+      }
+      submodules_.echo_controller = std::make_unique<EchoCanceller3>(
+          config, multichannel_config, proc_sample_rate_hz(),
+          num_reverse_channels(), num_proc_channels());
+    }
+
+    // Setup the storage for returning the linear AEC output.
+    if (config_.echo_canceller.export_linear_aec_output) {
+      constexpr int kLinearOutputRateHz = 16000;
+      capture_.linear_aec_output = std::make_unique<AudioBuffer>(
+          kLinearOutputRateHz, num_proc_channels(), kLinearOutputRateHz,
+          num_proc_channels(), kLinearOutputRateHz, num_proc_channels());
+    } else {
+      capture_.linear_aec_output.reset();
+    }
+
+    capture_nonlocked_.echo_controller_enabled = true;
+
+    submodules_.echo_control_mobile.reset();
+    aecm_render_signal_queue_.reset();
+    return;
+  }
+
+  submodules_.echo_controller.reset();
+  capture_nonlocked_.echo_controller_enabled = false;
+  capture_.linear_aec_output.reset();
+
+  if (!config_.echo_canceller.enabled) {
+    submodules_.echo_control_mobile.reset();
+    aecm_render_signal_queue_.reset();
+    return;
+  }
+
+  if (config_.echo_canceller.mobile_mode) {
+    // Create and activate AECM.
+    size_t max_element_size =
+        std::max(static_cast<size_t>(1),
+                 kMaxAllowedValuesOfSamplesPerBand *
+                     EchoControlMobileImpl::NumCancellersRequired(
+                         num_output_channels(), num_reverse_channels()));
+
+    std::vector<int16_t> template_queue_element(max_element_size);
+
+    aecm_render_signal_queue_.reset(
+        new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
+            kMaxNumFramesToBuffer, template_queue_element,
+            RenderQueueItemVerifier<int16_t>(max_element_size)));
+
+    aecm_render_queue_buffer_.resize(max_element_size);
+    aecm_capture_queue_buffer_.resize(max_element_size);
+
+    submodules_.echo_control_mobile.reset(new EchoControlMobileImpl());
+
+    submodules_.echo_control_mobile->Initialize(proc_split_sample_rate_hz(),
+                                                num_reverse_channels(),
+                                                num_output_channels());
+    return;
+  }
+
+  submodules_.echo_control_mobile.reset();
+  aecm_render_signal_queue_.reset();
+}
+
+void AudioProcessingImpl::InitializeGainController1() {
+  if (config_.gain_controller2.enabled &&
+      config_.gain_controller2.input_volume_controller.enabled &&
+      config_.gain_controller1.enabled &&
+      (config_.gain_controller1.mode ==
+           AudioProcessing::Config::GainController1::kAdaptiveAnalog ||
+       config_.gain_controller1.analog_gain_controller.enabled)) {
+    RTC_LOG(LS_ERROR) << "APM configuration not valid: "
+                      << "Multiple input volume controllers enabled.";
+  }
+
+  if (!config_.gain_controller1.enabled) {
+    submodules_.agc_manager.reset();
+    submodules_.gain_control.reset();
+    return;
+  }
+
+  RTC_HISTOGRAM_BOOLEAN(
+      "WebRTC.Audio.GainController.Analog.Enabled",
+      config_.gain_controller1.analog_gain_controller.enabled);
+
+  if (!submodules_.gain_control) {
+    submodules_.gain_control.reset(new GainControlImpl());
+  }
+
+  submodules_.gain_control->Initialize(num_proc_channels(),
+                                       proc_sample_rate_hz());
+  if (!config_.gain_controller1.analog_gain_controller.enabled) {
+    int error = submodules_.gain_control->set_mode(
+        Agc1ConfigModeToInterfaceMode(config_.gain_controller1.mode));
+    RTC_DCHECK_EQ(kNoError, error);
+    error = submodules_.gain_control->set_target_level_dbfs(
+        config_.gain_controller1.target_level_dbfs);
+    RTC_DCHECK_EQ(kNoError, error);
+    error = submodules_.gain_control->set_compression_gain_db(
+        config_.gain_controller1.compression_gain_db);
+    RTC_DCHECK_EQ(kNoError, error);
+    error = submodules_.gain_control->enable_limiter(
+        config_.gain_controller1.enable_limiter);
+    RTC_DCHECK_EQ(kNoError, error);
+    constexpr int kAnalogLevelMinimum = 0;
+    constexpr int kAnalogLevelMaximum = 255;
+    error = submodules_.gain_control->set_analog_level_limits(
+        kAnalogLevelMinimum, kAnalogLevelMaximum);
+    RTC_DCHECK_EQ(kNoError, error);
+
+    submodules_.agc_manager.reset();
+    return;
+  }
+
+  if (!submodules_.agc_manager.get() ||
+      submodules_.agc_manager->num_channels() !=
+          static_cast<int>(num_proc_channels())) {
+    int stream_analog_level = -1;
+    const bool re_creation = !!submodules_.agc_manager;
+    if (re_creation) {
+      stream_analog_level = submodules_.agc_manager->recommended_analog_level();
+    }
+    submodules_.agc_manager.reset(new AgcManagerDirect(
+        num_proc_channels(), config_.gain_controller1.analog_gain_controller));
+    if (re_creation) {
+      submodules_.agc_manager->set_stream_analog_level(stream_analog_level);
+    }
+  }
+  submodules_.agc_manager->Initialize();
+  submodules_.agc_manager->SetupDigitalGainControl(*submodules_.gain_control);
+  submodules_.agc_manager->HandleCaptureOutputUsedChange(
+      capture_.capture_output_used);
+}
+
+void AudioProcessingImpl::InitializeGainController2() {
+  if (!config_.gain_controller2.enabled) {
+    submodules_.gain_controller2.reset();
+    return;
+  }
+  // Override the input volume controller configuration if the AGC2 experiment
+  // is running and its parameters require to fully switch the gain control to
+  // AGC2.
+  const bool input_volume_controller_config_overridden =
+      gain_controller2_experiment_params_.has_value() &&
+      gain_controller2_experiment_params_->agc2_config.has_value();
+  const InputVolumeController::Config input_volume_controller_config =
+      input_volume_controller_config_overridden
+          ? gain_controller2_experiment_params_->agc2_config
+                ->input_volume_controller
+          : InputVolumeController::Config{};
+  // If the APM VAD sub-module is not used, let AGC2 use its internal VAD.
+  const bool use_internal_vad =
+      !UseApmVadSubModule(config_, gain_controller2_experiment_params_);
+  submodules_.gain_controller2 = std::make_unique<GainController2>(
+      config_.gain_controller2, input_volume_controller_config,
+      proc_fullband_sample_rate_hz(), num_proc_channels(), use_internal_vad);
+  submodules_.gain_controller2->SetCaptureOutputUsed(
+      capture_.capture_output_used);
+}
+
+void AudioProcessingImpl::InitializeVoiceActivityDetector() {
+  if (!UseApmVadSubModule(config_, gain_controller2_experiment_params_)) {
+    submodules_.voice_activity_detector.reset();
+    return;
+  }
+
+  if (!submodules_.voice_activity_detector) {
+    RTC_DCHECK(!!submodules_.gain_controller2);
+    // TODO(bugs.webrtc.org/13663): Cache CPU features in APM and use here.
+    submodules_.voice_activity_detector =
+        std::make_unique<VoiceActivityDetectorWrapper>(
+            submodules_.gain_controller2->GetCpuFeatures(),
+            proc_fullband_sample_rate_hz());
+  } else {
+    submodules_.voice_activity_detector->Initialize(
+        proc_fullband_sample_rate_hz());
+  }
+}
+
+void AudioProcessingImpl::InitializeNoiseSuppressor() {
+  submodules_.noise_suppressor.reset();
+
+  if (config_.noise_suppression.enabled) {
+    auto map_level =
+        [](AudioProcessing::Config::NoiseSuppression::Level level) {
+          using NoiseSuppresionConfig =
+              AudioProcessing::Config::NoiseSuppression;
+          switch (level) {
+            case NoiseSuppresionConfig::kLow:
+              return NsConfig::SuppressionLevel::k6dB;
+            case NoiseSuppresionConfig::kModerate:
+              return NsConfig::SuppressionLevel::k12dB;
+            case NoiseSuppresionConfig::kHigh:
+              return NsConfig::SuppressionLevel::k18dB;
+            case NoiseSuppresionConfig::kVeryHigh:
+              return NsConfig::SuppressionLevel::k21dB;
+          }
+          RTC_CHECK_NOTREACHED();
+        };
+
+    NsConfig cfg;
+    cfg.target_level = map_level(config_.noise_suppression.level);
+    submodules_.noise_suppressor = std::make_unique<NoiseSuppressor>(
+        cfg, proc_sample_rate_hz(), num_proc_channels());
+  }
+}
+
+void AudioProcessingImpl::InitializeCaptureLevelsAdjuster() {
+  if (config_.pre_amplifier.enabled ||
+      config_.capture_level_adjustment.enabled) {
+    // Use both the pre-amplifier and the capture level adjustment gains as
+    // pre-gains.
+    float pre_gain = 1.f;
+    if (config_.pre_amplifier.enabled) {
+      pre_gain *= config_.pre_amplifier.fixed_gain_factor;
+    }
+    if (config_.capture_level_adjustment.enabled) {
+      pre_gain *= config_.capture_level_adjustment.pre_gain_factor;
+    }
+
+    submodules_.capture_levels_adjuster =
+        std::make_unique<CaptureLevelsAdjuster>(
+            config_.capture_level_adjustment.analog_mic_gain_emulation.enabled,
+            config_.capture_level_adjustment.analog_mic_gain_emulation
+                .initial_level,
+            pre_gain, config_.capture_level_adjustment.post_gain_factor);
+  } else {
+    submodules_.capture_levels_adjuster.reset();
+  }
+}
+
+void AudioProcessingImpl::InitializeResidualEchoDetector() {
+  if (submodules_.echo_detector) {
+    submodules_.echo_detector->Initialize(
+        proc_fullband_sample_rate_hz(), 1,
+        formats_.render_processing_format.sample_rate_hz(), 1);
+  }
+}
+
+void AudioProcessingImpl::InitializeAnalyzer() {
+  if (submodules_.capture_analyzer) {
+    submodules_.capture_analyzer->Initialize(proc_fullband_sample_rate_hz(),
+                                             num_proc_channels());
+  }
+}
+
+void AudioProcessingImpl::InitializePostProcessor() {
+  if (submodules_.capture_post_processor) {
+    submodules_.capture_post_processor->Initialize(
+        proc_fullband_sample_rate_hz(), num_proc_channels());
+  }
+}
+
+void AudioProcessingImpl::InitializePreProcessor() {
+  if (submodules_.render_pre_processor) {
+    submodules_.render_pre_processor->Initialize(
+        formats_.render_processing_format.sample_rate_hz(),
+        formats_.render_processing_format.num_channels());
+  }
+}
+
+void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) {
+  if (!aec_dump_) {
+    return;
+  }
+
+  std::string experiments_description = "";
+  // TODO(peah): Add semicolon-separated concatenations of experiment
+  // descriptions for other submodules.
+  if (!!submodules_.capture_post_processor) {
+    experiments_description += "CapturePostProcessor;";
+  }
+  if (!!submodules_.render_pre_processor) {
+    experiments_description += "RenderPreProcessor;";
+  }
+  if (capture_nonlocked_.echo_controller_enabled) {
+    experiments_description += "EchoController;";
+  }
+  if (config_.gain_controller2.enabled) {
+    experiments_description += "GainController2;";
+  }
+
+  InternalAPMConfig apm_config;
+
+  apm_config.aec_enabled = config_.echo_canceller.enabled;
+  apm_config.aec_delay_agnostic_enabled = false;
+  apm_config.aec_extended_filter_enabled = false;
+  apm_config.aec_suppression_level = 0;
+
+  apm_config.aecm_enabled = !!submodules_.echo_control_mobile;
+  apm_config.aecm_comfort_noise_enabled =
+      submodules_.echo_control_mobile &&
+      submodules_.echo_control_mobile->is_comfort_noise_enabled();
+  apm_config.aecm_routing_mode =
+      submodules_.echo_control_mobile
+          ? static_cast<int>(submodules_.echo_control_mobile->routing_mode())
+          : 0;
+
+  apm_config.agc_enabled = !!submodules_.gain_control;
+
+  apm_config.agc_mode = submodules_.gain_control
+                            ? static_cast<int>(submodules_.gain_control->mode())
+                            : GainControl::kAdaptiveAnalog;
+  apm_config.agc_limiter_enabled =
+      submodules_.gain_control ? submodules_.gain_control->is_limiter_enabled()
+                               : false;
+  apm_config.noise_robust_agc_enabled = !!submodules_.agc_manager;
+
+  apm_config.hpf_enabled = config_.high_pass_filter.enabled;
+
+  apm_config.ns_enabled = config_.noise_suppression.enabled;
+  apm_config.ns_level = static_cast<int>(config_.noise_suppression.level);
+
+  apm_config.transient_suppression_enabled =
+      config_.transient_suppression.enabled;
+  apm_config.experiments_description = experiments_description;
+  apm_config.pre_amplifier_enabled = config_.pre_amplifier.enabled;
+  apm_config.pre_amplifier_fixed_gain_factor =
+      config_.pre_amplifier.fixed_gain_factor;
+
+  if (!forced && apm_config == apm_config_for_aec_dump_) {
+    return;
+  }
+  aec_dump_->WriteConfig(apm_config);
+  apm_config_for_aec_dump_ = apm_config;
+}
+
+void AudioProcessingImpl::RecordUnprocessedCaptureStream(
+    const float* const* src) {
+  RTC_DCHECK(aec_dump_);
+  WriteAecDumpConfigMessage(false);
+
+  const size_t channel_size = formats_.api_format.input_stream().num_frames();
+  const size_t num_channels = formats_.api_format.input_stream().num_channels();
+  aec_dump_->AddCaptureStreamInput(
+      AudioFrameView<const float>(src, num_channels, channel_size));
+  RecordAudioProcessingState();
+}
+
+void AudioProcessingImpl::RecordUnprocessedCaptureStream(
+    const int16_t* const data,
+    const StreamConfig& config) {
+  RTC_DCHECK(aec_dump_);
+  WriteAecDumpConfigMessage(false);
+
+  aec_dump_->AddCaptureStreamInput(data, config.num_channels(),
+                                   config.num_frames());
+  RecordAudioProcessingState();
+}
+
+void AudioProcessingImpl::RecordProcessedCaptureStream(
+    const float* const* processed_capture_stream) {
+  RTC_DCHECK(aec_dump_);
+
+  const size_t channel_size = formats_.api_format.output_stream().num_frames();
+  const size_t num_channels =
+      formats_.api_format.output_stream().num_channels();
+  aec_dump_->AddCaptureStreamOutput(AudioFrameView<const float>(
+      processed_capture_stream, num_channels, channel_size));
+  aec_dump_->WriteCaptureStreamMessage();
+}
+
+void AudioProcessingImpl::RecordProcessedCaptureStream(
+    const int16_t* const data,
+    const StreamConfig& config) {
+  RTC_DCHECK(aec_dump_);
+
+  aec_dump_->AddCaptureStreamOutput(data, config.num_channels(),
+                                    config.num_frames());
+  aec_dump_->WriteCaptureStreamMessage();
+}
+
+void AudioProcessingImpl::RecordAudioProcessingState() {
+  RTC_DCHECK(aec_dump_);
+  AecDump::AudioProcessingState audio_proc_state;
+  audio_proc_state.delay = capture_nonlocked_.stream_delay_ms;
+  audio_proc_state.drift = 0;
+  audio_proc_state.applied_input_volume = capture_.applied_input_volume;
+  audio_proc_state.keypress = capture_.key_pressed;
+  aec_dump_->AddAudioProcessingState(audio_proc_state);
+}
+
+AudioProcessingImpl::ApmCaptureState::ApmCaptureState()
+    : was_stream_delay_set(false),
+      capture_output_used(true),
+      capture_output_used_last_frame(true),
+      key_pressed(false),
+      capture_processing_format(kSampleRate16kHz),
+      split_rate(kSampleRate16kHz),
+      echo_path_gain_change(false),
+      prev_pre_adjustment_gain(-1.0f),
+      playout_volume(-1),
+      prev_playout_volume(-1),
+      applied_input_volume_changed(false) {}
+
+AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
+
+AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
+
+AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
+
+AudioProcessingImpl::ApmStatsReporter::ApmStatsReporter()
+    : stats_message_queue_(1) {}
+
+AudioProcessingImpl::ApmStatsReporter::~ApmStatsReporter() = default;
+
+AudioProcessingStats AudioProcessingImpl::ApmStatsReporter::GetStatistics() {
+  MutexLock lock_stats(&mutex_stats_);
+  bool new_stats_available = stats_message_queue_.Remove(&cached_stats_);
+  // If the message queue is full, return the cached stats.
+  static_cast<void>(new_stats_available);
+
+  return cached_stats_;
+}
+
+void AudioProcessingImpl::ApmStatsReporter::UpdateStatistics(
+    const AudioProcessingStats& new_stats) {
+  AudioProcessingStats stats_to_queue = new_stats;
+  bool stats_message_passed = stats_message_queue_.Insert(&stats_to_queue);
+  // If the message queue is full, discard the new stats.
+  static_cast<void>(stats_message_passed);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h
new file mode 100644
index 0000000000..fe80e0d912
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h
@@ -0,0 +1,603 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
+
+#include <stdio.h>
+
+#include <atomic>
+#include <list>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/function_view.h"
+#include "modules/audio_processing/aec3/echo_canceller3.h"
+#include "modules/audio_processing/agc/agc_manager_direct.h"
+#include "modules/audio_processing/agc/gain_control.h"
+#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h"
+#include "modules/audio_processing/echo_control_mobile_impl.h"
+#include "modules/audio_processing/gain_control_impl.h"
+#include "modules/audio_processing/gain_controller2.h"
+#include "modules/audio_processing/high_pass_filter.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "modules/audio_processing/include/audio_frame_proxies.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/include/audio_processing_statistics.h"
+#include "modules/audio_processing/ns/noise_suppressor.h"
+#include "modules/audio_processing/optionally_built_submodule_creators.h"
+#include "modules/audio_processing/render_queue_item_verifier.h"
+#include "modules/audio_processing/rms_level.h"
+#include "modules/audio_processing/transient/transient_suppressor.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/swap_queue.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioConverter;
+
+constexpr int RuntimeSettingQueueSize() {
+  return 100;
+}
+
+class AudioProcessingImpl : public AudioProcessing {
+ public:
+  // Methods forcing APM to run in a single-threaded manner.
+  // Acquires both the render and capture locks.
+  AudioProcessingImpl();
+  AudioProcessingImpl(const AudioProcessing::Config& config,
+                      std::unique_ptr<CustomProcessing> capture_post_processor,
+                      std::unique_ptr<CustomProcessing> render_pre_processor,
+                      std::unique_ptr<EchoControlFactory> echo_control_factory,
+                      rtc::scoped_refptr<EchoDetector> echo_detector,
+                      std::unique_ptr<CustomAudioAnalyzer> capture_analyzer);
+  ~AudioProcessingImpl() override;
+  int Initialize() override;
+  int Initialize(const ProcessingConfig& processing_config) override;
+  void ApplyConfig(const AudioProcessing::Config& config) override;
+  bool CreateAndAttachAecDump(absl::string_view file_name,
+                              int64_t max_log_size_bytes,
+                              rtc::TaskQueue* worker_queue) override;
+  bool CreateAndAttachAecDump(FILE* handle,
+                              int64_t max_log_size_bytes,
+                              rtc::TaskQueue* worker_queue) override;
+  // TODO(webrtc:5298) Deprecated variant.
+  void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
+  void DetachAecDump() override;
+  void SetRuntimeSetting(RuntimeSetting setting) override;
+  bool PostRuntimeSetting(RuntimeSetting setting) override;
+
+  // Capture-side exclusive methods possibly running APM in a
+  // multi-threaded manner. Acquire the capture lock.
+  int ProcessStream(const int16_t* const src,
+                    const StreamConfig& input_config,
+                    const StreamConfig& output_config,
+                    int16_t* const dest) override;
+  int ProcessStream(const float* const* src,
+                    const StreamConfig& input_config,
+                    const StreamConfig& output_config,
+                    float* const* dest) override;
+  bool GetLinearAecOutput(
+      rtc::ArrayView<std::array<float, 160>> linear_output) const override;
+  void set_output_will_be_muted(bool muted) override;
+  void HandleCaptureOutputUsedSetting(bool capture_output_used)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  int set_stream_delay_ms(int delay) override;
+  void set_stream_key_pressed(bool key_pressed) override;
+  void set_stream_analog_level(int level) override;
+  int recommended_stream_analog_level() const
+      RTC_LOCKS_EXCLUDED(mutex_capture_) override;
+
+  // Render-side exclusive methods possibly running APM in a
+  // multi-threaded manner. Acquire the render lock.
+  int ProcessReverseStream(const int16_t* const src,
+                           const StreamConfig& input_config,
+                           const StreamConfig& output_config,
+                           int16_t* const dest) override;
+  int AnalyzeReverseStream(const float* const* data,
+                           const StreamConfig& reverse_config) override;
+  int ProcessReverseStream(const float* const* src,
+                           const StreamConfig& input_config,
+                           const StreamConfig& output_config,
+                           float* const* dest) override;
+
+  // Methods only accessed from APM submodules or
+  // from AudioProcessing tests in a single-threaded manner.
+  // Hence there is no need for locks in these.
+  int proc_sample_rate_hz() const override;
+  int proc_split_sample_rate_hz() const override;
+  size_t num_input_channels() const override;
+  size_t num_proc_channels() const override;
+  size_t num_output_channels() const override;
+  size_t num_reverse_channels() const override;
+  int stream_delay_ms() const override;
+
+  AudioProcessingStats GetStatistics(bool has_remote_tracks) override {
+    return GetStatistics();
+  }
+  AudioProcessingStats GetStatistics() override {
+    return stats_reporter_.GetStatistics();
+  }
+
+  AudioProcessing::Config GetConfig() const override;
+
+ protected:
+  // Overridden in a mock.
+  virtual void InitializeLocked()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
+  void AssertLockedForTest()
+      RTC_ASSERT_EXCLUSIVE_LOCK(mutex_render_, mutex_capture_) {
+    mutex_render_.AssertHeld();
+    mutex_capture_.AssertHeld();
+  }
+
+ private:
+  // TODO(peah): These friend classes should be removed as soon as the new
+  // parameter setting scheme allows.
+  FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, DefaultBehavior);
+  FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, ValidConfigBehavior);
+  FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, InValidConfigBehavior);
+  FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
+                           ToggleTransientSuppressor);
+  FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
+                           ReinitializeTransientSuppressor);
+  FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
+                           BitexactWithDisabledModules);
+  FRIEND_TEST_ALL_PREFIXES(
+      AudioProcessingImplGainController2FieldTrialParametrizedTest,
+      ConfigAdjustedWhenExperimentEnabled);
+
+  void set_stream_analog_level_locked(int level)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  void UpdateRecommendedInputVolumeLocked()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  void OverrideSubmoduleCreationForTesting(
+      const ApmSubmoduleCreationOverrides& overrides);
+
+  // Class providing thread-safe message pipe functionality for
+  // `runtime_settings_`.
+  class RuntimeSettingEnqueuer {
+   public:
+    explicit RuntimeSettingEnqueuer(
+        SwapQueue<RuntimeSetting>* runtime_settings);
+    ~RuntimeSettingEnqueuer();
+
+    // Enqueue setting and return whether the setting was successfully enqueued.
+    bool Enqueue(RuntimeSetting setting);
+
+   private:
+    SwapQueue<RuntimeSetting>& runtime_settings_;
+  };
+
+  const std::unique_ptr<ApmDataDumper> data_dumper_;
+  static std::atomic<int> instance_count_;
+  const bool use_setup_specific_default_aec3_config_;
+
+  // Parameters for the "GainController2" experiment which determines whether
+  // the following APM sub-modules are created and, if so, their configurations:
+  // AGC2 (`gain_controller2`), AGC1 (`gain_control`, `agc_manager`) and TS
+  // (`transient_suppressor`).
+  // TODO(bugs.webrtc.org/7494): Remove when the "WebRTC-Audio-GainController2"
+  // field trial is removed.
+  struct GainController2ExperimentParams {
+    struct Agc2Config {
+      InputVolumeController::Config input_volume_controller;
+      AudioProcessing::Config::GainController2::AdaptiveDigital
+          adaptive_digital_controller;
+    };
+    // When `agc2_config` is specified, all gain control switches to AGC2 and
+    // the configuration is overridden.
+    absl::optional<Agc2Config> agc2_config;
+    // When true, the transient suppressor submodule is never created regardless
+    // of the APM configuration.
+    bool disallow_transient_suppressor_usage;
+  };
+  // Specified when the "WebRTC-Audio-GainController2" field trial is specified.
+  // TODO(bugs.webrtc.org/7494): Remove when the "WebRTC-Audio-GainController2"
+  // field trial is removed.
+  const absl::optional<GainController2ExperimentParams>
+      gain_controller2_experiment_params_;
+
+  // Parses the "WebRTC-Audio-GainController2" field trial. If disabled, returns
+  // an unspecified value.
+  static absl::optional<GainController2ExperimentParams>
+  GetGainController2ExperimentParams();
+
+  // When `experiment_params` is specified, returns an APM configuration
+  // modified according to the experiment parameters. Otherwise returns
+  // `config`.
+  static AudioProcessing::Config AdjustConfig(
+      const AudioProcessing::Config& config,
+      const absl::optional<GainController2ExperimentParams>& experiment_params);
+  // Returns true if the APM VAD sub-module should be used.
+  static bool UseApmVadSubModule(
+      const AudioProcessing::Config& config,
+      const absl::optional<GainController2ExperimentParams>& experiment_params);
+
+  TransientSuppressor::VadMode transient_suppressor_vad_mode_;
+
+  SwapQueue<RuntimeSetting> capture_runtime_settings_;
+  SwapQueue<RuntimeSetting> render_runtime_settings_;
+
+  RuntimeSettingEnqueuer capture_runtime_settings_enqueuer_;
+  RuntimeSettingEnqueuer render_runtime_settings_enqueuer_;
+
+  // EchoControl factory.
+  const std::unique_ptr<EchoControlFactory> echo_control_factory_;
+
+  class SubmoduleStates {
+   public:
+    SubmoduleStates(bool capture_post_processor_enabled,
+                    bool render_pre_processor_enabled,
+                    bool capture_analyzer_enabled);
+    // Updates the submodule state and returns true if it has changed.
+    bool Update(bool high_pass_filter_enabled,
+                bool mobile_echo_controller_enabled,
+                bool noise_suppressor_enabled,
+                bool adaptive_gain_controller_enabled,
+                bool gain_controller2_enabled,
+                bool voice_activity_detector_enabled,
+                bool gain_adjustment_enabled,
+                bool echo_controller_enabled,
+                bool transient_suppressor_enabled);
+    bool CaptureMultiBandSubModulesActive() const;
+    bool CaptureMultiBandProcessingPresent() const;
+    bool CaptureMultiBandProcessingActive(bool ec_processing_active) const;
+    bool CaptureFullBandProcessingActive() const;
+    bool CaptureAnalyzerActive() const;
+    bool RenderMultiBandSubModulesActive() const;
+    bool RenderFullBandProcessingActive() const;
+    bool RenderMultiBandProcessingActive() const;
+    bool HighPassFilteringRequired() const;
+
+   private:
+    const bool capture_post_processor_enabled_ = false;
+    const bool render_pre_processor_enabled_ = false;
+    const bool capture_analyzer_enabled_ = false;
+    bool high_pass_filter_enabled_ = false;
+    bool mobile_echo_controller_enabled_ = false;
+    bool noise_suppressor_enabled_ = false;
+    bool adaptive_gain_controller_enabled_ = false;
+    bool voice_activity_detector_enabled_ = false;
+    bool gain_controller2_enabled_ = false;
+    bool gain_adjustment_enabled_ = false;
+    bool echo_controller_enabled_ = false;
+    bool transient_suppressor_enabled_ = false;
+    bool first_update_ = true;
+  };
+
+  // Methods for modifying the formats struct that is used by both
+  // the render and capture threads. The check for whether modifications are
+  // needed is done while holding a single lock only, thereby avoiding that the
+  // capture thread blocks the render thread.
+  // Called by render: Holds the render lock when reading the format struct and
+  // acquires both locks if reinitialization is required.
+  void MaybeInitializeRender(const StreamConfig& input_config,
+                             const StreamConfig& output_config)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
+  // Called by capture: Acquires and releases the capture lock to read the
+  // format struct and acquires both locks if reinitialization is needed.
+  void MaybeInitializeCapture(const StreamConfig& input_config,
+                              const StreamConfig& output_config);
+
+  // Method for updating the state keeping track of the active submodules.
+  // Returns a bool indicating whether the state has changed.
+  bool UpdateActiveSubmoduleStates()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  // Methods requiring APM running in a single-threaded manner, requiring both
+  // the render and capture lock to be acquired.
+  void InitializeLocked(const ProcessingConfig& config)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
+  void InitializeResidualEchoDetector()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
+  void InitializeEchoController()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
+
+  // Initializations of capture-only sub-modules, requiring the capture lock
+  // already acquired.
+  void InitializeHighPassFilter(bool forced_reset)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  void InitializeGainController1() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  void InitializeTransientSuppressor()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  // Initializes the `GainController2` sub-module. If the sub-module is enabled,
+  // recreates it.
+  void InitializeGainController2() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  // Initializes the `VoiceActivityDetectorWrapper` sub-module. If the
+  // sub-module is enabled, recreates it. Call `InitializeGainController2()`
+  // first.
+  // TODO(bugs.webrtc.org/13663): Remove if TS is removed otherwise remove call
+  // order requirement - i.e., decouple from `InitializeGainController2()`.
+  void InitializeVoiceActivityDetector()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  void InitializeNoiseSuppressor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  void InitializeCaptureLevelsAdjuster()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  void InitializePostProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  void InitializeAnalyzer() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  // Initializations of render-only submodules, requiring the render lock
+  // already acquired.
+  void InitializePreProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
+
+  // Sample rate used for the fullband processing.
+  int proc_fullband_sample_rate_hz() const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  // Empties and handles the respective RuntimeSetting queues.
+  void HandleCaptureRuntimeSettings()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  void HandleRenderRuntimeSettings()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
+
+  void EmptyQueuedRenderAudio() RTC_LOCKS_EXCLUDED(mutex_capture_);
+  void EmptyQueuedRenderAudioLocked()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+  void AllocateRenderQueue()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
+  void QueueBandedRenderAudio(AudioBuffer* audio)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
+  void QueueNonbandedRenderAudio(AudioBuffer* audio)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
+
+  // Capture-side exclusive methods possibly running APM in a multi-threaded
+  // manner that are called with the render lock already acquired.
+  int ProcessCaptureStreamLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  // Render-side exclusive methods possibly running APM in a multi-threaded
+  // manner that are called with the render lock already acquired.
+  int AnalyzeReverseStreamLocked(const float* const* src,
+                                 const StreamConfig& input_config,
+                                 const StreamConfig& output_config)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
+  int ProcessRenderStreamLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
+
+  // Collects configuration settings from public and private
+  // submodules to be saved as an audioproc::Config message on the
+  // AecDump if it is attached.  If not `forced`, only writes the current
+  // config if it is different from the last saved one; if `forced`,
+  // writes the config regardless of the last saved.
+  void WriteAecDumpConfigMessage(bool forced)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  // Notifies attached AecDump of current configuration and capture data.
+  void RecordUnprocessedCaptureStream(const float* const* capture_stream)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  void RecordUnprocessedCaptureStream(const int16_t* const data,
+                                      const StreamConfig& config)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  // Notifies attached AecDump of current configuration and
+  // processed capture data and issues a capture stream recording
+  // request.
+  void RecordProcessedCaptureStream(
+      const float* const* processed_capture_stream)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  void RecordProcessedCaptureStream(const int16_t* const data,
+                                    const StreamConfig& config)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  // Notifies attached AecDump about current state (delay, drift, etc).
+  void RecordAudioProcessingState()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  // Ensures that overruns in the capture runtime settings queue is properly
+  // handled by the code, providing safe-fallbacks to mitigate the implications
+  // of any settings being missed.
+  void HandleOverrunInCaptureRuntimeSettingsQueue()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+
+  // AecDump instance used for optionally logging APM config, input
+  // and output to file in the AEC-dump format defined in debug.proto.
+  std::unique_ptr<AecDump> aec_dump_;
+
+  // Hold the last config written with AecDump for avoiding writing
+  // the same config twice.
+  InternalAPMConfig apm_config_for_aec_dump_ RTC_GUARDED_BY(mutex_capture_);
+
+  // Critical sections.
+  mutable Mutex mutex_render_ RTC_ACQUIRED_BEFORE(mutex_capture_);
+  mutable Mutex mutex_capture_;
+
+  // Struct containing the Config specifying the behavior of APM.
+  AudioProcessing::Config config_;
+
+  // Overrides for testing the exclusion of some submodules from the build.
+  ApmSubmoduleCreationOverrides submodule_creation_overrides_
+      RTC_GUARDED_BY(mutex_capture_);
+
+  // Class containing information about what submodules are active.
+  SubmoduleStates submodule_states_;
+
+  // Struct containing the pointers to the submodules.
+  struct Submodules {
+    Submodules(std::unique_ptr<CustomProcessing> capture_post_processor,
+               std::unique_ptr<CustomProcessing> render_pre_processor,
+               rtc::scoped_refptr<EchoDetector> echo_detector,
+               std::unique_ptr<CustomAudioAnalyzer> capture_analyzer)
+        : echo_detector(std::move(echo_detector)),
+          capture_post_processor(std::move(capture_post_processor)),
+          render_pre_processor(std::move(render_pre_processor)),
+          capture_analyzer(std::move(capture_analyzer)) {}
+    // Accessed internally from capture or during initialization.
+    const rtc::scoped_refptr<EchoDetector> echo_detector;
+    const std::unique_ptr<CustomProcessing> capture_post_processor;
+    const std::unique_ptr<CustomProcessing> render_pre_processor;
+    const std::unique_ptr<CustomAudioAnalyzer> capture_analyzer;
+    std::unique_ptr<AgcManagerDirect> agc_manager;
+    std::unique_ptr<GainControlImpl> gain_control;
+    std::unique_ptr<GainController2> gain_controller2;
+    std::unique_ptr<VoiceActivityDetectorWrapper> voice_activity_detector;
+    std::unique_ptr<HighPassFilter> high_pass_filter;
+    std::unique_ptr<EchoControl> echo_controller;
+    std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
+    std::unique_ptr<NoiseSuppressor> noise_suppressor;
+    std::unique_ptr<TransientSuppressor> transient_suppressor;
+    std::unique_ptr<CaptureLevelsAdjuster> capture_levels_adjuster;
+  } submodules_;
+
+  // State that is written to while holding both the render and capture locks
+  // but can be read without any lock being held.
+  // As this is only accessed internally of APM, and all internal methods in APM
+  // either are holding the render or capture locks, this construct is safe as
+  // it is not possible to read the variables while writing them.
+  struct ApmFormatState {
+    ApmFormatState()
+        :  // Format of processing streams at input/output call sites.
+          api_format({{{kSampleRate16kHz, 1},
+                       {kSampleRate16kHz, 1},
+                       {kSampleRate16kHz, 1},
+                       {kSampleRate16kHz, 1}}}),
+          render_processing_format(kSampleRate16kHz, 1) {}
+    ProcessingConfig api_format;
+    StreamConfig render_processing_format;
+  } formats_;
+
+  // APM constants.
+  const struct ApmConstants {
+    ApmConstants(bool multi_channel_render_support,
+                 bool multi_channel_capture_support,
+                 bool enforce_split_band_hpf,
+                 bool minimize_processing_for_unused_output,
+                 bool transient_suppressor_forced_off)
+        : multi_channel_render_support(multi_channel_render_support),
+          multi_channel_capture_support(multi_channel_capture_support),
+          enforce_split_band_hpf(enforce_split_band_hpf),
+          minimize_processing_for_unused_output(
+              minimize_processing_for_unused_output),
+          transient_suppressor_forced_off(transient_suppressor_forced_off) {}
+    bool multi_channel_render_support;
+    bool multi_channel_capture_support;
+    bool enforce_split_band_hpf;
+    bool minimize_processing_for_unused_output;
+    bool transient_suppressor_forced_off;
+  } constants_;
+
+  struct ApmCaptureState {
+    ApmCaptureState();
+    ~ApmCaptureState();
+    bool was_stream_delay_set;
+    bool capture_output_used;
+    bool capture_output_used_last_frame;
+    bool key_pressed;
+    std::unique_ptr<AudioBuffer> capture_audio;
+    std::unique_ptr<AudioBuffer> capture_fullband_audio;
+    std::unique_ptr<AudioBuffer> linear_aec_output;
+    // Only the rate and samples fields of capture_processing_format_ are used
+    // because the capture processing number of channels is mutable and is
+    // tracked by the capture_audio_.
+    StreamConfig capture_processing_format;
+    int split_rate;
+    bool echo_path_gain_change;
+    float prev_pre_adjustment_gain;
+    int playout_volume;
+    int prev_playout_volume;
+    AudioProcessingStats stats;
+    // Input volume applied on the audio input device when the audio is
+    // acquired. Unspecified when unknown.
+    absl::optional<int> applied_input_volume;
+    bool applied_input_volume_changed;
+    // Recommended input volume to apply on the audio input device the next time
+    // that audio is acquired. Unspecified when no input volume can be
+    // recommended.
+    absl::optional<int> recommended_input_volume;
+  } capture_ RTC_GUARDED_BY(mutex_capture_);
+
+  struct ApmCaptureNonLockedState {
+    ApmCaptureNonLockedState()
+        : capture_processing_format(kSampleRate16kHz),
+          split_rate(kSampleRate16kHz),
+          stream_delay_ms(0) {}
+    // Only the rate and samples fields of capture_processing_format_ are used
+    // because the forward processing number of channels is mutable and is
+    // tracked by the capture_audio_.
+    StreamConfig capture_processing_format;
+    int split_rate;
+    int stream_delay_ms;
+    bool echo_controller_enabled = false;
+  } capture_nonlocked_;
+
+  struct ApmRenderState {
+    ApmRenderState();
+    ~ApmRenderState();
+    std::unique_ptr<AudioConverter> render_converter;
+    std::unique_ptr<AudioBuffer> render_audio;
+  } render_ RTC_GUARDED_BY(mutex_render_);
+
+  // Class for statistics reporting. The class is thread-safe and no lock is
+  // needed when accessing it.
+  class ApmStatsReporter {
+   public:
+    ApmStatsReporter();
+    ~ApmStatsReporter();
+
+    // Returns the most recently reported statistics.
+    AudioProcessingStats GetStatistics();
+
+    // Update the cached statistics.
+    void UpdateStatistics(const AudioProcessingStats& new_stats);
+
+   private:
+    Mutex mutex_stats_;
+    AudioProcessingStats cached_stats_ RTC_GUARDED_BY(mutex_stats_);
+    SwapQueue<AudioProcessingStats> stats_message_queue_;
+  } stats_reporter_;
+
+  std::vector<int16_t> aecm_render_queue_buffer_ RTC_GUARDED_BY(mutex_render_);
+  std::vector<int16_t> aecm_capture_queue_buffer_
+      RTC_GUARDED_BY(mutex_capture_);
+
+  size_t agc_render_queue_element_max_size_ RTC_GUARDED_BY(mutex_render_)
+      RTC_GUARDED_BY(mutex_capture_) = 0;
+  std::vector<int16_t> agc_render_queue_buffer_ RTC_GUARDED_BY(mutex_render_);
+  std::vector<int16_t> agc_capture_queue_buffer_ RTC_GUARDED_BY(mutex_capture_);
+
+  size_t red_render_queue_element_max_size_ RTC_GUARDED_BY(mutex_render_)
+      RTC_GUARDED_BY(mutex_capture_) = 0;
+  std::vector<float> red_render_queue_buffer_ RTC_GUARDED_BY(mutex_render_);
+  std::vector<float> red_capture_queue_buffer_ RTC_GUARDED_BY(mutex_capture_);
+
+  RmsLevel capture_input_rms_ RTC_GUARDED_BY(mutex_capture_);
+  RmsLevel capture_output_rms_ RTC_GUARDED_BY(mutex_capture_);
+  int capture_rms_interval_counter_ RTC_GUARDED_BY(mutex_capture_) = 0;
+
+  InputVolumeStatsReporter applied_input_volume_stats_reporter_
+      RTC_GUARDED_BY(mutex_capture_);
+  InputVolumeStatsReporter recommended_input_volume_stats_reporter_
+      RTC_GUARDED_BY(mutex_capture_);
+
+  // Lock protection not needed.
+  std::unique_ptr<
+      SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
+      aecm_render_signal_queue_;
+  std::unique_ptr<
+      SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
+      agc_render_signal_queue_;
+  std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
+      red_render_signal_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
new file mode 100644
index 0000000000..3614b574df
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -0,0 +1,1012 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/audio_processing_impl.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "modules/audio_processing/test/test_utils.h"
+#include "rtc_base/event.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/random.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kMaxFrameSize = 480;
+constexpr TimeDelta kTestTimeOutLimit = TimeDelta::Minutes(10);
+
+class AudioProcessingImplLockTest;
+
+// Type of the render thread APM API call to use in the test.
+enum class RenderApiImpl {
+  ProcessReverseStreamImplInteger,
+  ProcessReverseStreamImplFloat,
+  AnalyzeReverseStreamImplFloat,
+};
+
+// Type of the capture thread APM API call to use in the test.
+enum class CaptureApiImpl { ProcessStreamImplInteger, ProcessStreamImplFloat };
+
+// The runtime parameter setting scheme to use in the test.
+enum class RuntimeParameterSettingScheme {
+  SparseStreamMetadataChangeScheme,
+  ExtremeStreamMetadataChangeScheme,
+  FixedMonoStreamMetadataScheme,
+  FixedStereoStreamMetadataScheme
+};
+
+// Variant of echo canceller settings to use in the test.
+enum class AecType {
+  BasicWebRtcAecSettings,
+  AecTurnedOff,
+  BasicWebRtcAecSettingsWithExtentedFilter,
+  BasicWebRtcAecSettingsWithDelayAgnosticAec,
+  BasicWebRtcAecSettingsWithAecMobile
+};
+
+// Thread-safe random number generator wrapper.
+class RandomGenerator {
+ public:
+  RandomGenerator() : rand_gen_(42U) {}
+
+  int RandInt(int min, int max) {
+    MutexLock lock(&mutex_);
+    return rand_gen_.Rand(min, max);
+  }
+
+  int RandInt(int max) {
+    MutexLock lock(&mutex_);
+    return rand_gen_.Rand(max);
+  }
+
+  float RandFloat() {
+    MutexLock lock(&mutex_);
+    return rand_gen_.Rand<float>();
+  }
+
+ private:
+  Mutex mutex_;
+  Random rand_gen_ RTC_GUARDED_BY(mutex_);
+};
+
+// Variables related to the audio data and formats.
+struct AudioFrameData {
+  explicit AudioFrameData(int max_frame_size) {
+    // Set up the two-dimensional arrays needed for the APM API calls.
+    input_framechannels.resize(2 * max_frame_size);
+    input_frame.resize(2);
+    input_frame[0] = &input_framechannels[0];
+    input_frame[1] = &input_framechannels[max_frame_size];
+
+    output_frame_channels.resize(2 * max_frame_size);
+    output_frame.resize(2);
+    output_frame[0] = &output_frame_channels[0];
+    output_frame[1] = &output_frame_channels[max_frame_size];
+
+    frame.resize(2 * max_frame_size);
+  }
+
+  std::vector<int16_t> frame;
+
+  std::vector<float*> output_frame;
+  std::vector<float> output_frame_channels;
+  std::vector<float*> input_frame;
+  std::vector<float> input_framechannels;
+
+  int input_sample_rate_hz = 16000;
+  int input_number_of_channels = 1;
+  int output_sample_rate_hz = 16000;
+  int output_number_of_channels = 1;
+};
+
+// The configuration for the test.
+struct TestConfig {
+  // Test case generator for the test configurations to use in the brief tests.
+  static std::vector<TestConfig> GenerateBriefTestConfigs() {
+    std::vector<TestConfig> test_configs;
+    AecType aec_types[] = {AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec,
+                           AecType::BasicWebRtcAecSettingsWithAecMobile};
+    for (auto aec_type : aec_types) {
+      TestConfig test_config;
+      test_config.aec_type = aec_type;
+
+      test_config.min_number_of_calls = 300;
+
+      // Perform tests only with the extreme runtime parameter setting scheme.
+      test_config.runtime_parameter_setting_scheme =
+          RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme;
+
+      // Only test 16 kHz for this test suite.
+      test_config.initial_sample_rate_hz = 16000;
+
+      // Create test config for the Int16 processing API function set.
+      test_config.render_api_function =
+          RenderApiImpl::ProcessReverseStreamImplInteger;
+      test_config.capture_api_function =
+          CaptureApiImpl::ProcessStreamImplInteger;
+      test_configs.push_back(test_config);
+
+      // Create test config for the StreamConfig processing API function set.
+      test_config.render_api_function =
+          RenderApiImpl::ProcessReverseStreamImplFloat;
+      test_config.capture_api_function = CaptureApiImpl::ProcessStreamImplFloat;
+      test_configs.push_back(test_config);
+    }
+
+    // Return the created test configurations.
+    return test_configs;
+  }
+
+  // Test case generator for the test configurations to use in the extensive
+  // tests.
+  static std::vector<TestConfig> GenerateExtensiveTestConfigs() {
+    // Lambda functions for the test config generation.
+    auto add_processing_apis = [](TestConfig test_config) {
+      struct AllowedApiCallCombinations {
+        RenderApiImpl render_api;
+        CaptureApiImpl capture_api;
+      };
+
+      const AllowedApiCallCombinations api_calls[] = {
+          {RenderApiImpl::ProcessReverseStreamImplInteger,
+           CaptureApiImpl::ProcessStreamImplInteger},
+          {RenderApiImpl::ProcessReverseStreamImplFloat,
+           CaptureApiImpl::ProcessStreamImplFloat},
+          {RenderApiImpl::AnalyzeReverseStreamImplFloat,
+           CaptureApiImpl::ProcessStreamImplFloat},
+          {RenderApiImpl::ProcessReverseStreamImplInteger,
+           CaptureApiImpl::ProcessStreamImplFloat},
+          {RenderApiImpl::ProcessReverseStreamImplFloat,
+           CaptureApiImpl::ProcessStreamImplInteger}};
+      std::vector<TestConfig> out;
+      for (auto api_call : api_calls) {
+        test_config.render_api_function = api_call.render_api;
+        test_config.capture_api_function = api_call.capture_api;
+        out.push_back(test_config);
+      }
+      return out;
+    };
+
+    auto add_aec_settings = [](const std::vector<TestConfig>& in) {
+      std::vector<TestConfig> out;
+      AecType aec_types[] = {
+          AecType::BasicWebRtcAecSettings, AecType::AecTurnedOff,
+          AecType::BasicWebRtcAecSettingsWithExtentedFilter,
+          AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec,
+          AecType::BasicWebRtcAecSettingsWithAecMobile};
+      for (auto test_config : in) {
+        // Due to a VisualStudio 2015 compiler issue, the internal loop
+        // variable here cannot override a previously defined name.
+        // In other words "type" cannot be named "aec_type" here.
+        // https://connect.microsoft.com/VisualStudio/feedback/details/2291755
+        for (auto type : aec_types) {
+          test_config.aec_type = type;
+          out.push_back(test_config);
+        }
+      }
+      return out;
+    };
+
+    auto add_settings_scheme = [](const std::vector<TestConfig>& in) {
+      std::vector<TestConfig> out;
+      RuntimeParameterSettingScheme schemes[] = {
+          RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme,
+          RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme,
+          RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme,
+          RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme};
+
+      for (auto test_config : in) {
+        for (auto scheme : schemes) {
+          test_config.runtime_parameter_setting_scheme = scheme;
+          out.push_back(test_config);
+        }
+      }
+      return out;
+    };
+
+    auto add_sample_rates = [](const std::vector<TestConfig>& in) {
+      const int sample_rates[] = {8000, 16000, 32000, 48000};
+
+      std::vector<TestConfig> out;
+      for (auto test_config : in) {
+        auto available_rates =
+            (test_config.aec_type ==
+                     AecType::BasicWebRtcAecSettingsWithAecMobile
+                 ? rtc::ArrayView<const int>(sample_rates, 2)
+                 : rtc::ArrayView<const int>(sample_rates));
+
+        for (auto rate : available_rates) {
+          test_config.initial_sample_rate_hz = rate;
+          out.push_back(test_config);
+        }
+      }
+      return out;
+    };
+
+    // Generate test configurations of the relevant combinations of the
+    // parameters to
+    // test.
+    TestConfig test_config;
+    test_config.min_number_of_calls = 10000;
+    return add_sample_rates(add_settings_scheme(
+        add_aec_settings(add_processing_apis(test_config))));
+  }
+
+  RenderApiImpl render_api_function =
+      RenderApiImpl::ProcessReverseStreamImplFloat;
+  CaptureApiImpl capture_api_function = CaptureApiImpl::ProcessStreamImplFloat;
+  RuntimeParameterSettingScheme runtime_parameter_setting_scheme =
+      RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme;
+  int initial_sample_rate_hz = 16000;
+  AecType aec_type = AecType::BasicWebRtcAecSettingsWithDelayAgnosticAec;
+  int min_number_of_calls = 300;
+};
+
+// Handler for the frame counters.
+class FrameCounters {
+ public:
+  void IncreaseRenderCounter() {
+    MutexLock lock(&mutex_);
+    render_count++;
+  }
+
+  void IncreaseCaptureCounter() {
+    MutexLock lock(&mutex_);
+    capture_count++;
+  }
+
+  int GetCaptureCounter() const {
+    MutexLock lock(&mutex_);
+    return capture_count;
+  }
+
+  int GetRenderCounter() const {
+    MutexLock lock(&mutex_);
+    return render_count;
+  }
+
+  int CaptureMinusRenderCounters() const {
+    MutexLock lock(&mutex_);
+    return capture_count - render_count;
+  }
+
+  int RenderMinusCaptureCounters() const {
+    return -CaptureMinusRenderCounters();
+  }
+
+  bool BothCountersExceedeThreshold(int threshold) {
+    MutexLock lock(&mutex_);
+    return (render_count > threshold && capture_count > threshold);
+  }
+
+ private:
+  mutable Mutex mutex_;
+  int render_count RTC_GUARDED_BY(mutex_) = 0;
+  int capture_count RTC_GUARDED_BY(mutex_) = 0;
+};
+
+// Class for handling the capture side processing.
+class CaptureProcessor {
+ public:
+  CaptureProcessor(int max_frame_size,
+                   RandomGenerator* rand_gen,
+                   rtc::Event* render_call_event,
+                   rtc::Event* capture_call_event,
+                   FrameCounters* shared_counters_state,
+                   const TestConfig* test_config,
+                   AudioProcessing* apm);
+  void Process();
+
+ private:
+  static constexpr int kMaxCallDifference = 10;
+  static constexpr float kCaptureInputFloatLevel = 0.03125f;
+  static constexpr int kCaptureInputFixLevel = 1024;
+
+  void PrepareFrame();
+  void CallApmCaptureSide();
+  void ApplyRuntimeSettingScheme();
+
+  RandomGenerator* const rand_gen_ = nullptr;
+  rtc::Event* const render_call_event_ = nullptr;
+  rtc::Event* const capture_call_event_ = nullptr;
+  FrameCounters* const frame_counters_ = nullptr;
+  const TestConfig* const test_config_ = nullptr;
+  AudioProcessing* const apm_ = nullptr;
+  AudioFrameData frame_data_;
+};
+
+// Class for handling the stats processing.
+class StatsProcessor {
+ public:
+  StatsProcessor(RandomGenerator* rand_gen,
+                 const TestConfig* test_config,
+                 AudioProcessing* apm);
+  void Process();
+
+ private:
+  RandomGenerator* rand_gen_ = nullptr;
+  const TestConfig* const test_config_ = nullptr;
+  AudioProcessing* apm_ = nullptr;
+};
+
+// Class for handling the render side processing.
+class RenderProcessor {
+ public:
+  RenderProcessor(int max_frame_size,
+                  RandomGenerator* rand_gen,
+                  rtc::Event* render_call_event,
+                  rtc::Event* capture_call_event,
+                  FrameCounters* shared_counters_state,
+                  const TestConfig* test_config,
+                  AudioProcessing* apm);
+  void Process();
+
+ private:
+  static constexpr int kMaxCallDifference = 10;
+  static constexpr int kRenderInputFixLevel = 16384;
+  static constexpr float kRenderInputFloatLevel = 0.5f;
+
+  void PrepareFrame();
+  void CallApmRenderSide();
+  void ApplyRuntimeSettingScheme();
+
+  RandomGenerator* const rand_gen_ = nullptr;
+  rtc::Event* const render_call_event_ = nullptr;
+  rtc::Event* const capture_call_event_ = nullptr;
+  FrameCounters* const frame_counters_ = nullptr;
+  const TestConfig* const test_config_ = nullptr;
+  AudioProcessing* const apm_ = nullptr;
+  AudioFrameData frame_data_;
+  bool first_render_call_ = true;
+};
+
+class AudioProcessingImplLockTest
+    : public ::testing::TestWithParam<TestConfig> {
+ public:
+  AudioProcessingImplLockTest();
+  bool RunTest();
+  bool MaybeEndTest();
+
+ private:
+  void SetUp() override;
+  void TearDown() override;
+
+  // Tests whether all the required render and capture side calls have been
+  // done.
+  bool TestDone() {
+    return frame_counters_.BothCountersExceedeThreshold(
+        test_config_.min_number_of_calls);
+  }
+
+  // Start the threads used in the test.
+  void StartThreads() {
+    const auto attributes =
+        rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
+    render_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (!MaybeEndTest())
+            render_thread_state_.Process();
+        },
+        "render", attributes);
+    capture_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (!MaybeEndTest()) {
+            capture_thread_state_.Process();
+          }
+        },
+        "capture", attributes);
+
+    stats_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (!MaybeEndTest())
+            stats_thread_state_.Process();
+        },
+        "stats", attributes);
+  }
+
+  // Event handlers for the test.
+  rtc::Event test_complete_;
+  rtc::Event render_call_event_;
+  rtc::Event capture_call_event_;
+
+  // Thread related variables.
+  mutable RandomGenerator rand_gen_;
+
+  const TestConfig test_config_;
+  rtc::scoped_refptr<AudioProcessing> apm_;
+  FrameCounters frame_counters_;
+  RenderProcessor render_thread_state_;
+  CaptureProcessor capture_thread_state_;
+  StatsProcessor stats_thread_state_;
+  rtc::PlatformThread render_thread_;
+  rtc::PlatformThread capture_thread_;
+  rtc::PlatformThread stats_thread_;
+};
+
+// Sleeps a random time between 0 and max_sleep milliseconds.
+void SleepRandomMs(int max_sleep, RandomGenerator* rand_gen) {
+  int sleeptime = rand_gen->RandInt(0, max_sleep);
+  SleepMs(sleeptime);
+}
+
+// Populates a float audio frame with random data.
+void PopulateAudioFrame(float** frame,
+                        float amplitude,
+                        size_t num_channels,
+                        size_t samples_per_channel,
+                        RandomGenerator* rand_gen) {
+  for (size_t ch = 0; ch < num_channels; ch++) {
+    for (size_t k = 0; k < samples_per_channel; k++) {
+      // Store random 16 bit quantized float number between +-amplitude.
+      frame[ch][k] = amplitude * (2 * rand_gen->RandFloat() - 1);
+    }
+  }
+}
+
+// Populates an integer audio frame with random data.
+void PopulateAudioFrame(float amplitude,
+                        size_t num_channels,
+                        size_t samples_per_channel,
+                        rtc::ArrayView<int16_t> frame,
+                        RandomGenerator* rand_gen) {
+  ASSERT_GT(amplitude, 0);
+  ASSERT_LE(amplitude, 32767);
+  for (size_t ch = 0; ch < num_channels; ch++) {
+    for (size_t k = 0; k < samples_per_channel; k++) {
+      // Store random 16 bit number between -(amplitude+1) and
+      // amplitude.
+      frame[k * ch] = rand_gen->RandInt(2 * amplitude + 1) - amplitude - 1;
+    }
+  }
+}
+
+AudioProcessing::Config GetApmTestConfig(AecType aec_type) {
+  AudioProcessing::Config apm_config;
+  apm_config.echo_canceller.enabled = aec_type != AecType::AecTurnedOff;
+  apm_config.echo_canceller.mobile_mode =
+      aec_type == AecType::BasicWebRtcAecSettingsWithAecMobile;
+  apm_config.gain_controller1.enabled = true;
+  apm_config.gain_controller1.mode =
+      AudioProcessing::Config::GainController1::kAdaptiveDigital;
+  apm_config.noise_suppression.enabled = true;
+  return apm_config;
+}
+
+AudioProcessingImplLockTest::AudioProcessingImplLockTest()
+    : test_config_(GetParam()),
+      apm_(AudioProcessingBuilderForTesting()
+               .SetConfig(GetApmTestConfig(test_config_.aec_type))
+               .Create()),
+      render_thread_state_(kMaxFrameSize,
+                           &rand_gen_,
+                           &render_call_event_,
+                           &capture_call_event_,
+                           &frame_counters_,
+                           &test_config_,
+                           apm_.get()),
+      capture_thread_state_(kMaxFrameSize,
+                            &rand_gen_,
+                            &render_call_event_,
+                            &capture_call_event_,
+                            &frame_counters_,
+                            &test_config_,
+                            apm_.get()),
+      stats_thread_state_(&rand_gen_, &test_config_, apm_.get()) {}
+
+// Run the test with a timeout.
+bool AudioProcessingImplLockTest::RunTest() {
+  StartThreads();
+  return test_complete_.Wait(kTestTimeOutLimit);
+}
+
+bool AudioProcessingImplLockTest::MaybeEndTest() {
+  if (HasFatalFailure() || TestDone()) {
+    test_complete_.Set();
+    return true;
+  }
+  return false;
+}
+
+void AudioProcessingImplLockTest::SetUp() {}
+
+void AudioProcessingImplLockTest::TearDown() {
+  render_call_event_.Set();
+  capture_call_event_.Set();
+}
+
+StatsProcessor::StatsProcessor(RandomGenerator* rand_gen,
+                               const TestConfig* test_config,
+                               AudioProcessing* apm)
+    : rand_gen_(rand_gen), test_config_(test_config), apm_(apm) {}
+
+// Implements the callback functionality for the statistics
+// collection thread.
+void StatsProcessor::Process() {
+  SleepRandomMs(100, rand_gen_);
+
+  AudioProcessing::Config apm_config = apm_->GetConfig();
+  if (test_config_->aec_type != AecType::AecTurnedOff) {
+    EXPECT_TRUE(apm_config.echo_canceller.enabled);
+    EXPECT_EQ(apm_config.echo_canceller.mobile_mode,
+              (test_config_->aec_type ==
+               AecType::BasicWebRtcAecSettingsWithAecMobile));
+  } else {
+    EXPECT_FALSE(apm_config.echo_canceller.enabled);
+  }
+  EXPECT_TRUE(apm_config.gain_controller1.enabled);
+  EXPECT_TRUE(apm_config.noise_suppression.enabled);
+
+  // The below return value is not testable.
+  apm_->GetStatistics();
+}
+
+CaptureProcessor::CaptureProcessor(int max_frame_size,
+                                   RandomGenerator* rand_gen,
+                                   rtc::Event* render_call_event,
+                                   rtc::Event* capture_call_event,
+                                   FrameCounters* shared_counters_state,
+                                   const TestConfig* test_config,
+                                   AudioProcessing* apm)
+    : rand_gen_(rand_gen),
+      render_call_event_(render_call_event),
+      capture_call_event_(capture_call_event),
+      frame_counters_(shared_counters_state),
+      test_config_(test_config),
+      apm_(apm),
+      frame_data_(max_frame_size) {}
+
+// Implements the callback functionality for the capture thread.
+void CaptureProcessor::Process() {
+  // Sleep a random time to simulate thread jitter.
+  SleepRandomMs(3, rand_gen_);
+
+  // Ensure that the number of render and capture calls do not
+  // differ too much.
+  if (frame_counters_->CaptureMinusRenderCounters() > kMaxCallDifference) {
+    render_call_event_->Wait(rtc::Event::kForever);
+  }
+
+  // Apply any specified capture side APM non-processing runtime calls.
+  ApplyRuntimeSettingScheme();
+
+  // Apply the capture side processing call.
+  CallApmCaptureSide();
+
+  // Increase the number of capture-side calls.
+  frame_counters_->IncreaseCaptureCounter();
+
+  // Flag to the render thread that another capture API call has occurred
+  // by triggering this threads call event.
+  capture_call_event_->Set();
+}
+
+// Prepares a frame with relevant audio data and metadata.
+void CaptureProcessor::PrepareFrame() {
+  // Restrict to a common fixed sample rate if the integer
+  // interface is used.
+  if (test_config_->capture_api_function ==
+      CaptureApiImpl::ProcessStreamImplInteger) {
+    frame_data_.input_sample_rate_hz = test_config_->initial_sample_rate_hz;
+    frame_data_.output_sample_rate_hz = test_config_->initial_sample_rate_hz;
+  }
+
+  // Prepare the audio data.
+  StreamConfig input_stream_config(frame_data_.input_sample_rate_hz,
+                                   frame_data_.input_number_of_channels);
+
+  PopulateAudioFrame(kCaptureInputFixLevel, input_stream_config.num_channels(),
+                     input_stream_config.num_frames(), frame_data_.frame,
+                     rand_gen_);
+
+  PopulateAudioFrame(&frame_data_.input_frame[0], kCaptureInputFloatLevel,
+                     input_stream_config.num_channels(),
+                     input_stream_config.num_frames(), rand_gen_);
+}
+
+// Applies the capture side processing API call.
+void CaptureProcessor::CallApmCaptureSide() {
+  // Prepare a proper capture side processing API call input.
+  PrepareFrame();
+
+  // Set the stream delay.
+  apm_->set_stream_delay_ms(30);
+
+  // Set the analog level.
+  apm_->set_stream_analog_level(80);
+
+  // Call the specified capture side API processing method.
+  StreamConfig input_stream_config(frame_data_.input_sample_rate_hz,
+                                   frame_data_.input_number_of_channels);
+  StreamConfig output_stream_config(frame_data_.output_sample_rate_hz,
+                                    frame_data_.output_number_of_channels);
+  int result = AudioProcessing::kNoError;
+  switch (test_config_->capture_api_function) {
+    case CaptureApiImpl::ProcessStreamImplInteger:
+      result =
+          apm_->ProcessStream(frame_data_.frame.data(), input_stream_config,
+                              output_stream_config, frame_data_.frame.data());
+      break;
+    case CaptureApiImpl::ProcessStreamImplFloat:
+      result = apm_->ProcessStream(&frame_data_.input_frame[0],
+                                   input_stream_config, output_stream_config,
+                                   &frame_data_.output_frame[0]);
+      break;
+    default:
+      FAIL();
+  }
+
+  // Retrieve the new analog level.
+  apm_->recommended_stream_analog_level();
+
+  // Check the return code for error.
+  ASSERT_EQ(AudioProcessing::kNoError, result);
+}
+
+// Applies any runtime capture APM API calls and audio stream characteristics
+// specified by the scheme for the test.
+void CaptureProcessor::ApplyRuntimeSettingScheme() {
+  const int capture_count_local = frame_counters_->GetCaptureCounter();
+
+  // Update the number of channels and sample rates for the input and output.
+  // Note that the counts frequencies for when to set parameters
+  // are set using prime numbers in order to ensure that the
+  // permutation scheme in the parameter setting changes.
+  switch (test_config_->runtime_parameter_setting_scheme) {
+    case RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme:
+      if (capture_count_local == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (capture_count_local % 11 == 0)
+        frame_data_.input_sample_rate_hz = 32000;
+      else if (capture_count_local % 73 == 0)
+        frame_data_.input_sample_rate_hz = 48000;
+      else if (capture_count_local % 89 == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (capture_count_local % 97 == 0)
+        frame_data_.input_sample_rate_hz = 8000;
+
+      if (capture_count_local == 0)
+        frame_data_.input_number_of_channels = 1;
+      else if (capture_count_local % 4 == 0)
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+
+      if (capture_count_local == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (capture_count_local % 5 == 0)
+        frame_data_.output_sample_rate_hz = 32000;
+      else if (capture_count_local % 47 == 0)
+        frame_data_.output_sample_rate_hz = 48000;
+      else if (capture_count_local % 53 == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (capture_count_local % 71 == 0)
+        frame_data_.output_sample_rate_hz = 8000;
+
+      if (capture_count_local == 0)
+        frame_data_.output_number_of_channels = 1;
+      else if (capture_count_local % 8 == 0)
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+      break;
+    case RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme:
+      if (capture_count_local % 2 == 0) {
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+      } else {
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.input_sample_rate_hz == 8000)
+          frame_data_.input_sample_rate_hz = 16000;
+        else if (frame_data_.input_sample_rate_hz == 16000)
+          frame_data_.input_sample_rate_hz = 32000;
+        else if (frame_data_.input_sample_rate_hz == 32000)
+          frame_data_.input_sample_rate_hz = 48000;
+        else if (frame_data_.input_sample_rate_hz == 48000)
+          frame_data_.input_sample_rate_hz = 8000;
+
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.output_sample_rate_hz == 8000)
+          frame_data_.output_sample_rate_hz = 16000;
+        else if (frame_data_.output_sample_rate_hz == 16000)
+          frame_data_.output_sample_rate_hz = 32000;
+        else if (frame_data_.output_sample_rate_hz == 32000)
+          frame_data_.output_sample_rate_hz = 48000;
+        else if (frame_data_.output_sample_rate_hz == 48000)
+          frame_data_.output_sample_rate_hz = 8000;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme:
+      if (capture_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme:
+      if (capture_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 2;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 2;
+      }
+      break;
+    default:
+      FAIL();
+  }
+
+  // Call any specified runtime APM setter and
+  // getter calls.
+  switch (test_config_->runtime_parameter_setting_scheme) {
+    case RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme:
+    case RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme:
+      break;
+    case RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme:
+    case RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme:
+      if (capture_count_local % 2 == 0) {
+        ASSERT_EQ(AudioProcessing::Error::kNoError,
+                  apm_->set_stream_delay_ms(30));
+        apm_->set_stream_key_pressed(true);
+      } else {
+        ASSERT_EQ(AudioProcessing::Error::kNoError,
+                  apm_->set_stream_delay_ms(50));
+        apm_->set_stream_key_pressed(false);
+      }
+      break;
+    default:
+      FAIL();
+  }
+
+  // Restric the number of output channels not to exceed
+  // the number of input channels.
+  frame_data_.output_number_of_channels =
+      std::min(frame_data_.output_number_of_channels,
+               frame_data_.input_number_of_channels);
+}
+
+RenderProcessor::RenderProcessor(int max_frame_size,
+                                 RandomGenerator* rand_gen,
+                                 rtc::Event* render_call_event,
+                                 rtc::Event* capture_call_event,
+                                 FrameCounters* shared_counters_state,
+                                 const TestConfig* test_config,
+                                 AudioProcessing* apm)
+    : rand_gen_(rand_gen),
+      render_call_event_(render_call_event),
+      capture_call_event_(capture_call_event),
+      frame_counters_(shared_counters_state),
+      test_config_(test_config),
+      apm_(apm),
+      frame_data_(max_frame_size) {}
+
+// Implements the callback functionality for the render thread.
+void RenderProcessor::Process() {
+  // Conditional wait to ensure that a capture call has been done
+  // before the first render call is performed (implicitly
+  // required by the APM API).
+  if (first_render_call_) {
+    capture_call_event_->Wait(rtc::Event::kForever);
+    first_render_call_ = false;
+  }
+
+  // Sleep a random time to simulate thread jitter.
+  SleepRandomMs(3, rand_gen_);
+
+  // Ensure that the number of render and capture calls do not
+  // differ too much.
+  if (frame_counters_->RenderMinusCaptureCounters() > kMaxCallDifference) {
+    capture_call_event_->Wait(rtc::Event::kForever);
+  }
+
+  // Apply any specified render side APM non-processing runtime calls.
+  ApplyRuntimeSettingScheme();
+
+  // Apply the render side processing call.
+  CallApmRenderSide();
+
+  // Increase the number of render-side calls.
+  frame_counters_->IncreaseRenderCounter();
+
+  // Flag to the capture thread that another render API call has occurred
+  // by triggering this threads call event.
+  render_call_event_->Set();
+}
+
+// Prepares the render side frame and the accompanying metadata
+// with the appropriate information.
+void RenderProcessor::PrepareFrame() {
+  // Restrict to a common fixed sample rate if the integer interface is
+  // used.
+  if ((test_config_->render_api_function ==
+       RenderApiImpl::ProcessReverseStreamImplInteger) ||
+      (test_config_->aec_type !=
+       AecType::BasicWebRtcAecSettingsWithAecMobile)) {
+    frame_data_.input_sample_rate_hz = test_config_->initial_sample_rate_hz;
+    frame_data_.output_sample_rate_hz = test_config_->initial_sample_rate_hz;
+  }
+
+  // Prepare the audio data.
+  StreamConfig input_stream_config(frame_data_.input_sample_rate_hz,
+                                   frame_data_.input_number_of_channels);
+
+  PopulateAudioFrame(kRenderInputFixLevel, input_stream_config.num_channels(),
+                     input_stream_config.num_frames(), frame_data_.frame,
+                     rand_gen_);
+
+  PopulateAudioFrame(&frame_data_.input_frame[0], kRenderInputFloatLevel,
+                     input_stream_config.num_channels(),
+                     input_stream_config.num_frames(), rand_gen_);
+}
+
+// Makes the render side processing API call.
+void RenderProcessor::CallApmRenderSide() {
+  // Prepare a proper render side processing API call input.
+  PrepareFrame();
+
+  // Call the specified render side API processing method.
+  StreamConfig input_stream_config(frame_data_.input_sample_rate_hz,
+                                   frame_data_.input_number_of_channels);
+  StreamConfig output_stream_config(frame_data_.output_sample_rate_hz,
+                                    frame_data_.output_number_of_channels);
+  int result = AudioProcessing::kNoError;
+  switch (test_config_->render_api_function) {
+    case RenderApiImpl::ProcessReverseStreamImplInteger:
+      result = apm_->ProcessReverseStream(
+          frame_data_.frame.data(), input_stream_config, output_stream_config,
+          frame_data_.frame.data());
+      break;
+    case RenderApiImpl::ProcessReverseStreamImplFloat:
+      result = apm_->ProcessReverseStream(
+          &frame_data_.input_frame[0], input_stream_config,
+          output_stream_config, &frame_data_.output_frame[0]);
+      break;
+    case RenderApiImpl::AnalyzeReverseStreamImplFloat:
+      result = apm_->AnalyzeReverseStream(&frame_data_.input_frame[0],
+                                          input_stream_config);
+      break;
+    default:
+      FAIL();
+  }
+
+  // Check the return code for error.
+  ASSERT_EQ(AudioProcessing::kNoError, result);
+}
+
+// Applies any render capture side APM API calls and audio stream
+// characteristics
+// specified by the scheme for the test.
+void RenderProcessor::ApplyRuntimeSettingScheme() {
+  const int render_count_local = frame_counters_->GetRenderCounter();
+
+  // Update the number of channels and sample rates for the input and output.
+  // Note that the counts frequencies for when to set parameters
+  // are set using prime numbers in order to ensure that the
+  // permutation scheme in the parameter setting changes.
+  switch (test_config_->runtime_parameter_setting_scheme) {
+    case RuntimeParameterSettingScheme::SparseStreamMetadataChangeScheme:
+      if (render_count_local == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (render_count_local % 47 == 0)
+        frame_data_.input_sample_rate_hz = 32000;
+      else if (render_count_local % 71 == 0)
+        frame_data_.input_sample_rate_hz = 48000;
+      else if (render_count_local % 79 == 0)
+        frame_data_.input_sample_rate_hz = 16000;
+      else if (render_count_local % 83 == 0)
+        frame_data_.input_sample_rate_hz = 8000;
+
+      if (render_count_local == 0)
+        frame_data_.input_number_of_channels = 1;
+      else if (render_count_local % 4 == 0)
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+
+      if (render_count_local == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (render_count_local % 17 == 0)
+        frame_data_.output_sample_rate_hz = 32000;
+      else if (render_count_local % 19 == 0)
+        frame_data_.output_sample_rate_hz = 48000;
+      else if (render_count_local % 29 == 0)
+        frame_data_.output_sample_rate_hz = 16000;
+      else if (render_count_local % 61 == 0)
+        frame_data_.output_sample_rate_hz = 8000;
+
+      if (render_count_local == 0)
+        frame_data_.output_number_of_channels = 1;
+      else if (render_count_local % 8 == 0)
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+      break;
+    case RuntimeParameterSettingScheme::ExtremeStreamMetadataChangeScheme:
+      if (render_count_local == 0) {
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+      } else {
+        frame_data_.input_number_of_channels =
+            (frame_data_.input_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.input_sample_rate_hz == 8000)
+          frame_data_.input_sample_rate_hz = 16000;
+        else if (frame_data_.input_sample_rate_hz == 16000)
+          frame_data_.input_sample_rate_hz = 32000;
+        else if (frame_data_.input_sample_rate_hz == 32000)
+          frame_data_.input_sample_rate_hz = 48000;
+        else if (frame_data_.input_sample_rate_hz == 48000)
+          frame_data_.input_sample_rate_hz = 8000;
+
+        frame_data_.output_number_of_channels =
+            (frame_data_.output_number_of_channels == 1 ? 2 : 1);
+        if (frame_data_.output_sample_rate_hz == 8000)
+          frame_data_.output_sample_rate_hz = 16000;
+        else if (frame_data_.output_sample_rate_hz == 16000)
+          frame_data_.output_sample_rate_hz = 32000;
+        else if (frame_data_.output_sample_rate_hz == 32000)
+          frame_data_.output_sample_rate_hz = 48000;
+        else if (frame_data_.output_sample_rate_hz == 48000)
+          frame_data_.output_sample_rate_hz = 8000;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedMonoStreamMetadataScheme:
+      if (render_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 1;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 1;
+      }
+      break;
+    case RuntimeParameterSettingScheme::FixedStereoStreamMetadataScheme:
+      if (render_count_local == 0) {
+        frame_data_.input_sample_rate_hz = 16000;
+        frame_data_.input_number_of_channels = 2;
+        frame_data_.output_sample_rate_hz = 16000;
+        frame_data_.output_number_of_channels = 2;
+      }
+      break;
+    default:
+      FAIL();
+  }
+
+  // Restric the number of output channels not to exceed
+  // the number of input channels.
+  frame_data_.output_number_of_channels =
+      std::min(frame_data_.output_number_of_channels,
+               frame_data_.input_number_of_channels);
+}
+
+}  // namespace
+
+TEST_P(AudioProcessingImplLockTest, LockTest) {
+  // Run test and verify that it did not time out.
+  ASSERT_TRUE(RunTest());
+}
+
+// Instantiate tests from the extreme test configuration set.
+INSTANTIATE_TEST_SUITE_P(
+    DISABLED_AudioProcessingImplLockExtensive,
+    AudioProcessingImplLockTest,
+    ::testing::ValuesIn(TestConfig::GenerateExtensiveTestConfigs()));
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioProcessingImplLockBrief,
+    AudioProcessingImplLockTest,
+    ::testing::ValuesIn(TestConfig::GenerateBriefTestConfigs()));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_unittest.cc
new file mode 100644
index 0000000000..7c12a07ed9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -0,0 +1,1569 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/audio_processing_impl.h"
+
+#include <algorithm>
+#include <array>
+#include <memory>
+#include <tuple>
+
+#include "absl/types/optional.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/optionally_built_submodule_creators.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+#include "modules/audio_processing/test/echo_control_mock.h"
+#include "modules/audio_processing/test/test_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/random.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Invoke;
+using ::testing::NotNull;
+
+class MockInitialize : public AudioProcessingImpl {
+ public:
+  MockInitialize() : AudioProcessingImpl() {}
+
+  MOCK_METHOD(void, InitializeLocked, (), (override));
+  void RealInitializeLocked() {
+    AssertLockedForTest();
+    AudioProcessingImpl::InitializeLocked();
+  }
+
+  MOCK_METHOD(void, AddRef, (), (const, override));
+  MOCK_METHOD(rtc::RefCountReleaseStatus, Release, (), (const, override));
+};
+
+// Creates MockEchoControl instances and provides a raw pointer access to
+// the next created one. The raw pointer is meant to be used with gmock.
+// Returning a pointer of the next created MockEchoControl instance is necessary
+// for the following reasons: (i) gmock expectations must be set before any call
+// occurs, (ii) APM is initialized the first time that
+// AudioProcessingImpl::ProcessStream() is called and the initialization leads
+// to the creation of a new EchoControl object.
+class MockEchoControlFactory : public EchoControlFactory {
+ public:
+  MockEchoControlFactory() : next_mock_(std::make_unique<MockEchoControl>()) {}
+  // Returns a pointer to the next MockEchoControl that this factory creates.
+  MockEchoControl* GetNext() const { return next_mock_.get(); }
+  std::unique_ptr<EchoControl> Create(int sample_rate_hz,
+                                      int num_render_channels,
+                                      int num_capture_channels) override {
+    std::unique_ptr<EchoControl> mock = std::move(next_mock_);
+    next_mock_ = std::make_unique<MockEchoControl>();
+    return mock;
+  }
+
+ private:
+  std::unique_ptr<MockEchoControl> next_mock_;
+};
+
+// Mocks EchoDetector and records the first samples of the last analyzed render
+// stream frame. Used to check what data is read by an EchoDetector
+// implementation injected into an APM.
+class TestEchoDetector : public EchoDetector {
+ public:
+  TestEchoDetector()
+      : analyze_render_audio_called_(false),
+        last_render_audio_first_sample_(0.f) {}
+  ~TestEchoDetector() override = default;
+  void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) override {
+    last_render_audio_first_sample_ = render_audio[0];
+    analyze_render_audio_called_ = true;
+  }
+  void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override {
+  }
+  void Initialize(int capture_sample_rate_hz,
+                  int num_capture_channels,
+                  int render_sample_rate_hz,
+                  int num_render_channels) override {}
+  EchoDetector::Metrics GetMetrics() const override { return {}; }
+  // Returns true if AnalyzeRenderAudio() has been called at least once.
+  bool analyze_render_audio_called() const {
+    return analyze_render_audio_called_;
+  }
+  // Returns the first sample of the last analyzed render frame.
+  float last_render_audio_first_sample() const {
+    return last_render_audio_first_sample_;
+  }
+
+ private:
+  bool analyze_render_audio_called_;
+  float last_render_audio_first_sample_;
+};
+
+// Mocks CustomProcessing and applies ProcessSample() to all the samples.
+// Meant to be injected into an APM to modify samples in a known and detectable
+// way.
+class TestRenderPreProcessor : public CustomProcessing {
+ public:
+  TestRenderPreProcessor() = default;
+  ~TestRenderPreProcessor() = default;
+  void Initialize(int sample_rate_hz, int num_channels) override {}
+  void Process(AudioBuffer* audio) override {
+    for (size_t k = 0; k < audio->num_channels(); ++k) {
+      rtc::ArrayView<float> channel_view(audio->channels()[k],
+                                         audio->num_frames());
+      std::transform(channel_view.begin(), channel_view.end(),
+                     channel_view.begin(), ProcessSample);
+    }
+  }
+  std::string ToString() const override { return "TestRenderPreProcessor"; }
+  void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting) override {}
+  // Modifies a sample. This member is used in Process() to modify a frame and
+  // it is publicly visible to enable tests.
+  static constexpr float ProcessSample(float x) { return 2.f * x; }
+};
+
+// Runs `apm` input processing for volume adjustments for `num_frames` random
+// frames starting from the volume `initial_volume`. This includes three steps:
+// 1) Set the input volume 2) Process the stream 3) Set the new recommended
+// input volume. Returns the new recommended input volume.
+int ProcessInputVolume(AudioProcessing& apm,
+                       int num_frames,
+                       int initial_volume) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  std::array<float, kSampleRateHz / 100> buffer;
+  float* channel_pointers[] = {buffer.data()};
+  StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
+                             /*num_channels=*/kNumChannels);
+  int recommended_input_volume = initial_volume;
+  for (int i = 0; i < num_frames; ++i) {
+    Random random_generator(2341U);
+    RandomizeSampleVector(&random_generator, buffer);
+
+    apm.set_stream_analog_level(recommended_input_volume);
+    apm.ProcessStream(channel_pointers, stream_config, stream_config,
+                      channel_pointers);
+    recommended_input_volume = apm.recommended_stream_analog_level();
+  }
+  return recommended_input_volume;
+}
+
+}  // namespace
+
+TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
+  MockInitialize mock;
+  ON_CALL(mock, InitializeLocked)
+      .WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked));
+
+  EXPECT_CALL(mock, InitializeLocked).Times(1);
+  mock.Initialize();
+
+  constexpr size_t kMaxSampleRateHz = 32000;
+  constexpr size_t kMaxNumChannels = 2;
+  std::array<int16_t, kMaxNumChannels * kMaxSampleRateHz / 100> frame;
+  frame.fill(0);
+  StreamConfig config(16000, 1);
+  // Call with the default parameters; there should be an init.
+  EXPECT_CALL(mock, InitializeLocked).Times(0);
+  EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data()));
+  EXPECT_NOERR(
+      mock.ProcessReverseStream(frame.data(), config, config, frame.data()));
+
+  // New sample rate. (Only impacts ProcessStream).
+  config = StreamConfig(32000, 1);
+  EXPECT_CALL(mock, InitializeLocked).Times(1);
+  EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data()));
+
+  // New number of channels.
+  config = StreamConfig(32000, 2);
+  EXPECT_CALL(mock, InitializeLocked).Times(2);
+  EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data()));
+  EXPECT_NOERR(
+      mock.ProcessReverseStream(frame.data(), config, config, frame.data()));
+
+  // A new sample rate passed to ProcessReverseStream should cause an init.
+  config = StreamConfig(16000, 2);
+  EXPECT_CALL(mock, InitializeLocked).Times(1);
+  EXPECT_NOERR(
+      mock.ProcessReverseStream(frame.data(), config, config, frame.data()));
+}
+
+TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) {
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting().Create();
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.pre_amplifier.enabled = true;
+  apm_config.pre_amplifier.fixed_gain_factor = 1.f;
+  apm->ApplyConfig(apm_config);
+
+  constexpr int kSampleRateHz = 48000;
+  constexpr int16_t kAudioLevel = 10000;
+  constexpr size_t kNumChannels = 2;
+
+  std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
+  StreamConfig config(kSampleRateHz, kNumChannels);
+  frame.fill(kAudioLevel);
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+  EXPECT_EQ(frame[100], kAudioLevel)
+      << "With factor 1, frame shouldn't be modified.";
+
+  constexpr float kGainFactor = 2.f;
+  apm->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCapturePreGain(kGainFactor));
+
+  // Process for two frames to have time to ramp up gain.
+  for (int i = 0; i < 2; ++i) {
+    frame.fill(kAudioLevel);
+    apm->ProcessStream(frame.data(), config, config, frame.data());
+  }
+  EXPECT_EQ(frame[100], kGainFactor * kAudioLevel)
+      << "Frame should be amplified.";
+}
+
+TEST(AudioProcessingImplTest,
+     LevelAdjustmentUpdateCapturePreGainRuntimeSetting) {
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting().Create();
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.capture_level_adjustment.enabled = true;
+  apm_config.capture_level_adjustment.pre_gain_factor = 1.f;
+  apm->ApplyConfig(apm_config);
+
+  constexpr int kSampleRateHz = 48000;
+  constexpr int16_t kAudioLevel = 10000;
+  constexpr size_t kNumChannels = 2;
+
+  std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
+  StreamConfig config(kSampleRateHz, kNumChannels);
+  frame.fill(kAudioLevel);
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+  EXPECT_EQ(frame[100], kAudioLevel)
+      << "With factor 1, frame shouldn't be modified.";
+
+  constexpr float kGainFactor = 2.f;
+  apm->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCapturePreGain(kGainFactor));
+
+  // Process for two frames to have time to ramp up gain.
+  for (int i = 0; i < 2; ++i) {
+    frame.fill(kAudioLevel);
+    apm->ProcessStream(frame.data(), config, config, frame.data());
+  }
+  EXPECT_EQ(frame[100], kGainFactor * kAudioLevel)
+      << "Frame should be amplified.";
+}
+
+TEST(AudioProcessingImplTest,
+     LevelAdjustmentUpdateCapturePostGainRuntimeSetting) {
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting().Create();
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.capture_level_adjustment.enabled = true;
+  apm_config.capture_level_adjustment.post_gain_factor = 1.f;
+  apm->ApplyConfig(apm_config);
+
+  constexpr int kSampleRateHz = 48000;
+  constexpr int16_t kAudioLevel = 10000;
+  constexpr size_t kNumChannels = 2;
+
+  std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
+  StreamConfig config(kSampleRateHz, kNumChannels);
+  frame.fill(kAudioLevel);
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+  EXPECT_EQ(frame[100], kAudioLevel)
+      << "With factor 1, frame shouldn't be modified.";
+
+  constexpr float kGainFactor = 2.f;
+  apm->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCapturePostGain(kGainFactor));
+
+  // Process for two frames to have time to ramp up gain.
+  for (int i = 0; i < 2; ++i) {
+    frame.fill(kAudioLevel);
+    apm->ProcessStream(frame.data(), config, config, frame.data());
+  }
+  EXPECT_EQ(frame[100], kGainFactor * kAudioLevel)
+      << "Frame should be amplified.";
+}
+
+TEST(AudioProcessingImplTest, EchoControllerObservesSetCaptureUsageChange) {
+  // Tests that the echo controller observes that the capture usage has been
+  // updated.
+  auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
+  const MockEchoControlFactory* echo_control_factory_ptr =
+      echo_control_factory.get();
+
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoControlFactory(std::move(echo_control_factory))
+          .Create();
+
+  constexpr int16_t kAudioLevel = 10000;
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 2;
+  std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
+  StreamConfig config(kSampleRateHz, kNumChannels);
+  frame.fill(kAudioLevel);
+
+  MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
+
+  // Ensure that SetCaptureOutputUsage is not called when no runtime settings
+  // are passed.
+  EXPECT_CALL(*echo_control_mock, SetCaptureOutputUsage(testing::_)).Times(0);
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+
+  // Ensure that SetCaptureOutputUsage is called with the right information when
+  // a runtime setting is passed.
+  EXPECT_CALL(*echo_control_mock,
+              SetCaptureOutputUsage(/*capture_output_used=*/false))
+      .Times(1);
+  EXPECT_TRUE(apm->PostRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
+          /*capture_output_used=*/false)));
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+
+  EXPECT_CALL(*echo_control_mock,
+              SetCaptureOutputUsage(/*capture_output_used=*/true))
+      .Times(1);
+  EXPECT_TRUE(apm->PostRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
+          /*capture_output_used=*/true)));
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+
+  // The number of positions to place items in the queue is equal to the queue
+  // size minus 1.
+  constexpr int kNumSlotsInQueue = RuntimeSettingQueueSize();
+
+  // Ensure that SetCaptureOutputUsage is called with the right information when
+  // many runtime settings are passed.
+  for (int k = 0; k < kNumSlotsInQueue - 1; ++k) {
+    EXPECT_TRUE(apm->PostRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
+            /*capture_output_used=*/false)));
+  }
+  EXPECT_CALL(*echo_control_mock,
+              SetCaptureOutputUsage(/*capture_output_used=*/false))
+      .Times(kNumSlotsInQueue - 1);
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+
+  // Ensure that SetCaptureOutputUsage is properly called with the fallback
+  // value when the runtime settings queue becomes full.
+  for (int k = 0; k < kNumSlotsInQueue; ++k) {
+    EXPECT_TRUE(apm->PostRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
+            /*capture_output_used=*/false)));
+  }
+  EXPECT_FALSE(apm->PostRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
+          /*capture_output_used=*/false)));
+  EXPECT_FALSE(apm->PostRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
+          /*capture_output_used=*/false)));
+  EXPECT_CALL(*echo_control_mock,
+              SetCaptureOutputUsage(/*capture_output_used=*/false))
+      .Times(kNumSlotsInQueue);
+  EXPECT_CALL(*echo_control_mock,
+              SetCaptureOutputUsage(/*capture_output_used=*/true))
+      .Times(1);
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+}
+
+TEST(AudioProcessingImplTest,
+     EchoControllerObservesPreAmplifierEchoPathGainChange) {
+  // Tests that the echo controller observes an echo path gain change when the
+  // pre-amplifier submodule changes the gain.
+  auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
+  const auto* echo_control_factory_ptr = echo_control_factory.get();
+
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoControlFactory(std::move(echo_control_factory))
+          .Create();
+  // Disable AGC.
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.gain_controller1.enabled = false;
+  apm_config.gain_controller2.enabled = false;
+  apm_config.pre_amplifier.enabled = true;
+  apm_config.pre_amplifier.fixed_gain_factor = 1.f;
+  apm->ApplyConfig(apm_config);
+
+  constexpr int16_t kAudioLevel = 10000;
+  constexpr size_t kSampleRateHz = 48000;
+  constexpr size_t kNumChannels = 2;
+  std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
+  StreamConfig config(kSampleRateHz, kNumChannels);
+  frame.fill(kAudioLevel);
+
+  MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
+
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
+      .Times(1);
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
+      .Times(1);
+  apm->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f));
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+}
+
+TEST(AudioProcessingImplTest,
+     EchoControllerObservesLevelAdjustmentPreGainEchoPathGainChange) {
+  // Tests that the echo controller observes an echo path gain change when the
+  // pre-amplifier submodule changes the gain.
+  auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
+  const auto* echo_control_factory_ptr = echo_control_factory.get();
+
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoControlFactory(std::move(echo_control_factory))
+          .Create();
+  // Disable AGC.
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.gain_controller1.enabled = false;
+  apm_config.gain_controller2.enabled = false;
+  apm_config.capture_level_adjustment.enabled = true;
+  apm_config.capture_level_adjustment.pre_gain_factor = 1.f;
+  apm->ApplyConfig(apm_config);
+
+  constexpr int16_t kAudioLevel = 10000;
+  constexpr size_t kSampleRateHz = 48000;
+  constexpr size_t kNumChannels = 2;
+  std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
+  StreamConfig config(kSampleRateHz, kNumChannels);
+  frame.fill(kAudioLevel);
+
+  MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
+
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
+      .Times(1);
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
+      .Times(1);
+  apm->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f));
+  apm->ProcessStream(frame.data(), config, config, frame.data());
+}
+
+TEST(AudioProcessingImplTest,
+     EchoControllerObservesAnalogAgc1EchoPathGainChange) {
+  // Tests that the echo controller observes an echo path gain change when the
+  // AGC1 analog adaptive submodule changes the analog gain.
+  auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
+  const auto* echo_control_factory_ptr = echo_control_factory.get();
+
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoControlFactory(std::move(echo_control_factory))
+          .Create();
+  webrtc::AudioProcessing::Config apm_config;
+  // Enable AGC1.
+  apm_config.gain_controller1.enabled = true;
+  apm_config.gain_controller1.analog_gain_controller.enabled = true;
+  apm_config.gain_controller2.enabled = false;
+  apm_config.pre_amplifier.enabled = false;
+  apm->ApplyConfig(apm_config);
+
+  constexpr int16_t kAudioLevel = 1000;
+  constexpr size_t kSampleRateHz = 48000;
+  constexpr size_t kNumChannels = 2;
+  std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
+  StreamConfig stream_config(kSampleRateHz, kNumChannels);
+  frame.fill(kAudioLevel);
+
+  MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
+
+  constexpr int kInitialStreamAnalogLevel = 123;
+  apm->set_stream_analog_level(kInitialStreamAnalogLevel);
+
+  // When the first fame is processed, no echo path gain change must be
+  // detected.
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
+      .Times(1);
+  apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
+
+  // Simulate the application of the recommended analog level.
+  int recommended_analog_level = apm->recommended_stream_analog_level();
+  if (recommended_analog_level == kInitialStreamAnalogLevel) {
+    // Force an analog gain change if it did not happen.
+    recommended_analog_level++;
+  }
+  apm->set_stream_analog_level(recommended_analog_level);
+
+  // After the first fame and with a stream analog level change, the echo path
+  // gain change must be detected.
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
+      .Times(1);
+  apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
+}
+
+// Tests that a stream is successfully processed when AGC2 adaptive digital is
+// used and when the field trial
+// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-Default/` is set.
+TEST(AudioProcessingImplTest,
+     ProcessWithAgc2AndTransientSuppressorVadModeDefault) {
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Disabled/");
+  auto apm = AudioProcessingBuilder()
+                 .SetConfig({.gain_controller1{.enabled = false}})
+                 .Create();
+  ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.gain_controller1.enabled = false;
+  apm_config.gain_controller2.enabled = true;
+  apm_config.gain_controller2.adaptive_digital.enabled = true;
+  apm_config.transient_suppression.enabled = true;
+  apm->ApplyConfig(apm_config);
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  std::array<float, kSampleRateHz / 100> buffer;
+  float* channel_pointers[] = {buffer.data()};
+  StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
+                             /*num_channels=*/kNumChannels);
+  Random random_generator(2341U);
+  constexpr int kFramesToProcess = 10;
+  for (int i = 0; i < kFramesToProcess; ++i) {
+    RandomizeSampleVector(&random_generator, buffer);
+    ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+  }
+}
+
+// Tests that a stream is successfully processed when AGC2 adaptive digital is
+// used and when the field trial
+// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad/` is set.
+TEST(AudioProcessingImplTest,
+     ProcessWithAgc2AndTransientSuppressorVadModeRnnVad) {
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
+  rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
+  ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.gain_controller1.enabled = false;
+  apm_config.gain_controller2.enabled = true;
+  apm_config.gain_controller2.adaptive_digital.enabled = true;
+  apm_config.transient_suppression.enabled = true;
+  apm->ApplyConfig(apm_config);
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  std::array<float, kSampleRateHz / 100> buffer;
+  float* channel_pointers[] = {buffer.data()};
+  StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
+                             /*num_channels=*/kNumChannels);
+  Random random_generator(2341U);
+  constexpr int kFramesToProcess = 10;
+  for (int i = 0; i < kFramesToProcess; ++i) {
+    RandomizeSampleVector(&random_generator, buffer);
+    ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+  }
+}
+
+TEST(AudioProcessingImplTest, EchoControllerObservesPlayoutVolumeChange) {
+  // Tests that the echo controller observes an echo path gain change when a
+  // playout volume change is reported.
+  auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
+  const auto* echo_control_factory_ptr = echo_control_factory.get();
+
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoControlFactory(std::move(echo_control_factory))
+          .Create();
+  // Disable AGC.
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.gain_controller1.enabled = false;
+  apm_config.gain_controller2.enabled = false;
+  apm->ApplyConfig(apm_config);
+
+  constexpr int16_t kAudioLevel = 10000;
+  constexpr size_t kSampleRateHz = 48000;
+  constexpr size_t kNumChannels = 2;
+  std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
+  StreamConfig stream_config(kSampleRateHz, kNumChannels);
+  frame.fill(kAudioLevel);
+
+  MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
+
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
+      .Times(1);
+  apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
+
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
+      .Times(1);
+  apm->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50));
+  apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
+
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
+      .Times(1);
+  apm->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50));
+  apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
+
+  EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
+  EXPECT_CALL(*echo_control_mock,
+              ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
+      .Times(1);
+  apm->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(100));
+  apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
+}
+
+TEST(AudioProcessingImplTest, RenderPreProcessorBeforeEchoDetector) {
+  // Make sure that signal changes caused by a render pre-processing sub-module
+  // take place before any echo detector analysis.
+  auto test_echo_detector = rtc::make_ref_counted<TestEchoDetector>();
+  std::unique_ptr<CustomProcessing> test_render_pre_processor(
+      new TestRenderPreProcessor());
+  // Create APM injecting the test echo detector and render pre-processor.
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoDetector(test_echo_detector)
+          .SetRenderPreProcessing(std::move(test_render_pre_processor))
+          .Create();
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.pre_amplifier.enabled = true;
+  apm->ApplyConfig(apm_config);
+
+  constexpr int16_t kAudioLevel = 1000;
+  constexpr int kSampleRateHz = 16000;
+  constexpr size_t kNumChannels = 1;
+  // Explicitly initialize APM to ensure no render frames are discarded.
+  const ProcessingConfig processing_config = {{
+      {kSampleRateHz, kNumChannels},
+      {kSampleRateHz, kNumChannels},
+      {kSampleRateHz, kNumChannels},
+      {kSampleRateHz, kNumChannels},
+  }};
+  apm->Initialize(processing_config);
+
+  std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
+  StreamConfig stream_config(kSampleRateHz, kNumChannels);
+
+  constexpr float kAudioLevelFloat = static_cast<float>(kAudioLevel);
+  constexpr float kExpectedPreprocessedAudioLevel =
+      TestRenderPreProcessor::ProcessSample(kAudioLevelFloat);
+  ASSERT_NE(kAudioLevelFloat, kExpectedPreprocessedAudioLevel);
+
+  // Analyze a render stream frame.
+  frame.fill(kAudioLevel);
+  ASSERT_EQ(AudioProcessing::Error::kNoError,
+            apm->ProcessReverseStream(frame.data(), stream_config,
+                                      stream_config, frame.data()));
+  // Trigger a call to in EchoDetector::AnalyzeRenderAudio() via
+  // ProcessStream().
+  frame.fill(kAudioLevel);
+  ASSERT_EQ(AudioProcessing::Error::kNoError,
+            apm->ProcessStream(frame.data(), stream_config, stream_config,
+                               frame.data()));
+  // Regardless of how the call to in EchoDetector::AnalyzeRenderAudio() is
+  // triggered, the line below checks that the call has occurred. If not, the
+  // APM implementation may have changed and this test might need to be adapted.
+  ASSERT_TRUE(test_echo_detector->analyze_render_audio_called());
+  // Check that the data read in EchoDetector::AnalyzeRenderAudio() is that
+  // produced by the render pre-processor.
+  EXPECT_EQ(kExpectedPreprocessedAudioLevel,
+            test_echo_detector->last_render_audio_first_sample());
+}
+
+// Disabling build-optional submodules and trying to enable them via the APM
+// config should be bit-exact with running APM with said submodules disabled.
+// This mainly tests that SetCreateOptionalSubmodulesForTesting has an effect.
+TEST(ApmWithSubmodulesExcludedTest, BitexactWithDisabledModules) {
+  auto apm = rtc::make_ref_counted<AudioProcessingImpl>();
+  ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
+
+  ApmSubmoduleCreationOverrides overrides;
+  overrides.transient_suppression = true;
+  apm->OverrideSubmoduleCreationForTesting(overrides);
+
+  AudioProcessing::Config apm_config = apm->GetConfig();
+  apm_config.transient_suppression.enabled = true;
+  apm->ApplyConfig(apm_config);
+
+  rtc::scoped_refptr<AudioProcessing> apm_reference =
+      AudioProcessingBuilder().Create();
+  apm_config = apm_reference->GetConfig();
+  apm_config.transient_suppression.enabled = false;
+  apm_reference->ApplyConfig(apm_config);
+
+  constexpr int kSampleRateHz = 16000;
+  constexpr int kNumChannels = 1;
+  std::array<float, kSampleRateHz / 100> buffer;
+  std::array<float, kSampleRateHz / 100> buffer_reference;
+  float* channel_pointers[] = {buffer.data()};
+  float* channel_pointers_reference[] = {buffer_reference.data()};
+  StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
+                             /*num_channels=*/kNumChannels);
+  Random random_generator(2341U);
+  constexpr int kFramesToProcessPerConfiguration = 10;
+
+  for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
+    RandomizeSampleVector(&random_generator, buffer);
+    std::copy(buffer.begin(), buffer.end(), buffer_reference.begin());
+    ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+    ASSERT_EQ(
+        apm_reference->ProcessStream(channel_pointers_reference, stream_config,
+                                     stream_config, channel_pointers_reference),
+        kNoErr);
+    for (int j = 0; j < kSampleRateHz / 100; ++j) {
+      EXPECT_EQ(buffer[j], buffer_reference[j]);
+    }
+  }
+}
+
+// Disable transient suppressor creation and run APM in ways that should trigger
+// calls to the transient suppressor API.
+TEST(ApmWithSubmodulesExcludedTest, ReinitializeTransientSuppressor) {
+  auto apm = rtc::make_ref_counted<AudioProcessingImpl>();
+  ASSERT_EQ(apm->Initialize(), kNoErr);
+
+  ApmSubmoduleCreationOverrides overrides;
+  overrides.transient_suppression = true;
+  apm->OverrideSubmoduleCreationForTesting(overrides);
+
+  AudioProcessing::Config config = apm->GetConfig();
+  config.transient_suppression.enabled = true;
+  apm->ApplyConfig(config);
+  // 960 samples per frame: 10 ms of <= 48 kHz audio with <= 2 channels.
+  float buffer[960];
+  float* channel_pointers[] = {&buffer[0], &buffer[480]};
+  Random random_generator(2341U);
+  constexpr int kFramesToProcessPerConfiguration = 3;
+
+  StreamConfig initial_stream_config(/*sample_rate_hz=*/16000,
+                                     /*num_channels=*/1);
+  for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
+    RandomizeSampleVector(&random_generator, buffer);
+    EXPECT_EQ(apm->ProcessStream(channel_pointers, initial_stream_config,
+                                 initial_stream_config, channel_pointers),
+              kNoErr);
+  }
+
+  StreamConfig stereo_stream_config(/*sample_rate_hz=*/16000,
+                                    /*num_channels=*/2);
+  for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
+    RandomizeSampleVector(&random_generator, buffer);
+    EXPECT_EQ(apm->ProcessStream(channel_pointers, stereo_stream_config,
+                                 stereo_stream_config, channel_pointers),
+              kNoErr);
+  }
+
+  StreamConfig high_sample_rate_stream_config(/*sample_rate_hz=*/48000,
+                                              /*num_channels=*/2);
+  for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
+    RandomizeSampleVector(&random_generator, buffer);
+    EXPECT_EQ(
+        apm->ProcessStream(channel_pointers, high_sample_rate_stream_config,
+                           high_sample_rate_stream_config, channel_pointers),
+        kNoErr);
+  }
+}
+
+// Disable transient suppressor creation and run APM in ways that should trigger
+// calls to the transient suppressor API.
+TEST(ApmWithSubmodulesExcludedTest, ToggleTransientSuppressor) {
+  auto apm = rtc::make_ref_counted<AudioProcessingImpl>();
+  ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
+
+  ApmSubmoduleCreationOverrides overrides;
+  overrides.transient_suppression = true;
+  apm->OverrideSubmoduleCreationForTesting(overrides);
+
+  //  960 samples per frame: 10 ms of <= 48 kHz audio with <= 2 channels.
+  float buffer[960];
+  float* channel_pointers[] = {&buffer[0], &buffer[480]};
+  Random random_generator(2341U);
+  constexpr int kFramesToProcessPerConfiguration = 3;
+  StreamConfig stream_config(/*sample_rate_hz=*/16000,
+                             /*num_channels=*/1);
+
+  AudioProcessing::Config config = apm->GetConfig();
+  config.transient_suppression.enabled = true;
+  apm->ApplyConfig(config);
+  for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
+    RandomizeSampleVector(&random_generator, buffer);
+    EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+  }
+
+  config = apm->GetConfig();
+  config.transient_suppression.enabled = false;
+  apm->ApplyConfig(config);
+  for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
+    RandomizeSampleVector(&random_generator, buffer);
+    EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+  }
+
+  config = apm->GetConfig();
+  config.transient_suppression.enabled = true;
+  apm->ApplyConfig(config);
+  for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
+    RandomizeSampleVector(&random_generator, buffer);
+    EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+  }
+}
+
+class StartupInputVolumeParameterizedTest
+    : public ::testing::TestWithParam<int> {};
+
+// Tests that, when no input volume controller is used, the startup input volume
+// is never modified.
+TEST_P(StartupInputVolumeParameterizedTest,
+       WithNoInputVolumeControllerStartupVolumeNotModified) {
+  webrtc::AudioProcessing::Config config;
+  config.gain_controller1.enabled = false;
+  config.gain_controller2.enabled = false;
+  auto apm = AudioProcessingBuilder().SetConfig(config).Create();
+
+  int startup_volume = GetParam();
+  int recommended_volume = ProcessInputVolume(
+      *apm, /*num_frames=*/1, /*initial_volume=*/startup_volume);
+  EXPECT_EQ(recommended_volume, startup_volume);
+}
+
+INSTANTIATE_TEST_SUITE_P(AudioProcessingImplTest,
+                         StartupInputVolumeParameterizedTest,
+                         ::testing::Values(0, 5, 15, 50, 100));
+
+// Tests that, when no input volume controller is used, the recommended input
+// volume always matches the applied one.
+TEST(AudioProcessingImplTest,
+     WithNoInputVolumeControllerAppliedAndRecommendedVolumesMatch) {
+  webrtc::AudioProcessing::Config config;
+  config.gain_controller1.enabled = false;
+  config.gain_controller2.enabled = false;
+  auto apm = AudioProcessingBuilder().SetConfig(config).Create();
+
+  Random rand_gen(42);
+  for (int i = 0; i < 32; ++i) {
+    SCOPED_TRACE(i);
+    int32_t applied_volume = rand_gen.Rand(/*low=*/0, /*high=*/255);
+    int recommended_volume =
+        ProcessInputVolume(*apm, /*num_frames=*/1, applied_volume);
+    EXPECT_EQ(recommended_volume, applied_volume);
+  }
+}
+
+class ApmInputVolumeControllerParametrizedTest
+    : public ::testing::TestWithParam<
+          std::tuple<int, int, AudioProcessing::Config>> {
+ protected:
+  ApmInputVolumeControllerParametrizedTest()
+      : sample_rate_hz_(std::get<0>(GetParam())),
+        num_channels_(std::get<1>(GetParam())),
+        channels_(num_channels_),
+        channel_pointers_(num_channels_) {
+    const int frame_size = sample_rate_hz_ / 100;
+    for (int c = 0; c < num_channels_; ++c) {
+      channels_[c].resize(frame_size);
+      channel_pointers_[c] = channels_[c].data();
+      std::fill(channels_[c].begin(), channels_[c].end(), 0.0f);
+    }
+  }
+
+  int sample_rate_hz() const { return sample_rate_hz_; }
+  int num_channels() const { return num_channels_; }
+  AudioProcessing::Config GetConfig() const { return std::get<2>(GetParam()); }
+
+  float* const* channel_pointers() { return channel_pointers_.data(); }
+
+ private:
+  const int sample_rate_hz_;
+  const int num_channels_;
+  std::vector<std::vector<float>> channels_;
+  std::vector<float*> channel_pointers_;
+};
+
+TEST_P(ApmInputVolumeControllerParametrizedTest,
+       EnforceMinInputVolumeAtStartupWithZeroVolume) {
+  const StreamConfig stream_config(sample_rate_hz(), num_channels());
+  auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
+
+  apm->set_stream_analog_level(0);
+  apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+                     channel_pointers());
+  EXPECT_GT(apm->recommended_stream_analog_level(), 0);
+}
+
+TEST_P(ApmInputVolumeControllerParametrizedTest,
+       EnforceMinInputVolumeAtStartupWithNonZeroVolume) {
+  const StreamConfig stream_config(sample_rate_hz(), num_channels());
+  auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
+
+  constexpr int kStartupVolume = 3;
+  apm->set_stream_analog_level(kStartupVolume);
+  apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+                     channel_pointers());
+  EXPECT_GT(apm->recommended_stream_analog_level(), kStartupVolume);
+}
+
+TEST_P(ApmInputVolumeControllerParametrizedTest,
+       EnforceMinInputVolumeAfterManualVolumeAdjustment) {
+  const auto config = GetConfig();
+  if (config.gain_controller1.enabled) {
+    // After a downward manual adjustment, AGC1 slowly converges to the minimum
+    // input volume.
+    GTEST_SKIP() << "Does not apply to AGC1";
+  }
+  const StreamConfig stream_config(sample_rate_hz(), num_channels());
+  auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
+
+  apm->set_stream_analog_level(20);
+  apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+                     channel_pointers());
+  constexpr int kManuallyAdjustedVolume = 3;
+  apm->set_stream_analog_level(kManuallyAdjustedVolume);
+  apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+                     channel_pointers());
+  EXPECT_GT(apm->recommended_stream_analog_level(), kManuallyAdjustedVolume);
+}
+
+TEST_P(ApmInputVolumeControllerParametrizedTest,
+       DoNotEnforceMinInputVolumeAtStartupWithHighVolume) {
+  const StreamConfig stream_config(sample_rate_hz(), num_channels());
+  auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
+
+  constexpr int kStartupVolume = 200;
+  apm->set_stream_analog_level(kStartupVolume);
+  apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+                     channel_pointers());
+  EXPECT_EQ(apm->recommended_stream_analog_level(), kStartupVolume);
+}
+
+TEST_P(ApmInputVolumeControllerParametrizedTest,
+       DoNotEnforceMinInputVolumeAfterManualVolumeAdjustmentToZero) {
+  const StreamConfig stream_config(sample_rate_hz(), num_channels());
+  auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
+
+  apm->set_stream_analog_level(100);
+  apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+                     channel_pointers());
+  apm->set_stream_analog_level(0);
+  apm->ProcessStream(channel_pointers(), stream_config, stream_config,
+                     channel_pointers());
+  EXPECT_EQ(apm->recommended_stream_analog_level(), 0);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioProcessingImplTest,
+    ApmInputVolumeControllerParametrizedTest,
+    ::testing::Combine(
+        ::testing::Values(8000, 16000, 32000, 48000),  // Sample rates.
+        ::testing::Values(1, 2),                       // Number of channels.
+        ::testing::Values(
+            // Full AGC1.
+            AudioProcessing::Config{
+                .gain_controller1 = {.enabled = true,
+                                     .analog_gain_controller =
+                                         {.enabled = true,
+                                          .enable_digital_adaptive = true}},
+                .gain_controller2 = {.enabled = false}},
+            // Hybrid AGC.
+            AudioProcessing::Config{
+                .gain_controller1 = {.enabled = true,
+                                     .analog_gain_controller =
+                                         {.enabled = true,
+                                          .enable_digital_adaptive = false}},
+                .gain_controller2 = {.enabled = true,
+                                     .adaptive_digital = {.enabled = true}}})));
+
+// When the input volume is not emulated and no input volume controller is
+// active, the recommended volume must always be the applied volume.
+TEST(AudioProcessingImplTest,
+     RecommendAppliedInputVolumeWithNoAgcWithNoEmulation) {
+  auto apm = AudioProcessingBuilder()
+                 .SetConfig({.capture_level_adjustment = {.enabled = false},
+                             .gain_controller1 = {.enabled = false}})
+                 .Create();
+
+  constexpr int kOneFrame = 1;
+  EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/123), 123);
+  EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/59), 59);
+  EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
+}
+
+// When the input volume is emulated, the recommended volume must always be the
+// applied volume and at any time it must not be that set in the input volume
+// emulator.
+// TODO(bugs.webrtc.org/14581): Enable when APM fixed to let this test pass.
+TEST(AudioProcessingImplTest,
+     DISABLED_RecommendAppliedInputVolumeWithNoAgcWithEmulation) {
+  auto apm =
+      AudioProcessingBuilder()
+          .SetConfig({.capture_level_adjustment = {.enabled = true,
+                                                   .analog_mic_gain_emulation{
+                                                       .enabled = true,
+                                                       .initial_level = 255}},
+                      .gain_controller1 = {.enabled = false}})
+          .Create();
+
+  constexpr int kOneFrame = 1;
+  EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/123), 123);
+  EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/59), 59);
+  EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
+}
+
+// Even if there is an enabled input volume controller, when the input volume is
+// emulated, the recommended volume is always the applied volume because the
+// active controller must only adjust the internally emulated volume and leave
+// the externally applied volume unchanged.
+// TODO(bugs.webrtc.org/14581): Enable when APM fixed to let this test pass.
+TEST(AudioProcessingImplTest,
+     DISABLED_RecommendAppliedInputVolumeWithAgcWithEmulation) {
+  auto apm =
+      AudioProcessingBuilder()
+          .SetConfig({.capture_level_adjustment = {.enabled = true,
+                                                   .analog_mic_gain_emulation{
+                                                       .enabled = true}},
+                      .gain_controller1 = {.enabled = true,
+                                           .analog_gain_controller{
+                                               .enabled = true,
+                                           }}})
+          .Create();
+
+  constexpr int kOneFrame = 1;
+  EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/123), 123);
+  EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/59), 59);
+  EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
+}
+
+TEST(AudioProcessingImplTest,
+     Agc2FieldTrialDoNotSwitchToFullAgc2WhenNoAgcIsActive) {
+  constexpr AudioProcessing::Config kOriginal{
+      .gain_controller1{.enabled = false},
+      .gain_controller2{.enabled = false},
+  };
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(kOriginal);
+  adjusted = apm->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
+}
+
+TEST(AudioProcessingImplTest,
+     Agc2FieldTrialDoNotSwitchToFullAgc2WithAgc1Agc2InputVolumeControllers) {
+  constexpr AudioProcessing::Config kOriginal{
+      .gain_controller1{.enabled = true,
+                        .analog_gain_controller{.enabled = true}},
+      .gain_controller2{.enabled = true,
+                        .input_volume_controller{.enabled = true}},
+  };
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(kOriginal);
+  adjusted = apm->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
+}
+
+class Agc2FieldTrialParametrizedTest
+    : public ::testing::TestWithParam<AudioProcessing::Config> {};
+
+TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfDisabled) {
+  const AudioProcessing::Config original = GetParam();
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Disabled/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(original);
+  adjusted = apm->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfNoOverride) {
+  const AudioProcessing::Config original = GetParam();
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,"
+      "switch_to_agc2:false,"
+      "disallow_transient_suppressor_usage:false/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(original);
+  adjusted = apm->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest, DoNotSwitchToFullAgc2) {
+  const AudioProcessing::Config original = GetParam();
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:false/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(original);
+  adjusted = apm->GetConfig();
+  EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
+  EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest, SwitchToFullAgc2) {
+  const AudioProcessing::Config original = GetParam();
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+  EXPECT_FALSE(adjusted.gain_controller1.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(original);
+  adjusted = apm->GetConfig();
+  EXPECT_FALSE(adjusted.gain_controller1.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest,
+       SwitchToFullAgc2AndOverrideInputVolumeControllerParameters) {
+  const AudioProcessing::Config original = GetParam();
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true,"
+      "min_input_volume:123,"
+      "clipped_level_min:20,"
+      "clipped_level_step:30,"
+      "clipped_ratio_threshold:0.4,"
+      "clipped_wait_frames:50,"
+      "enable_clipping_predictor:true,"
+      "target_range_max_dbfs:-6,"
+      "target_range_min_dbfs:-70,"
+      "update_input_volume_wait_frames:80,"
+      "speech_probability_threshold:0.9,"
+      "speech_ratio_threshold:1.0/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+  EXPECT_FALSE(adjusted.gain_controller1.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(original);
+  adjusted = apm->GetConfig();
+  EXPECT_FALSE(adjusted.gain_controller1.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest,
+       SwitchToFullAgc2AndOverrideAdaptiveDigitalControllerParameters) {
+  const AudioProcessing::Config original = GetParam();
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true,"
+      "headroom_db:10,"
+      "max_gain_db:20,"
+      "initial_gain_db:7,"
+      "max_gain_change_db_per_second:5,"
+      "max_output_noise_level_dbfs:-40/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
+  EXPECT_FALSE(adjusted.gain_controller1.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+  ASSERT_NE(adjusted.gain_controller2.adaptive_digital,
+            original.gain_controller2.adaptive_digital);
+  EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10);
+  EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20);
+  EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7);
+  EXPECT_EQ(
+      adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second,
+      5);
+  EXPECT_EQ(
+      adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs,
+      -40);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(original);
+  adjusted = apm->GetConfig();
+  EXPECT_FALSE(adjusted.gain_controller1.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
+  EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
+  ASSERT_NE(adjusted.gain_controller2.adaptive_digital,
+            original.gain_controller2.adaptive_digital);
+  EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10);
+  EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20);
+  EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7);
+  EXPECT_EQ(
+      adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second,
+      5);
+  EXPECT_EQ(
+      adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs,
+      -40);
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithTs) {
+  AudioProcessing::Config config = GetParam();
+  if (!config.transient_suppression.enabled) {
+    GTEST_SKIP() << "TS is disabled, skip.";
+  }
+
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Disabled/");
+  auto apm = AudioProcessingBuilder().SetConfig(config).Create();
+
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  std::array<float, kSampleRateHz / 100> buffer;
+  float* channel_pointers[] = {buffer.data()};
+  StreamConfig stream_config(kSampleRateHz, kNumChannels);
+  Random random_generator(2341U);
+  constexpr int kFramesToProcess = 10;
+  int volume = 100;
+  for (int i = 0; i < kFramesToProcess; ++i) {
+    SCOPED_TRACE(i);
+    RandomizeSampleVector(&random_generator, buffer);
+    apm->set_stream_analog_level(volume);
+    ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+    volume = apm->recommended_stream_analog_level();
+  }
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithoutTs) {
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,"
+      "switch_to_agc2:false,"
+      "disallow_transient_suppressor_usage:true/");
+  auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create();
+
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  std::array<float, kSampleRateHz / 100> buffer;
+  float* channel_pointers[] = {buffer.data()};
+  StreamConfig stream_config(kSampleRateHz, kNumChannels);
+  Random random_generator(2341U);
+  constexpr int kFramesToProcess = 10;
+  int volume = 100;
+  for (int i = 0; i < kFramesToProcess; ++i) {
+    SCOPED_TRACE(i);
+    RandomizeSampleVector(&random_generator, buffer);
+    apm->set_stream_analog_level(volume);
+    ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+    volume = apm->recommended_stream_analog_level();
+  }
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest,
+       ProcessSucceedsWhenSwitchToFullAgc2WithTs) {
+  AudioProcessing::Config config = GetParam();
+  if (!config.transient_suppression.enabled) {
+    GTEST_SKIP() << "TS is disabled, skip.";
+  }
+
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,"
+      "switch_to_agc2:true,"
+      "disallow_transient_suppressor_usage:false/");
+  auto apm = AudioProcessingBuilder().SetConfig(config).Create();
+
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  std::array<float, kSampleRateHz / 100> buffer;
+  float* channel_pointers[] = {buffer.data()};
+  StreamConfig stream_config(kSampleRateHz, kNumChannels);
+  Random random_generator(2341U);
+  constexpr int kFramesToProcess = 10;
+  int volume = 100;
+  for (int i = 0; i < kFramesToProcess; ++i) {
+    SCOPED_TRACE(i);
+    RandomizeSampleVector(&random_generator, buffer);
+    apm->set_stream_analog_level(volume);
+    ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+    volume = apm->recommended_stream_analog_level();
+  }
+}
+
+TEST_P(Agc2FieldTrialParametrizedTest,
+       ProcessSucceedsWhenSwitchToFullAgc2WithoutTs) {
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,"
+      "switch_to_agc2:true,"
+      "disallow_transient_suppressor_usage:true/");
+  auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create();
+
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  std::array<float, kSampleRateHz / 100> buffer;
+  float* channel_pointers[] = {buffer.data()};
+  StreamConfig stream_config(kSampleRateHz, kNumChannels);
+  Random random_generator(2341U);
+  constexpr int kFramesToProcess = 10;
+  int volume = 100;
+  for (int i = 0; i < kFramesToProcess; ++i) {
+    SCOPED_TRACE(i);
+    RandomizeSampleVector(&random_generator, buffer);
+    apm->set_stream_analog_level(volume);
+    ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
+                                 channel_pointers),
+              kNoErr);
+    volume = apm->recommended_stream_analog_level();
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioProcessingImplTest,
+    Agc2FieldTrialParametrizedTest,
+    ::testing::Values(
+        // Full AGC1, TS disabled.
+        AudioProcessing::Config{
+            .transient_suppression = {.enabled = false},
+            .gain_controller1 =
+                {.enabled = true,
+                 .analog_gain_controller = {.enabled = true,
+                                            .enable_digital_adaptive = true}},
+            .gain_controller2 = {.enabled = false}},
+        // Full AGC1, TS enabled.
+        AudioProcessing::Config{
+            .transient_suppression = {.enabled = true},
+            .gain_controller1 =
+                {.enabled = true,
+                 .analog_gain_controller = {.enabled = true,
+                                            .enable_digital_adaptive = true}},
+            .gain_controller2 = {.enabled = false}},
+        // Hybrid AGC, TS disabled.
+        AudioProcessing::Config{
+            .transient_suppression = {.enabled = false},
+            .gain_controller1 =
+                {.enabled = true,
+                 .analog_gain_controller = {.enabled = true,
+                                            .enable_digital_adaptive = false}},
+            .gain_controller2 = {.enabled = true,
+                                 .adaptive_digital = {.enabled = true}}},
+        // Hybrid AGC, TS enabled.
+        AudioProcessing::Config{
+            .transient_suppression = {.enabled = true},
+            .gain_controller1 =
+                {.enabled = true,
+                 .analog_gain_controller = {.enabled = true,
+                                            .enable_digital_adaptive = false}},
+            .gain_controller2 = {.enabled = true,
+                                 .adaptive_digital = {.enabled = true}}}));
+
+TEST(AudioProcessingImplTest, CanDisableTransientSuppressor) {
+  constexpr AudioProcessing::Config kOriginal = {
+      .transient_suppression = {.enabled = false}};
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+  EXPECT_FALSE(adjusted.transient_suppression.enabled);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(kOriginal);
+  adjusted = apm->GetConfig();
+  EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
+}
+
+TEST(AudioProcessingImplTest, CanEnableTs) {
+  constexpr AudioProcessing::Config kOriginal = {
+      .transient_suppression = {.enabled = true}};
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+  EXPECT_TRUE(adjusted.transient_suppression.enabled);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(kOriginal);
+  adjusted = apm->GetConfig();
+  EXPECT_TRUE(adjusted.transient_suppression.enabled);
+}
+
+TEST(AudioProcessingImplTest, CanDisableTsWithAgc2FieldTrialDisabled) {
+  constexpr AudioProcessing::Config kOriginal = {
+      .transient_suppression = {.enabled = false}};
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Disabled/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+  EXPECT_FALSE(adjusted.transient_suppression.enabled);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(kOriginal);
+  adjusted = apm->GetConfig();
+  EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
+}
+
+TEST(AudioProcessingImplTest, CanEnableTsWithAgc2FieldTrialDisabled) {
+  constexpr AudioProcessing::Config kOriginal = {
+      .transient_suppression = {.enabled = true}};
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Disabled/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+  EXPECT_TRUE(adjusted.transient_suppression.enabled);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(kOriginal);
+  adjusted = apm->GetConfig();
+  EXPECT_TRUE(adjusted.transient_suppression.enabled);
+}
+
+TEST(AudioProcessingImplTest,
+     CanDisableTsWithAgc2FieldTrialEnabledAndUsageAllowed) {
+  constexpr AudioProcessing::Config kOriginal = {
+      .transient_suppression = {.enabled = false}};
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,"
+      "disallow_transient_suppressor_usage:false/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+  EXPECT_FALSE(adjusted.transient_suppression.enabled);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(kOriginal);
+  adjusted = apm->GetConfig();
+  EXPECT_FALSE(adjusted.transient_suppression.enabled);
+}
+
+TEST(AudioProcessingImplTest,
+     CanEnableTsWithAgc2FieldTrialEnabledAndUsageAllowed) {
+  constexpr AudioProcessing::Config kOriginal = {
+      .transient_suppression = {.enabled = true}};
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,"
+      "disallow_transient_suppressor_usage:false/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+  EXPECT_TRUE(adjusted.transient_suppression.enabled);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(kOriginal);
+  adjusted = apm->GetConfig();
+  EXPECT_TRUE(adjusted.transient_suppression.enabled);
+}
+
+TEST(AudioProcessingImplTest,
+     CannotEnableTsWithAgc2FieldTrialEnabledAndUsageDisallowed) {
+  constexpr AudioProcessing::Config kOriginal = {
+      .transient_suppression = {.enabled = true}};
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-GainController2/Enabled,"
+      "disallow_transient_suppressor_usage:true/");
+
+  // Test config application via `AudioProcessing` ctor.
+  auto adjusted =
+      AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
+  EXPECT_FALSE(adjusted.transient_suppression.enabled);
+
+  // Test config application via `AudioProcessing::ApplyConfig()`.
+  auto apm = AudioProcessingBuilder().Create();
+  apm->ApplyConfig(kOriginal);
+  adjusted = apm->GetConfig();
+  EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_performance_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_performance_unittest.cc
new file mode 100644
index 0000000000..10d3d84951
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -0,0 +1,568 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <math.h>
+
+#include <algorithm>
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.h"
+#include "modules/audio_processing/audio_processing_impl.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "modules/audio_processing/test/test_utils.h"
+#include "rtc_base/event.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/random.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::GetGlobalMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Metric;
+using ::webrtc::test::Unit;
+
+class CallSimulator;
+
+// Type of the render thread APM API call to use in the test.
+enum class ProcessorType { kRender, kCapture };
+
+// Variant of APM processing settings to use in the test.
+enum class SettingsType {
+  kDefaultApmDesktop,
+  kDefaultApmMobile,
+  kAllSubmodulesTurnedOff,
+  kDefaultApmDesktopWithoutDelayAgnostic,
+  kDefaultApmDesktopWithoutExtendedFilter
+};
+
+// Variables related to the audio data and formats.
+struct AudioFrameData {
+  explicit AudioFrameData(size_t max_frame_size) {
+    // Set up the two-dimensional arrays needed for the APM API calls.
+    input_framechannels.resize(2 * max_frame_size);
+    input_frame.resize(2);
+    input_frame[0] = &input_framechannels[0];
+    input_frame[1] = &input_framechannels[max_frame_size];
+
+    output_frame_channels.resize(2 * max_frame_size);
+    output_frame.resize(2);
+    output_frame[0] = &output_frame_channels[0];
+    output_frame[1] = &output_frame_channels[max_frame_size];
+  }
+
+  std::vector<float> output_frame_channels;
+  std::vector<float*> output_frame;
+  std::vector<float> input_framechannels;
+  std::vector<float*> input_frame;
+  StreamConfig input_stream_config;
+  StreamConfig output_stream_config;
+};
+
+// The configuration for the test.
+struct SimulationConfig {
+  SimulationConfig(int sample_rate_hz, SettingsType simulation_settings)
+      : sample_rate_hz(sample_rate_hz),
+        simulation_settings(simulation_settings) {}
+
+  static std::vector<SimulationConfig> GenerateSimulationConfigs() {
+    std::vector<SimulationConfig> simulation_configs;
+#ifndef WEBRTC_ANDROID
+    const SettingsType desktop_settings[] = {
+        SettingsType::kDefaultApmDesktop, SettingsType::kAllSubmodulesTurnedOff,
+        SettingsType::kDefaultApmDesktopWithoutDelayAgnostic,
+        SettingsType::kDefaultApmDesktopWithoutExtendedFilter};
+
+    const int desktop_sample_rates[] = {8000, 16000, 32000, 48000};
+
+    for (auto sample_rate : desktop_sample_rates) {
+      for (auto settings : desktop_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+#endif
+
+    const SettingsType mobile_settings[] = {SettingsType::kDefaultApmMobile};
+
+    const int mobile_sample_rates[] = {8000, 16000};
+
+    for (auto sample_rate : mobile_sample_rates) {
+      for (auto settings : mobile_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+
+    return simulation_configs;
+  }
+
+  std::string SettingsDescription() const {
+    std::string description;
+    switch (simulation_settings) {
+      case SettingsType::kDefaultApmMobile:
+        description = "DefaultApmMobile";
+        break;
+      case SettingsType::kDefaultApmDesktop:
+        description = "DefaultApmDesktop";
+        break;
+      case SettingsType::kAllSubmodulesTurnedOff:
+        description = "AllSubmodulesOff";
+        break;
+      case SettingsType::kDefaultApmDesktopWithoutDelayAgnostic:
+        description = "DefaultApmDesktopWithoutDelayAgnostic";
+        break;
+      case SettingsType::kDefaultApmDesktopWithoutExtendedFilter:
+        description = "DefaultApmDesktopWithoutExtendedFilter";
+        break;
+    }
+    return description;
+  }
+
+  int sample_rate_hz = 16000;
+  SettingsType simulation_settings = SettingsType::kDefaultApmDesktop;
+};
+
+// Handler for the frame counters.
+class FrameCounters {
+ public:
+  void IncreaseRenderCounter() { render_count_.fetch_add(1); }
+
+  void IncreaseCaptureCounter() { capture_count_.fetch_add(1); }
+
+  int CaptureMinusRenderCounters() const {
+    // The return value will be approximate, but that's good enough since
+    // by the time we return the value, it's not guaranteed to be correct
+    // anyway.
+    return capture_count_.load(std::memory_order_acquire) -
+           render_count_.load(std::memory_order_acquire);
+  }
+
+  int RenderMinusCaptureCounters() const {
+    return -CaptureMinusRenderCounters();
+  }
+
+  bool BothCountersExceedeThreshold(int threshold) const {
+    // TODO(tommi): We could use an event to signal this so that we don't need
+    // to be polling from the main thread and possibly steal cycles.
+    const int capture_count = capture_count_.load(std::memory_order_acquire);
+    const int render_count = render_count_.load(std::memory_order_acquire);
+    return (render_count > threshold && capture_count > threshold);
+  }
+
+ private:
+  std::atomic<int> render_count_{0};
+  std::atomic<int> capture_count_{0};
+};
+
+// Class that represents a flag that can only be raised.
+class LockedFlag {
+ public:
+  bool get_flag() const { return flag_.load(std::memory_order_acquire); }
+
+  void set_flag() {
+    if (!get_flag()) {
+      // read-only operation to avoid affecting the cache-line.
+      int zero = 0;
+      flag_.compare_exchange_strong(zero, 1);
+    }
+  }
+
+ private:
+  std::atomic<int> flag_{0};
+};
+
+// Parent class for the thread processors.
+class TimedThreadApiProcessor {
+ public:
+  TimedThreadApiProcessor(ProcessorType processor_type,
+                          Random* rand_gen,
+                          FrameCounters* shared_counters_state,
+                          LockedFlag* capture_call_checker,
+                          CallSimulator* test_framework,
+                          const SimulationConfig* simulation_config,
+                          AudioProcessing* apm,
+                          int num_durations_to_store,
+                          float input_level,
+                          int num_channels)
+      : rand_gen_(rand_gen),
+        frame_counters_(shared_counters_state),
+        capture_call_checker_(capture_call_checker),
+        test_(test_framework),
+        simulation_config_(simulation_config),
+        apm_(apm),
+        frame_data_(kMaxFrameSize),
+        clock_(webrtc::Clock::GetRealTimeClock()),
+        num_durations_to_store_(num_durations_to_store),
+        api_call_durations_(num_durations_to_store_ - kNumInitializationFrames),
+        samples_count_(0),
+        input_level_(input_level),
+        processor_type_(processor_type),
+        num_channels_(num_channels) {}
+
+  // Implements the callback functionality for the threads.
+  bool Process();
+
+  // Method for printing out the simulation statistics.
+  void print_processor_statistics(absl::string_view processor_name) const {
+    const std::string modifier = "_api_call_duration";
+
+    const std::string sample_rate_name =
+        "_" + std::to_string(simulation_config_->sample_rate_hz) + "Hz";
+
+    GetGlobalMetricsLogger()->LogMetric(
+        "apm_timing" + sample_rate_name, processor_name, api_call_durations_,
+        Unit::kMilliseconds, ImprovementDirection::kNeitherIsBetter);
+  }
+
+  void AddDuration(int64_t duration) {
+    if (samples_count_ >= kNumInitializationFrames &&
+        samples_count_ < num_durations_to_store_) {
+      api_call_durations_.AddSample(duration);
+    }
+    samples_count_++;
+  }
+
+ private:
+  static const int kMaxCallDifference = 10;
+  static const int kMaxFrameSize = 480;
+  static const int kNumInitializationFrames = 5;
+
+  int ProcessCapture() {
+    // Set the stream delay.
+    apm_->set_stream_delay_ms(30);
+
+    // Call and time the specified capture side API processing method.
+    const int64_t start_time = clock_->TimeInMicroseconds();
+    const int result = apm_->ProcessStream(
+        &frame_data_.input_frame[0], frame_data_.input_stream_config,
+        frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+    const int64_t end_time = clock_->TimeInMicroseconds();
+
+    frame_counters_->IncreaseCaptureCounter();
+
+    AddDuration(end_time - start_time);
+
+    if (first_process_call_) {
+      // Flag that the capture side has been called at least once
+      // (needed to ensure that a capture call has been done
+      // before the first render call is performed (implicitly
+      // required by the APM API).
+      capture_call_checker_->set_flag();
+      first_process_call_ = false;
+    }
+    return result;
+  }
+
+  bool ReadyToProcessCapture() {
+    return (frame_counters_->CaptureMinusRenderCounters() <=
+            kMaxCallDifference);
+  }
+
+  int ProcessRender() {
+    // Call and time the specified render side API processing method.
+    const int64_t start_time = clock_->TimeInMicroseconds();
+    const int result = apm_->ProcessReverseStream(
+        &frame_data_.input_frame[0], frame_data_.input_stream_config,
+        frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+    const int64_t end_time = clock_->TimeInMicroseconds();
+    frame_counters_->IncreaseRenderCounter();
+
+    AddDuration(end_time - start_time);
+
+    return result;
+  }
+
+  bool ReadyToProcessRender() {
+    // Do not process until at least one capture call has been done.
+    // (implicitly required by the APM API).
+    if (first_process_call_ && !capture_call_checker_->get_flag()) {
+      return false;
+    }
+
+    // Ensure that the number of render and capture calls do not differ too
+    // much.
+    if (frame_counters_->RenderMinusCaptureCounters() > kMaxCallDifference) {
+      return false;
+    }
+
+    first_process_call_ = false;
+    return true;
+  }
+
+  void PrepareFrame() {
+    // Lambda function for populating a float multichannel audio frame
+    // with random data.
+    auto populate_audio_frame = [](float amplitude, size_t num_channels,
+                                   size_t samples_per_channel, Random* rand_gen,
+                                   float** frame) {
+      for (size_t ch = 0; ch < num_channels; ch++) {
+        for (size_t k = 0; k < samples_per_channel; k++) {
+          // Store random float number with a value between +-amplitude.
+          frame[ch][k] = amplitude * (2 * rand_gen->Rand<float>() - 1);
+        }
+      }
+    };
+
+    // Prepare the audio input data and metadata.
+    frame_data_.input_stream_config.set_sample_rate_hz(
+        simulation_config_->sample_rate_hz);
+    frame_data_.input_stream_config.set_num_channels(num_channels_);
+    populate_audio_frame(input_level_, num_channels_,
+                         (simulation_config_->sample_rate_hz *
+                          AudioProcessing::kChunkSizeMs / 1000),
+                         rand_gen_, &frame_data_.input_frame[0]);
+
+    // Prepare the float audio output data and metadata.
+    frame_data_.output_stream_config.set_sample_rate_hz(
+        simulation_config_->sample_rate_hz);
+    frame_data_.output_stream_config.set_num_channels(1);
+  }
+
+  bool ReadyToProcess() {
+    switch (processor_type_) {
+      case ProcessorType::kRender:
+        return ReadyToProcessRender();
+
+      case ProcessorType::kCapture:
+        return ReadyToProcessCapture();
+    }
+
+    // Should not be reached, but the return statement is needed for the code to
+    // build successfully on Android.
+    RTC_DCHECK_NOTREACHED();
+    return false;
+  }
+
+  Random* rand_gen_ = nullptr;
+  FrameCounters* frame_counters_ = nullptr;
+  LockedFlag* capture_call_checker_ = nullptr;
+  CallSimulator* test_ = nullptr;
+  const SimulationConfig* const simulation_config_ = nullptr;
+  AudioProcessing* apm_ = nullptr;
+  AudioFrameData frame_data_;
+  webrtc::Clock* clock_;
+  const size_t num_durations_to_store_;
+  SamplesStatsCounter api_call_durations_;
+  size_t samples_count_ = 0;
+  const float input_level_;
+  bool first_process_call_ = true;
+  const ProcessorType processor_type_;
+  const int num_channels_ = 1;
+};
+
+// Class for managing the test simulation.
+class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
+ public:
+  CallSimulator()
+      : rand_gen_(42U),
+        simulation_config_(static_cast<SimulationConfig>(GetParam())) {}
+
+  // Run the call simulation with a timeout.
+  bool Run() {
+    StartThreads();
+
+    bool result = test_complete_.Wait(kTestTimeout);
+
+    StopThreads();
+
+    render_thread_state_->print_processor_statistics(
+        simulation_config_.SettingsDescription() + "_render");
+    capture_thread_state_->print_processor_statistics(
+        simulation_config_.SettingsDescription() + "_capture");
+
+    return result;
+  }
+
+  // Tests whether all the required render and capture side calls have been
+  // done.
+  bool MaybeEndTest() {
+    if (frame_counters_.BothCountersExceedeThreshold(kMinNumFramesToProcess)) {
+      test_complete_.Set();
+      return true;
+    }
+    return false;
+  }
+
+ private:
+  static const float kCaptureInputFloatLevel;
+  static const float kRenderInputFloatLevel;
+  static const int kMinNumFramesToProcess = 150;
+  static constexpr TimeDelta kTestTimeout =
+      TimeDelta::Millis(3 * 10 * kMinNumFramesToProcess);
+
+  // Stop all running threads.
+  void StopThreads() {
+    render_thread_.Finalize();
+    capture_thread_.Finalize();
+  }
+
+  // Simulator and APM setup.
+  void SetUp() override {
+    // Lambda function for setting the default APM runtime settings for desktop.
+    auto set_default_desktop_apm_runtime_settings = [](AudioProcessing* apm) {
+      AudioProcessing::Config apm_config = apm->GetConfig();
+      apm_config.echo_canceller.enabled = true;
+      apm_config.echo_canceller.mobile_mode = false;
+      apm_config.noise_suppression.enabled = true;
+      apm_config.gain_controller1.enabled = true;
+      apm_config.gain_controller1.mode =
+          AudioProcessing::Config::GainController1::kAdaptiveDigital;
+      apm->ApplyConfig(apm_config);
+    };
+
+    // Lambda function for setting the default APM runtime settings for mobile.
+    auto set_default_mobile_apm_runtime_settings = [](AudioProcessing* apm) {
+      AudioProcessing::Config apm_config = apm->GetConfig();
+      apm_config.echo_canceller.enabled = true;
+      apm_config.echo_canceller.mobile_mode = true;
+      apm_config.noise_suppression.enabled = true;
+      apm_config.gain_controller1.mode =
+          AudioProcessing::Config::GainController1::kAdaptiveDigital;
+      apm->ApplyConfig(apm_config);
+    };
+
+    // Lambda function for turning off all of the APM runtime settings
+    // submodules.
+    auto turn_off_default_apm_runtime_settings = [](AudioProcessing* apm) {
+      AudioProcessing::Config apm_config = apm->GetConfig();
+      apm_config.echo_canceller.enabled = false;
+      apm_config.gain_controller1.enabled = false;
+      apm_config.noise_suppression.enabled = false;
+      apm->ApplyConfig(apm_config);
+    };
+
+    int num_capture_channels = 1;
+    switch (simulation_config_.simulation_settings) {
+      case SettingsType::kDefaultApmMobile: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        set_default_mobile_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultApmDesktop: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kAllSubmodulesTurnedOff: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        turn_off_default_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultApmDesktopWithoutDelayAgnostic: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultApmDesktopWithoutExtendedFilter: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        break;
+      }
+    }
+
+    render_thread_state_.reset(new TimedThreadApiProcessor(
+        ProcessorType::kRender, &rand_gen_, &frame_counters_,
+        &capture_call_checker_, this, &simulation_config_, apm_.get(),
+        kMinNumFramesToProcess, kRenderInputFloatLevel, 1));
+    capture_thread_state_.reset(new TimedThreadApiProcessor(
+        ProcessorType::kCapture, &rand_gen_, &frame_counters_,
+        &capture_call_checker_, this, &simulation_config_, apm_.get(),
+        kMinNumFramesToProcess, kCaptureInputFloatLevel, num_capture_channels));
+  }
+
+  // Start the threads used in the test.
+  void StartThreads() {
+    const auto attributes =
+        rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
+    render_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (render_thread_state_->Process()) {
+          }
+        },
+        "render", attributes);
+    capture_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (capture_thread_state_->Process()) {
+          }
+        },
+        "capture", attributes);
+  }
+
+  // Event handler for the test.
+  rtc::Event test_complete_;
+
+  // Thread related variables.
+  Random rand_gen_;
+
+  rtc::scoped_refptr<AudioProcessing> apm_;
+  const SimulationConfig simulation_config_;
+  FrameCounters frame_counters_;
+  LockedFlag capture_call_checker_;
+  std::unique_ptr<TimedThreadApiProcessor> render_thread_state_;
+  std::unique_ptr<TimedThreadApiProcessor> capture_thread_state_;
+  rtc::PlatformThread render_thread_;
+  rtc::PlatformThread capture_thread_;
+};
+
+// Implements the callback functionality for the threads.
+bool TimedThreadApiProcessor::Process() {
+  PrepareFrame();
+
+  // Wait in a spinlock manner until it is ok to start processing.
+  // Note that SleepMs is not applicable since it only allows sleeping
+  // on a millisecond basis which is too long.
+  // TODO(tommi): This loop may affect the performance of the test that it's
+  // meant to measure.  See if we could use events instead to signal readiness.
+  while (!ReadyToProcess()) {
+  }
+
+  int result = AudioProcessing::kNoError;
+  switch (processor_type_) {
+    case ProcessorType::kRender:
+      result = ProcessRender();
+      break;
+    case ProcessorType::kCapture:
+      result = ProcessCapture();
+      break;
+  }
+
+  EXPECT_EQ(result, AudioProcessing::kNoError);
+
+  return !test_->MaybeEndTest();
+}
+
+const float CallSimulator::kRenderInputFloatLevel = 0.5f;
+const float CallSimulator::kCaptureInputFloatLevel = 0.03125f;
+}  // anonymous namespace
+
+// TODO(peah): Reactivate once issue 7712 has been resolved.
+TEST_P(CallSimulator, DISABLED_ApiCallDurationTest) {
+  // Run test and verify that it did not time out.
+  EXPECT_TRUE(Run());
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioProcessingPerformanceTest,
+    CallSimulator,
+    ::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_statistics_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/audio_processing_statistics_gn/moz.build
new file mode 100644
index 0000000000..f0af20c335
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_statistics_gn/moz.build
@@ -0,0 +1,217 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_processing_statistics_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc
new file mode 100644
index 0000000000..e320e71405
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc
@@ -0,0 +1,3441 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/include/audio_processing.h"
+
+#include <math.h>
+#include <stdio.h>
+
+#include <algorithm>
+#include <cmath>
+#include <limits>
+#include <memory>
+#include <numeric>
+#include <queue>
+#include <string>
+
+#include "absl/flags/flag.h"
+#include "absl/strings/string_view.h"
+#include "api/audio/echo_detector_creator.h"
+#include "api/make_ref_counted.h"
+#include "common_audio/include/audio_util.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "common_audio/resampler/push_sinc_resampler.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "modules/audio_processing/audio_processing_impl.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "modules/audio_processing/test/protobuf_utils.h"
+#include "modules/audio_processing/test/test_utils.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/protobuf_utils.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/swap_queue.h"
+#include "rtc_base/system/arch.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/thread.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "modules/audio_processing/debug.pb.h"
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/test/unittest.pb.h"
+#else
+#include "modules/audio_processing/test/unittest.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+
+ABSL_FLAG(bool,
+          write_apm_ref_data,
+          false,
+          "Write ApmTest.Process results to file, instead of comparing results "
+          "to the existing reference data file.");
+
+namespace webrtc {
+namespace {
+
+// All sample rates used by APM internally during processing. Other input /
+// output rates are resampled to / from one of these.
+const int kProcessSampleRates[] = {16000, 32000, 48000};
+
+enum StreamDirection { kForward = 0, kReverse };
+
+void ConvertToFloat(const int16_t* int_data, ChannelBuffer<float>* cb) {
+  ChannelBuffer<int16_t> cb_int(cb->num_frames(), cb->num_channels());
+  Deinterleave(int_data, cb->num_frames(), cb->num_channels(),
+               cb_int.channels());
+  for (size_t i = 0; i < cb->num_channels(); ++i) {
+    S16ToFloat(cb_int.channels()[i], cb->num_frames(), cb->channels()[i]);
+  }
+}
+
+void ConvertToFloat(const Int16FrameData& frame, ChannelBuffer<float>* cb) {
+  ConvertToFloat(frame.data.data(), cb);
+}
+
+void MixStereoToMono(const float* stereo,
+                     float* mono,
+                     size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; ++i)
+    mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) / 2;
+}
+
+void MixStereoToMono(const int16_t* stereo,
+                     int16_t* mono,
+                     size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; ++i)
+    mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) >> 1;
+}
+
+void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
+    stereo[i * 2 + 1] = stereo[i * 2];
+  }
+}
+
+void VerifyChannelsAreEqual(const int16_t* stereo, size_t samples_per_channel) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
+    EXPECT_EQ(stereo[i * 2 + 1], stereo[i * 2]);
+  }
+}
+
+void SetFrameTo(Int16FrameData* frame, int16_t value) {
+  for (size_t i = 0; i < frame->samples_per_channel * frame->num_channels;
+       ++i) {
+    frame->data[i] = value;
+  }
+}
+
+void SetFrameTo(Int16FrameData* frame, int16_t left, int16_t right) {
+  ASSERT_EQ(2u, frame->num_channels);
+  for (size_t i = 0; i < frame->samples_per_channel * 2; i += 2) {
+    frame->data[i] = left;
+    frame->data[i + 1] = right;
+  }
+}
+
+void ScaleFrame(Int16FrameData* frame, float scale) {
+  for (size_t i = 0; i < frame->samples_per_channel * frame->num_channels;
+       ++i) {
+    frame->data[i] = FloatS16ToS16(frame->data[i] * scale);
+  }
+}
+
+bool FrameDataAreEqual(const Int16FrameData& frame1,
+                       const Int16FrameData& frame2) {
+  if (frame1.samples_per_channel != frame2.samples_per_channel) {
+    return false;
+  }
+  if (frame1.num_channels != frame2.num_channels) {
+    return false;
+  }
+  if (memcmp(
+          frame1.data.data(), frame2.data.data(),
+          frame1.samples_per_channel * frame1.num_channels * sizeof(int16_t))) {
+    return false;
+  }
+  return true;
+}
+
+rtc::ArrayView<int16_t> GetMutableFrameData(Int16FrameData* frame) {
+  int16_t* ptr = frame->data.data();
+  const size_t len = frame->samples_per_channel * frame->num_channels;
+  return rtc::ArrayView<int16_t>(ptr, len);
+}
+
+rtc::ArrayView<const int16_t> GetFrameData(const Int16FrameData& frame) {
+  const int16_t* ptr = frame.data.data();
+  const size_t len = frame.samples_per_channel * frame.num_channels;
+  return rtc::ArrayView<const int16_t>(ptr, len);
+}
+
+void EnableAllAPComponents(AudioProcessing* ap) {
+  AudioProcessing::Config apm_config = ap->GetConfig();
+  apm_config.echo_canceller.enabled = true;
+#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
+  apm_config.echo_canceller.mobile_mode = true;
+
+  apm_config.gain_controller1.enabled = true;
+  apm_config.gain_controller1.mode =
+      AudioProcessing::Config::GainController1::kAdaptiveDigital;
+#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+  apm_config.echo_canceller.mobile_mode = false;
+
+  apm_config.gain_controller1.enabled = true;
+  apm_config.gain_controller1.mode =
+      AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+#endif
+
+  apm_config.noise_suppression.enabled = true;
+
+  apm_config.high_pass_filter.enabled = true;
+  apm_config.pipeline.maximum_internal_processing_rate = 48000;
+  ap->ApplyConfig(apm_config);
+}
+
+// These functions are only used by ApmTest.Process.
+template <class T>
+T AbsValue(T a) {
+  return a > 0 ? a : -a;
+}
+
+int16_t MaxAudioFrame(const Int16FrameData& frame) {
+  const size_t length = frame.samples_per_channel * frame.num_channels;
+  int16_t max_data = AbsValue(frame.data[0]);
+  for (size_t i = 1; i < length; i++) {
+    max_data = std::max(max_data, AbsValue(frame.data[i]));
+  }
+
+  return max_data;
+}
+
+void OpenFileAndWriteMessage(absl::string_view filename,
+                             const MessageLite& msg) {
+  FILE* file = fopen(std::string(filename).c_str(), "wb");
+  ASSERT_TRUE(file != NULL);
+
+  int32_t size = rtc::checked_cast<int32_t>(msg.ByteSizeLong());
+  ASSERT_GT(size, 0);
+  std::unique_ptr<uint8_t[]> array(new uint8_t[size]);
+  ASSERT_TRUE(msg.SerializeToArray(array.get(), size));
+
+  ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
+  ASSERT_EQ(static_cast<size_t>(size),
+            fwrite(array.get(), sizeof(array[0]), size, file));
+  fclose(file);
+}
+
+std::string ResourceFilePath(absl::string_view name, int sample_rate_hz) {
+  rtc::StringBuilder ss;
+  // Resource files are all stereo.
+  ss << name << sample_rate_hz / 1000 << "_stereo";
+  return test::ResourcePath(ss.str(), "pcm");
+}
+
+// Temporary filenames unique to this process. Used to be able to run these
+// tests in parallel as each process needs to be running in isolation they can't
+// have competing filenames.
+std::map<std::string, std::string> temp_filenames;
+
+std::string OutputFilePath(absl::string_view name,
+                           int input_rate,
+                           int output_rate,
+                           int reverse_input_rate,
+                           int reverse_output_rate,
+                           size_t num_input_channels,
+                           size_t num_output_channels,
+                           size_t num_reverse_input_channels,
+                           size_t num_reverse_output_channels,
+                           StreamDirection file_direction) {
+  rtc::StringBuilder ss;
+  ss << name << "_i" << num_input_channels << "_" << input_rate / 1000 << "_ir"
+     << num_reverse_input_channels << "_" << reverse_input_rate / 1000 << "_";
+  if (num_output_channels == 1) {
+    ss << "mono";
+  } else if (num_output_channels == 2) {
+    ss << "stereo";
+  } else {
+    RTC_DCHECK_NOTREACHED();
+  }
+  ss << output_rate / 1000;
+  if (num_reverse_output_channels == 1) {
+    ss << "_rmono";
+  } else if (num_reverse_output_channels == 2) {
+    ss << "_rstereo";
+  } else {
+    RTC_DCHECK_NOTREACHED();
+  }
+  ss << reverse_output_rate / 1000;
+  ss << "_d" << file_direction << "_pcm";
+
+  std::string filename = ss.str();
+  if (temp_filenames[filename].empty())
+    temp_filenames[filename] = test::TempFilename(test::OutputPath(), filename);
+  return temp_filenames[filename];
+}
+
+void ClearTempFiles() {
+  for (auto& kv : temp_filenames)
+    remove(kv.second.c_str());
+}
+
+// Only remove "out" files. Keep "ref" files.
+void ClearTempOutFiles() {
+  for (auto it = temp_filenames.begin(); it != temp_filenames.end();) {
+    const std::string& filename = it->first;
+    if (filename.substr(0, 3).compare("out") == 0) {
+      remove(it->second.c_str());
+      temp_filenames.erase(it++);
+    } else {
+      it++;
+    }
+  }
+}
+
+void OpenFileAndReadMessage(absl::string_view filename, MessageLite* msg) {
+  FILE* file = fopen(std::string(filename).c_str(), "rb");
+  ASSERT_TRUE(file != NULL);
+  ReadMessageFromFile(file, msg);
+  fclose(file);
+}
+
+// Reads a 10 ms chunk (actually AudioProcessing::GetFrameSize() samples per
+// channel) of int16 interleaved audio from the given (assumed stereo) file,
+// converts to deinterleaved float (optionally downmixing) and returns the
+// result in `cb`. Returns false if the file ended (or on error) and true
+// otherwise.
+//
+// `int_data` and `float_data` are just temporary space that must be
+// sufficiently large to hold the 10 ms chunk.
+bool ReadChunk(FILE* file,
+               int16_t* int_data,
+               float* float_data,
+               ChannelBuffer<float>* cb) {
+  // The files always contain stereo audio.
+  size_t frame_size = cb->num_frames() * 2;
+  size_t read_count = fread(int_data, sizeof(int16_t), frame_size, file);
+  if (read_count != frame_size) {
+    // Check that the file really ended.
+    RTC_DCHECK(feof(file));
+    return false;  // This is expected.
+  }
+
+  S16ToFloat(int_data, frame_size, float_data);
+  if (cb->num_channels() == 1) {
+    MixStereoToMono(float_data, cb->channels()[0], cb->num_frames());
+  } else {
+    Deinterleave(float_data, cb->num_frames(), 2, cb->channels());
+  }
+
+  return true;
+}
+
+// Returns the reference file name that matches the current CPU
+// architecture/optimizations.
+std::string GetReferenceFilename() {
+#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
+  return test::ResourcePath("audio_processing/output_data_fixed", "pb");
+#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+  if (GetCPUInfo(kAVX2) != 0) {
+    return test::ResourcePath("audio_processing/output_data_float_avx2", "pb");
+  }
+  return test::ResourcePath("audio_processing/output_data_float", "pb");
+#endif
+}
+
+// Flag that can temporarily be enabled for local debugging to inspect
+// `ApmTest.VerifyDebugDump(Int|Float)` failures. Do not upload code changes
+// with this flag set to true.
+constexpr bool kDumpWhenExpectMessageEqFails = false;
+
+// Checks the debug constants values used in this file so that no code change is
+// submitted with values temporarily used for local debugging.
+TEST(ApmUnitTests, CheckDebugConstants) {
+  ASSERT_FALSE(kDumpWhenExpectMessageEqFails);
+}
+
+// Expects the equality of `actual` and `expected` by inspecting a hard-coded
+// subset of `audioproc::Stream` fields.
+void ExpectStreamFieldsEq(const audioproc::Stream& actual,
+                          const audioproc::Stream& expected) {
+  EXPECT_EQ(actual.input_data(), expected.input_data());
+  EXPECT_EQ(actual.output_data(), expected.output_data());
+  EXPECT_EQ(actual.delay(), expected.delay());
+  EXPECT_EQ(actual.drift(), expected.drift());
+  EXPECT_EQ(actual.applied_input_volume(), expected.applied_input_volume());
+  EXPECT_EQ(actual.keypress(), expected.keypress());
+}
+
+// Expects the equality of `actual` and `expected` by inspecting a hard-coded
+// subset of `audioproc::Event` fields.
+void ExpectEventFieldsEq(const audioproc::Event& actual,
+                         const audioproc::Event& expected) {
+  EXPECT_EQ(actual.type(), expected.type());
+  if (actual.type() != expected.type()) {
+    return;
+  }
+  switch (actual.type()) {
+    case audioproc::Event::STREAM:
+      ExpectStreamFieldsEq(actual.stream(), expected.stream());
+      break;
+    default:
+      // Not implemented.
+      break;
+  }
+}
+
+// Returns true if the `actual` and `expected` byte streams share the same size
+// and contain the same data. If they differ and `kDumpWhenExpectMessageEqFails`
+// is true, checks the equality of a subset of `audioproc::Event` (nested)
+// fields.
+bool ExpectMessageEq(rtc::ArrayView<const uint8_t> actual,
+                     rtc::ArrayView<const uint8_t> expected) {
+  EXPECT_EQ(actual.size(), expected.size());
+  if (actual.size() != expected.size()) {
+    return false;
+  }
+  if (memcmp(actual.data(), expected.data(), actual.size()) == 0) {
+    // Same message. No need to parse.
+    return true;
+  }
+  if (kDumpWhenExpectMessageEqFails) {
+    // Parse differing messages and expect equality to produce detailed error
+    // messages.
+    audioproc::Event event_actual, event_expected;
+    RTC_DCHECK(event_actual.ParseFromArray(actual.data(), actual.size()));
+    RTC_DCHECK(event_expected.ParseFromArray(expected.data(), expected.size()));
+    ExpectEventFieldsEq(event_actual, event_expected);
+  }
+  return false;
+}
+
+class ApmTest : public ::testing::Test {
+ protected:
+  ApmTest();
+  virtual void SetUp();
+  virtual void TearDown();
+
+  static void SetUpTestSuite() {}
+
+  static void TearDownTestSuite() { ClearTempFiles(); }
+
+  // Used to select between int and float interface tests.
+  enum Format { kIntFormat, kFloatFormat };
+
+  void Init(int sample_rate_hz,
+            int output_sample_rate_hz,
+            int reverse_sample_rate_hz,
+            size_t num_input_channels,
+            size_t num_output_channels,
+            size_t num_reverse_channels,
+            bool open_output_file);
+  void Init(AudioProcessing* ap);
+  void EnableAllComponents();
+  bool ReadFrame(FILE* file, Int16FrameData* frame);
+  bool ReadFrame(FILE* file, Int16FrameData* frame, ChannelBuffer<float>* cb);
+  void ReadFrameWithRewind(FILE* file, Int16FrameData* frame);
+  void ReadFrameWithRewind(FILE* file,
+                           Int16FrameData* frame,
+                           ChannelBuffer<float>* cb);
+  void ProcessDelayVerificationTest(int delay_ms,
+                                    int system_delay_ms,
+                                    int delay_min,
+                                    int delay_max);
+  void TestChangingChannelsInt16Interface(
+      size_t num_channels,
+      AudioProcessing::Error expected_return);
+  void TestChangingForwardChannels(size_t num_in_channels,
+                                   size_t num_out_channels,
+                                   AudioProcessing::Error expected_return);
+  void TestChangingReverseChannels(size_t num_rev_channels,
+                                   AudioProcessing::Error expected_return);
+  void RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate);
+  void RunManualVolumeChangeIsPossibleTest(int sample_rate);
+  void StreamParametersTest(Format format);
+  int ProcessStreamChooser(Format format);
+  int AnalyzeReverseStreamChooser(Format format);
+  void ProcessDebugDump(absl::string_view in_filename,
+                        absl::string_view out_filename,
+                        Format format,
+                        int max_size_bytes);
+  void VerifyDebugDumpTest(Format format);
+
+  const std::string output_path_;
+  const std::string ref_filename_;
+  rtc::scoped_refptr<AudioProcessing> apm_;
+  Int16FrameData frame_;
+  Int16FrameData revframe_;
+  std::unique_ptr<ChannelBuffer<float>> float_cb_;
+  std::unique_ptr<ChannelBuffer<float>> revfloat_cb_;
+  int output_sample_rate_hz_;
+  size_t num_output_channels_;
+  FILE* far_file_;
+  FILE* near_file_;
+  FILE* out_file_;
+};
+
+ApmTest::ApmTest()
+    : output_path_(test::OutputPath()),
+      ref_filename_(GetReferenceFilename()),
+      output_sample_rate_hz_(0),
+      num_output_channels_(0),
+      far_file_(NULL),
+      near_file_(NULL),
+      out_file_(NULL) {
+  apm_ = AudioProcessingBuilderForTesting().Create();
+  AudioProcessing::Config apm_config = apm_->GetConfig();
+  apm_config.gain_controller1.analog_gain_controller.enabled = false;
+  apm_config.pipeline.maximum_internal_processing_rate = 48000;
+  apm_->ApplyConfig(apm_config);
+}
+
+void ApmTest::SetUp() {
+  ASSERT_TRUE(apm_.get() != NULL);
+
+  Init(32000, 32000, 32000, 2, 2, 2, false);
+}
+
+void ApmTest::TearDown() {
+  if (far_file_) {
+    ASSERT_EQ(0, fclose(far_file_));
+  }
+  far_file_ = NULL;
+
+  if (near_file_) {
+    ASSERT_EQ(0, fclose(near_file_));
+  }
+  near_file_ = NULL;
+
+  if (out_file_) {
+    ASSERT_EQ(0, fclose(out_file_));
+  }
+  out_file_ = NULL;
+}
+
+void ApmTest::Init(AudioProcessing* ap) {
+  ASSERT_EQ(
+      kNoErr,
+      ap->Initialize({{{frame_.sample_rate_hz, frame_.num_channels},
+                       {output_sample_rate_hz_, num_output_channels_},
+                       {revframe_.sample_rate_hz, revframe_.num_channels},
+                       {revframe_.sample_rate_hz, revframe_.num_channels}}}));
+}
+
+void ApmTest::Init(int sample_rate_hz,
+                   int output_sample_rate_hz,
+                   int reverse_sample_rate_hz,
+                   size_t num_input_channels,
+                   size_t num_output_channels,
+                   size_t num_reverse_channels,
+                   bool open_output_file) {
+  SetContainerFormat(sample_rate_hz, num_input_channels, &frame_, &float_cb_);
+  output_sample_rate_hz_ = output_sample_rate_hz;
+  num_output_channels_ = num_output_channels;
+
+  SetContainerFormat(reverse_sample_rate_hz, num_reverse_channels, &revframe_,
+                     &revfloat_cb_);
+  Init(apm_.get());
+
+  if (far_file_) {
+    ASSERT_EQ(0, fclose(far_file_));
+  }
+  std::string filename = ResourceFilePath("far", sample_rate_hz);
+  far_file_ = fopen(filename.c_str(), "rb");
+  ASSERT_TRUE(far_file_ != NULL) << "Could not open file " << filename << "\n";
+
+  if (near_file_) {
+    ASSERT_EQ(0, fclose(near_file_));
+  }
+  filename = ResourceFilePath("near", sample_rate_hz);
+  near_file_ = fopen(filename.c_str(), "rb");
+  ASSERT_TRUE(near_file_ != NULL) << "Could not open file " << filename << "\n";
+
+  if (open_output_file) {
+    if (out_file_) {
+      ASSERT_EQ(0, fclose(out_file_));
+    }
+    filename = OutputFilePath(
+        "out", sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz,
+        reverse_sample_rate_hz, num_input_channels, num_output_channels,
+        num_reverse_channels, num_reverse_channels, kForward);
+    out_file_ = fopen(filename.c_str(), "wb");
+    ASSERT_TRUE(out_file_ != NULL)
+        << "Could not open file " << filename << "\n";
+  }
+}
+
+void ApmTest::EnableAllComponents() {
+  EnableAllAPComponents(apm_.get());
+}
+
+bool ApmTest::ReadFrame(FILE* file,
+                        Int16FrameData* frame,
+                        ChannelBuffer<float>* cb) {
+  // The files always contain stereo audio.
+  size_t frame_size = frame->samples_per_channel * 2;
+  size_t read_count =
+      fread(frame->data.data(), sizeof(int16_t), frame_size, file);
+  if (read_count != frame_size) {
+    // Check that the file really ended.
+    EXPECT_NE(0, feof(file));
+    return false;  // This is expected.
+  }
+
+  if (frame->num_channels == 1) {
+    MixStereoToMono(frame->data.data(), frame->data.data(),
+                    frame->samples_per_channel);
+  }
+
+  if (cb) {
+    ConvertToFloat(*frame, cb);
+  }
+  return true;
+}
+
+bool ApmTest::ReadFrame(FILE* file, Int16FrameData* frame) {
+  return ReadFrame(file, frame, NULL);
+}
+
+// If the end of the file has been reached, rewind it and attempt to read the
+// frame again.
+void ApmTest::ReadFrameWithRewind(FILE* file,
+                                  Int16FrameData* frame,
+                                  ChannelBuffer<float>* cb) {
+  if (!ReadFrame(near_file_, &frame_, cb)) {
+    rewind(near_file_);
+    ASSERT_TRUE(ReadFrame(near_file_, &frame_, cb));
+  }
+}
+
+void ApmTest::ReadFrameWithRewind(FILE* file, Int16FrameData* frame) {
+  ReadFrameWithRewind(file, frame, NULL);
+}
+
+int ApmTest::ProcessStreamChooser(Format format) {
+  if (format == kIntFormat) {
+    return apm_->ProcessStream(
+        frame_.data.data(),
+        StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+        StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+        frame_.data.data());
+  }
+  return apm_->ProcessStream(
+      float_cb_->channels(),
+      StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+      StreamConfig(output_sample_rate_hz_, num_output_channels_),
+      float_cb_->channels());
+}
+
+int ApmTest::AnalyzeReverseStreamChooser(Format format) {
+  if (format == kIntFormat) {
+    return apm_->ProcessReverseStream(
+        revframe_.data.data(),
+        StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+        StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+        revframe_.data.data());
+  }
+  return apm_->AnalyzeReverseStream(
+      revfloat_cb_->channels(),
+      StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels));
+}
+
+void ApmTest::ProcessDelayVerificationTest(int delay_ms,
+                                           int system_delay_ms,
+                                           int delay_min,
+                                           int delay_max) {
+  // The `revframe_` and `frame_` should include the proper frame information,
+  // hence can be used for extracting information.
+  Int16FrameData tmp_frame;
+  std::queue<Int16FrameData*> frame_queue;
+  bool causal = true;
+
+  tmp_frame.CopyFrom(revframe_);
+  SetFrameTo(&tmp_frame, 0);
+
+  EXPECT_EQ(apm_->kNoError, apm_->Initialize());
+  // Initialize the `frame_queue` with empty frames.
+  int frame_delay = delay_ms / 10;
+  while (frame_delay < 0) {
+    Int16FrameData* frame = new Int16FrameData();
+    frame->CopyFrom(tmp_frame);
+    frame_queue.push(frame);
+    frame_delay++;
+    causal = false;
+  }
+  while (frame_delay > 0) {
+    Int16FrameData* frame = new Int16FrameData();
+    frame->CopyFrom(tmp_frame);
+    frame_queue.push(frame);
+    frame_delay--;
+  }
+  // Run for 4.5 seconds, skipping statistics from the first 2.5 seconds.  We
+  // need enough frames with audio to have reliable estimates, but as few as
+  // possible to keep processing time down.  4.5 seconds seemed to be a good
+  // compromise for this recording.
+  for (int frame_count = 0; frame_count < 450; ++frame_count) {
+    Int16FrameData* frame = new Int16FrameData();
+    frame->CopyFrom(tmp_frame);
+    // Use the near end recording, since that has more speech in it.
+    ASSERT_TRUE(ReadFrame(near_file_, frame));
+    frame_queue.push(frame);
+    Int16FrameData* reverse_frame = frame;
+    Int16FrameData* process_frame = frame_queue.front();
+    if (!causal) {
+      reverse_frame = frame_queue.front();
+      // When we call ProcessStream() the frame is modified, so we can't use the
+      // pointer directly when things are non-causal. Use an intermediate frame
+      // and copy the data.
+      process_frame = &tmp_frame;
+      process_frame->CopyFrom(*frame);
+    }
+    EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(
+                                  reverse_frame->data.data(),
+                                  StreamConfig(reverse_frame->sample_rate_hz,
+                                               reverse_frame->num_channels),
+                                  StreamConfig(reverse_frame->sample_rate_hz,
+                                               reverse_frame->num_channels),
+                                  reverse_frame->data.data()));
+    EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(system_delay_ms));
+    EXPECT_EQ(apm_->kNoError,
+              apm_->ProcessStream(process_frame->data.data(),
+                                  StreamConfig(process_frame->sample_rate_hz,
+                                               process_frame->num_channels),
+                                  StreamConfig(process_frame->sample_rate_hz,
+                                               process_frame->num_channels),
+                                  process_frame->data.data()));
+    frame = frame_queue.front();
+    frame_queue.pop();
+    delete frame;
+
+    if (frame_count == 250) {
+      // Discard the first delay metrics to avoid convergence effects.
+      static_cast<void>(apm_->GetStatistics());
+    }
+  }
+
+  rewind(near_file_);
+  while (!frame_queue.empty()) {
+    Int16FrameData* frame = frame_queue.front();
+    frame_queue.pop();
+    delete frame;
+  }
+  // Calculate expected delay estimate and acceptable regions. Further,
+  // limit them w.r.t. AEC delay estimation support.
+  const size_t samples_per_ms =
+      rtc::SafeMin<size_t>(16u, frame_.samples_per_channel / 10);
+  const int expected_median =
+      rtc::SafeClamp<int>(delay_ms - system_delay_ms, delay_min, delay_max);
+  const int expected_median_high = rtc::SafeClamp<int>(
+      expected_median + rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min,
+      delay_max);
+  const int expected_median_low = rtc::SafeClamp<int>(
+      expected_median - rtc::dchecked_cast<int>(96 / samples_per_ms), delay_min,
+      delay_max);
+  // Verify delay metrics.
+  AudioProcessingStats stats = apm_->GetStatistics();
+  ASSERT_TRUE(stats.delay_median_ms.has_value());
+  int32_t median = *stats.delay_median_ms;
+  EXPECT_GE(expected_median_high, median);
+  EXPECT_LE(expected_median_low, median);
+}
+
+void ApmTest::StreamParametersTest(Format format) {
+  // No errors when the components are disabled.
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
+
+  // -- Missing AGC level --
+  AudioProcessing::Config apm_config = apm_->GetConfig();
+  apm_config.gain_controller1.enabled = true;
+  apm_->ApplyConfig(apm_config);
+  EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
+
+  // Resets after successful ProcessStream().
+  apm_->set_stream_analog_level(127);
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
+  EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
+
+  // Other stream parameters set correctly.
+  apm_config.echo_canceller.enabled = true;
+  apm_config.echo_canceller.mobile_mode = false;
+  apm_->ApplyConfig(apm_config);
+  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+  EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
+  apm_config.gain_controller1.enabled = false;
+  apm_->ApplyConfig(apm_config);
+
+  // -- Missing delay --
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
+
+  // Resets after successful ProcessStream().
+  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
+
+  // Other stream parameters set correctly.
+  apm_config.gain_controller1.enabled = true;
+  apm_->ApplyConfig(apm_config);
+  apm_->set_stream_analog_level(127);
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
+  apm_config.gain_controller1.enabled = false;
+  apm_->ApplyConfig(apm_config);
+
+  // -- No stream parameters --
+  EXPECT_EQ(apm_->kNoError, AnalyzeReverseStreamChooser(format));
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
+
+  // -- All there --
+  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
+  apm_->set_stream_analog_level(127);
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
+}
+
+TEST_F(ApmTest, StreamParametersInt) {
+  StreamParametersTest(kIntFormat);
+}
+
+TEST_F(ApmTest, StreamParametersFloat) {
+  StreamParametersTest(kFloatFormat);
+}
+
+void ApmTest::TestChangingChannelsInt16Interface(
+    size_t num_channels,
+    AudioProcessing::Error expected_return) {
+  frame_.num_channels = num_channels;
+
+  EXPECT_EQ(expected_return,
+            apm_->ProcessStream(
+                frame_.data.data(),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                frame_.data.data()));
+  EXPECT_EQ(expected_return,
+            apm_->ProcessReverseStream(
+                frame_.data.data(),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                frame_.data.data()));
+}
+
+void ApmTest::TestChangingForwardChannels(
+    size_t num_in_channels,
+    size_t num_out_channels,
+    AudioProcessing::Error expected_return) {
+  const StreamConfig input_stream = {frame_.sample_rate_hz, num_in_channels};
+  const StreamConfig output_stream = {output_sample_rate_hz_, num_out_channels};
+
+  EXPECT_EQ(expected_return,
+            apm_->ProcessStream(float_cb_->channels(), input_stream,
+                                output_stream, float_cb_->channels()));
+}
+
+void ApmTest::TestChangingReverseChannels(
+    size_t num_rev_channels,
+    AudioProcessing::Error expected_return) {
+  const ProcessingConfig processing_config = {
+      {{frame_.sample_rate_hz, apm_->num_input_channels()},
+       {output_sample_rate_hz_, apm_->num_output_channels()},
+       {frame_.sample_rate_hz, num_rev_channels},
+       {frame_.sample_rate_hz, num_rev_channels}}};
+
+  EXPECT_EQ(
+      expected_return,
+      apm_->ProcessReverseStream(
+          float_cb_->channels(), processing_config.reverse_input_stream(),
+          processing_config.reverse_output_stream(), float_cb_->channels()));
+}
+
+TEST_F(ApmTest, ChannelsInt16Interface) {
+  // Testing number of invalid and valid channels.
+  Init(16000, 16000, 16000, 4, 4, 4, false);
+
+  TestChangingChannelsInt16Interface(0, apm_->kBadNumberChannelsError);
+
+  for (size_t i = 1; i < 4; i++) {
+    TestChangingChannelsInt16Interface(i, kNoErr);
+    EXPECT_EQ(i, apm_->num_input_channels());
+  }
+}
+
+TEST_F(ApmTest, Channels) {
+  // Testing number of invalid and valid channels.
+  Init(16000, 16000, 16000, 4, 4, 4, false);
+
+  TestChangingForwardChannels(0, 1, apm_->kBadNumberChannelsError);
+  TestChangingReverseChannels(0, apm_->kBadNumberChannelsError);
+
+  for (size_t i = 1; i < 4; ++i) {
+    for (size_t j = 0; j < 1; ++j) {
+      // Output channels much be one or match input channels.
+      if (j == 1 || i == j) {
+        TestChangingForwardChannels(i, j, kNoErr);
+        TestChangingReverseChannels(i, kNoErr);
+
+        EXPECT_EQ(i, apm_->num_input_channels());
+        EXPECT_EQ(j, apm_->num_output_channels());
+        // The number of reverse channels used for processing to is always 1.
+        EXPECT_EQ(1u, apm_->num_reverse_channels());
+      } else {
+        TestChangingForwardChannels(i, j,
+                                    AudioProcessing::kBadNumberChannelsError);
+      }
+    }
+  }
+}
+
+TEST_F(ApmTest, SampleRatesInt) {
+  // Testing some valid sample rates.
+  for (int sample_rate : {8000, 12000, 16000, 32000, 44100, 48000, 96000}) {
+    SetContainerFormat(sample_rate, 2, &frame_, &float_cb_);
+    EXPECT_NOERR(ProcessStreamChooser(kIntFormat));
+  }
+}
+
+// This test repeatedly reconfigures the pre-amplifier in APM, processes a
+// number of frames, and checks that output signal has the right level.
+TEST_F(ApmTest, PreAmplifier) {
+  // Fill the audio frame with a sawtooth pattern.
+  rtc::ArrayView<int16_t> frame_data = GetMutableFrameData(&frame_);
+  const size_t samples_per_channel = frame_.samples_per_channel;
+  for (size_t i = 0; i < samples_per_channel; i++) {
+    for (size_t ch = 0; ch < frame_.num_channels; ++ch) {
+      frame_data[i + ch * samples_per_channel] = 10000 * ((i % 3) - 1);
+    }
+  }
+  // Cache the frame in tmp_frame.
+  Int16FrameData tmp_frame;
+  tmp_frame.CopyFrom(frame_);
+
+  auto compute_power = [](const Int16FrameData& frame) {
+    rtc::ArrayView<const int16_t> data = GetFrameData(frame);
+    return std::accumulate(data.begin(), data.end(), 0.0f,
+                           [](float a, float b) { return a + b * b; }) /
+           data.size() / 32768 / 32768;
+  };
+
+  const float input_power = compute_power(tmp_frame);
+  // Double-check that the input data is large compared to the error kEpsilon.
+  constexpr float kEpsilon = 1e-4f;
+  RTC_DCHECK_GE(input_power, 10 * kEpsilon);
+
+  // 1. Enable pre-amp with 0 dB gain.
+  AudioProcessing::Config config = apm_->GetConfig();
+  config.pre_amplifier.enabled = true;
+  config.pre_amplifier.fixed_gain_factor = 1.0f;
+  apm_->ApplyConfig(config);
+
+  for (int i = 0; i < 20; ++i) {
+    frame_.CopyFrom(tmp_frame);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
+  }
+  float output_power = compute_power(frame_);
+  EXPECT_NEAR(output_power, input_power, kEpsilon);
+  config = apm_->GetConfig();
+  EXPECT_EQ(config.pre_amplifier.fixed_gain_factor, 1.0f);
+
+  // 2. Change pre-amp gain via ApplyConfig.
+  config.pre_amplifier.fixed_gain_factor = 2.0f;
+  apm_->ApplyConfig(config);
+
+  for (int i = 0; i < 20; ++i) {
+    frame_.CopyFrom(tmp_frame);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
+  }
+  output_power = compute_power(frame_);
+  EXPECT_NEAR(output_power, 4 * input_power, kEpsilon);
+  config = apm_->GetConfig();
+  EXPECT_EQ(config.pre_amplifier.fixed_gain_factor, 2.0f);
+
+  // 3. Change pre-amp gain via a RuntimeSetting.
+  apm_->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCapturePreGain(1.5f));
+
+  for (int i = 0; i < 20; ++i) {
+    frame_.CopyFrom(tmp_frame);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
+  }
+  output_power = compute_power(frame_);
+  EXPECT_NEAR(output_power, 2.25 * input_power, kEpsilon);
+  config = apm_->GetConfig();
+  EXPECT_EQ(config.pre_amplifier.fixed_gain_factor, 1.5f);
+}
+
+// Ensures that the emulated analog mic gain functionality runs without
+// crashing.
+TEST_F(ApmTest, AnalogMicGainEmulation) {
+  // Fill the audio frame with a sawtooth pattern.
+  rtc::ArrayView<int16_t> frame_data = GetMutableFrameData(&frame_);
+  const size_t samples_per_channel = frame_.samples_per_channel;
+  for (size_t i = 0; i < samples_per_channel; i++) {
+    for (size_t ch = 0; ch < frame_.num_channels; ++ch) {
+      frame_data[i + ch * samples_per_channel] = 100 * ((i % 3) - 1);
+    }
+  }
+  // Cache the frame in tmp_frame.
+  Int16FrameData tmp_frame;
+  tmp_frame.CopyFrom(frame_);
+
+  // Enable the analog gain emulation.
+  AudioProcessing::Config config = apm_->GetConfig();
+  config.capture_level_adjustment.enabled = true;
+  config.capture_level_adjustment.analog_mic_gain_emulation.enabled = true;
+  config.capture_level_adjustment.analog_mic_gain_emulation.initial_level = 21;
+  config.gain_controller1.enabled = true;
+  config.gain_controller1.mode =
+      AudioProcessing::Config::GainController1::Mode::kAdaptiveAnalog;
+  config.gain_controller1.analog_gain_controller.enabled = true;
+  apm_->ApplyConfig(config);
+
+  // Process a number of frames to ensure that the code runs without crashes.
+  for (int i = 0; i < 20; ++i) {
+    frame_.CopyFrom(tmp_frame);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
+  }
+}
+
+// This test repeatedly reconfigures the capture level adjustment functionality
+// in APM, processes a number of frames, and checks that output signal has the
+// right level.
+TEST_F(ApmTest, CaptureLevelAdjustment) {
+  // Fill the audio frame with a sawtooth pattern.
+  rtc::ArrayView<int16_t> frame_data = GetMutableFrameData(&frame_);
+  const size_t samples_per_channel = frame_.samples_per_channel;
+  for (size_t i = 0; i < samples_per_channel; i++) {
+    for (size_t ch = 0; ch < frame_.num_channels; ++ch) {
+      frame_data[i + ch * samples_per_channel] = 100 * ((i % 3) - 1);
+    }
+  }
+  // Cache the frame in tmp_frame.
+  Int16FrameData tmp_frame;
+  tmp_frame.CopyFrom(frame_);
+
+  auto compute_power = [](const Int16FrameData& frame) {
+    rtc::ArrayView<const int16_t> data = GetFrameData(frame);
+    return std::accumulate(data.begin(), data.end(), 0.0f,
+                           [](float a, float b) { return a + b * b; }) /
+           data.size() / 32768 / 32768;
+  };
+
+  const float input_power = compute_power(tmp_frame);
+  // Double-check that the input data is large compared to the error kEpsilon.
+  constexpr float kEpsilon = 1e-20f;
+  RTC_DCHECK_GE(input_power, 10 * kEpsilon);
+
+  // 1. Enable pre-amp with 0 dB gain.
+  AudioProcessing::Config config = apm_->GetConfig();
+  config.capture_level_adjustment.enabled = true;
+  config.capture_level_adjustment.pre_gain_factor = 0.5f;
+  config.capture_level_adjustment.post_gain_factor = 4.f;
+  const float expected_output_power1 =
+      config.capture_level_adjustment.pre_gain_factor *
+      config.capture_level_adjustment.pre_gain_factor *
+      config.capture_level_adjustment.post_gain_factor *
+      config.capture_level_adjustment.post_gain_factor * input_power;
+  apm_->ApplyConfig(config);
+
+  for (int i = 0; i < 20; ++i) {
+    frame_.CopyFrom(tmp_frame);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
+  }
+  float output_power = compute_power(frame_);
+  EXPECT_NEAR(output_power, expected_output_power1, kEpsilon);
+  config = apm_->GetConfig();
+  EXPECT_EQ(config.capture_level_adjustment.pre_gain_factor, 0.5f);
+  EXPECT_EQ(config.capture_level_adjustment.post_gain_factor, 4.f);
+
+  // 2. Change pre-amp gain via ApplyConfig.
+  config.capture_level_adjustment.pre_gain_factor = 1.0f;
+  config.capture_level_adjustment.post_gain_factor = 2.f;
+  const float expected_output_power2 =
+      config.capture_level_adjustment.pre_gain_factor *
+      config.capture_level_adjustment.pre_gain_factor *
+      config.capture_level_adjustment.post_gain_factor *
+      config.capture_level_adjustment.post_gain_factor * input_power;
+  apm_->ApplyConfig(config);
+
+  for (int i = 0; i < 20; ++i) {
+    frame_.CopyFrom(tmp_frame);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
+  }
+  output_power = compute_power(frame_);
+  EXPECT_NEAR(output_power, expected_output_power2, kEpsilon);
+  config = apm_->GetConfig();
+  EXPECT_EQ(config.capture_level_adjustment.pre_gain_factor, 1.0f);
+  EXPECT_EQ(config.capture_level_adjustment.post_gain_factor, 2.f);
+
+  // 3. Change pre-amp gain via a RuntimeSetting.
+  constexpr float kPreGain3 = 0.5f;
+  constexpr float kPostGain3 = 3.f;
+  const float expected_output_power3 =
+      kPreGain3 * kPreGain3 * kPostGain3 * kPostGain3 * input_power;
+
+  apm_->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCapturePreGain(kPreGain3));
+  apm_->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCapturePostGain(kPostGain3));
+
+  for (int i = 0; i < 20; ++i) {
+    frame_.CopyFrom(tmp_frame);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kIntFormat));
+  }
+  output_power = compute_power(frame_);
+  EXPECT_NEAR(output_power, expected_output_power3, kEpsilon);
+  config = apm_->GetConfig();
+  EXPECT_EQ(config.capture_level_adjustment.pre_gain_factor, 0.5f);
+  EXPECT_EQ(config.capture_level_adjustment.post_gain_factor, 3.f);
+}
+
+TEST_F(ApmTest, GainControl) {
+  AudioProcessing::Config config = apm_->GetConfig();
+  config.gain_controller1.enabled = false;
+  apm_->ApplyConfig(config);
+  config.gain_controller1.enabled = true;
+  apm_->ApplyConfig(config);
+
+  // Testing gain modes
+  for (auto mode :
+       {AudioProcessing::Config::GainController1::kAdaptiveDigital,
+        AudioProcessing::Config::GainController1::kFixedDigital,
+        AudioProcessing::Config::GainController1::kAdaptiveAnalog}) {
+    config.gain_controller1.mode = mode;
+    apm_->ApplyConfig(config);
+    apm_->set_stream_analog_level(100);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
+  }
+
+  // Testing target levels
+  for (int target_level_dbfs : {0, 15, 31}) {
+    config.gain_controller1.target_level_dbfs = target_level_dbfs;
+    apm_->ApplyConfig(config);
+    apm_->set_stream_analog_level(100);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
+  }
+
+  // Testing compression gains
+  for (int compression_gain_db : {0, 10, 90}) {
+    config.gain_controller1.compression_gain_db = compression_gain_db;
+    apm_->ApplyConfig(config);
+    apm_->set_stream_analog_level(100);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
+  }
+
+  // Testing limiter off/on
+  for (bool enable : {false, true}) {
+    config.gain_controller1.enable_limiter = enable;
+    apm_->ApplyConfig(config);
+    apm_->set_stream_analog_level(100);
+    EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
+  }
+
+  // Testing level limits.
+  constexpr int kMinLevel = 0;
+  constexpr int kMaxLevel = 255;
+  apm_->set_stream_analog_level(kMinLevel);
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
+  apm_->set_stream_analog_level((kMinLevel + kMaxLevel) / 2);
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
+  apm_->set_stream_analog_level(kMaxLevel);
+  EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(kFloatFormat));
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+using ApmDeathTest = ApmTest;
+
+TEST_F(ApmDeathTest, GainControlDiesOnTooLowTargetLevelDbfs) {
+  auto config = apm_->GetConfig();
+  config.gain_controller1.enabled = true;
+  config.gain_controller1.target_level_dbfs = -1;
+  EXPECT_DEATH(apm_->ApplyConfig(config), "");
+}
+
+TEST_F(ApmDeathTest, GainControlDiesOnTooHighTargetLevelDbfs) {
+  auto config = apm_->GetConfig();
+  config.gain_controller1.enabled = true;
+  config.gain_controller1.target_level_dbfs = 32;
+  EXPECT_DEATH(apm_->ApplyConfig(config), "");
+}
+
+TEST_F(ApmDeathTest, GainControlDiesOnTooLowCompressionGainDb) {
+  auto config = apm_->GetConfig();
+  config.gain_controller1.enabled = true;
+  config.gain_controller1.compression_gain_db = -1;
+  EXPECT_DEATH(apm_->ApplyConfig(config), "");
+}
+
+TEST_F(ApmDeathTest, GainControlDiesOnTooHighCompressionGainDb) {
+  auto config = apm_->GetConfig();
+  config.gain_controller1.enabled = true;
+  config.gain_controller1.compression_gain_db = 91;
+  EXPECT_DEATH(apm_->ApplyConfig(config), "");
+}
+
+TEST_F(ApmDeathTest, ApmDiesOnTooLowAnalogLevel) {
+  auto config = apm_->GetConfig();
+  config.gain_controller1.enabled = true;
+  apm_->ApplyConfig(config);
+  EXPECT_DEATH(apm_->set_stream_analog_level(-1), "");
+}
+
+TEST_F(ApmDeathTest, ApmDiesOnTooHighAnalogLevel) {
+  auto config = apm_->GetConfig();
+  config.gain_controller1.enabled = true;
+  apm_->ApplyConfig(config);
+  EXPECT_DEATH(apm_->set_stream_analog_level(256), "");
+}
+#endif
+
+void ApmTest::RunQuantizedVolumeDoesNotGetStuckTest(int sample_rate) {
+  Init(sample_rate, sample_rate, sample_rate, 2, 2, 2, false);
+  auto config = apm_->GetConfig();
+  config.gain_controller1.enabled = true;
+  config.gain_controller1.mode =
+      AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+  apm_->ApplyConfig(config);
+
+  int out_analog_level = 0;
+  for (int i = 0; i < 2000; ++i) {
+    ReadFrameWithRewind(near_file_, &frame_);
+    // Ensure the audio is at a low level, so the AGC will try to increase it.
+    ScaleFrame(&frame_, 0.25);
+
+    // Always pass in the same volume.
+    apm_->set_stream_analog_level(100);
+    EXPECT_EQ(apm_->kNoError,
+              apm_->ProcessStream(
+                  frame_.data.data(),
+                  StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                  StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                  frame_.data.data()));
+    out_analog_level = apm_->recommended_stream_analog_level();
+  }
+
+  // Ensure the AGC is still able to reach the maximum.
+  EXPECT_EQ(255, out_analog_level);
+}
+
+// Verifies that despite volume slider quantization, the AGC can continue to
+// increase its volume.
+TEST_F(ApmTest, QuantizedVolumeDoesNotGetStuck) {
+  for (size_t sample_rate_hz : kProcessSampleRates) {
+    SCOPED_TRACE(::testing::Message() << "sample_rate_hz=" << sample_rate_hz);
+    RunQuantizedVolumeDoesNotGetStuckTest(sample_rate_hz);
+  }
+}
+
+void ApmTest::RunManualVolumeChangeIsPossibleTest(int sample_rate) {
+  Init(sample_rate, sample_rate, sample_rate, 2, 2, 2, false);
+  auto config = apm_->GetConfig();
+  config.gain_controller1.enabled = true;
+  config.gain_controller1.mode =
+      AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+  apm_->ApplyConfig(config);
+
+  int out_analog_level = 100;
+  for (int i = 0; i < 1000; ++i) {
+    ReadFrameWithRewind(near_file_, &frame_);
+    // Ensure the audio is at a low level, so the AGC will try to increase it.
+    ScaleFrame(&frame_, 0.25);
+
+    apm_->set_stream_analog_level(out_analog_level);
+    EXPECT_EQ(apm_->kNoError,
+              apm_->ProcessStream(
+                  frame_.data.data(),
+                  StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                  StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                  frame_.data.data()));
+    out_analog_level = apm_->recommended_stream_analog_level();
+  }
+
+  // Ensure the volume was raised.
+  EXPECT_GT(out_analog_level, 100);
+  int highest_level_reached = out_analog_level;
+  // Simulate a user manual volume change.
+  out_analog_level = 100;
+
+  for (int i = 0; i < 300; ++i) {
+    ReadFrameWithRewind(near_file_, &frame_);
+    ScaleFrame(&frame_, 0.25);
+
+    apm_->set_stream_analog_level(out_analog_level);
+    EXPECT_EQ(apm_->kNoError,
+              apm_->ProcessStream(
+                  frame_.data.data(),
+                  StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                  StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                  frame_.data.data()));
+    out_analog_level = apm_->recommended_stream_analog_level();
+    // Check that AGC respected the manually adjusted volume.
+    EXPECT_LT(out_analog_level, highest_level_reached);
+  }
+  // Check that the volume was still raised.
+  EXPECT_GT(out_analog_level, 100);
+}
+
+TEST_F(ApmTest, ManualVolumeChangeIsPossible) {
+  for (size_t sample_rate_hz : kProcessSampleRates) {
+    SCOPED_TRACE(::testing::Message() << "sample_rate_hz=" << sample_rate_hz);
+    RunManualVolumeChangeIsPossibleTest(sample_rate_hz);
+  }
+}
+
+TEST_F(ApmTest, HighPassFilter) {
+  // Turn HP filter on/off
+  AudioProcessing::Config apm_config;
+  apm_config.high_pass_filter.enabled = true;
+  apm_->ApplyConfig(apm_config);
+  apm_config.high_pass_filter.enabled = false;
+  apm_->ApplyConfig(apm_config);
+}
+
+TEST_F(ApmTest, AllProcessingDisabledByDefault) {
+  AudioProcessing::Config config = apm_->GetConfig();
+  EXPECT_FALSE(config.echo_canceller.enabled);
+  EXPECT_FALSE(config.high_pass_filter.enabled);
+  EXPECT_FALSE(config.gain_controller1.enabled);
+  EXPECT_FALSE(config.noise_suppression.enabled);
+}
+
+TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabledInt) {
+  // Test that ProcessStream simply copies input to output when all components
+  // are disabled.
+  // Runs over all processing rates, and some particularly common or special
+  // rates.
+  // - 8000 Hz: lowest sample rate seen in Chrome metrics,
+  // - 22050 Hz: APM input/output frames are not exactly 10 ms,
+  // - 44100 Hz: very common desktop sample rate.
+  constexpr int kSampleRatesHz[] = {8000, 16000, 22050, 32000, 44100, 48000};
+  for (size_t sample_rate_hz : kSampleRatesHz) {
+    SCOPED_TRACE(::testing::Message() << "sample_rate_hz=" << sample_rate_hz);
+    Init(sample_rate_hz, sample_rate_hz, sample_rate_hz, 2, 2, 2, false);
+    SetFrameTo(&frame_, 1000, 2000);
+    Int16FrameData frame_copy;
+    frame_copy.CopyFrom(frame_);
+    for (int j = 0; j < 1000; j++) {
+      EXPECT_EQ(apm_->kNoError,
+                apm_->ProcessStream(
+                    frame_.data.data(),
+                    StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                    StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                    frame_.data.data()));
+      EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
+      EXPECT_EQ(apm_->kNoError,
+                apm_->ProcessReverseStream(
+                    frame_.data.data(),
+                    StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                    StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                    frame_.data.data()));
+      EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
+    }
+  }
+}
+
+TEST_F(ApmTest, NoProcessingWhenAllComponentsDisabledFloat) {
+  // Test that ProcessStream simply copies input to output when all components
+  // are disabled.
+  const size_t kSamples = 160;
+  const int sample_rate = 16000;
+  const float src[kSamples] = {-1.0f, 0.0f, 1.0f};
+  float dest[kSamples] = {};
+
+  auto src_channels = &src[0];
+  auto dest_channels = &dest[0];
+
+  apm_ = AudioProcessingBuilderForTesting().Create();
+  EXPECT_NOERR(apm_->ProcessStream(&src_channels, StreamConfig(sample_rate, 1),
+                                   StreamConfig(sample_rate, 1),
+                                   &dest_channels));
+
+  for (size_t i = 0; i < kSamples; ++i) {
+    EXPECT_EQ(src[i], dest[i]);
+  }
+
+  // Same for ProcessReverseStream.
+  float rev_dest[kSamples] = {};
+  auto rev_dest_channels = &rev_dest[0];
+
+  StreamConfig input_stream = {sample_rate, 1};
+  StreamConfig output_stream = {sample_rate, 1};
+  EXPECT_NOERR(apm_->ProcessReverseStream(&src_channels, input_stream,
+                                          output_stream, &rev_dest_channels));
+
+  for (size_t i = 0; i < kSamples; ++i) {
+    EXPECT_EQ(src[i], rev_dest[i]);
+  }
+}
+
+TEST_F(ApmTest, IdenticalInputChannelsResultInIdenticalOutputChannels) {
+  EnableAllComponents();
+
+  for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) {
+    Init(kProcessSampleRates[i], kProcessSampleRates[i], kProcessSampleRates[i],
+         2, 2, 2, false);
+    int analog_level = 127;
+    ASSERT_EQ(0, feof(far_file_));
+    ASSERT_EQ(0, feof(near_file_));
+    while (ReadFrame(far_file_, &revframe_) && ReadFrame(near_file_, &frame_)) {
+      CopyLeftToRightChannel(revframe_.data.data(),
+                             revframe_.samples_per_channel);
+
+      ASSERT_EQ(
+          kNoErr,
+          apm_->ProcessReverseStream(
+              revframe_.data.data(),
+              StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+              StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+              revframe_.data.data()));
+
+      CopyLeftToRightChannel(frame_.data.data(), frame_.samples_per_channel);
+
+      ASSERT_EQ(kNoErr, apm_->set_stream_delay_ms(0));
+      apm_->set_stream_analog_level(analog_level);
+      ASSERT_EQ(kNoErr,
+                apm_->ProcessStream(
+                    frame_.data.data(),
+                    StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                    StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                    frame_.data.data()));
+      analog_level = apm_->recommended_stream_analog_level();
+
+      VerifyChannelsAreEqual(frame_.data.data(), frame_.samples_per_channel);
+    }
+    rewind(far_file_);
+    rewind(near_file_);
+  }
+}
+
+TEST_F(ApmTest, SplittingFilter) {
+  // Verify the filter is not active through undistorted audio when:
+  // 1. No components are enabled...
+  SetFrameTo(&frame_, 1000);
+  Int16FrameData frame_copy;
+  frame_copy.CopyFrom(frame_);
+  EXPECT_EQ(apm_->kNoError,
+            apm_->ProcessStream(
+                frame_.data.data(),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                frame_.data.data()));
+  EXPECT_EQ(apm_->kNoError,
+            apm_->ProcessStream(
+                frame_.data.data(),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                frame_.data.data()));
+  EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
+
+  // 2. Only the level estimator is enabled...
+  auto apm_config = apm_->GetConfig();
+  SetFrameTo(&frame_, 1000);
+  frame_copy.CopyFrom(frame_);
+  apm_->ApplyConfig(apm_config);
+  EXPECT_EQ(apm_->kNoError,
+            apm_->ProcessStream(
+                frame_.data.data(),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                frame_.data.data()));
+  EXPECT_EQ(apm_->kNoError,
+            apm_->ProcessStream(
+                frame_.data.data(),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                frame_.data.data()));
+  EXPECT_TRUE(FrameDataAreEqual(frame_, frame_copy));
+  apm_->ApplyConfig(apm_config);
+
+  // Check the test is valid. We should have distortion from the filter
+  // when AEC is enabled (which won't affect the audio).
+  apm_config.echo_canceller.enabled = true;
+  apm_config.echo_canceller.mobile_mode = false;
+  apm_->ApplyConfig(apm_config);
+  frame_.samples_per_channel = 320;
+  frame_.num_channels = 2;
+  frame_.sample_rate_hz = 32000;
+  SetFrameTo(&frame_, 1000);
+  frame_copy.CopyFrom(frame_);
+  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+  EXPECT_EQ(apm_->kNoError,
+            apm_->ProcessStream(
+                frame_.data.data(),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                frame_.data.data()));
+  EXPECT_FALSE(FrameDataAreEqual(frame_, frame_copy));
+}
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+void ApmTest::ProcessDebugDump(absl::string_view in_filename,
+                               absl::string_view out_filename,
+                               Format format,
+                               int max_size_bytes) {
+  TaskQueueForTest worker_queue("ApmTest_worker_queue");
+  FILE* in_file = fopen(std::string(in_filename).c_str(), "rb");
+  ASSERT_TRUE(in_file != NULL);
+  audioproc::Event event_msg;
+  bool first_init = true;
+
+  while (ReadMessageFromFile(in_file, &event_msg)) {
+    if (event_msg.type() == audioproc::Event::INIT) {
+      const audioproc::Init msg = event_msg.init();
+      int reverse_sample_rate = msg.sample_rate();
+      if (msg.has_reverse_sample_rate()) {
+        reverse_sample_rate = msg.reverse_sample_rate();
+      }
+      int output_sample_rate = msg.sample_rate();
+      if (msg.has_output_sample_rate()) {
+        output_sample_rate = msg.output_sample_rate();
+      }
+
+      Init(msg.sample_rate(), output_sample_rate, reverse_sample_rate,
+           msg.num_input_channels(), msg.num_output_channels(),
+           msg.num_reverse_channels(), false);
+      if (first_init) {
+        // AttachAecDump() writes an additional init message. Don't start
+        // recording until after the first init to avoid the extra message.
+        auto aec_dump =
+            AecDumpFactory::Create(out_filename, max_size_bytes, &worker_queue);
+        EXPECT_TRUE(aec_dump);
+        apm_->AttachAecDump(std::move(aec_dump));
+        first_init = false;
+      }
+
+    } else if (event_msg.type() == audioproc::Event::REVERSE_STREAM) {
+      const audioproc::ReverseStream msg = event_msg.reverse_stream();
+
+      if (msg.channel_size() > 0) {
+        ASSERT_EQ(revframe_.num_channels,
+                  static_cast<size_t>(msg.channel_size()));
+        for (int i = 0; i < msg.channel_size(); ++i) {
+          memcpy(revfloat_cb_->channels()[i], msg.channel(i).data(),
+                 msg.channel(i).size());
+        }
+      } else {
+        memcpy(revframe_.data.data(), msg.data().data(), msg.data().size());
+        if (format == kFloatFormat) {
+          // We're using an int16 input file; convert to float.
+          ConvertToFloat(revframe_, revfloat_cb_.get());
+        }
+      }
+      AnalyzeReverseStreamChooser(format);
+
+    } else if (event_msg.type() == audioproc::Event::STREAM) {
+      const audioproc::Stream msg = event_msg.stream();
+      // ProcessStream could have changed this for the output frame.
+      frame_.num_channels = apm_->num_input_channels();
+
+      apm_->set_stream_analog_level(msg.applied_input_volume());
+      EXPECT_NOERR(apm_->set_stream_delay_ms(msg.delay()));
+      if (msg.has_keypress()) {
+        apm_->set_stream_key_pressed(msg.keypress());
+      } else {
+        apm_->set_stream_key_pressed(true);
+      }
+
+      if (msg.input_channel_size() > 0) {
+        ASSERT_EQ(frame_.num_channels,
+                  static_cast<size_t>(msg.input_channel_size()));
+        for (int i = 0; i < msg.input_channel_size(); ++i) {
+          memcpy(float_cb_->channels()[i], msg.input_channel(i).data(),
+                 msg.input_channel(i).size());
+        }
+      } else {
+        memcpy(frame_.data.data(), msg.input_data().data(),
+               msg.input_data().size());
+        if (format == kFloatFormat) {
+          // We're using an int16 input file; convert to float.
+          ConvertToFloat(frame_, float_cb_.get());
+        }
+      }
+      ProcessStreamChooser(format);
+    }
+  }
+  apm_->DetachAecDump();
+  fclose(in_file);
+}
+
+void ApmTest::VerifyDebugDumpTest(Format format) {
+  rtc::ScopedFakeClock fake_clock;
+  const std::string in_filename = test::ResourcePath("ref03", "aecdump");
+  std::string format_string;
+  switch (format) {
+    case kIntFormat:
+      format_string = "_int";
+      break;
+    case kFloatFormat:
+      format_string = "_float";
+      break;
+  }
+  const std::string ref_filename = test::TempFilename(
+      test::OutputPath(), std::string("ref") + format_string + "_aecdump");
+  const std::string out_filename = test::TempFilename(
+      test::OutputPath(), std::string("out") + format_string + "_aecdump");
+  const std::string limited_filename = test::TempFilename(
+      test::OutputPath(), std::string("limited") + format_string + "_aecdump");
+  const size_t logging_limit_bytes = 100000;
+  // We expect at least this many bytes in the created logfile.
+  const size_t logging_expected_bytes = 95000;
+  EnableAllComponents();
+  ProcessDebugDump(in_filename, ref_filename, format, -1);
+  ProcessDebugDump(ref_filename, out_filename, format, -1);
+  ProcessDebugDump(ref_filename, limited_filename, format, logging_limit_bytes);
+
+  FILE* ref_file = fopen(ref_filename.c_str(), "rb");
+  FILE* out_file = fopen(out_filename.c_str(), "rb");
+  FILE* limited_file = fopen(limited_filename.c_str(), "rb");
+  ASSERT_TRUE(ref_file != NULL);
+  ASSERT_TRUE(out_file != NULL);
+  ASSERT_TRUE(limited_file != NULL);
+  std::unique_ptr<uint8_t[]> ref_bytes;
+  std::unique_ptr<uint8_t[]> out_bytes;
+  std::unique_ptr<uint8_t[]> limited_bytes;
+
+  size_t ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
+  size_t out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
+  size_t limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes);
+  size_t bytes_read = 0;
+  size_t bytes_read_limited = 0;
+  while (ref_size > 0 && out_size > 0) {
+    bytes_read += ref_size;
+    bytes_read_limited += limited_size;
+    EXPECT_EQ(ref_size, out_size);
+    EXPECT_GE(ref_size, limited_size);
+    EXPECT_TRUE(ExpectMessageEq(/*actual=*/{out_bytes.get(), out_size},
+                                /*expected=*/{ref_bytes.get(), ref_size}));
+    if (limited_size > 0) {
+      EXPECT_TRUE(
+          ExpectMessageEq(/*actual=*/{limited_bytes.get(), limited_size},
+                          /*expected=*/{ref_bytes.get(), ref_size}));
+    }
+    ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
+    out_size = ReadMessageBytesFromFile(out_file, &out_bytes);
+    limited_size = ReadMessageBytesFromFile(limited_file, &limited_bytes);
+  }
+  EXPECT_GT(bytes_read, 0u);
+  EXPECT_GT(bytes_read_limited, logging_expected_bytes);
+  EXPECT_LE(bytes_read_limited, logging_limit_bytes);
+  EXPECT_NE(0, feof(ref_file));
+  EXPECT_NE(0, feof(out_file));
+  EXPECT_NE(0, feof(limited_file));
+  ASSERT_EQ(0, fclose(ref_file));
+  ASSERT_EQ(0, fclose(out_file));
+  ASSERT_EQ(0, fclose(limited_file));
+  remove(ref_filename.c_str());
+  remove(out_filename.c_str());
+  remove(limited_filename.c_str());
+}
+
+TEST_F(ApmTest, VerifyDebugDumpInt) {
+  VerifyDebugDumpTest(kIntFormat);
+}
+
+TEST_F(ApmTest, VerifyDebugDumpFloat) {
+  VerifyDebugDumpTest(kFloatFormat);
+}
+#endif
+
+// TODO(andrew): expand test to verify output.
+TEST_F(ApmTest, DebugDump) {
+  TaskQueueForTest worker_queue("ApmTest_worker_queue");
+  const std::string filename =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  {
+    auto aec_dump = AecDumpFactory::Create("", -1, &worker_queue);
+    EXPECT_FALSE(aec_dump);
+  }
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  // Stopping without having started should be OK.
+  apm_->DetachAecDump();
+
+  auto aec_dump = AecDumpFactory::Create(filename, -1, &worker_queue);
+  EXPECT_TRUE(aec_dump);
+  apm_->AttachAecDump(std::move(aec_dump));
+  EXPECT_EQ(apm_->kNoError,
+            apm_->ProcessStream(
+                frame_.data.data(),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                frame_.data.data()));
+  EXPECT_EQ(apm_->kNoError,
+            apm_->ProcessReverseStream(
+                revframe_.data.data(),
+                StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+                StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+                revframe_.data.data()));
+  apm_->DetachAecDump();
+
+  // Verify the file has been written.
+  FILE* fid = fopen(filename.c_str(), "r");
+  ASSERT_TRUE(fid != NULL);
+
+  // Clean it up.
+  ASSERT_EQ(0, fclose(fid));
+  ASSERT_EQ(0, remove(filename.c_str()));
+#else
+  // Verify the file has NOT been written.
+  ASSERT_TRUE(fopen(filename.c_str(), "r") == NULL);
+#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
+}
+
+// TODO(andrew): expand test to verify output.
+TEST_F(ApmTest, DebugDumpFromFileHandle) {
+  TaskQueueForTest worker_queue("ApmTest_worker_queue");
+
+  const std::string filename =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  FileWrapper f = FileWrapper::OpenWriteOnly(filename);
+  ASSERT_TRUE(f.is_open());
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+  // Stopping without having started should be OK.
+  apm_->DetachAecDump();
+
+  auto aec_dump = AecDumpFactory::Create(std::move(f), -1, &worker_queue);
+  EXPECT_TRUE(aec_dump);
+  apm_->AttachAecDump(std::move(aec_dump));
+  EXPECT_EQ(apm_->kNoError,
+            apm_->ProcessReverseStream(
+                revframe_.data.data(),
+                StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+                StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+                revframe_.data.data()));
+  EXPECT_EQ(apm_->kNoError,
+            apm_->ProcessStream(
+                frame_.data.data(),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                frame_.data.data()));
+  apm_->DetachAecDump();
+
+  // Verify the file has been written.
+  FILE* fid = fopen(filename.c_str(), "r");
+  ASSERT_TRUE(fid != NULL);
+
+  // Clean it up.
+  ASSERT_EQ(0, fclose(fid));
+  ASSERT_EQ(0, remove(filename.c_str()));
+#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
+}
+
+// TODO(andrew): Add a test to process a few frames with different combinations
+// of enabled components.
+
+TEST_F(ApmTest, Process) {
+  GOOGLE_PROTOBUF_VERIFY_VERSION;
+  audioproc::OutputData ref_data;
+
+  if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
+    OpenFileAndReadMessage(ref_filename_, &ref_data);
+  } else {
+    const int kChannels[] = {1, 2};
+    // Write the desired tests to the protobuf reference file.
+    for (size_t i = 0; i < arraysize(kChannels); i++) {
+      for (size_t j = 0; j < arraysize(kChannels); j++) {
+        for (int sample_rate_hz : AudioProcessing::kNativeSampleRatesHz) {
+          audioproc::Test* test = ref_data.add_test();
+          test->set_num_reverse_channels(kChannels[i]);
+          test->set_num_input_channels(kChannels[j]);
+          test->set_num_output_channels(kChannels[j]);
+          test->set_sample_rate(sample_rate_hz);
+          test->set_use_aec_extended_filter(false);
+        }
+      }
+    }
+#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+    // To test the extended filter mode.
+    audioproc::Test* test = ref_data.add_test();
+    test->set_num_reverse_channels(2);
+    test->set_num_input_channels(2);
+    test->set_num_output_channels(2);
+    test->set_sample_rate(AudioProcessing::kSampleRate32kHz);
+    test->set_use_aec_extended_filter(true);
+#endif
+  }
+
+  for (int i = 0; i < ref_data.test_size(); i++) {
+    printf("Running test %d of %d...\n", i + 1, ref_data.test_size());
+
+    audioproc::Test* test = ref_data.mutable_test(i);
+    // TODO(ajm): We no longer allow different input and output channels. Skip
+    // these tests for now, but they should be removed from the set.
+    if (test->num_input_channels() != test->num_output_channels())
+      continue;
+
+    apm_ = AudioProcessingBuilderForTesting()
+               .SetEchoDetector(CreateEchoDetector())
+               .Create();
+    AudioProcessing::Config apm_config = apm_->GetConfig();
+    apm_config.gain_controller1.analog_gain_controller.enabled = false;
+    apm_->ApplyConfig(apm_config);
+
+    EnableAllComponents();
+
+    Init(test->sample_rate(), test->sample_rate(), test->sample_rate(),
+         static_cast<size_t>(test->num_input_channels()),
+         static_cast<size_t>(test->num_output_channels()),
+         static_cast<size_t>(test->num_reverse_channels()), true);
+
+    int frame_count = 0;
+    int analog_level = 127;
+    int analog_level_average = 0;
+    int max_output_average = 0;
+#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+    int stats_index = 0;
+#endif
+
+    while (ReadFrame(far_file_, &revframe_) && ReadFrame(near_file_, &frame_)) {
+      EXPECT_EQ(
+          apm_->kNoError,
+          apm_->ProcessReverseStream(
+              revframe_.data.data(),
+              StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+              StreamConfig(revframe_.sample_rate_hz, revframe_.num_channels),
+              revframe_.data.data()));
+
+      EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
+      apm_->set_stream_analog_level(analog_level);
+
+      EXPECT_EQ(apm_->kNoError,
+                apm_->ProcessStream(
+                    frame_.data.data(),
+                    StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                    StreamConfig(frame_.sample_rate_hz, frame_.num_channels),
+                    frame_.data.data()));
+
+      // Ensure the frame was downmixed properly.
+      EXPECT_EQ(static_cast<size_t>(test->num_output_channels()),
+                frame_.num_channels);
+
+      max_output_average += MaxAudioFrame(frame_);
+
+      analog_level = apm_->recommended_stream_analog_level();
+      analog_level_average += analog_level;
+      AudioProcessingStats stats = apm_->GetStatistics();
+
+      size_t frame_size = frame_.samples_per_channel * frame_.num_channels;
+      size_t write_count =
+          fwrite(frame_.data.data(), sizeof(int16_t), frame_size, out_file_);
+      ASSERT_EQ(frame_size, write_count);
+
+      // Reset in case of downmixing.
+      frame_.num_channels = static_cast<size_t>(test->num_input_channels());
+      frame_count++;
+
+#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+      const int kStatsAggregationFrameNum = 100;  // 1 second.
+      if (frame_count % kStatsAggregationFrameNum == 0) {
+        // Get echo and delay metrics.
+        AudioProcessingStats stats2 = apm_->GetStatistics();
+
+        // Echo metrics.
+        const float echo_return_loss = stats2.echo_return_loss.value_or(-1.0f);
+        const float echo_return_loss_enhancement =
+            stats2.echo_return_loss_enhancement.value_or(-1.0f);
+        const float residual_echo_likelihood =
+            stats2.residual_echo_likelihood.value_or(-1.0f);
+        const float residual_echo_likelihood_recent_max =
+            stats2.residual_echo_likelihood_recent_max.value_or(-1.0f);
+
+        if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
+          const audioproc::Test::EchoMetrics& reference =
+              test->echo_metrics(stats_index);
+          constexpr float kEpsilon = 0.01;
+          EXPECT_NEAR(echo_return_loss, reference.echo_return_loss(), kEpsilon);
+          EXPECT_NEAR(echo_return_loss_enhancement,
+                      reference.echo_return_loss_enhancement(), kEpsilon);
+          EXPECT_NEAR(residual_echo_likelihood,
+                      reference.residual_echo_likelihood(), kEpsilon);
+          EXPECT_NEAR(residual_echo_likelihood_recent_max,
+                      reference.residual_echo_likelihood_recent_max(),
+                      kEpsilon);
+          ++stats_index;
+        } else {
+          audioproc::Test::EchoMetrics* message_echo = test->add_echo_metrics();
+          message_echo->set_echo_return_loss(echo_return_loss);
+          message_echo->set_echo_return_loss_enhancement(
+              echo_return_loss_enhancement);
+          message_echo->set_residual_echo_likelihood(residual_echo_likelihood);
+          message_echo->set_residual_echo_likelihood_recent_max(
+              residual_echo_likelihood_recent_max);
+        }
+      }
+#endif  // defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE).
+    }
+    max_output_average /= frame_count;
+    analog_level_average /= frame_count;
+
+    if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
+      const int kIntNear = 1;
+      // All numbers being consistently higher on N7 compare to the reference
+      // data.
+      // TODO(bjornv): If we start getting more of these offsets on Android we
+      // should consider a different approach. Either using one slack for all,
+      // or generate a separate android reference.
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+      const int kMaxOutputAverageOffset = 9;
+      const int kMaxOutputAverageNear = 26;
+#else
+      const int kMaxOutputAverageOffset = 0;
+      const int kMaxOutputAverageNear = kIntNear;
+#endif
+      EXPECT_NEAR(test->analog_level_average(), analog_level_average, kIntNear);
+      EXPECT_NEAR(test->max_output_average(),
+                  max_output_average - kMaxOutputAverageOffset,
+                  kMaxOutputAverageNear);
+    } else {
+      test->set_analog_level_average(analog_level_average);
+      test->set_max_output_average(max_output_average);
+    }
+
+    rewind(far_file_);
+    rewind(near_file_);
+  }
+
+  if (absl::GetFlag(FLAGS_write_apm_ref_data)) {
+    OpenFileAndWriteMessage(ref_filename_, ref_data);
+  }
+}
+
+// Compares the reference and test arrays over a region around the expected
+// delay. Finds the highest SNR in that region and adds the variance and squared
+// error results to the supplied accumulators.
+void UpdateBestSNR(const float* ref,
+                   const float* test,
+                   size_t length,
+                   int expected_delay,
+                   double* variance_acc,
+                   double* sq_error_acc) {
+  RTC_CHECK_LT(expected_delay, length)
+      << "delay greater than signal length, cannot compute SNR";
+  double best_snr = std::numeric_limits<double>::min();
+  double best_variance = 0;
+  double best_sq_error = 0;
+  // Search over a region of nine samples around the expected delay.
+  for (int delay = std::max(expected_delay - 4, 0); delay <= expected_delay + 4;
+       ++delay) {
+    double sq_error = 0;
+    double variance = 0;
+    for (size_t i = 0; i < length - delay; ++i) {
+      double error = test[i + delay] - ref[i];
+      sq_error += error * error;
+      variance += ref[i] * ref[i];
+    }
+
+    if (sq_error == 0) {
+      *variance_acc += variance;
+      return;
+    }
+    double snr = variance / sq_error;
+    if (snr > best_snr) {
+      best_snr = snr;
+      best_variance = variance;
+      best_sq_error = sq_error;
+    }
+  }
+
+  *variance_acc += best_variance;
+  *sq_error_acc += best_sq_error;
+}
+
+// Used to test a multitude of sample rate and channel combinations. It works
+// by first producing a set of reference files (in SetUpTestCase) that are
+// assumed to be correct, as the used parameters are verified by other tests
+// in this collection. Primarily the reference files are all produced at
+// "native" rates which do not involve any resampling.
+
+// Each test pass produces an output file with a particular format. The output
+// is matched against the reference file closest to its internal processing
+// format. If necessary the output is resampled back to its process format.
+// Due to the resampling distortion, we don't expect identical results, but
+// enforce SNR thresholds which vary depending on the format. 0 is a special
+// case SNR which corresponds to inf, or zero error.
+typedef std::tuple<int, int, int, int, double, double> AudioProcessingTestData;
+class AudioProcessingTest
+    : public ::testing::TestWithParam<AudioProcessingTestData> {
+ public:
+  AudioProcessingTest()
+      : input_rate_(std::get<0>(GetParam())),
+        output_rate_(std::get<1>(GetParam())),
+        reverse_input_rate_(std::get<2>(GetParam())),
+        reverse_output_rate_(std::get<3>(GetParam())),
+        expected_snr_(std::get<4>(GetParam())),
+        expected_reverse_snr_(std::get<5>(GetParam())) {}
+
+  virtual ~AudioProcessingTest() {}
+
+  static void SetUpTestSuite() {
+    // Create all needed output reference files.
+    const size_t kNumChannels[] = {1, 2};
+    for (size_t i = 0; i < arraysize(kProcessSampleRates); ++i) {
+      for (size_t j = 0; j < arraysize(kNumChannels); ++j) {
+        for (size_t k = 0; k < arraysize(kNumChannels); ++k) {
+          // The reference files always have matching input and output channels.
+          ProcessFormat(kProcessSampleRates[i], kProcessSampleRates[i],
+                        kProcessSampleRates[i], kProcessSampleRates[i],
+                        kNumChannels[j], kNumChannels[j], kNumChannels[k],
+                        kNumChannels[k], "ref");
+        }
+      }
+    }
+  }
+
+  void TearDown() {
+    // Remove "out" files after each test.
+    ClearTempOutFiles();
+  }
+
+  static void TearDownTestSuite() { ClearTempFiles(); }
+
+  // Runs a process pass on files with the given parameters and dumps the output
+  // to a file specified with `output_file_prefix`. Both forward and reverse
+  // output streams are dumped.
+  static void ProcessFormat(int input_rate,
+                            int output_rate,
+                            int reverse_input_rate,
+                            int reverse_output_rate,
+                            size_t num_input_channels,
+                            size_t num_output_channels,
+                            size_t num_reverse_input_channels,
+                            size_t num_reverse_output_channels,
+                            absl::string_view output_file_prefix) {
+    AudioProcessing::Config apm_config;
+    apm_config.gain_controller1.analog_gain_controller.enabled = false;
+    rtc::scoped_refptr<AudioProcessing> ap =
+        AudioProcessingBuilderForTesting().SetConfig(apm_config).Create();
+
+    EnableAllAPComponents(ap.get());
+
+    ProcessingConfig processing_config = {
+        {{input_rate, num_input_channels},
+         {output_rate, num_output_channels},
+         {reverse_input_rate, num_reverse_input_channels},
+         {reverse_output_rate, num_reverse_output_channels}}};
+    ap->Initialize(processing_config);
+
+    FILE* far_file =
+        fopen(ResourceFilePath("far", reverse_input_rate).c_str(), "rb");
+    FILE* near_file = fopen(ResourceFilePath("near", input_rate).c_str(), "rb");
+    FILE* out_file = fopen(
+        OutputFilePath(
+            output_file_prefix, input_rate, output_rate, reverse_input_rate,
+            reverse_output_rate, num_input_channels, num_output_channels,
+            num_reverse_input_channels, num_reverse_output_channels, kForward)
+            .c_str(),
+        "wb");
+    FILE* rev_out_file = fopen(
+        OutputFilePath(
+            output_file_prefix, input_rate, output_rate, reverse_input_rate,
+            reverse_output_rate, num_input_channels, num_output_channels,
+            num_reverse_input_channels, num_reverse_output_channels, kReverse)
+            .c_str(),
+        "wb");
+    ASSERT_TRUE(far_file != NULL);
+    ASSERT_TRUE(near_file != NULL);
+    ASSERT_TRUE(out_file != NULL);
+    ASSERT_TRUE(rev_out_file != NULL);
+
+    ChannelBuffer<float> fwd_cb(AudioProcessing::GetFrameSize(input_rate),
+                                num_input_channels);
+    ChannelBuffer<float> rev_cb(
+        AudioProcessing::GetFrameSize(reverse_input_rate),
+        num_reverse_input_channels);
+    ChannelBuffer<float> out_cb(AudioProcessing::GetFrameSize(output_rate),
+                                num_output_channels);
+    ChannelBuffer<float> rev_out_cb(
+        AudioProcessing::GetFrameSize(reverse_output_rate),
+        num_reverse_output_channels);
+
+    // Temporary buffers.
+    const int max_length =
+        2 * std::max(std::max(out_cb.num_frames(), rev_out_cb.num_frames()),
+                     std::max(fwd_cb.num_frames(), rev_cb.num_frames()));
+    std::unique_ptr<float[]> float_data(new float[max_length]);
+    std::unique_ptr<int16_t[]> int_data(new int16_t[max_length]);
+
+    int analog_level = 127;
+    while (ReadChunk(far_file, int_data.get(), float_data.get(), &rev_cb) &&
+           ReadChunk(near_file, int_data.get(), float_data.get(), &fwd_cb)) {
+      EXPECT_NOERR(ap->ProcessReverseStream(
+          rev_cb.channels(), processing_config.reverse_input_stream(),
+          processing_config.reverse_output_stream(), rev_out_cb.channels()));
+
+      EXPECT_NOERR(ap->set_stream_delay_ms(0));
+      ap->set_stream_analog_level(analog_level);
+
+      EXPECT_NOERR(ap->ProcessStream(
+          fwd_cb.channels(), StreamConfig(input_rate, num_input_channels),
+          StreamConfig(output_rate, num_output_channels), out_cb.channels()));
+
+      // Dump forward output to file.
+      Interleave(out_cb.channels(), out_cb.num_frames(), out_cb.num_channels(),
+                 float_data.get());
+      size_t out_length = out_cb.num_channels() * out_cb.num_frames();
+
+      ASSERT_EQ(out_length, fwrite(float_data.get(), sizeof(float_data[0]),
+                                   out_length, out_file));
+
+      // Dump reverse output to file.
+      Interleave(rev_out_cb.channels(), rev_out_cb.num_frames(),
+                 rev_out_cb.num_channels(), float_data.get());
+      size_t rev_out_length =
+          rev_out_cb.num_channels() * rev_out_cb.num_frames();
+
+      ASSERT_EQ(rev_out_length, fwrite(float_data.get(), sizeof(float_data[0]),
+                                       rev_out_length, rev_out_file));
+
+      analog_level = ap->recommended_stream_analog_level();
+    }
+    fclose(far_file);
+    fclose(near_file);
+    fclose(out_file);
+    fclose(rev_out_file);
+  }
+
+ protected:
+  int input_rate_;
+  int output_rate_;
+  int reverse_input_rate_;
+  int reverse_output_rate_;
+  double expected_snr_;
+  double expected_reverse_snr_;
+};
+
+TEST_P(AudioProcessingTest, Formats) {
+  struct ChannelFormat {
+    int num_input;
+    int num_output;
+    int num_reverse_input;
+    int num_reverse_output;
+  };
+  ChannelFormat cf[] = {
+      {1, 1, 1, 1}, {1, 1, 2, 1}, {2, 1, 1, 1},
+      {2, 1, 2, 1}, {2, 2, 1, 1}, {2, 2, 2, 2},
+  };
+
+  for (size_t i = 0; i < arraysize(cf); ++i) {
+    ProcessFormat(input_rate_, output_rate_, reverse_input_rate_,
+                  reverse_output_rate_, cf[i].num_input, cf[i].num_output,
+                  cf[i].num_reverse_input, cf[i].num_reverse_output, "out");
+
+    // Verify output for both directions.
+    std::vector<StreamDirection> stream_directions;
+    stream_directions.push_back(kForward);
+    stream_directions.push_back(kReverse);
+    for (StreamDirection file_direction : stream_directions) {
+      const int in_rate = file_direction ? reverse_input_rate_ : input_rate_;
+      const int out_rate = file_direction ? reverse_output_rate_ : output_rate_;
+      const int out_num =
+          file_direction ? cf[i].num_reverse_output : cf[i].num_output;
+      const double expected_snr =
+          file_direction ? expected_reverse_snr_ : expected_snr_;
+
+      const int min_ref_rate = std::min(in_rate, out_rate);
+      int ref_rate;
+      if (min_ref_rate > 32000) {
+        ref_rate = 48000;
+      } else if (min_ref_rate > 16000) {
+        ref_rate = 32000;
+      } else {
+        ref_rate = 16000;
+      }
+
+      FILE* out_file = fopen(
+          OutputFilePath("out", input_rate_, output_rate_, reverse_input_rate_,
+                         reverse_output_rate_, cf[i].num_input,
+                         cf[i].num_output, cf[i].num_reverse_input,
+                         cf[i].num_reverse_output, file_direction)
+              .c_str(),
+          "rb");
+      // The reference files always have matching input and output channels.
+      FILE* ref_file =
+          fopen(OutputFilePath("ref", ref_rate, ref_rate, ref_rate, ref_rate,
+                               cf[i].num_output, cf[i].num_output,
+                               cf[i].num_reverse_output,
+                               cf[i].num_reverse_output, file_direction)
+                    .c_str(),
+                "rb");
+      ASSERT_TRUE(out_file != NULL);
+      ASSERT_TRUE(ref_file != NULL);
+
+      const size_t ref_length =
+          AudioProcessing::GetFrameSize(ref_rate) * out_num;
+      const size_t out_length =
+          AudioProcessing::GetFrameSize(out_rate) * out_num;
+      // Data from the reference file.
+      std::unique_ptr<float[]> ref_data(new float[ref_length]);
+      // Data from the output file.
+      std::unique_ptr<float[]> out_data(new float[out_length]);
+      // Data from the resampled output, in case the reference and output rates
+      // don't match.
+      std::unique_ptr<float[]> cmp_data(new float[ref_length]);
+
+      PushResampler<float> resampler;
+      resampler.InitializeIfNeeded(out_rate, ref_rate, out_num);
+
+      // Compute the resampling delay of the output relative to the reference,
+      // to find the region over which we should search for the best SNR.
+      float expected_delay_sec = 0;
+      if (in_rate != ref_rate) {
+        // Input resampling delay.
+        expected_delay_sec +=
+            PushSincResampler::AlgorithmicDelaySeconds(in_rate);
+      }
+      if (out_rate != ref_rate) {
+        // Output resampling delay.
+        expected_delay_sec +=
+            PushSincResampler::AlgorithmicDelaySeconds(ref_rate);
+        // Delay of converting the output back to its processing rate for
+        // testing.
+        expected_delay_sec +=
+            PushSincResampler::AlgorithmicDelaySeconds(out_rate);
+      }
+      // The delay is multiplied by the number of channels because
+      // UpdateBestSNR() computes the SNR over interleaved data without taking
+      // channels into account.
+      int expected_delay =
+          std::floor(expected_delay_sec * ref_rate + 0.5f) * out_num;
+
+      double variance = 0;
+      double sq_error = 0;
+      while (fread(out_data.get(), sizeof(out_data[0]), out_length, out_file) &&
+             fread(ref_data.get(), sizeof(ref_data[0]), ref_length, ref_file)) {
+        float* out_ptr = out_data.get();
+        if (out_rate != ref_rate) {
+          // Resample the output back to its internal processing rate if
+          // necessary.
+          ASSERT_EQ(ref_length,
+                    static_cast<size_t>(resampler.Resample(
+                        out_ptr, out_length, cmp_data.get(), ref_length)));
+          out_ptr = cmp_data.get();
+        }
+
+        // Update the `sq_error` and `variance` accumulators with the highest
+        // SNR of reference vs output.
+        UpdateBestSNR(ref_data.get(), out_ptr, ref_length, expected_delay,
+                      &variance, &sq_error);
+      }
+
+      std::cout << "(" << input_rate_ << ", " << output_rate_ << ", "
+                << reverse_input_rate_ << ", " << reverse_output_rate_ << ", "
+                << cf[i].num_input << ", " << cf[i].num_output << ", "
+                << cf[i].num_reverse_input << ", " << cf[i].num_reverse_output
+                << ", " << file_direction << "): ";
+      if (sq_error > 0) {
+        double snr = 10 * log10(variance / sq_error);
+        EXPECT_GE(snr, expected_snr);
+        EXPECT_NE(0, expected_snr);
+        std::cout << "SNR=" << snr << " dB" << std::endl;
+      } else {
+        std::cout << "SNR=inf dB" << std::endl;
+      }
+
+      fclose(out_file);
+      fclose(ref_file);
+    }
+  }
+}
+
+#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
+INSTANTIATE_TEST_SUITE_P(
+    CommonFormats,
+    AudioProcessingTest,
+    // Internal processing rates and the particularly common sample rate 44100
+    // Hz are tested in a grid of combinations (capture in, render in, out).
+    ::testing::Values(std::make_tuple(48000, 48000, 48000, 48000, 0, 0),
+                      std::make_tuple(48000, 48000, 32000, 48000, 40, 30),
+                      std::make_tuple(48000, 48000, 16000, 48000, 40, 20),
+                      std::make_tuple(48000, 44100, 48000, 44100, 20, 20),
+                      std::make_tuple(48000, 44100, 32000, 44100, 20, 15),
+                      std::make_tuple(48000, 44100, 16000, 44100, 20, 15),
+                      std::make_tuple(48000, 32000, 48000, 32000, 30, 35),
+                      std::make_tuple(48000, 32000, 32000, 32000, 30, 0),
+                      std::make_tuple(48000, 32000, 16000, 32000, 30, 20),
+                      std::make_tuple(48000, 16000, 48000, 16000, 25, 20),
+                      std::make_tuple(48000, 16000, 32000, 16000, 25, 20),
+                      std::make_tuple(48000, 16000, 16000, 16000, 25, 0),
+
+                      std::make_tuple(44100, 48000, 48000, 48000, 30, 0),
+                      std::make_tuple(44100, 48000, 32000, 48000, 30, 30),
+                      std::make_tuple(44100, 48000, 16000, 48000, 30, 20),
+                      std::make_tuple(44100, 44100, 48000, 44100, 20, 20),
+                      std::make_tuple(44100, 44100, 32000, 44100, 20, 15),
+                      std::make_tuple(44100, 44100, 16000, 44100, 20, 15),
+                      std::make_tuple(44100, 32000, 48000, 32000, 30, 35),
+                      std::make_tuple(44100, 32000, 32000, 32000, 30, 0),
+                      std::make_tuple(44100, 32000, 16000, 32000, 30, 20),
+                      std::make_tuple(44100, 16000, 48000, 16000, 25, 20),
+                      std::make_tuple(44100, 16000, 32000, 16000, 25, 20),
+                      std::make_tuple(44100, 16000, 16000, 16000, 25, 0),
+
+                      std::make_tuple(32000, 48000, 48000, 48000, 15, 0),
+                      std::make_tuple(32000, 48000, 32000, 48000, 15, 30),
+                      std::make_tuple(32000, 48000, 16000, 48000, 15, 20),
+                      std::make_tuple(32000, 44100, 48000, 44100, 19, 20),
+                      std::make_tuple(32000, 44100, 32000, 44100, 19, 15),
+                      std::make_tuple(32000, 44100, 16000, 44100, 19, 15),
+                      std::make_tuple(32000, 32000, 48000, 32000, 40, 35),
+                      std::make_tuple(32000, 32000, 32000, 32000, 0, 0),
+                      std::make_tuple(32000, 32000, 16000, 32000, 39, 20),
+                      std::make_tuple(32000, 16000, 48000, 16000, 25, 20),
+                      std::make_tuple(32000, 16000, 32000, 16000, 25, 20),
+                      std::make_tuple(32000, 16000, 16000, 16000, 25, 0),
+
+                      std::make_tuple(16000, 48000, 48000, 48000, 9, 0),
+                      std::make_tuple(16000, 48000, 32000, 48000, 9, 30),
+                      std::make_tuple(16000, 48000, 16000, 48000, 9, 20),
+                      std::make_tuple(16000, 44100, 48000, 44100, 15, 20),
+                      std::make_tuple(16000, 44100, 32000, 44100, 15, 15),
+                      std::make_tuple(16000, 44100, 16000, 44100, 15, 15),
+                      std::make_tuple(16000, 32000, 48000, 32000, 25, 35),
+                      std::make_tuple(16000, 32000, 32000, 32000, 25, 0),
+                      std::make_tuple(16000, 32000, 16000, 32000, 25, 20),
+                      std::make_tuple(16000, 16000, 48000, 16000, 39, 20),
+                      std::make_tuple(16000, 16000, 32000, 16000, 39, 20),
+                      std::make_tuple(16000, 16000, 16000, 16000, 0, 0),
+
+                      // Other sample rates are not tested exhaustively, to keep
+                      // the test runtime manageable.
+                      //
+                      // Testing most other sample rates logged by Chrome UMA:
+                      //  - WebRTC.AudioInputSampleRate
+                      //  - WebRTC.AudioOutputSampleRate
+                      // ApmConfiguration.HandlingOfRateCombinations covers
+                      // remaining sample rates.
+                      std::make_tuple(192000, 192000, 48000, 192000, 20, 40),
+                      std::make_tuple(176400, 176400, 48000, 176400, 20, 35),
+                      std::make_tuple(96000, 96000, 48000, 96000, 20, 40),
+                      std::make_tuple(88200, 88200, 48000, 88200, 20, 20),
+                      std::make_tuple(44100, 44100, 48000, 44100, 20, 20)));
+
+#elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
+INSTANTIATE_TEST_SUITE_P(
+    CommonFormats,
+    AudioProcessingTest,
+    ::testing::Values(std::make_tuple(48000, 48000, 48000, 48000, 19, 0),
+                      std::make_tuple(48000, 48000, 32000, 48000, 19, 30),
+                      std::make_tuple(48000, 48000, 16000, 48000, 19, 20),
+                      std::make_tuple(48000, 44100, 48000, 44100, 15, 20),
+                      std::make_tuple(48000, 44100, 32000, 44100, 15, 15),
+                      std::make_tuple(48000, 44100, 16000, 44100, 15, 15),
+                      std::make_tuple(48000, 32000, 48000, 32000, 19, 35),
+                      std::make_tuple(48000, 32000, 32000, 32000, 19, 0),
+                      std::make_tuple(48000, 32000, 16000, 32000, 19, 20),
+                      std::make_tuple(48000, 16000, 48000, 16000, 20, 20),
+                      std::make_tuple(48000, 16000, 32000, 16000, 20, 20),
+                      std::make_tuple(48000, 16000, 16000, 16000, 20, 0),
+
+                      std::make_tuple(44100, 48000, 48000, 48000, 15, 0),
+                      std::make_tuple(44100, 48000, 32000, 48000, 15, 30),
+                      std::make_tuple(44100, 48000, 16000, 48000, 15, 20),
+                      std::make_tuple(44100, 44100, 48000, 44100, 15, 20),
+                      std::make_tuple(44100, 44100, 32000, 44100, 15, 15),
+                      std::make_tuple(44100, 44100, 16000, 44100, 15, 15),
+                      std::make_tuple(44100, 32000, 48000, 32000, 18, 35),
+                      std::make_tuple(44100, 32000, 32000, 32000, 18, 0),
+                      std::make_tuple(44100, 32000, 16000, 32000, 18, 20),
+                      std::make_tuple(44100, 16000, 48000, 16000, 19, 20),
+                      std::make_tuple(44100, 16000, 32000, 16000, 19, 20),
+                      std::make_tuple(44100, 16000, 16000, 16000, 19, 0),
+
+                      std::make_tuple(32000, 48000, 48000, 48000, 17, 0),
+                      std::make_tuple(32000, 48000, 32000, 48000, 17, 30),
+                      std::make_tuple(32000, 48000, 16000, 48000, 17, 20),
+                      std::make_tuple(32000, 44100, 48000, 44100, 20, 20),
+                      std::make_tuple(32000, 44100, 32000, 44100, 20, 15),
+                      std::make_tuple(32000, 44100, 16000, 44100, 20, 15),
+                      std::make_tuple(32000, 32000, 48000, 32000, 27, 35),
+                      std::make_tuple(32000, 32000, 32000, 32000, 0, 0),
+                      std::make_tuple(32000, 32000, 16000, 32000, 30, 20),
+                      std::make_tuple(32000, 16000, 48000, 16000, 20, 20),
+                      std::make_tuple(32000, 16000, 32000, 16000, 20, 20),
+                      std::make_tuple(32000, 16000, 16000, 16000, 20, 0),
+
+                      std::make_tuple(16000, 48000, 48000, 48000, 11, 0),
+                      std::make_tuple(16000, 48000, 32000, 48000, 11, 30),
+                      std::make_tuple(16000, 48000, 16000, 48000, 11, 20),
+                      std::make_tuple(16000, 44100, 48000, 44100, 15, 20),
+                      std::make_tuple(16000, 44100, 32000, 44100, 15, 15),
+                      std::make_tuple(16000, 44100, 16000, 44100, 15, 15),
+                      std::make_tuple(16000, 32000, 48000, 32000, 24, 35),
+                      std::make_tuple(16000, 32000, 32000, 32000, 24, 0),
+                      std::make_tuple(16000, 32000, 16000, 32000, 25, 20),
+                      std::make_tuple(16000, 16000, 48000, 16000, 28, 20),
+                      std::make_tuple(16000, 16000, 32000, 16000, 28, 20),
+                      std::make_tuple(16000, 16000, 16000, 16000, 0, 0),
+
+                      std::make_tuple(192000, 192000, 48000, 192000, 20, 40),
+                      std::make_tuple(176400, 176400, 48000, 176400, 20, 35),
+                      std::make_tuple(96000, 96000, 48000, 96000, 20, 40),
+                      std::make_tuple(88200, 88200, 48000, 88200, 20, 20),
+                      std::make_tuple(44100, 44100, 48000, 44100, 20, 20)));
+#endif
+
+// Produces a scoped trace debug output.
+std::string ProduceDebugText(int render_input_sample_rate_hz,
+                             int render_output_sample_rate_hz,
+                             int capture_input_sample_rate_hz,
+                             int capture_output_sample_rate_hz,
+                             size_t render_input_num_channels,
+                             size_t render_output_num_channels,
+                             size_t capture_input_num_channels,
+                             size_t capture_output_num_channels) {
+  rtc::StringBuilder ss;
+  ss << "Sample rates:"
+        "\n Render input: "
+     << render_input_sample_rate_hz
+     << " Hz"
+        "\n Render output: "
+     << render_output_sample_rate_hz
+     << " Hz"
+        "\n Capture input: "
+     << capture_input_sample_rate_hz
+     << " Hz"
+        "\n Capture output: "
+     << capture_output_sample_rate_hz
+     << " Hz"
+        "\nNumber of channels:"
+        "\n Render input: "
+     << render_input_num_channels
+     << "\n Render output: " << render_output_num_channels
+     << "\n Capture input: " << capture_input_num_channels
+     << "\n Capture output: " << capture_output_num_channels;
+  return ss.Release();
+}
+
+// Validates that running the audio processing module using various combinations
+// of sample rates and number of channels works as intended.
+void RunApmRateAndChannelTest(
+    rtc::ArrayView<const int> sample_rates_hz,
+    rtc::ArrayView<const int> render_channel_counts,
+    rtc::ArrayView<const int> capture_channel_counts) {
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.pipeline.multi_channel_render = true;
+  apm_config.pipeline.multi_channel_capture = true;
+  apm_config.echo_canceller.enabled = true;
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting().SetConfig(apm_config).Create();
+
+  StreamConfig render_input_stream_config;
+  StreamConfig render_output_stream_config;
+  StreamConfig capture_input_stream_config;
+  StreamConfig capture_output_stream_config;
+
+  std::vector<float> render_input_frame_channels;
+  std::vector<float*> render_input_frame;
+  std::vector<float> render_output_frame_channels;
+  std::vector<float*> render_output_frame;
+  std::vector<float> capture_input_frame_channels;
+  std::vector<float*> capture_input_frame;
+  std::vector<float> capture_output_frame_channels;
+  std::vector<float*> capture_output_frame;
+
+  for (auto render_input_sample_rate_hz : sample_rates_hz) {
+    for (auto render_output_sample_rate_hz : sample_rates_hz) {
+      for (auto capture_input_sample_rate_hz : sample_rates_hz) {
+        for (auto capture_output_sample_rate_hz : sample_rates_hz) {
+          for (size_t render_input_num_channels : render_channel_counts) {
+            for (size_t capture_input_num_channels : capture_channel_counts) {
+              size_t render_output_num_channels = render_input_num_channels;
+              size_t capture_output_num_channels = capture_input_num_channels;
+              auto populate_audio_frame = [](int sample_rate_hz,
+                                             size_t num_channels,
+                                             StreamConfig* cfg,
+                                             std::vector<float>* channels_data,
+                                             std::vector<float*>* frame_data) {
+                cfg->set_sample_rate_hz(sample_rate_hz);
+                cfg->set_num_channels(num_channels);
+
+                size_t max_frame_size =
+                    AudioProcessing::GetFrameSize(sample_rate_hz);
+                channels_data->resize(num_channels * max_frame_size);
+                std::fill(channels_data->begin(), channels_data->end(), 0.5f);
+                frame_data->resize(num_channels);
+                for (size_t channel = 0; channel < num_channels; ++channel) {
+                  (*frame_data)[channel] =
+                      &(*channels_data)[channel * max_frame_size];
+                }
+              };
+
+              populate_audio_frame(
+                  render_input_sample_rate_hz, render_input_num_channels,
+                  &render_input_stream_config, &render_input_frame_channels,
+                  &render_input_frame);
+              populate_audio_frame(
+                  render_output_sample_rate_hz, render_output_num_channels,
+                  &render_output_stream_config, &render_output_frame_channels,
+                  &render_output_frame);
+              populate_audio_frame(
+                  capture_input_sample_rate_hz, capture_input_num_channels,
+                  &capture_input_stream_config, &capture_input_frame_channels,
+                  &capture_input_frame);
+              populate_audio_frame(
+                  capture_output_sample_rate_hz, capture_output_num_channels,
+                  &capture_output_stream_config, &capture_output_frame_channels,
+                  &capture_output_frame);
+
+              for (size_t frame = 0; frame < 2; ++frame) {
+                SCOPED_TRACE(ProduceDebugText(
+                    render_input_sample_rate_hz, render_output_sample_rate_hz,
+                    capture_input_sample_rate_hz, capture_output_sample_rate_hz,
+                    render_input_num_channels, render_output_num_channels,
+                    render_input_num_channels, capture_output_num_channels));
+
+                int result = apm->ProcessReverseStream(
+                    &render_input_frame[0], render_input_stream_config,
+                    render_output_stream_config, &render_output_frame[0]);
+                EXPECT_EQ(result, AudioProcessing::kNoError);
+                result = apm->ProcessStream(
+                    &capture_input_frame[0], capture_input_stream_config,
+                    capture_output_stream_config, &capture_output_frame[0]);
+                EXPECT_EQ(result, AudioProcessing::kNoError);
+              }
+            }
+          }
+        }
+      }
+    }
+  }
+}
+
+constexpr void Toggle(bool& b) {
+  b ^= true;
+}
+
+}  // namespace
+
+TEST(RuntimeSettingTest, TestDefaultCtor) {
+  auto s = AudioProcessing::RuntimeSetting();
+  EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
+}
+
+TEST(RuntimeSettingTest, TestUsageWithSwapQueue) {
+  SwapQueue<AudioProcessing::RuntimeSetting> q(1);
+  auto s = AudioProcessing::RuntimeSetting();
+  ASSERT_TRUE(q.Insert(&s));
+  ASSERT_TRUE(q.Remove(&s));
+  EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
+}
+
+TEST(ApmConfiguration, EnablePostProcessing) {
+  // Verify that apm uses a capture post processing module if one is provided.
+  auto mock_post_processor_ptr =
+      new ::testing::NiceMock<test::MockCustomProcessing>();
+  auto mock_post_processor =
+      std::unique_ptr<CustomProcessing>(mock_post_processor_ptr);
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetCapturePostProcessing(std::move(mock_post_processor))
+          .Create();
+
+  Int16FrameData audio;
+  audio.num_channels = 1;
+  SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
+
+  EXPECT_CALL(*mock_post_processor_ptr, Process(::testing::_)).Times(1);
+  apm->ProcessStream(audio.data.data(),
+                     StreamConfig(audio.sample_rate_hz, audio.num_channels),
+                     StreamConfig(audio.sample_rate_hz, audio.num_channels),
+                     audio.data.data());
+}
+
+TEST(ApmConfiguration, EnablePreProcessing) {
+  // Verify that apm uses a capture post processing module if one is provided.
+  auto mock_pre_processor_ptr =
+      new ::testing::NiceMock<test::MockCustomProcessing>();
+  auto mock_pre_processor =
+      std::unique_ptr<CustomProcessing>(mock_pre_processor_ptr);
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetRenderPreProcessing(std::move(mock_pre_processor))
+          .Create();
+
+  Int16FrameData audio;
+  audio.num_channels = 1;
+  SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
+
+  EXPECT_CALL(*mock_pre_processor_ptr, Process(::testing::_)).Times(1);
+  apm->ProcessReverseStream(
+      audio.data.data(), StreamConfig(audio.sample_rate_hz, audio.num_channels),
+      StreamConfig(audio.sample_rate_hz, audio.num_channels),
+      audio.data.data());
+}
+
+TEST(ApmConfiguration, EnableCaptureAnalyzer) {
+  // Verify that apm uses a capture analyzer if one is provided.
+  auto mock_capture_analyzer_ptr =
+      new ::testing::NiceMock<test::MockCustomAudioAnalyzer>();
+  auto mock_capture_analyzer =
+      std::unique_ptr<CustomAudioAnalyzer>(mock_capture_analyzer_ptr);
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetCaptureAnalyzer(std::move(mock_capture_analyzer))
+          .Create();
+
+  Int16FrameData audio;
+  audio.num_channels = 1;
+  SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
+
+  EXPECT_CALL(*mock_capture_analyzer_ptr, Analyze(::testing::_)).Times(1);
+  apm->ProcessStream(audio.data.data(),
+                     StreamConfig(audio.sample_rate_hz, audio.num_channels),
+                     StreamConfig(audio.sample_rate_hz, audio.num_channels),
+                     audio.data.data());
+}
+
+TEST(ApmConfiguration, PreProcessingReceivesRuntimeSettings) {
+  auto mock_pre_processor_ptr =
+      new ::testing::NiceMock<test::MockCustomProcessing>();
+  auto mock_pre_processor =
+      std::unique_ptr<CustomProcessing>(mock_pre_processor_ptr);
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetRenderPreProcessing(std::move(mock_pre_processor))
+          .Create();
+  apm->SetRuntimeSetting(
+      AudioProcessing::RuntimeSetting::CreateCustomRenderSetting(0));
+
+  // RuntimeSettings forwarded during 'Process*Stream' calls.
+  // Therefore we have to make one such call.
+  Int16FrameData audio;
+  audio.num_channels = 1;
+  SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
+
+  EXPECT_CALL(*mock_pre_processor_ptr, SetRuntimeSetting(::testing::_))
+      .Times(1);
+  apm->ProcessReverseStream(
+      audio.data.data(), StreamConfig(audio.sample_rate_hz, audio.num_channels),
+      StreamConfig(audio.sample_rate_hz, audio.num_channels),
+      audio.data.data());
+}
+
+class MyEchoControlFactory : public EchoControlFactory {
+ public:
+  std::unique_ptr<EchoControl> Create(int sample_rate_hz) {
+    auto ec = new test::MockEchoControl();
+    EXPECT_CALL(*ec, AnalyzeRender(::testing::_)).Times(1);
+    EXPECT_CALL(*ec, AnalyzeCapture(::testing::_)).Times(2);
+    EXPECT_CALL(*ec, ProcessCapture(::testing::_, ::testing::_, ::testing::_))
+        .Times(2);
+    return std::unique_ptr<EchoControl>(ec);
+  }
+
+  std::unique_ptr<EchoControl> Create(int sample_rate_hz,
+                                      int num_render_channels,
+                                      int num_capture_channels) {
+    return Create(sample_rate_hz);
+  }
+};
+
+TEST(ApmConfiguration, EchoControlInjection) {
+  // Verify that apm uses an injected echo controller if one is provided.
+  std::unique_ptr<EchoControlFactory> echo_control_factory(
+      new MyEchoControlFactory());
+
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoControlFactory(std::move(echo_control_factory))
+          .Create();
+
+  Int16FrameData audio;
+  audio.num_channels = 1;
+  SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
+  apm->ProcessStream(audio.data.data(),
+                     StreamConfig(audio.sample_rate_hz, audio.num_channels),
+                     StreamConfig(audio.sample_rate_hz, audio.num_channels),
+                     audio.data.data());
+  apm->ProcessReverseStream(
+      audio.data.data(), StreamConfig(audio.sample_rate_hz, audio.num_channels),
+      StreamConfig(audio.sample_rate_hz, audio.num_channels),
+      audio.data.data());
+  apm->ProcessStream(audio.data.data(),
+                     StreamConfig(audio.sample_rate_hz, audio.num_channels),
+                     StreamConfig(audio.sample_rate_hz, audio.num_channels),
+                     audio.data.data());
+}
+
+TEST(ApmConfiguration, EchoDetectorInjection) {
+  using ::testing::_;
+  rtc::scoped_refptr<test::MockEchoDetector> mock_echo_detector =
+      rtc::make_ref_counted<::testing::StrictMock<test::MockEchoDetector>>();
+  EXPECT_CALL(*mock_echo_detector,
+              Initialize(/*capture_sample_rate_hz=*/16000, _,
+                         /*render_sample_rate_hz=*/16000, _))
+      .Times(1);
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoDetector(mock_echo_detector)
+          .Create();
+
+  // The echo detector is included in processing when enabled.
+  EXPECT_CALL(*mock_echo_detector, AnalyzeRenderAudio(_))
+      .WillOnce([](rtc::ArrayView<const float> render_audio) {
+        EXPECT_EQ(render_audio.size(), 160u);
+      });
+  EXPECT_CALL(*mock_echo_detector, AnalyzeCaptureAudio(_))
+      .WillOnce([](rtc::ArrayView<const float> capture_audio) {
+        EXPECT_EQ(capture_audio.size(), 160u);
+      });
+  EXPECT_CALL(*mock_echo_detector, GetMetrics()).Times(1);
+
+  Int16FrameData frame;
+  frame.num_channels = 1;
+  SetFrameSampleRate(&frame, 16000);
+
+  apm->ProcessReverseStream(frame.data.data(), StreamConfig(16000, 1),
+                            StreamConfig(16000, 1), frame.data.data());
+  apm->ProcessStream(frame.data.data(), StreamConfig(16000, 1),
+                     StreamConfig(16000, 1), frame.data.data());
+
+  // When processing rates change, the echo detector is also reinitialized to
+  // match those.
+  EXPECT_CALL(*mock_echo_detector,
+              Initialize(/*capture_sample_rate_hz=*/48000, _,
+                         /*render_sample_rate_hz=*/16000, _))
+      .Times(1);
+  EXPECT_CALL(*mock_echo_detector,
+              Initialize(/*capture_sample_rate_hz=*/48000, _,
+                         /*render_sample_rate_hz=*/48000, _))
+      .Times(1);
+  EXPECT_CALL(*mock_echo_detector, AnalyzeRenderAudio(_))
+      .WillOnce([](rtc::ArrayView<const float> render_audio) {
+        EXPECT_EQ(render_audio.size(), 480u);
+      });
+  EXPECT_CALL(*mock_echo_detector, AnalyzeCaptureAudio(_))
+      .Times(2)
+      .WillRepeatedly([](rtc::ArrayView<const float> capture_audio) {
+        EXPECT_EQ(capture_audio.size(), 480u);
+      });
+  EXPECT_CALL(*mock_echo_detector, GetMetrics()).Times(2);
+
+  SetFrameSampleRate(&frame, 48000);
+  apm->ProcessStream(frame.data.data(), StreamConfig(48000, 1),
+                     StreamConfig(48000, 1), frame.data.data());
+  apm->ProcessReverseStream(frame.data.data(), StreamConfig(48000, 1),
+                            StreamConfig(48000, 1), frame.data.data());
+  apm->ProcessStream(frame.data.data(), StreamConfig(48000, 1),
+                     StreamConfig(48000, 1), frame.data.data());
+}
+
+rtc::scoped_refptr<AudioProcessing> CreateApm(bool mobile_aec) {
+  // Enable residual echo detection, for stats.
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoDetector(CreateEchoDetector())
+          .Create();
+  if (!apm) {
+    return apm;
+  }
+
+  ProcessingConfig processing_config = {
+      {{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
+
+  if (apm->Initialize(processing_config) != 0) {
+    return nullptr;
+  }
+
+  // Disable all components except for an AEC.
+  AudioProcessing::Config apm_config;
+  apm_config.high_pass_filter.enabled = false;
+  apm_config.gain_controller1.enabled = false;
+  apm_config.gain_controller2.enabled = false;
+  apm_config.echo_canceller.enabled = true;
+  apm_config.echo_canceller.mobile_mode = mobile_aec;
+  apm_config.noise_suppression.enabled = false;
+  apm->ApplyConfig(apm_config);
+  return apm;
+}
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_MAC)
+#define MAYBE_ApmStatistics DISABLED_ApmStatistics
+#else
+#define MAYBE_ApmStatistics ApmStatistics
+#endif
+
+TEST(MAYBE_ApmStatistics, AECEnabledTest) {
+  // Set up APM with AEC3 and process some audio.
+  rtc::scoped_refptr<AudioProcessing> apm = CreateApm(false);
+  ASSERT_TRUE(apm);
+  AudioProcessing::Config apm_config;
+  apm_config.echo_canceller.enabled = true;
+  apm->ApplyConfig(apm_config);
+
+  // Set up an audioframe.
+  Int16FrameData frame;
+  frame.num_channels = 1;
+  SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate32kHz);
+
+  // Fill the audio frame with a sawtooth pattern.
+  int16_t* ptr = frame.data.data();
+  for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
+    ptr[i] = 10000 * ((i % 3) - 1);
+  }
+
+  // Do some processing.
+  for (int i = 0; i < 200; i++) {
+    EXPECT_EQ(apm->ProcessReverseStream(
+                  frame.data.data(),
+                  StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                  StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                  frame.data.data()),
+              0);
+    EXPECT_EQ(apm->set_stream_delay_ms(0), 0);
+    EXPECT_EQ(apm->ProcessStream(
+                  frame.data.data(),
+                  StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                  StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                  frame.data.data()),
+              0);
+  }
+
+  // Test statistics interface.
+  AudioProcessingStats stats = apm->GetStatistics();
+  // We expect all statistics to be set and have a sensible value.
+  ASSERT_TRUE(stats.residual_echo_likelihood.has_value());
+  EXPECT_GE(*stats.residual_echo_likelihood, 0.0);
+  EXPECT_LE(*stats.residual_echo_likelihood, 1.0);
+  ASSERT_TRUE(stats.residual_echo_likelihood_recent_max.has_value());
+  EXPECT_GE(*stats.residual_echo_likelihood_recent_max, 0.0);
+  EXPECT_LE(*stats.residual_echo_likelihood_recent_max, 1.0);
+  ASSERT_TRUE(stats.echo_return_loss.has_value());
+  EXPECT_NE(*stats.echo_return_loss, -100.0);
+  ASSERT_TRUE(stats.echo_return_loss_enhancement.has_value());
+  EXPECT_NE(*stats.echo_return_loss_enhancement, -100.0);
+}
+
+TEST(MAYBE_ApmStatistics, AECMEnabledTest) {
+  // Set up APM with AECM and process some audio.
+  rtc::scoped_refptr<AudioProcessing> apm = CreateApm(true);
+  ASSERT_TRUE(apm);
+
+  // Set up an audioframe.
+  Int16FrameData frame;
+  frame.num_channels = 1;
+  SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate32kHz);
+
+  // Fill the audio frame with a sawtooth pattern.
+  int16_t* ptr = frame.data.data();
+  for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
+    ptr[i] = 10000 * ((i % 3) - 1);
+  }
+
+  // Do some processing.
+  for (int i = 0; i < 200; i++) {
+    EXPECT_EQ(apm->ProcessReverseStream(
+                  frame.data.data(),
+                  StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                  StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                  frame.data.data()),
+              0);
+    EXPECT_EQ(apm->set_stream_delay_ms(0), 0);
+    EXPECT_EQ(apm->ProcessStream(
+                  frame.data.data(),
+                  StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                  StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                  frame.data.data()),
+              0);
+  }
+
+  // Test statistics interface.
+  AudioProcessingStats stats = apm->GetStatistics();
+  // We expect only the residual echo detector statistics to be set and have a
+  // sensible value.
+  ASSERT_TRUE(stats.residual_echo_likelihood.has_value());
+  EXPECT_GE(*stats.residual_echo_likelihood, 0.0);
+  EXPECT_LE(*stats.residual_echo_likelihood, 1.0);
+  ASSERT_TRUE(stats.residual_echo_likelihood_recent_max.has_value());
+  EXPECT_GE(*stats.residual_echo_likelihood_recent_max, 0.0);
+  EXPECT_LE(*stats.residual_echo_likelihood_recent_max, 1.0);
+  EXPECT_FALSE(stats.echo_return_loss.has_value());
+  EXPECT_FALSE(stats.echo_return_loss_enhancement.has_value());
+}
+
+TEST(ApmStatistics, DoNotReportVoiceDetectedStat) {
+  ProcessingConfig processing_config = {
+      {{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
+
+  // Set up an audioframe.
+  Int16FrameData frame;
+  frame.num_channels = 1;
+  SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate32kHz);
+
+  // Fill the audio frame with a sawtooth pattern.
+  int16_t* ptr = frame.data.data();
+  for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
+    ptr[i] = 10000 * ((i % 3) - 1);
+  }
+
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting().Create();
+  apm->Initialize(processing_config);
+
+  // No metric should be reported.
+  EXPECT_EQ(
+      apm->ProcessStream(frame.data.data(),
+                         StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                         StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                         frame.data.data()),
+      0);
+  EXPECT_FALSE(apm->GetStatistics().voice_detected.has_value());
+}
+
+TEST(ApmStatistics, GetStatisticsReportsNoEchoDetectorStatsWhenDisabled) {
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting().Create();
+  Int16FrameData frame;
+  frame.num_channels = 1;
+  SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate32kHz);
+  ASSERT_EQ(
+      apm->ProcessStream(frame.data.data(),
+                         StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                         StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                         frame.data.data()),
+      0);
+  // Echo detector is disabled by default, no stats reported.
+  AudioProcessingStats stats = apm->GetStatistics();
+  EXPECT_FALSE(stats.residual_echo_likelihood.has_value());
+  EXPECT_FALSE(stats.residual_echo_likelihood_recent_max.has_value());
+}
+
+TEST(ApmStatistics, GetStatisticsReportsEchoDetectorStatsWhenEnabled) {
+  // Create APM with an echo detector injected.
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoDetector(CreateEchoDetector())
+          .Create();
+  Int16FrameData frame;
+  frame.num_channels = 1;
+  SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate32kHz);
+  // Echo detector enabled: Report stats.
+  ASSERT_EQ(
+      apm->ProcessStream(frame.data.data(),
+                         StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                         StreamConfig(frame.sample_rate_hz, frame.num_channels),
+                         frame.data.data()),
+      0);
+  AudioProcessingStats stats = apm->GetStatistics();
+  EXPECT_TRUE(stats.residual_echo_likelihood.has_value());
+  EXPECT_TRUE(stats.residual_echo_likelihood_recent_max.has_value());
+}
+
+TEST(ApmConfiguration, HandlingOfRateAndChannelCombinations) {
+  std::array<int, 3> sample_rates_hz = {16000, 32000, 48000};
+  std::array<int, 2> render_channel_counts = {1, 7};
+  std::array<int, 2> capture_channel_counts = {1, 7};
+  RunApmRateAndChannelTest(sample_rates_hz, render_channel_counts,
+                           capture_channel_counts);
+}
+
+TEST(ApmConfiguration, HandlingOfChannelCombinations) {
+  std::array<int, 1> sample_rates_hz = {48000};
+  std::array<int, 8> render_channel_counts = {1, 2, 3, 4, 5, 6, 7, 8};
+  std::array<int, 8> capture_channel_counts = {1, 2, 3, 4, 5, 6, 7, 8};
+  RunApmRateAndChannelTest(sample_rates_hz, render_channel_counts,
+                           capture_channel_counts);
+}
+
+TEST(ApmConfiguration, HandlingOfRateCombinations) {
+  // Test rates <= 96000 logged by Chrome UMA:
+  //  - WebRTC.AudioInputSampleRate
+  //  - WebRTC.AudioOutputSampleRate
+  // Higher rates are tested in AudioProcessingTest.Format, to keep the number
+  // of combinations in this test manageable.
+  std::array<int, 9> sample_rates_hz = {8000,  11025, 16000, 22050, 32000,
+                                        44100, 48000, 88200, 96000};
+  std::array<int, 1> render_channel_counts = {2};
+  std::array<int, 1> capture_channel_counts = {2};
+  RunApmRateAndChannelTest(sample_rates_hz, render_channel_counts,
+                           capture_channel_counts);
+}
+
+TEST(ApmConfiguration, SelfAssignment) {
+  // At some point memory sanitizer was complaining about self-assigment.
+  // Make sure we don't regress.
+  AudioProcessing::Config config;
+  AudioProcessing::Config* config2 = &config;
+  *config2 = *config2;  // Workaround -Wself-assign-overloaded
+  SUCCEED();  // Real success is absence of defects from asan/msan/ubsan.
+}
+
+TEST(AudioProcessing, GainController1ConfigEqual) {
+  AudioProcessing::Config::GainController1 a;
+  AudioProcessing::Config::GainController1 b;
+  EXPECT_EQ(a, b);
+
+  Toggle(a.enabled);
+  b.enabled = a.enabled;
+  EXPECT_EQ(a, b);
+
+  a.mode = AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital;
+  b.mode = a.mode;
+  EXPECT_EQ(a, b);
+
+  a.target_level_dbfs++;
+  b.target_level_dbfs = a.target_level_dbfs;
+  EXPECT_EQ(a, b);
+
+  a.compression_gain_db++;
+  b.compression_gain_db = a.compression_gain_db;
+  EXPECT_EQ(a, b);
+
+  Toggle(a.enable_limiter);
+  b.enable_limiter = a.enable_limiter;
+  EXPECT_EQ(a, b);
+
+  auto& a_analog = a.analog_gain_controller;
+  auto& b_analog = b.analog_gain_controller;
+
+  Toggle(a_analog.enabled);
+  b_analog.enabled = a_analog.enabled;
+  EXPECT_EQ(a, b);
+
+  a_analog.startup_min_volume++;
+  b_analog.startup_min_volume = a_analog.startup_min_volume;
+  EXPECT_EQ(a, b);
+
+  a_analog.clipped_level_min++;
+  b_analog.clipped_level_min = a_analog.clipped_level_min;
+  EXPECT_EQ(a, b);
+
+  Toggle(a_analog.enable_digital_adaptive);
+  b_analog.enable_digital_adaptive = a_analog.enable_digital_adaptive;
+  EXPECT_EQ(a, b);
+}
+
+// Checks that one differing parameter is sufficient to make two configs
+// different.
+TEST(AudioProcessing, GainController1ConfigNotEqual) {
+  AudioProcessing::Config::GainController1 a;
+  const AudioProcessing::Config::GainController1 b;
+
+  Toggle(a.enabled);
+  EXPECT_NE(a, b);
+  a = b;
+
+  a.mode = AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital;
+  EXPECT_NE(a, b);
+  a = b;
+
+  a.target_level_dbfs++;
+  EXPECT_NE(a, b);
+  a = b;
+
+  a.compression_gain_db++;
+  EXPECT_NE(a, b);
+  a = b;
+
+  Toggle(a.enable_limiter);
+  EXPECT_NE(a, b);
+  a = b;
+
+  auto& a_analog = a.analog_gain_controller;
+  const auto& b_analog = b.analog_gain_controller;
+
+  Toggle(a_analog.enabled);
+  EXPECT_NE(a, b);
+  a_analog = b_analog;
+
+  a_analog.startup_min_volume++;
+  EXPECT_NE(a, b);
+  a_analog = b_analog;
+
+  a_analog.clipped_level_min++;
+  EXPECT_NE(a, b);
+  a_analog = b_analog;
+
+  Toggle(a_analog.enable_digital_adaptive);
+  EXPECT_NE(a, b);
+  a_analog = b_analog;
+}
+
+TEST(AudioProcessing, GainController2ConfigEqual) {
+  AudioProcessing::Config::GainController2 a;
+  AudioProcessing::Config::GainController2 b;
+  EXPECT_EQ(a, b);
+
+  Toggle(a.enabled);
+  b.enabled = a.enabled;
+  EXPECT_EQ(a, b);
+
+  a.fixed_digital.gain_db += 1.0f;
+  b.fixed_digital.gain_db = a.fixed_digital.gain_db;
+  EXPECT_EQ(a, b);
+
+  auto& a_adaptive = a.adaptive_digital;
+  auto& b_adaptive = b.adaptive_digital;
+
+  Toggle(a_adaptive.enabled);
+  b_adaptive.enabled = a_adaptive.enabled;
+  EXPECT_EQ(a, b);
+
+  a_adaptive.headroom_db += 1.0f;
+  b_adaptive.headroom_db = a_adaptive.headroom_db;
+  EXPECT_EQ(a, b);
+
+  a_adaptive.max_gain_db += 1.0f;
+  b_adaptive.max_gain_db = a_adaptive.max_gain_db;
+  EXPECT_EQ(a, b);
+
+  a_adaptive.initial_gain_db += 1.0f;
+  b_adaptive.initial_gain_db = a_adaptive.initial_gain_db;
+  EXPECT_EQ(a, b);
+
+  a_adaptive.max_gain_change_db_per_second += 1.0f;
+  b_adaptive.max_gain_change_db_per_second =
+      a_adaptive.max_gain_change_db_per_second;
+  EXPECT_EQ(a, b);
+
+  a_adaptive.max_output_noise_level_dbfs += 1.0f;
+  b_adaptive.max_output_noise_level_dbfs =
+      a_adaptive.max_output_noise_level_dbfs;
+  EXPECT_EQ(a, b);
+}
+
+// Checks that one differing parameter is sufficient to make two configs
+// different.
+TEST(AudioProcessing, GainController2ConfigNotEqual) {
+  AudioProcessing::Config::GainController2 a;
+  const AudioProcessing::Config::GainController2 b;
+
+  Toggle(a.enabled);
+  EXPECT_NE(a, b);
+  a = b;
+
+  a.fixed_digital.gain_db += 1.0f;
+  EXPECT_NE(a, b);
+  a.fixed_digital = b.fixed_digital;
+
+  auto& a_adaptive = a.adaptive_digital;
+  const auto& b_adaptive = b.adaptive_digital;
+
+  Toggle(a_adaptive.enabled);
+  EXPECT_NE(a, b);
+  a_adaptive = b_adaptive;
+
+  a_adaptive.headroom_db += 1.0f;
+  EXPECT_NE(a, b);
+  a_adaptive = b_adaptive;
+
+  a_adaptive.max_gain_db += 1.0f;
+  EXPECT_NE(a, b);
+  a_adaptive = b_adaptive;
+
+  a_adaptive.initial_gain_db += 1.0f;
+  EXPECT_NE(a, b);
+  a_adaptive = b_adaptive;
+
+  a_adaptive.max_gain_change_db_per_second += 1.0f;
+  EXPECT_NE(a, b);
+  a_adaptive = b_adaptive;
+
+  a_adaptive.max_output_noise_level_dbfs += 1.0f;
+  EXPECT_NE(a, b);
+  a_adaptive = b_adaptive;
+}
+
+struct ApmFormatHandlingTestParams {
+  enum class ExpectedOutput {
+    kErrorAndUnmodified,
+    kErrorAndSilence,
+    kErrorAndCopyOfFirstChannel,
+    kErrorAndExactCopy,
+    kNoError
+  };
+
+  StreamConfig input_config;
+  StreamConfig output_config;
+  ExpectedOutput expected_output;
+};
+
+class ApmFormatHandlingTest
+    : public ::testing::TestWithParam<
+          std::tuple<StreamDirection, ApmFormatHandlingTestParams>> {
+ public:
+  ApmFormatHandlingTest()
+      : stream_direction_(std::get<0>(GetParam())),
+        test_params_(std::get<1>(GetParam())) {}
+
+ protected:
+  ::testing::Message ProduceDebugMessage() {
+    return ::testing::Message()
+           << "input sample_rate_hz="
+           << test_params_.input_config.sample_rate_hz()
+           << " num_channels=" << test_params_.input_config.num_channels()
+           << ", output sample_rate_hz="
+           << test_params_.output_config.sample_rate_hz()
+           << " num_channels=" << test_params_.output_config.num_channels()
+           << ", stream_direction=" << stream_direction_ << ", expected_output="
+           << static_cast<int>(test_params_.expected_output);
+  }
+
+  StreamDirection stream_direction_;
+  ApmFormatHandlingTestParams test_params_;
+};
+
+INSTANTIATE_TEST_SUITE_P(
+    FormatValidation,
+    ApmFormatHandlingTest,
+    testing::Combine(
+        ::testing::Values(kForward, kReverse),
+        ::testing::Values(
+            // Test cases with values on the boundary of legal ranges.
+            ApmFormatHandlingTestParams{
+                StreamConfig(16000, 1), StreamConfig(8000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kNoError},
+            ApmFormatHandlingTestParams{
+                StreamConfig(8000, 1), StreamConfig(16000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kNoError},
+            ApmFormatHandlingTestParams{
+                StreamConfig(384000, 1), StreamConfig(16000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kNoError},
+            ApmFormatHandlingTestParams{
+                StreamConfig(16000, 1), StreamConfig(384000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kNoError},
+            ApmFormatHandlingTestParams{
+                StreamConfig(16000, 2), StreamConfig(16000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kNoError},
+            ApmFormatHandlingTestParams{
+                StreamConfig(16000, 3), StreamConfig(16000, 3),
+                ApmFormatHandlingTestParams::ExpectedOutput::kNoError},
+
+            // Supported but incompatible formats.
+            ApmFormatHandlingTestParams{
+                StreamConfig(16000, 3), StreamConfig(16000, 2),
+                ApmFormatHandlingTestParams::ExpectedOutput::
+                    kErrorAndCopyOfFirstChannel},
+            ApmFormatHandlingTestParams{
+                StreamConfig(16000, 3), StreamConfig(16000, 4),
+                ApmFormatHandlingTestParams::ExpectedOutput::
+                    kErrorAndCopyOfFirstChannel},
+
+            // Unsupported format and input / output mismatch.
+            ApmFormatHandlingTestParams{
+                StreamConfig(7900, 1), StreamConfig(16000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndSilence},
+            ApmFormatHandlingTestParams{
+                StreamConfig(16000, 1), StreamConfig(7900, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndSilence},
+            ApmFormatHandlingTestParams{
+                StreamConfig(390000, 1), StreamConfig(16000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndSilence},
+            ApmFormatHandlingTestParams{
+                StreamConfig(16000, 1), StreamConfig(390000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndSilence},
+            ApmFormatHandlingTestParams{
+                StreamConfig(-16000, 1), StreamConfig(16000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndSilence},
+
+            // Unsupported format but input / output formats match.
+            ApmFormatHandlingTestParams{StreamConfig(7900, 1),
+                                        StreamConfig(7900, 1),
+                                        ApmFormatHandlingTestParams::
+                                            ExpectedOutput::kErrorAndExactCopy},
+            ApmFormatHandlingTestParams{StreamConfig(390000, 1),
+                                        StreamConfig(390000, 1),
+                                        ApmFormatHandlingTestParams::
+                                            ExpectedOutput::kErrorAndExactCopy},
+
+            // Unsupported but identical sample rate, channel mismatch.
+            ApmFormatHandlingTestParams{
+                StreamConfig(7900, 1), StreamConfig(7900, 2),
+                ApmFormatHandlingTestParams::ExpectedOutput::
+                    kErrorAndCopyOfFirstChannel},
+            ApmFormatHandlingTestParams{
+                StreamConfig(7900, 2), StreamConfig(7900, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::
+                    kErrorAndCopyOfFirstChannel},
+
+            // Test cases with meaningless output format.
+            ApmFormatHandlingTestParams{
+                StreamConfig(16000, 1), StreamConfig(-16000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::
+                    kErrorAndUnmodified},
+            ApmFormatHandlingTestParams{
+                StreamConfig(-16000, 1), StreamConfig(-16000, 1),
+                ApmFormatHandlingTestParams::ExpectedOutput::
+                    kErrorAndUnmodified})));
+
+TEST_P(ApmFormatHandlingTest, IntApi) {
+  SCOPED_TRACE(ProduceDebugMessage());
+
+  // Set up input and output data.
+  const size_t num_input_samples =
+      test_params_.input_config.num_channels() *
+      std::abs(test_params_.input_config.sample_rate_hz() / 100);
+  const size_t num_output_samples =
+      test_params_.output_config.num_channels() *
+      std::abs(test_params_.output_config.sample_rate_hz() / 100);
+  std::vector<int16_t> input_block(num_input_samples);
+  for (int i = 0; i < static_cast<int>(input_block.size()); ++i) {
+    input_block[i] = i;
+  }
+  std::vector<int16_t> output_block(num_output_samples);
+  constexpr int kUnlikelyOffset = 37;
+  for (int i = 0; i < static_cast<int>(output_block.size()); ++i) {
+    output_block[i] = i - kUnlikelyOffset;
+  }
+
+  // Call APM.
+  rtc::scoped_refptr<AudioProcessing> ap =
+      AudioProcessingBuilderForTesting().Create();
+  int error;
+  if (stream_direction_ == kForward) {
+    error = ap->ProcessStream(input_block.data(), test_params_.input_config,
+                              test_params_.output_config, output_block.data());
+  } else {
+    error = ap->ProcessReverseStream(
+        input_block.data(), test_params_.input_config,
+        test_params_.output_config, output_block.data());
+  }
+
+  // Check output.
+  switch (test_params_.expected_output) {
+    case ApmFormatHandlingTestParams::ExpectedOutput::kNoError:
+      EXPECT_EQ(error, AudioProcessing::kNoError);
+      break;
+    case ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndUnmodified:
+      EXPECT_NE(error, AudioProcessing::kNoError);
+      for (int i = 0; i < static_cast<int>(output_block.size()); ++i) {
+        EXPECT_EQ(output_block[i], i - kUnlikelyOffset);
+      }
+      break;
+    case ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndSilence:
+      EXPECT_NE(error, AudioProcessing::kNoError);
+      for (int i = 0; i < static_cast<int>(output_block.size()); ++i) {
+        EXPECT_EQ(output_block[i], 0);
+      }
+      break;
+    case ApmFormatHandlingTestParams::ExpectedOutput::
+        kErrorAndCopyOfFirstChannel:
+      EXPECT_NE(error, AudioProcessing::kNoError);
+      for (size_t ch = 0; ch < test_params_.output_config.num_channels();
+           ++ch) {
+        for (size_t i = 0; i < test_params_.output_config.num_frames(); ++i) {
+          EXPECT_EQ(
+              output_block[ch + i * test_params_.output_config.num_channels()],
+              static_cast<int16_t>(i *
+                                   test_params_.input_config.num_channels()));
+        }
+      }
+      break;
+    case ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndExactCopy:
+      EXPECT_NE(error, AudioProcessing::kNoError);
+      for (int i = 0; i < static_cast<int>(output_block.size()); ++i) {
+        EXPECT_EQ(output_block[i], i);
+      }
+      break;
+  }
+}
+
+TEST_P(ApmFormatHandlingTest, FloatApi) {
+  SCOPED_TRACE(ProduceDebugMessage());
+
+  // Set up input and output data.
+  const size_t input_samples_per_channel =
+      std::abs(test_params_.input_config.sample_rate_hz()) / 100;
+  const size_t output_samples_per_channel =
+      std::abs(test_params_.output_config.sample_rate_hz()) / 100;
+  const size_t input_num_channels = test_params_.input_config.num_channels();
+  const size_t output_num_channels = test_params_.output_config.num_channels();
+  ChannelBuffer<float> input_block(input_samples_per_channel,
+                                   input_num_channels);
+  ChannelBuffer<float> output_block(output_samples_per_channel,
+                                    output_num_channels);
+  for (size_t ch = 0; ch < input_num_channels; ++ch) {
+    for (size_t i = 0; i < input_samples_per_channel; ++i) {
+      input_block.channels()[ch][i] = ch + i * input_num_channels;
+    }
+  }
+  constexpr int kUnlikelyOffset = 37;
+  for (size_t ch = 0; ch < output_num_channels; ++ch) {
+    for (size_t i = 0; i < output_samples_per_channel; ++i) {
+      output_block.channels()[ch][i] =
+          ch + i * output_num_channels - kUnlikelyOffset;
+    }
+  }
+
+  // Call APM.
+  rtc::scoped_refptr<AudioProcessing> ap =
+      AudioProcessingBuilderForTesting().Create();
+  int error;
+  if (stream_direction_ == kForward) {
+    error =
+        ap->ProcessStream(input_block.channels(), test_params_.input_config,
+                          test_params_.output_config, output_block.channels());
+  } else {
+    error = ap->ProcessReverseStream(
+        input_block.channels(), test_params_.input_config,
+        test_params_.output_config, output_block.channels());
+  }
+
+  // Check output.
+  switch (test_params_.expected_output) {
+    case ApmFormatHandlingTestParams::ExpectedOutput::kNoError:
+      EXPECT_EQ(error, AudioProcessing::kNoError);
+      break;
+    case ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndUnmodified:
+      EXPECT_NE(error, AudioProcessing::kNoError);
+      for (size_t ch = 0; ch < output_num_channels; ++ch) {
+        for (size_t i = 0; i < output_samples_per_channel; ++i) {
+          EXPECT_EQ(output_block.channels()[ch][i],
+                    ch + i * output_num_channels - kUnlikelyOffset);
+        }
+      }
+      break;
+    case ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndSilence:
+      EXPECT_NE(error, AudioProcessing::kNoError);
+      for (size_t ch = 0; ch < output_num_channels; ++ch) {
+        for (size_t i = 0; i < output_samples_per_channel; ++i) {
+          EXPECT_EQ(output_block.channels()[ch][i], 0);
+        }
+      }
+      break;
+    case ApmFormatHandlingTestParams::ExpectedOutput::
+        kErrorAndCopyOfFirstChannel:
+      EXPECT_NE(error, AudioProcessing::kNoError);
+      for (size_t ch = 0; ch < output_num_channels; ++ch) {
+        for (size_t i = 0; i < output_samples_per_channel; ++i) {
+          EXPECT_EQ(output_block.channels()[ch][i],
+                    input_block.channels()[0][i]);
+        }
+      }
+      break;
+    case ApmFormatHandlingTestParams::ExpectedOutput::kErrorAndExactCopy:
+      EXPECT_NE(error, AudioProcessing::kNoError);
+      for (size_t ch = 0; ch < output_num_channels; ++ch) {
+        for (size_t i = 0; i < output_samples_per_channel; ++i) {
+          EXPECT_EQ(output_block.channels()[ch][i],
+                    input_block.channels()[ch][i]);
+        }
+      }
+      break;
+  }
+}
+
+TEST(ApmAnalyzeReverseStreamFormatTest, AnalyzeReverseStream) {
+  for (auto&& [input_config, expect_error] :
+       {std::tuple(StreamConfig(16000, 2), /*expect_error=*/false),
+        std::tuple(StreamConfig(8000, 1), /*expect_error=*/false),
+        std::tuple(StreamConfig(384000, 1), /*expect_error=*/false),
+        std::tuple(StreamConfig(7900, 1), /*expect_error=*/true),
+        std::tuple(StreamConfig(390000, 1), /*expect_error=*/true),
+        std::tuple(StreamConfig(16000, 0), /*expect_error=*/true),
+        std::tuple(StreamConfig(-16000, 0), /*expect_error=*/true)}) {
+    SCOPED_TRACE(::testing::Message()
+                 << "sample_rate_hz=" << input_config.sample_rate_hz()
+                 << " num_channels=" << input_config.num_channels());
+
+    // Set up input data.
+    ChannelBuffer<float> input_block(
+        std::abs(input_config.sample_rate_hz()) / 100,
+        input_config.num_channels());
+
+    // Call APM.
+    rtc::scoped_refptr<AudioProcessing> ap =
+        AudioProcessingBuilderForTesting().Create();
+    int error = ap->AnalyzeReverseStream(input_block.channels(), input_config);
+
+    // Check output.
+    if (expect_error) {
+      EXPECT_NE(error, AudioProcessing::kNoError);
+    } else {
+      EXPECT_EQ(error, AudioProcessing::kNoError);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/BUILD.gn
new file mode 100644
index 0000000000..e7ff8482f6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/BUILD.gn
@@ -0,0 +1,45 @@
+# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("capture_levels_adjuster") {
+  visibility = [ "*" ]
+
+  sources = [
+    "audio_samples_scaler.cc",
+    "audio_samples_scaler.h",
+    "capture_levels_adjuster.cc",
+    "capture_levels_adjuster.h",
+  ]
+
+  defines = []
+
+  deps = [
+    "..:audio_buffer",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:safe_minmax",
+  ]
+}
+
+rtc_library("capture_levels_adjuster_unittests") {
+  testonly = true
+
+  sources = [
+    "audio_samples_scaler_unittest.cc",
+    "capture_levels_adjuster_unittest.cc",
+  ]
+  deps = [
+    ":capture_levels_adjuster",
+    "..:audioproc_test_utils",
+    "../../../rtc_base:gunit_helpers",
+    "../../../rtc_base:stringutils",
+    "../../../test:test_support",
+  ]
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc
new file mode 100644
index 0000000000..cb2336b87d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h"
+
+#include <algorithm>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+AudioSamplesScaler::AudioSamplesScaler(float initial_gain)
+    : previous_gain_(initial_gain), target_gain_(initial_gain) {}
+
+void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) {
+  if (static_cast<int>(audio_buffer.num_frames()) != samples_per_channel_) {
+    // Update the members depending on audio-buffer length if needed.
+    RTC_DCHECK_GT(audio_buffer.num_frames(), 0);
+    samples_per_channel_ = static_cast<int>(audio_buffer.num_frames());
+    one_by_samples_per_channel_ = 1.f / samples_per_channel_;
+  }
+
+  if (target_gain_ == 1.f && previous_gain_ == target_gain_) {
+    // If only a gain of 1 is to be applied, do an early return without applying
+    // any gain.
+    return;
+  }
+
+  float gain = previous_gain_;
+  if (previous_gain_ == target_gain_) {
+    // Apply a non-changing gain.
+    for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
+      rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
+                                         samples_per_channel_);
+      for (float& sample : channel_view) {
+        sample *= gain;
+      }
+    }
+  } else {
+    const float increment =
+        (target_gain_ - previous_gain_) * one_by_samples_per_channel_;
+
+    if (increment > 0.f) {
+      // Apply an increasing gain.
+      for (size_t channel = 0; channel < audio_buffer.num_channels();
+           ++channel) {
+        gain = previous_gain_;
+        rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
+                                           samples_per_channel_);
+        for (float& sample : channel_view) {
+          gain = std::min(gain + increment, target_gain_);
+          sample *= gain;
+        }
+      }
+    } else {
+      // Apply a decreasing gain.
+      for (size_t channel = 0; channel < audio_buffer.num_channels();
+           ++channel) {
+        gain = previous_gain_;
+        rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
+                                           samples_per_channel_);
+        for (float& sample : channel_view) {
+          gain = std::max(gain + increment, target_gain_);
+          sample *= gain;
+        }
+      }
+    }
+  }
+  previous_gain_ = target_gain_;
+
+  // Saturate the samples to be in the S16 range.
+  for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
+    rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
+                                       samples_per_channel_);
+    for (float& sample : channel_view) {
+      constexpr float kMinFloatS16Value = -32768.f;
+      constexpr float kMaxFloatS16Value = 32767.f;
+      sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h
new file mode 100644
index 0000000000..2ae8533940
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_
+#define MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_
+
+#include <stddef.h>
+
+#include "modules/audio_processing/audio_buffer.h"
+
+namespace webrtc {
+
+// Handles and applies a gain to the samples in an audio buffer.
+// The gain is applied for each sample and any changes in the gain take effect
+// gradually (in a linear manner) over one frame.
+class AudioSamplesScaler {
+ public:
+  // C-tor. The supplied `initial_gain` is used immediately at the first call to
+  // Process(), i.e., in contrast to the gain supplied by SetGain(...) there is
+  // no gradual change to the `initial_gain`.
+  explicit AudioSamplesScaler(float initial_gain);
+  AudioSamplesScaler(const AudioSamplesScaler&) = delete;
+  AudioSamplesScaler& operator=(const AudioSamplesScaler&) = delete;
+
+  // Applies the specified gain to the audio in `audio_buffer`.
+  void Process(AudioBuffer& audio_buffer);
+
+  // Sets the gain to apply to each sample.
+  void SetGain(float gain) { target_gain_ = gain; }
+
+ private:
+  float previous_gain_ = 1.f;
+  float target_gain_ = 1.f;
+  int samples_per_channel_ = -1;
+  float one_by_samples_per_channel_ = -1.f;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler_unittest.cc b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler_unittest.cc
new file mode 100644
index 0000000000..6e5fc2cbe3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler_unittest.cc
@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h"
+
+#include <tuple>
+
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+float SampleValueForChannel(int channel) {
+  constexpr float kSampleBaseValue = 100.f;
+  constexpr float kSampleChannelOffset = 1.f;
+  return kSampleBaseValue + channel * kSampleChannelOffset;
+}
+
+void PopulateBuffer(AudioBuffer& audio_buffer) {
+  for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
+    test::FillBufferChannel(SampleValueForChannel(ch), ch, audio_buffer);
+  }
+}
+
+constexpr int kNumFramesToProcess = 10;
+
+class AudioSamplesScalerTest
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::tuple<int, int, float>> {
+ protected:
+  int sample_rate_hz() const { return std::get<0>(GetParam()); }
+  int num_channels() const { return std::get<1>(GetParam()); }
+  float initial_gain() const { return std::get<2>(GetParam()); }
+};
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioSamplesScalerTestSuite,
+    AudioSamplesScalerTest,
+    ::testing::Combine(::testing::Values(16000, 32000, 48000),
+                       ::testing::Values(1, 2, 4),
+                       ::testing::Values(0.1f, 1.f, 2.f, 4.f)));
+
+TEST_P(AudioSamplesScalerTest, InitialGainIsRespected) {
+  AudioSamplesScaler scaler(initial_gain());
+
+  AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(),
+                           num_channels(), sample_rate_hz(), num_channels());
+
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    PopulateBuffer(audio_buffer);
+    scaler.Process(audio_buffer);
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        initial_gain() * SampleValueForChannel(ch));
+      }
+    }
+  }
+}
+
+TEST_P(AudioSamplesScalerTest, VerifyGainAdjustment) {
+  const float higher_gain = initial_gain();
+  const float lower_gain = higher_gain / 2.f;
+
+  AudioSamplesScaler scaler(lower_gain);
+
+  AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(),
+                           num_channels(), sample_rate_hz(), num_channels());
+
+  // Allow the intial, lower, gain to take effect.
+  PopulateBuffer(audio_buffer);
+
+  scaler.Process(audio_buffer);
+
+  // Set the new, higher, gain.
+  scaler.SetGain(higher_gain);
+
+  // Ensure that the new, higher, gain is achieved gradually over one frame.
+  PopulateBuffer(audio_buffer);
+
+  scaler.Process(audio_buffer);
+  for (int ch = 0; ch < num_channels(); ++ch) {
+    for (size_t i = 0; i < audio_buffer.num_frames() - 1; ++i) {
+      EXPECT_LT(audio_buffer.channels_const()[ch][i],
+                higher_gain * SampleValueForChannel(ch));
+      EXPECT_LE(audio_buffer.channels_const()[ch][i],
+                audio_buffer.channels_const()[ch][i + 1]);
+    }
+    EXPECT_LE(audio_buffer.channels_const()[ch][audio_buffer.num_frames() - 1],
+              higher_gain * SampleValueForChannel(ch));
+  }
+
+  // Ensure that the new, higher, gain is achieved and stay unchanged.
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    PopulateBuffer(audio_buffer);
+    scaler.Process(audio_buffer);
+
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        higher_gain * SampleValueForChannel(ch));
+      }
+    }
+  }
+
+  // Set the new, lower, gain.
+  scaler.SetGain(lower_gain);
+
+  // Ensure that the new, lower, gain is achieved gradually over one frame.
+  PopulateBuffer(audio_buffer);
+  scaler.Process(audio_buffer);
+  for (int ch = 0; ch < num_channels(); ++ch) {
+    for (size_t i = 0; i < audio_buffer.num_frames() - 1; ++i) {
+      EXPECT_GT(audio_buffer.channels_const()[ch][i],
+                lower_gain * SampleValueForChannel(ch));
+      EXPECT_GE(audio_buffer.channels_const()[ch][i],
+                audio_buffer.channels_const()[ch][i + 1]);
+    }
+    EXPECT_GE(audio_buffer.channels_const()[ch][audio_buffer.num_frames() - 1],
+              lower_gain * SampleValueForChannel(ch));
+  }
+
+  // Ensure that the new, lower, gain is achieved and stay unchanged.
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    PopulateBuffer(audio_buffer);
+    scaler.Process(audio_buffer);
+
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        lower_gain * SampleValueForChannel(ch));
+      }
+    }
+  }
+}
+
+TEST(AudioSamplesScaler, UpwardsClamping) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  constexpr float kGain = 10.f;
+  constexpr float kMaxClampedSampleValue = 32767.f;
+  static_assert(kGain > 1.f, "");
+
+  AudioSamplesScaler scaler(kGain);
+
+  AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
+                           kNumChannels, kSampleRateHz, kNumChannels);
+
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
+      test::FillBufferChannel(
+          kMaxClampedSampleValue - audio_buffer.num_channels() + 1.f + ch, ch,
+          audio_buffer);
+    }
+
+    scaler.Process(audio_buffer);
+    for (int ch = 0; ch < kNumChannels; ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        kMaxClampedSampleValue);
+      }
+    }
+  }
+}
+
+TEST(AudioSamplesScaler, DownwardsClamping) {
+  constexpr int kSampleRateHz = 48000;
+  constexpr int kNumChannels = 1;
+  constexpr float kGain = 10.f;
+  constexpr float kMinClampedSampleValue = -32768.f;
+  static_assert(kGain > 1.f, "");
+
+  AudioSamplesScaler scaler(kGain);
+
+  AudioBuffer audio_buffer(kSampleRateHz, kNumChannels, kSampleRateHz,
+                           kNumChannels, kSampleRateHz, kNumChannels);
+
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
+      test::FillBufferChannel(
+          kMinClampedSampleValue + audio_buffer.num_channels() - 1.f + ch, ch,
+          audio_buffer);
+    }
+
+    scaler.Process(audio_buffer);
+    for (int ch = 0; ch < kNumChannels; ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        kMinClampedSampleValue);
+      }
+    }
+  }
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.cc b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.cc
new file mode 100644
index 0000000000..dfda582915
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.cc
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h"
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kMinAnalogMicGainLevel = 0;
+constexpr int kMaxAnalogMicGainLevel = 255;
+
+float ComputeLevelBasedGain(int emulated_analog_mic_gain_level) {
+  static_assert(
+      kMinAnalogMicGainLevel == 0,
+      "The minimum gain level must be 0 for the maths below to work.");
+  static_assert(kMaxAnalogMicGainLevel > 0,
+                "The minimum gain level must be larger than 0 for the maths "
+                "below to work.");
+  constexpr float kGainToLevelMultiplier = 1.f / kMaxAnalogMicGainLevel;
+
+  RTC_DCHECK_GE(emulated_analog_mic_gain_level, kMinAnalogMicGainLevel);
+  RTC_DCHECK_LE(emulated_analog_mic_gain_level, kMaxAnalogMicGainLevel);
+  return kGainToLevelMultiplier * emulated_analog_mic_gain_level;
+}
+
+float ComputePreGain(float pre_gain,
+                     int emulated_analog_mic_gain_level,
+                     bool emulated_analog_mic_gain_enabled) {
+  return emulated_analog_mic_gain_enabled
+             ? pre_gain * ComputeLevelBasedGain(emulated_analog_mic_gain_level)
+             : pre_gain;
+}
+
+}  // namespace
+
+CaptureLevelsAdjuster::CaptureLevelsAdjuster(
+    bool emulated_analog_mic_gain_enabled,
+    int emulated_analog_mic_gain_level,
+    float pre_gain,
+    float post_gain)
+    : emulated_analog_mic_gain_enabled_(emulated_analog_mic_gain_enabled),
+      emulated_analog_mic_gain_level_(emulated_analog_mic_gain_level),
+      pre_gain_(pre_gain),
+      pre_adjustment_gain_(ComputePreGain(pre_gain_,
+                                          emulated_analog_mic_gain_level_,
+                                          emulated_analog_mic_gain_enabled_)),
+      pre_scaler_(pre_adjustment_gain_),
+      post_scaler_(post_gain) {}
+
+void CaptureLevelsAdjuster::ApplyPreLevelAdjustment(AudioBuffer& audio_buffer) {
+  pre_scaler_.Process(audio_buffer);
+}
+
+void CaptureLevelsAdjuster::ApplyPostLevelAdjustment(
+    AudioBuffer& audio_buffer) {
+  post_scaler_.Process(audio_buffer);
+}
+
+void CaptureLevelsAdjuster::SetPreGain(float pre_gain) {
+  pre_gain_ = pre_gain;
+  UpdatePreAdjustmentGain();
+}
+
+void CaptureLevelsAdjuster::SetPostGain(float post_gain) {
+  post_scaler_.SetGain(post_gain);
+}
+
+void CaptureLevelsAdjuster::SetAnalogMicGainLevel(int level) {
+  RTC_DCHECK_GE(level, kMinAnalogMicGainLevel);
+  RTC_DCHECK_LE(level, kMaxAnalogMicGainLevel);
+  int clamped_level =
+      rtc::SafeClamp(level, kMinAnalogMicGainLevel, kMaxAnalogMicGainLevel);
+
+  emulated_analog_mic_gain_level_ = clamped_level;
+  UpdatePreAdjustmentGain();
+}
+
+void CaptureLevelsAdjuster::UpdatePreAdjustmentGain() {
+  pre_adjustment_gain_ =
+      ComputePreGain(pre_gain_, emulated_analog_mic_gain_level_,
+                     emulated_analog_mic_gain_enabled_);
+  pre_scaler_.SetGain(pre_adjustment_gain_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h
new file mode 100644
index 0000000000..38b68ad06c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_CAPTURE_LEVELS_ADJUSTER_H_
+#define MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_CAPTURE_LEVELS_ADJUSTER_H_
+
+#include <stddef.h>
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h"
+
+namespace webrtc {
+
+// Adjusts the level of the capture signal before and after all capture-side
+// processing is done using a combination of explicitly specified gains
+// and an emulated analog gain functionality where a specified analog level
+// results in an additional gain. The pre-adjustment is achieved by combining
+// the gain value `pre_gain` and the level `emulated_analog_mic_gain_level` to
+// form a combined gain of `pre_gain`*`emulated_analog_mic_gain_level`/255 which
+// is multiplied to each sample. The intention of the
+// `emulated_analog_mic_gain_level` is to be controlled by the analog AGC
+// functionality and to produce an emulated analog mic gain equal to
+// `emulated_analog_mic_gain_level`/255. The post level adjustment is achieved
+// by multiplying each sample with the value of `post_gain`. Any changes in the
+// gains take are done smoothly over one frame and the scaled samples are
+// clamped to fit into the allowed S16 sample range.
+class CaptureLevelsAdjuster {
+ public:
+  // C-tor. The values for the level and the gains must fulfill
+  // 0 <= emulated_analog_mic_gain_level <= 255.
+  // 0.f <= pre_gain.
+  // 0.f <= post_gain.
+  CaptureLevelsAdjuster(bool emulated_analog_mic_gain_enabled,
+                        int emulated_analog_mic_gain_level,
+                        float pre_gain,
+                        float post_gain);
+  CaptureLevelsAdjuster(const CaptureLevelsAdjuster&) = delete;
+  CaptureLevelsAdjuster& operator=(const CaptureLevelsAdjuster&) = delete;
+
+  // Adjusts the level of the signal. This should be called before any of the
+  // other processing is performed.
+  void ApplyPreLevelAdjustment(AudioBuffer& audio_buffer);
+
+  // Adjusts the level of the signal. This should be called after all of the
+  // other processing have been performed.
+  void ApplyPostLevelAdjustment(AudioBuffer& audio_buffer);
+
+  // Sets the gain to apply to each sample before any of the other processing is
+  // performed.
+  void SetPreGain(float pre_gain);
+
+  // Returns the total pre-adjustment gain applied, comprising both the pre_gain
+  // as well as the gain from the emulated analog mic, to each sample before any
+  // of the other processing is performed.
+  float GetPreAdjustmentGain() const { return pre_adjustment_gain_; }
+
+  // Sets the gain to apply to each sample after all of the other processing
+  // have been performed.
+  void SetPostGain(float post_gain);
+
+  // Sets the analog gain level to use for the emulated analog gain.
+  // `level` must be in the range [0...255].
+  void SetAnalogMicGainLevel(int level);
+
+  // Returns the current analog gain level used for the emulated analog gain.
+  int GetAnalogMicGainLevel() const { return emulated_analog_mic_gain_level_; }
+
+ private:
+  // Updates the value of `pre_adjustment_gain_` based on the supplied values
+  // for `pre_gain` and `emulated_analog_mic_gain_level_`.
+  void UpdatePreAdjustmentGain();
+
+  const bool emulated_analog_mic_gain_enabled_;
+  int emulated_analog_mic_gain_level_;
+  float pre_gain_;
+  float pre_adjustment_gain_;
+  AudioSamplesScaler pre_scaler_;
+  AudioSamplesScaler post_scaler_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_CAPTURE_LEVELS_ADJUSTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_gn/moz.build
new file mode 100644
index 0000000000..3e57c48b6f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.cc",
+    "/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("capture_levels_adjuster_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc
new file mode 100644
index 0000000000..1183441a14
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster_unittest.cc
@@ -0,0 +1,187 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h"
+
+#include <algorithm>
+#include <tuple>
+
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+float SampleValueForChannel(int channel) {
+  constexpr float kSampleBaseValue = 100.f;
+  constexpr float kSampleChannelOffset = 1.f;
+  return kSampleBaseValue + channel * kSampleChannelOffset;
+}
+
+void PopulateBuffer(AudioBuffer& audio_buffer) {
+  for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
+    test::FillBufferChannel(SampleValueForChannel(ch), ch, audio_buffer);
+  }
+}
+
+float ComputeExpectedSignalGainAfterApplyPreLevelAdjustment(
+    bool emulated_analog_mic_gain_enabled,
+    int emulated_analog_mic_gain_level,
+    float pre_gain) {
+  if (!emulated_analog_mic_gain_enabled) {
+    return pre_gain;
+  }
+  return pre_gain * std::min(emulated_analog_mic_gain_level, 255) / 255.f;
+}
+
+float ComputeExpectedSignalGainAfterApplyPostLevelAdjustment(
+    bool emulated_analog_mic_gain_enabled,
+    int emulated_analog_mic_gain_level,
+    float pre_gain,
+    float post_gain) {
+  return post_gain * ComputeExpectedSignalGainAfterApplyPreLevelAdjustment(
+                         emulated_analog_mic_gain_enabled,
+                         emulated_analog_mic_gain_level, pre_gain);
+}
+
+constexpr int kNumFramesToProcess = 10;
+
+class CaptureLevelsAdjusterTest
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<
+          std::tuple<int, int, bool, int, float, float>> {
+ protected:
+  int sample_rate_hz() const { return std::get<0>(GetParam()); }
+  int num_channels() const { return std::get<1>(GetParam()); }
+  bool emulated_analog_mic_gain_enabled() const {
+    return std::get<2>(GetParam());
+  }
+  int emulated_analog_mic_gain_level() const { return std::get<3>(GetParam()); }
+  float pre_gain() const { return std::get<4>(GetParam()); }
+  float post_gain() const { return std::get<5>(GetParam()); }
+};
+
+INSTANTIATE_TEST_SUITE_P(
+    CaptureLevelsAdjusterTestSuite,
+    CaptureLevelsAdjusterTest,
+    ::testing::Combine(::testing::Values(16000, 32000, 48000),
+                       ::testing::Values(1, 2, 4),
+                       ::testing::Values(false, true),
+                       ::testing::Values(21, 255),
+                       ::testing::Values(0.1f, 1.f, 4.f),
+                       ::testing::Values(0.1f, 1.f, 4.f)));
+
+TEST_P(CaptureLevelsAdjusterTest, InitialGainIsInstantlyAchieved) {
+  CaptureLevelsAdjuster adjuster(emulated_analog_mic_gain_enabled(),
+                                 emulated_analog_mic_gain_level(), pre_gain(),
+                                 post_gain());
+
+  AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(),
+                           num_channels(), sample_rate_hz(), num_channels());
+
+  const float expected_signal_gain_after_pre_gain =
+      ComputeExpectedSignalGainAfterApplyPreLevelAdjustment(
+          emulated_analog_mic_gain_enabled(), emulated_analog_mic_gain_level(),
+          pre_gain());
+  const float expected_signal_gain_after_post_level_adjustment =
+      ComputeExpectedSignalGainAfterApplyPostLevelAdjustment(
+          emulated_analog_mic_gain_enabled(), emulated_analog_mic_gain_level(),
+          pre_gain(), post_gain());
+
+  for (int frame = 0; frame < kNumFramesToProcess; ++frame) {
+    PopulateBuffer(audio_buffer);
+    adjuster.ApplyPreLevelAdjustment(audio_buffer);
+    EXPECT_FLOAT_EQ(adjuster.GetPreAdjustmentGain(),
+                    expected_signal_gain_after_pre_gain);
+
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(
+            audio_buffer.channels_const()[ch][i],
+            expected_signal_gain_after_pre_gain * SampleValueForChannel(ch));
+      }
+    }
+    adjuster.ApplyPostLevelAdjustment(audio_buffer);
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        expected_signal_gain_after_post_level_adjustment *
+                            SampleValueForChannel(ch));
+      }
+    }
+  }
+}
+
+TEST_P(CaptureLevelsAdjusterTest, NewGainsAreAchieved) {
+  const int lower_emulated_analog_mic_gain_level =
+      emulated_analog_mic_gain_level();
+  const float lower_pre_gain = pre_gain();
+  const float lower_post_gain = post_gain();
+  const int higher_emulated_analog_mic_gain_level =
+      std::min(lower_emulated_analog_mic_gain_level * 2, 255);
+  const float higher_pre_gain = lower_pre_gain * 2.f;
+  const float higher_post_gain = lower_post_gain * 2.f;
+
+  CaptureLevelsAdjuster adjuster(emulated_analog_mic_gain_enabled(),
+                                 lower_emulated_analog_mic_gain_level,
+                                 lower_pre_gain, lower_post_gain);
+
+  AudioBuffer audio_buffer(sample_rate_hz(), num_channels(), sample_rate_hz(),
+                           num_channels(), sample_rate_hz(), num_channels());
+
+  const float expected_signal_gain_after_pre_gain =
+      ComputeExpectedSignalGainAfterApplyPreLevelAdjustment(
+          emulated_analog_mic_gain_enabled(),
+          higher_emulated_analog_mic_gain_level, higher_pre_gain);
+  const float expected_signal_gain_after_post_level_adjustment =
+      ComputeExpectedSignalGainAfterApplyPostLevelAdjustment(
+          emulated_analog_mic_gain_enabled(),
+          higher_emulated_analog_mic_gain_level, higher_pre_gain,
+          higher_post_gain);
+
+  adjuster.SetPreGain(higher_pre_gain);
+  adjuster.SetPostGain(higher_post_gain);
+  adjuster.SetAnalogMicGainLevel(higher_emulated_analog_mic_gain_level);
+
+  PopulateBuffer(audio_buffer);
+  adjuster.ApplyPreLevelAdjustment(audio_buffer);
+  adjuster.ApplyPostLevelAdjustment(audio_buffer);
+  EXPECT_EQ(adjuster.GetAnalogMicGainLevel(),
+            higher_emulated_analog_mic_gain_level);
+
+  for (int frame = 1; frame < kNumFramesToProcess; ++frame) {
+    PopulateBuffer(audio_buffer);
+    adjuster.ApplyPreLevelAdjustment(audio_buffer);
+    EXPECT_FLOAT_EQ(adjuster.GetPreAdjustmentGain(),
+                    expected_signal_gain_after_pre_gain);
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(
+            audio_buffer.channels_const()[ch][i],
+            expected_signal_gain_after_pre_gain * SampleValueForChannel(ch));
+      }
+    }
+
+    adjuster.ApplyPostLevelAdjustment(audio_buffer);
+    for (int ch = 0; ch < num_channels(); ++ch) {
+      for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[ch][i],
+                        expected_signal_gain_after_post_level_adjustment *
+                            SampleValueForChannel(ch));
+      }
+    }
+
+    EXPECT_EQ(adjuster.GetAnalogMicGainLevel(),
+              higher_emulated_analog_mic_gain_level);
+  }
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/debug.proto b/third_party/libwebrtc/modules/audio_processing/debug.proto
new file mode 100644
index 0000000000..cc5efbc73c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/debug.proto
@@ -0,0 +1,115 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.audioproc;
+
+// Contains the format of input/output/reverse audio. An Init message is added
+// when any of the fields are changed.
+message Init {
+  optional int32 sample_rate = 1;
+  optional int32 device_sample_rate = 2 [deprecated=true];
+  optional int32 num_input_channels = 3;
+  optional int32 num_output_channels = 4;
+  optional int32 num_reverse_channels = 5;
+  optional int32 reverse_sample_rate = 6;
+  optional int32 output_sample_rate = 7;
+  optional int32 reverse_output_sample_rate = 8;
+  optional int32 num_reverse_output_channels = 9;
+  optional int64 timestamp_ms = 10;
+}
+
+// May contain interleaved or deinterleaved data, but don't store both formats.
+message ReverseStream {
+  // int16 interleaved data.
+  optional bytes data = 1;
+
+  // float deinterleaved data, where each repeated element points to a single
+  // channel buffer of data.
+  repeated bytes channel = 2;
+}
+
+// May contain interleaved or deinterleaved data, but don't store both formats.
+message Stream {
+  // int16 interleaved data.
+  optional bytes input_data = 1;
+  optional bytes output_data = 2;
+
+  optional int32 delay = 3;
+  optional sint32 drift = 4;
+  optional int32 applied_input_volume = 5;
+  optional bool keypress = 6;
+
+  // float deinterleaved data, where each repeated element points to a single
+  // channel buffer of data.
+  repeated bytes input_channel = 7;
+  repeated bytes output_channel = 8;
+}
+
+// Contains the configurations of various APM component. A Config message is
+// added when any of the fields are changed.
+message Config {
+  // Acoustic echo canceler.
+  optional bool aec_enabled = 1;
+  optional bool aec_delay_agnostic_enabled = 2;
+  optional bool aec_drift_compensation_enabled = 3;
+  optional bool aec_extended_filter_enabled = 4;
+  optional int32 aec_suppression_level = 5;
+  // Mobile AEC.
+  optional bool aecm_enabled = 6;
+  optional bool aecm_comfort_noise_enabled = 7 [deprecated = true];
+  optional int32 aecm_routing_mode = 8 [deprecated = true];
+  // Automatic gain controller.
+  optional bool agc_enabled = 9;
+  optional int32 agc_mode = 10;
+  optional bool agc_limiter_enabled = 11;
+  optional bool noise_robust_agc_enabled = 12;
+  // High pass filter.
+  optional bool hpf_enabled = 13;
+  // Noise suppression.
+  optional bool ns_enabled = 14;
+  optional int32 ns_level = 15;
+  // Transient suppression.
+  optional bool transient_suppression_enabled = 16;
+  // Semicolon-separated string containing experimental feature
+  // descriptions.
+  optional string experiments_description = 17;
+  reserved 18;  // Intelligibility enhancer enabled (deprecated).
+  // Pre amplifier.
+  optional bool pre_amplifier_enabled = 19;
+  optional float pre_amplifier_fixed_gain_factor = 20;
+
+  // Next field number 21.
+}
+
+message PlayoutAudioDeviceInfo {
+  optional int32 id = 1;
+  optional int32 max_volume = 2;
+}
+
+message RuntimeSetting {
+  optional float capture_pre_gain = 1;
+  optional float custom_render_processing_setting = 2;
+  optional float capture_fixed_post_gain = 3;
+  optional int32 playout_volume_change = 4;
+  optional PlayoutAudioDeviceInfo playout_audio_device_change = 5;
+  optional bool capture_output_used = 6;
+  optional float capture_post_gain = 7;
+}
+
+message Event {
+  enum Type {
+    INIT = 0;
+    REVERSE_STREAM = 1;
+    STREAM = 2;
+    CONFIG = 3;
+    UNKNOWN_EVENT = 4;
+    RUNTIME_SETTING = 5;
+  }
+
+  required Type type = 1;
+
+  optional Init init = 2;
+  optional ReverseStream reverse_stream = 3;
+  optional Stream stream = 4;
+  optional Config config = 5;
+  optional RuntimeSetting runtime_setting = 6;
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_bit_exact_unittest.cc b/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_bit_exact_unittest.cc
new file mode 100644
index 0000000000..f351811e08
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_bit_exact_unittest.cc
@@ -0,0 +1,221 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/echo_control_mobile_impl.h"
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+#include "modules/audio_processing/test/bitexactness_tools.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+// TODO(peah): Increase the number of frames to proces when the issue of
+// non repeatable test results have been found.
+const int kNumFramesToProcess = 200;
+
+void SetupComponent(int sample_rate_hz,
+                    EchoControlMobileImpl::RoutingMode routing_mode,
+                    bool comfort_noise_enabled,
+                    EchoControlMobileImpl* echo_control_mobile) {
+  echo_control_mobile->Initialize(
+      sample_rate_hz > 16000 ? 16000 : sample_rate_hz, 1, 1);
+  echo_control_mobile->set_routing_mode(routing_mode);
+  echo_control_mobile->enable_comfort_noise(comfort_noise_enabled);
+}
+
+void ProcessOneFrame(int sample_rate_hz,
+                     int stream_delay_ms,
+                     AudioBuffer* render_audio_buffer,
+                     AudioBuffer* capture_audio_buffer,
+                     EchoControlMobileImpl* echo_control_mobile) {
+  if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
+    render_audio_buffer->SplitIntoFrequencyBands();
+    capture_audio_buffer->SplitIntoFrequencyBands();
+  }
+
+  std::vector<int16_t> render_audio;
+  EchoControlMobileImpl::PackRenderAudioBuffer(
+      render_audio_buffer, 1, render_audio_buffer->num_channels(),
+      &render_audio);
+  echo_control_mobile->ProcessRenderAudio(render_audio);
+
+  echo_control_mobile->ProcessCaptureAudio(capture_audio_buffer,
+                                           stream_delay_ms);
+
+  if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
+    capture_audio_buffer->MergeFrequencyBands();
+  }
+}
+
+void RunBitexactnessTest(int sample_rate_hz,
+                         size_t num_channels,
+                         int stream_delay_ms,
+                         EchoControlMobileImpl::RoutingMode routing_mode,
+                         bool comfort_noise_enabled,
+                         const rtc::ArrayView<const float>& output_reference) {
+  EchoControlMobileImpl echo_control_mobile;
+  SetupComponent(sample_rate_hz, routing_mode, comfort_noise_enabled,
+                 &echo_control_mobile);
+
+  const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
+  const StreamConfig render_config(sample_rate_hz, num_channels);
+  AudioBuffer render_buffer(
+      render_config.sample_rate_hz(), render_config.num_channels(),
+      render_config.sample_rate_hz(), 1, render_config.sample_rate_hz(), 1);
+  test::InputAudioFile render_file(
+      test::GetApmRenderTestVectorFileName(sample_rate_hz));
+  std::vector<float> render_input(samples_per_channel * num_channels);
+
+  const StreamConfig capture_config(sample_rate_hz, num_channels);
+  AudioBuffer capture_buffer(
+      capture_config.sample_rate_hz(), capture_config.num_channels(),
+      capture_config.sample_rate_hz(), 1, capture_config.sample_rate_hz(), 1);
+  test::InputAudioFile capture_file(
+      test::GetApmCaptureTestVectorFileName(sample_rate_hz));
+  std::vector<float> capture_input(samples_per_channel * num_channels);
+
+  for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
+    ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
+                                   &render_file, render_input);
+    ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
+                                   &capture_file, capture_input);
+
+    test::CopyVectorToAudioBuffer(render_config, render_input, &render_buffer);
+    test::CopyVectorToAudioBuffer(capture_config, capture_input,
+                                  &capture_buffer);
+
+    ProcessOneFrame(sample_rate_hz, stream_delay_ms, &render_buffer,
+                    &capture_buffer, &echo_control_mobile);
+  }
+
+  // Extract and verify the test results.
+  std::vector<float> capture_output;
+  test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
+                                     &capture_output);
+
+  // Compare the output with the reference. Only the first values of the output
+  // from last frame processed are compared in order not having to specify all
+  // preceeding frames as testvectors. As the algorithm being tested has a
+  // memory, testing only the last frame implicitly also tests the preceeding
+  // frames.
+  const float kElementErrorBound = 1.0f / 32768.0f;
+  EXPECT_TRUE(test::VerifyDeinterleavedArray(
+      capture_config.num_frames(), capture_config.num_channels(),
+      output_reference, capture_output, kElementErrorBound));
+}
+
+}  // namespace
+
+// TODO(peah): Renable once the integer overflow issue in aecm_core.c:932:69
+// has been solved.
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono8kHz_LoudSpeakerPhone_CngOn_StreamDelay0) {
+  const float kOutputReference[] = {0.005280f, 0.002380f, -0.000427f};
+
+  RunBitexactnessTest(8000, 1, 0,
+                      EchoControlMobileImpl::RoutingMode::kLoudSpeakerphone,
+                      true, kOutputReference);
+}
+
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono16kHz_LoudSpeakerPhone_CngOn_StreamDelay0) {
+  const float kOutputReference[] = {0.003601f, 0.002991f, 0.001923f};
+  RunBitexactnessTest(16000, 1, 0,
+                      EchoControlMobileImpl::RoutingMode::kLoudSpeakerphone,
+                      true, kOutputReference);
+}
+
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono32kHz_LoudSpeakerPhone_CngOn_StreamDelay0) {
+  const float kOutputReference[] = {0.002258f, 0.002899f, 0.003906f};
+
+  RunBitexactnessTest(32000, 1, 0,
+                      EchoControlMobileImpl::RoutingMode::kLoudSpeakerphone,
+                      true, kOutputReference);
+}
+
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono48kHz_LoudSpeakerPhone_CngOn_StreamDelay0) {
+  const float kOutputReference[] = {-0.000046f, 0.000041f, 0.000249f};
+
+  RunBitexactnessTest(48000, 1, 0,
+                      EchoControlMobileImpl::RoutingMode::kLoudSpeakerphone,
+                      true, kOutputReference);
+}
+
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono16kHz_LoudSpeakerPhone_CngOff_StreamDelay0) {
+  const float kOutputReference[] = {0.000000f, 0.000000f, 0.000000f};
+
+  RunBitexactnessTest(16000, 1, 0,
+                      EchoControlMobileImpl::RoutingMode::kLoudSpeakerphone,
+                      false, kOutputReference);
+}
+
+// TODO(peah): Renable once the integer overflow issue in aecm_core.c:932:69
+// has been solved.
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono16kHz_LoudSpeakerPhone_CngOn_StreamDelay5) {
+  const float kOutputReference[] = {0.003693f, 0.002930f, 0.001801f};
+
+  RunBitexactnessTest(16000, 1, 5,
+                      EchoControlMobileImpl::RoutingMode::kLoudSpeakerphone,
+                      true, kOutputReference);
+}
+
+TEST(EchoControlMobileBitExactnessTest,
+     Mono16kHz_LoudSpeakerPhone_CngOn_StreamDelay10) {
+  const float kOutputReference[] = {-0.002380f, -0.002533f, -0.002563f};
+
+  RunBitexactnessTest(16000, 1, 10,
+                      EchoControlMobileImpl::RoutingMode::kLoudSpeakerphone,
+                      true, kOutputReference);
+}
+
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono16kHz_QuietEarpieceOrHeadset_CngOn_StreamDelay0) {
+  const float kOutputReference[] = {0.000397f, 0.000000f, -0.000305f};
+
+  RunBitexactnessTest(
+      16000, 1, 0, EchoControlMobileImpl::RoutingMode::kQuietEarpieceOrHeadset,
+      true, kOutputReference);
+}
+
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono16kHz_Earpiece_CngOn_StreamDelay0) {
+  const float kOutputReference[] = {0.002167f, 0.001617f, 0.001038f};
+
+  RunBitexactnessTest(16000, 1, 0,
+                      EchoControlMobileImpl::RoutingMode::kEarpiece, true,
+                      kOutputReference);
+}
+
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono16kHz_LoudEarpiece_CngOn_StreamDelay0) {
+  const float kOutputReference[] = {0.003540f, 0.002899f, 0.001862f};
+
+  RunBitexactnessTest(16000, 1, 0,
+                      EchoControlMobileImpl::RoutingMode::kLoudEarpiece, true,
+                      kOutputReference);
+}
+
+TEST(EchoControlMobileBitExactnessTest,
+     DISABLED_Mono16kHz_SpeakerPhone_CngOn_StreamDelay0) {
+  const float kOutputReference[] = {0.003632f, 0.003052f, 0.001984f};
+
+  RunBitexactnessTest(16000, 1, 0,
+                      EchoControlMobileImpl::RoutingMode::kSpeakerphone, true,
+                      kOutputReference);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_impl.cc b/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_impl.cc
new file mode 100644
index 0000000000..fa5cb8ffec
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_impl.cc
@@ -0,0 +1,287 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_control_mobile_impl.h"
+
+#include <string.h>
+
+#include <cstdint>
+
+#include "modules/audio_processing/aecm/echo_control_mobile.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+int16_t MapSetting(EchoControlMobileImpl::RoutingMode mode) {
+  switch (mode) {
+    case EchoControlMobileImpl::kQuietEarpieceOrHeadset:
+      return 0;
+    case EchoControlMobileImpl::kEarpiece:
+      return 1;
+    case EchoControlMobileImpl::kLoudEarpiece:
+      return 2;
+    case EchoControlMobileImpl::kSpeakerphone:
+      return 3;
+    case EchoControlMobileImpl::kLoudSpeakerphone:
+      return 4;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return -1;
+}
+
+AudioProcessing::Error MapError(int err) {
+  switch (err) {
+    case AECM_UNSUPPORTED_FUNCTION_ERROR:
+      return AudioProcessing::kUnsupportedFunctionError;
+    case AECM_NULL_POINTER_ERROR:
+      return AudioProcessing::kNullPointerError;
+    case AECM_BAD_PARAMETER_ERROR:
+      return AudioProcessing::kBadParameterError;
+    case AECM_BAD_PARAMETER_WARNING:
+      return AudioProcessing::kBadStreamParameterWarning;
+    default:
+      // AECM_UNSPECIFIED_ERROR
+      // AECM_UNINITIALIZED_ERROR
+      return AudioProcessing::kUnspecifiedError;
+  }
+}
+
+}  // namespace
+
+struct EchoControlMobileImpl::StreamProperties {
+  StreamProperties() = delete;
+  StreamProperties(int sample_rate_hz,
+                   size_t num_reverse_channels,
+                   size_t num_output_channels)
+      : sample_rate_hz(sample_rate_hz),
+        num_reverse_channels(num_reverse_channels),
+        num_output_channels(num_output_channels) {}
+
+  int sample_rate_hz;
+  size_t num_reverse_channels;
+  size_t num_output_channels;
+};
+
+class EchoControlMobileImpl::Canceller {
+ public:
+  Canceller() {
+    state_ = WebRtcAecm_Create();
+    RTC_CHECK(state_);
+  }
+
+  ~Canceller() {
+    RTC_DCHECK(state_);
+    WebRtcAecm_Free(state_);
+  }
+
+  Canceller(const Canceller&) = delete;
+  Canceller& operator=(const Canceller&) = delete;
+
+  void* state() {
+    RTC_DCHECK(state_);
+    return state_;
+  }
+
+  void Initialize(int sample_rate_hz) {
+    RTC_DCHECK(state_);
+    int error = WebRtcAecm_Init(state_, sample_rate_hz);
+    RTC_DCHECK_EQ(AudioProcessing::kNoError, error);
+  }
+
+ private:
+  void* state_;
+};
+
+EchoControlMobileImpl::EchoControlMobileImpl()
+    : routing_mode_(kSpeakerphone), comfort_noise_enabled_(false) {}
+
+EchoControlMobileImpl::~EchoControlMobileImpl() {}
+
+void EchoControlMobileImpl::ProcessRenderAudio(
+    rtc::ArrayView<const int16_t> packed_render_audio) {
+  RTC_DCHECK(stream_properties_);
+
+  size_t buffer_index = 0;
+  size_t num_frames_per_band =
+      packed_render_audio.size() / (stream_properties_->num_output_channels *
+                                    stream_properties_->num_reverse_channels);
+
+  for (auto& canceller : cancellers_) {
+    WebRtcAecm_BufferFarend(canceller->state(),
+                            &packed_render_audio[buffer_index],
+                            num_frames_per_band);
+
+    buffer_index += num_frames_per_band;
+  }
+}
+
+void EchoControlMobileImpl::PackRenderAudioBuffer(
+    const AudioBuffer* audio,
+    size_t num_output_channels,
+    size_t num_channels,
+    std::vector<int16_t>* packed_buffer) {
+  RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
+                audio->num_frames_per_band());
+  RTC_DCHECK_EQ(num_channels, audio->num_channels());
+
+  // The ordering convention must be followed to pass to the correct AECM.
+  packed_buffer->clear();
+  int render_channel = 0;
+  for (size_t i = 0; i < num_output_channels; i++) {
+    for (size_t j = 0; j < audio->num_channels(); j++) {
+      std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> data_to_buffer;
+      FloatS16ToS16(audio->split_bands_const(render_channel)[kBand0To8kHz],
+                    audio->num_frames_per_band(), data_to_buffer.data());
+
+      // Buffer the samples in the render queue.
+      packed_buffer->insert(
+          packed_buffer->end(), data_to_buffer.data(),
+          data_to_buffer.data() + audio->num_frames_per_band());
+      render_channel = (render_channel + 1) % audio->num_channels();
+    }
+  }
+}
+
+size_t EchoControlMobileImpl::NumCancellersRequired(
+    size_t num_output_channels,
+    size_t num_reverse_channels) {
+  return num_output_channels * num_reverse_channels;
+}
+
+int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio,
+                                               int stream_delay_ms) {
+  RTC_DCHECK(stream_properties_);
+  RTC_DCHECK_GE(160, audio->num_frames_per_band());
+  RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_output_channels);
+  RTC_DCHECK_GE(cancellers_.size(), stream_properties_->num_reverse_channels *
+                                        audio->num_channels());
+
+  int err = AudioProcessing::kNoError;
+
+  // The ordering convention must be followed to pass to the correct AECM.
+  size_t handle_index = 0;
+  for (size_t capture = 0; capture < audio->num_channels(); ++capture) {
+    // TODO(ajm): improve how this works, possibly inside AECM.
+    //            This is kind of hacked up.
+    RTC_DCHECK_LT(capture, low_pass_reference_.size());
+    const int16_t* noisy =
+        reference_copied_ ? low_pass_reference_[capture].data() : nullptr;
+
+    RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
+                  audio->num_frames_per_band());
+
+    std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> split_bands_data;
+    int16_t* split_bands = split_bands_data.data();
+    const int16_t* clean = split_bands_data.data();
+    if (audio->split_bands(capture)[kBand0To8kHz]) {
+      FloatS16ToS16(audio->split_bands(capture)[kBand0To8kHz],
+                    audio->num_frames_per_band(), split_bands_data.data());
+    } else {
+      clean = nullptr;
+      split_bands = nullptr;
+    }
+
+    if (noisy == NULL) {
+      noisy = clean;
+      clean = NULL;
+    }
+    for (size_t render = 0; render < stream_properties_->num_reverse_channels;
+         ++render) {
+      err = WebRtcAecm_Process(cancellers_[handle_index]->state(), noisy, clean,
+                               split_bands, audio->num_frames_per_band(),
+                               stream_delay_ms);
+
+      if (split_bands) {
+        S16ToFloatS16(split_bands, audio->num_frames_per_band(),
+                      audio->split_bands(capture)[kBand0To8kHz]);
+      }
+
+      if (err != AudioProcessing::kNoError) {
+        return MapError(err);
+      }
+
+      ++handle_index;
+    }
+    for (size_t band = 1u; band < audio->num_bands(); ++band) {
+      memset(audio->split_bands_f(capture)[band], 0,
+             audio->num_frames_per_band() *
+                 sizeof(audio->split_bands_f(capture)[band][0]));
+    }
+  }
+  return AudioProcessing::kNoError;
+}
+
+int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
+  if (MapSetting(mode) == -1) {
+    return AudioProcessing::kBadParameterError;
+  }
+  routing_mode_ = mode;
+  return Configure();
+}
+
+EchoControlMobileImpl::RoutingMode EchoControlMobileImpl::routing_mode() const {
+  return routing_mode_;
+}
+
+int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
+  comfort_noise_enabled_ = enable;
+  return Configure();
+}
+
+bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
+  return comfort_noise_enabled_;
+}
+
+void EchoControlMobileImpl::Initialize(int sample_rate_hz,
+                                       size_t num_reverse_channels,
+                                       size_t num_output_channels) {
+  low_pass_reference_.resize(num_output_channels);
+  for (auto& reference : low_pass_reference_) {
+    reference.fill(0);
+  }
+
+  stream_properties_.reset(new StreamProperties(
+      sample_rate_hz, num_reverse_channels, num_output_channels));
+
+  // AECM only supports 16 kHz or lower sample rates.
+  RTC_DCHECK_LE(stream_properties_->sample_rate_hz,
+                AudioProcessing::kSampleRate16kHz);
+
+  cancellers_.resize(
+      NumCancellersRequired(stream_properties_->num_output_channels,
+                            stream_properties_->num_reverse_channels));
+
+  for (auto& canceller : cancellers_) {
+    if (!canceller) {
+      canceller.reset(new Canceller());
+    }
+    canceller->Initialize(sample_rate_hz);
+  }
+  Configure();
+}
+
+int EchoControlMobileImpl::Configure() {
+  AecmConfig config;
+  config.cngMode = comfort_noise_enabled_;
+  config.echoMode = MapSetting(routing_mode_);
+  int error = AudioProcessing::kNoError;
+  for (auto& canceller : cancellers_) {
+    int handle_error = WebRtcAecm_set_config(canceller->state(), config);
+    if (handle_error != AudioProcessing::kNoError) {
+      error = handle_error;
+    }
+  }
+  return error;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_impl.h b/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_impl.h
new file mode 100644
index 0000000000..f7f2626a0e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_impl.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+class AudioBuffer;
+
+// The acoustic echo control for mobile (AECM) component is a low complexity
+// robust option intended for use on mobile devices.
+class EchoControlMobileImpl {
+ public:
+  EchoControlMobileImpl();
+
+  ~EchoControlMobileImpl();
+
+  // Recommended settings for particular audio routes. In general, the louder
+  // the echo is expected to be, the higher this value should be set. The
+  // preferred setting may vary from device to device.
+  enum RoutingMode {
+    kQuietEarpieceOrHeadset,
+    kEarpiece,
+    kLoudEarpiece,
+    kSpeakerphone,
+    kLoudSpeakerphone
+  };
+
+  // Sets echo control appropriate for the audio routing `mode` on the device.
+  // It can and should be updated during a call if the audio routing changes.
+  int set_routing_mode(RoutingMode mode);
+  RoutingMode routing_mode() const;
+
+  // Comfort noise replaces suppressed background noise to maintain a
+  // consistent signal level.
+  int enable_comfort_noise(bool enable);
+  bool is_comfort_noise_enabled() const;
+
+  void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
+  int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
+
+  void Initialize(int sample_rate_hz,
+                  size_t num_reverse_channels,
+                  size_t num_output_channels);
+
+  static void PackRenderAudioBuffer(const AudioBuffer* audio,
+                                    size_t num_output_channels,
+                                    size_t num_channels,
+                                    std::vector<int16_t>* packed_buffer);
+
+  static size_t NumCancellersRequired(size_t num_output_channels,
+                                      size_t num_reverse_channels);
+
+ private:
+  class Canceller;
+  struct StreamProperties;
+
+  int Configure();
+
+  RoutingMode routing_mode_;
+  bool comfort_noise_enabled_;
+
+  std::vector<std::unique_ptr<Canceller>> cancellers_;
+  std::unique_ptr<StreamProperties> stream_properties_;
+  std::vector<std::array<int16_t, 160>> low_pass_reference_;
+  bool reference_copied_ = false;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_unittest.cc b/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_unittest.cc
new file mode 100644
index 0000000000..ed0393043c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_control_mobile_unittest.cc
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <array>
+#include <vector>
+
+#include "modules/audio_processing/echo_control_mobile_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+TEST(EchoControlMobileTest, InterfaceConfiguration) {
+  EchoControlMobileImpl aecm;
+  aecm.Initialize(AudioProcessing::kSampleRate16kHz, 2, 2);
+
+  // Toggle routing modes
+  std::array<EchoControlMobileImpl::RoutingMode, 5> routing_modes = {
+      EchoControlMobileImpl::kQuietEarpieceOrHeadset,
+      EchoControlMobileImpl::kEarpiece,
+      EchoControlMobileImpl::kLoudEarpiece,
+      EchoControlMobileImpl::kSpeakerphone,
+      EchoControlMobileImpl::kLoudSpeakerphone,
+  };
+  for (auto mode : routing_modes) {
+    EXPECT_EQ(0, aecm.set_routing_mode(mode));
+    EXPECT_EQ(mode, aecm.routing_mode());
+  }
+
+  // Turn comfort noise off/on
+  EXPECT_EQ(0, aecm.enable_comfort_noise(false));
+  EXPECT_FALSE(aecm.is_comfort_noise_enabled());
+  EXPECT_EQ(0, aecm.enable_comfort_noise(true));
+  EXPECT_TRUE(aecm.is_comfort_noise_enabled());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/circular_buffer.cc b/third_party/libwebrtc/modules/audio_processing/echo_detector/circular_buffer.cc
new file mode 100644
index 0000000000..a6d10edfe2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/circular_buffer.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_detector/circular_buffer.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+CircularBuffer::CircularBuffer(size_t size) : buffer_(size) {}
+CircularBuffer::~CircularBuffer() = default;
+
+void CircularBuffer::Push(float value) {
+  buffer_[next_insertion_index_] = value;
+  ++next_insertion_index_;
+  next_insertion_index_ %= buffer_.size();
+  RTC_DCHECK_LT(next_insertion_index_, buffer_.size());
+  nr_elements_in_buffer_ = std::min(nr_elements_in_buffer_ + 1, buffer_.size());
+  RTC_DCHECK_LE(nr_elements_in_buffer_, buffer_.size());
+}
+
+absl::optional<float> CircularBuffer::Pop() {
+  if (nr_elements_in_buffer_ == 0) {
+    return absl::nullopt;
+  }
+  const size_t index =
+      (buffer_.size() + next_insertion_index_ - nr_elements_in_buffer_) %
+      buffer_.size();
+  RTC_DCHECK_LT(index, buffer_.size());
+  --nr_elements_in_buffer_;
+  return buffer_[index];
+}
+
+void CircularBuffer::Clear() {
+  std::fill(buffer_.begin(), buffer_.end(), 0.f);
+  next_insertion_index_ = 0;
+  nr_elements_in_buffer_ = 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/circular_buffer.h b/third_party/libwebrtc/modules/audio_processing/echo_detector/circular_buffer.h
new file mode 100644
index 0000000000..db1aeaebf6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/circular_buffer.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_CIRCULAR_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_CIRCULAR_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+// Ring buffer containing floating point values.
+struct CircularBuffer {
+ public:
+  explicit CircularBuffer(size_t size);
+  ~CircularBuffer();
+
+  void Push(float value);
+  absl::optional<float> Pop();
+  size_t Size() const { return nr_elements_in_buffer_; }
+  // This function fills the buffer with zeros, but does not change its size.
+  void Clear();
+
+ private:
+  std::vector<float> buffer_;
+  size_t next_insertion_index_ = 0;
+  // This is the number of elements that have been pushed into the circular
+  // buffer, not the allocated buffer size.
+  size_t nr_elements_in_buffer_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_CIRCULAR_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/circular_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
new file mode 100644
index 0000000000..7a234d4a55
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_detector/circular_buffer.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(CircularBufferTests, LessThanMaxTest) {
+  CircularBuffer test_buffer(3);
+  test_buffer.Push(1.f);
+  test_buffer.Push(2.f);
+  EXPECT_EQ(1.f, test_buffer.Pop());
+  EXPECT_EQ(2.f, test_buffer.Pop());
+}
+
+TEST(CircularBufferTests, FillTest) {
+  CircularBuffer test_buffer(3);
+  test_buffer.Push(1.f);
+  test_buffer.Push(2.f);
+  test_buffer.Push(3.f);
+  EXPECT_EQ(1.f, test_buffer.Pop());
+  EXPECT_EQ(2.f, test_buffer.Pop());
+  EXPECT_EQ(3.f, test_buffer.Pop());
+}
+
+TEST(CircularBufferTests, OverflowTest) {
+  CircularBuffer test_buffer(3);
+  test_buffer.Push(1.f);
+  test_buffer.Push(2.f);
+  test_buffer.Push(3.f);
+  test_buffer.Push(4.f);
+  // Because the circular buffer has a size of 3, the first insert should have
+  // been forgotten.
+  EXPECT_EQ(2.f, test_buffer.Pop());
+  EXPECT_EQ(3.f, test_buffer.Pop());
+  EXPECT_EQ(4.f, test_buffer.Pop());
+}
+
+TEST(CircularBufferTests, ReadFromEmpty) {
+  CircularBuffer test_buffer(3);
+  EXPECT_EQ(absl::nullopt, test_buffer.Pop());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/mean_variance_estimator.cc b/third_party/libwebrtc/modules/audio_processing/echo_detector/mean_variance_estimator.cc
new file mode 100644
index 0000000000..a9ebb8cd92
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/mean_variance_estimator.cc
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_detector/mean_variance_estimator.h"
+
+#include <cmath>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+// Parameter controlling the adaptation speed.
+constexpr float kAlpha = 0.001f;
+
+}  // namespace
+
+void MeanVarianceEstimator::Update(float value) {
+  mean_ = (1.f - kAlpha) * mean_ + kAlpha * value;
+  variance_ =
+      (1.f - kAlpha) * variance_ + kAlpha * (value - mean_) * (value - mean_);
+  RTC_DCHECK(std::isfinite(mean_));
+  RTC_DCHECK(std::isfinite(variance_));
+}
+
+float MeanVarianceEstimator::std_deviation() const {
+  RTC_DCHECK_GE(variance_, 0.f);
+  return sqrtf(variance_);
+}
+
+float MeanVarianceEstimator::mean() const {
+  return mean_;
+}
+
+void MeanVarianceEstimator::Clear() {
+  mean_ = 0.f;
+  variance_ = 0.f;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h b/third_party/libwebrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h
new file mode 100644
index 0000000000..7f793df1e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_MEAN_VARIANCE_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_MEAN_VARIANCE_ESTIMATOR_H_
+
+namespace webrtc {
+
+// This class iteratively estimates the mean and variance of a signal.
+class MeanVarianceEstimator {
+ public:
+  void Update(float value);
+  float std_deviation() const;
+  float mean() const;
+  void Clear();
+
+ private:
+  // Estimate of the expected value of the input values.
+  float mean_ = 0.f;
+  // Estimate of the variance of the input values.
+  float variance_ = 0.f;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_MEAN_VARIANCE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/mean_variance_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/echo_detector/mean_variance_estimator_unittest.cc
new file mode 100644
index 0000000000..8327d23e8a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/mean_variance_estimator_unittest.cc
@@ -0,0 +1,65 @@
+
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_detector/mean_variance_estimator.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(MeanVarianceEstimatorTests, InsertTwoValues) {
+  MeanVarianceEstimator test_estimator;
+  // Insert two values.
+  test_estimator.Update(3.f);
+  test_estimator.Update(5.f);
+
+  EXPECT_GT(test_estimator.mean(), 0.f);
+  EXPECT_GT(test_estimator.std_deviation(), 0.f);
+  // Test Clear method
+  test_estimator.Clear();
+  EXPECT_EQ(test_estimator.mean(), 0.f);
+  EXPECT_EQ(test_estimator.std_deviation(), 0.f);
+}
+
+TEST(MeanVarianceEstimatorTests, InsertZeroes) {
+  MeanVarianceEstimator test_estimator;
+  // Insert the same value many times.
+  for (size_t i = 0; i < 20000; i++) {
+    test_estimator.Update(0.f);
+  }
+  EXPECT_EQ(test_estimator.mean(), 0.f);
+  EXPECT_EQ(test_estimator.std_deviation(), 0.f);
+}
+
+TEST(MeanVarianceEstimatorTests, ConstantValueTest) {
+  MeanVarianceEstimator test_estimator;
+  for (size_t i = 0; i < 20000; i++) {
+    test_estimator.Update(3.f);
+  }
+  // The mean should be close to three, and the standard deviation should be
+  // close to zero.
+  EXPECT_NEAR(3.0f, test_estimator.mean(), 0.01f);
+  EXPECT_NEAR(0.0f, test_estimator.std_deviation(), 0.01f);
+}
+
+TEST(MeanVarianceEstimatorTests, AlternatingValueTest) {
+  MeanVarianceEstimator test_estimator;
+  for (size_t i = 0; i < 20000; i++) {
+    test_estimator.Update(1.f);
+    test_estimator.Update(-1.f);
+  }
+  // The mean should be close to zero, and the standard deviation should be
+  // close to one.
+  EXPECT_NEAR(0.0f, test_estimator.mean(), 0.01f);
+  EXPECT_NEAR(1.0f, test_estimator.std_deviation(), 0.01f);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/moving_max.cc b/third_party/libwebrtc/modules/audio_processing/echo_detector/moving_max.cc
new file mode 100644
index 0000000000..3054e98bd3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/moving_max.cc
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_detector/moving_max.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+// Parameter for controlling how fast the estimated maximum decays after the
+// previous maximum is no longer valid. With a value of 0.99, the maximum will
+// decay to 1% of its former value after 460 updates.
+constexpr float kDecayFactor = 0.99f;
+
+}  // namespace
+
+MovingMax::MovingMax(size_t window_size) : window_size_(window_size) {
+  RTC_DCHECK_GT(window_size, 0);
+}
+
+MovingMax::~MovingMax() {}
+
+void MovingMax::Update(float value) {
+  if (counter_ >= window_size_ - 1) {
+    max_value_ *= kDecayFactor;
+  } else {
+    ++counter_;
+  }
+  if (value > max_value_) {
+    max_value_ = value;
+    counter_ = 0;
+  }
+}
+
+float MovingMax::max() const {
+  return max_value_;
+}
+
+void MovingMax::Clear() {
+  max_value_ = 0.f;
+  counter_ = 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/moving_max.h b/third_party/libwebrtc/modules/audio_processing/echo_detector/moving_max.h
new file mode 100644
index 0000000000..f7d8ee8137
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/moving_max.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_MOVING_MAX_H_
+#define MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_MOVING_MAX_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+
+class MovingMax {
+ public:
+  explicit MovingMax(size_t window_size);
+  ~MovingMax();
+
+  void Update(float value);
+  float max() const;
+  // Reset all of the state in this class.
+  void Clear();
+
+ private:
+  float max_value_ = 0.f;
+  size_t counter_ = 0;
+  size_t window_size_ = 1;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_MOVING_MAX_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/moving_max_unittest.cc b/third_party/libwebrtc/modules/audio_processing/echo_detector/moving_max_unittest.cc
new file mode 100644
index 0000000000..9429127a2b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/moving_max_unittest.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_detector/moving_max.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Test if the maximum is correctly found.
+TEST(MovingMaxTests, SimpleTest) {
+  MovingMax test_moving_max(5);
+  test_moving_max.Update(1.0f);
+  test_moving_max.Update(1.1f);
+  test_moving_max.Update(1.9f);
+  test_moving_max.Update(1.87f);
+  test_moving_max.Update(1.89f);
+  EXPECT_EQ(1.9f, test_moving_max.max());
+}
+
+// Test if values fall out of the window when expected.
+TEST(MovingMaxTests, SlidingWindowTest) {
+  MovingMax test_moving_max(5);
+  test_moving_max.Update(1.0f);
+  test_moving_max.Update(1.9f);
+  test_moving_max.Update(1.7f);
+  test_moving_max.Update(1.87f);
+  test_moving_max.Update(1.89f);
+  test_moving_max.Update(1.3f);
+  test_moving_max.Update(1.2f);
+  EXPECT_LT(test_moving_max.max(), 1.9f);
+}
+
+// Test if Clear() works as expected.
+TEST(MovingMaxTests, ClearTest) {
+  MovingMax test_moving_max(5);
+  test_moving_max.Update(1.0f);
+  test_moving_max.Update(1.1f);
+  test_moving_max.Update(1.9f);
+  test_moving_max.Update(1.87f);
+  test_moving_max.Update(1.89f);
+  EXPECT_EQ(1.9f, test_moving_max.max());
+  test_moving_max.Clear();
+  EXPECT_EQ(0.f, test_moving_max.max());
+}
+
+// Test the decay of the estimated maximum.
+TEST(MovingMaxTests, DecayTest) {
+  MovingMax test_moving_max(1);
+  test_moving_max.Update(1.0f);
+  float previous_value = 1.0f;
+  for (int i = 0; i < 500; i++) {
+    test_moving_max.Update(0.0f);
+    EXPECT_LT(test_moving_max.max(), previous_value);
+    EXPECT_GT(test_moving_max.max(), 0.0f);
+    previous_value = test_moving_max.max();
+  }
+  EXPECT_LT(test_moving_max.max(), 0.01f);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc b/third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc
new file mode 100644
index 0000000000..8ec9fe9f0b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
+
+#include <math.h>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+// Parameter controlling the adaptation speed.
+constexpr float kAlpha = 0.001f;
+
+}  // namespace
+
+void NormalizedCovarianceEstimator::Update(float x,
+                                           float x_mean,
+                                           float x_sigma,
+                                           float y,
+                                           float y_mean,
+                                           float y_sigma) {
+  covariance_ =
+      (1.f - kAlpha) * covariance_ + kAlpha * (x - x_mean) * (y - y_mean);
+  normalized_cross_correlation_ = covariance_ / (x_sigma * y_sigma + .0001f);
+  RTC_DCHECK(isfinite(covariance_));
+  RTC_DCHECK(isfinite(normalized_cross_correlation_));
+}
+
+void NormalizedCovarianceEstimator::Clear() {
+  covariance_ = 0.f;
+  normalized_cross_correlation_ = 0.f;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h b/third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h
new file mode 100644
index 0000000000..e3c36d88ba
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_NORMALIZED_COVARIANCE_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_NORMALIZED_COVARIANCE_ESTIMATOR_H_
+
+namespace webrtc {
+
+// This class iteratively estimates the normalized covariance between two
+// signals.
+class NormalizedCovarianceEstimator {
+ public:
+  void Update(float x,
+              float x_mean,
+              float x_var,
+              float y,
+              float y_mean,
+              float y_var);
+  // This function returns an estimate of the Pearson product-moment correlation
+  // coefficient of the two signals.
+  float normalized_cross_correlation() const {
+    return normalized_cross_correlation_;
+  }
+  float covariance() const { return covariance_; }
+  // This function resets the estimated values to zero.
+  void Clear();
+
+ private:
+  float normalized_cross_correlation_ = 0.f;
+  // Estimate of the covariance value.
+  float covariance_ = 0.f;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_NORMALIZED_COVARIANCE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc
new file mode 100644
index 0000000000..89fb9383f6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc
@@ -0,0 +1,41 @@
+
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(NormalizedCovarianceEstimatorTests, IdenticalSignalTest) {
+  NormalizedCovarianceEstimator test_estimator;
+  for (size_t i = 0; i < 10000; i++) {
+    test_estimator.Update(1.f, 0.f, 1.f, 1.f, 0.f, 1.f);
+    test_estimator.Update(-1.f, 0.f, 1.f, -1.f, 0.f, 1.f);
+  }
+  // A normalized covariance value close to 1 is expected.
+  EXPECT_NEAR(1.f, test_estimator.normalized_cross_correlation(), 0.01f);
+  test_estimator.Clear();
+  EXPECT_EQ(0.f, test_estimator.normalized_cross_correlation());
+}
+
+TEST(NormalizedCovarianceEstimatorTests, OppositeSignalTest) {
+  NormalizedCovarianceEstimator test_estimator;
+  // Insert the same value many times.
+  for (size_t i = 0; i < 10000; i++) {
+    test_estimator.Update(1.f, 0.f, 1.f, -1.f, 0.f, 1.f);
+    test_estimator.Update(-1.f, 0.f, 1.f, 1.f, 0.f, 1.f);
+  }
+  // A normalized covariance value close to -1 is expected.
+  EXPECT_NEAR(-1.f, test_estimator.normalized_cross_correlation(), 0.01f);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/g3doc/audio_processing_module.md b/third_party/libwebrtc/modules/audio_processing/g3doc/audio_processing_module.md
new file mode 100644
index 0000000000..a77f62fbaf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/g3doc/audio_processing_module.md
@@ -0,0 +1,26 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'peah' reviewed: '2021-04-13'} *-->
+
+# Audio Processing Module (APM)
+
+## Overview
+
+The APM is responsible for applying speech enhancements effects to the
+microphone signal. These effects are required for VoIP calling and some
+examples include echo cancellation (AEC), noise suppression (NS) and
+automatic gain control (AGC).
+
+The API for APM resides in [`/modules/audio_processing/include`][https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_processing/include].
+APM is created using the [`AudioProcessingBuilder`][https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_processing/include/audio_processing.h]
+builder that allows it to be customized and configured.
+
+Some specific aspects of APM include that:
+*  APM is fully thread-safe in that it can be accessed concurrently from
+   different threads.
+*  APM handles for any input sample rates < 384 kHz and achieves this by
+   automatic reconfiguration whenever a new sample format is observed.
+*  APM handles any number of microphone channels and loudspeaker channels, with
+   the same automatic reconfiguration as for the sample rates.
+
+
+APM can either be used as part of the WebRTC native pipeline, or standalone.
diff --git a/third_party/libwebrtc/modules/audio_processing/gain_control_impl.cc b/third_party/libwebrtc/modules/audio_processing/gain_control_impl.cc
new file mode 100644
index 0000000000..edc49d1401
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_control_impl.cc
@@ -0,0 +1,373 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/gain_control_impl.h"
+
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/agc/legacy/gain_control.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+typedef void Handle;
+
+namespace {
+int16_t MapSetting(GainControl::Mode mode) {
+  switch (mode) {
+    case GainControl::kAdaptiveAnalog:
+      return kAgcModeAdaptiveAnalog;
+    case GainControl::kAdaptiveDigital:
+      return kAgcModeAdaptiveDigital;
+    case GainControl::kFixedDigital:
+      return kAgcModeFixedDigital;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return -1;
+}
+
+// Applies the sub-frame `gains` to all the bands in `out` and clamps the output
+// in the signed 16 bit range.
+void ApplyDigitalGain(const int32_t gains[11],
+                      size_t num_bands,
+                      float* const* out) {
+  constexpr float kScaling = 1.f / 65536.f;
+  constexpr int kNumSubSections = 16;
+  constexpr float kOneByNumSubSections = 1.f / kNumSubSections;
+
+  float gains_scaled[11];
+  for (int k = 0; k < 11; ++k) {
+    gains_scaled[k] = gains[k] * kScaling;
+  }
+
+  for (size_t b = 0; b < num_bands; ++b) {
+    float* out_band = out[b];
+    for (int k = 0, sample = 0; k < 10; ++k) {
+      const float delta =
+          (gains_scaled[k + 1] - gains_scaled[k]) * kOneByNumSubSections;
+      float gain = gains_scaled[k];
+      for (int n = 0; n < kNumSubSections; ++n, ++sample) {
+        RTC_DCHECK_EQ(k * kNumSubSections + n, sample);
+        out_band[sample] *= gain;
+        out_band[sample] =
+            std::min(32767.f, std::max(-32768.f, out_band[sample]));
+        gain += delta;
+      }
+    }
+  }
+}
+
+}  // namespace
+
+struct GainControlImpl::MonoAgcState {
+  MonoAgcState() {
+    state = WebRtcAgc_Create();
+    RTC_CHECK(state);
+  }
+
+  ~MonoAgcState() {
+    RTC_DCHECK(state);
+    WebRtcAgc_Free(state);
+  }
+
+  MonoAgcState(const MonoAgcState&) = delete;
+  MonoAgcState& operator=(const MonoAgcState&) = delete;
+  int32_t gains[11];
+  Handle* state;
+};
+
+int GainControlImpl::instance_counter_ = 0;
+
+GainControlImpl::GainControlImpl()
+    : data_dumper_(new ApmDataDumper(instance_counter_)),
+      mode_(kAdaptiveAnalog),
+      minimum_capture_level_(0),
+      maximum_capture_level_(255),
+      limiter_enabled_(true),
+      target_level_dbfs_(3),
+      compression_gain_db_(9),
+      analog_capture_level_(0),
+      was_analog_level_set_(false),
+      stream_is_saturated_(false) {}
+
+GainControlImpl::~GainControlImpl() = default;
+
+void GainControlImpl::ProcessRenderAudio(
+    rtc::ArrayView<const int16_t> packed_render_audio) {
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    WebRtcAgc_AddFarend(mono_agcs_[ch]->state, packed_render_audio.data(),
+                        packed_render_audio.size());
+  }
+}
+
+void GainControlImpl::PackRenderAudioBuffer(
+    const AudioBuffer& audio,
+    std::vector<int16_t>* packed_buffer) {
+  RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, audio.num_frames_per_band());
+  std::array<int16_t, AudioBuffer::kMaxSplitFrameLength>
+      mixed_16_kHz_render_data;
+  rtc::ArrayView<const int16_t> mixed_16_kHz_render(
+      mixed_16_kHz_render_data.data(), audio.num_frames_per_band());
+  if (audio.num_channels() == 1) {
+    FloatS16ToS16(audio.split_bands_const(0)[kBand0To8kHz],
+                  audio.num_frames_per_band(), mixed_16_kHz_render_data.data());
+  } else {
+    const int num_channels = static_cast<int>(audio.num_channels());
+    for (size_t i = 0; i < audio.num_frames_per_band(); ++i) {
+      int32_t sum = 0;
+      for (int ch = 0; ch < num_channels; ++ch) {
+        sum += FloatS16ToS16(audio.split_channels_const(kBand0To8kHz)[ch][i]);
+      }
+      mixed_16_kHz_render_data[i] = sum / num_channels;
+    }
+  }
+
+  packed_buffer->clear();
+  packed_buffer->insert(
+      packed_buffer->end(), mixed_16_kHz_render.data(),
+      (mixed_16_kHz_render.data() + audio.num_frames_per_band()));
+}
+
+int GainControlImpl::AnalyzeCaptureAudio(const AudioBuffer& audio) {
+  RTC_DCHECK(num_proc_channels_);
+  RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, audio.num_frames_per_band());
+  RTC_DCHECK_EQ(audio.num_channels(), *num_proc_channels_);
+  RTC_DCHECK_LE(*num_proc_channels_, mono_agcs_.size());
+
+  int16_t split_band_data[AudioBuffer::kMaxNumBands]
+                         [AudioBuffer::kMaxSplitFrameLength];
+  int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
+      split_band_data[0], split_band_data[1], split_band_data[2]};
+
+  if (mode_ == kAdaptiveAnalog) {
+    for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+      capture_levels_[ch] = analog_capture_level_;
+
+      audio.ExportSplitChannelData(ch, split_bands);
+
+      int err =
+          WebRtcAgc_AddMic(mono_agcs_[ch]->state, split_bands,
+                           audio.num_bands(), audio.num_frames_per_band());
+
+      if (err != AudioProcessing::kNoError) {
+        return AudioProcessing::kUnspecifiedError;
+      }
+    }
+  } else if (mode_ == kAdaptiveDigital) {
+    for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+      int32_t capture_level_out = 0;
+
+      audio.ExportSplitChannelData(ch, split_bands);
+
+      int err =
+          WebRtcAgc_VirtualMic(mono_agcs_[ch]->state, split_bands,
+                               audio.num_bands(), audio.num_frames_per_band(),
+                               analog_capture_level_, &capture_level_out);
+
+      capture_levels_[ch] = capture_level_out;
+
+      if (err != AudioProcessing::kNoError) {
+        return AudioProcessing::kUnspecifiedError;
+      }
+    }
+  }
+
+  return AudioProcessing::kNoError;
+}
+
+int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
+                                         bool stream_has_echo) {
+  if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
+    return AudioProcessing::kStreamParameterNotSetError;
+  }
+
+  RTC_DCHECK(num_proc_channels_);
+  RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
+                audio->num_frames_per_band());
+  RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
+
+  stream_is_saturated_ = false;
+  bool error_reported = false;
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    int16_t split_band_data[AudioBuffer::kMaxNumBands]
+                           [AudioBuffer::kMaxSplitFrameLength];
+    int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
+        split_band_data[0], split_band_data[1], split_band_data[2]};
+    audio->ExportSplitChannelData(ch, split_bands);
+
+    // The call to stream_has_echo() is ok from a deadlock perspective
+    // as the capture lock is allready held.
+    int32_t new_capture_level = 0;
+    uint8_t saturation_warning = 0;
+    int err_analyze = WebRtcAgc_Analyze(
+        mono_agcs_[ch]->state, split_bands, audio->num_bands(),
+        audio->num_frames_per_band(), capture_levels_[ch], &new_capture_level,
+        stream_has_echo, &saturation_warning, mono_agcs_[ch]->gains);
+    capture_levels_[ch] = new_capture_level;
+
+    error_reported = error_reported || err_analyze != AudioProcessing::kNoError;
+
+    stream_is_saturated_ = stream_is_saturated_ || saturation_warning == 1;
+  }
+
+  // Choose the minimun gain for application
+  size_t index_to_apply = 0;
+  for (size_t ch = 1; ch < mono_agcs_.size(); ++ch) {
+    if (mono_agcs_[index_to_apply]->gains[10] < mono_agcs_[ch]->gains[10]) {
+      index_to_apply = ch;
+    }
+  }
+
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    ApplyDigitalGain(mono_agcs_[index_to_apply]->gains, audio->num_bands(),
+                     audio->split_bands(ch));
+  }
+
+  RTC_DCHECK_LT(0ul, *num_proc_channels_);
+  if (mode_ == kAdaptiveAnalog) {
+    // Take the analog level to be the minimum accross all channels.
+    analog_capture_level_ = capture_levels_[0];
+    for (size_t ch = 1; ch < mono_agcs_.size(); ++ch) {
+      analog_capture_level_ =
+          std::min(analog_capture_level_, capture_levels_[ch]);
+    }
+  }
+
+  if (error_reported) {
+    return AudioProcessing::kUnspecifiedError;
+  }
+
+  was_analog_level_set_ = false;
+
+  return AudioProcessing::kNoError;
+}
+
+
+// TODO(ajm): ensure this is called under kAdaptiveAnalog.
+int GainControlImpl::set_stream_analog_level(int level) {
+  data_dumper_->DumpRaw("gain_control_set_stream_analog_level", 1, &level);
+
+  was_analog_level_set_ = true;
+  if (level < minimum_capture_level_ || level > maximum_capture_level_) {
+    return AudioProcessing::kBadParameterError;
+  }
+  analog_capture_level_ = level;
+
+  return AudioProcessing::kNoError;
+}
+
+int GainControlImpl::stream_analog_level() const {
+  data_dumper_->DumpRaw("gain_control_stream_analog_level", 1,
+                        &analog_capture_level_);
+  return analog_capture_level_;
+}
+
+int GainControlImpl::set_mode(Mode mode) {
+  if (MapSetting(mode) == -1) {
+    return AudioProcessing::kBadParameterError;
+  }
+
+  mode_ = mode;
+  RTC_DCHECK(num_proc_channels_);
+  RTC_DCHECK(sample_rate_hz_);
+  Initialize(*num_proc_channels_, *sample_rate_hz_);
+  return AudioProcessing::kNoError;
+}
+
+
+int GainControlImpl::set_analog_level_limits(int minimum, int maximum) {
+  if (minimum < 0 || maximum > 65535 || maximum < minimum) {
+    return AudioProcessing::kBadParameterError;
+  }
+
+  minimum_capture_level_ = minimum;
+  maximum_capture_level_ = maximum;
+
+  RTC_DCHECK(num_proc_channels_);
+  RTC_DCHECK(sample_rate_hz_);
+  Initialize(*num_proc_channels_, *sample_rate_hz_);
+  return AudioProcessing::kNoError;
+}
+
+
+int GainControlImpl::set_target_level_dbfs(int level) {
+  if (level > 31 || level < 0) {
+    return AudioProcessing::kBadParameterError;
+  }
+  target_level_dbfs_ = level;
+  return Configure();
+}
+
+int GainControlImpl::set_compression_gain_db(int gain) {
+  if (gain < 0 || gain > 90) {
+    RTC_LOG(LS_ERROR) << "set_compression_gain_db(" << gain << ") failed.";
+    return AudioProcessing::kBadParameterError;
+  }
+  compression_gain_db_ = gain;
+  return Configure();
+}
+
+int GainControlImpl::enable_limiter(bool enable) {
+  limiter_enabled_ = enable;
+  return Configure();
+}
+
+void GainControlImpl::Initialize(size_t num_proc_channels, int sample_rate_hz) {
+  data_dumper_->InitiateNewSetOfRecordings();
+
+  RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000 ||
+             sample_rate_hz == 48000);
+
+  num_proc_channels_ = num_proc_channels;
+  sample_rate_hz_ = sample_rate_hz;
+
+  mono_agcs_.resize(*num_proc_channels_);
+  capture_levels_.resize(*num_proc_channels_);
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    if (!mono_agcs_[ch]) {
+      mono_agcs_[ch].reset(new MonoAgcState());
+    }
+
+    int error = WebRtcAgc_Init(mono_agcs_[ch]->state, minimum_capture_level_,
+                               maximum_capture_level_, MapSetting(mode_),
+                               *sample_rate_hz_);
+    RTC_DCHECK_EQ(error, 0);
+    capture_levels_[ch] = analog_capture_level_;
+  }
+
+  Configure();
+}
+
+int GainControlImpl::Configure() {
+  WebRtcAgcConfig config;
+  // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
+  //            change the interface.
+  // RTC_DCHECK_LE(target_level_dbfs_, 0);
+  // config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
+  config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
+  config.compressionGaindB = static_cast<int16_t>(compression_gain_db_);
+  config.limiterEnable = limiter_enabled_;
+
+  int error = AudioProcessing::kNoError;
+  for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
+    int error_ch = WebRtcAgc_set_config(mono_agcs_[ch]->state, config);
+    if (error_ch != AudioProcessing::kNoError) {
+      error = error_ch;
+    }
+  }
+  return error;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/gain_control_impl.h b/third_party/libwebrtc/modules/audio_processing/gain_control_impl.h
new file mode 100644
index 0000000000..8aea8f2e95
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_control_impl.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/agc/gain_control.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioBuffer;
+
+class GainControlImpl : public GainControl {
+ public:
+  GainControlImpl();
+  GainControlImpl(const GainControlImpl&) = delete;
+  GainControlImpl& operator=(const GainControlImpl&) = delete;
+
+  ~GainControlImpl() override;
+
+  void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
+  int AnalyzeCaptureAudio(const AudioBuffer& audio);
+  int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
+
+  void Initialize(size_t num_proc_channels, int sample_rate_hz);
+
+  static void PackRenderAudioBuffer(const AudioBuffer& audio,
+                                    std::vector<int16_t>* packed_buffer);
+
+  // GainControl implementation.
+  int stream_analog_level() const override;
+  bool is_limiter_enabled() const override { return limiter_enabled_; }
+  Mode mode() const override { return mode_; }
+  int set_mode(Mode mode) override;
+  int compression_gain_db() const override { return compression_gain_db_; }
+  int set_analog_level_limits(int minimum, int maximum) override;
+  int set_compression_gain_db(int gain) override;
+  int set_target_level_dbfs(int level) override;
+  int enable_limiter(bool enable) override;
+  int set_stream_analog_level(int level) override;
+
+ private:
+  struct MonoAgcState;
+
+  // GainControl implementation.
+  int target_level_dbfs() const override { return target_level_dbfs_; }
+  int analog_level_minimum() const override { return minimum_capture_level_; }
+  int analog_level_maximum() const override { return maximum_capture_level_; }
+  bool stream_is_saturated() const override { return stream_is_saturated_; }
+
+  int Configure();
+
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+
+  Mode mode_;
+  int minimum_capture_level_;
+  int maximum_capture_level_;
+  bool limiter_enabled_;
+  int target_level_dbfs_;
+  int compression_gain_db_;
+  int analog_capture_level_ = 0;
+  bool was_analog_level_set_;
+  bool stream_is_saturated_;
+
+  std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_;
+  std::vector<int> capture_levels_;
+
+  absl::optional<size_t> num_proc_channels_;
+  absl::optional<int> sample_rate_hz_;
+
+  static int instance_counter_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/gain_control_unittest.cc b/third_party/libwebrtc/modules/audio_processing/gain_control_unittest.cc
new file mode 100644
index 0000000000..1662dc506f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_control_unittest.cc
@@ -0,0 +1,393 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/gain_control_impl.h"
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+#include "modules/audio_processing/test/bitexactness_tools.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+const int kNumFramesToProcess = 100;
+
+void ProcessOneFrame(int sample_rate_hz,
+                     AudioBuffer* render_audio_buffer,
+                     AudioBuffer* capture_audio_buffer,
+                     GainControlImpl* gain_controller) {
+  if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
+    render_audio_buffer->SplitIntoFrequencyBands();
+    capture_audio_buffer->SplitIntoFrequencyBands();
+  }
+
+  std::vector<int16_t> render_audio;
+  GainControlImpl::PackRenderAudioBuffer(*render_audio_buffer, &render_audio);
+  gain_controller->ProcessRenderAudio(render_audio);
+  gain_controller->AnalyzeCaptureAudio(*capture_audio_buffer);
+  gain_controller->ProcessCaptureAudio(capture_audio_buffer, false);
+
+  if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
+    capture_audio_buffer->MergeFrequencyBands();
+  }
+}
+
+void SetupComponent(int sample_rate_hz,
+                    GainControl::Mode mode,
+                    int target_level_dbfs,
+                    int stream_analog_level,
+                    int compression_gain_db,
+                    bool enable_limiter,
+                    int analog_level_min,
+                    int analog_level_max,
+                    GainControlImpl* gain_controller) {
+  gain_controller->Initialize(1, sample_rate_hz);
+  GainControl* gc = static_cast<GainControl*>(gain_controller);
+  gc->set_mode(mode);
+  gc->set_stream_analog_level(stream_analog_level);
+  gc->set_target_level_dbfs(target_level_dbfs);
+  gc->set_compression_gain_db(compression_gain_db);
+  gc->enable_limiter(enable_limiter);
+  gc->set_analog_level_limits(analog_level_min, analog_level_max);
+}
+
+void RunBitExactnessTest(int sample_rate_hz,
+                         size_t num_channels,
+                         GainControl::Mode mode,
+                         int target_level_dbfs,
+                         int stream_analog_level,
+                         int compression_gain_db,
+                         bool enable_limiter,
+                         int analog_level_min,
+                         int analog_level_max,
+                         int achieved_stream_analog_level_reference,
+                         rtc::ArrayView<const float> output_reference) {
+  GainControlImpl gain_controller;
+  SetupComponent(sample_rate_hz, mode, target_level_dbfs, stream_analog_level,
+                 compression_gain_db, enable_limiter, analog_level_min,
+                 analog_level_max, &gain_controller);
+
+  const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
+  const StreamConfig render_config(sample_rate_hz, num_channels);
+  AudioBuffer render_buffer(
+      render_config.sample_rate_hz(), render_config.num_channels(),
+      render_config.sample_rate_hz(), 1, render_config.sample_rate_hz(), 1);
+  test::InputAudioFile render_file(
+      test::GetApmRenderTestVectorFileName(sample_rate_hz));
+  std::vector<float> render_input(samples_per_channel * num_channels);
+
+  const StreamConfig capture_config(sample_rate_hz, num_channels);
+  AudioBuffer capture_buffer(
+      capture_config.sample_rate_hz(), capture_config.num_channels(),
+      capture_config.sample_rate_hz(), 1, capture_config.sample_rate_hz(), 1);
+  test::InputAudioFile capture_file(
+      test::GetApmCaptureTestVectorFileName(sample_rate_hz));
+  std::vector<float> capture_input(samples_per_channel * num_channels);
+
+  for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
+    ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
+                                   &render_file, render_input);
+    ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
+                                   &capture_file, capture_input);
+
+    test::CopyVectorToAudioBuffer(render_config, render_input, &render_buffer);
+    test::CopyVectorToAudioBuffer(capture_config, capture_input,
+                                  &capture_buffer);
+
+    ProcessOneFrame(sample_rate_hz, &render_buffer, &capture_buffer,
+                    &gain_controller);
+  }
+
+  // Extract and verify the test results.
+  std::vector<float> capture_output;
+  test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
+                                     &capture_output);
+
+  EXPECT_EQ(achieved_stream_analog_level_reference,
+            gain_controller.stream_analog_level());
+
+  // Compare the output with the reference. Only the first values of the output
+  // from last frame processed are compared in order not having to specify all
+  // preceeding frames as testvectors. As the algorithm being tested has a
+  // memory, testing only the last frame implicitly also tests the preceeding
+  // frames.
+  const float kElementErrorBound = 1.0f / 32768.0f;
+  EXPECT_TRUE(test::VerifyDeinterleavedArray(
+      capture_config.num_frames(), capture_config.num_channels(),
+      output_reference, capture_output, kElementErrorBound));
+}
+
+}  // namespace
+
+// TODO(peah): Activate all these tests for ARM and ARM64 once the issue on the
+// Chromium ARM and ARM64 boths have been identified. This is tracked in the
+// issue https://bugs.chromium.org/p/webrtc/issues/detail?id=5711.
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.006561f, -0.004608f, -0.002899f};
+  RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Stereo16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Stereo16kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.027313f, -0.015900f, -0.028107f,
+                                    -0.027313f, -0.015900f, -0.028107f};
+  RunBitExactnessTest(16000, 2, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono32kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono32kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.010162f, -0.009155f, -0.008301f};
+  RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono48kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono48kHz_AdaptiveAnalog_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.010162f, -0.009155f, -0.008301f};
+  RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.003967f, -0.002777f, -0.001770f};
+  RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Stereo16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Stereo16kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.015411f, -0.008972f, -0.015839f,
+                                    -0.015411f, -0.008972f, -0.015839f};
+  RunBitExactnessTest(16000, 2, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono32kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono32kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.006134f, -0.005524f, -0.005005f};
+  RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono48kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono48kHz_AdaptiveDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.006134f, -0.005524f, -0.005005};
+  RunBitExactnessTest(32000, 1, GainControl::Mode::kAdaptiveDigital, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.011749f, -0.008270f, -0.005219f};
+  RunBitExactnessTest(16000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Stereo16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Stereo16kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.048896f, -0.028479f, -0.050345f,
+                                    -0.048896f, -0.028479f, -0.050345f};
+  RunBitExactnessTest(16000, 2, GainControl::Mode::kFixedDigital, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono32kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono32kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.018158f, -0.016357f, -0.014832f};
+  RunBitExactnessTest(32000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono48kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono48kHz_FixedDigital_Tl10_SL50_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 50;
+  const float kOutputReference[] = {-0.018158f, -0.016357f, -0.014832f};
+  RunBitExactnessTest(32000, 1, GainControl::Mode::kFixedDigital, 10, 50, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono16kHz_AdaptiveAnalog_Tl10_SL10_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono16kHz_AdaptiveAnalog_Tl10_SL10_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 12;
+  const float kOutputReference[] = {-0.006561f, -0.004608f, -0.002899f};
+  RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 10, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono16kHz_AdaptiveAnalog_Tl10_SL100_CG5_Lim_AL70_80) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono16kHz_AdaptiveAnalog_Tl10_SL100_CG5_Lim_AL70_80) {
+#endif
+  const int kStreamAnalogLevelReference = 100;
+  const float kOutputReference[] = {-0.003998f, -0.002808f, -0.001770f};
+  RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveAnalog, 10, 100, 5,
+                      true, 70, 80, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono16kHz_AdaptiveDigital_Tl10_SL100_CG5_NoLim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL100_CG5_NoLim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 100;
+  const float kOutputReference[] = {-0.004028f, -0.002838f, -0.001770f};
+  RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 100, 5,
+                      false, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono16kHz_AdaptiveDigital_Tl40_SL100_CG5_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono16kHz_AdaptiveDigital_Tl40_SL100_CG5_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 100;
+  const float kOutputReference[] = {-0.008728f, -0.006134f, -0.003845f};
+  RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 40, 100, 5,
+                      true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
+      defined(WEBRTC_ANDROID))
+TEST(GainControlBitExactnessTest,
+     Mono16kHz_AdaptiveDigital_Tl10_SL100_CG30_Lim_AL0_100) {
+#else
+TEST(GainControlBitExactnessTest,
+     DISABLED_Mono16kHz_AdaptiveDigital_Tl10_SL100_CG30_Lim_AL0_100) {
+#endif
+  const int kStreamAnalogLevelReference = 100;
+  const float kOutputReference[] = {-0.005859f, -0.004120f, -0.002594f};
+  RunBitExactnessTest(16000, 1, GainControl::Mode::kAdaptiveDigital, 10, 100,
+                      30, true, 0, 100, kStreamAnalogLevelReference,
+                      kOutputReference);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/gain_controller2.cc b/third_party/libwebrtc/modules/audio_processing/gain_controller2.cc
new file mode 100644
index 0000000000..dd3521268d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_controller2.cc
@@ -0,0 +1,283 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/gain_controller2.h"
+
+#include <memory>
+#include <utility>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/agc2_common.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+using Agc2Config = AudioProcessing::Config::GainController2;
+using InputVolumeControllerConfig = InputVolumeController::Config;
+
+constexpr int kLogLimiterStatsPeriodMs = 30'000;
+constexpr int kFrameLengthMs = 10;
+constexpr int kLogLimiterStatsPeriodNumFrames =
+    kLogLimiterStatsPeriodMs / kFrameLengthMs;
+
+// Detects the available CPU features and applies any kill-switches.
+AvailableCpuFeatures GetAllowedCpuFeatures() {
+  AvailableCpuFeatures features = GetAvailableCpuFeatures();
+  if (field_trial::IsEnabled("WebRTC-Agc2SimdSse2KillSwitch")) {
+    features.sse2 = false;
+  }
+  if (field_trial::IsEnabled("WebRTC-Agc2SimdAvx2KillSwitch")) {
+    features.avx2 = false;
+  }
+  if (field_trial::IsEnabled("WebRTC-Agc2SimdNeonKillSwitch")) {
+    features.neon = false;
+  }
+  return features;
+}
+
+// Peak and RMS audio levels in dBFS.
+struct AudioLevels {
+  float peak_dbfs;
+  float rms_dbfs;
+};
+
+// Speech level info.
+struct SpeechLevel {
+  bool is_confident;
+  float rms_dbfs;
+};
+
+// Computes the audio levels for the first channel in `frame`.
+AudioLevels ComputeAudioLevels(AudioFrameView<float> frame,
+                               ApmDataDumper& data_dumper) {
+  float peak = 0.0f;
+  float rms = 0.0f;
+  for (const auto& x : frame.channel(0)) {
+    peak = std::max(std::fabs(x), peak);
+    rms += x * x;
+  }
+  AudioLevels levels{
+      FloatS16ToDbfs(peak),
+      FloatS16ToDbfs(std::sqrt(rms / frame.samples_per_channel()))};
+  data_dumper.DumpRaw("agc2_input_rms_dbfs", levels.rms_dbfs);
+  data_dumper.DumpRaw("agc2_input_peak_dbfs", levels.peak_dbfs);
+  return levels;
+}
+
+}  // namespace
+
+std::atomic<int> GainController2::instance_count_(0);
+
+GainController2::GainController2(
+    const Agc2Config& config,
+    const InputVolumeControllerConfig& input_volume_controller_config,
+    int sample_rate_hz,
+    int num_channels,
+    bool use_internal_vad)
+    : cpu_features_(GetAllowedCpuFeatures()),
+      data_dumper_(instance_count_.fetch_add(1) + 1),
+      fixed_gain_applier_(
+          /*hard_clip_samples=*/false,
+          /*initial_gain_factor=*/DbToRatio(config.fixed_digital.gain_db)),
+      limiter_(sample_rate_hz, &data_dumper_, /*histogram_name_prefix=*/"Agc2"),
+      calls_since_last_limiter_log_(0) {
+  RTC_DCHECK(Validate(config));
+  data_dumper_.InitiateNewSetOfRecordings();
+
+  if (config.input_volume_controller.enabled ||
+      config.adaptive_digital.enabled) {
+    // Create dependencies.
+    speech_level_estimator_ = std::make_unique<SpeechLevelEstimator>(
+        &data_dumper_, config.adaptive_digital, kAdjacentSpeechFramesThreshold);
+    if (use_internal_vad)
+      vad_ = std::make_unique<VoiceActivityDetectorWrapper>(
+          kVadResetPeriodMs, cpu_features_, sample_rate_hz);
+  }
+
+  if (config.input_volume_controller.enabled) {
+    // Create controller.
+    input_volume_controller_ = std::make_unique<InputVolumeController>(
+        num_channels, input_volume_controller_config);
+    // TODO(bugs.webrtc.org/7494): Call `Initialize` in ctor and remove method.
+    input_volume_controller_->Initialize();
+  }
+
+  if (config.adaptive_digital.enabled) {
+    // Create dependencies.
+    noise_level_estimator_ = CreateNoiseFloorEstimator(&data_dumper_);
+    saturation_protector_ = CreateSaturationProtector(
+        kSaturationProtectorInitialHeadroomDb, kAdjacentSpeechFramesThreshold,
+        &data_dumper_);
+    // Create controller.
+    adaptive_digital_controller_ =
+        std::make_unique<AdaptiveDigitalGainController>(
+            &data_dumper_, config.adaptive_digital,
+            kAdjacentSpeechFramesThreshold);
+  }
+}
+
+GainController2::~GainController2() = default;
+
+// TODO(webrtc:7494): Pass the flag also to the other components.
+void GainController2::SetCaptureOutputUsed(bool capture_output_used) {
+  if (input_volume_controller_) {
+    input_volume_controller_->HandleCaptureOutputUsedChange(
+        capture_output_used);
+  }
+}
+
+void GainController2::SetFixedGainDb(float gain_db) {
+  const float gain_factor = DbToRatio(gain_db);
+  if (fixed_gain_applier_.GetGainFactor() != gain_factor) {
+    // Reset the limiter to quickly react on abrupt level changes caused by
+    // large changes of the fixed gain.
+    limiter_.Reset();
+  }
+  fixed_gain_applier_.SetGainFactor(gain_factor);
+}
+
+void GainController2::Analyze(int applied_input_volume,
+                              const AudioBuffer& audio_buffer) {
+  recommended_input_volume_ = absl::nullopt;
+
+  RTC_DCHECK_GE(applied_input_volume, 0);
+  RTC_DCHECK_LE(applied_input_volume, 255);
+
+  if (input_volume_controller_) {
+    input_volume_controller_->AnalyzeInputAudio(applied_input_volume,
+                                                audio_buffer);
+  }
+}
+
+void GainController2::Process(absl::optional<float> speech_probability,
+                              bool input_volume_changed,
+                              AudioBuffer* audio) {
+  recommended_input_volume_ = absl::nullopt;
+
+  data_dumper_.DumpRaw("agc2_applied_input_volume_changed",
+                       input_volume_changed);
+  if (input_volume_changed) {
+    // Handle input volume changes.
+    if (speech_level_estimator_)
+      speech_level_estimator_->Reset();
+    if (saturation_protector_)
+      saturation_protector_->Reset();
+  }
+
+  AudioFrameView<float> float_frame(audio->channels(), audio->num_channels(),
+                                    audio->num_frames());
+  // Compute speech probability.
+  if (vad_) {
+    // When the VAD component runs, `speech_probability` should not be specified
+    // because APM should not run the same VAD twice (as an APM sub-module and
+    // internally in AGC2).
+    RTC_DCHECK(!speech_probability.has_value());
+    speech_probability = vad_->Analyze(float_frame);
+  }
+  if (speech_probability.has_value()) {
+    RTC_DCHECK_GE(*speech_probability, 0.0f);
+    RTC_DCHECK_LE(*speech_probability, 1.0f);
+  }
+  // The speech probability may not be defined at this step (e.g., when the
+  // fixed digital controller alone is enabled).
+  if (speech_probability.has_value())
+    data_dumper_.DumpRaw("agc2_speech_probability", *speech_probability);
+
+  // Compute audio, noise and speech levels.
+  AudioLevels audio_levels = ComputeAudioLevels(float_frame, data_dumper_);
+  absl::optional<float> noise_rms_dbfs;
+  if (noise_level_estimator_) {
+    // TODO(bugs.webrtc.org/7494): Pass `audio_levels` to remove duplicated
+    // computation in `noise_level_estimator_`.
+    noise_rms_dbfs = noise_level_estimator_->Analyze(float_frame);
+  }
+  absl::optional<SpeechLevel> speech_level;
+  if (speech_level_estimator_) {
+    RTC_DCHECK(speech_probability.has_value());
+    speech_level_estimator_->Update(
+        audio_levels.rms_dbfs, audio_levels.peak_dbfs, *speech_probability);
+    speech_level =
+        SpeechLevel{.is_confident = speech_level_estimator_->is_confident(),
+                    .rms_dbfs = speech_level_estimator_->level_dbfs()};
+  }
+
+  // Update the recommended input volume.
+  if (input_volume_controller_) {
+    RTC_DCHECK(speech_level.has_value());
+    RTC_DCHECK(speech_probability.has_value());
+    if (speech_probability.has_value()) {
+      recommended_input_volume_ =
+          input_volume_controller_->RecommendInputVolume(
+              *speech_probability,
+              speech_level->is_confident
+                  ? absl::optional<float>(speech_level->rms_dbfs)
+                  : absl::nullopt);
+    }
+  }
+
+  if (adaptive_digital_controller_) {
+    RTC_DCHECK(saturation_protector_);
+    RTC_DCHECK(speech_probability.has_value());
+    RTC_DCHECK(speech_level.has_value());
+    saturation_protector_->Analyze(*speech_probability, audio_levels.peak_dbfs,
+                                   speech_level->rms_dbfs);
+    float headroom_db = saturation_protector_->HeadroomDb();
+    data_dumper_.DumpRaw("agc2_headroom_db", headroom_db);
+    float limiter_envelope_dbfs = FloatS16ToDbfs(limiter_.LastAudioLevel());
+    data_dumper_.DumpRaw("agc2_limiter_envelope_dbfs", limiter_envelope_dbfs);
+    RTC_DCHECK(noise_rms_dbfs.has_value());
+    adaptive_digital_controller_->Process(
+        /*info=*/{.speech_probability = *speech_probability,
+                  .speech_level_dbfs = speech_level->rms_dbfs,
+                  .speech_level_reliable = speech_level->is_confident,
+                  .noise_rms_dbfs = *noise_rms_dbfs,
+                  .headroom_db = headroom_db,
+                  .limiter_envelope_dbfs = limiter_envelope_dbfs},
+        float_frame);
+  }
+
+  // TODO(bugs.webrtc.org/7494): Pass `audio_levels` to remove duplicated
+  // computation in `limiter_`.
+  fixed_gain_applier_.ApplyGain(float_frame);
+
+  limiter_.Process(float_frame);
+
+  // Periodically log limiter stats.
+  if (++calls_since_last_limiter_log_ == kLogLimiterStatsPeriodNumFrames) {
+    calls_since_last_limiter_log_ = 0;
+    InterpolatedGainCurve::Stats stats = limiter_.GetGainCurveStats();
+    RTC_LOG(LS_INFO) << "[AGC2] limiter stats"
+                     << " | identity: " << stats.look_ups_identity_region
+                     << " | knee: " << stats.look_ups_knee_region
+                     << " | limiter: " << stats.look_ups_limiter_region
+                     << " | saturation: " << stats.look_ups_saturation_region;
+  }
+}
+
+bool GainController2::Validate(
+    const AudioProcessing::Config::GainController2& config) {
+  const auto& fixed = config.fixed_digital;
+  const auto& adaptive = config.adaptive_digital;
+  return fixed.gain_db >= 0.0f && fixed.gain_db < 50.0f &&
+         adaptive.headroom_db >= 0.0f && adaptive.max_gain_db > 0.0f &&
+         adaptive.initial_gain_db >= 0.0f &&
+         adaptive.max_gain_change_db_per_second > 0.0f &&
+         adaptive.max_output_noise_level_dbfs <= 0.0f;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/gain_controller2.h b/third_party/libwebrtc/modules/audio_processing/gain_controller2.h
new file mode 100644
index 0000000000..43b5828d35
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_controller2.h
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
+#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
+
+#include <atomic>
+#include <memory>
+#include <string>
+
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
+#include "modules/audio_processing/agc2/cpu_features.h"
+#include "modules/audio_processing/agc2/gain_applier.h"
+#include "modules/audio_processing/agc2/input_volume_controller.h"
+#include "modules/audio_processing/agc2/limiter.h"
+#include "modules/audio_processing/agc2/noise_level_estimator.h"
+#include "modules/audio_processing/agc2/saturation_protector.h"
+#include "modules/audio_processing/agc2/speech_level_estimator.h"
+#include "modules/audio_processing/agc2/vad_wrapper.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+class AudioBuffer;
+
+// Gain Controller 2 aims to automatically adjust levels by acting on the
+// microphone gain and/or applying digital gain.
+class GainController2 {
+ public:
+  // Ctor. If `use_internal_vad` is true, an internal voice activity
+  // detector is used for digital adaptive gain.
+  GainController2(
+      const AudioProcessing::Config::GainController2& config,
+      const InputVolumeController::Config& input_volume_controller_config,
+      int sample_rate_hz,
+      int num_channels,
+      bool use_internal_vad);
+  GainController2(const GainController2&) = delete;
+  GainController2& operator=(const GainController2&) = delete;
+  ~GainController2();
+
+  // Sets the fixed digital gain.
+  void SetFixedGainDb(float gain_db);
+
+  // Updates the input volume controller about whether the capture output is
+  // used or not.
+  void SetCaptureOutputUsed(bool capture_output_used);
+
+  // Analyzes `audio_buffer` before `Process()` is called so that the analysis
+  // can be performed before digital processing operations take place (e.g.,
+  // echo cancellation). The analysis consists of input clipping detection and
+  // prediction (if enabled). The value of `applied_input_volume` is limited to
+  // [0, 255].
+  void Analyze(int applied_input_volume, const AudioBuffer& audio_buffer);
+
+  // Updates the recommended input volume, applies the adaptive digital and the
+  // fixed digital gains and runs a limiter on `audio`.
+  // When the internal VAD is not used, `speech_probability` should be specified
+  // and in the [0, 1] range. Otherwise ignores `speech_probability` and
+  // computes the speech probability via `vad_`.
+  // Handles input volume changes; if the caller cannot determine whether an
+  // input volume change occurred, set `input_volume_changed` to false.
+  void Process(absl::optional<float> speech_probability,
+               bool input_volume_changed,
+               AudioBuffer* audio);
+
+  static bool Validate(const AudioProcessing::Config::GainController2& config);
+
+  AvailableCpuFeatures GetCpuFeatures() const { return cpu_features_; }
+
+  absl::optional<int> recommended_input_volume() const {
+    return recommended_input_volume_;
+  }
+
+ private:
+  static std::atomic<int> instance_count_;
+  const AvailableCpuFeatures cpu_features_;
+  ApmDataDumper data_dumper_;
+
+  GainApplier fixed_gain_applier_;
+  std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_;
+  std::unique_ptr<VoiceActivityDetectorWrapper> vad_;
+  std::unique_ptr<SpeechLevelEstimator> speech_level_estimator_;
+  std::unique_ptr<InputVolumeController> input_volume_controller_;
+  // TODO(bugs.webrtc.org/7494): Rename to `CrestFactorEstimator`.
+  std::unique_ptr<SaturationProtector> saturation_protector_;
+  std::unique_ptr<AdaptiveDigitalGainController> adaptive_digital_controller_;
+  Limiter limiter_;
+
+  int calls_since_last_limiter_log_;
+
+  // TODO(bugs.webrtc.org/7494): Remove intermediate storing at this level once
+  // APM refactoring is completed.
+  // Recommended input volume from `InputVolumecontroller`. Non-empty after
+  // `Process()` if input volume controller is enabled and
+  // `InputVolumeController::Process()` has returned a non-empty value.
+  absl::optional<int> recommended_input_volume_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/gain_controller2_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/gain_controller2_gn/moz.build
new file mode 100644
index 0000000000..fcbe873d27
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_controller2_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/gain_controller2.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("gain_controller2_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/gain_controller2_unittest.cc b/third_party/libwebrtc/modules/audio_processing/gain_controller2_unittest.cc
new file mode 100644
index 0000000000..5023bab617
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/gain_controller2_unittest.cc
@@ -0,0 +1,615 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/gain_controller2.h"
+
+#include <algorithm>
+#include <cmath>
+#include <memory>
+#include <numeric>
+#include <tuple>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+#include "modules/audio_processing/test/bitexactness_tools.h"
+#include "rtc_base/checks.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using ::testing::Eq;
+using ::testing::Optional;
+
+using Agc2Config = AudioProcessing::Config::GainController2;
+using InputVolumeControllerConfig = InputVolumeController::Config;
+
+// Sets all the samples in `ab` to `value`.
+void SetAudioBufferSamples(float value, AudioBuffer& ab) {
+  for (size_t k = 0; k < ab.num_channels(); ++k) {
+    std::fill(ab.channels()[k], ab.channels()[k] + ab.num_frames(), value);
+  }
+}
+
+float RunAgc2WithConstantInput(GainController2& agc2,
+                               float input_level,
+                               int num_frames,
+                               int sample_rate_hz,
+                               int num_channels = 1,
+                               int applied_initial_volume = 0) {
+  const int num_samples = rtc::CheckedDivExact(sample_rate_hz, 100);
+  AudioBuffer ab(sample_rate_hz, num_channels, sample_rate_hz, num_channels,
+                 sample_rate_hz, num_channels);
+
+  // Give time to the level estimator to converge.
+  for (int i = 0; i < num_frames + 1; ++i) {
+    SetAudioBufferSamples(input_level, ab);
+    const auto applied_volume = agc2.recommended_input_volume();
+    agc2.Analyze(applied_volume.value_or(applied_initial_volume), ab);
+
+    agc2.Process(/*speech_probability=*/absl::nullopt,
+                 /*input_volume_changed=*/false, &ab);
+  }
+
+  // Return the last sample from the last processed frame.
+  return ab.channels()[0][num_samples - 1];
+}
+
+std::unique_ptr<GainController2> CreateAgc2FixedDigitalMode(
+    float fixed_gain_db,
+    int sample_rate_hz) {
+  Agc2Config config;
+  config.adaptive_digital.enabled = false;
+  config.fixed_digital.gain_db = fixed_gain_db;
+  EXPECT_TRUE(GainController2::Validate(config));
+  return std::make_unique<GainController2>(
+      config, InputVolumeControllerConfig{}, sample_rate_hz,
+      /*num_channels=*/1,
+      /*use_internal_vad=*/true);
+}
+
+constexpr InputVolumeControllerConfig kTestInputVolumeControllerConfig{
+    .clipped_level_min = 20,
+    .clipped_level_step = 30,
+    .clipped_ratio_threshold = 0.4,
+    .clipped_wait_frames = 50,
+    .enable_clipping_predictor = true,
+    .target_range_max_dbfs = -6,
+    .target_range_min_dbfs = -70,
+    .update_input_volume_wait_frames = 100,
+    .speech_probability_threshold = 0.9,
+    .speech_ratio_threshold = 1,
+};
+
+}  // namespace
+
+TEST(GainController2, CheckDefaultConfig) {
+  Agc2Config config;
+  EXPECT_TRUE(GainController2::Validate(config));
+}
+
+TEST(GainController2, CheckFixedDigitalConfig) {
+  Agc2Config config;
+  // Attenuation is not allowed.
+  config.fixed_digital.gain_db = -5.0f;
+  EXPECT_FALSE(GainController2::Validate(config));
+  // No gain is allowed.
+  config.fixed_digital.gain_db = 0.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+  // Positive gain is allowed.
+  config.fixed_digital.gain_db = 15.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+}
+
+TEST(GainController2, CheckHeadroomDb) {
+  Agc2Config config;
+  config.adaptive_digital.headroom_db = -1.0f;
+  EXPECT_FALSE(GainController2::Validate(config));
+  config.adaptive_digital.headroom_db = 0.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+  config.adaptive_digital.headroom_db = 5.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+}
+
+TEST(GainController2, CheckMaxGainDb) {
+  Agc2Config config;
+  config.adaptive_digital.max_gain_db = -1.0f;
+  EXPECT_FALSE(GainController2::Validate(config));
+  config.adaptive_digital.max_gain_db = 0.0f;
+  EXPECT_FALSE(GainController2::Validate(config));
+  config.adaptive_digital.max_gain_db = 5.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+}
+
+TEST(GainController2, CheckInitialGainDb) {
+  Agc2Config config;
+  config.adaptive_digital.initial_gain_db = -1.0f;
+  EXPECT_FALSE(GainController2::Validate(config));
+  config.adaptive_digital.initial_gain_db = 0.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+  config.adaptive_digital.initial_gain_db = 5.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+}
+
+TEST(GainController2, CheckAdaptiveDigitalMaxGainChangeSpeedConfig) {
+  Agc2Config config;
+  config.adaptive_digital.max_gain_change_db_per_second = -1.0f;
+  EXPECT_FALSE(GainController2::Validate(config));
+  config.adaptive_digital.max_gain_change_db_per_second = 0.0f;
+  EXPECT_FALSE(GainController2::Validate(config));
+  config.adaptive_digital.max_gain_change_db_per_second = 5.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+}
+
+TEST(GainController2, CheckAdaptiveDigitalMaxOutputNoiseLevelConfig) {
+  Agc2Config config;
+  config.adaptive_digital.max_output_noise_level_dbfs = 5.0f;
+  EXPECT_FALSE(GainController2::Validate(config));
+  config.adaptive_digital.max_output_noise_level_dbfs = 0.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+  config.adaptive_digital.max_output_noise_level_dbfs = -5.0f;
+  EXPECT_TRUE(GainController2::Validate(config));
+}
+
+TEST(GainController2,
+     CheckGetRecommendedInputVolumeWhenInputVolumeControllerNotEnabled) {
+  constexpr float kHighInputLevel = 32767.0f;
+  constexpr float kLowInputLevel = 1000.0f;
+  constexpr int kInitialInputVolume = 100;
+  constexpr int kNumChannels = 2;
+  constexpr int kNumFrames = 5;
+  constexpr int kSampleRateHz = 16000;
+
+  Agc2Config config;
+  config.input_volume_controller.enabled = false;
+
+  auto gain_controller = std::make_unique<GainController2>(
+      config, InputVolumeControllerConfig{}, kSampleRateHz, kNumChannels,
+      /*use_internal_vad=*/true);
+
+  EXPECT_FALSE(gain_controller->recommended_input_volume().has_value());
+
+  // Run AGC for a signal with no clipping or detected speech.
+  RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames,
+                           kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+  EXPECT_FALSE(gain_controller->recommended_input_volume().has_value());
+
+  // Run AGC for a signal with clipping.
+  RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames,
+                           kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+  EXPECT_FALSE(gain_controller->recommended_input_volume().has_value());
+}
+
+TEST(
+    GainController2,
+    CheckGetRecommendedInputVolumeWhenInputVolumeControllerNotEnabledAndSpecificConfigUsed) {
+  constexpr float kHighInputLevel = 32767.0f;
+  constexpr float kLowInputLevel = 1000.0f;
+  constexpr int kInitialInputVolume = 100;
+  constexpr int kNumChannels = 2;
+  constexpr int kNumFrames = 5;
+  constexpr int kSampleRateHz = 16000;
+
+  Agc2Config config;
+  config.input_volume_controller.enabled = false;
+
+  auto gain_controller = std::make_unique<GainController2>(
+      config, kTestInputVolumeControllerConfig, kSampleRateHz, kNumChannels,
+      /*use_internal_vad=*/true);
+
+  EXPECT_FALSE(gain_controller->recommended_input_volume().has_value());
+
+  // Run AGC for a signal with no clipping or detected speech.
+  RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames,
+                           kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+  EXPECT_FALSE(gain_controller->recommended_input_volume().has_value());
+
+  // Run AGC for a signal with clipping.
+  RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames,
+                           kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+  EXPECT_FALSE(gain_controller->recommended_input_volume().has_value());
+}
+
+TEST(GainController2,
+     CheckGetRecommendedInputVolumeWhenInputVolumeControllerEnabled) {
+  constexpr float kHighInputLevel = 32767.0f;
+  constexpr float kLowInputLevel = 1000.0f;
+  constexpr int kInitialInputVolume = 100;
+  constexpr int kNumChannels = 2;
+  constexpr int kNumFrames = 5;
+  constexpr int kSampleRateHz = 16000;
+
+  Agc2Config config;
+  config.input_volume_controller.enabled = true;
+  config.adaptive_digital.enabled = true;
+
+  auto gain_controller = std::make_unique<GainController2>(
+      config, InputVolumeControllerConfig{}, kSampleRateHz, kNumChannels,
+      /*use_internal_vad=*/true);
+
+  EXPECT_FALSE(gain_controller->recommended_input_volume().has_value());
+
+  // Run AGC for a signal with no clipping or detected speech.
+  RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames,
+                           kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+  EXPECT_TRUE(gain_controller->recommended_input_volume().has_value());
+
+  // Run AGC for a signal with clipping.
+  RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames,
+                           kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+  EXPECT_TRUE(gain_controller->recommended_input_volume().has_value());
+}
+
+TEST(
+    GainController2,
+    CheckGetRecommendedInputVolumeWhenInputVolumeControllerEnabledAndSpecificConfigUsed) {
+  constexpr float kHighInputLevel = 32767.0f;
+  constexpr float kLowInputLevel = 1000.0f;
+  constexpr int kInitialInputVolume = 100;
+  constexpr int kNumChannels = 2;
+  constexpr int kNumFrames = 5;
+  constexpr int kSampleRateHz = 16000;
+
+  Agc2Config config;
+  config.input_volume_controller.enabled = true;
+  config.adaptive_digital.enabled = true;
+
+  auto gain_controller = std::make_unique<GainController2>(
+      config, kTestInputVolumeControllerConfig, kSampleRateHz, kNumChannels,
+      /*use_internal_vad=*/true);
+
+  EXPECT_FALSE(gain_controller->recommended_input_volume().has_value());
+
+  // Run AGC for a signal with no clipping or detected speech.
+  RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames,
+                           kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+  EXPECT_TRUE(gain_controller->recommended_input_volume().has_value());
+
+  // Run AGC for a signal with clipping.
+  RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames,
+                           kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+  EXPECT_TRUE(gain_controller->recommended_input_volume().has_value());
+}
+
+// Checks that the default config is applied.
+TEST(GainController2, ApplyDefaultConfig) {
+  auto gain_controller2 = std::make_unique<GainController2>(
+      Agc2Config{}, InputVolumeControllerConfig{},
+      /*sample_rate_hz=*/16000, /*num_channels=*/2,
+      /*use_internal_vad=*/true);
+  EXPECT_TRUE(gain_controller2.get());
+}
+
+TEST(GainController2FixedDigital, GainShouldChangeOnSetGain) {
+  constexpr float kInputLevel = 1000.0f;
+  constexpr size_t kNumFrames = 5;
+  constexpr size_t kSampleRateHz = 8000;
+  constexpr float kGain0Db = 0.0f;
+  constexpr float kGain20Db = 20.0f;
+
+  auto agc2_fixed = CreateAgc2FixedDigitalMode(kGain0Db, kSampleRateHz);
+
+  // Signal level is unchanged with 0 db gain.
+  EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(*agc2_fixed, kInputLevel, kNumFrames,
+                                           kSampleRateHz),
+                  kInputLevel);
+
+  // +20 db should increase signal by a factor of 10.
+  agc2_fixed->SetFixedGainDb(kGain20Db);
+  EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(*agc2_fixed, kInputLevel, kNumFrames,
+                                           kSampleRateHz),
+                  kInputLevel * 10);
+}
+
+TEST(GainController2FixedDigital, ChangeFixedGainShouldBeFastAndTimeInvariant) {
+  // Number of frames required for the fixed gain controller to adapt on the
+  // input signal when the gain changes.
+  constexpr size_t kNumFrames = 5;
+
+  constexpr float kInputLevel = 1000.0f;
+  constexpr size_t kSampleRateHz = 8000;
+  constexpr float kGainDbLow = 0.0f;
+  constexpr float kGainDbHigh = 25.0f;
+  static_assert(kGainDbLow < kGainDbHigh, "");
+
+  auto agc2_fixed = CreateAgc2FixedDigitalMode(kGainDbLow, kSampleRateHz);
+
+  // Start with a lower gain.
+  const float output_level_pre = RunAgc2WithConstantInput(
+      *agc2_fixed, kInputLevel, kNumFrames, kSampleRateHz);
+
+  // Increase gain.
+  agc2_fixed->SetFixedGainDb(kGainDbHigh);
+  static_cast<void>(RunAgc2WithConstantInput(*agc2_fixed, kInputLevel,
+                                             kNumFrames, kSampleRateHz));
+
+  // Back to the lower gain.
+  agc2_fixed->SetFixedGainDb(kGainDbLow);
+  const float output_level_post = RunAgc2WithConstantInput(
+      *agc2_fixed, kInputLevel, kNumFrames, kSampleRateHz);
+
+  EXPECT_EQ(output_level_pre, output_level_post);
+}
+
+class FixedDigitalTest
+    : public ::testing::TestWithParam<std::tuple<float, float, int, bool>> {
+ protected:
+  float gain_db_min() const { return std::get<0>(GetParam()); }
+  float gain_db_max() const { return std::get<1>(GetParam()); }
+  int sample_rate_hz() const { return std::get<2>(GetParam()); }
+  bool saturation_expected() const { return std::get<3>(GetParam()); }
+};
+
+TEST_P(FixedDigitalTest, CheckSaturationBehaviorWithLimiter) {
+  for (const float gain_db : test::LinSpace(gain_db_min(), gain_db_max(), 10)) {
+    SCOPED_TRACE(gain_db);
+    auto agc2_fixed = CreateAgc2FixedDigitalMode(gain_db, sample_rate_hz());
+    const float processed_sample =
+        RunAgc2WithConstantInput(*agc2_fixed, /*input_level=*/32767.0f,
+                                 /*num_frames=*/5, sample_rate_hz());
+    if (saturation_expected()) {
+      EXPECT_FLOAT_EQ(processed_sample, 32767.0f);
+    } else {
+      EXPECT_LT(processed_sample, 32767.0f);
+    }
+  }
+}
+
+static_assert(test::kLimiterMaxInputLevelDbFs < 10, "");
+INSTANTIATE_TEST_SUITE_P(
+    GainController2,
+    FixedDigitalTest,
+    ::testing::Values(
+        // When gain < `test::kLimiterMaxInputLevelDbFs`, the limiter will not
+        // saturate the signal (at any sample rate).
+        std::make_tuple(0.1f,
+                        test::kLimiterMaxInputLevelDbFs - 0.01f,
+                        8000,
+                        false),
+        std::make_tuple(0.1,
+                        test::kLimiterMaxInputLevelDbFs - 0.01f,
+                        48000,
+                        false),
+        // When gain > `test::kLimiterMaxInputLevelDbFs`, the limiter will
+        // saturate the signal (at any sample rate).
+        std::make_tuple(test::kLimiterMaxInputLevelDbFs + 0.01f,
+                        10.0f,
+                        8000,
+                        true),
+        std::make_tuple(test::kLimiterMaxInputLevelDbFs + 0.01f,
+                        10.0f,
+                        48000,
+                        true)));
+
+// Processes a test audio file and checks that the gain applied at the end of
+// the recording is close to the expected value.
+TEST(GainController2, CheckFinalGainWithAdaptiveDigitalController) {
+  constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz;
+  constexpr int kStereo = 2;
+
+  // Create AGC2 enabling only the adaptive digital controller.
+  Agc2Config config;
+  config.fixed_digital.gain_db = 0.0f;
+  config.adaptive_digital.enabled = true;
+  GainController2 agc2(config, /*input_volume_controller_config=*/{},
+                       kSampleRateHz, kStereo,
+                       /*use_internal_vad=*/true);
+
+  test::InputAudioFile input_file(
+      test::GetApmCaptureTestVectorFileName(kSampleRateHz),
+      /*loop_at_end=*/true);
+  const StreamConfig stream_config(kSampleRateHz, kStereo);
+
+  // Init buffers.
+  constexpr int kFrameDurationMs = 10;
+  std::vector<float> frame(kStereo * stream_config.num_frames());
+  AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo,
+                           kSampleRateHz, kStereo);
+
+  // Simulate.
+  constexpr float kGainDb = -6.0f;
+  const float gain = std::pow(10.0f, kGainDb / 20.0f);
+  constexpr int kDurationMs = 10000;
+  constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs;
+  for (int i = 0; i < kNumFramesToProcess; ++i) {
+    ReadFloatSamplesFromStereoFile(stream_config.num_frames(),
+                                   stream_config.num_channels(), &input_file,
+                                   frame);
+    // Apply a fixed gain to the input audio.
+    for (float& x : frame) {
+      x *= gain;
+    }
+    test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
+    agc2.Process(/*speech_probability=*/absl::nullopt,
+                 /*input_volume_changed=*/false, &audio_buffer);
+  }
+
+  // Estimate the applied gain by processing a probing frame.
+  SetAudioBufferSamples(/*value=*/1.0f, audio_buffer);
+  agc2.Process(/*speech_probability=*/absl::nullopt,
+               /*input_volume_changed=*/false, &audio_buffer);
+  const float applied_gain_db =
+      20.0f * std::log10(audio_buffer.channels_const()[0][0]);
+
+  constexpr float kExpectedGainDb = 5.6f;
+  constexpr float kToleranceDb = 0.3f;
+  EXPECT_NEAR(applied_gain_db, kExpectedGainDb, kToleranceDb);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Checks that `GainController2` crashes in debug mode if it runs its internal
+// VAD and the speech probability values are provided by the caller.
+TEST(GainController2DeathTest,
+     DebugCrashIfUseInternalVadAndSpeechProbabilityGiven) {
+  constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz;
+  constexpr int kStereo = 2;
+  AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo,
+                           kSampleRateHz, kStereo);
+  // Create AGC2 so that the interval VAD is also created.
+  GainController2 agc2(/*config=*/{.adaptive_digital = {.enabled = true}},
+                       /*input_volume_controller_config=*/{}, kSampleRateHz,
+                       kStereo,
+                       /*use_internal_vad=*/true);
+
+  EXPECT_DEATH(agc2.Process(/*speech_probability=*/0.123f,
+                            /*input_volume_changed=*/false, &audio_buffer),
+               "");
+}
+#endif
+
+// Processes a test audio file and checks that the injected speech probability
+// is not ignored when the internal VAD is not used.
+TEST(GainController2,
+     CheckInjectedVadProbabilityUsedWithAdaptiveDigitalController) {
+  constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz;
+  constexpr int kStereo = 2;
+
+  // Create AGC2 enabling only the adaptive digital controller.
+  Agc2Config config;
+  config.fixed_digital.gain_db = 0.0f;
+  config.adaptive_digital.enabled = true;
+  GainController2 agc2(config, /*input_volume_controller_config=*/{},
+                       kSampleRateHz, kStereo,
+                       /*use_internal_vad=*/false);
+  GainController2 agc2_reference(config, /*input_volume_controller_config=*/{},
+                                 kSampleRateHz, kStereo,
+                                 /*use_internal_vad=*/true);
+
+  test::InputAudioFile input_file(
+      test::GetApmCaptureTestVectorFileName(kSampleRateHz),
+      /*loop_at_end=*/true);
+  const StreamConfig stream_config(kSampleRateHz, kStereo);
+
+  // Init buffers.
+  constexpr int kFrameDurationMs = 10;
+  std::vector<float> frame(kStereo * stream_config.num_frames());
+  AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo,
+                           kSampleRateHz, kStereo);
+  AudioBuffer audio_buffer_reference(kSampleRateHz, kStereo, kSampleRateHz,
+                                     kStereo, kSampleRateHz, kStereo);
+  // Simulate.
+  constexpr float kGainDb = -6.0f;
+  const float gain = std::pow(10.0f, kGainDb / 20.0f);
+  constexpr int kDurationMs = 10000;
+  constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs;
+  constexpr float kSpeechProbabilities[] = {1.0f, 0.3f};
+  constexpr float kEpsilon = 0.0001f;
+  bool all_samples_zero = true;
+  bool all_samples_equal = true;
+  for (int i = 0, j = 0; i < kNumFramesToProcess; ++i, j = 1 - j) {
+    ReadFloatSamplesFromStereoFile(stream_config.num_frames(),
+                                   stream_config.num_channels(), &input_file,
+                                   frame);
+    // Apply a fixed gain to the input audio.
+    for (float& x : frame) {
+      x *= gain;
+    }
+    test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
+    agc2.Process(kSpeechProbabilities[j], /*input_volume_changed=*/false,
+                 &audio_buffer);
+    test::CopyVectorToAudioBuffer(stream_config, frame,
+                                  &audio_buffer_reference);
+    agc2_reference.Process(/*speech_probability=*/absl::nullopt,
+                           /*input_volume_changed=*/false,
+                           &audio_buffer_reference);
+    // Check the output buffers.
+    for (int i = 0; i < kStereo; ++i) {
+      for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) {
+        all_samples_zero &=
+            fabs(audio_buffer.channels_const()[i][j]) < kEpsilon;
+        all_samples_equal &=
+            fabs(audio_buffer.channels_const()[i][j] -
+                 audio_buffer_reference.channels_const()[i][j]) < kEpsilon;
+      }
+    }
+  }
+  EXPECT_FALSE(all_samples_zero);
+  EXPECT_FALSE(all_samples_equal);
+}
+
+// Processes a test audio file and checks that the output is equal when
+// an injected speech probability from `VoiceActivityDetectorWrapper` and
+// the speech probability computed by the internal VAD are the same.
+TEST(GainController2,
+     CheckEqualResultFromInjectedVadProbabilityWithAdaptiveDigitalController) {
+  constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz;
+  constexpr int kStereo = 2;
+
+  // Create AGC2 enabling only the adaptive digital controller.
+  Agc2Config config;
+  config.fixed_digital.gain_db = 0.0f;
+  config.adaptive_digital.enabled = true;
+  GainController2 agc2(config, /*input_volume_controller_config=*/{},
+                       kSampleRateHz, kStereo,
+                       /*use_internal_vad=*/false);
+  GainController2 agc2_reference(config, /*input_volume_controller_config=*/{},
+                                 kSampleRateHz, kStereo,
+                                 /*use_internal_vad=*/true);
+  VoiceActivityDetectorWrapper vad(GetAvailableCpuFeatures(), kSampleRateHz);
+  test::InputAudioFile input_file(
+      test::GetApmCaptureTestVectorFileName(kSampleRateHz),
+      /*loop_at_end=*/true);
+  const StreamConfig stream_config(kSampleRateHz, kStereo);
+
+  // Init buffers.
+  constexpr int kFrameDurationMs = 10;
+  std::vector<float> frame(kStereo * stream_config.num_frames());
+  AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo,
+                           kSampleRateHz, kStereo);
+  AudioBuffer audio_buffer_reference(kSampleRateHz, kStereo, kSampleRateHz,
+                                     kStereo, kSampleRateHz, kStereo);
+
+  // Simulate.
+  constexpr float kGainDb = -6.0f;
+  const float gain = std::pow(10.0f, kGainDb / 20.0f);
+  constexpr int kDurationMs = 10000;
+  constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs;
+  for (int i = 0; i < kNumFramesToProcess; ++i) {
+    ReadFloatSamplesFromStereoFile(stream_config.num_frames(),
+                                   stream_config.num_channels(), &input_file,
+                                   frame);
+    // Apply a fixed gain to the input audio.
+    for (float& x : frame) {
+      x *= gain;
+    }
+    test::CopyVectorToAudioBuffer(stream_config, frame,
+                                  &audio_buffer_reference);
+    agc2_reference.Process(absl::nullopt, /*input_volume_changed=*/false,
+                           &audio_buffer_reference);
+    test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
+    float speech_probability = vad.Analyze(AudioFrameView<const float>(
+        audio_buffer.channels(), audio_buffer.num_channels(),
+        audio_buffer.num_frames()));
+    agc2.Process(speech_probability, /*input_volume_changed=*/false,
+                 &audio_buffer);
+    // Check the output buffer.
+    for (int i = 0; i < kStereo; ++i) {
+      for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) {
+        EXPECT_FLOAT_EQ(audio_buffer.channels_const()[i][j],
+                        audio_buffer_reference.channels_const()[i][j]);
+      }
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/high_pass_filter.cc b/third_party/libwebrtc/modules/audio_processing/high_pass_filter.cc
new file mode 100644
index 0000000000..3b4740f6a5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/high_pass_filter.cc
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/high_pass_filter.h"
+
+#include "api/array_view.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+// [B,A] = butter(2,100/8000,'high')
+constexpr CascadedBiQuadFilter::BiQuadCoefficients
+    kHighPassFilterCoefficients16kHz = {{0.97261f, -1.94523f, 0.97261f},
+                                        {-1.94448f, 0.94598f}};
+
+// [B,A] = butter(2,100/16000,'high')
+constexpr CascadedBiQuadFilter::BiQuadCoefficients
+    kHighPassFilterCoefficients32kHz = {{0.98621f, -1.97242f, 0.98621f},
+                                        {-1.97223f, 0.97261f}};
+
+// [B,A] = butter(2,100/24000,'high')
+constexpr CascadedBiQuadFilter::BiQuadCoefficients
+    kHighPassFilterCoefficients48kHz = {{0.99079f, -1.98157f, 0.99079f},
+                                        {-1.98149f, 0.98166f}};
+
+constexpr size_t kNumberOfHighPassBiQuads = 1;
+
+const CascadedBiQuadFilter::BiQuadCoefficients& ChooseCoefficients(
+    int sample_rate_hz) {
+  switch (sample_rate_hz) {
+    case 16000:
+      return kHighPassFilterCoefficients16kHz;
+    case 32000:
+      return kHighPassFilterCoefficients32kHz;
+    case 48000:
+      return kHighPassFilterCoefficients48kHz;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  RTC_DCHECK_NOTREACHED();
+  return kHighPassFilterCoefficients16kHz;
+}
+
+}  // namespace
+
+HighPassFilter::HighPassFilter(int sample_rate_hz, size_t num_channels)
+    : sample_rate_hz_(sample_rate_hz) {
+  filters_.resize(num_channels);
+  const auto& coefficients = ChooseCoefficients(sample_rate_hz_);
+  for (size_t k = 0; k < filters_.size(); ++k) {
+    filters_[k].reset(
+        new CascadedBiQuadFilter(coefficients, kNumberOfHighPassBiQuads));
+  }
+}
+
+HighPassFilter::~HighPassFilter() = default;
+
+void HighPassFilter::Process(AudioBuffer* audio, bool use_split_band_data) {
+  RTC_DCHECK(audio);
+  RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
+  if (use_split_band_data) {
+    for (size_t k = 0; k < audio->num_channels(); ++k) {
+      rtc::ArrayView<float> channel_data = rtc::ArrayView<float>(
+          audio->split_bands(k)[0], audio->num_frames_per_band());
+      filters_[k]->Process(channel_data);
+    }
+  } else {
+    for (size_t k = 0; k < audio->num_channels(); ++k) {
+      rtc::ArrayView<float> channel_data =
+          rtc::ArrayView<float>(&audio->channels()[k][0], audio->num_frames());
+      filters_[k]->Process(channel_data);
+    }
+  }
+}
+
+void HighPassFilter::Process(std::vector<std::vector<float>>* audio) {
+  RTC_DCHECK_EQ(filters_.size(), audio->size());
+  for (size_t k = 0; k < audio->size(); ++k) {
+    filters_[k]->Process((*audio)[k]);
+  }
+}
+
+void HighPassFilter::Reset() {
+  for (size_t k = 0; k < filters_.size(); ++k) {
+    filters_[k]->Reset();
+  }
+}
+
+void HighPassFilter::Reset(size_t num_channels) {
+  const size_t old_num_channels = filters_.size();
+  filters_.resize(num_channels);
+  if (filters_.size() < old_num_channels) {
+    Reset();
+  } else {
+    for (size_t k = 0; k < old_num_channels; ++k) {
+      filters_[k]->Reset();
+    }
+    const auto& coefficients = ChooseCoefficients(sample_rate_hz_);
+    for (size_t k = old_num_channels; k < filters_.size(); ++k) {
+      filters_[k].reset(
+          new CascadedBiQuadFilter(coefficients, kNumberOfHighPassBiQuads));
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/high_pass_filter.h b/third_party/libwebrtc/modules/audio_processing/high_pass_filter.h
new file mode 100644
index 0000000000..7e7c370cd1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/high_pass_filter.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/utility/cascaded_biquad_filter.h"
+
+namespace webrtc {
+
+class AudioBuffer;
+
+class HighPassFilter {
+ public:
+  HighPassFilter(int sample_rate_hz, size_t num_channels);
+  ~HighPassFilter();
+  HighPassFilter(const HighPassFilter&) = delete;
+  HighPassFilter& operator=(const HighPassFilter&) = delete;
+
+  void Process(AudioBuffer* audio, bool use_split_band_data);
+  void Process(std::vector<std::vector<float>>* audio);
+  void Reset();
+  void Reset(size_t num_channels);
+
+  int sample_rate_hz() const { return sample_rate_hz_; }
+  size_t num_channels() const { return filters_.size(); }
+
+ private:
+  const int sample_rate_hz_;
+  std::vector<std::unique_ptr<CascadedBiQuadFilter>> filters_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/high_pass_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/high_pass_filter_gn/moz.build
new file mode 100644
index 0000000000..4c8aac9c3d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/high_pass_filter_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/high_pass_filter.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("high_pass_filter_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/high_pass_filter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/high_pass_filter_unittest.cc
new file mode 100644
index 0000000000..9f3c8fe595
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/high_pass_filter_unittest.cc
@@ -0,0 +1,301 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/high_pass_filter.h"
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+#include "modules/audio_processing/test/bitexactness_tools.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+// Process one frame of data via the AudioBuffer interface and produce the
+// output.
+std::vector<float> ProcessOneFrameAsAudioBuffer(
+    const std::vector<float>& frame_input,
+    const StreamConfig& stream_config,
+    HighPassFilter* high_pass_filter) {
+  AudioBuffer audio_buffer(
+      stream_config.sample_rate_hz(), stream_config.num_channels(),
+      stream_config.sample_rate_hz(), stream_config.num_channels(),
+      stream_config.sample_rate_hz(), stream_config.num_channels());
+
+  test::CopyVectorToAudioBuffer(stream_config, frame_input, &audio_buffer);
+  high_pass_filter->Process(&audio_buffer, /*use_split_band_data=*/false);
+  std::vector<float> frame_output;
+  test::ExtractVectorFromAudioBuffer(stream_config, &audio_buffer,
+                                     &frame_output);
+  return frame_output;
+}
+
+// Process one frame of data via the vector interface and produce the output.
+std::vector<float> ProcessOneFrameAsVector(
+    const std::vector<float>& frame_input,
+    const StreamConfig& stream_config,
+    HighPassFilter* high_pass_filter) {
+  std::vector<std::vector<float>> process_vector(
+      stream_config.num_channels(),
+      std::vector<float>(stream_config.num_frames()));
+
+  for (size_t k = 0; k < stream_config.num_frames(); ++k) {
+    for (size_t channel = 0; channel < stream_config.num_channels();
+         ++channel) {
+      process_vector[channel][k] =
+          frame_input[k * stream_config.num_channels() + channel];
+    }
+  }
+
+  high_pass_filter->Process(&process_vector);
+
+  std::vector<float> output;
+  for (size_t k = 0; k < stream_config.num_frames(); ++k) {
+    for (size_t channel = 0; channel < stream_config.num_channels();
+         ++channel) {
+      output.push_back(process_vector[channel][k]);
+    }
+  }
+
+  return process_vector[0];
+}
+
+// Processes a specified amount of frames, verifies the results and reports
+// any errors.
+void RunBitexactnessTest(int num_channels,
+                         bool use_audio_buffer_interface,
+                         const std::vector<float>& input,
+                         const std::vector<float>& reference) {
+  const StreamConfig stream_config(16000, num_channels);
+  HighPassFilter high_pass_filter(16000, num_channels);
+
+  std::vector<float> output;
+  const size_t num_frames_to_process =
+      input.size() /
+      (stream_config.num_frames() * stream_config.num_channels());
+  for (size_t frame_no = 0; frame_no < num_frames_to_process; ++frame_no) {
+    std::vector<float> frame_input(
+        input.begin() + stream_config.num_frames() *
+                            stream_config.num_channels() * frame_no,
+        input.begin() + stream_config.num_frames() *
+                            stream_config.num_channels() * (frame_no + 1));
+    if (use_audio_buffer_interface) {
+      output = ProcessOneFrameAsAudioBuffer(frame_input, stream_config,
+                                            &high_pass_filter);
+    } else {
+      output = ProcessOneFrameAsVector(frame_input, stream_config,
+                                       &high_pass_filter);
+    }
+  }
+
+  // Form vector to compare the reference to. Only the last frame processed
+  // is compared in order not having to specify all preceeding frames as
+  // inputs. As the algorithm being tested has a memory, testing only
+  // the last frame implicitly also tests the preceeding frames.
+  const size_t reference_frame_length =
+      reference.size() / stream_config.num_channels();
+  std::vector<float> output_to_verify;
+  for (size_t channel_no = 0; channel_no < stream_config.num_channels();
+       ++channel_no) {
+    output_to_verify.insert(
+        output_to_verify.end(),
+        output.begin() + channel_no * stream_config.num_frames(),
+        output.begin() + channel_no * stream_config.num_frames() +
+            reference_frame_length);
+  }
+
+  const float kElementErrorBound = 1.0f / 32768.0f;
+  EXPECT_TRUE(test::VerifyDeinterleavedArray(
+      reference_frame_length, num_channels, reference, output_to_verify,
+      kElementErrorBound));
+}
+
+// Method for forming a vector out of an array.
+// TODO(peah): Remove once braced initialization is allowed.
+std::vector<float> CreateVector(const rtc::ArrayView<const float>& array_view) {
+  std::vector<float> v;
+  for (auto value : array_view) {
+    v.push_back(value);
+  }
+  return v;
+}
+}  // namespace
+
+TEST(HighPassFilterAccuracyTest, ResetWithAudioBufferInterface) {
+  const StreamConfig stream_config_stereo(16000, 2);
+  const StreamConfig stream_config_mono(16000, 1);
+  std::vector<float> x_mono(160, 1.f);
+  std::vector<float> x_stereo(320, 1.f);
+  HighPassFilter hpf(16000, 1);
+  std::vector<float> y =
+      ProcessOneFrameAsAudioBuffer(x_mono, stream_config_mono, &hpf);
+  hpf.Reset(2);
+  y = ProcessOneFrameAsAudioBuffer(x_stereo, stream_config_stereo, &hpf);
+  hpf.Reset(1);
+  y = ProcessOneFrameAsAudioBuffer(x_mono, stream_config_mono, &hpf);
+  hpf.Reset();
+  y = ProcessOneFrameAsAudioBuffer(x_mono, stream_config_mono, &hpf);
+}
+
+TEST(HighPassFilterAccuracyTest, ResetWithVectorInterface) {
+  const StreamConfig stream_config_stereo(16000, 2);
+  const StreamConfig stream_config_mono(16000, 1);
+  std::vector<float> x_mono(160, 1.f);
+  std::vector<float> x_stereo(320, 1.f);
+  HighPassFilter hpf(16000, 1);
+  std::vector<float> y =
+      ProcessOneFrameAsVector(x_mono, stream_config_mono, &hpf);
+  hpf.Reset(2);
+  y = ProcessOneFrameAsVector(x_stereo, stream_config_stereo, &hpf);
+  hpf.Reset(1);
+  y = ProcessOneFrameAsVector(x_mono, stream_config_mono, &hpf);
+  hpf.Reset();
+  y = ProcessOneFrameAsVector(x_mono, stream_config_mono, &hpf);
+}
+
+TEST(HighPassFilterAccuracyTest, MonoInitial) {
+  const float kReferenceInput[] = {
+      0.150254f,  0.512488f,  -0.631245f, 0.240938f,  0.089080f,  -0.365440f,
+      -0.121169f, 0.095748f,  1.000000f,  0.773932f,  -0.377232f, 0.848124f,
+      0.202718f,  -0.017621f, 0.199738f,  -0.057279f, -0.034693f, 0.416303f,
+      0.393761f,  0.396041f,  0.187653f,  -0.337438f, 0.200436f,  0.455577f,
+      0.136624f,  0.289150f,  0.203131f,  -0.084798f, 0.082124f,  -0.220010f,
+      0.248266f,  -0.320554f, -0.298701f, -0.226218f, -0.822794f, 0.401962f,
+      0.090876f,  -0.210968f, 0.382936f,  -0.478291f, -0.028572f, -0.067474f,
+      0.089204f,  0.087430f,  -0.241695f, -0.008398f, -0.046076f, 0.175416f,
+      0.305518f,  0.309992f,  -0.241352f, 0.021618f,  -0.339291f, -0.311173f,
+      -0.001914f, 0.428301f,  -0.215087f, 0.103784f,  -0.063041f, 0.312250f,
+      -0.304344f, 0.009098f,  0.154406f,  0.307571f,  0.431537f,  0.024014f,
+      -0.416832f, -0.207440f, -0.296664f, 0.656846f,  -0.172033f, 0.209054f,
+      -0.053772f, 0.248326f,  -0.213741f, -0.391871f, -0.397490f, 0.136428f,
+      -0.049568f, -0.054788f, 0.396633f,  0.081485f,  0.055279f,  0.443690f,
+      -0.224812f, 0.194675f,  0.233369f,  -0.068107f, 0.060270f,  -0.325801f,
+      -0.320801f, 0.029308f,  0.201837f,  0.722528f,  -0.186366f, 0.052351f,
+      -0.023053f, -0.540192f, -0.122671f, -0.501532f, 0.234847f,  -0.248165f,
+      0.027971f,  -0.152171f, 0.084820f,  -0.167764f, 0.136923f,  0.206619f,
+      0.478395f,  -0.054249f, -0.597574f, -0.234627f, 0.378548f,  -0.299619f,
+      0.268543f,  0.034666f,  0.401492f,  -0.547983f, -0.055248f, -0.337538f,
+      0.812657f,  0.230611f,  0.385360f,  -0.295713f, -0.130957f, -0.076143f,
+      0.306960f,  -0.077653f, 0.196049f,  -0.573390f, -0.098885f, -0.230155f,
+      -0.440716f, 0.141956f,  0.078802f,  0.009356f,  -0.372703f, 0.315083f,
+      0.097859f,  -0.083575f, 0.006397f,  -0.073216f, -0.489105f, -0.079827f,
+      -0.232329f, -0.273644f, -0.323162f, -0.149105f, -0.559646f, 0.269458f,
+      0.145333f,  -0.005597f, -0.009717f, -0.223051f, 0.284676f,  -0.037228f,
+      -0.199679f, 0.377651f,  -0.062813f, -0.164607f};
+  const float kReference[] = {0.146139f, 0.490336f,  -0.649520f, 0.233881f,
+                              0.073214f, -0.373256f, -0.115394f, 0.102109f,
+                              0.976217f, 0.702270f,  -0.457697f, 0.757116f};
+
+  for (bool use_audio_buffer_interface : {true, false}) {
+    RunBitexactnessTest(
+        1, use_audio_buffer_interface,
+        CreateVector(rtc::ArrayView<const float>(kReferenceInput)),
+        CreateVector(rtc::ArrayView<const float>(kReference)));
+  }
+}
+
+TEST(HighPassFilterAccuracyTest, MonoConverged) {
+  const float kReferenceInput[] = {
+      0.150254f,  0.512488f,  -0.631245f, 0.240938f,  0.089080f,  -0.365440f,
+      -0.121169f, 0.095748f,  1.000000f,  0.773932f,  -0.377232f, 0.848124f,
+      0.202718f,  -0.017621f, 0.199738f,  -0.057279f, -0.034693f, 0.416303f,
+      0.393761f,  0.396041f,  0.187653f,  -0.337438f, 0.200436f,  0.455577f,
+      0.136624f,  0.289150f,  0.203131f,  -0.084798f, 0.082124f,  -0.220010f,
+      0.248266f,  -0.320554f, -0.298701f, -0.226218f, -0.822794f, 0.401962f,
+      0.090876f,  -0.210968f, 0.382936f,  -0.478291f, -0.028572f, -0.067474f,
+      0.089204f,  0.087430f,  -0.241695f, -0.008398f, -0.046076f, 0.175416f,
+      0.305518f,  0.309992f,  -0.241352f, 0.021618f,  -0.339291f, -0.311173f,
+      -0.001914f, 0.428301f,  -0.215087f, 0.103784f,  -0.063041f, 0.312250f,
+      -0.304344f, 0.009098f,  0.154406f,  0.307571f,  0.431537f,  0.024014f,
+      -0.416832f, -0.207440f, -0.296664f, 0.656846f,  -0.172033f, 0.209054f,
+      -0.053772f, 0.248326f,  -0.213741f, -0.391871f, -0.397490f, 0.136428f,
+      -0.049568f, -0.054788f, 0.396633f,  0.081485f,  0.055279f,  0.443690f,
+      -0.224812f, 0.194675f,  0.233369f,  -0.068107f, 0.060270f,  -0.325801f,
+      -0.320801f, 0.029308f,  0.201837f,  0.722528f,  -0.186366f, 0.052351f,
+      -0.023053f, -0.540192f, -0.122671f, -0.501532f, 0.234847f,  -0.248165f,
+      0.027971f,  -0.152171f, 0.084820f,  -0.167764f, 0.136923f,  0.206619f,
+      0.478395f,  -0.054249f, -0.597574f, -0.234627f, 0.378548f,  -0.299619f,
+      0.268543f,  0.034666f,  0.401492f,  -0.547983f, -0.055248f, -0.337538f,
+      0.812657f,  0.230611f,  0.385360f,  -0.295713f, -0.130957f, -0.076143f,
+      0.306960f,  -0.077653f, 0.196049f,  -0.573390f, -0.098885f, -0.230155f,
+      -0.440716f, 0.141956f,  0.078802f,  0.009356f,  -0.372703f, 0.315083f,
+      0.097859f,  -0.083575f, 0.006397f,  -0.073216f, -0.489105f, -0.079827f,
+      -0.232329f, -0.273644f, -0.323162f, -0.149105f, -0.559646f, 0.269458f,
+      0.145333f,  -0.005597f, -0.009717f, -0.223051f, 0.284676f,  -0.037228f,
+      -0.199679f, 0.377651f,  -0.062813f, -0.164607f, -0.082091f, -0.236957f,
+      -0.313025f, 0.705903f,  0.462637f,  0.085942f,  -0.351308f, -0.241859f,
+      -0.049333f, 0.221165f,  -0.372235f, -0.651092f, -0.404957f, 0.093201f,
+      0.109366f,  0.126224f,  -0.036409f, 0.051333f,  -0.133063f, 0.240896f,
+      -0.380532f, 0.127160f,  -0.237176f, -0.093586f, 0.154478f,  0.290379f,
+      -0.312329f, 0.352297f,  0.184480f,  -0.018965f, -0.054555f, -0.060811f,
+      -0.084705f, 0.006440f,  0.014333f,  0.230847f,  0.426721f,  0.130481f,
+      -0.058605f, 0.174712f,  0.051204f,  -0.287773f, 0.265265f,  0.085810f,
+      0.037775f,  0.143988f,  0.073051f,  -0.263103f, -0.045366f, -0.040816f,
+      -0.148673f, 0.470072f,  -0.244727f, -0.135204f, -0.198973f, -0.328139f,
+      -0.053722f, -0.076590f, 0.427586f,  -0.069591f, -0.297399f, 0.448094f,
+      0.345037f,  -0.064170f, -0.420903f, -0.124253f, -0.043578f, 0.077149f,
+      -0.072983f, 0.123916f,  0.109517f,  -0.349508f, -0.264912f, -0.207106f,
+      -0.141912f, -0.089586f, 0.003485f,  -0.846518f, -0.127715f, 0.347208f,
+      -0.298095f, 0.260935f,  0.097899f,  -0.008106f, 0.050987f,  -0.437362f,
+      -0.023625f, 0.448230f,  0.027484f,  0.011562f,  -0.205167f, -0.008611f,
+      0.064930f,  0.119156f,  -0.104183f, -0.066078f, 0.565530f,  -0.631108f,
+      0.623029f,  0.094334f,  0.279472f,  -0.465059f, -0.164888f, -0.077706f,
+      0.118130f,  -0.466746f, 0.131800f,  -0.338936f, 0.018497f,  0.182304f,
+      0.091398f,  0.302547f,  0.281153f,  -0.181899f, 0.071836f,  -0.263911f,
+      -0.369380f, 0.258447f,  0.000014f,  -0.015347f, 0.254619f,  0.166159f,
+      0.097865f,  0.349389f,  0.259834f,  0.067003f,  -0.192925f, -0.182080f,
+      0.333139f,  -0.450434f, -0.006836f, -0.544615f, 0.285183f,  0.240811f,
+      0.000325f,  -0.019796f, -0.694804f, 0.162411f,  -0.612686f, -0.648134f,
+      0.022338f,  -0.265058f, 0.114993f,  0.189185f,  0.239697f,  -0.193148f,
+      0.125581f,  0.028122f,  0.230849f,  0.149832f,  0.250919f,  -0.036871f,
+      -0.041136f, 0.281627f,  -0.593466f, -0.141009f, -0.355074f, -0.106915f,
+      0.181276f,  0.230753f,  -0.283631f, -0.131643f, 0.038292f,  -0.081563f,
+      0.084345f,  0.111763f,  -0.259882f, -0.049416f, -0.595824f, 0.320077f,
+      -0.175802f, -0.336422f, -0.070966f, -0.399242f, -0.005829f, -0.156680f,
+      0.608591f,  0.318150f,  -0.697767f, 0.123331f,  -0.390716f, -0.071276f,
+      0.045943f,  0.208958f,  -0.076304f, 0.440505f,  -0.134400f, 0.091525f,
+      0.185763f,  0.023806f,  0.246186f,  0.090323f,  -0.219133f, -0.504520f,
+      0.519393f,  -0.168939f, 0.028884f,  0.157380f,  0.031745f,  -0.252830f,
+      -0.130705f, -0.034901f, 0.413302f,  -0.240559f, 0.219279f,  0.086246f,
+      -0.065353f, -0.295376f, -0.079405f, -0.024226f, -0.410629f, 0.053706f,
+      -0.229794f, -0.026336f, 0.093956f,  -0.252810f, -0.080555f, 0.097827f,
+      -0.513040f, 0.289508f,  0.677527f,  0.268109f,  -0.088244f, 0.119781f,
+      -0.289511f, 0.524778f,  0.262884f,  0.220028f,  -0.244767f, 0.089411f,
+      -0.156018f, -0.087030f, -0.159292f, -0.286646f, -0.253953f, -0.058657f,
+      -0.474756f, 0.169797f,  -0.032919f, 0.195384f,  0.075355f,  0.138131f,
+      -0.414465f, -0.285118f, -0.124915f, 0.030645f,  0.315431f,  -0.081032f,
+      0.352546f,  0.132860f,  0.328112f,  0.035476f,  -0.183550f, -0.413984f,
+      0.043452f,  0.228748f,  -0.081765f, -0.151125f, -0.086251f, -0.306448f,
+      -0.137774f, -0.050508f, 0.012811f,  -0.017824f, 0.170841f,  0.030549f,
+      0.506935f,  0.087197f,  0.504274f,  -0.202080f, 0.147146f,  -0.072728f,
+      0.167713f,  0.165977f,  -0.610894f, -0.370849f, -0.402698f, 0.112297f,
+      0.410855f,  -0.091330f, 0.227008f,  0.152454f,  -0.293884f, 0.111074f,
+      -0.210121f, 0.423728f,  -0.009101f, 0.457188f,  -0.118785f, 0.164720f,
+      -0.017547f, -0.565046f, -0.274461f, 0.171169f,  -0.015338f, -0.312635f,
+      -0.175044f, 0.069729f,  -0.277504f, 0.272454f,  -0.179049f, 0.505495f,
+      -0.301774f, 0.055664f,  -0.425058f, -0.202222f, -0.165787f, 0.112155f,
+      0.263284f,  0.083972f,  -0.104256f, 0.227892f,  0.223253f,  0.033592f,
+      0.159638f,  0.115358f,  -0.275811f, 0.212265f,  -0.183658f, -0.168768f};
+
+  const float kReference[] = {-0.248836f, -0.086982f, 0.083715f,  -0.036787f,
+                              0.127212f,  0.147464f,  -0.221733f, -0.004484f,
+                              -0.535107f, 0.385999f,  -0.116346f, -0.265302f};
+
+  for (bool use_audio_buffer_interface : {true, false}) {
+    RunBitexactnessTest(
+        1, use_audio_buffer_interface,
+        CreateVector(rtc::ArrayView<const float>(kReferenceInput)),
+        CreateVector(rtc::ArrayView<const float>(kReference)));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/include/aec_dump.cc b/third_party/libwebrtc/modules/audio_processing/include/aec_dump.cc
new file mode 100644
index 0000000000..8f788cb802
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/aec_dump.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/include/aec_dump.h"
+
+namespace webrtc {
+InternalAPMConfig::InternalAPMConfig() = default;
+InternalAPMConfig::InternalAPMConfig(const InternalAPMConfig&) = default;
+InternalAPMConfig::InternalAPMConfig(InternalAPMConfig&&) = default;
+InternalAPMConfig& InternalAPMConfig::operator=(const InternalAPMConfig&) =
+    default;
+
+bool InternalAPMConfig::operator==(const InternalAPMConfig& other) const {
+  return aec_enabled == other.aec_enabled &&
+         aec_delay_agnostic_enabled == other.aec_delay_agnostic_enabled &&
+         aec_drift_compensation_enabled ==
+             other.aec_drift_compensation_enabled &&
+         aec_extended_filter_enabled == other.aec_extended_filter_enabled &&
+         aec_suppression_level == other.aec_suppression_level &&
+         aecm_enabled == other.aecm_enabled &&
+         aecm_comfort_noise_enabled == other.aecm_comfort_noise_enabled &&
+         aecm_routing_mode == other.aecm_routing_mode &&
+         agc_enabled == other.agc_enabled && agc_mode == other.agc_mode &&
+         agc_limiter_enabled == other.agc_limiter_enabled &&
+         hpf_enabled == other.hpf_enabled && ns_enabled == other.ns_enabled &&
+         ns_level == other.ns_level &&
+         transient_suppression_enabled == other.transient_suppression_enabled &&
+         noise_robust_agc_enabled == other.noise_robust_agc_enabled &&
+         pre_amplifier_enabled == other.pre_amplifier_enabled &&
+         pre_amplifier_fixed_gain_factor ==
+             other.pre_amplifier_fixed_gain_factor &&
+         experiments_description == other.experiments_description;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/include/aec_dump.h b/third_party/libwebrtc/modules/audio_processing/include/aec_dump.h
new file mode 100644
index 0000000000..6f2eb64f3a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/aec_dump.h
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+// Struct for passing current config from APM without having to
+// include protobuf headers.
+struct InternalAPMConfig {
+  InternalAPMConfig();
+  InternalAPMConfig(const InternalAPMConfig&);
+  InternalAPMConfig(InternalAPMConfig&&);
+
+  InternalAPMConfig& operator=(const InternalAPMConfig&);
+  InternalAPMConfig& operator=(InternalAPMConfig&&) = delete;
+
+  bool operator==(const InternalAPMConfig& other) const;
+
+  bool aec_enabled = false;
+  bool aec_delay_agnostic_enabled = false;
+  bool aec_drift_compensation_enabled = false;
+  bool aec_extended_filter_enabled = false;
+  int aec_suppression_level = 0;
+  bool aecm_enabled = false;
+  bool aecm_comfort_noise_enabled = false;
+  int aecm_routing_mode = 0;
+  bool agc_enabled = false;
+  int agc_mode = 0;
+  bool agc_limiter_enabled = false;
+  bool hpf_enabled = false;
+  bool ns_enabled = false;
+  int ns_level = 0;
+  bool transient_suppression_enabled = false;
+  bool noise_robust_agc_enabled = false;
+  bool pre_amplifier_enabled = false;
+  float pre_amplifier_fixed_gain_factor = 1.f;
+  std::string experiments_description = "";
+};
+
+// An interface for recording configuration and input/output streams
+// of the Audio Processing Module. The recordings are called
+// 'aec-dumps' and are stored in a protobuf format defined in
+// debug.proto.
+// The Write* methods are always safe to call concurrently or
+// otherwise for all implementing subclasses. The intended mode of
+// operation is to create a protobuf object from the input, and send
+// it away to be written to file asynchronously.
+class AecDump {
+ public:
+  struct AudioProcessingState {
+    int delay;
+    int drift;
+    absl::optional<int> applied_input_volume;
+    bool keypress;
+  };
+
+  virtual ~AecDump() = default;
+
+  // Logs Event::Type INIT message.
+  virtual void WriteInitMessage(const ProcessingConfig& api_format,
+                                int64_t time_now_ms) = 0;
+  ABSL_DEPRECATED("")
+  void WriteInitMessage(const ProcessingConfig& api_format) {
+    WriteInitMessage(api_format, 0);
+  }
+
+  // Logs Event::Type STREAM message. To log an input/output pair,
+  // call the AddCapture* and AddAudioProcessingState methods followed
+  // by a WriteCaptureStreamMessage call.
+  virtual void AddCaptureStreamInput(
+      const AudioFrameView<const float>& src) = 0;
+  virtual void AddCaptureStreamOutput(
+      const AudioFrameView<const float>& src) = 0;
+  virtual void AddCaptureStreamInput(const int16_t* const data,
+                                     int num_channels,
+                                     int samples_per_channel) = 0;
+  virtual void AddCaptureStreamOutput(const int16_t* const data,
+                                      int num_channels,
+                                      int samples_per_channel) = 0;
+  virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
+  virtual void WriteCaptureStreamMessage() = 0;
+
+  // Logs Event::Type REVERSE_STREAM message.
+  virtual void WriteRenderStreamMessage(const int16_t* const data,
+                                        int num_channels,
+                                        int samples_per_channel) = 0;
+  virtual void WriteRenderStreamMessage(
+      const AudioFrameView<const float>& src) = 0;
+
+  virtual void WriteRuntimeSetting(
+      const AudioProcessing::RuntimeSetting& runtime_setting) = 0;
+
+  // Logs Event::Type CONFIG message.
+  virtual void WriteConfig(const InternalAPMConfig& config) = 0;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.cc b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.cc
new file mode 100644
index 0000000000..7cc4fb75e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/include/audio_frame_proxies.h"
+
+#include "api/audio/audio_frame.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
+  if (!frame || !ap) {
+    return AudioProcessing::Error::kNullPointerError;
+  }
+
+  StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_);
+  StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_);
+  RTC_DCHECK_EQ(frame->samples_per_channel(), input_config.num_frames());
+
+  int result = ap->ProcessStream(frame->data(), input_config, output_config,
+                                 frame->mutable_data());
+
+  AudioProcessingStats stats = ap->GetStatistics();
+
+  if (stats.voice_detected) {
+    frame->vad_activity_ = *stats.voice_detected
+                               ? AudioFrame::VADActivity::kVadActive
+                               : AudioFrame::VADActivity::kVadPassive;
+  }
+
+  return result;
+}
+
+int ProcessReverseAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
+  if (!frame || !ap) {
+    return AudioProcessing::Error::kNullPointerError;
+  }
+
+  // Must be a native rate.
+  if (frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate8kHz &&
+      frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate16kHz &&
+      frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate32kHz &&
+      frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate48kHz) {
+    return AudioProcessing::Error::kBadSampleRateError;
+  }
+
+  if (frame->num_channels_ <= 0) {
+    return AudioProcessing::Error::kBadNumberChannelsError;
+  }
+
+  StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_);
+  StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_);
+
+  int result = ap->ProcessReverseStream(frame->data(), input_config,
+                                        output_config, frame->mutable_data());
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.h b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.h
new file mode 100644
index 0000000000..5dd111ca2b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_PROXIES_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_PROXIES_H_
+
+namespace webrtc {
+
+class AudioFrame;
+class AudioProcessing;
+
+// Processes a 10 ms `frame` of the primary audio stream using the provided
+// AudioProcessing object. On the client-side, this is the near-end (or
+// captured) audio. The `sample_rate_hz_`, `num_channels_`, and
+// `samples_per_channel_` members of `frame` must be valid. If changed from the
+// previous call to this function, it will trigger an initialization of the
+// provided AudioProcessing object.
+// The function returns any error codes passed from the AudioProcessing
+// ProcessStream method.
+int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame);
+
+// Processes a 10 ms `frame` of the reverse direction audio stream using the
+// provided AudioProcessing object. The frame may be modified. On the
+// client-side, this is the far-end (or to be rendered) audio. The
+// `sample_rate_hz_`, `num_channels_`, and `samples_per_channel_` members of
+// `frame` must be valid. If changed from the previous call to this function, it
+// will trigger an initialization of the provided AudioProcessing object.
+// The function returns any error codes passed from the AudioProcessing
+// ProcessReverseStream method.
+int ProcessReverseAudioFrame(AudioProcessing* ap, AudioFrame* frame);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_PROXIES_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_frame_view.h b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_view.h
new file mode 100644
index 0000000000..164784a7cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_view.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Class to pass audio data in T** format, where T is a numeric type.
+template <class T>
+class AudioFrameView {
+ public:
+  // `num_channels` and `channel_size` describe the T**
+  // `audio_samples`. `audio_samples` is assumed to point to a
+  // two-dimensional |num_channels * channel_size| array of floats.
+  AudioFrameView(T* const* audio_samples, int num_channels, int channel_size)
+      : audio_samples_(audio_samples),
+        num_channels_(num_channels),
+        channel_size_(channel_size) {
+    RTC_DCHECK_GE(num_channels_, 0);
+    RTC_DCHECK_GE(channel_size_, 0);
+  }
+
+  // Implicit cast to allow converting Frame<float> to
+  // Frame<const float>.
+  template <class U>
+  AudioFrameView(AudioFrameView<U> other)
+      : audio_samples_(other.data()),
+        num_channels_(other.num_channels()),
+        channel_size_(other.samples_per_channel()) {}
+
+  AudioFrameView() = delete;
+
+  int num_channels() const { return num_channels_; }
+
+  int samples_per_channel() const { return channel_size_; }
+
+  rtc::ArrayView<T> channel(int idx) {
+    RTC_DCHECK_LE(0, idx);
+    RTC_DCHECK_LE(idx, num_channels_);
+    return rtc::ArrayView<T>(audio_samples_[idx], channel_size_);
+  }
+
+  rtc::ArrayView<const T> channel(int idx) const {
+    RTC_DCHECK_LE(0, idx);
+    RTC_DCHECK_LE(idx, num_channels_);
+    return rtc::ArrayView<const T>(audio_samples_[idx], channel_size_);
+  }
+
+  T* const* data() { return audio_samples_; }
+
+ private:
+  T* const* audio_samples_;
+  int num_channels_;
+  int channel_size_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.cc b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.cc
new file mode 100644
index 0000000000..13ddcc588a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.cc
@@ -0,0 +1,210 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/include/audio_processing.h"
+
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace {
+
+using Agc1Config = AudioProcessing::Config::GainController1;
+using Agc2Config = AudioProcessing::Config::GainController2;
+
+std::string NoiseSuppressionLevelToString(
+    const AudioProcessing::Config::NoiseSuppression::Level& level) {
+  switch (level) {
+    case AudioProcessing::Config::NoiseSuppression::Level::kLow:
+      return "Low";
+    case AudioProcessing::Config::NoiseSuppression::Level::kModerate:
+      return "Moderate";
+    case AudioProcessing::Config::NoiseSuppression::Level::kHigh:
+      return "High";
+    case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
+      return "VeryHigh";
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
+  switch (mode) {
+    case Agc1Config::Mode::kAdaptiveAnalog:
+      return "AdaptiveAnalog";
+    case Agc1Config::Mode::kAdaptiveDigital:
+      return "AdaptiveDigital";
+    case Agc1Config::Mode::kFixedDigital:
+      return "FixedDigital";
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+}  // namespace
+
+constexpr int AudioProcessing::kNativeSampleRatesHz[];
+
+void CustomProcessing::SetRuntimeSetting(
+    AudioProcessing::RuntimeSetting setting) {}
+
+bool Agc1Config::operator==(const Agc1Config& rhs) const {
+  const auto& analog_lhs = analog_gain_controller;
+  const auto& analog_rhs = rhs.analog_gain_controller;
+  return enabled == rhs.enabled && mode == rhs.mode &&
+         target_level_dbfs == rhs.target_level_dbfs &&
+         compression_gain_db == rhs.compression_gain_db &&
+         enable_limiter == rhs.enable_limiter &&
+         analog_lhs.enabled == analog_rhs.enabled &&
+         analog_lhs.startup_min_volume == analog_rhs.startup_min_volume &&
+         analog_lhs.clipped_level_min == analog_rhs.clipped_level_min &&
+         analog_lhs.enable_digital_adaptive ==
+             analog_rhs.enable_digital_adaptive &&
+         analog_lhs.clipped_level_step == analog_rhs.clipped_level_step &&
+         analog_lhs.clipped_ratio_threshold ==
+             analog_rhs.clipped_ratio_threshold &&
+         analog_lhs.clipped_wait_frames == analog_rhs.clipped_wait_frames &&
+         analog_lhs.clipping_predictor.mode ==
+             analog_rhs.clipping_predictor.mode &&
+         analog_lhs.clipping_predictor.window_length ==
+             analog_rhs.clipping_predictor.window_length &&
+         analog_lhs.clipping_predictor.reference_window_length ==
+             analog_rhs.clipping_predictor.reference_window_length &&
+         analog_lhs.clipping_predictor.reference_window_delay ==
+             analog_rhs.clipping_predictor.reference_window_delay &&
+         analog_lhs.clipping_predictor.clipping_threshold ==
+             analog_rhs.clipping_predictor.clipping_threshold &&
+         analog_lhs.clipping_predictor.crest_factor_margin ==
+             analog_rhs.clipping_predictor.crest_factor_margin &&
+         analog_lhs.clipping_predictor.use_predicted_step ==
+             analog_rhs.clipping_predictor.use_predicted_step;
+}
+
+bool Agc2Config::AdaptiveDigital::operator==(
+    const Agc2Config::AdaptiveDigital& rhs) const {
+  return enabled == rhs.enabled && headroom_db == rhs.headroom_db &&
+         max_gain_db == rhs.max_gain_db &&
+         initial_gain_db == rhs.initial_gain_db &&
+         max_gain_change_db_per_second == rhs.max_gain_change_db_per_second &&
+         max_output_noise_level_dbfs == rhs.max_output_noise_level_dbfs;
+}
+
+bool Agc2Config::InputVolumeController::operator==(
+    const Agc2Config::InputVolumeController& rhs) const {
+  return enabled == rhs.enabled;
+}
+
+bool Agc2Config::operator==(const Agc2Config& rhs) const {
+  return enabled == rhs.enabled &&
+         fixed_digital.gain_db == rhs.fixed_digital.gain_db &&
+         adaptive_digital == rhs.adaptive_digital &&
+         input_volume_controller == rhs.input_volume_controller;
+}
+
+bool AudioProcessing::Config::CaptureLevelAdjustment::operator==(
+    const AudioProcessing::Config::CaptureLevelAdjustment& rhs) const {
+  return enabled == rhs.enabled && pre_gain_factor == rhs.pre_gain_factor &&
+         post_gain_factor == rhs.post_gain_factor &&
+         analog_mic_gain_emulation == rhs.analog_mic_gain_emulation;
+}
+
+bool AudioProcessing::Config::CaptureLevelAdjustment::AnalogMicGainEmulation::
+operator==(const AudioProcessing::Config::CaptureLevelAdjustment::
+               AnalogMicGainEmulation& rhs) const {
+  return enabled == rhs.enabled && initial_level == rhs.initial_level;
+}
+
+std::string AudioProcessing::Config::ToString() const {
+  char buf[2048];
+  rtc::SimpleStringBuilder builder(buf);
+  builder << "AudioProcessing::Config{ "
+             "pipeline: { "
+             "maximum_internal_processing_rate: "
+          << pipeline.maximum_internal_processing_rate
+          << ", multi_channel_render: " << pipeline.multi_channel_render
+          << ", multi_channel_capture: " << pipeline.multi_channel_capture
+          << " }, pre_amplifier: { enabled: " << pre_amplifier.enabled
+          << ", fixed_gain_factor: " << pre_amplifier.fixed_gain_factor
+          << " },capture_level_adjustment: { enabled: "
+          << capture_level_adjustment.enabled
+          << ", pre_gain_factor: " << capture_level_adjustment.pre_gain_factor
+          << ", post_gain_factor: " << capture_level_adjustment.post_gain_factor
+          << ", analog_mic_gain_emulation: { enabled: "
+          << capture_level_adjustment.analog_mic_gain_emulation.enabled
+          << ", initial_level: "
+          << capture_level_adjustment.analog_mic_gain_emulation.initial_level
+          << " }}, high_pass_filter: { enabled: " << high_pass_filter.enabled
+          << " }, echo_canceller: { enabled: " << echo_canceller.enabled
+          << ", mobile_mode: " << echo_canceller.mobile_mode
+          << ", enforce_high_pass_filtering: "
+          << echo_canceller.enforce_high_pass_filtering
+          << " }, noise_suppression: { enabled: " << noise_suppression.enabled
+          << ", level: "
+          << NoiseSuppressionLevelToString(noise_suppression.level)
+          << " }, transient_suppression: { enabled: "
+          << transient_suppression.enabled
+          << " }, gain_controller1: { enabled: " << gain_controller1.enabled
+          << ", mode: " << GainController1ModeToString(gain_controller1.mode)
+          << ", target_level_dbfs: " << gain_controller1.target_level_dbfs
+          << ", compression_gain_db: " << gain_controller1.compression_gain_db
+          << ", enable_limiter: " << gain_controller1.enable_limiter
+          << ", analog_gain_controller { enabled: "
+          << gain_controller1.analog_gain_controller.enabled
+          << ", startup_min_volume: "
+          << gain_controller1.analog_gain_controller.startup_min_volume
+          << ", clipped_level_min: "
+          << gain_controller1.analog_gain_controller.clipped_level_min
+          << ", enable_digital_adaptive: "
+          << gain_controller1.analog_gain_controller.enable_digital_adaptive
+          << ", clipped_level_step: "
+          << gain_controller1.analog_gain_controller.clipped_level_step
+          << ", clipped_ratio_threshold: "
+          << gain_controller1.analog_gain_controller.clipped_ratio_threshold
+          << ", clipped_wait_frames: "
+          << gain_controller1.analog_gain_controller.clipped_wait_frames
+          << ", clipping_predictor:  { enabled: "
+          << gain_controller1.analog_gain_controller.clipping_predictor.enabled
+          << ", mode: "
+          << gain_controller1.analog_gain_controller.clipping_predictor.mode
+          << ", window_length: "
+          << gain_controller1.analog_gain_controller.clipping_predictor
+                 .window_length
+          << ", reference_window_length: "
+          << gain_controller1.analog_gain_controller.clipping_predictor
+                 .reference_window_length
+          << ", reference_window_delay: "
+          << gain_controller1.analog_gain_controller.clipping_predictor
+                 .reference_window_delay
+          << ", clipping_threshold: "
+          << gain_controller1.analog_gain_controller.clipping_predictor
+                 .clipping_threshold
+          << ", crest_factor_margin: "
+          << gain_controller1.analog_gain_controller.clipping_predictor
+                 .crest_factor_margin
+          << ", use_predicted_step: "
+          << gain_controller1.analog_gain_controller.clipping_predictor
+                 .use_predicted_step
+          << " }}}, gain_controller2: { enabled: " << gain_controller2.enabled
+          << ", fixed_digital: { gain_db: "
+          << gain_controller2.fixed_digital.gain_db
+          << " }, adaptive_digital: { enabled: "
+          << gain_controller2.adaptive_digital.enabled
+          << ", headroom_db: " << gain_controller2.adaptive_digital.headroom_db
+          << ", max_gain_db: " << gain_controller2.adaptive_digital.max_gain_db
+          << ", initial_gain_db: "
+          << gain_controller2.adaptive_digital.initial_gain_db
+          << ", max_gain_change_db_per_second: "
+          << gain_controller2.adaptive_digital.max_gain_change_db_per_second
+          << ", max_output_noise_level_dbfs: "
+          << gain_controller2.adaptive_digital.max_output_noise_level_dbfs
+          << " }, input_volume_control : { enabled "
+          << gain_controller2.input_volume_controller.enabled << "}}";
+  return builder.str();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h
new file mode 100644
index 0000000000..f613a38de1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h
@@ -0,0 +1,941 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
+
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#ifndef _USE_MATH_DEFINES
+#define _USE_MATH_DEFINES
+#endif
+
+#include <math.h>
+#include <stddef.h>  // size_t
+#include <stdio.h>   // FILE
+#include <string.h>
+
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "api/audio/echo_control.h"
+#include "api/scoped_refptr.h"
+#include "modules/audio_processing/include/audio_processing_statistics.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace rtc {
+class TaskQueue;
+}  // namespace rtc
+
+namespace webrtc {
+
+class AecDump;
+class AudioBuffer;
+
+class StreamConfig;
+class ProcessingConfig;
+
+class EchoDetector;
+class CustomAudioAnalyzer;
+class CustomProcessing;
+
+// The Audio Processing Module (APM) provides a collection of voice processing
+// components designed for real-time communications software.
+//
+// APM operates on two audio streams on a frame-by-frame basis. Frames of the
+// primary stream, on which all processing is applied, are passed to
+// `ProcessStream()`. Frames of the reverse direction stream are passed to
+// `ProcessReverseStream()`. On the client-side, this will typically be the
+// near-end (capture) and far-end (render) streams, respectively. APM should be
+// placed in the signal chain as close to the audio hardware abstraction layer
+// (HAL) as possible.
+//
+// On the server-side, the reverse stream will normally not be used, with
+// processing occurring on each incoming stream.
+//
+// Component interfaces follow a similar pattern and are accessed through
+// corresponding getters in APM. All components are disabled at create-time,
+// with default settings that are recommended for most situations. New settings
+// can be applied without enabling a component. Enabling a component triggers
+// memory allocation and initialization to allow it to start processing the
+// streams.
+//
+// Thread safety is provided with the following assumptions to reduce locking
+// overhead:
+//   1. The stream getters and setters are called from the same thread as
+//      ProcessStream(). More precisely, stream functions are never called
+//      concurrently with ProcessStream().
+//   2. Parameter getters are never called concurrently with the corresponding
+//      setter.
+//
+// APM accepts only linear PCM audio data in chunks of ~10 ms (see
+// AudioProcessing::GetFrameSize() for details) and sample rates ranging from
+// 8000 Hz to 384000 Hz. The int16 interfaces use interleaved data, while the
+// float interfaces use deinterleaved data.
+//
+// Usage example, omitting error checking:
+// rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
+//
+// AudioProcessing::Config config;
+// config.echo_canceller.enabled = true;
+// config.echo_canceller.mobile_mode = false;
+//
+// config.gain_controller1.enabled = true;
+// config.gain_controller1.mode =
+// AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+// config.gain_controller1.analog_level_minimum = 0;
+// config.gain_controller1.analog_level_maximum = 255;
+//
+// config.gain_controller2.enabled = true;
+//
+// config.high_pass_filter.enabled = true;
+//
+// apm->ApplyConfig(config)
+//
+// // Start a voice call...
+//
+// // ... Render frame arrives bound for the audio HAL ...
+// apm->ProcessReverseStream(render_frame);
+//
+// // ... Capture frame arrives from the audio HAL ...
+// // Call required set_stream_ functions.
+// apm->set_stream_delay_ms(delay_ms);
+// apm->set_stream_analog_level(analog_level);
+//
+// apm->ProcessStream(capture_frame);
+//
+// // Call required stream_ functions.
+// analog_level = apm->recommended_stream_analog_level();
+// has_voice = apm->stream_has_voice();
+//
+// // Repeat render and capture processing for the duration of the call...
+// // Start a new call...
+// apm->Initialize();
+//
+// // Close the application...
+// apm.reset();
+//
+class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
+ public:
+  // The struct below constitutes the new parameter scheme for the audio
+  // processing. It is being introduced gradually and until it is fully
+  // introduced, it is prone to change.
+  // TODO(peah): Remove this comment once the new config scheme is fully rolled
+  // out.
+  //
+  // The parameters and behavior of the audio processing module are controlled
+  // by changing the default values in the AudioProcessing::Config struct.
+  // The config is applied by passing the struct to the ApplyConfig method.
+  //
+  // This config is intended to be used during setup, and to enable/disable
+  // top-level processing effects. Use during processing may cause undesired
+  // submodule resets, affecting the audio quality. Use the RuntimeSetting
+  // construct for runtime configuration.
+  struct RTC_EXPORT Config {
+    // Sets the properties of the audio processing pipeline.
+    struct RTC_EXPORT Pipeline {
+      // Ways to downmix a multi-channel track to mono.
+      enum class DownmixMethod {
+        kAverageChannels,  // Average across channels.
+        kUseFirstChannel   // Use the first channel.
+      };
+
+      // Maximum allowed processing rate used internally. May only be set to
+      // 32000 or 48000 and any differing values will be treated as 48000.
+      int maximum_internal_processing_rate = 48000;
+      // Allow multi-channel processing of render audio.
+      bool multi_channel_render = false;
+      // Allow multi-channel processing of capture audio when AEC3 is active
+      // or a custom AEC is injected..
+      bool multi_channel_capture = false;
+      // Indicates how to downmix multi-channel capture audio to mono (when
+      // needed).
+      DownmixMethod capture_downmix_method = DownmixMethod::kAverageChannels;
+    } pipeline;
+
+    // Enabled the pre-amplifier. It amplifies the capture signal
+    // before any other processing is done.
+    // TODO(webrtc:5298): Deprecate and use the pre-gain functionality in
+    // capture_level_adjustment instead.
+    struct PreAmplifier {
+      bool enabled = false;
+      float fixed_gain_factor = 1.0f;
+    } pre_amplifier;
+
+    // Functionality for general level adjustment in the capture pipeline. This
+    // should not be used together with the legacy PreAmplifier functionality.
+    struct CaptureLevelAdjustment {
+      bool operator==(const CaptureLevelAdjustment& rhs) const;
+      bool operator!=(const CaptureLevelAdjustment& rhs) const {
+        return !(*this == rhs);
+      }
+      bool enabled = false;
+      // The `pre_gain_factor` scales the signal before any processing is done.
+      float pre_gain_factor = 1.0f;
+      // The `post_gain_factor` scales the signal after all processing is done.
+      float post_gain_factor = 1.0f;
+      struct AnalogMicGainEmulation {
+        bool operator==(const AnalogMicGainEmulation& rhs) const;
+        bool operator!=(const AnalogMicGainEmulation& rhs) const {
+          return !(*this == rhs);
+        }
+        bool enabled = false;
+        // Initial analog gain level to use for the emulated analog gain. Must
+        // be in the range [0...255].
+        int initial_level = 255;
+      } analog_mic_gain_emulation;
+    } capture_level_adjustment;
+
+    struct HighPassFilter {
+      bool enabled = false;
+      bool apply_in_full_band = true;
+    } high_pass_filter;
+
+    struct EchoCanceller {
+      bool enabled = false;
+      bool mobile_mode = false;
+      bool export_linear_aec_output = false;
+      // Enforce the highpass filter to be on (has no effect for the mobile
+      // mode).
+      bool enforce_high_pass_filtering = true;
+    } echo_canceller;
+
+    // Enables background noise suppression.
+    struct NoiseSuppression {
+      bool enabled = false;
+      enum Level { kLow, kModerate, kHigh, kVeryHigh };
+      Level level = kModerate;
+      bool analyze_linear_aec_output_when_available = false;
+    } noise_suppression;
+
+    // Enables transient suppression.
+    struct TransientSuppression {
+      bool enabled = false;
+    } transient_suppression;
+
+    // Enables automatic gain control (AGC) functionality.
+    // The automatic gain control (AGC) component brings the signal to an
+    // appropriate range. This is done by applying a digital gain directly and,
+    // in the analog mode, prescribing an analog gain to be applied at the audio
+    // HAL.
+    // Recommended to be enabled on the client-side.
+    struct RTC_EXPORT GainController1 {
+      bool operator==(const GainController1& rhs) const;
+      bool operator!=(const GainController1& rhs) const {
+        return !(*this == rhs);
+      }
+
+      bool enabled = false;
+      enum Mode {
+        // Adaptive mode intended for use if an analog volume control is
+        // available on the capture device. It will require the user to provide
+        // coupling between the OS mixer controls and AGC through the
+        // stream_analog_level() functions.
+        // It consists of an analog gain prescription for the audio device and a
+        // digital compression stage.
+        kAdaptiveAnalog,
+        // Adaptive mode intended for situations in which an analog volume
+        // control is unavailable. It operates in a similar fashion to the
+        // adaptive analog mode, but with scaling instead applied in the digital
+        // domain. As with the analog mode, it additionally uses a digital
+        // compression stage.
+        kAdaptiveDigital,
+        // Fixed mode which enables only the digital compression stage also used
+        // by the two adaptive modes.
+        // It is distinguished from the adaptive modes by considering only a
+        // short time-window of the input signal. It applies a fixed gain
+        // through most of the input level range, and compresses (gradually
+        // reduces gain with increasing level) the input signal at higher
+        // levels. This mode is preferred on embedded devices where the capture
+        // signal level is predictable, so that a known gain can be applied.
+        kFixedDigital
+      };
+      Mode mode = kAdaptiveAnalog;
+      // Sets the target peak level (or envelope) of the AGC in dBFs (decibels
+      // from digital full-scale). The convention is to use positive values. For
+      // instance, passing in a value of 3 corresponds to -3 dBFs, or a target
+      // level 3 dB below full-scale. Limited to [0, 31].
+      int target_level_dbfs = 3;
+      // Sets the maximum gain the digital compression stage may apply, in dB. A
+      // higher number corresponds to greater compression, while a value of 0
+      // will leave the signal uncompressed. Limited to [0, 90].
+      // For updates after APM setup, use a RuntimeSetting instead.
+      int compression_gain_db = 9;
+      // When enabled, the compression stage will hard limit the signal to the
+      // target level. Otherwise, the signal will be compressed but not limited
+      // above the target level.
+      bool enable_limiter = true;
+
+      // Enables the analog gain controller functionality.
+      struct AnalogGainController {
+        bool enabled = true;
+        // TODO(bugs.webrtc.org/7494): Deprecated. Stop using and remove.
+        int startup_min_volume = 0;
+        // Lowest analog microphone level that will be applied in response to
+        // clipping.
+        int clipped_level_min = 70;
+        // If true, an adaptive digital gain is applied.
+        bool enable_digital_adaptive = true;
+        // Amount the microphone level is lowered with every clipping event.
+        // Limited to (0, 255].
+        int clipped_level_step = 15;
+        // Proportion of clipped samples required to declare a clipping event.
+        // Limited to (0.f, 1.f).
+        float clipped_ratio_threshold = 0.1f;
+        // Time in frames to wait after a clipping event before checking again.
+        // Limited to values higher than 0.
+        int clipped_wait_frames = 300;
+
+        // Enables clipping prediction functionality.
+        struct ClippingPredictor {
+          bool enabled = false;
+          enum Mode {
+            // Clipping event prediction mode with fixed step estimation.
+            kClippingEventPrediction,
+            // Clipped peak estimation mode with adaptive step estimation.
+            kAdaptiveStepClippingPeakPrediction,
+            // Clipped peak estimation mode with fixed step estimation.
+            kFixedStepClippingPeakPrediction,
+          };
+          Mode mode = kClippingEventPrediction;
+          // Number of frames in the sliding analysis window.
+          int window_length = 5;
+          // Number of frames in the sliding reference window.
+          int reference_window_length = 5;
+          // Reference window delay (unit: number of frames).
+          int reference_window_delay = 5;
+          // Clipping prediction threshold (dBFS).
+          float clipping_threshold = -1.0f;
+          // Crest factor drop threshold (dB).
+          float crest_factor_margin = 3.0f;
+          // If true, the recommended clipped level step is used to modify the
+          // analog gain. Otherwise, the predictor runs without affecting the
+          // analog gain.
+          bool use_predicted_step = true;
+        } clipping_predictor;
+      } analog_gain_controller;
+    } gain_controller1;
+
+    // Parameters for AGC2, an Automatic Gain Control (AGC) sub-module which
+    // replaces the AGC sub-module parametrized by `gain_controller1`.
+    // AGC2 brings the captured audio signal to the desired level by combining
+    // three different controllers (namely, input volume controller, adapative
+    // digital controller and fixed digital controller) and a limiter.
+    // TODO(bugs.webrtc.org:7494): Name `GainController` when AGC1 removed.
+    struct RTC_EXPORT GainController2 {
+      bool operator==(const GainController2& rhs) const;
+      bool operator!=(const GainController2& rhs) const {
+        return !(*this == rhs);
+      }
+
+      // AGC2 must be created if and only if `enabled` is true.
+      bool enabled = false;
+
+      // Parameters for the input volume controller, which adjusts the input
+      // volume applied when the audio is captured (e.g., microphone volume on
+      // a soundcard, input volume on HAL).
+      struct InputVolumeController {
+        bool operator==(const InputVolumeController& rhs) const;
+        bool operator!=(const InputVolumeController& rhs) const {
+          return !(*this == rhs);
+        }
+        bool enabled = false;
+      } input_volume_controller;
+
+      // Parameters for the adaptive digital controller, which adjusts and
+      // applies a digital gain after echo cancellation and after noise
+      // suppression.
+      struct RTC_EXPORT AdaptiveDigital {
+        bool operator==(const AdaptiveDigital& rhs) const;
+        bool operator!=(const AdaptiveDigital& rhs) const {
+          return !(*this == rhs);
+        }
+        bool enabled = false;
+        float headroom_db = 6.0f;
+        float max_gain_db = 30.0f;
+        float initial_gain_db = 8.0f;
+        float max_gain_change_db_per_second = 3.0f;
+        float max_output_noise_level_dbfs = -50.0f;
+      } adaptive_digital;
+
+      // Parameters for the fixed digital controller, which applies a fixed
+      // digital gain after the adaptive digital controller and before the
+      // limiter.
+      struct FixedDigital {
+        // By setting `gain_db` to a value greater than zero, the limiter can be
+        // turned into a compressor that first applies a fixed gain.
+        float gain_db = 0.0f;
+      } fixed_digital;
+    } gain_controller2;
+
+    std::string ToString() const;
+  };
+
+  // Specifies the properties of a setting to be passed to AudioProcessing at
+  // runtime.
+  class RuntimeSetting {
+   public:
+    enum class Type {
+      kNotSpecified,
+      kCapturePreGain,
+      kCaptureCompressionGain,
+      kCaptureFixedPostGain,
+      kPlayoutVolumeChange,
+      kCustomRenderProcessingRuntimeSetting,
+      kPlayoutAudioDeviceChange,
+      kCapturePostGain,
+      kCaptureOutputUsed
+    };
+
+    // Play-out audio device properties.
+    struct PlayoutAudioDeviceInfo {
+      int id;          // Identifies the audio device.
+      int max_volume;  // Maximum play-out volume.
+    };
+
+    RuntimeSetting() : type_(Type::kNotSpecified), value_(0.0f) {}
+    ~RuntimeSetting() = default;
+
+    static RuntimeSetting CreateCapturePreGain(float gain) {
+      return {Type::kCapturePreGain, gain};
+    }
+
+    static RuntimeSetting CreateCapturePostGain(float gain) {
+      return {Type::kCapturePostGain, gain};
+    }
+
+    // Corresponds to Config::GainController1::compression_gain_db, but for
+    // runtime configuration.
+    static RuntimeSetting CreateCompressionGainDb(int gain_db) {
+      RTC_DCHECK_GE(gain_db, 0);
+      RTC_DCHECK_LE(gain_db, 90);
+      return {Type::kCaptureCompressionGain, static_cast<float>(gain_db)};
+    }
+
+    // Corresponds to Config::GainController2::fixed_digital::gain_db, but for
+    // runtime configuration.
+    static RuntimeSetting CreateCaptureFixedPostGain(float gain_db) {
+      RTC_DCHECK_GE(gain_db, 0.0f);
+      RTC_DCHECK_LE(gain_db, 90.0f);
+      return {Type::kCaptureFixedPostGain, gain_db};
+    }
+
+    // Creates a runtime setting to notify play-out (aka render) audio device
+    // changes.
+    static RuntimeSetting CreatePlayoutAudioDeviceChange(
+        PlayoutAudioDeviceInfo audio_device) {
+      return {Type::kPlayoutAudioDeviceChange, audio_device};
+    }
+
+    // Creates a runtime setting to notify play-out (aka render) volume changes.
+    // `volume` is the unnormalized volume, the maximum of which
+    static RuntimeSetting CreatePlayoutVolumeChange(int volume) {
+      return {Type::kPlayoutVolumeChange, volume};
+    }
+
+    static RuntimeSetting CreateCustomRenderSetting(float payload) {
+      return {Type::kCustomRenderProcessingRuntimeSetting, payload};
+    }
+
+    static RuntimeSetting CreateCaptureOutputUsedSetting(
+        bool capture_output_used) {
+      return {Type::kCaptureOutputUsed, capture_output_used};
+    }
+
+    Type type() const { return type_; }
+    // Getters do not return a value but instead modify the argument to protect
+    // from implicit casting.
+    void GetFloat(float* value) const {
+      RTC_DCHECK(value);
+      *value = value_.float_value;
+    }
+    void GetInt(int* value) const {
+      RTC_DCHECK(value);
+      *value = value_.int_value;
+    }
+    void GetBool(bool* value) const {
+      RTC_DCHECK(value);
+      *value = value_.bool_value;
+    }
+    void GetPlayoutAudioDeviceInfo(PlayoutAudioDeviceInfo* value) const {
+      RTC_DCHECK(value);
+      *value = value_.playout_audio_device_info;
+    }
+
+   private:
+    RuntimeSetting(Type id, float value) : type_(id), value_(value) {}
+    RuntimeSetting(Type id, int value) : type_(id), value_(value) {}
+    RuntimeSetting(Type id, PlayoutAudioDeviceInfo value)
+        : type_(id), value_(value) {}
+    Type type_;
+    union U {
+      U() {}
+      U(int value) : int_value(value) {}
+      U(float value) : float_value(value) {}
+      U(PlayoutAudioDeviceInfo value) : playout_audio_device_info(value) {}
+      float float_value;
+      int int_value;
+      bool bool_value;
+      PlayoutAudioDeviceInfo playout_audio_device_info;
+    } value_;
+  };
+
+  ~AudioProcessing() override {}
+
+  // Initializes internal states, while retaining all user settings. This
+  // should be called before beginning to process a new audio stream. However,
+  // it is not necessary to call before processing the first stream after
+  // creation.
+  //
+  // It is also not necessary to call if the audio parameters (sample
+  // rate and number of channels) have changed. Passing updated parameters
+  // directly to `ProcessStream()` and `ProcessReverseStream()` is permissible.
+  // If the parameters are known at init-time though, they may be provided.
+  // TODO(webrtc:5298): Change to return void.
+  virtual int Initialize() = 0;
+
+  // The int16 interfaces require:
+  //   - only `NativeRate`s be used
+  //   - that the input, output and reverse rates must match
+  //   - that `processing_config.output_stream()` matches
+  //     `processing_config.input_stream()`.
+  //
+  // The float interfaces accept arbitrary rates and support differing input and
+  // output layouts, but the output must have either one channel or the same
+  // number of channels as the input.
+  virtual int Initialize(const ProcessingConfig& processing_config) = 0;
+
+  // TODO(peah): This method is a temporary solution used to take control
+  // over the parameters in the audio processing module and is likely to change.
+  virtual void ApplyConfig(const Config& config) = 0;
+
+  // TODO(ajm): Only intended for internal use. Make private and friend the
+  // necessary classes?
+  virtual int proc_sample_rate_hz() const = 0;
+  virtual int proc_split_sample_rate_hz() const = 0;
+  virtual size_t num_input_channels() const = 0;
+  virtual size_t num_proc_channels() const = 0;
+  virtual size_t num_output_channels() const = 0;
+  virtual size_t num_reverse_channels() const = 0;
+
+  // Set to true when the output of AudioProcessing will be muted or in some
+  // other way not used. Ideally, the captured audio would still be processed,
+  // but some components may change behavior based on this information.
+  // Default false. This method takes a lock. To achieve this in a lock-less
+  // manner the PostRuntimeSetting can instead be used.
+  virtual void set_output_will_be_muted(bool muted) = 0;
+
+  // Enqueues a runtime setting.
+  virtual void SetRuntimeSetting(RuntimeSetting setting) = 0;
+
+  // Enqueues a runtime setting. Returns a bool indicating whether the
+  // enqueueing was successfull.
+  virtual bool PostRuntimeSetting(RuntimeSetting setting) = 0;
+
+  // Accepts and produces a ~10 ms frame of interleaved 16 bit integer audio as
+  // specified in `input_config` and `output_config`. `src` and `dest` may use
+  // the same memory, if desired.
+  virtual int ProcessStream(const int16_t* const src,
+                            const StreamConfig& input_config,
+                            const StreamConfig& output_config,
+                            int16_t* const dest) = 0;
+
+  // Accepts deinterleaved float audio with the range [-1, 1]. Each element of
+  // `src` points to a channel buffer, arranged according to `input_stream`. At
+  // output, the channels will be arranged according to `output_stream` in
+  // `dest`.
+  //
+  // The output must have one channel or as many channels as the input. `src`
+  // and `dest` may use the same memory, if desired.
+  virtual int ProcessStream(const float* const* src,
+                            const StreamConfig& input_config,
+                            const StreamConfig& output_config,
+                            float* const* dest) = 0;
+
+  // Accepts and produces a ~10 ms frame of interleaved 16 bit integer audio for
+  // the reverse direction audio stream as specified in `input_config` and
+  // `output_config`. `src` and `dest` may use the same memory, if desired.
+  virtual int ProcessReverseStream(const int16_t* const src,
+                                   const StreamConfig& input_config,
+                                   const StreamConfig& output_config,
+                                   int16_t* const dest) = 0;
+
+  // Accepts deinterleaved float audio with the range [-1, 1]. Each element of
+  // `data` points to a channel buffer, arranged according to `reverse_config`.
+  virtual int ProcessReverseStream(const float* const* src,
+                                   const StreamConfig& input_config,
+                                   const StreamConfig& output_config,
+                                   float* const* dest) = 0;
+
+  // Accepts deinterleaved float audio with the range [-1, 1]. Each element
+  // of `data` points to a channel buffer, arranged according to
+  // `reverse_config`.
+  virtual int AnalyzeReverseStream(const float* const* data,
+                                   const StreamConfig& reverse_config) = 0;
+
+  // Returns the most recently produced ~10 ms of the linear AEC output at a
+  // rate of 16 kHz. If there is more than one capture channel, a mono
+  // representation of the input is returned. Returns true/false to indicate
+  // whether an output returned.
+  virtual bool GetLinearAecOutput(
+      rtc::ArrayView<std::array<float, 160>> linear_output) const = 0;
+
+  // This must be called prior to ProcessStream() if and only if adaptive analog
+  // gain control is enabled, to pass the current analog level from the audio
+  // HAL. Must be within the range [0, 255].
+  virtual void set_stream_analog_level(int level) = 0;
+
+  // When an analog mode is set, this should be called after
+  // `set_stream_analog_level()` and `ProcessStream()` to obtain the recommended
+  // new analog level for the audio HAL. It is the user's responsibility to
+  // apply this level.
+  virtual int recommended_stream_analog_level() const = 0;
+
+  // This must be called if and only if echo processing is enabled.
+  //
+  // Sets the `delay` in ms between ProcessReverseStream() receiving a far-end
+  // frame and ProcessStream() receiving a near-end frame containing the
+  // corresponding echo. On the client-side this can be expressed as
+  //   delay = (t_render - t_analyze) + (t_process - t_capture)
+  // where,
+  //   - t_analyze is the time a frame is passed to ProcessReverseStream() and
+  //     t_render is the time the first sample of the same frame is rendered by
+  //     the audio hardware.
+  //   - t_capture is the time the first sample of a frame is captured by the
+  //     audio hardware and t_process is the time the same frame is passed to
+  //     ProcessStream().
+  virtual int set_stream_delay_ms(int delay) = 0;
+  virtual int stream_delay_ms() const = 0;
+
+  // Call to signal that a key press occurred (true) or did not occur (false)
+  // with this chunk of audio.
+  virtual void set_stream_key_pressed(bool key_pressed) = 0;
+
+  // Creates and attaches an webrtc::AecDump for recording debugging
+  // information.
+  // The `worker_queue` may not be null and must outlive the created
+  // AecDump instance. |max_log_size_bytes == -1| means the log size
+  // will be unlimited. `handle` may not be null. The AecDump takes
+  // responsibility for `handle` and closes it in the destructor. A
+  // return value of true indicates that the file has been
+  // sucessfully opened, while a value of false indicates that
+  // opening the file failed.
+  virtual bool CreateAndAttachAecDump(absl::string_view file_name,
+                                      int64_t max_log_size_bytes,
+                                      rtc::TaskQueue* worker_queue) = 0;
+  virtual bool CreateAndAttachAecDump(FILE* handle,
+                                      int64_t max_log_size_bytes,
+                                      rtc::TaskQueue* worker_queue) = 0;
+
+  // TODO(webrtc:5298) Deprecated variant.
+  // Attaches provided webrtc::AecDump for recording debugging
+  // information. Log file and maximum file size logic is supposed to
+  // be handled by implementing instance of AecDump. Calling this
+  // method when another AecDump is attached resets the active AecDump
+  // with a new one. This causes the d-tor of the earlier AecDump to
+  // be called. The d-tor call may block until all pending logging
+  // tasks are completed.
+  virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) = 0;
+
+  // If no AecDump is attached, this has no effect. If an AecDump is
+  // attached, it's destructor is called. The d-tor may block until
+  // all pending logging tasks are completed.
+  virtual void DetachAecDump() = 0;
+
+  // Get audio processing statistics.
+  virtual AudioProcessingStats GetStatistics() = 0;
+  // TODO(webrtc:5298) Deprecated variant. The `has_remote_tracks` argument
+  // should be set if there are active remote tracks (this would usually be true
+  // during a call). If there are no remote tracks some of the stats will not be
+  // set by AudioProcessing, because they only make sense if there is at least
+  // one remote track.
+  virtual AudioProcessingStats GetStatistics(bool has_remote_tracks) = 0;
+
+  // Returns the last applied configuration.
+  virtual AudioProcessing::Config GetConfig() const = 0;
+
+  enum Error {
+    // Fatal errors.
+    kNoError = 0,
+    kUnspecifiedError = -1,
+    kCreationFailedError = -2,
+    kUnsupportedComponentError = -3,
+    kUnsupportedFunctionError = -4,
+    kNullPointerError = -5,
+    kBadParameterError = -6,
+    kBadSampleRateError = -7,
+    kBadDataLengthError = -8,
+    kBadNumberChannelsError = -9,
+    kFileError = -10,
+    kStreamParameterNotSetError = -11,
+    kNotEnabledError = -12,
+
+    // Warnings are non-fatal.
+    // This results when a set_stream_ parameter is out of range. Processing
+    // will continue, but the parameter may have been truncated.
+    kBadStreamParameterWarning = -13
+  };
+
+  // Native rates supported by the integer interfaces.
+  enum NativeRate {
+    kSampleRate8kHz = 8000,
+    kSampleRate16kHz = 16000,
+    kSampleRate32kHz = 32000,
+    kSampleRate48kHz = 48000
+  };
+
+  // TODO(kwiberg): We currently need to support a compiler (Visual C++) that
+  // complains if we don't explicitly state the size of the array here. Remove
+  // the size when that's no longer the case.
+  static constexpr int kNativeSampleRatesHz[4] = {
+      kSampleRate8kHz, kSampleRate16kHz, kSampleRate32kHz, kSampleRate48kHz};
+  static constexpr size_t kNumNativeSampleRates =
+      arraysize(kNativeSampleRatesHz);
+  static constexpr int kMaxNativeSampleRateHz =
+      kNativeSampleRatesHz[kNumNativeSampleRates - 1];
+
+  // APM processes audio in chunks of about 10 ms. See GetFrameSize() for
+  // details.
+  static constexpr int kChunkSizeMs = 10;
+
+  // Returns floor(sample_rate_hz/100): the number of samples per channel used
+  // as input and output to the audio processing module in calls to
+  // ProcessStream, ProcessReverseStream, AnalyzeReverseStream, and
+  // GetLinearAecOutput.
+  //
+  // This is exactly 10 ms for sample rates divisible by 100. For example:
+  //  - 48000 Hz (480 samples per channel),
+  //  - 44100 Hz (441 samples per channel),
+  //  - 16000 Hz (160 samples per channel).
+  //
+  // Sample rates not divisible by 100 are received/produced in frames of
+  // approximately 10 ms. For example:
+  //  - 22050 Hz (220 samples per channel, or ~9.98 ms per frame),
+  //  - 11025 Hz (110 samples per channel, or ~9.98 ms per frame).
+  // These nondivisible sample rates yield lower audio quality compared to
+  // multiples of 100. Internal resampling to 10 ms frames causes a simulated
+  // clock drift effect which impacts the performance of (for example) echo
+  // cancellation.
+  static int GetFrameSize(int sample_rate_hz) { return sample_rate_hz / 100; }
+};
+
+class RTC_EXPORT AudioProcessingBuilder {
+ public:
+  AudioProcessingBuilder();
+  AudioProcessingBuilder(const AudioProcessingBuilder&) = delete;
+  AudioProcessingBuilder& operator=(const AudioProcessingBuilder&) = delete;
+  ~AudioProcessingBuilder();
+
+  // Sets the APM configuration.
+  AudioProcessingBuilder& SetConfig(const AudioProcessing::Config& config) {
+    config_ = config;
+    return *this;
+  }
+
+  // Sets the echo controller factory to inject when APM is created.
+  AudioProcessingBuilder& SetEchoControlFactory(
+      std::unique_ptr<EchoControlFactory> echo_control_factory) {
+    echo_control_factory_ = std::move(echo_control_factory);
+    return *this;
+  }
+
+  // Sets the capture post-processing sub-module to inject when APM is created.
+  AudioProcessingBuilder& SetCapturePostProcessing(
+      std::unique_ptr<CustomProcessing> capture_post_processing) {
+    capture_post_processing_ = std::move(capture_post_processing);
+    return *this;
+  }
+
+  // Sets the render pre-processing sub-module to inject when APM is created.
+  AudioProcessingBuilder& SetRenderPreProcessing(
+      std::unique_ptr<CustomProcessing> render_pre_processing) {
+    render_pre_processing_ = std::move(render_pre_processing);
+    return *this;
+  }
+
+  // Sets the echo detector to inject when APM is created.
+  AudioProcessingBuilder& SetEchoDetector(
+      rtc::scoped_refptr<EchoDetector> echo_detector) {
+    echo_detector_ = std::move(echo_detector);
+    return *this;
+  }
+
+  // Sets the capture analyzer sub-module to inject when APM is created.
+  AudioProcessingBuilder& SetCaptureAnalyzer(
+      std::unique_ptr<CustomAudioAnalyzer> capture_analyzer) {
+    capture_analyzer_ = std::move(capture_analyzer);
+    return *this;
+  }
+
+  // Creates an APM instance with the specified config or the default one if
+  // unspecified. Injects the specified components transferring the ownership
+  // to the newly created APM instance - i.e., except for the config, the
+  // builder is reset to its initial state.
+  rtc::scoped_refptr<AudioProcessing> Create();
+
+ private:
+  AudioProcessing::Config config_;
+  std::unique_ptr<EchoControlFactory> echo_control_factory_;
+  std::unique_ptr<CustomProcessing> capture_post_processing_;
+  std::unique_ptr<CustomProcessing> render_pre_processing_;
+  rtc::scoped_refptr<EchoDetector> echo_detector_;
+  std::unique_ptr<CustomAudioAnalyzer> capture_analyzer_;
+};
+
+class StreamConfig {
+ public:
+  // sample_rate_hz: The sampling rate of the stream.
+  // num_channels: The number of audio channels in the stream.
+  StreamConfig(int sample_rate_hz = 0, size_t num_channels = 0)
+      : sample_rate_hz_(sample_rate_hz),
+        num_channels_(num_channels),
+        num_frames_(calculate_frames(sample_rate_hz)) {}
+
+  void set_sample_rate_hz(int value) {
+    sample_rate_hz_ = value;
+    num_frames_ = calculate_frames(value);
+  }
+  void set_num_channels(size_t value) { num_channels_ = value; }
+
+  int sample_rate_hz() const { return sample_rate_hz_; }
+
+  // The number of channels in the stream.
+  size_t num_channels() const { return num_channels_; }
+
+  size_t num_frames() const { return num_frames_; }
+  size_t num_samples() const { return num_channels_ * num_frames_; }
+
+  bool operator==(const StreamConfig& other) const {
+    return sample_rate_hz_ == other.sample_rate_hz_ &&
+           num_channels_ == other.num_channels_;
+  }
+
+  bool operator!=(const StreamConfig& other) const { return !(*this == other); }
+
+ private:
+  static size_t calculate_frames(int sample_rate_hz) {
+    return static_cast<size_t>(AudioProcessing::GetFrameSize(sample_rate_hz));
+  }
+
+  int sample_rate_hz_;
+  size_t num_channels_;
+  size_t num_frames_;
+};
+
+class ProcessingConfig {
+ public:
+  enum StreamName {
+    kInputStream,
+    kOutputStream,
+    kReverseInputStream,
+    kReverseOutputStream,
+    kNumStreamNames,
+  };
+
+  const StreamConfig& input_stream() const {
+    return streams[StreamName::kInputStream];
+  }
+  const StreamConfig& output_stream() const {
+    return streams[StreamName::kOutputStream];
+  }
+  const StreamConfig& reverse_input_stream() const {
+    return streams[StreamName::kReverseInputStream];
+  }
+  const StreamConfig& reverse_output_stream() const {
+    return streams[StreamName::kReverseOutputStream];
+  }
+
+  StreamConfig& input_stream() { return streams[StreamName::kInputStream]; }
+  StreamConfig& output_stream() { return streams[StreamName::kOutputStream]; }
+  StreamConfig& reverse_input_stream() {
+    return streams[StreamName::kReverseInputStream];
+  }
+  StreamConfig& reverse_output_stream() {
+    return streams[StreamName::kReverseOutputStream];
+  }
+
+  bool operator==(const ProcessingConfig& other) const {
+    for (int i = 0; i < StreamName::kNumStreamNames; ++i) {
+      if (this->streams[i] != other.streams[i]) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  bool operator!=(const ProcessingConfig& other) const {
+    return !(*this == other);
+  }
+
+  StreamConfig streams[StreamName::kNumStreamNames];
+};
+
+// Experimental interface for a custom analysis submodule.
+class CustomAudioAnalyzer {
+ public:
+  // (Re-) Initializes the submodule.
+  virtual void Initialize(int sample_rate_hz, int num_channels) = 0;
+  // Analyzes the given capture or render signal.
+  virtual void Analyze(const AudioBuffer* audio) = 0;
+  // Returns a string representation of the module state.
+  virtual std::string ToString() const = 0;
+
+  virtual ~CustomAudioAnalyzer() {}
+};
+
+// Interface for a custom processing submodule.
+class CustomProcessing {
+ public:
+  // (Re-)Initializes the submodule.
+  virtual void Initialize(int sample_rate_hz, int num_channels) = 0;
+  // Processes the given capture or render signal.
+  virtual void Process(AudioBuffer* audio) = 0;
+  // Returns a string representation of the module state.
+  virtual std::string ToString() const = 0;
+  // Handles RuntimeSettings. TODO(webrtc:9262): make pure virtual
+  // after updating dependencies.
+  virtual void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting);
+
+  virtual ~CustomProcessing() {}
+};
+
+// Interface for an echo detector submodule.
+class EchoDetector : public rtc::RefCountInterface {
+ public:
+  // (Re-)Initializes the submodule.
+  virtual void Initialize(int capture_sample_rate_hz,
+                          int num_capture_channels,
+                          int render_sample_rate_hz,
+                          int num_render_channels) = 0;
+
+  // Analysis (not changing) of the first channel of the render signal.
+  virtual void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) = 0;
+
+  // Analysis (not changing) of the capture signal.
+  virtual void AnalyzeCaptureAudio(
+      rtc::ArrayView<const float> capture_audio) = 0;
+
+  struct Metrics {
+    absl::optional<double> echo_likelihood;
+    absl::optional<double> echo_likelihood_recent_max;
+  };
+
+  // Collect current metrics from the echo detector.
+  virtual Metrics GetMetrics() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.cc b/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.cc
new file mode 100644
index 0000000000..7139ee502e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.cc
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/include/audio_processing_statistics.h"
+
+namespace webrtc {
+
+AudioProcessingStats::AudioProcessingStats() = default;
+
+AudioProcessingStats::AudioProcessingStats(const AudioProcessingStats& other) =
+    default;
+
+AudioProcessingStats::~AudioProcessingStats() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h b/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h
new file mode 100644
index 0000000000..3b43319951
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+// This version of the stats uses Optionals, it will replace the regular
+// AudioProcessingStatistics struct.
+struct RTC_EXPORT AudioProcessingStats {
+  AudioProcessingStats();
+  AudioProcessingStats(const AudioProcessingStats& other);
+  ~AudioProcessingStats();
+
+  // Deprecated.
+  // TODO(bugs.webrtc.org/11226): Remove.
+  // True if voice is detected in the last capture frame, after processing.
+  // It is conservative in flagging audio as speech, with low likelihood of
+  // incorrectly flagging a frame as voice.
+  // Only reported if voice detection is enabled in AudioProcessing::Config.
+  absl::optional<bool> voice_detected;
+
+  // AEC Statistics.
+  // ERL = 10log_10(P_far / P_echo)
+  absl::optional<double> echo_return_loss;
+  // ERLE = 10log_10(P_echo / P_out)
+  absl::optional<double> echo_return_loss_enhancement;
+  // Fraction of time that the AEC linear filter is divergent, in a 1-second
+  // non-overlapped aggregation window.
+  absl::optional<double> divergent_filter_fraction;
+
+  // The delay metrics consists of the delay median and standard deviation. It
+  // also consists of the fraction of delay estimates that can make the echo
+  // cancellation perform poorly. The values are aggregated until the first
+  // call to `GetStatistics()` and afterwards aggregated and updated every
+  // second. Note that if there are several clients pulling metrics from
+  // `GetStatistics()` during a session the first call from any of them will
+  // change to one second aggregation window for all.
+  absl::optional<int32_t> delay_median_ms;
+  absl::optional<int32_t> delay_standard_deviation_ms;
+
+  // Residual echo detector likelihood.
+  absl::optional<double> residual_echo_likelihood;
+  // Maximum residual echo likelihood from the last time period.
+  absl::optional<double> residual_echo_likelihood_recent_max;
+
+  // The instantaneous delay estimate produced in the AEC. The unit is in
+  // milliseconds and the value is the instantaneous value at the time of the
+  // call to `GetStatistics()`.
+  absl::optional<int32_t> delay_ms;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h b/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h
new file mode 100644
index 0000000000..2ea1a865c3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -0,0 +1,178 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/include/audio_processing_statistics.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+namespace test {
+class MockCustomProcessing : public CustomProcessing {
+ public:
+  virtual ~MockCustomProcessing() {}
+  MOCK_METHOD(void,
+              Initialize,
+              (int sample_rate_hz, int num_channels),
+              (override));
+  MOCK_METHOD(void, Process, (AudioBuffer * audio), (override));
+  MOCK_METHOD(void,
+              SetRuntimeSetting,
+              (AudioProcessing::RuntimeSetting setting),
+              (override));
+  MOCK_METHOD(std::string, ToString, (), (const, override));
+};
+
+class MockCustomAudioAnalyzer : public CustomAudioAnalyzer {
+ public:
+  virtual ~MockCustomAudioAnalyzer() {}
+  MOCK_METHOD(void,
+              Initialize,
+              (int sample_rate_hz, int num_channels),
+              (override));
+  MOCK_METHOD(void, Analyze, (const AudioBuffer* audio), (override));
+  MOCK_METHOD(std::string, ToString, (), (const, override));
+};
+
+class MockEchoControl : public EchoControl {
+ public:
+  virtual ~MockEchoControl() {}
+  MOCK_METHOD(void, AnalyzeRender, (AudioBuffer * render), (override));
+  MOCK_METHOD(void, AnalyzeCapture, (AudioBuffer * capture), (override));
+  MOCK_METHOD(void,
+              ProcessCapture,
+              (AudioBuffer * capture, bool echo_path_change),
+              (override));
+  MOCK_METHOD(void,
+              ProcessCapture,
+              (AudioBuffer * capture,
+               AudioBuffer* linear_output,
+               bool echo_path_change),
+              (override));
+  MOCK_METHOD(Metrics, GetMetrics, (), (const, override));
+  MOCK_METHOD(void, SetAudioBufferDelay, (int delay_ms), (override));
+  MOCK_METHOD(bool, ActiveProcessing, (), (const, override));
+};
+
+class MockEchoDetector : public EchoDetector {
+ public:
+  virtual ~MockEchoDetector() {}
+  MOCK_METHOD(void,
+              Initialize,
+              (int capture_sample_rate_hz,
+               int num_capture_channels,
+               int render_sample_rate_hz,
+               int num_render_channels),
+              (override));
+  MOCK_METHOD(void,
+              AnalyzeRenderAudio,
+              (rtc::ArrayView<const float> render_audio),
+              (override));
+  MOCK_METHOD(void,
+              AnalyzeCaptureAudio,
+              (rtc::ArrayView<const float> capture_audio),
+              (override));
+  MOCK_METHOD(Metrics, GetMetrics, (), (const, override));
+};
+
+class MockAudioProcessing : public AudioProcessing {
+ public:
+  MockAudioProcessing() {}
+
+  virtual ~MockAudioProcessing() {}
+
+  MOCK_METHOD(int, Initialize, (), (override));
+  MOCK_METHOD(int,
+              Initialize,
+              (const ProcessingConfig& processing_config),
+              (override));
+  MOCK_METHOD(void, ApplyConfig, (const Config& config), (override));
+  MOCK_METHOD(int, proc_sample_rate_hz, (), (const, override));
+  MOCK_METHOD(int, proc_split_sample_rate_hz, (), (const, override));
+  MOCK_METHOD(size_t, num_input_channels, (), (const, override));
+  MOCK_METHOD(size_t, num_proc_channels, (), (const, override));
+  MOCK_METHOD(size_t, num_output_channels, (), (const, override));
+  MOCK_METHOD(size_t, num_reverse_channels, (), (const, override));
+  MOCK_METHOD(void, set_output_will_be_muted, (bool muted), (override));
+  MOCK_METHOD(void, SetRuntimeSetting, (RuntimeSetting setting), (override));
+  MOCK_METHOD(bool, PostRuntimeSetting, (RuntimeSetting setting), (override));
+  MOCK_METHOD(int,
+              ProcessStream,
+              (const int16_t* const src,
+               const StreamConfig& input_config,
+               const StreamConfig& output_config,
+               int16_t* const dest),
+              (override));
+  MOCK_METHOD(int,
+              ProcessStream,
+              (const float* const* src,
+               const StreamConfig& input_config,
+               const StreamConfig& output_config,
+               float* const* dest),
+              (override));
+  MOCK_METHOD(int,
+              ProcessReverseStream,
+              (const int16_t* const src,
+               const StreamConfig& input_config,
+               const StreamConfig& output_config,
+               int16_t* const dest),
+              (override));
+  MOCK_METHOD(int,
+              AnalyzeReverseStream,
+              (const float* const* data, const StreamConfig& reverse_config),
+              (override));
+  MOCK_METHOD(int,
+              ProcessReverseStream,
+              (const float* const* src,
+               const StreamConfig& input_config,
+               const StreamConfig& output_config,
+               float* const* dest),
+              (override));
+  MOCK_METHOD(bool,
+              GetLinearAecOutput,
+              ((rtc::ArrayView<std::array<float, 160>> linear_output)),
+              (const, override));
+  MOCK_METHOD(int, set_stream_delay_ms, (int delay), (override));
+  MOCK_METHOD(int, stream_delay_ms, (), (const, override));
+  MOCK_METHOD(void, set_stream_key_pressed, (bool key_pressed), (override));
+  MOCK_METHOD(void, set_stream_analog_level, (int), (override));
+  MOCK_METHOD(int, recommended_stream_analog_level, (), (const, override));
+  MOCK_METHOD(bool,
+              CreateAndAttachAecDump,
+              (absl::string_view file_name,
+               int64_t max_log_size_bytes,
+               rtc::TaskQueue* worker_queue),
+              (override));
+  MOCK_METHOD(bool,
+              CreateAndAttachAecDump,
+              (FILE * handle,
+               int64_t max_log_size_bytes,
+               rtc::TaskQueue* worker_queue),
+              (override));
+  MOCK_METHOD(void, AttachAecDump, (std::unique_ptr<AecDump>), (override));
+  MOCK_METHOD(void, DetachAecDump, (), (override));
+
+  MOCK_METHOD(AudioProcessingStats, GetStatistics, (), (override));
+  MOCK_METHOD(AudioProcessingStats, GetStatistics, (bool), (override));
+
+  MOCK_METHOD(AudioProcessing::Config, GetConfig, (), (const, override));
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc
new file mode 100644
index 0000000000..a15321ad48
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/strings/string_builder.h"
+
+// Check to verify that the define is properly set.
+#if !defined(WEBRTC_APM_DEBUG_DUMP) || \
+    (WEBRTC_APM_DEBUG_DUMP != 0 && WEBRTC_APM_DEBUG_DUMP != 1)
+#error "Set WEBRTC_APM_DEBUG_DUMP to either 0 or 1"
+#endif
+
+namespace webrtc {
+namespace {
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+
+#if defined(WEBRTC_WIN)
+constexpr char kPathDelimiter = '\\';
+#else
+constexpr char kPathDelimiter = '/';
+#endif
+
+std::string FormFileName(absl::string_view output_dir,
+                         absl::string_view name,
+                         int instance_index,
+                         int reinit_index,
+                         absl::string_view suffix) {
+#ifdef WEBRTC_WIN
+  char sep = '\\';
+#else
+  char sep = '/';
+#endif
+
+  std::stringstream ss;
+  std::string base = rtc::LogMessage::aec_debug_filename();
+  ss << base;
+
+  if (base.length() && base.back() != sep) {
+    ss << sep;
+  }
+
+  ss << name << "_" << instance_index << "-" << reinit_index << suffix;
+  return ss.str();
+}
+#endif
+
+}  // namespace
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+ApmDataDumper::ApmDataDumper(int instance_index)
+    : instance_index_(instance_index)
+    , debug_written_(0) {}
+#else
+ApmDataDumper::ApmDataDumper(int instance_index) {}
+#endif
+
+ApmDataDumper::~ApmDataDumper() = default;
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+bool ApmDataDumper::recording_activated_ = false;
+absl::optional<int> ApmDataDumper::dump_set_to_use_;
+char ApmDataDumper::output_dir_[] = "";
+
+FILE* ApmDataDumper::GetRawFile(absl::string_view name) {
+  std::string filename = FormFileName(output_dir_, name, instance_index_,
+                                      recording_set_index_, ".dat");
+  auto& f = raw_files_[filename];
+  if (!f) {
+    f.reset(fopen(filename.c_str(), "wb"));
+    RTC_CHECK(f.get()) << "Cannot write to " << filename << ".";
+  }
+  return f.get();
+}
+
+WavWriter* ApmDataDumper::GetWavFile(absl::string_view name,
+                                     int sample_rate_hz,
+                                     int num_channels,
+                                     WavFile::SampleFormat format) {
+  std::string filename = FormFileName(output_dir_, name, instance_index_,
+                                      recording_set_index_, ".wav");
+  auto& f = wav_files_[filename];
+  if (!f) {
+    f.reset(
+        new WavWriter(filename.c_str(), sample_rate_hz, num_channels, format));
+  }
+  return f.get();
+}
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h
new file mode 100644
index 0000000000..aa8496819b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/logging/apm_data_dumper.h
@@ -0,0 +1,452 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
+#define MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
+
+#include <stdint.h>
+#include <stdio.h>
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+#include <memory>
+#include <string>
+#include <unordered_map>
+#endif
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#if WEBRTC_APM_DEBUG_DUMP == 1
+#include "common_audio/wav_file.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_utils.h"
+#endif
+
+// Check to verify that the define is properly set.
+#if !defined(WEBRTC_APM_DEBUG_DUMP) || \
+    (WEBRTC_APM_DEBUG_DUMP != 0 && WEBRTC_APM_DEBUG_DUMP != 1)
+#error "Set WEBRTC_APM_DEBUG_DUMP to either 0 or 1"
+#endif
+
+namespace webrtc {
+
+#if WEBRTC_APM_DEBUG_DUMP == 1
+// Functor used to use as a custom deleter in the map of file pointers to raw
+// files.
+struct RawFileCloseFunctor {
+  void operator()(FILE* f) const { if (f) fclose(f); }
+};
+#endif
+
+// Class that handles dumping of variables into files.
+class ApmDataDumper {
+ public:
+  // Constructor that takes an instance index that may
+  // be used to distinguish data dumped from different
+  // instances of the code.
+  explicit ApmDataDumper(int instance_index);
+
+  ApmDataDumper() = delete;
+  ApmDataDumper(const ApmDataDumper&) = delete;
+  ApmDataDumper& operator=(const ApmDataDumper&) = delete;
+
+  ~ApmDataDumper();
+
+  // Activates or deactivate the dumping functionality.
+  static void SetActivated(bool activated) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    recording_activated_ = activated;
+#endif
+  }
+
+  // Returns whether dumping functionality is enabled/available.
+  static bool IsAvailable() {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    return true;
+#else
+    return false;
+#endif
+  }
+
+  // Default dump set.
+  static constexpr size_t kDefaultDumpSet = 0;
+
+  // Specifies what dump set to use. All dump commands with a different dump set
+  // than the one specified will be discarded. If not specificed, all dump sets
+  // will be used.
+  static void SetDumpSetToUse(int dump_set_to_use) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    dump_set_to_use_ = dump_set_to_use;
+#endif
+  }
+
+  // Set an optional output directory.
+  static void SetOutputDirectory(absl::string_view output_dir) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength);
+    rtc::strcpyn(output_dir_, output_dir.size(), output_dir);
+#endif
+  }
+
+  // Reinitializes the data dumping such that new versions
+  // of all files being dumped to are created.
+  void InitiateNewSetOfRecordings() {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    ++recording_set_index_;
+    debug_written_ = 0;
+#endif
+  }
+
+  // Methods for performing dumping of data of various types into
+  // various formats.
+  void DumpRaw(absl::string_view name,
+               double v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const double* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const double> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               float v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const float* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const float> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name, bool v, int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, static_cast<int16_t>(v));
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const bool* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        for (size_t k = 0; k < v_length; ++k) {
+          int16_t value = static_cast<int16_t>(v[k]);
+          fwrite(&value, sizeof(value), 1, file);
+        }
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const bool> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               int16_t v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const int16_t* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const int16_t> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               int32_t v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const int32_t* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(&v, sizeof(v), 1, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               size_t v_length,
+               const size_t* v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      FILE* file = GetRawFile(name);
+      if (file) {
+        fwrite(v, sizeof(v[0]), v_length, file);
+      }
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const int32_t> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpRaw(name, v.size(), v.data());
+    }
+#endif
+  }
+
+  void DumpRaw(absl::string_view name,
+               rtc::ArrayView<const size_t> v,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    DumpRaw(name, v.size(), v.data());
+#endif
+  }
+
+  void DumpWav(absl::string_view name,
+               size_t v_length,
+               const float* v,
+               int sample_rate_hz,
+               int num_channels,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels,
+                                   WavFile::SampleFormat::kFloat);
+      file->WriteSamples(v, v_length);
+      // Cheat and use aec_near as a stand-in for "size of the largest file"
+      // in the dump.  We're looking to limit the total time, and that's a
+      // reasonable stand-in.
+      if (strcmp(name, "aec_near") == 0) {
+       updateDebugWritten(v_length * sizeof(float));
+      }
+    }
+#endif
+  }
+
+  void DumpWav(absl::string_view name,
+               rtc::ArrayView<const float> v,
+               int sample_rate_hz,
+               int num_channels,
+               int dump_set = kDefaultDumpSet) {
+#if WEBRTC_APM_DEBUG_DUMP == 1
+    if (dump_set_to_use_ && *dump_set_to_use_ != dump_set)
+      return;
+
+    if (recording_activated_) {
+      DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
+    }
+#endif
+  }
+
+ private:
+#if WEBRTC_APM_DEBUG_DUMP == 1
+  static bool recording_activated_;
+  static absl::optional<int> dump_set_to_use_;
+  static constexpr size_t kOutputDirMaxLength = 1024;
+  static char output_dir_[kOutputDirMaxLength];
+  const int instance_index_;
+  int recording_set_index_ = 0;
+  std::unordered_map<std::string, std::unique_ptr<FILE, RawFileCloseFunctor>>
+      raw_files_;
+  std::unordered_map<std::string, std::unique_ptr<WavWriter>> wav_files_;
+
+  FILE* GetRawFile(absl::string_view name);
+  WavWriter* GetWavFile(absl::string_view name,
+                        int sample_rate_hz,
+                        int num_channels,
+                        WavFile::SampleFormat format);
+
+  uint32_t debug_written_ = 0;
+
+  void updateDebugWritten(uint32_t amount) {
+    debug_written_ += amount;
+    if (debug_written_ >= webrtc::Trace::aec_debug_size()) {
+      SetActivated(false);
+    }
+  }
+
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/ns/BUILD.gn
new file mode 100644
index 0000000000..8c2e9dba84
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/BUILD.gn
@@ -0,0 +1,104 @@
+# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_static_library("ns") {
+  visibility = [ "*" ]
+  configs += [ "..:apm_debug_dump" ]
+  sources = [
+    "fast_math.cc",
+    "fast_math.h",
+    "histograms.cc",
+    "histograms.h",
+    "noise_estimator.cc",
+    "noise_estimator.h",
+    "noise_suppressor.cc",
+    "noise_suppressor.h",
+    "ns_common.h",
+    "ns_config.h",
+    "ns_fft.cc",
+    "ns_fft.h",
+    "prior_signal_model.cc",
+    "prior_signal_model.h",
+    "prior_signal_model_estimator.cc",
+    "prior_signal_model_estimator.h",
+    "quantile_noise_estimator.cc",
+    "quantile_noise_estimator.h",
+    "signal_model.cc",
+    "signal_model.h",
+    "signal_model_estimator.cc",
+    "signal_model_estimator.h",
+    "speech_probability_estimator.cc",
+    "speech_probability_estimator.h",
+    "suppression_params.cc",
+    "suppression_params.h",
+    "wiener_filter.cc",
+    "wiener_filter.h",
+  ]
+
+  defines = []
+  if (rtc_build_with_neon && target_cpu != "arm64") {
+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
+    cflags = [ "-mfpu=neon" ]
+  }
+
+  deps = [
+    "..:apm_logging",
+    "..:audio_buffer",
+    "..:high_pass_filter",
+    "../../../api:array_view",
+    "../../../common_audio:common_audio_c",
+    "../../../common_audio/third_party/ooura:fft_size_128",
+    "../../../common_audio/third_party/ooura:fft_size_256",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base/system:arch",
+    "../../../system_wrappers",
+    "../../../system_wrappers:field_trial",
+    "../../../system_wrappers:metrics",
+    "../utility:cascaded_biquad_filter",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+if (rtc_include_tests) {
+  rtc_source_set("ns_unittests") {
+    testonly = true
+
+    configs += [ "..:apm_debug_dump" ]
+    sources = [ "noise_suppressor_unittest.cc" ]
+
+    deps = [
+      ":ns",
+      "..:apm_logging",
+      "..:audio_buffer",
+      "..:audio_processing",
+      "..:high_pass_filter",
+      "../../../api:array_view",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:safe_minmax",
+      "../../../rtc_base:stringutils",
+      "../../../rtc_base/system:arch",
+      "../../../system_wrappers",
+      "../../../test:test_support",
+      "../utility:cascaded_biquad_filter",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+
+    defines = []
+
+    if (rtc_enable_protobuf) {
+      sources += []
+    }
+
+    if (!build_with_chromium) {
+      deps += [ "..:audio_processing_unittests" ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/fast_math.cc b/third_party/libwebrtc/modules/audio_processing/ns/fast_math.cc
new file mode 100644
index 0000000000..d13110c43f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/fast_math.cc
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/fast_math.h"
+
+#include <math.h>
+#include <stdint.h>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+float FastLog2f(float in) {
+  RTC_DCHECK_GT(in, .0f);
+  // Read and interpret float as uint32_t and then cast to float.
+  // This is done to extract the exponent (bits 30 - 23).
+  // "Right shift" of the exponent is then performed by multiplying
+  // with the constant (1/2^23). Finally, we subtract a constant to
+  // remove the bias (https://en.wikipedia.org/wiki/Exponent_bias).
+  union {
+    float dummy;
+    uint32_t a;
+  } x = {in};
+  float out = x.a;
+  out *= 1.1920929e-7f;  // 1/2^23
+  out -= 126.942695f;    // Remove bias.
+  return out;
+}
+
+}  // namespace
+
+float SqrtFastApproximation(float f) {
+  // TODO(peah): Add fast approximate implementation.
+  return sqrtf(f);
+}
+
+float Pow2Approximation(float p) {
+  // TODO(peah): Add fast approximate implementation.
+  return powf(2.f, p);
+}
+
+float PowApproximation(float x, float p) {
+  return Pow2Approximation(p * FastLog2f(x));
+}
+
+float LogApproximation(float x) {
+  constexpr float kLogOf2 = 0.69314718056f;
+  return FastLog2f(x) * kLogOf2;
+}
+
+void LogApproximation(rtc::ArrayView<const float> x, rtc::ArrayView<float> y) {
+  for (size_t k = 0; k < x.size(); ++k) {
+    y[k] = LogApproximation(x[k]);
+  }
+}
+
+float ExpApproximation(float x) {
+  constexpr float kLog10Ofe = 0.4342944819f;
+  return PowApproximation(10.f, x * kLog10Ofe);
+}
+
+void ExpApproximation(rtc::ArrayView<const float> x, rtc::ArrayView<float> y) {
+  for (size_t k = 0; k < x.size(); ++k) {
+    y[k] = ExpApproximation(x[k]);
+  }
+}
+
+void ExpApproximationSignFlip(rtc::ArrayView<const float> x,
+                              rtc::ArrayView<float> y) {
+  for (size_t k = 0; k < x.size(); ++k) {
+    y[k] = ExpApproximation(-x[k]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/fast_math.h b/third_party/libwebrtc/modules/audio_processing/ns/fast_math.h
new file mode 100644
index 0000000000..0aefee940b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/fast_math.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_FAST_MATH_H_
+#define MODULES_AUDIO_PROCESSING_NS_FAST_MATH_H_
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Sqrt approximation.
+float SqrtFastApproximation(float f);
+
+// Log base conversion log(x) = log2(x)/log2(e).
+float LogApproximation(float x);
+void LogApproximation(rtc::ArrayView<const float> x, rtc::ArrayView<float> y);
+
+// 2^x approximation.
+float Pow2Approximation(float p);
+
+// x^p approximation.
+float PowApproximation(float x, float p);
+
+// e^x approximation.
+float ExpApproximation(float x);
+void ExpApproximation(rtc::ArrayView<const float> x, rtc::ArrayView<float> y);
+void ExpApproximationSignFlip(rtc::ArrayView<const float> x,
+                              rtc::ArrayView<float> y);
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_FAST_MATH_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/histograms.cc b/third_party/libwebrtc/modules/audio_processing/ns/histograms.cc
new file mode 100644
index 0000000000..1d4f4590d2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/histograms.cc
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/histograms.h"
+
+namespace webrtc {
+
+Histograms::Histograms() {
+  Clear();
+}
+
+void Histograms::Clear() {
+  lrt_.fill(0);
+  spectral_flatness_.fill(0);
+  spectral_diff_.fill(0);
+}
+
+void Histograms::Update(const SignalModel& features_) {
+  // Update the histogram for the LRT.
+  constexpr float kOneByBinSizeLrt = 1.f / kBinSizeLrt;
+  if (features_.lrt < kHistogramSize * kBinSizeLrt && features_.lrt >= 0.f) {
+    ++lrt_[kOneByBinSizeLrt * features_.lrt];
+  }
+
+  // Update histogram for the spectral flatness.
+  constexpr float kOneByBinSizeSpecFlat = 1.f / kBinSizeSpecFlat;
+  if (features_.spectral_flatness < kHistogramSize * kBinSizeSpecFlat &&
+      features_.spectral_flatness >= 0.f) {
+    ++spectral_flatness_[features_.spectral_flatness * kOneByBinSizeSpecFlat];
+  }
+
+  // Update histogram for the spectral difference.
+  constexpr float kOneByBinSizeSpecDiff = 1.f / kBinSizeSpecDiff;
+  if (features_.spectral_diff < kHistogramSize * kBinSizeSpecDiff &&
+      features_.spectral_diff >= 0.f) {
+    ++spectral_diff_[features_.spectral_diff * kOneByBinSizeSpecDiff];
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/histograms.h b/third_party/libwebrtc/modules/audio_processing/ns/histograms.h
new file mode 100644
index 0000000000..9640e743cf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/histograms.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_HISTOGRAMS_H_
+#define MODULES_AUDIO_PROCESSING_NS_HISTOGRAMS_H_
+
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/ns/ns_common.h"
+#include "modules/audio_processing/ns/signal_model.h"
+
+namespace webrtc {
+
+constexpr int kHistogramSize = 1000;
+
+// Class for handling the updating of histograms.
+class Histograms {
+ public:
+  Histograms();
+  Histograms(const Histograms&) = delete;
+  Histograms& operator=(const Histograms&) = delete;
+
+  // Clears the histograms.
+  void Clear();
+
+  // Extracts thresholds for feature parameters and updates the corresponding
+  // histogram.
+  void Update(const SignalModel& features_);
+
+  // Methods for accessing the histograms.
+  rtc::ArrayView<const int, kHistogramSize> get_lrt() const { return lrt_; }
+  rtc::ArrayView<const int, kHistogramSize> get_spectral_flatness() const {
+    return spectral_flatness_;
+  }
+  rtc::ArrayView<const int, kHistogramSize> get_spectral_diff() const {
+    return spectral_diff_;
+  }
+
+ private:
+  std::array<int, kHistogramSize> lrt_;
+  std::array<int, kHistogramSize> spectral_flatness_;
+  std::array<int, kHistogramSize> spectral_diff_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_HISTOGRAMS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/noise_estimator.cc b/third_party/libwebrtc/modules/audio_processing/ns/noise_estimator.cc
new file mode 100644
index 0000000000..5367545f25
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/noise_estimator.cc
@@ -0,0 +1,195 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/noise_estimator.h"
+
+#include <algorithm>
+
+#include "modules/audio_processing/ns/fast_math.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Log(i).
+constexpr std::array<float, 129> log_table = {
+    0.f,       0.f,       0.f,       0.f,       0.f,       1.609438f, 1.791759f,
+    1.945910f, 2.079442f, 2.197225f, 2.302585f, 2.397895f, 2.484907f, 2.564949f,
+    2.639057f, 2.708050f, 2.772589f, 2.833213f, 2.890372f, 2.944439f, 2.995732f,
+    3.044522f, 3.091043f, 3.135494f, 3.178054f, 3.218876f, 3.258097f, 3.295837f,
+    3.332205f, 3.367296f, 3.401197f, 3.433987f, 3.465736f, 3.496507f, 3.526361f,
+    3.555348f, 3.583519f, 3.610918f, 3.637586f, 3.663562f, 3.688879f, 3.713572f,
+    3.737669f, 3.761200f, 3.784190f, 3.806663f, 3.828641f, 3.850147f, 3.871201f,
+    3.891820f, 3.912023f, 3.931826f, 3.951244f, 3.970292f, 3.988984f, 4.007333f,
+    4.025352f, 4.043051f, 4.060443f, 4.077538f, 4.094345f, 4.110874f, 4.127134f,
+    4.143135f, 4.158883f, 4.174387f, 4.189655f, 4.204693f, 4.219508f, 4.234107f,
+    4.248495f, 4.262680f, 4.276666f, 4.290460f, 4.304065f, 4.317488f, 4.330733f,
+    4.343805f, 4.356709f, 4.369448f, 4.382027f, 4.394449f, 4.406719f, 4.418841f,
+    4.430817f, 4.442651f, 4.454347f, 4.465908f, 4.477337f, 4.488636f, 4.499810f,
+    4.510859f, 4.521789f, 4.532599f, 4.543295f, 4.553877f, 4.564348f, 4.574711f,
+    4.584968f, 4.595119f, 4.605170f, 4.615121f, 4.624973f, 4.634729f, 4.644391f,
+    4.653960f, 4.663439f, 4.672829f, 4.682131f, 4.691348f, 4.700480f, 4.709530f,
+    4.718499f, 4.727388f, 4.736198f, 4.744932f, 4.753591f, 4.762174f, 4.770685f,
+    4.779124f, 4.787492f, 4.795791f, 4.804021f, 4.812184f, 4.820282f, 4.828314f,
+    4.836282f, 4.844187f, 4.852030f};
+
+}  // namespace
+
+NoiseEstimator::NoiseEstimator(const SuppressionParams& suppression_params)
+    : suppression_params_(suppression_params) {
+  noise_spectrum_.fill(0.f);
+  prev_noise_spectrum_.fill(0.f);
+  conservative_noise_spectrum_.fill(0.f);
+  parametric_noise_spectrum_.fill(0.f);
+}
+
+void NoiseEstimator::PrepareAnalysis() {
+  std::copy(noise_spectrum_.begin(), noise_spectrum_.end(),
+            prev_noise_spectrum_.begin());
+}
+
+void NoiseEstimator::PreUpdate(
+    int32_t num_analyzed_frames,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    float signal_spectral_sum) {
+  quantile_noise_estimator_.Estimate(signal_spectrum, noise_spectrum_);
+
+  if (num_analyzed_frames < kShortStartupPhaseBlocks) {
+    // Compute simplified noise model during startup.
+    const size_t kStartBand = 5;
+    float sum_log_i_log_magn = 0.f;
+    float sum_log_i = 0.f;
+    float sum_log_i_square = 0.f;
+    float sum_log_magn = 0.f;
+    for (size_t i = kStartBand; i < kFftSizeBy2Plus1; ++i) {
+      float log_i = log_table[i];
+      sum_log_i += log_i;
+      sum_log_i_square += log_i * log_i;
+      float log_signal = LogApproximation(signal_spectrum[i]);
+      sum_log_magn += log_signal;
+      sum_log_i_log_magn += log_i * log_signal;
+    }
+
+    // Estimate the parameter for the level of the white noise.
+    constexpr float kOneByFftSizeBy2Plus1 = 1.f / kFftSizeBy2Plus1;
+    white_noise_level_ += signal_spectral_sum * kOneByFftSizeBy2Plus1 *
+                          suppression_params_.over_subtraction_factor;
+
+    // Estimate pink noise parameters.
+    float denom = sum_log_i_square * (kFftSizeBy2Plus1 - kStartBand) -
+                  sum_log_i * sum_log_i;
+    float num =
+        sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn;
+    RTC_DCHECK_NE(denom, 0.f);
+    float pink_noise_adjustment = num / denom;
+
+    // Constrain the estimated spectrum to be positive.
+    pink_noise_adjustment = std::max(pink_noise_adjustment, 0.f);
+    pink_noise_numerator_ += pink_noise_adjustment;
+    num = sum_log_i * sum_log_magn -
+          (kFftSizeBy2Plus1 - kStartBand) * sum_log_i_log_magn;
+    RTC_DCHECK_NE(denom, 0.f);
+    pink_noise_adjustment = num / denom;
+
+    // Constrain the pink noise power to be in the interval [0, 1].
+    pink_noise_adjustment = std::max(std::min(pink_noise_adjustment, 1.f), 0.f);
+
+    pink_noise_exp_ += pink_noise_adjustment;
+
+    const float one_by_num_analyzed_frames_plus_1 =
+        1.f / (num_analyzed_frames + 1.f);
+
+    // Calculate the frequency-independent parts of parametric noise estimate.
+    float parametric_exp = 0.f;
+    float parametric_num = 0.f;
+    if (pink_noise_exp_ > 0.f) {
+      // Use pink noise estimate.
+      parametric_num = ExpApproximation(pink_noise_numerator_ *
+                                        one_by_num_analyzed_frames_plus_1);
+      parametric_num *= num_analyzed_frames + 1.f;
+      parametric_exp = pink_noise_exp_ * one_by_num_analyzed_frames_plus_1;
+    }
+
+    constexpr float kOneByShortStartupPhaseBlocks =
+        1.f / kShortStartupPhaseBlocks;
+    for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+      // Estimate the background noise using the white and pink noise
+      // parameters.
+      if (pink_noise_exp_ == 0.f) {
+        // Use white noise estimate.
+        parametric_noise_spectrum_[i] = white_noise_level_;
+      } else {
+        // Use pink noise estimate.
+        float use_band = i < kStartBand ? kStartBand : i;
+        float denom = PowApproximation(use_band, parametric_exp);
+        RTC_DCHECK_NE(denom, 0.f);
+        parametric_noise_spectrum_[i] = parametric_num / denom;
+      }
+    }
+
+    // Weight quantile noise with modeled noise.
+    for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+      noise_spectrum_[i] *= num_analyzed_frames;
+      float tmp = parametric_noise_spectrum_[i] *
+                  (kShortStartupPhaseBlocks - num_analyzed_frames);
+      noise_spectrum_[i] += tmp * one_by_num_analyzed_frames_plus_1;
+      noise_spectrum_[i] *= kOneByShortStartupPhaseBlocks;
+    }
+  }
+}
+
+void NoiseEstimator::PostUpdate(
+    rtc::ArrayView<const float> speech_probability,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum) {
+  // Time-avg parameter for noise_spectrum update.
+  constexpr float kNoiseUpdate = 0.9f;
+
+  float gamma = kNoiseUpdate;
+  for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+    const float prob_speech = speech_probability[i];
+    const float prob_non_speech = 1.f - prob_speech;
+
+    // Temporary noise update used for speech frames if update value is less
+    // than previous.
+    float noise_update_tmp =
+        gamma * prev_noise_spectrum_[i] +
+        (1.f - gamma) * (prob_non_speech * signal_spectrum[i] +
+                         prob_speech * prev_noise_spectrum_[i]);
+
+    // Time-constant based on speech/noise_spectrum state.
+    float gamma_old = gamma;
+
+    // Increase gamma for frame likely to be seech.
+    constexpr float kProbRange = .2f;
+    gamma = prob_speech > kProbRange ? .99f : kNoiseUpdate;
+
+    // Conservative noise_spectrum update.
+    if (prob_speech < kProbRange) {
+      conservative_noise_spectrum_[i] +=
+          0.05f * (signal_spectrum[i] - conservative_noise_spectrum_[i]);
+    }
+
+    // Noise_spectrum update.
+    if (gamma == gamma_old) {
+      noise_spectrum_[i] = noise_update_tmp;
+    } else {
+      noise_spectrum_[i] =
+          gamma * prev_noise_spectrum_[i] +
+          (1.f - gamma) * (prob_non_speech * signal_spectrum[i] +
+                           prob_speech * prev_noise_spectrum_[i]);
+      // Allow for noise_spectrum update downwards: If noise_spectrum update
+      // decreases the noise_spectrum, it is safe, so allow it to happen.
+      noise_spectrum_[i] = std::min(noise_spectrum_[i], noise_update_tmp);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/noise_estimator.h b/third_party/libwebrtc/modules/audio_processing/ns/noise_estimator.h
new file mode 100644
index 0000000000..0c0466a679
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/noise_estimator.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_NOISE_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_NS_NOISE_ESTIMATOR_H_
+
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/ns/ns_common.h"
+#include "modules/audio_processing/ns/quantile_noise_estimator.h"
+#include "modules/audio_processing/ns/suppression_params.h"
+
+namespace webrtc {
+
+// Class for estimating the spectral characteristics of the noise in an incoming
+// signal.
+class NoiseEstimator {
+ public:
+  explicit NoiseEstimator(const SuppressionParams& suppression_params);
+
+  // Prepare the estimator for analysis of a new frame.
+  void PrepareAnalysis();
+
+  // Performs the first step of the estimator update.
+  void PreUpdate(int32_t num_analyzed_frames,
+                 rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+                 float signal_spectral_sum);
+
+  // Performs the second step of the estimator update.
+  void PostUpdate(
+      rtc::ArrayView<const float> speech_probability,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum);
+
+  // Returns the noise spectral estimate.
+  rtc::ArrayView<const float, kFftSizeBy2Plus1> get_noise_spectrum() const {
+    return noise_spectrum_;
+  }
+
+  // Returns the noise from the previous frame.
+  rtc::ArrayView<const float, kFftSizeBy2Plus1> get_prev_noise_spectrum()
+      const {
+    return prev_noise_spectrum_;
+  }
+
+  // Returns a noise spectral estimate based on white and pink noise parameters.
+  rtc::ArrayView<const float, kFftSizeBy2Plus1> get_parametric_noise_spectrum()
+      const {
+    return parametric_noise_spectrum_;
+  }
+  rtc::ArrayView<const float, kFftSizeBy2Plus1>
+  get_conservative_noise_spectrum() const {
+    return conservative_noise_spectrum_;
+  }
+
+ private:
+  const SuppressionParams& suppression_params_;
+  float white_noise_level_ = 0.f;
+  float pink_noise_numerator_ = 0.f;
+  float pink_noise_exp_ = 0.f;
+  std::array<float, kFftSizeBy2Plus1> prev_noise_spectrum_;
+  std::array<float, kFftSizeBy2Plus1> conservative_noise_spectrum_;
+  std::array<float, kFftSizeBy2Plus1> parametric_noise_spectrum_;
+  std::array<float, kFftSizeBy2Plus1> noise_spectrum_;
+  QuantileNoiseEstimator quantile_noise_estimator_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_NOISE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor.cc b/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor.cc
new file mode 100644
index 0000000000..d66faa6ed4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor.cc
@@ -0,0 +1,555 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/noise_suppressor.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include <algorithm>
+
+#include "modules/audio_processing/ns/fast_math.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Maps sample rate to number of bands.
+size_t NumBandsForRate(size_t sample_rate_hz) {
+  RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000 ||
+             sample_rate_hz == 48000);
+  return sample_rate_hz / 16000;
+}
+
+// Maximum number of channels for which the channel data is stored on
+// the stack. If the number of channels are larger than this, they are stored
+// using scratch memory that is pre-allocated on the heap. The reason for this
+// partitioning is not to waste heap space for handling the more common numbers
+// of channels, while at the same time not limiting the support for higher
+// numbers of channels by enforcing the channel data to be stored on the
+// stack using a fixed maximum value.
+constexpr size_t kMaxNumChannelsOnStack = 2;
+
+// Chooses the number of channels to store on the heap when that is required due
+// to the number of channels being larger than the pre-defined number
+// of channels to store on the stack.
+size_t NumChannelsOnHeap(size_t num_channels) {
+  return num_channels > kMaxNumChannelsOnStack ? num_channels : 0;
+}
+
+// Hybrib Hanning and flat window for the filterbank.
+constexpr std::array<float, 96> kBlocks160w256FirstHalf = {
+    0.00000000f, 0.01636173f, 0.03271908f, 0.04906767f, 0.06540313f,
+    0.08172107f, 0.09801714f, 0.11428696f, 0.13052619f, 0.14673047f,
+    0.16289547f, 0.17901686f, 0.19509032f, 0.21111155f, 0.22707626f,
+    0.24298018f, 0.25881905f, 0.27458862f, 0.29028468f, 0.30590302f,
+    0.32143947f, 0.33688985f, 0.35225005f, 0.36751594f, 0.38268343f,
+    0.39774847f, 0.41270703f, 0.42755509f, 0.44228869f, 0.45690388f,
+    0.47139674f, 0.48576339f, 0.50000000f, 0.51410274f, 0.52806785f,
+    0.54189158f, 0.55557023f, 0.56910015f, 0.58247770f, 0.59569930f,
+    0.60876143f, 0.62166057f, 0.63439328f, 0.64695615f, 0.65934582f,
+    0.67155895f, 0.68359230f, 0.69544264f, 0.70710678f, 0.71858162f,
+    0.72986407f, 0.74095113f, 0.75183981f, 0.76252720f, 0.77301045f,
+    0.78328675f, 0.79335334f, 0.80320753f, 0.81284668f, 0.82226822f,
+    0.83146961f, 0.84044840f, 0.84920218f, 0.85772861f, 0.86602540f,
+    0.87409034f, 0.88192126f, 0.88951608f, 0.89687274f, 0.90398929f,
+    0.91086382f, 0.91749450f, 0.92387953f, 0.93001722f, 0.93590593f,
+    0.94154407f, 0.94693013f, 0.95206268f, 0.95694034f, 0.96156180f,
+    0.96592583f, 0.97003125f, 0.97387698f, 0.97746197f, 0.98078528f,
+    0.98384601f, 0.98664333f, 0.98917651f, 0.99144486f, 0.99344778f,
+    0.99518473f, 0.99665524f, 0.99785892f, 0.99879546f, 0.99946459f,
+    0.99986614f};
+
+// Applies the filterbank window to a buffer.
+void ApplyFilterBankWindow(rtc::ArrayView<float, kFftSize> x) {
+  for (size_t i = 0; i < 96; ++i) {
+    x[i] = kBlocks160w256FirstHalf[i] * x[i];
+  }
+
+  for (size_t i = 161, k = 95; i < kFftSize; ++i, --k) {
+    RTC_DCHECK_NE(0, k);
+    x[i] = kBlocks160w256FirstHalf[k] * x[i];
+  }
+}
+
+// Extends a frame with previous data.
+void FormExtendedFrame(rtc::ArrayView<const float, kNsFrameSize> frame,
+                       rtc::ArrayView<float, kFftSize - kNsFrameSize> old_data,
+                       rtc::ArrayView<float, kFftSize> extended_frame) {
+  std::copy(old_data.begin(), old_data.end(), extended_frame.begin());
+  std::copy(frame.begin(), frame.end(),
+            extended_frame.begin() + old_data.size());
+  std::copy(extended_frame.end() - old_data.size(), extended_frame.end(),
+            old_data.begin());
+}
+
+// Uses overlap-and-add to produce an output frame.
+void OverlapAndAdd(rtc::ArrayView<const float, kFftSize> extended_frame,
+                   rtc::ArrayView<float, kOverlapSize> overlap_memory,
+                   rtc::ArrayView<float, kNsFrameSize> output_frame) {
+  for (size_t i = 0; i < kOverlapSize; ++i) {
+    output_frame[i] = overlap_memory[i] + extended_frame[i];
+  }
+  std::copy(extended_frame.begin() + kOverlapSize,
+            extended_frame.begin() + kNsFrameSize,
+            output_frame.begin() + kOverlapSize);
+  std::copy(extended_frame.begin() + kNsFrameSize, extended_frame.end(),
+            overlap_memory.begin());
+}
+
+// Produces a delayed frame.
+void DelaySignal(rtc::ArrayView<const float, kNsFrameSize> frame,
+                 rtc::ArrayView<float, kFftSize - kNsFrameSize> delay_buffer,
+                 rtc::ArrayView<float, kNsFrameSize> delayed_frame) {
+  constexpr size_t kSamplesFromFrame = kNsFrameSize - (kFftSize - kNsFrameSize);
+  std::copy(delay_buffer.begin(), delay_buffer.end(), delayed_frame.begin());
+  std::copy(frame.begin(), frame.begin() + kSamplesFromFrame,
+            delayed_frame.begin() + delay_buffer.size());
+
+  std::copy(frame.begin() + kSamplesFromFrame, frame.end(),
+            delay_buffer.begin());
+}
+
+// Computes the energy of an extended frame.
+float ComputeEnergyOfExtendedFrame(rtc::ArrayView<const float, kFftSize> x) {
+  float energy = 0.f;
+  for (float x_k : x) {
+    energy += x_k * x_k;
+  }
+
+  return energy;
+}
+
+// Computes the energy of an extended frame based on its subcomponents.
+float ComputeEnergyOfExtendedFrame(
+    rtc::ArrayView<const float, kNsFrameSize> frame,
+    rtc::ArrayView<float, kFftSize - kNsFrameSize> old_data) {
+  float energy = 0.f;
+  for (float v : old_data) {
+    energy += v * v;
+  }
+  for (float v : frame) {
+    energy += v * v;
+  }
+
+  return energy;
+}
+
+// Computes the magnitude spectrum based on an FFT output.
+void ComputeMagnitudeSpectrum(
+    rtc::ArrayView<const float, kFftSize> real,
+    rtc::ArrayView<const float, kFftSize> imag,
+    rtc::ArrayView<float, kFftSizeBy2Plus1> signal_spectrum) {
+  signal_spectrum[0] = fabsf(real[0]) + 1.f;
+  signal_spectrum[kFftSizeBy2Plus1 - 1] =
+      fabsf(real[kFftSizeBy2Plus1 - 1]) + 1.f;
+
+  for (size_t i = 1; i < kFftSizeBy2Plus1 - 1; ++i) {
+    signal_spectrum[i] =
+        SqrtFastApproximation(real[i] * real[i] + imag[i] * imag[i]) + 1.f;
+  }
+}
+
+// Compute prior and post SNR.
+void ComputeSnr(rtc::ArrayView<const float, kFftSizeBy2Plus1> filter,
+                rtc::ArrayView<const float> prev_signal_spectrum,
+                rtc::ArrayView<const float> signal_spectrum,
+                rtc::ArrayView<const float> prev_noise_spectrum,
+                rtc::ArrayView<const float> noise_spectrum,
+                rtc::ArrayView<float> prior_snr,
+                rtc::ArrayView<float> post_snr) {
+  for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+    // Previous post SNR.
+    // Previous estimate: based on previous frame with gain filter.
+    float prev_estimate = prev_signal_spectrum[i] /
+                          (prev_noise_spectrum[i] + 0.0001f) * filter[i];
+    // Post SNR.
+    if (signal_spectrum[i] > noise_spectrum[i]) {
+      post_snr[i] = signal_spectrum[i] / (noise_spectrum[i] + 0.0001f) - 1.f;
+    } else {
+      post_snr[i] = 0.f;
+    }
+    // The directed decision estimate of the prior SNR is a sum the current and
+    // previous estimates.
+    prior_snr[i] = 0.98f * prev_estimate + (1.f - 0.98f) * post_snr[i];
+  }
+}
+
+// Computes the attenuating gain for the noise suppression of the upper bands.
+float ComputeUpperBandsGain(
+    float minimum_attenuating_gain,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> filter,
+    rtc::ArrayView<const float> speech_probability,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> prev_analysis_signal_spectrum,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum) {
+  // Average speech prob and filter gain for the end of the lowest band.
+  constexpr int kNumAvgBins = 32;
+  constexpr float kOneByNumAvgBins = 1.f / kNumAvgBins;
+
+  float avg_prob_speech = 0.f;
+  float avg_filter_gain = 0.f;
+  for (size_t i = kFftSizeBy2Plus1 - kNumAvgBins - 1; i < kFftSizeBy2Plus1 - 1;
+       i++) {
+    avg_prob_speech += speech_probability[i];
+    avg_filter_gain += filter[i];
+  }
+  avg_prob_speech = avg_prob_speech * kOneByNumAvgBins;
+  avg_filter_gain = avg_filter_gain * kOneByNumAvgBins;
+
+  // If the speech was suppressed by a component between Analyze and Process, an
+  // example being by an AEC, it should not be considered speech for the purpose
+  // of high band suppression. To that end, the speech probability is scaled
+  // accordingly.
+  float sum_analysis_spectrum = 0.f;
+  float sum_processing_spectrum = 0.f;
+  for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+    sum_analysis_spectrum += prev_analysis_signal_spectrum[i];
+    sum_processing_spectrum += signal_spectrum[i];
+  }
+
+  // The magnitude spectrum computation enforces the spectrum to be strictly
+  // positive.
+  RTC_DCHECK_GT(sum_analysis_spectrum, 0.f);
+  avg_prob_speech *= sum_processing_spectrum / sum_analysis_spectrum;
+
+  // Compute gain based on speech probability.
+  float gain =
+      0.5f * (1.f + static_cast<float>(tanh(2.f * avg_prob_speech - 1.f)));
+
+  // Combine gain with low band gain.
+  if (avg_prob_speech >= 0.5f) {
+    gain = 0.25f * gain + 0.75f * avg_filter_gain;
+  } else {
+    gain = 0.5f * gain + 0.5f * avg_filter_gain;
+  }
+
+  // Make sure gain is within flooring range.
+  return std::min(std::max(gain, minimum_attenuating_gain), 1.f);
+}
+
+}  // namespace
+
+NoiseSuppressor::ChannelState::ChannelState(
+    const SuppressionParams& suppression_params,
+    size_t num_bands)
+    : wiener_filter(suppression_params),
+      noise_estimator(suppression_params),
+      process_delay_memory(num_bands > 1 ? num_bands - 1 : 0) {
+  analyze_analysis_memory.fill(0.f);
+  prev_analysis_signal_spectrum.fill(1.f);
+  process_analysis_memory.fill(0.f);
+  process_synthesis_memory.fill(0.f);
+  for (auto& d : process_delay_memory) {
+    d.fill(0.f);
+  }
+}
+
+NoiseSuppressor::NoiseSuppressor(const NsConfig& config,
+                                 size_t sample_rate_hz,
+                                 size_t num_channels)
+    : num_bands_(NumBandsForRate(sample_rate_hz)),
+      num_channels_(num_channels),
+      suppression_params_(config.target_level),
+      filter_bank_states_heap_(NumChannelsOnHeap(num_channels_)),
+      upper_band_gains_heap_(NumChannelsOnHeap(num_channels_)),
+      energies_before_filtering_heap_(NumChannelsOnHeap(num_channels_)),
+      gain_adjustments_heap_(NumChannelsOnHeap(num_channels_)),
+      channels_(num_channels_) {
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    channels_[ch] =
+        std::make_unique<ChannelState>(suppression_params_, num_bands_);
+  }
+}
+
+void NoiseSuppressor::AggregateWienerFilters(
+    rtc::ArrayView<float, kFftSizeBy2Plus1> filter) const {
+  rtc::ArrayView<const float, kFftSizeBy2Plus1> filter0 =
+      channels_[0]->wiener_filter.get_filter();
+  std::copy(filter0.begin(), filter0.end(), filter.begin());
+
+  for (size_t ch = 1; ch < num_channels_; ++ch) {
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> filter_ch =
+        channels_[ch]->wiener_filter.get_filter();
+
+    for (size_t k = 0; k < kFftSizeBy2Plus1; ++k) {
+      filter[k] = std::min(filter[k], filter_ch[k]);
+    }
+  }
+}
+
+void NoiseSuppressor::Analyze(const AudioBuffer& audio) {
+  // Prepare the noise estimator for the analysis stage.
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    channels_[ch]->noise_estimator.PrepareAnalysis();
+  }
+
+  // Check for zero frames.
+  bool zero_frame = true;
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    rtc::ArrayView<const float, kNsFrameSize> y_band0(
+        &audio.split_bands_const(ch)[0][0], kNsFrameSize);
+    float energy = ComputeEnergyOfExtendedFrame(
+        y_band0, channels_[ch]->analyze_analysis_memory);
+    if (energy > 0.f) {
+      zero_frame = false;
+      break;
+    }
+  }
+
+  if (zero_frame) {
+    // We want to avoid updating statistics in this case:
+    // Updating feature statistics when we have zeros only will cause
+    // thresholds to move towards zero signal situations. This in turn has the
+    // effect that once the signal is "turned on" (non-zero values) everything
+    // will be treated as speech and there is no noise suppression effect.
+    // Depending on the duration of the inactive signal it takes a
+    // considerable amount of time for the system to learn what is noise and
+    // what is speech.
+    return;
+  }
+
+  // Only update analysis counter for frames that are properly analyzed.
+  if (++num_analyzed_frames_ < 0) {
+    num_analyzed_frames_ = 0;
+  }
+
+  // Analyze all channels.
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    std::unique_ptr<ChannelState>& ch_p = channels_[ch];
+    rtc::ArrayView<const float, kNsFrameSize> y_band0(
+        &audio.split_bands_const(ch)[0][0], kNsFrameSize);
+
+    // Form an extended frame and apply analysis filter bank windowing.
+    std::array<float, kFftSize> extended_frame;
+    FormExtendedFrame(y_band0, ch_p->analyze_analysis_memory, extended_frame);
+    ApplyFilterBankWindow(extended_frame);
+
+    // Compute the magnitude spectrum.
+    std::array<float, kFftSize> real;
+    std::array<float, kFftSize> imag;
+    fft_.Fft(extended_frame, real, imag);
+
+    std::array<float, kFftSizeBy2Plus1> signal_spectrum;
+    ComputeMagnitudeSpectrum(real, imag, signal_spectrum);
+
+    // Compute energies.
+    float signal_energy = 0.f;
+    for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+      signal_energy += real[i] * real[i] + imag[i] * imag[i];
+    }
+    signal_energy /= kFftSizeBy2Plus1;
+
+    float signal_spectral_sum = 0.f;
+    for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+      signal_spectral_sum += signal_spectrum[i];
+    }
+
+    // Estimate the noise spectra and the probability estimates of speech
+    // presence.
+    ch_p->noise_estimator.PreUpdate(num_analyzed_frames_, signal_spectrum,
+                                    signal_spectral_sum);
+
+    std::array<float, kFftSizeBy2Plus1> post_snr;
+    std::array<float, kFftSizeBy2Plus1> prior_snr;
+    ComputeSnr(ch_p->wiener_filter.get_filter(),
+               ch_p->prev_analysis_signal_spectrum, signal_spectrum,
+               ch_p->noise_estimator.get_prev_noise_spectrum(),
+               ch_p->noise_estimator.get_noise_spectrum(), prior_snr, post_snr);
+
+    ch_p->speech_probability_estimator.Update(
+        num_analyzed_frames_, prior_snr, post_snr,
+        ch_p->noise_estimator.get_conservative_noise_spectrum(),
+        signal_spectrum, signal_spectral_sum, signal_energy);
+
+    ch_p->noise_estimator.PostUpdate(
+        ch_p->speech_probability_estimator.get_probability(), signal_spectrum);
+
+    // Store the magnitude spectrum to make it avalilable for the process
+    // method.
+    std::copy(signal_spectrum.begin(), signal_spectrum.end(),
+              ch_p->prev_analysis_signal_spectrum.begin());
+  }
+}
+
+void NoiseSuppressor::Process(AudioBuffer* audio) {
+  // Select the space for storing data during the processing.
+  std::array<FilterBankState, kMaxNumChannelsOnStack> filter_bank_states_stack;
+  rtc::ArrayView<FilterBankState> filter_bank_states(
+      filter_bank_states_stack.data(), num_channels_);
+  std::array<float, kMaxNumChannelsOnStack> upper_band_gains_stack;
+  rtc::ArrayView<float> upper_band_gains(upper_band_gains_stack.data(),
+                                         num_channels_);
+  std::array<float, kMaxNumChannelsOnStack> energies_before_filtering_stack;
+  rtc::ArrayView<float> energies_before_filtering(
+      energies_before_filtering_stack.data(), num_channels_);
+  std::array<float, kMaxNumChannelsOnStack> gain_adjustments_stack;
+  rtc::ArrayView<float> gain_adjustments(gain_adjustments_stack.data(),
+                                         num_channels_);
+  if (NumChannelsOnHeap(num_channels_) > 0) {
+    // If the stack-allocated space is too small, use the heap for storing the
+    // data.
+    filter_bank_states = rtc::ArrayView<FilterBankState>(
+        filter_bank_states_heap_.data(), num_channels_);
+    upper_band_gains =
+        rtc::ArrayView<float>(upper_band_gains_heap_.data(), num_channels_);
+    energies_before_filtering = rtc::ArrayView<float>(
+        energies_before_filtering_heap_.data(), num_channels_);
+    gain_adjustments =
+        rtc::ArrayView<float>(gain_adjustments_heap_.data(), num_channels_);
+  }
+
+  // Compute the suppression filters for all channels.
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    // Form an extended frame and apply analysis filter bank windowing.
+    rtc::ArrayView<float, kNsFrameSize> y_band0(&audio->split_bands(ch)[0][0],
+                                                kNsFrameSize);
+
+    FormExtendedFrame(y_band0, channels_[ch]->process_analysis_memory,
+                      filter_bank_states[ch].extended_frame);
+
+    ApplyFilterBankWindow(filter_bank_states[ch].extended_frame);
+
+    energies_before_filtering[ch] =
+        ComputeEnergyOfExtendedFrame(filter_bank_states[ch].extended_frame);
+
+    // Perform filter bank analysis and compute the magnitude spectrum.
+    fft_.Fft(filter_bank_states[ch].extended_frame, filter_bank_states[ch].real,
+             filter_bank_states[ch].imag);
+
+    std::array<float, kFftSizeBy2Plus1> signal_spectrum;
+    ComputeMagnitudeSpectrum(filter_bank_states[ch].real,
+                             filter_bank_states[ch].imag, signal_spectrum);
+
+    // Compute the frequency domain gain filter for noise attenuation.
+    channels_[ch]->wiener_filter.Update(
+        num_analyzed_frames_,
+        channels_[ch]->noise_estimator.get_noise_spectrum(),
+        channels_[ch]->noise_estimator.get_prev_noise_spectrum(),
+        channels_[ch]->noise_estimator.get_parametric_noise_spectrum(),
+        signal_spectrum);
+
+    if (num_bands_ > 1) {
+      // Compute the time-domain gain for attenuating the noise in the upper
+      // bands.
+
+      upper_band_gains[ch] = ComputeUpperBandsGain(
+          suppression_params_.minimum_attenuating_gain,
+          channels_[ch]->wiener_filter.get_filter(),
+          channels_[ch]->speech_probability_estimator.get_probability(),
+          channels_[ch]->prev_analysis_signal_spectrum, signal_spectrum);
+    }
+  }
+
+  // Only do the below processing if the output of the audio processing module
+  // is used.
+  if (!capture_output_used_) {
+    return;
+  }
+
+  // Aggregate the Wiener filters for all channels.
+  std::array<float, kFftSizeBy2Plus1> filter_data;
+  rtc::ArrayView<const float, kFftSizeBy2Plus1> filter = filter_data;
+  if (num_channels_ == 1) {
+    filter = channels_[0]->wiener_filter.get_filter();
+  } else {
+    AggregateWienerFilters(filter_data);
+  }
+
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    // Apply the filter to the lower band.
+    for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+      filter_bank_states[ch].real[i] *= filter[i];
+      filter_bank_states[ch].imag[i] *= filter[i];
+    }
+  }
+
+  // Perform filter bank synthesis
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    fft_.Ifft(filter_bank_states[ch].real, filter_bank_states[ch].imag,
+              filter_bank_states[ch].extended_frame);
+  }
+
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    const float energy_after_filtering =
+        ComputeEnergyOfExtendedFrame(filter_bank_states[ch].extended_frame);
+
+    // Apply synthesis window.
+    ApplyFilterBankWindow(filter_bank_states[ch].extended_frame);
+
+    // Compute the adjustment of the noise attenuation filter based on the
+    // effect of the attenuation.
+    gain_adjustments[ch] =
+        channels_[ch]->wiener_filter.ComputeOverallScalingFactor(
+            num_analyzed_frames_,
+            channels_[ch]->speech_probability_estimator.get_prior_probability(),
+            energies_before_filtering[ch], energy_after_filtering);
+  }
+
+  // Select and apply adjustment of the noise attenuation filter based on the
+  // effect of the attenuation.
+  float gain_adjustment = gain_adjustments[0];
+  for (size_t ch = 1; ch < num_channels_; ++ch) {
+    gain_adjustment = std::min(gain_adjustment, gain_adjustments[ch]);
+  }
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    for (size_t i = 0; i < kFftSize; ++i) {
+      filter_bank_states[ch].extended_frame[i] =
+          gain_adjustment * filter_bank_states[ch].extended_frame[i];
+    }
+  }
+
+  // Use overlap-and-add to form the output frame of the lowest band.
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    rtc::ArrayView<float, kNsFrameSize> y_band0(&audio->split_bands(ch)[0][0],
+                                                kNsFrameSize);
+    OverlapAndAdd(filter_bank_states[ch].extended_frame,
+                  channels_[ch]->process_synthesis_memory, y_band0);
+  }
+
+  if (num_bands_ > 1) {
+    // Select the noise attenuating gain to apply to the upper band.
+    float upper_band_gain = upper_band_gains[0];
+    for (size_t ch = 1; ch < num_channels_; ++ch) {
+      upper_band_gain = std::min(upper_band_gain, upper_band_gains[ch]);
+    }
+
+    // Process the upper bands.
+    for (size_t ch = 0; ch < num_channels_; ++ch) {
+      for (size_t b = 1; b < num_bands_; ++b) {
+        // Delay the upper bands to match the delay of the filterbank applied to
+        // the lowest band.
+        rtc::ArrayView<float, kNsFrameSize> y_band(
+            &audio->split_bands(ch)[b][0], kNsFrameSize);
+        std::array<float, kNsFrameSize> delayed_frame;
+        DelaySignal(y_band, channels_[ch]->process_delay_memory[b - 1],
+                    delayed_frame);
+
+        // Apply the time-domain noise-attenuating gain.
+        for (size_t j = 0; j < kNsFrameSize; j++) {
+          y_band[j] = upper_band_gain * delayed_frame[j];
+        }
+      }
+    }
+  }
+
+  // Limit the output the allowed range.
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    for (size_t b = 0; b < num_bands_; ++b) {
+      rtc::ArrayView<float, kNsFrameSize> y_band(&audio->split_bands(ch)[b][0],
+                                                 kNsFrameSize);
+      for (size_t j = 0; j < kNsFrameSize; j++) {
+        y_band[j] = std::min(std::max(y_band[j], -32768.f), 32767.f);
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor.h b/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor.h
new file mode 100644
index 0000000000..1e321cf4a2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSOR_H_
+#define MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/ns/noise_estimator.h"
+#include "modules/audio_processing/ns/ns_common.h"
+#include "modules/audio_processing/ns/ns_config.h"
+#include "modules/audio_processing/ns/ns_fft.h"
+#include "modules/audio_processing/ns/speech_probability_estimator.h"
+#include "modules/audio_processing/ns/wiener_filter.h"
+
+namespace webrtc {
+
+// Class for suppressing noise in a signal.
+class NoiseSuppressor {
+ public:
+  NoiseSuppressor(const NsConfig& config,
+                  size_t sample_rate_hz,
+                  size_t num_channels);
+  NoiseSuppressor(const NoiseSuppressor&) = delete;
+  NoiseSuppressor& operator=(const NoiseSuppressor&) = delete;
+
+  // Analyses the signal (typically applied before the AEC to avoid analyzing
+  // any comfort noise signal).
+  void Analyze(const AudioBuffer& audio);
+
+  // Applies noise suppression.
+  void Process(AudioBuffer* audio);
+
+  // Specifies whether the capture output will be used. The purpose of this is
+  // to allow the noise suppressor to deactivate some of the processing when the
+  // resulting output is anyway not used, for instance when the endpoint is
+  // muted.
+  void SetCaptureOutputUsage(bool capture_output_used) {
+    capture_output_used_ = capture_output_used;
+  }
+
+ private:
+  const size_t num_bands_;
+  const size_t num_channels_;
+  const SuppressionParams suppression_params_;
+  int32_t num_analyzed_frames_ = -1;
+  NrFft fft_;
+  bool capture_output_used_ = true;
+
+  struct ChannelState {
+    ChannelState(const SuppressionParams& suppression_params, size_t num_bands);
+
+    SpeechProbabilityEstimator speech_probability_estimator;
+    WienerFilter wiener_filter;
+    NoiseEstimator noise_estimator;
+    std::array<float, kFftSizeBy2Plus1> prev_analysis_signal_spectrum;
+    std::array<float, kFftSize - kNsFrameSize> analyze_analysis_memory;
+    std::array<float, kOverlapSize> process_analysis_memory;
+    std::array<float, kOverlapSize> process_synthesis_memory;
+    std::vector<std::array<float, kOverlapSize>> process_delay_memory;
+  };
+
+  struct FilterBankState {
+    std::array<float, kFftSize> real;
+    std::array<float, kFftSize> imag;
+    std::array<float, kFftSize> extended_frame;
+  };
+
+  std::vector<FilterBankState> filter_bank_states_heap_;
+  std::vector<float> upper_band_gains_heap_;
+  std::vector<float> energies_before_filtering_heap_;
+  std::vector<float> gain_adjustments_heap_;
+  std::vector<std::unique_ptr<ChannelState>> channels_;
+
+  // Aggregates the Wiener filters into a single filter to use.
+  void AggregateWienerFilters(
+      rtc::ArrayView<float, kFftSizeBy2Plus1> filter) const;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor_unittest.cc b/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor_unittest.cc
new file mode 100644
index 0000000000..28ea63ae40
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor_unittest.cc
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/noise_suppressor.h"
+
+#include <deque>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "rtc_base/strings/string_builder.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+std::string ProduceDebugText(int sample_rate_hz,
+                             size_t num_channels,
+                             NsConfig::SuppressionLevel level) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz << ", num_channels: " << num_channels
+     << ", level: " << static_cast<int>(level);
+  return ss.Release();
+}
+
+void PopulateInputFrameWithIdenticalChannels(size_t num_channels,
+                                             size_t num_bands,
+                                             size_t frame_index,
+                                             AudioBuffer* audio) {
+  for (size_t ch = 0; ch < num_channels; ++ch) {
+    for (size_t b = 0; b < num_bands; ++b) {
+      for (size_t i = 0; i < 160; ++i) {
+        float value = static_cast<int>(frame_index * 160 + i);
+        audio->split_bands(ch)[b][i] = (value > 0 ? 5000 * b + value : 0);
+      }
+    }
+  }
+}
+
+void VerifyIdenticalChannels(size_t num_channels,
+                             size_t num_bands,
+                             size_t frame_index,
+                             const AudioBuffer& audio) {
+  EXPECT_GT(num_channels, 1u);
+  for (size_t ch = 1; ch < num_channels; ++ch) {
+    for (size_t b = 0; b < num_bands; ++b) {
+      for (size_t i = 0; i < 160; ++i) {
+        EXPECT_EQ(audio.split_bands_const(ch)[b][i],
+                  audio.split_bands_const(0)[b][i]);
+      }
+    }
+  }
+}
+
+}  // namespace
+
+// Verifies that the same noise reduction effect is applied to all channels.
+TEST(NoiseSuppressor, IdenticalChannelEffects) {
+  for (auto rate : {16000, 32000, 48000}) {
+    for (auto num_channels : {1, 4, 8}) {
+      for (auto level :
+           {NsConfig::SuppressionLevel::k6dB, NsConfig::SuppressionLevel::k12dB,
+            NsConfig::SuppressionLevel::k18dB,
+            NsConfig::SuppressionLevel::k21dB}) {
+        SCOPED_TRACE(ProduceDebugText(rate, num_channels, level));
+
+        const size_t num_bands = rate / 16000;
+        // const int frame_length = rtc::CheckedDivExact(rate, 100);
+        AudioBuffer audio(rate, num_channels, rate, num_channels, rate,
+                          num_channels);
+        NsConfig cfg;
+        NoiseSuppressor ns(cfg, rate, num_channels);
+        for (size_t frame_index = 0; frame_index < 1000; ++frame_index) {
+          if (rate > 16000) {
+            audio.SplitIntoFrequencyBands();
+          }
+
+          PopulateInputFrameWithIdenticalChannels(num_channels, num_bands,
+                                                  frame_index, &audio);
+
+          ns.Analyze(audio);
+          ns.Process(&audio);
+          if (num_channels > 1) {
+            VerifyIdenticalChannels(num_channels, num_bands, frame_index,
+                                    audio);
+          }
+        }
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/ns_common.h b/third_party/libwebrtc/modules/audio_processing/ns/ns_common.h
new file mode 100644
index 0000000000..d6149f72a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/ns_common.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_NS_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_NS_NS_COMMON_H_
+
+#include <cstddef>
+
+namespace webrtc {
+
+constexpr size_t kFftSize = 256;
+constexpr size_t kFftSizeBy2Plus1 = kFftSize / 2 + 1;
+constexpr size_t kNsFrameSize = 160;
+constexpr size_t kOverlapSize = kFftSize - kNsFrameSize;
+
+constexpr int kShortStartupPhaseBlocks = 50;
+constexpr int kLongStartupPhaseBlocks = 200;
+constexpr int kFeatureUpdateWindowSize = 500;
+
+constexpr float kLtrFeatureThr = 0.5f;
+constexpr float kBinSizeLrt = 0.1f;
+constexpr float kBinSizeSpecFlat = 0.05f;
+constexpr float kBinSizeSpecDiff = 0.1f;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_NS_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/ns_config.h b/third_party/libwebrtc/modules/audio_processing/ns/ns_config.h
new file mode 100644
index 0000000000..0a285e9cea
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/ns_config.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_NS_CONFIG_H_
+#define MODULES_AUDIO_PROCESSING_NS_NS_CONFIG_H_
+
+namespace webrtc {
+
+// Config struct for the noise suppressor
+struct NsConfig {
+  enum class SuppressionLevel { k6dB, k12dB, k18dB, k21dB };
+  SuppressionLevel target_level = SuppressionLevel::k12dB;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_NS_CONFIG_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/ns_fft.cc b/third_party/libwebrtc/modules/audio_processing/ns/ns_fft.cc
new file mode 100644
index 0000000000..264c46972c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/ns_fft.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/ns_fft.h"
+
+#include "common_audio/third_party/ooura/fft_size_256/fft4g.h"
+
+namespace webrtc {
+
+NrFft::NrFft() : bit_reversal_state_(kFftSize / 2), tables_(kFftSize / 2) {
+  // Initialize WebRtc_rdt (setting (bit_reversal_state_[0] to 0 triggers
+  // initialization)
+  bit_reversal_state_[0] = 0.f;
+  std::array<float, kFftSize> tmp_buffer;
+  tmp_buffer.fill(0.f);
+  WebRtc_rdft(kFftSize, 1, tmp_buffer.data(), bit_reversal_state_.data(),
+              tables_.data());
+}
+
+void NrFft::Fft(rtc::ArrayView<float, kFftSize> time_data,
+                rtc::ArrayView<float, kFftSize> real,
+                rtc::ArrayView<float, kFftSize> imag) {
+  WebRtc_rdft(kFftSize, 1, time_data.data(), bit_reversal_state_.data(),
+              tables_.data());
+
+  imag[0] = 0;
+  real[0] = time_data[0];
+
+  imag[kFftSizeBy2Plus1 - 1] = 0;
+  real[kFftSizeBy2Plus1 - 1] = time_data[1];
+
+  for (size_t i = 1; i < kFftSizeBy2Plus1 - 1; ++i) {
+    real[i] = time_data[2 * i];
+    imag[i] = time_data[2 * i + 1];
+  }
+}
+
+void NrFft::Ifft(rtc::ArrayView<const float> real,
+                 rtc::ArrayView<const float> imag,
+                 rtc::ArrayView<float> time_data) {
+  time_data[0] = real[0];
+  time_data[1] = real[kFftSizeBy2Plus1 - 1];
+  for (size_t i = 1; i < kFftSizeBy2Plus1 - 1; ++i) {
+    time_data[2 * i] = real[i];
+    time_data[2 * i + 1] = imag[i];
+  }
+  WebRtc_rdft(kFftSize, -1, time_data.data(), bit_reversal_state_.data(),
+              tables_.data());
+
+  // Scale the output
+  constexpr float kScaling = 2.f / kFftSize;
+  for (float& d : time_data) {
+    d *= kScaling;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/ns_fft.h b/third_party/libwebrtc/modules/audio_processing/ns/ns_fft.h
new file mode 100644
index 0000000000..539251eef2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/ns_fft.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_NS_FFT_H_
+#define MODULES_AUDIO_PROCESSING_NS_NS_FFT_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/ns/ns_common.h"
+
+namespace webrtc {
+
+// Wrapper class providing 256 point FFT functionality.
+class NrFft {
+ public:
+  NrFft();
+  NrFft(const NrFft&) = delete;
+  NrFft& operator=(const NrFft&) = delete;
+
+  // Transforms the signal from time to frequency domain.
+  void Fft(rtc::ArrayView<float, kFftSize> time_data,
+           rtc::ArrayView<float, kFftSize> real,
+           rtc::ArrayView<float, kFftSize> imag);
+
+  // Transforms the signal from frequency to time domain.
+  void Ifft(rtc::ArrayView<const float> real,
+            rtc::ArrayView<const float> imag,
+            rtc::ArrayView<float> time_data);
+
+ private:
+  std::vector<size_t> bit_reversal_state_;
+  std::vector<float> tables_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_NS_FFT_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/ns_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/ns/ns_gn/moz.build
new file mode 100644
index 0000000000..12076eedcb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/ns_gn/moz.build
@@ -0,0 +1,245 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_APM_DEBUG_DUMP"] = "0"
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/ns/fast_math.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/histograms.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/noise_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/noise_suppressor.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/ns_fft.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/quantile_noise_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/signal_model.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/signal_model_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/speech_probability_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/suppression_params.cc",
+    "/third_party/libwebrtc/modules/audio_processing/ns/wiener_filter.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("ns_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model.cc b/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model.cc
new file mode 100644
index 0000000000..f25a1e2060
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model.cc
@@ -0,0 +1,18 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/prior_signal_model.h"
+
+namespace webrtc {
+
+PriorSignalModel::PriorSignalModel(float lrt_initial_value)
+    : lrt(lrt_initial_value) {}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model.h b/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model.h
new file mode 100644
index 0000000000..dcfa7ea709
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_PRIOR_SIGNAL_MODEL_H_
+#define MODULES_AUDIO_PROCESSING_NS_PRIOR_SIGNAL_MODEL_H_
+
+namespace webrtc {
+
+// Struct for storing the prior signal model parameters.
+struct PriorSignalModel {
+  explicit PriorSignalModel(float lrt_initial_value);
+  PriorSignalModel(const PriorSignalModel&) = delete;
+  PriorSignalModel& operator=(const PriorSignalModel&) = delete;
+
+  float lrt;
+  float flatness_threshold = .5f;
+  float template_diff_threshold = .5f;
+  float lrt_weighting = 1.f;
+  float flatness_weighting = 0.f;
+  float difference_weighting = 0.f;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_PRIOR_SIGNAL_MODEL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model_estimator.cc b/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model_estimator.cc
new file mode 100644
index 0000000000..c814658e57
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model_estimator.cc
@@ -0,0 +1,170 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/prior_signal_model_estimator.h"
+
+#include <math.h>
+#include <algorithm>
+
+#include "modules/audio_processing/ns/fast_math.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Identifies the first of the two largest peaks in the histogram.
+void FindFirstOfTwoLargestPeaks(
+    float bin_size,
+    rtc::ArrayView<const int, kHistogramSize> spectral_flatness,
+    float* peak_position,
+    int* peak_weight) {
+  RTC_DCHECK(peak_position);
+  RTC_DCHECK(peak_weight);
+
+  int peak_value = 0;
+  int secondary_peak_value = 0;
+  *peak_position = 0.f;
+  float secondary_peak_position = 0.f;
+  *peak_weight = 0;
+  int secondary_peak_weight = 0;
+
+  // Identify the two largest peaks.
+  for (int i = 0; i < kHistogramSize; ++i) {
+    const float bin_mid = (i + 0.5f) * bin_size;
+    if (spectral_flatness[i] > peak_value) {
+      // Found new "first" peak candidate.
+      secondary_peak_value = peak_value;
+      secondary_peak_weight = *peak_weight;
+      secondary_peak_position = *peak_position;
+
+      peak_value = spectral_flatness[i];
+      *peak_weight = spectral_flatness[i];
+      *peak_position = bin_mid;
+    } else if (spectral_flatness[i] > secondary_peak_value) {
+      // Found new "second" peak candidate.
+      secondary_peak_value = spectral_flatness[i];
+      secondary_peak_weight = spectral_flatness[i];
+      secondary_peak_position = bin_mid;
+    }
+  }
+
+  // Merge the peaks if they are close.
+  if ((fabs(secondary_peak_position - *peak_position) < 2 * bin_size) &&
+      (secondary_peak_weight > 0.5f * (*peak_weight))) {
+    *peak_weight += secondary_peak_weight;
+    *peak_position = 0.5f * (*peak_position + secondary_peak_position);
+  }
+}
+
+void UpdateLrt(rtc::ArrayView<const int, kHistogramSize> lrt_histogram,
+               float* prior_model_lrt,
+               bool* low_lrt_fluctuations) {
+  RTC_DCHECK(prior_model_lrt);
+  RTC_DCHECK(low_lrt_fluctuations);
+
+  float average = 0.f;
+  float average_compl = 0.f;
+  float average_squared = 0.f;
+  int count = 0;
+
+  for (int i = 0; i < 10; ++i) {
+    float bin_mid = (i + 0.5f) * kBinSizeLrt;
+    average += lrt_histogram[i] * bin_mid;
+    count += lrt_histogram[i];
+  }
+  if (count > 0) {
+    average = average / count;
+  }
+
+  for (int i = 0; i < kHistogramSize; ++i) {
+    float bin_mid = (i + 0.5f) * kBinSizeLrt;
+    average_squared += lrt_histogram[i] * bin_mid * bin_mid;
+    average_compl += lrt_histogram[i] * bin_mid;
+  }
+  constexpr float kOneFeatureUpdateWindowSize = 1.f / kFeatureUpdateWindowSize;
+  average_squared = average_squared * kOneFeatureUpdateWindowSize;
+  average_compl = average_compl * kOneFeatureUpdateWindowSize;
+
+  // Fluctuation limit of LRT feature.
+  *low_lrt_fluctuations = average_squared - average * average_compl < 0.05f;
+
+  // Get threshold for LRT feature.
+  constexpr float kMaxLrt = 1.f;
+  constexpr float kMinLrt = .2f;
+  if (*low_lrt_fluctuations) {
+    // Very low fluctuation, so likely noise.
+    *prior_model_lrt = kMaxLrt;
+  } else {
+    *prior_model_lrt = std::min(kMaxLrt, std::max(kMinLrt, 1.2f * average));
+  }
+}
+
+}  // namespace
+
+PriorSignalModelEstimator::PriorSignalModelEstimator(float lrt_initial_value)
+    : prior_model_(lrt_initial_value) {}
+
+// Extract thresholds for feature parameters and computes the threshold/weights.
+void PriorSignalModelEstimator::Update(const Histograms& histograms) {
+  bool low_lrt_fluctuations;
+  UpdateLrt(histograms.get_lrt(), &prior_model_.lrt, &low_lrt_fluctuations);
+
+  // For spectral flatness and spectral difference: compute the main peaks of
+  // the histograms.
+  float spectral_flatness_peak_position;
+  int spectral_flatness_peak_weight;
+  FindFirstOfTwoLargestPeaks(
+      kBinSizeSpecFlat, histograms.get_spectral_flatness(),
+      &spectral_flatness_peak_position, &spectral_flatness_peak_weight);
+
+  float spectral_diff_peak_position = 0.f;
+  int spectral_diff_peak_weight = 0;
+  FindFirstOfTwoLargestPeaks(kBinSizeSpecDiff, histograms.get_spectral_diff(),
+                             &spectral_diff_peak_position,
+                             &spectral_diff_peak_weight);
+
+  // Reject if weight of peaks is not large enough, or peak value too small.
+  // Peak limit for spectral flatness (varies between 0 and 1).
+  const int use_spec_flat = spectral_flatness_peak_weight < 0.3f * 500 ||
+                                    spectral_flatness_peak_position < 0.6f
+                                ? 0
+                                : 1;
+
+  // Reject if weight of peaks is not large enough or if fluctuation of the LRT
+  // feature are very low, indicating a noise state.
+  const int use_spec_diff =
+      spectral_diff_peak_weight < 0.3f * 500 || low_lrt_fluctuations ? 0 : 1;
+
+  // Update the model.
+  prior_model_.template_diff_threshold = 1.2f * spectral_diff_peak_position;
+  prior_model_.template_diff_threshold =
+      std::min(1.f, std::max(0.16f, prior_model_.template_diff_threshold));
+
+  float one_by_feature_sum = 1.f / (1.f + use_spec_flat + use_spec_diff);
+  prior_model_.lrt_weighting = one_by_feature_sum;
+
+  if (use_spec_flat == 1) {
+    prior_model_.flatness_threshold = 0.9f * spectral_flatness_peak_position;
+    prior_model_.flatness_threshold =
+        std::min(.95f, std::max(0.1f, prior_model_.flatness_threshold));
+    prior_model_.flatness_weighting = one_by_feature_sum;
+  } else {
+    prior_model_.flatness_weighting = 0.f;
+  }
+
+  if (use_spec_diff == 1) {
+    prior_model_.difference_weighting = one_by_feature_sum;
+  } else {
+    prior_model_.difference_weighting = 0.f;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model_estimator.h b/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model_estimator.h
new file mode 100644
index 0000000000..d178323dba
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/prior_signal_model_estimator.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_PRIOR_SIGNAL_MODEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_NS_PRIOR_SIGNAL_MODEL_ESTIMATOR_H_
+
+#include "modules/audio_processing/ns/histograms.h"
+#include "modules/audio_processing/ns/prior_signal_model.h"
+
+namespace webrtc {
+
+// Estimator of the prior signal model parameters.
+class PriorSignalModelEstimator {
+ public:
+  explicit PriorSignalModelEstimator(float lrt_initial_value);
+  PriorSignalModelEstimator(const PriorSignalModelEstimator&) = delete;
+  PriorSignalModelEstimator& operator=(const PriorSignalModelEstimator&) =
+      delete;
+
+  // Updates the model estimate.
+  void Update(const Histograms& h);
+
+  // Returns the estimated model.
+  const PriorSignalModel& get_prior_model() const { return prior_model_; }
+
+ private:
+  PriorSignalModel prior_model_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_PRIOR_SIGNAL_MODEL_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/quantile_noise_estimator.cc b/third_party/libwebrtc/modules/audio_processing/ns/quantile_noise_estimator.cc
new file mode 100644
index 0000000000..bab494ff21
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/quantile_noise_estimator.cc
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/quantile_noise_estimator.h"
+
+#include <algorithm>
+
+#include "modules/audio_processing/ns/fast_math.h"
+
+namespace webrtc {
+
+QuantileNoiseEstimator::QuantileNoiseEstimator() {
+  quantile_.fill(0.f);
+  density_.fill(0.3f);
+  log_quantile_.fill(8.f);
+
+  constexpr float kOneBySimult = 1.f / kSimult;
+  for (size_t i = 0; i < kSimult; ++i) {
+    counter_[i] = floor(kLongStartupPhaseBlocks * (i + 1.f) * kOneBySimult);
+  }
+}
+
+void QuantileNoiseEstimator::Estimate(
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    rtc::ArrayView<float, kFftSizeBy2Plus1> noise_spectrum) {
+  std::array<float, kFftSizeBy2Plus1> log_spectrum;
+  LogApproximation(signal_spectrum, log_spectrum);
+
+  int quantile_index_to_return = -1;
+  // Loop over simultaneous estimates.
+  for (int s = 0, k = 0; s < kSimult;
+       ++s, k += static_cast<int>(kFftSizeBy2Plus1)) {
+    const float one_by_counter_plus_1 = 1.f / (counter_[s] + 1.f);
+    for (int i = 0, j = k; i < static_cast<int>(kFftSizeBy2Plus1); ++i, ++j) {
+      // Update log quantile estimate.
+      const float delta = density_[j] > 1.f ? 40.f / density_[j] : 40.f;
+
+      const float multiplier = delta * one_by_counter_plus_1;
+      if (log_spectrum[i] > log_quantile_[j]) {
+        log_quantile_[j] += 0.25f * multiplier;
+      } else {
+        log_quantile_[j] -= 0.75f * multiplier;
+      }
+
+      // Update density estimate.
+      constexpr float kWidth = 0.01f;
+      constexpr float kOneByWidthPlus2 = 1.f / (2.f * kWidth);
+      if (fabs(log_spectrum[i] - log_quantile_[j]) < kWidth) {
+        density_[j] = (counter_[s] * density_[j] + kOneByWidthPlus2) *
+                      one_by_counter_plus_1;
+      }
+    }
+
+    if (counter_[s] >= kLongStartupPhaseBlocks) {
+      counter_[s] = 0;
+      if (num_updates_ >= kLongStartupPhaseBlocks) {
+        quantile_index_to_return = k;
+      }
+    }
+
+    ++counter_[s];
+  }
+
+  // Sequentially update the noise during startup.
+  if (num_updates_ < kLongStartupPhaseBlocks) {
+    // Use the last "s" to get noise during startup that differ from zero.
+    quantile_index_to_return = kFftSizeBy2Plus1 * (kSimult - 1);
+    ++num_updates_;
+  }
+
+  if (quantile_index_to_return >= 0) {
+    ExpApproximation(
+        rtc::ArrayView<const float>(&log_quantile_[quantile_index_to_return],
+                                    kFftSizeBy2Plus1),
+        quantile_);
+  }
+
+  std::copy(quantile_.begin(), quantile_.end(), noise_spectrum.begin());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/quantile_noise_estimator.h b/third_party/libwebrtc/modules/audio_processing/ns/quantile_noise_estimator.h
new file mode 100644
index 0000000000..67d1512209
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/quantile_noise_estimator.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_QUANTILE_NOISE_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_NS_QUANTILE_NOISE_ESTIMATOR_H_
+
+#include <math.h>
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/ns/ns_common.h"
+
+namespace webrtc {
+
+constexpr int kSimult = 3;
+
+// For quantile noise estimation.
+class QuantileNoiseEstimator {
+ public:
+  QuantileNoiseEstimator();
+  QuantileNoiseEstimator(const QuantileNoiseEstimator&) = delete;
+  QuantileNoiseEstimator& operator=(const QuantileNoiseEstimator&) = delete;
+
+  // Estimate noise.
+  void Estimate(rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+                rtc::ArrayView<float, kFftSizeBy2Plus1> noise_spectrum);
+
+ private:
+  std::array<float, kSimult * kFftSizeBy2Plus1> density_;
+  std::array<float, kSimult * kFftSizeBy2Plus1> log_quantile_;
+  std::array<float, kFftSizeBy2Plus1> quantile_;
+  std::array<int, kSimult> counter_;
+  int num_updates_ = 1;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_QUANTILE_NOISE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/signal_model.cc b/third_party/libwebrtc/modules/audio_processing/ns/signal_model.cc
new file mode 100644
index 0000000000..364bfd00d8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/signal_model.cc
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/signal_model.h"
+
+namespace webrtc {
+
+SignalModel::SignalModel() {
+  constexpr float kSfFeatureThr = 0.5f;
+
+  lrt = kLtrFeatureThr;
+  spectral_flatness = kSfFeatureThr;
+  spectral_diff = kSfFeatureThr;
+  avg_log_lrt.fill(kLtrFeatureThr);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/signal_model.h b/third_party/libwebrtc/modules/audio_processing/ns/signal_model.h
new file mode 100644
index 0000000000..6614d38a38
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/signal_model.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_SIGNAL_MODEL_H_
+#define MODULES_AUDIO_PROCESSING_NS_SIGNAL_MODEL_H_
+
+#include <array>
+
+#include "modules/audio_processing/ns/ns_common.h"
+
+namespace webrtc {
+
+struct SignalModel {
+  SignalModel();
+  SignalModel(const SignalModel&) = delete;
+  SignalModel& operator=(const SignalModel&) = delete;
+
+  float lrt;
+  float spectral_diff;
+  float spectral_flatness;
+  // Log LRT factor with time-smoothing.
+  std::array<float, kFftSizeBy2Plus1> avg_log_lrt;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_SIGNAL_MODEL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/signal_model_estimator.cc b/third_party/libwebrtc/modules/audio_processing/ns/signal_model_estimator.cc
new file mode 100644
index 0000000000..67dd3bb687
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/signal_model_estimator.cc
@@ -0,0 +1,175 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/signal_model_estimator.h"
+
+#include "modules/audio_processing/ns/fast_math.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr float kOneByFftSizeBy2Plus1 = 1.f / kFftSizeBy2Plus1;
+
+// Computes the difference measure between input spectrum and a template/learned
+// noise spectrum.
+float ComputeSpectralDiff(
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    float signal_spectral_sum,
+    float diff_normalization) {
+  // spectral_diff = var(signal_spectrum) - cov(signal_spectrum, magnAvgPause)^2
+  // / var(magnAvgPause)
+
+  // Compute average quantities.
+  float noise_average = 0.f;
+  for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+    // Conservative smooth noise spectrum from pause frames.
+    noise_average += conservative_noise_spectrum[i];
+  }
+  noise_average = noise_average * kOneByFftSizeBy2Plus1;
+  float signal_average = signal_spectral_sum * kOneByFftSizeBy2Plus1;
+
+  // Compute variance and covariance quantities.
+  float covariance = 0.f;
+  float noise_variance = 0.f;
+  float signal_variance = 0.f;
+  for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+    float signal_diff = signal_spectrum[i] - signal_average;
+    float noise_diff = conservative_noise_spectrum[i] - noise_average;
+    covariance += signal_diff * noise_diff;
+    noise_variance += noise_diff * noise_diff;
+    signal_variance += signal_diff * signal_diff;
+  }
+  covariance *= kOneByFftSizeBy2Plus1;
+  noise_variance *= kOneByFftSizeBy2Plus1;
+  signal_variance *= kOneByFftSizeBy2Plus1;
+
+  // Update of average magnitude spectrum.
+  float spectral_diff =
+      signal_variance - (covariance * covariance) / (noise_variance + 0.0001f);
+  // Normalize.
+  return spectral_diff / (diff_normalization + 0.0001f);
+}
+
+// Updates the spectral flatness based on the input spectrum.
+void UpdateSpectralFlatness(
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    float signal_spectral_sum,
+    float* spectral_flatness) {
+  RTC_DCHECK(spectral_flatness);
+
+  // Compute log of ratio of the geometric to arithmetic mean (handle the log(0)
+  // separately).
+  constexpr float kAveraging = 0.3f;
+  float avg_spect_flatness_num = 0.f;
+  for (size_t i = 1; i < kFftSizeBy2Plus1; ++i) {
+    if (signal_spectrum[i] == 0.f) {
+      *spectral_flatness -= kAveraging * (*spectral_flatness);
+      return;
+    }
+  }
+
+  for (size_t i = 1; i < kFftSizeBy2Plus1; ++i) {
+    avg_spect_flatness_num += LogApproximation(signal_spectrum[i]);
+  }
+
+  float avg_spect_flatness_denom = signal_spectral_sum - signal_spectrum[0];
+
+  avg_spect_flatness_denom = avg_spect_flatness_denom * kOneByFftSizeBy2Plus1;
+  avg_spect_flatness_num = avg_spect_flatness_num * kOneByFftSizeBy2Plus1;
+
+  float spectral_tmp =
+      ExpApproximation(avg_spect_flatness_num) / avg_spect_flatness_denom;
+
+  // Time-avg update of spectral flatness feature.
+  *spectral_flatness += kAveraging * (spectral_tmp - *spectral_flatness);
+}
+
+// Updates the log LRT measures.
+void UpdateSpectralLrt(rtc::ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
+                       rtc::ArrayView<const float, kFftSizeBy2Plus1> post_snr,
+                       rtc::ArrayView<float, kFftSizeBy2Plus1> avg_log_lrt,
+                       float* lrt) {
+  RTC_DCHECK(lrt);
+
+  for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+    float tmp1 = 1.f + 2.f * prior_snr[i];
+    float tmp2 = 2.f * prior_snr[i] / (tmp1 + 0.0001f);
+    float bessel_tmp = (post_snr[i] + 1.f) * tmp2;
+    avg_log_lrt[i] +=
+        .5f * (bessel_tmp - LogApproximation(tmp1) - avg_log_lrt[i]);
+  }
+
+  float log_lrt_time_avg_k_sum = 0.f;
+  for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+    log_lrt_time_avg_k_sum += avg_log_lrt[i];
+  }
+  *lrt = log_lrt_time_avg_k_sum * kOneByFftSizeBy2Plus1;
+}
+
+}  // namespace
+
+SignalModelEstimator::SignalModelEstimator()
+    : prior_model_estimator_(kLtrFeatureThr) {}
+
+void SignalModelEstimator::AdjustNormalization(int32_t num_analyzed_frames,
+                                               float signal_energy) {
+  diff_normalization_ *= num_analyzed_frames;
+  diff_normalization_ += signal_energy;
+  diff_normalization_ /= (num_analyzed_frames + 1);
+}
+
+// Update the noise features.
+void SignalModelEstimator::Update(
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> post_snr,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    float signal_spectral_sum,
+    float signal_energy) {
+  // Compute spectral flatness on input spectrum.
+  UpdateSpectralFlatness(signal_spectrum, signal_spectral_sum,
+                         &features_.spectral_flatness);
+
+  // Compute difference of input spectrum with learned/estimated noise spectrum.
+  float spectral_diff =
+      ComputeSpectralDiff(conservative_noise_spectrum, signal_spectrum,
+                          signal_spectral_sum, diff_normalization_);
+  // Compute time-avg update of difference feature.
+  features_.spectral_diff += 0.3f * (spectral_diff - features_.spectral_diff);
+
+  signal_energy_sum_ += signal_energy;
+
+  // Compute histograms for parameter decisions (thresholds and weights for
+  // features). Parameters are extracted periodically.
+  if (--histogram_analysis_counter_ > 0) {
+    histograms_.Update(features_);
+  } else {
+    // Compute model parameters.
+    prior_model_estimator_.Update(histograms_);
+
+    // Clear histograms for next update.
+    histograms_.Clear();
+
+    histogram_analysis_counter_ = kFeatureUpdateWindowSize;
+
+    // Update every window:
+    // Compute normalization for the spectral difference for next estimation.
+    signal_energy_sum_ = signal_energy_sum_ / kFeatureUpdateWindowSize;
+    diff_normalization_ = 0.5f * (signal_energy_sum_ + diff_normalization_);
+    signal_energy_sum_ = 0.f;
+  }
+
+  // Compute the LRT.
+  UpdateSpectralLrt(prior_snr, post_snr, features_.avg_log_lrt, &features_.lrt);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/signal_model_estimator.h b/third_party/libwebrtc/modules/audio_processing/ns/signal_model_estimator.h
new file mode 100644
index 0000000000..58ce00acbf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/signal_model_estimator.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_SIGNAL_MODEL_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_NS_SIGNAL_MODEL_ESTIMATOR_H_
+
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/ns/histograms.h"
+#include "modules/audio_processing/ns/ns_common.h"
+#include "modules/audio_processing/ns/prior_signal_model.h"
+#include "modules/audio_processing/ns/prior_signal_model_estimator.h"
+#include "modules/audio_processing/ns/signal_model.h"
+
+namespace webrtc {
+
+class SignalModelEstimator {
+ public:
+  SignalModelEstimator();
+  SignalModelEstimator(const SignalModelEstimator&) = delete;
+  SignalModelEstimator& operator=(const SignalModelEstimator&) = delete;
+
+  // Compute signal normalization during the initial startup phase.
+  void AdjustNormalization(int32_t num_analyzed_frames, float signal_energy);
+
+  void Update(
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> post_snr,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+      float signal_spectral_sum,
+      float signal_energy);
+
+  const PriorSignalModel& get_prior_model() const {
+    return prior_model_estimator_.get_prior_model();
+  }
+  const SignalModel& get_model() { return features_; }
+
+ private:
+  float diff_normalization_ = 0.f;
+  float signal_energy_sum_ = 0.f;
+  Histograms histograms_;
+  int histogram_analysis_counter_ = 500;
+  PriorSignalModelEstimator prior_model_estimator_;
+  SignalModel features_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_SIGNAL_MODEL_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/speech_probability_estimator.cc b/third_party/libwebrtc/modules/audio_processing/ns/speech_probability_estimator.cc
new file mode 100644
index 0000000000..fce9bc8e07
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/speech_probability_estimator.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/speech_probability_estimator.h"
+
+#include <math.h>
+#include <algorithm>
+
+#include "modules/audio_processing/ns/fast_math.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+SpeechProbabilityEstimator::SpeechProbabilityEstimator() {
+  speech_probability_.fill(0.f);
+}
+
+void SpeechProbabilityEstimator::Update(
+    int32_t num_analyzed_frames,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> post_snr,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+    float signal_spectral_sum,
+    float signal_energy) {
+  // Update models.
+  if (num_analyzed_frames < kLongStartupPhaseBlocks) {
+    signal_model_estimator_.AdjustNormalization(num_analyzed_frames,
+                                                signal_energy);
+  }
+  signal_model_estimator_.Update(prior_snr, post_snr,
+                                 conservative_noise_spectrum, signal_spectrum,
+                                 signal_spectral_sum, signal_energy);
+
+  const SignalModel& model = signal_model_estimator_.get_model();
+  const PriorSignalModel& prior_model =
+      signal_model_estimator_.get_prior_model();
+
+  // Width parameter in sigmoid map for prior model.
+  constexpr float kWidthPrior0 = 4.f;
+  // Width for pause region: lower range, so increase width in tanh map.
+  constexpr float kWidthPrior1 = 2.f * kWidthPrior0;
+
+  // Average LRT feature: use larger width in tanh map for pause regions.
+  float width_prior = model.lrt < prior_model.lrt ? kWidthPrior1 : kWidthPrior0;
+
+  // Compute indicator function: sigmoid map.
+  float indicator0 =
+      0.5f * (tanh(width_prior * (model.lrt - prior_model.lrt)) + 1.f);
+
+  // Spectral flatness feature: use larger width in tanh map for pause regions.
+  width_prior = model.spectral_flatness > prior_model.flatness_threshold
+                    ? kWidthPrior1
+                    : kWidthPrior0;
+
+  // Compute indicator function: sigmoid map.
+  float indicator1 =
+      0.5f * (tanh(1.f * width_prior *
+                   (prior_model.flatness_threshold - model.spectral_flatness)) +
+              1.f);
+
+  // For template spectrum-difference : use larger width in tanh map for pause
+  // regions.
+  width_prior = model.spectral_diff < prior_model.template_diff_threshold
+                    ? kWidthPrior1
+                    : kWidthPrior0;
+
+  // Compute indicator function: sigmoid map.
+  float indicator2 =
+      0.5f * (tanh(width_prior * (model.spectral_diff -
+                                  prior_model.template_diff_threshold)) +
+              1.f);
+
+  // Combine the indicator function with the feature weights.
+  float ind_prior = prior_model.lrt_weighting * indicator0 +
+                    prior_model.flatness_weighting * indicator1 +
+                    prior_model.difference_weighting * indicator2;
+
+  // Compute the prior probability.
+  prior_speech_prob_ += 0.1f * (ind_prior - prior_speech_prob_);
+
+  // Make sure probabilities are within range: keep floor to 0.01.
+  prior_speech_prob_ = std::max(std::min(prior_speech_prob_, 1.f), 0.01f);
+
+  // Final speech probability: combine prior model with LR factor:.
+  float gain_prior =
+      (1.f - prior_speech_prob_) / (prior_speech_prob_ + 0.0001f);
+
+  std::array<float, kFftSizeBy2Plus1> inv_lrt;
+  ExpApproximationSignFlip(model.avg_log_lrt, inv_lrt);
+  for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+    speech_probability_[i] = 1.f / (1.f + gain_prior * inv_lrt[i]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/speech_probability_estimator.h b/third_party/libwebrtc/modules/audio_processing/ns/speech_probability_estimator.h
new file mode 100644
index 0000000000..259c3b6776
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/speech_probability_estimator.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_SPEECH_PROBABILITY_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_NS_SPEECH_PROBABILITY_ESTIMATOR_H_
+
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/ns/ns_common.h"
+#include "modules/audio_processing/ns/signal_model_estimator.h"
+
+namespace webrtc {
+
+// Class for estimating the probability of speech.
+class SpeechProbabilityEstimator {
+ public:
+  SpeechProbabilityEstimator();
+  SpeechProbabilityEstimator(const SpeechProbabilityEstimator&) = delete;
+  SpeechProbabilityEstimator& operator=(const SpeechProbabilityEstimator&) =
+      delete;
+
+  // Compute speech probability.
+  void Update(
+      int32_t num_analyzed_frames,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> post_snr,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
+      float signal_spectral_sum,
+      float signal_energy);
+
+  float get_prior_probability() const { return prior_speech_prob_; }
+  rtc::ArrayView<const float> get_probability() { return speech_probability_; }
+
+ private:
+  SignalModelEstimator signal_model_estimator_;
+  float prior_speech_prob_ = .5f;
+  std::array<float, kFftSizeBy2Plus1> speech_probability_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_SPEECH_PROBABILITY_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/suppression_params.cc b/third_party/libwebrtc/modules/audio_processing/ns/suppression_params.cc
new file mode 100644
index 0000000000..7bf18346f9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/suppression_params.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/suppression_params.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+SuppressionParams::SuppressionParams(
+    NsConfig::SuppressionLevel suppression_level) {
+  switch (suppression_level) {
+    case NsConfig::SuppressionLevel::k6dB:
+      over_subtraction_factor = 1.f;
+      // 6 dB attenuation.
+      minimum_attenuating_gain = 0.5f;
+      use_attenuation_adjustment = false;
+      break;
+    case NsConfig::SuppressionLevel::k12dB:
+      over_subtraction_factor = 1.f;
+      // 12 dB attenuation.
+      minimum_attenuating_gain = 0.25f;
+      use_attenuation_adjustment = true;
+      break;
+    case NsConfig::SuppressionLevel::k18dB:
+      over_subtraction_factor = 1.1f;
+      // 18 dB attenuation.
+      minimum_attenuating_gain = 0.125f;
+      use_attenuation_adjustment = true;
+      break;
+    case NsConfig::SuppressionLevel::k21dB:
+      over_subtraction_factor = 1.25f;
+      // 20.9 dB attenuation.
+      minimum_attenuating_gain = 0.09f;
+      use_attenuation_adjustment = true;
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/suppression_params.h b/third_party/libwebrtc/modules/audio_processing/ns/suppression_params.h
new file mode 100644
index 0000000000..ad11977d81
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/suppression_params.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_SUPPRESSION_PARAMS_H_
+#define MODULES_AUDIO_PROCESSING_NS_SUPPRESSION_PARAMS_H_
+
+#include "modules/audio_processing/ns/ns_config.h"
+
+namespace webrtc {
+
+struct SuppressionParams {
+  explicit SuppressionParams(NsConfig::SuppressionLevel suppression_level);
+  SuppressionParams(const SuppressionParams&) = delete;
+  SuppressionParams& operator=(const SuppressionParams&) = delete;
+
+  float over_subtraction_factor;
+  float minimum_attenuating_gain;
+  bool use_attenuation_adjustment;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_SUPPRESSION_PARAMS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/wiener_filter.cc b/third_party/libwebrtc/modules/audio_processing/ns/wiener_filter.cc
new file mode 100644
index 0000000000..e14b7970d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/wiener_filter.cc
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/ns/wiener_filter.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include <algorithm>
+
+#include "modules/audio_processing/ns/fast_math.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+WienerFilter::WienerFilter(const SuppressionParams& suppression_params)
+    : suppression_params_(suppression_params) {
+  filter_.fill(1.f);
+  initial_spectral_estimate_.fill(0.f);
+  spectrum_prev_process_.fill(0.f);
+}
+
+void WienerFilter::Update(
+    int32_t num_analyzed_frames,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> noise_spectrum,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> prev_noise_spectrum,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> parametric_noise_spectrum,
+    rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum) {
+  for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+    // Previous estimate based on previous frame with gain filter.
+    float prev_tsa = spectrum_prev_process_[i] /
+                     (prev_noise_spectrum[i] + 0.0001f) * filter_[i];
+
+    // Current estimate.
+    float current_tsa;
+    if (signal_spectrum[i] > noise_spectrum[i]) {
+      current_tsa = signal_spectrum[i] / (noise_spectrum[i] + 0.0001f) - 1.f;
+    } else {
+      current_tsa = 0.f;
+    }
+
+    // Directed decision estimate is sum of two terms: current estimate and
+    // previous estimate.
+    float snr_prior = 0.98f * prev_tsa + (1.f - 0.98f) * current_tsa;
+    filter_[i] =
+        snr_prior / (suppression_params_.over_subtraction_factor + snr_prior);
+    filter_[i] = std::max(std::min(filter_[i], 1.f),
+                          suppression_params_.minimum_attenuating_gain);
+  }
+
+  if (num_analyzed_frames < kShortStartupPhaseBlocks) {
+    for (size_t i = 0; i < kFftSizeBy2Plus1; ++i) {
+      initial_spectral_estimate_[i] += signal_spectrum[i];
+      float filter_initial = initial_spectral_estimate_[i] -
+                             suppression_params_.over_subtraction_factor *
+                                 parametric_noise_spectrum[i];
+      filter_initial /= initial_spectral_estimate_[i] + 0.0001f;
+
+      filter_initial = std::max(std::min(filter_initial, 1.f),
+                                suppression_params_.minimum_attenuating_gain);
+
+      // Weight the two suppression filters.
+      constexpr float kOnyByShortStartupPhaseBlocks =
+          1.f / kShortStartupPhaseBlocks;
+      filter_initial *= kShortStartupPhaseBlocks - num_analyzed_frames;
+      filter_[i] *= num_analyzed_frames;
+      filter_[i] += filter_initial;
+      filter_[i] *= kOnyByShortStartupPhaseBlocks;
+    }
+  }
+
+  std::copy(signal_spectrum.begin(), signal_spectrum.end(),
+            spectrum_prev_process_.begin());
+}
+
+float WienerFilter::ComputeOverallScalingFactor(
+    int32_t num_analyzed_frames,
+    float prior_speech_probability,
+    float energy_before_filtering,
+    float energy_after_filtering) const {
+  if (!suppression_params_.use_attenuation_adjustment ||
+      num_analyzed_frames <= kLongStartupPhaseBlocks) {
+    return 1.f;
+  }
+
+  float gain = SqrtFastApproximation(energy_after_filtering /
+                                     (energy_before_filtering + 1.f));
+
+  // Scaling for new version. Threshold in final energy gain factor calculation.
+  constexpr float kBLim = 0.5f;
+  float scale_factor1 = 1.f;
+  if (gain > kBLim) {
+    scale_factor1 = 1.f + 1.3f * (gain - kBLim);
+    if (gain * scale_factor1 > 1.f) {
+      scale_factor1 = 1.f / gain;
+    }
+  }
+
+  float scale_factor2 = 1.f;
+  if (gain < kBLim) {
+    // Do not reduce scale too much for pause regions: attenuation here should
+    // be controlled by flooring.
+    gain = std::max(gain, suppression_params_.minimum_attenuating_gain);
+    scale_factor2 = 1.f - 0.3f * (kBLim - gain);
+  }
+
+  // Combine both scales with speech/noise prob: note prior
+  // (prior_speech_probability) is not frequency dependent.
+  return prior_speech_probability * scale_factor1 +
+         (1.f - prior_speech_probability) * scale_factor2;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/ns/wiener_filter.h b/third_party/libwebrtc/modules/audio_processing/ns/wiener_filter.h
new file mode 100644
index 0000000000..b55c5dc59d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/ns/wiener_filter.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_NS_WIENER_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_NS_WIENER_FILTER_H_
+
+#include <array>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/ns/ns_common.h"
+#include "modules/audio_processing/ns/suppression_params.h"
+
+namespace webrtc {
+
+// Estimates a Wiener-filter based frequency domain noise reduction filter.
+class WienerFilter {
+ public:
+  explicit WienerFilter(const SuppressionParams& suppression_params);
+  WienerFilter(const WienerFilter&) = delete;
+  WienerFilter& operator=(const WienerFilter&) = delete;
+
+  // Updates the filter estimate.
+  void Update(
+      int32_t num_analyzed_frames,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> noise_spectrum,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> prev_noise_spectrum,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> parametric_noise_spectrum,
+      rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum);
+
+  // Compute an overall gain scaling factor.
+  float ComputeOverallScalingFactor(int32_t num_analyzed_frames,
+                                    float prior_speech_probability,
+                                    float energy_before_filtering,
+                                    float energy_after_filtering) const;
+
+  // Returns the filter.
+  rtc::ArrayView<const float, kFftSizeBy2Plus1> get_filter() const {
+    return filter_;
+  }
+
+ private:
+  const SuppressionParams& suppression_params_;
+  std::array<float, kFftSizeBy2Plus1> spectrum_prev_process_;
+  std::array<float, kFftSizeBy2Plus1> initial_spectral_estimate_;
+  std::array<float, kFftSizeBy2Plus1> filter_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_NS_WIENER_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators.cc b/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators.cc
new file mode 100644
index 0000000000..cea5c837dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators.cc
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/optionally_built_submodule_creators.h"
+
+#include <memory>
+
+#include "modules/audio_processing/transient/transient_suppressor_impl.h"
+
+namespace webrtc {
+
+std::unique_ptr<TransientSuppressor> CreateTransientSuppressor(
+    const ApmSubmoduleCreationOverrides& overrides,
+    TransientSuppressor::VadMode vad_mode,
+    int sample_rate_hz,
+    int detection_rate_hz,
+    int num_channels) {
+#ifdef WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR
+  return nullptr;
+#else
+  if (overrides.transient_suppression) {
+    return nullptr;
+  }
+  return std::make_unique<TransientSuppressorImpl>(
+      vad_mode, sample_rate_hz, detection_rate_hz, num_channels);
+#endif
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators.h b/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators.h
new file mode 100644
index 0000000000..1be2743986
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_OPTIONALLY_BUILT_SUBMODULE_CREATORS_H_
+#define MODULES_AUDIO_PROCESSING_OPTIONALLY_BUILT_SUBMODULE_CREATORS_H_
+
+#include <memory>
+
+#include "modules/audio_processing/transient/transient_suppressor.h"
+
+namespace webrtc {
+
+// These overrides are only to be used for testing purposes.
+// Each flag emulates a preprocessor macro to exclude a submodule of APM from
+// the build, e.g. WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR. If the corresponding
+// flag `transient_suppression` is enabled, then the creators will return
+// nullptr instead of a submodule instance, as if the macro had been defined.
+struct ApmSubmoduleCreationOverrides {
+  bool transient_suppression = false;
+};
+
+// Creates a transient suppressor.
+// Will instead return nullptr if one of the following is true:
+// * WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR is defined
+// * The corresponding override in `overrides` is enabled.
+std::unique_ptr<TransientSuppressor> CreateTransientSuppressor(
+    const ApmSubmoduleCreationOverrides& overrides,
+    TransientSuppressor::VadMode vad_mode,
+    int sample_rate_hz,
+    int detection_rate_hz,
+    int num_channels);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_OPTIONALLY_BUILT_SUBMODULE_CREATORS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators_gn/moz.build
new file mode 100644
index 0000000000..3e4fa4ca51
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/optionally_built_submodule_creators.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("optionally_built_submodule_creators_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/render_queue_item_verifier.h b/third_party/libwebrtc/modules/audio_processing/render_queue_item_verifier.h
new file mode 100644
index 0000000000..b8aff4a107
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/render_queue_item_verifier.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H_
+#define MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H_
+
+#include <vector>
+
+namespace webrtc {
+
+// Functor to use when supplying a verifier function for the queue item
+// verifcation.
+template <typename T>
+class RenderQueueItemVerifier {
+ public:
+  explicit RenderQueueItemVerifier(size_t minimum_capacity)
+      : minimum_capacity_(minimum_capacity) {}
+
+  bool operator()(const std::vector<T>& v) const {
+    return v.capacity() >= minimum_capacity_;
+  }
+
+ private:
+  size_t minimum_capacity_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_RENDER_QUEUE_ITEM_VERIFIER_H__
diff --git a/third_party/libwebrtc/modules/audio_processing/residual_echo_detector.cc b/third_party/libwebrtc/modules/audio_processing/residual_echo_detector.cc
new file mode 100644
index 0000000000..2a564fc233
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/residual_echo_detector.cc
@@ -0,0 +1,205 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/residual_echo_detector.h"
+
+#include <algorithm>
+#include <numeric>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace {
+
+float Power(rtc::ArrayView<const float> input) {
+  if (input.empty()) {
+    return 0.f;
+  }
+  return std::inner_product(input.begin(), input.end(), input.begin(), 0.f) /
+         input.size();
+}
+
+constexpr size_t kLookbackFrames = 650;
+// TODO(ivoc): Verify the size of this buffer.
+constexpr size_t kRenderBufferSize = 30;
+constexpr float kAlpha = 0.001f;
+// 10 seconds of data, updated every 10 ms.
+constexpr size_t kAggregationBufferSize = 10 * 100;
+
+}  // namespace
+
+namespace webrtc {
+
+std::atomic<int> ResidualEchoDetector::instance_count_(0);
+
+ResidualEchoDetector::ResidualEchoDetector()
+    : data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
+      render_buffer_(kRenderBufferSize),
+      render_power_(kLookbackFrames),
+      render_power_mean_(kLookbackFrames),
+      render_power_std_dev_(kLookbackFrames),
+      covariances_(kLookbackFrames),
+      recent_likelihood_max_(kAggregationBufferSize) {}
+
+ResidualEchoDetector::~ResidualEchoDetector() = default;
+
+void ResidualEchoDetector::AnalyzeRenderAudio(
+    rtc::ArrayView<const float> render_audio) {
+  // Dump debug data assuming 48 kHz sample rate (if this assumption is not
+  // valid the dumped audio will need to be converted offline accordingly).
+  data_dumper_->DumpWav("ed_render", render_audio.size(), render_audio.data(),
+                        48000, 1);
+
+  if (render_buffer_.Size() == 0) {
+    frames_since_zero_buffer_size_ = 0;
+  } else if (frames_since_zero_buffer_size_ >= kRenderBufferSize) {
+    // This can happen in a few cases: at the start of a call, due to a glitch
+    // or due to clock drift. The excess capture value will be ignored.
+    // TODO(ivoc): Include how often this happens in APM stats.
+    render_buffer_.Pop();
+    frames_since_zero_buffer_size_ = 0;
+  }
+  ++frames_since_zero_buffer_size_;
+  float power = Power(render_audio);
+  render_buffer_.Push(power);
+}
+
+void ResidualEchoDetector::AnalyzeCaptureAudio(
+    rtc::ArrayView<const float> capture_audio) {
+  // Dump debug data assuming 48 kHz sample rate (if this assumption is not
+  // valid the dumped audio will need to be converted offline accordingly).
+  data_dumper_->DumpWav("ed_capture", capture_audio.size(),
+                        capture_audio.data(), 48000, 1);
+
+  if (first_process_call_) {
+    // On the first process call (so the start of a call), we must flush the
+    // render buffer, otherwise the render data will be delayed.
+    render_buffer_.Clear();
+    first_process_call_ = false;
+  }
+
+  // Get the next render value.
+  const absl::optional<float> buffered_render_power = render_buffer_.Pop();
+  if (!buffered_render_power) {
+    // This can happen in a few cases: at the start of a call, due to a glitch
+    // or due to clock drift. The excess capture value will be ignored.
+    // TODO(ivoc): Include how often this happens in APM stats.
+    return;
+  }
+  // Update the render statistics, and store the statistics in circular buffers.
+  render_statistics_.Update(*buffered_render_power);
+  RTC_DCHECK_LT(next_insertion_index_, kLookbackFrames);
+  render_power_[next_insertion_index_] = *buffered_render_power;
+  render_power_mean_[next_insertion_index_] = render_statistics_.mean();
+  render_power_std_dev_[next_insertion_index_] =
+      render_statistics_.std_deviation();
+
+  // Get the next capture value, update capture statistics and add the relevant
+  // values to the buffers.
+  const float capture_power = Power(capture_audio);
+  capture_statistics_.Update(capture_power);
+  const float capture_mean = capture_statistics_.mean();
+  const float capture_std_deviation = capture_statistics_.std_deviation();
+
+  // Update the covariance values and determine the new echo likelihood.
+  echo_likelihood_ = 0.f;
+  size_t read_index = next_insertion_index_;
+
+  int best_delay = -1;
+  for (size_t delay = 0; delay < covariances_.size(); ++delay) {
+    RTC_DCHECK_LT(read_index, render_power_.size());
+    covariances_[delay].Update(capture_power, capture_mean,
+                               capture_std_deviation, render_power_[read_index],
+                               render_power_mean_[read_index],
+                               render_power_std_dev_[read_index]);
+    read_index = read_index > 0 ? read_index - 1 : kLookbackFrames - 1;
+
+    if (covariances_[delay].normalized_cross_correlation() > echo_likelihood_) {
+      echo_likelihood_ = covariances_[delay].normalized_cross_correlation();
+      best_delay = static_cast<int>(delay);
+    }
+  }
+  // This is a temporary log message to help find the underlying cause for echo
+  // likelihoods > 1.0.
+  // TODO(ivoc): Remove once the issue is resolved.
+  if (echo_likelihood_ > 1.1f) {
+    // Make sure we don't spam the log.
+    if (log_counter_ < 5 && best_delay != -1) {
+      size_t read_index = kLookbackFrames + next_insertion_index_ - best_delay;
+      if (read_index >= kLookbackFrames) {
+        read_index -= kLookbackFrames;
+      }
+      RTC_DCHECK_LT(read_index, render_power_.size());
+      RTC_LOG_F(LS_ERROR) << "Echo detector internal state: {"
+                             "Echo likelihood: "
+                          << echo_likelihood_ << ", Best Delay: " << best_delay
+                          << ", Covariance: "
+                          << covariances_[best_delay].covariance()
+                          << ", Last capture power: " << capture_power
+                          << ", Capture mean: " << capture_mean
+                          << ", Capture_standard deviation: "
+                          << capture_std_deviation << ", Last render power: "
+                          << render_power_[read_index]
+                          << ", Render mean: " << render_power_mean_[read_index]
+                          << ", Render standard deviation: "
+                          << render_power_std_dev_[read_index]
+                          << ", Reliability: " << reliability_ << "}";
+      log_counter_++;
+    }
+  }
+  RTC_DCHECK_LT(echo_likelihood_, 1.1f);
+
+  reliability_ = (1.0f - kAlpha) * reliability_ + kAlpha * 1.0f;
+  echo_likelihood_ *= reliability_;
+  // This is a temporary fix to prevent echo likelihood values > 1.0.
+  // TODO(ivoc): Find the root cause of this issue and fix it.
+  echo_likelihood_ = std::min(echo_likelihood_, 1.0f);
+  int echo_percentage = static_cast<int>(echo_likelihood_ * 100);
+  RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ResidualEchoDetector.EchoLikelihood",
+                       echo_percentage, 0, 100, 100 /* number of bins */);
+
+  // Update the buffer of recent likelihood values.
+  recent_likelihood_max_.Update(echo_likelihood_);
+
+  // Update the next insertion index.
+  next_insertion_index_ = next_insertion_index_ < (kLookbackFrames - 1)
+                              ? next_insertion_index_ + 1
+                              : 0;
+}
+
+void ResidualEchoDetector::Initialize(int /*capture_sample_rate_hz*/,
+                                      int /*num_capture_channels*/,
+                                      int /*render_sample_rate_hz*/,
+                                      int /*num_render_channels*/) {
+  render_buffer_.Clear();
+  std::fill(render_power_.begin(), render_power_.end(), 0.f);
+  std::fill(render_power_mean_.begin(), render_power_mean_.end(), 0.f);
+  std::fill(render_power_std_dev_.begin(), render_power_std_dev_.end(), 0.f);
+  render_statistics_.Clear();
+  capture_statistics_.Clear();
+  recent_likelihood_max_.Clear();
+  for (auto& cov : covariances_) {
+    cov.Clear();
+  }
+  echo_likelihood_ = 0.f;
+  next_insertion_index_ = 0;
+  reliability_ = 0.f;
+}
+
+EchoDetector::Metrics ResidualEchoDetector::GetMetrics() const {
+  EchoDetector::Metrics metrics;
+  metrics.echo_likelihood = echo_likelihood_;
+  metrics.echo_likelihood_recent_max = recent_likelihood_max_.max();
+  return metrics;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/residual_echo_detector.h b/third_party/libwebrtc/modules/audio_processing/residual_echo_detector.h
new file mode 100644
index 0000000000..ac554b17c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/residual_echo_detector.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
+#define MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
+
+#include <atomic>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/echo_detector/circular_buffer.h"
+#include "modules/audio_processing/echo_detector/mean_variance_estimator.h"
+#include "modules/audio_processing/echo_detector/moving_max.h"
+#include "modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioBuffer;
+
+class ResidualEchoDetector : public EchoDetector {
+ public:
+  ResidualEchoDetector();
+  ~ResidualEchoDetector() override;
+
+  // This function should be called while holding the render lock.
+  void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) override;
+
+  // This function should be called while holding the capture lock.
+  void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override;
+
+  // This function should be called while holding the capture lock.
+  void Initialize(int capture_sample_rate_hz,
+                  int num_capture_channels,
+                  int render_sample_rate_hz,
+                  int num_render_channels) override;
+
+  // This function is for testing purposes only.
+  void SetReliabilityForTest(float value) { reliability_ = value; }
+
+  // This function should be called while holding the capture lock.
+  EchoDetector::Metrics GetMetrics() const override;
+
+ private:
+  static std::atomic<int> instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  // Keep track if the `Process` function has been previously called.
+  bool first_process_call_ = true;
+  // Buffer for storing the power of incoming farend buffers. This is needed for
+  // cases where calls to BufferFarend and Process are jittery.
+  CircularBuffer render_buffer_;
+  // Count how long ago it was that the size of `render_buffer_` was zero. This
+  // value is also reset to zero when clock drift is detected and a value from
+  // the renderbuffer is discarded, even though the buffer is not actually zero
+  // at that point. This is done to avoid repeatedly removing elements in this
+  // situation.
+  size_t frames_since_zero_buffer_size_ = 0;
+
+  // Circular buffers containing delayed versions of the power, mean and
+  // standard deviation, for calculating the delayed covariance values.
+  std::vector<float> render_power_;
+  std::vector<float> render_power_mean_;
+  std::vector<float> render_power_std_dev_;
+  // Covariance estimates for different delay values.
+  std::vector<NormalizedCovarianceEstimator> covariances_;
+  // Index where next element should be inserted in all of the above circular
+  // buffers.
+  size_t next_insertion_index_ = 0;
+
+  MeanVarianceEstimator render_statistics_;
+  MeanVarianceEstimator capture_statistics_;
+  // Current echo likelihood.
+  float echo_likelihood_ = 0.f;
+  // Reliability of the current likelihood.
+  float reliability_ = 0.f;
+  MovingMax recent_likelihood_max_;
+
+  int log_counter_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/residual_echo_detector_unittest.cc b/third_party/libwebrtc/modules/audio_processing/residual_echo_detector_unittest.cc
new file mode 100644
index 0000000000..d8c227a443
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/residual_echo_detector_unittest.cc
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/residual_echo_detector.h"
+
+#include <vector>
+
+#include "api/make_ref_counted.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(ResidualEchoDetectorTests, Echo) {
+  auto echo_detector = rtc::make_ref_counted<ResidualEchoDetector>();
+  echo_detector->SetReliabilityForTest(1.0f);
+  std::vector<float> ones(160, 1.f);
+  std::vector<float> zeros(160, 0.f);
+
+  // In this test the capture signal has a delay of 10 frames w.r.t. the render
+  // signal, but is otherwise identical. Both signals are periodic with a 20
+  // frame interval.
+  for (int i = 0; i < 1000; i++) {
+    if (i % 20 == 0) {
+      echo_detector->AnalyzeRenderAudio(ones);
+      echo_detector->AnalyzeCaptureAudio(zeros);
+    } else if (i % 20 == 10) {
+      echo_detector->AnalyzeRenderAudio(zeros);
+      echo_detector->AnalyzeCaptureAudio(ones);
+    } else {
+      echo_detector->AnalyzeRenderAudio(zeros);
+      echo_detector->AnalyzeCaptureAudio(zeros);
+    }
+  }
+  // We expect to detect echo with near certain likelihood.
+  auto ed_metrics = echo_detector->GetMetrics();
+  ASSERT_TRUE(ed_metrics.echo_likelihood);
+  EXPECT_NEAR(1.f, ed_metrics.echo_likelihood.value(), 0.01f);
+}
+
+TEST(ResidualEchoDetectorTests, NoEcho) {
+  auto echo_detector = rtc::make_ref_counted<ResidualEchoDetector>();
+  echo_detector->SetReliabilityForTest(1.0f);
+  std::vector<float> ones(160, 1.f);
+  std::vector<float> zeros(160, 0.f);
+
+  // In this test the capture signal is always zero, so no echo should be
+  // detected.
+  for (int i = 0; i < 1000; i++) {
+    if (i % 20 == 0) {
+      echo_detector->AnalyzeRenderAudio(ones);
+    } else {
+      echo_detector->AnalyzeRenderAudio(zeros);
+    }
+    echo_detector->AnalyzeCaptureAudio(zeros);
+  }
+  // We expect to not detect any echo.
+  auto ed_metrics = echo_detector->GetMetrics();
+  ASSERT_TRUE(ed_metrics.echo_likelihood);
+  EXPECT_NEAR(0.f, ed_metrics.echo_likelihood.value(), 0.01f);
+}
+
+TEST(ResidualEchoDetectorTests, EchoWithRenderClockDrift) {
+  auto echo_detector = rtc::make_ref_counted<ResidualEchoDetector>();
+  echo_detector->SetReliabilityForTest(1.0f);
+  std::vector<float> ones(160, 1.f);
+  std::vector<float> zeros(160, 0.f);
+
+  // In this test the capture signal has a delay of 10 frames w.r.t. the render
+  // signal, but is otherwise identical. Both signals are periodic with a 20
+  // frame interval. There is a simulated clock drift of 1% in this test, with
+  // the render side producing data slightly faster.
+  for (int i = 0; i < 1000; i++) {
+    if (i % 20 == 0) {
+      echo_detector->AnalyzeRenderAudio(ones);
+      echo_detector->AnalyzeCaptureAudio(zeros);
+    } else if (i % 20 == 10) {
+      echo_detector->AnalyzeRenderAudio(zeros);
+      echo_detector->AnalyzeCaptureAudio(ones);
+    } else {
+      echo_detector->AnalyzeRenderAudio(zeros);
+      echo_detector->AnalyzeCaptureAudio(zeros);
+    }
+    if (i % 100 == 0) {
+      // This is causing the simulated clock drift.
+      echo_detector->AnalyzeRenderAudio(zeros);
+    }
+  }
+  // We expect to detect echo with high likelihood. Clock drift is harder to
+  // correct on the render side than on the capture side. This is due to the
+  // render buffer, clock drift can only be discovered after a certain delay.
+  // A growing buffer can be caused by jitter or clock drift and it's not
+  // possible to make this decision right away. For this reason we only expect
+  // an echo likelihood of 75% in this test.
+  auto ed_metrics = echo_detector->GetMetrics();
+  ASSERT_TRUE(ed_metrics.echo_likelihood);
+  EXPECT_GT(ed_metrics.echo_likelihood.value(), 0.75f);
+}
+
+TEST(ResidualEchoDetectorTests, EchoWithCaptureClockDrift) {
+  auto echo_detector = rtc::make_ref_counted<ResidualEchoDetector>();
+  echo_detector->SetReliabilityForTest(1.0f);
+  std::vector<float> ones(160, 1.f);
+  std::vector<float> zeros(160, 0.f);
+
+  // In this test the capture signal has a delay of 10 frames w.r.t. the render
+  // signal, but is otherwise identical. Both signals are periodic with a 20
+  // frame interval. There is a simulated clock drift of 1% in this test, with
+  // the capture side producing data slightly faster.
+  for (int i = 0; i < 1000; i++) {
+    if (i % 20 == 0) {
+      echo_detector->AnalyzeRenderAudio(ones);
+      echo_detector->AnalyzeCaptureAudio(zeros);
+    } else if (i % 20 == 10) {
+      echo_detector->AnalyzeRenderAudio(zeros);
+      echo_detector->AnalyzeCaptureAudio(ones);
+    } else {
+      echo_detector->AnalyzeRenderAudio(zeros);
+      echo_detector->AnalyzeCaptureAudio(zeros);
+    }
+    if (i % 100 == 0) {
+      // This is causing the simulated clock drift.
+      echo_detector->AnalyzeCaptureAudio(zeros);
+    }
+  }
+  // We expect to detect echo with near certain likelihood.
+  auto ed_metrics = echo_detector->GetMetrics();
+  ASSERT_TRUE(ed_metrics.echo_likelihood);
+  EXPECT_NEAR(1.f, ed_metrics.echo_likelihood.value(), 0.01f);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/rms_level.cc b/third_party/libwebrtc/modules/audio_processing/rms_level.cc
new file mode 100644
index 0000000000..b0a45cb403
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/rms_level.cc
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/rms_level.h"
+
+#include <algorithm>
+#include <cmath>
+#include <numeric>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+static constexpr float kMaxSquaredLevel = 32768 * 32768;
+// kMinLevel is the level corresponding to kMinLevelDb, that is 10^(-127/10).
+static constexpr float kMinLevel = 1.995262314968883e-13f;
+
+// Calculates the normalized RMS value from a mean square value. The input
+// should be the sum of squared samples divided by the number of samples. The
+// value will be normalized to full range before computing the RMS, wich is
+// returned as a negated dBfs. That is, 0 is full amplitude while 127 is very
+// faint.
+int ComputeRms(float mean_square) {
+  if (mean_square <= kMinLevel * kMaxSquaredLevel) {
+    // Very faint; simply return the minimum value.
+    return RmsLevel::kMinLevelDb;
+  }
+  // Normalize by the max level.
+  const float mean_square_norm = mean_square / kMaxSquaredLevel;
+  RTC_DCHECK_GT(mean_square_norm, kMinLevel);
+  // 20log_10(x^0.5) = 10log_10(x)
+  const float rms = 10.f * std::log10(mean_square_norm);
+  RTC_DCHECK_LE(rms, 0.f);
+  RTC_DCHECK_GT(rms, -RmsLevel::kMinLevelDb);
+  // Return the negated value.
+  return static_cast<int>(-rms + 0.5f);
+}
+}  // namespace
+
+RmsLevel::RmsLevel() {
+  Reset();
+}
+
+RmsLevel::~RmsLevel() = default;
+
+void RmsLevel::Reset() {
+  sum_square_ = 0.f;
+  sample_count_ = 0;
+  max_sum_square_ = 0.f;
+  block_size_ = absl::nullopt;
+}
+
+void RmsLevel::Analyze(rtc::ArrayView<const int16_t> data) {
+  if (data.empty()) {
+    return;
+  }
+
+  CheckBlockSize(data.size());
+
+  const float sum_square =
+      std::accumulate(data.begin(), data.end(), 0.f,
+                      [](float a, int16_t b) { return a + b * b; });
+  RTC_DCHECK_GE(sum_square, 0.f);
+  sum_square_ += sum_square;
+  sample_count_ += data.size();
+
+  max_sum_square_ = std::max(max_sum_square_, sum_square);
+}
+
+void RmsLevel::Analyze(rtc::ArrayView<const float> data) {
+  if (data.empty()) {
+    return;
+  }
+
+  CheckBlockSize(data.size());
+
+  float sum_square = 0.f;
+
+  for (float data_k : data) {
+    int16_t tmp =
+        static_cast<int16_t>(std::min(std::max(data_k, -32768.f), 32767.f));
+    sum_square += tmp * tmp;
+  }
+  RTC_DCHECK_GE(sum_square, 0.f);
+  sum_square_ += sum_square;
+  sample_count_ += data.size();
+
+  max_sum_square_ = std::max(max_sum_square_, sum_square);
+}
+
+void RmsLevel::AnalyzeMuted(size_t length) {
+  CheckBlockSize(length);
+  sample_count_ += length;
+}
+
+int RmsLevel::Average() {
+  const bool have_samples = (sample_count_ != 0);
+  int rms = have_samples ? ComputeRms(sum_square_ / sample_count_)
+                         : RmsLevel::kMinLevelDb;
+
+  // To ensure that kMinLevelDb represents digital silence (muted audio
+  // sources) we'll check here if the sum_square is actually 0. If it's not
+  // we'll bump up the return value to `kInaudibleButNotMuted`.
+  // https://datatracker.ietf.org/doc/html/rfc6464
+  if (have_samples && rms == RmsLevel::kMinLevelDb && sum_square_ != 0.0f) {
+    rms = kInaudibleButNotMuted;
+  }
+
+  Reset();
+  return rms;
+}
+
+RmsLevel::Levels RmsLevel::AverageAndPeak() {
+  // Note that block_size_ should by design always be non-empty when
+  // sample_count_ != 0. Also, the * operator of absl::optional enforces this
+  // with a DCHECK.
+  Levels levels = (sample_count_ == 0)
+                      ? Levels{RmsLevel::kMinLevelDb, RmsLevel::kMinLevelDb}
+                      : Levels{ComputeRms(sum_square_ / sample_count_),
+                               ComputeRms(max_sum_square_ / *block_size_)};
+  Reset();
+  return levels;
+}
+
+void RmsLevel::CheckBlockSize(size_t block_size) {
+  if (block_size_ != block_size) {
+    Reset();
+    block_size_ = block_size;
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/rms_level.h b/third_party/libwebrtc/modules/audio_processing/rms_level.h
new file mode 100644
index 0000000000..fbece19ecd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/rms_level.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+#define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Computes the root mean square (RMS) level in dBFs (decibels from digital
+// full-scale) of audio data. The computation follows RFC 6465:
+// https://tools.ietf.org/html/rfc6465
+// with the intent that it can provide the RTP audio level indication.
+//
+// The expected approach is to provide constant-sized chunks of audio to
+// Analyze(). When enough chunks have been accumulated to form a packet, call
+// Average() to get the audio level indicator for the RTP header.
+class RmsLevel {
+ public:
+  struct Levels {
+    int average;
+    int peak;
+  };
+
+  enum : int { kMinLevelDb = 127, kInaudibleButNotMuted = 126 };
+
+  RmsLevel();
+  ~RmsLevel();
+
+  // Can be called to reset internal states, but is not required during normal
+  // operation.
+  void Reset();
+
+  // Pass each chunk of audio to Analyze() to accumulate the level.
+  void Analyze(rtc::ArrayView<const int16_t> data);
+  void Analyze(rtc::ArrayView<const float> data);
+
+  // If all samples with the given `length` have a magnitude of zero, this is
+  // a shortcut to avoid some computation.
+  void AnalyzeMuted(size_t length);
+
+  // Computes the RMS level over all data passed to Analyze() since the last
+  // call to Average(). The returned value is positive but should be interpreted
+  // as negative as per the RFC. It is constrained to [0, 127]. Resets the
+  // internal state to start a new measurement period.
+  int Average();
+
+  // Like Average() above, but also returns the RMS peak value. Resets the
+  // internal state to start a new measurement period.
+  Levels AverageAndPeak();
+
+ private:
+  // Compares `block_size` with `block_size_`. If they are different, calls
+  // Reset() and stores the new size.
+  void CheckBlockSize(size_t block_size);
+
+  float sum_square_;
+  size_t sample_count_;
+  float max_sum_square_;
+  absl::optional<size_t> block_size_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/rms_level_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/rms_level_gn/moz.build
new file mode 100644
index 0000000000..ec35734b6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/rms_level_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/rms_level.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rms_level_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/rms_level_unittest.cc b/third_party/libwebrtc/modules/audio_processing/rms_level_unittest.cc
new file mode 100644
index 0000000000..4cbad461e7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/rms_level_unittest.cc
@@ -0,0 +1,197 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#define _USE_MATH_DEFINES
+#include "modules/audio_processing/rms_level.h"
+
+#include <cmath>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+constexpr int kSampleRateHz = 48000;
+constexpr size_t kBlockSizeSamples = kSampleRateHz / 100;
+
+std::unique_ptr<RmsLevel> RunTest(rtc::ArrayView<const int16_t> input) {
+  std::unique_ptr<RmsLevel> level(new RmsLevel);
+  for (size_t n = 0; n + kBlockSizeSamples <= input.size();
+       n += kBlockSizeSamples) {
+    level->Analyze(input.subview(n, kBlockSizeSamples));
+  }
+  return level;
+}
+
+std::unique_ptr<RmsLevel> RunTest(rtc::ArrayView<const float> input) {
+  std::unique_ptr<RmsLevel> level(new RmsLevel);
+  for (size_t n = 0; n + kBlockSizeSamples <= input.size();
+       n += kBlockSizeSamples) {
+    level->Analyze(input.subview(n, kBlockSizeSamples));
+  }
+  return level;
+}
+
+std::vector<int16_t> CreateInt16Sinusoid(int frequency_hz,
+                                         int amplitude,
+                                         size_t num_samples) {
+  std::vector<int16_t> x(num_samples);
+  for (size_t n = 0; n < num_samples; ++n) {
+    x[n] = rtc::saturated_cast<int16_t>(
+        amplitude * std::sin(2 * M_PI * n * frequency_hz / kSampleRateHz));
+  }
+  return x;
+}
+
+std::vector<float> CreateFloatSinusoid(int frequency_hz,
+                                       int amplitude,
+                                       size_t num_samples) {
+  std::vector<int16_t> x16 =
+      CreateInt16Sinusoid(frequency_hz, amplitude, num_samples);
+  std::vector<float> x(x16.size());
+  for (size_t n = 0; n < x.size(); ++n) {
+    x[n] = x16[n];
+  }
+  return x;
+}
+
+}  // namespace
+
+TEST(RmsLevelTest, VerifyIndentityBetweenFloatAndFix) {
+  auto x_f = CreateFloatSinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto x_i = CreateFloatSinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level_f = RunTest(x_f);
+  auto level_i = RunTest(x_i);
+  int avg_i = level_i->Average();
+  int avg_f = level_f->Average();
+  EXPECT_EQ(3, avg_i);  // -3 dBFS
+  EXPECT_EQ(avg_f, avg_i);
+}
+
+TEST(RmsLevelTest, Run1000HzFullScale) {
+  auto x = CreateInt16Sinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  EXPECT_EQ(3, level->Average());  // -3 dBFS
+}
+
+TEST(RmsLevelTest, Run1000HzFullScaleAverageAndPeak) {
+  auto x = CreateInt16Sinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  auto stats = level->AverageAndPeak();
+  EXPECT_EQ(3, stats.average);  // -3 dBFS
+  EXPECT_EQ(3, stats.peak);
+}
+
+TEST(RmsLevelTest, Run1000HzHalfScale) {
+  auto x = CreateInt16Sinusoid(1000, INT16_MAX / 2, kSampleRateHz);
+  auto level = RunTest(x);
+  EXPECT_EQ(9, level->Average());  // -9 dBFS
+}
+
+TEST(RmsLevelTest, RunZeros) {
+  std::vector<int16_t> x(kSampleRateHz, 0);  // 1 second of pure silence.
+  auto level = RunTest(x);
+  EXPECT_EQ(127, level->Average());
+}
+
+TEST(RmsLevelTest, RunZerosAverageAndPeak) {
+  std::vector<int16_t> x(kSampleRateHz, 0);  // 1 second of pure silence.
+  auto level = RunTest(x);
+  auto stats = level->AverageAndPeak();
+  EXPECT_EQ(127, stats.average);
+  EXPECT_EQ(127, stats.peak);
+}
+
+TEST(RmsLevelTest, NoSamples) {
+  RmsLevel level;
+  EXPECT_EQ(127, level.Average());  // Return minimum if no samples are given.
+}
+
+TEST(RmsLevelTest, NoSamplesAverageAndPeak) {
+  RmsLevel level;
+  auto stats = level.AverageAndPeak();
+  EXPECT_EQ(127, stats.average);
+  EXPECT_EQ(127, stats.peak);
+}
+
+TEST(RmsLevelTest, PollTwice) {
+  auto x = CreateInt16Sinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  level->Average();
+  EXPECT_EQ(127, level->Average());  // Stats should be reset at this point.
+}
+
+TEST(RmsLevelTest, Reset) {
+  auto x = CreateInt16Sinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  level->Reset();
+  EXPECT_EQ(127, level->Average());  // Stats should be reset at this point.
+}
+
+// Inserts 1 second of full-scale sinusoid, followed by 1 second of muted.
+TEST(RmsLevelTest, ProcessMuted) {
+  auto x = CreateInt16Sinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  const size_t kBlocksPerSecond = rtc::CheckedDivExact(
+      static_cast<size_t>(kSampleRateHz), kBlockSizeSamples);
+  for (size_t i = 0; i < kBlocksPerSecond; ++i) {
+    level->AnalyzeMuted(kBlockSizeSamples);
+  }
+  EXPECT_EQ(6, level->Average());  // Average RMS halved due to the silence.
+}
+
+// Digital silence must yield 127 and anything else should yield 126 or lower.
+TEST(RmsLevelTest, OnlyDigitalSilenceIs127) {
+  std::vector<int16_t> test_buffer(kSampleRateHz, 0);
+  auto level = RunTest(test_buffer);
+  EXPECT_EQ(127, level->Average());
+  // Change one sample to something other than 0 to make the buffer not strictly
+  // represent digital silence.
+  test_buffer[0] = 1;
+  level = RunTest(test_buffer);
+  EXPECT_LT(level->Average(), 127);
+}
+
+// Inserts 1 second of half-scale sinusoid, follwed by 10 ms of full-scale, and
+// finally 1 second of half-scale again. Expect the average to be -9 dBFS due
+// to the vast majority of the signal being half-scale, and the peak to be
+// -3 dBFS.
+TEST(RmsLevelTest, RunHalfScaleAndInsertFullScale) {
+  auto half_scale = CreateInt16Sinusoid(1000, INT16_MAX / 2, kSampleRateHz);
+  auto full_scale = CreateInt16Sinusoid(1000, INT16_MAX, kSampleRateHz / 100);
+  auto x = half_scale;
+  x.insert(x.end(), full_scale.begin(), full_scale.end());
+  x.insert(x.end(), half_scale.begin(), half_scale.end());
+  ASSERT_EQ(static_cast<size_t>(2 * kSampleRateHz + kSampleRateHz / 100),
+            x.size());
+  auto level = RunTest(x);
+  auto stats = level->AverageAndPeak();
+  EXPECT_EQ(9, stats.average);
+  EXPECT_EQ(3, stats.peak);
+}
+
+TEST(RmsLevelTest, ResetOnBlockSizeChange) {
+  auto x = CreateInt16Sinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  // Create a new signal with half amplitude, but double block length.
+  auto y = CreateInt16Sinusoid(1000, INT16_MAX / 2, kBlockSizeSamples * 2);
+  level->Analyze(y);
+  auto stats = level->AverageAndPeak();
+  // Expect all stats to only be influenced by the last signal (y), since the
+  // changed block size should reset the stats.
+  EXPECT_EQ(9, stats.average);
+  EXPECT_EQ(9, stats.peak);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/splitting_filter.cc b/third_party/libwebrtc/modules/audio_processing/splitting_filter.cc
new file mode 100644
index 0000000000..d47090bc03
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/splitting_filter.cc
@@ -0,0 +1,144 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/splitting_filter.h"
+
+#include <array>
+
+#include "api/array_view.h"
+#include "common_audio/channel_buffer.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kSamplesPerBand = 160;
+constexpr size_t kTwoBandFilterSamplesPerFrame = 320;
+
+}  // namespace
+
+SplittingFilter::SplittingFilter(size_t num_channels,
+                                 size_t num_bands,
+                                 size_t num_frames)
+    : num_bands_(num_bands),
+      two_bands_states_(num_bands_ == 2 ? num_channels : 0),
+      three_band_filter_banks_(num_bands_ == 3 ? num_channels : 0) {
+  RTC_CHECK(num_bands_ == 2 || num_bands_ == 3);
+}
+
+SplittingFilter::~SplittingFilter() = default;
+
+void SplittingFilter::Analysis(const ChannelBuffer<float>* data,
+                               ChannelBuffer<float>* bands) {
+  RTC_DCHECK_EQ(num_bands_, bands->num_bands());
+  RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
+  RTC_DCHECK_EQ(data->num_frames(),
+                bands->num_frames_per_band() * bands->num_bands());
+  if (bands->num_bands() == 2) {
+    TwoBandsAnalysis(data, bands);
+  } else if (bands->num_bands() == 3) {
+    ThreeBandsAnalysis(data, bands);
+  }
+}
+
+void SplittingFilter::Synthesis(const ChannelBuffer<float>* bands,
+                                ChannelBuffer<float>* data) {
+  RTC_DCHECK_EQ(num_bands_, bands->num_bands());
+  RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
+  RTC_DCHECK_EQ(data->num_frames(),
+                bands->num_frames_per_band() * bands->num_bands());
+  if (bands->num_bands() == 2) {
+    TwoBandsSynthesis(bands, data);
+  } else if (bands->num_bands() == 3) {
+    ThreeBandsSynthesis(bands, data);
+  }
+}
+
+void SplittingFilter::TwoBandsAnalysis(const ChannelBuffer<float>* data,
+                                       ChannelBuffer<float>* bands) {
+  RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
+  RTC_DCHECK_EQ(data->num_frames(), kTwoBandFilterSamplesPerFrame);
+
+  for (size_t i = 0; i < two_bands_states_.size(); ++i) {
+    std::array<std::array<int16_t, kSamplesPerBand>, 2> bands16;
+    std::array<int16_t, kTwoBandFilterSamplesPerFrame> full_band16;
+    FloatS16ToS16(data->channels(0)[i], full_band16.size(), full_band16.data());
+    WebRtcSpl_AnalysisQMF(full_band16.data(), data->num_frames(),
+                          bands16[0].data(), bands16[1].data(),
+                          two_bands_states_[i].analysis_state1,
+                          two_bands_states_[i].analysis_state2);
+    S16ToFloatS16(bands16[0].data(), bands16[0].size(), bands->channels(0)[i]);
+    S16ToFloatS16(bands16[1].data(), bands16[1].size(), bands->channels(1)[i]);
+  }
+}
+
+void SplittingFilter::TwoBandsSynthesis(const ChannelBuffer<float>* bands,
+                                        ChannelBuffer<float>* data) {
+  RTC_DCHECK_LE(data->num_channels(), two_bands_states_.size());
+  RTC_DCHECK_EQ(data->num_frames(), kTwoBandFilterSamplesPerFrame);
+  for (size_t i = 0; i < data->num_channels(); ++i) {
+    std::array<std::array<int16_t, kSamplesPerBand>, 2> bands16;
+    std::array<int16_t, kTwoBandFilterSamplesPerFrame> full_band16;
+    FloatS16ToS16(bands->channels(0)[i], bands16[0].size(), bands16[0].data());
+    FloatS16ToS16(bands->channels(1)[i], bands16[1].size(), bands16[1].data());
+    WebRtcSpl_SynthesisQMF(bands16[0].data(), bands16[1].data(),
+                           bands->num_frames_per_band(), full_band16.data(),
+                           two_bands_states_[i].synthesis_state1,
+                           two_bands_states_[i].synthesis_state2);
+    S16ToFloatS16(full_band16.data(), full_band16.size(), data->channels(0)[i]);
+  }
+}
+
+void SplittingFilter::ThreeBandsAnalysis(const ChannelBuffer<float>* data,
+                                         ChannelBuffer<float>* bands) {
+  RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
+  RTC_DCHECK_LE(data->num_channels(), three_band_filter_banks_.size());
+  RTC_DCHECK_LE(data->num_channels(), bands->num_channels());
+  RTC_DCHECK_EQ(data->num_frames(), ThreeBandFilterBank::kFullBandSize);
+  RTC_DCHECK_EQ(bands->num_frames(), ThreeBandFilterBank::kFullBandSize);
+  RTC_DCHECK_EQ(bands->num_bands(), ThreeBandFilterBank::kNumBands);
+  RTC_DCHECK_EQ(bands->num_frames_per_band(),
+                ThreeBandFilterBank::kSplitBandSize);
+
+  for (size_t i = 0; i < three_band_filter_banks_.size(); ++i) {
+    three_band_filter_banks_[i].Analysis(
+        rtc::ArrayView<const float, ThreeBandFilterBank::kFullBandSize>(
+            data->channels_view()[i].data(),
+            ThreeBandFilterBank::kFullBandSize),
+        rtc::ArrayView<const rtc::ArrayView<float>,
+                       ThreeBandFilterBank::kNumBands>(
+            bands->bands_view(i).data(), ThreeBandFilterBank::kNumBands));
+  }
+}
+
+void SplittingFilter::ThreeBandsSynthesis(const ChannelBuffer<float>* bands,
+                                          ChannelBuffer<float>* data) {
+  RTC_DCHECK_LE(data->num_channels(), three_band_filter_banks_.size());
+  RTC_DCHECK_LE(data->num_channels(), bands->num_channels());
+  RTC_DCHECK_LE(data->num_channels(), three_band_filter_banks_.size());
+  RTC_DCHECK_EQ(data->num_frames(), ThreeBandFilterBank::kFullBandSize);
+  RTC_DCHECK_EQ(bands->num_frames(), ThreeBandFilterBank::kFullBandSize);
+  RTC_DCHECK_EQ(bands->num_bands(), ThreeBandFilterBank::kNumBands);
+  RTC_DCHECK_EQ(bands->num_frames_per_band(),
+                ThreeBandFilterBank::kSplitBandSize);
+
+  for (size_t i = 0; i < data->num_channels(); ++i) {
+    three_band_filter_banks_[i].Synthesis(
+        rtc::ArrayView<const rtc::ArrayView<float>,
+                       ThreeBandFilterBank::kNumBands>(
+            bands->bands_view(i).data(), ThreeBandFilterBank::kNumBands),
+        rtc::ArrayView<float, ThreeBandFilterBank::kFullBandSize>(
+            data->channels_view()[i].data(),
+            ThreeBandFilterBank::kFullBandSize));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/splitting_filter.h b/third_party/libwebrtc/modules/audio_processing/splitting_filter.h
new file mode 100644
index 0000000000..e578dd07c1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/splitting_filter.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_SPLITTING_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_SPLITTING_FILTER_H_
+
+#include <cstring>
+#include <memory>
+#include <vector>
+
+#include "common_audio/channel_buffer.h"
+#include "modules/audio_processing/three_band_filter_bank.h"
+
+namespace webrtc {
+
+struct TwoBandsStates {
+  TwoBandsStates() {
+    memset(analysis_state1, 0, sizeof(analysis_state1));
+    memset(analysis_state2, 0, sizeof(analysis_state2));
+    memset(synthesis_state1, 0, sizeof(synthesis_state1));
+    memset(synthesis_state2, 0, sizeof(synthesis_state2));
+  }
+
+  static const int kStateSize = 6;
+  int analysis_state1[kStateSize];
+  int analysis_state2[kStateSize];
+  int synthesis_state1[kStateSize];
+  int synthesis_state2[kStateSize];
+};
+
+// Splitting filter which is able to split into and merge from 2 or 3 frequency
+// bands. The number of channels needs to be provided at construction time.
+//
+// For each block, Analysis() is called to split into bands and then Synthesis()
+// to merge these bands again. The input and output signals are contained in
+// ChannelBuffers and for the different bands an array of ChannelBuffers is
+// used.
+class SplittingFilter {
+ public:
+  SplittingFilter(size_t num_channels, size_t num_bands, size_t num_frames);
+  ~SplittingFilter();
+
+  void Analysis(const ChannelBuffer<float>* data, ChannelBuffer<float>* bands);
+  void Synthesis(const ChannelBuffer<float>* bands, ChannelBuffer<float>* data);
+
+ private:
+  // Two-band analysis and synthesis work for 640 samples or less.
+  void TwoBandsAnalysis(const ChannelBuffer<float>* data,
+                        ChannelBuffer<float>* bands);
+  void TwoBandsSynthesis(const ChannelBuffer<float>* bands,
+                         ChannelBuffer<float>* data);
+  void ThreeBandsAnalysis(const ChannelBuffer<float>* data,
+                          ChannelBuffer<float>* bands);
+  void ThreeBandsSynthesis(const ChannelBuffer<float>* bands,
+                           ChannelBuffer<float>* data);
+  void InitBuffers();
+
+  const size_t num_bands_;
+  std::vector<TwoBandsStates> two_bands_states_;
+  std::vector<ThreeBandFilterBank> three_band_filter_banks_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_SPLITTING_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/splitting_filter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/splitting_filter_unittest.cc
new file mode 100644
index 0000000000..30fe4caf9c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/splitting_filter_unittest.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#define _USE_MATH_DEFINES
+
+#include "modules/audio_processing/splitting_filter.h"
+
+#include <cmath>
+
+#include "common_audio/channel_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+const size_t kSamplesPer16kHzChannel = 160;
+const size_t kSamplesPer48kHzChannel = 480;
+
+}  // namespace
+
+// Generates a signal from presence or absence of sine waves of different
+// frequencies.
+// Splits into 3 bands and checks their presence or absence.
+// Recombines the bands.
+// Calculates the delay.
+// Checks that the cross correlation of input and output is high enough at the
+// calculated delay.
+TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
+  static const int kChannels = 1;
+  static const int kSampleRateHz = 48000;
+  static const size_t kNumBands = 3;
+  static const int kFrequenciesHz[kNumBands] = {1000, 12000, 18000};
+  static const float kAmplitude = 8192.f;
+  static const size_t kChunks = 8;
+  SplittingFilter splitting_filter(kChannels, kNumBands,
+                                   kSamplesPer48kHzChannel);
+  ChannelBuffer<float> in_data(kSamplesPer48kHzChannel, kChannels, kNumBands);
+  ChannelBuffer<float> bands(kSamplesPer48kHzChannel, kChannels, kNumBands);
+  ChannelBuffer<float> out_data(kSamplesPer48kHzChannel, kChannels, kNumBands);
+  for (size_t i = 0; i < kChunks; ++i) {
+    // Input signal generation.
+    bool is_present[kNumBands];
+    memset(in_data.channels()[0], 0,
+           kSamplesPer48kHzChannel * sizeof(in_data.channels()[0][0]));
+    for (size_t j = 0; j < kNumBands; ++j) {
+      is_present[j] = i & (static_cast<size_t>(1) << j);
+      float amplitude = is_present[j] ? kAmplitude : 0.f;
+      for (size_t k = 0; k < kSamplesPer48kHzChannel; ++k) {
+        in_data.channels()[0][k] +=
+            amplitude * sin(2.f * M_PI * kFrequenciesHz[j] *
+                            (i * kSamplesPer48kHzChannel + k) / kSampleRateHz);
+      }
+    }
+    // Three band splitting filter.
+    splitting_filter.Analysis(&in_data, &bands);
+    // Energy calculation.
+    float energy[kNumBands];
+    for (size_t j = 0; j < kNumBands; ++j) {
+      energy[j] = 0.f;
+      for (size_t k = 0; k < kSamplesPer16kHzChannel; ++k) {
+        energy[j] += bands.channels(j)[0][k] * bands.channels(j)[0][k];
+      }
+      energy[j] /= kSamplesPer16kHzChannel;
+      if (is_present[j]) {
+        EXPECT_GT(energy[j], kAmplitude * kAmplitude / 4);
+      } else {
+        EXPECT_LT(energy[j], kAmplitude * kAmplitude / 4);
+      }
+    }
+    // Three band merge.
+    splitting_filter.Synthesis(&bands, &out_data);
+    // Delay and cross correlation estimation.
+    float xcorr = 0.f;
+    for (size_t delay = 0; delay < kSamplesPer48kHzChannel; ++delay) {
+      float tmpcorr = 0.f;
+      for (size_t j = delay; j < kSamplesPer48kHzChannel; ++j) {
+        tmpcorr += in_data.channels()[0][j - delay] * out_data.channels()[0][j];
+      }
+      tmpcorr /= kSamplesPer48kHzChannel;
+      if (tmpcorr > xcorr) {
+        xcorr = tmpcorr;
+      }
+    }
+    // High cross correlation check.
+    bool any_present = false;
+    for (size_t j = 0; j < kNumBands; ++j) {
+      any_present |= is_present[j];
+    }
+    if (any_present) {
+      EXPECT_GT(xcorr, kAmplitude * kAmplitude / 4);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.cc b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
new file mode 100644
index 0000000000..416e287751
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -0,0 +1,656 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/aec_dump_based_simulator.h"
+
+#include <iostream>
+#include <memory>
+
+#include "modules/audio_processing/echo_control_mobile_impl.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/test/aec_dump_based_simulator.h"
+#include "modules/audio_processing/test/protobuf_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+// Verify output bitexactness for the fixed interface.
+// TODO(peah): Check whether it would make sense to add a threshold
+// to use for checking the bitexactness in a soft manner.
+bool VerifyFixedBitExactness(const webrtc::audioproc::Stream& msg,
+                             const Int16Frame& frame) {
+  if (sizeof(frame.data[0]) * frame.data.size() != msg.output_data().size()) {
+    return false;
+  } else {
+    const int16_t* frame_data = frame.data.data();
+    for (int k = 0; k < frame.num_channels * frame.samples_per_channel; ++k) {
+      if (msg.output_data().data()[k] != frame_data[k]) {
+        return false;
+      }
+    }
+  }
+  return true;
+}
+
+// Verify output bitexactness for the float interface.
+bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg,
+                             const StreamConfig& out_config,
+                             const ChannelBuffer<float>& out_buf) {
+  if (static_cast<size_t>(msg.output_channel_size()) !=
+          out_config.num_channels() ||
+      msg.output_channel(0).size() != out_config.num_frames()) {
+    return false;
+  } else {
+    for (int ch = 0; ch < msg.output_channel_size(); ++ch) {
+      for (size_t sample = 0; sample < out_config.num_frames(); ++sample) {
+        if (msg.output_channel(ch).data()[sample] !=
+            out_buf.channels()[ch][sample]) {
+          return false;
+        }
+      }
+    }
+  }
+  return true;
+}
+
+// Selectively reads the next proto-buf message from dump-file or string input.
+// Returns a bool indicating whether a new message was available.
+bool ReadNextMessage(bool use_dump_file,
+                     FILE* dump_input_file,
+                     std::stringstream& input,
+                     webrtc::audioproc::Event& event_msg) {
+  if (use_dump_file) {
+    return ReadMessageFromFile(dump_input_file, &event_msg);
+  }
+  return ReadMessageFromString(&input, &event_msg);
+}
+
+}  // namespace
+
+AecDumpBasedSimulator::AecDumpBasedSimulator(
+    const SimulationSettings& settings,
+    rtc::scoped_refptr<AudioProcessing> audio_processing,
+    std::unique_ptr<AudioProcessingBuilder> ap_builder)
+    : AudioProcessingSimulator(settings,
+                               std::move(audio_processing),
+                               std::move(ap_builder)) {
+  MaybeOpenCallOrderFile();
+}
+
+AecDumpBasedSimulator::~AecDumpBasedSimulator() = default;
+
+void AecDumpBasedSimulator::PrepareProcessStreamCall(
+    const webrtc::audioproc::Stream& msg) {
+  if (msg.has_input_data()) {
+    // Fixed interface processing.
+    // Verify interface invariance.
+    RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
+              interface_used_ == InterfaceType::kNotSpecified);
+    interface_used_ = InterfaceType::kFixedInterface;
+
+    // Populate input buffer.
+    RTC_CHECK_EQ(sizeof(fwd_frame_.data[0]) * fwd_frame_.data.size(),
+                 msg.input_data().size());
+    memcpy(fwd_frame_.data.data(), msg.input_data().data(),
+           msg.input_data().size());
+  } else {
+    // Float interface processing.
+    // Verify interface invariance.
+    RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface ||
+              interface_used_ == InterfaceType::kNotSpecified);
+    interface_used_ = InterfaceType::kFloatInterface;
+
+    RTC_CHECK_EQ(in_buf_->num_channels(),
+                 static_cast<size_t>(msg.input_channel_size()));
+
+    // Populate input buffer.
+    for (size_t i = 0; i < in_buf_->num_channels(); ++i) {
+      RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
+                   msg.input_channel(i).size());
+      std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
+                  msg.input_channel(i).size());
+    }
+  }
+
+  if (artificial_nearend_buffer_reader_) {
+    if (artificial_nearend_buffer_reader_->Read(
+            artificial_nearend_buf_.get())) {
+      if (msg.has_input_data()) {
+        int16_t* fwd_frame_data = fwd_frame_.data.data();
+        for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
+          fwd_frame_data[k] = rtc::saturated_cast<int16_t>(
+              fwd_frame_data[k] +
+              static_cast<int16_t>(32767 *
+                                   artificial_nearend_buf_->channels()[0][k]));
+        }
+      } else {
+        for (int i = 0; i < msg.input_channel_size(); ++i) {
+          for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
+            in_buf_->channels()[i][k] +=
+                artificial_nearend_buf_->channels()[0][k];
+            in_buf_->channels()[i][k] = std::min(
+                32767.f, std::max(-32768.f, in_buf_->channels()[i][k]));
+          }
+        }
+      }
+    } else {
+      if (!artificial_nearend_eof_reported_) {
+        std::cout << "The artificial nearend file ended before the recording.";
+        artificial_nearend_eof_reported_ = true;
+      }
+    }
+  }
+
+  if (!settings_.use_stream_delay || *settings_.use_stream_delay) {
+    if (!settings_.stream_delay) {
+      if (msg.has_delay()) {
+        RTC_CHECK_EQ(AudioProcessing::kNoError,
+                     ap_->set_stream_delay_ms(msg.delay()));
+      }
+    } else {
+      RTC_CHECK_EQ(AudioProcessing::kNoError,
+                   ap_->set_stream_delay_ms(*settings_.stream_delay));
+    }
+  }
+
+  if (settings_.override_key_pressed.has_value()) {
+    // Key pressed state overridden.
+    ap_->set_stream_key_pressed(*settings_.override_key_pressed);
+  } else {
+    // Set the recorded key pressed state.
+    if (msg.has_keypress()) {
+      ap_->set_stream_key_pressed(msg.keypress());
+    }
+  }
+
+  // Set the applied input level if available.
+  aec_dump_applied_input_level_ =
+      msg.has_applied_input_volume()
+          ? absl::optional<int>(msg.applied_input_volume())
+          : absl::nullopt;
+}
+
+void AecDumpBasedSimulator::VerifyProcessStreamBitExactness(
+    const webrtc::audioproc::Stream& msg) {
+  if (bitexact_output_) {
+    if (interface_used_ == InterfaceType::kFixedInterface) {
+      bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_);
+    } else {
+      bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_);
+    }
+  }
+}
+
+void AecDumpBasedSimulator::PrepareReverseProcessStreamCall(
+    const webrtc::audioproc::ReverseStream& msg) {
+  if (msg.has_data()) {
+    // Fixed interface processing.
+    // Verify interface invariance.
+    RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
+              interface_used_ == InterfaceType::kNotSpecified);
+    interface_used_ = InterfaceType::kFixedInterface;
+
+    // Populate input buffer.
+    RTC_CHECK_EQ(sizeof(rev_frame_.data[0]) * rev_frame_.data.size(),
+                 msg.data().size());
+    memcpy(rev_frame_.data.data(), msg.data().data(), msg.data().size());
+  } else {
+    // Float interface processing.
+    // Verify interface invariance.
+    RTC_CHECK(interface_used_ == InterfaceType::kFloatInterface ||
+              interface_used_ == InterfaceType::kNotSpecified);
+    interface_used_ = InterfaceType::kFloatInterface;
+
+    RTC_CHECK_EQ(reverse_in_buf_->num_channels(),
+                 static_cast<size_t>(msg.channel_size()));
+
+    // Populate input buffer.
+    for (int i = 0; i < msg.channel_size(); ++i) {
+      RTC_CHECK_EQ(reverse_in_buf_->num_frames() *
+                       sizeof(*reverse_in_buf_->channels()[i]),
+                   msg.channel(i).size());
+      std::memcpy(reverse_in_buf_->channels()[i], msg.channel(i).data(),
+                  msg.channel(i).size());
+    }
+  }
+}
+
+void AecDumpBasedSimulator::Process() {
+  ConfigureAudioProcessor();
+
+  if (settings_.artificial_nearend_filename) {
+    std::unique_ptr<WavReader> artificial_nearend_file(
+        new WavReader(settings_.artificial_nearend_filename->c_str()));
+
+    RTC_CHECK_EQ(1, artificial_nearend_file->num_channels())
+        << "Only mono files for the artificial nearend are supported, "
+           "reverted to not using the artificial nearend file";
+
+    const int sample_rate_hz = artificial_nearend_file->sample_rate();
+    artificial_nearend_buffer_reader_.reset(
+        new ChannelBufferWavReader(std::move(artificial_nearend_file)));
+    artificial_nearend_buf_.reset(new ChannelBuffer<float>(
+        rtc::CheckedDivExact(sample_rate_hz, kChunksPerSecond), 1));
+  }
+
+  const bool use_dump_file = !settings_.aec_dump_input_string.has_value();
+  std::stringstream input;
+  if (use_dump_file) {
+    dump_input_file_ =
+        OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
+  } else {
+    input << settings_.aec_dump_input_string.value();
+  }
+
+  webrtc::audioproc::Event event_msg;
+  int capture_frames_since_init = 0;
+  int init_index = 0;
+  while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) {
+    SelectivelyToggleDataDumping(init_index, capture_frames_since_init);
+    HandleEvent(event_msg, capture_frames_since_init, init_index);
+
+    // Perfom an early exit if the init block to process has been fully
+    // processed
+    if (finished_processing_specified_init_block_) {
+      break;
+    }
+    RTC_CHECK(!settings_.init_to_process ||
+              *settings_.init_to_process >= init_index);
+  }
+
+  if (use_dump_file) {
+    fclose(dump_input_file_);
+  }
+
+  DetachAecDump();
+}
+
+void AecDumpBasedSimulator::Analyze() {
+  const bool use_dump_file = !settings_.aec_dump_input_string.has_value();
+  std::stringstream input;
+  if (use_dump_file) {
+    dump_input_file_ =
+        OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
+  } else {
+    input << settings_.aec_dump_input_string.value();
+  }
+
+  webrtc::audioproc::Event event_msg;
+  int num_capture_frames = 0;
+  int num_render_frames = 0;
+  int init_index = 0;
+  while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) {
+    if (event_msg.type() == webrtc::audioproc::Event::INIT) {
+      ++init_index;
+      constexpr float kNumFramesPerSecond = 100.f;
+      float capture_time_seconds = num_capture_frames / kNumFramesPerSecond;
+      float render_time_seconds = num_render_frames / kNumFramesPerSecond;
+
+      std::cout << "Inits:" << std::endl;
+      std::cout << init_index << ": -->" << std::endl;
+      std::cout << " Time:" << std::endl;
+      std::cout << "  Capture: " << capture_time_seconds << " s ("
+                << num_capture_frames << " frames) " << std::endl;
+      std::cout << "  Render: " << render_time_seconds << " s ("
+                << num_render_frames << " frames) " << std::endl;
+    } else if (event_msg.type() == webrtc::audioproc::Event::STREAM) {
+      ++num_capture_frames;
+    } else if (event_msg.type() == webrtc::audioproc::Event::REVERSE_STREAM) {
+      ++num_render_frames;
+    }
+  }
+
+  if (use_dump_file) {
+    fclose(dump_input_file_);
+  }
+}
+
+void AecDumpBasedSimulator::HandleEvent(
+    const webrtc::audioproc::Event& event_msg,
+    int& capture_frames_since_init,
+    int& init_index) {
+  switch (event_msg.type()) {
+    case webrtc::audioproc::Event::INIT:
+      RTC_CHECK(event_msg.has_init());
+      ++init_index;
+      capture_frames_since_init = 0;
+      HandleMessage(event_msg.init(), init_index);
+      break;
+    case webrtc::audioproc::Event::STREAM:
+      RTC_CHECK(event_msg.has_stream());
+      ++capture_frames_since_init;
+      HandleMessage(event_msg.stream());
+      break;
+    case webrtc::audioproc::Event::REVERSE_STREAM:
+      RTC_CHECK(event_msg.has_reverse_stream());
+      HandleMessage(event_msg.reverse_stream());
+      break;
+    case webrtc::audioproc::Event::CONFIG:
+      RTC_CHECK(event_msg.has_config());
+      HandleMessage(event_msg.config());
+      break;
+    case webrtc::audioproc::Event::RUNTIME_SETTING:
+      HandleMessage(event_msg.runtime_setting());
+      break;
+    case webrtc::audioproc::Event::UNKNOWN_EVENT:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
+void AecDumpBasedSimulator::HandleMessage(
+    const webrtc::audioproc::Config& msg) {
+  if (settings_.use_verbose_logging) {
+    std::cout << "Config at frame:" << std::endl;
+    std::cout << " Forward: " << get_num_process_stream_calls() << std::endl;
+    std::cout << " Reverse: " << get_num_reverse_process_stream_calls()
+              << std::endl;
+  }
+
+  if (!settings_.discard_all_settings_in_aecdump) {
+    if (settings_.use_verbose_logging) {
+      std::cout << "Setting used in config:" << std::endl;
+    }
+    AudioProcessing::Config apm_config = ap_->GetConfig();
+
+    if (msg.has_aec_enabled() || settings_.use_aec) {
+      bool enable = settings_.use_aec ? *settings_.use_aec : msg.aec_enabled();
+      apm_config.echo_canceller.enabled = enable;
+      if (settings_.use_verbose_logging) {
+        std::cout << " aec_enabled: " << (enable ? "true" : "false")
+                  << std::endl;
+      }
+    }
+
+    if (msg.has_aecm_enabled() || settings_.use_aecm) {
+      bool enable =
+          settings_.use_aecm ? *settings_.use_aecm : msg.aecm_enabled();
+      apm_config.echo_canceller.enabled |= enable;
+      apm_config.echo_canceller.mobile_mode = enable;
+      if (settings_.use_verbose_logging) {
+        std::cout << " aecm_enabled: " << (enable ? "true" : "false")
+                  << std::endl;
+      }
+    }
+
+    if (msg.has_aecm_comfort_noise_enabled() &&
+        msg.aecm_comfort_noise_enabled()) {
+      RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AECM comfort noise";
+    }
+
+    if (msg.has_aecm_routing_mode() &&
+        static_cast<webrtc::EchoControlMobileImpl::RoutingMode>(
+            msg.aecm_routing_mode()) != EchoControlMobileImpl::kSpeakerphone) {
+      RTC_LOG(LS_ERROR) << "Ignoring deprecated setting: AECM routing mode: "
+                        << msg.aecm_routing_mode();
+    }
+
+    if (msg.has_agc_enabled() || settings_.use_agc) {
+      bool enable = settings_.use_agc ? *settings_.use_agc : msg.agc_enabled();
+      apm_config.gain_controller1.enabled = enable;
+      if (settings_.use_verbose_logging) {
+        std::cout << " agc_enabled: " << (enable ? "true" : "false")
+                  << std::endl;
+      }
+    }
+
+    if (msg.has_agc_mode() || settings_.agc_mode) {
+      int mode = settings_.agc_mode ? *settings_.agc_mode : msg.agc_mode();
+      apm_config.gain_controller1.mode =
+          static_cast<webrtc::AudioProcessing::Config::GainController1::Mode>(
+              mode);
+      if (settings_.use_verbose_logging) {
+        std::cout << " agc_mode: " << mode << std::endl;
+      }
+    }
+
+    if (msg.has_agc_limiter_enabled() || settings_.use_agc_limiter) {
+      bool enable = settings_.use_agc_limiter ? *settings_.use_agc_limiter
+                                              : msg.agc_limiter_enabled();
+      apm_config.gain_controller1.enable_limiter = enable;
+      if (settings_.use_verbose_logging) {
+        std::cout << " agc_limiter_enabled: " << (enable ? "true" : "false")
+                  << std::endl;
+      }
+    }
+
+    if (settings_.use_agc2) {
+      bool enable = *settings_.use_agc2;
+      apm_config.gain_controller2.enabled = enable;
+      if (settings_.agc2_fixed_gain_db) {
+        apm_config.gain_controller2.fixed_digital.gain_db =
+            *settings_.agc2_fixed_gain_db;
+      }
+      if (settings_.use_verbose_logging) {
+        std::cout << " agc2_enabled: " << (enable ? "true" : "false")
+                  << std::endl;
+      }
+    }
+
+    if (msg.has_noise_robust_agc_enabled()) {
+      apm_config.gain_controller1.analog_gain_controller.enabled =
+          settings_.use_analog_agc ? *settings_.use_analog_agc
+                                   : msg.noise_robust_agc_enabled();
+      if (settings_.use_verbose_logging) {
+        std::cout << " noise_robust_agc_enabled: "
+                  << (msg.noise_robust_agc_enabled() ? "true" : "false")
+                  << std::endl;
+      }
+    }
+
+    if (msg.has_transient_suppression_enabled() || settings_.use_ts) {
+      bool enable = settings_.use_ts ? *settings_.use_ts
+                                     : msg.transient_suppression_enabled();
+      apm_config.transient_suppression.enabled = enable;
+      if (settings_.use_verbose_logging) {
+        std::cout << " transient_suppression_enabled: "
+                  << (enable ? "true" : "false") << std::endl;
+      }
+    }
+
+    if (msg.has_hpf_enabled() || settings_.use_hpf) {
+      bool enable = settings_.use_hpf ? *settings_.use_hpf : msg.hpf_enabled();
+      apm_config.high_pass_filter.enabled = enable;
+      if (settings_.use_verbose_logging) {
+        std::cout << " hpf_enabled: " << (enable ? "true" : "false")
+                  << std::endl;
+      }
+    }
+
+    if (msg.has_ns_enabled() || settings_.use_ns) {
+      bool enable = settings_.use_ns ? *settings_.use_ns : msg.ns_enabled();
+      apm_config.noise_suppression.enabled = enable;
+      if (settings_.use_verbose_logging) {
+        std::cout << " ns_enabled: " << (enable ? "true" : "false")
+                  << std::endl;
+      }
+    }
+
+    if (msg.has_ns_level() || settings_.ns_level) {
+      int level = settings_.ns_level ? *settings_.ns_level : msg.ns_level();
+      apm_config.noise_suppression.level =
+          static_cast<AudioProcessing::Config::NoiseSuppression::Level>(level);
+      if (settings_.use_verbose_logging) {
+        std::cout << " ns_level: " << level << std::endl;
+      }
+    }
+
+    if (msg.has_pre_amplifier_enabled() || settings_.use_pre_amplifier) {
+      const bool enable = settings_.use_pre_amplifier
+                              ? *settings_.use_pre_amplifier
+                              : msg.pre_amplifier_enabled();
+      apm_config.pre_amplifier.enabled = enable;
+    }
+
+    if (msg.has_pre_amplifier_fixed_gain_factor() ||
+        settings_.pre_amplifier_gain_factor) {
+      const float gain = settings_.pre_amplifier_gain_factor
+                             ? *settings_.pre_amplifier_gain_factor
+                             : msg.pre_amplifier_fixed_gain_factor();
+      apm_config.pre_amplifier.fixed_gain_factor = gain;
+    }
+
+    if (settings_.use_verbose_logging && msg.has_experiments_description() &&
+        !msg.experiments_description().empty()) {
+      std::cout << " experiments not included by default in the simulation: "
+                << msg.experiments_description() << std::endl;
+    }
+
+    ap_->ApplyConfig(apm_config);
+  }
+}
+
+void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg,
+                                          int init_index) {
+  RTC_CHECK(msg.has_sample_rate());
+  RTC_CHECK(msg.has_num_input_channels());
+  RTC_CHECK(msg.has_num_reverse_channels());
+  RTC_CHECK(msg.has_reverse_sample_rate());
+
+  // Do not perform the init if the init block to process is fully processed
+  if (settings_.init_to_process && *settings_.init_to_process < init_index) {
+    finished_processing_specified_init_block_ = true;
+  }
+
+  MaybeOpenCallOrderFile();
+
+  if (settings_.use_verbose_logging) {
+    std::cout << "Init at frame:" << std::endl;
+    std::cout << " Forward: " << get_num_process_stream_calls() << std::endl;
+    std::cout << " Reverse: " << get_num_reverse_process_stream_calls()
+              << std::endl;
+  }
+
+  int num_output_channels;
+  if (settings_.output_num_channels) {
+    num_output_channels = *settings_.output_num_channels;
+  } else {
+    num_output_channels = msg.has_num_output_channels()
+                              ? msg.num_output_channels()
+                              : msg.num_input_channels();
+  }
+
+  int output_sample_rate;
+  if (settings_.output_sample_rate_hz) {
+    output_sample_rate = *settings_.output_sample_rate_hz;
+  } else {
+    output_sample_rate = msg.has_output_sample_rate() ? msg.output_sample_rate()
+                                                      : msg.sample_rate();
+  }
+
+  int num_reverse_output_channels;
+  if (settings_.reverse_output_num_channels) {
+    num_reverse_output_channels = *settings_.reverse_output_num_channels;
+  } else {
+    num_reverse_output_channels = msg.has_num_reverse_output_channels()
+                                      ? msg.num_reverse_output_channels()
+                                      : msg.num_reverse_channels();
+  }
+
+  int reverse_output_sample_rate;
+  if (settings_.reverse_output_sample_rate_hz) {
+    reverse_output_sample_rate = *settings_.reverse_output_sample_rate_hz;
+  } else {
+    reverse_output_sample_rate = msg.has_reverse_output_sample_rate()
+                                     ? msg.reverse_output_sample_rate()
+                                     : msg.reverse_sample_rate();
+  }
+
+  SetupBuffersConfigsOutputs(
+      msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(),
+      reverse_output_sample_rate, msg.num_input_channels(), num_output_channels,
+      msg.num_reverse_channels(), num_reverse_output_channels);
+}
+
+void AecDumpBasedSimulator::HandleMessage(
+    const webrtc::audioproc::Stream& msg) {
+  if (call_order_output_file_) {
+    *call_order_output_file_ << "c";
+  }
+  PrepareProcessStreamCall(msg);
+  ProcessStream(interface_used_ == InterfaceType::kFixedInterface);
+  VerifyProcessStreamBitExactness(msg);
+}
+
+void AecDumpBasedSimulator::HandleMessage(
+    const webrtc::audioproc::ReverseStream& msg) {
+  if (call_order_output_file_) {
+    *call_order_output_file_ << "r";
+  }
+  PrepareReverseProcessStreamCall(msg);
+  ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
+}
+
+void AecDumpBasedSimulator::HandleMessage(
+    const webrtc::audioproc::RuntimeSetting& msg) {
+  RTC_CHECK(ap_.get());
+  if (msg.has_capture_pre_gain()) {
+    // Handle capture pre-gain runtime setting only if not overridden.
+    const bool pre_amplifier_overridden =
+        (!settings_.use_pre_amplifier || *settings_.use_pre_amplifier) &&
+        !settings_.pre_amplifier_gain_factor;
+    const bool capture_level_adjustment_overridden =
+        (!settings_.use_capture_level_adjustment ||
+         *settings_.use_capture_level_adjustment) &&
+        !settings_.pre_gain_factor;
+    if (pre_amplifier_overridden || capture_level_adjustment_overridden) {
+      ap_->SetRuntimeSetting(
+          AudioProcessing::RuntimeSetting::CreateCapturePreGain(
+              msg.capture_pre_gain()));
+    }
+  } else if (msg.has_capture_post_gain()) {
+    // Handle capture post-gain runtime setting only if not overridden.
+    if ((!settings_.use_capture_level_adjustment ||
+         *settings_.use_capture_level_adjustment) &&
+        !settings_.post_gain_factor) {
+      ap_->SetRuntimeSetting(
+          AudioProcessing::RuntimeSetting::CreateCapturePreGain(
+              msg.capture_pre_gain()));
+    }
+  } else if (msg.has_capture_fixed_post_gain()) {
+    // Handle capture fixed-post-gain runtime setting only if not overridden.
+    if ((!settings_.use_agc2 || *settings_.use_agc2) &&
+        !settings_.agc2_fixed_gain_db) {
+      ap_->SetRuntimeSetting(
+          AudioProcessing::RuntimeSetting::CreateCaptureFixedPostGain(
+              msg.capture_fixed_post_gain()));
+    }
+  } else if (msg.has_playout_volume_change()) {
+    ap_->SetRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(
+            msg.playout_volume_change()));
+  } else if (msg.has_playout_audio_device_change()) {
+    ap_->SetRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreatePlayoutAudioDeviceChange(
+            {msg.playout_audio_device_change().id(),
+             msg.playout_audio_device_change().max_volume()}));
+  } else if (msg.has_capture_output_used()) {
+    ap_->SetRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
+            msg.capture_output_used()));
+  }
+}
+
+void AecDumpBasedSimulator::MaybeOpenCallOrderFile() {
+  if (settings_.call_order_output_filename.has_value()) {
+    const std::string filename = settings_.store_intermediate_output
+                                     ? *settings_.call_order_output_filename +
+                                           "_" +
+                                           std::to_string(output_reset_counter_)
+                                     : *settings_.call_order_output_filename;
+    call_order_output_file_ = std::make_unique<std::ofstream>(filename);
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h
new file mode 100644
index 0000000000..e2c1f3e4ba
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/aec_dump_based_simulator.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
+#define MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
+
+#include <fstream>
+#include <string>
+
+#include "modules/audio_processing/test/audio_processing_simulator.h"
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "modules/audio_processing/debug.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+namespace test {
+
+// Used to perform an audio processing simulation from an aec dump.
+class AecDumpBasedSimulator final : public AudioProcessingSimulator {
+ public:
+  AecDumpBasedSimulator(const SimulationSettings& settings,
+                        rtc::scoped_refptr<AudioProcessing> audio_processing,
+                        std::unique_ptr<AudioProcessingBuilder> ap_builder);
+
+  AecDumpBasedSimulator() = delete;
+  AecDumpBasedSimulator(const AecDumpBasedSimulator&) = delete;
+  AecDumpBasedSimulator& operator=(const AecDumpBasedSimulator&) = delete;
+
+  ~AecDumpBasedSimulator() override;
+
+  // Processes the messages in the aecdump file.
+  void Process() override;
+
+  // Analyzes the data in the aecdump file and reports the resulting statistics.
+  void Analyze() override;
+
+ private:
+  void HandleEvent(const webrtc::audioproc::Event& event_msg,
+                   int& num_forward_chunks_processed,
+                   int& init_index);
+  void HandleMessage(const webrtc::audioproc::Init& msg, int init_index);
+  void HandleMessage(const webrtc::audioproc::Stream& msg);
+  void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
+  void HandleMessage(const webrtc::audioproc::Config& msg);
+  void HandleMessage(const webrtc::audioproc::RuntimeSetting& msg);
+  void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg);
+  void PrepareReverseProcessStreamCall(
+      const webrtc::audioproc::ReverseStream& msg);
+  void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg);
+  void MaybeOpenCallOrderFile();
+  enum InterfaceType {
+    kFixedInterface,
+    kFloatInterface,
+    kNotSpecified,
+  };
+
+  FILE* dump_input_file_;
+  std::unique_ptr<ChannelBuffer<float>> artificial_nearend_buf_;
+  std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_;
+  bool artificial_nearend_eof_reported_ = false;
+  InterfaceType interface_used_ = InterfaceType::kNotSpecified;
+  std::unique_ptr<std::ofstream> call_order_output_file_;
+  bool finished_processing_specified_init_block_ = false;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/AndroidManifest.xml b/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/AndroidManifest.xml
new file mode 100644
index 0000000000..c6063b3d76
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/AndroidManifest.xml
@@ -0,0 +1,30 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- BEGIN_INCLUDE(manifest) -->
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+        package="com.example.native_activity"
+        android:versionCode="1"
+        android:versionName="1.0">
+
+    <!-- This is the platform API where NativeActivity was introduced. -->
+    <uses-sdk android:minSdkVersion="8" />
+
+    <!-- This .apk has no Java code itself, so set hasCode to false. -->
+    <application android:label="@string/app_name" android:hasCode="false" android:debuggable="true">
+
+        <!-- Our activity is the built-in NativeActivity framework class.
+             This will take care of integrating with our NDK code. -->
+        <activity android:name="android.app.NativeActivity"
+                android:label="@string/app_name"
+                android:configChanges="orientation|keyboardHidden">
+            <!-- Tell NativeActivity the name of or .so -->
+            <meta-data android:name="android.app.lib_name"
+                    android:value="apmtest-activity" />
+            <intent-filter>
+                <action android:name="android.intent.action.MAIN" />
+                <category android:name="android.intent.category.LAUNCHER" />
+            </intent-filter>
+        </activity>
+    </application>
+
+</manifest> 
+<!-- END_INCLUDE(manifest) -->
diff --git a/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/default.properties b/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/default.properties
new file mode 100644
index 0000000000..9a2c9f6c88
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/default.properties
@@ -0,0 +1,11 @@
+# This file is automatically generated by Android Tools.
+# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
+# 
+# This file must be checked in Version Control Systems.
+# 
+# To customize properties used by the Ant build system use,
+# "build.properties", and override values to adapt the script to your
+# project structure.
+
+# Project target.
+target=android-9
diff --git a/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/jni/main.c b/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/jni/main.c
new file mode 100644
index 0000000000..2e19635683
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/jni/main.c
@@ -0,0 +1,307 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ */
+
+//BEGIN_INCLUDE(all)
+#include <jni.h>
+#include <errno.h>
+
+#include <EGL/egl.h>
+#include <GLES/gl.h>
+
+#include <android/sensor.h>
+#include <android/log.h>
+#include <android_native_app_glue.h>
+
+#define LOGI(...) ((void)__android_log_print(ANDROID_LOG_INFO, "native-activity", __VA_ARGS__))
+#define LOGW(...) ((void)__android_log_print(ANDROID_LOG_WARN, "native-activity", __VA_ARGS__))
+
+/**
+ * Our saved state data.
+ */
+struct saved_state {
+    float angle;
+    int32_t x;
+    int32_t y;
+};
+
+/**
+ * Shared state for our app.
+ */
+struct engine {
+    struct android_app* app;
+
+    ASensorManager* sensorManager;
+    const ASensor* accelerometerSensor;
+    ASensorEventQueue* sensorEventQueue;
+
+    int animating;
+    EGLDisplay display;
+    EGLSurface surface;
+    EGLContext context;
+    int32_t width;
+    int32_t height;
+    struct saved_state state;
+};
+
+/**
+ * Initialize an EGL context for the current display.
+ */
+static int engine_init_display(struct engine* engine) {
+    // initialize OpenGL ES and EGL
+
+    /*
+     * Here specify the attributes of the desired configuration.
+     * Below, we select an EGLConfig with at least 8 bits per color
+     * component compatible with on-screen windows
+     */
+    const EGLint attribs[] = {
+            EGL_SURFACE_TYPE, EGL_WINDOW_BIT,
+            EGL_BLUE_SIZE, 8,
+            EGL_GREEN_SIZE, 8,
+            EGL_RED_SIZE, 8,
+            EGL_NONE
+    };
+    EGLint w, h, dummy, format;
+    EGLint numConfigs;
+    EGLConfig config;
+    EGLSurface surface;
+    EGLContext context;
+
+    EGLDisplay display = eglGetDisplay(EGL_DEFAULT_DISPLAY);
+
+    eglInitialize(display, 0, 0);
+
+    /* Here, the application chooses the configuration it desires. In this
+     * sample, we have a very simplified selection process, where we pick
+     * the first EGLConfig that matches our criteria */
+    eglChooseConfig(display, attribs, &config, 1, &numConfigs);
+
+    /* EGL_NATIVE_VISUAL_ID is an attribute of the EGLConfig that is
+     * guaranteed to be accepted by ANativeWindow_setBuffersGeometry().
+     * As soon as we picked a EGLConfig, we can safely reconfigure the
+     * ANativeWindow buffers to match, using EGL_NATIVE_VISUAL_ID. */
+    eglGetConfigAttrib(display, config, EGL_NATIVE_VISUAL_ID, &format);
+
+    ANativeWindow_setBuffersGeometry(engine->app->window, 0, 0, format);
+
+    surface = eglCreateWindowSurface(display, config, engine->app->window, NULL);
+    context = eglCreateContext(display, config, NULL, NULL);
+
+    if (eglMakeCurrent(display, surface, surface, context) == EGL_FALSE) {
+        LOGW("Unable to eglMakeCurrent");
+        return -1;
+    }
+
+    eglQuerySurface(display, surface, EGL_WIDTH, &w);
+    eglQuerySurface(display, surface, EGL_HEIGHT, &h);
+
+    engine->display = display;
+    engine->context = context;
+    engine->surface = surface;
+    engine->width = w;
+    engine->height = h;
+    engine->state.angle = 0;
+
+    // Initialize GL state.
+    glHint(GL_PERSPECTIVE_CORRECTION_HINT, GL_FASTEST);
+    glEnable(GL_CULL_FACE);
+    glShadeModel(GL_SMOOTH);
+    glDisable(GL_DEPTH_TEST);
+
+    return 0;
+}
+
+/**
+ * Just the current frame in the display.
+ */
+static void engine_draw_frame(struct engine* engine) {
+    if (engine->display == NULL) {
+        // No display.
+        return;
+    }
+
+    // Just fill the screen with a color.
+    glClearColor(((float)engine->state.x)/engine->width, engine->state.angle,
+            ((float)engine->state.y)/engine->height, 1);
+    glClear(GL_COLOR_BUFFER_BIT);
+
+    eglSwapBuffers(engine->display, engine->surface);
+}
+
+/**
+ * Tear down the EGL context currently associated with the display.
+ */
+static void engine_term_display(struct engine* engine) {
+    if (engine->display != EGL_NO_DISPLAY) {
+        eglMakeCurrent(engine->display, EGL_NO_SURFACE, EGL_NO_SURFACE, EGL_NO_CONTEXT);
+        if (engine->context != EGL_NO_CONTEXT) {
+            eglDestroyContext(engine->display, engine->context);
+        }
+        if (engine->surface != EGL_NO_SURFACE) {
+            eglDestroySurface(engine->display, engine->surface);
+        }
+        eglTerminate(engine->display);
+    }
+    engine->animating = 0;
+    engine->display = EGL_NO_DISPLAY;
+    engine->context = EGL_NO_CONTEXT;
+    engine->surface = EGL_NO_SURFACE;
+}
+
+/**
+ * Process the next input event.
+ */
+static int32_t engine_handle_input(struct android_app* app, AInputEvent* event) {
+    struct engine* engine = (struct engine*)app->userData;
+    if (AInputEvent_getType(event) == AINPUT_EVENT_TYPE_MOTION) {
+        engine->animating = 1;
+        engine->state.x = AMotionEvent_getX(event, 0);
+        engine->state.y = AMotionEvent_getY(event, 0);
+        return 1;
+    }
+    return 0;
+}
+
+/**
+ * Process the next main command.
+ */
+static void engine_handle_cmd(struct android_app* app, int32_t cmd) {
+    struct engine* engine = (struct engine*)app->userData;
+    switch (cmd) {
+        case APP_CMD_SAVE_STATE:
+            // The system has asked us to save our current state.  Do so.
+            engine->app->savedState = malloc(sizeof(struct saved_state));
+            *((struct saved_state*)engine->app->savedState) = engine->state;
+            engine->app->savedStateSize = sizeof(struct saved_state);
+            break;
+        case APP_CMD_INIT_WINDOW:
+            // The window is being shown, get it ready.
+            if (engine->app->window != NULL) {
+                engine_init_display(engine);
+                engine_draw_frame(engine);
+            }
+            break;
+        case APP_CMD_TERM_WINDOW:
+            // The window is being hidden or closed, clean it up.
+            engine_term_display(engine);
+            break;
+        case APP_CMD_GAINED_FOCUS:
+            // When our app gains focus, we start monitoring the accelerometer.
+            if (engine->accelerometerSensor != NULL) {
+                ASensorEventQueue_enableSensor(engine->sensorEventQueue,
+                        engine->accelerometerSensor);
+                // We'd like to get 60 events per second (in us).
+                ASensorEventQueue_setEventRate(engine->sensorEventQueue,
+                        engine->accelerometerSensor, (1000L/60)*1000);
+            }
+            break;
+        case APP_CMD_LOST_FOCUS:
+            // When our app loses focus, we stop monitoring the accelerometer.
+            // This is to avoid consuming battery while not being used.
+            if (engine->accelerometerSensor != NULL) {
+                ASensorEventQueue_disableSensor(engine->sensorEventQueue,
+                        engine->accelerometerSensor);
+            }
+            // Also stop animating.
+            engine->animating = 0;
+            engine_draw_frame(engine);
+            break;
+    }
+}
+
+/**
+ * This is the main entry point of a native application that is using
+ * android_native_app_glue.  It runs in its own thread, with its own
+ * event loop for receiving input events and doing other things.
+ */
+void android_main(struct android_app* state) {
+    struct engine engine;
+
+    // Make sure glue isn't stripped.
+    app_dummy();
+
+    memset(&engine, 0, sizeof(engine));
+    state->userData = &engine;
+    state->onAppCmd = engine_handle_cmd;
+    state->onInputEvent = engine_handle_input;
+    engine.app = state;
+
+    // Prepare to monitor accelerometer
+    engine.sensorManager = ASensorManager_getInstance();
+    engine.accelerometerSensor = ASensorManager_getDefaultSensor(engine.sensorManager,
+            ASENSOR_TYPE_ACCELEROMETER);
+    engine.sensorEventQueue = ASensorManager_createEventQueue(engine.sensorManager,
+            state->looper, LOOPER_ID_USER, NULL, NULL);
+
+    if (state->savedState != NULL) {
+        // We are starting with a previous saved state; restore from it.
+        engine.state = *(struct saved_state*)state->savedState;
+    }
+
+    // loop waiting for stuff to do.
+
+    while (1) {
+        // Read all pending events.
+        int ident;
+        int events;
+        struct android_poll_source* source;
+
+        // If not animating, we will block forever waiting for events.
+        // If animating, we loop until all events are read, then continue
+        // to draw the next frame of animation.
+        while ((ident=ALooper_pollAll(engine.animating ? 0 : -1, NULL, &events,
+                (void**)&source)) >= 0) {
+
+            // Process this event.
+            if (source != NULL) {
+                source->process(state, source);
+            }
+
+            // If a sensor has data, process it now.
+            if (ident == LOOPER_ID_USER) {
+                if (engine.accelerometerSensor != NULL) {
+                    ASensorEvent event;
+                    while (ASensorEventQueue_getEvents(engine.sensorEventQueue,
+                            &event, 1) > 0) {
+                        LOGI("accelerometer: x=%f y=%f z=%f",
+                                event.acceleration.x, event.acceleration.y,
+                                event.acceleration.z);
+                    }
+                }
+            }
+
+            // Check if we are exiting.
+            if (state->destroyRequested != 0) {
+                engine_term_display(&engine);
+                return;
+            }
+        }
+
+        if (engine.animating) {
+            // Done with events; draw next animation frame.
+            engine.state.angle += .01f;
+            if (engine.state.angle > 1) {
+                engine.state.angle = 0;
+            }
+
+            // Drawing is throttled to the screen update rate, so there
+            // is no need to do timing here.
+            engine_draw_frame(&engine);
+        }
+    }
+}
+//END_INCLUDE(all)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/res/values/strings.xml b/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/res/values/strings.xml
new file mode 100644
index 0000000000..d0bd0f3051
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/android/apmtest/res/values/strings.xml
@@ -0,0 +1,4 @@
+<?xml version="1.0" encoding="utf-8"?>
+<resources>
+    <string name="app_name">apmtest</string>
+</resources>
diff --git a/third_party/libwebrtc/modules/audio_processing/test/api_call_statistics.cc b/third_party/libwebrtc/modules/audio_processing/test/api_call_statistics.cc
new file mode 100644
index 0000000000..ee8a308596
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/api_call_statistics.cc
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/api_call_statistics.h"
+
+#include <algorithm>
+#include <fstream>
+#include <iostream>
+#include <limits>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace test {
+
+void ApiCallStatistics::Add(int64_t duration_nanos, CallType call_type) {
+  calls_.push_back(CallData(duration_nanos, call_type));
+}
+
+void ApiCallStatistics::PrintReport() const {
+  int64_t min_render = std::numeric_limits<int64_t>::max();
+  int64_t min_capture = std::numeric_limits<int64_t>::max();
+  int64_t max_render = 0;
+  int64_t max_capture = 0;
+  int64_t sum_render = 0;
+  int64_t sum_capture = 0;
+  int64_t num_render = 0;
+  int64_t num_capture = 0;
+  int64_t avg_render = 0;
+  int64_t avg_capture = 0;
+
+  for (auto v : calls_) {
+    if (v.call_type == CallType::kRender) {
+      ++num_render;
+      min_render = std::min(min_render, v.duration_nanos);
+      max_render = std::max(max_render, v.duration_nanos);
+      sum_render += v.duration_nanos;
+    } else {
+      ++num_capture;
+      min_capture = std::min(min_capture, v.duration_nanos);
+      max_capture = std::max(max_capture, v.duration_nanos);
+      sum_capture += v.duration_nanos;
+    }
+  }
+  min_render /= rtc::kNumNanosecsPerMicrosec;
+  max_render /= rtc::kNumNanosecsPerMicrosec;
+  sum_render /= rtc::kNumNanosecsPerMicrosec;
+  min_capture /= rtc::kNumNanosecsPerMicrosec;
+  max_capture /= rtc::kNumNanosecsPerMicrosec;
+  sum_capture /= rtc::kNumNanosecsPerMicrosec;
+  avg_render = num_render > 0 ? sum_render / num_render : 0;
+  avg_capture = num_capture > 0 ? sum_capture / num_capture : 0;
+
+  std::cout << std::endl
+            << "Total time: " << (sum_capture + sum_render) * 1e-6 << " s"
+            << std::endl
+            << " Render API calls:" << std::endl
+            << "   min: " << min_render << " us" << std::endl
+            << "   max: " << max_render << " us" << std::endl
+            << "   avg: " << avg_render << " us" << std::endl
+            << " Capture API calls:" << std::endl
+            << "   min: " << min_capture << " us" << std::endl
+            << "   max: " << max_capture << " us" << std::endl
+            << "   avg: " << avg_capture << " us" << std::endl;
+}
+
+void ApiCallStatistics::WriteReportToFile(absl::string_view filename) const {
+  std::unique_ptr<std::ofstream> out =
+      std::make_unique<std::ofstream>(std::string(filename));
+  for (auto v : calls_) {
+    if (v.call_type == CallType::kRender) {
+      *out << "render, ";
+    } else {
+      *out << "capture, ";
+    }
+    *out << (v.duration_nanos / rtc::kNumNanosecsPerMicrosec) << std::endl;
+  }
+}
+
+ApiCallStatistics::CallData::CallData(int64_t duration_nanos,
+                                      CallType call_type)
+    : duration_nanos(duration_nanos), call_type(call_type) {}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/api_call_statistics.h b/third_party/libwebrtc/modules/audio_processing/test/api_call_statistics.h
new file mode 100644
index 0000000000..8fced104f9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/api_call_statistics.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_API_CALL_STATISTICS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_API_CALL_STATISTICS_H_
+
+#include <vector>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+// Collects statistics about the API call durations.
+class ApiCallStatistics {
+ public:
+  enum class CallType { kRender, kCapture };
+
+  // Adds a new datapoint.
+  void Add(int64_t duration_nanos, CallType call_type);
+
+  // Prints out a report of the statistics.
+  void PrintReport() const;
+
+  // Writes the call information to a file.
+  void WriteReportToFile(absl::string_view filename) const;
+
+ private:
+  struct CallData {
+    CallData(int64_t duration_nanos, CallType call_type);
+    int64_t duration_nanos;
+    CallType call_type;
+  };
+  std::vector<CallData> calls_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_API_CALL_STATISTICS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/apmtest.m b/third_party/libwebrtc/modules/audio_processing/test/apmtest.m
new file mode 100644
index 0000000000..1c8183c3ec
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/apmtest.m
@@ -0,0 +1,365 @@
+%
+%  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+%
+%  Use of this source code is governed by a BSD-style license
+%  that can be found in the LICENSE file in the root of the source
+%  tree. An additional intellectual property rights grant can be found
+%  in the file PATENTS.  All contributing project authors may
+%  be found in the AUTHORS file in the root of the source tree.
+%
+
+function apmtest(task, testname, filepath, casenumber, legacy)
+%APMTEST is a tool to process APM file sets and easily display the output.
+%   APMTEST(TASK, TESTNAME, CASENUMBER) performs one of several TASKs:
+%     'test'  Processes the files to produce test output.
+%     'list'  Prints a list of cases in the test set, preceded by their
+%             CASENUMBERs.
+%     'show'  Uses spclab to show the test case specified by the
+%             CASENUMBER parameter.
+%
+%   using a set of test files determined by TESTNAME:
+%     'all'   All tests.
+%     'apm'   The standard APM test set (default).
+%     'apmm'  The mobile APM test set.
+%     'aec'   The AEC test set.
+%     'aecm'  The AECM test set.
+%     'agc'   The AGC test set.
+%     'ns'    The NS test set.
+%     'vad'   The VAD test set.
+%
+%   FILEPATH specifies the path to the test data files.
+%
+%   CASENUMBER can be used to select a single test case. Omit CASENUMBER,
+%   or set to zero, to use all test cases.
+%
+
+if nargin < 5 || isempty(legacy)
+  % Set to true to run old VQE recordings.
+  legacy = false;
+end
+
+if nargin < 4 || isempty(casenumber)
+  casenumber = 0;
+end
+
+if nargin < 3 || isempty(filepath)
+  filepath = 'data/';
+end
+
+if nargin < 2 || isempty(testname)
+  testname = 'all';
+end
+
+if nargin < 1 || isempty(task)
+  task = 'test';
+end
+
+if ~strcmp(task, 'test') && ~strcmp(task, 'list') && ~strcmp(task, 'show')
+  error(['TASK ' task ' is not recognized']);
+end
+
+if casenumber == 0 && strcmp(task, 'show')
+  error(['CASENUMBER must be specified for TASK ' task]);
+end
+
+inpath = [filepath 'input/'];
+outpath = [filepath 'output/'];
+refpath = [filepath 'reference/'];
+
+if strcmp(testname, 'all')
+  tests = {'apm','apmm','aec','aecm','agc','ns','vad'};
+else
+  tests = {testname};
+end
+
+if legacy
+  progname = './test';
+else
+  progname = './process_test';
+end
+
+global farFile;
+global nearFile;
+global eventFile;
+global delayFile;
+global driftFile;
+
+if legacy
+  farFile = 'vqeFar.pcm';
+  nearFile = 'vqeNear.pcm';
+  eventFile = 'vqeEvent.dat';
+  delayFile = 'vqeBuf.dat';
+  driftFile = 'vqeDrift.dat';
+else
+  farFile = 'apm_far.pcm';
+  nearFile = 'apm_near.pcm';
+  eventFile = 'apm_event.dat';
+  delayFile = 'apm_delay.dat';
+  driftFile = 'apm_drift.dat';
+end
+
+simulateMode = false;
+nErr = 0;
+nCases = 0;
+for i=1:length(tests)
+  simulateMode = false;
+
+  if strcmp(tests{i}, 'apm')
+    testdir = ['apm/'];
+    outfile = ['out'];
+    if legacy
+      opt = ['-ec 1 -agc 2 -nc 2 -vad 3'];
+    else
+      opt = ['--no_progress -hpf' ...
+          ' -aec --drift_compensation -agc --fixed_digital' ...
+          ' -ns --ns_moderate -vad'];
+    end
+
+  elseif strcmp(tests{i}, 'apm-swb')
+    simulateMode = true;
+    testdir = ['apm-swb/'];
+    outfile = ['out'];
+    if legacy
+      opt = ['-fs 32000 -ec 1 -agc 2 -nc 2'];
+    else
+      opt = ['--no_progress -fs 32000 -hpf' ...
+          ' -aec --drift_compensation -agc --adaptive_digital' ...
+          ' -ns --ns_moderate -vad'];
+    end
+  elseif strcmp(tests{i}, 'apmm')
+    testdir = ['apmm/'];
+    outfile = ['out'];
+    opt = ['-aec --drift_compensation -agc --fixed_digital -hpf -ns ' ...
+        '--ns_moderate'];
+
+  else
+    error(['TESTNAME ' tests{i} ' is not recognized']);
+  end
+
+  inpathtest = [inpath testdir];
+  outpathtest = [outpath testdir];
+  refpathtest = [refpath testdir];
+
+  if ~exist(inpathtest,'dir')
+    error(['Input directory ' inpathtest ' does not exist']);
+  end
+
+  if ~exist(refpathtest,'dir')
+    warning(['Reference directory ' refpathtest ' does not exist']);
+  end
+
+  [status, errMsg] = mkdir(outpathtest);
+  if (status == 0)
+    error(errMsg);
+  end
+
+  [nErr, nCases] = recurseDir(inpathtest, outpathtest, refpathtest, outfile, ...
+      progname, opt, simulateMode, nErr, nCases, task, casenumber, legacy);
+
+  if strcmp(task, 'test') || strcmp(task, 'show')
+    system(['rm ' farFile]);
+    system(['rm ' nearFile]);
+    if simulateMode == false
+      system(['rm ' eventFile]);
+      system(['rm ' delayFile]);
+      system(['rm ' driftFile]);
+    end
+  end
+end
+
+if ~strcmp(task, 'list')
+  if nErr == 0
+    fprintf(1, '\nAll files are bit-exact to reference\n', nErr);
+  else
+    fprintf(1, '\n%d files are NOT bit-exact to reference\n', nErr);
+  end
+end
+
+
+function [nErrOut, nCases] = recurseDir(inpath, outpath, refpath, ...
+    outfile, progname, opt, simulateMode, nErr, nCases, task, casenumber, ...
+    legacy)
+
+global farFile;
+global nearFile;
+global eventFile;
+global delayFile;
+global driftFile;
+
+dirs = dir(inpath);
+nDirs = 0;
+nErrOut = nErr;
+for i=3:length(dirs) % skip . and ..
+  nDirs = nDirs + dirs(i).isdir;
+end
+
+
+if nDirs == 0
+  nCases = nCases + 1;
+
+  if casenumber == nCases || casenumber == 0
+
+    if strcmp(task, 'list')
+      fprintf([num2str(nCases) '. ' outfile '\n'])
+    else
+      vadoutfile = ['vad_' outfile '.dat'];
+      outfile = [outfile '.pcm'];
+
+      % Check for VAD test
+      vadTest = 0;
+      if ~isempty(findstr(opt, '-vad'))
+        vadTest = 1;
+        if legacy
+          opt = [opt ' ' outpath vadoutfile];
+        else
+          opt = [opt ' --vad_out_file ' outpath vadoutfile];
+        end
+      end
+
+      if exist([inpath 'vqeFar.pcm'])
+        system(['ln -s -f ' inpath 'vqeFar.pcm ' farFile]);
+      elseif exist([inpath 'apm_far.pcm'])
+        system(['ln -s -f ' inpath 'apm_far.pcm ' farFile]);
+      end
+
+      if exist([inpath 'vqeNear.pcm'])
+        system(['ln -s -f ' inpath 'vqeNear.pcm ' nearFile]);
+      elseif exist([inpath 'apm_near.pcm'])
+        system(['ln -s -f ' inpath 'apm_near.pcm ' nearFile]);
+      end
+
+      if exist([inpath 'vqeEvent.dat'])
+        system(['ln -s -f ' inpath 'vqeEvent.dat ' eventFile]);
+      elseif exist([inpath 'apm_event.dat'])
+        system(['ln -s -f ' inpath 'apm_event.dat ' eventFile]);
+      end
+
+      if exist([inpath 'vqeBuf.dat'])
+        system(['ln -s -f ' inpath 'vqeBuf.dat ' delayFile]);
+      elseif exist([inpath 'apm_delay.dat'])
+        system(['ln -s -f ' inpath 'apm_delay.dat ' delayFile]);
+      end
+
+      if exist([inpath 'vqeSkew.dat'])
+        system(['ln -s -f ' inpath 'vqeSkew.dat ' driftFile]);
+      elseif exist([inpath 'vqeDrift.dat'])
+        system(['ln -s -f ' inpath 'vqeDrift.dat ' driftFile]);
+      elseif exist([inpath 'apm_drift.dat'])
+        system(['ln -s -f ' inpath 'apm_drift.dat ' driftFile]);
+      end
+
+      if simulateMode == false
+        command = [progname ' -o ' outpath outfile ' ' opt];
+      else
+        if legacy
+          inputCmd = [' -in ' nearFile];
+        else
+          inputCmd = [' -i ' nearFile];
+        end
+
+        if exist([farFile])
+          if legacy
+            inputCmd = [' -if ' farFile inputCmd];
+          else
+            inputCmd = [' -ir ' farFile inputCmd];
+          end
+        end
+        command = [progname inputCmd ' -o ' outpath outfile ' ' opt];
+      end
+      % This prevents MATLAB from using its own C libraries.
+      shellcmd = ['bash -c "unset LD_LIBRARY_PATH;'];
+      fprintf([command '\n']);
+      [status, result] = system([shellcmd command '"']);
+      fprintf(result);
+
+      fprintf(['Reference file: ' refpath outfile '\n']);
+
+      if vadTest == 1
+        equal_to_ref = are_files_equal([outpath vadoutfile], ...
+                                       [refpath vadoutfile], ...
+                                       'int8');
+        if ~equal_to_ref
+          nErr = nErr + 1;
+        end
+      end
+
+      [equal_to_ref, diffvector] = are_files_equal([outpath outfile], ...
+                                                   [refpath outfile], ...
+                                                   'int16');
+      if ~equal_to_ref
+        nErr = nErr + 1;
+      end
+
+      if strcmp(task, 'show')
+        % Assume the last init gives the sample rate of interest.
+        str_idx = strfind(result, 'Sample rate:');
+        fs = str2num(result(str_idx(end) + 13:str_idx(end) + 17));
+        fprintf('Using %d Hz\n', fs);
+
+        if exist([farFile])
+          spclab(fs, farFile, nearFile, [refpath outfile], ...
+              [outpath outfile], diffvector);
+          %spclab(fs, diffvector);
+        else
+          spclab(fs, nearFile, [refpath outfile], [outpath outfile], ...
+              diffvector);
+          %spclab(fs, diffvector);
+        end
+      end
+    end
+  end
+else
+
+  for i=3:length(dirs)
+    if dirs(i).isdir
+      [nErr, nCases] = recurseDir([inpath dirs(i).name '/'], outpath, ...
+          refpath,[outfile '_' dirs(i).name], progname, opt, ...
+          simulateMode, nErr, nCases, task, casenumber, legacy);
+    end
+  end
+end
+nErrOut = nErr;
+
+function [are_equal, diffvector] = ...
+    are_files_equal(newfile, reffile, precision, diffvector)
+
+are_equal = false;
+diffvector = 0;
+if ~exist(newfile,'file')
+  warning(['Output file ' newfile ' does not exist']);  
+  return
+end
+
+if ~exist(reffile,'file')
+  warning(['Reference file ' reffile ' does not exist']);  
+  return
+end
+
+fid = fopen(newfile,'rb');
+new = fread(fid,inf,precision);
+fclose(fid);
+
+fid = fopen(reffile,'rb');
+ref = fread(fid,inf,precision);
+fclose(fid);
+
+if length(new) ~= length(ref)
+  warning('Reference is not the same length as output');
+  minlength = min(length(new), length(ref));
+  new = new(1:minlength);
+  ref = ref(1:minlength);
+end
+diffvector = new - ref;
+
+if isequal(new, ref)
+  fprintf([newfile ' is bit-exact to reference\n']);
+  are_equal = true;
+else
+  if isempty(new)
+    warning([newfile ' is empty']);
+    return
+  end
+  snr = snrseg(new,ref,80);
+  fprintf('\n');
+  are_equal = false;
+end
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.cc b/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.cc
new file mode 100644
index 0000000000..64fb9c7ab1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+
+#include <string.h>
+
+namespace webrtc {
+namespace test {
+
+void SetupFrame(const StreamConfig& stream_config,
+                std::vector<float*>* frame,
+                std::vector<float>* frame_samples) {
+  frame_samples->resize(stream_config.num_channels() *
+                        stream_config.num_frames());
+  frame->resize(stream_config.num_channels());
+  for (size_t ch = 0; ch < stream_config.num_channels(); ++ch) {
+    (*frame)[ch] = &(*frame_samples)[ch * stream_config.num_frames()];
+  }
+}
+
+void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
+                             rtc::ArrayView<const float> source,
+                             AudioBuffer* destination) {
+  std::vector<float*> input;
+  std::vector<float> input_samples;
+
+  SetupFrame(stream_config, &input, &input_samples);
+
+  RTC_CHECK_EQ(input_samples.size(), source.size());
+  memcpy(input_samples.data(), source.data(),
+         source.size() * sizeof(source[0]));
+
+  destination->CopyFrom(&input[0], stream_config);
+}
+
+void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config,
+                                  AudioBuffer* source,
+                                  std::vector<float>* destination) {
+  std::vector<float*> output;
+
+  SetupFrame(stream_config, &output, destination);
+
+  source->CopyTo(stream_config, &output[0]);
+}
+
+void FillBuffer(float value, AudioBuffer& audio_buffer) {
+  for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) {
+    FillBufferChannel(value, ch, audio_buffer);
+  }
+}
+
+void FillBufferChannel(float value, int channel, AudioBuffer& audio_buffer) {
+  RTC_CHECK_LT(channel, audio_buffer.num_channels());
+  for (size_t i = 0; i < audio_buffer.num_frames(); ++i) {
+    audio_buffer.channels()[channel][i] = value;
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.h b/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.h
new file mode 100644
index 0000000000..faac4bf9ff
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+namespace test {
+
+// Copies a vector into an audiobuffer.
+void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
+                             rtc::ArrayView<const float> source,
+                             AudioBuffer* destination);
+
+// Extracts a vector from an audiobuffer.
+void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config,
+                                  AudioBuffer* source,
+                                  std::vector<float>* destination);
+
+// Sets all values in `audio_buffer` to `value`.
+void FillBuffer(float value, AudioBuffer& audio_buffer);
+
+// Sets all values channel `channel` for `audio_buffer` to `value`.
+void FillBufferChannel(float value, int channel, AudioBuffer& audio_buffer);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_processing_builder_for_testing.cc b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_builder_for_testing.cc
new file mode 100644
index 0000000000..6bd266dc58
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_builder_for_testing.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/audio_processing/audio_processing_impl.h"
+
+namespace webrtc {
+
+AudioProcessingBuilderForTesting::AudioProcessingBuilderForTesting() = default;
+AudioProcessingBuilderForTesting::~AudioProcessingBuilderForTesting() = default;
+
+#ifdef WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE
+
+rtc::scoped_refptr<AudioProcessing> AudioProcessingBuilderForTesting::Create() {
+  return rtc::make_ref_counted<AudioProcessingImpl>(
+      config_, std::move(capture_post_processing_),
+      std::move(render_pre_processing_), std::move(echo_control_factory_),
+      std::move(echo_detector_), std::move(capture_analyzer_));
+}
+
+#else
+
+rtc::scoped_refptr<AudioProcessing> AudioProcessingBuilderForTesting::Create() {
+  AudioProcessingBuilder builder;
+  TransferOwnershipsToBuilder(&builder);
+  return builder.SetConfig(config_).Create();
+}
+
+#endif
+
+void AudioProcessingBuilderForTesting::TransferOwnershipsToBuilder(
+    AudioProcessingBuilder* builder) {
+  builder->SetCapturePostProcessing(std::move(capture_post_processing_));
+  builder->SetRenderPreProcessing(std::move(render_pre_processing_));
+  builder->SetEchoControlFactory(std::move(echo_control_factory_));
+  builder->SetEchoDetector(std::move(echo_detector_));
+  builder->SetCaptureAnalyzer(std::move(capture_analyzer_));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_processing_builder_for_testing.h b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_builder_for_testing.h
new file mode 100644
index 0000000000..e73706c1b6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_builder_for_testing.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_BUILDER_FOR_TESTING_H_
+#define MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_BUILDER_FOR_TESTING_H_
+
+#include <list>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+// Facilitates building of AudioProcessingImp for the tests.
+class AudioProcessingBuilderForTesting {
+ public:
+  AudioProcessingBuilderForTesting();
+  AudioProcessingBuilderForTesting(const AudioProcessingBuilderForTesting&) =
+      delete;
+  AudioProcessingBuilderForTesting& operator=(
+      const AudioProcessingBuilderForTesting&) = delete;
+  ~AudioProcessingBuilderForTesting();
+
+  // Sets the APM configuration.
+  AudioProcessingBuilderForTesting& SetConfig(
+      const AudioProcessing::Config& config) {
+    config_ = config;
+    return *this;
+  }
+
+  // Sets the echo controller factory to inject when APM is created.
+  AudioProcessingBuilderForTesting& SetEchoControlFactory(
+      std::unique_ptr<EchoControlFactory> echo_control_factory) {
+    echo_control_factory_ = std::move(echo_control_factory);
+    return *this;
+  }
+
+  // Sets the capture post-processing sub-module to inject when APM is created.
+  AudioProcessingBuilderForTesting& SetCapturePostProcessing(
+      std::unique_ptr<CustomProcessing> capture_post_processing) {
+    capture_post_processing_ = std::move(capture_post_processing);
+    return *this;
+  }
+
+  // Sets the render pre-processing sub-module to inject when APM is created.
+  AudioProcessingBuilderForTesting& SetRenderPreProcessing(
+      std::unique_ptr<CustomProcessing> render_pre_processing) {
+    render_pre_processing_ = std::move(render_pre_processing);
+    return *this;
+  }
+
+  // Sets the echo detector to inject when APM is created.
+  AudioProcessingBuilderForTesting& SetEchoDetector(
+      rtc::scoped_refptr<EchoDetector> echo_detector) {
+    echo_detector_ = std::move(echo_detector);
+    return *this;
+  }
+
+  // Sets the capture analyzer sub-module to inject when APM is created.
+  AudioProcessingBuilderForTesting& SetCaptureAnalyzer(
+      std::unique_ptr<CustomAudioAnalyzer> capture_analyzer) {
+    capture_analyzer_ = std::move(capture_analyzer);
+    return *this;
+  }
+
+  // Creates an APM instance with the specified config or the default one if
+  // unspecified. Injects the specified components transferring the ownership
+  // to the newly created APM instance - i.e., except for the config, the
+  // builder is reset to its initial state.
+  rtc::scoped_refptr<AudioProcessing> Create();
+
+ private:
+  // Transfers the ownership to a non-testing builder.
+  void TransferOwnershipsToBuilder(AudioProcessingBuilder* builder);
+
+  AudioProcessing::Config config_;
+  std::unique_ptr<EchoControlFactory> echo_control_factory_;
+  std::unique_ptr<CustomProcessing> capture_post_processing_;
+  std::unique_ptr<CustomProcessing> render_pre_processing_;
+  rtc::scoped_refptr<EchoDetector> echo_detector_;
+  std::unique_ptr<CustomAudioAnalyzer> capture_analyzer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_BUILDER_FOR_TESTING_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc
new file mode 100644
index 0000000000..7497d49fde
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -0,0 +1,630 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/audio_processing_simulator.h"
+
+#include <algorithm>
+#include <fstream>
+#include <iostream>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/echo_canceller3_config_json.h"
+#include "api/audio/echo_canceller3_factory.h"
+#include "api/audio/echo_detector_creator.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "modules/audio_processing/echo_control_mobile_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/test/fake_recording_device.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/json.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+// Helper for reading JSON from a file and parsing it to an AEC3 configuration.
+EchoCanceller3Config ReadAec3ConfigFromJsonFile(absl::string_view filename) {
+  std::string json_string;
+  std::string s;
+  std::ifstream f(std::string(filename).c_str());
+  if (f.fail()) {
+    std::cout << "Failed to open the file " << filename << std::endl;
+    RTC_CHECK_NOTREACHED();
+  }
+  while (std::getline(f, s)) {
+    json_string += s;
+  }
+
+  bool parsing_successful;
+  EchoCanceller3Config cfg;
+  Aec3ConfigFromJsonString(json_string, &cfg, &parsing_successful);
+  if (!parsing_successful) {
+    std::cout << "Parsing of json string failed: " << std::endl
+              << json_string << std::endl;
+    RTC_CHECK_NOTREACHED();
+  }
+  RTC_CHECK(EchoCanceller3Config::Validate(&cfg));
+
+  return cfg;
+}
+
+std::string GetIndexedOutputWavFilename(absl::string_view wav_name,
+                                        int counter) {
+  rtc::StringBuilder ss;
+  ss << wav_name.substr(0, wav_name.size() - 4) << "_" << counter
+     << wav_name.substr(wav_name.size() - 4);
+  return ss.Release();
+}
+
+void WriteEchoLikelihoodGraphFileHeader(std::ofstream* output_file) {
+  (*output_file) << "import numpy as np" << std::endl
+                 << "import matplotlib.pyplot as plt" << std::endl
+                 << "y = np.array([";
+}
+
+void WriteEchoLikelihoodGraphFileFooter(std::ofstream* output_file) {
+  (*output_file) << "])" << std::endl
+                 << "if __name__ == '__main__':" << std::endl
+                 << "  x = np.arange(len(y))*.01" << std::endl
+                 << "  plt.plot(x, y)" << std::endl
+                 << "  plt.ylabel('Echo likelihood')" << std::endl
+                 << "  plt.xlabel('Time (s)')" << std::endl
+                 << "  plt.show()" << std::endl;
+}
+
+// RAII class for execution time measurement. Updates the provided
+// ApiCallStatistics based on the time between ScopedTimer creation and
+// leaving the enclosing scope.
+class ScopedTimer {
+ public:
+  ScopedTimer(ApiCallStatistics* api_call_statistics,
+              ApiCallStatistics::CallType call_type)
+      : start_time_(rtc::TimeNanos()),
+        call_type_(call_type),
+        api_call_statistics_(api_call_statistics) {}
+
+  ~ScopedTimer() {
+    api_call_statistics_->Add(rtc::TimeNanos() - start_time_, call_type_);
+  }
+
+ private:
+  const int64_t start_time_;
+  const ApiCallStatistics::CallType call_type_;
+  ApiCallStatistics* const api_call_statistics_;
+};
+
+}  // namespace
+
+SimulationSettings::SimulationSettings() = default;
+SimulationSettings::SimulationSettings(const SimulationSettings&) = default;
+SimulationSettings::~SimulationSettings() = default;
+
+AudioProcessingSimulator::AudioProcessingSimulator(
+    const SimulationSettings& settings,
+    rtc::scoped_refptr<AudioProcessing> audio_processing,
+    std::unique_ptr<AudioProcessingBuilder> ap_builder)
+    : settings_(settings),
+      ap_(std::move(audio_processing)),
+      applied_input_volume_(settings.initial_mic_level),
+      fake_recording_device_(
+          settings.initial_mic_level,
+          settings_.simulate_mic_gain ? *settings.simulated_mic_kind : 0),
+      worker_queue_("file_writer_task_queue") {
+  RTC_CHECK(!settings_.dump_internal_data || WEBRTC_APM_DEBUG_DUMP == 1);
+  if (settings_.dump_start_frame || settings_.dump_end_frame) {
+    ApmDataDumper::SetActivated(!settings_.dump_start_frame);
+  } else {
+    ApmDataDumper::SetActivated(settings_.dump_internal_data);
+  }
+
+  if (settings_.dump_set_to_use) {
+    ApmDataDumper::SetDumpSetToUse(*settings_.dump_set_to_use);
+  }
+
+  if (settings_.dump_internal_data_output_dir.has_value()) {
+    ApmDataDumper::SetOutputDirectory(
+        settings_.dump_internal_data_output_dir.value());
+  }
+
+  if (settings_.ed_graph_output_filename &&
+      !settings_.ed_graph_output_filename->empty()) {
+    residual_echo_likelihood_graph_writer_.open(
+        *settings_.ed_graph_output_filename);
+    RTC_CHECK(residual_echo_likelihood_graph_writer_.is_open());
+    WriteEchoLikelihoodGraphFileHeader(&residual_echo_likelihood_graph_writer_);
+  }
+
+  if (settings_.simulate_mic_gain)
+    RTC_LOG(LS_VERBOSE) << "Simulating analog mic gain";
+
+  // Create the audio processing object.
+  RTC_CHECK(!(ap_ && ap_builder))
+      << "The AudioProcessing and the AudioProcessingBuilder cannot both be "
+         "specified at the same time.";
+
+  if (ap_) {
+    RTC_CHECK(!settings_.aec_settings_filename);
+    RTC_CHECK(!settings_.print_aec_parameter_values);
+  } else {
+    // Use specied builder if such is provided, otherwise create a new builder.
+    std::unique_ptr<AudioProcessingBuilder> builder =
+        !!ap_builder ? std::move(ap_builder)
+                     : std::make_unique<AudioProcessingBuilder>();
+
+    // Create and set an EchoCanceller3Factory if needed.
+    const bool use_aec = settings_.use_aec && *settings_.use_aec;
+    if (use_aec) {
+      EchoCanceller3Config cfg;
+      if (settings_.aec_settings_filename) {
+        if (settings_.use_verbose_logging) {
+          std::cout << "Reading AEC Parameters from JSON input." << std::endl;
+        }
+        cfg = ReadAec3ConfigFromJsonFile(*settings_.aec_settings_filename);
+      }
+
+      if (settings_.linear_aec_output_filename) {
+        cfg.filter.export_linear_aec_output = true;
+      }
+
+      if (settings_.print_aec_parameter_values) {
+        if (!settings_.use_quiet_output) {
+          std::cout << "AEC settings:" << std::endl;
+        }
+        std::cout << Aec3ConfigToJsonString(cfg) << std::endl;
+      }
+
+      auto echo_control_factory = std::make_unique<EchoCanceller3Factory>(cfg);
+      builder->SetEchoControlFactory(std::move(echo_control_factory));
+    }
+
+    if (settings_.use_ed && *settings.use_ed) {
+      builder->SetEchoDetector(CreateEchoDetector());
+    }
+
+    // Create an audio processing object.
+    ap_ = builder->Create();
+    RTC_CHECK(ap_);
+  }
+}
+
+AudioProcessingSimulator::~AudioProcessingSimulator() {
+  if (residual_echo_likelihood_graph_writer_.is_open()) {
+    WriteEchoLikelihoodGraphFileFooter(&residual_echo_likelihood_graph_writer_);
+    residual_echo_likelihood_graph_writer_.close();
+  }
+}
+
+void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
+  // Optionally simulate the input volume.
+  if (settings_.simulate_mic_gain) {
+    RTC_DCHECK(!settings_.use_analog_mic_gain_emulation);
+    // Set the input volume to simulate.
+    fake_recording_device_.SetMicLevel(applied_input_volume_);
+
+    if (settings_.aec_dump_input_filename &&
+        aec_dump_applied_input_level_.has_value()) {
+      // For AEC dumps, use the applied input level, if recorded, to "virtually
+      // restore" the capture signal level before the input volume was applied.
+      fake_recording_device_.SetUndoMicLevel(*aec_dump_applied_input_level_);
+    }
+
+    // Apply the input volume.
+    if (fixed_interface) {
+      fake_recording_device_.SimulateAnalogGain(fwd_frame_.data);
+    } else {
+      fake_recording_device_.SimulateAnalogGain(in_buf_.get());
+    }
+  }
+
+  // Let APM know which input volume was applied.
+  // Keep track of whether `set_stream_analog_level()` is called.
+  bool applied_input_volume_set = false;
+  if (settings_.simulate_mic_gain) {
+    // When the input volume is simulated, use the volume applied for
+    // simulation.
+    ap_->set_stream_analog_level(fake_recording_device_.MicLevel());
+    applied_input_volume_set = true;
+  } else if (!settings_.use_analog_mic_gain_emulation) {
+    // Ignore the recommended input volume stored in `applied_input_volume_` and
+    // instead notify APM with the recorded input volume (if available).
+    if (settings_.aec_dump_input_filename &&
+        aec_dump_applied_input_level_.has_value()) {
+      // The actually applied input volume is available in the AEC dump.
+      ap_->set_stream_analog_level(*aec_dump_applied_input_level_);
+      applied_input_volume_set = true;
+    } else if (!settings_.aec_dump_input_filename) {
+      // Wav files do not include any information about the actually applied
+      // input volume. Hence, use the recommended input volume stored in
+      // `applied_input_volume_`.
+      ap_->set_stream_analog_level(applied_input_volume_);
+      applied_input_volume_set = true;
+    }
+  }
+
+  // Post any scheduled runtime settings.
+  if (settings_.frame_for_sending_capture_output_used_false &&
+      *settings_.frame_for_sending_capture_output_used_false ==
+          static_cast<int>(num_process_stream_calls_)) {
+    ap_->PostRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(false));
+  }
+  if (settings_.frame_for_sending_capture_output_used_true &&
+      *settings_.frame_for_sending_capture_output_used_true ==
+          static_cast<int>(num_process_stream_calls_)) {
+    ap_->PostRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(true));
+  }
+
+  // Process the current audio frame.
+  if (fixed_interface) {
+    {
+      const auto st = ScopedTimer(&api_call_statistics_,
+                                  ApiCallStatistics::CallType::kCapture);
+      RTC_CHECK_EQ(
+          AudioProcessing::kNoError,
+          ap_->ProcessStream(fwd_frame_.data.data(), fwd_frame_.config,
+                             fwd_frame_.config, fwd_frame_.data.data()));
+    }
+    fwd_frame_.CopyTo(out_buf_.get());
+  } else {
+    const auto st = ScopedTimer(&api_call_statistics_,
+                                ApiCallStatistics::CallType::kCapture);
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->ProcessStream(in_buf_->channels(), in_config_,
+                                    out_config_, out_buf_->channels()));
+  }
+
+  // Retrieve the recommended input volume only if `set_stream_analog_level()`
+  // has been called to stick to the APM API contract.
+  if (applied_input_volume_set) {
+    applied_input_volume_ = ap_->recommended_stream_analog_level();
+  }
+
+  if (buffer_memory_writer_) {
+    RTC_CHECK(!buffer_file_writer_);
+    buffer_memory_writer_->Write(*out_buf_);
+  } else if (buffer_file_writer_) {
+    RTC_CHECK(!buffer_memory_writer_);
+    buffer_file_writer_->Write(*out_buf_);
+  }
+
+  if (linear_aec_output_file_writer_) {
+    bool output_available = ap_->GetLinearAecOutput(linear_aec_output_buf_);
+    RTC_CHECK(output_available);
+    RTC_CHECK_GT(linear_aec_output_buf_.size(), 0);
+    RTC_CHECK_EQ(linear_aec_output_buf_[0].size(), 160);
+    for (size_t k = 0; k < linear_aec_output_buf_[0].size(); ++k) {
+      for (size_t ch = 0; ch < linear_aec_output_buf_.size(); ++ch) {
+        RTC_CHECK_EQ(linear_aec_output_buf_[ch].size(), 160);
+        float sample = FloatToFloatS16(linear_aec_output_buf_[ch][k]);
+        linear_aec_output_file_writer_->WriteSamples(&sample, 1);
+      }
+    }
+  }
+
+  if (residual_echo_likelihood_graph_writer_.is_open()) {
+    auto stats = ap_->GetStatistics();
+    residual_echo_likelihood_graph_writer_
+        << stats.residual_echo_likelihood.value_or(-1.f) << ", ";
+  }
+
+  ++num_process_stream_calls_;
+}
+
+void AudioProcessingSimulator::ProcessReverseStream(bool fixed_interface) {
+  if (fixed_interface) {
+    {
+      const auto st = ScopedTimer(&api_call_statistics_,
+                                  ApiCallStatistics::CallType::kRender);
+      RTC_CHECK_EQ(
+          AudioProcessing::kNoError,
+          ap_->ProcessReverseStream(rev_frame_.data.data(), rev_frame_.config,
+                                    rev_frame_.config, rev_frame_.data.data()));
+    }
+    rev_frame_.CopyTo(reverse_out_buf_.get());
+  } else {
+    const auto st = ScopedTimer(&api_call_statistics_,
+                                ApiCallStatistics::CallType::kRender);
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->ProcessReverseStream(
+                     reverse_in_buf_->channels(), reverse_in_config_,
+                     reverse_out_config_, reverse_out_buf_->channels()));
+  }
+
+  if (reverse_buffer_file_writer_) {
+    reverse_buffer_file_writer_->Write(*reverse_out_buf_);
+  }
+
+  ++num_reverse_process_stream_calls_;
+}
+
+void AudioProcessingSimulator::SetupBuffersConfigsOutputs(
+    int input_sample_rate_hz,
+    int output_sample_rate_hz,
+    int reverse_input_sample_rate_hz,
+    int reverse_output_sample_rate_hz,
+    int input_num_channels,
+    int output_num_channels,
+    int reverse_input_num_channels,
+    int reverse_output_num_channels) {
+  in_config_ = StreamConfig(input_sample_rate_hz, input_num_channels);
+  in_buf_.reset(new ChannelBuffer<float>(
+      rtc::CheckedDivExact(input_sample_rate_hz, kChunksPerSecond),
+      input_num_channels));
+
+  reverse_in_config_ =
+      StreamConfig(reverse_input_sample_rate_hz, reverse_input_num_channels);
+  reverse_in_buf_.reset(new ChannelBuffer<float>(
+      rtc::CheckedDivExact(reverse_input_sample_rate_hz, kChunksPerSecond),
+      reverse_input_num_channels));
+
+  out_config_ = StreamConfig(output_sample_rate_hz, output_num_channels);
+  out_buf_.reset(new ChannelBuffer<float>(
+      rtc::CheckedDivExact(output_sample_rate_hz, kChunksPerSecond),
+      output_num_channels));
+
+  reverse_out_config_ =
+      StreamConfig(reverse_output_sample_rate_hz, reverse_output_num_channels);
+  reverse_out_buf_.reset(new ChannelBuffer<float>(
+      rtc::CheckedDivExact(reverse_output_sample_rate_hz, kChunksPerSecond),
+      reverse_output_num_channels));
+
+  fwd_frame_.SetFormat(input_sample_rate_hz, input_num_channels);
+  rev_frame_.SetFormat(reverse_input_sample_rate_hz,
+                       reverse_input_num_channels);
+
+  if (settings_.use_verbose_logging) {
+    rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE);
+
+    std::cout << "Sample rates:" << std::endl;
+    std::cout << " Forward input: " << input_sample_rate_hz << std::endl;
+    std::cout << " Forward output: " << output_sample_rate_hz << std::endl;
+    std::cout << " Reverse input: " << reverse_input_sample_rate_hz
+              << std::endl;
+    std::cout << " Reverse output: " << reverse_output_sample_rate_hz
+              << std::endl;
+    std::cout << "Number of channels: " << std::endl;
+    std::cout << " Forward input: " << input_num_channels << std::endl;
+    std::cout << " Forward output: " << output_num_channels << std::endl;
+    std::cout << " Reverse input: " << reverse_input_num_channels << std::endl;
+    std::cout << " Reverse output: " << reverse_output_num_channels
+              << std::endl;
+  }
+
+  SetupOutput();
+}
+
+void AudioProcessingSimulator::SelectivelyToggleDataDumping(
+    int init_index,
+    int capture_frames_since_init) const {
+  if (!(settings_.dump_start_frame || settings_.dump_end_frame)) {
+    return;
+  }
+
+  if (settings_.init_to_process && *settings_.init_to_process != init_index) {
+    return;
+  }
+
+  if (settings_.dump_start_frame &&
+      *settings_.dump_start_frame == capture_frames_since_init) {
+    ApmDataDumper::SetActivated(true);
+  }
+
+  if (settings_.dump_end_frame &&
+      *settings_.dump_end_frame == capture_frames_since_init) {
+    ApmDataDumper::SetActivated(false);
+  }
+}
+
+void AudioProcessingSimulator::SetupOutput() {
+  if (settings_.output_filename) {
+    std::string filename;
+    if (settings_.store_intermediate_output) {
+      filename = GetIndexedOutputWavFilename(*settings_.output_filename,
+                                             output_reset_counter_);
+    } else {
+      filename = *settings_.output_filename;
+    }
+
+    std::unique_ptr<WavWriter> out_file(
+        new WavWriter(filename, out_config_.sample_rate_hz(),
+                      static_cast<size_t>(out_config_.num_channels()),
+                      settings_.wav_output_format));
+    buffer_file_writer_.reset(new ChannelBufferWavWriter(std::move(out_file)));
+  } else if (settings_.aec_dump_input_string.has_value()) {
+    buffer_memory_writer_ = std::make_unique<ChannelBufferVectorWriter>(
+        settings_.processed_capture_samples);
+  }
+
+  if (settings_.linear_aec_output_filename) {
+    std::string filename;
+    if (settings_.store_intermediate_output) {
+      filename = GetIndexedOutputWavFilename(
+          *settings_.linear_aec_output_filename, output_reset_counter_);
+    } else {
+      filename = *settings_.linear_aec_output_filename;
+    }
+
+    linear_aec_output_file_writer_.reset(
+        new WavWriter(filename, 16000, out_config_.num_channels(),
+                      settings_.wav_output_format));
+
+    linear_aec_output_buf_.resize(out_config_.num_channels());
+  }
+
+  if (settings_.reverse_output_filename) {
+    std::string filename;
+    if (settings_.store_intermediate_output) {
+      filename = GetIndexedOutputWavFilename(*settings_.reverse_output_filename,
+                                             output_reset_counter_);
+    } else {
+      filename = *settings_.reverse_output_filename;
+    }
+
+    std::unique_ptr<WavWriter> reverse_out_file(
+        new WavWriter(filename, reverse_out_config_.sample_rate_hz(),
+                      static_cast<size_t>(reverse_out_config_.num_channels()),
+                      settings_.wav_output_format));
+    reverse_buffer_file_writer_.reset(
+        new ChannelBufferWavWriter(std::move(reverse_out_file)));
+  }
+
+  ++output_reset_counter_;
+}
+
+void AudioProcessingSimulator::DetachAecDump() {
+  if (settings_.aec_dump_output_filename) {
+    ap_->DetachAecDump();
+  }
+}
+
+void AudioProcessingSimulator::ConfigureAudioProcessor() {
+  AudioProcessing::Config apm_config;
+  if (settings_.use_ts) {
+    apm_config.transient_suppression.enabled = *settings_.use_ts != 0;
+  }
+  if (settings_.multi_channel_render) {
+    apm_config.pipeline.multi_channel_render = *settings_.multi_channel_render;
+  }
+
+  if (settings_.multi_channel_capture) {
+    apm_config.pipeline.multi_channel_capture =
+        *settings_.multi_channel_capture;
+  }
+
+  if (settings_.use_agc2) {
+    apm_config.gain_controller2.enabled = *settings_.use_agc2;
+    if (settings_.agc2_fixed_gain_db) {
+      apm_config.gain_controller2.fixed_digital.gain_db =
+          *settings_.agc2_fixed_gain_db;
+    }
+    if (settings_.agc2_use_adaptive_gain) {
+      apm_config.gain_controller2.adaptive_digital.enabled =
+          *settings_.agc2_use_adaptive_gain;
+    }
+  }
+  if (settings_.use_pre_amplifier) {
+    apm_config.pre_amplifier.enabled = *settings_.use_pre_amplifier;
+    if (settings_.pre_amplifier_gain_factor) {
+      apm_config.pre_amplifier.fixed_gain_factor =
+          *settings_.pre_amplifier_gain_factor;
+    }
+  }
+
+  if (settings_.use_analog_mic_gain_emulation) {
+    if (*settings_.use_analog_mic_gain_emulation) {
+      apm_config.capture_level_adjustment.enabled = true;
+      apm_config.capture_level_adjustment.analog_mic_gain_emulation.enabled =
+          true;
+    } else {
+      apm_config.capture_level_adjustment.analog_mic_gain_emulation.enabled =
+          false;
+    }
+  }
+  if (settings_.analog_mic_gain_emulation_initial_level) {
+    apm_config.capture_level_adjustment.analog_mic_gain_emulation
+        .initial_level = *settings_.analog_mic_gain_emulation_initial_level;
+  }
+
+  if (settings_.use_capture_level_adjustment) {
+    apm_config.capture_level_adjustment.enabled =
+        *settings_.use_capture_level_adjustment;
+  }
+  if (settings_.pre_gain_factor) {
+    apm_config.capture_level_adjustment.pre_gain_factor =
+        *settings_.pre_gain_factor;
+  }
+  if (settings_.post_gain_factor) {
+    apm_config.capture_level_adjustment.post_gain_factor =
+        *settings_.post_gain_factor;
+  }
+
+  const bool use_aec = settings_.use_aec && *settings_.use_aec;
+  const bool use_aecm = settings_.use_aecm && *settings_.use_aecm;
+  if (use_aec || use_aecm) {
+    apm_config.echo_canceller.enabled = true;
+    apm_config.echo_canceller.mobile_mode = use_aecm;
+  }
+  apm_config.echo_canceller.export_linear_aec_output =
+      !!settings_.linear_aec_output_filename;
+
+  if (settings_.use_hpf) {
+    apm_config.high_pass_filter.enabled = *settings_.use_hpf;
+  }
+
+  if (settings_.use_agc) {
+    apm_config.gain_controller1.enabled = *settings_.use_agc;
+  }
+  if (settings_.agc_mode) {
+    apm_config.gain_controller1.mode =
+        static_cast<webrtc::AudioProcessing::Config::GainController1::Mode>(
+            *settings_.agc_mode);
+  }
+  if (settings_.use_agc_limiter) {
+    apm_config.gain_controller1.enable_limiter = *settings_.use_agc_limiter;
+  }
+  if (settings_.agc_target_level) {
+    apm_config.gain_controller1.target_level_dbfs = *settings_.agc_target_level;
+  }
+  if (settings_.agc_compression_gain) {
+    apm_config.gain_controller1.compression_gain_db =
+        *settings_.agc_compression_gain;
+  }
+  if (settings_.use_analog_agc) {
+    apm_config.gain_controller1.analog_gain_controller.enabled =
+        *settings_.use_analog_agc;
+  }
+  if (settings_.analog_agc_use_digital_adaptive_controller) {
+    apm_config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
+        *settings_.analog_agc_use_digital_adaptive_controller;
+  }
+
+  if (settings_.maximum_internal_processing_rate) {
+    apm_config.pipeline.maximum_internal_processing_rate =
+        *settings_.maximum_internal_processing_rate;
+  }
+
+  if (settings_.use_ns) {
+    apm_config.noise_suppression.enabled = *settings_.use_ns;
+  }
+  if (settings_.ns_level) {
+    const int level = *settings_.ns_level;
+    RTC_CHECK_GE(level, 0);
+    RTC_CHECK_LE(level, 3);
+    apm_config.noise_suppression.level =
+        static_cast<AudioProcessing::Config::NoiseSuppression::Level>(level);
+  }
+  if (settings_.ns_analysis_on_linear_aec_output) {
+    apm_config.noise_suppression.analyze_linear_aec_output_when_available =
+        *settings_.ns_analysis_on_linear_aec_output;
+  }
+
+  ap_->ApplyConfig(apm_config);
+
+  if (settings_.use_ts) {
+    // Default to key pressed if activating the transient suppressor with
+    // continuous key events.
+    ap_->set_stream_key_pressed(*settings_.use_ts == 2);
+  }
+
+  if (settings_.aec_dump_output_filename) {
+    ap_->AttachAecDump(AecDumpFactory::Create(
+        *settings_.aec_dump_output_filename, -1, &worker_queue_));
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.h b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.h
new file mode 100644
index 0000000000..e40d818bd8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.h
@@ -0,0 +1,247 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
+#define MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
+
+#include <algorithm>
+#include <fstream>
+#include <limits>
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "common_audio/channel_buffer.h"
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/test/api_call_statistics.h"
+#include "modules/audio_processing/test/fake_recording_device.h"
+#include "modules/audio_processing/test/test_utils.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace test {
+
+static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
+
+struct Int16Frame {
+  void SetFormat(int sample_rate_hz, int num_channels) {
+    this->sample_rate_hz = sample_rate_hz;
+    samples_per_channel =
+        rtc::CheckedDivExact(sample_rate_hz, kChunksPerSecond);
+    this->num_channels = num_channels;
+    config = StreamConfig(sample_rate_hz, num_channels);
+    data.resize(num_channels * samples_per_channel);
+  }
+
+  void CopyTo(ChannelBuffer<float>* dest) {
+    RTC_DCHECK(dest);
+    RTC_CHECK_EQ(num_channels, dest->num_channels());
+    RTC_CHECK_EQ(samples_per_channel, dest->num_frames());
+    // Copy the data from the input buffer.
+    std::vector<float> tmp(samples_per_channel * num_channels);
+    S16ToFloat(data.data(), tmp.size(), tmp.data());
+    Deinterleave(tmp.data(), samples_per_channel, num_channels,
+                 dest->channels());
+  }
+
+  void CopyFrom(const ChannelBuffer<float>& src) {
+    RTC_CHECK_EQ(src.num_channels(), num_channels);
+    RTC_CHECK_EQ(src.num_frames(), samples_per_channel);
+    data.resize(num_channels * samples_per_channel);
+    int16_t* dest_data = data.data();
+    for (int ch = 0; ch < num_channels; ++ch) {
+      for (int sample = 0; sample < samples_per_channel; ++sample) {
+        dest_data[sample * num_channels + ch] =
+            src.channels()[ch][sample] * 32767;
+      }
+    }
+  }
+
+  int sample_rate_hz;
+  int samples_per_channel;
+  int num_channels;
+
+  StreamConfig config;
+
+  std::vector<int16_t> data;
+};
+
+// Holds all the parameters available for controlling the simulation.
+struct SimulationSettings {
+  SimulationSettings();
+  SimulationSettings(const SimulationSettings&);
+  ~SimulationSettings();
+  absl::optional<int> stream_delay;
+  absl::optional<bool> use_stream_delay;
+  absl::optional<int> output_sample_rate_hz;
+  absl::optional<int> output_num_channels;
+  absl::optional<int> reverse_output_sample_rate_hz;
+  absl::optional<int> reverse_output_num_channels;
+  absl::optional<std::string> output_filename;
+  absl::optional<std::string> reverse_output_filename;
+  absl::optional<std::string> input_filename;
+  absl::optional<std::string> reverse_input_filename;
+  absl::optional<std::string> artificial_nearend_filename;
+  absl::optional<std::string> linear_aec_output_filename;
+  absl::optional<bool> use_aec;
+  absl::optional<bool> use_aecm;
+  absl::optional<bool> use_ed;  // Residual Echo Detector.
+  absl::optional<std::string> ed_graph_output_filename;
+  absl::optional<bool> use_agc;
+  absl::optional<bool> use_agc2;
+  absl::optional<bool> use_pre_amplifier;
+  absl::optional<bool> use_capture_level_adjustment;
+  absl::optional<bool> use_analog_mic_gain_emulation;
+  absl::optional<bool> use_hpf;
+  absl::optional<bool> use_ns;
+  absl::optional<int> use_ts;
+  absl::optional<bool> use_analog_agc;
+  absl::optional<bool> use_all;
+  absl::optional<bool> analog_agc_use_digital_adaptive_controller;
+  absl::optional<int> agc_mode;
+  absl::optional<int> agc_target_level;
+  absl::optional<bool> use_agc_limiter;
+  absl::optional<int> agc_compression_gain;
+  absl::optional<bool> agc2_use_adaptive_gain;
+  absl::optional<float> agc2_fixed_gain_db;
+  absl::optional<float> pre_amplifier_gain_factor;
+  absl::optional<float> pre_gain_factor;
+  absl::optional<float> post_gain_factor;
+  absl::optional<float> analog_mic_gain_emulation_initial_level;
+  absl::optional<int> ns_level;
+  absl::optional<bool> ns_analysis_on_linear_aec_output;
+  absl::optional<bool> override_key_pressed;
+  absl::optional<int> maximum_internal_processing_rate;
+  int initial_mic_level;
+  bool simulate_mic_gain = false;
+  absl::optional<bool> multi_channel_render;
+  absl::optional<bool> multi_channel_capture;
+  absl::optional<int> simulated_mic_kind;
+  absl::optional<int> frame_for_sending_capture_output_used_false;
+  absl::optional<int> frame_for_sending_capture_output_used_true;
+  bool report_performance = false;
+  absl::optional<std::string> performance_report_output_filename;
+  bool report_bitexactness = false;
+  bool use_verbose_logging = false;
+  bool use_quiet_output = false;
+  bool discard_all_settings_in_aecdump = true;
+  absl::optional<std::string> aec_dump_input_filename;
+  absl::optional<std::string> aec_dump_output_filename;
+  bool fixed_interface = false;
+  bool store_intermediate_output = false;
+  bool print_aec_parameter_values = false;
+  bool dump_internal_data = false;
+  WavFile::SampleFormat wav_output_format = WavFile::SampleFormat::kInt16;
+  absl::optional<std::string> dump_internal_data_output_dir;
+  absl::optional<int> dump_set_to_use;
+  absl::optional<std::string> call_order_input_filename;
+  absl::optional<std::string> call_order_output_filename;
+  absl::optional<std::string> aec_settings_filename;
+  absl::optional<absl::string_view> aec_dump_input_string;
+  std::vector<float>* processed_capture_samples = nullptr;
+  bool analysis_only = false;
+  absl::optional<int> dump_start_frame;
+  absl::optional<int> dump_end_frame;
+  absl::optional<int> init_to_process;
+};
+
+// Provides common functionality for performing audioprocessing simulations.
+class AudioProcessingSimulator {
+ public:
+  AudioProcessingSimulator(const SimulationSettings& settings,
+                           rtc::scoped_refptr<AudioProcessing> audio_processing,
+                           std::unique_ptr<AudioProcessingBuilder> ap_builder);
+
+  AudioProcessingSimulator() = delete;
+  AudioProcessingSimulator(const AudioProcessingSimulator&) = delete;
+  AudioProcessingSimulator& operator=(const AudioProcessingSimulator&) = delete;
+
+  virtual ~AudioProcessingSimulator();
+
+  // Processes the data in the input.
+  virtual void Process() = 0;
+
+  // Returns the execution times of all AudioProcessing calls.
+  const ApiCallStatistics& GetApiCallStatistics() const {
+    return api_call_statistics_;
+  }
+
+  // Analyzes the data in the input and reports the resulting statistics.
+  virtual void Analyze() = 0;
+
+  // Reports whether the processed recording was bitexact.
+  bool OutputWasBitexact() { return bitexact_output_; }
+
+  size_t get_num_process_stream_calls() { return num_process_stream_calls_; }
+  size_t get_num_reverse_process_stream_calls() {
+    return num_reverse_process_stream_calls_;
+  }
+
+ protected:
+  void ProcessStream(bool fixed_interface);
+  void ProcessReverseStream(bool fixed_interface);
+  void ConfigureAudioProcessor();
+  void DetachAecDump();
+  void SetupBuffersConfigsOutputs(int input_sample_rate_hz,
+                                  int output_sample_rate_hz,
+                                  int reverse_input_sample_rate_hz,
+                                  int reverse_output_sample_rate_hz,
+                                  int input_num_channels,
+                                  int output_num_channels,
+                                  int reverse_input_num_channels,
+                                  int reverse_output_num_channels);
+  void SelectivelyToggleDataDumping(int init_index,
+                                    int capture_frames_since_init) const;
+
+  const SimulationSettings settings_;
+  rtc::scoped_refptr<AudioProcessing> ap_;
+
+  std::unique_ptr<ChannelBuffer<float>> in_buf_;
+  std::unique_ptr<ChannelBuffer<float>> out_buf_;
+  std::unique_ptr<ChannelBuffer<float>> reverse_in_buf_;
+  std::unique_ptr<ChannelBuffer<float>> reverse_out_buf_;
+  std::vector<std::array<float, 160>> linear_aec_output_buf_;
+  StreamConfig in_config_;
+  StreamConfig out_config_;
+  StreamConfig reverse_in_config_;
+  StreamConfig reverse_out_config_;
+  std::unique_ptr<ChannelBufferWavReader> buffer_reader_;
+  std::unique_ptr<ChannelBufferWavReader> reverse_buffer_reader_;
+  Int16Frame rev_frame_;
+  Int16Frame fwd_frame_;
+  bool bitexact_output_ = true;
+  absl::optional<int> aec_dump_applied_input_level_ = 0;
+
+ protected:
+  size_t output_reset_counter_ = 0;
+
+ private:
+  void SetupOutput();
+
+  size_t num_process_stream_calls_ = 0;
+  size_t num_reverse_process_stream_calls_ = 0;
+  std::unique_ptr<ChannelBufferWavWriter> buffer_file_writer_;
+  std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_file_writer_;
+  std::unique_ptr<ChannelBufferVectorWriter> buffer_memory_writer_;
+  std::unique_ptr<WavWriter> linear_aec_output_file_writer_;
+  ApiCallStatistics api_call_statistics_;
+  std::ofstream residual_echo_likelihood_graph_writer_;
+  int applied_input_volume_;
+  FakeRecordingDevice fake_recording_device_;
+
+  TaskQueueForTest worker_queue_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_SIMULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc b/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc
new file mode 100644
index 0000000000..c23ec74366
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.cc
@@ -0,0 +1,821 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/audioproc_float_impl.h"
+
+#include <string.h>
+
+#include <iostream>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/test/aec_dump_based_simulator.h"
+#include "modules/audio_processing/test/audio_processing_simulator.h"
+#include "modules/audio_processing/test/wav_based_simulator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/field_trial.h"
+
+constexpr int kParameterNotSpecifiedValue = -10000;
+
+ABSL_FLAG(std::string, dump_input, "", "Aec dump input filename");
+ABSL_FLAG(std::string, dump_output, "", "Aec dump output filename");
+ABSL_FLAG(std::string, i, "", "Forward stream input wav filename");
+ABSL_FLAG(std::string, o, "", "Forward stream output wav filename");
+ABSL_FLAG(std::string, ri, "", "Reverse stream input wav filename");
+ABSL_FLAG(std::string, ro, "", "Reverse stream output wav filename");
+ABSL_FLAG(std::string,
+          artificial_nearend,
+          "",
+          "Artificial nearend wav filename");
+ABSL_FLAG(std::string, linear_aec_output, "", "Linear AEC output wav filename");
+ABSL_FLAG(int,
+          output_num_channels,
+          kParameterNotSpecifiedValue,
+          "Number of forward stream output channels");
+ABSL_FLAG(int,
+          reverse_output_num_channels,
+          kParameterNotSpecifiedValue,
+          "Number of Reverse stream output channels");
+ABSL_FLAG(int,
+          output_sample_rate_hz,
+          kParameterNotSpecifiedValue,
+          "Forward stream output sample rate in Hz");
+ABSL_FLAG(int,
+          reverse_output_sample_rate_hz,
+          kParameterNotSpecifiedValue,
+          "Reverse stream output sample rate in Hz");
+ABSL_FLAG(bool,
+          fixed_interface,
+          false,
+          "Use the fixed interface when operating on wav files");
+ABSL_FLAG(int,
+          aec,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the echo canceller");
+ABSL_FLAG(int,
+          aecm,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the mobile echo controller");
+ABSL_FLAG(int,
+          ed,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the residual echo detector");
+ABSL_FLAG(std::string,
+          ed_graph,
+          "",
+          "Output filename for graph of echo likelihood");
+ABSL_FLAG(int,
+          agc,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the AGC");
+ABSL_FLAG(int,
+          agc2,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the AGC2");
+ABSL_FLAG(int,
+          pre_amplifier,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate(0) the pre amplifier");
+ABSL_FLAG(
+    int,
+    capture_level_adjustment,
+    kParameterNotSpecifiedValue,
+    "Activate (1) or deactivate(0) the capture level adjustment functionality");
+ABSL_FLAG(int,
+          analog_mic_gain_emulation,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate(0) the analog mic gain emulation in the "
+          "production (non-test) code.");
+ABSL_FLAG(int,
+          hpf,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the high-pass filter");
+ABSL_FLAG(int,
+          ns,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the noise suppressor");
+ABSL_FLAG(int,
+          ts,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the transient suppressor");
+ABSL_FLAG(int,
+          analog_agc,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the analog AGC");
+ABSL_FLAG(bool,
+          all_default,
+          false,
+          "Activate all of the default components (will be overridden by any "
+          "other settings)");
+ABSL_FLAG(int,
+          analog_agc_use_digital_adaptive_controller,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) digital adaptation in AGC1. "
+          "Digital adaptation is active by default.");
+ABSL_FLAG(int,
+          agc_mode,
+          kParameterNotSpecifiedValue,
+          "Specify the AGC mode (0-2)");
+ABSL_FLAG(int,
+          agc_target_level,
+          kParameterNotSpecifiedValue,
+          "Specify the AGC target level (0-31)");
+ABSL_FLAG(int,
+          agc_limiter,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the level estimator");
+ABSL_FLAG(int,
+          agc_compression_gain,
+          kParameterNotSpecifiedValue,
+          "Specify the AGC compression gain (0-90)");
+ABSL_FLAG(int,
+          agc2_enable_adaptive_gain,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) the AGC2 adaptive gain");
+ABSL_FLAG(float,
+          agc2_fixed_gain_db,
+          kParameterNotSpecifiedValue,
+          "AGC2 fixed gain (dB) to apply");
+ABSL_FLAG(float,
+          pre_amplifier_gain_factor,
+          kParameterNotSpecifiedValue,
+          "Pre-amplifier gain factor (linear) to apply");
+ABSL_FLAG(float,
+          pre_gain_factor,
+          kParameterNotSpecifiedValue,
+          "Pre-gain factor (linear) to apply in the capture level adjustment");
+ABSL_FLAG(float,
+          post_gain_factor,
+          kParameterNotSpecifiedValue,
+          "Post-gain factor (linear) to apply in the capture level adjustment");
+ABSL_FLAG(float,
+          analog_mic_gain_emulation_initial_level,
+          kParameterNotSpecifiedValue,
+          "Emulated analog mic level to apply initially in the production "
+          "(non-test) code.");
+ABSL_FLAG(int,
+          ns_level,
+          kParameterNotSpecifiedValue,
+          "Specify the NS level (0-3)");
+ABSL_FLAG(int,
+          ns_analysis_on_linear_aec_output,
+          kParameterNotSpecifiedValue,
+          "Specifies whether the noise suppression analysis is done on the "
+          "linear AEC output");
+ABSL_FLAG(int,
+          maximum_internal_processing_rate,
+          kParameterNotSpecifiedValue,
+          "Set a maximum internal processing rate (32000 or 48000) to override "
+          "the default rate");
+ABSL_FLAG(int,
+          stream_delay,
+          kParameterNotSpecifiedValue,
+          "Specify the stream delay in ms to use");
+ABSL_FLAG(int,
+          use_stream_delay,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) reporting the stream delay");
+ABSL_FLAG(int,
+          stream_drift_samples,
+          kParameterNotSpecifiedValue,
+          "Specify the number of stream drift samples to use");
+ABSL_FLAG(int,
+          initial_mic_level,
+          100,
+          "Initial mic level (0-255) for the analog mic gain simulation in the "
+          "test code");
+ABSL_FLAG(int,
+          simulate_mic_gain,
+          0,
+          "Activate (1) or deactivate(0) the analog mic gain simulation in the "
+          "test code");
+ABSL_FLAG(int,
+          multi_channel_render,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) multi-channel render processing in "
+          "APM pipeline");
+ABSL_FLAG(int,
+          multi_channel_capture,
+          kParameterNotSpecifiedValue,
+          "Activate (1) or deactivate (0) multi-channel capture processing in "
+          "APM pipeline");
+ABSL_FLAG(int,
+          simulated_mic_kind,
+          kParameterNotSpecifiedValue,
+          "Specify which microphone kind to use for microphone simulation");
+ABSL_FLAG(int,
+          override_key_pressed,
+          kParameterNotSpecifiedValue,
+          "Always set to true (1) or to false (0) the key press state. If "
+          "unspecified, false is set with Wav files or, with AEC dumps, the "
+          "recorded event is used.");
+ABSL_FLAG(int,
+          frame_for_sending_capture_output_used_false,
+          kParameterNotSpecifiedValue,
+          "Capture frame index for sending a runtime setting for that the "
+          "capture output is not used.");
+ABSL_FLAG(int,
+          frame_for_sending_capture_output_used_true,
+          kParameterNotSpecifiedValue,
+          "Capture frame index for sending a runtime setting for that the "
+          "capture output is used.");
+ABSL_FLAG(bool, performance_report, false, "Report the APM performance ");
+ABSL_FLAG(std::string,
+          performance_report_output_file,
+          "",
+          "Generate a CSV file with the API call durations");
+ABSL_FLAG(bool, verbose, false, "Produce verbose output");
+ABSL_FLAG(bool,
+          quiet,
+          false,
+          "Avoid producing information about the progress.");
+ABSL_FLAG(bool,
+          bitexactness_report,
+          false,
+          "Report bitexactness for aec dump result reproduction");
+ABSL_FLAG(bool,
+          discard_settings_in_aecdump,
+          false,
+          "Discard any config settings specified in the aec dump");
+ABSL_FLAG(bool,
+          store_intermediate_output,
+          false,
+          "Creates new output files after each init");
+ABSL_FLAG(std::string,
+          custom_call_order_file,
+          "",
+          "Custom process API call order file");
+ABSL_FLAG(std::string,
+          output_custom_call_order_file,
+          "",
+          "Generate custom process API call order file from AEC dump");
+ABSL_FLAG(bool,
+          print_aec_parameter_values,
+          false,
+          "Print parameter values used in AEC in JSON-format");
+ABSL_FLAG(std::string,
+          aec_settings,
+          "",
+          "File in JSON-format with custom AEC settings");
+ABSL_FLAG(bool,
+          dump_data,
+          false,
+          "Dump internal data during the call (requires build flag)");
+ABSL_FLAG(std::string,
+          dump_data_output_dir,
+          "",
+          "Internal data dump output directory");
+ABSL_FLAG(int,
+          dump_set_to_use,
+          kParameterNotSpecifiedValue,
+          "Specifies the dump set to use (if not all the dump sets will "
+          "be used");
+ABSL_FLAG(bool,
+          analyze,
+          false,
+          "Only analyze the call setup behavior (no processing)");
+ABSL_FLAG(float,
+          dump_start_seconds,
+          kParameterNotSpecifiedValue,
+          "Start of when to dump data (seconds).");
+ABSL_FLAG(float,
+          dump_end_seconds,
+          kParameterNotSpecifiedValue,
+          "End of when to dump data (seconds).");
+ABSL_FLAG(int,
+          dump_start_frame,
+          kParameterNotSpecifiedValue,
+          "Start of when to dump data (frames).");
+ABSL_FLAG(int,
+          dump_end_frame,
+          kParameterNotSpecifiedValue,
+          "End of when to dump data (frames).");
+ABSL_FLAG(int,
+          init_to_process,
+          kParameterNotSpecifiedValue,
+          "Init index to process.");
+
+ABSL_FLAG(bool,
+          float_wav_output,
+          false,
+          "Produce floating point wav output files.");
+
+ABSL_FLAG(std::string,
+          force_fieldtrials,
+          "",
+          "Field trials control experimental feature code which can be forced. "
+          "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
+          " will assign the group Enable to field trial WebRTC-FooFeature.");
+
+namespace webrtc {
+namespace test {
+namespace {
+
+const char kUsageDescription[] =
+    "Usage: audioproc_f [options] -i <input.wav>\n"
+    "                   or\n"
+    "       audioproc_f [options] -dump_input <aec_dump>\n"
+    "\n\n"
+    "Command-line tool to simulate a call using the audio "
+    "processing module, either based on wav files or "
+    "protobuf debug dump recordings.\n";
+
+void SetSettingIfSpecified(absl::string_view value,
+                           absl::optional<std::string>* parameter) {
+  if (value.compare("") != 0) {
+    *parameter = std::string(value);
+  }
+}
+
+void SetSettingIfSpecified(int value, absl::optional<int>* parameter) {
+  if (value != kParameterNotSpecifiedValue) {
+    *parameter = value;
+  }
+}
+
+void SetSettingIfSpecified(float value, absl::optional<float>* parameter) {
+  constexpr float kFloatParameterNotSpecifiedValue =
+      kParameterNotSpecifiedValue;
+  if (value != kFloatParameterNotSpecifiedValue) {
+    *parameter = value;
+  }
+}
+
+void SetSettingIfFlagSet(int32_t flag, absl::optional<bool>* parameter) {
+  if (flag == 0) {
+    *parameter = false;
+  } else if (flag == 1) {
+    *parameter = true;
+  }
+}
+
+SimulationSettings CreateSettings() {
+  SimulationSettings settings;
+  if (absl::GetFlag(FLAGS_all_default)) {
+    settings.use_ts = true;
+    settings.use_analog_agc = true;
+    settings.use_ns = true;
+    settings.use_hpf = true;
+    settings.use_agc = true;
+    settings.use_agc2 = false;
+    settings.use_pre_amplifier = false;
+    settings.use_aec = true;
+    settings.use_aecm = false;
+    settings.use_ed = false;
+  }
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_input),
+                        &settings.aec_dump_input_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_output),
+                        &settings.aec_dump_output_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_i), &settings.input_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_o), &settings.output_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_ri),
+                        &settings.reverse_input_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_ro),
+                        &settings.reverse_output_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_artificial_nearend),
+                        &settings.artificial_nearend_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_linear_aec_output),
+                        &settings.linear_aec_output_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_output_num_channels),
+                        &settings.output_num_channels);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_reverse_output_num_channels),
+                        &settings.reverse_output_num_channels);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_output_sample_rate_hz),
+                        &settings.output_sample_rate_hz);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_reverse_output_sample_rate_hz),
+                        &settings.reverse_output_sample_rate_hz);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_aec), &settings.use_aec);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_aecm), &settings.use_aecm);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_ed), &settings.use_ed);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_ed_graph),
+                        &settings.ed_graph_output_filename);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_agc), &settings.use_agc);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_agc2), &settings.use_agc2);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_pre_amplifier),
+                      &settings.use_pre_amplifier);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_capture_level_adjustment),
+                      &settings.use_capture_level_adjustment);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_mic_gain_emulation),
+                      &settings.use_analog_mic_gain_emulation);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_hpf), &settings.use_hpf);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_ns), &settings.use_ns);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_ts), &settings.use_ts);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc),
+                      &settings.use_analog_agc);
+  SetSettingIfFlagSet(
+      absl::GetFlag(FLAGS_analog_agc_use_digital_adaptive_controller),
+      &settings.analog_agc_use_digital_adaptive_controller);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_mode), &settings.agc_mode);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_target_level),
+                        &settings.agc_target_level);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_agc_limiter),
+                      &settings.use_agc_limiter);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_compression_gain),
+                        &settings.agc_compression_gain);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_agc2_enable_adaptive_gain),
+                      &settings.agc2_use_adaptive_gain);
+
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_agc2_fixed_gain_db),
+                        &settings.agc2_fixed_gain_db);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_pre_amplifier_gain_factor),
+                        &settings.pre_amplifier_gain_factor);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_pre_gain_factor),
+                        &settings.pre_gain_factor);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_post_gain_factor),
+                        &settings.post_gain_factor);
+  SetSettingIfSpecified(
+      absl::GetFlag(FLAGS_analog_mic_gain_emulation_initial_level),
+      &settings.analog_mic_gain_emulation_initial_level);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_ns_level), &settings.ns_level);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_ns_analysis_on_linear_aec_output),
+                      &settings.ns_analysis_on_linear_aec_output);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_maximum_internal_processing_rate),
+                        &settings.maximum_internal_processing_rate);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_stream_delay),
+                        &settings.stream_delay);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_use_stream_delay),
+                      &settings.use_stream_delay);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_custom_call_order_file),
+                        &settings.call_order_input_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_output_custom_call_order_file),
+                        &settings.call_order_output_filename);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_aec_settings),
+                        &settings.aec_settings_filename);
+  settings.initial_mic_level = absl::GetFlag(FLAGS_initial_mic_level);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_multi_channel_render),
+                      &settings.multi_channel_render);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_multi_channel_capture),
+                      &settings.multi_channel_capture);
+  settings.simulate_mic_gain = absl::GetFlag(FLAGS_simulate_mic_gain);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_simulated_mic_kind),
+                        &settings.simulated_mic_kind);
+  SetSettingIfFlagSet(absl::GetFlag(FLAGS_override_key_pressed),
+                      &settings.override_key_pressed);
+  SetSettingIfSpecified(
+      absl::GetFlag(FLAGS_frame_for_sending_capture_output_used_false),
+      &settings.frame_for_sending_capture_output_used_false);
+  SetSettingIfSpecified(
+      absl::GetFlag(FLAGS_frame_for_sending_capture_output_used_true),
+      &settings.frame_for_sending_capture_output_used_true);
+  settings.report_performance = absl::GetFlag(FLAGS_performance_report);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_performance_report_output_file),
+                        &settings.performance_report_output_filename);
+  settings.use_verbose_logging = absl::GetFlag(FLAGS_verbose);
+  settings.use_quiet_output = absl::GetFlag(FLAGS_quiet);
+  settings.report_bitexactness = absl::GetFlag(FLAGS_bitexactness_report);
+  settings.discard_all_settings_in_aecdump =
+      absl::GetFlag(FLAGS_discard_settings_in_aecdump);
+  settings.fixed_interface = absl::GetFlag(FLAGS_fixed_interface);
+  settings.store_intermediate_output =
+      absl::GetFlag(FLAGS_store_intermediate_output);
+  settings.print_aec_parameter_values =
+      absl::GetFlag(FLAGS_print_aec_parameter_values);
+  settings.dump_internal_data = absl::GetFlag(FLAGS_dump_data);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_data_output_dir),
+                        &settings.dump_internal_data_output_dir);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_set_to_use),
+                        &settings.dump_set_to_use);
+  settings.wav_output_format = absl::GetFlag(FLAGS_float_wav_output)
+                                   ? WavFile::SampleFormat::kFloat
+                                   : WavFile::SampleFormat::kInt16;
+
+  settings.analysis_only = absl::GetFlag(FLAGS_analyze);
+
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_start_frame),
+                        &settings.dump_start_frame);
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_end_frame),
+                        &settings.dump_end_frame);
+
+  constexpr int kFramesPerSecond = 100;
+  absl::optional<float> start_seconds;
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_start_seconds),
+                        &start_seconds);
+  if (start_seconds) {
+    settings.dump_start_frame = *start_seconds * kFramesPerSecond;
+  }
+
+  absl::optional<float> end_seconds;
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_dump_end_seconds), &end_seconds);
+  if (end_seconds) {
+    settings.dump_end_frame = *end_seconds * kFramesPerSecond;
+  }
+
+  SetSettingIfSpecified(absl::GetFlag(FLAGS_init_to_process),
+                        &settings.init_to_process);
+
+  return settings;
+}
+
+void ReportConditionalErrorAndExit(bool condition, absl::string_view message) {
+  if (condition) {
+    std::cerr << message << std::endl;
+    exit(1);
+  }
+}
+
+void PerformBasicParameterSanityChecks(
+    const SimulationSettings& settings,
+    bool pre_constructed_ap_provided,
+    bool pre_constructed_ap_builder_provided) {
+  if (settings.input_filename || settings.reverse_input_filename) {
+    ReportConditionalErrorAndExit(
+        !!settings.aec_dump_input_filename,
+        "Error: The aec dump file cannot be specified "
+        "together with input wav files!\n");
+
+    ReportConditionalErrorAndExit(
+        !!settings.aec_dump_input_string,
+        "Error: The aec dump input string cannot be specified "
+        "together with input wav files!\n");
+
+    ReportConditionalErrorAndExit(!!settings.artificial_nearend_filename,
+                                  "Error: The artificial nearend cannot be "
+                                  "specified together with input wav files!\n");
+
+    ReportConditionalErrorAndExit(!settings.input_filename,
+                                  "Error: When operating at wav files, the "
+                                  "input wav filename must be "
+                                  "specified!\n");
+
+    ReportConditionalErrorAndExit(
+        settings.reverse_output_filename && !settings.reverse_input_filename,
+        "Error: When operating at wav files, the reverse input wav filename "
+        "must be specified if the reverse output wav filename is specified!\n");
+  } else {
+    ReportConditionalErrorAndExit(
+        !settings.aec_dump_input_filename && !settings.aec_dump_input_string,
+        "Error: Either the aec dump input file, the wav "
+        "input file or the aec dump input string must be specified!\n");
+    ReportConditionalErrorAndExit(
+        settings.aec_dump_input_filename && settings.aec_dump_input_string,
+        "Error: The aec dump input file cannot be specified together with the "
+        "aec dump input string!\n");
+  }
+
+  ReportConditionalErrorAndExit(settings.use_aec && !(*settings.use_aec) &&
+                                    settings.linear_aec_output_filename,
+                                "Error: The linear AEC ouput filename cannot "
+                                "be specified without the AEC being active");
+
+  ReportConditionalErrorAndExit(
+      settings.use_aec && *settings.use_aec && settings.use_aecm &&
+          *settings.use_aecm,
+      "Error: The AEC and the AECM cannot be activated at the same time!\n");
+
+  ReportConditionalErrorAndExit(
+      settings.output_sample_rate_hz && *settings.output_sample_rate_hz <= 0,
+      "Error: --output_sample_rate_hz must be positive!\n");
+
+  ReportConditionalErrorAndExit(
+      settings.reverse_output_sample_rate_hz &&
+          settings.output_sample_rate_hz &&
+          *settings.output_sample_rate_hz <= 0,
+      "Error: --reverse_output_sample_rate_hz must be positive!\n");
+
+  ReportConditionalErrorAndExit(
+      settings.output_num_channels && *settings.output_num_channels <= 0,
+      "Error: --output_num_channels must be positive!\n");
+
+  ReportConditionalErrorAndExit(
+      settings.reverse_output_num_channels &&
+          *settings.reverse_output_num_channels <= 0,
+      "Error: --reverse_output_num_channels must be positive!\n");
+
+  ReportConditionalErrorAndExit(
+      settings.agc_target_level && ((*settings.agc_target_level) < 0 ||
+                                    (*settings.agc_target_level) > 31),
+      "Error: --agc_target_level must be specified between 0 and 31.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.agc_compression_gain && ((*settings.agc_compression_gain) < 0 ||
+                                        (*settings.agc_compression_gain) > 90),
+      "Error: --agc_compression_gain must be specified between 0 and 90.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.agc2_fixed_gain_db && ((*settings.agc2_fixed_gain_db) < 0 ||
+                                      (*settings.agc2_fixed_gain_db) > 90),
+      "Error: --agc2_fixed_gain_db must be specified between 0 and 90.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.ns_level &&
+          ((*settings.ns_level) < 0 || (*settings.ns_level) > 3),
+      "Error: --ns_level must be specified between 0 and 3.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.report_bitexactness && !settings.aec_dump_input_filename,
+      "Error: --bitexactness_report can only be used when operating on an "
+      "aecdump\n");
+
+  ReportConditionalErrorAndExit(
+      settings.call_order_input_filename && settings.aec_dump_input_filename,
+      "Error: --custom_call_order_file cannot be used when operating on an "
+      "aecdump\n");
+
+  ReportConditionalErrorAndExit(
+      (settings.initial_mic_level < 0 || settings.initial_mic_level > 255),
+      "Error: --initial_mic_level must be specified between 0 and 255.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.simulated_mic_kind && !settings.simulate_mic_gain,
+      "Error: --simulated_mic_kind cannot be specified when mic simulation is "
+      "disabled\n");
+
+  ReportConditionalErrorAndExit(
+      !settings.simulated_mic_kind && settings.simulate_mic_gain,
+      "Error: --simulated_mic_kind must be specified when mic simulation is "
+      "enabled\n");
+
+  // TODO(bugs.webrtc.org/7494): Document how the two settings below differ.
+  ReportConditionalErrorAndExit(
+      settings.simulate_mic_gain && settings.use_analog_mic_gain_emulation,
+      "Error: --simulate_mic_gain and --use_analog_mic_gain_emulation cannot "
+      "be enabled at the same time\n");
+
+  auto valid_wav_name = [](absl::string_view wav_file_name) {
+    if (wav_file_name.size() < 5) {
+      return false;
+    }
+    if ((wav_file_name.compare(wav_file_name.size() - 4, 4, ".wav") == 0) ||
+        (wav_file_name.compare(wav_file_name.size() - 4, 4, ".WAV") == 0)) {
+      return true;
+    }
+    return false;
+  };
+
+  ReportConditionalErrorAndExit(
+      settings.input_filename && (!valid_wav_name(*settings.input_filename)),
+      "Error: --i must be a valid .wav file name.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.output_filename && (!valid_wav_name(*settings.output_filename)),
+      "Error: --o must be a valid .wav file name.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.reverse_input_filename &&
+          (!valid_wav_name(*settings.reverse_input_filename)),
+      "Error: --ri must be a valid .wav file name.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.reverse_output_filename &&
+          (!valid_wav_name(*settings.reverse_output_filename)),
+      "Error: --ro must be a valid .wav file name.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.artificial_nearend_filename &&
+          !valid_wav_name(*settings.artificial_nearend_filename),
+      "Error: --artifical_nearend must be a valid .wav file name.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.linear_aec_output_filename &&
+          (!valid_wav_name(*settings.linear_aec_output_filename)),
+      "Error: --linear_aec_output must be a valid .wav file name.\n");
+
+  ReportConditionalErrorAndExit(
+      WEBRTC_APM_DEBUG_DUMP == 0 && settings.dump_internal_data,
+      "Error: --dump_data cannot be set without proper build support.\n");
+
+  ReportConditionalErrorAndExit(settings.init_to_process &&
+                                    *settings.init_to_process != 1 &&
+                                    !settings.aec_dump_input_filename,
+                                "Error: --init_to_process must be set to 1 for "
+                                "wav-file based simulations.\n");
+
+  ReportConditionalErrorAndExit(
+      !settings.init_to_process &&
+          (settings.dump_start_frame || settings.dump_end_frame),
+      "Error: --init_to_process must be set when specifying a start and/or end "
+      "frame for when to dump internal data.\n");
+
+  ReportConditionalErrorAndExit(
+      !settings.dump_internal_data &&
+          settings.dump_internal_data_output_dir.has_value(),
+      "Error: --dump_data_output_dir cannot be set without --dump_data.\n");
+
+  ReportConditionalErrorAndExit(
+      !settings.aec_dump_input_filename &&
+          settings.call_order_output_filename.has_value(),
+      "Error: --output_custom_call_order_file needs an AEC dump input file.\n");
+
+  ReportConditionalErrorAndExit(
+      (!settings.use_pre_amplifier || !(*settings.use_pre_amplifier)) &&
+          settings.pre_amplifier_gain_factor.has_value(),
+      "Error: --pre_amplifier_gain_factor needs --pre_amplifier to be "
+      "specified and set.\n");
+
+  ReportConditionalErrorAndExit(
+      pre_constructed_ap_provided && pre_constructed_ap_builder_provided,
+      "Error: The AudioProcessing and the AudioProcessingBuilder cannot both "
+      "be specified at the same time.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.aec_settings_filename && pre_constructed_ap_provided,
+      "Error: The aec_settings_filename cannot be specified when a "
+      "pre-constructed audio processing object is provided.\n");
+
+  ReportConditionalErrorAndExit(
+      settings.aec_settings_filename && pre_constructed_ap_provided,
+      "Error: The print_aec_parameter_values cannot be set when a "
+      "pre-constructed audio processing object is provided.\n");
+
+  if (settings.linear_aec_output_filename && pre_constructed_ap_provided) {
+    std::cout << "Warning: For the linear AEC output to be stored, this must "
+                 "be configured in the AEC that is part of the provided "
+                 "AudioProcessing object."
+              << std::endl;
+  }
+}
+
+int RunSimulation(rtc::scoped_refptr<AudioProcessing> audio_processing,
+                  std::unique_ptr<AudioProcessingBuilder> ap_builder,
+                  int argc,
+                  char* argv[],
+                  absl::string_view input_aecdump,
+                  std::vector<float>* processed_capture_samples) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  if (args.size() != 1) {
+    printf("%s", kUsageDescription);
+    return 1;
+  }
+  // InitFieldTrialsFromString stores the char*, so the char array must
+  // outlive the application.
+  const std::string field_trials = absl::GetFlag(FLAGS_force_fieldtrials);
+  webrtc::field_trial::InitFieldTrialsFromString(field_trials.c_str());
+
+  SimulationSettings settings = CreateSettings();
+  if (!input_aecdump.empty()) {
+    settings.aec_dump_input_string = input_aecdump;
+    settings.processed_capture_samples = processed_capture_samples;
+    RTC_CHECK(settings.processed_capture_samples);
+  }
+  PerformBasicParameterSanityChecks(settings, !!audio_processing, !!ap_builder);
+  std::unique_ptr<AudioProcessingSimulator> processor;
+
+  if (settings.aec_dump_input_filename || settings.aec_dump_input_string) {
+    processor.reset(new AecDumpBasedSimulator(
+        settings, std::move(audio_processing), std::move(ap_builder)));
+  } else {
+    processor.reset(new WavBasedSimulator(settings, std::move(audio_processing),
+                                          std::move(ap_builder)));
+  }
+
+  if (settings.analysis_only) {
+    processor->Analyze();
+  } else {
+    processor->Process();
+  }
+
+  if (settings.report_performance) {
+    processor->GetApiCallStatistics().PrintReport();
+  }
+  if (settings.performance_report_output_filename) {
+    processor->GetApiCallStatistics().WriteReportToFile(
+        *settings.performance_report_output_filename);
+  }
+
+  if (settings.report_bitexactness && settings.aec_dump_input_filename) {
+    if (processor->OutputWasBitexact()) {
+      std::cout << "The processing was bitexact.";
+    } else {
+      std::cout << "The processing was not bitexact.";
+    }
+  }
+
+  return 0;
+}
+
+}  // namespace
+
+int AudioprocFloatImpl(rtc::scoped_refptr<AudioProcessing> audio_processing,
+                       int argc,
+                       char* argv[]) {
+  return RunSimulation(
+      std::move(audio_processing), /*ap_builder=*/nullptr, argc, argv,
+      /*input_aecdump=*/"", /*processed_capture_samples=*/nullptr);
+}
+
+int AudioprocFloatImpl(std::unique_ptr<AudioProcessingBuilder> ap_builder,
+                       int argc,
+                       char* argv[],
+                       absl::string_view input_aecdump,
+                       std::vector<float>* processed_capture_samples) {
+  return RunSimulation(/*audio_processing=*/nullptr, std::move(ap_builder),
+                       argc, argv, input_aecdump, processed_capture_samples);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.h b/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.h
new file mode 100644
index 0000000000..5ed3aefab7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/audioproc_float_impl.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_
+
+#include <memory>
+
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+namespace test {
+
+// This function implements the audio processing simulation utility. Pass
+// `input_aecdump` to provide the content of an AEC dump file as a string; if
+// `input_aecdump` is not passed, a WAV or AEC input dump file must be specified
+// via the `argv` argument. Pass `processed_capture_samples` to write in it the
+// samples processed on the capture side; if `processed_capture_samples` is not
+// passed, the output file can optionally be specified via the `argv` argument.
+// Any audio_processing object specified in the input is used for the
+// simulation. Note that when the audio_processing object is specified all
+// functionality that relies on using the internal builder is deactivated,
+// since the AudioProcessing object is already created and the builder is not
+// used in the simulation.
+int AudioprocFloatImpl(rtc::scoped_refptr<AudioProcessing> audio_processing,
+                       int argc,
+                       char* argv[]);
+
+// This function implements the audio processing simulation utility. Pass
+// `input_aecdump` to provide the content of an AEC dump file as a string; if
+// `input_aecdump` is not passed, a WAV or AEC input dump file must be specified
+// via the `argv` argument. Pass `processed_capture_samples` to write in it the
+// samples processed on the capture side; if `processed_capture_samples` is not
+// passed, the output file can optionally be specified via the `argv` argument.
+int AudioprocFloatImpl(std::unique_ptr<AudioProcessingBuilder> ap_builder,
+                       int argc,
+                       char* argv[],
+                       absl::string_view input_aecdump,
+                       std::vector<float>* processed_capture_samples);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/bitexactness_tools.cc b/third_party/libwebrtc/modules/audio_processing/test/bitexactness_tools.cc
new file mode 100644
index 0000000000..0464345364
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/bitexactness_tools.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/bitexactness_tools.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <string>
+#include <vector>
+
+#include "api/array_view.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+
+std::string GetApmRenderTestVectorFileName(int sample_rate_hz) {
+  switch (sample_rate_hz) {
+    case 8000:
+      return ResourcePath("far8_stereo", "pcm");
+    case 16000:
+      return ResourcePath("far16_stereo", "pcm");
+    case 32000:
+      return ResourcePath("far32_stereo", "pcm");
+    case 48000:
+      return ResourcePath("far48_stereo", "pcm");
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  return "";
+}
+
+std::string GetApmCaptureTestVectorFileName(int sample_rate_hz) {
+  switch (sample_rate_hz) {
+    case 8000:
+      return ResourcePath("near8_stereo", "pcm");
+    case 16000:
+      return ResourcePath("near16_stereo", "pcm");
+    case 32000:
+      return ResourcePath("near32_stereo", "pcm");
+    case 48000:
+      return ResourcePath("near48_stereo", "pcm");
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  return "";
+}
+
+void ReadFloatSamplesFromStereoFile(size_t samples_per_channel,
+                                    size_t num_channels,
+                                    InputAudioFile* stereo_pcm_file,
+                                    rtc::ArrayView<float> data) {
+  RTC_DCHECK_LE(num_channels, 2);
+  RTC_DCHECK_EQ(data.size(), samples_per_channel * num_channels);
+  std::vector<int16_t> read_samples(samples_per_channel * 2);
+  stereo_pcm_file->Read(samples_per_channel * 2, read_samples.data());
+
+  // Convert samples to float and discard any channels not needed.
+  for (size_t sample = 0; sample < samples_per_channel; ++sample) {
+    for (size_t channel = 0; channel < num_channels; ++channel) {
+      data[sample * num_channels + channel] =
+          read_samples[sample * 2 + channel] / 32768.0f;
+    }
+  }
+}
+
+::testing::AssertionResult VerifyDeinterleavedArray(
+    size_t samples_per_channel,
+    size_t num_channels,
+    rtc::ArrayView<const float> reference,
+    rtc::ArrayView<const float> output,
+    float element_error_bound) {
+  // Form vectors to compare the reference to. Only the first values of the
+  // outputs are compared in order not having to specify all preceeding frames
+  // as testvectors.
+  const size_t reference_frame_length =
+      rtc::CheckedDivExact(reference.size(), num_channels);
+
+  std::vector<float> output_to_verify;
+  for (size_t channel_no = 0; channel_no < num_channels; ++channel_no) {
+    output_to_verify.insert(output_to_verify.end(),
+                            output.begin() + channel_no * samples_per_channel,
+                            output.begin() + channel_no * samples_per_channel +
+                                reference_frame_length);
+  }
+
+  return VerifyArray(reference, output_to_verify, element_error_bound);
+}
+
+::testing::AssertionResult VerifyArray(rtc::ArrayView<const float> reference,
+                                       rtc::ArrayView<const float> output,
+                                       float element_error_bound) {
+  // The vectors are deemed to be bitexact only if
+  // a) output have a size at least as long as the reference.
+  // b) the samples in the reference are bitexact with the corresponding samples
+  //    in the output.
+
+  bool equal = true;
+  if (output.size() < reference.size()) {
+    equal = false;
+  } else {
+    // Compare the first samples in the vectors.
+    for (size_t k = 0; k < reference.size(); ++k) {
+      if (fabs(output[k] - reference[k]) > element_error_bound) {
+        equal = false;
+        break;
+      }
+    }
+  }
+
+  if (equal) {
+    return ::testing::AssertionSuccess();
+  }
+
+  // Lambda function that produces a formatted string with the data in the
+  // vector.
+  auto print_vector_in_c_format = [](rtc::ArrayView<const float> v,
+                                     size_t num_values_to_print) {
+    std::string s = "{ ";
+    for (size_t k = 0; k < std::min(num_values_to_print, v.size()); ++k) {
+      s += std::to_string(v[k]) + "f";
+      s += (k < (num_values_to_print - 1)) ? ", " : "";
+    }
+    return s + " }";
+  };
+
+  // If the vectors are deemed not to be similar, return a report of the
+  // difference.
+  return ::testing::AssertionFailure()
+         << std::endl
+         << "    Actual values : "
+         << print_vector_in_c_format(output,
+                                     std::min(output.size(), reference.size()))
+         << std::endl
+         << "    Expected values: "
+         << print_vector_in_c_format(reference, reference.size()) << std::endl;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/bitexactness_tools.h b/third_party/libwebrtc/modules/audio_processing/test/bitexactness_tools.h
new file mode 100644
index 0000000000..2d3113276d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/bitexactness_tools.h
@@ -0,0 +1,56 @@
+
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_BITEXACTNESS_TOOLS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_BITEXACTNESS_TOOLS_H_
+
+#include <string>
+
+#include "api/array_view.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+// Returns test vector to use for the render signal in an
+// APM bitexactness test.
+std::string GetApmRenderTestVectorFileName(int sample_rate_hz);
+
+// Returns test vector to use for the capture signal in an
+// APM bitexactness test.
+std::string GetApmCaptureTestVectorFileName(int sample_rate_hz);
+
+// Extract float samples of up to two channels from a pcm file.
+void ReadFloatSamplesFromStereoFile(size_t samples_per_channel,
+                                    size_t num_channels,
+                                    InputAudioFile* stereo_pcm_file,
+                                    rtc::ArrayView<float> data);
+
+// Verifies a frame against a reference and returns the results as an
+// AssertionResult.
+::testing::AssertionResult VerifyDeinterleavedArray(
+    size_t samples_per_channel,
+    size_t num_channels,
+    rtc::ArrayView<const float> reference,
+    rtc::ArrayView<const float> output,
+    float element_error_bound);
+
+// Verifies a vector against a reference and returns the results as an
+// AssertionResult.
+::testing::AssertionResult VerifyArray(rtc::ArrayView<const float> reference,
+                                       rtc::ArrayView<const float> output,
+                                       float element_error_bound);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_BITEXACTNESS_TOOLS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
new file mode 100644
index 0000000000..2c3678092e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -0,0 +1,81 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../../webrtc.gni")
+
+if (!build_with_chromium) {
+  group("conversational_speech") {
+    testonly = true
+    deps = [ ":conversational_speech_generator" ]
+  }
+
+  rtc_executable("conversational_speech_generator") {
+    testonly = true
+    sources = [ "generator.cc" ]
+    deps = [
+      ":lib",
+      "../../../../test:fileutils",
+      "../../../../test:test_support",
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+    ]
+  }
+}
+
+rtc_library("lib") {
+  testonly = true
+  sources = [
+    "config.cc",
+    "config.h",
+    "multiend_call.cc",
+    "multiend_call.h",
+    "simulator.cc",
+    "simulator.h",
+    "timing.cc",
+    "timing.h",
+    "wavreader_abstract_factory.h",
+    "wavreader_factory.cc",
+    "wavreader_factory.h",
+    "wavreader_interface.h",
+  ]
+  deps = [
+    "../../../../api:array_view",
+    "../../../../common_audio",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:logging",
+    "../../../../rtc_base:safe_conversions",
+    "../../../../rtc_base:stringutils",
+    "../../../../test:fileutils",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  visibility = [ ":*" ]  # Only targets in this file can depend on this.
+}
+
+rtc_library("unittest") {
+  testonly = true
+  sources = [
+    "generator_unittest.cc",
+    "mock_wavreader.cc",
+    "mock_wavreader.h",
+    "mock_wavreader_factory.cc",
+    "mock_wavreader_factory.h",
+  ]
+  deps = [
+    ":lib",
+    "../../../../api:array_view",
+    "../../../../common_audio",
+    "../../../../rtc_base:logging",
+    "../../../../test:fileutils",
+    "../../../../test:test_support",
+    "//testing/gtest",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/OWNERS b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/OWNERS
new file mode 100644
index 0000000000..07cff405e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/OWNERS
@@ -0,0 +1,3 @@
+alessiob@webrtc.org
+henrik.lundin@webrtc.org
+peah@webrtc.org
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/README.md b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/README.md
new file mode 100644
index 0000000000..0fa66669e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/README.md
@@ -0,0 +1,74 @@
+# Conversational Speech generator tool
+
+Tool to generate multiple-end audio tracks to simulate conversational speech
+with two or more participants.
+
+The input to the tool is a directory containing a number of audio tracks and
+a text file indicating how to time the sequence of speech turns (see the Example
+section).
+
+Since the timing of the speaking turns is specified by the user, the generated
+tracks may not be suitable for testing scenarios in which there is unpredictable
+network delay (e.g., end-to-end RTC assessment).
+
+Instead, the generated pairs can be used when the delay is constant (obviously
+including the case in which there is no delay).
+For instance, echo cancellation in the APM module can be evaluated using two-end
+audio tracks as input and reverse input.
+
+By indicating negative and positive time offsets, one can reproduce cross-talk
+(aka double-talk) and silence in the conversation.
+
+### Example
+
+For each end, there is a set of audio tracks, e.g., a1, a2 and a3 (speaker A)
+and b1, b2 (speaker B).
+The text file with the timing information may look like this:
+
+```
+A a1 0
+B b1 0
+A a2 100
+B b2 -200
+A a3 0
+A a4 0
+```
+
+The first column indicates the speaker name, the second contains the audio track
+file names, and the third the offsets (in milliseconds) used to concatenate the
+chunks. An optional fourth column contains positive or negative integral gains
+in dB that will be applied to the tracks. It's possible to specify the gain for
+some turns but not for others. If the gain is left out, no gain is applied.
+
+Assume that all the audio tracks in the example above are 1000 ms long.
+The tool will then generate two tracks (A and B) that look like this:
+
+**Track A**
+```
+  a1 (1000 ms)
+  silence (1100 ms)
+  a2 (1000 ms)
+  silence (800 ms)
+  a3 (1000 ms)
+  a4 (1000 ms)
+```
+
+**Track B**
+```
+  silence (1000 ms)
+  b1 (1000 ms)
+  silence (900 ms)
+  b2 (1000 ms)
+  silence (2000 ms)
+```
+
+The two tracks can be also visualized as follows (one characheter represents
+100 ms, "." is silence and "*" is speech).
+
+```
+t: 0         1        2        3        4        5        6 (s)
+A: **********...........**********........********************
+B: ..........**********.........**********....................
+                                ^ 200 ms cross-talk
+        100 ms silence ^
+```
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/config.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/config.cc
new file mode 100644
index 0000000000..76d3de8108
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/config.cc
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/conversational_speech/config.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+const std::string& Config::audiotracks_path() const {
+  return audiotracks_path_;
+}
+
+const std::string& Config::timing_filepath() const {
+  return timing_filepath_;
+}
+
+const std::string& Config::output_path() const {
+  return output_path_;
+}
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/config.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/config.h
new file mode 100644
index 0000000000..5a847e06a2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/config.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_CONFIG_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_CONFIG_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+struct Config {
+  Config(absl::string_view audiotracks_path,
+         absl::string_view timing_filepath,
+         absl::string_view output_path)
+      : audiotracks_path_(audiotracks_path),
+        timing_filepath_(timing_filepath),
+        output_path_(output_path) {}
+
+  const std::string& audiotracks_path() const;
+  const std::string& timing_filepath() const;
+  const std::string& output_path() const;
+
+  const std::string audiotracks_path_;
+  const std::string timing_filepath_;
+  const std::string output_path_;
+};
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_CONFIG_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/generator.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/generator.cc
new file mode 100644
index 0000000000..d0bc2f2319
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/generator.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <iostream>
+#include <vector>
+
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "modules/audio_processing/test/conversational_speech/config.h"
+#include "modules/audio_processing/test/conversational_speech/multiend_call.h"
+#include "modules/audio_processing/test/conversational_speech/simulator.h"
+#include "modules/audio_processing/test/conversational_speech/timing.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_factory.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(std::string, i, "", "Directory containing the speech turn wav files");
+ABSL_FLAG(std::string, t, "", "Path to the timing text file");
+ABSL_FLAG(std::string, o, "", "Output wav files destination path");
+
+namespace webrtc {
+namespace test {
+namespace {
+
+const char kUsageDescription[] =
+    "Usage: conversational_speech_generator\n"
+    "          -i <path/to/source/audiotracks>\n"
+    "          -t <path/to/timing_file.txt>\n"
+    "          -o <output/path>\n"
+    "\n\n"
+    "Command-line tool to generate multiple-end audio tracks to simulate "
+    "conversational speech with two or more participants.\n";
+
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  if (args.size() != 1) {
+    printf("%s", kUsageDescription);
+    return 1;
+  }
+  RTC_CHECK(DirExists(absl::GetFlag(FLAGS_i)));
+  RTC_CHECK(FileExists(absl::GetFlag(FLAGS_t)));
+  RTC_CHECK(DirExists(absl::GetFlag(FLAGS_o)));
+
+  conversational_speech::Config config(
+      absl::GetFlag(FLAGS_i), absl::GetFlag(FLAGS_t), absl::GetFlag(FLAGS_o));
+
+  // Load timing.
+  std::vector<conversational_speech::Turn> timing =
+      conversational_speech::LoadTiming(config.timing_filepath());
+
+  // Parse timing and audio tracks.
+  auto wavreader_factory =
+      std::make_unique<conversational_speech::WavReaderFactory>();
+  conversational_speech::MultiEndCall multiend_call(
+      timing, config.audiotracks_path(), std::move(wavreader_factory));
+
+  // Generate output audio tracks.
+  auto generated_audiotrack_pairs =
+      conversational_speech::Simulate(multiend_call, config.output_path());
+
+  // Show paths to created audio tracks.
+  std::cout << "Output files:" << std::endl;
+  for (const auto& output_paths_entry : *generated_audiotrack_pairs) {
+    std::cout << "  speaker: " << output_paths_entry.first << std::endl;
+    std::cout << "    near end: " << output_paths_entry.second.near_end
+              << std::endl;
+    std::cout << "    far end: " << output_paths_entry.second.far_end
+              << std::endl;
+  }
+
+  return 0;
+}
+
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::main(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
new file mode 100644
index 0000000000..17714440d4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
@@ -0,0 +1,675 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file consists of unit tests for webrtc::test::conversational_speech
+// members. Part of them focus on accepting or rejecting different
+// conversational speech setups. A setup is defined by a set of audio tracks and
+// timing information).
+// The docstring at the beginning of each TEST(ConversationalSpeechTest,
+// MultiEndCallSetup*) function looks like the drawing below and indicates which
+// setup is tested.
+//
+//    Accept:
+//    A 0****.....
+//    B .....1****
+//
+// The drawing indicates the following:
+// - the illustrated setup should be accepted,
+// - there are two speakers (namely, A and B),
+// - A is the first speaking, B is the second one,
+// - each character after the speaker's letter indicates a time unit (e.g., 100
+//   ms),
+// - "*" indicates speaking, "." listening,
+// - numbers indicate the turn index in std::vector<Turn>.
+//
+// Note that the same speaker can appear in multiple lines in order to depict
+// cases in which there are wrong offsets leading to self cross-talk (which is
+// rejected).
+
+// MSVC++ requires this to be set before any other includes to get M_PI.
+#define _USE_MATH_DEFINES
+
+#include <stdio.h>
+
+#include <cmath>
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/test/conversational_speech/config.h"
+#include "modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
+#include "modules/audio_processing/test/conversational_speech/multiend_call.h"
+#include "modules/audio_processing/test/conversational_speech/simulator.h"
+#include "modules/audio_processing/test/conversational_speech/timing.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_factory.h"
+#include "rtc_base/logging.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using conversational_speech::LoadTiming;
+using conversational_speech::MockWavReaderFactory;
+using conversational_speech::MultiEndCall;
+using conversational_speech::SaveTiming;
+using conversational_speech::Turn;
+using conversational_speech::WavReaderFactory;
+
+const char* const audiotracks_path = "/path/to/audiotracks";
+const char* const timing_filepath = "/path/to/timing_file.txt";
+const char* const output_path = "/path/to/output_dir";
+
+const std::vector<Turn> expected_timing = {
+    {"A", "a1", 0, 0},    {"B", "b1", 0, 0}, {"A", "a2", 100, 0},
+    {"B", "b2", -200, 0}, {"A", "a3", 0, 0}, {"A", "a3", 0, 0},
+};
+const std::size_t kNumberOfTurns = expected_timing.size();
+
+// Default arguments for MockWavReaderFactory ctor.
+// Fake audio track parameters.
+constexpr int kDefaultSampleRate = 48000;
+const std::map<std::string, const MockWavReaderFactory::Params>
+    kDefaultMockWavReaderFactoryParamsMap = {
+        {"t300", {kDefaultSampleRate, 1u, 14400u}},   // Mono, 0.3 seconds.
+        {"t500", {kDefaultSampleRate, 1u, 24000u}},   // Mono, 0.5 seconds.
+        {"t1000", {kDefaultSampleRate, 1u, 48000u}},  // Mono, 1.0 seconds.
+        {"sr8000", {8000, 1u, 8000u}},     // 8kHz sample rate, mono, 1 second.
+        {"sr16000", {16000, 1u, 16000u}},  // 16kHz sample rate, mono, 1 second.
+        {"sr16000_stereo", {16000, 2u, 16000u}},  // Like sr16000, but stereo.
+};
+const MockWavReaderFactory::Params& kDefaultMockWavReaderFactoryParams =
+    kDefaultMockWavReaderFactoryParamsMap.at("t500");
+
+std::unique_ptr<MockWavReaderFactory> CreateMockWavReaderFactory() {
+  return std::unique_ptr<MockWavReaderFactory>(
+      new MockWavReaderFactory(kDefaultMockWavReaderFactoryParams,
+                               kDefaultMockWavReaderFactoryParamsMap));
+}
+
+void CreateSineWavFile(absl::string_view filepath,
+                       const MockWavReaderFactory::Params& params,
+                       float frequency = 440.0f) {
+  // Create samples.
+  constexpr double two_pi = 2.0 * M_PI;
+  std::vector<int16_t> samples(params.num_samples);
+  for (std::size_t i = 0; i < params.num_samples; ++i) {
+    // TODO(alessiob): the produced tone is not pure, improve.
+    samples[i] = std::lround(
+        32767.0f * std::sin(two_pi * i * frequency / params.sample_rate));
+  }
+
+  // Write samples.
+  WavWriter wav_writer(filepath, params.sample_rate, params.num_channels);
+  wav_writer.WriteSamples(samples.data(), params.num_samples);
+}
+
+// Parameters to generate audio tracks with CreateSineWavFile.
+struct SineAudioTrackParams {
+  MockWavReaderFactory::Params params;
+  float frequency;
+};
+
+// Creates a temporary directory in which sine audio tracks are written.
+std::string CreateTemporarySineAudioTracks(
+    const std::map<std::string, SineAudioTrackParams>& sine_tracks_params) {
+  // Create temporary directory.
+  std::string temp_directory =
+      OutputPath() + "TempConversationalSpeechAudioTracks";
+  CreateDir(temp_directory);
+
+  // Create sine tracks.
+  for (const auto& it : sine_tracks_params) {
+    const std::string temp_filepath = JoinFilename(temp_directory, it.first);
+    CreateSineWavFile(temp_filepath, it.second.params, it.second.frequency);
+  }
+
+  return temp_directory;
+}
+
+void CheckAudioTrackParams(const WavReaderFactory& wav_reader_factory,
+                           absl::string_view filepath,
+                           const MockWavReaderFactory::Params& expeted_params) {
+  auto wav_reader = wav_reader_factory.Create(filepath);
+  EXPECT_EQ(expeted_params.sample_rate, wav_reader->SampleRate());
+  EXPECT_EQ(expeted_params.num_channels, wav_reader->NumChannels());
+  EXPECT_EQ(expeted_params.num_samples, wav_reader->NumSamples());
+}
+
+void DeleteFolderAndContents(absl::string_view dir) {
+  if (!DirExists(dir)) {
+    return;
+  }
+  absl::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
+  EXPECT_TRUE(dir_content);
+  for (const auto& path : *dir_content) {
+    if (DirExists(path)) {
+      DeleteFolderAndContents(path);
+    } else if (FileExists(path)) {
+      // TODO(alessiob): Wrap with EXPECT_TRUE() once webrtc:7769 bug fixed.
+      RemoveFile(path);
+    } else {
+      FAIL();
+    }
+  }
+  // TODO(alessiob): Wrap with EXPECT_TRUE() once webrtc:7769 bug fixed.
+  RemoveDir(dir);
+}
+
+}  // namespace
+
+using ::testing::_;
+
+TEST(ConversationalSpeechTest, Settings) {
+  const conversational_speech::Config config(audiotracks_path, timing_filepath,
+                                             output_path);
+
+  // Test getters.
+  EXPECT_EQ(audiotracks_path, config.audiotracks_path());
+  EXPECT_EQ(timing_filepath, config.timing_filepath());
+  EXPECT_EQ(output_path, config.output_path());
+}
+
+TEST(ConversationalSpeechTest, TimingSaveLoad) {
+  // Save test timing.
+  const std::string temporary_filepath =
+      TempFilename(OutputPath(), "TempTimingTestFile");
+  SaveTiming(temporary_filepath, expected_timing);
+
+  // Create a std::vector<Turn> instance by loading from file.
+  std::vector<Turn> actual_timing = LoadTiming(temporary_filepath);
+  RemoveFile(temporary_filepath);
+
+  // Check size.
+  EXPECT_EQ(expected_timing.size(), actual_timing.size());
+
+  // Check Turn instances.
+  for (size_t index = 0; index < expected_timing.size(); ++index) {
+    EXPECT_EQ(expected_timing[index], actual_timing[index])
+        << "turn #" << index << " not matching";
+  }
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallCreate) {
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There are 5 unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(5);
+
+  // Inject the mock wav reader factory.
+  conversational_speech::MultiEndCall multiend_call(
+      expected_timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_TRUE(multiend_call.valid());
+
+  // Test.
+  EXPECT_EQ(2u, multiend_call.speaker_names().size());
+  EXPECT_EQ(5u, multiend_call.audiotrack_readers().size());
+  EXPECT_EQ(6u, multiend_call.speaking_turns().size());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupDifferentSampleRates) {
+  const std::vector<Turn> timing = {
+      {"A", "sr8000", 0, 0},
+      {"B", "sr16000", 0, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There are two unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(::testing::_)).Times(2);
+
+  MultiEndCall multiend_call(timing, audiotracks_path,
+                             std::move(mock_wavreader_factory));
+  EXPECT_FALSE(multiend_call.valid());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupMultipleChannels) {
+  const std::vector<Turn> timing = {
+      {"A", "sr16000_stereo", 0, 0},
+      {"B", "sr16000_stereo", 0, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There is one unique audio track to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(::testing::_)).Times(1);
+
+  MultiEndCall multiend_call(timing, audiotracks_path,
+                             std::move(mock_wavreader_factory));
+  EXPECT_FALSE(multiend_call.valid());
+}
+
+TEST(ConversationalSpeechTest,
+     MultiEndCallSetupDifferentSampleRatesAndMultipleNumChannels) {
+  const std::vector<Turn> timing = {
+      {"A", "sr8000", 0, 0},
+      {"B", "sr16000_stereo", 0, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There are two unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(::testing::_)).Times(2);
+
+  MultiEndCall multiend_call(timing, audiotracks_path,
+                             std::move(mock_wavreader_factory));
+  EXPECT_FALSE(multiend_call.valid());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupFirstOffsetNegative) {
+  const std::vector<Turn> timing = {
+      {"A", "t500", -100, 0},
+      {"B", "t500", 0, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There is one unique audio track to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(1);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_FALSE(multiend_call.valid());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupSimple) {
+  // Accept:
+  // A 0****.....
+  // B .....1****
+  constexpr std::size_t expected_duration = kDefaultSampleRate;
+  const std::vector<Turn> timing = {
+      {"A", "t500", 0, 0},
+      {"B", "t500", 0, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There is one unique audio track to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(1);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_TRUE(multiend_call.valid());
+
+  // Test.
+  EXPECT_EQ(2u, multiend_call.speaker_names().size());
+  EXPECT_EQ(1u, multiend_call.audiotrack_readers().size());
+  EXPECT_EQ(2u, multiend_call.speaking_turns().size());
+  EXPECT_EQ(expected_duration, multiend_call.total_duration_samples());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupPause) {
+  // Accept:
+  // A 0****.......
+  // B .......1****
+  constexpr std::size_t expected_duration = kDefaultSampleRate * 1.2;
+  const std::vector<Turn> timing = {
+      {"A", "t500", 0, 0},
+      {"B", "t500", 200, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There is one unique audio track to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(1);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_TRUE(multiend_call.valid());
+
+  // Test.
+  EXPECT_EQ(2u, multiend_call.speaker_names().size());
+  EXPECT_EQ(1u, multiend_call.audiotrack_readers().size());
+  EXPECT_EQ(2u, multiend_call.speaking_turns().size());
+  EXPECT_EQ(expected_duration, multiend_call.total_duration_samples());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalk) {
+  // Accept:
+  // A 0****....
+  // B ....1****
+  constexpr std::size_t expected_duration = kDefaultSampleRate * 0.9;
+  const std::vector<Turn> timing = {
+      {"A", "t500", 0, 0},
+      {"B", "t500", -100, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There is one unique audio track to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(1);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_TRUE(multiend_call.valid());
+
+  // Test.
+  EXPECT_EQ(2u, multiend_call.speaker_names().size());
+  EXPECT_EQ(1u, multiend_call.audiotrack_readers().size());
+  EXPECT_EQ(2u, multiend_call.speaking_turns().size());
+  EXPECT_EQ(expected_duration, multiend_call.total_duration_samples());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupInvalidOrder) {
+  // Reject:
+  // A ..0****
+  // B .1****.  The n-th turn cannot start before the (n-1)-th one.
+  const std::vector<Turn> timing = {
+      {"A", "t500", 200, 0},
+      {"B", "t500", -600, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There is one unique audio track to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(1);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_FALSE(multiend_call.valid());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkThree) {
+  // Accept:
+  // A 0****2****...
+  // B ...1*********
+  constexpr std::size_t expected_duration = kDefaultSampleRate * 1.3;
+  const std::vector<Turn> timing = {
+      {"A", "t500", 0, 0},
+      {"B", "t1000", -200, 0},
+      {"A", "t500", -800, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There are two unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(2);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_TRUE(multiend_call.valid());
+
+  // Test.
+  EXPECT_EQ(2u, multiend_call.speaker_names().size());
+  EXPECT_EQ(2u, multiend_call.audiotrack_readers().size());
+  EXPECT_EQ(3u, multiend_call.speaking_turns().size());
+  EXPECT_EQ(expected_duration, multiend_call.total_duration_samples());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupSelfCrossTalkNearInvalid) {
+  // Reject:
+  // A 0****......
+  // A ...1****...
+  // B ......2****
+  //      ^  Turn #1 overlaps with #0 which is from the same speaker.
+  const std::vector<Turn> timing = {
+      {"A", "t500", 0, 0},
+      {"A", "t500", -200, 0},
+      {"B", "t500", -200, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There is one unique audio track to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(1);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_FALSE(multiend_call.valid());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupSelfCrossTalkFarInvalid) {
+  // Reject:
+  // A 0*********
+  // B 1**.......
+  // C ...2**....
+  // A ......3**.
+  //         ^  Turn #3 overlaps with #0 which is from the same speaker.
+  const std::vector<Turn> timing = {
+      {"A", "t1000", 0, 0},
+      {"B", "t300", -1000, 0},
+      {"C", "t300", 0, 0},
+      {"A", "t300", 0, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There are two unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(2);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_FALSE(multiend_call.valid());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkMiddleValid) {
+  // Accept:
+  // A 0*********..
+  // B ..1****.....
+  // C .......2****
+  constexpr std::size_t expected_duration = kDefaultSampleRate * 1.2;
+  const std::vector<Turn> timing = {
+      {"A", "t1000", 0, 0},
+      {"B", "t500", -800, 0},
+      {"C", "t500", 0, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There are two unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(2);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_TRUE(multiend_call.valid());
+
+  // Test.
+  EXPECT_EQ(3u, multiend_call.speaker_names().size());
+  EXPECT_EQ(2u, multiend_call.audiotrack_readers().size());
+  EXPECT_EQ(3u, multiend_call.speaking_turns().size());
+  EXPECT_EQ(expected_duration, multiend_call.total_duration_samples());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkMiddleInvalid) {
+  // Reject:
+  // A 0*********
+  // B ..1****...
+  // C ....2****.
+  //       ^  Turn #2 overlaps both with #0 and #1 (cross-talk with 3+ speakers
+  //          not permitted).
+  const std::vector<Turn> timing = {
+      {"A", "t1000", 0, 0},
+      {"B", "t500", -800, 0},
+      {"C", "t500", -300, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There are two unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(2);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_FALSE(multiend_call.valid());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkMiddleAndPause) {
+  // Accept:
+  // A 0*********..
+  // B .2****......
+  // C .......3****
+  constexpr std::size_t expected_duration = kDefaultSampleRate * 1.2;
+  const std::vector<Turn> timing = {
+      {"A", "t1000", 0, 0},
+      {"B", "t500", -900, 0},
+      {"C", "t500", 100, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There are two unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(2);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_TRUE(multiend_call.valid());
+
+  // Test.
+  EXPECT_EQ(3u, multiend_call.speaker_names().size());
+  EXPECT_EQ(2u, multiend_call.audiotrack_readers().size());
+  EXPECT_EQ(3u, multiend_call.speaking_turns().size());
+  EXPECT_EQ(expected_duration, multiend_call.total_duration_samples());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkFullOverlapValid) {
+  // Accept:
+  // A 0****
+  // B 1****
+  const std::vector<Turn> timing = {
+      {"A", "t500", 0, 0},
+      {"B", "t500", -500, 0},
+  };
+  auto mock_wavreader_factory = CreateMockWavReaderFactory();
+
+  // There is one unique audio track to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(1);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_TRUE(multiend_call.valid());
+
+  // Test.
+  EXPECT_EQ(2u, multiend_call.speaker_names().size());
+  EXPECT_EQ(1u, multiend_call.audiotrack_readers().size());
+  EXPECT_EQ(2u, multiend_call.speaking_turns().size());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupLongSequence) {
+  // Accept:
+  // A 0****....3****.5**.
+  // B .....1****...4**...
+  // C ......2**.......6**..
+  constexpr std::size_t expected_duration = kDefaultSampleRate * 1.9;
+  const std::vector<Turn> timing = {
+      {"A", "t500", 0, 0},    {"B", "t500", 0, 0},    {"C", "t300", -400, 0},
+      {"A", "t500", 0, 0},    {"B", "t300", -100, 0}, {"A", "t300", -100, 0},
+      {"C", "t300", -200, 0},
+  };
+  auto mock_wavreader_factory = std::unique_ptr<MockWavReaderFactory>(
+      new MockWavReaderFactory(kDefaultMockWavReaderFactoryParams,
+                               kDefaultMockWavReaderFactoryParamsMap));
+
+  // There are two unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(2);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_TRUE(multiend_call.valid());
+
+  // Test.
+  EXPECT_EQ(3u, multiend_call.speaker_names().size());
+  EXPECT_EQ(2u, multiend_call.audiotrack_readers().size());
+  EXPECT_EQ(7u, multiend_call.speaking_turns().size());
+  EXPECT_EQ(expected_duration, multiend_call.total_duration_samples());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallSetupLongSequenceInvalid) {
+  // Reject:
+  // A 0****....3****.6**
+  // B .....1****...4**..
+  // C ......2**.....5**..
+  //                 ^ Turns #4, #5 and #6 overlapping (cross-talk with 3+
+  //                   speakers not permitted).
+  const std::vector<Turn> timing = {
+      {"A", "t500", 0, 0},    {"B", "t500", 0, 0},    {"C", "t300", -400, 0},
+      {"A", "t500", 0, 0},    {"B", "t300", -100, 0}, {"A", "t300", -200, 0},
+      {"C", "t300", -200, 0},
+  };
+  auto mock_wavreader_factory = std::unique_ptr<MockWavReaderFactory>(
+      new MockWavReaderFactory(kDefaultMockWavReaderFactoryParams,
+                               kDefaultMockWavReaderFactoryParamsMap));
+
+  // There are two unique audio tracks to read.
+  EXPECT_CALL(*mock_wavreader_factory, Create(_)).Times(2);
+
+  conversational_speech::MultiEndCall multiend_call(
+      timing, audiotracks_path, std::move(mock_wavreader_factory));
+  EXPECT_FALSE(multiend_call.valid());
+}
+
+TEST(ConversationalSpeechTest, MultiEndCallWavReaderAdaptorSine) {
+  // Parameters with which wav files are created.
+  constexpr int duration_seconds = 5;
+  const int sample_rates[] = {8000, 11025, 16000, 22050, 32000, 44100, 48000};
+
+  for (int sample_rate : sample_rates) {
+    const std::string temp_filename = OutputPath() + "TempSineWavFile_" +
+                                      std::to_string(sample_rate) + ".wav";
+
+    // Write wav file.
+    const std::size_t num_samples = duration_seconds * sample_rate;
+    MockWavReaderFactory::Params params = {sample_rate, 1u, num_samples};
+    CreateSineWavFile(temp_filename, params);
+
+    // Load wav file and check if params match.
+    WavReaderFactory wav_reader_factory;
+    MockWavReaderFactory::Params expeted_params = {sample_rate, 1u,
+                                                   num_samples};
+    CheckAudioTrackParams(wav_reader_factory, temp_filename, expeted_params);
+
+    // Clean up.
+    RemoveFile(temp_filename);
+  }
+}
+
+TEST(ConversationalSpeechTest, DISABLED_MultiEndCallSimulator) {
+  // Simulated call (one character corresponding to 500 ms):
+  // A 0*********...........2*********.....
+  // B ...........1*********.....3*********
+  const std::vector<Turn> expected_timing = {
+      {"A", "t5000_440.wav", 0, 0},
+      {"B", "t5000_880.wav", 500, 0},
+      {"A", "t5000_440.wav", 0, 0},
+      {"B", "t5000_880.wav", -2500, 0},
+  };
+  const std::size_t expected_duration_seconds = 18;
+
+  // Create temporary audio track files.
+  const int sample_rate = 16000;
+  const std::map<std::string, SineAudioTrackParams> sine_tracks_params = {
+      {"t5000_440.wav", {{sample_rate, 1u, sample_rate * 5}, 440.0}},
+      {"t5000_880.wav", {{sample_rate, 1u, sample_rate * 5}, 880.0}},
+  };
+  const std::string audiotracks_path =
+      CreateTemporarySineAudioTracks(sine_tracks_params);
+
+  // Set up the multi-end call.
+  auto wavreader_factory =
+      std::unique_ptr<WavReaderFactory>(new WavReaderFactory());
+  MultiEndCall multiend_call(expected_timing, audiotracks_path,
+                             std::move(wavreader_factory));
+
+  // Simulate the call.
+  std::string output_path = JoinFilename(audiotracks_path, "output");
+  CreateDir(output_path);
+  RTC_LOG(LS_VERBOSE) << "simulator output path: " << output_path;
+  auto generated_audiotrak_pairs =
+      conversational_speech::Simulate(multiend_call, output_path);
+  EXPECT_EQ(2u, generated_audiotrak_pairs->size());
+
+  // Check the output.
+  WavReaderFactory wav_reader_factory;
+  const MockWavReaderFactory::Params expeted_params = {
+      sample_rate, 1u, sample_rate * expected_duration_seconds};
+  for (const auto& it : *generated_audiotrak_pairs) {
+    RTC_LOG(LS_VERBOSE) << "checking far/near-end for <" << it.first << ">";
+    CheckAudioTrackParams(wav_reader_factory, it.second.near_end,
+                          expeted_params);
+    CheckAudioTrackParams(wav_reader_factory, it.second.far_end,
+                          expeted_params);
+  }
+
+  // Clean.
+  EXPECT_NO_FATAL_FAILURE(DeleteFolderAndContents(audiotracks_path));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc
new file mode 100644
index 0000000000..1263e938c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/conversational_speech/mock_wavreader.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+using ::testing::Return;
+
+MockWavReader::MockWavReader(int sample_rate,
+                             size_t num_channels,
+                             size_t num_samples)
+    : sample_rate_(sample_rate),
+      num_channels_(num_channels),
+      num_samples_(num_samples) {
+  ON_CALL(*this, SampleRate()).WillByDefault(Return(sample_rate_));
+  ON_CALL(*this, NumChannels()).WillByDefault(Return(num_channels_));
+  ON_CALL(*this, NumSamples()).WillByDefault(Return(num_samples_));
+}
+
+MockWavReader::~MockWavReader() = default;
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
new file mode 100644
index 0000000000..94e20b9ec6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_
+
+#include <cstddef>
+#include <string>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class MockWavReader : public WavReaderInterface {
+ public:
+  MockWavReader(int sample_rate, size_t num_channels, size_t num_samples);
+  ~MockWavReader();
+
+  // TODO(alessiob): use ON_CALL to return random samples if needed.
+  MOCK_METHOD(size_t, ReadFloatSamples, (rtc::ArrayView<float>), (override));
+  MOCK_METHOD(size_t, ReadInt16Samples, (rtc::ArrayView<int16_t>), (override));
+
+  MOCK_METHOD(int, SampleRate, (), (const, override));
+  MOCK_METHOD(size_t, NumChannels, (), (const, override));
+  MOCK_METHOD(size_t, NumSamples, (), (const, override));
+
+ private:
+  const int sample_rate_;
+  const size_t num_channels_;
+  const size_t num_samples_;
+};
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
new file mode 100644
index 0000000000..a377cce7e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/test/conversational_speech/mock_wavreader.h"
+#include "rtc_base/logging.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+using ::testing::_;
+using ::testing::Invoke;
+
+MockWavReaderFactory::MockWavReaderFactory(
+    const Params& default_params,
+    const std::map<std::string, const Params>& params)
+    : default_params_(default_params), audiotrack_names_params_(params) {
+  ON_CALL(*this, Create(_))
+      .WillByDefault(Invoke(this, &MockWavReaderFactory::CreateMock));
+}
+
+MockWavReaderFactory::MockWavReaderFactory(const Params& default_params)
+    : MockWavReaderFactory(default_params,
+                           std::map<std::string, const Params>{}) {}
+
+MockWavReaderFactory::~MockWavReaderFactory() = default;
+
+std::unique_ptr<WavReaderInterface> MockWavReaderFactory::CreateMock(
+    absl::string_view filepath) {
+  // Search the parameters corresponding to filepath.
+  size_t delimiter = filepath.find_last_of("/\\");  // Either windows or posix
+  std::string filename(filepath.substr(
+      delimiter == absl::string_view::npos ? 0 : delimiter + 1));
+  const auto it = audiotrack_names_params_.find(filename);
+
+  // If not found, use default parameters.
+  if (it == audiotrack_names_params_.end()) {
+    RTC_LOG(LS_VERBOSE) << "using default parameters for " << filepath;
+    return std::unique_ptr<WavReaderInterface>(new MockWavReader(
+        default_params_.sample_rate, default_params_.num_channels,
+        default_params_.num_samples));
+  }
+
+  // Found, use the audiotrack-specific parameters.
+  RTC_LOG(LS_VERBOSE) << "using ad-hoc parameters for " << filepath;
+  RTC_LOG(LS_VERBOSE) << "sample_rate " << it->second.sample_rate;
+  RTC_LOG(LS_VERBOSE) << "num_channels " << it->second.num_channels;
+  RTC_LOG(LS_VERBOSE) << "num_samples " << it->second.num_samples;
+  return std::unique_ptr<WavReaderInterface>(new MockWavReader(
+      it->second.sample_rate, it->second.num_channels, it->second.num_samples));
+}
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h
new file mode 100644
index 0000000000..bcc7f3069b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_FACTORY_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_FACTORY_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class MockWavReaderFactory : public WavReaderAbstractFactory {
+ public:
+  struct Params {
+    int sample_rate;
+    size_t num_channels;
+    size_t num_samples;
+  };
+
+  MockWavReaderFactory(const Params& default_params,
+                       const std::map<std::string, const Params>& params);
+  explicit MockWavReaderFactory(const Params& default_params);
+  ~MockWavReaderFactory();
+
+  MOCK_METHOD(std::unique_ptr<WavReaderInterface>,
+              Create,
+              (absl::string_view),
+              (const, override));
+
+ private:
+  // Creates a MockWavReader instance using the parameters in
+  // audiotrack_names_params_ if the entry corresponding to filepath exists,
+  // otherwise creates a MockWavReader instance using the default parameters.
+  std::unique_ptr<WavReaderInterface> CreateMock(absl::string_view filepath);
+
+  const Params& default_params_;
+  std::map<std::string, const Params> audiotrack_names_params_;
+};
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
new file mode 100644
index 0000000000..952114a78b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/conversational_speech/multiend_call.h"
+
+#include <algorithm>
+#include <iterator>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/logging.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+MultiEndCall::MultiEndCall(
+    rtc::ArrayView<const Turn> timing,
+    absl::string_view audiotracks_path,
+    std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory)
+    : timing_(timing),
+      audiotracks_path_(audiotracks_path),
+      wavreader_abstract_factory_(std::move(wavreader_abstract_factory)),
+      valid_(false) {
+  FindSpeakerNames();
+  if (CreateAudioTrackReaders())
+    valid_ = CheckTiming();
+}
+
+MultiEndCall::~MultiEndCall() = default;
+
+void MultiEndCall::FindSpeakerNames() {
+  RTC_DCHECK(speaker_names_.empty());
+  for (const Turn& turn : timing_) {
+    speaker_names_.emplace(turn.speaker_name);
+  }
+}
+
+bool MultiEndCall::CreateAudioTrackReaders() {
+  RTC_DCHECK(audiotrack_readers_.empty());
+  sample_rate_hz_ = 0;  // Sample rate will be set when reading the first track.
+  for (const Turn& turn : timing_) {
+    auto it = audiotrack_readers_.find(turn.audiotrack_file_name);
+    if (it != audiotrack_readers_.end())
+      continue;
+
+    const std::string audiotrack_file_path =
+        test::JoinFilename(audiotracks_path_, turn.audiotrack_file_name);
+
+    // Map the audiotrack file name to a new instance of WavReaderInterface.
+    std::unique_ptr<WavReaderInterface> wavreader =
+        wavreader_abstract_factory_->Create(
+            test::JoinFilename(audiotracks_path_, turn.audiotrack_file_name));
+
+    if (sample_rate_hz_ == 0) {
+      sample_rate_hz_ = wavreader->SampleRate();
+    } else if (sample_rate_hz_ != wavreader->SampleRate()) {
+      RTC_LOG(LS_ERROR)
+          << "All the audio tracks should have the same sample rate.";
+      return false;
+    }
+
+    if (wavreader->NumChannels() != 1) {
+      RTC_LOG(LS_ERROR) << "Only mono audio tracks supported.";
+      return false;
+    }
+
+    audiotrack_readers_.emplace(turn.audiotrack_file_name,
+                                std::move(wavreader));
+  }
+
+  return true;
+}
+
+bool MultiEndCall::CheckTiming() {
+  struct Interval {
+    size_t begin;
+    size_t end;
+  };
+  size_t number_of_turns = timing_.size();
+  auto millisecond_to_samples = [](int ms, int sr) -> int {
+    // Truncation may happen if the sampling rate is not an integer multiple
+    // of 1000 (e.g., 44100).
+    return ms * sr / 1000;
+  };
+  auto in_interval = [](size_t value, const Interval& interval) {
+    return interval.begin <= value && value < interval.end;
+  };
+  total_duration_samples_ = 0;
+  speaking_turns_.clear();
+
+  // Begin and end timestamps for the last two turns (unit: number of samples).
+  Interval second_last_turn = {0, 0};
+  Interval last_turn = {0, 0};
+
+  // Initialize map to store speaking turn indices of each speaker (used to
+  // detect self cross-talk).
+  std::map<std::string, std::vector<size_t>> speaking_turn_indices;
+  for (const std::string& speaker_name : speaker_names_) {
+    speaking_turn_indices.emplace(std::piecewise_construct,
+                                  std::forward_as_tuple(speaker_name),
+                                  std::forward_as_tuple());
+  }
+
+  // Parse turns.
+  for (size_t turn_index = 0; turn_index < number_of_turns; ++turn_index) {
+    const Turn& turn = timing_[turn_index];
+    auto it = audiotrack_readers_.find(turn.audiotrack_file_name);
+    RTC_CHECK(it != audiotrack_readers_.end())
+        << "Audio track reader not created";
+
+    // Begin and end timestamps for the current turn.
+    int offset_samples =
+        millisecond_to_samples(turn.offset, it->second->SampleRate());
+    std::size_t begin_timestamp = last_turn.end + offset_samples;
+    std::size_t end_timestamp = begin_timestamp + it->second->NumSamples();
+    RTC_LOG(LS_INFO) << "turn #" << turn_index << " " << begin_timestamp << "-"
+                     << end_timestamp << " ms";
+
+    // The order is invalid if the offset is negative and its absolute value is
+    // larger then the duration of the previous turn.
+    if (offset_samples < 0 &&
+        -offset_samples > static_cast<int>(last_turn.end - last_turn.begin)) {
+      RTC_LOG(LS_ERROR) << "invalid order";
+      return false;
+    }
+
+    // Cross-talk with 3 or more speakers occurs when the beginning of the
+    // current interval falls in the last two turns.
+    if (turn_index > 1 && in_interval(begin_timestamp, last_turn) &&
+        in_interval(begin_timestamp, second_last_turn)) {
+      RTC_LOG(LS_ERROR) << "cross-talk with 3+ speakers";
+      return false;
+    }
+
+    // Append turn.
+    speaking_turns_.emplace_back(turn.speaker_name, turn.audiotrack_file_name,
+                                 begin_timestamp, end_timestamp, turn.gain);
+
+    // Save speaking turn index for self cross-talk detection.
+    RTC_DCHECK_EQ(speaking_turns_.size(), turn_index + 1);
+    speaking_turn_indices[turn.speaker_name].push_back(turn_index);
+
+    // Update total duration of the consversational speech.
+    if (total_duration_samples_ < end_timestamp)
+      total_duration_samples_ = end_timestamp;
+
+    // Update and continue with next turn.
+    second_last_turn = last_turn;
+    last_turn.begin = begin_timestamp;
+    last_turn.end = end_timestamp;
+  }
+
+  // Detect self cross-talk.
+  for (const std::string& speaker_name : speaker_names_) {
+    RTC_LOG(LS_INFO) << "checking self cross-talk for <" << speaker_name << ">";
+
+    // Copy all turns for this speaker to new vector.
+    std::vector<SpeakingTurn> speaking_turns_for_name;
+    std::copy_if(speaking_turns_.begin(), speaking_turns_.end(),
+                 std::back_inserter(speaking_turns_for_name),
+                 [&speaker_name](const SpeakingTurn& st) {
+                   return st.speaker_name == speaker_name;
+                 });
+
+    // Check for overlap between adjacent elements.
+    // This is a sufficient condition for self cross-talk since the intervals
+    // are sorted by begin timestamp.
+    auto overlap = std::adjacent_find(
+        speaking_turns_for_name.begin(), speaking_turns_for_name.end(),
+        [](const SpeakingTurn& a, const SpeakingTurn& b) {
+          return a.end > b.begin;
+        });
+
+    if (overlap != speaking_turns_for_name.end()) {
+      RTC_LOG(LS_ERROR) << "Self cross-talk detected";
+      return false;
+    }
+  }
+
+  return true;
+}
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/multiend_call.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
new file mode 100644
index 0000000000..63283465fa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
+
+#include <stddef.h>
+
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/test/conversational_speech/timing.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class MultiEndCall {
+ public:
+  struct SpeakingTurn {
+    // Constructor required in order to use std::vector::emplace_back().
+    SpeakingTurn(absl::string_view new_speaker_name,
+                 absl::string_view new_audiotrack_file_name,
+                 size_t new_begin,
+                 size_t new_end,
+                 int gain)
+        : speaker_name(new_speaker_name),
+          audiotrack_file_name(new_audiotrack_file_name),
+          begin(new_begin),
+          end(new_end),
+          gain(gain) {}
+    std::string speaker_name;
+    std::string audiotrack_file_name;
+    size_t begin;
+    size_t end;
+    int gain;
+  };
+
+  MultiEndCall(
+      rtc::ArrayView<const Turn> timing,
+      absl::string_view audiotracks_path,
+      std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory);
+  ~MultiEndCall();
+
+  MultiEndCall(const MultiEndCall&) = delete;
+  MultiEndCall& operator=(const MultiEndCall&) = delete;
+
+  const std::set<std::string>& speaker_names() const { return speaker_names_; }
+  const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
+  audiotrack_readers() const {
+    return audiotrack_readers_;
+  }
+  bool valid() const { return valid_; }
+  int sample_rate() const { return sample_rate_hz_; }
+  size_t total_duration_samples() const { return total_duration_samples_; }
+  const std::vector<SpeakingTurn>& speaking_turns() const {
+    return speaking_turns_;
+  }
+
+ private:
+  // Finds unique speaker names.
+  void FindSpeakerNames();
+
+  // Creates one WavReader instance for each unique audiotrack. It returns false
+  // if the audio tracks do not have the same sample rate or if they are not
+  // mono.
+  bool CreateAudioTrackReaders();
+
+  // Validates the speaking turns timing information. Accepts cross-talk, but
+  // only up to 2 speakers. Rejects unordered turns and self cross-talk.
+  bool CheckTiming();
+
+  rtc::ArrayView<const Turn> timing_;
+  std::string audiotracks_path_;
+  std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory_;
+  std::set<std::string> speaker_names_;
+  std::map<std::string, std::unique_ptr<WavReaderInterface>>
+      audiotrack_readers_;
+  bool valid_;
+  int sample_rate_hz_;
+  size_t total_duration_samples_;
+  std::vector<SpeakingTurn> speaking_turns_;
+};
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/simulator.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/simulator.cc
new file mode 100644
index 0000000000..89bcd48d84
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/simulator.cc
@@ -0,0 +1,235 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/conversational_speech/simulator.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <memory>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "common_audio/include/audio_util.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using conversational_speech::MultiEndCall;
+using conversational_speech::SpeakerOutputFilePaths;
+using conversational_speech::WavReaderInterface;
+
+// Combines output path and speaker names to define the output file paths for
+// the near-end and far=end audio tracks.
+std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>>
+InitSpeakerOutputFilePaths(const std::set<std::string>& speaker_names,
+                           absl::string_view output_path) {
+  // Create map.
+  auto speaker_output_file_paths_map =
+      std::make_unique<std::map<std::string, SpeakerOutputFilePaths>>();
+
+  // Add near-end and far-end output paths into the map.
+  for (const auto& speaker_name : speaker_names) {
+    const std::string near_end_path =
+        test::JoinFilename(output_path, "s_" + speaker_name + "-near_end.wav");
+    RTC_LOG(LS_VERBOSE) << "The near-end audio track will be created in "
+                        << near_end_path << ".";
+
+    const std::string far_end_path =
+        test::JoinFilename(output_path, "s_" + speaker_name + "-far_end.wav");
+    RTC_LOG(LS_VERBOSE) << "The far-end audio track will be created in "
+                        << far_end_path << ".";
+
+    // Add to map.
+    speaker_output_file_paths_map->emplace(
+        std::piecewise_construct, std::forward_as_tuple(speaker_name),
+        std::forward_as_tuple(near_end_path, far_end_path));
+  }
+
+  return speaker_output_file_paths_map;
+}
+
+// Class that provides one WavWriter for the near-end and one for the far-end
+// output track of a speaker.
+class SpeakerWavWriters {
+ public:
+  SpeakerWavWriters(const SpeakerOutputFilePaths& output_file_paths,
+                    int sample_rate)
+      : near_end_wav_writer_(output_file_paths.near_end, sample_rate, 1u),
+        far_end_wav_writer_(output_file_paths.far_end, sample_rate, 1u) {}
+  WavWriter* near_end_wav_writer() { return &near_end_wav_writer_; }
+  WavWriter* far_end_wav_writer() { return &far_end_wav_writer_; }
+
+ private:
+  WavWriter near_end_wav_writer_;
+  WavWriter far_end_wav_writer_;
+};
+
+// Initializes one WavWriter instance for each speaker and both the near-end and
+// far-end output tracks.
+std::unique_ptr<std::map<std::string, SpeakerWavWriters>>
+InitSpeakersWavWriters(const std::map<std::string, SpeakerOutputFilePaths>&
+                           speaker_output_file_paths,
+                       int sample_rate) {
+  // Create map.
+  auto speaker_wav_writers_map =
+      std::make_unique<std::map<std::string, SpeakerWavWriters>>();
+
+  // Add SpeakerWavWriters instance into the map.
+  for (auto it = speaker_output_file_paths.begin();
+       it != speaker_output_file_paths.end(); ++it) {
+    speaker_wav_writers_map->emplace(
+        std::piecewise_construct, std::forward_as_tuple(it->first),
+        std::forward_as_tuple(it->second, sample_rate));
+  }
+
+  return speaker_wav_writers_map;
+}
+
+// Reads all the samples for each audio track.
+std::unique_ptr<std::map<std::string, std::vector<int16_t>>> PreloadAudioTracks(
+    const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
+        audiotrack_readers) {
+  // Create map.
+  auto audiotracks_map =
+      std::make_unique<std::map<std::string, std::vector<int16_t>>>();
+
+  // Add audio track vectors.
+  for (auto it = audiotrack_readers.begin(); it != audiotrack_readers.end();
+       ++it) {
+    // Add map entry.
+    audiotracks_map->emplace(std::piecewise_construct,
+                             std::forward_as_tuple(it->first),
+                             std::forward_as_tuple(it->second->NumSamples()));
+
+    // Read samples.
+    it->second->ReadInt16Samples(audiotracks_map->at(it->first));
+  }
+
+  return audiotracks_map;
+}
+
+// Writes all the values in `source_samples` via `wav_writer`. If the number of
+// previously written samples in `wav_writer` is less than `interval_begin`, it
+// adds zeros as left padding. The padding corresponds to intervals during which
+// a speaker is not active.
+void PadLeftWriteChunk(rtc::ArrayView<const int16_t> source_samples,
+                       size_t interval_begin,
+                       WavWriter* wav_writer) {
+  // Add left padding.
+  RTC_CHECK(wav_writer);
+  RTC_CHECK_GE(interval_begin, wav_writer->num_samples());
+  size_t padding_size = interval_begin - wav_writer->num_samples();
+  if (padding_size != 0) {
+    const std::vector<int16_t> padding(padding_size, 0);
+    wav_writer->WriteSamples(padding.data(), padding_size);
+  }
+
+  // Write source samples.
+  wav_writer->WriteSamples(source_samples.data(), source_samples.size());
+}
+
+// Appends zeros via `wav_writer`. The number of zeros is always non-negative
+// and equal to the difference between the previously written samples and
+// `pad_samples`.
+void PadRightWrite(WavWriter* wav_writer, size_t pad_samples) {
+  RTC_CHECK(wav_writer);
+  RTC_CHECK_GE(pad_samples, wav_writer->num_samples());
+  size_t padding_size = pad_samples - wav_writer->num_samples();
+  if (padding_size != 0) {
+    const std::vector<int16_t> padding(padding_size, 0);
+    wav_writer->WriteSamples(padding.data(), padding_size);
+  }
+}
+
+void ScaleSignal(rtc::ArrayView<const int16_t> source_samples,
+                 int gain,
+                 rtc::ArrayView<int16_t> output_samples) {
+  const float gain_linear = DbToRatio(gain);
+  RTC_DCHECK_EQ(source_samples.size(), output_samples.size());
+  std::transform(source_samples.begin(), source_samples.end(),
+                 output_samples.begin(), [gain_linear](int16_t x) -> int16_t {
+                   return rtc::saturated_cast<int16_t>(x * gain_linear);
+                 });
+}
+
+}  // namespace
+
+namespace conversational_speech {
+
+std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>> Simulate(
+    const MultiEndCall& multiend_call,
+    absl::string_view output_path) {
+  // Set output file paths and initialize wav writers.
+  const auto& speaker_names = multiend_call.speaker_names();
+  auto speaker_output_file_paths =
+      InitSpeakerOutputFilePaths(speaker_names, output_path);
+  auto speakers_wav_writers = InitSpeakersWavWriters(
+      *speaker_output_file_paths, multiend_call.sample_rate());
+
+  // Preload all the input audio tracks.
+  const auto& audiotrack_readers = multiend_call.audiotrack_readers();
+  auto audiotracks = PreloadAudioTracks(audiotrack_readers);
+
+  // TODO(alessiob): When speaker_names.size() == 2, near-end and far-end
+  // across the 2 speakers are symmetric; hence, the code below could be
+  // replaced by only creating the near-end or the far-end. However, this would
+  // require to split the unit tests and document the behavior in README.md.
+  // In practice, it should not be an issue since the files are not expected to
+  // be signinificant.
+
+  // Write near-end and far-end output tracks.
+  for (const auto& speaking_turn : multiend_call.speaking_turns()) {
+    const std::string& active_speaker_name = speaking_turn.speaker_name;
+    const auto source_audiotrack =
+        audiotracks->at(speaking_turn.audiotrack_file_name);
+    std::vector<int16_t> scaled_audiotrack(source_audiotrack.size());
+    ScaleSignal(source_audiotrack, speaking_turn.gain, scaled_audiotrack);
+
+    // Write active speaker's chunk to active speaker's near-end.
+    PadLeftWriteChunk(
+        scaled_audiotrack, speaking_turn.begin,
+        speakers_wav_writers->at(active_speaker_name).near_end_wav_writer());
+
+    // Write active speaker's chunk to other participants' far-ends.
+    for (const std::string& speaker_name : speaker_names) {
+      if (speaker_name == active_speaker_name)
+        continue;
+      PadLeftWriteChunk(
+          scaled_audiotrack, speaking_turn.begin,
+          speakers_wav_writers->at(speaker_name).far_end_wav_writer());
+    }
+  }
+
+  // Finalize all the output tracks with right padding.
+  // This is required to make all the output tracks duration equal.
+  size_t duration_samples = multiend_call.total_duration_samples();
+  for (const std::string& speaker_name : speaker_names) {
+    PadRightWrite(speakers_wav_writers->at(speaker_name).near_end_wav_writer(),
+                  duration_samples);
+    PadRightWrite(speakers_wav_writers->at(speaker_name).far_end_wav_writer(),
+                  duration_samples);
+  }
+
+  return speaker_output_file_paths;
+}
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/simulator.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/simulator.h
new file mode 100644
index 0000000000..2f311e16b3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/simulator.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_SIMULATOR_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_SIMULATOR_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/test/conversational_speech/multiend_call.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+struct SpeakerOutputFilePaths {
+  SpeakerOutputFilePaths(absl::string_view new_near_end,
+                         absl::string_view new_far_end)
+      : near_end(new_near_end), far_end(new_far_end) {}
+  // Paths to the near-end and far-end audio track files.
+  const std::string near_end;
+  const std::string far_end;
+};
+
+// Generates the near-end and far-end audio track pairs for each speaker.
+std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>> Simulate(
+    const MultiEndCall& multiend_call,
+    absl::string_view output_path);
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_SIMULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc
new file mode 100644
index 0000000000..95ec9f542e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/conversational_speech/timing.h"
+
+#include <fstream>
+#include <iostream>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/string_encode.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+bool Turn::operator==(const Turn& b) const {
+  return b.speaker_name == speaker_name &&
+         b.audiotrack_file_name == audiotrack_file_name && b.offset == offset &&
+         b.gain == gain;
+}
+
+std::vector<Turn> LoadTiming(absl::string_view timing_filepath) {
+  // Line parser.
+  auto parse_line = [](absl::string_view line) {
+    std::vector<absl::string_view> fields = rtc::split(line, ' ');
+    RTC_CHECK_GE(fields.size(), 3);
+    RTC_CHECK_LE(fields.size(), 4);
+    int gain = 0;
+    if (fields.size() == 4) {
+      gain = rtc::StringToNumber<int>(fields[3]).value_or(0);
+    }
+    return Turn(fields[0], fields[1],
+                rtc::StringToNumber<int>(fields[2]).value_or(0), gain);
+  };
+
+  // Init.
+  std::vector<Turn> timing;
+
+  // Parse lines.
+  std::string line;
+  std::ifstream infile(std::string{timing_filepath});
+  while (std::getline(infile, line)) {
+    if (line.empty())
+      continue;
+    timing.push_back(parse_line(line));
+  }
+  infile.close();
+
+  return timing;
+}
+
+void SaveTiming(absl::string_view timing_filepath,
+                rtc::ArrayView<const Turn> timing) {
+  std::ofstream outfile(std::string{timing_filepath});
+  RTC_CHECK(outfile.is_open());
+  for (const Turn& turn : timing) {
+    outfile << turn.speaker_name << " " << turn.audiotrack_file_name << " "
+            << turn.offset << " " << turn.gain << std::endl;
+  }
+  outfile.close();
+}
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.h
new file mode 100644
index 0000000000..9314f6fc43
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+struct Turn {
+  Turn(absl::string_view new_speaker_name,
+       absl::string_view new_audiotrack_file_name,
+       int new_offset,
+       int gain)
+      : speaker_name(new_speaker_name),
+        audiotrack_file_name(new_audiotrack_file_name),
+        offset(new_offset),
+        gain(gain) {}
+  bool operator==(const Turn& b) const;
+  std::string speaker_name;
+  std::string audiotrack_file_name;
+  int offset;
+  int gain;
+};
+
+// Loads a list of turns from a file.
+std::vector<Turn> LoadTiming(absl::string_view timing_filepath);
+
+// Writes a list of turns into a file.
+void SaveTiming(absl::string_view timing_filepath,
+                rtc::ArrayView<const Turn> timing);
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
new file mode 100644
index 0000000000..14ddfc7539
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_ABSTRACT_FACTORY_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_ABSTRACT_FACTORY_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class WavReaderAbstractFactory {
+ public:
+  virtual ~WavReaderAbstractFactory() = default;
+  virtual std::unique_ptr<WavReaderInterface> Create(
+      absl::string_view filepath) const = 0;
+};
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_ABSTRACT_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
new file mode 100644
index 0000000000..99b1686484
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/conversational_speech/wavreader_factory.h"
+
+#include <cstddef>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "common_audio/wav_file.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using conversational_speech::WavReaderInterface;
+
+class WavReaderAdaptor final : public WavReaderInterface {
+ public:
+  explicit WavReaderAdaptor(absl::string_view filepath)
+      : wav_reader_(filepath) {}
+  ~WavReaderAdaptor() override = default;
+
+  size_t ReadFloatSamples(rtc::ArrayView<float> samples) override {
+    return wav_reader_.ReadSamples(samples.size(), samples.begin());
+  }
+
+  size_t ReadInt16Samples(rtc::ArrayView<int16_t> samples) override {
+    return wav_reader_.ReadSamples(samples.size(), samples.begin());
+  }
+
+  int SampleRate() const override { return wav_reader_.sample_rate(); }
+
+  size_t NumChannels() const override { return wav_reader_.num_channels(); }
+
+  size_t NumSamples() const override { return wav_reader_.num_samples(); }
+
+ private:
+  WavReader wav_reader_;
+};
+
+}  // namespace
+
+namespace conversational_speech {
+
+WavReaderFactory::WavReaderFactory() = default;
+
+WavReaderFactory::~WavReaderFactory() = default;
+
+std::unique_ptr<WavReaderInterface> WavReaderFactory::Create(
+    absl::string_view filepath) const {
+  return std::unique_ptr<WavReaderAdaptor>(new WavReaderAdaptor(filepath));
+}
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h
new file mode 100644
index 0000000000..f2e5b61055
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_FACTORY_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_FACTORY_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class WavReaderFactory : public WavReaderAbstractFactory {
+ public:
+  WavReaderFactory();
+  ~WavReaderFactory() override;
+  std::unique_ptr<WavReaderInterface> Create(
+      absl::string_view filepath) const override;
+};
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
new file mode 100644
index 0000000000..c74f639461
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_INTERFACE_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_INTERFACE_H_
+
+#include <stddef.h>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class WavReaderInterface {
+ public:
+  virtual ~WavReaderInterface() = default;
+
+  // Returns the number of samples read.
+  virtual size_t ReadFloatSamples(rtc::ArrayView<float> samples) = 0;
+  virtual size_t ReadInt16Samples(rtc::ArrayView<int16_t> samples) = 0;
+
+  // Getters.
+  virtual int SampleRate() const = 0;
+  virtual size_t NumChannels() const = 0;
+  virtual size_t NumSamples() const = 0;
+};
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.cc b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.cc
new file mode 100644
index 0000000000..2f483f5077
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.cc
@@ -0,0 +1,250 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/debug_dump_replayer.h"
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "modules/audio_processing/test/protobuf_utils.h"
+#include "modules/audio_processing/test/runtime_setting_util.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
+                      const StreamConfig& config) {
+  auto& buffer_ref = *buffer;
+  if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
+      buffer_ref->num_channels() != config.num_channels()) {
+    buffer_ref.reset(
+        new ChannelBuffer<float>(config.num_frames(), config.num_channels()));
+  }
+}
+
+}  // namespace
+
+DebugDumpReplayer::DebugDumpReplayer()
+    : input_(nullptr),  // will be created upon usage.
+      reverse_(nullptr),
+      output_(nullptr),
+      apm_(nullptr),
+      debug_file_(nullptr) {}
+
+DebugDumpReplayer::~DebugDumpReplayer() {
+  if (debug_file_)
+    fclose(debug_file_);
+}
+
+bool DebugDumpReplayer::SetDumpFile(absl::string_view filename) {
+  debug_file_ = fopen(std::string(filename).c_str(), "rb");
+  LoadNextMessage();
+  return debug_file_;
+}
+
+// Get next event that has not run.
+absl::optional<audioproc::Event> DebugDumpReplayer::GetNextEvent() const {
+  if (!has_next_event_)
+    return absl::nullopt;
+  else
+    return next_event_;
+}
+
+// Run the next event. Returns the event type.
+bool DebugDumpReplayer::RunNextEvent() {
+  if (!has_next_event_)
+    return false;
+  switch (next_event_.type()) {
+    case audioproc::Event::INIT:
+      OnInitEvent(next_event_.init());
+      break;
+    case audioproc::Event::STREAM:
+      OnStreamEvent(next_event_.stream());
+      break;
+    case audioproc::Event::REVERSE_STREAM:
+      OnReverseStreamEvent(next_event_.reverse_stream());
+      break;
+    case audioproc::Event::CONFIG:
+      OnConfigEvent(next_event_.config());
+      break;
+    case audioproc::Event::RUNTIME_SETTING:
+      OnRuntimeSettingEvent(next_event_.runtime_setting());
+      break;
+    case audioproc::Event::UNKNOWN_EVENT:
+      // We do not expect to receive UNKNOWN event.
+      RTC_CHECK_NOTREACHED();
+  }
+  LoadNextMessage();
+  return true;
+}
+
+const ChannelBuffer<float>* DebugDumpReplayer::GetOutput() const {
+  return output_.get();
+}
+
+StreamConfig DebugDumpReplayer::GetOutputConfig() const {
+  return output_config_;
+}
+
+// OnInitEvent reset the input/output/reserve channel format.
+void DebugDumpReplayer::OnInitEvent(const audioproc::Init& msg) {
+  RTC_CHECK(msg.has_num_input_channels());
+  RTC_CHECK(msg.has_output_sample_rate());
+  RTC_CHECK(msg.has_num_output_channels());
+  RTC_CHECK(msg.has_reverse_sample_rate());
+  RTC_CHECK(msg.has_num_reverse_channels());
+
+  input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
+  output_config_ =
+      StreamConfig(msg.output_sample_rate(), msg.num_output_channels());
+  reverse_config_ =
+      StreamConfig(msg.reverse_sample_rate(), msg.num_reverse_channels());
+
+  MaybeResetBuffer(&input_, input_config_);
+  MaybeResetBuffer(&output_, output_config_);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+// OnStreamEvent replays an input signal and verifies the output.
+void DebugDumpReplayer::OnStreamEvent(const audioproc::Stream& msg) {
+  // APM should have been created.
+  RTC_CHECK(apm_.get());
+
+  if (msg.has_applied_input_volume()) {
+    apm_->set_stream_analog_level(msg.applied_input_volume());
+  }
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->set_stream_delay_ms(msg.delay()));
+
+  if (msg.has_keypress()) {
+    apm_->set_stream_key_pressed(msg.keypress());
+  } else {
+    apm_->set_stream_key_pressed(true);
+  }
+
+  RTC_CHECK_EQ(input_config_.num_channels(),
+               static_cast<size_t>(msg.input_channel_size()));
+  RTC_CHECK_EQ(input_config_.num_frames() * sizeof(float),
+               msg.input_channel(0).size());
+
+  for (int i = 0; i < msg.input_channel_size(); ++i) {
+    memcpy(input_->channels()[i], msg.input_channel(i).data(),
+           msg.input_channel(i).size());
+  }
+
+  RTC_CHECK_EQ(AudioProcessing::kNoError,
+               apm_->ProcessStream(input_->channels(), input_config_,
+                                   output_config_, output_->channels()));
+}
+
+void DebugDumpReplayer::OnReverseStreamEvent(
+    const audioproc::ReverseStream& msg) {
+  // APM should have been created.
+  RTC_CHECK(apm_.get());
+
+  RTC_CHECK_GT(msg.channel_size(), 0);
+  RTC_CHECK_EQ(reverse_config_.num_channels(),
+               static_cast<size_t>(msg.channel_size()));
+  RTC_CHECK_EQ(reverse_config_.num_frames() * sizeof(float),
+               msg.channel(0).size());
+
+  for (int i = 0; i < msg.channel_size(); ++i) {
+    memcpy(reverse_->channels()[i], msg.channel(i).data(),
+           msg.channel(i).size());
+  }
+
+  RTC_CHECK_EQ(
+      AudioProcessing::kNoError,
+      apm_->ProcessReverseStream(reverse_->channels(), reverse_config_,
+                                 reverse_config_, reverse_->channels()));
+}
+
+void DebugDumpReplayer::OnConfigEvent(const audioproc::Config& msg) {
+  MaybeRecreateApm(msg);
+  ConfigureApm(msg);
+}
+
+void DebugDumpReplayer::OnRuntimeSettingEvent(
+    const audioproc::RuntimeSetting& msg) {
+  RTC_CHECK(apm_.get());
+  ReplayRuntimeSetting(apm_.get(), msg);
+}
+
+void DebugDumpReplayer::MaybeRecreateApm(const audioproc::Config& msg) {
+  // These configurations cannot be changed on the fly.
+  RTC_CHECK(msg.has_aec_delay_agnostic_enabled());
+  RTC_CHECK(msg.has_aec_extended_filter_enabled());
+
+  // We only create APM once, since changes on these fields should not
+  // happen in current implementation.
+  if (!apm_.get()) {
+    apm_ = AudioProcessingBuilderForTesting().Create();
+  }
+}
+
+void DebugDumpReplayer::ConfigureApm(const audioproc::Config& msg) {
+  AudioProcessing::Config apm_config;
+
+  // AEC2/AECM configs.
+  RTC_CHECK(msg.has_aec_enabled());
+  RTC_CHECK(msg.has_aecm_enabled());
+  apm_config.echo_canceller.enabled = msg.aec_enabled() || msg.aecm_enabled();
+  apm_config.echo_canceller.mobile_mode = msg.aecm_enabled();
+
+  // HPF configs.
+  RTC_CHECK(msg.has_hpf_enabled());
+  apm_config.high_pass_filter.enabled = msg.hpf_enabled();
+
+  // Preamp configs.
+  RTC_CHECK(msg.has_pre_amplifier_enabled());
+  apm_config.pre_amplifier.enabled = msg.pre_amplifier_enabled();
+  apm_config.pre_amplifier.fixed_gain_factor =
+      msg.pre_amplifier_fixed_gain_factor();
+
+  // NS configs.
+  RTC_CHECK(msg.has_ns_enabled());
+  RTC_CHECK(msg.has_ns_level());
+  apm_config.noise_suppression.enabled = msg.ns_enabled();
+  apm_config.noise_suppression.level =
+      static_cast<AudioProcessing::Config::NoiseSuppression::Level>(
+          msg.ns_level());
+
+  // TS configs.
+  RTC_CHECK(msg.has_transient_suppression_enabled());
+  apm_config.transient_suppression.enabled =
+      msg.transient_suppression_enabled();
+
+  // AGC configs.
+  RTC_CHECK(msg.has_agc_enabled());
+  RTC_CHECK(msg.has_agc_mode());
+  RTC_CHECK(msg.has_agc_limiter_enabled());
+  apm_config.gain_controller1.enabled = msg.agc_enabled();
+  apm_config.gain_controller1.mode =
+      static_cast<AudioProcessing::Config::GainController1::Mode>(
+          msg.agc_mode());
+  apm_config.gain_controller1.enable_limiter = msg.agc_limiter_enabled();
+  RTC_CHECK(msg.has_noise_robust_agc_enabled());
+  apm_config.gain_controller1.analog_gain_controller.enabled =
+      msg.noise_robust_agc_enabled();
+
+  apm_->ApplyConfig(apm_config);
+}
+
+void DebugDumpReplayer::LoadNextMessage() {
+  has_next_event_ =
+      debug_file_ && ReadMessageFromFile(debug_file_, &next_event_);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h
new file mode 100644
index 0000000000..be21c68663
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h
@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
+#define MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/channel_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "modules/audio_processing/debug.pb.h"
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+namespace test {
+
+class DebugDumpReplayer {
+ public:
+  DebugDumpReplayer();
+  ~DebugDumpReplayer();
+
+  // Set dump file
+  bool SetDumpFile(absl::string_view filename);
+
+  // Return next event.
+  absl::optional<audioproc::Event> GetNextEvent() const;
+
+  // Run the next event. Returns true if succeeded.
+  bool RunNextEvent();
+
+  const ChannelBuffer<float>* GetOutput() const;
+  StreamConfig GetOutputConfig() const;
+
+ private:
+  // Following functions are facilities for replaying debug dumps.
+  void OnInitEvent(const audioproc::Init& msg);
+  void OnStreamEvent(const audioproc::Stream& msg);
+  void OnReverseStreamEvent(const audioproc::ReverseStream& msg);
+  void OnConfigEvent(const audioproc::Config& msg);
+  void OnRuntimeSettingEvent(const audioproc::RuntimeSetting& msg);
+
+  void MaybeRecreateApm(const audioproc::Config& msg);
+  void ConfigureApm(const audioproc::Config& msg);
+
+  void LoadNextMessage();
+
+  // Buffer for APM input/output.
+  std::unique_ptr<ChannelBuffer<float>> input_;
+  std::unique_ptr<ChannelBuffer<float>> reverse_;
+  std::unique_ptr<ChannelBuffer<float>> output_;
+
+  rtc::scoped_refptr<AudioProcessing> apm_;
+
+  FILE* debug_file_;
+
+  StreamConfig input_config_;
+  StreamConfig reverse_config_;
+  StreamConfig output_config_;
+
+  bool has_next_event_;
+  audioproc::Event next_event_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc
new file mode 100644
index 0000000000..cded5de217
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -0,0 +1,504 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>  // size_t
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/echo_canceller3_factory.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "modules/audio_processing/test/debug_dump_replayer.h"
+#include "modules/audio_processing/test/test_utils.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
+                      const StreamConfig& config) {
+  auto& buffer_ref = *buffer;
+  if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
+      buffer_ref->num_channels() != config.num_channels()) {
+    buffer_ref.reset(
+        new ChannelBuffer<float>(config.num_frames(), config.num_channels()));
+  }
+}
+
+class DebugDumpGenerator {
+ public:
+  DebugDumpGenerator(absl::string_view input_file_name,
+                     int input_rate_hz,
+                     int input_channels,
+                     absl::string_view reverse_file_name,
+                     int reverse_rate_hz,
+                     int reverse_channels,
+                     absl::string_view dump_file_name,
+                     bool enable_pre_amplifier);
+
+  // Constructor that uses default input files.
+  explicit DebugDumpGenerator(const AudioProcessing::Config& apm_config);
+
+  ~DebugDumpGenerator();
+
+  // Changes the sample rate of the input audio to the APM.
+  void SetInputRate(int rate_hz);
+
+  // Sets if converts stereo input signal to mono by discarding other channels.
+  void ForceInputMono(bool mono);
+
+  // Changes the sample rate of the reverse audio to the APM.
+  void SetReverseRate(int rate_hz);
+
+  // Sets if converts stereo reverse signal to mono by discarding other
+  // channels.
+  void ForceReverseMono(bool mono);
+
+  // Sets the required sample rate of the APM output.
+  void SetOutputRate(int rate_hz);
+
+  // Sets the required channels of the APM output.
+  void SetOutputChannels(int channels);
+
+  std::string dump_file_name() const { return dump_file_name_; }
+
+  void StartRecording();
+  void Process(size_t num_blocks);
+  void StopRecording();
+  AudioProcessing* apm() const { return apm_.get(); }
+
+ private:
+  static void ReadAndDeinterleave(ResampleInputAudioFile* audio,
+                                  int channels,
+                                  const StreamConfig& config,
+                                  float* const* buffer);
+
+  // APM input/output settings.
+  StreamConfig input_config_;
+  StreamConfig reverse_config_;
+  StreamConfig output_config_;
+
+  // Input file format.
+  const std::string input_file_name_;
+  ResampleInputAudioFile input_audio_;
+  const int input_file_channels_;
+
+  // Reverse file format.
+  const std::string reverse_file_name_;
+  ResampleInputAudioFile reverse_audio_;
+  const int reverse_file_channels_;
+
+  // Buffer for APM input/output.
+  std::unique_ptr<ChannelBuffer<float>> input_;
+  std::unique_ptr<ChannelBuffer<float>> reverse_;
+  std::unique_ptr<ChannelBuffer<float>> output_;
+
+  bool enable_pre_amplifier_;
+
+  TaskQueueForTest worker_queue_;
+  rtc::scoped_refptr<AudioProcessing> apm_;
+
+  const std::string dump_file_name_;
+};
+
+DebugDumpGenerator::DebugDumpGenerator(absl::string_view input_file_name,
+                                       int input_rate_hz,
+                                       int input_channels,
+                                       absl::string_view reverse_file_name,
+                                       int reverse_rate_hz,
+                                       int reverse_channels,
+                                       absl::string_view dump_file_name,
+                                       bool enable_pre_amplifier)
+    : input_config_(input_rate_hz, input_channels),
+      reverse_config_(reverse_rate_hz, reverse_channels),
+      output_config_(input_rate_hz, input_channels),
+      input_audio_(input_file_name, input_rate_hz, input_rate_hz),
+      input_file_channels_(input_channels),
+      reverse_audio_(reverse_file_name, reverse_rate_hz, reverse_rate_hz),
+      reverse_file_channels_(reverse_channels),
+      input_(new ChannelBuffer<float>(input_config_.num_frames(),
+                                      input_config_.num_channels())),
+      reverse_(new ChannelBuffer<float>(reverse_config_.num_frames(),
+                                        reverse_config_.num_channels())),
+      output_(new ChannelBuffer<float>(output_config_.num_frames(),
+                                       output_config_.num_channels())),
+      enable_pre_amplifier_(enable_pre_amplifier),
+      worker_queue_("debug_dump_generator_worker_queue"),
+      dump_file_name_(dump_file_name) {
+  AudioProcessingBuilderForTesting apm_builder;
+  apm_ = apm_builder.Create();
+}
+
+DebugDumpGenerator::DebugDumpGenerator(
+    const AudioProcessing::Config& apm_config)
+    : DebugDumpGenerator(ResourcePath("near32_stereo", "pcm"),
+                         32000,
+                         2,
+                         ResourcePath("far32_stereo", "pcm"),
+                         32000,
+                         2,
+                         TempFilename(OutputPath(), "debug_aec"),
+                         apm_config.pre_amplifier.enabled) {
+  apm_->ApplyConfig(apm_config);
+}
+
+DebugDumpGenerator::~DebugDumpGenerator() {
+  remove(dump_file_name_.c_str());
+}
+
+void DebugDumpGenerator::SetInputRate(int rate_hz) {
+  input_audio_.set_output_rate_hz(rate_hz);
+  input_config_.set_sample_rate_hz(rate_hz);
+  MaybeResetBuffer(&input_, input_config_);
+}
+
+void DebugDumpGenerator::ForceInputMono(bool mono) {
+  const int channels = mono ? 1 : input_file_channels_;
+  input_config_.set_num_channels(channels);
+  MaybeResetBuffer(&input_, input_config_);
+}
+
+void DebugDumpGenerator::SetReverseRate(int rate_hz) {
+  reverse_audio_.set_output_rate_hz(rate_hz);
+  reverse_config_.set_sample_rate_hz(rate_hz);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+void DebugDumpGenerator::ForceReverseMono(bool mono) {
+  const int channels = mono ? 1 : reverse_file_channels_;
+  reverse_config_.set_num_channels(channels);
+  MaybeResetBuffer(&reverse_, reverse_config_);
+}
+
+void DebugDumpGenerator::SetOutputRate(int rate_hz) {
+  output_config_.set_sample_rate_hz(rate_hz);
+  MaybeResetBuffer(&output_, output_config_);
+}
+
+void DebugDumpGenerator::SetOutputChannels(int channels) {
+  output_config_.set_num_channels(channels);
+  MaybeResetBuffer(&output_, output_config_);
+}
+
+void DebugDumpGenerator::StartRecording() {
+  apm_->AttachAecDump(
+      AecDumpFactory::Create(dump_file_name_.c_str(), -1, &worker_queue_));
+}
+
+void DebugDumpGenerator::Process(size_t num_blocks) {
+  for (size_t i = 0; i < num_blocks; ++i) {
+    ReadAndDeinterleave(&reverse_audio_, reverse_file_channels_,
+                        reverse_config_, reverse_->channels());
+    ReadAndDeinterleave(&input_audio_, input_file_channels_, input_config_,
+                        input_->channels());
+    RTC_CHECK_EQ(AudioProcessing::kNoError, apm_->set_stream_delay_ms(100));
+    apm_->set_stream_analog_level(100);
+    if (enable_pre_amplifier_) {
+      apm_->SetRuntimeSetting(
+          AudioProcessing::RuntimeSetting::CreateCapturePreGain(1 + i % 10));
+    }
+    apm_->set_stream_key_pressed(i % 10 == 9);
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 apm_->ProcessStream(input_->channels(), input_config_,
+                                     output_config_, output_->channels()));
+
+    RTC_CHECK_EQ(
+        AudioProcessing::kNoError,
+        apm_->ProcessReverseStream(reverse_->channels(), reverse_config_,
+                                   reverse_config_, reverse_->channels()));
+  }
+}
+
+void DebugDumpGenerator::StopRecording() {
+  apm_->DetachAecDump();
+}
+
+void DebugDumpGenerator::ReadAndDeinterleave(ResampleInputAudioFile* audio,
+                                             int channels,
+                                             const StreamConfig& config,
+                                             float* const* buffer) {
+  const size_t num_frames = config.num_frames();
+  const int out_channels = config.num_channels();
+
+  std::vector<int16_t> signal(channels * num_frames);
+
+  audio->Read(num_frames * channels, &signal[0]);
+
+  // We only allow reducing number of channels by discarding some channels.
+  RTC_CHECK_LE(out_channels, channels);
+  for (int channel = 0; channel < out_channels; ++channel) {
+    for (size_t i = 0; i < num_frames; ++i) {
+      buffer[channel][i] = S16ToFloat(signal[i * channels + channel]);
+    }
+  }
+}
+
+}  // namespace
+
+class DebugDumpTest : public ::testing::Test {
+ public:
+  // VerifyDebugDump replays a debug dump using APM and verifies that the result
+  // is bit-exact-identical to the output channel in the dump. This is only
+  // guaranteed if the debug dump is started on the first frame.
+  void VerifyDebugDump(absl::string_view in_filename);
+
+ private:
+  DebugDumpReplayer debug_dump_replayer_;
+};
+
+void DebugDumpTest::VerifyDebugDump(absl::string_view in_filename) {
+  ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(in_filename));
+
+  while (const absl::optional<audioproc::Event> event =
+             debug_dump_replayer_.GetNextEvent()) {
+    debug_dump_replayer_.RunNextEvent();
+    if (event->type() == audioproc::Event::STREAM) {
+      const audioproc::Stream* msg = &event->stream();
+      const StreamConfig output_config = debug_dump_replayer_.GetOutputConfig();
+      const ChannelBuffer<float>* output = debug_dump_replayer_.GetOutput();
+      // Check that output of APM is bit-exact to the output in the dump.
+      ASSERT_EQ(output_config.num_channels(),
+                static_cast<size_t>(msg->output_channel_size()));
+      ASSERT_EQ(output_config.num_frames() * sizeof(float),
+                msg->output_channel(0).size());
+      for (int i = 0; i < msg->output_channel_size(); ++i) {
+        ASSERT_EQ(0,
+                  memcmp(output->channels()[i], msg->output_channel(i).data(),
+                         msg->output_channel(i).size()));
+      }
+    }
+  }
+}
+
+TEST_F(DebugDumpTest, SimpleCase) {
+  DebugDumpGenerator generator(/*apm_config=*/{});
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ChangeInputFormat) {
+  DebugDumpGenerator generator(/*apm_config=*/{});
+
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetInputRate(48000);
+
+  generator.ForceInputMono(true);
+  // Number of output channel should not be larger than that of input. APM will
+  // fail otherwise.
+  generator.SetOutputChannels(1);
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ChangeReverseFormat) {
+  DebugDumpGenerator generator(/*apm_config=*/{});
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetReverseRate(48000);
+  generator.ForceReverseMono(true);
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ChangeOutputFormat) {
+  DebugDumpGenerator generator(/*apm_config=*/{});
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetOutputRate(48000);
+  generator.SetOutputChannels(1);
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleAec) {
+  AudioProcessing::Config apm_config;
+  apm_config.echo_canceller.enabled = true;
+  DebugDumpGenerator generator(apm_config);
+  generator.StartRecording();
+  generator.Process(100);
+
+  apm_config.echo_canceller.enabled = false;
+  generator.apm()->ApplyConfig(apm_config);
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringInclusive) {
+  AudioProcessing::Config apm_config;
+  apm_config.echo_canceller.enabled = true;
+  apm_config.gain_controller1.analog_gain_controller.enabled = true;
+  apm_config.gain_controller1.analog_gain_controller.startup_min_volume = 0;
+  DebugDumpGenerator generator(apm_config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+
+  DebugDumpReplayer debug_dump_replayer_;
+
+  ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
+
+  while (const absl::optional<audioproc::Event> event =
+             debug_dump_replayer_.GetNextEvent()) {
+    debug_dump_replayer_.RunNextEvent();
+    if (event->type() == audioproc::Event::CONFIG) {
+      const audioproc::Config* msg = &event->config();
+      ASSERT_TRUE(msg->has_experiments_description());
+      EXPECT_PRED_FORMAT2(::testing::IsSubstring, "EchoController",
+                          msg->experiments_description().c_str());
+    }
+  }
+}
+
+TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringExclusive) {
+  AudioProcessing::Config apm_config;
+  apm_config.echo_canceller.enabled = true;
+  DebugDumpGenerator generator(apm_config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+
+  DebugDumpReplayer debug_dump_replayer_;
+
+  ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
+
+  while (const absl::optional<audioproc::Event> event =
+             debug_dump_replayer_.GetNextEvent()) {
+    debug_dump_replayer_.RunNextEvent();
+    if (event->type() == audioproc::Event::CONFIG) {
+      const audioproc::Config* msg = &event->config();
+      ASSERT_TRUE(msg->has_experiments_description());
+      EXPECT_PRED_FORMAT2(::testing::IsNotSubstring,
+                          "AgcClippingLevelExperiment",
+                          msg->experiments_description().c_str());
+    }
+  }
+}
+
+TEST_F(DebugDumpTest, VerifyAec3ExperimentalString) {
+  AudioProcessing::Config apm_config;
+  apm_config.echo_canceller.enabled = true;
+  DebugDumpGenerator generator(apm_config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+
+  DebugDumpReplayer debug_dump_replayer_;
+
+  ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
+
+  while (const absl::optional<audioproc::Event> event =
+             debug_dump_replayer_.GetNextEvent()) {
+    debug_dump_replayer_.RunNextEvent();
+    if (event->type() == audioproc::Event::CONFIG) {
+      const audioproc::Config* msg = &event->config();
+      ASSERT_TRUE(msg->has_experiments_description());
+      EXPECT_PRED_FORMAT2(::testing::IsSubstring, "EchoController",
+                          msg->experiments_description().c_str());
+    }
+  }
+}
+
+TEST_F(DebugDumpTest, VerifyEmptyExperimentalString) {
+  DebugDumpGenerator generator(/*apm_config=*/{});
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+
+  DebugDumpReplayer debug_dump_replayer_;
+
+  ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
+
+  while (const absl::optional<audioproc::Event> event =
+             debug_dump_replayer_.GetNextEvent()) {
+    debug_dump_replayer_.RunNextEvent();
+    if (event->type() == audioproc::Event::CONFIG) {
+      const audioproc::Config* msg = &event->config();
+      ASSERT_TRUE(msg->has_experiments_description());
+      EXPECT_EQ(0u, msg->experiments_description().size());
+    }
+  }
+}
+
+// AGC is not supported on Android or iOS.
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+#define MAYBE_ToggleAgc DISABLED_ToggleAgc
+#else
+#define MAYBE_ToggleAgc ToggleAgc
+#endif
+TEST_F(DebugDumpTest, MAYBE_ToggleAgc) {
+  DebugDumpGenerator generator(/*apm_config=*/{});
+  generator.StartRecording();
+  generator.Process(100);
+
+  AudioProcessing::Config apm_config = generator.apm()->GetConfig();
+  apm_config.gain_controller1.enabled = !apm_config.gain_controller1.enabled;
+  generator.apm()->ApplyConfig(apm_config);
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, ToggleNs) {
+  DebugDumpGenerator generator(/*apm_config=*/{});
+  generator.StartRecording();
+  generator.Process(100);
+
+  AudioProcessing::Config apm_config = generator.apm()->GetConfig();
+  apm_config.noise_suppression.enabled = !apm_config.noise_suppression.enabled;
+  generator.apm()->ApplyConfig(apm_config);
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, TransientSuppressionOn) {
+  DebugDumpGenerator generator(/*apm_config=*/{});
+
+  AudioProcessing::Config apm_config = generator.apm()->GetConfig();
+  apm_config.transient_suppression.enabled = true;
+  generator.apm()->ApplyConfig(apm_config);
+
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+TEST_F(DebugDumpTest, PreAmplifierIsOn) {
+  AudioProcessing::Config apm_config;
+  apm_config.pre_amplifier.enabled = true;
+  DebugDumpGenerator generator(apm_config);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(generator.dump_file_name());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.cc b/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.cc
new file mode 100644
index 0000000000..1d36b954f9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.cc
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void RandomizeSampleVector(Random* random_generator, rtc::ArrayView<float> v) {
+  RandomizeSampleVector(random_generator, v,
+                        /*amplitude=*/32767.f);
+}
+
+void RandomizeSampleVector(Random* random_generator,
+                           rtc::ArrayView<float> v,
+                           float amplitude) {
+  for (auto& v_k : v) {
+    v_k = 2 * amplitude * random_generator->Rand<float>() - amplitude;
+  }
+}
+
+template <typename T>
+void DelayBuffer<T>::Delay(rtc::ArrayView<const T> x,
+                           rtc::ArrayView<T> x_delayed) {
+  RTC_DCHECK_EQ(x.size(), x_delayed.size());
+  if (buffer_.empty()) {
+    std::copy(x.begin(), x.end(), x_delayed.begin());
+  } else {
+    for (size_t k = 0; k < x.size(); ++k) {
+      x_delayed[k] = buffer_[next_insert_index_];
+      buffer_[next_insert_index_] = x[k];
+      next_insert_index_ = (next_insert_index_ + 1) % buffer_.size();
+    }
+  }
+}
+
+template class DelayBuffer<float>;
+template class DelayBuffer<int>;
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.h b/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.h
new file mode 100644
index 0000000000..0d70cd39c6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_ECHO_CANCELLER_TEST_TOOLS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_ECHO_CANCELLER_TEST_TOOLS_H_
+
+#include <algorithm>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+
+// Randomizes the elements in a vector with values -32767.f:32767.f.
+void RandomizeSampleVector(Random* random_generator, rtc::ArrayView<float> v);
+
+// Randomizes the elements in a vector with values -amplitude:amplitude.
+void RandomizeSampleVector(Random* random_generator,
+                           rtc::ArrayView<float> v,
+                           float amplitude);
+
+// Class for delaying a signal a fixed number of samples.
+template <typename T>
+class DelayBuffer {
+ public:
+  explicit DelayBuffer(size_t delay) : buffer_(delay) {}
+  ~DelayBuffer() = default;
+
+  // Produces a delayed signal copy of x.
+  void Delay(rtc::ArrayView<const T> x, rtc::ArrayView<T> x_delayed);
+
+ private:
+  std::vector<T> buffer_;
+  size_t next_insert_index_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_ECHO_CANCELLER_TEST_TOOLS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools_unittest.cc b/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools_unittest.cc
new file mode 100644
index 0000000000..164d28fa16
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/echo_canceller_test_tools_unittest.cc
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/echo_canceller_test_tools.h"
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/random.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(EchoCancellerTestTools, FloatDelayBuffer) {
+  constexpr size_t kDelay = 10;
+  DelayBuffer<float> delay_buffer(kDelay);
+  std::vector<float> v(1000, 0.f);
+  for (size_t k = 0; k < v.size(); ++k) {
+    v[k] = k;
+  }
+  std::vector<float> v_delayed = v;
+  constexpr size_t kBlockSize = 50;
+  for (size_t k = 0; k < rtc::CheckedDivExact(v.size(), kBlockSize); ++k) {
+    delay_buffer.Delay(
+        rtc::ArrayView<const float>(&v[k * kBlockSize], kBlockSize),
+        rtc::ArrayView<float>(&v_delayed[k * kBlockSize], kBlockSize));
+  }
+  for (size_t k = kDelay; k < v.size(); ++k) {
+    EXPECT_EQ(v[k - kDelay], v_delayed[k]);
+  }
+}
+
+TEST(EchoCancellerTestTools, IntDelayBuffer) {
+  constexpr size_t kDelay = 10;
+  DelayBuffer<int> delay_buffer(kDelay);
+  std::vector<int> v(1000, 0);
+  for (size_t k = 0; k < v.size(); ++k) {
+    v[k] = k;
+  }
+  std::vector<int> v_delayed = v;
+  const size_t kBlockSize = 50;
+  for (size_t k = 0; k < rtc::CheckedDivExact(v.size(), kBlockSize); ++k) {
+    delay_buffer.Delay(
+        rtc::ArrayView<const int>(&v[k * kBlockSize], kBlockSize),
+        rtc::ArrayView<int>(&v_delayed[k * kBlockSize], kBlockSize));
+  }
+  for (size_t k = kDelay; k < v.size(); ++k) {
+    EXPECT_EQ(v[k - kDelay], v_delayed[k]);
+  }
+}
+
+TEST(EchoCancellerTestTools, RandomizeSampleVector) {
+  Random random_generator(42U);
+  std::vector<float> v(50, 0.f);
+  std::vector<float> v_ref = v;
+  RandomizeSampleVector(&random_generator, v);
+  EXPECT_NE(v, v_ref);
+  v_ref = v;
+  RandomizeSampleVector(&random_generator, v);
+  EXPECT_NE(v, v_ref);
+}
+
+TEST(EchoCancellerTestTools, RandomizeSampleVectorWithAmplitude) {
+  Random random_generator(42U);
+  std::vector<float> v(50, 0.f);
+  RandomizeSampleVector(&random_generator, v, 1000.f);
+  EXPECT_GE(1000.f, *std::max_element(v.begin(), v.end()));
+  EXPECT_LE(-1000.f, *std::min_element(v.begin(), v.end()));
+  RandomizeSampleVector(&random_generator, v, 100.f);
+  EXPECT_GE(100.f, *std::max_element(v.begin(), v.end()));
+  EXPECT_LE(-100.f, *std::min_element(v.begin(), v.end()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/echo_control_mock.h b/third_party/libwebrtc/modules/audio_processing/test/echo_control_mock.h
new file mode 100644
index 0000000000..763d6e4f0b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/echo_control_mock.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_ECHO_CONTROL_MOCK_H_
+#define MODULES_AUDIO_PROCESSING_TEST_ECHO_CONTROL_MOCK_H_
+
+#include "api/audio/echo_control.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class AudioBuffer;
+
+class MockEchoControl : public EchoControl {
+ public:
+  MOCK_METHOD(void, AnalyzeRender, (AudioBuffer * render), (override));
+  MOCK_METHOD(void, AnalyzeCapture, (AudioBuffer * capture), (override));
+  MOCK_METHOD(void,
+              ProcessCapture,
+              (AudioBuffer * capture, bool echo_path_change),
+              (override));
+  MOCK_METHOD(void,
+              ProcessCapture,
+              (AudioBuffer * capture,
+               AudioBuffer* linear_output,
+               bool echo_path_change),
+              (override));
+  MOCK_METHOD(EchoControl::Metrics, GetMetrics, (), (const, override));
+  MOCK_METHOD(void, SetAudioBufferDelay, (int delay_ms), (override));
+  MOCK_METHOD(void,
+              SetCaptureOutputUsage,
+              (bool capture_output_used),
+              (override));
+  MOCK_METHOD(bool, ActiveProcessing, (), (const, override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_ECHO_CONTROL_MOCK_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.cc b/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.cc
new file mode 100644
index 0000000000..3fd80b2ede
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.cc
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/fake_recording_device.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/agc2/gain_map_internal.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+constexpr float kFloatSampleMin = -32768.f;
+constexpr float kFloatSampleMax = 32767.0f;
+
+}  // namespace
+
+// Abstract class for the different fake recording devices.
+class FakeRecordingDeviceWorker {
+ public:
+  explicit FakeRecordingDeviceWorker(const int initial_mic_level)
+      : mic_level_(initial_mic_level) {}
+  int mic_level() const { return mic_level_; }
+  void set_mic_level(const int level) { mic_level_ = level; }
+  void set_undo_mic_level(const int level) { undo_mic_level_ = level; }
+  virtual ~FakeRecordingDeviceWorker() = default;
+  virtual void ModifyBufferInt16(rtc::ArrayView<int16_t> buffer) = 0;
+  virtual void ModifyBufferFloat(ChannelBuffer<float>* buffer) = 0;
+
+ protected:
+  // Mic level to simulate.
+  int mic_level_;
+  // Optional mic level to undo.
+  absl::optional<int> undo_mic_level_;
+};
+
+namespace {
+
+// Identity fake recording device. The samples are not modified, which is
+// equivalent to a constant gain curve at 1.0 - only used for testing.
+class FakeRecordingDeviceIdentity final : public FakeRecordingDeviceWorker {
+ public:
+  explicit FakeRecordingDeviceIdentity(const int initial_mic_level)
+      : FakeRecordingDeviceWorker(initial_mic_level) {}
+  ~FakeRecordingDeviceIdentity() override = default;
+  void ModifyBufferInt16(rtc::ArrayView<int16_t> buffer) override {}
+  void ModifyBufferFloat(ChannelBuffer<float>* buffer) override {}
+};
+
+// Linear fake recording device. The gain curve is a linear function mapping the
+// mic levels range [0, 255] to [0.0, 1.0].
+class FakeRecordingDeviceLinear final : public FakeRecordingDeviceWorker {
+ public:
+  explicit FakeRecordingDeviceLinear(const int initial_mic_level)
+      : FakeRecordingDeviceWorker(initial_mic_level) {}
+  ~FakeRecordingDeviceLinear() override = default;
+  void ModifyBufferInt16(rtc::ArrayView<int16_t> buffer) override {
+    const size_t number_of_samples = buffer.size();
+    int16_t* data = buffer.data();
+    // If an undo level is specified, virtually restore the unmodified
+    // microphone level; otherwise simulate the mic gain only.
+    const float divisor =
+        (undo_mic_level_ && *undo_mic_level_ > 0) ? *undo_mic_level_ : 255.f;
+    for (size_t i = 0; i < number_of_samples; ++i) {
+      data[i] = rtc::saturated_cast<int16_t>(data[i] * mic_level_ / divisor);
+    }
+  }
+  void ModifyBufferFloat(ChannelBuffer<float>* buffer) override {
+    // If an undo level is specified, virtually restore the unmodified
+    // microphone level; otherwise simulate the mic gain only.
+    const float divisor =
+        (undo_mic_level_ && *undo_mic_level_ > 0) ? *undo_mic_level_ : 255.f;
+    for (size_t c = 0; c < buffer->num_channels(); ++c) {
+      for (size_t i = 0; i < buffer->num_frames(); ++i) {
+        buffer->channels()[c][i] =
+            rtc::SafeClamp(buffer->channels()[c][i] * mic_level_ / divisor,
+                           kFloatSampleMin, kFloatSampleMax);
+      }
+    }
+  }
+};
+
+float ComputeAgcLinearFactor(const absl::optional<int>& undo_mic_level,
+                             int mic_level) {
+  // If an undo level is specified, virtually restore the unmodified
+  // microphone level; otherwise simulate the mic gain only.
+  const int undo_level =
+      (undo_mic_level && *undo_mic_level > 0) ? *undo_mic_level : 100;
+  return DbToRatio(kGainMap[mic_level] - kGainMap[undo_level]);
+}
+
+// Roughly dB-scale fake recording device. Valid levels are [0, 255]. The mic
+// applies a gain from kGainMap in agc/gain_map_internal.h.
+class FakeRecordingDeviceAgc final : public FakeRecordingDeviceWorker {
+ public:
+  explicit FakeRecordingDeviceAgc(const int initial_mic_level)
+      : FakeRecordingDeviceWorker(initial_mic_level) {}
+  ~FakeRecordingDeviceAgc() override = default;
+  void ModifyBufferInt16(rtc::ArrayView<int16_t> buffer) override {
+    const float scaling_factor =
+        ComputeAgcLinearFactor(undo_mic_level_, mic_level_);
+    const size_t number_of_samples = buffer.size();
+    int16_t* data = buffer.data();
+    for (size_t i = 0; i < number_of_samples; ++i) {
+      data[i] = rtc::saturated_cast<int16_t>(data[i] * scaling_factor);
+    }
+  }
+  void ModifyBufferFloat(ChannelBuffer<float>* buffer) override {
+    const float scaling_factor =
+        ComputeAgcLinearFactor(undo_mic_level_, mic_level_);
+    for (size_t c = 0; c < buffer->num_channels(); ++c) {
+      for (size_t i = 0; i < buffer->num_frames(); ++i) {
+        buffer->channels()[c][i] =
+            rtc::SafeClamp(buffer->channels()[c][i] * scaling_factor,
+                           kFloatSampleMin, kFloatSampleMax);
+      }
+    }
+  }
+};
+
+}  // namespace
+
+FakeRecordingDevice::FakeRecordingDevice(int initial_mic_level,
+                                         int device_kind) {
+  switch (device_kind) {
+    case 0:
+      worker_ =
+          std::make_unique<FakeRecordingDeviceIdentity>(initial_mic_level);
+      break;
+    case 1:
+      worker_ = std::make_unique<FakeRecordingDeviceLinear>(initial_mic_level);
+      break;
+    case 2:
+      worker_ = std::make_unique<FakeRecordingDeviceAgc>(initial_mic_level);
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+}
+
+FakeRecordingDevice::~FakeRecordingDevice() = default;
+
+int FakeRecordingDevice::MicLevel() const {
+  RTC_CHECK(worker_);
+  return worker_->mic_level();
+}
+
+void FakeRecordingDevice::SetMicLevel(const int level) {
+  RTC_CHECK(worker_);
+  if (level != worker_->mic_level())
+    RTC_LOG(LS_INFO) << "Simulate mic level update: " << level;
+  worker_->set_mic_level(level);
+}
+
+void FakeRecordingDevice::SetUndoMicLevel(const int level) {
+  RTC_DCHECK(worker_);
+  // TODO(alessiob): The behavior with undo level equal to zero is not clear yet
+  // and will be defined in future CLs once more FakeRecordingDeviceWorker
+  // implementations need to be added.
+  RTC_CHECK(level > 0) << "Zero undo mic level is unsupported";
+  worker_->set_undo_mic_level(level);
+}
+
+void FakeRecordingDevice::SimulateAnalogGain(rtc::ArrayView<int16_t> buffer) {
+  RTC_DCHECK(worker_);
+  worker_->ModifyBufferInt16(buffer);
+}
+
+void FakeRecordingDevice::SimulateAnalogGain(ChannelBuffer<float>* buffer) {
+  RTC_DCHECK(worker_);
+  worker_->ModifyBufferFloat(buffer);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.h b/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.h
new file mode 100644
index 0000000000..da3c0cf794
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_FAKE_RECORDING_DEVICE_H_
+#define MODULES_AUDIO_PROCESSING_TEST_FAKE_RECORDING_DEVICE_H_
+
+#include <algorithm>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "common_audio/channel_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeRecordingDeviceWorker;
+
+// Class for simulating a microphone with analog gain.
+//
+// The intended modes of operation are the following:
+//
+// FakeRecordingDevice fake_mic(255, 1);
+//
+// fake_mic.SetMicLevel(170);
+// fake_mic.SimulateAnalogGain(buffer);
+//
+// When the mic level to undo is known:
+//
+// fake_mic.SetMicLevel(170);
+// fake_mic.SetUndoMicLevel(30);
+// fake_mic.SimulateAnalogGain(buffer);
+//
+// The second option virtually restores the unmodified microphone level. Calling
+// SimulateAnalogGain() will first "undo" the gain applied by the real
+// microphone (e.g., 30).
+class FakeRecordingDevice final {
+ public:
+  FakeRecordingDevice(int initial_mic_level, int device_kind);
+  ~FakeRecordingDevice();
+
+  int MicLevel() const;
+  void SetMicLevel(int level);
+  void SetUndoMicLevel(int level);
+
+  // Simulates the analog gain.
+  // If `real_device_level` is a valid level, the unmodified mic signal is
+  // virtually restored. To skip the latter step set `real_device_level` to
+  // an empty value.
+  void SimulateAnalogGain(rtc::ArrayView<int16_t> buffer);
+
+  // Simulates the analog gain.
+  // If `real_device_level` is a valid level, the unmodified mic signal is
+  // virtually restored. To skip the latter step set `real_device_level` to
+  // an empty value.
+  void SimulateAnalogGain(ChannelBuffer<float>* buffer);
+
+ private:
+  // Fake recording device worker.
+  std::unique_ptr<FakeRecordingDeviceWorker> worker_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_FAKE_RECORDING_DEVICE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device_unittest.cc b/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device_unittest.cc
new file mode 100644
index 0000000000..2ac8b1dc48
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device_unittest.cc
@@ -0,0 +1,231 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/fake_recording_device.h"
+
+#include <cmath>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr int kInitialMicLevel = 100;
+
+// TODO(alessiob): Add new fake recording device kind values here as they are
+// added in FakeRecordingDevice::FakeRecordingDevice.
+const std::vector<int> kFakeRecDeviceKinds = {0, 1, 2};
+
+const std::vector<std::vector<float>> kTestMultiChannelSamples{
+    std::vector<float>{-10.f, -1.f, -0.1f, 0.f, 0.1f, 1.f, 10.f}};
+
+// Writes samples into ChannelBuffer<float>.
+void WritesDataIntoChannelBuffer(const std::vector<std::vector<float>>& data,
+                                 ChannelBuffer<float>* buff) {
+  EXPECT_EQ(data.size(), buff->num_channels());
+  EXPECT_EQ(data[0].size(), buff->num_frames());
+  for (size_t c = 0; c < buff->num_channels(); ++c) {
+    for (size_t f = 0; f < buff->num_frames(); ++f) {
+      buff->channels()[c][f] = data[c][f];
+    }
+  }
+}
+
+std::unique_ptr<ChannelBuffer<float>> CreateChannelBufferWithData(
+    const std::vector<std::vector<float>>& data) {
+  auto buff =
+      std::make_unique<ChannelBuffer<float>>(data[0].size(), data.size());
+  WritesDataIntoChannelBuffer(data, buff.get());
+  return buff;
+}
+
+// Checks that the samples modified using monotonic level values are also
+// monotonic.
+void CheckIfMonotoneSamplesModules(const ChannelBuffer<float>* prev,
+                                   const ChannelBuffer<float>* curr) {
+  RTC_DCHECK_EQ(prev->num_channels(), curr->num_channels());
+  RTC_DCHECK_EQ(prev->num_frames(), curr->num_frames());
+  bool valid = true;
+  for (size_t i = 0; i < prev->num_channels(); ++i) {
+    for (size_t j = 0; j < prev->num_frames(); ++j) {
+      valid = std::fabs(prev->channels()[i][j]) <=
+              std::fabs(curr->channels()[i][j]);
+      if (!valid) {
+        break;
+      }
+    }
+    if (!valid) {
+      break;
+    }
+  }
+  EXPECT_TRUE(valid);
+}
+
+// Checks that the samples in each pair have the same sign unless the sample in
+// `dst` is zero (because of zero gain).
+void CheckSameSign(const ChannelBuffer<float>* src,
+                   const ChannelBuffer<float>* dst) {
+  RTC_DCHECK_EQ(src->num_channels(), dst->num_channels());
+  RTC_DCHECK_EQ(src->num_frames(), dst->num_frames());
+  const auto fsgn = [](float x) { return ((x < 0) ? -1 : (x > 0) ? 1 : 0); };
+  bool valid = true;
+  for (size_t i = 0; i < src->num_channels(); ++i) {
+    for (size_t j = 0; j < src->num_frames(); ++j) {
+      valid = dst->channels()[i][j] == 0.0f ||
+              fsgn(src->channels()[i][j]) == fsgn(dst->channels()[i][j]);
+      if (!valid) {
+        break;
+      }
+    }
+    if (!valid) {
+      break;
+    }
+  }
+  EXPECT_TRUE(valid);
+}
+
+std::string FakeRecordingDeviceKindToString(int fake_rec_device_kind) {
+  rtc::StringBuilder ss;
+  ss << "fake recording device: " << fake_rec_device_kind;
+  return ss.Release();
+}
+
+std::string AnalogLevelToString(int level) {
+  rtc::StringBuilder ss;
+  ss << "analog level: " << level;
+  return ss.Release();
+}
+
+}  // namespace
+
+TEST(FakeRecordingDevice, CheckHelperFunctions) {
+  constexpr size_t kC = 0;  // Channel index.
+  constexpr size_t kS = 1;  // Sample index.
+
+  // Check read.
+  auto buff = CreateChannelBufferWithData(kTestMultiChannelSamples);
+  for (size_t c = 0; c < kTestMultiChannelSamples.size(); ++c) {
+    for (size_t s = 0; s < kTestMultiChannelSamples[0].size(); ++s) {
+      EXPECT_EQ(kTestMultiChannelSamples[c][s], buff->channels()[c][s]);
+    }
+  }
+
+  // Check write.
+  buff->channels()[kC][kS] = -5.0f;
+  RTC_DCHECK_NE(buff->channels()[kC][kS], kTestMultiChannelSamples[kC][kS]);
+
+  // Check reset.
+  WritesDataIntoChannelBuffer(kTestMultiChannelSamples, buff.get());
+  EXPECT_EQ(buff->channels()[kC][kS], kTestMultiChannelSamples[kC][kS]);
+}
+
+// Implicitly checks that changes to the mic and undo levels are visible to the
+// FakeRecordingDeviceWorker implementation are injected in FakeRecordingDevice.
+TEST(FakeRecordingDevice, TestWorkerAbstractClass) {
+  FakeRecordingDevice fake_recording_device(kInitialMicLevel, 1);
+
+  auto buff1 = CreateChannelBufferWithData(kTestMultiChannelSamples);
+  fake_recording_device.SetMicLevel(100);
+  fake_recording_device.SimulateAnalogGain(buff1.get());
+
+  auto buff2 = CreateChannelBufferWithData(kTestMultiChannelSamples);
+  fake_recording_device.SetMicLevel(200);
+  fake_recording_device.SimulateAnalogGain(buff2.get());
+
+  for (size_t c = 0; c < kTestMultiChannelSamples.size(); ++c) {
+    for (size_t s = 0; s < kTestMultiChannelSamples[0].size(); ++s) {
+      EXPECT_LE(std::abs(buff1->channels()[c][s]),
+                std::abs(buff2->channels()[c][s]));
+    }
+  }
+
+  auto buff3 = CreateChannelBufferWithData(kTestMultiChannelSamples);
+  fake_recording_device.SetMicLevel(200);
+  fake_recording_device.SetUndoMicLevel(100);
+  fake_recording_device.SimulateAnalogGain(buff3.get());
+
+  for (size_t c = 0; c < kTestMultiChannelSamples.size(); ++c) {
+    for (size_t s = 0; s < kTestMultiChannelSamples[0].size(); ++s) {
+      EXPECT_LE(std::abs(buff1->channels()[c][s]),
+                std::abs(buff3->channels()[c][s]));
+      EXPECT_LE(std::abs(buff2->channels()[c][s]),
+                std::abs(buff3->channels()[c][s]));
+    }
+  }
+}
+
+TEST(FakeRecordingDevice, GainCurveShouldBeMonotone) {
+  // Create input-output buffers.
+  auto buff_prev = CreateChannelBufferWithData(kTestMultiChannelSamples);
+  auto buff_curr = CreateChannelBufferWithData(kTestMultiChannelSamples);
+
+  // Test different mappings.
+  for (auto fake_rec_device_kind : kFakeRecDeviceKinds) {
+    SCOPED_TRACE(FakeRecordingDeviceKindToString(fake_rec_device_kind));
+    FakeRecordingDevice fake_recording_device(kInitialMicLevel,
+                                              fake_rec_device_kind);
+    // TODO(alessiob): The test below is designed for state-less recording
+    // devices. If, for instance, a device has memory, the test might need
+    // to be redesigned (e.g., re-initialize fake recording device).
+
+    // Apply lowest analog level.
+    WritesDataIntoChannelBuffer(kTestMultiChannelSamples, buff_prev.get());
+    fake_recording_device.SetMicLevel(0);
+    fake_recording_device.SimulateAnalogGain(buff_prev.get());
+
+    // Increment analog level to check monotonicity.
+    for (int i = 1; i <= 255; ++i) {
+      SCOPED_TRACE(AnalogLevelToString(i));
+      WritesDataIntoChannelBuffer(kTestMultiChannelSamples, buff_curr.get());
+      fake_recording_device.SetMicLevel(i);
+      fake_recording_device.SimulateAnalogGain(buff_curr.get());
+      CheckIfMonotoneSamplesModules(buff_prev.get(), buff_curr.get());
+
+      // Update prev.
+      buff_prev.swap(buff_curr);
+    }
+  }
+}
+
+TEST(FakeRecordingDevice, GainCurveShouldNotChangeSign) {
+  // Create view on original samples.
+  std::unique_ptr<const ChannelBuffer<float>> buff_orig =
+      CreateChannelBufferWithData(kTestMultiChannelSamples);
+
+  // Create output buffer.
+  auto buff = CreateChannelBufferWithData(kTestMultiChannelSamples);
+
+  // Test different mappings.
+  for (auto fake_rec_device_kind : kFakeRecDeviceKinds) {
+    SCOPED_TRACE(FakeRecordingDeviceKindToString(fake_rec_device_kind));
+    FakeRecordingDevice fake_recording_device(kInitialMicLevel,
+                                              fake_rec_device_kind);
+
+    // TODO(alessiob): The test below is designed for state-less recording
+    // devices. If, for instance, a device has memory, the test might need
+    // to be redesigned (e.g., re-initialize fake recording device).
+    for (int i = 0; i <= 255; ++i) {
+      SCOPED_TRACE(AnalogLevelToString(i));
+      WritesDataIntoChannelBuffer(kTestMultiChannelSamples, buff.get());
+      fake_recording_device.SetMicLevel(i);
+      fake_recording_device.SimulateAnalogGain(buff.get());
+      CheckSameSign(buff_orig.get(), buff.get());
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/performance_timer.cc b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.cc
new file mode 100644
index 0000000000..1a82258903
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.cc
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/performance_timer.h"
+
+#include <math.h>
+
+#include <numeric>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+PerformanceTimer::PerformanceTimer(int num_frames_to_process)
+    : clock_(webrtc::Clock::GetRealTimeClock()) {
+  timestamps_us_.reserve(num_frames_to_process);
+}
+
+PerformanceTimer::~PerformanceTimer() = default;
+
+void PerformanceTimer::StartTimer() {
+  start_timestamp_us_ = clock_->TimeInMicroseconds();
+}
+
+void PerformanceTimer::StopTimer() {
+  RTC_DCHECK(start_timestamp_us_);
+  timestamps_us_.push_back(clock_->TimeInMicroseconds() - *start_timestamp_us_);
+}
+
+double PerformanceTimer::GetDurationAverage() const {
+  return GetDurationAverage(0);
+}
+
+double PerformanceTimer::GetDurationStandardDeviation() const {
+  return GetDurationStandardDeviation(0);
+}
+
+double PerformanceTimer::GetDurationAverage(
+    size_t number_of_warmup_samples) const {
+  RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples);
+  const size_t number_of_samples =
+      timestamps_us_.size() - number_of_warmup_samples;
+  return static_cast<double>(
+             std::accumulate(timestamps_us_.begin() + number_of_warmup_samples,
+                             timestamps_us_.end(), static_cast<int64_t>(0))) /
+         number_of_samples;
+}
+
+double PerformanceTimer::GetDurationStandardDeviation(
+    size_t number_of_warmup_samples) const {
+  RTC_DCHECK_GT(timestamps_us_.size(), number_of_warmup_samples);
+  const size_t number_of_samples =
+      timestamps_us_.size() - number_of_warmup_samples;
+  RTC_DCHECK_GT(number_of_samples, 0);
+  double average_duration = GetDurationAverage(number_of_warmup_samples);
+
+  double variance = std::accumulate(
+      timestamps_us_.begin() + number_of_warmup_samples, timestamps_us_.end(),
+      0.0, [average_duration](const double& a, const int64_t& b) {
+        return a + (b - average_duration) * (b - average_duration);
+      });
+
+  return sqrt(variance / number_of_samples);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/performance_timer.h b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.h
new file mode 100644
index 0000000000..5375ba74e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/performance_timer.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
+#define MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace test {
+
+class PerformanceTimer {
+ public:
+  explicit PerformanceTimer(int num_frames_to_process);
+  ~PerformanceTimer();
+
+  void StartTimer();
+  void StopTimer();
+
+  double GetDurationAverage() const;
+  double GetDurationStandardDeviation() const;
+
+  // These methods are the same as those above, but they ignore the first
+  // `number_of_warmup_samples` measurements.
+  double GetDurationAverage(size_t number_of_warmup_samples) const;
+  double GetDurationStandardDeviation(size_t number_of_warmup_samples) const;
+
+ private:
+  webrtc::Clock* clock_;
+  absl::optional<int64_t> start_timestamp_us_;
+  std::vector<int64_t> timestamps_us_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_PERFORMANCE_TIMER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.cc b/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.cc
new file mode 100644
index 0000000000..75574961b0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/protobuf_utils.h"
+
+#include <memory>
+
+#include "rtc_base/system/arch.h"
+
+namespace {
+// Allocates new memory in the memory owned by the unique_ptr to fit the raw
+// message and returns the number of bytes read when having a string stream as
+// input.
+size_t ReadMessageBytesFromString(std::stringstream* input,
+                                  std::unique_ptr<uint8_t[]>* bytes) {
+  int32_t size = 0;
+  input->read(reinterpret_cast<char*>(&size), sizeof(int32_t));
+  int32_t size_read = input->gcount();
+  if (size_read != sizeof(int32_t))
+    return 0;
+  if (size <= 0)
+    return 0;
+
+  *bytes = std::make_unique<uint8_t[]>(size);
+  input->read(reinterpret_cast<char*>(bytes->get()),
+              size * sizeof((*bytes)[0]));
+  size_read = input->gcount();
+  return size_read == size ? size : 0;
+}
+}  // namespace
+
+namespace webrtc {
+
+size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes) {
+// The "wire format" for the size is little-endian. Assume we're running on
+// a little-endian machine.
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Need to convert messsage from little-endian."
+#endif
+  int32_t size = 0;
+  if (fread(&size, sizeof(size), 1, file) != 1)
+    return 0;
+  if (size <= 0)
+    return 0;
+
+  *bytes = std::make_unique<uint8_t[]>(size);
+  return fread(bytes->get(), sizeof((*bytes)[0]), size, file);
+}
+
+// Returns true on success, false on error or end-of-file.
+bool ReadMessageFromFile(FILE* file, MessageLite* msg) {
+  std::unique_ptr<uint8_t[]> bytes;
+  size_t size = ReadMessageBytesFromFile(file, &bytes);
+  if (!size)
+    return false;
+
+  msg->Clear();
+  return msg->ParseFromArray(bytes.get(), size);
+}
+
+// Returns true on success, false on error or end of string stream.
+bool ReadMessageFromString(std::stringstream* input, MessageLite* msg) {
+  std::unique_ptr<uint8_t[]> bytes;
+  size_t size = ReadMessageBytesFromString(input, &bytes);
+  if (!size)
+    return false;
+
+  msg->Clear();
+  return msg->ParseFromArray(bytes.get(), size);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.h b/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.h
new file mode 100644
index 0000000000..b9c2e819f9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/protobuf_utils.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_
+
+#include <memory>
+#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
+
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/protobuf_utils.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "modules/audio_processing/debug.pb.h"
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+
+// Allocates new memory in the unique_ptr to fit the raw message and returns the
+// number of bytes read.
+size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes);
+
+// Returns true on success, false on error or end-of-file.
+bool ReadMessageFromFile(FILE* file, MessageLite* msg);
+
+// Returns true on success, false on error or end of string stream.
+bool ReadMessageFromString(
+    std::stringstream* input,  // no-presubmit-check TODO(webrtc:8982)
+    MessageLite* msg);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn
new file mode 100644
index 0000000000..e53a829623
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn
@@ -0,0 +1,170 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../../webrtc.gni")
+
+if (!build_with_chromium) {
+  group("py_quality_assessment") {
+    testonly = true
+    deps = [
+      ":scripts",
+      ":unit_tests",
+    ]
+  }
+
+  copy("scripts") {
+    testonly = true
+    sources = [
+      "README.md",
+      "apm_quality_assessment.py",
+      "apm_quality_assessment.sh",
+      "apm_quality_assessment_boxplot.py",
+      "apm_quality_assessment_export.py",
+      "apm_quality_assessment_gencfgs.py",
+      "apm_quality_assessment_optimize.py",
+    ]
+    outputs = [ "$root_build_dir/py_quality_assessment/{{source_file_part}}" ]
+    deps = [
+      ":apm_configs",
+      ":lib",
+      ":output",
+      "../../../../resources/audio_processing/test/py_quality_assessment:probing_signals",
+      "../../../../rtc_tools:audioproc_f",
+    ]
+  }
+
+  copy("apm_configs") {
+    testonly = true
+    sources = [ "apm_configs/default.json" ]
+    visibility = [ ":*" ]  # Only targets in this file can depend on this.
+    outputs = [
+      "$root_build_dir/py_quality_assessment/apm_configs/{{source_file_part}}",
+    ]
+  }  # apm_configs
+
+  copy("lib") {
+    testonly = true
+    sources = [
+      "quality_assessment/__init__.py",
+      "quality_assessment/annotations.py",
+      "quality_assessment/audioproc_wrapper.py",
+      "quality_assessment/collect_data.py",
+      "quality_assessment/data_access.py",
+      "quality_assessment/echo_path_simulation.py",
+      "quality_assessment/echo_path_simulation_factory.py",
+      "quality_assessment/eval_scores.py",
+      "quality_assessment/eval_scores_factory.py",
+      "quality_assessment/evaluation.py",
+      "quality_assessment/exceptions.py",
+      "quality_assessment/export.py",
+      "quality_assessment/export_unittest.py",
+      "quality_assessment/external_vad.py",
+      "quality_assessment/input_mixer.py",
+      "quality_assessment/input_signal_creator.py",
+      "quality_assessment/results.css",
+      "quality_assessment/results.js",
+      "quality_assessment/signal_processing.py",
+      "quality_assessment/simulation.py",
+      "quality_assessment/test_data_generation.py",
+      "quality_assessment/test_data_generation_factory.py",
+    ]
+    visibility = [ ":*" ]  # Only targets in this file can depend on this.
+    outputs = [ "$root_build_dir/py_quality_assessment/quality_assessment/{{source_file_part}}" ]
+    deps = [ "../../../../resources/audio_processing/test/py_quality_assessment:noise_tracks" ]
+  }
+
+  copy("output") {
+    testonly = true
+    sources = [ "output/README.md" ]
+    visibility = [ ":*" ]  # Only targets in this file can depend on this.
+    outputs =
+        [ "$root_build_dir/py_quality_assessment/output/{{source_file_part}}" ]
+  }
+
+  group("unit_tests") {
+    testonly = true
+    visibility = [ ":*" ]  # Only targets in this file can depend on this.
+    deps = [
+      ":apm_vad",
+      ":fake_polqa",
+      ":lib_unit_tests",
+      ":scripts_unit_tests",
+      ":vad",
+    ]
+  }
+
+  rtc_executable("fake_polqa") {
+    testonly = true
+    sources = [ "quality_assessment/fake_polqa.cc" ]
+    visibility = [ ":*" ]  # Only targets in this file can depend on this.
+    output_dir = "${root_out_dir}/py_quality_assessment/quality_assessment"
+    deps = [
+      "../../../../rtc_base:checks",
+      "//third_party/abseil-cpp/absl/strings",
+    ]
+  }
+
+  rtc_executable("vad") {
+    testonly = true
+    sources = [ "quality_assessment/vad.cc" ]
+    deps = [
+      "../../../../common_audio",
+      "../../../../rtc_base:logging",
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+    ]
+  }
+
+  rtc_executable("apm_vad") {
+    testonly = true
+    sources = [ "quality_assessment/apm_vad.cc" ]
+    deps = [
+      "../..",
+      "../../../../common_audio",
+      "../../../../rtc_base:logging",
+      "../../vad",
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+    ]
+  }
+
+  rtc_executable("sound_level") {
+    testonly = true
+    sources = [ "quality_assessment/sound_level.cc" ]
+    deps = [
+      "../..",
+      "../../../../common_audio",
+      "../../../../rtc_base:logging",
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+    ]
+  }
+
+  copy("lib_unit_tests") {
+    testonly = true
+    sources = [
+      "quality_assessment/annotations_unittest.py",
+      "quality_assessment/echo_path_simulation_unittest.py",
+      "quality_assessment/eval_scores_unittest.py",
+      "quality_assessment/fake_external_vad.py",
+      "quality_assessment/input_mixer_unittest.py",
+      "quality_assessment/signal_processing_unittest.py",
+      "quality_assessment/simulation_unittest.py",
+      "quality_assessment/test_data_generation_unittest.py",
+    ]
+    visibility = [ ":*" ]  # Only targets in this file can depend on this.
+    outputs = [ "$root_build_dir/py_quality_assessment/quality_assessment/{{source_file_part}}" ]
+  }
+
+  copy("scripts_unit_tests") {
+    testonly = true
+    sources = [ "apm_quality_assessment_unittest.py" ]
+    visibility = [ ":*" ]  # Only targets in this file can depend on this.
+    outputs = [ "$root_build_dir/py_quality_assessment/{{source_file_part}}" ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/OWNERS b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/OWNERS
new file mode 100644
index 0000000000..9f56bb830d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/OWNERS
@@ -0,0 +1,5 @@
+aleloi@webrtc.org
+alessiob@webrtc.org
+henrik.lundin@webrtc.org
+ivoc@webrtc.org
+peah@webrtc.org
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/README.md b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/README.md
new file mode 100644
index 0000000000..4156112df2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/README.md
@@ -0,0 +1,125 @@
+# APM Quality Assessment tool
+
+Python wrapper of APM simulators (e.g., `audioproc_f`) with which quality
+assessment can be automatized. The tool allows to simulate different noise
+conditions, input signals, APM configurations and it computes different scores.
+Once the scores are computed, the results can be easily exported to an HTML page
+which allows to listen to the APM input and output signals and also the
+reference one used for evaluation.
+
+## Dependencies
+ - OS: Linux
+ - Python 2.7
+ - Python libraries: enum34, numpy, scipy, pydub (0.17.0+), pandas (0.20.1+),
+                     pyquery (1.2+), jsmin (2.2+), csscompressor (0.9.4)
+ - It is recommended that a dedicated Python environment is used
+   - install `virtualenv`
+   - `$ sudo apt-get install python-virtualenv`
+   - setup a new Python environment (e.g., `my_env`)
+   - `$ cd ~ && virtualenv my_env`
+   - activate the new Python environment
+   - `$ source ~/my_env/bin/activate`
+   - add dependcies via `pip`
+   - `(my_env)$ pip install enum34 numpy pydub scipy pandas pyquery jsmin \`
+                `csscompressor`
+ - PolqaOem64 (see http://www.polqa.info/)
+    - Tested with POLQA Library v1.180 / P863 v2.400
+ - Aachen Impulse Response (AIR) Database
+    - Download https://www2.iks.rwth-aachen.de/air/air_database_release_1_4.zip
+ - Input probing signals and noise tracks (you can make your own dataset - *1)
+
+## Build
+ - Compile WebRTC
+ - Go to `out/Default/py_quality_assessment` and check that
+   `apm_quality_assessment.py` exists
+
+## Unit tests
+ - Compile WebRTC
+ - Go to `out/Default/py_quality_assessment`
+ - Run `python -m unittest discover -p "*_unittest.py"`
+
+## First time setup
+ - Deploy PolqaOem64 and set the `POLQA_PATH` environment variable
+   - e.g., `$ export POLQA_PATH=/var/opt/PolqaOem64`
+ - Deploy the AIR Database and set the `AECHEN_IR_DATABASE_PATH` environment
+ variable
+   - e.g., `$ export AECHEN_IR_DATABASE_PATH=/var/opt/AIR_1_4`
+ - Deploy probing signal tracks into
+   - `out/Default/py_quality_assessment/probing_signals` (*1)
+ - Deploy noise tracks into
+   - `out/Default/py_quality_assessment/noise_tracks` (*1, *2)
+
+(*1) You can use custom files as long as they are mono tracks sampled at 48kHz
+encoded in the 16 bit signed format (it is recommended that the tracks are
+converted and exported with Audacity).
+
+## Usage (scores computation)
+ - Go to `out/Default/py_quality_assessment`
+ - Check the `apm_quality_assessment.sh` as an example script to parallelize the
+   experiments
+ - Adjust the script according to your preferences (e.g., output path)
+ - Run `apm_quality_assessment.sh`
+ - The script will end by opening the browser and showing ALL the computed
+   scores
+
+## Usage (export reports)
+Showing all the results at once can be confusing. You therefore may want to
+export separate reports. In this case, you can use the
+`apm_quality_assessment_export.py` script as follows:
+
+ - Set `--output_dir, -o` to the same value used in `apm_quality_assessment.sh`
+ - Use regular expressions to select/filter out scores by
+    - APM configurations: `--config_names, -c`
+    - capture signals: `--capture_names, -i`
+    - render signals: `--render_names, -r`
+    - echo simulator: `--echo_simulator_names, -e`
+    - test data generators: `--test_data_generators, -t`
+    - scores: `--eval_scores, -s`
+ - Assign a suffix to the report name using `-f <suffix>`
+
+For instance:
+
+```
+$ ./apm_quality_assessment_export.py \
+  -o output/ \
+  -c "(^default$)|(.*AE.*)" \
+  -t \(white_noise\) \
+  -s \(polqa\) \
+  -f echo
+```
+
+## Usage (boxplot)
+After generating stats, it can help to visualize how a score depends on a
+certain APM simulator parameter. The `apm_quality_assessment_boxplot.py` script
+helps with that, producing plots similar to [this
+one](https://matplotlib.org/mpl_examples/pylab_examples/boxplot_demo_06.png).
+
+Suppose some scores come from running the APM simulator `audioproc_f` with
+or without the level controller: `--lc=1` or `--lc=0`. Then two boxplots
+side by side can be generated with
+
+```
+$ ./apm_quality_assessment_boxplot.py \
+      -o /path/to/output
+      -v <score_name>
+      -n /path/to/dir/with/apm_configs
+      -z lc
+```
+
+## Troubleshooting
+The input wav file must be:
+  - sampled at a sample rate that is a multiple of 100 (required by POLQA)
+  - in the 16 bit format (required by `audioproc_f`)
+  - encoded in the Microsoft WAV signed 16 bit PCM format (Audacity default
+    when exporting)
+
+Depending on the license, the POLQA tool may take “breaks” as a way to limit the
+throughput. When this happens, the APM Quality Assessment tool is slowed down.
+For more details about this limitation, check Section 10.9.1 in the POLQA manual
+v.1.18.
+
+In case of issues with the POLQA score computation, check
+`py_quality_assessment/eval_scores.py` and adapt
+`PolqaScore._parse_output_file()`.
+The code can be also fixed directly into the build directory (namely,
+`out/Default/py_quality_assessment/eval_scores.py`).
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_configs/default.json b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_configs/default.json
new file mode 100644
index 0000000000..5c3277bac0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_configs/default.json
@@ -0,0 +1 @@
+{"-all_default": null}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.py
new file mode 100755
index 0000000000..e067ecb692
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.py
@@ -0,0 +1,217 @@
+#!/usr/bin/env python
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Perform APM module quality assessment on one or more input files using one or
+   more APM simulator configuration files and one or more test data generators.
+
+Usage: apm_quality_assessment.py -i audio1.wav [audio2.wav ...]
+                                 -c cfg1.json [cfg2.json ...]
+                                 -n white [echo ...]
+                                 -e audio_level [polqa ...]
+                                 -o /path/to/output
+"""
+
+import argparse
+import logging
+import os
+import sys
+
+import quality_assessment.audioproc_wrapper as audioproc_wrapper
+import quality_assessment.echo_path_simulation as echo_path_simulation
+import quality_assessment.eval_scores as eval_scores
+import quality_assessment.evaluation as evaluation
+import quality_assessment.eval_scores_factory as eval_scores_factory
+import quality_assessment.external_vad as external_vad
+import quality_assessment.test_data_generation as test_data_generation
+import quality_assessment.test_data_generation_factory as  \
+    test_data_generation_factory
+import quality_assessment.simulation as simulation
+
+_ECHO_PATH_SIMULATOR_NAMES = (
+    echo_path_simulation.EchoPathSimulator.REGISTERED_CLASSES)
+_TEST_DATA_GENERATOR_CLASSES = (
+    test_data_generation.TestDataGenerator.REGISTERED_CLASSES)
+_TEST_DATA_GENERATORS_NAMES = _TEST_DATA_GENERATOR_CLASSES.keys()
+_EVAL_SCORE_WORKER_CLASSES = eval_scores.EvaluationScore.REGISTERED_CLASSES
+_EVAL_SCORE_WORKER_NAMES = _EVAL_SCORE_WORKER_CLASSES.keys()
+
+_DEFAULT_CONFIG_FILE = 'apm_configs/default.json'
+
+_POLQA_BIN_NAME = 'PolqaOem64'
+
+
+def _InstanceArgumentsParser():
+    """Arguments parser factory.
+  """
+    parser = argparse.ArgumentParser(description=(
+        'Perform APM module quality assessment on one or more input files using '
+        'one or more APM simulator configuration files and one or more '
+        'test data generators.'))
+
+    parser.add_argument('-c',
+                        '--config_files',
+                        nargs='+',
+                        required=False,
+                        help=('path to the configuration files defining the '
+                              'arguments with which the APM simulator tool is '
+                              'called'),
+                        default=[_DEFAULT_CONFIG_FILE])
+
+    parser.add_argument(
+        '-i',
+        '--capture_input_files',
+        nargs='+',
+        required=True,
+        help='path to the capture input wav files (one or more)')
+
+    parser.add_argument('-r',
+                        '--render_input_files',
+                        nargs='+',
+                        required=False,
+                        help=('path to the render input wav files; either '
+                              'omitted or one file for each file in '
+                              '--capture_input_files (files will be paired by '
+                              'index)'),
+                        default=None)
+
+    parser.add_argument('-p',
+                        '--echo_path_simulator',
+                        required=False,
+                        help=('custom echo path simulator name; required if '
+                              '--render_input_files is specified'),
+                        choices=_ECHO_PATH_SIMULATOR_NAMES,
+                        default=echo_path_simulation.NoEchoPathSimulator.NAME)
+
+    parser.add_argument('-t',
+                        '--test_data_generators',
+                        nargs='+',
+                        required=False,
+                        help='custom list of test data generators to use',
+                        choices=_TEST_DATA_GENERATORS_NAMES,
+                        default=_TEST_DATA_GENERATORS_NAMES)
+
+    parser.add_argument('--additive_noise_tracks_path', required=False,
+                        help='path to the wav files for the additive',
+                        default=test_data_generation.  \
+                                AdditiveNoiseTestDataGenerator.  \
+                                DEFAULT_NOISE_TRACKS_PATH)
+
+    parser.add_argument('-e',
+                        '--eval_scores',
+                        nargs='+',
+                        required=False,
+                        help='custom list of evaluation scores to use',
+                        choices=_EVAL_SCORE_WORKER_NAMES,
+                        default=_EVAL_SCORE_WORKER_NAMES)
+
+    parser.add_argument('-o',
+                        '--output_dir',
+                        required=False,
+                        help=('base path to the output directory in which the '
+                              'output wav files and the evaluation outcomes '
+                              'are saved'),
+                        default='output')
+
+    parser.add_argument('--polqa_path',
+                        required=True,
+                        help='path to the POLQA tool')
+
+    parser.add_argument('--air_db_path',
+                        required=True,
+                        help='path to the Aechen IR database')
+
+    parser.add_argument('--apm_sim_path', required=False,
+                        help='path to the APM simulator tool',
+                        default=audioproc_wrapper.  \
+                                AudioProcWrapper.  \
+                                DEFAULT_APM_SIMULATOR_BIN_PATH)
+
+    parser.add_argument('--echo_metric_tool_bin_path',
+                        required=False,
+                        help=('path to the echo metric binary '
+                              '(required for the echo eval score)'),
+                        default=None)
+
+    parser.add_argument(
+        '--copy_with_identity_generator',
+        required=False,
+        help=('If true, the identity test data generator makes a '
+              'copy of the clean speech input file.'),
+        default=False)
+
+    parser.add_argument('--external_vad_paths',
+                        nargs='+',
+                        required=False,
+                        help=('Paths to external VAD programs. Each must take'
+                              '\'-i <wav file> -o <output>\' inputs'),
+                        default=[])
+
+    parser.add_argument('--external_vad_names',
+                        nargs='+',
+                        required=False,
+                        help=('Keys to the vad paths. Must be different and '
+                              'as many as the paths.'),
+                        default=[])
+
+    return parser
+
+
+def _ValidateArguments(args, parser):
+    if args.capture_input_files and args.render_input_files and (len(
+            args.capture_input_files) != len(args.render_input_files)):
+        parser.error(
+            '--render_input_files and --capture_input_files must be lists '
+            'having the same length')
+        sys.exit(1)
+
+    if args.render_input_files and not args.echo_path_simulator:
+        parser.error(
+            'when --render_input_files is set, --echo_path_simulator is '
+            'also required')
+        sys.exit(1)
+
+    if len(args.external_vad_names) != len(args.external_vad_paths):
+        parser.error('If provided, --external_vad_paths and '
+                     '--external_vad_names must '
+                     'have the same number of arguments.')
+        sys.exit(1)
+
+
+def main():
+    # TODO(alessiob): level = logging.INFO once debugged.
+    logging.basicConfig(level=logging.DEBUG)
+    parser = _InstanceArgumentsParser()
+    args = parser.parse_args()
+    _ValidateArguments(args, parser)
+
+    simulator = simulation.ApmModuleSimulator(
+        test_data_generator_factory=(
+            test_data_generation_factory.TestDataGeneratorFactory(
+                aechen_ir_database_path=args.air_db_path,
+                noise_tracks_path=args.additive_noise_tracks_path,
+                copy_with_identity=args.copy_with_identity_generator)),
+        evaluation_score_factory=eval_scores_factory.
+        EvaluationScoreWorkerFactory(
+            polqa_tool_bin_path=os.path.join(args.polqa_path, _POLQA_BIN_NAME),
+            echo_metric_tool_bin_path=args.echo_metric_tool_bin_path),
+        ap_wrapper=audioproc_wrapper.AudioProcWrapper(args.apm_sim_path),
+        evaluator=evaluation.ApmModuleEvaluator(),
+        external_vads=external_vad.ExternalVad.ConstructVadDict(
+            args.external_vad_paths, args.external_vad_names))
+    simulator.Run(config_filepaths=args.config_files,
+                  capture_input_filepaths=args.capture_input_files,
+                  render_input_filepaths=args.render_input_files,
+                  echo_path_simulator_name=args.echo_path_simulator,
+                  test_data_generator_names=args.test_data_generators,
+                  eval_score_names=args.eval_scores,
+                  output_dir=args.output_dir)
+    sys.exit(0)
+
+
+if __name__ == '__main__':
+    main()
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.sh b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.sh
new file mode 100755
index 0000000000..aa563ee26b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment.sh
@@ -0,0 +1,91 @@
+#!/bin/bash
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# Path to the POLQA tool.
+if [ -z ${POLQA_PATH} ]; then  # Check if defined.
+  # Default location.
+  export POLQA_PATH='/var/opt/PolqaOem64'
+fi
+if [ -d "${POLQA_PATH}" ]; then
+  echo "POLQA found in ${POLQA_PATH}"
+else
+  echo "POLQA not found in ${POLQA_PATH}"
+  exit 1
+fi
+
+# Path to the Aechen IR database.
+if [ -z ${AECHEN_IR_DATABASE_PATH} ]; then  # Check if defined.
+  # Default location.
+  export AECHEN_IR_DATABASE_PATH='/var/opt/AIR_1_4'
+fi
+if [ -d "${AECHEN_IR_DATABASE_PATH}" ]; then
+  echo "AIR database found in ${AECHEN_IR_DATABASE_PATH}"
+else
+  echo "AIR database not found in ${AECHEN_IR_DATABASE_PATH}"
+  exit 1
+fi
+
+# Customize probing signals, test data generators and scores if needed.
+CAPTURE_SIGNALS=(probing_signals/*.wav)
+TEST_DATA_GENERATORS=( \
+    "identity" \
+    "white_noise" \
+    # "environmental_noise" \
+    # "reverberation" \
+)
+SCORES=( \
+    # "polqa" \
+    "audio_level_peak" \
+    "audio_level_mean" \
+)
+OUTPUT_PATH=output
+
+# Generate standard APM config files.
+chmod +x apm_quality_assessment_gencfgs.py
+./apm_quality_assessment_gencfgs.py
+
+# Customize APM configurations if needed.
+APM_CONFIGS=(apm_configs/*.json)
+
+# Add output path if missing.
+if [ ! -d ${OUTPUT_PATH} ]; then
+  mkdir ${OUTPUT_PATH}
+fi
+
+# Start one process for each "probing signal"-"test data source" pair.
+chmod +x apm_quality_assessment.py
+for capture_signal_filepath in "${CAPTURE_SIGNALS[@]}" ; do
+  probing_signal_name="$(basename $capture_signal_filepath)"
+  probing_signal_name="${probing_signal_name%.*}"
+  for test_data_gen_name in "${TEST_DATA_GENERATORS[@]}" ; do
+    LOG_FILE="${OUTPUT_PATH}/apm_qa-${probing_signal_name}-"`
+             `"${test_data_gen_name}.log"
+    echo "Starting ${probing_signal_name} ${test_data_gen_name} "`
+         `"(see ${LOG_FILE})"
+    ./apm_quality_assessment.py \
+        --polqa_path ${POLQA_PATH}\
+        --air_db_path ${AECHEN_IR_DATABASE_PATH}\
+        -i ${capture_signal_filepath} \
+        -o ${OUTPUT_PATH} \
+        -t ${test_data_gen_name} \
+        -c "${APM_CONFIGS[@]}" \
+        -e "${SCORES[@]}" > $LOG_FILE 2>&1 &
+  done
+done
+
+# Join Python processes running apm_quality_assessment.py.
+wait
+
+# Export results.
+chmod +x ./apm_quality_assessment_export.py
+./apm_quality_assessment_export.py -o ${OUTPUT_PATH}
+
+# Show results in the browser.
+RESULTS_FILE="$(realpath ${OUTPUT_PATH}/results.html)"
+sensible-browser "file://${RESULTS_FILE}" > /dev/null 2>&1 &
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py
new file mode 100644
index 0000000000..c425885b95
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_boxplot.py
@@ -0,0 +1,154 @@
+#!/usr/bin/env python
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Shows boxplots of given score for different values of selected
+parameters. Can be used to compare scores by audioproc_f flag.
+
+Usage: apm_quality_assessment_boxplot.py -o /path/to/output
+                                         -v polqa
+                                         -n /path/to/dir/with/apm_configs
+                                         -z audioproc_f_arg1 [arg2 ...]
+
+Arguments --config_names, --render_names, --echo_simulator_names,
+--test_data_generators, --eval_scores can be used to filter the data
+used for plotting.
+"""
+
+import collections
+import logging
+import matplotlib.pyplot as plt
+import os
+
+import quality_assessment.data_access as data_access
+import quality_assessment.collect_data as collect_data
+
+
+def InstanceArgumentsParser():
+    """Arguments parser factory.
+  """
+    parser = collect_data.InstanceArgumentsParser()
+    parser.description = (
+        'Shows boxplot of given score for different values of selected'
+        'parameters. Can be used to compare scores by audioproc_f flag')
+
+    parser.add_argument('-v',
+                        '--eval_score',
+                        required=True,
+                        help=('Score name for constructing boxplots'))
+
+    parser.add_argument(
+        '-n',
+        '--config_dir',
+        required=False,
+        help=('path to the folder with the configuration files'),
+        default='apm_configs')
+
+    parser.add_argument('-z',
+                        '--params_to_plot',
+                        required=True,
+                        nargs='+',
+                        help=('audioproc_f parameter values'
+                              'by which to group scores (no leading dash)'))
+
+    return parser
+
+
+def FilterScoresByParams(data_frame, filter_params, score_name, config_dir):
+    """Filters data on the values of one or more parameters.
+
+  Args:
+    data_frame: pandas.DataFrame of all used input data.
+
+    filter_params: each config of the input data is assumed to have
+      exactly one parameter from `filter_params` defined. Every value
+      of the parameters in `filter_params` is a key in the returned
+      dict; the associated value is all cells of the data with that
+      value of the parameter.
+
+    score_name: Name of score which value is boxplotted. Currently cannot do
+      more than one value.
+
+    config_dir: path to dir with APM configs.
+
+  Returns: dictionary, key is a param value, result is all scores for
+    that param value (see `filter_params` for explanation).
+  """
+    results = collections.defaultdict(dict)
+    config_names = data_frame['apm_config'].drop_duplicates().values.tolist()
+
+    for config_name in config_names:
+        config_json = data_access.AudioProcConfigFile.Load(
+            os.path.join(config_dir, config_name + '.json'))
+        data_with_config = data_frame[data_frame.apm_config == config_name]
+        data_cell_scores = data_with_config[data_with_config.eval_score_name ==
+                                            score_name]
+
+        # Exactly one of `params_to_plot` must match:
+        (matching_param, ) = [
+            x for x in filter_params if '-' + x in config_json
+        ]
+
+        # Add scores for every track to the result.
+        for capture_name in data_cell_scores.capture:
+            result_score = float(data_cell_scores[data_cell_scores.capture ==
+                                                  capture_name].score)
+            config_dict = results[config_json['-' + matching_param]]
+            if capture_name not in config_dict:
+                config_dict[capture_name] = {}
+
+            config_dict[capture_name][matching_param] = result_score
+
+    return results
+
+
+def _FlattenToScoresList(config_param_score_dict):
+    """Extracts a list of scores from input data structure.
+
+  Args:
+    config_param_score_dict: of the form {'capture_name':
+    {'param_name' : score_value,.. } ..}
+
+  Returns: Plain list of all score value present in input data
+    structure
+  """
+    result = []
+    for capture_name in config_param_score_dict:
+        result += list(config_param_score_dict[capture_name].values())
+    return result
+
+
+def main():
+    # Init.
+    # TODO(alessiob): INFO once debugged.
+    logging.basicConfig(level=logging.DEBUG)
+    parser = InstanceArgumentsParser()
+    args = parser.parse_args()
+
+    # Get the scores.
+    src_path = collect_data.ConstructSrcPath(args)
+    logging.debug(src_path)
+    scores_data_frame = collect_data.FindScores(src_path, args)
+
+    # Filter the data by `args.params_to_plot`
+    scores_filtered = FilterScoresByParams(scores_data_frame,
+                                           args.params_to_plot,
+                                           args.eval_score, args.config_dir)
+
+    data_list = sorted(scores_filtered.items())
+    data_values = [_FlattenToScoresList(x) for (_, x) in data_list]
+    data_labels = [x for (x, _) in data_list]
+
+    _, axes = plt.subplots(nrows=1, ncols=1, figsize=(6, 6))
+    axes.boxplot(data_values, labels=data_labels)
+    axes.set_ylabel(args.eval_score)
+    axes.set_xlabel('/'.join(args.params_to_plot))
+    plt.show()
+
+
+if __name__ == "__main__":
+    main()
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_export.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_export.py
new file mode 100755
index 0000000000..c20accb9dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_export.py
@@ -0,0 +1,63 @@
+#!/usr/bin/env python
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Export the scores computed by the apm_quality_assessment.py script into an
+   HTML file.
+"""
+
+import logging
+import os
+import sys
+
+import quality_assessment.collect_data as collect_data
+import quality_assessment.export as export
+
+
+def _BuildOutputFilename(filename_suffix):
+    """Builds the filename for the exported file.
+
+  Args:
+    filename_suffix: suffix for the output file name.
+
+  Returns:
+    A string.
+  """
+    if filename_suffix is None:
+        return 'results.html'
+    return 'results-{}.html'.format(filename_suffix)
+
+
+def main():
+    # Init.
+    logging.basicConfig(
+        level=logging.DEBUG)  # TODO(alessio): INFO once debugged.
+    parser = collect_data.InstanceArgumentsParser()
+    parser.add_argument('-f',
+                        '--filename_suffix',
+                        help=('suffix of the exported file'))
+    parser.description = ('Exports pre-computed APM module quality assessment '
+                          'results into HTML tables')
+    args = parser.parse_args()
+
+    # Get the scores.
+    src_path = collect_data.ConstructSrcPath(args)
+    logging.debug(src_path)
+    scores_data_frame = collect_data.FindScores(src_path, args)
+
+    # Export.
+    output_filepath = os.path.join(args.output_dir,
+                                   _BuildOutputFilename(args.filename_suffix))
+    exporter = export.HtmlExport(output_filepath)
+    exporter.Export(scores_data_frame)
+
+    logging.info('output file successfully written in %s', output_filepath)
+    sys.exit(0)
+
+
+if __name__ == '__main__':
+    main()
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py
new file mode 100755
index 0000000000..ca80f85bd1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_gencfgs.py
@@ -0,0 +1,128 @@
+#!/usr/bin/env python
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Generate .json files with which the APM module can be tested using the
+   apm_quality_assessment.py script and audioproc_f as APM simulator.
+"""
+
+import logging
+import os
+
+import quality_assessment.data_access as data_access
+
+OUTPUT_PATH = os.path.abspath('apm_configs')
+
+
+def _GenerateDefaultOverridden(config_override):
+    """Generates one or more APM overriden configurations.
+
+  For each item in config_override, it overrides the default configuration and
+  writes a new APM configuration file.
+
+  The default settings are loaded via "-all_default".
+  Check "src/modules/audio_processing/test/audioproc_float.cc" and search
+  for "if (FLAG_all_default) {".
+
+  For instance, in 55eb6d621489730084927868fed195d3645a9ec9 the default is this:
+  settings.use_aec = rtc::Optional<bool>(true);
+  settings.use_aecm = rtc::Optional<bool>(false);
+  settings.use_agc = rtc::Optional<bool>(true);
+  settings.use_bf = rtc::Optional<bool>(false);
+  settings.use_ed = rtc::Optional<bool>(false);
+  settings.use_hpf = rtc::Optional<bool>(true);
+  settings.use_le = rtc::Optional<bool>(true);
+  settings.use_ns = rtc::Optional<bool>(true);
+  settings.use_ts = rtc::Optional<bool>(true);
+  settings.use_vad = rtc::Optional<bool>(true);
+
+  Args:
+    config_override: dict of APM configuration file names as keys; the values
+                     are dict instances encoding the audioproc_f flags.
+  """
+    for config_filename in config_override:
+        config = config_override[config_filename]
+        config['-all_default'] = None
+
+        config_filepath = os.path.join(
+            OUTPUT_PATH, 'default-{}.json'.format(config_filename))
+        logging.debug('config file <%s> | %s', config_filepath, config)
+
+        data_access.AudioProcConfigFile.Save(config_filepath, config)
+        logging.info('config file created: <%s>', config_filepath)
+
+
+def _GenerateAllDefaultButOne():
+    """Disables the flags enabled by default one-by-one.
+  """
+    config_sets = {
+        'no_AEC': {
+            '-aec': 0,
+        },
+        'no_AGC': {
+            '-agc': 0,
+        },
+        'no_HP_filter': {
+            '-hpf': 0,
+        },
+        'no_level_estimator': {
+            '-le': 0,
+        },
+        'no_noise_suppressor': {
+            '-ns': 0,
+        },
+        'no_transient_suppressor': {
+            '-ts': 0,
+        },
+        'no_vad': {
+            '-vad': 0,
+        },
+    }
+    _GenerateDefaultOverridden(config_sets)
+
+
+def _GenerateAllDefaultPlusOne():
+    """Enables the flags disabled by default one-by-one.
+  """
+    config_sets = {
+        'with_AECM': {
+            '-aec': 0,
+            '-aecm': 1,
+        },  # AEC and AECM are exclusive.
+        'with_AGC_limiter': {
+            '-agc_limiter': 1,
+        },
+        'with_AEC_delay_agnostic': {
+            '-delay_agnostic': 1,
+        },
+        'with_drift_compensation': {
+            '-drift_compensation': 1,
+        },
+        'with_residual_echo_detector': {
+            '-ed': 1,
+        },
+        'with_AEC_extended_filter': {
+            '-extended_filter': 1,
+        },
+        'with_LC': {
+            '-lc': 1,
+        },
+        'with_refined_adaptive_filter': {
+            '-refined_adaptive_filter': 1,
+        },
+    }
+    _GenerateDefaultOverridden(config_sets)
+
+
+def main():
+    logging.basicConfig(level=logging.INFO)
+    _GenerateAllDefaultPlusOne()
+    _GenerateAllDefaultButOne()
+
+
+if __name__ == '__main__':
+    main()
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py
new file mode 100644
index 0000000000..ecae2ed995
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py
@@ -0,0 +1,189 @@
+#!/usr/bin/env python
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Finds the APM configuration that maximizes a provided metric by
+parsing the output generated apm_quality_assessment.py.
+"""
+
+from __future__ import division
+
+import collections
+import logging
+import os
+
+import quality_assessment.data_access as data_access
+import quality_assessment.collect_data as collect_data
+
+
+def _InstanceArgumentsParser():
+    """Arguments parser factory. Extends the arguments from 'collect_data'
+  with a few extra for selecting what parameters to optimize for.
+  """
+    parser = collect_data.InstanceArgumentsParser()
+    parser.description = (
+        'Rudimentary optimization of a function over different parameter'
+        'combinations.')
+
+    parser.add_argument(
+        '-n',
+        '--config_dir',
+        required=False,
+        help=('path to the folder with the configuration files'),
+        default='apm_configs')
+
+    parser.add_argument('-p',
+                        '--params',
+                        required=True,
+                        nargs='+',
+                        help=('parameters to parse from the config files in'
+                              'config_dir'))
+
+    parser.add_argument(
+        '-z',
+        '--params_not_to_optimize',
+        required=False,
+        nargs='+',
+        default=[],
+        help=('parameters from `params` not to be optimized for'))
+
+    return parser
+
+
+def _ConfigurationAndScores(data_frame, params, params_not_to_optimize,
+                            config_dir):
+    """Returns a list of all configurations and scores.
+
+  Args:
+    data_frame: A pandas data frame with the scores and config name
+                returned by _FindScores.
+    params: The parameter names to parse from configs the config
+            directory
+
+    params_not_to_optimize: The parameter names which shouldn't affect
+                            the optimal parameter
+                            selection. E.g., fixed settings and not
+                            tunable parameters.
+
+    config_dir: Path to folder with config files.
+
+  Returns:
+    Dictionary of the form
+    {param_combination: [{params: {param1: value1, ...},
+                          scores: {score1: value1, ...}}]}.
+
+    The key `param_combination` runs over all parameter combinations
+    of the parameters in `params` and not in
+    `params_not_to_optimize`. A corresponding value is a list of all
+    param combinations for params in `params_not_to_optimize` and
+    their scores.
+  """
+    results = collections.defaultdict(list)
+    config_names = data_frame['apm_config'].drop_duplicates().values.tolist()
+    score_names = data_frame['eval_score_name'].drop_duplicates(
+    ).values.tolist()
+
+    # Normalize the scores
+    normalization_constants = {}
+    for score_name in score_names:
+        scores = data_frame[data_frame.eval_score_name == score_name].score
+        normalization_constants[score_name] = max(scores)
+
+    params_to_optimize = [p for p in params if p not in params_not_to_optimize]
+    param_combination = collections.namedtuple("ParamCombination",
+                                               params_to_optimize)
+
+    for config_name in config_names:
+        config_json = data_access.AudioProcConfigFile.Load(
+            os.path.join(config_dir, config_name + ".json"))
+        scores = {}
+        data_cell = data_frame[data_frame.apm_config == config_name]
+        for score_name in score_names:
+            data_cell_scores = data_cell[data_cell.eval_score_name ==
+                                         score_name].score
+            scores[score_name] = sum(data_cell_scores) / len(data_cell_scores)
+            scores[score_name] /= normalization_constants[score_name]
+
+        result = {'scores': scores, 'params': {}}
+        config_optimize_params = {}
+        for param in params:
+            if param in params_to_optimize:
+                config_optimize_params[param] = config_json['-' + param]
+            else:
+                result['params'][param] = config_json['-' + param]
+
+        current_param_combination = param_combination(**config_optimize_params)
+        results[current_param_combination].append(result)
+    return results
+
+
+def _FindOptimalParameter(configs_and_scores, score_weighting):
+    """Finds the config producing the maximal score.
+
+  Args:
+    configs_and_scores: structure of the form returned by
+                        _ConfigurationAndScores
+
+    score_weighting: a function to weight together all score values of
+                     the form [{params: {param1: value1, ...}, scores:
+                                {score1: value1, ...}}] into a numeric
+                     value
+  Returns:
+    the config that has the largest values of `score_weighting` applied
+    to its scores.
+  """
+
+    min_score = float('+inf')
+    best_params = None
+    for config in configs_and_scores:
+        scores_and_params = configs_and_scores[config]
+        current_score = score_weighting(scores_and_params)
+        if current_score < min_score:
+            min_score = current_score
+            best_params = config
+            logging.debug("Score: %f", current_score)
+            logging.debug("Config: %s", str(config))
+    return best_params
+
+
+def _ExampleWeighting(scores_and_configs):
+    """Example argument to `_FindOptimalParameter`
+  Args:
+    scores_and_configs: a list of configs and scores, in the form
+                        described in _FindOptimalParameter
+  Returns:
+    numeric value, the sum of all scores
+  """
+    res = 0
+    for score_config in scores_and_configs:
+        res += sum(score_config['scores'].values())
+    return res
+
+
+def main():
+    # Init.
+    # TODO(alessiob): INFO once debugged.
+    logging.basicConfig(level=logging.DEBUG)
+    parser = _InstanceArgumentsParser()
+    args = parser.parse_args()
+
+    # Get the scores.
+    src_path = collect_data.ConstructSrcPath(args)
+    logging.debug('Src path <%s>', src_path)
+    scores_data_frame = collect_data.FindScores(src_path, args)
+    all_scores = _ConfigurationAndScores(scores_data_frame, args.params,
+                                         args.params_not_to_optimize,
+                                         args.config_dir)
+
+    opt_param = _FindOptimalParameter(all_scores, _ExampleWeighting)
+
+    logging.info('Optimal parameter combination: <%s>', opt_param)
+    logging.info('It\'s score values: <%s>', all_scores[opt_param])
+
+
+if __name__ == "__main__":
+    main()
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_unittest.py
new file mode 100644
index 0000000000..80338c1373
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_unittest.py
@@ -0,0 +1,28 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Unit tests for the apm_quality_assessment module.
+"""
+
+import sys
+import unittest
+
+import mock
+
+import apm_quality_assessment
+
+
+class TestSimulationScript(unittest.TestCase):
+    """Unit tests for the apm_quality_assessment module.
+  """
+
+    def testMain(self):
+        # Exit with error code if no arguments are passed.
+        with self.assertRaises(SystemExit) as cm, mock.patch.object(
+                sys, 'argv', ['apm_quality_assessment.py']):
+            apm_quality_assessment.main()
+        self.assertGreater(cm.exception.code, 0)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/output/README.md b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/output/README.md
new file mode 100644
index 0000000000..66e2a1c848
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/output/README.md
@@ -0,0 +1 @@
+You can use this folder for the output generated by the apm_quality_assessment scripts.
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/__init__.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/__init__.py
new file mode 100644
index 0000000000..b870dfaef3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/__init__.py
@@ -0,0 +1,7 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py
new file mode 100644
index 0000000000..93a8248397
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py
@@ -0,0 +1,296 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Extraction of annotations from audio files.
+"""
+
+from __future__ import division
+import logging
+import os
+import shutil
+import struct
+import subprocess
+import sys
+import tempfile
+
+try:
+    import numpy as np
+except ImportError:
+    logging.critical('Cannot import the third-party Python package numpy')
+    sys.exit(1)
+
+from . import external_vad
+from . import exceptions
+from . import signal_processing
+
+
+class AudioAnnotationsExtractor(object):
+    """Extracts annotations from audio files.
+  """
+
+    class VadType(object):
+        ENERGY_THRESHOLD = 1  # TODO(alessiob): Consider switching to P56 standard.
+        WEBRTC_COMMON_AUDIO = 2  # common_audio/vad/include/vad.h
+        WEBRTC_APM = 4  # modules/audio_processing/vad/vad.h
+
+        def __init__(self, value):
+            if (not isinstance(value, int)) or not 0 <= value <= 7:
+                raise exceptions.InitializationException('Invalid vad type: ' +
+                                                         value)
+            self._value = value
+
+        def Contains(self, vad_type):
+            return self._value | vad_type == self._value
+
+        def __str__(self):
+            vads = []
+            if self.Contains(self.ENERGY_THRESHOLD):
+                vads.append("energy")
+            if self.Contains(self.WEBRTC_COMMON_AUDIO):
+                vads.append("common_audio")
+            if self.Contains(self.WEBRTC_APM):
+                vads.append("apm")
+            return "VadType({})".format(", ".join(vads))
+
+    _OUTPUT_FILENAME_TEMPLATE = '{}annotations.npz'
+
+    # Level estimation params.
+    _ONE_DB_REDUCTION = np.power(10.0, -1.0 / 20.0)
+    _LEVEL_FRAME_SIZE_MS = 1.0
+    # The time constants in ms indicate the time it takes for the level estimate
+    # to go down/up by 1 db if the signal is zero.
+    _LEVEL_ATTACK_MS = 5.0
+    _LEVEL_DECAY_MS = 20.0
+
+    # VAD params.
+    _VAD_THRESHOLD = 1
+    _VAD_WEBRTC_PATH = os.path.join(os.path.dirname(os.path.abspath(__file__)),
+                                    os.pardir, os.pardir)
+    _VAD_WEBRTC_COMMON_AUDIO_PATH = os.path.join(_VAD_WEBRTC_PATH, 'vad')
+
+    _VAD_WEBRTC_APM_PATH = os.path.join(_VAD_WEBRTC_PATH, 'apm_vad')
+
+    def __init__(self, vad_type, external_vads=None):
+        self._signal = None
+        self._level = None
+        self._level_frame_size = None
+        self._common_audio_vad = None
+        self._energy_vad = None
+        self._apm_vad_probs = None
+        self._apm_vad_rms = None
+        self._vad_frame_size = None
+        self._vad_frame_size_ms = None
+        self._c_attack = None
+        self._c_decay = None
+
+        self._vad_type = self.VadType(vad_type)
+        logging.info('VADs used for annotations: ' + str(self._vad_type))
+
+        if external_vads is None:
+            external_vads = {}
+        self._external_vads = external_vads
+
+        assert len(self._external_vads) == len(external_vads), (
+            'The external VAD names must be unique.')
+        for vad in external_vads.values():
+            if not isinstance(vad, external_vad.ExternalVad):
+                raise exceptions.InitializationException('Invalid vad type: ' +
+                                                         str(type(vad)))
+            logging.info('External VAD used for annotation: ' + str(vad.name))
+
+        assert os.path.exists(self._VAD_WEBRTC_COMMON_AUDIO_PATH), \
+          self._VAD_WEBRTC_COMMON_AUDIO_PATH
+        assert os.path.exists(self._VAD_WEBRTC_APM_PATH), \
+          self._VAD_WEBRTC_APM_PATH
+
+    @classmethod
+    def GetOutputFileNameTemplate(cls):
+        return cls._OUTPUT_FILENAME_TEMPLATE
+
+    def GetLevel(self):
+        return self._level
+
+    def GetLevelFrameSize(self):
+        return self._level_frame_size
+
+    @classmethod
+    def GetLevelFrameSizeMs(cls):
+        return cls._LEVEL_FRAME_SIZE_MS
+
+    def GetVadOutput(self, vad_type):
+        if vad_type == self.VadType.ENERGY_THRESHOLD:
+            return self._energy_vad
+        elif vad_type == self.VadType.WEBRTC_COMMON_AUDIO:
+            return self._common_audio_vad
+        elif vad_type == self.VadType.WEBRTC_APM:
+            return (self._apm_vad_probs, self._apm_vad_rms)
+        else:
+            raise exceptions.InitializationException('Invalid vad type: ' +
+                                                     vad_type)
+
+    def GetVadFrameSize(self):
+        return self._vad_frame_size
+
+    def GetVadFrameSizeMs(self):
+        return self._vad_frame_size_ms
+
+    def Extract(self, filepath):
+        # Load signal.
+        self._signal = signal_processing.SignalProcessingUtils.LoadWav(
+            filepath)
+        if self._signal.channels != 1:
+            raise NotImplementedError(
+                'Multiple-channel annotations not implemented')
+
+        # Level estimation params.
+        self._level_frame_size = int(self._signal.frame_rate / 1000 *
+                                     (self._LEVEL_FRAME_SIZE_MS))
+        self._c_attack = 0.0 if self._LEVEL_ATTACK_MS == 0 else (
+            self._ONE_DB_REDUCTION**(self._LEVEL_FRAME_SIZE_MS /
+                                     self._LEVEL_ATTACK_MS))
+        self._c_decay = 0.0 if self._LEVEL_DECAY_MS == 0 else (
+            self._ONE_DB_REDUCTION**(self._LEVEL_FRAME_SIZE_MS /
+                                     self._LEVEL_DECAY_MS))
+
+        # Compute level.
+        self._LevelEstimation()
+
+        # Ideal VAD output, it requires clean speech with high SNR as input.
+        if self._vad_type.Contains(self.VadType.ENERGY_THRESHOLD):
+            # Naive VAD based on level thresholding.
+            vad_threshold = np.percentile(self._level, self._VAD_THRESHOLD)
+            self._energy_vad = np.uint8(self._level > vad_threshold)
+            self._vad_frame_size = self._level_frame_size
+            self._vad_frame_size_ms = self._LEVEL_FRAME_SIZE_MS
+        if self._vad_type.Contains(self.VadType.WEBRTC_COMMON_AUDIO):
+            # WebRTC common_audio/ VAD.
+            self._RunWebRtcCommonAudioVad(filepath, self._signal.frame_rate)
+        if self._vad_type.Contains(self.VadType.WEBRTC_APM):
+            # WebRTC modules/audio_processing/ VAD.
+            self._RunWebRtcApmVad(filepath)
+        for extvad_name in self._external_vads:
+            self._external_vads[extvad_name].Run(filepath)
+
+    def Save(self, output_path, annotation_name=""):
+        ext_kwargs = {
+            'extvad_conf-' + ext_vad:
+            self._external_vads[ext_vad].GetVadOutput()
+            for ext_vad in self._external_vads
+        }
+        np.savez_compressed(file=os.path.join(
+            output_path,
+            self.GetOutputFileNameTemplate().format(annotation_name)),
+                            level=self._level,
+                            level_frame_size=self._level_frame_size,
+                            level_frame_size_ms=self._LEVEL_FRAME_SIZE_MS,
+                            vad_output=self._common_audio_vad,
+                            vad_energy_output=self._energy_vad,
+                            vad_frame_size=self._vad_frame_size,
+                            vad_frame_size_ms=self._vad_frame_size_ms,
+                            vad_probs=self._apm_vad_probs,
+                            vad_rms=self._apm_vad_rms,
+                            **ext_kwargs)
+
+    def _LevelEstimation(self):
+        # Read samples.
+        samples = signal_processing.SignalProcessingUtils.AudioSegmentToRawData(
+            self._signal).astype(np.float32) / 32768.0
+        num_frames = len(samples) // self._level_frame_size
+        num_samples = num_frames * self._level_frame_size
+
+        # Envelope.
+        self._level = np.max(np.reshape(np.abs(samples[:num_samples]),
+                                        (num_frames, self._level_frame_size)),
+                             axis=1)
+        assert len(self._level) == num_frames
+
+        # Envelope smoothing.
+        smooth = lambda curr, prev, k: (1 - k) * curr + k * prev
+        self._level[0] = smooth(self._level[0], 0.0, self._c_attack)
+        for i in range(1, num_frames):
+            self._level[i] = smooth(
+                self._level[i], self._level[i - 1], self._c_attack if
+                (self._level[i] > self._level[i - 1]) else self._c_decay)
+
+    def _RunWebRtcCommonAudioVad(self, wav_file_path, sample_rate):
+        self._common_audio_vad = None
+        self._vad_frame_size = None
+
+        # Create temporary output path.
+        tmp_path = tempfile.mkdtemp()
+        output_file_path = os.path.join(
+            tmp_path,
+            os.path.split(wav_file_path)[1] + '_vad.tmp')
+
+        # Call WebRTC VAD.
+        try:
+            subprocess.call([
+                self._VAD_WEBRTC_COMMON_AUDIO_PATH, '-i', wav_file_path, '-o',
+                output_file_path
+            ],
+                            cwd=self._VAD_WEBRTC_PATH)
+
+            # Read bytes.
+            with open(output_file_path, 'rb') as f:
+                raw_data = f.read()
+
+            # Parse side information.
+            self._vad_frame_size_ms = struct.unpack('B', raw_data[0])[0]
+            self._vad_frame_size = self._vad_frame_size_ms * sample_rate / 1000
+            assert self._vad_frame_size_ms in [10, 20, 30]
+            extra_bits = struct.unpack('B', raw_data[-1])[0]
+            assert 0 <= extra_bits <= 8
+
+            # Init VAD vector.
+            num_bytes = len(raw_data)
+            num_frames = 8 * (num_bytes -
+                              2) - extra_bits  # 8 frames for each byte.
+            self._common_audio_vad = np.zeros(num_frames, np.uint8)
+
+            # Read VAD decisions.
+            for i, byte in enumerate(raw_data[1:-1]):
+                byte = struct.unpack('B', byte)[0]
+                for j in range(8 if i < num_bytes - 3 else (8 - extra_bits)):
+                    self._common_audio_vad[i * 8 + j] = int(byte & 1)
+                    byte = byte >> 1
+        except Exception as e:
+            logging.error('Error while running the WebRTC VAD (' + e.message +
+                          ')')
+        finally:
+            if os.path.exists(tmp_path):
+                shutil.rmtree(tmp_path)
+
+    def _RunWebRtcApmVad(self, wav_file_path):
+        # Create temporary output path.
+        tmp_path = tempfile.mkdtemp()
+        output_file_path_probs = os.path.join(
+            tmp_path,
+            os.path.split(wav_file_path)[1] + '_vad_probs.tmp')
+        output_file_path_rms = os.path.join(
+            tmp_path,
+            os.path.split(wav_file_path)[1] + '_vad_rms.tmp')
+
+        # Call WebRTC VAD.
+        try:
+            subprocess.call([
+                self._VAD_WEBRTC_APM_PATH, '-i', wav_file_path, '-o_probs',
+                output_file_path_probs, '-o_rms', output_file_path_rms
+            ],
+                            cwd=self._VAD_WEBRTC_PATH)
+
+            # Parse annotations.
+            self._apm_vad_probs = np.fromfile(output_file_path_probs,
+                                              np.double)
+            self._apm_vad_rms = np.fromfile(output_file_path_rms, np.double)
+            assert len(self._apm_vad_rms) == len(self._apm_vad_probs)
+
+        except Exception as e:
+            logging.error('Error while running the WebRTC APM VAD (' +
+                          e.message + ')')
+        finally:
+            if os.path.exists(tmp_path):
+                shutil.rmtree(tmp_path)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations_unittest.py
new file mode 100644
index 0000000000..8230208808
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations_unittest.py
@@ -0,0 +1,160 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Unit tests for the annotations module.
+"""
+
+from __future__ import division
+import logging
+import os
+import shutil
+import tempfile
+import unittest
+
+import numpy as np
+
+from . import annotations
+from . import external_vad
+from . import input_signal_creator
+from . import signal_processing
+
+
+class TestAnnotationsExtraction(unittest.TestCase):
+    """Unit tests for the annotations module.
+  """
+
+    _CLEAN_TMP_OUTPUT = True
+    _DEBUG_PLOT_VAD = False
+    _VAD_TYPE_CLASS = annotations.AudioAnnotationsExtractor.VadType
+    _ALL_VAD_TYPES = (_VAD_TYPE_CLASS.ENERGY_THRESHOLD
+                      | _VAD_TYPE_CLASS.WEBRTC_COMMON_AUDIO
+                      | _VAD_TYPE_CLASS.WEBRTC_APM)
+
+    def setUp(self):
+        """Create temporary folder."""
+        self._tmp_path = tempfile.mkdtemp()
+        self._wav_file_path = os.path.join(self._tmp_path, 'tone.wav')
+        pure_tone, _ = input_signal_creator.InputSignalCreator.Create(
+            'pure_tone', [440, 1000])
+        signal_processing.SignalProcessingUtils.SaveWav(
+            self._wav_file_path, pure_tone)
+        self._sample_rate = pure_tone.frame_rate
+
+    def tearDown(self):
+        """Recursively delete temporary folder."""
+        if self._CLEAN_TMP_OUTPUT:
+            shutil.rmtree(self._tmp_path)
+        else:
+            logging.warning(self.id() + ' did not clean the temporary path ' +
+                            (self._tmp_path))
+
+    def testFrameSizes(self):
+        e = annotations.AudioAnnotationsExtractor(self._ALL_VAD_TYPES)
+        e.Extract(self._wav_file_path)
+        samples_to_ms = lambda n, sr: 1000 * n // sr
+        self.assertEqual(
+            samples_to_ms(e.GetLevelFrameSize(), self._sample_rate),
+            e.GetLevelFrameSizeMs())
+        self.assertEqual(samples_to_ms(e.GetVadFrameSize(), self._sample_rate),
+                         e.GetVadFrameSizeMs())
+
+    def testVoiceActivityDetectors(self):
+        for vad_type_value in range(0, self._ALL_VAD_TYPES + 1):
+            vad_type = self._VAD_TYPE_CLASS(vad_type_value)
+            e = annotations.AudioAnnotationsExtractor(vad_type=vad_type_value)
+            e.Extract(self._wav_file_path)
+            if vad_type.Contains(self._VAD_TYPE_CLASS.ENERGY_THRESHOLD):
+                # pylint: disable=unpacking-non-sequence
+                vad_output = e.GetVadOutput(
+                    self._VAD_TYPE_CLASS.ENERGY_THRESHOLD)
+                self.assertGreater(len(vad_output), 0)
+                self.assertGreaterEqual(
+                    float(np.sum(vad_output)) / len(vad_output), 0.95)
+
+            if vad_type.Contains(self._VAD_TYPE_CLASS.WEBRTC_COMMON_AUDIO):
+                # pylint: disable=unpacking-non-sequence
+                vad_output = e.GetVadOutput(
+                    self._VAD_TYPE_CLASS.WEBRTC_COMMON_AUDIO)
+                self.assertGreater(len(vad_output), 0)
+                self.assertGreaterEqual(
+                    float(np.sum(vad_output)) / len(vad_output), 0.95)
+
+            if vad_type.Contains(self._VAD_TYPE_CLASS.WEBRTC_APM):
+                # pylint: disable=unpacking-non-sequence
+                (vad_probs,
+                 vad_rms) = e.GetVadOutput(self._VAD_TYPE_CLASS.WEBRTC_APM)
+                self.assertGreater(len(vad_probs), 0)
+                self.assertGreater(len(vad_rms), 0)
+                self.assertGreaterEqual(
+                    float(np.sum(vad_probs)) / len(vad_probs), 0.5)
+                self.assertGreaterEqual(
+                    float(np.sum(vad_rms)) / len(vad_rms), 20000)
+
+            if self._DEBUG_PLOT_VAD:
+                frame_times_s = lambda num_frames, frame_size_ms: np.arange(
+                    num_frames).astype(np.float32) * frame_size_ms / 1000.0
+                level = e.GetLevel()
+                t_level = frame_times_s(num_frames=len(level),
+                                        frame_size_ms=e.GetLevelFrameSizeMs())
+                t_vad = frame_times_s(num_frames=len(vad_output),
+                                      frame_size_ms=e.GetVadFrameSizeMs())
+                import matplotlib.pyplot as plt
+                plt.figure()
+                plt.hold(True)
+                plt.plot(t_level, level)
+                plt.plot(t_vad, vad_output * np.max(level), '.')
+                plt.show()
+
+    def testSaveLoad(self):
+        e = annotations.AudioAnnotationsExtractor(self._ALL_VAD_TYPES)
+        e.Extract(self._wav_file_path)
+        e.Save(self._tmp_path, "fake-annotation")
+
+        data = np.load(
+            os.path.join(
+                self._tmp_path,
+                e.GetOutputFileNameTemplate().format("fake-annotation")))
+        np.testing.assert_array_equal(e.GetLevel(), data['level'])
+        self.assertEqual(np.float32, data['level'].dtype)
+        np.testing.assert_array_equal(
+            e.GetVadOutput(self._VAD_TYPE_CLASS.ENERGY_THRESHOLD),
+            data['vad_energy_output'])
+        np.testing.assert_array_equal(
+            e.GetVadOutput(self._VAD_TYPE_CLASS.WEBRTC_COMMON_AUDIO),
+            data['vad_output'])
+        np.testing.assert_array_equal(
+            e.GetVadOutput(self._VAD_TYPE_CLASS.WEBRTC_APM)[0],
+            data['vad_probs'])
+        np.testing.assert_array_equal(
+            e.GetVadOutput(self._VAD_TYPE_CLASS.WEBRTC_APM)[1],
+            data['vad_rms'])
+        self.assertEqual(np.uint8, data['vad_energy_output'].dtype)
+        self.assertEqual(np.float64, data['vad_probs'].dtype)
+        self.assertEqual(np.float64, data['vad_rms'].dtype)
+
+    def testEmptyExternalShouldNotCrash(self):
+        for vad_type_value in range(0, self._ALL_VAD_TYPES + 1):
+            annotations.AudioAnnotationsExtractor(vad_type_value, {})
+
+    def testFakeExternalSaveLoad(self):
+        def FakeExternalFactory():
+            return external_vad.ExternalVad(
+                os.path.join(os.path.dirname(os.path.abspath(__file__)),
+                             'fake_external_vad.py'), 'fake')
+
+        for vad_type_value in range(0, self._ALL_VAD_TYPES + 1):
+            e = annotations.AudioAnnotationsExtractor(
+                vad_type_value, {'fake': FakeExternalFactory()})
+            e.Extract(self._wav_file_path)
+            e.Save(self._tmp_path, annotation_name="fake-annotation")
+            data = np.load(
+                os.path.join(
+                    self._tmp_path,
+                    e.GetOutputFileNameTemplate().format("fake-annotation")))
+            self.assertEqual(np.float32, data['extvad_conf-fake'].dtype)
+            np.testing.assert_almost_equal(np.arange(100, dtype=np.float32),
+                                           data['extvad_conf-fake'])
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_configs/default.json b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_configs/default.json
new file mode 100644
index 0000000000..5c3277bac0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_configs/default.json
@@ -0,0 +1 @@
+{"-all_default": null}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc
new file mode 100644
index 0000000000..73ce4ed3f7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc
@@ -0,0 +1,96 @@
+// Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+//
+// Use of this source code is governed by a BSD-style license
+// that can be found in the LICENSE file in the root of the source
+// tree. An additional intellectual property rights grant can be found
+// in the file PATENTS.  All contributing project authors may
+// be found in the AUTHORS file in the root of the source tree.
+
+#include <array>
+#include <fstream>
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/vad/voice_activity_detector.h"
+#include "rtc_base/logging.h"
+
+ABSL_FLAG(std::string, i, "", "Input wav file");
+ABSL_FLAG(std::string, o_probs, "", "VAD probabilities output file");
+ABSL_FLAG(std::string, o_rms, "", "VAD output file");
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr uint8_t kAudioFrameLengthMilliseconds = 10;
+constexpr int kMaxSampleRate = 48000;
+constexpr size_t kMaxFrameLen =
+    kAudioFrameLengthMilliseconds * kMaxSampleRate / 1000;
+
+int main(int argc, char* argv[]) {
+  absl::ParseCommandLine(argc, argv);
+  const std::string input_file = absl::GetFlag(FLAGS_i);
+  const std::string output_probs_file = absl::GetFlag(FLAGS_o_probs);
+  const std::string output_file = absl::GetFlag(FLAGS_o_rms);
+  // Open wav input file and check properties.
+  WavReader wav_reader(input_file);
+  if (wav_reader.num_channels() != 1) {
+    RTC_LOG(LS_ERROR) << "Only mono wav files supported";
+    return 1;
+  }
+  if (wav_reader.sample_rate() > kMaxSampleRate) {
+    RTC_LOG(LS_ERROR) << "Beyond maximum sample rate (" << kMaxSampleRate
+                      << ")";
+    return 1;
+  }
+  const size_t audio_frame_len = rtc::CheckedDivExact(
+      kAudioFrameLengthMilliseconds * wav_reader.sample_rate(), 1000);
+  if (audio_frame_len > kMaxFrameLen) {
+    RTC_LOG(LS_ERROR) << "The frame size and/or the sample rate are too large.";
+    return 1;
+  }
+
+  // Create output file and write header.
+  std::ofstream out_probs_file(output_probs_file, std::ofstream::binary);
+  std::ofstream out_rms_file(output_file, std::ofstream::binary);
+
+  // Run VAD and write decisions.
+  VoiceActivityDetector vad;
+  std::array<int16_t, kMaxFrameLen> samples;
+
+  while (true) {
+    // Process frame.
+    const auto read_samples =
+        wav_reader.ReadSamples(audio_frame_len, samples.data());
+    if (read_samples < audio_frame_len) {
+      break;
+    }
+    vad.ProcessChunk(samples.data(), audio_frame_len, wav_reader.sample_rate());
+    // Write output.
+    auto probs = vad.chunkwise_voice_probabilities();
+    auto rms = vad.chunkwise_rms();
+    RTC_CHECK_EQ(probs.size(), rms.size());
+    RTC_CHECK_EQ(sizeof(double), 8);
+
+    for (const auto& p : probs) {
+      out_probs_file.write(reinterpret_cast<const char*>(&p), 8);
+    }
+    for (const auto& r : rms) {
+      out_rms_file.write(reinterpret_cast<const char*>(&r), 8);
+    }
+  }
+
+  out_probs_file.close();
+  out_rms_file.close();
+  return 0;
+}
+
+}  // namespace
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::main(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/audioproc_wrapper.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/audioproc_wrapper.py
new file mode 100644
index 0000000000..04aeaa95b9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/audioproc_wrapper.py
@@ -0,0 +1,100 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Class implementing a wrapper for APM simulators.
+"""
+
+import cProfile
+import logging
+import os
+import subprocess
+
+from . import data_access
+from . import exceptions
+
+
+class AudioProcWrapper(object):
+    """Wrapper for APM simulators.
+  """
+
+    DEFAULT_APM_SIMULATOR_BIN_PATH = os.path.abspath(
+        os.path.join(os.pardir, 'audioproc_f'))
+    OUTPUT_FILENAME = 'output.wav'
+
+    def __init__(self, simulator_bin_path):
+        """Ctor.
+
+    Args:
+      simulator_bin_path: path to the APM simulator binary.
+    """
+        self._simulator_bin_path = simulator_bin_path
+        self._config = None
+        self._output_signal_filepath = None
+
+        # Profiler instance to measure running time.
+        self._profiler = cProfile.Profile()
+
+    @property
+    def output_filepath(self):
+        return self._output_signal_filepath
+
+    def Run(self,
+            config_filepath,
+            capture_input_filepath,
+            output_path,
+            render_input_filepath=None):
+        """Runs APM simulator.
+
+    Args:
+      config_filepath: path to the configuration file specifying the arguments
+                       for the APM simulator.
+      capture_input_filepath: path to the capture audio track input file (aka
+                              forward or near-end).
+      output_path: path of the audio track output file.
+      render_input_filepath: path to the render audio track input file (aka
+                             reverse or far-end).
+    """
+        # Init.
+        self._output_signal_filepath = os.path.join(output_path,
+                                                    self.OUTPUT_FILENAME)
+        profiling_stats_filepath = os.path.join(output_path, 'profiling.stats')
+
+        # Skip if the output has already been generated.
+        if os.path.exists(self._output_signal_filepath) and os.path.exists(
+                profiling_stats_filepath):
+            return
+
+        # Load configuration.
+        self._config = data_access.AudioProcConfigFile.Load(config_filepath)
+
+        # Set remaining parameters.
+        if not os.path.exists(capture_input_filepath):
+            raise exceptions.FileNotFoundError(
+                'cannot find capture input file')
+        self._config['-i'] = capture_input_filepath
+        self._config['-o'] = self._output_signal_filepath
+        if render_input_filepath is not None:
+            if not os.path.exists(render_input_filepath):
+                raise exceptions.FileNotFoundError(
+                    'cannot find render input file')
+            self._config['-ri'] = render_input_filepath
+
+        # Build arguments list.
+        args = [self._simulator_bin_path]
+        for param_name in self._config:
+            args.append(param_name)
+            if self._config[param_name] is not None:
+                args.append(str(self._config[param_name]))
+        logging.debug(' '.join(args))
+
+        # Run.
+        self._profiler.enable()
+        subprocess.call(args)
+        self._profiler.disable()
+
+        # Save profiling stats.
+        self._profiler.dump_stats(profiling_stats_filepath)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/collect_data.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/collect_data.py
new file mode 100644
index 0000000000..38aac0cbe2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/collect_data.py
@@ -0,0 +1,243 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Imports a filtered subset of the scores and configurations computed
+by apm_quality_assessment.py into a pandas data frame.
+"""
+
+import argparse
+import glob
+import logging
+import os
+import re
+import sys
+
+try:
+    import pandas as pd
+except ImportError:
+    logging.critical('Cannot import the third-party Python package pandas')
+    sys.exit(1)
+
+from . import data_access as data_access
+from . import simulation as sim
+
+# Compiled regular expressions used to extract score descriptors.
+RE_CONFIG_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixApmConfig() +
+                            r'(.+)')
+RE_CAPTURE_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixCapture() +
+                             r'(.+)')
+RE_RENDER_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixRender() + r'(.+)')
+RE_ECHO_SIM_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixEchoSimulator() +
+                              r'(.+)')
+RE_TEST_DATA_GEN_NAME = re.compile(
+    sim.ApmModuleSimulator.GetPrefixTestDataGenerator() + r'(.+)')
+RE_TEST_DATA_GEN_PARAMS = re.compile(
+    sim.ApmModuleSimulator.GetPrefixTestDataGeneratorParameters() + r'(.+)')
+RE_SCORE_NAME = re.compile(sim.ApmModuleSimulator.GetPrefixScore() +
+                           r'(.+)(\..+)')
+
+
+def InstanceArgumentsParser():
+    """Arguments parser factory.
+  """
+    parser = argparse.ArgumentParser(
+        description=('Override this description in a user script by changing'
+                     ' `parser.description` of the returned parser.'))
+
+    parser.add_argument('-o',
+                        '--output_dir',
+                        required=True,
+                        help=('the same base path used with the '
+                              'apm_quality_assessment tool'))
+
+    parser.add_argument(
+        '-c',
+        '--config_names',
+        type=re.compile,
+        help=('regular expression to filter the APM configuration'
+              ' names'))
+
+    parser.add_argument(
+        '-i',
+        '--capture_names',
+        type=re.compile,
+        help=('regular expression to filter the capture signal '
+              'names'))
+
+    parser.add_argument('-r',
+                        '--render_names',
+                        type=re.compile,
+                        help=('regular expression to filter the render signal '
+                              'names'))
+
+    parser.add_argument(
+        '-e',
+        '--echo_simulator_names',
+        type=re.compile,
+        help=('regular expression to filter the echo simulator '
+              'names'))
+
+    parser.add_argument('-t',
+                        '--test_data_generators',
+                        type=re.compile,
+                        help=('regular expression to filter the test data '
+                              'generator names'))
+
+    parser.add_argument(
+        '-s',
+        '--eval_scores',
+        type=re.compile,
+        help=('regular expression to filter the evaluation score '
+              'names'))
+
+    return parser
+
+
+def _GetScoreDescriptors(score_filepath):
+    """Extracts a score descriptor from the given score file path.
+
+  Args:
+    score_filepath: path to the score file.
+
+  Returns:
+    A tuple of strings (APM configuration name, capture audio track name,
+    render audio track name, echo simulator name, test data generator name,
+    test data generator parameters as string, evaluation score name).
+  """
+    fields = score_filepath.split(os.sep)[-7:]
+    extract_name = lambda index, reg_expr: (reg_expr.match(fields[index]).
+                                            groups(0)[0])
+    return (
+        extract_name(0, RE_CONFIG_NAME),
+        extract_name(1, RE_CAPTURE_NAME),
+        extract_name(2, RE_RENDER_NAME),
+        extract_name(3, RE_ECHO_SIM_NAME),
+        extract_name(4, RE_TEST_DATA_GEN_NAME),
+        extract_name(5, RE_TEST_DATA_GEN_PARAMS),
+        extract_name(6, RE_SCORE_NAME),
+    )
+
+
+def _ExcludeScore(config_name, capture_name, render_name, echo_simulator_name,
+                  test_data_gen_name, score_name, args):
+    """Decides whether excluding a score.
+
+  A set of optional regular expressions in args is used to determine if the
+  score should be excluded (depending on its |*_name| descriptors).
+
+  Args:
+    config_name: APM configuration name.
+    capture_name: capture audio track name.
+    render_name: render audio track name.
+    echo_simulator_name: echo simulator name.
+    test_data_gen_name: test data generator name.
+    score_name: evaluation score name.
+    args: parsed arguments.
+
+  Returns:
+    A boolean.
+  """
+    value_regexpr_pairs = [
+        (config_name, args.config_names),
+        (capture_name, args.capture_names),
+        (render_name, args.render_names),
+        (echo_simulator_name, args.echo_simulator_names),
+        (test_data_gen_name, args.test_data_generators),
+        (score_name, args.eval_scores),
+    ]
+
+    # Score accepted if each value matches the corresponding regular expression.
+    for value, regexpr in value_regexpr_pairs:
+        if regexpr is None:
+            continue
+        if not regexpr.match(value):
+            return True
+
+    return False
+
+
+def FindScores(src_path, args):
+    """Given a search path, find scores and return a DataFrame object.
+
+  Args:
+    src_path: Search path pattern.
+    args: parsed arguments.
+
+  Returns:
+    A DataFrame object.
+  """
+    # Get scores.
+    scores = []
+    for score_filepath in glob.iglob(src_path):
+        # Extract score descriptor fields from the path.
+        (config_name, capture_name, render_name, echo_simulator_name,
+         test_data_gen_name, test_data_gen_params,
+         score_name) = _GetScoreDescriptors(score_filepath)
+
+        # Ignore the score if required.
+        if _ExcludeScore(config_name, capture_name, render_name,
+                         echo_simulator_name, test_data_gen_name, score_name,
+                         args):
+            logging.info('ignored score: %s %s %s %s %s %s', config_name,
+                         capture_name, render_name, echo_simulator_name,
+                         test_data_gen_name, score_name)
+            continue
+
+        # Read metadata and score.
+        metadata = data_access.Metadata.LoadAudioTestDataPaths(
+            os.path.split(score_filepath)[0])
+        score = data_access.ScoreFile.Load(score_filepath)
+
+        # Add a score with its descriptor fields.
+        scores.append((
+            metadata['clean_capture_input_filepath'],
+            metadata['echo_free_capture_filepath'],
+            metadata['echo_filepath'],
+            metadata['render_filepath'],
+            metadata['capture_filepath'],
+            metadata['apm_output_filepath'],
+            metadata['apm_reference_filepath'],
+            config_name,
+            capture_name,
+            render_name,
+            echo_simulator_name,
+            test_data_gen_name,
+            test_data_gen_params,
+            score_name,
+            score,
+        ))
+
+    return pd.DataFrame(data=scores,
+                        columns=(
+                            'clean_capture_input_filepath',
+                            'echo_free_capture_filepath',
+                            'echo_filepath',
+                            'render_filepath',
+                            'capture_filepath',
+                            'apm_output_filepath',
+                            'apm_reference_filepath',
+                            'apm_config',
+                            'capture',
+                            'render',
+                            'echo_simulator',
+                            'test_data_gen',
+                            'test_data_gen_params',
+                            'eval_score_name',
+                            'score',
+                        ))
+
+
+def ConstructSrcPath(args):
+    return os.path.join(
+        args.output_dir,
+        sim.ApmModuleSimulator.GetPrefixApmConfig() + '*',
+        sim.ApmModuleSimulator.GetPrefixCapture() + '*',
+        sim.ApmModuleSimulator.GetPrefixRender() + '*',
+        sim.ApmModuleSimulator.GetPrefixEchoSimulator() + '*',
+        sim.ApmModuleSimulator.GetPrefixTestDataGenerator() + '*',
+        sim.ApmModuleSimulator.GetPrefixTestDataGeneratorParameters() + '*',
+        sim.ApmModuleSimulator.GetPrefixScore() + '*')
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/data_access.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/data_access.py
new file mode 100644
index 0000000000..c1aebb67f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/data_access.py
@@ -0,0 +1,154 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Data access utility functions and classes.
+"""
+
+import json
+import os
+
+
+def MakeDirectory(path):
+    """Makes a directory recursively without rising exceptions if existing.
+
+  Args:
+    path: path to the directory to be created.
+  """
+    if os.path.exists(path):
+        return
+    os.makedirs(path)
+
+
+class Metadata(object):
+    """Data access class to save and load metadata.
+  """
+
+    def __init__(self):
+        pass
+
+    _GENERIC_METADATA_SUFFIX = '.mdata'
+    _AUDIO_TEST_DATA_FILENAME = 'audio_test_data.json'
+
+    @classmethod
+    def LoadFileMetadata(cls, filepath):
+        """Loads generic metadata linked to a file.
+
+    Args:
+      filepath: path to the metadata file to read.
+
+    Returns:
+      A dict.
+    """
+        with open(filepath + cls._GENERIC_METADATA_SUFFIX) as f:
+            return json.load(f)
+
+    @classmethod
+    def SaveFileMetadata(cls, filepath, metadata):
+        """Saves generic metadata linked to a file.
+
+    Args:
+      filepath: path to the metadata file to write.
+      metadata: a dict.
+    """
+        with open(filepath + cls._GENERIC_METADATA_SUFFIX, 'w') as f:
+            json.dump(metadata, f)
+
+    @classmethod
+    def LoadAudioTestDataPaths(cls, metadata_path):
+        """Loads the input and the reference audio track paths.
+
+    Args:
+      metadata_path: path to the directory containing the metadata file.
+
+    Returns:
+      Tuple with the paths to the input and output audio tracks.
+    """
+        metadata_filepath = os.path.join(metadata_path,
+                                         cls._AUDIO_TEST_DATA_FILENAME)
+        with open(metadata_filepath) as f:
+            return json.load(f)
+
+    @classmethod
+    def SaveAudioTestDataPaths(cls, output_path, **filepaths):
+        """Saves the input and the reference audio track paths.
+
+    Args:
+      output_path: path to the directory containing the metadata file.
+
+    Keyword Args:
+      filepaths: collection of audio track file paths to save.
+    """
+        output_filepath = os.path.join(output_path,
+                                       cls._AUDIO_TEST_DATA_FILENAME)
+        with open(output_filepath, 'w') as f:
+            json.dump(filepaths, f)
+
+
+class AudioProcConfigFile(object):
+    """Data access to load/save APM simulator argument lists.
+
+  The arguments stored in the config files are used to control the APM flags.
+  """
+
+    def __init__(self):
+        pass
+
+    @classmethod
+    def Load(cls, filepath):
+        """Loads a configuration file for an APM simulator.
+
+    Args:
+      filepath: path to the configuration file.
+
+    Returns:
+      A dict containing the configuration.
+    """
+        with open(filepath) as f:
+            return json.load(f)
+
+    @classmethod
+    def Save(cls, filepath, config):
+        """Saves a configuration file for an APM simulator.
+
+    Args:
+      filepath: path to the configuration file.
+      config: a dict containing the configuration.
+    """
+        with open(filepath, 'w') as f:
+            json.dump(config, f)
+
+
+class ScoreFile(object):
+    """Data access class to save and load float scalar scores.
+  """
+
+    def __init__(self):
+        pass
+
+    @classmethod
+    def Load(cls, filepath):
+        """Loads a score from file.
+
+    Args:
+      filepath: path to the score file.
+
+    Returns:
+      A float encoding the score.
+    """
+        with open(filepath) as f:
+            return float(f.readline().strip())
+
+    @classmethod
+    def Save(cls, filepath, score):
+        """Saves a score into a file.
+
+    Args:
+      filepath: path to the score file.
+      score: float encoding the score.
+    """
+        with open(filepath, 'w') as f:
+            f.write('{0:f}\n'.format(score))
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation.py
new file mode 100644
index 0000000000..65903ea32d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation.py
@@ -0,0 +1,136 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Echo path simulation module.
+"""
+
+import hashlib
+import os
+
+from . import signal_processing
+
+
+class EchoPathSimulator(object):
+    """Abstract class for the echo path simulators.
+
+  In general, an echo path simulator is a function of the render signal and
+  simulates the propagation of the latter into the microphone (e.g., due to
+  mechanical or electrical paths).
+  """
+
+    NAME = None
+    REGISTERED_CLASSES = {}
+
+    def __init__(self):
+        pass
+
+    def Simulate(self, output_path):
+        """Creates the echo signal and stores it in an audio file (abstract method).
+
+    Args:
+      output_path: Path in which any output can be saved.
+
+    Returns:
+      Path to the generated audio track file or None if no echo is present.
+    """
+        raise NotImplementedError()
+
+    @classmethod
+    def RegisterClass(cls, class_to_register):
+        """Registers an EchoPathSimulator implementation.
+
+    Decorator to automatically register the classes that extend
+    EchoPathSimulator.
+    Example usage:
+
+    @EchoPathSimulator.RegisterClass
+    class NoEchoPathSimulator(EchoPathSimulator):
+      pass
+    """
+        cls.REGISTERED_CLASSES[class_to_register.NAME] = class_to_register
+        return class_to_register
+
+
+@EchoPathSimulator.RegisterClass
+class NoEchoPathSimulator(EchoPathSimulator):
+    """Simulates absence of echo."""
+
+    NAME = 'noecho'
+
+    def __init__(self):
+        EchoPathSimulator.__init__(self)
+
+    def Simulate(self, output_path):
+        return None
+
+
+@EchoPathSimulator.RegisterClass
+class LinearEchoPathSimulator(EchoPathSimulator):
+    """Simulates linear echo path.
+
+  This class applies a given impulse response to the render input and then it
+  sums the signal to the capture input signal.
+  """
+
+    NAME = 'linear'
+
+    def __init__(self, render_input_filepath, impulse_response):
+        """
+    Args:
+      render_input_filepath: Render audio track file.
+      impulse_response: list or numpy vector of float values.
+    """
+        EchoPathSimulator.__init__(self)
+        self._render_input_filepath = render_input_filepath
+        self._impulse_response = impulse_response
+
+    def Simulate(self, output_path):
+        """Simulates linear echo path."""
+        # Form the file name with a hash of the impulse response.
+        impulse_response_hash = hashlib.sha256(
+            str(self._impulse_response).encode('utf-8', 'ignore')).hexdigest()
+        echo_filepath = os.path.join(
+            output_path, 'linear_echo_{}.wav'.format(impulse_response_hash))
+
+        # If the simulated echo audio track file does not exists, create it.
+        if not os.path.exists(echo_filepath):
+            render = signal_processing.SignalProcessingUtils.LoadWav(
+                self._render_input_filepath)
+            echo = signal_processing.SignalProcessingUtils.ApplyImpulseResponse(
+                render, self._impulse_response)
+            signal_processing.SignalProcessingUtils.SaveWav(
+                echo_filepath, echo)
+
+        return echo_filepath
+
+
+@EchoPathSimulator.RegisterClass
+class RecordedEchoPathSimulator(EchoPathSimulator):
+    """Uses recorded echo.
+
+  This class uses the clean capture input file name to build the file name of
+  the corresponding recording containing echo (a predefined suffix is used).
+  Such a file is expected to be already existing.
+  """
+
+    NAME = 'recorded'
+
+    _FILE_NAME_SUFFIX = '_echo'
+
+    def __init__(self, render_input_filepath):
+        EchoPathSimulator.__init__(self)
+        self._render_input_filepath = render_input_filepath
+
+    def Simulate(self, output_path):
+        """Uses recorded echo path."""
+        path, file_name_ext = os.path.split(self._render_input_filepath)
+        file_name, file_ext = os.path.splitext(file_name_ext)
+        echo_filepath = os.path.join(
+            path, '{}{}{}'.format(file_name, self._FILE_NAME_SUFFIX, file_ext))
+        assert os.path.exists(echo_filepath), (
+            'cannot find the echo audio track file {}'.format(echo_filepath))
+        return echo_filepath
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_factory.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_factory.py
new file mode 100644
index 0000000000..4b46b36b47
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_factory.py
@@ -0,0 +1,48 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Echo path simulation factory module.
+"""
+
+import numpy as np
+
+from . import echo_path_simulation
+
+
+class EchoPathSimulatorFactory(object):
+
+    # TODO(alessiob): Replace 20 ms delay (at 48 kHz sample rate) with a more
+    # realistic impulse response.
+    _LINEAR_ECHO_IMPULSE_RESPONSE = np.array([0.0] * (20 * 48) + [0.15])
+
+    def __init__(self):
+        pass
+
+    @classmethod
+    def GetInstance(cls, echo_path_simulator_class, render_input_filepath):
+        """Creates an EchoPathSimulator instance given a class object.
+
+    Args:
+      echo_path_simulator_class: EchoPathSimulator class object (not an
+                                 instance).
+      render_input_filepath: Path to the render audio track file.
+
+    Returns:
+      An EchoPathSimulator instance.
+    """
+        assert render_input_filepath is not None or (
+            echo_path_simulator_class ==
+            echo_path_simulation.NoEchoPathSimulator)
+
+        if echo_path_simulator_class == echo_path_simulation.NoEchoPathSimulator:
+            return echo_path_simulation.NoEchoPathSimulator()
+        elif echo_path_simulator_class == (
+                echo_path_simulation.LinearEchoPathSimulator):
+            return echo_path_simulation.LinearEchoPathSimulator(
+                render_input_filepath, cls._LINEAR_ECHO_IMPULSE_RESPONSE)
+        else:
+            return echo_path_simulator_class(render_input_filepath)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_unittest.py
new file mode 100644
index 0000000000..b6cc8abdde
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/echo_path_simulation_unittest.py
@@ -0,0 +1,82 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Unit tests for the echo path simulation module.
+"""
+
+import shutil
+import os
+import tempfile
+import unittest
+
+import pydub
+
+from . import echo_path_simulation
+from . import echo_path_simulation_factory
+from . import signal_processing
+
+
+class TestEchoPathSimulators(unittest.TestCase):
+    """Unit tests for the eval_scores module.
+  """
+
+    def setUp(self):
+        """Creates temporary data."""
+        self._tmp_path = tempfile.mkdtemp()
+
+        # Create and save white noise.
+        silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000)
+        white_noise = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            silence)
+        self._audio_track_num_samples = (
+            signal_processing.SignalProcessingUtils.CountSamples(white_noise))
+        self._audio_track_filepath = os.path.join(self._tmp_path,
+                                                  'white_noise.wav')
+        signal_processing.SignalProcessingUtils.SaveWav(
+            self._audio_track_filepath, white_noise)
+
+        # Make a copy the white noise audio track file; it will be used by
+        # echo_path_simulation.RecordedEchoPathSimulator.
+        shutil.copy(self._audio_track_filepath,
+                    os.path.join(self._tmp_path, 'white_noise_echo.wav'))
+
+    def tearDown(self):
+        """Recursively deletes temporary folders."""
+        shutil.rmtree(self._tmp_path)
+
+    def testRegisteredClasses(self):
+        # Check that there is at least one registered echo path simulator.
+        registered_classes = (
+            echo_path_simulation.EchoPathSimulator.REGISTERED_CLASSES)
+        self.assertIsInstance(registered_classes, dict)
+        self.assertGreater(len(registered_classes), 0)
+
+        # Instance factory.
+        factory = echo_path_simulation_factory.EchoPathSimulatorFactory()
+
+        # Try each registered echo path simulator.
+        for echo_path_simulator_name in registered_classes:
+            simulator = factory.GetInstance(
+                echo_path_simulator_class=registered_classes[
+                    echo_path_simulator_name],
+                render_input_filepath=self._audio_track_filepath)
+
+            echo_filepath = simulator.Simulate(self._tmp_path)
+            if echo_filepath is None:
+                self.assertEqual(echo_path_simulation.NoEchoPathSimulator.NAME,
+                                 echo_path_simulator_name)
+                # No other tests in this case.
+                continue
+
+            # Check that the echo audio track file exists and its length is greater or
+            # equal to that of the render audio track.
+            self.assertTrue(os.path.exists(echo_filepath))
+            echo = signal_processing.SignalProcessingUtils.LoadWav(
+                echo_filepath)
+            self.assertGreaterEqual(
+                signal_processing.SignalProcessingUtils.CountSamples(echo),
+                self._audio_track_num_samples)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py
new file mode 100644
index 0000000000..59c5f74be4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores.py
@@ -0,0 +1,427 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Evaluation score abstract class and implementations.
+"""
+
+from __future__ import division
+import logging
+import os
+import re
+import subprocess
+import sys
+
+try:
+    import numpy as np
+except ImportError:
+    logging.critical('Cannot import the third-party Python package numpy')
+    sys.exit(1)
+
+from . import data_access
+from . import exceptions
+from . import signal_processing
+
+
+class EvaluationScore(object):
+
+    NAME = None
+    REGISTERED_CLASSES = {}
+
+    def __init__(self, score_filename_prefix):
+        self._score_filename_prefix = score_filename_prefix
+        self._input_signal_metadata = None
+        self._reference_signal = None
+        self._reference_signal_filepath = None
+        self._tested_signal = None
+        self._tested_signal_filepath = None
+        self._output_filepath = None
+        self._score = None
+        self._render_signal_filepath = None
+
+    @classmethod
+    def RegisterClass(cls, class_to_register):
+        """Registers an EvaluationScore implementation.
+
+    Decorator to automatically register the classes that extend EvaluationScore.
+    Example usage:
+
+    @EvaluationScore.RegisterClass
+    class AudioLevelScore(EvaluationScore):
+      pass
+    """
+        cls.REGISTERED_CLASSES[class_to_register.NAME] = class_to_register
+        return class_to_register
+
+    @property
+    def output_filepath(self):
+        return self._output_filepath
+
+    @property
+    def score(self):
+        return self._score
+
+    def SetInputSignalMetadata(self, metadata):
+        """Sets input signal metadata.
+
+    Args:
+      metadata: dict instance.
+    """
+        self._input_signal_metadata = metadata
+
+    def SetReferenceSignalFilepath(self, filepath):
+        """Sets the path to the audio track used as reference signal.
+
+    Args:
+      filepath: path to the reference audio track.
+    """
+        self._reference_signal_filepath = filepath
+
+    def SetTestedSignalFilepath(self, filepath):
+        """Sets the path to the audio track used as test signal.
+
+    Args:
+      filepath: path to the test audio track.
+    """
+        self._tested_signal_filepath = filepath
+
+    def SetRenderSignalFilepath(self, filepath):
+        """Sets the path to the audio track used as render signal.
+
+    Args:
+      filepath: path to the test audio track.
+    """
+        self._render_signal_filepath = filepath
+
+    def Run(self, output_path):
+        """Extracts the score for the set test data pair.
+
+    Args:
+      output_path: path to the directory where the output is written.
+    """
+        self._output_filepath = os.path.join(
+            output_path, self._score_filename_prefix + self.NAME + '.txt')
+        try:
+            # If the score has already been computed, load.
+            self._LoadScore()
+            logging.debug('score found and loaded')
+        except IOError:
+            # Compute the score.
+            logging.debug('score not found, compute')
+            self._Run(output_path)
+
+    def _Run(self, output_path):
+        # Abstract method.
+        raise NotImplementedError()
+
+    def _LoadReferenceSignal(self):
+        assert self._reference_signal_filepath is not None
+        self._reference_signal = signal_processing.SignalProcessingUtils.LoadWav(
+            self._reference_signal_filepath)
+
+    def _LoadTestedSignal(self):
+        assert self._tested_signal_filepath is not None
+        self._tested_signal = signal_processing.SignalProcessingUtils.LoadWav(
+            self._tested_signal_filepath)
+
+    def _LoadScore(self):
+        return data_access.ScoreFile.Load(self._output_filepath)
+
+    def _SaveScore(self):
+        return data_access.ScoreFile.Save(self._output_filepath, self._score)
+
+
+@EvaluationScore.RegisterClass
+class AudioLevelPeakScore(EvaluationScore):
+    """Peak audio level score.
+
+  Defined as the difference between the peak audio level of the tested and
+  the reference signals.
+
+  Unit: dB
+  Ideal: 0 dB
+  Worst case: +/-inf dB
+  """
+
+    NAME = 'audio_level_peak'
+
+    def __init__(self, score_filename_prefix):
+        EvaluationScore.__init__(self, score_filename_prefix)
+
+    def _Run(self, output_path):
+        self._LoadReferenceSignal()
+        self._LoadTestedSignal()
+        self._score = self._tested_signal.dBFS - self._reference_signal.dBFS
+        self._SaveScore()
+
+
+@EvaluationScore.RegisterClass
+class MeanAudioLevelScore(EvaluationScore):
+    """Mean audio level score.
+
+  Defined as the difference between the mean audio level of the tested and
+  the reference signals.
+
+  Unit: dB
+  Ideal: 0 dB
+  Worst case: +/-inf dB
+  """
+
+    NAME = 'audio_level_mean'
+
+    def __init__(self, score_filename_prefix):
+        EvaluationScore.__init__(self, score_filename_prefix)
+
+    def _Run(self, output_path):
+        self._LoadReferenceSignal()
+        self._LoadTestedSignal()
+
+        dbfs_diffs_sum = 0.0
+        seconds = min(len(self._tested_signal), len(
+            self._reference_signal)) // 1000
+        for t in range(seconds):
+            t0 = t * seconds
+            t1 = t0 + seconds
+            dbfs_diffs_sum += (self._tested_signal[t0:t1].dBFS -
+                               self._reference_signal[t0:t1].dBFS)
+        self._score = dbfs_diffs_sum / float(seconds)
+        self._SaveScore()
+
+
+@EvaluationScore.RegisterClass
+class EchoMetric(EvaluationScore):
+    """Echo score.
+
+  Proportion of detected echo.
+
+  Unit: ratio
+  Ideal: 0
+  Worst case: 1
+  """
+
+    NAME = 'echo_metric'
+
+    def __init__(self, score_filename_prefix, echo_detector_bin_filepath):
+        EvaluationScore.__init__(self, score_filename_prefix)
+
+        # POLQA binary file path.
+        self._echo_detector_bin_filepath = echo_detector_bin_filepath
+        if not os.path.exists(self._echo_detector_bin_filepath):
+            logging.error('cannot find EchoMetric tool binary file')
+            raise exceptions.FileNotFoundError()
+
+        self._echo_detector_bin_path, _ = os.path.split(
+            self._echo_detector_bin_filepath)
+
+    def _Run(self, output_path):
+        echo_detector_out_filepath = os.path.join(output_path,
+                                                  'echo_detector.out')
+        if os.path.exists(echo_detector_out_filepath):
+            os.unlink(echo_detector_out_filepath)
+
+        logging.debug("Render signal filepath: %s",
+                      self._render_signal_filepath)
+        if not os.path.exists(self._render_signal_filepath):
+            logging.error(
+                "Render input required for evaluating the echo metric.")
+
+        args = [
+            self._echo_detector_bin_filepath, '--output_file',
+            echo_detector_out_filepath, '--', '-i',
+            self._tested_signal_filepath, '-ri', self._render_signal_filepath
+        ]
+        logging.debug(' '.join(args))
+        subprocess.call(args, cwd=self._echo_detector_bin_path)
+
+        # Parse Echo detector tool output and extract the score.
+        self._score = self._ParseOutputFile(echo_detector_out_filepath)
+        self._SaveScore()
+
+    @classmethod
+    def _ParseOutputFile(cls, echo_metric_file_path):
+        """
+    Parses the POLQA tool output formatted as a table ('-t' option).
+
+    Args:
+      polqa_out_filepath: path to the POLQA tool output file.
+
+    Returns:
+      The score as a number in [0, 1].
+    """
+        with open(echo_metric_file_path) as f:
+            return float(f.read())
+
+
+@EvaluationScore.RegisterClass
+class PolqaScore(EvaluationScore):
+    """POLQA score.
+
+  See http://www.polqa.info/.
+
+  Unit: MOS
+  Ideal: 4.5
+  Worst case: 1.0
+  """
+
+    NAME = 'polqa'
+
+    def __init__(self, score_filename_prefix, polqa_bin_filepath):
+        EvaluationScore.__init__(self, score_filename_prefix)
+
+        # POLQA binary file path.
+        self._polqa_bin_filepath = polqa_bin_filepath
+        if not os.path.exists(self._polqa_bin_filepath):
+            logging.error('cannot find POLQA tool binary file')
+            raise exceptions.FileNotFoundError()
+
+        # Path to the POLQA directory with binary and license files.
+        self._polqa_tool_path, _ = os.path.split(self._polqa_bin_filepath)
+
+    def _Run(self, output_path):
+        polqa_out_filepath = os.path.join(output_path, 'polqa.out')
+        if os.path.exists(polqa_out_filepath):
+            os.unlink(polqa_out_filepath)
+
+        args = [
+            self._polqa_bin_filepath,
+            '-t',
+            '-q',
+            '-Overwrite',
+            '-Ref',
+            self._reference_signal_filepath,
+            '-Test',
+            self._tested_signal_filepath,
+            '-LC',
+            'NB',
+            '-Out',
+            polqa_out_filepath,
+        ]
+        logging.debug(' '.join(args))
+        subprocess.call(args, cwd=self._polqa_tool_path)
+
+        # Parse POLQA tool output and extract the score.
+        polqa_output = self._ParseOutputFile(polqa_out_filepath)
+        self._score = float(polqa_output['PolqaScore'])
+
+        self._SaveScore()
+
+    @classmethod
+    def _ParseOutputFile(cls, polqa_out_filepath):
+        """
+    Parses the POLQA tool output formatted as a table ('-t' option).
+
+    Args:
+      polqa_out_filepath: path to the POLQA tool output file.
+
+    Returns:
+      A dict.
+    """
+        data = []
+        with open(polqa_out_filepath) as f:
+            for line in f:
+                line = line.strip()
+                if len(line) == 0 or line.startswith('*'):
+                    # Ignore comments.
+                    continue
+                # Read fields.
+                data.append(re.split(r'\t+', line))
+
+        # Two rows expected (header and values).
+        assert len(data) == 2, 'Cannot parse POLQA output'
+        number_of_fields = len(data[0])
+        assert number_of_fields == len(data[1])
+
+        # Build and return a dictionary with field names (header) as keys and the
+        # corresponding field values as values.
+        return {
+            data[0][index]: data[1][index]
+            for index in range(number_of_fields)
+        }
+
+
+@EvaluationScore.RegisterClass
+class TotalHarmonicDistorsionScore(EvaluationScore):
+    """Total harmonic distorsion plus noise score.
+
+  Total harmonic distorsion plus noise score.
+  See "https://en.wikipedia.org/wiki/Total_harmonic_distortion#THD.2BN".
+
+  Unit: -.
+  Ideal: 0.
+  Worst case: +inf
+  """
+
+    NAME = 'thd'
+
+    def __init__(self, score_filename_prefix):
+        EvaluationScore.__init__(self, score_filename_prefix)
+        self._input_frequency = None
+
+    def _Run(self, output_path):
+        self._CheckInputSignal()
+
+        self._LoadTestedSignal()
+        if self._tested_signal.channels != 1:
+            raise exceptions.EvaluationScoreException(
+                'unsupported number of channels')
+        samples = signal_processing.SignalProcessingUtils.AudioSegmentToRawData(
+            self._tested_signal)
+
+        # Init.
+        num_samples = len(samples)
+        duration = len(self._tested_signal) / 1000.0
+        scaling = 2.0 / num_samples
+        max_freq = self._tested_signal.frame_rate / 2
+        f0_freq = float(self._input_frequency)
+        t = np.linspace(0, duration, num_samples)
+
+        # Analyze harmonics.
+        b_terms = []
+        n = 1
+        while f0_freq * n < max_freq:
+            x_n = np.sum(
+                samples * np.sin(2.0 * np.pi * n * f0_freq * t)) * scaling
+            y_n = np.sum(
+                samples * np.cos(2.0 * np.pi * n * f0_freq * t)) * scaling
+            b_terms.append(np.sqrt(x_n**2 + y_n**2))
+            n += 1
+
+        output_without_fundamental = samples - b_terms[0] * np.sin(
+            2.0 * np.pi * f0_freq * t)
+        distortion_and_noise = np.sqrt(
+            np.sum(output_without_fundamental**2) * np.pi * scaling)
+
+        # TODO(alessiob): Fix or remove if not needed.
+        # thd = np.sqrt(np.sum(b_terms[1:]**2)) / b_terms[0]
+
+        # TODO(alessiob): Check the range of `thd_plus_noise` and update the class
+        # docstring above if accordingly.
+        thd_plus_noise = distortion_and_noise / b_terms[0]
+
+        self._score = thd_plus_noise
+        self._SaveScore()
+
+    def _CheckInputSignal(self):
+        # Check input signal and get properties.
+        try:
+            if self._input_signal_metadata['signal'] != 'pure_tone':
+                raise exceptions.EvaluationScoreException(
+                    'The THD score requires a pure tone as input signal')
+            self._input_frequency = self._input_signal_metadata['frequency']
+            if self._input_signal_metadata[
+                    'test_data_gen_name'] != 'identity' or (
+                        self._input_signal_metadata['test_data_gen_config'] !=
+                        'default'):
+                raise exceptions.EvaluationScoreException(
+                    'The THD score cannot be used with any test data generator other '
+                    'than "identity"')
+        except TypeError:
+            raise exceptions.EvaluationScoreException(
+                'The THD score requires an input signal with associated metadata'
+            )
+        except KeyError:
+            raise exceptions.EvaluationScoreException(
+                'Invalid input signal metadata to compute the THD score')
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_factory.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_factory.py
new file mode 100644
index 0000000000..5749a8924b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_factory.py
@@ -0,0 +1,55 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""EvaluationScore factory class.
+"""
+
+import logging
+
+from . import exceptions
+from . import eval_scores
+
+
+class EvaluationScoreWorkerFactory(object):
+    """Factory class used to instantiate evaluation score workers.
+
+  The ctor gets the parametrs that are used to instatiate the evaluation score
+  workers.
+  """
+
+    def __init__(self, polqa_tool_bin_path, echo_metric_tool_bin_path):
+        self._score_filename_prefix = None
+        self._polqa_tool_bin_path = polqa_tool_bin_path
+        self._echo_metric_tool_bin_path = echo_metric_tool_bin_path
+
+    def SetScoreFilenamePrefix(self, prefix):
+        self._score_filename_prefix = prefix
+
+    def GetInstance(self, evaluation_score_class):
+        """Creates an EvaluationScore instance given a class object.
+
+    Args:
+      evaluation_score_class: EvaluationScore class object (not an instance).
+
+    Returns:
+      An EvaluationScore instance.
+    """
+        if self._score_filename_prefix is None:
+            raise exceptions.InitializationException(
+                'The score file name prefix for evaluation score workers is not set'
+            )
+        logging.debug('factory producing a %s evaluation score',
+                      evaluation_score_class)
+
+        if evaluation_score_class == eval_scores.PolqaScore:
+            return eval_scores.PolqaScore(self._score_filename_prefix,
+                                          self._polqa_tool_bin_path)
+        elif evaluation_score_class == eval_scores.EchoMetric:
+            return eval_scores.EchoMetric(self._score_filename_prefix,
+                                          self._echo_metric_tool_bin_path)
+        else:
+            return evaluation_score_class(self._score_filename_prefix)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_unittest.py
new file mode 100644
index 0000000000..12e043320e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/eval_scores_unittest.py
@@ -0,0 +1,137 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Unit tests for the eval_scores module.
+"""
+
+import os
+import shutil
+import tempfile
+import unittest
+
+import pydub
+
+from . import data_access
+from . import eval_scores
+from . import eval_scores_factory
+from . import signal_processing
+
+
+class TestEvalScores(unittest.TestCase):
+    """Unit tests for the eval_scores module.
+  """
+
+    def setUp(self):
+        """Create temporary output folder and two audio track files."""
+        self._output_path = tempfile.mkdtemp()
+
+        # Create fake reference and tested (i.e., APM output) audio track files.
+        silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000)
+        fake_reference_signal = (signal_processing.SignalProcessingUtils.
+                                 GenerateWhiteNoise(silence))
+        fake_tested_signal = (signal_processing.SignalProcessingUtils.
+                              GenerateWhiteNoise(silence))
+
+        # Save fake audio tracks.
+        self._fake_reference_signal_filepath = os.path.join(
+            self._output_path, 'fake_ref.wav')
+        signal_processing.SignalProcessingUtils.SaveWav(
+            self._fake_reference_signal_filepath, fake_reference_signal)
+        self._fake_tested_signal_filepath = os.path.join(
+            self._output_path, 'fake_test.wav')
+        signal_processing.SignalProcessingUtils.SaveWav(
+            self._fake_tested_signal_filepath, fake_tested_signal)
+
+    def tearDown(self):
+        """Recursively delete temporary folder."""
+        shutil.rmtree(self._output_path)
+
+    def testRegisteredClasses(self):
+        # Evaluation score names to exclude (tested separately).
+        exceptions = ['thd', 'echo_metric']
+
+        # Preliminary check.
+        self.assertTrue(os.path.exists(self._output_path))
+
+        # Check that there is at least one registered evaluation score worker.
+        registered_classes = eval_scores.EvaluationScore.REGISTERED_CLASSES
+        self.assertIsInstance(registered_classes, dict)
+        self.assertGreater(len(registered_classes), 0)
+
+        # Instance evaluation score workers factory with fake dependencies.
+        eval_score_workers_factory = (
+            eval_scores_factory.EvaluationScoreWorkerFactory(
+                polqa_tool_bin_path=os.path.join(
+                    os.path.dirname(os.path.abspath(__file__)), 'fake_polqa'),
+                echo_metric_tool_bin_path=None))
+        eval_score_workers_factory.SetScoreFilenamePrefix('scores-')
+
+        # Try each registered evaluation score worker.
+        for eval_score_name in registered_classes:
+            if eval_score_name in exceptions:
+                continue
+
+            # Instance evaluation score worker.
+            eval_score_worker = eval_score_workers_factory.GetInstance(
+                registered_classes[eval_score_name])
+
+            # Set fake input metadata and reference and test file paths, then run.
+            eval_score_worker.SetReferenceSignalFilepath(
+                self._fake_reference_signal_filepath)
+            eval_score_worker.SetTestedSignalFilepath(
+                self._fake_tested_signal_filepath)
+            eval_score_worker.Run(self._output_path)
+
+            # Check output.
+            score = data_access.ScoreFile.Load(
+                eval_score_worker.output_filepath)
+            self.assertTrue(isinstance(score, float))
+
+    def testTotalHarmonicDistorsionScore(self):
+        # Init.
+        pure_tone_freq = 5000.0
+        eval_score_worker = eval_scores.TotalHarmonicDistorsionScore('scores-')
+        eval_score_worker.SetInputSignalMetadata({
+            'signal':
+            'pure_tone',
+            'frequency':
+            pure_tone_freq,
+            'test_data_gen_name':
+            'identity',
+            'test_data_gen_config':
+            'default',
+        })
+        template = pydub.AudioSegment.silent(duration=1000, frame_rate=48000)
+
+        # Create 3 test signals: pure tone, pure tone + white noise, white noise
+        # only.
+        pure_tone = signal_processing.SignalProcessingUtils.GeneratePureTone(
+            template, pure_tone_freq)
+        white_noise = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            template)
+        noisy_tone = signal_processing.SignalProcessingUtils.MixSignals(
+            pure_tone, white_noise)
+
+        # Compute scores for increasingly distorted pure tone signals.
+        scores = [None, None, None]
+        for index, tested_signal in enumerate(
+            [pure_tone, noisy_tone, white_noise]):
+            # Save signal.
+            tmp_filepath = os.path.join(self._output_path, 'tmp_thd.wav')
+            signal_processing.SignalProcessingUtils.SaveWav(
+                tmp_filepath, tested_signal)
+
+            # Compute score.
+            eval_score_worker.SetTestedSignalFilepath(tmp_filepath)
+            eval_score_worker.Run(self._output_path)
+            scores[index] = eval_score_worker.score
+
+            # Remove output file to avoid caching.
+            os.remove(eval_score_worker.output_filepath)
+
+        # Validate scores (lowest score with a pure tone).
+        self.assertTrue(all([scores[i + 1] > scores[i] for i in range(2)]))
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/evaluation.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/evaluation.py
new file mode 100644
index 0000000000..2599085329
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/evaluation.py
@@ -0,0 +1,57 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Evaluator of the APM module.
+"""
+
+import logging
+
+
+class ApmModuleEvaluator(object):
+    """APM evaluator class.
+  """
+
+    def __init__(self):
+        pass
+
+    @classmethod
+    def Run(cls, evaluation_score_workers, apm_input_metadata,
+            apm_output_filepath, reference_input_filepath,
+            render_input_filepath, output_path):
+        """Runs the evaluation.
+
+    Iterates over the given evaluation score workers.
+
+    Args:
+      evaluation_score_workers: list of EvaluationScore instances.
+      apm_input_metadata: dictionary with metadata of the APM input.
+      apm_output_filepath: path to the audio track file with the APM output.
+      reference_input_filepath: path to the reference audio track file.
+      output_path: output path.
+
+    Returns:
+      A dict of evaluation score name and score pairs.
+    """
+        # Init.
+        scores = {}
+
+        for evaluation_score_worker in evaluation_score_workers:
+            logging.info('   computing <%s> score',
+                         evaluation_score_worker.NAME)
+            evaluation_score_worker.SetInputSignalMetadata(apm_input_metadata)
+            evaluation_score_worker.SetReferenceSignalFilepath(
+                reference_input_filepath)
+            evaluation_score_worker.SetTestedSignalFilepath(
+                apm_output_filepath)
+            evaluation_score_worker.SetRenderSignalFilepath(
+                render_input_filepath)
+
+            evaluation_score_worker.Run(output_path)
+            scores[
+                evaluation_score_worker.NAME] = evaluation_score_worker.score
+
+        return scores
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/exceptions.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/exceptions.py
new file mode 100644
index 0000000000..893901d359
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/exceptions.py
@@ -0,0 +1,45 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Exception classes.
+"""
+
+
+class FileNotFoundError(Exception):
+    """File not found exception.
+  """
+    pass
+
+
+class SignalProcessingException(Exception):
+    """Signal processing exception.
+  """
+    pass
+
+
+class InputMixerException(Exception):
+    """Input mixer exception.
+  """
+    pass
+
+
+class InputSignalCreatorException(Exception):
+    """Input signal creator exception.
+  """
+    pass
+
+
+class EvaluationScoreException(Exception):
+    """Evaluation score exception.
+  """
+    pass
+
+
+class InitializationException(Exception):
+    """Initialization exception.
+  """
+    pass
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py
new file mode 100644
index 0000000000..fe3a6c7cb9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export.py
@@ -0,0 +1,426 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import functools
+import hashlib
+import logging
+import os
+import re
+import sys
+
+try:
+    import csscompressor
+except ImportError:
+    logging.critical(
+        'Cannot import the third-party Python package csscompressor')
+    sys.exit(1)
+
+try:
+    import jsmin
+except ImportError:
+    logging.critical('Cannot import the third-party Python package jsmin')
+    sys.exit(1)
+
+
+class HtmlExport(object):
+    """HTML exporter class for APM quality scores."""
+
+    _NEW_LINE = '\n'
+
+    # CSS and JS file paths.
+    _PATH = os.path.dirname(os.path.realpath(__file__))
+    _CSS_FILEPATH = os.path.join(_PATH, 'results.css')
+    _CSS_MINIFIED = True
+    _JS_FILEPATH = os.path.join(_PATH, 'results.js')
+    _JS_MINIFIED = True
+
+    def __init__(self, output_filepath):
+        self._scores_data_frame = None
+        self._output_filepath = output_filepath
+
+    def Export(self, scores_data_frame):
+        """Exports scores into an HTML file.
+
+    Args:
+      scores_data_frame: DataFrame instance.
+    """
+        self._scores_data_frame = scores_data_frame
+        html = [
+            '<html>',
+            self._BuildHeader(),
+            ('<script type="text/javascript">'
+             '(function () {'
+             'window.addEventListener(\'load\', function () {'
+             'var inspector = new AudioInspector();'
+             '});'
+             '})();'
+             '</script>'), '<body>',
+            self._BuildBody(), '</body>', '</html>'
+        ]
+        self._Save(self._output_filepath, self._NEW_LINE.join(html))
+
+    def _BuildHeader(self):
+        """Builds the <head> section of the HTML file.
+
+    The header contains the page title and either embedded or linked CSS and JS
+    files.
+
+    Returns:
+      A string with <head>...</head> HTML.
+    """
+        html = ['<head>', '<title>Results</title>']
+
+        # Add Material Design hosted libs.
+        html.append('<link rel="stylesheet" href="http://fonts.googleapis.com/'
+                    'css?family=Roboto:300,400,500,700" type="text/css">')
+        html.append(
+            '<link rel="stylesheet" href="https://fonts.googleapis.com/'
+            'icon?family=Material+Icons">')
+        html.append(
+            '<link rel="stylesheet" href="https://code.getmdl.io/1.3.0/'
+            'material.indigo-pink.min.css">')
+        html.append('<script defer src="https://code.getmdl.io/1.3.0/'
+                    'material.min.js"></script>')
+
+        # Embed custom JavaScript and CSS files.
+        html.append('<script>')
+        with open(self._JS_FILEPATH) as f:
+            html.append(
+                jsmin.jsmin(f.read()) if self._JS_MINIFIED else (
+                    f.read().rstrip()))
+        html.append('</script>')
+        html.append('<style>')
+        with open(self._CSS_FILEPATH) as f:
+            html.append(
+                csscompressor.compress(f.read()) if self._CSS_MINIFIED else (
+                    f.read().rstrip()))
+        html.append('</style>')
+
+        html.append('</head>')
+
+        return self._NEW_LINE.join(html)
+
+    def _BuildBody(self):
+        """Builds the content of the <body> section."""
+        score_names = self._scores_data_frame[
+            'eval_score_name'].drop_duplicates().values.tolist()
+
+        html = [
+            ('<div class="mdl-layout mdl-js-layout mdl-layout--fixed-header '
+             'mdl-layout--fixed-tabs">'),
+            '<header class="mdl-layout__header">',
+            '<div class="mdl-layout__header-row">',
+            '<span class="mdl-layout-title">APM QA results ({})</span>'.format(
+                self._output_filepath),
+            '</div>',
+        ]
+
+        # Tab selectors.
+        html.append('<div class="mdl-layout__tab-bar mdl-js-ripple-effect">')
+        for tab_index, score_name in enumerate(score_names):
+            is_active = tab_index == 0
+            html.append('<a href="#score-tab-{}" class="mdl-layout__tab{}">'
+                        '{}</a>'.format(tab_index,
+                                        ' is-active' if is_active else '',
+                                        self._FormatName(score_name)))
+        html.append('</div>')
+
+        html.append('</header>')
+        html.append(
+            '<main class="mdl-layout__content" style="overflow-x: auto;">')
+
+        # Tabs content.
+        for tab_index, score_name in enumerate(score_names):
+            html.append('<section class="mdl-layout__tab-panel{}" '
+                        'id="score-tab-{}">'.format(
+                            ' is-active' if is_active else '', tab_index))
+            html.append('<div class="page-content">')
+            html.append(
+                self._BuildScoreTab(score_name, ('s{}'.format(tab_index), )))
+            html.append('</div>')
+            html.append('</section>')
+
+        html.append('</main>')
+        html.append('</div>')
+
+        # Add snackbar for notifications.
+        html.append(
+            '<div id="snackbar" aria-live="assertive" aria-atomic="true"'
+            ' aria-relevant="text" class="mdl-snackbar mdl-js-snackbar">'
+            '<div class="mdl-snackbar__text"></div>'
+            '<button type="button" class="mdl-snackbar__action"></button>'
+            '</div>')
+
+        return self._NEW_LINE.join(html)
+
+    def _BuildScoreTab(self, score_name, anchor_data):
+        """Builds the content of a tab."""
+        # Find unique values.
+        scores = self._scores_data_frame[
+            self._scores_data_frame.eval_score_name == score_name]
+        apm_configs = sorted(self._FindUniqueTuples(scores, ['apm_config']))
+        test_data_gen_configs = sorted(
+            self._FindUniqueTuples(scores,
+                                   ['test_data_gen', 'test_data_gen_params']))
+
+        html = [
+            '<div class="mdl-grid">',
+            '<div class="mdl-layout-spacer"></div>',
+            '<div class="mdl-cell mdl-cell--10-col">',
+            ('<table class="mdl-data-table mdl-js-data-table mdl-shadow--2dp" '
+             'style="width: 100%;">'),
+        ]
+
+        # Header.
+        html.append('<thead><tr><th>APM config / Test data generator</th>')
+        for test_data_gen_info in test_data_gen_configs:
+            html.append('<th>{} {}</th>'.format(
+                self._FormatName(test_data_gen_info[0]),
+                test_data_gen_info[1]))
+        html.append('</tr></thead>')
+
+        # Body.
+        html.append('<tbody>')
+        for apm_config in apm_configs:
+            html.append('<tr><td>' + self._FormatName(apm_config[0]) + '</td>')
+            for test_data_gen_info in test_data_gen_configs:
+                dialog_id = self._ScoreStatsInspectorDialogId(
+                    score_name, apm_config[0], test_data_gen_info[0],
+                    test_data_gen_info[1])
+                html.append(
+                    '<td onclick="openScoreStatsInspector(\'{}\')">{}</td>'.
+                    format(
+                        dialog_id,
+                        self._BuildScoreTableCell(score_name,
+                                                  test_data_gen_info[0],
+                                                  test_data_gen_info[1],
+                                                  apm_config[0])))
+            html.append('</tr>')
+        html.append('</tbody>')
+
+        html.append(
+            '</table></div><div class="mdl-layout-spacer"></div></div>')
+
+        html.append(
+            self._BuildScoreStatsInspectorDialogs(score_name, apm_configs,
+                                                  test_data_gen_configs,
+                                                  anchor_data))
+
+        return self._NEW_LINE.join(html)
+
+    def _BuildScoreTableCell(self, score_name, test_data_gen,
+                             test_data_gen_params, apm_config):
+        """Builds the content of a table cell for a score table."""
+        scores = self._SliceDataForScoreTableCell(score_name, apm_config,
+                                                  test_data_gen,
+                                                  test_data_gen_params)
+        stats = self._ComputeScoreStats(scores)
+
+        html = []
+        items_id_prefix = (score_name + test_data_gen + test_data_gen_params +
+                           apm_config)
+        if stats['count'] == 1:
+            # Show the only available score.
+            item_id = hashlib.md5(items_id_prefix.encode('utf-8')).hexdigest()
+            html.append('<div id="single-value-{0}">{1:f}</div>'.format(
+                item_id, scores['score'].mean()))
+            html.append(
+                '<div class="mdl-tooltip" data-mdl-for="single-value-{}">{}'
+                '</div>'.format(item_id, 'single value'))
+        else:
+            # Show stats.
+            for stat_name in ['min', 'max', 'mean', 'std dev']:
+                item_id = hashlib.md5(
+                    (items_id_prefix + stat_name).encode('utf-8')).hexdigest()
+                html.append('<div id="stats-{0}">{1:f}</div>'.format(
+                    item_id, stats[stat_name]))
+                html.append(
+                    '<div class="mdl-tooltip" data-mdl-for="stats-{}">{}'
+                    '</div>'.format(item_id, stat_name))
+
+        return self._NEW_LINE.join(html)
+
+    def _BuildScoreStatsInspectorDialogs(self, score_name, apm_configs,
+                                         test_data_gen_configs, anchor_data):
+        """Builds a set of score stats inspector dialogs."""
+        html = []
+        for apm_config in apm_configs:
+            for test_data_gen_info in test_data_gen_configs:
+                dialog_id = self._ScoreStatsInspectorDialogId(
+                    score_name, apm_config[0], test_data_gen_info[0],
+                    test_data_gen_info[1])
+
+                html.append('<dialog class="mdl-dialog" id="{}" '
+                            'style="width: 40%;">'.format(dialog_id))
+
+                # Content.
+                html.append('<div class="mdl-dialog__content">')
+                html.append(
+                    '<h6><strong>APM config preset</strong>: {}<br/>'
+                    '<strong>Test data generator</strong>: {} ({})</h6>'.
+                    format(self._FormatName(apm_config[0]),
+                           self._FormatName(test_data_gen_info[0]),
+                           test_data_gen_info[1]))
+                html.append(
+                    self._BuildScoreStatsInspectorDialog(
+                        score_name, apm_config[0], test_data_gen_info[0],
+                        test_data_gen_info[1], anchor_data + (dialog_id, )))
+                html.append('</div>')
+
+                # Actions.
+                html.append('<div class="mdl-dialog__actions">')
+                html.append('<button type="button" class="mdl-button" '
+                            'onclick="closeScoreStatsInspector()">'
+                            'Close</button>')
+                html.append('</div>')
+
+                html.append('</dialog>')
+
+        return self._NEW_LINE.join(html)
+
+    def _BuildScoreStatsInspectorDialog(self, score_name, apm_config,
+                                        test_data_gen, test_data_gen_params,
+                                        anchor_data):
+        """Builds one score stats inspector dialog."""
+        scores = self._SliceDataForScoreTableCell(score_name, apm_config,
+                                                  test_data_gen,
+                                                  test_data_gen_params)
+
+        capture_render_pairs = sorted(
+            self._FindUniqueTuples(scores, ['capture', 'render']))
+        echo_simulators = sorted(
+            self._FindUniqueTuples(scores, ['echo_simulator']))
+
+        html = [
+            '<table class="mdl-data-table mdl-js-data-table mdl-shadow--2dp">'
+        ]
+
+        # Header.
+        html.append('<thead><tr><th>Capture-Render / Echo simulator</th>')
+        for echo_simulator in echo_simulators:
+            html.append('<th>' + self._FormatName(echo_simulator[0]) + '</th>')
+        html.append('</tr></thead>')
+
+        # Body.
+        html.append('<tbody>')
+        for row, (capture, render) in enumerate(capture_render_pairs):
+            html.append('<tr><td><div>{}</div><div>{}</div></td>'.format(
+                capture, render))
+            for col, echo_simulator in enumerate(echo_simulators):
+                score_tuple = self._SliceDataForScoreStatsTableCell(
+                    scores, capture, render, echo_simulator[0])
+                cell_class = 'r{}c{}'.format(row, col)
+                html.append('<td class="single-score-cell {}">{}</td>'.format(
+                    cell_class,
+                    self._BuildScoreStatsInspectorTableCell(
+                        score_tuple, anchor_data + (cell_class, ))))
+            html.append('</tr>')
+        html.append('</tbody>')
+
+        html.append('</table>')
+
+        # Placeholder for the audio inspector.
+        html.append('<div class="audio-inspector-placeholder"></div>')
+
+        return self._NEW_LINE.join(html)
+
+    def _BuildScoreStatsInspectorTableCell(self, score_tuple, anchor_data):
+        """Builds the content of a cell of a score stats inspector."""
+        anchor = '&'.join(anchor_data)
+        html = [('<div class="v">{}</div>'
+                 '<button class="mdl-button mdl-js-button mdl-button--icon"'
+                 ' data-anchor="{}">'
+                 '<i class="material-icons mdl-color-text--blue-grey">link</i>'
+                 '</button>').format(score_tuple.score, anchor)]
+
+        # Add all the available file paths as hidden data.
+        for field_name in score_tuple.keys():
+            if field_name.endswith('_filepath'):
+                html.append(
+                    '<input type="hidden" name="{}" value="{}">'.format(
+                        field_name, score_tuple[field_name]))
+
+        return self._NEW_LINE.join(html)
+
+    def _SliceDataForScoreTableCell(self, score_name, apm_config,
+                                    test_data_gen, test_data_gen_params):
+        """Slices `self._scores_data_frame` to extract the data for a tab."""
+        masks = []
+        masks.append(self._scores_data_frame.eval_score_name == score_name)
+        masks.append(self._scores_data_frame.apm_config == apm_config)
+        masks.append(self._scores_data_frame.test_data_gen == test_data_gen)
+        masks.append(self._scores_data_frame.test_data_gen_params ==
+                     test_data_gen_params)
+        mask = functools.reduce((lambda i1, i2: i1 & i2), masks)
+        del masks
+        return self._scores_data_frame[mask]
+
+    @classmethod
+    def _SliceDataForScoreStatsTableCell(cls, scores, capture, render,
+                                         echo_simulator):
+        """Slices `scores` to extract the data for a tab."""
+        masks = []
+
+        masks.append(scores.capture == capture)
+        masks.append(scores.render == render)
+        masks.append(scores.echo_simulator == echo_simulator)
+        mask = functools.reduce((lambda i1, i2: i1 & i2), masks)
+        del masks
+
+        sliced_data = scores[mask]
+        assert len(sliced_data) == 1, 'single score is expected'
+        return sliced_data.iloc[0]
+
+    @classmethod
+    def _FindUniqueTuples(cls, data_frame, fields):
+        """Slices `data_frame` to a list of fields and finds unique tuples."""
+        return data_frame[fields].drop_duplicates().values.tolist()
+
+    @classmethod
+    def _ComputeScoreStats(cls, data_frame):
+        """Computes score stats."""
+        scores = data_frame['score']
+        return {
+            'count': scores.count(),
+            'min': scores.min(),
+            'max': scores.max(),
+            'mean': scores.mean(),
+            'std dev': scores.std(),
+        }
+
+    @classmethod
+    def _ScoreStatsInspectorDialogId(cls, score_name, apm_config,
+                                     test_data_gen, test_data_gen_params):
+        """Assigns a unique name to a dialog."""
+        return 'score-stats-dialog-' + hashlib.md5(
+            'score-stats-inspector-{}-{}-{}-{}'.format(
+                score_name, apm_config, test_data_gen,
+                test_data_gen_params).encode('utf-8')).hexdigest()
+
+    @classmethod
+    def _Save(cls, output_filepath, html):
+        """Writes the HTML file.
+
+    Args:
+      output_filepath: output file path.
+      html: string with the HTML content.
+    """
+        with open(output_filepath, 'w') as f:
+            f.write(html)
+
+    @classmethod
+    def _FormatName(cls, name):
+        """Formats a name.
+
+    Args:
+      name: a string.
+
+    Returns:
+      A copy of name in which underscores and dashes are replaced with a space.
+    """
+        return re.sub(r'[_\-]', ' ', name)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export_unittest.py
new file mode 100644
index 0000000000..412aa7c4e7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/export_unittest.py
@@ -0,0 +1,86 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Unit tests for the export module.
+"""
+
+import logging
+import os
+import shutil
+import tempfile
+import unittest
+
+import pyquery as pq
+
+from . import audioproc_wrapper
+from . import collect_data
+from . import eval_scores_factory
+from . import evaluation
+from . import export
+from . import simulation
+from . import test_data_generation_factory
+
+
+class TestExport(unittest.TestCase):
+    """Unit tests for the export module.
+  """
+
+    _CLEAN_TMP_OUTPUT = True
+
+    def setUp(self):
+        """Creates temporary data to export."""
+        self._tmp_path = tempfile.mkdtemp()
+
+        # Run a fake experiment to produce data to export.
+        simulator = simulation.ApmModuleSimulator(
+            test_data_generator_factory=(
+                test_data_generation_factory.TestDataGeneratorFactory(
+                    aechen_ir_database_path='',
+                    noise_tracks_path='',
+                    copy_with_identity=False)),
+            evaluation_score_factory=(
+                eval_scores_factory.EvaluationScoreWorkerFactory(
+                    polqa_tool_bin_path=os.path.join(
+                        os.path.dirname(os.path.abspath(__file__)),
+                        'fake_polqa'),
+                    echo_metric_tool_bin_path=None)),
+            ap_wrapper=audioproc_wrapper.AudioProcWrapper(
+                audioproc_wrapper.AudioProcWrapper.
+                DEFAULT_APM_SIMULATOR_BIN_PATH),
+            evaluator=evaluation.ApmModuleEvaluator())
+        simulator.Run(
+            config_filepaths=['apm_configs/default.json'],
+            capture_input_filepaths=[
+                os.path.join(self._tmp_path, 'pure_tone-440_1000.wav'),
+                os.path.join(self._tmp_path, 'pure_tone-880_1000.wav'),
+            ],
+            test_data_generator_names=['identity', 'white_noise'],
+            eval_score_names=['audio_level_peak', 'audio_level_mean'],
+            output_dir=self._tmp_path)
+
+        # Export results.
+        p = collect_data.InstanceArgumentsParser()
+        args = p.parse_args(['--output_dir', self._tmp_path])
+        src_path = collect_data.ConstructSrcPath(args)
+        self._data_to_export = collect_data.FindScores(src_path, args)
+
+    def tearDown(self):
+        """Recursively deletes temporary folders."""
+        if self._CLEAN_TMP_OUTPUT:
+            shutil.rmtree(self._tmp_path)
+        else:
+            logging.warning(self.id() + ' did not clean the temporary path ' +
+                            (self._tmp_path))
+
+    def testCreateHtmlReport(self):
+        fn_out = os.path.join(self._tmp_path, 'results.html')
+        exporter = export.HtmlExport(fn_out)
+        exporter.Export(self._data_to_export)
+
+        document = pq.PyQuery(filename=fn_out)
+        self.assertIsInstance(document, pq.PyQuery)
+        # TODO(alessiob): Use PyQuery API to check the HTML file.
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/external_vad.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/external_vad.py
new file mode 100644
index 0000000000..a7db7b4840
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/external_vad.py
@@ -0,0 +1,75 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+from __future__ import division
+
+import logging
+import os
+import subprocess
+import shutil
+import sys
+import tempfile
+
+try:
+    import numpy as np
+except ImportError:
+    logging.critical('Cannot import the third-party Python package numpy')
+    sys.exit(1)
+
+from . import signal_processing
+
+
+class ExternalVad(object):
+    def __init__(self, path_to_binary, name):
+        """Args:
+       path_to_binary: path to binary that accepts '-i <wav>', '-o
+          <float probabilities>'. There must be one float value per
+          10ms audio
+       name: a name to identify the external VAD. Used for saving
+          the output as extvad_output-<name>.
+    """
+        self._path_to_binary = path_to_binary
+        self.name = name
+        assert os.path.exists(self._path_to_binary), (self._path_to_binary)
+        self._vad_output = None
+
+    def Run(self, wav_file_path):
+        _signal = signal_processing.SignalProcessingUtils.LoadWav(
+            wav_file_path)
+        if _signal.channels != 1:
+            raise NotImplementedError('Multiple-channel'
+                                      ' annotations not implemented')
+        if _signal.frame_rate != 48000:
+            raise NotImplementedError('Frame rates '
+                                      'other than 48000 not implemented')
+
+        tmp_path = tempfile.mkdtemp()
+        try:
+            output_file_path = os.path.join(tmp_path, self.name + '_vad.tmp')
+            subprocess.call([
+                self._path_to_binary, '-i', wav_file_path, '-o',
+                output_file_path
+            ])
+            self._vad_output = np.fromfile(output_file_path, np.float32)
+        except Exception as e:
+            logging.error('Error while running the ' + self.name + ' VAD (' +
+                          e.message + ')')
+        finally:
+            if os.path.exists(tmp_path):
+                shutil.rmtree(tmp_path)
+
+    def GetVadOutput(self):
+        assert self._vad_output is not None
+        return self._vad_output
+
+    @classmethod
+    def ConstructVadDict(cls, vad_paths, vad_names):
+        external_vads = {}
+        for path, name in zip(vad_paths, vad_names):
+            external_vads[name] = ExternalVad(path, name)
+        return external_vads
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_external_vad.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_external_vad.py
new file mode 100755
index 0000000000..f679f8c94a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_external_vad.py
@@ -0,0 +1,25 @@
+#!/usr/bin/python
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+import argparse
+import numpy as np
+
+
+def main():
+    parser = argparse.ArgumentParser()
+    parser.add_argument('-i', required=True)
+    parser.add_argument('-o', required=True)
+
+    args = parser.parse_args()
+
+    array = np.arange(100, dtype=np.float32)
+    array.tofile(open(args.o, 'w'))
+
+
+if __name__ == '__main__':
+    main()
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc
new file mode 100644
index 0000000000..6f3b2d1dd7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <fstream>
+#include <iostream>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+const char* const kErrorMessage = "-Out /path/to/output/file is mandatory";
+
+// Writes fake output intended to be parsed by
+// quality_assessment.eval_scores.PolqaScore.
+void WriteOutputFile(absl::string_view output_file_path) {
+  RTC_CHECK_NE(output_file_path, "");
+  std::ofstream out(std::string{output_file_path});
+  RTC_CHECK(!out.bad());
+  out << "* Fake Polqa output" << std::endl;
+  out << "FakeField1\tPolqaScore\tFakeField2" << std::endl;
+  out << "FakeValue1\t3.25\tFakeValue2" << std::endl;
+  out.close();
+}
+
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  // Find "-Out" and use its next argument as output file path.
+  RTC_CHECK_GE(argc, 3) << kErrorMessage;
+  const std::string kSoughtFlagName = "-Out";
+  for (int i = 1; i < argc - 1; ++i) {
+    if (kSoughtFlagName.compare(argv[i]) == 0) {
+      WriteOutputFile(argv[i + 1]);
+      return 0;
+    }
+  }
+  RTC_FATAL() << kErrorMessage;
+}
+
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::main(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py
new file mode 100644
index 0000000000..af022bd461
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer.py
@@ -0,0 +1,97 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Input mixer module.
+"""
+
+import logging
+import os
+
+from . import exceptions
+from . import signal_processing
+
+
+class ApmInputMixer(object):
+    """Class to mix a set of audio segments down to the APM input."""
+
+    _HARD_CLIPPING_LOG_MSG = 'hard clipping detected in the  mixed signal'
+
+    def __init__(self):
+        pass
+
+    @classmethod
+    def HardClippingLogMessage(cls):
+        """Returns the log message used when hard clipping is detected in the mix.
+
+    This method is mainly intended to be used by the unit tests.
+    """
+        return cls._HARD_CLIPPING_LOG_MSG
+
+    @classmethod
+    def Mix(cls, output_path, capture_input_filepath, echo_filepath):
+        """Mixes capture and echo.
+
+    Creates the overall capture input for APM by mixing the "echo-free" capture
+    signal with the echo signal (e.g., echo simulated via the
+    echo_path_simulation module).
+
+    The echo signal cannot be shorter than the capture signal and the generated
+    mix will have the same duration of the capture signal. The latter property
+    is enforced in order to let the input of APM and the reference signal
+    created by TestDataGenerator have the same length (required for the
+    evaluation step).
+
+    Hard-clipping may occur in the mix; a warning is raised when this happens.
+
+    If `echo_filepath` is None, nothing is done and `capture_input_filepath` is
+    returned.
+
+    Args:
+      speech: AudioSegment instance.
+      echo_path: AudioSegment instance or None.
+
+    Returns:
+      Path to the mix audio track file.
+    """
+        if echo_filepath is None:
+            return capture_input_filepath
+
+        # Build the mix output file name as a function of the echo file name.
+        # This ensures that if the internal parameters of the echo path simulator
+        # change, no erroneous cache hit occurs.
+        echo_file_name, _ = os.path.splitext(os.path.split(echo_filepath)[1])
+        capture_input_file_name, _ = os.path.splitext(
+            os.path.split(capture_input_filepath)[1])
+        mix_filepath = os.path.join(
+            output_path,
+            'mix_capture_{}_{}.wav'.format(capture_input_file_name,
+                                           echo_file_name))
+
+        # Create the mix if not done yet.
+        mix = None
+        if not os.path.exists(mix_filepath):
+            echo_free_capture = signal_processing.SignalProcessingUtils.LoadWav(
+                capture_input_filepath)
+            echo = signal_processing.SignalProcessingUtils.LoadWav(
+                echo_filepath)
+
+            if signal_processing.SignalProcessingUtils.CountSamples(echo) < (
+                    signal_processing.SignalProcessingUtils.CountSamples(
+                        echo_free_capture)):
+                raise exceptions.InputMixerException(
+                    'echo cannot be shorter than capture')
+
+            mix = echo_free_capture.overlay(echo)
+            signal_processing.SignalProcessingUtils.SaveWav(mix_filepath, mix)
+
+        # Check if hard clipping occurs.
+        if mix is None:
+            mix = signal_processing.SignalProcessingUtils.LoadWav(mix_filepath)
+        if signal_processing.SignalProcessingUtils.DetectHardClipping(mix):
+            logging.warning(cls._HARD_CLIPPING_LOG_MSG)
+
+        return mix_filepath
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer_unittest.py
new file mode 100644
index 0000000000..4fd5e4f1ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_mixer_unittest.py
@@ -0,0 +1,140 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Unit tests for the input mixer module.
+"""
+
+import logging
+import os
+import shutil
+import tempfile
+import unittest
+
+import mock
+
+from . import exceptions
+from . import input_mixer
+from . import signal_processing
+
+
+class TestApmInputMixer(unittest.TestCase):
+    """Unit tests for the ApmInputMixer class.
+  """
+
+    # Audio track file names created in setUp().
+    _FILENAMES = ['capture', 'echo_1', 'echo_2', 'shorter', 'longer']
+
+    # Target peak power level (dBFS) of each audio track file created in setUp().
+    # These values are hand-crafted in order to make saturation happen when
+    # capture and echo_2 are mixed and the contrary for capture and echo_1.
+    # None means that the power is not changed.
+    _MAX_PEAK_POWER_LEVELS = [-10.0, -5.0, 0.0, None, None]
+
+    # Audio track file durations in milliseconds.
+    _DURATIONS = [1000, 1000, 1000, 800, 1200]
+
+    _SAMPLE_RATE = 48000
+
+    def setUp(self):
+        """Creates temporary data."""
+        self._tmp_path = tempfile.mkdtemp()
+
+        # Create audio track files.
+        self._audio_tracks = {}
+        for filename, peak_power, duration in zip(self._FILENAMES,
+                                                  self._MAX_PEAK_POWER_LEVELS,
+                                                  self._DURATIONS):
+            audio_track_filepath = os.path.join(self._tmp_path,
+                                                '{}.wav'.format(filename))
+
+            # Create a pure tone with the target peak power level.
+            template = signal_processing.SignalProcessingUtils.GenerateSilence(
+                duration=duration, sample_rate=self._SAMPLE_RATE)
+            signal = signal_processing.SignalProcessingUtils.GeneratePureTone(
+                template)
+            if peak_power is not None:
+                signal = signal.apply_gain(-signal.max_dBFS + peak_power)
+
+            signal_processing.SignalProcessingUtils.SaveWav(
+                audio_track_filepath, signal)
+            self._audio_tracks[filename] = {
+                'filepath':
+                audio_track_filepath,
+                'num_samples':
+                signal_processing.SignalProcessingUtils.CountSamples(signal)
+            }
+
+    def tearDown(self):
+        """Recursively deletes temporary folders."""
+        shutil.rmtree(self._tmp_path)
+
+    def testCheckMixSameDuration(self):
+        """Checks the duration when mixing capture and echo with same duration."""
+        mix_filepath = input_mixer.ApmInputMixer.Mix(
+            self._tmp_path, self._audio_tracks['capture']['filepath'],
+            self._audio_tracks['echo_1']['filepath'])
+        self.assertTrue(os.path.exists(mix_filepath))
+
+        mix = signal_processing.SignalProcessingUtils.LoadWav(mix_filepath)
+        self.assertEqual(
+            self._audio_tracks['capture']['num_samples'],
+            signal_processing.SignalProcessingUtils.CountSamples(mix))
+
+    def testRejectShorterEcho(self):
+        """Rejects echo signals that are shorter than the capture signal."""
+        try:
+            _ = input_mixer.ApmInputMixer.Mix(
+                self._tmp_path, self._audio_tracks['capture']['filepath'],
+                self._audio_tracks['shorter']['filepath'])
+            self.fail('no exception raised')
+        except exceptions.InputMixerException:
+            pass
+
+    def testCheckMixDurationWithLongerEcho(self):
+        """Checks the duration when mixing an echo longer than the capture."""
+        mix_filepath = input_mixer.ApmInputMixer.Mix(
+            self._tmp_path, self._audio_tracks['capture']['filepath'],
+            self._audio_tracks['longer']['filepath'])
+        self.assertTrue(os.path.exists(mix_filepath))
+
+        mix = signal_processing.SignalProcessingUtils.LoadWav(mix_filepath)
+        self.assertEqual(
+            self._audio_tracks['capture']['num_samples'],
+            signal_processing.SignalProcessingUtils.CountSamples(mix))
+
+    def testCheckOutputFileNamesConflict(self):
+        """Checks that different echo files lead to different output file names."""
+        mix1_filepath = input_mixer.ApmInputMixer.Mix(
+            self._tmp_path, self._audio_tracks['capture']['filepath'],
+            self._audio_tracks['echo_1']['filepath'])
+        self.assertTrue(os.path.exists(mix1_filepath))
+
+        mix2_filepath = input_mixer.ApmInputMixer.Mix(
+            self._tmp_path, self._audio_tracks['capture']['filepath'],
+            self._audio_tracks['echo_2']['filepath'])
+        self.assertTrue(os.path.exists(mix2_filepath))
+
+        self.assertNotEqual(mix1_filepath, mix2_filepath)
+
+    def testHardClippingLogExpected(self):
+        """Checks that hard clipping warning is raised when occurring."""
+        logging.warning = mock.MagicMock(name='warning')
+        _ = input_mixer.ApmInputMixer.Mix(
+            self._tmp_path, self._audio_tracks['capture']['filepath'],
+            self._audio_tracks['echo_2']['filepath'])
+        logging.warning.assert_called_once_with(
+            input_mixer.ApmInputMixer.HardClippingLogMessage())
+
+    def testHardClippingLogNotExpected(self):
+        """Checks that hard clipping warning is not raised when not occurring."""
+        logging.warning = mock.MagicMock(name='warning')
+        _ = input_mixer.ApmInputMixer.Mix(
+            self._tmp_path, self._audio_tracks['capture']['filepath'],
+            self._audio_tracks['echo_1']['filepath'])
+        self.assertNotIn(
+            mock.call(input_mixer.ApmInputMixer.HardClippingLogMessage()),
+            logging.warning.call_args_list)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py
new file mode 100644
index 0000000000..b64fdcca89
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py
@@ -0,0 +1,68 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Input signal creator module.
+"""
+
+from . import exceptions
+from . import signal_processing
+
+
+class InputSignalCreator(object):
+    """Input signal creator class.
+  """
+
+    @classmethod
+    def Create(cls, name, raw_params):
+        """Creates a input signal and its metadata.
+
+    Args:
+      name: Input signal creator name.
+      raw_params: Tuple of parameters to pass to the specific signal creator.
+
+    Returns:
+      (AudioSegment, dict) tuple.
+    """
+        try:
+            signal = {}
+            params = {}
+
+            if name == 'pure_tone':
+                params['frequency'] = float(raw_params[0])
+                params['duration'] = int(raw_params[1])
+                signal = cls._CreatePureTone(params['frequency'],
+                                             params['duration'])
+            else:
+                raise exceptions.InputSignalCreatorException(
+                    'Invalid input signal creator name')
+
+            # Complete metadata.
+            params['signal'] = name
+
+            return signal, params
+        except (TypeError, AssertionError) as e:
+            raise exceptions.InputSignalCreatorException(
+                'Invalid signal creator parameters: {}'.format(e))
+
+    @classmethod
+    def _CreatePureTone(cls, frequency, duration):
+        """
+    Generates a pure tone at 48000 Hz.
+
+    Args:
+      frequency: Float in (0-24000] (Hz).
+      duration: Integer (milliseconds).
+
+    Returns:
+      AudioSegment instance.
+    """
+        assert 0 < frequency <= 24000
+        assert duration > 0
+        template = signal_processing.SignalProcessingUtils.GenerateSilence(
+            duration)
+        return signal_processing.SignalProcessingUtils.GeneratePureTone(
+            template, frequency)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.css b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.css
new file mode 100644
index 0000000000..2f406bb002
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.css
@@ -0,0 +1,32 @@
+/* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS.  All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+td.selected-score {
+  background-color: #DDD;
+}
+
+td.single-score-cell{
+  text-align: center;
+}
+
+.audio-inspector {
+  text-align: center;
+}
+
+.audio-inspector div{
+  margin-bottom: 0;
+  padding-bottom: 0;
+  padding-top: 0;
+}
+
+.audio-inspector div div{
+  margin-bottom: 0;
+  padding-bottom: 0;
+  padding-top: 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.js b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.js
new file mode 100644
index 0000000000..8e47411058
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/results.js
@@ -0,0 +1,376 @@
+// Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+//
+// Use of this source code is governed by a BSD-style license
+// that can be found in the LICENSE file in the root of the source
+// tree. An additional intellectual property rights grant can be found
+// in the file PATENTS.  All contributing project authors may
+// be found in the AUTHORS file in the root of the source tree.
+
+/**
+ * Opens the score stats inspector dialog.
+ * @param {String} dialogId: identifier of the dialog to show.
+ * @return {DOMElement} The dialog element that has been opened.
+ */
+function openScoreStatsInspector(dialogId) {
+  var dialog = document.getElementById(dialogId);
+  dialog.showModal();
+  return dialog;
+}
+
+/**
+ * Closes the score stats inspector dialog.
+ */
+function closeScoreStatsInspector() {
+  var dialog = document.querySelector('dialog[open]');
+  if (dialog == null)
+    return;
+  dialog.close();
+}
+
+/**
+ * Audio inspector class.
+ * @constructor
+ */
+function AudioInspector() {
+  console.debug('Creating an AudioInspector instance.');
+  this.audioPlayer_ = new Audio();
+  this.metadata_ = {};
+  this.currentScore_ = null;
+  this.audioInspector_ = null;
+  this.snackbarContainer_ = document.querySelector('#snackbar');
+
+  // Get base URL without anchors.
+  this.baseUrl_ = window.location.href;
+  var index = this.baseUrl_.indexOf('#');
+  if (index > 0)
+    this.baseUrl_ = this.baseUrl_.substr(0, index)
+  console.info('Base URL set to "' + window.location.href + '".');
+
+  window.event.stopPropagation();
+  this.createTextAreasForCopy_();
+  this.createAudioInspector_();
+  this.initializeEventHandlers_();
+
+  // When MDL is ready, parse the anchor (if any) to show the requested
+  // experiment.
+  var self = this;
+  document.querySelectorAll('header a')[0].addEventListener(
+      'mdl-componentupgraded', function() {
+    if (!self.parseWindowAnchor()) {
+      // If not experiment is requested, open the first section.
+      console.info('No anchor parsing, opening the first section.');
+      document.querySelectorAll('header a > span')[0].click();
+    }
+  });
+}
+
+/**
+ * Parse the anchor in the window URL.
+ * @return {bool} True if the parsing succeeded.
+ */
+AudioInspector.prototype.parseWindowAnchor = function() {
+  var index = location.href.indexOf('#');
+  if (index == -1) {
+    console.debug('No # found in the URL.');
+    return false;
+  }
+
+  var anchor = location.href.substr(index - location.href.length + 1);
+  console.info('Anchor changed: "' + anchor + '".');
+
+  var parts = anchor.split('&');
+  if (parts.length != 3) {
+    console.info('Ignoring anchor with invalid number of fields.');
+    return false;
+  }
+
+  var openDialog = document.querySelector('dialog[open]');
+  try {
+    // Open the requested dialog if not already open.
+    if (!openDialog || openDialog.id != parts[1]) {
+      !openDialog || openDialog.close();
+      document.querySelectorAll('header a > span')[
+          parseInt(parts[0].substr(1))].click();
+      openDialog = openScoreStatsInspector(parts[1]);
+    }
+
+    // Trigger click on cell.
+    var cell = openDialog.querySelector('td.' + parts[2]);
+    cell.focus();
+    cell.click();
+
+    this.showNotification_('Experiment selected.');
+    return true;
+  } catch (e) {
+    this.showNotification_('Cannot select experiment :(');
+    console.error('Exception caught while selecting experiment: "' + e + '".');
+  }
+
+  return false;
+}
+
+/**
+ * Set up the inspector for a new score.
+ * @param {DOMElement} element: Element linked to the selected score.
+ */
+AudioInspector.prototype.selectedScoreChange = function(element) {
+  if (this.currentScore_ == element) { return; }
+  if (this.currentScore_ != null) {
+    this.currentScore_.classList.remove('selected-score');
+  }
+  this.currentScore_ = element;
+  this.currentScore_.classList.add('selected-score');
+  this.stopAudio();
+
+  // Read metadata.
+  var matches = element.querySelectorAll('input[type=hidden]');
+  this.metadata_ = {};
+  for (var index = 0; index < matches.length; ++index) {
+    this.metadata_[matches[index].name] = matches[index].value;
+  }
+
+  // Show the audio inspector interface.
+  var container = element.parentNode.parentNode.parentNode.parentNode;
+  var audioInspectorPlaceholder = container.querySelector(
+      '.audio-inspector-placeholder');
+  this.moveInspector_(audioInspectorPlaceholder);
+};
+
+/**
+ * Stop playing audio.
+ */
+AudioInspector.prototype.stopAudio = function() {
+  console.info('Pausing audio play out.');
+  this.audioPlayer_.pause();
+};
+
+/**
+ * Show a text message using the snackbar.
+ */
+AudioInspector.prototype.showNotification_ = function(text) {
+  try {
+    this.snackbarContainer_.MaterialSnackbar.showSnackbar({
+        message: text, timeout: 2000});
+  } catch (e) {
+    // Fallback to an alert.
+    alert(text);
+    console.warn('Cannot use snackbar: "' + e + '"');
+  }
+}
+
+/**
+ * Move the audio inspector DOM node into the given parent.
+ * @param {DOMElement} newParentNode: New parent for the inspector.
+ */
+AudioInspector.prototype.moveInspector_ = function(newParentNode) {
+  newParentNode.appendChild(this.audioInspector_);
+};
+
+/**
+ * Play audio file from url.
+ * @param {string} metadataFieldName: Metadata field name.
+ */
+AudioInspector.prototype.playAudio = function(metadataFieldName) {
+  if (this.metadata_[metadataFieldName] == undefined) { return; }
+  if (this.metadata_[metadataFieldName] == 'None') {
+    alert('The selected stream was not used during the experiment.');
+    return;
+  }
+  this.stopAudio();
+  this.audioPlayer_.src = this.metadata_[metadataFieldName];
+  console.debug('Audio source URL: "' + this.audioPlayer_.src + '"');
+  this.audioPlayer_.play();
+  console.info('Playing out audio.');
+};
+
+/**
+ * Create hidden text areas to copy URLs.
+ *
+ * For each dialog, one text area is created since it is not possible to select
+ * text on a text area outside of the active dialog.
+ */
+AudioInspector.prototype.createTextAreasForCopy_ = function() {
+  var self = this;
+  document.querySelectorAll('dialog.mdl-dialog').forEach(function(element) {
+    var textArea = document.createElement("textarea");
+    textArea.classList.add('url-copy');
+    textArea.style.position = 'fixed';
+    textArea.style.bottom = 0;
+    textArea.style.left = 0;
+    textArea.style.width = '2em';
+    textArea.style.height = '2em';
+    textArea.style.border = 'none';
+    textArea.style.outline = 'none';
+    textArea.style.boxShadow = 'none';
+    textArea.style.background = 'transparent';
+    textArea.style.fontSize = '6px';
+    element.appendChild(textArea);
+  });
+}
+
+/**
+ * Create audio inspector.
+ */
+AudioInspector.prototype.createAudioInspector_ = function() {
+  var buttonIndex = 0;
+  function getButtonHtml(icon, toolTipText, caption, metadataFieldName) {
+    var buttonId = 'audioInspectorButton' + buttonIndex++;
+    html = caption == null ? '' : caption;
+    html += '<button class="mdl-button mdl-js-button mdl-button--icon ' +
+                'mdl-js-ripple-effect" id="' + buttonId + '">' +
+              '<i class="material-icons">' + icon + '</i>' +
+              '<div class="mdl-tooltip" data-mdl-for="' + buttonId + '">' +
+                 toolTipText +
+              '</div>';
+    if (metadataFieldName != null) {
+      html += '<input type="hidden" value="' + metadataFieldName + '">'
+    }
+    html += '</button>'
+
+    return html;
+  }
+
+  // TODO(alessiob): Add timeline and highlight current track by changing icon
+  // color.
+
+  this.audioInspector_ = document.createElement('div');
+  this.audioInspector_.classList.add('audio-inspector');
+  this.audioInspector_.innerHTML =
+      '<div class="mdl-grid">' +
+        '<div class="mdl-layout-spacer"></div>' +
+        '<div class="mdl-cell mdl-cell--2-col">' +
+          getButtonHtml('play_arrow', 'Simulated echo', 'E<sub>in</sub>',
+                        'echo_filepath') +
+        '</div>' +
+        '<div class="mdl-cell mdl-cell--2-col">' +
+          getButtonHtml('stop', 'Stop playing [S]', null, '__stop__') +
+        '</div>' +
+        '<div class="mdl-cell mdl-cell--2-col">' +
+          getButtonHtml('play_arrow', 'Render stream', 'R<sub>in</sub>',
+                        'render_filepath') +
+        '</div>' +
+        '<div class="mdl-layout-spacer"></div>' +
+      '</div>' +
+      '<div class="mdl-grid">' +
+        '<div class="mdl-layout-spacer"></div>' +
+        '<div class="mdl-cell mdl-cell--2-col">' +
+          getButtonHtml('play_arrow', 'Capture stream (APM input) [1]',
+                        'Y\'<sub>in</sub>', 'capture_filepath') +
+        '</div>' +
+        '<div class="mdl-cell mdl-cell--2-col"><strong>APM</strong></div>' +
+        '<div class="mdl-cell mdl-cell--2-col">' +
+          getButtonHtml('play_arrow', 'APM output [2]', 'Y<sub>out</sub>',
+                        'apm_output_filepath') +
+        '</div>' +
+        '<div class="mdl-layout-spacer"></div>' +
+      '</div>' +
+      '<div class="mdl-grid">' +
+        '<div class="mdl-layout-spacer"></div>' +
+        '<div class="mdl-cell mdl-cell--2-col">' +
+          getButtonHtml('play_arrow', 'Echo-free capture stream',
+                        'Y<sub>in</sub>', 'echo_free_capture_filepath') +
+        '</div>' +
+        '<div class="mdl-cell mdl-cell--2-col">' +
+          getButtonHtml('play_arrow', 'Clean capture stream',
+                        'Y<sub>clean</sub>', 'clean_capture_input_filepath') +
+        '</div>' +
+        '<div class="mdl-cell mdl-cell--2-col">' +
+          getButtonHtml('play_arrow', 'APM reference [3]', 'Y<sub>ref</sub>',
+                        'apm_reference_filepath') +
+        '</div>' +
+        '<div class="mdl-layout-spacer"></div>' +
+      '</div>';
+
+  // Add an invisible node as initial container for the audio inspector.
+  var parent = document.createElement('div');
+  parent.style.display = 'none';
+  this.moveInspector_(parent);
+  document.body.appendChild(parent);
+};
+
+/**
+ * Initialize event handlers.
+ */
+AudioInspector.prototype.initializeEventHandlers_ = function() {
+  var self = this;
+
+  // Score cells.
+  document.querySelectorAll('td.single-score-cell').forEach(function(element) {
+    element.onclick = function() {
+      self.selectedScoreChange(this);
+    }
+  });
+
+  // Copy anchor URLs icons.
+  if (document.queryCommandSupported('copy')) {
+    document.querySelectorAll('td.single-score-cell button').forEach(
+        function(element) {
+      element.onclick = function() {
+        // Find the text area in the dialog.
+        var textArea = element.closest('dialog').querySelector(
+            'textarea.url-copy');
+
+        // Copy.
+        textArea.value = self.baseUrl_ + '#' + element.getAttribute(
+            'data-anchor');
+        textArea.select();
+        try {
+          if (!document.execCommand('copy'))
+            throw 'Copy returned false';
+          self.showNotification_('Experiment URL copied.');
+        } catch (e) {
+          self.showNotification_('Cannot copy experiment URL :(');
+          console.error(e);
+        }
+      }
+    });
+  } else {
+    self.showNotification_(
+        'The copy command is disabled. URL copy is not enabled.');
+  }
+
+  // Audio inspector buttons.
+  this.audioInspector_.querySelectorAll('button').forEach(function(element) {
+    var target = element.querySelector('input[type=hidden]');
+    if (target == null) { return; }
+    element.onclick = function() {
+      if (target.value == '__stop__') {
+        self.stopAudio();
+      } else {
+        self.playAudio(target.value);
+      }
+    };
+  });
+
+  // Dialog close handlers.
+  var dialogs = document.querySelectorAll('dialog').forEach(function(element) {
+    element.onclose = function() {
+      self.stopAudio();
+    }
+  });
+
+  // Keyboard shortcuts.
+  window.onkeyup = function(e) {
+    var key = e.keyCode ? e.keyCode : e.which;
+    switch (key) {
+      case 49:  // 1.
+        self.playAudio('capture_filepath');
+        break;
+      case 50:  // 2.
+        self.playAudio('apm_output_filepath');
+        break;
+      case 51:  // 3.
+        self.playAudio('apm_reference_filepath');
+        break;
+      case 83:  // S.
+      case 115:  // s.
+        self.stopAudio();
+        break;
+    }
+  };
+
+  // Hash change.
+  window.onhashchange = function(e) {
+    self.parseWindowAnchor();
+  }
+};
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py
new file mode 100644
index 0000000000..95e801903d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing.py
@@ -0,0 +1,359 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Signal processing utility module.
+"""
+
+import array
+import logging
+import os
+import sys
+import enum
+
+try:
+    import numpy as np
+except ImportError:
+    logging.critical('Cannot import the third-party Python package numpy')
+    sys.exit(1)
+
+try:
+    import pydub
+    import pydub.generators
+except ImportError:
+    logging.critical('Cannot import the third-party Python package pydub')
+    sys.exit(1)
+
+try:
+    import scipy.signal
+    import scipy.fftpack
+except ImportError:
+    logging.critical('Cannot import the third-party Python package scipy')
+    sys.exit(1)
+
+from . import exceptions
+
+
+class SignalProcessingUtils(object):
+    """Collection of signal processing utilities.
+  """
+
+    @enum.unique
+    class MixPadding(enum.Enum):
+        NO_PADDING = 0
+        ZERO_PADDING = 1
+        LOOP = 2
+
+    def __init__(self):
+        pass
+
+    @classmethod
+    def LoadWav(cls, filepath, channels=1):
+        """Loads wav file.
+
+    Args:
+      filepath: path to the wav audio track file to load.
+      channels: number of channels (downmixing to mono by default).
+
+    Returns:
+      AudioSegment instance.
+    """
+        if not os.path.exists(filepath):
+            logging.error('cannot find the <%s> audio track file', filepath)
+            raise exceptions.FileNotFoundError()
+        return pydub.AudioSegment.from_file(filepath,
+                                            format='wav',
+                                            channels=channels)
+
+    @classmethod
+    def SaveWav(cls, output_filepath, signal):
+        """Saves wav file.
+
+    Args:
+      output_filepath: path to the wav audio track file to save.
+      signal: AudioSegment instance.
+    """
+        return signal.export(output_filepath, format='wav')
+
+    @classmethod
+    def CountSamples(cls, signal):
+        """Number of samples per channel.
+
+    Args:
+      signal: AudioSegment instance.
+
+    Returns:
+      An integer.
+    """
+        number_of_samples = len(signal.get_array_of_samples())
+        assert signal.channels > 0
+        assert number_of_samples % signal.channels == 0
+        return number_of_samples / signal.channels
+
+    @classmethod
+    def GenerateSilence(cls, duration=1000, sample_rate=48000):
+        """Generates silence.
+
+    This method can also be used to create a template AudioSegment instance.
+    A template can then be used with other Generate*() methods accepting an
+    AudioSegment instance as argument.
+
+    Args:
+      duration: duration in ms.
+      sample_rate: sample rate.
+
+    Returns:
+      AudioSegment instance.
+    """
+        return pydub.AudioSegment.silent(duration, sample_rate)
+
+    @classmethod
+    def GeneratePureTone(cls, template, frequency=440.0):
+        """Generates a pure tone.
+
+    The pure tone is generated with the same duration and in the same format of
+    the given template signal.
+
+    Args:
+      template: AudioSegment instance.
+      frequency: Frequency of the pure tone in Hz.
+
+    Return:
+      AudioSegment instance.
+    """
+        if frequency > template.frame_rate >> 1:
+            raise exceptions.SignalProcessingException('Invalid frequency')
+
+        generator = pydub.generators.Sine(sample_rate=template.frame_rate,
+                                          bit_depth=template.sample_width * 8,
+                                          freq=frequency)
+
+        return generator.to_audio_segment(duration=len(template), volume=0.0)
+
+    @classmethod
+    def GenerateWhiteNoise(cls, template):
+        """Generates white noise.
+
+    The white noise is generated with the same duration and in the same format
+    of the given template signal.
+
+    Args:
+      template: AudioSegment instance.
+
+    Return:
+      AudioSegment instance.
+    """
+        generator = pydub.generators.WhiteNoise(
+            sample_rate=template.frame_rate,
+            bit_depth=template.sample_width * 8)
+        return generator.to_audio_segment(duration=len(template), volume=0.0)
+
+    @classmethod
+    def AudioSegmentToRawData(cls, signal):
+        samples = signal.get_array_of_samples()
+        if samples.typecode != 'h':
+            raise exceptions.SignalProcessingException(
+                'Unsupported samples type')
+        return np.array(signal.get_array_of_samples(), np.int16)
+
+    @classmethod
+    def Fft(cls, signal, normalize=True):
+        if signal.channels != 1:
+            raise NotImplementedError('multiple-channel FFT not implemented')
+        x = cls.AudioSegmentToRawData(signal).astype(np.float32)
+        if normalize:
+            x /= max(abs(np.max(x)), 1.0)
+        y = scipy.fftpack.fft(x)
+        return y[:len(y) / 2]
+
+    @classmethod
+    def DetectHardClipping(cls, signal, threshold=2):
+        """Detects hard clipping.
+
+    Hard clipping is simply detected by counting samples that touch either the
+    lower or upper bound too many times in a row (according to `threshold`).
+    The presence of a single sequence of samples meeting such property is enough
+    to label the signal as hard clipped.
+
+    Args:
+      signal: AudioSegment instance.
+      threshold: minimum number of samples at full-scale in a row.
+
+    Returns:
+      True if hard clipping is detect, False otherwise.
+    """
+        if signal.channels != 1:
+            raise NotImplementedError(
+                'multiple-channel clipping not implemented')
+        if signal.sample_width != 2:  # Note that signal.sample_width is in bytes.
+            raise exceptions.SignalProcessingException(
+                'hard-clipping detection only supported for 16 bit samples')
+        samples = cls.AudioSegmentToRawData(signal)
+
+        # Detect adjacent clipped samples.
+        samples_type_info = np.iinfo(samples.dtype)
+        mask_min = samples == samples_type_info.min
+        mask_max = samples == samples_type_info.max
+
+        def HasLongSequence(vector, min_legth=threshold):
+            """Returns True if there are one or more long sequences of True flags."""
+            seq_length = 0
+            for b in vector:
+                seq_length = seq_length + 1 if b else 0
+                if seq_length >= min_legth:
+                    return True
+            return False
+
+        return HasLongSequence(mask_min) or HasLongSequence(mask_max)
+
+    @classmethod
+    def ApplyImpulseResponse(cls, signal, impulse_response):
+        """Applies an impulse response to a signal.
+
+    Args:
+      signal: AudioSegment instance.
+      impulse_response: list or numpy vector of float values.
+
+    Returns:
+      AudioSegment instance.
+    """
+        # Get samples.
+        assert signal.channels == 1, (
+            'multiple-channel recordings not supported')
+        samples = signal.get_array_of_samples()
+
+        # Convolve.
+        logging.info(
+            'applying %d order impulse response to a signal lasting %d ms',
+            len(impulse_response), len(signal))
+        convolved_samples = scipy.signal.fftconvolve(in1=samples,
+                                                     in2=impulse_response,
+                                                     mode='full').astype(
+                                                         np.int16)
+        logging.info('convolution computed')
+
+        # Cast.
+        convolved_samples = array.array(signal.array_type, convolved_samples)
+
+        # Verify.
+        logging.debug('signal length: %d samples', len(samples))
+        logging.debug('convolved signal length: %d samples',
+                      len(convolved_samples))
+        assert len(convolved_samples) > len(samples)
+
+        # Generate convolved signal AudioSegment instance.
+        convolved_signal = pydub.AudioSegment(data=convolved_samples,
+                                              metadata={
+                                                  'sample_width':
+                                                  signal.sample_width,
+                                                  'frame_rate':
+                                                  signal.frame_rate,
+                                                  'frame_width':
+                                                  signal.frame_width,
+                                                  'channels': signal.channels,
+                                              })
+        assert len(convolved_signal) > len(signal)
+
+        return convolved_signal
+
+    @classmethod
+    def Normalize(cls, signal):
+        """Normalizes a signal.
+
+    Args:
+      signal: AudioSegment instance.
+
+    Returns:
+      An AudioSegment instance.
+    """
+        return signal.apply_gain(-signal.max_dBFS)
+
+    @classmethod
+    def Copy(cls, signal):
+        """Makes a copy os a signal.
+
+    Args:
+      signal: AudioSegment instance.
+
+    Returns:
+      An AudioSegment instance.
+    """
+        return pydub.AudioSegment(data=signal.get_array_of_samples(),
+                                  metadata={
+                                      'sample_width': signal.sample_width,
+                                      'frame_rate': signal.frame_rate,
+                                      'frame_width': signal.frame_width,
+                                      'channels': signal.channels,
+                                  })
+
+    @classmethod
+    def MixSignals(cls,
+                   signal,
+                   noise,
+                   target_snr=0.0,
+                   pad_noise=MixPadding.NO_PADDING):
+        """Mixes `signal` and `noise` with a target SNR.
+
+    Mix `signal` and `noise` with a desired SNR by scaling `noise`.
+    If the target SNR is +/- infinite, a copy of signal/noise is returned.
+    If `signal` is shorter than `noise`, the length of the mix equals that of
+    `signal`. Otherwise, the mix length depends on whether padding is applied.
+    When padding is not applied, that is `pad_noise` is set to NO_PADDING
+    (default), the mix length equals that of `noise` - i.e., `signal` is
+    truncated. Otherwise, `noise` is extended and the resulting mix has the same
+    length of `signal`.
+
+    Args:
+      signal: AudioSegment instance (signal).
+      noise: AudioSegment instance (noise).
+      target_snr: float, numpy.Inf or -numpy.Inf (dB).
+      pad_noise: SignalProcessingUtils.MixPadding, default: NO_PADDING.
+
+    Returns:
+      An AudioSegment instance.
+    """
+        # Handle infinite target SNR.
+        if target_snr == -np.Inf:
+            # Return a copy of noise.
+            logging.warning('SNR = -Inf, returning noise')
+            return cls.Copy(noise)
+        elif target_snr == np.Inf:
+            # Return a copy of signal.
+            logging.warning('SNR = +Inf, returning signal')
+            return cls.Copy(signal)
+
+        # Check signal and noise power.
+        signal_power = float(signal.dBFS)
+        noise_power = float(noise.dBFS)
+        if signal_power == -np.Inf:
+            logging.error('signal has -Inf power, cannot mix')
+            raise exceptions.SignalProcessingException(
+                'cannot mix a signal with -Inf power')
+        if noise_power == -np.Inf:
+            logging.error('noise has -Inf power, cannot mix')
+            raise exceptions.SignalProcessingException(
+                'cannot mix a signal with -Inf power')
+
+        # Mix.
+        gain_db = signal_power - noise_power - target_snr
+        signal_duration = len(signal)
+        noise_duration = len(noise)
+        if signal_duration <= noise_duration:
+            # Ignore `pad_noise`, `noise` is truncated if longer that `signal`, the
+            # mix will have the same length of `signal`.
+            return signal.overlay(noise.apply_gain(gain_db))
+        elif pad_noise == cls.MixPadding.NO_PADDING:
+            # `signal` is longer than `noise`, but no padding is applied to `noise`.
+            # Truncate `signal`.
+            return noise.overlay(signal, gain_during_overlay=gain_db)
+        elif pad_noise == cls.MixPadding.ZERO_PADDING:
+            # TODO(alessiob): Check that this works as expected.
+            return signal.overlay(noise.apply_gain(gain_db))
+        elif pad_noise == cls.MixPadding.LOOP:
+            # `signal` is longer than `noise`, extend `noise` by looping.
+            return signal.overlay(noise.apply_gain(gain_db), loop=True)
+        else:
+            raise exceptions.SignalProcessingException('invalid padding type')
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing_unittest.py
new file mode 100644
index 0000000000..881fb66800
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/signal_processing_unittest.py
@@ -0,0 +1,183 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Unit tests for the signal_processing module.
+"""
+
+import unittest
+
+import numpy as np
+import pydub
+
+from . import exceptions
+from . import signal_processing
+
+
+class TestSignalProcessing(unittest.TestCase):
+    """Unit tests for the signal_processing module.
+  """
+
+    def testMixSignals(self):
+        # Generate a template signal with which white noise can be generated.
+        silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000)
+
+        # Generate two distinct AudioSegment instances with 1 second of white noise.
+        signal = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            silence)
+        noise = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            silence)
+
+        # Extract samples.
+        signal_samples = signal.get_array_of_samples()
+        noise_samples = noise.get_array_of_samples()
+
+        # Test target SNR -Inf (noise expected).
+        mix_neg_inf = signal_processing.SignalProcessingUtils.MixSignals(
+            signal, noise, -np.Inf)
+        self.assertTrue(len(noise), len(mix_neg_inf))  # Check duration.
+        mix_neg_inf_samples = mix_neg_inf.get_array_of_samples()
+        self.assertTrue(  # Check samples.
+            all([x == y for x, y in zip(noise_samples, mix_neg_inf_samples)]))
+
+        # Test target SNR 0.0 (different data expected).
+        mix_0 = signal_processing.SignalProcessingUtils.MixSignals(
+            signal, noise, 0.0)
+        self.assertTrue(len(signal), len(mix_0))  # Check duration.
+        self.assertTrue(len(noise), len(mix_0))
+        mix_0_samples = mix_0.get_array_of_samples()
+        self.assertTrue(
+            any([x != y for x, y in zip(signal_samples, mix_0_samples)]))
+        self.assertTrue(
+            any([x != y for x, y in zip(noise_samples, mix_0_samples)]))
+
+        # Test target SNR +Inf (signal expected).
+        mix_pos_inf = signal_processing.SignalProcessingUtils.MixSignals(
+            signal, noise, np.Inf)
+        self.assertTrue(len(signal), len(mix_pos_inf))  # Check duration.
+        mix_pos_inf_samples = mix_pos_inf.get_array_of_samples()
+        self.assertTrue(  # Check samples.
+            all([x == y for x, y in zip(signal_samples, mix_pos_inf_samples)]))
+
+    def testMixSignalsMinInfPower(self):
+        silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000)
+        signal = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            silence)
+
+        with self.assertRaises(exceptions.SignalProcessingException):
+            _ = signal_processing.SignalProcessingUtils.MixSignals(
+                signal, silence, 0.0)
+
+        with self.assertRaises(exceptions.SignalProcessingException):
+            _ = signal_processing.SignalProcessingUtils.MixSignals(
+                silence, signal, 0.0)
+
+    def testMixSignalNoiseDifferentLengths(self):
+        # Test signals.
+        shorter = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            pydub.AudioSegment.silent(duration=1000, frame_rate=8000))
+        longer = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            pydub.AudioSegment.silent(duration=2000, frame_rate=8000))
+
+        # When the signal is shorter than the noise, the mix length always equals
+        # that of the signal regardless of whether padding is applied.
+        # No noise padding, length of signal less than that of noise.
+        mix = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=shorter,
+            noise=longer,
+            pad_noise=signal_processing.SignalProcessingUtils.MixPadding.
+            NO_PADDING)
+        self.assertEqual(len(shorter), len(mix))
+        # With noise padding, length of signal less than that of noise.
+        mix = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=shorter,
+            noise=longer,
+            pad_noise=signal_processing.SignalProcessingUtils.MixPadding.
+            ZERO_PADDING)
+        self.assertEqual(len(shorter), len(mix))
+
+        # When the signal is longer than the noise, the mix length depends on
+        # whether padding is applied.
+        # No noise padding, length of signal greater than that of noise.
+        mix = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=longer,
+            noise=shorter,
+            pad_noise=signal_processing.SignalProcessingUtils.MixPadding.
+            NO_PADDING)
+        self.assertEqual(len(shorter), len(mix))
+        # With noise padding, length of signal greater than that of noise.
+        mix = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=longer,
+            noise=shorter,
+            pad_noise=signal_processing.SignalProcessingUtils.MixPadding.
+            ZERO_PADDING)
+        self.assertEqual(len(longer), len(mix))
+
+    def testMixSignalNoisePaddingTypes(self):
+        # Test signals.
+        shorter = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            pydub.AudioSegment.silent(duration=1000, frame_rate=8000))
+        longer = signal_processing.SignalProcessingUtils.GeneratePureTone(
+            pydub.AudioSegment.silent(duration=2000, frame_rate=8000), 440.0)
+
+        # Zero padding: expect pure tone only in 1-2s.
+        mix_zero_pad = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=longer,
+            noise=shorter,
+            target_snr=-6,
+            pad_noise=signal_processing.SignalProcessingUtils.MixPadding.
+            ZERO_PADDING)
+
+        # Loop: expect pure tone plus noise in 1-2s.
+        mix_loop = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=longer,
+            noise=shorter,
+            target_snr=-6,
+            pad_noise=signal_processing.SignalProcessingUtils.MixPadding.LOOP)
+
+        def Energy(signal):
+            samples = signal_processing.SignalProcessingUtils.AudioSegmentToRawData(
+                signal).astype(np.float32)
+            return np.sum(samples * samples)
+
+        e_mix_zero_pad = Energy(mix_zero_pad[-1000:])
+        e_mix_loop = Energy(mix_loop[-1000:])
+        self.assertLess(0, e_mix_zero_pad)
+        self.assertLess(e_mix_zero_pad, e_mix_loop)
+
+    def testMixSignalSnr(self):
+        # Test signals.
+        tone_low = signal_processing.SignalProcessingUtils.GeneratePureTone(
+            pydub.AudioSegment.silent(duration=64, frame_rate=8000), 250.0)
+        tone_high = signal_processing.SignalProcessingUtils.GeneratePureTone(
+            pydub.AudioSegment.silent(duration=64, frame_rate=8000), 3000.0)
+
+        def ToneAmplitudes(mix):
+            """Returns the amplitude of the coefficients #16 and #192, which
+         correspond to the tones at 250 and 3k Hz respectively."""
+            mix_fft = np.absolute(
+                signal_processing.SignalProcessingUtils.Fft(mix))
+            return mix_fft[16], mix_fft[192]
+
+        mix = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=tone_low, noise=tone_high, target_snr=-6)
+        ampl_low, ampl_high = ToneAmplitudes(mix)
+        self.assertLess(ampl_low, ampl_high)
+
+        mix = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=tone_high, noise=tone_low, target_snr=-6)
+        ampl_low, ampl_high = ToneAmplitudes(mix)
+        self.assertLess(ampl_high, ampl_low)
+
+        mix = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=tone_low, noise=tone_high, target_snr=6)
+        ampl_low, ampl_high = ToneAmplitudes(mix)
+        self.assertLess(ampl_high, ampl_low)
+
+        mix = signal_processing.SignalProcessingUtils.MixSignals(
+            signal=tone_high, noise=tone_low, target_snr=6)
+        ampl_low, ampl_high = ToneAmplitudes(mix)
+        self.assertLess(ampl_low, ampl_high)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py
new file mode 100644
index 0000000000..69b3a1624e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation.py
@@ -0,0 +1,446 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""APM module simulator.
+"""
+
+import logging
+import os
+
+from . import annotations
+from . import data_access
+from . import echo_path_simulation
+from . import echo_path_simulation_factory
+from . import eval_scores
+from . import exceptions
+from . import input_mixer
+from . import input_signal_creator
+from . import signal_processing
+from . import test_data_generation
+
+
+class ApmModuleSimulator(object):
+    """Audio processing module (APM) simulator class.
+  """
+
+    _TEST_DATA_GENERATOR_CLASSES = (
+        test_data_generation.TestDataGenerator.REGISTERED_CLASSES)
+    _EVAL_SCORE_WORKER_CLASSES = eval_scores.EvaluationScore.REGISTERED_CLASSES
+
+    _PREFIX_APM_CONFIG = 'apmcfg-'
+    _PREFIX_CAPTURE = 'capture-'
+    _PREFIX_RENDER = 'render-'
+    _PREFIX_ECHO_SIMULATOR = 'echosim-'
+    _PREFIX_TEST_DATA_GEN = 'datagen-'
+    _PREFIX_TEST_DATA_GEN_PARAMS = 'datagen_params-'
+    _PREFIX_SCORE = 'score-'
+
+    def __init__(self,
+                 test_data_generator_factory,
+                 evaluation_score_factory,
+                 ap_wrapper,
+                 evaluator,
+                 external_vads=None):
+        if external_vads is None:
+            external_vads = {}
+        self._test_data_generator_factory = test_data_generator_factory
+        self._evaluation_score_factory = evaluation_score_factory
+        self._audioproc_wrapper = ap_wrapper
+        self._evaluator = evaluator
+        self._annotator = annotations.AudioAnnotationsExtractor(
+            annotations.AudioAnnotationsExtractor.VadType.ENERGY_THRESHOLD
+            | annotations.AudioAnnotationsExtractor.VadType.WEBRTC_COMMON_AUDIO
+            | annotations.AudioAnnotationsExtractor.VadType.WEBRTC_APM,
+            external_vads)
+
+        # Init.
+        self._test_data_generator_factory.SetOutputDirectoryPrefix(
+            self._PREFIX_TEST_DATA_GEN_PARAMS)
+        self._evaluation_score_factory.SetScoreFilenamePrefix(
+            self._PREFIX_SCORE)
+
+        # Properties for each run.
+        self._base_output_path = None
+        self._output_cache_path = None
+        self._test_data_generators = None
+        self._evaluation_score_workers = None
+        self._config_filepaths = None
+        self._capture_input_filepaths = None
+        self._render_input_filepaths = None
+        self._echo_path_simulator_class = None
+
+    @classmethod
+    def GetPrefixApmConfig(cls):
+        return cls._PREFIX_APM_CONFIG
+
+    @classmethod
+    def GetPrefixCapture(cls):
+        return cls._PREFIX_CAPTURE
+
+    @classmethod
+    def GetPrefixRender(cls):
+        return cls._PREFIX_RENDER
+
+    @classmethod
+    def GetPrefixEchoSimulator(cls):
+        return cls._PREFIX_ECHO_SIMULATOR
+
+    @classmethod
+    def GetPrefixTestDataGenerator(cls):
+        return cls._PREFIX_TEST_DATA_GEN
+
+    @classmethod
+    def GetPrefixTestDataGeneratorParameters(cls):
+        return cls._PREFIX_TEST_DATA_GEN_PARAMS
+
+    @classmethod
+    def GetPrefixScore(cls):
+        return cls._PREFIX_SCORE
+
+    def Run(self,
+            config_filepaths,
+            capture_input_filepaths,
+            test_data_generator_names,
+            eval_score_names,
+            output_dir,
+            render_input_filepaths=None,
+            echo_path_simulator_name=(
+                echo_path_simulation.NoEchoPathSimulator.NAME)):
+        """Runs the APM simulation.
+
+    Initializes paths and required instances, then runs all the simulations.
+    The render input can be optionally added. If added, the number of capture
+    input audio tracks and the number of render input audio tracks have to be
+    equal. The two lists are used to form pairs of capture and render input.
+
+    Args:
+      config_filepaths: set of APM configuration files to test.
+      capture_input_filepaths: set of capture input audio track files to test.
+      test_data_generator_names: set of test data generator names to test.
+      eval_score_names: set of evaluation score names to test.
+      output_dir: base path to the output directory for wav files and outcomes.
+      render_input_filepaths: set of render input audio track files to test.
+      echo_path_simulator_name: name of the echo path simulator to use when
+                                render input is provided.
+    """
+        assert render_input_filepaths is None or (
+            len(capture_input_filepaths) == len(render_input_filepaths)), (
+                'render input set size not matching input set size')
+        assert render_input_filepaths is None or echo_path_simulator_name in (
+            echo_path_simulation.EchoPathSimulator.REGISTERED_CLASSES), (
+                'invalid echo path simulator')
+        self._base_output_path = os.path.abspath(output_dir)
+
+        # Output path used to cache the data shared across simulations.
+        self._output_cache_path = os.path.join(self._base_output_path,
+                                               '_cache')
+
+        # Instance test data generators.
+        self._test_data_generators = [
+            self._test_data_generator_factory.GetInstance(
+                test_data_generators_class=(
+                    self._TEST_DATA_GENERATOR_CLASSES[name]))
+            for name in (test_data_generator_names)
+        ]
+
+        # Instance evaluation score workers.
+        self._evaluation_score_workers = [
+            self._evaluation_score_factory.GetInstance(
+                evaluation_score_class=self._EVAL_SCORE_WORKER_CLASSES[name])
+            for (name) in eval_score_names
+        ]
+
+        # Set APM configuration file paths.
+        self._config_filepaths = self._CreatePathsCollection(config_filepaths)
+
+        # Set probing signal file paths.
+        if render_input_filepaths is None:
+            # Capture input only.
+            self._capture_input_filepaths = self._CreatePathsCollection(
+                capture_input_filepaths)
+            self._render_input_filepaths = None
+        else:
+            # Set both capture and render input signals.
+            self._SetTestInputSignalFilePaths(capture_input_filepaths,
+                                              render_input_filepaths)
+
+        # Set the echo path simulator class.
+        self._echo_path_simulator_class = (
+            echo_path_simulation.EchoPathSimulator.
+            REGISTERED_CLASSES[echo_path_simulator_name])
+
+        self._SimulateAll()
+
+    def _SimulateAll(self):
+        """Runs all the simulations.
+
+    Iterates over the combinations of APM configurations, probing signals, and
+    test data generators. This method is mainly responsible for the creation of
+    the cache and output directories required in order to call _Simulate().
+    """
+        without_render_input = self._render_input_filepaths is None
+
+        # Try different APM config files.
+        for config_name in self._config_filepaths:
+            config_filepath = self._config_filepaths[config_name]
+
+            # Try different capture-render pairs.
+            for capture_input_name in self._capture_input_filepaths:
+                # Output path for the capture signal annotations.
+                capture_annotations_cache_path = os.path.join(
+                    self._output_cache_path,
+                    self._PREFIX_CAPTURE + capture_input_name)
+                data_access.MakeDirectory(capture_annotations_cache_path)
+
+                # Capture.
+                capture_input_filepath = self._capture_input_filepaths[
+                    capture_input_name]
+                if not os.path.exists(capture_input_filepath):
+                    # If the input signal file does not exist, try to create using the
+                    # available input signal creators.
+                    self._CreateInputSignal(capture_input_filepath)
+                assert os.path.exists(capture_input_filepath)
+                self._ExtractCaptureAnnotations(
+                    capture_input_filepath, capture_annotations_cache_path)
+
+                # Render and simulated echo path (optional).
+                render_input_filepath = None if without_render_input else (
+                    self._render_input_filepaths[capture_input_name])
+                render_input_name = '(none)' if without_render_input else (
+                    self._ExtractFileName(render_input_filepath))
+                echo_path_simulator = (echo_path_simulation_factory.
+                                       EchoPathSimulatorFactory.GetInstance(
+                                           self._echo_path_simulator_class,
+                                           render_input_filepath))
+
+                # Try different test data generators.
+                for test_data_generators in self._test_data_generators:
+                    logging.info(
+                        'APM config preset: <%s>, capture: <%s>, render: <%s>,'
+                        'test data generator: <%s>,  echo simulator: <%s>',
+                        config_name, capture_input_name, render_input_name,
+                        test_data_generators.NAME, echo_path_simulator.NAME)
+
+                    # Output path for the generated test data.
+                    test_data_cache_path = os.path.join(
+                        capture_annotations_cache_path,
+                        self._PREFIX_TEST_DATA_GEN + test_data_generators.NAME)
+                    data_access.MakeDirectory(test_data_cache_path)
+                    logging.debug('test data cache path: <%s>',
+                                  test_data_cache_path)
+
+                    # Output path for the echo simulator and APM input mixer output.
+                    echo_test_data_cache_path = os.path.join(
+                        test_data_cache_path,
+                        'echosim-{}'.format(echo_path_simulator.NAME))
+                    data_access.MakeDirectory(echo_test_data_cache_path)
+                    logging.debug('echo test data cache path: <%s>',
+                                  echo_test_data_cache_path)
+
+                    # Full output path.
+                    output_path = os.path.join(
+                        self._base_output_path,
+                        self._PREFIX_APM_CONFIG + config_name,
+                        self._PREFIX_CAPTURE + capture_input_name,
+                        self._PREFIX_RENDER + render_input_name,
+                        self._PREFIX_ECHO_SIMULATOR + echo_path_simulator.NAME,
+                        self._PREFIX_TEST_DATA_GEN + test_data_generators.NAME)
+                    data_access.MakeDirectory(output_path)
+                    logging.debug('output path: <%s>', output_path)
+
+                    self._Simulate(test_data_generators,
+                                   capture_input_filepath,
+                                   render_input_filepath, test_data_cache_path,
+                                   echo_test_data_cache_path, output_path,
+                                   config_filepath, echo_path_simulator)
+
+    @staticmethod
+    def _CreateInputSignal(input_signal_filepath):
+        """Creates a missing input signal file.
+
+    The file name is parsed to extract input signal creator and params. If a
+    creator is matched and the parameters are valid, a new signal is generated
+    and written in `input_signal_filepath`.
+
+    Args:
+      input_signal_filepath: Path to the input signal audio file to write.
+
+    Raises:
+      InputSignalCreatorException
+    """
+        filename = os.path.splitext(
+            os.path.split(input_signal_filepath)[-1])[0]
+        filename_parts = filename.split('-')
+
+        if len(filename_parts) < 2:
+            raise exceptions.InputSignalCreatorException(
+                'Cannot parse input signal file name')
+
+        signal, metadata = input_signal_creator.InputSignalCreator.Create(
+            filename_parts[0], filename_parts[1].split('_'))
+
+        signal_processing.SignalProcessingUtils.SaveWav(
+            input_signal_filepath, signal)
+        data_access.Metadata.SaveFileMetadata(input_signal_filepath, metadata)
+
+    def _ExtractCaptureAnnotations(self,
+                                   input_filepath,
+                                   output_path,
+                                   annotation_name=""):
+        self._annotator.Extract(input_filepath)
+        self._annotator.Save(output_path, annotation_name)
+
+    def _Simulate(self, test_data_generators, clean_capture_input_filepath,
+                  render_input_filepath, test_data_cache_path,
+                  echo_test_data_cache_path, output_path, config_filepath,
+                  echo_path_simulator):
+        """Runs a single set of simulation.
+
+    Simulates a given combination of APM configuration, probing signal, and
+    test data generator. It iterates over the test data generator
+    internal configurations.
+
+    Args:
+      test_data_generators: TestDataGenerator instance.
+      clean_capture_input_filepath: capture input audio track file to be
+                                    processed by a test data generator and
+                                    not affected by echo.
+      render_input_filepath: render input audio track file to test.
+      test_data_cache_path: path for the generated test audio track files.
+      echo_test_data_cache_path: path for the echo simulator.
+      output_path: base output path for the test data generator.
+      config_filepath: APM configuration file to test.
+      echo_path_simulator: EchoPathSimulator instance.
+    """
+        # Generate pairs of noisy input and reference signal files.
+        test_data_generators.Generate(
+            input_signal_filepath=clean_capture_input_filepath,
+            test_data_cache_path=test_data_cache_path,
+            base_output_path=output_path)
+
+        # Extract metadata linked to the clean input file (if any).
+        apm_input_metadata = None
+        try:
+            apm_input_metadata = data_access.Metadata.LoadFileMetadata(
+                clean_capture_input_filepath)
+        except IOError as e:
+            apm_input_metadata = {}
+        apm_input_metadata['test_data_gen_name'] = test_data_generators.NAME
+        apm_input_metadata['test_data_gen_config'] = None
+
+        # For each test data pair, simulate a call and evaluate.
+        for config_name in test_data_generators.config_names:
+            logging.info(' - test data generator config: <%s>', config_name)
+            apm_input_metadata['test_data_gen_config'] = config_name
+
+            # Paths to the test data generator output.
+            # Note that the reference signal does not depend on the render input
+            # which is optional.
+            noisy_capture_input_filepath = (
+                test_data_generators.noisy_signal_filepaths[config_name])
+            reference_signal_filepath = (
+                test_data_generators.reference_signal_filepaths[config_name])
+
+            # Output path for the evaluation (e.g., APM output file).
+            evaluation_output_path = test_data_generators.apm_output_paths[
+                config_name]
+
+            # Paths to the APM input signals.
+            echo_path_filepath = echo_path_simulator.Simulate(
+                echo_test_data_cache_path)
+            apm_input_filepath = input_mixer.ApmInputMixer.Mix(
+                echo_test_data_cache_path, noisy_capture_input_filepath,
+                echo_path_filepath)
+
+            # Extract annotations for the APM input mix.
+            apm_input_basepath, apm_input_filename = os.path.split(
+                apm_input_filepath)
+            self._ExtractCaptureAnnotations(
+                apm_input_filepath, apm_input_basepath,
+                os.path.splitext(apm_input_filename)[0] + '-')
+
+            # Simulate a call using APM.
+            self._audioproc_wrapper.Run(
+                config_filepath=config_filepath,
+                capture_input_filepath=apm_input_filepath,
+                render_input_filepath=render_input_filepath,
+                output_path=evaluation_output_path)
+
+            try:
+                # Evaluate.
+                self._evaluator.Run(
+                    evaluation_score_workers=self._evaluation_score_workers,
+                    apm_input_metadata=apm_input_metadata,
+                    apm_output_filepath=self._audioproc_wrapper.
+                    output_filepath,
+                    reference_input_filepath=reference_signal_filepath,
+                    render_input_filepath=render_input_filepath,
+                    output_path=evaluation_output_path,
+                )
+
+                # Save simulation metadata.
+                data_access.Metadata.SaveAudioTestDataPaths(
+                    output_path=evaluation_output_path,
+                    clean_capture_input_filepath=clean_capture_input_filepath,
+                    echo_free_capture_filepath=noisy_capture_input_filepath,
+                    echo_filepath=echo_path_filepath,
+                    render_filepath=render_input_filepath,
+                    capture_filepath=apm_input_filepath,
+                    apm_output_filepath=self._audioproc_wrapper.
+                    output_filepath,
+                    apm_reference_filepath=reference_signal_filepath,
+                    apm_config_filepath=config_filepath,
+                )
+            except exceptions.EvaluationScoreException as e:
+                logging.warning('the evaluation failed: %s', e.message)
+                continue
+
+    def _SetTestInputSignalFilePaths(self, capture_input_filepaths,
+                                     render_input_filepaths):
+        """Sets input and render input file paths collections.
+
+    Pairs the input and render input files by storing the file paths into two
+    collections. The key is the file name of the input file.
+
+    Args:
+      capture_input_filepaths: list of file paths.
+      render_input_filepaths: list of file paths.
+    """
+        self._capture_input_filepaths = {}
+        self._render_input_filepaths = {}
+        assert len(capture_input_filepaths) == len(render_input_filepaths)
+        for capture_input_filepath, render_input_filepath in zip(
+                capture_input_filepaths, render_input_filepaths):
+            name = self._ExtractFileName(capture_input_filepath)
+            self._capture_input_filepaths[name] = os.path.abspath(
+                capture_input_filepath)
+            self._render_input_filepaths[name] = os.path.abspath(
+                render_input_filepath)
+
+    @classmethod
+    def _CreatePathsCollection(cls, filepaths):
+        """Creates a collection of file paths.
+
+    Given a list of file paths, makes a collection with one item for each file
+    path. The value is absolute path, the key is the file name without
+    extenstion.
+
+    Args:
+      filepaths: list of file paths.
+
+    Returns:
+      A dict.
+    """
+        filepaths_collection = {}
+        for filepath in filepaths:
+            name = cls._ExtractFileName(filepath)
+            filepaths_collection[name] = os.path.abspath(filepath)
+        return filepaths_collection
+
+    @classmethod
+    def _ExtractFileName(cls, filepath):
+        return os.path.splitext(os.path.split(filepath)[-1])[0]
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation_unittest.py
new file mode 100644
index 0000000000..78ca17f589
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/simulation_unittest.py
@@ -0,0 +1,203 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Unit tests for the simulation module.
+"""
+
+import logging
+import os
+import shutil
+import tempfile
+import unittest
+
+import mock
+import pydub
+
+from . import audioproc_wrapper
+from . import eval_scores_factory
+from . import evaluation
+from . import external_vad
+from . import signal_processing
+from . import simulation
+from . import test_data_generation_factory
+
+
+class TestApmModuleSimulator(unittest.TestCase):
+    """Unit tests for the ApmModuleSimulator class.
+  """
+
+    def setUp(self):
+        """Create temporary folders and fake audio track."""
+        self._output_path = tempfile.mkdtemp()
+        self._tmp_path = tempfile.mkdtemp()
+
+        silence = pydub.AudioSegment.silent(duration=1000, frame_rate=48000)
+        fake_signal = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            silence)
+        self._fake_audio_track_path = os.path.join(self._output_path,
+                                                   'fake.wav')
+        signal_processing.SignalProcessingUtils.SaveWav(
+            self._fake_audio_track_path, fake_signal)
+
+    def tearDown(self):
+        """Recursively delete temporary folders."""
+        shutil.rmtree(self._output_path)
+        shutil.rmtree(self._tmp_path)
+
+    def testSimulation(self):
+        # Instance dependencies to mock and inject.
+        ap_wrapper = audioproc_wrapper.AudioProcWrapper(
+            audioproc_wrapper.AudioProcWrapper.DEFAULT_APM_SIMULATOR_BIN_PATH)
+        evaluator = evaluation.ApmModuleEvaluator()
+        ap_wrapper.Run = mock.MagicMock(name='Run')
+        evaluator.Run = mock.MagicMock(name='Run')
+
+        # Instance non-mocked dependencies.
+        test_data_generator_factory = (
+            test_data_generation_factory.TestDataGeneratorFactory(
+                aechen_ir_database_path='',
+                noise_tracks_path='',
+                copy_with_identity=False))
+        evaluation_score_factory = eval_scores_factory.EvaluationScoreWorkerFactory(
+            polqa_tool_bin_path=os.path.join(os.path.dirname(__file__),
+                                             'fake_polqa'),
+            echo_metric_tool_bin_path=None)
+
+        # Instance simulator.
+        simulator = simulation.ApmModuleSimulator(
+            test_data_generator_factory=test_data_generator_factory,
+            evaluation_score_factory=evaluation_score_factory,
+            ap_wrapper=ap_wrapper,
+            evaluator=evaluator,
+            external_vads={
+                'fake':
+                external_vad.ExternalVad(
+                    os.path.join(os.path.dirname(__file__),
+                                 'fake_external_vad.py'), 'fake')
+            })
+
+        # What to simulate.
+        config_files = ['apm_configs/default.json']
+        input_files = [self._fake_audio_track_path]
+        test_data_generators = ['identity', 'white_noise']
+        eval_scores = ['audio_level_mean', 'polqa']
+
+        # Run all simulations.
+        simulator.Run(config_filepaths=config_files,
+                      capture_input_filepaths=input_files,
+                      test_data_generator_names=test_data_generators,
+                      eval_score_names=eval_scores,
+                      output_dir=self._output_path)
+
+        # Check.
+        # TODO(alessiob): Once the TestDataGenerator classes can be configured by
+        # the client code (e.g., number of SNR pairs for the white noise test data
+        # generator), the exact number of calls to ap_wrapper.Run and evaluator.Run
+        # is known; use that with assertEqual.
+        min_number_of_simulations = len(config_files) * len(input_files) * len(
+            test_data_generators)
+        self.assertGreaterEqual(len(ap_wrapper.Run.call_args_list),
+                                min_number_of_simulations)
+        self.assertGreaterEqual(len(evaluator.Run.call_args_list),
+                                min_number_of_simulations)
+
+    def testInputSignalCreation(self):
+        # Instance simulator.
+        simulator = simulation.ApmModuleSimulator(
+            test_data_generator_factory=(
+                test_data_generation_factory.TestDataGeneratorFactory(
+                    aechen_ir_database_path='',
+                    noise_tracks_path='',
+                    copy_with_identity=False)),
+            evaluation_score_factory=(
+                eval_scores_factory.EvaluationScoreWorkerFactory(
+                    polqa_tool_bin_path=os.path.join(os.path.dirname(__file__),
+                                                     'fake_polqa'),
+                    echo_metric_tool_bin_path=None)),
+            ap_wrapper=audioproc_wrapper.AudioProcWrapper(
+                audioproc_wrapper.AudioProcWrapper.
+                DEFAULT_APM_SIMULATOR_BIN_PATH),
+            evaluator=evaluation.ApmModuleEvaluator())
+
+        # Inexistent input files to be silently created.
+        input_files = [
+            os.path.join(self._tmp_path, 'pure_tone-440_1000.wav'),
+            os.path.join(self._tmp_path, 'pure_tone-1000_500.wav'),
+        ]
+        self.assertFalse(
+            any([os.path.exists(input_file) for input_file in (input_files)]))
+
+        # The input files are created during the simulation.
+        simulator.Run(config_filepaths=['apm_configs/default.json'],
+                      capture_input_filepaths=input_files,
+                      test_data_generator_names=['identity'],
+                      eval_score_names=['audio_level_peak'],
+                      output_dir=self._output_path)
+        self.assertTrue(
+            all([os.path.exists(input_file) for input_file in (input_files)]))
+
+    def testPureToneGenerationWithTotalHarmonicDistorsion(self):
+        logging.warning = mock.MagicMock(name='warning')
+
+        # Instance simulator.
+        simulator = simulation.ApmModuleSimulator(
+            test_data_generator_factory=(
+                test_data_generation_factory.TestDataGeneratorFactory(
+                    aechen_ir_database_path='',
+                    noise_tracks_path='',
+                    copy_with_identity=False)),
+            evaluation_score_factory=(
+                eval_scores_factory.EvaluationScoreWorkerFactory(
+                    polqa_tool_bin_path=os.path.join(os.path.dirname(__file__),
+                                                     'fake_polqa'),
+                    echo_metric_tool_bin_path=None)),
+            ap_wrapper=audioproc_wrapper.AudioProcWrapper(
+                audioproc_wrapper.AudioProcWrapper.
+                DEFAULT_APM_SIMULATOR_BIN_PATH),
+            evaluator=evaluation.ApmModuleEvaluator())
+
+        # What to simulate.
+        config_files = ['apm_configs/default.json']
+        input_files = [os.path.join(self._tmp_path, 'pure_tone-440_1000.wav')]
+        eval_scores = ['thd']
+
+        # Should work.
+        simulator.Run(config_filepaths=config_files,
+                      capture_input_filepaths=input_files,
+                      test_data_generator_names=['identity'],
+                      eval_score_names=eval_scores,
+                      output_dir=self._output_path)
+        self.assertFalse(logging.warning.called)
+
+        # Warning expected.
+        simulator.Run(
+            config_filepaths=config_files,
+            capture_input_filepaths=input_files,
+            test_data_generator_names=['white_noise'],  # Not allowed with THD.
+            eval_score_names=eval_scores,
+            output_dir=self._output_path)
+        logging.warning.assert_called_with('the evaluation failed: %s', (
+            'The THD score cannot be used with any test data generator other than '
+            '"identity"'))
+
+    #   # Init.
+    #   generator = test_data_generation.IdentityTestDataGenerator('tmp')
+    #   input_signal_filepath = os.path.join(
+    #       self._test_data_cache_path, 'pure_tone-440_1000.wav')
+
+    #   # Check that the input signal is generated.
+    #   self.assertFalse(os.path.exists(input_signal_filepath))
+    #   generator.Generate(
+    #       input_signal_filepath=input_signal_filepath,
+    #       test_data_cache_path=self._test_data_cache_path,
+    #       base_output_path=self._base_output_path)
+    #   self.assertTrue(os.path.exists(input_signal_filepath))
+
+    #   # Check input signal properties.
+    #   input_signal = signal_processing.SignalProcessingUtils.LoadWav(
+    #       input_signal_filepath)
+    #   self.assertEqual(1000, len(input_signal))
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
new file mode 100644
index 0000000000..1f24d9d370
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
@@ -0,0 +1,127 @@
+// Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+//
+// Use of this source code is governed by a BSD-style license
+// that can be found in the LICENSE file in the root of the source
+// tree. An additional intellectual property rights grant can be found
+// in the file PATENTS.  All contributing project authors may
+// be found in the AUTHORS file in the root of the source tree.
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <fstream>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "common_audio/include/audio_util.h"
+#include "common_audio/wav_file.h"
+#include "rtc_base/logging.h"
+
+ABSL_FLAG(std::string, i, "", "Input wav file");
+ABSL_FLAG(std::string, oc, "", "Config output file");
+ABSL_FLAG(std::string, ol, "", "Levels output file");
+ABSL_FLAG(float, a, 5.f, "Attack (ms)");
+ABSL_FLAG(float, d, 20.f, "Decay (ms)");
+ABSL_FLAG(int, f, 10, "Frame length (ms)");
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr int kMaxSampleRate = 48000;
+constexpr uint8_t kMaxFrameLenMs = 30;
+constexpr size_t kMaxFrameLen = kMaxFrameLenMs * kMaxSampleRate / 1000;
+
+const double kOneDbReduction = DbToRatio(-1.0);
+
+int main(int argc, char* argv[]) {
+  absl::ParseCommandLine(argc, argv);
+  // Check parameters.
+  if (absl::GetFlag(FLAGS_f) < 1 || absl::GetFlag(FLAGS_f) > kMaxFrameLenMs) {
+    RTC_LOG(LS_ERROR) << "Invalid frame length (min: 1, max: " << kMaxFrameLenMs
+                      << ")";
+    return 1;
+  }
+  if (absl::GetFlag(FLAGS_a) < 0 || absl::GetFlag(FLAGS_d) < 0) {
+    RTC_LOG(LS_ERROR) << "Attack and decay must be non-negative";
+    return 1;
+  }
+
+  // Open wav input file and check properties.
+  const std::string input_file = absl::GetFlag(FLAGS_i);
+  const std::string config_output_file = absl::GetFlag(FLAGS_oc);
+  const std::string levels_output_file = absl::GetFlag(FLAGS_ol);
+  WavReader wav_reader(input_file);
+  if (wav_reader.num_channels() != 1) {
+    RTC_LOG(LS_ERROR) << "Only mono wav files supported";
+    return 1;
+  }
+  if (wav_reader.sample_rate() > kMaxSampleRate) {
+    RTC_LOG(LS_ERROR) << "Beyond maximum sample rate (" << kMaxSampleRate
+                      << ")";
+    return 1;
+  }
+
+  // Map from milliseconds to samples.
+  const size_t audio_frame_length = rtc::CheckedDivExact(
+      absl::GetFlag(FLAGS_f) * wav_reader.sample_rate(), 1000);
+  auto time_const = [](double c) {
+    return std::pow(kOneDbReduction, absl::GetFlag(FLAGS_f) / c);
+  };
+  const float attack =
+      absl::GetFlag(FLAGS_a) == 0.0 ? 0.0 : time_const(absl::GetFlag(FLAGS_a));
+  const float decay =
+      absl::GetFlag(FLAGS_d) == 0.0 ? 0.0 : time_const(absl::GetFlag(FLAGS_d));
+
+  // Write config to file.
+  std::ofstream out_config(config_output_file);
+  out_config << "{"
+                "'frame_len_ms': "
+             << absl::GetFlag(FLAGS_f)
+             << ", "
+                "'attack_ms': "
+             << absl::GetFlag(FLAGS_a)
+             << ", "
+                "'decay_ms': "
+             << absl::GetFlag(FLAGS_d) << "}\n";
+  out_config.close();
+
+  // Measure level frame-by-frame.
+  std::ofstream out_levels(levels_output_file, std::ofstream::binary);
+  std::array<int16_t, kMaxFrameLen> samples;
+  float level_prev = 0.f;
+  while (true) {
+    // Process frame.
+    const auto read_samples =
+        wav_reader.ReadSamples(audio_frame_length, samples.data());
+    if (read_samples < audio_frame_length)
+      break;  // EOF.
+
+    // Frame peak level.
+    std::transform(samples.begin(), samples.begin() + audio_frame_length,
+                   samples.begin(), [](int16_t s) { return std::abs(s); });
+    const int16_t peak_level = *std::max_element(
+        samples.cbegin(), samples.cbegin() + audio_frame_length);
+    const float level_curr = static_cast<float>(peak_level) / 32768.f;
+
+    // Temporal smoothing.
+    auto smooth = [&level_prev, &level_curr](float c) {
+      return (1.0 - c) * level_curr + c * level_prev;
+    };
+    level_prev = smooth(level_curr > level_prev ? attack : decay);
+
+    // Write output.
+    out_levels.write(reinterpret_cast<const char*>(&level_prev), sizeof(float));
+  }
+  out_levels.close();
+
+  return 0;
+}
+
+}  // namespace
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::main(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation.py
new file mode 100644
index 0000000000..7e86faccec
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation.py
@@ -0,0 +1,526 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Test data generators producing signals pairs intended to be used to
+test the APM module. Each pair consists of a noisy input and a reference signal.
+The former is used as APM input and it is generated by adding noise to a
+clean audio track. The reference is the expected APM output.
+
+Throughout this file, the following naming convention is used:
+  - input signal: the clean signal (e.g., speech),
+  - noise signal: the noise to be summed up to the input signal (e.g., white
+    noise, Gaussian noise),
+  - noisy signal: input + noise.
+The noise signal may or may not be a function of the clean signal. For
+instance, white noise is independently generated, whereas reverberation is
+obtained by convolving the input signal with an impulse response.
+"""
+
+import logging
+import os
+import shutil
+import sys
+
+try:
+    import scipy.io
+except ImportError:
+    logging.critical('Cannot import the third-party Python package scipy')
+    sys.exit(1)
+
+from . import data_access
+from . import exceptions
+from . import signal_processing
+
+
+class TestDataGenerator(object):
+    """Abstract class responsible for the generation of noisy signals.
+
+  Given a clean signal, it generates two streams named noisy signal and
+  reference. The former is the clean signal deteriorated by the noise source,
+  the latter goes through the same deterioration process, but more "gently".
+  Noisy signal and reference are produced so that the reference is the signal
+  expected at the output of the APM module when the latter is fed with the noisy
+  signal.
+
+  An test data generator generates one or more pairs.
+  """
+
+    NAME = None
+    REGISTERED_CLASSES = {}
+
+    def __init__(self, output_directory_prefix):
+        self._output_directory_prefix = output_directory_prefix
+        # Init dictionaries with one entry for each test data generator
+        # configuration (e.g., different SNRs).
+        # Noisy audio track files (stored separately in a cache folder).
+        self._noisy_signal_filepaths = None
+        # Path to be used for the APM simulation output files.
+        self._apm_output_paths = None
+        # Reference audio track files (stored separately in a cache folder).
+        self._reference_signal_filepaths = None
+        self.Clear()
+
+    @classmethod
+    def RegisterClass(cls, class_to_register):
+        """Registers a TestDataGenerator implementation.
+
+    Decorator to automatically register the classes that extend
+    TestDataGenerator.
+    Example usage:
+
+    @TestDataGenerator.RegisterClass
+    class IdentityGenerator(TestDataGenerator):
+      pass
+    """
+        cls.REGISTERED_CLASSES[class_to_register.NAME] = class_to_register
+        return class_to_register
+
+    @property
+    def config_names(self):
+        return self._noisy_signal_filepaths.keys()
+
+    @property
+    def noisy_signal_filepaths(self):
+        return self._noisy_signal_filepaths
+
+    @property
+    def apm_output_paths(self):
+        return self._apm_output_paths
+
+    @property
+    def reference_signal_filepaths(self):
+        return self._reference_signal_filepaths
+
+    def Generate(self, input_signal_filepath, test_data_cache_path,
+                 base_output_path):
+        """Generates a set of noisy input and reference audiotrack file pairs.
+
+    This method initializes an empty set of pairs and calls the _Generate()
+    method implemented in a concrete class.
+
+    Args:
+      input_signal_filepath: path to the clean input audio track file.
+      test_data_cache_path: path to the cache of the generated audio track
+                            files.
+      base_output_path: base path where output is written.
+    """
+        self.Clear()
+        self._Generate(input_signal_filepath, test_data_cache_path,
+                       base_output_path)
+
+    def Clear(self):
+        """Clears the generated output path dictionaries.
+    """
+        self._noisy_signal_filepaths = {}
+        self._apm_output_paths = {}
+        self._reference_signal_filepaths = {}
+
+    def _Generate(self, input_signal_filepath, test_data_cache_path,
+                  base_output_path):
+        """Abstract method to be implemented in each concrete class.
+    """
+        raise NotImplementedError()
+
+    def _AddNoiseSnrPairs(self, base_output_path, noisy_mix_filepaths,
+                          snr_value_pairs):
+        """Adds noisy-reference signal pairs.
+
+    Args:
+      base_output_path: noisy tracks base output path.
+      noisy_mix_filepaths: nested dictionary of noisy signal paths organized
+                           by noisy track name and SNR level.
+      snr_value_pairs: list of SNR pairs.
+    """
+        for noise_track_name in noisy_mix_filepaths:
+            for snr_noisy, snr_refence in snr_value_pairs:
+                config_name = '{0}_{1:d}_{2:d}_SNR'.format(
+                    noise_track_name, snr_noisy, snr_refence)
+                output_path = self._MakeDir(base_output_path, config_name)
+                self._AddNoiseReferenceFilesPair(
+                    config_name=config_name,
+                    noisy_signal_filepath=noisy_mix_filepaths[noise_track_name]
+                    [snr_noisy],
+                    reference_signal_filepath=noisy_mix_filepaths[
+                        noise_track_name][snr_refence],
+                    output_path=output_path)
+
+    def _AddNoiseReferenceFilesPair(self, config_name, noisy_signal_filepath,
+                                    reference_signal_filepath, output_path):
+        """Adds one noisy-reference signal pair.
+
+    Args:
+      config_name: name of the APM configuration.
+      noisy_signal_filepath: path to noisy audio track file.
+      reference_signal_filepath: path to reference audio track file.
+      output_path: APM output path.
+    """
+        assert config_name not in self._noisy_signal_filepaths
+        self._noisy_signal_filepaths[config_name] = os.path.abspath(
+            noisy_signal_filepath)
+        self._apm_output_paths[config_name] = os.path.abspath(output_path)
+        self._reference_signal_filepaths[config_name] = os.path.abspath(
+            reference_signal_filepath)
+
+    def _MakeDir(self, base_output_path, test_data_generator_config_name):
+        output_path = os.path.join(
+            base_output_path,
+            self._output_directory_prefix + test_data_generator_config_name)
+        data_access.MakeDirectory(output_path)
+        return output_path
+
+
+@TestDataGenerator.RegisterClass
+class IdentityTestDataGenerator(TestDataGenerator):
+    """Generator that adds no noise.
+
+  Both the noisy and the reference signals are the input signal.
+  """
+
+    NAME = 'identity'
+
+    def __init__(self, output_directory_prefix, copy_with_identity):
+        TestDataGenerator.__init__(self, output_directory_prefix)
+        self._copy_with_identity = copy_with_identity
+
+    @property
+    def copy_with_identity(self):
+        return self._copy_with_identity
+
+    def _Generate(self, input_signal_filepath, test_data_cache_path,
+                  base_output_path):
+        config_name = 'default'
+        output_path = self._MakeDir(base_output_path, config_name)
+
+        if self._copy_with_identity:
+            input_signal_filepath_new = os.path.join(
+                test_data_cache_path,
+                os.path.split(input_signal_filepath)[1])
+            logging.info('copying ' + input_signal_filepath + ' to ' +
+                         (input_signal_filepath_new))
+            shutil.copy(input_signal_filepath, input_signal_filepath_new)
+            input_signal_filepath = input_signal_filepath_new
+
+        self._AddNoiseReferenceFilesPair(
+            config_name=config_name,
+            noisy_signal_filepath=input_signal_filepath,
+            reference_signal_filepath=input_signal_filepath,
+            output_path=output_path)
+
+
+@TestDataGenerator.RegisterClass
+class WhiteNoiseTestDataGenerator(TestDataGenerator):
+    """Generator that adds white noise.
+  """
+
+    NAME = 'white_noise'
+
+    # Each pair indicates the clean vs. noisy and reference vs. noisy SNRs.
+    # The reference (second value of each pair) always has a lower amount of noise
+    # - i.e., the SNR is 10 dB higher.
+    _SNR_VALUE_PAIRS = [
+        [20, 30],  # Smallest noise.
+        [10, 20],
+        [5, 15],
+        [0, 10],  # Largest noise.
+    ]
+
+    _NOISY_SIGNAL_FILENAME_TEMPLATE = 'noise_{0:d}_SNR.wav'
+
+    def __init__(self, output_directory_prefix):
+        TestDataGenerator.__init__(self, output_directory_prefix)
+
+    def _Generate(self, input_signal_filepath, test_data_cache_path,
+                  base_output_path):
+        # Load the input signal.
+        input_signal = signal_processing.SignalProcessingUtils.LoadWav(
+            input_signal_filepath)
+
+        # Create the noise track.
+        noise_signal = signal_processing.SignalProcessingUtils.GenerateWhiteNoise(
+            input_signal)
+
+        # Create the noisy mixes (once for each unique SNR value).
+        noisy_mix_filepaths = {}
+        snr_values = set(
+            [snr for pair in self._SNR_VALUE_PAIRS for snr in pair])
+        for snr in snr_values:
+            noisy_signal_filepath = os.path.join(
+                test_data_cache_path,
+                self._NOISY_SIGNAL_FILENAME_TEMPLATE.format(snr))
+
+            # Create and save if not done.
+            if not os.path.exists(noisy_signal_filepath):
+                # Create noisy signal.
+                noisy_signal = signal_processing.SignalProcessingUtils.MixSignals(
+                    input_signal, noise_signal, snr)
+
+                # Save.
+                signal_processing.SignalProcessingUtils.SaveWav(
+                    noisy_signal_filepath, noisy_signal)
+
+            # Add file to the collection of mixes.
+            noisy_mix_filepaths[snr] = noisy_signal_filepath
+
+        # Add all the noisy-reference signal pairs.
+        for snr_noisy, snr_refence in self._SNR_VALUE_PAIRS:
+            config_name = '{0:d}_{1:d}_SNR'.format(snr_noisy, snr_refence)
+            output_path = self._MakeDir(base_output_path, config_name)
+            self._AddNoiseReferenceFilesPair(
+                config_name=config_name,
+                noisy_signal_filepath=noisy_mix_filepaths[snr_noisy],
+                reference_signal_filepath=noisy_mix_filepaths[snr_refence],
+                output_path=output_path)
+
+
+# TODO(alessiob): remove comment when class implemented.
+# @TestDataGenerator.RegisterClass
+class NarrowBandNoiseTestDataGenerator(TestDataGenerator):
+    """Generator that adds narrow-band noise.
+  """
+
+    NAME = 'narrow_band_noise'
+
+    def __init__(self, output_directory_prefix):
+        TestDataGenerator.__init__(self, output_directory_prefix)
+
+    def _Generate(self, input_signal_filepath, test_data_cache_path,
+                  base_output_path):
+        # TODO(alessiob): implement.
+        pass
+
+
+@TestDataGenerator.RegisterClass
+class AdditiveNoiseTestDataGenerator(TestDataGenerator):
+    """Generator that adds noise loops.
+
+  This generator uses all the wav files in a given path (default: noise_tracks/)
+  and mixes them to the clean speech with different target SNRs (hard-coded).
+  """
+
+    NAME = 'additive_noise'
+    _NOISY_SIGNAL_FILENAME_TEMPLATE = '{0}_{1:d}_SNR.wav'
+
+    DEFAULT_NOISE_TRACKS_PATH = os.path.join(os.path.dirname(__file__),
+                                             os.pardir, 'noise_tracks')
+
+    # TODO(alessiob): Make the list of SNR pairs customizable.
+    # Each pair indicates the clean vs. noisy and reference vs. noisy SNRs.
+    # The reference (second value of each pair) always has a lower amount of noise
+    # - i.e., the SNR is 10 dB higher.
+    _SNR_VALUE_PAIRS = [
+        [20, 30],  # Smallest noise.
+        [10, 20],
+        [5, 15],
+        [0, 10],  # Largest noise.
+    ]
+
+    def __init__(self, output_directory_prefix, noise_tracks_path):
+        TestDataGenerator.__init__(self, output_directory_prefix)
+        self._noise_tracks_path = noise_tracks_path
+        self._noise_tracks_file_names = [
+            n for n in os.listdir(self._noise_tracks_path)
+            if n.lower().endswith('.wav')
+        ]
+        if len(self._noise_tracks_file_names) == 0:
+            raise exceptions.InitializationException(
+                'No wav files found in the noise tracks path %s' %
+                (self._noise_tracks_path))
+
+    def _Generate(self, input_signal_filepath, test_data_cache_path,
+                  base_output_path):
+        """Generates test data pairs using environmental noise.
+
+    For each noise track and pair of SNR values, the following two audio tracks
+    are created: the noisy signal and the reference signal. The former is
+    obtained by mixing the (clean) input signal to the corresponding noise
+    track enforcing the target SNR.
+    """
+        # Init.
+        snr_values = set(
+            [snr for pair in self._SNR_VALUE_PAIRS for snr in pair])
+
+        # Load the input signal.
+        input_signal = signal_processing.SignalProcessingUtils.LoadWav(
+            input_signal_filepath)
+
+        noisy_mix_filepaths = {}
+        for noise_track_filename in self._noise_tracks_file_names:
+            # Load the noise track.
+            noise_track_name, _ = os.path.splitext(noise_track_filename)
+            noise_track_filepath = os.path.join(self._noise_tracks_path,
+                                                noise_track_filename)
+            if not os.path.exists(noise_track_filepath):
+                logging.error('cannot find the <%s> noise track',
+                              noise_track_filename)
+                raise exceptions.FileNotFoundError()
+
+            noise_signal = signal_processing.SignalProcessingUtils.LoadWav(
+                noise_track_filepath)
+
+            # Create the noisy mixes (once for each unique SNR value).
+            noisy_mix_filepaths[noise_track_name] = {}
+            for snr in snr_values:
+                noisy_signal_filepath = os.path.join(
+                    test_data_cache_path,
+                    self._NOISY_SIGNAL_FILENAME_TEMPLATE.format(
+                        noise_track_name, snr))
+
+                # Create and save if not done.
+                if not os.path.exists(noisy_signal_filepath):
+                    # Create noisy signal.
+                    noisy_signal = signal_processing.SignalProcessingUtils.MixSignals(
+                        input_signal,
+                        noise_signal,
+                        snr,
+                        pad_noise=signal_processing.SignalProcessingUtils.
+                        MixPadding.LOOP)
+
+                    # Save.
+                    signal_processing.SignalProcessingUtils.SaveWav(
+                        noisy_signal_filepath, noisy_signal)
+
+                # Add file to the collection of mixes.
+                noisy_mix_filepaths[noise_track_name][
+                    snr] = noisy_signal_filepath
+
+        # Add all the noise-SNR pairs.
+        self._AddNoiseSnrPairs(base_output_path, noisy_mix_filepaths,
+                               self._SNR_VALUE_PAIRS)
+
+
+@TestDataGenerator.RegisterClass
+class ReverberationTestDataGenerator(TestDataGenerator):
+    """Generator that adds reverberation noise.
+
+  TODO(alessiob): Make this class more generic since the impulse response can be
+  anything (not just reverberation); call it e.g.,
+  ConvolutionalNoiseTestDataGenerator.
+  """
+
+    NAME = 'reverberation'
+
+    _IMPULSE_RESPONSES = {
+        'lecture': 'air_binaural_lecture_0_0_1.mat',  # Long echo.
+        'booth': 'air_binaural_booth_0_0_1.mat',  # Short echo.
+    }
+    _MAX_IMPULSE_RESPONSE_LENGTH = None
+
+    # Each pair indicates the clean vs. noisy and reference vs. noisy SNRs.
+    # The reference (second value of each pair) always has a lower amount of noise
+    # - i.e., the SNR is 5 dB higher.
+    _SNR_VALUE_PAIRS = [
+        [3, 8],  # Smallest noise.
+        [-3, 2],  # Largest noise.
+    ]
+
+    _NOISE_TRACK_FILENAME_TEMPLATE = '{0}.wav'
+    _NOISY_SIGNAL_FILENAME_TEMPLATE = '{0}_{1:d}_SNR.wav'
+
+    def __init__(self, output_directory_prefix, aechen_ir_database_path):
+        TestDataGenerator.__init__(self, output_directory_prefix)
+        self._aechen_ir_database_path = aechen_ir_database_path
+
+    def _Generate(self, input_signal_filepath, test_data_cache_path,
+                  base_output_path):
+        """Generates test data pairs using reverberation noise.
+
+    For each impulse response, one noise track is created. For each impulse
+    response and pair of SNR values, the following 2 audio tracks are
+    created: the noisy signal and the reference signal. The former is
+    obtained by mixing the (clean) input signal to the corresponding noise
+    track enforcing the target SNR.
+    """
+        # Init.
+        snr_values = set(
+            [snr for pair in self._SNR_VALUE_PAIRS for snr in pair])
+
+        # Load the input signal.
+        input_signal = signal_processing.SignalProcessingUtils.LoadWav(
+            input_signal_filepath)
+
+        noisy_mix_filepaths = {}
+        for impulse_response_name in self._IMPULSE_RESPONSES:
+            noise_track_filename = self._NOISE_TRACK_FILENAME_TEMPLATE.format(
+                impulse_response_name)
+            noise_track_filepath = os.path.join(test_data_cache_path,
+                                                noise_track_filename)
+            noise_signal = None
+            try:
+                # Load noise track.
+                noise_signal = signal_processing.SignalProcessingUtils.LoadWav(
+                    noise_track_filepath)
+            except exceptions.FileNotFoundError:
+                # Generate noise track by applying the impulse response.
+                impulse_response_filepath = os.path.join(
+                    self._aechen_ir_database_path,
+                    self._IMPULSE_RESPONSES[impulse_response_name])
+                noise_signal = self._GenerateNoiseTrack(
+                    noise_track_filepath, input_signal,
+                    impulse_response_filepath)
+            assert noise_signal is not None
+
+            # Create the noisy mixes (once for each unique SNR value).
+            noisy_mix_filepaths[impulse_response_name] = {}
+            for snr in snr_values:
+                noisy_signal_filepath = os.path.join(
+                    test_data_cache_path,
+                    self._NOISY_SIGNAL_FILENAME_TEMPLATE.format(
+                        impulse_response_name, snr))
+
+                # Create and save if not done.
+                if not os.path.exists(noisy_signal_filepath):
+                    # Create noisy signal.
+                    noisy_signal = signal_processing.SignalProcessingUtils.MixSignals(
+                        input_signal, noise_signal, snr)
+
+                    # Save.
+                    signal_processing.SignalProcessingUtils.SaveWav(
+                        noisy_signal_filepath, noisy_signal)
+
+                # Add file to the collection of mixes.
+                noisy_mix_filepaths[impulse_response_name][
+                    snr] = noisy_signal_filepath
+
+        # Add all the noise-SNR pairs.
+        self._AddNoiseSnrPairs(base_output_path, noisy_mix_filepaths,
+                               self._SNR_VALUE_PAIRS)
+
+    def _GenerateNoiseTrack(self, noise_track_filepath, input_signal,
+                            impulse_response_filepath):
+        """Generates noise track.
+
+    Generate a signal by convolving input_signal with the impulse response in
+    impulse_response_filepath; then save to noise_track_filepath.
+
+    Args:
+      noise_track_filepath: output file path for the noise track.
+      input_signal: (clean) input signal samples.
+      impulse_response_filepath: impulse response file path.
+
+    Returns:
+      AudioSegment instance.
+    """
+        # Load impulse response.
+        data = scipy.io.loadmat(impulse_response_filepath)
+        impulse_response = data['h_air'].flatten()
+        if self._MAX_IMPULSE_RESPONSE_LENGTH is not None:
+            logging.info('truncating impulse response from %d to %d samples',
+                         len(impulse_response),
+                         self._MAX_IMPULSE_RESPONSE_LENGTH)
+            impulse_response = impulse_response[:self.
+                                                _MAX_IMPULSE_RESPONSE_LENGTH]
+
+        # Apply impulse response.
+        processed_signal = (
+            signal_processing.SignalProcessingUtils.ApplyImpulseResponse(
+                input_signal, impulse_response))
+
+        # Save.
+        signal_processing.SignalProcessingUtils.SaveWav(
+            noise_track_filepath, processed_signal)
+
+        return processed_signal
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_factory.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_factory.py
new file mode 100644
index 0000000000..948888e775
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_factory.py
@@ -0,0 +1,71 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""TestDataGenerator factory class.
+"""
+
+import logging
+
+from . import exceptions
+from . import test_data_generation
+
+
+class TestDataGeneratorFactory(object):
+    """Factory class used to create test data generators.
+
+  Usage: Create a factory passing parameters to the ctor with which the
+  generators will be produced.
+  """
+
+    def __init__(self, aechen_ir_database_path, noise_tracks_path,
+                 copy_with_identity):
+        """Ctor.
+
+    Args:
+      aechen_ir_database_path: Path to the Aechen Impulse Response database.
+      noise_tracks_path: Path to the noise tracks to add.
+      copy_with_identity: Flag indicating whether the identity generator has to
+                          make copies of the clean speech input files.
+    """
+        self._output_directory_prefix = None
+        self._aechen_ir_database_path = aechen_ir_database_path
+        self._noise_tracks_path = noise_tracks_path
+        self._copy_with_identity = copy_with_identity
+
+    def SetOutputDirectoryPrefix(self, prefix):
+        self._output_directory_prefix = prefix
+
+    def GetInstance(self, test_data_generators_class):
+        """Creates an TestDataGenerator instance given a class object.
+
+    Args:
+      test_data_generators_class: TestDataGenerator class object (not an
+                                  instance).
+
+    Returns:
+      TestDataGenerator instance.
+    """
+        if self._output_directory_prefix is None:
+            raise exceptions.InitializationException(
+                'The output directory prefix for test data generators is not set'
+            )
+        logging.debug('factory producing %s', test_data_generators_class)
+
+        if test_data_generators_class == (
+                test_data_generation.IdentityTestDataGenerator):
+            return test_data_generation.IdentityTestDataGenerator(
+                self._output_directory_prefix, self._copy_with_identity)
+        elif test_data_generators_class == (
+                test_data_generation.ReverberationTestDataGenerator):
+            return test_data_generation.ReverberationTestDataGenerator(
+                self._output_directory_prefix, self._aechen_ir_database_path)
+        elif test_data_generators_class == (
+                test_data_generation.AdditiveNoiseTestDataGenerator):
+            return test_data_generation.AdditiveNoiseTestDataGenerator(
+                self._output_directory_prefix, self._noise_tracks_path)
+        else:
+            return test_data_generators_class(self._output_directory_prefix)
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py
new file mode 100644
index 0000000000..f75098ae2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/test_data_generation_unittest.py
@@ -0,0 +1,207 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+"""Unit tests for the test_data_generation module.
+"""
+
+import os
+import shutil
+import tempfile
+import unittest
+
+import numpy as np
+import scipy.io
+
+from . import test_data_generation
+from . import test_data_generation_factory
+from . import signal_processing
+
+
+class TestTestDataGenerators(unittest.TestCase):
+    """Unit tests for the test_data_generation module.
+  """
+
+    def setUp(self):
+        """Create temporary folders."""
+        self._base_output_path = tempfile.mkdtemp()
+        self._test_data_cache_path = tempfile.mkdtemp()
+        self._fake_air_db_path = tempfile.mkdtemp()
+
+        # Fake AIR DB impulse responses.
+        # TODO(alessiob): ReverberationTestDataGenerator will change to allow custom
+        # impulse responses. When changed, the coupling below between
+        # impulse_response_mat_file_names and
+        # ReverberationTestDataGenerator._IMPULSE_RESPONSES can be removed.
+        impulse_response_mat_file_names = [
+            'air_binaural_lecture_0_0_1.mat',
+            'air_binaural_booth_0_0_1.mat',
+        ]
+        for impulse_response_mat_file_name in impulse_response_mat_file_names:
+            data = {'h_air': np.random.rand(1, 1000).astype('<f8')}
+            scipy.io.savemat(
+                os.path.join(self._fake_air_db_path,
+                             impulse_response_mat_file_name), data)
+
+    def tearDown(self):
+        """Recursively delete temporary folders."""
+        shutil.rmtree(self._base_output_path)
+        shutil.rmtree(self._test_data_cache_path)
+        shutil.rmtree(self._fake_air_db_path)
+
+    def testTestDataGenerators(self):
+        # Preliminary check.
+        self.assertTrue(os.path.exists(self._base_output_path))
+        self.assertTrue(os.path.exists(self._test_data_cache_path))
+
+        # Check that there is at least one registered test data generator.
+        registered_classes = (
+            test_data_generation.TestDataGenerator.REGISTERED_CLASSES)
+        self.assertIsInstance(registered_classes, dict)
+        self.assertGreater(len(registered_classes), 0)
+
+        # Instance generators factory.
+        generators_factory = test_data_generation_factory.TestDataGeneratorFactory(
+            aechen_ir_database_path=self._fake_air_db_path,
+            noise_tracks_path=test_data_generation.  \
+                              AdditiveNoiseTestDataGenerator.  \
+                              DEFAULT_NOISE_TRACKS_PATH,
+            copy_with_identity=False)
+        generators_factory.SetOutputDirectoryPrefix('datagen-')
+
+        # Use a simple input file as clean input signal.
+        input_signal_filepath = os.path.join(os.getcwd(), 'probing_signals',
+                                             'tone-880.wav')
+        self.assertTrue(os.path.exists(input_signal_filepath))
+
+        # Load input signal.
+        input_signal = signal_processing.SignalProcessingUtils.LoadWav(
+            input_signal_filepath)
+
+        # Try each registered test data generator.
+        for generator_name in registered_classes:
+            # Instance test data generator.
+            generator = generators_factory.GetInstance(
+                registered_classes[generator_name])
+
+            # Generate the noisy input - reference pairs.
+            generator.Generate(input_signal_filepath=input_signal_filepath,
+                               test_data_cache_path=self._test_data_cache_path,
+                               base_output_path=self._base_output_path)
+
+            # Perform checks.
+            self._CheckGeneratedPairsListSizes(generator)
+            self._CheckGeneratedPairsSignalDurations(generator, input_signal)
+            self._CheckGeneratedPairsOutputPaths(generator)
+
+    def testTestidentityDataGenerator(self):
+        # Preliminary check.
+        self.assertTrue(os.path.exists(self._base_output_path))
+        self.assertTrue(os.path.exists(self._test_data_cache_path))
+
+        # Use a simple input file as clean input signal.
+        input_signal_filepath = os.path.join(os.getcwd(), 'probing_signals',
+                                             'tone-880.wav')
+        self.assertTrue(os.path.exists(input_signal_filepath))
+
+        def GetNoiseReferenceFilePaths(identity_generator):
+            noisy_signal_filepaths = identity_generator.noisy_signal_filepaths
+            reference_signal_filepaths = identity_generator.reference_signal_filepaths
+            assert noisy_signal_filepaths.keys(
+            ) == reference_signal_filepaths.keys()
+            assert len(noisy_signal_filepaths.keys()) == 1
+            key = noisy_signal_filepaths.keys()[0]
+            return noisy_signal_filepaths[key], reference_signal_filepaths[key]
+
+        # Test the `copy_with_identity` flag.
+        for copy_with_identity in [False, True]:
+            # Instance the generator through the factory.
+            factory = test_data_generation_factory.TestDataGeneratorFactory(
+                aechen_ir_database_path='',
+                noise_tracks_path='',
+                copy_with_identity=copy_with_identity)
+            factory.SetOutputDirectoryPrefix('datagen-')
+            generator = factory.GetInstance(
+                test_data_generation.IdentityTestDataGenerator)
+            # Check `copy_with_identity` is set correctly.
+            self.assertEqual(copy_with_identity, generator.copy_with_identity)
+
+            # Generate test data and extract the paths to the noise and the reference
+            # files.
+            generator.Generate(input_signal_filepath=input_signal_filepath,
+                               test_data_cache_path=self._test_data_cache_path,
+                               base_output_path=self._base_output_path)
+            noisy_signal_filepath, reference_signal_filepath = (
+                GetNoiseReferenceFilePaths(generator))
+
+            # Check that a copy is made if and only if `copy_with_identity` is True.
+            if copy_with_identity:
+                self.assertNotEqual(noisy_signal_filepath,
+                                    input_signal_filepath)
+                self.assertNotEqual(reference_signal_filepath,
+                                    input_signal_filepath)
+            else:
+                self.assertEqual(noisy_signal_filepath, input_signal_filepath)
+                self.assertEqual(reference_signal_filepath,
+                                 input_signal_filepath)
+
+    def _CheckGeneratedPairsListSizes(self, generator):
+        config_names = generator.config_names
+        number_of_pairs = len(config_names)
+        self.assertEqual(number_of_pairs,
+                         len(generator.noisy_signal_filepaths))
+        self.assertEqual(number_of_pairs, len(generator.apm_output_paths))
+        self.assertEqual(number_of_pairs,
+                         len(generator.reference_signal_filepaths))
+
+    def _CheckGeneratedPairsSignalDurations(self, generator, input_signal):
+        """Checks duration of the generated signals.
+
+    Checks that the noisy input and the reference tracks are audio files
+    with duration equal to or greater than that of the input signal.
+
+    Args:
+      generator: TestDataGenerator instance.
+      input_signal: AudioSegment instance.
+    """
+        input_signal_length = (
+            signal_processing.SignalProcessingUtils.CountSamples(input_signal))
+
+        # Iterate over the noisy signal - reference pairs.
+        for config_name in generator.config_names:
+            # Load the noisy input file.
+            noisy_signal_filepath = generator.noisy_signal_filepaths[
+                config_name]
+            noisy_signal = signal_processing.SignalProcessingUtils.LoadWav(
+                noisy_signal_filepath)
+
+            # Check noisy input signal length.
+            noisy_signal_length = (signal_processing.SignalProcessingUtils.
+                                   CountSamples(noisy_signal))
+            self.assertGreaterEqual(noisy_signal_length, input_signal_length)
+
+            # Load the reference file.
+            reference_signal_filepath = generator.reference_signal_filepaths[
+                config_name]
+            reference_signal = signal_processing.SignalProcessingUtils.LoadWav(
+                reference_signal_filepath)
+
+            # Check noisy input signal length.
+            reference_signal_length = (signal_processing.SignalProcessingUtils.
+                                       CountSamples(reference_signal))
+            self.assertGreaterEqual(reference_signal_length,
+                                    input_signal_length)
+
+    def _CheckGeneratedPairsOutputPaths(self, generator):
+        """Checks that the output path created by the generator exists.
+
+    Args:
+      generator: TestDataGenerator instance.
+    """
+        # Iterate over the noisy signal - reference pairs.
+        for config_name in generator.config_names:
+            output_path = generator.apm_output_paths[config_name]
+            self.assertTrue(os.path.exists(output_path))
diff --git a/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
new file mode 100644
index 0000000000..b47f6221cb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
@@ -0,0 +1,103 @@
+// Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+//
+// Use of this source code is governed by a BSD-style license
+// that can be found in the LICENSE file in the root of the source
+// tree. An additional intellectual property rights grant can be found
+// in the file PATENTS.  All contributing project authors may
+// be found in the AUTHORS file in the root of the source tree.
+
+#include "common_audio/vad/include/vad.h"
+
+#include <array>
+#include <fstream>
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "common_audio/wav_file.h"
+#include "rtc_base/logging.h"
+
+ABSL_FLAG(std::string, i, "", "Input wav file");
+ABSL_FLAG(std::string, o, "", "VAD output file");
+
+namespace webrtc {
+namespace test {
+namespace {
+
+// The allowed values are 10, 20 or 30 ms.
+constexpr uint8_t kAudioFrameLengthMilliseconds = 30;
+constexpr int kMaxSampleRate = 48000;
+constexpr size_t kMaxFrameLen =
+    kAudioFrameLengthMilliseconds * kMaxSampleRate / 1000;
+
+constexpr uint8_t kBitmaskBuffSize = 8;
+
+int main(int argc, char* argv[]) {
+  absl::ParseCommandLine(argc, argv);
+  const std::string input_file = absl::GetFlag(FLAGS_i);
+  const std::string output_file = absl::GetFlag(FLAGS_o);
+  // Open wav input file and check properties.
+  WavReader wav_reader(input_file);
+  if (wav_reader.num_channels() != 1) {
+    RTC_LOG(LS_ERROR) << "Only mono wav files supported";
+    return 1;
+  }
+  if (wav_reader.sample_rate() > kMaxSampleRate) {
+    RTC_LOG(LS_ERROR) << "Beyond maximum sample rate (" << kMaxSampleRate
+                      << ")";
+    return 1;
+  }
+  const size_t audio_frame_length = rtc::CheckedDivExact(
+      kAudioFrameLengthMilliseconds * wav_reader.sample_rate(), 1000);
+  if (audio_frame_length > kMaxFrameLen) {
+    RTC_LOG(LS_ERROR) << "The frame size and/or the sample rate are too large.";
+    return 1;
+  }
+
+  // Create output file and write header.
+  std::ofstream out_file(output_file, std::ofstream::binary);
+  const char audio_frame_length_ms = kAudioFrameLengthMilliseconds;
+  out_file.write(&audio_frame_length_ms, 1);  // Header.
+
+  // Run VAD and write decisions.
+  std::unique_ptr<Vad> vad = CreateVad(Vad::Aggressiveness::kVadNormal);
+  std::array<int16_t, kMaxFrameLen> samples;
+  char buff = 0;     // Buffer to write one bit per frame.
+  uint8_t next = 0;  // Points to the next bit to write in `buff`.
+  while (true) {
+    // Process frame.
+    const auto read_samples =
+        wav_reader.ReadSamples(audio_frame_length, samples.data());
+    if (read_samples < audio_frame_length)
+      break;
+    const auto is_speech = vad->VoiceActivity(
+        samples.data(), audio_frame_length, wav_reader.sample_rate());
+
+    // Write output.
+    buff = is_speech ? buff | (1 << next) : buff & ~(1 << next);
+    if (++next == kBitmaskBuffSize) {
+      out_file.write(&buff, 1);  // Flush.
+      buff = 0;                  // Reset.
+      next = 0;
+    }
+  }
+
+  // Finalize.
+  char extra_bits = 0;
+  if (next > 0) {
+    extra_bits = kBitmaskBuffSize - next;
+    out_file.write(&buff, 1);  // Flush.
+  }
+  out_file.write(&extra_bits, 1);
+  out_file.close();
+
+  return 0;
+}
+
+}  // namespace
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::main(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.cc b/third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.cc
new file mode 100644
index 0000000000..4899d2d459
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/runtime_setting_util.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void ReplayRuntimeSetting(AudioProcessing* apm,
+                          const webrtc::audioproc::RuntimeSetting& setting) {
+  RTC_CHECK(apm);
+  // TODO(bugs.webrtc.org/9138): Add ability to handle different types
+  // of settings. Currently CapturePreGain, CaptureFixedPostGain and
+  // PlayoutVolumeChange are supported.
+  RTC_CHECK(setting.has_capture_pre_gain() ||
+            setting.has_capture_fixed_post_gain() ||
+            setting.has_playout_volume_change());
+
+  if (setting.has_capture_pre_gain()) {
+    apm->SetRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreateCapturePreGain(
+            setting.capture_pre_gain()));
+  } else if (setting.has_capture_fixed_post_gain()) {
+    apm->SetRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreateCaptureFixedPostGain(
+            setting.capture_fixed_post_gain()));
+  } else if (setting.has_playout_volume_change()) {
+    apm->SetRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(
+            setting.playout_volume_change()));
+  } else if (setting.has_playout_audio_device_change()) {
+    apm->SetRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreatePlayoutAudioDeviceChange(
+            {setting.playout_audio_device_change().id(),
+             setting.playout_audio_device_change().max_volume()}));
+  } else if (setting.has_capture_output_used()) {
+    apm->SetRuntimeSetting(
+        AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
+            setting.capture_output_used()));
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.h b/third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.h
new file mode 100644
index 0000000000..d8cbe82076
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/runtime_setting_util.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_RUNTIME_SETTING_UTIL_H_
+#define MODULES_AUDIO_PROCESSING_TEST_RUNTIME_SETTING_UTIL_H_
+
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/test/protobuf_utils.h"
+
+namespace webrtc {
+
+void ReplayRuntimeSetting(AudioProcessing* apm,
+                          const webrtc::audioproc::RuntimeSetting& setting);
+}
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_RUNTIME_SETTING_UTIL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.cc b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.cc
new file mode 100644
index 0000000000..458f6ced76
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/simulator_buffers.h"
+
+#include "modules/audio_processing/test/audio_buffer_tools.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+SimulatorBuffers::SimulatorBuffers(int render_input_sample_rate_hz,
+                                   int capture_input_sample_rate_hz,
+                                   int render_output_sample_rate_hz,
+                                   int capture_output_sample_rate_hz,
+                                   size_t num_render_input_channels,
+                                   size_t num_capture_input_channels,
+                                   size_t num_render_output_channels,
+                                   size_t num_capture_output_channels) {
+  Random rand_gen(42);
+  CreateConfigAndBuffer(render_input_sample_rate_hz, num_render_input_channels,
+                        &rand_gen, &render_input_buffer, &render_input_config,
+                        &render_input, &render_input_samples);
+
+  CreateConfigAndBuffer(render_output_sample_rate_hz,
+                        num_render_output_channels, &rand_gen,
+                        &render_output_buffer, &render_output_config,
+                        &render_output, &render_output_samples);
+
+  CreateConfigAndBuffer(capture_input_sample_rate_hz,
+                        num_capture_input_channels, &rand_gen,
+                        &capture_input_buffer, &capture_input_config,
+                        &capture_input, &capture_input_samples);
+
+  CreateConfigAndBuffer(capture_output_sample_rate_hz,
+                        num_capture_output_channels, &rand_gen,
+                        &capture_output_buffer, &capture_output_config,
+                        &capture_output, &capture_output_samples);
+
+  UpdateInputBuffers();
+}
+
+SimulatorBuffers::~SimulatorBuffers() = default;
+
+void SimulatorBuffers::CreateConfigAndBuffer(
+    int sample_rate_hz,
+    size_t num_channels,
+    Random* rand_gen,
+    std::unique_ptr<AudioBuffer>* buffer,
+    StreamConfig* config,
+    std::vector<float*>* buffer_data,
+    std::vector<float>* buffer_data_samples) {
+  int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
+  *config = StreamConfig(sample_rate_hz, num_channels);
+  buffer->reset(
+      new AudioBuffer(config->sample_rate_hz(), config->num_channels(),
+                      config->sample_rate_hz(), config->num_channels(),
+                      config->sample_rate_hz(), config->num_channels()));
+
+  buffer_data_samples->resize(samples_per_channel * num_channels);
+  for (auto& v : *buffer_data_samples) {
+    v = rand_gen->Rand<float>();
+  }
+
+  buffer_data->resize(num_channels);
+  for (size_t ch = 0; ch < num_channels; ++ch) {
+    (*buffer_data)[ch] = &(*buffer_data_samples)[ch * samples_per_channel];
+  }
+}
+
+void SimulatorBuffers::UpdateInputBuffers() {
+  test::CopyVectorToAudioBuffer(capture_input_config, capture_input_samples,
+                                capture_input_buffer.get());
+  test::CopyVectorToAudioBuffer(render_input_config, render_input_samples,
+                                render_input_buffer.get());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h
new file mode 100644
index 0000000000..36dcf301a2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/simulator_buffers.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
+
+#include <memory>
+#include <vector>
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+namespace test {
+
+struct SimulatorBuffers {
+  SimulatorBuffers(int render_input_sample_rate_hz,
+                   int capture_input_sample_rate_hz,
+                   int render_output_sample_rate_hz,
+                   int capture_output_sample_rate_hz,
+                   size_t num_render_input_channels,
+                   size_t num_capture_input_channels,
+                   size_t num_render_output_channels,
+                   size_t num_capture_output_channels);
+  ~SimulatorBuffers();
+
+  void CreateConfigAndBuffer(int sample_rate_hz,
+                             size_t num_channels,
+                             Random* rand_gen,
+                             std::unique_ptr<AudioBuffer>* buffer,
+                             StreamConfig* config,
+                             std::vector<float*>* buffer_data,
+                             std::vector<float>* buffer_data_samples);
+
+  void UpdateInputBuffers();
+
+  std::unique_ptr<AudioBuffer> render_input_buffer;
+  std::unique_ptr<AudioBuffer> capture_input_buffer;
+  std::unique_ptr<AudioBuffer> render_output_buffer;
+  std::unique_ptr<AudioBuffer> capture_output_buffer;
+  StreamConfig render_input_config;
+  StreamConfig capture_input_config;
+  StreamConfig render_output_config;
+  StreamConfig capture_output_config;
+  std::vector<float*> render_input;
+  std::vector<float> render_input_samples;
+  std::vector<float*> capture_input;
+  std::vector<float> capture_input_samples;
+  std::vector<float*> render_output;
+  std::vector<float> render_output_samples;
+  std::vector<float*> capture_output;
+  std::vector<float> capture_output_samples;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/test_utils.cc b/third_party/libwebrtc/modules/audio_processing/test/test_utils.cc
new file mode 100644
index 0000000000..9aeebe5155
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/test_utils.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/test_utils.h"
+
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+
+ChannelBufferWavReader::ChannelBufferWavReader(std::unique_ptr<WavReader> file)
+    : file_(std::move(file)) {}
+
+ChannelBufferWavReader::~ChannelBufferWavReader() = default;
+
+bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
+  interleaved_.resize(buffer->size());
+  if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
+      interleaved_.size()) {
+    return false;
+  }
+
+  FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
+               buffer->channels());
+  return true;
+}
+
+ChannelBufferWavWriter::ChannelBufferWavWriter(std::unique_ptr<WavWriter> file)
+    : file_(std::move(file)) {}
+
+ChannelBufferWavWriter::~ChannelBufferWavWriter() = default;
+
+void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
+  interleaved_.resize(buffer.size());
+  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
+             &interleaved_[0]);
+  FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  file_->WriteSamples(&interleaved_[0], interleaved_.size());
+}
+
+ChannelBufferVectorWriter::ChannelBufferVectorWriter(std::vector<float>* output)
+    : output_(output) {
+  RTC_DCHECK(output_);
+}
+
+ChannelBufferVectorWriter::~ChannelBufferVectorWriter() = default;
+
+void ChannelBufferVectorWriter::Write(const ChannelBuffer<float>& buffer) {
+  // Account for sample rate changes throughout a simulation.
+  interleaved_buffer_.resize(buffer.size());
+  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
+             interleaved_buffer_.data());
+  size_t old_size = output_->size();
+  output_->resize(old_size + interleaved_buffer_.size());
+  FloatToFloatS16(interleaved_buffer_.data(), interleaved_buffer_.size(),
+                  output_->data() + old_size);
+}
+
+FILE* OpenFile(absl::string_view filename, absl::string_view mode) {
+  std::string filename_str(filename);
+  FILE* file = fopen(filename_str.c_str(), std::string(mode).c_str());
+  if (!file) {
+    printf("Unable to open file %s\n", filename_str.c_str());
+    exit(1);
+  }
+  return file;
+}
+
+void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz) {
+  frame->sample_rate_hz = sample_rate_hz;
+  frame->samples_per_channel =
+      AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/test_utils.h b/third_party/libwebrtc/modules/audio_processing/test/test_utils.h
new file mode 100644
index 0000000000..bf82f9d66d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/test_utils.h
@@ -0,0 +1,170 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
+#define MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
+
+#include <math.h>
+
+#include <iterator>
+#include <limits>
+#include <memory>
+#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/channel_buffer.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
+#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
+
+// Encapsulates samples and metadata for an integer frame.
+struct Int16FrameData {
+  // Max data size that matches the data size of the AudioFrame class, providing
+  // storage for 8 channels of 96 kHz data.
+  static const int kMaxDataSizeSamples = 7680;
+
+  Int16FrameData() {
+    sample_rate_hz = 0;
+    num_channels = 0;
+    samples_per_channel = 0;
+    data.fill(0);
+  }
+
+  void CopyFrom(const Int16FrameData& src) {
+    samples_per_channel = src.samples_per_channel;
+    sample_rate_hz = src.sample_rate_hz;
+    num_channels = src.num_channels;
+
+    const size_t length = samples_per_channel * num_channels;
+    RTC_CHECK_LE(length, kMaxDataSizeSamples);
+    memcpy(data.data(), src.data.data(), sizeof(int16_t) * length);
+  }
+  std::array<int16_t, kMaxDataSizeSamples> data;
+  int32_t sample_rate_hz;
+  size_t num_channels;
+  size_t samples_per_channel;
+};
+
+// Reads ChannelBuffers from a provided WavReader.
+class ChannelBufferWavReader final {
+ public:
+  explicit ChannelBufferWavReader(std::unique_ptr<WavReader> file);
+  ~ChannelBufferWavReader();
+
+  ChannelBufferWavReader(const ChannelBufferWavReader&) = delete;
+  ChannelBufferWavReader& operator=(const ChannelBufferWavReader&) = delete;
+
+  // Reads data from the file according to the `buffer` format. Returns false if
+  // a full buffer can't be read from the file.
+  bool Read(ChannelBuffer<float>* buffer);
+
+ private:
+  std::unique_ptr<WavReader> file_;
+  std::vector<float> interleaved_;
+};
+
+// Writes ChannelBuffers to a provided WavWriter.
+class ChannelBufferWavWriter final {
+ public:
+  explicit ChannelBufferWavWriter(std::unique_ptr<WavWriter> file);
+  ~ChannelBufferWavWriter();
+
+  ChannelBufferWavWriter(const ChannelBufferWavWriter&) = delete;
+  ChannelBufferWavWriter& operator=(const ChannelBufferWavWriter&) = delete;
+
+  void Write(const ChannelBuffer<float>& buffer);
+
+ private:
+  std::unique_ptr<WavWriter> file_;
+  std::vector<float> interleaved_;
+};
+
+// Takes a pointer to a vector. Allows appending the samples of channel buffers
+// to the given vector, by interleaving the samples and converting them to float
+// S16.
+class ChannelBufferVectorWriter final {
+ public:
+  explicit ChannelBufferVectorWriter(std::vector<float>* output);
+  ChannelBufferVectorWriter(const ChannelBufferVectorWriter&) = delete;
+  ChannelBufferVectorWriter& operator=(const ChannelBufferVectorWriter&) =
+      delete;
+  ~ChannelBufferVectorWriter();
+
+  // Creates an interleaved copy of `buffer`, converts the samples to float S16
+  // and appends the result to output_.
+  void Write(const ChannelBuffer<float>& buffer);
+
+ private:
+  std::vector<float> interleaved_buffer_;
+  std::vector<float>* output_;
+};
+
+// Exits on failure; do not use in unit tests.
+FILE* OpenFile(absl::string_view filename, absl::string_view mode);
+
+void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz);
+
+template <typename T>
+void SetContainerFormat(int sample_rate_hz,
+                        size_t num_channels,
+                        Int16FrameData* frame,
+                        std::unique_ptr<ChannelBuffer<T> >* cb) {
+  SetFrameSampleRate(frame, sample_rate_hz);
+  frame->num_channels = num_channels;
+  cb->reset(new ChannelBuffer<T>(frame->samples_per_channel, num_channels));
+}
+
+template <typename T>
+float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) {
+  float mse = 0;
+  float mean = 0;
+  *variance = 0;
+  for (size_t i = 0; i < length; ++i) {
+    T error = ref[i] - test[i];
+    mse += error * error;
+    *variance += ref[i] * ref[i];
+    mean += ref[i];
+  }
+  mse /= length;
+  *variance /= length;
+  mean /= length;
+  *variance -= mean * mean;
+
+  float snr = 100;  // We assign 100 dB to the zero-error case.
+  if (mse > 0)
+    snr = 10 * log10(*variance / mse);
+  return snr;
+}
+
+// Returns a vector<T> parsed from whitespace delimited values in to_parse,
+// or an empty vector if the string could not be parsed.
+template <typename T>
+std::vector<T> ParseList(absl::string_view to_parse) {
+  std::vector<T> values;
+
+  std::istringstream str(  // no-presubmit-check TODO(webrtc:8982)
+      std::string{to_parse});
+  std::copy(
+      std::istream_iterator<T>(str),  // no-presubmit-check TODO(webrtc:8982)
+      std::istream_iterator<T>(),     // no-presubmit-check TODO(webrtc:8982)
+      std::back_inserter(values));
+
+  return values;
+}
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/test/unittest.proto b/third_party/libwebrtc/modules/audio_processing/test/unittest.proto
new file mode 100644
index 0000000000..07d1cda6c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/unittest.proto
@@ -0,0 +1,48 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.audioproc;
+
+message Test {
+  optional int32 num_reverse_channels = 1;
+  optional int32 num_input_channels = 2;
+  optional int32 num_output_channels = 3;
+  optional int32 sample_rate = 4;
+
+  message Frame {
+  }
+
+  repeated Frame frame = 5;
+
+  optional int32 analog_level_average = 6;
+  optional int32 max_output_average = 7;
+  optional int32 has_voice_count = 9;
+  optional int32 is_saturated_count = 10;
+
+  message EchoMetrics {
+    optional float echo_return_loss = 1;
+    optional float echo_return_loss_enhancement = 2;
+    optional float divergent_filter_fraction = 3;
+    optional float residual_echo_likelihood = 4;
+    optional float residual_echo_likelihood_recent_max = 5;
+  }
+
+  repeated EchoMetrics echo_metrics = 11;
+
+  message DelayMetrics {
+    optional int32 median = 1;
+    optional int32 std = 2;
+  }
+
+  repeated DelayMetrics delay_metrics = 12;
+
+  optional float rms_dbfs_average = 13;
+
+  optional float ns_speech_probability_average = 14;
+
+  optional bool use_aec_extended_filter = 15;
+}
+
+message OutputData {
+  repeated Test test = 1;
+}
+
diff --git a/third_party/libwebrtc/modules/audio_processing/test/wav_based_simulator.cc b/third_party/libwebrtc/modules/audio_processing/test/wav_based_simulator.cc
new file mode 100644
index 0000000000..ee87f9e1a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/wav_based_simulator.cc
@@ -0,0 +1,202 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/test/wav_based_simulator.h"
+
+#include <stdio.h>
+
+#include <iostream>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "modules/audio_processing/test/test_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/file_wrapper.h"
+
+namespace webrtc {
+namespace test {
+
+std::vector<WavBasedSimulator::SimulationEventType>
+WavBasedSimulator::GetCustomEventChain(absl::string_view filename) {
+  std::vector<WavBasedSimulator::SimulationEventType> call_chain;
+  FileWrapper file_wrapper = FileWrapper::OpenReadOnly(filename);
+
+  RTC_CHECK(file_wrapper.is_open())
+      << "Could not open the custom call order file, reverting "
+         "to using the default call order";
+
+  char c;
+  size_t num_read = file_wrapper.Read(&c, sizeof(char));
+  while (num_read > 0) {
+    switch (c) {
+      case 'r':
+        call_chain.push_back(SimulationEventType::kProcessReverseStream);
+        break;
+      case 'c':
+        call_chain.push_back(SimulationEventType::kProcessStream);
+        break;
+      case '\n':
+        break;
+      default:
+        RTC_FATAL() << "Incorrect custom call order file";
+    }
+
+    num_read = file_wrapper.Read(&c, sizeof(char));
+  }
+
+  return call_chain;
+}
+
+WavBasedSimulator::WavBasedSimulator(
+    const SimulationSettings& settings,
+    rtc::scoped_refptr<AudioProcessing> audio_processing,
+    std::unique_ptr<AudioProcessingBuilder> ap_builder)
+    : AudioProcessingSimulator(settings,
+                               std::move(audio_processing),
+                               std::move(ap_builder)) {
+  if (settings_.call_order_input_filename) {
+    call_chain_ = WavBasedSimulator::GetCustomEventChain(
+        *settings_.call_order_input_filename);
+  } else {
+    call_chain_ = WavBasedSimulator::GetDefaultEventChain();
+  }
+}
+
+WavBasedSimulator::~WavBasedSimulator() = default;
+
+std::vector<WavBasedSimulator::SimulationEventType>
+WavBasedSimulator::GetDefaultEventChain() {
+  std::vector<WavBasedSimulator::SimulationEventType> call_chain(2);
+  call_chain[0] = SimulationEventType::kProcessStream;
+  call_chain[1] = SimulationEventType::kProcessReverseStream;
+  return call_chain;
+}
+
+void WavBasedSimulator::PrepareProcessStreamCall() {
+  if (settings_.fixed_interface) {
+    fwd_frame_.CopyFrom(*in_buf_);
+  }
+  ap_->set_stream_key_pressed(settings_.override_key_pressed.value_or(false));
+
+  if (!settings_.use_stream_delay || *settings_.use_stream_delay) {
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 ap_->set_stream_delay_ms(
+                     settings_.stream_delay ? *settings_.stream_delay : 0));
+  }
+}
+
+void WavBasedSimulator::PrepareReverseProcessStreamCall() {
+  if (settings_.fixed_interface) {
+    rev_frame_.CopyFrom(*reverse_in_buf_);
+  }
+}
+
+void WavBasedSimulator::Process() {
+  ConfigureAudioProcessor();
+
+  Initialize();
+
+  bool samples_left_to_process = true;
+  int call_chain_index = 0;
+  int capture_frames_since_init = 0;
+  constexpr int kInitIndex = 1;
+  while (samples_left_to_process) {
+    switch (call_chain_[call_chain_index]) {
+      case SimulationEventType::kProcessStream:
+        SelectivelyToggleDataDumping(kInitIndex, capture_frames_since_init);
+
+        samples_left_to_process = HandleProcessStreamCall();
+        ++capture_frames_since_init;
+        break;
+      case SimulationEventType::kProcessReverseStream:
+        if (settings_.reverse_input_filename) {
+          samples_left_to_process = HandleProcessReverseStreamCall();
+        }
+        break;
+      default:
+        RTC_CHECK_NOTREACHED();
+    }
+
+    call_chain_index = (call_chain_index + 1) % call_chain_.size();
+  }
+
+  DetachAecDump();
+}
+
+void WavBasedSimulator::Analyze() {
+  std::cout << "Inits:" << std::endl;
+  std::cout << "1: -->" << std::endl;
+  std::cout << " Time:" << std::endl;
+  std::cout << "  Capture: 0 s (0 frames) " << std::endl;
+  std::cout << "  Render: 0 s (0 frames)" << std::endl;
+}
+
+bool WavBasedSimulator::HandleProcessStreamCall() {
+  bool samples_left_to_process = buffer_reader_->Read(in_buf_.get());
+  if (samples_left_to_process) {
+    PrepareProcessStreamCall();
+    ProcessStream(settings_.fixed_interface);
+  }
+  return samples_left_to_process;
+}
+
+bool WavBasedSimulator::HandleProcessReverseStreamCall() {
+  bool samples_left_to_process =
+      reverse_buffer_reader_->Read(reverse_in_buf_.get());
+  if (samples_left_to_process) {
+    PrepareReverseProcessStreamCall();
+    ProcessReverseStream(settings_.fixed_interface);
+  }
+  return samples_left_to_process;
+}
+
+void WavBasedSimulator::Initialize() {
+  std::unique_ptr<WavReader> in_file(
+      new WavReader(settings_.input_filename->c_str()));
+  int input_sample_rate_hz = in_file->sample_rate();
+  int input_num_channels = in_file->num_channels();
+  buffer_reader_.reset(new ChannelBufferWavReader(std::move(in_file)));
+
+  int output_sample_rate_hz = settings_.output_sample_rate_hz
+                                  ? *settings_.output_sample_rate_hz
+                                  : input_sample_rate_hz;
+  int output_num_channels = settings_.output_num_channels
+                                ? *settings_.output_num_channels
+                                : input_num_channels;
+
+  int reverse_sample_rate_hz = 48000;
+  int reverse_num_channels = 1;
+  int reverse_output_sample_rate_hz = 48000;
+  int reverse_output_num_channels = 1;
+  if (settings_.reverse_input_filename) {
+    std::unique_ptr<WavReader> reverse_in_file(
+        new WavReader(settings_.reverse_input_filename->c_str()));
+    reverse_sample_rate_hz = reverse_in_file->sample_rate();
+    reverse_num_channels = reverse_in_file->num_channels();
+    reverse_buffer_reader_.reset(
+        new ChannelBufferWavReader(std::move(reverse_in_file)));
+
+    reverse_output_sample_rate_hz =
+        settings_.reverse_output_sample_rate_hz
+            ? *settings_.reverse_output_sample_rate_hz
+            : reverse_sample_rate_hz;
+    reverse_output_num_channels = settings_.reverse_output_num_channels
+                                      ? *settings_.reverse_output_num_channels
+                                      : reverse_num_channels;
+  }
+
+  SetupBuffersConfigsOutputs(
+      input_sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz,
+      reverse_output_sample_rate_hz, input_num_channels, output_num_channels,
+      reverse_num_channels, reverse_output_num_channels);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/test/wav_based_simulator.h b/third_party/libwebrtc/modules/audio_processing/test/wav_based_simulator.h
new file mode 100644
index 0000000000..44e9ee2b7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/wav_based_simulator.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TEST_WAV_BASED_SIMULATOR_H_
+#define MODULES_AUDIO_PROCESSING_TEST_WAV_BASED_SIMULATOR_H_
+
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_processing/test/audio_processing_simulator.h"
+
+namespace webrtc {
+namespace test {
+
+// Used to perform an audio processing simulation from wav files.
+class WavBasedSimulator final : public AudioProcessingSimulator {
+ public:
+  WavBasedSimulator(const SimulationSettings& settings,
+                    rtc::scoped_refptr<AudioProcessing> audio_processing,
+                    std::unique_ptr<AudioProcessingBuilder> ap_builder);
+
+  WavBasedSimulator() = delete;
+  WavBasedSimulator(const WavBasedSimulator&) = delete;
+  WavBasedSimulator& operator=(const WavBasedSimulator&) = delete;
+
+  ~WavBasedSimulator() override;
+
+  // Processes the WAV input.
+  void Process() override;
+
+  // Only analyzes the data for the simulation, instead of perform any
+  // processing.
+  void Analyze() override;
+
+ private:
+  enum SimulationEventType {
+    kProcessStream,
+    kProcessReverseStream,
+  };
+
+  void Initialize();
+  bool HandleProcessStreamCall();
+  bool HandleProcessReverseStreamCall();
+  void PrepareProcessStreamCall();
+  void PrepareReverseProcessStreamCall();
+  static std::vector<SimulationEventType> GetDefaultEventChain();
+  static std::vector<SimulationEventType> GetCustomEventChain(
+      absl::string_view filename);
+
+  std::vector<SimulationEventType> call_chain_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_WAV_BASED_SIMULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/three_band_filter_bank.cc b/third_party/libwebrtc/modules/audio_processing/three_band_filter_bank.cc
new file mode 100644
index 0000000000..bd1c50477a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/three_band_filter_bank.cc
@@ -0,0 +1,278 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// An implementation of a 3-band FIR filter-bank with DCT modulation, similar to
+// the proposed in "Multirate Signal Processing for Communication Systems" by
+// Fredric J Harris.
+//
+// The idea is to take a heterodyne system and change the order of the
+// components to get something which is efficient to implement digitally.
+//
+// It is possible to separate the filter using the noble identity as follows:
+//
+// H(z) = H0(z^3) + z^-1 * H1(z^3) + z^-2 * H2(z^3)
+//
+// This is used in the analysis stage to first downsample serial to parallel
+// and then filter each branch with one of these polyphase decompositions of the
+// lowpass prototype. Because each filter is only a modulation of the prototype,
+// it is enough to multiply each coefficient by the respective cosine value to
+// shift it to the desired band. But because the cosine period is 12 samples,
+// it requires separating the prototype even further using the noble identity.
+// After filtering and modulating for each band, the output of all filters is
+// accumulated to get the downsampled bands.
+//
+// A similar logic can be applied to the synthesis stage.
+
+#include "modules/audio_processing/three_band_filter_bank.h"
+
+#include <array>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+// Factors to take into account when choosing `kFilterSize`:
+//   1. Higher `kFilterSize`, means faster transition, which ensures less
+//      aliasing. This is especially important when there is non-linear
+//      processing between the splitting and merging.
+//   2. The delay that this filter bank introduces is
+//      `kNumBands` * `kSparsity` * `kFilterSize` / 2, so it increases linearly
+//      with `kFilterSize`.
+//   3. The computation complexity also increases linearly with `kFilterSize`.
+
+// The Matlab code to generate these `kFilterCoeffs` is:
+//
+// N = kNumBands * kSparsity * kFilterSize - 1;
+// h = fir1(N, 1 / (2 * kNumBands), kaiser(N + 1, 3.5));
+// reshape(h, kNumBands * kSparsity, kFilterSize);
+//
+// The code below uses the values of kFilterSize, kNumBands and kSparsity
+// specified in the header.
+
+// Because the total bandwidth of the lower and higher band is double the middle
+// one (because of the spectrum parity), the low-pass prototype is half the
+// bandwidth of 1 / (2 * `kNumBands`) and is then shifted with cosine modulation
+// to the right places.
+// A Kaiser window is used because of its flexibility and the alpha is set to
+// 3.5, since that sets a stop band attenuation of 40dB ensuring a fast
+// transition.
+
+constexpr int kSubSampling = ThreeBandFilterBank::kNumBands;
+constexpr int kDctSize = ThreeBandFilterBank::kNumBands;
+static_assert(ThreeBandFilterBank::kNumBands *
+                      ThreeBandFilterBank::kSplitBandSize ==
+                  ThreeBandFilterBank::kFullBandSize,
+              "The full band must be split in equally sized subbands");
+
+const float
+    kFilterCoeffs[ThreeBandFilterBank::kNumNonZeroFilters][kFilterSize] = {
+        {-0.00047749f, -0.00496888f, +0.16547118f, +0.00425496f},
+        {-0.00173287f, -0.01585778f, +0.14989004f, +0.00994113f},
+        {-0.00304815f, -0.02536082f, +0.12154542f, +0.01157993f},
+        {-0.00346946f, -0.02587886f, +0.04760441f, +0.00607594f},
+        {-0.00154717f, -0.01136076f, +0.01387458f, +0.00186353f},
+        {+0.00186353f, +0.01387458f, -0.01136076f, -0.00154717f},
+        {+0.00607594f, +0.04760441f, -0.02587886f, -0.00346946f},
+        {+0.00983212f, +0.08543175f, -0.02982767f, -0.00383509f},
+        {+0.00994113f, +0.14989004f, -0.01585778f, -0.00173287f},
+        {+0.00425496f, +0.16547118f, -0.00496888f, -0.00047749f}};
+
+constexpr int kZeroFilterIndex1 = 3;
+constexpr int kZeroFilterIndex2 = 9;
+
+const float kDctModulation[ThreeBandFilterBank::kNumNonZeroFilters][kDctSize] =
+    {{2.f, 2.f, 2.f},
+     {1.73205077f, 0.f, -1.73205077f},
+     {1.f, -2.f, 1.f},
+     {-1.f, 2.f, -1.f},
+     {-1.73205077f, 0.f, 1.73205077f},
+     {-2.f, -2.f, -2.f},
+     {-1.73205077f, 0.f, 1.73205077f},
+     {-1.f, 2.f, -1.f},
+     {1.f, -2.f, 1.f},
+     {1.73205077f, 0.f, -1.73205077f}};
+
+// Filters the input signal `in` with the filter `filter` using a shift by
+// `in_shift`, taking into account the previous state.
+void FilterCore(
+    rtc::ArrayView<const float, kFilterSize> filter,
+    rtc::ArrayView<const float, ThreeBandFilterBank::kSplitBandSize> in,
+    const int in_shift,
+    rtc::ArrayView<float, ThreeBandFilterBank::kSplitBandSize> out,
+    rtc::ArrayView<float, kMemorySize> state) {
+  constexpr int kMaxInShift = (kStride - 1);
+  RTC_DCHECK_GE(in_shift, 0);
+  RTC_DCHECK_LE(in_shift, kMaxInShift);
+  std::fill(out.begin(), out.end(), 0.f);
+
+  for (int k = 0; k < in_shift; ++k) {
+    for (int i = 0, j = kMemorySize + k - in_shift; i < kFilterSize;
+         ++i, j -= kStride) {
+      out[k] += state[j] * filter[i];
+    }
+  }
+
+  for (int k = in_shift, shift = 0; k < kFilterSize * kStride; ++k, ++shift) {
+    RTC_DCHECK_GE(shift, 0);
+    const int loop_limit = std::min(kFilterSize, 1 + (shift >> kStrideLog2));
+    for (int i = 0, j = shift; i < loop_limit; ++i, j -= kStride) {
+      out[k] += in[j] * filter[i];
+    }
+    for (int i = loop_limit, j = kMemorySize + shift - loop_limit * kStride;
+         i < kFilterSize; ++i, j -= kStride) {
+      out[k] += state[j] * filter[i];
+    }
+  }
+
+  for (int k = kFilterSize * kStride, shift = kFilterSize * kStride - in_shift;
+       k < ThreeBandFilterBank::kSplitBandSize; ++k, ++shift) {
+    for (int i = 0, j = shift; i < kFilterSize; ++i, j -= kStride) {
+      out[k] += in[j] * filter[i];
+    }
+  }
+
+  // Update current state.
+  std::copy(in.begin() + ThreeBandFilterBank::kSplitBandSize - kMemorySize,
+            in.end(), state.begin());
+}
+
+}  // namespace
+
+// Because the low-pass filter prototype has half bandwidth it is possible to
+// use a DCT to shift it in both directions at the same time, to the center
+// frequencies [1 / 12, 3 / 12, 5 / 12].
+ThreeBandFilterBank::ThreeBandFilterBank() {
+  RTC_DCHECK_EQ(state_analysis_.size(), kNumNonZeroFilters);
+  RTC_DCHECK_EQ(state_synthesis_.size(), kNumNonZeroFilters);
+  for (int k = 0; k < kNumNonZeroFilters; ++k) {
+    RTC_DCHECK_EQ(state_analysis_[k].size(), kMemorySize);
+    RTC_DCHECK_EQ(state_synthesis_[k].size(), kMemorySize);
+
+    state_analysis_[k].fill(0.f);
+    state_synthesis_[k].fill(0.f);
+  }
+}
+
+ThreeBandFilterBank::~ThreeBandFilterBank() = default;
+
+// The analysis can be separated in these steps:
+//   1. Serial to parallel downsampling by a factor of `kNumBands`.
+//   2. Filtering of `kSparsity` different delayed signals with polyphase
+//      decomposition of the low-pass prototype filter and upsampled by a factor
+//      of `kSparsity`.
+//   3. Modulating with cosines and accumulating to get the desired band.
+void ThreeBandFilterBank::Analysis(
+    rtc::ArrayView<const float, kFullBandSize> in,
+    rtc::ArrayView<const rtc::ArrayView<float>, ThreeBandFilterBank::kNumBands>
+        out) {
+  // Initialize the output to zero.
+  for (int band = 0; band < ThreeBandFilterBank::kNumBands; ++band) {
+    RTC_DCHECK_EQ(out[band].size(), kSplitBandSize);
+    std::fill(out[band].begin(), out[band].end(), 0);
+  }
+
+  for (int downsampling_index = 0; downsampling_index < kSubSampling;
+       ++downsampling_index) {
+    // Downsample to form the filter input.
+    std::array<float, kSplitBandSize> in_subsampled;
+    for (int k = 0; k < kSplitBandSize; ++k) {
+      in_subsampled[k] =
+          in[(kSubSampling - 1) - downsampling_index + kSubSampling * k];
+    }
+
+    for (int in_shift = 0; in_shift < kStride; ++in_shift) {
+      // Choose filter, skip zero filters.
+      const int index = downsampling_index + in_shift * kSubSampling;
+      if (index == kZeroFilterIndex1 || index == kZeroFilterIndex2) {
+        continue;
+      }
+      const int filter_index =
+          index < kZeroFilterIndex1
+              ? index
+              : (index < kZeroFilterIndex2 ? index - 1 : index - 2);
+
+      rtc::ArrayView<const float, kFilterSize> filter(
+          kFilterCoeffs[filter_index]);
+      rtc::ArrayView<const float, kDctSize> dct_modulation(
+          kDctModulation[filter_index]);
+      rtc::ArrayView<float, kMemorySize> state(state_analysis_[filter_index]);
+
+      // Filter.
+      std::array<float, kSplitBandSize> out_subsampled;
+      FilterCore(filter, in_subsampled, in_shift, out_subsampled, state);
+
+      // Band and modulate the output.
+      for (int band = 0; band < ThreeBandFilterBank::kNumBands; ++band) {
+        float* out_band = out[band].data();
+        for (int n = 0; n < kSplitBandSize; ++n) {
+          out_band[n] += dct_modulation[band] * out_subsampled[n];
+        }
+      }
+    }
+  }
+}
+
+// The synthesis can be separated in these steps:
+//   1. Modulating with cosines.
+//   2. Filtering each one with a polyphase decomposition of the low-pass
+//      prototype filter upsampled by a factor of `kSparsity` and accumulating
+//      `kSparsity` signals with different delays.
+//   3. Parallel to serial upsampling by a factor of `kNumBands`.
+void ThreeBandFilterBank::Synthesis(
+    rtc::ArrayView<const rtc::ArrayView<float>, ThreeBandFilterBank::kNumBands>
+        in,
+    rtc::ArrayView<float, kFullBandSize> out) {
+  std::fill(out.begin(), out.end(), 0);
+  for (int upsampling_index = 0; upsampling_index < kSubSampling;
+       ++upsampling_index) {
+    for (int in_shift = 0; in_shift < kStride; ++in_shift) {
+      // Choose filter, skip zero filters.
+      const int index = upsampling_index + in_shift * kSubSampling;
+      if (index == kZeroFilterIndex1 || index == kZeroFilterIndex2) {
+        continue;
+      }
+      const int filter_index =
+          index < kZeroFilterIndex1
+              ? index
+              : (index < kZeroFilterIndex2 ? index - 1 : index - 2);
+
+      rtc::ArrayView<const float, kFilterSize> filter(
+          kFilterCoeffs[filter_index]);
+      rtc::ArrayView<const float, kDctSize> dct_modulation(
+          kDctModulation[filter_index]);
+      rtc::ArrayView<float, kMemorySize> state(state_synthesis_[filter_index]);
+
+      // Prepare filter input by modulating the banded input.
+      std::array<float, kSplitBandSize> in_subsampled;
+      std::fill(in_subsampled.begin(), in_subsampled.end(), 0.f);
+      for (int band = 0; band < ThreeBandFilterBank::kNumBands; ++band) {
+        RTC_DCHECK_EQ(in[band].size(), kSplitBandSize);
+        const float* in_band = in[band].data();
+        for (int n = 0; n < kSplitBandSize; ++n) {
+          in_subsampled[n] += dct_modulation[band] * in_band[n];
+        }
+      }
+
+      // Filter.
+      std::array<float, kSplitBandSize> out_subsampled;
+      FilterCore(filter, in_subsampled, in_shift, out_subsampled, state);
+
+      // Upsample.
+      constexpr float kUpsamplingScaling = kSubSampling;
+      for (int k = 0; k < kSplitBandSize; ++k) {
+        out[upsampling_index + kSubSampling * k] +=
+            kUpsamplingScaling * out_subsampled[k];
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/three_band_filter_bank.h b/third_party/libwebrtc/modules/audio_processing/three_band_filter_bank.h
new file mode 100644
index 0000000000..db66caba4a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/three_band_filter_bank.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_THREE_BAND_FILTER_BANK_H_
+#define MODULES_AUDIO_PROCESSING_THREE_BAND_FILTER_BANK_H_
+
+#include <array>
+#include <cstring>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+constexpr int kSparsity = 4;
+constexpr int kStrideLog2 = 2;
+constexpr int kStride = 1 << kStrideLog2;
+constexpr int kNumZeroFilters = 2;
+constexpr int kFilterSize = 4;
+constexpr int kMemorySize = kFilterSize * kStride - 1;
+static_assert(kMemorySize == 15,
+              "The memory size must be sufficient to provide memory for the "
+              "shifted filters");
+
+// An implementation of a 3-band FIR filter-bank with DCT modulation, similar to
+// the proposed in "Multirate Signal Processing for Communication Systems" by
+// Fredric J Harris.
+// The low-pass filter prototype has these characteristics:
+// * Pass-band ripple = 0.3dB
+// * Pass-band frequency = 0.147 (7kHz at 48kHz)
+// * Stop-band attenuation = 40dB
+// * Stop-band frequency = 0.192 (9.2kHz at 48kHz)
+// * Delay = 24 samples (500us at 48kHz)
+// * Linear phase
+// This filter bank does not satisfy perfect reconstruction. The SNR after
+// analysis and synthesis (with no processing in between) is approximately 9.5dB
+// depending on the input signal after compensating for the delay.
+class ThreeBandFilterBank final {
+ public:
+  static const int kNumBands = 3;
+  static const int kFullBandSize = 480;
+  static const int kSplitBandSize =
+      ThreeBandFilterBank::kFullBandSize / ThreeBandFilterBank::kNumBands;
+  static const int kNumNonZeroFilters =
+      kSparsity * ThreeBandFilterBank::kNumBands - kNumZeroFilters;
+
+  ThreeBandFilterBank();
+  ~ThreeBandFilterBank();
+
+  // Splits `in` of size kFullBandSize into 3 downsampled frequency bands in
+  // `out`, each of size 160.
+  void Analysis(rtc::ArrayView<const float, kFullBandSize> in,
+                rtc::ArrayView<const rtc::ArrayView<float>, kNumBands> out);
+
+  // Merges the 3 downsampled frequency bands in `in`, each of size 160, into
+  // `out`, which is of size kFullBandSize.
+  void Synthesis(rtc::ArrayView<const rtc::ArrayView<float>, kNumBands> in,
+                 rtc::ArrayView<float, kFullBandSize> out);
+
+ private:
+  std::array<std::array<float, kMemorySize>, kNumNonZeroFilters>
+      state_analysis_;
+  std::array<std::array<float, kMemorySize>, kNumNonZeroFilters>
+      state_synthesis_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_THREE_BAND_FILTER_BANK_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/transient/BUILD.gn
new file mode 100644
index 0000000000..41aeab0abe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/BUILD.gn
@@ -0,0 +1,133 @@
+# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("transient_suppressor_api") {
+  sources = [ "transient_suppressor.h" ]
+}
+
+rtc_library("transient_suppressor_impl") {
+  visibility = [
+    ":click_annotate",
+    ":transient_suppression_test",
+    ":transient_suppression_unittests",
+    "..:optionally_built_submodule_creators",
+  ]
+  sources = [
+    "common.h",
+    "daubechies_8_wavelet_coeffs.h",
+    "dyadic_decimator.h",
+    "moving_moments.cc",
+    "moving_moments.h",
+    "transient_detector.cc",
+    "transient_detector.h",
+    "transient_suppressor_impl.cc",
+    "transient_suppressor_impl.h",
+    "windows_private.h",
+    "wpd_node.cc",
+    "wpd_node.h",
+    "wpd_tree.cc",
+    "wpd_tree.h",
+  ]
+  deps = [
+    ":transient_suppressor_api",
+    ":voice_probability_delay_unit",
+    "../../../common_audio:common_audio",
+    "../../../common_audio:common_audio_c",
+    "../../../common_audio:fir_filter",
+    "../../../common_audio:fir_filter_factory",
+    "../../../common_audio/third_party/ooura:fft_size_256",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:gtest_prod",
+    "../../../rtc_base:logging",
+  ]
+}
+
+rtc_library("voice_probability_delay_unit") {
+  sources = [
+    "voice_probability_delay_unit.cc",
+    "voice_probability_delay_unit.h",
+  ]
+  deps = [ "../../../rtc_base:checks" ]
+}
+
+if (rtc_include_tests) {
+  if (!build_with_chromium) {
+    rtc_executable("click_annotate") {
+      testonly = true
+      sources = [
+        "click_annotate.cc",
+        "file_utils.cc",
+        "file_utils.h",
+      ]
+      deps = [
+        ":transient_suppressor_impl",
+        "..:audio_processing",
+        "../../../rtc_base/system:file_wrapper",
+        "../../../system_wrappers",
+      ]
+    }
+
+    rtc_executable("transient_suppression_test") {
+      testonly = true
+      sources = [
+        "file_utils.cc",
+        "file_utils.h",
+        "transient_suppression_test.cc",
+        "voice_probability_delay_unit_unittest.cc",
+      ]
+      deps = [
+        ":transient_suppressor_api",
+        ":transient_suppressor_impl",
+        ":voice_probability_delay_unit",
+        "..:audio_processing",
+        "../../../common_audio",
+        "../../../rtc_base/system:file_wrapper",
+        "../../../system_wrappers",
+        "../../../test:fileutils",
+        "../../../test:test_support",
+        "../agc:level_estimation",
+        "//testing/gtest",
+        "//third_party/abseil-cpp/absl/flags:flag",
+        "//third_party/abseil-cpp/absl/flags:parse",
+        "//third_party/abseil-cpp/absl/types:optional",
+      ]
+    }
+  }
+
+  rtc_library("transient_suppression_unittests") {
+    testonly = true
+    sources = [
+      "dyadic_decimator_unittest.cc",
+      "file_utils.cc",
+      "file_utils.h",
+      "file_utils_unittest.cc",
+      "moving_moments_unittest.cc",
+      "transient_detector_unittest.cc",
+      "transient_suppressor_unittest.cc",
+      "voice_probability_delay_unit_unittest.cc",
+      "wpd_node_unittest.cc",
+      "wpd_tree_unittest.cc",
+    ]
+    deps = [
+      ":transient_suppressor_api",
+      ":transient_suppressor_impl",
+      ":voice_probability_delay_unit",
+      "../../../rtc_base:stringutils",
+      "../../../rtc_base/system:file_wrapper",
+      "../../../test:fileutils",
+      "../../../test:test_support",
+      "//testing/gtest",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/click_annotate.cc b/third_party/libwebrtc/modules/audio_processing/transient/click_annotate.cc
new file mode 100644
index 0000000000..f3f040f9aa
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/click_annotate.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cfloat>
+#include <cstdio>
+#include <cstdlib>
+#include <memory>
+#include <vector>
+
+#include "modules/audio_processing/transient/file_utils.h"
+#include "modules/audio_processing/transient/transient_detector.h"
+#include "rtc_base/system/file_wrapper.h"
+
+using webrtc::FileWrapper;
+using webrtc::TransientDetector;
+
+// Application to generate a RTP timing file.
+// Opens the PCM file and divides the signal in frames.
+// Creates a send times array, one for each step.
+// Each block that contains a transient, has an infinite send time.
+// The resultant array is written to a DAT file
+// Returns -1 on error or `lost_packets` otherwise.
+int main(int argc, char* argv[]) {
+  if (argc != 5) {
+    printf("\n%s - Application to generate a RTP timing file.\n\n", argv[0]);
+    printf("%s PCMfile DATfile chunkSize sampleRate\n\n", argv[0]);
+    printf("Opens the PCMfile with sampleRate in Hertz.\n");
+    printf("Creates a send times array, one for each chunkSize ");
+    printf("milliseconds step.\n");
+    printf("Each block that contains a transient, has an infinite send time. ");
+    printf("The resultant array is written to a DATfile.\n\n");
+    return 0;
+  }
+
+  FileWrapper pcm_file = FileWrapper::OpenReadOnly(argv[1]);
+  if (!pcm_file.is_open()) {
+    printf("\nThe %s could not be opened.\n\n", argv[1]);
+    return -1;
+  }
+
+  FileWrapper dat_file = FileWrapper::OpenWriteOnly(argv[2]);
+  if (!dat_file.is_open()) {
+    printf("\nThe %s could not be opened.\n\n", argv[2]);
+    return -1;
+  }
+
+  int chunk_size_ms = atoi(argv[3]);
+  if (chunk_size_ms <= 0) {
+    printf("\nThe chunkSize must be a positive integer\n\n");
+    return -1;
+  }
+
+  int sample_rate_hz = atoi(argv[4]);
+  if (sample_rate_hz <= 0) {
+    printf("\nThe sampleRate must be a positive integer\n\n");
+    return -1;
+  }
+
+  TransientDetector detector(sample_rate_hz);
+  int lost_packets = 0;
+  size_t audio_buffer_length = chunk_size_ms * sample_rate_hz / 1000;
+  std::unique_ptr<float[]> audio_buffer(new float[audio_buffer_length]);
+  std::vector<float> send_times;
+
+  // Read first buffer from the PCM test file.
+  size_t file_samples_read = ReadInt16FromFileToFloatBuffer(
+      &pcm_file, audio_buffer_length, audio_buffer.get());
+  for (int time = 0; file_samples_read > 0; time += chunk_size_ms) {
+    // Pad the rest of the buffer with zeros.
+    for (size_t i = file_samples_read; i < audio_buffer_length; ++i) {
+      audio_buffer[i] = 0.0;
+    }
+    float value =
+        detector.Detect(audio_buffer.get(), audio_buffer_length, NULL, 0);
+    if (value < 0.5f) {
+      value = time;
+    } else {
+      value = FLT_MAX;
+      ++lost_packets;
+    }
+    send_times.push_back(value);
+
+    // Read next buffer from the PCM test file.
+    file_samples_read = ReadInt16FromFileToFloatBuffer(
+        &pcm_file, audio_buffer_length, audio_buffer.get());
+  }
+
+  size_t floats_written =
+      WriteFloatBufferToFile(&dat_file, send_times.size(), &send_times[0]);
+
+  if (floats_written == 0) {
+    printf("\nThe send times could not be written to DAT file\n\n");
+    return -1;
+  }
+
+  pcm_file.Close();
+  dat_file.Close();
+
+  return lost_packets;
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/common.h b/third_party/libwebrtc/modules/audio_processing/transient/common.h
new file mode 100644
index 0000000000..63c9a7b315
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/common.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_
+namespace webrtc {
+namespace ts {
+
+static const float kPi = 3.14159265358979323846f;
+static const int kChunkSizeMs = 10;
+enum {
+  kSampleRate8kHz = 8000,
+  kSampleRate16kHz = 16000,
+  kSampleRate32kHz = 32000,
+  kSampleRate48kHz = 48000
+};
+
+}  // namespace ts
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h b/third_party/libwebrtc/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h
new file mode 100644
index 0000000000..92233bfd74
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This header file defines the coefficients of the FIR based approximation of
+// the Meyer Wavelet
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_DAUBECHIES_8_WAVELET_COEFFS_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_DAUBECHIES_8_WAVELET_COEFFS_H_
+
+// Decomposition coefficients Daubechies 8.
+
+namespace webrtc {
+
+const int kDaubechies8CoefficientsLength = 16;
+
+const float kDaubechies8HighPassCoefficients[kDaubechies8CoefficientsLength] = {
+    -5.44158422430816093862e-02f, 3.12871590914465924627e-01f,
+    -6.75630736298012846142e-01f, 5.85354683654869090148e-01f,
+    1.58291052560238926228e-02f,  -2.84015542962428091389e-01f,
+    -4.72484573997972536787e-04f, 1.28747426620186011803e-01f,
+    1.73693010020221083600e-02f,  -4.40882539310647192377e-02f,
+    -1.39810279170155156436e-02f, 8.74609404701565465445e-03f,
+    4.87035299301066034600e-03f,  -3.91740372995977108837e-04f,
+    -6.75449405998556772109e-04f, -1.17476784002281916305e-04f};
+
+const float kDaubechies8LowPassCoefficients[kDaubechies8CoefficientsLength] = {
+    -1.17476784002281916305e-04f, 6.75449405998556772109e-04f,
+    -3.91740372995977108837e-04f, -4.87035299301066034600e-03f,
+    8.74609404701565465445e-03f,  1.39810279170155156436e-02f,
+    -4.40882539310647192377e-02f, -1.73693010020221083600e-02f,
+    1.28747426620186011803e-01f,  4.72484573997972536787e-04f,
+    -2.84015542962428091389e-01f, -1.58291052560238926228e-02f,
+    5.85354683654869090148e-01f,  6.75630736298012846142e-01f,
+    3.12871590914465924627e-01f,  5.44158422430816093862e-02f};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_DAUBECHIES_8_WAVELET_COEFFS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/dyadic_decimator.h b/third_party/libwebrtc/modules/audio_processing/transient/dyadic_decimator.h
new file mode 100644
index 0000000000..52467e8c25
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/dyadic_decimator.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
+
+#include <cstdlib>
+
+// Provides a set of static methods to perform dyadic decimations.
+
+namespace webrtc {
+
+// Returns the proper length of the output buffer that you should use for the
+// given `in_length` and decimation `odd_sequence`.
+// Return -1 on error.
+inline size_t GetOutLengthToDyadicDecimate(size_t in_length,
+                                           bool odd_sequence) {
+  size_t out_length = in_length / 2;
+
+  if (in_length % 2 == 1 && !odd_sequence) {
+    ++out_length;
+  }
+
+  return out_length;
+}
+
+// Performs a dyadic decimation: removes every odd/even member of a sequence
+// halving its overall length.
+// Arguments:
+//    in: array of `in_length`.
+//    odd_sequence: If false, the odd members will be removed (1, 3, 5, ...);
+//                  if true, the even members will be removed (0, 2, 4, ...).
+//    out: array of `out_length`. `out_length` must be large enough to
+//         hold the decimated output. The necessary length can be provided by
+//         GetOutLengthToDyadicDecimate().
+//         Must be previously allocated.
+// Returns the number of output samples, -1 on error.
+template <typename T>
+static size_t DyadicDecimate(const T* in,
+                             size_t in_length,
+                             bool odd_sequence,
+                             T* out,
+                             size_t out_length) {
+  size_t half_length = GetOutLengthToDyadicDecimate(in_length, odd_sequence);
+
+  if (!in || !out || in_length <= 0 || out_length < half_length) {
+    return 0;
+  }
+
+  size_t output_samples = 0;
+  size_t index_adjustment = odd_sequence ? 1 : 0;
+  for (output_samples = 0; output_samples < half_length; ++output_samples) {
+    out[output_samples] = in[output_samples * 2 + index_adjustment];
+  }
+
+  return output_samples;
+}
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_DYADIC_DECIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/dyadic_decimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/transient/dyadic_decimator_unittest.cc
new file mode 100644
index 0000000000..e4776d694f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/dyadic_decimator_unittest.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/dyadic_decimator.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+static const size_t kEvenBufferLength = 6;
+static const size_t kOddBufferLength = 5;
+static const size_t kOutBufferLength = 3;
+
+int16_t const test_buffer_even_len[] = {0, 1, 2, 3, 4, 5};
+int16_t const test_buffer_odd_len[] = {0, 1, 2, 3, 4};
+int16_t test_buffer_out[kOutBufferLength];
+
+TEST(DyadicDecimatorTest, GetOutLengthToDyadicDecimate) {
+  EXPECT_EQ(3u, GetOutLengthToDyadicDecimate(6, false));
+  EXPECT_EQ(3u, GetOutLengthToDyadicDecimate(6, true));
+  EXPECT_EQ(3u, GetOutLengthToDyadicDecimate(5, false));
+  EXPECT_EQ(2u, GetOutLengthToDyadicDecimate(5, true));
+}
+
+TEST(DyadicDecimatorTest, DyadicDecimateErrorValues) {
+  size_t out_samples = 0;
+
+  out_samples = DyadicDecimate(static_cast<int16_t*>(NULL), kEvenBufferLength,
+                               false,  // Even sequence.
+                               test_buffer_out, kOutBufferLength);
+  EXPECT_EQ(0u, out_samples);
+
+  out_samples = DyadicDecimate(test_buffer_even_len, kEvenBufferLength,
+                               false,  // Even sequence.
+                               static_cast<int16_t*>(NULL), kOutBufferLength);
+  EXPECT_EQ(0u, out_samples);
+
+  // Less than required `out_length`.
+  out_samples = DyadicDecimate(test_buffer_even_len, kEvenBufferLength,
+                               false,  // Even sequence.
+                               test_buffer_out, 2);
+  EXPECT_EQ(0u, out_samples);
+}
+
+TEST(DyadicDecimatorTest, DyadicDecimateEvenLengthEvenSequence) {
+  size_t expected_out_samples =
+      GetOutLengthToDyadicDecimate(kEvenBufferLength, false);
+
+  size_t out_samples = DyadicDecimate(test_buffer_even_len, kEvenBufferLength,
+                                      false,  // Even sequence.
+                                      test_buffer_out, kOutBufferLength);
+
+  EXPECT_EQ(expected_out_samples, out_samples);
+
+  EXPECT_EQ(0, test_buffer_out[0]);
+  EXPECT_EQ(2, test_buffer_out[1]);
+  EXPECT_EQ(4, test_buffer_out[2]);
+}
+
+TEST(DyadicDecimatorTest, DyadicDecimateEvenLengthOddSequence) {
+  size_t expected_out_samples =
+      GetOutLengthToDyadicDecimate(kEvenBufferLength, true);
+
+  size_t out_samples = DyadicDecimate(test_buffer_even_len, kEvenBufferLength,
+                                      true,  // Odd sequence.
+                                      test_buffer_out, kOutBufferLength);
+
+  EXPECT_EQ(expected_out_samples, out_samples);
+
+  EXPECT_EQ(1, test_buffer_out[0]);
+  EXPECT_EQ(3, test_buffer_out[1]);
+  EXPECT_EQ(5, test_buffer_out[2]);
+}
+
+TEST(DyadicDecimatorTest, DyadicDecimateOddLengthEvenSequence) {
+  size_t expected_out_samples =
+      GetOutLengthToDyadicDecimate(kOddBufferLength, false);
+
+  size_t out_samples = DyadicDecimate(test_buffer_odd_len, kOddBufferLength,
+                                      false,  // Even sequence.
+                                      test_buffer_out, kOutBufferLength);
+
+  EXPECT_EQ(expected_out_samples, out_samples);
+
+  EXPECT_EQ(0, test_buffer_out[0]);
+  EXPECT_EQ(2, test_buffer_out[1]);
+  EXPECT_EQ(4, test_buffer_out[2]);
+}
+
+TEST(DyadicDecimatorTest, DyadicDecimateOddLengthOddSequence) {
+  size_t expected_out_samples =
+      GetOutLengthToDyadicDecimate(kOddBufferLength, true);
+
+  size_t out_samples = DyadicDecimate(test_buffer_odd_len, kOddBufferLength,
+                                      true,  // Odd sequence.
+                                      test_buffer_out, kOutBufferLength);
+
+  EXPECT_EQ(expected_out_samples, out_samples);
+
+  EXPECT_EQ(1, test_buffer_out[0]);
+  EXPECT_EQ(3, test_buffer_out[1]);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/file_utils.cc b/third_party/libwebrtc/modules/audio_processing/transient/file_utils.cc
new file mode 100644
index 0000000000..58f99325d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/file_utils.cc
@@ -0,0 +1,257 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/file_utils.h"
+
+#include <memory>
+
+#include "rtc_base/system/file_wrapper.h"
+
+namespace webrtc {
+
+int ConvertByteArrayToFloat(const uint8_t bytes[4], float* out) {
+  if (!bytes || !out) {
+    return -1;
+  }
+
+  uint32_t binary_value = 0;
+  for (int i = 3; i >= 0; --i) {
+    binary_value <<= 8;
+    binary_value += bytes[i];
+  }
+
+  *out = bit_cast<float>(binary_value);
+
+  return 0;
+}
+
+int ConvertByteArrayToDouble(const uint8_t bytes[8], double* out) {
+  if (!bytes || !out) {
+    return -1;
+  }
+
+  uint64_t binary_value = 0;
+  for (int i = 7; i >= 0; --i) {
+    binary_value <<= 8;
+    binary_value += bytes[i];
+  }
+
+  *out = bit_cast<double>(binary_value);
+
+  return 0;
+}
+
+int ConvertFloatToByteArray(float value, uint8_t out_bytes[4]) {
+  if (!out_bytes) {
+    return -1;
+  }
+
+  uint32_t binary_value = bit_cast<uint32_t>(value);
+  for (size_t i = 0; i < 4; ++i) {
+    out_bytes[i] = binary_value;
+    binary_value >>= 8;
+  }
+
+  return 0;
+}
+
+int ConvertDoubleToByteArray(double value, uint8_t out_bytes[8]) {
+  if (!out_bytes) {
+    return -1;
+  }
+
+  uint64_t binary_value = bit_cast<uint64_t>(value);
+  for (size_t i = 0; i < 8; ++i) {
+    out_bytes[i] = binary_value;
+    binary_value >>= 8;
+  }
+
+  return 0;
+}
+
+size_t ReadInt16BufferFromFile(FileWrapper* file,
+                               size_t length,
+                               int16_t* buffer) {
+  if (!file || !file->is_open() || !buffer || length <= 0) {
+    return 0;
+  }
+
+  std::unique_ptr<uint8_t[]> byte_array(new uint8_t[2]);
+
+  size_t int16s_read = 0;
+
+  while (int16s_read < length) {
+    size_t bytes_read = file->Read(byte_array.get(), 2);
+    if (bytes_read < 2) {
+      break;
+    }
+    int16_t value = byte_array[1];
+    value <<= 8;
+    value += byte_array[0];
+    buffer[int16s_read] = value;
+    ++int16s_read;
+  }
+
+  return int16s_read;
+}
+
+size_t ReadInt16FromFileToFloatBuffer(FileWrapper* file,
+                                      size_t length,
+                                      float* buffer) {
+  if (!file || !file->is_open() || !buffer || length <= 0) {
+    return 0;
+  }
+
+  std::unique_ptr<int16_t[]> buffer16(new int16_t[length]);
+
+  size_t int16s_read = ReadInt16BufferFromFile(file, length, buffer16.get());
+
+  for (size_t i = 0; i < int16s_read; ++i) {
+    buffer[i] = buffer16[i];
+  }
+
+  return int16s_read;
+}
+
+size_t ReadInt16FromFileToDoubleBuffer(FileWrapper* file,
+                                       size_t length,
+                                       double* buffer) {
+  if (!file || !file->is_open() || !buffer || length <= 0) {
+    return 0;
+  }
+
+  std::unique_ptr<int16_t[]> buffer16(new int16_t[length]);
+
+  size_t int16s_read = ReadInt16BufferFromFile(file, length, buffer16.get());
+
+  for (size_t i = 0; i < int16s_read; ++i) {
+    buffer[i] = buffer16[i];
+  }
+
+  return int16s_read;
+}
+
+size_t ReadFloatBufferFromFile(FileWrapper* file,
+                               size_t length,
+                               float* buffer) {
+  if (!file || !file->is_open() || !buffer || length <= 0) {
+    return 0;
+  }
+
+  std::unique_ptr<uint8_t[]> byte_array(new uint8_t[4]);
+
+  size_t floats_read = 0;
+
+  while (floats_read < length) {
+    size_t bytes_read = file->Read(byte_array.get(), 4);
+    if (bytes_read < 4) {
+      break;
+    }
+    ConvertByteArrayToFloat(byte_array.get(), &buffer[floats_read]);
+    ++floats_read;
+  }
+
+  return floats_read;
+}
+
+size_t ReadDoubleBufferFromFile(FileWrapper* file,
+                                size_t length,
+                                double* buffer) {
+  if (!file || !file->is_open() || !buffer || length <= 0) {
+    return 0;
+  }
+
+  std::unique_ptr<uint8_t[]> byte_array(new uint8_t[8]);
+
+  size_t doubles_read = 0;
+
+  while (doubles_read < length) {
+    size_t bytes_read = file->Read(byte_array.get(), 8);
+    if (bytes_read < 8) {
+      break;
+    }
+    ConvertByteArrayToDouble(byte_array.get(), &buffer[doubles_read]);
+    ++doubles_read;
+  }
+
+  return doubles_read;
+}
+
+size_t WriteInt16BufferToFile(FileWrapper* file,
+                              size_t length,
+                              const int16_t* buffer) {
+  if (!file || !file->is_open() || !buffer || length <= 0) {
+    return 0;
+  }
+
+  std::unique_ptr<uint8_t[]> byte_array(new uint8_t[2]);
+
+  size_t int16s_written = 0;
+
+  for (int16s_written = 0; int16s_written < length; ++int16s_written) {
+    // Get byte representation.
+    byte_array[0] = buffer[int16s_written] & 0xFF;
+    byte_array[1] = (buffer[int16s_written] >> 8) & 0xFF;
+
+    file->Write(byte_array.get(), 2);
+  }
+
+  file->Flush();
+
+  return int16s_written;
+}
+
+size_t WriteFloatBufferToFile(FileWrapper* file,
+                              size_t length,
+                              const float* buffer) {
+  if (!file || !file->is_open() || !buffer || length <= 0) {
+    return 0;
+  }
+
+  std::unique_ptr<uint8_t[]> byte_array(new uint8_t[4]);
+
+  size_t floats_written = 0;
+
+  for (floats_written = 0; floats_written < length; ++floats_written) {
+    // Get byte representation.
+    ConvertFloatToByteArray(buffer[floats_written], byte_array.get());
+
+    file->Write(byte_array.get(), 4);
+  }
+
+  file->Flush();
+
+  return floats_written;
+}
+
+size_t WriteDoubleBufferToFile(FileWrapper* file,
+                               size_t length,
+                               const double* buffer) {
+  if (!file || !file->is_open() || !buffer || length <= 0) {
+    return 0;
+  }
+
+  std::unique_ptr<uint8_t[]> byte_array(new uint8_t[8]);
+
+  size_t doubles_written = 0;
+
+  for (doubles_written = 0; doubles_written < length; ++doubles_written) {
+    // Get byte representation.
+    ConvertDoubleToByteArray(buffer[doubles_written], byte_array.get());
+
+    file->Write(byte_array.get(), 8);
+  }
+
+  file->Flush();
+
+  return doubles_written;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/file_utils.h b/third_party/libwebrtc/modules/audio_processing/transient/file_utils.h
new file mode 100644
index 0000000000..b748337773
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/file_utils.h
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_FILE_UTILS_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_FILE_UTILS_H_
+
+#include <string.h>
+
+#include "rtc_base/system/file_wrapper.h"
+
+namespace webrtc {
+
+// This is a copy of the cast included in the Chromium codebase here:
+// http://cs.chromium.org/src/third_party/cld/base/casts.h
+template <class Dest, class Source>
+inline Dest bit_cast(const Source& source) {
+  // A compile error here means your Dest and Source have different sizes.
+  static_assert(sizeof(Dest) == sizeof(Source),
+                "Dest and Source have different sizes");
+
+  Dest dest;
+  memcpy(&dest, &source, sizeof(dest));
+  return dest;
+}
+
+// Converts the byte array with binary float representation to float.
+// Bytes must be in little-endian order.
+// Returns 0 if correct, -1 on error.
+int ConvertByteArrayToFloat(const uint8_t bytes[4], float* out);
+
+// Converts the byte array with binary double representation to double.
+// Bytes must be in little-endian order.
+// Returns 0 if correct, -1 on error.
+int ConvertByteArrayToDouble(const uint8_t bytes[8], double* out);
+
+// Converts a float to a byte array with binary float representation.
+// Bytes will be in little-endian order.
+// Returns 0 if correct, -1 on error.
+int ConvertFloatToByteArray(float value, uint8_t out_bytes[4]);
+
+// Converts a double to a byte array with binary double representation.
+// Bytes will be in little-endian order.
+// Returns 0 if correct, -1 on error.
+int ConvertDoubleToByteArray(double value, uint8_t out_bytes[8]);
+
+// Reads `length` 16-bit integers from `file` to `buffer`.
+// `file` must be previously opened.
+// Returns the number of 16-bit integers read or -1 on error.
+size_t ReadInt16BufferFromFile(FileWrapper* file,
+                               size_t length,
+                               int16_t* buffer);
+
+// Reads `length` 16-bit integers from `file` and stores those values
+// (converting them) in `buffer`.
+// `file` must be previously opened.
+// Returns the number of 16-bit integers read or -1 on error.
+size_t ReadInt16FromFileToFloatBuffer(FileWrapper* file,
+                                      size_t length,
+                                      float* buffer);
+
+// Reads `length` 16-bit integers from `file` and stores those values
+// (converting them) in `buffer`.
+// `file` must be previously opened.
+// Returns the number of 16-bit integers read or -1 on error.
+size_t ReadInt16FromFileToDoubleBuffer(FileWrapper* file,
+                                       size_t length,
+                                       double* buffer);
+
+// Reads `length` floats in binary representation (4 bytes) from `file` to
+// `buffer`.
+// `file` must be previously opened.
+// Returns the number of floats read or -1 on error.
+size_t ReadFloatBufferFromFile(FileWrapper* file, size_t length, float* buffer);
+
+// Reads `length` doubles in binary representation (8 bytes) from `file` to
+// `buffer`.
+// `file` must be previously opened.
+// Returns the number of doubles read or -1 on error.
+size_t ReadDoubleBufferFromFile(FileWrapper* file,
+                                size_t length,
+                                double* buffer);
+
+// Writes `length` 16-bit integers from `buffer` in binary representation (2
+// bytes) to `file`. It flushes `file`, so after this call there are no
+// writings pending.
+// `file` must be previously opened.
+// Returns the number of doubles written or -1 on error.
+size_t WriteInt16BufferToFile(FileWrapper* file,
+                              size_t length,
+                              const int16_t* buffer);
+
+// Writes `length` floats from `buffer` in binary representation (4 bytes) to
+// `file`. It flushes `file`, so after this call there are no writtings pending.
+// `file` must be previously opened.
+// Returns the number of doubles written or -1 on error.
+size_t WriteFloatBufferToFile(FileWrapper* file,
+                              size_t length,
+                              const float* buffer);
+
+// Writes `length` doubles from `buffer` in binary representation (8 bytes) to
+// `file`. It flushes `file`, so after this call there are no writings pending.
+// `file` must be previously opened.
+// Returns the number of doubles written or -1 on error.
+size_t WriteDoubleBufferToFile(FileWrapper* file,
+                               size_t length,
+                               const double* buffer);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_FILE_UTILS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/file_utils_unittest.cc b/third_party/libwebrtc/modules/audio_processing/transient/file_utils_unittest.cc
new file mode 100644
index 0000000000..a9dddb1eda
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/file_utils_unittest.cc
@@ -0,0 +1,501 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/file_utils.h"
+
+#include <string.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+static const uint8_t kPiBytesf[4] = {0xDB, 0x0F, 0x49, 0x40};
+static const uint8_t kEBytesf[4] = {0x54, 0xF8, 0x2D, 0x40};
+static const uint8_t kAvogadroBytesf[4] = {0x2F, 0x0C, 0xFF, 0x66};
+
+static const uint8_t kPiBytes[8] = {0x18, 0x2D, 0x44, 0x54,
+                                    0xFB, 0x21, 0x09, 0x40};
+static const uint8_t kEBytes[8] = {0x69, 0x57, 0x14, 0x8B,
+                                   0x0A, 0xBF, 0x05, 0x40};
+static const uint8_t kAvogadroBytes[8] = {0xF4, 0xBC, 0xA8, 0xDF,
+                                          0x85, 0xE1, 0xDF, 0x44};
+
+static const double kPi = 3.14159265358979323846;
+static const double kE = 2.71828182845904523536;
+static const double kAvogadro = 602214100000000000000000.0;
+
+class TransientFileUtilsTest : public ::testing::Test {
+ protected:
+  TransientFileUtilsTest()
+      : kTestFileName(
+            test::ResourcePath("audio_processing/transient/double-utils",
+                               "dat")),
+        kTestFileNamef(
+            test::ResourcePath("audio_processing/transient/float-utils",
+                               "dat")) {}
+
+  ~TransientFileUtilsTest() override { CleanupTempFiles(); }
+
+  std::string CreateTempFilename(absl::string_view dir,
+                                 absl::string_view prefix) {
+    std::string filename = test::TempFilename(dir, prefix);
+    temp_filenames_.push_back(filename);
+    return filename;
+  }
+
+  void CleanupTempFiles() {
+    for (const std::string& filename : temp_filenames_) {
+      remove(filename.c_str());
+    }
+    temp_filenames_.clear();
+  }
+
+  // This file (used in some tests) contains binary data. The data correspond to
+  // the double representation of the constants: Pi, E, and the Avogadro's
+  // Number;
+  // appended in that order.
+  const std::string kTestFileName;
+
+  // This file (used in some tests) contains binary data. The data correspond to
+  // the float representation of the constants: Pi, E, and the Avogadro's
+  // Number;
+  // appended in that order.
+  const std::string kTestFileNamef;
+
+  // List of temporary filenames created by CreateTempFilename.
+  std::vector<std::string> temp_filenames_;
+};
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertByteArrayToFloat DISABLED_ConvertByteArrayToFloat
+#else
+#define MAYBE_ConvertByteArrayToFloat ConvertByteArrayToFloat
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToFloat) {
+  float value = 0.0;
+
+  EXPECT_EQ(0, ConvertByteArrayToFloat(kPiBytesf, &value));
+  EXPECT_FLOAT_EQ(kPi, value);
+
+  EXPECT_EQ(0, ConvertByteArrayToFloat(kEBytesf, &value));
+  EXPECT_FLOAT_EQ(kE, value);
+
+  EXPECT_EQ(0, ConvertByteArrayToFloat(kAvogadroBytesf, &value));
+  EXPECT_FLOAT_EQ(kAvogadro, value);
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertByteArrayToDouble DISABLED_ConvertByteArrayToDouble
+#else
+#define MAYBE_ConvertByteArrayToDouble ConvertByteArrayToDouble
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertByteArrayToDouble) {
+  double value = 0.0;
+
+  EXPECT_EQ(0, ConvertByteArrayToDouble(kPiBytes, &value));
+  EXPECT_DOUBLE_EQ(kPi, value);
+
+  EXPECT_EQ(0, ConvertByteArrayToDouble(kEBytes, &value));
+  EXPECT_DOUBLE_EQ(kE, value);
+
+  EXPECT_EQ(0, ConvertByteArrayToDouble(kAvogadroBytes, &value));
+  EXPECT_DOUBLE_EQ(kAvogadro, value);
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertFloatToByteArray DISABLED_ConvertFloatToByteArray
+#else
+#define MAYBE_ConvertFloatToByteArray ConvertFloatToByteArray
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertFloatToByteArray) {
+  std::unique_ptr<uint8_t[]> bytes(new uint8_t[4]);
+
+  EXPECT_EQ(0, ConvertFloatToByteArray(kPi, bytes.get()));
+  EXPECT_EQ(0, memcmp(bytes.get(), kPiBytesf, 4));
+
+  EXPECT_EQ(0, ConvertFloatToByteArray(kE, bytes.get()));
+  EXPECT_EQ(0, memcmp(bytes.get(), kEBytesf, 4));
+
+  EXPECT_EQ(0, ConvertFloatToByteArray(kAvogadro, bytes.get()));
+  EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytesf, 4));
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ConvertDoubleToByteArray DISABLED_ConvertDoubleToByteArray
+#else
+#define MAYBE_ConvertDoubleToByteArray ConvertDoubleToByteArray
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ConvertDoubleToByteArray) {
+  std::unique_ptr<uint8_t[]> bytes(new uint8_t[8]);
+
+  EXPECT_EQ(0, ConvertDoubleToByteArray(kPi, bytes.get()));
+  EXPECT_EQ(0, memcmp(bytes.get(), kPiBytes, 8));
+
+  EXPECT_EQ(0, ConvertDoubleToByteArray(kE, bytes.get()));
+  EXPECT_EQ(0, memcmp(bytes.get(), kEBytes, 8));
+
+  EXPECT_EQ(0, ConvertDoubleToByteArray(kAvogadro, bytes.get()));
+  EXPECT_EQ(0, memcmp(bytes.get(), kAvogadroBytes, 8));
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16BufferFromFile DISABLED_ReadInt16BufferFromFile
+#else
+#define MAYBE_ReadInt16BufferFromFile ReadInt16BufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16BufferFromFile) {
+  std::string test_filename = kTestFileName;
+
+  FileWrapper file = FileWrapper::OpenReadOnly(test_filename);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kTestFileName.c_str();
+
+  const size_t kBufferLength = 12;
+  std::unique_ptr<int16_t[]> buffer(new int16_t[kBufferLength]);
+
+  EXPECT_EQ(kBufferLength,
+            ReadInt16BufferFromFile(&file, kBufferLength, buffer.get()));
+  EXPECT_EQ(22377, buffer[4]);
+  EXPECT_EQ(16389, buffer[7]);
+  EXPECT_EQ(17631, buffer[kBufferLength - 1]);
+
+  file.Rewind();
+
+  // The next test is for checking the case where there are not as much data as
+  // needed in the file, but reads to the end, and it returns the number of
+  // int16s read.
+  const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
+  buffer.reset(new int16_t[kBufferLenghtLargerThanFile]);
+  EXPECT_EQ(kBufferLength,
+            ReadInt16BufferFromFile(&file, kBufferLenghtLargerThanFile,
+                                    buffer.get()));
+  EXPECT_EQ(11544, buffer[0]);
+  EXPECT_EQ(22377, buffer[4]);
+  EXPECT_EQ(16389, buffer[7]);
+  EXPECT_EQ(17631, buffer[kBufferLength - 1]);
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16FromFileToFloatBuffer \
+  DISABLED_ReadInt16FromFileToFloatBuffer
+#else
+#define MAYBE_ReadInt16FromFileToFloatBuffer ReadInt16FromFileToFloatBuffer
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToFloatBuffer) {
+  std::string test_filename = kTestFileName;
+
+  FileWrapper file = FileWrapper::OpenReadOnly(test_filename);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kTestFileName.c_str();
+
+  const size_t kBufferLength = 12;
+  std::unique_ptr<float[]> buffer(new float[kBufferLength]);
+
+  EXPECT_EQ(kBufferLength,
+            ReadInt16FromFileToFloatBuffer(&file, kBufferLength, buffer.get()));
+
+  EXPECT_DOUBLE_EQ(11544, buffer[0]);
+  EXPECT_DOUBLE_EQ(22377, buffer[4]);
+  EXPECT_DOUBLE_EQ(16389, buffer[7]);
+  EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
+
+  file.Rewind();
+
+  // The next test is for checking the case where there are not as much data as
+  // needed in the file, but reads to the end, and it returns the number of
+  // int16s read.
+  const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
+  buffer.reset(new float[kBufferLenghtLargerThanFile]);
+  EXPECT_EQ(kBufferLength,
+            ReadInt16FromFileToFloatBuffer(&file, kBufferLenghtLargerThanFile,
+                                           buffer.get()));
+  EXPECT_DOUBLE_EQ(11544, buffer[0]);
+  EXPECT_DOUBLE_EQ(22377, buffer[4]);
+  EXPECT_DOUBLE_EQ(16389, buffer[7]);
+  EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadInt16FromFileToDoubleBuffer \
+  DISABLED_ReadInt16FromFileToDoubleBuffer
+#else
+#define MAYBE_ReadInt16FromFileToDoubleBuffer ReadInt16FromFileToDoubleBuffer
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToDoubleBuffer) {
+  std::string test_filename = kTestFileName;
+
+  FileWrapper file = FileWrapper::OpenReadOnly(test_filename);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kTestFileName.c_str();
+
+  const size_t kBufferLength = 12;
+  std::unique_ptr<double[]> buffer(new double[kBufferLength]);
+
+  EXPECT_EQ(kBufferLength, ReadInt16FromFileToDoubleBuffer(&file, kBufferLength,
+                                                           buffer.get()));
+  EXPECT_DOUBLE_EQ(11544, buffer[0]);
+  EXPECT_DOUBLE_EQ(22377, buffer[4]);
+  EXPECT_DOUBLE_EQ(16389, buffer[7]);
+  EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
+
+  file.Rewind();
+
+  // The next test is for checking the case where there are not as much data as
+  // needed in the file, but reads to the end, and it returns the number of
+  // int16s read.
+  const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
+  buffer.reset(new double[kBufferLenghtLargerThanFile]);
+  EXPECT_EQ(kBufferLength,
+            ReadInt16FromFileToDoubleBuffer(&file, kBufferLenghtLargerThanFile,
+                                            buffer.get()));
+  EXPECT_DOUBLE_EQ(11544, buffer[0]);
+  EXPECT_DOUBLE_EQ(22377, buffer[4]);
+  EXPECT_DOUBLE_EQ(16389, buffer[7]);
+  EXPECT_DOUBLE_EQ(17631, buffer[kBufferLength - 1]);
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadFloatBufferFromFile DISABLED_ReadFloatBufferFromFile
+#else
+#define MAYBE_ReadFloatBufferFromFile ReadFloatBufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadFloatBufferFromFile) {
+  std::string test_filename = kTestFileNamef;
+
+  FileWrapper file = FileWrapper::OpenReadOnly(test_filename);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kTestFileNamef.c_str();
+
+  const size_t kBufferLength = 3;
+  std::unique_ptr<float[]> buffer(new float[kBufferLength]);
+
+  EXPECT_EQ(kBufferLength,
+            ReadFloatBufferFromFile(&file, kBufferLength, buffer.get()));
+  EXPECT_FLOAT_EQ(kPi, buffer[0]);
+  EXPECT_FLOAT_EQ(kE, buffer[1]);
+  EXPECT_FLOAT_EQ(kAvogadro, buffer[2]);
+
+  file.Rewind();
+
+  // The next test is for checking the case where there are not as much data as
+  // needed in the file, but reads to the end, and it returns the number of
+  // doubles read.
+  const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
+  buffer.reset(new float[kBufferLenghtLargerThanFile]);
+  EXPECT_EQ(kBufferLength,
+            ReadFloatBufferFromFile(&file, kBufferLenghtLargerThanFile,
+                                    buffer.get()));
+  EXPECT_FLOAT_EQ(kPi, buffer[0]);
+  EXPECT_FLOAT_EQ(kE, buffer[1]);
+  EXPECT_FLOAT_EQ(kAvogadro, buffer[2]);
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ReadDoubleBufferFromFile DISABLED_ReadDoubleBufferFromFile
+#else
+#define MAYBE_ReadDoubleBufferFromFile ReadDoubleBufferFromFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ReadDoubleBufferFromFile) {
+  std::string test_filename = kTestFileName;
+
+  FileWrapper file = FileWrapper::OpenReadOnly(test_filename);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kTestFileName.c_str();
+
+  const size_t kBufferLength = 3;
+  std::unique_ptr<double[]> buffer(new double[kBufferLength]);
+
+  EXPECT_EQ(kBufferLength,
+            ReadDoubleBufferFromFile(&file, kBufferLength, buffer.get()));
+  EXPECT_DOUBLE_EQ(kPi, buffer[0]);
+  EXPECT_DOUBLE_EQ(kE, buffer[1]);
+  EXPECT_DOUBLE_EQ(kAvogadro, buffer[2]);
+
+  file.Rewind();
+
+  // The next test is for checking the case where there are not as much data as
+  // needed in the file, but reads to the end, and it returns the number of
+  // doubles read.
+  const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
+  buffer.reset(new double[kBufferLenghtLargerThanFile]);
+  EXPECT_EQ(kBufferLength,
+            ReadDoubleBufferFromFile(&file, kBufferLenghtLargerThanFile,
+                                     buffer.get()));
+  EXPECT_DOUBLE_EQ(kPi, buffer[0]);
+  EXPECT_DOUBLE_EQ(kE, buffer[1]);
+  EXPECT_DOUBLE_EQ(kAvogadro, buffer[2]);
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteInt16BufferToFile DISABLED_WriteInt16BufferToFile
+#else
+#define MAYBE_WriteInt16BufferToFile WriteInt16BufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteInt16BufferToFile) {
+  std::string kOutFileName =
+      CreateTempFilename(test::OutputPath(), "utils_test");
+
+  FileWrapper file = FileWrapper::OpenWriteOnly(kOutFileName);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kOutFileName.c_str();
+
+  const size_t kBufferLength = 3;
+  std::unique_ptr<int16_t[]> written_buffer(new int16_t[kBufferLength]);
+  std::unique_ptr<int16_t[]> read_buffer(new int16_t[kBufferLength]);
+
+  written_buffer[0] = 1;
+  written_buffer[1] = 2;
+  written_buffer[2] = 3;
+
+  EXPECT_EQ(kBufferLength,
+            WriteInt16BufferToFile(&file, kBufferLength, written_buffer.get()));
+
+  file.Close();
+
+  file = FileWrapper::OpenReadOnly(kOutFileName);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kOutFileName.c_str();
+
+  EXPECT_EQ(kBufferLength,
+            ReadInt16BufferFromFile(&file, kBufferLength, read_buffer.get()));
+  EXPECT_EQ(0, memcmp(written_buffer.get(), read_buffer.get(),
+                      kBufferLength * sizeof(written_buffer[0])));
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteFloatBufferToFile DISABLED_WriteFloatBufferToFile
+#else
+#define MAYBE_WriteFloatBufferToFile WriteFloatBufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteFloatBufferToFile) {
+  std::string kOutFileName =
+      CreateTempFilename(test::OutputPath(), "utils_test");
+
+  FileWrapper file = FileWrapper::OpenWriteOnly(kOutFileName);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kOutFileName.c_str();
+
+  const size_t kBufferLength = 3;
+  std::unique_ptr<float[]> written_buffer(new float[kBufferLength]);
+  std::unique_ptr<float[]> read_buffer(new float[kBufferLength]);
+
+  written_buffer[0] = static_cast<float>(kPi);
+  written_buffer[1] = static_cast<float>(kE);
+  written_buffer[2] = static_cast<float>(kAvogadro);
+
+  EXPECT_EQ(kBufferLength,
+            WriteFloatBufferToFile(&file, kBufferLength, written_buffer.get()));
+
+  file.Close();
+
+  file = FileWrapper::OpenReadOnly(kOutFileName);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kOutFileName.c_str();
+
+  EXPECT_EQ(kBufferLength,
+            ReadFloatBufferFromFile(&file, kBufferLength, read_buffer.get()));
+  EXPECT_EQ(0, memcmp(written_buffer.get(), read_buffer.get(),
+                      kBufferLength * sizeof(written_buffer[0])));
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_WriteDoubleBufferToFile DISABLED_WriteDoubleBufferToFile
+#else
+#define MAYBE_WriteDoubleBufferToFile WriteDoubleBufferToFile
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_WriteDoubleBufferToFile) {
+  std::string kOutFileName =
+      CreateTempFilename(test::OutputPath(), "utils_test");
+
+  FileWrapper file = FileWrapper::OpenWriteOnly(kOutFileName);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kOutFileName.c_str();
+
+  const size_t kBufferLength = 3;
+  std::unique_ptr<double[]> written_buffer(new double[kBufferLength]);
+  std::unique_ptr<double[]> read_buffer(new double[kBufferLength]);
+
+  written_buffer[0] = kPi;
+  written_buffer[1] = kE;
+  written_buffer[2] = kAvogadro;
+
+  EXPECT_EQ(kBufferLength, WriteDoubleBufferToFile(&file, kBufferLength,
+                                                   written_buffer.get()));
+
+  file.Close();
+
+  file = FileWrapper::OpenReadOnly(kOutFileName);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kOutFileName.c_str();
+
+  EXPECT_EQ(kBufferLength,
+            ReadDoubleBufferFromFile(&file, kBufferLength, read_buffer.get()));
+  EXPECT_EQ(0, memcmp(written_buffer.get(), read_buffer.get(),
+                      kBufferLength * sizeof(written_buffer[0])));
+}
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_ExpectedErrorReturnValues DISABLED_ExpectedErrorReturnValues
+#else
+#define MAYBE_ExpectedErrorReturnValues ExpectedErrorReturnValues
+#endif
+TEST_F(TransientFileUtilsTest, MAYBE_ExpectedErrorReturnValues) {
+  std::string test_filename = kTestFileName;
+
+  double value;
+  std::unique_ptr<int16_t[]> int16_buffer(new int16_t[1]);
+  std::unique_ptr<double[]> double_buffer(new double[1]);
+  FileWrapper file;
+
+  EXPECT_EQ(-1, ConvertByteArrayToDouble(NULL, &value));
+  EXPECT_EQ(-1, ConvertByteArrayToDouble(kPiBytes, NULL));
+
+  EXPECT_EQ(-1, ConvertDoubleToByteArray(kPi, NULL));
+
+  // Tests with file not opened.
+  EXPECT_EQ(0u, ReadInt16BufferFromFile(&file, 1, int16_buffer.get()));
+  EXPECT_EQ(0u, ReadInt16FromFileToDoubleBuffer(&file, 1, double_buffer.get()));
+  EXPECT_EQ(0u, ReadDoubleBufferFromFile(&file, 1, double_buffer.get()));
+  EXPECT_EQ(0u, WriteInt16BufferToFile(&file, 1, int16_buffer.get()));
+  EXPECT_EQ(0u, WriteDoubleBufferToFile(&file, 1, double_buffer.get()));
+
+  file = FileWrapper::OpenReadOnly(test_filename);
+  ASSERT_TRUE(file.is_open()) << "File could not be opened:\n"
+                              << kTestFileName.c_str();
+
+  EXPECT_EQ(0u, ReadInt16BufferFromFile(NULL, 1, int16_buffer.get()));
+  EXPECT_EQ(0u, ReadInt16BufferFromFile(&file, 1, NULL));
+  EXPECT_EQ(0u, ReadInt16BufferFromFile(&file, 0, int16_buffer.get()));
+
+  EXPECT_EQ(0u, ReadInt16FromFileToDoubleBuffer(NULL, 1, double_buffer.get()));
+  EXPECT_EQ(0u, ReadInt16FromFileToDoubleBuffer(&file, 1, NULL));
+  EXPECT_EQ(0u, ReadInt16FromFileToDoubleBuffer(&file, 0, double_buffer.get()));
+
+  EXPECT_EQ(0u, ReadDoubleBufferFromFile(NULL, 1, double_buffer.get()));
+  EXPECT_EQ(0u, ReadDoubleBufferFromFile(&file, 1, NULL));
+  EXPECT_EQ(0u, ReadDoubleBufferFromFile(&file, 0, double_buffer.get()));
+
+  EXPECT_EQ(0u, WriteInt16BufferToFile(NULL, 1, int16_buffer.get()));
+  EXPECT_EQ(0u, WriteInt16BufferToFile(&file, 1, NULL));
+  EXPECT_EQ(0u, WriteInt16BufferToFile(&file, 0, int16_buffer.get()));
+
+  EXPECT_EQ(0u, WriteDoubleBufferToFile(NULL, 1, double_buffer.get()));
+  EXPECT_EQ(0u, WriteDoubleBufferToFile(&file, 1, NULL));
+  EXPECT_EQ(0u, WriteDoubleBufferToFile(&file, 0, double_buffer.get()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/moving_moments.cc b/third_party/libwebrtc/modules/audio_processing/transient/moving_moments.cc
new file mode 100644
index 0000000000..83810bfe3c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/moving_moments.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/moving_moments.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+MovingMoments::MovingMoments(size_t length)
+    : length_(length), queue_(), sum_(0.0), sum_of_squares_(0.0) {
+  RTC_DCHECK_GT(length, 0);
+  for (size_t i = 0; i < length; ++i) {
+    queue_.push(0.0);
+  }
+}
+
+MovingMoments::~MovingMoments() {}
+
+void MovingMoments::CalculateMoments(const float* in,
+                                     size_t in_length,
+                                     float* first,
+                                     float* second) {
+  RTC_DCHECK(in);
+  RTC_DCHECK_GT(in_length, 0);
+  RTC_DCHECK(first);
+  RTC_DCHECK(second);
+
+  for (size_t i = 0; i < in_length; ++i) {
+    const float old_value = queue_.front();
+    queue_.pop();
+    queue_.push(in[i]);
+
+    sum_ += in[i] - old_value;
+    sum_of_squares_ += in[i] * in[i] - old_value * old_value;
+    first[i] = sum_ / length_;
+    second[i] = std::max(0.f, sum_of_squares_ / length_);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/moving_moments.h b/third_party/libwebrtc/modules/audio_processing/transient/moving_moments.h
new file mode 100644
index 0000000000..70451dcb71
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/moving_moments.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_MOVING_MOMENTS_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_MOVING_MOMENTS_H_
+
+#include <stddef.h>
+
+#include <queue>
+
+namespace webrtc {
+
+// Calculates the first and second moments for each value of a buffer taking
+// into account a given number of previous values.
+// It preserves its state, so it can be multiple-called.
+// TODO(chadan): Implement a function that takes a buffer of first moments and a
+// buffer of second moments; and calculates the variances. When needed.
+// TODO(chadan): Add functionality to update with a buffer but only output are
+// the last values of the moments. When needed.
+class MovingMoments {
+ public:
+  // Creates a Moving Moments object, that uses the last `length` values
+  // (including the new value introduced in every new calculation).
+  explicit MovingMoments(size_t length);
+  ~MovingMoments();
+
+  // Calculates the new values using `in`. Results will be in the out buffers.
+  // `first` and `second` must be allocated with at least `in_length`.
+  void CalculateMoments(const float* in,
+                        size_t in_length,
+                        float* first,
+                        float* second);
+
+ private:
+  size_t length_;
+  // A queue holding the `length_` latest input values.
+  std::queue<float> queue_;
+  // Sum of the values of the queue.
+  float sum_;
+  // Sum of the squares of the values of the queue.
+  float sum_of_squares_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_MOVING_MOMENTS_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/moving_moments_unittest.cc b/third_party/libwebrtc/modules/audio_processing/transient/moving_moments_unittest.cc
new file mode 100644
index 0000000000..b0e613e7ab
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/moving_moments_unittest.cc
@@ -0,0 +1,207 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/moving_moments.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+static const float kTolerance = 0.0001f;
+
+class MovingMomentsTest : public ::testing::Test {
+ protected:
+  static const size_t kMovingMomentsBufferLength = 5;
+  static const size_t kMaxOutputLength = 20;  // Valid for this tests only.
+
+  virtual void SetUp();
+  // Calls CalculateMoments and verifies that it produces the expected
+  // outputs.
+  void CalculateMomentsAndVerify(const float* input,
+                                 size_t input_length,
+                                 const float* expected_mean,
+                                 const float* expected_mean_squares);
+
+  std::unique_ptr<MovingMoments> moving_moments_;
+  float output_mean_[kMaxOutputLength];
+  float output_mean_squares_[kMaxOutputLength];
+};
+
+const size_t MovingMomentsTest::kMaxOutputLength;
+
+void MovingMomentsTest::SetUp() {
+  moving_moments_.reset(new MovingMoments(kMovingMomentsBufferLength));
+}
+
+void MovingMomentsTest::CalculateMomentsAndVerify(
+    const float* input,
+    size_t input_length,
+    const float* expected_mean,
+    const float* expected_mean_squares) {
+  ASSERT_LE(input_length, kMaxOutputLength);
+
+  moving_moments_->CalculateMoments(input, input_length, output_mean_,
+                                    output_mean_squares_);
+
+  for (size_t i = 1; i < input_length; ++i) {
+    EXPECT_NEAR(expected_mean[i], output_mean_[i], kTolerance);
+    EXPECT_NEAR(expected_mean_squares[i], output_mean_squares_[i], kTolerance);
+  }
+}
+
+TEST_F(MovingMomentsTest, CorrectMomentsOfAnAllZerosBuffer) {
+  const float kInput[] = {0.f, 0.f, 0.f, 0.f, 0.f};
+  const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
+
+  const float expected_mean[kInputLength] = {0.f, 0.f, 0.f, 0.f, 0.f};
+  const float expected_mean_squares[kInputLength] = {0.f, 0.f, 0.f, 0.f, 0.f};
+
+  CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
+                            expected_mean_squares);
+}
+
+TEST_F(MovingMomentsTest, CorrectMomentsOfAConstantBuffer) {
+  const float kInput[] = {5.f, 5.f, 5.f, 5.f, 5.f, 5.f, 5.f, 5.f, 5.f, 5.f};
+  const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
+
+  const float expected_mean[kInputLength] = {1.f, 2.f, 3.f, 4.f, 5.f,
+                                             5.f, 5.f, 5.f, 5.f, 5.f};
+  const float expected_mean_squares[kInputLength] = {
+      5.f, 10.f, 15.f, 20.f, 25.f, 25.f, 25.f, 25.f, 25.f, 25.f};
+
+  CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
+                            expected_mean_squares);
+}
+
+TEST_F(MovingMomentsTest, CorrectMomentsOfAnIncreasingBuffer) {
+  const float kInput[] = {1.f, 2.f, 3.f, 4.f, 5.f, 6.f, 7.f, 8.f, 9.f};
+  const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
+
+  const float expected_mean[kInputLength] = {0.2f, 0.6f, 1.2f, 2.f, 3.f,
+                                             4.f,  5.f,  6.f,  7.f};
+  const float expected_mean_squares[kInputLength] = {
+      0.2f, 1.f, 2.8f, 6.f, 11.f, 18.f, 27.f, 38.f, 51.f};
+
+  CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
+                            expected_mean_squares);
+}
+
+TEST_F(MovingMomentsTest, CorrectMomentsOfADecreasingBuffer) {
+  const float kInput[] = {-1.f, -2.f, -3.f, -4.f, -5.f, -6.f, -7.f, -8.f, -9.f};
+  const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
+
+  const float expected_mean[kInputLength] = {-0.2f, -0.6f, -1.2f, -2.f, -3.f,
+                                             -4.f,  -5.f,  -6.f,  -7.f};
+  const float expected_mean_squares[kInputLength] = {
+      0.2f, 1.f, 2.8f, 6.f, 11.f, 18.f, 27.f, 38.f, 51.f};
+
+  CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
+                            expected_mean_squares);
+}
+
+TEST_F(MovingMomentsTest, CorrectMomentsOfAZeroMeanSequence) {
+  const size_t kMovingMomentsBufferLength = 4;
+  moving_moments_.reset(new MovingMoments(kMovingMomentsBufferLength));
+  const float kInput[] = {1.f,  -1.f, 1.f,  -1.f, 1.f,
+                          -1.f, 1.f,  -1.f, 1.f,  -1.f};
+  const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
+
+  const float expected_mean[kInputLength] = {0.25f, 0.f, 0.25f, 0.f, 0.f,
+                                             0.f,   0.f, 0.f,   0.f, 0.f};
+  const float expected_mean_squares[kInputLength] = {
+      0.25f, 0.5f, 0.75f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f};
+
+  CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
+                            expected_mean_squares);
+}
+
+TEST_F(MovingMomentsTest, CorrectMomentsOfAnArbitraryBuffer) {
+  const float kInput[] = {0.2f,  0.3f,  0.5f,  0.7f, 0.11f,
+                          0.13f, 0.17f, 0.19f, 0.23f};
+  const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
+
+  const float expected_mean[kInputLength] = {
+      0.04f, 0.1f, 0.2f, 0.34f, 0.362f, 0.348f, 0.322f, 0.26f, 0.166f};
+  const float expected_mean_squares[kInputLength] = {0.008f,  0.026f,  0.076f,
+                                                     0.174f,  0.1764f, 0.1718f,
+                                                     0.1596f, 0.1168f, 0.0294f};
+
+  CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
+                            expected_mean_squares);
+}
+
+TEST_F(MovingMomentsTest, MutipleCalculateMomentsCalls) {
+  const float kInputFirstCall[] = {0.2f,  0.3f,  0.5f,  0.7f, 0.11f,
+                                   0.13f, 0.17f, 0.19f, 0.23f};
+  const size_t kInputFirstCallLength =
+      sizeof(kInputFirstCall) / sizeof(kInputFirstCall[0]);
+  const float kInputSecondCall[] = {0.29f, 0.31f};
+  const size_t kInputSecondCallLength =
+      sizeof(kInputSecondCall) / sizeof(kInputSecondCall[0]);
+  const float kInputThirdCall[] = {0.37f, 0.41f, 0.43f, 0.47f};
+  const size_t kInputThirdCallLength =
+      sizeof(kInputThirdCall) / sizeof(kInputThirdCall[0]);
+
+  const float expected_mean_first_call[kInputFirstCallLength] = {
+      0.04f, 0.1f, 0.2f, 0.34f, 0.362f, 0.348f, 0.322f, 0.26f, 0.166f};
+  const float expected_mean_squares_first_call[kInputFirstCallLength] = {
+      0.008f,  0.026f,  0.076f,  0.174f, 0.1764f,
+      0.1718f, 0.1596f, 0.1168f, 0.0294f};
+
+  const float expected_mean_second_call[kInputSecondCallLength] = {0.202f,
+                                                                   0.238f};
+  const float expected_mean_squares_second_call[kInputSecondCallLength] = {
+      0.0438f, 0.0596f};
+
+  const float expected_mean_third_call[kInputThirdCallLength] = {
+      0.278f, 0.322f, 0.362f, 0.398f};
+  const float expected_mean_squares_third_call[kInputThirdCallLength] = {
+      0.0812f, 0.1076f, 0.134f, 0.1614f};
+
+  CalculateMomentsAndVerify(kInputFirstCall, kInputFirstCallLength,
+                            expected_mean_first_call,
+                            expected_mean_squares_first_call);
+
+  CalculateMomentsAndVerify(kInputSecondCall, kInputSecondCallLength,
+                            expected_mean_second_call,
+                            expected_mean_squares_second_call);
+
+  CalculateMomentsAndVerify(kInputThirdCall, kInputThirdCallLength,
+                            expected_mean_third_call,
+                            expected_mean_squares_third_call);
+}
+
+TEST_F(MovingMomentsTest, VerifySampleBasedVsBlockBasedCalculation) {
+  const float kInput[] = {0.2f,  0.3f,  0.5f,  0.7f, 0.11f,
+                          0.13f, 0.17f, 0.19f, 0.23f};
+  const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
+
+  float output_mean_block_based[kInputLength];
+  float output_mean_squares_block_based[kInputLength];
+
+  float output_mean_sample_based;
+  float output_mean_squares_sample_based;
+
+  moving_moments_->CalculateMoments(kInput, kInputLength,
+                                    output_mean_block_based,
+                                    output_mean_squares_block_based);
+  moving_moments_.reset(new MovingMoments(kMovingMomentsBufferLength));
+  for (size_t i = 0; i < kInputLength; ++i) {
+    moving_moments_->CalculateMoments(&kInput[i], 1, &output_mean_sample_based,
+                                      &output_mean_squares_sample_based);
+    EXPECT_FLOAT_EQ(output_mean_block_based[i], output_mean_sample_based);
+    EXPECT_FLOAT_EQ(output_mean_squares_block_based[i],
+                    output_mean_squares_sample_based);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/test/plotDetection.m b/third_party/libwebrtc/modules/audio_processing/transient/test/plotDetection.m
new file mode 100644
index 0000000000..8e12ab920b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/test/plotDetection.m
@@ -0,0 +1,22 @@
+%
+%  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+%
+%  Use of this source code is governed by a BSD-style license
+%  that can be found in the LICENSE file in the root of the source
+%  tree. An additional intellectual property rights grant can be found
+%  in the file PATENTS.  All contributing project authors may
+%  be found in the AUTHORS file in the root of the source tree.
+%
+
+function [] = plotDetection(PCMfile, DATfile, fs, chunkSize)
+%[] = plotDetection(PCMfile, DATfile, fs, chunkSize)
+%
+%Plots the signal alongside the detection values.
+%
+%PCMfile: The file of the input signal in PCM format.
+%DATfile: The file containing the detection values in binary float format.
+%fs: The sample rate of the signal in Hertz.
+%chunkSize: The chunk size used to compute the detection values in seconds.
+[x, tx] = readPCM(PCMfile, fs);
+[d, td] = readDetection(DATfile, fs, chunkSize);
+plot(tx, x, td, d);
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/test/readDetection.m b/third_party/libwebrtc/modules/audio_processing/transient/test/readDetection.m
new file mode 100644
index 0000000000..832bf31ec8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/test/readDetection.m
@@ -0,0 +1,26 @@
+%
+%  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+%
+%  Use of this source code is governed by a BSD-style license
+%  that can be found in the LICENSE file in the root of the source
+%  tree. An additional intellectual property rights grant can be found
+%  in the file PATENTS.  All contributing project authors may
+%  be found in the AUTHORS file in the root of the source tree.
+%
+
+function [d, t] = readDetection(file, fs, chunkSize)
+%[d, t] = readDetection(file, fs, chunkSize)
+%
+%Reads a detection signal from a DAT file.
+%
+%d: The detection signal.
+%t: The respective time vector.
+%
+%file: The DAT file where the detection signal is stored in float format.
+%fs: The signal sample rate in Hertz.
+%chunkSize: The chunk size used for the detection in seconds.
+fid = fopen(file);
+d = fread(fid, inf, 'float');
+fclose(fid);
+t = 0:(1 / fs):(length(d) * chunkSize - 1 / fs);
+d = d(floor(t / chunkSize) + 1);
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/test/readPCM.m b/third_party/libwebrtc/modules/audio_processing/transient/test/readPCM.m
new file mode 100644
index 0000000000..cd3cef8a3c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/test/readPCM.m
@@ -0,0 +1,26 @@
+%
+%  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+%
+%  Use of this source code is governed by a BSD-style license
+%  that can be found in the LICENSE file in the root of the source
+%  tree. An additional intellectual property rights grant can be found
+%  in the file PATENTS.  All contributing project authors may
+%  be found in the AUTHORS file in the root of the source tree.
+%
+
+function [x, t] = readPCM(file, fs)
+%[x, t] = readPCM(file, fs)
+%
+%Reads a signal from a PCM file.
+%
+%x: The read signal after normalization.
+%t: The respective time vector.
+%
+%file: The PCM file where the signal is stored in int16 format.
+%fs: The signal sample rate in Hertz.
+fid = fopen(file);
+x = fread(fid, inf, 'int16');
+fclose(fid);
+x = x - mean(x);
+x = x / max(abs(x));
+t = 0:(1 / fs):((length(x) - 1) / fs);
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_detector.cc b/third_party/libwebrtc/modules/audio_processing/transient/transient_detector.cc
new file mode 100644
index 0000000000..5c35505368
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_detector.cc
@@ -0,0 +1,176 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/transient_detector.h"
+
+#include <float.h>
+#include <string.h>
+
+#include <algorithm>
+#include <cmath>
+
+#include "modules/audio_processing/transient/common.h"
+#include "modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h"
+#include "modules/audio_processing/transient/moving_moments.h"
+#include "modules/audio_processing/transient/wpd_node.h"
+#include "modules/audio_processing/transient/wpd_tree.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+static const int kTransientLengthMs = 30;
+static const int kChunksAtStartupLeftToDelete =
+    kTransientLengthMs / ts::kChunkSizeMs;
+static const float kDetectThreshold = 16.f;
+
+TransientDetector::TransientDetector(int sample_rate_hz)
+    : samples_per_chunk_(sample_rate_hz * ts::kChunkSizeMs / 1000),
+      last_first_moment_(),
+      last_second_moment_(),
+      chunks_at_startup_left_to_delete_(kChunksAtStartupLeftToDelete),
+      reference_energy_(1.f),
+      using_reference_(false) {
+  RTC_DCHECK(sample_rate_hz == ts::kSampleRate8kHz ||
+             sample_rate_hz == ts::kSampleRate16kHz ||
+             sample_rate_hz == ts::kSampleRate32kHz ||
+             sample_rate_hz == ts::kSampleRate48kHz);
+  int samples_per_transient = sample_rate_hz * kTransientLengthMs / 1000;
+  // Adjustment to avoid data loss while downsampling, making
+  // `samples_per_chunk_` and `samples_per_transient` always divisible by
+  // `kLeaves`.
+  samples_per_chunk_ -= samples_per_chunk_ % kLeaves;
+  samples_per_transient -= samples_per_transient % kLeaves;
+
+  tree_leaves_data_length_ = samples_per_chunk_ / kLeaves;
+  wpd_tree_.reset(new WPDTree(samples_per_chunk_,
+                              kDaubechies8HighPassCoefficients,
+                              kDaubechies8LowPassCoefficients,
+                              kDaubechies8CoefficientsLength, kLevels));
+  for (size_t i = 0; i < kLeaves; ++i) {
+    moving_moments_[i].reset(
+        new MovingMoments(samples_per_transient / kLeaves));
+  }
+
+  first_moments_.reset(new float[tree_leaves_data_length_]);
+  second_moments_.reset(new float[tree_leaves_data_length_]);
+
+  for (int i = 0; i < kChunksAtStartupLeftToDelete; ++i) {
+    previous_results_.push_back(0.f);
+  }
+}
+
+TransientDetector::~TransientDetector() {}
+
+float TransientDetector::Detect(const float* data,
+                                size_t data_length,
+                                const float* reference_data,
+                                size_t reference_length) {
+  RTC_DCHECK(data);
+  RTC_DCHECK_EQ(samples_per_chunk_, data_length);
+
+  // TODO(aluebs): Check if these errors can logically happen and if not assert
+  // on them.
+  if (wpd_tree_->Update(data, samples_per_chunk_) != 0) {
+    return -1.f;
+  }
+
+  float result = 0.f;
+
+  for (size_t i = 0; i < kLeaves; ++i) {
+    WPDNode* leaf = wpd_tree_->NodeAt(kLevels, i);
+
+    moving_moments_[i]->CalculateMoments(leaf->data(), tree_leaves_data_length_,
+                                         first_moments_.get(),
+                                         second_moments_.get());
+
+    // Add value delayed (Use the last moments from the last call to Detect).
+    float unbiased_data = leaf->data()[0] - last_first_moment_[i];
+    result +=
+        unbiased_data * unbiased_data / (last_second_moment_[i] + FLT_MIN);
+
+    // Add new values.
+    for (size_t j = 1; j < tree_leaves_data_length_; ++j) {
+      unbiased_data = leaf->data()[j] - first_moments_[j - 1];
+      result +=
+          unbiased_data * unbiased_data / (second_moments_[j - 1] + FLT_MIN);
+    }
+
+    last_first_moment_[i] = first_moments_[tree_leaves_data_length_ - 1];
+    last_second_moment_[i] = second_moments_[tree_leaves_data_length_ - 1];
+  }
+
+  result /= tree_leaves_data_length_;
+
+  result *= ReferenceDetectionValue(reference_data, reference_length);
+
+  if (chunks_at_startup_left_to_delete_ > 0) {
+    chunks_at_startup_left_to_delete_--;
+    result = 0.f;
+  }
+
+  if (result >= kDetectThreshold) {
+    result = 1.f;
+  } else {
+    // Get proportional value.
+    // Proportion achieved with a squared raised cosine function with domain
+    // [0, kDetectThreshold) and image [0, 1), it's always increasing.
+    const float horizontal_scaling = ts::kPi / kDetectThreshold;
+    const float kHorizontalShift = ts::kPi;
+    const float kVerticalScaling = 0.5f;
+    const float kVerticalShift = 1.f;
+
+    result = (std::cos(result * horizontal_scaling + kHorizontalShift) +
+              kVerticalShift) *
+             kVerticalScaling;
+    result *= result;
+  }
+
+  previous_results_.pop_front();
+  previous_results_.push_back(result);
+
+  // In the current implementation we return the max of the current result and
+  // the previous results, so the high results have a width equals to
+  // `transient_length`.
+  return *std::max_element(previous_results_.begin(), previous_results_.end());
+}
+
+// Looks for the highest slope and compares it with the previous ones.
+// An exponential transformation takes this to the [0, 1] range. This value is
+// multiplied by the detection result to avoid false positives.
+float TransientDetector::ReferenceDetectionValue(const float* data,
+                                                 size_t length) {
+  if (data == NULL) {
+    using_reference_ = false;
+    return 1.f;
+  }
+  static const float kEnergyRatioThreshold = 0.2f;
+  static const float kReferenceNonLinearity = 20.f;
+  static const float kMemory = 0.99f;
+  float reference_energy = 0.f;
+  for (size_t i = 1; i < length; ++i) {
+    reference_energy += data[i] * data[i];
+  }
+  if (reference_energy == 0.f) {
+    using_reference_ = false;
+    return 1.f;
+  }
+  RTC_DCHECK_NE(0, reference_energy_);
+  float result = 1.f / (1.f + std::exp(kReferenceNonLinearity *
+                                       (kEnergyRatioThreshold -
+                                        reference_energy / reference_energy_)));
+  reference_energy_ =
+      kMemory * reference_energy_ + (1.f - kMemory) * reference_energy;
+
+  using_reference_ = true;
+
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_detector.h b/third_party/libwebrtc/modules/audio_processing/transient/transient_detector.h
new file mode 100644
index 0000000000..a3dbb7ffde
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_detector.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_DETECTOR_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_DETECTOR_H_
+
+#include <stddef.h>
+
+#include <deque>
+#include <memory>
+
+#include "modules/audio_processing/transient/moving_moments.h"
+#include "modules/audio_processing/transient/wpd_tree.h"
+
+namespace webrtc {
+
+// This is an implementation of the transient detector described in "Causal
+// Wavelet based transient detector".
+// Calculates the log-likelihood of a transient to happen on a signal at any
+// given time based on the previous samples; it uses a WPD tree to analyze the
+// signal.  It preserves its state, so it can be multiple-called.
+class TransientDetector {
+ public:
+  // TODO(chadan): The only supported wavelet is Daubechies 8 using a WPD tree
+  // of 3 levels. Make an overloaded constructor to allow different wavelets and
+  // depths of the tree. When needed.
+
+  // Creates a wavelet based transient detector.
+  TransientDetector(int sample_rate_hz);
+
+  ~TransientDetector();
+
+  // Calculates the log-likelihood of the existence of a transient in `data`.
+  // `data_length` has to be equal to `samples_per_chunk_`.
+  // Returns a value between 0 and 1, as a non linear representation of this
+  // likelihood.
+  // Returns a negative value on error.
+  float Detect(const float* data,
+               size_t data_length,
+               const float* reference_data,
+               size_t reference_length);
+
+  bool using_reference() { return using_reference_; }
+
+ private:
+  float ReferenceDetectionValue(const float* data, size_t length);
+
+  static const size_t kLevels = 3;
+  static const size_t kLeaves = 1 << kLevels;
+
+  size_t samples_per_chunk_;
+
+  std::unique_ptr<WPDTree> wpd_tree_;
+  size_t tree_leaves_data_length_;
+
+  // A MovingMoments object is needed for each leaf in the WPD tree.
+  std::unique_ptr<MovingMoments> moving_moments_[kLeaves];
+
+  std::unique_ptr<float[]> first_moments_;
+  std::unique_ptr<float[]> second_moments_;
+
+  // Stores the last calculated moments from the previous detection.
+  float last_first_moment_[kLeaves];
+  float last_second_moment_[kLeaves];
+
+  // We keep track of the previous results from the previous chunks, so it can
+  // be used to effectively give results according to the `transient_length`.
+  std::deque<float> previous_results_;
+
+  // Number of chunks that are going to return only zeros at the beginning of
+  // the detection. It helps to avoid infs and nans due to the lack of
+  // information.
+  int chunks_at_startup_left_to_delete_;
+
+  float reference_energy_;
+
+  bool using_reference_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_detector_unittest.cc b/third_party/libwebrtc/modules/audio_processing/transient/transient_detector_unittest.cc
new file mode 100644
index 0000000000..a7364626fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_detector_unittest.cc
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/transient_detector.h"
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_processing/transient/common.h"
+#include "modules/audio_processing/transient/file_utils.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+static const int kSampleRatesHz[] = {ts::kSampleRate8kHz, ts::kSampleRate16kHz,
+                                     ts::kSampleRate32kHz,
+                                     ts::kSampleRate48kHz};
+static const size_t kNumberOfSampleRates =
+    sizeof(kSampleRatesHz) / sizeof(*kSampleRatesHz);
+
+// This test is for the correctness of the transient detector.
+// Checks the results comparing them with the ones stored in the detect files in
+// the directory: resources/audio_processing/transient/
+// The files contain all the results in double precision (Little endian).
+// The audio files used with different sample rates are stored in the same
+// directory.
+#if defined(WEBRTC_IOS)
+TEST(TransientDetectorTest, DISABLED_CorrectnessBasedOnFiles) {
+#else
+TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
+#endif
+  for (size_t i = 0; i < kNumberOfSampleRates; ++i) {
+    int sample_rate_hz = kSampleRatesHz[i];
+
+    // Prepare detect file.
+    rtc::StringBuilder detect_file_name;
+    detect_file_name << "audio_processing/transient/detect"
+                     << (sample_rate_hz / 1000) << "kHz";
+
+    FileWrapper detect_file = FileWrapper::OpenReadOnly(
+        test::ResourcePath(detect_file_name.str(), "dat"));
+
+    bool file_opened = detect_file.is_open();
+    ASSERT_TRUE(file_opened) << "File could not be opened.\n"
+                             << detect_file_name.str().c_str();
+
+    // Prepare audio file.
+    rtc::StringBuilder audio_file_name;
+    audio_file_name << "audio_processing/transient/audio"
+                    << (sample_rate_hz / 1000) << "kHz";
+
+    FileWrapper audio_file = FileWrapper::OpenReadOnly(
+        test::ResourcePath(audio_file_name.str(), "pcm"));
+
+    // Create detector.
+    TransientDetector detector(sample_rate_hz);
+
+    const size_t buffer_length = sample_rate_hz * ts::kChunkSizeMs / 1000;
+    std::unique_ptr<float[]> buffer(new float[buffer_length]);
+
+    const float kTolerance = 0.02f;
+
+    size_t frames_read = 0;
+
+    while (ReadInt16FromFileToFloatBuffer(&audio_file, buffer_length,
+                                          buffer.get()) == buffer_length) {
+      ++frames_read;
+
+      float detector_value =
+          detector.Detect(buffer.get(), buffer_length, NULL, 0);
+      double file_value;
+      ASSERT_EQ(1u, ReadDoubleBufferFromFile(&detect_file, 1, &file_value))
+          << "Detect test file is malformed.\n";
+
+      // Compare results with data from the matlab test file.
+      EXPECT_NEAR(file_value, detector_value, kTolerance)
+          << "Frame: " << frames_read;
+    }
+
+    detect_file.Close();
+    audio_file.Close();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppression_test.cc b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppression_test.cc
new file mode 100644
index 0000000000..2d8baf9416
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppression_test.cc
@@ -0,0 +1,238 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc/agc.h"
+#include "modules/audio_processing/transient/transient_suppressor.h"
+#include "modules/audio_processing/transient/transient_suppressor_impl.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(std::string, in_file_name, "", "PCM file that contains the signal.");
+ABSL_FLAG(std::string,
+          detection_file_name,
+          "",
+          "PCM file that contains the detection signal.");
+ABSL_FLAG(std::string,
+          reference_file_name,
+          "",
+          "PCM file that contains the reference signal.");
+
+ABSL_FLAG(int,
+          chunk_size_ms,
+          10,
+          "Time between each chunk of samples in milliseconds.");
+
+ABSL_FLAG(int,
+          sample_rate_hz,
+          16000,
+          "Sampling frequency of the signal in Hertz.");
+ABSL_FLAG(int,
+          detection_rate_hz,
+          0,
+          "Sampling frequency of the detection signal in Hertz.");
+
+ABSL_FLAG(int, num_channels, 1, "Number of channels.");
+
+namespace webrtc {
+
+const char kUsage[] =
+    "\nDetects and suppresses transients from file.\n\n"
+    "This application loads the signal from the in_file_name with a specific\n"
+    "num_channels and sample_rate_hz, the detection signal from the\n"
+    "detection_file_name with a specific detection_rate_hz, and the reference\n"
+    "signal from the reference_file_name with sample_rate_hz, divides them\n"
+    "into chunk_size_ms blocks, computes its voice value and depending on the\n"
+    "voice_threshold does the respective restoration. You can always get the\n"
+    "all-voiced or all-unvoiced cases by setting the voice_threshold to 0 or\n"
+    "1 respectively.\n\n";
+
+// Read next buffers from the test files (signed 16-bit host-endian PCM
+// format). audio_buffer has int16 samples, detection_buffer has float samples
+// with range [-32768,32767], and reference_buffer has float samples with range
+// [-1,1]. Return true iff all the buffers were filled completely.
+bool ReadBuffers(FILE* in_file,
+                 size_t audio_buffer_size,
+                 int num_channels,
+                 int16_t* audio_buffer,
+                 FILE* detection_file,
+                 size_t detection_buffer_size,
+                 float* detection_buffer,
+                 FILE* reference_file,
+                 float* reference_buffer) {
+  std::unique_ptr<int16_t[]> tmpbuf;
+  int16_t* read_ptr = audio_buffer;
+  if (num_channels > 1) {
+    tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]);
+    read_ptr = tmpbuf.get();
+  }
+  if (fread(read_ptr, sizeof(*read_ptr), num_channels * audio_buffer_size,
+            in_file) != num_channels * audio_buffer_size) {
+    return false;
+  }
+  // De-interleave.
+  if (num_channels > 1) {
+    for (int i = 0; i < num_channels; ++i) {
+      for (size_t j = 0; j < audio_buffer_size; ++j) {
+        audio_buffer[i * audio_buffer_size + j] =
+            read_ptr[i + j * num_channels];
+      }
+    }
+  }
+  if (detection_file) {
+    std::unique_ptr<int16_t[]> ibuf(new int16_t[detection_buffer_size]);
+    if (fread(ibuf.get(), sizeof(ibuf[0]), detection_buffer_size,
+              detection_file) != detection_buffer_size)
+      return false;
+    for (size_t i = 0; i < detection_buffer_size; ++i)
+      detection_buffer[i] = ibuf[i];
+  }
+  if (reference_file) {
+    std::unique_ptr<int16_t[]> ibuf(new int16_t[audio_buffer_size]);
+    if (fread(ibuf.get(), sizeof(ibuf[0]), audio_buffer_size, reference_file) !=
+        audio_buffer_size)
+      return false;
+    S16ToFloat(ibuf.get(), audio_buffer_size, reference_buffer);
+  }
+  return true;
+}
+
+// Write a number of samples to an open signed 16-bit host-endian PCM file.
+static void WritePCM(FILE* f,
+                     size_t num_samples,
+                     int num_channels,
+                     const float* buffer) {
+  std::unique_ptr<int16_t[]> ibuf(new int16_t[num_channels * num_samples]);
+  // Interleave.
+  for (int i = 0; i < num_channels; ++i) {
+    for (size_t j = 0; j < num_samples; ++j) {
+      ibuf[i + j * num_channels] = FloatS16ToS16(buffer[i * num_samples + j]);
+    }
+  }
+  fwrite(ibuf.get(), sizeof(ibuf[0]), num_channels * num_samples, f);
+}
+
+// This application tests the transient suppression by providing a processed
+// PCM file, which has to be listened to in order to evaluate the
+// performance.
+// It gets an audio file, and its voice gain information, and the suppressor
+// process it giving the output file "suppressed_keystrokes.pcm".
+void void_main() {
+  // TODO(aluebs): Remove all FileWrappers.
+  // Prepare the input file.
+  FILE* in_file = fopen(absl::GetFlag(FLAGS_in_file_name).c_str(), "rb");
+  ASSERT_TRUE(in_file != NULL);
+
+  // Prepare the detection file.
+  FILE* detection_file = NULL;
+  if (!absl::GetFlag(FLAGS_detection_file_name).empty()) {
+    detection_file =
+        fopen(absl::GetFlag(FLAGS_detection_file_name).c_str(), "rb");
+  }
+
+  // Prepare the reference file.
+  FILE* reference_file = NULL;
+  if (!absl::GetFlag(FLAGS_reference_file_name).empty()) {
+    reference_file =
+        fopen(absl::GetFlag(FLAGS_reference_file_name).c_str(), "rb");
+  }
+
+  // Prepare the output file.
+  std::string out_file_name = test::OutputPath() + "suppressed_keystrokes.pcm";
+  FILE* out_file = fopen(out_file_name.c_str(), "wb");
+  ASSERT_TRUE(out_file != NULL);
+
+  int detection_rate_hz = absl::GetFlag(FLAGS_detection_rate_hz);
+  if (detection_rate_hz == 0) {
+    detection_rate_hz = absl::GetFlag(FLAGS_sample_rate_hz);
+  }
+
+  Agc agc;
+
+  TransientSuppressorImpl suppressor(TransientSuppressor::VadMode::kDefault,
+                                     absl::GetFlag(FLAGS_sample_rate_hz),
+                                     detection_rate_hz,
+                                     absl::GetFlag(FLAGS_num_channels));
+
+  const size_t audio_buffer_size = absl::GetFlag(FLAGS_chunk_size_ms) *
+                                   absl::GetFlag(FLAGS_sample_rate_hz) / 1000;
+  const size_t detection_buffer_size =
+      absl::GetFlag(FLAGS_chunk_size_ms) * detection_rate_hz / 1000;
+
+  // int16 and float variants of the same data.
+  std::unique_ptr<int16_t[]> audio_buffer_i(
+      new int16_t[absl::GetFlag(FLAGS_num_channels) * audio_buffer_size]);
+  std::unique_ptr<float[]> audio_buffer_f(
+      new float[absl::GetFlag(FLAGS_num_channels) * audio_buffer_size]);
+
+  std::unique_ptr<float[]> detection_buffer, reference_buffer;
+
+  if (detection_file)
+    detection_buffer.reset(new float[detection_buffer_size]);
+  if (reference_file)
+    reference_buffer.reset(new float[audio_buffer_size]);
+
+  while (ReadBuffers(
+      in_file, audio_buffer_size, absl::GetFlag(FLAGS_num_channels),
+      audio_buffer_i.get(), detection_file, detection_buffer_size,
+      detection_buffer.get(), reference_file, reference_buffer.get())) {
+    agc.Process({audio_buffer_i.get(), audio_buffer_size});
+
+    for (size_t i = 0;
+         i < absl::GetFlag(FLAGS_num_channels) * audio_buffer_size; ++i) {
+      audio_buffer_f[i] = audio_buffer_i[i];
+    }
+
+    suppressor.Suppress(audio_buffer_f.get(), audio_buffer_size,
+                        absl::GetFlag(FLAGS_num_channels),
+                        detection_buffer.get(), detection_buffer_size,
+                        reference_buffer.get(), audio_buffer_size,
+                        agc.voice_probability(), true);
+
+    // Write result to out file.
+    WritePCM(out_file, audio_buffer_size, absl::GetFlag(FLAGS_num_channels),
+             audio_buffer_f.get());
+  }
+
+  fclose(in_file);
+  if (detection_file) {
+    fclose(detection_file);
+  }
+  if (reference_file) {
+    fclose(reference_file);
+  }
+  fclose(out_file);
+}
+
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  if (args.size() != 1) {
+    printf("%s", webrtc::kUsage);
+    return 1;
+  }
+  RTC_CHECK_GT(absl::GetFlag(FLAGS_chunk_size_ms), 0);
+  RTC_CHECK_GT(absl::GetFlag(FLAGS_sample_rate_hz), 0);
+  RTC_CHECK_GT(absl::GetFlag(FLAGS_num_channels), 0);
+
+  webrtc::void_main();
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor.h b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor.h
new file mode 100644
index 0000000000..ecb3c3baab
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
+
+#include <cstddef>
+
+namespace webrtc {
+
+// Detects transients in an audio stream and suppress them using a simple
+// restoration algorithm that attenuates unexpected spikes in the spectrum.
+class TransientSuppressor {
+ public:
+  // Type of VAD used by the caller to compute the `voice_probability` argument
+  // `Suppress()`.
+  enum class VadMode {
+    // By default, `TransientSuppressor` assumes that `voice_probability` is
+    // computed by `AgcManagerDirect`.
+    kDefault = 0,
+    // Use this mode when `TransientSuppressor` must assume that
+    // `voice_probability` is computed by the RNN VAD.
+    kRnnVad,
+    // Use this mode to let `TransientSuppressor::Suppressor()` ignore
+    // `voice_probability` and behave as if voice information is unavailable
+    // (regardless of the passed value).
+    kNoVad,
+  };
+
+  virtual ~TransientSuppressor() {}
+
+  virtual void Initialize(int sample_rate_hz,
+                          int detector_rate_hz,
+                          int num_channels) = 0;
+
+  // Processes a `data` chunk, and returns it with keystrokes suppressed from
+  // it. The float format is assumed to be int16 ranged. If there are more than
+  // one channel, the chunks are concatenated one after the other in `data`.
+  // `data_length` must be equal to `data_length_`.
+  // `num_channels` must be equal to `num_channels_`.
+  // A sub-band, ideally the higher, can be used as `detection_data`. If it is
+  // NULL, `data` is used for the detection too. The `detection_data` is always
+  // assumed mono.
+  // If a reference signal (e.g. keyboard microphone) is available, it can be
+  // passed in as `reference_data`. It is assumed mono and must have the same
+  // length as `data`. NULL is accepted if unavailable.
+  // This suppressor performs better if voice information is available.
+  // `voice_probability` is the probability of voice being present in this chunk
+  // of audio. If voice information is not available, `voice_probability` must
+  // always be set to 1.
+  // `key_pressed` determines if a key was pressed on this audio chunk.
+  // Returns a delayed version of `voice_probability` according to the
+  // algorithmic delay introduced by this method. In this way, the modified
+  // `data` and the returned voice probability will be temporally aligned.
+  virtual float Suppress(float* data,
+                         size_t data_length,
+                         int num_channels,
+                         const float* detection_data,
+                         size_t detection_length,
+                         const float* reference_data,
+                         size_t reference_length,
+                         float voice_probability,
+                         bool key_pressed) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_api_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_api_gn/moz.build
new file mode 100644
index 0000000000..5988e89a6a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_api_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("transient_suppressor_api_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl.cc b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl.cc
new file mode 100644
index 0000000000..90428464e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl.cc
@@ -0,0 +1,455 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/transient_suppressor_impl.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cmath>
+#include <complex>
+#include <deque>
+#include <limits>
+#include <set>
+#include <string>
+
+#include "common_audio/include/audio_util.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/third_party/ooura/fft_size_256/fft4g.h"
+#include "modules/audio_processing/transient/common.h"
+#include "modules/audio_processing/transient/transient_detector.h"
+#include "modules/audio_processing/transient/transient_suppressor.h"
+#include "modules/audio_processing/transient/windows_private.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+static const float kMeanIIRCoefficient = 0.5f;
+
+// TODO(aluebs): Check if these values work also for 48kHz.
+static const size_t kMinVoiceBin = 3;
+static const size_t kMaxVoiceBin = 60;
+
+namespace {
+
+float ComplexMagnitude(float a, float b) {
+  return std::abs(a) + std::abs(b);
+}
+
+std::string GetVadModeLabel(TransientSuppressor::VadMode vad_mode) {
+  switch (vad_mode) {
+    case TransientSuppressor::VadMode::kDefault:
+      return "default";
+    case TransientSuppressor::VadMode::kRnnVad:
+      return "RNN VAD";
+    case TransientSuppressor::VadMode::kNoVad:
+      return "no VAD";
+  }
+}
+
+}  // namespace
+
+TransientSuppressorImpl::TransientSuppressorImpl(VadMode vad_mode,
+                                                 int sample_rate_hz,
+                                                 int detector_rate_hz,
+                                                 int num_channels)
+    : vad_mode_(vad_mode),
+      voice_probability_delay_unit_(/*delay_num_samples=*/0, sample_rate_hz),
+      analyzed_audio_is_silent_(false),
+      data_length_(0),
+      detection_length_(0),
+      analysis_length_(0),
+      buffer_delay_(0),
+      complex_analysis_length_(0),
+      num_channels_(0),
+      window_(NULL),
+      detector_smoothed_(0.f),
+      keypress_counter_(0),
+      chunks_since_keypress_(0),
+      detection_enabled_(false),
+      suppression_enabled_(false),
+      use_hard_restoration_(false),
+      chunks_since_voice_change_(0),
+      seed_(182),
+      using_reference_(false) {
+  RTC_LOG(LS_INFO) << "VAD mode: " << GetVadModeLabel(vad_mode_);
+  Initialize(sample_rate_hz, detector_rate_hz, num_channels);
+}
+
+TransientSuppressorImpl::~TransientSuppressorImpl() {}
+
+void TransientSuppressorImpl::Initialize(int sample_rate_hz,
+                                         int detection_rate_hz,
+                                         int num_channels) {
+  RTC_DCHECK(sample_rate_hz == ts::kSampleRate8kHz ||
+             sample_rate_hz == ts::kSampleRate16kHz ||
+             sample_rate_hz == ts::kSampleRate32kHz ||
+             sample_rate_hz == ts::kSampleRate48kHz);
+  RTC_DCHECK(detection_rate_hz == ts::kSampleRate8kHz ||
+             detection_rate_hz == ts::kSampleRate16kHz ||
+             detection_rate_hz == ts::kSampleRate32kHz ||
+             detection_rate_hz == ts::kSampleRate48kHz);
+  RTC_DCHECK_GT(num_channels, 0);
+
+  switch (sample_rate_hz) {
+    case ts::kSampleRate8kHz:
+      analysis_length_ = 128u;
+      window_ = kBlocks80w128;
+      break;
+    case ts::kSampleRate16kHz:
+      analysis_length_ = 256u;
+      window_ = kBlocks160w256;
+      break;
+    case ts::kSampleRate32kHz:
+      analysis_length_ = 512u;
+      window_ = kBlocks320w512;
+      break;
+    case ts::kSampleRate48kHz:
+      analysis_length_ = 1024u;
+      window_ = kBlocks480w1024;
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return;
+  }
+
+  detector_.reset(new TransientDetector(detection_rate_hz));
+  data_length_ = sample_rate_hz * ts::kChunkSizeMs / 1000;
+  RTC_DCHECK_LE(data_length_, analysis_length_);
+  buffer_delay_ = analysis_length_ - data_length_;
+
+  voice_probability_delay_unit_.Initialize(/*delay_num_samples=*/buffer_delay_,
+                                           sample_rate_hz);
+
+  complex_analysis_length_ = analysis_length_ / 2 + 1;
+  RTC_DCHECK_GE(complex_analysis_length_, kMaxVoiceBin);
+  num_channels_ = num_channels;
+  in_buffer_.reset(new float[analysis_length_ * num_channels_]);
+  memset(in_buffer_.get(), 0,
+         analysis_length_ * num_channels_ * sizeof(in_buffer_[0]));
+  detection_length_ = detection_rate_hz * ts::kChunkSizeMs / 1000;
+  detection_buffer_.reset(new float[detection_length_]);
+  memset(detection_buffer_.get(), 0,
+         detection_length_ * sizeof(detection_buffer_[0]));
+  out_buffer_.reset(new float[analysis_length_ * num_channels_]);
+  memset(out_buffer_.get(), 0,
+         analysis_length_ * num_channels_ * sizeof(out_buffer_[0]));
+  // ip[0] must be zero to trigger initialization using rdft().
+  size_t ip_length = 2 + sqrtf(analysis_length_);
+  ip_.reset(new size_t[ip_length]());
+  memset(ip_.get(), 0, ip_length * sizeof(ip_[0]));
+  wfft_.reset(new float[complex_analysis_length_ - 1]);
+  memset(wfft_.get(), 0, (complex_analysis_length_ - 1) * sizeof(wfft_[0]));
+  spectral_mean_.reset(new float[complex_analysis_length_ * num_channels_]);
+  memset(spectral_mean_.get(), 0,
+         complex_analysis_length_ * num_channels_ * sizeof(spectral_mean_[0]));
+  fft_buffer_.reset(new float[analysis_length_ + 2]);
+  memset(fft_buffer_.get(), 0, (analysis_length_ + 2) * sizeof(fft_buffer_[0]));
+  magnitudes_.reset(new float[complex_analysis_length_]);
+  memset(magnitudes_.get(), 0,
+         complex_analysis_length_ * sizeof(magnitudes_[0]));
+  mean_factor_.reset(new float[complex_analysis_length_]);
+
+  static const float kFactorHeight = 10.f;
+  static const float kLowSlope = 1.f;
+  static const float kHighSlope = 0.3f;
+  for (size_t i = 0; i < complex_analysis_length_; ++i) {
+    mean_factor_[i] =
+        kFactorHeight /
+            (1.f + std::exp(kLowSlope * static_cast<int>(i - kMinVoiceBin))) +
+        kFactorHeight /
+            (1.f + std::exp(kHighSlope * static_cast<int>(kMaxVoiceBin - i)));
+  }
+  detector_smoothed_ = 0.f;
+  keypress_counter_ = 0;
+  chunks_since_keypress_ = 0;
+  detection_enabled_ = false;
+  suppression_enabled_ = false;
+  use_hard_restoration_ = false;
+  chunks_since_voice_change_ = 0;
+  seed_ = 182;
+  using_reference_ = false;
+}
+
+float TransientSuppressorImpl::Suppress(float* data,
+                                        size_t data_length,
+                                        int num_channels,
+                                        const float* detection_data,
+                                        size_t detection_length,
+                                        const float* reference_data,
+                                        size_t reference_length,
+                                        float voice_probability,
+                                        bool key_pressed) {
+  if (!data || data_length != data_length_ || num_channels != num_channels_ ||
+      detection_length != detection_length_ || voice_probability < 0 ||
+      voice_probability > 1) {
+    // The audio is not modified, so the voice probability is returned as is
+    // (delay not applied).
+    return voice_probability;
+  }
+
+  UpdateKeypress(key_pressed);
+  UpdateBuffers(data);
+
+  if (detection_enabled_) {
+    UpdateRestoration(voice_probability);
+
+    if (!detection_data) {
+      // Use the input data  of the first channel if special detection data is
+      // not supplied.
+      detection_data = &in_buffer_[buffer_delay_];
+    }
+
+    float detector_result = detector_->Detect(detection_data, detection_length,
+                                              reference_data, reference_length);
+    if (detector_result < 0) {
+      // The audio is not modified, so the voice probability is returned as is
+      // (delay not applied).
+      return voice_probability;
+    }
+
+    using_reference_ = detector_->using_reference();
+
+    // `detector_smoothed_` follows the `detector_result` when this last one is
+    // increasing, but has an exponential decaying tail to be able to suppress
+    // the ringing of keyclicks.
+    float smooth_factor = using_reference_ ? 0.6 : 0.1;
+    detector_smoothed_ = detector_result >= detector_smoothed_
+                             ? detector_result
+                             : smooth_factor * detector_smoothed_ +
+                                   (1 - smooth_factor) * detector_result;
+
+    for (int i = 0; i < num_channels_; ++i) {
+      Suppress(&in_buffer_[i * analysis_length_],
+               &spectral_mean_[i * complex_analysis_length_],
+               &out_buffer_[i * analysis_length_]);
+    }
+  }
+
+  // If the suppression isn't enabled, we use the in buffer to delay the signal
+  // appropriately. This also gives time for the out buffer to be refreshed with
+  // new data between detection and suppression getting enabled.
+  for (int i = 0; i < num_channels_; ++i) {
+    memcpy(&data[i * data_length_],
+           suppression_enabled_ ? &out_buffer_[i * analysis_length_]
+                                : &in_buffer_[i * analysis_length_],
+           data_length_ * sizeof(*data));
+  }
+
+  // The audio has been modified, return the delayed voice probability.
+  return voice_probability_delay_unit_.Delay(voice_probability);
+}
+
+// This should only be called when detection is enabled. UpdateBuffers() must
+// have been called. At return, `out_buffer_` will be filled with the
+// processed output.
+void TransientSuppressorImpl::Suppress(float* in_ptr,
+                                       float* spectral_mean,
+                                       float* out_ptr) {
+  // Go to frequency domain.
+  for (size_t i = 0; i < analysis_length_; ++i) {
+    // TODO(aluebs): Rename windows
+    fft_buffer_[i] = in_ptr[i] * window_[i];
+  }
+
+  WebRtc_rdft(analysis_length_, 1, fft_buffer_.get(), ip_.get(), wfft_.get());
+
+  // Since WebRtc_rdft puts R[n/2] in fft_buffer_[1], we move it to the end
+  // for convenience.
+  fft_buffer_[analysis_length_] = fft_buffer_[1];
+  fft_buffer_[analysis_length_ + 1] = 0.f;
+  fft_buffer_[1] = 0.f;
+
+  for (size_t i = 0; i < complex_analysis_length_; ++i) {
+    magnitudes_[i] =
+        ComplexMagnitude(fft_buffer_[i * 2], fft_buffer_[i * 2 + 1]);
+  }
+  // Restore audio if necessary.
+  if (suppression_enabled_) {
+    if (use_hard_restoration_) {
+      HardRestoration(spectral_mean);
+    } else {
+      SoftRestoration(spectral_mean);
+    }
+  }
+
+  // Update the spectral mean.
+  for (size_t i = 0; i < complex_analysis_length_; ++i) {
+    spectral_mean[i] = (1 - kMeanIIRCoefficient) * spectral_mean[i] +
+                       kMeanIIRCoefficient * magnitudes_[i];
+  }
+
+  // Back to time domain.
+  // Put R[n/2] back in fft_buffer_[1].
+  fft_buffer_[1] = fft_buffer_[analysis_length_];
+
+  WebRtc_rdft(analysis_length_, -1, fft_buffer_.get(), ip_.get(), wfft_.get());
+  const float fft_scaling = 2.f / analysis_length_;
+
+  for (size_t i = 0; i < analysis_length_; ++i) {
+    out_ptr[i] += fft_buffer_[i] * window_[i] * fft_scaling;
+  }
+}
+
+void TransientSuppressorImpl::UpdateKeypress(bool key_pressed) {
+  const int kKeypressPenalty = 1000 / ts::kChunkSizeMs;
+  const int kIsTypingThreshold = 1000 / ts::kChunkSizeMs;
+  const int kChunksUntilNotTyping = 4000 / ts::kChunkSizeMs;  // 4 seconds.
+
+  if (key_pressed) {
+    keypress_counter_ += kKeypressPenalty;
+    chunks_since_keypress_ = 0;
+    detection_enabled_ = true;
+  }
+  keypress_counter_ = std::max(0, keypress_counter_ - 1);
+
+  if (keypress_counter_ > kIsTypingThreshold) {
+    if (!suppression_enabled_) {
+      RTC_LOG(LS_INFO) << "[ts] Transient suppression is now enabled.";
+    }
+    suppression_enabled_ = true;
+    keypress_counter_ = 0;
+  }
+
+  if (detection_enabled_ && ++chunks_since_keypress_ > kChunksUntilNotTyping) {
+    if (suppression_enabled_) {
+      RTC_LOG(LS_INFO) << "[ts] Transient suppression is now disabled.";
+    }
+    detection_enabled_ = false;
+    suppression_enabled_ = false;
+    keypress_counter_ = 0;
+  }
+}
+
+void TransientSuppressorImpl::UpdateRestoration(float voice_probability) {
+  bool not_voiced;
+  switch (vad_mode_) {
+    case TransientSuppressor::VadMode::kDefault: {
+      constexpr float kVoiceThreshold = 0.02f;
+      not_voiced = voice_probability < kVoiceThreshold;
+      break;
+    }
+    case TransientSuppressor::VadMode::kRnnVad: {
+      constexpr float kVoiceThreshold = 0.7f;
+      not_voiced = voice_probability < kVoiceThreshold;
+      break;
+    }
+    case TransientSuppressor::VadMode::kNoVad:
+      // Always assume that voice is detected.
+      not_voiced = false;
+      break;
+  }
+
+  if (not_voiced == use_hard_restoration_) {
+    chunks_since_voice_change_ = 0;
+  } else {
+    ++chunks_since_voice_change_;
+
+    // Number of 10 ms frames to wait to transition to and from hard
+    // restoration.
+    constexpr int kHardRestorationOffsetDelay = 3;
+    constexpr int kHardRestorationOnsetDelay = 80;
+
+    if ((use_hard_restoration_ &&
+         chunks_since_voice_change_ > kHardRestorationOffsetDelay) ||
+        (!use_hard_restoration_ &&
+         chunks_since_voice_change_ > kHardRestorationOnsetDelay)) {
+      use_hard_restoration_ = not_voiced;
+      chunks_since_voice_change_ = 0;
+    }
+  }
+}
+
+// Shift buffers to make way for new data. Must be called after
+// `detection_enabled_` is updated by UpdateKeypress().
+void TransientSuppressorImpl::UpdateBuffers(float* data) {
+  // TODO(aluebs): Change to ring buffer.
+  memmove(in_buffer_.get(), &in_buffer_[data_length_],
+          (buffer_delay_ + (num_channels_ - 1) * analysis_length_) *
+              sizeof(in_buffer_[0]));
+  // Copy new chunk to buffer.
+  for (int i = 0; i < num_channels_; ++i) {
+    memcpy(&in_buffer_[buffer_delay_ + i * analysis_length_],
+           &data[i * data_length_], data_length_ * sizeof(*data));
+  }
+  if (detection_enabled_) {
+    // Shift previous chunk in out buffer.
+    memmove(out_buffer_.get(), &out_buffer_[data_length_],
+            (buffer_delay_ + (num_channels_ - 1) * analysis_length_) *
+                sizeof(out_buffer_[0]));
+    // Initialize new chunk in out buffer.
+    for (int i = 0; i < num_channels_; ++i) {
+      memset(&out_buffer_[buffer_delay_ + i * analysis_length_], 0,
+             data_length_ * sizeof(out_buffer_[0]));
+    }
+  }
+}
+
+// Restores the unvoiced signal if a click is present.
+// Attenuates by a certain factor every peak in the `fft_buffer_` that exceeds
+// the spectral mean. The attenuation depends on `detector_smoothed_`.
+// If a restoration takes place, the `magnitudes_` are updated to the new value.
+void TransientSuppressorImpl::HardRestoration(float* spectral_mean) {
+  const float detector_result =
+      1.f - std::pow(1.f - detector_smoothed_, using_reference_ ? 200.f : 50.f);
+  // To restore, we get the peaks in the spectrum. If higher than the previous
+  // spectral mean we adjust them.
+  for (size_t i = 0; i < complex_analysis_length_; ++i) {
+    if (magnitudes_[i] > spectral_mean[i] && magnitudes_[i] > 0) {
+      // RandU() generates values on [0, int16::max()]
+      const float phase = 2 * ts::kPi * WebRtcSpl_RandU(&seed_) /
+                          std::numeric_limits<int16_t>::max();
+      const float scaled_mean = detector_result * spectral_mean[i];
+
+      fft_buffer_[i * 2] = (1 - detector_result) * fft_buffer_[i * 2] +
+                           scaled_mean * cosf(phase);
+      fft_buffer_[i * 2 + 1] = (1 - detector_result) * fft_buffer_[i * 2 + 1] +
+                               scaled_mean * sinf(phase);
+      magnitudes_[i] = magnitudes_[i] -
+                       detector_result * (magnitudes_[i] - spectral_mean[i]);
+    }
+  }
+}
+
+// Restores the voiced signal if a click is present.
+// Attenuates by a certain factor every peak in the `fft_buffer_` that exceeds
+// the spectral mean and that is lower than some function of the current block
+// frequency mean. The attenuation depends on `detector_smoothed_`.
+// If a restoration takes place, the `magnitudes_` are updated to the new value.
+void TransientSuppressorImpl::SoftRestoration(float* spectral_mean) {
+  // Get the spectral magnitude mean of the current block.
+  float block_frequency_mean = 0;
+  for (size_t i = kMinVoiceBin; i < kMaxVoiceBin; ++i) {
+    block_frequency_mean += magnitudes_[i];
+  }
+  block_frequency_mean /= (kMaxVoiceBin - kMinVoiceBin);
+
+  // To restore, we get the peaks in the spectrum. If higher than the
+  // previous spectral mean and lower than a factor of the block mean
+  // we adjust them. The factor is a double sigmoid that has a minimum in the
+  // voice frequency range (300Hz - 3kHz).
+  for (size_t i = 0; i < complex_analysis_length_; ++i) {
+    if (magnitudes_[i] > spectral_mean[i] && magnitudes_[i] > 0 &&
+        (using_reference_ ||
+         magnitudes_[i] < block_frequency_mean * mean_factor_[i])) {
+      const float new_magnitude =
+          magnitudes_[i] -
+          detector_smoothed_ * (magnitudes_[i] - spectral_mean[i]);
+      const float magnitude_ratio = new_magnitude / magnitudes_[i];
+
+      fft_buffer_[i * 2] *= magnitude_ratio;
+      fft_buffer_[i * 2 + 1] *= magnitude_ratio;
+      magnitudes_[i] = new_magnitude;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl.h b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl.h
new file mode 100644
index 0000000000..4005a16b0a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl.h
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_IMPL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "modules/audio_processing/transient/transient_suppressor.h"
+#include "modules/audio_processing/transient/voice_probability_delay_unit.h"
+#include "rtc_base/gtest_prod_util.h"
+
+namespace webrtc {
+
+class TransientDetector;
+
+// Detects transients in an audio stream and suppress them using a simple
+// restoration algorithm that attenuates unexpected spikes in the spectrum.
+class TransientSuppressorImpl : public TransientSuppressor {
+ public:
+  TransientSuppressorImpl(VadMode vad_mode,
+                          int sample_rate_hz,
+                          int detector_rate_hz,
+                          int num_channels);
+  ~TransientSuppressorImpl() override;
+
+  void Initialize(int sample_rate_hz,
+                  int detector_rate_hz,
+                  int num_channels) override;
+
+  float Suppress(float* data,
+                 size_t data_length,
+                 int num_channels,
+                 const float* detection_data,
+                 size_t detection_length,
+                 const float* reference_data,
+                 size_t reference_length,
+                 float voice_probability,
+                 bool key_pressed) override;
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(TransientSuppressorVadModeParametrization,
+                           TypingDetectionLogicWorksAsExpectedForMono);
+  void Suppress(float* in_ptr, float* spectral_mean, float* out_ptr);
+
+  void UpdateKeypress(bool key_pressed);
+  void UpdateRestoration(float voice_probability);
+
+  void UpdateBuffers(float* data);
+
+  void HardRestoration(float* spectral_mean);
+  void SoftRestoration(float* spectral_mean);
+
+  const VadMode vad_mode_;
+  VoiceProbabilityDelayUnit voice_probability_delay_unit_;
+
+  std::unique_ptr<TransientDetector> detector_;
+
+  bool analyzed_audio_is_silent_;
+
+  size_t data_length_;
+  size_t detection_length_;
+  size_t analysis_length_;
+  size_t buffer_delay_;
+  size_t complex_analysis_length_;
+  int num_channels_;
+  // Input buffer where the original samples are stored.
+  std::unique_ptr<float[]> in_buffer_;
+  std::unique_ptr<float[]> detection_buffer_;
+  // Output buffer where the restored samples are stored.
+  std::unique_ptr<float[]> out_buffer_;
+
+  // Arrays for fft.
+  std::unique_ptr<size_t[]> ip_;
+  std::unique_ptr<float[]> wfft_;
+
+  std::unique_ptr<float[]> spectral_mean_;
+
+  // Stores the data for the fft.
+  std::unique_ptr<float[]> fft_buffer_;
+
+  std::unique_ptr<float[]> magnitudes_;
+
+  const float* window_;
+
+  std::unique_ptr<float[]> mean_factor_;
+
+  float detector_smoothed_;
+
+  int keypress_counter_;
+  int chunks_since_keypress_;
+  bool detection_enabled_;
+  bool suppression_enabled_;
+
+  bool use_hard_restoration_;
+  int chunks_since_voice_change_;
+
+  uint32_t seed_;
+
+  bool using_reference_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl_gn/moz.build
new file mode 100644
index 0000000000..ee6b82a2f6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl_gn/moz.build
@@ -0,0 +1,236 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/transient/moving_moments.cc",
+    "/third_party/libwebrtc/modules/audio_processing/transient/transient_detector.cc",
+    "/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_impl.cc",
+    "/third_party/libwebrtc/modules/audio_processing/transient/wpd_node.cc",
+    "/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("transient_suppressor_impl_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_unittest.cc b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_unittest.cc
new file mode 100644
index 0000000000..ab48504af6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/transient_suppressor_unittest.cc
@@ -0,0 +1,175 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/transient_suppressor.h"
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/transient/common.h"
+#include "modules/audio_processing/transient/transient_suppressor_impl.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+constexpr int kMono = 1;
+
+// Returns the index of the first non-zero sample in `samples` or an unspecified
+// value if no value is zero.
+absl::optional<int> FindFirstNonZeroSample(const std::vector<float>& samples) {
+  for (size_t i = 0; i < samples.size(); ++i) {
+    if (samples[i] != 0.0f) {
+      return i;
+    }
+  }
+  return absl::nullopt;
+}
+
+}  // namespace
+
+class TransientSuppressorVadModeParametrization
+    : public ::testing::TestWithParam<TransientSuppressor::VadMode> {};
+
+TEST_P(TransientSuppressorVadModeParametrization,
+       TypingDetectionLogicWorksAsExpectedForMono) {
+  TransientSuppressorImpl ts(GetParam(), ts::kSampleRate16kHz,
+                             ts::kSampleRate16kHz, kMono);
+
+  // Each key-press enables detection.
+  EXPECT_FALSE(ts.detection_enabled_);
+  ts.UpdateKeypress(true);
+  EXPECT_TRUE(ts.detection_enabled_);
+
+  // It takes four seconds without any key-press to disable the detection
+  for (int time_ms = 0; time_ms < 3990; time_ms += ts::kChunkSizeMs) {
+    ts.UpdateKeypress(false);
+    EXPECT_TRUE(ts.detection_enabled_);
+  }
+  ts.UpdateKeypress(false);
+  EXPECT_FALSE(ts.detection_enabled_);
+
+  // Key-presses that are more than a second apart from each other don't enable
+  // suppression.
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_FALSE(ts.suppression_enabled_);
+    ts.UpdateKeypress(true);
+    EXPECT_TRUE(ts.detection_enabled_);
+    EXPECT_FALSE(ts.suppression_enabled_);
+    for (int time_ms = 0; time_ms < 990; time_ms += ts::kChunkSizeMs) {
+      ts.UpdateKeypress(false);
+      EXPECT_TRUE(ts.detection_enabled_);
+      EXPECT_FALSE(ts.suppression_enabled_);
+    }
+    ts.UpdateKeypress(false);
+  }
+
+  // Two consecutive key-presses is enough to enable the suppression.
+  ts.UpdateKeypress(true);
+  EXPECT_FALSE(ts.suppression_enabled_);
+  ts.UpdateKeypress(true);
+  EXPECT_TRUE(ts.suppression_enabled_);
+
+  // Key-presses that are less than a second apart from each other don't disable
+  // detection nor suppression.
+  for (int i = 0; i < 100; ++i) {
+    for (int time_ms = 0; time_ms < 1000; time_ms += ts::kChunkSizeMs) {
+      ts.UpdateKeypress(false);
+      EXPECT_TRUE(ts.detection_enabled_);
+      EXPECT_TRUE(ts.suppression_enabled_);
+    }
+    ts.UpdateKeypress(true);
+    EXPECT_TRUE(ts.detection_enabled_);
+    EXPECT_TRUE(ts.suppression_enabled_);
+  }
+
+  // It takes four seconds without any key-press to disable the detection and
+  // suppression.
+  for (int time_ms = 0; time_ms < 3990; time_ms += ts::kChunkSizeMs) {
+    ts.UpdateKeypress(false);
+    EXPECT_TRUE(ts.detection_enabled_);
+    EXPECT_TRUE(ts.suppression_enabled_);
+  }
+  for (int time_ms = 0; time_ms < 1000; time_ms += ts::kChunkSizeMs) {
+    ts.UpdateKeypress(false);
+    EXPECT_FALSE(ts.detection_enabled_);
+    EXPECT_FALSE(ts.suppression_enabled_);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    TransientSuppressorImplTest,
+    TransientSuppressorVadModeParametrization,
+    ::testing::Values(TransientSuppressor::VadMode::kDefault,
+                      TransientSuppressor::VadMode::kRnnVad,
+                      TransientSuppressor::VadMode::kNoVad));
+
+class TransientSuppressorSampleRateParametrization
+    : public ::testing::TestWithParam<int> {};
+
+// Checks that voice probability and processed audio data are temporally aligned
+// after `Suppress()` is called.
+TEST_P(TransientSuppressorSampleRateParametrization,
+       CheckAudioAndVoiceProbabilityTemporallyAligned) {
+  const int sample_rate_hz = GetParam();
+  TransientSuppressorImpl ts(TransientSuppressor::VadMode::kDefault,
+                             sample_rate_hz,
+                             /*detection_rate_hz=*/sample_rate_hz, kMono);
+
+  const int frame_size = sample_rate_hz * ts::kChunkSizeMs / 1000;
+  std::vector<float> frame(frame_size);
+
+  constexpr int kMaxAttempts = 3;
+  for (int i = 0; i < kMaxAttempts; ++i) {
+    SCOPED_TRACE(i);
+
+    // Call `Suppress()` on frames of non-zero audio samples.
+    std::fill(frame.begin(), frame.end(), 1000.0f);
+    float delayed_voice_probability = ts.Suppress(
+        frame.data(), frame.size(), kMono, /*detection_data=*/nullptr,
+        /*detection_length=*/frame_size, /*reference_data=*/nullptr,
+        /*reference_length=*/frame_size, /*voice_probability=*/1.0f,
+        /*key_pressed=*/false);
+
+    // Detect the algorithmic delay of `TransientSuppressorImpl`.
+    absl::optional<int> frame_delay = FindFirstNonZeroSample(frame);
+
+    // Check that the delayed voice probability is delayed according to the
+    // measured delay.
+    if (frame_delay.has_value()) {
+      if (*frame_delay == 0) {
+        // When the delay is a multiple integer of the frame duration,
+        // `Suppress()` returns a copy of a previously observed voice
+        // probability value.
+        EXPECT_EQ(delayed_voice_probability, 1.0f);
+      } else {
+        // Instead, when the delay is fractional, `Suppress()` returns an
+        // interpolated value. Since the exact value depends on the
+        // interpolation method, we only check that the delayed voice
+        // probability is not zero as it must converge towards the previoulsy
+        // observed value.
+        EXPECT_GT(delayed_voice_probability, 0.0f);
+      }
+      break;
+    } else {
+      // The algorithmic delay is longer than the duration of a single frame.
+      // Until the delay is detected, the delayed voice probability is zero.
+      EXPECT_EQ(delayed_voice_probability, 0.0f);
+    }
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(TransientSuppressorImplTest,
+                         TransientSuppressorSampleRateParametrization,
+                         ::testing::Values(ts::kSampleRate8kHz,
+                                           ts::kSampleRate16kHz,
+                                           ts::kSampleRate32kHz,
+                                           ts::kSampleRate48kHz));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit.cc b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit.cc
new file mode 100644
index 0000000000..27b2b42b38
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/voice_probability_delay_unit.h"
+
+#include <array>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+VoiceProbabilityDelayUnit::VoiceProbabilityDelayUnit(int delay_num_samples,
+                                                     int sample_rate_hz) {
+  Initialize(delay_num_samples, sample_rate_hz);
+}
+
+void VoiceProbabilityDelayUnit::Initialize(int delay_num_samples,
+                                           int sample_rate_hz) {
+  RTC_DCHECK_GE(delay_num_samples, 0);
+  RTC_DCHECK_LE(delay_num_samples, sample_rate_hz / 50)
+      << "The implementation does not support delays greater than 20 ms.";
+  int frame_size = rtc::CheckedDivExact(sample_rate_hz, 100);  // 10 ms.
+  if (delay_num_samples <= frame_size) {
+    weights_[0] = 0.0f;
+    weights_[1] = static_cast<float>(delay_num_samples) / frame_size;
+    weights_[2] =
+        static_cast<float>(frame_size - delay_num_samples) / frame_size;
+  } else {
+    delay_num_samples -= frame_size;
+    weights_[0] = static_cast<float>(delay_num_samples) / frame_size;
+    weights_[1] =
+        static_cast<float>(frame_size - delay_num_samples) / frame_size;
+    weights_[2] = 0.0f;
+  }
+
+  // Resets the delay unit.
+  last_probabilities_.fill(0.0f);
+}
+
+float VoiceProbabilityDelayUnit::Delay(float voice_probability) {
+  float weighted_probability = weights_[0] * last_probabilities_[0] +
+                               weights_[1] * last_probabilities_[1] +
+                               weights_[2] * voice_probability;
+  last_probabilities_[0] = last_probabilities_[1];
+  last_probabilities_[1] = voice_probability;
+  return weighted_probability;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit.h b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit.h
new file mode 100644
index 0000000000..05961663e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
+
+#include <array>
+
+namespace webrtc {
+
+// Iteratively produces a sequence of delayed voice probability values given a
+// fixed delay between 0 and 20 ms and given a sequence of voice probability
+// values observed every 10 ms. Supports fractional delays, that are delays
+// which are not a multiple integer of 10 ms. Applies interpolation with
+// fractional delays; otherwise, returns a previously observed value according
+// to the given fixed delay.
+class VoiceProbabilityDelayUnit {
+ public:
+  // Ctor. `delay_num_samples` is the delay in number of samples and it must be
+  // non-negative and less than 20 ms.
+  VoiceProbabilityDelayUnit(int delay_num_samples, int sample_rate_hz);
+
+  // Handles delay and sample rate changes and resets the delay unit.
+  void Initialize(int delay_num_samples, int sample_rate_hz);
+
+  // Observes `voice_probability` and returns a delayed voice probability.
+  float Delay(float voice_probability);
+
+ private:
+  std::array<float, 3> weights_;
+  std::array<float, 2> last_probabilities_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_VOICE_PROBABILITY_DELAY_UNIT_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_gn/moz.build
new file mode 100644
index 0000000000..e2abcb8490
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("voice_probability_delay_unit_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_unittest.cc b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_unittest.cc
new file mode 100644
index 0000000000..04848e6f2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/voice_probability_delay_unit_unittest.cc
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/voice_probability_delay_unit.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+// Checks that with zero delay, the observed value is immediately returned as
+// delayed value.
+TEST(VoiceProbabilityDelayUnit, NoDelay) {
+  VoiceProbabilityDelayUnit delay_unit(/*delay_num_samples=*/0,
+                                       /*sample_rate_hz=*/48000);
+  constexpr int kMax = 5;
+  for (int i = 0; i <= kMax; ++i) {
+    SCOPED_TRACE(i);
+    float voice_probability = static_cast<float>(i) / kMax;
+    EXPECT_EQ(voice_probability, delay_unit.Delay(voice_probability));
+  }
+}
+
+// Checks that with integer delays, an exact copy of a previously observed value
+// is returned.
+TEST(VoiceProbabilityDelayUnit, IntegerDelay) {
+  VoiceProbabilityDelayUnit delay_unit_10ms(/*delay_num_samples=*/480,
+                                            /*sample_rate_hz=*/48000);
+  delay_unit_10ms.Delay(0.125f);
+  EXPECT_EQ(0.125f, delay_unit_10ms.Delay(0.9f));
+
+  VoiceProbabilityDelayUnit delay_unit_20ms(/*delay_num_samples=*/960,
+                                            /*sample_rate_hz=*/48000);
+  delay_unit_20ms.Delay(0.125f);
+  delay_unit_20ms.Delay(0.8f);
+  EXPECT_EQ(0.125f, delay_unit_20ms.Delay(0.9f));
+}
+
+// Checks that with a fractional delay < 10 ms, interpolation is applied.
+TEST(VoiceProbabilityDelayUnit, FractionalDelayLessThan10ms) {
+  // Create delay unit with fractional delay of 6 ms.
+  VoiceProbabilityDelayUnit delay_unit(/*delay_num_samples=*/288,
+                                       /*sample_rate_hz=*/48000);
+  //  frame 0
+  // --------- frame 1
+  //          ---------
+  //    0000001111
+  delay_unit.Delay(1.0f);
+  EXPECT_FLOAT_EQ(0.68f, delay_unit.Delay(0.2f));
+}
+
+// Checks that with a fractional delay > 10 ms, interpolation is applied.
+TEST(VoiceProbabilityDelayUnit, FractionalDelayGreaterThan10ms) {
+  // Create delay unit with fractional delay of 14 ms.
+  VoiceProbabilityDelayUnit delay_unit(/*delay_num_samples=*/672,
+                                       /*sample_rate_hz=*/48000);
+  //  frame 0
+  // --------- frame 1
+  //          --------- frame 2
+  //                   ---------
+  //      0000111111
+  delay_unit.Delay(1.0f);
+  delay_unit.Delay(0.2f);
+  EXPECT_FLOAT_EQ(0.52f, delay_unit.Delay(1.0f));
+}
+
+// Checks that `Initialize()` resets the delay unit.
+TEST(VoiceProbabilityDelayUnit, InitializeResetsDelayUnit) {
+  VoiceProbabilityDelayUnit delay_unit(/*delay_num_samples=*/960,
+                                       /*sample_rate_hz=*/48000);
+  delay_unit.Delay(1.0f);
+  delay_unit.Delay(0.9f);
+
+  delay_unit.Initialize(/*delay_num_samples=*/160, /*sample_rate_hz=*/8000);
+  EXPECT_EQ(0.0f, delay_unit.Delay(0.1f));
+  EXPECT_EQ(0.0f, delay_unit.Delay(0.2f));
+  EXPECT_EQ(0.1f, delay_unit.Delay(0.3f));
+}
+
+// Checks that `Initialize()` handles delay changes.
+TEST(VoiceProbabilityDelayUnit, InitializeHandlesDelayChanges) {
+  // Start with a 20 ms delay.
+  VoiceProbabilityDelayUnit delay_unit(/*delay_num_samples=*/960,
+                                       /*sample_rate_hz=*/48000);
+  delay_unit.Delay(1.0f);
+  delay_unit.Delay(0.9f);
+
+  // Lower the delay to 10 ms.
+  delay_unit.Initialize(/*delay_num_samples=*/80, /*sample_rate_hz=*/8000);
+  EXPECT_EQ(0.0f, delay_unit.Delay(0.1f));
+  EXPECT_EQ(0.1f, delay_unit.Delay(0.2f));
+
+  // Increase the delay to 15 ms.
+  delay_unit.Initialize(/*delay_num_samples=*/120, /*sample_rate_hz=*/8000);
+  EXPECT_EQ(0.0f, delay_unit.Delay(0.1f));
+  EXPECT_EQ(0.05f, delay_unit.Delay(0.2f));
+  EXPECT_EQ(0.15f, delay_unit.Delay(0.3f));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/windows_private.h b/third_party/libwebrtc/modules/audio_processing/transient/windows_private.h
new file mode 100644
index 0000000000..54e3c25785
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/windows_private.h
@@ -0,0 +1,557 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_WINDOWS_PRIVATE_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_WINDOWS_PRIVATE_H_
+
+namespace webrtc {
+
+// Hanning window for 4ms 16kHz
+static const float kHanning64w128[128] = {
+    0.00000000000000f, 0.02454122852291f, 0.04906767432742f, 0.07356456359967f,
+    0.09801714032956f, 0.12241067519922f, 0.14673047445536f, 0.17096188876030f,
+    0.19509032201613f, 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
+    0.29028467725446f, 0.31368174039889f, 0.33688985339222f, 0.35989503653499f,
+    0.38268343236509f, 0.40524131400499f, 0.42755509343028f, 0.44961132965461f,
+    0.47139673682600f, 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
+    0.55557023301960f, 0.57580819141785f, 0.59569930449243f, 0.61523159058063f,
+    0.63439328416365f, 0.65317284295378f, 0.67155895484702f, 0.68954054473707f,
+    0.70710678118655f, 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
+    0.77301045336274f, 0.78834642762661f, 0.80320753148064f, 0.81758481315158f,
+    0.83146961230255f, 0.84485356524971f, 0.85772861000027f, 0.87008699110871f,
+    0.88192126434835f, 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
+    0.92387953251129f, 0.93299279883474f, 0.94154406518302f, 0.94952818059304f,
+    0.95694033573221f, 0.96377606579544f, 0.97003125319454f, 0.97570213003853f,
+    0.98078528040323f, 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
+    0.99518472667220f, 0.99729045667869f, 0.99879545620517f, 0.99969881869620f,
+    1.00000000000000f, 0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
+    0.99518472667220f, 0.99247953459871f, 0.98917650996478f, 0.98527764238894f,
+    0.98078528040323f, 0.97570213003853f, 0.97003125319454f, 0.96377606579544f,
+    0.95694033573221f, 0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
+    0.92387953251129f, 0.91420975570353f, 0.90398929312344f, 0.89322430119552f,
+    0.88192126434835f, 0.87008699110871f, 0.85772861000027f, 0.84485356524971f,
+    0.83146961230255f, 0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
+    0.77301045336274f, 0.75720884650648f, 0.74095112535496f, 0.72424708295147f,
+    0.70710678118655f, 0.68954054473707f, 0.67155895484702f, 0.65317284295378f,
+    0.63439328416365f, 0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
+    0.55557023301960f, 0.53499761988710f, 0.51410274419322f, 0.49289819222978f,
+    0.47139673682600f, 0.44961132965461f, 0.42755509343028f, 0.40524131400499f,
+    0.38268343236509f, 0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
+    0.29028467725446f, 0.26671275747490f, 0.24298017990326f, 0.21910124015687f,
+    0.19509032201613f, 0.17096188876030f, 0.14673047445536f, 0.12241067519922f,
+    0.09801714032956f, 0.07356456359967f, 0.04906767432742f, 0.02454122852291f};
+
+// hybrib Hanning & flat window
+static const float kBlocks80w128[128] = {
+    0.00000000f, 0.03271908f, 0.06540313f, 0.09801714f, 0.13052619f,
+    0.16289547f, 0.19509032f, 0.22707626f, 0.25881905f, 0.29028468f,
+    0.32143947f, 0.35225005f, 0.38268343f, 0.41270703f, 0.44228869f,
+    0.47139674f, 0.50000000f, 0.52806785f, 0.55557023f, 0.58247770f,
+    0.60876143f, 0.63439328f, 0.65934582f, 0.68359230f, 0.70710678f,
+    0.72986407f, 0.75183981f, 0.77301045f, 0.79335334f, 0.81284668f,
+    0.83146961f, 0.84920218f, 0.86602540f, 0.88192126f, 0.89687274f,
+    0.91086382f, 0.92387953f, 0.93590593f, 0.94693013f, 0.95694034f,
+    0.96592583f, 0.97387698f, 0.98078528f, 0.98664333f, 0.99144486f,
+    0.99518473f, 0.99785892f, 0.99946459f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 0.99946459f, 0.99785892f, 0.99518473f, 0.99144486f,
+    0.98664333f, 0.98078528f, 0.97387698f, 0.96592583f, 0.95694034f,
+    0.94693013f, 0.93590593f, 0.92387953f, 0.91086382f, 0.89687274f,
+    0.88192126f, 0.86602540f, 0.84920218f, 0.83146961f, 0.81284668f,
+    0.79335334f, 0.77301045f, 0.75183981f, 0.72986407f, 0.70710678f,
+    0.68359230f, 0.65934582f, 0.63439328f, 0.60876143f, 0.58247770f,
+    0.55557023f, 0.52806785f, 0.50000000f, 0.47139674f, 0.44228869f,
+    0.41270703f, 0.38268343f, 0.35225005f, 0.32143947f, 0.29028468f,
+    0.25881905f, 0.22707626f, 0.19509032f, 0.16289547f, 0.13052619f,
+    0.09801714f, 0.06540313f, 0.03271908f};
+
+// hybrib Hanning & flat window
+static const float kBlocks160w256[256] = {
+    0.00000000f, 0.01636173f, 0.03271908f, 0.04906767f, 0.06540313f,
+    0.08172107f, 0.09801714f, 0.11428696f, 0.13052619f, 0.14673047f,
+    0.16289547f, 0.17901686f, 0.19509032f, 0.21111155f, 0.22707626f,
+    0.24298018f, 0.25881905f, 0.27458862f, 0.29028468f, 0.30590302f,
+    0.32143947f, 0.33688985f, 0.35225005f, 0.36751594f, 0.38268343f,
+    0.39774847f, 0.41270703f, 0.42755509f, 0.44228869f, 0.45690388f,
+    0.47139674f, 0.48576339f, 0.50000000f, 0.51410274f, 0.52806785f,
+    0.54189158f, 0.55557023f, 0.56910015f, 0.58247770f, 0.59569930f,
+    0.60876143f, 0.62166057f, 0.63439328f, 0.64695615f, 0.65934582f,
+    0.67155895f, 0.68359230f, 0.69544264f, 0.70710678f, 0.71858162f,
+    0.72986407f, 0.74095113f, 0.75183981f, 0.76252720f, 0.77301045f,
+    0.78328675f, 0.79335334f, 0.80320753f, 0.81284668f, 0.82226822f,
+    0.83146961f, 0.84044840f, 0.84920218f, 0.85772861f, 0.86602540f,
+    0.87409034f, 0.88192126f, 0.88951608f, 0.89687274f, 0.90398929f,
+    0.91086382f, 0.91749450f, 0.92387953f, 0.93001722f, 0.93590593f,
+    0.94154407f, 0.94693013f, 0.95206268f, 0.95694034f, 0.96156180f,
+    0.96592583f, 0.97003125f, 0.97387698f, 0.97746197f, 0.98078528f,
+    0.98384601f, 0.98664333f, 0.98917651f, 0.99144486f, 0.99344778f,
+    0.99518473f, 0.99665524f, 0.99785892f, 0.99879546f, 0.99946459f,
+    0.99986614f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 0.99986614f, 0.99946459f, 0.99879546f, 0.99785892f,
+    0.99665524f, 0.99518473f, 0.99344778f, 0.99144486f, 0.98917651f,
+    0.98664333f, 0.98384601f, 0.98078528f, 0.97746197f, 0.97387698f,
+    0.97003125f, 0.96592583f, 0.96156180f, 0.95694034f, 0.95206268f,
+    0.94693013f, 0.94154407f, 0.93590593f, 0.93001722f, 0.92387953f,
+    0.91749450f, 0.91086382f, 0.90398929f, 0.89687274f, 0.88951608f,
+    0.88192126f, 0.87409034f, 0.86602540f, 0.85772861f, 0.84920218f,
+    0.84044840f, 0.83146961f, 0.82226822f, 0.81284668f, 0.80320753f,
+    0.79335334f, 0.78328675f, 0.77301045f, 0.76252720f, 0.75183981f,
+    0.74095113f, 0.72986407f, 0.71858162f, 0.70710678f, 0.69544264f,
+    0.68359230f, 0.67155895f, 0.65934582f, 0.64695615f, 0.63439328f,
+    0.62166057f, 0.60876143f, 0.59569930f, 0.58247770f, 0.56910015f,
+    0.55557023f, 0.54189158f, 0.52806785f, 0.51410274f, 0.50000000f,
+    0.48576339f, 0.47139674f, 0.45690388f, 0.44228869f, 0.42755509f,
+    0.41270703f, 0.39774847f, 0.38268343f, 0.36751594f, 0.35225005f,
+    0.33688985f, 0.32143947f, 0.30590302f, 0.29028468f, 0.27458862f,
+    0.25881905f, 0.24298018f, 0.22707626f, 0.21111155f, 0.19509032f,
+    0.17901686f, 0.16289547f, 0.14673047f, 0.13052619f, 0.11428696f,
+    0.09801714f, 0.08172107f, 0.06540313f, 0.04906767f, 0.03271908f,
+    0.01636173f};
+
+// hybrib Hanning & flat window: for 20ms
+static const float kBlocks320w512[512] = {
+    0.00000000f, 0.00818114f, 0.01636173f, 0.02454123f, 0.03271908f,
+    0.04089475f, 0.04906767f, 0.05723732f, 0.06540313f, 0.07356456f,
+    0.08172107f, 0.08987211f, 0.09801714f, 0.10615561f, 0.11428696f,
+    0.12241068f, 0.13052619f, 0.13863297f, 0.14673047f, 0.15481816f,
+    0.16289547f, 0.17096189f, 0.17901686f, 0.18705985f, 0.19509032f,
+    0.20310773f, 0.21111155f, 0.21910124f, 0.22707626f, 0.23503609f,
+    0.24298018f, 0.25090801f, 0.25881905f, 0.26671276f, 0.27458862f,
+    0.28244610f, 0.29028468f, 0.29810383f, 0.30590302f, 0.31368174f,
+    0.32143947f, 0.32917568f, 0.33688985f, 0.34458148f, 0.35225005f,
+    0.35989504f, 0.36751594f, 0.37511224f, 0.38268343f, 0.39022901f,
+    0.39774847f, 0.40524131f, 0.41270703f, 0.42014512f, 0.42755509f,
+    0.43493645f, 0.44228869f, 0.44961133f, 0.45690388f, 0.46416584f,
+    0.47139674f, 0.47859608f, 0.48576339f, 0.49289819f, 0.50000000f,
+    0.50706834f, 0.51410274f, 0.52110274f, 0.52806785f, 0.53499762f,
+    0.54189158f, 0.54874927f, 0.55557023f, 0.56235401f, 0.56910015f,
+    0.57580819f, 0.58247770f, 0.58910822f, 0.59569930f, 0.60225052f,
+    0.60876143f, 0.61523159f, 0.62166057f, 0.62804795f, 0.63439328f,
+    0.64069616f, 0.64695615f, 0.65317284f, 0.65934582f, 0.66547466f,
+    0.67155895f, 0.67759830f, 0.68359230f, 0.68954054f, 0.69544264f,
+    0.70129818f, 0.70710678f, 0.71286806f, 0.71858162f, 0.72424708f,
+    0.72986407f, 0.73543221f, 0.74095113f, 0.74642045f, 0.75183981f,
+    0.75720885f, 0.76252720f, 0.76779452f, 0.77301045f, 0.77817464f,
+    0.78328675f, 0.78834643f, 0.79335334f, 0.79830715f, 0.80320753f,
+    0.80805415f, 0.81284668f, 0.81758481f, 0.82226822f, 0.82689659f,
+    0.83146961f, 0.83598698f, 0.84044840f, 0.84485357f, 0.84920218f,
+    0.85349396f, 0.85772861f, 0.86190585f, 0.86602540f, 0.87008699f,
+    0.87409034f, 0.87803519f, 0.88192126f, 0.88574831f, 0.88951608f,
+    0.89322430f, 0.89687274f, 0.90046115f, 0.90398929f, 0.90745693f,
+    0.91086382f, 0.91420976f, 0.91749450f, 0.92071783f, 0.92387953f,
+    0.92697940f, 0.93001722f, 0.93299280f, 0.93590593f, 0.93875641f,
+    0.94154407f, 0.94426870f, 0.94693013f, 0.94952818f, 0.95206268f,
+    0.95453345f, 0.95694034f, 0.95928317f, 0.96156180f, 0.96377607f,
+    0.96592583f, 0.96801094f, 0.97003125f, 0.97198664f, 0.97387698f,
+    0.97570213f, 0.97746197f, 0.97915640f, 0.98078528f, 0.98234852f,
+    0.98384601f, 0.98527764f, 0.98664333f, 0.98794298f, 0.98917651f,
+    0.99034383f, 0.99144486f, 0.99247953f, 0.99344778f, 0.99434953f,
+    0.99518473f, 0.99595331f, 0.99665524f, 0.99729046f, 0.99785892f,
+    0.99836060f, 0.99879546f, 0.99916346f, 0.99946459f, 0.99969882f,
+    0.99986614f, 0.99996653f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 0.99996653f, 0.99986614f, 0.99969882f, 0.99946459f,
+    0.99916346f, 0.99879546f, 0.99836060f, 0.99785892f, 0.99729046f,
+    0.99665524f, 0.99595331f, 0.99518473f, 0.99434953f, 0.99344778f,
+    0.99247953f, 0.99144486f, 0.99034383f, 0.98917651f, 0.98794298f,
+    0.98664333f, 0.98527764f, 0.98384601f, 0.98234852f, 0.98078528f,
+    0.97915640f, 0.97746197f, 0.97570213f, 0.97387698f, 0.97198664f,
+    0.97003125f, 0.96801094f, 0.96592583f, 0.96377607f, 0.96156180f,
+    0.95928317f, 0.95694034f, 0.95453345f, 0.95206268f, 0.94952818f,
+    0.94693013f, 0.94426870f, 0.94154407f, 0.93875641f, 0.93590593f,
+    0.93299280f, 0.93001722f, 0.92697940f, 0.92387953f, 0.92071783f,
+    0.91749450f, 0.91420976f, 0.91086382f, 0.90745693f, 0.90398929f,
+    0.90046115f, 0.89687274f, 0.89322430f, 0.88951608f, 0.88574831f,
+    0.88192126f, 0.87803519f, 0.87409034f, 0.87008699f, 0.86602540f,
+    0.86190585f, 0.85772861f, 0.85349396f, 0.84920218f, 0.84485357f,
+    0.84044840f, 0.83598698f, 0.83146961f, 0.82689659f, 0.82226822f,
+    0.81758481f, 0.81284668f, 0.80805415f, 0.80320753f, 0.79830715f,
+    0.79335334f, 0.78834643f, 0.78328675f, 0.77817464f, 0.77301045f,
+    0.76779452f, 0.76252720f, 0.75720885f, 0.75183981f, 0.74642045f,
+    0.74095113f, 0.73543221f, 0.72986407f, 0.72424708f, 0.71858162f,
+    0.71286806f, 0.70710678f, 0.70129818f, 0.69544264f, 0.68954054f,
+    0.68359230f, 0.67759830f, 0.67155895f, 0.66547466f, 0.65934582f,
+    0.65317284f, 0.64695615f, 0.64069616f, 0.63439328f, 0.62804795f,
+    0.62166057f, 0.61523159f, 0.60876143f, 0.60225052f, 0.59569930f,
+    0.58910822f, 0.58247770f, 0.57580819f, 0.56910015f, 0.56235401f,
+    0.55557023f, 0.54874927f, 0.54189158f, 0.53499762f, 0.52806785f,
+    0.52110274f, 0.51410274f, 0.50706834f, 0.50000000f, 0.49289819f,
+    0.48576339f, 0.47859608f, 0.47139674f, 0.46416584f, 0.45690388f,
+    0.44961133f, 0.44228869f, 0.43493645f, 0.42755509f, 0.42014512f,
+    0.41270703f, 0.40524131f, 0.39774847f, 0.39022901f, 0.38268343f,
+    0.37511224f, 0.36751594f, 0.35989504f, 0.35225005f, 0.34458148f,
+    0.33688985f, 0.32917568f, 0.32143947f, 0.31368174f, 0.30590302f,
+    0.29810383f, 0.29028468f, 0.28244610f, 0.27458862f, 0.26671276f,
+    0.25881905f, 0.25090801f, 0.24298018f, 0.23503609f, 0.22707626f,
+    0.21910124f, 0.21111155f, 0.20310773f, 0.19509032f, 0.18705985f,
+    0.17901686f, 0.17096189f, 0.16289547f, 0.15481816f, 0.14673047f,
+    0.13863297f, 0.13052619f, 0.12241068f, 0.11428696f, 0.10615561f,
+    0.09801714f, 0.08987211f, 0.08172107f, 0.07356456f, 0.06540313f,
+    0.05723732f, 0.04906767f, 0.04089475f, 0.03271908f, 0.02454123f,
+    0.01636173f, 0.00818114f};
+
+// Hanning window: for 15ms at 16kHz with symmetric zeros
+static const float kBlocks240w512[512] = {
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00654494f, 0.01308960f, 0.01963369f,
+    0.02617695f, 0.03271908f, 0.03925982f, 0.04579887f, 0.05233596f,
+    0.05887080f, 0.06540313f, 0.07193266f, 0.07845910f, 0.08498218f,
+    0.09150162f, 0.09801714f, 0.10452846f, 0.11103531f, 0.11753740f,
+    0.12403446f, 0.13052620f, 0.13701233f, 0.14349262f, 0.14996676f,
+    0.15643448f, 0.16289547f, 0.16934951f, 0.17579629f, 0.18223552f,
+    0.18866697f, 0.19509032f, 0.20150533f, 0.20791170f, 0.21430916f,
+    0.22069745f, 0.22707628f, 0.23344538f, 0.23980446f, 0.24615330f,
+    0.25249159f, 0.25881904f, 0.26513544f, 0.27144045f, 0.27773386f,
+    0.28401536f, 0.29028466f, 0.29654160f, 0.30278578f, 0.30901700f,
+    0.31523499f, 0.32143945f, 0.32763019f, 0.33380687f, 0.33996925f,
+    0.34611708f, 0.35225007f, 0.35836795f, 0.36447051f, 0.37055743f,
+    0.37662852f, 0.38268346f, 0.38872197f, 0.39474389f, 0.40074885f,
+    0.40673664f, 0.41270703f, 0.41865975f, 0.42459452f, 0.43051112f,
+    0.43640924f, 0.44228873f, 0.44814920f, 0.45399052f, 0.45981237f,
+    0.46561453f, 0.47139674f, 0.47715878f, 0.48290035f, 0.48862126f,
+    0.49432120f, 0.50000000f, 0.50565743f, 0.51129311f, 0.51690692f,
+    0.52249855f, 0.52806789f, 0.53361452f, 0.53913832f, 0.54463905f,
+    0.55011642f, 0.55557024f, 0.56100029f, 0.56640625f, 0.57178795f,
+    0.57714522f, 0.58247769f, 0.58778524f, 0.59306765f, 0.59832460f,
+    0.60355598f, 0.60876143f, 0.61394083f, 0.61909395f, 0.62422055f,
+    0.62932038f, 0.63439333f, 0.63943899f, 0.64445734f, 0.64944810f,
+    0.65441096f, 0.65934587f, 0.66425246f, 0.66913062f, 0.67398012f,
+    0.67880076f, 0.68359232f, 0.68835455f, 0.69308740f, 0.69779050f,
+    0.70246369f, 0.70710677f, 0.71171963f, 0.71630198f, 0.72085363f,
+    0.72537440f, 0.72986406f, 0.73432255f, 0.73874950f, 0.74314487f,
+    0.74750835f, 0.75183982f, 0.75613910f, 0.76040596f, 0.76464027f,
+    0.76884186f, 0.77301043f, 0.77714598f, 0.78124821f, 0.78531694f,
+    0.78935206f, 0.79335338f, 0.79732066f, 0.80125386f, 0.80515265f,
+    0.80901700f, 0.81284672f, 0.81664157f, 0.82040149f, 0.82412618f,
+    0.82781565f, 0.83146966f, 0.83508795f, 0.83867061f, 0.84221727f,
+    0.84572780f, 0.84920216f, 0.85264021f, 0.85604161f, 0.85940641f,
+    0.86273444f, 0.86602545f, 0.86927933f, 0.87249607f, 0.87567532f,
+    0.87881714f, 0.88192129f, 0.88498765f, 0.88801610f, 0.89100653f,
+    0.89395881f, 0.89687276f, 0.89974827f, 0.90258533f, 0.90538365f,
+    0.90814316f, 0.91086388f, 0.91354549f, 0.91618794f, 0.91879123f,
+    0.92135513f, 0.92387950f, 0.92636442f, 0.92880958f, 0.93121493f,
+    0.93358046f, 0.93590593f, 0.93819135f, 0.94043654f, 0.94264150f,
+    0.94480604f, 0.94693011f, 0.94901365f, 0.95105654f, 0.95305866f,
+    0.95501995f, 0.95694035f, 0.95881975f, 0.96065807f, 0.96245527f,
+    0.96421117f, 0.96592581f, 0.96759909f, 0.96923089f, 0.97082120f,
+    0.97236991f, 0.97387701f, 0.97534233f, 0.97676587f, 0.97814763f,
+    0.97948742f, 0.98078531f, 0.98204112f, 0.98325491f, 0.98442656f,
+    0.98555607f, 0.98664331f, 0.98768836f, 0.98869103f, 0.98965138f,
+    0.99056935f, 0.99144489f, 0.99227792f, 0.99306846f, 0.99381649f,
+    0.99452192f, 0.99518472f, 0.99580491f, 0.99638247f, 0.99691731f,
+    0.99740952f, 0.99785894f, 0.99826562f, 0.99862951f, 0.99895066f,
+    0.99922901f, 0.99946457f, 0.99965733f, 0.99980724f, 0.99991435f,
+    0.99997860f, 1.00000000f, 0.99997860f, 0.99991435f, 0.99980724f,
+    0.99965733f, 0.99946457f, 0.99922901f, 0.99895066f, 0.99862951f,
+    0.99826562f, 0.99785894f, 0.99740946f, 0.99691731f, 0.99638247f,
+    0.99580491f, 0.99518472f, 0.99452192f, 0.99381644f, 0.99306846f,
+    0.99227792f, 0.99144489f, 0.99056935f, 0.98965138f, 0.98869103f,
+    0.98768836f, 0.98664331f, 0.98555607f, 0.98442656f, 0.98325491f,
+    0.98204112f, 0.98078525f, 0.97948742f, 0.97814757f, 0.97676587f,
+    0.97534227f, 0.97387695f, 0.97236991f, 0.97082120f, 0.96923089f,
+    0.96759909f, 0.96592581f, 0.96421117f, 0.96245521f, 0.96065807f,
+    0.95881969f, 0.95694029f, 0.95501995f, 0.95305860f, 0.95105648f,
+    0.94901365f, 0.94693011f, 0.94480604f, 0.94264150f, 0.94043654f,
+    0.93819129f, 0.93590593f, 0.93358046f, 0.93121493f, 0.92880952f,
+    0.92636436f, 0.92387950f, 0.92135507f, 0.91879123f, 0.91618794f,
+    0.91354543f, 0.91086382f, 0.90814310f, 0.90538365f, 0.90258527f,
+    0.89974827f, 0.89687276f, 0.89395875f, 0.89100647f, 0.88801610f,
+    0.88498759f, 0.88192123f, 0.87881714f, 0.87567532f, 0.87249595f,
+    0.86927933f, 0.86602539f, 0.86273432f, 0.85940641f, 0.85604161f,
+    0.85264009f, 0.84920216f, 0.84572780f, 0.84221715f, 0.83867055f,
+    0.83508795f, 0.83146954f, 0.82781565f, 0.82412612f, 0.82040137f,
+    0.81664157f, 0.81284660f, 0.80901700f, 0.80515265f, 0.80125374f,
+    0.79732066f, 0.79335332f, 0.78935200f, 0.78531694f, 0.78124815f,
+    0.77714586f, 0.77301049f, 0.76884180f, 0.76464021f, 0.76040596f,
+    0.75613904f, 0.75183970f, 0.74750835f, 0.74314481f, 0.73874938f,
+    0.73432249f, 0.72986400f, 0.72537428f, 0.72085363f, 0.71630186f,
+    0.71171951f, 0.70710677f, 0.70246363f, 0.69779032f, 0.69308734f,
+    0.68835449f, 0.68359220f, 0.67880070f, 0.67398006f, 0.66913044f,
+    0.66425240f, 0.65934575f, 0.65441096f, 0.64944804f, 0.64445722f,
+    0.63943905f, 0.63439327f, 0.62932026f, 0.62422055f, 0.61909389f,
+    0.61394072f, 0.60876143f, 0.60355592f, 0.59832448f, 0.59306765f,
+    0.58778518f, 0.58247757f, 0.57714522f, 0.57178789f, 0.56640613f,
+    0.56100023f, 0.55557019f, 0.55011630f, 0.54463905f, 0.53913826f,
+    0.53361434f, 0.52806783f, 0.52249849f, 0.51690674f, 0.51129305f,
+    0.50565726f, 0.50000006f, 0.49432117f, 0.48862115f, 0.48290038f,
+    0.47715873f, 0.47139663f, 0.46561456f, 0.45981231f, 0.45399037f,
+    0.44814920f, 0.44228864f, 0.43640912f, 0.43051112f, 0.42459446f,
+    0.41865960f, 0.41270703f, 0.40673658f, 0.40074870f, 0.39474386f,
+    0.38872188f, 0.38268328f, 0.37662849f, 0.37055734f, 0.36447033f,
+    0.35836792f, 0.35224995f, 0.34611690f, 0.33996922f, 0.33380675f,
+    0.32763001f, 0.32143945f, 0.31523487f, 0.30901679f, 0.30278572f,
+    0.29654145f, 0.29028472f, 0.28401530f, 0.27773371f, 0.27144048f,
+    0.26513538f, 0.25881892f, 0.25249159f, 0.24615324f, 0.23980433f,
+    0.23344538f, 0.22707619f, 0.22069728f, 0.21430916f, 0.20791161f,
+    0.20150517f, 0.19509031f, 0.18866688f, 0.18223536f, 0.17579627f,
+    0.16934940f, 0.16289529f, 0.15643445f, 0.14996666f, 0.14349243f,
+    0.13701232f, 0.13052608f, 0.12403426f, 0.11753736f, 0.11103519f,
+    0.10452849f, 0.09801710f, 0.09150149f, 0.08498220f, 0.07845904f,
+    0.07193252f, 0.06540315f, 0.05887074f, 0.05233581f, 0.04579888f,
+    0.03925974f, 0.03271893f, 0.02617695f, 0.01963361f, 0.01308943f,
+    0.00654493f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f};
+
+// Hanning window: for 30ms with 1024 fft with symmetric zeros at 16kHz
+static const float kBlocks480w1024[1024] = {
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00327249f, 0.00654494f,
+    0.00981732f, 0.01308960f, 0.01636173f, 0.01963369f, 0.02290544f,
+    0.02617695f, 0.02944817f, 0.03271908f, 0.03598964f, 0.03925982f,
+    0.04252957f, 0.04579887f, 0.04906768f, 0.05233596f, 0.05560368f,
+    0.05887080f, 0.06213730f, 0.06540313f, 0.06866825f, 0.07193266f,
+    0.07519628f, 0.07845910f, 0.08172107f, 0.08498218f, 0.08824237f,
+    0.09150162f, 0.09475989f, 0.09801714f, 0.10127335f, 0.10452846f,
+    0.10778246f, 0.11103531f, 0.11428697f, 0.11753740f, 0.12078657f,
+    0.12403446f, 0.12728101f, 0.13052620f, 0.13376999f, 0.13701233f,
+    0.14025325f, 0.14349262f, 0.14673047f, 0.14996676f, 0.15320145f,
+    0.15643448f, 0.15966582f, 0.16289547f, 0.16612339f, 0.16934951f,
+    0.17257382f, 0.17579629f, 0.17901687f, 0.18223552f, 0.18545224f,
+    0.18866697f, 0.19187967f, 0.19509032f, 0.19829889f, 0.20150533f,
+    0.20470962f, 0.20791170f, 0.21111156f, 0.21430916f, 0.21750447f,
+    0.22069745f, 0.22388805f, 0.22707628f, 0.23026206f, 0.23344538f,
+    0.23662618f, 0.23980446f, 0.24298020f, 0.24615330f, 0.24932377f,
+    0.25249159f, 0.25565669f, 0.25881904f, 0.26197866f, 0.26513544f,
+    0.26828939f, 0.27144045f, 0.27458861f, 0.27773386f, 0.28087610f,
+    0.28401536f, 0.28715158f, 0.29028466f, 0.29341471f, 0.29654160f,
+    0.29966527f, 0.30278578f, 0.30590302f, 0.30901700f, 0.31212768f,
+    0.31523499f, 0.31833893f, 0.32143945f, 0.32453656f, 0.32763019f,
+    0.33072028f, 0.33380687f, 0.33688986f, 0.33996925f, 0.34304500f,
+    0.34611708f, 0.34918544f, 0.35225007f, 0.35531089f, 0.35836795f,
+    0.36142117f, 0.36447051f, 0.36751595f, 0.37055743f, 0.37359497f,
+    0.37662852f, 0.37965801f, 0.38268346f, 0.38570479f, 0.38872197f,
+    0.39173502f, 0.39474389f, 0.39774847f, 0.40074885f, 0.40374491f,
+    0.40673664f, 0.40972406f, 0.41270703f, 0.41568562f, 0.41865975f,
+    0.42162940f, 0.42459452f, 0.42755508f, 0.43051112f, 0.43346250f,
+    0.43640924f, 0.43935132f, 0.44228873f, 0.44522133f, 0.44814920f,
+    0.45107228f, 0.45399052f, 0.45690390f, 0.45981237f, 0.46271592f,
+    0.46561453f, 0.46850815f, 0.47139674f, 0.47428030f, 0.47715878f,
+    0.48003215f, 0.48290035f, 0.48576337f, 0.48862126f, 0.49147385f,
+    0.49432120f, 0.49716330f, 0.50000000f, 0.50283140f, 0.50565743f,
+    0.50847799f, 0.51129311f, 0.51410276f, 0.51690692f, 0.51970553f,
+    0.52249855f, 0.52528602f, 0.52806789f, 0.53084403f, 0.53361452f,
+    0.53637928f, 0.53913832f, 0.54189163f, 0.54463905f, 0.54738063f,
+    0.55011642f, 0.55284631f, 0.55557024f, 0.55828828f, 0.56100029f,
+    0.56370628f, 0.56640625f, 0.56910014f, 0.57178795f, 0.57446963f,
+    0.57714522f, 0.57981455f, 0.58247769f, 0.58513463f, 0.58778524f,
+    0.59042960f, 0.59306765f, 0.59569931f, 0.59832460f, 0.60094351f,
+    0.60355598f, 0.60616195f, 0.60876143f, 0.61135441f, 0.61394083f,
+    0.61652070f, 0.61909395f, 0.62166059f, 0.62422055f, 0.62677383f,
+    0.62932038f, 0.63186020f, 0.63439333f, 0.63691956f, 0.63943899f,
+    0.64195162f, 0.64445734f, 0.64695615f, 0.64944810f, 0.65193301f,
+    0.65441096f, 0.65688187f, 0.65934587f, 0.66180271f, 0.66425246f,
+    0.66669512f, 0.66913062f, 0.67155898f, 0.67398012f, 0.67639405f,
+    0.67880076f, 0.68120021f, 0.68359232f, 0.68597710f, 0.68835455f,
+    0.69072467f, 0.69308740f, 0.69544262f, 0.69779050f, 0.70013082f,
+    0.70246369f, 0.70478904f, 0.70710677f, 0.70941699f, 0.71171963f,
+    0.71401459f, 0.71630198f, 0.71858168f, 0.72085363f, 0.72311789f,
+    0.72537440f, 0.72762316f, 0.72986406f, 0.73209721f, 0.73432255f,
+    0.73653996f, 0.73874950f, 0.74095118f, 0.74314487f, 0.74533057f,
+    0.74750835f, 0.74967808f, 0.75183982f, 0.75399351f, 0.75613910f,
+    0.75827658f, 0.76040596f, 0.76252723f, 0.76464027f, 0.76674515f,
+    0.76884186f, 0.77093029f, 0.77301043f, 0.77508241f, 0.77714598f,
+    0.77920127f, 0.78124821f, 0.78328675f, 0.78531694f, 0.78733873f,
+    0.78935206f, 0.79135692f, 0.79335338f, 0.79534125f, 0.79732066f,
+    0.79929149f, 0.80125386f, 0.80320752f, 0.80515265f, 0.80708915f,
+    0.80901700f, 0.81093621f, 0.81284672f, 0.81474853f, 0.81664157f,
+    0.81852591f, 0.82040149f, 0.82226825f, 0.82412618f, 0.82597536f,
+    0.82781565f, 0.82964706f, 0.83146966f, 0.83328325f, 0.83508795f,
+    0.83688378f, 0.83867061f, 0.84044838f, 0.84221727f, 0.84397703f,
+    0.84572780f, 0.84746957f, 0.84920216f, 0.85092574f, 0.85264021f,
+    0.85434544f, 0.85604161f, 0.85772866f, 0.85940641f, 0.86107504f,
+    0.86273444f, 0.86438453f, 0.86602545f, 0.86765707f, 0.86927933f,
+    0.87089235f, 0.87249607f, 0.87409031f, 0.87567532f, 0.87725097f,
+    0.87881714f, 0.88037390f, 0.88192129f, 0.88345921f, 0.88498765f,
+    0.88650668f, 0.88801610f, 0.88951612f, 0.89100653f, 0.89248741f,
+    0.89395881f, 0.89542055f, 0.89687276f, 0.89831537f, 0.89974827f,
+    0.90117162f, 0.90258533f, 0.90398932f, 0.90538365f, 0.90676826f,
+    0.90814316f, 0.90950841f, 0.91086388f, 0.91220951f, 0.91354549f,
+    0.91487163f, 0.91618794f, 0.91749454f, 0.91879123f, 0.92007810f,
+    0.92135513f, 0.92262226f, 0.92387950f, 0.92512691f, 0.92636442f,
+    0.92759192f, 0.92880958f, 0.93001723f, 0.93121493f, 0.93240267f,
+    0.93358046f, 0.93474817f, 0.93590593f, 0.93705362f, 0.93819135f,
+    0.93931901f, 0.94043654f, 0.94154406f, 0.94264150f, 0.94372880f,
+    0.94480604f, 0.94587320f, 0.94693011f, 0.94797695f, 0.94901365f,
+    0.95004016f, 0.95105654f, 0.95206273f, 0.95305866f, 0.95404440f,
+    0.95501995f, 0.95598525f, 0.95694035f, 0.95788521f, 0.95881975f,
+    0.95974404f, 0.96065807f, 0.96156180f, 0.96245527f, 0.96333838f,
+    0.96421117f, 0.96507370f, 0.96592581f, 0.96676767f, 0.96759909f,
+    0.96842021f, 0.96923089f, 0.97003126f, 0.97082120f, 0.97160077f,
+    0.97236991f, 0.97312868f, 0.97387701f, 0.97461486f, 0.97534233f,
+    0.97605932f, 0.97676587f, 0.97746199f, 0.97814763f, 0.97882277f,
+    0.97948742f, 0.98014158f, 0.98078531f, 0.98141843f, 0.98204112f,
+    0.98265332f, 0.98325491f, 0.98384601f, 0.98442656f, 0.98499662f,
+    0.98555607f, 0.98610497f, 0.98664331f, 0.98717111f, 0.98768836f,
+    0.98819500f, 0.98869103f, 0.98917651f, 0.98965138f, 0.99011570f,
+    0.99056935f, 0.99101239f, 0.99144489f, 0.99186671f, 0.99227792f,
+    0.99267852f, 0.99306846f, 0.99344778f, 0.99381649f, 0.99417448f,
+    0.99452192f, 0.99485862f, 0.99518472f, 0.99550015f, 0.99580491f,
+    0.99609905f, 0.99638247f, 0.99665523f, 0.99691731f, 0.99716878f,
+    0.99740952f, 0.99763954f, 0.99785894f, 0.99806762f, 0.99826562f,
+    0.99845290f, 0.99862951f, 0.99879545f, 0.99895066f, 0.99909520f,
+    0.99922901f, 0.99935216f, 0.99946457f, 0.99956632f, 0.99965733f,
+    0.99973762f, 0.99980724f, 0.99986613f, 0.99991435f, 0.99995178f,
+    0.99997860f, 0.99999464f, 1.00000000f, 0.99999464f, 0.99997860f,
+    0.99995178f, 0.99991435f, 0.99986613f, 0.99980724f, 0.99973762f,
+    0.99965733f, 0.99956632f, 0.99946457f, 0.99935216f, 0.99922901f,
+    0.99909520f, 0.99895066f, 0.99879545f, 0.99862951f, 0.99845290f,
+    0.99826562f, 0.99806762f, 0.99785894f, 0.99763954f, 0.99740946f,
+    0.99716872f, 0.99691731f, 0.99665523f, 0.99638247f, 0.99609905f,
+    0.99580491f, 0.99550015f, 0.99518472f, 0.99485862f, 0.99452192f,
+    0.99417448f, 0.99381644f, 0.99344778f, 0.99306846f, 0.99267852f,
+    0.99227792f, 0.99186671f, 0.99144489f, 0.99101239f, 0.99056935f,
+    0.99011564f, 0.98965138f, 0.98917651f, 0.98869103f, 0.98819494f,
+    0.98768836f, 0.98717111f, 0.98664331f, 0.98610497f, 0.98555607f,
+    0.98499656f, 0.98442656f, 0.98384601f, 0.98325491f, 0.98265326f,
+    0.98204112f, 0.98141843f, 0.98078525f, 0.98014158f, 0.97948742f,
+    0.97882277f, 0.97814757f, 0.97746193f, 0.97676587f, 0.97605932f,
+    0.97534227f, 0.97461486f, 0.97387695f, 0.97312862f, 0.97236991f,
+    0.97160077f, 0.97082120f, 0.97003126f, 0.96923089f, 0.96842015f,
+    0.96759909f, 0.96676761f, 0.96592581f, 0.96507365f, 0.96421117f,
+    0.96333838f, 0.96245521f, 0.96156180f, 0.96065807f, 0.95974404f,
+    0.95881969f, 0.95788515f, 0.95694029f, 0.95598525f, 0.95501995f,
+    0.95404440f, 0.95305860f, 0.95206267f, 0.95105648f, 0.95004016f,
+    0.94901365f, 0.94797695f, 0.94693011f, 0.94587314f, 0.94480604f,
+    0.94372880f, 0.94264150f, 0.94154406f, 0.94043654f, 0.93931895f,
+    0.93819129f, 0.93705362f, 0.93590593f, 0.93474817f, 0.93358046f,
+    0.93240267f, 0.93121493f, 0.93001723f, 0.92880952f, 0.92759192f,
+    0.92636436f, 0.92512691f, 0.92387950f, 0.92262226f, 0.92135507f,
+    0.92007804f, 0.91879123f, 0.91749448f, 0.91618794f, 0.91487157f,
+    0.91354543f, 0.91220951f, 0.91086382f, 0.90950835f, 0.90814310f,
+    0.90676820f, 0.90538365f, 0.90398932f, 0.90258527f, 0.90117157f,
+    0.89974827f, 0.89831525f, 0.89687276f, 0.89542055f, 0.89395875f,
+    0.89248741f, 0.89100647f, 0.88951600f, 0.88801610f, 0.88650662f,
+    0.88498759f, 0.88345915f, 0.88192123f, 0.88037384f, 0.87881714f,
+    0.87725091f, 0.87567532f, 0.87409031f, 0.87249595f, 0.87089223f,
+    0.86927933f, 0.86765701f, 0.86602539f, 0.86438447f, 0.86273432f,
+    0.86107504f, 0.85940641f, 0.85772860f, 0.85604161f, 0.85434544f,
+    0.85264009f, 0.85092574f, 0.84920216f, 0.84746951f, 0.84572780f,
+    0.84397697f, 0.84221715f, 0.84044844f, 0.83867055f, 0.83688372f,
+    0.83508795f, 0.83328319f, 0.83146954f, 0.82964706f, 0.82781565f,
+    0.82597530f, 0.82412612f, 0.82226813f, 0.82040137f, 0.81852591f,
+    0.81664157f, 0.81474847f, 0.81284660f, 0.81093609f, 0.80901700f,
+    0.80708915f, 0.80515265f, 0.80320752f, 0.80125374f, 0.79929143f,
+    0.79732066f, 0.79534125f, 0.79335332f, 0.79135686f, 0.78935200f,
+    0.78733861f, 0.78531694f, 0.78328675f, 0.78124815f, 0.77920121f,
+    0.77714586f, 0.77508223f, 0.77301049f, 0.77093029f, 0.76884180f,
+    0.76674509f, 0.76464021f, 0.76252711f, 0.76040596f, 0.75827658f,
+    0.75613904f, 0.75399339f, 0.75183970f, 0.74967796f, 0.74750835f,
+    0.74533057f, 0.74314481f, 0.74095106f, 0.73874938f, 0.73653996f,
+    0.73432249f, 0.73209721f, 0.72986400f, 0.72762305f, 0.72537428f,
+    0.72311789f, 0.72085363f, 0.71858162f, 0.71630186f, 0.71401453f,
+    0.71171951f, 0.70941705f, 0.70710677f, 0.70478898f, 0.70246363f,
+    0.70013070f, 0.69779032f, 0.69544268f, 0.69308734f, 0.69072461f,
+    0.68835449f, 0.68597704f, 0.68359220f, 0.68120021f, 0.67880070f,
+    0.67639399f, 0.67398006f, 0.67155886f, 0.66913044f, 0.66669512f,
+    0.66425240f, 0.66180259f, 0.65934575f, 0.65688181f, 0.65441096f,
+    0.65193301f, 0.64944804f, 0.64695609f, 0.64445722f, 0.64195150f,
+    0.63943905f, 0.63691956f, 0.63439327f, 0.63186014f, 0.62932026f,
+    0.62677372f, 0.62422055f, 0.62166059f, 0.61909389f, 0.61652064f,
+    0.61394072f, 0.61135429f, 0.60876143f, 0.60616189f, 0.60355592f,
+    0.60094339f, 0.59832448f, 0.59569913f, 0.59306765f, 0.59042960f,
+    0.58778518f, 0.58513451f, 0.58247757f, 0.57981461f, 0.57714522f,
+    0.57446963f, 0.57178789f, 0.56910002f, 0.56640613f, 0.56370628f,
+    0.56100023f, 0.55828822f, 0.55557019f, 0.55284619f, 0.55011630f,
+    0.54738069f, 0.54463905f, 0.54189152f, 0.53913826f, 0.53637916f,
+    0.53361434f, 0.53084403f, 0.52806783f, 0.52528596f, 0.52249849f,
+    0.51970541f, 0.51690674f, 0.51410276f, 0.51129305f, 0.50847787f,
+    0.50565726f, 0.50283122f, 0.50000006f, 0.49716327f, 0.49432117f,
+    0.49147379f, 0.48862115f, 0.48576325f, 0.48290038f, 0.48003212f,
+    0.47715873f, 0.47428021f, 0.47139663f, 0.46850798f, 0.46561456f,
+    0.46271589f, 0.45981231f, 0.45690379f, 0.45399037f, 0.45107210f,
+    0.44814920f, 0.44522130f, 0.44228864f, 0.43935123f, 0.43640912f,
+    0.43346232f, 0.43051112f, 0.42755505f, 0.42459446f, 0.42162928f,
+    0.41865960f, 0.41568545f, 0.41270703f, 0.40972400f, 0.40673658f,
+    0.40374479f, 0.40074870f, 0.39774850f, 0.39474386f, 0.39173496f,
+    0.38872188f, 0.38570464f, 0.38268328f, 0.37965804f, 0.37662849f,
+    0.37359491f, 0.37055734f, 0.36751580f, 0.36447033f, 0.36142117f,
+    0.35836792f, 0.35531086f, 0.35224995f, 0.34918529f, 0.34611690f,
+    0.34304500f, 0.33996922f, 0.33688980f, 0.33380675f, 0.33072016f,
+    0.32763001f, 0.32453656f, 0.32143945f, 0.31833887f, 0.31523487f,
+    0.31212750f, 0.30901679f, 0.30590302f, 0.30278572f, 0.29966521f,
+    0.29654145f, 0.29341453f, 0.29028472f, 0.28715155f, 0.28401530f,
+    0.28087601f, 0.27773371f, 0.27458847f, 0.27144048f, 0.26828936f,
+    0.26513538f, 0.26197854f, 0.25881892f, 0.25565651f, 0.25249159f,
+    0.24932374f, 0.24615324f, 0.24298008f, 0.23980433f, 0.23662600f,
+    0.23344538f, 0.23026201f, 0.22707619f, 0.22388794f, 0.22069728f,
+    0.21750426f, 0.21430916f, 0.21111152f, 0.20791161f, 0.20470949f,
+    0.20150517f, 0.19829892f, 0.19509031f, 0.19187963f, 0.18866688f,
+    0.18545210f, 0.18223536f, 0.17901689f, 0.17579627f, 0.17257376f,
+    0.16934940f, 0.16612324f, 0.16289529f, 0.15966584f, 0.15643445f,
+    0.15320137f, 0.14996666f, 0.14673033f, 0.14349243f, 0.14025325f,
+    0.13701232f, 0.13376991f, 0.13052608f, 0.12728085f, 0.12403426f,
+    0.12078657f, 0.11753736f, 0.11428688f, 0.11103519f, 0.10778230f,
+    0.10452849f, 0.10127334f, 0.09801710f, 0.09475980f, 0.09150149f,
+    0.08824220f, 0.08498220f, 0.08172106f, 0.07845904f, 0.07519618f,
+    0.07193252f, 0.06866808f, 0.06540315f, 0.06213728f, 0.05887074f,
+    0.05560357f, 0.05233581f, 0.04906749f, 0.04579888f, 0.04252954f,
+    0.03925974f, 0.03598953f, 0.03271893f, 0.02944798f, 0.02617695f,
+    0.02290541f, 0.01963361f, 0.01636161f, 0.01308943f, 0.00981712f,
+    0.00654493f, 0.00327244f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_WINDOWS_PRIVATE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/wpd_node.cc b/third_party/libwebrtc/modules/audio_processing/transient/wpd_node.cc
new file mode 100644
index 0000000000..2e0ee7e5b7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/wpd_node.cc
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/wpd_node.h"
+
+#include <math.h>
+#include <string.h>
+
+#include "common_audio/fir_filter.h"
+#include "common_audio/fir_filter_factory.h"
+#include "modules/audio_processing/transient/dyadic_decimator.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+WPDNode::WPDNode(size_t length,
+                 const float* coefficients,
+                 size_t coefficients_length)
+    :  // The data buffer has parent data length to be able to contain and
+       // filter it.
+      data_(new float[2 * length + 1]),
+      length_(length),
+      filter_(
+          CreateFirFilter(coefficients, coefficients_length, 2 * length + 1)) {
+  RTC_DCHECK_GT(length, 0);
+  RTC_DCHECK(coefficients);
+  RTC_DCHECK_GT(coefficients_length, 0);
+  memset(data_.get(), 0.f, (2 * length + 1) * sizeof(data_[0]));
+}
+
+WPDNode::~WPDNode() {}
+
+int WPDNode::Update(const float* parent_data, size_t parent_data_length) {
+  if (!parent_data || (parent_data_length / 2) != length_) {
+    return -1;
+  }
+
+  // Filter data.
+  filter_->Filter(parent_data, parent_data_length, data_.get());
+
+  // Decimate data.
+  const bool kOddSequence = true;
+  size_t output_samples = DyadicDecimate(data_.get(), parent_data_length,
+                                         kOddSequence, data_.get(), length_);
+  if (output_samples != length_) {
+    return -1;
+  }
+
+  // Get abs to all values.
+  for (size_t i = 0; i < length_; ++i) {
+    data_[i] = fabs(data_[i]);
+  }
+
+  return 0;
+}
+
+int WPDNode::set_data(const float* new_data, size_t length) {
+  if (!new_data || length != length_) {
+    return -1;
+  }
+  memcpy(data_.get(), new_data, length * sizeof(data_[0]));
+  return 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/wpd_node.h b/third_party/libwebrtc/modules/audio_processing/transient/wpd_node.h
new file mode 100644
index 0000000000..41614fab0f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/wpd_node.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_NODE_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_NODE_H_
+
+#include <memory>
+
+namespace webrtc {
+
+class FIRFilter;
+
+// A single node of a Wavelet Packet Decomposition (WPD) tree.
+class WPDNode {
+ public:
+  // Creates a WPDNode. The data vector will contain zeros. The filter will have
+  // the coefficients provided.
+  WPDNode(size_t length, const float* coefficients, size_t coefficients_length);
+  ~WPDNode();
+
+  // Updates the node data. `parent_data` / 2 must be equals to `length_`.
+  // Returns 0 if correct, and -1 otherwise.
+  int Update(const float* parent_data, size_t parent_data_length);
+
+  const float* data() const { return data_.get(); }
+  // Returns 0 if correct, and -1 otherwise.
+  int set_data(const float* new_data, size_t length);
+  size_t length() const { return length_; }
+
+ private:
+  std::unique_ptr<float[]> data_;
+  size_t length_;
+  std::unique_ptr<FIRFilter> filter_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_NODE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/wpd_node_unittest.cc b/third_party/libwebrtc/modules/audio_processing/transient/wpd_node_unittest.cc
new file mode 100644
index 0000000000..5f9238255c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/wpd_node_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/wpd_node.h"
+
+#include <string.h>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+static const size_t kDataLength = 5;
+static const float kTolerance = 0.0001f;
+
+static const size_t kParentDataLength = kDataLength * 2;
+static const float kParentData[kParentDataLength] = {1.f, 2.f, 3.f, 4.f, 5.f,
+                                                     6.f, 7.f, 8.f, 9.f, 10.f};
+
+static const float kCoefficients[] = {0.2f, -0.3f, 0.5f, -0.7f, 0.11f};
+static const size_t kCoefficientsLength =
+    sizeof(kCoefficients) / sizeof(kCoefficients[0]);
+
+TEST(WPDNodeTest, Accessors) {
+  WPDNode node(kDataLength, kCoefficients, kCoefficientsLength);
+  EXPECT_EQ(0, node.set_data(kParentData, kDataLength));
+  EXPECT_EQ(0, memcmp(node.data(), kParentData,
+                      kDataLength * sizeof(node.data()[0])));
+}
+
+TEST(WPDNodeTest, UpdateThatOnlyDecimates) {
+  const float kIndentyCoefficient = 1.f;
+  WPDNode node(kDataLength, &kIndentyCoefficient, 1);
+  EXPECT_EQ(0, node.Update(kParentData, kParentDataLength));
+  for (size_t i = 0; i < kDataLength; ++i) {
+    EXPECT_FLOAT_EQ(kParentData[i * 2 + 1], node.data()[i]);
+  }
+}
+
+TEST(WPDNodeTest, UpdateWithArbitraryDataAndArbitraryFilter) {
+  WPDNode node(kDataLength, kCoefficients, kCoefficientsLength);
+  EXPECT_EQ(0, node.Update(kParentData, kParentDataLength));
+  EXPECT_NEAR(0.1f, node.data()[0], kTolerance);
+  EXPECT_NEAR(0.2f, node.data()[1], kTolerance);
+  EXPECT_NEAR(0.18f, node.data()[2], kTolerance);
+  EXPECT_NEAR(0.56f, node.data()[3], kTolerance);
+  EXPECT_NEAR(0.94f, node.data()[4], kTolerance);
+}
+
+TEST(WPDNodeTest, ExpectedErrorReturnValue) {
+  WPDNode node(kDataLength, kCoefficients, kCoefficientsLength);
+  EXPECT_EQ(-1, node.Update(kParentData, kParentDataLength - 1));
+  EXPECT_EQ(-1, node.Update(NULL, kParentDataLength));
+  EXPECT_EQ(-1, node.set_data(kParentData, kDataLength - 1));
+  EXPECT_EQ(-1, node.set_data(NULL, kDataLength));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree.cc b/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree.cc
new file mode 100644
index 0000000000..c8aa615881
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/wpd_tree.h"
+
+#include <string.h>
+
+#include "modules/audio_processing/transient/wpd_node.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+WPDTree::WPDTree(size_t data_length,
+                 const float* high_pass_coefficients,
+                 const float* low_pass_coefficients,
+                 size_t coefficients_length,
+                 int levels)
+    : data_length_(data_length),
+      levels_(levels),
+      num_nodes_((1 << (levels + 1)) - 1) {
+  RTC_DCHECK_GT(data_length, (static_cast<size_t>(1) << levels));
+  RTC_DCHECK(high_pass_coefficients);
+  RTC_DCHECK(low_pass_coefficients);
+  RTC_DCHECK_GT(levels, 0);
+  // Size is 1 more, so we can use the array as 1-based. nodes_[0] is never
+  // allocated.
+  nodes_.reset(new std::unique_ptr<WPDNode>[num_nodes_ + 1]);
+
+  // Create the first node
+  const float kRootCoefficient = 1.f;  // Identity Coefficient.
+  nodes_[1].reset(new WPDNode(data_length, &kRootCoefficient, 1));
+  // Variables used to create the rest of the nodes.
+  size_t index = 1;
+  size_t index_left_child = 0;
+  size_t index_right_child = 0;
+
+  int num_nodes_at_curr_level = 0;
+
+  // Branching each node in each level to create its children. The last level is
+  // not branched (all the nodes of that level are leaves).
+  for (int current_level = 0; current_level < levels; ++current_level) {
+    num_nodes_at_curr_level = 1 << current_level;
+    for (int i = 0; i < num_nodes_at_curr_level; ++i) {
+      index = (1 << current_level) + i;
+      // Obtain the index of the current node children.
+      index_left_child = index * 2;
+      index_right_child = index_left_child + 1;
+      nodes_[index_left_child].reset(new WPDNode(nodes_[index]->length() / 2,
+                                                 low_pass_coefficients,
+                                                 coefficients_length));
+      nodes_[index_right_child].reset(new WPDNode(nodes_[index]->length() / 2,
+                                                  high_pass_coefficients,
+                                                  coefficients_length));
+    }
+  }
+}
+
+WPDTree::~WPDTree() {}
+
+WPDNode* WPDTree::NodeAt(int level, int index) {
+  if (level < 0 || level > levels_ || index < 0 || index >= 1 << level) {
+    return NULL;
+  }
+
+  return nodes_[(1 << level) + index].get();
+}
+
+int WPDTree::Update(const float* data, size_t data_length) {
+  if (!data || data_length != data_length_) {
+    return -1;
+  }
+
+  // Update the root node.
+  int update_result = nodes_[1]->set_data(data, data_length);
+  if (update_result != 0) {
+    return -1;
+  }
+
+  // Variables used to update the rest of the nodes.
+  size_t index = 1;
+  size_t index_left_child = 0;
+  size_t index_right_child = 0;
+
+  int num_nodes_at_curr_level = 0;
+
+  for (int current_level = 0; current_level < levels_; ++current_level) {
+    num_nodes_at_curr_level = 1 << current_level;
+    for (int i = 0; i < num_nodes_at_curr_level; ++i) {
+      index = (1 << current_level) + i;
+      // Obtain the index of the current node children.
+      index_left_child = index * 2;
+      index_right_child = index_left_child + 1;
+
+      update_result = nodes_[index_left_child]->Update(nodes_[index]->data(),
+                                                       nodes_[index]->length());
+      if (update_result != 0) {
+        return -1;
+      }
+
+      update_result = nodes_[index_right_child]->Update(
+          nodes_[index]->data(), nodes_[index]->length());
+      if (update_result != 0) {
+        return -1;
+      }
+    }
+  }
+
+  return 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree.h b/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree.h
new file mode 100644
index 0000000000..13cb8d9c2f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_TREE_H_
+#define MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_TREE_H_
+
+#include <stddef.h>
+
+#include <memory>
+
+#include "modules/audio_processing/transient/wpd_node.h"
+
+namespace webrtc {
+
+// Tree of a Wavelet Packet Decomposition (WPD).
+//
+// The root node contains all the data provided; for each node in the tree, the
+// left child contains the approximation coefficients extracted from the node,
+// and the right child contains the detail coefficients.
+// It preserves its state, so it can be multiple-called.
+//
+// The number of nodes in the tree will be 2 ^ levels - 1.
+//
+// Implementation details: Since the tree always will be a complete binary tree,
+// it is implemented using a single linear array instead of managing the
+// relationships in each node. For convience is better to use a array that
+// starts in 1 (instead of 0). Taking that into account, the following formulas
+// apply:
+// Root node index: 1.
+// Node(Level, Index in that level): 2 ^ Level + (Index in that level).
+// Left Child: Current node index * 2.
+// Right Child: Current node index * 2 + 1.
+// Parent: Current Node Index / 2 (Integer division).
+class WPDTree {
+ public:
+  // Creates a WPD tree using the data length and coefficients provided.
+  WPDTree(size_t data_length,
+          const float* high_pass_coefficients,
+          const float* low_pass_coefficients,
+          size_t coefficients_length,
+          int levels);
+  ~WPDTree();
+
+  // Returns the number of nodes at any given level.
+  static int NumberOfNodesAtLevel(int level) { return 1 << level; }
+
+  // Returns a pointer to the node at the given level and index(of that level).
+  // Level goes from 0 to levels().
+  // Index goes from 0 to the number of NumberOfNodesAtLevel(level) - 1.
+  //
+  // You can use the following formulas to get any node within the tree:
+  // Notation: (Level, Index of node in that level).
+  // Root node: (0/0).
+  // Left Child: (Current node level + 1, Current node index * 2).
+  // Right Child: (Current node level + 1, Current node index * 2 + 1).
+  // Parent: (Current node level - 1, Current node index / 2) (Integer division)
+  //
+  // If level or index are out of bounds the function will return NULL.
+  WPDNode* NodeAt(int level, int index);
+
+  // Updates all the nodes of the tree with the new data. `data_length` must be
+  // teh same that was used for the creation of the tree.
+  // Returns 0 if correct, and -1 otherwise.
+  int Update(const float* data, size_t data_length);
+
+  // Returns the total number of levels below the root. Root is cosidered level
+  // 0.
+  int levels() const { return levels_; }
+
+  // Returns the total number of nodes.
+  int num_nodes() const { return num_nodes_; }
+
+  // Returns the total number of leaves.
+  int num_leaves() const { return 1 << levels_; }
+
+ private:
+  size_t data_length_;
+  int levels_;
+  int num_nodes_;
+  std::unique_ptr<std::unique_ptr<WPDNode>[]> nodes_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_TREE_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree_unittest.cc b/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
new file mode 100644
index 0000000000..bf3ff987d7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
@@ -0,0 +1,177 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/transient/wpd_tree.h"
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h"
+#include "modules/audio_processing/transient/file_utils.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TEST(WPDTreeTest, Construction) {
+  const size_t kTestBufferSize = 100;
+  const int kLevels = 5;
+  const int kExpectedNumberOfNodes = (1 << (kLevels + 1)) - 1;
+
+  float test_buffer[kTestBufferSize];
+  memset(test_buffer, 0.f, kTestBufferSize * sizeof(*test_buffer));
+  float test_coefficients[] = {1.f, 2.f, 3.f, 4.f, 5.f};
+  const size_t kTestCoefficientsLength =
+      sizeof(test_coefficients) / sizeof(test_coefficients[0]);
+  WPDTree tree(kTestBufferSize, test_coefficients, test_coefficients,
+               kTestCoefficientsLength, kLevels);
+  ASSERT_EQ(kExpectedNumberOfNodes, tree.num_nodes());
+  // Checks for NodeAt(level, index).
+  int nodes_at_level = 0;
+  for (int level = 0; level <= kLevels; ++level) {
+    nodes_at_level = 1 << level;
+    for (int i = 0; i < nodes_at_level; ++i) {
+      ASSERT_TRUE(NULL != tree.NodeAt(level, i));
+    }
+    // Out of bounds.
+    EXPECT_EQ(NULL, tree.NodeAt(level, -1));
+    EXPECT_EQ(NULL, tree.NodeAt(level, -12));
+    EXPECT_EQ(NULL, tree.NodeAt(level, nodes_at_level));
+    EXPECT_EQ(NULL, tree.NodeAt(level, nodes_at_level + 5));
+  }
+  // Out of bounds.
+  EXPECT_EQ(NULL, tree.NodeAt(-1, 0));
+  EXPECT_EQ(NULL, tree.NodeAt(-12, 0));
+  EXPECT_EQ(NULL, tree.NodeAt(kLevels + 1, 0));
+  EXPECT_EQ(NULL, tree.NodeAt(kLevels + 5, 0));
+  // Checks for Update().
+  EXPECT_EQ(0, tree.Update(test_buffer, kTestBufferSize));
+  EXPECT_EQ(-1, tree.Update(NULL, kTestBufferSize));
+  EXPECT_EQ(-1, tree.Update(test_buffer, kTestBufferSize - 1));
+}
+
+// This test is for the correctness of the tree.
+// Checks the results from the Matlab equivalent, it is done comparing the
+// results that are stored in the output files from Matlab.
+// It also writes the results in its own set of files in the out directory.
+// Matlab and output files contain all the results in double precision (Little
+// endian) appended.
+#if defined(WEBRTC_IOS)
+TEST(WPDTreeTest, DISABLED_CorrectnessBasedOnMatlabFiles) {
+#else
+TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
+#endif
+  // 10 ms at 16000 Hz.
+  const size_t kTestBufferSize = 160;
+  const int kLevels = 3;
+  const int kLeaves = 1 << kLevels;
+  const size_t kLeavesSamples = kTestBufferSize >> kLevels;
+  // Create tree with Discrete Meyer Wavelet Coefficients.
+  WPDTree tree(kTestBufferSize, kDaubechies8HighPassCoefficients,
+               kDaubechies8LowPassCoefficients, kDaubechies8CoefficientsLength,
+               kLevels);
+  // Allocate and open all matlab and out files.
+  FileWrapper matlab_files_data[kLeaves];
+  FileWrapper out_files_data[kLeaves];
+
+  for (int i = 0; i < kLeaves; ++i) {
+    // Matlab files.
+    rtc::StringBuilder matlab_stream;
+    matlab_stream << "audio_processing/transient/wpd" << i;
+    std::string matlab_string = test::ResourcePath(matlab_stream.str(), "dat");
+    matlab_files_data[i] = FileWrapper::OpenReadOnly(matlab_string);
+
+    bool file_opened = matlab_files_data[i].is_open();
+    ASSERT_TRUE(file_opened) << "File could not be opened.\n" << matlab_string;
+
+    // Out files.
+    rtc::StringBuilder out_stream;
+    out_stream << test::OutputPath() << "wpd_" << i << ".out";
+    std::string out_string = out_stream.str();
+
+    out_files_data[i] = FileWrapper::OpenWriteOnly(out_string);
+
+    file_opened = out_files_data[i].is_open();
+    ASSERT_TRUE(file_opened) << "File could not be opened.\n" << out_string;
+  }
+
+  // Prepare the test file.
+  std::string test_file_name = test::ResourcePath(
+      "audio_processing/transient/ajm-macbook-1-spke16m", "pcm");
+
+  FileWrapper test_file = FileWrapper::OpenReadOnly(test_file_name);
+
+  bool file_opened = test_file.is_open();
+  ASSERT_TRUE(file_opened) << "File could not be opened.\n" << test_file_name;
+
+  float test_buffer[kTestBufferSize];
+
+  // Only the first frames of the audio file are tested. The matlab files also
+  // only contains information about the first frames.
+  const size_t kMaxFramesToTest = 100;
+  const float kTolerance = 0.03f;
+
+  size_t frames_read = 0;
+
+  // Read first buffer from the PCM test file.
+  size_t file_samples_read =
+      ReadInt16FromFileToFloatBuffer(&test_file, kTestBufferSize, test_buffer);
+  while (file_samples_read > 0 && frames_read < kMaxFramesToTest) {
+    ++frames_read;
+
+    if (file_samples_read < kTestBufferSize) {
+      // Pad the rest of the buffer with zeros.
+      for (size_t i = file_samples_read; i < kTestBufferSize; ++i) {
+        test_buffer[i] = 0.0;
+      }
+    }
+    tree.Update(test_buffer, kTestBufferSize);
+    double matlab_buffer[kTestBufferSize];
+
+    // Compare results with data from the matlab test files.
+    for (int i = 0; i < kLeaves; ++i) {
+      // Compare data values
+      size_t matlab_samples_read = ReadDoubleBufferFromFile(
+          &matlab_files_data[i], kLeavesSamples, matlab_buffer);
+
+      ASSERT_EQ(kLeavesSamples, matlab_samples_read)
+          << "Matlab test files are malformed.\n"
+             "File: 3_"
+          << i;
+      // Get output data from the corresponding node
+      const float* node_data = tree.NodeAt(kLevels, i)->data();
+      // Compare with matlab files.
+      for (size_t j = 0; j < kLeavesSamples; ++j) {
+        EXPECT_NEAR(matlab_buffer[j], node_data[j], kTolerance)
+            << "\nLeaf: " << i << "\nSample: " << j
+            << "\nFrame: " << frames_read - 1;
+      }
+
+      // Write results to out files.
+      WriteFloatBufferToFile(&out_files_data[i], kLeavesSamples, node_data);
+    }
+
+    // Read next buffer from the PCM test file.
+    file_samples_read = ReadInt16FromFileToFloatBuffer(
+        &test_file, kTestBufferSize, test_buffer);
+  }
+
+  // Close all matlab and out files.
+  for (int i = 0; i < kLeaves; ++i) {
+    matlab_files_data[i].Close();
+    out_files_data[i].Close();
+  }
+
+  test_file.Close();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/utility/BUILD.gn
new file mode 100644
index 0000000000..4851e77b03
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/BUILD.gn
@@ -0,0 +1,79 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("cascaded_biquad_filter") {
+  sources = [
+    "cascaded_biquad_filter.cc",
+    "cascaded_biquad_filter.h",
+  ]
+  deps = [
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+  ]
+}
+
+rtc_library("legacy_delay_estimator") {
+  sources = [
+    "delay_estimator.cc",
+    "delay_estimator.h",
+    "delay_estimator_internal.h",
+    "delay_estimator_wrapper.cc",
+    "delay_estimator_wrapper.h",
+  ]
+  deps = [ "../../../rtc_base:checks" ]
+}
+
+rtc_library("pffft_wrapper") {
+  visibility = [ "../*" ]
+  sources = [
+    "pffft_wrapper.cc",
+    "pffft_wrapper.h",
+  ]
+  deps = [
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "//third_party/pffft",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("cascaded_biquad_filter_unittest") {
+    testonly = true
+
+    sources = [ "cascaded_biquad_filter_unittest.cc" ]
+    deps = [
+      ":cascaded_biquad_filter",
+      "../../../test:test_support",
+      "//testing/gtest",
+    ]
+  }
+
+  rtc_library("legacy_delay_estimator_unittest") {
+    testonly = true
+
+    sources = [ "delay_estimator_unittest.cc" ]
+    deps = [
+      ":legacy_delay_estimator",
+      "../../../test:test_support",
+      "//testing/gtest",
+    ]
+  }
+
+  rtc_library("pffft_wrapper_unittest") {
+    testonly = true
+    sources = [ "pffft_wrapper_unittest.cc" ]
+    deps = [
+      ":pffft_wrapper",
+      "../../../test:test_support",
+      "//testing/gtest",
+      "//third_party/pffft",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/DEPS b/third_party/libwebrtc/modules/audio_processing/utility/DEPS
new file mode 100644
index 0000000000..c72d810b24
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/DEPS
@@ -0,0 +1,3 @@
+include_rules = [
+  "+third_party/pffft",
+]
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter.cc b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter.cc
new file mode 100644
index 0000000000..0d236ce0be
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/utility/cascaded_biquad_filter.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+CascadedBiQuadFilter::BiQuadParam::BiQuadParam(std::complex<float> zero,
+                                               std::complex<float> pole,
+                                               float gain,
+                                               bool mirror_zero_along_i_axis)
+    : zero(zero),
+      pole(pole),
+      gain(gain),
+      mirror_zero_along_i_axis(mirror_zero_along_i_axis) {}
+
+CascadedBiQuadFilter::BiQuadParam::BiQuadParam(const BiQuadParam&) = default;
+
+CascadedBiQuadFilter::BiQuad::BiQuad(
+    const CascadedBiQuadFilter::BiQuadParam& param)
+    : x(), y() {
+  float z_r = std::real(param.zero);
+  float z_i = std::imag(param.zero);
+  float p_r = std::real(param.pole);
+  float p_i = std::imag(param.pole);
+  float gain = param.gain;
+
+  if (param.mirror_zero_along_i_axis) {
+    // Assuming zeroes at z_r and -z_r.
+    RTC_DCHECK(z_i == 0.f);
+    coefficients.b[0] = gain * 1.f;
+    coefficients.b[1] = 0.f;
+    coefficients.b[2] = gain * -(z_r * z_r);
+  } else {
+    // Assuming zeros at (z_r + z_i*i) and (z_r - z_i*i).
+    coefficients.b[0] = gain * 1.f;
+    coefficients.b[1] = gain * -2.f * z_r;
+    coefficients.b[2] = gain * (z_r * z_r + z_i * z_i);
+  }
+
+  // Assuming poles at (p_r + p_i*i) and (p_r - p_i*i).
+  coefficients.a[0] = -2.f * p_r;
+  coefficients.a[1] = p_r * p_r + p_i * p_i;
+}
+
+void CascadedBiQuadFilter::BiQuad::BiQuad::Reset() {
+  x[0] = x[1] = y[0] = y[1] = 0.f;
+}
+
+CascadedBiQuadFilter::CascadedBiQuadFilter(
+    const CascadedBiQuadFilter::BiQuadCoefficients& coefficients,
+    size_t num_biquads)
+    : biquads_(num_biquads, BiQuad(coefficients)) {}
+
+CascadedBiQuadFilter::CascadedBiQuadFilter(
+    const std::vector<CascadedBiQuadFilter::BiQuadParam>& biquad_params) {
+  for (const auto& param : biquad_params) {
+    biquads_.push_back(BiQuad(param));
+  }
+}
+
+CascadedBiQuadFilter::~CascadedBiQuadFilter() = default;
+
+void CascadedBiQuadFilter::Process(rtc::ArrayView<const float> x,
+                                   rtc::ArrayView<float> y) {
+  if (biquads_.size() > 0) {
+    ApplyBiQuad(x, y, &biquads_[0]);
+    for (size_t k = 1; k < biquads_.size(); ++k) {
+      ApplyBiQuad(y, y, &biquads_[k]);
+    }
+  } else {
+    std::copy(x.begin(), x.end(), y.begin());
+  }
+}
+
+void CascadedBiQuadFilter::Process(rtc::ArrayView<float> y) {
+  for (auto& biquad : biquads_) {
+    ApplyBiQuad(y, y, &biquad);
+  }
+}
+
+void CascadedBiQuadFilter::Reset() {
+  for (auto& biquad : biquads_) {
+    biquad.Reset();
+  }
+}
+
+void CascadedBiQuadFilter::ApplyBiQuad(rtc::ArrayView<const float> x,
+                                       rtc::ArrayView<float> y,
+                                       CascadedBiQuadFilter::BiQuad* biquad) {
+  RTC_DCHECK_EQ(x.size(), y.size());
+  const float c_a_0 = biquad->coefficients.a[0];
+  const float c_a_1 = biquad->coefficients.a[1];
+  const float c_b_0 = biquad->coefficients.b[0];
+  const float c_b_1 = biquad->coefficients.b[1];
+  const float c_b_2 = biquad->coefficients.b[2];
+  float m_x_0 = biquad->x[0];
+  float m_x_1 = biquad->x[1];
+  float m_y_0 = biquad->y[0];
+  float m_y_1 = biquad->y[1];
+  for (size_t k = 0; k < x.size(); ++k) {
+    const float tmp = x[k];
+    y[k] = c_b_0 * tmp + c_b_1 * m_x_0 + c_b_2 * m_x_1 - c_a_0 * m_y_0 -
+           c_a_1 * m_y_1;
+    m_x_1 = m_x_0;
+    m_x_0 = tmp;
+    m_y_1 = m_y_0;
+    m_y_0 = y[k];
+  }
+  biquad->x[0] = m_x_0;
+  biquad->x[1] = m_x_1;
+  biquad->y[0] = m_y_0;
+  biquad->y[1] = m_y_1;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter.h b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter.h
new file mode 100644
index 0000000000..120b52aa57
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_UTILITY_CASCADED_BIQUAD_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_UTILITY_CASCADED_BIQUAD_FILTER_H_
+
+#include <stddef.h>
+
+#include <complex>
+#include <vector>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Applies a number of biquads in a cascaded manner. The filter implementation
+// is direct form 1.
+class CascadedBiQuadFilter {
+ public:
+  struct BiQuadParam {
+    BiQuadParam(std::complex<float> zero,
+                std::complex<float> pole,
+                float gain,
+                bool mirror_zero_along_i_axis = false);
+    explicit BiQuadParam(const BiQuadParam&);
+    std::complex<float> zero;
+    std::complex<float> pole;
+    float gain;
+    bool mirror_zero_along_i_axis;
+  };
+
+  struct BiQuadCoefficients {
+    float b[3];
+    float a[2];
+  };
+
+  struct BiQuad {
+    explicit BiQuad(const BiQuadCoefficients& coefficients)
+        : coefficients(coefficients), x(), y() {}
+    explicit BiQuad(const CascadedBiQuadFilter::BiQuadParam& param);
+    void Reset();
+    BiQuadCoefficients coefficients;
+    float x[2];
+    float y[2];
+  };
+
+  CascadedBiQuadFilter(
+      const CascadedBiQuadFilter::BiQuadCoefficients& coefficients,
+      size_t num_biquads);
+  explicit CascadedBiQuadFilter(
+      const std::vector<CascadedBiQuadFilter::BiQuadParam>& biquad_params);
+  ~CascadedBiQuadFilter();
+  CascadedBiQuadFilter(const CascadedBiQuadFilter&) = delete;
+  CascadedBiQuadFilter& operator=(const CascadedBiQuadFilter&) = delete;
+
+  // Applies the biquads on the values in x in order to form the output in y.
+  void Process(rtc::ArrayView<const float> x, rtc::ArrayView<float> y);
+  // Applies the biquads on the values in y in an in-place manner.
+  void Process(rtc::ArrayView<float> y);
+  // Resets the filter to its initial state.
+  void Reset();
+
+ private:
+  void ApplyBiQuad(rtc::ArrayView<const float> x,
+                   rtc::ArrayView<float> y,
+                   CascadedBiQuadFilter::BiQuad* biquad);
+
+  std::vector<BiQuad> biquads_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_UTILITY_CASCADED_BIQUAD_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_gn/moz.build
new file mode 100644
index 0000000000..8cf3aaefeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("cascaded_biquad_filter_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc
new file mode 100644
index 0000000000..ff7022dba4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/cascaded_biquad_filter_unittest.cc
@@ -0,0 +1,157 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/utility/cascaded_biquad_filter.h"
+
+#include <vector>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// Coefficients for a second order Butterworth high-pass filter with cutoff
+// frequency 100 Hz.
+const CascadedBiQuadFilter::BiQuadCoefficients kHighPassFilterCoefficients = {
+    {0.97261f, -1.94523f, 0.97261f},
+    {-1.94448f, 0.94598f}};
+
+const CascadedBiQuadFilter::BiQuadCoefficients kTransparentCoefficients = {
+    {1.f, 0.f, 0.f},
+    {0.f, 0.f}};
+
+const CascadedBiQuadFilter::BiQuadCoefficients kBlockingCoefficients = {
+    {0.f, 0.f, 0.f},
+    {0.f, 0.f}};
+
+std::vector<float> CreateInputWithIncreasingValues(size_t vector_length) {
+  std::vector<float> v(vector_length);
+  for (size_t k = 0; k < v.size(); ++k) {
+    v[k] = k;
+  }
+  return v;
+}
+
+}  // namespace
+
+// Verifies that the filter applies an effect which removes the input signal.
+// The test also verifies that the in-place Process API call works as intended.
+TEST(CascadedBiquadFilter, BlockingConfiguration) {
+  std::vector<float> values = CreateInputWithIncreasingValues(1000);
+
+  CascadedBiQuadFilter filter(kBlockingCoefficients, 1);
+  filter.Process(values);
+
+  EXPECT_EQ(std::vector<float>(1000, 0.f), values);
+}
+
+// Verifies that the filter is able to form a zero-mean output from a
+// non-zeromean input signal when coefficients for a high-pass filter are
+// applied. The test also verifies that the filter works with multiple biquads.
+TEST(CascadedBiquadFilter, HighPassConfiguration) {
+  std::vector<float> values(1000);
+  for (size_t k = 0; k < values.size(); ++k) {
+    values[k] = 1.f;
+  }
+
+  CascadedBiQuadFilter filter(kHighPassFilterCoefficients, 2);
+  filter.Process(values);
+
+  for (size_t k = values.size() / 2; k < values.size(); ++k) {
+    EXPECT_NEAR(0.f, values[k], 1e-4);
+  }
+}
+
+// Verifies that the reset functionality works as intended.
+TEST(CascadedBiquadFilter, HighPassConfigurationResetFunctionality) {
+  CascadedBiQuadFilter filter(kHighPassFilterCoefficients, 2);
+
+  std::vector<float> values1(100, 1.f);
+  filter.Process(values1);
+
+  filter.Reset();
+
+  std::vector<float> values2(100, 1.f);
+  filter.Process(values2);
+
+  for (size_t k = 0; k < values1.size(); ++k) {
+    EXPECT_EQ(values1[k], values2[k]);
+  }
+}
+
+// Verifies that the filter is able to produce a transparent effect with no
+// impact on the data when the proper coefficients are applied. The test also
+// verifies that the non-in-place Process API call works as intended.
+TEST(CascadedBiquadFilter, TransparentConfiguration) {
+  const std::vector<float> input = CreateInputWithIncreasingValues(1000);
+  std::vector<float> output(input.size());
+
+  CascadedBiQuadFilter filter(kTransparentCoefficients, 1);
+  filter.Process(input, output);
+
+  EXPECT_EQ(input, output);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Verifies that the check of the lengths for the input and output works for the
+// non-in-place call.
+TEST(CascadedBiquadFilterDeathTest, InputSizeCheckVerification) {
+  const std::vector<float> input = CreateInputWithIncreasingValues(10);
+  std::vector<float> output(input.size() - 1);
+
+  CascadedBiQuadFilter filter(kTransparentCoefficients, 1);
+  EXPECT_DEATH(filter.Process(input, output), "");
+}
+#endif
+
+// Verifies the conversion from zero, pole, gain to filter coefficients for
+// lowpass filter.
+TEST(CascadedBiquadFilter, BiQuadParamLowPass) {
+  CascadedBiQuadFilter::BiQuadParam param(
+      {-1.0f, 0.0f}, {0.23146901f, 0.39514232f}, 0.1866943331163784f);
+  CascadedBiQuadFilter::BiQuad filter(param);
+  const float epsilon = 1e-6f;
+  EXPECT_NEAR(filter.coefficients.b[0], 0.18669433f, epsilon);
+  EXPECT_NEAR(filter.coefficients.b[1], 0.37338867f, epsilon);
+  EXPECT_NEAR(filter.coefficients.b[2], 0.18669433f, epsilon);
+  EXPECT_NEAR(filter.coefficients.a[0], -0.46293803f, epsilon);
+  EXPECT_NEAR(filter.coefficients.a[1], 0.20971536f, epsilon);
+}
+
+// Verifies the conversion from zero, pole, gain to filter coefficients for
+// highpass filter.
+TEST(CascadedBiquadFilter, BiQuadParamHighPass) {
+  CascadedBiQuadFilter::BiQuadParam param(
+      {1.0f, 0.0f}, {0.72712179f, 0.21296904f}, 0.75707637533388494f);
+  CascadedBiQuadFilter::BiQuad filter(param);
+  const float epsilon = 1e-6f;
+  EXPECT_NEAR(filter.coefficients.b[0], 0.75707638f, epsilon);
+  EXPECT_NEAR(filter.coefficients.b[1], -1.51415275f, epsilon);
+  EXPECT_NEAR(filter.coefficients.b[2], 0.75707638f, epsilon);
+  EXPECT_NEAR(filter.coefficients.a[0], -1.45424359f, epsilon);
+  EXPECT_NEAR(filter.coefficients.a[1], 0.57406192f, epsilon);
+}
+
+// Verifies the conversion from zero, pole, gain to filter coefficients for
+// bandpass filter.
+TEST(CascadedBiquadFilter, BiQuadParamBandPass) {
+  CascadedBiQuadFilter::BiQuadParam param(
+      {1.0f, 0.0f}, {1.11022302e-16f, 0.71381051f}, 0.2452372752527856f, true);
+  CascadedBiQuadFilter::BiQuad filter(param);
+  const float epsilon = 1e-6f;
+  EXPECT_NEAR(filter.coefficients.b[0], 0.24523728f, epsilon);
+  EXPECT_NEAR(filter.coefficients.b[1], 0.f, epsilon);
+  EXPECT_NEAR(filter.coefficients.b[2], -0.24523728f, epsilon);
+  EXPECT_NEAR(filter.coefficients.a[0], -2.22044605e-16f, epsilon);
+  EXPECT_NEAR(filter.coefficients.a[1], 5.09525449e-01f, epsilon);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator.cc b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator.cc
new file mode 100644
index 0000000000..6868392f6f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator.cc
@@ -0,0 +1,708 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/utility/delay_estimator.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Number of right shifts for scaling is linearly depending on number of bits in
+// the far-end binary spectrum.
+static const int kShiftsAtZero = 13;  // Right shifts at zero binary spectrum.
+static const int kShiftsLinearSlope = 3;
+
+static const int32_t kProbabilityOffset = 1024;      // 2 in Q9.
+static const int32_t kProbabilityLowerLimit = 8704;  // 17 in Q9.
+static const int32_t kProbabilityMinSpread = 2816;   // 5.5 in Q9.
+
+// Robust validation settings
+static const float kHistogramMax = 3000.f;
+static const float kLastHistogramMax = 250.f;
+static const float kMinHistogramThreshold = 1.5f;
+static const int kMinRequiredHits = 10;
+static const int kMaxHitsWhenPossiblyNonCausal = 10;
+static const int kMaxHitsWhenPossiblyCausal = 1000;
+static const float kQ14Scaling = 1.f / (1 << 14);  // Scaling by 2^14 to get Q0.
+static const float kFractionSlope = 0.05f;
+static const float kMinFractionWhenPossiblyCausal = 0.5f;
+static const float kMinFractionWhenPossiblyNonCausal = 0.25f;
+
+}  // namespace
+
+// Counts and returns number of bits of a 32-bit word.
+static int BitCount(uint32_t u32) {
+  uint32_t tmp =
+      u32 - ((u32 >> 1) & 033333333333) - ((u32 >> 2) & 011111111111);
+  tmp = ((tmp + (tmp >> 3)) & 030707070707);
+  tmp = (tmp + (tmp >> 6));
+  tmp = (tmp + (tmp >> 12) + (tmp >> 24)) & 077;
+
+  return ((int)tmp);
+}
+
+// Compares the `binary_vector` with all rows of the `binary_matrix` and counts
+// per row the number of times they have the same value.
+//
+// Inputs:
+//      - binary_vector     : binary "vector" stored in a long
+//      - binary_matrix     : binary "matrix" stored as a vector of long
+//      - matrix_size       : size of binary "matrix"
+//
+// Output:
+//      - bit_counts        : "Vector" stored as a long, containing for each
+//                            row the number of times the matrix row and the
+//                            input vector have the same value
+//
+static void BitCountComparison(uint32_t binary_vector,
+                               const uint32_t* binary_matrix,
+                               int matrix_size,
+                               int32_t* bit_counts) {
+  int n = 0;
+
+  // Compare `binary_vector` with all rows of the `binary_matrix`
+  for (; n < matrix_size; n++) {
+    bit_counts[n] = (int32_t)BitCount(binary_vector ^ binary_matrix[n]);
+  }
+}
+
+// Collects necessary statistics for the HistogramBasedValidation().  This
+// function has to be called prior to calling HistogramBasedValidation().  The
+// statistics updated and used by the HistogramBasedValidation() are:
+//  1. the number of `candidate_hits`, which states for how long we have had the
+//     same `candidate_delay`
+//  2. the `histogram` of candidate delays over time.  This histogram is
+//     weighted with respect to a reliability measure and time-varying to cope
+//     with possible delay shifts.
+// For further description see commented code.
+//
+// Inputs:
+//  - candidate_delay   : The delay to validate.
+//  - valley_depth_q14  : The cost function has a valley/minimum at the
+//                        `candidate_delay` location.  `valley_depth_q14` is the
+//                        cost function difference between the minimum and
+//                        maximum locations.  The value is in the Q14 domain.
+//  - valley_level_q14  : Is the cost function value at the minimum, in Q14.
+static void UpdateRobustValidationStatistics(BinaryDelayEstimator* self,
+                                             int candidate_delay,
+                                             int32_t valley_depth_q14,
+                                             int32_t valley_level_q14) {
+  const float valley_depth = valley_depth_q14 * kQ14Scaling;
+  float decrease_in_last_set = valley_depth;
+  const int max_hits_for_slow_change = (candidate_delay < self->last_delay)
+                                           ? kMaxHitsWhenPossiblyNonCausal
+                                           : kMaxHitsWhenPossiblyCausal;
+  int i = 0;
+
+  RTC_DCHECK_EQ(self->history_size, self->farend->history_size);
+  // Reset `candidate_hits` if we have a new candidate.
+  if (candidate_delay != self->last_candidate_delay) {
+    self->candidate_hits = 0;
+    self->last_candidate_delay = candidate_delay;
+  }
+  self->candidate_hits++;
+
+  // The `histogram` is updated differently across the bins.
+  // 1. The `candidate_delay` histogram bin is increased with the
+  //    `valley_depth`, which is a simple measure of how reliable the
+  //    `candidate_delay` is.  The histogram is not increased above
+  //    `kHistogramMax`.
+  self->histogram[candidate_delay] += valley_depth;
+  if (self->histogram[candidate_delay] > kHistogramMax) {
+    self->histogram[candidate_delay] = kHistogramMax;
+  }
+  // 2. The histogram bins in the neighborhood of `candidate_delay` are
+  //    unaffected.  The neighborhood is defined as x + {-2, -1, 0, 1}.
+  // 3. The histogram bins in the neighborhood of `last_delay` are decreased
+  //    with `decrease_in_last_set`.  This value equals the difference between
+  //    the cost function values at the locations `candidate_delay` and
+  //    `last_delay` until we reach `max_hits_for_slow_change` consecutive hits
+  //    at the `candidate_delay`.  If we exceed this amount of hits the
+  //    `candidate_delay` is a "potential" candidate and we start decreasing
+  //    these histogram bins more rapidly with `valley_depth`.
+  if (self->candidate_hits < max_hits_for_slow_change) {
+    decrease_in_last_set =
+        (self->mean_bit_counts[self->compare_delay] - valley_level_q14) *
+        kQ14Scaling;
+  }
+  // 4. All other bins are decreased with `valley_depth`.
+  // TODO(bjornv): Investigate how to make this loop more efficient.  Split up
+  // the loop?  Remove parts that doesn't add too much.
+  for (i = 0; i < self->history_size; ++i) {
+    int is_in_last_set = (i >= self->last_delay - 2) &&
+                         (i <= self->last_delay + 1) && (i != candidate_delay);
+    int is_in_candidate_set =
+        (i >= candidate_delay - 2) && (i <= candidate_delay + 1);
+    self->histogram[i] -=
+        decrease_in_last_set * is_in_last_set +
+        valley_depth * (!is_in_last_set && !is_in_candidate_set);
+    // 5. No histogram bin can go below 0.
+    if (self->histogram[i] < 0) {
+      self->histogram[i] = 0;
+    }
+  }
+}
+
+// Validates the `candidate_delay`, estimated in WebRtc_ProcessBinarySpectrum(),
+// based on a mix of counting concurring hits with a modified histogram
+// of recent delay estimates.  In brief a candidate is valid (returns 1) if it
+// is the most likely according to the histogram.  There are a couple of
+// exceptions that are worth mentioning:
+//  1. If the `candidate_delay` < `last_delay` it can be that we are in a
+//     non-causal state, breaking a possible echo control algorithm.  Hence, we
+//     open up for a quicker change by allowing the change even if the
+//     `candidate_delay` is not the most likely one according to the histogram.
+//  2. There's a minimum number of hits (kMinRequiredHits) and the histogram
+//     value has to reached a minimum (kMinHistogramThreshold) to be valid.
+//  3. The action is also depending on the filter length used for echo control.
+//     If the delay difference is larger than what the filter can capture, we
+//     also move quicker towards a change.
+// For further description see commented code.
+//
+// Input:
+//  - candidate_delay     : The delay to validate.
+//
+// Return value:
+//  - is_histogram_valid  : 1 - The `candidate_delay` is valid.
+//                          0 - Otherwise.
+static int HistogramBasedValidation(const BinaryDelayEstimator* self,
+                                    int candidate_delay) {
+  float fraction = 1.f;
+  float histogram_threshold = self->histogram[self->compare_delay];
+  const int delay_difference = candidate_delay - self->last_delay;
+  int is_histogram_valid = 0;
+
+  // The histogram based validation of `candidate_delay` is done by comparing
+  // the `histogram` at bin `candidate_delay` with a `histogram_threshold`.
+  // This `histogram_threshold` equals a `fraction` of the `histogram` at bin
+  // `last_delay`.  The `fraction` is a piecewise linear function of the
+  // `delay_difference` between the `candidate_delay` and the `last_delay`
+  // allowing for a quicker move if
+  //  i) a potential echo control filter can not handle these large differences.
+  // ii) keeping `last_delay` instead of updating to `candidate_delay` could
+  //     force an echo control into a non-causal state.
+  // We further require the histogram to have reached a minimum value of
+  // `kMinHistogramThreshold`.  In addition, we also require the number of
+  // `candidate_hits` to be more than `kMinRequiredHits` to remove spurious
+  // values.
+
+  // Calculate a comparison histogram value (`histogram_threshold`) that is
+  // depending on the distance between the `candidate_delay` and `last_delay`.
+  // TODO(bjornv): How much can we gain by turning the fraction calculation
+  // into tables?
+  if (delay_difference > self->allowed_offset) {
+    fraction = 1.f - kFractionSlope * (delay_difference - self->allowed_offset);
+    fraction = (fraction > kMinFractionWhenPossiblyCausal
+                    ? fraction
+                    : kMinFractionWhenPossiblyCausal);
+  } else if (delay_difference < 0) {
+    fraction =
+        kMinFractionWhenPossiblyNonCausal - kFractionSlope * delay_difference;
+    fraction = (fraction > 1.f ? 1.f : fraction);
+  }
+  histogram_threshold *= fraction;
+  histogram_threshold =
+      (histogram_threshold > kMinHistogramThreshold ? histogram_threshold
+                                                    : kMinHistogramThreshold);
+
+  is_histogram_valid =
+      (self->histogram[candidate_delay] >= histogram_threshold) &&
+      (self->candidate_hits > kMinRequiredHits);
+
+  return is_histogram_valid;
+}
+
+// Performs a robust validation of the `candidate_delay` estimated in
+// WebRtc_ProcessBinarySpectrum().  The algorithm takes the
+// `is_instantaneous_valid` and the `is_histogram_valid` and combines them
+// into a robust validation.  The HistogramBasedValidation() has to be called
+// prior to this call.
+// For further description on how the combination is done, see commented code.
+//
+// Inputs:
+//  - candidate_delay         : The delay to validate.
+//  - is_instantaneous_valid  : The instantaneous validation performed in
+//                              WebRtc_ProcessBinarySpectrum().
+//  - is_histogram_valid      : The histogram based validation.
+//
+// Return value:
+//  - is_robust               : 1 - The candidate_delay is valid according to a
+//                                  combination of the two inputs.
+//                            : 0 - Otherwise.
+static int RobustValidation(const BinaryDelayEstimator* self,
+                            int candidate_delay,
+                            int is_instantaneous_valid,
+                            int is_histogram_valid) {
+  int is_robust = 0;
+
+  // The final robust validation is based on the two algorithms; 1) the
+  // `is_instantaneous_valid` and 2) the histogram based with result stored in
+  // `is_histogram_valid`.
+  //   i) Before we actually have a valid estimate (`last_delay` == -2), we say
+  //      a candidate is valid if either algorithm states so
+  //      (`is_instantaneous_valid` OR `is_histogram_valid`).
+  is_robust =
+      (self->last_delay < 0) && (is_instantaneous_valid || is_histogram_valid);
+  //  ii) Otherwise, we need both algorithms to be certain
+  //      (`is_instantaneous_valid` AND `is_histogram_valid`)
+  is_robust |= is_instantaneous_valid && is_histogram_valid;
+  // iii) With one exception, i.e., the histogram based algorithm can overrule
+  //      the instantaneous one if `is_histogram_valid` = 1 and the histogram
+  //      is significantly strong.
+  is_robust |= is_histogram_valid &&
+               (self->histogram[candidate_delay] > self->last_delay_histogram);
+
+  return is_robust;
+}
+
+void WebRtc_FreeBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self) {
+  if (self == NULL) {
+    return;
+  }
+
+  free(self->binary_far_history);
+  self->binary_far_history = NULL;
+
+  free(self->far_bit_counts);
+  self->far_bit_counts = NULL;
+
+  free(self);
+}
+
+BinaryDelayEstimatorFarend* WebRtc_CreateBinaryDelayEstimatorFarend(
+    int history_size) {
+  BinaryDelayEstimatorFarend* self = NULL;
+
+  if (history_size > 1) {
+    // Sanity conditions fulfilled.
+    self = static_cast<BinaryDelayEstimatorFarend*>(
+        malloc(sizeof(BinaryDelayEstimatorFarend)));
+  }
+  if (self == NULL) {
+    return NULL;
+  }
+
+  self->history_size = 0;
+  self->binary_far_history = NULL;
+  self->far_bit_counts = NULL;
+  if (WebRtc_AllocateFarendBufferMemory(self, history_size) == 0) {
+    WebRtc_FreeBinaryDelayEstimatorFarend(self);
+    self = NULL;
+  }
+  return self;
+}
+
+int WebRtc_AllocateFarendBufferMemory(BinaryDelayEstimatorFarend* self,
+                                      int history_size) {
+  RTC_DCHECK(self);
+  // (Re-)Allocate memory for history buffers.
+  self->binary_far_history = static_cast<uint32_t*>(
+      realloc(self->binary_far_history,
+              history_size * sizeof(*self->binary_far_history)));
+  self->far_bit_counts = static_cast<int*>(realloc(
+      self->far_bit_counts, history_size * sizeof(*self->far_bit_counts)));
+  if ((self->binary_far_history == NULL) || (self->far_bit_counts == NULL)) {
+    history_size = 0;
+  }
+  // Fill with zeros if we have expanded the buffers.
+  if (history_size > self->history_size) {
+    int size_diff = history_size - self->history_size;
+    memset(&self->binary_far_history[self->history_size], 0,
+           sizeof(*self->binary_far_history) * size_diff);
+    memset(&self->far_bit_counts[self->history_size], 0,
+           sizeof(*self->far_bit_counts) * size_diff);
+  }
+  self->history_size = history_size;
+
+  return self->history_size;
+}
+
+void WebRtc_InitBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self) {
+  RTC_DCHECK(self);
+  memset(self->binary_far_history, 0, sizeof(uint32_t) * self->history_size);
+  memset(self->far_bit_counts, 0, sizeof(int) * self->history_size);
+}
+
+void WebRtc_SoftResetBinaryDelayEstimatorFarend(
+    BinaryDelayEstimatorFarend* self,
+    int delay_shift) {
+  int abs_shift = abs(delay_shift);
+  int shift_size = 0;
+  int dest_index = 0;
+  int src_index = 0;
+  int padding_index = 0;
+
+  RTC_DCHECK(self);
+  shift_size = self->history_size - abs_shift;
+  RTC_DCHECK_GT(shift_size, 0);
+  if (delay_shift == 0) {
+    return;
+  } else if (delay_shift > 0) {
+    dest_index = abs_shift;
+  } else if (delay_shift < 0) {
+    src_index = abs_shift;
+    padding_index = shift_size;
+  }
+
+  // Shift and zero pad buffers.
+  memmove(&self->binary_far_history[dest_index],
+          &self->binary_far_history[src_index],
+          sizeof(*self->binary_far_history) * shift_size);
+  memset(&self->binary_far_history[padding_index], 0,
+         sizeof(*self->binary_far_history) * abs_shift);
+  memmove(&self->far_bit_counts[dest_index], &self->far_bit_counts[src_index],
+          sizeof(*self->far_bit_counts) * shift_size);
+  memset(&self->far_bit_counts[padding_index], 0,
+         sizeof(*self->far_bit_counts) * abs_shift);
+}
+
+void WebRtc_AddBinaryFarSpectrum(BinaryDelayEstimatorFarend* handle,
+                                 uint32_t binary_far_spectrum) {
+  RTC_DCHECK(handle);
+  // Shift binary spectrum history and insert current `binary_far_spectrum`.
+  memmove(&(handle->binary_far_history[1]), &(handle->binary_far_history[0]),
+          (handle->history_size - 1) * sizeof(uint32_t));
+  handle->binary_far_history[0] = binary_far_spectrum;
+
+  // Shift history of far-end binary spectrum bit counts and insert bit count
+  // of current `binary_far_spectrum`.
+  memmove(&(handle->far_bit_counts[1]), &(handle->far_bit_counts[0]),
+          (handle->history_size - 1) * sizeof(int));
+  handle->far_bit_counts[0] = BitCount(binary_far_spectrum);
+}
+
+void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self) {
+  if (self == NULL) {
+    return;
+  }
+
+  free(self->mean_bit_counts);
+  self->mean_bit_counts = NULL;
+
+  free(self->bit_counts);
+  self->bit_counts = NULL;
+
+  free(self->binary_near_history);
+  self->binary_near_history = NULL;
+
+  free(self->histogram);
+  self->histogram = NULL;
+
+  // BinaryDelayEstimator does not have ownership of `farend`, hence we do not
+  // free the memory here. That should be handled separately by the user.
+  self->farend = NULL;
+
+  free(self);
+}
+
+BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator(
+    BinaryDelayEstimatorFarend* farend,
+    int max_lookahead) {
+  BinaryDelayEstimator* self = NULL;
+
+  if ((farend != NULL) && (max_lookahead >= 0)) {
+    // Sanity conditions fulfilled.
+    self = static_cast<BinaryDelayEstimator*>(
+        malloc(sizeof(BinaryDelayEstimator)));
+  }
+  if (self == NULL) {
+    return NULL;
+  }
+
+  self->farend = farend;
+  self->near_history_size = max_lookahead + 1;
+  self->history_size = 0;
+  self->robust_validation_enabled = 0;  // Disabled by default.
+  self->allowed_offset = 0;
+
+  self->lookahead = max_lookahead;
+
+  // Allocate memory for spectrum and history buffers.
+  self->mean_bit_counts = NULL;
+  self->bit_counts = NULL;
+  self->histogram = NULL;
+  self->binary_near_history = static_cast<uint32_t*>(
+      malloc((max_lookahead + 1) * sizeof(*self->binary_near_history)));
+  if (self->binary_near_history == NULL ||
+      WebRtc_AllocateHistoryBufferMemory(self, farend->history_size) == 0) {
+    WebRtc_FreeBinaryDelayEstimator(self);
+    self = NULL;
+  }
+
+  return self;
+}
+
+int WebRtc_AllocateHistoryBufferMemory(BinaryDelayEstimator* self,
+                                       int history_size) {
+  BinaryDelayEstimatorFarend* far = self->farend;
+  // (Re-)Allocate memory for spectrum and history buffers.
+  if (history_size != far->history_size) {
+    // Only update far-end buffers if we need.
+    history_size = WebRtc_AllocateFarendBufferMemory(far, history_size);
+  }
+  // The extra array element in `mean_bit_counts` and `histogram` is a dummy
+  // element only used while `last_delay` == -2, i.e., before we have a valid
+  // estimate.
+  self->mean_bit_counts = static_cast<int32_t*>(
+      realloc(self->mean_bit_counts,
+              (history_size + 1) * sizeof(*self->mean_bit_counts)));
+  self->bit_counts = static_cast<int32_t*>(
+      realloc(self->bit_counts, history_size * sizeof(*self->bit_counts)));
+  self->histogram = static_cast<float*>(
+      realloc(self->histogram, (history_size + 1) * sizeof(*self->histogram)));
+
+  if ((self->mean_bit_counts == NULL) || (self->bit_counts == NULL) ||
+      (self->histogram == NULL)) {
+    history_size = 0;
+  }
+  // Fill with zeros if we have expanded the buffers.
+  if (history_size > self->history_size) {
+    int size_diff = history_size - self->history_size;
+    memset(&self->mean_bit_counts[self->history_size], 0,
+           sizeof(*self->mean_bit_counts) * size_diff);
+    memset(&self->bit_counts[self->history_size], 0,
+           sizeof(*self->bit_counts) * size_diff);
+    memset(&self->histogram[self->history_size], 0,
+           sizeof(*self->histogram) * size_diff);
+  }
+  self->history_size = history_size;
+
+  return self->history_size;
+}
+
+void WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* self) {
+  int i = 0;
+  RTC_DCHECK(self);
+
+  memset(self->bit_counts, 0, sizeof(int32_t) * self->history_size);
+  memset(self->binary_near_history, 0,
+         sizeof(uint32_t) * self->near_history_size);
+  for (i = 0; i <= self->history_size; ++i) {
+    self->mean_bit_counts[i] = (20 << 9);  // 20 in Q9.
+    self->histogram[i] = 0.f;
+  }
+  self->minimum_probability = kMaxBitCountsQ9;          // 32 in Q9.
+  self->last_delay_probability = (int)kMaxBitCountsQ9;  // 32 in Q9.
+
+  // Default return value if we're unable to estimate. -1 is used for errors.
+  self->last_delay = -2;
+
+  self->last_candidate_delay = -2;
+  self->compare_delay = self->history_size;
+  self->candidate_hits = 0;
+  self->last_delay_histogram = 0.f;
+}
+
+int WebRtc_SoftResetBinaryDelayEstimator(BinaryDelayEstimator* self,
+                                         int delay_shift) {
+  int lookahead = 0;
+  RTC_DCHECK(self);
+  lookahead = self->lookahead;
+  self->lookahead -= delay_shift;
+  if (self->lookahead < 0) {
+    self->lookahead = 0;
+  }
+  if (self->lookahead > self->near_history_size - 1) {
+    self->lookahead = self->near_history_size - 1;
+  }
+  return lookahead - self->lookahead;
+}
+
+int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self,
+                                 uint32_t binary_near_spectrum) {
+  int i = 0;
+  int candidate_delay = -1;
+  int valid_candidate = 0;
+
+  int32_t value_best_candidate = kMaxBitCountsQ9;
+  int32_t value_worst_candidate = 0;
+  int32_t valley_depth = 0;
+
+  RTC_DCHECK(self);
+  if (self->farend->history_size != self->history_size) {
+    // Non matching history sizes.
+    return -1;
+  }
+  if (self->near_history_size > 1) {
+    // If we apply lookahead, shift near-end binary spectrum history. Insert
+    // current `binary_near_spectrum` and pull out the delayed one.
+    memmove(&(self->binary_near_history[1]), &(self->binary_near_history[0]),
+            (self->near_history_size - 1) * sizeof(uint32_t));
+    self->binary_near_history[0] = binary_near_spectrum;
+    binary_near_spectrum = self->binary_near_history[self->lookahead];
+  }
+
+  // Compare with delayed spectra and store the `bit_counts` for each delay.
+  BitCountComparison(binary_near_spectrum, self->farend->binary_far_history,
+                     self->history_size, self->bit_counts);
+
+  // Update `mean_bit_counts`, which is the smoothed version of `bit_counts`.
+  for (i = 0; i < self->history_size; i++) {
+    // `bit_counts` is constrained to [0, 32], meaning we can smooth with a
+    // factor up to 2^26. We use Q9.
+    int32_t bit_count = (self->bit_counts[i] << 9);  // Q9.
+
+    // Update `mean_bit_counts` only when far-end signal has something to
+    // contribute. If `far_bit_counts` is zero the far-end signal is weak and
+    // we likely have a poor echo condition, hence don't update.
+    if (self->farend->far_bit_counts[i] > 0) {
+      // Make number of right shifts piecewise linear w.r.t. `far_bit_counts`.
+      int shifts = kShiftsAtZero;
+      shifts -= (kShiftsLinearSlope * self->farend->far_bit_counts[i]) >> 4;
+      WebRtc_MeanEstimatorFix(bit_count, shifts, &(self->mean_bit_counts[i]));
+    }
+  }
+
+  // Find `candidate_delay`, `value_best_candidate` and `value_worst_candidate`
+  // of `mean_bit_counts`.
+  for (i = 0; i < self->history_size; i++) {
+    if (self->mean_bit_counts[i] < value_best_candidate) {
+      value_best_candidate = self->mean_bit_counts[i];
+      candidate_delay = i;
+    }
+    if (self->mean_bit_counts[i] > value_worst_candidate) {
+      value_worst_candidate = self->mean_bit_counts[i];
+    }
+  }
+  valley_depth = value_worst_candidate - value_best_candidate;
+
+  // The `value_best_candidate` is a good indicator on the probability of
+  // `candidate_delay` being an accurate delay (a small `value_best_candidate`
+  // means a good binary match). In the following sections we make a decision
+  // whether to update `last_delay` or not.
+  // 1) If the difference bit counts between the best and the worst delay
+  //    candidates is too small we consider the situation to be unreliable and
+  //    don't update `last_delay`.
+  // 2) If the situation is reliable we update `last_delay` if the value of the
+  //    best candidate delay has a value less than
+  //     i) an adaptive threshold `minimum_probability`, or
+  //    ii) this corresponding value `last_delay_probability`, but updated at
+  //        this time instant.
+
+  // Update `minimum_probability`.
+  if ((self->minimum_probability > kProbabilityLowerLimit) &&
+      (valley_depth > kProbabilityMinSpread)) {
+    // The "hard" threshold can't be lower than 17 (in Q9).
+    // The valley in the curve also has to be distinct, i.e., the
+    // difference between `value_worst_candidate` and `value_best_candidate` has
+    // to be large enough.
+    int32_t threshold = value_best_candidate + kProbabilityOffset;
+    if (threshold < kProbabilityLowerLimit) {
+      threshold = kProbabilityLowerLimit;
+    }
+    if (self->minimum_probability > threshold) {
+      self->minimum_probability = threshold;
+    }
+  }
+  // Update `last_delay_probability`.
+  // We use a Markov type model, i.e., a slowly increasing level over time.
+  self->last_delay_probability++;
+  // Validate `candidate_delay`.  We have a reliable instantaneous delay
+  // estimate if
+  //  1) The valley is distinct enough (`valley_depth` > `kProbabilityOffset`)
+  // and
+  //  2) The depth of the valley is deep enough
+  //      (`value_best_candidate` < `minimum_probability`)
+  //     and deeper than the best estimate so far
+  //      (`value_best_candidate` < `last_delay_probability`)
+  valid_candidate = ((valley_depth > kProbabilityOffset) &&
+                     ((value_best_candidate < self->minimum_probability) ||
+                      (value_best_candidate < self->last_delay_probability)));
+
+  // Check for nonstationary farend signal.
+  const bool non_stationary_farend =
+      std::any_of(self->farend->far_bit_counts,
+                  self->farend->far_bit_counts + self->history_size,
+                  [](int a) { return a > 0; });
+
+  if (non_stationary_farend) {
+    // Only update the validation statistics when the farend is nonstationary
+    // as the underlying estimates are otherwise frozen.
+    UpdateRobustValidationStatistics(self, candidate_delay, valley_depth,
+                                     value_best_candidate);
+  }
+
+  if (self->robust_validation_enabled) {
+    int is_histogram_valid = HistogramBasedValidation(self, candidate_delay);
+    valid_candidate = RobustValidation(self, candidate_delay, valid_candidate,
+                                       is_histogram_valid);
+  }
+
+  // Only update the delay estimate when the farend is nonstationary and when
+  // a valid delay candidate is available.
+  if (non_stationary_farend && valid_candidate) {
+    if (candidate_delay != self->last_delay) {
+      self->last_delay_histogram =
+          (self->histogram[candidate_delay] > kLastHistogramMax
+               ? kLastHistogramMax
+               : self->histogram[candidate_delay]);
+      // Adjust the histogram if we made a change to `last_delay`, though it was
+      // not the most likely one according to the histogram.
+      if (self->histogram[candidate_delay] <
+          self->histogram[self->compare_delay]) {
+        self->histogram[self->compare_delay] = self->histogram[candidate_delay];
+      }
+    }
+    self->last_delay = candidate_delay;
+    if (value_best_candidate < self->last_delay_probability) {
+      self->last_delay_probability = value_best_candidate;
+    }
+    self->compare_delay = self->last_delay;
+  }
+
+  return self->last_delay;
+}
+
+int WebRtc_binary_last_delay(BinaryDelayEstimator* self) {
+  RTC_DCHECK(self);
+  return self->last_delay;
+}
+
+float WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self) {
+  float quality = 0;
+  RTC_DCHECK(self);
+
+  if (self->robust_validation_enabled) {
+    // Simply a linear function of the histogram height at delay estimate.
+    quality = self->histogram[self->compare_delay] / kHistogramMax;
+  } else {
+    // Note that `last_delay_probability` states how deep the minimum of the
+    // cost function is, so it is rather an error probability.
+    quality = (float)(kMaxBitCountsQ9 - self->last_delay_probability) /
+              kMaxBitCountsQ9;
+    if (quality < 0) {
+      quality = 0;
+    }
+  }
+  return quality;
+}
+
+void WebRtc_MeanEstimatorFix(int32_t new_value,
+                             int factor,
+                             int32_t* mean_value) {
+  int32_t diff = new_value - *mean_value;
+
+  // mean_new = mean_value + ((new_value - mean_value) >> factor);
+  if (diff < 0) {
+    diff = -((-diff) >> factor);
+  } else {
+    diff = (diff >> factor);
+  }
+  *mean_value += diff;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator.h b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator.h
new file mode 100644
index 0000000000..b6fc36a759
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator.h
@@ -0,0 +1,257 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Performs delay estimation on binary converted spectra.
+// The return value is  0 - OK and -1 - Error, unless otherwise stated.
+
+#ifndef MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+static const int32_t kMaxBitCountsQ9 = (32 << 9);  // 32 matching bits in Q9.
+
+typedef struct {
+  // Pointer to bit counts.
+  int* far_bit_counts;
+  // Binary history variables.
+  uint32_t* binary_far_history;
+  int history_size;
+} BinaryDelayEstimatorFarend;
+
+typedef struct {
+  // Pointer to bit counts.
+  int32_t* mean_bit_counts;
+  // Array only used locally in ProcessBinarySpectrum() but whose size is
+  // determined at run-time.
+  int32_t* bit_counts;
+
+  // Binary history variables.
+  uint32_t* binary_near_history;
+  int near_history_size;
+  int history_size;
+
+  // Delay estimation variables.
+  int32_t minimum_probability;
+  int last_delay_probability;
+
+  // Delay memory.
+  int last_delay;
+
+  // Robust validation
+  int robust_validation_enabled;
+  int allowed_offset;
+  int last_candidate_delay;
+  int compare_delay;
+  int candidate_hits;
+  float* histogram;
+  float last_delay_histogram;
+
+  // For dynamically changing the lookahead when using SoftReset...().
+  int lookahead;
+
+  // Far-end binary spectrum history buffer etc.
+  BinaryDelayEstimatorFarend* farend;
+} BinaryDelayEstimator;
+
+// Releases the memory allocated by
+// WebRtc_CreateBinaryDelayEstimatorFarend(...).
+// Input:
+//    - self              : Pointer to the binary delay estimation far-end
+//                          instance which is the return value of
+//                          WebRtc_CreateBinaryDelayEstimatorFarend().
+//
+void WebRtc_FreeBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self);
+
+// Allocates the memory needed by the far-end part of the binary delay
+// estimation. The memory needs to be initialized separately through
+// WebRtc_InitBinaryDelayEstimatorFarend(...).
+//
+// Inputs:
+//      - history_size    : Size of the far-end binary spectrum history.
+//
+// Return value:
+//      - BinaryDelayEstimatorFarend*
+//                        : Created `handle`. If the memory can't be allocated
+//                          or if any of the input parameters are invalid NULL
+//                          is returned.
+//
+BinaryDelayEstimatorFarend* WebRtc_CreateBinaryDelayEstimatorFarend(
+    int history_size);
+
+// Re-allocates the buffers.
+//
+// Inputs:
+//      - self            : Pointer to the binary estimation far-end instance
+//                          which is the return value of
+//                          WebRtc_CreateBinaryDelayEstimatorFarend().
+//      - history_size    : Size of the far-end binary spectrum history.
+//
+// Return value:
+//      - history_size    : The history size allocated.
+int WebRtc_AllocateFarendBufferMemory(BinaryDelayEstimatorFarend* self,
+                                      int history_size);
+
+// Initializes the delay estimation far-end instance created with
+// WebRtc_CreateBinaryDelayEstimatorFarend(...).
+//
+// Input:
+//    - self              : Pointer to the delay estimation far-end instance.
+//
+// Output:
+//    - self              : Initialized far-end instance.
+//
+void WebRtc_InitBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self);
+
+// Soft resets the delay estimation far-end instance created with
+// WebRtc_CreateBinaryDelayEstimatorFarend(...).
+//
+// Input:
+//    - delay_shift   : The amount of blocks to shift history buffers.
+//
+void WebRtc_SoftResetBinaryDelayEstimatorFarend(
+    BinaryDelayEstimatorFarend* self,
+    int delay_shift);
+
+// Adds the binary far-end spectrum to the internal far-end history buffer. This
+// spectrum is used as reference when calculating the delay using
+// WebRtc_ProcessBinarySpectrum().
+//
+// Inputs:
+//    - self                  : Pointer to the delay estimation far-end
+//                              instance.
+//    - binary_far_spectrum   : Far-end binary spectrum.
+//
+// Output:
+//    - self                  : Updated far-end instance.
+//
+void WebRtc_AddBinaryFarSpectrum(BinaryDelayEstimatorFarend* self,
+                                 uint32_t binary_far_spectrum);
+
+// Releases the memory allocated by WebRtc_CreateBinaryDelayEstimator(...).
+//
+// Note that BinaryDelayEstimator utilizes BinaryDelayEstimatorFarend, but does
+// not take ownership of it, hence the BinaryDelayEstimator has to be torn down
+// before the far-end.
+//
+// Input:
+//    - self              : Pointer to the binary delay estimation instance
+//                          which is the return value of
+//                          WebRtc_CreateBinaryDelayEstimator().
+//
+void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self);
+
+// Allocates the memory needed by the binary delay estimation. The memory needs
+// to be initialized separately through WebRtc_InitBinaryDelayEstimator(...).
+//
+// See WebRtc_CreateDelayEstimator(..) in delay_estimator_wrapper.c for detailed
+// description.
+BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator(
+    BinaryDelayEstimatorFarend* farend,
+    int max_lookahead);
+
+// Re-allocates `history_size` dependent buffers. The far-end buffers will be
+// updated at the same time if needed.
+//
+// Input:
+//      - self            : Pointer to the binary estimation instance which is
+//                          the return value of
+//                          WebRtc_CreateBinaryDelayEstimator().
+//      - history_size    : Size of the history buffers.
+//
+// Return value:
+//      - history_size    : The history size allocated.
+int WebRtc_AllocateHistoryBufferMemory(BinaryDelayEstimator* self,
+                                       int history_size);
+
+// Initializes the delay estimation instance created with
+// WebRtc_CreateBinaryDelayEstimator(...).
+//
+// Input:
+//    - self              : Pointer to the delay estimation instance.
+//
+// Output:
+//    - self              : Initialized instance.
+//
+void WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* self);
+
+// Soft resets the delay estimation instance created with
+// WebRtc_CreateBinaryDelayEstimator(...).
+//
+// Input:
+//    - delay_shift   : The amount of blocks to shift history buffers.
+//
+// Return value:
+//    - actual_shifts : The actual number of shifts performed.
+//
+int WebRtc_SoftResetBinaryDelayEstimator(BinaryDelayEstimator* self,
+                                         int delay_shift);
+
+// Estimates and returns the delay between the binary far-end and binary near-
+// end spectra. It is assumed the binary far-end spectrum has been added using
+// WebRtc_AddBinaryFarSpectrum() prior to this call. The value will be offset by
+// the lookahead (i.e. the lookahead should be subtracted from the returned
+// value).
+//
+// Inputs:
+//    - self                  : Pointer to the delay estimation instance.
+//    - binary_near_spectrum  : Near-end binary spectrum of the current block.
+//
+// Output:
+//    - self                  : Updated instance.
+//
+// Return value:
+//    - delay                 :  >= 0 - Calculated delay value.
+//                              -2    - Insufficient data for estimation.
+//
+int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self,
+                                 uint32_t binary_near_spectrum);
+
+// Returns the last calculated delay updated by the function
+// WebRtc_ProcessBinarySpectrum(...).
+//
+// Input:
+//    - self                  : Pointer to the delay estimation instance.
+//
+// Return value:
+//    - delay                 :  >= 0 - Last calculated delay value
+//                              -2    - Insufficient data for estimation.
+//
+int WebRtc_binary_last_delay(BinaryDelayEstimator* self);
+
+// Returns the estimation quality of the last calculated delay updated by the
+// function WebRtc_ProcessBinarySpectrum(...). The estimation quality is a value
+// in the interval [0, 1].  The higher the value, the better the quality.
+//
+// Return value:
+//    - delay_quality         :  >= 0 - Estimation quality of last calculated
+//                                      delay value.
+float WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self);
+
+// Updates the `mean_value` recursively with a step size of 2^-`factor`. This
+// function is used internally in the Binary Delay Estimator as well as the
+// Fixed point wrapper.
+//
+// Inputs:
+//    - new_value             : The new value the mean should be updated with.
+//    - factor                : The step size, in number of right shifts.
+//
+// Input/Output:
+//    - mean_value            : Pointer to the mean value.
+//
+void WebRtc_MeanEstimatorFix(int32_t new_value,
+                             int factor,
+                             int32_t* mean_value);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_internal.h b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_internal.h
new file mode 100644
index 0000000000..891e20027d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_internal.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Header file including the delay estimator handle used for testing.
+
+#ifndef MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_INTERNAL_H_
+
+#include "modules/audio_processing/utility/delay_estimator.h"
+
+namespace webrtc {
+
+typedef union {
+  float float_;
+  int32_t int32_;
+} SpectrumType;
+
+typedef struct {
+  // Pointers to mean values of spectrum.
+  SpectrumType* mean_far_spectrum;
+  // `mean_far_spectrum` initialization indicator.
+  int far_spectrum_initialized;
+
+  int spectrum_size;
+
+  // Far-end part of binary spectrum based delay estimation.
+  BinaryDelayEstimatorFarend* binary_farend;
+} DelayEstimatorFarend;
+
+typedef struct {
+  // Pointers to mean values of spectrum.
+  SpectrumType* mean_near_spectrum;
+  // `mean_near_spectrum` initialization indicator.
+  int near_spectrum_initialized;
+
+  int spectrum_size;
+
+  // Binary spectrum based delay estimator
+  BinaryDelayEstimator* binary_handle;
+} DelayEstimator;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_INTERNAL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_unittest.cc b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
new file mode 100644
index 0000000000..6052612ef3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
@@ -0,0 +1,621 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/utility/delay_estimator.h"
+
+#include "modules/audio_processing/utility/delay_estimator_internal.h"
+#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kSpectrumSize = 65;
+// Delay history sizes.
+constexpr int kMaxDelay = 100;
+constexpr int kLookahead = 10;
+constexpr int kHistorySize = kMaxDelay + kLookahead;
+// Length of binary spectrum sequence.
+constexpr int kSequenceLength = 400;
+
+const int kDifferentHistorySize = 3;
+const int kDifferentLookahead = 1;
+
+const int kEnable[] = {0, 1};
+const size_t kSizeEnable = sizeof(kEnable) / sizeof(*kEnable);
+
+class DelayEstimatorTest : public ::testing::Test {
+ protected:
+  DelayEstimatorTest();
+  void SetUp() override;
+  void TearDown() override;
+
+  void Init();
+  void InitBinary();
+  void VerifyDelay(BinaryDelayEstimator* binary_handle, int offset, int delay);
+  void RunBinarySpectra(BinaryDelayEstimator* binary1,
+                        BinaryDelayEstimator* binary2,
+                        int near_offset,
+                        int lookahead_offset,
+                        int far_offset);
+  void RunBinarySpectraTest(int near_offset,
+                            int lookahead_offset,
+                            int ref_robust_validation,
+                            int robust_validation);
+
+  void* handle_;
+  DelayEstimator* self_;
+  void* farend_handle_;
+  DelayEstimatorFarend* farend_self_;
+  BinaryDelayEstimator* binary_;
+  BinaryDelayEstimatorFarend* binary_farend_;
+  int spectrum_size_;
+  // Dummy input spectra.
+  float far_f_[kSpectrumSize];
+  float near_f_[kSpectrumSize];
+  uint16_t far_u16_[kSpectrumSize];
+  uint16_t near_u16_[kSpectrumSize];
+  uint32_t binary_spectrum_[kSequenceLength + kHistorySize];
+};
+
+DelayEstimatorTest::DelayEstimatorTest()
+    : handle_(NULL),
+      self_(NULL),
+      farend_handle_(NULL),
+      farend_self_(NULL),
+      binary_(NULL),
+      binary_farend_(NULL),
+      spectrum_size_(kSpectrumSize) {
+  // Dummy input data are set with more or less arbitrary non-zero values.
+  memset(far_f_, 1, sizeof(far_f_));
+  memset(near_f_, 2, sizeof(near_f_));
+  memset(far_u16_, 1, sizeof(far_u16_));
+  memset(near_u16_, 2, sizeof(near_u16_));
+  // Construct a sequence of binary spectra used to verify delay estimate. The
+  // `kSequenceLength` has to be long enough for the delay estimation to leave
+  // the initialized state.
+  binary_spectrum_[0] = 1;
+  for (int i = 1; i < (kSequenceLength + kHistorySize); i++) {
+    binary_spectrum_[i] = 3 * binary_spectrum_[i - 1];
+  }
+}
+
+void DelayEstimatorTest::SetUp() {
+  farend_handle_ =
+      WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, kHistorySize);
+  ASSERT_TRUE(farend_handle_ != NULL);
+  farend_self_ = reinterpret_cast<DelayEstimatorFarend*>(farend_handle_);
+  handle_ = WebRtc_CreateDelayEstimator(farend_handle_, kLookahead);
+  ASSERT_TRUE(handle_ != NULL);
+  self_ = reinterpret_cast<DelayEstimator*>(handle_);
+  binary_farend_ = WebRtc_CreateBinaryDelayEstimatorFarend(kHistorySize);
+  ASSERT_TRUE(binary_farend_ != NULL);
+  binary_ = WebRtc_CreateBinaryDelayEstimator(binary_farend_, kLookahead);
+  ASSERT_TRUE(binary_ != NULL);
+}
+
+void DelayEstimatorTest::TearDown() {
+  WebRtc_FreeDelayEstimator(handle_);
+  handle_ = NULL;
+  self_ = NULL;
+  WebRtc_FreeDelayEstimatorFarend(farend_handle_);
+  farend_handle_ = NULL;
+  farend_self_ = NULL;
+  WebRtc_FreeBinaryDelayEstimator(binary_);
+  binary_ = NULL;
+  WebRtc_FreeBinaryDelayEstimatorFarend(binary_farend_);
+  binary_farend_ = NULL;
+}
+
+void DelayEstimatorTest::Init() {
+  // Initialize Delay Estimator
+  EXPECT_EQ(0, WebRtc_InitDelayEstimatorFarend(farend_handle_));
+  EXPECT_EQ(0, WebRtc_InitDelayEstimator(handle_));
+  // Verify initialization.
+  EXPECT_EQ(0, farend_self_->far_spectrum_initialized);
+  EXPECT_EQ(0, self_->near_spectrum_initialized);
+  EXPECT_EQ(-2, WebRtc_last_delay(handle_));  // Delay in initial state.
+  EXPECT_FLOAT_EQ(0, WebRtc_last_delay_quality(handle_));  // Zero quality.
+}
+
+void DelayEstimatorTest::InitBinary() {
+  // Initialize Binary Delay Estimator (far-end part).
+  WebRtc_InitBinaryDelayEstimatorFarend(binary_farend_);
+  // Initialize Binary Delay Estimator
+  WebRtc_InitBinaryDelayEstimator(binary_);
+  // Verify initialization. This does not guarantee a complete check, since
+  // `last_delay` may be equal to -2 before initialization if done on the fly.
+  EXPECT_EQ(-2, binary_->last_delay);
+}
+
+void DelayEstimatorTest::VerifyDelay(BinaryDelayEstimator* binary_handle,
+                                     int offset,
+                                     int delay) {
+  // Verify that we WebRtc_binary_last_delay() returns correct delay.
+  EXPECT_EQ(delay, WebRtc_binary_last_delay(binary_handle));
+
+  if (delay != -2) {
+    // Verify correct delay estimate. In the non-causal case the true delay
+    // is equivalent with the `offset`.
+    EXPECT_EQ(offset, delay);
+  }
+}
+
+void DelayEstimatorTest::RunBinarySpectra(BinaryDelayEstimator* binary1,
+                                          BinaryDelayEstimator* binary2,
+                                          int near_offset,
+                                          int lookahead_offset,
+                                          int far_offset) {
+  int different_validations =
+      binary1->robust_validation_enabled ^ binary2->robust_validation_enabled;
+  WebRtc_InitBinaryDelayEstimatorFarend(binary_farend_);
+  WebRtc_InitBinaryDelayEstimator(binary1);
+  WebRtc_InitBinaryDelayEstimator(binary2);
+  // Verify initialization. This does not guarantee a complete check, since
+  // `last_delay` may be equal to -2 before initialization if done on the fly.
+  EXPECT_EQ(-2, binary1->last_delay);
+  EXPECT_EQ(-2, binary2->last_delay);
+  for (int i = kLookahead; i < (kSequenceLength + kLookahead); i++) {
+    WebRtc_AddBinaryFarSpectrum(binary_farend_,
+                                binary_spectrum_[i + far_offset]);
+    int delay_1 = WebRtc_ProcessBinarySpectrum(binary1, binary_spectrum_[i]);
+    int delay_2 = WebRtc_ProcessBinarySpectrum(
+        binary2, binary_spectrum_[i - near_offset]);
+
+    VerifyDelay(binary1, far_offset + kLookahead, delay_1);
+    VerifyDelay(binary2,
+                far_offset + kLookahead + lookahead_offset + near_offset,
+                delay_2);
+    // Expect the two delay estimates to be offset by `lookahead_offset` +
+    // `near_offset` when we have left the initial state.
+    if ((delay_1 != -2) && (delay_2 != -2)) {
+      EXPECT_EQ(delay_1, delay_2 - lookahead_offset - near_offset);
+    }
+    // For the case of identical signals `delay_1` and `delay_2` should match
+    // all the time, unless one of them has robust validation turned on.  In
+    // that case the robust validation leaves the initial state faster.
+    if ((near_offset == 0) && (lookahead_offset == 0)) {
+      if (!different_validations) {
+        EXPECT_EQ(delay_1, delay_2);
+      } else {
+        if (binary1->robust_validation_enabled) {
+          EXPECT_GE(delay_1, delay_2);
+        } else {
+          EXPECT_GE(delay_2, delay_1);
+        }
+      }
+    }
+  }
+  // Verify that we have left the initialized state.
+  EXPECT_NE(-2, WebRtc_binary_last_delay(binary1));
+  EXPECT_LT(0, WebRtc_binary_last_delay_quality(binary1));
+  EXPECT_NE(-2, WebRtc_binary_last_delay(binary2));
+  EXPECT_LT(0, WebRtc_binary_last_delay_quality(binary2));
+}
+
+void DelayEstimatorTest::RunBinarySpectraTest(int near_offset,
+                                              int lookahead_offset,
+                                              int ref_robust_validation,
+                                              int robust_validation) {
+  BinaryDelayEstimator* binary2 = WebRtc_CreateBinaryDelayEstimator(
+      binary_farend_, kLookahead + lookahead_offset);
+  // Verify the delay for both causal and non-causal systems. For causal systems
+  // the delay is equivalent with a positive `offset` of the far-end sequence.
+  // For non-causal systems the delay is equivalent with a negative `offset` of
+  // the far-end sequence.
+  binary_->robust_validation_enabled = ref_robust_validation;
+  binary2->robust_validation_enabled = robust_validation;
+  for (int offset = -kLookahead;
+       offset < kMaxDelay - lookahead_offset - near_offset; offset++) {
+    RunBinarySpectra(binary_, binary2, near_offset, lookahead_offset, offset);
+  }
+  WebRtc_FreeBinaryDelayEstimator(binary2);
+  binary2 = NULL;
+  binary_->robust_validation_enabled = 0;  // Reset reference.
+}
+
+TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) {
+  // In this test we verify correct error returns on invalid API calls.
+
+  // WebRtc_CreateDelayEstimatorFarend() and WebRtc_CreateDelayEstimator()
+  // should return a NULL pointer on invalid input values.
+  // Make sure we have a non-NULL value at start, so we can detect NULL after
+  // create failure.
+  void* handle = farend_handle_;
+  handle = WebRtc_CreateDelayEstimatorFarend(33, kHistorySize);
+  EXPECT_TRUE(handle == NULL);
+  handle = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, 1);
+  EXPECT_TRUE(handle == NULL);
+
+  handle = handle_;
+  handle = WebRtc_CreateDelayEstimator(NULL, kLookahead);
+  EXPECT_TRUE(handle == NULL);
+  handle = WebRtc_CreateDelayEstimator(farend_handle_, -1);
+  EXPECT_TRUE(handle == NULL);
+
+  // WebRtc_InitDelayEstimatorFarend() and WebRtc_InitDelayEstimator() should
+  // return -1 if we have a NULL pointer as `handle`.
+  EXPECT_EQ(-1, WebRtc_InitDelayEstimatorFarend(NULL));
+  EXPECT_EQ(-1, WebRtc_InitDelayEstimator(NULL));
+
+  // WebRtc_AddFarSpectrumFloat() should return -1 if we have:
+  // 1) NULL pointer as `handle`.
+  // 2) NULL pointer as far-end spectrum.
+  // 3) Incorrect spectrum size.
+  EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(NULL, far_f_, spectrum_size_));
+  // Use `farend_handle_` which is properly created at SetUp().
+  EXPECT_EQ(-1,
+            WebRtc_AddFarSpectrumFloat(farend_handle_, NULL, spectrum_size_));
+  EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_,
+                                           spectrum_size_ + 1));
+
+  // WebRtc_AddFarSpectrumFix() should return -1 if we have:
+  // 1) NULL pointer as `handle`.
+  // 2) NULL pointer as far-end spectrum.
+  // 3) Incorrect spectrum size.
+  // 4) Too high precision in far-end spectrum (Q-domain > 15).
+  EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(NULL, far_u16_, spectrum_size_, 0));
+  EXPECT_EQ(-1,
+            WebRtc_AddFarSpectrumFix(farend_handle_, NULL, spectrum_size_, 0));
+  EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_,
+                                         spectrum_size_ + 1, 0));
+  EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_,
+                                         spectrum_size_, 16));
+
+  // WebRtc_set_history_size() should return -1 if:
+  // 1) `handle` is a NULL.
+  // 2) `history_size` <= 1.
+  EXPECT_EQ(-1, WebRtc_set_history_size(NULL, 1));
+  EXPECT_EQ(-1, WebRtc_set_history_size(handle_, 1));
+  // WebRtc_history_size() should return -1 if:
+  // 1) NULL pointer input.
+  EXPECT_EQ(-1, WebRtc_history_size(NULL));
+  // 2) there is a mismatch between history size.
+  void* tmp_handle = WebRtc_CreateDelayEstimator(farend_handle_, kHistorySize);
+  EXPECT_EQ(0, WebRtc_InitDelayEstimator(tmp_handle));
+  EXPECT_EQ(kDifferentHistorySize,
+            WebRtc_set_history_size(tmp_handle, kDifferentHistorySize));
+  EXPECT_EQ(kDifferentHistorySize, WebRtc_history_size(tmp_handle));
+  EXPECT_EQ(kHistorySize, WebRtc_set_history_size(handle_, kHistorySize));
+  EXPECT_EQ(-1, WebRtc_history_size(tmp_handle));
+
+  // WebRtc_set_lookahead() should return -1 if we try a value outside the
+  /// buffer.
+  EXPECT_EQ(-1, WebRtc_set_lookahead(handle_, kLookahead + 1));
+  EXPECT_EQ(-1, WebRtc_set_lookahead(handle_, -1));
+
+  // WebRtc_set_allowed_offset() should return -1 if we have:
+  // 1) NULL pointer as `handle`.
+  // 2) `allowed_offset` < 0.
+  EXPECT_EQ(-1, WebRtc_set_allowed_offset(NULL, 0));
+  EXPECT_EQ(-1, WebRtc_set_allowed_offset(handle_, -1));
+
+  EXPECT_EQ(-1, WebRtc_get_allowed_offset(NULL));
+
+  // WebRtc_enable_robust_validation() should return -1 if we have:
+  // 1) NULL pointer as `handle`.
+  // 2) Incorrect `enable` value (not 0 or 1).
+  EXPECT_EQ(-1, WebRtc_enable_robust_validation(NULL, kEnable[0]));
+  EXPECT_EQ(-1, WebRtc_enable_robust_validation(handle_, -1));
+  EXPECT_EQ(-1, WebRtc_enable_robust_validation(handle_, 2));
+
+  // WebRtc_is_robust_validation_enabled() should return -1 if we have NULL
+  // pointer as `handle`.
+  EXPECT_EQ(-1, WebRtc_is_robust_validation_enabled(NULL));
+
+  // WebRtc_DelayEstimatorProcessFloat() should return -1 if we have:
+  // 1) NULL pointer as `handle`.
+  // 2) NULL pointer as near-end spectrum.
+  // 3) Incorrect spectrum size.
+  // 4) Non matching history sizes if multiple delay estimators using the same
+  //    far-end reference.
+  EXPECT_EQ(-1,
+            WebRtc_DelayEstimatorProcessFloat(NULL, near_f_, spectrum_size_));
+  // Use `handle_` which is properly created at SetUp().
+  EXPECT_EQ(-1,
+            WebRtc_DelayEstimatorProcessFloat(handle_, NULL, spectrum_size_));
+  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(handle_, near_f_,
+                                                  spectrum_size_ + 1));
+  // `tmp_handle` is already in a non-matching state.
+  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(tmp_handle, near_f_,
+                                                  spectrum_size_));
+
+  // WebRtc_DelayEstimatorProcessFix() should return -1 if we have:
+  // 1) NULL pointer as `handle`.
+  // 2) NULL pointer as near-end spectrum.
+  // 3) Incorrect spectrum size.
+  // 4) Too high precision in near-end spectrum (Q-domain > 15).
+  // 5) Non matching history sizes if multiple delay estimators using the same
+  //    far-end reference.
+  EXPECT_EQ(
+      -1, WebRtc_DelayEstimatorProcessFix(NULL, near_u16_, spectrum_size_, 0));
+  EXPECT_EQ(-1,
+            WebRtc_DelayEstimatorProcessFix(handle_, NULL, spectrum_size_, 0));
+  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_,
+                                                spectrum_size_ + 1, 0));
+  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_,
+                                                spectrum_size_, 16));
+  // `tmp_handle` is already in a non-matching state.
+  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(tmp_handle, near_u16_,
+                                                spectrum_size_, 0));
+  WebRtc_FreeDelayEstimator(tmp_handle);
+
+  // WebRtc_last_delay() should return -1 if we have a NULL pointer as `handle`.
+  EXPECT_EQ(-1, WebRtc_last_delay(NULL));
+
+  // Free any local memory if needed.
+  WebRtc_FreeDelayEstimator(handle);
+}
+
+TEST_F(DelayEstimatorTest, VerifyAllowedOffset) {
+  // Is set to zero by default.
+  EXPECT_EQ(0, WebRtc_get_allowed_offset(handle_));
+  for (int i = 1; i >= 0; i--) {
+    EXPECT_EQ(0, WebRtc_set_allowed_offset(handle_, i));
+    EXPECT_EQ(i, WebRtc_get_allowed_offset(handle_));
+    Init();
+    // Unaffected over a reset.
+    EXPECT_EQ(i, WebRtc_get_allowed_offset(handle_));
+  }
+}
+
+TEST_F(DelayEstimatorTest, VerifyEnableRobustValidation) {
+  // Disabled by default.
+  EXPECT_EQ(0, WebRtc_is_robust_validation_enabled(handle_));
+  for (size_t i = 0; i < kSizeEnable; ++i) {
+    EXPECT_EQ(0, WebRtc_enable_robust_validation(handle_, kEnable[i]));
+    EXPECT_EQ(kEnable[i], WebRtc_is_robust_validation_enabled(handle_));
+    Init();
+    // Unaffected over a reset.
+    EXPECT_EQ(kEnable[i], WebRtc_is_robust_validation_enabled(handle_));
+  }
+}
+
+TEST_F(DelayEstimatorTest, InitializedSpectrumAfterProcess) {
+  // In this test we verify that the mean spectra are initialized after first
+  // time we call WebRtc_AddFarSpectrum() and Process() respectively. The test
+  // also verifies the state is not left for zero spectra.
+  const float kZerosFloat[kSpectrumSize] = {0.0};
+  const uint16_t kZerosU16[kSpectrumSize] = {0};
+
+  // For floating point operations, process one frame and verify initialization
+  // flag.
+  Init();
+  EXPECT_EQ(0, WebRtc_AddFarSpectrumFloat(farend_handle_, kZerosFloat,
+                                          spectrum_size_));
+  EXPECT_EQ(0, farend_self_->far_spectrum_initialized);
+  EXPECT_EQ(0,
+            WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_, spectrum_size_));
+  EXPECT_EQ(1, farend_self_->far_spectrum_initialized);
+  EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFloat(handle_, kZerosFloat,
+                                                  spectrum_size_));
+  EXPECT_EQ(0, self_->near_spectrum_initialized);
+  EXPECT_EQ(
+      -2, WebRtc_DelayEstimatorProcessFloat(handle_, near_f_, spectrum_size_));
+  EXPECT_EQ(1, self_->near_spectrum_initialized);
+
+  // For fixed point operations, process one frame and verify initialization
+  // flag.
+  Init();
+  EXPECT_EQ(0, WebRtc_AddFarSpectrumFix(farend_handle_, kZerosU16,
+                                        spectrum_size_, 0));
+  EXPECT_EQ(0, farend_self_->far_spectrum_initialized);
+  EXPECT_EQ(
+      0, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_, spectrum_size_, 0));
+  EXPECT_EQ(1, farend_self_->far_spectrum_initialized);
+  EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFix(handle_, kZerosU16,
+                                                spectrum_size_, 0));
+  EXPECT_EQ(0, self_->near_spectrum_initialized);
+  EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_,
+                                                spectrum_size_, 0));
+  EXPECT_EQ(1, self_->near_spectrum_initialized);
+}
+
+TEST_F(DelayEstimatorTest, CorrectLastDelay) {
+  // In this test we verify that we get the correct last delay upon valid call.
+  // We simply process the same data until we leave the initialized state
+  // (`last_delay` = -2). Then we compare the Process() output with the
+  // last_delay() call.
+
+  // TODO(bjornv): Update quality values for robust validation.
+  int last_delay = 0;
+  // Floating point operations.
+  Init();
+  for (int i = 0; i < 200; i++) {
+    EXPECT_EQ(
+        0, WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_, spectrum_size_));
+    last_delay =
+        WebRtc_DelayEstimatorProcessFloat(handle_, near_f_, spectrum_size_);
+    if (last_delay != -2) {
+      EXPECT_EQ(last_delay, WebRtc_last_delay(handle_));
+      if (!WebRtc_is_robust_validation_enabled(handle_)) {
+        EXPECT_FLOAT_EQ(7203.f / kMaxBitCountsQ9,
+                        WebRtc_last_delay_quality(handle_));
+      }
+      break;
+    }
+  }
+  // Verify that we have left the initialized state.
+  EXPECT_NE(-2, WebRtc_last_delay(handle_));
+  EXPECT_LT(0, WebRtc_last_delay_quality(handle_));
+
+  // Fixed point operations.
+  Init();
+  for (int i = 0; i < 200; i++) {
+    EXPECT_EQ(0, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_,
+                                          spectrum_size_, 0));
+    last_delay =
+        WebRtc_DelayEstimatorProcessFix(handle_, near_u16_, spectrum_size_, 0);
+    if (last_delay != -2) {
+      EXPECT_EQ(last_delay, WebRtc_last_delay(handle_));
+      if (!WebRtc_is_robust_validation_enabled(handle_)) {
+        EXPECT_FLOAT_EQ(7203.f / kMaxBitCountsQ9,
+                        WebRtc_last_delay_quality(handle_));
+      }
+      break;
+    }
+  }
+  // Verify that we have left the initialized state.
+  EXPECT_NE(-2, WebRtc_last_delay(handle_));
+  EXPECT_LT(0, WebRtc_last_delay_quality(handle_));
+}
+
+TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfBinaryEstimatorFarend) {
+  // In this test we verify correct output on invalid API calls to the Binary
+  // Delay Estimator (far-end part).
+
+  BinaryDelayEstimatorFarend* binary = binary_farend_;
+  // WebRtc_CreateBinaryDelayEstimatorFarend() should return -1 if the input
+  // history size is less than 2. This is to make sure the buffer shifting
+  // applies properly.
+  // Make sure we have a non-NULL value at start, so we can detect NULL after
+  // create failure.
+  binary = WebRtc_CreateBinaryDelayEstimatorFarend(1);
+  EXPECT_TRUE(binary == NULL);
+}
+
+TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfBinaryEstimator) {
+  // In this test we verify correct output on invalid API calls to the Binary
+  // Delay Estimator.
+
+  BinaryDelayEstimator* binary_handle = binary_;
+  // WebRtc_CreateBinaryDelayEstimator() should return -1 if we have a NULL
+  // pointer as `binary_farend` or invalid input values. Upon failure, the
+  // `binary_handle` should be NULL.
+  // Make sure we have a non-NULL value at start, so we can detect NULL after
+  // create failure.
+  binary_handle = WebRtc_CreateBinaryDelayEstimator(NULL, kLookahead);
+  EXPECT_TRUE(binary_handle == NULL);
+  binary_handle = WebRtc_CreateBinaryDelayEstimator(binary_farend_, -1);
+  EXPECT_TRUE(binary_handle == NULL);
+}
+
+TEST_F(DelayEstimatorTest, MeanEstimatorFix) {
+  // In this test we verify that we update the mean value in correct direction
+  // only. With "direction" we mean increase or decrease.
+
+  int32_t mean_value = 4000;
+  int32_t mean_value_before = mean_value;
+  int32_t new_mean_value = mean_value * 2;
+
+  // Increasing `mean_value`.
+  WebRtc_MeanEstimatorFix(new_mean_value, 10, &mean_value);
+  EXPECT_LT(mean_value_before, mean_value);
+  EXPECT_GT(new_mean_value, mean_value);
+
+  // Decreasing `mean_value`.
+  new_mean_value = mean_value / 2;
+  mean_value_before = mean_value;
+  WebRtc_MeanEstimatorFix(new_mean_value, 10, &mean_value);
+  EXPECT_GT(mean_value_before, mean_value);
+  EXPECT_LT(new_mean_value, mean_value);
+}
+
+TEST_F(DelayEstimatorTest, ExactDelayEstimateMultipleNearSameSpectrum) {
+  // In this test we verify that we get the correct delay estimates if we shift
+  // the signal accordingly. We create two Binary Delay Estimators and feed them
+  // with the same signals, so they should output the same results.
+  // We verify both causal and non-causal delays.
+  // For these noise free signals, the robust validation should not have an
+  // impact, hence we turn robust validation on/off for both reference and
+  // delayed near end.
+
+  for (size_t i = 0; i < kSizeEnable; ++i) {
+    for (size_t j = 0; j < kSizeEnable; ++j) {
+      RunBinarySpectraTest(0, 0, kEnable[i], kEnable[j]);
+    }
+  }
+}
+
+TEST_F(DelayEstimatorTest, ExactDelayEstimateMultipleNearDifferentSpectrum) {
+  // In this test we use the same setup as above, but we now feed the two Binary
+  // Delay Estimators with different signals, so they should output different
+  // results.
+  // For these noise free signals, the robust validation should not have an
+  // impact, hence we turn robust validation on/off for both reference and
+  // delayed near end.
+
+  const int kNearOffset = 1;
+  for (size_t i = 0; i < kSizeEnable; ++i) {
+    for (size_t j = 0; j < kSizeEnable; ++j) {
+      RunBinarySpectraTest(kNearOffset, 0, kEnable[i], kEnable[j]);
+    }
+  }
+}
+
+TEST_F(DelayEstimatorTest, ExactDelayEstimateMultipleNearDifferentLookahead) {
+  // In this test we use the same setup as above, feeding the two Binary
+  // Delay Estimators with the same signals. The difference is that we create
+  // them with different lookahead.
+  // For these noise free signals, the robust validation should not have an
+  // impact, hence we turn robust validation on/off for both reference and
+  // delayed near end.
+
+  const int kLookaheadOffset = 1;
+  for (size_t i = 0; i < kSizeEnable; ++i) {
+    for (size_t j = 0; j < kSizeEnable; ++j) {
+      RunBinarySpectraTest(0, kLookaheadOffset, kEnable[i], kEnable[j]);
+    }
+  }
+}
+
+TEST_F(DelayEstimatorTest, AllowedOffsetNoImpactWhenRobustValidationDisabled) {
+  // The same setup as in ExactDelayEstimateMultipleNearSameSpectrum with the
+  // difference that `allowed_offset` is set for the reference binary delay
+  // estimator.
+
+  binary_->allowed_offset = 10;
+  RunBinarySpectraTest(0, 0, 0, 0);
+  binary_->allowed_offset = 0;  // Reset reference.
+}
+
+TEST_F(DelayEstimatorTest, VerifyLookaheadAtCreate) {
+  void* farend_handle =
+      WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, kMaxDelay);
+  ASSERT_TRUE(farend_handle != NULL);
+  void* handle = WebRtc_CreateDelayEstimator(farend_handle, kLookahead);
+  ASSERT_TRUE(handle != NULL);
+  EXPECT_EQ(kLookahead, WebRtc_lookahead(handle));
+  WebRtc_FreeDelayEstimator(handle);
+  WebRtc_FreeDelayEstimatorFarend(farend_handle);
+}
+
+TEST_F(DelayEstimatorTest, VerifyLookaheadIsSetAndKeptAfterInit) {
+  EXPECT_EQ(kLookahead, WebRtc_lookahead(handle_));
+  EXPECT_EQ(kDifferentLookahead,
+            WebRtc_set_lookahead(handle_, kDifferentLookahead));
+  EXPECT_EQ(kDifferentLookahead, WebRtc_lookahead(handle_));
+  EXPECT_EQ(0, WebRtc_InitDelayEstimatorFarend(farend_handle_));
+  EXPECT_EQ(kDifferentLookahead, WebRtc_lookahead(handle_));
+  EXPECT_EQ(0, WebRtc_InitDelayEstimator(handle_));
+  EXPECT_EQ(kDifferentLookahead, WebRtc_lookahead(handle_));
+}
+
+TEST_F(DelayEstimatorTest, VerifyHistorySizeAtCreate) {
+  EXPECT_EQ(kHistorySize, WebRtc_history_size(handle_));
+}
+
+TEST_F(DelayEstimatorTest, VerifyHistorySizeIsSetAndKeptAfterInit) {
+  EXPECT_EQ(kHistorySize, WebRtc_history_size(handle_));
+  EXPECT_EQ(kDifferentHistorySize,
+            WebRtc_set_history_size(handle_, kDifferentHistorySize));
+  EXPECT_EQ(kDifferentHistorySize, WebRtc_history_size(handle_));
+  EXPECT_EQ(0, WebRtc_InitDelayEstimator(handle_));
+  EXPECT_EQ(kDifferentHistorySize, WebRtc_history_size(handle_));
+  EXPECT_EQ(0, WebRtc_InitDelayEstimatorFarend(farend_handle_));
+  EXPECT_EQ(kDifferentHistorySize, WebRtc_history_size(handle_));
+}
+
+// TODO(bjornv): Add tests for SoftReset...(...).
+
+}  // namespace
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
new file mode 100644
index 0000000000..3b1409cc0b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
@@ -0,0 +1,489 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "modules/audio_processing/utility/delay_estimator.h"
+#include "modules/audio_processing/utility/delay_estimator_internal.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Only bit `kBandFirst` through bit `kBandLast` are processed and
+// `kBandFirst` - `kBandLast` must be < 32.
+constexpr int kBandFirst = 12;
+constexpr int kBandLast = 43;
+
+static __inline uint32_t SetBit(uint32_t in, int pos) {
+  uint32_t mask = (1 << pos);
+  uint32_t out = (in | mask);
+
+  return out;
+}
+
+// Calculates the mean recursively. Same version as WebRtc_MeanEstimatorFix(),
+// but for float.
+//
+// Inputs:
+//    - new_value             : New additional value.
+//    - scale                 : Scale for smoothing (should be less than 1.0).
+//
+// Input/Output:
+//    - mean_value            : Pointer to the mean value for updating.
+//
+static void MeanEstimatorFloat(float new_value,
+                               float scale,
+                               float* mean_value) {
+  RTC_DCHECK_LT(scale, 1.0f);
+  *mean_value += (new_value - *mean_value) * scale;
+}
+
+// Computes the binary spectrum by comparing the input `spectrum` with a
+// `threshold_spectrum`. Float and fixed point versions.
+//
+// Inputs:
+//      - spectrum            : Spectrum of which the binary spectrum should be
+//                              calculated.
+//      - threshold_spectrum  : Threshold spectrum with which the input
+//                              spectrum is compared.
+// Return:
+//      - out                 : Binary spectrum.
+//
+static uint32_t BinarySpectrumFix(const uint16_t* spectrum,
+                                  SpectrumType* threshold_spectrum,
+                                  int q_domain,
+                                  int* threshold_initialized) {
+  int i = kBandFirst;
+  uint32_t out = 0;
+
+  RTC_DCHECK_LT(q_domain, 16);
+
+  if (!(*threshold_initialized)) {
+    // Set the `threshold_spectrum` to half the input `spectrum` as starting
+    // value. This speeds up the convergence.
+    for (i = kBandFirst; i <= kBandLast; i++) {
+      if (spectrum[i] > 0) {
+        // Convert input spectrum from Q(`q_domain`) to Q15.
+        int32_t spectrum_q15 = ((int32_t)spectrum[i]) << (15 - q_domain);
+        threshold_spectrum[i].int32_ = (spectrum_q15 >> 1);
+        *threshold_initialized = 1;
+      }
+    }
+  }
+  for (i = kBandFirst; i <= kBandLast; i++) {
+    // Convert input spectrum from Q(`q_domain`) to Q15.
+    int32_t spectrum_q15 = ((int32_t)spectrum[i]) << (15 - q_domain);
+    // Update the `threshold_spectrum`.
+    WebRtc_MeanEstimatorFix(spectrum_q15, 6, &(threshold_spectrum[i].int32_));
+    // Convert `spectrum` at current frequency bin to a binary value.
+    if (spectrum_q15 > threshold_spectrum[i].int32_) {
+      out = SetBit(out, i - kBandFirst);
+    }
+  }
+
+  return out;
+}
+
+static uint32_t BinarySpectrumFloat(const float* spectrum,
+                                    SpectrumType* threshold_spectrum,
+                                    int* threshold_initialized) {
+  int i = kBandFirst;
+  uint32_t out = 0;
+  const float kScale = 1 / 64.0;
+
+  if (!(*threshold_initialized)) {
+    // Set the `threshold_spectrum` to half the input `spectrum` as starting
+    // value. This speeds up the convergence.
+    for (i = kBandFirst; i <= kBandLast; i++) {
+      if (spectrum[i] > 0.0f) {
+        threshold_spectrum[i].float_ = (spectrum[i] / 2);
+        *threshold_initialized = 1;
+      }
+    }
+  }
+
+  for (i = kBandFirst; i <= kBandLast; i++) {
+    // Update the `threshold_spectrum`.
+    MeanEstimatorFloat(spectrum[i], kScale, &(threshold_spectrum[i].float_));
+    // Convert `spectrum` at current frequency bin to a binary value.
+    if (spectrum[i] > threshold_spectrum[i].float_) {
+      out = SetBit(out, i - kBandFirst);
+    }
+  }
+
+  return out;
+}
+
+void WebRtc_FreeDelayEstimatorFarend(void* handle) {
+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
+
+  if (handle == NULL) {
+    return;
+  }
+
+  free(self->mean_far_spectrum);
+  self->mean_far_spectrum = NULL;
+
+  WebRtc_FreeBinaryDelayEstimatorFarend(self->binary_farend);
+  self->binary_farend = NULL;
+
+  free(self);
+}
+
+void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) {
+  DelayEstimatorFarend* self = NULL;
+
+  // Check if the sub band used in the delay estimation is small enough to fit
+  // the binary spectra in a uint32_t.
+  static_assert(kBandLast - kBandFirst < 32, "");
+
+  if (spectrum_size >= kBandLast) {
+    self = static_cast<DelayEstimatorFarend*>(
+        malloc(sizeof(DelayEstimatorFarend)));
+  }
+
+  if (self != NULL) {
+    int memory_fail = 0;
+
+    // Allocate memory for the binary far-end spectrum handling.
+    self->binary_farend = WebRtc_CreateBinaryDelayEstimatorFarend(history_size);
+    memory_fail |= (self->binary_farend == NULL);
+
+    // Allocate memory for spectrum buffers.
+    self->mean_far_spectrum = static_cast<SpectrumType*>(
+        malloc(spectrum_size * sizeof(SpectrumType)));
+    memory_fail |= (self->mean_far_spectrum == NULL);
+
+    self->spectrum_size = spectrum_size;
+
+    if (memory_fail) {
+      WebRtc_FreeDelayEstimatorFarend(self);
+      self = NULL;
+    }
+  }
+
+  return self;
+}
+
+int WebRtc_InitDelayEstimatorFarend(void* handle) {
+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
+
+  if (self == NULL) {
+    return -1;
+  }
+
+  // Initialize far-end part of binary delay estimator.
+  WebRtc_InitBinaryDelayEstimatorFarend(self->binary_farend);
+
+  // Set averaged far and near end spectra to zero.
+  memset(self->mean_far_spectrum, 0,
+         sizeof(SpectrumType) * self->spectrum_size);
+  // Reset initialization indicators.
+  self->far_spectrum_initialized = 0;
+
+  return 0;
+}
+
+void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift) {
+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
+  RTC_DCHECK(self);
+  WebRtc_SoftResetBinaryDelayEstimatorFarend(self->binary_farend, delay_shift);
+}
+
+int WebRtc_AddFarSpectrumFix(void* handle,
+                             const uint16_t* far_spectrum,
+                             int spectrum_size,
+                             int far_q) {
+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
+  uint32_t binary_spectrum = 0;
+
+  if (self == NULL) {
+    return -1;
+  }
+  if (far_spectrum == NULL) {
+    // Empty far end spectrum.
+    return -1;
+  }
+  if (spectrum_size != self->spectrum_size) {
+    // Data sizes don't match.
+    return -1;
+  }
+  if (far_q > 15) {
+    // If `far_q` is larger than 15 we cannot guarantee no wrap around.
+    return -1;
+  }
+
+  // Get binary spectrum.
+  binary_spectrum = BinarySpectrumFix(far_spectrum, self->mean_far_spectrum,
+                                      far_q, &(self->far_spectrum_initialized));
+  WebRtc_AddBinaryFarSpectrum(self->binary_farend, binary_spectrum);
+
+  return 0;
+}
+
+int WebRtc_AddFarSpectrumFloat(void* handle,
+                               const float* far_spectrum,
+                               int spectrum_size) {
+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
+  uint32_t binary_spectrum = 0;
+
+  if (self == NULL) {
+    return -1;
+  }
+  if (far_spectrum == NULL) {
+    // Empty far end spectrum.
+    return -1;
+  }
+  if (spectrum_size != self->spectrum_size) {
+    // Data sizes don't match.
+    return -1;
+  }
+
+  // Get binary spectrum.
+  binary_spectrum = BinarySpectrumFloat(far_spectrum, self->mean_far_spectrum,
+                                        &(self->far_spectrum_initialized));
+  WebRtc_AddBinaryFarSpectrum(self->binary_farend, binary_spectrum);
+
+  return 0;
+}
+
+void WebRtc_FreeDelayEstimator(void* handle) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+
+  if (handle == NULL) {
+    return;
+  }
+
+  free(self->mean_near_spectrum);
+  self->mean_near_spectrum = NULL;
+
+  WebRtc_FreeBinaryDelayEstimator(self->binary_handle);
+  self->binary_handle = NULL;
+
+  free(self);
+}
+
+void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead) {
+  DelayEstimator* self = NULL;
+  DelayEstimatorFarend* farend = (DelayEstimatorFarend*)farend_handle;
+
+  if (farend_handle != NULL) {
+    self = static_cast<DelayEstimator*>(malloc(sizeof(DelayEstimator)));
+  }
+
+  if (self != NULL) {
+    int memory_fail = 0;
+
+    // Allocate memory for the farend spectrum handling.
+    self->binary_handle =
+        WebRtc_CreateBinaryDelayEstimator(farend->binary_farend, max_lookahead);
+    memory_fail |= (self->binary_handle == NULL);
+
+    // Allocate memory for spectrum buffers.
+    self->mean_near_spectrum = static_cast<SpectrumType*>(
+        malloc(farend->spectrum_size * sizeof(SpectrumType)));
+    memory_fail |= (self->mean_near_spectrum == NULL);
+
+    self->spectrum_size = farend->spectrum_size;
+
+    if (memory_fail) {
+      WebRtc_FreeDelayEstimator(self);
+      self = NULL;
+    }
+  }
+
+  return self;
+}
+
+int WebRtc_InitDelayEstimator(void* handle) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+
+  if (self == NULL) {
+    return -1;
+  }
+
+  // Initialize binary delay estimator.
+  WebRtc_InitBinaryDelayEstimator(self->binary_handle);
+
+  // Set averaged far and near end spectra to zero.
+  memset(self->mean_near_spectrum, 0,
+         sizeof(SpectrumType) * self->spectrum_size);
+  // Reset initialization indicators.
+  self->near_spectrum_initialized = 0;
+
+  return 0;
+}
+
+int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+  RTC_DCHECK(self);
+  return WebRtc_SoftResetBinaryDelayEstimator(self->binary_handle, delay_shift);
+}
+
+int WebRtc_set_history_size(void* handle, int history_size) {
+  DelayEstimator* self = static_cast<DelayEstimator*>(handle);
+
+  if ((self == NULL) || (history_size <= 1)) {
+    return -1;
+  }
+  return WebRtc_AllocateHistoryBufferMemory(self->binary_handle, history_size);
+}
+
+int WebRtc_history_size(const void* handle) {
+  const DelayEstimator* self = static_cast<const DelayEstimator*>(handle);
+
+  if (self == NULL) {
+    return -1;
+  }
+  if (self->binary_handle->farend->history_size !=
+      self->binary_handle->history_size) {
+    // Non matching history sizes.
+    return -1;
+  }
+  return self->binary_handle->history_size;
+}
+
+int WebRtc_set_lookahead(void* handle, int lookahead) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+  RTC_DCHECK(self);
+  RTC_DCHECK(self->binary_handle);
+  if ((lookahead > self->binary_handle->near_history_size - 1) ||
+      (lookahead < 0)) {
+    return -1;
+  }
+  self->binary_handle->lookahead = lookahead;
+  return self->binary_handle->lookahead;
+}
+
+int WebRtc_lookahead(void* handle) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+  RTC_DCHECK(self);
+  RTC_DCHECK(self->binary_handle);
+  return self->binary_handle->lookahead;
+}
+
+int WebRtc_set_allowed_offset(void* handle, int allowed_offset) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+
+  if ((self == NULL) || (allowed_offset < 0)) {
+    return -1;
+  }
+  self->binary_handle->allowed_offset = allowed_offset;
+  return 0;
+}
+
+int WebRtc_get_allowed_offset(const void* handle) {
+  const DelayEstimator* self = (const DelayEstimator*)handle;
+
+  if (self == NULL) {
+    return -1;
+  }
+  return self->binary_handle->allowed_offset;
+}
+
+int WebRtc_enable_robust_validation(void* handle, int enable) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+
+  if (self == NULL) {
+    return -1;
+  }
+  if ((enable < 0) || (enable > 1)) {
+    return -1;
+  }
+  RTC_DCHECK(self->binary_handle);
+  self->binary_handle->robust_validation_enabled = enable;
+  return 0;
+}
+
+int WebRtc_is_robust_validation_enabled(const void* handle) {
+  const DelayEstimator* self = (const DelayEstimator*)handle;
+
+  if (self == NULL) {
+    return -1;
+  }
+  return self->binary_handle->robust_validation_enabled;
+}
+
+int WebRtc_DelayEstimatorProcessFix(void* handle,
+                                    const uint16_t* near_spectrum,
+                                    int spectrum_size,
+                                    int near_q) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+  uint32_t binary_spectrum = 0;
+
+  if (self == NULL) {
+    return -1;
+  }
+  if (near_spectrum == NULL) {
+    // Empty near end spectrum.
+    return -1;
+  }
+  if (spectrum_size != self->spectrum_size) {
+    // Data sizes don't match.
+    return -1;
+  }
+  if (near_q > 15) {
+    // If `near_q` is larger than 15 we cannot guarantee no wrap around.
+    return -1;
+  }
+
+  // Get binary spectra.
+  binary_spectrum =
+      BinarySpectrumFix(near_spectrum, self->mean_near_spectrum, near_q,
+                        &(self->near_spectrum_initialized));
+
+  return WebRtc_ProcessBinarySpectrum(self->binary_handle, binary_spectrum);
+}
+
+int WebRtc_DelayEstimatorProcessFloat(void* handle,
+                                      const float* near_spectrum,
+                                      int spectrum_size) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+  uint32_t binary_spectrum = 0;
+
+  if (self == NULL) {
+    return -1;
+  }
+  if (near_spectrum == NULL) {
+    // Empty near end spectrum.
+    return -1;
+  }
+  if (spectrum_size != self->spectrum_size) {
+    // Data sizes don't match.
+    return -1;
+  }
+
+  // Get binary spectrum.
+  binary_spectrum = BinarySpectrumFloat(near_spectrum, self->mean_near_spectrum,
+                                        &(self->near_spectrum_initialized));
+
+  return WebRtc_ProcessBinarySpectrum(self->binary_handle, binary_spectrum);
+}
+
+int WebRtc_last_delay(void* handle) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+
+  if (self == NULL) {
+    return -1;
+  }
+
+  return WebRtc_binary_last_delay(self->binary_handle);
+}
+
+float WebRtc_last_delay_quality(void* handle) {
+  DelayEstimator* self = (DelayEstimator*)handle;
+  RTC_DCHECK(self);
+  return WebRtc_binary_last_delay_quality(self->binary_handle);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_wrapper.h b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_wrapper.h
new file mode 100644
index 0000000000..a90cbe31cb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_wrapper.h
@@ -0,0 +1,248 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Performs delay estimation on block by block basis.
+// The return value is  0 - OK and -1 - Error, unless otherwise stated.
+
+#ifndef MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
+#define MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+// Releases the memory allocated by WebRtc_CreateDelayEstimatorFarend(...)
+void WebRtc_FreeDelayEstimatorFarend(void* handle);
+
+// Allocates the memory needed by the far-end part of the delay estimation. The
+// memory needs to be initialized separately through
+// WebRtc_InitDelayEstimatorFarend(...).
+//
+// Inputs:
+//  - spectrum_size     : Size of the spectrum used both in far-end and
+//                        near-end. Used to allocate memory for spectrum
+//                        specific buffers.
+//  - history_size      : The far-end history buffer size. A change in buffer
+//                        size can be forced with WebRtc_set_history_size().
+//                        Note that the maximum delay which can be estimated is
+//                        determined together with WebRtc_set_lookahead().
+//
+// Return value:
+//  - void*             : Created `handle`. If the memory can't be allocated or
+//                        if any of the input parameters are invalid NULL is
+//                        returned.
+void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size);
+
+// Initializes the far-end part of the delay estimation instance returned by
+// WebRtc_CreateDelayEstimatorFarend(...)
+int WebRtc_InitDelayEstimatorFarend(void* handle);
+
+// Soft resets the far-end part of the delay estimation instance returned by
+// WebRtc_CreateDelayEstimatorFarend(...).
+// Input:
+//      - delay_shift   : The amount of blocks to shift history buffers.
+void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift);
+
+// Adds the far-end spectrum to the far-end history buffer. This spectrum is
+// used as reference when calculating the delay using
+// WebRtc_ProcessSpectrum().
+//
+// Inputs:
+//    - far_spectrum    : Far-end spectrum.
+//    - spectrum_size   : The size of the data arrays (same for both far- and
+//                        near-end).
+//    - far_q           : The Q-domain of the far-end data.
+//
+// Output:
+//    - handle          : Updated far-end instance.
+//
+int WebRtc_AddFarSpectrumFix(void* handle,
+                             const uint16_t* far_spectrum,
+                             int spectrum_size,
+                             int far_q);
+
+// See WebRtc_AddFarSpectrumFix() for description.
+int WebRtc_AddFarSpectrumFloat(void* handle,
+                               const float* far_spectrum,
+                               int spectrum_size);
+
+// Releases the memory allocated by WebRtc_CreateDelayEstimator(...)
+void WebRtc_FreeDelayEstimator(void* handle);
+
+// Allocates the memory needed by the delay estimation. The memory needs to be
+// initialized separately through WebRtc_InitDelayEstimator(...).
+//
+// Inputs:
+//      - farend_handle : Pointer to the far-end part of the delay estimation
+//                        instance created prior to this call using
+//                        WebRtc_CreateDelayEstimatorFarend().
+//
+//                        Note that WebRtc_CreateDelayEstimator does not take
+//                        ownership of `farend_handle`, which has to be torn
+//                        down properly after this instance.
+//
+//      - max_lookahead : Maximum amount of non-causal lookahead allowed. The
+//                        actual amount of lookahead used can be controlled by
+//                        WebRtc_set_lookahead(...). The default `lookahead` is
+//                        set to `max_lookahead` at create time. Use
+//                        WebRtc_set_lookahead(...) before start if a different
+//                        value is desired.
+//
+//                        Using lookahead can detect cases in which a near-end
+//                        signal occurs before the corresponding far-end signal.
+//                        It will delay the estimate for the current block by an
+//                        equal amount, and the returned values will be offset
+//                        by it.
+//
+//                        A value of zero is the typical no-lookahead case.
+//                        This also represents the minimum delay which can be
+//                        estimated.
+//
+//                        Note that the effective range of delay estimates is
+//                        [-`lookahead`,... ,`history_size`-`lookahead`)
+//                        where `history_size` is set through
+//                        WebRtc_set_history_size().
+//
+// Return value:
+//      - void*         : Created `handle`. If the memory can't be allocated or
+//                        if any of the input parameters are invalid NULL is
+//                        returned.
+void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead);
+
+// Initializes the delay estimation instance returned by
+// WebRtc_CreateDelayEstimator(...)
+int WebRtc_InitDelayEstimator(void* handle);
+
+// Soft resets the delay estimation instance returned by
+// WebRtc_CreateDelayEstimator(...)
+// Input:
+//      - delay_shift   : The amount of blocks to shift history buffers.
+//
+// Return value:
+//      - actual_shifts : The actual number of shifts performed.
+int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift);
+
+// Sets the effective `history_size` used. Valid values from 2. We simply need
+// at least two delays to compare to perform an estimate. If `history_size` is
+// changed, buffers are reallocated filling in with zeros if necessary.
+// Note that changing the `history_size` affects both buffers in far-end and
+// near-end. Hence it is important to change all DelayEstimators that use the
+// same reference far-end, to the same `history_size` value.
+// Inputs:
+//  - handle            : Pointer to the delay estimation instance.
+//  - history_size      : Effective history size to be used.
+// Return value:
+//  - new_history_size  : The new history size used. If the memory was not able
+//                        to be allocated 0 is returned.
+int WebRtc_set_history_size(void* handle, int history_size);
+
+// Returns the history_size currently used.
+// Input:
+//      - handle        : Pointer to the delay estimation instance.
+int WebRtc_history_size(const void* handle);
+
+// Sets the amount of `lookahead` to use. Valid values are [0, max_lookahead]
+// where `max_lookahead` was set at create time through
+// WebRtc_CreateDelayEstimator(...).
+//
+// Input:
+//      - handle        : Pointer to the delay estimation instance.
+//      - lookahead     : The amount of lookahead to be used.
+//
+// Return value:
+//      - new_lookahead : The actual amount of lookahead set, unless `handle` is
+//                        a NULL pointer or `lookahead` is invalid, for which an
+//                        error is returned.
+int WebRtc_set_lookahead(void* handle, int lookahead);
+
+// Returns the amount of lookahead we currently use.
+// Input:
+//      - handle        : Pointer to the delay estimation instance.
+int WebRtc_lookahead(void* handle);
+
+// Sets the `allowed_offset` used in the robust validation scheme.  If the
+// delay estimator is used in an echo control component, this parameter is
+// related to the filter length.  In principle `allowed_offset` should be set to
+// the echo control filter length minus the expected echo duration, i.e., the
+// delay offset the echo control can handle without quality regression.  The
+// default value, used if not set manually, is zero.  Note that `allowed_offset`
+// has to be non-negative.
+// Inputs:
+//  - handle            : Pointer to the delay estimation instance.
+//  - allowed_offset    : The amount of delay offset, measured in partitions,
+//                        the echo control filter can handle.
+int WebRtc_set_allowed_offset(void* handle, int allowed_offset);
+
+// Returns the `allowed_offset` in number of partitions.
+int WebRtc_get_allowed_offset(const void* handle);
+
+// Enables/Disables a robust validation functionality in the delay estimation.
+// This is by default set to disabled at create time.  The state is preserved
+// over a reset.
+// Inputs:
+//      - handle        : Pointer to the delay estimation instance.
+//      - enable        : Enable (1) or disable (0) this feature.
+int WebRtc_enable_robust_validation(void* handle, int enable);
+
+// Returns 1 if robust validation is enabled and 0 if disabled.
+int WebRtc_is_robust_validation_enabled(const void* handle);
+
+// Estimates and returns the delay between the far-end and near-end blocks. The
+// value will be offset by the lookahead (i.e. the lookahead should be
+// subtracted from the returned value).
+// Inputs:
+//      - handle        : Pointer to the delay estimation instance.
+//      - near_spectrum : Pointer to the near-end spectrum data of the current
+//                        block.
+//      - spectrum_size : The size of the data arrays (same for both far- and
+//                        near-end).
+//      - near_q        : The Q-domain of the near-end data.
+//
+// Output:
+//      - handle        : Updated instance.
+//
+// Return value:
+//      - delay         :  >= 0 - Calculated delay value.
+//                        -1    - Error.
+//                        -2    - Insufficient data for estimation.
+int WebRtc_DelayEstimatorProcessFix(void* handle,
+                                    const uint16_t* near_spectrum,
+                                    int spectrum_size,
+                                    int near_q);
+
+// See WebRtc_DelayEstimatorProcessFix() for description.
+int WebRtc_DelayEstimatorProcessFloat(void* handle,
+                                      const float* near_spectrum,
+                                      int spectrum_size);
+
+// Returns the last calculated delay updated by the function
+// WebRtc_DelayEstimatorProcess(...).
+//
+// Input:
+//      - handle        : Pointer to the delay estimation instance.
+//
+// Return value:
+//      - delay         : >= 0  - Last calculated delay value.
+//                        -1    - Error.
+//                        -2    - Insufficient data for estimation.
+int WebRtc_last_delay(void* handle);
+
+// Returns the estimation quality/probability of the last calculated delay
+// updated by the function WebRtc_DelayEstimatorProcess(...). The estimation
+// quality is a value in the interval [0, 1]. The higher the value, the better
+// the quality.
+//
+// Return value:
+//      - delay_quality : >= 0  - Estimation quality of last calculated delay.
+float WebRtc_last_delay_quality(void* handle);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/legacy_delay_estimator_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/utility/legacy_delay_estimator_gn/moz.build
new file mode 100644
index 0000000000..11294e2aef
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/legacy_delay_estimator_gn/moz.build
@@ -0,0 +1,222 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator.cc",
+    "/third_party/libwebrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("legacy_delay_estimator_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper.cc b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper.cc
new file mode 100644
index 0000000000..88642fb12b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/utility/pffft_wrapper.h"
+
+#include "rtc_base/checks.h"
+#include "third_party/pffft/src/pffft.h"
+
+namespace webrtc {
+namespace {
+
+size_t GetBufferSize(size_t fft_size, Pffft::FftType fft_type) {
+  return fft_size * (fft_type == Pffft::FftType::kReal ? 1 : 2);
+}
+
+float* AllocatePffftBuffer(size_t size) {
+  return static_cast<float*>(pffft_aligned_malloc(size * sizeof(float)));
+}
+
+}  // namespace
+
+Pffft::FloatBuffer::FloatBuffer(size_t fft_size, FftType fft_type)
+    : size_(GetBufferSize(fft_size, fft_type)),
+      data_(AllocatePffftBuffer(size_)) {}
+
+Pffft::FloatBuffer::~FloatBuffer() {
+  pffft_aligned_free(data_);
+}
+
+rtc::ArrayView<const float> Pffft::FloatBuffer::GetConstView() const {
+  return {data_, size_};
+}
+
+rtc::ArrayView<float> Pffft::FloatBuffer::GetView() {
+  return {data_, size_};
+}
+
+Pffft::Pffft(size_t fft_size, FftType fft_type)
+    : fft_size_(fft_size),
+      fft_type_(fft_type),
+      pffft_status_(pffft_new_setup(
+          fft_size_,
+          fft_type == Pffft::FftType::kReal ? PFFFT_REAL : PFFFT_COMPLEX)),
+      scratch_buffer_(
+          AllocatePffftBuffer(GetBufferSize(fft_size_, fft_type_))) {
+  RTC_DCHECK(pffft_status_);
+  RTC_DCHECK(scratch_buffer_);
+}
+
+Pffft::~Pffft() {
+  pffft_destroy_setup(pffft_status_);
+  pffft_aligned_free(scratch_buffer_);
+}
+
+bool Pffft::IsValidFftSize(size_t fft_size, FftType fft_type) {
+  if (fft_size == 0) {
+    return false;
+  }
+  // PFFFT only supports transforms for inputs of length N of the form
+  // N = (2^a)*(3^b)*(5^c) where b >=0 and c >= 0 and a >= 5 for the real FFT
+  // and a >= 4 for the complex FFT.
+  constexpr int kFactors[] = {2, 3, 5};
+  int factorization[] = {0, 0, 0};
+  int n = static_cast<int>(fft_size);
+  for (int i = 0; i < 3; ++i) {
+    while (n % kFactors[i] == 0) {
+      n = n / kFactors[i];
+      factorization[i]++;
+    }
+  }
+  int a_min = (fft_type == Pffft::FftType::kReal) ? 5 : 4;
+  return factorization[0] >= a_min && n == 1;
+}
+
+bool Pffft::IsSimdEnabled() {
+  return pffft_simd_size() > 1;
+}
+
+std::unique_ptr<Pffft::FloatBuffer> Pffft::CreateBuffer() const {
+  // Cannot use make_unique from absl because Pffft is the only friend of
+  // Pffft::FloatBuffer.
+  std::unique_ptr<Pffft::FloatBuffer> buffer(
+      new Pffft::FloatBuffer(fft_size_, fft_type_));
+  return buffer;
+}
+
+void Pffft::ForwardTransform(const FloatBuffer& in,
+                             FloatBuffer* out,
+                             bool ordered) {
+  RTC_DCHECK_EQ(in.size(), GetBufferSize(fft_size_, fft_type_));
+  RTC_DCHECK_EQ(in.size(), out->size());
+  RTC_DCHECK(scratch_buffer_);
+  if (ordered) {
+    pffft_transform_ordered(pffft_status_, in.const_data(), out->data(),
+                            scratch_buffer_, PFFFT_FORWARD);
+  } else {
+    pffft_transform(pffft_status_, in.const_data(), out->data(),
+                    scratch_buffer_, PFFFT_FORWARD);
+  }
+}
+
+void Pffft::BackwardTransform(const FloatBuffer& in,
+                              FloatBuffer* out,
+                              bool ordered) {
+  RTC_DCHECK_EQ(in.size(), GetBufferSize(fft_size_, fft_type_));
+  RTC_DCHECK_EQ(in.size(), out->size());
+  RTC_DCHECK(scratch_buffer_);
+  if (ordered) {
+    pffft_transform_ordered(pffft_status_, in.const_data(), out->data(),
+                            scratch_buffer_, PFFFT_BACKWARD);
+  } else {
+    pffft_transform(pffft_status_, in.const_data(), out->data(),
+                    scratch_buffer_, PFFFT_BACKWARD);
+  }
+}
+
+void Pffft::FrequencyDomainConvolve(const FloatBuffer& fft_x,
+                                    const FloatBuffer& fft_y,
+                                    FloatBuffer* out,
+                                    float scaling) {
+  RTC_DCHECK_EQ(fft_x.size(), GetBufferSize(fft_size_, fft_type_));
+  RTC_DCHECK_EQ(fft_x.size(), fft_y.size());
+  RTC_DCHECK_EQ(fft_x.size(), out->size());
+  pffft_zconvolve_accumulate(pffft_status_, fft_x.const_data(),
+                             fft_y.const_data(), out->data(), scaling);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper.h b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper.h
new file mode 100644
index 0000000000..983c2fd1bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper.h
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_UTILITY_PFFFT_WRAPPER_H_
+#define MODULES_AUDIO_PROCESSING_UTILITY_PFFFT_WRAPPER_H_
+
+#include <memory>
+
+#include "api/array_view.h"
+
+// Forward declaration.
+struct PFFFT_Setup;
+
+namespace webrtc {
+
+// Pretty-Fast Fast Fourier Transform (PFFFT) wrapper class.
+// Not thread safe.
+class Pffft {
+ public:
+  enum class FftType { kReal, kComplex };
+
+  // 1D floating point buffer used as input/output data type for the FFT ops.
+  // It must be constructed using Pffft::CreateBuffer().
+  class FloatBuffer {
+   public:
+    FloatBuffer(const FloatBuffer&) = delete;
+    FloatBuffer& operator=(const FloatBuffer&) = delete;
+    ~FloatBuffer();
+
+    rtc::ArrayView<const float> GetConstView() const;
+    rtc::ArrayView<float> GetView();
+
+   private:
+    friend class Pffft;
+    FloatBuffer(size_t fft_size, FftType fft_type);
+    const float* const_data() const { return data_; }
+    float* data() { return data_; }
+    size_t size() const { return size_; }
+
+    const size_t size_;
+    float* const data_;
+  };
+
+  // TODO(https://crbug.com/webrtc/9577): Consider adding a factory and making
+  // the ctor private.
+  // static std::unique_ptr<Pffft> Create(size_t fft_size,
+  // FftType fft_type); Ctor. `fft_size` must be a supported size (see
+  // Pffft::IsValidFftSize()). If not supported, the code will crash.
+  Pffft(size_t fft_size, FftType fft_type);
+  Pffft(const Pffft&) = delete;
+  Pffft& operator=(const Pffft&) = delete;
+  ~Pffft();
+
+  // Returns true if the FFT size is supported.
+  static bool IsValidFftSize(size_t fft_size, FftType fft_type);
+
+  // Returns true if SIMD code optimizations are being used.
+  static bool IsSimdEnabled();
+
+  // Creates a buffer of the right size.
+  std::unique_ptr<FloatBuffer> CreateBuffer() const;
+
+  // TODO(https://crbug.com/webrtc/9577): Overload with rtc::ArrayView args.
+  // Computes the forward fast Fourier transform.
+  void ForwardTransform(const FloatBuffer& in, FloatBuffer* out, bool ordered);
+  // Computes the backward fast Fourier transform.
+  void BackwardTransform(const FloatBuffer& in, FloatBuffer* out, bool ordered);
+
+  // Multiplies the frequency components of `fft_x` and `fft_y` and accumulates
+  // them into `out`. The arrays must have been obtained with
+  // ForwardTransform(..., /*ordered=*/false) - i.e., `fft_x` and `fft_y` must
+  // not be ordered.
+  void FrequencyDomainConvolve(const FloatBuffer& fft_x,
+                               const FloatBuffer& fft_y,
+                               FloatBuffer* out,
+                               float scaling = 1.f);
+
+ private:
+  const size_t fft_size_;
+  const FftType fft_type_;
+  PFFFT_Setup* pffft_status_;
+  float* const scratch_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_UTILITY_PFFFT_WRAPPER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_gn/moz.build
new file mode 100644
index 0000000000..02898359bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("pffft_wrapper_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_unittest.cc b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_unittest.cc
new file mode 100644
index 0000000000..2ad6849cd4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/utility/pffft_wrapper_unittest.cc
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/utility/pffft_wrapper.h"
+
+#include <algorithm>
+#include <cstdlib>
+#include <memory>
+
+#include "test/gtest.h"
+#include "third_party/pffft/src/pffft.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr size_t kMaxValidSizeCheck = 1024;
+
+static constexpr int kFftSizes[] = {
+    16,  32,      64,  96,  128,  160,  192,  256,  288,  384,   5 * 96, 512,
+    576, 5 * 128, 800, 864, 1024, 2048, 2592, 4000, 4096, 12000, 36864};
+
+void CreatePffftWrapper(size_t fft_size, Pffft::FftType fft_type) {
+  Pffft pffft_wrapper(fft_size, fft_type);
+}
+
+float* AllocateScratchBuffer(size_t fft_size, bool complex_fft) {
+  return static_cast<float*>(
+      pffft_aligned_malloc(fft_size * (complex_fft ? 2 : 1) * sizeof(float)));
+}
+
+double frand() {
+  return std::rand() / static_cast<double>(RAND_MAX);
+}
+
+void ExpectArrayViewsEquality(rtc::ArrayView<const float> a,
+                              rtc::ArrayView<const float> b) {
+  ASSERT_EQ(a.size(), b.size());
+  for (size_t i = 0; i < a.size(); ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_EQ(a[i], b[i]);
+  }
+}
+
+// Compares the output of the PFFFT C++ wrapper to that of the C PFFFT.
+// Bit-exactness is expected.
+void PffftValidateWrapper(size_t fft_size, bool complex_fft) {
+  // Always use the same seed to avoid flakiness.
+  std::srand(0);
+
+  // Init PFFFT.
+  PFFFT_Setup* pffft_status =
+      pffft_new_setup(fft_size, complex_fft ? PFFFT_COMPLEX : PFFFT_REAL);
+  ASSERT_TRUE(pffft_status) << "FFT size (" << fft_size << ") not supported.";
+  size_t num_floats = fft_size * (complex_fft ? 2 : 1);
+  int num_bytes = static_cast<int>(num_floats) * sizeof(float);
+  float* in = static_cast<float*>(pffft_aligned_malloc(num_bytes));
+  float* out = static_cast<float*>(pffft_aligned_malloc(num_bytes));
+  float* scratch = AllocateScratchBuffer(fft_size, complex_fft);
+
+  // Init PFFFT C++ wrapper.
+  Pffft::FftType fft_type =
+      complex_fft ? Pffft::FftType::kComplex : Pffft::FftType::kReal;
+  ASSERT_TRUE(Pffft::IsValidFftSize(fft_size, fft_type));
+  Pffft pffft_wrapper(fft_size, fft_type);
+  auto in_wrapper = pffft_wrapper.CreateBuffer();
+  auto out_wrapper = pffft_wrapper.CreateBuffer();
+
+  // Input and output buffers views.
+  rtc::ArrayView<float> in_view(in, num_floats);
+  rtc::ArrayView<float> out_view(out, num_floats);
+  auto in_wrapper_view = in_wrapper->GetView();
+  EXPECT_EQ(in_wrapper_view.size(), num_floats);
+  auto out_wrapper_view = out_wrapper->GetConstView();
+  EXPECT_EQ(out_wrapper_view.size(), num_floats);
+
+  // Random input data.
+  for (size_t i = 0; i < num_floats; ++i) {
+    in_wrapper_view[i] = in[i] = static_cast<float>(frand() * 2.0 - 1.0);
+  }
+
+  // Forward transform.
+  pffft_transform(pffft_status, in, out, scratch, PFFFT_FORWARD);
+  pffft_wrapper.ForwardTransform(*in_wrapper, out_wrapper.get(),
+                                 /*ordered=*/false);
+  ExpectArrayViewsEquality(out_view, out_wrapper_view);
+
+  // Copy the FFT results into the input buffers to compute the backward FFT.
+  std::copy(out_view.begin(), out_view.end(), in_view.begin());
+  std::copy(out_wrapper_view.begin(), out_wrapper_view.end(),
+            in_wrapper_view.begin());
+
+  // Backward transform.
+  pffft_transform(pffft_status, in, out, scratch, PFFFT_BACKWARD);
+  pffft_wrapper.BackwardTransform(*in_wrapper, out_wrapper.get(),
+                                  /*ordered=*/false);
+  ExpectArrayViewsEquality(out_view, out_wrapper_view);
+
+  pffft_destroy_setup(pffft_status);
+  pffft_aligned_free(in);
+  pffft_aligned_free(out);
+  pffft_aligned_free(scratch);
+}
+
+}  // namespace
+
+TEST(PffftTest, CreateWrapperWithValidSize) {
+  for (size_t fft_size = 0; fft_size < kMaxValidSizeCheck; ++fft_size) {
+    SCOPED_TRACE(fft_size);
+    if (Pffft::IsValidFftSize(fft_size, Pffft::FftType::kReal)) {
+      CreatePffftWrapper(fft_size, Pffft::FftType::kReal);
+    }
+    if (Pffft::IsValidFftSize(fft_size, Pffft::FftType::kComplex)) {
+      CreatePffftWrapper(fft_size, Pffft::FftType::kComplex);
+    }
+  }
+}
+
+#if !defined(NDEBUG) && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+class PffftInvalidSizeDeathTest : public ::testing::Test,
+                                  public ::testing::WithParamInterface<size_t> {
+};
+
+TEST_P(PffftInvalidSizeDeathTest, DoNotCreateRealWrapper) {
+  size_t fft_size = GetParam();
+  ASSERT_FALSE(Pffft::IsValidFftSize(fft_size, Pffft::FftType::kReal));
+  EXPECT_DEATH(CreatePffftWrapper(fft_size, Pffft::FftType::kReal), "");
+}
+
+TEST_P(PffftInvalidSizeDeathTest, DoNotCreateComplexWrapper) {
+  size_t fft_size = GetParam();
+  ASSERT_FALSE(Pffft::IsValidFftSize(fft_size, Pffft::FftType::kComplex));
+  EXPECT_DEATH(CreatePffftWrapper(fft_size, Pffft::FftType::kComplex), "");
+}
+
+INSTANTIATE_TEST_SUITE_P(PffftTest,
+                         PffftInvalidSizeDeathTest,
+                         ::testing::Values(17,
+                                           33,
+                                           65,
+                                           97,
+                                           129,
+                                           161,
+                                           193,
+                                           257,
+                                           289,
+                                           385,
+                                           481,
+                                           513,
+                                           577,
+                                           641,
+                                           801,
+                                           865,
+                                           1025));
+
+#endif
+
+// TODO(https://crbug.com/webrtc/9577): Enable once SIMD is always enabled.
+TEST(PffftTest, DISABLED_CheckSimd) {
+  EXPECT_TRUE(Pffft::IsSimdEnabled());
+}
+
+TEST(PffftTest, FftBitExactness) {
+  for (int fft_size : kFftSizes) {
+    SCOPED_TRACE(fft_size);
+    if (fft_size != 16) {
+      PffftValidateWrapper(fft_size, /*complex_fft=*/false);
+    }
+    PffftValidateWrapper(fft_size, /*complex_fft=*/true);
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/vad/BUILD.gn
new file mode 100644
index 0000000000..71e079d3a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/BUILD.gn
@@ -0,0 +1,69 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+rtc_library("vad") {
+  visibility = [
+    "../*",
+    "../../../rtc_tools:*",
+  ]
+  sources = [
+    "common.h",
+    "gmm.cc",
+    "gmm.h",
+    "noise_gmm_tables.h",
+    "pitch_based_vad.cc",
+    "pitch_based_vad.h",
+    "pitch_internal.cc",
+    "pitch_internal.h",
+    "pole_zero_filter.cc",
+    "pole_zero_filter.h",
+    "standalone_vad.cc",
+    "standalone_vad.h",
+    "vad_audio_proc.cc",
+    "vad_audio_proc.h",
+    "vad_audio_proc_internal.h",
+    "vad_circular_buffer.cc",
+    "vad_circular_buffer.h",
+    "voice_activity_detector.cc",
+    "voice_activity_detector.h",
+    "voice_gmm_tables.h",
+  ]
+  deps = [
+    "../../../audio/utility:audio_frame_operations",
+    "../../../common_audio",
+    "../../../common_audio:common_audio_c",
+    "../../../common_audio/third_party/ooura:fft_size_256",
+    "../../../rtc_base:checks",
+    "../../audio_coding:isac_vad",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("vad_unittests") {
+    testonly = true
+    sources = [
+      "gmm_unittest.cc",
+      "pitch_based_vad_unittest.cc",
+      "pitch_internal_unittest.cc",
+      "pole_zero_filter_unittest.cc",
+      "standalone_vad_unittest.cc",
+      "vad_audio_proc_unittest.cc",
+      "vad_circular_buffer_unittest.cc",
+      "voice_activity_detector_unittest.cc",
+    ]
+    deps = [
+      ":vad",
+      "../../../common_audio",
+      "../../../test:fileutils",
+      "../../../test:test_support",
+      "//testing/gmock",
+      "//testing/gtest",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/common.h b/third_party/libwebrtc/modules/audio_processing/vad/common.h
new file mode 100644
index 0000000000..b5a5fb385b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/common.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
+#define MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
+
+#include <stddef.h>
+
+static const int kSampleRateHz = 16000;
+static const size_t kLength10Ms = kSampleRateHz / 100;
+static const size_t kMaxNumFrames = 4;
+
+struct AudioFeatures {
+  double log_pitch_gain[kMaxNumFrames];
+  double pitch_lag_hz[kMaxNumFrames];
+  double spectral_peak[kMaxNumFrames];
+  double rms[kMaxNumFrames];
+  size_t num_frames;
+  bool silence;
+};
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/gmm.cc b/third_party/libwebrtc/modules/audio_processing/vad/gmm.cc
new file mode 100644
index 0000000000..3b8764c4d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/gmm.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/gmm.h"
+
+#include <math.h>
+
+namespace webrtc {
+
+static const int kMaxDimension = 10;
+
+static void RemoveMean(const double* in,
+                       const double* mean_vec,
+                       int dimension,
+                       double* out) {
+  for (int n = 0; n < dimension; ++n)
+    out[n] = in[n] - mean_vec[n];
+}
+
+static double ComputeExponent(const double* in,
+                              const double* covar_inv,
+                              int dimension) {
+  double q = 0;
+  for (int i = 0; i < dimension; ++i) {
+    double v = 0;
+    for (int j = 0; j < dimension; j++)
+      v += (*covar_inv++) * in[j];
+    q += v * in[i];
+  }
+  q *= -0.5;
+  return q;
+}
+
+double EvaluateGmm(const double* x, const GmmParameters& gmm_parameters) {
+  if (gmm_parameters.dimension > kMaxDimension) {
+    return -1;  // This is invalid pdf so the caller can check this.
+  }
+  double f = 0;
+  double v[kMaxDimension];
+  const double* mean_vec = gmm_parameters.mean;
+  const double* covar_inv = gmm_parameters.covar_inverse;
+
+  for (int n = 0; n < gmm_parameters.num_mixtures; n++) {
+    RemoveMean(x, mean_vec, gmm_parameters.dimension, v);
+    double q = ComputeExponent(v, covar_inv, gmm_parameters.dimension) +
+               gmm_parameters.weight[n];
+    f += exp(q);
+    mean_vec += gmm_parameters.dimension;
+    covar_inv += gmm_parameters.dimension * gmm_parameters.dimension;
+  }
+  return f;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/gmm.h b/third_party/libwebrtc/modules/audio_processing/vad/gmm.h
new file mode 100644
index 0000000000..d9d68ecfdc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/gmm.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_GMM_H_
+#define MODULES_AUDIO_PROCESSING_VAD_GMM_H_
+
+namespace webrtc {
+
+// A structure that specifies a GMM.
+// A GMM is formulated as
+//  f(x) = w[0] * mixture[0] + w[1] * mixture[1] + ... +
+//         w[num_mixtures - 1] * mixture[num_mixtures - 1];
+// Where a 'mixture' is a Gaussian density.
+
+struct GmmParameters {
+  // weight[n] = log(w[n]) - `dimension`/2 * log(2*pi) - 1/2 * log(det(cov[n]));
+  // where cov[n] is the covariance matrix of mixture n;
+  const double* weight;
+  // pointer to the first element of a `num_mixtures`x`dimension` matrix
+  // where kth row is the mean of the kth mixture.
+  const double* mean;
+  // pointer to the first element of a `num_mixtures`x`dimension`x`dimension`
+  // 3D-matrix, where the kth 2D-matrix is the inverse of the covariance
+  // matrix of the kth mixture.
+  const double* covar_inverse;
+  // Dimensionality of the mixtures.
+  int dimension;
+  // number of the mixtures.
+  int num_mixtures;
+};
+
+// Evaluate the given GMM, according to `gmm_parameters`, at the given point
+// `x`. If the dimensionality of the given GMM is larger that the maximum
+// acceptable dimension by the following function -1 is returned.
+double EvaluateGmm(const double* x, const GmmParameters& gmm_parameters);
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_PROCESSING_VAD_GMM_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/gmm_unittest.cc b/third_party/libwebrtc/modules/audio_processing/vad/gmm_unittest.cc
new file mode 100644
index 0000000000..d895afab7b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/gmm_unittest.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/gmm.h"
+
+#include <math.h>
+
+#include "modules/audio_processing/vad/noise_gmm_tables.h"
+#include "modules/audio_processing/vad/voice_gmm_tables.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(GmmTest, EvaluateGmm) {
+  GmmParameters noise_gmm;
+  GmmParameters voice_gmm;
+
+  // Setup noise GMM.
+  noise_gmm.dimension = kNoiseGmmDim;
+  noise_gmm.num_mixtures = kNoiseGmmNumMixtures;
+  noise_gmm.weight = kNoiseGmmWeights;
+  noise_gmm.mean = &kNoiseGmmMean[0][0];
+  noise_gmm.covar_inverse = &kNoiseGmmCovarInverse[0][0][0];
+
+  // Setup voice GMM.
+  voice_gmm.dimension = kVoiceGmmDim;
+  voice_gmm.num_mixtures = kVoiceGmmNumMixtures;
+  voice_gmm.weight = kVoiceGmmWeights;
+  voice_gmm.mean = &kVoiceGmmMean[0][0];
+  voice_gmm.covar_inverse = &kVoiceGmmCovarInverse[0][0][0];
+
+  // Test vectors. These are the mean of the GMM means.
+  const double kXVoice[kVoiceGmmDim] = {-1.35893162459863, 602.862491970368,
+                                        178.022069191324};
+  const double kXNoise[kNoiseGmmDim] = {-2.33443722724409, 2827.97828765184,
+                                        141.114178166812};
+
+  // Expected pdf values. These values are computed in MATLAB using EvalGmm.m
+  const double kPdfNoise = 1.88904409403101e-07;
+  const double kPdfVoice = 1.30453996982266e-06;
+
+  // Relative error should be smaller that the following value.
+  const double kAcceptedRelativeErr = 1e-10;
+
+  // Test Voice.
+  double pdf = EvaluateGmm(kXVoice, voice_gmm);
+  EXPECT_GT(pdf, 0);
+  double relative_error = fabs(pdf - kPdfVoice) / kPdfVoice;
+  EXPECT_LE(relative_error, kAcceptedRelativeErr);
+
+  // Test Noise.
+  pdf = EvaluateGmm(kXNoise, noise_gmm);
+  EXPECT_GT(pdf, 0);
+  relative_error = fabs(pdf - kPdfNoise) / kPdfNoise;
+  EXPECT_LE(relative_error, kAcceptedRelativeErr);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/noise_gmm_tables.h b/third_party/libwebrtc/modules/audio_processing/vad/noise_gmm_tables.h
new file mode 100644
index 0000000000..944a5401cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/noise_gmm_tables.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// GMM tables for inactive segments. Generated by MakeGmmTables.m.
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
+#define MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
+
+namespace webrtc {
+
+static const int kNoiseGmmNumMixtures = 12;
+static const int kNoiseGmmDim = 3;
+
+static const double
+    kNoiseGmmCovarInverse[kNoiseGmmNumMixtures][kNoiseGmmDim][kNoiseGmmDim] = {
+        {{7.36219567592941e+00, 4.83060785179861e-03, 1.23335151497610e-02},
+         {4.83060785179861e-03, 1.65289507047817e-04, -2.41490588169997e-04},
+         {1.23335151497610e-02, -2.41490588169997e-04, 6.59472060689382e-03}},
+        {{8.70265239309140e+00, -5.30636201431086e-04, 5.44014966585347e-03},
+         {-5.30636201431086e-04, 3.11095453521008e-04, -1.86287206836035e-04},
+         {5.44014966585347e-03, -1.86287206836035e-04, 6.29493388790744e-04}},
+        {{4.53467851955055e+00, -3.92977536695197e-03, -2.46521420693317e-03},
+         {-3.92977536695197e-03, 4.94650752632750e-05, -1.08587438501826e-05},
+         {-2.46521420693317e-03, -1.08587438501826e-05, 9.28793975422261e-05}},
+        {{9.26817997114275e-01, -4.03976069276753e-04, -3.56441427392165e-03},
+         {-4.03976069276753e-04, 2.51976251631430e-06, 1.46914206734572e-07},
+         {-3.56441427392165e-03, 1.46914206734572e-07, 8.19914567685373e-05}},
+        {{7.61715986787441e+00, -1.54889041216888e-04, 2.41756280071656e-02},
+         {-1.54889041216888e-04, 3.50282550461672e-07, -6.27251196972490e-06},
+         {2.41756280071656e-02, -6.27251196972490e-06, 1.45061847649872e-02}},
+        {{8.31193642663158e+00, -3.84070508164323e-04, -3.09750630821876e-02},
+         {-3.84070508164323e-04, 3.80433432277336e-07, -1.14321142836636e-06},
+         {-3.09750630821876e-02, -1.14321142836636e-06, 8.35091486289997e-04}},
+        {{9.67283151270894e-01, 5.82465812445039e-05, -3.18350798617053e-03},
+         {5.82465812445039e-05, 2.23762672000318e-07, -7.74196587408623e-07},
+         {-3.18350798617053e-03, -7.74196587408623e-07, 3.85120938338325e-04}},
+        {{8.28066236985388e+00, 5.87634508319763e-05, 6.99303090891743e-03},
+         {5.87634508319763e-05, 2.93746018618058e-07, 3.40843332882272e-07},
+         {6.99303090891743e-03, 3.40843332882272e-07, 1.99379171190344e-04}},
+        {{6.07488998675646e+00, -1.11494526618473e-02, 5.10013111123381e-03},
+         {-1.11494526618473e-02, 6.99238879921751e-04, 5.36718550370870e-05},
+         {5.10013111123381e-03, 5.36718550370870e-05, 5.26909853276753e-04}},
+        {{6.90492021419175e+00, 4.20639355257863e-04, -2.38612752336481e-03},
+         {4.20639355257863e-04, 3.31246767338153e-06, -2.42052288150859e-08},
+         {-2.38612752336481e-03, -2.42052288150859e-08, 4.46608368363412e-04}},
+        {{1.31069150869715e+01, -1.73718583865670e-04, -1.97591814508578e-02},
+         {-1.73718583865670e-04, 2.80451716300124e-07, 9.96570755379865e-07},
+         {-1.97591814508578e-02, 9.96570755379865e-07, 2.41361900868847e-03}},
+        {{4.69566344239814e+00, -2.61077567563690e-04, 5.26359000761433e-03},
+         {-2.61077567563690e-04, 1.82420859823767e-06, -7.83645887541601e-07},
+         {5.26359000761433e-03, -7.83645887541601e-07, 1.33586288288802e-02}}};
+
+static const double kNoiseGmmMean[kNoiseGmmNumMixtures][kNoiseGmmDim] = {
+    {-2.01386094766163e+00, 1.69702162045397e+02, 7.41715804872181e+01},
+    {-1.94684591777290e+00, 1.42398396732668e+02, 1.64186321157831e+02},
+    {-2.29319297562437e+00, 3.86415425589868e+02, 2.13452215267125e+02},
+    {-3.25487177070268e+00, 1.08668712553616e+03, 2.33119949467419e+02},
+    {-2.13159632447467e+00, 4.83821702557717e+03, 6.86786166673740e+01},
+    {-2.26171410780526e+00, 4.79420193982422e+03, 1.53222513286450e+02},
+    {-3.32166740703185e+00, 4.35161135834358e+03, 1.33206448431316e+02},
+    {-2.19290322814343e+00, 3.98325506609408e+03, 2.13249167359934e+02},
+    {-2.02898459255404e+00, 7.37039893155007e+03, 1.12518527491926e+02},
+    {-2.26150236399500e+00, 1.54896745196145e+03, 1.49717357868579e+02},
+    {-2.00417668301790e+00, 3.82434760310304e+03, 1.07438913004312e+02},
+    {-2.30193040814533e+00, 1.43953696546439e+03, 7.04085275122649e+01}};
+
+static const double kNoiseGmmWeights[kNoiseGmmNumMixtures] = {
+    -1.09422832086193e+01, -1.10847897513425e+01, -1.36767587732187e+01,
+    -1.79789356118641e+01, -1.42830169160894e+01, -1.56500228061379e+01,
+    -1.83124990950113e+01, -1.69979436177477e+01, -1.12329424387828e+01,
+    -1.41311785780639e+01, -1.47171861448585e+01, -1.35963362781839e+01};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad.cc b/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad.cc
new file mode 100644
index 0000000000..68e60dc66a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad.cc
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/pitch_based_vad.h"
+
+#include <string.h>
+
+#include "modules/audio_processing/vad/common.h"
+#include "modules/audio_processing/vad/noise_gmm_tables.h"
+#include "modules/audio_processing/vad/vad_circular_buffer.h"
+#include "modules/audio_processing/vad/voice_gmm_tables.h"
+
+namespace webrtc {
+
+static_assert(kNoiseGmmDim == kVoiceGmmDim,
+              "noise and voice gmm dimension not equal");
+
+// These values should match MATLAB counterparts for unit-tests to pass.
+static const int kPosteriorHistorySize = 500;  // 5 sec of 10 ms frames.
+static const double kInitialPriorProbability = 0.3;
+static const int kTransientWidthThreshold = 7;
+static const double kLowProbabilityThreshold = 0.2;
+
+static double LimitProbability(double p) {
+  const double kLimHigh = 0.99;
+  const double kLimLow = 0.01;
+
+  if (p > kLimHigh)
+    p = kLimHigh;
+  else if (p < kLimLow)
+    p = kLimLow;
+  return p;
+}
+
+PitchBasedVad::PitchBasedVad()
+    : p_prior_(kInitialPriorProbability),
+      circular_buffer_(VadCircularBuffer::Create(kPosteriorHistorySize)) {
+  // Setup noise GMM.
+  noise_gmm_.dimension = kNoiseGmmDim;
+  noise_gmm_.num_mixtures = kNoiseGmmNumMixtures;
+  noise_gmm_.weight = kNoiseGmmWeights;
+  noise_gmm_.mean = &kNoiseGmmMean[0][0];
+  noise_gmm_.covar_inverse = &kNoiseGmmCovarInverse[0][0][0];
+
+  // Setup voice GMM.
+  voice_gmm_.dimension = kVoiceGmmDim;
+  voice_gmm_.num_mixtures = kVoiceGmmNumMixtures;
+  voice_gmm_.weight = kVoiceGmmWeights;
+  voice_gmm_.mean = &kVoiceGmmMean[0][0];
+  voice_gmm_.covar_inverse = &kVoiceGmmCovarInverse[0][0][0];
+}
+
+PitchBasedVad::~PitchBasedVad() {}
+
+int PitchBasedVad::VoicingProbability(const AudioFeatures& features,
+                                      double* p_combined) {
+  double p;
+  double gmm_features[3];
+  double pdf_features_given_voice;
+  double pdf_features_given_noise;
+  // These limits are the same in matlab implementation 'VoicingProbGMM().'
+  const double kLimLowLogPitchGain = -2.0;
+  const double kLimHighLogPitchGain = -0.9;
+  const double kLimLowSpectralPeak = 200;
+  const double kLimHighSpectralPeak = 2000;
+  const double kEps = 1e-12;
+  for (size_t n = 0; n < features.num_frames; n++) {
+    gmm_features[0] = features.log_pitch_gain[n];
+    gmm_features[1] = features.spectral_peak[n];
+    gmm_features[2] = features.pitch_lag_hz[n];
+
+    pdf_features_given_voice = EvaluateGmm(gmm_features, voice_gmm_);
+    pdf_features_given_noise = EvaluateGmm(gmm_features, noise_gmm_);
+
+    if (features.spectral_peak[n] < kLimLowSpectralPeak ||
+        features.spectral_peak[n] > kLimHighSpectralPeak ||
+        features.log_pitch_gain[n] < kLimLowLogPitchGain) {
+      pdf_features_given_voice = kEps * pdf_features_given_noise;
+    } else if (features.log_pitch_gain[n] > kLimHighLogPitchGain) {
+      pdf_features_given_noise = kEps * pdf_features_given_voice;
+    }
+
+    p = p_prior_ * pdf_features_given_voice /
+        (pdf_features_given_voice * p_prior_ +
+         pdf_features_given_noise * (1 - p_prior_));
+
+    p = LimitProbability(p);
+
+    // Combine pitch-based probability with standalone probability, before
+    // updating prior probabilities.
+    double prod_active = p * p_combined[n];
+    double prod_inactive = (1 - p) * (1 - p_combined[n]);
+    p_combined[n] = prod_active / (prod_active + prod_inactive);
+
+    if (UpdatePrior(p_combined[n]) < 0)
+      return -1;
+    // Limit prior probability. With a zero prior probability the posterior
+    // probability is always zero.
+    p_prior_ = LimitProbability(p_prior_);
+  }
+  return 0;
+}
+
+int PitchBasedVad::UpdatePrior(double p) {
+  circular_buffer_->Insert(p);
+  if (circular_buffer_->RemoveTransient(kTransientWidthThreshold,
+                                        kLowProbabilityThreshold) < 0)
+    return -1;
+  p_prior_ = circular_buffer_->Mean();
+  return 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad.h b/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad.h
new file mode 100644
index 0000000000..fa3abc2d28
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
+#define MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
+
+#include <memory>
+
+#include "modules/audio_processing/vad/common.h"
+#include "modules/audio_processing/vad/gmm.h"
+
+namespace webrtc {
+
+class VadCircularBuffer;
+
+// Computes the probability of the input audio frame to be active given
+// the corresponding pitch-gain and lag of the frame.
+class PitchBasedVad {
+ public:
+  PitchBasedVad();
+  ~PitchBasedVad();
+
+  // Compute pitch-based voicing probability, given the features.
+  //   features: a structure containing features required for computing voicing
+  //             probabilities.
+  //
+  //   p_combined: an array which contains the combined activity probabilities
+  //               computed prior to the call of this function. The method,
+  //               then, computes the voicing probabilities and combine them
+  //               with the given values. The result are returned in `p`.
+  int VoicingProbability(const AudioFeatures& features, double* p_combined);
+
+ private:
+  int UpdatePrior(double p);
+
+  // TODO(turajs): maybe defining this at a higher level (maybe enum) so that
+  // all the code recognize it as "no-error."
+  static const int kNoError = 0;
+
+  GmmParameters noise_gmm_;
+  GmmParameters voice_gmm_;
+
+  double p_prior_;
+
+  std::unique_ptr<VadCircularBuffer> circular_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_PITCH_BASED_VAD_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad_unittest.cc b/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad_unittest.cc
new file mode 100644
index 0000000000..4a8331a769
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad_unittest.cc
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/pitch_based_vad.h"
+
+#include <math.h>
+#include <stdio.h>
+
+#include <string>
+
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TEST(PitchBasedVadTest, VoicingProbabilityTest) {
+  std::string spectral_peak_file_name =
+      test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat");
+  FILE* spectral_peak_file = fopen(spectral_peak_file_name.c_str(), "rb");
+  ASSERT_TRUE(spectral_peak_file != NULL);
+
+  std::string pitch_gain_file_name =
+      test::ResourcePath("audio_processing/agc/agc_pitch_gain", "dat");
+  FILE* pitch_gain_file = fopen(pitch_gain_file_name.c_str(), "rb");
+  ASSERT_TRUE(pitch_gain_file != NULL);
+
+  std::string pitch_lag_file_name =
+      test::ResourcePath("audio_processing/agc/agc_pitch_lag", "dat");
+  FILE* pitch_lag_file = fopen(pitch_lag_file_name.c_str(), "rb");
+  ASSERT_TRUE(pitch_lag_file != NULL);
+
+  std::string voicing_prob_file_name =
+      test::ResourcePath("audio_processing/agc/agc_voicing_prob", "dat");
+  FILE* voicing_prob_file = fopen(voicing_prob_file_name.c_str(), "rb");
+  ASSERT_TRUE(voicing_prob_file != NULL);
+
+  PitchBasedVad vad_;
+
+  double reference_activity_probability;
+
+  AudioFeatures audio_features;
+  memset(&audio_features, 0, sizeof(audio_features));
+  audio_features.num_frames = 1;
+  while (fread(audio_features.spectral_peak,
+               sizeof(audio_features.spectral_peak[0]), 1,
+               spectral_peak_file) == 1u) {
+    double p;
+    ASSERT_EQ(1u, fread(audio_features.log_pitch_gain,
+                        sizeof(audio_features.log_pitch_gain[0]), 1,
+                        pitch_gain_file));
+    ASSERT_EQ(1u,
+              fread(audio_features.pitch_lag_hz,
+                    sizeof(audio_features.pitch_lag_hz[0]), 1, pitch_lag_file));
+    ASSERT_EQ(1u, fread(&reference_activity_probability,
+                        sizeof(reference_activity_probability), 1,
+                        voicing_prob_file));
+
+    p = 0.5;  // Initialize to the neutral value for combining probabilities.
+    EXPECT_EQ(0, vad_.VoicingProbability(audio_features, &p));
+    EXPECT_NEAR(p, reference_activity_probability, 0.01);
+  }
+
+  fclose(spectral_peak_file);
+  fclose(pitch_gain_file);
+  fclose(pitch_lag_file);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal.cc b/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal.cc
new file mode 100644
index 0000000000..8f86918644
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal.cc
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/pitch_internal.h"
+
+#include <cmath>
+
+namespace webrtc {
+
+// A 4-to-3 linear interpolation.
+// The interpolation constants are derived as following:
+// Input pitch parameters are updated every 7.5 ms. Within a 30-ms interval
+// we are interested in pitch parameters of 0-5 ms, 10-15ms and 20-25ms. This is
+// like interpolating 4-to-6 and keep the odd samples.
+// The reason behind this is that LPC coefficients are computed for the first
+// half of each 10ms interval.
+static void PitchInterpolation(double old_val, const double* in, double* out) {
+  out[0] = 1. / 6. * old_val + 5. / 6. * in[0];
+  out[1] = 5. / 6. * in[1] + 1. / 6. * in[2];
+  out[2] = 0.5 * in[2] + 0.5 * in[3];
+}
+
+void GetSubframesPitchParameters(int sampling_rate_hz,
+                                 double* gains,
+                                 double* lags,
+                                 int num_in_frames,
+                                 int num_out_frames,
+                                 double* log_old_gain,
+                                 double* old_lag,
+                                 double* log_pitch_gain,
+                                 double* pitch_lag_hz) {
+  // Gain interpolation is in log-domain, also returned in log-domain.
+  for (int n = 0; n < num_in_frames; n++)
+    gains[n] = log(gains[n] + 1e-12);
+
+  // Interpolate lags and gains.
+  PitchInterpolation(*log_old_gain, gains, log_pitch_gain);
+  *log_old_gain = gains[num_in_frames - 1];
+  PitchInterpolation(*old_lag, lags, pitch_lag_hz);
+  *old_lag = lags[num_in_frames - 1];
+
+  // Convert pitch-lags to Hertz.
+  for (int n = 0; n < num_out_frames; n++) {
+    pitch_lag_hz[n] = (sampling_rate_hz) / (pitch_lag_hz[n]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal.h b/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal.h
new file mode 100644
index 0000000000..e382c1fbde
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
+
+namespace webrtc {
+
+// TODO(turajs): Write a description of this function. Also be consistent with
+// usage of `sampling_rate_hz` vs `kSamplingFreqHz`.
+void GetSubframesPitchParameters(int sampling_rate_hz,
+                                 double* gains,
+                                 double* lags,
+                                 int num_in_frames,
+                                 int num_out_frames,
+                                 double* log_old_gain,
+                                 double* old_lag,
+                                 double* log_pitch_gain,
+                                 double* pitch_lag_hz);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_PITCH_INTERNAL_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal_unittest.cc b/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal_unittest.cc
new file mode 100644
index 0000000000..c851421ba7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal_unittest.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/pitch_internal.h"
+
+#include <math.h>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(PitchInternalTest, test) {
+  const int kSamplingRateHz = 8000;
+  const int kNumInputParameters = 4;
+  const int kNumOutputParameters = 3;
+  // Inputs
+  double log_old_gain = log(0.5);
+  double gains[] = {0.6, 0.2, 0.5, 0.4};
+
+  double old_lag = 70;
+  double lags[] = {90, 111, 122, 50};
+
+  // Expected outputs
+  double expected_log_pitch_gain[] = {-0.541212549898316, -1.45672279045507,
+                                      -0.80471895621705};
+  double expected_log_old_gain = log(gains[kNumInputParameters - 1]);
+
+  double expected_pitch_lag_hz[] = {92.3076923076923, 70.9010339734121,
+                                    93.0232558139535};
+  double expected_old_lag = lags[kNumInputParameters - 1];
+
+  double log_pitch_gain[kNumOutputParameters];
+  double pitch_lag_hz[kNumInputParameters];
+
+  GetSubframesPitchParameters(kSamplingRateHz, gains, lags, kNumInputParameters,
+                              kNumOutputParameters, &log_old_gain, &old_lag,
+                              log_pitch_gain, pitch_lag_hz);
+
+  for (int n = 0; n < 3; n++) {
+    EXPECT_NEAR(pitch_lag_hz[n], expected_pitch_lag_hz[n], 1e-6);
+    EXPECT_NEAR(log_pitch_gain[n], expected_log_pitch_gain[n], 1e-8);
+  }
+  EXPECT_NEAR(old_lag, expected_old_lag, 1e-6);
+  EXPECT_NEAR(log_old_gain, expected_log_old_gain, 1e-8);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter.cc b/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter.cc
new file mode 100644
index 0000000000..e7a611309c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/pole_zero_filter.h"
+
+#include <string.h>
+
+#include <algorithm>
+
+namespace webrtc {
+
+PoleZeroFilter* PoleZeroFilter::Create(const float* numerator_coefficients,
+                                       size_t order_numerator,
+                                       const float* denominator_coefficients,
+                                       size_t order_denominator) {
+  if (order_numerator > kMaxFilterOrder ||
+      order_denominator > kMaxFilterOrder || denominator_coefficients[0] == 0 ||
+      numerator_coefficients == NULL || denominator_coefficients == NULL)
+    return NULL;
+  return new PoleZeroFilter(numerator_coefficients, order_numerator,
+                            denominator_coefficients, order_denominator);
+}
+
+PoleZeroFilter::PoleZeroFilter(const float* numerator_coefficients,
+                               size_t order_numerator,
+                               const float* denominator_coefficients,
+                               size_t order_denominator)
+    : past_input_(),
+      past_output_(),
+      numerator_coefficients_(),
+      denominator_coefficients_(),
+      order_numerator_(order_numerator),
+      order_denominator_(order_denominator),
+      highest_order_(std::max(order_denominator, order_numerator)) {
+  memcpy(numerator_coefficients_, numerator_coefficients,
+         sizeof(numerator_coefficients_[0]) * (order_numerator_ + 1));
+  memcpy(denominator_coefficients_, denominator_coefficients,
+         sizeof(denominator_coefficients_[0]) * (order_denominator_ + 1));
+
+  if (denominator_coefficients_[0] != 1) {
+    for (size_t n = 0; n <= order_numerator_; n++)
+      numerator_coefficients_[n] /= denominator_coefficients_[0];
+    for (size_t n = 0; n <= order_denominator_; n++)
+      denominator_coefficients_[n] /= denominator_coefficients_[0];
+  }
+}
+
+template <typename T>
+static float FilterArPast(const T* past,
+                          size_t order,
+                          const float* coefficients) {
+  float sum = 0.0f;
+  size_t past_index = order - 1;
+  for (size_t k = 1; k <= order; k++, past_index--)
+    sum += coefficients[k] * past[past_index];
+  return sum;
+}
+
+int PoleZeroFilter::Filter(const int16_t* in,
+                           size_t num_input_samples,
+                           float* output) {
+  if (in == NULL || output == NULL)
+    return -1;
+  // This is the typical case, just a memcpy.
+  const size_t k = std::min(num_input_samples, highest_order_);
+  size_t n;
+  for (n = 0; n < k; n++) {
+    output[n] = in[n] * numerator_coefficients_[0];
+    output[n] += FilterArPast(&past_input_[n], order_numerator_,
+                              numerator_coefficients_);
+    output[n] -= FilterArPast(&past_output_[n], order_denominator_,
+                              denominator_coefficients_);
+
+    past_input_[n + order_numerator_] = in[n];
+    past_output_[n + order_denominator_] = output[n];
+  }
+  if (highest_order_ < num_input_samples) {
+    for (size_t m = 0; n < num_input_samples; n++, m++) {
+      output[n] = in[n] * numerator_coefficients_[0];
+      output[n] +=
+          FilterArPast(&in[m], order_numerator_, numerator_coefficients_);
+      output[n] -= FilterArPast(&output[m], order_denominator_,
+                                denominator_coefficients_);
+    }
+    // Record into the past signal.
+    memcpy(past_input_, &in[num_input_samples - order_numerator_],
+           sizeof(in[0]) * order_numerator_);
+    memcpy(past_output_, &output[num_input_samples - order_denominator_],
+           sizeof(output[0]) * order_denominator_);
+  } else {
+    // Odd case that the length of the input is shorter that filter order.
+    memmove(past_input_, &past_input_[num_input_samples],
+            order_numerator_ * sizeof(past_input_[0]));
+    memmove(past_output_, &past_output_[num_input_samples],
+            order_denominator_ * sizeof(past_output_[0]));
+  }
+  return 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter.h b/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter.h
new file mode 100644
index 0000000000..11a05114d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+class PoleZeroFilter {
+ public:
+  ~PoleZeroFilter() {}
+
+  static PoleZeroFilter* Create(const float* numerator_coefficients,
+                                size_t order_numerator,
+                                const float* denominator_coefficients,
+                                size_t order_denominator);
+
+  int Filter(const int16_t* in, size_t num_input_samples, float* output);
+
+ private:
+  PoleZeroFilter(const float* numerator_coefficients,
+                 size_t order_numerator,
+                 const float* denominator_coefficients,
+                 size_t order_denominator);
+
+  static const int kMaxFilterOrder = 24;
+
+  int16_t past_input_[kMaxFilterOrder * 2];
+  float past_output_[kMaxFilterOrder * 2];
+
+  float numerator_coefficients_[kMaxFilterOrder + 1];
+  float denominator_coefficients_[kMaxFilterOrder + 1];
+
+  size_t order_numerator_;
+  size_t order_denominator_;
+  size_t highest_order_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc b/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc
new file mode 100644
index 0000000000..8088b40125
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/pole_zero_filter.h"
+
+#include <math.h>
+#include <stdio.h>
+
+#include <memory>
+
+#include "modules/audio_processing/vad/vad_audio_proc_internal.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+static const int kInputSamples = 50;
+
+static const int16_t kInput[kInputSamples] = {
+    -2136,  -7116, 10715,  2464,   3164,   8139,   11393, 24013, -32117, -5544,
+    -27740, 10181, 14190,  -24055, -15912, 17393,  6359,  -9950, -13894, 32432,
+    -23944, 3437,  -8381,  19768,  3087,   -19795, -5920, 13310, 1407,   3876,
+    4059,   3524,  -23130, 19121,  -27900, -24840, 4089,  21422, -3625,  3015,
+    -11236, 28856, 13424,  6571,   -19761, -6361,  15821, -9469, 29727,  32229};
+
+static const float kReferenceOutput[kInputSamples] = {
+    -2082.230472f,  -6878.572941f,  10697.090871f,  2358.373952f,
+    2973.936512f,   7738.580650f,   10690.803213f,  22687.091576f,
+    -32676.684717f, -5879.621684f,  -27359.297432f, 10368.735888f,
+    13994.584604f,  -23676.126249f, -15078.250390f, 17818.253338f,
+    6577.743123f,   -9498.369315f,  -13073.651079f, 32460.026588f,
+    -23391.849347f, 3953.805667f,   -7667.761363f,  19995.153447f,
+    3185.575477f,   -19207.365160f, -5143.103201f,  13756.317237f,
+    1779.654794f,   4142.269755f,   4209.475034f,   3572.991789f,
+    -22509.089546f, 19307.878964f,  -27060.439759f, -23319.042810f,
+    5547.685267f,   22312.718676f,  -2707.309027f,  3852.358490f,
+    -10135.510093f, 29241.509970f,  13394.397233f,  6340.721417f,
+    -19510.207905f, -5908.442086f,  15882.301634f,  -9211.335255f,
+    29253.056735f,  30874.443046f};
+
+class PoleZeroFilterTest : public ::testing::Test {
+ protected:
+  PoleZeroFilterTest()
+      : my_filter_(PoleZeroFilter::Create(kCoeffNumerator,
+                                          kFilterOrder,
+                                          kCoeffDenominator,
+                                          kFilterOrder)) {}
+
+  ~PoleZeroFilterTest() override {}
+
+  void FilterSubframes(int num_subframes);
+
+ private:
+  void TestClean();
+  std::unique_ptr<PoleZeroFilter> my_filter_;
+};
+
+void PoleZeroFilterTest::FilterSubframes(int num_subframes) {
+  float output[kInputSamples];
+  const int num_subframe_samples = kInputSamples / num_subframes;
+  EXPECT_EQ(num_subframe_samples * num_subframes, kInputSamples);
+
+  for (int n = 0; n < num_subframes; n++) {
+    my_filter_->Filter(&kInput[n * num_subframe_samples], num_subframe_samples,
+                       &output[n * num_subframe_samples]);
+  }
+  for (int n = 0; n < kInputSamples; n++) {
+    EXPECT_NEAR(output[n], kReferenceOutput[n], 1);
+  }
+}
+
+TEST_F(PoleZeroFilterTest, OneSubframe) {
+  FilterSubframes(1);
+}
+
+TEST_F(PoleZeroFilterTest, TwoSubframes) {
+  FilterSubframes(2);
+}
+
+TEST_F(PoleZeroFilterTest, FiveSubframes) {
+  FilterSubframes(5);
+}
+
+TEST_F(PoleZeroFilterTest, TenSubframes) {
+  FilterSubframes(10);
+}
+
+TEST_F(PoleZeroFilterTest, TwentyFiveSubframes) {
+  FilterSubframes(25);
+}
+
+TEST_F(PoleZeroFilterTest, FiftySubframes) {
+  FilterSubframes(50);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad.cc b/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad.cc
new file mode 100644
index 0000000000..1397668eb4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/standalone_vad.h"
+
+#include <string.h>
+
+#include "common_audio/vad/include/webrtc_vad.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+static const int kDefaultStandaloneVadMode = 3;
+
+StandaloneVad::StandaloneVad(VadInst* vad)
+    : vad_(vad), buffer_(), index_(0), mode_(kDefaultStandaloneVadMode) {}
+
+StandaloneVad::~StandaloneVad() {
+  WebRtcVad_Free(vad_);
+}
+
+StandaloneVad* StandaloneVad::Create() {
+  VadInst* vad = WebRtcVad_Create();
+  if (!vad)
+    return nullptr;
+
+  int err = WebRtcVad_Init(vad);
+  err |= WebRtcVad_set_mode(vad, kDefaultStandaloneVadMode);
+  if (err != 0) {
+    WebRtcVad_Free(vad);
+    return nullptr;
+  }
+  return new StandaloneVad(vad);
+}
+
+int StandaloneVad::AddAudio(const int16_t* data, size_t length) {
+  if (length != kLength10Ms)
+    return -1;
+
+  if (index_ + length > kLength10Ms * kMaxNum10msFrames)
+    // Reset the buffer if it's full.
+    // TODO(ajm): Instead, consider just processing every 10 ms frame. Then we
+    // can forgo the buffering.
+    index_ = 0;
+
+  memcpy(&buffer_[index_], data, sizeof(int16_t) * length);
+  index_ += length;
+  return 0;
+}
+
+int StandaloneVad::GetActivity(double* p, size_t length_p) {
+  if (index_ == 0)
+    return -1;
+
+  const size_t num_frames = index_ / kLength10Ms;
+  if (num_frames > length_p)
+    return -1;
+  RTC_DCHECK_EQ(0, WebRtcVad_ValidRateAndFrameLength(kSampleRateHz, index_));
+
+  int activity = WebRtcVad_Process(vad_, kSampleRateHz, buffer_, index_);
+  if (activity < 0)
+    return -1;
+  else if (activity == 0)
+    p[0] = 0.01;  // Arbitrary but small and non-zero.
+  else
+    p[0] = 0.5;  // 0.5 is neutral values when combinned by other probabilities.
+  for (size_t n = 1; n < num_frames; n++)
+    p[n] = p[0];
+  // Reset the buffer to start from the beginning.
+  index_ = 0;
+  return activity;
+}
+
+int StandaloneVad::set_mode(int mode) {
+  if (mode < 0 || mode > 3)
+    return -1;
+  if (WebRtcVad_set_mode(vad_, mode) != 0)
+    return -1;
+
+  mode_ = mode;
+  return 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad.h b/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad.h
new file mode 100644
index 0000000000..b08463374e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
+#define MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "common_audio/vad/include/webrtc_vad.h"
+#include "modules/audio_processing/vad/common.h"
+
+namespace webrtc {
+
+class StandaloneVad {
+ public:
+  static StandaloneVad* Create();
+  ~StandaloneVad();
+
+  // Outputs
+  //   p: a buffer where probabilities are written to.
+  //   length_p: number of elements of `p`.
+  //
+  // return value:
+  //    -1: if no audio is stored or VAD returns error.
+  //     0: in success.
+  // In case of error the content of `activity` is unchanged.
+  //
+  // Note that due to a high false-positive (VAD decision is active while the
+  // processed audio is just background noise) rate, stand-alone VAD is used as
+  // a one-sided indicator. The activity probability is 0.5 if the frame is
+  // classified as active, and the probability is 0.01 if the audio is
+  // classified as passive. In this way, when probabilities are combined, the
+  // effect of the stand-alone VAD is neutral if the input is classified as
+  // active.
+  int GetActivity(double* p, size_t length_p);
+
+  // Expecting 10 ms of 16 kHz audio to be pushed in.
+  int AddAudio(const int16_t* data, size_t length);
+
+  // Set aggressiveness of VAD, 0 is the least aggressive and 3 is the most
+  // aggressive mode. Returns -1 if the input is less than 0 or larger than 3,
+  // otherwise 0 is returned.
+  int set_mode(int mode);
+  // Get the agressiveness of the current VAD.
+  int mode() const { return mode_; }
+
+ private:
+  explicit StandaloneVad(VadInst* vad);
+
+  static const size_t kMaxNum10msFrames = 3;
+
+  // TODO(turajs): Is there a way to use scoped-pointer here?
+  VadInst* vad_;
+  int16_t buffer_[kMaxNum10msFrames * kLength10Ms];
+  size_t index_;
+  int mode_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad_unittest.cc b/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
new file mode 100644
index 0000000000..0fa2ed78b1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/standalone_vad.h"
+
+#include <string.h>
+
+#include <memory>
+
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TEST(StandaloneVadTest, Api) {
+  std::unique_ptr<StandaloneVad> vad(StandaloneVad::Create());
+  int16_t data[kLength10Ms] = {0};
+
+  // Valid frame length (for 32 kHz rate), but not what the VAD is expecting.
+  EXPECT_EQ(-1, vad->AddAudio(data, 320));
+
+  const size_t kMaxNumFrames = 3;
+  double p[kMaxNumFrames];
+  for (size_t n = 0; n < kMaxNumFrames; n++)
+    EXPECT_EQ(0, vad->AddAudio(data, kLength10Ms));
+
+  // Pretend `p` is shorter that it should be.
+  EXPECT_EQ(-1, vad->GetActivity(p, kMaxNumFrames - 1));
+
+  EXPECT_EQ(0, vad->GetActivity(p, kMaxNumFrames));
+
+  // Ask for activity when buffer is empty.
+  EXPECT_EQ(-1, vad->GetActivity(p, kMaxNumFrames));
+
+  // Should reset and result in one buffer.
+  for (size_t n = 0; n < kMaxNumFrames + 1; n++)
+    EXPECT_EQ(0, vad->AddAudio(data, kLength10Ms));
+  EXPECT_EQ(0, vad->GetActivity(p, 1));
+
+  // Wrong modes
+  EXPECT_EQ(-1, vad->set_mode(-1));
+  EXPECT_EQ(-1, vad->set_mode(4));
+
+  // Valid mode.
+  const int kMode = 2;
+  EXPECT_EQ(0, vad->set_mode(kMode));
+  EXPECT_EQ(kMode, vad->mode());
+}
+
+#if defined(WEBRTC_IOS)
+TEST(StandaloneVadTest, DISABLED_ActivityDetection) {
+#else
+TEST(StandaloneVadTest, ActivityDetection) {
+#endif
+  std::unique_ptr<StandaloneVad> vad(StandaloneVad::Create());
+  const size_t kDataLength = kLength10Ms;
+  int16_t data[kDataLength] = {0};
+
+  FILE* pcm_file =
+      fopen(test::ResourcePath("audio_processing/agc/agc_audio", "pcm").c_str(),
+            "rb");
+  ASSERT_TRUE(pcm_file != NULL);
+
+  FILE* reference_file = fopen(
+      test::ResourcePath("audio_processing/agc/agc_vad", "dat").c_str(), "rb");
+  ASSERT_TRUE(reference_file != NULL);
+
+  // Reference activities are prepared with 0 aggressiveness.
+  ASSERT_EQ(0, vad->set_mode(0));
+
+  // Stand-alone VAD can operate on 1, 2 or 3 frames of length 10 ms. The
+  // reference file is created for 30 ms frame.
+  const int kNumVadFramesToProcess = 3;
+  int num_frames = 0;
+  while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) {
+    vad->AddAudio(data, kDataLength);
+    num_frames++;
+    if (num_frames == kNumVadFramesToProcess) {
+      num_frames = 0;
+      int referece_activity;
+      double p[kNumVadFramesToProcess];
+      EXPECT_EQ(1u, fread(&referece_activity, sizeof(referece_activity), 1,
+                          reference_file));
+      int activity = vad->GetActivity(p, kNumVadFramesToProcess);
+      EXPECT_EQ(referece_activity, activity);
+      if (activity != 0) {
+        // When active, probabilities are set to 0.5.
+        for (int n = 0; n < kNumVadFramesToProcess; n++)
+          EXPECT_EQ(0.5, p[n]);
+      } else {
+        // When inactive, probabilities are set to 0.01.
+        for (int n = 0; n < kNumVadFramesToProcess; n++)
+          EXPECT_EQ(0.01, p[n]);
+      }
+    }
+  }
+  fclose(reference_file);
+  fclose(pcm_file);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc.cc b/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc.cc
new file mode 100644
index 0000000000..aaf8214d7c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc.cc
@@ -0,0 +1,275 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/vad_audio_proc.h"
+
+#include <math.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "common_audio/third_party/ooura/fft_size_256/fft4g.h"
+#include "modules/audio_processing/vad/pitch_internal.h"
+#include "modules/audio_processing/vad/pole_zero_filter.h"
+#include "modules/audio_processing/vad/vad_audio_proc_internal.h"
+#include "rtc_base/checks.h"
+extern "C" {
+#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h"
+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+}
+
+namespace webrtc {
+
+// The following structures are declared anonymous in iSAC's structs.h. To
+// forward declare them, we use this derived class trick.
+struct VadAudioProc::PitchAnalysisStruct : public ::PitchAnalysisStruct {};
+struct VadAudioProc::PreFiltBankstr : public ::PreFiltBankstr {};
+
+static constexpr float kFrequencyResolution =
+    kSampleRateHz / static_cast<float>(VadAudioProc::kDftSize);
+static constexpr int kSilenceRms = 5;
+
+// TODO(turajs): Make a Create or Init for VadAudioProc.
+VadAudioProc::VadAudioProc()
+    : audio_buffer_(),
+      num_buffer_samples_(kNumPastSignalSamples),
+      log_old_gain_(-2),
+      old_lag_(50),  // Arbitrary but valid as pitch-lag (in samples).
+      pitch_analysis_handle_(new PitchAnalysisStruct),
+      pre_filter_handle_(new PreFiltBankstr),
+      high_pass_filter_(PoleZeroFilter::Create(kCoeffNumerator,
+                                               kFilterOrder,
+                                               kCoeffDenominator,
+                                               kFilterOrder)) {
+  static_assert(kNumPastSignalSamples + kNumSubframeSamples ==
+                    sizeof(kLpcAnalWin) / sizeof(kLpcAnalWin[0]),
+                "lpc analysis window incorrect size");
+  static_assert(kLpcOrder + 1 == sizeof(kCorrWeight) / sizeof(kCorrWeight[0]),
+                "correlation weight incorrect size");
+
+  // TODO(turajs): Are we doing too much in the constructor?
+  float data[kDftSize];
+  // Make FFT to initialize.
+  ip_[0] = 0;
+  WebRtc_rdft(kDftSize, 1, data, ip_, w_fft_);
+  // TODO(turajs): Need to initialize high-pass filter.
+
+  // Initialize iSAC components.
+  WebRtcIsac_InitPreFilterbank(pre_filter_handle_.get());
+  WebRtcIsac_InitPitchAnalysis(pitch_analysis_handle_.get());
+}
+
+VadAudioProc::~VadAudioProc() {}
+
+void VadAudioProc::ResetBuffer() {
+  memcpy(audio_buffer_, &audio_buffer_[kNumSamplesToProcess],
+         sizeof(audio_buffer_[0]) * kNumPastSignalSamples);
+  num_buffer_samples_ = kNumPastSignalSamples;
+}
+
+int VadAudioProc::ExtractFeatures(const int16_t* frame,
+                                  size_t length,
+                                  AudioFeatures* features) {
+  features->num_frames = 0;
+  if (length != kNumSubframeSamples) {
+    return -1;
+  }
+
+  // High-pass filter to remove the DC component and very low frequency content.
+  // We have experienced that this high-pass filtering improves voice/non-voiced
+  // classification.
+  if (high_pass_filter_->Filter(frame, kNumSubframeSamples,
+                                &audio_buffer_[num_buffer_samples_]) != 0) {
+    return -1;
+  }
+
+  num_buffer_samples_ += kNumSubframeSamples;
+  if (num_buffer_samples_ < kBufferLength) {
+    return 0;
+  }
+  RTC_DCHECK_EQ(num_buffer_samples_, kBufferLength);
+  features->num_frames = kNum10msSubframes;
+  features->silence = false;
+
+  Rms(features->rms, kMaxNumFrames);
+  for (size_t i = 0; i < kNum10msSubframes; ++i) {
+    if (features->rms[i] < kSilenceRms) {
+      // PitchAnalysis can cause NaNs in the pitch gain if it's fed silence.
+      // Bail out here instead.
+      features->silence = true;
+      ResetBuffer();
+      return 0;
+    }
+  }
+
+  PitchAnalysis(features->log_pitch_gain, features->pitch_lag_hz,
+                kMaxNumFrames);
+  FindFirstSpectralPeaks(features->spectral_peak, kMaxNumFrames);
+  ResetBuffer();
+  return 0;
+}
+
+// Computes |kLpcOrder + 1| correlation coefficients.
+void VadAudioProc::SubframeCorrelation(double* corr,
+                                       size_t length_corr,
+                                       size_t subframe_index) {
+  RTC_DCHECK_GE(length_corr, kLpcOrder + 1);
+  double windowed_audio[kNumSubframeSamples + kNumPastSignalSamples];
+  size_t buffer_index = subframe_index * kNumSubframeSamples;
+
+  for (size_t n = 0; n < kNumSubframeSamples + kNumPastSignalSamples; n++)
+    windowed_audio[n] = audio_buffer_[buffer_index++] * kLpcAnalWin[n];
+
+  WebRtcIsac_AutoCorr(corr, windowed_audio,
+                      kNumSubframeSamples + kNumPastSignalSamples, kLpcOrder);
+}
+
+// Compute `kNum10msSubframes` sets of LPC coefficients, one per 10 ms input.
+// The analysis window is 15 ms long and it is centered on the first half of
+// each 10ms sub-frame. This is equivalent to computing LPC coefficients for the
+// first half of each 10 ms subframe.
+void VadAudioProc::GetLpcPolynomials(double* lpc, size_t length_lpc) {
+  RTC_DCHECK_GE(length_lpc, kNum10msSubframes * (kLpcOrder + 1));
+  double corr[kLpcOrder + 1];
+  double reflec_coeff[kLpcOrder];
+  for (size_t i = 0, offset_lpc = 0; i < kNum10msSubframes;
+       i++, offset_lpc += kLpcOrder + 1) {
+    SubframeCorrelation(corr, kLpcOrder + 1, i);
+    corr[0] *= 1.0001;
+    // This makes Lev-Durb a bit more stable.
+    for (size_t k = 0; k < kLpcOrder + 1; k++) {
+      corr[k] *= kCorrWeight[k];
+    }
+    WebRtcIsac_LevDurb(&lpc[offset_lpc], reflec_coeff, corr, kLpcOrder);
+  }
+}
+
+// Fit a second order curve to these 3 points and find the location of the
+// extremum. The points are inverted before curve fitting.
+static float QuadraticInterpolation(float prev_val,
+                                    float curr_val,
+                                    float next_val) {
+  // Doing the interpolation in |1 / A(z)|^2.
+  float fractional_index = 0;
+  next_val = 1.0f / next_val;
+  prev_val = 1.0f / prev_val;
+  curr_val = 1.0f / curr_val;
+
+  fractional_index =
+      -(next_val - prev_val) * 0.5f / (next_val + prev_val - 2.f * curr_val);
+  RTC_DCHECK_LT(fabs(fractional_index), 1);
+  return fractional_index;
+}
+
+// 1 / A(z), where A(z) is defined by `lpc` is a model of the spectral envelope
+// of the input signal. The local maximum of the spectral envelope corresponds
+// with the local minimum of A(z). It saves complexity, as we save one
+// inversion. Furthermore, we find the first local maximum of magnitude squared,
+// to save on one square root.
+void VadAudioProc::FindFirstSpectralPeaks(double* f_peak,
+                                          size_t length_f_peak) {
+  RTC_DCHECK_GE(length_f_peak, kNum10msSubframes);
+  double lpc[kNum10msSubframes * (kLpcOrder + 1)];
+  // For all sub-frames.
+  GetLpcPolynomials(lpc, kNum10msSubframes * (kLpcOrder + 1));
+
+  const size_t kNumDftCoefficients = kDftSize / 2 + 1;
+  float data[kDftSize];
+
+  for (size_t i = 0; i < kNum10msSubframes; i++) {
+    // Convert to float with zero pad.
+    memset(data, 0, sizeof(data));
+    for (size_t n = 0; n < kLpcOrder + 1; n++) {
+      data[n] = static_cast<float>(lpc[i * (kLpcOrder + 1) + n]);
+    }
+    // Transform to frequency domain.
+    WebRtc_rdft(kDftSize, 1, data, ip_, w_fft_);
+
+    size_t index_peak = 0;
+    float prev_magn_sqr = data[0] * data[0];
+    float curr_magn_sqr = data[2] * data[2] + data[3] * data[3];
+    float next_magn_sqr;
+    bool found_peak = false;
+    for (size_t n = 2; n < kNumDftCoefficients - 1; n++) {
+      next_magn_sqr =
+          data[2 * n] * data[2 * n] + data[2 * n + 1] * data[2 * n + 1];
+      if (curr_magn_sqr < prev_magn_sqr && curr_magn_sqr < next_magn_sqr) {
+        found_peak = true;
+        index_peak = n - 1;
+        break;
+      }
+      prev_magn_sqr = curr_magn_sqr;
+      curr_magn_sqr = next_magn_sqr;
+    }
+    float fractional_index = 0;
+    if (!found_peak) {
+      // Checking if |kNumDftCoefficients - 1| is the local minimum.
+      next_magn_sqr = data[1] * data[1];
+      if (curr_magn_sqr < prev_magn_sqr && curr_magn_sqr < next_magn_sqr) {
+        index_peak = kNumDftCoefficients - 1;
+      }
+    } else {
+      // A peak is found, do a simple quadratic interpolation to get a more
+      // accurate estimate of the peak location.
+      fractional_index =
+          QuadraticInterpolation(prev_magn_sqr, curr_magn_sqr, next_magn_sqr);
+    }
+    f_peak[i] = (index_peak + fractional_index) * kFrequencyResolution;
+  }
+}
+
+// Using iSAC functions to estimate pitch gains & lags.
+void VadAudioProc::PitchAnalysis(double* log_pitch_gains,
+                                 double* pitch_lags_hz,
+                                 size_t length) {
+  // TODO(turajs): This can be "imported" from iSAC & and the next two
+  // constants.
+  RTC_DCHECK_GE(length, kNum10msSubframes);
+  const int kNumPitchSubframes = 4;
+  double gains[kNumPitchSubframes];
+  double lags[kNumPitchSubframes];
+
+  const int kNumSubbandFrameSamples = 240;
+  const int kNumLookaheadSamples = 24;
+
+  float lower[kNumSubbandFrameSamples];
+  float upper[kNumSubbandFrameSamples];
+  double lower_lookahead[kNumSubbandFrameSamples];
+  double upper_lookahead[kNumSubbandFrameSamples];
+  double lower_lookahead_pre_filter[kNumSubbandFrameSamples +
+                                    kNumLookaheadSamples];
+
+  // Split signal to lower and upper bands
+  WebRtcIsac_SplitAndFilterFloat(&audio_buffer_[kNumPastSignalSamples], lower,
+                                 upper, lower_lookahead, upper_lookahead,
+                                 pre_filter_handle_.get());
+  WebRtcIsac_PitchAnalysis(lower_lookahead, lower_lookahead_pre_filter,
+                           pitch_analysis_handle_.get(), lags, gains);
+
+  // Lags are computed on lower-band signal with sampling rate half of the
+  // input signal.
+  GetSubframesPitchParameters(
+      kSampleRateHz / 2, gains, lags, kNumPitchSubframes, kNum10msSubframes,
+      &log_old_gain_, &old_lag_, log_pitch_gains, pitch_lags_hz);
+}
+
+void VadAudioProc::Rms(double* rms, size_t length_rms) {
+  RTC_DCHECK_GE(length_rms, kNum10msSubframes);
+  size_t offset = kNumPastSignalSamples;
+  for (size_t i = 0; i < kNum10msSubframes; i++) {
+    rms[i] = 0;
+    for (size_t n = 0; n < kNumSubframeSamples; n++, offset++)
+      rms[i] += audio_buffer_[offset] * audio_buffer_[offset];
+    rms[i] = sqrt(rms[i] / kNumSubframeSamples);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc.h b/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc.h
new file mode 100644
index 0000000000..cbdd707129
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc.h
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "modules/audio_processing/vad/common.h"  // AudioFeatures, kSampleR...
+
+namespace webrtc {
+
+class PoleZeroFilter;
+
+class VadAudioProc {
+ public:
+  // Forward declare iSAC structs.
+  struct PitchAnalysisStruct;
+  struct PreFiltBankstr;
+
+  VadAudioProc();
+  ~VadAudioProc();
+
+  int ExtractFeatures(const int16_t* audio_frame,
+                      size_t length,
+                      AudioFeatures* audio_features);
+
+  static constexpr size_t kDftSize = 512;
+
+ private:
+  void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, size_t length);
+  void SubframeCorrelation(double* corr,
+                           size_t length_corr,
+                           size_t subframe_index);
+  void GetLpcPolynomials(double* lpc, size_t length_lpc);
+  void FindFirstSpectralPeaks(double* f_peak, size_t length_f_peak);
+  void Rms(double* rms, size_t length_rms);
+  void ResetBuffer();
+
+  // To compute spectral peak we perform LPC analysis to get spectral envelope.
+  // For every 30 ms we compute 3 spectral peak there for 3 LPC analysis.
+  // LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame
+  // we need 5 ms of past signal to create the input of LPC analysis.
+  static constexpr size_t kNumPastSignalSamples =
+      static_cast<size_t>(kSampleRateHz / 200);
+
+  // TODO(turajs): maybe defining this at a higher level (maybe enum) so that
+  // all the code recognize it as "no-error."
+  static constexpr int kNoError = 0;
+
+  static constexpr size_t kNum10msSubframes = 3;
+  static constexpr size_t kNumSubframeSamples =
+      static_cast<size_t>(kSampleRateHz / 100);
+  // Samples in 30 ms @ given sampling rate.
+  static constexpr size_t kNumSamplesToProcess =
+      size_t{kNum10msSubframes} * kNumSubframeSamples;
+  static constexpr size_t kBufferLength =
+      size_t{kNumPastSignalSamples} + kNumSamplesToProcess;
+  static constexpr size_t kIpLength = kDftSize >> 1;
+  static constexpr size_t kWLength = kDftSize >> 1;
+  static constexpr size_t kLpcOrder = 16;
+
+  size_t ip_[kIpLength];
+  float w_fft_[kWLength];
+
+  // A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ).
+  float audio_buffer_[kBufferLength];
+  size_t num_buffer_samples_;
+
+  double log_old_gain_;
+  double old_lag_;
+
+  std::unique_ptr<PitchAnalysisStruct> pitch_analysis_handle_;
+  std::unique_ptr<PreFiltBankstr> pre_filter_handle_;
+  std::unique_ptr<PoleZeroFilter> high_pass_filter_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc_internal.h b/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc_internal.h
new file mode 100644
index 0000000000..93589affe8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc_internal.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_INTERNAL_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_INTERNAL_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+
+// These values should match MATLAB counterparts for unit-tests to pass.
+static const double kCorrWeight[] = {
+    1.000000, 0.985000, 0.970225, 0.955672, 0.941337, 0.927217,
+    0.913308, 0.899609, 0.886115, 0.872823, 0.859730, 0.846834,
+    0.834132, 0.821620, 0.809296, 0.797156, 0.785199};
+
+static const double kLpcAnalWin[] = {
+    0.00000000, 0.01314436, 0.02628645, 0.03942400, 0.05255473, 0.06567639,
+    0.07878670, 0.09188339, 0.10496421, 0.11802689, 0.13106918, 0.14408883,
+    0.15708358, 0.17005118, 0.18298941, 0.19589602, 0.20876878, 0.22160547,
+    0.23440387, 0.24716177, 0.25987696, 0.27254725, 0.28517045, 0.29774438,
+    0.31026687, 0.32273574, 0.33514885, 0.34750406, 0.35979922, 0.37203222,
+    0.38420093, 0.39630327, 0.40833713, 0.42030043, 0.43219112, 0.44400713,
+    0.45574642, 0.46740697, 0.47898676, 0.49048379, 0.50189608, 0.51322164,
+    0.52445853, 0.53560481, 0.54665854, 0.55761782, 0.56848075, 0.57924546,
+    0.58991008, 0.60047278, 0.61093173, 0.62128512, 0.63153117, 0.64166810,
+    0.65169416, 0.66160761, 0.67140676, 0.68108990, 0.69065536, 0.70010148,
+    0.70942664, 0.71862923, 0.72770765, 0.73666033, 0.74548573, 0.75418233,
+    0.76274862, 0.77118312, 0.77948437, 0.78765094, 0.79568142, 0.80357442,
+    0.81132858, 0.81894256, 0.82641504, 0.83374472, 0.84093036, 0.84797069,
+    0.85486451, 0.86161063, 0.86820787, 0.87465511, 0.88095122, 0.88709512,
+    0.89308574, 0.89892206, 0.90460306, 0.91012776, 0.91549520, 0.92070447,
+    0.92575465, 0.93064488, 0.93537432, 0.93994213, 0.94434755, 0.94858979,
+    0.95266814, 0.95658189, 0.96033035, 0.96391289, 0.96732888, 0.97057773,
+    0.97365889, 0.97657181, 0.97931600, 0.98189099, 0.98429632, 0.98653158,
+    0.98859639, 0.99049038, 0.99221324, 0.99376466, 0.99514438, 0.99635215,
+    0.99738778, 0.99825107, 0.99894188, 0.99946010, 0.99980562, 0.99997840,
+    0.99997840, 0.99980562, 0.99946010, 0.99894188, 0.99825107, 0.99738778,
+    0.99635215, 0.99514438, 0.99376466, 0.99221324, 0.99049038, 0.98859639,
+    0.98653158, 0.98429632, 0.98189099, 0.97931600, 0.97657181, 0.97365889,
+    0.97057773, 0.96732888, 0.96391289, 0.96033035, 0.95658189, 0.95266814,
+    0.94858979, 0.94434755, 0.93994213, 0.93537432, 0.93064488, 0.92575465,
+    0.92070447, 0.91549520, 0.91012776, 0.90460306, 0.89892206, 0.89308574,
+    0.88709512, 0.88095122, 0.87465511, 0.86820787, 0.86161063, 0.85486451,
+    0.84797069, 0.84093036, 0.83374472, 0.82641504, 0.81894256, 0.81132858,
+    0.80357442, 0.79568142, 0.78765094, 0.77948437, 0.77118312, 0.76274862,
+    0.75418233, 0.74548573, 0.73666033, 0.72770765, 0.71862923, 0.70942664,
+    0.70010148, 0.69065536, 0.68108990, 0.67140676, 0.66160761, 0.65169416,
+    0.64166810, 0.63153117, 0.62128512, 0.61093173, 0.60047278, 0.58991008,
+    0.57924546, 0.56848075, 0.55761782, 0.54665854, 0.53560481, 0.52445853,
+    0.51322164, 0.50189608, 0.49048379, 0.47898676, 0.46740697, 0.45574642,
+    0.44400713, 0.43219112, 0.42030043, 0.40833713, 0.39630327, 0.38420093,
+    0.37203222, 0.35979922, 0.34750406, 0.33514885, 0.32273574, 0.31026687,
+    0.29774438, 0.28517045, 0.27254725, 0.25987696, 0.24716177, 0.23440387,
+    0.22160547, 0.20876878, 0.19589602, 0.18298941, 0.17005118, 0.15708358,
+    0.14408883, 0.13106918, 0.11802689, 0.10496421, 0.09188339, 0.07878670,
+    0.06567639, 0.05255473, 0.03942400, 0.02628645, 0.01314436, 0.00000000};
+
+static const size_t kFilterOrder = 2;
+static const float kCoeffNumerator[kFilterOrder + 1] = {0.974827f, -1.949650f,
+                                                        0.974827f};
+static const float kCoeffDenominator[kFilterOrder + 1] = {1.0f, -1.971999f,
+                                                          0.972457f};
+
+static_assert(kFilterOrder + 1 ==
+                  sizeof(kCoeffNumerator) / sizeof(kCoeffNumerator[0]),
+              "numerator coefficients incorrect size");
+static_assert(kFilterOrder + 1 ==
+                  sizeof(kCoeffDenominator) / sizeof(kCoeffDenominator[0]),
+              "denominator coefficients incorrect size");
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROCESSING_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
new file mode 100644
index 0000000000..0afed84c35
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// We don't test the value of pitch gain and lags as they are created by iSAC
+// routines. However, interpolation of pitch-gain and lags is in a separate
+// class and has its own unit-test.
+
+#include "modules/audio_processing/vad/vad_audio_proc.h"
+
+#include <math.h>
+#include <stdio.h>
+
+#include <string>
+
+#include "modules/audio_processing/vad/common.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TEST(AudioProcessingTest, DISABLED_ComputingFirstSpectralPeak) {
+  VadAudioProc audioproc;
+
+  std::string peak_file_name =
+      test::ResourcePath("audio_processing/agc/agc_spectral_peak", "dat");
+  FILE* peak_file = fopen(peak_file_name.c_str(), "rb");
+  ASSERT_TRUE(peak_file != NULL);
+
+  std::string pcm_file_name =
+      test::ResourcePath("audio_processing/agc/agc_audio", "pcm");
+  FILE* pcm_file = fopen(pcm_file_name.c_str(), "rb");
+  ASSERT_TRUE(pcm_file != NULL);
+
+  // Read 10 ms audio in each iteration.
+  const size_t kDataLength = kLength10Ms;
+  int16_t data[kDataLength] = {0};
+  AudioFeatures features;
+  double sp[kMaxNumFrames];
+  while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) {
+    audioproc.ExtractFeatures(data, kDataLength, &features);
+    if (features.num_frames > 0) {
+      ASSERT_LT(features.num_frames, kMaxNumFrames);
+      // Read reference values.
+      const size_t num_frames = features.num_frames;
+      ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file));
+      for (size_t n = 0; n < features.num_frames; n++)
+        EXPECT_NEAR(features.spectral_peak[n], sp[n], 3);
+    }
+  }
+
+  fclose(peak_file);
+  fclose(pcm_file);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer.cc b/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer.cc
new file mode 100644
index 0000000000..31f14d7f64
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/vad_circular_buffer.h"
+
+#include <stdlib.h>
+
+namespace webrtc {
+
+VadCircularBuffer::VadCircularBuffer(int buffer_size)
+    : buffer_(new double[buffer_size]),
+      is_full_(false),
+      index_(0),
+      buffer_size_(buffer_size),
+      sum_(0) {}
+
+VadCircularBuffer::~VadCircularBuffer() {}
+
+void VadCircularBuffer::Reset() {
+  is_full_ = false;
+  index_ = 0;
+  sum_ = 0;
+}
+
+VadCircularBuffer* VadCircularBuffer::Create(int buffer_size) {
+  if (buffer_size <= 0)
+    return NULL;
+  return new VadCircularBuffer(buffer_size);
+}
+
+double VadCircularBuffer::Oldest() const {
+  if (!is_full_)
+    return buffer_[0];
+  else
+    return buffer_[index_];
+}
+
+double VadCircularBuffer::Mean() {
+  double m;
+  if (is_full_) {
+    m = sum_ / buffer_size_;
+  } else {
+    if (index_ > 0)
+      m = sum_ / index_;
+    else
+      m = 0;
+  }
+  return m;
+}
+
+void VadCircularBuffer::Insert(double value) {
+  if (is_full_) {
+    sum_ -= buffer_[index_];
+  }
+  sum_ += value;
+  buffer_[index_] = value;
+  index_++;
+  if (index_ >= buffer_size_) {
+    is_full_ = true;
+    index_ = 0;
+  }
+}
+int VadCircularBuffer::BufferLevel() {
+  if (is_full_)
+    return buffer_size_;
+  return index_;
+}
+
+int VadCircularBuffer::Get(int index, double* value) const {
+  int err = ConvertToLinearIndex(&index);
+  if (err < 0)
+    return -1;
+  *value = buffer_[index];
+  return 0;
+}
+
+int VadCircularBuffer::Set(int index, double value) {
+  int err = ConvertToLinearIndex(&index);
+  if (err < 0)
+    return -1;
+
+  sum_ -= buffer_[index];
+  buffer_[index] = value;
+  sum_ += value;
+  return 0;
+}
+
+int VadCircularBuffer::ConvertToLinearIndex(int* index) const {
+  if (*index < 0 || *index >= buffer_size_)
+    return -1;
+
+  if (!is_full_ && *index >= index_)
+    return -1;
+
+  *index = index_ - 1 - *index;
+  if (*index < 0)
+    *index += buffer_size_;
+  return 0;
+}
+
+int VadCircularBuffer::RemoveTransient(int width_threshold,
+                                       double val_threshold) {
+  if (!is_full_ && index_ < width_threshold + 2)
+    return 0;
+
+  int index_1 = 0;
+  int index_2 = width_threshold + 1;
+  double v = 0;
+  if (Get(index_1, &v) < 0)
+    return -1;
+  if (v < val_threshold) {
+    Set(index_1, 0);
+    int index;
+    for (index = index_2; index > index_1; index--) {
+      if (Get(index, &v) < 0)
+        return -1;
+      if (v < val_threshold)
+        break;
+    }
+    for (; index > index_1; index--) {
+      if (Set(index, 0.0) < 0)
+        return -1;
+    }
+  }
+  return 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer.h b/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer.h
new file mode 100644
index 0000000000..c1806f9e83
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_CIRCULAR_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VAD_CIRCULAR_BUFFER_H_
+
+#include <memory>
+
+namespace webrtc {
+
+// A circular buffer tailored to the need of this project. It stores last
+// K samples of the input, and keeps track of the mean of the last samples.
+//
+// It is used in class "PitchBasedActivity" to keep track of posterior
+// probabilities in the past few seconds. The posterior probabilities are used
+// to recursively update prior probabilities.
+class VadCircularBuffer {
+ public:
+  static VadCircularBuffer* Create(int buffer_size);
+  ~VadCircularBuffer();
+
+  // If buffer is wrapped around.
+  bool is_full() const { return is_full_; }
+  // Get the oldest entry in the buffer.
+  double Oldest() const;
+  // Insert new value into the buffer.
+  void Insert(double value);
+  // Reset buffer, forget the past, start fresh.
+  void Reset();
+
+  // The mean value of the elements in the buffer. The return value is zero if
+  // buffer is empty, i.e. no value is inserted.
+  double Mean();
+  // Remove transients. If the values exceed `val_threshold` for a period
+  // shorter then or equal to `width_threshold`, then that period is considered
+  // transient and set to zero.
+  int RemoveTransient(int width_threshold, double val_threshold);
+
+ private:
+  explicit VadCircularBuffer(int buffer_size);
+  // Get previous values. |index = 0| corresponds to the most recent
+  // insertion. |index = 1| is the one before the most recent insertion, and
+  // so on.
+  int Get(int index, double* value) const;
+  // Set a given position to `value`. `index` is interpreted as above.
+  int Set(int index, double value);
+  // Return the number of valid elements in the buffer.
+  int BufferLevel();
+
+  // Convert an index with the interpretation as get() method to the
+  // corresponding linear index.
+  int ConvertToLinearIndex(int* index) const;
+
+  std::unique_ptr<double[]> buffer_;
+  bool is_full_;
+  int index_;
+  int buffer_size_;
+  double sum_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VAD_CIRCULAR_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer_unittest.cc
new file mode 100644
index 0000000000..efbd70d9d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer_unittest.cc
@@ -0,0 +1,134 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/vad_circular_buffer.h"
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+static const int kWidthThreshold = 7;
+static const double kValThreshold = 1.0;
+static const int kLongBuffSize = 100;
+static const int kShortBuffSize = 10;
+
+static void InsertSequentially(int k, VadCircularBuffer* circular_buffer) {
+  double mean_val;
+  for (int n = 1; n <= k; n++) {
+    EXPECT_TRUE(!circular_buffer->is_full());
+    circular_buffer->Insert(n);
+    mean_val = circular_buffer->Mean();
+    EXPECT_EQ((n + 1.0) / 2., mean_val);
+  }
+}
+
+static void Insert(double value,
+                   int num_insertion,
+                   VadCircularBuffer* circular_buffer) {
+  for (int n = 0; n < num_insertion; n++)
+    circular_buffer->Insert(value);
+}
+
+static void InsertZeros(int num_zeros, VadCircularBuffer* circular_buffer) {
+  Insert(0.0, num_zeros, circular_buffer);
+}
+
+TEST(VadCircularBufferTest, GeneralTest) {
+  std::unique_ptr<VadCircularBuffer> circular_buffer(
+      VadCircularBuffer::Create(kShortBuffSize));
+  double mean_val;
+
+  // Mean should return zero if nothing is inserted.
+  mean_val = circular_buffer->Mean();
+  EXPECT_DOUBLE_EQ(0.0, mean_val);
+  InsertSequentially(kShortBuffSize, circular_buffer.get());
+
+  // Should be full.
+  EXPECT_TRUE(circular_buffer->is_full());
+  // Correct update after being full.
+  for (int n = 1; n < kShortBuffSize; n++) {
+    circular_buffer->Insert(n);
+    mean_val = circular_buffer->Mean();
+    EXPECT_DOUBLE_EQ((kShortBuffSize + 1.) / 2., mean_val);
+    EXPECT_TRUE(circular_buffer->is_full());
+  }
+
+  // Check reset. This should be like starting fresh.
+  circular_buffer->Reset();
+  mean_val = circular_buffer->Mean();
+  EXPECT_DOUBLE_EQ(0, mean_val);
+  InsertSequentially(kShortBuffSize, circular_buffer.get());
+  EXPECT_TRUE(circular_buffer->is_full());
+}
+
+TEST(VadCircularBufferTest, TransientsRemoval) {
+  std::unique_ptr<VadCircularBuffer> circular_buffer(
+      VadCircularBuffer::Create(kLongBuffSize));
+  // Let the first transient be in wrap-around.
+  InsertZeros(kLongBuffSize - kWidthThreshold / 2, circular_buffer.get());
+
+  double push_val = kValThreshold;
+  double mean_val;
+  for (int k = kWidthThreshold; k >= 1; k--) {
+    Insert(push_val, k, circular_buffer.get());
+    circular_buffer->Insert(0);
+    mean_val = circular_buffer->Mean();
+    EXPECT_DOUBLE_EQ(k * push_val / kLongBuffSize, mean_val);
+    circular_buffer->RemoveTransient(kWidthThreshold, kValThreshold);
+    mean_val = circular_buffer->Mean();
+    EXPECT_DOUBLE_EQ(0, mean_val);
+  }
+}
+
+TEST(VadCircularBufferTest, TransientDetection) {
+  std::unique_ptr<VadCircularBuffer> circular_buffer(
+      VadCircularBuffer::Create(kLongBuffSize));
+  // Let the first transient be in wrap-around.
+  int num_insertion = kLongBuffSize - kWidthThreshold / 2;
+  InsertZeros(num_insertion, circular_buffer.get());
+
+  double push_val = 2;
+  // This is longer than a transient and shouldn't be removed.
+  int num_non_zero_elements = kWidthThreshold + 1;
+  Insert(push_val, num_non_zero_elements, circular_buffer.get());
+
+  double mean_val = circular_buffer->Mean();
+  EXPECT_DOUBLE_EQ(num_non_zero_elements * push_val / kLongBuffSize, mean_val);
+  circular_buffer->Insert(0);
+  EXPECT_EQ(0,
+            circular_buffer->RemoveTransient(kWidthThreshold, kValThreshold));
+  mean_val = circular_buffer->Mean();
+  EXPECT_DOUBLE_EQ(num_non_zero_elements * push_val / kLongBuffSize, mean_val);
+
+  // A transient right after a non-transient, should be removed and mean is
+  // not changed.
+  num_insertion = 3;
+  Insert(push_val, num_insertion, circular_buffer.get());
+  circular_buffer->Insert(0);
+  EXPECT_EQ(0,
+            circular_buffer->RemoveTransient(kWidthThreshold, kValThreshold));
+  mean_val = circular_buffer->Mean();
+  EXPECT_DOUBLE_EQ(num_non_zero_elements * push_val / kLongBuffSize, mean_val);
+
+  // Last input is larger than threshold, although the sequence is short but
+  // it shouldn't be considered transient.
+  Insert(push_val, num_insertion, circular_buffer.get());
+  num_non_zero_elements += num_insertion;
+  EXPECT_EQ(0,
+            circular_buffer->RemoveTransient(kWidthThreshold, kValThreshold));
+  mean_val = circular_buffer->Mean();
+  EXPECT_DOUBLE_EQ(num_non_zero_elements * push_val / kLongBuffSize, mean_val);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/vad_gn/moz.build b/third_party/libwebrtc/modules/audio_processing/vad/vad_gn/moz.build
new file mode 100644
index 0000000000..0e76427c6e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/vad_gn/moz.build
@@ -0,0 +1,239 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/audio_processing/vad/gmm.cc",
+    "/third_party/libwebrtc/modules/audio_processing/vad/pitch_based_vad.cc",
+    "/third_party/libwebrtc/modules/audio_processing/vad/pitch_internal.cc",
+    "/third_party/libwebrtc/modules/audio_processing/vad/pole_zero_filter.cc",
+    "/third_party/libwebrtc/modules/audio_processing/vad/standalone_vad.cc",
+    "/third_party/libwebrtc/modules/audio_processing/vad/vad_audio_proc.cc",
+    "/third_party/libwebrtc/modules/audio_processing/vad/vad_circular_buffer.cc",
+    "/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("vad_gn")
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector.cc b/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector.cc
new file mode 100644
index 0000000000..02023d6a72
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector.cc
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/voice_activity_detector.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+const size_t kNumChannels = 1;
+
+const double kDefaultVoiceValue = 1.0;
+const double kNeutralProbability = 0.5;
+const double kLowProbability = 0.01;
+
+}  // namespace
+
+VoiceActivityDetector::VoiceActivityDetector()
+    : last_voice_probability_(kDefaultVoiceValue),
+      standalone_vad_(StandaloneVad::Create()) {}
+
+VoiceActivityDetector::~VoiceActivityDetector() = default;
+
+// Because ISAC has a different chunk length, it updates
+// `chunkwise_voice_probabilities_` and `chunkwise_rms_` when there is new data.
+// Otherwise it clears them.
+void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
+                                         size_t length,
+                                         int sample_rate_hz) {
+  RTC_DCHECK_EQ(length, sample_rate_hz / 100);
+  // TODO(bugs.webrtc.org/7494): Remove resampling and force 16 kHz audio.
+  // Resample to the required rate.
+  const int16_t* resampled_ptr = audio;
+  if (sample_rate_hz != kSampleRateHz) {
+    RTC_CHECK_EQ(
+        resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
+        0);
+    resampler_.Push(audio, length, resampled_, kLength10Ms, length);
+    resampled_ptr = resampled_;
+  }
+  RTC_DCHECK_EQ(length, kLength10Ms);
+
+  // Each chunk needs to be passed into `standalone_vad_`, because internally it
+  // buffers the audio and processes it all at once when GetActivity() is
+  // called.
+  RTC_CHECK_EQ(standalone_vad_->AddAudio(resampled_ptr, length), 0);
+
+  audio_processing_.ExtractFeatures(resampled_ptr, length, &features_);
+
+  chunkwise_voice_probabilities_.resize(features_.num_frames);
+  chunkwise_rms_.resize(features_.num_frames);
+  std::copy(features_.rms, features_.rms + chunkwise_rms_.size(),
+            chunkwise_rms_.begin());
+  if (features_.num_frames > 0) {
+    if (features_.silence) {
+      // The other features are invalid, so set the voice probabilities to an
+      // arbitrary low value.
+      std::fill(chunkwise_voice_probabilities_.begin(),
+                chunkwise_voice_probabilities_.end(), kLowProbability);
+    } else {
+      std::fill(chunkwise_voice_probabilities_.begin(),
+                chunkwise_voice_probabilities_.end(), kNeutralProbability);
+      RTC_CHECK_GE(
+          standalone_vad_->GetActivity(&chunkwise_voice_probabilities_[0],
+                                       chunkwise_voice_probabilities_.size()),
+          0);
+      RTC_CHECK_GE(pitch_based_vad_.VoicingProbability(
+                       features_, &chunkwise_voice_probabilities_[0]),
+                   0);
+    }
+    last_voice_probability_ = chunkwise_voice_probabilities_.back();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector.h b/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector.h
new file mode 100644
index 0000000000..92b9a8c208
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "common_audio/resampler/include/resampler.h"
+#include "modules/audio_processing/vad/common.h"
+#include "modules/audio_processing/vad/pitch_based_vad.h"
+#include "modules/audio_processing/vad/standalone_vad.h"
+#include "modules/audio_processing/vad/vad_audio_proc.h"
+
+namespace webrtc {
+
+// A Voice Activity Detector (VAD) that combines the voice probability from the
+// StandaloneVad and PitchBasedVad to get a more robust estimation.
+class VoiceActivityDetector {
+ public:
+  VoiceActivityDetector();
+  ~VoiceActivityDetector();
+
+  // Processes each audio chunk and estimates the voice probability.
+  // TODO(bugs.webrtc.org/7494): Switch to rtc::ArrayView and remove
+  // `sample_rate_hz`.
+  void ProcessChunk(const int16_t* audio, size_t length, int sample_rate_hz);
+
+  // Returns a vector of voice probabilities for each chunk. It can be empty for
+  // some chunks, but it catches up afterwards returning multiple values at
+  // once.
+  const std::vector<double>& chunkwise_voice_probabilities() const {
+    return chunkwise_voice_probabilities_;
+  }
+
+  // Returns a vector of RMS values for each chunk. It has the same length as
+  // chunkwise_voice_probabilities().
+  const std::vector<double>& chunkwise_rms() const { return chunkwise_rms_; }
+
+  // Returns the last voice probability, regardless of the internal
+  // implementation, although it has a few chunks of delay.
+  float last_voice_probability() const { return last_voice_probability_; }
+
+ private:
+  // TODO(aluebs): Change these to float.
+  std::vector<double> chunkwise_voice_probabilities_;
+  std::vector<double> chunkwise_rms_;
+
+  float last_voice_probability_;
+
+  Resampler resampler_;
+  VadAudioProc audio_processing_;
+
+  std::unique_ptr<StandaloneVad> standalone_vad_;
+  PitchBasedVad pitch_based_vad_;
+
+  int16_t resampled_[kLength10Ms];
+  AudioFeatures features_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector_unittest.cc b/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector_unittest.cc
new file mode 100644
index 0000000000..80f21c8db0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/voice_activity_detector_unittest.cc
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/vad/voice_activity_detector.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace {
+
+const int kStartTimeSec = 16;
+const float kMeanSpeechProbability = 0.3f;
+const float kMaxNoiseProbability = 0.1f;
+const size_t kNumChunks = 300u;
+const size_t kNumChunksPerIsacBlock = 3;
+
+void GenerateNoise(std::vector<int16_t>* data) {
+  for (size_t i = 0; i < data->size(); ++i) {
+    // std::rand returns between 0 and RAND_MAX, but this will work because it
+    // wraps into some random place.
+    (*data)[i] = std::rand();
+  }
+}
+
+}  // namespace
+
+TEST(VoiceActivityDetectorTest, ConstructorSetsDefaultValues) {
+  const float kDefaultVoiceValue = 1.f;
+
+  VoiceActivityDetector vad;
+
+  std::vector<double> p = vad.chunkwise_voice_probabilities();
+  std::vector<double> rms = vad.chunkwise_rms();
+
+  EXPECT_EQ(p.size(), 0u);
+  EXPECT_EQ(rms.size(), 0u);
+
+  EXPECT_FLOAT_EQ(vad.last_voice_probability(), kDefaultVoiceValue);
+}
+
+TEST(VoiceActivityDetectorTest, Speech16kHzHasHighVoiceProbabilities) {
+  const int kSampleRateHz = 16000;
+  const int kLength10Ms = kSampleRateHz / 100;
+
+  VoiceActivityDetector vad;
+
+  std::vector<int16_t> data(kLength10Ms);
+  float mean_probability = 0.f;
+
+  FILE* pcm_file =
+      fopen(test::ResourcePath("audio_processing/transient/audio16kHz", "pcm")
+                .c_str(),
+            "rb");
+  ASSERT_TRUE(pcm_file != nullptr);
+  // The silences in the file are skipped to get a more robust voice probability
+  // for speech.
+  ASSERT_EQ(fseek(pcm_file, kStartTimeSec * kSampleRateHz * sizeof(data[0]),
+                  SEEK_SET),
+            0);
+
+  size_t num_chunks = 0;
+  while (fread(&data[0], sizeof(data[0]), data.size(), pcm_file) ==
+         data.size()) {
+    vad.ProcessChunk(&data[0], data.size(), kSampleRateHz);
+
+    mean_probability += vad.last_voice_probability();
+
+    ++num_chunks;
+  }
+
+  mean_probability /= num_chunks;
+
+  EXPECT_GT(mean_probability, kMeanSpeechProbability);
+}
+
+TEST(VoiceActivityDetectorTest, Speech32kHzHasHighVoiceProbabilities) {
+  const int kSampleRateHz = 32000;
+  const int kLength10Ms = kSampleRateHz / 100;
+
+  VoiceActivityDetector vad;
+
+  std::vector<int16_t> data(kLength10Ms);
+  float mean_probability = 0.f;
+
+  FILE* pcm_file =
+      fopen(test::ResourcePath("audio_processing/transient/audio32kHz", "pcm")
+                .c_str(),
+            "rb");
+  ASSERT_TRUE(pcm_file != nullptr);
+  // The silences in the file are skipped to get a more robust voice probability
+  // for speech.
+  ASSERT_EQ(fseek(pcm_file, kStartTimeSec * kSampleRateHz * sizeof(data[0]),
+                  SEEK_SET),
+            0);
+
+  size_t num_chunks = 0;
+  while (fread(&data[0], sizeof(data[0]), data.size(), pcm_file) ==
+         data.size()) {
+    vad.ProcessChunk(&data[0], data.size(), kSampleRateHz);
+
+    mean_probability += vad.last_voice_probability();
+
+    ++num_chunks;
+  }
+
+  mean_probability /= num_chunks;
+
+  EXPECT_GT(mean_probability, kMeanSpeechProbability);
+}
+
+TEST(VoiceActivityDetectorTest, Noise16kHzHasLowVoiceProbabilities) {
+  VoiceActivityDetector vad;
+
+  std::vector<int16_t> data(kLength10Ms);
+  float max_probability = 0.f;
+
+  std::srand(42);
+
+  for (size_t i = 0; i < kNumChunks; ++i) {
+    GenerateNoise(&data);
+
+    vad.ProcessChunk(&data[0], data.size(), kSampleRateHz);
+
+    // Before the |vad has enough data to process an ISAC block it will return
+    // the default value, 1.f, which would ruin the `max_probability` value.
+    if (i > kNumChunksPerIsacBlock) {
+      max_probability = std::max(max_probability, vad.last_voice_probability());
+    }
+  }
+
+  EXPECT_LT(max_probability, kMaxNoiseProbability);
+}
+
+TEST(VoiceActivityDetectorTest, Noise32kHzHasLowVoiceProbabilities) {
+  VoiceActivityDetector vad;
+
+  std::vector<int16_t> data(2 * kLength10Ms);
+  float max_probability = 0.f;
+
+  std::srand(42);
+
+  for (size_t i = 0; i < kNumChunks; ++i) {
+    GenerateNoise(&data);
+
+    vad.ProcessChunk(&data[0], data.size(), 2 * kSampleRateHz);
+
+    // Before the |vad has enough data to process an ISAC block it will return
+    // the default value, 1.f, which would ruin the `max_probability` value.
+    if (i > kNumChunksPerIsacBlock) {
+      max_probability = std::max(max_probability, vad.last_voice_probability());
+    }
+  }
+
+  EXPECT_LT(max_probability, kMaxNoiseProbability);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/vad/voice_gmm_tables.h b/third_party/libwebrtc/modules/audio_processing/vad/voice_gmm_tables.h
new file mode 100644
index 0000000000..ef4ad7e21e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/vad/voice_gmm_tables.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// GMM tables for active segments. Generated by MakeGmmTables.m.
+
+#ifndef MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_
+#define MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_
+
+static const int kVoiceGmmNumMixtures = 12;
+static const int kVoiceGmmDim = 3;
+
+static const double
+    kVoiceGmmCovarInverse[kVoiceGmmNumMixtures][kVoiceGmmDim][kVoiceGmmDim] = {
+        {{1.83673825579513e+00, -8.09791637570095e-04, 4.60106414365986e-03},
+         {-8.09791637570095e-04, 8.89351738394608e-04, -9.80188953277734e-04},
+         {4.60106414365986e-03, -9.80188953277734e-04, 1.38706060206582e-03}},
+        {{6.76228912850703e+01, -1.98893120119660e-02, -3.53548357253551e-03},
+         {-1.98893120119660e-02, 3.96216858500530e-05, -4.08492938394097e-05},
+         {-3.53548357253551e-03, -4.08492938394097e-05, 9.31864352856416e-04}},
+        {{9.98612435944558e+00, -5.27880954316893e-03, -6.30342541619017e-03},
+         {-5.27880954316893e-03, 4.54359480225226e-05, 6.30804591626044e-05},
+         {-6.30342541619017e-03, 6.30804591626044e-05, 5.36466441382942e-04}},
+        {{3.39917474216349e+01, -1.56213579433191e-03, -4.01459014990225e-02},
+         {-1.56213579433191e-03, 6.40415424897724e-05, 6.20076342427833e-05},
+         {-4.01459014990225e-02, 6.20076342427833e-05, 3.51199070103063e-03}},
+        {{1.34545062271428e+01, -7.94513610147144e-03, -5.34401019341728e-02},
+         {-7.94513610147144e-03, 1.16511820098649e-04, 4.66063702069293e-05},
+         {-5.34401019341728e-02, 4.66063702069293e-05, 2.72354323774163e-03}},
+        {{1.08557844314806e+02, -1.54885805673668e-02, -1.88029692674851e-02},
+         {-1.54885805673668e-02, 1.16404042786406e-04, 6.45579292702802e-06},
+         {-1.88029692674851e-02, 6.45579292702802e-06, 4.32330478391416e-04}},
+        {{8.22940066541450e+01, -1.15903110231303e-02, -4.92166764865343e-02},
+         {-1.15903110231303e-02, 7.42510742165261e-05, 3.73007314191290e-06},
+         {-4.92166764865343e-02, 3.73007314191290e-06, 3.64005221593244e-03}},
+        {{2.31133605685660e+00, -7.83261568950254e-04, 7.45744012346313e-04},
+         {-7.83261568950254e-04, 1.29460648214142e-05, -2.22774455093730e-06},
+         {7.45744012346313e-04, -2.22774455093730e-06, 1.05117294093010e-04}},
+        {{3.78767849189611e+02, 1.57759761011568e-03, -2.08551217988774e-02},
+         {1.57759761011568e-03, 4.76066236886865e-05, -2.33977412299324e-05},
+         {-2.08551217988774e-02, -2.33977412299324e-05, 5.24261005371196e-04}},
+        {{6.98580096506135e-01, -5.13850255217378e-04, -4.01124551717056e-04},
+         {-5.13850255217378e-04, 1.40501021984840e-06, -2.09496928716569e-06},
+         {-4.01124551717056e-04, -2.09496928716569e-06, 2.82879357740037e-04}},
+        {{2.62770945162399e+00, -2.31825753241430e-03, -5.30447217466318e-03},
+         {-2.31825753241430e-03, 4.59108572227649e-05, 7.67631886355405e-05},
+         {-5.30447217466318e-03, 7.67631886355405e-05, 2.28521601674098e-03}},
+        {{1.89940391362152e+02, -4.23280856852379e-03, -2.70608873541399e-02},
+         {-4.23280856852379e-03, 6.77547582742563e-05, 2.69154203800467e-05},
+         {-2.70608873541399e-02, 2.69154203800467e-05, 3.88574543373470e-03}}};
+
+static const double kVoiceGmmMean[kVoiceGmmNumMixtures][kVoiceGmmDim] = {
+    {-2.15020241646536e+00, 4.97079062999877e+02, 4.77078119504505e+02},
+    {-8.92097680029190e-01, 5.92064964199921e+02, 1.81045145941059e+02},
+    {-1.29435784144398e+00, 4.98450293410611e+02, 1.71991263804064e+02},
+    {-1.03925228397884e+00, 4.99511274321571e+02, 1.05838336539105e+02},
+    {-1.29229047206129e+00, 4.15026762566707e+02, 1.12861119017125e+02},
+    {-7.88748114599810e-01, 4.48739336688113e+02, 1.89784216956337e+02},
+    {-8.77777402332642e-01, 4.86620285054533e+02, 1.13477708016491e+02},
+    {-2.06465957063057e+00, 6.33385049870607e+02, 2.32758546796149e+02},
+    {-6.98893789231685e-01, 5.93622051503385e+02, 1.92536982473203e+02},
+    {-2.55901217508894e+00, 1.55914919756205e+03, 1.39769980835570e+02},
+    {-1.92070024165837e+00, 4.87983940444185e+02, 1.02745468128289e+02},
+    {-7.29187507662854e-01, 5.22717685022855e+02, 1.16377942283991e+02}};
+
+static const double kVoiceGmmWeights[kVoiceGmmNumMixtures] = {
+    -1.39789694361035e+01, -1.19527720202104e+01, -1.32396317929055e+01,
+    -1.09436815209238e+01, -1.13440027478149e+01, -1.12200721834504e+01,
+    -1.02537324043693e+01, -1.60789861938302e+01, -1.03394494048344e+01,
+    -1.83207938586818e+01, -1.31186044948288e+01, -9.52479998673554e+00};
+#endif  // MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/BUILD.gn b/third_party/libwebrtc/modules/congestion_controller/BUILD.gn
new file mode 100644
index 0000000000..33f5508137
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/BUILD.gn
@@ -0,0 +1,68 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+config("bwe_test_logging") {
+  if (rtc_enable_bwe_test_logging) {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
+  } else {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
+  }
+}
+
+rtc_library("congestion_controller") {
+  visibility = [ "*" ]
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "include/receive_side_congestion_controller.h",
+    "receive_side_congestion_controller.cc",
+    "remb_throttler.cc",
+    "remb_throttler.h",
+  ]
+
+  deps = [
+    "../../api/transport:network_control",
+    "../../api/units:data_rate",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base/synchronization:mutex",
+    "../pacing",
+    "../remote_bitrate_estimator",
+    "../rtp_rtcp:rtp_rtcp_format",
+  ]
+}
+
+if (rtc_include_tests && !build_with_chromium) {
+  rtc_library("congestion_controller_unittests") {
+    testonly = true
+
+    sources = [
+      "receive_side_congestion_controller_unittest.cc",
+      "remb_throttler_unittest.cc",
+    ]
+    deps = [
+      ":congestion_controller",
+      "../../api/test/network_emulation",
+      "../../api/test/network_emulation:create_cross_traffic",
+      "../../api/units:data_rate",
+      "../../api/units:time_delta",
+      "../../api/units:timestamp",
+      "../../system_wrappers",
+      "../../test:test_support",
+      "../../test/scenario",
+      "../pacing",
+      "goog_cc:estimators",
+      "goog_cc:goog_cc_unittests",
+      "pcc:pcc_unittests",
+      "rtp:congestion_controller_unittests",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/congestion_controller/DEPS b/third_party/libwebrtc/modules/congestion_controller/DEPS
new file mode 100644
index 0000000000..2ed9952e22
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+  "+logging/rtc_event_log",
+  "+system_wrappers",
+  "+video",
+]
diff --git a/third_party/libwebrtc/modules/congestion_controller/OWNERS b/third_party/libwebrtc/modules/congestion_controller/OWNERS
new file mode 100644
index 0000000000..9a836bad06
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/OWNERS
@@ -0,0 +1,7 @@
+danilchap@webrtc.org
+linderborg@webrtc.org
+stefan@webrtc.org
+terelius@webrtc.org
+mflodman@webrtc.org
+yinwa@webrtc.org
+perkj@webrtc.org
diff --git a/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build
new file mode 100644
index 0000000000..9633f1ef48
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build
@@ -0,0 +1,234 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/remb_throttler.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("congestion_controller_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/BUILD.gn b/third_party/libwebrtc/modules/congestion_controller/goog_cc/BUILD.gn
new file mode 100644
index 0000000000..150201e1bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/BUILD.gn
@@ -0,0 +1,369 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+config("bwe_test_logging") {
+  if (rtc_enable_bwe_test_logging) {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
+  } else {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
+  }
+}
+
+rtc_library("goog_cc") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "goog_cc_network_control.cc",
+    "goog_cc_network_control.h",
+  ]
+
+  deps = [
+    ":alr_detector",
+    ":delay_based_bwe",
+    ":estimators",
+    ":loss_based_bwe_v2",
+    ":probe_controller",
+    ":pushback_controller",
+    ":send_side_bwe",
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:field_trial_based_config",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../logging:rtc_event_pacing",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base/experiments:alr_experiment",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../rtc_base/experiments:rate_control_settings",
+    "../../../system_wrappers",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("link_capacity_estimator") {
+  sources = [
+    "link_capacity_estimator.cc",
+    "link_capacity_estimator.h",
+  ]
+  deps = [
+    "../../../api/units:data_rate",
+    "../../../rtc_base:safe_minmax",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("pushback_controller") {
+  sources = [
+    "congestion_window_pushback_controller.cc",
+    "congestion_window_pushback_controller.h",
+  ]
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_size",
+    "../../../rtc_base:checks",
+    "../../../rtc_base/experiments:rate_control_settings",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("alr_detector") {
+  sources = [
+    "alr_detector.cc",
+    "alr_detector.h",
+  ]
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:field_trial_based_config",
+    "../../../logging:rtc_event_pacing",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:timeutils",
+    "../../../rtc_base/experiments:alr_experiment",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../pacing:interval_budget",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+rtc_library("estimators") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "acknowledged_bitrate_estimator.cc",
+    "acknowledged_bitrate_estimator.h",
+    "acknowledged_bitrate_estimator_interface.cc",
+    "acknowledged_bitrate_estimator_interface.h",
+    "bitrate_estimator.cc",
+    "bitrate_estimator.h",
+    "delay_increase_detector_interface.h",
+    "probe_bitrate_estimator.cc",
+    "probe_bitrate_estimator.h",
+    "robust_throughput_estimator.cc",
+    "robust_throughput_estimator.h",
+    "trendline_estimator.cc",
+    "trendline_estimator.h",
+  ]
+
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:macromagic",
+    "../../../rtc_base:rtc_numerics",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("loss_based_bwe_v2") {
+  sources = [
+    "loss_based_bwe_v2.cc",
+    "loss_based_bwe_v2.h",
+  ]
+  deps = [
+    "../../../api:array_view",
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:logging",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("loss_based_bwe_v1") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "loss_based_bandwidth_estimation.cc",
+    "loss_based_bandwidth_estimation.h",
+  ]
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:checks",
+    "../../../rtc_base/experiments:field_trial_parser",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_library("send_side_bwe") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "send_side_bandwidth_estimation.cc",
+    "send_side_bandwidth_estimation.h",
+  ]
+  deps = [
+    ":loss_based_bwe_v1",
+    ":loss_based_bwe_v2",
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../system_wrappers:field_trial",
+    "../../../system_wrappers:metrics",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("delay_based_bwe") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "delay_based_bwe.cc",
+    "delay_based_bwe.h",
+    "inter_arrival_delta.cc",
+    "inter_arrival_delta.h",
+  ]
+
+  deps = [
+    ":estimators",
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:network_control",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:race_checker",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../system_wrappers:metrics",
+    "../../pacing",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("probe_controller") {
+  sources = [
+    "probe_controller.cc",
+    "probe_controller.h",
+  ]
+
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../logging:rtc_event_pacing",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:macromagic",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../system_wrappers:metrics",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("test_goog_cc_printer") {
+    testonly = true
+    sources = [
+      "test/goog_cc_printer.cc",
+      "test/goog_cc_printer.h",
+    ]
+    deps = [
+      ":alr_detector",
+      ":delay_based_bwe",
+      ":estimators",
+      ":goog_cc",
+      "../../../api/rtc_event_log",
+      "../../../api/transport:goog_cc",
+      "../../../api/transport:network_control",
+      "../../../api/units:timestamp",
+      "../../../rtc_base:checks",
+      "../../../test/logging:log_writer",
+      "../../remote_bitrate_estimator",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+  if (!build_with_chromium) {
+    rtc_library("goog_cc_unittests") {
+      testonly = true
+
+      sources = [
+        "acknowledged_bitrate_estimator_unittest.cc",
+        "alr_detector_unittest.cc",
+        "congestion_window_pushback_controller_unittest.cc",
+        "delay_based_bwe_unittest.cc",
+        "delay_based_bwe_unittest_helper.cc",
+        "delay_based_bwe_unittest_helper.h",
+        "goog_cc_network_control_unittest.cc",
+        "loss_based_bwe_v2_test.cc",
+        "probe_bitrate_estimator_unittest.cc",
+        "probe_controller_unittest.cc",
+        "robust_throughput_estimator_unittest.cc",
+        "send_side_bandwidth_estimation_unittest.cc",
+        "trendline_estimator_unittest.cc",
+      ]
+      deps = [
+        ":alr_detector",
+        ":delay_based_bwe",
+        ":estimators",
+        ":goog_cc",
+        ":loss_based_bwe_v2",
+        ":probe_controller",
+        ":pushback_controller",
+        ":send_side_bwe",
+        "../../../api:field_trials_view",
+        "../../../api:network_state_predictor_api",
+        "../../../api/rtc_event_log",
+        "../../../api/test/network_emulation",
+        "../../../api/test/network_emulation:create_cross_traffic",
+        "../../../api/transport:field_trial_based_config",
+        "../../../api/transport:goog_cc",
+        "../../../api/transport:network_control",
+        "../../../api/units:data_rate",
+        "../../../api/units:data_size",
+        "../../../api/units:time_delta",
+        "../../../api/units:timestamp",
+        "../../../logging:mocks",
+        "../../../logging:rtc_event_bwe",
+        "../../../rtc_base:checks",
+        "../../../rtc_base:logging",
+        "../../../rtc_base:random",
+        "../../../rtc_base:rtc_base_tests_utils",
+        "../../../rtc_base:stringutils",
+        "../../../rtc_base/experiments:alr_experiment",
+        "../../../system_wrappers",
+        "../../../test:explicit_key_value_config",
+        "../../../test:field_trial",
+        "../../../test:test_support",
+        "../../../test/scenario",
+        "../../pacing",
+        "//testing/gmock",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
new file mode 100644
index 0000000000..08b42a8168
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
+    const FieldTrialsView* key_value_config)
+    : AcknowledgedBitrateEstimator(
+          key_value_config,
+          std::make_unique<BitrateEstimator>(key_value_config)) {}
+
+AcknowledgedBitrateEstimator::~AcknowledgedBitrateEstimator() {}
+
+AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
+    const FieldTrialsView* key_value_config,
+    std::unique_ptr<BitrateEstimator> bitrate_estimator)
+    : in_alr_(false), bitrate_estimator_(std::move(bitrate_estimator)) {}
+
+void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
+    const std::vector<PacketResult>& packet_feedback_vector) {
+  RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
+                            packet_feedback_vector.end(),
+                            PacketResult::ReceiveTimeOrder()));
+  for (const auto& packet : packet_feedback_vector) {
+    if (alr_ended_time_ && packet.sent_packet.send_time > *alr_ended_time_) {
+      bitrate_estimator_->ExpectFastRateChange();
+      alr_ended_time_.reset();
+    }
+    DataSize acknowledged_estimate = packet.sent_packet.size;
+    acknowledged_estimate += packet.sent_packet.prior_unacked_data;
+    bitrate_estimator_->Update(packet.receive_time, acknowledged_estimate,
+                               in_alr_);
+  }
+}
+
+absl::optional<DataRate> AcknowledgedBitrateEstimator::bitrate() const {
+  return bitrate_estimator_->bitrate();
+}
+
+absl::optional<DataRate> AcknowledgedBitrateEstimator::PeekRate() const {
+  return bitrate_estimator_->PeekRate();
+}
+
+void AcknowledgedBitrateEstimator::SetAlrEndedTime(Timestamp alr_ended_time) {
+  alr_ended_time_.emplace(alr_ended_time);
+}
+
+void AcknowledgedBitrateEstimator::SetAlr(bool in_alr) {
+  in_alr_ = in_alr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
new file mode 100644
index 0000000000..d10846ab3a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
+
+namespace webrtc {
+
+class AcknowledgedBitrateEstimator
+    : public AcknowledgedBitrateEstimatorInterface {
+ public:
+  AcknowledgedBitrateEstimator(
+      const FieldTrialsView* key_value_config,
+      std::unique_ptr<BitrateEstimator> bitrate_estimator);
+
+  explicit AcknowledgedBitrateEstimator(
+      const FieldTrialsView* key_value_config);
+  ~AcknowledgedBitrateEstimator() override;
+
+  void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) override;
+  absl::optional<DataRate> bitrate() const override;
+  absl::optional<DataRate> PeekRate() const override;
+  void SetAlr(bool in_alr) override;
+  void SetAlrEndedTime(Timestamp alr_ended_time) override;
+
+ private:
+  absl::optional<Timestamp> alr_ended_time_;
+  bool in_alr_;
+  std::unique_ptr<BitrateEstimator> bitrate_estimator_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.cc
new file mode 100644
index 0000000000..c043353a7a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.cc
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+
+#include <algorithm>
+
+#include "api/units/time_delta.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/robust_throughput_estimator.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+constexpr char RobustThroughputEstimatorSettings::kKey[];
+
+RobustThroughputEstimatorSettings::RobustThroughputEstimatorSettings(
+    const FieldTrialsView* key_value_config) {
+  Parser()->Parse(
+      key_value_config->Lookup(RobustThroughputEstimatorSettings::kKey));
+  if (window_packets < 10 || 1000 < window_packets) {
+    RTC_LOG(LS_WARNING) << "Window size must be between 10 and 1000 packets";
+    window_packets = 20;
+  }
+  if (max_window_packets < 10 || 1000 < max_window_packets) {
+    RTC_LOG(LS_WARNING)
+        << "Max window size must be between 10 and 1000 packets";
+    max_window_packets = 500;
+  }
+  max_window_packets = std::max(max_window_packets, window_packets);
+
+  if (required_packets < 10 || 1000 < required_packets) {
+    RTC_LOG(LS_WARNING) << "Required number of initial packets must be between "
+                           "10 and 1000 packets";
+    required_packets = 10;
+  }
+  required_packets = std::min(required_packets, window_packets);
+
+  if (min_window_duration < TimeDelta::Millis(100) ||
+      TimeDelta::Millis(3000) < min_window_duration) {
+    RTC_LOG(LS_WARNING) << "Window duration must be between 100 and 3000 ms";
+    min_window_duration = TimeDelta::Millis(750);
+  }
+  if (max_window_duration < TimeDelta::Seconds(1) ||
+      TimeDelta::Seconds(15) < max_window_duration) {
+    RTC_LOG(LS_WARNING) << "Max window duration must be between 1 and 15 s";
+    max_window_duration = TimeDelta::Seconds(5);
+  }
+  min_window_duration = std::min(min_window_duration, max_window_duration);
+
+  if (unacked_weight < 0.0 || 1.0 < unacked_weight) {
+    RTC_LOG(LS_WARNING)
+        << "Weight for prior unacked size must be between 0 and 1.";
+    unacked_weight = 1.0;
+  }
+}
+
+std::unique_ptr<StructParametersParser>
+RobustThroughputEstimatorSettings::Parser() {
+  return StructParametersParser::Create(
+      "enabled", &enabled,                          //
+      "window_packets", &window_packets,            //
+      "max_window_packets", &max_window_packets,    //
+      "window_duration", &min_window_duration,      //
+      "max_window_duration", &max_window_duration,  //
+      "required_packets", &required_packets,        //
+      "unacked_weight", &unacked_weight);
+}
+
+AcknowledgedBitrateEstimatorInterface::
+    ~AcknowledgedBitrateEstimatorInterface() {}
+
+std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+AcknowledgedBitrateEstimatorInterface::Create(
+    const FieldTrialsView* key_value_config) {
+  RobustThroughputEstimatorSettings simplified_estimator_settings(
+      key_value_config);
+  if (simplified_estimator_settings.enabled) {
+    return std::make_unique<RobustThroughputEstimator>(
+        simplified_estimator_settings);
+  }
+  return std::make_unique<AcknowledgedBitrateEstimator>(key_value_config);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h
new file mode 100644
index 0000000000..515af1efc9
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+struct RobustThroughputEstimatorSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-RobustThroughputEstimatorSettings";
+
+  RobustThroughputEstimatorSettings() = delete;
+  explicit RobustThroughputEstimatorSettings(
+      const FieldTrialsView* key_value_config);
+
+  bool enabled = false;  // Set to true to use RobustThroughputEstimator.
+
+  // The estimator keeps the smallest window containing at least
+  // `window_packets` and at least the packets received during the last
+  // `min_window_duration` milliseconds.
+  // (This means that it may store more than `window_packets` at high bitrates,
+  // and a longer duration than `min_window_duration` at low bitrates.)
+  // However, if will never store more than kMaxPackets (for performance
+  // reasons), and never longer than max_window_duration (to avoid very old
+  // packets influencing the estimate for example when sending is paused).
+  unsigned window_packets = 20;
+  unsigned max_window_packets = 500;
+  TimeDelta min_window_duration = TimeDelta::Seconds(1);
+  TimeDelta max_window_duration = TimeDelta::Seconds(5);
+
+  // The estimator window requires at least `required_packets` packets
+  // to produce an estimate.
+  unsigned required_packets = 10;
+
+  // If audio packets aren't included in allocation (i.e. the
+  // estimated available bandwidth is divided only among the video
+  // streams), then `unacked_weight` should be set to 0.
+  // If audio packets are included in allocation, but not in bandwidth
+  // estimation (i.e. they don't have transport-wide sequence numbers,
+  // but we nevertheless divide the estimated available bandwidth among
+  // both audio and video streams), then `unacked_weight` should be set to 1.
+  // If all packets have transport-wide sequence numbers, then the value
+  // of `unacked_weight` doesn't matter.
+  double unacked_weight = 1.0;
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class AcknowledgedBitrateEstimatorInterface {
+ public:
+  static std::unique_ptr<AcknowledgedBitrateEstimatorInterface> Create(
+      const FieldTrialsView* key_value_config);
+  virtual ~AcknowledgedBitrateEstimatorInterface();
+
+  virtual void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) = 0;
+  virtual absl::optional<DataRate> bitrate() const = 0;
+  virtual absl::optional<DataRate> PeekRate() const = 0;
+  virtual void SetAlr(bool in_alr) = 0;
+  virtual void SetAlrEndedTime(Timestamp alr_ended_time) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
new file mode 100644
index 0000000000..e5b733b119
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/transport/field_trial_based_config.h"
+#include "rtc_base/fake_clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::InSequence;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace webrtc {
+
+namespace {
+
+constexpr int64_t kFirstArrivalTimeMs = 10;
+constexpr int64_t kFirstSendTimeMs = 10;
+constexpr uint16_t kSequenceNumber = 1;
+constexpr size_t kPayloadSize = 10;
+
+class MockBitrateEstimator : public BitrateEstimator {
+ public:
+  using BitrateEstimator::BitrateEstimator;
+  MOCK_METHOD(void,
+              Update,
+              (Timestamp at_time, DataSize data_size, bool in_alr),
+              (override));
+  MOCK_METHOD(absl::optional<DataRate>, bitrate, (), (const, override));
+  MOCK_METHOD(void, ExpectFastRateChange, (), (override));
+};
+
+struct AcknowledgedBitrateEstimatorTestStates {
+  FieldTrialBasedConfig field_trial_config;
+  std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator;
+  MockBitrateEstimator* mock_bitrate_estimator;
+};
+
+AcknowledgedBitrateEstimatorTestStates CreateTestStates() {
+  AcknowledgedBitrateEstimatorTestStates states;
+  auto mock_bitrate_estimator =
+      std::make_unique<MockBitrateEstimator>(&states.field_trial_config);
+  states.mock_bitrate_estimator = mock_bitrate_estimator.get();
+  states.acknowledged_bitrate_estimator =
+      std::make_unique<AcknowledgedBitrateEstimator>(
+          &states.field_trial_config, std::move(mock_bitrate_estimator));
+  return states;
+}
+
+std::vector<PacketResult> CreateFeedbackVector() {
+  std::vector<PacketResult> packet_feedback_vector(2);
+  packet_feedback_vector[0].receive_time =
+      Timestamp::Millis(kFirstArrivalTimeMs);
+  packet_feedback_vector[0].sent_packet.send_time =
+      Timestamp::Millis(kFirstSendTimeMs);
+  packet_feedback_vector[0].sent_packet.sequence_number = kSequenceNumber;
+  packet_feedback_vector[0].sent_packet.size = DataSize::Bytes(kPayloadSize);
+  packet_feedback_vector[1].receive_time =
+      Timestamp::Millis(kFirstArrivalTimeMs + 10);
+  packet_feedback_vector[1].sent_packet.send_time =
+      Timestamp::Millis(kFirstSendTimeMs + 10);
+  packet_feedback_vector[1].sent_packet.sequence_number = kSequenceNumber;
+  packet_feedback_vector[1].sent_packet.size =
+      DataSize::Bytes(kPayloadSize + 10);
+  return packet_feedback_vector;
+}
+
+}  // anonymous namespace
+
+TEST(TestAcknowledgedBitrateEstimator, UpdateBandwidth) {
+  auto states = CreateTestStates();
+  auto packet_feedback_vector = CreateFeedbackVector();
+  {
+    InSequence dummy;
+    EXPECT_CALL(*states.mock_bitrate_estimator,
+                Update(packet_feedback_vector[0].receive_time,
+                       packet_feedback_vector[0].sent_packet.size,
+                       /*in_alr*/ false))
+        .Times(1);
+    EXPECT_CALL(*states.mock_bitrate_estimator,
+                Update(packet_feedback_vector[1].receive_time,
+                       packet_feedback_vector[1].sent_packet.size,
+                       /*in_alr*/ false))
+        .Times(1);
+  }
+  states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+      packet_feedback_vector);
+}
+
+TEST(TestAcknowledgedBitrateEstimator, ExpectFastRateChangeWhenLeftAlr) {
+  auto states = CreateTestStates();
+  auto packet_feedback_vector = CreateFeedbackVector();
+  {
+    InSequence dummy;
+    EXPECT_CALL(*states.mock_bitrate_estimator,
+                Update(packet_feedback_vector[0].receive_time,
+                       packet_feedback_vector[0].sent_packet.size,
+                       /*in_alr*/ false))
+        .Times(1);
+    EXPECT_CALL(*states.mock_bitrate_estimator, ExpectFastRateChange())
+        .Times(1);
+    EXPECT_CALL(*states.mock_bitrate_estimator,
+                Update(packet_feedback_vector[1].receive_time,
+                       packet_feedback_vector[1].sent_packet.size,
+                       /*in_alr*/ false))
+        .Times(1);
+  }
+  states.acknowledged_bitrate_estimator->SetAlrEndedTime(
+      Timestamp::Millis(kFirstArrivalTimeMs + 1));
+  states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+      packet_feedback_vector);
+}
+
+TEST(TestAcknowledgedBitrateEstimator, ReturnBitrate) {
+  auto states = CreateTestStates();
+  absl::optional<DataRate> return_value = DataRate::KilobitsPerSec(42);
+  EXPECT_CALL(*states.mock_bitrate_estimator, bitrate())
+      .Times(1)
+      .WillOnce(Return(return_value));
+  EXPECT_EQ(return_value, states.acknowledged_bitrate_estimator->bitrate());
+}
+
+}  // namespace webrtc*/
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.cc
new file mode 100644
index 0000000000..f1e649b7cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+
+#include <cstdint>
+#include <cstdio>
+#include <memory>
+
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+AlrDetectorConfig GetConfigFromTrials(const FieldTrialsView* key_value_config) {
+  RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled(*key_value_config));
+  absl::optional<AlrExperimentSettings> experiment_settings =
+      AlrExperimentSettings::CreateFromFieldTrial(
+          *key_value_config,
+          AlrExperimentSettings::kScreenshareProbingBweExperimentName);
+  if (!experiment_settings) {
+    experiment_settings = AlrExperimentSettings::CreateFromFieldTrial(
+        *key_value_config,
+        AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
+  }
+  AlrDetectorConfig conf;
+  if (experiment_settings) {
+    conf.bandwidth_usage_ratio =
+        experiment_settings->alr_bandwidth_usage_percent / 100.0;
+    conf.start_budget_level_ratio =
+        experiment_settings->alr_start_budget_level_percent / 100.0;
+    conf.stop_budget_level_ratio =
+        experiment_settings->alr_stop_budget_level_percent / 100.0;
+  }
+  conf.Parser()->Parse(
+      key_value_config->Lookup("WebRTC-AlrDetectorParameters"));
+  return conf;
+}
+}  //  namespace
+
+std::unique_ptr<StructParametersParser> AlrDetectorConfig::Parser() {
+  return StructParametersParser::Create(   //
+      "bw_usage", &bandwidth_usage_ratio,  //
+      "start", &start_budget_level_ratio,  //
+      "stop", &stop_budget_level_ratio);
+}
+
+AlrDetector::AlrDetector(AlrDetectorConfig config, RtcEventLog* event_log)
+    : conf_(config), alr_budget_(0, true), event_log_(event_log) {}
+
+AlrDetector::AlrDetector(const FieldTrialsView* key_value_config)
+    : AlrDetector(GetConfigFromTrials(key_value_config), nullptr) {}
+
+AlrDetector::AlrDetector(const FieldTrialsView* key_value_config,
+                         RtcEventLog* event_log)
+    : AlrDetector(GetConfigFromTrials(key_value_config), event_log) {}
+AlrDetector::~AlrDetector() {}
+
+void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t send_time_ms) {
+  if (!last_send_time_ms_.has_value()) {
+    last_send_time_ms_ = send_time_ms;
+    // Since the duration for sending the bytes is unknwon, return without
+    // updating alr state.
+    return;
+  }
+  int64_t delta_time_ms = send_time_ms - *last_send_time_ms_;
+  last_send_time_ms_ = send_time_ms;
+
+  alr_budget_.UseBudget(bytes_sent);
+  alr_budget_.IncreaseBudget(delta_time_ms);
+  bool state_changed = false;
+  if (alr_budget_.budget_ratio() > conf_.start_budget_level_ratio &&
+      !alr_started_time_ms_) {
+    alr_started_time_ms_.emplace(rtc::TimeMillis());
+    state_changed = true;
+  } else if (alr_budget_.budget_ratio() < conf_.stop_budget_level_ratio &&
+             alr_started_time_ms_) {
+    state_changed = true;
+    alr_started_time_ms_.reset();
+  }
+  if (event_log_ && state_changed) {
+    event_log_->Log(
+        std::make_unique<RtcEventAlrState>(alr_started_time_ms_.has_value()));
+  }
+}
+
+void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
+  RTC_DCHECK(bitrate_bps);
+  int target_rate_kbps =
+      static_cast<double>(bitrate_bps) * conf_.bandwidth_usage_ratio / 1000;
+  alr_budget_.set_target_rate_kbps(target_rate_kbps);
+}
+
+absl::optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
+    const {
+  return alr_started_time_ms_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.h
new file mode 100644
index 0000000000..5e7a3e1075
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "modules/pacing/interval_budget.h"
+#include "rtc_base/experiments/alr_experiment.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+struct AlrDetectorConfig {
+  // Sent traffic ratio as a function of network capacity used to determine
+  // application-limited region. ALR region start when bandwidth usage drops
+  // below kAlrStartUsageRatio and ends when it raises above
+  // kAlrEndUsageRatio. NOTE: This is intentionally conservative at the moment
+  // until BW adjustments of application limited region is fine tuned.
+  double bandwidth_usage_ratio = 0.65;
+  double start_budget_level_ratio = 0.80;
+  double stop_budget_level_ratio = 0.50;
+  std::unique_ptr<StructParametersParser> Parser();
+};
+// Application limited region detector is a class that utilizes signals of
+// elapsed time and bytes sent to estimate whether network traffic is
+// currently limited by the application's ability to generate traffic.
+//
+// AlrDetector provides a signal that can be utilized to adjust
+// estimate bandwidth.
+// Note: This class is not thread-safe.
+class AlrDetector {
+ public:
+  AlrDetector(AlrDetectorConfig config, RtcEventLog* event_log);
+  explicit AlrDetector(const FieldTrialsView* key_value_config);
+  AlrDetector(const FieldTrialsView* key_value_config, RtcEventLog* event_log);
+  ~AlrDetector();
+
+  void OnBytesSent(size_t bytes_sent, int64_t send_time_ms);
+
+  // Set current estimated bandwidth.
+  void SetEstimatedBitrate(int bitrate_bps);
+
+  // Returns time in milliseconds when the current application-limited region
+  // started or empty result if the sender is currently not application-limited.
+  absl::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
+
+ private:
+  friend class GoogCcStatePrinter;
+  const AlrDetectorConfig conf_;
+
+  absl::optional<int64_t> last_send_time_ms_;
+
+  IntervalBudget alr_budget_;
+  absl::optional<int64_t> alr_started_time_ms_;
+
+  RtcEventLog* event_log_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build
new file mode 100644
index 0000000000..f6520bf358
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("alr_detector_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
new file mode 100644
index 0000000000..eac19d0081
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
@@ -0,0 +1,206 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+
+#include "api/transport/field_trial_based_config.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/alr_experiment.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace {
+
+constexpr int kEstimatedBitrateBps = 300000;
+
+}  // namespace
+
+namespace webrtc {
+namespace {
+class SimulateOutgoingTrafficIn {
+ public:
+  explicit SimulateOutgoingTrafficIn(AlrDetector* alr_detector,
+                                     int64_t* timestamp_ms)
+      : alr_detector_(alr_detector), timestamp_ms_(timestamp_ms) {
+    RTC_CHECK(alr_detector_);
+  }
+
+  SimulateOutgoingTrafficIn& ForTimeMs(int time_ms) {
+    interval_ms_ = time_ms;
+    ProduceTraffic();
+    return *this;
+  }
+
+  SimulateOutgoingTrafficIn& AtPercentOfEstimatedBitrate(int usage_percentage) {
+    usage_percentage_.emplace(usage_percentage);
+    ProduceTraffic();
+    return *this;
+  }
+
+ private:
+  void ProduceTraffic() {
+    if (!interval_ms_ || !usage_percentage_)
+      return;
+    const int kTimeStepMs = 10;
+    for (int t = 0; t < *interval_ms_; t += kTimeStepMs) {
+      *timestamp_ms_ += kTimeStepMs;
+      alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
+                                     kTimeStepMs / (8 * 100 * 1000),
+                                 *timestamp_ms_);
+    }
+    int remainder_ms = *interval_ms_ % kTimeStepMs;
+    if (remainder_ms > 0) {
+      *timestamp_ms_ += kTimeStepMs;
+      alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
+                                     remainder_ms / (8 * 100 * 1000),
+                                 *timestamp_ms_);
+    }
+  }
+  AlrDetector* const alr_detector_;
+  int64_t* timestamp_ms_;
+  absl::optional<int> interval_ms_;
+  absl::optional<int> usage_percentage_;
+};
+}  // namespace
+
+TEST(AlrDetectorTest, AlrDetection) {
+  FieldTrialBasedConfig field_trials;
+  int64_t timestamp_ms = 1000;
+  AlrDetector alr_detector(&field_trials);
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps);
+
+  // Start in non-ALR state.
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Stay in non-ALR state when usage is close to 100%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(90);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Verify that we ALR starts when bitrate drops below 20%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1500)
+      .AtPercentOfEstimatedBitrate(20);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Verify that ALR ends when usage is above 65%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(4000)
+      .AtPercentOfEstimatedBitrate(100);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+}
+
+TEST(AlrDetectorTest, ShortSpike) {
+  FieldTrialBasedConfig field_trials;
+  int64_t timestamp_ms = 1000;
+  AlrDetector alr_detector(&field_trials);
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps);
+  // Start in non-ALR state.
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Verify that we ALR starts when bitrate drops below 20%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(20);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Verify that we stay in ALR region even after a short bitrate spike.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(100)
+      .AtPercentOfEstimatedBitrate(150);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // ALR ends when usage is above 65%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(3000)
+      .AtPercentOfEstimatedBitrate(100);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+}
+
+TEST(AlrDetectorTest, BandwidthEstimateChanges) {
+  FieldTrialBasedConfig field_trials;
+  int64_t timestamp_ms = 1000;
+  AlrDetector alr_detector(&field_trials);
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps);
+
+  // Start in non-ALR state.
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // ALR starts when bitrate drops below 20%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(20);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // When bandwidth estimate drops the detector should stay in ALR mode and quit
+  // it shortly afterwards as the sender continues sending the same amount of
+  // traffic. This is necessary to ensure that ProbeController can still react
+  // to the BWE drop by initiating a new probe.
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps / 5);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(50);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+}
+
+TEST(AlrDetectorTest, ParseControlFieldTrial) {
+  webrtc::test::ScopedFieldTrials scoped_field_trial(
+      "WebRTC-ProbingScreenshareBwe/Control/");
+  absl::optional<AlrExperimentSettings> parsed_params =
+      AlrExperimentSettings::CreateFromFieldTrial(
+          FieldTrialBasedConfig(), "WebRTC-ProbingScreenshareBwe");
+  EXPECT_FALSE(static_cast<bool>(parsed_params));
+}
+
+TEST(AlrDetectorTest, ParseActiveFieldTrial) {
+  webrtc::test::ScopedFieldTrials scoped_field_trial(
+      "WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/");
+  absl::optional<AlrExperimentSettings> parsed_params =
+      AlrExperimentSettings::CreateFromFieldTrial(
+          FieldTrialBasedConfig(), "WebRTC-ProbingScreenshareBwe");
+  ASSERT_TRUE(static_cast<bool>(parsed_params));
+  EXPECT_EQ(1.1f, parsed_params->pacing_factor);
+  EXPECT_EQ(2875, parsed_params->max_paced_queue_time);
+  EXPECT_EQ(85, parsed_params->alr_bandwidth_usage_percent);
+  EXPECT_EQ(20, parsed_params->alr_start_budget_level_percent);
+  EXPECT_EQ(-20, parsed_params->alr_stop_budget_level_percent);
+  EXPECT_EQ(1, parsed_params->group_id);
+}
+
+TEST(AlrDetectorTest, ParseAlrSpecificFieldTrial) {
+  webrtc::test::ScopedFieldTrials scoped_field_trial(
+      "WebRTC-AlrDetectorParameters/"
+      "bw_usage:90%,start:0%,stop:-10%/");
+  FieldTrialBasedConfig field_trials;
+  AlrDetector alr_detector(&field_trials);
+  int64_t timestamp_ms = 1000;
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps);
+
+  // Start in non-ALR state.
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // ALR does not start at 100% utilization.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(100);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // ALR does start at 85% utilization.
+  // Overused 10% above so it should take about 2s to reach a budget level of
+  // 0%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(2100)
+      .AtPercentOfEstimatedBitrate(85);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
new file mode 100644
index 0000000000..9c68e48886
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
@@ -0,0 +1,166 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <cmath>
+#include <string>
+
+#include "api/units/data_rate.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kInitialRateWindowMs = 500;
+constexpr int kRateWindowMs = 150;
+constexpr int kMinRateWindowMs = 150;
+constexpr int kMaxRateWindowMs = 1000;
+
+const char kBweThroughputWindowConfig[] = "WebRTC-BweThroughputWindowConfig";
+
+}  // namespace
+
+BitrateEstimator::BitrateEstimator(const FieldTrialsView* key_value_config)
+    : sum_(0),
+      initial_window_ms_("initial_window_ms",
+                         kInitialRateWindowMs,
+                         kMinRateWindowMs,
+                         kMaxRateWindowMs),
+      noninitial_window_ms_("window_ms",
+                            kRateWindowMs,
+                            kMinRateWindowMs,
+                            kMaxRateWindowMs),
+      uncertainty_scale_("scale", 10.0),
+      uncertainty_scale_in_alr_("scale_alr", uncertainty_scale_),
+      small_sample_uncertainty_scale_("scale_small", uncertainty_scale_),
+      small_sample_threshold_("small_thresh", DataSize::Zero()),
+      uncertainty_symmetry_cap_("symmetry_cap", DataRate::Zero()),
+      estimate_floor_("floor", DataRate::Zero()),
+      current_window_ms_(0),
+      prev_time_ms_(-1),
+      bitrate_estimate_kbps_(-1.0f),
+      bitrate_estimate_var_(50.0f) {
+  // E.g WebRTC-BweThroughputWindowConfig/initial_window_ms:350,window_ms:250/
+  ParseFieldTrial(
+      {&initial_window_ms_, &noninitial_window_ms_, &uncertainty_scale_,
+       &uncertainty_scale_in_alr_, &small_sample_uncertainty_scale_,
+       &small_sample_threshold_, &uncertainty_symmetry_cap_, &estimate_floor_},
+      key_value_config->Lookup(kBweThroughputWindowConfig));
+}
+
+BitrateEstimator::~BitrateEstimator() = default;
+
+void BitrateEstimator::Update(Timestamp at_time, DataSize amount, bool in_alr) {
+  int rate_window_ms = noninitial_window_ms_.Get();
+  // We use a larger window at the beginning to get a more stable sample that
+  // we can use to initialize the estimate.
+  if (bitrate_estimate_kbps_ < 0.f)
+    rate_window_ms = initial_window_ms_.Get();
+  bool is_small_sample = false;
+  float bitrate_sample_kbps = UpdateWindow(at_time.ms(), amount.bytes(),
+                                           rate_window_ms, &is_small_sample);
+  if (bitrate_sample_kbps < 0.0f)
+    return;
+  if (bitrate_estimate_kbps_ < 0.0f) {
+    // This is the very first sample we get. Use it to initialize the estimate.
+    bitrate_estimate_kbps_ = bitrate_sample_kbps;
+    return;
+  }
+  // Optionally use higher uncertainty for very small samples to avoid dropping
+  // estimate and for samples obtained in ALR.
+  float scale = uncertainty_scale_;
+  if (is_small_sample && bitrate_sample_kbps < bitrate_estimate_kbps_) {
+    scale = small_sample_uncertainty_scale_;
+  } else if (in_alr && bitrate_sample_kbps < bitrate_estimate_kbps_) {
+    // Optionally use higher uncertainty for samples obtained during ALR.
+    scale = uncertainty_scale_in_alr_;
+  }
+  // Define the sample uncertainty as a function of how far away it is from the
+  // current estimate. With low values of uncertainty_symmetry_cap_ we add more
+  // uncertainty to increases than to decreases. For higher values we approach
+  // symmetry.
+  float sample_uncertainty =
+      scale * std::abs(bitrate_estimate_kbps_ - bitrate_sample_kbps) /
+      (bitrate_estimate_kbps_ +
+       std::min(bitrate_sample_kbps,
+                uncertainty_symmetry_cap_.Get().kbps<float>()));
+
+  float sample_var = sample_uncertainty * sample_uncertainty;
+  // Update a bayesian estimate of the rate, weighting it lower if the sample
+  // uncertainty is large.
+  // The bitrate estimate uncertainty is increased with each update to model
+  // that the bitrate changes over time.
+  float pred_bitrate_estimate_var = bitrate_estimate_var_ + 5.f;
+  bitrate_estimate_kbps_ = (sample_var * bitrate_estimate_kbps_ +
+                            pred_bitrate_estimate_var * bitrate_sample_kbps) /
+                           (sample_var + pred_bitrate_estimate_var);
+  bitrate_estimate_kbps_ =
+      std::max(bitrate_estimate_kbps_, estimate_floor_.Get().kbps<float>());
+  bitrate_estimate_var_ = sample_var * pred_bitrate_estimate_var /
+                          (sample_var + pred_bitrate_estimate_var);
+  BWE_TEST_LOGGING_PLOT(1, "acknowledged_bitrate", at_time.ms(),
+                        bitrate_estimate_kbps_ * 1000);
+}
+
+float BitrateEstimator::UpdateWindow(int64_t now_ms,
+                                     int bytes,
+                                     int rate_window_ms,
+                                     bool* is_small_sample) {
+  RTC_DCHECK(is_small_sample != nullptr);
+  // Reset if time moves backwards.
+  if (now_ms < prev_time_ms_) {
+    prev_time_ms_ = -1;
+    sum_ = 0;
+    current_window_ms_ = 0;
+  }
+  if (prev_time_ms_ >= 0) {
+    current_window_ms_ += now_ms - prev_time_ms_;
+    // Reset if nothing has been received for more than a full window.
+    if (now_ms - prev_time_ms_ > rate_window_ms) {
+      sum_ = 0;
+      current_window_ms_ %= rate_window_ms;
+    }
+  }
+  prev_time_ms_ = now_ms;
+  float bitrate_sample = -1.0f;
+  if (current_window_ms_ >= rate_window_ms) {
+    *is_small_sample = sum_ < small_sample_threshold_->bytes();
+    bitrate_sample = 8.0f * sum_ / static_cast<float>(rate_window_ms);
+    current_window_ms_ -= rate_window_ms;
+    sum_ = 0;
+  }
+  sum_ += bytes;
+  return bitrate_sample;
+}
+
+absl::optional<DataRate> BitrateEstimator::bitrate() const {
+  if (bitrate_estimate_kbps_ < 0.f)
+    return absl::nullopt;
+  return DataRate::KilobitsPerSec(bitrate_estimate_kbps_);
+}
+
+absl::optional<DataRate> BitrateEstimator::PeekRate() const {
+  if (current_window_ms_ > 0)
+    return DataSize::Bytes(sum_) / TimeDelta::Millis(current_window_ms_);
+  return absl::nullopt;
+}
+
+void BitrateEstimator::ExpectFastRateChange() {
+  // By setting the bitrate-estimate variance to a higher value we allow the
+  // bitrate to change fast for the next few samples.
+  bitrate_estimate_var_ += 200;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
new file mode 100644
index 0000000000..a6f985800e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+// Computes a bayesian estimate of the throughput given acks containing
+// the arrival time and payload size. Samples which are far from the current
+// estimate or are based on few packets are given a smaller weight, as they
+// are considered to be more likely to have been caused by, e.g., delay spikes
+// unrelated to congestion.
+class BitrateEstimator {
+ public:
+  explicit BitrateEstimator(const FieldTrialsView* key_value_config);
+  virtual ~BitrateEstimator();
+  virtual void Update(Timestamp at_time, DataSize amount, bool in_alr);
+
+  virtual absl::optional<DataRate> bitrate() const;
+  absl::optional<DataRate> PeekRate() const;
+
+  virtual void ExpectFastRateChange();
+
+ private:
+  float UpdateWindow(int64_t now_ms,
+                     int bytes,
+                     int rate_window_ms,
+                     bool* is_small_sample);
+  int sum_;
+  FieldTrialConstrained<int> initial_window_ms_;
+  FieldTrialConstrained<int> noninitial_window_ms_;
+  FieldTrialParameter<double> uncertainty_scale_;
+  FieldTrialParameter<double> uncertainty_scale_in_alr_;
+  FieldTrialParameter<double> small_sample_uncertainty_scale_;
+  FieldTrialParameter<DataSize> small_sample_threshold_;
+  FieldTrialParameter<DataRate> uncertainty_symmetry_cap_;
+  FieldTrialParameter<DataRate> estimate_floor_;
+  int64_t current_window_ms_;
+  int64_t prev_time_ms_;
+  float bitrate_estimate_kbps_;
+  float bitrate_estimate_var_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc
new file mode 100644
index 0000000000..2f188f30ca
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
+
+#include <inttypes.h>
+#include <stdio.h>
+
+#include <algorithm>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+
+namespace webrtc {
+
+CongestionWindowPushbackController::CongestionWindowPushbackController(
+    const FieldTrialsView* key_value_config)
+    : add_pacing_(
+          absl::StartsWith(key_value_config->Lookup(
+                               "WebRTC-AddPacingToCongestionWindowPushback"),
+                           "Enabled")),
+      min_pushback_target_bitrate_bps_(
+          RateControlSettings::ParseFromKeyValueConfig(key_value_config)
+              .CongestionWindowMinPushbackTargetBitrateBps()),
+      current_data_window_(
+          RateControlSettings::ParseFromKeyValueConfig(key_value_config)
+              .CongestionWindowInitialDataWindow()) {}
+
+void CongestionWindowPushbackController::UpdateOutstandingData(
+    int64_t outstanding_bytes) {
+  outstanding_bytes_ = outstanding_bytes;
+}
+void CongestionWindowPushbackController::UpdatePacingQueue(
+    int64_t pacing_bytes) {
+  pacing_bytes_ = pacing_bytes;
+}
+
+void CongestionWindowPushbackController::SetDataWindow(DataSize data_window) {
+  current_data_window_ = data_window;
+}
+
+uint32_t CongestionWindowPushbackController::UpdateTargetBitrate(
+    uint32_t bitrate_bps) {
+  if (!current_data_window_ || current_data_window_->IsZero())
+    return bitrate_bps;
+  int64_t total_bytes = outstanding_bytes_;
+  if (add_pacing_)
+    total_bytes += pacing_bytes_;
+  double fill_ratio =
+      total_bytes / static_cast<double>(current_data_window_->bytes());
+  if (fill_ratio > 1.5) {
+    encoding_rate_ratio_ *= 0.9;
+  } else if (fill_ratio > 1) {
+    encoding_rate_ratio_ *= 0.95;
+  } else if (fill_ratio < 0.1) {
+    encoding_rate_ratio_ = 1.0;
+  } else {
+    encoding_rate_ratio_ *= 1.05;
+    encoding_rate_ratio_ = std::min(encoding_rate_ratio_, 1.0);
+  }
+  uint32_t adjusted_target_bitrate_bps =
+      static_cast<uint32_t>(bitrate_bps * encoding_rate_ratio_);
+
+  // Do not adjust below the minimum pushback bitrate but do obey if the
+  // original estimate is below it.
+  bitrate_bps = adjusted_target_bitrate_bps < min_pushback_target_bitrate_bps_
+                    ? std::min(bitrate_bps, min_pushback_target_bitrate_bps_)
+                    : adjusted_target_bitrate_bps;
+  return bitrate_bps;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h
new file mode 100644
index 0000000000..ea9ed97c3d
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_size.h"
+
+namespace webrtc {
+
+// This class enables pushback from congestion window directly to video encoder.
+// When the congestion window is filling up, the video encoder target bitrate
+// will be reduced accordingly to accommodate the network changes. To avoid
+// pausing video too frequently, a minimum encoder target bitrate threshold is
+// used to prevent video pause due to a full congestion window.
+class CongestionWindowPushbackController {
+ public:
+  explicit CongestionWindowPushbackController(
+      const FieldTrialsView* key_value_config);
+  void UpdateOutstandingData(int64_t outstanding_bytes);
+  void UpdatePacingQueue(int64_t pacing_bytes);
+  uint32_t UpdateTargetBitrate(uint32_t bitrate_bps);
+  void SetDataWindow(DataSize data_window);
+
+ private:
+  const bool add_pacing_;
+  const uint32_t min_pushback_target_bitrate_bps_;
+  absl::optional<DataSize> current_data_window_;
+  int64_t outstanding_bytes_ = 0;
+  int64_t pacing_bytes_ = 0;
+  double encoding_rate_ratio_ = 1.0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller_unittest.cc
new file mode 100644
index 0000000000..62dde02323
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
+
+#include <memory>
+
+#include "api/transport/field_trial_based_config.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+
+namespace webrtc {
+namespace test {
+
+class CongestionWindowPushbackControllerTest : public ::testing::Test {
+ public:
+  CongestionWindowPushbackControllerTest() {
+    cwnd_controller_.reset(
+        new CongestionWindowPushbackController(&field_trial_config_));
+  }
+
+ protected:
+  FieldTrialBasedConfig field_trial_config_;
+
+  std::unique_ptr<CongestionWindowPushbackController> cwnd_controller_;
+};
+
+TEST_F(CongestionWindowPushbackControllerTest, FullCongestionWindow) {
+  cwnd_controller_->UpdateOutstandingData(100000);
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(50000));
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(72000u, bitrate_bps);
+
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(50000));
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(static_cast<uint32_t>(72000 * 0.9 * 0.9), bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, NormalCongestionWindow) {
+  cwnd_controller_->UpdateOutstandingData(199999);
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(200000));
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(80000u, bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, LowBitrate) {
+  cwnd_controller_->UpdateOutstandingData(100000);
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(50000));
+
+  uint32_t bitrate_bps = 35000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(static_cast<uint32_t>(35000 * 0.9), bitrate_bps);
+
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(20000));
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(30000u, bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, NoPushbackOnDataWindowUnset) {
+  cwnd_controller_->UpdateOutstandingData(1e8);  // Large number
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(80000u, bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, PushbackOnInititialDataWindow) {
+  test::ScopedFieldTrials trials("WebRTC-CongestionWindow/InitWin:100000/");
+  cwnd_controller_.reset(
+      new CongestionWindowPushbackController(&field_trial_config_));
+  cwnd_controller_->UpdateOutstandingData(1e8);  // Large number
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_GT(80000u, bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, PushbackDropFrame) {
+  test::ScopedFieldTrials trials("WebRTC-CongestionWindow/DropFrame:true/");
+  cwnd_controller_.reset(
+      new CongestionWindowPushbackController(&field_trial_config_));
+  cwnd_controller_->UpdateOutstandingData(1e8);  // Large number
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(50000));
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_GT(80000u, bitrate_bps);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
new file mode 100644
index 0000000000..07ac599148
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
@@ -0,0 +1,305 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstdio>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/units/time_delta.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+constexpr TimeDelta kStreamTimeOut = TimeDelta::Seconds(2);
+constexpr TimeDelta kSendTimeGroupLength = TimeDelta::Millis(5);
+
+// This ssrc is used to fulfill the current API but will be removed
+// after the API has been changed.
+constexpr uint32_t kFixedSsrc = 0;
+}  // namespace
+
+constexpr char BweSeparateAudioPacketsSettings::kKey[];
+
+BweSeparateAudioPacketsSettings::BweSeparateAudioPacketsSettings(
+    const FieldTrialsView* key_value_config) {
+  Parser()->Parse(
+      key_value_config->Lookup(BweSeparateAudioPacketsSettings::kKey));
+}
+
+std::unique_ptr<StructParametersParser>
+BweSeparateAudioPacketsSettings::Parser() {
+  return StructParametersParser::Create(      //
+      "enabled", &enabled,                    //
+      "packet_threshold", &packet_threshold,  //
+      "time_threshold", &time_threshold);
+}
+
+DelayBasedBwe::Result::Result()
+    : updated(false),
+      probe(false),
+      target_bitrate(DataRate::Zero()),
+      recovered_from_overuse(false) {}
+
+DelayBasedBwe::DelayBasedBwe(const FieldTrialsView* key_value_config,
+                             RtcEventLog* event_log,
+                             NetworkStatePredictor* network_state_predictor)
+    : event_log_(event_log),
+      key_value_config_(key_value_config),
+      separate_audio_(key_value_config),
+      audio_packets_since_last_video_(0),
+      last_video_packet_recv_time_(Timestamp::MinusInfinity()),
+      network_state_predictor_(network_state_predictor),
+      video_delay_detector_(
+          new TrendlineEstimator(key_value_config_, network_state_predictor_)),
+      audio_delay_detector_(
+          new TrendlineEstimator(key_value_config_, network_state_predictor_)),
+      active_delay_detector_(video_delay_detector_.get()),
+      last_seen_packet_(Timestamp::MinusInfinity()),
+      uma_recorded_(false),
+      rate_control_(key_value_config, /*send_side=*/true),
+      prev_bitrate_(DataRate::Zero()),
+      prev_state_(BandwidthUsage::kBwNormal) {
+  RTC_LOG(LS_INFO)
+      << "Initialized DelayBasedBwe with separate audio overuse detection"
+      << separate_audio_.Parser()->Encode();
+}
+
+DelayBasedBwe::~DelayBasedBwe() {}
+
+DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
+    const TransportPacketsFeedback& msg,
+    absl::optional<DataRate> acked_bitrate,
+    absl::optional<DataRate> probe_bitrate,
+    absl::optional<NetworkStateEstimate> network_estimate,
+    bool in_alr) {
+  RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
+
+  auto packet_feedback_vector = msg.SortedByReceiveTime();
+  // TODO(holmer): An empty feedback vector here likely means that
+  // all acks were too late and that the send time history had
+  // timed out. We should reduce the rate when this occurs.
+  if (packet_feedback_vector.empty()) {
+    RTC_LOG(LS_WARNING) << "Very late feedback received.";
+    return DelayBasedBwe::Result();
+  }
+
+  if (!uma_recorded_) {
+    RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
+                              BweNames::kSendSideTransportSeqNum,
+                              BweNames::kBweNamesMax);
+    uma_recorded_ = true;
+  }
+  bool delayed_feedback = true;
+  bool recovered_from_overuse = false;
+  BandwidthUsage prev_detector_state = active_delay_detector_->State();
+  for (const auto& packet_feedback : packet_feedback_vector) {
+    delayed_feedback = false;
+    IncomingPacketFeedback(packet_feedback, msg.feedback_time);
+    if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
+        active_delay_detector_->State() == BandwidthUsage::kBwNormal) {
+      recovered_from_overuse = true;
+    }
+    prev_detector_state = active_delay_detector_->State();
+  }
+
+  if (delayed_feedback) {
+    // TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
+    // against building very large network queues.
+    return Result();
+  }
+  rate_control_.SetInApplicationLimitedRegion(in_alr);
+  rate_control_.SetNetworkStateEstimate(network_estimate);
+  return MaybeUpdateEstimate(acked_bitrate, probe_bitrate,
+                             std::move(network_estimate),
+                             recovered_from_overuse, in_alr, msg.feedback_time);
+}
+
+void DelayBasedBwe::IncomingPacketFeedback(const PacketResult& packet_feedback,
+                                           Timestamp at_time) {
+  // Reset if the stream has timed out.
+  if (last_seen_packet_.IsInfinite() ||
+      at_time - last_seen_packet_ > kStreamTimeOut) {
+    video_inter_arrival_delta_ =
+        std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
+    audio_inter_arrival_delta_ =
+        std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
+
+    video_delay_detector_.reset(
+        new TrendlineEstimator(key_value_config_, network_state_predictor_));
+    audio_delay_detector_.reset(
+        new TrendlineEstimator(key_value_config_, network_state_predictor_));
+    active_delay_detector_ = video_delay_detector_.get();
+  }
+  last_seen_packet_ = at_time;
+
+  // As an alternative to ignoring small packets, we can separate audio and
+  // video packets for overuse detection.
+  DelayIncreaseDetectorInterface* delay_detector_for_packet =
+      video_delay_detector_.get();
+  if (separate_audio_.enabled) {
+    if (packet_feedback.sent_packet.audio) {
+      delay_detector_for_packet = audio_delay_detector_.get();
+      audio_packets_since_last_video_++;
+      if (audio_packets_since_last_video_ > separate_audio_.packet_threshold &&
+          packet_feedback.receive_time - last_video_packet_recv_time_ >
+              separate_audio_.time_threshold) {
+        active_delay_detector_ = audio_delay_detector_.get();
+      }
+    } else {
+      audio_packets_since_last_video_ = 0;
+      last_video_packet_recv_time_ =
+          std::max(last_video_packet_recv_time_, packet_feedback.receive_time);
+      active_delay_detector_ = video_delay_detector_.get();
+    }
+  }
+  DataSize packet_size = packet_feedback.sent_packet.size;
+
+  TimeDelta send_delta = TimeDelta::Zero();
+  TimeDelta recv_delta = TimeDelta::Zero();
+  int size_delta = 0;
+
+  InterArrivalDelta* inter_arrival_for_packet =
+      (separate_audio_.enabled && packet_feedback.sent_packet.audio)
+          ? audio_inter_arrival_delta_.get()
+          : video_inter_arrival_delta_.get();
+  bool calculated_deltas = inter_arrival_for_packet->ComputeDeltas(
+      packet_feedback.sent_packet.send_time, packet_feedback.receive_time,
+      at_time, packet_size.bytes(), &send_delta, &recv_delta, &size_delta);
+
+  delay_detector_for_packet->Update(recv_delta.ms<double>(),
+                                    send_delta.ms<double>(),
+                                    packet_feedback.sent_packet.send_time.ms(),
+                                    packet_feedback.receive_time.ms(),
+                                    packet_size.bytes(), calculated_deltas);
+}
+
+DataRate DelayBasedBwe::TriggerOveruse(Timestamp at_time,
+                                       absl::optional<DataRate> link_capacity) {
+  RateControlInput input(BandwidthUsage::kBwOverusing, link_capacity);
+  return rate_control_.Update(&input, at_time);
+}
+
+DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
+    absl::optional<DataRate> acked_bitrate,
+    absl::optional<DataRate> probe_bitrate,
+    absl::optional<NetworkStateEstimate> state_estimate,
+    bool recovered_from_overuse,
+    bool in_alr,
+    Timestamp at_time) {
+  Result result;
+
+  // Currently overusing the bandwidth.
+  if (active_delay_detector_->State() == BandwidthUsage::kBwOverusing) {
+    if (acked_bitrate &&
+        rate_control_.TimeToReduceFurther(at_time, *acked_bitrate)) {
+      result.updated =
+          UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
+    } else if (!acked_bitrate && rate_control_.ValidEstimate() &&
+               rate_control_.InitialTimeToReduceFurther(at_time)) {
+      // Overusing before we have a measured acknowledged bitrate. Reduce send
+      // rate by 50% every 200 ms.
+      // TODO(tschumim): Improve this and/or the acknowledged bitrate estimator
+      // so that we (almost) always have a bitrate estimate.
+      rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, at_time);
+      result.updated = true;
+      result.probe = false;
+      result.target_bitrate = rate_control_.LatestEstimate();
+    }
+  } else {
+    if (probe_bitrate) {
+      result.probe = true;
+      result.updated = true;
+      rate_control_.SetEstimate(*probe_bitrate, at_time);
+      result.target_bitrate = rate_control_.LatestEstimate();
+    } else {
+      result.updated =
+          UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
+      result.recovered_from_overuse = recovered_from_overuse;
+    }
+  }
+  BandwidthUsage detector_state = active_delay_detector_->State();
+  if ((result.updated && prev_bitrate_ != result.target_bitrate) ||
+      detector_state != prev_state_) {
+    DataRate bitrate = result.updated ? result.target_bitrate : prev_bitrate_;
+
+    BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", at_time.ms(), bitrate.bps());
+
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventBweUpdateDelayBased>(
+          bitrate.bps(), detector_state));
+    }
+
+    prev_bitrate_ = bitrate;
+    prev_state_ = detector_state;
+  }
+
+  result.delay_detector_state = detector_state;
+  return result;
+}
+
+bool DelayBasedBwe::UpdateEstimate(Timestamp at_time,
+                                   absl::optional<DataRate> acked_bitrate,
+                                   DataRate* target_rate) {
+  const RateControlInput input(active_delay_detector_->State(), acked_bitrate);
+  *target_rate = rate_control_.Update(&input, at_time);
+  return rate_control_.ValidEstimate();
+}
+
+void DelayBasedBwe::OnRttUpdate(TimeDelta avg_rtt) {
+  rate_control_.SetRtt(avg_rtt);
+}
+
+bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
+                                   DataRate* bitrate) const {
+  // Currently accessed from both the process thread (see
+  // ModuleRtpRtcpImpl::Process()) and the configuration thread (see
+  // Call::GetStats()). Should in the future only be accessed from a single
+  // thread.
+  RTC_DCHECK(ssrcs);
+  RTC_DCHECK(bitrate);
+  if (!rate_control_.ValidEstimate())
+    return false;
+
+  *ssrcs = {kFixedSsrc};
+  *bitrate = rate_control_.LatestEstimate();
+  return true;
+}
+
+void DelayBasedBwe::SetStartBitrate(DataRate start_bitrate) {
+  RTC_LOG(LS_INFO) << "BWE Setting start bitrate to: "
+                   << ToString(start_bitrate);
+  rate_control_.SetStartBitrate(start_bitrate);
+}
+
+void DelayBasedBwe::SetMinBitrate(DataRate min_bitrate) {
+  // Called from both the configuration thread and the network thread. Shouldn't
+  // be called from the network thread in the future.
+  rate_control_.SetMinBitrate(min_bitrate);
+}
+
+TimeDelta DelayBasedBwe::GetExpectedBwePeriod() const {
+  return rate_control_.GetExpectedBandwidthPeriod();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
new file mode 100644
index 0000000000..e91a1dff54
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
+#include "modules/congestion_controller/goog_cc/inter_arrival_delta.h"
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "modules/remote_bitrate_estimator/inter_arrival.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/race_checker.h"
+
+namespace webrtc {
+class RtcEventLog;
+
+struct BweSeparateAudioPacketsSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-SeparateAudioPackets";
+
+  BweSeparateAudioPacketsSettings() = default;
+  explicit BweSeparateAudioPacketsSettings(
+      const FieldTrialsView* key_value_config);
+
+  bool enabled = false;
+  int packet_threshold = 10;
+  TimeDelta time_threshold = TimeDelta::Seconds(1);
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class DelayBasedBwe {
+ public:
+  struct Result {
+    Result();
+    ~Result() = default;
+    bool updated;
+    bool probe;
+    DataRate target_bitrate = DataRate::Zero();
+    bool recovered_from_overuse;
+    BandwidthUsage delay_detector_state;
+  };
+
+  explicit DelayBasedBwe(const FieldTrialsView* key_value_config,
+                         RtcEventLog* event_log,
+                         NetworkStatePredictor* network_state_predictor);
+
+  DelayBasedBwe() = delete;
+  DelayBasedBwe(const DelayBasedBwe&) = delete;
+  DelayBasedBwe& operator=(const DelayBasedBwe&) = delete;
+
+  virtual ~DelayBasedBwe();
+
+  Result IncomingPacketFeedbackVector(
+      const TransportPacketsFeedback& msg,
+      absl::optional<DataRate> acked_bitrate,
+      absl::optional<DataRate> probe_bitrate,
+      absl::optional<NetworkStateEstimate> network_estimate,
+      bool in_alr);
+  void OnRttUpdate(TimeDelta avg_rtt);
+  bool LatestEstimate(std::vector<uint32_t>* ssrcs, DataRate* bitrate) const;
+  void SetStartBitrate(DataRate start_bitrate);
+  void SetMinBitrate(DataRate min_bitrate);
+  TimeDelta GetExpectedBwePeriod() const;
+  DataRate TriggerOveruse(Timestamp at_time,
+                          absl::optional<DataRate> link_capacity);
+  DataRate last_estimate() const { return prev_bitrate_; }
+  BandwidthUsage last_state() const { return prev_state_; }
+
+ private:
+  friend class GoogCcStatePrinter;
+  void IncomingPacketFeedback(const PacketResult& packet_feedback,
+                              Timestamp at_time);
+  Result MaybeUpdateEstimate(
+      absl::optional<DataRate> acked_bitrate,
+      absl::optional<DataRate> probe_bitrate,
+      absl::optional<NetworkStateEstimate> state_estimate,
+      bool recovered_from_overuse,
+      bool in_alr,
+      Timestamp at_time);
+  // Updates the current remote rate estimate and returns true if a valid
+  // estimate exists.
+  bool UpdateEstimate(Timestamp at_time,
+                      absl::optional<DataRate> acked_bitrate,
+                      DataRate* target_rate);
+
+  rtc::RaceChecker network_race_;
+  RtcEventLog* const event_log_;
+  const FieldTrialsView* const key_value_config_;
+
+  // Alternatively, run two separate overuse detectors for audio and video,
+  // and fall back to the audio one if we haven't seen a video packet in a
+  // while.
+  BweSeparateAudioPacketsSettings separate_audio_;
+  int64_t audio_packets_since_last_video_;
+  Timestamp last_video_packet_recv_time_;
+
+  NetworkStatePredictor* network_state_predictor_;
+  std::unique_ptr<InterArrival> video_inter_arrival_;
+  std::unique_ptr<InterArrivalDelta> video_inter_arrival_delta_;
+  std::unique_ptr<DelayIncreaseDetectorInterface> video_delay_detector_;
+  std::unique_ptr<InterArrival> audio_inter_arrival_;
+  std::unique_ptr<InterArrivalDelta> audio_inter_arrival_delta_;
+  std::unique_ptr<DelayIncreaseDetectorInterface> audio_delay_detector_;
+  DelayIncreaseDetectorInterface* active_delay_detector_;
+
+  Timestamp last_seen_packet_;
+  bool uma_recorded_;
+  AimdRateControl rate_control_;
+  DataRate prev_bitrate_;
+  BandwidthUsage prev_state_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build
new file mode 100644
index 0000000000..15f5a583ef
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build
@@ -0,0 +1,234 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("delay_based_bwe_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
new file mode 100644
index 0000000000..b7dc6aae47
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
@@ -0,0 +1,309 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+
+#include <string>
+
+#include "api/transport/network_types.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kNumProbesCluster0 = 5;
+constexpr int kNumProbesCluster1 = 8;
+const PacedPacketInfo kPacingInfo0(0, kNumProbesCluster0, 2000);
+const PacedPacketInfo kPacingInfo1(1, kNumProbesCluster1, 4000);
+constexpr float kTargetUtilizationFraction = 0.95f;
+}  // namespace
+
+TEST_F(DelayBasedBweTest, ProbeDetection) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  for (int i = 0; i < kNumProbesCluster0; ++i) {
+    clock_.AdvanceTimeMilliseconds(10);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, now_ms, 1000, kPacingInfo0);
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+
+  // Second burst sent at 8 * 1000 / 5 = 1600 kbps.
+  for (int i = 0; i < kNumProbesCluster1; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, now_ms, 1000, kPacingInfo1);
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_GT(bitrate_observer_.latest_bitrate(), 1500000u);
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionNonPacedPackets) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps, but with every other packet
+  // not being paced which could mess things up.
+  for (int i = 0; i < kNumProbesCluster0; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, now_ms, 1000, kPacingInfo0);
+    // Non-paced packet, arriving 5 ms after.
+    clock_.AdvanceTimeMilliseconds(5);
+    IncomingFeedback(now_ms, now_ms, 100, PacedPacketInfo());
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_GT(bitrate_observer_.latest_bitrate(), 800000u);
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionFasterArrival) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  // Arriving at 8 * 1000 / 5 = 1600 kbps.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kNumProbesCluster0; ++i) {
+    clock_.AdvanceTimeMilliseconds(1);
+    send_time_ms += 10;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, send_time_ms, 1000, kPacingInfo0);
+  }
+
+  EXPECT_FALSE(bitrate_observer_.updated());
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrival) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 5 = 1600 kbps.
+  // Arriving at 8 * 1000 / 7 = 1142 kbps.
+  // Since the receive rate is significantly below the send rate, we expect to
+  // use 95% of the estimated capacity.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kNumProbesCluster1; ++i) {
+    clock_.AdvanceTimeMilliseconds(7);
+    send_time_ms += 5;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, send_time_ms, 1000, kPacingInfo1);
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
+              kTargetUtilizationFraction * 1140000u, 10000u);
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrivalHighBitrate) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // Burst sent at 8 * 1000 / 1 = 8000 kbps.
+  // Arriving at 8 * 1000 / 2 = 4000 kbps.
+  // Since the receive rate is significantly below the send rate, we expect to
+  // use 95% of the estimated capacity.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kNumProbesCluster1; ++i) {
+    clock_.AdvanceTimeMilliseconds(2);
+    send_time_ms += 1;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, send_time_ms, 1000, kPacingInfo1);
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
+              kTargetUtilizationFraction * 4000000u, 10000u);
+}
+
+TEST_F(DelayBasedBweTest, GetExpectedBwePeriodMs) {
+  auto default_interval = bitrate_estimator_->GetExpectedBwePeriod();
+  EXPECT_GT(default_interval.ms(), 0);
+  CapacityDropTestHelper(1, true, 333, 0);
+  auto interval = bitrate_estimator_->GetExpectedBwePeriod();
+  EXPECT_GT(interval.ms(), 0);
+  EXPECT_NE(interval.ms(), default_interval.ms());
+}
+
+TEST_F(DelayBasedBweTest, InitialBehavior) {
+  InitialBehaviorTestHelper(730000);
+}
+
+TEST_F(DelayBasedBweTest, RateIncreaseReordering) {
+  RateIncreaseReorderingTestHelper(730000);
+}
+TEST_F(DelayBasedBweTest, RateIncreaseRtpTimestamps) {
+  RateIncreaseRtpTimestampsTestHelper(622);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropOneStream) {
+  CapacityDropTestHelper(1, false, 300, 0);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropPosOffsetChange) {
+  CapacityDropTestHelper(1, false, 867, 30000);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropNegOffsetChange) {
+  CapacityDropTestHelper(1, false, 933, -30000);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropOneStreamWrap) {
+  CapacityDropTestHelper(1, true, 333, 0);
+}
+
+TEST_F(DelayBasedBweTest, TestTimestampGrouping) {
+  TestTimestampGroupingTestHelper();
+}
+
+TEST_F(DelayBasedBweTest, TestShortTimeoutAndWrap) {
+  // Simulate a client leaving and rejoining the call after 35 seconds. This
+  // will make abs send time wrap, so if streams aren't timed out properly
+  // the next 30 seconds of packets will be out of order.
+  TestWrappingHelper(35);
+}
+
+TEST_F(DelayBasedBweTest, TestLongTimeoutAndWrap) {
+  // Simulate a client leaving and rejoining the call after some multiple of
+  // 64 seconds later. This will cause a zero difference in abs send times due
+  // to the wrap, but a big difference in arrival time, if streams aren't
+  // properly timed out.
+  TestWrappingHelper(10 * 64);
+}
+
+TEST_F(DelayBasedBweTest, TestInitialOveruse) {
+  const DataRate kStartBitrate = DataRate::KilobitsPerSec(300);
+  const DataRate kInitialCapacity = DataRate::KilobitsPerSec(200);
+  const uint32_t kDummySsrc = 0;
+  // High FPS to ensure that we send a lot of packets in a short time.
+  const int kFps = 90;
+
+  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate.bps()));
+  stream_generator_->set_capacity_bps(kInitialCapacity.bps());
+
+  // Needed to initialize the AimdRateControl.
+  bitrate_estimator_->SetStartBitrate(kStartBitrate);
+
+  // Produce 30 frames (in 1/3 second) and give them to the estimator.
+  int64_t bitrate_bps = kStartBitrate.bps();
+  bool seen_overuse = false;
+  for (int i = 0; i < 30; ++i) {
+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
+    // The purpose of this test is to ensure that we back down even if we don't
+    // have any acknowledged bitrate estimate yet. Hence, if the test works
+    // as expected, we should not have a measured bitrate yet.
+    EXPECT_FALSE(acknowledged_bitrate_estimator_->bitrate().has_value());
+    if (overuse) {
+      EXPECT_TRUE(bitrate_observer_.updated());
+      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate.bps() / 2,
+                  15000);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      seen_overuse = true;
+      break;
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+  }
+  EXPECT_TRUE(seen_overuse);
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate.bps() / 2,
+              15000);
+}
+
+TEST_F(DelayBasedBweTest, TestTimestampPrecisionHandling) {
+  // This test does some basic checks to make sure that timestamps with higher
+  // than millisecond precision are handled properly and do not cause any
+  // problems in the estimator. Specifically, previously reported in
+  // webrtc:14023 and described in more details there, the rounding to the
+  // nearest milliseconds caused discrepancy in the accumulated delay. This lead
+  // to false-positive overuse detection.
+  // Technical details of the test:
+  // Send times(ms): 0.000,  9.725, 20.000, 29.725, 40.000, 49.725, ...
+  // Recv times(ms): 0.500, 10.000, 20.500, 30.000, 40.500, 50.000, ...
+  // Send deltas(ms):   9.750,  10.250,  9.750, 10.250,  9.750, ...
+  // Recv deltas(ms):   9.500,  10.500,  9.500, 10.500,  9.500, ...
+  // There is no delay building up between the send times and the receive times,
+  // therefore this case should never lead to an overuse detection. However, if
+  // the time deltas were accidentally rounded to the nearest milliseconds, then
+  // all the send deltas would be equal to 10ms while some recv deltas would
+  // round up to 11ms which would lead in a false illusion of delay build up.
+  uint32_t last_bitrate = bitrate_observer_.latest_bitrate();
+  for (int i = 0; i < 1000; ++i) {
+    clock_.AdvanceTimeMicroseconds(500);
+    IncomingFeedback(clock_.CurrentTime(),
+                     clock_.CurrentTime() - TimeDelta::Micros(500), 1000,
+                     PacedPacketInfo());
+    clock_.AdvanceTimeMicroseconds(9500);
+    IncomingFeedback(clock_.CurrentTime(),
+                     clock_.CurrentTime() - TimeDelta::Micros(250), 1000,
+                     PacedPacketInfo());
+    clock_.AdvanceTimeMicroseconds(10000);
+
+    // The bitrate should never decrease in this test.
+    EXPECT_LE(last_bitrate, bitrate_observer_.latest_bitrate());
+    last_bitrate = bitrate_observer_.latest_bitrate();
+  }
+}
+
+class DelayBasedBweTestWithBackoffTimeoutExperiment : public DelayBasedBweTest {
+ public:
+  DelayBasedBweTestWithBackoffTimeoutExperiment()
+      : DelayBasedBweTest(
+            "WebRTC-BweAimdRateControlConfig/initial_backoff_interval:200ms/") {
+  }
+};
+
+// This test subsumes and improves DelayBasedBweTest.TestInitialOveruse above.
+TEST_F(DelayBasedBweTestWithBackoffTimeoutExperiment, TestInitialOveruse) {
+  const DataRate kStartBitrate = DataRate::KilobitsPerSec(300);
+  const DataRate kInitialCapacity = DataRate::KilobitsPerSec(200);
+  const uint32_t kDummySsrc = 0;
+  // High FPS to ensure that we send a lot of packets in a short time.
+  const int kFps = 90;
+
+  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate.bps()));
+  stream_generator_->set_capacity_bps(kInitialCapacity.bps());
+
+  // Needed to initialize the AimdRateControl.
+  bitrate_estimator_->SetStartBitrate(kStartBitrate);
+
+  // Produce 30 frames (in 1/3 second) and give them to the estimator.
+  int64_t bitrate_bps = kStartBitrate.bps();
+  bool seen_overuse = false;
+  for (int frames = 0; frames < 30 && !seen_overuse; ++frames) {
+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
+    // The purpose of this test is to ensure that we back down even if we don't
+    // have any acknowledged bitrate estimate yet. Hence, if the test works
+    // as expected, we should not have a measured bitrate yet.
+    EXPECT_FALSE(acknowledged_bitrate_estimator_->bitrate().has_value());
+    if (overuse) {
+      EXPECT_TRUE(bitrate_observer_.updated());
+      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate.bps() / 2,
+                  15000);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      seen_overuse = true;
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+  }
+  EXPECT_TRUE(seen_overuse);
+  // Continue generating an additional 15 frames (equivalent to 167 ms) and
+  // verify that we don't back down further.
+  for (int frames = 0; frames < 15 && seen_overuse; ++frames) {
+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
+    EXPECT_FALSE(overuse);
+    if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      EXPECT_GE(bitrate_bps, kStartBitrate.bps() / 2 - 15000);
+      EXPECT_LE(bitrate_bps, kInitialCapacity.bps() + 15000);
+      bitrate_observer_.Reset();
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
new file mode 100644
index 0000000000..8618a7814e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
@@ -0,0 +1,529 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+constexpr size_t kMtu = 1200;
+constexpr uint32_t kAcceptedBitrateErrorBps = 50000;
+
+// Number of packets needed before we have a valid estimate.
+constexpr int kNumInitialPackets = 2;
+
+constexpr int kInitialProbingPackets = 5;
+
+namespace test {
+
+void TestBitrateObserver::OnReceiveBitrateChanged(uint32_t bitrate) {
+  latest_bitrate_ = bitrate;
+  updated_ = true;
+}
+
+RtpStream::RtpStream(int fps, int bitrate_bps)
+    : fps_(fps), bitrate_bps_(bitrate_bps), next_rtp_time_(0) {
+  RTC_CHECK_GT(fps_, 0);
+}
+
+// Generates a new frame for this stream. If called too soon after the
+// previous frame, no frame will be generated. The frame is split into
+// packets.
+int64_t RtpStream::GenerateFrame(int64_t time_now_us,
+                                 std::vector<PacketResult>* packets) {
+  if (time_now_us < next_rtp_time_) {
+    return next_rtp_time_;
+  }
+  RTC_CHECK(packets != NULL);
+  size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
+  size_t n_packets =
+      std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
+  size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
+  for (size_t i = 0; i < n_packets; ++i) {
+    PacketResult packet;
+    packet.sent_packet.send_time =
+        Timestamp::Micros(time_now_us + kSendSideOffsetUs);
+    packet.sent_packet.size = DataSize::Bytes(payload_size);
+    packets->push_back(packet);
+  }
+  next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
+  return next_rtp_time_;
+}
+
+// The send-side time when the next frame can be generated.
+int64_t RtpStream::next_rtp_time() const {
+  return next_rtp_time_;
+}
+
+void RtpStream::set_bitrate_bps(int bitrate_bps) {
+  ASSERT_GE(bitrate_bps, 0);
+  bitrate_bps_ = bitrate_bps;
+}
+
+int RtpStream::bitrate_bps() const {
+  return bitrate_bps_;
+}
+
+bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs,
+                        const std::unique_ptr<RtpStream>& rhs) {
+  return lhs->next_rtp_time_ < rhs->next_rtp_time_;
+}
+
+StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
+    : capacity_(capacity), prev_arrival_time_us_(time_now) {}
+
+StreamGenerator::~StreamGenerator() = default;
+
+// Add a new stream.
+void StreamGenerator::AddStream(RtpStream* stream) {
+  streams_.push_back(std::unique_ptr<RtpStream>(stream));
+}
+
+// Set the link capacity.
+void StreamGenerator::set_capacity_bps(int capacity_bps) {
+  ASSERT_GT(capacity_bps, 0);
+  capacity_ = capacity_bps;
+}
+
+// Divides `bitrate_bps` among all streams. The allocated bitrate per stream
+// is decided by the current allocation ratios.
+void StreamGenerator::SetBitrateBps(int bitrate_bps) {
+  ASSERT_GE(streams_.size(), 0u);
+  int total_bitrate_before = 0;
+  for (const auto& stream : streams_) {
+    total_bitrate_before += stream->bitrate_bps();
+  }
+  int64_t bitrate_before = 0;
+  int total_bitrate_after = 0;
+  for (const auto& stream : streams_) {
+    bitrate_before += stream->bitrate_bps();
+    int64_t bitrate_after =
+        (bitrate_before * bitrate_bps + total_bitrate_before / 2) /
+        total_bitrate_before;
+    stream->set_bitrate_bps(bitrate_after - total_bitrate_after);
+    total_bitrate_after += stream->bitrate_bps();
+  }
+  ASSERT_EQ(bitrate_before, total_bitrate_before);
+  EXPECT_EQ(total_bitrate_after, bitrate_bps);
+}
+
+// TODO(holmer): Break out the channel simulation part from this class to make
+// it possible to simulate different types of channels.
+int64_t StreamGenerator::GenerateFrame(std::vector<PacketResult>* packets,
+                                       int64_t time_now_us) {
+  RTC_CHECK(packets != NULL);
+  RTC_CHECK(packets->empty());
+  RTC_CHECK_GT(capacity_, 0);
+  auto it =
+      std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+  (*it)->GenerateFrame(time_now_us, packets);
+  for (PacketResult& packet : *packets) {
+    int capacity_bpus = capacity_ / 1000;
+    int64_t required_network_time_us =
+        (8 * 1000 * packet.sent_packet.size.bytes() + capacity_bpus / 2) /
+        capacity_bpus;
+    prev_arrival_time_us_ =
+        std::max(time_now_us + required_network_time_us,
+                 prev_arrival_time_us_ + required_network_time_us);
+    packet.receive_time = Timestamp::Micros(prev_arrival_time_us_);
+  }
+  it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+  return std::max((*it)->next_rtp_time(), time_now_us);
+}
+}  // namespace test
+
+DelayBasedBweTest::DelayBasedBweTest() : DelayBasedBweTest("") {}
+
+DelayBasedBweTest::DelayBasedBweTest(absl::string_view field_trial_string)
+    : field_trial(
+          std::make_unique<test::ScopedFieldTrials>(field_trial_string)),
+      clock_(100000000),
+      acknowledged_bitrate_estimator_(
+          AcknowledgedBitrateEstimatorInterface::Create(&field_trial_config_)),
+      probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)),
+      bitrate_estimator_(
+          new DelayBasedBwe(&field_trial_config_, nullptr, nullptr)),
+      stream_generator_(new test::StreamGenerator(1e6,  // Capacity.
+                                                  clock_.TimeInMicroseconds())),
+      arrival_time_offset_ms_(0),
+      first_update_(true) {}
+
+DelayBasedBweTest::~DelayBasedBweTest() {}
+
+void DelayBasedBweTest::AddDefaultStream() {
+  stream_generator_->AddStream(new test::RtpStream(30, 3e5));
+}
+
+const uint32_t DelayBasedBweTest::kDefaultSsrc = 0;
+
+void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
+                                         int64_t send_time_ms,
+                                         size_t payload_size) {
+  IncomingFeedback(arrival_time_ms, send_time_ms, payload_size,
+                   PacedPacketInfo());
+}
+
+void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
+                                         int64_t send_time_ms,
+                                         size_t payload_size,
+                                         const PacedPacketInfo& pacing_info) {
+  RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
+  IncomingFeedback(Timestamp::Millis(arrival_time_ms + arrival_time_offset_ms_),
+                   Timestamp::Millis(send_time_ms), payload_size, pacing_info);
+}
+
+void DelayBasedBweTest::IncomingFeedback(Timestamp receive_time,
+                                         Timestamp send_time,
+                                         size_t payload_size,
+                                         const PacedPacketInfo& pacing_info) {
+  PacketResult packet;
+  packet.receive_time = receive_time;
+  packet.sent_packet.send_time = send_time;
+  packet.sent_packet.size = DataSize::Bytes(payload_size);
+  packet.sent_packet.pacing_info = pacing_info;
+  if (packet.sent_packet.pacing_info.probe_cluster_id !=
+      PacedPacketInfo::kNotAProbe)
+    probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
+
+  TransportPacketsFeedback msg;
+  msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());
+  msg.packet_feedbacks.push_back(packet);
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
+      msg.SortedByReceiveTime());
+  DelayBasedBwe::Result result =
+      bitrate_estimator_->IncomingPacketFeedbackVector(
+          msg, acknowledged_bitrate_estimator_->bitrate(),
+          probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
+          /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
+  if (result.updated) {
+    bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
+  }
+}
+
+// Generates a frame of packets belonging to a stream at a given bitrate and
+// with a given ssrc. The stream is pushed through a very simple simulated
+// network, and is then given to the receive-side bandwidth estimator.
+// Returns true if an over-use was seen, false otherwise.
+// The StreamGenerator::updated() should be used to check for any changes in
+// target bitrate after the call to this function.
+bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
+                                                uint32_t bitrate_bps) {
+  stream_generator_->SetBitrateBps(bitrate_bps);
+  std::vector<PacketResult> packets;
+
+  int64_t next_time_us =
+      stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
+  if (packets.empty())
+    return false;
+
+  bool overuse = false;
+  bitrate_observer_.Reset();
+  clock_.AdvanceTimeMicroseconds(packets.back().receive_time.us() -
+                                 clock_.TimeInMicroseconds());
+  for (auto& packet : packets) {
+    RTC_CHECK_GE(packet.receive_time.ms() + arrival_time_offset_ms_, 0);
+    packet.receive_time += TimeDelta::Millis(arrival_time_offset_ms_);
+
+    if (packet.sent_packet.pacing_info.probe_cluster_id !=
+        PacedPacketInfo::kNotAProbe)
+      probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
+  }
+
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
+  TransportPacketsFeedback msg;
+  msg.packet_feedbacks = packets;
+  msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());
+
+  DelayBasedBwe::Result result =
+      bitrate_estimator_->IncomingPacketFeedbackVector(
+          msg, acknowledged_bitrate_estimator_->bitrate(),
+          probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
+          /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
+  if (result.updated) {
+    bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
+    if (!first_update_ && result.target_bitrate.bps() < bitrate_bps)
+      overuse = true;
+    first_update_ = false;
+  }
+
+  clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
+  return overuse;
+}
+
+// Run the bandwidth estimator with a stream of `number_of_frames` frames, or
+// until it reaches `target_bitrate`.
+// Can for instance be used to run the estimator for some time to get it
+// into a steady state.
+uint32_t DelayBasedBweTest::SteadyStateRun(uint32_t ssrc,
+                                           int max_number_of_frames,
+                                           uint32_t start_bitrate,
+                                           uint32_t min_bitrate,
+                                           uint32_t max_bitrate,
+                                           uint32_t target_bitrate) {
+  uint32_t bitrate_bps = start_bitrate;
+  bool bitrate_update_seen = false;
+  // Produce `number_of_frames` frames and give them to the estimator.
+  for (int i = 0; i < max_number_of_frames; ++i) {
+    bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
+    if (overuse) {
+      EXPECT_LT(bitrate_observer_.latest_bitrate(), max_bitrate);
+      EXPECT_GT(bitrate_observer_.latest_bitrate(), min_bitrate);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_update_seen = true;
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+    if (bitrate_update_seen && bitrate_bps > target_bitrate) {
+      break;
+    }
+  }
+  EXPECT_TRUE(bitrate_update_seen);
+  return bitrate_bps;
+}
+
+void DelayBasedBweTest::InitialBehaviorTestHelper(
+    uint32_t expected_converge_bitrate) {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const PacedPacketInfo kPacingInfo(0, 5, 5000);
+  DataRate bitrate = DataRate::Zero();
+  int64_t send_time_ms = 0;
+  std::vector<uint32_t> ssrcs;
+  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+  EXPECT_EQ(0u, ssrcs.size());
+  clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+  EXPECT_FALSE(bitrate_observer_.updated());
+  bitrate_observer_.Reset();
+  clock_.AdvanceTimeMilliseconds(1000);
+  // Inserting packets for 5 seconds to get a valid estimate.
+  for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+    // NOTE!!! If the following line is moved under the if case then this test
+    //         wont work on windows realease bots.
+    PacedPacketInfo pacing_info =
+        i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
+
+    if (i == kNumInitialPackets) {
+      EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+      EXPECT_EQ(0u, ssrcs.size());
+      EXPECT_FALSE(bitrate_observer_.updated());
+      bitrate_observer_.Reset();
+    }
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
+                     pacing_info);
+    clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
+    send_time_ms += kFrameIntervalMs;
+  }
+  EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+  ASSERT_EQ(1u, ssrcs.size());
+  EXPECT_EQ(kDefaultSsrc, ssrcs.front());
+  EXPECT_NEAR(expected_converge_bitrate, bitrate.bps(),
+              kAcceptedBitrateErrorBps);
+  EXPECT_TRUE(bitrate_observer_.updated());
+  bitrate_observer_.Reset();
+  EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate.bps());
+}
+
+void DelayBasedBweTest::RateIncreaseReorderingTestHelper(
+    uint32_t expected_bitrate_bps) {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const PacedPacketInfo kPacingInfo(0, 5, 5000);
+  int64_t send_time_ms = 0;
+  // Inserting packets for five seconds to get a valid estimate.
+  for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+    // NOTE!!! If the following line is moved under the if case then this test
+    //         wont work on windows realease bots.
+    PacedPacketInfo pacing_info =
+        i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
+
+    // TODO(sprang): Remove this hack once the single stream estimator is gone,
+    // as it doesn't do anything in Process().
+    if (i == kNumInitialPackets) {
+      // Process after we have enough frames to get a valid input rate estimate.
+
+      EXPECT_FALSE(bitrate_observer_.updated());  // No valid estimate.
+    }
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
+                     pacing_info);
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
+              kAcceptedBitrateErrorBps);
+  for (int i = 0; i < 10; ++i) {
+    clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+    send_time_ms += 2 * kFrameIntervalMs;
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+    IncomingFeedback(clock_.TimeInMilliseconds(),
+                     send_time_ms - kFrameIntervalMs, 1000);
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
+              kAcceptedBitrateErrorBps);
+}
+
+// Make sure we initially increase the bitrate as expected.
+void DelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper(
+    int expected_iterations) {
+  // This threshold corresponds approximately to increasing linearly with
+  // bitrate(i) = 1.04 * bitrate(i-1) + 1000
+  // until bitrate(i) > 500000, with bitrate(1) ~= 30000.
+  uint32_t bitrate_bps = 30000;
+  int iterations = 0;
+  AddDefaultStream();
+  // Feed the estimator with a stream of packets and verify that it reaches
+  // 500 kbps at the expected time.
+  while (bitrate_bps < 5e5) {
+    bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+    if (overuse) {
+      EXPECT_GT(bitrate_observer_.latest_bitrate(), bitrate_bps);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+    ++iterations;
+  }
+  ASSERT_EQ(expected_iterations, iterations);
+}
+
+void DelayBasedBweTest::CapacityDropTestHelper(
+    int number_of_streams,
+    bool wrap_time_stamp,
+    uint32_t expected_bitrate_drop_delta,
+    int64_t receiver_clock_offset_change_ms) {
+  const int kFramerate = 30;
+  const int kStartBitrate = 900e3;
+  const int kMinExpectedBitrate = 800e3;
+  const int kMaxExpectedBitrate = 1100e3;
+  const uint32_t kInitialCapacityBps = 1000e3;
+  const uint32_t kReducedCapacityBps = 500e3;
+
+  int steady_state_time = 0;
+  if (number_of_streams <= 1) {
+    steady_state_time = 10;
+    AddDefaultStream();
+  } else {
+    steady_state_time = 10 * number_of_streams;
+    int bitrate_sum = 0;
+    int kBitrateDenom = number_of_streams * (number_of_streams - 1);
+    for (int i = 0; i < number_of_streams; i++) {
+      // First stream gets half available bitrate, while the rest share the
+      // remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
+      int bitrate = kStartBitrate / 2;
+      if (i > 0) {
+        bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
+      }
+      stream_generator_->AddStream(new test::RtpStream(kFramerate, bitrate));
+      bitrate_sum += bitrate;
+    }
+    ASSERT_EQ(bitrate_sum, kStartBitrate);
+  }
+
+  // Run in steady state to make the estimator converge.
+  stream_generator_->set_capacity_bps(kInitialCapacityBps);
+  uint32_t bitrate_bps = SteadyStateRun(
+      kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
+      kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
+  EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 180000u);
+  bitrate_observer_.Reset();
+
+  // Add an offset to make sure the BWE can handle it.
+  arrival_time_offset_ms_ += receiver_clock_offset_change_ms;
+
+  // Reduce the capacity and verify the decrease time.
+  stream_generator_->set_capacity_bps(kReducedCapacityBps);
+  int64_t overuse_start_time = clock_.TimeInMilliseconds();
+  int64_t bitrate_drop_time = -1;
+  for (int i = 0; i < 100 * number_of_streams; ++i) {
+    GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+    if (bitrate_drop_time == -1 &&
+        bitrate_observer_.latest_bitrate() <= kReducedCapacityBps) {
+      bitrate_drop_time = clock_.TimeInMilliseconds();
+    }
+    if (bitrate_observer_.updated())
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+  }
+
+  EXPECT_NEAR(expected_bitrate_drop_delta,
+              bitrate_drop_time - overuse_start_time, 33);
+}
+
+void DelayBasedBweTest::TestTimestampGroupingTestHelper() {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  int64_t send_time_ms = 0;
+  // Initial set of frames to increase the bitrate. 6 seconds to have enough
+  // time for the first estimate to be generated and for Process() to be called.
+  for (int i = 0; i <= 6 * kFramerate; ++i) {
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_GE(bitrate_observer_.latest_bitrate(), 400000u);
+
+  // Insert batches of frames which were sent very close in time. Also simulate
+  // capacity over-use to see that we back off correctly.
+  const int kTimestampGroupLength = 15;
+  for (int i = 0; i < 100; ++i) {
+    for (int j = 0; j < kTimestampGroupLength; ++j) {
+      // Insert `kTimestampGroupLength` frames with just 1 timestamp ticks in
+      // between. Should be treated as part of the same group by the estimator.
+      IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 100);
+      clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
+      send_time_ms += 1;
+    }
+    // Increase time until next batch to simulate over-use.
+    clock_.AdvanceTimeMilliseconds(10);
+    send_time_ms += kFrameIntervalMs - kTimestampGroupLength;
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  // Should have reduced the estimate.
+  EXPECT_LT(bitrate_observer_.latest_bitrate(), 400000u);
+}
+
+void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
+  const int kFramerate = 100;
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  int64_t send_time_ms = 0;
+
+  for (size_t i = 0; i < 3000; ++i) {
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  DataRate bitrate_before = DataRate::Zero();
+  std::vector<uint32_t> ssrcs;
+  bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before);
+
+  clock_.AdvanceTimeMilliseconds(silence_time_s * 1000);
+  send_time_ms += silence_time_s * 1000;
+
+  for (size_t i = 0; i < 24; ++i) {
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+    clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  DataRate bitrate_after = DataRate::Zero();
+  bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
+  EXPECT_LT(bitrate_after, bitrate_before);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
new file mode 100644
index 0000000000..d56fe892d5
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
@@ -0,0 +1,189 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/network_types.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "system_wrappers/include/clock.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+class TestBitrateObserver {
+ public:
+  TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
+  ~TestBitrateObserver() {}
+
+  void OnReceiveBitrateChanged(uint32_t bitrate);
+
+  void Reset() { updated_ = false; }
+
+  bool updated() const { return updated_; }
+
+  uint32_t latest_bitrate() const { return latest_bitrate_; }
+
+ private:
+  bool updated_;
+  uint32_t latest_bitrate_;
+};
+
+class RtpStream {
+ public:
+  enum { kSendSideOffsetUs = 1000000 };
+
+  RtpStream(int fps, int bitrate_bps);
+
+  RtpStream(const RtpStream&) = delete;
+  RtpStream& operator=(const RtpStream&) = delete;
+
+  // Generates a new frame for this stream. If called too soon after the
+  // previous frame, no frame will be generated. The frame is split into
+  // packets.
+  int64_t GenerateFrame(int64_t time_now_us,
+                        std::vector<PacketResult>* packets);
+
+  // The send-side time when the next frame can be generated.
+  int64_t next_rtp_time() const;
+
+  void set_bitrate_bps(int bitrate_bps);
+
+  int bitrate_bps() const;
+
+  static bool Compare(const std::unique_ptr<RtpStream>& lhs,
+                      const std::unique_ptr<RtpStream>& rhs);
+
+ private:
+  int fps_;
+  int bitrate_bps_;
+  int64_t next_rtp_time_;
+};
+
+class StreamGenerator {
+ public:
+  StreamGenerator(int capacity, int64_t time_now);
+  ~StreamGenerator();
+
+  StreamGenerator(const StreamGenerator&) = delete;
+  StreamGenerator& operator=(const StreamGenerator&) = delete;
+
+  // Add a new stream.
+  void AddStream(RtpStream* stream);
+
+  // Set the link capacity.
+  void set_capacity_bps(int capacity_bps);
+
+  // Divides `bitrate_bps` among all streams. The allocated bitrate per stream
+  // is decided by the initial allocation ratios.
+  void SetBitrateBps(int bitrate_bps);
+
+  // Set the RTP timestamp offset for the stream identified by `ssrc`.
+  void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset);
+
+  // TODO(holmer): Break out the channel simulation part from this class to make
+  // it possible to simulate different types of channels.
+  int64_t GenerateFrame(std::vector<PacketResult>* packets,
+                        int64_t time_now_us);
+
+ private:
+  // Capacity of the simulated channel in bits per second.
+  int capacity_;
+  // The time when the last packet arrived.
+  int64_t prev_arrival_time_us_;
+  // All streams being transmitted on this simulated channel.
+  std::vector<std::unique_ptr<RtpStream>> streams_;
+};
+}  // namespace test
+
+class DelayBasedBweTest : public ::testing::Test {
+ public:
+  DelayBasedBweTest();
+  explicit DelayBasedBweTest(absl::string_view field_trial_string);
+  ~DelayBasedBweTest() override;
+
+ protected:
+  void AddDefaultStream();
+
+  // Helpers to insert a single packet into the delay-based BWE.
+  void IncomingFeedback(int64_t arrival_time_ms,
+                        int64_t send_time_ms,
+                        size_t payload_size);
+  void IncomingFeedback(int64_t arrival_time_ms,
+                        int64_t send_time_ms,
+                        size_t payload_size,
+                        const PacedPacketInfo& pacing_info);
+  void IncomingFeedback(Timestamp receive_time,
+                        Timestamp send_time,
+                        size_t payload_size,
+                        const PacedPacketInfo& pacing_info);
+
+  // Generates a frame of packets belonging to a stream at a given bitrate and
+  // with a given ssrc. The stream is pushed through a very simple simulated
+  // network, and is then given to the receive-side bandwidth estimator.
+  // Returns true if an over-use was seen, false otherwise.
+  // The StreamGenerator::updated() should be used to check for any changes in
+  // target bitrate after the call to this function.
+  bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps);
+
+  // Run the bandwidth estimator with a stream of `number_of_frames` frames, or
+  // until it reaches `target_bitrate`.
+  // Can for instance be used to run the estimator for some time to get it
+  // into a steady state.
+  uint32_t SteadyStateRun(uint32_t ssrc,
+                          int number_of_frames,
+                          uint32_t start_bitrate,
+                          uint32_t min_bitrate,
+                          uint32_t max_bitrate,
+                          uint32_t target_bitrate);
+
+  void TestTimestampGroupingTestHelper();
+
+  void TestWrappingHelper(int silence_time_s);
+
+  void InitialBehaviorTestHelper(uint32_t expected_converge_bitrate);
+  void RateIncreaseReorderingTestHelper(uint32_t expected_bitrate);
+  void RateIncreaseRtpTimestampsTestHelper(int expected_iterations);
+  void CapacityDropTestHelper(int number_of_streams,
+                              bool wrap_time_stamp,
+                              uint32_t expected_bitrate_drop_delta,
+                              int64_t receiver_clock_offset_change_ms);
+
+  static const uint32_t kDefaultSsrc;
+  FieldTrialBasedConfig field_trial_config_;
+
+  std::unique_ptr<test::ScopedFieldTrials>
+      field_trial;        // Must be initialized first.
+  SimulatedClock clock_;  // Time at the receiver.
+  test::TestBitrateObserver bitrate_observer_;
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      acknowledged_bitrate_estimator_;
+  const std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
+  std::unique_ptr<DelayBasedBwe> bitrate_estimator_;
+  std::unique_ptr<test::StreamGenerator> stream_generator_;
+  int64_t arrival_time_offset_ms_;
+  bool first_update_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h
new file mode 100644
index 0000000000..fc12cff7d5
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
+
+#include <stdint.h>
+
+#include "api/network_state_predictor.h"
+
+namespace webrtc {
+
+class DelayIncreaseDetectorInterface {
+ public:
+  DelayIncreaseDetectorInterface() {}
+  virtual ~DelayIncreaseDetectorInterface() {}
+
+  DelayIncreaseDetectorInterface(const DelayIncreaseDetectorInterface&) =
+      delete;
+  DelayIncreaseDetectorInterface& operator=(
+      const DelayIncreaseDetectorInterface&) = delete;
+
+  // Update the detector with a new sample. The deltas should represent deltas
+  // between timestamp groups as defined by the InterArrival class.
+  virtual void Update(double recv_delta_ms,
+                      double send_delta_ms,
+                      int64_t send_time_ms,
+                      int64_t arrival_time_ms,
+                      size_t packet_size,
+                      bool calculated_deltas) = 0;
+
+  virtual BandwidthUsage State() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build
new file mode 100644
index 0000000000..b15062af41
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build
@@ -0,0 +1,238 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("estimators_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build
new file mode 100644
index 0000000000..973d4bf32f
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("goog_cc_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
new file mode 100644
index 0000000000..3a9de8c4dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
@@ -0,0 +1,725 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
+
+#include <inttypes.h>
+#include <stdio.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <numeric>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "api/network_state_predictor.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_remote_estimate.h"
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+// From RTCPSender video report interval.
+constexpr TimeDelta kLossUpdateInterval = TimeDelta::Millis(1000);
+
+// Pacing-rate relative to our target send rate.
+// Multiplicative factor that is applied to the target bitrate to calculate
+// the number of bytes that can be transmitted per interval.
+// Increasing this factor will result in lower delays in cases of bitrate
+// overshoots from the encoder.
+constexpr float kDefaultPaceMultiplier = 2.5f;
+
+// If the probe result is far below the current throughput estimate
+// it's unlikely that the probe is accurate, so we don't want to drop too far.
+// However, if we actually are overusing, we want to drop to something slightly
+// below the current throughput estimate to drain the network queues.
+constexpr double kProbeDropThroughputFraction = 0.85;
+
+bool IsEnabled(const FieldTrialsView* config, absl::string_view key) {
+  return absl::StartsWith(config->Lookup(key), "Enabled");
+}
+
+bool IsNotDisabled(const FieldTrialsView* config, absl::string_view key) {
+  return !absl::StartsWith(config->Lookup(key), "Disabled");
+}
+
+BandwidthLimitedCause GetBandwidthLimitedCause(
+    LossBasedState loss_based_state,
+    BandwidthUsage bandwidth_usage,
+    bool not_probe_if_delay_increased) {
+  if (not_probe_if_delay_increased &&
+      (bandwidth_usage == BandwidthUsage::kBwOverusing ||
+       bandwidth_usage == BandwidthUsage::kBwUnderusing)) {
+    return BandwidthLimitedCause::kDelayBasedLimitedDelayIncreased;
+  }
+  switch (loss_based_state) {
+    case LossBasedState::kDecreasing:
+      return BandwidthLimitedCause::kLossLimitedBweDecreasing;
+    case LossBasedState::kIncreasing:
+      return BandwidthLimitedCause::kLossLimitedBweIncreasing;
+    default:
+      return BandwidthLimitedCause::kDelayBasedLimited;
+  }
+}
+
+}  // namespace
+
+GoogCcNetworkController::GoogCcNetworkController(NetworkControllerConfig config,
+                                                 GoogCcConfig goog_cc_config)
+    : key_value_config_(config.key_value_config ? config.key_value_config
+                                                : &trial_based_config_),
+      event_log_(config.event_log),
+      packet_feedback_only_(goog_cc_config.feedback_only),
+      safe_reset_on_route_change_("Enabled"),
+      safe_reset_acknowledged_rate_("ack"),
+      use_min_allocatable_as_lower_bound_(
+          IsNotDisabled(key_value_config_, "WebRTC-Bwe-MinAllocAsLowerBound")),
+      ignore_probes_lower_than_network_estimate_(IsNotDisabled(
+          key_value_config_,
+          "WebRTC-Bwe-IgnoreProbesLowerThanNetworkStateEstimate")),
+      limit_probes_lower_than_throughput_estimate_(
+          IsEnabled(key_value_config_,
+                    "WebRTC-Bwe-LimitProbesLowerThanThroughputEstimate")),
+      rate_control_settings_(
+          RateControlSettings::ParseFromKeyValueConfig(key_value_config_)),
+      pace_at_max_of_bwe_and_lower_link_capacity_(
+          IsEnabled(key_value_config_,
+                    "WebRTC-Bwe-PaceAtMaxOfBweAndLowerLinkCapacity")),
+      probe_controller_(
+          new ProbeController(key_value_config_, config.event_log)),
+      congestion_window_pushback_controller_(
+          rate_control_settings_.UseCongestionWindowPushback()
+              ? std::make_unique<CongestionWindowPushbackController>(
+                    key_value_config_)
+              : nullptr),
+      bandwidth_estimation_(
+          std::make_unique<SendSideBandwidthEstimation>(key_value_config_,
+                                                        event_log_)),
+      alr_detector_(
+          std::make_unique<AlrDetector>(key_value_config_, config.event_log)),
+      probe_bitrate_estimator_(new ProbeBitrateEstimator(config.event_log)),
+      network_estimator_(std::move(goog_cc_config.network_state_estimator)),
+      network_state_predictor_(
+          std::move(goog_cc_config.network_state_predictor)),
+      delay_based_bwe_(new DelayBasedBwe(key_value_config_,
+                                         event_log_,
+                                         network_state_predictor_.get())),
+      acknowledged_bitrate_estimator_(
+          AcknowledgedBitrateEstimatorInterface::Create(key_value_config_)),
+      initial_config_(config),
+      last_loss_based_target_rate_(*config.constraints.starting_rate),
+      last_pushback_target_rate_(last_loss_based_target_rate_),
+      last_stable_target_rate_(last_loss_based_target_rate_),
+      pacing_factor_(config.stream_based_config.pacing_factor.value_or(
+          kDefaultPaceMultiplier)),
+      min_total_allocated_bitrate_(
+          config.stream_based_config.min_total_allocated_bitrate.value_or(
+              DataRate::Zero())),
+      max_padding_rate_(config.stream_based_config.max_padding_rate.value_or(
+          DataRate::Zero())) {
+  RTC_DCHECK(config.constraints.at_time.IsFinite());
+  ParseFieldTrial(
+      {&safe_reset_on_route_change_, &safe_reset_acknowledged_rate_},
+      key_value_config_->Lookup("WebRTC-Bwe-SafeResetOnRouteChange"));
+  if (delay_based_bwe_)
+    delay_based_bwe_->SetMinBitrate(kCongestionControllerMinBitrate);
+}
+
+GoogCcNetworkController::~GoogCcNetworkController() {}
+
+NetworkControlUpdate GoogCcNetworkController::OnNetworkAvailability(
+    NetworkAvailability msg) {
+  NetworkControlUpdate update;
+  update.probe_cluster_configs = probe_controller_->OnNetworkAvailability(msg);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnNetworkRouteChange(
+    NetworkRouteChange msg) {
+  if (safe_reset_on_route_change_) {
+    absl::optional<DataRate> estimated_bitrate;
+    if (safe_reset_acknowledged_rate_) {
+      estimated_bitrate = acknowledged_bitrate_estimator_->bitrate();
+      if (!estimated_bitrate)
+        estimated_bitrate = acknowledged_bitrate_estimator_->PeekRate();
+    } else {
+      estimated_bitrate = bandwidth_estimation_->target_rate();
+    }
+    if (estimated_bitrate) {
+      if (msg.constraints.starting_rate) {
+        msg.constraints.starting_rate =
+            std::min(*msg.constraints.starting_rate, *estimated_bitrate);
+      } else {
+        msg.constraints.starting_rate = estimated_bitrate;
+      }
+    }
+  }
+
+  acknowledged_bitrate_estimator_ =
+      AcknowledgedBitrateEstimatorInterface::Create(key_value_config_);
+  probe_bitrate_estimator_.reset(new ProbeBitrateEstimator(event_log_));
+  if (network_estimator_)
+    network_estimator_->OnRouteChange(msg);
+  delay_based_bwe_.reset(new DelayBasedBwe(key_value_config_, event_log_,
+                                           network_state_predictor_.get()));
+  bandwidth_estimation_->OnRouteChange();
+  probe_controller_->Reset(msg.at_time);
+  NetworkControlUpdate update;
+  update.probe_cluster_configs = ResetConstraints(msg.constraints);
+  MaybeTriggerOnNetworkChanged(&update, msg.at_time);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnProcessInterval(
+    ProcessInterval msg) {
+  NetworkControlUpdate update;
+  if (initial_config_) {
+    update.probe_cluster_configs =
+        ResetConstraints(initial_config_->constraints);
+    update.pacer_config = GetPacingRates(msg.at_time);
+
+    if (initial_config_->stream_based_config.requests_alr_probing) {
+      probe_controller_->EnablePeriodicAlrProbing(
+          *initial_config_->stream_based_config.requests_alr_probing);
+    }
+    absl::optional<DataRate> total_bitrate =
+        initial_config_->stream_based_config.max_total_allocated_bitrate;
+    if (total_bitrate) {
+      auto probes = probe_controller_->OnMaxTotalAllocatedBitrate(
+          *total_bitrate, msg.at_time);
+      update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
+                                          probes.begin(), probes.end());
+    }
+    initial_config_.reset();
+  }
+  if (congestion_window_pushback_controller_ && msg.pacer_queue) {
+    congestion_window_pushback_controller_->UpdatePacingQueue(
+        msg.pacer_queue->bytes());
+  }
+  bandwidth_estimation_->UpdateEstimate(msg.at_time);
+  absl::optional<int64_t> start_time_ms =
+      alr_detector_->GetApplicationLimitedRegionStartTime();
+  probe_controller_->SetAlrStartTimeMs(start_time_ms);
+
+  auto probes = probe_controller_->Process(msg.at_time);
+  update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
+                                      probes.begin(), probes.end());
+
+  if (rate_control_settings_.UseCongestionWindow() &&
+      last_packet_received_time_.IsFinite() && !feedback_max_rtts_.empty()) {
+    UpdateCongestionWindowSize();
+  }
+  if (congestion_window_pushback_controller_ && current_data_window_) {
+    congestion_window_pushback_controller_->SetDataWindow(
+        *current_data_window_);
+  } else {
+    update.congestion_window = current_data_window_;
+  }
+  MaybeTriggerOnNetworkChanged(&update, msg.at_time);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnRemoteBitrateReport(
+    RemoteBitrateReport msg) {
+  if (packet_feedback_only_) {
+    RTC_LOG(LS_ERROR) << "Received REMB for packet feedback only GoogCC";
+    return NetworkControlUpdate();
+  }
+  bandwidth_estimation_->UpdateReceiverEstimate(msg.receive_time,
+                                                msg.bandwidth);
+  BWE_TEST_LOGGING_PLOT(1, "REMB_kbps", msg.receive_time.ms(),
+                        msg.bandwidth.bps() / 1000);
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnRoundTripTimeUpdate(
+    RoundTripTimeUpdate msg) {
+  if (packet_feedback_only_ || msg.smoothed)
+    return NetworkControlUpdate();
+  RTC_DCHECK(!msg.round_trip_time.IsZero());
+  if (delay_based_bwe_)
+    delay_based_bwe_->OnRttUpdate(msg.round_trip_time);
+  bandwidth_estimation_->UpdateRtt(msg.round_trip_time, msg.receive_time);
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnSentPacket(
+    SentPacket sent_packet) {
+  alr_detector_->OnBytesSent(sent_packet.size.bytes(),
+                             sent_packet.send_time.ms());
+  acknowledged_bitrate_estimator_->SetAlr(
+      alr_detector_->GetApplicationLimitedRegionStartTime().has_value());
+
+  if (!first_packet_sent_) {
+    first_packet_sent_ = true;
+    // Initialize feedback time to send time to allow estimation of RTT until
+    // first feedback is received.
+    bandwidth_estimation_->UpdatePropagationRtt(sent_packet.send_time,
+                                                TimeDelta::Zero());
+  }
+  bandwidth_estimation_->OnSentPacket(sent_packet);
+
+  if (congestion_window_pushback_controller_) {
+    congestion_window_pushback_controller_->UpdateOutstandingData(
+        sent_packet.data_in_flight.bytes());
+    NetworkControlUpdate update;
+    MaybeTriggerOnNetworkChanged(&update, sent_packet.send_time);
+    return update;
+  } else {
+    return NetworkControlUpdate();
+  }
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnReceivedPacket(
+    ReceivedPacket received_packet) {
+  last_packet_received_time_ = received_packet.receive_time;
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnStreamsConfig(
+    StreamsConfig msg) {
+  NetworkControlUpdate update;
+  if (msg.requests_alr_probing) {
+    probe_controller_->EnablePeriodicAlrProbing(*msg.requests_alr_probing);
+  }
+  if (msg.max_total_allocated_bitrate) {
+    update.probe_cluster_configs =
+        probe_controller_->OnMaxTotalAllocatedBitrate(
+            *msg.max_total_allocated_bitrate, msg.at_time);
+  }
+
+  bool pacing_changed = false;
+  if (msg.pacing_factor && *msg.pacing_factor != pacing_factor_) {
+    pacing_factor_ = *msg.pacing_factor;
+    pacing_changed = true;
+  }
+  if (msg.min_total_allocated_bitrate &&
+      *msg.min_total_allocated_bitrate != min_total_allocated_bitrate_) {
+    min_total_allocated_bitrate_ = *msg.min_total_allocated_bitrate;
+    pacing_changed = true;
+
+    if (use_min_allocatable_as_lower_bound_) {
+      ClampConstraints();
+      delay_based_bwe_->SetMinBitrate(min_data_rate_);
+      bandwidth_estimation_->SetMinMaxBitrate(min_data_rate_, max_data_rate_);
+    }
+  }
+  if (msg.max_padding_rate && *msg.max_padding_rate != max_padding_rate_) {
+    max_padding_rate_ = *msg.max_padding_rate;
+    pacing_changed = true;
+  }
+
+  if (pacing_changed)
+    update.pacer_config = GetPacingRates(msg.at_time);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnTargetRateConstraints(
+    TargetRateConstraints constraints) {
+  NetworkControlUpdate update;
+  update.probe_cluster_configs = ResetConstraints(constraints);
+  MaybeTriggerOnNetworkChanged(&update, constraints.at_time);
+  return update;
+}
+
+void GoogCcNetworkController::ClampConstraints() {
+  // TODO(holmer): We should make sure the default bitrates are set to 10 kbps,
+  // and that we don't try to set the min bitrate to 0 from any applications.
+  // The congestion controller should allow a min bitrate of 0.
+  min_data_rate_ = std::max(min_target_rate_, kCongestionControllerMinBitrate);
+  if (use_min_allocatable_as_lower_bound_) {
+    min_data_rate_ = std::max(min_data_rate_, min_total_allocated_bitrate_);
+  }
+  if (max_data_rate_ < min_data_rate_) {
+    RTC_LOG(LS_WARNING) << "max bitrate smaller than min bitrate";
+    max_data_rate_ = min_data_rate_;
+  }
+  if (starting_rate_ && starting_rate_ < min_data_rate_) {
+    RTC_LOG(LS_WARNING) << "start bitrate smaller than min bitrate";
+    starting_rate_ = min_data_rate_;
+  }
+}
+
+std::vector<ProbeClusterConfig> GoogCcNetworkController::ResetConstraints(
+    TargetRateConstraints new_constraints) {
+  min_target_rate_ = new_constraints.min_data_rate.value_or(DataRate::Zero());
+  max_data_rate_ =
+      new_constraints.max_data_rate.value_or(DataRate::PlusInfinity());
+  starting_rate_ = new_constraints.starting_rate;
+  ClampConstraints();
+
+  bandwidth_estimation_->SetBitrates(starting_rate_, min_data_rate_,
+                                     max_data_rate_, new_constraints.at_time);
+
+  if (starting_rate_)
+    delay_based_bwe_->SetStartBitrate(*starting_rate_);
+  delay_based_bwe_->SetMinBitrate(min_data_rate_);
+
+  return probe_controller_->SetBitrates(
+      min_data_rate_, starting_rate_.value_or(DataRate::Zero()), max_data_rate_,
+      new_constraints.at_time);
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnTransportLossReport(
+    TransportLossReport msg) {
+  if (packet_feedback_only_)
+    return NetworkControlUpdate();
+  int64_t total_packets_delta =
+      msg.packets_received_delta + msg.packets_lost_delta;
+  bandwidth_estimation_->UpdatePacketsLost(
+      msg.packets_lost_delta, total_packets_delta, msg.receive_time);
+  return NetworkControlUpdate();
+}
+
+void GoogCcNetworkController::UpdateCongestionWindowSize() {
+  TimeDelta min_feedback_max_rtt = TimeDelta::Millis(
+      *std::min_element(feedback_max_rtts_.begin(), feedback_max_rtts_.end()));
+
+  const DataSize kMinCwnd = DataSize::Bytes(2 * 1500);
+  TimeDelta time_window =
+      min_feedback_max_rtt +
+      TimeDelta::Millis(
+          rate_control_settings_.GetCongestionWindowAdditionalTimeMs());
+
+  DataSize data_window = last_loss_based_target_rate_ * time_window;
+  if (current_data_window_) {
+    data_window =
+        std::max(kMinCwnd, (data_window + current_data_window_.value()) / 2);
+  } else {
+    data_window = std::max(kMinCwnd, data_window);
+  }
+  current_data_window_ = data_window;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
+    TransportPacketsFeedback report) {
+  if (report.packet_feedbacks.empty()) {
+    // TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
+    // against building very large network queues.
+    return NetworkControlUpdate();
+  }
+
+  if (congestion_window_pushback_controller_) {
+    congestion_window_pushback_controller_->UpdateOutstandingData(
+        report.data_in_flight.bytes());
+  }
+  TimeDelta max_feedback_rtt = TimeDelta::MinusInfinity();
+  TimeDelta min_propagation_rtt = TimeDelta::PlusInfinity();
+  Timestamp max_recv_time = Timestamp::MinusInfinity();
+
+  std::vector<PacketResult> feedbacks = report.ReceivedWithSendInfo();
+  for (const auto& feedback : feedbacks)
+    max_recv_time = std::max(max_recv_time, feedback.receive_time);
+
+  for (const auto& feedback : feedbacks) {
+    TimeDelta feedback_rtt =
+        report.feedback_time - feedback.sent_packet.send_time;
+    TimeDelta min_pending_time = max_recv_time - feedback.receive_time;
+    TimeDelta propagation_rtt = feedback_rtt - min_pending_time;
+    max_feedback_rtt = std::max(max_feedback_rtt, feedback_rtt);
+    min_propagation_rtt = std::min(min_propagation_rtt, propagation_rtt);
+  }
+
+  if (max_feedback_rtt.IsFinite()) {
+    feedback_max_rtts_.push_back(max_feedback_rtt.ms());
+    const size_t kMaxFeedbackRttWindow = 32;
+    if (feedback_max_rtts_.size() > kMaxFeedbackRttWindow)
+      feedback_max_rtts_.pop_front();
+    // TODO(srte): Use time since last unacknowledged packet.
+    bandwidth_estimation_->UpdatePropagationRtt(report.feedback_time,
+                                                min_propagation_rtt);
+  }
+  if (packet_feedback_only_) {
+    if (!feedback_max_rtts_.empty()) {
+      int64_t sum_rtt_ms =
+          std::accumulate(feedback_max_rtts_.begin(), feedback_max_rtts_.end(),
+                          static_cast<int64_t>(0));
+      int64_t mean_rtt_ms = sum_rtt_ms / feedback_max_rtts_.size();
+      if (delay_based_bwe_)
+        delay_based_bwe_->OnRttUpdate(TimeDelta::Millis(mean_rtt_ms));
+    }
+
+    TimeDelta feedback_min_rtt = TimeDelta::PlusInfinity();
+    for (const auto& packet_feedback : feedbacks) {
+      TimeDelta pending_time = packet_feedback.receive_time - max_recv_time;
+      TimeDelta rtt = report.feedback_time -
+                      packet_feedback.sent_packet.send_time - pending_time;
+      // Value used for predicting NACK round trip time in FEC controller.
+      feedback_min_rtt = std::min(rtt, feedback_min_rtt);
+    }
+    if (feedback_min_rtt.IsFinite()) {
+      bandwidth_estimation_->UpdateRtt(feedback_min_rtt, report.feedback_time);
+    }
+
+    expected_packets_since_last_loss_update_ +=
+        report.PacketsWithFeedback().size();
+    for (const auto& packet_feedback : report.PacketsWithFeedback()) {
+      if (!packet_feedback.IsReceived())
+        lost_packets_since_last_loss_update_ += 1;
+    }
+    if (report.feedback_time > next_loss_update_) {
+      next_loss_update_ = report.feedback_time + kLossUpdateInterval;
+      bandwidth_estimation_->UpdatePacketsLost(
+          lost_packets_since_last_loss_update_,
+          expected_packets_since_last_loss_update_, report.feedback_time);
+      expected_packets_since_last_loss_update_ = 0;
+      lost_packets_since_last_loss_update_ = 0;
+    }
+  }
+  absl::optional<int64_t> alr_start_time =
+      alr_detector_->GetApplicationLimitedRegionStartTime();
+
+  if (previously_in_alr_ && !alr_start_time.has_value()) {
+    int64_t now_ms = report.feedback_time.ms();
+    acknowledged_bitrate_estimator_->SetAlrEndedTime(report.feedback_time);
+    probe_controller_->SetAlrEndedTimeMs(now_ms);
+  }
+  previously_in_alr_ = alr_start_time.has_value();
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
+      report.SortedByReceiveTime());
+  auto acknowledged_bitrate = acknowledged_bitrate_estimator_->bitrate();
+  bandwidth_estimation_->SetAcknowledgedRate(acknowledged_bitrate,
+                                             report.feedback_time);
+  for (const auto& feedback : report.SortedByReceiveTime()) {
+    if (feedback.sent_packet.pacing_info.probe_cluster_id !=
+        PacedPacketInfo::kNotAProbe) {
+      probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(feedback);
+    }
+  }
+
+  if (network_estimator_) {
+    network_estimator_->OnTransportPacketsFeedback(report);
+    auto prev_estimate = estimate_;
+    estimate_ = network_estimator_->GetCurrentEstimate();
+    // TODO(srte): Make OnTransportPacketsFeedback signal whether the state
+    // changed to avoid the need for this check.
+    if (estimate_ && (!prev_estimate || estimate_->last_feed_time !=
+                                            prev_estimate->last_feed_time)) {
+      event_log_->Log(std::make_unique<RtcEventRemoteEstimate>(
+          estimate_->link_capacity_lower, estimate_->link_capacity_upper));
+      probe_controller_->SetNetworkStateEstimate(*estimate_);
+    }
+  }
+  absl::optional<DataRate> probe_bitrate =
+      probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate();
+  if (ignore_probes_lower_than_network_estimate_ && probe_bitrate &&
+      estimate_ && *probe_bitrate < delay_based_bwe_->last_estimate() &&
+      *probe_bitrate < estimate_->link_capacity_lower) {
+    probe_bitrate.reset();
+  }
+  if (limit_probes_lower_than_throughput_estimate_ && probe_bitrate &&
+      acknowledged_bitrate) {
+    // Limit the backoff to something slightly below the acknowledged
+    // bitrate. ("Slightly below" because we want to drain the queues
+    // if we are actually overusing.)
+    // The acknowledged bitrate shouldn't normally be higher than the delay
+    // based estimate, but it could happen e.g. due to packet bursts or
+    // encoder overshoot. We use std::min to ensure that a probe result
+    // below the current BWE never causes an increase.
+    DataRate limit =
+        std::min(delay_based_bwe_->last_estimate(),
+                 *acknowledged_bitrate * kProbeDropThroughputFraction);
+    probe_bitrate = std::max(*probe_bitrate, limit);
+  }
+
+  NetworkControlUpdate update;
+  bool recovered_from_overuse = false;
+
+  DelayBasedBwe::Result result;
+  result = delay_based_bwe_->IncomingPacketFeedbackVector(
+      report, acknowledged_bitrate, probe_bitrate, estimate_,
+      alr_start_time.has_value());
+
+  if (result.updated) {
+    if (result.probe) {
+      bandwidth_estimation_->SetSendBitrate(result.target_bitrate,
+                                            report.feedback_time);
+    }
+    // Since SetSendBitrate now resets the delay-based estimate, we have to
+    // call UpdateDelayBasedEstimate after SetSendBitrate.
+    bandwidth_estimation_->UpdateDelayBasedEstimate(report.feedback_time,
+                                                    result.target_bitrate);
+  }
+  bandwidth_estimation_->UpdateLossBasedEstimator(
+      report, result.delay_detector_state, probe_bitrate,
+      estimate_ ? estimate_->link_capacity_upper : DataRate::PlusInfinity());
+  if (result.updated) {
+    // Update the estimate in the ProbeController, in case we want to probe.
+    MaybeTriggerOnNetworkChanged(&update, report.feedback_time);
+  }
+
+  recovered_from_overuse = result.recovered_from_overuse;
+
+  if (recovered_from_overuse) {
+    probe_controller_->SetAlrStartTimeMs(alr_start_time);
+    auto probes = probe_controller_->RequestProbe(report.feedback_time);
+    update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
+                                        probes.begin(), probes.end());
+  }
+
+  // No valid RTT could be because send-side BWE isn't used, in which case
+  // we don't try to limit the outstanding packets.
+  if (rate_control_settings_.UseCongestionWindow() &&
+      max_feedback_rtt.IsFinite()) {
+    UpdateCongestionWindowSize();
+  }
+  if (congestion_window_pushback_controller_ && current_data_window_) {
+    congestion_window_pushback_controller_->SetDataWindow(
+        *current_data_window_);
+  } else {
+    update.congestion_window = current_data_window_;
+  }
+
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnNetworkStateEstimate(
+    NetworkStateEstimate msg) {
+  estimate_ = msg;
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::GetNetworkState(
+    Timestamp at_time) const {
+  NetworkControlUpdate update;
+  update.target_rate = TargetTransferRate();
+  update.target_rate->network_estimate.at_time = at_time;
+  update.target_rate->network_estimate.loss_rate_ratio =
+      last_estimated_fraction_loss_.value_or(0) / 255.0;
+  update.target_rate->network_estimate.round_trip_time =
+      last_estimated_round_trip_time_;
+  update.target_rate->network_estimate.bwe_period =
+      delay_based_bwe_->GetExpectedBwePeriod();
+
+  update.target_rate->at_time = at_time;
+  update.target_rate->target_rate = last_pushback_target_rate_;
+  update.target_rate->stable_target_rate =
+      bandwidth_estimation_->GetEstimatedLinkCapacity();
+  update.pacer_config = GetPacingRates(at_time);
+  update.congestion_window = current_data_window_;
+  return update;
+}
+
+void GoogCcNetworkController::MaybeTriggerOnNetworkChanged(
+    NetworkControlUpdate* update,
+    Timestamp at_time) {
+  uint8_t fraction_loss = bandwidth_estimation_->fraction_loss();
+  TimeDelta round_trip_time = bandwidth_estimation_->round_trip_time();
+  DataRate loss_based_target_rate = bandwidth_estimation_->target_rate();
+  DataRate pushback_target_rate = loss_based_target_rate;
+
+  BWE_TEST_LOGGING_PLOT(1, "fraction_loss_%", at_time.ms(),
+                        (fraction_loss * 100) / 256);
+  BWE_TEST_LOGGING_PLOT(1, "rtt_ms", at_time.ms(), round_trip_time.ms());
+  BWE_TEST_LOGGING_PLOT(1, "Target_bitrate_kbps", at_time.ms(),
+                        loss_based_target_rate.kbps());
+
+  double cwnd_reduce_ratio = 0.0;
+  if (congestion_window_pushback_controller_) {
+    int64_t pushback_rate =
+        congestion_window_pushback_controller_->UpdateTargetBitrate(
+            loss_based_target_rate.bps());
+    pushback_rate = std::max<int64_t>(bandwidth_estimation_->GetMinBitrate(),
+                                      pushback_rate);
+    pushback_target_rate = DataRate::BitsPerSec(pushback_rate);
+    if (rate_control_settings_.UseCongestionWindowDropFrameOnly()) {
+      cwnd_reduce_ratio = static_cast<double>(loss_based_target_rate.bps() -
+                                              pushback_target_rate.bps()) /
+                          loss_based_target_rate.bps();
+    }
+  }
+  DataRate stable_target_rate =
+      bandwidth_estimation_->GetEstimatedLinkCapacity();
+  stable_target_rate = std::min(stable_target_rate, pushback_target_rate);
+
+  if ((loss_based_target_rate != last_loss_based_target_rate_) ||
+      (fraction_loss != last_estimated_fraction_loss_) ||
+      (round_trip_time != last_estimated_round_trip_time_) ||
+      (pushback_target_rate != last_pushback_target_rate_) ||
+      (stable_target_rate != last_stable_target_rate_)) {
+    last_loss_based_target_rate_ = loss_based_target_rate;
+    last_pushback_target_rate_ = pushback_target_rate;
+    last_estimated_fraction_loss_ = fraction_loss;
+    last_estimated_round_trip_time_ = round_trip_time;
+    last_stable_target_rate_ = stable_target_rate;
+
+    alr_detector_->SetEstimatedBitrate(loss_based_target_rate.bps());
+
+    TimeDelta bwe_period = delay_based_bwe_->GetExpectedBwePeriod();
+
+    TargetTransferRate target_rate_msg;
+    target_rate_msg.at_time = at_time;
+    if (rate_control_settings_.UseCongestionWindowDropFrameOnly()) {
+      target_rate_msg.target_rate = loss_based_target_rate;
+      target_rate_msg.cwnd_reduce_ratio = cwnd_reduce_ratio;
+    } else {
+      target_rate_msg.target_rate = pushback_target_rate;
+    }
+    target_rate_msg.stable_target_rate = stable_target_rate;
+    target_rate_msg.network_estimate.at_time = at_time;
+    target_rate_msg.network_estimate.round_trip_time = round_trip_time;
+    target_rate_msg.network_estimate.loss_rate_ratio = fraction_loss / 255.0f;
+    target_rate_msg.network_estimate.bwe_period = bwe_period;
+
+    update->target_rate = target_rate_msg;
+
+    auto probes = probe_controller_->SetEstimatedBitrate(
+        loss_based_target_rate,
+        GetBandwidthLimitedCause(
+            bandwidth_estimation_->loss_based_state(),
+            delay_based_bwe_->last_state(),
+            probe_controller_->DontProbeIfDelayIncreased()),
+        at_time);
+    update->probe_cluster_configs.insert(update->probe_cluster_configs.end(),
+                                         probes.begin(), probes.end());
+    update->pacer_config = GetPacingRates(at_time);
+    RTC_LOG(LS_VERBOSE) << "bwe " << at_time.ms() << " pushback_target_bps="
+                        << last_pushback_target_rate_.bps()
+                        << " estimate_bps=" << loss_based_target_rate.bps();
+  }
+}
+
+PacerConfig GoogCcNetworkController::GetPacingRates(Timestamp at_time) const {
+  // Pacing rate is based on target rate before congestion window pushback,
+  // because we don't want to build queues in the pacer when pushback occurs.
+  DataRate pacing_rate = DataRate::Zero();
+  if (pace_at_max_of_bwe_and_lower_link_capacity_ && estimate_) {
+    pacing_rate =
+        std::max({min_total_allocated_bitrate_, estimate_->link_capacity_lower,
+                  last_loss_based_target_rate_}) *
+        pacing_factor_;
+  } else {
+    pacing_rate =
+        std::max(min_total_allocated_bitrate_, last_loss_based_target_rate_) *
+        pacing_factor_;
+  }
+  DataRate padding_rate =
+      std::min(max_padding_rate_, last_pushback_target_rate_);
+  PacerConfig msg;
+  msg.at_time = at_time;
+  msg.time_window = TimeDelta::Seconds(1);
+  msg.data_window = pacing_rate * msg.time_window;
+  msg.pad_window = padding_rate * msg.time_window;
+  return msg;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h
new file mode 100644
index 0000000000..37a064e37c
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
+
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+
+namespace webrtc {
+struct GoogCcConfig {
+  std::unique_ptr<NetworkStateEstimator> network_state_estimator = nullptr;
+  std::unique_ptr<NetworkStatePredictor> network_state_predictor = nullptr;
+  bool feedback_only = false;
+};
+
+class GoogCcNetworkController : public NetworkControllerInterface {
+ public:
+  GoogCcNetworkController(NetworkControllerConfig config,
+                          GoogCcConfig goog_cc_config);
+
+  GoogCcNetworkController() = delete;
+  GoogCcNetworkController(const GoogCcNetworkController&) = delete;
+  GoogCcNetworkController& operator=(const GoogCcNetworkController&) = delete;
+
+  ~GoogCcNetworkController() override;
+
+  // NetworkControllerInterface
+  NetworkControlUpdate OnNetworkAvailability(NetworkAvailability msg) override;
+  NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange msg) override;
+  NetworkControlUpdate OnProcessInterval(ProcessInterval msg) override;
+  NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport msg) override;
+  NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override;
+  NetworkControlUpdate OnSentPacket(SentPacket msg) override;
+  NetworkControlUpdate OnReceivedPacket(ReceivedPacket msg) override;
+  NetworkControlUpdate OnStreamsConfig(StreamsConfig msg) override;
+  NetworkControlUpdate OnTargetRateConstraints(
+      TargetRateConstraints msg) override;
+  NetworkControlUpdate OnTransportLossReport(TransportLossReport msg) override;
+  NetworkControlUpdate OnTransportPacketsFeedback(
+      TransportPacketsFeedback msg) override;
+  NetworkControlUpdate OnNetworkStateEstimate(
+      NetworkStateEstimate msg) override;
+
+  NetworkControlUpdate GetNetworkState(Timestamp at_time) const;
+
+ private:
+  friend class GoogCcStatePrinter;
+  std::vector<ProbeClusterConfig> ResetConstraints(
+      TargetRateConstraints new_constraints);
+  void ClampConstraints();
+  void MaybeTriggerOnNetworkChanged(NetworkControlUpdate* update,
+                                    Timestamp at_time);
+  void UpdateCongestionWindowSize();
+  PacerConfig GetPacingRates(Timestamp at_time) const;
+  const FieldTrialBasedConfig trial_based_config_;
+
+  const FieldTrialsView* const key_value_config_;
+  RtcEventLog* const event_log_;
+  const bool packet_feedback_only_;
+  FieldTrialFlag safe_reset_on_route_change_;
+  FieldTrialFlag safe_reset_acknowledged_rate_;
+  const bool use_min_allocatable_as_lower_bound_;
+  const bool ignore_probes_lower_than_network_estimate_;
+  const bool limit_probes_lower_than_throughput_estimate_;
+  const RateControlSettings rate_control_settings_;
+  const bool pace_at_max_of_bwe_and_lower_link_capacity_;
+
+  const std::unique_ptr<ProbeController> probe_controller_;
+  const std::unique_ptr<CongestionWindowPushbackController>
+      congestion_window_pushback_controller_;
+
+  std::unique_ptr<SendSideBandwidthEstimation> bandwidth_estimation_;
+  std::unique_ptr<AlrDetector> alr_detector_;
+  std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
+  std::unique_ptr<NetworkStateEstimator> network_estimator_;
+  std::unique_ptr<NetworkStatePredictor> network_state_predictor_;
+  std::unique_ptr<DelayBasedBwe> delay_based_bwe_;
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      acknowledged_bitrate_estimator_;
+
+  absl::optional<NetworkControllerConfig> initial_config_;
+
+  DataRate min_target_rate_ = DataRate::Zero();
+  DataRate min_data_rate_ = DataRate::Zero();
+  DataRate max_data_rate_ = DataRate::PlusInfinity();
+  absl::optional<DataRate> starting_rate_;
+
+  bool first_packet_sent_ = false;
+
+  absl::optional<NetworkStateEstimate> estimate_;
+
+  Timestamp next_loss_update_ = Timestamp::MinusInfinity();
+  int lost_packets_since_last_loss_update_ = 0;
+  int expected_packets_since_last_loss_update_ = 0;
+
+  std::deque<int64_t> feedback_max_rtts_;
+
+  DataRate last_loss_based_target_rate_;
+  DataRate last_pushback_target_rate_;
+  DataRate last_stable_target_rate_;
+
+  absl::optional<uint8_t> last_estimated_fraction_loss_ = 0;
+  TimeDelta last_estimated_round_trip_time_ = TimeDelta::PlusInfinity();
+  Timestamp last_packet_received_time_ = Timestamp::MinusInfinity();
+
+  double pacing_factor_;
+  DataRate min_total_allocated_bitrate_;
+  DataRate max_padding_rate_;
+
+  bool previously_in_alr_ = false;
+
+  absl::optional<DataSize> current_data_window_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
new file mode 100644
index 0000000000..7e051f505b
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
@@ -0,0 +1,934 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <queue>
+
+#include "absl/strings/string_view.h"
+#include "api/test/network_emulation/create_cross_traffic.h"
+#include "api/test/network_emulation/cross_traffic.h"
+#include "api/transport/goog_cc_factory.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+using ::testing::NiceMock;
+
+namespace webrtc {
+namespace test {
+namespace {
+// Count dips from a constant high bandwidth level within a short window.
+int CountBandwidthDips(std::queue<DataRate> bandwidth_history,
+                       DataRate threshold) {
+  if (bandwidth_history.empty())
+    return true;
+  DataRate first = bandwidth_history.front();
+  bandwidth_history.pop();
+
+  int dips = 0;
+  bool state_high = true;
+  while (!bandwidth_history.empty()) {
+    if (bandwidth_history.front() + threshold < first && state_high) {
+      ++dips;
+      state_high = false;
+    } else if (bandwidth_history.front() == first) {
+      state_high = true;
+    } else if (bandwidth_history.front() > first) {
+      // If this is toggling we will catch it later when front becomes first.
+      state_high = false;
+    }
+    bandwidth_history.pop();
+  }
+  return dips;
+}
+GoogCcNetworkControllerFactory CreateFeedbackOnlyFactory() {
+  GoogCcFactoryConfig config;
+  config.feedback_only = true;
+  return GoogCcNetworkControllerFactory(std::move(config));
+}
+
+const uint32_t kInitialBitrateKbps = 60;
+const DataRate kInitialBitrate = DataRate::KilobitsPerSec(kInitialBitrateKbps);
+const float kDefaultPacingRate = 2.5f;
+
+CallClient* CreateVideoSendingClient(
+    Scenario* s,
+    CallClientConfig config,
+    std::vector<EmulatedNetworkNode*> send_link,
+    std::vector<EmulatedNetworkNode*> return_link) {
+  auto* client = s->CreateClient("send", std::move(config));
+  auto* route = s->CreateRoutes(client, send_link,
+                                s->CreateClient("return", CallClientConfig()),
+                                return_link);
+  s->CreateVideoStream(route->forward(), VideoStreamConfig());
+  return client;
+}
+
+NetworkRouteChange CreateRouteChange(
+    Timestamp time,
+    absl::optional<DataRate> start_rate = absl::nullopt,
+    absl::optional<DataRate> min_rate = absl::nullopt,
+    absl::optional<DataRate> max_rate = absl::nullopt) {
+  NetworkRouteChange route_change;
+  route_change.at_time = time;
+  route_change.constraints.at_time = time;
+  route_change.constraints.min_data_rate = min_rate;
+  route_change.constraints.max_data_rate = max_rate;
+  route_change.constraints.starting_rate = start_rate;
+  return route_change;
+}
+
+PacketResult CreatePacketResult(Timestamp arrival_time,
+                                Timestamp send_time,
+                                size_t payload_size,
+                                PacedPacketInfo pacing_info) {
+  PacketResult packet_result;
+  packet_result.sent_packet = SentPacket();
+  packet_result.sent_packet.send_time = send_time;
+  packet_result.sent_packet.size = DataSize::Bytes(payload_size);
+  packet_result.sent_packet.pacing_info = pacing_info;
+  packet_result.receive_time = arrival_time;
+  return packet_result;
+}
+
+// Simulate sending packets and receiving transport feedback during
+// `runtime_ms`.
+absl::optional<DataRate> PacketTransmissionAndFeedbackBlock(
+    NetworkControllerInterface* controller,
+    int64_t runtime_ms,
+    int64_t delay,
+    Timestamp& current_time) {
+  NetworkControlUpdate update;
+  absl::optional<DataRate> target_bitrate;
+  int64_t delay_buildup = 0;
+  int64_t start_time_ms = current_time.ms();
+  while (current_time.ms() - start_time_ms < runtime_ms) {
+    constexpr size_t kPayloadSize = 1000;
+    PacketResult packet =
+        CreatePacketResult(current_time + TimeDelta::Millis(delay_buildup),
+                           current_time, kPayloadSize, PacedPacketInfo());
+    delay_buildup += delay;
+    update = controller->OnSentPacket(packet.sent_packet);
+    if (update.target_rate) {
+      target_bitrate = update.target_rate->target_rate;
+    }
+    TransportPacketsFeedback feedback;
+    feedback.feedback_time = packet.receive_time;
+    feedback.packet_feedbacks.push_back(packet);
+    update = controller->OnTransportPacketsFeedback(feedback);
+    if (update.target_rate) {
+      target_bitrate = update.target_rate->target_rate;
+    }
+    current_time += TimeDelta::Millis(50);
+    update = controller->OnProcessInterval({.at_time = current_time});
+    if (update.target_rate) {
+      target_bitrate = update.target_rate->target_rate;
+    }
+  }
+  return target_bitrate;
+}
+
+// Scenarios:
+
+void UpdatesTargetRateBasedOnLinkCapacity(absl::string_view test_name = "") {
+  auto factory = CreateFeedbackOnlyFactory();
+  Scenario s("googcc_unit/target_capacity" + std::string(test_name), false);
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(1500);
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  auto send_net = s.CreateMutableSimulationNode([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(500);
+    c->delay = TimeDelta::Millis(100);
+    c->loss_rate = 0.0;
+  });
+  auto ret_net = s.CreateMutableSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  StatesPrinter* truth = s.CreatePrinter(
+      "send.truth.txt", TimeDelta::PlusInfinity(), {send_net->ConfigPrinter()});
+
+  auto* client = CreateVideoSendingClient(&s, config, {send_net->node()},
+                                          {ret_net->node()});
+
+  truth->PrintRow();
+  s.RunFor(TimeDelta::Seconds(25));
+  truth->PrintRow();
+  EXPECT_NEAR(client->target_rate().kbps(), 450, 100);
+
+  send_net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(800);
+    c->delay = TimeDelta::Millis(100);
+  });
+
+  truth->PrintRow();
+  s.RunFor(TimeDelta::Seconds(20));
+  truth->PrintRow();
+  EXPECT_NEAR(client->target_rate().kbps(), 750, 150);
+
+  send_net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(100);
+    c->delay = TimeDelta::Millis(200);
+  });
+  ret_net->UpdateConfig(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(200); });
+
+  truth->PrintRow();
+  s.RunFor(TimeDelta::Seconds(50));
+  truth->PrintRow();
+  EXPECT_NEAR(client->target_rate().kbps(), 90, 25);
+}
+
+DataRate RunRembDipScenario(absl::string_view test_name) {
+  Scenario s(test_name);
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(2000);
+  net_conf.delay = TimeDelta::Millis(50);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  });
+  auto send_net = {s.CreateSimulationNode(net_conf)};
+  auto ret_net = {s.CreateSimulationNode(net_conf)};
+  auto* route = s.CreateRoutes(
+      client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+
+  s.RunFor(TimeDelta::Seconds(10));
+  EXPECT_GT(client->send_bandwidth().kbps(), 1500);
+
+  DataRate RembLimit = DataRate::KilobitsPerSec(250);
+  client->SetRemoteBitrate(RembLimit);
+  s.RunFor(TimeDelta::Seconds(1));
+  EXPECT_EQ(client->send_bandwidth(), RembLimit);
+
+  DataRate RembLimitLifted = DataRate::KilobitsPerSec(10000);
+  client->SetRemoteBitrate(RembLimitLifted);
+  s.RunFor(TimeDelta::Seconds(10));
+
+  return client->send_bandwidth();
+}
+
+}  // namespace
+
+class NetworkControllerTestFixture {
+ public:
+  NetworkControllerTestFixture() : factory_() {}
+
+  std::unique_ptr<NetworkControllerInterface> CreateController() {
+    NetworkControllerConfig config = InitialConfig();
+    std::unique_ptr<NetworkControllerInterface> controller =
+        factory_.Create(config);
+    return controller;
+  }
+
+ private:
+  NetworkControllerConfig InitialConfig(
+      int starting_bandwidth_kbps = kInitialBitrateKbps,
+      int min_data_rate_kbps = 0,
+      int max_data_rate_kbps = 5 * kInitialBitrateKbps) {
+    NetworkControllerConfig config;
+    config.constraints.at_time = Timestamp::Zero();
+    config.constraints.min_data_rate =
+        DataRate::KilobitsPerSec(min_data_rate_kbps);
+    config.constraints.max_data_rate =
+        DataRate::KilobitsPerSec(max_data_rate_kbps);
+    config.constraints.starting_rate =
+        DataRate::KilobitsPerSec(starting_bandwidth_kbps);
+    config.event_log = &event_log_;
+    return config;
+  }
+
+  NiceMock<MockRtcEventLog> event_log_;
+  GoogCcNetworkControllerFactory factory_;
+};
+
+TEST(GoogCcNetworkControllerTest, InitializeTargetRateOnFirstProcessInterval) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+
+  NetworkControlUpdate update =
+      controller->OnProcessInterval({.at_time = Timestamp::Millis(123456)});
+
+  EXPECT_EQ(update.target_rate->target_rate, kInitialBitrate);
+  EXPECT_EQ(update.pacer_config->data_rate(),
+            kInitialBitrate * kDefaultPacingRate);
+  EXPECT_EQ(update.probe_cluster_configs[0].target_data_rate,
+            kInitialBitrate * 3);
+  EXPECT_EQ(update.probe_cluster_configs[1].target_data_rate,
+            kInitialBitrate * 5);
+}
+
+TEST(GoogCcNetworkControllerTest, ReactsToChangedNetworkConditions) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  Timestamp current_time = Timestamp::Millis(123);
+  NetworkControlUpdate update =
+      controller->OnProcessInterval({.at_time = current_time});
+  update = controller->OnRemoteBitrateReport(
+      {.receive_time = current_time, .bandwidth = kInitialBitrate * 2});
+
+  current_time += TimeDelta::Millis(25);
+  update = controller->OnProcessInterval({.at_time = current_time});
+  EXPECT_EQ(update.target_rate->target_rate, kInitialBitrate * 2);
+  EXPECT_EQ(update.pacer_config->data_rate(),
+            kInitialBitrate * 2 * kDefaultPacingRate);
+
+  update = controller->OnRemoteBitrateReport(
+      {.receive_time = current_time, .bandwidth = kInitialBitrate});
+  current_time += TimeDelta::Millis(25);
+  update = controller->OnProcessInterval({.at_time = current_time});
+  EXPECT_EQ(update.target_rate->target_rate, kInitialBitrate);
+  EXPECT_EQ(update.pacer_config->data_rate(),
+            kInitialBitrate * kDefaultPacingRate);
+}
+
+TEST(GoogCcNetworkControllerTest, OnNetworkRouteChanged) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  Timestamp current_time = Timestamp::Millis(123);
+  DataRate new_bitrate = DataRate::BitsPerSec(200000);
+  NetworkControlUpdate update = controller->OnNetworkRouteChange(
+      CreateRouteChange(current_time, new_bitrate));
+  EXPECT_EQ(update.target_rate->target_rate, new_bitrate);
+  EXPECT_EQ(update.pacer_config->data_rate(), new_bitrate * kDefaultPacingRate);
+  EXPECT_EQ(update.probe_cluster_configs.size(), 2u);
+
+  // If the bitrate is reset to -1, the new starting bitrate will be
+  // the minimum default bitrate.
+  const DataRate kDefaultMinBitrate = DataRate::KilobitsPerSec(5);
+  update = controller->OnNetworkRouteChange(CreateRouteChange(current_time));
+  EXPECT_EQ(update.target_rate->target_rate, kDefaultMinBitrate);
+  EXPECT_NEAR(update.pacer_config->data_rate().bps<double>(),
+              kDefaultMinBitrate.bps<double>() * kDefaultPacingRate, 10);
+  EXPECT_EQ(update.probe_cluster_configs.size(), 2u);
+}
+
+TEST(GoogCcNetworkControllerTest, ProbeOnRouteChange) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  Timestamp current_time = Timestamp::Millis(123);
+  NetworkControlUpdate update = controller->OnNetworkRouteChange(
+      CreateRouteChange(current_time, 2 * kInitialBitrate, DataRate::Zero(),
+                        20 * kInitialBitrate));
+
+  EXPECT_TRUE(update.pacer_config.has_value());
+  EXPECT_EQ(update.target_rate->target_rate, kInitialBitrate * 2);
+  EXPECT_EQ(update.probe_cluster_configs.size(), 2u);
+  EXPECT_EQ(update.probe_cluster_configs[0].target_data_rate,
+            kInitialBitrate * 6);
+  EXPECT_EQ(update.probe_cluster_configs[1].target_data_rate,
+            kInitialBitrate * 12);
+
+  update = controller->OnProcessInterval({.at_time = current_time});
+}
+
+// Bandwidth estimation is updated when feedbacks are received.
+// Feedbacks which show an increasing delay cause the estimation to be reduced.
+TEST(GoogCcNetworkControllerTest, UpdatesDelayBasedEstimate) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  const int64_t kRunTimeMs = 6000;
+  Timestamp current_time = Timestamp::Millis(123);
+
+  // The test must run and insert packets/feedback long enough that the
+  // BWE computes a valid estimate. This is first done in an environment which
+  // simulates no bandwidth limitation, and therefore not built-up delay.
+  absl::optional<DataRate> target_bitrate_before_delay =
+      PacketTransmissionAndFeedbackBlock(controller.get(), kRunTimeMs, 0,
+                                         current_time);
+  ASSERT_TRUE(target_bitrate_before_delay.has_value());
+
+  // Repeat, but this time with a building delay, and make sure that the
+  // estimation is adjusted downwards.
+  absl::optional<DataRate> target_bitrate_after_delay =
+      PacketTransmissionAndFeedbackBlock(controller.get(), kRunTimeMs, 50,
+                                         current_time);
+  EXPECT_LT(*target_bitrate_after_delay, *target_bitrate_before_delay);
+}
+
+TEST(GoogCcNetworkControllerTest, PaceAtMaxOfLowerLinkCapacityAndBwe) {
+  ScopedFieldTrials trial(
+      "WebRTC-Bwe-PaceAtMaxOfBweAndLowerLinkCapacity/Enabled/");
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  Timestamp current_time = Timestamp::Millis(123);
+  NetworkControlUpdate update =
+      controller->OnProcessInterval({.at_time = current_time});
+  current_time += TimeDelta::Millis(100);
+  NetworkStateEstimate network_estimate = {.link_capacity_lower =
+                                               10 * kInitialBitrate};
+  update = controller->OnNetworkStateEstimate(network_estimate);
+  // OnNetworkStateEstimate does not trigger processing a new estimate. So add a
+  // dummy loss report to trigger a BWE update in the next process interval.
+  TransportLossReport loss_report;
+  loss_report.start_time = current_time;
+  loss_report.end_time = current_time;
+  loss_report.receive_time = current_time;
+  loss_report.packets_received_delta = 50;
+  loss_report.packets_lost_delta = 1;
+  update = controller->OnTransportLossReport(loss_report);
+  update = controller->OnProcessInterval({.at_time = current_time});
+  ASSERT_TRUE(update.pacer_config);
+  ASSERT_TRUE(update.target_rate);
+  ASSERT_LT(update.target_rate->target_rate,
+            network_estimate.link_capacity_lower);
+  EXPECT_EQ(update.pacer_config->data_rate().kbps(),
+            network_estimate.link_capacity_lower.kbps() * kDefaultPacingRate);
+
+  current_time += TimeDelta::Millis(100);
+  // Set a low link capacity estimate and verify that pacing rate is set
+  // relative to loss based/delay based estimate.
+  network_estimate = {.link_capacity_lower = 0.5 * kInitialBitrate};
+  update = controller->OnNetworkStateEstimate(network_estimate);
+  // Again, we need to inject a dummy loss report to trigger an update of the
+  // BWE in the next process interval.
+  loss_report.start_time = current_time;
+  loss_report.end_time = current_time;
+  loss_report.receive_time = current_time;
+  loss_report.packets_received_delta = 50;
+  loss_report.packets_lost_delta = 0;
+  update = controller->OnTransportLossReport(loss_report);
+  update = controller->OnProcessInterval({.at_time = current_time});
+  ASSERT_TRUE(update.target_rate);
+  ASSERT_GT(update.target_rate->target_rate,
+            network_estimate.link_capacity_lower);
+  EXPECT_EQ(update.pacer_config->data_rate().kbps(),
+            update.target_rate->target_rate.kbps() * kDefaultPacingRate);
+}
+
+// Test congestion window pushback on network delay happens.
+TEST(GoogCcScenario, CongestionWindowPushbackOnNetworkDelay) {
+  auto factory = CreateFeedbackOnlyFactory();
+  ScopedFieldTrials trial(
+      "WebRTC-CongestionWindow/QueueSize:800,MinBitrate:30000/");
+  Scenario s("googcc_unit/cwnd_on_delay", false);
+  auto send_net =
+      s.CreateMutableSimulationNode([=](NetworkSimulationConfig* c) {
+        c->bandwidth = DataRate::KilobitsPerSec(1000);
+        c->delay = TimeDelta::Millis(100);
+      });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  // Start high so bandwidth drop has max effect.
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(2000);
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+
+  auto* client = CreateVideoSendingClient(&s, std::move(config),
+                                          {send_net->node()}, {ret_net});
+
+  s.RunFor(TimeDelta::Seconds(10));
+  send_net->PauseTransmissionUntil(s.Now() + TimeDelta::Seconds(10));
+  s.RunFor(TimeDelta::Seconds(3));
+
+  // After 3 seconds without feedback from any sent packets, we expect that the
+  // target rate is reduced to the minimum pushback threshold
+  // kDefaultMinPushbackTargetBitrateBps, which is defined as 30 kbps in
+  // congestion_window_pushback_controller.
+  EXPECT_LT(client->target_rate().kbps(), 40);
+}
+
+// Test congestion window pushback on network delay happens.
+TEST(GoogCcScenario, CongestionWindowPushbackDropFrameOnNetworkDelay) {
+  auto factory = CreateFeedbackOnlyFactory();
+  ScopedFieldTrials trial(
+      "WebRTC-CongestionWindow/QueueSize:800,MinBitrate:30000,DropFrame:true/");
+  Scenario s("googcc_unit/cwnd_on_delay", false);
+  auto send_net =
+      s.CreateMutableSimulationNode([=](NetworkSimulationConfig* c) {
+        c->bandwidth = DataRate::KilobitsPerSec(1000);
+        c->delay = TimeDelta::Millis(100);
+      });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  // Start high so bandwidth drop has max effect.
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(2000);
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+
+  auto* client = CreateVideoSendingClient(&s, std::move(config),
+                                          {send_net->node()}, {ret_net});
+
+  s.RunFor(TimeDelta::Seconds(10));
+  send_net->PauseTransmissionUntil(s.Now() + TimeDelta::Seconds(10));
+  s.RunFor(TimeDelta::Seconds(3));
+
+  // As the dropframe is set, after 3 seconds without feedback from any sent
+  // packets, we expect that the target rate is not reduced by congestion
+  // window.
+  EXPECT_GT(client->target_rate().kbps(), 300);
+}
+
+TEST(GoogCcScenario, PaddingRateLimitedByCongestionWindowInTrial) {
+  ScopedFieldTrials trial(
+      "WebRTC-CongestionWindow/QueueSize:200,MinBitrate:30000/");
+
+  Scenario s("googcc_unit/padding_limited", false);
+  auto send_net =
+      s.CreateMutableSimulationNode([=](NetworkSimulationConfig* c) {
+        c->bandwidth = DataRate::KilobitsPerSec(1000);
+        c->delay = TimeDelta::Millis(100);
+      });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  CallClientConfig config;
+  // Start high so bandwidth drop has max effect.
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(2000);
+  auto* client = s.CreateClient("send", config);
+  auto* route =
+      s.CreateRoutes(client, {send_net->node()},
+                     s.CreateClient("return", CallClientConfig()), {ret_net});
+  VideoStreamConfig video;
+  video.stream.pad_to_rate = config.transport.rates.max_rate;
+  s.CreateVideoStream(route->forward(), video);
+
+  // Run for a few seconds to allow the controller to stabilize.
+  s.RunFor(TimeDelta::Seconds(10));
+
+  // Check that padding rate matches target rate.
+  EXPECT_NEAR(client->padding_rate().kbps(), client->target_rate().kbps(), 1);
+
+  // Check this is also the case when congestion window pushback kicks in.
+  send_net->PauseTransmissionUntil(s.Now() + TimeDelta::Seconds(1));
+  EXPECT_NEAR(client->padding_rate().kbps(), client->target_rate().kbps(), 1);
+}
+
+TEST(GoogCcScenario, LimitsToFloorIfRttIsHighInTrial) {
+  // The field trial limits maximum RTT to 2 seconds, higher RTT means that the
+  // controller backs off until it reaches the minimum configured bitrate. This
+  // allows the RTT to recover faster than the regular control mechanism would
+  // achieve.
+  const DataRate kBandwidthFloor = DataRate::KilobitsPerSec(50);
+  ScopedFieldTrials trial("WebRTC-Bwe-MaxRttLimit/limit:2s,floor:" +
+                          std::to_string(kBandwidthFloor.kbps()) + "kbps/");
+  // In the test case, we limit the capacity and add a cross traffic packet
+  // burst that blocks media from being sent. This causes the RTT to quickly
+  // increase above the threshold in the trial.
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(100);
+  const TimeDelta kBufferBloatDuration = TimeDelta::Seconds(10);
+  Scenario s("googcc_unit/limit_trial", false);
+  auto send_net = s.CreateSimulationNode([=](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(100);
+  });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  CallClientConfig config;
+  config.transport.rates.start_rate = kLinkCapacity;
+
+  auto* client = CreateVideoSendingClient(&s, config, {send_net}, {ret_net});
+  // Run for a few seconds to allow the controller to stabilize.
+  s.RunFor(TimeDelta::Seconds(10));
+  const DataSize kBloatPacketSize = DataSize::Bytes(1000);
+  const int kBloatPacketCount =
+      static_cast<int>(kBufferBloatDuration * kLinkCapacity / kBloatPacketSize);
+  // This will cause the RTT to be large for a while.
+  s.TriggerPacketBurst({send_net}, kBloatPacketCount, kBloatPacketSize.bytes());
+  // Wait to allow the high RTT to be detected and acted upon.
+  s.RunFor(TimeDelta::Seconds(6));
+  // By now the target rate should have dropped to the minimum configured rate.
+  EXPECT_NEAR(client->target_rate().kbps(), kBandwidthFloor.kbps(), 5);
+}
+
+TEST(GoogCcScenario, UpdatesTargetRateBasedOnLinkCapacity) {
+  UpdatesTargetRateBasedOnLinkCapacity();
+}
+
+TEST(GoogCcScenario, StableEstimateDoesNotVaryInSteadyState) {
+  auto factory = CreateFeedbackOnlyFactory();
+  Scenario s("googcc_unit/stable_target", false);
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(500);
+  net_conf.delay = TimeDelta::Millis(100);
+  auto send_net = s.CreateSimulationNode(net_conf);
+  auto ret_net = s.CreateSimulationNode(net_conf);
+
+  auto* client = CreateVideoSendingClient(&s, config, {send_net}, {ret_net});
+  // Run for a while to allow the estimate to stabilize.
+  s.RunFor(TimeDelta::Seconds(30));
+  DataRate min_stable_target = DataRate::PlusInfinity();
+  DataRate max_stable_target = DataRate::MinusInfinity();
+  DataRate min_target = DataRate::PlusInfinity();
+  DataRate max_target = DataRate::MinusInfinity();
+
+  // Measure variation in steady state.
+  for (int i = 0; i < 20; ++i) {
+    auto stable_target_rate = client->stable_target_rate();
+    auto target_rate = client->target_rate();
+    EXPECT_LE(stable_target_rate, target_rate);
+
+    min_stable_target = std::min(min_stable_target, stable_target_rate);
+    max_stable_target = std::max(max_stable_target, stable_target_rate);
+    min_target = std::min(min_target, target_rate);
+    max_target = std::max(max_target, target_rate);
+    s.RunFor(TimeDelta::Seconds(1));
+  }
+  // We should expect drops by at least 15% (default backoff.)
+  EXPECT_LT(min_target / max_target, 0.85);
+  // We should expect the stable target to be more stable than the immediate one
+  EXPECT_GE(min_stable_target / max_stable_target, min_target / max_target);
+}
+
+TEST(GoogCcScenario, LossBasedControlUpdatesTargetRateBasedOnLinkCapacity) {
+  ScopedFieldTrials trial("WebRTC-Bwe-LossBasedControl/Enabled/");
+  // TODO(srte): Should the behavior be unaffected at low loss rates?
+  UpdatesTargetRateBasedOnLinkCapacity("_loss_based");
+}
+
+TEST(GoogCcScenario, LossBasedControlDoesModestBackoffToHighLoss) {
+  ScopedFieldTrials trial("WebRTC-Bwe-LossBasedControl/Enabled/");
+  Scenario s("googcc_unit/high_loss_channel", false);
+  CallClientConfig config;
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(1500);
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  auto send_net = s.CreateSimulationNode([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(2000);
+    c->delay = TimeDelta::Millis(200);
+    c->loss_rate = 0.1;
+  });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(200); });
+
+  auto* client = CreateVideoSendingClient(&s, config, {send_net}, {ret_net});
+
+  s.RunFor(TimeDelta::Seconds(120));
+  // Without LossBasedControl trial, bandwidth drops to ~10 kbps.
+  EXPECT_GT(client->target_rate().kbps(), 100);
+}
+
+DataRate AverageBitrateAfterCrossInducedLoss(absl::string_view name) {
+  Scenario s(name, false);
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
+  net_conf.delay = TimeDelta::Millis(100);
+  // Short queue length means that we'll induce loss when sudden TCP traffic
+  // spikes are induced. This corresponds to ca 200 ms for a packet size of 1000
+  // bytes. Such limited buffers are common on for instance wifi routers.
+  net_conf.packet_queue_length_limit = 25;
+
+  auto send_net = {s.CreateSimulationNode(net_conf)};
+  auto ret_net = {s.CreateSimulationNode(net_conf)};
+
+  auto* client = s.CreateClient("send", CallClientConfig());
+  auto* callee = s.CreateClient("return", CallClientConfig());
+  auto* route = s.CreateRoutes(client, send_net, callee, ret_net);
+  // TODO(srte): Make this work with RTX enabled or remove it.
+  auto* video = s.CreateVideoStream(route->forward(), [](VideoStreamConfig* c) {
+    c->stream.use_rtx = false;
+  });
+  s.RunFor(TimeDelta::Seconds(10));
+  for (int i = 0; i < 4; ++i) {
+    // Sends TCP cross traffic inducing loss.
+    auto* tcp_traffic = s.net()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
+        s.net()->CreateRoute(send_net), s.net()->CreateRoute(ret_net),
+        FakeTcpConfig()));
+    s.RunFor(TimeDelta::Seconds(2));
+    // Allow the ccongestion controller to recover.
+    s.net()->StopCrossTraffic(tcp_traffic);
+    s.RunFor(TimeDelta::Seconds(20));
+  }
+
+  // Querying the video stats from within the expected runtime environment
+  // (i.e. the TQ that belongs to the CallClient, not the Scenario TQ that
+  // we're currently on).
+  VideoReceiveStreamInterface::Stats video_receive_stats;
+  auto* video_stream = video->receive();
+  callee->SendTask([&video_stream, &video_receive_stats]() {
+    video_receive_stats = video_stream->GetStats();
+  });
+  return DataSize::Bytes(
+             video_receive_stats.rtp_stats.packet_counter.TotalBytes()) /
+         s.TimeSinceStart();
+}
+
+TEST(GoogCcScenario, MaintainsLowRateInSafeResetTrial) {
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(200);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(300);
+
+  ScopedFieldTrials trial("WebRTC-Bwe-SafeResetOnRouteChange/Enabled/");
+  Scenario s("googcc_unit/safe_reset_low");
+  auto* send_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(10);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {send_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to stabilize.
+  s.RunFor(TimeDelta::Millis(500));
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kLinkCapacity.kbps(), 50);
+  s.ChangeRoute(route->forward(), {send_net});
+  // Allow new settings to propagate.
+  s.RunFor(TimeDelta::Millis(100));
+  // Under the trial, the target should be unchanged for low rates.
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kLinkCapacity.kbps(), 50);
+}
+
+TEST(GoogCcScenario, CutsHighRateInSafeResetTrial) {
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(1000);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(300);
+
+  ScopedFieldTrials trial("WebRTC-Bwe-SafeResetOnRouteChange/Enabled/");
+  Scenario s("googcc_unit/safe_reset_high_cut");
+  auto send_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(50);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {send_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to stabilize.
+  s.RunFor(TimeDelta::Millis(500));
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kLinkCapacity.kbps(), 300);
+  s.ChangeRoute(route->forward(), {send_net});
+  // Allow new settings to propagate.
+  s.RunFor(TimeDelta::Millis(50));
+  // Under the trial, the target should be reset from high values.
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kStartRate.kbps(), 30);
+}
+
+TEST(GoogCcScenario, DetectsHighRateInSafeResetTrial) {
+  ScopedFieldTrials trial("WebRTC-Bwe-SafeResetOnRouteChange/Enabled,ack/");
+  const DataRate kInitialLinkCapacity = DataRate::KilobitsPerSec(200);
+  const DataRate kNewLinkCapacity = DataRate::KilobitsPerSec(800);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(300);
+
+  Scenario s("googcc_unit/safe_reset_high_detect");
+  auto* initial_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kInitialLinkCapacity;
+    c->delay = TimeDelta::Millis(50);
+  });
+  auto* new_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kNewLinkCapacity;
+    c->delay = TimeDelta::Millis(50);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {initial_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to stabilize.
+  s.RunFor(TimeDelta::Millis(2000));
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kInitialLinkCapacity.kbps(), 50);
+  s.ChangeRoute(route->forward(), {new_net});
+  // Allow new settings to propagate, but not probes to be received.
+  s.RunFor(TimeDelta::Millis(50));
+  // Under the field trial, the target rate should be unchanged since it's lower
+  // than the starting rate.
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kInitialLinkCapacity.kbps(), 50);
+  // However, probing should have made us detect the higher rate.
+  // NOTE: This test causes high loss rate, and the loss-based estimator reduces
+  // the bitrate, making the test fail if we wait longer than one second here.
+  s.RunFor(TimeDelta::Millis(1000));
+  EXPECT_GT(client->send_bandwidth().kbps(), kNewLinkCapacity.kbps() - 300);
+}
+
+TEST(GoogCcScenario, TargetRateReducedOnPacingBufferBuildupInTrial) {
+  // Configure strict pacing to ensure build-up.
+  ScopedFieldTrials trial(
+      "WebRTC-CongestionWindow/QueueSize:100,MinBitrate:30000/"
+      "WebRTC-Video-Pacing/factor:1.0/"
+      "WebRTC-AddPacingToCongestionWindowPushback/Enabled/");
+
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(1000);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(1000);
+
+  Scenario s("googcc_unit/pacing_buffer_buildup");
+  auto* net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(50);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow some time for the buffer to build up.
+  s.RunFor(TimeDelta::Seconds(5));
+
+  // Without trial, pacer delay reaches ~250 ms.
+  EXPECT_LT(client->GetStats().pacer_delay_ms, 150);
+}
+
+TEST(GoogCcScenario, NoBandwidthTogglingInLossControlTrial) {
+  ScopedFieldTrials trial("WebRTC-Bwe-LossBasedControl/Enabled/");
+  Scenario s("googcc_unit/no_toggling");
+  auto* send_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(2000);
+    c->loss_rate = 0.2;
+    c->delay = TimeDelta::Millis(10);
+  });
+
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  });
+  auto* route = s.CreateRoutes(
+      client, {send_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to initialize.
+  s.RunFor(TimeDelta::Millis(250));
+
+  std::queue<DataRate> bandwidth_history;
+  const TimeDelta step = TimeDelta::Millis(50);
+  for (TimeDelta time = TimeDelta::Zero(); time < TimeDelta::Millis(2000);
+       time += step) {
+    s.RunFor(step);
+    const TimeDelta window = TimeDelta::Millis(500);
+    if (bandwidth_history.size() >= window / step)
+      bandwidth_history.pop();
+    bandwidth_history.push(client->send_bandwidth());
+    EXPECT_LT(
+        CountBandwidthDips(bandwidth_history, DataRate::KilobitsPerSec(100)),
+        2);
+  }
+}
+
+TEST(GoogCcScenario, NoRttBackoffCollapseWhenVideoStops) {
+  ScopedFieldTrials trial("WebRTC-Bwe-MaxRttLimit/limit:2s/");
+  Scenario s("googcc_unit/rttbackoff_video_stop");
+  auto* send_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(2000);
+    c->delay = TimeDelta::Millis(100);
+  });
+
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  });
+  auto* route = s.CreateRoutes(
+      client, {send_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  auto* video = s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to initialize, then stop video.
+  s.RunFor(TimeDelta::Seconds(1));
+  video->send()->Stop();
+  s.RunFor(TimeDelta::Seconds(4));
+  EXPECT_GT(client->send_bandwidth().kbps(), 1000);
+}
+
+TEST(GoogCcScenario, NoCrashOnVeryLateFeedback) {
+  Scenario s;
+  auto ret_net = s.CreateMutableSimulationNode(NetworkSimulationConfig());
+  auto* route = s.CreateRoutes(
+      s.CreateClient("send", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())},
+      s.CreateClient("return", CallClientConfig()), {ret_net->node()});
+  auto* video = s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  s.RunFor(TimeDelta::Seconds(5));
+  // Delay feedback by several minutes. This will cause removal of the send time
+  // history for the packets as long as kSendTimeHistoryWindow is configured for
+  // a shorter time span.
+  ret_net->PauseTransmissionUntil(s.Now() + TimeDelta::Seconds(300));
+  // Stopping video stream while waiting to save test execution time.
+  video->send()->Stop();
+  s.RunFor(TimeDelta::Seconds(299));
+  // Starting to cause addition of new packet to history, which cause old
+  // packets to be removed.
+  video->send()->Start();
+  // Runs until the lost packets are received. We expect that this will run
+  // without causing any runtime failures.
+  s.RunFor(TimeDelta::Seconds(2));
+}
+
+TEST(GoogCcScenario, IsFairToTCP) {
+  Scenario s("googcc_unit/tcp_fairness");
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
+  net_conf.delay = TimeDelta::Millis(50);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  });
+  auto send_net = {s.CreateSimulationNode(net_conf)};
+  auto ret_net = {s.CreateSimulationNode(net_conf)};
+  auto* route = s.CreateRoutes(
+      client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  s.net()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
+      s.net()->CreateRoute(send_net), s.net()->CreateRoute(ret_net),
+      FakeTcpConfig()));
+  s.RunFor(TimeDelta::Seconds(10));
+
+  // Currently only testing for the upper limit as we in practice back out
+  // quite a lot in this scenario. If this behavior is fixed, we should add a
+  // lower bound to ensure it stays fixed.
+  EXPECT_LT(client->send_bandwidth().kbps(), 750);
+}
+
+TEST(GoogCcScenario, FastRampupOnRembCapLifted) {
+  DataRate final_estimate =
+      RunRembDipScenario("googcc_unit/default_fast_rampup_on_remb_cap_lifted");
+  EXPECT_GT(final_estimate.kbps(), 1500);
+}
+
+TEST(GoogCcScenario, FallbackToLossBasedBweWithoutPacketFeedback) {
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(1000);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(1000);
+
+  Scenario s("googcc_unit/high_loss_channel", false);
+  auto* net = s.CreateMutableSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(100);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {net->node()}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+
+  // Create a config without packet feedback.
+  VideoStreamConfig video_config;
+  video_config.stream.packet_feedback = false;
+  s.CreateVideoStream(route->forward(), video_config);
+
+  s.RunFor(TimeDelta::Seconds(20));
+  // Bandwith does not backoff because network is normal.
+  EXPECT_GE(client->target_rate().kbps(), 500);
+
+  // Update the network to create high loss ratio
+  net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->loss_rate = 0.15;
+  });
+  s.RunFor(TimeDelta::Seconds(20));
+
+  // Bandwidth decreases thanks to loss based bwe v0.
+  EXPECT_LE(client->target_rate().kbps(), 300);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.cc
new file mode 100644
index 0000000000..2d50d08e6a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.cc
@@ -0,0 +1,140 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/inter_arrival_delta.h"
+
+#include <algorithm>
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+static constexpr TimeDelta kBurstDeltaThreshold = TimeDelta::Millis(5);
+static constexpr TimeDelta kMaxBurstDuration = TimeDelta::Millis(100);
+constexpr TimeDelta InterArrivalDelta::kArrivalTimeOffsetThreshold;
+
+InterArrivalDelta::InterArrivalDelta(TimeDelta send_time_group_length)
+    : send_time_group_length_(send_time_group_length),
+      current_timestamp_group_(),
+      prev_timestamp_group_(),
+      num_consecutive_reordered_packets_(0) {}
+
+bool InterArrivalDelta::ComputeDeltas(Timestamp send_time,
+                                      Timestamp arrival_time,
+                                      Timestamp system_time,
+                                      size_t packet_size,
+                                      TimeDelta* send_time_delta,
+                                      TimeDelta* arrival_time_delta,
+                                      int* packet_size_delta) {
+  bool calculated_deltas = false;
+  if (current_timestamp_group_.IsFirstPacket()) {
+    // We don't have enough data to update the filter, so we store it until we
+    // have two frames of data to process.
+    current_timestamp_group_.send_time = send_time;
+    current_timestamp_group_.first_send_time = send_time;
+    current_timestamp_group_.first_arrival = arrival_time;
+  } else if (current_timestamp_group_.first_send_time > send_time) {
+    // Reordered packet.
+    return false;
+  } else if (NewTimestampGroup(arrival_time, send_time)) {
+    // First packet of a later send burst, the previous packets sample is ready.
+    if (prev_timestamp_group_.complete_time.IsFinite()) {
+      *send_time_delta =
+          current_timestamp_group_.send_time - prev_timestamp_group_.send_time;
+      *arrival_time_delta = current_timestamp_group_.complete_time -
+                            prev_timestamp_group_.complete_time;
+
+      TimeDelta system_time_delta = current_timestamp_group_.last_system_time -
+                                    prev_timestamp_group_.last_system_time;
+
+      if (*arrival_time_delta - system_time_delta >=
+          kArrivalTimeOffsetThreshold) {
+        RTC_LOG(LS_WARNING)
+            << "The arrival time clock offset has changed (diff = "
+            << arrival_time_delta->ms() - system_time_delta.ms()
+            << " ms), resetting.";
+        Reset();
+        return false;
+      }
+      if (*arrival_time_delta < TimeDelta::Zero()) {
+        // The group of packets has been reordered since receiving its local
+        // arrival timestamp.
+        ++num_consecutive_reordered_packets_;
+        if (num_consecutive_reordered_packets_ >= kReorderedResetThreshold) {
+          RTC_LOG(LS_WARNING)
+              << "Packets between send burst arrived out of order, resetting:"
+              << " arrival_time_delta_ms=" << arrival_time_delta->ms()
+              << ", send_time_delta_ms=" << send_time_delta->ms();
+          Reset();
+        }
+        return false;
+      } else {
+        num_consecutive_reordered_packets_ = 0;
+      }
+      *packet_size_delta = static_cast<int>(current_timestamp_group_.size) -
+                           static_cast<int>(prev_timestamp_group_.size);
+      calculated_deltas = true;
+    }
+    prev_timestamp_group_ = current_timestamp_group_;
+    // The new timestamp is now the current frame.
+    current_timestamp_group_.first_send_time = send_time;
+    current_timestamp_group_.send_time = send_time;
+    current_timestamp_group_.first_arrival = arrival_time;
+    current_timestamp_group_.size = 0;
+  } else {
+    current_timestamp_group_.send_time =
+        std::max(current_timestamp_group_.send_time, send_time);
+  }
+  // Accumulate the frame size.
+  current_timestamp_group_.size += packet_size;
+  current_timestamp_group_.complete_time = arrival_time;
+  current_timestamp_group_.last_system_time = system_time;
+
+  return calculated_deltas;
+}
+
+// Assumes that `timestamp` is not reordered compared to
+// `current_timestamp_group_`.
+bool InterArrivalDelta::NewTimestampGroup(Timestamp arrival_time,
+                                          Timestamp send_time) const {
+  if (current_timestamp_group_.IsFirstPacket()) {
+    return false;
+  } else if (BelongsToBurst(arrival_time, send_time)) {
+    return false;
+  } else {
+    return send_time - current_timestamp_group_.first_send_time >
+           send_time_group_length_;
+  }
+}
+
+bool InterArrivalDelta::BelongsToBurst(Timestamp arrival_time,
+                                       Timestamp send_time) const {
+  RTC_DCHECK(current_timestamp_group_.complete_time.IsFinite());
+  TimeDelta arrival_time_delta =
+      arrival_time - current_timestamp_group_.complete_time;
+  TimeDelta send_time_delta = send_time - current_timestamp_group_.send_time;
+  if (send_time_delta.IsZero())
+    return true;
+  TimeDelta propagation_delta = arrival_time_delta - send_time_delta;
+  if (propagation_delta < TimeDelta::Zero() &&
+      arrival_time_delta <= kBurstDeltaThreshold &&
+      arrival_time - current_timestamp_group_.first_arrival < kMaxBurstDuration)
+    return true;
+  return false;
+}
+
+void InterArrivalDelta::Reset() {
+  num_consecutive_reordered_packets_ = 0;
+  current_timestamp_group_ = SendTimeGroup();
+  prev_timestamp_group_ = SendTimeGroup();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.h
new file mode 100644
index 0000000000..4046590eeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.h
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_INTER_ARRIVAL_DELTA_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_INTER_ARRIVAL_DELTA_H_
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+
+// Helper class to compute the inter-arrival time delta and the size delta
+// between two send bursts. This code is branched from
+// modules/remote_bitrate_estimator/inter_arrival.
+class InterArrivalDelta {
+ public:
+  // After this many packet groups received out of order InterArrival will
+  // reset, assuming that clocks have made a jump.
+  static constexpr int kReorderedResetThreshold = 3;
+  static constexpr TimeDelta kArrivalTimeOffsetThreshold =
+      TimeDelta::Seconds(3);
+
+  // A send time group is defined as all packets with a send time which are at
+  // most send_time_group_length older than the first timestamp in that
+  // group.
+  explicit InterArrivalDelta(TimeDelta send_time_group_length);
+
+  InterArrivalDelta() = delete;
+  InterArrivalDelta(const InterArrivalDelta&) = delete;
+  InterArrivalDelta& operator=(const InterArrivalDelta&) = delete;
+
+  // This function returns true if a delta was computed, or false if the current
+  // group is still incomplete or if only one group has been completed.
+  // `send_time` is the send time.
+  // `arrival_time` is the time at which the packet arrived.
+  // `packet_size` is the size of the packet.
+  // `timestamp_delta` (output) is the computed send time delta.
+  // `arrival_time_delta` (output) is the computed arrival-time delta.
+  // `packet_size_delta` (output) is the computed size delta.
+  bool ComputeDeltas(Timestamp send_time,
+                     Timestamp arrival_time,
+                     Timestamp system_time,
+                     size_t packet_size,
+                     TimeDelta* send_time_delta,
+                     TimeDelta* arrival_time_delta,
+                     int* packet_size_delta);
+
+ private:
+  struct SendTimeGroup {
+    SendTimeGroup()
+        : size(0),
+          first_send_time(Timestamp::MinusInfinity()),
+          send_time(Timestamp::MinusInfinity()),
+          first_arrival(Timestamp::MinusInfinity()),
+          complete_time(Timestamp::MinusInfinity()),
+          last_system_time(Timestamp::MinusInfinity()) {}
+
+    bool IsFirstPacket() const { return complete_time.IsInfinite(); }
+
+    size_t size;
+    Timestamp first_send_time;
+    Timestamp send_time;
+    Timestamp first_arrival;
+    Timestamp complete_time;
+    Timestamp last_system_time;
+  };
+
+  // Returns true if the last packet was the end of the current batch and the
+  // packet with `send_time` is the first of a new batch.
+  bool NewTimestampGroup(Timestamp arrival_time, Timestamp send_time) const;
+
+  bool BelongsToBurst(Timestamp arrival_time, Timestamp send_time) const;
+
+  void Reset();
+
+  const TimeDelta send_time_group_length_;
+  SendTimeGroup current_timestamp_group_;
+  SendTimeGroup prev_timestamp_group_;
+  int num_consecutive_reordered_packets_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_INTER_ARRIVAL_DELTA_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc
new file mode 100644
index 0000000000..9fd537a422
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc
@@ -0,0 +1,77 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/goog_cc/link_capacity_estimator.h"
+
+#include <algorithm>
+
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+LinkCapacityEstimator::LinkCapacityEstimator() {}
+
+DataRate LinkCapacityEstimator::UpperBound() const {
+  if (estimate_kbps_.has_value())
+    return DataRate::KilobitsPerSec(estimate_kbps_.value() +
+                                    3 * deviation_estimate_kbps());
+  return DataRate::Infinity();
+}
+
+DataRate LinkCapacityEstimator::LowerBound() const {
+  if (estimate_kbps_.has_value())
+    return DataRate::KilobitsPerSec(
+        std::max(0.0, estimate_kbps_.value() - 3 * deviation_estimate_kbps()));
+  return DataRate::Zero();
+}
+
+void LinkCapacityEstimator::Reset() {
+  estimate_kbps_.reset();
+}
+
+void LinkCapacityEstimator::OnOveruseDetected(DataRate acknowledged_rate) {
+  Update(acknowledged_rate, 0.05);
+}
+
+void LinkCapacityEstimator::OnProbeRate(DataRate probe_rate) {
+  Update(probe_rate, 0.5);
+}
+
+void LinkCapacityEstimator::Update(DataRate capacity_sample, double alpha) {
+  double sample_kbps = capacity_sample.kbps();
+  if (!estimate_kbps_.has_value()) {
+    estimate_kbps_ = sample_kbps;
+  } else {
+    estimate_kbps_ = (1 - alpha) * estimate_kbps_.value() + alpha * sample_kbps;
+  }
+  // Estimate the variance of the link capacity estimate and normalize the
+  // variance with the link capacity estimate.
+  const double norm = std::max(estimate_kbps_.value(), 1.0);
+  double error_kbps = estimate_kbps_.value() - sample_kbps;
+  deviation_kbps_ =
+      (1 - alpha) * deviation_kbps_ + alpha * error_kbps * error_kbps / norm;
+  // 0.4 ~= 14 kbit/s at 500 kbit/s
+  // 2.5f ~= 35 kbit/s at 500 kbit/s
+  deviation_kbps_ = rtc::SafeClamp(deviation_kbps_, 0.4f, 2.5f);
+}
+
+bool LinkCapacityEstimator::has_estimate() const {
+  return estimate_kbps_.has_value();
+}
+
+DataRate LinkCapacityEstimator::estimate() const {
+  return DataRate::KilobitsPerSec(*estimate_kbps_);
+}
+
+double LinkCapacityEstimator::deviation_estimate_kbps() const {
+  // Calculate the max bit rate std dev given the normalized
+  // variance and the current throughput bitrate. The standard deviation will
+  // only be used if estimate_kbps_ has a value.
+  return sqrt(deviation_kbps_ * estimate_kbps_.value());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.h
new file mode 100644
index 0000000000..aa23491d9d
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
+
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+
+namespace webrtc {
+class LinkCapacityEstimator {
+ public:
+  LinkCapacityEstimator();
+  DataRate UpperBound() const;
+  DataRate LowerBound() const;
+  void Reset();
+  void OnOveruseDetected(DataRate acknowledged_rate);
+  void OnProbeRate(DataRate probe_rate);
+  bool has_estimate() const;
+  DataRate estimate() const;
+
+ private:
+  friend class GoogCcStatePrinter;
+  void Update(DataRate capacity_sample, double alpha);
+
+  double deviation_estimate_kbps() const;
+  absl::optional<double> estimate_kbps_;
+  double deviation_kbps_ = 0.4;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build
new file mode 100644
index 0000000000..829ff47d87
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("link_capacity_estimator_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc
new file mode 100644
index 0000000000..7524c84d92
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc
@@ -0,0 +1,260 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h"
+
+#include <algorithm>
+#include <string>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+namespace {
+const char kBweLossBasedControl[] = "WebRTC-Bwe-LossBasedControl";
+
+// Expecting RTCP feedback to be sent with roughly 1s intervals, a 5s gap
+// indicates a channel outage.
+constexpr TimeDelta kMaxRtcpFeedbackInterval = TimeDelta::Millis(5000);
+
+// Increase slower when RTT is high.
+double GetIncreaseFactor(const LossBasedControlConfig& config, TimeDelta rtt) {
+  // Clamp the RTT
+  if (rtt < config.increase_low_rtt) {
+    rtt = config.increase_low_rtt;
+  } else if (rtt > config.increase_high_rtt) {
+    rtt = config.increase_high_rtt;
+  }
+  auto rtt_range = config.increase_high_rtt.Get() - config.increase_low_rtt;
+  if (rtt_range <= TimeDelta::Zero()) {
+    RTC_DCHECK_NOTREACHED();  // Only on misconfiguration.
+    return config.min_increase_factor;
+  }
+  auto rtt_offset = rtt - config.increase_low_rtt;
+  auto relative_offset = std::max(0.0, std::min(rtt_offset / rtt_range, 1.0));
+  auto factor_range = config.max_increase_factor - config.min_increase_factor;
+  return config.min_increase_factor + (1 - relative_offset) * factor_range;
+}
+
+double LossFromBitrate(DataRate bitrate,
+                       DataRate loss_bandwidth_balance,
+                       double exponent) {
+  if (loss_bandwidth_balance >= bitrate)
+    return 1.0;
+  return pow(loss_bandwidth_balance / bitrate, exponent);
+}
+
+DataRate BitrateFromLoss(double loss,
+                         DataRate loss_bandwidth_balance,
+                         double exponent) {
+  if (exponent <= 0) {
+    RTC_DCHECK_NOTREACHED();
+    return DataRate::Infinity();
+  }
+  if (loss < 1e-5)
+    return DataRate::Infinity();
+  return loss_bandwidth_balance * pow(loss, -1.0 / exponent);
+}
+
+double ExponentialUpdate(TimeDelta window, TimeDelta interval) {
+  // Use the convention that exponential window length (which is really
+  // infinite) is the time it takes to dampen to 1/e.
+  if (window <= TimeDelta::Zero()) {
+    RTC_DCHECK_NOTREACHED();
+    return 1.0f;
+  }
+  return 1.0f - exp(interval / window * -1.0);
+}
+
+bool IsEnabled(const webrtc::FieldTrialsView& key_value_config,
+               absl::string_view name) {
+  return absl::StartsWith(key_value_config.Lookup(name), "Enabled");
+}
+
+}  // namespace
+
+LossBasedControlConfig::LossBasedControlConfig(
+    const FieldTrialsView* key_value_config)
+    : enabled(IsEnabled(*key_value_config, kBweLossBasedControl)),
+      min_increase_factor("min_incr", 1.02),
+      max_increase_factor("max_incr", 1.08),
+      increase_low_rtt("incr_low_rtt", TimeDelta::Millis(200)),
+      increase_high_rtt("incr_high_rtt", TimeDelta::Millis(800)),
+      decrease_factor("decr", 0.99),
+      loss_window("loss_win", TimeDelta::Millis(800)),
+      loss_max_window("loss_max_win", TimeDelta::Millis(800)),
+      acknowledged_rate_max_window("ackrate_max_win", TimeDelta::Millis(800)),
+      increase_offset("incr_offset", DataRate::BitsPerSec(1000)),
+      loss_bandwidth_balance_increase("balance_incr",
+                                      DataRate::KilobitsPerSec(0.5)),
+      loss_bandwidth_balance_decrease("balance_decr",
+                                      DataRate::KilobitsPerSec(4)),
+      loss_bandwidth_balance_reset("balance_reset",
+                                   DataRate::KilobitsPerSec(0.1)),
+      loss_bandwidth_balance_exponent("exponent", 0.5),
+      allow_resets("resets", false),
+      decrease_interval("decr_intvl", TimeDelta::Millis(300)),
+      loss_report_timeout("timeout", TimeDelta::Millis(6000)) {
+  ParseFieldTrial(
+      {&min_increase_factor, &max_increase_factor, &increase_low_rtt,
+       &increase_high_rtt, &decrease_factor, &loss_window, &loss_max_window,
+       &acknowledged_rate_max_window, &increase_offset,
+       &loss_bandwidth_balance_increase, &loss_bandwidth_balance_decrease,
+       &loss_bandwidth_balance_reset, &loss_bandwidth_balance_exponent,
+       &allow_resets, &decrease_interval, &loss_report_timeout},
+      key_value_config->Lookup(kBweLossBasedControl));
+}
+LossBasedControlConfig::LossBasedControlConfig(const LossBasedControlConfig&) =
+    default;
+LossBasedControlConfig::~LossBasedControlConfig() = default;
+
+LossBasedBandwidthEstimation::LossBasedBandwidthEstimation(
+    const FieldTrialsView* key_value_config)
+    : config_(key_value_config),
+      average_loss_(0),
+      average_loss_max_(0),
+      loss_based_bitrate_(DataRate::Zero()),
+      acknowledged_bitrate_max_(DataRate::Zero()),
+      acknowledged_bitrate_last_update_(Timestamp::MinusInfinity()),
+      time_last_decrease_(Timestamp::MinusInfinity()),
+      has_decreased_since_last_loss_report_(false),
+      last_loss_packet_report_(Timestamp::MinusInfinity()),
+      last_loss_ratio_(0) {}
+
+void LossBasedBandwidthEstimation::UpdateLossStatistics(
+    const std::vector<PacketResult>& packet_results,
+    Timestamp at_time) {
+  if (packet_results.empty()) {
+    RTC_DCHECK_NOTREACHED();
+    return;
+  }
+  int loss_count = 0;
+  for (const auto& pkt : packet_results) {
+    loss_count += !pkt.IsReceived() ? 1 : 0;
+  }
+  last_loss_ratio_ = static_cast<double>(loss_count) / packet_results.size();
+  const TimeDelta time_passed = last_loss_packet_report_.IsFinite()
+                                    ? at_time - last_loss_packet_report_
+                                    : TimeDelta::Seconds(1);
+  last_loss_packet_report_ = at_time;
+  has_decreased_since_last_loss_report_ = false;
+
+  average_loss_ += ExponentialUpdate(config_.loss_window, time_passed) *
+                   (last_loss_ratio_ - average_loss_);
+  if (average_loss_ > average_loss_max_) {
+    average_loss_max_ = average_loss_;
+  } else {
+    average_loss_max_ +=
+        ExponentialUpdate(config_.loss_max_window, time_passed) *
+        (average_loss_ - average_loss_max_);
+  }
+}
+
+void LossBasedBandwidthEstimation::UpdateAcknowledgedBitrate(
+    DataRate acknowledged_bitrate,
+    Timestamp at_time) {
+  const TimeDelta time_passed =
+      acknowledged_bitrate_last_update_.IsFinite()
+          ? at_time - acknowledged_bitrate_last_update_
+          : TimeDelta::Seconds(1);
+  acknowledged_bitrate_last_update_ = at_time;
+  if (acknowledged_bitrate > acknowledged_bitrate_max_) {
+    acknowledged_bitrate_max_ = acknowledged_bitrate;
+  } else {
+    acknowledged_bitrate_max_ -=
+        ExponentialUpdate(config_.acknowledged_rate_max_window, time_passed) *
+        (acknowledged_bitrate_max_ - acknowledged_bitrate);
+  }
+}
+
+DataRate LossBasedBandwidthEstimation::Update(Timestamp at_time,
+                                              DataRate min_bitrate,
+                                              DataRate wanted_bitrate,
+                                              TimeDelta last_round_trip_time) {
+  if (loss_based_bitrate_.IsZero()) {
+    loss_based_bitrate_ = wanted_bitrate;
+  }
+
+  // Only increase if loss has been low for some time.
+  const double loss_estimate_for_increase = average_loss_max_;
+  // Avoid multiple decreases from averaging over one loss spike.
+  const double loss_estimate_for_decrease =
+      std::min(average_loss_, last_loss_ratio_);
+  const bool allow_decrease =
+      !has_decreased_since_last_loss_report_ &&
+      (at_time - time_last_decrease_ >=
+       last_round_trip_time + config_.decrease_interval);
+  // If packet lost reports are too old, dont increase bitrate.
+  const bool loss_report_valid =
+      at_time - last_loss_packet_report_ < 1.2 * kMaxRtcpFeedbackInterval;
+
+  if (loss_report_valid && config_.allow_resets &&
+      loss_estimate_for_increase < loss_reset_threshold()) {
+    loss_based_bitrate_ = wanted_bitrate;
+  } else if (loss_report_valid &&
+             loss_estimate_for_increase < loss_increase_threshold()) {
+    // Increase bitrate by RTT-adaptive ratio.
+    DataRate new_increased_bitrate =
+        min_bitrate * GetIncreaseFactor(config_, last_round_trip_time) +
+        config_.increase_offset;
+    // The bitrate that would make the loss "just high enough".
+    const DataRate new_increased_bitrate_cap = BitrateFromLoss(
+        loss_estimate_for_increase, config_.loss_bandwidth_balance_increase,
+        config_.loss_bandwidth_balance_exponent);
+    new_increased_bitrate =
+        std::min(new_increased_bitrate, new_increased_bitrate_cap);
+    loss_based_bitrate_ = std::max(new_increased_bitrate, loss_based_bitrate_);
+  } else if (loss_estimate_for_decrease > loss_decrease_threshold() &&
+             allow_decrease) {
+    // The bitrate that would make the loss "just acceptable".
+    const DataRate new_decreased_bitrate_floor = BitrateFromLoss(
+        loss_estimate_for_decrease, config_.loss_bandwidth_balance_decrease,
+        config_.loss_bandwidth_balance_exponent);
+    DataRate new_decreased_bitrate =
+        std::max(decreased_bitrate(), new_decreased_bitrate_floor);
+    if (new_decreased_bitrate < loss_based_bitrate_) {
+      time_last_decrease_ = at_time;
+      has_decreased_since_last_loss_report_ = true;
+      loss_based_bitrate_ = new_decreased_bitrate;
+    }
+  }
+  return loss_based_bitrate_;
+}
+
+void LossBasedBandwidthEstimation::Initialize(DataRate bitrate) {
+  loss_based_bitrate_ = bitrate;
+  average_loss_ = 0;
+  average_loss_max_ = 0;
+}
+
+double LossBasedBandwidthEstimation::loss_reset_threshold() const {
+  return LossFromBitrate(loss_based_bitrate_,
+                         config_.loss_bandwidth_balance_reset,
+                         config_.loss_bandwidth_balance_exponent);
+}
+
+double LossBasedBandwidthEstimation::loss_increase_threshold() const {
+  return LossFromBitrate(loss_based_bitrate_,
+                         config_.loss_bandwidth_balance_increase,
+                         config_.loss_bandwidth_balance_exponent);
+}
+
+double LossBasedBandwidthEstimation::loss_decrease_threshold() const {
+  return LossFromBitrate(loss_based_bitrate_,
+                         config_.loss_bandwidth_balance_decrease,
+                         config_.loss_bandwidth_balance_exponent);
+}
+
+DataRate LossBasedBandwidthEstimation::decreased_bitrate() const {
+  return config_.decrease_factor * acknowledged_bitrate_max_;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h
new file mode 100644
index 0000000000..9f69caba89
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h
@@ -0,0 +1,97 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
+
+#include <vector>
+
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+struct LossBasedControlConfig {
+  explicit LossBasedControlConfig(const FieldTrialsView* key_value_config);
+  LossBasedControlConfig(const LossBasedControlConfig&);
+  LossBasedControlConfig& operator=(const LossBasedControlConfig&) = default;
+  ~LossBasedControlConfig();
+  bool enabled;
+  FieldTrialParameter<double> min_increase_factor;
+  FieldTrialParameter<double> max_increase_factor;
+  FieldTrialParameter<TimeDelta> increase_low_rtt;
+  FieldTrialParameter<TimeDelta> increase_high_rtt;
+  FieldTrialParameter<double> decrease_factor;
+  FieldTrialParameter<TimeDelta> loss_window;
+  FieldTrialParameter<TimeDelta> loss_max_window;
+  FieldTrialParameter<TimeDelta> acknowledged_rate_max_window;
+  FieldTrialParameter<DataRate> increase_offset;
+  FieldTrialParameter<DataRate> loss_bandwidth_balance_increase;
+  FieldTrialParameter<DataRate> loss_bandwidth_balance_decrease;
+  FieldTrialParameter<DataRate> loss_bandwidth_balance_reset;
+  FieldTrialParameter<double> loss_bandwidth_balance_exponent;
+  FieldTrialParameter<bool> allow_resets;
+  FieldTrialParameter<TimeDelta> decrease_interval;
+  FieldTrialParameter<TimeDelta> loss_report_timeout;
+};
+
+// Estimates an upper BWE limit based on loss.
+// It requires knowledge about lost packets and acknowledged bitrate.
+// Ie, this class require transport feedback.
+class LossBasedBandwidthEstimation {
+ public:
+  explicit LossBasedBandwidthEstimation(
+      const FieldTrialsView* key_value_config);
+  // Returns the new estimate.
+  DataRate Update(Timestamp at_time,
+                  DataRate min_bitrate,
+                  DataRate wanted_bitrate,
+                  TimeDelta last_round_trip_time);
+  void UpdateAcknowledgedBitrate(DataRate acknowledged_bitrate,
+                                 Timestamp at_time);
+  void Initialize(DataRate bitrate);
+  bool Enabled() const { return config_.enabled; }
+  // Returns true if LossBasedBandwidthEstimation is enabled and have
+  // received loss statistics. Ie, this class require transport feedback.
+  bool InUse() const {
+    return Enabled() && last_loss_packet_report_.IsFinite();
+  }
+  void UpdateLossStatistics(const std::vector<PacketResult>& packet_results,
+                            Timestamp at_time);
+  DataRate GetEstimate() const { return loss_based_bitrate_; }
+
+ private:
+  friend class GoogCcStatePrinter;
+  void Reset(DataRate bitrate);
+  double loss_increase_threshold() const;
+  double loss_decrease_threshold() const;
+  double loss_reset_threshold() const;
+
+  DataRate decreased_bitrate() const;
+
+  const LossBasedControlConfig config_;
+  double average_loss_;
+  double average_loss_max_;
+  DataRate loss_based_bitrate_;
+  DataRate acknowledged_bitrate_max_;
+  Timestamp acknowledged_bitrate_last_update_;
+  Timestamp time_last_decrease_;
+  bool has_decreased_since_last_loss_report_;
+  Timestamp last_loss_packet_report_;
+  double last_loss_ratio_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build
new file mode 100644
index 0000000000..aae4b82520
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build
@@ -0,0 +1,226 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("loss_based_bwe_v1_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
new file mode 100644
index 0000000000..b6efdeee9e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
@@ -0,0 +1,1080 @@
+/*
+ *  Copyright 2021 The WebRTC project authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+
+#include <algorithm>
+#include <cmath>
+#include <complex>
+#include <cstddef>
+#include <cstdlib>
+#include <limits>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "rtc_base/experiments/field_trial_list.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+bool IsValid(DataRate datarate) {
+  return datarate.IsFinite();
+}
+
+bool IsValid(absl::optional<DataRate> datarate) {
+  return datarate.has_value() && IsValid(datarate.value());
+}
+
+bool IsValid(Timestamp timestamp) {
+  return timestamp.IsFinite();
+}
+
+struct PacketResultsSummary {
+  int num_packets = 0;
+  int num_lost_packets = 0;
+  DataSize total_size = DataSize::Zero();
+  Timestamp first_send_time = Timestamp::PlusInfinity();
+  Timestamp last_send_time = Timestamp::MinusInfinity();
+};
+
+// Returns a `PacketResultsSummary` where `first_send_time` is `PlusInfinity,
+// and `last_send_time` is `MinusInfinity`, if `packet_results` is empty.
+PacketResultsSummary GetPacketResultsSummary(
+    rtc::ArrayView<const PacketResult> packet_results) {
+  PacketResultsSummary packet_results_summary;
+
+  packet_results_summary.num_packets = packet_results.size();
+  for (const PacketResult& packet : packet_results) {
+    if (!packet.IsReceived()) {
+      packet_results_summary.num_lost_packets++;
+    }
+    packet_results_summary.total_size += packet.sent_packet.size;
+    packet_results_summary.first_send_time = std::min(
+        packet_results_summary.first_send_time, packet.sent_packet.send_time);
+    packet_results_summary.last_send_time = std::max(
+        packet_results_summary.last_send_time, packet.sent_packet.send_time);
+  }
+
+  return packet_results_summary;
+}
+
+double GetLossProbability(double inherent_loss,
+                          DataRate loss_limited_bandwidth,
+                          DataRate sending_rate) {
+  if (inherent_loss < 0.0 || inherent_loss > 1.0) {
+    RTC_LOG(LS_WARNING) << "The inherent loss must be in [0,1]: "
+                        << inherent_loss;
+    inherent_loss = std::min(std::max(inherent_loss, 0.0), 1.0);
+  }
+  if (!sending_rate.IsFinite()) {
+    RTC_LOG(LS_WARNING) << "The sending rate must be finite: "
+                        << ToString(sending_rate);
+  }
+  if (!loss_limited_bandwidth.IsFinite()) {
+    RTC_LOG(LS_WARNING) << "The loss limited bandwidth must be finite: "
+                        << ToString(loss_limited_bandwidth);
+  }
+
+  double loss_probability = inherent_loss;
+  if (IsValid(sending_rate) && IsValid(loss_limited_bandwidth) &&
+      (sending_rate > loss_limited_bandwidth)) {
+    loss_probability += (1 - inherent_loss) *
+                        (sending_rate - loss_limited_bandwidth) / sending_rate;
+  }
+  return std::min(std::max(loss_probability, 1.0e-6), 1.0 - 1.0e-6);
+}
+
+}  // namespace
+
+LossBasedBweV2::LossBasedBweV2(const FieldTrialsView* key_value_config)
+    : config_(CreateConfig(key_value_config)) {
+  if (!config_.has_value()) {
+    RTC_LOG(LS_VERBOSE) << "The configuration does not specify that the "
+                           "estimator should be enabled, disabling it.";
+    return;
+  }
+  if (!IsConfigValid()) {
+    RTC_LOG(LS_WARNING)
+        << "The configuration is not valid, disabling the estimator.";
+    config_.reset();
+    return;
+  }
+
+  current_estimate_.inherent_loss = config_->initial_inherent_loss_estimate;
+  observations_.resize(config_->observation_window_size);
+  temporal_weights_.resize(config_->observation_window_size);
+  instant_upper_bound_temporal_weights_.resize(
+      config_->observation_window_size);
+  CalculateTemporalWeights();
+}
+
+bool LossBasedBweV2::IsEnabled() const {
+  return config_.has_value();
+}
+
+bool LossBasedBweV2::IsReady() const {
+  return IsEnabled() && IsValid(current_estimate_.loss_limited_bandwidth) &&
+         num_observations_ > 0;
+}
+
+LossBasedBweV2::Result LossBasedBweV2::GetLossBasedResult() const {
+  Result result;
+  result.state = current_state_;
+  if (!IsReady()) {
+    if (!IsEnabled()) {
+      RTC_LOG(LS_WARNING)
+          << "The estimator must be enabled before it can be used.";
+    } else {
+      if (!IsValid(current_estimate_.loss_limited_bandwidth)) {
+        RTC_LOG(LS_WARNING)
+            << "The estimator must be initialized before it can be used.";
+      }
+      if (num_observations_ <= 0) {
+        RTC_LOG(LS_WARNING) << "The estimator must receive enough loss "
+                               "statistics before it can be used.";
+      }
+    }
+    result.bandwidth_estimate = IsValid(delay_based_estimate_)
+                                    ? delay_based_estimate_
+                                    : DataRate::PlusInfinity();
+    return result;
+  }
+
+  if (IsValid(delay_based_estimate_)) {
+    result.bandwidth_estimate =
+        std::min({current_estimate_.loss_limited_bandwidth,
+                  GetInstantUpperBound(), delay_based_estimate_});
+  } else {
+    result.bandwidth_estimate = std::min(
+        current_estimate_.loss_limited_bandwidth, GetInstantUpperBound());
+  }
+  return result;
+}
+
+void LossBasedBweV2::SetAcknowledgedBitrate(DataRate acknowledged_bitrate) {
+  if (IsValid(acknowledged_bitrate)) {
+    acknowledged_bitrate_ = acknowledged_bitrate;
+  } else {
+    RTC_LOG(LS_WARNING) << "The acknowledged bitrate must be finite: "
+                        << ToString(acknowledged_bitrate);
+  }
+}
+
+void LossBasedBweV2::SetBandwidthEstimate(DataRate bandwidth_estimate) {
+  if (IsValid(bandwidth_estimate)) {
+    current_estimate_.loss_limited_bandwidth = bandwidth_estimate;
+  } else {
+    RTC_LOG(LS_WARNING) << "The bandwidth estimate must be finite: "
+                        << ToString(bandwidth_estimate);
+  }
+}
+
+void LossBasedBweV2::SetMinMaxBitrate(DataRate min_bitrate,
+                                      DataRate max_bitrate) {
+  if (IsValid(min_bitrate)) {
+    min_bitrate_ = min_bitrate;
+  } else {
+    RTC_LOG(LS_WARNING) << "The min bitrate must be finite: "
+                        << ToString(min_bitrate);
+  }
+
+  if (IsValid(max_bitrate)) {
+    max_bitrate_ = max_bitrate;
+  } else {
+    RTC_LOG(LS_WARNING) << "The max bitrate must be finite: "
+                        << ToString(max_bitrate);
+  }
+}
+
+void LossBasedBweV2::SetProbeBitrate(absl::optional<DataRate> probe_bitrate) {
+  if (probe_bitrate.has_value() && IsValid(probe_bitrate.value())) {
+    if (!IsValid(probe_bitrate_) || probe_bitrate_ > probe_bitrate.value()) {
+      probe_bitrate_ = probe_bitrate.value();
+    }
+  }
+}
+
+void LossBasedBweV2::UpdateBandwidthEstimate(
+    rtc::ArrayView<const PacketResult> packet_results,
+    DataRate delay_based_estimate,
+    BandwidthUsage delay_detector_state,
+    absl::optional<DataRate> probe_bitrate,
+    DataRate upper_link_capacity) {
+  delay_based_estimate_ = delay_based_estimate;
+  upper_link_capacity_ = upper_link_capacity;
+  if (!IsEnabled()) {
+    RTC_LOG(LS_WARNING)
+        << "The estimator must be enabled before it can be used.";
+    return;
+  }
+  SetProbeBitrate(probe_bitrate);
+  if (packet_results.empty()) {
+    RTC_LOG(LS_VERBOSE)
+        << "The estimate cannot be updated without any loss statistics.";
+    return;
+  }
+
+  if (!PushBackObservation(packet_results, delay_detector_state)) {
+    return;
+  }
+
+  if (!IsValid(current_estimate_.loss_limited_bandwidth)) {
+    RTC_LOG(LS_VERBOSE)
+        << "The estimator must be initialized before it can be used.";
+    return;
+  }
+
+  ChannelParameters best_candidate = current_estimate_;
+  double objective_max = std::numeric_limits<double>::lowest();
+  for (ChannelParameters candidate : GetCandidates()) {
+    NewtonsMethodUpdate(candidate);
+
+    const double candidate_objective = GetObjective(candidate);
+    if (candidate_objective > objective_max) {
+      objective_max = candidate_objective;
+      best_candidate = candidate;
+    }
+  }
+  if (best_candidate.loss_limited_bandwidth <
+      current_estimate_.loss_limited_bandwidth) {
+    last_time_estimate_reduced_ = last_send_time_most_recent_observation_;
+  }
+
+  // Do not increase the estimate if the average loss is greater than current
+  // inherent loss.
+  if (GetAverageReportedLossRatio() > best_candidate.inherent_loss &&
+      config_->not_increase_if_inherent_loss_less_than_average_loss &&
+      current_estimate_.loss_limited_bandwidth <
+          best_candidate.loss_limited_bandwidth) {
+    best_candidate.loss_limited_bandwidth =
+        current_estimate_.loss_limited_bandwidth;
+  }
+
+  if (IsBandwidthLimitedDueToLoss()) {
+    // Bound the estimate increase if:
+    // 1. The estimate has been increased for less than
+    // `delayed_increase_window` ago, and
+    // 2. The best candidate is greater than bandwidth_limit_in_current_window.
+    if (recovering_after_loss_timestamp_.IsFinite() &&
+        recovering_after_loss_timestamp_ + config_->delayed_increase_window >
+            last_send_time_most_recent_observation_ &&
+        best_candidate.loss_limited_bandwidth >
+            bandwidth_limit_in_current_window_) {
+      best_candidate.loss_limited_bandwidth =
+          bandwidth_limit_in_current_window_;
+    }
+
+    bool increasing_when_loss_limited =
+        IsEstimateIncreasingWhenLossLimited(best_candidate);
+    // Bound the best candidate by the acked bitrate unless there is a recent
+    // probe result.
+    if (increasing_when_loss_limited && !IsValid(probe_bitrate_) &&
+        IsValid(acknowledged_bitrate_)) {
+      best_candidate.loss_limited_bandwidth =
+          IsValid(best_candidate.loss_limited_bandwidth)
+              ? std::min(best_candidate.loss_limited_bandwidth,
+                         config_->bandwidth_rampup_upper_bound_factor *
+                             (*acknowledged_bitrate_))
+              : config_->bandwidth_rampup_upper_bound_factor *
+                    (*acknowledged_bitrate_);
+    }
+
+    // Use probe bitrate as the estimate as probe bitrate is trusted to be
+    // correct. After being used, the probe bitrate is reset.
+    if (config_->probe_integration_enabled && IsValid(probe_bitrate_)) {
+      best_candidate.loss_limited_bandwidth =
+          std::min(probe_bitrate_, best_candidate.loss_limited_bandwidth);
+      probe_bitrate_ = DataRate::MinusInfinity();
+    }
+  }
+
+  if (IsEstimateIncreasingWhenLossLimited(best_candidate) &&
+      best_candidate.loss_limited_bandwidth < delay_based_estimate) {
+    current_state_ = LossBasedState::kIncreasing;
+  } else if (best_candidate.loss_limited_bandwidth < delay_based_estimate_) {
+    current_state_ = LossBasedState::kDecreasing;
+  } else if (best_candidate.loss_limited_bandwidth >= delay_based_estimate_) {
+    current_state_ = LossBasedState::kDelayBasedEstimate;
+  }
+  current_estimate_ = best_candidate;
+
+  if (IsBandwidthLimitedDueToLoss() &&
+      (recovering_after_loss_timestamp_.IsInfinite() ||
+       recovering_after_loss_timestamp_ + config_->delayed_increase_window <
+           last_send_time_most_recent_observation_)) {
+    bandwidth_limit_in_current_window_ =
+        std::max(kCongestionControllerMinBitrate,
+                 current_estimate_.loss_limited_bandwidth *
+                     config_->max_increase_factor);
+    recovering_after_loss_timestamp_ = last_send_time_most_recent_observation_;
+  }
+}
+
+bool LossBasedBweV2::IsEstimateIncreasingWhenLossLimited(
+    const ChannelParameters& best_candidate) {
+  return (current_estimate_.loss_limited_bandwidth <
+              best_candidate.loss_limited_bandwidth ||
+          (current_estimate_.loss_limited_bandwidth ==
+               best_candidate.loss_limited_bandwidth &&
+           current_state_ == LossBasedState::kIncreasing)) &&
+         IsBandwidthLimitedDueToLoss();
+}
+
+// Returns a `LossBasedBweV2::Config` iff the `key_value_config` specifies a
+// configuration for the `LossBasedBweV2` which is explicitly enabled.
+absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig(
+    const FieldTrialsView* key_value_config) {
+  FieldTrialParameter<bool> enabled("Enabled", true);
+  FieldTrialParameter<double> bandwidth_rampup_upper_bound_factor(
+      "BwRampupUpperBoundFactor", 1000000.0);
+  FieldTrialParameter<double> rampup_acceleration_max_factor(
+      "BwRampupAccelMaxFactor", 0.0);
+  FieldTrialParameter<TimeDelta> rampup_acceleration_maxout_time(
+      "BwRampupAccelMaxoutTime", TimeDelta::Seconds(60));
+  FieldTrialList<double> candidate_factors("CandidateFactors",
+                                           {1.02, 1.0, 0.95});
+  FieldTrialParameter<double> higher_bandwidth_bias_factor("HigherBwBiasFactor",
+                                                           0.0002);
+  FieldTrialParameter<double> higher_log_bandwidth_bias_factor(
+      "HigherLogBwBiasFactor", 0.02);
+  FieldTrialParameter<double> inherent_loss_lower_bound(
+      "InherentLossLowerBound", 1.0e-3);
+  FieldTrialParameter<double> loss_threshold_of_high_bandwidth_preference(
+      "LossThresholdOfHighBandwidthPreference", 0.15);
+  FieldTrialParameter<double> bandwidth_preference_smoothing_factor(
+      "BandwidthPreferenceSmoothingFactor", 0.002);
+  FieldTrialParameter<DataRate> inherent_loss_upper_bound_bandwidth_balance(
+      "InherentLossUpperBoundBwBalance", DataRate::KilobitsPerSec(75.0));
+  FieldTrialParameter<double> inherent_loss_upper_bound_offset(
+      "InherentLossUpperBoundOffset", 0.05);
+  FieldTrialParameter<double> initial_inherent_loss_estimate(
+      "InitialInherentLossEstimate", 0.01);
+  FieldTrialParameter<int> newton_iterations("NewtonIterations", 1);
+  FieldTrialParameter<double> newton_step_size("NewtonStepSize", 0.75);
+  FieldTrialParameter<bool> append_acknowledged_rate_candidate(
+      "AckedRateCandidate", true);
+  FieldTrialParameter<bool> append_delay_based_estimate_candidate(
+      "DelayBasedCandidate", true);
+  FieldTrialParameter<TimeDelta> observation_duration_lower_bound(
+      "ObservationDurationLowerBound", TimeDelta::Millis(250));
+  FieldTrialParameter<int> observation_window_size("ObservationWindowSize", 20);
+  FieldTrialParameter<double> sending_rate_smoothing_factor(
+      "SendingRateSmoothingFactor", 0.0);
+  FieldTrialParameter<double> instant_upper_bound_temporal_weight_factor(
+      "InstantUpperBoundTemporalWeightFactor", 0.9);
+  FieldTrialParameter<DataRate> instant_upper_bound_bandwidth_balance(
+      "InstantUpperBoundBwBalance", DataRate::KilobitsPerSec(75.0));
+  FieldTrialParameter<double> instant_upper_bound_loss_offset(
+      "InstantUpperBoundLossOffset", 0.05);
+  FieldTrialParameter<double> temporal_weight_factor("TemporalWeightFactor",
+                                                     0.9);
+  FieldTrialParameter<double> bandwidth_backoff_lower_bound_factor(
+      "BwBackoffLowerBoundFactor", 1.0);
+  FieldTrialParameter<bool> trendline_integration_enabled(
+      "TrendlineIntegrationEnabled", false);
+  FieldTrialParameter<int> trendline_observations_window_size(
+      "TrendlineObservationsWindowSize", 20);
+  FieldTrialParameter<double> max_increase_factor("MaxIncreaseFactor", 1.3);
+  FieldTrialParameter<TimeDelta> delayed_increase_window(
+      "DelayedIncreaseWindow", TimeDelta::Millis(300));
+  FieldTrialParameter<bool> use_acked_bitrate_only_when_overusing(
+      "UseAckedBitrateOnlyWhenOverusing", false);
+  FieldTrialParameter<bool>
+      not_increase_if_inherent_loss_less_than_average_loss(
+          "NotIncreaseIfInherentLossLessThanAverageLoss", true);
+  FieldTrialParameter<double> high_loss_rate_threshold("HighLossRateThreshold",
+                                                       1.0);
+  FieldTrialParameter<DataRate> bandwidth_cap_at_high_loss_rate(
+      "BandwidthCapAtHighLossRate", DataRate::KilobitsPerSec(500.0));
+  FieldTrialParameter<double> slope_of_bwe_high_loss_func(
+      "SlopeOfBweHighLossFunc", 1000);
+  FieldTrialParameter<bool> probe_integration_enabled("ProbeIntegrationEnabled",
+                                                      false);
+  FieldTrialParameter<bool> bound_by_upper_link_capacity_when_loss_limited(
+      "BoundByUpperLinkCapacityWhenLossLimited", true);
+  if (key_value_config) {
+    ParseFieldTrial({&enabled,
+                     &bandwidth_rampup_upper_bound_factor,
+                     &rampup_acceleration_max_factor,
+                     &rampup_acceleration_maxout_time,
+                     &candidate_factors,
+                     &higher_bandwidth_bias_factor,
+                     &higher_log_bandwidth_bias_factor,
+                     &inherent_loss_lower_bound,
+                     &loss_threshold_of_high_bandwidth_preference,
+                     &bandwidth_preference_smoothing_factor,
+                     &inherent_loss_upper_bound_bandwidth_balance,
+                     &inherent_loss_upper_bound_offset,
+                     &initial_inherent_loss_estimate,
+                     &newton_iterations,
+                     &newton_step_size,
+                     &append_acknowledged_rate_candidate,
+                     &append_delay_based_estimate_candidate,
+                     &observation_duration_lower_bound,
+                     &observation_window_size,
+                     &sending_rate_smoothing_factor,
+                     &instant_upper_bound_temporal_weight_factor,
+                     &instant_upper_bound_bandwidth_balance,
+                     &instant_upper_bound_loss_offset,
+                     &temporal_weight_factor,
+                     &bandwidth_backoff_lower_bound_factor,
+                     &trendline_integration_enabled,
+                     &trendline_observations_window_size,
+                     &max_increase_factor,
+                     &delayed_increase_window,
+                     &use_acked_bitrate_only_when_overusing,
+                     &not_increase_if_inherent_loss_less_than_average_loss,
+                     &probe_integration_enabled,
+                     &high_loss_rate_threshold,
+                     &bandwidth_cap_at_high_loss_rate,
+                     &slope_of_bwe_high_loss_func,
+                     &bound_by_upper_link_capacity_when_loss_limited},
+                    key_value_config->Lookup("WebRTC-Bwe-LossBasedBweV2"));
+  }
+
+  absl::optional<Config> config;
+  if (!enabled.Get()) {
+    return config;
+  }
+  config.emplace(Config());
+  config->bandwidth_rampup_upper_bound_factor =
+      bandwidth_rampup_upper_bound_factor.Get();
+  config->rampup_acceleration_max_factor = rampup_acceleration_max_factor.Get();
+  config->rampup_acceleration_maxout_time =
+      rampup_acceleration_maxout_time.Get();
+  config->candidate_factors = candidate_factors.Get();
+  config->higher_bandwidth_bias_factor = higher_bandwidth_bias_factor.Get();
+  config->higher_log_bandwidth_bias_factor =
+      higher_log_bandwidth_bias_factor.Get();
+  config->inherent_loss_lower_bound = inherent_loss_lower_bound.Get();
+  config->loss_threshold_of_high_bandwidth_preference =
+      loss_threshold_of_high_bandwidth_preference.Get();
+  config->bandwidth_preference_smoothing_factor =
+      bandwidth_preference_smoothing_factor.Get();
+  config->inherent_loss_upper_bound_bandwidth_balance =
+      inherent_loss_upper_bound_bandwidth_balance.Get();
+  config->inherent_loss_upper_bound_offset =
+      inherent_loss_upper_bound_offset.Get();
+  config->initial_inherent_loss_estimate = initial_inherent_loss_estimate.Get();
+  config->newton_iterations = newton_iterations.Get();
+  config->newton_step_size = newton_step_size.Get();
+  config->append_acknowledged_rate_candidate =
+      append_acknowledged_rate_candidate.Get();
+  config->append_delay_based_estimate_candidate =
+      append_delay_based_estimate_candidate.Get();
+  config->observation_duration_lower_bound =
+      observation_duration_lower_bound.Get();
+  config->observation_window_size = observation_window_size.Get();
+  config->sending_rate_smoothing_factor = sending_rate_smoothing_factor.Get();
+  config->instant_upper_bound_temporal_weight_factor =
+      instant_upper_bound_temporal_weight_factor.Get();
+  config->instant_upper_bound_bandwidth_balance =
+      instant_upper_bound_bandwidth_balance.Get();
+  config->instant_upper_bound_loss_offset =
+      instant_upper_bound_loss_offset.Get();
+  config->temporal_weight_factor = temporal_weight_factor.Get();
+  config->bandwidth_backoff_lower_bound_factor =
+      bandwidth_backoff_lower_bound_factor.Get();
+  config->trendline_integration_enabled = trendline_integration_enabled.Get();
+  config->trendline_observations_window_size =
+      trendline_observations_window_size.Get();
+  config->max_increase_factor = max_increase_factor.Get();
+  config->delayed_increase_window = delayed_increase_window.Get();
+  config->use_acked_bitrate_only_when_overusing =
+      use_acked_bitrate_only_when_overusing.Get();
+  config->not_increase_if_inherent_loss_less_than_average_loss =
+      not_increase_if_inherent_loss_less_than_average_loss.Get();
+  config->high_loss_rate_threshold = high_loss_rate_threshold.Get();
+  config->bandwidth_cap_at_high_loss_rate =
+      bandwidth_cap_at_high_loss_rate.Get();
+  config->slope_of_bwe_high_loss_func = slope_of_bwe_high_loss_func.Get();
+  config->probe_integration_enabled = probe_integration_enabled.Get();
+  config->bound_by_upper_link_capacity_when_loss_limited =
+      bound_by_upper_link_capacity_when_loss_limited.Get();
+
+  return config;
+}
+
+bool LossBasedBweV2::IsConfigValid() const {
+  if (!config_.has_value()) {
+    return false;
+  }
+
+  bool valid = true;
+
+  if (config_->bandwidth_rampup_upper_bound_factor <= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The bandwidth rampup upper bound factor must be greater than 1: "
+        << config_->bandwidth_rampup_upper_bound_factor;
+    valid = false;
+  }
+  if (config_->rampup_acceleration_max_factor < 0.0) {
+    RTC_LOG(LS_WARNING)
+        << "The rampup acceleration max factor must be non-negative.: "
+        << config_->rampup_acceleration_max_factor;
+    valid = false;
+  }
+  if (config_->rampup_acceleration_maxout_time <= TimeDelta::Zero()) {
+    RTC_LOG(LS_WARNING)
+        << "The rampup acceleration maxout time must be above zero: "
+        << config_->rampup_acceleration_maxout_time.seconds();
+    valid = false;
+  }
+  for (double candidate_factor : config_->candidate_factors) {
+    if (candidate_factor <= 0.0) {
+      RTC_LOG(LS_WARNING) << "All candidate factors must be greater than zero: "
+                          << candidate_factor;
+      valid = false;
+    }
+  }
+
+  // Ensure that the configuration allows generation of at least one candidate
+  // other than the current estimate.
+  if (!config_->append_acknowledged_rate_candidate &&
+      !config_->append_delay_based_estimate_candidate &&
+      !absl::c_any_of(config_->candidate_factors,
+                      [](double cf) { return cf != 1.0; })) {
+    RTC_LOG(LS_WARNING)
+        << "The configuration does not allow generating candidates. Specify "
+           "a candidate factor other than 1.0, allow the acknowledged rate "
+           "to be a candidate, and/or allow the delay based estimate to be a "
+           "candidate.";
+    valid = false;
+  }
+
+  if (config_->higher_bandwidth_bias_factor < 0.0) {
+    RTC_LOG(LS_WARNING)
+        << "The higher bandwidth bias factor must be non-negative: "
+        << config_->higher_bandwidth_bias_factor;
+    valid = false;
+  }
+  if (config_->inherent_loss_lower_bound < 0.0 ||
+      config_->inherent_loss_lower_bound >= 1.0) {
+    RTC_LOG(LS_WARNING) << "The inherent loss lower bound must be in [0, 1): "
+                        << config_->inherent_loss_lower_bound;
+    valid = false;
+  }
+  if (config_->loss_threshold_of_high_bandwidth_preference < 0.0 ||
+      config_->loss_threshold_of_high_bandwidth_preference >= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The loss threshold of high bandwidth preference must be in [0, 1): "
+        << config_->loss_threshold_of_high_bandwidth_preference;
+    valid = false;
+  }
+  if (config_->bandwidth_preference_smoothing_factor <= 0.0 ||
+      config_->bandwidth_preference_smoothing_factor > 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The bandwidth preference smoothing factor must be in (0, 1]: "
+        << config_->bandwidth_preference_smoothing_factor;
+    valid = false;
+  }
+  if (config_->inherent_loss_upper_bound_bandwidth_balance <=
+      DataRate::Zero()) {
+    RTC_LOG(LS_WARNING)
+        << "The inherent loss upper bound bandwidth balance "
+           "must be positive: "
+        << ToString(config_->inherent_loss_upper_bound_bandwidth_balance);
+    valid = false;
+  }
+  if (config_->inherent_loss_upper_bound_offset <
+          config_->inherent_loss_lower_bound ||
+      config_->inherent_loss_upper_bound_offset >= 1.0) {
+    RTC_LOG(LS_WARNING) << "The inherent loss upper bound must be greater "
+                           "than or equal to the inherent "
+                           "loss lower bound, which is "
+                        << config_->inherent_loss_lower_bound
+                        << ", and less than 1: "
+                        << config_->inherent_loss_upper_bound_offset;
+    valid = false;
+  }
+  if (config_->initial_inherent_loss_estimate < 0.0 ||
+      config_->initial_inherent_loss_estimate >= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The initial inherent loss estimate must be in [0, 1): "
+        << config_->initial_inherent_loss_estimate;
+    valid = false;
+  }
+  if (config_->newton_iterations <= 0) {
+    RTC_LOG(LS_WARNING) << "The number of Newton iterations must be positive: "
+                        << config_->newton_iterations;
+    valid = false;
+  }
+  if (config_->newton_step_size <= 0.0) {
+    RTC_LOG(LS_WARNING) << "The Newton step size must be positive: "
+                        << config_->newton_step_size;
+    valid = false;
+  }
+  if (config_->observation_duration_lower_bound <= TimeDelta::Zero()) {
+    RTC_LOG(LS_WARNING)
+        << "The observation duration lower bound must be positive: "
+        << ToString(config_->observation_duration_lower_bound);
+    valid = false;
+  }
+  if (config_->observation_window_size < 2) {
+    RTC_LOG(LS_WARNING) << "The observation window size must be at least 2: "
+                        << config_->observation_window_size;
+    valid = false;
+  }
+  if (config_->sending_rate_smoothing_factor < 0.0 ||
+      config_->sending_rate_smoothing_factor >= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The sending rate smoothing factor must be in [0, 1): "
+        << config_->sending_rate_smoothing_factor;
+    valid = false;
+  }
+  if (config_->instant_upper_bound_temporal_weight_factor <= 0.0 ||
+      config_->instant_upper_bound_temporal_weight_factor > 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The instant upper bound temporal weight factor must be in (0, 1]"
+        << config_->instant_upper_bound_temporal_weight_factor;
+    valid = false;
+  }
+  if (config_->instant_upper_bound_bandwidth_balance <= DataRate::Zero()) {
+    RTC_LOG(LS_WARNING)
+        << "The instant upper bound bandwidth balance must be positive: "
+        << ToString(config_->instant_upper_bound_bandwidth_balance);
+    valid = false;
+  }
+  if (config_->instant_upper_bound_loss_offset < 0.0 ||
+      config_->instant_upper_bound_loss_offset >= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The instant upper bound loss offset must be in [0, 1): "
+        << config_->instant_upper_bound_loss_offset;
+    valid = false;
+  }
+  if (config_->temporal_weight_factor <= 0.0 ||
+      config_->temporal_weight_factor > 1.0) {
+    RTC_LOG(LS_WARNING) << "The temporal weight factor must be in (0, 1]: "
+                        << config_->temporal_weight_factor;
+    valid = false;
+  }
+  if (config_->bandwidth_backoff_lower_bound_factor > 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The bandwidth backoff lower bound factor must not be greater than "
+           "1: "
+        << config_->bandwidth_backoff_lower_bound_factor;
+    valid = false;
+  }
+  if (config_->trendline_observations_window_size < 1) {
+    RTC_LOG(LS_WARNING) << "The trendline window size must be at least 1: "
+                        << config_->trendline_observations_window_size;
+    valid = false;
+  }
+  if (config_->max_increase_factor <= 0.0) {
+    RTC_LOG(LS_WARNING) << "The maximum increase factor must be positive: "
+                        << config_->max_increase_factor;
+    valid = false;
+  }
+  if (config_->delayed_increase_window <= TimeDelta::Zero()) {
+    RTC_LOG(LS_WARNING) << "The delayed increase window must be positive: "
+                        << config_->delayed_increase_window.ms();
+    valid = false;
+  }
+  if (config_->high_loss_rate_threshold <= 0.0 ||
+      config_->high_loss_rate_threshold > 1.0) {
+    RTC_LOG(LS_WARNING) << "The high loss rate threshold must be in (0, 1]: "
+                        << config_->high_loss_rate_threshold;
+    valid = false;
+  }
+  return valid;
+}
+
+double LossBasedBweV2::GetAverageReportedLossRatio() const {
+  if (num_observations_ <= 0) {
+    return 0.0;
+  }
+
+  double num_packets = 0;
+  double num_lost_packets = 0;
+  for (const Observation& observation : observations_) {
+    if (!observation.IsInitialized()) {
+      continue;
+    }
+
+    double instant_temporal_weight =
+        instant_upper_bound_temporal_weights_[(num_observations_ - 1) -
+                                              observation.id];
+    num_packets += instant_temporal_weight * observation.num_packets;
+    num_lost_packets += instant_temporal_weight * observation.num_lost_packets;
+  }
+
+  return num_lost_packets / num_packets;
+}
+
+DataRate LossBasedBweV2::GetCandidateBandwidthUpperBound() const {
+  DataRate candidate_bandwidth_upper_bound = max_bitrate_;
+  if (IsBandwidthLimitedDueToLoss() &&
+      IsValid(bandwidth_limit_in_current_window_)) {
+    candidate_bandwidth_upper_bound = bandwidth_limit_in_current_window_;
+  }
+
+  if (config_->trendline_integration_enabled) {
+    candidate_bandwidth_upper_bound =
+        std::min(GetInstantUpperBound(), candidate_bandwidth_upper_bound);
+    if (IsValid(delay_based_estimate_)) {
+      candidate_bandwidth_upper_bound =
+          std::min(delay_based_estimate_, candidate_bandwidth_upper_bound);
+    }
+  }
+
+  if (!acknowledged_bitrate_.has_value())
+    return candidate_bandwidth_upper_bound;
+
+  if (config_->rampup_acceleration_max_factor > 0.0) {
+    const TimeDelta time_since_bandwidth_reduced = std::min(
+        config_->rampup_acceleration_maxout_time,
+        std::max(TimeDelta::Zero(), last_send_time_most_recent_observation_ -
+                                        last_time_estimate_reduced_));
+    const double rampup_acceleration = config_->rampup_acceleration_max_factor *
+                                       time_since_bandwidth_reduced /
+                                       config_->rampup_acceleration_maxout_time;
+
+    candidate_bandwidth_upper_bound +=
+        rampup_acceleration * (*acknowledged_bitrate_);
+  }
+  return candidate_bandwidth_upper_bound;
+}
+
+std::vector<LossBasedBweV2::ChannelParameters> LossBasedBweV2::GetCandidates()
+    const {
+  std::vector<DataRate> bandwidths;
+  bool can_increase_bitrate = TrendlineEsimateAllowBitrateIncrease();
+  for (double candidate_factor : config_->candidate_factors) {
+    if (!can_increase_bitrate && candidate_factor > 1.0) {
+      continue;
+    }
+    bandwidths.push_back(candidate_factor *
+                         current_estimate_.loss_limited_bandwidth);
+  }
+
+  if (acknowledged_bitrate_.has_value() &&
+      config_->append_acknowledged_rate_candidate &&
+      TrendlineEsimateAllowEmergencyBackoff()) {
+    bandwidths.push_back(*acknowledged_bitrate_ *
+                         config_->bandwidth_backoff_lower_bound_factor);
+  }
+
+  if (IsValid(delay_based_estimate_) &&
+      config_->append_delay_based_estimate_candidate) {
+    if (can_increase_bitrate &&
+        delay_based_estimate_ > current_estimate_.loss_limited_bandwidth) {
+      bandwidths.push_back(delay_based_estimate_);
+    }
+  }
+
+  const DataRate candidate_bandwidth_upper_bound =
+      GetCandidateBandwidthUpperBound();
+
+  std::vector<ChannelParameters> candidates;
+  candidates.resize(bandwidths.size());
+  for (size_t i = 0; i < bandwidths.size(); ++i) {
+    ChannelParameters candidate = current_estimate_;
+    if (config_->trendline_integration_enabled) {
+      candidate.loss_limited_bandwidth =
+          std::min(bandwidths[i], candidate_bandwidth_upper_bound);
+    } else {
+      candidate.loss_limited_bandwidth = std::min(
+          bandwidths[i], std::max(current_estimate_.loss_limited_bandwidth,
+                                  candidate_bandwidth_upper_bound));
+    }
+    candidate.inherent_loss = GetFeasibleInherentLoss(candidate);
+    candidates[i] = candidate;
+  }
+  return candidates;
+}
+
+LossBasedBweV2::Derivatives LossBasedBweV2::GetDerivatives(
+    const ChannelParameters& channel_parameters) const {
+  Derivatives derivatives;
+
+  for (const Observation& observation : observations_) {
+    if (!observation.IsInitialized()) {
+      continue;
+    }
+
+    double loss_probability = GetLossProbability(
+        channel_parameters.inherent_loss,
+        channel_parameters.loss_limited_bandwidth, observation.sending_rate);
+
+    double temporal_weight =
+        temporal_weights_[(num_observations_ - 1) - observation.id];
+
+    derivatives.first +=
+        temporal_weight *
+        ((observation.num_lost_packets / loss_probability) -
+         (observation.num_received_packets / (1.0 - loss_probability)));
+    derivatives.second -=
+        temporal_weight *
+        ((observation.num_lost_packets / std::pow(loss_probability, 2)) +
+         (observation.num_received_packets /
+          std::pow(1.0 - loss_probability, 2)));
+  }
+
+  if (derivatives.second >= 0.0) {
+    RTC_LOG(LS_ERROR) << "The second derivative is mathematically guaranteed "
+                         "to be negative but is "
+                      << derivatives.second << ".";
+    derivatives.second = -1.0e-6;
+  }
+
+  return derivatives;
+}
+
+double LossBasedBweV2::GetFeasibleInherentLoss(
+    const ChannelParameters& channel_parameters) const {
+  return std::min(
+      std::max(channel_parameters.inherent_loss,
+               config_->inherent_loss_lower_bound),
+      GetInherentLossUpperBound(channel_parameters.loss_limited_bandwidth));
+}
+
+double LossBasedBweV2::GetInherentLossUpperBound(DataRate bandwidth) const {
+  if (bandwidth.IsZero()) {
+    return 1.0;
+  }
+
+  double inherent_loss_upper_bound =
+      config_->inherent_loss_upper_bound_offset +
+      config_->inherent_loss_upper_bound_bandwidth_balance / bandwidth;
+  return std::min(inherent_loss_upper_bound, 1.0);
+}
+
+double LossBasedBweV2::AdjustBiasFactor(double loss_rate,
+                                        double bias_factor) const {
+  return bias_factor *
+         (config_->loss_threshold_of_high_bandwidth_preference - loss_rate) /
+         (config_->bandwidth_preference_smoothing_factor +
+          std::abs(config_->loss_threshold_of_high_bandwidth_preference -
+                   loss_rate));
+}
+
+double LossBasedBweV2::GetHighBandwidthBias(DataRate bandwidth) const {
+  if (IsValid(bandwidth)) {
+    const double average_reported_loss_ratio = GetAverageReportedLossRatio();
+    return AdjustBiasFactor(average_reported_loss_ratio,
+                            config_->higher_bandwidth_bias_factor) *
+               bandwidth.kbps() +
+           AdjustBiasFactor(average_reported_loss_ratio,
+                            config_->higher_log_bandwidth_bias_factor) *
+               std::log(1.0 + bandwidth.kbps());
+  }
+  return 0.0;
+}
+
+double LossBasedBweV2::GetObjective(
+    const ChannelParameters& channel_parameters) const {
+  double objective = 0.0;
+
+  const double high_bandwidth_bias =
+      GetHighBandwidthBias(channel_parameters.loss_limited_bandwidth);
+
+  for (const Observation& observation : observations_) {
+    if (!observation.IsInitialized()) {
+      continue;
+    }
+
+    double loss_probability = GetLossProbability(
+        channel_parameters.inherent_loss,
+        channel_parameters.loss_limited_bandwidth, observation.sending_rate);
+
+    double temporal_weight =
+        temporal_weights_[(num_observations_ - 1) - observation.id];
+
+    objective +=
+        temporal_weight *
+        ((observation.num_lost_packets * std::log(loss_probability)) +
+         (observation.num_received_packets * std::log(1.0 - loss_probability)));
+    objective +=
+        temporal_weight * high_bandwidth_bias * observation.num_packets;
+  }
+
+  return objective;
+}
+
+DataRate LossBasedBweV2::GetSendingRate(
+    DataRate instantaneous_sending_rate) const {
+  if (num_observations_ <= 0) {
+    return instantaneous_sending_rate;
+  }
+
+  const int most_recent_observation_idx =
+      (num_observations_ - 1) % config_->observation_window_size;
+  const Observation& most_recent_observation =
+      observations_[most_recent_observation_idx];
+  DataRate sending_rate_previous_observation =
+      most_recent_observation.sending_rate;
+
+  return config_->sending_rate_smoothing_factor *
+             sending_rate_previous_observation +
+         (1.0 - config_->sending_rate_smoothing_factor) *
+             instantaneous_sending_rate;
+}
+
+DataRate LossBasedBweV2::GetInstantUpperBound() const {
+  return cached_instant_upper_bound_.value_or(max_bitrate_);
+}
+
+void LossBasedBweV2::CalculateInstantUpperBound() {
+  DataRate instant_limit = max_bitrate_;
+  const double average_reported_loss_ratio = GetAverageReportedLossRatio();
+  if (average_reported_loss_ratio > config_->instant_upper_bound_loss_offset) {
+    instant_limit = config_->instant_upper_bound_bandwidth_balance /
+                    (average_reported_loss_ratio -
+                     config_->instant_upper_bound_loss_offset);
+    if (average_reported_loss_ratio > config_->high_loss_rate_threshold) {
+      instant_limit = std::min(
+          instant_limit, DataRate::KilobitsPerSec(std::max(
+                             static_cast<double>(min_bitrate_.kbps()),
+                             config_->bandwidth_cap_at_high_loss_rate.kbps() -
+                                 config_->slope_of_bwe_high_loss_func *
+                                     average_reported_loss_ratio)));
+    }
+  }
+
+  if (IsBandwidthLimitedDueToLoss()) {
+    if (IsValid(upper_link_capacity_) &&
+        config_->bound_by_upper_link_capacity_when_loss_limited) {
+      instant_limit = std::min(instant_limit, upper_link_capacity_);
+    }
+  }
+  cached_instant_upper_bound_ = instant_limit;
+}
+
+void LossBasedBweV2::CalculateTemporalWeights() {
+  for (int i = 0; i < config_->observation_window_size; ++i) {
+    temporal_weights_[i] = std::pow(config_->temporal_weight_factor, i);
+    instant_upper_bound_temporal_weights_[i] =
+        std::pow(config_->instant_upper_bound_temporal_weight_factor, i);
+  }
+}
+
+void LossBasedBweV2::NewtonsMethodUpdate(
+    ChannelParameters& channel_parameters) const {
+  if (num_observations_ <= 0) {
+    return;
+  }
+
+  for (int i = 0; i < config_->newton_iterations; ++i) {
+    const Derivatives derivatives = GetDerivatives(channel_parameters);
+    channel_parameters.inherent_loss -=
+        config_->newton_step_size * derivatives.first / derivatives.second;
+    channel_parameters.inherent_loss =
+        GetFeasibleInherentLoss(channel_parameters);
+  }
+}
+
+bool LossBasedBweV2::TrendlineEsimateAllowBitrateIncrease() const {
+  if (!config_->trendline_integration_enabled) {
+    return true;
+  }
+
+  for (const auto& detector_state : delay_detector_states_) {
+    if (detector_state == BandwidthUsage::kBwOverusing ||
+        detector_state == BandwidthUsage::kBwUnderusing) {
+      return false;
+    }
+  }
+  return true;
+}
+
+bool LossBasedBweV2::TrendlineEsimateAllowEmergencyBackoff() const {
+  if (!config_->trendline_integration_enabled) {
+    return true;
+  }
+
+  if (!config_->use_acked_bitrate_only_when_overusing) {
+    return true;
+  }
+
+  for (const auto& detector_state : delay_detector_states_) {
+    if (detector_state == BandwidthUsage::kBwOverusing) {
+      return true;
+    }
+  }
+
+  return false;
+}
+
+bool LossBasedBweV2::PushBackObservation(
+    rtc::ArrayView<const PacketResult> packet_results,
+    BandwidthUsage delay_detector_state) {
+  delay_detector_states_.push_front(delay_detector_state);
+  if (static_cast<int>(delay_detector_states_.size()) >
+      config_->trendline_observations_window_size) {
+    delay_detector_states_.pop_back();
+  }
+
+  if (packet_results.empty()) {
+    return false;
+  }
+
+  PacketResultsSummary packet_results_summary =
+      GetPacketResultsSummary(packet_results);
+
+  partial_observation_.num_packets += packet_results_summary.num_packets;
+  partial_observation_.num_lost_packets +=
+      packet_results_summary.num_lost_packets;
+  partial_observation_.size += packet_results_summary.total_size;
+
+  // This is the first packet report we have received.
+  if (!IsValid(last_send_time_most_recent_observation_)) {
+    last_send_time_most_recent_observation_ =
+        packet_results_summary.first_send_time;
+  }
+
+  const Timestamp last_send_time = packet_results_summary.last_send_time;
+  const TimeDelta observation_duration =
+      last_send_time - last_send_time_most_recent_observation_;
+  // Too small to be meaningful.
+  if (observation_duration <= TimeDelta::Zero() ||
+      (observation_duration < config_->observation_duration_lower_bound &&
+       (delay_detector_state != BandwidthUsage::kBwOverusing ||
+        !config_->trendline_integration_enabled))) {
+    return false;
+  }
+
+  last_send_time_most_recent_observation_ = last_send_time;
+
+  Observation observation;
+  observation.num_packets = partial_observation_.num_packets;
+  observation.num_lost_packets = partial_observation_.num_lost_packets;
+  observation.num_received_packets =
+      observation.num_packets - observation.num_lost_packets;
+  observation.sending_rate =
+      GetSendingRate(partial_observation_.size / observation_duration);
+  observation.id = num_observations_++;
+  observations_[observation.id % config_->observation_window_size] =
+      observation;
+
+  partial_observation_ = PartialObservation();
+
+  CalculateInstantUpperBound();
+  return true;
+}
+
+bool LossBasedBweV2::IsBandwidthLimitedDueToLoss() const {
+  return current_state_ != LossBasedState::kDelayBasedEstimate;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
new file mode 100644
index 0000000000..9ff9cb74c6
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
@@ -0,0 +1,203 @@
+/*
+ *  Copyright 2021 The WebRTC project authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_
+
+#include <cstddef>
+#include <deque>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+
+// State of the loss based estimate, which can be either increasing/decreasing
+// when network is loss limited, or equal to the delay based estimate.
+enum class LossBasedState {
+  kIncreasing = 0,
+  kDecreasing = 1,
+  kDelayBasedEstimate = 2
+};
+
+class LossBasedBweV2 {
+ public:
+  struct Result {
+    ~Result() = default;
+    DataRate bandwidth_estimate = DataRate::Zero();
+    LossBasedState state = LossBasedState::kDelayBasedEstimate;
+  };
+  // Creates a disabled `LossBasedBweV2` if the
+  // `key_value_config` is not valid.
+  explicit LossBasedBweV2(const FieldTrialsView* key_value_config);
+
+  LossBasedBweV2(const LossBasedBweV2&) = delete;
+  LossBasedBweV2& operator=(const LossBasedBweV2&) = delete;
+
+  ~LossBasedBweV2() = default;
+
+  bool IsEnabled() const;
+  // Returns true iff a BWE can be calculated, i.e., the estimator has been
+  // initialized with a BWE and then has received enough `PacketResult`s.
+  bool IsReady() const;
+
+  // Returns `DataRate::PlusInfinity` if no BWE can be calculated.
+  Result GetLossBasedResult() const;
+
+  void SetAcknowledgedBitrate(DataRate acknowledged_bitrate);
+  void SetBandwidthEstimate(DataRate bandwidth_estimate);
+  void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate);
+  void UpdateBandwidthEstimate(
+      rtc::ArrayView<const PacketResult> packet_results,
+      DataRate delay_based_estimate,
+      BandwidthUsage delay_detector_state,
+      absl::optional<DataRate> probe_bitrate,
+      DataRate upper_link_capacity);
+
+ private:
+  struct ChannelParameters {
+    double inherent_loss = 0.0;
+    DataRate loss_limited_bandwidth = DataRate::MinusInfinity();
+  };
+
+  struct Config {
+    double bandwidth_rampup_upper_bound_factor = 0.0;
+    double rampup_acceleration_max_factor = 0.0;
+    TimeDelta rampup_acceleration_maxout_time = TimeDelta::Zero();
+    std::vector<double> candidate_factors;
+    double higher_bandwidth_bias_factor = 0.0;
+    double higher_log_bandwidth_bias_factor = 0.0;
+    double inherent_loss_lower_bound = 0.0;
+    double loss_threshold_of_high_bandwidth_preference = 0.0;
+    double bandwidth_preference_smoothing_factor = 0.0;
+    DataRate inherent_loss_upper_bound_bandwidth_balance =
+        DataRate::MinusInfinity();
+    double inherent_loss_upper_bound_offset = 0.0;
+    double initial_inherent_loss_estimate = 0.0;
+    int newton_iterations = 0;
+    double newton_step_size = 0.0;
+    bool append_acknowledged_rate_candidate = true;
+    bool append_delay_based_estimate_candidate = false;
+    TimeDelta observation_duration_lower_bound = TimeDelta::Zero();
+    int observation_window_size = 0;
+    double sending_rate_smoothing_factor = 0.0;
+    double instant_upper_bound_temporal_weight_factor = 0.0;
+    DataRate instant_upper_bound_bandwidth_balance = DataRate::MinusInfinity();
+    double instant_upper_bound_loss_offset = 0.0;
+    double temporal_weight_factor = 0.0;
+    double bandwidth_backoff_lower_bound_factor = 0.0;
+    bool trendline_integration_enabled = false;
+    int trendline_observations_window_size = 0;
+    double max_increase_factor = 0.0;
+    TimeDelta delayed_increase_window = TimeDelta::Zero();
+    bool use_acked_bitrate_only_when_overusing = false;
+    bool not_increase_if_inherent_loss_less_than_average_loss = false;
+    double high_loss_rate_threshold = 1.0;
+    DataRate bandwidth_cap_at_high_loss_rate = DataRate::MinusInfinity();
+    double slope_of_bwe_high_loss_func = 1000.0;
+    bool probe_integration_enabled = false;
+    bool bound_by_upper_link_capacity_when_loss_limited = false;
+  };
+
+  struct Derivatives {
+    double first = 0.0;
+    double second = 0.0;
+  };
+
+  struct Observation {
+    bool IsInitialized() const { return id != -1; }
+
+    int num_packets = 0;
+    int num_lost_packets = 0;
+    int num_received_packets = 0;
+    DataRate sending_rate = DataRate::MinusInfinity();
+    int id = -1;
+  };
+
+  struct PartialObservation {
+    int num_packets = 0;
+    int num_lost_packets = 0;
+    DataSize size = DataSize::Zero();
+  };
+
+  static absl::optional<Config> CreateConfig(
+      const FieldTrialsView* key_value_config);
+  bool IsConfigValid() const;
+
+  // Returns `0.0` if not enough loss statistics have been received.
+  double GetAverageReportedLossRatio() const;
+  std::vector<ChannelParameters> GetCandidates() const;
+  DataRate GetCandidateBandwidthUpperBound() const;
+  Derivatives GetDerivatives(const ChannelParameters& channel_parameters) const;
+  double GetFeasibleInherentLoss(
+      const ChannelParameters& channel_parameters) const;
+  double GetInherentLossUpperBound(DataRate bandwidth) const;
+  double AdjustBiasFactor(double loss_rate, double bias_factor) const;
+  double GetHighBandwidthBias(DataRate bandwidth) const;
+  double GetObjective(const ChannelParameters& channel_parameters) const;
+  DataRate GetSendingRate(DataRate instantaneous_sending_rate) const;
+  DataRate GetInstantUpperBound() const;
+  void CalculateInstantUpperBound();
+
+  void CalculateTemporalWeights();
+  void NewtonsMethodUpdate(ChannelParameters& channel_parameters) const;
+
+  // Returns false if there exists a kBwOverusing or kBwUnderusing in the
+  // window.
+  bool TrendlineEsimateAllowBitrateIncrease() const;
+
+  // Returns true if there exists an overusing state in the window.
+  bool TrendlineEsimateAllowEmergencyBackoff() const;
+
+  // Returns false if no observation was created.
+  bool PushBackObservation(rtc::ArrayView<const PacketResult> packet_results,
+                           BandwidthUsage delay_detector_state);
+  void UpdateTrendlineEstimator(
+      const std::vector<PacketResult>& packet_feedbacks,
+      Timestamp at_time);
+  void UpdateDelayDetector(BandwidthUsage delay_detector_state);
+  bool IsEstimateIncreasingWhenLossLimited(
+      const ChannelParameters& best_candidate);
+  bool IsBandwidthLimitedDueToLoss() const;
+  void SetProbeBitrate(absl::optional<DataRate> probe_bitrate);
+
+  absl::optional<DataRate> acknowledged_bitrate_;
+  absl::optional<Config> config_;
+  ChannelParameters current_estimate_;
+  int num_observations_ = 0;
+  std::vector<Observation> observations_;
+  PartialObservation partial_observation_;
+  Timestamp last_send_time_most_recent_observation_ = Timestamp::PlusInfinity();
+  Timestamp last_time_estimate_reduced_ = Timestamp::MinusInfinity();
+  absl::optional<DataRate> cached_instant_upper_bound_;
+  std::vector<double> instant_upper_bound_temporal_weights_;
+  std::vector<double> temporal_weights_;
+  std::deque<BandwidthUsage> delay_detector_states_;
+  Timestamp recovering_after_loss_timestamp_ = Timestamp::MinusInfinity();
+  DataRate bandwidth_limit_in_current_window_ = DataRate::PlusInfinity();
+  DataRate min_bitrate_ = DataRate::KilobitsPerSec(1);
+  DataRate max_bitrate_ = DataRate::PlusInfinity();
+  LossBasedState current_state_ = LossBasedState::kDelayBasedEstimate;
+  DataRate probe_bitrate_ = DataRate::PlusInfinity();
+  DataRate delay_based_estimate_ = DataRate::PlusInfinity();
+  DataRate upper_link_capacity_ = DataRate::PlusInfinity();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build
new file mode 100644
index 0000000000..5728e0c4b2
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("loss_based_bwe_v2_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc
new file mode 100644
index 0000000000..c303c29d68
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc
@@ -0,0 +1,1526 @@
+/*
+ *  Copyright 2021 The WebRTC project authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+
+#include <string>
+#include <vector>
+
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using ::webrtc::test::ExplicitKeyValueConfig;
+
+constexpr TimeDelta kObservationDurationLowerBound = TimeDelta::Millis(200);
+constexpr TimeDelta kDelayedIncreaseWindow = TimeDelta::Millis(300);
+constexpr double kMaxIncreaseFactor = 1.5;
+
+class LossBasedBweV2Test : public ::testing::TestWithParam<bool> {
+ protected:
+  std::string Config(bool enabled,
+                     bool valid,
+                     bool trendline_integration_enabled) {
+    char buffer[1024];
+    rtc::SimpleStringBuilder config_string(buffer);
+
+    config_string << "WebRTC-Bwe-LossBasedBweV2/";
+
+    if (enabled) {
+      config_string << "Enabled:true";
+    } else {
+      config_string << "Enabled:false";
+    }
+
+    if (valid) {
+      config_string << ",BwRampupUpperBoundFactor:1.2";
+    } else {
+      config_string << ",BwRampupUpperBoundFactor:0.0";
+    }
+
+    if (trendline_integration_enabled) {
+      config_string << ",TrendlineIntegrationEnabled:true";
+    } else {
+      config_string << ",TrendlineIntegrationEnabled:false";
+    }
+
+    config_string
+        << ",CandidateFactors:1.1|1.0|0.95,HigherBwBiasFactor:0.01,"
+           "DelayBasedCandidate:true,"
+           "InherentLossLowerBound:0.001,InherentLossUpperBoundBwBalance:"
+           "14kbps,"
+           "InherentLossUpperBoundOffset:0.9,InitialInherentLossEstimate:0.01,"
+           "NewtonIterations:2,NewtonStepSize:0.4,ObservationWindowSize:15,"
+           "SendingRateSmoothingFactor:0.01,"
+           "InstantUpperBoundTemporalWeightFactor:0.97,"
+           "InstantUpperBoundBwBalance:90kbps,"
+           "InstantUpperBoundLossOffset:0.1,TemporalWeightFactor:0.98";
+
+    config_string.AppendFormat(
+        ",ObservationDurationLowerBound:%dms",
+        static_cast<int>(kObservationDurationLowerBound.ms()));
+    config_string.AppendFormat(",MaxIncreaseFactor:%f", kMaxIncreaseFactor);
+    config_string.AppendFormat(",DelayedIncreaseWindow:%dms",
+                               static_cast<int>(kDelayedIncreaseWindow.ms()));
+
+    config_string << "/";
+
+    return config_string.str();
+  }
+
+  std::vector<PacketResult> CreatePacketResultsWithReceivedPackets(
+      Timestamp first_packet_timestamp) {
+    std::vector<PacketResult> enough_feedback(2);
+    enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[1].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[0].sent_packet.send_time = first_packet_timestamp;
+    enough_feedback[1].sent_packet.send_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[0].receive_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[1].receive_time =
+        first_packet_timestamp + 2 * kObservationDurationLowerBound;
+    return enough_feedback;
+  }
+
+  std::vector<PacketResult> CreatePacketResultsWith10pLossRate(
+      Timestamp first_packet_timestamp) {
+    std::vector<PacketResult> enough_feedback(10);
+    enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+    for (unsigned i = 0; i < enough_feedback.size(); ++i) {
+      enough_feedback[i].sent_packet.size = DataSize::Bytes(15'000);
+      enough_feedback[i].sent_packet.send_time =
+          first_packet_timestamp +
+          static_cast<int>(i) * kObservationDurationLowerBound;
+      enough_feedback[i].receive_time =
+          first_packet_timestamp +
+          static_cast<int>(i + 1) * kObservationDurationLowerBound;
+    }
+    enough_feedback[9].receive_time = Timestamp::PlusInfinity();
+    return enough_feedback;
+  }
+
+  std::vector<PacketResult> CreatePacketResultsWith50pLossRate(
+      Timestamp first_packet_timestamp) {
+    std::vector<PacketResult> enough_feedback(2);
+    enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[1].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[0].sent_packet.send_time = first_packet_timestamp;
+    enough_feedback[1].sent_packet.send_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[0].receive_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[1].receive_time = Timestamp::PlusInfinity();
+    return enough_feedback;
+  }
+
+  std::vector<PacketResult> CreatePacketResultsWith100pLossRate(
+      Timestamp first_packet_timestamp) {
+    std::vector<PacketResult> enough_feedback(2);
+    enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[1].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[0].sent_packet.send_time = first_packet_timestamp;
+    enough_feedback[1].sent_packet.send_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[0].receive_time = Timestamp::PlusInfinity();
+    enough_feedback[1].receive_time = Timestamp::PlusInfinity();
+    return enough_feedback;
+  }
+};
+
+TEST_P(LossBasedBweV2Test, EnabledWhenGivenValidConfigurationValues) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  EXPECT_TRUE(loss_based_bandwidth_estimator.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test, DisabledWhenGivenDisabledConfiguration) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/false, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test, DisabledWhenGivenNonValidConfigurationValues) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/false,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test, DisabledWhenGivenNonPositiveCandidateFactor) {
+  ExplicitKeyValueConfig key_value_config_negative_candidate_factor(
+      "WebRTC-Bwe-LossBasedBweV2/Enabled:true,CandidateFactors:-1.3|1.1/");
+  LossBasedBweV2 loss_based_bandwidth_estimator_1(
+      &key_value_config_negative_candidate_factor);
+  EXPECT_FALSE(loss_based_bandwidth_estimator_1.IsEnabled());
+
+  ExplicitKeyValueConfig key_value_config_zero_candidate_factor(
+      "WebRTC-Bwe-LossBasedBweV2/Enabled:true,CandidateFactors:0.0|1.1/");
+  LossBasedBweV2 loss_based_bandwidth_estimator_2(
+      &key_value_config_zero_candidate_factor);
+  EXPECT_FALSE(loss_based_bandwidth_estimator_2.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test,
+       DisabledWhenGivenConfigurationThatDoesNotAllowGeneratingCandidates) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.0,AckedRateCandidate:false,"
+      "DelayBasedCandidate:false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test, ReturnsDelayBasedEstimateWhenDisabled) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/false, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      /*packet_results=*/{},
+      /*delay_based_estimate=*/DataRate::KilobitsPerSec(100),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(100));
+}
+
+TEST_P(LossBasedBweV2Test,
+       ReturnsDelayBasedEstimateWhenWhenGivenNonValidConfigurationValues) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/false,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      /*packet_results=*/{},
+      /*delay_based_estimate=*/DataRate::KilobitsPerSec(100),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(100));
+}
+
+TEST_P(LossBasedBweV2Test,
+       BandwidthEstimateGivenInitializationAndThenFeedback) {
+  std::vector<PacketResult> enough_feedback =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_TRUE(loss_based_bandwidth_estimator.IsReady());
+  EXPECT_TRUE(loss_based_bandwidth_estimator.GetLossBasedResult()
+                  .bandwidth_estimate.IsFinite());
+}
+
+TEST_P(LossBasedBweV2Test, NoBandwidthEstimateGivenNoInitialization) {
+  std::vector<PacketResult> enough_feedback =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsReady());
+  EXPECT_TRUE(loss_based_bandwidth_estimator.GetLossBasedResult()
+                  .bandwidth_estimate.IsPlusInfinity());
+}
+
+TEST_P(LossBasedBweV2Test, NoBandwidthEstimateGivenNotEnoughFeedback) {
+  // Create packet results where the observation duration is less than the lower
+  // bound.
+  PacketResult not_enough_feedback[2];
+  not_enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+  not_enough_feedback[1].sent_packet.size = DataSize::Bytes(15'000);
+  not_enough_feedback[0].sent_packet.send_time = Timestamp::Zero();
+  not_enough_feedback[1].sent_packet.send_time =
+      Timestamp::Zero() + kObservationDurationLowerBound / 2;
+  not_enough_feedback[0].receive_time =
+      Timestamp::Zero() + kObservationDurationLowerBound / 2;
+  not_enough_feedback[1].receive_time =
+      Timestamp::Zero() + kObservationDurationLowerBound;
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsReady());
+  EXPECT_TRUE(loss_based_bandwidth_estimator.GetLossBasedResult()
+                  .bandwidth_estimate.IsPlusInfinity());
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      not_enough_feedback, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsReady());
+  EXPECT_TRUE(loss_based_bandwidth_estimator.GetLossBasedResult()
+                  .bandwidth_estimate.IsPlusInfinity());
+}
+
+TEST_P(LossBasedBweV2Test,
+       SetValueIsTheEstimateUntilAdditionalFeedbackHasBeenReceived) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_NE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_NE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+TEST_P(LossBasedBweV2Test,
+       SetAcknowledgedBitrateOnlyAffectsTheBweWhenAdditionalFeedbackIsGiven) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator_1(&key_value_config);
+  LossBasedBweV2 loss_based_bandwidth_estimator_2(&key_value_config);
+
+  loss_based_bandwidth_estimator_1.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator_2.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator_1.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  loss_based_bandwidth_estimator_2.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator_1.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(660));
+
+  loss_based_bandwidth_estimator_1.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(900));
+
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator_1.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(660));
+
+  loss_based_bandwidth_estimator_1.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  loss_based_bandwidth_estimator_2.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_NE(
+      loss_based_bandwidth_estimator_1.GetLossBasedResult().bandwidth_estimate,
+      loss_based_bandwidth_estimator_2.GetLossBasedResult().bandwidth_estimate);
+}
+
+TEST_P(LossBasedBweV2Test,
+       BandwidthEstimateIsCappedToBeTcpFairGivenTooHighLossRate) {
+  std::vector<PacketResult> enough_feedback_no_received_packets =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_no_received_packets,
+      /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(100));
+}
+
+TEST_P(LossBasedBweV2Test, BandwidthEstimateNotIncreaseWhenNetworkUnderusing) {
+  if (!GetParam()) {
+    GTEST_SKIP() << "This test should run only if "
+                    "trendline_integration_enabled is enabled";
+  }
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwUnderusing, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_LE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_LE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+// When network is normal, estimate can increase but never be higher than
+// the delay based estimate.
+TEST_P(LossBasedBweV2Test,
+       BandwidthEstimateCappedByDelayBasedEstimateWhenNetworkNormal) {
+  // Create two packet results, network is in normal state, 100% packets are
+  // received, and no delay increase.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // If the delay based estimate is infinity, then loss based estimate increases
+  // and not bounded by delay based estimate.
+  EXPECT_GT(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::KilobitsPerSec(500),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // If the delay based estimate is not infinity, then loss based estimate is
+  // bounded by delay based estimate.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(500));
+}
+
+// When loss based bwe receives a strong signal of overusing and an increase in
+// loss rate, it should acked bitrate for emegency backoff.
+TEST_P(LossBasedBweV2Test, UseAckedBitrateForEmegencyBackOff) {
+  // Create two packet results, first packet has 50% loss rate, second packet
+  // has 100% loss rate.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  DataRate acked_bitrate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_bitrate);
+  // Update estimate when network is overusing, and 50% loss rate.
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwOverusing,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // Update estimate again when network is continuously overusing, and 100%
+  // loss rate.
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwOverusing,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // The estimate bitrate now is backed off based on acked bitrate.
+  EXPECT_LE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      acked_bitrate);
+}
+
+// When receiving the same packet feedback, loss based bwe ignores the feedback
+// and returns the current estimate.
+TEST_P(LossBasedBweV2Test, NoBweChangeIfObservationDurationUnchanged) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_1 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  // Use the same feedback and check if the estimate is unchanged.
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_EQ(estimate_2, estimate_1);
+}
+
+// When receiving feedback of packets that were sent within an observation
+// duration, and network is in the normal state, loss based bwe returns the
+// current estimate.
+TEST_P(LossBasedBweV2Test,
+       NoBweChangeIfObservationDurationIsSmallAndNetworkNormal) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound - TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_1 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_EQ(estimate_2, estimate_1);
+}
+
+// When receiving feedback of packets that were sent within an observation
+// duration, and network is in the underusing state, loss based bwe returns the
+// current estimate.
+TEST_P(LossBasedBweV2Test,
+       NoBweIncreaseIfObservationDurationIsSmallAndNetworkUnderusing) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound - TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_1 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwUnderusing, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_LE(estimate_2, estimate_1);
+}
+
+// When receiving feedback of packets that were sent within an observation
+// duration, network is overusing, and trendline integration is enabled, loss
+// based bwe updates its estimate.
+TEST_P(LossBasedBweV2Test,
+       UpdateEstimateIfObservationDurationIsSmallAndNetworkOverusing) {
+  if (!GetParam()) {
+    GTEST_SKIP() << "This test should run only if "
+                    "trendline_integration_enabled is enabled";
+  }
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound - TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_1 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwOverusing,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_LT(estimate_2, estimate_1);
+}
+
+TEST_P(LossBasedBweV2Test,
+       IncreaseToDelayBasedEstimateIfNoLossOrDelayIncrease) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      delay_based_estimate);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      delay_based_estimate);
+}
+
+TEST_P(LossBasedBweV2Test,
+       IncreaseByMaxIncreaseFactorAfterLossBasedBweBacksOff) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1.5,BwRampupUpperBoundFactor:"
+      "2.0,NotIncreaseIfInherentLossLessThanAverageLoss:false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  LossBasedBweV2::Result result_at_loss =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+
+  // Network recovers after loss.
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  LossBasedBweV2::Result result_after_recovery =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_EQ(result_after_recovery.bandwidth_estimate,
+            result_at_loss.bandwidth_estimate * 1.5);
+}
+
+TEST_P(LossBasedBweV2Test,
+       LossBasedStateIsDelayBasedEstimateAfterNetworkRecovering) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:100|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:100,"
+      "BwRampupUpperBoundFactor:"
+      "2.0,NotIncreaseIfInherentLossLessThanAverageLoss:false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(600);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  ASSERT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDecreasing);
+
+  // Network recovers after loss.
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDelayBasedEstimate);
+
+  // Network recovers continuing.
+  std::vector<PacketResult> enough_feedback_3 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound * 2);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_3, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDelayBasedEstimate);
+}
+
+TEST_P(LossBasedBweV2Test,
+       LossBasedStateIsNotDelayBasedEstimateIfDelayBasedEsimtateInfinite) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:100|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:100,"
+      "BwRampupUpperBoundFactor:"
+      "2.0/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::PlusInfinity();
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  ASSERT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDecreasing);
+
+  // Network recovers after loss.
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_NE(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDelayBasedEstimate);
+}
+
+// After loss based bwe backs off, the next estimate is capped by
+// a factor of acked bitrate.
+TEST_P(LossBasedBweV2Test,
+       IncreaseByFactorOfAckedBitrateAfterLossBasedBweBacksOff) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,LossThresholdOfHighBandwidthPreference:0.99,"
+      "BwRampupUpperBoundFactor:1.2,"
+      "InherentLossUpperBoundOffset:0.9,ObservationDurationLowerBound:200ms/");
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Change the acked bitrate to make sure that the estimate is bounded by a
+  // factor of acked bitrate.
+  DataRate acked_bitrate = DataRate::KilobitsPerSec(50);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_bitrate);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // The estimate is capped by acked_bitrate * BwRampupUpperBoundFactor.
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_EQ(estimate_2, acked_bitrate * 1.2);
+}
+
+// After loss based bwe backs off, the estimate is bounded during the delayed
+// window.
+TEST_P(LossBasedBweV2Test,
+       EstimateBitrateIsBoundedDuringDelayedWindowAfterLossBasedBweBacksOff) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kDelayedIncreaseWindow - TimeDelta::Millis(2));
+  std::vector<PacketResult> enough_feedback_3 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kDelayedIncreaseWindow - TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // Increase the acknowledged bitrate to make sure that the estimate is not
+  // capped too low.
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(5000));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // The estimate is capped by current_estimate * kMaxIncreaseFactor because
+  // it recently backed off.
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_3, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // The latest estimate is the same as the previous estimate since the sent
+  // packets were sent within the DelayedIncreaseWindow.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      estimate_2);
+}
+
+// The estimate is not bounded after the delayed increase window.
+TEST_P(LossBasedBweV2Test, KeepIncreasingEstimateAfterDelayedIncreaseWindow) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kDelayedIncreaseWindow - TimeDelta::Millis(1));
+  std::vector<PacketResult> enough_feedback_3 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kDelayedIncreaseWindow + TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // Increase the acknowledged bitrate to make sure that the estimate is not
+  // capped too low.
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(5000));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // The estimate is capped by current_estimate * kMaxIncreaseFactor because it
+  // recently backed off.
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_3, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // The estimate can continue increasing after the DelayedIncreaseWindow.
+  EXPECT_GE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      estimate_2);
+}
+
+TEST_P(LossBasedBweV2Test, NotIncreaseIfInherentLossLessThanAverageLoss) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,"
+      "NotIncreaseIfInherentLossLessThanAverageLoss:true/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_10p_loss_1 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_10p_loss_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Do not increase the bitrate because inherent loss is less than average loss
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+TEST_P(LossBasedBweV2Test,
+       SelectHighBandwidthCandidateIfLossRateIsLessThanThreshold) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|0.8,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "20,NotIncreaseIfInherentLossLessThanAverageLoss:false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_10p_loss_1 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_10p_loss_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Because LossThresholdOfHighBandwidthPreference is 20%, the average loss is
+  // 10%, bandwidth estimate should increase.
+  EXPECT_GT(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+TEST_P(LossBasedBweV2Test,
+       SelectLowBandwidthCandidateIfLossRateIsIsHigherThanThreshold) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|0.8,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_10p_loss_1 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_10p_loss_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Because LossThresholdOfHighBandwidthPreference is 5%, the average loss is
+  // 10%, bandwidth estimate should decrease.
+  EXPECT_LT(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+TEST_P(LossBasedBweV2Test, UseProbeResultWhenRecoveringFromLoss) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1000,"
+      "BwRampupUpperBoundFactor:2.0,ProbeIntegrationEnabled:true/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Network recovers after loss.
+  DataRate probe_estimate = DataRate::KilobitsPerSec(300);
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      probe_estimate, /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  LossBasedBweV2::Result result_after_recovery =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_EQ(result_after_recovery.bandwidth_estimate, probe_estimate);
+}
+
+// If BoundByUpperLinkCapacityWhenLossLimited is enabled, the estimate is
+// bounded by the upper link capacity when bandwidth is loss limited.
+TEST_P(LossBasedBweV2Test, BoundEstimateByUpperLinkCapacityWhenLossLimited) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1000,"
+      "BwRampupUpperBoundFactor:2.0,BoundByUpperLinkCapacityWhenLossLimited:"
+      "true/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Network recovers after loss.
+  DataRate upper_link_capacity = DataRate::KilobitsPerSec(10);
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt, upper_link_capacity);
+
+  LossBasedBweV2::Result result_after_recovery =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_EQ(result_after_recovery.bandwidth_estimate, upper_link_capacity);
+}
+
+// If BoundByUpperLinkCapacityWhenLossLimited is enabled, the estimate is not
+// bounded by the upper link capacity when bandwidth is not loss limited.
+TEST_P(LossBasedBweV2Test,
+       NotBoundEstimateByUpperLinkCapacityWhenNotLossLimited) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1000,"
+      "BwRampupUpperBoundFactor:2.0,BoundByUpperLinkCapacityWhenLossLimited:"
+      "true/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create a normal network without loss
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  DataRate upper_link_capacity = DataRate::KilobitsPerSec(10);
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt, upper_link_capacity);
+
+  LossBasedBweV2::Result loss_based_result =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_GT(loss_based_result.bandwidth_estimate, upper_link_capacity);
+}
+
+// If BoundByUpperLinkCapacityWhenLossLimited is disabled, the estimate is not
+// bounded by the upper link capacity.
+TEST_P(LossBasedBweV2Test, NotBoundEstimateByUpperLinkCapacity) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1000,"
+      "BwRampupUpperBoundFactor:2.0,BoundByUpperLinkCapacityWhenLossLimited:"
+      "false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Network recovers after loss.
+  DataRate upper_link_capacity = DataRate::KilobitsPerSec(10);
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt, upper_link_capacity);
+
+  LossBasedBweV2::Result result_after_recovery =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_GT(result_after_recovery.bandwidth_estimate, upper_link_capacity);
+}
+
+TEST_P(LossBasedBweV2Test,
+       StricterBoundUsingHighLossRateThresholdAt10pLossRate) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.0,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05,HighLossRateThreshold:0.09/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000000));
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_10p_loss_1 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_10p_loss_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // At 10% loss rate and high loss rate threshold to be 10%, cap the estimate
+  // to be 500 * 1000-0.1 = 400kbps.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(400));
+}
+
+TEST_P(LossBasedBweV2Test,
+       StricterBoundUsingHighLossRateThresholdAt50pLossRate) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.0,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05,HighLossRateThreshold:0.3/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000000));
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_50p_loss_1 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_50p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_50p_loss_2 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_50p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // At 50% loss rate and high loss rate threshold to be 30%, cap the estimate
+  // to be the min bitrate.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(10));
+}
+
+TEST_P(LossBasedBweV2Test,
+       StricterBoundUsingHighLossRateThresholdAt100pLossRate) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.0,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05,HighLossRateThreshold:0.3/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000000));
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_100p_loss_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_100p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_100p_loss_2 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_100p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // At 100% loss rate and high loss rate threshold to be 30%, cap the estimate
+  // to be the min bitrate.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(10));
+}
+
+TEST_P(LossBasedBweV2Test, EstimateRecoversAfterHighLoss) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.1|1.0|0.9,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05,HighLossRateThreshold:0.3/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000000));
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_100p_loss_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_100p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Make sure that the estimate is set to min bitrate because of 100% loss
+  // rate.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(10));
+
+  // Create some feedbacks with 0 loss rate to simulate network recovering.
+  std::vector<PacketResult> enough_feedback_0p_loss_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_0p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_0p_loss_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound * 2);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_0p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // The estimate increases as network recovers.
+  EXPECT_GT(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(10));
+}
+
+TEST_P(LossBasedBweV2Test, EstimateIsNotHigherThanMaxBitrate) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000));
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(1000));
+  std::vector<PacketResult> enough_feedback =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_LE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(1000));
+}
+
+INSTANTIATE_TEST_SUITE_P(LossBasedBweV2Tests,
+                         LossBasedBweV2Test,
+                         ::testing::Bool());
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
new file mode 100644
index 0000000000..a94f653157
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
@@ -0,0 +1,201 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace {
+// The minumum number of probes we need to receive feedback about in percent
+// in order to have a valid estimate.
+constexpr double kMinReceivedProbesRatio = .80;
+
+// The minumum number of bytes we need to receive feedback about in percent
+// in order to have a valid estimate.
+constexpr double kMinReceivedBytesRatio = .80;
+
+// The maximum |receive rate| / |send rate| ratio for a valid estimate.
+constexpr float kMaxValidRatio = 2.0f;
+
+// The minimum |receive rate| / |send rate| ratio assuming that the link is
+// not saturated, i.e. we assume that we will receive at least
+// kMinRatioForUnsaturatedLink * |send rate| if |send rate| is less than the
+// link capacity.
+constexpr float kMinRatioForUnsaturatedLink = 0.9f;
+
+// The target utilization of the link. If we know true link capacity
+// we'd like to send at 95% of that rate.
+constexpr float kTargetUtilizationFraction = 0.95f;
+
+// The maximum time period over which the cluster history is retained.
+// This is also the maximum time period beyond which a probing burst is not
+// expected to last.
+constexpr TimeDelta kMaxClusterHistory = TimeDelta::Seconds(1);
+
+// The maximum time interval between first and the last probe on a cluster
+// on the sender side as well as the receive side.
+constexpr TimeDelta kMaxProbeInterval = TimeDelta::Seconds(1);
+
+}  // namespace
+
+ProbeBitrateEstimator::ProbeBitrateEstimator(RtcEventLog* event_log)
+    : event_log_(event_log) {}
+
+ProbeBitrateEstimator::~ProbeBitrateEstimator() = default;
+
+absl::optional<DataRate> ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
+    const PacketResult& packet_feedback) {
+  int cluster_id = packet_feedback.sent_packet.pacing_info.probe_cluster_id;
+  RTC_DCHECK_NE(cluster_id, PacedPacketInfo::kNotAProbe);
+
+  EraseOldClusters(packet_feedback.receive_time);
+
+  AggregatedCluster* cluster = &clusters_[cluster_id];
+
+  if (packet_feedback.sent_packet.send_time < cluster->first_send) {
+    cluster->first_send = packet_feedback.sent_packet.send_time;
+  }
+  if (packet_feedback.sent_packet.send_time > cluster->last_send) {
+    cluster->last_send = packet_feedback.sent_packet.send_time;
+    cluster->size_last_send = packet_feedback.sent_packet.size;
+  }
+  if (packet_feedback.receive_time < cluster->first_receive) {
+    cluster->first_receive = packet_feedback.receive_time;
+    cluster->size_first_receive = packet_feedback.sent_packet.size;
+  }
+  if (packet_feedback.receive_time > cluster->last_receive) {
+    cluster->last_receive = packet_feedback.receive_time;
+  }
+  cluster->size_total += packet_feedback.sent_packet.size;
+  cluster->num_probes += 1;
+
+  RTC_DCHECK_GT(
+      packet_feedback.sent_packet.pacing_info.probe_cluster_min_probes, 0);
+  RTC_DCHECK_GT(packet_feedback.sent_packet.pacing_info.probe_cluster_min_bytes,
+                0);
+
+  int min_probes =
+      packet_feedback.sent_packet.pacing_info.probe_cluster_min_probes *
+      kMinReceivedProbesRatio;
+  DataSize min_size =
+      DataSize::Bytes(
+          packet_feedback.sent_packet.pacing_info.probe_cluster_min_bytes) *
+      kMinReceivedBytesRatio;
+  if (cluster->num_probes < min_probes || cluster->size_total < min_size)
+    return absl::nullopt;
+
+  TimeDelta send_interval = cluster->last_send - cluster->first_send;
+  TimeDelta receive_interval = cluster->last_receive - cluster->first_receive;
+
+  if (send_interval <= TimeDelta::Zero() || send_interval > kMaxProbeInterval ||
+      receive_interval <= TimeDelta::Zero() ||
+      receive_interval > kMaxProbeInterval) {
+    RTC_LOG(LS_INFO) << "Probing unsuccessful, invalid send/receive interval"
+                        " [cluster id: "
+                     << cluster_id
+                     << "] [send interval: " << ToString(send_interval)
+                     << "]"
+                        " [receive interval: "
+                     << ToString(receive_interval) << "]";
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
+          cluster_id, ProbeFailureReason::kInvalidSendReceiveInterval));
+    }
+    return absl::nullopt;
+  }
+  // Since the `send_interval` does not include the time it takes to actually
+  // send the last packet the size of the last sent packet should not be
+  // included when calculating the send bitrate.
+  RTC_DCHECK_GT(cluster->size_total, cluster->size_last_send);
+  DataSize send_size = cluster->size_total - cluster->size_last_send;
+  DataRate send_rate = send_size / send_interval;
+
+  // Since the `receive_interval` does not include the time it takes to
+  // actually receive the first packet the size of the first received packet
+  // should not be included when calculating the receive bitrate.
+  RTC_DCHECK_GT(cluster->size_total, cluster->size_first_receive);
+  DataSize receive_size = cluster->size_total - cluster->size_first_receive;
+  DataRate receive_rate = receive_size / receive_interval;
+
+  double ratio = receive_rate / send_rate;
+  if (ratio > kMaxValidRatio) {
+    RTC_LOG(LS_INFO) << "Probing unsuccessful, receive/send ratio too high"
+                        " [cluster id: "
+                     << cluster_id << "] [send: " << ToString(send_size)
+                     << " / " << ToString(send_interval) << " = "
+                     << ToString(send_rate)
+                     << "]"
+                        " [receive: "
+                     << ToString(receive_size) << " / "
+                     << ToString(receive_interval) << " = "
+                     << ToString(receive_rate)
+                     << " ]"
+                        " [ratio: "
+                     << ToString(receive_rate) << " / " << ToString(send_rate)
+                     << " = " << ratio << " > kMaxValidRatio ("
+                     << kMaxValidRatio << ")]";
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
+          cluster_id, ProbeFailureReason::kInvalidSendReceiveRatio));
+    }
+    return absl::nullopt;
+  }
+  RTC_LOG(LS_INFO) << "Probing successful"
+                      " [cluster id: "
+                   << cluster_id << "] [send: " << ToString(send_size) << " / "
+                   << ToString(send_interval) << " = " << ToString(send_rate)
+                   << " ]"
+                      " [receive: "
+                   << ToString(receive_size) << " / "
+                   << ToString(receive_interval) << " = "
+                   << ToString(receive_rate) << "]";
+
+  DataRate res = std::min(send_rate, receive_rate);
+  // If we're receiving at significantly lower bitrate than we were sending at,
+  // it suggests that we've found the true capacity of the link. In this case,
+  // set the target bitrate slightly lower to not immediately overuse.
+  if (receive_rate < kMinRatioForUnsaturatedLink * send_rate) {
+    RTC_DCHECK_GT(send_rate, receive_rate);
+    res = kTargetUtilizationFraction * receive_rate;
+  }
+  if (event_log_) {
+    event_log_->Log(
+        std::make_unique<RtcEventProbeResultSuccess>(cluster_id, res.bps()));
+  }
+  estimated_data_rate_ = res;
+  return estimated_data_rate_;
+}
+
+absl::optional<DataRate>
+ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrate() {
+  absl::optional<DataRate> estimated_data_rate = estimated_data_rate_;
+  estimated_data_rate_.reset();
+  return estimated_data_rate;
+}
+
+void ProbeBitrateEstimator::EraseOldClusters(Timestamp timestamp) {
+  for (auto it = clusters_.begin(); it != clusters_.end();) {
+    if (it->second.last_receive + kMaxClusterHistory < timestamp) {
+      it = clusters_.erase(it);
+    } else {
+      ++it;
+    }
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
new file mode 100644
index 0000000000..d5a523b7f3
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
+
+#include <limits>
+#include <map>
+
+#include "absl/types/optional.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+
+namespace webrtc {
+class RtcEventLog;
+
+class ProbeBitrateEstimator {
+ public:
+  explicit ProbeBitrateEstimator(RtcEventLog* event_log);
+  ~ProbeBitrateEstimator();
+
+  // Should be called for every probe packet we receive feedback about.
+  // Returns the estimated bitrate if the probe completes a valid cluster.
+  absl::optional<DataRate> HandleProbeAndEstimateBitrate(
+      const PacketResult& packet_feedback);
+
+  absl::optional<DataRate> FetchAndResetLastEstimatedBitrate();
+
+ private:
+  struct AggregatedCluster {
+    int num_probes = 0;
+    Timestamp first_send = Timestamp::PlusInfinity();
+    Timestamp last_send = Timestamp::MinusInfinity();
+    Timestamp first_receive = Timestamp::PlusInfinity();
+    Timestamp last_receive = Timestamp::MinusInfinity();
+    DataSize size_last_send = DataSize::Zero();
+    DataSize size_first_receive = DataSize::Zero();
+    DataSize size_total = DataSize::Zero();
+  };
+
+  // Erases old cluster data that was seen before `timestamp`.
+  void EraseOldClusters(Timestamp timestamp);
+
+  std::map<int, AggregatedCluster> clusters_;
+  RtcEventLog* const event_log_;
+  absl::optional<DataRate> estimated_data_rate_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
new file mode 100644
index 0000000000..6b4146d2bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
@@ -0,0 +1,228 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+
+#include <stddef.h>
+
+#include "api/transport/network_types.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kDefaultMinProbes = 5;
+constexpr int kDefaultMinBytes = 5000;
+constexpr float kTargetUtilizationFraction = 0.95f;
+}  // anonymous namespace
+
+class TestProbeBitrateEstimator : public ::testing::Test {
+ public:
+  TestProbeBitrateEstimator() : probe_bitrate_estimator_(nullptr) {}
+
+  // TODO(philipel): Use PacedPacketInfo when ProbeBitrateEstimator is rewritten
+  //                 to use that information.
+  void AddPacketFeedback(int probe_cluster_id,
+                         size_t size_bytes,
+                         int64_t send_time_ms,
+                         int64_t arrival_time_ms,
+                         int min_probes = kDefaultMinProbes,
+                         int min_bytes = kDefaultMinBytes) {
+    const Timestamp kReferenceTime = Timestamp::Seconds(1000);
+    PacketResult feedback;
+    feedback.sent_packet.send_time =
+        kReferenceTime + TimeDelta::Millis(send_time_ms);
+    feedback.sent_packet.size = DataSize::Bytes(size_bytes);
+    feedback.sent_packet.pacing_info =
+        PacedPacketInfo(probe_cluster_id, min_probes, min_bytes);
+    feedback.receive_time = kReferenceTime + TimeDelta::Millis(arrival_time_ms);
+    measured_data_rate_ =
+        probe_bitrate_estimator_.HandleProbeAndEstimateBitrate(feedback);
+  }
+
+ protected:
+  absl::optional<DataRate> measured_data_rate_;
+  ProbeBitrateEstimator probe_bitrate_estimator_;
+};
+
+TEST_F(TestProbeBitrateEstimator, OneCluster) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 30, 40);
+
+  EXPECT_NEAR(measured_data_rate_->bps(), 800000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, OneClusterTooFewProbes) {
+  AddPacketFeedback(0, 2000, 0, 10);
+  AddPacketFeedback(0, 2000, 10, 20);
+  AddPacketFeedback(0, 2000, 20, 30);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, OneClusterTooFewBytes) {
+  const int kMinBytes = 6000;
+  AddPacketFeedback(0, 800, 0, 10, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 800, 10, 20, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 800, 20, 30, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 800, 30, 40, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 800, 40, 50, kDefaultMinProbes, kMinBytes);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, SmallCluster) {
+  const int kMinBytes = 1000;
+  AddPacketFeedback(0, 150, 0, 10, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 10, 20, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 20, 30, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 30, 40, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 40, 50, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 50, 60, kDefaultMinProbes, kMinBytes);
+  EXPECT_NEAR(measured_data_rate_->bps(), 120000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, LargeCluster) {
+  const int kMinProbes = 30;
+  const int kMinBytes = 312500;
+
+  int64_t send_time = 0;
+  int64_t receive_time = 5;
+  for (int i = 0; i < 25; ++i) {
+    AddPacketFeedback(0, 12500, send_time, receive_time, kMinProbes, kMinBytes);
+    ++send_time;
+    ++receive_time;
+  }
+  EXPECT_NEAR(measured_data_rate_->bps(), 100000000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, FastReceive) {
+  AddPacketFeedback(0, 1000, 0, 15);
+  AddPacketFeedback(0, 1000, 10, 30);
+  AddPacketFeedback(0, 1000, 20, 35);
+  AddPacketFeedback(0, 1000, 30, 40);
+
+  EXPECT_NEAR(measured_data_rate_->bps(), 800000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, TooFastReceive) {
+  AddPacketFeedback(0, 1000, 0, 19);
+  AddPacketFeedback(0, 1000, 10, 22);
+  AddPacketFeedback(0, 1000, 20, 25);
+  AddPacketFeedback(0, 1000, 40, 27);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, SlowReceive) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 40);
+  AddPacketFeedback(0, 1000, 20, 70);
+  AddPacketFeedback(0, 1000, 30, 85);
+  // Expected send rate = 800 kbps, expected receive rate = 320 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 320000,
+              10);
+}
+
+TEST_F(TestProbeBitrateEstimator, BurstReceive) {
+  AddPacketFeedback(0, 1000, 0, 50);
+  AddPacketFeedback(0, 1000, 10, 50);
+  AddPacketFeedback(0, 1000, 20, 50);
+  AddPacketFeedback(0, 1000, 40, 50);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, MultipleClusters) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 40, 60);
+  // Expected send rate = 600 kbps, expected receive rate = 480 kbps.
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 480000,
+              10);
+
+  AddPacketFeedback(0, 1000, 50, 60);
+  // Expected send rate = 640 kbps, expected receive rate = 640 kbps.
+  EXPECT_NEAR(measured_data_rate_->bps(), 640000, 10);
+
+  AddPacketFeedback(1, 1000, 60, 70);
+  AddPacketFeedback(1, 1000, 65, 77);
+  AddPacketFeedback(1, 1000, 70, 84);
+  AddPacketFeedback(1, 1000, 75, 90);
+  // Expected send rate = 1600 kbps, expected receive rate = 1200 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 1200000,
+              10);
+}
+
+TEST_F(TestProbeBitrateEstimator, IgnoreOldClusters) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+
+  AddPacketFeedback(1, 1000, 60, 70);
+  AddPacketFeedback(1, 1000, 65, 77);
+  AddPacketFeedback(1, 1000, 70, 84);
+  AddPacketFeedback(1, 1000, 75, 90);
+  // Expected send rate = 1600 kbps, expected receive rate = 1200 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 1200000,
+              10);
+
+  // Coming in 6s later
+  AddPacketFeedback(0, 1000, 40 + 6000, 60 + 6000);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, IgnoreSizeLastSendPacket) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 30, 40);
+  AddPacketFeedback(0, 1500, 40, 50);
+  // Expected send rate = 800 kbps, expected receive rate = 900 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), 800000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, IgnoreSizeFirstReceivePacket) {
+  AddPacketFeedback(0, 1500, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 30, 40);
+  // Expected send rate = 933 kbps, expected receive rate = 800 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 800000,
+              10);
+}
+
+TEST_F(TestProbeBitrateEstimator, NoLastEstimatedBitrateBps) {
+  EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrate());
+}
+
+TEST_F(TestProbeBitrateEstimator, FetchLastEstimatedBitrateBps) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 30, 40);
+
+  auto estimated_bitrate =
+      probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrate();
+  EXPECT_TRUE(estimated_bitrate);
+  EXPECT_NEAR(estimated_bitrate->bps(), 800000, 10);
+  EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrate());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.cc
new file mode 100644
index 0000000000..1af943c4cb
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.cc
@@ -0,0 +1,558 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+
+#include <algorithm>
+#include <initializer_list>
+#include <memory>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+// Maximum waiting time from the time of initiating probing to getting
+// the measured results back.
+constexpr TimeDelta kMaxWaitingTimeForProbingResult = TimeDelta::Seconds(1);
+
+// Default probing bitrate limit. Applied only when the application didn't
+// specify max bitrate.
+constexpr DataRate kDefaultMaxProbingBitrate = DataRate::KilobitsPerSec(5000);
+
+// If the bitrate drops to a factor `kBitrateDropThreshold` or lower
+// and we recover within `kBitrateDropTimeoutMs`, then we'll send
+// a probe at a fraction `kProbeFractionAfterDrop` of the original bitrate.
+constexpr double kBitrateDropThreshold = 0.66;
+constexpr TimeDelta kBitrateDropTimeout = TimeDelta::Seconds(5);
+constexpr double kProbeFractionAfterDrop = 0.85;
+
+// Timeout for probing after leaving ALR. If the bitrate drops significantly,
+// (as determined by the delay based estimator) and we leave ALR, then we will
+// send a probe if we recover within `kLeftAlrTimeoutMs` ms.
+constexpr TimeDelta kAlrEndedTimeout = TimeDelta::Seconds(3);
+
+// The expected uncertainty of probe result (as a fraction of the target probe
+// This is a limit on how often probing can be done when there is a BW
+// drop detected in ALR.
+constexpr TimeDelta kMinTimeBetweenAlrProbes = TimeDelta::Seconds(5);
+
+// bitrate). Used to avoid probing if the probe bitrate is close to our current
+// estimate.
+constexpr double kProbeUncertainty = 0.05;
+
+// Use probing to recover faster after large bitrate estimate drops.
+constexpr char kBweRapidRecoveryExperiment[] =
+    "WebRTC-BweRapidRecoveryExperiment";
+
+void MaybeLogProbeClusterCreated(RtcEventLog* event_log,
+                                 const ProbeClusterConfig& probe) {
+  RTC_DCHECK(event_log);
+  if (!event_log) {
+    return;
+  }
+
+  DataSize min_data_size = probe.target_data_rate * probe.target_duration;
+  event_log->Log(std::make_unique<RtcEventProbeClusterCreated>(
+      probe.id, probe.target_data_rate.bps(), probe.target_probe_count,
+      min_data_size.bytes()));
+}
+
+}  // namespace
+
+ProbeControllerConfig::ProbeControllerConfig(
+    const FieldTrialsView* key_value_config)
+    : first_exponential_probe_scale("p1", 3.0),
+      second_exponential_probe_scale("p2", 6.0),
+      further_exponential_probe_scale("step_size", 2),
+      further_probe_threshold("further_probe_threshold", 0.7),
+      alr_probing_interval("alr_interval", TimeDelta::Seconds(5)),
+      alr_probe_scale("alr_scale", 2),
+      network_state_estimate_probing_interval("network_state_interval",
+                                              TimeDelta::PlusInfinity()),
+      probe_if_estimate_lower_than_network_state_estimate_ratio(
+          "est_lower_than_network_ratio",
+          0),
+      estimate_lower_than_network_state_estimate_probing_interval(
+          "est_lower_than_network_interval",
+          TimeDelta::Seconds(3)),
+      network_state_probe_scale("network_state_scale", 1.0),
+      network_state_probe_duration("network_state_probe_duration",
+                                   TimeDelta::Millis(15)),
+
+      probe_on_max_allocated_bitrate_change("probe_max_allocation", true),
+      first_allocation_probe_scale("alloc_p1", 1),
+      second_allocation_probe_scale("alloc_p2", 2),
+      allocation_allow_further_probing("alloc_probe_further", false),
+      allocation_probe_max("alloc_probe_max", DataRate::PlusInfinity()),
+      min_probe_packets_sent("min_probe_packets_sent", 5),
+      min_probe_duration("min_probe_duration", TimeDelta::Millis(15)),
+      limit_probe_target_rate_to_loss_bwe("limit_probe_target_rate_to_loss_bwe",
+                                          false),
+      loss_limited_probe_scale("loss_limited_scale", 1.5),
+      skip_if_estimate_larger_than_fraction_of_max(
+          "skip_if_est_larger_than_fraction_of_max",
+          0.0),
+      not_probe_if_delay_increased("not_probe_if_delay_increased", false) {
+  ParseFieldTrial({&first_exponential_probe_scale,
+                   &second_exponential_probe_scale,
+                   &further_exponential_probe_scale,
+                   &further_probe_threshold,
+                   &alr_probing_interval,
+                   &alr_probe_scale,
+                   &probe_on_max_allocated_bitrate_change,
+                   &first_allocation_probe_scale,
+                   &second_allocation_probe_scale,
+                   &allocation_allow_further_probing,
+                   &min_probe_duration,
+                   &network_state_estimate_probing_interval,
+                   &probe_if_estimate_lower_than_network_state_estimate_ratio,
+                   &estimate_lower_than_network_state_estimate_probing_interval,
+                   &network_state_probe_scale,
+                   &network_state_probe_duration,
+                   &min_probe_packets_sent,
+                   &limit_probe_target_rate_to_loss_bwe,
+                   &loss_limited_probe_scale,
+                   &skip_if_estimate_larger_than_fraction_of_max,
+                   &not_probe_if_delay_increased},
+                  key_value_config->Lookup("WebRTC-Bwe-ProbingConfiguration"));
+
+  // Specialized keys overriding subsets of WebRTC-Bwe-ProbingConfiguration
+  ParseFieldTrial(
+      {&first_exponential_probe_scale, &second_exponential_probe_scale},
+      key_value_config->Lookup("WebRTC-Bwe-InitialProbing"));
+  ParseFieldTrial({&further_exponential_probe_scale, &further_probe_threshold},
+                  key_value_config->Lookup("WebRTC-Bwe-ExponentialProbing"));
+  ParseFieldTrial(
+      {&alr_probing_interval, &alr_probe_scale, &loss_limited_probe_scale},
+      key_value_config->Lookup("WebRTC-Bwe-AlrProbing"));
+  ParseFieldTrial(
+      {&first_allocation_probe_scale, &second_allocation_probe_scale,
+       &allocation_allow_further_probing, &allocation_probe_max},
+      key_value_config->Lookup("WebRTC-Bwe-AllocationProbing"));
+  ParseFieldTrial({&min_probe_packets_sent, &min_probe_duration},
+                  key_value_config->Lookup("WebRTC-Bwe-ProbingBehavior"));
+}
+
+ProbeControllerConfig::ProbeControllerConfig(const ProbeControllerConfig&) =
+    default;
+ProbeControllerConfig::~ProbeControllerConfig() = default;
+
+ProbeController::ProbeController(const FieldTrialsView* key_value_config,
+                                 RtcEventLog* event_log)
+    : enable_periodic_alr_probing_(false),
+      in_rapid_recovery_experiment_(absl::StartsWith(
+          key_value_config->Lookup(kBweRapidRecoveryExperiment),
+          "Enabled")),
+      event_log_(event_log),
+      config_(ProbeControllerConfig(key_value_config)) {
+  Reset(Timestamp::Zero());
+}
+
+ProbeController::~ProbeController() {}
+
+std::vector<ProbeClusterConfig> ProbeController::SetBitrates(
+    DataRate min_bitrate,
+    DataRate start_bitrate,
+    DataRate max_bitrate,
+    Timestamp at_time) {
+  if (start_bitrate > DataRate::Zero()) {
+    start_bitrate_ = start_bitrate;
+    estimated_bitrate_ = start_bitrate;
+  } else if (start_bitrate_.IsZero()) {
+    start_bitrate_ = min_bitrate;
+  }
+
+  // The reason we use the variable `old_max_bitrate_pbs` is because we
+  // need to set `max_bitrate_` before we call InitiateProbing.
+  DataRate old_max_bitrate = max_bitrate_;
+  max_bitrate_ =
+      max_bitrate.IsFinite() ? max_bitrate : kDefaultMaxProbingBitrate;
+
+  switch (state_) {
+    case State::kInit:
+      if (network_available_)
+        return InitiateExponentialProbing(at_time);
+      break;
+
+    case State::kWaitingForProbingResult:
+      break;
+
+    case State::kProbingComplete:
+      // If the new max bitrate is higher than both the old max bitrate and the
+      // estimate then initiate probing.
+      if (!estimated_bitrate_.IsZero() && old_max_bitrate < max_bitrate_ &&
+          estimated_bitrate_ < max_bitrate_) {
+        return InitiateProbing(at_time, {max_bitrate_}, false);
+      }
+      break;
+  }
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::OnMaxTotalAllocatedBitrate(
+    DataRate max_total_allocated_bitrate,
+    Timestamp at_time) {
+  const bool in_alr = alr_start_time_.has_value();
+  const bool allow_allocation_probe = in_alr;
+
+  if (config_.probe_on_max_allocated_bitrate_change &&
+      state_ == State::kProbingComplete &&
+      max_total_allocated_bitrate != max_total_allocated_bitrate_ &&
+      estimated_bitrate_ < max_bitrate_ &&
+      estimated_bitrate_ < max_total_allocated_bitrate &&
+      allow_allocation_probe) {
+    max_total_allocated_bitrate_ = max_total_allocated_bitrate;
+
+    if (!config_.first_allocation_probe_scale)
+      return std::vector<ProbeClusterConfig>();
+
+    DataRate first_probe_rate = max_total_allocated_bitrate *
+                                config_.first_allocation_probe_scale.Value();
+    DataRate probe_cap = config_.allocation_probe_max.Get();
+    first_probe_rate = std::min(first_probe_rate, probe_cap);
+    std::vector<DataRate> probes = {first_probe_rate};
+    if (config_.second_allocation_probe_scale) {
+      DataRate second_probe_rate =
+          max_total_allocated_bitrate *
+          config_.second_allocation_probe_scale.Value();
+      second_probe_rate = std::min(second_probe_rate, probe_cap);
+      if (second_probe_rate > first_probe_rate)
+        probes.push_back(second_probe_rate);
+    }
+    return InitiateProbing(at_time, probes,
+                           config_.allocation_allow_further_probing.Get());
+  }
+  max_total_allocated_bitrate_ = max_total_allocated_bitrate;
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::OnNetworkAvailability(
+    NetworkAvailability msg) {
+  network_available_ = msg.network_available;
+
+  if (!network_available_ && state_ == State::kWaitingForProbingResult) {
+    state_ = State::kProbingComplete;
+    min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  }
+
+  if (network_available_ && state_ == State::kInit && !start_bitrate_.IsZero())
+    return InitiateExponentialProbing(msg.at_time);
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::InitiateExponentialProbing(
+    Timestamp at_time) {
+  RTC_DCHECK(network_available_);
+  RTC_DCHECK(state_ == State::kInit);
+  RTC_DCHECK_GT(start_bitrate_, DataRate::Zero());
+
+  // When probing at 1.8 Mbps ( 6x 300), this represents a threshold of
+  // 1.2 Mbps to continue probing.
+  std::vector<DataRate> probes = {config_.first_exponential_probe_scale *
+                                  start_bitrate_};
+  if (config_.second_exponential_probe_scale &&
+      config_.second_exponential_probe_scale.GetOptional().value() > 0) {
+    probes.push_back(config_.second_exponential_probe_scale.Value() *
+                     start_bitrate_);
+  }
+  return InitiateProbing(at_time, probes, true);
+}
+
+std::vector<ProbeClusterConfig> ProbeController::SetEstimatedBitrate(
+    DataRate bitrate,
+    BandwidthLimitedCause bandwidth_limited_cause,
+    Timestamp at_time) {
+  bandwidth_limited_cause_ = bandwidth_limited_cause;
+  if (bitrate < kBitrateDropThreshold * estimated_bitrate_) {
+    time_of_last_large_drop_ = at_time;
+    bitrate_before_last_large_drop_ = estimated_bitrate_;
+  }
+  estimated_bitrate_ = bitrate;
+
+  if (state_ == State::kWaitingForProbingResult) {
+    // Continue probing if probing results indicate channel has greater
+    // capacity.
+    DataRate network_state_estimate_probe_further_limit =
+        config_.network_state_estimate_probing_interval->IsFinite() &&
+                network_estimate_
+            ? network_estimate_->link_capacity_upper *
+                  config_.further_probe_threshold
+            : DataRate::PlusInfinity();
+    RTC_LOG(LS_INFO) << "Measured bitrate: " << bitrate
+                     << " Minimum to probe further: "
+                     << min_bitrate_to_probe_further_ << " upper limit: "
+                     << network_state_estimate_probe_further_limit;
+
+    if (bitrate > min_bitrate_to_probe_further_ &&
+        bitrate <= network_state_estimate_probe_further_limit) {
+      return InitiateProbing(
+          at_time, {config_.further_exponential_probe_scale * bitrate}, true);
+    }
+  }
+  return {};
+}
+
+void ProbeController::EnablePeriodicAlrProbing(bool enable) {
+  enable_periodic_alr_probing_ = enable;
+}
+
+void ProbeController::SetAlrStartTimeMs(
+    absl::optional<int64_t> alr_start_time_ms) {
+  if (alr_start_time_ms) {
+    alr_start_time_ = Timestamp::Millis(*alr_start_time_ms);
+  } else {
+    alr_start_time_ = absl::nullopt;
+  }
+}
+void ProbeController::SetAlrEndedTimeMs(int64_t alr_end_time_ms) {
+  alr_end_time_.emplace(Timestamp::Millis(alr_end_time_ms));
+}
+
+std::vector<ProbeClusterConfig> ProbeController::RequestProbe(
+    Timestamp at_time) {
+  // Called once we have returned to normal state after a large drop in
+  // estimated bandwidth. The current response is to initiate a single probe
+  // session (if not already probing) at the previous bitrate.
+  //
+  // If the probe session fails, the assumption is that this drop was a
+  // real one from a competing flow or a network change.
+  bool in_alr = alr_start_time_.has_value();
+  bool alr_ended_recently =
+      (alr_end_time_.has_value() &&
+       at_time - alr_end_time_.value() < kAlrEndedTimeout);
+  if (in_alr || alr_ended_recently || in_rapid_recovery_experiment_) {
+    if (state_ == State::kProbingComplete) {
+      DataRate suggested_probe =
+          kProbeFractionAfterDrop * bitrate_before_last_large_drop_;
+      DataRate min_expected_probe_result =
+          (1 - kProbeUncertainty) * suggested_probe;
+      TimeDelta time_since_drop = at_time - time_of_last_large_drop_;
+      TimeDelta time_since_probe = at_time - last_bwe_drop_probing_time_;
+      if (min_expected_probe_result > estimated_bitrate_ &&
+          time_since_drop < kBitrateDropTimeout &&
+          time_since_probe > kMinTimeBetweenAlrProbes) {
+        RTC_LOG(LS_INFO) << "Detected big bandwidth drop, start probing.";
+        // Track how often we probe in response to bandwidth drop in ALR.
+        RTC_HISTOGRAM_COUNTS_10000(
+            "WebRTC.BWE.BweDropProbingIntervalInS",
+            (at_time - last_bwe_drop_probing_time_).seconds());
+        last_bwe_drop_probing_time_ = at_time;
+        return InitiateProbing(at_time, {suggested_probe}, false);
+      }
+    }
+  }
+  return std::vector<ProbeClusterConfig>();
+}
+
+void ProbeController::SetNetworkStateEstimate(
+    webrtc::NetworkStateEstimate estimate) {
+  network_estimate_ = estimate;
+}
+
+void ProbeController::Reset(Timestamp at_time) {
+  network_available_ = true;
+  bandwidth_limited_cause_ = BandwidthLimitedCause::kDelayBasedLimited;
+  state_ = State::kInit;
+  min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  time_last_probing_initiated_ = Timestamp::Zero();
+  estimated_bitrate_ = DataRate::Zero();
+  network_estimate_ = absl::nullopt;
+  start_bitrate_ = DataRate::Zero();
+  max_bitrate_ = kDefaultMaxProbingBitrate;
+  Timestamp now = at_time;
+  last_bwe_drop_probing_time_ = now;
+  alr_end_time_.reset();
+  time_of_last_large_drop_ = now;
+  bitrate_before_last_large_drop_ = DataRate::Zero();
+  max_total_allocated_bitrate_ = DataRate::Zero();
+}
+
+bool ProbeController::TimeForAlrProbe(Timestamp at_time) const {
+  if (enable_periodic_alr_probing_ && alr_start_time_) {
+    Timestamp next_probe_time =
+        std::max(*alr_start_time_, time_last_probing_initiated_) +
+        config_.alr_probing_interval;
+    return at_time >= next_probe_time;
+  }
+  return false;
+}
+
+bool ProbeController::TimeForNetworkStateProbe(Timestamp at_time) const {
+  if (!network_estimate_ ||
+      network_estimate_->link_capacity_upper.IsInfinite()) {
+    return false;
+  }
+
+  bool probe_due_to_low_estimate =
+      bandwidth_limited_cause_ == BandwidthLimitedCause::kDelayBasedLimited &&
+      estimated_bitrate_ <
+          config_.probe_if_estimate_lower_than_network_state_estimate_ratio *
+              network_estimate_->link_capacity_upper;
+  if (probe_due_to_low_estimate &&
+      config_.estimate_lower_than_network_state_estimate_probing_interval
+          ->IsFinite()) {
+    Timestamp next_probe_time =
+        time_last_probing_initiated_ +
+        config_.estimate_lower_than_network_state_estimate_probing_interval;
+    return at_time >= next_probe_time;
+  }
+
+  bool periodic_probe =
+      estimated_bitrate_ < network_estimate_->link_capacity_upper;
+  if (periodic_probe &&
+      config_.network_state_estimate_probing_interval->IsFinite()) {
+    Timestamp next_probe_time = time_last_probing_initiated_ +
+                                config_.network_state_estimate_probing_interval;
+    return at_time >= next_probe_time;
+  }
+
+  return false;
+}
+
+std::vector<ProbeClusterConfig> ProbeController::Process(Timestamp at_time) {
+  if (at_time - time_last_probing_initiated_ >
+      kMaxWaitingTimeForProbingResult) {
+    if (state_ == State::kWaitingForProbingResult) {
+      RTC_LOG(LS_INFO) << "kWaitingForProbingResult: timeout";
+      state_ = State::kProbingComplete;
+      min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+    }
+  }
+  if (estimated_bitrate_.IsZero() || state_ != State::kProbingComplete) {
+    return {};
+  }
+  if (TimeForAlrProbe(at_time) || TimeForNetworkStateProbe(at_time)) {
+    return InitiateProbing(
+        at_time, {estimated_bitrate_ * config_.alr_probe_scale}, true);
+  }
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::InitiateProbing(
+    Timestamp now,
+    std::vector<DataRate> bitrates_to_probe,
+    bool probe_further) {
+  if (config_.skip_if_estimate_larger_than_fraction_of_max > 0) {
+    DataRate network_estimate = network_estimate_
+                                    ? network_estimate_->link_capacity_upper
+                                    : DataRate::PlusInfinity();
+    DataRate max_probe_rate =
+        max_total_allocated_bitrate_.IsZero()
+            ? max_bitrate_
+            : std::min(max_total_allocated_bitrate_, max_bitrate_);
+    if (std::min(network_estimate, estimated_bitrate_) >
+        config_.skip_if_estimate_larger_than_fraction_of_max * max_probe_rate) {
+      state_ = State::kProbingComplete;
+      min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+      return {};
+    }
+  }
+
+  DataRate max_probe_bitrate = max_bitrate_;
+  if (max_total_allocated_bitrate_ > DataRate::Zero()) {
+    // If a max allocated bitrate has been configured, allow probing up to 2x
+    // that rate. This allows some overhead to account for bursty streams,
+    // which otherwise would have to ramp up when the overshoot is already in
+    // progress.
+    // It also avoids minor quality reduction caused by probes often being
+    // received at slightly less than the target probe bitrate.
+    max_probe_bitrate =
+        std::min(max_probe_bitrate, max_total_allocated_bitrate_ * 2);
+  }
+
+  DataRate estimate_capped_bitrate = DataRate::PlusInfinity();
+  if (config_.limit_probe_target_rate_to_loss_bwe) {
+    switch (bandwidth_limited_cause_) {
+      case BandwidthLimitedCause::kLossLimitedBweDecreasing:
+        // If bandwidth estimate is decreasing because of packet loss, do not
+        // send probes.
+        return {};
+      case BandwidthLimitedCause::kLossLimitedBweIncreasing:
+        estimate_capped_bitrate =
+            std::min(max_probe_bitrate,
+                     estimated_bitrate_ * config_.loss_limited_probe_scale);
+        break;
+      case BandwidthLimitedCause::kDelayBasedLimited:
+        break;
+      default:
+        break;
+    }
+  }
+  if (config_.not_probe_if_delay_increased &&
+      bandwidth_limited_cause_ ==
+          BandwidthLimitedCause::kDelayBasedLimitedDelayIncreased) {
+    return {};
+  }
+
+  if (config_.network_state_estimate_probing_interval->IsFinite() &&
+      network_estimate_ && network_estimate_->link_capacity_upper.IsFinite()) {
+    if (network_estimate_->link_capacity_upper.IsZero()) {
+      RTC_LOG(LS_INFO) << "Not sending probe, Network state estimate is zero";
+      return {};
+    }
+    estimate_capped_bitrate =
+        std::min({estimate_capped_bitrate, max_probe_bitrate,
+                  network_estimate_->link_capacity_upper *
+                      config_.network_state_probe_scale});
+  }
+
+  std::vector<ProbeClusterConfig> pending_probes;
+  for (DataRate bitrate : bitrates_to_probe) {
+    RTC_DCHECK(!bitrate.IsZero());
+
+    bitrate = std::min(bitrate, estimate_capped_bitrate);
+    if (bitrate > max_probe_bitrate) {
+      bitrate = max_probe_bitrate;
+      probe_further = false;
+    }
+
+    ProbeClusterConfig config;
+    config.at_time = now;
+    config.target_data_rate = bitrate;
+    if (network_estimate_ &&
+        config_.network_state_estimate_probing_interval->IsFinite()) {
+      config.target_duration = config_.network_state_probe_duration;
+    } else {
+      config.target_duration = config_.min_probe_duration;
+    }
+
+    config.target_probe_count = config_.min_probe_packets_sent;
+    config.id = next_probe_cluster_id_;
+    next_probe_cluster_id_++;
+    MaybeLogProbeClusterCreated(event_log_, config);
+    pending_probes.push_back(config);
+  }
+  time_last_probing_initiated_ = now;
+  if (probe_further) {
+    state_ = State::kWaitingForProbingResult;
+    // Dont expect probe results to be larger than a fraction of the actual
+    // probe rate.
+    min_bitrate_to_probe_further_ =
+        std::min(estimate_capped_bitrate, (*(bitrates_to_probe.end() - 1))) *
+        config_.further_probe_threshold;
+  } else {
+    state_ = State::kProbingComplete;
+    min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  }
+  return pending_probes;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.h
new file mode 100644
index 0000000000..aa8b526ab0
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.h
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
+
+#include <stdint.h>
+
+#include <initializer_list>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+struct ProbeControllerConfig {
+  explicit ProbeControllerConfig(const FieldTrialsView* key_value_config);
+  ProbeControllerConfig(const ProbeControllerConfig&);
+  ProbeControllerConfig& operator=(const ProbeControllerConfig&) = default;
+  ~ProbeControllerConfig();
+
+  // These parameters configure the initial probes. First we send one or two
+  // probes of sizes p1 * start_bitrate_ and p2 * start_bitrate_.
+  // Then whenever we get a bitrate estimate of at least further_probe_threshold
+  // times the size of the last sent probe we'll send another one of size
+  // step_size times the new estimate.
+  FieldTrialParameter<double> first_exponential_probe_scale;
+  FieldTrialOptional<double> second_exponential_probe_scale;
+  FieldTrialParameter<double> further_exponential_probe_scale;
+  FieldTrialParameter<double> further_probe_threshold;
+
+  // Configures how often we send ALR probes and how big they are.
+  FieldTrialParameter<TimeDelta> alr_probing_interval;
+  FieldTrialParameter<double> alr_probe_scale;
+
+  // Configures how often we send probes if NetworkStateEstimate is available.
+  FieldTrialParameter<TimeDelta> network_state_estimate_probing_interval;
+  // Periodically probe as long as the the ratio beteeen current estimate and
+  // NetworkStateEstimate is lower then this.
+  FieldTrialParameter<double>
+      probe_if_estimate_lower_than_network_state_estimate_ratio;
+  FieldTrialParameter<TimeDelta>
+      estimate_lower_than_network_state_estimate_probing_interval;
+  FieldTrialParameter<double> network_state_probe_scale;
+  // Overrides min_probe_duration if network_state_estimate_probing_interval
+  // is set and a network state estimate is known.
+  FieldTrialParameter<TimeDelta> network_state_probe_duration;
+
+  // Configures the probes emitted by changed to the allocated bitrate.
+  FieldTrialParameter<bool> probe_on_max_allocated_bitrate_change;
+  FieldTrialOptional<double> first_allocation_probe_scale;
+  FieldTrialOptional<double> second_allocation_probe_scale;
+  FieldTrialFlag allocation_allow_further_probing;
+  FieldTrialParameter<DataRate> allocation_probe_max;
+
+  // The minimum number probing packets used.
+  FieldTrialParameter<int> min_probe_packets_sent;
+  // The minimum probing duration.
+  FieldTrialParameter<TimeDelta> min_probe_duration;
+  // Periodically probe when bandwidth estimate is loss limited.
+  FieldTrialParameter<bool> limit_probe_target_rate_to_loss_bwe;
+  FieldTrialParameter<double> loss_limited_probe_scale;
+  // Dont send a probe if min(estimate, network state estimate) is larger than
+  // this fraction of the set max bitrate.
+  FieldTrialParameter<double> skip_if_estimate_larger_than_fraction_of_max;
+  // Do not send probes if network is either overusing or underusing.
+  FieldTrialParameter<bool> not_probe_if_delay_increased;
+};
+
+// Reason that bandwidth estimate is limited. Bandwidth estimate can be limited
+// by either delay based bwe, or loss based bwe when it increases/decreases the
+// estimate.
+enum class BandwidthLimitedCause {
+  kLossLimitedBweIncreasing = 0,
+  kLossLimitedBweDecreasing = 1,
+  kDelayBasedLimited = 2,
+  kDelayBasedLimitedDelayIncreased = 3,
+};
+
+// This class controls initiation of probing to estimate initial channel
+// capacity. There is also support for probing during a session when max
+// bitrate is adjusted by an application.
+class ProbeController {
+ public:
+  explicit ProbeController(const FieldTrialsView* key_value_config,
+                           RtcEventLog* event_log);
+  ~ProbeController();
+
+  ProbeController(const ProbeController&) = delete;
+  ProbeController& operator=(const ProbeController&) = delete;
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> SetBitrates(
+      DataRate min_bitrate,
+      DataRate start_bitrate,
+      DataRate max_bitrate,
+      Timestamp at_time);
+
+  // The total bitrate, as opposed to the max bitrate, is the sum of the
+  // configured bitrates for all active streams.
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig>
+  OnMaxTotalAllocatedBitrate(DataRate max_total_allocated_bitrate,
+                             Timestamp at_time);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> OnNetworkAvailability(
+      NetworkAvailability msg);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> SetEstimatedBitrate(
+      DataRate bitrate,
+      BandwidthLimitedCause bandwidth_limited_cause,
+      Timestamp at_time);
+
+  void EnablePeriodicAlrProbing(bool enable);
+
+  void SetAlrStartTimeMs(absl::optional<int64_t> alr_start_time);
+  void SetAlrEndedTimeMs(int64_t alr_end_time);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> RequestProbe(
+      Timestamp at_time);
+
+  void SetNetworkStateEstimate(webrtc::NetworkStateEstimate estimate);
+
+  // Resets the ProbeController to a state equivalent to as if it was just
+  // created EXCEPT for `enable_periodic_alr_probing_`.
+  void Reset(Timestamp at_time);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> Process(
+      Timestamp at_time);
+
+  // Gets the value of field trial not_probe_if_delay_increased.
+  bool DontProbeIfDelayIncreased() {
+    return config_.not_probe_if_delay_increased;
+  }
+
+ private:
+  enum class State {
+    // Initial state where no probing has been triggered yet.
+    kInit,
+    // Waiting for probing results to continue further probing.
+    kWaitingForProbingResult,
+    // Probing is complete.
+    kProbingComplete,
+  };
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig>
+  InitiateExponentialProbing(Timestamp at_time);
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> InitiateProbing(
+      Timestamp now,
+      std::vector<DataRate> bitrates_to_probe,
+      bool probe_further);
+  bool TimeForAlrProbe(Timestamp at_time) const;
+  bool TimeForNetworkStateProbe(Timestamp at_time) const;
+
+  bool network_available_;
+  BandwidthLimitedCause bandwidth_limited_cause_ =
+      BandwidthLimitedCause::kDelayBasedLimited;
+  State state_;
+  DataRate min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  Timestamp time_last_probing_initiated_ = Timestamp::MinusInfinity();
+  DataRate estimated_bitrate_ = DataRate::Zero();
+  absl::optional<webrtc::NetworkStateEstimate> network_estimate_;
+  DataRate start_bitrate_ = DataRate::Zero();
+  DataRate max_bitrate_ = DataRate::PlusInfinity();
+  Timestamp last_bwe_drop_probing_time_ = Timestamp::Zero();
+  absl::optional<Timestamp> alr_start_time_;
+  absl::optional<Timestamp> alr_end_time_;
+  bool enable_periodic_alr_probing_;
+  Timestamp time_of_last_large_drop_ = Timestamp::MinusInfinity();
+  DataRate bitrate_before_last_large_drop_ = DataRate::Zero();
+  DataRate max_total_allocated_bitrate_ = DataRate::Zero();
+
+  const bool in_rapid_recovery_experiment_;
+  RtcEventLog* event_log_;
+
+  int32_t next_probe_cluster_id_ = 1;
+
+  ProbeControllerConfig config_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build
new file mode 100644
index 0000000000..4f4f573cd9
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("probe_controller_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
new file mode 100644
index 0000000000..e6a5c8ceef
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
@@ -0,0 +1,1131 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+
+#include <memory>
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::NiceMock;
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+constexpr DataRate kMinBitrate = DataRate::BitsPerSec(100);
+constexpr DataRate kStartBitrate = DataRate::BitsPerSec(300);
+constexpr DataRate kMaxBitrate = DataRate::BitsPerSec(10000);
+
+constexpr TimeDelta kExponentialProbingTimeout = TimeDelta::Seconds(5);
+
+constexpr TimeDelta kAlrProbeInterval = TimeDelta::Seconds(5);
+constexpr TimeDelta kAlrEndedTimeout = TimeDelta::Seconds(3);
+constexpr TimeDelta kBitrateDropTimeout = TimeDelta::Seconds(5);
+}  // namespace
+
+class ProbeControllerFixture {
+ public:
+  explicit ProbeControllerFixture(absl::string_view field_trials = "")
+      : field_trial_config_(field_trials), clock_(100000000L) {}
+
+  std::unique_ptr<ProbeController> CreateController() {
+    return std::make_unique<ProbeController>(&field_trial_config_,
+                                             &mock_rtc_event_log);
+  }
+
+  Timestamp CurrentTime() { return clock_.CurrentTime(); }
+  void AdvanceTime(TimeDelta delta) { clock_.AdvanceTime(delta); }
+
+  ExplicitKeyValueConfig field_trial_config_;
+  SimulatedClock clock_;
+  NiceMock<MockRtcEventLog> mock_rtc_event_log;
+};
+
+TEST(ProbeControllerTest, InitiatesProbingAtStart) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_GE(probes.size(), 2u);
+}
+
+TEST(ProbeControllerTest, SetsDefaultTargetDurationAndTargetProbeCount) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  std::vector<ProbeClusterConfig> probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_GE(probes.size(), 2u);
+
+  EXPECT_EQ(probes[0].target_duration, TimeDelta::Millis(15));
+  EXPECT_EQ(probes[0].target_probe_count, 5);
+}
+
+TEST(ProbeControllerTest,
+     FieldTrialsOverrideDefaultTargetDurationAndTargetProbeCount) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingBehavior/"
+      "min_probe_packets_sent:2,min_probe_duration:123ms/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  std::vector<ProbeClusterConfig> probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_GE(probes.size(), 2u);
+
+  EXPECT_EQ(probes[0].target_duration, TimeDelta::Millis(123));
+  EXPECT_EQ(probes[0].target_probe_count, 2);
+}
+
+TEST(ProbeControllerTest, ProbeOnlyWhenNetworkIsUp) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->OnNetworkAvailability(
+      {.at_time = fixture.CurrentTime(), .network_available = false});
+  probes = probe_controller->SetBitrates(kMinBitrate, kStartBitrate,
+                                         kMaxBitrate, fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+  probes = probe_controller->OnNetworkAvailability(
+      {.at_time = fixture.CurrentTime(), .network_available = true});
+  EXPECT_GE(probes.size(), 2u);
+}
+
+TEST(ProbeControllerTest, CanConfigureInitialProbeRateFactor) {
+  ProbeControllerFixture fixture("WebRTC-Bwe-ProbingConfiguration/p1:2,p2:3/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  EXPECT_EQ(probes[0].target_data_rate, kStartBitrate * 2);
+  EXPECT_EQ(probes[1].target_data_rate, kStartBitrate * 3);
+}
+
+TEST(ProbeControllerTest, DisableSecondInitialProbeIfRateFactorZero) {
+  ProbeControllerFixture fixture("WebRTC-Bwe-ProbingConfiguration/p1:2,p2:0/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, kStartBitrate * 2);
+}
+
+TEST(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncrease) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  // Long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate + DataRate::BitsPerSec(100),
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), kMaxBitrate.bps() + 100);
+}
+
+TEST(ProbeControllerTest, ProbesOnMaxAllocatedBitrateIncreaseOnlyWhenInAlr) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate - DataRate::BitsPerSec(1),
+      BandwidthLimitedCause::kDelayBasedLimited, fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  // Probe when in alr.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      kMaxBitrate + DataRate::BitsPerSec(1), fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  EXPECT_EQ(probes.at(0).target_data_rate, kMaxBitrate);
+
+  // Do not probe when not in alr.
+  probe_controller->SetAlrStartTimeMs(absl::nullopt);
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      kMaxBitrate + DataRate::BitsPerSec(2), fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, CanDisableProbingOnMaxTotalAllocatedBitrateIncrease) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "probe_max_allocation:false/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate - DataRate::BitsPerSec(1),
+      BandwidthLimitedCause::kDelayBasedLimited, fixture.CurrentTime());
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+
+  // Do no probe, since probe_max_allocation:false.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      kMaxBitrate + DataRate::BitsPerSec(1), fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncreaseAtMaxBitrate) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  // Long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate + DataRate::BitsPerSec(100),
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate,
+            kMaxBitrate + DataRate::BitsPerSec(100));
+}
+
+TEST(ProbeControllerTest, TestExponentialProbing) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+
+  // Repeated probe should only be sent when estimated bitrate climbs above
+  // 0.7 * 6 * kStartBitrate = 1260.
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1000), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1800), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 2 * 1800);
+}
+
+TEST(ProbeControllerTest, TestExponentialProbingTimeout) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  // Advance far enough to cause a time out in waiting for probing result.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1800), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, RequestProbeInAlr) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_GE(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probes = probe_controller->RequestProbe(fixture.CurrentTime());
+
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 0.85 * 500);
+}
+
+TEST(ProbeControllerTest, RequestProbeWhenAlrEndedRecently) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(absl::nullopt);
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probe_controller->SetAlrEndedTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrEndedTimeout - TimeDelta::Millis(1));
+  probes = probe_controller->RequestProbe(fixture.CurrentTime());
+
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 0.85 * 500);
+}
+
+TEST(ProbeControllerTest, RequestProbeWhenAlrNotEndedRecently) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(absl::nullopt);
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probe_controller->SetAlrEndedTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrEndedTimeout + TimeDelta::Millis(1));
+  probes = probe_controller->RequestProbe(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, RequestProbeWhenBweDropNotRecent) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  fixture.AdvanceTime(kBitrateDropTimeout + TimeDelta::Millis(1));
+  probes = probe_controller->RequestProbe(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, PeriodicProbing) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  Timestamp start_time = fixture.CurrentTime();
+
+  // Expect the controller to send a new probe after 5s has passed.
+  probe_controller->SetAlrStartTimeMs(start_time.ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 1000);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  // The following probe should be sent at 10s into ALR.
+  probe_controller->SetAlrStartTimeMs(start_time.ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(4));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  probe_controller->SetAlrStartTimeMs(start_time.ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, PeriodicProbingAfterReset) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  Timestamp alr_start_time = fixture.CurrentTime();
+
+  probe_controller->SetAlrStartTimeMs(alr_start_time.ms());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->Reset(fixture.CurrentTime());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  // Since bitrates are not yet set, no probe is sent event though we are in ALR
+  // mode.
+  EXPECT_TRUE(probes.empty());
+
+  probes = probe_controller->SetBitrates(kMinBitrate, kStartBitrate,
+                                         kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+
+  // Make sure we use `kStartBitrateBps` as the estimated bitrate
+  // until SetEstimatedBitrate is called with an updated estimate.
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, kStartBitrate * 2);
+}
+
+TEST(ProbeControllerTest, TestExponentialProbingOverflow) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  const DataRate kMbpsMultiplier = DataRate::KilobitsPerSec(1000);
+  auto probes = probe_controller->SetBitrates(kMinBitrate, 10 * kMbpsMultiplier,
+                                              100 * kMbpsMultiplier,
+                                              fixture.CurrentTime());
+  // Verify that probe bitrate is capped at the specified max bitrate.
+  probes = probe_controller->SetEstimatedBitrate(
+      60 * kMbpsMultiplier, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, 100 * kMbpsMultiplier);
+  // Verify that repeated probes aren't sent.
+  probes = probe_controller->SetEstimatedBitrate(
+      100 * kMbpsMultiplier, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, TestAllocatedBitrateCap) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  const DataRate kMbpsMultiplier = DataRate::KilobitsPerSec(1000);
+  const DataRate kMaxBitrate = 100 * kMbpsMultiplier;
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, 10 * kMbpsMultiplier, kMaxBitrate, fixture.CurrentTime());
+
+  // Configure ALR for periodic probing.
+  probe_controller->EnablePeriodicAlrProbing(true);
+  Timestamp alr_start_time = fixture.CurrentTime();
+  probe_controller->SetAlrStartTimeMs(alr_start_time.ms());
+
+  DataRate estimated_bitrate = kMaxBitrate / 10;
+  probes = probe_controller->SetEstimatedBitrate(
+      estimated_bitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  // Set a max allocated bitrate below the current estimate.
+  DataRate max_allocated = estimated_bitrate - 1 * kMbpsMultiplier;
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(max_allocated,
+                                                        fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());  // No probe since lower than current max.
+
+  // Probes such as ALR capped at 2x the max allocation limit.
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, 2 * max_allocated);
+
+  // Remove allocation limit.
+  EXPECT_TRUE(
+      probe_controller
+          ->OnMaxTotalAllocatedBitrate(DataRate::Zero(), fixture.CurrentTime())
+          .empty());
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, estimated_bitrate * 2);
+}
+
+TEST(ProbeControllerTest, ConfigurableProbingFieldTrial) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "p1:2,p2:5,step_size:3,further_probe_threshold:0.8,"
+      "alloc_p1:2,alloc_p2,min_probe_packets_sent:2/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(kMinBitrate, kStartBitrate,
+                                              DataRate::KilobitsPerSec(5000),
+                                              fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 600);
+  EXPECT_EQ(probes[0].target_probe_count, 2);
+  EXPECT_EQ(probes[1].target_data_rate.bps(), 1500);
+  EXPECT_EQ(probes[1].target_probe_count, 2);
+
+  // Repeated probe should only be sent when estimated bitrate climbs above
+  // 0.8 * 5 * kStartBitrateBps = 1200.
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1100), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 0u);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 3 * 1250);
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      DataRate::KilobitsPerSec(200), fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 400'000);
+}
+
+TEST(ProbeControllerTest, LimitAlrProbeWhenLossBasedBweLimited) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // Expect the controller to send a new probe after 5s has passed.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500),
+      BandwidthLimitedCause::kLossLimitedBweIncreasing, fixture.CurrentTime());
+  fixture.AdvanceTime(TimeDelta::Seconds(6));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, 1.5 * DataRate::BitsPerSec(500));
+
+  probes = probe_controller->SetEstimatedBitrate(
+      1.5 * DataRate::BitsPerSec(500),
+      BandwidthLimitedCause::kDelayBasedLimited, fixture.CurrentTime());
+  fixture.AdvanceTime(TimeDelta::Seconds(6));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_GT(probes[0].target_data_rate, 1.5 * 1.5 * DataRate::BitsPerSec(500));
+}
+
+TEST(ProbeControllerTest, PeriodicProbeAtUpperNetworkStateEstimate) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/network_state_interval:5s/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(5000), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // Expect the controller to send a new probe after 5s has passed.
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = DataRate::KilobitsPerSec(6);
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+}
+
+TEST(ProbeControllerTest,
+     LimitProbeAtUpperNetworkStateEstimateIfLossBasedLimited) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // Expect the controller to send a new probe after 5s has passed.
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = DataRate::BitsPerSec(700);
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500),
+      BandwidthLimitedCause::kLossLimitedBweIncreasing, fixture.CurrentTime());
+  // Expect the controller to send a new probe after 5s has passed.
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_EQ(probes[0].target_data_rate, DataRate::BitsPerSec(700));
+}
+
+TEST(ProbeControllerTest, AlrProbesLimitedByNetworkStateEstimate) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/network_state_interval:5s/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::KilobitsPerSec(6), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, kMaxBitrate);
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = DataRate::BitsPerSec(8000);
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+}
+
+TEST(ProbeControllerTest, CanSetLongerProbeDurationAfterNetworkStateEstimate) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,network_state_probe_duration:100ms/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::KilobitsPerSec(5), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_LT(probes[0].target_duration, TimeDelta::Millis(100));
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = DataRate::KilobitsPerSec(6);
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_duration, TimeDelta::Millis(100));
+}
+
+TEST(ProbeControllerTest, ProbeInAlrIfLossBasedIncreasing) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kLossLimitedBweIncreasing,
+      fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  // Probe when in alr.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes.at(0).target_data_rate, 1.5 * kStartBitrate);
+}
+
+TEST(ProbeControllerTest, ProbeFurtherInAlrIfLossBasedIncreasing) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kLossLimitedBweIncreasing,
+      fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  // Probe when in alr.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  ASSERT_EQ(probes.at(0).target_data_rate, 1.5 * kStartBitrate);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      1.5 * kStartBitrate, BandwidthLimitedCause::kLossLimitedBweIncreasing,
+      fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, 1.5 * 1.5 * kStartBitrate);
+}
+
+TEST(ProbeControllerTest, NotProbeWhenInAlrIfLossBasedDecreases) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kLossLimitedBweDecreasing,
+      fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  // Not probe in alr when loss based estimate decreases.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, NotProbeIfLossBasedIncreasingOutsideAlr) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kLossLimitedBweIncreasing,
+      fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  probe_controller->SetAlrStartTimeMs(absl::nullopt);
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, ProbeFurtherWhenLossBasedIsSameAsDelayBasedEstimate) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 5 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  DataRate probe_target_rate = probes[0].target_data_rate;
+  EXPECT_LT(probe_target_rate, state_estimate.link_capacity_upper);
+  // Expect that more probes are sent if BWE is the same as delay based
+  // estimate.
+  probes = probe_controller->SetEstimatedBitrate(
+      probe_target_rate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_EQ(probes[0].target_data_rate, 2 * probe_target_rate);
+}
+
+TEST(ProbeControllerTest, ProbeIfEstimateLowerThanNetworkStateEstimate) {
+  // Periodic probe every 1 second if estimate is lower than 50% of the
+  // NetworkStateEstimate.
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/est_lower_than_network_interval:1s,"
+      "est_lower_than_network_ratio:0.5,limit_probe_"
+      "target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  state_estimate.link_capacity_upper = kStartBitrate * 3;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_GT(probes[0].target_data_rate, kStartBitrate);
+
+  // If network state not increased, send another probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+
+  // Stop probing if estimate increase. We might probe further here though.
+  probes = probe_controller->SetEstimatedBitrate(
+      2 * kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // No more periodic probes.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, DontProbeFurtherWhenLossLimited) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 3 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+  EXPECT_LT(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+  // Expect that no more probes are sent immediately if BWE is loss limited.
+  probes = probe_controller->SetEstimatedBitrate(
+      probes[0].target_data_rate,
+      BandwidthLimitedCause::kLossLimitedBweDecreasing, fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, ProbeFurtherWhenDelayBasedLimited) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 3 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+  EXPECT_LT(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+  // Since the probe was successfull, expect to continue probing.
+  probes = probe_controller->SetEstimatedBitrate(
+      probes[0].target_data_rate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+  EXPECT_EQ(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+}
+
+TEST(ProbeControllerTest,
+     ProbeFurtherIfNetworkStateEstimateIncreaseAfterProbeSent) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 1.2 * probes[0].target_data_rate / 2;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  // No immediate further probing since probe result is low.
+  probes = probe_controller->SetEstimatedBitrate(
+      probes[0].target_data_rate / 2, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_LE(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+  // If the network state estimate increase above the threshold to probe
+  // further, and the probe suceeed, expect a new probe.
+  state_estimate.link_capacity_upper = 3 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  probes = probe_controller->SetEstimatedBitrate(
+      probes[0].target_data_rate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+
+  // But no more probes if estimate is close to the link capacity.
+  probes = probe_controller->SetEstimatedBitrate(
+      state_estimate.link_capacity_upper * 0.9,
+      BandwidthLimitedCause::kDelayBasedLimited, fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, SkipAlrProbeIfEstimateLargerThanMaxProbe) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "skip_if_est_larger_than_fraction_of_max:0.9/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  // But if the max rate increase, A new probe is sent.
+  probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, 2 * kMaxBitrate, fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+}
+
+TEST(ProbeControllerTest,
+     SkipAlrProbeIfEstimateLargerThanFractionOfMaxAllocated) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "skip_if_est_larger_than_fraction_of_max:1.0/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate / 2, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(kMaxBitrate / 2,
+                                                        fixture.CurrentTime());
+  // No probes since total allocated is not higher than the current estimate.
+  EXPECT_TRUE(probes.empty());
+  fixture.AdvanceTime(TimeDelta::Seconds(2));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  // But if the max allocated increase, A new probe is sent.
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      kMaxBitrate / 2 + DataRate::BitsPerSec(1), fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+}
+
+TEST(ProbeControllerTest, SkipNetworkStateProbeIfEstimateLargerThanMaxProbe) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:2s,skip_if_est_larger_than_fraction_of_max:0.9/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = 2 * kMaxBitrate});
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, SendsProbeIfNetworkStateEstimateLowerThanMaxProbe) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:2s,skip_if_est_larger_than_fraction_of_max:0.9,"
+      "/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = 2 * kMaxBitrate});
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  // Need to wait at least two seconds before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(2100));
+
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = 2 * kStartBitrate});
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+}
+
+TEST(ProbeControllerTest, DontSendProbeIfNetworkStateEstimateIsZero) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = kStartBitrate});
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = DataRate::Zero()});
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+  fixture.AdvanceTime(TimeDelta::Seconds(6));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, DontProbeIfDelayIncreased) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,not_probe_if_delay_increased:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 3 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimitedDelayIncreased,
+      fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build
new file mode 100644
index 0000000000..e189363d3e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("pushback_controller_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc
new file mode 100644
index 0000000000..792a93d41e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc
@@ -0,0 +1,189 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/robust_throughput_estimator.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <utility>
+
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+RobustThroughputEstimator::RobustThroughputEstimator(
+    const RobustThroughputEstimatorSettings& settings)
+    : settings_(settings),
+      latest_discarded_send_time_(Timestamp::MinusInfinity()) {
+  RTC_DCHECK(settings.enabled);
+}
+
+RobustThroughputEstimator::~RobustThroughputEstimator() {}
+
+bool RobustThroughputEstimator::FirstPacketOutsideWindow() {
+  if (window_.empty())
+    return false;
+  if (window_.size() > settings_.max_window_packets)
+    return true;
+  TimeDelta current_window_duration =
+      window_.back().receive_time - window_.front().receive_time;
+  if (current_window_duration > settings_.max_window_duration)
+    return true;
+  if (window_.size() > settings_.window_packets &&
+      current_window_duration > settings_.min_window_duration) {
+    return true;
+  }
+  return false;
+}
+
+void RobustThroughputEstimator::IncomingPacketFeedbackVector(
+    const std::vector<PacketResult>& packet_feedback_vector) {
+  RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
+                            packet_feedback_vector.end(),
+                            PacketResult::ReceiveTimeOrder()));
+  for (const auto& packet : packet_feedback_vector) {
+    // Ignore packets without valid send or receive times.
+    // (This should not happen in production since lost packets are filtered
+    // out before passing the feedback vector to the throughput estimator.
+    // However, explicitly handling this case makes the estimator more robust
+    // and avoids a hard-to-detect bad state.)
+    if (packet.receive_time.IsInfinite() ||
+        packet.sent_packet.send_time.IsInfinite()) {
+      continue;
+    }
+
+    // Insert the new packet.
+    window_.push_back(packet);
+    window_.back().sent_packet.prior_unacked_data =
+        window_.back().sent_packet.prior_unacked_data *
+        settings_.unacked_weight;
+    // In most cases, receive timestamps should already be in order, but in the
+    // rare case where feedback packets have been reordered, we do some swaps to
+    // ensure that the window is sorted.
+    for (size_t i = window_.size() - 1;
+         i > 0 && window_[i].receive_time < window_[i - 1].receive_time; i--) {
+      std::swap(window_[i], window_[i - 1]);
+    }
+  }
+
+  // Remove old packets.
+  while (FirstPacketOutsideWindow()) {
+    latest_discarded_send_time_ = std::max(
+        latest_discarded_send_time_, window_.front().sent_packet.send_time);
+    window_.pop_front();
+  }
+}
+
+absl::optional<DataRate> RobustThroughputEstimator::bitrate() const {
+  if (window_.empty() || window_.size() < settings_.required_packets)
+    return absl::nullopt;
+
+  TimeDelta largest_recv_gap(TimeDelta::Zero());
+  TimeDelta second_largest_recv_gap(TimeDelta::Zero());
+  for (size_t i = 1; i < window_.size(); i++) {
+    // Find receive time gaps.
+    TimeDelta gap = window_[i].receive_time - window_[i - 1].receive_time;
+    if (gap > largest_recv_gap) {
+      second_largest_recv_gap = largest_recv_gap;
+      largest_recv_gap = gap;
+    } else if (gap > second_largest_recv_gap) {
+      second_largest_recv_gap = gap;
+    }
+  }
+
+  Timestamp first_send_time = Timestamp::PlusInfinity();
+  Timestamp last_send_time = Timestamp::MinusInfinity();
+  Timestamp first_recv_time = Timestamp::PlusInfinity();
+  Timestamp last_recv_time = Timestamp::MinusInfinity();
+  DataSize recv_size = DataSize::Bytes(0);
+  DataSize send_size = DataSize::Bytes(0);
+  DataSize first_recv_size = DataSize::Bytes(0);
+  DataSize last_send_size = DataSize::Bytes(0);
+  size_t num_sent_packets_in_window = 0;
+  for (const auto& packet : window_) {
+    if (packet.receive_time < first_recv_time) {
+      first_recv_time = packet.receive_time;
+      first_recv_size =
+          packet.sent_packet.size + packet.sent_packet.prior_unacked_data;
+    }
+    last_recv_time = std::max(last_recv_time, packet.receive_time);
+    recv_size += packet.sent_packet.size;
+    recv_size += packet.sent_packet.prior_unacked_data;
+
+    if (packet.sent_packet.send_time < latest_discarded_send_time_) {
+      // If we have dropped packets from the window that were sent after
+      // this packet, then this packet was reordered. Ignore it from
+      // the send rate computation (since the send time may be very far
+      // in the past, leading to underestimation of the send rate.)
+      // However, ignoring packets creates a risk that we end up without
+      // any packets left to compute a send rate.
+      continue;
+    }
+    if (packet.sent_packet.send_time > last_send_time) {
+      last_send_time = packet.sent_packet.send_time;
+      last_send_size =
+          packet.sent_packet.size + packet.sent_packet.prior_unacked_data;
+    }
+    first_send_time = std::min(first_send_time, packet.sent_packet.send_time);
+
+    send_size += packet.sent_packet.size;
+    send_size += packet.sent_packet.prior_unacked_data;
+    ++num_sent_packets_in_window;
+  }
+
+  // Suppose a packet of size S is sent every T milliseconds.
+  // A window of N packets would contain N*S bytes, but the time difference
+  // between the first and the last packet would only be (N-1)*T. Thus, we
+  // need to remove the size of one packet to get the correct rate of S/T.
+  // Which packet to remove (if the packets have varying sizes),
+  // depends on the network model.
+  // Suppose that 2 packets with sizes s1 and s2, are received at times t1
+  // and t2, respectively. If the packets were transmitted back to back over
+  // a bottleneck with rate capacity r, then we'd expect t2 = t1 + r * s2.
+  // Thus, r = (t2-t1) / s2, so the size of the first packet doesn't affect
+  // the difference between t1 and t2.
+  // Analoguously, if the first packet is sent at time t1 and the sender
+  // paces the packets at rate r, then the second packet can be sent at time
+  // t2 = t1 + r * s1. Thus, the send rate estimate r = (t2-t1) / s1 doesn't
+  // depend on the size of the last packet.
+  recv_size -= first_recv_size;
+  send_size -= last_send_size;
+
+  // Remove the largest gap by replacing it by the second largest gap.
+  // This is to ensure that spurious "delay spikes" (i.e. when the
+  // network stops transmitting packets for a short period, followed
+  // by a burst of delayed packets), don't cause the estimate to drop.
+  // This could cause an overestimation, which we guard against by
+  // never returning an estimate above the send rate.
+  RTC_DCHECK(first_recv_time.IsFinite());
+  RTC_DCHECK(last_recv_time.IsFinite());
+  TimeDelta recv_duration = (last_recv_time - first_recv_time) -
+                            largest_recv_gap + second_largest_recv_gap;
+  recv_duration = std::max(recv_duration, TimeDelta::Millis(1));
+
+  if (num_sent_packets_in_window < settings_.required_packets) {
+    // Too few send times to calculate a reliable send rate.
+    return recv_size / recv_duration;
+  }
+
+  RTC_DCHECK(first_send_time.IsFinite());
+  RTC_DCHECK(last_send_time.IsFinite());
+  TimeDelta send_duration = last_send_time - first_send_time;
+  send_duration = std::max(send_duration, TimeDelta::Millis(1));
+
+  return std::min(send_size / send_duration, recv_size / recv_duration);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.h
new file mode 100644
index 0000000000..9d89856496
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_
+
+#include <deque>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+
+namespace webrtc {
+
+class RobustThroughputEstimator : public AcknowledgedBitrateEstimatorInterface {
+ public:
+  explicit RobustThroughputEstimator(
+      const RobustThroughputEstimatorSettings& settings);
+  ~RobustThroughputEstimator() override;
+
+  void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) override;
+
+  absl::optional<DataRate> bitrate() const override;
+
+  absl::optional<DataRate> PeekRate() const override { return bitrate(); }
+  void SetAlr(bool /*in_alr*/) override {}
+  void SetAlrEndedTime(Timestamp /*alr_ended_time*/) override {}
+
+ private:
+  bool FirstPacketOutsideWindow();
+
+  const RobustThroughputEstimatorSettings settings_;
+  std::deque<PacketResult> window_;
+  Timestamp latest_discarded_send_time_ = Timestamp::MinusInfinity();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator_unittest.cc
new file mode 100644
index 0000000000..95ac525640
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator_unittest.cc
@@ -0,0 +1,427 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/robust_throughput_estimator.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <algorithm>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+RobustThroughputEstimatorSettings CreateRobustThroughputEstimatorSettings(
+    absl::string_view field_trial_string) {
+  test::ExplicitKeyValueConfig trials(field_trial_string);
+  RobustThroughputEstimatorSettings settings(&trials);
+  return settings;
+}
+
+class FeedbackGenerator {
+ public:
+  std::vector<PacketResult> CreateFeedbackVector(size_t number_of_packets,
+                                                 DataSize packet_size,
+                                                 DataRate send_rate,
+                                                 DataRate recv_rate) {
+    std::vector<PacketResult> packet_feedback_vector(number_of_packets);
+    for (size_t i = 0; i < number_of_packets; i++) {
+      packet_feedback_vector[i].sent_packet.send_time = send_clock_;
+      packet_feedback_vector[i].sent_packet.sequence_number = sequence_number_;
+      packet_feedback_vector[i].sent_packet.size = packet_size;
+      send_clock_ += packet_size / send_rate;
+      recv_clock_ += packet_size / recv_rate;
+      sequence_number_ += 1;
+      packet_feedback_vector[i].receive_time = recv_clock_;
+    }
+    return packet_feedback_vector;
+  }
+
+  Timestamp CurrentReceiveClock() { return recv_clock_; }
+
+  void AdvanceReceiveClock(TimeDelta delta) { recv_clock_ += delta; }
+
+  void AdvanceSendClock(TimeDelta delta) { send_clock_ += delta; }
+
+ private:
+  Timestamp send_clock_ = Timestamp::Millis(100000);
+  Timestamp recv_clock_ = Timestamp::Millis(10000);
+  uint16_t sequence_number_ = 100;
+};
+
+TEST(RobustThroughputEstimatorTest, InitialEstimate) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  // No estimate until the estimator has enough data.
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(9, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  EXPECT_FALSE(throughput_estimator.bitrate().has_value());
+
+  // Estimate once `required_packets` packets have been received.
+  packet_feedback = feedback_generator.CreateFeedbackVector(
+      1, DataSize::Bytes(1000), send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+
+  // Estimate remains stable when send and receive rates are stable.
+  packet_feedback = feedback_generator.CreateFeedbackVector(
+      15, DataSize::Bytes(1000), send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+}
+
+TEST(RobustThroughputEstimatorTest, EstimateAdapts) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+
+  // 1 second, 800kbps, estimate is stable.
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+  for (int i = 0; i < 10; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+
+  // 1 second, 1600kbps, estimate increases
+  send_rate = DataRate::BytesPerSec(200000);
+  recv_rate = DataRate::BytesPerSec(200000);
+  for (int i = 0; i < 20; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_GE(throughput.value(), DataRate::BytesPerSec(100000));
+    EXPECT_LE(throughput.value(), send_rate);
+  }
+
+  // 1 second, 1600kbps, estimate is stable
+  for (int i = 0; i < 20; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+
+  // 1 second, 400kbps, estimate decreases
+  send_rate = DataRate::BytesPerSec(50000);
+  recv_rate = DataRate::BytesPerSec(50000);
+  for (int i = 0; i < 5; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_LE(throughput.value(), DataRate::BytesPerSec(200000));
+    EXPECT_GE(throughput.value(), send_rate);
+  }
+
+  // 1 second, 400kbps, estimate is stable
+  send_rate = DataRate::BytesPerSec(50000);
+  recv_rate = DataRate::BytesPerSec(50000);
+  for (int i = 0; i < 5; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+}
+
+TEST(RobustThroughputEstimatorTest, CappedByReceiveRate) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(25000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              recv_rate.bytes_per_sec<double>(),
+              0.05 * recv_rate.bytes_per_sec<double>());  // Allow 5% error
+}
+
+TEST(RobustThroughputEstimatorTest, CappedBySendRate) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(50000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              send_rate.bytes_per_sec<double>(),
+              0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+}
+
+TEST(RobustThroughputEstimatorTest, DelaySpike) {
+  FeedbackGenerator feedback_generator;
+  // This test uses a 500ms window to amplify the effect
+  // of a delay spike.
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true,window_duration:500ms/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+
+  // Delay spike. 25 packets sent, but none received.
+  feedback_generator.AdvanceReceiveClock(TimeDelta::Millis(250));
+
+  // Deliver all of the packets during the next 50 ms. (During this time,
+  // we'll have sent an additional 5 packets, so we need to receive 30
+  // packets at 1000 bytes each in 50 ms, i.e. 600000 bytes per second).
+  recv_rate = DataRate::BytesPerSec(600000);
+  // Estimate should not drop.
+  for (int i = 0; i < 30; ++i) {
+    packet_feedback = feedback_generator.CreateFeedbackVector(
+        1, DataSize::Bytes(1000), send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+                send_rate.bytes_per_sec<double>(),
+                0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+  }
+
+  // Delivery at normal rate. When the packets received before the gap
+  // has left the estimator's window, the receive rate will be high, but the
+  // estimate should be capped by the send rate.
+  recv_rate = DataRate::BytesPerSec(100000);
+  for (int i = 0; i < 20; ++i) {
+    packet_feedback = feedback_generator.CreateFeedbackVector(
+        5, DataSize::Bytes(1000), send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+                send_rate.bytes_per_sec<double>(),
+                0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+  }
+}
+
+TEST(RobustThroughputEstimatorTest, HighLoss) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+
+  // 50% loss
+  for (size_t i = 0; i < packet_feedback.size(); i++) {
+    if (i % 2 == 1) {
+      packet_feedback[i].receive_time = Timestamp::PlusInfinity();
+    }
+  }
+
+  std::sort(packet_feedback.begin(), packet_feedback.end(),
+            PacketResult::ReceiveTimeOrder());
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              send_rate.bytes_per_sec<double>() / 2,
+              0.05 * send_rate.bytes_per_sec<double>() / 2);  // Allow 5% error
+}
+
+TEST(RobustThroughputEstimatorTest, ReorderedFeedback) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+
+  std::vector<PacketResult> delayed_feedback =
+      feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  packet_feedback = feedback_generator.CreateFeedbackVector(
+      10, DataSize::Bytes(1000), send_rate, recv_rate);
+
+  // Since we're missing some feedback, it's expected that the
+  // estimate will drop.
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_LT(throughput.value(), send_rate);
+
+  // But it should completely recover as soon as we get the feedback.
+  throughput_estimator.IncomingPacketFeedbackVector(delayed_feedback);
+  throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+
+  // It should then remain stable (as if the feedbacks weren't reordered.)
+  for (int i = 0; i < 10; ++i) {
+    packet_feedback = feedback_generator.CreateFeedbackVector(
+        15, DataSize::Bytes(1000), send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+}
+
+TEST(RobustThroughputEstimatorTest, DeepReordering) {
+  FeedbackGenerator feedback_generator;
+  // This test uses a 500ms window to amplify the
+  // effect of reordering.
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true,window_duration:500ms/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> delayed_packets =
+      feedback_generator.CreateFeedbackVector(1, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+
+  for (int i = 0; i < 10; i++) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+
+  // Delayed packet arrives ~1 second after it should have.
+  // Since the window is 500 ms, the delayed packet was sent ~500
+  // ms before the second oldest packet. However, the send rate
+  // should not drop.
+  delayed_packets.front().receive_time =
+      feedback_generator.CurrentReceiveClock();
+  throughput_estimator.IncomingPacketFeedbackVector(delayed_packets);
+  auto throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              send_rate.bytes_per_sec<double>(),
+              0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+
+  // Thoughput should stay stable.
+  for (int i = 0; i < 10; i++) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+                send_rate.bytes_per_sec<double>(),
+                0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+  }
+}
+
+TEST(RobustThroughputEstimatorTest, StreamPausedAndResumed) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  EXPECT_TRUE(throughput.has_value());
+  double expected_bytes_per_sec = 100 * 1000.0;
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              expected_bytes_per_sec,
+              0.05 * expected_bytes_per_sec);  // Allow 5% error
+
+  // No packets sent or feedback received for 60s.
+  feedback_generator.AdvanceSendClock(TimeDelta::Seconds(60));
+  feedback_generator.AdvanceReceiveClock(TimeDelta::Seconds(60));
+
+  // Resume sending packets at the same rate as before. The estimate
+  // will initially be invalid, due to lack of recent data.
+  packet_feedback = feedback_generator.CreateFeedbackVector(
+      5, DataSize::Bytes(1000), send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  throughput = throughput_estimator.bitrate();
+  EXPECT_FALSE(throughput.has_value());
+
+  // But be back to the normal level once we have enough data.
+  for (int i = 0; i < 4; ++i) {
+    packet_feedback = feedback_generator.CreateFeedbackVector(
+        5, DataSize::Bytes(1000), send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
new file mode 100644
index 0000000000..1e4db1ffaf
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
@@ -0,0 +1,695 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
+
+#include <algorithm>
+#include <cstdio>
+#include <limits>
+#include <memory>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+constexpr TimeDelta kBweIncreaseInterval = TimeDelta::Millis(1000);
+constexpr TimeDelta kBweDecreaseInterval = TimeDelta::Millis(300);
+constexpr TimeDelta kStartPhase = TimeDelta::Millis(2000);
+constexpr TimeDelta kBweConverganceTime = TimeDelta::Millis(20000);
+constexpr int kLimitNumPackets = 20;
+constexpr DataRate kDefaultMaxBitrate = DataRate::BitsPerSec(1000000000);
+constexpr TimeDelta kLowBitrateLogPeriod = TimeDelta::Millis(10000);
+constexpr TimeDelta kRtcEventLogPeriod = TimeDelta::Millis(5000);
+// Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals.
+constexpr TimeDelta kMaxRtcpFeedbackInterval = TimeDelta::Millis(5000);
+
+constexpr float kDefaultLowLossThreshold = 0.02f;
+constexpr float kDefaultHighLossThreshold = 0.1f;
+constexpr DataRate kDefaultBitrateThreshold = DataRate::Zero();
+
+struct UmaRampUpMetric {
+  const char* metric_name;
+  int bitrate_kbps;
+};
+
+const UmaRampUpMetric kUmaRampupMetrics[] = {
+    {"WebRTC.BWE.RampUpTimeTo500kbpsInMs", 500},
+    {"WebRTC.BWE.RampUpTimeTo1000kbpsInMs", 1000},
+    {"WebRTC.BWE.RampUpTimeTo2000kbpsInMs", 2000}};
+const size_t kNumUmaRampupMetrics =
+    sizeof(kUmaRampupMetrics) / sizeof(kUmaRampupMetrics[0]);
+
+const char kBweLosExperiment[] = "WebRTC-BweLossExperiment";
+
+bool BweLossExperimentIsEnabled() {
+  std::string experiment_string =
+      webrtc::field_trial::FindFullName(kBweLosExperiment);
+  // The experiment is enabled iff the field trial string begins with "Enabled".
+  return absl::StartsWith(experiment_string, "Enabled");
+}
+
+bool ReadBweLossExperimentParameters(float* low_loss_threshold,
+                                     float* high_loss_threshold,
+                                     uint32_t* bitrate_threshold_kbps) {
+  RTC_DCHECK(low_loss_threshold);
+  RTC_DCHECK(high_loss_threshold);
+  RTC_DCHECK(bitrate_threshold_kbps);
+  std::string experiment_string =
+      webrtc::field_trial::FindFullName(kBweLosExperiment);
+  int parsed_values =
+      sscanf(experiment_string.c_str(), "Enabled-%f,%f,%u", low_loss_threshold,
+             high_loss_threshold, bitrate_threshold_kbps);
+  if (parsed_values == 3) {
+    RTC_CHECK_GT(*low_loss_threshold, 0.0f)
+        << "Loss threshold must be greater than 0.";
+    RTC_CHECK_LE(*low_loss_threshold, 1.0f)
+        << "Loss threshold must be less than or equal to 1.";
+    RTC_CHECK_GT(*high_loss_threshold, 0.0f)
+        << "Loss threshold must be greater than 0.";
+    RTC_CHECK_LE(*high_loss_threshold, 1.0f)
+        << "Loss threshold must be less than or equal to 1.";
+    RTC_CHECK_LE(*low_loss_threshold, *high_loss_threshold)
+        << "The low loss threshold must be less than or equal to the high loss "
+           "threshold.";
+    RTC_CHECK_GE(*bitrate_threshold_kbps, 0)
+        << "Bitrate threshold can't be negative.";
+    RTC_CHECK_LT(*bitrate_threshold_kbps,
+                 std::numeric_limits<int>::max() / 1000)
+        << "Bitrate must be smaller enough to avoid overflows.";
+    return true;
+  }
+  RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweLossExperiment "
+                         "experiment from field trial string. Using default.";
+  *low_loss_threshold = kDefaultLowLossThreshold;
+  *high_loss_threshold = kDefaultHighLossThreshold;
+  *bitrate_threshold_kbps = kDefaultBitrateThreshold.kbps();
+  return false;
+}
+}  // namespace
+
+LinkCapacityTracker::LinkCapacityTracker()
+    : tracking_rate("rate", TimeDelta::Seconds(10)) {
+  ParseFieldTrial({&tracking_rate},
+                  field_trial::FindFullName("WebRTC-Bwe-LinkCapacity"));
+}
+
+LinkCapacityTracker::~LinkCapacityTracker() {}
+
+void LinkCapacityTracker::UpdateDelayBasedEstimate(
+    Timestamp at_time,
+    DataRate delay_based_bitrate) {
+  if (delay_based_bitrate < last_delay_based_estimate_) {
+    capacity_estimate_bps_ =
+        std::min(capacity_estimate_bps_, delay_based_bitrate.bps<double>());
+    last_link_capacity_update_ = at_time;
+  }
+  last_delay_based_estimate_ = delay_based_bitrate;
+}
+
+void LinkCapacityTracker::OnStartingRate(DataRate start_rate) {
+  if (last_link_capacity_update_.IsInfinite())
+    capacity_estimate_bps_ = start_rate.bps<double>();
+}
+
+void LinkCapacityTracker::OnRateUpdate(absl::optional<DataRate> acknowledged,
+                                       DataRate target,
+                                       Timestamp at_time) {
+  if (!acknowledged)
+    return;
+  DataRate acknowledged_target = std::min(*acknowledged, target);
+  if (acknowledged_target.bps() > capacity_estimate_bps_) {
+    TimeDelta delta = at_time - last_link_capacity_update_;
+    double alpha = delta.IsFinite() ? exp(-(delta / tracking_rate.Get())) : 0;
+    capacity_estimate_bps_ = alpha * capacity_estimate_bps_ +
+                             (1 - alpha) * acknowledged_target.bps<double>();
+  }
+  last_link_capacity_update_ = at_time;
+}
+
+void LinkCapacityTracker::OnRttBackoff(DataRate backoff_rate,
+                                       Timestamp at_time) {
+  capacity_estimate_bps_ =
+      std::min(capacity_estimate_bps_, backoff_rate.bps<double>());
+  last_link_capacity_update_ = at_time;
+}
+
+DataRate LinkCapacityTracker::estimate() const {
+  return DataRate::BitsPerSec(capacity_estimate_bps_);
+}
+
+RttBasedBackoff::RttBasedBackoff(const FieldTrialsView* key_value_config)
+    : disabled_("Disabled"),
+      configured_limit_("limit", TimeDelta::Seconds(3)),
+      drop_fraction_("fraction", 0.8),
+      drop_interval_("interval", TimeDelta::Seconds(1)),
+      bandwidth_floor_("floor", DataRate::KilobitsPerSec(5)),
+      rtt_limit_(TimeDelta::PlusInfinity()),
+      // By initializing this to plus infinity, we make sure that we never
+      // trigger rtt backoff unless packet feedback is enabled.
+      last_propagation_rtt_update_(Timestamp::PlusInfinity()),
+      last_propagation_rtt_(TimeDelta::Zero()),
+      last_packet_sent_(Timestamp::MinusInfinity()) {
+  ParseFieldTrial({&disabled_, &configured_limit_, &drop_fraction_,
+                   &drop_interval_, &bandwidth_floor_},
+                  key_value_config->Lookup("WebRTC-Bwe-MaxRttLimit"));
+  if (!disabled_) {
+    rtt_limit_ = configured_limit_.Get();
+  }
+}
+
+void RttBasedBackoff::UpdatePropagationRtt(Timestamp at_time,
+                                           TimeDelta propagation_rtt) {
+  last_propagation_rtt_update_ = at_time;
+  last_propagation_rtt_ = propagation_rtt;
+}
+
+TimeDelta RttBasedBackoff::CorrectedRtt(Timestamp at_time) const {
+  TimeDelta time_since_rtt = at_time - last_propagation_rtt_update_;
+  TimeDelta timeout_correction = time_since_rtt;
+  // Avoid timeout when no packets are being sent.
+  TimeDelta time_since_packet_sent = at_time - last_packet_sent_;
+  timeout_correction =
+      std::max(time_since_rtt - time_since_packet_sent, TimeDelta::Zero());
+  return timeout_correction + last_propagation_rtt_;
+}
+
+RttBasedBackoff::~RttBasedBackoff() = default;
+
+SendSideBandwidthEstimation::SendSideBandwidthEstimation(
+    const FieldTrialsView* key_value_config,
+    RtcEventLog* event_log)
+    : rtt_backoff_(key_value_config),
+      lost_packets_since_last_loss_update_(0),
+      expected_packets_since_last_loss_update_(0),
+      current_target_(DataRate::Zero()),
+      last_logged_target_(DataRate::Zero()),
+      min_bitrate_configured_(kCongestionControllerMinBitrate),
+      max_bitrate_configured_(kDefaultMaxBitrate),
+      last_low_bitrate_log_(Timestamp::MinusInfinity()),
+      has_decreased_since_last_fraction_loss_(false),
+      last_loss_feedback_(Timestamp::MinusInfinity()),
+      last_loss_packet_report_(Timestamp::MinusInfinity()),
+      last_fraction_loss_(0),
+      last_logged_fraction_loss_(0),
+      last_round_trip_time_(TimeDelta::Zero()),
+      receiver_limit_(DataRate::PlusInfinity()),
+      delay_based_limit_(DataRate::PlusInfinity()),
+      time_last_decrease_(Timestamp::MinusInfinity()),
+      first_report_time_(Timestamp::MinusInfinity()),
+      initially_lost_packets_(0),
+      bitrate_at_2_seconds_(DataRate::Zero()),
+      uma_update_state_(kNoUpdate),
+      uma_rtt_state_(kNoUpdate),
+      rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
+      event_log_(event_log),
+      last_rtc_event_log_(Timestamp::MinusInfinity()),
+      low_loss_threshold_(kDefaultLowLossThreshold),
+      high_loss_threshold_(kDefaultHighLossThreshold),
+      bitrate_threshold_(kDefaultBitrateThreshold),
+      loss_based_bandwidth_estimator_v1_(key_value_config),
+      loss_based_bandwidth_estimator_v2_(key_value_config),
+      loss_based_state_(LossBasedState::kDelayBasedEstimate),
+      disable_receiver_limit_caps_only_("Disabled") {
+  RTC_DCHECK(event_log);
+  if (BweLossExperimentIsEnabled()) {
+    uint32_t bitrate_threshold_kbps;
+    if (ReadBweLossExperimentParameters(&low_loss_threshold_,
+                                        &high_loss_threshold_,
+                                        &bitrate_threshold_kbps)) {
+      RTC_LOG(LS_INFO) << "Enabled BweLossExperiment with parameters "
+                       << low_loss_threshold_ << ", " << high_loss_threshold_
+                       << ", " << bitrate_threshold_kbps;
+      bitrate_threshold_ = DataRate::KilobitsPerSec(bitrate_threshold_kbps);
+    }
+  }
+  ParseFieldTrial({&disable_receiver_limit_caps_only_},
+                  key_value_config->Lookup("WebRTC-Bwe-ReceiverLimitCapsOnly"));
+  if (LossBasedBandwidthEstimatorV2Enabled()) {
+    loss_based_bandwidth_estimator_v2_.SetMinMaxBitrate(
+        min_bitrate_configured_, max_bitrate_configured_);
+  }
+}
+
+SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
+
+void SendSideBandwidthEstimation::OnRouteChange() {
+  lost_packets_since_last_loss_update_ = 0;
+  expected_packets_since_last_loss_update_ = 0;
+  current_target_ = DataRate::Zero();
+  min_bitrate_configured_ = kCongestionControllerMinBitrate;
+  max_bitrate_configured_ = kDefaultMaxBitrate;
+  last_low_bitrate_log_ = Timestamp::MinusInfinity();
+  has_decreased_since_last_fraction_loss_ = false;
+  last_loss_feedback_ = Timestamp::MinusInfinity();
+  last_loss_packet_report_ = Timestamp::MinusInfinity();
+  last_fraction_loss_ = 0;
+  last_logged_fraction_loss_ = 0;
+  last_round_trip_time_ = TimeDelta::Zero();
+  receiver_limit_ = DataRate::PlusInfinity();
+  delay_based_limit_ = DataRate::PlusInfinity();
+  time_last_decrease_ = Timestamp::MinusInfinity();
+  first_report_time_ = Timestamp::MinusInfinity();
+  initially_lost_packets_ = 0;
+  bitrate_at_2_seconds_ = DataRate::Zero();
+  uma_update_state_ = kNoUpdate;
+  uma_rtt_state_ = kNoUpdate;
+  last_rtc_event_log_ = Timestamp::MinusInfinity();
+}
+
+void SendSideBandwidthEstimation::SetBitrates(
+    absl::optional<DataRate> send_bitrate,
+    DataRate min_bitrate,
+    DataRate max_bitrate,
+    Timestamp at_time) {
+  SetMinMaxBitrate(min_bitrate, max_bitrate);
+  if (send_bitrate) {
+    link_capacity_.OnStartingRate(*send_bitrate);
+    SetSendBitrate(*send_bitrate, at_time);
+  }
+}
+
+void SendSideBandwidthEstimation::SetSendBitrate(DataRate bitrate,
+                                                 Timestamp at_time) {
+  RTC_DCHECK_GT(bitrate, DataRate::Zero());
+  // Reset to avoid being capped by the estimate.
+  delay_based_limit_ = DataRate::PlusInfinity();
+  UpdateTargetBitrate(bitrate, at_time);
+  // Clear last sent bitrate history so the new value can be used directly
+  // and not capped.
+  min_bitrate_history_.clear();
+}
+
+void SendSideBandwidthEstimation::SetMinMaxBitrate(DataRate min_bitrate,
+                                                   DataRate max_bitrate) {
+  min_bitrate_configured_ =
+      std::max(min_bitrate, kCongestionControllerMinBitrate);
+  if (max_bitrate > DataRate::Zero() && max_bitrate.IsFinite()) {
+    max_bitrate_configured_ = std::max(min_bitrate_configured_, max_bitrate);
+  } else {
+    max_bitrate_configured_ = kDefaultMaxBitrate;
+  }
+  loss_based_bandwidth_estimator_v2_.SetMinMaxBitrate(min_bitrate_configured_,
+                                                      max_bitrate_configured_);
+}
+
+int SendSideBandwidthEstimation::GetMinBitrate() const {
+  return min_bitrate_configured_.bps<int>();
+}
+
+DataRate SendSideBandwidthEstimation::target_rate() const {
+  DataRate target = current_target_;
+  if (!disable_receiver_limit_caps_only_)
+    target = std::min(target, receiver_limit_);
+  return std::max(min_bitrate_configured_, target);
+}
+
+LossBasedState SendSideBandwidthEstimation::loss_based_state() const {
+  return loss_based_state_;
+}
+
+DataRate SendSideBandwidthEstimation::GetEstimatedLinkCapacity() const {
+  return link_capacity_.estimate();
+}
+
+void SendSideBandwidthEstimation::UpdateReceiverEstimate(Timestamp at_time,
+                                                         DataRate bandwidth) {
+  // TODO(srte): Ensure caller passes PlusInfinity, not zero, to represent no
+  // limitation.
+  receiver_limit_ = bandwidth.IsZero() ? DataRate::PlusInfinity() : bandwidth;
+  ApplyTargetLimits(at_time);
+}
+
+void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(Timestamp at_time,
+                                                           DataRate bitrate) {
+  link_capacity_.UpdateDelayBasedEstimate(at_time, bitrate);
+  // TODO(srte): Ensure caller passes PlusInfinity, not zero, to represent no
+  // limitation.
+  delay_based_limit_ = bitrate.IsZero() ? DataRate::PlusInfinity() : bitrate;
+  ApplyTargetLimits(at_time);
+}
+
+void SendSideBandwidthEstimation::SetAcknowledgedRate(
+    absl::optional<DataRate> acknowledged_rate,
+    Timestamp at_time) {
+  acknowledged_rate_ = acknowledged_rate;
+  if (!acknowledged_rate.has_value()) {
+    return;
+  }
+  if (LossBasedBandwidthEstimatorV1Enabled()) {
+    loss_based_bandwidth_estimator_v1_.UpdateAcknowledgedBitrate(
+        *acknowledged_rate, at_time);
+  }
+  if (LossBasedBandwidthEstimatorV2Enabled()) {
+    loss_based_bandwidth_estimator_v2_.SetAcknowledgedBitrate(
+        *acknowledged_rate);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateLossBasedEstimator(
+    const TransportPacketsFeedback& report,
+    BandwidthUsage delay_detector_state,
+    absl::optional<DataRate> probe_bitrate,
+    DataRate upper_link_capacity) {
+  if (LossBasedBandwidthEstimatorV1Enabled()) {
+    loss_based_bandwidth_estimator_v1_.UpdateLossStatistics(
+        report.packet_feedbacks, report.feedback_time);
+  }
+  if (LossBasedBandwidthEstimatorV2Enabled()) {
+    loss_based_bandwidth_estimator_v2_.UpdateBandwidthEstimate(
+        report.packet_feedbacks, delay_based_limit_, delay_detector_state,
+        probe_bitrate, upper_link_capacity);
+    UpdateEstimate(report.feedback_time);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdatePacketsLost(int64_t packets_lost,
+                                                    int64_t number_of_packets,
+                                                    Timestamp at_time) {
+  last_loss_feedback_ = at_time;
+  if (first_report_time_.IsInfinite())
+    first_report_time_ = at_time;
+
+  // Check sequence number diff and weight loss report
+  if (number_of_packets > 0) {
+    int64_t expected =
+        expected_packets_since_last_loss_update_ + number_of_packets;
+
+    // Don't generate a loss rate until it can be based on enough packets.
+    if (expected < kLimitNumPackets) {
+      // Accumulate reports.
+      expected_packets_since_last_loss_update_ = expected;
+      lost_packets_since_last_loss_update_ += packets_lost;
+      return;
+    }
+
+    has_decreased_since_last_fraction_loss_ = false;
+    int64_t lost_q8 =
+        std::max<int64_t>(lost_packets_since_last_loss_update_ + packets_lost,
+                          0)
+        << 8;
+    last_fraction_loss_ = std::min<int>(lost_q8 / expected, 255);
+
+    // Reset accumulators.
+    lost_packets_since_last_loss_update_ = 0;
+    expected_packets_since_last_loss_update_ = 0;
+    last_loss_packet_report_ = at_time;
+    UpdateEstimate(at_time);
+  }
+
+  UpdateUmaStatsPacketsLost(at_time, packets_lost);
+}
+
+void SendSideBandwidthEstimation::UpdateUmaStatsPacketsLost(Timestamp at_time,
+                                                            int packets_lost) {
+  DataRate bitrate_kbps =
+      DataRate::KilobitsPerSec((current_target_.bps() + 500) / 1000);
+  for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
+    if (!rampup_uma_stats_updated_[i] &&
+        bitrate_kbps.kbps() >= kUmaRampupMetrics[i].bitrate_kbps) {
+      RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
+                                   (at_time - first_report_time_).ms());
+      rampup_uma_stats_updated_[i] = true;
+    }
+  }
+  if (IsInStartPhase(at_time)) {
+    initially_lost_packets_ += packets_lost;
+  } else if (uma_update_state_ == kNoUpdate) {
+    uma_update_state_ = kFirstDone;
+    bitrate_at_2_seconds_ = bitrate_kbps;
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
+                         initially_lost_packets_, 0, 100, 50);
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
+                         bitrate_at_2_seconds_.kbps(), 0, 2000, 50);
+  } else if (uma_update_state_ == kFirstDone &&
+             at_time - first_report_time_ >= kBweConverganceTime) {
+    uma_update_state_ = kDone;
+    int bitrate_diff_kbps = std::max(
+        bitrate_at_2_seconds_.kbps<int>() - bitrate_kbps.kbps<int>(), 0);
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps,
+                         0, 2000, 50);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateRtt(TimeDelta rtt, Timestamp at_time) {
+  // Update RTT if we were able to compute an RTT based on this RTCP.
+  // FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
+  if (rtt > TimeDelta::Zero())
+    last_round_trip_time_ = rtt;
+
+  if (!IsInStartPhase(at_time) && uma_rtt_state_ == kNoUpdate) {
+    uma_rtt_state_ = kDone;
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", rtt.ms<int>(), 0, 2000, 50);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) {
+  if (rtt_backoff_.CorrectedRtt(at_time) > rtt_backoff_.rtt_limit_) {
+    if (at_time - time_last_decrease_ >= rtt_backoff_.drop_interval_ &&
+        current_target_ > rtt_backoff_.bandwidth_floor_) {
+      time_last_decrease_ = at_time;
+      DataRate new_bitrate =
+          std::max(current_target_ * rtt_backoff_.drop_fraction_,
+                   rtt_backoff_.bandwidth_floor_.Get());
+      link_capacity_.OnRttBackoff(new_bitrate, at_time);
+      UpdateTargetBitrate(new_bitrate, at_time);
+      return;
+    }
+    // TODO(srte): This is likely redundant in most cases.
+    ApplyTargetLimits(at_time);
+    return;
+  }
+
+  // We trust the REMB and/or delay-based estimate during the first 2 seconds if
+  // we haven't had any packet loss reported, to allow startup bitrate probing.
+  if (last_fraction_loss_ == 0 && IsInStartPhase(at_time)) {
+    DataRate new_bitrate = current_target_;
+    // TODO(srte): We should not allow the new_bitrate to be larger than the
+    // receiver limit here.
+    if (receiver_limit_.IsFinite())
+      new_bitrate = std::max(receiver_limit_, new_bitrate);
+    if (delay_based_limit_.IsFinite())
+      new_bitrate = std::max(delay_based_limit_, new_bitrate);
+    if (LossBasedBandwidthEstimatorV1Enabled()) {
+      loss_based_bandwidth_estimator_v1_.Initialize(new_bitrate);
+    }
+    if (LossBasedBandwidthEstimatorV2Enabled()) {
+      loss_based_bandwidth_estimator_v2_.SetBandwidthEstimate(new_bitrate);
+    }
+
+    if (new_bitrate != current_target_) {
+      min_bitrate_history_.clear();
+      if (LossBasedBandwidthEstimatorV1Enabled()) {
+        min_bitrate_history_.push_back(std::make_pair(at_time, new_bitrate));
+      } else {
+        min_bitrate_history_.push_back(
+            std::make_pair(at_time, current_target_));
+      }
+      UpdateTargetBitrate(new_bitrate, at_time);
+      return;
+    }
+  }
+  UpdateMinHistory(at_time);
+  if (last_loss_packet_report_.IsInfinite()) {
+    // No feedback received.
+    // TODO(srte): This is likely redundant in most cases.
+    ApplyTargetLimits(at_time);
+    return;
+  }
+
+  if (LossBasedBandwidthEstimatorV1ReadyForUse()) {
+    DataRate new_bitrate = loss_based_bandwidth_estimator_v1_.Update(
+        at_time, min_bitrate_history_.front().second, delay_based_limit_,
+        last_round_trip_time_);
+    UpdateTargetBitrate(new_bitrate, at_time);
+    return;
+  }
+
+  if (LossBasedBandwidthEstimatorV2ReadyForUse()) {
+    LossBasedBweV2::Result result =
+        loss_based_bandwidth_estimator_v2_.GetLossBasedResult();
+    loss_based_state_ = result.state;
+    UpdateTargetBitrate(result.bandwidth_estimate, at_time);
+    return;
+  }
+
+  TimeDelta time_since_loss_packet_report = at_time - last_loss_packet_report_;
+  if (time_since_loss_packet_report < 1.2 * kMaxRtcpFeedbackInterval) {
+    // We only care about loss above a given bitrate threshold.
+    float loss = last_fraction_loss_ / 256.0f;
+    // We only make decisions based on loss when the bitrate is above a
+    // threshold. This is a crude way of handling loss which is uncorrelated
+    // to congestion.
+    if (current_target_ < bitrate_threshold_ || loss <= low_loss_threshold_) {
+      // Loss < 2%: Increase rate by 8% of the min bitrate in the last
+      // kBweIncreaseInterval.
+      // Note that by remembering the bitrate over the last second one can
+      // rampup up one second faster than if only allowed to start ramping
+      // at 8% per second rate now. E.g.:
+      //   If sending a constant 100kbps it can rampup immediately to 108kbps
+      //   whenever a receiver report is received with lower packet loss.
+      //   If instead one would do: current_bitrate_ *= 1.08^(delta time),
+      //   it would take over one second since the lower packet loss to achieve
+      //   108kbps.
+      DataRate new_bitrate = DataRate::BitsPerSec(
+          min_bitrate_history_.front().second.bps() * 1.08 + 0.5);
+
+      // Add 1 kbps extra, just to make sure that we do not get stuck
+      // (gives a little extra increase at low rates, negligible at higher
+      // rates).
+      new_bitrate += DataRate::BitsPerSec(1000);
+      UpdateTargetBitrate(new_bitrate, at_time);
+      return;
+    } else if (current_target_ > bitrate_threshold_) {
+      if (loss <= high_loss_threshold_) {
+        // Loss between 2% - 10%: Do nothing.
+      } else {
+        // Loss > 10%: Limit the rate decreases to once a kBweDecreaseInterval
+        // + rtt.
+        if (!has_decreased_since_last_fraction_loss_ &&
+            (at_time - time_last_decrease_) >=
+                (kBweDecreaseInterval + last_round_trip_time_)) {
+          time_last_decrease_ = at_time;
+
+          // Reduce rate:
+          //   newRate = rate * (1 - 0.5*lossRate);
+          //   where packetLoss = 256*lossRate;
+          DataRate new_bitrate = DataRate::BitsPerSec(
+              (current_target_.bps() *
+               static_cast<double>(512 - last_fraction_loss_)) /
+              512.0);
+          has_decreased_since_last_fraction_loss_ = true;
+          UpdateTargetBitrate(new_bitrate, at_time);
+          return;
+        }
+      }
+    }
+  }
+  // TODO(srte): This is likely redundant in most cases.
+  ApplyTargetLimits(at_time);
+}
+
+void SendSideBandwidthEstimation::UpdatePropagationRtt(
+    Timestamp at_time,
+    TimeDelta propagation_rtt) {
+  rtt_backoff_.UpdatePropagationRtt(at_time, propagation_rtt);
+}
+
+void SendSideBandwidthEstimation::OnSentPacket(const SentPacket& sent_packet) {
+  // Only feedback-triggering packets will be reported here.
+  rtt_backoff_.last_packet_sent_ = sent_packet.send_time;
+}
+
+bool SendSideBandwidthEstimation::IsInStartPhase(Timestamp at_time) const {
+  return first_report_time_.IsInfinite() ||
+         at_time - first_report_time_ < kStartPhase;
+}
+
+void SendSideBandwidthEstimation::UpdateMinHistory(Timestamp at_time) {
+  // Remove old data points from history.
+  // Since history precision is in ms, add one so it is able to increase
+  // bitrate if it is off by as little as 0.5ms.
+  while (!min_bitrate_history_.empty() &&
+         at_time - min_bitrate_history_.front().first + TimeDelta::Millis(1) >
+             kBweIncreaseInterval) {
+    min_bitrate_history_.pop_front();
+  }
+
+  // Typical minimum sliding-window algorithm: Pop values higher than current
+  // bitrate before pushing it.
+  while (!min_bitrate_history_.empty() &&
+         current_target_ <= min_bitrate_history_.back().second) {
+    min_bitrate_history_.pop_back();
+  }
+
+  min_bitrate_history_.push_back(std::make_pair(at_time, current_target_));
+}
+
+DataRate SendSideBandwidthEstimation::GetUpperLimit() const {
+  DataRate upper_limit = delay_based_limit_;
+  if (disable_receiver_limit_caps_only_)
+    upper_limit = std::min(upper_limit, receiver_limit_);
+  return std::min(upper_limit, max_bitrate_configured_);
+}
+
+void SendSideBandwidthEstimation::MaybeLogLowBitrateWarning(DataRate bitrate,
+                                                            Timestamp at_time) {
+  if (at_time - last_low_bitrate_log_ > kLowBitrateLogPeriod) {
+    RTC_LOG(LS_WARNING) << "Estimated available bandwidth " << ToString(bitrate)
+                        << " is below configured min bitrate "
+                        << ToString(min_bitrate_configured_) << ".";
+    last_low_bitrate_log_ = at_time;
+  }
+}
+
+void SendSideBandwidthEstimation::MaybeLogLossBasedEvent(Timestamp at_time) {
+  if (current_target_ != last_logged_target_ ||
+      last_fraction_loss_ != last_logged_fraction_loss_ ||
+      at_time - last_rtc_event_log_ > kRtcEventLogPeriod) {
+    event_log_->Log(std::make_unique<RtcEventBweUpdateLossBased>(
+        current_target_.bps(), last_fraction_loss_,
+        expected_packets_since_last_loss_update_));
+    last_logged_fraction_loss_ = last_fraction_loss_;
+    last_logged_target_ = current_target_;
+    last_rtc_event_log_ = at_time;
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateTargetBitrate(DataRate new_bitrate,
+                                                      Timestamp at_time) {
+  new_bitrate = std::min(new_bitrate, GetUpperLimit());
+  if (new_bitrate < min_bitrate_configured_) {
+    MaybeLogLowBitrateWarning(new_bitrate, at_time);
+    new_bitrate = min_bitrate_configured_;
+  }
+  current_target_ = new_bitrate;
+  MaybeLogLossBasedEvent(at_time);
+  link_capacity_.OnRateUpdate(acknowledged_rate_, current_target_, at_time);
+}
+
+void SendSideBandwidthEstimation::ApplyTargetLimits(Timestamp at_time) {
+  UpdateTargetBitrate(current_target_, at_time);
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV1Enabled() const {
+  return loss_based_bandwidth_estimator_v1_.Enabled() &&
+         !LossBasedBandwidthEstimatorV2Enabled();
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV1ReadyForUse()
+    const {
+  return LossBasedBandwidthEstimatorV1Enabled() &&
+         loss_based_bandwidth_estimator_v1_.InUse();
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2Enabled() const {
+  return loss_based_bandwidth_estimator_v2_.IsEnabled();
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2ReadyForUse()
+    const {
+  return LossBasedBandwidthEstimatorV2Enabled() &&
+         loss_based_bandwidth_estimator_v2_.IsReady();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
new file mode 100644
index 0000000000..77510236d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
@@ -0,0 +1,210 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ *  FEC and NACK added bitrate is handled outside class
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
+
+#include <stdint.h>
+
+#include <deque>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class LinkCapacityTracker {
+ public:
+  LinkCapacityTracker();
+  ~LinkCapacityTracker();
+  // Call when a new delay-based estimate is available.
+  void UpdateDelayBasedEstimate(Timestamp at_time,
+                                DataRate delay_based_bitrate);
+  void OnStartingRate(DataRate start_rate);
+  void OnRateUpdate(absl::optional<DataRate> acknowledged,
+                    DataRate target,
+                    Timestamp at_time);
+  void OnRttBackoff(DataRate backoff_rate, Timestamp at_time);
+  DataRate estimate() const;
+
+ private:
+  FieldTrialParameter<TimeDelta> tracking_rate;
+  double capacity_estimate_bps_ = 0;
+  Timestamp last_link_capacity_update_ = Timestamp::MinusInfinity();
+  DataRate last_delay_based_estimate_ = DataRate::PlusInfinity();
+};
+
+class RttBasedBackoff {
+ public:
+  explicit RttBasedBackoff(const FieldTrialsView* key_value_config);
+  ~RttBasedBackoff();
+  void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
+  TimeDelta CorrectedRtt(Timestamp at_time) const;
+
+  FieldTrialFlag disabled_;
+  FieldTrialParameter<TimeDelta> configured_limit_;
+  FieldTrialParameter<double> drop_fraction_;
+  FieldTrialParameter<TimeDelta> drop_interval_;
+  FieldTrialParameter<DataRate> bandwidth_floor_;
+
+ public:
+  TimeDelta rtt_limit_;
+  Timestamp last_propagation_rtt_update_;
+  TimeDelta last_propagation_rtt_;
+  Timestamp last_packet_sent_;
+};
+
+class SendSideBandwidthEstimation {
+ public:
+  SendSideBandwidthEstimation() = delete;
+  SendSideBandwidthEstimation(const FieldTrialsView* key_value_config,
+                              RtcEventLog* event_log);
+  ~SendSideBandwidthEstimation();
+
+  void OnRouteChange();
+
+  DataRate target_rate() const;
+  LossBasedState loss_based_state() const;
+  uint8_t fraction_loss() const { return last_fraction_loss_; }
+  TimeDelta round_trip_time() const { return last_round_trip_time_; }
+
+  DataRate GetEstimatedLinkCapacity() const;
+  // Call periodically to update estimate.
+  void UpdateEstimate(Timestamp at_time);
+  void OnSentPacket(const SentPacket& sent_packet);
+  void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
+
+  // Call when we receive a RTCP message with TMMBR or REMB.
+  void UpdateReceiverEstimate(Timestamp at_time, DataRate bandwidth);
+
+  // Call when a new delay-based estimate is available.
+  void UpdateDelayBasedEstimate(Timestamp at_time, DataRate bitrate);
+
+  // Call when we receive a RTCP message with a ReceiveBlock.
+  void UpdatePacketsLost(int64_t packets_lost,
+                         int64_t number_of_packets,
+                         Timestamp at_time);
+
+  // Call when we receive a RTCP message with a ReceiveBlock.
+  void UpdateRtt(TimeDelta rtt, Timestamp at_time);
+
+  void SetBitrates(absl::optional<DataRate> send_bitrate,
+                   DataRate min_bitrate,
+                   DataRate max_bitrate,
+                   Timestamp at_time);
+  void SetSendBitrate(DataRate bitrate, Timestamp at_time);
+  void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate);
+  int GetMinBitrate() const;
+  void SetAcknowledgedRate(absl::optional<DataRate> acknowledged_rate,
+                           Timestamp at_time);
+  void UpdateLossBasedEstimator(const TransportPacketsFeedback& report,
+                                BandwidthUsage delay_detector_state,
+                                absl::optional<DataRate> probe_bitrate,
+                                DataRate upper_link_capacity);
+
+ private:
+  friend class GoogCcStatePrinter;
+
+  enum UmaState { kNoUpdate, kFirstDone, kDone };
+
+  bool IsInStartPhase(Timestamp at_time) const;
+
+  void UpdateUmaStatsPacketsLost(Timestamp at_time, int packets_lost);
+
+  // Updates history of min bitrates.
+  // After this method returns min_bitrate_history_.front().second contains the
+  // min bitrate used during last kBweIncreaseIntervalMs.
+  void UpdateMinHistory(Timestamp at_time);
+
+  // Gets the upper limit for the target bitrate. This is the minimum of the
+  // delay based limit, the receiver limit and the loss based controller limit.
+  DataRate GetUpperLimit() const;
+  // Prints a warning if `bitrate` if sufficiently long time has past since last
+  // warning.
+  void MaybeLogLowBitrateWarning(DataRate bitrate, Timestamp at_time);
+  // Stores an update to the event log if the loss rate has changed, the target
+  // has changed, or sufficient time has passed since last stored event.
+  void MaybeLogLossBasedEvent(Timestamp at_time);
+
+  // Cap `bitrate` to [min_bitrate_configured_, max_bitrate_configured_] and
+  // set `current_bitrate_` to the capped value and updates the event log.
+  void UpdateTargetBitrate(DataRate bitrate, Timestamp at_time);
+  // Applies lower and upper bounds to the current target rate.
+  // TODO(srte): This seems to be called even when limits haven't changed, that
+  // should be cleaned up.
+  void ApplyTargetLimits(Timestamp at_time);
+
+  bool LossBasedBandwidthEstimatorV1Enabled() const;
+  bool LossBasedBandwidthEstimatorV2Enabled() const;
+
+  bool LossBasedBandwidthEstimatorV1ReadyForUse() const;
+  bool LossBasedBandwidthEstimatorV2ReadyForUse() const;
+
+  RttBasedBackoff rtt_backoff_;
+  LinkCapacityTracker link_capacity_;
+
+  std::deque<std::pair<Timestamp, DataRate> > min_bitrate_history_;
+
+  // incoming filters
+  int lost_packets_since_last_loss_update_;
+  int expected_packets_since_last_loss_update_;
+
+  absl::optional<DataRate> acknowledged_rate_;
+  DataRate current_target_;
+  DataRate last_logged_target_;
+  DataRate min_bitrate_configured_;
+  DataRate max_bitrate_configured_;
+  Timestamp last_low_bitrate_log_;
+
+  bool has_decreased_since_last_fraction_loss_;
+  Timestamp last_loss_feedback_;
+  Timestamp last_loss_packet_report_;
+  uint8_t last_fraction_loss_;
+  uint8_t last_logged_fraction_loss_;
+  TimeDelta last_round_trip_time_;
+
+  // The max bitrate as set by the receiver in the call. This is typically
+  // signalled using the REMB RTCP message and is used when we don't have any
+  // send side delay based estimate.
+  DataRate receiver_limit_;
+  DataRate delay_based_limit_;
+  Timestamp time_last_decrease_;
+  Timestamp first_report_time_;
+  int initially_lost_packets_;
+  DataRate bitrate_at_2_seconds_;
+  UmaState uma_update_state_;
+  UmaState uma_rtt_state_;
+  std::vector<bool> rampup_uma_stats_updated_;
+  RtcEventLog* const event_log_;
+  Timestamp last_rtc_event_log_;
+  float low_loss_threshold_;
+  float high_loss_threshold_;
+  DataRate bitrate_threshold_;
+  LossBasedBandwidthEstimation loss_based_bandwidth_estimator_v1_;
+  LossBasedBweV2 loss_based_bandwidth_estimator_v2_;
+  LossBasedState loss_based_state_;
+  FieldTrialFlag disable_receiver_limit_caps_only_;
+};
+}  // namespace webrtc
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation_unittest.cc
new file mode 100644
index 0000000000..17d1aa1ada
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation_unittest.cc
@@ -0,0 +1,206 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
+
+#include "api/rtc_event_log/rtc_event.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+MATCHER(LossBasedBweUpdateWithBitrateOnly, "") {
+  if (arg->GetType() != RtcEvent::Type::BweUpdateLossBased) {
+    return false;
+  }
+  auto bwe_event = static_cast<RtcEventBweUpdateLossBased*>(arg);
+  return bwe_event->bitrate_bps() > 0 && bwe_event->fraction_loss() == 0;
+}
+
+MATCHER(LossBasedBweUpdateWithBitrateAndLossFraction, "") {
+  if (arg->GetType() != RtcEvent::Type::BweUpdateLossBased) {
+    return false;
+  }
+  auto bwe_event = static_cast<RtcEventBweUpdateLossBased*>(arg);
+  return bwe_event->bitrate_bps() > 0 && bwe_event->fraction_loss() > 0;
+}
+
+void TestProbing(bool use_delay_based) {
+  ::testing::NiceMock<MockRtcEventLog> event_log;
+  test::ExplicitKeyValueConfig key_value_config("");
+  SendSideBandwidthEstimation bwe(&key_value_config, &event_log);
+  int64_t now_ms = 0;
+  bwe.SetMinMaxBitrate(DataRate::BitsPerSec(100000),
+                       DataRate::BitsPerSec(1500000));
+  bwe.SetSendBitrate(DataRate::BitsPerSec(200000), Timestamp::Millis(now_ms));
+
+  const int kRembBps = 1000000;
+  const int kSecondRembBps = kRembBps + 500000;
+
+  bwe.UpdatePacketsLost(/*packets_lost=*/0, /*number_of_packets=*/1,
+                        Timestamp::Millis(now_ms));
+  bwe.UpdateRtt(TimeDelta::Millis(50), Timestamp::Millis(now_ms));
+
+  // Initial REMB applies immediately.
+  if (use_delay_based) {
+    bwe.UpdateDelayBasedEstimate(Timestamp::Millis(now_ms),
+                                 DataRate::BitsPerSec(kRembBps));
+  } else {
+    bwe.UpdateReceiverEstimate(Timestamp::Millis(now_ms),
+                               DataRate::BitsPerSec(kRembBps));
+  }
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+  EXPECT_EQ(kRembBps, bwe.target_rate().bps());
+
+  // Second REMB doesn't apply immediately.
+  now_ms += 2001;
+  if (use_delay_based) {
+    bwe.UpdateDelayBasedEstimate(Timestamp::Millis(now_ms),
+                                 DataRate::BitsPerSec(kSecondRembBps));
+  } else {
+    bwe.UpdateReceiverEstimate(Timestamp::Millis(now_ms),
+                               DataRate::BitsPerSec(kSecondRembBps));
+  }
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+  EXPECT_EQ(kRembBps, bwe.target_rate().bps());
+}
+
+TEST(SendSideBweTest, InitialRembWithProbing) {
+  TestProbing(false);
+}
+
+TEST(SendSideBweTest, InitialDelayBasedBweWithProbing) {
+  TestProbing(true);
+}
+
+TEST(SendSideBweTest, DoesntReapplyBitrateDecreaseWithoutFollowingRemb) {
+  MockRtcEventLog event_log;
+  EXPECT_CALL(event_log, LogProxy(LossBasedBweUpdateWithBitrateOnly()))
+      .Times(1);
+  EXPECT_CALL(event_log,
+              LogProxy(LossBasedBweUpdateWithBitrateAndLossFraction()))
+      .Times(1);
+  test::ExplicitKeyValueConfig key_value_config("");
+  SendSideBandwidthEstimation bwe(&key_value_config, &event_log);
+  static const int kMinBitrateBps = 100000;
+  static const int kInitialBitrateBps = 1000000;
+  int64_t now_ms = 1000;
+  bwe.SetMinMaxBitrate(DataRate::BitsPerSec(kMinBitrateBps),
+                       DataRate::BitsPerSec(1500000));
+  bwe.SetSendBitrate(DataRate::BitsPerSec(kInitialBitrateBps),
+                     Timestamp::Millis(now_ms));
+
+  static const uint8_t kFractionLoss = 128;
+  static const int64_t kRttMs = 50;
+  now_ms += 10000;
+
+  EXPECT_EQ(kInitialBitrateBps, bwe.target_rate().bps());
+  EXPECT_EQ(0, bwe.fraction_loss());
+  EXPECT_EQ(0, bwe.round_trip_time().ms());
+
+  // Signal heavy loss to go down in bitrate.
+  bwe.UpdatePacketsLost(/*packets_lost=*/50, /*number_of_packets=*/100,
+                        Timestamp::Millis(now_ms));
+  bwe.UpdateRtt(TimeDelta::Millis(kRttMs), Timestamp::Millis(now_ms));
+
+  // Trigger an update 2 seconds later to not be rate limited.
+  now_ms += 1000;
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+  EXPECT_LT(bwe.target_rate().bps(), kInitialBitrateBps);
+  // Verify that the obtained bitrate isn't hitting the min bitrate, or this
+  // test doesn't make sense. If this ever happens, update the thresholds or
+  // loss rates so that it doesn't hit min bitrate after one bitrate update.
+  EXPECT_GT(bwe.target_rate().bps(), kMinBitrateBps);
+  EXPECT_EQ(kFractionLoss, bwe.fraction_loss());
+  EXPECT_EQ(kRttMs, bwe.round_trip_time().ms());
+
+  // Triggering an update shouldn't apply further downgrade nor upgrade since
+  // there's no intermediate receiver block received indicating whether this is
+  // currently good or not.
+  int last_bitrate_bps = bwe.target_rate().bps();
+  // Trigger an update 2 seconds later to not be rate limited (but it still
+  // shouldn't update).
+  now_ms += 1000;
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+
+  EXPECT_EQ(last_bitrate_bps, bwe.target_rate().bps());
+  // The old loss rate should still be applied though.
+  EXPECT_EQ(kFractionLoss, bwe.fraction_loss());
+  EXPECT_EQ(kRttMs, bwe.round_trip_time().ms());
+}
+
+TEST(SendSideBweTest, SettingSendBitrateOverridesDelayBasedEstimate) {
+  ::testing::NiceMock<MockRtcEventLog> event_log;
+  test::ExplicitKeyValueConfig key_value_config("");
+  SendSideBandwidthEstimation bwe(&key_value_config, &event_log);
+  static const int kMinBitrateBps = 10000;
+  static const int kMaxBitrateBps = 10000000;
+  static const int kInitialBitrateBps = 300000;
+  static const int kDelayBasedBitrateBps = 350000;
+  static const int kForcedHighBitrate = 2500000;
+
+  int64_t now_ms = 0;
+
+  bwe.SetMinMaxBitrate(DataRate::BitsPerSec(kMinBitrateBps),
+                       DataRate::BitsPerSec(kMaxBitrateBps));
+  bwe.SetSendBitrate(DataRate::BitsPerSec(kInitialBitrateBps),
+                     Timestamp::Millis(now_ms));
+
+  bwe.UpdateDelayBasedEstimate(Timestamp::Millis(now_ms),
+                               DataRate::BitsPerSec(kDelayBasedBitrateBps));
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+  EXPECT_GE(bwe.target_rate().bps(), kInitialBitrateBps);
+  EXPECT_LE(bwe.target_rate().bps(), kDelayBasedBitrateBps);
+
+  bwe.SetSendBitrate(DataRate::BitsPerSec(kForcedHighBitrate),
+                     Timestamp::Millis(now_ms));
+  EXPECT_EQ(bwe.target_rate().bps(), kForcedHighBitrate);
+}
+
+TEST(RttBasedBackoff, DefaultEnabled) {
+  test::ExplicitKeyValueConfig key_value_config("");
+  RttBasedBackoff rtt_backoff(&key_value_config);
+  EXPECT_TRUE(rtt_backoff.rtt_limit_.IsFinite());
+}
+
+TEST(RttBasedBackoff, CanBeDisabled) {
+  test::ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-MaxRttLimit/Disabled/");
+  RttBasedBackoff rtt_backoff(&key_value_config);
+  EXPECT_TRUE(rtt_backoff.rtt_limit_.IsPlusInfinity());
+}
+
+TEST(SendSideBweTest, FractionLossIsNotOverflowed) {
+  MockRtcEventLog event_log;
+  test::ExplicitKeyValueConfig key_value_config("");
+  SendSideBandwidthEstimation bwe(&key_value_config, &event_log);
+  static const int kMinBitrateBps = 100000;
+  static const int kInitialBitrateBps = 1000000;
+  int64_t now_ms = 1000;
+  bwe.SetMinMaxBitrate(DataRate::BitsPerSec(kMinBitrateBps),
+                       DataRate::BitsPerSec(1500000));
+  bwe.SetSendBitrate(DataRate::BitsPerSec(kInitialBitrateBps),
+                     Timestamp::Millis(now_ms));
+
+  now_ms += 10000;
+
+  EXPECT_EQ(kInitialBitrateBps, bwe.target_rate().bps());
+  EXPECT_EQ(0, bwe.fraction_loss());
+
+  // Signal negative loss.
+  bwe.UpdatePacketsLost(/*packets_lost=*/-1, /*number_of_packets=*/100,
+                        Timestamp::Millis(now_ms));
+  EXPECT_EQ(0, bwe.fraction_loss());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build
new file mode 100644
index 0000000000..c93e412e83
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("send_side_bwe_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
new file mode 100644
index 0000000000..6a8849ed6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
@@ -0,0 +1,200 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/goog_cc/test/goog_cc_printer.h"
+
+#include <math.h>
+
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+void WriteTypedValue(RtcEventLogOutput* out, int value) {
+  LogWriteFormat(out, "%i", value);
+}
+void WriteTypedValue(RtcEventLogOutput* out, double value) {
+  LogWriteFormat(out, "%.6f", value);
+}
+void WriteTypedValue(RtcEventLogOutput* out, absl::optional<DataRate> value) {
+  LogWriteFormat(out, "%.0f", value ? value->bytes_per_sec<double>() : NAN);
+}
+void WriteTypedValue(RtcEventLogOutput* out, absl::optional<DataSize> value) {
+  LogWriteFormat(out, "%.0f", value ? value->bytes<double>() : NAN);
+}
+void WriteTypedValue(RtcEventLogOutput* out, absl::optional<TimeDelta> value) {
+  LogWriteFormat(out, "%.3f", value ? value->seconds<double>() : NAN);
+}
+void WriteTypedValue(RtcEventLogOutput* out, absl::optional<Timestamp> value) {
+  LogWriteFormat(out, "%.3f", value ? value->seconds<double>() : NAN);
+}
+
+template <typename F>
+class TypedFieldLogger : public FieldLogger {
+ public:
+  TypedFieldLogger(absl::string_view name, F&& getter)
+      : name_(name), getter_(std::forward<F>(getter)) {}
+  const std::string& name() const override { return name_; }
+  void WriteValue(RtcEventLogOutput* out) override {
+    WriteTypedValue(out, getter_());
+  }
+
+ private:
+  std::string name_;
+  F getter_;
+};
+
+template <typename F>
+FieldLogger* Log(absl::string_view name, F&& getter) {
+  return new TypedFieldLogger<F>(name, std::forward<F>(getter));
+}
+
+}  // namespace
+GoogCcStatePrinter::GoogCcStatePrinter() {
+  for (auto* logger : CreateLoggers()) {
+    loggers_.emplace_back(logger);
+  }
+}
+
+std::deque<FieldLogger*> GoogCcStatePrinter::CreateLoggers() {
+  auto stable_estimate = [this] {
+    return DataRate::KilobitsPerSec(
+        controller_->delay_based_bwe_->rate_control_.link_capacity_
+            .estimate_kbps_.value_or(-INFINITY));
+  };
+  auto rate_control_state = [this] {
+    return static_cast<int>(
+        controller_->delay_based_bwe_->rate_control_.rate_control_state_);
+  };
+  auto trend = [this] {
+    return reinterpret_cast<TrendlineEstimator*>(
+        controller_->delay_based_bwe_->active_delay_detector_);
+  };
+  auto acknowledged_rate = [this] {
+    return controller_->acknowledged_bitrate_estimator_->bitrate();
+  };
+  auto loss_cont = [&] {
+    return &controller_->bandwidth_estimation_
+                ->loss_based_bandwidth_estimator_v1_;
+  };
+  std::deque<FieldLogger*> loggers({
+      Log("time", [=] { return target_.at_time; }),
+      Log("rtt", [=] { return target_.network_estimate.round_trip_time; }),
+      Log("target", [=] { return target_.target_rate; }),
+      Log("stable_target", [=] { return target_.stable_target_rate; }),
+      Log("pacing", [=] { return pacing_.data_rate(); }),
+      Log("padding", [=] { return pacing_.pad_rate(); }),
+      Log("window", [=] { return congestion_window_; }),
+      Log("rate_control_state", [=] { return rate_control_state(); }),
+      Log("stable_estimate", [=] { return stable_estimate(); }),
+      Log("trendline", [=] { return trend()->prev_trend_; }),
+      Log("trendline_modified_offset",
+          [=] { return trend()->prev_modified_trend_; }),
+      Log("trendline_offset_threshold", [=] { return trend()->threshold_; }),
+      Log("acknowledged_rate", [=] { return acknowledged_rate(); }),
+      Log("est_capacity", [=] { return est_.link_capacity; }),
+      Log("est_capacity_dev", [=] { return est_.link_capacity_std_dev; }),
+      Log("est_capacity_min", [=] { return est_.link_capacity_min; }),
+      Log("est_cross_traffic", [=] { return est_.cross_traffic_ratio; }),
+      Log("est_cross_delay", [=] { return est_.cross_delay_rate; }),
+      Log("est_spike_delay", [=] { return est_.spike_delay_rate; }),
+      Log("est_pre_buffer", [=] { return est_.pre_link_buffer_delay; }),
+      Log("est_post_buffer", [=] { return est_.post_link_buffer_delay; }),
+      Log("est_propagation", [=] { return est_.propagation_delay; }),
+      Log("loss_ratio", [=] { return loss_cont()->last_loss_ratio_; }),
+      Log("loss_average", [=] { return loss_cont()->average_loss_; }),
+      Log("loss_average_max", [=] { return loss_cont()->average_loss_max_; }),
+      Log("loss_thres_inc",
+          [=] { return loss_cont()->loss_increase_threshold(); }),
+      Log("loss_thres_dec",
+          [=] { return loss_cont()->loss_decrease_threshold(); }),
+      Log("loss_dec_rate", [=] { return loss_cont()->decreased_bitrate(); }),
+      Log("loss_based_rate", [=] { return loss_cont()->loss_based_bitrate_; }),
+      Log("loss_ack_rate",
+          [=] { return loss_cont()->acknowledged_bitrate_max_; }),
+      Log("data_window", [=] { return controller_->current_data_window_; }),
+      Log("pushback_target",
+          [=] { return controller_->last_pushback_target_rate_; }),
+  });
+  return loggers;
+}
+GoogCcStatePrinter::~GoogCcStatePrinter() = default;
+
+void GoogCcStatePrinter::PrintHeaders(RtcEventLogOutput* log) {
+  int ix = 0;
+  for (const auto& logger : loggers_) {
+    if (ix++)
+      log->Write(" ");
+    log->Write(logger->name());
+  }
+  log->Write("\n");
+  log->Flush();
+}
+
+void GoogCcStatePrinter::PrintState(RtcEventLogOutput* log,
+                                    GoogCcNetworkController* controller,
+                                    Timestamp at_time) {
+  controller_ = controller;
+  auto state_update = controller_->GetNetworkState(at_time);
+  target_ = state_update.target_rate.value();
+  pacing_ = state_update.pacer_config.value();
+  if (state_update.congestion_window)
+    congestion_window_ = *state_update.congestion_window;
+  if (controller_->network_estimator_) {
+    est_ = controller_->network_estimator_->GetCurrentEstimate().value_or(
+        NetworkStateEstimate());
+  }
+
+  int ix = 0;
+  for (const auto& logger : loggers_) {
+    if (ix++)
+      log->Write(" ");
+    logger->WriteValue(log);
+  }
+
+  log->Write("\n");
+  log->Flush();
+}
+
+GoogCcDebugFactory::GoogCcDebugFactory()
+    : GoogCcDebugFactory(GoogCcFactoryConfig()) {}
+
+GoogCcDebugFactory::GoogCcDebugFactory(GoogCcFactoryConfig config)
+    : GoogCcNetworkControllerFactory(std::move(config)) {}
+
+std::unique_ptr<NetworkControllerInterface> GoogCcDebugFactory::Create(
+    NetworkControllerConfig config) {
+  RTC_CHECK(controller_ == nullptr);
+  auto controller = GoogCcNetworkControllerFactory::Create(config);
+  controller_ = static_cast<GoogCcNetworkController*>(controller.get());
+  return controller;
+}
+
+void GoogCcDebugFactory::PrintState(const Timestamp at_time) {
+  if (controller_ && log_writer_) {
+    printer_.PrintState(log_writer_.get(), controller_, at_time);
+  }
+}
+
+void GoogCcDebugFactory::AttachWriter(
+    std::unique_ptr<RtcEventLogOutput> log_writer) {
+  if (log_writer) {
+    log_writer_ = std::move(log_writer);
+    printer_.PrintHeaders(log_writer_.get());
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h
new file mode 100644
index 0000000000..16fa657e71
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_TEST_GOOG_CC_PRINTER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_TEST_GOOG_CC_PRINTER_H_
+
+#include <deque>
+#include <memory>
+#include <string>
+
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/goog_cc_factory.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
+#include "test/logging/log_writer.h"
+
+namespace webrtc {
+
+class FieldLogger {
+ public:
+  virtual ~FieldLogger() = default;
+  virtual const std::string& name() const = 0;
+  virtual void WriteValue(RtcEventLogOutput* out) = 0;
+};
+
+class GoogCcStatePrinter {
+ public:
+  GoogCcStatePrinter();
+  GoogCcStatePrinter(const GoogCcStatePrinter&) = delete;
+  GoogCcStatePrinter& operator=(const GoogCcStatePrinter&) = delete;
+  ~GoogCcStatePrinter();
+
+  void PrintHeaders(RtcEventLogOutput* log);
+  void PrintState(RtcEventLogOutput* log,
+                  GoogCcNetworkController* controller,
+                  Timestamp at_time);
+
+ private:
+  std::deque<FieldLogger*> CreateLoggers();
+  std::deque<std::unique_ptr<FieldLogger>> loggers_;
+
+  GoogCcNetworkController* controller_ = nullptr;
+  TargetTransferRate target_;
+  PacerConfig pacing_;
+  DataSize congestion_window_ = DataSize::PlusInfinity();
+  NetworkStateEstimate est_;
+};
+
+class GoogCcDebugFactory : public GoogCcNetworkControllerFactory {
+ public:
+  GoogCcDebugFactory();
+  explicit GoogCcDebugFactory(GoogCcFactoryConfig config);
+  std::unique_ptr<NetworkControllerInterface> Create(
+      NetworkControllerConfig config) override;
+
+  void PrintState(Timestamp at_time);
+
+  void AttachWriter(std::unique_ptr<RtcEventLogOutput> log_writer);
+
+ private:
+  GoogCcStatePrinter printer_;
+  GoogCcNetworkController* controller_ = nullptr;
+  std::unique_ptr<RtcEventLogOutput> log_writer_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_TEST_GOOG_CC_PRINTER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc
new file mode 100644
index 0000000000..88182d4f80
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc
@@ -0,0 +1,332 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/network_state_predictor.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+namespace {
+
+// Parameters for linear least squares fit of regression line to noisy data.
+constexpr double kDefaultTrendlineSmoothingCoeff = 0.9;
+constexpr double kDefaultTrendlineThresholdGain = 4.0;
+const char kBweWindowSizeInPacketsExperiment[] =
+    "WebRTC-BweWindowSizeInPackets";
+
+size_t ReadTrendlineFilterWindowSize(const FieldTrialsView* key_value_config) {
+  std::string experiment_string =
+      key_value_config->Lookup(kBweWindowSizeInPacketsExperiment);
+  size_t window_size;
+  int parsed_values =
+      sscanf(experiment_string.c_str(), "Enabled-%zu", &window_size);
+  if (parsed_values == 1) {
+    if (window_size > 1)
+      return window_size;
+    RTC_LOG(LS_WARNING) << "Window size must be greater than 1.";
+  }
+  RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweWindowSizeInPackets"
+                         " experiment from field trial string. Using default.";
+  return TrendlineEstimatorSettings::kDefaultTrendlineWindowSize;
+}
+
+absl::optional<double> LinearFitSlope(
+    const std::deque<TrendlineEstimator::PacketTiming>& packets) {
+  RTC_DCHECK(packets.size() >= 2);
+  // Compute the "center of mass".
+  double sum_x = 0;
+  double sum_y = 0;
+  for (const auto& packet : packets) {
+    sum_x += packet.arrival_time_ms;
+    sum_y += packet.smoothed_delay_ms;
+  }
+  double x_avg = sum_x / packets.size();
+  double y_avg = sum_y / packets.size();
+  // Compute the slope k = \sum (x_i-x_avg)(y_i-y_avg) / \sum (x_i-x_avg)^2
+  double numerator = 0;
+  double denominator = 0;
+  for (const auto& packet : packets) {
+    double x = packet.arrival_time_ms;
+    double y = packet.smoothed_delay_ms;
+    numerator += (x - x_avg) * (y - y_avg);
+    denominator += (x - x_avg) * (x - x_avg);
+  }
+  if (denominator == 0)
+    return absl::nullopt;
+  return numerator / denominator;
+}
+
+absl::optional<double> ComputeSlopeCap(
+    const std::deque<TrendlineEstimator::PacketTiming>& packets,
+    const TrendlineEstimatorSettings& settings) {
+  RTC_DCHECK(1 <= settings.beginning_packets &&
+             settings.beginning_packets < packets.size());
+  RTC_DCHECK(1 <= settings.end_packets &&
+             settings.end_packets < packets.size());
+  RTC_DCHECK(settings.beginning_packets + settings.end_packets <=
+             packets.size());
+  TrendlineEstimator::PacketTiming early = packets[0];
+  for (size_t i = 1; i < settings.beginning_packets; ++i) {
+    if (packets[i].raw_delay_ms < early.raw_delay_ms)
+      early = packets[i];
+  }
+  size_t late_start = packets.size() - settings.end_packets;
+  TrendlineEstimator::PacketTiming late = packets[late_start];
+  for (size_t i = late_start + 1; i < packets.size(); ++i) {
+    if (packets[i].raw_delay_ms < late.raw_delay_ms)
+      late = packets[i];
+  }
+  if (late.arrival_time_ms - early.arrival_time_ms < 1) {
+    return absl::nullopt;
+  }
+  return (late.raw_delay_ms - early.raw_delay_ms) /
+             (late.arrival_time_ms - early.arrival_time_ms) +
+         settings.cap_uncertainty;
+}
+
+constexpr double kMaxAdaptOffsetMs = 15.0;
+constexpr double kOverUsingTimeThreshold = 10;
+constexpr int kMinNumDeltas = 60;
+constexpr int kDeltaCounterMax = 1000;
+
+}  // namespace
+
+constexpr char TrendlineEstimatorSettings::kKey[];
+
+TrendlineEstimatorSettings::TrendlineEstimatorSettings(
+    const FieldTrialsView* key_value_config) {
+  if (absl::StartsWith(
+          key_value_config->Lookup(kBweWindowSizeInPacketsExperiment),
+          "Enabled")) {
+    window_size = ReadTrendlineFilterWindowSize(key_value_config);
+  }
+  Parser()->Parse(key_value_config->Lookup(TrendlineEstimatorSettings::kKey));
+  if (window_size < 10 || 200 < window_size) {
+    RTC_LOG(LS_WARNING) << "Window size must be between 10 and 200 packets";
+    window_size = kDefaultTrendlineWindowSize;
+  }
+  if (enable_cap) {
+    if (beginning_packets < 1 || end_packets < 1 ||
+        beginning_packets > window_size || end_packets > window_size) {
+      RTC_LOG(LS_WARNING) << "Size of beginning and end must be between 1 and "
+                          << window_size;
+      enable_cap = false;
+      beginning_packets = end_packets = 0;
+      cap_uncertainty = 0.0;
+    }
+    if (beginning_packets + end_packets > window_size) {
+      RTC_LOG(LS_WARNING)
+          << "Size of beginning plus end can't exceed the window size";
+      enable_cap = false;
+      beginning_packets = end_packets = 0;
+      cap_uncertainty = 0.0;
+    }
+    if (cap_uncertainty < 0.0 || 0.025 < cap_uncertainty) {
+      RTC_LOG(LS_WARNING) << "Cap uncertainty must be between 0 and 0.025";
+      cap_uncertainty = 0.0;
+    }
+  }
+}
+
+std::unique_ptr<StructParametersParser> TrendlineEstimatorSettings::Parser() {
+  return StructParametersParser::Create("sort", &enable_sort,  //
+                                        "cap", &enable_cap,    //
+                                        "beginning_packets",
+                                        &beginning_packets,                   //
+                                        "end_packets", &end_packets,          //
+                                        "cap_uncertainty", &cap_uncertainty,  //
+                                        "window_size", &window_size);
+}
+
+TrendlineEstimator::TrendlineEstimator(
+    const FieldTrialsView* key_value_config,
+    NetworkStatePredictor* network_state_predictor)
+    : settings_(key_value_config),
+      smoothing_coef_(kDefaultTrendlineSmoothingCoeff),
+      threshold_gain_(kDefaultTrendlineThresholdGain),
+      num_of_deltas_(0),
+      first_arrival_time_ms_(-1),
+      accumulated_delay_(0),
+      smoothed_delay_(0),
+      delay_hist_(),
+      k_up_(0.0087),
+      k_down_(0.039),
+      overusing_time_threshold_(kOverUsingTimeThreshold),
+      threshold_(12.5),
+      prev_modified_trend_(NAN),
+      last_update_ms_(-1),
+      prev_trend_(0.0),
+      time_over_using_(-1),
+      overuse_counter_(0),
+      hypothesis_(BandwidthUsage::kBwNormal),
+      hypothesis_predicted_(BandwidthUsage::kBwNormal),
+      network_state_predictor_(network_state_predictor) {
+  RTC_LOG(LS_INFO)
+      << "Using Trendline filter for delay change estimation with settings "
+      << settings_.Parser()->Encode() << " and "
+      << (network_state_predictor_ ? "injected" : "no")
+      << " network state predictor";
+}
+
+TrendlineEstimator::~TrendlineEstimator() {}
+
+void TrendlineEstimator::UpdateTrendline(double recv_delta_ms,
+                                         double send_delta_ms,
+                                         int64_t send_time_ms,
+                                         int64_t arrival_time_ms,
+                                         size_t packet_size) {
+  const double delta_ms = recv_delta_ms - send_delta_ms;
+  ++num_of_deltas_;
+  num_of_deltas_ = std::min(num_of_deltas_, kDeltaCounterMax);
+  if (first_arrival_time_ms_ == -1)
+    first_arrival_time_ms_ = arrival_time_ms;
+
+  // Exponential backoff filter.
+  accumulated_delay_ += delta_ms;
+  BWE_TEST_LOGGING_PLOT(1, "accumulated_delay_ms", arrival_time_ms,
+                        accumulated_delay_);
+  smoothed_delay_ = smoothing_coef_ * smoothed_delay_ +
+                    (1 - smoothing_coef_) * accumulated_delay_;
+  BWE_TEST_LOGGING_PLOT(1, "smoothed_delay_ms", arrival_time_ms,
+                        smoothed_delay_);
+
+  // Maintain packet window
+  delay_hist_.emplace_back(
+      static_cast<double>(arrival_time_ms - first_arrival_time_ms_),
+      smoothed_delay_, accumulated_delay_);
+  if (settings_.enable_sort) {
+    for (size_t i = delay_hist_.size() - 1;
+         i > 0 &&
+         delay_hist_[i].arrival_time_ms < delay_hist_[i - 1].arrival_time_ms;
+         --i) {
+      std::swap(delay_hist_[i], delay_hist_[i - 1]);
+    }
+  }
+  if (delay_hist_.size() > settings_.window_size)
+    delay_hist_.pop_front();
+
+  // Simple linear regression.
+  double trend = prev_trend_;
+  if (delay_hist_.size() == settings_.window_size) {
+    // Update trend_ if it is possible to fit a line to the data. The delay
+    // trend can be seen as an estimate of (send_rate - capacity)/capacity.
+    // 0 < trend < 1   ->  the delay increases, queues are filling up
+    //   trend == 0    ->  the delay does not change
+    //   trend < 0     ->  the delay decreases, queues are being emptied
+    trend = LinearFitSlope(delay_hist_).value_or(trend);
+    if (settings_.enable_cap) {
+      absl::optional<double> cap = ComputeSlopeCap(delay_hist_, settings_);
+      // We only use the cap to filter out overuse detections, not
+      // to detect additional underuses.
+      if (trend >= 0 && cap.has_value() && trend > cap.value()) {
+        trend = cap.value();
+      }
+    }
+  }
+  BWE_TEST_LOGGING_PLOT(1, "trendline_slope", arrival_time_ms, trend);
+
+  Detect(trend, send_delta_ms, arrival_time_ms);
+}
+
+void TrendlineEstimator::Update(double recv_delta_ms,
+                                double send_delta_ms,
+                                int64_t send_time_ms,
+                                int64_t arrival_time_ms,
+                                size_t packet_size,
+                                bool calculated_deltas) {
+  if (calculated_deltas) {
+    UpdateTrendline(recv_delta_ms, send_delta_ms, send_time_ms, arrival_time_ms,
+                    packet_size);
+  }
+  if (network_state_predictor_) {
+    hypothesis_predicted_ = network_state_predictor_->Update(
+        send_time_ms, arrival_time_ms, hypothesis_);
+  }
+}
+
+BandwidthUsage TrendlineEstimator::State() const {
+  return network_state_predictor_ ? hypothesis_predicted_ : hypothesis_;
+}
+
+void TrendlineEstimator::Detect(double trend, double ts_delta, int64_t now_ms) {
+  if (num_of_deltas_ < 2) {
+    hypothesis_ = BandwidthUsage::kBwNormal;
+    return;
+  }
+  const double modified_trend =
+      std::min(num_of_deltas_, kMinNumDeltas) * trend * threshold_gain_;
+  prev_modified_trend_ = modified_trend;
+  BWE_TEST_LOGGING_PLOT(1, "T", now_ms, modified_trend);
+  BWE_TEST_LOGGING_PLOT(1, "threshold", now_ms, threshold_);
+  if (modified_trend > threshold_) {
+    if (time_over_using_ == -1) {
+      // Initialize the timer. Assume that we've been
+      // over-using half of the time since the previous
+      // sample.
+      time_over_using_ = ts_delta / 2;
+    } else {
+      // Increment timer
+      time_over_using_ += ts_delta;
+    }
+    overuse_counter_++;
+    if (time_over_using_ > overusing_time_threshold_ && overuse_counter_ > 1) {
+      if (trend >= prev_trend_) {
+        time_over_using_ = 0;
+        overuse_counter_ = 0;
+        hypothesis_ = BandwidthUsage::kBwOverusing;
+      }
+    }
+  } else if (modified_trend < -threshold_) {
+    time_over_using_ = -1;
+    overuse_counter_ = 0;
+    hypothesis_ = BandwidthUsage::kBwUnderusing;
+  } else {
+    time_over_using_ = -1;
+    overuse_counter_ = 0;
+    hypothesis_ = BandwidthUsage::kBwNormal;
+  }
+  prev_trend_ = trend;
+  UpdateThreshold(modified_trend, now_ms);
+}
+
+void TrendlineEstimator::UpdateThreshold(double modified_trend,
+                                         int64_t now_ms) {
+  if (last_update_ms_ == -1)
+    last_update_ms_ = now_ms;
+
+  if (fabs(modified_trend) > threshold_ + kMaxAdaptOffsetMs) {
+    // Avoid adapting the threshold to big latency spikes, caused e.g.,
+    // by a sudden capacity drop.
+    last_update_ms_ = now_ms;
+    return;
+  }
+
+  const double k = fabs(modified_trend) < threshold_ ? k_down_ : k_up_;
+  const int64_t kMaxTimeDeltaMs = 100;
+  int64_t time_delta_ms = std::min(now_ms - last_update_ms_, kMaxTimeDeltaMs);
+  threshold_ += k * (fabs(modified_trend) - threshold_) * time_delta_ms;
+  threshold_ = rtc::SafeClamp(threshold_, 6.f, 600.f);
+  last_update_ms_ = now_ms;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
new file mode 100644
index 0000000000..ffda25df74
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
@@ -0,0 +1,125 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+#include <utility>
+
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+struct TrendlineEstimatorSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-TrendlineEstimatorSettings";
+  static constexpr unsigned kDefaultTrendlineWindowSize = 20;
+
+  TrendlineEstimatorSettings() = delete;
+  explicit TrendlineEstimatorSettings(const FieldTrialsView* key_value_config);
+
+  // Sort the packets in the window. Should be redundant,
+  // but then almost no cost.
+  bool enable_sort = false;
+
+  // Cap the trendline slope based on the minimum delay seen
+  // in the beginning_packets and end_packets respectively.
+  bool enable_cap = false;
+  unsigned beginning_packets = 7;
+  unsigned end_packets = 7;
+  double cap_uncertainty = 0.0;
+
+  // Size (in packets) of the window.
+  unsigned window_size = kDefaultTrendlineWindowSize;
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class TrendlineEstimator : public DelayIncreaseDetectorInterface {
+ public:
+  TrendlineEstimator(const FieldTrialsView* key_value_config,
+                     NetworkStatePredictor* network_state_predictor);
+
+  ~TrendlineEstimator() override;
+
+  TrendlineEstimator(const TrendlineEstimator&) = delete;
+  TrendlineEstimator& operator=(const TrendlineEstimator&) = delete;
+
+  // Update the estimator with a new sample. The deltas should represent deltas
+  // between timestamp groups as defined by the InterArrival class.
+  void Update(double recv_delta_ms,
+              double send_delta_ms,
+              int64_t send_time_ms,
+              int64_t arrival_time_ms,
+              size_t packet_size,
+              bool calculated_deltas) override;
+
+  void UpdateTrendline(double recv_delta_ms,
+                       double send_delta_ms,
+                       int64_t send_time_ms,
+                       int64_t arrival_time_ms,
+                       size_t packet_size);
+
+  BandwidthUsage State() const override;
+
+  struct PacketTiming {
+    PacketTiming(double arrival_time_ms,
+                 double smoothed_delay_ms,
+                 double raw_delay_ms)
+        : arrival_time_ms(arrival_time_ms),
+          smoothed_delay_ms(smoothed_delay_ms),
+          raw_delay_ms(raw_delay_ms) {}
+    double arrival_time_ms;
+    double smoothed_delay_ms;
+    double raw_delay_ms;
+  };
+
+ private:
+  friend class GoogCcStatePrinter;
+  void Detect(double trend, double ts_delta, int64_t now_ms);
+
+  void UpdateThreshold(double modified_offset, int64_t now_ms);
+
+  // Parameters.
+  TrendlineEstimatorSettings settings_;
+  const double smoothing_coef_;
+  const double threshold_gain_;
+  // Used by the existing threshold.
+  int num_of_deltas_;
+  // Keep the arrival times small by using the change from the first packet.
+  int64_t first_arrival_time_ms_;
+  // Exponential backoff filtering.
+  double accumulated_delay_;
+  double smoothed_delay_;
+  // Linear least squares regression.
+  std::deque<PacketTiming> delay_hist_;
+
+  const double k_up_;
+  const double k_down_;
+  double overusing_time_threshold_;
+  double threshold_;
+  double prev_modified_trend_;
+  int64_t last_update_ms_;
+  double prev_trend_;
+  double time_over_using_;
+  int overuse_counter_;
+  BandwidthUsage hypothesis_;
+  BandwidthUsage hypothesis_predicted_;
+  NetworkStatePredictor* network_state_predictor_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc
new file mode 100644
index 0000000000..b0195abdf5
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
+
+#include <algorithm>
+#include <numeric>
+#include <vector>
+
+#include "api/transport/field_trial_based_config.h"
+#include "rtc_base/random.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+class PacketTimeGenerator {
+ public:
+  PacketTimeGenerator(int64_t initial_clock, double time_between_packets)
+      : initial_clock_(initial_clock),
+        time_between_packets_(time_between_packets),
+        packets_(0) {}
+  int64_t operator()() {
+    return initial_clock_ + time_between_packets_ * packets_++;
+  }
+
+ private:
+  const int64_t initial_clock_;
+  const double time_between_packets_;
+  size_t packets_;
+};
+
+class TrendlineEstimatorTest : public testing::Test {
+ public:
+  TrendlineEstimatorTest()
+      : send_times(kPacketCount),
+        recv_times(kPacketCount),
+        packet_sizes(kPacketCount),
+        config(),
+        estimator(&config, nullptr),
+        count(1) {
+    std::fill(packet_sizes.begin(), packet_sizes.end(), kPacketSizeBytes);
+  }
+
+  void RunTestUntilStateChange() {
+    RTC_DCHECK_EQ(send_times.size(), kPacketCount);
+    RTC_DCHECK_EQ(recv_times.size(), kPacketCount);
+    RTC_DCHECK_EQ(packet_sizes.size(), kPacketCount);
+    RTC_DCHECK_GE(count, 1);
+    RTC_DCHECK_LT(count, kPacketCount);
+
+    auto initial_state = estimator.State();
+    for (; count < kPacketCount; count++) {
+      double recv_delta = recv_times[count] - recv_times[count - 1];
+      double send_delta = send_times[count] - send_times[count - 1];
+      estimator.Update(recv_delta, send_delta, send_times[count],
+                       recv_times[count], packet_sizes[count], true);
+      if (estimator.State() != initial_state) {
+        return;
+      }
+    }
+  }
+
+ protected:
+  const size_t kPacketCount = 25;
+  const size_t kPacketSizeBytes = 1200;
+  std::vector<int64_t> send_times;
+  std::vector<int64_t> recv_times;
+  std::vector<size_t> packet_sizes;
+  const FieldTrialBasedConfig config;
+  TrendlineEstimator estimator;
+  size_t count;
+};
+}  // namespace
+
+TEST_F(TrendlineEstimatorTest, Normal) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          20 /*delivered at the same pace*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+TEST_F(TrendlineEstimatorTest, Overusing) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          1.1 * 20 /*10% slower delivery*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+TEST_F(TrendlineEstimatorTest, Underusing) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          0.85 * 20 /*15% faster delivery*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwUnderusing);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwUnderusing);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+TEST_F(TrendlineEstimatorTest, IncludesSmallPacketsByDefault) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          1.1 * 20 /*10% slower delivery*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  std::fill(packet_sizes.begin(), packet_sizes.end(), 100);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/include/receive_side_congestion_controller.h b/third_party/libwebrtc/modules/congestion_controller/include/receive_side_congestion_controller.h
new file mode 100644
index 0000000000..7696396016
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/include/receive_side_congestion_controller.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/transport/network_control.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "modules/congestion_controller/remb_throttler.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+class RemoteBitrateEstimator;
+
+// This class represents the congestion control state for receive
+// streams. For send side bandwidth estimation, this is simply
+// relaying for each received RTP packet back to the sender. While for
+// receive side bandwidth estimation, we do the estimation locally and
+// send our results back to the sender.
+class ReceiveSideCongestionController : public CallStatsObserver {
+ public:
+  ReceiveSideCongestionController(
+      Clock* clock,
+      RemoteEstimatorProxy::TransportFeedbackSender feedback_sender,
+      RembThrottler::RembSender remb_sender,
+      NetworkStateEstimator* network_state_estimator);
+
+  ~ReceiveSideCongestionController() override {}
+
+  virtual void OnReceivedPacket(int64_t arrival_time_ms,
+                                size_t payload_size,
+                                const RTPHeader& header);
+
+  void SetSendPeriodicFeedback(bool send_periodic_feedback);
+
+  // Implements CallStatsObserver.
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+
+  // This is send bitrate, used to control the rate of feedback messages.
+  void OnBitrateChanged(int bitrate_bps);
+
+  // Ensures the remote party is notified of the receive bitrate no larger than
+  // `bitrate` using RTCP REMB.
+  void SetMaxDesiredReceiveBitrate(DataRate bitrate);
+
+  void SetTransportOverhead(DataSize overhead_per_packet);
+
+  // Returns latest receive side bandwidth estimation.
+  // Returns zero if receive side bandwidth estimation is unavailable.
+  DataRate LatestReceiveSideEstimate() const;
+
+  // Removes stream from receive side bandwidth estimation.
+  // Noop if receive side bwe is not used or stream doesn't participate in it.
+  void RemoveStream(uint32_t ssrc);
+
+  // Runs periodic tasks if it is time to run them, returns time until next
+  // call to `MaybeProcess` should be non idle.
+  TimeDelta MaybeProcess();
+
+ private:
+  void PickEstimatorFromHeader(const RTPHeader& header)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  void PickEstimator() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock& clock_;
+  RembThrottler remb_throttler_;
+  RemoteEstimatorProxy remote_estimator_proxy_;
+
+  mutable Mutex mutex_;
+  std::unique_ptr<RemoteBitrateEstimator> rbe_ RTC_GUARDED_BY(mutex_);
+  bool using_absolute_send_time_ RTC_GUARDED_BY(mutex_);
+  uint32_t packets_since_absolute_send_time_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/BUILD.gn b/third_party/libwebrtc/modules/congestion_controller/pcc/BUILD.gn
new file mode 100644
index 0000000000..85b12b3771
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/BUILD.gn
@@ -0,0 +1,123 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("pcc") {
+  sources = [
+    "pcc_factory.cc",
+    "pcc_factory.h",
+  ]
+  deps = [
+    ":pcc_controller",
+    "../../../api/transport:network_control",
+    "../../../api/units:time_delta",
+  ]
+}
+
+rtc_library("pcc_controller") {
+  sources = [
+    "pcc_network_controller.cc",
+    "pcc_network_controller.h",
+  ]
+  deps = [
+    ":bitrate_controller",
+    ":monitor_interval",
+    ":rtt_tracker",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:random",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("monitor_interval") {
+  sources = [
+    "monitor_interval.cc",
+    "monitor_interval.h",
+  ]
+  deps = [
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:logging",
+  ]
+}
+
+rtc_library("rtt_tracker") {
+  sources = [
+    "rtt_tracker.cc",
+    "rtt_tracker.h",
+  ]
+  deps = [
+    "../../../api/transport:network_control",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+  ]
+}
+
+rtc_library("utility_function") {
+  sources = [
+    "utility_function.cc",
+    "utility_function.h",
+  ]
+  deps = [
+    ":monitor_interval",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../rtc_base:checks",
+  ]
+}
+
+rtc_library("bitrate_controller") {
+  sources = [
+    "bitrate_controller.cc",
+    "bitrate_controller.h",
+  ]
+  deps = [
+    ":monitor_interval",
+    ":utility_function",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+if (rtc_include_tests && !build_with_chromium) {
+  rtc_library("pcc_unittests") {
+    testonly = true
+    sources = [
+      "bitrate_controller_unittest.cc",
+      "monitor_interval_unittest.cc",
+      "pcc_network_controller_unittest.cc",
+      "rtt_tracker_unittest.cc",
+      "utility_function_unittest.cc",
+    ]
+    deps = [
+      ":bitrate_controller",
+      ":monitor_interval",
+      ":pcc",
+      ":pcc_controller",
+      ":rtt_tracker",
+      ":utility_function",
+      "../../../api/transport:network_control",
+      "../../../api/units:data_rate",
+      "../../../api/units:data_size",
+      "../../../api/units:time_delta",
+      "../../../api/units:timestamp",
+      "../../../test:test_support",
+      "../../../test/scenario",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.cc
new file mode 100644
index 0000000000..16b8e6966f
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.cc
@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/bitrate_controller.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstdlib>
+#include <memory>
+#include <utility>
+#include <vector>
+
+
+namespace webrtc {
+namespace pcc {
+
+PccBitrateController::PccBitrateController(double initial_conversion_factor,
+                                           double initial_dynamic_boundary,
+                                           double dynamic_boundary_increment,
+                                           double rtt_gradient_coefficient,
+                                           double loss_coefficient,
+                                           double throughput_coefficient,
+                                           double throughput_power,
+                                           double rtt_gradient_threshold,
+                                           double delay_gradient_negative_bound)
+    : PccBitrateController(initial_conversion_factor,
+                           initial_dynamic_boundary,
+                           dynamic_boundary_increment,
+                           std::make_unique<ModifiedVivaceUtilityFunction>(
+                               rtt_gradient_coefficient,
+                               loss_coefficient,
+                               throughput_coefficient,
+                               throughput_power,
+                               rtt_gradient_threshold,
+                               delay_gradient_negative_bound)) {}
+
+PccBitrateController::PccBitrateController(
+    double initial_conversion_factor,
+    double initial_dynamic_boundary,
+    double dynamic_boundary_increment,
+    std::unique_ptr<PccUtilityFunctionInterface> utility_function)
+    : consecutive_boundary_adjustments_number_(0),
+      initial_dynamic_boundary_(initial_dynamic_boundary),
+      dynamic_boundary_increment_(dynamic_boundary_increment),
+      utility_function_(std::move(utility_function)),
+      step_size_adjustments_number_(0),
+      initial_conversion_factor_(initial_conversion_factor) {}
+
+PccBitrateController::~PccBitrateController() = default;
+
+double PccBitrateController::ComputeStepSize(double utility_gradient) {
+  // Computes number of consecutive step size adjustments.
+  if (utility_gradient > 0) {
+    step_size_adjustments_number_ =
+        std::max<int64_t>(step_size_adjustments_number_ + 1, 1);
+  } else if (utility_gradient < 0) {
+    step_size_adjustments_number_ =
+        std::min<int64_t>(step_size_adjustments_number_ - 1, -1);
+  } else {
+    step_size_adjustments_number_ = 0;
+  }
+  // Computes step size amplifier.
+  int64_t step_size_amplifier = 1;
+  if (std::abs(step_size_adjustments_number_) <= 3) {
+    step_size_amplifier =
+        std::max<int64_t>(std::abs(step_size_adjustments_number_), 1);
+  } else {
+    step_size_amplifier = 2 * std::abs(step_size_adjustments_number_) - 3;
+  }
+  return step_size_amplifier * initial_conversion_factor_;
+}
+
+double PccBitrateController::ApplyDynamicBoundary(double rate_change,
+                                                  double bitrate) {
+  double rate_change_abs = std::abs(rate_change);
+  int64_t rate_change_sign = (rate_change > 0) ? 1 : -1;
+  if (consecutive_boundary_adjustments_number_ * rate_change_sign < 0) {
+    consecutive_boundary_adjustments_number_ = 0;
+  }
+  double dynamic_change_boundary =
+      initial_dynamic_boundary_ +
+      std::abs(consecutive_boundary_adjustments_number_) *
+          dynamic_boundary_increment_;
+  double boundary = bitrate * dynamic_change_boundary;
+  if (rate_change_abs > boundary) {
+    consecutive_boundary_adjustments_number_ += rate_change_sign;
+    return boundary * rate_change_sign;
+  }
+  // Rate change smaller than boundary. Reset boundary to the smallest possible
+  // that would allow the change.
+  while (rate_change_abs <= boundary &&
+         consecutive_boundary_adjustments_number_ * rate_change_sign > 0) {
+    consecutive_boundary_adjustments_number_ -= rate_change_sign;
+    dynamic_change_boundary =
+        initial_dynamic_boundary_ +
+        std::abs(consecutive_boundary_adjustments_number_) *
+            dynamic_boundary_increment_;
+    boundary = bitrate * dynamic_change_boundary;
+  }
+  consecutive_boundary_adjustments_number_ += rate_change_sign;
+  return rate_change;
+}
+
+absl::optional<DataRate>
+PccBitrateController::ComputeRateUpdateForSlowStartMode(
+    const PccMonitorInterval& monitor_interval) {
+  double utility_value = utility_function_->Compute(monitor_interval);
+  if (previous_utility_.has_value() && utility_value <= previous_utility_) {
+    return absl::nullopt;
+  }
+  previous_utility_ = utility_value;
+  return monitor_interval.GetTargetSendingRate();
+}
+
+DataRate PccBitrateController::ComputeRateUpdateForOnlineLearningMode(
+    const std::vector<PccMonitorInterval>& intervals,
+    DataRate bandwith_estimate) {
+  double first_utility = utility_function_->Compute(intervals[0]);
+  double second_utility = utility_function_->Compute(intervals[1]);
+  double first_bitrate_bps = intervals[0].GetTargetSendingRate().bps();
+  double second_bitrate_bps = intervals[1].GetTargetSendingRate().bps();
+  double gradient = (first_utility - second_utility) /
+                    (first_bitrate_bps - second_bitrate_bps);
+  double rate_change_bps = gradient * ComputeStepSize(gradient);  // delta_r
+  rate_change_bps =
+      ApplyDynamicBoundary(rate_change_bps, bandwith_estimate.bps());
+  return DataRate::BitsPerSec(
+      std::max(0.0, bandwith_estimate.bps() + rate_change_bps));
+}
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.h b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.h
new file mode 100644
index 0000000000..fadeea1b55
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_BITRATE_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_BITRATE_CONTROLLER_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+#include "modules/congestion_controller/pcc/utility_function.h"
+
+namespace webrtc {
+namespace pcc {
+
+class PccBitrateController {
+ public:
+  PccBitrateController(double initial_conversion_factor,
+                       double initial_dynamic_boundary,
+                       double dynamic_boundary_increment,
+                       double rtt_gradient_coefficient,
+                       double loss_coefficient,
+                       double throughput_coefficient,
+                       double throughput_power,
+                       double rtt_gradient_threshold,
+                       double delay_gradient_negative_bound);
+
+  PccBitrateController(
+      double initial_conversion_factor,
+      double initial_dynamic_boundary,
+      double dynamic_boundary_increment,
+      std::unique_ptr<PccUtilityFunctionInterface> utility_function);
+
+  absl::optional<DataRate> ComputeRateUpdateForSlowStartMode(
+      const PccMonitorInterval& monitor_interval);
+
+  DataRate ComputeRateUpdateForOnlineLearningMode(
+      const std::vector<PccMonitorInterval>& block,
+      DataRate bandwidth_estimate);
+
+  ~PccBitrateController();
+
+ private:
+  double ApplyDynamicBoundary(double rate_change, double bitrate);
+  double ComputeStepSize(double utility_gradient);
+
+  // Dynamic boundary variables:
+  int64_t consecutive_boundary_adjustments_number_;
+  const double initial_dynamic_boundary_;
+  const double dynamic_boundary_increment_;
+
+  const std::unique_ptr<PccUtilityFunctionInterface> utility_function_;
+  // Step Size variables:
+  int64_t step_size_adjustments_number_;
+  const double initial_conversion_factor_;
+
+  absl::optional<double> previous_utility_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_BITRATE_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller_unittest.cc
new file mode 100644
index 0000000000..957d99b1de
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller_unittest.cc
@@ -0,0 +1,303 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/bitrate_controller.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace pcc {
+namespace test {
+namespace {
+constexpr double kInitialConversionFactor = 1;
+constexpr double kInitialDynamicBoundary = 0.05;
+constexpr double kDynamicBoundaryIncrement = 0.1;
+
+constexpr double kDelayGradientCoefficient = 900;
+constexpr double kLossCoefficient = 11.35;
+constexpr double kThroughputCoefficient = 500 * 1000;
+constexpr double kThroughputPower = 0.99;
+constexpr double kDelayGradientThreshold = 0.01;
+constexpr double kDelayGradientNegativeBound = 10;
+
+const DataRate kTargetSendingRate = DataRate::KilobitsPerSec(300);
+const double kEpsilon = 0.05;
+const Timestamp kStartTime = Timestamp::Micros(0);
+const TimeDelta kPacketsDelta = TimeDelta::Millis(1);
+const TimeDelta kIntervalDuration = TimeDelta::Millis(1000);
+const TimeDelta kDefaultRtt = TimeDelta::Millis(1000);
+const DataSize kDefaultDataSize = DataSize::Bytes(100);
+
+std::vector<PacketResult> CreatePacketResults(
+    const std::vector<Timestamp>& packets_send_times,
+    const std::vector<Timestamp>& packets_received_times = {},
+    const std::vector<DataSize>& packets_sizes = {}) {
+  std::vector<PacketResult> packet_results;
+  PacketResult packet_result;
+  SentPacket sent_packet;
+  for (size_t i = 0; i < packets_send_times.size(); ++i) {
+    sent_packet.send_time = packets_send_times[i];
+    if (packets_sizes.empty()) {
+      sent_packet.size = kDefaultDataSize;
+    } else {
+      sent_packet.size = packets_sizes[i];
+    }
+    packet_result.sent_packet = sent_packet;
+    if (packets_received_times.empty()) {
+      packet_result.receive_time = packets_send_times[i] + kDefaultRtt;
+    } else {
+      packet_result.receive_time = packets_received_times[i];
+    }
+    packet_results.push_back(packet_result);
+  }
+  return packet_results;
+}
+
+class MockUtilityFunction : public PccUtilityFunctionInterface {
+ public:
+  MOCK_METHOD(double,
+              Compute,
+              (const PccMonitorInterval& monitor_interval),
+              (const, override));
+};
+
+}  // namespace
+
+TEST(PccBitrateControllerTest, IncreaseRateWhenNoChangesForTestBitrates) {
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, kDelayGradientCoefficient, kLossCoefficient,
+      kThroughputCoefficient, kThroughputPower, kDelayGradientThreshold,
+      kDelayGradientNegativeBound);
+  VivaceUtilityFunction utility_function(
+      kDelayGradientCoefficient, kLossCoefficient, kThroughputCoefficient,
+      kThroughputPower, kDelayGradientThreshold, kDelayGradientNegativeBound);
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + kPacketsDelta,
+                           kStartTime + kIntervalDuration + kPacketsDelta,
+                           kStartTime + 3 * kIntervalDuration},
+                          {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + kPacketsDelta,
+                           kStartTime + kIntervalDuration + kPacketsDelta,
+                           kStartTime + 3 * kIntervalDuration},
+                          {}, {}));
+  // For both of the monitor intervals there were no change in rtt gradient
+  // and in packet loss. Since the only difference is in the sending rate,
+  // the higher sending rate should be chosen by congestion controller.
+  EXPECT_GT(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps());
+}
+
+TEST(PccBitrateControllerTest, NoChangesWhenUtilityFunctionDoesntChange) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(2)
+      .WillOnce(::testing::Return(100))
+      .WillOnce(::testing::Return(100));
+
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  // Because we don't have any packets inside of monitor intervals, utility
+  // function should be zero for both of them and the sending rate should not
+  // change.
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps());
+}
+
+TEST(PccBitrateControllerTest, NoBoundaryWhenSmallGradient) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  constexpr double kFirstMonitorIntervalUtility = 0;
+  const double kSecondMonitorIntervalUtility =
+      2 * kTargetSendingRate.bps() * kEpsilon;
+
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(2)
+      .WillOnce(::testing::Return(kFirstMonitorIntervalUtility))
+      .WillOnce(::testing::Return(kSecondMonitorIntervalUtility));
+
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+
+  double gradient =
+      (kFirstMonitorIntervalUtility - kSecondMonitorIntervalUtility) /
+      (kTargetSendingRate.bps() * 2 * kEpsilon);
+  // When the gradient is small we don't hit the dynamic boundary.
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps() + kInitialConversionFactor * gradient);
+}
+
+TEST(PccBitrateControllerTest, FaceBoundaryWhenLargeGradient) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  constexpr double kFirstMonitorIntervalUtility = 0;
+  const double kSecondMonitorIntervalUtility =
+      10 * kInitialDynamicBoundary * kTargetSendingRate.bps() * 2 *
+      kTargetSendingRate.bps() * kEpsilon;
+
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(4)
+      .WillOnce(::testing::Return(kFirstMonitorIntervalUtility))
+      .WillOnce(::testing::Return(kSecondMonitorIntervalUtility))
+      .WillOnce(::testing::Return(kFirstMonitorIntervalUtility))
+      .WillOnce(::testing::Return(kSecondMonitorIntervalUtility));
+
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  // The utility function gradient is too big and we hit the dynamic boundary.
+  EXPECT_EQ(bitrate_controller.ComputeRateUpdateForOnlineLearningMode(
+                monitor_block, kTargetSendingRate),
+            kTargetSendingRate * (1 - kInitialDynamicBoundary));
+  // For the second time we hit the dynamic boundary in the same direction, the
+  // boundary should increase.
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps() *
+                (1 - kInitialDynamicBoundary - kDynamicBoundaryIncrement));
+}
+
+TEST(PccBitrateControllerTest, SlowStartMode) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  constexpr double kFirstUtilityFunction = 1000;
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(4)
+      // For first 3 calls we expect to stay in the SLOW_START mode and double
+      // the sending rate since the utility function increases its value. For
+      // the last call utility function decreases its value, this means that
+      // we should not double the sending rate and exit SLOW_START mode.
+      .WillOnce(::testing::Return(kFirstUtilityFunction))
+      .WillOnce(::testing::Return(kFirstUtilityFunction + 1))
+      .WillOnce(::testing::Return(kFirstUtilityFunction + 2))
+      .WillOnce(::testing::Return(kFirstUtilityFunction + 1));
+
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  std::vector<PccMonitorInterval> monitor_block{PccMonitorInterval(
+      2 * kTargetSendingRate, kStartTime, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  EXPECT_EQ(
+      bitrate_controller.ComputeRateUpdateForSlowStartMode(monitor_block[0]),
+      kTargetSendingRate * 2);
+  EXPECT_EQ(
+      bitrate_controller.ComputeRateUpdateForSlowStartMode(monitor_block[0]),
+      kTargetSendingRate * 2);
+  EXPECT_EQ(
+      bitrate_controller.ComputeRateUpdateForSlowStartMode(monitor_block[0]),
+      kTargetSendingRate * 2);
+  EXPECT_EQ(
+      bitrate_controller.ComputeRateUpdateForSlowStartMode(monitor_block[0]),
+      absl::nullopt);
+}
+
+TEST(PccBitrateControllerTest, StepSizeIncrease) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  constexpr double kFirstMiUtilityFunction = 0;
+  const double kSecondMiUtilityFunction =
+      2 * kTargetSendingRate.bps() * kEpsilon;
+
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(4)
+      .WillOnce(::testing::Return(kFirstMiUtilityFunction))
+      .WillOnce(::testing::Return(kSecondMiUtilityFunction))
+      .WillOnce(::testing::Return(kFirstMiUtilityFunction))
+      .WillOnce(::testing::Return(kSecondMiUtilityFunction));
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+
+  double gradient = (kFirstMiUtilityFunction - kSecondMiUtilityFunction) /
+                    (kTargetSendingRate.bps() * 2 * kEpsilon);
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  // If we are moving in the same direction - the step size should increase.
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps() + kInitialConversionFactor * gradient);
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps() + 2 * kInitialConversionFactor * gradient);
+}
+
+}  // namespace test
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.cc
new file mode 100644
index 0000000000..de1e2d5e69
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+
+#include <stddef.h>
+
+#include <cmath>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace pcc {
+
+PccMonitorInterval::PccMonitorInterval(DataRate target_sending_rate,
+                                       Timestamp start_time,
+                                       TimeDelta duration)
+    : target_sending_rate_(target_sending_rate),
+      start_time_(start_time),
+      interval_duration_(duration),
+      received_packets_size_(DataSize::Zero()),
+      feedback_collection_done_(false) {}
+
+PccMonitorInterval::~PccMonitorInterval() = default;
+
+PccMonitorInterval::PccMonitorInterval(const PccMonitorInterval& other) =
+    default;
+
+void PccMonitorInterval::OnPacketsFeedback(
+    const std::vector<PacketResult>& packets_results) {
+  for (const PacketResult& packet_result : packets_results) {
+    if (packet_result.sent_packet.send_time <= start_time_) {
+      continue;
+    }
+    // Here we assume that if some packets are reordered with packets sent
+    // after the end of the monitor interval, then they are lost. (Otherwise
+    // it is not clear how long should we wait for packets feedback to arrive).
+    if (packet_result.sent_packet.send_time >
+        start_time_ + interval_duration_) {
+      feedback_collection_done_ = true;
+      return;
+    }
+    if (!packet_result.IsReceived()) {
+      lost_packets_sent_time_.push_back(packet_result.sent_packet.send_time);
+    } else {
+      received_packets_.push_back(
+          {packet_result.receive_time - packet_result.sent_packet.send_time,
+           packet_result.sent_packet.send_time});
+      received_packets_size_ += packet_result.sent_packet.size;
+    }
+  }
+}
+
+// For the formula used in computations see formula for "slope" in the second
+// method:
+// https://www.johndcook.com/blog/2008/10/20/comparing-two-ways-to-fit-a-line-to-data/
+double PccMonitorInterval::ComputeDelayGradient(
+    double delay_gradient_threshold) const {
+  // Early return to prevent division by 0 in case all packets are sent at the
+  // same time.
+  if (received_packets_.empty() || received_packets_.front().sent_time ==
+                                       received_packets_.back().sent_time) {
+    return 0;
+  }
+  double sum_times = 0;
+  for (const ReceivedPacket& packet : received_packets_) {
+    double time_delta_us =
+        (packet.sent_time - received_packets_[0].sent_time).us();
+    sum_times += time_delta_us;
+  }
+  double sum_squared_scaled_time_deltas = 0;
+  double sum_scaled_time_delta_dot_delay = 0;
+  for (const ReceivedPacket& packet : received_packets_) {
+    double time_delta_us =
+        (packet.sent_time - received_packets_[0].sent_time).us();
+    double delay = packet.delay.us();
+    double scaled_time_delta_us =
+        time_delta_us - sum_times / received_packets_.size();
+    sum_squared_scaled_time_deltas +=
+        scaled_time_delta_us * scaled_time_delta_us;
+    sum_scaled_time_delta_dot_delay += scaled_time_delta_us * delay;
+  }
+  double rtt_gradient =
+      sum_scaled_time_delta_dot_delay / sum_squared_scaled_time_deltas;
+  if (std::abs(rtt_gradient) < delay_gradient_threshold)
+    rtt_gradient = 0;
+  return rtt_gradient;
+}
+
+bool PccMonitorInterval::IsFeedbackCollectionDone() const {
+  return feedback_collection_done_;
+}
+
+Timestamp PccMonitorInterval::GetEndTime() const {
+  return start_time_ + interval_duration_;
+}
+
+double PccMonitorInterval::GetLossRate() const {
+  size_t packets_lost = lost_packets_sent_time_.size();
+  size_t packets_received = received_packets_.size();
+  if (packets_lost == 0)
+    return 0;
+  return static_cast<double>(packets_lost) / (packets_lost + packets_received);
+}
+
+DataRate PccMonitorInterval::GetTargetSendingRate() const {
+  return target_sending_rate_;
+}
+
+DataRate PccMonitorInterval::GetTransmittedPacketsRate() const {
+  if (received_packets_.empty()) {
+    return target_sending_rate_;
+  }
+  Timestamp receive_time_of_first_packet =
+      received_packets_.front().sent_time + received_packets_.front().delay;
+  Timestamp receive_time_of_last_packet =
+      received_packets_.back().sent_time + received_packets_.back().delay;
+  if (receive_time_of_first_packet == receive_time_of_last_packet) {
+    RTC_LOG(LS_WARNING)
+        << "All packets in monitor interval were received at the same time.";
+    return target_sending_rate_;
+  }
+  return received_packets_size_ /
+         (receive_time_of_last_packet - receive_time_of_first_packet);
+}
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.h b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.h
new file mode 100644
index 0000000000..51bd0f068a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_MONITOR_INTERVAL_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_MONITOR_INTERVAL_H_
+
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+namespace pcc {
+
+// PCC divides time into consecutive monitor intervals which are used to test
+// consequences for performance of sending at a certain rate.
+class PccMonitorInterval {
+ public:
+  PccMonitorInterval(DataRate target_sending_rate,
+                     Timestamp start_time,
+                     TimeDelta duration);
+  ~PccMonitorInterval();
+  PccMonitorInterval(const PccMonitorInterval& other);
+  void OnPacketsFeedback(const std::vector<PacketResult>& packets_results);
+  // Returns true if got complete information about packets.
+  // Notice, this only happens when received feedback about the first packet
+  // which were sent after the end of the monitor interval. If such event
+  // doesn't occur, we don't mind anyway and stay in the same state.
+  bool IsFeedbackCollectionDone() const;
+  Timestamp GetEndTime() const;
+
+  double GetLossRate() const;
+  // Estimates the gradient using linear regression on the 2-dimensional
+  // dataset (sampled packets delay, time of sampling).
+  double ComputeDelayGradient(double delay_gradient_threshold) const;
+  DataRate GetTargetSendingRate() const;
+  // How fast receiving side gets packets.
+  DataRate GetTransmittedPacketsRate() const;
+
+ private:
+  struct ReceivedPacket {
+    TimeDelta delay;
+    Timestamp sent_time;
+  };
+  // Target bitrate used to generate and pace the outgoing packets.
+  // Actually sent bitrate might not match the target exactly.
+  DataRate target_sending_rate_;
+  // Start time is not included into interval while end time is included.
+  Timestamp start_time_;
+  TimeDelta interval_duration_;
+  // Vectors below updates while receiving feedback.
+  std::vector<ReceivedPacket> received_packets_;
+  std::vector<Timestamp> lost_packets_sent_time_;
+  DataSize received_packets_size_;
+  bool feedback_collection_done_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_MONITOR_INTERVAL_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval_unittest.cc
new file mode 100644
index 0000000000..aaff57bd2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval_unittest.cc
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+
+#include <stddef.h>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace pcc {
+namespace test {
+namespace {
+const DataRate kTargetSendingRate = DataRate::KilobitsPerSec(300);
+const Timestamp kStartTime = Timestamp::Micros(0);
+const TimeDelta kPacketsDelta = TimeDelta::Millis(1);
+const TimeDelta kIntervalDuration = TimeDelta::Millis(100);
+const TimeDelta kDefaultDelay = TimeDelta::Millis(100);
+const DataSize kDefaultPacketSize = DataSize::Bytes(100);
+constexpr double kDelayGradientThreshold = 0.01;
+
+std::vector<PacketResult> CreatePacketResults(
+    const std::vector<Timestamp>& packets_send_times,
+    const std::vector<Timestamp>& packets_received_times = {},
+    const std::vector<DataSize>& packets_sizes = {}) {
+  std::vector<PacketResult> packet_results;
+  for (size_t i = 0; i < packets_send_times.size(); ++i) {
+    SentPacket sent_packet;
+    sent_packet.send_time = packets_send_times[i];
+    if (packets_sizes.empty()) {
+      sent_packet.size = kDefaultPacketSize;
+    } else {
+      sent_packet.size = packets_sizes[i];
+    }
+    PacketResult packet_result;
+    packet_result.sent_packet = sent_packet;
+    if (packets_received_times.empty()) {
+      packet_result.receive_time = packets_send_times[i] + kDefaultDelay;
+    } else {
+      packet_result.receive_time = packets_received_times[i];
+    }
+    packet_results.push_back(packet_result);
+  }
+  return packet_results;
+}
+
+}  // namespace
+
+TEST(PccMonitorIntervalTest, InitialValuesAreEqualToOnesSetInConstructor) {
+  PccMonitorInterval interval{kTargetSendingRate, kStartTime,
+                              kIntervalDuration};
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), false);
+  EXPECT_EQ(interval.GetEndTime(), kStartTime + kIntervalDuration);
+  EXPECT_EQ(interval.GetTargetSendingRate(), kTargetSendingRate);
+}
+
+TEST(PccMonitorIntervalTest, IndicatesDoneWhenFeedbackReceivedAfterInterval) {
+  PccMonitorInterval interval{kTargetSendingRate, kStartTime,
+                              kIntervalDuration};
+  interval.OnPacketsFeedback(CreatePacketResults({kStartTime}));
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), false);
+  interval.OnPacketsFeedback(
+      CreatePacketResults({kStartTime, kStartTime + kIntervalDuration}));
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), false);
+  interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kIntervalDuration, kStartTime + 2 * kIntervalDuration}));
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), true);
+}
+
+TEST(PccMonitorIntervalTest, LossRateIsOneThirdIfLostOnePacketOutOfThree) {
+  PccMonitorInterval interval{kTargetSendingRate, kStartTime,
+                              kIntervalDuration};
+  std::vector<Timestamp> start_times = {
+      kStartTime, kStartTime + 0.1 * kIntervalDuration,
+      kStartTime + 0.5 * kIntervalDuration, kStartTime + kIntervalDuration,
+      kStartTime + 2 * kIntervalDuration};
+  std::vector<Timestamp> end_times = {
+      kStartTime + 2 * kIntervalDuration, kStartTime + 2 * kIntervalDuration,
+      Timestamp::PlusInfinity(), kStartTime + 2 * kIntervalDuration,
+      kStartTime + 4 * kIntervalDuration};
+  std::vector<DataSize> packet_sizes = {
+      kDefaultPacketSize, 2 * kDefaultPacketSize, 3 * kDefaultPacketSize,
+      4 * kDefaultPacketSize, 5 * kDefaultPacketSize};
+  std::vector<PacketResult> packet_results =
+      CreatePacketResults(start_times, end_times, packet_sizes);
+  interval.OnPacketsFeedback(packet_results);
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), true);
+
+  EXPECT_DOUBLE_EQ(interval.GetLossRate(), 1. / 3);
+}
+
+TEST(PccMonitorIntervalTest, DelayGradientIsZeroIfNoChangeInPacketDelay) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 3 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay + 2 * kPacketsDelta,
+       Timestamp::PlusInfinity()},
+      {}));
+  // Delay gradient should be zero, because both received packets have the
+  // same one way delay.
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 0);
+}
+
+TEST(PccMonitorIntervalTest,
+     DelayGradientIsZeroWhenOnePacketSentInMonitorInterval) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, kStartTime + 3 * kIntervalDuration}, {}));
+  // Only one received packet belongs to the monitor_interval, delay gradient
+  // should be zero in this case.
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 0);
+}
+
+TEST(PccMonitorIntervalTest, DelayGradientIsOne) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 3 * kPacketsDelta, kStartTime + 3 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + 4 * kPacketsDelta + kDefaultDelay,
+       kStartTime + 3 * kIntervalDuration},
+      {}));
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 1);
+}
+
+TEST(PccMonitorIntervalTest, DelayGradientIsMinusOne) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 5 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay, kStartTime + 3 * kIntervalDuration},
+      {}));
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), -1);
+}
+
+TEST(PccMonitorIntervalTest,
+     DelayGradientIsZeroIfItSmallerWhenGradientThreshold) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + kPacketsDelta,
+       kStartTime + 102 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay + kPacketsDelta,
+       kStartTime + 3 * kIntervalDuration},
+      {}));
+  // Delay gradient is less than 0.01 hence should be treated as zero.
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 0);
+}
+
+TEST(PccMonitorIntervalTest,
+     DelayGradientIsZeroWhenAllPacketsSentAtTheSameTime) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + kPacketsDelta,
+       kStartTime + kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay + kPacketsDelta,
+       kStartTime + 3 * kIntervalDuration},
+      {}));
+  // If all packets were sent at the same time, then delay gradient should be
+  // zero.
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 0);
+}
+
+}  // namespace test
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.cc
new file mode 100644
index 0000000000..c35c6e8ab2
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/pcc_factory.h"
+
+#include <memory>
+
+#include "modules/congestion_controller/pcc/pcc_network_controller.h"
+
+namespace webrtc {
+
+PccNetworkControllerFactory::PccNetworkControllerFactory() {}
+
+std::unique_ptr<NetworkControllerInterface> PccNetworkControllerFactory::Create(
+    NetworkControllerConfig config) {
+  return std::make_unique<pcc::PccNetworkController>(config);
+}
+
+TimeDelta PccNetworkControllerFactory::GetProcessInterval() const {
+  return TimeDelta::PlusInfinity();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.h b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.h
new file mode 100644
index 0000000000..bb70d7a499
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_PCC_FACTORY_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_PCC_FACTORY_H_
+
+#include <memory>
+
+#include "api/transport/network_control.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+class PccNetworkControllerFactory : public NetworkControllerFactoryInterface {
+ public:
+  PccNetworkControllerFactory();
+  std::unique_ptr<NetworkControllerInterface> Create(
+      NetworkControllerConfig config) override;
+  TimeDelta GetProcessInterval() const override;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_PCC_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.cc
new file mode 100644
index 0000000000..8653470955
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.cc
@@ -0,0 +1,391 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/pcc_network_controller.h"
+
+#include <algorithm>
+
+#include "absl/types/optional.h"
+#include "api/units/data_size.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace pcc {
+namespace {
+constexpr int64_t kInitialRttMs = 200;
+constexpr int64_t kInitialBandwidthKbps = 300;
+constexpr double kMonitorIntervalDurationRatio = 1;
+constexpr double kDefaultSamplingStep = 0.05;
+constexpr double kTimeoutRatio = 2;
+constexpr double kAlphaForRtt = 0.9;
+constexpr double kSlowStartModeIncrease = 1.5;
+
+constexpr double kAlphaForPacketInterval = 0.9;
+constexpr int64_t kMinPacketsNumberPerInterval = 20;
+const TimeDelta kMinDurationOfMonitorInterval = TimeDelta::Millis(50);
+const TimeDelta kStartupDuration = TimeDelta::Millis(500);
+constexpr double kMinRateChangeBps = 4000;
+constexpr DataRate kMinRateHaveMultiplicativeRateChange = DataRate::BitsPerSec(
+    static_cast<int64_t>(kMinRateChangeBps / kDefaultSamplingStep));
+
+// Bitrate controller constants.
+constexpr double kInitialConversionFactor = 5;
+constexpr double kInitialDynamicBoundary = 0.1;
+constexpr double kDynamicBoundaryIncrement = 0.1;
+// Utility function parameters.
+constexpr double kRttGradientCoefficientBps = 0.005;
+constexpr double kLossCoefficientBps = 10;
+constexpr double kThroughputCoefficient = 0.001;
+constexpr double kThroughputPower = 0.9;
+constexpr double kRttGradientThreshold = 0.01;
+constexpr double kDelayGradientNegativeBound = 0.1;
+
+constexpr int64_t kNumberOfPacketsToKeep = 20;
+const uint64_t kRandomSeed = 100;
+}  // namespace
+
+PccNetworkController::PccNetworkController(NetworkControllerConfig config)
+    : start_time_(Timestamp::PlusInfinity()),
+      last_sent_packet_time_(Timestamp::PlusInfinity()),
+      smoothed_packets_sending_interval_(TimeDelta::Zero()),
+      mode_(Mode::kStartup),
+      default_bandwidth_(DataRate::KilobitsPerSec(kInitialBandwidthKbps)),
+      bandwidth_estimate_(default_bandwidth_),
+      rtt_tracker_(TimeDelta::Millis(kInitialRttMs), kAlphaForRtt),
+      monitor_interval_timeout_(TimeDelta::Millis(kInitialRttMs) *
+                                kTimeoutRatio),
+      monitor_interval_length_strategy_(MonitorIntervalLengthStrategy::kFixed),
+      monitor_interval_duration_ratio_(kMonitorIntervalDurationRatio),
+      sampling_step_(kDefaultSamplingStep),
+      monitor_interval_timeout_ratio_(kTimeoutRatio),
+      min_packets_number_per_interval_(kMinPacketsNumberPerInterval),
+      bitrate_controller_(kInitialConversionFactor,
+                          kInitialDynamicBoundary,
+                          kDynamicBoundaryIncrement,
+                          kRttGradientCoefficientBps,
+                          kLossCoefficientBps,
+                          kThroughputCoefficient,
+                          kThroughputPower,
+                          kRttGradientThreshold,
+                          kDelayGradientNegativeBound),
+      monitor_intervals_duration_(TimeDelta::Zero()),
+      complete_feedback_monitor_interval_number_(0),
+      random_generator_(kRandomSeed) {
+  if (config.constraints.starting_rate) {
+    default_bandwidth_ = *config.constraints.starting_rate;
+    bandwidth_estimate_ = default_bandwidth_;
+  }
+}
+
+PccNetworkController::~PccNetworkController() {}
+
+NetworkControlUpdate PccNetworkController::CreateRateUpdate(
+    Timestamp at_time) const {
+  DataRate sending_rate = DataRate::Zero();
+  if (monitor_intervals_.empty() ||
+      (monitor_intervals_.size() >= monitor_intervals_bitrates_.size() &&
+       at_time >= monitor_intervals_.back().GetEndTime())) {
+    sending_rate = bandwidth_estimate_;
+  } else {
+    sending_rate = monitor_intervals_.back().GetTargetSendingRate();
+  }
+  // Set up config when sending rate is computed.
+  NetworkControlUpdate update;
+
+  // Set up target rate to encoder.
+  TargetTransferRate target_rate_msg;
+  target_rate_msg.at_time = at_time;
+  target_rate_msg.network_estimate.at_time = at_time;
+  target_rate_msg.network_estimate.round_trip_time = rtt_tracker_.GetRtt();
+  // TODO(koloskova): Add correct estimate.
+  target_rate_msg.network_estimate.loss_rate_ratio = 0;
+  target_rate_msg.network_estimate.bwe_period =
+      monitor_interval_duration_ratio_ * rtt_tracker_.GetRtt();
+
+  target_rate_msg.target_rate = sending_rate;
+  update.target_rate = target_rate_msg;
+
+  // Set up pacing/padding target rate.
+  PacerConfig pacer_config;
+  pacer_config.at_time = at_time;
+  pacer_config.time_window = TimeDelta::Millis(1);
+  pacer_config.data_window = sending_rate * pacer_config.time_window;
+  pacer_config.pad_window = sending_rate * pacer_config.time_window;
+
+  update.pacer_config = pacer_config;
+  return update;
+}
+
+NetworkControlUpdate PccNetworkController::OnSentPacket(SentPacket msg) {
+  // Start new monitor interval if previous has finished.
+  // Monitor interval is initialized in OnProcessInterval function.
+  if (start_time_.IsInfinite()) {
+    start_time_ = msg.send_time;
+    monitor_intervals_duration_ = kStartupDuration;
+    monitor_intervals_bitrates_ = {bandwidth_estimate_};
+    monitor_intervals_.emplace_back(bandwidth_estimate_, msg.send_time,
+                                    monitor_intervals_duration_);
+    complete_feedback_monitor_interval_number_ = 0;
+  }
+  if (last_sent_packet_time_.IsFinite()) {
+    smoothed_packets_sending_interval_ =
+        (msg.send_time - last_sent_packet_time_) * kAlphaForPacketInterval +
+        (1 - kAlphaForPacketInterval) * smoothed_packets_sending_interval_;
+  }
+  last_sent_packet_time_ = msg.send_time;
+  if (!monitor_intervals_.empty() &&
+      msg.send_time >= monitor_intervals_.back().GetEndTime() &&
+      monitor_intervals_bitrates_.size() > monitor_intervals_.size()) {
+    // Start new monitor interval.
+    monitor_intervals_.emplace_back(
+        monitor_intervals_bitrates_[monitor_intervals_.size()], msg.send_time,
+        monitor_intervals_duration_);
+  }
+  if (IsTimeoutExpired(msg.send_time)) {
+    DataSize received_size = DataSize::Zero();
+    for (size_t i = 1; i < last_received_packets_.size(); ++i) {
+      received_size += last_received_packets_[i].sent_packet.size;
+    }
+    TimeDelta sending_time = TimeDelta::Zero();
+    if (last_received_packets_.size() > 0)
+      sending_time = last_received_packets_.back().receive_time -
+                     last_received_packets_.front().receive_time;
+    DataRate receiving_rate = bandwidth_estimate_;
+    if (sending_time > TimeDelta::Zero())
+      receiving_rate = received_size / sending_time;
+    bandwidth_estimate_ =
+        std::min<DataRate>(bandwidth_estimate_ * 0.5, receiving_rate);
+    if (mode_ == Mode::kSlowStart)
+      mode_ = Mode::kOnlineLearning;
+  }
+  if (mode_ == Mode::kStartup &&
+      msg.send_time - start_time_ >= kStartupDuration) {
+    DataSize received_size = DataSize::Zero();
+    for (size_t i = 1; i < last_received_packets_.size(); ++i) {
+      received_size += last_received_packets_[i].sent_packet.size;
+    }
+    TimeDelta sending_time = TimeDelta::Zero();
+    if (last_received_packets_.size() > 0)
+      sending_time = last_received_packets_.back().receive_time -
+                     last_received_packets_.front().receive_time;
+    DataRate receiving_rate = bandwidth_estimate_;
+    if (sending_time > TimeDelta::Zero())
+      receiving_rate = received_size / sending_time;
+    bandwidth_estimate_ = receiving_rate;
+    monitor_intervals_.clear();
+    mode_ = Mode::kSlowStart;
+    monitor_intervals_duration_ = ComputeMonitorIntervalsDuration();
+    monitor_intervals_bitrates_ = {bandwidth_estimate_};
+    monitor_intervals_.emplace_back(bandwidth_estimate_, msg.send_time,
+                                    monitor_intervals_duration_);
+    bandwidth_estimate_ = bandwidth_estimate_ * (1 / kSlowStartModeIncrease);
+    complete_feedback_monitor_interval_number_ = 0;
+    return CreateRateUpdate(msg.send_time);
+  }
+  if (IsFeedbackCollectionDone() || IsTimeoutExpired(msg.send_time)) {
+    // Creating new monitor intervals.
+    monitor_intervals_.clear();
+    monitor_interval_timeout_ =
+        rtt_tracker_.GetRtt() * monitor_interval_timeout_ratio_;
+    monitor_intervals_duration_ = ComputeMonitorIntervalsDuration();
+    complete_feedback_monitor_interval_number_ = 0;
+    // Compute bitrates and start first monitor interval.
+    if (mode_ == Mode::kSlowStart) {
+      monitor_intervals_bitrates_ = {kSlowStartModeIncrease *
+                                     bandwidth_estimate_};
+      monitor_intervals_.emplace_back(
+          kSlowStartModeIncrease * bandwidth_estimate_, msg.send_time,
+          monitor_intervals_duration_);
+    } else {
+      RTC_DCHECK(mode_ == Mode::kOnlineLearning || mode_ == Mode::kDoubleCheck);
+      monitor_intervals_.clear();
+      int64_t sign = 2 * (random_generator_.Rand(0, 1) % 2) - 1;
+      RTC_DCHECK_GE(sign, -1);
+      RTC_DCHECK_LE(sign, 1);
+      if (bandwidth_estimate_ >= kMinRateHaveMultiplicativeRateChange) {
+        monitor_intervals_bitrates_ = {
+            bandwidth_estimate_ * (1 + sign * sampling_step_),
+            bandwidth_estimate_ * (1 - sign * sampling_step_)};
+      } else {
+        monitor_intervals_bitrates_ = {
+            DataRate::BitsPerSec(std::max<double>(
+                bandwidth_estimate_.bps() + sign * kMinRateChangeBps, 0)),
+            DataRate::BitsPerSec(std::max<double>(
+                bandwidth_estimate_.bps() - sign * kMinRateChangeBps, 0))};
+      }
+      monitor_intervals_.emplace_back(monitor_intervals_bitrates_[0],
+                                      msg.send_time,
+                                      monitor_intervals_duration_);
+    }
+  }
+  return CreateRateUpdate(msg.send_time);
+}
+
+TimeDelta PccNetworkController::ComputeMonitorIntervalsDuration() const {
+  TimeDelta monitor_intervals_duration = TimeDelta::Zero();
+  if (monitor_interval_length_strategy_ ==
+      MonitorIntervalLengthStrategy::kAdaptive) {
+    monitor_intervals_duration = std::max(
+        rtt_tracker_.GetRtt() * monitor_interval_duration_ratio_,
+        smoothed_packets_sending_interval_ * min_packets_number_per_interval_);
+  } else {
+    RTC_DCHECK(monitor_interval_length_strategy_ ==
+               MonitorIntervalLengthStrategy::kFixed);
+    monitor_intervals_duration =
+        smoothed_packets_sending_interval_ * min_packets_number_per_interval_;
+  }
+  monitor_intervals_duration =
+      std::max(kMinDurationOfMonitorInterval, monitor_intervals_duration);
+  return monitor_intervals_duration;
+}
+
+bool PccNetworkController::IsTimeoutExpired(Timestamp current_time) const {
+  if (complete_feedback_monitor_interval_number_ >= monitor_intervals_.size()) {
+    return false;
+  }
+  return current_time -
+             monitor_intervals_[complete_feedback_monitor_interval_number_]
+                 .GetEndTime() >=
+         monitor_interval_timeout_;
+}
+
+bool PccNetworkController::IsFeedbackCollectionDone() const {
+  return complete_feedback_monitor_interval_number_ >=
+         monitor_intervals_bitrates_.size();
+}
+
+NetworkControlUpdate PccNetworkController::OnTransportPacketsFeedback(
+    TransportPacketsFeedback msg) {
+  if (msg.packet_feedbacks.empty())
+    return NetworkControlUpdate();
+  // Save packets to last_received_packets_ array.
+  for (const PacketResult& packet_result : msg.ReceivedWithSendInfo()) {
+    last_received_packets_.push_back(packet_result);
+  }
+  while (last_received_packets_.size() > kNumberOfPacketsToKeep) {
+    last_received_packets_.pop_front();
+  }
+  rtt_tracker_.OnPacketsFeedback(msg.PacketsWithFeedback(), msg.feedback_time);
+  // Skip rate update in case when online learning mode just started, but
+  // corresponding monitor intervals were not started yet.
+  if (mode_ == Mode::kOnlineLearning &&
+      monitor_intervals_bitrates_.size() < 2) {
+    return NetworkControlUpdate();
+  }
+  if (!IsFeedbackCollectionDone() && !monitor_intervals_.empty()) {
+    while (complete_feedback_monitor_interval_number_ <
+           monitor_intervals_.size()) {
+      monitor_intervals_[complete_feedback_monitor_interval_number_]
+          .OnPacketsFeedback(msg.PacketsWithFeedback());
+      if (!monitor_intervals_[complete_feedback_monitor_interval_number_]
+               .IsFeedbackCollectionDone())
+        break;
+      ++complete_feedback_monitor_interval_number_;
+    }
+  }
+  if (IsFeedbackCollectionDone()) {
+    if (mode_ == Mode::kDoubleCheck) {
+      mode_ = Mode::kOnlineLearning;
+    } else if (NeedDoubleCheckMeasurments()) {
+      mode_ = Mode::kDoubleCheck;
+    }
+    if (mode_ != Mode::kDoubleCheck)
+      UpdateSendingRateAndMode();
+  }
+  return NetworkControlUpdate();
+}
+
+bool PccNetworkController::NeedDoubleCheckMeasurments() const {
+  if (mode_ == Mode::kSlowStart) {
+    return false;
+  }
+  double first_loss_rate = monitor_intervals_[0].GetLossRate();
+  double second_loss_rate = monitor_intervals_[1].GetLossRate();
+  DataRate first_bitrate = monitor_intervals_[0].GetTargetSendingRate();
+  DataRate second_bitrate = monitor_intervals_[1].GetTargetSendingRate();
+  if ((first_bitrate.bps() - second_bitrate.bps()) *
+          (first_loss_rate - second_loss_rate) <
+      0) {
+    return true;
+  }
+  return false;
+}
+
+void PccNetworkController::UpdateSendingRateAndMode() {
+  if (monitor_intervals_.empty() || !IsFeedbackCollectionDone()) {
+    return;
+  }
+  if (mode_ == Mode::kSlowStart) {
+    DataRate old_bandwidth_estimate = bandwidth_estimate_;
+    bandwidth_estimate_ =
+        bitrate_controller_
+            .ComputeRateUpdateForSlowStartMode(monitor_intervals_[0])
+            .value_or(bandwidth_estimate_);
+    if (bandwidth_estimate_ <= old_bandwidth_estimate)
+      mode_ = Mode::kOnlineLearning;
+  } else {
+    RTC_DCHECK(mode_ == Mode::kOnlineLearning);
+    bandwidth_estimate_ =
+        bitrate_controller_.ComputeRateUpdateForOnlineLearningMode(
+            monitor_intervals_, bandwidth_estimate_);
+  }
+}
+
+NetworkControlUpdate PccNetworkController::OnNetworkAvailability(
+    NetworkAvailability msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnNetworkRouteChange(
+    NetworkRouteChange msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnProcessInterval(
+    ProcessInterval msg) {
+  return CreateRateUpdate(msg.at_time);
+}
+
+NetworkControlUpdate PccNetworkController::OnTargetRateConstraints(
+    TargetRateConstraints msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnRemoteBitrateReport(
+    RemoteBitrateReport) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnRoundTripTimeUpdate(
+    RoundTripTimeUpdate) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnTransportLossReport(
+    TransportLossReport) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnStreamsConfig(StreamsConfig msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnReceivedPacket(
+    ReceivedPacket msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnNetworkStateEstimate(
+    NetworkStateEstimate msg) {
+  return NetworkControlUpdate();
+}
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.h b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.h
new file mode 100644
index 0000000000..e5f65dd7d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.h
@@ -0,0 +1,125 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_PCC_NETWORK_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_PCC_NETWORK_CONTROLLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <deque>
+#include <vector>
+
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/pcc/bitrate_controller.h"
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+#include "modules/congestion_controller/pcc/rtt_tracker.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+namespace pcc {
+
+// PCC (Performance-oriented Congestion Control) Vivace is a congestion
+// control algorithm based on online (convex) optimization in machine learning.
+// It divides time into consecutive Monitor Intervals (MI) to test sending
+// rates r(1 + eps), r(1 - eps) for the current sending rate r.
+// At the end of each MI it computes utility function to transform the
+// performance statistics into a numerical value. Then it updates current
+// sending rate using gradient ascent to maximize utility function.
+class PccNetworkController : public NetworkControllerInterface {
+ public:
+  enum class Mode {
+    kStartup,
+    // Slow start phase of PCC doubles sending rate each monitor interval.
+    kSlowStart,
+    // After getting the first decrease in utility function PCC exits slow start
+    // and enters the online learning phase.
+    kOnlineLearning,
+    // If we got that sending with the lower rate resulted in higher packet
+    // loss, then the measurements are unreliable and we need to double check
+    // them.
+    kDoubleCheck
+  };
+
+  enum class MonitorIntervalLengthStrategy {
+    // Monitor interval length adaptive when it is proportional to packets RTT.
+    kAdaptive,
+    // Monitor interval length is fixed when it is equal to the time of sending
+    // predefined amount of packets (kMinPacketsNumberPerInterval).
+    kFixed
+  };
+
+  explicit PccNetworkController(NetworkControllerConfig config);
+  ~PccNetworkController() override;
+
+  // NetworkControllerInterface
+  NetworkControlUpdate OnNetworkAvailability(NetworkAvailability msg) override;
+  NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange msg) override;
+  NetworkControlUpdate OnProcessInterval(ProcessInterval msg) override;
+  NetworkControlUpdate OnSentPacket(SentPacket msg) override;
+  NetworkControlUpdate OnTargetRateConstraints(
+      TargetRateConstraints msg) override;
+  NetworkControlUpdate OnTransportPacketsFeedback(
+      TransportPacketsFeedback msg) override;
+
+  // Part of remote bitrate estimation api, not implemented for PCC
+  NetworkControlUpdate OnStreamsConfig(StreamsConfig msg) override;
+  NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport msg) override;
+  NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override;
+  NetworkControlUpdate OnTransportLossReport(TransportLossReport msg) override;
+  NetworkControlUpdate OnReceivedPacket(ReceivedPacket msg) override;
+  NetworkControlUpdate OnNetworkStateEstimate(
+      NetworkStateEstimate msg) override;
+
+ private:
+  void UpdateSendingRateAndMode();
+  NetworkControlUpdate CreateRateUpdate(Timestamp at_time) const;
+  TimeDelta ComputeMonitorIntervalsDuration() const;
+  bool NeedDoubleCheckMeasurments() const;
+  bool IsTimeoutExpired(Timestamp current_time) const;
+  bool IsFeedbackCollectionDone() const;
+
+  Timestamp start_time_;
+  Timestamp last_sent_packet_time_;
+  TimeDelta smoothed_packets_sending_interval_;
+  Mode mode_;
+
+  // Default value used for initializing bandwidth.
+  DataRate default_bandwidth_;
+  // Current estimate r.
+  DataRate bandwidth_estimate_;
+
+  RttTracker rtt_tracker_;
+  TimeDelta monitor_interval_timeout_;
+  const MonitorIntervalLengthStrategy monitor_interval_length_strategy_;
+  const double monitor_interval_duration_ratio_;
+  const double sampling_step_;  // Epsilon.
+  const double monitor_interval_timeout_ratio_;
+  const int64_t min_packets_number_per_interval_;
+
+  PccBitrateController bitrate_controller_;
+
+  std::vector<PccMonitorInterval> monitor_intervals_;
+  std::vector<DataRate> monitor_intervals_bitrates_;
+  TimeDelta monitor_intervals_duration_;
+  size_t complete_feedback_monitor_interval_number_;
+
+  webrtc::Random random_generator_;
+  std::deque<PacketResult> last_received_packets_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_PCC_NETWORK_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc
new file mode 100644
index 0000000000..c98680c785
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/pcc_network_controller.h"
+
+#include <memory>
+
+#include "modules/congestion_controller/pcc/pcc_factory.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+using ::testing::AllOf;
+using ::testing::Field;
+using ::testing::Ge;
+using ::testing::Le;
+using ::testing::Matcher;
+using ::testing::Property;
+
+namespace webrtc {
+namespace test {
+namespace {
+
+const DataRate kInitialBitrate = DataRate::KilobitsPerSec(60);
+const Timestamp kDefaultStartTime = Timestamp::Millis(10000000);
+
+constexpr double kDataRateMargin = 0.20;
+constexpr double kMinDataRateFactor = 1 - kDataRateMargin;
+constexpr double kMaxDataRateFactor = 1 + kDataRateMargin;
+inline Matcher<TargetTransferRate> TargetRateCloseTo(DataRate rate) {
+  DataRate min_data_rate = rate * kMinDataRateFactor;
+  DataRate max_data_rate = rate * kMaxDataRateFactor;
+  return Field(&TargetTransferRate::target_rate,
+               AllOf(Ge(min_data_rate), Le(max_data_rate)));
+}
+
+NetworkControllerConfig InitialConfig(
+    int starting_bandwidth_kbps = kInitialBitrate.kbps(),
+    int min_data_rate_kbps = 0,
+    int max_data_rate_kbps = 5 * kInitialBitrate.kbps()) {
+  NetworkControllerConfig config;
+  config.constraints.at_time = kDefaultStartTime;
+  config.constraints.min_data_rate =
+      DataRate::KilobitsPerSec(min_data_rate_kbps);
+  config.constraints.max_data_rate =
+      DataRate::KilobitsPerSec(max_data_rate_kbps);
+  config.constraints.starting_rate =
+      DataRate::KilobitsPerSec(starting_bandwidth_kbps);
+  return config;
+}
+
+ProcessInterval InitialProcessInterval() {
+  ProcessInterval process_interval;
+  process_interval.at_time = kDefaultStartTime;
+  return process_interval;
+}
+
+}  // namespace
+
+TEST(PccNetworkControllerTest, SendsConfigurationOnFirstProcess) {
+  std::unique_ptr<NetworkControllerInterface> controller_;
+  controller_.reset(new pcc::PccNetworkController(InitialConfig()));
+
+  NetworkControlUpdate update =
+      controller_->OnProcessInterval(InitialProcessInterval());
+  EXPECT_THAT(*update.target_rate, TargetRateCloseTo(kInitialBitrate));
+  EXPECT_THAT(*update.pacer_config,
+              Property(&PacerConfig::data_rate, Ge(kInitialBitrate)));
+}
+
+TEST(PccNetworkControllerTest, UpdatesTargetSendRate) {
+  PccNetworkControllerFactory factory;
+  Scenario s("pcc_unit/updates_rate", false);
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(1500);
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  auto send_net = s.CreateMutableSimulationNode([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(500);
+    c->delay = TimeDelta::Millis(100);
+  });
+  auto ret_net = s.CreateMutableSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+
+  auto* client = s.CreateClient("send", config);
+  auto* route = s.CreateRoutes(client, {send_net->node()},
+                               s.CreateClient("return", CallClientConfig()),
+                               {ret_net->node()});
+  VideoStreamConfig video;
+  video.stream.use_rtx = false;
+  s.CreateVideoStream(route->forward(), video);
+  s.RunFor(TimeDelta::Seconds(30));
+  EXPECT_NEAR(client->target_rate().kbps(), 450, 100);
+  send_net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(800);
+    c->delay = TimeDelta::Millis(100);
+  });
+  s.RunFor(TimeDelta::Seconds(20));
+  EXPECT_NEAR(client->target_rate().kbps(), 750, 150);
+  send_net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(200);
+    c->delay = TimeDelta::Millis(200);
+  });
+  ret_net->UpdateConfig(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(200); });
+  s.RunFor(TimeDelta::Seconds(35));
+  EXPECT_NEAR(client->target_rate().kbps(), 170, 50);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.cc
new file mode 100644
index 0000000000..af9dc8f11b
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/rtt_tracker.h"
+
+#include <algorithm>
+
+namespace webrtc {
+namespace pcc {
+
+RttTracker::RttTracker(TimeDelta initial_rtt, double alpha)
+    : rtt_estimate_(initial_rtt), alpha_(alpha) {}
+
+void RttTracker::OnPacketsFeedback(
+    const std::vector<PacketResult>& packet_feedbacks,
+    Timestamp feedback_received_time) {
+  TimeDelta packet_rtt = TimeDelta::MinusInfinity();
+  for (const PacketResult& packet_result : packet_feedbacks) {
+    if (!packet_result.IsReceived())
+      continue;
+    packet_rtt = std::max<TimeDelta>(
+        packet_rtt,
+        feedback_received_time - packet_result.sent_packet.send_time);
+  }
+  if (packet_rtt.IsFinite())
+    rtt_estimate_ = (1 - alpha_) * rtt_estimate_ + alpha_ * packet_rtt;
+}
+
+TimeDelta RttTracker::GetRtt() const {
+  return rtt_estimate_;
+}
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.h b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.h
new file mode 100644
index 0000000000..94033cd511
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_RTT_TRACKER_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_RTT_TRACKER_H_
+
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+namespace pcc {
+
+class RttTracker {
+ public:
+  RttTracker(TimeDelta initial_rtt, double alpha);
+  // Updates RTT estimate.
+  void OnPacketsFeedback(const std::vector<PacketResult>& packet_feedbacks,
+                         Timestamp feedback_received_time);
+  TimeDelta GetRtt() const;
+
+ private:
+  TimeDelta rtt_estimate_;
+  double alpha_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_RTT_TRACKER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker_unittest.cc
new file mode 100644
index 0000000000..7d90e86822
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker_unittest.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/rtt_tracker.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace pcc {
+namespace test {
+namespace {
+const TimeDelta kInitialRtt = TimeDelta::Micros(10);
+constexpr double kAlpha = 0.9;
+const Timestamp kStartTime = Timestamp::Seconds(0);
+
+PacketResult GetPacketWithRtt(TimeDelta rtt) {
+  SentPacket packet;
+  packet.send_time = kStartTime;
+  PacketResult packet_result;
+  packet_result.sent_packet = packet;
+  if (rtt.IsFinite()) {
+    packet_result.receive_time = kStartTime + rtt;
+  } else {
+    packet_result.receive_time = Timestamp::PlusInfinity();
+  }
+  return packet_result;
+}
+}  // namespace
+
+TEST(PccRttTrackerTest, InitialValue) {
+  RttTracker tracker{kInitialRtt, kAlpha};
+  EXPECT_EQ(kInitialRtt, tracker.GetRtt());
+  for (int i = 0; i < 100; ++i) {
+    tracker.OnPacketsFeedback({GetPacketWithRtt(kInitialRtt)},
+                              kStartTime + kInitialRtt);
+  }
+  EXPECT_EQ(kInitialRtt, tracker.GetRtt());
+}
+
+TEST(PccRttTrackerTest, DoNothingWhenPacketIsLost) {
+  RttTracker tracker{kInitialRtt, kAlpha};
+  tracker.OnPacketsFeedback({GetPacketWithRtt(TimeDelta::PlusInfinity())},
+                            kStartTime + kInitialRtt);
+  EXPECT_EQ(tracker.GetRtt(), kInitialRtt);
+}
+
+TEST(PccRttTrackerTest, ChangeInRtt) {
+  RttTracker tracker{kInitialRtt, kAlpha};
+  const TimeDelta kNewRtt = TimeDelta::Micros(100);
+  tracker.OnPacketsFeedback({GetPacketWithRtt(kNewRtt)}, kStartTime + kNewRtt);
+  EXPECT_GT(tracker.GetRtt(), kInitialRtt);
+  EXPECT_LE(tracker.GetRtt(), kNewRtt);
+  for (int i = 0; i < 100; ++i) {
+    tracker.OnPacketsFeedback({GetPacketWithRtt(kNewRtt)},
+                              kStartTime + kNewRtt);
+  }
+  const TimeDelta absolute_error = TimeDelta::Micros(1);
+  EXPECT_NEAR(tracker.GetRtt().us(), kNewRtt.us(), absolute_error.us());
+  EXPECT_LE(tracker.GetRtt(), kNewRtt);
+}
+
+}  // namespace test
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.cc
new file mode 100644
index 0000000000..006a2fccd9
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/utility_function.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "api/units/data_rate.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace pcc {
+
+VivaceUtilityFunction::VivaceUtilityFunction(
+    double delay_gradient_coefficient,
+    double loss_coefficient,
+    double throughput_coefficient,
+    double throughput_power,
+    double delay_gradient_threshold,
+    double delay_gradient_negative_bound)
+    : delay_gradient_coefficient_(delay_gradient_coefficient),
+      loss_coefficient_(loss_coefficient),
+      throughput_power_(throughput_power),
+      throughput_coefficient_(throughput_coefficient),
+      delay_gradient_threshold_(delay_gradient_threshold),
+      delay_gradient_negative_bound_(delay_gradient_negative_bound) {
+  RTC_DCHECK_GE(delay_gradient_negative_bound_, 0);
+}
+
+double VivaceUtilityFunction::Compute(
+    const PccMonitorInterval& monitor_interval) const {
+  RTC_DCHECK(monitor_interval.IsFeedbackCollectionDone());
+  double bitrate = monitor_interval.GetTargetSendingRate().bps();
+  double loss_rate = monitor_interval.GetLossRate();
+  double rtt_gradient =
+      monitor_interval.ComputeDelayGradient(delay_gradient_threshold_);
+  rtt_gradient = std::max(rtt_gradient, -delay_gradient_negative_bound_);
+  return (throughput_coefficient_ * std::pow(bitrate, throughput_power_)) -
+         (delay_gradient_coefficient_ * bitrate * rtt_gradient) -
+         (loss_coefficient_ * bitrate * loss_rate);
+}
+
+VivaceUtilityFunction::~VivaceUtilityFunction() = default;
+
+ModifiedVivaceUtilityFunction::ModifiedVivaceUtilityFunction(
+    double delay_gradient_coefficient,
+    double loss_coefficient,
+    double throughput_coefficient,
+    double throughput_power,
+    double delay_gradient_threshold,
+    double delay_gradient_negative_bound)
+    : delay_gradient_coefficient_(delay_gradient_coefficient),
+      loss_coefficient_(loss_coefficient),
+      throughput_power_(throughput_power),
+      throughput_coefficient_(throughput_coefficient),
+      delay_gradient_threshold_(delay_gradient_threshold),
+      delay_gradient_negative_bound_(delay_gradient_negative_bound) {
+  RTC_DCHECK_GE(delay_gradient_negative_bound_, 0);
+}
+
+double ModifiedVivaceUtilityFunction::Compute(
+    const PccMonitorInterval& monitor_interval) const {
+  RTC_DCHECK(monitor_interval.IsFeedbackCollectionDone());
+  double bitrate = monitor_interval.GetTargetSendingRate().bps();
+  double loss_rate = monitor_interval.GetLossRate();
+  double rtt_gradient =
+      monitor_interval.ComputeDelayGradient(delay_gradient_threshold_);
+  rtt_gradient = std::max(rtt_gradient, -delay_gradient_negative_bound_);
+  return (throughput_coefficient_ * std::pow(bitrate, throughput_power_) *
+          bitrate) -
+         (delay_gradient_coefficient_ * bitrate * bitrate * rtt_gradient) -
+         (loss_coefficient_ * bitrate * bitrate * loss_rate);
+}
+
+ModifiedVivaceUtilityFunction::~ModifiedVivaceUtilityFunction() = default;
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.h b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.h
new file mode 100644
index 0000000000..98bb0744c1
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.h
@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_UTILITY_FUNCTION_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_UTILITY_FUNCTION_H_
+
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+
+namespace webrtc {
+namespace pcc {
+
+// Utility function is used by PCC to transform the performance statistics
+// (sending rate, loss rate, packets latency) gathered at one monitor interval
+// into a numerical value.
+// https://www.usenix.org/conference/nsdi18/presentation/dong
+class PccUtilityFunctionInterface {
+ public:
+  virtual double Compute(const PccMonitorInterval& monitor_interval) const = 0;
+  virtual ~PccUtilityFunctionInterface() = default;
+};
+
+// Vivace utility function were suggested in the paper "PCC Vivace:
+// Online-Learning Congestion Control", Mo Dong et all.
+class VivaceUtilityFunction : public PccUtilityFunctionInterface {
+ public:
+  VivaceUtilityFunction(double delay_gradient_coefficient,
+                        double loss_coefficient,
+                        double throughput_coefficient,
+                        double throughput_power,
+                        double delay_gradient_threshold,
+                        double delay_gradient_negative_bound);
+  double Compute(const PccMonitorInterval& monitor_interval) const override;
+  ~VivaceUtilityFunction() override;
+
+ private:
+  const double delay_gradient_coefficient_;
+  const double loss_coefficient_;
+  const double throughput_power_;
+  const double throughput_coefficient_;
+  const double delay_gradient_threshold_;
+  const double delay_gradient_negative_bound_;
+};
+
+// This utility function were obtained by tuning Vivace utility function.
+// The main difference is that gradient of modified utilify funtion (as well as
+// rate updates) scales proportionally to the sending rate which leads to
+// better performance in case of single sender.
+class ModifiedVivaceUtilityFunction : public PccUtilityFunctionInterface {
+ public:
+  ModifiedVivaceUtilityFunction(double delay_gradient_coefficient,
+                                double loss_coefficient,
+                                double throughput_coefficient,
+                                double throughput_power,
+                                double delay_gradient_threshold,
+                                double delay_gradient_negative_bound);
+  double Compute(const PccMonitorInterval& monitor_interval) const override;
+  ~ModifiedVivaceUtilityFunction() override;
+
+ private:
+  const double delay_gradient_coefficient_;
+  const double loss_coefficient_;
+  const double throughput_power_;
+  const double throughput_coefficient_;
+  const double delay_gradient_threshold_;
+  const double delay_gradient_negative_bound_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_UTILITY_FUNCTION_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function_unittest.cc
new file mode 100644
index 0000000000..19b2d15920
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function_unittest.cc
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/utility_function.h"
+
+#include <stddef.h>
+
+#include <cmath>
+#include <type_traits>
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace pcc {
+namespace test {
+namespace {
+constexpr double kLossCoefficient = 11.35;
+constexpr double kThroughputPower = 0.9;
+constexpr double kThroughputCoefficient = 1;
+constexpr double kDelayGradientNegativeBound = 10;
+
+const Timestamp kStartTime = Timestamp::Micros(0);
+const TimeDelta kPacketsDelta = TimeDelta::Millis(1);
+const TimeDelta kIntervalDuration = TimeDelta::Millis(100);
+const DataRate kSendingBitrate = DataRate::BitsPerSec(1000);
+
+const DataSize kDefaultDataSize = DataSize::Bytes(100);
+const TimeDelta kDefaultDelay = TimeDelta::Millis(100);
+
+std::vector<PacketResult> CreatePacketResults(
+    const std::vector<Timestamp>& packets_send_times,
+    const std::vector<Timestamp>& packets_received_times = {},
+    const std::vector<DataSize>& packets_sizes = {}) {
+  std::vector<PacketResult> packet_results;
+  PacketResult packet_result;
+  SentPacket sent_packet;
+  for (size_t i = 0; i < packets_send_times.size(); ++i) {
+    sent_packet.send_time = packets_send_times[i];
+    if (packets_sizes.empty()) {
+      sent_packet.size = kDefaultDataSize;
+    } else {
+      sent_packet.size = packets_sizes[i];
+    }
+    packet_result.sent_packet = sent_packet;
+    if (packets_received_times.empty()) {
+      packet_result.receive_time = packets_send_times[i] + kDefaultDelay;
+    } else {
+      packet_result.receive_time = packets_received_times[i];
+    }
+    packet_results.push_back(packet_result);
+  }
+  return packet_results;
+}
+
+}  // namespace
+
+TEST(PccVivaceUtilityFunctionTest,
+     UtilityIsThroughputTermIfAllRestCoefficientsAreZero) {
+  VivaceUtilityFunction utility_function(0, 0, kThroughputCoefficient,
+                                         kThroughputPower, 0,
+                                         kDelayGradientNegativeBound);
+  PccMonitorInterval monitor_interval(kSendingBitrate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 3 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kPacketsDelta + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay + 3 * kPacketsDelta,
+       Timestamp::PlusInfinity()},
+      {kDefaultDataSize, kDefaultDataSize, kDefaultDataSize,
+       kDefaultDataSize}));
+  EXPECT_DOUBLE_EQ(utility_function.Compute(monitor_interval),
+                   kThroughputCoefficient *
+                       std::pow(kSendingBitrate.bps(), kThroughputPower));
+}
+
+TEST(PccVivaceUtilityFunctionTest,
+     LossTermIsNonZeroIfLossCoefficientIsNonZero) {
+  VivaceUtilityFunction utility_function(
+      0, kLossCoefficient, kThroughputCoefficient, kThroughputPower, 0,
+      kDelayGradientNegativeBound);
+  PccMonitorInterval monitor_interval(kSendingBitrate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 5 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay, kStartTime + 3 * kIntervalDuration},
+      {}));
+  // The second packet was lost.
+  EXPECT_DOUBLE_EQ(utility_function.Compute(monitor_interval),
+                   kThroughputCoefficient *
+                           std::pow(kSendingBitrate.bps(), kThroughputPower) -
+                       kLossCoefficient * kSendingBitrate.bps() *
+                           monitor_interval.GetLossRate());
+}
+
+}  // namespace test
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc
new file mode 100644
index 0000000000..e43b020f6e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc
@@ -0,0 +1,134 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+
+#include "api/units/data_rate.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+static const uint32_t kTimeOffsetSwitchThreshold = 30;
+}  // namespace
+
+void ReceiveSideCongestionController::OnRttUpdate(int64_t avg_rtt_ms,
+                                                  int64_t max_rtt_ms) {
+  MutexLock lock(&mutex_);
+  rbe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
+}
+
+void ReceiveSideCongestionController::RemoveStream(uint32_t ssrc) {
+  MutexLock lock(&mutex_);
+  rbe_->RemoveStream(ssrc);
+}
+
+DataRate ReceiveSideCongestionController::LatestReceiveSideEstimate() const {
+  MutexLock lock(&mutex_);
+  return rbe_->LatestEstimate();
+}
+
+void ReceiveSideCongestionController::PickEstimatorFromHeader(
+    const RTPHeader& header) {
+  if (header.extension.hasAbsoluteSendTime) {
+    // If we see AST in header, switch RBE strategy immediately.
+    if (!using_absolute_send_time_) {
+      RTC_LOG(LS_INFO)
+          << "WrappingBitrateEstimator: Switching to absolute send time RBE.";
+      using_absolute_send_time_ = true;
+      PickEstimator();
+    }
+    packets_since_absolute_send_time_ = 0;
+  } else {
+    // When we don't see AST, wait for a few packets before going back to TOF.
+    if (using_absolute_send_time_) {
+      ++packets_since_absolute_send_time_;
+      if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
+        RTC_LOG(LS_INFO)
+            << "WrappingBitrateEstimator: Switching to transmission "
+               "time offset RBE.";
+        using_absolute_send_time_ = false;
+        PickEstimator();
+      }
+    }
+  }
+}
+
+// Instantiate RBE for Time Offset or Absolute Send Time extensions.
+void ReceiveSideCongestionController::PickEstimator() {
+  if (using_absolute_send_time_) {
+    rbe_ = std::make_unique<RemoteBitrateEstimatorAbsSendTime>(&remb_throttler_,
+                                                               &clock_);
+  } else {
+    rbe_ = std::make_unique<RemoteBitrateEstimatorSingleStream>(
+        &remb_throttler_, &clock_);
+  }
+}
+
+ReceiveSideCongestionController::ReceiveSideCongestionController(
+    Clock* clock,
+    RemoteEstimatorProxy::TransportFeedbackSender feedback_sender,
+    RembThrottler::RembSender remb_sender,
+    NetworkStateEstimator* network_state_estimator)
+    : clock_(*clock),
+      remb_throttler_(std::move(remb_sender), clock),
+      remote_estimator_proxy_(std::move(feedback_sender),
+                              network_state_estimator),
+      rbe_(new RemoteBitrateEstimatorSingleStream(&remb_throttler_, clock)),
+      using_absolute_send_time_(false),
+      packets_since_absolute_send_time_(0) {}
+
+void ReceiveSideCongestionController::OnReceivedPacket(
+    int64_t arrival_time_ms,
+    size_t payload_size,
+    const RTPHeader& header) {
+  remote_estimator_proxy_.IncomingPacket(arrival_time_ms, payload_size, header);
+  if (!header.extension.hasTransportSequenceNumber) {
+    // Receive-side BWE.
+    MutexLock lock(&mutex_);
+    PickEstimatorFromHeader(header);
+    rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
+  }
+}
+
+void ReceiveSideCongestionController::SetSendPeriodicFeedback(
+    bool send_periodic_feedback) {
+  remote_estimator_proxy_.SetSendPeriodicFeedback(send_periodic_feedback);
+}
+
+void ReceiveSideCongestionController::OnBitrateChanged(int bitrate_bps) {
+  remote_estimator_proxy_.OnBitrateChanged(bitrate_bps);
+}
+
+TimeDelta ReceiveSideCongestionController::MaybeProcess() {
+  Timestamp now = clock_.CurrentTime();
+  mutex_.Lock();
+  TimeDelta time_until_rbe = rbe_->Process();
+  mutex_.Unlock();
+  TimeDelta time_until_rep = remote_estimator_proxy_.Process(now);
+  TimeDelta time_until = std::min(time_until_rbe, time_until_rep);
+  return std::max(time_until, TimeDelta::Zero());
+}
+
+void ReceiveSideCongestionController::SetMaxDesiredReceiveBitrate(
+    DataRate bitrate) {
+  remb_throttler_.SetMaxDesiredReceiveBitrate(bitrate);
+}
+
+void ReceiveSideCongestionController::SetTransportOverhead(
+    DataSize overhead_per_packet) {
+  remote_estimator_proxy_.SetTransportOverhead(overhead_per_packet);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller_unittest.cc
new file mode 100644
index 0000000000..f2fd6d11d7
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller_unittest.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+
+#include "api/test/network_emulation/create_cross_traffic.h"
+#include "api/test/network_emulation/cross_traffic.h"
+#include "modules/pacing/packet_router.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::ElementsAre;
+using ::testing::MockFunction;
+
+namespace webrtc {
+
+namespace {
+
+// Helper to convert some time format to resolution used in absolute send time
+// header extension, rounded upwards. `t` is the time to convert, in some
+// resolution. `denom` is the value to divide `t` by to get whole seconds,
+// e.g. `denom` = 1000 if `t` is in milliseconds.
+uint32_t AbsSendTime(int64_t t, int64_t denom) {
+  return (((t << 18) + (denom >> 1)) / denom) & 0x00fffffful;
+}
+
+const uint32_t kInitialBitrateBps = 60000;
+
+}  // namespace
+
+namespace test {
+
+TEST(ReceiveSideCongestionControllerTest, SendsRembWithAbsSendTime) {
+  MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
+      feedback_sender;
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  SimulatedClock clock_(123456);
+
+  ReceiveSideCongestionController controller(
+      &clock_, feedback_sender.AsStdFunction(), remb_sender.AsStdFunction(),
+      nullptr);
+
+  size_t payload_size = 1000;
+  RTPHeader header;
+  header.ssrc = 0x11eb21c;
+  header.extension.hasAbsoluteSendTime = true;
+
+  EXPECT_CALL(remb_sender, Call(_, ElementsAre(header.ssrc))).Times(AtLeast(1));
+
+  for (int i = 0; i < 10; ++i) {
+    clock_.AdvanceTimeMilliseconds((1000 * payload_size) / kInitialBitrateBps);
+    int64_t now_ms = clock_.TimeInMilliseconds();
+    header.extension.absoluteSendTime = AbsSendTime(now_ms, 1000);
+    controller.OnReceivedPacket(now_ms, payload_size, header);
+  }
+}
+
+TEST(ReceiveSideCongestionControllerTest,
+     SendsRembAfterSetMaxDesiredReceiveBitrate) {
+  MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
+      feedback_sender;
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  SimulatedClock clock_(123456);
+
+  ReceiveSideCongestionController controller(
+      &clock_, feedback_sender.AsStdFunction(), remb_sender.AsStdFunction(),
+      nullptr);
+  EXPECT_CALL(remb_sender, Call(123, _));
+  controller.SetMaxDesiredReceiveBitrate(DataRate::BitsPerSec(123));
+}
+
+TEST(ReceiveSideCongestionControllerTest, ConvergesToCapacity) {
+  Scenario s("receive_cc_unit/converge");
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
+  net_conf.delay = TimeDelta::Millis(50);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  });
+
+  auto* route = s.CreateRoutes(client, {s.CreateSimulationNode(net_conf)},
+                               s.CreateClient("return", CallClientConfig()),
+                               {s.CreateSimulationNode(net_conf)});
+  VideoStreamConfig video;
+  video.stream.packet_feedback = false;
+  s.CreateVideoStream(route->forward(), video);
+  s.RunFor(TimeDelta::Seconds(30));
+  EXPECT_NEAR(client->send_bandwidth().kbps(), 900, 150);
+}
+
+TEST(ReceiveSideCongestionControllerTest, IsFairToTCP) {
+  Scenario s("receive_cc_unit/tcp_fairness");
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
+  net_conf.delay = TimeDelta::Millis(50);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  });
+  auto send_net = {s.CreateSimulationNode(net_conf)};
+  auto ret_net = {s.CreateSimulationNode(net_conf)};
+  auto* route = s.CreateRoutes(
+      client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
+  VideoStreamConfig video;
+  video.stream.packet_feedback = false;
+  s.CreateVideoStream(route->forward(), video);
+  s.net()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
+      s.net()->CreateRoute(send_net), s.net()->CreateRoute(ret_net),
+      FakeTcpConfig()));
+  s.RunFor(TimeDelta::Seconds(30));
+  // For some reason we get outcompeted by TCP here, this should probably be
+  // fixed and a lower bound should be added to the test.
+  EXPECT_LT(client->send_bandwidth().kbps(), 750);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/remb_throttler.cc b/third_party/libwebrtc/modules/congestion_controller/remb_throttler.cc
new file mode 100644
index 0000000000..fcc30af9a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/remb_throttler.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/remb_throttler.h"
+
+#include <algorithm>
+#include <utility>
+
+namespace webrtc {
+
+namespace {
+constexpr TimeDelta kRembSendInterval = TimeDelta::Millis(200);
+}  // namespace
+
+RembThrottler::RembThrottler(RembSender remb_sender, Clock* clock)
+    : remb_sender_(std::move(remb_sender)),
+      clock_(clock),
+      last_remb_time_(Timestamp::MinusInfinity()),
+      last_send_remb_bitrate_(DataRate::PlusInfinity()),
+      max_remb_bitrate_(DataRate::PlusInfinity()) {}
+
+void RembThrottler::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                                            uint32_t bitrate_bps) {
+  DataRate receive_bitrate = DataRate::BitsPerSec(bitrate_bps);
+  Timestamp now = clock_->CurrentTime();
+  {
+    MutexLock lock(&mutex_);
+    // % threshold for if we should send a new REMB asap.
+    const int64_t kSendThresholdPercent = 103;
+    if (receive_bitrate * kSendThresholdPercent / 100 >
+            last_send_remb_bitrate_ &&
+        now < last_remb_time_ + kRembSendInterval) {
+      return;
+    }
+    last_remb_time_ = now;
+    last_send_remb_bitrate_ = receive_bitrate;
+    receive_bitrate = std::min(last_send_remb_bitrate_, max_remb_bitrate_);
+  }
+  remb_sender_(receive_bitrate.bps(), ssrcs);
+}
+
+void RembThrottler::SetMaxDesiredReceiveBitrate(DataRate bitrate) {
+  Timestamp now = clock_->CurrentTime();
+  {
+    MutexLock lock(&mutex_);
+    max_remb_bitrate_ = bitrate;
+    if (now - last_remb_time_ < kRembSendInterval &&
+        !last_send_remb_bitrate_.IsZero() &&
+        last_send_remb_bitrate_ <= max_remb_bitrate_) {
+      return;
+    }
+  }
+  remb_sender_(bitrate.bps(), /*ssrcs=*/{});
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/remb_throttler.h b/third_party/libwebrtc/modules/congestion_controller/remb_throttler.h
new file mode 100644
index 0000000000..85292cbc09
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/remb_throttler.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_REMB_THROTTLER_H_
+#define MODULES_CONGESTION_CONTROLLER_REMB_THROTTLER_H_
+
+#include <functional>
+#include <vector>
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// RembThrottler is a helper class used for throttling RTCP REMB messages.
+// Throttles small changes to the received BWE within 200ms.
+class RembThrottler : public RemoteBitrateObserver {
+ public:
+  using RembSender =
+      std::function<void(int64_t bitrate_bps, std::vector<uint32_t> ssrcs)>;
+  RembThrottler(RembSender remb_sender, Clock* clock);
+
+  // Ensures the remote party is notified of the receive bitrate no larger than
+  // `bitrate` using RTCP REMB.
+  void SetMaxDesiredReceiveBitrate(DataRate bitrate);
+
+  // Implements RemoteBitrateObserver;
+  // Called every time there is a new bitrate estimate for a receive channel
+  // group. This call will trigger a new RTCP REMB packet if the bitrate
+  // estimate has decreased or if no RTCP REMB packet has been sent for
+  // a certain time interval.
+  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                               uint32_t bitrate_bps) override;
+
+ private:
+  const RembSender remb_sender_;
+  Clock* const clock_;
+  mutable Mutex mutex_;
+  Timestamp last_remb_time_ RTC_GUARDED_BY(mutex_);
+  DataRate last_send_remb_bitrate_ RTC_GUARDED_BY(mutex_);
+  DataRate max_remb_bitrate_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_CONGESTION_CONTROLLER_REMB_THROTTLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/remb_throttler_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/remb_throttler_unittest.cc
new file mode 100644
index 0000000000..3f8df8a7bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/remb_throttler_unittest.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/remb_throttler.h"
+
+#include <vector>
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::MockFunction;
+
+TEST(RembThrottlerTest, CallRembSenderOnFirstReceiveBitrateChange) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call(12345, std::vector<uint32_t>({1, 2, 3})));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/12345);
+}
+
+TEST(RembThrottlerTest, ThrottlesSmallReceiveBitrateDecrease) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call);
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/12346);
+  clock.AdvanceTime(TimeDelta::Millis(100));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/12345);
+
+  EXPECT_CALL(remb_sender, Call(12345, _));
+  clock.AdvanceTime(TimeDelta::Millis(101));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/12345);
+}
+
+TEST(RembThrottlerTest, DoNotThrottleLargeReceiveBitrateDecrease) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call(2345, _));
+  EXPECT_CALL(remb_sender, Call(1234, _));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/2345);
+  clock.AdvanceTime(TimeDelta::Millis(1));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/1234);
+}
+
+TEST(RembThrottlerTest, ThrottlesReceiveBitrateIncrease) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call);
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/1234);
+  clock.AdvanceTime(TimeDelta::Millis(100));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/2345);
+
+  // Updates 200ms after previous callback is not throttled.
+  EXPECT_CALL(remb_sender, Call(2345, _));
+  clock.AdvanceTime(TimeDelta::Millis(101));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/2345);
+}
+
+TEST(RembThrottlerTest, CallRembSenderOnSetMaxDesiredReceiveBitrate) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+  EXPECT_CALL(remb_sender, Call(1234, _));
+  remb_throttler.SetMaxDesiredReceiveBitrate(DataRate::BitsPerSec(1234));
+}
+
+TEST(RembThrottlerTest, CallRembSenderWithMinOfMaxDesiredAndOnReceivedBitrate) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call(1234, _));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/1234);
+  clock.AdvanceTime(TimeDelta::Millis(1));
+  remb_throttler.SetMaxDesiredReceiveBitrate(DataRate::BitsPerSec(4567));
+
+  clock.AdvanceTime(TimeDelta::Millis(200));
+  EXPECT_CALL(remb_sender, Call(4567, _));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/5678);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn b/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn
new file mode 100644
index 0000000000..cd13332b7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn
@@ -0,0 +1,100 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+config("bwe_test_logging") {
+  if (rtc_enable_bwe_test_logging) {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
+  } else {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
+  }
+}
+
+rtc_library("control_handler") {
+  visibility = [ "*" ]
+  sources = [
+    "control_handler.cc",
+    "control_handler.h",
+  ]
+
+  deps = [
+    "../../../api:sequence_checker",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base/system:no_unique_address",
+    "../../../system_wrappers:field_trial",
+    "../../pacing",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+rtc_library("transport_feedback") {
+  visibility = [ "*" ]
+  sources = [
+    "transport_feedback_adapter.cc",
+    "transport_feedback_adapter.h",
+    "transport_feedback_demuxer.cc",
+    "transport_feedback_demuxer.h",
+  ]
+
+  deps = [
+    "../..:module_api_public",
+    "../../../api:sequence_checker",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_size",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:macromagic",
+    "../../../rtc_base:network_route",
+    "../../../rtc_base:rtc_numerics",
+    "../../../rtc_base/network:sent_packet",
+    "../../../rtc_base/synchronization:mutex",
+    "../../../rtc_base/system:no_unique_address",
+    "../../../system_wrappers",
+    "../../../system_wrappers:field_trial",
+    "../../rtp_rtcp:rtp_rtcp_format",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("congestion_controller_unittests") {
+    testonly = true
+
+    sources = [
+      "transport_feedback_adapter_unittest.cc",
+      "transport_feedback_demuxer_unittest.cc",
+    ]
+    deps = [
+      ":transport_feedback",
+      "../:congestion_controller",
+      "../../../api/transport:network_control",
+      "../../../logging:mocks",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:safe_conversions",
+      "../../../rtc_base/network:sent_packet",
+      "../../../system_wrappers",
+      "../../../test:field_trial",
+      "../../../test:test_support",
+      "../../pacing",
+      "../../remote_bitrate_estimator",
+      "../../rtp_rtcp:rtp_rtcp_format",
+      "//testing/gmock",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc
new file mode 100644
index 0000000000..ffa373aeba
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/rtp/control_handler.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "api/units/data_rate.h"
+#include "modules/pacing/pacing_controller.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+// By default, pacer emergency stops encoder when buffer reaches a high level.
+bool IsPacerEmergencyStopDisabled() {
+  return field_trial::IsEnabled("WebRTC-DisablePacerEmergencyStop");
+}
+
+}  // namespace
+CongestionControlHandler::CongestionControlHandler()
+    : disable_pacer_emergency_stop_(IsPacerEmergencyStopDisabled()) {
+  sequenced_checker_.Detach();
+}
+
+CongestionControlHandler::~CongestionControlHandler() {}
+
+void CongestionControlHandler::SetTargetRate(
+    TargetTransferRate new_target_rate) {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  RTC_CHECK(new_target_rate.at_time.IsFinite());
+  last_incoming_ = new_target_rate;
+}
+
+void CongestionControlHandler::SetNetworkAvailability(bool network_available) {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  network_available_ = network_available;
+}
+
+void CongestionControlHandler::SetPacerQueue(TimeDelta expected_queue_time) {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  pacer_expected_queue_ms_ = expected_queue_time.ms();
+}
+
+absl::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  if (!last_incoming_.has_value())
+    return absl::nullopt;
+  TargetTransferRate new_outgoing = *last_incoming_;
+  DataRate log_target_rate = new_outgoing.target_rate;
+  bool pause_encoding = false;
+  if (!network_available_) {
+    pause_encoding = true;
+  } else if (!disable_pacer_emergency_stop_ &&
+             pacer_expected_queue_ms_ >
+                 PacingController::kMaxExpectedQueueLength.ms()) {
+    pause_encoding = true;
+  }
+  if (pause_encoding)
+    new_outgoing.target_rate = DataRate::Zero();
+  if (!last_reported_ ||
+      last_reported_->target_rate != new_outgoing.target_rate ||
+      (!new_outgoing.target_rate.IsZero() &&
+       (last_reported_->network_estimate.loss_rate_ratio !=
+            new_outgoing.network_estimate.loss_rate_ratio ||
+        last_reported_->network_estimate.round_trip_time !=
+            new_outgoing.network_estimate.round_trip_time))) {
+    if (encoder_paused_in_last_report_ != pause_encoding)
+      RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: "
+                       << ToString(log_target_rate) << ".";
+    encoder_paused_in_last_report_ = pause_encoding;
+    last_reported_ = new_outgoing;
+    return new_outgoing;
+  }
+  return absl::nullopt;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h
new file mode 100644
index 0000000000..d8e7263a02
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
+#define MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace webrtc {
+// This is used to observe the network controller state and route calls to
+// the proper handler. It also keeps cached values for safe asynchronous use.
+// This makes sure that things running on the worker queue can't access state
+// in RtpTransportControllerSend, which would risk causing data race on
+// destruction unless members are properly ordered.
+class CongestionControlHandler {
+ public:
+  CongestionControlHandler();
+  ~CongestionControlHandler();
+
+  CongestionControlHandler(const CongestionControlHandler&) = delete;
+  CongestionControlHandler& operator=(const CongestionControlHandler&) = delete;
+
+  void SetTargetRate(TargetTransferRate new_target_rate);
+  void SetNetworkAvailability(bool network_available);
+  void SetPacerQueue(TimeDelta expected_queue_time);
+  absl::optional<TargetTransferRate> GetUpdate();
+
+ private:
+  absl::optional<TargetTransferRate> last_incoming_;
+  absl::optional<TargetTransferRate> last_reported_;
+  bool network_available_ = true;
+  bool encoder_paused_in_last_report_ = false;
+
+  const bool disable_pacer_emergency_stop_;
+  int64_t pacer_expected_queue_ms_ = 0;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequenced_checker_;
+};
+}  // namespace webrtc
+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build
new file mode 100644
index 0000000000..6ee156d2f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("control_handler_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
new file mode 100644
index 0000000000..e83d09d263
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
@@ -0,0 +1,275 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
+
+#include <stdlib.h>
+
+#include <algorithm>
+#include <cmath>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+constexpr TimeDelta kSendTimeHistoryWindow = TimeDelta::Seconds(60);
+
+void InFlightBytesTracker::AddInFlightPacketBytes(
+    const PacketFeedback& packet) {
+  RTC_DCHECK(packet.sent.send_time.IsFinite());
+  auto it = in_flight_data_.find(packet.network_route);
+  if (it != in_flight_data_.end()) {
+    it->second += packet.sent.size;
+  } else {
+    in_flight_data_.insert({packet.network_route, packet.sent.size});
+  }
+}
+
+void InFlightBytesTracker::RemoveInFlightPacketBytes(
+    const PacketFeedback& packet) {
+  if (packet.sent.send_time.IsInfinite())
+    return;
+  auto it = in_flight_data_.find(packet.network_route);
+  if (it != in_flight_data_.end()) {
+    RTC_DCHECK_GE(it->second, packet.sent.size);
+    it->second -= packet.sent.size;
+    if (it->second.IsZero())
+      in_flight_data_.erase(it);
+  }
+}
+
+DataSize InFlightBytesTracker::GetOutstandingData(
+    const rtc::NetworkRoute& network_route) const {
+  auto it = in_flight_data_.find(network_route);
+  if (it != in_flight_data_.end()) {
+    return it->second;
+  } else {
+    return DataSize::Zero();
+  }
+}
+
+// Comparator for consistent map with NetworkRoute as key.
+bool InFlightBytesTracker::NetworkRouteComparator::operator()(
+    const rtc::NetworkRoute& a,
+    const rtc::NetworkRoute& b) const {
+  if (a.local.network_id() != b.local.network_id())
+    return a.local.network_id() < b.local.network_id();
+  if (a.remote.network_id() != b.remote.network_id())
+    return a.remote.network_id() < b.remote.network_id();
+
+  if (a.local.adapter_id() != b.local.adapter_id())
+    return a.local.adapter_id() < b.local.adapter_id();
+  if (a.remote.adapter_id() != b.remote.adapter_id())
+    return a.remote.adapter_id() < b.remote.adapter_id();
+
+  if (a.local.uses_turn() != b.local.uses_turn())
+    return a.local.uses_turn() < b.local.uses_turn();
+  if (a.remote.uses_turn() != b.remote.uses_turn())
+    return a.remote.uses_turn() < b.remote.uses_turn();
+
+  return a.connected < b.connected;
+}
+
+TransportFeedbackAdapter::TransportFeedbackAdapter() = default;
+
+
+void TransportFeedbackAdapter::AddPacket(const RtpPacketSendInfo& packet_info,
+                                         size_t overhead_bytes,
+                                         Timestamp creation_time) {
+  PacketFeedback packet;
+  packet.creation_time = creation_time;
+  packet.sent.sequence_number =
+      seq_num_unwrapper_.Unwrap(packet_info.transport_sequence_number);
+  packet.sent.size = DataSize::Bytes(packet_info.length + overhead_bytes);
+  packet.sent.audio = packet_info.packet_type == RtpPacketMediaType::kAudio;
+  packet.network_route = network_route_;
+  packet.sent.pacing_info = packet_info.pacing_info;
+
+  while (!history_.empty() &&
+         creation_time - history_.begin()->second.creation_time >
+             kSendTimeHistoryWindow) {
+    // TODO(sprang): Warn if erasing (too many) old items?
+    if (history_.begin()->second.sent.sequence_number > last_ack_seq_num_)
+      in_flight_.RemoveInFlightPacketBytes(history_.begin()->second);
+    history_.erase(history_.begin());
+  }
+  history_.insert(std::make_pair(packet.sent.sequence_number, packet));
+}
+
+absl::optional<SentPacket> TransportFeedbackAdapter::ProcessSentPacket(
+    const rtc::SentPacket& sent_packet) {
+  auto send_time = Timestamp::Millis(sent_packet.send_time_ms);
+  // TODO(srte): Only use one way to indicate that packet feedback is used.
+  if (sent_packet.info.included_in_feedback || sent_packet.packet_id != -1) {
+    int64_t unwrapped_seq_num =
+        seq_num_unwrapper_.Unwrap(sent_packet.packet_id);
+    auto it = history_.find(unwrapped_seq_num);
+    if (it != history_.end()) {
+      bool packet_retransmit = it->second.sent.send_time.IsFinite();
+      it->second.sent.send_time = send_time;
+      last_send_time_ = std::max(last_send_time_, send_time);
+      // TODO(srte): Don't do this on retransmit.
+      if (!pending_untracked_size_.IsZero()) {
+        if (send_time < last_untracked_send_time_)
+          RTC_LOG(LS_WARNING)
+              << "appending acknowledged data for out of order packet. (Diff: "
+              << ToString(last_untracked_send_time_ - send_time) << " ms.)";
+        it->second.sent.prior_unacked_data += pending_untracked_size_;
+        pending_untracked_size_ = DataSize::Zero();
+      }
+      if (!packet_retransmit) {
+        if (it->second.sent.sequence_number > last_ack_seq_num_)
+          in_flight_.AddInFlightPacketBytes(it->second);
+        it->second.sent.data_in_flight = GetOutstandingData();
+        return it->second.sent;
+      }
+    }
+  } else if (sent_packet.info.included_in_allocation) {
+    if (send_time < last_send_time_) {
+      RTC_LOG(LS_WARNING) << "ignoring untracked data for out of order packet.";
+    }
+    pending_untracked_size_ +=
+        DataSize::Bytes(sent_packet.info.packet_size_bytes);
+    last_untracked_send_time_ = std::max(last_untracked_send_time_, send_time);
+  }
+  return absl::nullopt;
+}
+
+absl::optional<TransportPacketsFeedback>
+TransportFeedbackAdapter::ProcessTransportFeedback(
+    const rtcp::TransportFeedback& feedback,
+    Timestamp feedback_receive_time) {
+  if (feedback.GetPacketStatusCount() == 0) {
+    RTC_LOG(LS_INFO) << "Empty transport feedback packet received.";
+    return absl::nullopt;
+  }
+
+  TransportPacketsFeedback msg;
+  msg.feedback_time = feedback_receive_time;
+
+  msg.prior_in_flight = in_flight_.GetOutstandingData(network_route_);
+  msg.packet_feedbacks =
+      ProcessTransportFeedbackInner(feedback, feedback_receive_time);
+  if (msg.packet_feedbacks.empty())
+    return absl::nullopt;
+
+  auto it = history_.find(last_ack_seq_num_);
+  if (it != history_.end()) {
+    msg.first_unacked_send_time = it->second.sent.send_time;
+  }
+  msg.data_in_flight = in_flight_.GetOutstandingData(network_route_);
+
+  return msg;
+}
+
+void TransportFeedbackAdapter::SetNetworkRoute(
+    const rtc::NetworkRoute& network_route) {
+  network_route_ = network_route;
+}
+
+DataSize TransportFeedbackAdapter::GetOutstandingData() const {
+  return in_flight_.GetOutstandingData(network_route_);
+}
+
+std::vector<PacketResult>
+TransportFeedbackAdapter::ProcessTransportFeedbackInner(
+    const rtcp::TransportFeedback& feedback,
+    Timestamp feedback_receive_time) {
+  // Add timestamp deltas to a local time base selected on first packet arrival.
+  // This won't be the true time base, but makes it easier to manually inspect
+  // time stamps.
+  if (last_timestamp_.IsInfinite()) {
+    current_offset_ = feedback_receive_time;
+  } else {
+    // TODO(srte): We shouldn't need to do rounding here.
+    const TimeDelta delta = feedback.GetBaseDelta(last_timestamp_)
+                                .RoundDownTo(TimeDelta::Millis(1));
+    // Protect against assigning current_offset_ negative value.
+    if (delta < Timestamp::Zero() - current_offset_) {
+      RTC_LOG(LS_WARNING) << "Unexpected feedback timestamp received.";
+      current_offset_ = feedback_receive_time;
+    } else {
+      current_offset_ += delta;
+    }
+  }
+  last_timestamp_ = feedback.BaseTime();
+
+  std::vector<PacketResult> packet_result_vector;
+  packet_result_vector.reserve(feedback.GetPacketStatusCount());
+
+  size_t failed_lookups = 0;
+  size_t ignored = 0;
+
+  feedback.ForAllPackets([&](uint16_t sequence_number,
+                             TimeDelta delta_since_base) {
+    int64_t seq_num = seq_num_unwrapper_.Unwrap(sequence_number);
+
+    if (seq_num > last_ack_seq_num_) {
+      // Starts at history_.begin() if last_ack_seq_num_ < 0, since any valid
+      // sequence number is >= 0.
+      for (auto it = history_.upper_bound(last_ack_seq_num_);
+           it != history_.upper_bound(seq_num); ++it) {
+        in_flight_.RemoveInFlightPacketBytes(it->second);
+      }
+      last_ack_seq_num_ = seq_num;
+    }
+
+    auto it = history_.find(seq_num);
+    if (it == history_.end()) {
+      ++failed_lookups;
+      return;
+    }
+
+    if (it->second.sent.send_time.IsInfinite()) {
+      // TODO(srte): Fix the tests that makes this happen and make this a
+      // DCHECK.
+      RTC_DLOG(LS_ERROR)
+          << "Received feedback before packet was indicated as sent";
+      return;
+    }
+
+    PacketFeedback packet_feedback = it->second;
+    if (delta_since_base.IsFinite()) {
+      packet_feedback.receive_time =
+          current_offset_ + delta_since_base.RoundDownTo(TimeDelta::Millis(1));
+      // Note: Lost packets are not removed from history because they might be
+      // reported as received by a later feedback.
+      history_.erase(it);
+    }
+    if (packet_feedback.network_route == network_route_) {
+      PacketResult result;
+      result.sent_packet = packet_feedback.sent;
+      result.receive_time = packet_feedback.receive_time;
+      packet_result_vector.push_back(result);
+    } else {
+      ++ignored;
+    }
+  });
+
+  if (failed_lookups > 0) {
+    RTC_LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
+                        << " packet" << (failed_lookups > 1 ? "s" : "")
+                        << ". Send time history too small?";
+  }
+  if (ignored > 0) {
+    RTC_LOG(LS_INFO) << "Ignoring " << ignored
+                     << " packets because they were sent on a different route.";
+  }
+
+  return packet_result_vector;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h
new file mode 100644
index 0000000000..7b1243b64b
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
+#define MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
+
+#include <deque>
+#include <map>
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "api/transport/network_types.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/network_route.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+struct PacketFeedback {
+  PacketFeedback() = default;
+  // Time corresponding to when this object was created.
+  Timestamp creation_time = Timestamp::MinusInfinity();
+  SentPacket sent;
+  // Time corresponding to when the packet was received. Timestamped with the
+  // receiver's clock. For unreceived packet, Timestamp::PlusInfinity() is
+  // used.
+  Timestamp receive_time = Timestamp::PlusInfinity();
+
+  // The network route that this packet is associated with.
+  rtc::NetworkRoute network_route;
+};
+
+class InFlightBytesTracker {
+ public:
+  void AddInFlightPacketBytes(const PacketFeedback& packet);
+  void RemoveInFlightPacketBytes(const PacketFeedback& packet);
+  DataSize GetOutstandingData(const rtc::NetworkRoute& network_route) const;
+
+ private:
+  struct NetworkRouteComparator {
+    bool operator()(const rtc::NetworkRoute& a,
+                    const rtc::NetworkRoute& b) const;
+  };
+  std::map<rtc::NetworkRoute, DataSize, NetworkRouteComparator> in_flight_data_;
+};
+
+class TransportFeedbackAdapter {
+ public:
+  TransportFeedbackAdapter();
+
+  void AddPacket(const RtpPacketSendInfo& packet_info,
+                 size_t overhead_bytes,
+                 Timestamp creation_time);
+  absl::optional<SentPacket> ProcessSentPacket(
+      const rtc::SentPacket& sent_packet);
+
+  absl::optional<TransportPacketsFeedback> ProcessTransportFeedback(
+      const rtcp::TransportFeedback& feedback,
+      Timestamp feedback_receive_time);
+
+  void SetNetworkRoute(const rtc::NetworkRoute& network_route);
+
+  DataSize GetOutstandingData() const;
+
+ private:
+  enum class SendTimeHistoryStatus { kNotAdded, kOk, kDuplicate };
+
+  std::vector<PacketResult> ProcessTransportFeedbackInner(
+      const rtcp::TransportFeedback& feedback,
+      Timestamp feedback_receive_time);
+
+  DataSize pending_untracked_size_ = DataSize::Zero();
+  Timestamp last_send_time_ = Timestamp::MinusInfinity();
+  Timestamp last_untracked_send_time_ = Timestamp::MinusInfinity();
+  RtpSequenceNumberUnwrapper seq_num_unwrapper_;
+  std::map<int64_t, PacketFeedback> history_;
+
+  // Sequence numbers are never negative, using -1 as it always < a real
+  // sequence number.
+  int64_t last_ack_seq_num_ = -1;
+  InFlightBytesTracker in_flight_;
+
+  Timestamp current_offset_ = Timestamp::MinusInfinity();
+  Timestamp last_timestamp_ = Timestamp::MinusInfinity();
+
+  rtc::NetworkRoute network_route_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
new file mode 100644
index 0000000000..14a2b13831
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
@@ -0,0 +1,407 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
+
+#include <limits>
+#include <memory>
+#include <vector>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/clock.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+
+namespace webrtc {
+
+namespace {
+constexpr uint32_t kSsrc = 8492;
+const PacedPacketInfo kPacingInfo0(0, 5, 2000);
+const PacedPacketInfo kPacingInfo1(1, 8, 4000);
+const PacedPacketInfo kPacingInfo2(2, 14, 7000);
+const PacedPacketInfo kPacingInfo3(3, 20, 10000);
+const PacedPacketInfo kPacingInfo4(4, 22, 10000);
+
+void ComparePacketFeedbackVectors(const std::vector<PacketResult>& truth,
+                                  const std::vector<PacketResult>& input) {
+  ASSERT_EQ(truth.size(), input.size());
+  size_t len = truth.size();
+  // truth contains the input data for the test, and input is what will be
+  // sent to the bandwidth estimator. truth.arrival_tims_ms is used to
+  // populate the transport feedback messages. As these times may be changed
+  // (because of resolution limits in the packets, and because of the time
+  // base adjustment performed by the TransportFeedbackAdapter at the first
+  // packet, the truth[x].arrival_time and input[x].arrival_time may not be
+  // equal. However, the difference must be the same for all x.
+  TimeDelta arrival_time_delta = truth[0].receive_time - input[0].receive_time;
+  for (size_t i = 0; i < len; ++i) {
+    RTC_CHECK(truth[i].IsReceived());
+    if (input[i].IsReceived()) {
+      EXPECT_EQ(truth[i].receive_time - input[i].receive_time,
+                arrival_time_delta);
+    }
+    EXPECT_EQ(truth[i].sent_packet.send_time, input[i].sent_packet.send_time);
+    EXPECT_EQ(truth[i].sent_packet.sequence_number,
+              input[i].sent_packet.sequence_number);
+    EXPECT_EQ(truth[i].sent_packet.size, input[i].sent_packet.size);
+    EXPECT_EQ(truth[i].sent_packet.pacing_info,
+              input[i].sent_packet.pacing_info);
+  }
+}
+
+PacketResult CreatePacket(int64_t receive_time_ms,
+                          int64_t send_time_ms,
+                          int64_t sequence_number,
+                          size_t payload_size,
+                          const PacedPacketInfo& pacing_info) {
+  PacketResult res;
+  res.receive_time = Timestamp::Millis(receive_time_ms);
+  res.sent_packet.send_time = Timestamp::Millis(send_time_ms);
+  res.sent_packet.sequence_number = sequence_number;
+  res.sent_packet.size = DataSize::Bytes(payload_size);
+  res.sent_packet.pacing_info = pacing_info;
+  return res;
+}
+
+class MockStreamFeedbackObserver : public webrtc::StreamFeedbackObserver {
+ public:
+  MOCK_METHOD(void,
+              OnPacketFeedbackVector,
+              (std::vector<StreamPacketInfo> packet_feedback_vector),
+              (override));
+};
+
+}  // namespace
+
+class TransportFeedbackAdapterTest : public ::testing::Test {
+ public:
+  TransportFeedbackAdapterTest() : clock_(0) {}
+
+  virtual ~TransportFeedbackAdapterTest() {}
+
+  virtual void SetUp() { adapter_.reset(new TransportFeedbackAdapter()); }
+
+  virtual void TearDown() { adapter_.reset(); }
+
+ protected:
+  void OnReceivedEstimatedBitrate(uint32_t bitrate) {}
+
+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
+                                    int64_t rtt,
+                                    int64_t now_ms) {}
+
+  void OnSentPacket(const PacketResult& packet_feedback) {
+    RtpPacketSendInfo packet_info;
+    packet_info.media_ssrc = kSsrc;
+    packet_info.transport_sequence_number =
+        packet_feedback.sent_packet.sequence_number;
+    packet_info.rtp_sequence_number = 0;
+    packet_info.length = packet_feedback.sent_packet.size.bytes();
+    packet_info.pacing_info = packet_feedback.sent_packet.pacing_info;
+    packet_info.packet_type = RtpPacketMediaType::kVideo;
+    adapter_->AddPacket(RtpPacketSendInfo(packet_info), 0u,
+                        clock_.CurrentTime());
+    adapter_->ProcessSentPacket(rtc::SentPacket(
+        packet_feedback.sent_packet.sequence_number,
+        packet_feedback.sent_packet.send_time.ms(), rtc::PacketInfo()));
+  }
+
+  SimulatedClock clock_;
+  std::unique_ptr<TransportFeedbackAdapter> adapter_;
+};
+
+TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
+  std::vector<PacketResult> packets;
+  packets.push_back(CreatePacket(100, 200, 0, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(110, 210, 1, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(120, 220, 2, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(130, 230, 3, 1500, kPacingInfo1));
+  packets.push_back(CreatePacket(140, 240, 4, 1500, kPacingInfo1));
+
+  for (const auto& packet : packets)
+    OnSentPacket(packet);
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(packets[0].sent_packet.sequence_number,
+                   packets[0].receive_time);
+
+  for (const auto& packet : packets) {
+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sent_packet.sequence_number,
+                                           packet.receive_time));
+  }
+
+  feedback.Build();
+
+  auto result =
+      adapter_->ProcessTransportFeedback(feedback, clock_.CurrentTime());
+  ComparePacketFeedbackVectors(packets, result->packet_feedbacks);
+}
+
+TEST_F(TransportFeedbackAdapterTest, FeedbackVectorReportsUnreceived) {
+  std::vector<PacketResult> sent_packets = {
+      CreatePacket(100, 220, 0, 1500, kPacingInfo0),
+      CreatePacket(110, 210, 1, 1500, kPacingInfo0),
+      CreatePacket(120, 220, 2, 1500, kPacingInfo0),
+      CreatePacket(130, 230, 3, 1500, kPacingInfo0),
+      CreatePacket(140, 240, 4, 1500, kPacingInfo0),
+      CreatePacket(150, 250, 5, 1500, kPacingInfo0),
+      CreatePacket(160, 260, 6, 1500, kPacingInfo0)};
+
+  for (const auto& packet : sent_packets)
+    OnSentPacket(packet);
+
+  // Note: Important to include the last packet, as only unreceived packets in
+  // between received packets can be inferred.
+  std::vector<PacketResult> received_packets = {
+      sent_packets[0], sent_packets[2], sent_packets[6]};
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(received_packets[0].sent_packet.sequence_number,
+                   received_packets[0].receive_time);
+
+  for (const auto& packet : received_packets) {
+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sent_packet.sequence_number,
+                                           packet.receive_time));
+  }
+
+  feedback.Build();
+
+  auto res = adapter_->ProcessTransportFeedback(feedback, clock_.CurrentTime());
+  ComparePacketFeedbackVectors(sent_packets, res->packet_feedbacks);
+}
+
+TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
+  std::vector<PacketResult> packets;
+  packets.push_back(CreatePacket(100, 200, 0, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(110, 210, 1, 1500, kPacingInfo1));
+  packets.push_back(CreatePacket(120, 220, 2, 1500, kPacingInfo2));
+  packets.push_back(CreatePacket(130, 230, 3, 1500, kPacingInfo3));
+  packets.push_back(CreatePacket(140, 240, 4, 1500, kPacingInfo4));
+
+  const uint16_t kSendSideDropBefore = 1;
+  const uint16_t kReceiveSideDropAfter = 3;
+
+  for (const auto& packet : packets) {
+    if (packet.sent_packet.sequence_number >= kSendSideDropBefore)
+      OnSentPacket(packet);
+  }
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(packets[0].sent_packet.sequence_number,
+                   packets[0].receive_time);
+
+  for (const auto& packet : packets) {
+    if (packet.sent_packet.sequence_number <= kReceiveSideDropAfter) {
+      EXPECT_TRUE(feedback.AddReceivedPacket(packet.sent_packet.sequence_number,
+                                             packet.receive_time));
+    }
+  }
+
+  feedback.Build();
+
+  std::vector<PacketResult> expected_packets(
+      packets.begin() + kSendSideDropBefore,
+      packets.begin() + kReceiveSideDropAfter + 1);
+  // Packets that have timed out on the send-side have lost the
+  // information stored on the send-side. And they will not be reported to
+  // observers since we won't know that they come from the same networks.
+
+  auto res = adapter_->ProcessTransportFeedback(feedback, clock_.CurrentTime());
+  ComparePacketFeedbackVectors(expected_packets, res->packet_feedbacks);
+}
+
+TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
+  TimeDelta kHighArrivalTime =
+      rtcp::TransportFeedback::kDeltaTick * (1 << 8) * ((1 << 23) - 1);
+  std::vector<PacketResult> packets;
+  packets.push_back(CreatePacket(kHighArrivalTime.ms() + 64, 210, 0, 1500,
+                                 PacedPacketInfo()));
+  packets.push_back(CreatePacket(kHighArrivalTime.ms() - 64, 210, 1, 1500,
+                                 PacedPacketInfo()));
+  packets.push_back(
+      CreatePacket(kHighArrivalTime.ms(), 220, 2, 1500, PacedPacketInfo()));
+
+  for (const auto& packet : packets)
+    OnSentPacket(packet);
+
+  for (size_t i = 0; i < packets.size(); ++i) {
+    std::unique_ptr<rtcp::TransportFeedback> feedback(
+        new rtcp::TransportFeedback());
+    feedback->SetBase(packets[i].sent_packet.sequence_number,
+                      packets[i].receive_time);
+
+    EXPECT_TRUE(feedback->AddReceivedPacket(
+        packets[i].sent_packet.sequence_number, packets[i].receive_time));
+
+    rtc::Buffer raw_packet = feedback->Build();
+    feedback = rtcp::TransportFeedback::ParseFrom(raw_packet.data(),
+                                                  raw_packet.size());
+
+    std::vector<PacketResult> expected_packets;
+    expected_packets.push_back(packets[i]);
+
+    auto res = adapter_->ProcessTransportFeedback(*feedback.get(),
+                                                  clock_.CurrentTime());
+    ComparePacketFeedbackVectors(expected_packets, res->packet_feedbacks);
+  }
+}
+
+TEST_F(TransportFeedbackAdapterTest, HandlesArrivalReordering) {
+  std::vector<PacketResult> packets;
+  packets.push_back(CreatePacket(120, 200, 0, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(110, 210, 1, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(100, 220, 2, 1500, kPacingInfo0));
+
+  for (const auto& packet : packets)
+    OnSentPacket(packet);
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(packets[0].sent_packet.sequence_number,
+                   packets[0].receive_time);
+
+  for (const auto& packet : packets) {
+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sent_packet.sequence_number,
+                                           packet.receive_time));
+  }
+
+  feedback.Build();
+
+  // Adapter keeps the packets ordered by sequence number (which is itself
+  // assigned by the order of transmission). Reordering by some other criteria,
+  // eg. arrival time, is up to the observers.
+  auto res = adapter_->ProcessTransportFeedback(feedback, clock_.CurrentTime());
+  ComparePacketFeedbackVectors(packets, res->packet_feedbacks);
+}
+
+TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
+  std::vector<PacketResult> sent_packets;
+  // TODO(srte): Consider using us resolution in the constants.
+  const TimeDelta kSmallDelta = (rtcp::TransportFeedback::kDeltaTick * 0xFF)
+                                    .RoundDownTo(TimeDelta::Millis(1));
+  const TimeDelta kLargePositiveDelta = (rtcp::TransportFeedback::kDeltaTick *
+                                         std::numeric_limits<int16_t>::max())
+                                            .RoundDownTo(TimeDelta::Millis(1));
+  const TimeDelta kLargeNegativeDelta = (rtcp::TransportFeedback::kDeltaTick *
+                                         std::numeric_limits<int16_t>::min())
+                                            .RoundDownTo(TimeDelta::Millis(1));
+
+  PacketResult packet_feedback;
+  packet_feedback.sent_packet.sequence_number = 1;
+  packet_feedback.sent_packet.send_time = Timestamp::Millis(100);
+  packet_feedback.receive_time = Timestamp::Millis(200);
+  packet_feedback.sent_packet.size = DataSize::Bytes(1500);
+  sent_packets.push_back(packet_feedback);
+
+  // TODO(srte): This rounding maintains previous behavior, but should ot be
+  // required.
+  packet_feedback.sent_packet.send_time += kSmallDelta;
+  packet_feedback.receive_time += kSmallDelta;
+  ++packet_feedback.sent_packet.sequence_number;
+  sent_packets.push_back(packet_feedback);
+
+  packet_feedback.sent_packet.send_time += kLargePositiveDelta;
+  packet_feedback.receive_time += kLargePositiveDelta;
+  ++packet_feedback.sent_packet.sequence_number;
+  sent_packets.push_back(packet_feedback);
+
+  packet_feedback.sent_packet.send_time += kLargeNegativeDelta;
+  packet_feedback.receive_time += kLargeNegativeDelta;
+  ++packet_feedback.sent_packet.sequence_number;
+  sent_packets.push_back(packet_feedback);
+
+  // Too large, delta - will need two feedback messages.
+  packet_feedback.sent_packet.send_time +=
+      kLargePositiveDelta + TimeDelta::Millis(1);
+  packet_feedback.receive_time += kLargePositiveDelta + TimeDelta::Millis(1);
+  ++packet_feedback.sent_packet.sequence_number;
+
+  // Packets will be added to send history.
+  for (const auto& packet : sent_packets)
+    OnSentPacket(packet);
+  OnSentPacket(packet_feedback);
+
+  // Create expected feedback and send into adapter.
+  std::unique_ptr<rtcp::TransportFeedback> feedback(
+      new rtcp::TransportFeedback());
+  feedback->SetBase(sent_packets[0].sent_packet.sequence_number,
+                    sent_packets[0].receive_time);
+
+  for (const auto& packet : sent_packets) {
+    EXPECT_TRUE(feedback->AddReceivedPacket(packet.sent_packet.sequence_number,
+                                            packet.receive_time));
+  }
+  EXPECT_FALSE(
+      feedback->AddReceivedPacket(packet_feedback.sent_packet.sequence_number,
+                                  packet_feedback.receive_time));
+
+  rtc::Buffer raw_packet = feedback->Build();
+  feedback =
+      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size());
+
+  std::vector<PacketResult> received_feedback;
+
+  EXPECT_TRUE(feedback.get() != nullptr);
+  auto res =
+      adapter_->ProcessTransportFeedback(*feedback.get(), clock_.CurrentTime());
+  ComparePacketFeedbackVectors(sent_packets, res->packet_feedbacks);
+
+  // Create a new feedback message and add the trailing item.
+  feedback.reset(new rtcp::TransportFeedback());
+  feedback->SetBase(packet_feedback.sent_packet.sequence_number,
+                    packet_feedback.receive_time);
+  EXPECT_TRUE(
+      feedback->AddReceivedPacket(packet_feedback.sent_packet.sequence_number,
+                                  packet_feedback.receive_time));
+  raw_packet = feedback->Build();
+  feedback =
+      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size());
+
+  EXPECT_TRUE(feedback.get() != nullptr);
+  {
+    auto res = adapter_->ProcessTransportFeedback(*feedback.get(),
+                                                  clock_.CurrentTime());
+    std::vector<PacketResult> expected_packets;
+    expected_packets.push_back(packet_feedback);
+    ComparePacketFeedbackVectors(expected_packets, res->packet_feedbacks);
+  }
+}
+
+TEST_F(TransportFeedbackAdapterTest, IgnoreDuplicatePacketSentCalls) {
+  auto packet = CreatePacket(100, 200, 0, 1500, kPacingInfo0);
+
+  // Add a packet and then mark it as sent.
+  RtpPacketSendInfo packet_info;
+  packet_info.media_ssrc = kSsrc;
+  packet_info.transport_sequence_number = packet.sent_packet.sequence_number;
+  packet_info.length = packet.sent_packet.size.bytes();
+  packet_info.pacing_info = packet.sent_packet.pacing_info;
+  packet_info.packet_type = RtpPacketMediaType::kVideo;
+  adapter_->AddPacket(packet_info, 0u, clock_.CurrentTime());
+  absl::optional<SentPacket> sent_packet = adapter_->ProcessSentPacket(
+      rtc::SentPacket(packet.sent_packet.sequence_number,
+                      packet.sent_packet.send_time.ms(), rtc::PacketInfo()));
+  EXPECT_TRUE(sent_packet.has_value());
+
+  // Call ProcessSentPacket() again with the same sequence number. This packet
+  // has already been marked as sent and the call should be ignored.
+  absl::optional<SentPacket> duplicate_packet = adapter_->ProcessSentPacket(
+      rtc::SentPacket(packet.sent_packet.sequence_number,
+                      packet.sent_packet.send_time.ms(), rtc::PacketInfo()));
+  EXPECT_FALSE(duplicate_packet.has_value());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.cc
new file mode 100644
index 0000000000..469c21434a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.cc
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/rtp/transport_feedback_demuxer.h"
+#include "absl/algorithm/container.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+
+namespace webrtc {
+namespace {
+static const size_t kMaxPacketsInHistory = 5000;
+}
+
+TransportFeedbackDemuxer::TransportFeedbackDemuxer() {
+  // In case the construction thread is different from where the registration
+  // and callbacks occur, detach from the construction thread.
+  observer_checker_.Detach();
+}
+
+void TransportFeedbackDemuxer::RegisterStreamFeedbackObserver(
+    std::vector<uint32_t> ssrcs,
+    StreamFeedbackObserver* observer) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+  RTC_DCHECK(observer);
+  RTC_DCHECK(absl::c_find_if(observers_, [=](const auto& pair) {
+               return pair.second == observer;
+             }) == observers_.end());
+  observers_.push_back({ssrcs, observer});
+}
+
+void TransportFeedbackDemuxer::DeRegisterStreamFeedbackObserver(
+    StreamFeedbackObserver* observer) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+  RTC_DCHECK(observer);
+  const auto it = absl::c_find_if(
+      observers_, [=](const auto& pair) { return pair.second == observer; });
+  RTC_DCHECK(it != observers_.end());
+  observers_.erase(it);
+}
+
+void TransportFeedbackDemuxer::AddPacket(const RtpPacketSendInfo& packet_info) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+
+  StreamFeedbackObserver::StreamPacketInfo info;
+  info.ssrc = packet_info.media_ssrc;
+  info.rtp_sequence_number = packet_info.rtp_sequence_number;
+  info.received = false;
+  info.is_retransmission =
+      packet_info.packet_type == RtpPacketMediaType::kRetransmission;
+  history_.insert(
+      {seq_num_unwrapper_.Unwrap(packet_info.transport_sequence_number), info});
+
+  while (history_.size() > kMaxPacketsInHistory) {
+    history_.erase(history_.begin());
+  }
+}
+
+void TransportFeedbackDemuxer::OnTransportFeedback(
+    const rtcp::TransportFeedback& feedback) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+
+  std::vector<StreamFeedbackObserver::StreamPacketInfo> stream_feedbacks;
+  feedback.ForAllPackets(
+      [&](uint16_t sequence_number, TimeDelta delta_since_base) {
+        RTC_DCHECK_RUN_ON(&observer_checker_);
+        auto it = history_.find(seq_num_unwrapper_.PeekUnwrap(sequence_number));
+        if (it != history_.end()) {
+          auto packet_info = it->second;
+          packet_info.received = delta_since_base.IsFinite();
+          stream_feedbacks.push_back(std::move(packet_info));
+          if (delta_since_base.IsFinite())
+            history_.erase(it);
+        }
+      });
+
+  for (auto& observer : observers_) {
+    std::vector<StreamFeedbackObserver::StreamPacketInfo> selected_feedback;
+    for (const auto& packet_info : stream_feedbacks) {
+      if (absl::c_count(observer.first, packet_info.ssrc) > 0) {
+        selected_feedback.push_back(packet_info);
+      }
+    }
+    if (!selected_feedback.empty()) {
+      observer.second->OnPacketFeedbackVector(std::move(selected_feedback));
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h
new file mode 100644
index 0000000000..278c144b61
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_
+#define MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_
+
+#include <map>
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace webrtc {
+
+// Implementation of StreamFeedbackProvider that provides a way for
+// implementations of StreamFeedbackObserver to register for feedback callbacks
+// for a given set of SSRCs.
+// Registration methods need to be called from the same execution context
+// (thread or task queue) and callbacks to
+// StreamFeedbackObserver::OnPacketFeedbackVector will be made in that same
+// context.
+// TODO(tommi): This appears to be the only implementation of this interface.
+// Do we need the interface?
+class TransportFeedbackDemuxer final : public StreamFeedbackProvider {
+ public:
+  TransportFeedbackDemuxer();
+
+  // Implements StreamFeedbackProvider interface
+  void RegisterStreamFeedbackObserver(
+      std::vector<uint32_t> ssrcs,
+      StreamFeedbackObserver* observer) override;
+  void DeRegisterStreamFeedbackObserver(
+      StreamFeedbackObserver* observer) override;
+  void AddPacket(const RtpPacketSendInfo& packet_info);
+  void OnTransportFeedback(const rtcp::TransportFeedback& feedback);
+
+ private:
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker observer_checker_;
+  RtpSequenceNumberUnwrapper seq_num_unwrapper_
+      RTC_GUARDED_BY(&observer_checker_);
+  std::map<int64_t, StreamFeedbackObserver::StreamPacketInfo> history_
+      RTC_GUARDED_BY(&observer_checker_);
+
+  // Maps a set of ssrcs to corresponding observer. Vectors are used rather than
+  // set/map to ensure that the processing order is consistent independently of
+  // the randomized ssrcs.
+  std::vector<std::pair<std::vector<uint32_t>, StreamFeedbackObserver*>>
+      observers_ RTC_GUARDED_BY(&observer_checker_);
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc
new file mode 100644
index 0000000000..52d8018bff
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/rtp/transport_feedback_demuxer.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::AllOf;
+using ::testing::ElementsAre;
+using ::testing::Field;
+using PacketInfo = StreamFeedbackObserver::StreamPacketInfo;
+
+static constexpr uint32_t kSsrc = 8492;
+
+class MockStreamFeedbackObserver : public webrtc::StreamFeedbackObserver {
+ public:
+  MOCK_METHOD(void,
+              OnPacketFeedbackVector,
+              (std::vector<StreamPacketInfo> packet_feedback_vector),
+              (override));
+};
+
+RtpPacketSendInfo CreatePacket(uint32_t ssrc,
+                               uint16_t rtp_sequence_number,
+                               int64_t transport_sequence_number,
+                               bool is_retransmission) {
+  RtpPacketSendInfo res;
+  res.media_ssrc = ssrc;
+  res.transport_sequence_number = transport_sequence_number;
+  res.rtp_sequence_number = rtp_sequence_number;
+  res.packet_type = is_retransmission ? RtpPacketMediaType::kRetransmission
+                                      : RtpPacketMediaType::kVideo;
+  return res;
+}
+}  // namespace
+
+TEST(TransportFeedbackDemuxerTest, ObserverSanity) {
+  TransportFeedbackDemuxer demuxer;
+  MockStreamFeedbackObserver mock;
+  demuxer.RegisterStreamFeedbackObserver({kSsrc}, &mock);
+
+  const uint16_t kRtpStartSeq = 55;
+  const int64_t kTransportStartSeq = 1;
+  demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq, kTransportStartSeq,
+                                 /*is_retransmit=*/false));
+  demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq + 1,
+                                 kTransportStartSeq + 1,
+                                 /*is_retransmit=*/false));
+  demuxer.AddPacket(CreatePacket(
+      kSsrc, kRtpStartSeq + 2, kTransportStartSeq + 2, /*is_retransmit=*/true));
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(kTransportStartSeq, Timestamp::Millis(1));
+  ASSERT_TRUE(
+      feedback.AddReceivedPacket(kTransportStartSeq, Timestamp::Millis(1)));
+  // Drop middle packet.
+  ASSERT_TRUE(
+      feedback.AddReceivedPacket(kTransportStartSeq + 2, Timestamp::Millis(3)));
+
+  EXPECT_CALL(
+      mock, OnPacketFeedbackVector(ElementsAre(
+                AllOf(Field(&PacketInfo::received, true),
+                      Field(&PacketInfo::ssrc, kSsrc),
+                      Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq),
+                      Field(&PacketInfo::is_retransmission, false)),
+                AllOf(Field(&PacketInfo::received, false),
+                      Field(&PacketInfo::ssrc, kSsrc),
+                      Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 1),
+                      Field(&PacketInfo::is_retransmission, false)),
+                AllOf(Field(&PacketInfo::received, true),
+                      Field(&PacketInfo::ssrc, kSsrc),
+                      Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 2),
+                      Field(&PacketInfo::is_retransmission, true)))));
+  demuxer.OnTransportFeedback(feedback);
+
+  demuxer.DeRegisterStreamFeedbackObserver(&mock);
+
+  demuxer.AddPacket(
+      CreatePacket(kSsrc, kRtpStartSeq + 3, kTransportStartSeq + 3, false));
+  rtcp::TransportFeedback second_feedback;
+  second_feedback.SetBase(kTransportStartSeq + 3, Timestamp::Millis(4));
+  ASSERT_TRUE(second_feedback.AddReceivedPacket(kTransportStartSeq + 3,
+                                                Timestamp::Millis(4)));
+
+  EXPECT_CALL(mock, OnPacketFeedbackVector).Times(0);
+  demuxer.OnTransportFeedback(second_feedback);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build
new file mode 100644
index 0000000000..eb5a2a87e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("transport_feedback_gn")
diff --git a/third_party/libwebrtc/modules/desktop_capture/BUILD.gn b/third_party/libwebrtc/modules/desktop_capture/BUILD.gn
new file mode 100644
index 0000000000..060d4e8200
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/BUILD.gn
@@ -0,0 +1,652 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//build/config/linux/gtk/gtk.gni")
+import("//build/config/ui.gni")
+import("../../webrtc.gni")
+
+use_desktop_capture_differ_sse2 = target_cpu == "x86" || target_cpu == "x64"
+
+config("x11_config") {
+  if (rtc_use_x11_extensions) {
+    defines = [ "WEBRTC_USE_X11" ]
+  }
+}
+
+rtc_library("primitives") {
+  visibility = [ "*" ]
+  sources = [
+    "desktop_capture_types.h",
+    "desktop_frame.cc",
+    "desktop_frame.h",
+    "desktop_geometry.cc",
+    "desktop_geometry.h",
+    "desktop_region.cc",
+    "desktop_region.h",
+    "shared_desktop_frame.cc",
+    "shared_desktop_frame.h",
+    "shared_memory.cc",
+    "shared_memory.h",
+  ]
+
+  deps = [
+    "../../api:scoped_refptr",
+    "../../rtc_base:checks",
+    "../../rtc_base:refcount",
+    "../../rtc_base/system:rtc_export",
+    "//third_party/libyuv",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("desktop_capture_modules_tests") {
+    testonly = true
+
+    defines = []
+    sources = []
+    deps = [
+      ":desktop_capture",
+      "../../api:function_view",
+      "../../api:scoped_refptr",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:platform_thread",
+      "../../rtc_base:random",
+      "../../rtc_base:timeutils",
+    ]
+    if (rtc_desktop_capture_supported) {
+      deps += [
+        ":desktop_capture_mock",
+        ":primitives",
+        ":screen_drawer",
+        "../../rtc_base/third_party/base64",
+        "../../system_wrappers",
+        "../../test:test_support",
+        "../../test:video_test_support",
+      ]
+      sources += [
+        "screen_capturer_integration_test.cc",
+        "screen_drawer_unittest.cc",
+        "window_finder_unittest.cc",
+      ]
+
+      if ((is_linux || is_chromeos) && rtc_use_pipewire) {
+        configs += [ "../portal:gio" ]
+      }
+
+      public_configs = [ ":x11_config" ]
+
+      if (is_win) {
+        deps += [ "../../rtc_base/win:windows_version" ]
+      }
+    }
+  }
+
+  if ((is_linux || is_chromeos) && rtc_use_pipewire) {
+    rtc_test("shared_screencast_stream_test") {
+      testonly = true
+
+      sources = [
+        "linux/wayland/shared_screencast_stream_unittest.cc",
+        "linux/wayland/test/test_screencast_stream_provider.cc",
+        "linux/wayland/test/test_screencast_stream_provider.h",
+      ]
+
+      configs += [ "../portal:pipewire_all" ]
+
+      deps = [
+        ":desktop_capture",
+        ":desktop_capture_mock",
+        ":primitives",
+        "../../rtc_base:checks",
+        "../../rtc_base:logging",
+        "../../rtc_base:random",
+        "../../rtc_base:timeutils",
+        "../portal",
+
+        # TODO(bugs.webrtc.org/9987): Remove this dep on rtc_base:rtc_base once
+        # rtc_base:threading is fully defined.
+        "../../rtc_base:task_queue_for_test",
+        "../../rtc_base:threading",
+        "../../system_wrappers",
+        "../../test:test_main",
+        "../../test:test_support",
+        "//api/units:time_delta",
+        "//rtc_base:rtc_event",
+      ]
+
+      data = [ "../../third_party/pipewire" ]
+      public_configs = [ "../portal:pipewire_config" ]
+    }
+  }
+
+  rtc_library("desktop_capture_unittests") {
+    testonly = true
+
+    defines = []
+    sources = [
+      "blank_detector_desktop_capturer_wrapper_unittest.cc",
+      "cropped_desktop_frame_unittest.cc",
+      "desktop_and_cursor_composer_unittest.cc",
+      "desktop_capturer_differ_wrapper_unittest.cc",
+      "desktop_frame_rotation_unittest.cc",
+      "desktop_frame_unittest.cc",
+      "desktop_geometry_unittest.cc",
+      "desktop_region_unittest.cc",
+      "differ_block_unittest.cc",
+      "fallback_desktop_capturer_wrapper_unittest.cc",
+      "mouse_cursor_monitor_unittest.cc",
+      "rgba_color_unittest.cc",
+      "test_utils.cc",
+      "test_utils.h",
+      "test_utils_unittest.cc",
+    ]
+
+    if ((is_linux || is_chromeos) && rtc_use_pipewire) {
+      configs += [ "../portal:gio" ]
+    }
+
+    deps = [
+      ":desktop_capture",
+      ":desktop_capture_mock",
+      ":primitives",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:random",
+      "../../rtc_base:timeutils",
+
+      # TODO(bugs.webrtc.org/9987): Remove this dep on rtc_base:rtc_base once
+      # rtc_base:threading is fully defined.
+      "../../rtc_base:task_queue_for_test",
+      "../../rtc_base:threading",
+      "../../system_wrappers",
+      "../../test:test_support",
+    ]
+
+    if (is_win) {
+      sources += [
+        "win/cursor_unittest.cc",
+        "win/cursor_unittest_resources.h",
+        "win/cursor_unittest_resources.rc",
+        "win/screen_capture_utils_unittest.cc",
+        "win/screen_capturer_win_directx_unittest.cc",
+        "win/test_support/test_window.cc",
+        "win/test_support/test_window.h",
+        "win/window_capture_utils_unittest.cc",
+      ]
+      deps += [
+        "../../rtc_base/win:scoped_com_initializer",
+        "../../rtc_base/win:windows_version",
+      ]
+    }
+
+    if (rtc_desktop_capture_supported) {
+      sources += [
+        "screen_capturer_helper_unittest.cc",
+        "screen_capturer_unittest.cc",
+        "window_capturer_unittest.cc",
+      ]
+      if (is_mac) {
+        sources += [ "screen_capturer_mac_unittest.cc" ]
+      }
+      if (rtc_enable_win_wgc) {
+        sources += [
+          "win/wgc_capture_source_unittest.cc",
+          "win/wgc_capturer_win_unittest.cc",
+        ]
+      }
+      deps += [
+        ":desktop_capture_mock",
+        "../../system_wrappers:metrics",
+      ]
+      public_configs = [ ":x11_config" ]
+    }
+  }
+
+  rtc_library("screen_drawer") {
+    testonly = true
+
+    sources = [
+      "screen_drawer.cc",
+      "screen_drawer.h",
+    ]
+
+    if (is_linux || is_chromeos) {
+      sources += [ "screen_drawer_linux.cc" ]
+      libs = [ "X11" ]
+    }
+
+    if (is_mac) {
+      sources += [ "screen_drawer_mac.cc" ]
+    }
+
+    if (is_win) {
+      sources += [ "screen_drawer_win.cc" ]
+    }
+
+    deps = [
+      ":desktop_capture",
+      ":primitives",
+      "../../api:scoped_refptr",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../system_wrappers",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+
+    if (is_posix || is_fuchsia) {
+      sources += [
+        "screen_drawer_lock_posix.cc",
+        "screen_drawer_lock_posix.h",
+      ]
+    }
+  }
+
+  rtc_library("desktop_capture_mock") {
+    testonly = true
+
+    sources = [
+      "mock_desktop_capturer_callback.cc",
+      "mock_desktop_capturer_callback.h",
+    ]
+
+    if ((is_linux || is_chromeos) && rtc_use_pipewire) {
+      configs += [ "../portal:gio" ]
+    }
+
+    deps = [
+      ":desktop_capture",
+      ":primitives",
+      "../../test:test_support",
+    ]
+  }
+}
+
+# TODO(bugs.webrtc.org/14187): remove when all users are gone
+if ((is_linux || is_chromeos) && rtc_use_pipewire) {
+  config("pipewire_config") {
+    configs = [ "../portal:pipewire_config" ]
+  }
+}
+
+rtc_library("desktop_capture") {
+  visibility = [ "*" ]
+  defines = []
+  deps = []
+  public_configs = [ ":x11_config" ]
+  sources = [
+    "blank_detector_desktop_capturer_wrapper.cc",
+    "blank_detector_desktop_capturer_wrapper.h",
+    "cropped_desktop_frame.cc",
+    "cropped_desktop_frame.h",
+    "cropping_window_capturer.cc",
+    "cropping_window_capturer.h",
+    "delegated_source_list_controller.h",
+    "desktop_and_cursor_composer.cc",
+    "desktop_and_cursor_composer.h",
+    "desktop_capture_metrics_helper.cc",
+    "desktop_capture_metrics_helper.h",
+    "desktop_capture_options.cc",
+    "desktop_capture_options.h",
+    "desktop_capturer.cc",
+    "desktop_capturer.h",
+    "desktop_capturer_differ_wrapper.cc",
+    "desktop_capturer_differ_wrapper.h",
+    "desktop_capturer_wrapper.cc",
+    "desktop_capturer_wrapper.h",
+    "desktop_frame_generator.cc",
+    "desktop_frame_generator.h",
+    "desktop_frame_rotation.cc",
+    "desktop_frame_rotation.h",
+    "differ_block.cc",
+    "differ_block.h",
+    "fake_desktop_capturer.cc",
+    "fake_desktop_capturer.h",
+    "fallback_desktop_capturer_wrapper.cc",
+    "fallback_desktop_capturer_wrapper.h",
+    "full_screen_application_handler.cc",
+    "full_screen_application_handler.h",
+    "full_screen_window_detector.cc",
+    "full_screen_window_detector.h",
+    "mouse_cursor.cc",
+    "mouse_cursor.h",
+    "mouse_cursor_monitor.h",
+    "resolution_tracker.cc",
+    "resolution_tracker.h",
+    "rgba_color.cc",
+    "rgba_color.h",
+    "screen_capture_frame_queue.h",
+    "screen_capturer_helper.cc",
+    "screen_capturer_helper.h",
+    "window_finder.cc",
+    "window_finder.h",
+  ]
+  if (is_linux && !is_castos && rtc_use_pipewire) {
+    sources += [ "desktop_capture_metadata.h" ]
+  }
+  if (is_mac) {
+    sources += [
+      "mac/desktop_configuration.h",
+      "mac/desktop_configuration_monitor.cc",
+      "mac/desktop_configuration_monitor.h",
+      "mac/full_screen_mac_application_handler.cc",
+      "mac/full_screen_mac_application_handler.h",
+      "mac/window_list_utils.cc",
+      "mac/window_list_utils.h",
+    ]
+    deps += [ ":desktop_capture_objc" ]
+  }
+  if (rtc_use_x11_extensions || rtc_use_pipewire) {
+    sources += [
+      "mouse_cursor_monitor_linux.cc",
+      "screen_capturer_linux.cc",
+      "window_capturer_linux.cc",
+    ]
+  }
+
+  if (rtc_use_x11_extensions) {
+    sources += [
+      "linux/x11/mouse_cursor_monitor_x11.cc",
+      "linux/x11/mouse_cursor_monitor_x11.h",
+      "linux/x11/screen_capturer_x11.cc",
+      "linux/x11/screen_capturer_x11.h",
+      "linux/x11/shared_x_display.cc",
+      "linux/x11/shared_x_display.h",
+      "linux/x11/window_capturer_x11.cc",
+      "linux/x11/window_capturer_x11.h",
+      "linux/x11/window_finder_x11.cc",
+      "linux/x11/window_finder_x11.h",
+      "linux/x11/window_list_utils.cc",
+      "linux/x11/window_list_utils.h",
+      "linux/x11/x_atom_cache.cc",
+      "linux/x11/x_atom_cache.h",
+      "linux/x11/x_error_trap.cc",
+      "linux/x11/x_error_trap.h",
+      "linux/x11/x_server_pixel_buffer.cc",
+      "linux/x11/x_server_pixel_buffer.h",
+      "linux/x11/x_window_property.cc",
+      "linux/x11/x_window_property.h",
+    ]
+    libs = [
+      "X11",
+      "Xcomposite",
+      "Xdamage",
+      "Xext",
+      "Xfixes",
+      "Xrender",
+      "Xrandr",
+      "Xtst",
+    ]
+  }
+
+  if (!is_win && !is_mac && !rtc_use_x11_extensions && !rtc_use_pipewire &&
+      !is_fuchsia) {
+    sources += [
+      "mouse_cursor_monitor_null.cc",
+      "screen_capturer_null.cc",
+      "window_capturer_null.cc",
+    ]
+  }
+
+  deps += [
+    ":primitives",
+    "../../api:function_view",
+    "../../api:make_ref_counted",
+    "../../api:refcountedbase",
+    "../../api:scoped_refptr",
+    "../../api:sequence_checker",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:random",
+    "../../rtc_base:stringutils",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:arch",
+    "../../rtc_base/system:no_unique_address",
+    "../../rtc_base/system:rtc_export",
+    "../../system_wrappers",
+    "../../system_wrappers:metrics",
+  ]
+
+  if (is_fuchsia) {
+    sources += [
+      "mouse_cursor_monitor_null.cc",
+      "screen_capturer_fuchsia.cc",
+      "screen_capturer_fuchsia.h",
+      "window_capturer_null.cc",
+    ]
+    deps += [
+      "../../rtc_base:divide_round",
+      "//third_party/fuchsia-sdk/sdk/fidl/fuchsia.sysmem",
+      "//third_party/fuchsia-sdk/sdk/fidl/fuchsia.ui.composition",
+      "//third_party/fuchsia-sdk/sdk/fidl/fuchsia.ui.scenic",
+      "//third_party/fuchsia-sdk/sdk/pkg/scenic_cpp",
+      "//third_party/fuchsia-sdk/sdk/pkg/sys_cpp",
+    ]
+  }
+
+  if (is_win) {
+    sources += [
+      "cropping_window_capturer_win.cc",
+      "desktop_frame_win.cc",
+      "desktop_frame_win.h",
+      "mouse_cursor_monitor_win.cc",
+      "screen_capturer_win.cc",
+      "win/cursor.cc",
+      "win/cursor.h",
+      "win/d3d_device.cc",
+      "win/d3d_device.h",
+      "win/desktop.cc",
+      "win/desktop.h",
+      "win/desktop_capture_utils.cc",
+      "win/desktop_capture_utils.h",
+      "win/display_configuration_monitor.cc",
+      "win/display_configuration_monitor.h",
+      "win/dxgi_adapter_duplicator.cc",
+      "win/dxgi_adapter_duplicator.h",
+      "win/dxgi_context.cc",
+      "win/dxgi_context.h",
+      "win/dxgi_duplicator_controller.cc",
+      "win/dxgi_duplicator_controller.h",
+      "win/dxgi_frame.cc",
+      "win/dxgi_frame.h",
+      "win/dxgi_output_duplicator.cc",
+      "win/dxgi_output_duplicator.h",
+      "win/dxgi_texture.cc",
+      "win/dxgi_texture.h",
+      "win/dxgi_texture_mapping.cc",
+      "win/dxgi_texture_mapping.h",
+      "win/dxgi_texture_staging.cc",
+      "win/dxgi_texture_staging.h",
+      "win/full_screen_win_application_handler.cc",
+      "win/full_screen_win_application_handler.h",
+      "win/scoped_gdi_object.h",
+      "win/scoped_thread_desktop.cc",
+      "win/scoped_thread_desktop.h",
+      "win/screen_capture_utils.cc",
+      "win/screen_capture_utils.h",
+      "win/screen_capturer_win_directx.cc",
+      "win/screen_capturer_win_directx.h",
+      "win/screen_capturer_win_gdi.cc",
+      "win/screen_capturer_win_gdi.h",
+      "win/screen_capturer_win_magnifier.cc",
+      "win/screen_capturer_win_magnifier.h",
+      "win/selected_window_context.cc",
+      "win/selected_window_context.h",
+      "win/window_capture_utils.cc",
+      "win/window_capture_utils.h",
+      "win/window_capturer_win_gdi.cc",
+      "win/window_capturer_win_gdi.h",
+      "window_capturer_win.cc",
+      "window_finder_win.cc",
+      "window_finder_win.h",
+    ]
+    libs = [
+      "d3d11.lib",
+      "dxgi.lib",
+    ]
+    deps += [
+      "../../rtc_base:win32",
+      "../../rtc_base/win:create_direct3d_device",
+      "../../rtc_base/win:get_activation_factory",
+      "../../rtc_base/win:windows_version",
+    ]
+  }
+
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+
+  if (rtc_use_x11_extensions) {
+    deps += [ "../../rtc_base:sanitizer" ]
+  }
+
+  deps += [ "//third_party/libyuv" ]
+
+  if (use_desktop_capture_differ_sse2) {
+    deps += [ ":desktop_capture_differ_sse2" ]
+  }
+
+  if (rtc_use_pipewire) {
+    sources += [
+      "linux/wayland/base_capturer_pipewire.cc",
+      "linux/wayland/base_capturer_pipewire.h",
+      "linux/wayland/egl_dmabuf.cc",
+      "linux/wayland/egl_dmabuf.h",
+      "linux/wayland/mouse_cursor_monitor_pipewire.cc",
+      "linux/wayland/mouse_cursor_monitor_pipewire.h",
+      "linux/wayland/portal_request_response.h",
+      "linux/wayland/restore_token_manager.cc",
+      "linux/wayland/restore_token_manager.h",
+      "linux/wayland/scoped_glib.h",
+      "linux/wayland/screen_capture_portal_interface.cc",
+      "linux/wayland/screen_capture_portal_interface.h",
+      "linux/wayland/screencast_portal.cc",
+      "linux/wayland/screencast_portal.h",
+      "linux/wayland/screencast_stream_utils.cc",
+      "linux/wayland/screencast_stream_utils.h",
+      "linux/wayland/shared_screencast_stream.cc",
+      "linux/wayland/shared_screencast_stream.h",
+      "linux/wayland/xdg_desktop_portal_utils.h",
+      "linux/wayland/xdg_session_details.h",
+    ]
+
+    configs += [ "../portal:pipewire_all" ]
+
+    public_configs += [ "../portal:pipewire_config" ]
+
+    deps += [
+      "../../rtc_base:sanitizer",
+      "../portal",
+    ]
+  }
+
+  if (rtc_enable_win_wgc) {
+    sources += [
+      "win/wgc_capture_session.cc",
+      "win/wgc_capture_session.h",
+      "win/wgc_capture_source.cc",
+      "win/wgc_capture_source.h",
+      "win/wgc_capturer_win.cc",
+      "win/wgc_capturer_win.h",
+      "win/wgc_desktop_frame.cc",
+      "win/wgc_desktop_frame.h",
+    ]
+    libs += [ "dwmapi.lib" ]
+    deps += [
+      "../../rtc_base:rtc_event",
+      "../../rtc_base/win:hstring",
+    ]
+  }
+}
+
+if (is_mac) {
+  rtc_library("desktop_capture_objc") {
+    # This target, needs to be separated from ":desktop_capture" because
+    # that is the C++ part of the target while this one is the Obj-C++ part.
+    # Aside from this, both represent a "desktop_capture" target.
+    # This target separation based on programming languages introduces a
+    # dependency cycle between ":desktop_capture" and
+    # ":desktop_capture_objc".
+    # To break this, ":desktop_capture_objc" shares some .h files with
+    # ":desktop_capture" but when external targets need one of these
+    # headers, they should depend on ":desktop_capture" and consider
+    # this target as private.
+    visibility = [ ":desktop_capture" ]
+    sources = [
+      "delegated_source_list_controller.h",
+      "desktop_capture_options.h",
+      "desktop_capturer.h",
+      "full_screen_application_handler.h",
+      "full_screen_window_detector.h",
+      "mac/desktop_configuration.h",
+      "mac/desktop_configuration.mm",
+      "mac/desktop_configuration_monitor.h",
+      "mac/desktop_frame_cgimage.h",
+      "mac/desktop_frame_cgimage.mm",
+      "mac/desktop_frame_iosurface.h",
+      "mac/desktop_frame_iosurface.mm",
+      "mac/desktop_frame_provider.h",
+      "mac/desktop_frame_provider.mm",
+      "mac/screen_capturer_mac.h",
+      "mac/screen_capturer_mac.mm",
+      "mac/window_list_utils.h",
+      "mouse_cursor.h",
+      "mouse_cursor_monitor.h",
+      "mouse_cursor_monitor_mac.mm",
+      "screen_capture_frame_queue.h",
+      "screen_capturer_darwin.mm",
+      "screen_capturer_helper.h",
+      "window_capturer_mac.mm",
+      "window_finder.h",
+      "window_finder_mac.h",
+      "window_finder_mac.mm",
+    ]
+    deps = [
+      ":primitives",
+      "../../api:function_view",
+      "../../api:refcountedbase",
+      "../../api:scoped_refptr",
+      "../../api:sequence_checker",
+      "../../rtc_base:checks",
+      "../../rtc_base:event_tracer",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:timeutils",
+      "../../rtc_base/synchronization:mutex",
+      "../../rtc_base/system:rtc_export",
+      "../../sdk:helpers_objc",
+    ]
+    frameworks = [
+      "AppKit.framework",
+      "IOKit.framework",
+      "IOSurface.framework",
+    ]
+  }
+}
+
+if (use_desktop_capture_differ_sse2) {
+  # Have to be compiled as a separate target because it needs to be compiled
+  # with SSE2 enabled.
+  rtc_library("desktop_capture_differ_sse2") {
+    visibility = [ ":*" ]
+    sources = [
+      "differ_vector_sse2.cc",
+      "differ_vector_sse2.h",
+    ]
+
+    if (is_posix || is_fuchsia) {
+      cflags = [ "-msse2" ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/desktop_capture/DEPS b/third_party/libwebrtc/modules/desktop_capture/DEPS
new file mode 100644
index 0000000000..8c894c4430
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/DEPS
@@ -0,0 +1,19 @@
+include_rules = [
+  "+system_wrappers",
+  "+third_party/libyuv",
+]
+
+specific_include_rules = {
+  "desktop_frame_cgimage\.h": [
+    "+sdk/objc",
+  ],
+  "desktop_frame_iosurface\.h": [
+    "+sdk/objc",
+  ],
+  "desktop_frame_provider\.h": [
+    "+sdk/objc",
+  ],
+  "screen_capturer_mac\.mm": [
+    "+sdk/objc",
+  ],
+}
diff --git a/third_party/libwebrtc/modules/desktop_capture/OWNERS b/third_party/libwebrtc/modules/desktop_capture/OWNERS
new file mode 100644
index 0000000000..e3bc32ee5c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/OWNERS
@@ -0,0 +1,2 @@
+alcooper@chromium.org
+mfoltz@chromium.org
diff --git a/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.cc b/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.cc
new file mode 100644
index 0000000000..8e56ffc3fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.cc
@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h"
+
+#include <stdint.h>
+
+#include <utility>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+BlankDetectorDesktopCapturerWrapper::BlankDetectorDesktopCapturerWrapper(
+    std::unique_ptr<DesktopCapturer> capturer,
+    RgbaColor blank_pixel,
+    bool check_per_capture)
+    : capturer_(std::move(capturer)),
+      blank_pixel_(blank_pixel),
+      check_per_capture_(check_per_capture) {
+  RTC_DCHECK(capturer_);
+}
+
+BlankDetectorDesktopCapturerWrapper::~BlankDetectorDesktopCapturerWrapper() =
+    default;
+
+void BlankDetectorDesktopCapturerWrapper::Start(
+    DesktopCapturer::Callback* callback) {
+  callback_ = callback;
+  capturer_->Start(this);
+}
+
+void BlankDetectorDesktopCapturerWrapper::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  capturer_->SetSharedMemoryFactory(std::move(shared_memory_factory));
+}
+
+void BlankDetectorDesktopCapturerWrapper::CaptureFrame() {
+  RTC_DCHECK(callback_);
+  capturer_->CaptureFrame();
+}
+
+void BlankDetectorDesktopCapturerWrapper::SetExcludedWindow(WindowId window) {
+  capturer_->SetExcludedWindow(window);
+}
+
+bool BlankDetectorDesktopCapturerWrapper::GetSourceList(SourceList* sources) {
+  return capturer_->GetSourceList(sources);
+}
+
+bool BlankDetectorDesktopCapturerWrapper::SelectSource(SourceId id) {
+  if (check_per_capture_) {
+    // If we start capturing a new source, we must reset these members
+    // so we don't short circuit the blank detection logic.
+    is_first_frame_ = true;
+    non_blank_frame_received_ = false;
+  }
+
+  return capturer_->SelectSource(id);
+}
+
+bool BlankDetectorDesktopCapturerWrapper::FocusOnSelectedSource() {
+  return capturer_->FocusOnSelectedSource();
+}
+
+bool BlankDetectorDesktopCapturerWrapper::IsOccluded(const DesktopVector& pos) {
+  return capturer_->IsOccluded(pos);
+}
+
+void BlankDetectorDesktopCapturerWrapper::OnCaptureResult(
+    Result result,
+    std::unique_ptr<DesktopFrame> frame) {
+  RTC_DCHECK(callback_);
+  if (result != Result::SUCCESS || non_blank_frame_received_) {
+    callback_->OnCaptureResult(result, std::move(frame));
+    return;
+  }
+
+  if (!frame) {
+    // Capturer can call the blank detector with empty frame. Blank
+    // detector regards it as a blank frame.
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY,
+                               std::unique_ptr<DesktopFrame>());
+    return;
+  }
+
+  // If nothing has been changed in current frame, we do not need to check it
+  // again.
+  if (!frame->updated_region().is_empty() || is_first_frame_) {
+    last_frame_is_blank_ = IsBlankFrame(*frame);
+    is_first_frame_ = false;
+  }
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.DesktopCapture.BlankFrameDetected",
+                        last_frame_is_blank_);
+  if (!last_frame_is_blank_) {
+    non_blank_frame_received_ = true;
+    callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
+    return;
+  }
+
+  callback_->OnCaptureResult(Result::ERROR_TEMPORARY,
+                             std::unique_ptr<DesktopFrame>());
+}
+
+bool BlankDetectorDesktopCapturerWrapper::IsBlankFrame(
+    const DesktopFrame& frame) const {
+  // We will check 7489 pixels for a frame with 1024 x 768 resolution.
+  for (int i = 0; i < frame.size().width() * frame.size().height(); i += 105) {
+    const int x = i % frame.size().width();
+    const int y = i / frame.size().width();
+    if (!IsBlankPixel(frame, x, y)) {
+      return false;
+    }
+  }
+
+  // We are verifying the pixel in the center as well.
+  return IsBlankPixel(frame, frame.size().width() / 2,
+                      frame.size().height() / 2);
+}
+
+bool BlankDetectorDesktopCapturerWrapper::IsBlankPixel(
+    const DesktopFrame& frame,
+    int x,
+    int y) const {
+  uint8_t* pixel_data = frame.GetFrameDataAtPos(DesktopVector(x, y));
+  return RgbaColor(pixel_data) == blank_pixel_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h b/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h
new file mode 100644
index 0000000000..d10f9cf725
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_BLANK_DETECTOR_DESKTOP_CAPTURER_WRAPPER_H_
+#define MODULES_DESKTOP_CAPTURE_BLANK_DETECTOR_DESKTOP_CAPTURER_WRAPPER_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/rgba_color.h"
+#include "modules/desktop_capture/shared_memory.h"
+
+namespace webrtc {
+
+// A DesktopCapturer wrapper detects the return value of its owned
+// DesktopCapturer implementation. If sampled pixels returned by the
+// DesktopCapturer implementation all equal to the blank pixel, this wrapper
+// returns ERROR_TEMPORARY. If the DesktopCapturer implementation fails for too
+// many times, this wrapper returns ERROR_PERMANENT.
+class BlankDetectorDesktopCapturerWrapper final
+    : public DesktopCapturer,
+      public DesktopCapturer::Callback {
+ public:
+  // Creates BlankDetectorDesktopCapturerWrapper. BlankDesktopCapturerWrapper
+  // takes ownership of `capturer`. The `blank_pixel` is the unmodified color
+  // returned by the `capturer`.
+  BlankDetectorDesktopCapturerWrapper(std::unique_ptr<DesktopCapturer> capturer,
+                                      RgbaColor blank_pixel,
+                                      bool check_per_capture = false);
+  ~BlankDetectorDesktopCapturerWrapper() override;
+
+  // DesktopCapturer interface.
+  void Start(DesktopCapturer::Callback* callback) override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  void CaptureFrame() override;
+  void SetExcludedWindow(WindowId window) override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  bool IsOccluded(const DesktopVector& pos) override;
+
+ private:
+  // DesktopCapturer::Callback interface.
+  void OnCaptureResult(Result result,
+                       std::unique_ptr<DesktopFrame> frame) override;
+
+  bool IsBlankFrame(const DesktopFrame& frame) const;
+
+  // Detects whether pixel at (x, y) equals to `blank_pixel_`.
+  bool IsBlankPixel(const DesktopFrame& frame, int x, int y) const;
+
+  const std::unique_ptr<DesktopCapturer> capturer_;
+  const RgbaColor blank_pixel_;
+
+  // Whether a non-blank frame has been received.
+  bool non_blank_frame_received_ = false;
+
+  // Whether the last frame is blank.
+  bool last_frame_is_blank_ = false;
+
+  // Whether current frame is the first frame.
+  bool is_first_frame_ = true;
+
+  // Blank inspection is made per capture instead of once for all
+  // screens or windows.
+  bool check_per_capture_ = false;
+
+  DesktopCapturer::Callback* callback_ = nullptr;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_BLANK_DETECTOR_DESKTOP_CAPTURER_WRAPPER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper_unittest.cc
new file mode 100644
index 0000000000..25a81edd89
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper_unittest.cc
@@ -0,0 +1,165 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_frame_generator.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/fake_desktop_capturer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class BlankDetectorDesktopCapturerWrapperTest
+    : public ::testing::Test,
+      public DesktopCapturer::Callback {
+ public:
+  BlankDetectorDesktopCapturerWrapperTest();
+  ~BlankDetectorDesktopCapturerWrapperTest() override;
+
+ protected:
+  void PerfTest(DesktopCapturer* capturer);
+
+  const int frame_width_ = 1024;
+  const int frame_height_ = 768;
+  std::unique_ptr<BlankDetectorDesktopCapturerWrapper> wrapper_;
+  DesktopCapturer* capturer_ = nullptr;
+  BlackWhiteDesktopFramePainter painter_;
+  int num_frames_captured_ = 0;
+  DesktopCapturer::Result last_result_ = DesktopCapturer::Result::SUCCESS;
+  std::unique_ptr<DesktopFrame> last_frame_;
+
+ private:
+  // DesktopCapturer::Callback interface.
+  void OnCaptureResult(DesktopCapturer::Result result,
+                       std::unique_ptr<DesktopFrame> frame) override;
+
+  PainterDesktopFrameGenerator frame_generator_;
+};
+
+BlankDetectorDesktopCapturerWrapperTest::
+    BlankDetectorDesktopCapturerWrapperTest() {
+  frame_generator_.size()->set(frame_width_, frame_height_);
+  frame_generator_.set_desktop_frame_painter(&painter_);
+  std::unique_ptr<DesktopCapturer> capturer(new FakeDesktopCapturer());
+  FakeDesktopCapturer* fake_capturer =
+      static_cast<FakeDesktopCapturer*>(capturer.get());
+  fake_capturer->set_frame_generator(&frame_generator_);
+  capturer_ = fake_capturer;
+  wrapper_.reset(new BlankDetectorDesktopCapturerWrapper(
+      std::move(capturer), RgbaColor(0, 0, 0, 0)));
+  wrapper_->Start(this);
+}
+
+BlankDetectorDesktopCapturerWrapperTest::
+    ~BlankDetectorDesktopCapturerWrapperTest() = default;
+
+void BlankDetectorDesktopCapturerWrapperTest::OnCaptureResult(
+    DesktopCapturer::Result result,
+    std::unique_ptr<DesktopFrame> frame) {
+  last_result_ = result;
+  last_frame_ = std::move(frame);
+  num_frames_captured_++;
+}
+
+void BlankDetectorDesktopCapturerWrapperTest::PerfTest(
+    DesktopCapturer* capturer) {
+  for (int i = 0; i < 10000; i++) {
+    capturer->CaptureFrame();
+    ASSERT_EQ(num_frames_captured_, i + 1);
+  }
+}
+
+TEST_F(BlankDetectorDesktopCapturerWrapperTest, ShouldDetectBlankFrame) {
+  wrapper_->CaptureFrame();
+  ASSERT_EQ(num_frames_captured_, 1);
+  ASSERT_EQ(last_result_, DesktopCapturer::Result::ERROR_TEMPORARY);
+  ASSERT_FALSE(last_frame_);
+}
+
+TEST_F(BlankDetectorDesktopCapturerWrapperTest, ShouldPassBlankDetection) {
+  painter_.updated_region()->AddRect(DesktopRect::MakeXYWH(0, 0, 100, 100));
+  wrapper_->CaptureFrame();
+  ASSERT_EQ(num_frames_captured_, 1);
+  ASSERT_EQ(last_result_, DesktopCapturer::Result::SUCCESS);
+  ASSERT_TRUE(last_frame_);
+
+  painter_.updated_region()->AddRect(
+      DesktopRect::MakeXYWH(frame_width_ - 100, frame_height_ - 100, 100, 100));
+  wrapper_->CaptureFrame();
+  ASSERT_EQ(num_frames_captured_, 2);
+  ASSERT_EQ(last_result_, DesktopCapturer::Result::SUCCESS);
+  ASSERT_TRUE(last_frame_);
+
+  painter_.updated_region()->AddRect(
+      DesktopRect::MakeXYWH(0, frame_height_ - 100, 100, 100));
+  wrapper_->CaptureFrame();
+  ASSERT_EQ(num_frames_captured_, 3);
+  ASSERT_EQ(last_result_, DesktopCapturer::Result::SUCCESS);
+  ASSERT_TRUE(last_frame_);
+
+  painter_.updated_region()->AddRect(
+      DesktopRect::MakeXYWH(frame_width_ - 100, 0, 100, 100));
+  wrapper_->CaptureFrame();
+  ASSERT_EQ(num_frames_captured_, 4);
+  ASSERT_EQ(last_result_, DesktopCapturer::Result::SUCCESS);
+  ASSERT_TRUE(last_frame_);
+
+  painter_.updated_region()->AddRect(DesktopRect::MakeXYWH(
+      (frame_width_ >> 1) - 50, (frame_height_ >> 1) - 50, 100, 100));
+  wrapper_->CaptureFrame();
+  ASSERT_EQ(num_frames_captured_, 5);
+  ASSERT_EQ(last_result_, DesktopCapturer::Result::SUCCESS);
+  ASSERT_TRUE(last_frame_);
+}
+
+TEST_F(BlankDetectorDesktopCapturerWrapperTest,
+       ShouldNotCheckAfterANonBlankFrameReceived) {
+  wrapper_->CaptureFrame();
+  ASSERT_EQ(num_frames_captured_, 1);
+  ASSERT_EQ(last_result_, DesktopCapturer::Result::ERROR_TEMPORARY);
+  ASSERT_FALSE(last_frame_);
+
+  painter_.updated_region()->AddRect(
+      DesktopRect::MakeXYWH(frame_width_ - 100, 0, 100, 100));
+  wrapper_->CaptureFrame();
+  ASSERT_EQ(num_frames_captured_, 2);
+  ASSERT_EQ(last_result_, DesktopCapturer::Result::SUCCESS);
+  ASSERT_TRUE(last_frame_);
+
+  for (int i = 0; i < 100; i++) {
+    wrapper_->CaptureFrame();
+    ASSERT_EQ(num_frames_captured_, i + 3);
+    ASSERT_EQ(last_result_, DesktopCapturer::Result::SUCCESS);
+    ASSERT_TRUE(last_frame_);
+  }
+}
+
+// There is no perceptible impact by using BlankDetectorDesktopCapturerWrapper.
+// i.e. less than 0.2ms per frame.
+// [ OK ] DISABLED_Performance (10210 ms)
+// [ OK ] DISABLED_PerformanceComparison (8791 ms)
+TEST_F(BlankDetectorDesktopCapturerWrapperTest, DISABLED_Performance) {
+  PerfTest(wrapper_.get());
+}
+
+TEST_F(BlankDetectorDesktopCapturerWrapperTest,
+       DISABLED_PerformanceComparison) {
+  capturer_->Start(this);
+  PerfTest(capturer_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame.cc b/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame.cc
new file mode 100644
index 0000000000..54488b7d62
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/cropped_desktop_frame.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/desktop_capture/desktop_region.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// A DesktopFrame that is a sub-rect of another DesktopFrame.
+class CroppedDesktopFrame : public DesktopFrame {
+ public:
+  CroppedDesktopFrame(std::unique_ptr<DesktopFrame> frame,
+                      const DesktopRect& rect);
+
+  CroppedDesktopFrame(const CroppedDesktopFrame&) = delete;
+  CroppedDesktopFrame& operator=(const CroppedDesktopFrame&) = delete;
+
+ private:
+  const std::unique_ptr<DesktopFrame> frame_;
+};
+
+std::unique_ptr<DesktopFrame> CreateCroppedDesktopFrame(
+    std::unique_ptr<DesktopFrame> frame,
+    const DesktopRect& rect) {
+  RTC_DCHECK(frame);
+
+  DesktopRect intersection = DesktopRect::MakeSize(frame->size());
+  intersection.IntersectWith(rect);
+  if (intersection.is_empty()) {
+    return nullptr;
+  }
+
+  if (frame->size().equals(rect.size())) {
+    return frame;
+  }
+
+  return std::unique_ptr<DesktopFrame>(
+      new CroppedDesktopFrame(std::move(frame), intersection));
+}
+
+CroppedDesktopFrame::CroppedDesktopFrame(std::unique_ptr<DesktopFrame> frame,
+                                         const DesktopRect& rect)
+    : DesktopFrame(rect.size(),
+                   frame->stride(),
+                   frame->GetFrameDataAtPos(rect.top_left()),
+                   frame->shared_memory()),
+      frame_(std::move(frame)) {
+  MoveFrameInfoFrom(frame_.get());
+  set_top_left(frame_->top_left().add(rect.top_left()));
+  mutable_updated_region()->IntersectWith(rect);
+  mutable_updated_region()->Translate(-rect.left(), -rect.top());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame.h b/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame.h
new file mode 100644
index 0000000000..5c672c7d32
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_CROPPED_DESKTOP_FRAME_H_
+#define MODULES_DESKTOP_CAPTURE_CROPPED_DESKTOP_FRAME_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Creates a DesktopFrame to contain only the area of `rect` in the original
+// `frame`.
+// `frame` should not be nullptr. `rect` is in `frame` coordinate, i.e.
+// `frame`->top_left() does not impact the area of `rect`.
+// Returns nullptr frame if `rect` is not contained by the bounds of `frame`.
+std::unique_ptr<DesktopFrame> RTC_EXPORT
+CreateCroppedDesktopFrame(std::unique_ptr<DesktopFrame> frame,
+                          const DesktopRect& rect);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_CROPPED_DESKTOP_FRAME_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame_unittest.cc
new file mode 100644
index 0000000000..9becf69636
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame_unittest.cc
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/cropped_desktop_frame.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+std::unique_ptr<DesktopFrame> CreateTestFrame() {
+  return std::make_unique<BasicDesktopFrame>(DesktopSize(10, 20));
+}
+
+TEST(CroppedDesktopFrameTest, DoNotCreateWrapperIfSizeIsNotChanged) {
+  std::unique_ptr<DesktopFrame> original = CreateTestFrame();
+  // owned by `original` and CroppedDesktopFrame.
+  DesktopFrame* raw_original = original.get();
+  std::unique_ptr<DesktopFrame> cropped = CreateCroppedDesktopFrame(
+      std::move(original), DesktopRect::MakeWH(10, 20));
+  ASSERT_EQ(cropped.get(), raw_original);
+}
+
+TEST(CroppedDesktopFrameTest, CropWhenPartiallyOutOfBounds) {
+  std::unique_ptr<DesktopFrame> cropped =
+      CreateCroppedDesktopFrame(CreateTestFrame(), DesktopRect::MakeWH(11, 10));
+  ASSERT_NE(nullptr, cropped);
+  ASSERT_EQ(cropped->size().width(), 10);
+  ASSERT_EQ(cropped->size().height(), 10);
+  ASSERT_EQ(cropped->top_left().x(), 0);
+  ASSERT_EQ(cropped->top_left().y(), 0);
+}
+
+TEST(CroppedDesktopFrameTest, ReturnNullIfCropRegionIsOutOfBounds) {
+  std::unique_ptr<DesktopFrame> frame = CreateTestFrame();
+  frame->set_top_left(DesktopVector(100, 200));
+  ASSERT_EQ(nullptr,
+            CreateCroppedDesktopFrame(
+                std::move(frame), DesktopRect::MakeLTRB(101, 203, 109, 218)));
+}
+
+TEST(CroppedDesktopFrameTest, CropASubArea) {
+  std::unique_ptr<DesktopFrame> cropped = CreateCroppedDesktopFrame(
+      CreateTestFrame(), DesktopRect::MakeLTRB(1, 2, 9, 19));
+  ASSERT_EQ(cropped->size().width(), 8);
+  ASSERT_EQ(cropped->size().height(), 17);
+  ASSERT_EQ(cropped->top_left().x(), 1);
+  ASSERT_EQ(cropped->top_left().y(), 2);
+}
+
+TEST(CroppedDesktopFrameTest, SetTopLeft) {
+  std::unique_ptr<DesktopFrame> frame = CreateTestFrame();
+  frame->set_top_left(DesktopVector(100, 200));
+  frame = CreateCroppedDesktopFrame(std::move(frame),
+                                    DesktopRect::MakeLTRB(1, 3, 9, 18));
+  ASSERT_EQ(frame->size().width(), 8);
+  ASSERT_EQ(frame->size().height(), 15);
+  ASSERT_EQ(frame->top_left().x(), 101);
+  ASSERT_EQ(frame->top_left().y(), 203);
+}
+
+TEST(CroppedDesktopFrameTest, InitializedWithZeros) {
+  std::unique_ptr<DesktopFrame> frame = CreateTestFrame();
+  const DesktopVector frame_origin = frame->top_left();
+  const DesktopSize frame_size = frame->size();
+  std::unique_ptr<DesktopFrame> cropped = CreateCroppedDesktopFrame(
+      std::move(frame), DesktopRect::MakeOriginSize(frame_origin, frame_size));
+  for (int j = 0; j < cropped->size().height(); ++j) {
+    for (int i = 0; i < cropped->stride(); ++i) {
+      ASSERT_EQ(cropped->data()[i + j * cropped->stride()], 0);
+    }
+  }
+}
+
+TEST(CroppedDesktopFrameTest, IccProfile) {
+  const uint8_t fake_icc_profile_data_array[] = {0x1a, 0x00, 0x2b, 0x00,
+                                                 0x3c, 0x00, 0x4d};
+  const std::vector<uint8_t> icc_profile(
+      fake_icc_profile_data_array,
+      fake_icc_profile_data_array + sizeof(fake_icc_profile_data_array));
+
+  std::unique_ptr<DesktopFrame> frame = CreateTestFrame();
+  EXPECT_EQ(frame->icc_profile().size(), 0UL);
+
+  frame->set_icc_profile(icc_profile);
+  EXPECT_EQ(frame->icc_profile().size(), 7UL);
+  EXPECT_EQ(frame->icc_profile(), icc_profile);
+
+  frame = CreateCroppedDesktopFrame(std::move(frame),
+                                    DesktopRect::MakeLTRB(2, 2, 8, 18));
+  EXPECT_EQ(frame->icc_profile().size(), 7UL);
+  EXPECT_EQ(frame->icc_profile(), icc_profile);
+
+  std::unique_ptr<SharedDesktopFrame> shared =
+      SharedDesktopFrame::Wrap(std::move(frame));
+  EXPECT_EQ(shared->icc_profile().size(), 7UL);
+  EXPECT_EQ(shared->icc_profile(), icc_profile);
+
+  std::unique_ptr<DesktopFrame> shared_other = shared->Share();
+  EXPECT_EQ(shared_other->icc_profile().size(), 7UL);
+  EXPECT_EQ(shared_other->icc_profile(), icc_profile);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer.cc b/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer.cc
new file mode 100644
index 0000000000..5e0faaade9
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/cropping_window_capturer.h"
+
+#include <stddef.h>
+
+#include <utility>
+
+#include "modules/desktop_capture/cropped_desktop_frame.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+CroppingWindowCapturer::CroppingWindowCapturer(
+    const DesktopCaptureOptions& options)
+    : options_(options),
+      callback_(NULL),
+      window_capturer_(DesktopCapturer::CreateRawWindowCapturer(options)),
+      selected_window_(kNullWindowId),
+      excluded_window_(kNullWindowId) {}
+
+CroppingWindowCapturer::~CroppingWindowCapturer() {}
+
+void CroppingWindowCapturer::Start(DesktopCapturer::Callback* callback) {
+  callback_ = callback;
+  window_capturer_->Start(callback);
+}
+
+void CroppingWindowCapturer::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  window_capturer_->SetSharedMemoryFactory(std::move(shared_memory_factory));
+}
+
+void CroppingWindowCapturer::CaptureFrame() {
+  if (ShouldUseScreenCapturer()) {
+    if (!screen_capturer_.get()) {
+      screen_capturer_ = DesktopCapturer::CreateRawScreenCapturer(options_);
+      if (excluded_window_) {
+        screen_capturer_->SetExcludedWindow(excluded_window_);
+      }
+      screen_capturer_->Start(this);
+    }
+    screen_capturer_->CaptureFrame();
+  } else {
+    window_capturer_->CaptureFrame();
+  }
+}
+
+void CroppingWindowCapturer::SetExcludedWindow(WindowId window) {
+  excluded_window_ = window;
+  if (screen_capturer_.get()) {
+    screen_capturer_->SetExcludedWindow(window);
+  }
+}
+
+bool CroppingWindowCapturer::GetSourceList(SourceList* sources) {
+  return window_capturer_->GetSourceList(sources);
+}
+
+bool CroppingWindowCapturer::SelectSource(SourceId id) {
+  if (window_capturer_->SelectSource(id)) {
+    selected_window_ = id;
+    return true;
+  }
+  return false;
+}
+
+bool CroppingWindowCapturer::FocusOnSelectedSource() {
+  return window_capturer_->FocusOnSelectedSource();
+}
+
+void CroppingWindowCapturer::OnCaptureResult(
+    DesktopCapturer::Result result,
+    std::unique_ptr<DesktopFrame> screen_frame) {
+  if (!ShouldUseScreenCapturer()) {
+    RTC_LOG(LS_INFO) << "Window no longer on top when ScreenCapturer finishes";
+    window_capturer_->CaptureFrame();
+    return;
+  }
+
+  if (result != Result::SUCCESS) {
+    RTC_LOG(LS_WARNING) << "ScreenCapturer failed to capture a frame";
+    callback_->OnCaptureResult(result, nullptr);
+    return;
+  }
+
+  DesktopRect window_rect = GetWindowRectInVirtualScreen();
+  if (window_rect.is_empty()) {
+    RTC_LOG(LS_WARNING) << "Window rect is empty";
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+
+  std::unique_ptr<DesktopFrame> cropped_frame =
+      CreateCroppedDesktopFrame(std::move(screen_frame), window_rect);
+
+  if (!cropped_frame) {
+    RTC_LOG(LS_WARNING) << "Window is outside of the captured display";
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+
+  callback_->OnCaptureResult(Result::SUCCESS, std::move(cropped_frame));
+}
+
+bool CroppingWindowCapturer::IsOccluded(const DesktopVector& pos) {
+  // Returns true if either capturer returns true.
+  if (window_capturer_->IsOccluded(pos)) {
+    return true;
+  }
+  if (screen_capturer_ != nullptr && screen_capturer_->IsOccluded(pos)) {
+    return true;
+  }
+  return false;
+}
+
+#if !defined(WEBRTC_WIN)
+// CroppingWindowCapturer is implemented only for windows. On other platforms
+// the regular window capturer is used.
+// static
+std::unique_ptr<DesktopCapturer> CroppingWindowCapturer::CreateCapturer(
+    const DesktopCaptureOptions& options) {
+  return DesktopCapturer::CreateWindowCapturer(options);
+}
+#endif
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer.h b/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer.h
new file mode 100644
index 0000000000..56478030b1
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer.h
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_CROPPING_WINDOW_CAPTURER_H_
+#define MODULES_DESKTOP_CAPTURE_CROPPING_WINDOW_CAPTURER_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/shared_memory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// WindowCapturer implementation that uses a screen capturer to capture the
+// whole screen and crops the video frame to the window area when the captured
+// window is on top.
+class RTC_EXPORT CroppingWindowCapturer : public DesktopCapturer,
+                                          public DesktopCapturer::Callback {
+ public:
+  static std::unique_ptr<DesktopCapturer> CreateCapturer(
+      const DesktopCaptureOptions& options);
+
+  ~CroppingWindowCapturer() override;
+
+  // DesktopCapturer implementation.
+  void Start(DesktopCapturer::Callback* callback) override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  void CaptureFrame() override;
+  void SetExcludedWindow(WindowId window) override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  bool IsOccluded(const DesktopVector& pos) override;
+
+  // DesktopCapturer::Callback implementation, passed to `screen_capturer_` to
+  // intercept the capture result.
+  void OnCaptureResult(DesktopCapturer::Result result,
+                       std::unique_ptr<DesktopFrame> frame) override;
+
+ protected:
+  explicit CroppingWindowCapturer(const DesktopCaptureOptions& options);
+
+  // The platform implementation should override these methods.
+
+  // Returns true if it is OK to capture the whole screen and crop to the
+  // selected window, i.e. the selected window is opaque, rectangular, and not
+  // occluded.
+  virtual bool ShouldUseScreenCapturer() = 0;
+
+  // Returns the window area relative to the top left of the virtual screen
+  // within the bounds of the virtual screen. This function should return the
+  // DesktopRect in full desktop coordinates, i.e. the top-left monitor starts
+  // from (0, 0).
+  virtual DesktopRect GetWindowRectInVirtualScreen() = 0;
+
+  WindowId selected_window() const { return selected_window_; }
+  WindowId excluded_window() const { return excluded_window_; }
+  DesktopCapturer* window_capturer() const { return window_capturer_.get(); }
+
+ private:
+  DesktopCaptureOptions options_;
+  DesktopCapturer::Callback* callback_;
+  std::unique_ptr<DesktopCapturer> window_capturer_;
+  std::unique_ptr<DesktopCapturer> screen_capturer_;
+  SourceId selected_window_;
+  WindowId excluded_window_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_CROPPING_WINDOW_CAPTURER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer_win.cc b/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer_win.cc
new file mode 100644
index 0000000000..ab2f807d33
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer_win.cc
@@ -0,0 +1,335 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/cropping_window_capturer.h"
+#include "modules/desktop_capture/desktop_capturer_differ_wrapper.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "modules/desktop_capture/win/selected_window_context.h"
+#include "modules/desktop_capture/win/window_capture_utils.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+#include "rtc_base/win/windows_version.h"
+
+namespace webrtc {
+
+namespace {
+
+// Used to pass input data for verifying the selected window is on top.
+struct TopWindowVerifierContext : public SelectedWindowContext {
+  TopWindowVerifierContext(HWND selected_window,
+                           HWND excluded_window,
+                           DesktopRect selected_window_rect,
+                           WindowCaptureHelperWin* window_capture_helper)
+      : SelectedWindowContext(selected_window,
+                              selected_window_rect,
+                              window_capture_helper),
+        excluded_window(excluded_window) {
+    RTC_DCHECK_NE(selected_window, excluded_window);
+  }
+
+  // Determines whether the selected window is on top (not occluded by any
+  // windows except for those it owns or any excluded window).
+  bool IsTopWindow() {
+    if (!IsSelectedWindowValid()) {
+      return false;
+    }
+
+    // Enumerate all top-level windows above the selected window in Z-order,
+    // checking whether any overlaps it. This uses FindWindowEx rather than
+    // EnumWindows because the latter excludes certain system windows (e.g. the
+    // Start menu & other taskbar menus) that should be detected here to avoid
+    // inadvertent capture.
+    int num_retries = 0;
+    while (true) {
+      HWND hwnd = nullptr;
+      while ((hwnd = FindWindowEx(nullptr, hwnd, nullptr, nullptr))) {
+        if (hwnd == selected_window()) {
+          // Windows are enumerated in top-down Z-order, so we can stop
+          // enumerating upon reaching the selected window & report it's on top.
+          return true;
+        }
+
+        // Ignore the excluded window.
+        if (hwnd == excluded_window) {
+          continue;
+        }
+
+        // Ignore windows that aren't visible on the current desktop.
+        if (!window_capture_helper()->IsWindowVisibleOnCurrentDesktop(hwnd)) {
+          continue;
+        }
+
+        // Ignore Chrome notification windows, especially the notification for
+        // the ongoing window sharing. Notes:
+        // - This only works with notifications from Chrome, not other Apps.
+        // - All notifications from Chrome will be ignored.
+        // - This may cause part or whole of notification window being cropped
+        // into the capturing of the target window if there is overlapping.
+        if (window_capture_helper()->IsWindowChromeNotification(hwnd)) {
+          continue;
+        }
+
+        // Ignore windows owned by the selected window since we want to capture
+        // them.
+        if (IsWindowOwnedBySelectedWindow(hwnd)) {
+          continue;
+        }
+
+        // Check whether this window intersects with the selected window.
+        if (IsWindowOverlappingSelectedWindow(hwnd)) {
+          // If intersection is not empty, the selected window is not on top.
+          return false;
+        }
+      }
+
+      DWORD lastError = GetLastError();
+      if (lastError == ERROR_SUCCESS) {
+        // The enumeration completed successfully without finding the selected
+        // window (which may have been closed).
+        RTC_LOG(LS_WARNING) << "Failed to find selected window (only expected "
+                               "if it was closed)";
+        RTC_DCHECK(!IsWindow(selected_window()));
+        return false;
+      } else if (lastError == ERROR_INVALID_WINDOW_HANDLE) {
+        // This error may occur if a window is closed around the time it's
+        // enumerated; retry the enumeration in this case up to 10 times
+        // (this should be a rare race & unlikely to recur).
+        if (++num_retries <= 10) {
+          RTC_LOG(LS_WARNING) << "Enumeration failed due to race with a window "
+                                 "closing; retrying - retry #"
+                              << num_retries;
+          continue;
+        } else {
+          RTC_LOG(LS_ERROR)
+              << "Exhausted retry allowance around window enumeration failures "
+                 "due to races with windows closing";
+        }
+      }
+
+      // The enumeration failed with an unexpected error (or more repeats of
+      // an infrequently-expected error than anticipated). After logging this &
+      // firing an assert when enabled, report that the selected window isn't
+      // topmost to avoid inadvertent capture of other windows.
+      RTC_LOG(LS_ERROR) << "Failed to enumerate windows: " << lastError;
+      RTC_DCHECK_NOTREACHED();
+      return false;
+    }
+  }
+
+  const HWND excluded_window;
+};
+
+class CroppingWindowCapturerWin : public CroppingWindowCapturer {
+ public:
+  explicit CroppingWindowCapturerWin(const DesktopCaptureOptions& options)
+      : CroppingWindowCapturer(options),
+        enumerate_current_process_windows_(
+            options.enumerate_current_process_windows()),
+        full_screen_window_detector_(options.full_screen_window_detector()) {}
+
+  void CaptureFrame() override;
+
+ private:
+  bool ShouldUseScreenCapturer() override;
+  DesktopRect GetWindowRectInVirtualScreen() override;
+
+  // Returns either selected by user sourceId or sourceId provided by
+  // FullScreenWindowDetector
+  WindowId GetWindowToCapture() const;
+
+  // The region from GetWindowRgn in the desktop coordinate if the region is
+  // rectangular, or the rect from GetWindowRect if the region is not set.
+  DesktopRect window_region_rect_;
+
+  WindowCaptureHelperWin window_capture_helper_;
+
+  bool enumerate_current_process_windows_;
+
+  rtc::scoped_refptr<FullScreenWindowDetector> full_screen_window_detector_;
+
+  // Used to make sure that we only log the usage of fullscreen detection once.
+  mutable bool fullscreen_usage_logged_ = false;
+};
+
+void CroppingWindowCapturerWin::CaptureFrame() {
+  DesktopCapturer* win_capturer = window_capturer();
+  if (win_capturer) {
+    // Feed the actual list of windows into full screen window detector.
+    if (full_screen_window_detector_) {
+      full_screen_window_detector_->UpdateWindowListIfNeeded(
+          selected_window(), [this](DesktopCapturer::SourceList* sources) {
+            // Get the list of top level windows, including ones with empty
+            // title. win_capturer_->GetSourceList can't be used here
+            // cause it filters out the windows with empty titles and
+            // it uses responsiveness check which could lead to performance
+            // issues.
+            SourceList result;
+            int window_list_flags =
+                enumerate_current_process_windows_
+                    ? GetWindowListFlags::kNone
+                    : GetWindowListFlags::kIgnoreCurrentProcessWindows;
+
+            if (!webrtc::GetWindowList(window_list_flags, &result))
+              return false;
+
+            // Filter out windows not visible on current desktop
+            auto it = std::remove_if(
+                result.begin(), result.end(), [this](const auto& source) {
+                  HWND hwnd = reinterpret_cast<HWND>(source.id);
+                  return !window_capture_helper_
+                              .IsWindowVisibleOnCurrentDesktop(hwnd);
+                });
+            result.erase(it, result.end());
+
+            sources->swap(result);
+            return true;
+          });
+    }
+    win_capturer->SelectSource(GetWindowToCapture());
+  }
+
+  CroppingWindowCapturer::CaptureFrame();
+}
+
+bool CroppingWindowCapturerWin::ShouldUseScreenCapturer() {
+  if (rtc::rtc_win::GetVersion() < rtc::rtc_win::Version::VERSION_WIN8 &&
+      window_capture_helper_.IsAeroEnabled()) {
+    return false;
+  }
+
+  const HWND selected = reinterpret_cast<HWND>(GetWindowToCapture());
+  // Check if the window is visible on current desktop.
+  if (!window_capture_helper_.IsWindowVisibleOnCurrentDesktop(selected)) {
+    return false;
+  }
+
+  // Check if the window is a translucent layered window.
+  const LONG window_ex_style = GetWindowLong(selected, GWL_EXSTYLE);
+  if (window_ex_style & WS_EX_LAYERED) {
+    COLORREF color_ref_key = 0;
+    BYTE alpha = 0;
+    DWORD flags = 0;
+
+    // GetLayeredWindowAttributes fails if the window was setup with
+    // UpdateLayeredWindow. We have no way to know the opacity of the window in
+    // that case. This happens for Stiky Note (crbug/412726).
+    if (!GetLayeredWindowAttributes(selected, &color_ref_key, &alpha, &flags))
+      return false;
+
+    // UpdateLayeredWindow is the only way to set per-pixel alpha and will cause
+    // the previous GetLayeredWindowAttributes to fail. So we only need to check
+    // the window wide color key or alpha.
+    if ((flags & LWA_COLORKEY) || ((flags & LWA_ALPHA) && (alpha < 255))) {
+      return false;
+    }
+  }
+
+  if (!GetWindowRect(selected, &window_region_rect_)) {
+    return false;
+  }
+
+  DesktopRect content_rect;
+  if (!GetWindowContentRect(selected, &content_rect)) {
+    return false;
+  }
+
+  DesktopRect region_rect;
+  // Get the window region and check if it is rectangular.
+  const int region_type =
+      GetWindowRegionTypeWithBoundary(selected, &region_rect);
+
+  // Do not use the screen capturer if the region is empty or not rectangular.
+  if (region_type == COMPLEXREGION || region_type == NULLREGION) {
+    return false;
+  }
+
+  if (region_type == SIMPLEREGION) {
+    // The `region_rect` returned from GetRgnBox() is always in window
+    // coordinate.
+    region_rect.Translate(window_region_rect_.left(),
+                          window_region_rect_.top());
+    // MSDN: The window region determines the area *within* the window where the
+    // system permits drawing.
+    // https://msdn.microsoft.com/en-us/library/windows/desktop/dd144950(v=vs.85).aspx.
+    //
+    // `region_rect` should always be inside of `window_region_rect_`. So after
+    // the intersection, `window_region_rect_` == `region_rect`. If so, what's
+    // the point of the intersecting operations? Why cannot we directly retrieve
+    // `window_region_rect_` from GetWindowRegionTypeWithBoundary() function?
+    // TODO(zijiehe): Figure out the purpose of these intersections.
+    window_region_rect_.IntersectWith(region_rect);
+    content_rect.IntersectWith(region_rect);
+  }
+
+  // Check if the client area is out of the screen area. When the window is
+  // maximized, only its client area is visible in the screen, the border will
+  // be hidden. So we are using `content_rect` here.
+  if (!GetFullscreenRect().ContainsRect(content_rect)) {
+    return false;
+  }
+
+  // Check if the window is occluded by any other window, excluding the child
+  // windows, context menus, and `excluded_window_`.
+  // `content_rect` is preferred, see the comments on
+  // IsWindowIntersectWithSelectedWindow().
+  TopWindowVerifierContext context(selected,
+                                   reinterpret_cast<HWND>(excluded_window()),
+                                   content_rect, &window_capture_helper_);
+  return context.IsTopWindow();
+}
+
+DesktopRect CroppingWindowCapturerWin::GetWindowRectInVirtualScreen() {
+  TRACE_EVENT0("webrtc",
+               "CroppingWindowCapturerWin::GetWindowRectInVirtualScreen");
+  DesktopRect window_rect;
+  HWND hwnd = reinterpret_cast<HWND>(GetWindowToCapture());
+  if (!GetCroppedWindowRect(hwnd, /*avoid_cropping_border*/ false, &window_rect,
+                            /*original_rect*/ nullptr)) {
+    RTC_LOG(LS_WARNING) << "Failed to get window info: " << GetLastError();
+    return window_rect;
+  }
+  window_rect.IntersectWith(window_region_rect_);
+
+  // Convert `window_rect` to be relative to the top-left of the virtual screen.
+  DesktopRect screen_rect(GetFullscreenRect());
+  window_rect.IntersectWith(screen_rect);
+  window_rect.Translate(-screen_rect.left(), -screen_rect.top());
+  return window_rect;
+}
+
+WindowId CroppingWindowCapturerWin::GetWindowToCapture() const {
+  const auto selected_source = selected_window();
+  const auto full_screen_source =
+      full_screen_window_detector_
+          ? full_screen_window_detector_->FindFullScreenWindow(selected_source)
+          : 0;
+  if (full_screen_source && full_screen_source != selected_source &&
+      !fullscreen_usage_logged_) {
+    fullscreen_usage_logged_ = true;
+    LogDesktopCapturerFullscreenDetectorUsage();
+  }
+  return full_screen_source ? full_screen_source : selected_source;
+}
+
+}  // namespace
+
+// static
+std::unique_ptr<DesktopCapturer> CroppingWindowCapturer::CreateCapturer(
+    const DesktopCaptureOptions& options) {
+  std::unique_ptr<DesktopCapturer> capturer(
+      new CroppingWindowCapturerWin(options));
+  if (capturer && options.detect_updated_region()) {
+    capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
+  }
+
+  return capturer;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/delegated_source_list_controller.h b/third_party/libwebrtc/modules/desktop_capture/delegated_source_list_controller.h
new file mode 100644
index 0000000000..cada7dc817
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/delegated_source_list_controller.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DELEGATED_SOURCE_LIST_CONTROLLER_H_
+#define MODULES_DESKTOP_CAPTURE_DELEGATED_SOURCE_LIST_CONTROLLER_H_
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// A controller to be implemented and returned by
+// GetDelegatedSourceListController in capturers that require showing their own
+// source list and managing user selection there. Apart from ensuring the
+// visibility of the source list, these capturers should largely be interacted
+// with the same as a normal capturer, though there may be some caveats for
+// some DesktopCapturer methods. See GetDelegatedSourceListController for more
+// information.
+class RTC_EXPORT DelegatedSourceListController {
+ public:
+  // Notifications that can be used to help drive any UI that the consumer may
+  // want to show around this source list (e.g. if an consumer shows their own
+  // UI in addition to the delegated source list).
+  class Observer {
+   public:
+    // Called after the user has made a selection in the delegated source list.
+    // Note that the consumer will still need to get the source out of the
+    // capturer by calling GetSourceList.
+    virtual void OnSelection() = 0;
+
+    // Called when there is any user action that cancels the source selection.
+    virtual void OnCancelled() = 0;
+
+    // Called when there is a system error that cancels the source selection.
+    virtual void OnError() = 0;
+
+   protected:
+    virtual ~Observer() {}
+  };
+
+  // Observer must remain valid until the owning DesktopCapturer is destroyed.
+  // Only one Observer is allowed at a time, and may be cleared by passing
+  // nullptr.
+  virtual void Observe(Observer* observer) = 0;
+
+  // Used to prompt the capturer to show the delegated source list. If the
+  // source list is already visible, this will be a no-op. Must be called after
+  // starting the DesktopCapturer.
+  //
+  // Note that any selection from a previous invocation of the source list may
+  // be cleared when this method is called.
+  virtual void EnsureVisible() = 0;
+
+  // Used to prompt the capturer to hide the delegated source list. If the
+  // source list is already hidden, this will be a no-op.
+  virtual void EnsureHidden() = 0;
+
+ protected:
+  virtual ~DelegatedSourceListController() {}
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DELEGATED_SOURCE_LIST_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
new file mode 100644
index 0000000000..dd688ac5f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
@@ -0,0 +1,260 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_and_cursor_composer.h"
+
+#include <stdint.h>
+#include <string.h>
+
+#include <memory>
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Helper function that blends one image into another. Source image must be
+// pre-multiplied with the alpha channel. Destination is assumed to be opaque.
+void AlphaBlend(uint8_t* dest,
+                int dest_stride,
+                const uint8_t* src,
+                int src_stride,
+                const DesktopSize& size) {
+  for (int y = 0; y < size.height(); ++y) {
+    for (int x = 0; x < size.width(); ++x) {
+      uint32_t base_alpha = 255 - src[x * DesktopFrame::kBytesPerPixel + 3];
+      if (base_alpha == 255) {
+        continue;
+      } else if (base_alpha == 0) {
+        memcpy(dest + x * DesktopFrame::kBytesPerPixel,
+               src + x * DesktopFrame::kBytesPerPixel,
+               DesktopFrame::kBytesPerPixel);
+      } else {
+        dest[x * DesktopFrame::kBytesPerPixel] =
+            dest[x * DesktopFrame::kBytesPerPixel] * base_alpha / 255 +
+            src[x * DesktopFrame::kBytesPerPixel];
+        dest[x * DesktopFrame::kBytesPerPixel + 1] =
+            dest[x * DesktopFrame::kBytesPerPixel + 1] * base_alpha / 255 +
+            src[x * DesktopFrame::kBytesPerPixel + 1];
+        dest[x * DesktopFrame::kBytesPerPixel + 2] =
+            dest[x * DesktopFrame::kBytesPerPixel + 2] * base_alpha / 255 +
+            src[x * DesktopFrame::kBytesPerPixel + 2];
+      }
+    }
+    src += src_stride;
+    dest += dest_stride;
+  }
+}
+
+// DesktopFrame wrapper that draws mouse on a frame and restores original
+// content before releasing the underlying frame.
+class DesktopFrameWithCursor : public DesktopFrame {
+ public:
+  // Takes ownership of `frame`.
+  DesktopFrameWithCursor(std::unique_ptr<DesktopFrame> frame,
+                         const MouseCursor& cursor,
+                         const DesktopVector& position,
+                         const DesktopRect& previous_cursor_rect,
+                         bool cursor_changed);
+  ~DesktopFrameWithCursor() override;
+
+  DesktopFrameWithCursor(const DesktopFrameWithCursor&) = delete;
+  DesktopFrameWithCursor& operator=(const DesktopFrameWithCursor&) = delete;
+
+  DesktopRect cursor_rect() const { return cursor_rect_; }
+
+ private:
+  const std::unique_ptr<DesktopFrame> original_frame_;
+
+  DesktopVector restore_position_;
+  std::unique_ptr<DesktopFrame> restore_frame_;
+  DesktopRect cursor_rect_;
+};
+
+DesktopFrameWithCursor::DesktopFrameWithCursor(
+    std::unique_ptr<DesktopFrame> frame,
+    const MouseCursor& cursor,
+    const DesktopVector& position,
+    const DesktopRect& previous_cursor_rect,
+    bool cursor_changed)
+    : DesktopFrame(frame->size(),
+                   frame->stride(),
+                   frame->data(),
+                   frame->shared_memory()),
+      original_frame_(std::move(frame)) {
+  MoveFrameInfoFrom(original_frame_.get());
+
+  DesktopVector image_pos = position.subtract(cursor.hotspot());
+  cursor_rect_ = DesktopRect::MakeSize(cursor.image()->size());
+  cursor_rect_.Translate(image_pos);
+  DesktopVector cursor_origin = cursor_rect_.top_left();
+  cursor_rect_.IntersectWith(DesktopRect::MakeSize(size()));
+
+  if (!previous_cursor_rect.equals(cursor_rect_)) {
+    mutable_updated_region()->AddRect(cursor_rect_);
+    // TODO(crbug:1323241) Update this code to properly handle the case where
+    // |previous_cursor_rect| is outside of the boundaries of |frame|.
+    // Any boundary check has to take into account the fact that
+    // |previous_cursor_rect| can be in DPI or in pixels, based on the platform
+    // we're running on.
+    mutable_updated_region()->AddRect(previous_cursor_rect);
+  } else if (cursor_changed) {
+    mutable_updated_region()->AddRect(cursor_rect_);
+  }
+
+  if (cursor_rect_.is_empty())
+    return;
+
+  // Copy original screen content under cursor to `restore_frame_`.
+  restore_position_ = cursor_rect_.top_left();
+  restore_frame_.reset(new BasicDesktopFrame(cursor_rect_.size()));
+  restore_frame_->CopyPixelsFrom(*this, cursor_rect_.top_left(),
+                                 DesktopRect::MakeSize(restore_frame_->size()));
+
+  // Blit the cursor.
+  uint8_t* cursor_rect_data =
+      reinterpret_cast<uint8_t*>(data()) + cursor_rect_.top() * stride() +
+      cursor_rect_.left() * DesktopFrame::kBytesPerPixel;
+  DesktopVector origin_shift = cursor_rect_.top_left().subtract(cursor_origin);
+  AlphaBlend(cursor_rect_data, stride(),
+             cursor.image()->data() +
+                 origin_shift.y() * cursor.image()->stride() +
+                 origin_shift.x() * DesktopFrame::kBytesPerPixel,
+             cursor.image()->stride(), cursor_rect_.size());
+}
+
+DesktopFrameWithCursor::~DesktopFrameWithCursor() {
+  // Restore original content of the frame.
+  if (restore_frame_) {
+    DesktopRect target_rect = DesktopRect::MakeSize(restore_frame_->size());
+    target_rect.Translate(restore_position_);
+    CopyPixelsFrom(restore_frame_->data(), restore_frame_->stride(),
+                   target_rect);
+  }
+}
+
+}  // namespace
+
+DesktopAndCursorComposer::DesktopAndCursorComposer(
+    std::unique_ptr<DesktopCapturer> desktop_capturer,
+    const DesktopCaptureOptions& options)
+    : DesktopAndCursorComposer(desktop_capturer.release(),
+                               MouseCursorMonitor::Create(options).release()) {}
+
+DesktopAndCursorComposer::DesktopAndCursorComposer(
+    DesktopCapturer* desktop_capturer,
+    MouseCursorMonitor* mouse_monitor)
+    : desktop_capturer_(desktop_capturer), mouse_monitor_(mouse_monitor) {
+  RTC_DCHECK(desktop_capturer_);
+}
+
+DesktopAndCursorComposer::~DesktopAndCursorComposer() = default;
+
+std::unique_ptr<DesktopAndCursorComposer>
+DesktopAndCursorComposer::CreateWithoutMouseCursorMonitor(
+    std::unique_ptr<DesktopCapturer> desktop_capturer) {
+  return std::unique_ptr<DesktopAndCursorComposer>(
+      new DesktopAndCursorComposer(desktop_capturer.release(), nullptr));
+}
+
+void DesktopAndCursorComposer::Start(DesktopCapturer::Callback* callback) {
+  callback_ = callback;
+  if (mouse_monitor_)
+    mouse_monitor_->Init(this, MouseCursorMonitor::SHAPE_AND_POSITION);
+  desktop_capturer_->Start(this);
+}
+
+void DesktopAndCursorComposer::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  desktop_capturer_->SetSharedMemoryFactory(std::move(shared_memory_factory));
+}
+
+void DesktopAndCursorComposer::CaptureFrame() {
+  if (mouse_monitor_)
+    mouse_monitor_->Capture();
+  desktop_capturer_->CaptureFrame();
+}
+
+void DesktopAndCursorComposer::SetExcludedWindow(WindowId window) {
+  desktop_capturer_->SetExcludedWindow(window);
+}
+
+bool DesktopAndCursorComposer::GetSourceList(SourceList* sources) {
+  return desktop_capturer_->GetSourceList(sources);
+}
+
+bool DesktopAndCursorComposer::SelectSource(SourceId id) {
+  return desktop_capturer_->SelectSource(id);
+}
+
+bool DesktopAndCursorComposer::FocusOnSelectedSource() {
+  return desktop_capturer_->FocusOnSelectedSource();
+}
+
+bool DesktopAndCursorComposer::IsOccluded(const DesktopVector& pos) {
+  return desktop_capturer_->IsOccluded(pos);
+}
+
+#if defined(WEBRTC_USE_GIO)
+DesktopCaptureMetadata DesktopAndCursorComposer::GetMetadata() {
+  return desktop_capturer_->GetMetadata();
+}
+#endif  // defined(WEBRTC_USE_GIO)
+
+void DesktopAndCursorComposer::OnCaptureResult(
+    DesktopCapturer::Result result,
+    std::unique_ptr<DesktopFrame> frame) {
+  if (frame && cursor_) {
+    if (!frame->may_contain_cursor() &&
+        frame->rect().Contains(cursor_position_) &&
+        !desktop_capturer_->IsOccluded(cursor_position_)) {
+      DesktopVector relative_position =
+          cursor_position_.subtract(frame->top_left());
+#if defined(WEBRTC_MAC) || defined(CHROMEOS)
+      // On OSX, the logical(DIP) and physical coordinates are used mixingly.
+      // For example, the captured cursor has its size in physical pixels(2x)
+      // and location in logical(DIP) pixels on Retina monitor. This will cause
+      // problem when the desktop is mixed with Retina and non-Retina monitors.
+      // So we use DIP pixel for all location info and compensate with the scale
+      // factor of current frame to the `relative_position`.
+      const float scale = frame->scale_factor();
+      relative_position.set(relative_position.x() * scale,
+                            relative_position.y() * scale);
+#endif
+      auto frame_with_cursor = std::make_unique<DesktopFrameWithCursor>(
+          std::move(frame), *cursor_, relative_position, previous_cursor_rect_,
+          cursor_changed_);
+      previous_cursor_rect_ = frame_with_cursor->cursor_rect();
+      cursor_changed_ = false;
+      frame = std::move(frame_with_cursor);
+      frame->set_may_contain_cursor(true);
+    }
+  }
+
+  callback_->OnCaptureResult(result, std::move(frame));
+}
+
+void DesktopAndCursorComposer::OnMouseCursor(MouseCursor* cursor) {
+  cursor_changed_ = true;
+  cursor_.reset(cursor);
+}
+
+void DesktopAndCursorComposer::OnMouseCursorPosition(
+    const DesktopVector& position) {
+  cursor_position_ = position;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer.h b/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer.h
new file mode 100644
index 0000000000..a078b3eeef
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer.h
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_AND_CURSOR_COMPOSER_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_AND_CURSOR_COMPOSER_H_
+
+#include <memory>
+#if defined(WEBRTC_USE_GIO)
+#include "modules/desktop_capture/desktop_capture_metadata.h"
+#endif  // defined(WEBRTC_USE_GIO)
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+#include "modules/desktop_capture/shared_memory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// A wrapper for DesktopCapturer that also captures mouse using specified
+// MouseCursorMonitor and renders it on the generated streams.
+class RTC_EXPORT DesktopAndCursorComposer
+    : public DesktopCapturer,
+      public DesktopCapturer::Callback,
+      public MouseCursorMonitor::Callback {
+ public:
+  // Creates a new composer that captures mouse cursor using
+  // MouseCursorMonitor::Create(options) and renders it into the frames
+  // generated by `desktop_capturer`.
+  DesktopAndCursorComposer(std::unique_ptr<DesktopCapturer> desktop_capturer,
+                           const DesktopCaptureOptions& options);
+
+  ~DesktopAndCursorComposer() override;
+
+  DesktopAndCursorComposer(const DesktopAndCursorComposer&) = delete;
+  DesktopAndCursorComposer& operator=(const DesktopAndCursorComposer&) = delete;
+
+  // Creates a new composer that relies on an external source for cursor shape
+  // and position information via the MouseCursorMonitor::Callback interface.
+  static std::unique_ptr<DesktopAndCursorComposer>
+  CreateWithoutMouseCursorMonitor(
+      std::unique_ptr<DesktopCapturer> desktop_capturer);
+
+  // DesktopCapturer interface.
+  void Start(DesktopCapturer::Callback* callback) override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  void CaptureFrame() override;
+  void SetExcludedWindow(WindowId window) override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  bool IsOccluded(const DesktopVector& pos) override;
+#if defined(WEBRTC_USE_GIO)
+  DesktopCaptureMetadata GetMetadata() override;
+#endif  // defined(WEBRTC_USE_GIO)
+
+  // MouseCursorMonitor::Callback interface.
+  void OnMouseCursor(MouseCursor* cursor) override;
+  void OnMouseCursorPosition(const DesktopVector& position) override;
+
+ private:
+  // Allows test cases to use a fake MouseCursorMonitor implementation.
+  friend class DesktopAndCursorComposerTest;
+
+  // Constructor to delegate both deprecated and new constructors and allows
+  // test cases to use a fake MouseCursorMonitor implementation.
+  DesktopAndCursorComposer(DesktopCapturer* desktop_capturer,
+                           MouseCursorMonitor* mouse_monitor);
+
+  // DesktopCapturer::Callback interface.
+  void OnCaptureResult(DesktopCapturer::Result result,
+                       std::unique_ptr<DesktopFrame> frame) override;
+
+  const std::unique_ptr<DesktopCapturer> desktop_capturer_;
+  const std::unique_ptr<MouseCursorMonitor> mouse_monitor_;
+
+  DesktopCapturer::Callback* callback_;
+
+  std::unique_ptr<MouseCursor> cursor_;
+  DesktopVector cursor_position_;
+  DesktopRect previous_cursor_rect_;
+  bool cursor_changed_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_AND_CURSOR_COMPOSER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
new file mode 100644
index 0000000000..179e002bc5
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
@@ -0,0 +1,479 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_and_cursor_composer.h"
+
+#include <stdint.h>
+#include <string.h>
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "rtc_base/arraysize.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using testing::ElementsAre;
+
+const int kFrameXCoord = 100;
+const int kFrameYCoord = 200;
+const int kScreenWidth = 100;
+const int kScreenHeight = 100;
+const int kCursorWidth = 10;
+const int kCursorHeight = 10;
+
+const int kTestCursorSize = 3;
+const uint32_t kTestCursorData[kTestCursorSize][kTestCursorSize] = {
+    {
+        0xffffffff,
+        0x99990000,
+        0xaa222222,
+    },
+    {
+        0x88008800,
+        0xaa0000aa,
+        0xaa333333,
+    },
+    {
+        0x00000000,
+        0xaa0000aa,
+        0xaa333333,
+    },
+};
+
+uint32_t GetFakeFramePixelValue(const DesktopVector& p) {
+  uint32_t r = 100 + p.x();
+  uint32_t g = 100 + p.y();
+  uint32_t b = 100 + p.x() + p.y();
+  return b + (g << 8) + (r << 16) + 0xff000000;
+}
+
+uint32_t GetFramePixel(const DesktopFrame& frame, const DesktopVector& pos) {
+  return *reinterpret_cast<uint32_t*>(frame.GetFrameDataAtPos(pos));
+}
+
+// Blends two pixel values taking into account alpha.
+uint32_t BlendPixels(uint32_t dest, uint32_t src) {
+  uint8_t alpha = 255 - ((src & 0xff000000) >> 24);
+  uint32_t r =
+      ((dest & 0x00ff0000) >> 16) * alpha / 255 + ((src & 0x00ff0000) >> 16);
+  uint32_t g =
+      ((dest & 0x0000ff00) >> 8) * alpha / 255 + ((src & 0x0000ff00) >> 8);
+  uint32_t b = (dest & 0x000000ff) * alpha / 255 + (src & 0x000000ff);
+  return b + (g << 8) + (r << 16) + 0xff000000;
+}
+
+DesktopFrame* CreateTestFrame(int width = kScreenWidth,
+                              int height = kScreenHeight) {
+  DesktopFrame* frame = new BasicDesktopFrame(DesktopSize(width, height));
+  uint32_t* data = reinterpret_cast<uint32_t*>(frame->data());
+  for (int y = 0; y < height; ++y) {
+    for (int x = 0; x < width; ++x) {
+      *(data++) = GetFakeFramePixelValue(DesktopVector(x, y));
+    }
+  }
+  return frame;
+}
+
+MouseCursor* CreateTestCursor(DesktopVector hotspot) {
+  std::unique_ptr<DesktopFrame> image(
+      new BasicDesktopFrame(DesktopSize(kCursorWidth, kCursorHeight)));
+  uint32_t* data = reinterpret_cast<uint32_t*>(image->data());
+  // Set four pixels near the hotspot and leave all other blank.
+  for (int y = 0; y < kTestCursorSize; ++y) {
+    for (int x = 0; x < kTestCursorSize; ++x) {
+      data[(hotspot.y() + y) * kCursorWidth + (hotspot.x() + x)] =
+          kTestCursorData[y][x];
+    }
+  }
+  return new MouseCursor(image.release(), hotspot);
+}
+
+class FakeScreenCapturer : public DesktopCapturer {
+ public:
+  FakeScreenCapturer() {}
+
+  void Start(Callback* callback) override { callback_ = callback; }
+
+  void CaptureFrame() override {
+    callback_->OnCaptureResult(
+        next_frame_ ? Result::SUCCESS : Result::ERROR_TEMPORARY,
+        std::move(next_frame_));
+  }
+
+  void SetNextFrame(std::unique_ptr<DesktopFrame> next_frame) {
+    next_frame_ = std::move(next_frame);
+  }
+
+  bool IsOccluded(const DesktopVector& pos) override { return is_occluded_; }
+
+  void set_is_occluded(bool value) { is_occluded_ = value; }
+
+ private:
+  Callback* callback_ = nullptr;
+
+  std::unique_ptr<DesktopFrame> next_frame_;
+  bool is_occluded_ = false;
+};
+
+class FakeMouseMonitor : public MouseCursorMonitor {
+ public:
+  FakeMouseMonitor() : changed_(true) {}
+
+  void SetState(CursorState state, const DesktopVector& pos) {
+    state_ = state;
+    position_ = pos;
+  }
+
+  void SetHotspot(const DesktopVector& hotspot) {
+    if (!hotspot_.equals(hotspot))
+      changed_ = true;
+    hotspot_ = hotspot;
+  }
+
+  void Init(Callback* callback, Mode mode) override { callback_ = callback; }
+
+  void Capture() override {
+    if (changed_) {
+      callback_->OnMouseCursor(CreateTestCursor(hotspot_));
+    }
+    callback_->OnMouseCursorPosition(position_);
+  }
+
+ private:
+  Callback* callback_;
+  CursorState state_;
+  DesktopVector position_;
+  DesktopVector hotspot_;
+  bool changed_;
+};
+
+void VerifyFrame(const DesktopFrame& frame,
+                 MouseCursorMonitor::CursorState state,
+                 const DesktopVector& pos) {
+  // Verify that all other pixels are set to their original values.
+  DesktopRect image_rect =
+      DesktopRect::MakeWH(kTestCursorSize, kTestCursorSize);
+  image_rect.Translate(pos);
+
+  for (int y = 0; y < kScreenHeight; ++y) {
+    for (int x = 0; x < kScreenWidth; ++x) {
+      DesktopVector p(x, y);
+      if (state == MouseCursorMonitor::INSIDE && image_rect.Contains(p)) {
+        EXPECT_EQ(BlendPixels(GetFakeFramePixelValue(p),
+                              kTestCursorData[y - pos.y()][x - pos.x()]),
+                  GetFramePixel(frame, p));
+      } else {
+        EXPECT_EQ(GetFakeFramePixelValue(p), GetFramePixel(frame, p));
+      }
+    }
+  }
+}
+
+}  // namespace
+
+bool operator==(const DesktopRect& left, const DesktopRect& right) {
+  return left.equals(right);
+}
+
+std::ostream& operator<<(std::ostream& out, const DesktopRect& rect) {
+  out << "{" << rect.left() << "+" << rect.top() << "-" << rect.width() << "x"
+      << rect.height() << "}";
+  return out;
+}
+
+class DesktopAndCursorComposerTest : public ::testing::Test,
+                                     public DesktopCapturer::Callback {
+ public:
+  explicit DesktopAndCursorComposerTest(bool include_cursor = true)
+      : fake_screen_(new FakeScreenCapturer()),
+        fake_cursor_(include_cursor ? new FakeMouseMonitor() : nullptr),
+        blender_(fake_screen_, fake_cursor_) {
+    blender_.Start(this);
+  }
+
+  // DesktopCapturer::Callback interface
+  void OnCaptureResult(DesktopCapturer::Result result,
+                       std::unique_ptr<DesktopFrame> frame) override {
+    frame_ = std::move(frame);
+  }
+
+ protected:
+  // Owned by `blender_`.
+  FakeScreenCapturer* fake_screen_;
+  FakeMouseMonitor* fake_cursor_;
+
+  DesktopAndCursorComposer blender_;
+  std::unique_ptr<DesktopFrame> frame_;
+};
+
+class DesktopAndCursorComposerNoCursorMonitorTest
+    : public DesktopAndCursorComposerTest {
+ public:
+  DesktopAndCursorComposerNoCursorMonitorTest()
+      : DesktopAndCursorComposerTest(false) {}
+};
+
+TEST_F(DesktopAndCursorComposerTest, CursorShouldBeIgnoredIfNoFrameCaptured) {
+  struct {
+    int x, y;
+    int hotspot_x, hotspot_y;
+    bool inside;
+  } tests[] = {
+      {0, 0, 0, 0, true},    {50, 50, 0, 0, true},   {100, 50, 0, 0, true},
+      {50, 100, 0, 0, true}, {100, 100, 0, 0, true}, {0, 0, 2, 5, true},
+      {1, 1, 2, 5, true},    {50, 50, 2, 5, true},   {100, 100, 2, 5, true},
+      {0, 0, 5, 2, true},    {50, 50, 5, 2, true},   {100, 100, 5, 2, true},
+      {0, 0, 0, 0, false},
+  };
+
+  for (size_t i = 0; i < arraysize(tests); i++) {
+    SCOPED_TRACE(i);
+
+    DesktopVector hotspot(tests[i].hotspot_x, tests[i].hotspot_y);
+    fake_cursor_->SetHotspot(hotspot);
+
+    MouseCursorMonitor::CursorState state = tests[i].inside
+                                                ? MouseCursorMonitor::INSIDE
+                                                : MouseCursorMonitor::OUTSIDE;
+    DesktopVector pos(tests[i].x, tests[i].y);
+    fake_cursor_->SetState(state, pos);
+
+    std::unique_ptr<SharedDesktopFrame> frame(
+        SharedDesktopFrame::Wrap(CreateTestFrame()));
+
+    blender_.CaptureFrame();
+    // If capturer captured nothing, then cursor should be ignored, not matter
+    // its state or position.
+    EXPECT_EQ(frame_, nullptr);
+  }
+}
+
+TEST_F(DesktopAndCursorComposerTest, CursorShouldBeIgnoredIfFrameMayContainIt) {
+  // We can't use a shared frame because we need to detect modifications
+  // compared to a control.
+  std::unique_ptr<DesktopFrame> control_frame(CreateTestFrame());
+  control_frame->set_top_left(DesktopVector(kFrameXCoord, kFrameYCoord));
+
+  struct {
+    int x;
+    int y;
+    bool may_contain_cursor;
+  } tests[] = {
+      {100, 200, true},
+      {100, 200, false},
+      {150, 250, true},
+      {150, 250, false},
+  };
+
+  for (size_t i = 0; i < arraysize(tests); i++) {
+    SCOPED_TRACE(i);
+
+    std::unique_ptr<DesktopFrame> frame(CreateTestFrame());
+    frame->set_top_left(DesktopVector(kFrameXCoord, kFrameYCoord));
+    frame->set_may_contain_cursor(tests[i].may_contain_cursor);
+    fake_screen_->SetNextFrame(std::move(frame));
+
+    const DesktopVector abs_pos(tests[i].x, tests[i].y);
+    fake_cursor_->SetState(MouseCursorMonitor::INSIDE, abs_pos);
+    blender_.CaptureFrame();
+
+    // If the frame may already have contained the cursor, then `CaptureFrame()`
+    // should not have modified it, so it should be the same as the control.
+    EXPECT_TRUE(frame_);
+    const DesktopVector rel_pos(abs_pos.subtract(control_frame->top_left()));
+    if (tests[i].may_contain_cursor) {
+      EXPECT_EQ(
+          *reinterpret_cast<uint32_t*>(frame_->GetFrameDataAtPos(rel_pos)),
+          *reinterpret_cast<uint32_t*>(
+              control_frame->GetFrameDataAtPos(rel_pos)));
+
+    } else {
+      // `CaptureFrame()` should have modified the frame to have the cursor.
+      EXPECT_NE(
+          *reinterpret_cast<uint32_t*>(frame_->GetFrameDataAtPos(rel_pos)),
+          *reinterpret_cast<uint32_t*>(
+              control_frame->GetFrameDataAtPos(rel_pos)));
+      EXPECT_TRUE(frame_->may_contain_cursor());
+    }
+  }
+}
+
+TEST_F(DesktopAndCursorComposerTest,
+       CursorShouldBeIgnoredIfItIsOutOfDesktopFrame) {
+  std::unique_ptr<SharedDesktopFrame> frame(
+      SharedDesktopFrame::Wrap(CreateTestFrame()));
+  frame->set_top_left(DesktopVector(kFrameXCoord, kFrameYCoord));
+  // The frame covers (100, 200) - (200, 300).
+
+  struct {
+    int x;
+    int y;
+  } tests[] = {
+      {0, 0},    {50, 50},         {50, 150},      {100, 150}, {50, 200},
+      {99, 200}, {100, 199},       {200, 300},     {200, 299}, {199, 300},
+      {-1, -1},  {-10000, -10000}, {10000, 10000},
+  };
+  for (size_t i = 0; i < arraysize(tests); i++) {
+    SCOPED_TRACE(i);
+
+    fake_screen_->SetNextFrame(frame->Share());
+    // The CursorState is ignored when using absolute cursor position.
+    fake_cursor_->SetState(MouseCursorMonitor::OUTSIDE,
+                           DesktopVector(tests[i].x, tests[i].y));
+    blender_.CaptureFrame();
+    VerifyFrame(*frame_, MouseCursorMonitor::OUTSIDE, DesktopVector(0, 0));
+  }
+}
+
+TEST_F(DesktopAndCursorComposerTest, IsOccludedShouldBeConsidered) {
+  std::unique_ptr<SharedDesktopFrame> frame(
+      SharedDesktopFrame::Wrap(CreateTestFrame()));
+  frame->set_top_left(DesktopVector(kFrameXCoord, kFrameYCoord));
+  // The frame covers (100, 200) - (200, 300).
+
+  struct {
+    int x;
+    int y;
+  } tests[] = {
+      {100, 200}, {101, 200}, {100, 201}, {101, 201}, {150, 250}, {199, 299},
+  };
+  fake_screen_->set_is_occluded(true);
+  for (size_t i = 0; i < arraysize(tests); i++) {
+    SCOPED_TRACE(i);
+
+    fake_screen_->SetNextFrame(frame->Share());
+    // The CursorState is ignored when using absolute cursor position.
+    fake_cursor_->SetState(MouseCursorMonitor::OUTSIDE,
+                           DesktopVector(tests[i].x, tests[i].y));
+    blender_.CaptureFrame();
+    VerifyFrame(*frame_, MouseCursorMonitor::OUTSIDE, DesktopVector());
+  }
+}
+
+TEST_F(DesktopAndCursorComposerTest, CursorIncluded) {
+  std::unique_ptr<SharedDesktopFrame> frame(
+      SharedDesktopFrame::Wrap(CreateTestFrame()));
+  frame->set_top_left(DesktopVector(kFrameXCoord, kFrameYCoord));
+  // The frame covers (100, 200) - (200, 300).
+
+  struct {
+    int x;
+    int y;
+  } tests[] = {
+      {100, 200}, {101, 200}, {100, 201}, {101, 201}, {150, 250}, {199, 299},
+  };
+  for (size_t i = 0; i < arraysize(tests); i++) {
+    SCOPED_TRACE(i);
+
+    const DesktopVector abs_pos(tests[i].x, tests[i].y);
+    const DesktopVector rel_pos(abs_pos.subtract(frame->top_left()));
+
+    fake_screen_->SetNextFrame(frame->Share());
+    // The CursorState is ignored when using absolute cursor position.
+    fake_cursor_->SetState(MouseCursorMonitor::OUTSIDE, abs_pos);
+    blender_.CaptureFrame();
+    VerifyFrame(*frame_, MouseCursorMonitor::INSIDE, rel_pos);
+
+    // Verify that the cursor is erased before the frame buffer is returned to
+    // the screen capturer.
+    frame_.reset();
+    VerifyFrame(*frame, MouseCursorMonitor::OUTSIDE, DesktopVector());
+  }
+}
+
+TEST_F(DesktopAndCursorComposerNoCursorMonitorTest,
+       UpdatedRegionIncludesOldAndNewCursorRectsIfMoved) {
+  std::unique_ptr<SharedDesktopFrame> frame(
+      SharedDesktopFrame::Wrap(CreateTestFrame()));
+  DesktopRect first_cursor_rect;
+  {
+    // Block to scope test_cursor, which is invalidated by OnMouseCursor.
+    MouseCursor* test_cursor = CreateTestCursor(DesktopVector(0, 0));
+    first_cursor_rect = DesktopRect::MakeSize(test_cursor->image()->size());
+    blender_.OnMouseCursor(test_cursor);
+  }
+  blender_.OnMouseCursorPosition(DesktopVector(0, 0));
+  fake_screen_->SetNextFrame(frame->Share());
+  blender_.CaptureFrame();
+
+  DesktopVector cursor_move_offset(1, 1);
+  DesktopRect second_cursor_rect = first_cursor_rect;
+  second_cursor_rect.Translate(cursor_move_offset);
+  blender_.OnMouseCursorPosition(cursor_move_offset);
+  fake_screen_->SetNextFrame(frame->Share());
+  blender_.CaptureFrame();
+
+  EXPECT_TRUE(frame->updated_region().is_empty());
+  DesktopRegion expected_region;
+  expected_region.AddRect(first_cursor_rect);
+  expected_region.AddRect(second_cursor_rect);
+  EXPECT_TRUE(frame_->updated_region().Equals(expected_region));
+}
+
+TEST_F(DesktopAndCursorComposerNoCursorMonitorTest,
+       UpdatedRegionIncludesOldAndNewCursorRectsIfShapeChanged) {
+  std::unique_ptr<SharedDesktopFrame> frame(
+      SharedDesktopFrame::Wrap(CreateTestFrame()));
+  DesktopRect first_cursor_rect;
+  {
+    // Block to scope test_cursor, which is invalidated by OnMouseCursor.
+    MouseCursor* test_cursor = CreateTestCursor(DesktopVector(0, 0));
+    first_cursor_rect = DesktopRect::MakeSize(test_cursor->image()->size());
+    blender_.OnMouseCursor(test_cursor);
+  }
+  blender_.OnMouseCursorPosition(DesktopVector(0, 0));
+  fake_screen_->SetNextFrame(frame->Share());
+  blender_.CaptureFrame();
+
+  // Create a second cursor, the same shape as the first. Since the code doesn't
+  // compare the cursor pixels, this is sufficient, and avoids needing two test
+  // cursor bitmaps.
+  DesktopRect second_cursor_rect;
+  {
+    MouseCursor* test_cursor = CreateTestCursor(DesktopVector(0, 0));
+    second_cursor_rect = DesktopRect::MakeSize(test_cursor->image()->size());
+    blender_.OnMouseCursor(test_cursor);
+  }
+  fake_screen_->SetNextFrame(frame->Share());
+  blender_.CaptureFrame();
+
+  EXPECT_TRUE(frame->updated_region().is_empty());
+  DesktopRegion expected_region;
+  expected_region.AddRect(first_cursor_rect);
+  expected_region.AddRect(second_cursor_rect);
+  EXPECT_TRUE(frame_->updated_region().Equals(expected_region));
+}
+
+TEST_F(DesktopAndCursorComposerNoCursorMonitorTest,
+       UpdatedRegionUnchangedIfCursorUnchanged) {
+  std::unique_ptr<SharedDesktopFrame> frame(
+      SharedDesktopFrame::Wrap(CreateTestFrame()));
+  blender_.OnMouseCursor(CreateTestCursor(DesktopVector(0, 0)));
+  blender_.OnMouseCursorPosition(DesktopVector(0, 0));
+  fake_screen_->SetNextFrame(frame->Share());
+  blender_.CaptureFrame();
+  fake_screen_->SetNextFrame(frame->Share());
+  blender_.CaptureFrame();
+
+  EXPECT_TRUE(frame->updated_region().is_empty());
+  EXPECT_TRUE(frame_->updated_region().is_empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capture_differ_sse2_gn/moz.build b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_differ_sse2_gn/moz.build
new file mode 100644
index 0000000000..06e178901e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_differ_sse2_gn/moz.build
@@ -0,0 +1,153 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_AVX2"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/desktop_capture/differ_vector_sse2.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2",
+        "-msse2"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+Library("desktop_capture_differ_sse2_gn")
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capture_gn/moz.build b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_gn/moz.build
new file mode 100644
index 0000000000..8c56b6b8e5
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_gn/moz.build
@@ -0,0 +1,678 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+if CONFIG["MOZ_WIDGET_TOOLKIT"] == "gtk":
+    CXXFLAGS += CONFIG["MOZ_GTK3_CFLAGS"]
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/media/libyuv/",
+    "/media/libyuv/libyuv/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/cropped_desktop_frame.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_and_cursor_composer.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metrics_helper.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_capture_options.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_capturer.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_wrapper.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_frame_generator.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/differ_block.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/fake_desktop_capturer.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/full_screen_application_handler.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/full_screen_window_detector.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/mouse_cursor.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/resolution_tracker.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/rgba_color.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/window_finder.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/mac/full_screen_mac_application_handler.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/mac/window_list_utils.cc"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_linux.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/screen_capturer_linux.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/window_capturer_linux.cc"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["WEBRTC_USE_X11"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "X11",
+        "Xcomposite",
+        "Xdamage",
+        "Xext",
+        "Xfixes",
+        "Xrandr",
+        "Xrender",
+        "Xtst"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_linux.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/screen_capturer_linux.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/window_capturer_linux.cc"
+    ]
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "d3d11",
+        "dxgi",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/cropping_window_capturer_win.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/desktop_frame_win.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/screen_capturer_win.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/cursor.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/d3d_device.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/desktop.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/desktop_capture_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/display_configuration_monitor.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/dxgi_context.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/dxgi_frame.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/dxgi_output_duplicator.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_mapping.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_staging.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/full_screen_win_application_handler.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/scoped_thread_desktop.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/selected_window_context.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/win/window_capturer_win_gdi.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/window_capturer_win.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/window_finder_win.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+    DEFINES["WEBRTC_USE_PIPEWIRE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+    LOCAL_INCLUDES += [
+        "/gfx/angle/checkout/include/",
+        "/third_party/drm/drm/",
+        "/third_party/drm/drm/include/",
+        "/third_party/drm/drm/include/libdrm/",
+        "/third_party/gbm/gbm/",
+        "/third_party/libepoxy/libepoxy/include/",
+        "/third_party/pipewire/"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["WEBRTC_USE_PIPEWIRE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+    LOCAL_INCLUDES += [
+        "/gfx/angle/checkout/include/",
+        "/third_party/drm/drm/",
+        "/third_party/drm/drm/include/",
+        "/third_party/drm/drm/include/libdrm/",
+        "/third_party/gbm/gbm/",
+        "/third_party/libepoxy/libepoxy/include/",
+        "/third_party/pipewire/"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["WEBRTC_USE_PIPEWIRE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+    LOCAL_INCLUDES += [
+        "/gfx/angle/checkout/include/",
+        "/third_party/drm/drm/",
+        "/third_party/drm/drm/include/",
+        "/third_party/drm/drm/include/libdrm/",
+        "/third_party/gbm/gbm/",
+        "/third_party/libepoxy/libepoxy/include/",
+        "/third_party/pipewire/"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "ppc64":
+
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_USE_X11"] = True
+
+    OS_LIBS += [
+        "X11",
+        "Xcomposite",
+        "Xdamage",
+        "Xext",
+        "Xfixes",
+        "Xrandr",
+        "Xrender",
+        "Xtst"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "riscv64":
+
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_USE_X11"] = True
+
+    OS_LIBS += [
+        "X11",
+        "Xcomposite",
+        "Xdamage",
+        "Xext",
+        "Xfixes",
+        "Xrandr",
+        "Xrender",
+        "Xtst"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["WEBRTC_USE_PIPEWIRE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+    LOCAL_INCLUDES += [
+        "/gfx/angle/checkout/include/",
+        "/third_party/drm/drm/",
+        "/third_party/drm/drm/include/",
+        "/third_party/drm/drm/include/libdrm/",
+        "/third_party/gbm/gbm/",
+        "/third_party/libepoxy/libepoxy/include/",
+        "/third_party/pipewire/"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["WEBRTC_USE_PIPEWIRE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+    LOCAL_INCLUDES += [
+        "/gfx/angle/checkout/include/",
+        "/third_party/drm/drm/",
+        "/third_party/drm/drm/include/",
+        "/third_party/drm/drm/include/libdrm/",
+        "/third_party/gbm/gbm/",
+        "/third_party/libepoxy/libepoxy/include/",
+        "/third_party/pipewire/"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["WEBRTC_USE_PIPEWIRE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+    LOCAL_INCLUDES += [
+        "/gfx/angle/checkout/include/",
+        "/third_party/drm/drm/",
+        "/third_party/drm/drm/include/",
+        "/third_party/drm/drm/include/libdrm/",
+        "/third_party/gbm/gbm/",
+        "/third_party/libepoxy/libepoxy/include/",
+        "/third_party/pipewire/"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_USE_X11"] = True
+
+    OS_LIBS += [
+        "X11",
+        "Xcomposite",
+        "Xdamage",
+        "Xext",
+        "Xfixes",
+        "Xrandr",
+        "Xrender",
+        "Xtst"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_USE_X11"] = True
+
+    OS_LIBS += [
+        "X11",
+        "Xcomposite",
+        "Xdamage",
+        "Xext",
+        "Xfixes",
+        "Xrandr",
+        "Xrender",
+        "Xtst"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "mips32" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_USE_X11"] = True
+
+    OS_LIBS += [
+        "X11",
+        "Xcomposite",
+        "Xdamage",
+        "Xext",
+        "Xfixes",
+        "Xrandr",
+        "Xrender",
+        "Xtst"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "mips64" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_USE_X11"] = True
+
+    OS_LIBS += [
+        "X11",
+        "Xcomposite",
+        "Xdamage",
+        "Xext",
+        "Xfixes",
+        "Xrandr",
+        "Xrender",
+        "Xtst"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_USE_X11"] = True
+
+    OS_LIBS += [
+        "X11",
+        "Xcomposite",
+        "Xdamage",
+        "Xext",
+        "Xfixes",
+        "Xrandr",
+        "Xrender",
+        "Xtst"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_USE_X11"] = True
+
+    OS_LIBS += [
+        "X11",
+        "Xcomposite",
+        "Xdamage",
+        "Xext",
+        "Xfixes",
+        "Xrandr",
+        "Xrender",
+        "Xtst"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc",
+        "/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc"
+    ]
+
+Library("desktop_capture_gn")
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metadata.h b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metadata.h
new file mode 100644
index 0000000000..49a20e729c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metadata.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_METADATA_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_METADATA_H_
+
+#if defined(WEBRTC_USE_GIO)
+#include "modules/portal/xdg_session_details.h"
+#endif  // defined(WEBRTC_USE_GIO)
+
+namespace webrtc {
+
+// Container for the metadata associated with a desktop capturer.
+struct DesktopCaptureMetadata {
+#if defined(WEBRTC_USE_GIO)
+  // Details about the XDG desktop session handle (used by wayland
+  // implementation in remoting)
+  xdg_portal::SessionDetails session_details;
+#endif  // defined(WEBRTC_USE_GIO)
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_METADATA_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metrics_helper.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metrics_helper.cc
new file mode 100644
index 0000000000..6b741ef4bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metrics_helper.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_capture_metrics_helper.h"
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+// This enum is logged via UMA so entries should not be reordered or have their
+// values changed. This should also be kept in sync with the values in the
+// DesktopCapturerId namespace.
+enum class SequentialDesktopCapturerId {
+  kUnknown = 0,
+  kWgcCapturerWin = 1,
+  kScreenCapturerWinMagnifier = 2,
+  kWindowCapturerWinGdi = 3,
+  kScreenCapturerWinGdi = 4,
+  kScreenCapturerWinDirectx = 5,
+  kMaxValue = kScreenCapturerWinDirectx
+};
+}  // namespace
+
+void RecordCapturerImpl(uint32_t capturer_id) {
+  SequentialDesktopCapturerId sequential_id;
+  switch (capturer_id) {
+    case DesktopCapturerId::kWgcCapturerWin:
+      sequential_id = SequentialDesktopCapturerId::kWgcCapturerWin;
+      break;
+    case DesktopCapturerId::kScreenCapturerWinMagnifier:
+      sequential_id = SequentialDesktopCapturerId::kScreenCapturerWinMagnifier;
+      break;
+    case DesktopCapturerId::kWindowCapturerWinGdi:
+      sequential_id = SequentialDesktopCapturerId::kWindowCapturerWinGdi;
+      break;
+    case DesktopCapturerId::kScreenCapturerWinGdi:
+      sequential_id = SequentialDesktopCapturerId::kScreenCapturerWinGdi;
+      break;
+    case DesktopCapturerId::kScreenCapturerWinDirectx:
+      sequential_id = SequentialDesktopCapturerId::kScreenCapturerWinDirectx;
+      break;
+    case DesktopCapturerId::kUnknown:
+    default:
+      sequential_id = SequentialDesktopCapturerId::kUnknown;
+  }
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.DesktopCapture.Win.DesktopCapturerImpl",
+      static_cast<int>(sequential_id),
+      static_cast<int>(SequentialDesktopCapturerId::kMaxValue));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metrics_helper.h b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metrics_helper.h
new file mode 100644
index 0000000000..37542b84bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_metrics_helper.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_METRICS_HELPER_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_METRICS_HELPER_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+void RecordCapturerImpl(uint32_t capturer_id);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_METRICS_HELPER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capture_objc_gn/moz.build b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_objc_gn/moz.build
new file mode 100644
index 0000000000..2706bed4a6
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_objc_gn/moz.build
@@ -0,0 +1,77 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+CMMFLAGS += [
+    "-fobjc-arc"
+]
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MAC"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_POSIX"] = True
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+DEFINES["__STDC_CONSTANT_MACROS"] = True
+DEFINES["__STDC_FORMAT_MACROS"] = True
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/sdk/objc/",
+    "/third_party/libwebrtc/sdk/objc/base/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration.mm",
+    "/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_cgimage.mm",
+    "/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_iosurface.mm",
+    "/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.mm",
+    "/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.mm",
+    "/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm",
+    "/third_party/libwebrtc/modules/desktop_capture/screen_capturer_darwin.mm",
+    "/third_party/libwebrtc/modules/desktop_capture/window_capturer_mac.mm",
+    "/third_party/libwebrtc/modules/desktop_capture/window_finder_mac.mm"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+Library("desktop_capture_objc_gn")
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capture_options.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_options.cc
new file mode 100644
index 0000000000..22c59ef4cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_options.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+
+#include "api/make_ref_counted.h"
+
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+#include "modules/desktop_capture/mac/full_screen_mac_application_handler.h"
+#elif defined(WEBRTC_WIN)
+#include "modules/desktop_capture/win/full_screen_win_application_handler.h"
+#endif
+#if defined(WEBRTC_USE_PIPEWIRE)
+#include "modules/desktop_capture/linux/wayland/shared_screencast_stream.h"
+#endif
+
+namespace webrtc {
+
+DesktopCaptureOptions::DesktopCaptureOptions() {}
+DesktopCaptureOptions::DesktopCaptureOptions(
+    const DesktopCaptureOptions& options) = default;
+DesktopCaptureOptions::DesktopCaptureOptions(DesktopCaptureOptions&& options) =
+    default;
+DesktopCaptureOptions::~DesktopCaptureOptions() {}
+
+DesktopCaptureOptions& DesktopCaptureOptions::operator=(
+    const DesktopCaptureOptions& options) = default;
+DesktopCaptureOptions& DesktopCaptureOptions::operator=(
+    DesktopCaptureOptions&& options) = default;
+
+// static
+DesktopCaptureOptions DesktopCaptureOptions::CreateDefault() {
+  DesktopCaptureOptions result;
+#if defined(WEBRTC_USE_X11)
+  result.set_x_display(SharedXDisplay::CreateDefault());
+#endif
+#if defined(WEBRTC_USE_PIPEWIRE)
+  result.set_screencast_stream(SharedScreenCastStream::CreateDefault());
+#endif
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+  result.set_configuration_monitor(
+      rtc::make_ref_counted<DesktopConfigurationMonitor>());
+  result.set_full_screen_window_detector(
+      rtc::make_ref_counted<FullScreenWindowDetector>(
+          CreateFullScreenMacApplicationHandler));
+#elif defined(WEBRTC_WIN)
+  result.set_full_screen_window_detector(
+      rtc::make_ref_counted<FullScreenWindowDetector>(
+          CreateFullScreenWinApplicationHandler));
+#endif
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capture_options.h b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_options.h
new file mode 100644
index 0000000000..67dffee08a
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_options.h
@@ -0,0 +1,262 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_OPTIONS_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_OPTIONS_H_
+
+#include "api/scoped_refptr.h"
+#include "rtc_base/system/rtc_export.h"
+
+#if defined(WEBRTC_USE_X11)
+#include "modules/desktop_capture/linux/x11/shared_x_display.h"
+#endif
+
+#if defined(WEBRTC_USE_PIPEWIRE)
+#include "modules/desktop_capture/linux/wayland/shared_screencast_stream.h"
+#endif
+
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
+#endif
+
+#include "modules/desktop_capture/full_screen_window_detector.h"
+
+namespace webrtc {
+
+// An object that stores initialization parameters for screen and window
+// capturers.
+class RTC_EXPORT DesktopCaptureOptions {
+ public:
+  // Returns instance of DesktopCaptureOptions with default parameters. On Linux
+  // also initializes X window connection. x_display() will be set to null if
+  // X11 connection failed (e.g. DISPLAY isn't set).
+  static DesktopCaptureOptions CreateDefault();
+
+  DesktopCaptureOptions();
+  DesktopCaptureOptions(const DesktopCaptureOptions& options);
+  DesktopCaptureOptions(DesktopCaptureOptions&& options);
+  ~DesktopCaptureOptions();
+
+  DesktopCaptureOptions& operator=(const DesktopCaptureOptions& options);
+  DesktopCaptureOptions& operator=(DesktopCaptureOptions&& options);
+
+#if defined(WEBRTC_USE_X11)
+  const rtc::scoped_refptr<SharedXDisplay>& x_display() const {
+    return x_display_;
+  }
+  void set_x_display(rtc::scoped_refptr<SharedXDisplay> x_display) {
+    x_display_ = x_display;
+  }
+#endif
+
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+  // TODO(zijiehe): Remove both DesktopConfigurationMonitor and
+  // FullScreenChromeWindowDetector out of DesktopCaptureOptions. It's not
+  // reasonable for external consumers to set these two parameters.
+  const rtc::scoped_refptr<DesktopConfigurationMonitor>& configuration_monitor()
+      const {
+    return configuration_monitor_;
+  }
+  // If nullptr is set, ScreenCapturer won't work and WindowCapturer may return
+  // inaccurate result from IsOccluded() function.
+  void set_configuration_monitor(
+      rtc::scoped_refptr<DesktopConfigurationMonitor> m) {
+    configuration_monitor_ = m;
+  }
+
+  bool allow_iosurface() const { return allow_iosurface_; }
+  void set_allow_iosurface(bool allow) { allow_iosurface_ = allow; }
+#endif
+
+  const rtc::scoped_refptr<FullScreenWindowDetector>&
+  full_screen_window_detector() const {
+    return full_screen_window_detector_;
+  }
+  void set_full_screen_window_detector(
+      rtc::scoped_refptr<FullScreenWindowDetector> detector) {
+    full_screen_window_detector_ = detector;
+  }
+
+  // Flag indicating that the capturer should use screen change notifications.
+  // Enables/disables use of XDAMAGE in the X11 capturer.
+  bool use_update_notifications() const { return use_update_notifications_; }
+  void set_use_update_notifications(bool use_update_notifications) {
+    use_update_notifications_ = use_update_notifications;
+  }
+
+  // Flag indicating if desktop effects (e.g. Aero) should be disabled when the
+  // capturer is active. Currently used only on Windows.
+  bool disable_effects() const { return disable_effects_; }
+  void set_disable_effects(bool disable_effects) {
+    disable_effects_ = disable_effects;
+  }
+
+  // Flag that should be set if the consumer uses updated_region() and the
+  // capturer should try to provide correct updated_region() for the frames it
+  // generates (e.g. by comparing each frame with the previous one).
+  bool detect_updated_region() const { return detect_updated_region_; }
+  void set_detect_updated_region(bool detect_updated_region) {
+    detect_updated_region_ = detect_updated_region;
+  }
+
+  // Indicates that the capturer should try to include the cursor in the frame.
+  // If it is able to do so it will set `DesktopFrame::may_contain_cursor()`.
+  // Not all capturers will support including the cursor. If this value is false
+  // or the cursor otherwise cannot be included in the frame, then cursor
+  // metadata will be sent, though the capturer may choose to always send cursor
+  // metadata.
+  bool prefer_cursor_embedded() const { return prefer_cursor_embedded_; }
+  void set_prefer_cursor_embedded(bool prefer_cursor_embedded) {
+    prefer_cursor_embedded_ = prefer_cursor_embedded;
+  }
+
+#if defined(WEBRTC_WIN)
+  // Enumerating windows owned by the current process on Windows has some
+  // complications due to |GetWindowText*()| APIs potentially causing a
+  // deadlock (see the comments in the `GetWindowListHandler()` function in
+  // window_capture_utils.cc for more details on the deadlock).
+  // To avoid this issue, consumers can either ensure that the thread that runs
+  // their message loop never waits on `GetSourceList()`, or they can set this
+  // flag to false which will prevent windows running in the current process
+  // from being enumerated and included in the results. Consumers can still
+  // provide the WindowId for their own windows to `SelectSource()` and capture
+  // them.
+  bool enumerate_current_process_windows() const {
+    return enumerate_current_process_windows_;
+  }
+  void set_enumerate_current_process_windows(
+      bool enumerate_current_process_windows) {
+    enumerate_current_process_windows_ = enumerate_current_process_windows;
+  }
+
+  bool allow_use_magnification_api() const {
+    return allow_use_magnification_api_;
+  }
+  void set_allow_use_magnification_api(bool allow) {
+    allow_use_magnification_api_ = allow;
+  }
+  // Allowing directx based capturer or not, this capturer works on windows 7
+  // with platform update / windows 8 or upper.
+  bool allow_directx_capturer() const { return allow_directx_capturer_; }
+  void set_allow_directx_capturer(bool enabled) {
+    allow_directx_capturer_ = enabled;
+  }
+
+  // Flag that may be set to allow use of the cropping window capturer (which
+  // captures the screen & crops that to the window region in some cases). An
+  // advantage of using this is significantly higher capture frame rates than
+  // capturing the window directly. A disadvantage of using this is the
+  // possibility of capturing unrelated content (e.g. overlapping windows that
+  // aren't detected properly, or neighboring regions when moving/resizing the
+  // captured window). Note: this flag influences the behavior of calls to
+  // DesktopCapturer::CreateWindowCapturer; calls to
+  // CroppingWindowCapturer::CreateCapturer ignore the flag (treat it as true).
+  bool allow_cropping_window_capturer() const {
+    return allow_cropping_window_capturer_;
+  }
+  void set_allow_cropping_window_capturer(bool allow) {
+    allow_cropping_window_capturer_ = allow;
+  }
+
+#if defined(RTC_ENABLE_WIN_WGC)
+  // This flag enables the WGC capturer for both window and screen capture.
+  // This capturer should offer similar or better performance than the cropping
+  // capturer without the disadvantages listed above. However, the WGC capturer
+  // is only available on Windows 10 version 1809 (Redstone 5) and up. This flag
+  // will have no affect on older versions.
+  // If set, and running a supported version of Win10, this flag will take
+  // precedence over the cropping, directx, and magnification flags.
+  bool allow_wgc_capturer() const { return allow_wgc_capturer_; }
+  void set_allow_wgc_capturer(bool allow) { allow_wgc_capturer_ = allow; }
+
+  // This flag enables the WGC capturer for fallback capturer.
+  // The flag is useful when the first capturer (eg. WindowCapturerWinGdi) is
+  // unreliable in certain devices where WGC is supported, but not used by
+  // default.
+  bool allow_wgc_capturer_fallback() const {
+    return allow_wgc_capturer_fallback_;
+  }
+  void set_allow_wgc_capturer_fallback(bool allow) {
+    allow_wgc_capturer_fallback_ = allow;
+  }
+#endif  // defined(RTC_ENABLE_WIN_WGC)
+#endif  // defined(WEBRTC_WIN)
+
+#if defined(WEBRTC_USE_PIPEWIRE)
+  bool allow_pipewire() const { return allow_pipewire_; }
+  void set_allow_pipewire(bool allow) { allow_pipewire_ = allow; }
+
+  const rtc::scoped_refptr<SharedScreenCastStream>& screencast_stream() const {
+    return screencast_stream_;
+  }
+  void set_screencast_stream(
+      rtc::scoped_refptr<SharedScreenCastStream> stream) {
+    screencast_stream_ = stream;
+  }
+
+  void set_width(uint32_t width) { width_ = width; }
+  uint32_t get_width() const { return width_; }
+
+  void set_height(uint32_t height) { height_ = height; }
+  uint32_t get_height() const { return height_; }
+
+  void set_pipewire_use_damage_region(bool use_damage_regions) {
+    pipewire_use_damage_region_ = use_damage_regions;
+  }
+  bool pipewire_use_damage_region() const {
+    return pipewire_use_damage_region_;
+  }
+#endif
+
+ private:
+#if defined(WEBRTC_USE_X11)
+  rtc::scoped_refptr<SharedXDisplay> x_display_;
+#endif
+#if defined(WEBRTC_USE_PIPEWIRE)
+  // An instance of shared PipeWire ScreenCast stream we share between
+  // BaseCapturerPipeWire and MouseCursorMonitorPipeWire as cursor information
+  // is sent together with screen content.
+  rtc::scoped_refptr<SharedScreenCastStream> screencast_stream_;
+#endif
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+  rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_;
+  bool allow_iosurface_ = false;
+#endif
+
+  rtc::scoped_refptr<FullScreenWindowDetector> full_screen_window_detector_;
+
+#if defined(WEBRTC_WIN)
+  bool enumerate_current_process_windows_ = true;
+  bool allow_use_magnification_api_ = false;
+  bool allow_directx_capturer_ = false;
+  bool allow_cropping_window_capturer_ = false;
+#if defined(RTC_ENABLE_WIN_WGC)
+  bool allow_wgc_capturer_ = false;
+  bool allow_wgc_capturer_fallback_ = false;
+#endif
+#endif
+#if defined(WEBRTC_USE_X11)
+  bool use_update_notifications_ = false;
+#else
+  bool use_update_notifications_ = true;
+#endif
+  bool disable_effects_ = true;
+  bool detect_updated_region_ = false;
+  bool prefer_cursor_embedded_ = false;
+#if defined(WEBRTC_USE_PIPEWIRE)
+  bool allow_pipewire_ = false;
+  bool pipewire_use_damage_region_ = true;
+  uint32_t width_ = 0;
+  uint32_t height_ = 0;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_OPTIONS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capture_types.h b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_types.h
new file mode 100644
index 0000000000..e777a45f92
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capture_types.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_TYPES_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_TYPES_H_
+
+#include <stdint.h>
+
+#ifdef XP_WIN      // Moving this into the global namespace
+typedef int pid_t; // matching what used to be in
+#endif             // video_capture_defines.h
+
+namespace webrtc {
+
+enum class CaptureType { kWindow, kScreen, kAnyScreenContent };
+
+// Type used to identify windows on the desktop. Values are platform-specific:
+//   - On Windows: HWND cast to intptr_t.
+//   - On Linux (with X11): X11 Window (unsigned long) type cast to intptr_t.
+//   - On OSX: integer window number.
+typedef intptr_t WindowId;
+
+const WindowId kNullWindowId = 0;
+
+const int64_t kInvalidDisplayId = -1;
+
+// Type used to identify screens on the desktop. Values are platform-specific:
+//   - On Windows: integer display device index.
+//   - On OSX: CGDirectDisplayID cast to intptr_t.
+//   - On Linux (with X11): TBD.
+//   - On ChromeOS: display::Display::id() is an int64_t.
+// On Windows, ScreenId is implementation dependent: sending a ScreenId from one
+// implementation to another usually won't work correctly.
+#if defined(CHROMEOS)
+typedef int64_t ScreenId;
+#else
+typedef intptr_t ScreenId;
+#endif
+
+// The screen id corresponds to all screen combined together.
+const ScreenId kFullDesktopScreenId = -1;
+
+const ScreenId kInvalidScreenId = -2;
+
+// Integers to attach to each DesktopFrame to differentiate the generator of
+// the frame. The entries in this namespace should remain in sync with the
+// SequentialDesktopCapturerId enum, which is logged via UMA.
+// `kScreenCapturerWinGdi` and `kScreenCapturerWinDirectx` values are preserved
+// to maintain compatibility
+namespace DesktopCapturerId {
+constexpr uint32_t CreateFourCC(char a, char b, char c, char d) {
+  return ((static_cast<uint32_t>(a)) | (static_cast<uint32_t>(b) << 8) |
+          (static_cast<uint32_t>(c) << 16) | (static_cast<uint32_t>(d) << 24));
+}
+
+constexpr uint32_t kUnknown = 0;
+constexpr uint32_t kWgcCapturerWin = 1;
+constexpr uint32_t kScreenCapturerWinMagnifier = 2;
+constexpr uint32_t kWindowCapturerWinGdi = 3;
+constexpr uint32_t kScreenCapturerWinGdi = CreateFourCC('G', 'D', 'I', ' ');
+constexpr uint32_t kScreenCapturerWinDirectx = CreateFourCC('D', 'X', 'G', 'I');
+constexpr uint32_t kX11CapturerLinux = CreateFourCC('X', '1', '1', ' ');
+constexpr uint32_t kWaylandCapturerLinux = CreateFourCC('W', 'L', ' ', ' ');
+}  // namespace DesktopCapturerId
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURE_TYPES_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capturer.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer.cc
new file mode 100644
index 0000000000..7f601821fc
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer.cc
@@ -0,0 +1,144 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_capturer.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include <cstring>
+#include <utility>
+
+#include "modules/desktop_capture/cropping_window_capturer.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer_differ_wrapper.h"
+#include "system_wrappers/include/metrics.h"
+
+#if defined(RTC_ENABLE_WIN_WGC)
+#include "modules/desktop_capture/win/wgc_capturer_win.h"
+#include "rtc_base/win/windows_version.h"
+#endif  // defined(RTC_ENABLE_WIN_WGC)
+
+#if defined(WEBRTC_USE_PIPEWIRE)
+#include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
+#endif
+
+namespace webrtc {
+
+void LogDesktopCapturerFullscreenDetectorUsage() {
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Screenshare.DesktopCapturerFullscreenDetector",
+                        true);
+}
+
+DesktopCapturer::~DesktopCapturer() = default;
+
+DelegatedSourceListController*
+DesktopCapturer::GetDelegatedSourceListController() {
+  return nullptr;
+}
+
+void DesktopCapturer::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {}
+
+void DesktopCapturer::SetExcludedWindow(WindowId window) {}
+
+bool DesktopCapturer::GetSourceList(SourceList* sources) {
+  return true;
+}
+
+bool DesktopCapturer::SelectSource(SourceId id) {
+  return false;
+}
+
+bool DesktopCapturer::FocusOnSelectedSource() {
+  return false;
+}
+
+bool DesktopCapturer::IsOccluded(const DesktopVector& pos) {
+  return false;
+}
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateWindowCapturer(
+    const DesktopCaptureOptions& options) {
+#if defined(RTC_ENABLE_WIN_WGC)
+  if (options.allow_wgc_capturer() && IsWgcSupported(CaptureType::kWindow)) {
+    return WgcCapturerWin::CreateRawWindowCapturer(options);
+  }
+#endif  // defined(RTC_ENABLE_WIN_WGC)
+
+#if defined(WEBRTC_WIN)
+  if (options.allow_cropping_window_capturer()) {
+    return CroppingWindowCapturer::CreateCapturer(options);
+  }
+#endif  // defined(WEBRTC_WIN)
+
+  std::unique_ptr<DesktopCapturer> capturer = CreateRawWindowCapturer(options);
+  if (capturer && options.detect_updated_region()) {
+    capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
+  }
+
+  return capturer;
+}
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateGenericCapturer(
+    const DesktopCaptureOptions& options) {
+  std::unique_ptr<DesktopCapturer> capturer = CreateRawGenericCapturer(options);
+  if (capturer && options.detect_updated_region()) {
+    capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
+  }
+
+  return capturer;
+}
+
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawGenericCapturer(
+    const DesktopCaptureOptions& options) {
+#if defined(WEBRTC_USE_PIPEWIRE)
+  if (options.allow_pipewire() && DesktopCapturer::IsRunningUnderWayland()) {
+    return std::make_unique<BaseCapturerPipeWire>(options,
+                                                  CaptureType::kAnyScreenContent);
+  }
+#endif  // defined(WEBRTC_USE_PIPEWIRE)
+
+  return nullptr;
+}
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateScreenCapturer(
+    const DesktopCaptureOptions& options) {
+#if defined(RTC_ENABLE_WIN_WGC)
+  if (options.allow_wgc_capturer() && IsWgcSupported(CaptureType::kScreen)) {
+    return WgcCapturerWin::CreateRawScreenCapturer(options);
+  }
+#endif  // defined(RTC_ENABLE_WIN_WGC)
+
+  std::unique_ptr<DesktopCapturer> capturer = CreateRawScreenCapturer(options);
+  if (capturer && options.detect_updated_region()) {
+    capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
+  }
+
+  return capturer;
+}
+
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+bool DesktopCapturer::IsRunningUnderWayland() {
+  const char* xdg_session_type = getenv("XDG_SESSION_TYPE");
+  if (!xdg_session_type || strncmp(xdg_session_type, "wayland", 7) != 0)
+    return false;
+
+  if (!(getenv("WAYLAND_DISPLAY")))
+    return false;
+
+  return true;
+}
+#endif  // defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capturer.h b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer.h
new file mode 100644
index 0000000000..5c3420710f
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer.h
@@ -0,0 +1,219 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+#include <type_traits>
+#include <vector>
+
+// TODO(alcooper): Update include usage in downstream consumers and then change
+// this to a forward declaration.
+#include "modules/desktop_capture/delegated_source_list_controller.h"
+#if defined(WEBRTC_USE_GIO)
+#include "modules/desktop_capture/desktop_capture_metadata.h"
+#endif  // defined(WEBRTC_USE_GIO)
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/shared_memory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+void RTC_EXPORT LogDesktopCapturerFullscreenDetectorUsage();
+
+class DesktopCaptureOptions;
+class DesktopFrame;
+
+// Abstract interface for screen and window capturers.
+class RTC_EXPORT DesktopCapturer {
+ public:
+  enum class Result {
+    // The frame was captured successfully.
+    SUCCESS,
+
+    // There was a temporary error. The caller should continue calling
+    // CaptureFrame(), in the expectation that it will eventually recover.
+    ERROR_TEMPORARY,
+
+    // Capture has failed and will keep failing if the caller tries calling
+    // CaptureFrame() again.
+    ERROR_PERMANENT,
+
+    MAX_VALUE = ERROR_PERMANENT
+  };
+
+  // Interface that must be implemented by the DesktopCapturer consumers.
+  class Callback {
+   public:
+    // Called after a frame has been captured. `frame` is not nullptr if and
+    // only if `result` is SUCCESS.
+    virtual void OnCaptureResult(Result result,
+                                 std::unique_ptr<DesktopFrame> frame) = 0;
+
+   protected:
+    virtual ~Callback() {}
+  };
+
+#if defined(CHROMEOS)
+  typedef int64_t SourceId;
+#else
+  typedef intptr_t SourceId;
+#endif
+
+  static_assert(std::is_same<SourceId, ScreenId>::value,
+                "SourceId should be a same type as ScreenId.");
+
+  struct Source {
+    // The unique id to represent a Source of current DesktopCapturer.
+    SourceId id;
+    pid_t pid;
+
+    // Title of the window or screen in UTF-8 encoding, maybe empty. This field
+    // should not be used to identify a source.
+    std::string title;
+
+#if defined(CHROMEOS)
+    // TODO(https://crbug.com/1369162): Remove or refactor this value.
+    WindowId in_process_id = kNullWindowId;
+#endif
+
+    // The display's unique ID. If no ID is defined, it will hold the value
+    // kInvalidDisplayId.
+    int64_t display_id = kInvalidDisplayId;
+  };
+
+  typedef std::vector<Source> SourceList;
+
+  virtual ~DesktopCapturer();
+
+  // Called at the beginning of a capturing session. `callback` must remain
+  // valid until capturer is destroyed.
+  virtual void Start(Callback* callback) = 0;
+
+  // Returns a valid pointer if the capturer requires the user to make a
+  // selection from a source list provided by the capturer.
+  // Returns nullptr if the capturer does not provide a UI for the user to make
+  // a selection.
+  //
+  // Callers should not take ownership of the returned pointer, but it is
+  // guaranteed to be valid as long as the desktop_capturer is valid.
+  // Note that consumers should still use GetSourceList and SelectSource, but
+  // their behavior may be modified if this returns a value. See those methods
+  // for a more in-depth discussion of those potential modifications.
+  virtual DelegatedSourceListController* GetDelegatedSourceListController();
+
+  // Sets SharedMemoryFactory that will be used to create buffers for the
+  // captured frames. The factory can be invoked on a thread other than the one
+  // where CaptureFrame() is called. It will be destroyed on the same thread.
+  // Shared memory is currently supported only by some DesktopCapturer
+  // implementations.
+  virtual void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory);
+
+  // Captures next frame, and involve callback provided by Start() function.
+  // Pending capture requests are canceled when DesktopCapturer is deleted.
+  virtual void CaptureFrame() = 0;
+
+  // Sets the window to be excluded from the captured image in the future
+  // Capture calls. Used to exclude the screenshare notification window for
+  // screen capturing.
+  virtual void SetExcludedWindow(WindowId window);
+
+  // TODO(zijiehe): Following functions should be pure virtual. The default
+  // implementations are for backward compatibility only. Remove default
+  // implementations once all DesktopCapturer implementations in Chromium have
+  // implemented these functions.
+
+  // Gets a list of sources current capturer supports. Returns false in case of
+  // a failure.
+  // For DesktopCapturer implementations to capture screens, this function
+  // should return monitors.
+  // For DesktopCapturer implementations to capture windows, this function
+  // should only return root windows owned by applications.
+  //
+  // Note that capturers who use a delegated source list will return a
+  // SourceList with exactly one value, but it may not be viable for capture
+  // (e.g. CaptureFrame will return ERROR_TEMPORARY) until a selection has been
+  // made.
+  virtual bool GetSourceList(SourceList* sources);
+
+  // Selects a source to be captured. Returns false in case of a failure (e.g.
+  // if there is no source with the specified type and id.)
+  //
+  // Note that some capturers with delegated source lists may also support
+  // selecting a SourceID that is not in the returned source list as a form of
+  // restore token.
+  virtual bool SelectSource(SourceId id);
+
+  // Brings the selected source to the front and sets the input focus on it.
+  // Returns false in case of a failure or no source has been selected or the
+  // implementation does not support this functionality.
+  virtual bool FocusOnSelectedSource();
+
+  // Returns true if the `pos` on the selected source is covered by other
+  // elements on the display, and is not visible to the users.
+  // `pos` is in full desktop coordinates, i.e. the top-left monitor always
+  // starts from (0, 0).
+  // The return value if `pos` is out of the scope of the source is undefined.
+  virtual bool IsOccluded(const DesktopVector& pos);
+
+  // Creates a DesktopCapturer instance which targets to capture windows and screens.
+  static std::unique_ptr<DesktopCapturer> CreateGenericCapturer(
+      const DesktopCaptureOptions& options);
+
+  // Creates a DesktopCapturer instance which targets to capture windows.
+  static std::unique_ptr<DesktopCapturer> CreateWindowCapturer(
+      const DesktopCaptureOptions& options);
+
+  // Creates a DesktopCapturer instance which targets to capture screens.
+  static std::unique_ptr<DesktopCapturer> CreateScreenCapturer(
+      const DesktopCaptureOptions& options);
+
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+  static bool IsRunningUnderWayland();
+
+  virtual void UpdateResolution(uint32_t width, uint32_t height) {}
+#endif  // defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+
+#if defined(WEBRTC_USE_GIO)
+  // Populates implementation specific metadata into the passed in pointer.
+  // Classes can choose to override it or use the default no-op implementation.
+  virtual DesktopCaptureMetadata GetMetadata() { return {}; }
+#endif  // defined(WEBRTC_USE_GIO)
+
+ protected:
+  // CroppingWindowCapturer needs to create raw capturers without wrappers, so
+  // the following two functions are protected.
+
+  // Creates a platform specific DesktopCapturer instance which targets to
+  // capture windows and screens.
+  static std::unique_ptr<DesktopCapturer> CreateRawGenericCapturer(
+      const DesktopCaptureOptions& options);
+
+  // Creates a platform specific DesktopCapturer instance which targets to
+  // capture windows.
+  static std::unique_ptr<DesktopCapturer> CreateRawWindowCapturer(
+      const DesktopCaptureOptions& options);
+
+  // Creates a platform specific DesktopCapturer instance which targets to
+  // capture screens.
+  static std::unique_ptr<DesktopCapturer> CreateRawScreenCapturer(
+      const DesktopCaptureOptions& options);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper.cc
new file mode 100644
index 0000000000..77543e4060
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper.cc
@@ -0,0 +1,232 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_capturer_differ_wrapper.h"
+
+#include <stdint.h>
+#include <string.h>
+
+#include <utility>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/differ_block.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+// Returns true if (0, 0) - (`width`, `height`) vector in `old_buffer` and
+// `new_buffer` are equal. `width` should be less than 32
+// (defined by kBlockSize), otherwise BlockDifference() should be used.
+bool PartialBlockDifference(const uint8_t* old_buffer,
+                            const uint8_t* new_buffer,
+                            int width,
+                            int height,
+                            int stride) {
+  RTC_DCHECK_LT(width, kBlockSize);
+  const int width_bytes = width * DesktopFrame::kBytesPerPixel;
+  for (int i = 0; i < height; i++) {
+    if (memcmp(old_buffer, new_buffer, width_bytes) != 0) {
+      return true;
+    }
+    old_buffer += stride;
+    new_buffer += stride;
+  }
+  return false;
+}
+
+// Compares columns in the range of [`left`, `right`), in a row in the
+// range of [`top`, `top` + `height`), starts from `old_buffer` and
+// `new_buffer`, and outputs updated regions into `output`. `stride` is the
+// DesktopFrame::stride().
+void CompareRow(const uint8_t* old_buffer,
+                const uint8_t* new_buffer,
+                const int left,
+                const int right,
+                const int top,
+                const int bottom,
+                const int stride,
+                DesktopRegion* const output) {
+  const int block_x_offset = kBlockSize * DesktopFrame::kBytesPerPixel;
+  const int width = right - left;
+  const int height = bottom - top;
+  const int block_count = (width - 1) / kBlockSize;
+  const int last_block_width = width - block_count * kBlockSize;
+  RTC_DCHECK_GT(last_block_width, 0);
+  RTC_DCHECK_LE(last_block_width, kBlockSize);
+
+  // The first block-column in a continuous dirty area in current block-row.
+  int first_dirty_x_block = -1;
+
+  // We always need to add dirty area into `output` in the last block, so handle
+  // it separatedly.
+  for (int x = 0; x < block_count; x++) {
+    if (BlockDifference(old_buffer, new_buffer, height, stride)) {
+      if (first_dirty_x_block == -1) {
+        // This is the first dirty block in a continuous dirty area.
+        first_dirty_x_block = x;
+      }
+    } else if (first_dirty_x_block != -1) {
+      // The block on the left is the last dirty block in a continuous
+      // dirty area.
+      output->AddRect(
+          DesktopRect::MakeLTRB(first_dirty_x_block * kBlockSize + left, top,
+                                x * kBlockSize + left, bottom));
+      first_dirty_x_block = -1;
+    }
+    old_buffer += block_x_offset;
+    new_buffer += block_x_offset;
+  }
+
+  bool last_block_diff;
+  if (last_block_width < kBlockSize) {
+    // The last one is a partial vector.
+    last_block_diff = PartialBlockDifference(old_buffer, new_buffer,
+                                             last_block_width, height, stride);
+  } else {
+    last_block_diff = BlockDifference(old_buffer, new_buffer, height, stride);
+  }
+  if (last_block_diff) {
+    if (first_dirty_x_block == -1) {
+      first_dirty_x_block = block_count;
+    }
+    output->AddRect(DesktopRect::MakeLTRB(
+        first_dirty_x_block * kBlockSize + left, top, right, bottom));
+  } else if (first_dirty_x_block != -1) {
+    output->AddRect(
+        DesktopRect::MakeLTRB(first_dirty_x_block * kBlockSize + left, top,
+                              block_count * kBlockSize + left, bottom));
+  }
+}
+
+// Compares `rect` area in `old_frame` and `new_frame`, and outputs dirty
+// regions into `output`.
+void CompareFrames(const DesktopFrame& old_frame,
+                   const DesktopFrame& new_frame,
+                   DesktopRect rect,
+                   DesktopRegion* const output) {
+  RTC_DCHECK(old_frame.size().equals(new_frame.size()));
+  RTC_DCHECK_EQ(old_frame.stride(), new_frame.stride());
+  rect.IntersectWith(DesktopRect::MakeSize(old_frame.size()));
+
+  const int y_block_count = (rect.height() - 1) / kBlockSize;
+  const int last_y_block_height = rect.height() - y_block_count * kBlockSize;
+  // Offset from the start of one block-row to the next.
+  const int block_y_stride = old_frame.stride() * kBlockSize;
+  const uint8_t* prev_block_row_start =
+      old_frame.GetFrameDataAtPos(rect.top_left());
+  const uint8_t* curr_block_row_start =
+      new_frame.GetFrameDataAtPos(rect.top_left());
+
+  int top = rect.top();
+  // The last row may have a different height, so we handle it separately.
+  for (int y = 0; y < y_block_count; y++) {
+    CompareRow(prev_block_row_start, curr_block_row_start, rect.left(),
+               rect.right(), top, top + kBlockSize, old_frame.stride(), output);
+    top += kBlockSize;
+    prev_block_row_start += block_y_stride;
+    curr_block_row_start += block_y_stride;
+  }
+  CompareRow(prev_block_row_start, curr_block_row_start, rect.left(),
+             rect.right(), top, top + last_y_block_height, old_frame.stride(),
+             output);
+}
+
+}  // namespace
+
+DesktopCapturerDifferWrapper::DesktopCapturerDifferWrapper(
+    std::unique_ptr<DesktopCapturer> base_capturer)
+    : base_capturer_(std::move(base_capturer)) {
+  RTC_DCHECK(base_capturer_);
+}
+
+DesktopCapturerDifferWrapper::~DesktopCapturerDifferWrapper() {}
+
+void DesktopCapturerDifferWrapper::Start(DesktopCapturer::Callback* callback) {
+  callback_ = callback;
+  base_capturer_->Start(this);
+}
+
+void DesktopCapturerDifferWrapper::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  base_capturer_->SetSharedMemoryFactory(std::move(shared_memory_factory));
+}
+
+void DesktopCapturerDifferWrapper::CaptureFrame() {
+  base_capturer_->CaptureFrame();
+}
+
+void DesktopCapturerDifferWrapper::SetExcludedWindow(WindowId window) {
+  base_capturer_->SetExcludedWindow(window);
+}
+
+bool DesktopCapturerDifferWrapper::GetSourceList(SourceList* sources) {
+  return base_capturer_->GetSourceList(sources);
+}
+
+bool DesktopCapturerDifferWrapper::SelectSource(SourceId id) {
+  return base_capturer_->SelectSource(id);
+}
+
+bool DesktopCapturerDifferWrapper::FocusOnSelectedSource() {
+  return base_capturer_->FocusOnSelectedSource();
+}
+
+bool DesktopCapturerDifferWrapper::IsOccluded(const DesktopVector& pos) {
+  return base_capturer_->IsOccluded(pos);
+}
+
+#if defined(WEBRTC_USE_GIO)
+DesktopCaptureMetadata DesktopCapturerDifferWrapper::GetMetadata() {
+  return base_capturer_->GetMetadata();
+}
+#endif  // defined(WEBRTC_USE_GIO)
+
+void DesktopCapturerDifferWrapper::OnCaptureResult(
+    Result result,
+    std::unique_ptr<DesktopFrame> input_frame) {
+  int64_t start_time_nanos = rtc::TimeNanos();
+  if (!input_frame) {
+    callback_->OnCaptureResult(result, nullptr);
+    return;
+  }
+  RTC_DCHECK(result == Result::SUCCESS);
+
+  std::unique_ptr<SharedDesktopFrame> frame =
+      SharedDesktopFrame::Wrap(std::move(input_frame));
+  if (last_frame_ && (last_frame_->size().width() != frame->size().width() ||
+                      last_frame_->size().height() != frame->size().height() ||
+                      last_frame_->stride() != frame->stride())) {
+    last_frame_.reset();
+  }
+
+  if (last_frame_) {
+    DesktopRegion hints;
+    hints.Swap(frame->mutable_updated_region());
+    for (DesktopRegion::Iterator it(hints); !it.IsAtEnd(); it.Advance()) {
+      CompareFrames(*last_frame_, *frame, it.rect(),
+                    frame->mutable_updated_region());
+    }
+  } else {
+    frame->mutable_updated_region()->SetRect(
+        DesktopRect::MakeSize(frame->size()));
+  }
+  last_frame_ = frame->Share();
+
+  frame->set_capture_time_ms(frame->capture_time_ms() +
+                             (rtc::TimeNanos() - start_time_nanos) /
+                                 rtc::kNumNanosecsPerMillisec);
+  callback_->OnCaptureResult(result, std::move(frame));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper.h b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper.h
new file mode 100644
index 0000000000..6ebb5d7bc3
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_DIFFER_WRAPPER_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_DIFFER_WRAPPER_H_
+
+#include <memory>
+#if defined(WEBRTC_USE_GIO)
+#include "modules/desktop_capture/desktop_capture_metadata.h"
+#endif  // defined(WEBRTC_USE_GIO)
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "modules/desktop_capture/shared_memory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// DesktopCapturer wrapper that calculates updated_region() by comparing frames
+// content. This class always expects the underlying DesktopCapturer
+// implementation returns a superset of updated regions in DestkopFrame. If a
+// DesktopCapturer implementation does not know the updated region, it should
+// set updated_region() to full frame.
+//
+// This class marks entire frame as updated if the frame size or frame stride
+// has been changed.
+class RTC_EXPORT DesktopCapturerDifferWrapper
+    : public DesktopCapturer,
+      public DesktopCapturer::Callback {
+ public:
+  // Creates a DesktopCapturerDifferWrapper with a DesktopCapturer
+  // implementation, and takes its ownership.
+  explicit DesktopCapturerDifferWrapper(
+      std::unique_ptr<DesktopCapturer> base_capturer);
+
+  ~DesktopCapturerDifferWrapper() override;
+
+  // DesktopCapturer interface.
+  void Start(DesktopCapturer::Callback* callback) override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  void CaptureFrame() override;
+  void SetExcludedWindow(WindowId window) override;
+  bool GetSourceList(SourceList* screens) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  bool IsOccluded(const DesktopVector& pos) override;
+#if defined(WEBRTC_USE_GIO)
+  DesktopCaptureMetadata GetMetadata() override;
+#endif  // defined(WEBRTC_USE_GIO)
+ private:
+  // DesktopCapturer::Callback interface.
+  void OnCaptureResult(Result result,
+                       std::unique_ptr<DesktopFrame> frame) override;
+
+  const std::unique_ptr<DesktopCapturer> base_capturer_;
+  DesktopCapturer::Callback* callback_;
+  std::unique_ptr<SharedDesktopFrame> last_frame_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_DIFFER_WRAPPER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper_unittest.cc
new file mode 100644
index 0000000000..9ccef3cc10
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_differ_wrapper_unittest.cc
@@ -0,0 +1,291 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_capturer_differ_wrapper.h"
+
+#include <initializer_list>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/differ_block.h"
+#include "modules/desktop_capture/fake_desktop_capturer.h"
+#include "modules/desktop_capture/mock_desktop_capturer_callback.h"
+#include "rtc_base/random.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// Compares and asserts `frame`.updated_region() equals to `rects`. This
+// function does not care about the order of the `rects` and it does not expect
+// DesktopRegion to return an exact area for each rectangle in `rects`.
+template <template <typename, typename...> class T = std::initializer_list,
+          typename... Rect>
+void AssertUpdatedRegionIs(const DesktopFrame& frame,
+                           const T<DesktopRect, Rect...>& rects) {
+  DesktopRegion region;
+  for (const auto& rect : rects) {
+    region.AddRect(rect);
+  }
+  ASSERT_TRUE(frame.updated_region().Equals(region));
+}
+
+// Compares and asserts `frame`.updated_region() covers all rectangles in
+// `rects`, but does not cover areas other than a kBlockSize expansion. This
+// function does not care about the order of the `rects`, and it does not expect
+// DesktopRegion to return an exact area of each rectangle in `rects`.
+template <template <typename, typename...> class T = std::initializer_list,
+          typename... Rect>
+void AssertUpdatedRegionCovers(const DesktopFrame& frame,
+                               const T<DesktopRect, Rect...>& rects) {
+  DesktopRegion region;
+  for (const auto& rect : rects) {
+    region.AddRect(rect);
+  }
+
+  // Intersect of `rects` and `frame`.updated_region() should be `rects`. i.e.
+  // `frame`.updated_region() should be a superset of `rects`.
+  DesktopRegion intersect(region);
+  intersect.IntersectWith(frame.updated_region());
+  ASSERT_TRUE(region.Equals(intersect));
+
+  // Difference between `rects` and `frame`.updated_region() should not cover
+  // areas which have larger than twice of kBlockSize width and height.
+  //
+  // Explanation of the 'twice' of kBlockSize (indeed kBlockSize * 2 - 2) is
+  // following,
+  // (Each block in the following grid is a 8 x 8 pixels area. X means the real
+  // updated area, m means the updated area marked by
+  // DesktopCapturerDifferWrapper.)
+  // +---+---+---+---+---+---+---+---+
+  // | X | m | m | m | m | m | m | m |
+  // +---+---+---+---+---+---+---+---+
+  // | m | m | m | m | m | m | m | m |
+  // +---+---+---+---+---+---+---+---+
+  // | m | m | m | m | m | m | m | m |
+  // +---+---+---+---+---+---+---+---+
+  // | m | m | m | m | m | m | m | X |
+  // +---+---+---+---+---+---+---+---+
+  // The top left [0, 0] - [8, 8] and right bottom [56, 24] - [64, 32] blocks of
+  // this area are updated. But since DesktopCapturerDifferWrapper compares
+  // 32 x 32 blocks by default, this entire area is marked as updated. So the
+  // [8, 8] - [56, 32] is expected to be covered in the difference.
+  //
+  // But if [0, 0] - [8, 8] and [64, 24] - [72, 32] blocks are updated,
+  // +---+---+---+---+---+---+---+---+---+---+---+---+
+  // | X | m | m | m |   |   |   |   | m | m | m | m |
+  // +---+---+---+---+---+---+---+---+---+---+---+---+
+  // | m | m | m | m |   |   |   |   | m | m | m | m |
+  // +---+---+---+---+---+---+---+---+---+---+---+---+
+  // | m | m | m | m |   |   |   |   | m | m | m | m |
+  // +---+---+---+---+---+---+---+---+---+---+---+---+
+  // | m | m | m | m |   |   |   |   | X | m | m | m |
+  // +---+---+---+---+---+---+---+---+---+---+---+---+
+  // the [8, 8] - [64, 32] is not expected to be covered in the difference. As
+  // DesktopCapturerDifferWrapper should only mark [0, 0] - [32, 32] and
+  // [64, 0] - [96, 32] as updated.
+  DesktopRegion differ(frame.updated_region());
+  differ.Subtract(region);
+  for (DesktopRegion::Iterator it(differ); !it.IsAtEnd(); it.Advance()) {
+    ASSERT_TRUE(it.rect().width() <= kBlockSize * 2 - 2 ||
+                it.rect().height() <= kBlockSize * 2 - 2);
+  }
+}
+
+// Executes a DesktopCapturerDifferWrapper::Capture() and compares its output
+// DesktopFrame::updated_region() with `updated_region` if `check_result` is
+// true. If `exactly_match` is true, AssertUpdatedRegionIs() will be used,
+// otherwise AssertUpdatedRegionCovers() will be used.
+template <template <typename, typename...> class T = std::initializer_list,
+          typename... Rect>
+void ExecuteDifferWrapperCase(BlackWhiteDesktopFramePainter* frame_painter,
+                              DesktopCapturerDifferWrapper* capturer,
+                              MockDesktopCapturerCallback* callback,
+                              const T<DesktopRect, Rect...>& updated_region,
+                              bool check_result,
+                              bool exactly_match) {
+  EXPECT_CALL(*callback, OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS,
+                                            ::testing::_))
+      .Times(1)
+      .WillOnce(
+          ::testing::Invoke([&updated_region, check_result, exactly_match](
+                                DesktopCapturer::Result result,
+                                std::unique_ptr<DesktopFrame>* frame) {
+            ASSERT_EQ(result, DesktopCapturer::Result::SUCCESS);
+            if (check_result) {
+              if (exactly_match) {
+                AssertUpdatedRegionIs(**frame, updated_region);
+              } else {
+                AssertUpdatedRegionCovers(**frame, updated_region);
+              }
+            }
+          }));
+  for (const auto& rect : updated_region) {
+    frame_painter->updated_region()->AddRect(rect);
+  }
+  capturer->CaptureFrame();
+}
+
+// Executes a DesktopCapturerDifferWrapper::Capture(), if updated_region() is
+// not set, this function will reset DesktopCapturerDifferWrapper internal
+// DesktopFrame into black.
+void ExecuteCapturer(DesktopCapturerDifferWrapper* capturer,
+                     MockDesktopCapturerCallback* callback) {
+  EXPECT_CALL(*callback, OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS,
+                                            ::testing::_))
+      .Times(1);
+  capturer->CaptureFrame();
+}
+
+void ExecuteDifferWrapperTest(bool with_hints,
+                              bool enlarge_updated_region,
+                              bool random_updated_region,
+                              bool check_result) {
+  const bool updated_region_should_exactly_match =
+      with_hints && !enlarge_updated_region && !random_updated_region;
+  BlackWhiteDesktopFramePainter frame_painter;
+  PainterDesktopFrameGenerator frame_generator;
+  frame_generator.set_desktop_frame_painter(&frame_painter);
+  std::unique_ptr<FakeDesktopCapturer> fake(new FakeDesktopCapturer());
+  fake->set_frame_generator(&frame_generator);
+  DesktopCapturerDifferWrapper capturer(std::move(fake));
+  MockDesktopCapturerCallback callback;
+  frame_generator.set_provide_updated_region_hints(with_hints);
+  frame_generator.set_enlarge_updated_region(enlarge_updated_region);
+  frame_generator.set_add_random_updated_region(random_updated_region);
+
+  capturer.Start(&callback);
+
+  EXPECT_CALL(callback, OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS,
+                                           ::testing::_))
+      .Times(1)
+      .WillOnce(::testing::Invoke([](DesktopCapturer::Result result,
+                                     std::unique_ptr<DesktopFrame>* frame) {
+        ASSERT_EQ(result, DesktopCapturer::Result::SUCCESS);
+        AssertUpdatedRegionIs(**frame,
+                              {DesktopRect::MakeSize((*frame)->size())});
+      }));
+  capturer.CaptureFrame();
+
+  ExecuteDifferWrapperCase(&frame_painter, &capturer, &callback,
+                           {DesktopRect::MakeLTRB(100, 100, 200, 200),
+                            DesktopRect::MakeLTRB(300, 300, 400, 400)},
+                           check_result, updated_region_should_exactly_match);
+  ExecuteCapturer(&capturer, &callback);
+
+  ExecuteDifferWrapperCase(
+      &frame_painter, &capturer, &callback,
+      {DesktopRect::MakeLTRB(0, 0, 40, 40),
+       DesktopRect::MakeLTRB(0, frame_generator.size()->height() - 40, 40,
+                             frame_generator.size()->height()),
+       DesktopRect::MakeLTRB(frame_generator.size()->width() - 40, 0,
+                             frame_generator.size()->width(), 40),
+       DesktopRect::MakeLTRB(frame_generator.size()->width() - 40,
+                             frame_generator.size()->height() - 40,
+                             frame_generator.size()->width(),
+                             frame_generator.size()->height())},
+      check_result, updated_region_should_exactly_match);
+
+  Random random(rtc::TimeMillis());
+  // Fuzzing tests.
+  for (int i = 0; i < 1000; i++) {
+    if (enlarge_updated_region) {
+      frame_generator.set_enlarge_range(random.Rand(1, 50));
+    }
+    frame_generator.size()->set(random.Rand(500, 2000), random.Rand(500, 2000));
+    ExecuteCapturer(&capturer, &callback);
+    std::vector<DesktopRect> updated_region;
+    for (int j = random.Rand(50); j >= 0; j--) {
+      // At least a 1 x 1 updated region.
+      const int left = random.Rand(0, frame_generator.size()->width() - 2);
+      const int top = random.Rand(0, frame_generator.size()->height() - 2);
+      const int right = random.Rand(left + 1, frame_generator.size()->width());
+      const int bottom = random.Rand(top + 1, frame_generator.size()->height());
+      updated_region.push_back(DesktopRect::MakeLTRB(left, top, right, bottom));
+    }
+    ExecuteDifferWrapperCase(&frame_painter, &capturer, &callback,
+                             updated_region, check_result,
+                             updated_region_should_exactly_match);
+  }
+}
+
+}  // namespace
+
+TEST(DesktopCapturerDifferWrapperTest, CaptureWithoutHints) {
+  ExecuteDifferWrapperTest(false, false, false, true);
+}
+
+TEST(DesktopCapturerDifferWrapperTest, CaptureWithHints) {
+  ExecuteDifferWrapperTest(true, false, false, true);
+}
+
+TEST(DesktopCapturerDifferWrapperTest, CaptureWithEnlargedHints) {
+  ExecuteDifferWrapperTest(true, true, false, true);
+}
+
+TEST(DesktopCapturerDifferWrapperTest, CaptureWithRandomHints) {
+  ExecuteDifferWrapperTest(true, false, true, true);
+}
+
+TEST(DesktopCapturerDifferWrapperTest, CaptureWithEnlargedAndRandomHints) {
+  ExecuteDifferWrapperTest(true, true, true, true);
+}
+
+// When hints are provided, DesktopCapturerDifferWrapper has a slightly better
+// performance in current configuration, but not so significant. Following is
+// one run result.
+// [ RUN      ] DISABLED_CaptureWithoutHintsPerf
+// [       OK ] DISABLED_CaptureWithoutHintsPerf (7118 ms)
+// [ RUN      ] DISABLED_CaptureWithHintsPerf
+// [       OK ] DISABLED_CaptureWithHintsPerf (5580 ms)
+// [ RUN      ] DISABLED_CaptureWithEnlargedHintsPerf
+// [       OK ] DISABLED_CaptureWithEnlargedHintsPerf (5974 ms)
+// [ RUN      ] DISABLED_CaptureWithRandomHintsPerf
+// [       OK ] DISABLED_CaptureWithRandomHintsPerf (6184 ms)
+// [ RUN      ] DISABLED_CaptureWithEnlargedAndRandomHintsPerf
+// [       OK ] DISABLED_CaptureWithEnlargedAndRandomHintsPerf (6347 ms)
+TEST(DesktopCapturerDifferWrapperTest, DISABLED_CaptureWithoutHintsPerf) {
+  int64_t started = rtc::TimeMillis();
+  ExecuteDifferWrapperTest(false, false, false, false);
+  ASSERT_LE(rtc::TimeMillis() - started, 15000);
+}
+
+TEST(DesktopCapturerDifferWrapperTest, DISABLED_CaptureWithHintsPerf) {
+  int64_t started = rtc::TimeMillis();
+  ExecuteDifferWrapperTest(true, false, false, false);
+  ASSERT_LE(rtc::TimeMillis() - started, 15000);
+}
+
+TEST(DesktopCapturerDifferWrapperTest, DISABLED_CaptureWithEnlargedHintsPerf) {
+  int64_t started = rtc::TimeMillis();
+  ExecuteDifferWrapperTest(true, true, false, false);
+  ASSERT_LE(rtc::TimeMillis() - started, 15000);
+}
+
+TEST(DesktopCapturerDifferWrapperTest, DISABLED_CaptureWithRandomHintsPerf) {
+  int64_t started = rtc::TimeMillis();
+  ExecuteDifferWrapperTest(true, false, true, false);
+  ASSERT_LE(rtc::TimeMillis() - started, 15000);
+}
+
+TEST(DesktopCapturerDifferWrapperTest,
+     DISABLED_CaptureWithEnlargedAndRandomHintsPerf) {
+  int64_t started = rtc::TimeMillis();
+  ExecuteDifferWrapperTest(true, true, true, false);
+  ASSERT_LE(rtc::TimeMillis() - started, 15000);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_wrapper.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_wrapper.cc
new file mode 100644
index 0000000000..4bbdd6c94f
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_wrapper.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_capturer_wrapper.h"
+
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+DesktopCapturerWrapper::DesktopCapturerWrapper(
+    std::unique_ptr<DesktopCapturer> base_capturer)
+    : base_capturer_(std::move(base_capturer)) {
+  RTC_DCHECK(base_capturer_);
+}
+
+DesktopCapturerWrapper::~DesktopCapturerWrapper() = default;
+
+void DesktopCapturerWrapper::Start(Callback* callback) {
+  base_capturer_->Start(callback);
+}
+
+void DesktopCapturerWrapper::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  base_capturer_->SetSharedMemoryFactory(std::move(shared_memory_factory));
+}
+
+void DesktopCapturerWrapper::CaptureFrame() {
+  base_capturer_->CaptureFrame();
+}
+
+void DesktopCapturerWrapper::SetExcludedWindow(WindowId window) {
+  base_capturer_->SetExcludedWindow(window);
+}
+
+bool DesktopCapturerWrapper::GetSourceList(SourceList* sources) {
+  return base_capturer_->GetSourceList(sources);
+}
+
+bool DesktopCapturerWrapper::SelectSource(SourceId id) {
+  return base_capturer_->SelectSource(id);
+}
+
+bool DesktopCapturerWrapper::FocusOnSelectedSource() {
+  return base_capturer_->FocusOnSelectedSource();
+}
+
+bool DesktopCapturerWrapper::IsOccluded(const DesktopVector& pos) {
+  return base_capturer_->IsOccluded(pos);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_wrapper.h b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_wrapper.h
new file mode 100644
index 0000000000..e0f50d79e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_capturer_wrapper.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_WRAPPER_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_WRAPPER_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/shared_memory.h"
+
+namespace webrtc {
+
+// Wraps a DesktopCapturer and forwards all the function calls to it.
+class DesktopCapturerWrapper : public DesktopCapturer {
+ public:
+  explicit DesktopCapturerWrapper(
+      std::unique_ptr<DesktopCapturer> base_capturer);
+  ~DesktopCapturerWrapper() override;
+
+  // DesktopCapturer implementations.
+  void Start(Callback* callback) override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  void CaptureFrame() override;
+  void SetExcludedWindow(WindowId window) override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  bool IsOccluded(const DesktopVector& pos) override;
+
+ protected:
+  // Guaranteed to be valid.
+  const std::unique_ptr<DesktopCapturer> base_capturer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_WRAPPER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_frame.cc
new file mode 100644
index 0000000000..d6f574588c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame.cc
@@ -0,0 +1,205 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_frame.h"
+
+#include <string.h>
+
+#include <cmath>
+#include <memory>
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "rtc_base/checks.h"
+#include "third_party/libyuv/include/libyuv/planar_functions.h"
+
+namespace webrtc {
+
+DesktopFrame::DesktopFrame(DesktopSize size,
+                           int stride,
+                           uint8_t* data,
+                           SharedMemory* shared_memory)
+    : data_(data),
+      shared_memory_(shared_memory),
+      size_(size),
+      stride_(stride),
+      capture_time_ms_(0),
+      capturer_id_(DesktopCapturerId::kUnknown) {
+  RTC_DCHECK(size_.width() >= 0);
+  RTC_DCHECK(size_.height() >= 0);
+}
+
+DesktopFrame::~DesktopFrame() = default;
+
+void DesktopFrame::CopyPixelsFrom(const uint8_t* src_buffer,
+                                  int src_stride,
+                                  const DesktopRect& dest_rect) {
+  RTC_CHECK(DesktopRect::MakeSize(size()).ContainsRect(dest_rect));
+
+  uint8_t* dest = GetFrameDataAtPos(dest_rect.top_left());
+  libyuv::CopyPlane(src_buffer, src_stride, dest, stride(),
+                    DesktopFrame::kBytesPerPixel * dest_rect.width(),
+                    dest_rect.height());
+}
+
+void DesktopFrame::CopyPixelsFrom(const DesktopFrame& src_frame,
+                                  const DesktopVector& src_pos,
+                                  const DesktopRect& dest_rect) {
+  RTC_CHECK(DesktopRect::MakeSize(src_frame.size())
+                .ContainsRect(
+                    DesktopRect::MakeOriginSize(src_pos, dest_rect.size())));
+
+  CopyPixelsFrom(src_frame.GetFrameDataAtPos(src_pos), src_frame.stride(),
+                 dest_rect);
+}
+
+bool DesktopFrame::CopyIntersectingPixelsFrom(const DesktopFrame& src_frame,
+                                              double horizontal_scale,
+                                              double vertical_scale) {
+  const DesktopVector& origin = top_left();
+  const DesktopVector& src_frame_origin = src_frame.top_left();
+
+  DesktopVector src_frame_offset = src_frame_origin.subtract(origin);
+
+  // Determine the intersection, first adjusting its origin to account for any
+  // DPI scaling.
+  DesktopRect intersection_rect = src_frame.rect();
+  if (horizontal_scale != 1.0 || vertical_scale != 1.0) {
+    DesktopVector origin_adjustment(
+        static_cast<int>(
+            std::round((horizontal_scale - 1.0) * src_frame_offset.x())),
+        static_cast<int>(
+            std::round((vertical_scale - 1.0) * src_frame_offset.y())));
+
+    intersection_rect.Translate(origin_adjustment);
+
+    src_frame_offset = src_frame_offset.add(origin_adjustment);
+  }
+
+  intersection_rect.IntersectWith(rect());
+  if (intersection_rect.is_empty()) {
+    return false;
+  }
+
+  // Translate the intersection rect to be relative to the outer rect.
+  intersection_rect.Translate(-origin.x(), -origin.y());
+
+  // Determine source position for the copy (offsets of outer frame from
+  // source origin, if positive).
+  int32_t src_pos_x = std::max(0, -src_frame_offset.x());
+  int32_t src_pos_y = std::max(0, -src_frame_offset.y());
+
+  CopyPixelsFrom(src_frame, DesktopVector(src_pos_x, src_pos_y),
+                 intersection_rect);
+  return true;
+}
+
+DesktopRect DesktopFrame::rect() const {
+  const float scale = scale_factor();
+  // Only scale the size.
+  return DesktopRect::MakeXYWH(top_left().x(), top_left().y(),
+                               size().width() / scale, size().height() / scale);
+}
+
+float DesktopFrame::scale_factor() const {
+  float scale = 1.0f;
+
+#if defined(WEBRTC_MAC) || defined(CHROMEOS)
+  // At least on Windows the logical and physical pixel are the same
+  // See http://crbug.com/948362.
+  if (!dpi().is_zero() && dpi().x() == dpi().y())
+    scale = dpi().x() / kStandardDPI;
+#endif
+
+  return scale;
+}
+
+uint8_t* DesktopFrame::GetFrameDataAtPos(const DesktopVector& pos) const {
+  return data() + stride() * pos.y() + DesktopFrame::kBytesPerPixel * pos.x();
+}
+
+void DesktopFrame::CopyFrameInfoFrom(const DesktopFrame& other) {
+  set_dpi(other.dpi());
+  set_capture_time_ms(other.capture_time_ms());
+  set_capturer_id(other.capturer_id());
+  *mutable_updated_region() = other.updated_region();
+  set_top_left(other.top_left());
+  set_icc_profile(other.icc_profile());
+}
+
+void DesktopFrame::MoveFrameInfoFrom(DesktopFrame* other) {
+  set_dpi(other->dpi());
+  set_capture_time_ms(other->capture_time_ms());
+  set_capturer_id(other->capturer_id());
+  mutable_updated_region()->Swap(other->mutable_updated_region());
+  set_top_left(other->top_left());
+  set_icc_profile(other->icc_profile());
+}
+
+BasicDesktopFrame::BasicDesktopFrame(DesktopSize size)
+    : DesktopFrame(size,
+                   kBytesPerPixel * size.width(),
+                   new uint8_t[kBytesPerPixel * size.width() * size.height()](),
+                   nullptr) {}
+
+BasicDesktopFrame::~BasicDesktopFrame() {
+  delete[] data_;
+}
+
+// static
+DesktopFrame* BasicDesktopFrame::CopyOf(const DesktopFrame& frame) {
+  DesktopFrame* result = new BasicDesktopFrame(frame.size());
+  // TODO(crbug.com/1330019): Temporary workaround for a known libyuv crash when
+  // the height or width is 0. Remove this once this change has been merged.
+  if (frame.size().width() && frame.size().height()) {
+    libyuv::CopyPlane(frame.data(), frame.stride(), result->data(),
+                      result->stride(), frame.size().width() * kBytesPerPixel,
+                      frame.size().height());
+  }
+  result->CopyFrameInfoFrom(frame);
+  return result;
+}
+
+// static
+std::unique_ptr<DesktopFrame> SharedMemoryDesktopFrame::Create(
+    DesktopSize size,
+    SharedMemoryFactory* shared_memory_factory) {
+  RTC_DCHECK(shared_memory_factory);
+
+  size_t buffer_size = size.height() * size.width() * kBytesPerPixel;
+  std::unique_ptr<SharedMemory> shared_memory =
+      shared_memory_factory->CreateSharedMemory(buffer_size);
+  if (!shared_memory)
+    return nullptr;
+
+  return std::make_unique<SharedMemoryDesktopFrame>(
+      size, size.width() * kBytesPerPixel, std::move(shared_memory));
+}
+
+SharedMemoryDesktopFrame::SharedMemoryDesktopFrame(DesktopSize size,
+                                                   int stride,
+                                                   SharedMemory* shared_memory)
+    : DesktopFrame(size,
+                   stride,
+                   reinterpret_cast<uint8_t*>(shared_memory->data()),
+                   shared_memory) {}
+
+SharedMemoryDesktopFrame::SharedMemoryDesktopFrame(
+    DesktopSize size,
+    int stride,
+    std::unique_ptr<SharedMemory> shared_memory)
+    : SharedMemoryDesktopFrame(size, stride, shared_memory.release()) {}
+
+SharedMemoryDesktopFrame::~SharedMemoryDesktopFrame() {
+  delete shared_memory_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame.h b/third_party/libwebrtc/modules/desktop_capture/desktop_frame.h
new file mode 100644
index 0000000000..3ee1867e70
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame.h
@@ -0,0 +1,227 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/shared_memory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+const float kStandardDPI = 96.0f;
+
+// DesktopFrame represents a video frame captured from the screen.
+class RTC_EXPORT DesktopFrame {
+ public:
+  // DesktopFrame objects always hold BGRA data.
+  static const int kBytesPerPixel = 4;
+
+  virtual ~DesktopFrame();
+
+  DesktopFrame(const DesktopFrame&) = delete;
+  DesktopFrame& operator=(const DesktopFrame&) = delete;
+
+  // Returns the rectangle in full desktop coordinates to indicate it covers
+  // the area of top_left() to top_letf() + size() / scale_factor().
+  DesktopRect rect() const;
+
+  // Returns the scale factor from DIPs to physical pixels of the frame.
+  // Assumes same scale in both X and Y directions at present.
+  float scale_factor() const;
+
+  // Size of the frame. In physical coordinates, mapping directly from the
+  // underlying buffer.
+  const DesktopSize& size() const { return size_; }
+
+  // The top-left of the frame in full desktop coordinates. E.g. the top left
+  // monitor should start from (0, 0). The desktop coordinates may be scaled by
+  // OS, but this is always consistent with the MouseCursorMonitor.
+  const DesktopVector& top_left() const { return top_left_; }
+  void set_top_left(const DesktopVector& top_left) { top_left_ = top_left; }
+
+  // Distance in the buffer between two neighboring rows in bytes.
+  int stride() const { return stride_; }
+
+  // Data buffer used for the frame.
+  uint8_t* data() const { return data_; }
+
+  // SharedMemory used for the buffer or NULL if memory is allocated on the
+  // heap. The result is guaranteed to be deleted only after the frame is
+  // deleted (classes that inherit from DesktopFrame must ensure it).
+  SharedMemory* shared_memory() const { return shared_memory_; }
+
+  // Indicates region of the screen that has changed since the previous frame.
+  const DesktopRegion& updated_region() const { return updated_region_; }
+  DesktopRegion* mutable_updated_region() { return &updated_region_; }
+
+  // DPI of the screen being captured. May be set to zero, e.g. if DPI is
+  // unknown.
+  const DesktopVector& dpi() const { return dpi_; }
+  void set_dpi(const DesktopVector& dpi) { dpi_ = dpi; }
+
+  // Indicates if this frame may have the mouse cursor in it. Capturers that
+  // support cursor capture may set this to true. If the cursor was
+  // outside of the captured area, this may be true even though the cursor is
+  // not in the image.
+  bool may_contain_cursor() const { return may_contain_cursor_; }
+  void set_may_contain_cursor(bool may_contain_cursor) {
+    may_contain_cursor_ = may_contain_cursor;
+  }
+
+  // Time taken to capture the frame in milliseconds.
+  int64_t capture_time_ms() const { return capture_time_ms_; }
+  void set_capture_time_ms(int64_t time_ms) { capture_time_ms_ = time_ms; }
+
+  // Copies pixels from a buffer or another frame. `dest_rect` rect must lay
+  // within bounds of this frame.
+  void CopyPixelsFrom(const uint8_t* src_buffer,
+                      int src_stride,
+                      const DesktopRect& dest_rect);
+  void CopyPixelsFrom(const DesktopFrame& src_frame,
+                      const DesktopVector& src_pos,
+                      const DesktopRect& dest_rect);
+
+  // Copies pixels from another frame, with the copied & overwritten regions
+  // representing the intersection between the two frames. Returns true if
+  // pixels were copied, or false if there's no intersection. The scale factors
+  // represent the ratios between pixel space & offset coordinate space (e.g.
+  // 2.0 would indicate the frames are scaled down by 50% for display, so any
+  // offset between their origins should be doubled).
+  bool CopyIntersectingPixelsFrom(const DesktopFrame& src_frame,
+                                  double horizontal_scale,
+                                  double vertical_scale);
+
+  // A helper to return the data pointer of a frame at the specified position.
+  uint8_t* GetFrameDataAtPos(const DesktopVector& pos) const;
+
+  // The DesktopCapturer implementation which generates current DesktopFrame.
+  // Not all DesktopCapturer implementations set this field; it's set to
+  // kUnknown by default.
+  uint32_t capturer_id() const { return capturer_id_; }
+  void set_capturer_id(uint32_t capturer_id) { capturer_id_ = capturer_id; }
+
+  // Copies various information from `other`. Anything initialized in
+  // constructor are not copied.
+  // This function is usually used when sharing a source DesktopFrame with
+  // several clients: the original DesktopFrame should be kept unchanged. For
+  // example, BasicDesktopFrame::CopyOf() and SharedDesktopFrame::Share().
+  void CopyFrameInfoFrom(const DesktopFrame& other);
+
+  // Copies various information from `other`. Anything initialized in
+  // constructor are not copied. Not like CopyFrameInfoFrom() function, this
+  // function uses swap or move constructor to avoid data copy. It won't break
+  // the `other`, but some of its information may be missing after this
+  // operation. E.g. other->updated_region_;
+  // This function is usually used when wrapping a DesktopFrame: the wrapper
+  // instance takes the ownership of `other`, so other components cannot access
+  // `other` anymore. For example, CroppedDesktopFrame and
+  // DesktopFrameWithCursor.
+  void MoveFrameInfoFrom(DesktopFrame* other);
+
+  // Set and get the ICC profile of the frame data pixels. Useful to build the
+  // a ColorSpace object from clients of webrtc library like chromium. The
+  // format of an ICC profile is defined in the following specification
+  // http://www.color.org/specification/ICC1v43_2010-12.pdf.
+  const std::vector<uint8_t>& icc_profile() const { return icc_profile_; }
+  void set_icc_profile(const std::vector<uint8_t>& icc_profile) {
+    icc_profile_ = icc_profile;
+  }
+
+ protected:
+  DesktopFrame(DesktopSize size,
+               int stride,
+               uint8_t* data,
+               SharedMemory* shared_memory);
+
+  // Ownership of the buffers is defined by the classes that inherit from this
+  // class. They must guarantee that the buffer is not deleted before the frame
+  // is deleted.
+  uint8_t* const data_;
+  SharedMemory* const shared_memory_;
+
+ private:
+  const DesktopSize size_;
+  const int stride_;
+
+  DesktopRegion updated_region_;
+  DesktopVector top_left_;
+  DesktopVector dpi_;
+  bool may_contain_cursor_ = false;
+  int64_t capture_time_ms_;
+  uint32_t capturer_id_;
+  std::vector<uint8_t> icc_profile_;
+};
+
+// A DesktopFrame that stores data in the heap.
+class RTC_EXPORT BasicDesktopFrame : public DesktopFrame {
+ public:
+  // The entire data buffer used for the frame is initialized with zeros.
+  explicit BasicDesktopFrame(DesktopSize size);
+
+  ~BasicDesktopFrame() override;
+
+  BasicDesktopFrame(const BasicDesktopFrame&) = delete;
+  BasicDesktopFrame& operator=(const BasicDesktopFrame&) = delete;
+
+  // Creates a BasicDesktopFrame that contains copy of `frame`.
+  // TODO(zijiehe): Return std::unique_ptr<DesktopFrame>
+  static DesktopFrame* CopyOf(const DesktopFrame& frame);
+};
+
+// A DesktopFrame that stores data in shared memory.
+class RTC_EXPORT SharedMemoryDesktopFrame : public DesktopFrame {
+ public:
+  // May return nullptr if `shared_memory_factory` failed to create a
+  // SharedMemory instance.
+  // `shared_memory_factory` should not be nullptr.
+  static std::unique_ptr<DesktopFrame> Create(
+      DesktopSize size,
+      SharedMemoryFactory* shared_memory_factory);
+
+  // Takes ownership of `shared_memory`.
+  // Deprecated, use the next constructor.
+  SharedMemoryDesktopFrame(DesktopSize size,
+                           int stride,
+                           SharedMemory* shared_memory);
+
+  // Preferred.
+  SharedMemoryDesktopFrame(DesktopSize size,
+                           int stride,
+                           std::unique_ptr<SharedMemory> shared_memory);
+
+  ~SharedMemoryDesktopFrame() override;
+
+  SharedMemoryDesktopFrame(const SharedMemoryDesktopFrame&) = delete;
+  SharedMemoryDesktopFrame& operator=(const SharedMemoryDesktopFrame&) = delete;
+
+ private:
+  // Avoid unexpected order of parameter evaluation.
+  // Executing both std::unique_ptr<T>::operator->() and
+  // std::unique_ptr<T>::release() in the member initializer list is not safe.
+  // Depends on the order of parameter evaluation,
+  // std::unique_ptr<T>::operator->() may trigger assertion failure if it has
+  // been evaluated after std::unique_ptr<T>::release(). By using this
+  // constructor, std::unique_ptr<T>::operator->() won't be involved anymore.
+  SharedMemoryDesktopFrame(DesktopRect rect,
+                           int stride,
+                           SharedMemory* shared_memory);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame_generator.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_generator.cc
new file mode 100644
index 0000000000..b5dfc28e46
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_generator.cc
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_frame_generator.h"
+
+#include <stdint.h>
+#include <string.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/rgba_color.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/random.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+// Sets `updated_region` to `frame`. If `enlarge_updated_region` is
+// true, this function will randomly enlarge each DesktopRect in
+// `updated_region`. But the enlarged DesktopRegion won't excceed the
+// frame->size(). If `add_random_updated_region` is true, several random
+// rectangles will also be included in `frame`.
+void SetUpdatedRegion(DesktopFrame* frame,
+                      const DesktopRegion& updated_region,
+                      bool enlarge_updated_region,
+                      int enlarge_range,
+                      bool add_random_updated_region) {
+  const DesktopRect screen_rect = DesktopRect::MakeSize(frame->size());
+  Random random(rtc::TimeMicros());
+  frame->mutable_updated_region()->Clear();
+  for (DesktopRegion::Iterator it(updated_region); !it.IsAtEnd();
+       it.Advance()) {
+    DesktopRect rect = it.rect();
+    if (enlarge_updated_region && enlarge_range > 0) {
+      rect.Extend(random.Rand(enlarge_range), random.Rand(enlarge_range),
+                  random.Rand(enlarge_range), random.Rand(enlarge_range));
+      rect.IntersectWith(screen_rect);
+    }
+    frame->mutable_updated_region()->AddRect(rect);
+  }
+
+  if (add_random_updated_region) {
+    for (int i = random.Rand(10); i >= 0; i--) {
+      // At least a 1 x 1 updated region.
+      const int left = random.Rand(0, frame->size().width() - 2);
+      const int top = random.Rand(0, frame->size().height() - 2);
+      const int right = random.Rand(left + 1, frame->size().width());
+      const int bottom = random.Rand(top + 1, frame->size().height());
+      frame->mutable_updated_region()->AddRect(
+          DesktopRect::MakeLTRB(left, top, right, bottom));
+    }
+  }
+}
+
+// Paints pixels in `rect` of `frame` to `color`.
+void PaintRect(DesktopFrame* frame, DesktopRect rect, RgbaColor rgba_color) {
+  static_assert(DesktopFrame::kBytesPerPixel == sizeof(uint32_t),
+                "kBytesPerPixel should be 4.");
+  RTC_DCHECK_GE(frame->size().width(), rect.right());
+  RTC_DCHECK_GE(frame->size().height(), rect.bottom());
+  uint32_t color = rgba_color.ToUInt32();
+  uint8_t* row = frame->GetFrameDataAtPos(rect.top_left());
+  for (int i = 0; i < rect.height(); i++) {
+    uint32_t* column = reinterpret_cast<uint32_t*>(row);
+    for (int j = 0; j < rect.width(); j++) {
+      column[j] = color;
+    }
+    row += frame->stride();
+  }
+}
+
+// Paints pixels in `region` of `frame` to `color`.
+void PaintRegion(DesktopFrame* frame,
+                 DesktopRegion* region,
+                 RgbaColor rgba_color) {
+  region->IntersectWith(DesktopRect::MakeSize(frame->size()));
+  for (DesktopRegion::Iterator it(*region); !it.IsAtEnd(); it.Advance()) {
+    PaintRect(frame, it.rect(), rgba_color);
+  }
+}
+
+}  // namespace
+
+DesktopFrameGenerator::DesktopFrameGenerator() {}
+DesktopFrameGenerator::~DesktopFrameGenerator() {}
+
+DesktopFramePainter::DesktopFramePainter() {}
+DesktopFramePainter::~DesktopFramePainter() {}
+
+PainterDesktopFrameGenerator::PainterDesktopFrameGenerator()
+    : size_(1024, 768),
+      return_frame_(true),
+      provide_updated_region_hints_(false),
+      enlarge_updated_region_(false),
+      enlarge_range_(20),
+      add_random_updated_region_(false),
+      painter_(nullptr) {}
+PainterDesktopFrameGenerator::~PainterDesktopFrameGenerator() {}
+
+std::unique_ptr<DesktopFrame> PainterDesktopFrameGenerator::GetNextFrame(
+    SharedMemoryFactory* factory) {
+  if (!return_frame_) {
+    return nullptr;
+  }
+
+  std::unique_ptr<DesktopFrame> frame = std::unique_ptr<DesktopFrame>(
+      factory ? SharedMemoryDesktopFrame::Create(size_, factory).release()
+              : new BasicDesktopFrame(size_));
+  if (painter_) {
+    DesktopRegion updated_region;
+    if (!painter_->Paint(frame.get(), &updated_region)) {
+      return nullptr;
+    }
+
+    if (provide_updated_region_hints_) {
+      SetUpdatedRegion(frame.get(), updated_region, enlarge_updated_region_,
+                       enlarge_range_, add_random_updated_region_);
+    } else {
+      frame->mutable_updated_region()->SetRect(
+          DesktopRect::MakeSize(frame->size()));
+    }
+  }
+
+  return frame;
+}
+
+DesktopSize* PainterDesktopFrameGenerator::size() {
+  return &size_;
+}
+
+void PainterDesktopFrameGenerator::set_return_frame(bool return_frame) {
+  return_frame_ = return_frame;
+}
+
+void PainterDesktopFrameGenerator::set_provide_updated_region_hints(
+    bool provide_updated_region_hints) {
+  provide_updated_region_hints_ = provide_updated_region_hints;
+}
+
+void PainterDesktopFrameGenerator::set_enlarge_updated_region(
+    bool enlarge_updated_region) {
+  enlarge_updated_region_ = enlarge_updated_region;
+}
+
+void PainterDesktopFrameGenerator::set_enlarge_range(int enlarge_range) {
+  enlarge_range_ = enlarge_range;
+}
+
+void PainterDesktopFrameGenerator::set_add_random_updated_region(
+    bool add_random_updated_region) {
+  add_random_updated_region_ = add_random_updated_region;
+}
+
+void PainterDesktopFrameGenerator::set_desktop_frame_painter(
+    DesktopFramePainter* painter) {
+  painter_ = painter;
+}
+
+BlackWhiteDesktopFramePainter::BlackWhiteDesktopFramePainter() {}
+BlackWhiteDesktopFramePainter::~BlackWhiteDesktopFramePainter() {}
+
+DesktopRegion* BlackWhiteDesktopFramePainter::updated_region() {
+  return &updated_region_;
+}
+
+bool BlackWhiteDesktopFramePainter::Paint(DesktopFrame* frame,
+                                          DesktopRegion* updated_region) {
+  RTC_DCHECK(updated_region->is_empty());
+  memset(frame->data(), 0, frame->stride() * frame->size().height());
+  PaintRegion(frame, &updated_region_, RgbaColor(0xFFFFFFFF));
+  updated_region_.Swap(updated_region);
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame_generator.h b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_generator.h
new file mode 100644
index 0000000000..3dedee9344
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_generator.h
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_GENERATOR_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_GENERATOR_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/shared_memory.h"
+
+namespace webrtc {
+
+// An interface to generate a DesktopFrame.
+class DesktopFrameGenerator {
+ public:
+  DesktopFrameGenerator();
+  virtual ~DesktopFrameGenerator();
+
+  virtual std::unique_ptr<DesktopFrame> GetNextFrame(
+      SharedMemoryFactory* factory) = 0;
+};
+
+// An interface to paint a DesktopFrame. This interface is used by
+// PainterDesktopFrameGenerator.
+class DesktopFramePainter {
+ public:
+  DesktopFramePainter();
+  virtual ~DesktopFramePainter();
+
+  virtual bool Paint(DesktopFrame* frame, DesktopRegion* updated_region) = 0;
+};
+
+// An implementation of DesktopFrameGenerator to take care about the
+// DesktopFrame size, filling updated_region(), etc, but leaves the real
+// painting work to a DesktopFramePainter implementation.
+class PainterDesktopFrameGenerator final : public DesktopFrameGenerator {
+ public:
+  PainterDesktopFrameGenerator();
+  ~PainterDesktopFrameGenerator() override;
+
+  std::unique_ptr<DesktopFrame> GetNextFrame(
+      SharedMemoryFactory* factory) override;
+
+  // Sets the size of the frame which will be returned in next GetNextFrame()
+  // call.
+  DesktopSize* size();
+
+  // Decides whether BaseDesktopFrameGenerator returns a frame in next Capture()
+  // callback. If return_frame_ is true, BaseDesktopFrameGenerator will create a
+  // frame according to both size_ and SharedMemoryFactory input, and uses
+  // Paint() function to paint it.
+  void set_return_frame(bool return_frame);
+
+  // Decides whether MockScreenCapturer returns a frame with updated regions.
+  // MockScreenCapturer will keep DesktopFrame::updated_region() empty if this
+  // field is false.
+  void set_provide_updated_region_hints(bool provide_updated_region_hints);
+
+  // Decides whether MockScreenCapturer randomly enlarges updated regions in the
+  // DesktopFrame. Set this field to true to simulate an inaccurate updated
+  // regions' return from OS APIs.
+  void set_enlarge_updated_region(bool enlarge_updated_region);
+
+  // The range to enlarge a updated region if `enlarge_updated_region_` is true.
+  // If this field is less than zero, it will be treated as zero, and
+  // `enlarge_updated_region_` will be ignored.
+  void set_enlarge_range(int enlarge_range);
+
+  // Decides whether BaseDesktopFrameGenerator randomly add some updated regions
+  // in the DesktopFrame. Set this field to true to simulate an inaccurate
+  // updated regions' return from OS APIs.
+  void set_add_random_updated_region(bool add_random_updated_region);
+
+  // Sets the painter object to do the real painting work, if no `painter_` has
+  // been set to this instance, the DesktopFrame returned by GetNextFrame()
+  // function will keep in an undefined but valid state.
+  // PainterDesktopFrameGenerator does not take ownership of the `painter`.
+  void set_desktop_frame_painter(DesktopFramePainter* painter);
+
+ private:
+  DesktopSize size_;
+  bool return_frame_;
+  bool provide_updated_region_hints_;
+  bool enlarge_updated_region_;
+  int enlarge_range_;
+  bool add_random_updated_region_;
+  DesktopFramePainter* painter_;
+};
+
+// An implementation of DesktopFramePainter to paint black on
+// mutable_updated_region(), and white elsewhere.
+class BlackWhiteDesktopFramePainter final : public DesktopFramePainter {
+ public:
+  BlackWhiteDesktopFramePainter();
+  ~BlackWhiteDesktopFramePainter() override;
+
+  // The black regions of the frame which will be returned in next Paint()
+  // call. BlackWhiteDesktopFramePainter will draw a white frame, with black
+  // in the updated_region_. Each Paint() call will consume updated_region_.
+  DesktopRegion* updated_region();
+
+  // DesktopFramePainter interface.
+  bool Paint(DesktopFrame* frame, DesktopRegion* updated_region) override;
+
+ private:
+  DesktopRegion updated_region_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation.cc
new file mode 100644
index 0000000000..f2745cf70b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation.cc
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_frame_rotation.h"
+
+#include "rtc_base/checks.h"
+#include "third_party/libyuv/include/libyuv/rotate_argb.h"
+
+namespace webrtc {
+
+namespace {
+
+libyuv::RotationMode ToLibyuvRotationMode(Rotation rotation) {
+  switch (rotation) {
+    case Rotation::CLOCK_WISE_0:
+      return libyuv::kRotate0;
+    case Rotation::CLOCK_WISE_90:
+      return libyuv::kRotate90;
+    case Rotation::CLOCK_WISE_180:
+      return libyuv::kRotate180;
+    case Rotation::CLOCK_WISE_270:
+      return libyuv::kRotate270;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return libyuv::kRotate0;
+}
+
+DesktopRect RotateAndOffsetRect(DesktopRect rect,
+                                DesktopSize size,
+                                Rotation rotation,
+                                DesktopVector offset) {
+  DesktopRect result = RotateRect(rect, size, rotation);
+  result.Translate(offset);
+  return result;
+}
+
+}  // namespace
+
+Rotation ReverseRotation(Rotation rotation) {
+  switch (rotation) {
+    case Rotation::CLOCK_WISE_0:
+      return rotation;
+    case Rotation::CLOCK_WISE_90:
+      return Rotation::CLOCK_WISE_270;
+    case Rotation::CLOCK_WISE_180:
+      return Rotation::CLOCK_WISE_180;
+    case Rotation::CLOCK_WISE_270:
+      return Rotation::CLOCK_WISE_90;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return Rotation::CLOCK_WISE_0;
+}
+
+DesktopSize RotateSize(DesktopSize size, Rotation rotation) {
+  switch (rotation) {
+    case Rotation::CLOCK_WISE_0:
+    case Rotation::CLOCK_WISE_180:
+      return size;
+    case Rotation::CLOCK_WISE_90:
+    case Rotation::CLOCK_WISE_270:
+      return DesktopSize(size.height(), size.width());
+  }
+  RTC_DCHECK_NOTREACHED();
+  return DesktopSize();
+}
+
+DesktopRect RotateRect(DesktopRect rect, DesktopSize size, Rotation rotation) {
+  switch (rotation) {
+    case Rotation::CLOCK_WISE_0:
+      return rect;
+    case Rotation::CLOCK_WISE_90:
+      return DesktopRect::MakeXYWH(size.height() - rect.bottom(), rect.left(),
+                                   rect.height(), rect.width());
+    case Rotation::CLOCK_WISE_180:
+      return DesktopRect::MakeXYWH(size.width() - rect.right(),
+                                   size.height() - rect.bottom(), rect.width(),
+                                   rect.height());
+    case Rotation::CLOCK_WISE_270:
+      return DesktopRect::MakeXYWH(rect.top(), size.width() - rect.right(),
+                                   rect.height(), rect.width());
+  }
+  RTC_DCHECK_NOTREACHED();
+  return DesktopRect();
+}
+
+void RotateDesktopFrame(const DesktopFrame& source,
+                        const DesktopRect& source_rect,
+                        const Rotation& rotation,
+                        const DesktopVector& target_offset,
+                        DesktopFrame* target) {
+  RTC_DCHECK(target);
+  RTC_DCHECK(DesktopRect::MakeSize(source.size()).ContainsRect(source_rect));
+  // The rectangle in `target`.
+  const DesktopRect target_rect =
+      RotateAndOffsetRect(source_rect, source.size(), rotation, target_offset);
+  RTC_DCHECK(DesktopRect::MakeSize(target->size()).ContainsRect(target_rect));
+
+  if (target_rect.is_empty()) {
+    return;
+  }
+
+  int result = libyuv::ARGBRotate(
+      source.GetFrameDataAtPos(source_rect.top_left()), source.stride(),
+      target->GetFrameDataAtPos(target_rect.top_left()), target->stride(),
+      source_rect.width(), source_rect.height(),
+      ToLibyuvRotationMode(rotation));
+  RTC_DCHECK_EQ(result, 0);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation.h b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation.h
new file mode 100644
index 0000000000..6b51b2f883
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_ROTATION_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_ROTATION_H_
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+
+namespace webrtc {
+
+// Represents the rotation of a DesktopFrame.
+enum class Rotation {
+  CLOCK_WISE_0,
+  CLOCK_WISE_90,
+  CLOCK_WISE_180,
+  CLOCK_WISE_270,
+};
+
+// Rotates input DesktopFrame `source`, copies pixel in an unrotated rectangle
+// `source_rect` into the target rectangle of another DesktopFrame `target`.
+// Target rectangle here is the rotated `source_rect` plus `target_offset`.
+// `rotation` specifies `source` to `target` rotation. `source_rect` is in
+// `source` coordinate. `target_offset` is in `target` coordinate.
+// This function triggers check failure if `source` does not cover the
+// `source_rect`, or `target` does not cover the rotated `rect`.
+void RotateDesktopFrame(const DesktopFrame& source,
+                        const DesktopRect& source_rect,
+                        const Rotation& rotation,
+                        const DesktopVector& target_offset,
+                        DesktopFrame* target);
+
+// Returns a reverse rotation of `rotation`.
+Rotation ReverseRotation(Rotation rotation);
+
+// Returns a rotated DesktopSize of `size`.
+DesktopSize RotateSize(DesktopSize size, Rotation rotation);
+
+// Returns a rotated DesktopRect of `rect`. The `size` represents the size of
+// the DesktopFrame which `rect` belongs in.
+DesktopRect RotateRect(DesktopRect rect, DesktopSize size, Rotation rotation);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_ROTATION_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation_unittest.cc
new file mode 100644
index 0000000000..782ca63e61
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_rotation_unittest.cc
@@ -0,0 +1,449 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_frame_rotation.h"
+
+#include <stdint.h>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/test_utils.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// A DesktopFrame implementation which stores data in an external int array.
+class ArrayDesktopFrame : public DesktopFrame {
+ public:
+  ArrayDesktopFrame(DesktopSize size, uint32_t* data);
+  ~ArrayDesktopFrame() override;
+};
+
+ArrayDesktopFrame::ArrayDesktopFrame(DesktopSize size, uint32_t* data)
+    : DesktopFrame(size,
+                   size.width() * kBytesPerPixel,
+                   reinterpret_cast<uint8_t*>(data),
+                   nullptr) {}
+
+ArrayDesktopFrame::~ArrayDesktopFrame() = default;
+
+}  // namespace
+
+TEST(DesktopFrameRotationTest, CopyRect3x4) {
+  // A DesktopFrame of 4-pixel width by 3-pixel height.
+  static uint32_t frame_pixels[] = {
+      0, 1, 2,  3,   //
+      4, 5, 6,  7,   //
+      8, 9, 10, 11,  //
+  };
+  ArrayDesktopFrame frame(DesktopSize(4, 3), frame_pixels);
+
+  {
+    BasicDesktopFrame target(DesktopSize(4, 3));
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_0, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(frame, target));
+  }
+
+  // After Rotating clock-wise 90 degree
+  {
+    static uint32_t expected_pixels[] = {
+        8,  4, 0,  //
+        9,  5, 1,  //
+        10, 6, 2,  //
+        11, 7, 3,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(3, 4), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(3, 4));
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_90, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  // After Rotating clock-wise 180 degree
+  {
+    static uint32_t expected_pixels[] = {
+        11, 10, 9, 8,  //
+        7,  6,  5, 4,  //
+        3,  2,  1, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(4, 3), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(4, 3));
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_180, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  // After Rotating clock-wise 270 degree
+  {
+    static uint32_t expected_pixels[] = {
+        3, 7, 11,  //
+        2, 6, 10,  //
+        1, 5, 9,   //
+        0, 4, 8,   //
+    };
+    ArrayDesktopFrame expected(DesktopSize(3, 4), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(3, 4));
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_270, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+}
+
+TEST(DesktopFrameRotationTest, CopyRect3x5) {
+  // A DesktopFrame of 5-pixel width by 3-pixel height.
+  static uint32_t frame_pixels[] = {
+      0,  1,  2,  3,  4,   //
+      5,  6,  7,  8,  9,   //
+      10, 11, 12, 13, 14,  //
+  };
+  ArrayDesktopFrame frame(DesktopSize(5, 3), frame_pixels);
+
+  {
+    BasicDesktopFrame target(DesktopSize(5, 3));
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_0, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, frame));
+  }
+
+  // After Rotating clock-wise 90 degree
+  {
+    static uint32_t expected_pixels[] = {
+        10, 5, 0,  //
+        11, 6, 1,  //
+        12, 7, 2,  //
+        13, 8, 3,  //
+        14, 9, 4,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(3, 5), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(3, 5));
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_90, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  // After Rotating clock-wise 180 degree
+  {
+    static uint32_t expected_pixels[]{
+        14, 13, 12, 11, 10,  //
+        9,  8,  7,  6,  5,   //
+        4,  3,  2,  1,  0,   //
+    };
+    ArrayDesktopFrame expected(DesktopSize(5, 3), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(5, 3));
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_180, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  // After Rotating clock-wise 270 degree
+  {
+    static uint32_t expected_pixels[] = {
+        4, 9, 14,  //
+        3, 8, 13,  //
+        2, 7, 12,  //
+        1, 6, 11,  //
+        0, 5, 10,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(3, 5), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(3, 5));
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_270, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+}
+
+TEST(DesktopFrameRotationTest, PartialCopyRect3x5) {
+  // A DesktopFrame of 5-pixel width by 3-pixel height.
+  static uint32_t frame_pixels[] = {
+      0,  1,  2,  3,  4,   //
+      5,  6,  7,  8,  9,   //
+      10, 11, 12, 13, 14,  //
+  };
+  ArrayDesktopFrame frame(DesktopSize(5, 3), frame_pixels);
+
+  {
+    static uint32_t expected_pixels[] = {
+        0, 0, 0, 0, 0,  //
+        0, 6, 7, 8, 0,  //
+        0, 0, 0, 0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(5, 3), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(5, 3));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeXYWH(1, 1, 3, 1),
+                       Rotation::CLOCK_WISE_0, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  {
+    static uint32_t expected_pixels[] = {
+        0, 1, 2, 3, 0,  //
+        0, 6, 7, 8, 0,  //
+        0, 0, 0, 0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(5, 3), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(5, 3));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeXYWH(1, 0, 3, 2),
+                       Rotation::CLOCK_WISE_0, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  // After Rotating clock-wise 90 degree
+  {
+    static uint32_t expected_pixels[] = {
+        0, 0, 0,  //
+        0, 6, 0,  //
+        0, 7, 0,  //
+        0, 8, 0,  //
+        0, 0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(3, 5), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(3, 5));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeXYWH(1, 1, 3, 1),
+                       Rotation::CLOCK_WISE_90, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  {
+    static uint32_t expected_pixels[] = {
+        0,  0, 0,  //
+        11, 6, 0,  //
+        12, 7, 0,  //
+        13, 8, 0,  //
+        0,  0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(3, 5), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(3, 5));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeXYWH(1, 1, 3, 2),
+                       Rotation::CLOCK_WISE_90, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  // After Rotating clock-wise 180 degree
+  {
+    static uint32_t expected_pixels[] = {
+        0, 0, 0, 0, 0,  //
+        0, 8, 7, 6, 0,  //
+        0, 0, 0, 0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(5, 3), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(5, 3));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeXYWH(1, 1, 3, 1),
+                       Rotation::CLOCK_WISE_180, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  {
+    static uint32_t expected_pixels[] = {
+        0, 13, 12, 11, 0,  //
+        0, 8,  7,  6,  0,  //
+        0, 0,  0,  0,  0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(5, 3), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(5, 3));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeXYWH(1, 1, 3, 2),
+                       Rotation::CLOCK_WISE_180, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  // After Rotating clock-wise 270 degree
+  {
+    static uint32_t expected_pixels[] = {
+        0, 0, 0,  //
+        0, 8, 0,  //
+        0, 7, 0,  //
+        0, 6, 0,  //
+        0, 0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(3, 5), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(3, 5));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeXYWH(1, 1, 3, 1),
+                       Rotation::CLOCK_WISE_270, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+
+  {
+    static uint32_t expected_pixels[] = {
+        0, 0, 0,  //
+        3, 8, 0,  //
+        2, 7, 0,  //
+        1, 6, 0,  //
+        0, 0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(3, 5), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(3, 5));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeXYWH(1, 0, 3, 2),
+                       Rotation::CLOCK_WISE_270, DesktopVector(), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+  }
+}
+
+TEST(DesktopFrameRotationTest, WithOffset) {
+  // A DesktopFrame of 4-pixel width by 3-pixel height.
+  static uint32_t frame_pixels[] = {
+      0, 1, 2,  3,   //
+      4, 5, 6,  7,   //
+      8, 9, 10, 11,  //
+  };
+  ArrayDesktopFrame frame(DesktopSize(4, 3), frame_pixels);
+
+  {
+    static uint32_t expected_pixels[] = {
+        0, 0, 0, 0,  0,  0, 0, 0,  //
+        0, 0, 1, 2,  3,  0, 0, 0,  //
+        0, 4, 5, 6,  7,  0, 0, 0,  //
+        0, 8, 9, 10, 11, 0, 0, 0,  //
+        0, 0, 0, 0,  0,  0, 0, 0,  //
+        0, 0, 0, 0,  0,  0, 0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(8, 6), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(8, 6));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_0, DesktopVector(1, 1), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+    target.mutable_updated_region()->Subtract(
+        DesktopRect::MakeOriginSize(DesktopVector(1, 1), frame.size()));
+    ASSERT_TRUE(target.updated_region().is_empty());
+  }
+
+  {
+    static uint32_t expected_pixels[] = {
+        0, 0,  0,  0, 0, 0, 0, 0,  //
+        0, 11, 10, 9, 8, 0, 0, 0,  //
+        0, 7,  6,  5, 4, 0, 0, 0,  //
+        0, 3,  2,  1, 0, 0, 0, 0,  //
+        0, 0,  0,  0, 0, 0, 0, 0,  //
+        0, 0,  0,  0, 0, 0, 0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(8, 6), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(8, 6));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_180, DesktopVector(1, 1), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+    target.mutable_updated_region()->Subtract(
+        DesktopRect::MakeOriginSize(DesktopVector(1, 1), frame.size()));
+    ASSERT_TRUE(target.updated_region().is_empty());
+  }
+
+  {
+    static uint32_t expected_pixels[] = {
+        0, 0,  0, 0, 0, 0,  //
+        0, 8,  4, 0, 0, 0,  //
+        0, 9,  5, 1, 0, 0,  //
+        0, 10, 6, 2, 0, 0,  //
+        0, 11, 7, 3, 0, 0,  //
+        0, 0,  0, 0, 0, 0,  //
+        0, 0,  0, 0, 0, 0,  //
+        0, 0,  0, 0, 0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(6, 8), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(6, 8));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_90, DesktopVector(1, 1), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+    target.mutable_updated_region()->Subtract(
+        DesktopRect::MakeXYWH(1, 1, 3, 4));
+    ASSERT_TRUE(target.updated_region().is_empty());
+  }
+
+  {
+    static uint32_t expected_pixels[] = {
+        0, 0, 0, 0,  0, 0,  //
+        0, 3, 7, 11, 0, 0,  //
+        0, 2, 6, 10, 0, 0,  //
+        0, 1, 5, 9,  0, 0,  //
+        0, 0, 4, 8,  0, 0,  //
+        0, 0, 0, 0,  0, 0,  //
+        0, 0, 0, 0,  0, 0,  //
+        0, 0, 0, 0,  0, 0,  //
+    };
+    ArrayDesktopFrame expected(DesktopSize(6, 8), expected_pixels);
+
+    BasicDesktopFrame target(DesktopSize(6, 8));
+    ClearDesktopFrame(&target);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_270, DesktopVector(1, 1), &target);
+    ASSERT_TRUE(DesktopFrameDataEquals(target, expected));
+    target.mutable_updated_region()->Subtract(
+        DesktopRect::MakeXYWH(1, 1, 3, 4));
+    ASSERT_TRUE(target.updated_region().is_empty());
+  }
+}
+
+// On a typical machine (Intel(R) Xeon(R) E5-1650 v3 @ 3.50GHz, with O2
+// optimization, the following case uses ~1.4s to finish. It means entirely
+// rotating one 2048 x 1536 frame, which is a large enough number to cover most
+// of desktop computer users, uses around 14ms.
+TEST(DesktopFrameRotationTest, DISABLED_PerformanceTest) {
+  BasicDesktopFrame frame(DesktopSize(2048, 1536));
+  BasicDesktopFrame target(DesktopSize(1536, 2048));
+  BasicDesktopFrame target2(DesktopSize(2048, 1536));
+  for (int i = 0; i < 100; i++) {
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_90, DesktopVector(), &target);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_270, DesktopVector(), &target);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_0, DesktopVector(), &target2);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_180, DesktopVector(), &target2);
+  }
+}
+
+// On a typical machine (Intel(R) Xeon(R) E5-1650 v3 @ 3.50GHz, with O2
+// optimization, the following case uses ~6.7s to finish. It means entirely
+// rotating one 4096 x 3072 frame uses around 67ms.
+TEST(DesktopFrameRotationTest, DISABLED_PerformanceTestOnLargeScreen) {
+  BasicDesktopFrame frame(DesktopSize(4096, 3072));
+  BasicDesktopFrame target(DesktopSize(3072, 4096));
+  BasicDesktopFrame target2(DesktopSize(4096, 3072));
+  for (int i = 0; i < 100; i++) {
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_90, DesktopVector(), &target);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_270, DesktopVector(), &target);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_0, DesktopVector(), &target2);
+    RotateDesktopFrame(frame, DesktopRect::MakeSize(frame.size()),
+                       Rotation::CLOCK_WISE_180, DesktopVector(), &target2);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_unittest.cc
new file mode 100644
index 0000000000..ce0cbb45f5
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_unittest.cc
@@ -0,0 +1,336 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_frame.h"
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/test_utils.h"
+#include "rtc_base/arraysize.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+std::unique_ptr<DesktopFrame> CreateTestFrame(DesktopRect rect,
+                                              int pixels_value) {
+  DesktopSize size = rect.size();
+  auto frame = std::make_unique<BasicDesktopFrame>(size);
+  frame->set_top_left(rect.top_left());
+  memset(frame->data(), pixels_value, frame->stride() * size.height());
+  return frame;
+}
+
+struct TestData {
+  const char* description;
+  DesktopRect dest_frame_rect;
+  DesktopRect src_frame_rect;
+  double horizontal_scale;
+  double vertical_scale;
+  DesktopRect expected_overlap_rect;
+};
+
+void RunTest(const TestData& test) {
+  // Copy a source frame with all bits set into a dest frame with none set.
+  auto dest_frame = CreateTestFrame(test.dest_frame_rect, 0);
+  auto src_frame = CreateTestFrame(test.src_frame_rect, 0xff);
+
+  dest_frame->CopyIntersectingPixelsFrom(
+      *src_frame, test.horizontal_scale, test.vertical_scale);
+
+  // Translate the expected overlap rect to be relative to the dest frame/rect.
+  DesktopVector dest_frame_origin = test.dest_frame_rect.top_left();
+  DesktopRect relative_expected_overlap_rect = test.expected_overlap_rect;
+  relative_expected_overlap_rect.Translate(-dest_frame_origin.x(),
+                                           -dest_frame_origin.y());
+
+  // Confirm bits are now set in the dest frame if & only if they fall in the
+  // expected range.
+  for (int y = 0; y < dest_frame->size().height(); ++y) {
+    SCOPED_TRACE(y);
+
+    for (int x = 0; x < dest_frame->size().width(); ++x) {
+      SCOPED_TRACE(x);
+
+      DesktopVector point(x, y);
+      uint8_t* data = dest_frame->GetFrameDataAtPos(point);
+      uint32_t pixel_value = *reinterpret_cast<uint32_t*>(data);
+      bool was_copied = pixel_value == 0xffffffff;
+      ASSERT_TRUE(was_copied || pixel_value == 0);
+
+      bool expected_to_be_copied =
+          relative_expected_overlap_rect.Contains(point);
+
+      ASSERT_EQ(was_copied, expected_to_be_copied);
+    }
+  }
+}
+
+void RunTests(const TestData* tests, int num_tests) {
+  for (int i = 0; i < num_tests; i++) {
+    const TestData& test = tests[i];
+
+    SCOPED_TRACE(test.description);
+
+    RunTest(test);
+  }
+}
+
+}  // namespace
+
+TEST(DesktopFrameTest, CopyIntersectingPixelsMatchingRects) {
+  const TestData tests[] = {
+    {"0 origin",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(0, 0, 2, 2)},
+
+    {"Negative origin",
+     DesktopRect::MakeXYWH(-1, -1, 2, 2),
+     DesktopRect::MakeXYWH(-1, -1, 2, 2),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(-1, -1, 2, 2)}
+  };
+
+  RunTests(tests, arraysize(tests));
+}
+
+TEST(DesktopFrameTest, CopyIntersectingPixelsMatchingRectsScaled) {
+  // The scale factors shouldn't affect matching rects (they're only applied
+  // to any difference between the origins)
+  const TestData tests[] = {
+    {"0 origin 2x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     2.0, 2.0,
+     DesktopRect::MakeXYWH(0, 0, 2, 2)},
+
+    {"0 origin 0.5x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     0.5, 0.5,
+     DesktopRect::MakeXYWH(0, 0, 2, 2)},
+
+    {"Negative origin 2x",
+     DesktopRect::MakeXYWH(-1, -1, 2, 2),
+     DesktopRect::MakeXYWH(-1, -1, 2, 2),
+     2.0, 2.0,
+     DesktopRect::MakeXYWH(-1, -1, 2, 2)},
+
+    {"Negative origin 0.5x",
+     DesktopRect::MakeXYWH(-1, -1, 2, 2),
+     DesktopRect::MakeXYWH(-1, -1, 2, 2),
+     0.5, 0.5,
+     DesktopRect::MakeXYWH(-1, -1, 2, 2)}
+  };
+
+  RunTests(tests, arraysize(tests));
+}
+
+TEST(DesktopFrameTest, CopyIntersectingPixelsFullyContainedRects) {
+  const TestData tests[] = {
+    {"0 origin top left",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, 0, 1, 1),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(0, 0, 1, 1)},
+
+    {"0 origin bottom right",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(1, 1, 1, 1),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(1, 1, 1, 1)},
+
+    {"Negative origin bottom left",
+     DesktopRect::MakeXYWH(-1, -1, 2, 2),
+     DesktopRect::MakeXYWH(-1, 0, 1, 1),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(-1, 0, 1, 1)}
+  };
+
+  RunTests(tests, arraysize(tests));
+}
+
+TEST(DesktopFrameTest, CopyIntersectingPixelsFullyContainedRectsScaled) {
+  const TestData tests[] = {
+    {"0 origin top left 2x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, 0, 1, 1),
+     2.0, 2.0,
+     DesktopRect::MakeXYWH(0, 0, 1, 1)},
+
+    {"0 origin top left 0.5x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, 0, 1, 1),
+     0.5, 0.5,
+     DesktopRect::MakeXYWH(0, 0, 1, 1)},
+
+    {"0 origin bottom left 2x",
+     DesktopRect::MakeXYWH(0, 0, 4, 4),
+     DesktopRect::MakeXYWH(1, 1, 2, 2),
+     2.0, 2.0,
+     DesktopRect::MakeXYWH(2, 2, 2, 2)},
+
+    {"0 origin bottom middle 2x/1x",
+     DesktopRect::MakeXYWH(0, 0, 4, 3),
+     DesktopRect::MakeXYWH(1, 1, 2, 2),
+     2.0, 1.0,
+     DesktopRect::MakeXYWH(2, 1, 2, 2)},
+
+    {"0 origin middle 0.5x",
+     DesktopRect::MakeXYWH(0, 0, 3, 3),
+     DesktopRect::MakeXYWH(2, 2, 1, 1),
+     0.5, 0.5,
+     DesktopRect::MakeXYWH(1, 1, 1, 1)},
+
+    {"Negative origin bottom left 2x",
+     DesktopRect::MakeXYWH(-1, -1, 3, 3),
+     DesktopRect::MakeXYWH(-1, 0, 1, 1),
+     2.0, 2.0,
+     DesktopRect::MakeXYWH(-1, 1, 1, 1)},
+
+    {"Negative origin near middle 0.5x",
+     DesktopRect::MakeXYWH(-2, -2, 2, 2),
+     DesktopRect::MakeXYWH(0, 0, 1, 1),
+     0.5, 0.5,
+     DesktopRect::MakeXYWH(-1, -1, 1, 1)}
+  };
+
+  RunTests(tests, arraysize(tests));
+}
+
+
+TEST(DesktopFrameTest, CopyIntersectingPixelsPartiallyContainedRects) {
+  const TestData tests[] = {
+    {"Top left",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(-1, -1, 2, 2),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(0, 0, 1, 1)},
+
+    {"Top right",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(1, -1, 2, 2),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(1, 0, 1, 1)},
+
+    {"Bottom right",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(1, 1, 2, 2),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(1, 1, 1, 1)},
+
+    {"Bottom left",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(-1, 1, 2, 2),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(0, 1, 1, 1)}
+  };
+
+  RunTests(tests, arraysize(tests));
+}
+
+TEST(DesktopFrameTest, CopyIntersectingPixelsPartiallyContainedRectsScaled) {
+  const TestData tests[] = {
+    {"Top left 2x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(-1, -1, 3, 3),
+     2.0, 2.0,
+     DesktopRect::MakeXYWH(0, 0, 1, 1)},
+
+    {"Top right 0.5x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(2, -2, 2, 2),
+     0.5, 0.5,
+     DesktopRect::MakeXYWH(1, 0, 1, 1)},
+
+    {"Bottom right 2x",
+     DesktopRect::MakeXYWH(0, 0, 3, 3),
+     DesktopRect::MakeXYWH(-1, 1, 3, 3),
+     2.0, 2.0,
+     DesktopRect::MakeXYWH(0, 2, 1, 1)},
+
+    {"Bottom left 0.5x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(-2, 2, 2, 2),
+     0.5, 0.5,
+     DesktopRect::MakeXYWH(0, 1, 1, 1)}
+  };
+
+  RunTests(tests, arraysize(tests));
+}
+
+
+TEST(DesktopFrameTest, CopyIntersectingPixelsUncontainedRects) {
+  const TestData tests[] = {
+    {"Left",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(-1, 0, 1, 2),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(0, 0, 0, 0)},
+
+    {"Top",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, -1, 2, 1),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(0, 0, 0, 0)},
+
+    {"Right",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(2, 0, 1, 2),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(0, 0, 0, 0)},
+
+
+    {"Bottom",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, 2, 2, 1),
+     1.0, 1.0,
+     DesktopRect::MakeXYWH(0, 0, 0, 0)}
+  };
+
+  RunTests(tests, arraysize(tests));
+}
+
+TEST(DesktopFrameTest, CopyIntersectingPixelsUncontainedRectsScaled) {
+  const TestData tests[] = {
+    {"Left 2x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(-1, 0, 2, 2),
+     2.0, 2.0,
+     DesktopRect::MakeXYWH(0, 0, 0, 0)},
+
+    {"Top 0.5x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, -2, 2, 1),
+     0.5, 0.5,
+     DesktopRect::MakeXYWH(0, 0, 0, 0)},
+
+    {"Right 2x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(1, 0, 1, 2),
+     2.0, 2.0,
+     DesktopRect::MakeXYWH(0, 0, 0, 0)},
+
+
+    {"Bottom 0.5x",
+     DesktopRect::MakeXYWH(0, 0, 2, 2),
+     DesktopRect::MakeXYWH(0, 4, 2, 1),
+     0.5, 0.5,
+     DesktopRect::MakeXYWH(0, 0, 0, 0)}
+  };
+
+  RunTests(tests, arraysize(tests));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame_win.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_win.cc
new file mode 100644
index 0000000000..262ebbdec0
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_win.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_frame_win.h"
+
+#include <utility>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+DesktopFrameWin::DesktopFrameWin(DesktopSize size,
+                                 int stride,
+                                 uint8_t* data,
+                                 std::unique_ptr<SharedMemory> shared_memory,
+                                 HBITMAP bitmap)
+    : DesktopFrame(size, stride, data, shared_memory.get()),
+      bitmap_(bitmap),
+      owned_shared_memory_(std::move(shared_memory)) {}
+
+DesktopFrameWin::~DesktopFrameWin() {
+  DeleteObject(bitmap_);
+}
+
+// static
+std::unique_ptr<DesktopFrameWin> DesktopFrameWin::Create(
+    DesktopSize size,
+    SharedMemoryFactory* shared_memory_factory,
+    HDC hdc) {
+  int bytes_per_row = size.width() * kBytesPerPixel;
+  int buffer_size = bytes_per_row * size.height();
+
+  // Describe a device independent bitmap (DIB) that is the size of the desktop.
+  BITMAPINFO bmi = {};
+  bmi.bmiHeader.biHeight = -size.height();
+  bmi.bmiHeader.biWidth = size.width();
+  bmi.bmiHeader.biPlanes = 1;
+  bmi.bmiHeader.biBitCount = DesktopFrameWin::kBytesPerPixel * 8;
+  bmi.bmiHeader.biSize = sizeof(bmi.bmiHeader);
+  bmi.bmiHeader.biSizeImage = bytes_per_row * size.height();
+
+  std::unique_ptr<SharedMemory> shared_memory;
+  HANDLE section_handle = nullptr;
+  if (shared_memory_factory) {
+    shared_memory = shared_memory_factory->CreateSharedMemory(buffer_size);
+    if (!shared_memory) {
+      RTC_LOG(LS_WARNING) << "Failed to allocate shared memory";
+      return nullptr;
+    }
+    section_handle = shared_memory->handle();
+  }
+  void* data = nullptr;
+  HBITMAP bitmap =
+      CreateDIBSection(hdc, &bmi, DIB_RGB_COLORS, &data, section_handle, 0);
+  if (!bitmap) {
+    RTC_LOG(LS_WARNING) << "Failed to allocate new window frame "
+                        << GetLastError();
+    return nullptr;
+  }
+
+  return std::unique_ptr<DesktopFrameWin>(
+      new DesktopFrameWin(size, bytes_per_row, reinterpret_cast<uint8_t*>(data),
+                          std::move(shared_memory), bitmap));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_frame_win.h b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_win.h
new file mode 100644
index 0000000000..f8faad6777
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_frame_win.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_WIN_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_WIN_H_
+
+#include <windows.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+
+namespace webrtc {
+
+// DesktopFrame implementation used by screen and window captures on Windows.
+// Frame data is stored in a GDI bitmap.
+class DesktopFrameWin : public DesktopFrame {
+ public:
+  ~DesktopFrameWin() override;
+
+  DesktopFrameWin(const DesktopFrameWin&) = delete;
+  DesktopFrameWin& operator=(const DesktopFrameWin&) = delete;
+
+  static std::unique_ptr<DesktopFrameWin>
+  Create(DesktopSize size, SharedMemoryFactory* shared_memory_factory, HDC hdc);
+
+  HBITMAP bitmap() { return bitmap_; }
+
+ private:
+  DesktopFrameWin(DesktopSize size,
+                  int stride,
+                  uint8_t* data,
+                  std::unique_ptr<SharedMemory> shared_memory,
+                  HBITMAP bitmap);
+
+  HBITMAP bitmap_;
+  std::unique_ptr<SharedMemory> owned_shared_memory_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_WIN_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_geometry.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_geometry.cc
new file mode 100644
index 0000000000..e0a5d7af83
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_geometry.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_geometry.h"
+
+#include <algorithm>
+#include <cmath>
+
+namespace webrtc {
+
+bool DesktopRect::Contains(const DesktopVector& point) const {
+  return point.x() >= left() && point.x() < right() && point.y() >= top() &&
+         point.y() < bottom();
+}
+
+bool DesktopRect::ContainsRect(const DesktopRect& rect) const {
+  return rect.left() >= left() && rect.right() <= right() &&
+         rect.top() >= top() && rect.bottom() <= bottom();
+}
+
+void DesktopRect::IntersectWith(const DesktopRect& rect) {
+  left_ = std::max(left(), rect.left());
+  top_ = std::max(top(), rect.top());
+  right_ = std::min(right(), rect.right());
+  bottom_ = std::min(bottom(), rect.bottom());
+  if (is_empty()) {
+    left_ = 0;
+    top_ = 0;
+    right_ = 0;
+    bottom_ = 0;
+  }
+}
+
+void DesktopRect::UnionWith(const DesktopRect& rect) {
+  if (is_empty()) {
+    *this = rect;
+    return;
+  }
+
+  if (rect.is_empty()) {
+    return;
+  }
+
+  left_ = std::min(left(), rect.left());
+  top_ = std::min(top(), rect.top());
+  right_ = std::max(right(), rect.right());
+  bottom_ = std::max(bottom(), rect.bottom());
+}
+
+void DesktopRect::Translate(int32_t dx, int32_t dy) {
+  left_ += dx;
+  top_ += dy;
+  right_ += dx;
+  bottom_ += dy;
+}
+
+void DesktopRect::Extend(int32_t left_offset,
+                         int32_t top_offset,
+                         int32_t right_offset,
+                         int32_t bottom_offset) {
+  left_ -= left_offset;
+  top_ -= top_offset;
+  right_ += right_offset;
+  bottom_ += bottom_offset;
+}
+
+void DesktopRect::Scale(double horizontal, double vertical) {
+  right_ += static_cast<int>(std::round(width() * (horizontal - 1)));
+  bottom_ += static_cast<int>(std::round(height() * (vertical - 1)));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_geometry.h b/third_party/libwebrtc/modules/desktop_capture/desktop_geometry.h
new file mode 100644
index 0000000000..691455df57
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_geometry.h
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_GEOMETRY_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_GEOMETRY_H_
+
+#include <stdint.h>
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// A vector in the 2D integer space. E.g. can be used to represent screen DPI.
+class DesktopVector {
+ public:
+  DesktopVector() : x_(0), y_(0) {}
+  DesktopVector(int32_t x, int32_t y) : x_(x), y_(y) {}
+
+  int32_t x() const { return x_; }
+  int32_t y() const { return y_; }
+  bool is_zero() const { return x_ == 0 && y_ == 0; }
+
+  bool equals(const DesktopVector& other) const {
+    return x_ == other.x_ && y_ == other.y_;
+  }
+
+  void set(int32_t x, int32_t y) {
+    x_ = x;
+    y_ = y;
+  }
+
+  DesktopVector add(const DesktopVector& other) const {
+    return DesktopVector(x() + other.x(), y() + other.y());
+  }
+  DesktopVector subtract(const DesktopVector& other) const {
+    return DesktopVector(x() - other.x(), y() - other.y());
+  }
+
+  DesktopVector operator-() const { return DesktopVector(-x_, -y_); }
+
+ private:
+  int32_t x_;
+  int32_t y_;
+};
+
+// Type used to represent screen/window size.
+class DesktopSize {
+ public:
+  DesktopSize() : width_(0), height_(0) {}
+  DesktopSize(int32_t width, int32_t height) : width_(width), height_(height) {}
+
+  int32_t width() const { return width_; }
+  int32_t height() const { return height_; }
+
+  bool is_empty() const { return width_ <= 0 || height_ <= 0; }
+
+  bool equals(const DesktopSize& other) const {
+    return width_ == other.width_ && height_ == other.height_;
+  }
+
+  void set(int32_t width, int32_t height) {
+    width_ = width;
+    height_ = height;
+  }
+
+ private:
+  int32_t width_;
+  int32_t height_;
+};
+
+// Represents a rectangle on the screen.
+class RTC_EXPORT DesktopRect {
+ public:
+  static DesktopRect MakeSize(const DesktopSize& size) {
+    return DesktopRect(0, 0, size.width(), size.height());
+  }
+  static DesktopRect MakeWH(int32_t width, int32_t height) {
+    return DesktopRect(0, 0, width, height);
+  }
+  static DesktopRect MakeXYWH(int32_t x,
+                              int32_t y,
+                              int32_t width,
+                              int32_t height) {
+    return DesktopRect(x, y, x + width, y + height);
+  }
+  static DesktopRect MakeLTRB(int32_t left,
+                              int32_t top,
+                              int32_t right,
+                              int32_t bottom) {
+    return DesktopRect(left, top, right, bottom);
+  }
+  static DesktopRect MakeOriginSize(const DesktopVector& origin,
+                                    const DesktopSize& size) {
+    return MakeXYWH(origin.x(), origin.y(), size.width(), size.height());
+  }
+
+  DesktopRect() : left_(0), top_(0), right_(0), bottom_(0) {}
+
+  int32_t left() const { return left_; }
+  int32_t top() const { return top_; }
+  int32_t right() const { return right_; }
+  int32_t bottom() const { return bottom_; }
+  int32_t width() const { return right_ - left_; }
+  int32_t height() const { return bottom_ - top_; }
+
+  void set_width(int32_t width) { right_ = left_ + width; }
+  void set_height(int32_t height) { bottom_ = top_ + height; }
+
+  DesktopVector top_left() const { return DesktopVector(left_, top_); }
+  DesktopSize size() const { return DesktopSize(width(), height()); }
+
+  bool is_empty() const { return left_ >= right_ || top_ >= bottom_; }
+
+  bool equals(const DesktopRect& other) const {
+    return left_ == other.left_ && top_ == other.top_ &&
+           right_ == other.right_ && bottom_ == other.bottom_;
+  }
+
+  // Returns true if `point` lies within the rectangle boundaries.
+  bool Contains(const DesktopVector& point) const;
+
+  // Returns true if `rect` lies within the boundaries of this rectangle.
+  bool ContainsRect(const DesktopRect& rect) const;
+
+  // Finds intersection with `rect`.
+  void IntersectWith(const DesktopRect& rect);
+
+  // Extends the rectangle to cover `rect`. If `this` is empty, replaces `this`
+  // with `rect`; if `rect` is empty, this function takes no effect.
+  void UnionWith(const DesktopRect& rect);
+
+  // Adds (dx, dy) to the position of the rectangle.
+  void Translate(int32_t dx, int32_t dy);
+  void Translate(DesktopVector d) { Translate(d.x(), d.y()); }
+
+  // Enlarges current DesktopRect by subtracting `left_offset` and `top_offset`
+  // from `left_` and `top_`, and adding `right_offset` and `bottom_offset` to
+  // `right_` and `bottom_`. This function does not normalize the result, so
+  // `left_` and `top_` may be less than zero or larger than `right_` and
+  // `bottom_`.
+  void Extend(int32_t left_offset,
+              int32_t top_offset,
+              int32_t right_offset,
+              int32_t bottom_offset);
+
+  // Scales current DesktopRect. This function does not impact the `top_` and
+  // `left_`.
+  void Scale(double horizontal, double vertical);
+
+ private:
+  DesktopRect(int32_t left, int32_t top, int32_t right, int32_t bottom)
+      : left_(left), top_(top), right_(right), bottom_(bottom) {}
+
+  int32_t left_;
+  int32_t top_;
+  int32_t right_;
+  int32_t bottom_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_GEOMETRY_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_geometry_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_geometry_unittest.cc
new file mode 100644
index 0000000000..f4a07fa46b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_geometry_unittest.cc
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_geometry.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(DesktopRectTest, UnionBetweenTwoNonEmptyRects) {
+  DesktopRect rect = DesktopRect::MakeLTRB(1, 1, 2, 2);
+  rect.UnionWith(DesktopRect::MakeLTRB(-2, -2, -1, -1));
+  ASSERT_TRUE(rect.equals(DesktopRect::MakeLTRB(-2, -2, 2, 2)));
+}
+
+TEST(DesktopRectTest, UnionWithEmptyRect) {
+  DesktopRect rect = DesktopRect::MakeWH(1, 1);
+  rect.UnionWith(DesktopRect());
+  ASSERT_TRUE(rect.equals(DesktopRect::MakeWH(1, 1)));
+
+  rect = DesktopRect::MakeXYWH(1, 1, 2, 2);
+  rect.UnionWith(DesktopRect());
+  ASSERT_TRUE(rect.equals(DesktopRect::MakeXYWH(1, 1, 2, 2)));
+
+  rect = DesktopRect::MakeXYWH(1, 1, 2, 2);
+  rect.UnionWith(DesktopRect::MakeXYWH(3, 3, 0, 0));
+  ASSERT_TRUE(rect.equals(DesktopRect::MakeXYWH(1, 1, 2, 2)));
+}
+
+TEST(DesktopRectTest, EmptyRectUnionWithNonEmptyOne) {
+  DesktopRect rect;
+  rect.UnionWith(DesktopRect::MakeWH(1, 1));
+  ASSERT_TRUE(rect.equals(DesktopRect::MakeWH(1, 1)));
+
+  rect = DesktopRect();
+  rect.UnionWith(DesktopRect::MakeXYWH(1, 1, 2, 2));
+  ASSERT_TRUE(rect.equals(DesktopRect::MakeXYWH(1, 1, 2, 2)));
+
+  rect = DesktopRect::MakeXYWH(3, 3, 0, 0);
+  rect.UnionWith(DesktopRect::MakeXYWH(1, 1, 2, 2));
+  ASSERT_TRUE(rect.equals(DesktopRect::MakeXYWH(1, 1, 2, 2)));
+}
+
+TEST(DesktopRectTest, EmptyRectUnionWithEmptyOne) {
+  DesktopRect rect;
+  rect.UnionWith(DesktopRect());
+  ASSERT_TRUE(rect.is_empty());
+
+  rect = DesktopRect::MakeXYWH(1, 1, 0, 0);
+  rect.UnionWith(DesktopRect());
+  ASSERT_TRUE(rect.is_empty());
+
+  rect = DesktopRect();
+  rect.UnionWith(DesktopRect::MakeXYWH(1, 1, 0, 0));
+  ASSERT_TRUE(rect.is_empty());
+
+  rect = DesktopRect::MakeXYWH(1, 1, 0, 0);
+  rect.UnionWith(DesktopRect::MakeXYWH(-1, -1, 0, 0));
+  ASSERT_TRUE(rect.is_empty());
+}
+
+TEST(DesktopRectTest, Scale) {
+  DesktopRect rect = DesktopRect::MakeXYWH(100, 100, 100, 100);
+  rect.Scale(1.1, 1.1);
+  ASSERT_EQ(rect.top(), 100);
+  ASSERT_EQ(rect.left(), 100);
+  ASSERT_EQ(rect.width(), 110);
+  ASSERT_EQ(rect.height(), 110);
+
+  rect = DesktopRect::MakeXYWH(100, 100, 100, 100);
+  rect.Scale(0.01, 0.01);
+  ASSERT_EQ(rect.top(), 100);
+  ASSERT_EQ(rect.left(), 100);
+  ASSERT_EQ(rect.width(), 1);
+  ASSERT_EQ(rect.height(), 1);
+
+  rect = DesktopRect::MakeXYWH(100, 100, 100, 100);
+  rect.Scale(1.1, 0.9);
+  ASSERT_EQ(rect.top(), 100);
+  ASSERT_EQ(rect.left(), 100);
+  ASSERT_EQ(rect.width(), 110);
+  ASSERT_EQ(rect.height(), 90);
+
+  rect = DesktopRect::MakeXYWH(0, 0, 100, 100);
+  rect.Scale(1.1, 1.1);
+  ASSERT_EQ(rect.top(), 0);
+  ASSERT_EQ(rect.left(), 0);
+  ASSERT_EQ(rect.width(), 110);
+  ASSERT_EQ(rect.height(), 110);
+
+  rect = DesktopRect::MakeXYWH(0, 100, 100, 100);
+  rect.Scale(1.1, 1.1);
+  ASSERT_EQ(rect.top(), 100);
+  ASSERT_EQ(rect.left(), 0);
+  ASSERT_EQ(rect.width(), 110);
+  ASSERT_EQ(rect.height(), 110);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_region.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_region.cc
new file mode 100644
index 0000000000..2c87c11eb3
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_region.cc
@@ -0,0 +1,567 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_region.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+DesktopRegion::RowSpan::RowSpan(int32_t left, int32_t right)
+    : left(left), right(right) {}
+
+DesktopRegion::Row::Row(const Row&) = default;
+DesktopRegion::Row::Row(Row&&) = default;
+
+DesktopRegion::Row::Row(int32_t top, int32_t bottom)
+    : top(top), bottom(bottom) {}
+
+DesktopRegion::Row::~Row() {}
+
+DesktopRegion::DesktopRegion() {}
+
+DesktopRegion::DesktopRegion(const DesktopRect& rect) {
+  AddRect(rect);
+}
+
+DesktopRegion::DesktopRegion(const DesktopRect* rects, int count) {
+  AddRects(rects, count);
+}
+
+DesktopRegion::DesktopRegion(const DesktopRegion& other) {
+  *this = other;
+}
+
+DesktopRegion::~DesktopRegion() {
+  Clear();
+}
+
+DesktopRegion& DesktopRegion::operator=(const DesktopRegion& other) {
+  Clear();
+  rows_ = other.rows_;
+  for (Rows::iterator it = rows_.begin(); it != rows_.end(); ++it) {
+    // Copy each row.
+    Row* row = it->second;
+    it->second = new Row(*row);
+  }
+  return *this;
+}
+
+bool DesktopRegion::Equals(const DesktopRegion& region) const {
+  // Iterate over rows of the tow regions and compare each row.
+  Rows::const_iterator it1 = rows_.begin();
+  Rows::const_iterator it2 = region.rows_.begin();
+  while (it1 != rows_.end()) {
+    if (it2 == region.rows_.end() || it1->first != it2->first ||
+        it1->second->top != it2->second->top ||
+        it1->second->bottom != it2->second->bottom ||
+        it1->second->spans != it2->second->spans) {
+      return false;
+    }
+    ++it1;
+    ++it2;
+  }
+  return it2 == region.rows_.end();
+}
+
+void DesktopRegion::Clear() {
+  for (Rows::iterator row = rows_.begin(); row != rows_.end(); ++row) {
+    delete row->second;
+  }
+  rows_.clear();
+}
+
+void DesktopRegion::SetRect(const DesktopRect& rect) {
+  Clear();
+  AddRect(rect);
+}
+
+void DesktopRegion::AddRect(const DesktopRect& rect) {
+  if (rect.is_empty())
+    return;
+
+  // Top of the part of the `rect` that hasn't been inserted yet. Increased as
+  // we iterate over the rows until it reaches `rect.bottom()`.
+  int top = rect.top();
+
+  // Iterate over all rows that may intersect with `rect` and add new rows when
+  // necessary.
+  Rows::iterator row = rows_.upper_bound(top);
+  while (top < rect.bottom()) {
+    if (row == rows_.end() || top < row->second->top) {
+      // If `top` is above the top of the current `row` then add a new row above
+      // the current one.
+      int32_t bottom = rect.bottom();
+      if (row != rows_.end() && row->second->top < bottom)
+        bottom = row->second->top;
+      row = rows_.insert(row, Rows::value_type(bottom, new Row(top, bottom)));
+    } else if (top > row->second->top) {
+      // If the `top` falls in the middle of the `row` then split `row` into
+      // two, at `top`, and leave `row` referring to the lower of the two,
+      // ready to insert a new span into.
+      RTC_DCHECK_LE(top, row->second->bottom);
+      Rows::iterator new_row = rows_.insert(
+          row, Rows::value_type(top, new Row(row->second->top, top)));
+      row->second->top = top;
+      new_row->second->spans = row->second->spans;
+    }
+
+    if (rect.bottom() < row->second->bottom) {
+      // If the bottom of the `rect` falls in the middle of the `row` split
+      // `row` into two, at `top`, and leave `row` referring to the upper of
+      // the two, ready to insert a new span into.
+      Rows::iterator new_row = rows_.insert(
+          row, Rows::value_type(rect.bottom(), new Row(top, rect.bottom())));
+      row->second->top = rect.bottom();
+      new_row->second->spans = row->second->spans;
+      row = new_row;
+    }
+
+    // Add a new span to the current row.
+    AddSpanToRow(row->second, rect.left(), rect.right());
+    top = row->second->bottom;
+
+    MergeWithPrecedingRow(row);
+
+    // Move to the next row.
+    ++row;
+  }
+
+  if (row != rows_.end())
+    MergeWithPrecedingRow(row);
+}
+
+void DesktopRegion::AddRects(const DesktopRect* rects, int count) {
+  for (int i = 0; i < count; ++i) {
+    AddRect(rects[i]);
+  }
+}
+
+void DesktopRegion::MergeWithPrecedingRow(Rows::iterator row) {
+  RTC_DCHECK(row != rows_.end());
+
+  if (row != rows_.begin()) {
+    Rows::iterator previous_row = row;
+    previous_row--;
+
+    // If `row` and `previous_row` are next to each other and contain the same
+    // set of spans then they can be merged.
+    if (previous_row->second->bottom == row->second->top &&
+        previous_row->second->spans == row->second->spans) {
+      row->second->top = previous_row->second->top;
+      delete previous_row->second;
+      rows_.erase(previous_row);
+    }
+  }
+}
+
+void DesktopRegion::AddRegion(const DesktopRegion& region) {
+  // TODO(sergeyu): This function is not optimized - potentially it can iterate
+  // over rows of the two regions similar to how it works in Intersect().
+  for (Iterator it(region); !it.IsAtEnd(); it.Advance()) {
+    AddRect(it.rect());
+  }
+}
+
+void DesktopRegion::Intersect(const DesktopRegion& region1,
+                              const DesktopRegion& region2) {
+  Clear();
+
+  Rows::const_iterator it1 = region1.rows_.begin();
+  Rows::const_iterator end1 = region1.rows_.end();
+  Rows::const_iterator it2 = region2.rows_.begin();
+  Rows::const_iterator end2 = region2.rows_.end();
+  if (it1 == end1 || it2 == end2)
+    return;
+
+  while (it1 != end1 && it2 != end2) {
+    // Arrange for `it1` to always be the top-most of the rows.
+    if (it2->second->top < it1->second->top) {
+      std::swap(it1, it2);
+      std::swap(end1, end2);
+    }
+
+    // Skip `it1` if it doesn't intersect `it2` at all.
+    if (it1->second->bottom <= it2->second->top) {
+      ++it1;
+      continue;
+    }
+
+    // Top of the `it1` row is above the top of `it2`, so top of the
+    // intersection is always the top of `it2`.
+    int32_t top = it2->second->top;
+    int32_t bottom = std::min(it1->second->bottom, it2->second->bottom);
+
+    Rows::iterator new_row = rows_.insert(
+        rows_.end(), Rows::value_type(bottom, new Row(top, bottom)));
+    IntersectRows(it1->second->spans, it2->second->spans,
+                  &new_row->second->spans);
+    if (new_row->second->spans.empty()) {
+      delete new_row->second;
+      rows_.erase(new_row);
+    } else {
+      MergeWithPrecedingRow(new_row);
+    }
+
+    // If `it1` was completely consumed, move to the next one.
+    if (it1->second->bottom == bottom)
+      ++it1;
+    // If `it2` was completely consumed, move to the next one.
+    if (it2->second->bottom == bottom)
+      ++it2;
+  }
+}
+
+// static
+void DesktopRegion::IntersectRows(const RowSpanSet& set1,
+                                  const RowSpanSet& set2,
+                                  RowSpanSet* output) {
+  RowSpanSet::const_iterator it1 = set1.begin();
+  RowSpanSet::const_iterator end1 = set1.end();
+  RowSpanSet::const_iterator it2 = set2.begin();
+  RowSpanSet::const_iterator end2 = set2.end();
+  RTC_DCHECK(it1 != end1 && it2 != end2);
+
+  do {
+    // Arrange for `it1` to always be the left-most of the spans.
+    if (it2->left < it1->left) {
+      std::swap(it1, it2);
+      std::swap(end1, end2);
+    }
+
+    // Skip `it1` if it doesn't intersect `it2` at all.
+    if (it1->right <= it2->left) {
+      ++it1;
+      continue;
+    }
+
+    int32_t left = it2->left;
+    int32_t right = std::min(it1->right, it2->right);
+    RTC_DCHECK_LT(left, right);
+
+    output->push_back(RowSpan(left, right));
+
+    // If `it1` was completely consumed, move to the next one.
+    if (it1->right == right)
+      ++it1;
+    // If `it2` was completely consumed, move to the next one.
+    if (it2->right == right)
+      ++it2;
+  } while (it1 != end1 && it2 != end2);
+}
+
+void DesktopRegion::IntersectWith(const DesktopRegion& region) {
+  DesktopRegion old_region;
+  Swap(&old_region);
+  Intersect(old_region, region);
+}
+
+void DesktopRegion::IntersectWith(const DesktopRect& rect) {
+  DesktopRegion region;
+  region.AddRect(rect);
+  IntersectWith(region);
+}
+
+void DesktopRegion::Subtract(const DesktopRegion& region) {
+  if (region.rows_.empty())
+    return;
+
+  // `row_b` refers to the current row being subtracted.
+  Rows::const_iterator row_b = region.rows_.begin();
+
+  // Current vertical position at which subtraction is happening.
+  int top = row_b->second->top;
+
+  // `row_a` refers to the current row we are subtracting from. Skip all rows
+  // above `top`.
+  Rows::iterator row_a = rows_.upper_bound(top);
+
+  // Step through rows of the both regions subtracting content of `row_b` from
+  // `row_a`.
+  while (row_a != rows_.end() && row_b != region.rows_.end()) {
+    // Skip `row_a` if it doesn't intersect with the `row_b`.
+    if (row_a->second->bottom <= top) {
+      // Each output row is merged with previously-processed rows before further
+      // rows are processed.
+      MergeWithPrecedingRow(row_a);
+      ++row_a;
+      continue;
+    }
+
+    if (top > row_a->second->top) {
+      // If `top` falls in the middle of `row_a` then split `row_a` into two, at
+      // `top`, and leave `row_a` referring to the lower of the two, ready to
+      // subtract spans from.
+      RTC_DCHECK_LE(top, row_a->second->bottom);
+      Rows::iterator new_row = rows_.insert(
+          row_a, Rows::value_type(top, new Row(row_a->second->top, top)));
+      row_a->second->top = top;
+      new_row->second->spans = row_a->second->spans;
+    } else if (top < row_a->second->top) {
+      // If the `top` is above `row_a` then skip the range between `top` and
+      // top of `row_a` because it's empty.
+      top = row_a->second->top;
+      if (top >= row_b->second->bottom) {
+        ++row_b;
+        if (row_b != region.rows_.end())
+          top = row_b->second->top;
+        continue;
+      }
+    }
+
+    if (row_b->second->bottom < row_a->second->bottom) {
+      // If the bottom of `row_b` falls in the middle of the `row_a` split
+      // `row_a` into two, at `top`, and leave `row_a` referring to the upper of
+      // the two, ready to subtract spans from.
+      int bottom = row_b->second->bottom;
+      Rows::iterator new_row =
+          rows_.insert(row_a, Rows::value_type(bottom, new Row(top, bottom)));
+      row_a->second->top = bottom;
+      new_row->second->spans = row_a->second->spans;
+      row_a = new_row;
+    }
+
+    // At this point the vertical range covered by `row_a` lays within the
+    // range covered by `row_b`. Subtract `row_b` spans from `row_a`.
+    RowSpanSet new_spans;
+    SubtractRows(row_a->second->spans, row_b->second->spans, &new_spans);
+    new_spans.swap(row_a->second->spans);
+    top = row_a->second->bottom;
+
+    if (top >= row_b->second->bottom) {
+      ++row_b;
+      if (row_b != region.rows_.end())
+        top = row_b->second->top;
+    }
+
+    // Check if the row is empty after subtraction and delete it. Otherwise move
+    // to the next one.
+    if (row_a->second->spans.empty()) {
+      Rows::iterator row_to_delete = row_a;
+      ++row_a;
+      delete row_to_delete->second;
+      rows_.erase(row_to_delete);
+    } else {
+      MergeWithPrecedingRow(row_a);
+      ++row_a;
+    }
+  }
+
+  if (row_a != rows_.end())
+    MergeWithPrecedingRow(row_a);
+}
+
+void DesktopRegion::Subtract(const DesktopRect& rect) {
+  DesktopRegion region;
+  region.AddRect(rect);
+  Subtract(region);
+}
+
+void DesktopRegion::Translate(int32_t dx, int32_t dy) {
+  Rows new_rows;
+
+  for (Rows::iterator it = rows_.begin(); it != rows_.end(); ++it) {
+    Row* row = it->second;
+
+    row->top += dy;
+    row->bottom += dy;
+
+    if (dx != 0) {
+      // Translate each span.
+      for (RowSpanSet::iterator span = row->spans.begin();
+           span != row->spans.end(); ++span) {
+        span->left += dx;
+        span->right += dx;
+      }
+    }
+
+    if (dy != 0)
+      new_rows.insert(new_rows.end(), Rows::value_type(row->bottom, row));
+  }
+
+  if (dy != 0)
+    new_rows.swap(rows_);
+}
+
+void DesktopRegion::Swap(DesktopRegion* region) {
+  rows_.swap(region->rows_);
+}
+
+// static
+bool DesktopRegion::CompareSpanRight(const RowSpan& r, int32_t value) {
+  return r.right < value;
+}
+
+// static
+bool DesktopRegion::CompareSpanLeft(const RowSpan& r, int32_t value) {
+  return r.left < value;
+}
+
+// static
+void DesktopRegion::AddSpanToRow(Row* row, int left, int right) {
+  // First check if the new span is located to the right of all existing spans.
+  // This is an optimization to avoid binary search in the case when rectangles
+  // are inserted sequentially from left to right.
+  if (row->spans.empty() || left > row->spans.back().right) {
+    row->spans.push_back(RowSpan(left, right));
+    return;
+  }
+
+  // Find the first span that ends at or after `left`.
+  RowSpanSet::iterator start = std::lower_bound(
+      row->spans.begin(), row->spans.end(), left, CompareSpanRight);
+  RTC_DCHECK(start < row->spans.end());
+
+  // Find the first span that starts after `right`.
+  RowSpanSet::iterator end =
+      std::lower_bound(start, row->spans.end(), right + 1, CompareSpanLeft);
+  if (end == row->spans.begin()) {
+    // There are no overlaps. Just insert the new span at the beginning.
+    row->spans.insert(row->spans.begin(), RowSpan(left, right));
+    return;
+  }
+
+  // Move end to the left, so that it points the last span that ends at or
+  // before `right`.
+  end--;
+
+  // At this point [start, end] is the range of spans that intersect with the
+  // new one.
+  if (end < start) {
+    // There are no overlaps. Just insert the new span at the correct position.
+    row->spans.insert(start, RowSpan(left, right));
+    return;
+  }
+
+  left = std::min(left, start->left);
+  right = std::max(right, end->right);
+
+  // Replace range [start, end] with the new span.
+  *start = RowSpan(left, right);
+  ++start;
+  ++end;
+  if (start < end)
+    row->spans.erase(start, end);
+}
+
+// static
+bool DesktopRegion::IsSpanInRow(const Row& row, const RowSpan& span) {
+  // Find the first span that starts at or after `span.left` and then check if
+  // it's the same span.
+  RowSpanSet::const_iterator it = std::lower_bound(
+      row.spans.begin(), row.spans.end(), span.left, CompareSpanLeft);
+  return it != row.spans.end() && *it == span;
+}
+
+// static
+void DesktopRegion::SubtractRows(const RowSpanSet& set_a,
+                                 const RowSpanSet& set_b,
+                                 RowSpanSet* output) {
+  RTC_DCHECK(!set_a.empty() && !set_b.empty());
+
+  RowSpanSet::const_iterator it_b = set_b.begin();
+
+  // Iterate over all spans in `set_a` adding parts of it that do not intersect
+  // with `set_b` to the `output`.
+  for (RowSpanSet::const_iterator it_a = set_a.begin(); it_a != set_a.end();
+       ++it_a) {
+    // If there is no intersection then append the current span and continue.
+    if (it_b == set_b.end() || it_a->right < it_b->left) {
+      output->push_back(*it_a);
+      continue;
+    }
+
+    // Iterate over `set_b` spans that may intersect with `it_a`.
+    int pos = it_a->left;
+    while (it_b != set_b.end() && it_b->left < it_a->right) {
+      if (it_b->left > pos)
+        output->push_back(RowSpan(pos, it_b->left));
+      if (it_b->right > pos) {
+        pos = it_b->right;
+        if (pos >= it_a->right)
+          break;
+      }
+      ++it_b;
+    }
+    if (pos < it_a->right)
+      output->push_back(RowSpan(pos, it_a->right));
+  }
+}
+
+DesktopRegion::Iterator::Iterator(const DesktopRegion& region)
+    : region_(region),
+      row_(region.rows_.begin()),
+      previous_row_(region.rows_.end()) {
+  if (!IsAtEnd()) {
+    RTC_DCHECK_GT(row_->second->spans.size(), 0);
+    row_span_ = row_->second->spans.begin();
+    UpdateCurrentRect();
+  }
+}
+
+DesktopRegion::Iterator::~Iterator() {}
+
+bool DesktopRegion::Iterator::IsAtEnd() const {
+  return row_ == region_.rows_.end();
+}
+
+void DesktopRegion::Iterator::Advance() {
+  RTC_DCHECK(!IsAtEnd());
+
+  while (true) {
+    ++row_span_;
+    if (row_span_ == row_->second->spans.end()) {
+      previous_row_ = row_;
+      ++row_;
+      if (row_ != region_.rows_.end()) {
+        RTC_DCHECK_GT(row_->second->spans.size(), 0);
+        row_span_ = row_->second->spans.begin();
+      }
+    }
+
+    if (IsAtEnd())
+      return;
+
+    // If the same span exists on the previous row then skip it, as we've
+    // already returned this span merged into the previous one, via
+    // UpdateCurrentRect().
+    if (previous_row_ != region_.rows_.end() &&
+        previous_row_->second->bottom == row_->second->top &&
+        IsSpanInRow(*previous_row_->second, *row_span_)) {
+      continue;
+    }
+
+    break;
+  }
+
+  RTC_DCHECK(!IsAtEnd());
+  UpdateCurrentRect();
+}
+
+void DesktopRegion::Iterator::UpdateCurrentRect() {
+  // Merge the current rectangle with the matching spans from later rows.
+  int bottom;
+  Rows::const_iterator bottom_row = row_;
+  Rows::const_iterator previous;
+  do {
+    bottom = bottom_row->second->bottom;
+    previous = bottom_row;
+    ++bottom_row;
+  } while (bottom_row != region_.rows_.end() &&
+           previous->second->bottom == bottom_row->second->top &&
+           IsSpanInRow(*bottom_row->second, *row_span_));
+  rect_ = DesktopRect::MakeLTRB(row_span_->left, row_->second->top,
+                                row_span_->right, bottom);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_region.h b/third_party/libwebrtc/modules/desktop_capture/desktop_region.h
new file mode 100644
index 0000000000..ae9d8a0ba9
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_region.h
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_REGION_H_
+#define MODULES_DESKTOP_CAPTURE_DESKTOP_REGION_H_
+
+#include <stdint.h>
+
+#include <map>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// DesktopRegion represents a region of the screen or window.
+//
+// Internally each region is stored as a set of rows where each row contains one
+// or more rectangles aligned vertically.
+class RTC_EXPORT DesktopRegion {
+ private:
+  // The following private types need to be declared first because they are used
+  // in the public Iterator.
+
+  // RowSpan represents a horizontal span withing a single row.
+  struct RowSpan {
+    RowSpan(int32_t left, int32_t right);
+
+    // Used by std::vector<>.
+    bool operator==(const RowSpan& that) const {
+      return left == that.left && right == that.right;
+    }
+
+    int32_t left;
+    int32_t right;
+  };
+
+  typedef std::vector<RowSpan> RowSpanSet;
+
+  // Row represents a single row of a region. A row is set of rectangles that
+  // have the same vertical position.
+  struct Row {
+    Row(const Row&);
+    Row(Row&&);
+    Row(int32_t top, int32_t bottom);
+    ~Row();
+
+    int32_t top;
+    int32_t bottom;
+
+    RowSpanSet spans;
+  };
+
+  // Type used to store list of rows in the region. The bottom position of row
+  // is used as the key so that rows are always ordered by their position. The
+  // map stores pointers to make Translate() more efficient.
+  typedef std::map<int, Row*> Rows;
+
+ public:
+  // Iterator that can be used to iterate over rectangles of a DesktopRegion.
+  // The region must not be mutated while the iterator is used.
+  class RTC_EXPORT Iterator {
+   public:
+    explicit Iterator(const DesktopRegion& target);
+    ~Iterator();
+
+    bool IsAtEnd() const;
+    void Advance();
+
+    const DesktopRect& rect() const { return rect_; }
+
+   private:
+    const DesktopRegion& region_;
+
+    // Updates `rect_` based on the current `row_` and `row_span_`. If
+    // `row_span_` matches spans on consecutive rows then they are also merged
+    // into `rect_`, to generate more efficient output.
+    void UpdateCurrentRect();
+
+    Rows::const_iterator row_;
+    Rows::const_iterator previous_row_;
+    RowSpanSet::const_iterator row_span_;
+    DesktopRect rect_;
+  };
+
+  DesktopRegion();
+  explicit DesktopRegion(const DesktopRect& rect);
+  DesktopRegion(const DesktopRect* rects, int count);
+  DesktopRegion(const DesktopRegion& other);
+  ~DesktopRegion();
+
+  DesktopRegion& operator=(const DesktopRegion& other);
+
+  bool is_empty() const { return rows_.empty(); }
+
+  bool Equals(const DesktopRegion& region) const;
+
+  // Reset the region to be empty.
+  void Clear();
+
+  // Reset region to contain just `rect`.
+  void SetRect(const DesktopRect& rect);
+
+  // Adds specified rect(s) or region to the region.
+  void AddRect(const DesktopRect& rect);
+  void AddRects(const DesktopRect* rects, int count);
+  void AddRegion(const DesktopRegion& region);
+
+  // Finds intersection of two regions and stores them in the current region.
+  void Intersect(const DesktopRegion& region1, const DesktopRegion& region2);
+
+  // Same as above but intersects content of the current region with `region`.
+  void IntersectWith(const DesktopRegion& region);
+
+  // Clips the region by the `rect`.
+  void IntersectWith(const DesktopRect& rect);
+
+  // Subtracts `region` from the current content of the region.
+  void Subtract(const DesktopRegion& region);
+
+  // Subtracts `rect` from the current content of the region.
+  void Subtract(const DesktopRect& rect);
+
+  // Adds (dx, dy) to the position of the region.
+  void Translate(int32_t dx, int32_t dy);
+
+  void Swap(DesktopRegion* region);
+
+ private:
+  // Comparison functions used for std::lower_bound(). Compare left or right
+  // edges withs a given `value`.
+  static bool CompareSpanLeft(const RowSpan& r, int32_t value);
+  static bool CompareSpanRight(const RowSpan& r, int32_t value);
+
+  // Adds a new span to the row, coalescing spans if necessary.
+  static void AddSpanToRow(Row* row, int32_t left, int32_t right);
+
+  // Returns true if the `span` exists in the given `row`.
+  static bool IsSpanInRow(const Row& row, const RowSpan& rect);
+
+  // Calculates the intersection of two sets of spans.
+  static void IntersectRows(const RowSpanSet& set1,
+                            const RowSpanSet& set2,
+                            RowSpanSet* output);
+
+  static void SubtractRows(const RowSpanSet& set_a,
+                           const RowSpanSet& set_b,
+                           RowSpanSet* output);
+
+  // Merges `row` with the row above it if they contain the same spans. Doesn't
+  // do anything if called with `row` set to rows_.begin() (i.e. first row of
+  // the region). If the rows were merged `row` remains a valid iterator to the
+  // merged row.
+  void MergeWithPrecedingRow(Rows::iterator row);
+
+  Rows rows_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_REGION_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/desktop_region_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/desktop_region_unittest.cc
new file mode 100644
index 0000000000..b8bd78e990
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/desktop_region_unittest.cc
@@ -0,0 +1,834 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_region.h"
+
+#include <stdlib.h>
+
+#include <algorithm>
+#include <cstdint>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+int RadmonInt(int max) {
+  return (rand() / 256) % max;
+}
+
+void CompareRegion(const DesktopRegion& region,
+                   const DesktopRect rects[],
+                   int rects_size) {
+  DesktopRegion::Iterator it(region);
+  for (int i = 0; i < rects_size; ++i) {
+    SCOPED_TRACE(i);
+    ASSERT_FALSE(it.IsAtEnd());
+    EXPECT_TRUE(it.rect().equals(rects[i]))
+        << it.rect().left() << "-" << it.rect().right() << "."
+        << it.rect().top() << "-" << it.rect().bottom() << " "
+        << rects[i].left() << "-" << rects[i].right() << "." << rects[i].top()
+        << "-" << rects[i].bottom();
+    it.Advance();
+  }
+  EXPECT_TRUE(it.IsAtEnd());
+}
+
+}  // namespace
+
+// Verify that regions are empty when created.
+TEST(DesktopRegionTest, Empty) {
+  DesktopRegion r;
+  CompareRegion(r, NULL, 0);
+}
+
+// Verify that empty rectangles are ignored.
+TEST(DesktopRegionTest, AddEmpty) {
+  DesktopRegion r;
+  DesktopRect rect = DesktopRect::MakeXYWH(1, 2, 0, 0);
+  r.AddRect(rect);
+  CompareRegion(r, NULL, 0);
+}
+
+// Verify that regions with a single rectangles are handled properly.
+TEST(DesktopRegionTest, SingleRect) {
+  DesktopRegion r;
+  DesktopRect rect = DesktopRect::MakeXYWH(1, 2, 3, 4);
+  r.AddRect(rect);
+  CompareRegion(r, &rect, 1);
+}
+
+// Verify that non-overlapping rectangles are not merged.
+TEST(DesktopRegionTest, NonOverlappingRects) {
+  struct Case {
+    int count;
+    DesktopRect rects[4];
+  } cases[] = {
+      {1, {DesktopRect::MakeXYWH(10, 10, 10, 10)}},
+      {2,
+       {DesktopRect::MakeXYWH(10, 10, 10, 10),
+        DesktopRect::MakeXYWH(30, 10, 10, 15)}},
+      {2,
+       {DesktopRect::MakeXYWH(10, 10, 10, 10),
+        DesktopRect::MakeXYWH(10, 30, 10, 5)}},
+      {3,
+       {DesktopRect::MakeXYWH(10, 10, 10, 9),
+        DesktopRect::MakeXYWH(30, 10, 15, 10),
+        DesktopRect::MakeXYWH(10, 30, 8, 10)}},
+      {4,
+       {DesktopRect::MakeXYWH(0, 0, 30, 10),
+        DesktopRect::MakeXYWH(40, 0, 10, 30),
+        DesktopRect::MakeXYWH(0, 20, 10, 30),
+        DesktopRect::MakeXYWH(20, 40, 30, 10)}},
+      {4,
+       {DesktopRect::MakeXYWH(0, 0, 10, 100),
+        DesktopRect::MakeXYWH(20, 10, 30, 10),
+        DesktopRect::MakeXYWH(20, 30, 30, 10),
+        DesktopRect::MakeXYWH(20, 50, 30, 10)}},
+  };
+
+  for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
+    SCOPED_TRACE(i);
+
+    DesktopRegion r;
+
+    for (int j = 0; j < cases[i].count; ++j) {
+      r.AddRect(cases[i].rects[j]);
+    }
+    CompareRegion(r, cases[i].rects, cases[i].count);
+
+    SCOPED_TRACE("Reverse");
+
+    // Try inserting rects in reverse order.
+    r.Clear();
+    for (int j = cases[i].count - 1; j >= 0; --j) {
+      r.AddRect(cases[i].rects[j]);
+    }
+    CompareRegion(r, cases[i].rects, cases[i].count);
+  }
+}
+
+TEST(DesktopRegionTest, TwoRects) {
+  struct Case {
+    DesktopRect input_rect1;
+    DesktopRect input_rect2;
+    int expected_count;
+    DesktopRect expected_rects[3];
+  } cases[] = {
+      // Touching rectangles that merge into one.
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(0, 100, 100, 200),
+       1,
+       {DesktopRect::MakeLTRB(0, 100, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(100, 0, 200, 100),
+       1,
+       {DesktopRect::MakeLTRB(100, 0, 200, 200)}},
+
+      // Rectangles touching on the vertical edge.
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(0, 150, 100, 250),
+       3,
+       {DesktopRect::MakeLTRB(100, 100, 200, 150),
+        DesktopRect::MakeLTRB(0, 150, 200, 200),
+        DesktopRect::MakeLTRB(0, 200, 100, 250)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(0, 50, 100, 150),
+       3,
+       {DesktopRect::MakeLTRB(0, 50, 100, 100),
+        DesktopRect::MakeLTRB(0, 100, 200, 150),
+        DesktopRect::MakeLTRB(100, 150, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(0, 120, 100, 180),
+       3,
+       {DesktopRect::MakeLTRB(100, 100, 200, 120),
+        DesktopRect::MakeLTRB(0, 120, 200, 180),
+        DesktopRect::MakeLTRB(100, 180, 200, 200)}},
+
+      // Rectangles touching on the horizontal edge.
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(150, 0, 250, 100),
+       2,
+       {DesktopRect::MakeLTRB(150, 0, 250, 100),
+        DesktopRect::MakeLTRB(100, 100, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(50, 0, 150, 100),
+       2,
+       {DesktopRect::MakeLTRB(50, 0, 150, 100),
+        DesktopRect::MakeLTRB(100, 100, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(120, 0, 180, 100),
+       2,
+       {DesktopRect::MakeLTRB(120, 0, 180, 100),
+        DesktopRect::MakeLTRB(100, 100, 200, 200)}},
+
+      // Overlapping rectangles.
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(50, 50, 150, 150),
+       3,
+       {DesktopRect::MakeLTRB(50, 50, 150, 100),
+        DesktopRect::MakeLTRB(50, 100, 200, 150),
+        DesktopRect::MakeLTRB(100, 150, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(150, 50, 250, 150),
+       3,
+       {DesktopRect::MakeLTRB(150, 50, 250, 100),
+        DesktopRect::MakeLTRB(100, 100, 250, 150),
+        DesktopRect::MakeLTRB(100, 150, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(0, 120, 150, 180),
+       3,
+       {DesktopRect::MakeLTRB(100, 100, 200, 120),
+        DesktopRect::MakeLTRB(0, 120, 200, 180),
+        DesktopRect::MakeLTRB(100, 180, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(120, 0, 180, 150),
+       2,
+       {DesktopRect::MakeLTRB(120, 0, 180, 100),
+        DesktopRect::MakeLTRB(100, 100, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 0, 200, 300),
+       DesktopRect::MakeLTRB(0, 100, 300, 200),
+       3,
+       {DesktopRect::MakeLTRB(100, 0, 200, 100),
+        DesktopRect::MakeLTRB(0, 100, 300, 200),
+        DesktopRect::MakeLTRB(100, 200, 200, 300)}},
+
+      // One rectangle enclosing another.
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(150, 150, 180, 180),
+       1,
+       {DesktopRect::MakeLTRB(100, 100, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(100, 100, 180, 180),
+       1,
+       {DesktopRect::MakeLTRB(100, 100, 200, 200)}},
+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
+       DesktopRect::MakeLTRB(150, 150, 200, 200),
+       1,
+       {DesktopRect::MakeLTRB(100, 100, 200, 200)}},
+  };
+
+  for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
+    SCOPED_TRACE(i);
+
+    DesktopRegion r;
+
+    r.AddRect(cases[i].input_rect1);
+    r.AddRect(cases[i].input_rect2);
+    CompareRegion(r, cases[i].expected_rects, cases[i].expected_count);
+
+    SCOPED_TRACE("Reverse");
+
+    // Run the same test with rectangles inserted in reverse order.
+    r.Clear();
+    r.AddRect(cases[i].input_rect2);
+    r.AddRect(cases[i].input_rect1);
+    CompareRegion(r, cases[i].expected_rects, cases[i].expected_count);
+  }
+}
+
+// Verify that DesktopRegion::AddRectToRow() works correctly by creating a row
+// of not overlapping rectangles and insert an overlapping rectangle into the
+// row at different positions. Result is verified by building a map of the
+// region in an array and comparing it with the expected values.
+TEST(DesktopRegionTest, SameRow) {
+  const int kMapWidth = 50;
+  const int kLastRectSizes[] = {3, 27};
+
+  DesktopRegion base_region;
+  bool base_map[kMapWidth] = {
+      false,
+  };
+
+  base_region.AddRect(DesktopRect::MakeXYWH(5, 0, 5, 1));
+  std::fill_n(base_map + 5, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(15, 0, 5, 1));
+  std::fill_n(base_map + 15, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(25, 0, 5, 1));
+  std::fill_n(base_map + 25, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(35, 0, 5, 1));
+  std::fill_n(base_map + 35, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(45, 0, 5, 1));
+  std::fill_n(base_map + 45, 5, true);
+
+  for (size_t i = 0; i < sizeof(kLastRectSizes) / sizeof(kLastRectSizes[0]);
+       i++) {
+    int last_rect_size = kLastRectSizes[i];
+    for (int x = 0; x < kMapWidth - last_rect_size; x++) {
+      SCOPED_TRACE(x);
+
+      DesktopRegion r = base_region;
+      r.AddRect(DesktopRect::MakeXYWH(x, 0, last_rect_size, 1));
+
+      bool expected_map[kMapWidth];
+      std::copy(base_map, base_map + kMapWidth, expected_map);
+      std::fill_n(expected_map + x, last_rect_size, true);
+
+      bool map[kMapWidth] = {
+          false,
+      };
+
+      int pos = -1;
+      for (DesktopRegion::Iterator it(r); !it.IsAtEnd(); it.Advance()) {
+        EXPECT_GT(it.rect().left(), pos);
+        pos = it.rect().right();
+        std::fill_n(map + it.rect().left(), it.rect().width(), true);
+      }
+
+      EXPECT_TRUE(std::equal(map, map + kMapWidth, expected_map));
+    }
+  }
+}
+
+TEST(DesktopRegionTest, ComplexRegions) {
+  struct Case {
+    int input_count;
+    DesktopRect input_rects[4];
+    int expected_count;
+    DesktopRect expected_rects[6];
+  } cases[] = {
+      {3,
+       {
+           DesktopRect::MakeLTRB(100, 100, 200, 200),
+           DesktopRect::MakeLTRB(0, 100, 100, 200),
+           DesktopRect::MakeLTRB(310, 110, 320, 120),
+       },
+       2,
+       {DesktopRect::MakeLTRB(0, 100, 200, 200),
+        DesktopRect::MakeLTRB(310, 110, 320, 120)}},
+      {3,
+       {DesktopRect::MakeLTRB(100, 100, 200, 200),
+        DesktopRect::MakeLTRB(50, 50, 150, 150),
+        DesktopRect::MakeLTRB(300, 125, 350, 175)},
+       4,
+       {DesktopRect::MakeLTRB(50, 50, 150, 100),
+        DesktopRect::MakeLTRB(50, 100, 200, 150),
+        DesktopRect::MakeLTRB(300, 125, 350, 175),
+        DesktopRect::MakeLTRB(100, 150, 200, 200)}},
+      {4,
+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
+        DesktopRect::MakeLTRB(10, 10, 40, 40),
+        DesktopRect::MakeLTRB(20, 20, 50, 50),
+        DesktopRect::MakeLTRB(50, 0, 65, 15)},
+       6,
+       {DesktopRect::MakeLTRB(0, 0, 30, 10),
+        DesktopRect::MakeLTRB(50, 0, 65, 15),
+        DesktopRect::MakeLTRB(0, 10, 40, 20),
+        DesktopRect::MakeLTRB(0, 20, 50, 30),
+        DesktopRect::MakeLTRB(10, 30, 50, 40),
+        DesktopRect::MakeLTRB(20, 40, 50, 50)}},
+      {3,
+       {DesktopRect::MakeLTRB(10, 10, 40, 20),
+        DesktopRect::MakeLTRB(10, 30, 40, 40),
+        DesktopRect::MakeLTRB(10, 20, 40, 30)},
+       1,
+       {DesktopRect::MakeLTRB(10, 10, 40, 40)}},
+  };
+
+  for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
+    SCOPED_TRACE(i);
+
+    DesktopRegion r;
+    r.AddRects(cases[i].input_rects, cases[i].input_count);
+    CompareRegion(r, cases[i].expected_rects, cases[i].expected_count);
+
+    // Try inserting rectangles in reverse order.
+    r.Clear();
+    for (int j = cases[i].input_count - 1; j >= 0; --j) {
+      r.AddRect(cases[i].input_rects[j]);
+    }
+    CompareRegion(r, cases[i].expected_rects, cases[i].expected_count);
+  }
+}
+
+TEST(DesktopRegionTest, Equals) {
+  struct Region {
+    int count;
+    DesktopRect rects[4];
+    int id;
+  } regions[] = {
+      // Same region with one of the rectangles 1 pixel wider/taller.
+      {2,
+       {DesktopRect::MakeLTRB(0, 100, 200, 200),
+        DesktopRect::MakeLTRB(310, 110, 320, 120)},
+       0},
+      {2,
+       {DesktopRect::MakeLTRB(0, 100, 201, 200),
+        DesktopRect::MakeLTRB(310, 110, 320, 120)},
+       1},
+      {2,
+       {DesktopRect::MakeLTRB(0, 100, 200, 201),
+        DesktopRect::MakeLTRB(310, 110, 320, 120)},
+       2},
+
+      // Same region with one of the rectangles shifted horizontally and
+      // vertically.
+      {4,
+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
+        DesktopRect::MakeLTRB(10, 10, 40, 40),
+        DesktopRect::MakeLTRB(20, 20, 50, 50),
+        DesktopRect::MakeLTRB(50, 0, 65, 15)},
+       3},
+      {4,
+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
+        DesktopRect::MakeLTRB(10, 10, 40, 40),
+        DesktopRect::MakeLTRB(20, 20, 50, 50),
+        DesktopRect::MakeLTRB(50, 1, 65, 16)},
+       4},
+      {4,
+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
+        DesktopRect::MakeLTRB(10, 10, 40, 40),
+        DesktopRect::MakeLTRB(20, 20, 50, 50),
+        DesktopRect::MakeLTRB(51, 0, 66, 15)},
+       5},
+
+      // Same region defined by a different set of rectangles - one of the
+      // rectangle is split horizontally into two.
+      {3,
+       {DesktopRect::MakeLTRB(100, 100, 200, 200),
+        DesktopRect::MakeLTRB(50, 50, 150, 150),
+        DesktopRect::MakeLTRB(300, 125, 350, 175)},
+       6},
+      {4,
+       {DesktopRect::MakeLTRB(100, 100, 200, 200),
+        DesktopRect::MakeLTRB(50, 50, 100, 150),
+        DesktopRect::MakeLTRB(100, 50, 150, 150),
+        DesktopRect::MakeLTRB(300, 125, 350, 175)},
+       6},
+
+      // Rectangle region defined by a set of rectangles that merge into one.
+      {3,
+       {DesktopRect::MakeLTRB(10, 10, 40, 20),
+        DesktopRect::MakeLTRB(10, 30, 40, 40),
+        DesktopRect::MakeLTRB(10, 20, 40, 30)},
+       7},
+      {1, {DesktopRect::MakeLTRB(10, 10, 40, 40)}, 7},
+  };
+  int kTotalRegions = sizeof(regions) / sizeof(Region);
+
+  for (int i = 0; i < kTotalRegions; ++i) {
+    SCOPED_TRACE(i);
+
+    DesktopRegion r1(regions[i].rects, regions[i].count);
+    for (int j = 0; j < kTotalRegions; ++j) {
+      SCOPED_TRACE(j);
+
+      DesktopRegion r2(regions[j].rects, regions[j].count);
+      EXPECT_EQ(regions[i].id == regions[j].id, r1.Equals(r2));
+    }
+  }
+}
+
+TEST(DesktopRegionTest, Translate) {
+  struct Case {
+    int input_count;
+    DesktopRect input_rects[4];
+    int dx;
+    int dy;
+    int expected_count;
+    DesktopRect expected_rects[5];
+  } cases[] = {
+      {3,
+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
+        DesktopRect::MakeLTRB(10, 10, 40, 40),
+        DesktopRect::MakeLTRB(20, 20, 50, 50)},
+       3,
+       5,
+       5,
+       {DesktopRect::MakeLTRB(3, 5, 33, 15),
+        DesktopRect::MakeLTRB(3, 15, 43, 25),
+        DesktopRect::MakeLTRB(3, 25, 53, 35),
+        DesktopRect::MakeLTRB(13, 35, 53, 45),
+        DesktopRect::MakeLTRB(23, 45, 53, 55)}},
+  };
+
+  for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
+    SCOPED_TRACE(i);
+
+    DesktopRegion r(cases[i].input_rects, cases[i].input_count);
+    r.Translate(cases[i].dx, cases[i].dy);
+    CompareRegion(r, cases[i].expected_rects, cases[i].expected_count);
+  }
+}
+
+TEST(DesktopRegionTest, Intersect) {
+  struct Case {
+    int input1_count;
+    DesktopRect input1_rects[4];
+    int input2_count;
+    DesktopRect input2_rects[4];
+    int expected_count;
+    DesktopRect expected_rects[5];
+  } cases[] = {
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(50, 50, 150, 150)},
+       1,
+       {DesktopRect::MakeLTRB(50, 50, 100, 100)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(100, 0, 200, 300)},
+       1,
+       {DesktopRect::MakeLTRB(0, 100, 300, 200)},
+       1,
+       {DesktopRect::MakeLTRB(100, 100, 200, 200)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       2,
+       {DesktopRect::MakeLTRB(50, 10, 150, 30),
+        DesktopRect::MakeLTRB(50, 30, 160, 50)},
+       1,
+       {DesktopRect::MakeLTRB(50, 10, 100, 50)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       2,
+       {DesktopRect::MakeLTRB(50, 10, 150, 30),
+        DesktopRect::MakeLTRB(50, 30, 90, 50)},
+       2,
+       {DesktopRect::MakeLTRB(50, 10, 100, 30),
+        DesktopRect::MakeLTRB(50, 30, 90, 50)}},
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(100, 50, 200, 200)},
+       0,
+       {}},
+  };
+
+  for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
+    SCOPED_TRACE(i);
+
+    DesktopRegion r1(cases[i].input1_rects, cases[i].input1_count);
+    DesktopRegion r2(cases[i].input2_rects, cases[i].input2_count);
+
+    DesktopRegion r;
+    r.Intersect(r1, r2);
+
+    CompareRegion(r, cases[i].expected_rects, cases[i].expected_count);
+  }
+}
+
+TEST(DesktopRegionTest, Subtract) {
+  struct Case {
+    int input1_count;
+    DesktopRect input1_rects[4];
+    int input2_count;
+    DesktopRect input2_rects[4];
+    int expected_count;
+    DesktopRect expected_rects[5];
+  } cases[] = {
+      // Subtract one rect from another.
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(50, 50, 150, 150)},
+       2,
+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
+        DesktopRect::MakeLTRB(0, 50, 50, 100)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(-50, -50, 50, 50)},
+       2,
+       {DesktopRect::MakeLTRB(50, 0, 100, 50),
+        DesktopRect::MakeLTRB(0, 50, 100, 100)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(-50, 50, 50, 150)},
+       2,
+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
+        DesktopRect::MakeLTRB(50, 50, 100, 100)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(50, 50, 150, 70)},
+       3,
+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
+        DesktopRect::MakeLTRB(0, 50, 50, 70),
+        DesktopRect::MakeLTRB(0, 70, 100, 100)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(50, 50, 70, 70)},
+       4,
+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
+        DesktopRect::MakeLTRB(0, 50, 50, 70),
+        DesktopRect::MakeLTRB(70, 50, 100, 70),
+        DesktopRect::MakeLTRB(0, 70, 100, 100)}},
+
+      // Empty result.
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       0,
+       {}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(-10, -10, 110, 110)},
+       0,
+       {}},
+
+      {2,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100),
+        DesktopRect::MakeLTRB(50, 50, 150, 150)},
+       2,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100),
+        DesktopRect::MakeLTRB(50, 50, 150, 150)},
+       0,
+       {}},
+
+      // One rect out of disjoint set.
+      {3,
+       {DesktopRect::MakeLTRB(0, 0, 10, 10),
+        DesktopRect::MakeLTRB(20, 20, 30, 30),
+        DesktopRect::MakeLTRB(40, 0, 50, 10)},
+       1,
+       {DesktopRect::MakeLTRB(20, 20, 30, 30)},
+       2,
+       {DesktopRect::MakeLTRB(0, 0, 10, 10),
+        DesktopRect::MakeLTRB(40, 0, 50, 10)}},
+
+      // Row merging.
+      {3,
+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
+        DesktopRect::MakeLTRB(0, 50, 150, 70),
+        DesktopRect::MakeLTRB(0, 70, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(100, 50, 150, 70)},
+       1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
+
+      // No-op subtraction.
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(100, 0, 200, 100)},
+       1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(-100, 0, 0, 100)},
+       1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(0, 100, 0, 200)},
+       1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
+
+      {1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
+       1,
+       {DesktopRect::MakeLTRB(0, -100, 100, 0)},
+       1,
+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
+  };
+
+  for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
+    SCOPED_TRACE(i);
+
+    DesktopRegion r1(cases[i].input1_rects, cases[i].input1_count);
+    DesktopRegion r2(cases[i].input2_rects, cases[i].input2_count);
+
+    r1.Subtract(r2);
+
+    CompareRegion(r1, cases[i].expected_rects, cases[i].expected_count);
+  }
+}
+
+// Verify that DesktopRegion::SubtractRows() works correctly by creating a row
+// of not overlapping rectangles and subtracting a set of rectangle. Result
+// is verified by building a map of the region in an array and comparing it with
+// the expected values.
+TEST(DesktopRegionTest, SubtractRectOnSameRow) {
+  const int kMapWidth = 50;
+
+  struct SpanSet {
+    int count;
+    struct Range {
+      int start;
+      int end;
+    } spans[3];
+  } span_sets[] = {
+      {1, {{0, 3}}},
+      {1, {{0, 5}}},
+      {1, {{0, 7}}},
+      {1, {{0, 12}}},
+      {2, {{0, 3}, {4, 5}, {6, 16}}},
+  };
+
+  DesktopRegion base_region;
+  bool base_map[kMapWidth] = {
+      false,
+  };
+
+  base_region.AddRect(DesktopRect::MakeXYWH(5, 0, 5, 1));
+  std::fill_n(base_map + 5, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(15, 0, 5, 1));
+  std::fill_n(base_map + 15, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(25, 0, 5, 1));
+  std::fill_n(base_map + 25, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(35, 0, 5, 1));
+  std::fill_n(base_map + 35, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(45, 0, 5, 1));
+  std::fill_n(base_map + 45, 5, true);
+
+  for (size_t i = 0; i < sizeof(span_sets) / sizeof(span_sets[0]); i++) {
+    SCOPED_TRACE(i);
+    SpanSet& span_set = span_sets[i];
+    int span_set_end = span_set.spans[span_set.count - 1].end;
+    for (int x = 0; x < kMapWidth - span_set_end; ++x) {
+      SCOPED_TRACE(x);
+
+      DesktopRegion r = base_region;
+
+      bool expected_map[kMapWidth];
+      std::copy(base_map, base_map + kMapWidth, expected_map);
+
+      DesktopRegion region2;
+      for (int span = 0; span < span_set.count; span++) {
+        std::fill_n(x + expected_map + span_set.spans[span].start,
+                    span_set.spans[span].end - span_set.spans[span].start,
+                    false);
+        region2.AddRect(DesktopRect::MakeLTRB(x + span_set.spans[span].start, 0,
+                                              x + span_set.spans[span].end, 1));
+      }
+      r.Subtract(region2);
+
+      bool map[kMapWidth] = {
+          false,
+      };
+
+      int pos = -1;
+      for (DesktopRegion::Iterator it(r); !it.IsAtEnd(); it.Advance()) {
+        EXPECT_GT(it.rect().left(), pos);
+        pos = it.rect().right();
+        std::fill_n(map + it.rect().left(), it.rect().width(), true);
+      }
+
+      EXPECT_TRUE(std::equal(map, map + kMapWidth, expected_map));
+    }
+  }
+}
+
+// Verify that DesktopRegion::Subtract() works correctly by creating a column of
+// not overlapping rectangles and subtracting a set of rectangle on the same
+// column. Result is verified by building a map of the region in an array and
+// comparing it with the expected values.
+TEST(DesktopRegionTest, SubtractRectOnSameCol) {
+  const int kMapHeight = 50;
+
+  struct SpanSet {
+    int count;
+    struct Range {
+      int start;
+      int end;
+    } spans[3];
+  } span_sets[] = {
+      {1, {{0, 3}}},
+      {1, {{0, 5}}},
+      {1, {{0, 7}}},
+      {1, {{0, 12}}},
+      {2, {{0, 3}, {4, 5}, {6, 16}}},
+  };
+
+  DesktopRegion base_region;
+  bool base_map[kMapHeight] = {
+      false,
+  };
+
+  base_region.AddRect(DesktopRect::MakeXYWH(0, 5, 1, 5));
+  std::fill_n(base_map + 5, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(0, 15, 1, 5));
+  std::fill_n(base_map + 15, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(0, 25, 1, 5));
+  std::fill_n(base_map + 25, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(0, 35, 1, 5));
+  std::fill_n(base_map + 35, 5, true);
+  base_region.AddRect(DesktopRect::MakeXYWH(0, 45, 1, 5));
+  std::fill_n(base_map + 45, 5, true);
+
+  for (size_t i = 0; i < sizeof(span_sets) / sizeof(span_sets[0]); i++) {
+    SCOPED_TRACE(i);
+    SpanSet& span_set = span_sets[i];
+    int span_set_end = span_set.spans[span_set.count - 1].end;
+    for (int y = 0; y < kMapHeight - span_set_end; ++y) {
+      SCOPED_TRACE(y);
+
+      DesktopRegion r = base_region;
+
+      bool expected_map[kMapHeight];
+      std::copy(base_map, base_map + kMapHeight, expected_map);
+
+      DesktopRegion region2;
+      for (int span = 0; span < span_set.count; span++) {
+        std::fill_n(y + expected_map + span_set.spans[span].start,
+                    span_set.spans[span].end - span_set.spans[span].start,
+                    false);
+        region2.AddRect(DesktopRect::MakeLTRB(0, y + span_set.spans[span].start,
+                                              1, y + span_set.spans[span].end));
+      }
+      r.Subtract(region2);
+
+      bool map[kMapHeight] = {
+          false,
+      };
+
+      int pos = -1;
+      for (DesktopRegion::Iterator it(r); !it.IsAtEnd(); it.Advance()) {
+        EXPECT_GT(it.rect().top(), pos);
+        pos = it.rect().bottom();
+        std::fill_n(map + it.rect().top(), it.rect().height(), true);
+      }
+
+      for (int j = 0; j < kMapHeight; j++) {
+        EXPECT_EQ(expected_map[j], map[j]) << "j = " << j;
+      }
+    }
+  }
+}
+
+TEST(DesktopRegionTest, DISABLED_Performance) {
+  for (int c = 0; c < 1000; ++c) {
+    DesktopRegion r;
+    for (int i = 0; i < 10; ++i) {
+      r.AddRect(
+          DesktopRect::MakeXYWH(RadmonInt(1000), RadmonInt(1000), 200, 200));
+    }
+
+    for (int i = 0; i < 1000; ++i) {
+      r.AddRect(DesktopRect::MakeXYWH(RadmonInt(1000), RadmonInt(1000),
+                                      5 + RadmonInt(10) * 5,
+                                      5 + RadmonInt(10) * 5));
+    }
+
+    // Iterate over the rectangles.
+    for (DesktopRegion::Iterator it(r); !it.IsAtEnd(); it.Advance()) {
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/differ_block.cc b/third_party/libwebrtc/modules/desktop_capture/differ_block.cc
new file mode 100644
index 0000000000..c51d33e41e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/differ_block.cc
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/differ_block.h"
+
+#include <string.h>
+
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+
+// This needs to be after rtc_base/system/arch.h which defines
+// architecture macros.
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+#include "modules/desktop_capture/differ_vector_sse2.h"
+#endif
+
+namespace webrtc {
+
+namespace {
+
+bool VectorDifference_C(const uint8_t* image1, const uint8_t* image2) {
+  return memcmp(image1, image2, kBlockSize * kBytesPerPixel) != 0;
+}
+
+}  // namespace
+
+bool VectorDifference(const uint8_t* image1, const uint8_t* image2) {
+  static bool (*diff_proc)(const uint8_t*, const uint8_t*) = nullptr;
+
+  if (!diff_proc) {
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+    bool have_sse2 = GetCPUInfo(kSSE2) != 0;
+    // For x86 processors, check if SSE2 is supported.
+    if (have_sse2 && kBlockSize == 32) {
+      diff_proc = &VectorDifference_SSE2_W32;
+    } else if (have_sse2 && kBlockSize == 16) {
+      diff_proc = &VectorDifference_SSE2_W16;
+    } else {
+      diff_proc = &VectorDifference_C;
+    }
+#else
+    // For other processors, always use C version.
+    // TODO(hclam): Implement a NEON version.
+    diff_proc = &VectorDifference_C;
+#endif
+  }
+
+  return diff_proc(image1, image2);
+}
+
+bool BlockDifference(const uint8_t* image1,
+                     const uint8_t* image2,
+                     int height,
+                     int stride) {
+  for (int i = 0; i < height; i++) {
+    if (VectorDifference(image1, image2)) {
+      return true;
+    }
+    image1 += stride;
+    image2 += stride;
+  }
+  return false;
+}
+
+bool BlockDifference(const uint8_t* image1, const uint8_t* image2, int stride) {
+  return BlockDifference(image1, image2, kBlockSize, stride);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/differ_block.h b/third_party/libwebrtc/modules/desktop_capture/differ_block.h
new file mode 100644
index 0000000000..6c71e214e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/differ_block.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_DIFFER_BLOCK_H_
+#define MODULES_DESKTOP_CAPTURE_DIFFER_BLOCK_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+// Size (in pixels) of each square block used for diffing. This must be a
+// multiple of sizeof(uint64)/8.
+const int kBlockSize = 32;
+
+// Format: BGRA 32 bit.
+const int kBytesPerPixel = 4;
+
+// Low level function to compare 2 vectors of pixels of size kBlockSize. Returns
+// whether the blocks differ.
+bool VectorDifference(const uint8_t* image1, const uint8_t* image2);
+
+// Low level function to compare 2 blocks of pixels of size
+// (kBlockSize, `height`).  Returns whether the blocks differ.
+bool BlockDifference(const uint8_t* image1,
+                     const uint8_t* image2,
+                     int height,
+                     int stride);
+
+// Low level function to compare 2 blocks of pixels of size
+// (kBlockSize, kBlockSize).  Returns whether the blocks differ.
+bool BlockDifference(const uint8_t* image1, const uint8_t* image2, int stride);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DIFFER_BLOCK_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/differ_block_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/differ_block_unittest.cc
new file mode 100644
index 0000000000..aa454c872d
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/differ_block_unittest.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/differ_block.h"
+
+#include <string.h>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Run 900 times to mimic 1280x720.
+// TODO(fbarchard): Remove benchmark once performance is non-issue.
+static const int kTimesToRun = 900;
+
+static void GenerateData(uint8_t* data, int size) {
+  for (int i = 0; i < size; ++i) {
+    data[i] = i;
+  }
+}
+
+// Memory buffer large enough for 2 blocks aligned to 16 bytes.
+static const int kSizeOfBlock = kBlockSize * kBlockSize * kBytesPerPixel;
+uint8_t block_buffer[kSizeOfBlock * 2 + 16];
+
+void PrepareBuffers(uint8_t*& block1, uint8_t*& block2) {
+  block1 = reinterpret_cast<uint8_t*>(
+      (reinterpret_cast<uintptr_t>(&block_buffer[0]) + 15) & ~15);
+  GenerateData(block1, kSizeOfBlock);
+  block2 = block1 + kSizeOfBlock;
+  memcpy(block2, block1, kSizeOfBlock);
+}
+
+TEST(BlockDifferenceTestSame, BlockDifference) {
+  uint8_t* block1;
+  uint8_t* block2;
+  PrepareBuffers(block1, block2);
+
+  // These blocks should match.
+  for (int i = 0; i < kTimesToRun; ++i) {
+    int result = BlockDifference(block1, block2, kBlockSize * kBytesPerPixel);
+    EXPECT_EQ(0, result);
+  }
+}
+
+TEST(BlockDifferenceTestLast, BlockDifference) {
+  uint8_t* block1;
+  uint8_t* block2;
+  PrepareBuffers(block1, block2);
+  block2[kSizeOfBlock - 2] += 1;
+
+  for (int i = 0; i < kTimesToRun; ++i) {
+    int result = BlockDifference(block1, block2, kBlockSize * kBytesPerPixel);
+    EXPECT_EQ(1, result);
+  }
+}
+
+TEST(BlockDifferenceTestMid, BlockDifference) {
+  uint8_t* block1;
+  uint8_t* block2;
+  PrepareBuffers(block1, block2);
+  block2[kSizeOfBlock / 2 + 1] += 1;
+
+  for (int i = 0; i < kTimesToRun; ++i) {
+    int result = BlockDifference(block1, block2, kBlockSize * kBytesPerPixel);
+    EXPECT_EQ(1, result);
+  }
+}
+
+TEST(BlockDifferenceTestFirst, BlockDifference) {
+  uint8_t* block1;
+  uint8_t* block2;
+  PrepareBuffers(block1, block2);
+  block2[0] += 1;
+
+  for (int i = 0; i < kTimesToRun; ++i) {
+    int result = BlockDifference(block1, block2, kBlockSize * kBytesPerPixel);
+    EXPECT_EQ(1, result);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/differ_vector_sse2.cc b/third_party/libwebrtc/modules/desktop_capture/differ_vector_sse2.cc
new file mode 100644
index 0000000000..1c8b602d71
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/differ_vector_sse2.cc
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/differ_vector_sse2.h"
+
+#if defined(_MSC_VER)
+#include <intrin.h>
+#else
+#include <emmintrin.h>
+#include <mmintrin.h>
+#endif
+
+namespace webrtc {
+
+extern bool VectorDifference_SSE2_W16(const uint8_t* image1,
+                                      const uint8_t* image2) {
+  __m128i acc = _mm_setzero_si128();
+  __m128i v0;
+  __m128i v1;
+  __m128i sad;
+  const __m128i* i1 = reinterpret_cast<const __m128i*>(image1);
+  const __m128i* i2 = reinterpret_cast<const __m128i*>(image2);
+  v0 = _mm_loadu_si128(i1);
+  v1 = _mm_loadu_si128(i2);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 1);
+  v1 = _mm_loadu_si128(i2 + 1);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 2);
+  v1 = _mm_loadu_si128(i2 + 2);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 3);
+  v1 = _mm_loadu_si128(i2 + 3);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+
+  // This essential means sad = acc >> 64. We only care about the lower 16
+  // bits.
+  sad = _mm_shuffle_epi32(acc, 0xEE);
+  sad = _mm_adds_epu16(sad, acc);
+  return _mm_cvtsi128_si32(sad) != 0;
+}
+
+extern bool VectorDifference_SSE2_W32(const uint8_t* image1,
+                                      const uint8_t* image2) {
+  __m128i acc = _mm_setzero_si128();
+  __m128i v0;
+  __m128i v1;
+  __m128i sad;
+  const __m128i* i1 = reinterpret_cast<const __m128i*>(image1);
+  const __m128i* i2 = reinterpret_cast<const __m128i*>(image2);
+  v0 = _mm_loadu_si128(i1);
+  v1 = _mm_loadu_si128(i2);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 1);
+  v1 = _mm_loadu_si128(i2 + 1);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 2);
+  v1 = _mm_loadu_si128(i2 + 2);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 3);
+  v1 = _mm_loadu_si128(i2 + 3);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 4);
+  v1 = _mm_loadu_si128(i2 + 4);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 5);
+  v1 = _mm_loadu_si128(i2 + 5);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 6);
+  v1 = _mm_loadu_si128(i2 + 6);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+  v0 = _mm_loadu_si128(i1 + 7);
+  v1 = _mm_loadu_si128(i2 + 7);
+  sad = _mm_sad_epu8(v0, v1);
+  acc = _mm_adds_epu16(acc, sad);
+
+  // This essential means sad = acc >> 64. We only care about the lower 16
+  // bits.
+  sad = _mm_shuffle_epi32(acc, 0xEE);
+  sad = _mm_adds_epu16(sad, acc);
+  return _mm_cvtsi128_si32(sad) != 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/differ_vector_sse2.h b/third_party/libwebrtc/modules/desktop_capture/differ_vector_sse2.h
new file mode 100644
index 0000000000..a3c297eb9d
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/differ_vector_sse2.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This header file is used only differ_block.h. It defines the SSE2 rountines
+// for finding vector difference.
+
+#ifndef MODULES_DESKTOP_CAPTURE_DIFFER_VECTOR_SSE2_H_
+#define MODULES_DESKTOP_CAPTURE_DIFFER_VECTOR_SSE2_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+// Find vector difference of dimension 16.
+extern bool VectorDifference_SSE2_W16(const uint8_t* image1,
+                                      const uint8_t* image2);
+
+// Find vector difference of dimension 32.
+extern bool VectorDifference_SSE2_W32(const uint8_t* image1,
+                                      const uint8_t* image2);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_DIFFER_VECTOR_SSE2_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/fake_desktop_capturer.cc b/third_party/libwebrtc/modules/desktop_capture/fake_desktop_capturer.cc
new file mode 100644
index 0000000000..67149bfcb9
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/fake_desktop_capturer.cc
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/fake_desktop_capturer.h"
+
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_frame.h"
+
+namespace webrtc {
+
+FakeDesktopCapturer::FakeDesktopCapturer() = default;
+FakeDesktopCapturer::~FakeDesktopCapturer() = default;
+
+void FakeDesktopCapturer::set_result(DesktopCapturer::Result result) {
+  result_ = result;
+}
+
+int FakeDesktopCapturer::num_frames_captured() const {
+  return num_frames_captured_;
+}
+
+int FakeDesktopCapturer::num_capture_attempts() const {
+  return num_capture_attempts_;
+}
+
+// Uses the `generator` provided as DesktopFrameGenerator, FakeDesktopCapturer
+// does
+// not take the ownership of `generator`.
+void FakeDesktopCapturer::set_frame_generator(
+    DesktopFrameGenerator* generator) {
+  generator_ = generator;
+}
+
+void FakeDesktopCapturer::Start(DesktopCapturer::Callback* callback) {
+  callback_ = callback;
+}
+
+void FakeDesktopCapturer::CaptureFrame() {
+  num_capture_attempts_++;
+  if (generator_) {
+    if (result_ != DesktopCapturer::Result::SUCCESS) {
+      callback_->OnCaptureResult(result_, nullptr);
+      return;
+    }
+
+    std::unique_ptr<DesktopFrame> frame(
+        generator_->GetNextFrame(shared_memory_factory_.get()));
+    if (frame) {
+      num_frames_captured_++;
+      callback_->OnCaptureResult(result_, std::move(frame));
+    } else {
+      callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_TEMPORARY,
+                                 nullptr);
+    }
+    return;
+  }
+  callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_PERMANENT, nullptr);
+}
+
+void FakeDesktopCapturer::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  shared_memory_factory_ = std::move(shared_memory_factory);
+}
+
+bool FakeDesktopCapturer::GetSourceList(DesktopCapturer::SourceList* sources) {
+  sources->push_back({kWindowId, 1, "A-Fake-DesktopCapturer-Window"});
+  sources->push_back({kScreenId, 1});
+  return true;
+}
+
+bool FakeDesktopCapturer::SelectSource(DesktopCapturer::SourceId id) {
+  return id == kWindowId || id == kScreenId || id == kFullDesktopScreenId;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/fake_desktop_capturer.h b/third_party/libwebrtc/modules/desktop_capture/fake_desktop_capturer.h
new file mode 100644
index 0000000000..086e6df0e2
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/fake_desktop_capturer.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_FAKE_DESKTOP_CAPTURER_H_
+#define MODULES_DESKTOP_CAPTURE_FAKE_DESKTOP_CAPTURER_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame_generator.h"
+#include "modules/desktop_capture/shared_memory.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// A fake implementation of DesktopCapturer or its derived interfaces to
+// generate DesktopFrame for testing purpose.
+//
+// Consumers can provide a FrameGenerator instance to generate instances of
+// DesktopFrame to return for each Capture() function call.
+// If no FrameGenerator provided, FakeDesktopCapturer will always return a
+// nullptr DesktopFrame.
+//
+// Double buffering is guaranteed by the FrameGenerator. FrameGenerator
+// implements in desktop_frame_generator.h guarantee double buffering, they
+// creates a new instance of DesktopFrame each time.
+class RTC_EXPORT FakeDesktopCapturer : public DesktopCapturer {
+ public:
+  FakeDesktopCapturer();
+  ~FakeDesktopCapturer() override;
+
+  // Decides the result which will be returned in next Capture() callback.
+  void set_result(DesktopCapturer::Result result);
+
+  // Uses the `generator` provided as DesktopFrameGenerator, FakeDesktopCapturer
+  // does not take the ownership of `generator`.
+  void set_frame_generator(DesktopFrameGenerator* generator);
+
+  // Count of DesktopFrame(s) have been returned by this instance. This field
+  // would never be negative.
+  int num_frames_captured() const;
+
+  // Count of CaptureFrame() calls have been made. This field would never be
+  // negative.
+  int num_capture_attempts() const;
+
+  // DesktopCapturer interface
+  void Start(DesktopCapturer::Callback* callback) override;
+  void CaptureFrame() override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  bool GetSourceList(DesktopCapturer::SourceList* sources) override;
+  bool SelectSource(DesktopCapturer::SourceId id) override;
+
+ private:
+  static constexpr DesktopCapturer::SourceId kWindowId = 1378277495;
+  static constexpr DesktopCapturer::SourceId kScreenId = 1378277496;
+
+  DesktopCapturer::Callback* callback_ = nullptr;
+  std::unique_ptr<SharedMemoryFactory> shared_memory_factory_;
+  DesktopCapturer::Result result_ = Result::SUCCESS;
+  DesktopFrameGenerator* generator_ = nullptr;
+  int num_frames_captured_ = 0;
+  int num_capture_attempts_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_FAKE_DESKTOP_CAPTURER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.cc b/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.cc
new file mode 100644
index 0000000000..61fdb416f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.cc
@@ -0,0 +1,183 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/fallback_desktop_capturer_wrapper.h"
+
+#include <stddef.h>
+
+#include <utility>
+
+#include "api/sequence_checker.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+// Implementation to share a SharedMemoryFactory between DesktopCapturer
+// instances. This class is designed for synchronized DesktopCapturer
+// implementations only.
+class SharedMemoryFactoryProxy : public SharedMemoryFactory {
+ public:
+  // Users should maintain the lifetime of `factory` to ensure it overlives
+  // current instance.
+  static std::unique_ptr<SharedMemoryFactory> Create(
+      SharedMemoryFactory* factory);
+  ~SharedMemoryFactoryProxy() override;
+
+  // Forwards CreateSharedMemory() calls to `factory_`. Users should always call
+  // this function in one thread. Users should not call this function after the
+  // SharedMemoryFactory which current instance created from has been destroyed.
+  std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) override;
+
+ private:
+  explicit SharedMemoryFactoryProxy(SharedMemoryFactory* factory);
+
+  SharedMemoryFactory* factory_ = nullptr;
+  SequenceChecker thread_checker_;
+};
+
+}  // namespace
+
+SharedMemoryFactoryProxy::SharedMemoryFactoryProxy(
+    SharedMemoryFactory* factory) {
+  RTC_DCHECK(factory);
+  factory_ = factory;
+}
+
+// static
+std::unique_ptr<SharedMemoryFactory> SharedMemoryFactoryProxy::Create(
+    SharedMemoryFactory* factory) {
+  return std::unique_ptr<SharedMemoryFactory>(
+      new SharedMemoryFactoryProxy(factory));
+}
+
+SharedMemoryFactoryProxy::~SharedMemoryFactoryProxy() = default;
+
+std::unique_ptr<SharedMemory> SharedMemoryFactoryProxy::CreateSharedMemory(
+    size_t size) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return factory_->CreateSharedMemory(size);
+}
+
+FallbackDesktopCapturerWrapper::FallbackDesktopCapturerWrapper(
+    std::unique_ptr<DesktopCapturer> main_capturer,
+    std::unique_ptr<DesktopCapturer> secondary_capturer)
+    : main_capturer_(std::move(main_capturer)),
+      secondary_capturer_(std::move(secondary_capturer)) {
+  RTC_DCHECK(main_capturer_);
+  RTC_DCHECK(secondary_capturer_);
+}
+
+FallbackDesktopCapturerWrapper::~FallbackDesktopCapturerWrapper() = default;
+
+void FallbackDesktopCapturerWrapper::Start(
+    DesktopCapturer::Callback* callback) {
+  callback_ = callback;
+  // FallbackDesktopCapturerWrapper catchs the callback of the main capturer,
+  // and checks its return value to decide whether the secondary capturer should
+  // be involved.
+  main_capturer_->Start(this);
+  // For the secondary capturer, we do not have a backup plan anymore, so
+  // FallbackDesktopCapturerWrapper won't check its return value any more. It
+  // will directly return to the input `callback`.
+  secondary_capturer_->Start(callback);
+}
+
+void FallbackDesktopCapturerWrapper::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  shared_memory_factory_ = std::move(shared_memory_factory);
+  if (shared_memory_factory_) {
+    main_capturer_->SetSharedMemoryFactory(
+        SharedMemoryFactoryProxy::Create(shared_memory_factory_.get()));
+    secondary_capturer_->SetSharedMemoryFactory(
+        SharedMemoryFactoryProxy::Create(shared_memory_factory_.get()));
+  } else {
+    main_capturer_->SetSharedMemoryFactory(
+        std::unique_ptr<SharedMemoryFactory>());
+    secondary_capturer_->SetSharedMemoryFactory(
+        std::unique_ptr<SharedMemoryFactory>());
+  }
+}
+
+void FallbackDesktopCapturerWrapper::CaptureFrame() {
+  RTC_DCHECK(callback_);
+  if (main_capturer_permanent_error_) {
+    secondary_capturer_->CaptureFrame();
+  } else {
+    main_capturer_->CaptureFrame();
+  }
+}
+
+void FallbackDesktopCapturerWrapper::SetExcludedWindow(WindowId window) {
+  main_capturer_->SetExcludedWindow(window);
+  secondary_capturer_->SetExcludedWindow(window);
+}
+
+bool FallbackDesktopCapturerWrapper::GetSourceList(SourceList* sources) {
+  if (main_capturer_permanent_error_) {
+    return secondary_capturer_->GetSourceList(sources);
+  }
+  return main_capturer_->GetSourceList(sources);
+}
+
+bool FallbackDesktopCapturerWrapper::SelectSource(SourceId id) {
+  if (main_capturer_permanent_error_) {
+    return secondary_capturer_->SelectSource(id);
+  }
+  const bool main_capturer_result = main_capturer_->SelectSource(id);
+  RTC_HISTOGRAM_BOOLEAN(
+      "WebRTC.DesktopCapture.PrimaryCapturerSelectSourceError",
+      main_capturer_result);
+  if (!main_capturer_result) {
+    main_capturer_permanent_error_ = true;
+  }
+
+  return secondary_capturer_->SelectSource(id);
+}
+
+bool FallbackDesktopCapturerWrapper::FocusOnSelectedSource() {
+  if (main_capturer_permanent_error_) {
+    return secondary_capturer_->FocusOnSelectedSource();
+  }
+  return main_capturer_->FocusOnSelectedSource() ||
+         secondary_capturer_->FocusOnSelectedSource();
+}
+
+bool FallbackDesktopCapturerWrapper::IsOccluded(const DesktopVector& pos) {
+  // Returns true if either capturer returns true.
+  if (main_capturer_permanent_error_) {
+    return secondary_capturer_->IsOccluded(pos);
+  }
+  return main_capturer_->IsOccluded(pos) ||
+         secondary_capturer_->IsOccluded(pos);
+}
+
+void FallbackDesktopCapturerWrapper::OnCaptureResult(
+    Result result,
+    std::unique_ptr<DesktopFrame> frame) {
+  RTC_DCHECK(callback_);
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.DesktopCapture.PrimaryCapturerError",
+                        result != Result::SUCCESS);
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.DesktopCapture.PrimaryCapturerPermanentError",
+                        result == Result::ERROR_PERMANENT);
+  if (result == Result::SUCCESS) {
+    callback_->OnCaptureResult(result, std::move(frame));
+    return;
+  }
+
+  if (result == Result::ERROR_PERMANENT) {
+    main_capturer_permanent_error_ = true;
+  }
+  secondary_capturer_->CaptureFrame();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.h b/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.h
new file mode 100644
index 0000000000..2855eae7ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_FALLBACK_DESKTOP_CAPTURER_WRAPPER_H_
+#define MODULES_DESKTOP_CAPTURE_FALLBACK_DESKTOP_CAPTURER_WRAPPER_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/shared_memory.h"
+
+namespace webrtc {
+
+// A DesktopCapturer wrapper owns two DesktopCapturer implementations. If the
+// main DesktopCapturer fails, it uses the secondary one instead. Two capturers
+// are expected to return same SourceList, and the meaning of each SourceId is
+// identical, otherwise FallbackDesktopCapturerWrapper may return frames from
+// different sources. Using asynchronized DesktopCapturer implementations with
+// SharedMemoryFactory is not supported, and may result crash or assertion
+// failure.
+class FallbackDesktopCapturerWrapper final : public DesktopCapturer,
+                                             public DesktopCapturer::Callback {
+ public:
+  FallbackDesktopCapturerWrapper(
+      std::unique_ptr<DesktopCapturer> main_capturer,
+      std::unique_ptr<DesktopCapturer> secondary_capturer);
+  ~FallbackDesktopCapturerWrapper() override;
+
+  // DesktopCapturer interface.
+  void Start(DesktopCapturer::Callback* callback) override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  void CaptureFrame() override;
+  void SetExcludedWindow(WindowId window) override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  bool IsOccluded(const DesktopVector& pos) override;
+
+ private:
+  // DesktopCapturer::Callback interface.
+  void OnCaptureResult(Result result,
+                       std::unique_ptr<DesktopFrame> frame) override;
+
+  const std::unique_ptr<DesktopCapturer> main_capturer_;
+  const std::unique_ptr<DesktopCapturer> secondary_capturer_;
+  std::unique_ptr<SharedMemoryFactory> shared_memory_factory_;
+  bool main_capturer_permanent_error_ = false;
+  DesktopCapturer::Callback* callback_ = nullptr;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_FALLBACK_DESKTOP_CAPTURER_WRAPPER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper_unittest.cc
new file mode 100644
index 0000000000..de66386434
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper_unittest.cc
@@ -0,0 +1,207 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/fallback_desktop_capturer_wrapper.h"
+
+#include <stddef.h>
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame_generator.h"
+#include "modules/desktop_capture/fake_desktop_capturer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+std::unique_ptr<DesktopCapturer> CreateDesktopCapturer(
+    PainterDesktopFrameGenerator* frame_generator) {
+  std::unique_ptr<FakeDesktopCapturer> capturer(new FakeDesktopCapturer());
+  capturer->set_frame_generator(frame_generator);
+  return std::move(capturer);
+}
+
+class FakeSharedMemory : public SharedMemory {
+ public:
+  explicit FakeSharedMemory(size_t size);
+  ~FakeSharedMemory() override;
+
+ private:
+  static int next_id_;
+};
+
+// static
+int FakeSharedMemory::next_id_ = 0;
+
+FakeSharedMemory::FakeSharedMemory(size_t size)
+    : SharedMemory(new char[size], size, 0, next_id_++) {}
+
+FakeSharedMemory::~FakeSharedMemory() {
+  delete[] static_cast<char*>(data_);
+}
+
+class FakeSharedMemoryFactory : public SharedMemoryFactory {
+ public:
+  FakeSharedMemoryFactory() = default;
+  ~FakeSharedMemoryFactory() override = default;
+
+  std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) override;
+};
+
+std::unique_ptr<SharedMemory> FakeSharedMemoryFactory::CreateSharedMemory(
+    size_t size) {
+  return std::unique_ptr<SharedMemory>(new FakeSharedMemory(size));
+}
+
+}  // namespace
+
+class FallbackDesktopCapturerWrapperTest : public ::testing::Test,
+                                           public DesktopCapturer::Callback {
+ public:
+  FallbackDesktopCapturerWrapperTest();
+  ~FallbackDesktopCapturerWrapperTest() override = default;
+
+ protected:
+  std::vector<std::pair<DesktopCapturer::Result, bool>> results_;
+  FakeDesktopCapturer* main_capturer_ = nullptr;
+  FakeDesktopCapturer* secondary_capturer_ = nullptr;
+  std::unique_ptr<FallbackDesktopCapturerWrapper> wrapper_;
+
+ private:
+  // DesktopCapturer::Callback interface
+  void OnCaptureResult(DesktopCapturer::Result result,
+                       std::unique_ptr<DesktopFrame> frame) override;
+  PainterDesktopFrameGenerator frame_generator_;
+};
+
+FallbackDesktopCapturerWrapperTest::FallbackDesktopCapturerWrapperTest() {
+  frame_generator_.size()->set(1024, 768);
+  std::unique_ptr<DesktopCapturer> main_capturer =
+      CreateDesktopCapturer(&frame_generator_);
+  std::unique_ptr<DesktopCapturer> secondary_capturer =
+      CreateDesktopCapturer(&frame_generator_);
+  main_capturer_ = static_cast<FakeDesktopCapturer*>(main_capturer.get());
+  secondary_capturer_ =
+      static_cast<FakeDesktopCapturer*>(secondary_capturer.get());
+  wrapper_.reset(new FallbackDesktopCapturerWrapper(
+      std::move(main_capturer), std::move(secondary_capturer)));
+  wrapper_->Start(this);
+}
+
+void FallbackDesktopCapturerWrapperTest::OnCaptureResult(
+    DesktopCapturer::Result result,
+    std::unique_ptr<DesktopFrame> frame) {
+  results_.emplace_back(result, !!frame);
+}
+
+TEST_F(FallbackDesktopCapturerWrapperTest, MainNeverFailed) {
+  wrapper_->CaptureFrame();
+  ASSERT_EQ(main_capturer_->num_capture_attempts(), 1);
+  ASSERT_EQ(main_capturer_->num_frames_captured(), 1);
+  ASSERT_EQ(secondary_capturer_->num_capture_attempts(), 0);
+  ASSERT_EQ(secondary_capturer_->num_frames_captured(), 0);
+  ASSERT_EQ(results_.size(), 1U);
+  ASSERT_EQ(results_[0],
+            std::make_pair(DesktopCapturer::Result::SUCCESS, true));
+}
+
+TEST_F(FallbackDesktopCapturerWrapperTest, MainFailedTemporarily) {
+  wrapper_->CaptureFrame();
+  main_capturer_->set_result(DesktopCapturer::Result::ERROR_TEMPORARY);
+  wrapper_->CaptureFrame();
+  main_capturer_->set_result(DesktopCapturer::Result::SUCCESS);
+  wrapper_->CaptureFrame();
+
+  ASSERT_EQ(main_capturer_->num_capture_attempts(), 3);
+  ASSERT_EQ(main_capturer_->num_frames_captured(), 2);
+  ASSERT_EQ(secondary_capturer_->num_capture_attempts(), 1);
+  ASSERT_EQ(secondary_capturer_->num_frames_captured(), 1);
+  ASSERT_EQ(results_.size(), 3U);
+  for (int i = 0; i < 3; i++) {
+    ASSERT_EQ(results_[i],
+              std::make_pair(DesktopCapturer::Result::SUCCESS, true));
+  }
+}
+
+TEST_F(FallbackDesktopCapturerWrapperTest, MainFailedPermanently) {
+  wrapper_->CaptureFrame();
+  main_capturer_->set_result(DesktopCapturer::Result::ERROR_PERMANENT);
+  wrapper_->CaptureFrame();
+  main_capturer_->set_result(DesktopCapturer::Result::SUCCESS);
+  wrapper_->CaptureFrame();
+
+  ASSERT_EQ(main_capturer_->num_capture_attempts(), 2);
+  ASSERT_EQ(main_capturer_->num_frames_captured(), 1);
+  ASSERT_EQ(secondary_capturer_->num_capture_attempts(), 2);
+  ASSERT_EQ(secondary_capturer_->num_frames_captured(), 2);
+  ASSERT_EQ(results_.size(), 3U);
+  for (int i = 0; i < 3; i++) {
+    ASSERT_EQ(results_[i],
+              std::make_pair(DesktopCapturer::Result::SUCCESS, true));
+  }
+}
+
+TEST_F(FallbackDesktopCapturerWrapperTest, BothFailed) {
+  wrapper_->CaptureFrame();
+  main_capturer_->set_result(DesktopCapturer::Result::ERROR_PERMANENT);
+  wrapper_->CaptureFrame();
+  main_capturer_->set_result(DesktopCapturer::Result::SUCCESS);
+  wrapper_->CaptureFrame();
+  secondary_capturer_->set_result(DesktopCapturer::Result::ERROR_TEMPORARY);
+  wrapper_->CaptureFrame();
+  secondary_capturer_->set_result(DesktopCapturer::Result::ERROR_PERMANENT);
+  wrapper_->CaptureFrame();
+  wrapper_->CaptureFrame();
+
+  ASSERT_EQ(main_capturer_->num_capture_attempts(), 2);
+  ASSERT_EQ(main_capturer_->num_frames_captured(), 1);
+  ASSERT_EQ(secondary_capturer_->num_capture_attempts(), 5);
+  ASSERT_EQ(secondary_capturer_->num_frames_captured(), 2);
+  ASSERT_EQ(results_.size(), 6U);
+  for (int i = 0; i < 3; i++) {
+    ASSERT_EQ(results_[i],
+              std::make_pair(DesktopCapturer::Result::SUCCESS, true));
+  }
+  ASSERT_EQ(results_[3],
+            std::make_pair(DesktopCapturer::Result::ERROR_TEMPORARY, false));
+  ASSERT_EQ(results_[4],
+            std::make_pair(DesktopCapturer::Result::ERROR_PERMANENT, false));
+  ASSERT_EQ(results_[5],
+            std::make_pair(DesktopCapturer::Result::ERROR_PERMANENT, false));
+}
+
+TEST_F(FallbackDesktopCapturerWrapperTest, WithSharedMemory) {
+  wrapper_->SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory>(new FakeSharedMemoryFactory()));
+  wrapper_->CaptureFrame();
+  main_capturer_->set_result(DesktopCapturer::Result::ERROR_TEMPORARY);
+  wrapper_->CaptureFrame();
+  main_capturer_->set_result(DesktopCapturer::Result::SUCCESS);
+  wrapper_->CaptureFrame();
+  main_capturer_->set_result(DesktopCapturer::Result::ERROR_PERMANENT);
+  wrapper_->CaptureFrame();
+  wrapper_->CaptureFrame();
+
+  ASSERT_EQ(main_capturer_->num_capture_attempts(), 4);
+  ASSERT_EQ(main_capturer_->num_frames_captured(), 2);
+  ASSERT_EQ(secondary_capturer_->num_capture_attempts(), 3);
+  ASSERT_EQ(secondary_capturer_->num_frames_captured(), 3);
+  ASSERT_EQ(results_.size(), 5U);
+  for (int i = 0; i < 5; i++) {
+    ASSERT_EQ(results_[i],
+              std::make_pair(DesktopCapturer::Result::SUCCESS, true));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/full_screen_application_handler.cc b/third_party/libwebrtc/modules/desktop_capture/full_screen_application_handler.cc
new file mode 100644
index 0000000000..e0975570ba
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/full_screen_application_handler.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/full_screen_application_handler.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+FullScreenApplicationHandler::FullScreenApplicationHandler(
+    DesktopCapturer::SourceId sourceId)
+    : source_id_(sourceId) {}
+
+DesktopCapturer::SourceId FullScreenApplicationHandler::FindFullScreenWindow(
+    const DesktopCapturer::SourceList&,
+    int64_t) const {
+  return 0;
+}
+
+DesktopCapturer::SourceId FullScreenApplicationHandler::GetSourceId() const {
+  return source_id_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/full_screen_application_handler.h b/third_party/libwebrtc/modules/desktop_capture/full_screen_application_handler.h
new file mode 100644
index 0000000000..b7e097a474
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/full_screen_application_handler.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_FULL_SCREEN_APPLICATION_HANDLER_H_
+#define MODULES_DESKTOP_CAPTURE_FULL_SCREEN_APPLICATION_HANDLER_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+
+namespace webrtc {
+
+// Base class for application specific handler to check criteria for switch to
+// full-screen mode and find if possible the full-screen window to share.
+// Supposed to be created and owned by platform specific
+// FullScreenWindowDetector.
+class FullScreenApplicationHandler {
+ public:
+  virtual ~FullScreenApplicationHandler() {}
+
+  FullScreenApplicationHandler(const FullScreenApplicationHandler&) = delete;
+  FullScreenApplicationHandler& operator=(const FullScreenApplicationHandler&) =
+      delete;
+
+  explicit FullScreenApplicationHandler(DesktopCapturer::SourceId sourceId);
+
+  // Returns the full-screen window in place of the original window if all the
+  // criteria are met, or 0 if no such window found.
+  virtual DesktopCapturer::SourceId FindFullScreenWindow(
+      const DesktopCapturer::SourceList& window_list,
+      int64_t timestamp) const;
+
+  // Returns source id of original window associated with
+  // FullScreenApplicationHandler
+  DesktopCapturer::SourceId GetSourceId() const;
+
+ private:
+  const DesktopCapturer::SourceId source_id_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_FULL_SCREEN_APPLICATION_HANDLER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/full_screen_window_detector.cc b/third_party/libwebrtc/modules/desktop_capture/full_screen_window_detector.cc
new file mode 100644
index 0000000000..d0bc9c7ca6
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/full_screen_window_detector.cc
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/full_screen_window_detector.h"
+#include "modules/desktop_capture/full_screen_application_handler.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+FullScreenWindowDetector::FullScreenWindowDetector(
+    ApplicationHandlerFactory application_handler_factory)
+    : application_handler_factory_(application_handler_factory),
+      last_update_time_ms_(0),
+      previous_source_id_(0),
+      no_handler_source_id_(0) {}
+
+DesktopCapturer::SourceId FullScreenWindowDetector::FindFullScreenWindow(
+    DesktopCapturer::SourceId original_source_id) {
+  if (app_handler_ == nullptr ||
+      app_handler_->GetSourceId() != original_source_id) {
+    return 0;
+  }
+  return app_handler_->FindFullScreenWindow(window_list_, last_update_time_ms_);
+}
+
+void FullScreenWindowDetector::UpdateWindowListIfNeeded(
+    DesktopCapturer::SourceId original_source_id,
+    rtc::FunctionView<bool(DesktopCapturer::SourceList*)> get_sources) {
+  const bool skip_update = previous_source_id_ != original_source_id;
+  previous_source_id_ = original_source_id;
+
+  // Here is an attempt to avoid redundant creating application handler in case
+  // when an instance of WindowCapturer is used to generate a thumbnail to show
+  // in picker by calling SelectSource and CaptureFrame for every available
+  // source.
+  if (skip_update) {
+    return;
+  }
+
+  CreateApplicationHandlerIfNeeded(original_source_id);
+  if (app_handler_ == nullptr) {
+    // There is no FullScreenApplicationHandler specific for
+    // current application
+    return;
+  }
+
+  constexpr int64_t kUpdateIntervalMs = 500;
+
+  if ((rtc::TimeMillis() - last_update_time_ms_) <= kUpdateIntervalMs) {
+    return;
+  }
+
+  DesktopCapturer::SourceList window_list;
+  if (get_sources(&window_list)) {
+    last_update_time_ms_ = rtc::TimeMillis();
+    window_list_.swap(window_list);
+  }
+}
+
+void FullScreenWindowDetector::CreateApplicationHandlerIfNeeded(
+    DesktopCapturer::SourceId source_id) {
+  if (no_handler_source_id_ == source_id) {
+    return;
+  }
+
+  if (app_handler_ == nullptr || app_handler_->GetSourceId() != source_id) {
+    app_handler_ = application_handler_factory_
+                       ? application_handler_factory_(source_id)
+                       : nullptr;
+  }
+
+  if (app_handler_ == nullptr) {
+    no_handler_source_id_ = source_id;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/full_screen_window_detector.h b/third_party/libwebrtc/modules/desktop_capture/full_screen_window_detector.h
new file mode 100644
index 0000000000..998b720d90
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/full_screen_window_detector.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_FULL_SCREEN_WINDOW_DETECTOR_H_
+#define MODULES_DESKTOP_CAPTURE_FULL_SCREEN_WINDOW_DETECTOR_H_
+
+#include <memory>
+
+#include "api/function_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/full_screen_application_handler.h"
+
+namespace webrtc {
+
+// This is a way to handle switch to full-screen mode for application in some
+// specific cases:
+// - Chrome on MacOS creates a new window in full-screen mode to
+//   show a tab full-screen and minimizes the old window.
+// - PowerPoint creates new windows in full-screen mode when user goes to
+//   presentation mode (Slide Show Window, Presentation Window).
+//
+// To continue capturing in these cases, we try to find the new full-screen
+// window using criteria provided by application specific
+// FullScreenApplicationHandler.
+
+class FullScreenWindowDetector
+    : public rtc::RefCountedNonVirtual<FullScreenWindowDetector> {
+ public:
+  using ApplicationHandlerFactory =
+      std::function<std::unique_ptr<FullScreenApplicationHandler>(
+          DesktopCapturer::SourceId sourceId)>;
+
+  FullScreenWindowDetector(
+      ApplicationHandlerFactory application_handler_factory);
+
+  FullScreenWindowDetector(const FullScreenWindowDetector&) = delete;
+  FullScreenWindowDetector& operator=(const FullScreenWindowDetector&) = delete;
+
+  // Returns the full-screen window in place of the original window if all the
+  // criteria provided by FullScreenApplicationHandler are met, or 0 if no such
+  // window found.
+  DesktopCapturer::SourceId FindFullScreenWindow(
+      DesktopCapturer::SourceId original_source_id);
+
+  // The caller should call this function periodically, implementation will
+  // update internal state no often than twice per second
+  void UpdateWindowListIfNeeded(
+      DesktopCapturer::SourceId original_source_id,
+      rtc::FunctionView<bool(DesktopCapturer::SourceList*)> get_sources);
+
+  static rtc::scoped_refptr<FullScreenWindowDetector>
+  CreateFullScreenWindowDetector();
+
+ protected:
+  std::unique_ptr<FullScreenApplicationHandler> app_handler_;
+
+ private:
+  void CreateApplicationHandlerIfNeeded(DesktopCapturer::SourceId source_id);
+
+  ApplicationHandlerFactory application_handler_factory_;
+
+  int64_t last_update_time_ms_;
+  DesktopCapturer::SourceId previous_source_id_;
+
+  // Save the source id when we fail to create an instance of
+  // CreateApplicationHandlerIfNeeded to avoid redundant attempt to do it again.
+  DesktopCapturer::SourceId no_handler_source_id_;
+
+  DesktopCapturer::SourceList window_list_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_FULL_SCREEN_WINDOW_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc
new file mode 100644
index 0000000000..cf4f7dc9aa
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc
@@ -0,0 +1,230 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/linux/wayland/restore_token_manager.h"
+#include "modules/portal/pipewire_utils.h"
+#include "modules/portal/xdg_desktop_portal_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+namespace {
+
+using xdg_portal::RequestResponse;
+using xdg_portal::ScreenCapturePortalInterface;
+using xdg_portal::SessionDetails;
+
+}  // namespace
+
+// static
+bool BaseCapturerPipeWire::IsSupported() {
+  // Unfortunately, the best way we have to check if PipeWire is available is
+  // to try to initialize it.
+  // InitializePipeWire should prevent us from repeatedly initializing PipeWire,
+  // but we also don't really expect support to change without the application
+  // restarting.
+  static bool supported =
+      DesktopCapturer::IsRunningUnderWayland() && InitializePipeWire();
+  return supported;
+}
+
+BaseCapturerPipeWire::BaseCapturerPipeWire(const DesktopCaptureOptions& options,
+                                           CaptureType type)
+    : BaseCapturerPipeWire(options,
+                           std::make_unique<ScreenCastPortal>(type, this)) {
+  is_screencast_portal_ = true;
+}
+
+BaseCapturerPipeWire::BaseCapturerPipeWire(
+    const DesktopCaptureOptions& options,
+    std::unique_ptr<ScreenCapturePortalInterface> portal)
+    : options_(options),
+      is_screencast_portal_(false),
+      portal_(std::move(portal)) {
+  source_id_ = RestoreTokenManager::GetInstance().GetUnusedId();
+  options_.screencast_stream()->SetUseDamageRegion(
+      options_.pipewire_use_damage_region());
+}
+
+BaseCapturerPipeWire::~BaseCapturerPipeWire() {
+  options_.screencast_stream()->StopScreenCastStream();
+}
+
+void BaseCapturerPipeWire::OnScreenCastRequestResult(RequestResponse result,
+                                                     uint32_t stream_node_id,
+                                                     int fd) {
+  is_portal_open_ = false;
+
+  // Reset the value of capturer_failed_ in case we succeed below. If we fail,
+  // then it'll set it to the right value again soon enough.
+  capturer_failed_ = false;
+  if (result != RequestResponse::kSuccess ||
+      !options_.screencast_stream()->StartScreenCastStream(
+          stream_node_id, fd, options_.get_width(), options_.get_height(),
+          options_.prefer_cursor_embedded())) {
+    capturer_failed_ = true;
+    RTC_LOG(LS_ERROR) << "ScreenCastPortal failed: "
+                      << static_cast<uint>(result);
+  } else if (ScreenCastPortal* screencast_portal = GetScreenCastPortal()) {
+    if (!screencast_portal->RestoreToken().empty()) {
+      RestoreTokenManager::GetInstance().AddToken(
+          source_id_, screencast_portal->RestoreToken());
+    }
+  }
+
+  if (!delegated_source_list_observer_)
+    return;
+
+  switch (result) {
+    case RequestResponse::kUnknown:
+      RTC_DCHECK_NOTREACHED();
+      break;
+    case RequestResponse::kSuccess:
+      delegated_source_list_observer_->OnSelection();
+      break;
+    case RequestResponse::kUserCancelled:
+      delegated_source_list_observer_->OnCancelled();
+      break;
+    case RequestResponse::kError:
+      delegated_source_list_observer_->OnError();
+      break;
+  }
+}
+
+void BaseCapturerPipeWire::OnScreenCastSessionClosed() {
+  if (!capturer_failed_) {
+    options_.screencast_stream()->StopScreenCastStream();
+  }
+}
+
+void BaseCapturerPipeWire::UpdateResolution(uint32_t width, uint32_t height) {
+  if (!capturer_failed_) {
+    options_.screencast_stream()->UpdateScreenCastStreamResolution(width,
+                                                                   height);
+  }
+}
+
+void BaseCapturerPipeWire::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+
+  callback_ = callback;
+
+  if (ScreenCastPortal* screencast_portal = GetScreenCastPortal()) {
+    screencast_portal->SetPersistMode(
+        ScreenCastPortal::PersistMode::kTransient);
+    if (selected_source_id_) {
+      screencast_portal->SetRestoreToken(
+          RestoreTokenManager::GetInstance().TakeToken(selected_source_id_));
+    }
+  }
+
+  is_portal_open_ = true;
+  portal_->Start();
+}
+
+void BaseCapturerPipeWire::CaptureFrame() {
+  TRACE_EVENT0("webrtc", "BaseCapturerPipeWire::CaptureFrame");
+  if (capturer_failed_) {
+    // This could be recoverable if the source list is re-summoned; but for our
+    // purposes this is fine, since it requires intervention to resolve and
+    // essentially starts a new capture.
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
+  std::unique_ptr<DesktopFrame> frame =
+      options_.screencast_stream()->CaptureFrame();
+
+  if (!frame || !frame->data()) {
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+
+  // TODO(julien.isorce): http://crbug.com/945468. Set the icc profile on
+  // the frame, see ScreenCapturerX11::CaptureFrame.
+
+  frame->set_capturer_id(DesktopCapturerId::kWaylandCapturerLinux);
+  frame->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
+                             rtc::kNumNanosecsPerMillisec);
+  callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
+}
+
+bool BaseCapturerPipeWire::GetSourceList(SourceList* sources) {
+  RTC_DCHECK(sources->size() == 0);
+  // List of available screens is already presented by the xdg-desktop-portal,
+  // so we just need a (valid) source id for any callers to pass around, even
+  // though it doesn't mean anything to us. Until the user selects a source in
+  // xdg-desktop-portal we'll just end up returning empty frames. Note that "0"
+  // is often treated as a null/placeholder id, so we shouldn't use that.
+  // TODO(https://crbug.com/1297671): Reconsider type of ID when plumbing
+  // token that will enable stream re-use.
+  sources->push_back({source_id_});
+  return true;
+}
+
+bool BaseCapturerPipeWire::SelectSource(SourceId id) {
+  // Screen selection is handled by the xdg-desktop-portal.
+  selected_source_id_ = id;
+  return true;
+}
+
+DelegatedSourceListController*
+BaseCapturerPipeWire::GetDelegatedSourceListController() {
+  return this;
+}
+
+void BaseCapturerPipeWire::Observe(Observer* observer) {
+  RTC_DCHECK(!delegated_source_list_observer_ || !observer);
+  delegated_source_list_observer_ = observer;
+}
+
+void BaseCapturerPipeWire::EnsureVisible() {
+  RTC_DCHECK(callback_);
+  if (is_portal_open_)
+    return;
+
+  // Clear any previously selected state/capture
+  portal_->Stop();
+  options_.screencast_stream()->StopScreenCastStream();
+
+  // Get a new source id to reflect that the source has changed.
+  source_id_ = RestoreTokenManager::GetInstance().GetUnusedId();
+
+  is_portal_open_ = true;
+  portal_->Start();
+}
+
+void BaseCapturerPipeWire::EnsureHidden() {
+  if (!is_portal_open_)
+    return;
+
+  is_portal_open_ = false;
+  portal_->Stop();
+}
+
+SessionDetails BaseCapturerPipeWire::GetSessionDetails() {
+  return portal_->GetSessionDetails();
+}
+
+ScreenCastPortal* BaseCapturerPipeWire::GetScreenCastPortal() {
+  return is_screencast_portal_ ? static_cast<ScreenCastPortal*>(portal_.get())
+                               : nullptr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.h
new file mode 100644
index 0000000000..4b5cdc4a65
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_BASE_CAPTURER_PIPEWIRE_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_BASE_CAPTURER_PIPEWIRE_H_
+
+#include "modules/desktop_capture/delegated_source_list_controller.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/linux/wayland/screen_capture_portal_interface.h"
+#include "modules/desktop_capture/linux/wayland/screencast_portal.h"
+#include "modules/desktop_capture/linux/wayland/shared_screencast_stream.h"
+#include "modules/portal/portal_request_response.h"
+#include "modules/portal/xdg_desktop_portal_utils.h"
+#include "modules/portal/xdg_session_details.h"
+
+namespace webrtc {
+
+class RTC_EXPORT BaseCapturerPipeWire
+    : public DesktopCapturer,
+      public DelegatedSourceListController,
+      public ScreenCastPortal::PortalNotifier {
+ public:
+  // Returns whether or not the current system can support capture via PipeWire.
+  // This will only be true on Wayland systems that also have PipeWire
+  // available, and thus may require dlopening PipeWire to determine if it is
+  // available.
+  static bool IsSupported();
+
+  BaseCapturerPipeWire(const DesktopCaptureOptions& options, CaptureType type);
+  BaseCapturerPipeWire(
+      const DesktopCaptureOptions& options,
+      std::unique_ptr<xdg_portal::ScreenCapturePortalInterface> portal);
+  ~BaseCapturerPipeWire() override;
+
+  BaseCapturerPipeWire(const BaseCapturerPipeWire&) = delete;
+  BaseCapturerPipeWire& operator=(const BaseCapturerPipeWire&) = delete;
+
+  // DesktopCapturer interface.
+  void Start(Callback* delegate) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  DelegatedSourceListController* GetDelegatedSourceListController() override;
+
+  // DelegatedSourceListController
+  void Observe(Observer* observer) override;
+  void EnsureVisible() override;
+  void EnsureHidden() override;
+
+  // ScreenCastPortal::PortalNotifier interface.
+  void OnScreenCastRequestResult(xdg_portal::RequestResponse result,
+                                 uint32_t stream_node_id,
+                                 int fd) override;
+  void OnScreenCastSessionClosed() override;
+  void UpdateResolution(uint32_t width, uint32_t height) override;
+
+  xdg_portal::SessionDetails GetSessionDetails();
+
+ private:
+  ScreenCastPortal* GetScreenCastPortal();
+
+  DesktopCaptureOptions options_ = {};
+  Callback* callback_ = nullptr;
+  bool capturer_failed_ = false;
+  bool is_screencast_portal_ = false;
+  bool is_portal_open_ = false;
+
+  Observer* delegated_source_list_observer_ = nullptr;
+
+  // SourceId that is selected using SelectSource() and that we previously
+  // returned in GetSourceList(). This should be a SourceId that has a restore
+  // token associated with it and can be restored if we have required version
+  // of xdg-desktop-portal.
+  SourceId selected_source_id_ = 0;
+  // SourceID we randomly generate and that is returned in GetSourceList() as
+  // available source that will later get assigned to a restore token in order
+  // to be restored later using SelectSource().
+  SourceId source_id_ = 0;
+  std::unique_ptr<xdg_portal::ScreenCapturePortalInterface> portal_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_BASE_CAPTURER_PIPEWIRE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
new file mode 100644
index 0000000000..6a019c64b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
@@ -0,0 +1,781 @@
+/*
+ *  Copyright 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/wayland/egl_dmabuf.h"
+
+#include <asm/ioctl.h>
+#include <dlfcn.h>
+#include <fcntl.h>
+#include <gdk/gdk.h>
+#include <libdrm/drm_fourcc.h>
+#include <linux/types.h>
+#include <spa/param/video/format-utils.h>
+#include <unistd.h>
+#include <xf86drm.h>
+
+#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/sanitizer.h"
+#include "rtc_base/string_encode.h"
+
+namespace webrtc {
+
+// EGL
+typedef EGLBoolean (*eglBindAPI_func)(EGLenum api);
+typedef EGLContext (*eglCreateContext_func)(EGLDisplay dpy,
+                                            EGLConfig config,
+                                            EGLContext share_context,
+                                            const EGLint* attrib_list);
+typedef EGLBoolean (*eglDestroyContext_func)(EGLDisplay display,
+                                             EGLContext context);
+typedef EGLBoolean (*eglTerminate_func)(EGLDisplay display);
+typedef EGLImageKHR (*eglCreateImageKHR_func)(EGLDisplay dpy,
+                                              EGLContext ctx,
+                                              EGLenum target,
+                                              EGLClientBuffer buffer,
+                                              const EGLint* attrib_list);
+typedef EGLBoolean (*eglDestroyImageKHR_func)(EGLDisplay dpy,
+                                              EGLImageKHR image);
+typedef EGLint (*eglGetError_func)(void);
+typedef void* (*eglGetProcAddress_func)(const char*);
+typedef EGLDisplay (*eglGetPlatformDisplayEXT_func)(EGLenum platform,
+                                                    void* native_display,
+                                                    const EGLint* attrib_list);
+typedef EGLDisplay (*eglGetPlatformDisplay_func)(EGLenum platform,
+                                                 void* native_display,
+                                                 const EGLAttrib* attrib_list);
+
+typedef EGLBoolean (*eglInitialize_func)(EGLDisplay dpy,
+                                         EGLint* major,
+                                         EGLint* minor);
+typedef EGLBoolean (*eglMakeCurrent_func)(EGLDisplay dpy,
+                                          EGLSurface draw,
+                                          EGLSurface read,
+                                          EGLContext ctx);
+typedef EGLBoolean (*eglQueryDmaBufFormatsEXT_func)(EGLDisplay dpy,
+                                                    EGLint max_formats,
+                                                    EGLint* formats,
+                                                    EGLint* num_formats);
+typedef EGLBoolean (*eglQueryDmaBufModifiersEXT_func)(EGLDisplay dpy,
+                                                      EGLint format,
+                                                      EGLint max_modifiers,
+                                                      EGLuint64KHR* modifiers,
+                                                      EGLBoolean* external_only,
+                                                      EGLint* num_modifiers);
+typedef const char* (*eglQueryString_func)(EGLDisplay dpy, EGLint name);
+typedef void (*glEGLImageTargetTexture2DOES_func)(GLenum target,
+                                                  GLeglImageOES image);
+
+// This doesn't follow naming conventions in WebRTC, where the naming
+// should look like e.g. egl_bind_api instead of EglBindAPI, however
+// we named them according to the exported functions they map to for
+// consistency.
+eglBindAPI_func EglBindAPI = nullptr;
+eglCreateContext_func EglCreateContext = nullptr;
+eglDestroyContext_func EglDestroyContext = nullptr;
+eglTerminate_func EglTerminate = nullptr;
+eglCreateImageKHR_func EglCreateImageKHR = nullptr;
+eglDestroyImageKHR_func EglDestroyImageKHR = nullptr;
+eglGetError_func EglGetError = nullptr;
+eglGetProcAddress_func EglGetProcAddress = nullptr;
+eglGetPlatformDisplayEXT_func EglGetPlatformDisplayEXT = nullptr;
+eglGetPlatformDisplay_func EglGetPlatformDisplay = nullptr;
+eglInitialize_func EglInitialize = nullptr;
+eglMakeCurrent_func EglMakeCurrent = nullptr;
+eglQueryDmaBufFormatsEXT_func EglQueryDmaBufFormatsEXT = nullptr;
+eglQueryDmaBufModifiersEXT_func EglQueryDmaBufModifiersEXT = nullptr;
+eglQueryString_func EglQueryString = nullptr;
+glEGLImageTargetTexture2DOES_func GlEGLImageTargetTexture2DOES = nullptr;
+
+// GL
+typedef void (*glBindTexture_func)(GLenum target, GLuint texture);
+typedef void (*glDeleteTextures_func)(GLsizei n, const GLuint* textures);
+typedef void (*glGenTextures_func)(GLsizei n, GLuint* textures);
+typedef GLenum (*glGetError_func)(void);
+typedef const GLubyte* (*glGetString_func)(GLenum name);
+typedef void (*glReadPixels_func)(GLint x,
+                                  GLint y,
+                                  GLsizei width,
+                                  GLsizei height,
+                                  GLenum format,
+                                  GLenum type,
+                                  void* data);
+typedef void (*glGenFramebuffers_func)(GLsizei n, GLuint* ids);
+typedef void (*glDeleteFramebuffers_func)(GLsizei n,
+                                          const GLuint* framebuffers);
+typedef void (*glBindFramebuffer_func)(GLenum target, GLuint framebuffer);
+typedef void (*glFramebufferTexture2D_func)(GLenum target,
+                                            GLenum attachment,
+                                            GLenum textarget,
+                                            GLuint texture,
+                                            GLint level);
+typedef GLenum (*glCheckFramebufferStatus_func)(GLenum target);
+typedef void (*glTexParameteri_func)(GLenum target, GLenum pname, GLint param);
+typedef void* (*glXGetProcAddressARB_func)(const char*);
+
+// This doesn't follow naming conventions in WebRTC, where the naming
+// should look like e.g. egl_bind_api instead of EglBindAPI, however
+// we named them according to the exported functions they map to for
+// consistency.
+glBindTexture_func GlBindTexture = nullptr;
+glDeleteTextures_func GlDeleteTextures = nullptr;
+glGenTextures_func GlGenTextures = nullptr;
+glGetError_func GlGetError = nullptr;
+glGetString_func GlGetString = nullptr;
+glReadPixels_func GlReadPixels = nullptr;
+glGenFramebuffers_func GlGenFramebuffers = nullptr;
+glDeleteFramebuffers_func GlDeleteFramebuffers = nullptr;
+glBindFramebuffer_func GlBindFramebuffer = nullptr;
+glFramebufferTexture2D_func GlFramebufferTexture2D = nullptr;
+glCheckFramebufferStatus_func GlCheckFramebufferStatus = nullptr;
+glTexParameteri_func GlTexParameteri = nullptr;
+glXGetProcAddressARB_func GlXGetProcAddressARB = nullptr;
+
+static const std::string FormatGLError(GLenum err) {
+  switch (err) {
+    case GL_NO_ERROR:
+      return "GL_NO_ERROR";
+    case GL_INVALID_ENUM:
+      return "GL_INVALID_ENUM";
+    case GL_INVALID_VALUE:
+      return "GL_INVALID_VALUE";
+    case GL_INVALID_OPERATION:
+      return "GL_INVALID_OPERATION";
+    case GL_STACK_OVERFLOW:
+      return "GL_STACK_OVERFLOW";
+    case GL_STACK_UNDERFLOW:
+      return "GL_STACK_UNDERFLOW";
+    case GL_OUT_OF_MEMORY:
+      return "GL_OUT_OF_MEMORY";
+    default:
+      return "GL error code: " + std::to_string(err);
+  }
+}
+
+static const std::string FormatEGLError(EGLint err) {
+  switch (err) {
+    case EGL_NOT_INITIALIZED:
+      return "EGL_NOT_INITIALIZED";
+    case EGL_BAD_ACCESS:
+      return "EGL_BAD_ACCESS";
+    case EGL_BAD_ALLOC:
+      return "EGL_BAD_ALLOC";
+    case EGL_BAD_ATTRIBUTE:
+      return "EGL_BAD_ATTRIBUTE";
+    case EGL_BAD_CONTEXT:
+      return "EGL_BAD_CONTEXT";
+    case EGL_BAD_CONFIG:
+      return "EGL_BAD_CONFIG";
+    case EGL_BAD_CURRENT_SURFACE:
+      return "EGL_BAD_CURRENT_SURFACE";
+    case EGL_BAD_DISPLAY:
+      return "EGL_BAD_DISPLAY";
+    case EGL_BAD_SURFACE:
+      return "EGL_BAD_SURFACE";
+    case EGL_BAD_MATCH:
+      return "EGL_BAD_MATCH";
+    case EGL_BAD_PARAMETER:
+      return "EGL_BAD_PARAMETER";
+    case EGL_BAD_NATIVE_PIXMAP:
+      return "EGL_BAD_NATIVE_PIXMAP";
+    case EGL_BAD_NATIVE_WINDOW:
+      return "EGL_BAD_NATIVE_WINDOW";
+    case EGL_CONTEXT_LOST:
+      return "EGL_CONTEXT_LOST";
+    default:
+      return "EGL error code: " + std::to_string(err);
+  }
+}
+
+static uint32_t SpaPixelFormatToDrmFormat(uint32_t spa_format) {
+  switch (spa_format) {
+    case SPA_VIDEO_FORMAT_RGBA:
+      return DRM_FORMAT_ABGR8888;
+    case SPA_VIDEO_FORMAT_RGBx:
+      return DRM_FORMAT_XBGR8888;
+    case SPA_VIDEO_FORMAT_BGRA:
+      return DRM_FORMAT_ARGB8888;
+    case SPA_VIDEO_FORMAT_BGRx:
+      return DRM_FORMAT_XRGB8888;
+    default:
+      return DRM_FORMAT_INVALID;
+  }
+}
+
+static void CloseLibrary(void* library) {
+  if (library) {
+    dlclose(library);
+    library = nullptr;
+  }
+}
+
+static bool IsWaylandDisplay() {
+  static auto sGdkWaylandDisplayGetType =
+    (GType (*)(void))dlsym(RTLD_DEFAULT, "gdk_wayland_display_get_type");
+  if (!sGdkWaylandDisplayGetType) {
+    return false;
+  }
+  return (G_TYPE_CHECK_INSTANCE_TYPE ((gdk_display_get_default()),
+                                      sGdkWaylandDisplayGetType()));
+}
+
+static bool IsX11Display() {
+  static auto sGdkX11DisplayGetType =
+    (GType (*)(void))dlsym(RTLD_DEFAULT, "gdk_x11_display_get_type");
+  if (!sGdkX11DisplayGetType) {
+    return false;
+  }
+  return (G_TYPE_CHECK_INSTANCE_TYPE ((gdk_display_get_default()),
+                                      sGdkX11DisplayGetType()));
+}
+
+static void* g_lib_egl = nullptr;
+
+RTC_NO_SANITIZE("cfi-icall")
+static bool OpenEGL() {
+  g_lib_egl = dlopen("libEGL.so.1", RTLD_NOW | RTLD_GLOBAL);
+  if (g_lib_egl) {
+    EglGetProcAddress =
+        (eglGetProcAddress_func)dlsym(g_lib_egl, "eglGetProcAddress");
+    return EglGetProcAddress;
+  }
+
+  return false;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+static bool LoadEGL() {
+  if (OpenEGL()) {
+    EglBindAPI = (eglBindAPI_func)EglGetProcAddress("eglBindAPI");
+    EglCreateContext =
+        (eglCreateContext_func)EglGetProcAddress("eglCreateContext");
+    EglDestroyContext =
+        (eglDestroyContext_func)EglGetProcAddress("eglDestroyContext");
+    EglTerminate = (eglTerminate_func)EglGetProcAddress("eglTerminate");
+    EglCreateImageKHR =
+        (eglCreateImageKHR_func)EglGetProcAddress("eglCreateImageKHR");
+    EglDestroyImageKHR =
+        (eglDestroyImageKHR_func)EglGetProcAddress("eglDestroyImageKHR");
+    EglGetError = (eglGetError_func)EglGetProcAddress("eglGetError");
+    EglGetPlatformDisplayEXT = (eglGetPlatformDisplayEXT_func)EglGetProcAddress(
+        "eglGetPlatformDisplayEXT");
+    EglGetPlatformDisplay =
+        (eglGetPlatformDisplay_func)EglGetProcAddress("eglGetPlatformDisplay");
+    EglInitialize = (eglInitialize_func)EglGetProcAddress("eglInitialize");
+    EglMakeCurrent = (eglMakeCurrent_func)EglGetProcAddress("eglMakeCurrent");
+    EglQueryString = (eglQueryString_func)EglGetProcAddress("eglQueryString");
+    GlEGLImageTargetTexture2DOES =
+        (glEGLImageTargetTexture2DOES_func)EglGetProcAddress(
+            "glEGLImageTargetTexture2DOES");
+
+    return EglBindAPI && EglCreateContext && EglCreateImageKHR &&
+           EglTerminate && EglDestroyContext && EglDestroyImageKHR &&
+           EglGetError && EglGetPlatformDisplayEXT && EglGetPlatformDisplay &&
+           EglInitialize && EglMakeCurrent && EglQueryString &&
+           GlEGLImageTargetTexture2DOES;
+  }
+
+  return false;
+}
+
+static void* g_lib_gl = nullptr;
+
+RTC_NO_SANITIZE("cfi-icall")
+static bool OpenGL() {
+  std::vector<std::string> names = {"libGL.so.1", "libGL.so"};
+  for (const std::string& name : names) {
+    g_lib_gl = dlopen(name.c_str(), RTLD_NOW | RTLD_GLOBAL);
+    if (g_lib_gl) {
+      GlXGetProcAddressARB =
+          (glXGetProcAddressARB_func)dlsym(g_lib_gl, "glXGetProcAddressARB");
+      return GlXGetProcAddressARB;
+    }
+  }
+
+  return false;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+static bool LoadGL() {
+  if (OpenGL()) {
+    GlGetString = (glGetString_func)GlXGetProcAddressARB("glGetString");
+    if (!GlGetString) {
+      return false;
+    }
+
+    GlBindTexture = (glBindTexture_func)GlXGetProcAddressARB("glBindTexture");
+    GlDeleteTextures =
+        (glDeleteTextures_func)GlXGetProcAddressARB("glDeleteTextures");
+    GlGenTextures = (glGenTextures_func)GlXGetProcAddressARB("glGenTextures");
+    GlGetError = (glGetError_func)GlXGetProcAddressARB("glGetError");
+    GlReadPixels = (glReadPixels_func)GlXGetProcAddressARB("glReadPixels");
+    GlGenFramebuffers =
+        (glGenFramebuffers_func)GlXGetProcAddressARB("glGenFramebuffers");
+    GlDeleteFramebuffers =
+        (glDeleteFramebuffers_func)GlXGetProcAddressARB("glDeleteFramebuffers");
+    GlBindFramebuffer =
+        (glBindFramebuffer_func)GlXGetProcAddressARB("glBindFramebuffer");
+    GlFramebufferTexture2D = (glFramebufferTexture2D_func)GlXGetProcAddressARB(
+        "glFramebufferTexture2D");
+    GlCheckFramebufferStatus =
+        (glCheckFramebufferStatus_func)GlXGetProcAddressARB(
+            "glCheckFramebufferStatus");
+
+    GlTexParameteri =
+        (glTexParameteri_func)GlXGetProcAddressARB("glTexParameteri");
+
+    return GlBindTexture && GlDeleteTextures && GlGenTextures && GlGetError &&
+           GlReadPixels && GlGenFramebuffers && GlDeleteFramebuffers &&
+           GlBindFramebuffer && GlFramebufferTexture2D &&
+           GlCheckFramebufferStatus && GlTexParameteri;
+  }
+
+  return false;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+EglDmaBuf::EglDmaBuf() {
+  if (!LoadEGL()) {
+    RTC_LOG(LS_ERROR) << "Unable to load EGL entry functions.";
+    CloseLibrary(g_lib_egl);
+    return;
+  }
+
+  if (!LoadGL()) {
+    RTC_LOG(LS_ERROR) << "Failed to load OpenGL entry functions.";
+    CloseLibrary(g_lib_gl);
+    return;
+  }
+
+  if (!GetClientExtensions(EGL_NO_DISPLAY, EGL_EXTENSIONS)) {
+    return;
+  }
+
+  bool has_platform_base_ext = false;
+  bool has_platform_gbm_ext = false;
+  bool has_khr_platform_gbm_ext = false;
+
+  for (const auto& extension : egl_.extensions) {
+    if (extension == "EGL_EXT_platform_base") {
+      has_platform_base_ext = true;
+      continue;
+    } else if (extension == "EGL_MESA_platform_gbm") {
+      has_platform_gbm_ext = true;
+      continue;
+    } else if (extension == "EGL_KHR_platform_gbm") {
+      has_khr_platform_gbm_ext = true;
+      continue;
+    }
+  }
+
+  if (!has_platform_base_ext || !has_platform_gbm_ext ||
+      !has_khr_platform_gbm_ext) {
+    RTC_LOG(LS_ERROR) << "One of required EGL extensions is missing";
+    return;
+  }
+
+  if (IsWaylandDisplay()) {
+    egl_.display = EglGetPlatformDisplay(EGL_PLATFORM_WAYLAND_KHR,
+                                         (void*)EGL_DEFAULT_DISPLAY, nullptr);
+  } else if (IsX11Display()) {
+    egl_.display = EglGetPlatformDisplay(EGL_PLATFORM_X11_KHR,
+                                         (void*)EGL_DEFAULT_DISPLAY, nullptr);
+  }
+
+  if (egl_.display == EGL_NO_DISPLAY) {
+    RTC_LOG(LS_ERROR) << "Failed to obtain default EGL display: "
+                      << FormatEGLError(EglGetError()) << "\n"
+                      << "Defaulting to using first available render node";
+    absl::optional<std::string> render_node = GetRenderNode();
+    if (!render_node) {
+      return;
+    }
+
+    drm_fd_ = open(render_node->c_str(), O_RDWR);
+
+    if (drm_fd_ < 0) {
+      RTC_LOG(LS_ERROR) << "Failed to open drm render node: "
+                        << strerror(errno);
+      return;
+    }
+
+    gbm_device_ = gbm_create_device(drm_fd_);
+
+    if (!gbm_device_) {
+      RTC_LOG(LS_ERROR) << "Cannot create GBM device: " << strerror(errno);
+      close(drm_fd_);
+      return;
+    }
+
+    // Use eglGetPlatformDisplayEXT() to get the display pointer
+    // if the implementation supports it.
+    egl_.display =
+        EglGetPlatformDisplayEXT(EGL_PLATFORM_GBM_KHR, gbm_device_, nullptr);
+  }
+
+  if (egl_.display == EGL_NO_DISPLAY) {
+    RTC_LOG(LS_ERROR) << "Error during obtaining EGL display: "
+                      << FormatEGLError(EglGetError());
+    return;
+  }
+
+  EGLint major, minor;
+  if (EglInitialize(egl_.display, &major, &minor) == EGL_FALSE) {
+    RTC_LOG(LS_ERROR) << "Error during eglInitialize: "
+                      << FormatEGLError(EglGetError());
+    return;
+  }
+
+  if (EglBindAPI(EGL_OPENGL_API) == EGL_FALSE) {
+    RTC_LOG(LS_ERROR) << "bind OpenGL API failed";
+    return;
+  }
+
+  egl_.context =
+      EglCreateContext(egl_.display, nullptr, EGL_NO_CONTEXT, nullptr);
+
+  if (egl_.context == EGL_NO_CONTEXT) {
+    RTC_LOG(LS_ERROR) << "Couldn't create EGL context: "
+                      << FormatGLError(EglGetError());
+    return;
+  }
+
+  if (!GetClientExtensions(egl_.display, EGL_EXTENSIONS)) {
+    return;
+  }
+
+  bool has_image_dma_buf_import_modifiers_ext = false;
+
+  for (const auto& extension : egl_.extensions) {
+    if (extension == "EGL_EXT_image_dma_buf_import") {
+      has_image_dma_buf_import_ext_ = true;
+      continue;
+    } else if (extension == "EGL_EXT_image_dma_buf_import_modifiers") {
+      has_image_dma_buf_import_modifiers_ext = true;
+      continue;
+    }
+  }
+
+  if (has_image_dma_buf_import_ext_ && has_image_dma_buf_import_modifiers_ext) {
+    EglQueryDmaBufFormatsEXT = (eglQueryDmaBufFormatsEXT_func)EglGetProcAddress(
+        "eglQueryDmaBufFormatsEXT");
+    EglQueryDmaBufModifiersEXT =
+        (eglQueryDmaBufModifiersEXT_func)EglGetProcAddress(
+            "eglQueryDmaBufModifiersEXT");
+  }
+
+  RTC_LOG(LS_INFO) << "Egl initialization succeeded";
+  egl_initialized_ = true;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+EglDmaBuf::~EglDmaBuf() {
+  if (gbm_device_) {
+    gbm_device_destroy(gbm_device_);
+    close(drm_fd_);
+  }
+
+  if (egl_.context != EGL_NO_CONTEXT) {
+    EglDestroyContext(egl_.display, egl_.context);
+  }
+
+  if (egl_.display != EGL_NO_DISPLAY) {
+    EglTerminate(egl_.display);
+  }
+
+  if (fbo_) {
+    GlDeleteFramebuffers(1, &fbo_);
+  }
+
+  if (texture_) {
+    GlDeleteTextures(1, &texture_);
+  }
+
+  // BUG: crbug.com/1290566
+  // Closing libEGL.so.1 when using NVidia drivers causes a crash
+  // when EglGetPlatformDisplayEXT() is used, at least this one is enough
+  // to be called to make it crash.
+  // It also looks that libepoxy and glad don't dlclose it either
+  // CloseLibrary(g_lib_egl);
+  // CloseLibrary(g_lib_gl);
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+bool EglDmaBuf::GetClientExtensions(EGLDisplay dpy, EGLint name) {
+  // Get the list of client extensions
+  const char* client_extensions_cstring = EglQueryString(dpy, name);
+  if (!client_extensions_cstring) {
+    // If eglQueryString() returned NULL, the implementation doesn't support
+    // EGL_EXT_client_extensions. Expect an EGL_BAD_DISPLAY error.
+    RTC_LOG(LS_ERROR) << "No client extensions defined! "
+                      << FormatEGLError(EglGetError());
+    return false;
+  }
+
+  std::vector<absl::string_view> client_extensions =
+      rtc::split(client_extensions_cstring, ' ');
+  for (const auto& extension : client_extensions) {
+    egl_.extensions.push_back(std::string(extension));
+  }
+
+  return true;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+bool EglDmaBuf::ImageFromDmaBuf(const DesktopSize& size,
+                                uint32_t format,
+                                const std::vector<PlaneData>& plane_datas,
+                                uint64_t modifier,
+                                const DesktopVector& offset,
+                                const DesktopSize& buffer_size,
+                                uint8_t* data) {
+  if (!egl_initialized_) {
+    return false;
+  }
+
+  if (plane_datas.size() <= 0) {
+    RTC_LOG(LS_ERROR) << "Failed to process buffer: invalid number of planes";
+    return false;
+  }
+
+  EGLint attribs[47];
+  int atti = 0;
+
+  attribs[atti++] = EGL_WIDTH;
+  attribs[atti++] = static_cast<EGLint>(size.width());
+  attribs[atti++] = EGL_HEIGHT;
+  attribs[atti++] = static_cast<EGLint>(size.height());
+  attribs[atti++] = EGL_LINUX_DRM_FOURCC_EXT;
+  attribs[atti++] = SpaPixelFormatToDrmFormat(format);
+
+  if (plane_datas.size() > 0) {
+    attribs[atti++] = EGL_DMA_BUF_PLANE0_FD_EXT;
+    attribs[atti++] = plane_datas[0].fd;
+    attribs[atti++] = EGL_DMA_BUF_PLANE0_OFFSET_EXT;
+    attribs[atti++] = plane_datas[0].offset;
+    attribs[atti++] = EGL_DMA_BUF_PLANE0_PITCH_EXT;
+    attribs[atti++] = plane_datas[0].stride;
+
+    if (modifier != DRM_FORMAT_MOD_INVALID) {
+      attribs[atti++] = EGL_DMA_BUF_PLANE0_MODIFIER_LO_EXT;
+      attribs[atti++] = modifier & 0xFFFFFFFF;
+      attribs[atti++] = EGL_DMA_BUF_PLANE0_MODIFIER_HI_EXT;
+      attribs[atti++] = modifier >> 32;
+    }
+  }
+
+  if (plane_datas.size() > 1) {
+    attribs[atti++] = EGL_DMA_BUF_PLANE1_FD_EXT;
+    attribs[atti++] = plane_datas[1].fd;
+    attribs[atti++] = EGL_DMA_BUF_PLANE1_OFFSET_EXT;
+    attribs[atti++] = plane_datas[1].offset;
+    attribs[atti++] = EGL_DMA_BUF_PLANE1_PITCH_EXT;
+    attribs[atti++] = plane_datas[1].stride;
+
+    if (modifier != DRM_FORMAT_MOD_INVALID) {
+      attribs[atti++] = EGL_DMA_BUF_PLANE1_MODIFIER_LO_EXT;
+      attribs[atti++] = modifier & 0xFFFFFFFF;
+      attribs[atti++] = EGL_DMA_BUF_PLANE1_MODIFIER_HI_EXT;
+      attribs[atti++] = modifier >> 32;
+    }
+  }
+
+  if (plane_datas.size() > 2) {
+    attribs[atti++] = EGL_DMA_BUF_PLANE2_FD_EXT;
+    attribs[atti++] = plane_datas[2].fd;
+    attribs[atti++] = EGL_DMA_BUF_PLANE2_OFFSET_EXT;
+    attribs[atti++] = plane_datas[2].offset;
+    attribs[atti++] = EGL_DMA_BUF_PLANE2_PITCH_EXT;
+    attribs[atti++] = plane_datas[2].stride;
+
+    if (modifier != DRM_FORMAT_MOD_INVALID) {
+      attribs[atti++] = EGL_DMA_BUF_PLANE2_MODIFIER_LO_EXT;
+      attribs[atti++] = modifier & 0xFFFFFFFF;
+      attribs[atti++] = EGL_DMA_BUF_PLANE2_MODIFIER_HI_EXT;
+      attribs[atti++] = modifier >> 32;
+    }
+  }
+
+  if (plane_datas.size() > 3) {
+    attribs[atti++] = EGL_DMA_BUF_PLANE3_FD_EXT;
+    attribs[atti++] = plane_datas[3].fd;
+    attribs[atti++] = EGL_DMA_BUF_PLANE3_OFFSET_EXT;
+    attribs[atti++] = plane_datas[3].offset;
+    attribs[atti++] = EGL_DMA_BUF_PLANE3_PITCH_EXT;
+    attribs[atti++] = plane_datas[3].stride;
+
+    if (modifier != DRM_FORMAT_MOD_INVALID) {
+      attribs[atti++] = EGL_DMA_BUF_PLANE3_MODIFIER_LO_EXT;
+      attribs[atti++] = modifier & 0xFFFFFFFF;
+      attribs[atti++] = EGL_DMA_BUF_PLANE3_MODIFIER_HI_EXT;
+      attribs[atti++] = modifier >> 32;
+    }
+  }
+
+  attribs[atti++] = EGL_NONE;
+
+  // bind context to render thread
+  EglMakeCurrent(egl_.display, EGL_NO_SURFACE, EGL_NO_SURFACE, egl_.context);
+
+  // create EGL image from attribute list
+  EGLImageKHR image = EglCreateImageKHR(
+      egl_.display, EGL_NO_CONTEXT, EGL_LINUX_DMA_BUF_EXT, nullptr, attribs);
+
+  if (image == EGL_NO_IMAGE) {
+    RTC_LOG(LS_ERROR) << "Failed to record frame: Error creating EGLImage - "
+                      << FormatEGLError(EglGetError());
+    return false;
+  }
+
+  // create GL 2D texture for framebuffer
+  if (!texture_) {
+    GlGenTextures(1, &texture_);
+    GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
+    GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_NEAREST);
+    GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+    GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+  }
+  GlBindTexture(GL_TEXTURE_2D, texture_);
+  GlEGLImageTargetTexture2DOES(GL_TEXTURE_2D, image);
+
+  if (!fbo_) {
+    GlGenFramebuffers(1, &fbo_);
+  }
+
+  GlBindFramebuffer(GL_FRAMEBUFFER, fbo_);
+  GlFramebufferTexture2D(GL_FRAMEBUFFER, GL_COLOR_ATTACHMENT0, GL_TEXTURE_2D,
+                         texture_, 0);
+  if (GlCheckFramebufferStatus(GL_FRAMEBUFFER) != GL_FRAMEBUFFER_COMPLETE) {
+    RTC_LOG(LS_ERROR) << "Failed to bind DMA buf framebuffer";
+    EglDestroyImageKHR(egl_.display, image);
+    return false;
+  }
+
+  GLenum gl_format = GL_BGRA;
+  switch (format) {
+    case SPA_VIDEO_FORMAT_RGBx:
+      gl_format = GL_RGBA;
+      break;
+    case SPA_VIDEO_FORMAT_RGBA:
+      gl_format = GL_RGBA;
+      break;
+    case SPA_VIDEO_FORMAT_BGRx:
+      gl_format = GL_BGRA;
+      break;
+    default:
+      gl_format = GL_BGRA;
+      break;
+  }
+
+  GlReadPixels(offset.x(), offset.y(), buffer_size.width(),
+               buffer_size.height(), gl_format, GL_UNSIGNED_BYTE, data);
+
+  const GLenum error = GlGetError();
+  if (error) {
+    RTC_LOG(LS_ERROR) << "Failed to get image from DMA buffer.";
+  }
+
+  EglDestroyImageKHR(egl_.display, image);
+
+  return !error;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+std::vector<uint64_t> EglDmaBuf::QueryDmaBufModifiers(uint32_t format) {
+  if (!egl_initialized_) {
+    return {};
+  }
+
+  // Explicit modifiers not supported, return just DRM_FORMAT_MOD_INVALID as we
+  // can still use modifier-less DMA-BUFs if we have required extension
+  if (EglQueryDmaBufFormatsEXT == nullptr ||
+      EglQueryDmaBufModifiersEXT == nullptr) {
+    return has_image_dma_buf_import_ext_
+               ? std::vector<uint64_t>{DRM_FORMAT_MOD_INVALID}
+               : std::vector<uint64_t>{};
+  }
+
+  uint32_t drm_format = SpaPixelFormatToDrmFormat(format);
+  // Should never happen as it's us who controls the list of supported formats
+  RTC_DCHECK(drm_format != DRM_FORMAT_INVALID);
+
+  EGLint count = 0;
+  EGLBoolean success =
+      EglQueryDmaBufFormatsEXT(egl_.display, 0, nullptr, &count);
+
+  if (!success || !count) {
+    RTC_LOG(LS_WARNING) << "Cannot query the number of formats.";
+    return {DRM_FORMAT_MOD_INVALID};
+  }
+
+  std::vector<uint32_t> formats(count);
+  if (!EglQueryDmaBufFormatsEXT(egl_.display, count,
+                                reinterpret_cast<EGLint*>(formats.data()),
+                                &count)) {
+    RTC_LOG(LS_WARNING) << "Cannot query a list of formats.";
+    return {DRM_FORMAT_MOD_INVALID};
+  }
+
+  if (std::find(formats.begin(), formats.end(), drm_format) == formats.end()) {
+    RTC_LOG(LS_WARNING) << "Format " << drm_format
+                        << " not supported for modifiers.";
+    return {DRM_FORMAT_MOD_INVALID};
+  }
+
+  success = EglQueryDmaBufModifiersEXT(egl_.display, drm_format, 0, nullptr,
+                                       nullptr, &count);
+
+  if (!success || !count) {
+    RTC_LOG(LS_WARNING) << "Cannot query the number of modifiers.";
+    return {DRM_FORMAT_MOD_INVALID};
+  }
+
+  std::vector<uint64_t> modifiers(count);
+  if (!EglQueryDmaBufModifiersEXT(egl_.display, drm_format, count,
+                                  modifiers.data(), nullptr, &count)) {
+    RTC_LOG(LS_WARNING) << "Cannot query a list of modifiers.";
+  }
+
+  // Support modifier-less buffers
+  modifiers.push_back(DRM_FORMAT_MOD_INVALID);
+  return modifiers;
+}
+
+absl::optional<std::string> EglDmaBuf::GetRenderNode() {
+  int max_devices = drmGetDevices2(0, nullptr, 0);
+  if (max_devices <= 0) {
+    RTC_LOG(LS_ERROR) << "drmGetDevices2() has not found any devices (errno="
+                      << -max_devices << ")";
+    return absl::nullopt;
+  }
+
+  std::vector<drmDevicePtr> devices(max_devices);
+  int ret = drmGetDevices2(0, devices.data(), max_devices);
+  if (ret < 0) {
+    RTC_LOG(LS_ERROR) << "drmGetDevices2() returned an error " << ret;
+    return absl::nullopt;
+  }
+
+  std::string render_node;
+
+  for (const drmDevicePtr& device : devices) {
+    if (device->available_nodes & (1 << DRM_NODE_RENDER)) {
+      render_node = device->nodes[DRM_NODE_RENDER];
+      break;
+    }
+  }
+
+  drmFreeDevices(devices.data(), ret);
+  return render_node;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.h
new file mode 100644
index 0000000000..22a8f5ab52
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/egl_dmabuf.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_EGL_DMABUF_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_EGL_DMABUF_H_
+
+#include <epoxy/egl.h>
+#include <epoxy/gl.h>
+#include <gbm.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+
+namespace webrtc {
+
+class EglDmaBuf {
+ public:
+  struct EGLStruct {
+    std::vector<std::string> extensions;
+    EGLDisplay display = EGL_NO_DISPLAY;
+    EGLContext context = EGL_NO_CONTEXT;
+  };
+
+  struct PlaneData {
+    int32_t fd;
+    uint32_t stride;
+    uint32_t offset;
+  };
+
+  EglDmaBuf();
+  ~EglDmaBuf();
+
+  // Returns whether the image was successfully imported from
+  // given DmaBuf and its parameters
+  bool ImageFromDmaBuf(const DesktopSize& size,
+                       uint32_t format,
+                       const std::vector<PlaneData>& plane_datas,
+                       uint64_t modifiers,
+                       const DesktopVector& offset,
+                       const DesktopSize& buffer_size,
+                       uint8_t* data);
+  std::vector<uint64_t> QueryDmaBufModifiers(uint32_t format);
+
+  bool IsEglInitialized() const { return egl_initialized_; }
+
+ private:
+  bool GetClientExtensions(EGLDisplay dpy, EGLint name);
+
+  bool egl_initialized_ = false;
+  bool has_image_dma_buf_import_ext_ = false;
+  int32_t drm_fd_ = -1;               // for GBM buffer mmap
+  gbm_device* gbm_device_ = nullptr;  // for passed GBM buffer retrieval
+
+  GLuint fbo_ = 0;
+  GLuint texture_ = 0;
+  EGLStruct egl_;
+
+  absl::optional<std::string> GetRenderNode();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_EGL_DMABUF_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc
new file mode 100644
index 0000000000..3d33b0fbb8
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.h"
+
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+MouseCursorMonitorPipeWire::MouseCursorMonitorPipeWire(
+    const DesktopCaptureOptions& options)
+    : options_(options) {
+  sequence_checker_.Detach();
+}
+
+MouseCursorMonitorPipeWire::~MouseCursorMonitorPipeWire() {}
+
+void MouseCursorMonitorPipeWire::Init(Callback* callback, Mode mode) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+
+  callback_ = callback;
+  mode_ = mode;
+}
+
+void MouseCursorMonitorPipeWire::Capture() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  RTC_DCHECK(callback_);
+
+  std::unique_ptr<MouseCursor> mouse_cursor =
+      options_.screencast_stream()->CaptureCursor();
+
+  if (mouse_cursor && mouse_cursor->image()->data()) {
+    callback_->OnMouseCursor(mouse_cursor.release());
+  }
+
+  if (mode_ == SHAPE_AND_POSITION) {
+    absl::optional<DesktopVector> mouse_cursor_position =
+        options_.screencast_stream()->CaptureCursorPosition();
+    if (mouse_cursor_position) {
+      callback_->OnMouseCursorPosition(mouse_cursor_position.value());
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.h
new file mode 100644
index 0000000000..da670bece9
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_MOUSE_CURSOR_MONITOR_PIPEWIRE_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_MOUSE_CURSOR_MONITOR_PIPEWIRE_H_
+
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/linux/wayland/shared_screencast_stream.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace webrtc {
+
+class MouseCursorMonitorPipeWire : public MouseCursorMonitor {
+ public:
+  explicit MouseCursorMonitorPipeWire(const DesktopCaptureOptions& options);
+  ~MouseCursorMonitorPipeWire() override;
+
+  // MouseCursorMonitor:
+  void Init(Callback* callback, Mode mode) override;
+  void Capture() override;
+
+  DesktopCaptureOptions options_ RTC_GUARDED_BY(sequence_checker_);
+  Callback* callback_ RTC_GUARDED_BY(sequence_checker_) = nullptr;
+  Mode mode_ RTC_GUARDED_BY(sequence_checker_) = SHAPE_AND_POSITION;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_MOUSE_CURSOR_MONITOR_PIPEWIRE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/portal_request_response.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/portal_request_response.h
new file mode 100644
index 0000000000..2589479347
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/portal_request_response.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_PORTAL_REQUEST_RESPONSE_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_PORTAL_REQUEST_RESPONSE_H_
+
+// TODO(bugs.webrtc.org/14187): remove when all users are gone
+#include "modules/portal/portal_request_response.h"
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_PORTAL_REQUEST_RESPONSE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.cc
new file mode 100644
index 0000000000..5ca9b957a9
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.cc
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/wayland/restore_token_manager.h"
+
+namespace webrtc {
+
+// static
+RestoreTokenManager& RestoreTokenManager::GetInstance() {
+  static webrtc::RestoreTokenManager* manager = new RestoreTokenManager();
+  return *manager;
+}
+
+void RestoreTokenManager::AddToken(DesktopCapturer::SourceId id,
+                                   const std::string& token) {
+  restore_tokens_.insert({id, token});
+}
+
+std::string RestoreTokenManager::TakeToken(DesktopCapturer::SourceId id) {
+  std::string token = restore_tokens_[id];
+  // Remove the token as it cannot be used anymore
+  restore_tokens_.erase(id);
+  return token;
+}
+
+DesktopCapturer::SourceId RestoreTokenManager::GetUnusedId() {
+  return ++last_source_id_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.h
new file mode 100644
index 0000000000..174bef121f
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/restore_token_manager.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_RESTORE_TOKEN_MANAGER_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_RESTORE_TOKEN_MANAGER_H_
+
+#include <mutex>
+#include <string>
+#include <unordered_map>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+
+namespace webrtc {
+
+class RestoreTokenManager {
+ public:
+  RestoreTokenManager(const RestoreTokenManager& manager) = delete;
+  RestoreTokenManager& operator=(const RestoreTokenManager& manager) = delete;
+
+  static RestoreTokenManager& GetInstance();
+
+  void AddToken(DesktopCapturer::SourceId id, const std::string& token);
+  std::string TakeToken(DesktopCapturer::SourceId id);
+
+  // Returns a source ID which does not have any token associated with it yet.
+  DesktopCapturer::SourceId GetUnusedId();
+
+ private:
+  RestoreTokenManager() = default;
+  ~RestoreTokenManager() = default;
+
+  DesktopCapturer::SourceId last_source_id_ = 0;
+
+  std::unordered_map<DesktopCapturer::SourceId, std::string> restore_tokens_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_RESTORE_TOKEN_MANAGER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/scoped_glib.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/scoped_glib.h
new file mode 100644
index 0000000000..1361f84328
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/scoped_glib.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCOPED_GLIB_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCOPED_GLIB_H_
+
+// TODO(bugs.webrtc.org/14187): remove when all users are gone
+#include "modules/portal/scoped_glib.h"
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCOPED_GLIB_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.cc
new file mode 100644
index 0000000000..1c7cc379df
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.cc
@@ -0,0 +1,127 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/desktop_capture/linux/wayland/screen_capture_portal_interface.h"
+
+#include <string>
+
+#include "modules/portal/xdg_desktop_portal_utils.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace xdg_portal {
+
+void ScreenCapturePortalInterface::RequestSessionUsingProxy(
+    GAsyncResult* result) {
+  Scoped<GError> error;
+  GDBusProxy* proxy = g_dbus_proxy_new_finish(result, error.receive());
+  if (!proxy) {
+    // Ignore the error caused by user cancelling the request via `cancellable_`
+    if (g_error_matches(error.get(), G_IO_ERROR, G_IO_ERROR_CANCELLED))
+      return;
+    RTC_LOG(LS_ERROR) << "Failed to get a proxy for the portal: "
+                      << error->message;
+    OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  RTC_LOG(LS_INFO) << "Successfully created proxy for the portal.";
+  RequestSession(proxy);
+}
+
+void ScreenCapturePortalInterface::OnSessionRequestResult(
+    GDBusProxy* proxy,
+    GAsyncResult* result) {
+  Scoped<GError> error;
+  Scoped<GVariant> variant(
+      g_dbus_proxy_call_finish(proxy, result, error.receive()));
+  if (!variant) {
+    // Ignore the error caused by user cancelling the request via `cancellable_`
+    if (g_error_matches(error.get(), G_IO_ERROR, G_IO_ERROR_CANCELLED))
+      return;
+    RTC_LOG(LS_ERROR) << "Failed to request session: " << error->message;
+    OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  RTC_LOG(LS_INFO) << "Initializing the session.";
+
+  Scoped<char> handle;
+  g_variant_get_child(variant.get(), /*index=*/0, /*format_string=*/"o",
+                      &handle);
+  if (!handle) {
+    RTC_LOG(LS_ERROR) << "Failed to initialize the session.";
+    OnPortalDone(RequestResponse::kError);
+    return;
+  }
+}
+
+void ScreenCapturePortalInterface::RegisterSessionClosedSignalHandler(
+    const SessionClosedSignalHandler session_close_signal_handler,
+    GVariant* parameters,
+    GDBusConnection* connection,
+    std::string& session_handle,
+    guint& session_closed_signal_id) {
+  uint32_t portal_response = 2;
+  Scoped<GVariant> response_data;
+  g_variant_get(parameters, /*format_string=*/"(u@a{sv})", &portal_response,
+                response_data.receive());
+
+  if (RequestResponseFromPortalResponse(portal_response) !=
+      RequestResponse::kSuccess) {
+    RTC_LOG(LS_ERROR) << "Failed to request the session subscription.";
+    OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  Scoped<GVariant> g_session_handle(
+      g_variant_lookup_value(response_data.get(), /*key=*/"session_handle",
+                             /*expected_type=*/nullptr));
+  session_handle = g_variant_get_string(
+      /*value=*/g_session_handle.get(), /*length=*/nullptr);
+
+  if (session_handle.empty()) {
+    RTC_LOG(LS_ERROR) << "Could not get session handle despite valid response";
+    OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  session_closed_signal_id = g_dbus_connection_signal_subscribe(
+      connection, kDesktopBusName, kSessionInterfaceName, /*member=*/"Closed",
+      session_handle.c_str(), /*arg0=*/nullptr, G_DBUS_SIGNAL_FLAGS_NONE,
+      session_close_signal_handler, this, /*user_data_free_func=*/nullptr);
+}
+
+void ScreenCapturePortalInterface::OnStartRequestResult(GDBusProxy* proxy,
+                                                        GAsyncResult* result) {
+  Scoped<GError> error;
+  Scoped<GVariant> variant(
+      g_dbus_proxy_call_finish(proxy, result, error.receive()));
+  if (!variant) {
+    if (g_error_matches(error.get(), G_IO_ERROR, G_IO_ERROR_CANCELLED))
+      return;
+    RTC_LOG(LS_ERROR) << "Failed to start the portal session: "
+                      << error->message;
+    OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  Scoped<char> handle;
+  g_variant_get_child(variant.get(), 0, "o", handle.receive());
+  if (!handle) {
+    RTC_LOG(LS_ERROR) << "Failed to initialize the start portal session.";
+    OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  RTC_LOG(LS_INFO) << "Subscribed to the start signal.";
+}
+
+}  // namespace xdg_portal
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.h
new file mode 100644
index 0000000000..7953c6470c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screen_capture_portal_interface.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCREEN_CAPTURE_PORTAL_INTERFACE_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCREEN_CAPTURE_PORTAL_INTERFACE_H_
+
+#include <gio/gio.h>
+
+#include <string>
+
+#include "modules/portal/portal_request_response.h"
+#include "modules/portal/scoped_glib.h"
+#include "modules/portal/xdg_desktop_portal_utils.h"
+#include "modules/portal/xdg_session_details.h"
+
+namespace webrtc {
+namespace xdg_portal {
+
+using SessionClosedSignalHandler = void (*)(GDBusConnection*,
+                                            const char*,
+                                            const char*,
+                                            const char*,
+                                            const char*,
+                                            GVariant*,
+                                            gpointer);
+
+// A base class for XDG desktop portals that can capture desktop/screen.
+// Note: downstream clients inherit from this class so it is advisable to
+// provide a default implementation of any new virtual methods that may be added
+// to this class.
+class RTC_EXPORT ScreenCapturePortalInterface {
+ public:
+  virtual ~ScreenCapturePortalInterface() {}
+  // Gets details about the session such as session handle.
+  virtual xdg_portal::SessionDetails GetSessionDetails() { return {}; }
+  // Starts the portal setup.
+  virtual void Start() {}
+
+  // Stops and cleans up the portal.
+  virtual void Stop() {}
+
+  // Notifies observers about the success/fail state of the portal
+  // request/response.
+  virtual void OnPortalDone(xdg_portal::RequestResponse result) {}
+  // Sends a create session request to the portal.
+  virtual void RequestSession(GDBusProxy* proxy) {}
+
+  // Following methods should not be made virtual as they share a common
+  // implementation between portals.
+
+  // Requests portal session using the proxy object.
+  void RequestSessionUsingProxy(GAsyncResult* result);
+  // Handles the session request result.
+  void OnSessionRequestResult(GDBusProxy* proxy, GAsyncResult* result);
+  // Subscribes to session close signal and sets up a handler for it.
+  void RegisterSessionClosedSignalHandler(
+      const SessionClosedSignalHandler session_close_signal_handler,
+      GVariant* parameters,
+      GDBusConnection* connection,
+      std::string& session_handle,
+      guint& session_closed_signal_id);
+  // Handles the result of session start request.
+  void OnStartRequestResult(GDBusProxy* proxy, GAsyncResult* result);
+};
+
+}  // namespace xdg_portal
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCREEN_CAPTURE_PORTAL_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc
new file mode 100644
index 0000000000..e7aaee001b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc
@@ -0,0 +1,471 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/wayland/screencast_portal.h"
+
+#include <gio/gunixfdlist.h>
+#include <glib-object.h>
+
+#include "modules/portal/scoped_glib.h"
+#include "modules/portal/xdg_desktop_portal_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+using xdg_portal::kScreenCastInterfaceName;
+using xdg_portal::PrepareSignalHandle;
+using xdg_portal::RequestResponse;
+using xdg_portal::RequestSessionProxy;
+using xdg_portal::SetupRequestResponseSignal;
+using xdg_portal::SetupSessionRequestHandlers;
+using xdg_portal::StartSessionRequest;
+using xdg_portal::TearDownSession;
+using xdg_portal::RequestResponseFromPortalResponse;
+
+}  // namespace
+
+// static
+ScreenCastPortal::CaptureSourceType ScreenCastPortal::ToCaptureSourceType(
+    CaptureType type) {
+  switch (type) {
+    case CaptureType::kScreen:
+      return ScreenCastPortal::CaptureSourceType::kScreen;
+    case CaptureType::kWindow:
+      return ScreenCastPortal::CaptureSourceType::kWindow;
+    case CaptureType::kAnyScreenContent:
+      return ScreenCastPortal::CaptureSourceType::kAnyScreenContent;
+  }
+}
+
+ScreenCastPortal::ScreenCastPortal(CaptureType type, PortalNotifier* notifier)
+    : ScreenCastPortal(type,
+                       notifier,
+                       OnProxyRequested,
+                       OnSourcesRequestResponseSignal,
+                       this) {}
+
+ScreenCastPortal::ScreenCastPortal(
+    CaptureType type,
+    PortalNotifier* notifier,
+    ProxyRequestResponseHandler proxy_request_response_handler,
+    SourcesRequestResponseSignalHandler sources_request_response_signal_handler,
+    gpointer user_data,
+    bool prefer_cursor_embedded)
+    : notifier_(notifier),
+      capture_source_type_(ToCaptureSourceType(type)),
+      cursor_mode_(prefer_cursor_embedded ? CursorMode::kEmbedded
+                                          : CursorMode::kMetadata),
+      proxy_request_response_handler_(proxy_request_response_handler),
+      sources_request_response_signal_handler_(
+          sources_request_response_signal_handler),
+      user_data_(user_data) {}
+
+ScreenCastPortal::~ScreenCastPortal() {
+  Stop();
+}
+
+void ScreenCastPortal::Stop() {
+  UnsubscribeSignalHandlers();
+  TearDownSession(std::move(session_handle_), proxy_, cancellable_,
+                  connection_);
+  session_handle_ = "";
+  cancellable_ = nullptr;
+  proxy_ = nullptr;
+  restore_token_ = "";
+
+  if (pw_fd_ != -1) {
+    close(pw_fd_);
+    pw_fd_ = -1;
+  }
+}
+
+void ScreenCastPortal::UnsubscribeSignalHandlers() {
+  if (start_request_signal_id_) {
+    g_dbus_connection_signal_unsubscribe(connection_, start_request_signal_id_);
+    start_request_signal_id_ = 0;
+  }
+
+  if (sources_request_signal_id_) {
+    g_dbus_connection_signal_unsubscribe(connection_,
+                                         sources_request_signal_id_);
+    sources_request_signal_id_ = 0;
+  }
+
+  if (session_request_signal_id_) {
+    g_dbus_connection_signal_unsubscribe(connection_,
+                                         session_request_signal_id_);
+    session_request_signal_id_ = 0;
+  }
+}
+
+void ScreenCastPortal::SetSessionDetails(
+    const xdg_portal::SessionDetails& session_details) {
+  if (session_details.proxy) {
+    proxy_ = session_details.proxy;
+    connection_ = g_dbus_proxy_get_connection(proxy_);
+  }
+  if (session_details.cancellable) {
+    cancellable_ = session_details.cancellable;
+  }
+  if (!session_details.session_handle.empty()) {
+    session_handle_ = session_details.session_handle;
+  }
+  if (session_details.pipewire_stream_node_id) {
+    pw_stream_node_id_ = session_details.pipewire_stream_node_id;
+  }
+}
+
+void ScreenCastPortal::Start() {
+  cancellable_ = g_cancellable_new();
+  RequestSessionProxy(kScreenCastInterfaceName, proxy_request_response_handler_,
+                      cancellable_, this);
+}
+
+xdg_portal::SessionDetails ScreenCastPortal::GetSessionDetails() {
+  return {};  // No-op
+}
+
+void ScreenCastPortal::OnPortalDone(RequestResponse result) {
+  notifier_->OnScreenCastRequestResult(result, pw_stream_node_id_, pw_fd_);
+  if (result != RequestResponse::kSuccess) {
+    Stop();
+  }
+}
+
+// static
+void ScreenCastPortal::OnProxyRequested(GObject* gobject,
+                                        GAsyncResult* result,
+                                        gpointer user_data) {
+  static_cast<ScreenCastPortal*>(user_data)->RequestSessionUsingProxy(result);
+}
+
+void ScreenCastPortal::RequestSession(GDBusProxy* proxy) {
+  proxy_ = proxy;
+  connection_ = g_dbus_proxy_get_connection(proxy_);
+  SetupSessionRequestHandlers(
+      "webrtc", OnSessionRequested, OnSessionRequestResponseSignal, connection_,
+      proxy_, cancellable_, portal_handle_, session_request_signal_id_, this);
+}
+
+// static
+void ScreenCastPortal::OnSessionRequested(GDBusProxy* proxy,
+                                          GAsyncResult* result,
+                                          gpointer user_data) {
+  static_cast<ScreenCastPortal*>(user_data)->OnSessionRequestResult(proxy,
+                                                                    result);
+}
+
+// static
+void ScreenCastPortal::OnSessionRequestResponseSignal(
+    GDBusConnection* connection,
+    const char* sender_name,
+    const char* object_path,
+    const char* interface_name,
+    const char* signal_name,
+    GVariant* parameters,
+    gpointer user_data) {
+  ScreenCastPortal* that = static_cast<ScreenCastPortal*>(user_data);
+  RTC_DCHECK(that);
+  that->RegisterSessionClosedSignalHandler(
+      OnSessionClosedSignal, parameters, that->connection_,
+      that->session_handle_, that->session_closed_signal_id_);
+
+  // Do not continue if we don't get session_handle back. The call above will
+  // already notify the capturer there is a failure, but we would still continue
+  // to make following request and crash on that.
+  if (!that->session_handle_.empty()) {
+    that->SourcesRequest();
+  }
+}
+
+// static
+void ScreenCastPortal::OnSessionClosedSignal(GDBusConnection* connection,
+                                             const char* sender_name,
+                                             const char* object_path,
+                                             const char* interface_name,
+                                             const char* signal_name,
+                                             GVariant* parameters,
+                                             gpointer user_data) {
+  ScreenCastPortal* that = static_cast<ScreenCastPortal*>(user_data);
+  RTC_DCHECK(that);
+
+  RTC_LOG(LS_INFO) << "Received closed signal from session.";
+
+  that->notifier_->OnScreenCastSessionClosed();
+
+  // Unsubscribe from the signal and free the session handle to avoid calling
+  // Session::Close from the destructor since it's already closed
+  g_dbus_connection_signal_unsubscribe(that->connection_,
+                                       that->session_closed_signal_id_);
+}
+
+void ScreenCastPortal::SourcesRequest() {
+  GVariantBuilder builder;
+  Scoped<char> variant_string;
+
+  g_variant_builder_init(&builder, G_VARIANT_TYPE_VARDICT);
+  // We want to record monitor content.
+  g_variant_builder_add(
+      &builder, "{sv}", "types",
+      g_variant_new_uint32(static_cast<uint32_t>(capture_source_type_)));
+  // We don't want to allow selection of multiple sources.
+  g_variant_builder_add(&builder, "{sv}", "multiple",
+                        g_variant_new_boolean(false));
+
+  Scoped<GVariant> cursorModesVariant(
+      g_dbus_proxy_get_cached_property(proxy_, "AvailableCursorModes"));
+  if (cursorModesVariant.get()) {
+    uint32_t modes = 0;
+    g_variant_get(cursorModesVariant.get(), "u", &modes);
+    // Make request only if this mode is advertised by the portal
+    // implementation.
+    if (modes & static_cast<uint32_t>(cursor_mode_)) {
+      g_variant_builder_add(
+          &builder, "{sv}", "cursor_mode",
+          g_variant_new_uint32(static_cast<uint32_t>(cursor_mode_)));
+    }
+  }
+
+  Scoped<GVariant> versionVariant(
+      g_dbus_proxy_get_cached_property(proxy_, "version"));
+  if (versionVariant.get()) {
+    uint32_t version = 0;
+    g_variant_get(versionVariant.get(), "u", &version);
+    // Make request only if xdg-desktop-portal has required API version
+    if (version >= 4) {
+      g_variant_builder_add(
+          &builder, "{sv}", "persist_mode",
+          g_variant_new_uint32(static_cast<uint32_t>(persist_mode_)));
+      if (!restore_token_.empty()) {
+        g_variant_builder_add(&builder, "{sv}", "restore_token",
+                              g_variant_new_string(restore_token_.c_str()));
+      }
+    }
+  }
+
+  variant_string = g_strdup_printf("webrtc%d", g_random_int_range(0, G_MAXINT));
+  g_variant_builder_add(&builder, "{sv}", "handle_token",
+                        g_variant_new_string(variant_string.get()));
+
+  sources_handle_ = PrepareSignalHandle(variant_string.get(), connection_);
+  sources_request_signal_id_ = SetupRequestResponseSignal(
+      sources_handle_.c_str(), sources_request_response_signal_handler_,
+      user_data_, connection_);
+
+  RTC_LOG(LS_INFO) << "Requesting sources from the screen cast session.";
+  g_dbus_proxy_call(
+      proxy_, "SelectSources",
+      g_variant_new("(oa{sv})", session_handle_.c_str(), &builder),
+      G_DBUS_CALL_FLAGS_NONE, /*timeout=*/-1, cancellable_,
+      reinterpret_cast<GAsyncReadyCallback>(OnSourcesRequested), this);
+}
+
+// static
+void ScreenCastPortal::OnSourcesRequested(GDBusProxy* proxy,
+                                          GAsyncResult* result,
+                                          gpointer user_data) {
+  ScreenCastPortal* that = static_cast<ScreenCastPortal*>(user_data);
+  RTC_DCHECK(that);
+
+  Scoped<GError> error;
+  Scoped<GVariant> variant(
+      g_dbus_proxy_call_finish(proxy, result, error.receive()));
+  if (!variant) {
+    if (g_error_matches(error.get(), G_IO_ERROR, G_IO_ERROR_CANCELLED))
+      return;
+    RTC_LOG(LS_ERROR) << "Failed to request the sources: " << error->message;
+    that->OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  RTC_LOG(LS_INFO) << "Sources requested from the screen cast session.";
+
+  Scoped<char> handle;
+  g_variant_get_child(variant.get(), 0, "o", handle.receive());
+  if (!handle) {
+    RTC_LOG(LS_ERROR) << "Failed to initialize the screen cast session.";
+    if (that->sources_request_signal_id_) {
+      g_dbus_connection_signal_unsubscribe(that->connection_,
+                                           that->sources_request_signal_id_);
+      that->sources_request_signal_id_ = 0;
+    }
+    that->OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  RTC_LOG(LS_INFO) << "Subscribed to sources signal.";
+}
+
+// static
+void ScreenCastPortal::OnSourcesRequestResponseSignal(
+    GDBusConnection* connection,
+    const char* sender_name,
+    const char* object_path,
+    const char* interface_name,
+    const char* signal_name,
+    GVariant* parameters,
+    gpointer user_data) {
+  ScreenCastPortal* that = static_cast<ScreenCastPortal*>(user_data);
+  RTC_DCHECK(that);
+
+  RTC_LOG(LS_INFO) << "Received sources signal from session.";
+
+  uint32_t portal_response;
+  g_variant_get(parameters, "(u@a{sv})", &portal_response, nullptr);
+  if (portal_response) {
+    RTC_LOG(LS_ERROR)
+        << "Failed to select sources for the screen cast session.";
+    that->OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  that->StartRequest();
+}
+
+void ScreenCastPortal::StartRequest() {
+  StartSessionRequest("webrtc", session_handle_, OnStartRequestResponseSignal,
+                      OnStartRequested, proxy_, connection_, cancellable_,
+                      start_request_signal_id_, start_handle_, this);
+}
+
+// static
+void ScreenCastPortal::OnStartRequested(GDBusProxy* proxy,
+                                        GAsyncResult* result,
+                                        gpointer user_data) {
+  static_cast<ScreenCastPortal*>(user_data)->OnStartRequestResult(proxy,
+                                                                  result);
+}
+
+// static
+void ScreenCastPortal::OnStartRequestResponseSignal(GDBusConnection* connection,
+                                                    const char* sender_name,
+                                                    const char* object_path,
+                                                    const char* interface_name,
+                                                    const char* signal_name,
+                                                    GVariant* parameters,
+                                                    gpointer user_data) {
+  ScreenCastPortal* that = static_cast<ScreenCastPortal*>(user_data);
+  RTC_DCHECK(that);
+
+  RTC_LOG(LS_INFO) << "Start signal received.";
+  uint32_t portal_response;
+  Scoped<GVariant> response_data;
+  Scoped<GVariantIter> iter;
+  Scoped<char> restore_token;
+  g_variant_get(parameters, "(u@a{sv})", &portal_response,
+                response_data.receive());
+  if (portal_response || !response_data) {
+    RTC_LOG(LS_ERROR) << "Failed to start the screen cast session.";
+    that->OnPortalDone(RequestResponseFromPortalResponse(portal_response));
+    return;
+  }
+
+  // Array of PipeWire streams. See
+  // https://github.com/flatpak/xdg-desktop-portal/blob/master/data/org.freedesktop.portal.ScreenCast.xml
+  // documentation for <method name="Start">.
+  if (g_variant_lookup(response_data.get(), "streams", "a(ua{sv})",
+                       iter.receive())) {
+    Scoped<GVariant> variant;
+
+    while (g_variant_iter_next(iter.get(), "@(ua{sv})", variant.receive())) {
+      uint32_t stream_id;
+      uint32_t type;
+      Scoped<GVariant> options;
+
+      g_variant_get(variant.get(), "(u@a{sv})", &stream_id, options.receive());
+      RTC_DCHECK(options.get());
+
+      if (g_variant_lookup(options.get(), "source_type", "u", &type)) {
+        that->capture_source_type_ =
+            static_cast<ScreenCastPortal::CaptureSourceType>(type);
+      }
+
+      that->pw_stream_node_id_ = stream_id;
+
+      break;
+    }
+  }
+
+  if (g_variant_lookup(response_data.get(), "restore_token", "s",
+                       restore_token.receive())) {
+    that->restore_token_ = restore_token.get();
+  }
+
+  that->OpenPipeWireRemote();
+}
+
+uint32_t ScreenCastPortal::pipewire_stream_node_id() {
+  return pw_stream_node_id_;
+}
+
+void ScreenCastPortal::SetPersistMode(ScreenCastPortal::PersistMode mode) {
+  persist_mode_ = mode;
+}
+
+void ScreenCastPortal::SetRestoreToken(const std::string& token) {
+  restore_token_ = token;
+}
+
+std::string ScreenCastPortal::RestoreToken() const {
+  return restore_token_;
+}
+
+void ScreenCastPortal::OpenPipeWireRemote() {
+  GVariantBuilder builder;
+  g_variant_builder_init(&builder, G_VARIANT_TYPE_VARDICT);
+
+  RTC_LOG(LS_INFO) << "Opening the PipeWire remote.";
+
+  g_dbus_proxy_call_with_unix_fd_list(
+      proxy_, "OpenPipeWireRemote",
+      g_variant_new("(oa{sv})", session_handle_.c_str(), &builder),
+      G_DBUS_CALL_FLAGS_NONE, /*timeout=*/-1, /*fd_list=*/nullptr, cancellable_,
+      reinterpret_cast<GAsyncReadyCallback>(OnOpenPipeWireRemoteRequested),
+      this);
+}
+
+// static
+void ScreenCastPortal::OnOpenPipeWireRemoteRequested(GDBusProxy* proxy,
+                                                     GAsyncResult* result,
+                                                     gpointer user_data) {
+  ScreenCastPortal* that = static_cast<ScreenCastPortal*>(user_data);
+  RTC_DCHECK(that);
+
+  Scoped<GError> error;
+  Scoped<GUnixFDList> outlist;
+  Scoped<GVariant> variant(g_dbus_proxy_call_with_unix_fd_list_finish(
+      proxy, outlist.receive(), result, error.receive()));
+  if (!variant) {
+    if (g_error_matches(error.get(), G_IO_ERROR, G_IO_ERROR_CANCELLED))
+      return;
+    RTC_LOG(LS_ERROR) << "Failed to open the PipeWire remote: "
+                      << error->message;
+    that->OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  int32_t index;
+  g_variant_get(variant.get(), "(h)", &index);
+
+  that->pw_fd_ = g_unix_fd_list_get(outlist.get(), index, error.receive());
+
+  if (that->pw_fd_ == -1) {
+    RTC_LOG(LS_ERROR) << "Failed to get file descriptor from the list: "
+                      << error->message;
+    that->OnPortalDone(RequestResponse::kError);
+    return;
+  }
+
+  that->OnPortalDone(RequestResponse::kSuccess);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.h
new file mode 100644
index 0000000000..37e55815e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.h
@@ -0,0 +1,218 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCREENCAST_PORTAL_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCREENCAST_PORTAL_H_
+
+#include <gio/gio.h>
+
+#include <string>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/linux/wayland/screen_capture_portal_interface.h"
+#include "modules/portal/portal_request_response.h"
+#include "modules/portal/xdg_desktop_portal_utils.h"
+#include "modules/portal/xdg_session_details.h"
+
+namespace webrtc {
+
+class RTC_EXPORT ScreenCastPortal
+    : public xdg_portal::ScreenCapturePortalInterface {
+ public:
+  using ProxyRequestResponseHandler = void (*)(GObject* object,
+                                               GAsyncResult* result,
+                                               gpointer user_data);
+
+  using SourcesRequestResponseSignalHandler =
+      void (*)(GDBusConnection* connection,
+               const char* sender_name,
+               const char* object_path,
+               const char* interface_name,
+               const char* signal_name,
+               GVariant* parameters,
+               gpointer user_data);
+
+  // Values are set based on cursor mode property in
+  // xdg-desktop-portal/screencast
+  // https://github.com/flatpak/xdg-desktop-portal/blob/master/data/org.freedesktop.portal.ScreenCast.xml
+  enum class CursorMode : uint32_t {
+    // Mouse cursor will not be included in any form
+    kHidden = 0b01,
+    // Mouse cursor will be part of the screen content
+    kEmbedded = 0b10,
+    // Mouse cursor information will be send separately in form of metadata
+    kMetadata = 0b100
+  };
+
+  // Values are set based on persist mode property in
+  // xdg-desktop-portal/screencast
+  // https://github.com/flatpak/xdg-desktop-portal/blob/master/data/org.freedesktop.portal.ScreenCast.xml
+  enum class PersistMode : uint32_t {
+    // Do not allow to restore stream
+    kDoNotPersist = 0b00,
+    // The restore token is valid as long as the application is alive. It's
+    // stored in memory and revoked when the application closes its DBus
+    // connection
+    kTransient = 0b01,
+    // The restore token is stored in disk and is valid until the user manually
+    // revokes it
+    kPersistent = 0b10
+  };
+
+  // Interface that must be implemented by the ScreenCastPortal consumers.
+  class PortalNotifier {
+   public:
+    virtual void OnScreenCastRequestResult(xdg_portal::RequestResponse result,
+                                           uint32_t stream_node_id,
+                                           int fd) = 0;
+    virtual void OnScreenCastSessionClosed() = 0;
+
+   protected:
+    PortalNotifier() = default;
+    virtual ~PortalNotifier() = default;
+  };
+
+  ScreenCastPortal(CaptureType type, PortalNotifier* notifier);
+  ScreenCastPortal(CaptureType type,
+                   PortalNotifier* notifier,
+                   ProxyRequestResponseHandler proxy_request_response_handler,
+                   SourcesRequestResponseSignalHandler
+                       sources_request_response_signal_handler,
+                   gpointer user_data,
+                   // TODO(chromium:1291247): Remove the default option once
+                   // downstream has been adjusted.
+                   bool prefer_cursor_embedded = false);
+
+  ~ScreenCastPortal();
+
+  // Initialize ScreenCastPortal with series of DBus calls where we try to
+  // obtain all the required information, like PipeWire file descriptor and
+  // PipeWire stream node ID.
+  //
+  // The observer will return whether the communication with xdg-desktop-portal
+  // was successful and only then you will be able to get all the required
+  // information in order to continue working with PipeWire.
+  void Start() override;
+  void Stop() override;
+  xdg_portal::SessionDetails GetSessionDetails() override;
+
+  // Method to notify the reason for failure of a portal request.
+  void OnPortalDone(xdg_portal::RequestResponse result) override;
+
+  // Sends a create session request to the portal.
+  void RequestSession(GDBusProxy* proxy) override;
+
+  // Set of methods leveraged by remote desktop portal to setup a common session
+  // with screen cast portal.
+  void SetSessionDetails(const xdg_portal::SessionDetails& session_details);
+  uint32_t pipewire_stream_node_id();
+  void SourcesRequest();
+  void OpenPipeWireRemote();
+
+  // ScreenCast specific methods for stream restoration
+  void SetPersistMode(ScreenCastPortal::PersistMode mode);
+  void SetRestoreToken(const std::string& token);
+  std::string RestoreToken() const;
+
+ private:
+  // Values are set based on source type property in
+  // xdg-desktop-portal/screencast
+  // https://github.com/flatpak/xdg-desktop-portal/blob/master/data/org.freedesktop.portal.ScreenCast.xml
+  enum class CaptureSourceType : uint32_t {
+    kScreen = 0b01,
+    kWindow = 0b10,
+    kAnyScreenContent = kScreen | kWindow
+  };
+  static CaptureSourceType ToCaptureSourceType(CaptureType type);
+
+  PortalNotifier* notifier_;
+
+  // A PipeWire stream ID of stream we will be connecting to
+  uint32_t pw_stream_node_id_ = 0;
+  // A file descriptor of PipeWire socket
+  int pw_fd_ = -1;
+  // Restore token that can be used to restore previous session
+  std::string restore_token_;
+
+  CaptureSourceType capture_source_type_ =
+      ScreenCastPortal::CaptureSourceType::kScreen;
+
+  CursorMode cursor_mode_ = CursorMode::kMetadata;
+
+  PersistMode persist_mode_ = ScreenCastPortal::PersistMode::kDoNotPersist;
+
+  ProxyRequestResponseHandler proxy_request_response_handler_;
+  SourcesRequestResponseSignalHandler sources_request_response_signal_handler_;
+  gpointer user_data_;
+
+  GDBusConnection* connection_ = nullptr;
+  GDBusProxy* proxy_ = nullptr;
+  GCancellable* cancellable_ = nullptr;
+  std::string portal_handle_;
+  std::string session_handle_;
+  std::string sources_handle_;
+  std::string start_handle_;
+  guint session_request_signal_id_ = 0;
+  guint sources_request_signal_id_ = 0;
+  guint start_request_signal_id_ = 0;
+  guint session_closed_signal_id_ = 0;
+
+  void UnsubscribeSignalHandlers();
+  static void OnProxyRequested(GObject* object,
+                               GAsyncResult* result,
+                               gpointer user_data);
+  static void OnSessionRequested(GDBusProxy* proxy,
+                                 GAsyncResult* result,
+                                 gpointer user_data);
+  static void OnSessionRequestResponseSignal(GDBusConnection* connection,
+                                             const char* sender_name,
+                                             const char* object_path,
+                                             const char* interface_name,
+                                             const char* signal_name,
+                                             GVariant* parameters,
+                                             gpointer user_data);
+  static void OnSessionClosedSignal(GDBusConnection* connection,
+                                    const char* sender_name,
+                                    const char* object_path,
+                                    const char* interface_name,
+                                    const char* signal_name,
+                                    GVariant* parameters,
+                                    gpointer user_data);
+  static void OnSourcesRequested(GDBusProxy* proxy,
+                                 GAsyncResult* result,
+                                 gpointer user_data);
+  static void OnSourcesRequestResponseSignal(GDBusConnection* connection,
+                                             const char* sender_name,
+                                             const char* object_path,
+                                             const char* interface_name,
+                                             const char* signal_name,
+                                             GVariant* parameters,
+                                             gpointer user_data);
+
+  void StartRequest();
+  static void OnStartRequested(GDBusProxy* proxy,
+                               GAsyncResult* result,
+                               gpointer user_data);
+  static void OnStartRequestResponseSignal(GDBusConnection* connection,
+                                           const char* sender_name,
+                                           const char* object_path,
+                                           const char* interface_name,
+                                           const char* signal_name,
+                                           GVariant* parameters,
+                                           gpointer user_data);
+
+  static void OnOpenPipeWireRemoteRequested(GDBusProxy* proxy,
+                                            GAsyncResult* result,
+                                            gpointer user_data);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCREENCAST_PORTAL_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc
new file mode 100644
index 0000000000..0c4900d1cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.cc
@@ -0,0 +1,123 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/wayland/screencast_stream_utils.h"
+
+#include <libdrm/drm_fourcc.h>
+#include <pipewire/pipewire.h>
+#include <spa/param/video/format-utils.h>
+
+#include <string>
+
+#include "rtc_base/string_to_number.h"
+
+#if !PW_CHECK_VERSION(0, 3, 29)
+#define SPA_POD_PROP_FLAG_MANDATORY (1u << 3)
+#endif
+#if !PW_CHECK_VERSION(0, 3, 33)
+#define SPA_POD_PROP_FLAG_DONT_FIXATE (1u << 4)
+#endif
+
+namespace webrtc {
+
+PipeWireVersion PipeWireVersion::Parse(const absl::string_view& version) {
+  std::vector<absl::string_view> parsed_version = rtc::split(version, '.');
+
+  if (parsed_version.size() != 3) {
+    return {};
+  }
+
+  absl::optional<int> major = rtc::StringToNumber<int>(parsed_version.at(0));
+  absl::optional<int> minor = rtc::StringToNumber<int>(parsed_version.at(1));
+  absl::optional<int> micro = rtc::StringToNumber<int>(parsed_version.at(2));
+
+  // Return invalid version if we failed to parse it
+  if (!major || !minor || !micro) {
+    return {};
+  }
+
+  return {major.value(), minor.value(), micro.value()};
+}
+
+bool PipeWireVersion::operator>=(const PipeWireVersion& other) {
+  if (!major && !minor && !micro) {
+    return false;
+  }
+
+  return std::tie(major, minor, micro) >=
+         std::tie(other.major, other.minor, other.micro);
+}
+
+bool PipeWireVersion::operator<=(const PipeWireVersion& other) {
+  if (!major && !minor && !micro) {
+    return false;
+  }
+
+  return std::tie(major, minor, micro) <=
+         std::tie(other.major, other.minor, other.micro);
+}
+
+spa_pod* BuildFormat(spa_pod_builder* builder,
+                     uint32_t format,
+                     const std::vector<uint64_t>& modifiers,
+                     const struct spa_rectangle* resolution) {
+  spa_pod_frame frames[2];
+  spa_rectangle pw_min_screen_bounds = spa_rectangle{1, 1};
+  spa_rectangle pw_max_screen_bounds = spa_rectangle{UINT32_MAX, UINT32_MAX};
+
+  spa_pod_builder_push_object(builder, &frames[0], SPA_TYPE_OBJECT_Format,
+                              SPA_PARAM_EnumFormat);
+  spa_pod_builder_add(builder, SPA_FORMAT_mediaType,
+                      SPA_POD_Id(SPA_MEDIA_TYPE_video), 0);
+  spa_pod_builder_add(builder, SPA_FORMAT_mediaSubtype,
+                      SPA_POD_Id(SPA_MEDIA_SUBTYPE_raw), 0);
+  spa_pod_builder_add(builder, SPA_FORMAT_VIDEO_format, SPA_POD_Id(format), 0);
+
+  if (modifiers.size()) {
+    if (modifiers.size() == 1 && modifiers[0] == DRM_FORMAT_MOD_INVALID) {
+      spa_pod_builder_prop(builder, SPA_FORMAT_VIDEO_modifier,
+                           SPA_POD_PROP_FLAG_MANDATORY);
+      spa_pod_builder_long(builder, modifiers[0]);
+    } else {
+      spa_pod_builder_prop(
+          builder, SPA_FORMAT_VIDEO_modifier,
+          SPA_POD_PROP_FLAG_MANDATORY | SPA_POD_PROP_FLAG_DONT_FIXATE);
+      spa_pod_builder_push_choice(builder, &frames[1], SPA_CHOICE_Enum, 0);
+
+      // modifiers from the array
+      bool first = true;
+      for (int64_t val : modifiers) {
+        spa_pod_builder_long(builder, val);
+        // Add the first modifier twice as the very first value is the default
+        // option
+        if (first) {
+          spa_pod_builder_long(builder, val);
+          first = false;
+        }
+      }
+      spa_pod_builder_pop(builder, &frames[1]);
+    }
+  }
+
+  if (resolution) {
+    spa_pod_builder_add(builder, SPA_FORMAT_VIDEO_size,
+                        SPA_POD_Rectangle(resolution), 0);
+  } else {
+    spa_pod_builder_add(builder, SPA_FORMAT_VIDEO_size,
+                        SPA_POD_CHOICE_RANGE_Rectangle(&pw_min_screen_bounds,
+                                                       &pw_min_screen_bounds,
+                                                       &pw_max_screen_bounds),
+                        0);
+  }
+
+  return static_cast<spa_pod*>(spa_pod_builder_pop(builder, &frames[0]));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.h
new file mode 100644
index 0000000000..e04d7db931
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_stream_utils.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCREENCAST_STREAM_UTILS_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCREENCAST_STREAM_UTILS_H_
+
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "rtc_base/string_encode.h"
+
+struct spa_pod;
+struct spa_pod_builder;
+struct spa_rectangle;
+
+namespace webrtc {
+
+struct PipeWireVersion {
+  static PipeWireVersion Parse(const absl::string_view& version);
+
+  // Returns whether current version is newer or same as required version
+  bool operator>=(const PipeWireVersion& other);
+  // Returns whether current version is older or same as required version
+  bool operator<=(const PipeWireVersion& other);
+
+  int major = 0;
+  int minor = 0;
+  int micro = 0;
+};
+
+// Returns a spa_pod used to build PipeWire stream format using given
+// arguments. Modifiers are optional value and when present they will be
+// used with SPA_POD_PROP_FLAG_MANDATORY and SPA_POD_PROP_FLAG_DONT_FIXATE
+// flags.
+spa_pod* BuildFormat(spa_pod_builder* builder,
+                     uint32_t format,
+                     const std::vector<uint64_t>& modifiers,
+                     const struct spa_rectangle* resolution);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SCREENCAST_STREAM_UTILS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
new file mode 100644
index 0000000000..49b6484510
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
@@ -0,0 +1,965 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/wayland/shared_screencast_stream.h"
+
+#include <fcntl.h>
+#include <libdrm/drm_fourcc.h>
+#include <pipewire/pipewire.h>
+#include <spa/param/video/format-utils.h>
+#include <sys/mman.h>
+
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "modules/desktop_capture/linux/wayland/egl_dmabuf.h"
+#include "modules/desktop_capture/linux/wayland/screencast_stream_utils.h"
+#include "modules/portal/pipewire_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/sanitizer.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+const int kBytesPerPixel = 4;
+const int kVideoDamageRegionCount = 16;
+
+constexpr int kCursorBpp = 4;
+constexpr int CursorMetaSize(int w, int h) {
+  return (sizeof(struct spa_meta_cursor) + sizeof(struct spa_meta_bitmap) +
+          w * h * kCursorBpp);
+}
+
+constexpr PipeWireVersion kDmaBufModifierMinVersion = {0, 3, 33};
+constexpr PipeWireVersion kDropSingleModifierMinVersion = {0, 3, 40};
+
+class ScopedBuf {
+ public:
+  ScopedBuf() {}
+  ScopedBuf(uint8_t* map, int map_size, int fd)
+      : map_(map), map_size_(map_size), fd_(fd) {}
+  ~ScopedBuf() {
+    if (map_ != MAP_FAILED) {
+      munmap(map_, map_size_);
+    }
+  }
+
+  explicit operator bool() { return map_ != MAP_FAILED; }
+
+  void initialize(uint8_t* map, int map_size, int fd) {
+    map_ = map;
+    map_size_ = map_size;
+    fd_ = fd;
+  }
+
+  uint8_t* get() { return map_; }
+
+ protected:
+  uint8_t* map_ = static_cast<uint8_t*>(MAP_FAILED);
+  int map_size_;
+  int fd_;
+};
+
+class SharedScreenCastStreamPrivate {
+ public:
+  SharedScreenCastStreamPrivate();
+  ~SharedScreenCastStreamPrivate();
+
+  bool StartScreenCastStream(uint32_t stream_node_id,
+                             int fd,
+                             uint32_t width = 0,
+                             uint32_t height = 0,
+                             bool is_cursor_embedded = false);
+  void UpdateScreenCastStreamResolution(uint32_t width, uint32_t height);
+  void SetUseDamageRegion(bool use_damage_region) {
+    use_damage_region_ = use_damage_region;
+  }
+  void SetObserver(SharedScreenCastStream::Observer* observer) {
+    observer_ = observer;
+  }
+  void StopScreenCastStream();
+  std::unique_ptr<SharedDesktopFrame> CaptureFrame();
+  std::unique_ptr<MouseCursor> CaptureCursor();
+  DesktopVector CaptureCursorPosition();
+
+ private:
+  // Stops the streams and cleans up any in-use elements.
+  void StopAndCleanupStream();
+
+  SharedScreenCastStream::Observer* observer_ = nullptr;
+
+  // Track damage region updates that were reported since the last time
+  // frame was captured
+  DesktopRegion damage_region_;
+
+  uint32_t pw_stream_node_id_ = 0;
+
+  DesktopSize stream_size_ = {};
+  DesktopSize frame_size_;
+
+  webrtc::Mutex queue_lock_;
+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_
+      RTC_GUARDED_BY(&queue_lock_);
+  std::unique_ptr<MouseCursor> mouse_cursor_;
+  DesktopVector mouse_cursor_position_ = DesktopVector(-1, -1);
+
+  int64_t modifier_;
+  std::unique_ptr<EglDmaBuf> egl_dmabuf_;
+  // List of modifiers we query as supported by the graphics card/driver
+  std::vector<uint64_t> modifiers_;
+
+  // PipeWire types
+  struct pw_context* pw_context_ = nullptr;
+  struct pw_core* pw_core_ = nullptr;
+  struct pw_stream* pw_stream_ = nullptr;
+  struct pw_thread_loop* pw_main_loop_ = nullptr;
+  struct spa_source* renegotiate_ = nullptr;
+
+  spa_hook spa_core_listener_;
+  spa_hook spa_stream_listener_;
+
+  // A number used to verify all previous methods and the resulting
+  // events have been handled.
+  int server_version_sync_ = 0;
+  // Version of the running PipeWire server we communicate with
+  PipeWireVersion pw_server_version_;
+  // Version of the library used to run our code
+  PipeWireVersion pw_client_version_;
+
+  // Resolution parameters.
+  uint32_t width_ = 0;
+  uint32_t height_ = 0;
+  webrtc::Mutex resolution_lock_;
+  // Resolution changes are processed during buffer processing.
+  bool pending_resolution_change_ RTC_GUARDED_BY(&resolution_lock_) = false;
+
+  bool use_damage_region_ = true;
+
+  // Specifies whether the pipewire stream has been initialized with a request
+  // to embed cursor into the captured frames.
+  bool is_cursor_embedded_ = false;
+
+  // event handlers
+  pw_core_events pw_core_events_ = {};
+  pw_stream_events pw_stream_events_ = {};
+
+  struct spa_video_info_raw spa_video_format_;
+
+  void ProcessBuffer(pw_buffer* buffer);
+  bool ProcessMemFDBuffer(pw_buffer* buffer,
+                          DesktopFrame& frame,
+                          const DesktopVector& offset);
+  bool ProcessDMABuffer(pw_buffer* buffer,
+                        DesktopFrame& frame,
+                        const DesktopVector& offset);
+  void ConvertRGBxToBGRx(uint8_t* frame, uint32_t size);
+
+  // PipeWire callbacks
+  static void OnCoreError(void* data,
+                          uint32_t id,
+                          int seq,
+                          int res,
+                          const char* message);
+  static void OnCoreDone(void* user_data, uint32_t id, int seq);
+  static void OnCoreInfo(void* user_data, const pw_core_info* info);
+  static void OnStreamParamChanged(void* data,
+                                   uint32_t id,
+                                   const struct spa_pod* format);
+  static void OnStreamStateChanged(void* data,
+                                   pw_stream_state old_state,
+                                   pw_stream_state state,
+                                   const char* error_message);
+  static void OnStreamProcess(void* data);
+  // This will be invoked in case we fail to process DMA-BUF PW buffer using
+  // negotiated stream parameters (modifier). We will drop the modifier we
+  // failed to use and try to use a different one or fallback to shared memory
+  // buffers.
+  static void OnRenegotiateFormat(void* data, uint64_t);
+};
+
+void SharedScreenCastStreamPrivate::OnCoreError(void* data,
+                                                uint32_t id,
+                                                int seq,
+                                                int res,
+                                                const char* message) {
+  SharedScreenCastStreamPrivate* stream =
+      static_cast<SharedScreenCastStreamPrivate*>(data);
+  RTC_DCHECK(stream);
+
+  RTC_LOG(LS_ERROR) << "PipeWire remote error: " << message;
+  pw_thread_loop_signal(stream->pw_main_loop_, false);
+}
+
+void SharedScreenCastStreamPrivate::OnCoreInfo(void* data,
+                                               const pw_core_info* info) {
+  SharedScreenCastStreamPrivate* stream =
+      static_cast<SharedScreenCastStreamPrivate*>(data);
+  RTC_DCHECK(stream);
+
+  stream->pw_server_version_ = PipeWireVersion::Parse(info->version);
+}
+
+void SharedScreenCastStreamPrivate::OnCoreDone(void* data,
+                                               uint32_t id,
+                                               int seq) {
+  const SharedScreenCastStreamPrivate* stream =
+      static_cast<SharedScreenCastStreamPrivate*>(data);
+  RTC_DCHECK(stream);
+
+  if (id == PW_ID_CORE && stream->server_version_sync_ == seq) {
+    pw_thread_loop_signal(stream->pw_main_loop_, false);
+  }
+}
+
+// static
+void SharedScreenCastStreamPrivate::OnStreamStateChanged(
+    void* data,
+    pw_stream_state old_state,
+    pw_stream_state state,
+    const char* error_message) {
+  SharedScreenCastStreamPrivate* that =
+      static_cast<SharedScreenCastStreamPrivate*>(data);
+  RTC_DCHECK(that);
+
+  switch (state) {
+    case PW_STREAM_STATE_ERROR:
+      RTC_LOG(LS_ERROR) << "PipeWire stream state error: " << error_message;
+      break;
+    case PW_STREAM_STATE_PAUSED:
+      if (that->observer_ && old_state != PW_STREAM_STATE_STREAMING) {
+        that->observer_->OnStreamConfigured();
+      }
+      break;
+    case PW_STREAM_STATE_STREAMING:
+    case PW_STREAM_STATE_UNCONNECTED:
+    case PW_STREAM_STATE_CONNECTING:
+      break;
+  }
+}
+
+// static
+void SharedScreenCastStreamPrivate::OnStreamParamChanged(
+    void* data,
+    uint32_t id,
+    const struct spa_pod* format) {
+  SharedScreenCastStreamPrivate* that =
+      static_cast<SharedScreenCastStreamPrivate*>(data);
+  RTC_DCHECK(that);
+
+  RTC_LOG(LS_INFO) << "PipeWire stream format changed.";
+  if (!format || id != SPA_PARAM_Format) {
+    return;
+  }
+
+  spa_format_video_raw_parse(format, &that->spa_video_format_);
+
+  auto width = that->spa_video_format_.size.width;
+  auto height = that->spa_video_format_.size.height;
+  auto stride = SPA_ROUND_UP_N(width * kBytesPerPixel, 4);
+  auto size = height * stride;
+
+  that->stream_size_ = DesktopSize(width, height);
+
+  uint8_t buffer[1024] = {};
+  auto builder = spa_pod_builder{buffer, sizeof(buffer)};
+
+  // Setup buffers and meta header for new format.
+
+  // When SPA_FORMAT_VIDEO_modifier is present we can use DMA-BUFs as
+  // the server announces support for it.
+  // See https://github.com/PipeWire/pipewire/blob/master/doc/dma-buf.dox
+  const bool has_modifier =
+      spa_pod_find_prop(format, nullptr, SPA_FORMAT_VIDEO_modifier);
+  that->modifier_ =
+      has_modifier ? that->spa_video_format_.modifier : DRM_FORMAT_MOD_INVALID;
+  std::vector<const spa_pod*> params;
+  const int buffer_types =
+      has_modifier
+          ? (1 << SPA_DATA_DmaBuf) | (1 << SPA_DATA_MemFd)
+          : (1 << SPA_DATA_MemFd);
+
+  params.push_back(reinterpret_cast<spa_pod*>(spa_pod_builder_add_object(
+      &builder, SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
+      SPA_PARAM_BUFFERS_size, SPA_POD_Int(size), SPA_PARAM_BUFFERS_stride,
+      SPA_POD_Int(stride), SPA_PARAM_BUFFERS_buffers,
+      SPA_POD_CHOICE_RANGE_Int(8, 1, 32), SPA_PARAM_BUFFERS_dataType,
+      SPA_POD_CHOICE_FLAGS_Int(buffer_types))));
+  params.push_back(reinterpret_cast<spa_pod*>(spa_pod_builder_add_object(
+      &builder, SPA_TYPE_OBJECT_ParamMeta, SPA_PARAM_Meta, SPA_PARAM_META_type,
+      SPA_POD_Id(SPA_META_Header), SPA_PARAM_META_size,
+      SPA_POD_Int(sizeof(struct spa_meta_header)))));
+  params.push_back(reinterpret_cast<spa_pod*>(spa_pod_builder_add_object(
+      &builder, SPA_TYPE_OBJECT_ParamMeta, SPA_PARAM_Meta, SPA_PARAM_META_type,
+      SPA_POD_Id(SPA_META_VideoCrop), SPA_PARAM_META_size,
+      SPA_POD_Int(sizeof(struct spa_meta_region)))));
+  params.push_back(reinterpret_cast<spa_pod*>(spa_pod_builder_add_object(
+      &builder, SPA_TYPE_OBJECT_ParamMeta, SPA_PARAM_Meta, SPA_PARAM_META_type,
+      SPA_POD_Id(SPA_META_Cursor), SPA_PARAM_META_size,
+      SPA_POD_CHOICE_RANGE_Int(CursorMetaSize(64, 64), CursorMetaSize(1, 1),
+                               CursorMetaSize(384, 384)))));
+  params.push_back(reinterpret_cast<spa_pod*>(spa_pod_builder_add_object(
+      &builder, SPA_TYPE_OBJECT_ParamMeta, SPA_PARAM_Meta, SPA_PARAM_META_type,
+      SPA_POD_Id(SPA_META_VideoDamage), SPA_PARAM_META_size,
+      SPA_POD_CHOICE_RANGE_Int(
+          sizeof(struct spa_meta_region) * kVideoDamageRegionCount,
+          sizeof(struct spa_meta_region) * 1,
+          sizeof(struct spa_meta_region) * kVideoDamageRegionCount))));
+
+  pw_stream_update_params(that->pw_stream_, params.data(), params.size());
+}
+
+// static
+void SharedScreenCastStreamPrivate::OnStreamProcess(void* data) {
+  SharedScreenCastStreamPrivate* that =
+      static_cast<SharedScreenCastStreamPrivate*>(data);
+  RTC_DCHECK(that);
+
+  struct pw_buffer* next_buffer;
+  struct pw_buffer* buffer = nullptr;
+
+  next_buffer = pw_stream_dequeue_buffer(that->pw_stream_);
+  while (next_buffer) {
+    buffer = next_buffer;
+    next_buffer = pw_stream_dequeue_buffer(that->pw_stream_);
+
+    if (next_buffer) {
+      pw_stream_queue_buffer(that->pw_stream_, buffer);
+    }
+  }
+
+  if (!buffer) {
+    return;
+  }
+
+  that->ProcessBuffer(buffer);
+
+  pw_stream_queue_buffer(that->pw_stream_, buffer);
+}
+
+void SharedScreenCastStreamPrivate::OnRenegotiateFormat(void* data, uint64_t) {
+  SharedScreenCastStreamPrivate* that =
+      static_cast<SharedScreenCastStreamPrivate*>(data);
+  RTC_DCHECK(that);
+
+  {
+    PipeWireThreadLoopLock thread_loop_lock(that->pw_main_loop_);
+
+    uint8_t buffer[2048] = {};
+
+    spa_pod_builder builder = spa_pod_builder{buffer, sizeof(buffer)};
+
+    std::vector<const spa_pod*> params;
+    struct spa_rectangle resolution =
+        SPA_RECTANGLE(that->width_, that->height_);
+
+    webrtc::MutexLock lock(&that->resolution_lock_);
+    for (uint32_t format : {SPA_VIDEO_FORMAT_BGRA, SPA_VIDEO_FORMAT_RGBA,
+                            SPA_VIDEO_FORMAT_BGRx, SPA_VIDEO_FORMAT_RGBx}) {
+      if (!that->modifiers_.empty()) {
+        params.push_back(BuildFormat(
+            &builder, format, that->modifiers_,
+            that->pending_resolution_change_ ? &resolution : nullptr));
+      }
+      params.push_back(BuildFormat(
+          &builder, format, /*modifiers=*/{},
+          that->pending_resolution_change_ ? &resolution : nullptr));
+    }
+
+    pw_stream_update_params(that->pw_stream_, params.data(), params.size());
+    that->pending_resolution_change_ = false;
+  }
+}
+
+SharedScreenCastStreamPrivate::SharedScreenCastStreamPrivate() {}
+
+SharedScreenCastStreamPrivate::~SharedScreenCastStreamPrivate() {
+  StopAndCleanupStream();
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+bool SharedScreenCastStreamPrivate::StartScreenCastStream(
+    uint32_t stream_node_id,
+    int fd,
+    uint32_t width,
+    uint32_t height,
+    bool is_cursor_embedded) {
+  width_ = width;
+  height_ = height;
+  is_cursor_embedded_ = is_cursor_embedded;
+  if (!InitializePipeWire()) {
+    RTC_LOG(LS_ERROR) << "Unable to open PipeWire library";
+    return false;
+  }
+  egl_dmabuf_ = std::make_unique<EglDmaBuf>();
+
+  pw_stream_node_id_ = stream_node_id;
+
+  pw_init(/*argc=*/nullptr, /*argc=*/nullptr);
+
+  pw_main_loop_ = pw_thread_loop_new("pipewire-main-loop", nullptr);
+
+  pw_context_ =
+      pw_context_new(pw_thread_loop_get_loop(pw_main_loop_), nullptr, 0);
+  if (!pw_context_) {
+    RTC_LOG(LS_ERROR) << "Failed to create PipeWire context";
+    return false;
+  }
+
+  if (pw_thread_loop_start(pw_main_loop_) < 0) {
+    RTC_LOG(LS_ERROR) << "Failed to start main PipeWire loop";
+    return false;
+  }
+
+  pw_client_version_ = PipeWireVersion::Parse(pw_get_library_version());
+
+  // Initialize event handlers, remote end and stream-related.
+  pw_core_events_.version = PW_VERSION_CORE_EVENTS;
+  pw_core_events_.info = &OnCoreInfo;
+  pw_core_events_.done = &OnCoreDone;
+  pw_core_events_.error = &OnCoreError;
+
+  pw_stream_events_.version = PW_VERSION_STREAM_EVENTS;
+  pw_stream_events_.state_changed = &OnStreamStateChanged;
+  pw_stream_events_.param_changed = &OnStreamParamChanged;
+  pw_stream_events_.process = &OnStreamProcess;
+
+  {
+    PipeWireThreadLoopLock thread_loop_lock(pw_main_loop_);
+
+    if (fd >= 0) {
+      pw_core_ = pw_context_connect_fd(
+          pw_context_, fcntl(fd, F_DUPFD_CLOEXEC, 0), nullptr, 0);
+    } else {
+      pw_core_ = pw_context_connect(pw_context_, nullptr, 0);
+    }
+
+    if (!pw_core_) {
+      RTC_LOG(LS_ERROR) << "Failed to connect PipeWire context";
+      return false;
+    }
+
+    pw_core_add_listener(pw_core_, &spa_core_listener_, &pw_core_events_, this);
+
+    // Add an event that can be later invoked by pw_loop_signal_event()
+    renegotiate_ = pw_loop_add_event(pw_thread_loop_get_loop(pw_main_loop_),
+                                     OnRenegotiateFormat, this);
+
+    server_version_sync_ =
+        pw_core_sync(pw_core_, PW_ID_CORE, server_version_sync_);
+
+    pw_thread_loop_wait(pw_main_loop_);
+
+    pw_properties* reuseProps =
+        pw_properties_new_string("pipewire.client.reuse=1");
+    pw_stream_ = pw_stream_new(pw_core_, "webrtc-consume-stream", reuseProps);
+
+    if (!pw_stream_) {
+      RTC_LOG(LS_ERROR) << "Failed to create PipeWire stream";
+      return false;
+    }
+
+    pw_stream_add_listener(pw_stream_, &spa_stream_listener_,
+                           &pw_stream_events_, this);
+    uint8_t buffer[2048] = {};
+
+    spa_pod_builder builder = spa_pod_builder{buffer, sizeof(buffer)};
+
+    std::vector<const spa_pod*> params;
+    const bool has_required_pw_client_version =
+        pw_client_version_ >= kDmaBufModifierMinVersion;
+    const bool has_required_pw_server_version =
+        pw_server_version_ >= kDmaBufModifierMinVersion;
+    struct spa_rectangle resolution;
+    bool set_resolution = false;
+    if (width && height) {
+      resolution = SPA_RECTANGLE(width, height);
+      set_resolution = true;
+    }
+    for (uint32_t format : {SPA_VIDEO_FORMAT_BGRA, SPA_VIDEO_FORMAT_RGBA,
+                            SPA_VIDEO_FORMAT_BGRx, SPA_VIDEO_FORMAT_RGBx}) {
+      // Modifiers can be used with PipeWire >= 0.3.33
+      if (has_required_pw_client_version && has_required_pw_server_version) {
+        modifiers_ = egl_dmabuf_->QueryDmaBufModifiers(format);
+
+        if (!modifiers_.empty()) {
+          params.push_back(BuildFormat(&builder, format, modifiers_,
+                                       set_resolution ? &resolution : nullptr));
+        }
+      }
+
+      params.push_back(BuildFormat(&builder, format, /*modifiers=*/{},
+                                   set_resolution ? &resolution : nullptr));
+    }
+
+    if (pw_stream_connect(pw_stream_, PW_DIRECTION_INPUT, pw_stream_node_id_,
+                          PW_STREAM_FLAG_AUTOCONNECT, params.data(),
+                          params.size()) != 0) {
+      RTC_LOG(LS_ERROR) << "Could not connect receiving stream.";
+      return false;
+    }
+
+    RTC_LOG(LS_INFO) << "PipeWire remote opened.";
+  }
+  return true;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+void SharedScreenCastStreamPrivate::UpdateScreenCastStreamResolution(
+    uint32_t width,
+    uint32_t height) {
+  if (!width || !height) {
+    RTC_LOG(LS_WARNING) << "Bad resolution specified: " << width << "x"
+                        << height;
+    return;
+  }
+  if (!pw_main_loop_) {
+    RTC_LOG(LS_WARNING) << "No main pipewire loop, ignoring resolution change";
+    return;
+  }
+  if (!renegotiate_) {
+    RTC_LOG(LS_WARNING) << "Can not renegotiate stream params, ignoring "
+                        << "resolution change";
+    return;
+  }
+  if (width_ != width || height_ != height) {
+    width_ = width;
+    height_ = height;
+    {
+      webrtc::MutexLock lock(&resolution_lock_);
+      pending_resolution_change_ = true;
+    }
+    pw_loop_signal_event(pw_thread_loop_get_loop(pw_main_loop_), renegotiate_);
+  }
+}
+
+void SharedScreenCastStreamPrivate::StopScreenCastStream() {
+  StopAndCleanupStream();
+}
+
+void SharedScreenCastStreamPrivate::StopAndCleanupStream() {
+  // We get buffers on the PipeWire thread, but this is called from the capturer
+  // thread, so we need to wait on and stop the pipewire thread before we
+  // disconnect the stream so that we can guarantee we aren't in the middle of
+  // processing a new frame.
+
+  // Even if we *do* somehow have the other objects without a pipewire thread,
+  // destroying them without a thread causes a crash.
+  if (!pw_main_loop_)
+    return;
+
+  // While we can stop the thread now, we cannot destroy it until we've cleaned
+  // up the other members.
+  pw_thread_loop_wait(pw_main_loop_);
+  pw_thread_loop_stop(pw_main_loop_);
+
+  if (pw_stream_) {
+    pw_stream_disconnect(pw_stream_);
+    pw_stream_destroy(pw_stream_);
+    pw_stream_ = nullptr;
+
+    {
+      webrtc::MutexLock lock(&queue_lock_);
+      queue_.Reset();
+    }
+  }
+
+  if (pw_core_) {
+    pw_core_disconnect(pw_core_);
+    pw_core_ = nullptr;
+  }
+
+  if (pw_context_) {
+    pw_context_destroy(pw_context_);
+    pw_context_ = nullptr;
+  }
+
+  pw_thread_loop_destroy(pw_main_loop_);
+  pw_main_loop_ = nullptr;
+}
+
+std::unique_ptr<SharedDesktopFrame>
+SharedScreenCastStreamPrivate::CaptureFrame() {
+  webrtc::MutexLock lock(&queue_lock_);
+
+  if (!pw_stream_ || !queue_.current_frame()) {
+    return std::unique_ptr<SharedDesktopFrame>{};
+  }
+
+  std::unique_ptr<SharedDesktopFrame> frame = queue_.current_frame()->Share();
+  if (use_damage_region_) {
+    frame->mutable_updated_region()->Swap(&damage_region_);
+    damage_region_.Clear();
+  }
+
+  return frame;
+}
+
+std::unique_ptr<MouseCursor> SharedScreenCastStreamPrivate::CaptureCursor() {
+  if (!mouse_cursor_) {
+    return nullptr;
+  }
+
+  return std::move(mouse_cursor_);
+}
+
+DesktopVector SharedScreenCastStreamPrivate::CaptureCursorPosition() {
+  return mouse_cursor_position_;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
+  spa_buffer* spa_buffer = buffer->buffer;
+
+  // Try to update the mouse cursor first, because it can be the only
+  // information carried by the buffer
+  {
+    const struct spa_meta_cursor* cursor =
+        static_cast<struct spa_meta_cursor*>(spa_buffer_find_meta_data(
+            spa_buffer, SPA_META_Cursor, sizeof(*cursor)));
+    if (cursor && spa_meta_cursor_is_valid(cursor)) {
+      struct spa_meta_bitmap* bitmap = nullptr;
+
+      if (cursor->bitmap_offset)
+        bitmap =
+            SPA_MEMBER(cursor, cursor->bitmap_offset, struct spa_meta_bitmap);
+
+      if (bitmap && bitmap->size.width > 0 && bitmap->size.height > 0) {
+        const uint8_t* bitmap_data =
+            SPA_MEMBER(bitmap, bitmap->offset, uint8_t);
+        BasicDesktopFrame* mouse_frame = new BasicDesktopFrame(
+            DesktopSize(bitmap->size.width, bitmap->size.height));
+        mouse_frame->CopyPixelsFrom(
+            bitmap_data, bitmap->stride,
+            DesktopRect::MakeWH(bitmap->size.width, bitmap->size.height));
+        mouse_cursor_ = std::make_unique<MouseCursor>(
+            mouse_frame, DesktopVector(cursor->hotspot.x, cursor->hotspot.y));
+
+        if (observer_) {
+          observer_->OnCursorShapeChanged();
+        }
+      }
+      mouse_cursor_position_.set(cursor->position.x, cursor->position.y);
+
+      if (observer_) {
+        observer_->OnCursorPositionChanged();
+      }
+    }
+  }
+
+  if (spa_buffer->datas[0].chunk->size == 0) {
+    return;
+  }
+
+  // Use SPA_META_VideoCrop metadata to get the frame size. KDE and GNOME do
+  // handle screen/window sharing differently. KDE/KWin doesn't use
+  // SPA_META_VideoCrop metadata and when sharing a window, it always sets
+  // stream size to size of the window. With that we just allocate the
+  // DesktopFrame using the size of the stream itself. GNOME/Mutter
+  // always sets stream size to the size of the whole screen, even when sharing
+  // a window. To get the real window size we have to use SPA_META_VideoCrop
+  // metadata. This gives us the size we need in order to allocate the
+  // DesktopFrame.
+
+  struct spa_meta_region* videocrop_metadata =
+      static_cast<struct spa_meta_region*>(spa_buffer_find_meta_data(
+          spa_buffer, SPA_META_VideoCrop, sizeof(*videocrop_metadata)));
+
+  // Video size from metadata is bigger than an actual video stream size.
+  // The metadata are wrong or we should up-scale the video...in both cases
+  // just quit now.
+  if (videocrop_metadata &&
+      (videocrop_metadata->region.size.width >
+           static_cast<uint32_t>(stream_size_.width()) ||
+       videocrop_metadata->region.size.height >
+           static_cast<uint32_t>(stream_size_.height()))) {
+    RTC_LOG(LS_ERROR) << "Stream metadata sizes are wrong!";
+
+    if (observer_) {
+      observer_->OnFailedToProcessBuffer();
+    }
+
+    return;
+  }
+
+  // Use SPA_META_VideoCrop metadata to get the DesktopFrame size in case
+  // a windows is shared and it represents just a small portion of the
+  // stream itself. This will be for example used in case of GNOME (Mutter)
+  // where the stream will have the size of the screen itself, but we care
+  // only about smaller portion representing the window inside.
+  bool videocrop_metadata_use = false;
+  const struct spa_rectangle* videocrop_metadata_size =
+      videocrop_metadata ? &videocrop_metadata->region.size : nullptr;
+
+  if (videocrop_metadata_size && videocrop_metadata_size->width != 0 &&
+      videocrop_metadata_size->height != 0 &&
+      (static_cast<int>(videocrop_metadata_size->width) <
+           stream_size_.width() ||
+       static_cast<int>(videocrop_metadata_size->height) <
+           stream_size_.height())) {
+    videocrop_metadata_use = true;
+  }
+
+  if (videocrop_metadata_use) {
+    frame_size_ = DesktopSize(videocrop_metadata_size->width,
+                              videocrop_metadata_size->height);
+  } else {
+    frame_size_ = stream_size_;
+  }
+
+  // Get the position of the video crop within the stream. Just double-check
+  // that the position doesn't exceed the size of the stream itself.
+  // NOTE: Currently it looks there is no implementation using this.
+  uint32_t y_offset =
+      videocrop_metadata_use &&
+              (videocrop_metadata->region.position.y + frame_size_.height() <=
+               stream_size_.height())
+          ? videocrop_metadata->region.position.y
+          : 0;
+  uint32_t x_offset =
+      videocrop_metadata_use &&
+              (videocrop_metadata->region.position.x + frame_size_.width() <=
+               stream_size_.width())
+          ? videocrop_metadata->region.position.x
+          : 0;
+  DesktopVector offset = DesktopVector(x_offset, y_offset);
+
+  webrtc::MutexLock lock(&queue_lock_);
+
+  queue_.MoveToNextFrame();
+  if (queue_.current_frame() && queue_.current_frame()->IsShared()) {
+    RTC_DLOG(LS_WARNING) << "Overwriting frame that is still shared";
+
+    if (observer_) {
+      observer_->OnFailedToProcessBuffer();
+    }
+  }
+
+  if (!queue_.current_frame() ||
+      !queue_.current_frame()->size().equals(frame_size_)) {
+    std::unique_ptr<DesktopFrame> frame(new BasicDesktopFrame(
+        DesktopSize(frame_size_.width(), frame_size_.height())));
+    queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(std::move(frame)));
+  }
+
+  bool bufferProcessed = false;
+  if (spa_buffer->datas[0].type == SPA_DATA_MemFd) {
+    bufferProcessed =
+        ProcessMemFDBuffer(buffer, *queue_.current_frame(), offset);
+  } else if (spa_buffer->datas[0].type == SPA_DATA_DmaBuf) {
+    bufferProcessed = ProcessDMABuffer(buffer, *queue_.current_frame(), offset);
+  }
+
+  if (!bufferProcessed) {
+    if (observer_) {
+      observer_->OnFailedToProcessBuffer();
+    }
+    return;
+  }
+
+  if (spa_video_format_.format == SPA_VIDEO_FORMAT_RGBx ||
+      spa_video_format_.format == SPA_VIDEO_FORMAT_RGBA) {
+    uint8_t* tmp_src = queue_.current_frame()->data();
+    for (int i = 0; i < frame_size_.height(); ++i) {
+      // If both sides decided to go with the RGBx format we need to convert
+      // it to BGRx to match color format expected by WebRTC.
+      ConvertRGBxToBGRx(tmp_src, queue_.current_frame()->stride());
+      tmp_src += queue_.current_frame()->stride();
+    }
+  }
+
+  if (observer_) {
+    observer_->OnDesktopFrameChanged();
+  }
+
+  if (use_damage_region_) {
+    const struct spa_meta* video_damage = static_cast<struct spa_meta*>(
+        spa_buffer_find_meta(spa_buffer, SPA_META_VideoDamage));
+    if (video_damage) {
+      spa_meta_region* meta_region;
+
+      queue_.current_frame()->mutable_updated_region()->Clear();
+
+      spa_meta_for_each(meta_region, video_damage) {
+        // Skip empty regions
+        if (meta_region->region.size.width == 0 ||
+            meta_region->region.size.height == 0) {
+          continue;
+        }
+
+        damage_region_.AddRect(DesktopRect::MakeXYWH(
+            meta_region->region.position.x, meta_region->region.position.y,
+            meta_region->region.size.width, meta_region->region.size.height));
+      }
+    } else {
+      damage_region_.SetRect(
+          DesktopRect::MakeSize(queue_.current_frame()->size()));
+    }
+  } else {
+    queue_.current_frame()->mutable_updated_region()->SetRect(
+        DesktopRect::MakeSize(queue_.current_frame()->size()));
+  }
+  queue_.current_frame()->set_may_contain_cursor(is_cursor_embedded_);
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+bool SharedScreenCastStreamPrivate::ProcessMemFDBuffer(
+    pw_buffer* buffer,
+    DesktopFrame& frame,
+    const DesktopVector& offset) {
+  spa_buffer* spa_buffer = buffer->buffer;
+  ScopedBuf map;
+  uint8_t* src = nullptr;
+
+  map.initialize(
+      static_cast<uint8_t*>(
+          mmap(nullptr,
+               spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
+               PROT_READ, MAP_PRIVATE, spa_buffer->datas[0].fd, 0)),
+      spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
+      spa_buffer->datas[0].fd);
+
+  if (!map) {
+    RTC_LOG(LS_ERROR) << "Failed to mmap the memory: " << std::strerror(errno);
+    return false;
+  }
+
+  src = SPA_MEMBER(map.get(), spa_buffer->datas[0].mapoffset, uint8_t);
+
+  uint32_t buffer_stride = spa_buffer->datas[0].chunk->stride;
+  uint32_t src_stride = buffer_stride;
+
+  uint8_t* updated_src =
+      src + (src_stride * offset.y()) + (kBytesPerPixel * offset.x());
+
+  frame.CopyPixelsFrom(
+      updated_src, (src_stride - (kBytesPerPixel * offset.x())),
+      DesktopRect::MakeWH(frame.size().width(), frame.size().height()));
+
+  return true;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+bool SharedScreenCastStreamPrivate::ProcessDMABuffer(
+    pw_buffer* buffer,
+    DesktopFrame& frame,
+    const DesktopVector& offset) {
+  spa_buffer* spa_buffer = buffer->buffer;
+
+  const uint n_planes = spa_buffer->n_datas;
+
+  if (!n_planes) {
+    return false;
+  }
+
+  std::vector<EglDmaBuf::PlaneData> plane_datas;
+  for (uint32_t i = 0; i < n_planes; ++i) {
+    EglDmaBuf::PlaneData data = {
+        static_cast<int32_t>(spa_buffer->datas[i].fd),
+        static_cast<uint32_t>(spa_buffer->datas[i].chunk->stride),
+        static_cast<uint32_t>(spa_buffer->datas[i].chunk->offset)};
+    plane_datas.push_back(data);
+  }
+
+  const bool imported = egl_dmabuf_->ImageFromDmaBuf(
+      stream_size_, spa_video_format_.format, plane_datas, modifier_, offset,
+      frame.size(), frame.data());
+  if (!imported) {
+    RTC_LOG(LS_ERROR) << "Dropping DMA-BUF modifier: " << modifier_
+                      << " and trying to renegotiate stream parameters";
+
+    if (pw_server_version_ >= kDropSingleModifierMinVersion) {
+      modifiers_.erase(
+          std::remove(modifiers_.begin(), modifiers_.end(), modifier_),
+          modifiers_.end());
+    } else {
+      modifiers_.clear();
+    }
+
+    pw_loop_signal_event(pw_thread_loop_get_loop(pw_main_loop_), renegotiate_);
+    return false;
+  }
+
+  return true;
+}
+
+void SharedScreenCastStreamPrivate::ConvertRGBxToBGRx(uint8_t* frame,
+                                                      uint32_t size) {
+  for (uint32_t i = 0; i < size; i += 4) {
+    uint8_t tempR = frame[i];
+    uint8_t tempB = frame[i + 2];
+    frame[i] = tempB;
+    frame[i + 2] = tempR;
+  }
+}
+
+SharedScreenCastStream::SharedScreenCastStream()
+    : private_(std::make_unique<SharedScreenCastStreamPrivate>()) {}
+
+SharedScreenCastStream::~SharedScreenCastStream() {}
+
+rtc::scoped_refptr<SharedScreenCastStream>
+SharedScreenCastStream::CreateDefault() {
+  // Explicit new, to access non-public constructor.
+  return rtc::scoped_refptr<SharedScreenCastStream>(
+      new SharedScreenCastStream());
+}
+
+bool SharedScreenCastStream::StartScreenCastStream(uint32_t stream_node_id) {
+  return private_->StartScreenCastStream(stream_node_id, -1);
+}
+
+bool SharedScreenCastStream::StartScreenCastStream(uint32_t stream_node_id,
+                                                   int fd,
+                                                   uint32_t width,
+                                                   uint32_t height,
+                                                   bool is_cursor_embedded) {
+  return private_->StartScreenCastStream(stream_node_id, fd, width, height,
+                                         is_cursor_embedded);
+}
+
+void SharedScreenCastStream::UpdateScreenCastStreamResolution(uint32_t width,
+                                                              uint32_t height) {
+  private_->UpdateScreenCastStreamResolution(width, height);
+}
+
+void SharedScreenCastStream::SetUseDamageRegion(bool use_damage_region) {
+  private_->SetUseDamageRegion(use_damage_region);
+}
+
+void SharedScreenCastStream::SetObserver(
+    SharedScreenCastStream::Observer* observer) {
+  private_->SetObserver(observer);
+}
+
+void SharedScreenCastStream::StopScreenCastStream() {
+  private_->StopScreenCastStream();
+}
+
+std::unique_ptr<SharedDesktopFrame> SharedScreenCastStream::CaptureFrame() {
+  return private_->CaptureFrame();
+}
+
+std::unique_ptr<MouseCursor> SharedScreenCastStream::CaptureCursor() {
+  return private_->CaptureCursor();
+}
+
+absl::optional<DesktopVector> SharedScreenCastStream::CaptureCursorPosition() {
+  DesktopVector position = private_->CaptureCursorPosition();
+
+  // Consider only (x >= 0 and y >= 0) a valid position
+  if (position.x() < 0 || position.y() < 0) {
+    return absl::nullopt;
+  }
+
+  return position;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.h
new file mode 100644
index 0000000000..9cdd3d89be
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SHARED_SCREENCAST_STREAM_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SHARED_SCREENCAST_STREAM_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/screen_capture_frame_queue.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class SharedScreenCastStreamPrivate;
+
+class RTC_EXPORT SharedScreenCastStream
+    : public rtc::RefCountedNonVirtual<SharedScreenCastStream> {
+ public:
+  class Observer {
+   public:
+    virtual void OnCursorPositionChanged() = 0;
+    virtual void OnCursorShapeChanged() = 0;
+    virtual void OnDesktopFrameChanged() = 0;
+    virtual void OnFailedToProcessBuffer() = 0;
+    virtual void OnStreamConfigured() = 0;
+
+   protected:
+    Observer() = default;
+    virtual ~Observer() = default;
+  };
+
+  static rtc::scoped_refptr<SharedScreenCastStream> CreateDefault();
+
+  bool StartScreenCastStream(uint32_t stream_node_id);
+  bool StartScreenCastStream(uint32_t stream_node_id,
+                             int fd,
+                             uint32_t width = 0,
+                             uint32_t height = 0,
+                             bool is_cursor_embedded = false);
+  void UpdateScreenCastStreamResolution(uint32_t width, uint32_t height);
+  void SetUseDamageRegion(bool use_damage_region);
+  void SetObserver(SharedScreenCastStream::Observer* observer);
+  void StopScreenCastStream();
+
+  // Below functions return the most recent information we get from a
+  // PipeWire buffer on each Process() callback. This assumes that we
+  // managed to successfuly connect to a PipeWire stream provided by the
+  // compositor (based on stream parameters). The cursor data are obtained
+  // from spa_meta_cursor stream metadata and therefore the cursor is not
+  // part of actual screen/window frame.
+
+  // Returns the most recent screen/window frame we obtained from PipeWire
+  // buffer. Will return an empty frame in case we didn't manage to get a frame
+  // from PipeWire buffer.
+  std::unique_ptr<SharedDesktopFrame> CaptureFrame();
+
+  // Returns the most recent mouse cursor image. Will return an nullptr cursor
+  // in case we didn't manage to get a cursor from PipeWire buffer. NOTE: the
+  // cursor image might not be updated on every cursor location change, but
+  // actually only when its shape changes.
+  std::unique_ptr<MouseCursor> CaptureCursor();
+
+  // Returns the most recent mouse cursor position. Will not return a value in
+  // case we didn't manage to get it from PipeWire buffer.
+  absl::optional<DesktopVector> CaptureCursorPosition();
+
+  ~SharedScreenCastStream();
+
+ protected:
+  SharedScreenCastStream();
+
+ private:
+  friend class SharedScreenCastStreamPrivate;
+
+  SharedScreenCastStream(const SharedScreenCastStream&) = delete;
+  SharedScreenCastStream& operator=(const SharedScreenCastStream&) = delete;
+
+  std::unique_ptr<SharedScreenCastStreamPrivate> private_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_SHARED_SCREENCAST_STREAM_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream_unittest.cc
new file mode 100644
index 0000000000..1de5f19013
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/shared_screencast_stream_unittest.cc
@@ -0,0 +1,160 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/wayland/shared_screencast_stream.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/units/time_delta.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/linux/wayland/test/test_screencast_stream_provider.h"
+#include "modules/desktop_capture/rgba_color.h"
+#include "rtc_base/event.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Ge;
+using ::testing::Invoke;
+
+namespace webrtc {
+
+constexpr TimeDelta kShortWait = TimeDelta::Seconds(5);
+constexpr TimeDelta kLongWait = TimeDelta::Seconds(15);
+
+constexpr int kBytesPerPixel = 4;
+constexpr int32_t kWidth = 800;
+constexpr int32_t kHeight = 640;
+
+class PipeWireStreamTest : public ::testing::Test,
+                           public TestScreenCastStreamProvider::Observer,
+                           public SharedScreenCastStream::Observer {
+ public:
+  PipeWireStreamTest() = default;
+  ~PipeWireStreamTest() = default;
+
+  // FakeScreenCastPortal::Observer
+  MOCK_METHOD(void, OnBufferAdded, (), (override));
+  MOCK_METHOD(void, OnFrameRecorded, (), (override));
+  MOCK_METHOD(void, OnStreamReady, (uint32_t stream_node_id), (override));
+  MOCK_METHOD(void, OnStartStreaming, (), (override));
+  MOCK_METHOD(void, OnStopStreaming, (), (override));
+
+  // SharedScreenCastStream::Observer
+  MOCK_METHOD(void, OnCursorPositionChanged, (), (override));
+  MOCK_METHOD(void, OnCursorShapeChanged, (), (override));
+  MOCK_METHOD(void, OnDesktopFrameChanged, (), (override));
+  MOCK_METHOD(void, OnFailedToProcessBuffer, (), (override));
+  MOCK_METHOD(void, OnStreamConfigured, (), (override));
+
+  void SetUp() override {
+    shared_screencast_stream_ = SharedScreenCastStream::CreateDefault();
+    shared_screencast_stream_->SetObserver(this);
+    test_screencast_stream_provider_ =
+        std::make_unique<TestScreenCastStreamProvider>(this, kWidth, kHeight);
+  }
+
+  void StartScreenCastStream(uint32_t stream_node_id) {
+    shared_screencast_stream_->StartScreenCastStream(stream_node_id);
+  }
+
+ protected:
+  uint recorded_frames_ = 0;
+  bool streaming_ = false;
+  std::unique_ptr<TestScreenCastStreamProvider>
+      test_screencast_stream_provider_;
+  rtc::scoped_refptr<SharedScreenCastStream> shared_screencast_stream_;
+};
+
+TEST_F(PipeWireStreamTest, TestPipeWire) {
+  // Set expectations for PipeWire to successfully connect both streams
+  rtc::Event waitConnectEvent;
+  rtc::Event waitStartStreamingEvent;
+
+  EXPECT_CALL(*this, OnStreamReady(_))
+      .WillOnce(Invoke(this, &PipeWireStreamTest::StartScreenCastStream));
+  EXPECT_CALL(*this, OnStreamConfigured).WillOnce([&waitConnectEvent] {
+    waitConnectEvent.Set();
+  });
+  EXPECT_CALL(*this, OnBufferAdded).Times(AtLeast(3));
+  EXPECT_CALL(*this, OnStartStreaming).WillOnce([&waitStartStreamingEvent] {
+    waitStartStreamingEvent.Set();
+  });
+
+  // Give it some time to connect, the order between these shouldn't matter, but
+  // we need to be sure we are connected before we proceed to work with frames.
+  waitConnectEvent.Wait(kLongWait);
+
+  // Wait until we start streaming
+  waitStartStreamingEvent.Wait(kShortWait);
+
+  rtc::Event frameRetrievedEvent;
+  EXPECT_CALL(*this, OnFrameRecorded).Times(3);
+  EXPECT_CALL(*this, OnDesktopFrameChanged)
+      .WillRepeatedly([&frameRetrievedEvent] { frameRetrievedEvent.Set(); });
+
+  // Record a frame in FakePipeWireStream
+  RgbaColor red_color(0, 0, 255);
+  test_screencast_stream_provider_->RecordFrame(red_color);
+
+  // Retrieve a frame from SharedScreenCastStream
+  frameRetrievedEvent.Wait(kShortWait);
+  std::unique_ptr<SharedDesktopFrame> frame =
+      shared_screencast_stream_->CaptureFrame();
+
+  // Check frame parameters
+  ASSERT_NE(frame, nullptr);
+  ASSERT_NE(frame->data(), nullptr);
+  EXPECT_EQ(frame->rect().width(), kWidth);
+  EXPECT_EQ(frame->rect().height(), kHeight);
+  EXPECT_EQ(frame->stride(), frame->rect().width() * kBytesPerPixel);
+  EXPECT_EQ(RgbaColor(frame->data()), red_color);
+
+  // Test DesktopFrameQueue
+  RgbaColor green_color(0, 255, 0);
+  test_screencast_stream_provider_->RecordFrame(green_color);
+  frameRetrievedEvent.Wait(kShortWait);
+  std::unique_ptr<SharedDesktopFrame> frame2 =
+      shared_screencast_stream_->CaptureFrame();
+  ASSERT_NE(frame2, nullptr);
+  ASSERT_NE(frame2->data(), nullptr);
+  EXPECT_EQ(frame2->rect().width(), kWidth);
+  EXPECT_EQ(frame2->rect().height(), kHeight);
+  EXPECT_EQ(frame2->stride(), frame->rect().width() * kBytesPerPixel);
+  EXPECT_EQ(RgbaColor(frame2->data()), green_color);
+
+  // Thanks to DesktopFrameQueue we should be able to have two frames shared
+  EXPECT_EQ(frame->IsShared(), true);
+  EXPECT_EQ(frame2->IsShared(), true);
+  EXPECT_NE(frame->data(), frame2->data());
+
+  // This should result into overwriting a frame in use
+  rtc::Event frameRecordedEvent;
+  RgbaColor blue_color(255, 0, 0);
+  EXPECT_CALL(*this, OnFailedToProcessBuffer).WillOnce([&frameRecordedEvent] {
+    frameRecordedEvent.Set();
+  });
+
+  test_screencast_stream_provider_->RecordFrame(blue_color);
+  frameRecordedEvent.Wait(kShortWait);
+
+  // First frame should be now overwritten with blue color
+  frameRetrievedEvent.Wait(kShortWait);
+  EXPECT_EQ(RgbaColor(frame->data()), blue_color);
+
+  // Test disconnection from stream
+  EXPECT_CALL(*this, OnStopStreaming);
+  shared_screencast_stream_->StopScreenCastStream();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/test/test_screencast_stream_provider.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/test/test_screencast_stream_provider.cc
new file mode 100644
index 0000000000..3b829959ac
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/test/test_screencast_stream_provider.cc
@@ -0,0 +1,361 @@
+
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/wayland/test/test_screencast_stream_provider.h"
+
+#include <fcntl.h>
+#include <sys/mman.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "modules/portal/pipewire_utils.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+constexpr int kBytesPerPixel = 4;
+
+TestScreenCastStreamProvider::TestScreenCastStreamProvider(Observer* observer,
+                                                           uint32_t width,
+                                                           uint32_t height)
+    : observer_(observer), width_(width), height_(height) {
+  if (!InitializePipeWire()) {
+    RTC_LOG(LS_ERROR) << "Unable to open PipeWire";
+    return;
+  }
+
+  pw_init(/*argc=*/nullptr, /*argc=*/nullptr);
+
+  pw_main_loop_ = pw_thread_loop_new("pipewire-test-main-loop", nullptr);
+
+  pw_context_ =
+      pw_context_new(pw_thread_loop_get_loop(pw_main_loop_), nullptr, 0);
+  if (!pw_context_) {
+    RTC_LOG(LS_ERROR) << "PipeWire test: Failed to create PipeWire context";
+    return;
+  }
+
+  if (pw_thread_loop_start(pw_main_loop_) < 0) {
+    RTC_LOG(LS_ERROR) << "PipeWire test: Failed to start main PipeWire loop";
+    return;
+  }
+
+  // Initialize event handlers, remote end and stream-related.
+  pw_core_events_.version = PW_VERSION_CORE_EVENTS;
+  pw_core_events_.error = &OnCoreError;
+
+  pw_stream_events_.version = PW_VERSION_STREAM_EVENTS;
+  pw_stream_events_.add_buffer = &OnStreamAddBuffer;
+  pw_stream_events_.remove_buffer = &OnStreamRemoveBuffer;
+  pw_stream_events_.state_changed = &OnStreamStateChanged;
+  pw_stream_events_.param_changed = &OnStreamParamChanged;
+
+  {
+    PipeWireThreadLoopLock thread_loop_lock(pw_main_loop_);
+
+    pw_core_ = pw_context_connect(pw_context_, nullptr, 0);
+    if (!pw_core_) {
+      RTC_LOG(LS_ERROR) << "PipeWire test: Failed to connect PipeWire context";
+      return;
+    }
+
+    pw_core_add_listener(pw_core_, &spa_core_listener_, &pw_core_events_, this);
+
+    pw_stream_ = pw_stream_new(pw_core_, "webrtc-test-stream", nullptr);
+
+    if (!pw_stream_) {
+      RTC_LOG(LS_ERROR) << "PipeWire test: Failed to create PipeWire stream";
+      return;
+    }
+
+    pw_stream_add_listener(pw_stream_, &spa_stream_listener_,
+                           &pw_stream_events_, this);
+    uint8_t buffer[2048] = {};
+
+    spa_pod_builder builder = spa_pod_builder{buffer, sizeof(buffer)};
+
+    std::vector<const spa_pod*> params;
+
+    spa_rectangle resolution =
+        SPA_RECTANGLE(uint32_t(width_), uint32_t(height_));
+    params.push_back(BuildFormat(&builder, SPA_VIDEO_FORMAT_BGRx,
+                                 /*modifiers=*/{}, &resolution));
+
+    auto flags =
+        pw_stream_flags(PW_STREAM_FLAG_DRIVER | PW_STREAM_FLAG_ALLOC_BUFFERS);
+    if (pw_stream_connect(pw_stream_, PW_DIRECTION_OUTPUT, SPA_ID_INVALID,
+                          flags, params.data(), params.size()) != 0) {
+      RTC_LOG(LS_ERROR) << "PipeWire test: Could not connect receiving stream.";
+      pw_stream_destroy(pw_stream_);
+      pw_stream_ = nullptr;
+      return;
+    }
+  }
+
+  return;
+}
+
+TestScreenCastStreamProvider::~TestScreenCastStreamProvider() {
+  if (pw_main_loop_) {
+    pw_thread_loop_stop(pw_main_loop_);
+  }
+
+  if (pw_stream_) {
+    pw_stream_destroy(pw_stream_);
+  }
+
+  if (pw_core_) {
+    pw_core_disconnect(pw_core_);
+  }
+
+  if (pw_context_) {
+    pw_context_destroy(pw_context_);
+  }
+
+  if (pw_main_loop_) {
+    pw_thread_loop_destroy(pw_main_loop_);
+  }
+}
+
+void TestScreenCastStreamProvider::RecordFrame(RgbaColor rgba_color) {
+  const char* error;
+  if (pw_stream_get_state(pw_stream_, &error) != PW_STREAM_STATE_STREAMING) {
+    if (error) {
+      RTC_LOG(LS_ERROR)
+          << "PipeWire test: Failed to record frame: stream is not active: "
+          << error;
+    }
+  }
+
+  struct pw_buffer* buffer = pw_stream_dequeue_buffer(pw_stream_);
+  if (!buffer) {
+    RTC_LOG(LS_ERROR) << "PipeWire test: No available buffer";
+    return;
+  }
+
+  struct spa_buffer* spa_buffer = buffer->buffer;
+  struct spa_data* spa_data = spa_buffer->datas;
+  uint8_t* data = static_cast<uint8_t*>(spa_data->data);
+  if (!data) {
+    RTC_LOG(LS_ERROR)
+        << "PipeWire test: Failed to record frame: invalid buffer data";
+    pw_stream_queue_buffer(pw_stream_, buffer);
+    return;
+  }
+
+  const int stride = SPA_ROUND_UP_N(width_ * kBytesPerPixel, 4);
+
+  spa_data->chunk->offset = 0;
+  spa_data->chunk->size = height_ * stride;
+  spa_data->chunk->stride = stride;
+
+  uint32_t color = rgba_color.ToUInt32();
+  for (uint32_t i = 0; i < height_; i++) {
+    uint32_t* column = reinterpret_cast<uint32_t*>(data);
+    for (uint32_t j = 0; j < width_; j++) {
+      column[j] = color;
+    }
+    data += stride;
+  }
+
+  pw_stream_queue_buffer(pw_stream_, buffer);
+  if (observer_) {
+    observer_->OnFrameRecorded();
+  }
+}
+
+void TestScreenCastStreamProvider::StartStreaming() {
+  if (pw_stream_ && pw_node_id_ != 0) {
+    pw_stream_set_active(pw_stream_, true);
+  }
+}
+
+void TestScreenCastStreamProvider::StopStreaming() {
+  if (pw_stream_ && pw_node_id_ != 0) {
+    pw_stream_set_active(pw_stream_, false);
+  }
+}
+
+// static
+void TestScreenCastStreamProvider::OnCoreError(void* data,
+                                               uint32_t id,
+                                               int seq,
+                                               int res,
+                                               const char* message) {
+  TestScreenCastStreamProvider* that =
+      static_cast<TestScreenCastStreamProvider*>(data);
+  RTC_DCHECK(that);
+
+  RTC_LOG(LS_ERROR) << "PipeWire test: PipeWire remote error: " << message;
+}
+
+// static
+void TestScreenCastStreamProvider::OnStreamStateChanged(
+    void* data,
+    pw_stream_state old_state,
+    pw_stream_state state,
+    const char* error_message) {
+  TestScreenCastStreamProvider* that =
+      static_cast<TestScreenCastStreamProvider*>(data);
+  RTC_DCHECK(that);
+
+  switch (state) {
+    case PW_STREAM_STATE_ERROR:
+      RTC_LOG(LS_ERROR) << "PipeWire test: PipeWire stream state error: "
+                        << error_message;
+      break;
+    case PW_STREAM_STATE_PAUSED:
+      if (that->pw_node_id_ == 0 && that->pw_stream_) {
+        that->pw_node_id_ = pw_stream_get_node_id(that->pw_stream_);
+        that->observer_->OnStreamReady(that->pw_node_id_);
+      } else {
+        // Stop streaming
+        that->is_streaming_ = false;
+        that->observer_->OnStopStreaming();
+      }
+      break;
+    case PW_STREAM_STATE_STREAMING:
+      // Start streaming
+      that->is_streaming_ = true;
+      that->observer_->OnStartStreaming();
+      break;
+    case PW_STREAM_STATE_CONNECTING:
+      break;
+    case PW_STREAM_STATE_UNCONNECTED:
+      if (that->is_streaming_) {
+        // Stop streaming
+        that->is_streaming_ = false;
+        that->observer_->OnStopStreaming();
+      }
+      break;
+  }
+}
+
+// static
+void TestScreenCastStreamProvider::OnStreamParamChanged(
+    void* data,
+    uint32_t id,
+    const struct spa_pod* format) {
+  TestScreenCastStreamProvider* that =
+      static_cast<TestScreenCastStreamProvider*>(data);
+  RTC_DCHECK(that);
+
+  RTC_LOG(LS_INFO) << "PipeWire test: PipeWire stream format changed.";
+  if (!format || id != SPA_PARAM_Format) {
+    return;
+  }
+
+  spa_format_video_raw_parse(format, &that->spa_video_format_);
+
+  auto stride = SPA_ROUND_UP_N(that->width_ * kBytesPerPixel, 4);
+
+  uint8_t buffer[1024] = {};
+  auto builder = spa_pod_builder{buffer, sizeof(buffer)};
+
+  // Setup buffers and meta header for new format.
+
+  std::vector<const spa_pod*> params;
+  const int buffer_types = (1 << SPA_DATA_MemFd);
+  spa_rectangle resolution = SPA_RECTANGLE(that->width_, that->height_);
+
+  params.push_back(reinterpret_cast<spa_pod*>(spa_pod_builder_add_object(
+      &builder, SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
+      SPA_FORMAT_VIDEO_size, SPA_POD_Rectangle(&resolution),
+      SPA_PARAM_BUFFERS_buffers, SPA_POD_CHOICE_RANGE_Int(16, 2, 16),
+      SPA_PARAM_BUFFERS_blocks, SPA_POD_Int(1), SPA_PARAM_BUFFERS_stride,
+      SPA_POD_Int(stride), SPA_PARAM_BUFFERS_size,
+      SPA_POD_Int(stride * that->height_), SPA_PARAM_BUFFERS_align,
+      SPA_POD_Int(16), SPA_PARAM_BUFFERS_dataType,
+      SPA_POD_CHOICE_FLAGS_Int(buffer_types))));
+  params.push_back(reinterpret_cast<spa_pod*>(spa_pod_builder_add_object(
+      &builder, SPA_TYPE_OBJECT_ParamMeta, SPA_PARAM_Meta, SPA_PARAM_META_type,
+      SPA_POD_Id(SPA_META_Header), SPA_PARAM_META_size,
+      SPA_POD_Int(sizeof(struct spa_meta_header)))));
+
+  pw_stream_update_params(that->pw_stream_, params.data(), params.size());
+}
+
+// static
+void TestScreenCastStreamProvider::OnStreamAddBuffer(void* data,
+                                                     pw_buffer* buffer) {
+  TestScreenCastStreamProvider* that =
+      static_cast<TestScreenCastStreamProvider*>(data);
+  RTC_DCHECK(that);
+
+  struct spa_data* spa_data = buffer->buffer->datas;
+
+  spa_data->mapoffset = 0;
+  spa_data->flags = SPA_DATA_FLAG_READWRITE;
+
+  if (!(spa_data[0].type & (1 << SPA_DATA_MemFd))) {
+    RTC_LOG(LS_ERROR)
+        << "PipeWire test: Client doesn't support memfd buffer data type";
+    return;
+  }
+
+  const int stride = SPA_ROUND_UP_N(that->width_ * kBytesPerPixel, 4);
+  spa_data->maxsize = stride * that->height_;
+  spa_data->type = SPA_DATA_MemFd;
+  spa_data->fd =
+      memfd_create("pipewire-test-memfd", MFD_CLOEXEC | MFD_ALLOW_SEALING);
+  if (spa_data->fd == -1) {
+    RTC_LOG(LS_ERROR) << "PipeWire test: Can't create memfd";
+    return;
+  }
+
+  spa_data->mapoffset = 0;
+
+  if (ftruncate(spa_data->fd, spa_data->maxsize) < 0) {
+    RTC_LOG(LS_ERROR) << "PipeWire test: Can't truncate to"
+                      << spa_data->maxsize;
+    return;
+  }
+
+  unsigned int seals = F_SEAL_GROW | F_SEAL_SHRINK | F_SEAL_SEAL;
+  if (fcntl(spa_data->fd, F_ADD_SEALS, seals) == -1) {
+    RTC_LOG(LS_ERROR) << "PipeWire test: Failed to add seals";
+  }
+
+  spa_data->data = mmap(nullptr, spa_data->maxsize, PROT_READ | PROT_WRITE,
+                        MAP_SHARED, spa_data->fd, spa_data->mapoffset);
+  if (spa_data->data == MAP_FAILED) {
+    RTC_LOG(LS_ERROR) << "PipeWire test: Failed to mmap memory";
+  } else {
+    that->observer_->OnBufferAdded();
+    RTC_LOG(LS_INFO) << "PipeWire test: Memfd created successfully: "
+                     << spa_data->data << spa_data->maxsize;
+  }
+}
+
+// static
+void TestScreenCastStreamProvider::OnStreamRemoveBuffer(void* data,
+                                                        pw_buffer* buffer) {
+  TestScreenCastStreamProvider* that =
+      static_cast<TestScreenCastStreamProvider*>(data);
+  RTC_DCHECK(that);
+
+  struct spa_buffer* spa_buffer = buffer->buffer;
+  struct spa_data* spa_data = spa_buffer->datas;
+  if (spa_data && spa_data->type == SPA_DATA_MemFd) {
+    munmap(spa_data->data, spa_data->maxsize);
+    close(spa_data->fd);
+  }
+}
+
+uint32_t TestScreenCastStreamProvider::PipeWireNodeId() {
+  return pw_node_id_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/test/test_screencast_stream_provider.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/test/test_screencast_stream_provider.h
new file mode 100644
index 0000000000..d893aa63ab
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/test/test_screencast_stream_provider.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_TEST_TEST_SCREENCAST_STREAM_PROVIDER_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_TEST_TEST_SCREENCAST_STREAM_PROVIDER_H_
+
+#include <pipewire/pipewire.h>
+#include <spa/param/video/format-utils.h>
+
+#include "modules/desktop_capture/linux/wayland/screencast_stream_utils.h"
+#include "modules/desktop_capture/rgba_color.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+
+class TestScreenCastStreamProvider {
+ public:
+  class Observer {
+   public:
+    virtual void OnBufferAdded() = 0;
+    virtual void OnFrameRecorded() = 0;
+    virtual void OnStreamReady(uint32_t stream_node_id) = 0;
+    virtual void OnStartStreaming() = 0;
+    virtual void OnStopStreaming() = 0;
+
+   protected:
+    Observer() = default;
+    virtual ~Observer() = default;
+  };
+
+  explicit TestScreenCastStreamProvider(Observer* observer,
+                                        uint32_t width,
+                                        uint32_t height);
+  ~TestScreenCastStreamProvider();
+
+  uint32_t PipeWireNodeId();
+
+  void RecordFrame(RgbaColor rgba_color);
+  void StartStreaming();
+  void StopStreaming();
+
+ private:
+  Observer* observer_;
+
+  // Resolution parameters.
+  uint32_t width_ = 0;
+  uint32_t height_ = 0;
+
+  bool is_streaming_ = false;
+  uint32_t pw_node_id_ = 0;
+
+  // PipeWire types
+  struct pw_context* pw_context_ = nullptr;
+  struct pw_core* pw_core_ = nullptr;
+  struct pw_stream* pw_stream_ = nullptr;
+  struct pw_thread_loop* pw_main_loop_ = nullptr;
+
+  spa_hook spa_core_listener_;
+  spa_hook spa_stream_listener_;
+
+  // event handlers
+  pw_core_events pw_core_events_ = {};
+  pw_stream_events pw_stream_events_ = {};
+
+  struct spa_video_info_raw spa_video_format_;
+
+  // PipeWire callbacks
+  static void OnCoreError(void* data,
+                          uint32_t id,
+                          int seq,
+                          int res,
+                          const char* message);
+  static void OnStreamAddBuffer(void* data, pw_buffer* buffer);
+  static void OnStreamRemoveBuffer(void* data, pw_buffer* buffer);
+  static void OnStreamParamChanged(void* data,
+                                   uint32_t id,
+                                   const struct spa_pod* format);
+  static void OnStreamStateChanged(void* data,
+                                   pw_stream_state old_state,
+                                   pw_stream_state state,
+                                   const char* error_message);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_TEST_TEST_SCREENCAST_STREAM_PROVIDER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/xdg_desktop_portal_utils.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/xdg_desktop_portal_utils.h
new file mode 100644
index 0000000000..b213e50308
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/xdg_desktop_portal_utils.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_XDG_DESKTOP_PORTAL_UTILS_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_XDG_DESKTOP_PORTAL_UTILS_H_
+
+// TODO(bugs.webrtc.org/14187): remove when all users are gone
+#include "modules/portal/xdg_desktop_portal_utils.h"
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_XDG_DESKTOP_PORTAL_UTILS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/xdg_session_details.h b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/xdg_session_details.h
new file mode 100644
index 0000000000..9feff5bdf7
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/xdg_session_details.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_XDG_SESSION_DETAILS_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_XDG_SESSION_DETAILS_H_
+
+// TODO(bugs.webrtc.org/14187): remove when all users are gone
+#include "modules/portal/xdg_session_details.h"
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WAYLAND_XDG_SESSION_DETAILS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc
new file mode 100644
index 0000000000..d4b85af6bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.cc
@@ -0,0 +1,258 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.h"
+
+#include <X11/Xlib.h>
+#include <X11/extensions/Xfixes.h>
+#include <X11/extensions/xfixeswire.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include <algorithm>
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/linux/x11/x_error_trap.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace {
+
+// WindowCapturer returns window IDs of X11 windows with WM_STATE attribute.
+// These windows may not be immediate children of the root window, because
+// window managers may re-parent them to add decorations. However,
+// XQueryPointer() expects to be passed children of the root. This function
+// searches up the list of the windows to find the root child that corresponds
+// to `window`.
+Window GetTopLevelWindow(Display* display, Window window) {
+  webrtc::XErrorTrap error_trap(display);
+  while (true) {
+    // If the window is in WithdrawnState then look at all of its children.
+    ::Window root, parent;
+    ::Window* children;
+    unsigned int num_children;
+    if (!XQueryTree(display, window, &root, &parent, &children,
+                    &num_children)) {
+      RTC_LOG(LS_ERROR) << "Failed to query for child windows although window"
+                           "does not have a valid WM_STATE.";
+      return None;
+    }
+    if (children)
+      XFree(children);
+
+    if (parent == root)
+      break;
+
+    window = parent;
+  }
+
+  return window;
+}
+
+}  // namespace
+
+namespace webrtc {
+
+MouseCursorMonitorX11::MouseCursorMonitorX11(
+    const DesktopCaptureOptions& options,
+    Window window)
+    : x_display_(options.x_display()),
+      callback_(NULL),
+      mode_(SHAPE_AND_POSITION),
+      window_(window),
+      have_xfixes_(false),
+      xfixes_event_base_(-1),
+      xfixes_error_base_(-1) {
+  // Set a default initial cursor shape in case XFixes is not present.
+  const int kSize = 5;
+  std::unique_ptr<DesktopFrame> default_cursor(
+      new BasicDesktopFrame(DesktopSize(kSize, kSize)));
+  const uint8_t pixels[kSize * kSize] = {
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0xff,
+      0x00, 0x00, 0xff, 0xff, 0xff, 0x00, 0x00, 0xff, 0xff,
+      0xff, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+  uint8_t* ptr = default_cursor->data();
+  for (int y = 0; y < kSize; ++y) {
+    for (int x = 0; x < kSize; ++x) {
+      *ptr++ = pixels[kSize * y + x];
+      *ptr++ = pixels[kSize * y + x];
+      *ptr++ = pixels[kSize * y + x];
+      *ptr++ = 0xff;
+    }
+  }
+  DesktopVector hotspot(2, 2);
+  cursor_shape_.reset(new MouseCursor(default_cursor.release(), hotspot));
+}
+
+MouseCursorMonitorX11::~MouseCursorMonitorX11() {
+  if (have_xfixes_) {
+    x_display_->RemoveEventHandler(xfixes_event_base_ + XFixesCursorNotify,
+                                   this);
+  }
+}
+
+void MouseCursorMonitorX11::Init(Callback* callback, Mode mode) {
+  // Init can be called only if not started
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+
+  callback_ = callback;
+  mode_ = mode;
+
+  have_xfixes_ =
+      XFixesQueryExtension(display(), &xfixes_event_base_, &xfixes_error_base_);
+
+  if (have_xfixes_) {
+    // Register for changes to the cursor shape.
+    XErrorTrap error_trap(display());
+    XFixesSelectCursorInput(display(), window_, XFixesDisplayCursorNotifyMask);
+    x_display_->AddEventHandler(xfixes_event_base_ + XFixesCursorNotify, this);
+
+    CaptureCursor();
+  } else {
+    RTC_LOG(LS_INFO) << "X server does not support XFixes.";
+  }
+}
+
+void MouseCursorMonitorX11::Capture() {
+  RTC_DCHECK(callback_);
+
+  // Process X11 events in case XFixes has sent cursor notification.
+  x_display_->ProcessPendingXEvents();
+
+  // cursor_shape_| is set only if we were notified of a cursor shape change.
+  if (cursor_shape_.get())
+    callback_->OnMouseCursor(cursor_shape_.release());
+
+  // Get cursor position if necessary.
+  if (mode_ == SHAPE_AND_POSITION) {
+    int root_x;
+    int root_y;
+    int win_x;
+    int win_y;
+    Window root_window;
+    Window child_window;
+    unsigned int mask;
+
+    CursorState state;
+    {
+      XErrorTrap error_trap(display());
+      Bool result =
+          XQueryPointer(display(), window_, &root_window, &child_window,
+                        &root_x, &root_y, &win_x, &win_y, &mask);
+      if (!result || error_trap.GetLastErrorAndDisable() != 0) {
+        state = OUTSIDE;
+      } else {
+        // In screen mode (window_ == root_window) the mouse is always inside.
+        // XQueryPointer() sets `child_window` to None if the cursor is outside
+        // `window_`.
+        state =
+            (window_ == root_window || child_window != None) ? INSIDE : OUTSIDE;
+      }
+    }
+
+    // As the comments to GetTopLevelWindow() above indicate, in window capture,
+    // the cursor position capture happens in `window_`, while the frame catpure
+    // happens in `child_window`. These two windows are not alwyas same, as
+    // window manager may add some decorations to the `window_`. So translate
+    // the coordinate in `window_` to the coordinate space of `child_window`.
+    if (window_ != root_window && state == INSIDE) {
+      int translated_x, translated_y;
+      Window unused;
+      if (XTranslateCoordinates(display(), window_, child_window, win_x, win_y,
+                                &translated_x, &translated_y, &unused)) {
+        win_x = translated_x;
+        win_y = translated_y;
+      }
+    }
+
+    // X11 always starts the coordinate from (0, 0), so we do not need to
+    // translate here.
+    callback_->OnMouseCursorPosition(DesktopVector(root_x, root_y));
+  }
+}
+
+bool MouseCursorMonitorX11::HandleXEvent(const XEvent& event) {
+  if (have_xfixes_ && event.type == xfixes_event_base_ + XFixesCursorNotify) {
+    const XFixesCursorNotifyEvent* cursor_event =
+        reinterpret_cast<const XFixesCursorNotifyEvent*>(&event);
+    if (cursor_event->subtype == XFixesDisplayCursorNotify) {
+      CaptureCursor();
+    }
+    // Return false, even if the event has been handled, because there might be
+    // other listeners for cursor notifications.
+  }
+  return false;
+}
+
+void MouseCursorMonitorX11::CaptureCursor() {
+  RTC_DCHECK(have_xfixes_);
+
+  XFixesCursorImage* img;
+  {
+    XErrorTrap error_trap(display());
+    img = XFixesGetCursorImage(display());
+    if (!img || error_trap.GetLastErrorAndDisable() != 0)
+      return;
+  }
+
+  std::unique_ptr<DesktopFrame> image(
+      new BasicDesktopFrame(DesktopSize(img->width, img->height)));
+
+  // Xlib stores 32-bit data in longs, even if longs are 64-bits long.
+  unsigned long* src = img->pixels;  // NOLINT(runtime/int)
+  uint32_t* dst = reinterpret_cast<uint32_t*>(image->data());
+  uint32_t* dst_end = dst + (img->width * img->height);
+  while (dst < dst_end) {
+    *dst++ = static_cast<uint32_t>(*src++);
+  }
+
+  DesktopVector hotspot(std::min(img->width, img->xhot),
+                        std::min(img->height, img->yhot));
+
+  XFree(img);
+
+  cursor_shape_.reset(new MouseCursor(image.release(), hotspot));
+}
+
+// static
+MouseCursorMonitor* MouseCursorMonitorX11::CreateForWindow(
+    const DesktopCaptureOptions& options,
+    WindowId window) {
+  if (!options.x_display())
+    return NULL;
+  window = GetTopLevelWindow(options.x_display()->display(), window);
+  if (window == None)
+    return NULL;
+  return new MouseCursorMonitorX11(options, window);
+}
+
+MouseCursorMonitor* MouseCursorMonitorX11::CreateForScreen(
+    const DesktopCaptureOptions& options,
+    ScreenId screen) {
+  if (!options.x_display())
+    return NULL;
+  return new MouseCursorMonitorX11(
+      options, DefaultRootWindow(options.x_display()->display()));
+}
+
+std::unique_ptr<MouseCursorMonitor> MouseCursorMonitorX11::Create(
+    const DesktopCaptureOptions& options) {
+  return std::unique_ptr<MouseCursorMonitor>(
+      CreateForScreen(options, kFullDesktopScreenId));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.h
new file mode 100644
index 0000000000..980d254a0a
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_MOUSE_CURSOR_MONITOR_X11_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_MOUSE_CURSOR_MONITOR_X11_H_
+
+#include <X11/X.h>
+
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/linux/x11/shared_x_display.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+
+namespace webrtc {
+
+class MouseCursorMonitorX11 : public MouseCursorMonitor,
+                              public SharedXDisplay::XEventHandler {
+ public:
+  MouseCursorMonitorX11(const DesktopCaptureOptions& options, Window window);
+  ~MouseCursorMonitorX11() override;
+
+  static MouseCursorMonitor* CreateForWindow(
+      const DesktopCaptureOptions& options,
+      WindowId window);
+  static MouseCursorMonitor* CreateForScreen(
+      const DesktopCaptureOptions& options,
+      ScreenId screen);
+  static std::unique_ptr<MouseCursorMonitor> Create(
+      const DesktopCaptureOptions& options);
+
+  void Init(Callback* callback, Mode mode) override;
+  void Capture() override;
+
+ private:
+  // SharedXDisplay::XEventHandler interface.
+  bool HandleXEvent(const XEvent& event) override;
+
+  Display* display() { return x_display_->display(); }
+
+  // Captures current cursor shape and stores it in `cursor_shape_`.
+  void CaptureCursor();
+
+  rtc::scoped_refptr<SharedXDisplay> x_display_;
+  Callback* callback_;
+  Mode mode_;
+  Window window_;
+
+  bool have_xfixes_;
+  int xfixes_event_base_;
+  int xfixes_error_base_;
+
+  std::unique_ptr<MouseCursor> cursor_shape_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_MOUSE_CURSOR_MONITOR_X11_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc
new file mode 100644
index 0000000000..fa6334e8ba
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.cc
@@ -0,0 +1,517 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/screen_capturer_x11.h"
+
+#include <X11/Xlib.h>
+#include <X11/extensions/Xdamage.h>
+#include <X11/extensions/Xfixes.h>
+#include <X11/extensions/damagewire.h>
+#include <dlfcn.h>
+#include <stdint.h>
+#include <string.h>
+
+#include <memory>
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/linux/x11/x_server_pixel_buffer.h"
+#include "modules/desktop_capture/screen_capture_frame_queue.h"
+#include "modules/desktop_capture/screen_capturer_helper.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/sanitizer.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+ScreenCapturerX11::ScreenCapturerX11() {
+  helper_.SetLogGridSize(4);
+}
+
+ScreenCapturerX11::~ScreenCapturerX11() {
+  options_.x_display()->RemoveEventHandler(ConfigureNotify, this);
+  if (use_damage_) {
+    options_.x_display()->RemoveEventHandler(damage_event_base_ + XDamageNotify,
+                                             this);
+  }
+  if (use_randr_) {
+    options_.x_display()->RemoveEventHandler(
+        randr_event_base_ + RRScreenChangeNotify, this);
+  }
+  DeinitXlib();
+}
+
+bool ScreenCapturerX11::Init(const DesktopCaptureOptions& options) {
+  TRACE_EVENT0("webrtc", "ScreenCapturerX11::Init");
+  options_ = options;
+
+  atom_cache_ = std::make_unique<XAtomCache>(display());
+
+  root_window_ = RootWindow(display(), DefaultScreen(display()));
+  if (root_window_ == BadValue) {
+    RTC_LOG(LS_ERROR) << "Unable to get the root window";
+    DeinitXlib();
+    return false;
+  }
+
+  gc_ = XCreateGC(display(), root_window_, 0, NULL);
+  if (gc_ == NULL) {
+    RTC_LOG(LS_ERROR) << "Unable to get graphics context";
+    DeinitXlib();
+    return false;
+  }
+
+  options_.x_display()->AddEventHandler(ConfigureNotify, this);
+
+  // Check for XFixes extension. This is required for cursor shape
+  // notifications, and for our use of XDamage.
+  if (XFixesQueryExtension(display(), &xfixes_event_base_,
+                           &xfixes_error_base_)) {
+    has_xfixes_ = true;
+  } else {
+    RTC_LOG(LS_INFO) << "X server does not support XFixes.";
+  }
+
+  // Register for changes to the dimensions of the root window.
+  XSelectInput(display(), root_window_, StructureNotifyMask);
+
+  if (!x_server_pixel_buffer_.Init(atom_cache_.get(),
+                                   DefaultRootWindow(display()))) {
+    RTC_LOG(LS_ERROR) << "Failed to initialize pixel buffer.";
+    return false;
+  }
+
+  if (options_.use_update_notifications()) {
+    InitXDamage();
+  }
+
+  InitXrandr();
+
+  // Default source set here so that selected_monitor_rect_ is sized correctly.
+  SelectSource(kFullDesktopScreenId);
+
+  return true;
+}
+
+void ScreenCapturerX11::InitXDamage() {
+  // Our use of XDamage requires XFixes.
+  if (!has_xfixes_) {
+    return;
+  }
+
+  // Check for XDamage extension.
+  if (!XDamageQueryExtension(display(), &damage_event_base_,
+                             &damage_error_base_)) {
+    RTC_LOG(LS_INFO) << "X server does not support XDamage.";
+    return;
+  }
+
+  // TODO(lambroslambrou): Disable DAMAGE in situations where it is known
+  // to fail, such as when Desktop Effects are enabled, with graphics
+  // drivers (nVidia, ATI) that fail to report DAMAGE notifications
+  // properly.
+
+  // Request notifications every time the screen becomes damaged.
+  damage_handle_ =
+      XDamageCreate(display(), root_window_, XDamageReportNonEmpty);
+  if (!damage_handle_) {
+    RTC_LOG(LS_ERROR) << "Unable to initialize XDamage.";
+    return;
+  }
+
+  // Create an XFixes server-side region to collate damage into.
+  damage_region_ = XFixesCreateRegion(display(), 0, 0);
+  if (!damage_region_) {
+    XDamageDestroy(display(), damage_handle_);
+    RTC_LOG(LS_ERROR) << "Unable to create XFixes region.";
+    return;
+  }
+
+  options_.x_display()->AddEventHandler(damage_event_base_ + XDamageNotify,
+                                        this);
+
+  use_damage_ = true;
+  RTC_LOG(LS_INFO) << "Using XDamage extension.";
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+void ScreenCapturerX11::InitXrandr() {
+  int major_version = 0;
+  int minor_version = 0;
+  int error_base_ignored = 0;
+  if (XRRQueryExtension(display(), &randr_event_base_, &error_base_ignored) &&
+      XRRQueryVersion(display(), &major_version, &minor_version)) {
+    if (major_version > 1 || (major_version == 1 && minor_version >= 5)) {
+      // Dynamically link XRRGetMonitors and XRRFreeMonitors as a workaround
+      // to avoid a dependency issue with Debian 8.
+      get_monitors_ = reinterpret_cast<get_monitors_func>(
+          dlsym(RTLD_DEFAULT, "XRRGetMonitors"));
+      free_monitors_ = reinterpret_cast<free_monitors_func>(
+          dlsym(RTLD_DEFAULT, "XRRFreeMonitors"));
+      if (get_monitors_ && free_monitors_) {
+        use_randr_ = true;
+        RTC_LOG(LS_INFO) << "Using XRandR extension v" << major_version << '.'
+                         << minor_version << '.';
+        monitors_ =
+            get_monitors_(display(), root_window_, true, &num_monitors_);
+
+        // Register for screen change notifications
+        XRRSelectInput(display(), root_window_, RRScreenChangeNotifyMask);
+        options_.x_display()->AddEventHandler(
+            randr_event_base_ + RRScreenChangeNotify, this);
+      } else {
+        RTC_LOG(LS_ERROR) << "Unable to link XRandR monitor functions.";
+      }
+    } else {
+      RTC_LOG(LS_ERROR) << "XRandR entension is older than v1.5.";
+    }
+  } else {
+    RTC_LOG(LS_ERROR) << "X server does not support XRandR.";
+  }
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+void ScreenCapturerX11::UpdateMonitors() {
+  // The queue should be reset whenever |selected_monitor_rect_| changes, so
+  // that the DCHECKs in CaptureScreen() are satisfied.
+  queue_.Reset();
+
+  if (monitors_) {
+    free_monitors_(monitors_);
+    monitors_ = nullptr;
+  }
+
+  monitors_ = get_monitors_(display(), root_window_, true, &num_monitors_);
+
+  if (selected_monitor_name_) {
+    if (selected_monitor_name_ == static_cast<Atom>(kFullDesktopScreenId)) {
+      selected_monitor_rect_ =
+          DesktopRect::MakeSize(x_server_pixel_buffer_.window_size());
+      return;
+    }
+
+    for (int i = 0; i < num_monitors_; ++i) {
+      XRRMonitorInfo& m = monitors_[i];
+      if (selected_monitor_name_ == m.name) {
+        RTC_LOG(LS_INFO) << "XRandR monitor " << m.name << " rect updated.";
+        selected_monitor_rect_ =
+            DesktopRect::MakeXYWH(m.x, m.y, m.width, m.height);
+        const auto& pixel_buffer_rect = x_server_pixel_buffer_.window_rect();
+        if (!pixel_buffer_rect.ContainsRect(selected_monitor_rect_)) {
+          // This is never expected to happen, but crop the rectangle anyway
+          // just in case the server returns inconsistent information.
+          // CaptureScreen() expects `selected_monitor_rect_` to lie within
+          // the pixel-buffer's rectangle.
+          RTC_LOG(LS_WARNING)
+              << "Cropping selected monitor rect to fit the pixel-buffer.";
+          selected_monitor_rect_.IntersectWith(pixel_buffer_rect);
+        }
+        return;
+      }
+    }
+
+    // The selected monitor is not connected anymore
+    RTC_LOG(LS_INFO) << "XRandR selected monitor " << selected_monitor_name_
+                     << " lost.";
+    selected_monitor_rect_ = DesktopRect::MakeWH(0, 0);
+  }
+}
+
+void ScreenCapturerX11::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+
+  callback_ = callback;
+}
+
+void ScreenCapturerX11::CaptureFrame() {
+  TRACE_EVENT0("webrtc", "ScreenCapturerX11::CaptureFrame");
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+  queue_.MoveToNextFrame();
+  if (queue_.current_frame() && queue_.current_frame()->IsShared()) {
+    RTC_DLOG(LS_WARNING) << "Overwriting frame that is still shared.";
+  }
+
+  // Process XEvents for XDamage and cursor shape tracking.
+  options_.x_display()->ProcessPendingXEvents();
+
+  // ProcessPendingXEvents() may call ScreenConfigurationChanged() which
+  // reinitializes `x_server_pixel_buffer_`. Check if the pixel buffer is still
+  // in a good shape.
+  if (!x_server_pixel_buffer_.is_initialized()) {
+    // We failed to initialize pixel buffer.
+    RTC_LOG(LS_ERROR) << "Pixel buffer is not initialized.";
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+
+  // Allocate the current frame buffer only if it is not already allocated.
+  // Note that we can't reallocate other buffers at this point, since the caller
+  // may still be reading from them.
+  if (!queue_.current_frame()) {
+    std::unique_ptr<DesktopFrame> frame(
+        new BasicDesktopFrame(selected_monitor_rect_.size()));
+
+    // We set the top-left of the frame so the mouse cursor will be composited
+    // properly, and our frame buffer will not be overrun while blitting.
+    frame->set_top_left(selected_monitor_rect_.top_left());
+    queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(std::move(frame)));
+  }
+
+  std::unique_ptr<DesktopFrame> result = CaptureScreen();
+  if (!result) {
+    RTC_LOG(LS_WARNING) << "Temporarily failed to capture screen.";
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+
+  last_invalid_region_ = result->updated_region();
+  result->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
+                              rtc::kNumNanosecsPerMillisec);
+  result->set_capturer_id(DesktopCapturerId::kX11CapturerLinux);
+  callback_->OnCaptureResult(Result::SUCCESS, std::move(result));
+}
+
+bool ScreenCapturerX11::GetSourceList(SourceList* sources) {
+  RTC_DCHECK(sources->size() == 0);
+  if (!use_randr_) {
+    sources->push_back({});
+    return true;
+  }
+
+  // Ensure that `monitors_` is updated with changes that may have happened
+  // between calls to GetSourceList().
+  options_.x_display()->ProcessPendingXEvents();
+
+  for (int i = 0; i < num_monitors_; ++i) {
+    XRRMonitorInfo& m = monitors_[i];
+    char* monitor_title = XGetAtomName(display(), m.name);
+
+    // Note name is an X11 Atom used to id the monitor.
+    sources->push_back({static_cast<SourceId>(m.name), 0, monitor_title});
+    XFree(monitor_title);
+  }
+
+  return true;
+}
+
+bool ScreenCapturerX11::SelectSource(SourceId id) {
+  // Prevent the reuse of any frame buffers allocated for a previously selected
+  // source. This is required to stop crashes, or old data from appearing in
+  // a captured frame, when the new source is sized differently then the source
+  // that was selected at the time a reused frame buffer was created.
+  queue_.Reset();
+
+  if (!use_randr_ || id == kFullDesktopScreenId) {
+    selected_monitor_name_ = kFullDesktopScreenId;
+    selected_monitor_rect_ =
+        DesktopRect::MakeSize(x_server_pixel_buffer_.window_size());
+    return true;
+  }
+
+  for (int i = 0; i < num_monitors_; ++i) {
+    if (id == static_cast<SourceId>(monitors_[i].name)) {
+      RTC_LOG(LS_INFO) << "XRandR selected source: " << id;
+      XRRMonitorInfo& m = monitors_[i];
+      selected_monitor_name_ = m.name;
+      selected_monitor_rect_ =
+          DesktopRect::MakeXYWH(m.x, m.y, m.width, m.height);
+      const auto& pixel_buffer_rect = x_server_pixel_buffer_.window_rect();
+      if (!pixel_buffer_rect.ContainsRect(selected_monitor_rect_)) {
+        RTC_LOG(LS_WARNING)
+            << "Cropping selected monitor rect to fit the pixel-buffer.";
+        selected_monitor_rect_.IntersectWith(pixel_buffer_rect);
+      }
+      return true;
+    }
+  }
+  return false;
+}
+
+bool ScreenCapturerX11::HandleXEvent(const XEvent& event) {
+  if (use_damage_ && (event.type == damage_event_base_ + XDamageNotify)) {
+    const XDamageNotifyEvent* damage_event =
+        reinterpret_cast<const XDamageNotifyEvent*>(&event);
+    if (damage_event->damage != damage_handle_)
+      return false;
+    RTC_DCHECK(damage_event->level == XDamageReportNonEmpty);
+    return true;
+  } else if (use_randr_ &&
+             event.type == randr_event_base_ + RRScreenChangeNotify) {
+    XRRUpdateConfiguration(const_cast<XEvent*>(&event));
+    UpdateMonitors();
+    RTC_LOG(LS_INFO) << "XRandR screen change event received.";
+    return false;
+  } else if (event.type == ConfigureNotify) {
+    ScreenConfigurationChanged();
+    return false;
+  }
+  return false;
+}
+
+std::unique_ptr<DesktopFrame> ScreenCapturerX11::CaptureScreen() {
+  std::unique_ptr<SharedDesktopFrame> frame = queue_.current_frame()->Share();
+  RTC_DCHECK(selected_monitor_rect_.size().equals(frame->size()));
+  RTC_DCHECK(selected_monitor_rect_.top_left().equals(frame->top_left()));
+
+  // Pass the screen size to the helper, so it can clip the invalid region if it
+  // expands that region to a grid. Note that the helper operates in the
+  // DesktopFrame coordinate system where the top-left pixel is (0, 0), even for
+  // a monitor with non-zero offset relative to `x_server_pixel_buffer_`.
+  helper_.set_size_most_recent(frame->size());
+
+  // In the DAMAGE case, ensure the frame is up-to-date with the previous frame
+  // if any.  If there isn't a previous frame, that means a screen-resolution
+  // change occurred, and `invalid_rects` will be updated to include the whole
+  // screen.
+  if (use_damage_ && queue_.previous_frame())
+    SynchronizeFrame();
+
+  DesktopRegion* updated_region = frame->mutable_updated_region();
+
+  x_server_pixel_buffer_.Synchronize();
+  if (use_damage_ && queue_.previous_frame()) {
+    // Atomically fetch and clear the damage region.
+    XDamageSubtract(display(), damage_handle_, None, damage_region_);
+    int rects_num = 0;
+    XRectangle bounds;
+    XRectangle* rects = XFixesFetchRegionAndBounds(display(), damage_region_,
+                                                   &rects_num, &bounds);
+    for (int i = 0; i < rects_num; ++i) {
+      auto damage_rect = DesktopRect::MakeXYWH(rects[i].x, rects[i].y,
+                                               rects[i].width, rects[i].height);
+
+      // Damage-regions are relative to `x_server_pixel_buffer`, so convert the
+      // region to DesktopFrame coordinates where the top-left is always (0, 0),
+      // before adding to the frame's updated_region. `helper_` also operates in
+      // DesktopFrame coordinates, and it will take care of cropping away any
+      // damage-regions that lie outside the selected monitor.
+      damage_rect.Translate(-frame->top_left());
+      updated_region->AddRect(damage_rect);
+    }
+    XFree(rects);
+    helper_.InvalidateRegion(*updated_region);
+
+    // Capture the damaged portions of the desktop.
+    helper_.TakeInvalidRegion(updated_region);
+
+    for (DesktopRegion::Iterator it(*updated_region); !it.IsAtEnd();
+         it.Advance()) {
+      auto rect = it.rect();
+      rect.Translate(frame->top_left());
+      if (!x_server_pixel_buffer_.CaptureRect(rect, frame.get()))
+        return nullptr;
+    }
+  } else {
+    // Doing full-screen polling, or this is the first capture after a
+    // screen-resolution change.  In either case, need a full-screen capture.
+    if (!x_server_pixel_buffer_.CaptureRect(selected_monitor_rect_,
+                                            frame.get())) {
+      return nullptr;
+    }
+    updated_region->SetRect(DesktopRect::MakeSize(frame->size()));
+  }
+
+  return std::move(frame);
+}
+
+void ScreenCapturerX11::ScreenConfigurationChanged() {
+  TRACE_EVENT0("webrtc", "ScreenCapturerX11::ScreenConfigurationChanged");
+  // Make sure the frame buffers will be reallocated.
+  queue_.Reset();
+
+  helper_.ClearInvalidRegion();
+  if (!x_server_pixel_buffer_.Init(atom_cache_.get(),
+                                   DefaultRootWindow(display()))) {
+    RTC_LOG(LS_ERROR) << "Failed to initialize pixel buffer after screen "
+                         "configuration change.";
+  }
+
+  if (use_randr_) {
+    // Adding/removing RANDR monitors can generate a ConfigureNotify event
+    // without generating any RRScreenChangeNotify event. So it is important to
+    // update the monitors here even if the screen resolution hasn't changed.
+    UpdateMonitors();
+  } else {
+    selected_monitor_rect_ =
+        DesktopRect::MakeSize(x_server_pixel_buffer_.window_size());
+  }
+}
+
+void ScreenCapturerX11::SynchronizeFrame() {
+  // Synchronize the current buffer with the previous one since we do not
+  // capture the entire desktop. Note that encoder may be reading from the
+  // previous buffer at this time so thread access complaints are false
+  // positives.
+
+  // TODO(hclam): We can reduce the amount of copying here by subtracting
+  // `capturer_helper_`s region from `last_invalid_region_`.
+  // http://crbug.com/92354
+  RTC_DCHECK(queue_.previous_frame());
+
+  DesktopFrame* current = queue_.current_frame();
+  DesktopFrame* last = queue_.previous_frame();
+  RTC_DCHECK(current != last);
+  for (DesktopRegion::Iterator it(last_invalid_region_); !it.IsAtEnd();
+       it.Advance()) {
+    const DesktopRect& r = it.rect();
+    current->CopyPixelsFrom(*last, r.top_left(), r);
+  }
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+void ScreenCapturerX11::DeinitXlib() {
+  if (monitors_) {
+    free_monitors_(monitors_);
+    monitors_ = nullptr;
+  }
+
+  if (gc_) {
+    XFreeGC(display(), gc_);
+    gc_ = nullptr;
+  }
+
+  x_server_pixel_buffer_.Release();
+
+  if (display()) {
+    if (damage_handle_) {
+      XDamageDestroy(display(), damage_handle_);
+      damage_handle_ = 0;
+    }
+
+    if (damage_region_) {
+      XFixesDestroyRegion(display(), damage_region_);
+      damage_region_ = 0;
+    }
+  }
+}
+
+// static
+std::unique_ptr<DesktopCapturer> ScreenCapturerX11::CreateRawScreenCapturer(
+    const DesktopCaptureOptions& options) {
+  if (!options.x_display())
+    return nullptr;
+
+  std::unique_ptr<ScreenCapturerX11> capturer(new ScreenCapturerX11());
+  if (!capturer.get()->Init(options)) {
+    return nullptr;
+  }
+
+  return std::move(capturer);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.h
new file mode 100644
index 0000000000..d2a437aaa2
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/screen_capturer_x11.h
@@ -0,0 +1,147 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_SCREEN_CAPTURER_X11_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_SCREEN_CAPTURER_X11_H_
+
+#include <X11/X.h>
+#include <X11/Xlib.h>
+#include <X11/extensions/Xdamage.h>
+#include <X11/extensions/Xfixes.h>
+#include <X11/extensions/Xrandr.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/linux/x11/shared_x_display.h"
+#include "modules/desktop_capture/linux/x11/x_atom_cache.h"
+#include "modules/desktop_capture/linux/x11/x_server_pixel_buffer.h"
+#include "modules/desktop_capture/screen_capture_frame_queue.h"
+#include "modules/desktop_capture/screen_capturer_helper.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+
+namespace webrtc {
+
+// A class to perform video frame capturing for Linux on X11.
+//
+// If XDamage is used, this class sets DesktopFrame::updated_region() according
+// to the areas reported by XDamage. Otherwise this class does not detect
+// DesktopFrame::updated_region(), the field is always set to the entire frame
+// rectangle. ScreenCapturerDifferWrapper should be used if that functionality
+// is necessary.
+class ScreenCapturerX11 : public DesktopCapturer,
+                          public SharedXDisplay::XEventHandler {
+ public:
+  ScreenCapturerX11();
+  ~ScreenCapturerX11() override;
+
+  ScreenCapturerX11(const ScreenCapturerX11&) = delete;
+  ScreenCapturerX11& operator=(const ScreenCapturerX11&) = delete;
+
+  static std::unique_ptr<DesktopCapturer> CreateRawScreenCapturer(
+      const DesktopCaptureOptions& options);
+
+  // TODO(ajwong): Do we really want this to be synchronous?
+  bool Init(const DesktopCaptureOptions& options);
+
+  // DesktopCapturer interface.
+  void Start(Callback* delegate) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+
+ private:
+  Display* display() { return options_.x_display()->display(); }
+
+  // SharedXDisplay::XEventHandler interface.
+  bool HandleXEvent(const XEvent& event) override;
+
+  void InitXDamage();
+  void InitXrandr();
+  void UpdateMonitors();
+
+  // Capture screen pixels to the current buffer in the queue. In the DAMAGE
+  // case, the ScreenCapturerHelper already holds the list of invalid rectangles
+  // from HandleXEvent(). In the non-DAMAGE case, this captures the
+  // whole screen, then calculates some invalid rectangles that include any
+  // differences between this and the previous capture.
+  std::unique_ptr<DesktopFrame> CaptureScreen();
+
+  // Called when the screen configuration is changed.
+  void ScreenConfigurationChanged();
+
+  // Synchronize the current buffer with `last_buffer_`, by copying pixels from
+  // the area of `last_invalid_rects`.
+  // Note this only works on the assumption that kNumBuffers == 2, as
+  // `last_invalid_rects` holds the differences from the previous buffer and
+  // the one prior to that (which will then be the current buffer).
+  void SynchronizeFrame();
+
+  void DeinitXlib();
+
+  DesktopCaptureOptions options_;
+
+  Callback* callback_ = nullptr;
+
+  // X11 graphics context.
+  GC gc_ = nullptr;
+  Window root_window_ = BadValue;
+
+  // XRandR 1.5 monitors.
+  bool use_randr_ = false;
+  int randr_event_base_ = 0;
+  XRRMonitorInfo* monitors_ = nullptr;
+  int num_monitors_ = 0;
+  DesktopRect selected_monitor_rect_;
+  // selected_monitor_name_ will be changed to kFullDesktopScreenId
+  // by a call to SelectSource() at the end of Init() because
+  // selected_monitor_rect_ should be updated as well.
+  // Setting it to kFullDesktopScreenId here might be misleading.
+  Atom selected_monitor_name_ = 0;
+  typedef XRRMonitorInfo* (*get_monitors_func)(Display*, Window, Bool, int*);
+  typedef void (*free_monitors_func)(XRRMonitorInfo*);
+  get_monitors_func get_monitors_ = nullptr;
+  free_monitors_func free_monitors_ = nullptr;
+
+  // XFixes.
+  bool has_xfixes_ = false;
+  int xfixes_event_base_ = -1;
+  int xfixes_error_base_ = -1;
+
+  // XDamage information.
+  bool use_damage_ = false;
+  Damage damage_handle_ = 0;
+  int damage_event_base_ = -1;
+  int damage_error_base_ = -1;
+  XserverRegion damage_region_ = 0;
+
+  // Access to the X Server's pixel buffer.
+  XServerPixelBuffer x_server_pixel_buffer_;
+
+  // A thread-safe list of invalid rectangles, and the size of the most
+  // recently captured screen.
+  ScreenCapturerHelper helper_;
+
+  // Queue of the frames buffers.
+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
+
+  // Invalid region from the previous capture. This is used to synchronize the
+  // current with the last buffer used.
+  DesktopRegion last_invalid_region_;
+
+  std::unique_ptr<XAtomCache> atom_cache_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_SCREEN_CAPTURER_X11_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc
new file mode 100644
index 0000000000..d690b0e2ba
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.cc
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/shared_x_display.h"
+
+#include <X11/Xlib.h>
+#include <X11/extensions/XTest.h>
+
+#include <algorithm>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+SharedXDisplay::SharedXDisplay(Display* display) : display_(display) {
+  RTC_DCHECK(display_);
+}
+
+SharedXDisplay::~SharedXDisplay() {
+  RTC_DCHECK(event_handlers_.empty());
+  XCloseDisplay(display_);
+}
+
+// static
+rtc::scoped_refptr<SharedXDisplay> SharedXDisplay::Create(
+    absl::string_view display_name) {
+  Display* display = XOpenDisplay(
+      display_name.empty() ? NULL : std::string(display_name).c_str());
+  if (!display) {
+    RTC_LOG(LS_ERROR) << "Unable to open display";
+    return nullptr;
+  }
+  return rtc::scoped_refptr<SharedXDisplay>(new SharedXDisplay(display));
+}
+
+// static
+rtc::scoped_refptr<SharedXDisplay> SharedXDisplay::CreateDefault() {
+  return Create(std::string());
+}
+
+void SharedXDisplay::AddEventHandler(int type, XEventHandler* handler) {
+  MutexLock lock(&mutex_);
+  event_handlers_[type].push_back(handler);
+}
+
+void SharedXDisplay::RemoveEventHandler(int type, XEventHandler* handler) {
+  MutexLock lock(&mutex_);
+  EventHandlersMap::iterator handlers = event_handlers_.find(type);
+  if (handlers == event_handlers_.end())
+    return;
+
+  std::vector<XEventHandler*>::iterator new_end =
+      std::remove(handlers->second.begin(), handlers->second.end(), handler);
+  handlers->second.erase(new_end, handlers->second.end());
+
+  // Check if no handlers left for this event.
+  if (handlers->second.empty())
+    event_handlers_.erase(handlers);
+}
+
+void SharedXDisplay::ProcessPendingXEvents() {
+  // Hold reference to `this` to prevent it from being destroyed while
+  // processing events.
+  rtc::scoped_refptr<SharedXDisplay> self(this);
+
+  // Protect access to `event_handlers_` after incrementing the refcount for
+  // `this` to ensure the instance is still valid when the lock is acquired.
+  MutexLock lock(&mutex_);
+
+  // Find the number of events that are outstanding "now."  We don't just loop
+  // on XPending because we want to guarantee this terminates.
+  int events_to_process = XPending(display());
+  XEvent e;
+
+  for (int i = 0; i < events_to_process; i++) {
+    XNextEvent(display(), &e);
+    EventHandlersMap::iterator handlers = event_handlers_.find(e.type);
+    if (handlers == event_handlers_.end())
+      continue;
+    for (std::vector<XEventHandler*>::iterator it = handlers->second.begin();
+         it != handlers->second.end(); ++it) {
+      if ((*it)->HandleXEvent(e))
+        break;
+    }
+  }
+}
+
+void SharedXDisplay::IgnoreXServerGrabs() {
+  int test_event_base = 0;
+  int test_error_base = 0;
+  int major = 0;
+  int minor = 0;
+  if (XTestQueryExtension(display(), &test_event_base, &test_error_base, &major,
+                          &minor)) {
+    XTestGrabControl(display(), true);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.h
new file mode 100644
index 0000000000..c05fc46546
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/shared_x_display.h
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_SHARED_X_DISPLAY_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_SHARED_X_DISPLAY_H_
+
+#include <map>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/rtc_export.h"
+#include "rtc_base/thread_annotations.h"
+
+// Including Xlib.h will involve evil defines (Bool, Status, True, False), which
+// easily conflict with other headers.
+typedef struct _XDisplay Display;
+typedef union _XEvent XEvent;
+
+namespace webrtc {
+
+// A ref-counted object to store XDisplay connection.
+class RTC_EXPORT SharedXDisplay
+    : public rtc::RefCountedNonVirtual<SharedXDisplay> {
+ public:
+  class XEventHandler {
+   public:
+    virtual ~XEventHandler() {}
+
+    // Processes XEvent. Returns true if the event has been handled.
+    virtual bool HandleXEvent(const XEvent& event) = 0;
+  };
+
+  // Creates a new X11 Display for the `display_name`. NULL is returned if X11
+  // connection failed. Equivalent to CreateDefault() when `display_name` is
+  // empty.
+  static rtc::scoped_refptr<SharedXDisplay> Create(
+      absl::string_view display_name);
+
+  // Creates X11 Display connection for the default display (e.g. specified in
+  // DISPLAY). NULL is returned if X11 connection failed.
+  static rtc::scoped_refptr<SharedXDisplay> CreateDefault();
+
+  Display* display() { return display_; }
+
+  // Adds a new event `handler` for XEvent's of `type`.
+  void AddEventHandler(int type, XEventHandler* handler);
+
+  // Removes event `handler` added using `AddEventHandler`. Doesn't do anything
+  // if `handler` is not registered.
+  void RemoveEventHandler(int type, XEventHandler* handler);
+
+  // Processes pending XEvents, calling corresponding event handlers.
+  void ProcessPendingXEvents();
+
+  void IgnoreXServerGrabs();
+
+  ~SharedXDisplay();
+
+  SharedXDisplay(const SharedXDisplay&) = delete;
+  SharedXDisplay& operator=(const SharedXDisplay&) = delete;
+
+ protected:
+  // Takes ownership of `display`.
+  explicit SharedXDisplay(Display* display);
+
+ private:
+  typedef std::map<int, std::vector<XEventHandler*> > EventHandlersMap;
+
+  Display* display_;
+
+  Mutex mutex_;
+
+  EventHandlersMap event_handlers_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_SHARED_X_DISPLAY_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc
new file mode 100644
index 0000000000..2e17d44c0a
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.cc
@@ -0,0 +1,256 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/window_capturer_x11.h"
+
+#include <X11/Xutil.h>
+#include <X11/extensions/Xcomposite.h>
+#include <X11/extensions/composite.h>
+#include <string.h>
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/linux/x11/shared_x_display.h"
+#include "modules/desktop_capture/linux/x11/window_finder_x11.h"
+#include "modules/desktop_capture/linux/x11/window_list_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+WindowCapturerX11::WindowCapturerX11(const DesktopCaptureOptions& options)
+    : x_display_(options.x_display()),
+      atom_cache_(display()),
+      window_finder_(&atom_cache_) {
+  int event_base, error_base, major_version, minor_version;
+  if (XCompositeQueryExtension(display(), &event_base, &error_base) &&
+      XCompositeQueryVersion(display(), &major_version, &minor_version) &&
+      // XCompositeNameWindowPixmap() requires version 0.2
+      (major_version > 0 || minor_version >= 2)) {
+    has_composite_extension_ = true;
+  } else {
+    RTC_LOG(LS_INFO) << "Xcomposite extension not available or too old.";
+  }
+
+  x_display_->AddEventHandler(ConfigureNotify, this);
+}
+
+WindowCapturerX11::~WindowCapturerX11() {
+  x_display_->RemoveEventHandler(ConfigureNotify, this);
+}
+
+bool WindowCapturerX11::GetSourceList(SourceList* sources) {
+  return GetWindowList(&atom_cache_, [this, sources](::Window window) {
+    Source w;
+    w.id = window;
+    w.pid = (pid_t)GetWindowProcessID(window);
+    if (this->GetWindowTitle(window, &w.title)) {
+      sources->push_back(w);
+    }
+    return true;
+  });
+}
+
+bool WindowCapturerX11::SelectSource(SourceId id) {
+  if (!x_server_pixel_buffer_.Init(&atom_cache_, id))
+    return false;
+
+  // Tell the X server to send us window resizing events.
+  XSelectInput(display(), id, StructureNotifyMask);
+
+  selected_window_ = id;
+
+  // In addition to needing X11 server-side support for Xcomposite, it actually
+  // needs to be turned on for the window. If the user has modern
+  // hardware/drivers but isn't using a compositing window manager, that won't
+  // be the case. Here we automatically turn it on.
+
+  // Redirect drawing to an offscreen buffer (ie, turn on compositing). X11
+  // remembers who has requested this and will turn it off for us when we exit.
+  XCompositeRedirectWindow(display(), id, CompositeRedirectAutomatic);
+
+  return true;
+}
+
+bool WindowCapturerX11::FocusOnSelectedSource() {
+  if (!selected_window_)
+    return false;
+
+  unsigned int num_children;
+  ::Window* children;
+  ::Window parent;
+  ::Window root;
+  // Find the root window to pass event to.
+  int status = XQueryTree(display(), selected_window_, &root, &parent,
+                          &children, &num_children);
+  if (status == 0) {
+    RTC_LOG(LS_ERROR) << "Failed to query for the root window.";
+    return false;
+  }
+
+  if (children)
+    XFree(children);
+
+  XRaiseWindow(display(), selected_window_);
+
+  // Some window managers (e.g., metacity in GNOME) consider it illegal to
+  // raise a window without also giving it input focus with
+  // _NET_ACTIVE_WINDOW, so XRaiseWindow() on its own isn't enough.
+  Atom atom = XInternAtom(display(), "_NET_ACTIVE_WINDOW", True);
+  if (atom != None) {
+    XEvent xev;
+    xev.xclient.type = ClientMessage;
+    xev.xclient.serial = 0;
+    xev.xclient.send_event = True;
+    xev.xclient.window = selected_window_;
+    xev.xclient.message_type = atom;
+
+    // The format member is set to 8, 16, or 32 and specifies whether the
+    // data should be viewed as a list of bytes, shorts, or longs.
+    xev.xclient.format = 32;
+
+    memset(xev.xclient.data.l, 0, sizeof(xev.xclient.data.l));
+
+    XSendEvent(display(), root, False,
+               SubstructureRedirectMask | SubstructureNotifyMask, &xev);
+  }
+  XFlush(display());
+  return true;
+}
+
+void WindowCapturerX11::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+
+  callback_ = callback;
+}
+
+void WindowCapturerX11::CaptureFrame() {
+  TRACE_EVENT0("webrtc", "WindowCapturerX11::CaptureFrame");
+  x_display_->ProcessPendingXEvents();
+
+  if (!x_server_pixel_buffer_.IsWindowValid()) {
+    RTC_LOG(LS_ERROR) << "The window is no longer valid.";
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+
+  if (!has_composite_extension_) {
+    // Without the Xcomposite extension we capture when the whole window is
+    // visible on screen and not covered by any other window. This is not
+    // something we want so instead, just bail out.
+    RTC_LOG(LS_ERROR) << "No Xcomposite extension detected.";
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+
+  if (GetWindowState(&atom_cache_, selected_window_) == IconicState) {
+    // Window is in minimized. Return a 1x1 frame as same as OSX/Win does.
+    std::unique_ptr<DesktopFrame> frame(
+        new BasicDesktopFrame(DesktopSize(1, 1)));
+    callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
+    return;
+  }
+
+  std::unique_ptr<DesktopFrame> frame(
+      new BasicDesktopFrame(x_server_pixel_buffer_.window_size()));
+
+  x_server_pixel_buffer_.Synchronize();
+  if (!x_server_pixel_buffer_.CaptureRect(DesktopRect::MakeSize(frame->size()),
+                                          frame.get())) {
+    RTC_LOG(LS_WARNING) << "Temporarily failed to capture winodw.";
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+
+  frame->mutable_updated_region()->SetRect(
+      DesktopRect::MakeSize(frame->size()));
+  frame->set_top_left(x_server_pixel_buffer_.window_rect().top_left());
+  frame->set_capturer_id(DesktopCapturerId::kX11CapturerLinux);
+
+  callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
+}
+
+bool WindowCapturerX11::IsOccluded(const DesktopVector& pos) {
+  return window_finder_.GetWindowUnderPoint(pos) !=
+         static_cast<WindowId>(selected_window_);
+}
+
+bool WindowCapturerX11::HandleXEvent(const XEvent& event) {
+  if (event.type == ConfigureNotify) {
+    XConfigureEvent xce = event.xconfigure;
+    if (xce.window == selected_window_) {
+      if (!DesktopRectFromXAttributes(xce).equals(
+              x_server_pixel_buffer_.window_rect())) {
+        if (!x_server_pixel_buffer_.Init(&atom_cache_, selected_window_)) {
+          RTC_LOG(LS_ERROR)
+              << "Failed to initialize pixel buffer after resizing.";
+        }
+      }
+    }
+  }
+
+  // Always returns false, so other observers can still receive the events.
+  return false;
+}
+
+bool WindowCapturerX11::GetWindowTitle(::Window window, std::string* title) {
+  int status;
+  bool result = false;
+  XTextProperty window_name;
+  window_name.value = nullptr;
+  if (window) {
+    status = XGetWMName(display(), window, &window_name);
+    if (status && window_name.value && window_name.nitems) {
+      int cnt;
+      char** list = nullptr;
+      status =
+          Xutf8TextPropertyToTextList(display(), &window_name, &list, &cnt);
+      if (status >= Success && cnt && *list) {
+        if (cnt > 1) {
+          RTC_LOG(LS_INFO) << "Window has " << cnt
+                           << " text properties, only using the first one.";
+        }
+        *title = *list;
+        result = true;
+      }
+      if (list)
+        XFreeStringList(list);
+    }
+    if (window_name.value)
+      XFree(window_name.value);
+  }
+  return result;
+}
+
+int WindowCapturerX11::GetWindowProcessID(::Window window) {
+  // Get _NET_WM_PID property of the window.
+  Atom process_atom = XInternAtom(display(), "_NET_WM_PID", True);
+  XWindowProperty<uint32_t> process_id(display(), window, process_atom);
+
+  return process_id.is_valid() ? *process_id.data() : 0;
+}
+
+// static
+std::unique_ptr<DesktopCapturer> WindowCapturerX11::CreateRawWindowCapturer(
+    const DesktopCaptureOptions& options) {
+  if (!options.x_display())
+    return nullptr;
+  return std::unique_ptr<DesktopCapturer>(new WindowCapturerX11(options));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.h
new file mode 100644
index 0000000000..cfd29eca66
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_capturer_x11.h
@@ -0,0 +1,78 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_WINDOW_CAPTURER_X11_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_WINDOW_CAPTURER_X11_H_
+
+#include <X11/X.h>
+#include <X11/Xlib.h>
+
+#include <memory>
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/linux/x11/shared_x_display.h"
+#include "modules/desktop_capture/linux/x11/x_window_property.h"
+#include "modules/desktop_capture/linux/x11/window_finder_x11.h"
+#include "modules/desktop_capture/linux/x11/x_atom_cache.h"
+#include "modules/desktop_capture/linux/x11/x_server_pixel_buffer.h"
+
+namespace webrtc {
+
+class WindowCapturerX11 : public DesktopCapturer,
+                          public SharedXDisplay::XEventHandler {
+ public:
+  explicit WindowCapturerX11(const DesktopCaptureOptions& options);
+  ~WindowCapturerX11() override;
+
+  WindowCapturerX11(const WindowCapturerX11&) = delete;
+  WindowCapturerX11& operator=(const WindowCapturerX11&) = delete;
+
+  static std::unique_ptr<DesktopCapturer> CreateRawWindowCapturer(
+      const DesktopCaptureOptions& options);
+
+  // DesktopCapturer interface.
+  void Start(Callback* callback) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  bool IsOccluded(const DesktopVector& pos) override;
+
+  // SharedXDisplay::XEventHandler interface.
+  bool HandleXEvent(const XEvent& event) override;
+
+ private:
+  Display* display() { return x_display_->display(); }
+
+  // Returns window title for the specified X `window`.
+  bool GetWindowTitle(::Window window, std::string* title);
+
+  // Returns the id of the owning process.
+  int GetWindowProcessID(::Window window);
+
+  Callback* callback_ = nullptr;
+
+  rtc::scoped_refptr<SharedXDisplay> x_display_;
+
+  bool has_composite_extension_ = false;
+
+  ::Window selected_window_ = 0;
+  XServerPixelBuffer x_server_pixel_buffer_;
+  XAtomCache atom_cache_;
+  WindowFinderX11 window_finder_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_WINDOW_CAPTURER_X11_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc
new file mode 100644
index 0000000000..dec17ab51f
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.cc
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/window_finder_x11.h"
+
+#include <X11/X.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/linux/x11/window_list_utils.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+WindowFinderX11::WindowFinderX11(XAtomCache* cache) : cache_(cache) {
+  RTC_DCHECK(cache_);
+}
+
+WindowFinderX11::~WindowFinderX11() = default;
+
+WindowId WindowFinderX11::GetWindowUnderPoint(DesktopVector point) {
+  WindowId id = kNullWindowId;
+  GetWindowList(cache_, [&id, this, point](::Window window) {
+    DesktopRect rect;
+    if (GetWindowRect(this->cache_->display(), window, &rect) &&
+        rect.Contains(point)) {
+      id = window;
+      return false;
+    }
+    return true;
+  });
+  return id;
+}
+
+// static
+std::unique_ptr<WindowFinder> WindowFinder::Create(
+    const WindowFinder::Options& options) {
+  if (options.cache == nullptr) {
+    return nullptr;
+  }
+
+  return std::make_unique<WindowFinderX11>(options.cache);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.h
new file mode 100644
index 0000000000..91de876417
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_finder_x11.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_WINDOW_FINDER_X11_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_WINDOW_FINDER_X11_H_
+
+#include "modules/desktop_capture/window_finder.h"
+
+namespace webrtc {
+
+class XAtomCache;
+
+// The implementation of WindowFinder for X11.
+class WindowFinderX11 final : public WindowFinder {
+ public:
+  explicit WindowFinderX11(XAtomCache* cache);
+  ~WindowFinderX11() override;
+
+  // WindowFinder implementation.
+  WindowId GetWindowUnderPoint(DesktopVector point) override;
+
+ private:
+  XAtomCache* const cache_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_WINDOW_FINDER_X11_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc
new file mode 100644
index 0000000000..ff2d467e29
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.cc
@@ -0,0 +1,198 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/window_list_utils.h"
+
+#include <X11/Xlib.h>
+#include <X11/Xutil.h>
+#include <string.h>
+
+#include <algorithm>
+
+#include "modules/desktop_capture/linux/x11/x_error_trap.h"
+#include "modules/desktop_capture/linux/x11/x_window_property.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+class DeferXFree {
+ public:
+  explicit DeferXFree(void* data) : data_(data) {}
+  ~DeferXFree();
+
+ private:
+  void* const data_;
+};
+
+DeferXFree::~DeferXFree() {
+  if (data_)
+    XFree(data_);
+}
+
+// Iterates through `window` hierarchy to find first visible window, i.e. one
+// that has WM_STATE property set to NormalState.
+// See http://tronche.com/gui/x/icccm/sec-4.html#s-4.1.3.1 .
+::Window GetApplicationWindow(XAtomCache* cache, ::Window window) {
+  int32_t state = GetWindowState(cache, window);
+  if (state == NormalState) {
+    // Window has WM_STATE==NormalState. Return it.
+    return window;
+  } else if (state == IconicState) {
+    // Window is in minimized. Skip it.
+    return 0;
+  }
+
+  RTC_DCHECK_EQ(state, WithdrawnState);
+  // If the window is in WithdrawnState then look at all of its children.
+  ::Window root, parent;
+  ::Window* children;
+  unsigned int num_children;
+  if (!XQueryTree(cache->display(), window, &root, &parent, &children,
+                  &num_children)) {
+    RTC_LOG(LS_ERROR) << "Failed to query for child windows although window"
+                         "does not have a valid WM_STATE.";
+    return 0;
+  }
+  ::Window app_window = 0;
+  for (unsigned int i = 0; i < num_children; ++i) {
+    app_window = GetApplicationWindow(cache, children[i]);
+    if (app_window)
+      break;
+  }
+
+  if (children)
+    XFree(children);
+  return app_window;
+}
+
+// Returns true if the `window` is a desktop element.
+bool IsDesktopElement(XAtomCache* cache, ::Window window) {
+  RTC_DCHECK(cache);
+  if (window == 0)
+    return false;
+
+  // First look for _NET_WM_WINDOW_TYPE. The standard
+  // (http://standards.freedesktop.org/wm-spec/latest/ar01s05.html#id2760306)
+  // says this hint *should* be present on all windows, and we use the existence
+  // of _NET_WM_WINDOW_TYPE_NORMAL in the property to indicate a window is not
+  // a desktop element (that is, only "normal" windows should be shareable).
+  XWindowProperty<uint32_t> window_type(cache->display(), window,
+                                        cache->WindowType());
+  if (window_type.is_valid() && window_type.size() > 0) {
+    uint32_t* end = window_type.data() + window_type.size();
+    bool is_normal =
+        (end != std::find(window_type.data(), end, cache->WindowTypeNormal()));
+    return !is_normal;
+  }
+
+  // Fall back on using the hint.
+  XClassHint class_hint;
+  Status status = XGetClassHint(cache->display(), window, &class_hint);
+  if (status == 0) {
+    // No hints, assume this is a normal application window.
+    return false;
+  }
+
+  DeferXFree free_res_name(class_hint.res_name);
+  DeferXFree free_res_class(class_hint.res_class);
+  return strcmp("gnome-panel", class_hint.res_name) == 0 ||
+         strcmp("desktop_window", class_hint.res_name) == 0;
+}
+
+}  // namespace
+
+int32_t GetWindowState(XAtomCache* cache, ::Window window) {
+  // Get WM_STATE property of the window.
+  XWindowProperty<uint32_t> window_state(cache->display(), window,
+                                         cache->WmState());
+
+  // WM_STATE is considered to be set to WithdrawnState when it missing.
+  return window_state.is_valid() ? *window_state.data() : WithdrawnState;
+}
+
+bool GetWindowList(XAtomCache* cache,
+                   rtc::FunctionView<bool(::Window)> on_window) {
+  RTC_DCHECK(cache);
+  RTC_DCHECK(on_window);
+  ::Display* const display = cache->display();
+
+  int failed_screens = 0;
+  const int num_screens = XScreenCount(display);
+  for (int screen = 0; screen < num_screens; screen++) {
+    ::Window root_window = XRootWindow(display, screen);
+    ::Window parent;
+    ::Window* children;
+    unsigned int num_children;
+    {
+      XErrorTrap error_trap(display);
+      if (XQueryTree(display, root_window, &root_window, &parent, &children,
+                     &num_children) == 0 ||
+          error_trap.GetLastErrorAndDisable() != 0) {
+        failed_screens++;
+        RTC_LOG(LS_ERROR) << "Failed to query for child windows for screen "
+                          << screen;
+        continue;
+      }
+    }
+
+    DeferXFree free_children(children);
+
+    for (unsigned int i = 0; i < num_children; i++) {
+      // Iterates in reverse order to return windows from front to back.
+      ::Window app_window =
+          GetApplicationWindow(cache, children[num_children - 1 - i]);
+      if (app_window && !IsDesktopElement(cache, app_window)) {
+        if (!on_window(app_window)) {
+          return true;
+        }
+      }
+    }
+  }
+
+  return failed_screens < num_screens;
+}
+
+bool GetWindowRect(::Display* display,
+                   ::Window window,
+                   DesktopRect* rect,
+                   XWindowAttributes* attributes /* = nullptr */) {
+  XWindowAttributes local_attributes;
+  int offset_x;
+  int offset_y;
+  if (attributes == nullptr) {
+    attributes = &local_attributes;
+  }
+
+  {
+    XErrorTrap error_trap(display);
+    if (!XGetWindowAttributes(display, window, attributes) ||
+        error_trap.GetLastErrorAndDisable() != 0) {
+      return false;
+    }
+  }
+  *rect = DesktopRectFromXAttributes(*attributes);
+
+  {
+    XErrorTrap error_trap(display);
+    ::Window child;
+    if (!XTranslateCoordinates(display, window, attributes->root, -rect->left(),
+                               -rect->top(), &offset_x, &offset_y, &child) ||
+        error_trap.GetLastErrorAndDisable() != 0) {
+      return false;
+    }
+  }
+  rect->Translate(offset_x, offset_y);
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.h
new file mode 100644
index 0000000000..923842df14
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/window_list_utils.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_WINDOW_LIST_UTILS_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_WINDOW_LIST_UTILS_H_
+
+#include <X11/X.h>
+#include <X11/Xlib.h>
+#include <stdint.h>
+
+#include "api/function_view.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/linux/x11/x_atom_cache.h"
+
+namespace webrtc {
+
+// Synchronously iterates all on-screen windows in `cache`.display() in
+// decreasing z-order and sends them one-by-one to `on_window` function before
+// GetWindowList() returns. If `on_window` returns false, this function ignores
+// other windows and returns immediately. GetWindowList() returns false if
+// native APIs failed. If multiple screens are attached to the `display`, this
+// function returns false only when native APIs failed on all screens. Menus,
+// panels and minimized windows will be ignored.
+bool GetWindowList(XAtomCache* cache,
+                   rtc::FunctionView<bool(::Window)> on_window);
+
+// Returns WM_STATE property of the `window`. This function returns
+// WithdrawnState if the `window` is missing.
+int32_t GetWindowState(XAtomCache* cache, ::Window window);
+
+// Returns the rectangle of the `window` in the coordinates of `display`. This
+// function returns false if native APIs failed. If `attributes` is provided, it
+// will be filled with the attributes of `window`. The `rect` is in system
+// coordinate, i.e. the primary monitor always starts from (0, 0).
+bool GetWindowRect(::Display* display,
+                   ::Window window,
+                   DesktopRect* rect,
+                   XWindowAttributes* attributes = nullptr);
+
+// Creates a DesktopRect from `attributes`.
+template <typename T>
+DesktopRect DesktopRectFromXAttributes(const T& attributes) {
+  return DesktopRect::MakeXYWH(attributes.x, attributes.y, attributes.width,
+                               attributes.height);
+}
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_WINDOW_LIST_UTILS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc
new file mode 100644
index 0000000000..157ba8b8fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/x_atom_cache.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+XAtomCache::XAtomCache(::Display* display) : display_(display) {
+  RTC_DCHECK(display_);
+}
+
+XAtomCache::~XAtomCache() = default;
+
+::Display* XAtomCache::display() const {
+  return display_;
+}
+
+Atom XAtomCache::WmState() {
+  return CreateIfNotExist(&wm_state_, "WM_STATE");
+}
+
+Atom XAtomCache::WindowType() {
+  return CreateIfNotExist(&window_type_, "_NET_WM_WINDOW_TYPE");
+}
+
+Atom XAtomCache::WindowTypeNormal() {
+  return CreateIfNotExist(&window_type_normal_, "_NET_WM_WINDOW_TYPE_NORMAL");
+}
+
+Atom XAtomCache::IccProfile() {
+  return CreateIfNotExist(&icc_profile_, "_ICC_PROFILE");
+}
+
+Atom XAtomCache::CreateIfNotExist(Atom* atom, const char* name) {
+  RTC_DCHECK(atom);
+  if (*atom == None) {
+    *atom = XInternAtom(display(), name, True);
+  }
+  return *atom;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.h
new file mode 100644
index 0000000000..39d957e98b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_atom_cache.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_X_ATOM_CACHE_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_X_ATOM_CACHE_H_
+
+#include <X11/X.h>
+#include <X11/Xlib.h>
+
+namespace webrtc {
+
+// A cache of Atom. Each Atom object is created on demand.
+class XAtomCache final {
+ public:
+  explicit XAtomCache(::Display* display);
+  ~XAtomCache();
+
+  ::Display* display() const;
+
+  Atom WmState();
+  Atom WindowType();
+  Atom WindowTypeNormal();
+  Atom IccProfile();
+
+ private:
+  // If |*atom| is None, this function uses XInternAtom() to retrieve an Atom.
+  Atom CreateIfNotExist(Atom* atom, const char* name);
+
+  ::Display* const display_;
+  Atom wm_state_ = None;
+  Atom window_type_ = None;
+  Atom window_type_normal_ = None;
+  Atom icc_profile_ = None;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_X_ATOM_CACHE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc
new file mode 100644
index 0000000000..3314dd286c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/x_error_trap.h"
+
+#include <stddef.h>
+
+#include <limits>
+
+#include "rtc_base/checks.h"
+
+
+namespace webrtc {
+
+Bool XErrorTrap::XServerErrorHandler(Display* display, xReply* rep,
+                                     char* /* buf */, int /* len */,
+                                     XPointer data) {
+  XErrorTrap* self = reinterpret_cast<XErrorTrap*>(data);
+  if (rep->generic.type != X_Error ||
+      // Overflow-safe last_request_read <= last_ignored_request_ for skipping
+      // async replies from requests before XErrorTrap was created.
+      self->last_ignored_request_ - display->last_request_read <
+      std::numeric_limits<unsigned long>::max() >> 1)
+    return False;
+  self->last_xserver_error_code_ = rep->error.errorCode;
+  return True;
+}
+
+XErrorTrap::XErrorTrap(Display* display)
+    : display_(display),
+      last_xserver_error_code_(0),
+      enabled_(true) {
+  // Use async_handlers instead of XSetErrorHandler().  async_handlers can
+  // remain in place and then be safely removed at the right time even if a
+  // handler change happens concurrently on another thread.  async_handlers
+  // are processed first and so can prevent errors reaching the global
+  // XSetErrorHandler handler.  They also will not see errors from or affect
+  // handling of errors on other Displays, which may be processed on other
+  // threads.
+  LockDisplay(display);
+  async_handler_.next = display->async_handlers;
+  async_handler_.handler = XServerErrorHandler;
+  async_handler_.data = reinterpret_cast<XPointer>(this);
+  display->async_handlers = &async_handler_;
+  last_ignored_request_ = display->request;
+  UnlockDisplay(display);
+}
+
+int XErrorTrap::GetLastErrorAndDisable() {
+  assert(enabled_);
+  enabled_ = false;
+  LockDisplay(display_);
+  DeqAsyncHandler(display_, &async_handler_);
+  UnlockDisplay(display_);
+  return last_xserver_error_code_;
+}
+
+XErrorTrap::~XErrorTrap() {
+  if (enabled_)
+    GetLastErrorAndDisable();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.h
new file mode 100644
index 0000000000..df7e86bf03
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_error_trap.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_X_ERROR_TRAP_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_X_ERROR_TRAP_H_
+
+#include <X11/Xlibint.h>
+#undef max // Xlibint.h defines this and it breaks std::max
+#undef min // Xlibint.h defines this and it breaks std::min
+
+namespace webrtc {
+
+// Helper class that registers X Window error handler. Caller can use
+// GetLastErrorAndDisable() to get the last error that was caught, if any.
+// An XErrorTrap may be constructed on any thread, but errors are collected
+// from all threads and so |display| should be used only on one thread.
+// Other Displays are unaffected.
+class XErrorTrap {
+ public:
+  explicit XErrorTrap(Display* display);
+  ~XErrorTrap();
+
+  XErrorTrap(const XErrorTrap&) = delete;
+  XErrorTrap& operator=(const XErrorTrap&) = delete;
+
+  // Returns last error and removes unregisters the error handler.
+  // Must not be called more than once.
+  int GetLastErrorAndDisable();
+
+ private:
+  static Bool XServerErrorHandler(Display* display, xReply* rep,
+                                  char* /* buf */, int /* len */,
+                                  XPointer data);
+
+  _XAsyncHandler async_handler_;
+  Display* display_;
+  unsigned long last_ignored_request_;
+  int last_xserver_error_code_;
+  bool enabled_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_X_ERROR_TRAP_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc
new file mode 100644
index 0000000000..fd6fc7daf4
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.cc
@@ -0,0 +1,379 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/x_server_pixel_buffer.h"
+
+#include <X11/Xutil.h>
+#include <stdint.h>
+#include <string.h>
+#include <sys/ipc.h>
+#include <sys/shm.h>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/linux/x11/window_list_utils.h"
+#include "modules/desktop_capture/linux/x11/x_error_trap.h"
+#include "modules/desktop_capture/linux/x11/x_window_property.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// Returns the number of bits `mask` has to be shifted left so its last
+// (most-significant) bit set becomes the most-significant bit of the word.
+// When `mask` is 0 the function returns 31.
+uint32_t MaskToShift(uint32_t mask) {
+  int shift = 0;
+  if ((mask & 0xffff0000u) == 0) {
+    mask <<= 16;
+    shift += 16;
+  }
+  if ((mask & 0xff000000u) == 0) {
+    mask <<= 8;
+    shift += 8;
+  }
+  if ((mask & 0xf0000000u) == 0) {
+    mask <<= 4;
+    shift += 4;
+  }
+  if ((mask & 0xc0000000u) == 0) {
+    mask <<= 2;
+    shift += 2;
+  }
+  if ((mask & 0x80000000u) == 0)
+    shift += 1;
+
+  return shift;
+}
+
+// Returns true if `image` is in RGB format.
+bool IsXImageRGBFormat(XImage* image) {
+  return image->bits_per_pixel == 32 && image->red_mask == 0xff0000 &&
+         image->green_mask == 0xff00 && image->blue_mask == 0xff;
+}
+
+// We expose two forms of blitting to handle variations in the pixel format.
+// In FastBlit(), the operation is effectively a memcpy.
+void FastBlit(XImage* x_image,
+              uint8_t* src_pos,
+              const DesktopRect& rect,
+              DesktopFrame* frame) {
+  RTC_DCHECK_LE(frame->top_left().x(), rect.left());
+  RTC_DCHECK_LE(frame->top_left().y(), rect.top());
+
+  int src_stride = x_image->bytes_per_line;
+  int dst_x = rect.left() - frame->top_left().x();
+  int dst_y = rect.top() - frame->top_left().y();
+
+  uint8_t* dst_pos = frame->data() + frame->stride() * dst_y;
+  dst_pos += dst_x * DesktopFrame::kBytesPerPixel;
+
+  int height = rect.height();
+  int row_bytes = rect.width() * DesktopFrame::kBytesPerPixel;
+  for (int y = 0; y < height; ++y) {
+    memcpy(dst_pos, src_pos, row_bytes);
+    src_pos += src_stride;
+    dst_pos += frame->stride();
+  }
+}
+
+void SlowBlit(XImage* x_image,
+              uint8_t* src_pos,
+              const DesktopRect& rect,
+              DesktopFrame* frame) {
+  RTC_DCHECK_LE(frame->top_left().x(), rect.left());
+  RTC_DCHECK_LE(frame->top_left().y(), rect.top());
+
+  int src_stride = x_image->bytes_per_line;
+  int dst_x = rect.left() - frame->top_left().x();
+  int dst_y = rect.top() - frame->top_left().y();
+  int width = rect.width(), height = rect.height();
+
+  uint32_t red_mask = x_image->red_mask;
+  uint32_t green_mask = x_image->red_mask;
+  uint32_t blue_mask = x_image->blue_mask;
+
+  uint32_t red_shift = MaskToShift(red_mask);
+  uint32_t green_shift = MaskToShift(green_mask);
+  uint32_t blue_shift = MaskToShift(blue_mask);
+
+  int bits_per_pixel = x_image->bits_per_pixel;
+
+  uint8_t* dst_pos = frame->data() + frame->stride() * dst_y;
+  dst_pos += dst_x * DesktopFrame::kBytesPerPixel;
+  // TODO(hclam): Optimize, perhaps using MMX code or by converting to
+  // YUV directly.
+  // TODO(sergeyu): This code doesn't handle XImage byte order properly and
+  // won't work with 24bpp images. Fix it.
+  for (int y = 0; y < height; y++) {
+    uint32_t* dst_pos_32 = reinterpret_cast<uint32_t*>(dst_pos);
+    uint32_t* src_pos_32 = reinterpret_cast<uint32_t*>(src_pos);
+    uint16_t* src_pos_16 = reinterpret_cast<uint16_t*>(src_pos);
+    for (int x = 0; x < width; x++) {
+      // Dereference through an appropriately-aligned pointer.
+      uint32_t pixel;
+      if (bits_per_pixel == 32) {
+        pixel = src_pos_32[x];
+      } else if (bits_per_pixel == 16) {
+        pixel = src_pos_16[x];
+      } else {
+        pixel = src_pos[x];
+      }
+      uint32_t r = (pixel & red_mask) << red_shift;
+      uint32_t g = (pixel & green_mask) << green_shift;
+      uint32_t b = (pixel & blue_mask) << blue_shift;
+      // Write as 32-bit RGB.
+      dst_pos_32[x] =
+          ((r >> 8) & 0xff0000) | ((g >> 16) & 0xff00) | ((b >> 24) & 0xff);
+    }
+    dst_pos += frame->stride();
+    src_pos += src_stride;
+  }
+}
+
+}  // namespace
+
+XServerPixelBuffer::XServerPixelBuffer() {}
+
+XServerPixelBuffer::~XServerPixelBuffer() {
+  Release();
+}
+
+void XServerPixelBuffer::Release() {
+  if (x_image_) {
+    XDestroyImage(x_image_);
+    x_image_ = nullptr;
+  }
+  if (x_shm_image_) {
+    XDestroyImage(x_shm_image_);
+    x_shm_image_ = nullptr;
+  }
+  if (shm_pixmap_) {
+    XFreePixmap(display_, shm_pixmap_);
+    shm_pixmap_ = 0;
+  }
+  if (shm_gc_) {
+    XFreeGC(display_, shm_gc_);
+    shm_gc_ = nullptr;
+  }
+
+  ReleaseSharedMemorySegment();
+
+  window_ = 0;
+}
+
+void XServerPixelBuffer::ReleaseSharedMemorySegment() {
+  if (!shm_segment_info_)
+    return;
+  if (shm_segment_info_->shmaddr != nullptr)
+    shmdt(shm_segment_info_->shmaddr);
+  if (shm_segment_info_->shmid != -1)
+    shmctl(shm_segment_info_->shmid, IPC_RMID, 0);
+  delete shm_segment_info_;
+  shm_segment_info_ = nullptr;
+}
+
+bool XServerPixelBuffer::Init(XAtomCache* cache, Window window) {
+  Release();
+  display_ = cache->display();
+
+  XWindowAttributes attributes;
+  if (!GetWindowRect(display_, window, &window_rect_, &attributes)) {
+    return false;
+  }
+
+  if (cache->IccProfile() != None) {
+    // `window` is the root window when doing screen capture.
+    XWindowProperty<uint8_t> icc_profile_property(cache->display(), window,
+                                                  cache->IccProfile());
+    if (icc_profile_property.is_valid() && icc_profile_property.size() > 0) {
+      icc_profile_ = std::vector<uint8_t>(
+          icc_profile_property.data(),
+          icc_profile_property.data() + icc_profile_property.size());
+    } else {
+      RTC_LOG(LS_WARNING) << "Failed to get icc profile";
+    }
+  }
+
+  window_ = window;
+  InitShm(attributes);
+
+  return true;
+}
+
+void XServerPixelBuffer::InitShm(const XWindowAttributes& attributes) {
+  Visual* default_visual = attributes.visual;
+  int default_depth = attributes.depth;
+
+  int major, minor;
+  Bool have_pixmaps;
+  if (!XShmQueryVersion(display_, &major, &minor, &have_pixmaps)) {
+    // Shared memory not supported. CaptureRect will use the XImage API instead.
+    return;
+  }
+
+  bool using_shm = false;
+  shm_segment_info_ = new XShmSegmentInfo;
+  shm_segment_info_->shmid = -1;
+  shm_segment_info_->shmaddr = nullptr;
+  shm_segment_info_->readOnly = False;
+  x_shm_image_ = XShmCreateImage(display_, default_visual, default_depth,
+                                 ZPixmap, 0, shm_segment_info_,
+                                 window_rect_.width(), window_rect_.height());
+  if (x_shm_image_) {
+    shm_segment_info_->shmid =
+        shmget(IPC_PRIVATE, x_shm_image_->bytes_per_line * x_shm_image_->height,
+               IPC_CREAT | 0600);
+    if (shm_segment_info_->shmid != -1) {
+      void* shmat_result = shmat(shm_segment_info_->shmid, 0, 0);
+      if (shmat_result != reinterpret_cast<void*>(-1)) {
+        shm_segment_info_->shmaddr = reinterpret_cast<char*>(shmat_result);
+        x_shm_image_->data = shm_segment_info_->shmaddr;
+
+        XErrorTrap error_trap(display_);
+        using_shm = XShmAttach(display_, shm_segment_info_);
+        XSync(display_, False);
+        if (error_trap.GetLastErrorAndDisable() != 0)
+          using_shm = false;
+        if (using_shm) {
+          RTC_LOG(LS_VERBOSE)
+              << "Using X shared memory segment " << shm_segment_info_->shmid;
+        }
+      }
+    } else {
+      RTC_LOG(LS_WARNING) << "Failed to get shared memory segment. "
+                             "Performance may be degraded.";
+    }
+  }
+
+  if (!using_shm) {
+    RTC_LOG(LS_WARNING)
+        << "Not using shared memory. Performance may be degraded.";
+    ReleaseSharedMemorySegment();
+    return;
+  }
+
+  if (have_pixmaps)
+    have_pixmaps = InitPixmaps(default_depth);
+
+  shmctl(shm_segment_info_->shmid, IPC_RMID, 0);
+  shm_segment_info_->shmid = -1;
+
+  RTC_LOG(LS_VERBOSE) << "Using X shared memory extension v" << major << "."
+                      << minor << " with" << (have_pixmaps ? "" : "out")
+                      << " pixmaps.";
+}
+
+bool XServerPixelBuffer::InitPixmaps(int depth) {
+  if (XShmPixmapFormat(display_) != ZPixmap)
+    return false;
+
+  {
+    XErrorTrap error_trap(display_);
+    shm_pixmap_ = XShmCreatePixmap(
+        display_, window_, shm_segment_info_->shmaddr, shm_segment_info_,
+        window_rect_.width(), window_rect_.height(), depth);
+    XSync(display_, False);
+    if (error_trap.GetLastErrorAndDisable() != 0) {
+      // `shm_pixmap_` is not not valid because the request was not processed
+      // by the X Server, so zero it.
+      shm_pixmap_ = 0;
+      return false;
+    }
+  }
+
+  {
+    XErrorTrap error_trap(display_);
+    XGCValues shm_gc_values;
+    shm_gc_values.subwindow_mode = IncludeInferiors;
+    shm_gc_values.graphics_exposures = False;
+    shm_gc_ = XCreateGC(display_, window_,
+                        GCSubwindowMode | GCGraphicsExposures, &shm_gc_values);
+    XSync(display_, False);
+    if (error_trap.GetLastErrorAndDisable() != 0) {
+      XFreePixmap(display_, shm_pixmap_);
+      shm_pixmap_ = 0;
+      shm_gc_ = 0;  // See shm_pixmap_ comment above.
+      return false;
+    }
+  }
+
+  return true;
+}
+
+bool XServerPixelBuffer::IsWindowValid() const {
+  XWindowAttributes attributes;
+  {
+    XErrorTrap error_trap(display_);
+    if (!XGetWindowAttributes(display_, window_, &attributes) ||
+        error_trap.GetLastErrorAndDisable() != 0) {
+      return false;
+    }
+  }
+  return true;
+}
+
+void XServerPixelBuffer::Synchronize() {
+  if (shm_segment_info_ && !shm_pixmap_) {
+    // XShmGetImage can fail if the display is being reconfigured.
+    XErrorTrap error_trap(display_);
+    // XShmGetImage fails if the window is partially out of screen.
+    xshm_get_image_succeeded_ =
+        XShmGetImage(display_, window_, x_shm_image_, 0, 0, AllPlanes);
+  }
+}
+
+bool XServerPixelBuffer::CaptureRect(const DesktopRect& rect,
+                                     DesktopFrame* frame) {
+  RTC_DCHECK_LE(rect.right(), window_rect_.width());
+  RTC_DCHECK_LE(rect.bottom(), window_rect_.height());
+
+  XImage* image;
+  uint8_t* data;
+
+  if (shm_segment_info_ && (shm_pixmap_ || xshm_get_image_succeeded_)) {
+    if (shm_pixmap_) {
+      XCopyArea(display_, window_, shm_pixmap_, shm_gc_, rect.left(),
+                rect.top(), rect.width(), rect.height(), rect.left(),
+                rect.top());
+      XSync(display_, False);
+    }
+
+    image = x_shm_image_;
+    data = reinterpret_cast<uint8_t*>(image->data) +
+           rect.top() * image->bytes_per_line +
+           rect.left() * image->bits_per_pixel / 8;
+
+  } else {
+    if (x_image_)
+      XDestroyImage(x_image_);
+    x_image_ = XGetImage(display_, window_, rect.left(), rect.top(),
+                         rect.width(), rect.height(), AllPlanes, ZPixmap);
+    if (!x_image_)
+      return false;
+
+    image = x_image_;
+    data = reinterpret_cast<uint8_t*>(image->data);
+  }
+
+  if (IsXImageRGBFormat(image)) {
+    FastBlit(image, data, rect, frame);
+  } else {
+    SlowBlit(image, data, rect, frame);
+  }
+
+  if (!icc_profile_.empty())
+    frame->set_icc_profile(icc_profile_);
+
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.h
new file mode 100644
index 0000000000..38af3a3e76
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_server_pixel_buffer.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Don't include this file in any .h files because it pulls in some X headers.
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_X_SERVER_PIXEL_BUFFER_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_X_SERVER_PIXEL_BUFFER_H_
+
+#include <X11/Xutil.h>
+#include <X11/extensions/XShm.h>
+
+#include <memory>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+
+namespace webrtc {
+
+class DesktopFrame;
+class XAtomCache;
+
+// A class to allow the X server's pixel buffer to be accessed as efficiently
+// as possible.
+class XServerPixelBuffer {
+ public:
+  XServerPixelBuffer();
+  ~XServerPixelBuffer();
+
+  XServerPixelBuffer(const XServerPixelBuffer&) = delete;
+  XServerPixelBuffer& operator=(const XServerPixelBuffer&) = delete;
+
+  void Release();
+
+  // Allocate (or reallocate) the pixel buffer for `window`. Returns false in
+  // case of an error (e.g. window doesn't exist).
+  bool Init(XAtomCache* cache, Window window);
+
+  bool is_initialized() { return window_ != 0; }
+
+  // Returns the size of the window the buffer was initialized for.
+  DesktopSize window_size() { return window_rect_.size(); }
+
+  // Returns the rectangle of the window the buffer was initialized for.
+  const DesktopRect& window_rect() { return window_rect_; }
+
+  // Returns true if the window can be found.
+  bool IsWindowValid() const;
+
+  // If shared memory is being used without pixmaps, synchronize this pixel
+  // buffer with the root window contents (otherwise, this is a no-op).
+  // This is to avoid doing a full-screen capture for each individual
+  // rectangle in the capture list, when it only needs to be done once at the
+  // beginning.
+  void Synchronize();
+
+  // Capture the specified rectangle and stores it in the `frame`. In the case
+  // where the full-screen data is captured by Synchronize(), this simply
+  // returns the pointer without doing any more work. The caller must ensure
+  // that `rect` is not larger than window_size().
+  bool CaptureRect(const DesktopRect& rect, DesktopFrame* frame);
+
+ private:
+  void ReleaseSharedMemorySegment();
+
+  void InitShm(const XWindowAttributes& attributes);
+  bool InitPixmaps(int depth);
+
+  Display* display_ = nullptr;
+  Window window_ = 0;
+  DesktopRect window_rect_;
+  XImage* x_image_ = nullptr;
+  XShmSegmentInfo* shm_segment_info_ = nullptr;
+  XImage* x_shm_image_ = nullptr;
+  Pixmap shm_pixmap_ = 0;
+  GC shm_gc_ = nullptr;
+  bool xshm_get_image_succeeded_ = false;
+  std::vector<uint8_t> icc_profile_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_X_SERVER_PIXEL_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc
new file mode 100644
index 0000000000..5e16dac404
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.cc
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/linux/x11/x_window_property.h"
+
+namespace webrtc {
+
+XWindowPropertyBase::XWindowPropertyBase(Display* display,
+                                         Window window,
+                                         Atom property,
+                                         int expected_size) {
+  const int kBitsPerByte = 8;
+  Atom actual_type;
+  int actual_format;
+  unsigned long bytes_after;  // NOLINT: type required by XGetWindowProperty
+  int status = XGetWindowProperty(display, window, property, 0L, ~0L, False,
+                                  AnyPropertyType, &actual_type, &actual_format,
+                                  &size_, &bytes_after, &data_);
+  if (status != Success) {
+    data_ = nullptr;
+    return;
+  }
+  if ((expected_size * kBitsPerByte) != actual_format) {
+    size_ = 0;
+    return;
+  }
+
+  is_valid_ = true;
+}
+
+XWindowPropertyBase::~XWindowPropertyBase() {
+  if (data_)
+    XFree(data_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.h b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.h
new file mode 100644
index 0000000000..28dfb97311
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/x11/x_window_property.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X11_X_WINDOW_PROPERTY_H_
+#define MODULES_DESKTOP_CAPTURE_LINUX_X11_X_WINDOW_PROPERTY_H_
+
+#include <X11/X.h>
+#include <X11/Xlib.h>
+
+namespace webrtc {
+
+class XWindowPropertyBase {
+ public:
+  XWindowPropertyBase(Display* display,
+                      Window window,
+                      Atom property,
+                      int expected_size);
+  virtual ~XWindowPropertyBase();
+
+  XWindowPropertyBase(const XWindowPropertyBase&) = delete;
+  XWindowPropertyBase& operator=(const XWindowPropertyBase&) = delete;
+
+  // True if we got properly value successfully.
+  bool is_valid() const { return is_valid_; }
+
+  // Size and value of the property.
+  size_t size() const { return size_; }
+
+ protected:
+  unsigned char* data_ = nullptr;
+
+ private:
+  bool is_valid_ = false;
+  unsigned long size_ = 0;  // NOLINT: type required by XGetWindowProperty
+};
+
+// Convenience wrapper for XGetWindowProperty() results.
+template <class PropertyType>
+class XWindowProperty : public XWindowPropertyBase {
+ public:
+  XWindowProperty(Display* display, const Window window, const Atom property)
+      : XWindowPropertyBase(display, window, property, sizeof(PropertyType)) {}
+  ~XWindowProperty() override = default;
+
+  XWindowProperty(const XWindowProperty&) = delete;
+  XWindowProperty& operator=(const XWindowProperty&) = delete;
+
+  const PropertyType* data() const {
+    return reinterpret_cast<PropertyType*>(data_);
+  }
+  PropertyType* data() { return reinterpret_cast<PropertyType*>(data_); }
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X11_X_WINDOW_PROPERTY_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration.h b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration.h
new file mode 100644
index 0000000000..2ad5474e44
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration.h
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_CONFIGURATION_H_
+#define MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_CONFIGURATION_H_
+
+#include <ApplicationServices/ApplicationServices.h>
+
+#include <vector>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Describes the configuration of a specific display.
+struct MacDisplayConfiguration {
+  MacDisplayConfiguration();
+  MacDisplayConfiguration(const MacDisplayConfiguration& other);
+  MacDisplayConfiguration(MacDisplayConfiguration&& other);
+  ~MacDisplayConfiguration();
+
+  MacDisplayConfiguration& operator=(const MacDisplayConfiguration& other);
+  MacDisplayConfiguration& operator=(MacDisplayConfiguration&& other);
+
+  // Cocoa identifier for this display.
+  CGDirectDisplayID id = 0;
+
+  // Bounds of this display in Density-Independent Pixels (DIPs).
+  DesktopRect bounds;
+
+  // Bounds of this display in physical pixels.
+  DesktopRect pixel_bounds;
+
+  // Scale factor from DIPs to physical pixels.
+  float dip_to_pixel_scale = 1.0f;
+
+  // Display type, built-in or external.
+  bool is_builtin;
+};
+
+typedef std::vector<MacDisplayConfiguration> MacDisplayConfigurations;
+
+// Describes the configuration of the whole desktop.
+struct RTC_EXPORT MacDesktopConfiguration {
+  // Used to request bottom-up or top-down coordinates.
+  enum Origin { BottomLeftOrigin, TopLeftOrigin };
+
+  MacDesktopConfiguration();
+  MacDesktopConfiguration(const MacDesktopConfiguration& other);
+  MacDesktopConfiguration(MacDesktopConfiguration&& other);
+  ~MacDesktopConfiguration();
+
+  MacDesktopConfiguration& operator=(const MacDesktopConfiguration& other);
+  MacDesktopConfiguration& operator=(MacDesktopConfiguration&& other);
+
+  // Returns the desktop & display configurations.
+  // If BottomLeftOrigin is used, the output is in Cocoa-style "bottom-up"
+  // (the origin is the bottom-left of the primary monitor, and coordinates
+  // increase as you move up the screen). Otherwise, the configuration will be
+  // converted to follow top-left coordinate system as Windows and X11.
+  static MacDesktopConfiguration GetCurrent(Origin origin);
+
+  // Returns true if the given desktop configuration equals this one.
+  bool Equals(const MacDesktopConfiguration& other);
+
+  // If `id` corresponds to the built-in display, return its configuration,
+  // otherwise return the configuration for the display with the specified id,
+  // or nullptr if no such display exists.
+  const MacDisplayConfiguration* FindDisplayConfigurationById(
+      CGDirectDisplayID id);
+
+  // Bounds of the desktop excluding monitors with DPI settings different from
+  // the main monitor. In Density-Independent Pixels (DIPs).
+  DesktopRect bounds;
+
+  // Same as bounds, but expressed in physical pixels.
+  DesktopRect pixel_bounds;
+
+  // Scale factor from DIPs to physical pixels.
+  float dip_to_pixel_scale = 1.0f;
+
+  // Configurations of the displays making up the desktop area.
+  MacDisplayConfigurations displays;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_CONFIGURATION_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration.mm b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration.mm
new file mode 100644
index 0000000000..93fb3f6226
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration.mm
@@ -0,0 +1,189 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+
+#include <math.h>
+#include <algorithm>
+#include <Cocoa/Cocoa.h>
+
+#include "rtc_base/checks.h"
+
+#if !defined(MAC_OS_X_VERSION_10_7) || \
+    MAC_OS_X_VERSION_MIN_REQUIRED < MAC_OS_X_VERSION_10_7
+
+@interface NSScreen (LionAPI)
+- (CGFloat)backingScaleFactor;
+- (NSRect)convertRectToBacking:(NSRect)aRect;
+@end
+
+#endif  // MAC_OS_X_VERSION_10_7
+
+namespace webrtc {
+
+namespace {
+
+DesktopRect NSRectToDesktopRect(const NSRect& ns_rect) {
+  return DesktopRect::MakeLTRB(
+      static_cast<int>(floor(ns_rect.origin.x)),
+      static_cast<int>(floor(ns_rect.origin.y)),
+      static_cast<int>(ceil(ns_rect.origin.x + ns_rect.size.width)),
+      static_cast<int>(ceil(ns_rect.origin.y + ns_rect.size.height)));
+}
+
+// Inverts the position of `rect` from bottom-up coordinates to top-down,
+// relative to `bounds`.
+void InvertRectYOrigin(const DesktopRect& bounds,
+                       DesktopRect* rect) {
+  RTC_DCHECK_EQ(bounds.top(), 0);
+  *rect = DesktopRect::MakeXYWH(
+      rect->left(), bounds.bottom() - rect->bottom(),
+      rect->width(), rect->height());
+}
+
+MacDisplayConfiguration GetConfigurationForScreen(NSScreen* screen) {
+  MacDisplayConfiguration display_config;
+
+  // Fetch the NSScreenNumber, which is also the CGDirectDisplayID.
+  NSDictionary* device_description = [screen deviceDescription];
+  display_config.id = static_cast<CGDirectDisplayID>(
+      [[device_description objectForKey:@"NSScreenNumber"] intValue]);
+
+  // Determine the display's logical & physical dimensions.
+  NSRect ns_bounds = [screen frame];
+  display_config.bounds = NSRectToDesktopRect(ns_bounds);
+
+  // If the host is running Mac OS X 10.7+ or later, query the scaling factor
+  // between logical and physical (aka "backing") pixels, otherwise assume 1:1.
+  if ([screen respondsToSelector:@selector(backingScaleFactor)] &&
+      [screen respondsToSelector:@selector(convertRectToBacking:)]) {
+    display_config.dip_to_pixel_scale = [screen backingScaleFactor];
+    NSRect ns_pixel_bounds = [screen convertRectToBacking: ns_bounds];
+    display_config.pixel_bounds = NSRectToDesktopRect(ns_pixel_bounds);
+  } else {
+    display_config.pixel_bounds = display_config.bounds;
+  }
+
+  // Determine if the display is built-in or external.
+  display_config.is_builtin = CGDisplayIsBuiltin(display_config.id);
+
+  return display_config;
+}
+
+}  // namespace
+
+MacDisplayConfiguration::MacDisplayConfiguration() = default;
+MacDisplayConfiguration::MacDisplayConfiguration(
+    const MacDisplayConfiguration& other) = default;
+MacDisplayConfiguration::MacDisplayConfiguration(
+    MacDisplayConfiguration&& other) = default;
+MacDisplayConfiguration::~MacDisplayConfiguration() = default;
+
+MacDisplayConfiguration& MacDisplayConfiguration::operator=(
+    const MacDisplayConfiguration& other) = default;
+MacDisplayConfiguration& MacDisplayConfiguration::operator=(
+    MacDisplayConfiguration&& other) = default;
+
+MacDesktopConfiguration::MacDesktopConfiguration() = default;
+MacDesktopConfiguration::MacDesktopConfiguration(
+    const MacDesktopConfiguration& other) = default;
+MacDesktopConfiguration::MacDesktopConfiguration(
+    MacDesktopConfiguration&& other) = default;
+MacDesktopConfiguration::~MacDesktopConfiguration() = default;
+
+MacDesktopConfiguration& MacDesktopConfiguration::operator=(
+    const MacDesktopConfiguration& other) = default;
+MacDesktopConfiguration& MacDesktopConfiguration::operator=(
+    MacDesktopConfiguration&& other) = default;
+
+// static
+MacDesktopConfiguration MacDesktopConfiguration::GetCurrent(Origin origin) {
+  MacDesktopConfiguration desktop_config;
+
+  NSArray* screens = [NSScreen screens];
+  RTC_DCHECK(screens);
+
+  // Iterator over the monitors, adding the primary monitor and monitors whose
+  // DPI match that of the primary monitor.
+  for (NSUInteger i = 0; i < [screens count]; ++i) {
+    MacDisplayConfiguration display_config =
+        GetConfigurationForScreen([screens objectAtIndex: i]);
+
+    if (i == 0)
+      desktop_config.dip_to_pixel_scale = display_config.dip_to_pixel_scale;
+
+    // Cocoa uses bottom-up coordinates, so if the caller wants top-down then
+    // we need to invert the positions of secondary monitors relative to the
+    // primary one (the primary monitor's position is (0,0) in both systems).
+    if (i > 0 && origin == TopLeftOrigin) {
+      InvertRectYOrigin(desktop_config.displays[0].bounds,
+                        &display_config.bounds);
+      // `display_bounds` is density dependent, so we need to convert the
+      // primay monitor's position into the secondary monitor's density context.
+      float scaling_factor = display_config.dip_to_pixel_scale /
+          desktop_config.displays[0].dip_to_pixel_scale;
+      DesktopRect primary_bounds = DesktopRect::MakeLTRB(
+          desktop_config.displays[0].pixel_bounds.left() * scaling_factor,
+          desktop_config.displays[0].pixel_bounds.top() * scaling_factor,
+          desktop_config.displays[0].pixel_bounds.right() * scaling_factor,
+          desktop_config.displays[0].pixel_bounds.bottom() * scaling_factor);
+      InvertRectYOrigin(primary_bounds, &display_config.pixel_bounds);
+    }
+
+    // Add the display to the configuration.
+    desktop_config.displays.push_back(display_config);
+
+    // Update the desktop bounds to account for this display, unless the current
+    // display uses different DPI settings.
+    if (display_config.dip_to_pixel_scale ==
+        desktop_config.dip_to_pixel_scale) {
+      desktop_config.bounds.UnionWith(display_config.bounds);
+      desktop_config.pixel_bounds.UnionWith(display_config.pixel_bounds);
+    }
+  }
+
+  return desktop_config;
+}
+
+// For convenience of comparing MacDisplayConfigurations in
+// MacDesktopConfiguration::Equals.
+bool operator==(const MacDisplayConfiguration& left,
+                const MacDisplayConfiguration& right) {
+  return left.id == right.id &&
+      left.bounds.equals(right.bounds) &&
+      left.pixel_bounds.equals(right.pixel_bounds) &&
+      left.dip_to_pixel_scale == right.dip_to_pixel_scale;
+}
+
+bool MacDesktopConfiguration::Equals(const MacDesktopConfiguration& other) {
+  return bounds.equals(other.bounds) &&
+      pixel_bounds.equals(other.pixel_bounds) &&
+      dip_to_pixel_scale == other.dip_to_pixel_scale &&
+      displays == other.displays;
+}
+
+const MacDisplayConfiguration*
+MacDesktopConfiguration::FindDisplayConfigurationById(
+    CGDirectDisplayID id) {
+  bool is_builtin = CGDisplayIsBuiltin(id);
+  for (MacDisplayConfigurations::const_iterator it = displays.begin();
+      it != displays.end(); ++it) {
+    // The MBP having both discrete and integrated graphic cards will do
+    // automate graphics switching by default. When it switches from discrete to
+    // integrated one, the current display ID of the built-in display will
+    // change and this will cause screen capture stops.
+    // So make screen capture of built-in display continuing even if its display
+    // ID is changed.
+    if ((is_builtin && it->is_builtin) || (!is_builtin && it->id == id)) return &(*it);
+  }
+  return NULL;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
new file mode 100644
index 0000000000..048a679ecc
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
+
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+DesktopConfigurationMonitor::DesktopConfigurationMonitor() {
+  CGError err = CGDisplayRegisterReconfigurationCallback(
+      DesktopConfigurationMonitor::DisplaysReconfiguredCallback, this);
+  if (err != kCGErrorSuccess)
+    RTC_LOG(LS_ERROR) << "CGDisplayRegisterReconfigurationCallback " << err;
+  MutexLock lock(&desktop_configuration_lock_);
+  desktop_configuration_ = MacDesktopConfiguration::GetCurrent(
+      MacDesktopConfiguration::TopLeftOrigin);
+}
+
+DesktopConfigurationMonitor::~DesktopConfigurationMonitor() {
+  CGError err = CGDisplayRemoveReconfigurationCallback(
+      DesktopConfigurationMonitor::DisplaysReconfiguredCallback, this);
+  if (err != kCGErrorSuccess)
+    RTC_LOG(LS_ERROR) << "CGDisplayRemoveReconfigurationCallback " << err;
+}
+
+MacDesktopConfiguration DesktopConfigurationMonitor::desktop_configuration() {
+  MutexLock lock(&desktop_configuration_lock_);
+  return desktop_configuration_;
+}
+
+// static
+// This method may be called on any system thread.
+void DesktopConfigurationMonitor::DisplaysReconfiguredCallback(
+    CGDirectDisplayID display,
+    CGDisplayChangeSummaryFlags flags,
+    void* user_parameter) {
+  DesktopConfigurationMonitor* monitor =
+      reinterpret_cast<DesktopConfigurationMonitor*>(user_parameter);
+  monitor->DisplaysReconfigured(display, flags);
+}
+
+void DesktopConfigurationMonitor::DisplaysReconfigured(
+    CGDirectDisplayID display,
+    CGDisplayChangeSummaryFlags flags) {
+  TRACE_EVENT0("webrtc", "DesktopConfigurationMonitor::DisplaysReconfigured");
+  RTC_LOG(LS_INFO) << "DisplaysReconfigured: "
+                      "DisplayID "
+                   << display << "; ChangeSummaryFlags " << flags;
+
+  if (flags & kCGDisplayBeginConfigurationFlag) {
+    reconfiguring_displays_.insert(display);
+    return;
+  }
+
+  reconfiguring_displays_.erase(display);
+  if (reconfiguring_displays_.empty()) {
+    MutexLock lock(&desktop_configuration_lock_);
+    desktop_configuration_ = MacDesktopConfiguration::GetCurrent(
+        MacDesktopConfiguration::TopLeftOrigin);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
new file mode 100644
index 0000000000..747295a538
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_CONFIGURATION_MONITOR_H_
+#define MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_CONFIGURATION_MONITOR_H_
+
+#include <ApplicationServices/ApplicationServices.h>
+
+#include <memory>
+#include <set>
+
+#include "api/ref_counted_base.h"
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// The class provides functions to synchronize capturing and display
+// reconfiguring across threads, and the up-to-date MacDesktopConfiguration.
+class DesktopConfigurationMonitor final
+    : public rtc::RefCountedNonVirtual<DesktopConfigurationMonitor> {
+ public:
+  DesktopConfigurationMonitor();
+  ~DesktopConfigurationMonitor();
+
+  DesktopConfigurationMonitor(const DesktopConfigurationMonitor&) = delete;
+  DesktopConfigurationMonitor& operator=(const DesktopConfigurationMonitor&) =
+      delete;
+
+  // Returns the current desktop configuration.
+  MacDesktopConfiguration desktop_configuration();
+
+ private:
+  static void DisplaysReconfiguredCallback(CGDirectDisplayID display,
+                                           CGDisplayChangeSummaryFlags flags,
+                                           void* user_parameter);
+  void DisplaysReconfigured(CGDirectDisplayID display,
+                            CGDisplayChangeSummaryFlags flags);
+
+  Mutex desktop_configuration_lock_;
+  MacDesktopConfiguration desktop_configuration_
+      RTC_GUARDED_BY(&desktop_configuration_lock_);
+  std::set<CGDirectDisplayID> reconfiguring_displays_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_CONFIGURATION_MONITOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_cgimage.h b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_cgimage.h
new file mode 100644
index 0000000000..d6279f9b36
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_cgimage.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_CGIMAGE_H_
+#define MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_CGIMAGE_H_
+
+#include <CoreGraphics/CoreGraphics.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "sdk/objc/helpers/scoped_cftyperef.h"
+
+namespace webrtc {
+
+class DesktopFrameCGImage final : public DesktopFrame {
+ public:
+  // Create an image containing a snapshot of the display at the time this is
+  // being called.
+  static std::unique_ptr<DesktopFrameCGImage> CreateForDisplay(
+      CGDirectDisplayID display_id);
+
+  // Create an image containing a snaphot of the given window at the time this
+  // is being called. This also works when the window is overlapped or in
+  // another workspace.
+  static std::unique_ptr<DesktopFrameCGImage> CreateForWindow(
+      CGWindowID window_id);
+
+  ~DesktopFrameCGImage() override;
+
+  DesktopFrameCGImage(const DesktopFrameCGImage&) = delete;
+  DesktopFrameCGImage& operator=(const DesktopFrameCGImage&) = delete;
+
+ private:
+  static std::unique_ptr<DesktopFrameCGImage> CreateFromCGImage(
+      rtc::ScopedCFTypeRef<CGImageRef> cg_image);
+
+  // This constructor expects `cg_image` to hold a non-null CGImageRef.
+  DesktopFrameCGImage(DesktopSize size,
+                      int stride,
+                      uint8_t* data,
+                      rtc::ScopedCFTypeRef<CGImageRef> cg_image,
+                      rtc::ScopedCFTypeRef<CFDataRef> cg_data);
+
+  const rtc::ScopedCFTypeRef<CGImageRef> cg_image_;
+  const rtc::ScopedCFTypeRef<CFDataRef> cg_data_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_CGIMAGE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_cgimage.mm b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_cgimage.mm
new file mode 100644
index 0000000000..0fb69b272d
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_cgimage.mm
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mac/desktop_frame_cgimage.h"
+
+#include <AvailabilityMacros.h>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+// static
+std::unique_ptr<DesktopFrameCGImage> DesktopFrameCGImage::CreateForDisplay(
+    CGDirectDisplayID display_id) {
+  // Create an image containing a snapshot of the display.
+  rtc::ScopedCFTypeRef<CGImageRef> cg_image(CGDisplayCreateImage(display_id));
+  if (!cg_image) {
+    return nullptr;
+  }
+
+  return DesktopFrameCGImage::CreateFromCGImage(cg_image);
+}
+
+// static
+std::unique_ptr<DesktopFrameCGImage> DesktopFrameCGImage::CreateForWindow(CGWindowID window_id) {
+  rtc::ScopedCFTypeRef<CGImageRef> cg_image(
+      CGWindowListCreateImage(CGRectNull,
+                              kCGWindowListOptionIncludingWindow,
+                              window_id,
+                              kCGWindowImageBoundsIgnoreFraming));
+  if (!cg_image) {
+    return nullptr;
+  }
+
+  return DesktopFrameCGImage::CreateFromCGImage(cg_image);
+}
+
+// static
+std::unique_ptr<DesktopFrameCGImage> DesktopFrameCGImage::CreateFromCGImage(
+    rtc::ScopedCFTypeRef<CGImageRef> cg_image) {
+  // Verify that the image has 32-bit depth.
+  int bits_per_pixel = CGImageGetBitsPerPixel(cg_image.get());
+  if (bits_per_pixel / 8 != DesktopFrame::kBytesPerPixel) {
+    RTC_LOG(LS_ERROR) << "CGDisplayCreateImage() returned imaged with " << bits_per_pixel
+                      << " bits per pixel. Only 32-bit depth is supported.";
+    return nullptr;
+  }
+
+  // Request access to the raw pixel data via the image's DataProvider.
+  CGDataProviderRef cg_provider = CGImageGetDataProvider(cg_image.get());
+  RTC_DCHECK(cg_provider);
+
+  // CGDataProviderCopyData returns a new data object containing a copy of the provider’s
+  // data.
+  rtc::ScopedCFTypeRef<CFDataRef> cg_data(CGDataProviderCopyData(cg_provider));
+  RTC_DCHECK(cg_data);
+
+  // CFDataGetBytePtr returns a read-only pointer to the bytes of a CFData object.
+  uint8_t* data = const_cast<uint8_t*>(CFDataGetBytePtr(cg_data.get()));
+  RTC_DCHECK(data);
+
+  DesktopSize size(CGImageGetWidth(cg_image.get()), CGImageGetHeight(cg_image.get()));
+  int stride = CGImageGetBytesPerRow(cg_image.get());
+
+  std::unique_ptr<DesktopFrameCGImage> frame(
+      new DesktopFrameCGImage(size, stride, data, cg_image, cg_data));
+
+  CGColorSpaceRef cg_color_space = CGImageGetColorSpace(cg_image.get());
+  if (cg_color_space) {
+#if !defined(MAC_OS_X_VERSION_10_13) || MAC_OS_X_VERSION_MIN_REQUIRED < MAC_OS_X_VERSION_10_13
+    rtc::ScopedCFTypeRef<CFDataRef> cf_icc_profile(CGColorSpaceCopyICCProfile(cg_color_space));
+#else
+    rtc::ScopedCFTypeRef<CFDataRef> cf_icc_profile(CGColorSpaceCopyICCData(cg_color_space));
+#endif
+    if (cf_icc_profile) {
+      const uint8_t* data_as_byte =
+          reinterpret_cast<const uint8_t*>(CFDataGetBytePtr(cf_icc_profile.get()));
+      const size_t data_size = CFDataGetLength(cf_icc_profile.get());
+      if (data_as_byte && data_size > 0) {
+        frame->set_icc_profile(std::vector<uint8_t>(data_as_byte, data_as_byte + data_size));
+      }
+    }
+  }
+
+  return frame;
+}
+
+DesktopFrameCGImage::DesktopFrameCGImage(DesktopSize size,
+                                         int stride,
+                                         uint8_t* data,
+                                         rtc::ScopedCFTypeRef<CGImageRef> cg_image,
+                                         rtc::ScopedCFTypeRef<CFDataRef> cg_data)
+    : DesktopFrame(size, stride, data, nullptr), cg_image_(cg_image), cg_data_(cg_data) {
+  RTC_DCHECK(cg_image_);
+  RTC_DCHECK(cg_data_);
+}
+
+DesktopFrameCGImage::~DesktopFrameCGImage() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_iosurface.h b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_iosurface.h
new file mode 100644
index 0000000000..73da0f693c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_iosurface.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_IOSURFACE_H_
+#define MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_IOSURFACE_H_
+
+#include <CoreGraphics/CoreGraphics.h>
+#include <IOSurface/IOSurface.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "sdk/objc/helpers/scoped_cftyperef.h"
+
+namespace webrtc {
+
+class DesktopFrameIOSurface final : public DesktopFrame {
+ public:
+  // Lock an IOSurfaceRef containing a snapshot of a display. Return NULL if
+  // failed to lock.
+  static std::unique_ptr<DesktopFrameIOSurface> Wrap(
+      rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface);
+
+  ~DesktopFrameIOSurface() override;
+
+  DesktopFrameIOSurface(const DesktopFrameIOSurface&) = delete;
+  DesktopFrameIOSurface& operator=(const DesktopFrameIOSurface&) = delete;
+
+ private:
+  // This constructor expects `io_surface` to hold a non-null IOSurfaceRef.
+  explicit DesktopFrameIOSurface(rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface);
+
+  const rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_IOSURFACE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_iosurface.mm b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_iosurface.mm
new file mode 100644
index 0000000000..b59b319db9
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_iosurface.mm
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mac/desktop_frame_iosurface.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+// static
+std::unique_ptr<DesktopFrameIOSurface> DesktopFrameIOSurface::Wrap(
+    rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface) {
+  if (!io_surface) {
+    return nullptr;
+  }
+
+  IOSurfaceIncrementUseCount(io_surface.get());
+  IOReturn status = IOSurfaceLock(io_surface.get(), kIOSurfaceLockReadOnly, nullptr);
+  if (status != kIOReturnSuccess) {
+    RTC_LOG(LS_ERROR) << "Failed to lock the IOSurface with status " << status;
+    IOSurfaceDecrementUseCount(io_surface.get());
+    return nullptr;
+  }
+
+  // Verify that the image has 32-bit depth.
+  int bytes_per_pixel = IOSurfaceGetBytesPerElement(io_surface.get());
+  if (bytes_per_pixel != DesktopFrame::kBytesPerPixel) {
+    RTC_LOG(LS_ERROR) << "CGDisplayStream handler returned IOSurface with " << (8 * bytes_per_pixel)
+                      << " bits per pixel. Only 32-bit depth is supported.";
+    IOSurfaceUnlock(io_surface.get(), kIOSurfaceLockReadOnly, nullptr);
+    IOSurfaceDecrementUseCount(io_surface.get());
+    return nullptr;
+  }
+
+  return std::unique_ptr<DesktopFrameIOSurface>(new DesktopFrameIOSurface(io_surface));
+}
+
+DesktopFrameIOSurface::DesktopFrameIOSurface(rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface)
+    : DesktopFrame(
+          DesktopSize(IOSurfaceGetWidth(io_surface.get()), IOSurfaceGetHeight(io_surface.get())),
+          IOSurfaceGetBytesPerRow(io_surface.get()),
+          static_cast<uint8_t*>(IOSurfaceGetBaseAddress(io_surface.get())),
+          nullptr),
+      io_surface_(io_surface) {
+  RTC_DCHECK(io_surface_);
+}
+
+DesktopFrameIOSurface::~DesktopFrameIOSurface() {
+  IOSurfaceUnlock(io_surface_.get(), kIOSurfaceLockReadOnly, nullptr);
+  IOSurfaceDecrementUseCount(io_surface_.get());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.h b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.h
new file mode 100644
index 0000000000..aad28d2f30
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_PROVIDER_H_
+#define MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_PROVIDER_H_
+
+#include <CoreGraphics/CoreGraphics.h>
+#include <IOSurface/IOSurface.h>
+
+#include <map>
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "sdk/objc/helpers/scoped_cftyperef.h"
+
+namespace webrtc {
+
+class DesktopFrameProvider {
+ public:
+  explicit DesktopFrameProvider(bool allow_iosurface);
+  ~DesktopFrameProvider();
+
+  DesktopFrameProvider(const DesktopFrameProvider&) = delete;
+  DesktopFrameProvider& operator=(const DesktopFrameProvider&) = delete;
+
+  // The caller takes ownership of the returned desktop frame. Otherwise
+  // returns null if `display_id` is invalid or not ready. Note that this
+  // function does not remove the frame from the internal container. Caller
+  // has to call the Release function.
+  std::unique_ptr<DesktopFrame> TakeLatestFrameForDisplay(
+      CGDirectDisplayID display_id);
+
+  // OS sends the latest IOSurfaceRef through
+  // CGDisplayStreamFrameAvailableHandler callback; we store it here.
+  void InvalidateIOSurface(CGDirectDisplayID display_id,
+                           rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface);
+
+  // Expected to be called before stopping the CGDisplayStreamRef streams.
+  void Release();
+
+ private:
+  SequenceChecker thread_checker_;
+  const bool allow_iosurface_;
+
+  // Most recent IOSurface that contains a capture of matching display.
+  std::map<CGDirectDisplayID, std::unique_ptr<SharedDesktopFrame>> io_surfaces_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_PROVIDER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.mm b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.mm
new file mode 100644
index 0000000000..009504a22b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/desktop_frame_provider.mm
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mac/desktop_frame_provider.h"
+
+#include <utility>
+
+#include "modules/desktop_capture/mac/desktop_frame_cgimage.h"
+#include "modules/desktop_capture/mac/desktop_frame_iosurface.h"
+
+namespace webrtc {
+
+DesktopFrameProvider::DesktopFrameProvider(bool allow_iosurface)
+    : allow_iosurface_(allow_iosurface) {
+  thread_checker_.Detach();
+}
+
+DesktopFrameProvider::~DesktopFrameProvider() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  Release();
+}
+
+std::unique_ptr<DesktopFrame> DesktopFrameProvider::TakeLatestFrameForDisplay(
+    CGDirectDisplayID display_id) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (!allow_iosurface_ || !io_surfaces_[display_id]) {
+    // Regenerate a snapshot. If iosurface is on it will be empty until the
+    // stream handler is called.
+    return DesktopFrameCGImage::CreateForDisplay(display_id);
+  }
+
+  return io_surfaces_[display_id]->Share();
+}
+
+void DesktopFrameProvider::InvalidateIOSurface(CGDirectDisplayID display_id,
+                                               rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (!allow_iosurface_) {
+    return;
+  }
+
+  std::unique_ptr<DesktopFrameIOSurface> desktop_frame_iosurface =
+      DesktopFrameIOSurface::Wrap(io_surface);
+
+  io_surfaces_[display_id] = desktop_frame_iosurface ?
+      SharedDesktopFrame::Wrap(std::move(desktop_frame_iosurface)) :
+      nullptr;
+}
+
+void DesktopFrameProvider::Release() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  if (!allow_iosurface_) {
+    return;
+  }
+
+  io_surfaces_.clear();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/full_screen_mac_application_handler.cc b/third_party/libwebrtc/modules/desktop_capture/mac/full_screen_mac_application_handler.cc
new file mode 100644
index 0000000000..45cd3223d2
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/full_screen_mac_application_handler.cc
@@ -0,0 +1,238 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mac/full_screen_mac_application_handler.h"
+
+#include <libproc.h>
+
+#include <algorithm>
+#include <functional>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/function_view.h"
+#include "modules/desktop_capture/mac/window_list_utils.h"
+
+namespace webrtc {
+namespace {
+
+static constexpr const char* kPowerPointSlideShowTitles[] = {
+    "PowerPoint-Bildschirmpräsentation",
+    "Προβολή παρουσίασης PowerPoint",
+    "PowerPoint スライド ショー",
+    "PowerPoint Slide Show",
+    "PowerPoint 幻灯片放映",
+    "Presentación de PowerPoint",
+    "PowerPoint-slideshow",
+    "Presentazione di PowerPoint",
+    "Prezentácia programu PowerPoint",
+    "Apresentação do PowerPoint",
+    "PowerPoint-bildspel",
+    "Prezentace v aplikaci PowerPoint",
+    "PowerPoint 슬라이드 쇼",
+    "PowerPoint-lysbildefremvisning",
+    "PowerPoint-vetítés",
+    "PowerPoint Slayt Gösterisi",
+    "Pokaz slajdów programu PowerPoint",
+    "PowerPoint 投影片放映",
+    "Демонстрация PowerPoint",
+    "Diaporama PowerPoint",
+    "PowerPoint-diaesitys",
+    "Peragaan Slide PowerPoint",
+    "PowerPoint-diavoorstelling",
+    "การนำเสนอสไลด์ PowerPoint",
+    "Apresentação de slides do PowerPoint",
+    "הצגת שקופיות של PowerPoint",
+    "عرض شرائح في PowerPoint"};
+
+class FullScreenMacApplicationHandler : public FullScreenApplicationHandler {
+ public:
+  using TitlePredicate =
+      std::function<bool(absl::string_view, absl::string_view)>;
+
+  FullScreenMacApplicationHandler(DesktopCapturer::SourceId sourceId,
+                                  TitlePredicate title_predicate,
+                                  bool ignore_original_window)
+      : FullScreenApplicationHandler(sourceId),
+        title_predicate_(title_predicate),
+        owner_pid_(GetWindowOwnerPid(sourceId)),
+        ignore_original_window_(ignore_original_window) {}
+
+ protected:
+  using CachePredicate =
+      rtc::FunctionView<bool(const DesktopCapturer::Source&)>;
+
+  void InvalidateCacheIfNeeded(const DesktopCapturer::SourceList& source_list,
+                               int64_t timestamp,
+                               CachePredicate predicate) const {
+    if (timestamp != cache_timestamp_) {
+      cache_sources_.clear();
+      std::copy_if(source_list.begin(), source_list.end(),
+                   std::back_inserter(cache_sources_), predicate);
+      cache_timestamp_ = timestamp;
+    }
+  }
+
+  WindowId FindFullScreenWindowWithSamePid(
+      const DesktopCapturer::SourceList& source_list,
+      int64_t timestamp) const {
+    InvalidateCacheIfNeeded(source_list, timestamp,
+                            [&](const DesktopCapturer::Source& src) {
+                              return src.id != GetSourceId() &&
+                                     GetWindowOwnerPid(src.id) == owner_pid_;
+                            });
+    if (cache_sources_.empty())
+      return kCGNullWindowID;
+
+    const auto original_window = GetSourceId();
+    const std::string title = GetWindowTitle(original_window);
+
+    // We can ignore any windows with empty titles cause regardless type of
+    // application it's impossible to verify that full screen window and
+    // original window are related to the same document.
+    if (title.empty())
+      return kCGNullWindowID;
+
+    MacDesktopConfiguration desktop_config =
+        MacDesktopConfiguration::GetCurrent(
+            MacDesktopConfiguration::TopLeftOrigin);
+
+    const auto it = std::find_if(
+        cache_sources_.begin(), cache_sources_.end(),
+        [&](const DesktopCapturer::Source& src) {
+          const std::string window_title = GetWindowTitle(src.id);
+
+          if (window_title.empty())
+            return false;
+
+          if (title_predicate_ && !title_predicate_(title, window_title))
+            return false;
+
+          return IsWindowFullScreen(desktop_config, src.id);
+        });
+
+    return it != cache_sources_.end() ? it->id : 0;
+  }
+
+  DesktopCapturer::SourceId FindFullScreenWindow(
+      const DesktopCapturer::SourceList& source_list,
+      int64_t timestamp) const override {
+    return !ignore_original_window_ && IsWindowOnScreen(GetSourceId())
+               ? 0
+               : FindFullScreenWindowWithSamePid(source_list, timestamp);
+  }
+
+ protected:
+  const TitlePredicate title_predicate_;
+  const int owner_pid_;
+  const bool ignore_original_window_;
+  mutable int64_t cache_timestamp_ = 0;
+  mutable DesktopCapturer::SourceList cache_sources_;
+};
+
+bool equal_title_predicate(absl::string_view original_title,
+                           absl::string_view title) {
+  return original_title == title;
+}
+
+bool slide_show_title_predicate(absl::string_view original_title,
+                                absl::string_view title) {
+  if (title.find(original_title) == absl::string_view::npos)
+    return false;
+
+  for (const char* pp_slide_title : kPowerPointSlideShowTitles) {
+    if (absl::StartsWith(title, pp_slide_title))
+      return true;
+  }
+  return false;
+}
+
+class OpenOfficeApplicationHandler : public FullScreenMacApplicationHandler {
+ public:
+  OpenOfficeApplicationHandler(DesktopCapturer::SourceId sourceId)
+      : FullScreenMacApplicationHandler(sourceId, nullptr, false) {}
+
+  DesktopCapturer::SourceId FindFullScreenWindow(
+      const DesktopCapturer::SourceList& source_list,
+      int64_t timestamp) const override {
+    InvalidateCacheIfNeeded(source_list, timestamp,
+                            [&](const DesktopCapturer::Source& src) {
+                              return GetWindowOwnerPid(src.id) == owner_pid_;
+                            });
+
+    const auto original_window = GetSourceId();
+    const std::string original_title = GetWindowTitle(original_window);
+
+    // Check if we have only one document window, otherwise it's not possible
+    // to securely match a document window and a slide show window which has
+    // empty title.
+    if (std::any_of(cache_sources_.begin(), cache_sources_.end(),
+                    [&original_title](const DesktopCapturer::Source& src) {
+                      return src.title.length() && src.title != original_title;
+                    })) {
+      return kCGNullWindowID;
+    }
+
+    MacDesktopConfiguration desktop_config =
+        MacDesktopConfiguration::GetCurrent(
+            MacDesktopConfiguration::TopLeftOrigin);
+
+    // Looking for slide show window,
+    // it must be a full screen window with empty title
+    const auto slide_show_window = std::find_if(
+        cache_sources_.begin(), cache_sources_.end(), [&](const auto& src) {
+          return src.title.empty() &&
+                 IsWindowFullScreen(desktop_config, src.id);
+        });
+
+    if (slide_show_window == cache_sources_.end()) {
+      return kCGNullWindowID;
+    }
+
+    return slide_show_window->id;
+  }
+};
+
+}  // namespace
+
+std::unique_ptr<FullScreenApplicationHandler>
+CreateFullScreenMacApplicationHandler(DesktopCapturer::SourceId sourceId) {
+  std::unique_ptr<FullScreenApplicationHandler> result;
+  int pid = GetWindowOwnerPid(sourceId);
+  char buffer[PROC_PIDPATHINFO_MAXSIZE];
+  int path_length = proc_pidpath(pid, buffer, sizeof(buffer));
+  if (path_length > 0) {
+    const char* last_slash = strrchr(buffer, '/');
+    const std::string name{last_slash ? last_slash + 1 : buffer};
+    const std::string owner_name = GetWindowOwnerName(sourceId);
+    FullScreenMacApplicationHandler::TitlePredicate predicate = nullptr;
+    bool ignore_original_window = false;
+    if (name.find("Google Chrome") == 0 || name == "Chromium") {
+      predicate = equal_title_predicate;
+    } else if (name == "Microsoft PowerPoint") {
+      predicate = slide_show_title_predicate;
+      ignore_original_window = true;
+    } else if (name == "Keynote") {
+      predicate = equal_title_predicate;
+    } else if (owner_name == "OpenOffice") {
+      return std::make_unique<OpenOfficeApplicationHandler>(sourceId);
+    }
+
+    if (predicate) {
+      result.reset(new FullScreenMacApplicationHandler(sourceId, predicate,
+                                                       ignore_original_window));
+    }
+  }
+
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/full_screen_mac_application_handler.h b/third_party/libwebrtc/modules/desktop_capture/mac/full_screen_mac_application_handler.h
new file mode 100644
index 0000000000..f795a22030
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/full_screen_mac_application_handler.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MAC_FULL_SCREEN_MAC_APPLICATION_HANDLER_H_
+#define MODULES_DESKTOP_CAPTURE_MAC_FULL_SCREEN_MAC_APPLICATION_HANDLER_H_
+
+#include <memory>
+#include "modules/desktop_capture/full_screen_application_handler.h"
+
+namespace webrtc {
+
+std::unique_ptr<FullScreenApplicationHandler>
+CreateFullScreenMacApplicationHandler(DesktopCapturer::SourceId sourceId);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MAC_FULL_SCREEN_MAC_APPLICATION_HANDLER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.h b/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.h
new file mode 100644
index 0000000000..7be05cc639
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.h
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MAC_SCREEN_CAPTURER_MAC_H_
+#define MODULES_DESKTOP_CAPTURE_MAC_SCREEN_CAPTURER_MAC_H_
+
+#include <CoreGraphics/CoreGraphics.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
+#include "modules/desktop_capture/mac/desktop_frame_provider.h"
+#include "modules/desktop_capture/screen_capture_frame_queue.h"
+#include "modules/desktop_capture/screen_capturer_helper.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+
+namespace webrtc {
+
+class DisplayStreamManager;
+
+// A class to perform video frame capturing for mac.
+class ScreenCapturerMac final : public DesktopCapturer {
+ public:
+  ScreenCapturerMac(
+      rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
+      bool detect_updated_region,
+      bool allow_iosurface);
+  ~ScreenCapturerMac() override;
+
+  ScreenCapturerMac(const ScreenCapturerMac&) = delete;
+  ScreenCapturerMac& operator=(const ScreenCapturerMac&) = delete;
+
+  // TODO(julien.isorce): Remove Init() or make it private.
+  bool Init();
+
+  // DesktopCapturer interface.
+  void Start(Callback* callback) override;
+  void CaptureFrame() override;
+  void SetExcludedWindow(WindowId window) override;
+  bool GetSourceList(SourceList* screens) override;
+  bool SelectSource(SourceId id) override;
+
+ private:
+  // Returns false if the selected screen is no longer valid.
+  bool CgBlit(const DesktopFrame& frame, const DesktopRegion& region);
+
+  // Called when the screen configuration is changed.
+  void ScreenConfigurationChanged();
+
+  bool RegisterRefreshAndMoveHandlers();
+  void UnregisterRefreshAndMoveHandlers();
+
+  void ScreenRefresh(CGDirectDisplayID display_id,
+                     CGRectCount count,
+                     const CGRect* rect_array,
+                     DesktopVector display_origin,
+                     IOSurfaceRef io_surface);
+  void ReleaseBuffers();
+
+  std::unique_ptr<DesktopFrame> CreateFrame();
+
+  const bool detect_updated_region_;
+
+  Callback* callback_ = nullptr;
+
+  // Queue of the frames buffers.
+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
+
+  // Current display configuration.
+  MacDesktopConfiguration desktop_config_;
+
+  // Currently selected display, or 0 if the full desktop is selected. On OS X
+  // 10.6 and before, this is always 0.
+  CGDirectDisplayID current_display_ = 0;
+
+  // The physical pixel bounds of the current screen.
+  DesktopRect screen_pixel_bounds_;
+
+  // The dip to physical pixel scale of the current screen.
+  float dip_to_pixel_scale_ = 1.0f;
+
+  // A thread-safe list of invalid rectangles, and the size of the most
+  // recently captured screen.
+  ScreenCapturerHelper helper_;
+
+  // Contains an invalid region from the previous capture.
+  DesktopRegion last_invalid_region_;
+
+  // Monitoring display reconfiguration.
+  rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor_;
+
+  CGWindowID excluded_window_ = 0;
+
+  // List of streams, one per screen.
+  std::vector<CGDisplayStreamRef> display_streams_;
+
+  // Container holding latest state of the snapshot per displays.
+  DesktopFrameProvider desktop_frame_provider_;
+
+  // Start, CaptureFrame and destructor have to called in the same thread.
+  SequenceChecker thread_checker_;
+
+  // Used to force CaptureFrame to update it's screen configuration
+  // and reregister event handlers. This ensure that this
+  // occurs on the ScreenCapture thread. Read and written from
+  // both the VideoCapture thread and ScreenCapture thread.
+  // Protected by desktop_config_monitor_.
+  bool update_screen_configuration_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MAC_SCREEN_CAPTURER_MAC_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.mm b/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.mm
new file mode 100644
index 0000000000..76fec13a39
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/screen_capturer_mac.mm
@@ -0,0 +1,633 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <utility>
+
+#include "modules/desktop_capture/mac/screen_capturer_mac.h"
+
+#include "modules/desktop_capture/mac/desktop_frame_provider.h"
+#include "modules/desktop_capture/mac/window_list_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "sdk/objc/helpers/scoped_cftyperef.h"
+
+// All these symbols have incorrect availability annotations in the 13.3 SDK.
+// These have the correct annotation. See https://crbug.com/1431897.
+// TODO(thakis): Remove this once FB12109479 is fixed and we updated to an SDK
+// with the fix.
+
+static CGDisplayStreamRef __nullable
+    wrapCGDisplayStreamCreate(CGDirectDisplayID display,
+                              size_t outputWidth,
+                              size_t outputHeight,
+                              int32_t pixelFormat,
+                              CFDictionaryRef __nullable properties,
+                              CGDisplayStreamFrameAvailableHandler __nullable handler)
+        CG_AVAILABLE_BUT_DEPRECATED(
+            10.8,
+            14.0,
+            "Please use ScreenCaptureKit API's initWithFilter:configuration:delegate: instead") {
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wunguarded-availability-new"
+  return CGDisplayStreamCreate(
+      display, outputWidth, outputHeight, pixelFormat, properties, handler);
+#pragma clang diagnostic pop
+}
+
+static CFRunLoopSourceRef __nullable
+    wrapCGDisplayStreamGetRunLoopSource(CGDisplayStreamRef cg_nullable displayStream)
+        CG_AVAILABLE_BUT_DEPRECATED(10.8,
+                                    14.0,
+                                    "There is no direct replacement for this function. Please use "
+                                    "ScreenCaptureKit API's SCStream to replace CGDisplayStream") {
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wunguarded-availability-new"
+  return CGDisplayStreamGetRunLoopSource(displayStream);
+#pragma clang diagnostic pop
+}
+
+static CGError wrapCGDisplayStreamStart(CGDisplayStreamRef cg_nullable displayStream)
+    CG_AVAILABLE_BUT_DEPRECATED(10.8,
+                                14.0,
+                                "Please use ScreenCaptureKit API's "
+                                "startCaptureWithCompletionHandler: to start a stream instead") {
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wunguarded-availability-new"
+  return CGDisplayStreamStart(displayStream);
+#pragma clang diagnostic pop
+}
+
+static CGError wrapCGDisplayStreamStop(CGDisplayStreamRef cg_nullable displayStream)
+    CG_AVAILABLE_BUT_DEPRECATED(10.8,
+                                14.0,
+                                "Please use ScreenCaptureKit API's "
+                                "stopCaptureWithCompletionHandler: to stop a stream instead") {
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wunguarded-availability-new"
+  return CGDisplayStreamStop(displayStream);
+#pragma clang diagnostic pop
+}
+
+static CFStringRef wrapkCGDisplayStreamShowCursor() CG_AVAILABLE_BUT_DEPRECATED(
+    10.8,
+    14.0,
+    "Please use ScreenCaptureKit API's SCStreamConfiguration showsCursor property instead") {
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wunguarded-availability-new"
+  return kCGDisplayStreamShowCursor;
+#pragma clang diagnostic pop
+}
+
+static const CGRect* __nullable
+    wrapCGDisplayStreamUpdateGetRects(CGDisplayStreamUpdateRef __nullable updateRef,
+                                      CGDisplayStreamUpdateRectType rectType,
+                                      size_t* rectCount)
+        CG_AVAILABLE_BUT_DEPRECATED(10.8,
+                                    14.0,
+                                    "Please use ScreenCaptureKit API's SCStreamFrameInfo with "
+                                    "SCStreamFrameInfoContentRect instead") {
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wunguarded-availability-new"
+  return CGDisplayStreamUpdateGetRects(updateRef, rectType, rectCount);
+#pragma clang diagnostic pop
+}
+
+namespace webrtc {
+
+namespace {
+
+// Scales all coordinates of a rect by a specified factor.
+DesktopRect ScaleAndRoundCGRect(const CGRect& rect, float scale) {
+  return DesktopRect::MakeLTRB(static_cast<int>(floor(rect.origin.x * scale)),
+                               static_cast<int>(floor(rect.origin.y * scale)),
+                               static_cast<int>(ceil((rect.origin.x + rect.size.width) * scale)),
+                               static_cast<int>(ceil((rect.origin.y + rect.size.height) * scale)));
+}
+
+// Copy pixels in the `rect` from `src_place` to `dest_plane`. `rect` should be
+// relative to the origin of `src_plane` and `dest_plane`.
+void CopyRect(const uint8_t* src_plane,
+              int src_plane_stride,
+              uint8_t* dest_plane,
+              int dest_plane_stride,
+              int bytes_per_pixel,
+              const DesktopRect& rect) {
+  // Get the address of the starting point.
+  const int src_y_offset = src_plane_stride * rect.top();
+  const int dest_y_offset = dest_plane_stride * rect.top();
+  const int x_offset = bytes_per_pixel * rect.left();
+  src_plane += src_y_offset + x_offset;
+  dest_plane += dest_y_offset + x_offset;
+
+  // Copy pixels in the rectangle line by line.
+  const int bytes_per_line = bytes_per_pixel * rect.width();
+  const int height = rect.height();
+  for (int i = 0; i < height; ++i) {
+    memcpy(dest_plane, src_plane, bytes_per_line);
+    src_plane += src_plane_stride;
+    dest_plane += dest_plane_stride;
+  }
+}
+
+// Returns an array of CGWindowID for all the on-screen windows except
+// `window_to_exclude`, or NULL if the window is not found or it fails. The
+// caller should release the returned CFArrayRef.
+CFArrayRef CreateWindowListWithExclusion(CGWindowID window_to_exclude) {
+  if (!window_to_exclude) return nullptr;
+
+  CFArrayRef all_windows =
+      CGWindowListCopyWindowInfo(kCGWindowListOptionOnScreenOnly, kCGNullWindowID);
+  if (!all_windows) return nullptr;
+
+  CFMutableArrayRef returned_array =
+      CFArrayCreateMutable(nullptr, CFArrayGetCount(all_windows), nullptr);
+
+  bool found = false;
+  for (CFIndex i = 0; i < CFArrayGetCount(all_windows); ++i) {
+    CFDictionaryRef window =
+        reinterpret_cast<CFDictionaryRef>(CFArrayGetValueAtIndex(all_windows, i));
+
+    CGWindowID id = GetWindowId(window);
+    if (id == window_to_exclude) {
+      found = true;
+      continue;
+    }
+    CFArrayAppendValue(returned_array, reinterpret_cast<void*>(id));
+  }
+  CFRelease(all_windows);
+
+  if (!found) {
+    CFRelease(returned_array);
+    returned_array = nullptr;
+  }
+  return returned_array;
+}
+
+// Returns the bounds of `window` in physical pixels, enlarged by a small amount
+// on four edges to take account of the border/shadow effects.
+DesktopRect GetExcludedWindowPixelBounds(CGWindowID window, float dip_to_pixel_scale) {
+  // The amount of pixels to add to the actual window bounds to take into
+  // account of the border/shadow effects.
+  static const int kBorderEffectSize = 20;
+  CGRect rect;
+  CGWindowID ids[1];
+  ids[0] = window;
+
+  CFArrayRef window_id_array =
+      CFArrayCreate(nullptr, reinterpret_cast<const void**>(&ids), 1, nullptr);
+  CFArrayRef window_array = CGWindowListCreateDescriptionFromArray(window_id_array);
+
+  if (CFArrayGetCount(window_array) > 0) {
+    CFDictionaryRef win =
+        reinterpret_cast<CFDictionaryRef>(CFArrayGetValueAtIndex(window_array, 0));
+    CFDictionaryRef bounds_ref =
+        reinterpret_cast<CFDictionaryRef>(CFDictionaryGetValue(win, kCGWindowBounds));
+    CGRectMakeWithDictionaryRepresentation(bounds_ref, &rect);
+  }
+
+  CFRelease(window_id_array);
+  CFRelease(window_array);
+
+  rect.origin.x -= kBorderEffectSize;
+  rect.origin.y -= kBorderEffectSize;
+  rect.size.width += kBorderEffectSize * 2;
+  rect.size.height += kBorderEffectSize * 2;
+  // `rect` is in DIP, so convert to physical pixels.
+  return ScaleAndRoundCGRect(rect, dip_to_pixel_scale);
+}
+
+// Create an image of the given region using the given `window_list`.
+// `pixel_bounds` should be in the primary display's coordinate in physical
+// pixels.
+rtc::ScopedCFTypeRef<CGImageRef> CreateExcludedWindowRegionImage(const DesktopRect& pixel_bounds,
+                                                                 float dip_to_pixel_scale,
+                                                                 CFArrayRef window_list) {
+  CGRect window_bounds;
+  // The origin is in DIP while the size is in physical pixels. That's what
+  // CGWindowListCreateImageFromArray expects.
+  window_bounds.origin.x = pixel_bounds.left() / dip_to_pixel_scale;
+  window_bounds.origin.y = pixel_bounds.top() / dip_to_pixel_scale;
+  window_bounds.size.width = pixel_bounds.width();
+  window_bounds.size.height = pixel_bounds.height();
+
+  return rtc::ScopedCFTypeRef<CGImageRef>(
+      CGWindowListCreateImageFromArray(window_bounds, window_list, kCGWindowImageDefault));
+}
+
+}  // namespace
+
+ScreenCapturerMac::ScreenCapturerMac(
+    rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
+    bool detect_updated_region,
+    bool allow_iosurface)
+    : detect_updated_region_(detect_updated_region),
+      desktop_config_monitor_(desktop_config_monitor),
+      desktop_frame_provider_(allow_iosurface) {
+  RTC_LOG(LS_INFO) << "Allow IOSurface: " << allow_iosurface;
+  thread_checker_.Detach();
+}
+
+ScreenCapturerMac::~ScreenCapturerMac() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  ReleaseBuffers();
+  UnregisterRefreshAndMoveHandlers();
+}
+
+bool ScreenCapturerMac::Init() {
+  TRACE_EVENT0("webrtc", "ScreenCapturerMac::Init");
+  desktop_config_ = desktop_config_monitor_->desktop_configuration();
+  return true;
+}
+
+void ScreenCapturerMac::ReleaseBuffers() {
+  // The buffers might be in use by the encoder, so don't delete them here.
+  // Instead, mark them as "needs update"; next time the buffers are used by
+  // the capturer, they will be recreated if necessary.
+  queue_.Reset();
+}
+
+void ScreenCapturerMac::Start(Callback* callback) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+  TRACE_EVENT_INSTANT1(
+      "webrtc", "ScreenCapturermac::Start", "target display id ", current_display_);
+
+  callback_ = callback;
+  update_screen_configuration_ = false;
+  // Start and operate CGDisplayStream handler all from capture thread.
+  if (!RegisterRefreshAndMoveHandlers()) {
+    RTC_LOG(LS_ERROR) << "Failed to register refresh and move handlers.";
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+  ScreenConfigurationChanged();
+}
+
+void ScreenCapturerMac::CaptureFrame() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  TRACE_EVENT0("webrtc", "creenCapturerMac::CaptureFrame");
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+  queue_.MoveToNextFrame();
+  if (queue_.current_frame() && queue_.current_frame()->IsShared()) {
+    RTC_DLOG(LS_WARNING) << "Overwriting frame that is still shared.";
+  }
+
+  MacDesktopConfiguration new_config = desktop_config_monitor_->desktop_configuration();
+  if (update_screen_configuration_ || !desktop_config_.Equals(new_config)) {
+    update_screen_configuration_ = false;
+    desktop_config_ = new_config;
+    // If the display configuraiton has changed then refresh capturer data
+    // structures. Occasionally, the refresh and move handlers are lost when
+    // the screen mode changes, so re-register them here.
+    UnregisterRefreshAndMoveHandlers();
+    if (!RegisterRefreshAndMoveHandlers()) {
+      RTC_LOG(LS_ERROR) << "Failed to register refresh and move handlers.";
+      callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+      return;
+    }
+    ScreenConfigurationChanged();
+  }
+
+  // When screen is zoomed in/out, OSX only updates the part of Rects currently
+  // displayed on screen, with relative location to current top-left on screen.
+  // This will cause problems when we copy the dirty regions to the captured
+  // image. So we invalidate the whole screen to copy all the screen contents.
+  // With CGI method, the zooming will be ignored and the whole screen contents
+  // will be captured as before.
+  // With IOSurface method, the zoomed screen contents will be captured.
+  if (UAZoomEnabled()) {
+    helper_.InvalidateScreen(screen_pixel_bounds_.size());
+  }
+
+  DesktopRegion region;
+  helper_.TakeInvalidRegion(&region);
+
+  // If the current buffer is from an older generation then allocate a new one.
+  // Note that we can't reallocate other buffers at this point, since the caller
+  // may still be reading from them.
+  if (!queue_.current_frame()) queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(CreateFrame()));
+
+  DesktopFrame* current_frame = queue_.current_frame();
+
+  if (!CgBlit(*current_frame, region)) {
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+  std::unique_ptr<DesktopFrame> new_frame = queue_.current_frame()->Share();
+  if (detect_updated_region_) {
+    *new_frame->mutable_updated_region() = region;
+  } else {
+    new_frame->mutable_updated_region()->AddRect(DesktopRect::MakeSize(new_frame->size()));
+  }
+
+  if (current_display_) {
+    const MacDisplayConfiguration* config =
+        desktop_config_.FindDisplayConfigurationById(current_display_);
+    if (config) {
+      new_frame->set_top_left(
+          config->bounds.top_left().subtract(desktop_config_.bounds.top_left()));
+    }
+  }
+
+  helper_.set_size_most_recent(new_frame->size());
+
+  new_frame->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
+                                 rtc::kNumNanosecsPerMillisec);
+  callback_->OnCaptureResult(Result::SUCCESS, std::move(new_frame));
+}
+
+void ScreenCapturerMac::SetExcludedWindow(WindowId window) {
+  excluded_window_ = window;
+}
+
+bool ScreenCapturerMac::GetSourceList(SourceList* screens) {
+  RTC_DCHECK(screens->size() == 0);
+
+  for (MacDisplayConfigurations::iterator it = desktop_config_.displays.begin();
+       it != desktop_config_.displays.end();
+       ++it) {
+    Source value = {it->id, 0, std::string()};
+    screens->push_back(value);
+  }
+  return true;
+}
+
+bool ScreenCapturerMac::SelectSource(SourceId id) {
+  if (id == kFullDesktopScreenId) {
+    current_display_ = 0;
+  } else {
+    const MacDisplayConfiguration* config =
+        desktop_config_.FindDisplayConfigurationById(static_cast<CGDirectDisplayID>(id));
+    if (!config) return false;
+    current_display_ = config->id;
+  }
+
+  ScreenConfigurationChanged();
+  return true;
+}
+
+bool ScreenCapturerMac::CgBlit(const DesktopFrame& frame, const DesktopRegion& region) {
+  // If not all screen region is dirty, copy the entire contents of the previous capture buffer,
+  // to capture over.
+  if (queue_.previous_frame() && !region.Equals(DesktopRegion(screen_pixel_bounds_))) {
+    memcpy(frame.data(), queue_.previous_frame()->data(), frame.stride() * frame.size().height());
+  }
+
+  MacDisplayConfigurations displays_to_capture;
+  if (current_display_) {
+    // Capturing a single screen. Note that the screen id may change when
+    // screens are added or removed.
+    const MacDisplayConfiguration* config =
+        desktop_config_.FindDisplayConfigurationById(current_display_);
+    if (config) {
+      displays_to_capture.push_back(*config);
+    } else {
+      RTC_LOG(LS_ERROR) << "The selected screen cannot be found for capturing.";
+      return false;
+    }
+  } else {
+    // Capturing the whole desktop.
+    displays_to_capture = desktop_config_.displays;
+  }
+
+  // Create the window list once for all displays.
+  CFArrayRef window_list = CreateWindowListWithExclusion(excluded_window_);
+
+  for (size_t i = 0; i < displays_to_capture.size(); ++i) {
+    const MacDisplayConfiguration& display_config = displays_to_capture[i];
+
+    // Capturing mixed-DPI on one surface is hard, so we only return displays
+    // that match the "primary" display's DPI. The primary display is always
+    // the first in the list.
+    if (i > 0 && display_config.dip_to_pixel_scale != displays_to_capture[0].dip_to_pixel_scale) {
+      continue;
+    }
+    // Determine the display's position relative to the desktop, in pixels.
+    DesktopRect display_bounds = display_config.pixel_bounds;
+    display_bounds.Translate(-screen_pixel_bounds_.left(), -screen_pixel_bounds_.top());
+
+    // Determine which parts of the blit region, if any, lay within the monitor.
+    DesktopRegion copy_region = region;
+    copy_region.IntersectWith(display_bounds);
+    if (copy_region.is_empty()) continue;
+
+    // Translate the region to be copied into display-relative coordinates.
+    copy_region.Translate(-display_bounds.left(), -display_bounds.top());
+
+    DesktopRect excluded_window_bounds;
+    rtc::ScopedCFTypeRef<CGImageRef> excluded_image;
+    if (excluded_window_ && window_list) {
+      // Get the region of the excluded window relative the primary display.
+      excluded_window_bounds =
+          GetExcludedWindowPixelBounds(excluded_window_, display_config.dip_to_pixel_scale);
+      excluded_window_bounds.IntersectWith(display_config.pixel_bounds);
+
+      // Create the image under the excluded window first, because it's faster
+      // than captuing the whole display.
+      if (!excluded_window_bounds.is_empty()) {
+        excluded_image = CreateExcludedWindowRegionImage(
+            excluded_window_bounds, display_config.dip_to_pixel_scale, window_list);
+      }
+    }
+
+    std::unique_ptr<DesktopFrame> frame_source =
+        desktop_frame_provider_.TakeLatestFrameForDisplay(display_config.id);
+    if (!frame_source) {
+      continue;
+    }
+
+    const uint8_t* display_base_address = frame_source->data();
+    int src_bytes_per_row = frame_source->stride();
+    RTC_DCHECK(display_base_address);
+
+    // `frame_source` size may be different from display_bounds in case the screen was
+    // resized recently.
+    copy_region.IntersectWith(frame_source->rect());
+
+    // Copy the dirty region from the display buffer into our desktop buffer.
+    uint8_t* out_ptr = frame.GetFrameDataAtPos(display_bounds.top_left());
+    for (DesktopRegion::Iterator it(copy_region); !it.IsAtEnd(); it.Advance()) {
+      CopyRect(display_base_address,
+               src_bytes_per_row,
+               out_ptr,
+               frame.stride(),
+               DesktopFrame::kBytesPerPixel,
+               it.rect());
+    }
+
+    if (excluded_image) {
+      CGDataProviderRef provider = CGImageGetDataProvider(excluded_image.get());
+      rtc::ScopedCFTypeRef<CFDataRef> excluded_image_data(CGDataProviderCopyData(provider));
+      RTC_DCHECK(excluded_image_data);
+      display_base_address = CFDataGetBytePtr(excluded_image_data.get());
+      src_bytes_per_row = CGImageGetBytesPerRow(excluded_image.get());
+
+      // Translate the bounds relative to the desktop, because `frame` data
+      // starts from the desktop top-left corner.
+      DesktopRect window_bounds_relative_to_desktop(excluded_window_bounds);
+      window_bounds_relative_to_desktop.Translate(-screen_pixel_bounds_.left(),
+                                                  -screen_pixel_bounds_.top());
+
+      DesktopRect rect_to_copy = DesktopRect::MakeSize(excluded_window_bounds.size());
+      rect_to_copy.IntersectWith(DesktopRect::MakeWH(CGImageGetWidth(excluded_image.get()),
+                                                     CGImageGetHeight(excluded_image.get())));
+
+      if (CGImageGetBitsPerPixel(excluded_image.get()) / 8 == DesktopFrame::kBytesPerPixel) {
+        CopyRect(display_base_address,
+                 src_bytes_per_row,
+                 frame.GetFrameDataAtPos(window_bounds_relative_to_desktop.top_left()),
+                 frame.stride(),
+                 DesktopFrame::kBytesPerPixel,
+                 rect_to_copy);
+      }
+    }
+  }
+  if (window_list) CFRelease(window_list);
+  return true;
+}
+
+void ScreenCapturerMac::ScreenConfigurationChanged() {
+  if (current_display_) {
+    const MacDisplayConfiguration* config =
+        desktop_config_.FindDisplayConfigurationById(current_display_);
+    screen_pixel_bounds_ = config ? config->pixel_bounds : DesktopRect();
+    dip_to_pixel_scale_ = config ? config->dip_to_pixel_scale : 1.0f;
+  } else {
+    screen_pixel_bounds_ = desktop_config_.pixel_bounds;
+    dip_to_pixel_scale_ = desktop_config_.dip_to_pixel_scale;
+  }
+
+  // Release existing buffers, which will be of the wrong size.
+  ReleaseBuffers();
+
+  // Clear the dirty region, in case the display is down-sizing.
+  helper_.ClearInvalidRegion();
+
+  // Re-mark the entire desktop as dirty.
+  helper_.InvalidateScreen(screen_pixel_bounds_.size());
+
+  // Make sure the frame buffers will be reallocated.
+  queue_.Reset();
+}
+
+bool ScreenCapturerMac::RegisterRefreshAndMoveHandlers() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  desktop_config_ = desktop_config_monitor_->desktop_configuration();
+  for (const auto& config : desktop_config_.displays) {
+    size_t pixel_width = config.pixel_bounds.width();
+    size_t pixel_height = config.pixel_bounds.height();
+    if (pixel_width == 0 || pixel_height == 0) continue;
+    CGDirectDisplayID display_id = config.id;
+    DesktopVector display_origin = config.pixel_bounds.top_left();
+
+    CGDisplayStreamFrameAvailableHandler handler = ^(CGDisplayStreamFrameStatus status,
+                                                     uint64_t display_time,
+                                                     IOSurfaceRef frame_surface,
+                                                     CGDisplayStreamUpdateRef updateRef) {
+      RTC_DCHECK(thread_checker_.IsCurrent());
+      if (status == kCGDisplayStreamFrameStatusStopped) return;
+
+      // Only pay attention to frame updates.
+      if (status != kCGDisplayStreamFrameStatusFrameComplete) return;
+
+      size_t count = 0;
+      const CGRect* rects =
+          wrapCGDisplayStreamUpdateGetRects(updateRef, kCGDisplayStreamUpdateDirtyRects, &count);
+      if (count != 0) {
+        // According to CGDisplayStream.h, it's safe to call
+        // CGDisplayStreamStop() from within the callback.
+        ScreenRefresh(display_id, count, rects, display_origin, frame_surface);
+      }
+    };
+
+    rtc::ScopedCFTypeRef<CFDictionaryRef> properties_dict(
+        CFDictionaryCreate(kCFAllocatorDefault,
+                           (const void*[]){wrapkCGDisplayStreamShowCursor()},
+                           (const void*[]){kCFBooleanFalse},
+                           1,
+                           &kCFTypeDictionaryKeyCallBacks,
+                           &kCFTypeDictionaryValueCallBacks));
+
+    CGDisplayStreamRef display_stream = wrapCGDisplayStreamCreate(
+        display_id, pixel_width, pixel_height, 'BGRA', properties_dict.get(), handler);
+
+    if (display_stream) {
+      CGError error = wrapCGDisplayStreamStart(display_stream);
+      if (error != kCGErrorSuccess) return false;
+
+      CFRunLoopSourceRef source = wrapCGDisplayStreamGetRunLoopSource(display_stream);
+      CFRunLoopAddSource(CFRunLoopGetCurrent(), source, kCFRunLoopCommonModes);
+      display_streams_.push_back(display_stream);
+    }
+  }
+
+  return true;
+}
+
+void ScreenCapturerMac::UnregisterRefreshAndMoveHandlers() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+
+  for (CGDisplayStreamRef stream : display_streams_) {
+    CFRunLoopSourceRef source = wrapCGDisplayStreamGetRunLoopSource(stream);
+    CFRunLoopRemoveSource(CFRunLoopGetCurrent(), source, kCFRunLoopCommonModes);
+    wrapCGDisplayStreamStop(stream);
+    CFRelease(stream);
+  }
+  display_streams_.clear();
+
+  // Release obsolete io surfaces.
+  desktop_frame_provider_.Release();
+}
+
+void ScreenCapturerMac::ScreenRefresh(CGDirectDisplayID display_id,
+                                      CGRectCount count,
+                                      const CGRect* rect_array,
+                                      DesktopVector display_origin,
+                                      IOSurfaceRef io_surface) {
+  if (screen_pixel_bounds_.is_empty()) ScreenConfigurationChanged();
+
+  // The refresh rects are in display coordinates. We want to translate to
+  // framebuffer coordinates. If a specific display is being captured, then no
+  // change is necessary. If all displays are being captured, then we want to
+  // translate by the origin of the display.
+  DesktopVector translate_vector;
+  if (!current_display_) translate_vector = display_origin;
+
+  DesktopRegion region;
+  for (CGRectCount i = 0; i < count; ++i) {
+    // All rects are already in physical pixel coordinates.
+    DesktopRect rect = DesktopRect::MakeXYWH(rect_array[i].origin.x,
+                                             rect_array[i].origin.y,
+                                             rect_array[i].size.width,
+                                             rect_array[i].size.height);
+
+    rect.Translate(translate_vector);
+
+    region.AddRect(rect);
+  }
+  // Always having the latest iosurface before invalidating a region.
+  // See https://bugs.chromium.org/p/webrtc/issues/detail?id=8652 for details.
+  desktop_frame_provider_.InvalidateIOSurface(
+      display_id, rtc::ScopedCFTypeRef<IOSurfaceRef>(io_surface, rtc::RetainPolicy::RETAIN));
+  helper_.InvalidateRegion(region);
+}
+
+std::unique_ptr<DesktopFrame> ScreenCapturerMac::CreateFrame() {
+  std::unique_ptr<DesktopFrame> frame(new BasicDesktopFrame(screen_pixel_bounds_.size()));
+  frame->set_dpi(
+      DesktopVector(kStandardDPI * dip_to_pixel_scale_, kStandardDPI * dip_to_pixel_scale_));
+  return frame;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/window_list_utils.cc b/third_party/libwebrtc/modules/desktop_capture/mac/window_list_utils.cc
new file mode 100644
index 0000000000..989ec7ea54
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/window_list_utils.cc
@@ -0,0 +1,430 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mac/window_list_utils.h"
+
+#include <ApplicationServices/ApplicationServices.h>
+
+#include <algorithm>
+#include <cmath>
+#include <iterator>
+#include <limits>
+#include <list>
+#include <map>
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+static_assert(static_cast<webrtc::WindowId>(kCGNullWindowID) ==
+                  webrtc::kNullWindowId,
+              "kNullWindowId needs to equal to kCGNullWindowID.");
+
+namespace webrtc {
+
+namespace {
+
+// WindowName of the status indicator dot shown since Monterey in the taskbar.
+// Testing on 12.2.1 shows this is independent of system language setting.
+const CFStringRef kStatusIndicator = CFSTR("StatusIndicator");
+const CFStringRef kStatusIndicatorOwnerName = CFSTR("Window Server");
+
+bool ToUtf8(const CFStringRef str16, std::string* str8) {
+  size_t maxlen = CFStringGetMaximumSizeForEncoding(CFStringGetLength(str16),
+                                                    kCFStringEncodingUTF8) +
+                  1;
+  std::unique_ptr<char[]> buffer(new char[maxlen]);
+  if (!buffer ||
+      !CFStringGetCString(str16, buffer.get(), maxlen, kCFStringEncodingUTF8)) {
+    return false;
+  }
+  str8->assign(buffer.get());
+  return true;
+}
+
+// Get CFDictionaryRef from `id` and call `on_window` against it. This function
+// returns false if native APIs fail, typically it indicates that the `id` does
+// not represent a window. `on_window` will not be called if false is returned
+// from this function.
+bool GetWindowRef(CGWindowID id,
+                  rtc::FunctionView<void(CFDictionaryRef)> on_window) {
+  RTC_DCHECK(on_window);
+
+  // TODO(zijiehe): `id` is a 32-bit integer, casting it to an array seems not
+  // safe enough. Maybe we should create a new
+  // const void* arr[] = {
+  //   reinterpret_cast<void*>(id) }
+  // };
+  CFArrayRef window_id_array =
+      CFArrayCreate(NULL, reinterpret_cast<const void**>(&id), 1, NULL);
+  CFArrayRef window_array =
+      CGWindowListCreateDescriptionFromArray(window_id_array);
+
+  bool result = false;
+  // TODO(zijiehe): CFArrayGetCount(window_array) should always return 1.
+  // Otherwise, we should treat it as failure.
+  if (window_array && CFArrayGetCount(window_array)) {
+    on_window(reinterpret_cast<CFDictionaryRef>(
+        CFArrayGetValueAtIndex(window_array, 0)));
+    result = true;
+  }
+
+  if (window_array) {
+    CFRelease(window_array);
+  }
+  CFRelease(window_id_array);
+  return result;
+}
+
+}  // namespace
+
+bool GetWindowList(rtc::FunctionView<bool(CFDictionaryRef)> on_window,
+                   bool ignore_minimized,
+                   bool only_zero_layer) {
+  RTC_DCHECK(on_window);
+
+  // Only get on screen, non-desktop windows.
+  // According to
+  // https://developer.apple.com/documentation/coregraphics/cgwindowlistoption/1454105-optiononscreenonly
+  // , when kCGWindowListOptionOnScreenOnly is used, the order of windows are in
+  // decreasing z-order.
+  CFArrayRef window_array = CGWindowListCopyWindowInfo(
+      kCGWindowListOptionOnScreenOnly | kCGWindowListExcludeDesktopElements,
+      kCGNullWindowID);
+  if (!window_array)
+    return false;
+
+  MacDesktopConfiguration desktop_config = MacDesktopConfiguration::GetCurrent(
+      MacDesktopConfiguration::TopLeftOrigin);
+
+  // Check windows to make sure they have an id, title, and use window layer
+  // other than 0.
+  CFIndex count = CFArrayGetCount(window_array);
+  for (CFIndex i = 0; i < count; i++) {
+    CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
+        CFArrayGetValueAtIndex(window_array, i));
+    if (!window) {
+      continue;
+    }
+
+    CFNumberRef window_id = reinterpret_cast<CFNumberRef>(
+        CFDictionaryGetValue(window, kCGWindowNumber));
+    if (!window_id) {
+      continue;
+    }
+
+    CFNumberRef window_layer = reinterpret_cast<CFNumberRef>(
+        CFDictionaryGetValue(window, kCGWindowLayer));
+    if (!window_layer) {
+      continue;
+    }
+
+    // Skip windows with layer!=0 (menu, dock).
+    int layer;
+    if (!CFNumberGetValue(window_layer, kCFNumberIntType, &layer)) {
+      continue;
+    }
+    if (only_zero_layer && layer != 0) {
+      continue;
+    }
+
+    // Skip windows that are minimized and not full screen.
+    if (ignore_minimized && !IsWindowOnScreen(window) &&
+        !IsWindowFullScreen(desktop_config, window)) {
+      continue;
+    }
+
+    // If window title is empty, only consider it if it is either on screen or
+    // fullscreen.
+    CFStringRef window_title = reinterpret_cast<CFStringRef>(
+        CFDictionaryGetValue(window, kCGWindowName));
+    if (!window_title && !IsWindowOnScreen(window) &&
+        !IsWindowFullScreen(desktop_config, window)) {
+      continue;
+    }
+
+    CFStringRef window_owner_name = reinterpret_cast<CFStringRef>(
+        CFDictionaryGetValue(window, kCGWindowOwnerName));
+    // Ignore the red dot status indicator shown in the stats bar. Unlike the
+    // rest of the system UI it has a window_layer of 0, so was otherwise
+    // included. See crbug.com/1297731.
+    if (window_title && CFEqual(window_title, kStatusIndicator) &&
+        window_owner_name &&
+        CFEqual(window_owner_name, kStatusIndicatorOwnerName)) {
+      continue;
+    }
+
+    if (!on_window(window)) {
+      break;
+    }
+  }
+
+  CFRelease(window_array);
+  return true;
+}
+
+bool GetWindowList(DesktopCapturer::SourceList* windows,
+                   bool ignore_minimized,
+                   bool only_zero_layer) {
+  // Use a std::list so that iterators are preversed upon insertion and
+  // deletion.
+  std::list<DesktopCapturer::Source> sources;
+  std::map<int, std::list<DesktopCapturer::Source>::const_iterator> pid_itr_map;
+  const bool ret = GetWindowList(
+      [&sources, &pid_itr_map](CFDictionaryRef window) {
+        WindowId window_id = GetWindowId(window);
+        if (window_id != kNullWindowId) {
+          const std::string title = GetWindowTitle(window);
+          const int pid = GetWindowOwnerPid(window);
+          // Check if window for the same pid have been already inserted.
+          std::map<int,
+                   std::list<DesktopCapturer::Source>::const_iterator>::iterator
+              itr = pid_itr_map.find(pid);
+
+          // Only consider empty titles if the app has no other window with a
+          // proper title.
+          if (title.empty()) {
+            std::string owner_name = GetWindowOwnerName(window);
+
+            // At this time we do not know if there will be other windows
+            // for the same pid unless they have been already inserted, hence
+            // the check in the map. Also skip the window if owner name is
+            // empty too.
+            if (!owner_name.empty() && (itr == pid_itr_map.end())) {
+              sources.push_back(DesktopCapturer::Source{window_id, pid, owner_name});
+              RTC_DCHECK(!sources.empty());
+              // Get an iterator on the last valid element in the source list.
+              std::list<DesktopCapturer::Source>::const_iterator last_source =
+                  --sources.end();
+              pid_itr_map.insert(
+                  std::pair<int,
+                            std::list<DesktopCapturer::Source>::const_iterator>(
+                      pid, last_source));
+            }
+          } else {
+            sources.push_back(DesktopCapturer::Source{window_id, pid, title});
+            // Once the window with empty title has been removed no other empty
+            // windows are allowed for the same pid.
+            if (itr != pid_itr_map.end() && (itr->second != sources.end())) {
+              sources.erase(itr->second);
+              // sdt::list::end() never changes during the lifetime of that
+              // list.
+              itr->second = sources.end();
+            }
+          }
+        }
+        return true;
+      },
+      ignore_minimized, only_zero_layer);
+
+  if (!ret)
+    return false;
+
+  RTC_DCHECK(windows);
+  windows->reserve(windows->size() + sources.size());
+  std::copy(std::begin(sources), std::end(sources),
+            std::back_inserter(*windows));
+
+  return true;
+}
+
+// Returns true if the window is occupying a full screen.
+bool IsWindowFullScreen(const MacDesktopConfiguration& desktop_config,
+                        CFDictionaryRef window) {
+  bool fullscreen = false;
+  CFDictionaryRef bounds_ref = reinterpret_cast<CFDictionaryRef>(
+      CFDictionaryGetValue(window, kCGWindowBounds));
+
+  CGRect bounds;
+  if (bounds_ref &&
+      CGRectMakeWithDictionaryRepresentation(bounds_ref, &bounds)) {
+    for (MacDisplayConfigurations::const_iterator it =
+             desktop_config.displays.begin();
+         it != desktop_config.displays.end(); it++) {
+      if (it->bounds.equals(
+              DesktopRect::MakeXYWH(bounds.origin.x, bounds.origin.y,
+                                    bounds.size.width, bounds.size.height))) {
+        fullscreen = true;
+        break;
+      }
+    }
+  }
+
+  return fullscreen;
+}
+
+bool IsWindowFullScreen(const MacDesktopConfiguration& desktop_config,
+                        CGWindowID id) {
+  bool fullscreen = false;
+  GetWindowRef(id, [&](CFDictionaryRef window) {
+    fullscreen = IsWindowFullScreen(desktop_config, window);
+  });
+  return fullscreen;
+}
+
+bool IsWindowOnScreen(CFDictionaryRef window) {
+  CFBooleanRef on_screen = reinterpret_cast<CFBooleanRef>(
+      CFDictionaryGetValue(window, kCGWindowIsOnscreen));
+  return on_screen != NULL && CFBooleanGetValue(on_screen);
+}
+
+bool IsWindowOnScreen(CGWindowID id) {
+  bool on_screen;
+  if (GetWindowRef(id, [&on_screen](CFDictionaryRef window) {
+        on_screen = IsWindowOnScreen(window);
+      })) {
+    return on_screen;
+  }
+  return false;
+}
+
+std::string GetWindowTitle(CFDictionaryRef window) {
+  CFStringRef title = reinterpret_cast<CFStringRef>(
+      CFDictionaryGetValue(window, kCGWindowName));
+  std::string result;
+  if (title && ToUtf8(title, &result)) {
+    return result;
+  }
+
+  return std::string();
+}
+
+std::string GetWindowTitle(CGWindowID id) {
+  std::string title;
+  if (GetWindowRef(id, [&title](CFDictionaryRef window) {
+        title = GetWindowTitle(window);
+      })) {
+    return title;
+  }
+  return std::string();
+}
+
+std::string GetWindowOwnerName(CFDictionaryRef window) {
+  CFStringRef owner_name = reinterpret_cast<CFStringRef>(
+      CFDictionaryGetValue(window, kCGWindowOwnerName));
+  std::string result;
+  if (owner_name && ToUtf8(owner_name, &result)) {
+    return result;
+  }
+  return std::string();
+}
+
+std::string GetWindowOwnerName(CGWindowID id) {
+  std::string owner_name;
+  if (GetWindowRef(id, [&owner_name](CFDictionaryRef window) {
+        owner_name = GetWindowOwnerName(window);
+      })) {
+    return owner_name;
+  }
+  return std::string();
+}
+
+WindowId GetWindowId(CFDictionaryRef window) {
+  CFNumberRef window_id = reinterpret_cast<CFNumberRef>(
+      CFDictionaryGetValue(window, kCGWindowNumber));
+  if (!window_id) {
+    return kNullWindowId;
+  }
+
+  // Note: WindowId is 64-bit on 64-bit system, but CGWindowID is always 32-bit.
+  // CFNumberGetValue() fills only top 32 bits, so we should use CGWindowID to
+  // receive the window id.
+  CGWindowID id;
+  if (!CFNumberGetValue(window_id, kCFNumberIntType, &id)) {
+    return kNullWindowId;
+  }
+
+  return id;
+}
+
+int GetWindowOwnerPid(CFDictionaryRef window) {
+  CFNumberRef window_pid = reinterpret_cast<CFNumberRef>(
+      CFDictionaryGetValue(window, kCGWindowOwnerPID));
+  if (!window_pid) {
+    return 0;
+  }
+
+  int pid;
+  if (!CFNumberGetValue(window_pid, kCFNumberIntType, &pid)) {
+    return 0;
+  }
+
+  return pid;
+}
+
+int GetWindowOwnerPid(CGWindowID id) {
+  int pid;
+  if (GetWindowRef(id, [&pid](CFDictionaryRef window) {
+        pid = GetWindowOwnerPid(window);
+      })) {
+    return pid;
+  }
+  return 0;
+}
+
+float GetScaleFactorAtPosition(const MacDesktopConfiguration& desktop_config,
+                               DesktopVector position) {
+  // Find the dpi to physical pixel scale for the screen where the mouse cursor
+  // is.
+  for (auto it = desktop_config.displays.begin();
+       it != desktop_config.displays.end(); ++it) {
+    if (it->bounds.Contains(position)) {
+      return it->dip_to_pixel_scale;
+    }
+  }
+  return 1;
+}
+
+float GetWindowScaleFactor(CGWindowID id, DesktopSize size) {
+  DesktopRect window_bounds = GetWindowBounds(id);
+  float scale = 1.0f;
+
+  if (!window_bounds.is_empty() && !size.is_empty()) {
+    float scale_x = size.width() / window_bounds.width();
+    float scale_y = size.height() / window_bounds.height();
+    // Currently the scale in X and Y directions must be same.
+    if ((std::fabs(scale_x - scale_y) <=
+         std::numeric_limits<float>::epsilon() * std::max(scale_x, scale_y)) &&
+        scale_x > 0.0f) {
+      scale = scale_x;
+    }
+  }
+
+  return scale;
+}
+
+DesktopRect GetWindowBounds(CFDictionaryRef window) {
+  CFDictionaryRef window_bounds = reinterpret_cast<CFDictionaryRef>(
+      CFDictionaryGetValue(window, kCGWindowBounds));
+  if (!window_bounds) {
+    return DesktopRect();
+  }
+
+  CGRect gc_window_rect;
+  if (!CGRectMakeWithDictionaryRepresentation(window_bounds, &gc_window_rect)) {
+    return DesktopRect();
+  }
+
+  return DesktopRect::MakeXYWH(gc_window_rect.origin.x, gc_window_rect.origin.y,
+                               gc_window_rect.size.width,
+                               gc_window_rect.size.height);
+}
+
+DesktopRect GetWindowBounds(CGWindowID id) {
+  DesktopRect result;
+  if (GetWindowRef(id, [&result](CFDictionaryRef window) {
+        result = GetWindowBounds(window);
+      })) {
+    return result;
+  }
+  return DesktopRect();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mac/window_list_utils.h b/third_party/libwebrtc/modules/desktop_capture/mac/window_list_utils.h
new file mode 100644
index 0000000000..a9b1e7007c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mac/window_list_utils.h
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MAC_WINDOW_LIST_UTILS_H_
+#define MODULES_DESKTOP_CAPTURE_MAC_WINDOW_LIST_UTILS_H_
+
+#include <ApplicationServices/ApplicationServices.h>
+
+#include <string>
+#include "api/function_view.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+
+namespace webrtc {
+
+// Iterates all on-screen windows in decreasing z-order and sends them
+// one-by-one to `on_window` function. If `on_window` returns false, this
+// function returns immediately. GetWindowList() returns false if native APIs
+// failed. Menus, dock (if `only_zero_layer`), minimized windows (if
+// `ignore_minimized` is true) and any windows which do not have a valid window
+// id or title will be ignored.
+bool GetWindowList(rtc::FunctionView<bool(CFDictionaryRef)> on_window,
+                   bool ignore_minimized,
+                   bool only_zero_layer);
+
+// Another helper function to get the on-screen windows.
+bool GetWindowList(DesktopCapturer::SourceList* windows,
+                   bool ignore_minimized,
+                   bool only_zero_layer);
+
+// Returns true if the window is occupying a full screen.
+bool IsWindowFullScreen(const MacDesktopConfiguration& desktop_config,
+                        CFDictionaryRef window);
+
+// Returns true if the window is occupying a full screen.
+bool IsWindowFullScreen(const MacDesktopConfiguration& desktop_config,
+                        CGWindowID id);
+
+// Returns true if the `window` is on screen. This function returns false if
+// native APIs fail.
+bool IsWindowOnScreen(CFDictionaryRef window);
+
+// Returns true if the window is on screen. This function returns false if
+// native APIs fail or `id` cannot be found.
+bool IsWindowOnScreen(CGWindowID id);
+
+// Returns utf-8 encoded title of `window`. If `window` is not a window or no
+// valid title can be retrieved, this function returns an empty string.
+std::string GetWindowTitle(CFDictionaryRef window);
+
+// Returns utf-8 encoded title of window `id`. If `id` cannot be found or no
+// valid title can be retrieved, this function returns an empty string.
+std::string GetWindowTitle(CGWindowID id);
+
+// Returns utf-8 encoded owner name of `window`. If `window` is not a window or
+// if no valid owner name can be retrieved, returns an empty string.
+std::string GetWindowOwnerName(CFDictionaryRef window);
+
+// Returns utf-8 encoded owner name of the given window `id`. If `id` cannot be
+// found or if no valid owner name can be retrieved, returns an empty string.
+std::string GetWindowOwnerName(CGWindowID id);
+
+// Returns id of `window`. If `window` is not a window or the window id cannot
+// be retrieved, this function returns kNullWindowId.
+WindowId GetWindowId(CFDictionaryRef window);
+
+// Returns the pid of the process owning `window`. Return 0 if `window` is not
+// a window or no valid owner can be retrieved.
+int GetWindowOwnerPid(CFDictionaryRef window);
+
+// Returns the pid of the process owning the window `id`. Return 0 if `id`
+// cannot be found or no valid owner can be retrieved.
+int GetWindowOwnerPid(CGWindowID id);
+
+// Returns the DIP to physical pixel scale at `position`. `position` is in
+// *unscaled* system coordinate, i.e. it's device-independent and the primary
+// monitor starts from (0, 0). If `position` is out of the system display, this
+// function returns 1.
+float GetScaleFactorAtPosition(const MacDesktopConfiguration& desktop_config,
+                               DesktopVector position);
+
+// Returns the DIP to physical pixel scale factor of the window with `id`.
+// The bounds of the window with `id` is in DIP coordinates and `size` is the
+// CGImage size of the window with `id` in physical coordinates. Comparing them
+// can give the current scale factor.
+// If the window overlaps multiple monitors, OS will decide on which monitor the
+// window is displayed and use its scale factor to the window. So this method
+// still works.
+float GetWindowScaleFactor(CGWindowID id, DesktopSize size);
+
+// Returns the bounds of `window`. If `window` is not a window or the bounds
+// cannot be retrieved, this function returns an empty DesktopRect. The returned
+// DesktopRect is in system coordinate, i.e. the primary monitor always starts
+// from (0, 0).
+// Deprecated: This function should be avoided in favor of the overload with
+// MacDesktopConfiguration.
+DesktopRect GetWindowBounds(CFDictionaryRef window);
+
+// Returns the bounds of window with `id`. If `id` does not represent a window
+// or the bounds cannot be retrieved, this function returns an empty
+// DesktopRect. The returned DesktopRect is in system coordinates.
+// Deprecated: This function should be avoided in favor of the overload with
+// MacDesktopConfiguration.
+DesktopRect GetWindowBounds(CGWindowID id);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MAC_WINDOW_LIST_UTILS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mock_desktop_capturer_callback.cc b/third_party/libwebrtc/modules/desktop_capture/mock_desktop_capturer_callback.cc
new file mode 100644
index 0000000000..de77d99e18
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mock_desktop_capturer_callback.cc
@@ -0,0 +1,23 @@
+/* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mock_desktop_capturer_callback.h"
+
+namespace webrtc {
+
+MockDesktopCapturerCallback::MockDesktopCapturerCallback() = default;
+MockDesktopCapturerCallback::~MockDesktopCapturerCallback() = default;
+
+void MockDesktopCapturerCallback::OnCaptureResult(
+    DesktopCapturer::Result result,
+    std::unique_ptr<DesktopFrame> frame) {
+  OnCaptureResultPtr(result, &frame);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mock_desktop_capturer_callback.h b/third_party/libwebrtc/modules/desktop_capture/mock_desktop_capturer_callback.h
new file mode 100644
index 0000000000..bb15ceaf4a
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mock_desktop_capturer_callback.h
@@ -0,0 +1,39 @@
+/* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MOCK_DESKTOP_CAPTURER_CALLBACK_H_
+#define MODULES_DESKTOP_CAPTURE_MOCK_DESKTOP_CAPTURER_CALLBACK_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDesktopCapturerCallback : public DesktopCapturer::Callback {
+ public:
+  MockDesktopCapturerCallback();
+  ~MockDesktopCapturerCallback() override;
+
+  MockDesktopCapturerCallback(const MockDesktopCapturerCallback&) = delete;
+  MockDesktopCapturerCallback& operator=(const MockDesktopCapturerCallback&) =
+      delete;
+
+  MOCK_METHOD(void,
+              OnCaptureResultPtr,
+              (DesktopCapturer::Result result,
+               std::unique_ptr<DesktopFrame>* frame));
+  void OnCaptureResult(DesktopCapturer::Result result,
+                       std::unique_ptr<DesktopFrame> frame) final;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MOCK_DESKTOP_CAPTURER_CALLBACK_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mouse_cursor.cc b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor.cc
new file mode 100644
index 0000000000..e826552b0f
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor.cc
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mouse_cursor.h"
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+MouseCursor::MouseCursor() {}
+
+MouseCursor::MouseCursor(DesktopFrame* image, const DesktopVector& hotspot)
+    : image_(image), hotspot_(hotspot) {
+  RTC_DCHECK(0 <= hotspot_.x() && hotspot_.x() <= image_->size().width());
+  RTC_DCHECK(0 <= hotspot_.y() && hotspot_.y() <= image_->size().height());
+}
+
+MouseCursor::~MouseCursor() {}
+
+// static
+MouseCursor* MouseCursor::CopyOf(const MouseCursor& cursor) {
+  return cursor.image()
+             ? new MouseCursor(BasicDesktopFrame::CopyOf(*cursor.image()),
+                               cursor.hotspot())
+             : new MouseCursor();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mouse_cursor.h b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor.h
new file mode 100644
index 0000000000..2dd793179b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_H_
+#define MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class RTC_EXPORT MouseCursor {
+ public:
+  MouseCursor();
+
+  // Takes ownership of `image`. `hotspot` must be within `image` boundaries.
+  MouseCursor(DesktopFrame* image, const DesktopVector& hotspot);
+
+  ~MouseCursor();
+
+  MouseCursor(const MouseCursor&) = delete;
+  MouseCursor& operator=(const MouseCursor&) = delete;
+
+  static MouseCursor* CopyOf(const MouseCursor& cursor);
+
+  void set_image(DesktopFrame* image) { image_.reset(image); }
+  const DesktopFrame* image() const { return image_.get(); }
+
+  void set_hotspot(const DesktopVector& hotspot) { hotspot_ = hotspot; }
+  const DesktopVector& hotspot() const { return hotspot_; }
+
+ private:
+  std::unique_ptr<DesktopFrame> image_;
+  DesktopVector hotspot_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor.h b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor.h
new file mode 100644
index 0000000000..ad134df36d
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_MONITOR_H_
+#define MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_MONITOR_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class DesktopCaptureOptions;
+class DesktopFrame;
+class MouseCursor;
+
+// Captures mouse shape and position.
+class MouseCursorMonitor {
+ public:
+  // Deprecated: CursorState will not be provided.
+  enum CursorState {
+    // Cursor on top of the window including window decorations.
+    INSIDE,
+
+    // Cursor is outside of the window.
+    OUTSIDE,
+  };
+
+  enum Mode {
+    // Capture only shape of the mouse cursor, but not position.
+    SHAPE_ONLY,
+
+    // Capture both, mouse cursor shape and position.
+    SHAPE_AND_POSITION,
+  };
+
+  // Callback interface used to pass current mouse cursor position and shape.
+  class Callback {
+   public:
+    // Called in response to Capture() when the cursor shape has changed. Must
+    // take ownership of `cursor`.
+    virtual void OnMouseCursor(MouseCursor* cursor) = 0;
+
+    // Called in response to Capture(). `position` indicates cursor position
+    // relative to the `window` specified in the constructor.
+    // Deprecated: use the following overload instead.
+    virtual void OnMouseCursorPosition(CursorState state,
+                                       const DesktopVector& position) {}
+
+    // Called in response to Capture(). `position` indicates cursor absolute
+    // position on the system in fullscreen coordinate, i.e. the top-left
+    // monitor always starts from (0, 0).
+    // The coordinates of the position is controlled by OS, but it's always
+    // consistent with DesktopFrame.rect().top_left().
+    // TODO(zijiehe): Ensure all implementations return the absolute position.
+    // TODO(zijiehe): Current this overload works correctly only when capturing
+    // mouse cursor against fullscreen.
+    virtual void OnMouseCursorPosition(const DesktopVector& position) {}
+
+   protected:
+    virtual ~Callback() {}
+  };
+
+  virtual ~MouseCursorMonitor() {}
+
+  // Creates a capturer that notifies of mouse cursor events while the cursor is
+  // over the specified window.
+  //
+  // Deprecated: use Create() function.
+  static MouseCursorMonitor* CreateForWindow(
+      const DesktopCaptureOptions& options,
+      WindowId window);
+
+  // Creates a capturer that monitors the mouse cursor shape and position over
+  // the specified screen.
+  //
+  // Deprecated: use Create() function.
+  static RTC_EXPORT MouseCursorMonitor* CreateForScreen(
+      const DesktopCaptureOptions& options,
+      ScreenId screen);
+
+  // Creates a capturer that monitors the mouse cursor shape and position across
+  // the entire desktop. The capturer ensures that the top-left monitor starts
+  // from (0, 0).
+  static RTC_EXPORT std::unique_ptr<MouseCursorMonitor> Create(
+      const DesktopCaptureOptions& options);
+
+  // Initializes the monitor with the `callback`, which must remain valid until
+  // capturer is destroyed.
+  virtual void Init(Callback* callback, Mode mode) = 0;
+
+  // Captures current cursor shape and position (depending on the `mode` passed
+  // to Init()). Calls Callback::OnMouseCursor() if cursor shape has
+  // changed since the last call (or when Capture() is called for the first
+  // time) and then Callback::OnMouseCursorPosition() if mode is set to
+  // SHAPE_AND_POSITION.
+  virtual void Capture() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_MONITOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_linux.cc b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_linux.cc
new file mode 100644
index 0000000000..b44500d4e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_linux.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+
+#if defined(WEBRTC_USE_X11)
+#include "modules/desktop_capture/linux/x11/mouse_cursor_monitor_x11.h"
+#endif  // defined(WEBRTC_USE_X11)
+
+#if defined(WEBRTC_USE_PIPEWIRE)
+#include "modules/desktop_capture/linux/wayland/mouse_cursor_monitor_pipewire.h"
+#endif  // defined(WEBRTC_USE_PIPEWIRE)
+
+namespace webrtc {
+
+// static
+MouseCursorMonitor* MouseCursorMonitor::CreateForWindow(
+    const DesktopCaptureOptions& options,
+    WindowId window) {
+#if defined(WEBRTC_USE_X11)
+  return MouseCursorMonitorX11::CreateForWindow(options, window);
+#else
+  return nullptr;
+#endif  // defined(WEBRTC_USE_X11)
+}
+
+// static
+MouseCursorMonitor* MouseCursorMonitor::CreateForScreen(
+    const DesktopCaptureOptions& options,
+    ScreenId screen) {
+#if defined(WEBRTC_USE_X11)
+  return MouseCursorMonitorX11::CreateForScreen(options, screen);
+#else
+  return nullptr;
+#endif  // defined(WEBRTC_USE_X11)
+}
+
+// static
+std::unique_ptr<MouseCursorMonitor> MouseCursorMonitor::Create(
+    const DesktopCaptureOptions& options) {
+#if defined(WEBRTC_USE_PIPEWIRE)
+  if (options.allow_pipewire() && DesktopCapturer::IsRunningUnderWayland() &&
+      options.screencast_stream()) {
+    return std::make_unique<MouseCursorMonitorPipeWire>(options);
+  }
+#endif  // defined(WEBRTC_USE_PIPEWIRE)
+
+#if defined(WEBRTC_USE_X11)
+  return MouseCursorMonitorX11::Create(options);
+#else
+  return nullptr;
+#endif  // defined(WEBRTC_USE_X11)
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
new file mode 100644
index 0000000000..512103ab5e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
@@ -0,0 +1,213 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+
+
+#include <memory>
+
+#include <ApplicationServices/ApplicationServices.h>
+#include <Cocoa/Cocoa.h>
+#include <CoreFoundation/CoreFoundation.h>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
+#include "modules/desktop_capture/mac/window_list_utils.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+CGImageRef CreateScaledCGImage(CGImageRef image, int width, int height) {
+  // Create context, keeping original image properties.
+  CGColorSpaceRef colorspace = CGImageGetColorSpace(image);
+  CGContextRef context = CGBitmapContextCreate(nullptr,
+                                               width,
+                                               height,
+                                               CGImageGetBitsPerComponent(image),
+                                               width * DesktopFrame::kBytesPerPixel,
+                                               colorspace,
+                                               CGImageGetBitmapInfo(image));
+
+  if (!context) return nil;
+
+  // Draw image to context, resizing it.
+  CGContextDrawImage(context, CGRectMake(0, 0, width, height), image);
+  // Extract resulting image from context.
+  CGImageRef imgRef = CGBitmapContextCreateImage(context);
+  CGContextRelease(context);
+
+  return imgRef;
+}
+}  // namespace
+
+class MouseCursorMonitorMac : public MouseCursorMonitor {
+ public:
+  MouseCursorMonitorMac(const DesktopCaptureOptions& options,
+                        CGWindowID window_id,
+                        ScreenId screen_id);
+  ~MouseCursorMonitorMac() override;
+
+  void Init(Callback* callback, Mode mode) override;
+  void Capture() override;
+
+ private:
+  static void DisplaysReconfiguredCallback(CGDirectDisplayID display,
+                                           CGDisplayChangeSummaryFlags flags,
+                                           void *user_parameter);
+  void DisplaysReconfigured(CGDirectDisplayID display,
+                            CGDisplayChangeSummaryFlags flags);
+
+  void CaptureImage(float scale);
+
+  rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_;
+  CGWindowID window_id_;
+  ScreenId screen_id_;
+  Callback* callback_ = NULL;
+  Mode mode_;
+  __strong NSImage* last_cursor_ = NULL;
+};
+
+MouseCursorMonitorMac::MouseCursorMonitorMac(const DesktopCaptureOptions& options,
+                                             CGWindowID window_id,
+                                             ScreenId screen_id)
+    : configuration_monitor_(options.configuration_monitor()),
+      window_id_(window_id),
+      screen_id_(screen_id),
+      mode_(SHAPE_AND_POSITION) {
+  RTC_DCHECK(window_id == kCGNullWindowID || screen_id == kInvalidScreenId);
+}
+
+MouseCursorMonitorMac::~MouseCursorMonitorMac() {}
+
+void MouseCursorMonitorMac::Init(Callback* callback, Mode mode) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+
+  callback_ = callback;
+  mode_ = mode;
+}
+
+void MouseCursorMonitorMac::Capture() {
+  RTC_DCHECK(callback_);
+
+  CGEventRef event = CGEventCreate(NULL);
+  CGPoint gc_position = CGEventGetLocation(event);
+  CFRelease(event);
+
+  DesktopVector position(gc_position.x, gc_position.y);
+
+  MacDesktopConfiguration configuration =
+      configuration_monitor_->desktop_configuration();
+  float scale = GetScaleFactorAtPosition(configuration, position);
+
+  CaptureImage(scale);
+
+  if (mode_ != SHAPE_AND_POSITION)
+    return;
+
+  // Always report cursor position in DIP pixel.
+  callback_->OnMouseCursorPosition(
+      position.subtract(configuration.bounds.top_left()));
+}
+
+void MouseCursorMonitorMac::CaptureImage(float scale) {
+  NSCursor* nscursor = [NSCursor currentSystemCursor];
+
+  NSImage* nsimage = [nscursor image];
+  if (nsimage == nil || !nsimage.isValid) {
+    return;
+  }
+  NSSize nssize = [nsimage size];  // DIP size
+
+  // No need to caputre cursor image if it's unchanged since last capture.
+  if (last_cursor_ && [[nsimage TIFFRepresentation] isEqual:[last_cursor_ TIFFRepresentation]]) return;
+  last_cursor_ = nsimage;
+
+  DesktopSize size(round(nssize.width * scale),
+                   round(nssize.height * scale));  // Pixel size
+  NSPoint nshotspot = [nscursor hotSpot];
+  DesktopVector hotspot(
+      std::max(0,
+               std::min(size.width(), static_cast<int>(nshotspot.x * scale))),
+      std::max(0,
+               std::min(size.height(), static_cast<int>(nshotspot.y * scale))));
+  CGImageRef cg_image =
+      [nsimage CGImageForProposedRect:NULL context:nil hints:nil];
+  if (!cg_image)
+    return;
+
+  // Before 10.12, OSX may report 1X cursor on Retina screen. (See
+  // crbug.com/632995.) After 10.12, OSX may report 2X cursor on non-Retina
+  // screen. (See crbug.com/671436.) So scaling the cursor if needed.
+  CGImageRef scaled_cg_image = nil;
+  if (CGImageGetWidth(cg_image) != static_cast<size_t>(size.width())) {
+    scaled_cg_image = CreateScaledCGImage(cg_image, size.width(), size.height());
+    if (scaled_cg_image != nil) {
+      cg_image = scaled_cg_image;
+    }
+  }
+  if (CGImageGetBitsPerPixel(cg_image) != DesktopFrame::kBytesPerPixel * 8 ||
+      CGImageGetWidth(cg_image) != static_cast<size_t>(size.width()) ||
+      CGImageGetBitsPerComponent(cg_image) != 8) {
+    if (scaled_cg_image != nil) CGImageRelease(scaled_cg_image);
+    return;
+  }
+
+  CGDataProviderRef provider = CGImageGetDataProvider(cg_image);
+  CFDataRef image_data_ref = CGDataProviderCopyData(provider);
+  if (image_data_ref == NULL) {
+    if (scaled_cg_image != nil) CGImageRelease(scaled_cg_image);
+    return;
+  }
+
+  const uint8_t* src_data =
+      reinterpret_cast<const uint8_t*>(CFDataGetBytePtr(image_data_ref));
+
+  // Create a MouseCursor that describes the cursor and pass it to
+  // the client.
+  std::unique_ptr<DesktopFrame> image(
+      new BasicDesktopFrame(DesktopSize(size.width(), size.height())));
+
+  int src_stride = CGImageGetBytesPerRow(cg_image);
+  image->CopyPixelsFrom(src_data, src_stride, DesktopRect::MakeSize(size));
+
+  CFRelease(image_data_ref);
+  if (scaled_cg_image != nil) CGImageRelease(scaled_cg_image);
+
+  std::unique_ptr<MouseCursor> cursor(
+      new MouseCursor(image.release(), hotspot));
+
+  callback_->OnMouseCursor(cursor.release());
+}
+
+MouseCursorMonitor* MouseCursorMonitor::CreateForWindow(
+    const DesktopCaptureOptions& options, WindowId window) {
+  return new MouseCursorMonitorMac(options, window, kInvalidScreenId);
+}
+
+MouseCursorMonitor* MouseCursorMonitor::CreateForScreen(
+    const DesktopCaptureOptions& options,
+    ScreenId screen) {
+  return new MouseCursorMonitorMac(options, kCGNullWindowID, screen);
+}
+
+std::unique_ptr<MouseCursorMonitor> MouseCursorMonitor::Create(
+    const DesktopCaptureOptions& options) {
+  return std::unique_ptr<MouseCursorMonitor>(
+      CreateForScreen(options, kFullDesktopScreenId));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_null.cc b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_null.cc
new file mode 100644
index 0000000000..ab1bc2fa33
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_null.cc
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+
+namespace webrtc {
+
+MouseCursorMonitor* MouseCursorMonitor::CreateForWindow(
+    const DesktopCaptureOptions& options,
+    WindowId window) {
+  return NULL;
+}
+
+MouseCursorMonitor* MouseCursorMonitor::CreateForScreen(
+    const DesktopCaptureOptions& options,
+    ScreenId screen) {
+  return NULL;
+}
+
+std::unique_ptr<MouseCursorMonitor> MouseCursorMonitor::Create(
+    const DesktopCaptureOptions& options) {
+  return std::unique_ptr<MouseCursorMonitor>(
+      CreateForScreen(options, kFullDesktopScreenId));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc
new file mode 100644
index 0000000000..f771276a2b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+
+#include <stddef.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class MouseCursorMonitorTest : public ::testing::Test,
+                               public MouseCursorMonitor::Callback {
+ public:
+  MouseCursorMonitorTest() : position_received_(false) {}
+
+  // MouseCursorMonitor::Callback interface
+  void OnMouseCursor(MouseCursor* cursor_image) override {
+    cursor_image_.reset(cursor_image);
+  }
+
+  void OnMouseCursorPosition(const DesktopVector& position) override {
+    position_ = position;
+    position_received_ = true;
+  }
+
+ protected:
+  std::unique_ptr<MouseCursor> cursor_image_;
+  DesktopVector position_;
+  bool position_received_;
+};
+
+// TODO(sergeyu): On Mac we need to initialize NSApplication before running the
+// tests. Figure out how to do that without breaking other tests in
+// modules_unittests and enable these tests on Mac.
+// https://code.google.com/p/webrtc/issues/detail?id=2532
+//
+// Disabled on Windows due to flake, see:
+// https://code.google.com/p/webrtc/issues/detail?id=3408
+// Disabled on Linux due to flake, see:
+// https://code.google.com/p/webrtc/issues/detail?id=3245
+#if !defined(WEBRTC_MAC) && !defined(WEBRTC_WIN) && !defined(WEBRTC_LINUX)
+#define MAYBE(x) x
+#else
+#define MAYBE(x) DISABLED_##x
+#endif
+
+TEST_F(MouseCursorMonitorTest, MAYBE(FromScreen)) {
+  std::unique_ptr<MouseCursorMonitor> capturer(
+      MouseCursorMonitor::CreateForScreen(
+          DesktopCaptureOptions::CreateDefault(),
+          webrtc::kFullDesktopScreenId));
+  RTC_DCHECK(capturer.get());
+  capturer->Init(this, MouseCursorMonitor::SHAPE_AND_POSITION);
+  capturer->Capture();
+
+  EXPECT_TRUE(cursor_image_.get());
+  EXPECT_GE(cursor_image_->hotspot().x(), 0);
+  EXPECT_LE(cursor_image_->hotspot().x(),
+            cursor_image_->image()->size().width());
+  EXPECT_GE(cursor_image_->hotspot().y(), 0);
+  EXPECT_LE(cursor_image_->hotspot().y(),
+            cursor_image_->image()->size().height());
+
+  EXPECT_TRUE(position_received_);
+}
+
+TEST_F(MouseCursorMonitorTest, MAYBE(FromWindow)) {
+  DesktopCaptureOptions options = DesktopCaptureOptions::CreateDefault();
+
+  // First get list of windows.
+  std::unique_ptr<DesktopCapturer> window_capturer(
+      DesktopCapturer::CreateWindowCapturer(options));
+
+  // If window capturing is not supported then skip this test.
+  if (!window_capturer.get())
+    return;
+
+  DesktopCapturer::SourceList sources;
+  EXPECT_TRUE(window_capturer->GetSourceList(&sources));
+
+  // Iterate over all windows and try capturing mouse cursor for each of them.
+  for (size_t i = 0; i < sources.size(); ++i) {
+    cursor_image_.reset();
+    position_received_ = false;
+
+    std::unique_ptr<MouseCursorMonitor> capturer(
+        MouseCursorMonitor::CreateForWindow(
+            DesktopCaptureOptions::CreateDefault(), sources[i].id));
+    RTC_DCHECK(capturer.get());
+
+    capturer->Init(this, MouseCursorMonitor::SHAPE_AND_POSITION);
+    capturer->Capture();
+
+    EXPECT_TRUE(cursor_image_.get());
+    EXPECT_TRUE(position_received_);
+  }
+}
+
+// Make sure that OnMouseCursorPosition() is not called in the SHAPE_ONLY mode.
+TEST_F(MouseCursorMonitorTest, MAYBE(ShapeOnly)) {
+  std::unique_ptr<MouseCursorMonitor> capturer(
+      MouseCursorMonitor::CreateForScreen(
+          DesktopCaptureOptions::CreateDefault(),
+          webrtc::kFullDesktopScreenId));
+  RTC_DCHECK(capturer.get());
+  capturer->Init(this, MouseCursorMonitor::SHAPE_ONLY);
+  capturer->Capture();
+
+  EXPECT_TRUE(cursor_image_.get());
+  EXPECT_FALSE(position_received_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
new file mode 100644
index 0000000000..18ef43eeb4
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
@@ -0,0 +1,220 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/mouse_cursor_monitor.h"
+#include "modules/desktop_capture/win/cursor.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "modules/desktop_capture/win/window_capture_utils.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+bool IsSameCursorShape(const CURSORINFO& left, const CURSORINFO& right) {
+  // If the cursors are not showing, we do not care the hCursor handle.
+  return left.flags == right.flags &&
+         (left.flags != CURSOR_SHOWING || left.hCursor == right.hCursor);
+}
+
+}  // namespace
+
+class MouseCursorMonitorWin : public MouseCursorMonitor {
+ public:
+  explicit MouseCursorMonitorWin(HWND window);
+  explicit MouseCursorMonitorWin(ScreenId screen);
+  ~MouseCursorMonitorWin() override;
+
+  void Init(Callback* callback, Mode mode) override;
+  void Capture() override;
+
+ private:
+  // Get the rect of the currently selected screen, relative to the primary
+  // display's top-left. If the screen is disabled or disconnected, or any error
+  // happens, an empty rect is returned.
+  DesktopRect GetScreenRect();
+
+  HWND window_;
+  ScreenId screen_;
+
+  Callback* callback_;
+  Mode mode_;
+
+  HDC desktop_dc_;
+
+  // The last CURSORINFO (converted to MouseCursor) we have sent to the client.
+  CURSORINFO last_cursor_;
+};
+
+MouseCursorMonitorWin::MouseCursorMonitorWin(HWND window)
+    : window_(window),
+      screen_(kInvalidScreenId),
+      callback_(NULL),
+      mode_(SHAPE_AND_POSITION),
+      desktop_dc_(NULL) {
+  memset(&last_cursor_, 0, sizeof(CURSORINFO));
+}
+
+MouseCursorMonitorWin::MouseCursorMonitorWin(ScreenId screen)
+    : window_(NULL),
+      screen_(screen),
+      callback_(NULL),
+      mode_(SHAPE_AND_POSITION),
+      desktop_dc_(NULL) {
+  RTC_DCHECK_GE(screen, kFullDesktopScreenId);
+  memset(&last_cursor_, 0, sizeof(CURSORINFO));
+}
+
+MouseCursorMonitorWin::~MouseCursorMonitorWin() {
+  if (desktop_dc_)
+    ReleaseDC(NULL, desktop_dc_);
+}
+
+void MouseCursorMonitorWin::Init(Callback* callback, Mode mode) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+  RTC_DCHECK(IsGUIThread(false));
+
+  callback_ = callback;
+  mode_ = mode;
+
+  desktop_dc_ = GetDC(NULL);
+}
+
+void MouseCursorMonitorWin::Capture() {
+// TODO: Bug 1666266. Commented out to pass new tests added in bug 1634044.
+//  RTC_DCHECK(IsGUIThread(false));
+  RTC_DCHECK(callback_);
+
+  CURSORINFO cursor_info;
+  cursor_info.cbSize = sizeof(CURSORINFO);
+  if (!GetCursorInfo(&cursor_info)) {
+    RTC_LOG_F(LS_ERROR) << "Unable to get cursor info. Error = "
+                        << GetLastError();
+    return;
+  }
+
+  if (!IsSameCursorShape(cursor_info, last_cursor_)) {
+    // Mozilla - CURSOR_SUPPRESSED is win8 and above; so we seem not to be able to see the symbol
+    if (cursor_info.flags != CURSOR_SHOWING) {
+      // The cursor is intentionally hidden now, send an empty bitmap.
+      last_cursor_ = cursor_info;
+      callback_->OnMouseCursor(new MouseCursor(
+          new BasicDesktopFrame(DesktopSize()), DesktopVector()));
+    } else {
+      // According to MSDN https://goo.gl/u6gyuC, HCURSOR instances returned by
+      // functions other than CreateCursor do not need to be actively destroyed.
+      // And CloseHandle function (https://goo.gl/ja5ycW) does not close a
+      // cursor, so assume a HCURSOR does not need to be closed.
+      if (cursor_info.flags == 0) {
+        // Host machine does not have a hardware mouse attached, we will send a
+        // default one instead.
+        // Note, Windows automatically caches cursor resource, so we do not need
+        // to cache the result of LoadCursor.
+        cursor_info.hCursor = LoadCursor(nullptr, IDC_ARROW);
+      }
+      std::unique_ptr<MouseCursor> cursor(
+          CreateMouseCursorFromHCursor(desktop_dc_, cursor_info.hCursor));
+      if (cursor) {
+        last_cursor_ = cursor_info;
+        callback_->OnMouseCursor(cursor.release());
+      }
+    }
+  }
+
+  if (mode_ != SHAPE_AND_POSITION)
+    return;
+
+  // CURSORINFO::ptScreenPos is in full desktop coordinate.
+  DesktopVector position(cursor_info.ptScreenPos.x, cursor_info.ptScreenPos.y);
+  bool inside = cursor_info.flags == CURSOR_SHOWING;
+
+  if (window_) {
+    DesktopRect original_rect;
+    DesktopRect cropped_rect;
+    if (!GetCroppedWindowRect(window_, /*avoid_cropping_border*/ false,
+                              &cropped_rect, &original_rect)) {
+      position.set(0, 0);
+      inside = false;
+    } else {
+      if (inside) {
+        HWND windowUnderCursor = WindowFromPoint(cursor_info.ptScreenPos);
+        inside = windowUnderCursor
+                     ? (window_ == GetAncestor(windowUnderCursor, GA_ROOT))
+                     : false;
+      }
+      position = position.subtract(cropped_rect.top_left());
+    }
+  } else {
+    RTC_DCHECK_NE(screen_, kInvalidScreenId);
+    DesktopRect rect = GetScreenRect();
+    if (inside)
+      inside = rect.Contains(position);
+    position = position.subtract(rect.top_left());
+  }
+
+  callback_->OnMouseCursorPosition(position);
+}
+
+DesktopRect MouseCursorMonitorWin::GetScreenRect() {
+  RTC_DCHECK(IsGUIThread(false));
+  RTC_DCHECK_NE(screen_, kInvalidScreenId);
+  if (screen_ == kFullDesktopScreenId) {
+    return DesktopRect::MakeXYWH(GetSystemMetrics(SM_XVIRTUALSCREEN),
+                                 GetSystemMetrics(SM_YVIRTUALSCREEN),
+                                 GetSystemMetrics(SM_CXVIRTUALSCREEN),
+                                 GetSystemMetrics(SM_CYVIRTUALSCREEN));
+  }
+  DISPLAY_DEVICE device;
+  device.cb = sizeof(device);
+  BOOL result = EnumDisplayDevices(NULL, screen_, &device, 0);
+  if (!result)
+    return DesktopRect();
+
+  DEVMODE device_mode;
+  device_mode.dmSize = sizeof(device_mode);
+  device_mode.dmDriverExtra = 0;
+  result = EnumDisplaySettingsEx(device.DeviceName, ENUM_CURRENT_SETTINGS,
+                                 &device_mode, 0);
+  if (!result)
+    return DesktopRect();
+
+  return DesktopRect::MakeXYWH(
+      device_mode.dmPosition.x, device_mode.dmPosition.y,
+      device_mode.dmPelsWidth, device_mode.dmPelsHeight);
+}
+
+MouseCursorMonitor* MouseCursorMonitor::CreateForWindow(
+    const DesktopCaptureOptions& options,
+    WindowId window) {
+  return new MouseCursorMonitorWin(reinterpret_cast<HWND>(window));
+}
+
+MouseCursorMonitor* MouseCursorMonitor::CreateForScreen(
+    const DesktopCaptureOptions& options,
+    ScreenId screen) {
+  return new MouseCursorMonitorWin(screen);
+}
+
+std::unique_ptr<MouseCursorMonitor> MouseCursorMonitor::Create(
+    const DesktopCaptureOptions& options) {
+  return std::unique_ptr<MouseCursorMonitor>(
+      CreateForScreen(options, kFullDesktopScreenId));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/primitives_gn/moz.build b/third_party/libwebrtc/modules/desktop_capture/primitives_gn/moz.build
new file mode 100644
index 0000000000..59c2ba037b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/primitives_gn/moz.build
@@ -0,0 +1,214 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/media/libyuv/",
+    "/media/libyuv/libyuv/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_frame.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_geometry.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/desktop_region.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/shared_desktop_frame.cc",
+    "/third_party/libwebrtc/modules/desktop_capture/shared_memory.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "ppc64":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "riscv64":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "mips32" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "mips64" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+Library("primitives_gn")
diff --git a/third_party/libwebrtc/modules/desktop_capture/resolution_tracker.cc b/third_party/libwebrtc/modules/desktop_capture/resolution_tracker.cc
new file mode 100644
index 0000000000..9639d627fa
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/resolution_tracker.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/resolution_tracker.h"
+
+namespace webrtc {
+
+bool ResolutionTracker::SetResolution(DesktopSize size) {
+  if (!initialized_) {
+    initialized_ = true;
+    last_size_ = size;
+    return false;
+  }
+
+  if (last_size_.equals(size)) {
+    return false;
+  }
+
+  last_size_ = size;
+  return true;
+}
+
+void ResolutionTracker::Reset() {
+  initialized_ = false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/resolution_tracker.h b/third_party/libwebrtc/modules/desktop_capture/resolution_tracker.h
new file mode 100644
index 0000000000..8fe9d61862
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/resolution_tracker.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_RESOLUTION_TRACKER_H_
+#define MODULES_DESKTOP_CAPTURE_RESOLUTION_TRACKER_H_
+
+#include "modules/desktop_capture/desktop_geometry.h"
+
+namespace webrtc {
+
+class ResolutionTracker final {
+ public:
+  // Sets the resolution to `size`. Returns true if a previous size was recorded
+  // and differs from `size`.
+  bool SetResolution(DesktopSize size);
+
+  // Resets to the initial state.
+  void Reset();
+
+ private:
+  DesktopSize last_size_;
+  bool initialized_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_RESOLUTION_TRACKER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/rgba_color.cc b/third_party/libwebrtc/modules/desktop_capture/rgba_color.cc
new file mode 100644
index 0000000000..362928a474
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/rgba_color.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/rgba_color.h"
+
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+
+namespace {
+
+bool AlphaEquals(uint8_t i, uint8_t j) {
+  // On Linux and Windows 8 or early version, '0' was returned for alpha channel
+  // from capturer APIs, on Windows 10, '255' was returned. So a workaround is
+  // to treat 0 as 255.
+  return i == j || ((i == 0 || i == 255) && (j == 0 || j == 255));
+}
+
+}  // namespace
+
+RgbaColor::RgbaColor(uint8_t blue, uint8_t green, uint8_t red, uint8_t alpha) {
+  this->blue = blue;
+  this->green = green;
+  this->red = red;
+  this->alpha = alpha;
+}
+
+RgbaColor::RgbaColor(uint8_t blue, uint8_t green, uint8_t red)
+    : RgbaColor(blue, green, red, 0xff) {}
+
+RgbaColor::RgbaColor(const uint8_t* bgra)
+    : RgbaColor(bgra[0], bgra[1], bgra[2], bgra[3]) {}
+
+RgbaColor::RgbaColor(uint32_t bgra)
+    : RgbaColor(reinterpret_cast<uint8_t*>(&bgra)) {}
+
+bool RgbaColor::operator==(const RgbaColor& right) const {
+  return blue == right.blue && green == right.green && red == right.red &&
+         AlphaEquals(alpha, right.alpha);
+}
+
+bool RgbaColor::operator!=(const RgbaColor& right) const {
+  return !(*this == right);
+}
+
+uint32_t RgbaColor::ToUInt32() const {
+#if defined(WEBRTC_ARCH_LITTLE_ENDIAN)
+  return blue | (green << 8) | (red << 16) | (alpha << 24);
+#else
+  return (blue << 24) | (green << 16) | (red << 8) | alpha;
+#endif
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/rgba_color.h b/third_party/libwebrtc/modules/desktop_capture/rgba_color.h
new file mode 100644
index 0000000000..2b13430a45
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/rgba_color.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_RGBA_COLOR_H_
+#define MODULES_DESKTOP_CAPTURE_RGBA_COLOR_H_
+
+#include <stdint.h>
+
+#include "modules/desktop_capture/desktop_frame.h"
+
+namespace webrtc {
+
+// A four-byte structure to store a color in BGRA format. This structure also
+// provides functions to be created from uint8_t array, say,
+// DesktopFrame::data(). It always uses BGRA order for internal storage to match
+// DesktopFrame::data().
+struct RgbaColor final {
+  // Creates a color with BGRA channels.
+  RgbaColor(uint8_t blue, uint8_t green, uint8_t red, uint8_t alpha);
+
+  // Creates a color with BGR channels, and set alpha channel to 255 (opaque).
+  RgbaColor(uint8_t blue, uint8_t green, uint8_t red);
+
+  // Creates a color from four-byte in BGRA order, i.e. DesktopFrame::data().
+  explicit RgbaColor(const uint8_t* bgra);
+
+  // Creates a color from BGRA channels in a uint format. Consumers should make
+  // sure the memory order of the uint32_t is always BGRA from left to right, no
+  // matter the system endian. This function creates an equivalent RgbaColor
+  // instance from the ToUInt32() result of another RgbaColor instance.
+  explicit RgbaColor(uint32_t bgra);
+
+  // Returns true if `this` and `right` is the same color.
+  bool operator==(const RgbaColor& right) const;
+
+  // Returns true if `this` and `right` are different colors.
+  bool operator!=(const RgbaColor& right) const;
+
+  uint32_t ToUInt32() const;
+
+  uint8_t blue;
+  uint8_t green;
+  uint8_t red;
+  uint8_t alpha;
+};
+static_assert(
+    DesktopFrame::kBytesPerPixel == sizeof(RgbaColor),
+    "A pixel in DesktopFrame should be safe to be represented by a RgbaColor");
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_RGBA_COLOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/rgba_color_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/rgba_color_unittest.cc
new file mode 100644
index 0000000000..48e40594b1
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/rgba_color_unittest.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/rgba_color.h"
+
+#include <cstdint>
+#include <vector>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(RgbaColorTest, ConvertFromAndToUInt32) {
+  static const std::vector<uint32_t> cases{
+      0,         1000,       2693,       3725,       4097,      12532,
+      19902,     27002,      27723,      30944,      65535,     65536,
+      231194,    255985,     322871,     883798,     9585200,   12410056,
+      12641940,  30496970,   105735668,  110117847,  482769275, 542368468,
+      798173396, 2678656711, 3231043200, UINT32_MAX,
+  };
+
+  for (uint32_t value : cases) {
+    RgbaColor left(value);
+    ASSERT_EQ(left.ToUInt32(), value);
+    RgbaColor right(left);
+    ASSERT_EQ(left.ToUInt32(), right.ToUInt32());
+  }
+}
+
+TEST(RgbaColorTest, AlphaChannelEquality) {
+  RgbaColor left(10, 10, 10, 0);
+  RgbaColor right(10, 10, 10, 255);
+  ASSERT_EQ(left, right);
+  right.alpha = 128;
+  ASSERT_NE(left, right);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capture_frame_queue.h b/third_party/libwebrtc/modules/desktop_capture/screen_capture_frame_queue.h
new file mode 100644
index 0000000000..46e19da77e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capture_frame_queue.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURE_FRAME_QUEUE_H_
+#define MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURE_FRAME_QUEUE_H_
+
+#include <memory>
+
+namespace webrtc {
+
+// Represents a queue of reusable video frames. Provides access to the 'current'
+// frame - the frame that the caller is working with at the moment, and to the
+// 'previous' frame - the predecessor of the current frame swapped by
+// MoveToNextFrame() call, if any.
+//
+// The caller is expected to (re)allocate frames if current_frame() returns
+// NULL. The caller can mark all frames in the queue for reallocation (when,
+// say, frame dimensions change). The queue records which frames need updating
+// which the caller can query.
+//
+// Frame consumer is expected to never hold more than kQueueLength frames
+// created by this function and it should release the earliest one before trying
+// to capture a new frame (i.e. before MoveToNextFrame() is called).
+template <typename FrameType>
+class ScreenCaptureFrameQueue {
+ public:
+  ScreenCaptureFrameQueue() = default;
+  ~ScreenCaptureFrameQueue() = default;
+
+  ScreenCaptureFrameQueue(const ScreenCaptureFrameQueue&) = delete;
+  ScreenCaptureFrameQueue& operator=(const ScreenCaptureFrameQueue&) = delete;
+
+  // Moves to the next frame in the queue, moving the 'current' frame to become
+  // the 'previous' one.
+  void MoveToNextFrame() { current_ = (current_ + 1) % kQueueLength; }
+
+  // Replaces the current frame with a new one allocated by the caller. The
+  // existing frame (if any) is destroyed. Takes ownership of `frame`.
+  void ReplaceCurrentFrame(std::unique_ptr<FrameType> frame) {
+    frames_[current_] = std::move(frame);
+  }
+
+  // Marks all frames obsolete and resets the previous frame pointer. No
+  // frames are freed though as the caller can still access them.
+  void Reset() {
+    for (int i = 0; i < kQueueLength; i++) {
+      frames_[i].reset();
+    }
+    current_ = 0;
+  }
+
+  FrameType* current_frame() const { return frames_[current_].get(); }
+
+  FrameType* previous_frame() const {
+    return frames_[(current_ + kQueueLength - 1) % kQueueLength].get();
+  }
+
+ private:
+  // Index of the current frame.
+  int current_ = 0;
+
+  static const int kQueueLength = 2;
+  std::unique_ptr<FrameType> frames_[kQueueLength];
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURE_FRAME_QUEUE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_darwin.mm b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_darwin.mm
new file mode 100644
index 0000000000..d5a7bb0522
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_darwin.mm
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/desktop_capture/mac/screen_capturer_mac.h"
+
+namespace webrtc {
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
+    const DesktopCaptureOptions& options) {
+  if (!options.configuration_monitor()) {
+    return nullptr;
+  }
+
+  std::unique_ptr<ScreenCapturerMac> capturer(new ScreenCapturerMac(
+      options.configuration_monitor(), options.detect_updated_region(), options.allow_iosurface()));
+  if (!capturer.get()->Init()) {
+    return nullptr;
+  }
+
+  return capturer;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_fuchsia.cc b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_fuchsia.cc
new file mode 100644
index 0000000000..19c4c735ad
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_fuchsia.cc
@@ -0,0 +1,417 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/screen_capturer_fuchsia.h"
+
+#include <fuchsia/sysmem/cpp/fidl.h>
+#include <fuchsia/ui/composition/cpp/fidl.h>
+#include <fuchsia/ui/scenic/cpp/fidl.h>
+#include <lib/sys/cpp/component_context.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/fallback_desktop_capturer_wrapper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/divide_round.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+static constexpr uint32_t kMinBufferCount = 2;
+static constexpr uint32_t kFuchsiaBytesPerPixel = 4;
+static constexpr DesktopCapturer::SourceId kFuchsiaScreenId = 1;
+// 500 milliseconds
+static constexpr zx::duration kEventDelay = zx::msec(500);
+static constexpr fuchsia::sysmem::ColorSpaceType kSRGBColorSpace =
+    fuchsia::sysmem::ColorSpaceType::SRGB;
+static constexpr fuchsia::sysmem::PixelFormatType kBGRA32PixelFormatType =
+    fuchsia::sysmem::PixelFormatType::BGRA32;
+
+// Round |value| up to the closest multiple of |multiple|
+size_t RoundUpToMultiple(size_t value, size_t multiple) {
+  return DivideRoundUp(value, multiple) * multiple;
+}
+
+}  // namespace
+
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
+    const DesktopCaptureOptions& options) {
+  if (ScreenCapturerFuchsia::CheckRequirements()) {
+    std::unique_ptr<ScreenCapturerFuchsia> capturer(
+        new ScreenCapturerFuchsia());
+    return capturer;
+  }
+  return nullptr;
+}
+
+ScreenCapturerFuchsia::ScreenCapturerFuchsia()
+    : component_context_(sys::ComponentContext::Create()) {
+  RTC_DCHECK(CheckRequirements());
+}
+
+ScreenCapturerFuchsia::~ScreenCapturerFuchsia() {
+  // unmap virtual memory mapped pointers
+  uint32_t virt_mem_bytes =
+      buffer_collection_info_.settings.buffer_settings.size_bytes;
+  for (uint32_t buffer_index = 0;
+       buffer_index < buffer_collection_info_.buffer_count; buffer_index++) {
+    uintptr_t address =
+        reinterpret_cast<uintptr_t>(virtual_memory_mapped_addrs_[buffer_index]);
+    zx_status_t status = zx::vmar::root_self()->unmap(address, virt_mem_bytes);
+    RTC_DCHECK(status == ZX_OK);
+  }
+}
+
+// TODO(fxbug.dev/100303): Remove this function when Flatland is the only API.
+bool ScreenCapturerFuchsia::CheckRequirements() {
+  std::unique_ptr<sys::ComponentContext> component_context =
+      sys::ComponentContext::Create();
+  fuchsia::ui::scenic::ScenicSyncPtr scenic;
+  zx_status_t status = component_context->svc()->Connect(scenic.NewRequest());
+  if (status != ZX_OK) {
+    RTC_LOG(LS_ERROR) << "Failed to connect to Scenic: " << status;
+    return false;
+  }
+
+  bool scenic_uses_flatland = false;
+  scenic->UsesFlatland(&scenic_uses_flatland);
+  if (!scenic_uses_flatland) {
+    RTC_LOG(LS_ERROR) << "Screen capture not supported without Flatland.";
+  }
+
+  return scenic_uses_flatland;
+}
+
+void ScreenCapturerFuchsia::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+  callback_ = callback;
+
+  fatal_error_ = false;
+
+  SetupBuffers();
+}
+
+void ScreenCapturerFuchsia::CaptureFrame() {
+  if (fatal_error_) {
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+  zx::event event;
+  zx::event dup;
+  zx_status_t status = zx::event::create(0, &event);
+  if (status != ZX_OK) {
+    RTC_LOG(LS_ERROR) << "Failed to create event: " << status;
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+  event.duplicate(ZX_RIGHT_SAME_RIGHTS, &dup);
+
+  fuchsia::ui::composition::GetNextFrameArgs next_frame_args;
+  next_frame_args.set_event(std::move(dup));
+
+  fuchsia::ui::composition::ScreenCapture_GetNextFrame_Result result;
+  screen_capture_->GetNextFrame(std::move(next_frame_args), &result);
+  if (result.is_err()) {
+    RTC_LOG(LS_ERROR) << "fuchsia.ui.composition.GetNextFrame() failed: "
+                      << result.err() << "\n";
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+
+  status = event.wait_one(ZX_EVENT_SIGNALED, zx::deadline_after(kEventDelay),
+                          nullptr);
+  if (status != ZX_OK) {
+    RTC_LOG(LS_ERROR) << "Timed out waiting for ScreenCapture to render frame: "
+                      << status;
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+  uint32_t buffer_index = result.response().buffer_id();
+
+  // TODO(bugs.webrtc.org/14097): Use SharedMemoryDesktopFrame and
+  // ScreenCaptureFrameQueue
+  std::unique_ptr<BasicDesktopFrame> frame(
+      new BasicDesktopFrame(DesktopSize(width_, height_)));
+
+  uint32_t pixels_per_row = GetPixelsPerRow(
+      buffer_collection_info_.settings.image_format_constraints);
+  uint32_t stride = kFuchsiaBytesPerPixel * pixels_per_row;
+  frame->CopyPixelsFrom(virtual_memory_mapped_addrs_[buffer_index], stride,
+                        DesktopRect::MakeWH(width_, height_));
+  // Mark the whole screen as having been updated.
+  frame->mutable_updated_region()->SetRect(
+      DesktopRect::MakeWH(width_, height_));
+
+  fuchsia::ui::composition::ScreenCapture_ReleaseFrame_Result release_result;
+  screen_capture_->ReleaseFrame(buffer_index, &release_result);
+  if (release_result.is_err()) {
+    RTC_LOG(LS_ERROR) << "fuchsia.ui.composition.ReleaseFrame() failed: "
+                      << release_result.err();
+  }
+
+  int capture_time_ms = (rtc::TimeNanos() - capture_start_time_nanos) /
+                        rtc::kNumNanosecsPerMillisec;
+  frame->set_capture_time_ms(capture_time_ms);
+  callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
+}
+
+bool ScreenCapturerFuchsia::GetSourceList(SourceList* screens) {
+  RTC_DCHECK(screens->size() == 0);
+  // Fuchsia only supports single monitor display at this point
+  screens->push_back({kFuchsiaScreenId, std::string("Fuchsia monitor")});
+  return true;
+}
+
+bool ScreenCapturerFuchsia::SelectSource(SourceId id) {
+  if (id == kFuchsiaScreenId || id == kFullDesktopScreenId) {
+    return true;
+  }
+  return false;
+}
+
+fuchsia::sysmem::BufferCollectionConstraints
+ScreenCapturerFuchsia::GetBufferConstraints() {
+  fuchsia::sysmem::BufferCollectionConstraints constraints;
+  constraints.usage.cpu =
+      fuchsia::sysmem::cpuUsageRead | fuchsia::sysmem::cpuUsageWrite;
+  constraints.min_buffer_count = kMinBufferCount;
+
+  constraints.has_buffer_memory_constraints = true;
+  constraints.buffer_memory_constraints.ram_domain_supported = true;
+  constraints.buffer_memory_constraints.cpu_domain_supported = true;
+
+  constraints.image_format_constraints_count = 1;
+  fuchsia::sysmem::ImageFormatConstraints& image_constraints =
+      constraints.image_format_constraints[0];
+  image_constraints.color_spaces_count = 1;
+  image_constraints.color_space[0] =
+      fuchsia::sysmem::ColorSpace{.type = kSRGBColorSpace};
+  image_constraints.pixel_format.type = kBGRA32PixelFormatType;
+  image_constraints.pixel_format.has_format_modifier = true;
+  image_constraints.pixel_format.format_modifier.value =
+      fuchsia::sysmem::FORMAT_MODIFIER_LINEAR;
+
+  image_constraints.required_min_coded_width = width_;
+  image_constraints.required_min_coded_height = height_;
+  image_constraints.required_max_coded_width = width_;
+  image_constraints.required_max_coded_height = height_;
+
+  image_constraints.bytes_per_row_divisor = kFuchsiaBytesPerPixel;
+
+  return constraints;
+}
+
+void ScreenCapturerFuchsia::SetupBuffers() {
+  fuchsia::ui::scenic::ScenicSyncPtr scenic;
+  zx_status_t status = component_context_->svc()->Connect(scenic.NewRequest());
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "Failed to connect to Scenic: " << status;
+    return;
+  }
+
+  fuchsia::ui::gfx::DisplayInfo display_info;
+  status = scenic->GetDisplayInfo(&display_info);
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "Failed to connect to get display dimensions: "
+                      << status;
+    return;
+  }
+  width_ = display_info.width_in_px;
+  height_ = display_info.height_in_px;
+
+  status = component_context_->svc()->Connect(sysmem_allocator_.NewRequest());
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "Failed to connect to Sysmem Allocator: " << status;
+    return;
+  }
+
+  fuchsia::sysmem::BufferCollectionTokenSyncPtr sysmem_token;
+  status =
+      sysmem_allocator_->AllocateSharedCollection(sysmem_token.NewRequest());
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR)
+        << "fuchsia.sysmem.Allocator.AllocateSharedCollection() failed: "
+        << status;
+    return;
+  }
+
+  fuchsia::sysmem::BufferCollectionTokenSyncPtr flatland_token;
+  status = sysmem_token->Duplicate(ZX_RIGHT_SAME_RIGHTS,
+                                   flatland_token.NewRequest());
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR)
+        << "fuchsia.sysmem.BufferCollectionToken.Duplicate() failed: "
+        << status;
+    return;
+  }
+
+  status = sysmem_token->Sync();
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "fuchsia.sysmem.BufferCollectionToken.Sync() failed: "
+                      << status;
+    return;
+  }
+
+  status = sysmem_allocator_->BindSharedCollection(std::move(sysmem_token),
+                                                   collection_.NewRequest());
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR)
+        << "fuchsia.sysmem.Allocator.BindSharedCollection() failed: " << status;
+    return;
+  }
+
+  status = collection_->SetConstraints(/*has_constraints=*/true,
+                                       GetBufferConstraints());
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR)
+        << "fuchsia.sysmem.BufferCollection.SetConstraints() failed: "
+        << status;
+    return;
+  }
+
+  fuchsia::ui::composition::BufferCollectionImportToken import_token;
+  fuchsia::ui::composition::BufferCollectionExportToken export_token;
+  status = zx::eventpair::create(0, &export_token.value, &import_token.value);
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR)
+        << "Failed to create BufferCollection import and export tokens: "
+        << status;
+    return;
+  }
+
+  status = component_context_->svc()->Connect(flatland_allocator_.NewRequest());
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "Failed to connect to Flatland Allocator: " << status;
+    return;
+  }
+
+  fuchsia::ui::composition::RegisterBufferCollectionArgs buffer_collection_args;
+  buffer_collection_args.set_export_token(std::move(export_token));
+  buffer_collection_args.set_buffer_collection_token(std::move(flatland_token));
+  buffer_collection_args.set_usage(
+      fuchsia::ui::composition::RegisterBufferCollectionUsage::SCREENSHOT);
+
+  fuchsia::ui::composition::Allocator_RegisterBufferCollection_Result
+      buffer_collection_result;
+  flatland_allocator_->RegisterBufferCollection(
+      std::move(buffer_collection_args), &buffer_collection_result);
+  if (buffer_collection_result.is_err()) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "fuchsia.ui.composition.Allocator."
+                         "RegisterBufferCollection() failed.";
+    return;
+  }
+
+  zx_status_t allocation_status;
+  status = collection_->WaitForBuffersAllocated(&allocation_status,
+                                                &buffer_collection_info_);
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "Failed to wait for buffer collection info: "
+                      << status;
+    return;
+  }
+  if (allocation_status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "Failed to allocate buffer collection: " << status;
+    return;
+  }
+  status = collection_->Close();
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "Failed to close buffer collection token: " << status;
+    return;
+  }
+
+  status = component_context_->svc()->Connect(screen_capture_.NewRequest());
+  if (status != ZX_OK) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR) << "Failed to connect to Screen Capture: " << status;
+    return;
+  }
+
+  // Configure buffers in ScreenCapture client.
+  fuchsia::ui::composition::ScreenCaptureConfig configure_args;
+  configure_args.set_import_token(std::move(import_token));
+  configure_args.set_buffer_count(buffer_collection_info_.buffer_count);
+  configure_args.set_size({width_, height_});
+
+  fuchsia::ui::composition::ScreenCapture_Configure_Result configure_result;
+  screen_capture_->Configure(std::move(configure_args), &configure_result);
+  if (configure_result.is_err()) {
+    fatal_error_ = true;
+    RTC_LOG(LS_ERROR)
+        << "fuchsia.ui.composition.ScreenCapture.Configure() failed: "
+        << configure_result.err();
+    return;
+  }
+
+  // We have a collection of virtual memory objects which the ScreenCapture
+  // client will write the frame data to when requested. We map each of these
+  // onto a pointer stored in virtual_memory_mapped_addrs_ which we can use to
+  // access this data.
+  uint32_t virt_mem_bytes =
+      buffer_collection_info_.settings.buffer_settings.size_bytes;
+  RTC_DCHECK(virt_mem_bytes > 0);
+  for (uint32_t buffer_index = 0;
+       buffer_index < buffer_collection_info_.buffer_count; buffer_index++) {
+    const zx::vmo& virt_mem = buffer_collection_info_.buffers[buffer_index].vmo;
+    virtual_memory_mapped_addrs_[buffer_index] = nullptr;
+    auto status = zx::vmar::root_self()->map(
+        ZX_VM_PERM_READ, /*vmar_offset*/ 0, virt_mem,
+        /*vmo_offset*/ 0, virt_mem_bytes,
+        reinterpret_cast<uintptr_t*>(
+            &virtual_memory_mapped_addrs_[buffer_index]));
+    if (status != ZX_OK) {
+      fatal_error_ = true;
+      RTC_LOG(LS_ERROR) << "Failed to map virtual memory: " << status;
+      return;
+    }
+  }
+}
+
+uint32_t ScreenCapturerFuchsia::GetPixelsPerRow(
+    const fuchsia::sysmem::ImageFormatConstraints& constraints) {
+  uint32_t stride = RoundUpToMultiple(
+      std::max(constraints.min_bytes_per_row, width_ * kFuchsiaBytesPerPixel),
+      constraints.bytes_per_row_divisor);
+  uint32_t pixels_per_row = stride / kFuchsiaBytesPerPixel;
+
+  return pixels_per_row;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_fuchsia.h b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_fuchsia.h
new file mode 100644
index 0000000000..444930963f
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_fuchsia.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURER_FUCHSIA_H_
+#define MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURER_FUCHSIA_H_
+
+#include <fuchsia/sysmem/cpp/fidl.h>
+#include <fuchsia/ui/composition/cpp/fidl.h>
+#include <lib/sys/cpp/component_context.h>
+
+#include <memory>
+#include <unordered_map>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+
+namespace webrtc {
+
+class ScreenCapturerFuchsia final : public DesktopCapturer {
+ public:
+  ScreenCapturerFuchsia();
+  ~ScreenCapturerFuchsia() override;
+
+  static bool CheckRequirements();
+
+  // DesktopCapturer interface.
+  void Start(Callback* callback) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* screens) override;
+  bool SelectSource(SourceId id) override;
+
+ private:
+  fuchsia::sysmem::BufferCollectionConstraints GetBufferConstraints();
+  void SetupBuffers();
+  uint32_t GetPixelsPerRow(
+      const fuchsia::sysmem::ImageFormatConstraints& constraints);
+
+  Callback* callback_ = nullptr;
+
+  std::unique_ptr<sys::ComponentContext> component_context_;
+  fuchsia::sysmem::AllocatorSyncPtr sysmem_allocator_;
+  fuchsia::ui::composition::AllocatorSyncPtr flatland_allocator_;
+  fuchsia::ui::composition::ScreenCaptureSyncPtr screen_capture_;
+  fuchsia::sysmem::BufferCollectionSyncPtr collection_;
+  fuchsia::sysmem::BufferCollectionInfo_2 buffer_collection_info_;
+  std::unordered_map<uint32_t, uint8_t*> virtual_memory_mapped_addrs_;
+
+  bool fatal_error_;
+
+  // Dimensions of the screen we are capturing
+  uint32_t width_;
+  uint32_t height_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURER_FUCHSIA_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper.cc b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper.cc
new file mode 100644
index 0000000000..f8261a90b0
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper.cc
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/screen_capturer_helper.h"
+
+
+namespace webrtc {
+
+void ScreenCapturerHelper::ClearInvalidRegion() {
+  MutexLock scoped_invalid_region_lock(&invalid_region_mutex_);
+  invalid_region_.Clear();
+}
+
+void ScreenCapturerHelper::InvalidateRegion(
+    const DesktopRegion& invalid_region) {
+  MutexLock scoped_invalid_region_lock(&invalid_region_mutex_);
+  invalid_region_.AddRegion(invalid_region);
+}
+
+void ScreenCapturerHelper::InvalidateScreen(const DesktopSize& size) {
+  MutexLock scoped_invalid_region_lock(&invalid_region_mutex_);
+  invalid_region_.AddRect(DesktopRect::MakeSize(size));
+}
+
+void ScreenCapturerHelper::TakeInvalidRegion(DesktopRegion* invalid_region) {
+  invalid_region->Clear();
+
+  {
+    MutexLock scoped_invalid_region_lock(&invalid_region_mutex_);
+    invalid_region->Swap(&invalid_region_);
+  }
+
+  if (log_grid_size_ > 0) {
+    DesktopRegion expanded_region;
+    ExpandToGrid(*invalid_region, log_grid_size_, &expanded_region);
+    expanded_region.Swap(invalid_region);
+
+    invalid_region->IntersectWith(DesktopRect::MakeSize(size_most_recent_));
+  }
+}
+
+void ScreenCapturerHelper::SetLogGridSize(int log_grid_size) {
+  log_grid_size_ = log_grid_size;
+}
+
+const DesktopSize& ScreenCapturerHelper::size_most_recent() const {
+  return size_most_recent_;
+}
+
+void ScreenCapturerHelper::set_size_most_recent(const DesktopSize& size) {
+  size_most_recent_ = size;
+}
+
+// Returns the largest multiple of `n` that is <= `x`.
+// `n` must be a power of 2. `nMask` is ~(`n` - 1).
+static int DownToMultiple(int x, int nMask) {
+  return (x & nMask);
+}
+
+// Returns the smallest multiple of `n` that is >= `x`.
+// `n` must be a power of 2. `nMask` is ~(`n` - 1).
+static int UpToMultiple(int x, int n, int nMask) {
+  return ((x + n - 1) & nMask);
+}
+
+void ScreenCapturerHelper::ExpandToGrid(const DesktopRegion& region,
+                                        int log_grid_size,
+                                        DesktopRegion* result) {
+  RTC_DCHECK_GE(log_grid_size, 1);
+  int grid_size = 1 << log_grid_size;
+  int grid_size_mask = ~(grid_size - 1);
+
+  result->Clear();
+  for (DesktopRegion::Iterator it(region); !it.IsAtEnd(); it.Advance()) {
+    int left = DownToMultiple(it.rect().left(), grid_size_mask);
+    int right = UpToMultiple(it.rect().right(), grid_size, grid_size_mask);
+    int top = DownToMultiple(it.rect().top(), grid_size_mask);
+    int bottom = UpToMultiple(it.rect().bottom(), grid_size, grid_size_mask);
+    result->AddRect(DesktopRect::MakeLTRB(left, top, right, bottom));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper.h b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper.h
new file mode 100644
index 0000000000..cd7fa689c0
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURER_HELPER_H_
+#define MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURER_HELPER_H_
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// ScreenCapturerHelper is intended to be used by an implementation of the
+// ScreenCapturer interface. It maintains a thread-safe invalid region, and
+// the size of the most recently captured screen, on behalf of the
+// ScreenCapturer that owns it.
+class ScreenCapturerHelper {
+ public:
+  ScreenCapturerHelper() = default;
+  ~ScreenCapturerHelper() = default;
+
+  ScreenCapturerHelper(const ScreenCapturerHelper&) = delete;
+  ScreenCapturerHelper& operator=(const ScreenCapturerHelper&) = delete;
+
+  // Clear out the invalid region.
+  void ClearInvalidRegion();
+
+  // Invalidate the specified region.
+  void InvalidateRegion(const DesktopRegion& invalid_region);
+
+  // Invalidate the entire screen, of a given size.
+  void InvalidateScreen(const DesktopSize& size);
+
+  // Copies current invalid region to `invalid_region` clears invalid region
+  // storage for the next frame.
+  void TakeInvalidRegion(DesktopRegion* invalid_region);
+
+  // Access the size of the most recently captured screen.
+  const DesktopSize& size_most_recent() const;
+  void set_size_most_recent(const DesktopSize& size);
+
+  // Lossy compression can result in color values leaking between pixels in one
+  // block. If part of a block changes, then unchanged parts of that block can
+  // be changed in the compressed output. So we need to re-render an entire
+  // block whenever part of the block changes.
+  //
+  // If `log_grid_size` is >= 1, then this function makes TakeInvalidRegion()
+  // produce an invalid region expanded so that its vertices lie on a grid of
+  // size 2 ^ `log_grid_size`. The expanded region is then clipped to the size
+  // of the most recently captured screen, as previously set by
+  // set_size_most_recent().
+  // If `log_grid_size` is <= 0, then the invalid region is not expanded.
+  void SetLogGridSize(int log_grid_size);
+
+  // Expands a region so that its vertices all lie on a grid.
+  // The grid size must be >= 2, so `log_grid_size` must be >= 1.
+  static void ExpandToGrid(const DesktopRegion& region,
+                           int log_grid_size,
+                           DesktopRegion* result);
+
+ private:
+  // A region that has been manually invalidated (through InvalidateRegion).
+  // These will be returned as dirty_region in the capture data during the next
+  // capture.
+  DesktopRegion invalid_region_ RTC_GUARDED_BY(invalid_region_mutex_);
+
+  // A lock protecting `invalid_region_` across threads.
+  Mutex invalid_region_mutex_;
+
+  // The size of the most recently captured screen.
+  DesktopSize size_most_recent_;
+
+  // The log (base 2) of the size of the grid to which the invalid region is
+  // expanded.
+  // If the value is <= 0, then the invalid region is not expanded to a grid.
+  int log_grid_size_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_SCREEN_CAPTURER_HELPER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper_unittest.cc
new file mode 100644
index 0000000000..165bbe42de
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_helper_unittest.cc
@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/screen_capturer_helper.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class ScreenCapturerHelperTest : public ::testing::Test {
+ protected:
+  ScreenCapturerHelper capturer_helper_;
+};
+
+TEST_F(ScreenCapturerHelperTest, ClearInvalidRegion) {
+  DesktopRegion region(DesktopRect::MakeXYWH(1, 2, 3, 4));
+  capturer_helper_.InvalidateRegion(region);
+  capturer_helper_.ClearInvalidRegion();
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(region.is_empty());
+}
+
+TEST_F(ScreenCapturerHelperTest, InvalidateRegion) {
+  DesktopRegion region;
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(region.is_empty());
+
+  region.SetRect(DesktopRect::MakeXYWH(1, 2, 3, 4));
+  capturer_helper_.InvalidateRegion(region);
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(DesktopRegion(DesktopRect::MakeXYWH(1, 2, 3, 4)).Equals(region));
+
+  capturer_helper_.InvalidateRegion(
+      DesktopRegion(DesktopRect::MakeXYWH(1, 2, 3, 4)));
+  capturer_helper_.InvalidateRegion(
+      DesktopRegion(DesktopRect::MakeXYWH(4, 2, 3, 4)));
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(DesktopRegion(DesktopRect::MakeXYWH(1, 2, 6, 4)).Equals(region));
+}
+
+TEST_F(ScreenCapturerHelperTest, InvalidateScreen) {
+  DesktopRegion region;
+  capturer_helper_.InvalidateScreen(DesktopSize(12, 34));
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(DesktopRegion(DesktopRect::MakeWH(12, 34)).Equals(region));
+}
+
+TEST_F(ScreenCapturerHelperTest, SizeMostRecent) {
+  EXPECT_TRUE(capturer_helper_.size_most_recent().is_empty());
+  capturer_helper_.set_size_most_recent(DesktopSize(12, 34));
+  EXPECT_TRUE(DesktopSize(12, 34).equals(capturer_helper_.size_most_recent()));
+}
+
+TEST_F(ScreenCapturerHelperTest, SetLogGridSize) {
+  capturer_helper_.set_size_most_recent(DesktopSize(10, 10));
+
+  DesktopRegion region;
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(DesktopRegion().Equals(region));
+
+  capturer_helper_.InvalidateRegion(
+      DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)));
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)).Equals(region));
+
+  capturer_helper_.SetLogGridSize(-1);
+  capturer_helper_.InvalidateRegion(
+      DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)));
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)).Equals(region));
+
+  capturer_helper_.SetLogGridSize(0);
+  capturer_helper_.InvalidateRegion(
+      DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)));
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)).Equals(region));
+
+  capturer_helper_.SetLogGridSize(1);
+  capturer_helper_.InvalidateRegion(
+      DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)));
+  capturer_helper_.TakeInvalidRegion(&region);
+
+  EXPECT_TRUE(DesktopRegion(DesktopRect::MakeXYWH(6, 6, 2, 2)).Equals(region));
+
+  capturer_helper_.SetLogGridSize(2);
+  capturer_helper_.InvalidateRegion(
+      DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)));
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(DesktopRegion(DesktopRect::MakeXYWH(4, 4, 4, 4)).Equals(region));
+
+  capturer_helper_.SetLogGridSize(0);
+  capturer_helper_.InvalidateRegion(
+      DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)));
+  capturer_helper_.TakeInvalidRegion(&region);
+  EXPECT_TRUE(DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)).Equals(region));
+}
+
+void TestExpandRegionToGrid(const DesktopRegion& region,
+                            int log_grid_size,
+                            const DesktopRegion& expanded_region_expected) {
+  DesktopRegion expanded_region1;
+  ScreenCapturerHelper::ExpandToGrid(region, log_grid_size, &expanded_region1);
+  EXPECT_TRUE(expanded_region_expected.Equals(expanded_region1));
+
+  DesktopRegion expanded_region2;
+  ScreenCapturerHelper::ExpandToGrid(expanded_region1, log_grid_size,
+                                     &expanded_region2);
+  EXPECT_TRUE(expanded_region1.Equals(expanded_region2));
+}
+
+void TestExpandRectToGrid(int l,
+                          int t,
+                          int r,
+                          int b,
+                          int log_grid_size,
+                          int lExpanded,
+                          int tExpanded,
+                          int rExpanded,
+                          int bExpanded) {
+  TestExpandRegionToGrid(DesktopRegion(DesktopRect::MakeLTRB(l, t, r, b)),
+                         log_grid_size,
+                         DesktopRegion(DesktopRect::MakeLTRB(
+                             lExpanded, tExpanded, rExpanded, bExpanded)));
+}
+
+TEST_F(ScreenCapturerHelperTest, ExpandToGrid) {
+  const int kLogGridSize = 4;
+  const int kGridSize = 1 << kLogGridSize;
+  for (int i = -2; i <= 2; i++) {
+    int x = i * kGridSize;
+    for (int j = -2; j <= 2; j++) {
+      int y = j * kGridSize;
+      TestExpandRectToGrid(x + 0, y + 0, x + 1, y + 1, kLogGridSize, x + 0,
+                           y + 0, x + kGridSize, y + kGridSize);
+      TestExpandRectToGrid(x + 0, y + kGridSize - 1, x + 1, y + kGridSize,
+                           kLogGridSize, x + 0, y + 0, x + kGridSize,
+                           y + kGridSize);
+      TestExpandRectToGrid(x + kGridSize - 1, y + kGridSize - 1, x + kGridSize,
+                           y + kGridSize, kLogGridSize, x + 0, y + 0,
+                           x + kGridSize, y + kGridSize);
+      TestExpandRectToGrid(x + kGridSize - 1, y + 0, x + kGridSize, y + 1,
+                           kLogGridSize, x + 0, y + 0, x + kGridSize,
+                           y + kGridSize);
+      TestExpandRectToGrid(x - 1, y + 0, x + 1, y + 1, kLogGridSize,
+                           x - kGridSize, y + 0, x + kGridSize, y + kGridSize);
+      TestExpandRectToGrid(x - 1, y - 1, x + 1, y + 0, kLogGridSize,
+                           x - kGridSize, y - kGridSize, x + kGridSize, y);
+      TestExpandRectToGrid(x + 0, y - 1, x + 1, y + 1, kLogGridSize, x,
+                           y - kGridSize, x + kGridSize, y + kGridSize);
+      TestExpandRectToGrid(x - 1, y - 1, x + 0, y + 1, kLogGridSize,
+                           x - kGridSize, y - kGridSize, x, y + kGridSize);
+
+      // Construct a region consisting of 3 pixels and verify that it's expanded
+      // properly to 3 squares that are kGridSize by kGridSize.
+      for (int q = 0; q < 4; ++q) {
+        DesktopRegion region;
+        DesktopRegion expanded_region_expected;
+
+        if (q != 0) {
+          region.AddRect(DesktopRect::MakeXYWH(x - 1, y - 1, 1, 1));
+          expanded_region_expected.AddRect(DesktopRect::MakeXYWH(
+              x - kGridSize, y - kGridSize, kGridSize, kGridSize));
+        }
+        if (q != 1) {
+          region.AddRect(DesktopRect::MakeXYWH(x, y - 1, 1, 1));
+          expanded_region_expected.AddRect(
+              DesktopRect::MakeXYWH(x, y - kGridSize, kGridSize, kGridSize));
+        }
+        if (q != 2) {
+          region.AddRect(DesktopRect::MakeXYWH(x - 1, y, 1, 1));
+          expanded_region_expected.AddRect(
+              DesktopRect::MakeXYWH(x - kGridSize, y, kGridSize, kGridSize));
+        }
+        if (q != 3) {
+          region.AddRect(DesktopRect::MakeXYWH(x, y, 1, 1));
+          expanded_region_expected.AddRect(
+              DesktopRect::MakeXYWH(x, y, kGridSize, kGridSize));
+        }
+
+        TestExpandRegionToGrid(region, kLogGridSize, expanded_region_expected);
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_integration_test.cc b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_integration_test.cc
new file mode 100644
index 0000000000..b33427ad42
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_integration_test.cc
@@ -0,0 +1,380 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include <algorithm>
+#include <initializer_list>
+#include <iostream>  // TODO(zijiehe): Remove once flaky has been resolved.
+#include <memory>
+#include <utility>
+
+// TODO(zijiehe): Remove once flaky has been resolved.
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/mock_desktop_capturer_callback.h"
+#include "modules/desktop_capture/rgba_color.h"
+#include "modules/desktop_capture/screen_drawer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/third_party/base64/base64.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+#if defined(WEBRTC_WIN)
+#include "modules/desktop_capture/win/screen_capturer_win_directx.h"
+#include "rtc_base/win/windows_version.h"
+#endif  // defined(WEBRTC_WIN)
+
+using ::testing::_;
+
+namespace webrtc {
+
+namespace {
+
+ACTION_P2(SaveCaptureResult, result, dest) {
+  *result = arg0;
+  *dest = std::move(*arg1);
+}
+
+// Returns true if color in `rect` of `frame` is `color`.
+bool ArePixelsColoredBy(const DesktopFrame& frame,
+                        DesktopRect rect,
+                        RgbaColor color,
+                        bool may_partially_draw) {
+  if (!may_partially_draw) {
+    // updated_region() should cover the painted area.
+    DesktopRegion updated_region(frame.updated_region());
+    updated_region.IntersectWith(rect);
+    if (!updated_region.Equals(DesktopRegion(rect))) {
+      return false;
+    }
+  }
+
+  // Color in the `rect` should be `color`.
+  uint8_t* row = frame.GetFrameDataAtPos(rect.top_left());
+  for (int i = 0; i < rect.height(); i++) {
+    uint8_t* column = row;
+    for (int j = 0; j < rect.width(); j++) {
+      if (color != RgbaColor(column)) {
+        return false;
+      }
+      column += DesktopFrame::kBytesPerPixel;
+    }
+    row += frame.stride();
+  }
+  return true;
+}
+
+}  // namespace
+
+class ScreenCapturerIntegrationTest : public ::testing::Test {
+ public:
+  void SetUp() override {
+    capturer_ = DesktopCapturer::CreateScreenCapturer(
+        DesktopCaptureOptions::CreateDefault());
+  }
+
+ protected:
+  void TestCaptureUpdatedRegion(
+      std::initializer_list<DesktopCapturer*> capturers) {
+    RTC_DCHECK(capturers.size() > 0);
+// A large enough area for the tests, which should be able to be fulfilled
+// by most systems.
+#if defined(WEBRTC_WIN)
+    // On Windows, an interesting warning window may pop up randomly. The root
+    // cause is still under investigation, so reduce the test area to work
+    // around. Bug https://bugs.chromium.org/p/webrtc/issues/detail?id=6666.
+    const int kTestArea = 416;
+#else
+    const int kTestArea = 512;
+#endif
+    const int kRectSize = 32;
+    std::unique_ptr<ScreenDrawer> drawer = ScreenDrawer::Create();
+    if (!drawer || drawer->DrawableRegion().is_empty()) {
+      RTC_LOG(LS_WARNING)
+          << "No ScreenDrawer implementation for current platform.";
+      return;
+    }
+    if (drawer->DrawableRegion().width() < kTestArea ||
+        drawer->DrawableRegion().height() < kTestArea) {
+      RTC_LOG(LS_WARNING)
+          << "ScreenDrawer::DrawableRegion() is too small for the "
+             "CaptureUpdatedRegion tests.";
+      return;
+    }
+
+    for (DesktopCapturer* capturer : capturers) {
+      capturer->Start(&callback_);
+    }
+
+    // Draw a set of `kRectSize` by `kRectSize` rectangles at (`i`, `i`), or
+    // `i` by `i` rectangles at (`kRectSize`, `kRectSize`). One of (controlled
+    // by `c`) its primary colors is `i`, and the other two are 0x7f. So we
+    // won't draw a black or white rectangle.
+    for (int c = 0; c < 3; c++) {
+      // A fixed size rectangle.
+      for (int i = 0; i < kTestArea - kRectSize; i += 16) {
+        DesktopRect rect = DesktopRect::MakeXYWH(i, i, kRectSize, kRectSize);
+        rect.Translate(drawer->DrawableRegion().top_left());
+        RgbaColor color((c == 0 ? (i & 0xff) : 0x7f),
+                        (c == 1 ? (i & 0xff) : 0x7f),
+                        (c == 2 ? (i & 0xff) : 0x7f));
+        // Fail fast.
+        ASSERT_NO_FATAL_FAILURE(
+            TestCaptureOneFrame(capturers, drawer.get(), rect, color));
+      }
+
+      // A variable-size rectangle.
+      for (int i = 0; i < kTestArea - kRectSize; i += 16) {
+        DesktopRect rect = DesktopRect::MakeXYWH(kRectSize, kRectSize, i, i);
+        rect.Translate(drawer->DrawableRegion().top_left());
+        RgbaColor color((c == 0 ? (i & 0xff) : 0x7f),
+                        (c == 1 ? (i & 0xff) : 0x7f),
+                        (c == 2 ? (i & 0xff) : 0x7f));
+        // Fail fast.
+        ASSERT_NO_FATAL_FAILURE(
+            TestCaptureOneFrame(capturers, drawer.get(), rect, color));
+      }
+    }
+  }
+
+  void TestCaptureUpdatedRegion() {
+    TestCaptureUpdatedRegion({capturer_.get()});
+  }
+
+#if defined(WEBRTC_WIN)
+  // Enable allow_directx_capturer in DesktopCaptureOptions, but let
+  // DesktopCapturer::CreateScreenCapturer() to decide whether a DirectX
+  // capturer should be used.
+  void MaybeCreateDirectxCapturer() {
+    DesktopCaptureOptions options(DesktopCaptureOptions::CreateDefault());
+    options.set_allow_directx_capturer(true);
+    capturer_ = DesktopCapturer::CreateScreenCapturer(options);
+  }
+
+  bool CreateDirectxCapturer() {
+    if (!ScreenCapturerWinDirectx::IsSupported()) {
+      RTC_LOG(LS_WARNING) << "Directx capturer is not supported";
+      return false;
+    }
+
+    MaybeCreateDirectxCapturer();
+    return true;
+  }
+
+  void CreateMagnifierCapturer() {
+    DesktopCaptureOptions options(DesktopCaptureOptions::CreateDefault());
+    options.set_allow_use_magnification_api(true);
+    capturer_ = DesktopCapturer::CreateScreenCapturer(options);
+  }
+#endif  // defined(WEBRTC_WIN)
+
+  std::unique_ptr<DesktopCapturer> capturer_;
+  MockDesktopCapturerCallback callback_;
+
+ private:
+  // Repeats capturing the frame by using `capturers` one-by-one for 600 times,
+  // typically 30 seconds, until they succeeded captured a `color` rectangle at
+  // `rect`. This function uses `drawer`->WaitForPendingDraws() between two
+  // attempts to wait for the screen to update.
+  void TestCaptureOneFrame(std::vector<DesktopCapturer*> capturers,
+                           ScreenDrawer* drawer,
+                           DesktopRect rect,
+                           RgbaColor color) {
+    const int wait_capture_round = 600;
+    drawer->Clear();
+    size_t succeeded_capturers = 0;
+    for (int i = 0; i < wait_capture_round; i++) {
+      drawer->DrawRectangle(rect, color);
+      drawer->WaitForPendingDraws();
+      for (size_t j = 0; j < capturers.size(); j++) {
+        if (capturers[j] == nullptr) {
+          // DesktopCapturer should return an empty updated_region() if no
+          // update detected. So we won't test it again if it has captured the
+          // rectangle we drew.
+          continue;
+        }
+        std::unique_ptr<DesktopFrame> frame = CaptureFrame(capturers[j]);
+        if (!frame) {
+          // CaptureFrame() has triggered an assertion failure already, we only
+          // need to return here.
+          return;
+        }
+
+        if (ArePixelsColoredBy(*frame, rect, color,
+                               drawer->MayDrawIncompleteShapes())) {
+          capturers[j] = nullptr;
+          succeeded_capturers++;
+        }
+        // The following else if statement is for debugging purpose only, which
+        // should be removed after flaky of ScreenCapturerIntegrationTest has
+        // been resolved.
+        else if (i == wait_capture_round - 1) {
+          std::string result;
+          rtc::Base64::EncodeFromArray(
+              frame->data(), frame->size().height() * frame->stride(), &result);
+          std::cout << frame->size().width() << " x " << frame->size().height()
+                    << std::endl;
+          // Split the entire string (can be over 4M) into several lines to
+          // avoid browser from sticking.
+          static const size_t kLineLength = 32768;
+          const char* result_end = result.c_str() + result.length();
+          for (const char* it = result.c_str(); it < result_end;
+               it += kLineLength) {
+            const size_t max_length = result_end - it;
+            std::cout << std::string(it, std::min(kLineLength, max_length))
+                      << std::endl;
+          }
+          std::cout << "Failed to capture rectangle " << rect.left() << " x "
+                    << rect.top() << " - " << rect.right() << " x "
+                    << rect.bottom() << " with color ("
+                    << static_cast<int>(color.red) << ", "
+                    << static_cast<int>(color.green) << ", "
+                    << static_cast<int>(color.blue) << ", "
+                    << static_cast<int>(color.alpha) << ")" << std::endl;
+          ASSERT_TRUE(false) << "ScreenCapturerIntegrationTest may be flaky. "
+                                "Please kindly FYI the broken link to "
+                                "zijiehe@chromium.org for investigation. If "
+                                "the failure continually happens, but I have "
+                                "not responded as quick as expected, disable "
+                                "*all* tests in "
+                                "screen_capturer_integration_test.cc to "
+                                "unblock other developers.";
+        }
+      }
+
+      if (succeeded_capturers == capturers.size()) {
+        break;
+      }
+    }
+
+    ASSERT_EQ(succeeded_capturers, capturers.size());
+  }
+
+  // Expects `capturer` to successfully capture a frame, and returns it.
+  std::unique_ptr<DesktopFrame> CaptureFrame(DesktopCapturer* capturer) {
+    for (int i = 0; i < 10; i++) {
+      std::unique_ptr<DesktopFrame> frame;
+      DesktopCapturer::Result result;
+      EXPECT_CALL(callback_, OnCaptureResultPtr(_, _))
+          .WillOnce(SaveCaptureResult(&result, &frame));
+      capturer->CaptureFrame();
+      ::testing::Mock::VerifyAndClearExpectations(&callback_);
+      if (result == DesktopCapturer::Result::SUCCESS) {
+        EXPECT_TRUE(frame);
+        return frame;
+      } else {
+        EXPECT_FALSE(frame);
+      }
+    }
+
+    EXPECT_TRUE(false);
+    return nullptr;
+  }
+};
+
+#if defined(WEBRTC_WIN)
+// ScreenCapturerWinGdi randomly returns blank screen, the root cause is still
+// unknown. Bug, https://bugs.chromium.org/p/webrtc/issues/detail?id=6843.
+#define MAYBE_CaptureUpdatedRegion DISABLED_CaptureUpdatedRegion
+#else
+#define MAYBE_CaptureUpdatedRegion CaptureUpdatedRegion
+#endif
+TEST_F(ScreenCapturerIntegrationTest, MAYBE_CaptureUpdatedRegion) {
+  TestCaptureUpdatedRegion();
+}
+
+#if defined(WEBRTC_WIN)
+// ScreenCapturerWinGdi randomly returns blank screen, the root cause is still
+// unknown. Bug, https://bugs.chromium.org/p/webrtc/issues/detail?id=6843.
+#define MAYBE_TwoCapturers DISABLED_TwoCapturers
+#else
+#define MAYBE_TwoCapturers TwoCapturers
+#endif
+TEST_F(ScreenCapturerIntegrationTest, MAYBE_TwoCapturers) {
+  std::unique_ptr<DesktopCapturer> capturer2 = std::move(capturer_);
+  SetUp();
+  TestCaptureUpdatedRegion({capturer_.get(), capturer2.get()});
+}
+
+#if defined(WEBRTC_WIN)
+
+// Windows cannot capture contents on VMs hosted in GCE. See bug
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=8153.
+TEST_F(ScreenCapturerIntegrationTest,
+       DISABLED_CaptureUpdatedRegionWithDirectxCapturer) {
+  if (!CreateDirectxCapturer()) {
+    return;
+  }
+
+  TestCaptureUpdatedRegion();
+}
+
+TEST_F(ScreenCapturerIntegrationTest, DISABLED_TwoDirectxCapturers) {
+  if (!CreateDirectxCapturer()) {
+    return;
+  }
+
+  std::unique_ptr<DesktopCapturer> capturer2 = std::move(capturer_);
+  RTC_CHECK(CreateDirectxCapturer());
+  TestCaptureUpdatedRegion({capturer_.get(), capturer2.get()});
+}
+
+TEST_F(ScreenCapturerIntegrationTest,
+       DISABLED_CaptureUpdatedRegionWithMagnifierCapturer) {
+  // On Windows 8 or later, magnifier APIs return a frame with a border on test
+  // environment, so disable these tests.
+  // Bug https://bugs.chromium.org/p/webrtc/issues/detail?id=6844
+  // TODO(zijiehe): Find the root cause of the border and failure, which cannot
+  // reproduce on my dev machine.
+  if (rtc::rtc_win::GetVersion() >= rtc::rtc_win::Version::VERSION_WIN8) {
+    return;
+  }
+  CreateMagnifierCapturer();
+  TestCaptureUpdatedRegion();
+}
+
+TEST_F(ScreenCapturerIntegrationTest, DISABLED_TwoMagnifierCapturers) {
+  // On Windows 8 or later, magnifier APIs return a frame with a border on test
+  // environment, so disable these tests.
+  // Bug https://bugs.chromium.org/p/webrtc/issues/detail?id=6844
+  // TODO(zijiehe): Find the root cause of the border and failure, which cannot
+  // reproduce on my dev machine.
+  if (rtc::rtc_win::GetVersion() >= rtc::rtc_win::Version::VERSION_WIN8) {
+    return;
+  }
+  CreateMagnifierCapturer();
+  std::unique_ptr<DesktopCapturer> capturer2 = std::move(capturer_);
+  CreateMagnifierCapturer();
+  TestCaptureUpdatedRegion({capturer_.get(), capturer2.get()});
+}
+
+TEST_F(ScreenCapturerIntegrationTest,
+       DISABLED_MaybeCaptureUpdatedRegionWithDirectxCapturer) {
+  if (rtc::rtc_win::GetVersion() < rtc::rtc_win::Version::VERSION_WIN8) {
+    // ScreenCapturerWinGdi randomly returns blank screen, the root cause is
+    // still unknown. Bug,
+    // https://bugs.chromium.org/p/webrtc/issues/detail?id=6843.
+    // On Windows 7 or early version, MaybeCreateDirectxCapturer() always
+    // creates GDI capturer.
+    return;
+  }
+  // Even DirectX capturer is not supported in current system, we should be able
+  // to select a usable capturer.
+  MaybeCreateDirectxCapturer();
+  TestCaptureUpdatedRegion();
+}
+
+#endif  // defined(WEBRTC_WIN)
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_linux.cc b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_linux.cc
new file mode 100644
index 0000000000..44993837e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_linux.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+
+#if defined(WEBRTC_USE_PIPEWIRE)
+#include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
+#endif  // defined(WEBRTC_USE_PIPEWIRE)
+
+#if defined(WEBRTC_USE_X11)
+#include "modules/desktop_capture/linux/x11/screen_capturer_x11.h"
+#endif  // defined(WEBRTC_USE_X11)
+
+namespace webrtc {
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
+    const DesktopCaptureOptions& options) {
+#if defined(WEBRTC_USE_PIPEWIRE)
+  if (options.allow_pipewire() && BaseCapturerPipeWire::IsSupported()) {
+    return std::make_unique<BaseCapturerPipeWire>(options,
+                                                  CaptureType::kScreen);
+  }
+#endif  // defined(WEBRTC_USE_PIPEWIRE)
+
+#if defined(WEBRTC_USE_X11)
+  if (!DesktopCapturer::IsRunningUnderWayland())
+    return ScreenCapturerX11::CreateRawScreenCapturer(options);
+#endif  // defined(WEBRTC_USE_X11)
+
+  return nullptr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc
new file mode 100644
index 0000000000..96e844066a
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <ApplicationServices/ApplicationServices.h>
+
+#include <memory>
+#include <ostream>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+#include "modules/desktop_capture/mock_desktop_capturer_callback.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::Return;
+
+namespace webrtc {
+
+class ScreenCapturerMacTest : public ::testing::Test {
+ public:
+  // Verifies that the whole screen is initially dirty.
+  void CaptureDoneCallback1(DesktopCapturer::Result result,
+                            std::unique_ptr<DesktopFrame>* frame);
+
+  // Verifies that a rectangle explicitly marked as dirty is propagated
+  // correctly.
+  void CaptureDoneCallback2(DesktopCapturer::Result result,
+                            std::unique_ptr<DesktopFrame>* frame);
+
+ protected:
+  void SetUp() override {
+    capturer_ = DesktopCapturer::CreateScreenCapturer(
+        DesktopCaptureOptions::CreateDefault());
+  }
+
+  std::unique_ptr<DesktopCapturer> capturer_;
+  MockDesktopCapturerCallback callback_;
+};
+
+void ScreenCapturerMacTest::CaptureDoneCallback1(
+    DesktopCapturer::Result result,
+    std::unique_ptr<DesktopFrame>* frame) {
+  EXPECT_EQ(result, DesktopCapturer::Result::SUCCESS);
+
+  MacDesktopConfiguration config = MacDesktopConfiguration::GetCurrent(
+      MacDesktopConfiguration::BottomLeftOrigin);
+
+  // Verify that the region contains full frame.
+  DesktopRegion::Iterator it((*frame)->updated_region());
+  EXPECT_TRUE(!it.IsAtEnd() && it.rect().equals(config.pixel_bounds));
+}
+
+void ScreenCapturerMacTest::CaptureDoneCallback2(
+    DesktopCapturer::Result result,
+    std::unique_ptr<DesktopFrame>* frame) {
+  EXPECT_EQ(result, DesktopCapturer::Result::SUCCESS);
+
+  MacDesktopConfiguration config = MacDesktopConfiguration::GetCurrent(
+      MacDesktopConfiguration::BottomLeftOrigin);
+  int width = config.pixel_bounds.width();
+  int height = config.pixel_bounds.height();
+
+  EXPECT_EQ(width, (*frame)->size().width());
+  EXPECT_EQ(height, (*frame)->size().height());
+  EXPECT_TRUE((*frame)->data() != NULL);
+  // Depending on the capture method, the screen may be flipped or not, so
+  // the stride may be positive or negative.
+  EXPECT_EQ(static_cast<int>(sizeof(uint32_t) * width),
+            abs((*frame)->stride()));
+}
+
+TEST_F(ScreenCapturerMacTest, Capture) {
+  EXPECT_CALL(callback_,
+              OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS, _))
+      .Times(2)
+      .WillOnce(Invoke(this, &ScreenCapturerMacTest::CaptureDoneCallback1))
+      .WillOnce(Invoke(this, &ScreenCapturerMacTest::CaptureDoneCallback2));
+
+  SCOPED_TRACE("");
+  capturer_->Start(&callback_);
+
+  // Check that we get an initial full-screen updated.
+  capturer_->CaptureFrame();
+
+  // Check that subsequent dirty rects are propagated correctly.
+  capturer_->CaptureFrame();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_null.cc b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_null.cc
new file mode 100644
index 0000000000..6b1ccb322e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_null.cc
@@ -0,0 +1,21 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_capturer.h"
+
+namespace webrtc {
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
+    const DesktopCaptureOptions& options) {
+  return nullptr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_unittest.cc
new file mode 100644
index 0000000000..8f5fe631f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_unittest.cc
@@ -0,0 +1,224 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/mock_desktop_capturer_callback.h"
+#include "rtc_base/logging.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+#if defined(WEBRTC_WIN)
+#include "modules/desktop_capture/win/screen_capturer_win_directx.h"
+#endif  // defined(WEBRTC_WIN)
+
+using ::testing::_;
+
+const int kTestSharedMemoryId = 123;
+
+namespace webrtc {
+
+class ScreenCapturerTest : public ::testing::Test {
+ public:
+  void SetUp() override {
+    capturer_ = DesktopCapturer::CreateScreenCapturer(
+        DesktopCaptureOptions::CreateDefault());
+    ASSERT_TRUE(capturer_);
+  }
+
+ protected:
+#if defined(WEBRTC_WIN)
+  // Enable allow_directx_capturer in DesktopCaptureOptions, but let
+  // DesktopCapturer::CreateScreenCapturer to decide whether a DirectX capturer
+  // should be used.
+  void MaybeCreateDirectxCapturer() {
+    DesktopCaptureOptions options(DesktopCaptureOptions::CreateDefault());
+    options.set_allow_directx_capturer(true);
+    capturer_ = DesktopCapturer::CreateScreenCapturer(options);
+  }
+
+  bool CreateDirectxCapturer() {
+    if (!ScreenCapturerWinDirectx::IsSupported()) {
+      RTC_LOG(LS_WARNING) << "Directx capturer is not supported";
+      return false;
+    }
+
+    MaybeCreateDirectxCapturer();
+    return true;
+  }
+
+  void CreateMagnifierCapturer() {
+    DesktopCaptureOptions options(DesktopCaptureOptions::CreateDefault());
+    options.set_allow_use_magnification_api(true);
+    capturer_ = DesktopCapturer::CreateScreenCapturer(options);
+  }
+#endif  // defined(WEBRTC_WIN)
+
+  std::unique_ptr<DesktopCapturer> capturer_;
+  MockDesktopCapturerCallback callback_;
+};
+
+class FakeSharedMemory : public SharedMemory {
+ public:
+  FakeSharedMemory(char* buffer, size_t size)
+      : SharedMemory(buffer, size, 0, kTestSharedMemoryId), buffer_(buffer) {}
+  ~FakeSharedMemory() override { delete[] buffer_; }
+
+  FakeSharedMemory(const FakeSharedMemory&) = delete;
+  FakeSharedMemory& operator=(const FakeSharedMemory&) = delete;
+
+ private:
+  char* buffer_;
+};
+
+class FakeSharedMemoryFactory : public SharedMemoryFactory {
+ public:
+  FakeSharedMemoryFactory() {}
+  ~FakeSharedMemoryFactory() override {}
+
+  FakeSharedMemoryFactory(const FakeSharedMemoryFactory&) = delete;
+  FakeSharedMemoryFactory& operator=(const FakeSharedMemoryFactory&) = delete;
+
+  std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) override {
+    return std::unique_ptr<SharedMemory>(
+        new FakeSharedMemory(new char[size], size));
+  }
+};
+
+ACTION_P(SaveUniquePtrArg, dest) {
+  *dest = std::move(*arg1);
+}
+
+// TODO(bugs.webrtc.org/12950): Re-enable when libc++ issue is fixed.
+#if defined(WEBRTC_LINUX) && defined(MEMORY_SANITIZER)
+#define MAYBE_GetScreenListAndSelectScreen DISABLED_GetScreenListAndSelectScreen
+#else
+#define MAYBE_GetScreenListAndSelectScreen GetScreenListAndSelectScreen
+#endif
+TEST_F(ScreenCapturerTest, MAYBE_GetScreenListAndSelectScreen) {
+  webrtc::DesktopCapturer::SourceList screens;
+  EXPECT_TRUE(capturer_->GetSourceList(&screens));
+  for (const auto& screen : screens) {
+    EXPECT_TRUE(capturer_->SelectSource(screen.id));
+  }
+}
+
+// Flaky on Linux. See: crbug.com/webrtc/7830
+#if defined(WEBRTC_LINUX)
+#define MAYBE_StartCapturer DISABLED_StartCaptuerer
+#else
+#define MAYBE_StartCapturer StartCapturer
+#endif
+TEST_F(ScreenCapturerTest, MAYBE_StartCapturer) {
+  capturer_->Start(&callback_);
+}
+
+#if defined(WEBRTC_LINUX)
+#define MAYBE_Capture DISABLED_Capture
+#else
+#define MAYBE_Capture Capture
+#endif
+TEST_F(ScreenCapturerTest, MAYBE_Capture) {
+  // Assume that Start() treats the screen as invalid initially.
+  std::unique_ptr<DesktopFrame> frame;
+  EXPECT_CALL(callback_,
+              OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS, _))
+      .WillOnce(SaveUniquePtrArg(&frame));
+
+  capturer_->Start(&callback_);
+  capturer_->CaptureFrame();
+
+  ASSERT_TRUE(frame);
+  EXPECT_GT(frame->size().width(), 0);
+  EXPECT_GT(frame->size().height(), 0);
+  EXPECT_GE(frame->stride(),
+            frame->size().width() * DesktopFrame::kBytesPerPixel);
+  EXPECT_TRUE(frame->shared_memory() == NULL);
+
+  // Verify that the region contains whole screen.
+  EXPECT_FALSE(frame->updated_region().is_empty());
+  DesktopRegion::Iterator it(frame->updated_region());
+  ASSERT_TRUE(!it.IsAtEnd());
+  EXPECT_TRUE(it.rect().equals(DesktopRect::MakeSize(frame->size())));
+  it.Advance();
+  EXPECT_TRUE(it.IsAtEnd());
+}
+
+#if defined(WEBRTC_WIN)
+
+TEST_F(ScreenCapturerTest, UseSharedBuffers) {
+  std::unique_ptr<DesktopFrame> frame;
+  EXPECT_CALL(callback_,
+              OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS, _))
+      .WillOnce(SaveUniquePtrArg(&frame));
+
+  capturer_->Start(&callback_);
+  capturer_->SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory>(new FakeSharedMemoryFactory()));
+  capturer_->CaptureFrame();
+
+  ASSERT_TRUE(frame);
+  ASSERT_TRUE(frame->shared_memory());
+  EXPECT_EQ(frame->shared_memory()->id(), kTestSharedMemoryId);
+}
+
+TEST_F(ScreenCapturerTest, UseMagnifier) {
+  CreateMagnifierCapturer();
+  std::unique_ptr<DesktopFrame> frame;
+  EXPECT_CALL(callback_,
+              OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS, _))
+      .WillOnce(SaveUniquePtrArg(&frame));
+
+  capturer_->Start(&callback_);
+  capturer_->CaptureFrame();
+  ASSERT_TRUE(frame);
+}
+
+TEST_F(ScreenCapturerTest, UseDirectxCapturer) {
+  if (!CreateDirectxCapturer()) {
+    return;
+  }
+
+  std::unique_ptr<DesktopFrame> frame;
+  EXPECT_CALL(callback_,
+              OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS, _))
+      .WillOnce(SaveUniquePtrArg(&frame));
+
+  capturer_->Start(&callback_);
+  capturer_->CaptureFrame();
+  ASSERT_TRUE(frame);
+}
+
+TEST_F(ScreenCapturerTest, UseDirectxCapturerWithSharedBuffers) {
+  if (!CreateDirectxCapturer()) {
+    return;
+  }
+
+  std::unique_ptr<DesktopFrame> frame;
+  EXPECT_CALL(callback_,
+              OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS, _))
+      .WillOnce(SaveUniquePtrArg(&frame));
+
+  capturer_->Start(&callback_);
+  capturer_->SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory>(new FakeSharedMemoryFactory()));
+  capturer_->CaptureFrame();
+  ASSERT_TRUE(frame);
+  ASSERT_TRUE(frame->shared_memory());
+  EXPECT_EQ(frame->shared_memory()->id(), kTestSharedMemoryId);
+}
+
+#endif  // defined(WEBRTC_WIN)
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_capturer_win.cc b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_win.cc
new file mode 100644
index 0000000000..b5935dc316
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_capturer_win.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <utility>
+
+#include "modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/fallback_desktop_capturer_wrapper.h"
+#include "modules/desktop_capture/rgba_color.h"
+#include "modules/desktop_capture/win/screen_capturer_win_directx.h"
+#include "modules/desktop_capture/win/screen_capturer_win_gdi.h"
+#include "modules/desktop_capture/win/screen_capturer_win_magnifier.h"
+
+namespace webrtc {
+
+namespace {
+
+std::unique_ptr<DesktopCapturer> CreateScreenCapturerWinDirectx() {
+  std::unique_ptr<DesktopCapturer> capturer(new ScreenCapturerWinDirectx());
+  capturer.reset(new BlankDetectorDesktopCapturerWrapper(
+      std::move(capturer), RgbaColor(0, 0, 0, 0)));
+  return capturer;
+}
+
+}  // namespace
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
+    const DesktopCaptureOptions& options) {
+  std::unique_ptr<DesktopCapturer> capturer(new ScreenCapturerWinGdi(options));
+  if (options.allow_directx_capturer()) {
+    // `dxgi_duplicator_controller` should be alive in this scope to ensure it
+    // won't unload DxgiDuplicatorController.
+    auto dxgi_duplicator_controller = DxgiDuplicatorController::Instance();
+    if (ScreenCapturerWinDirectx::IsSupported()) {
+      capturer.reset(new FallbackDesktopCapturerWrapper(
+          CreateScreenCapturerWinDirectx(), std::move(capturer)));
+    }
+  }
+
+  if (options.allow_use_magnification_api()) {
+    // ScreenCapturerWinMagnifier cannot work on Windows XP or earlier, as well
+    // as 64-bit only Windows, and it may randomly crash on multi-screen
+    // systems. So we may need to fallback to use original capturer.
+    capturer.reset(new FallbackDesktopCapturerWrapper(
+        std::unique_ptr<DesktopCapturer>(new ScreenCapturerWinMagnifier()),
+        std::move(capturer)));
+  }
+
+  return capturer;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_drawer.cc b/third_party/libwebrtc/modules/desktop_capture/screen_drawer.cc
new file mode 100644
index 0000000000..6460f19f65
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_drawer.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/screen_drawer.h"
+
+namespace webrtc {
+
+namespace {
+std::unique_ptr<ScreenDrawerLock> g_screen_drawer_lock;
+}  // namespace
+
+ScreenDrawerLock::ScreenDrawerLock() = default;
+ScreenDrawerLock::~ScreenDrawerLock() = default;
+
+ScreenDrawer::ScreenDrawer() {
+  g_screen_drawer_lock = ScreenDrawerLock::Create();
+}
+
+ScreenDrawer::~ScreenDrawer() {
+  g_screen_drawer_lock.reset();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_drawer.h b/third_party/libwebrtc/modules/desktop_capture/screen_drawer.h
new file mode 100644
index 0000000000..ad7c0ad8d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_drawer.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_SCREEN_DRAWER_H_
+#define MODULES_DESKTOP_CAPTURE_SCREEN_DRAWER_H_
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/rgba_color.h"
+
+namespace webrtc {
+
+// A cross-process lock to ensure only one ScreenDrawer can be used at a certain
+// time.
+class ScreenDrawerLock {
+ public:
+  virtual ~ScreenDrawerLock();
+
+  static std::unique_ptr<ScreenDrawerLock> Create();
+
+ protected:
+  ScreenDrawerLock();
+};
+
+// A set of basic platform dependent functions to draw various shapes on the
+// screen.
+class ScreenDrawer {
+ public:
+  // Creates a ScreenDrawer for the current platform, returns nullptr if no
+  // ScreenDrawer implementation available.
+  // If the implementation cannot guarantee two ScreenDrawer instances won't
+  // impact each other, this function may block current thread until another
+  // ScreenDrawer has been destroyed.
+  static std::unique_ptr<ScreenDrawer> Create();
+
+  ScreenDrawer();
+  virtual ~ScreenDrawer();
+
+  // Returns the region inside which DrawRectangle() function are expected to
+  // work, in capturer coordinates (assuming ScreenCapturer::SelectScreen has
+  // not been called). This region may exclude regions of the screen reserved by
+  // the OS for things like menu bars or app launchers. The DesktopRect is in
+  // system coordinate, i.e. the primary monitor always starts from (0, 0).
+  virtual DesktopRect DrawableRegion() = 0;
+
+  // Draws a rectangle to cover `rect` with `color`. Note, rect.bottom() and
+  // rect.right() two lines are not included. The part of `rect` which is out of
+  // DrawableRegion() will be ignored.
+  virtual void DrawRectangle(DesktopRect rect, RgbaColor color) = 0;
+
+  // Clears all content on the screen by filling the area with black.
+  virtual void Clear() = 0;
+
+  // Blocks current thread until OS finishes previous DrawRectangle() actions.
+  // ScreenCapturer should be able to capture the changes after this function
+  // finish.
+  virtual void WaitForPendingDraws() = 0;
+
+  // Returns true if incomplete shapes previous actions required may be drawn on
+  // the screen after a WaitForPendingDraws() call. i.e. Though the complete
+  // shapes will eventually be drawn on the screen, due to some OS limitations,
+  // these shapes may be partially appeared sometimes.
+  virtual bool MayDrawIncompleteShapes() = 0;
+
+  // Returns the id of the drawer window. This function returns kNullWindowId if
+  // the implementation does not draw on a window of the system.
+  virtual WindowId window_id() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_SCREEN_DRAWER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_drawer_linux.cc b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_linux.cc
new file mode 100644
index 0000000000..fce036b4aa
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_linux.cc
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <X11/X.h>
+#include <X11/Xlib.h>
+#include <string.h>
+
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/linux/x11/shared_x_display.h"
+#include "modules/desktop_capture/rgba_color.h"
+#include "modules/desktop_capture/screen_drawer.h"
+#include "modules/desktop_capture/screen_drawer_lock_posix.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/sleep.h"
+
+namespace webrtc {
+
+namespace {
+
+// A ScreenDrawer implementation for X11.
+class ScreenDrawerLinux : public ScreenDrawer {
+ public:
+  ScreenDrawerLinux();
+  ~ScreenDrawerLinux() override;
+
+  // ScreenDrawer interface.
+  DesktopRect DrawableRegion() override;
+  void DrawRectangle(DesktopRect rect, RgbaColor color) override;
+  void Clear() override;
+  void WaitForPendingDraws() override;
+  bool MayDrawIncompleteShapes() override;
+  WindowId window_id() const override;
+
+ private:
+  // Bring the window to the front, this can help to avoid the impact from other
+  // windows or shadow effect.
+  void BringToFront();
+
+  rtc::scoped_refptr<SharedXDisplay> display_;
+  int screen_num_;
+  DesktopRect rect_;
+  Window window_;
+  GC context_;
+  Colormap colormap_;
+};
+
+ScreenDrawerLinux::ScreenDrawerLinux() {
+  display_ = SharedXDisplay::CreateDefault();
+  RTC_CHECK(display_.get());
+  screen_num_ = DefaultScreen(display_->display());
+  XWindowAttributes root_attributes;
+  if (!XGetWindowAttributes(display_->display(),
+                            RootWindow(display_->display(), screen_num_),
+                            &root_attributes)) {
+    RTC_DCHECK_NOTREACHED() << "Failed to get root window size.";
+  }
+  window_ = XCreateSimpleWindow(
+      display_->display(), RootWindow(display_->display(), screen_num_), 0, 0,
+      root_attributes.width, root_attributes.height, 0,
+      BlackPixel(display_->display(), screen_num_),
+      BlackPixel(display_->display(), screen_num_));
+  XSelectInput(display_->display(), window_, StructureNotifyMask);
+  XMapWindow(display_->display(), window_);
+  while (true) {
+    XEvent event;
+    XNextEvent(display_->display(), &event);
+    if (event.type == MapNotify) {
+      break;
+    }
+  }
+  XFlush(display_->display());
+  Window child;
+  int x, y;
+  if (!XTranslateCoordinates(display_->display(), window_,
+                             RootWindow(display_->display(), screen_num_), 0, 0,
+                             &x, &y, &child)) {
+    RTC_DCHECK_NOTREACHED() << "Failed to get window position.";
+  }
+  // Some window manager does not allow a window to cover two or more monitors.
+  // So if the window is on the first monitor of a two-monitor system, the
+  // second half won't be able to show up without changing configurations of WM,
+  // and its DrawableRegion() is not accurate.
+  rect_ = DesktopRect::MakeLTRB(x, y, root_attributes.width,
+                                root_attributes.height);
+  context_ = DefaultGC(display_->display(), screen_num_);
+  colormap_ = DefaultColormap(display_->display(), screen_num_);
+  BringToFront();
+  // Wait for window animations.
+  SleepMs(200);
+}
+
+ScreenDrawerLinux::~ScreenDrawerLinux() {
+  XUnmapWindow(display_->display(), window_);
+  XDestroyWindow(display_->display(), window_);
+}
+
+DesktopRect ScreenDrawerLinux::DrawableRegion() {
+  return rect_;
+}
+
+void ScreenDrawerLinux::DrawRectangle(DesktopRect rect, RgbaColor color) {
+  rect.Translate(-rect_.left(), -rect_.top());
+  XColor xcolor;
+  // X11 does not support Alpha.
+  // X11 uses 16 bits for each primary color, so we need to slightly normalize
+  // a 8 bits channel to 16 bits channel, by setting the low 8 bits as its high
+  // 8 bits to avoid a mismatch of color returned by capturer.
+  xcolor.red = (color.red << 8) + color.red;
+  xcolor.green = (color.green << 8) + color.green;
+  xcolor.blue = (color.blue << 8) + color.blue;
+  xcolor.flags = DoRed | DoGreen | DoBlue;
+  XAllocColor(display_->display(), colormap_, &xcolor);
+  XSetForeground(display_->display(), context_, xcolor.pixel);
+  XFillRectangle(display_->display(), window_, context_, rect.left(),
+                 rect.top(), rect.width(), rect.height());
+  XFlush(display_->display());
+}
+
+void ScreenDrawerLinux::Clear() {
+  DrawRectangle(rect_, RgbaColor(0, 0, 0));
+}
+
+// TODO(zijiehe): Find the right signal from X11 to indicate the finish of all
+// pending paintings.
+void ScreenDrawerLinux::WaitForPendingDraws() {
+  SleepMs(50);
+}
+
+bool ScreenDrawerLinux::MayDrawIncompleteShapes() {
+  return true;
+}
+
+WindowId ScreenDrawerLinux::window_id() const {
+  return window_;
+}
+
+void ScreenDrawerLinux::BringToFront() {
+  Atom state_above = XInternAtom(display_->display(), "_NET_WM_STATE_ABOVE", 1);
+  Atom window_state = XInternAtom(display_->display(), "_NET_WM_STATE", 1);
+  if (state_above == None || window_state == None) {
+    // Fallback to use XRaiseWindow, it's not reliable if two windows are both
+    // raise itself to the top.
+    XRaiseWindow(display_->display(), window_);
+    return;
+  }
+
+  XEvent event;
+  memset(&event, 0, sizeof(event));
+  event.type = ClientMessage;
+  event.xclient.window = window_;
+  event.xclient.message_type = window_state;
+  event.xclient.format = 32;
+  event.xclient.data.l[0] = 1;  // _NET_WM_STATE_ADD
+  event.xclient.data.l[1] = state_above;
+  XSendEvent(display_->display(), RootWindow(display_->display(), screen_num_),
+             False, SubstructureRedirectMask | SubstructureNotifyMask, &event);
+}
+
+}  // namespace
+
+// static
+std::unique_ptr<ScreenDrawerLock> ScreenDrawerLock::Create() {
+  return std::make_unique<ScreenDrawerLockPosix>();
+}
+
+// static
+std::unique_ptr<ScreenDrawer> ScreenDrawer::Create() {
+  if (SharedXDisplay::CreateDefault().get()) {
+    return std::make_unique<ScreenDrawerLinux>();
+  }
+  return nullptr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_drawer_lock_posix.cc b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_lock_posix.cc
new file mode 100644
index 0000000000..28cb501fe7
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_lock_posix.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/screen_drawer_lock_posix.h"
+
+#include <fcntl.h>
+#include <sys/stat.h>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// A uuid as the name of semaphore.
+static constexpr char kSemaphoreName[] = "GSDL54fe5552804711e6a7253f429a";
+
+}  // namespace
+
+ScreenDrawerLockPosix::ScreenDrawerLockPosix()
+    : ScreenDrawerLockPosix(kSemaphoreName) {}
+
+ScreenDrawerLockPosix::ScreenDrawerLockPosix(const char* name) {
+  semaphore_ = sem_open(name, O_CREAT, S_IRWXU | S_IRWXG | S_IRWXO, 1);
+  if (semaphore_ == SEM_FAILED) {
+    RTC_LOG_ERRNO(LS_ERROR) << "Failed to create named semaphore with " << name;
+    RTC_DCHECK_NOTREACHED();
+  }
+
+  sem_wait(semaphore_);
+}
+
+ScreenDrawerLockPosix::~ScreenDrawerLockPosix() {
+  if (semaphore_ == SEM_FAILED) {
+    return;
+  }
+
+  sem_post(semaphore_);
+  sem_close(semaphore_);
+  // sem_unlink a named semaphore won't wait until other clients to release the
+  // sem_t. So if a new process starts, it will sem_open a different kernel
+  // object with the same name and eventually breaks the cross-process lock.
+}
+
+// static
+void ScreenDrawerLockPosix::Unlink(absl::string_view name) {
+  sem_unlink(std::string(name).c_str());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_drawer_lock_posix.h b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_lock_posix.h
new file mode 100644
index 0000000000..13899b2d75
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_lock_posix.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_SCREEN_DRAWER_LOCK_POSIX_H_
+#define MODULES_DESKTOP_CAPTURE_SCREEN_DRAWER_LOCK_POSIX_H_
+
+#include <semaphore.h>
+
+#include "absl/strings/string_view.h"
+#include "modules/desktop_capture/screen_drawer.h"
+
+namespace webrtc {
+
+class ScreenDrawerLockPosix final : public ScreenDrawerLock {
+ public:
+  ScreenDrawerLockPosix();
+  // Provides a name other than the default one for test only.
+  explicit ScreenDrawerLockPosix(const char* name);
+  ~ScreenDrawerLockPosix() override;
+
+  // Unlinks the named semaphore actively. This will remove the sem_t object in
+  // the system and allow others to create a different sem_t object with the
+  // same/ name.
+  static void Unlink(absl::string_view name);
+
+ private:
+  sem_t* semaphore_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_SCREEN_DRAWER_LOCK_POSIX_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_drawer_mac.cc b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_mac.cc
new file mode 100644
index 0000000000..17719e4439
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_mac.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// TODO(zijiehe): Implement ScreenDrawerMac
+
+#include <memory>
+
+#include "modules/desktop_capture/screen_drawer.h"
+#include "modules/desktop_capture/screen_drawer_lock_posix.h"
+
+namespace webrtc {
+
+// static
+std::unique_ptr<ScreenDrawerLock> ScreenDrawerLock::Create() {
+  return std::make_unique<ScreenDrawerLockPosix>();
+}
+
+// static
+std::unique_ptr<ScreenDrawer> ScreenDrawer::Create() {
+  return nullptr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_drawer_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_unittest.cc
new file mode 100644
index 0000000000..584770dbf8
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_unittest.cc
@@ -0,0 +1,160 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/screen_drawer.h"
+
+#include <stdint.h>
+
+#include <atomic>
+#include <memory>
+
+#include "api/function_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/random.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+
+#if defined(WEBRTC_POSIX)
+#include "modules/desktop_capture/screen_drawer_lock_posix.h"
+#endif
+
+namespace webrtc {
+
+namespace {
+
+void TestScreenDrawerLock(
+    rtc::FunctionView<std::unique_ptr<ScreenDrawerLock>()> ctor) {
+  constexpr int kLockDurationMs = 100;
+
+  std::atomic<bool> created(false);
+  std::atomic<bool> ready(false);
+
+  class Task {
+   public:
+    Task(std::atomic<bool>* created,
+         const std::atomic<bool>& ready,
+         rtc::FunctionView<std::unique_ptr<ScreenDrawerLock>()> ctor)
+        : created_(created), ready_(ready), ctor_(ctor) {}
+
+    ~Task() = default;
+
+    void RunTask() {
+      std::unique_ptr<ScreenDrawerLock> lock = ctor_();
+      ASSERT_TRUE(!!lock);
+      created_->store(true);
+      // Wait for the main thread to get the signal of created_.
+      while (!ready_.load()) {
+        SleepMs(1);
+      }
+      // At this point, main thread should begin to create a second lock. Though
+      // it's still possible the second lock won't be created before the
+      // following sleep has been finished, the possibility will be
+      // significantly reduced.
+      const int64_t current_ms = rtc::TimeMillis();
+      // SleepMs() may return early. See
+      // https://cs.chromium.org/chromium/src/third_party/webrtc/system_wrappers/include/sleep.h?rcl=4a604c80cecce18aff6fc5e16296d04675312d83&l=20
+      // But we need to ensure at least 100 ms has been passed before unlocking
+      // `lock`.
+      while (rtc::TimeMillis() - current_ms < kLockDurationMs) {
+        SleepMs(kLockDurationMs - (rtc::TimeMillis() - current_ms));
+      }
+    }
+
+   private:
+    std::atomic<bool>* const created_;
+    const std::atomic<bool>& ready_;
+    const rtc::FunctionView<std::unique_ptr<ScreenDrawerLock>()> ctor_;
+  } task(&created, ready, ctor);
+
+  auto lock_thread = rtc::PlatformThread::SpawnJoinable(
+      [&task] { task.RunTask(); }, "lock_thread");
+
+  // Wait for the first lock in Task::RunTask() to be created.
+  // TODO(zijiehe): Find a better solution to wait for the creation of the first
+  // lock. See
+  // https://chromium-review.googlesource.com/c/607688/13/webrtc/modules/desktop_capture/screen_drawer_unittest.cc
+  while (!created.load()) {
+    SleepMs(1);
+  }
+
+  const int64_t start_ms = rtc::TimeMillis();
+  ready.store(true);
+  // This is unlikely to fail, but just in case current thread is too laggy and
+  // cause the SleepMs() in RunTask() to finish before we creating another lock.
+  ASSERT_GT(kLockDurationMs, rtc::TimeMillis() - start_ms);
+  ctor();
+  ASSERT_LE(kLockDurationMs, rtc::TimeMillis() - start_ms);
+}
+
+}  // namespace
+
+// These are a set of manual test cases, as we do not have an automatical way to
+// detect whether a ScreenDrawer on a certain platform works well without
+// ScreenCapturer(s). So you may execute these test cases with
+// --gtest_also_run_disabled_tests --gtest_filter=ScreenDrawerTest.*.
+TEST(ScreenDrawerTest, DISABLED_DrawRectangles) {
+  std::unique_ptr<ScreenDrawer> drawer = ScreenDrawer::Create();
+  if (!drawer) {
+    RTC_LOG(LS_WARNING)
+        << "No ScreenDrawer implementation for current platform.";
+    return;
+  }
+
+  if (drawer->DrawableRegion().is_empty()) {
+    RTC_LOG(LS_WARNING)
+        << "ScreenDrawer of current platform does not provide a "
+           "non-empty DrawableRegion().";
+    return;
+  }
+
+  DesktopRect rect = drawer->DrawableRegion();
+  Random random(rtc::TimeMicros());
+  for (int i = 0; i < 100; i++) {
+    // Make sure we at least draw one pixel.
+    int left = random.Rand(rect.left(), rect.right() - 2);
+    int top = random.Rand(rect.top(), rect.bottom() - 2);
+    drawer->DrawRectangle(
+        DesktopRect::MakeLTRB(left, top, random.Rand(left + 1, rect.right()),
+                              random.Rand(top + 1, rect.bottom())),
+        RgbaColor(random.Rand<uint8_t>(), random.Rand<uint8_t>(),
+                  random.Rand<uint8_t>(), random.Rand<uint8_t>()));
+
+    if (i == 50) {
+      SleepMs(10000);
+    }
+  }
+
+  SleepMs(10000);
+}
+
+#if defined(THREAD_SANITIZER)  // bugs.webrtc.org/10019
+#define MAYBE_TwoScreenDrawerLocks DISABLED_TwoScreenDrawerLocks
+#else
+#define MAYBE_TwoScreenDrawerLocks TwoScreenDrawerLocks
+#endif
+TEST(ScreenDrawerTest, MAYBE_TwoScreenDrawerLocks) {
+#if defined(WEBRTC_POSIX)
+  // ScreenDrawerLockPosix won't be able to unlink the named semaphore. So use a
+  // different semaphore name here to avoid deadlock.
+  const char* semaphore_name = "GSDL8784541a812011e788ff67427b";
+  ScreenDrawerLockPosix::Unlink(semaphore_name);
+
+  TestScreenDrawerLock([semaphore_name]() {
+    return std::make_unique<ScreenDrawerLockPosix>(semaphore_name);
+  });
+#elif defined(WEBRTC_WIN)
+  TestScreenDrawerLock([]() { return ScreenDrawerLock::Create(); });
+#endif
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/screen_drawer_win.cc b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_win.cc
new file mode 100644
index 0000000000..7cf634fe89
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/screen_drawer_win.cc
@@ -0,0 +1,209 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <windows.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/screen_drawer.h"
+#include "system_wrappers/include/sleep.h"
+
+namespace webrtc {
+
+namespace {
+
+static constexpr TCHAR kMutexName[] =
+    TEXT("Local\\ScreenDrawerWin-da834f82-8044-11e6-ac81-73dcdd1c1869");
+
+class ScreenDrawerLockWin : public ScreenDrawerLock {
+ public:
+  ScreenDrawerLockWin();
+  ~ScreenDrawerLockWin() override;
+
+ private:
+  HANDLE mutex_;
+};
+
+ScreenDrawerLockWin::ScreenDrawerLockWin() {
+  while (true) {
+    mutex_ = CreateMutex(NULL, FALSE, kMutexName);
+    if (GetLastError() != ERROR_ALREADY_EXISTS && mutex_ != NULL) {
+      break;
+    } else {
+      if (mutex_) {
+        CloseHandle(mutex_);
+      }
+      SleepMs(1000);
+    }
+  }
+}
+
+ScreenDrawerLockWin::~ScreenDrawerLockWin() {
+  CloseHandle(mutex_);
+}
+
+DesktopRect GetScreenRect() {
+  HDC hdc = GetDC(NULL);
+  DesktopRect rect = DesktopRect::MakeWH(GetDeviceCaps(hdc, HORZRES),
+                                         GetDeviceCaps(hdc, VERTRES));
+  ReleaseDC(NULL, hdc);
+  return rect;
+}
+
+HWND CreateDrawerWindow(DesktopRect rect) {
+  HWND hwnd = CreateWindowA(
+      "STATIC", "DrawerWindow", WS_POPUPWINDOW | WS_VISIBLE, rect.left(),
+      rect.top(), rect.width(), rect.height(), NULL, NULL, NULL, NULL);
+  SetForegroundWindow(hwnd);
+  return hwnd;
+}
+
+COLORREF ColorToRef(RgbaColor color) {
+  // Windows device context does not support alpha.
+  return RGB(color.red, color.green, color.blue);
+}
+
+// A ScreenDrawer implementation for Windows.
+class ScreenDrawerWin : public ScreenDrawer {
+ public:
+  ScreenDrawerWin();
+  ~ScreenDrawerWin() override;
+
+  // ScreenDrawer interface.
+  DesktopRect DrawableRegion() override;
+  void DrawRectangle(DesktopRect rect, RgbaColor color) override;
+  void Clear() override;
+  void WaitForPendingDraws() override;
+  bool MayDrawIncompleteShapes() override;
+  WindowId window_id() const override;
+
+ private:
+  // Bring the window to the front, this can help to avoid the impact from other
+  // windows or shadow effects.
+  void BringToFront();
+
+  // Draw a line with `color`.
+  void DrawLine(DesktopVector start, DesktopVector end, RgbaColor color);
+
+  // Draw a dot with `color`.
+  void DrawDot(DesktopVector vect, RgbaColor color);
+
+  const DesktopRect rect_;
+  HWND window_;
+  HDC hdc_;
+};
+
+ScreenDrawerWin::ScreenDrawerWin()
+    : ScreenDrawer(),
+      rect_(GetScreenRect()),
+      window_(CreateDrawerWindow(rect_)),
+      hdc_(GetWindowDC(window_)) {
+  // We do not need to handle any messages for the `window_`, so disable Windows
+  // from processing windows ghosting feature.
+  DisableProcessWindowsGhosting();
+
+  // Always use stock pen (DC_PEN) and brush (DC_BRUSH).
+  SelectObject(hdc_, GetStockObject(DC_PEN));
+  SelectObject(hdc_, GetStockObject(DC_BRUSH));
+  BringToFront();
+}
+
+ScreenDrawerWin::~ScreenDrawerWin() {
+  ReleaseDC(NULL, hdc_);
+  DestroyWindow(window_);
+  // Unfortunately there is no EnableProcessWindowsGhosting() API.
+}
+
+DesktopRect ScreenDrawerWin::DrawableRegion() {
+  return rect_;
+}
+
+void ScreenDrawerWin::DrawRectangle(DesktopRect rect, RgbaColor color) {
+  if (rect.width() == 1 && rect.height() == 1) {
+    // Rectangle function cannot draw a 1 pixel rectangle.
+    DrawDot(rect.top_left(), color);
+    return;
+  }
+
+  if (rect.width() == 1 || rect.height() == 1) {
+    // Rectangle function cannot draw a 1 pixel rectangle.
+    DrawLine(rect.top_left(), DesktopVector(rect.right(), rect.bottom()),
+             color);
+    return;
+  }
+
+  SetDCBrushColor(hdc_, ColorToRef(color));
+  SetDCPenColor(hdc_, ColorToRef(color));
+  Rectangle(hdc_, rect.left(), rect.top(), rect.right(), rect.bottom());
+}
+
+void ScreenDrawerWin::Clear() {
+  DrawRectangle(rect_, RgbaColor(0, 0, 0));
+}
+
+// TODO(zijiehe): Find the right signal to indicate the finish of all pending
+// paintings.
+void ScreenDrawerWin::WaitForPendingDraws() {
+  BringToFront();
+  SleepMs(50);
+}
+
+bool ScreenDrawerWin::MayDrawIncompleteShapes() {
+  return true;
+}
+
+WindowId ScreenDrawerWin::window_id() const {
+  return reinterpret_cast<WindowId>(window_);
+}
+
+void ScreenDrawerWin::DrawLine(DesktopVector start,
+                               DesktopVector end,
+                               RgbaColor color) {
+  POINT points[2];
+  points[0].x = start.x();
+  points[0].y = start.y();
+  points[1].x = end.x();
+  points[1].y = end.y();
+  SetDCPenColor(hdc_, ColorToRef(color));
+  Polyline(hdc_, points, 2);
+}
+
+void ScreenDrawerWin::DrawDot(DesktopVector vect, RgbaColor color) {
+  SetPixel(hdc_, vect.x(), vect.y(), ColorToRef(color));
+}
+
+void ScreenDrawerWin::BringToFront() {
+  if (SetWindowPos(window_, HWND_TOPMOST, 0, 0, 0, 0,
+                   SWP_NOMOVE | SWP_NOSIZE) != FALSE) {
+    return;
+  }
+
+  long ex_style = GetWindowLong(window_, GWL_EXSTYLE);
+  ex_style |= WS_EX_TOPMOST;
+  if (SetWindowLong(window_, GWL_EXSTYLE, ex_style) != 0) {
+    return;
+  }
+
+  BringWindowToTop(window_);
+}
+
+}  // namespace
+
+// static
+std::unique_ptr<ScreenDrawerLock> ScreenDrawerLock::Create() {
+  return std::unique_ptr<ScreenDrawerLock>(new ScreenDrawerLockWin());
+}
+
+// static
+std::unique_ptr<ScreenDrawer> ScreenDrawer::Create() {
+  return std::unique_ptr<ScreenDrawer>(new ScreenDrawerWin());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/shared_desktop_frame.cc b/third_party/libwebrtc/modules/desktop_capture/shared_desktop_frame.cc
new file mode 100644
index 0000000000..e374038cbc
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/shared_desktop_frame.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/shared_desktop_frame.h"
+
+#include <memory>
+#include <type_traits>
+#include <utility>
+
+namespace webrtc {
+
+SharedDesktopFrame::~SharedDesktopFrame() {}
+
+// static
+std::unique_ptr<SharedDesktopFrame> SharedDesktopFrame::Wrap(
+    std::unique_ptr<DesktopFrame> desktop_frame) {
+  return std::unique_ptr<SharedDesktopFrame>(new SharedDesktopFrame(
+      rtc::scoped_refptr<Core>(new Core(std::move(desktop_frame)))));
+}
+
+SharedDesktopFrame* SharedDesktopFrame::Wrap(DesktopFrame* desktop_frame) {
+  return Wrap(std::unique_ptr<DesktopFrame>(desktop_frame)).release();
+}
+
+DesktopFrame* SharedDesktopFrame::GetUnderlyingFrame() {
+  return core_->get();
+}
+
+bool SharedDesktopFrame::ShareFrameWith(const SharedDesktopFrame& other) const {
+  return core_->get() == other.core_->get();
+}
+
+std::unique_ptr<SharedDesktopFrame> SharedDesktopFrame::Share() {
+  std::unique_ptr<SharedDesktopFrame> result(new SharedDesktopFrame(core_));
+  result->CopyFrameInfoFrom(*this);
+  return result;
+}
+
+bool SharedDesktopFrame::IsShared() {
+  return !core_->HasOneRef();
+}
+
+SharedDesktopFrame::SharedDesktopFrame(rtc::scoped_refptr<Core> core)
+    : DesktopFrame((*core)->size(),
+                   (*core)->stride(),
+                   (*core)->data(),
+                   (*core)->shared_memory()),
+      core_(core) {
+  CopyFrameInfoFrom(*(core_->get()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/shared_desktop_frame.h b/third_party/libwebrtc/modules/desktop_capture/shared_desktop_frame.h
new file mode 100644
index 0000000000..c6f52247f4
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/shared_desktop_frame.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_SHARED_DESKTOP_FRAME_H_
+#define MODULES_DESKTOP_CAPTURE_SHARED_DESKTOP_FRAME_H_
+
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "rtc_base/ref_counted_object.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// SharedDesktopFrame is a DesktopFrame that may have multiple instances all
+// sharing the same buffer.
+class RTC_EXPORT SharedDesktopFrame final : public DesktopFrame {
+ public:
+  ~SharedDesktopFrame() override;
+
+  SharedDesktopFrame(const SharedDesktopFrame&) = delete;
+  SharedDesktopFrame& operator=(const SharedDesktopFrame&) = delete;
+
+  static std::unique_ptr<SharedDesktopFrame> Wrap(
+      std::unique_ptr<DesktopFrame> desktop_frame);
+
+  // Deprecated.
+  // TODO(sergeyu): remove this method.
+  static SharedDesktopFrame* Wrap(DesktopFrame* desktop_frame);
+
+  // Deprecated. Clients do not need to know the underlying DesktopFrame
+  // instance.
+  // TODO(zijiehe): Remove this method.
+  // Returns the underlying instance of DesktopFrame.
+  DesktopFrame* GetUnderlyingFrame();
+
+  // Returns whether `this` and `other` share the underlying DesktopFrame.
+  bool ShareFrameWith(const SharedDesktopFrame& other) const;
+
+  // Creates a clone of this object.
+  std::unique_ptr<SharedDesktopFrame> Share();
+
+  // Checks if the frame is currently shared. If it returns false it's
+  // guaranteed that there are no clones of the object.
+  bool IsShared();
+
+ private:
+  typedef rtc::FinalRefCountedObject<std::unique_ptr<DesktopFrame>> Core;
+
+  SharedDesktopFrame(rtc::scoped_refptr<Core> core);
+
+  const rtc::scoped_refptr<Core> core_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_SHARED_DESKTOP_FRAME_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/shared_memory.cc b/third_party/libwebrtc/modules/desktop_capture/shared_memory.cc
new file mode 100644
index 0000000000..b4ff78b2a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/shared_memory.cc
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/shared_memory.h"
+
+namespace webrtc {
+
+#if defined(WEBRTC_WIN)
+const SharedMemory::Handle SharedMemory::kInvalidHandle = NULL;
+#else
+const SharedMemory::Handle SharedMemory::kInvalidHandle = -1;
+#endif
+
+SharedMemory::SharedMemory(void* data, size_t size, Handle handle, int id)
+    : data_(data), size_(size), handle_(handle), id_(id) {}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/shared_memory.h b/third_party/libwebrtc/modules/desktop_capture/shared_memory.h
new file mode 100644
index 0000000000..a7add4447b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/shared_memory.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_SHARED_MEMORY_H_
+#define MODULES_DESKTOP_CAPTURE_SHARED_MEMORY_H_
+
+#include <stddef.h>
+
+#if defined(WEBRTC_WIN)
+// Forward declare HANDLE in a windows.h compatible way so that we can avoid
+// including windows.h.
+typedef void* HANDLE;
+#endif
+
+#include <memory>
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// SharedMemory is a base class for shared memory. It stores all required
+// parameters of the buffer, but doesn't have any logic to allocate or destroy
+// the actual buffer. DesktopCapturer consumers that need to use shared memory
+// for video frames must extend this class with creation and destruction logic
+// specific for the target platform and then call
+// DesktopCapturer::SetSharedMemoryFactory().
+class RTC_EXPORT SharedMemory {
+ public:
+#if defined(WEBRTC_WIN)
+  typedef HANDLE Handle;
+  static const Handle kInvalidHandle;
+#else
+  typedef int Handle;
+  static const Handle kInvalidHandle;
+#endif
+
+  void* data() const { return data_; }
+  size_t size() const { return size_; }
+
+  // Platform-specific handle of the buffer.
+  Handle handle() const { return handle_; }
+
+  // Integer identifier that can be used used by consumers of DesktopCapturer
+  // interface to identify shared memory buffers it created.
+  int id() const { return id_; }
+
+  virtual ~SharedMemory() {}
+
+  SharedMemory(const SharedMemory&) = delete;
+  SharedMemory& operator=(const SharedMemory&) = delete;
+
+ protected:
+  SharedMemory(void* data, size_t size, Handle handle, int id);
+
+  void* const data_;
+  const size_t size_;
+  const Handle handle_;
+  const int id_;
+};
+
+// Interface used to create SharedMemory instances.
+class SharedMemoryFactory {
+ public:
+  SharedMemoryFactory() {}
+  virtual ~SharedMemoryFactory() {}
+
+  SharedMemoryFactory(const SharedMemoryFactory&) = delete;
+  SharedMemoryFactory& operator=(const SharedMemoryFactory&) = delete;
+
+  virtual std::unique_ptr<SharedMemory> CreateSharedMemory(size_t size) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_SHARED_MEMORY_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/test_utils.cc b/third_party/libwebrtc/modules/desktop_capture/test_utils.cc
new file mode 100644
index 0000000000..9483bf41ea
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/test_utils.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/test_utils.h"
+
+#include <stdint.h>
+#include <string.h>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void ClearDesktopFrame(DesktopFrame* frame) {
+  RTC_DCHECK(frame);
+  uint8_t* data = frame->data();
+  for (int i = 0; i < frame->size().height(); i++) {
+    memset(data, 0, frame->size().width() * DesktopFrame::kBytesPerPixel);
+    data += frame->stride();
+  }
+}
+
+bool DesktopFrameDataEquals(const DesktopFrame& left,
+                            const DesktopFrame& right) {
+  if (!left.size().equals(right.size())) {
+    return false;
+  }
+
+  const uint8_t* left_array = left.data();
+  const uint8_t* right_array = right.data();
+  for (int i = 0; i < left.size().height(); i++) {
+    if (memcmp(left_array, right_array,
+               DesktopFrame::kBytesPerPixel * left.size().width()) != 0) {
+      return false;
+    }
+    left_array += left.stride();
+    right_array += right.stride();
+  }
+
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/test_utils.h b/third_party/libwebrtc/modules/desktop_capture/test_utils.h
new file mode 100644
index 0000000000..8669fecba3
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/test_utils.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_TEST_UTILS_H_
+#define MODULES_DESKTOP_CAPTURE_TEST_UTILS_H_
+
+#include "modules/desktop_capture/desktop_frame.h"
+
+namespace webrtc {
+
+// Clears a DesktopFrame `frame` by setting its data() into 0.
+void ClearDesktopFrame(DesktopFrame* frame);
+
+// Compares size() and data() of two DesktopFrames `left` and `right`.
+bool DesktopFrameDataEquals(const DesktopFrame& left,
+                            const DesktopFrame& right);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_TEST_UTILS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/test_utils_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/test_utils_unittest.cc
new file mode 100644
index 0000000000..c1326f01cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/test_utils_unittest.cc
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/test_utils.h"
+
+#include <stdint.h>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/rgba_color.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+void PaintDesktopFrame(DesktopFrame* frame,
+                       DesktopVector pos,
+                       RgbaColor color) {
+  ASSERT_TRUE(frame);
+  ASSERT_TRUE(DesktopRect::MakeSize(frame->size()).Contains(pos));
+  *reinterpret_cast<uint32_t*>(frame->GetFrameDataAtPos(pos)) =
+      color.ToUInt32();
+}
+
+// A DesktopFrame implementation to store data in heap, but the stide is
+// doubled.
+class DoubleSizeDesktopFrame : public DesktopFrame {
+ public:
+  explicit DoubleSizeDesktopFrame(DesktopSize size);
+  ~DoubleSizeDesktopFrame() override;
+};
+
+DoubleSizeDesktopFrame::DoubleSizeDesktopFrame(DesktopSize size)
+    : DesktopFrame(
+          size,
+          kBytesPerPixel * size.width() * 2,
+          new uint8_t[kBytesPerPixel * size.width() * size.height() * 2],
+          nullptr) {}
+
+DoubleSizeDesktopFrame::~DoubleSizeDesktopFrame() {
+  delete[] data_;
+}
+
+}  // namespace
+
+TEST(TestUtilsTest, BasicDataEqualsCases) {
+  BasicDesktopFrame frame(DesktopSize(4, 4));
+  for (int i = 0; i < 4; i++) {
+    for (int j = 0; j < 4; j++) {
+      PaintDesktopFrame(&frame, DesktopVector(i, j), RgbaColor(4U * j + i));
+    }
+  }
+
+  ASSERT_TRUE(DesktopFrameDataEquals(frame, frame));
+  BasicDesktopFrame other(DesktopSize(4, 4));
+  for (int i = 0; i < 4; i++) {
+    for (int j = 0; j < 4; j++) {
+      PaintDesktopFrame(&other, DesktopVector(i, j), RgbaColor(4U * j + i));
+    }
+  }
+  ASSERT_TRUE(DesktopFrameDataEquals(frame, other));
+  PaintDesktopFrame(&other, DesktopVector(2, 2), RgbaColor(0U));
+  ASSERT_FALSE(DesktopFrameDataEquals(frame, other));
+}
+
+TEST(TestUtilsTest, DifferentSizeShouldNotEqual) {
+  BasicDesktopFrame frame(DesktopSize(4, 4));
+  for (int i = 0; i < 4; i++) {
+    for (int j = 0; j < 4; j++) {
+      PaintDesktopFrame(&frame, DesktopVector(i, j), RgbaColor(4U * j + i));
+    }
+  }
+
+  BasicDesktopFrame other(DesktopSize(2, 8));
+  for (int i = 0; i < 2; i++) {
+    for (int j = 0; j < 8; j++) {
+      PaintDesktopFrame(&other, DesktopVector(i, j), RgbaColor(2U * j + i));
+    }
+  }
+
+  ASSERT_FALSE(DesktopFrameDataEquals(frame, other));
+}
+
+TEST(TestUtilsTest, DifferentStrideShouldBeComparable) {
+  BasicDesktopFrame frame(DesktopSize(4, 4));
+  for (int i = 0; i < 4; i++) {
+    for (int j = 0; j < 4; j++) {
+      PaintDesktopFrame(&frame, DesktopVector(i, j), RgbaColor(4U * j + i));
+    }
+  }
+
+  ASSERT_TRUE(DesktopFrameDataEquals(frame, frame));
+  DoubleSizeDesktopFrame other(DesktopSize(4, 4));
+  for (int i = 0; i < 4; i++) {
+    for (int j = 0; j < 4; j++) {
+      PaintDesktopFrame(&other, DesktopVector(i, j), RgbaColor(4U * j + i));
+    }
+  }
+  ASSERT_TRUE(DesktopFrameDataEquals(frame, other));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor.cc b/third_party/libwebrtc/modules/desktop_capture/win/cursor.cc
new file mode 100644
index 0000000000..1d645098e2
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor.cc
@@ -0,0 +1,233 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/cursor.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/win/scoped_gdi_object.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+
+namespace {
+
+#if defined(WEBRTC_ARCH_LITTLE_ENDIAN)
+
+#define RGBA(r, g, b, a)                                   \
+  ((((a) << 24) & 0xff000000) | (((b) << 16) & 0xff0000) | \
+   (((g) << 8) & 0xff00) | ((r)&0xff))
+
+#else  // !defined(WEBRTC_ARCH_LITTLE_ENDIAN)
+
+#define RGBA(r, g, b, a)                                   \
+  ((((r) << 24) & 0xff000000) | (((g) << 16) & 0xff0000) | \
+   (((b) << 8) & 0xff00) | ((a)&0xff))
+
+#endif  // !defined(WEBRTC_ARCH_LITTLE_ENDIAN)
+
+const int kBytesPerPixel = DesktopFrame::kBytesPerPixel;
+
+// Pixel colors used when generating cursor outlines.
+const uint32_t kPixelRgbaBlack = RGBA(0, 0, 0, 0xff);
+const uint32_t kPixelRgbaWhite = RGBA(0xff, 0xff, 0xff, 0xff);
+const uint32_t kPixelRgbaTransparent = RGBA(0, 0, 0, 0);
+
+const uint32_t kPixelRgbWhite = RGB(0xff, 0xff, 0xff);
+
+// Expands the cursor shape to add a white outline for visibility against
+// dark backgrounds.
+void AddCursorOutline(int width, int height, uint32_t* data) {
+  for (int y = 0; y < height; y++) {
+    for (int x = 0; x < width; x++) {
+      // If this is a transparent pixel (bgr == 0 and alpha = 0), check the
+      // neighbor pixels to see if this should be changed to an outline pixel.
+      if (*data == kPixelRgbaTransparent) {
+        // Change to white pixel if any neighbors (top, bottom, left, right)
+        // are black.
+        if ((y > 0 && data[-width] == kPixelRgbaBlack) ||
+            (y < height - 1 && data[width] == kPixelRgbaBlack) ||
+            (x > 0 && data[-1] == kPixelRgbaBlack) ||
+            (x < width - 1 && data[1] == kPixelRgbaBlack)) {
+          *data = kPixelRgbaWhite;
+        }
+      }
+      data++;
+    }
+  }
+}
+
+// Premultiplies RGB components of the pixel data in the given image by
+// the corresponding alpha components.
+void AlphaMul(uint32_t* data, int width, int height) {
+  static_assert(sizeof(uint32_t) == kBytesPerPixel,
+                "size of uint32 should be the number of bytes per pixel");
+
+  for (uint32_t* data_end = data + width * height; data != data_end; ++data) {
+    RGBQUAD* from = reinterpret_cast<RGBQUAD*>(data);
+    RGBQUAD* to = reinterpret_cast<RGBQUAD*>(data);
+    to->rgbBlue =
+        (static_cast<uint16_t>(from->rgbBlue) * from->rgbReserved) / 0xff;
+    to->rgbGreen =
+        (static_cast<uint16_t>(from->rgbGreen) * from->rgbReserved) / 0xff;
+    to->rgbRed =
+        (static_cast<uint16_t>(from->rgbRed) * from->rgbReserved) / 0xff;
+  }
+}
+
+// Scans a 32bpp bitmap looking for any pixels with non-zero alpha component.
+// Returns true if non-zero alpha is found. `stride` is expressed in pixels.
+bool HasAlphaChannel(const uint32_t* data, int stride, int width, int height) {
+  const RGBQUAD* plane = reinterpret_cast<const RGBQUAD*>(data);
+  for (int y = 0; y < height; ++y) {
+    for (int x = 0; x < width; ++x) {
+      if (plane->rgbReserved != 0)
+        return true;
+      plane += 1;
+    }
+    plane += stride - width;
+  }
+
+  return false;
+}
+
+}  // namespace
+
+MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor) {
+  ICONINFO iinfo;
+  if (!GetIconInfo(cursor, &iinfo)) {
+    RTC_LOG_F(LS_ERROR) << "Unable to get cursor icon info. Error = "
+                        << GetLastError();
+    return NULL;
+  }
+
+  int hotspot_x = iinfo.xHotspot;
+  int hotspot_y = iinfo.yHotspot;
+
+  // Make sure the bitmaps will be freed.
+  win::ScopedBitmap scoped_mask(iinfo.hbmMask);
+  win::ScopedBitmap scoped_color(iinfo.hbmColor);
+  bool is_color = iinfo.hbmColor != NULL;
+
+  // Get `scoped_mask` dimensions.
+  BITMAP bitmap_info;
+  if (!GetObject(scoped_mask, sizeof(bitmap_info), &bitmap_info)) {
+    RTC_LOG_F(LS_ERROR) << "Unable to get bitmap info. Error = "
+                        << GetLastError();
+    return NULL;
+  }
+
+  int width = bitmap_info.bmWidth;
+  int height = bitmap_info.bmHeight;
+  std::unique_ptr<uint32_t[]> mask_data(new uint32_t[width * height]);
+
+  // Get pixel data from `scoped_mask` converting it to 32bpp along the way.
+  // GetDIBits() sets the alpha component of every pixel to 0.
+  BITMAPV5HEADER bmi = {0};
+  bmi.bV5Size = sizeof(bmi);
+  bmi.bV5Width = width;
+  bmi.bV5Height = -height;  // request a top-down bitmap.
+  bmi.bV5Planes = 1;
+  bmi.bV5BitCount = kBytesPerPixel * 8;
+  bmi.bV5Compression = BI_RGB;
+  bmi.bV5AlphaMask = 0xff000000;
+  bmi.bV5CSType = LCS_WINDOWS_COLOR_SPACE;
+  bmi.bV5Intent = LCS_GM_BUSINESS;
+  if (!GetDIBits(dc, scoped_mask, 0, height, mask_data.get(),
+                 reinterpret_cast<BITMAPINFO*>(&bmi), DIB_RGB_COLORS)) {
+    RTC_LOG_F(LS_ERROR) << "Unable to get bitmap bits. Error = "
+                        << GetLastError();
+    return NULL;
+  }
+
+  uint32_t* mask_plane = mask_data.get();
+  std::unique_ptr<DesktopFrame> image(
+      new BasicDesktopFrame(DesktopSize(width, height)));
+  bool has_alpha = false;
+
+  if (is_color) {
+    image.reset(new BasicDesktopFrame(DesktopSize(width, height)));
+    // Get the pixels from the color bitmap.
+    if (!GetDIBits(dc, scoped_color, 0, height, image->data(),
+                   reinterpret_cast<BITMAPINFO*>(&bmi), DIB_RGB_COLORS)) {
+      RTC_LOG_F(LS_ERROR) << "Unable to get bitmap bits. Error = "
+                          << GetLastError();
+      return NULL;
+    }
+
+    // GetDIBits() does not provide any indication whether the bitmap has alpha
+    // channel, so we use HasAlphaChannel() below to find it out.
+    has_alpha = HasAlphaChannel(reinterpret_cast<uint32_t*>(image->data()),
+                                width, width, height);
+  } else {
+    // For non-color cursors, the mask contains both an AND and an XOR mask and
+    // the height includes both. Thus, the width is correct, but we need to
+    // divide by 2 to get the correct mask height.
+    height /= 2;
+
+    image.reset(new BasicDesktopFrame(DesktopSize(width, height)));
+
+    // The XOR mask becomes the color bitmap.
+    memcpy(image->data(), mask_plane + (width * height),
+           image->stride() * height);
+  }
+
+  // Reconstruct transparency from the mask if the color image does not has
+  // alpha channel.
+  if (!has_alpha) {
+    bool add_outline = false;
+    uint32_t* dst = reinterpret_cast<uint32_t*>(image->data());
+    uint32_t* mask = mask_plane;
+    for (int y = 0; y < height; y++) {
+      for (int x = 0; x < width; x++) {
+        // The two bitmaps combine as follows:
+        //  mask  color   Windows Result   Our result    RGB   Alpha
+        //   0     00      Black            Black         00    ff
+        //   0     ff      White            White         ff    ff
+        //   1     00      Screen           Transparent   00    00
+        //   1     ff      Reverse-screen   Black         00    ff
+        //
+        // Since we don't support XOR cursors, we replace the "Reverse Screen"
+        // with black. In this case, we also add an outline around the cursor
+        // so that it is visible against a dark background.
+        if (*mask == kPixelRgbWhite) {
+          if (*dst != 0) {
+            add_outline = true;
+            *dst = kPixelRgbaBlack;
+          } else {
+            *dst = kPixelRgbaTransparent;
+          }
+        } else {
+          *dst = kPixelRgbaBlack ^ *dst;
+        }
+
+        ++dst;
+        ++mask;
+      }
+    }
+    if (add_outline) {
+      AddCursorOutline(width, height,
+                       reinterpret_cast<uint32_t*>(image->data()));
+    }
+  }
+
+  // Pre-multiply the resulting pixels since MouseCursor uses premultiplied
+  // images.
+  AlphaMul(reinterpret_cast<uint32_t*>(image->data()), width, height);
+
+  return new MouseCursor(image.release(), DesktopVector(hotspot_x, hotspot_y));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor.h b/third_party/libwebrtc/modules/desktop_capture/win/cursor.h
new file mode 100644
index 0000000000..54d78164a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_CURSOR_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_CURSOR_H_
+
+#include <windows.h>
+
+namespace webrtc {
+
+class MouseCursor;
+
+// Converts an HCURSOR into a `MouseCursor` instance.
+MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_CURSOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/1_24bpp.cur b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/1_24bpp.cur
new file mode 100644
index 0000000000..27702b825c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/1_24bpp.cur
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/1_32bpp.cur b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/1_32bpp.cur
new file mode 100644
index 0000000000..7e0d8596da
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/1_32bpp.cur
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/1_8bpp.cur b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/1_8bpp.cur
new file mode 100644
index 0000000000..fefb09e1a1
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/1_8bpp.cur
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/2_1bpp.cur b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/2_1bpp.cur
new file mode 100644
index 0000000000..4f8a094f31
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/2_1bpp.cur
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/2_32bpp.cur b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/2_32bpp.cur
new file mode 100644
index 0000000000..ac9cdbfbb3
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/2_32bpp.cur
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/3_32bpp.cur b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/3_32bpp.cur
new file mode 100644
index 0000000000..efdbee5415
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/3_32bpp.cur
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/3_4bpp.cur b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/3_4bpp.cur
new file mode 100644
index 0000000000..9678d55446
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_test_data/3_4bpp.cur
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/win/cursor_unittest.cc
new file mode 100644
index 0000000000..23f5d89571
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_unittest.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/cursor.h"
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/win/cursor_unittest_resources.h"
+#include "modules/desktop_capture/win/scoped_gdi_object.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+namespace {
+
+// Loads `left` from resources, converts it to a `MouseCursor` instance and
+// compares pixels with `right`. Returns true of MouseCursor bits match `right`.
+// `right` must be a 32bpp cursor with alpha channel.
+bool ConvertToMouseShapeAndCompare(unsigned left, unsigned right) {
+  HMODULE instance = GetModuleHandle(NULL);
+
+  // Load `left` from the EXE module's resources.
+  win::ScopedCursor cursor(reinterpret_cast<HCURSOR>(
+      LoadImage(instance, MAKEINTRESOURCE(left), IMAGE_CURSOR, 0, 0, 0)));
+  EXPECT_TRUE(cursor != NULL);
+
+  // Convert `cursor` to `mouse_shape`.
+  HDC dc = GetDC(NULL);
+  std::unique_ptr<MouseCursor> mouse_shape(
+      CreateMouseCursorFromHCursor(dc, cursor));
+  ReleaseDC(NULL, dc);
+
+  EXPECT_TRUE(mouse_shape.get());
+
+  // Load `right`.
+  cursor.Set(reinterpret_cast<HCURSOR>(
+      LoadImage(instance, MAKEINTRESOURCE(right), IMAGE_CURSOR, 0, 0, 0)));
+
+  ICONINFO iinfo;
+  EXPECT_TRUE(GetIconInfo(cursor, &iinfo));
+  EXPECT_TRUE(iinfo.hbmColor);
+
+  // Make sure the bitmaps will be freed.
+  win::ScopedBitmap scoped_mask(iinfo.hbmMask);
+  win::ScopedBitmap scoped_color(iinfo.hbmColor);
+
+  // Get `scoped_color` dimensions.
+  BITMAP bitmap_info;
+  EXPECT_TRUE(GetObject(scoped_color, sizeof(bitmap_info), &bitmap_info));
+
+  int width = bitmap_info.bmWidth;
+  int height = bitmap_info.bmHeight;
+  EXPECT_TRUE(DesktopSize(width, height).equals(mouse_shape->image()->size()));
+
+  // Get the pixels from `scoped_color`.
+  int size = width * height;
+  std::unique_ptr<uint32_t[]> data(new uint32_t[size]);
+  EXPECT_TRUE(GetBitmapBits(scoped_color, size * sizeof(uint32_t), data.get()));
+
+  // Compare the 32bpp image in `mouse_shape` with the one loaded from `right`.
+  return memcmp(data.get(), mouse_shape->image()->data(),
+                size * sizeof(uint32_t)) == 0;
+}
+
+}  // namespace
+
+TEST(MouseCursorTest, MatchCursors) {
+  EXPECT_TRUE(
+      ConvertToMouseShapeAndCompare(IDD_CURSOR1_24BPP, IDD_CURSOR1_32BPP));
+
+  EXPECT_TRUE(
+      ConvertToMouseShapeAndCompare(IDD_CURSOR1_8BPP, IDD_CURSOR1_32BPP));
+
+  EXPECT_TRUE(
+      ConvertToMouseShapeAndCompare(IDD_CURSOR2_1BPP, IDD_CURSOR2_32BPP));
+
+  EXPECT_TRUE(
+      ConvertToMouseShapeAndCompare(IDD_CURSOR3_4BPP, IDD_CURSOR3_32BPP));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_unittest_resources.h b/third_party/libwebrtc/modules/desktop_capture/win/cursor_unittest_resources.h
new file mode 100644
index 0000000000..f583554d68
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_unittest_resources.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_CURSOR_UNITTEST_RESOURCES_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_CURSOR_UNITTEST_RESOURCES_H_
+
+#define IDD_CURSOR1_24BPP 101
+#define IDD_CURSOR1_32BPP 102
+#define IDD_CURSOR1_8BPP 103
+
+#define IDD_CURSOR2_1BPP 104
+#define IDD_CURSOR2_32BPP 105
+
+#define IDD_CURSOR3_4BPP 106
+#define IDD_CURSOR3_32BPP 107
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_CURSOR_UNITTEST_RESOURCES_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/cursor_unittest_resources.rc b/third_party/libwebrtc/modules/desktop_capture/win/cursor_unittest_resources.rc
new file mode 100644
index 0000000000..90073791c9
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/cursor_unittest_resources.rc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/cursor_unittest_resources.h"
+
+// These cursors are matched with their less than  32bpp counterparts below.
+IDD_CURSOR1_32BPP CURSOR "cursor_test_data/1_32bpp.cur"
+IDD_CURSOR2_32BPP CURSOR "cursor_test_data/2_32bpp.cur"
+IDD_CURSOR3_32BPP CURSOR "cursor_test_data/3_32bpp.cur"
+
+// Matches IDD_CURSOR1_32BPP.
+IDD_CURSOR1_24BPP CURSOR "cursor_test_data/1_24bpp.cur"
+
+// Matches IDD_CURSOR1_32BPP.
+IDD_CURSOR1_8BPP CURSOR "cursor_test_data/1_8bpp.cur"
+
+// Matches IDD_CURSOR2_32BPP.
+IDD_CURSOR2_1BPP CURSOR "cursor_test_data/2_1bpp.cur"
+
+// Matches IDD_CURSOR3_32BPP.
+IDD_CURSOR3_4BPP CURSOR "cursor_test_data/3_4bpp.cur"
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/d3d_device.cc b/third_party/libwebrtc/modules/desktop_capture/win/d3d_device.cc
new file mode 100644
index 0000000000..3d46117501
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/d3d_device.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/d3d_device.h"
+
+#include <utility>
+
+#include "modules/desktop_capture/win/desktop_capture_utils.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+using Microsoft::WRL::ComPtr;
+
+D3dDevice::D3dDevice() = default;
+D3dDevice::D3dDevice(const D3dDevice& other) = default;
+D3dDevice::D3dDevice(D3dDevice&& other) = default;
+D3dDevice::~D3dDevice() = default;
+
+bool D3dDevice::Initialize(const ComPtr<IDXGIAdapter>& adapter) {
+  dxgi_adapter_ = adapter;
+  if (!dxgi_adapter_) {
+    RTC_LOG(LS_WARNING) << "An empty IDXGIAdapter instance has been received.";
+    return false;
+  }
+
+  D3D_FEATURE_LEVEL feature_level;
+  // Default feature levels contain D3D 9.1 through D3D 11.0.
+  _com_error error = D3D11CreateDevice(
+      adapter.Get(), D3D_DRIVER_TYPE_UNKNOWN, nullptr,
+      D3D11_CREATE_DEVICE_BGRA_SUPPORT | D3D11_CREATE_DEVICE_SINGLETHREADED,
+      nullptr, 0, D3D11_SDK_VERSION, d3d_device_.GetAddressOf(), &feature_level,
+      context_.GetAddressOf());
+  if (error.Error() != S_OK || !d3d_device_ || !context_) {
+    RTC_LOG(LS_WARNING) << "D3D11CreateDevice returned: "
+                        << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+
+  if (feature_level < D3D_FEATURE_LEVEL_11_0) {
+    RTC_LOG(LS_WARNING)
+        << "D3D11CreateDevice returned an instance without DirectX 11 support, "
+        << "level " << feature_level << ". Following initialization may fail.";
+    // D3D_FEATURE_LEVEL_11_0 is not officially documented on MSDN to be a
+    // requirement of Dxgi duplicator APIs.
+  }
+
+  error = d3d_device_.As(&dxgi_device_);
+  if (error.Error() != S_OK || !dxgi_device_) {
+    RTC_LOG(LS_WARNING)
+        << "ID3D11Device is not an implementation of IDXGIDevice, "
+        << "this usually means the system does not support DirectX "
+        << "11. Error received: "
+        << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+
+  return true;
+}
+
+// static
+std::vector<D3dDevice> D3dDevice::EnumDevices() {
+  ComPtr<IDXGIFactory1> factory;
+  _com_error error =
+      CreateDXGIFactory1(__uuidof(IDXGIFactory1),
+                         reinterpret_cast<void**>(factory.GetAddressOf()));
+  if (error.Error() != S_OK || !factory) {
+    RTC_LOG(LS_WARNING) << "Cannot create IDXGIFactory1: "
+                        << desktop_capture::utils::ComErrorToString(error);
+    return std::vector<D3dDevice>();
+  }
+
+  std::vector<D3dDevice> result;
+  for (int i = 0;; i++) {
+    ComPtr<IDXGIAdapter> adapter;
+    error = factory->EnumAdapters(i, adapter.GetAddressOf());
+    if (error.Error() == S_OK) {
+      D3dDevice device;
+      if (device.Initialize(adapter)) {
+        result.push_back(std::move(device));
+      }
+    } else if (error.Error() == DXGI_ERROR_NOT_FOUND) {
+      break;
+    } else {
+      RTC_LOG(LS_WARNING)
+          << "IDXGIFactory1::EnumAdapters returned an unexpected error: "
+          << desktop_capture::utils::ComErrorToString(error);
+    }
+  }
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/d3d_device.h b/third_party/libwebrtc/modules/desktop_capture/win/d3d_device.h
new file mode 100644
index 0000000000..aeb9d1823a
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/d3d_device.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_D3D_DEVICE_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_D3D_DEVICE_H_
+
+#include <comdef.h>
+#include <d3d11.h>
+#include <dxgi.h>
+#include <wrl/client.h>
+
+#include <vector>
+
+namespace webrtc {
+
+// A wrapper of ID3D11Device and its corresponding context and IDXGIAdapter.
+// This class represents one video card in the system.
+class D3dDevice {
+ public:
+  D3dDevice(const D3dDevice& other);
+  D3dDevice(D3dDevice&& other);
+  ~D3dDevice();
+
+  ID3D11Device* d3d_device() const { return d3d_device_.Get(); }
+
+  ID3D11DeviceContext* context() const { return context_.Get(); }
+
+  IDXGIDevice* dxgi_device() const { return dxgi_device_.Get(); }
+
+  IDXGIAdapter* dxgi_adapter() const { return dxgi_adapter_.Get(); }
+
+  // Returns all D3dDevice instances on the system. Returns an empty vector if
+  // anything wrong.
+  static std::vector<D3dDevice> EnumDevices();
+
+ private:
+  // Instances of D3dDevice should only be created by EnumDevices() static
+  // function.
+  D3dDevice();
+
+  // Initializes the D3dDevice from an IDXGIAdapter.
+  bool Initialize(const Microsoft::WRL::ComPtr<IDXGIAdapter>& adapter);
+
+  Microsoft::WRL::ComPtr<ID3D11Device> d3d_device_;
+  Microsoft::WRL::ComPtr<ID3D11DeviceContext> context_;
+  Microsoft::WRL::ComPtr<IDXGIDevice> dxgi_device_;
+  Microsoft::WRL::ComPtr<IDXGIAdapter> dxgi_adapter_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_D3D_DEVICE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/desktop.cc b/third_party/libwebrtc/modules/desktop_capture/win/desktop.cc
new file mode 100644
index 0000000000..4a671dd9ae
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/desktop.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/desktop.h"
+
+#include <vector>
+
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+
+namespace webrtc {
+
+Desktop::Desktop(HDESK desktop, bool own) : desktop_(desktop), own_(own) {}
+
+Desktop::~Desktop() {
+  if (own_ && desktop_ != NULL) {
+    if (!::CloseDesktop(desktop_)) {
+      RTC_LOG(LS_ERROR) << "Failed to close the owned desktop handle: "
+                        << GetLastError();
+    }
+  }
+}
+
+bool Desktop::GetName(std::wstring* desktop_name_out) const {
+  if (desktop_ == NULL)
+    return false;
+
+  DWORD length = 0;
+  int rv = GetUserObjectInformationW(desktop_, UOI_NAME, NULL, 0, &length);
+  if (rv || GetLastError() != ERROR_INSUFFICIENT_BUFFER)
+    abort();
+
+  length /= sizeof(WCHAR);
+  std::vector<WCHAR> buffer(length);
+  if (!GetUserObjectInformationW(desktop_, UOI_NAME, &buffer[0],
+                                 length * sizeof(WCHAR), &length)) {
+    RTC_LOG(LS_ERROR) << "Failed to query the desktop name: " << GetLastError();
+    return false;
+  }
+
+  desktop_name_out->assign(&buffer[0], length / sizeof(WCHAR));
+  return true;
+}
+
+bool Desktop::IsSame(const Desktop& other) const {
+  std::wstring name;
+  if (!GetName(&name))
+    return false;
+
+  std::wstring other_name;
+  if (!other.GetName(&other_name))
+    return false;
+
+  return name == other_name;
+}
+
+bool Desktop::SetThreadDesktop() const {
+  if (!::SetThreadDesktop(desktop_)) {
+    RTC_LOG(LS_ERROR) << "Failed to assign the desktop to the current thread: "
+                      << GetLastError();
+    return false;
+  }
+
+  return true;
+}
+
+Desktop* Desktop::GetDesktop(const WCHAR* desktop_name) {
+  ACCESS_MASK desired_access = DESKTOP_CREATEMENU | DESKTOP_CREATEWINDOW |
+                               DESKTOP_ENUMERATE | DESKTOP_HOOKCONTROL |
+                               DESKTOP_WRITEOBJECTS | DESKTOP_READOBJECTS |
+                               DESKTOP_SWITCHDESKTOP | GENERIC_WRITE;
+  HDESK desktop = OpenDesktopW(desktop_name, 0, FALSE, desired_access);
+  if (desktop == NULL) {
+    RTC_LOG(LS_ERROR) << "Failed to open the desktop '"
+                      << rtc::ToUtf8(desktop_name) << "': " << GetLastError();
+    return NULL;
+  }
+
+  return new Desktop(desktop, true);
+}
+
+Desktop* Desktop::GetInputDesktop() {
+  HDESK desktop = OpenInputDesktop(
+      0, FALSE, GENERIC_READ | GENERIC_WRITE | GENERIC_EXECUTE);
+  if (desktop == NULL)
+    return NULL;
+
+  return new Desktop(desktop, true);
+}
+
+Desktop* Desktop::GetThreadDesktop() {
+  HDESK desktop = ::GetThreadDesktop(GetCurrentThreadId());
+  if (desktop == NULL) {
+    RTC_LOG(LS_ERROR)
+        << "Failed to retrieve the handle of the desktop assigned to "
+           "the current thread: "
+        << GetLastError();
+    return NULL;
+  }
+
+  return new Desktop(desktop, false);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/desktop.h b/third_party/libwebrtc/modules/desktop_capture/win/desktop.h
new file mode 100644
index 0000000000..01bed8592d
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/desktop.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DESKTOP_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DESKTOP_H_
+
+#include <windows.h>
+
+#include <string>
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class RTC_EXPORT Desktop {
+ public:
+  ~Desktop();
+
+  Desktop(const Desktop&) = delete;
+  Desktop& operator=(const Desktop&) = delete;
+
+  // Returns the name of the desktop represented by the object. Return false if
+  // quering the name failed for any reason.
+  bool GetName(std::wstring* desktop_name_out) const;
+
+  // Returns true if `other` has the same name as this desktop. Returns false
+  // in any other case including failing Win32 APIs and uninitialized desktop
+  // handles.
+  bool IsSame(const Desktop& other) const;
+
+  // Assigns the desktop to the current thread. Returns false is the operation
+  // failed for any reason.
+  bool SetThreadDesktop() const;
+
+  // Returns the desktop by its name or NULL if an error occurs.
+  static Desktop* GetDesktop(const wchar_t* desktop_name);
+
+  // Returns the desktop currently receiving user input or NULL if an error
+  // occurs.
+  static Desktop* GetInputDesktop();
+
+  // Returns the desktop currently assigned to the calling thread or NULL if
+  // an error occurs.
+  static Desktop* GetThreadDesktop();
+
+ private:
+  Desktop(HDESK desktop, bool own);
+
+  // The desktop handle.
+  HDESK desktop_;
+
+  // True if `desktop_` must be closed on teardown.
+  bool own_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DESKTOP_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/desktop_capture_utils.cc b/third_party/libwebrtc/modules/desktop_capture/win/desktop_capture_utils.cc
new file mode 100644
index 0000000000..476ddc4aba
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/desktop_capture_utils.cc
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/desktop_capture_utils.h"
+
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace desktop_capture {
+namespace utils {
+
+// Generates a human-readable string from a COM error.
+std::string ComErrorToString(const _com_error& error) {
+  char buffer[1024];
+  rtc::SimpleStringBuilder string_builder(buffer);
+  // Use _bstr_t to simplify the wchar to char conversion for ErrorMessage().
+  _bstr_t error_message(error.ErrorMessage());
+  string_builder.AppendFormat("HRESULT: 0x%08X, Message: %s", error.Error(),
+                              static_cast<const char*>(error_message));
+  return string_builder.str();
+}
+
+}  // namespace utils
+}  // namespace desktop_capture
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/desktop_capture_utils.h b/third_party/libwebrtc/modules/desktop_capture/win/desktop_capture_utils.h
new file mode 100644
index 0000000000..ebf31419ce
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/desktop_capture_utils.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DESKTOP_CAPTURE_UTILS_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DESKTOP_CAPTURE_UTILS_H_
+
+#include <comdef.h>
+
+#include <string>
+
+namespace webrtc {
+namespace desktop_capture {
+namespace utils {
+
+// Generates a human-readable string from a COM error.
+std::string ComErrorToString(const _com_error& error);
+
+}  // namespace utils
+}  // namespace desktop_capture
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DESKTOP_CAPTURE_UTILS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/display_configuration_monitor.cc b/third_party/libwebrtc/modules/desktop_capture/win/display_configuration_monitor.cc
new file mode 100644
index 0000000000..52d89214d8
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/display_configuration_monitor.cc
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/display_configuration_monitor.h"
+
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+
+namespace webrtc {
+
+bool DisplayConfigurationMonitor::IsChanged() {
+  DesktopRect rect = GetFullscreenRect();
+  if (!initialized_) {
+    initialized_ = true;
+    rect_ = rect;
+    return false;
+  }
+
+  if (rect.equals(rect_)) {
+    return false;
+  }
+
+  rect_ = rect;
+  return true;
+}
+
+void DisplayConfigurationMonitor::Reset() {
+  initialized_ = false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/display_configuration_monitor.h b/third_party/libwebrtc/modules/desktop_capture/win/display_configuration_monitor.h
new file mode 100644
index 0000000000..39c211cfbe
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/display_configuration_monitor.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DISPLAY_CONFIGURATION_MONITOR_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DISPLAY_CONFIGURATION_MONITOR_H_
+
+#include "modules/desktop_capture/desktop_geometry.h"
+
+namespace webrtc {
+
+// A passive monitor to detect the change of display configuration on a Windows
+// system.
+// TODO(zijiehe): Also check for pixel format changes.
+class DisplayConfigurationMonitor {
+ public:
+  // Checks whether the change of display configuration has happened after last
+  // IsChanged() call. This function won't return true for the first time after
+  // constructor or Reset() call.
+  bool IsChanged();
+
+  // Resets to the initial state.
+  void Reset();
+
+ private:
+  DesktopRect rect_;
+  bool initialized_ = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DISPLAY_CONFIGURATION_MONITOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.cc b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.cc
new file mode 100644
index 0000000000..88ec4e25bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.cc
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/dxgi_adapter_duplicator.h"
+
+#include <comdef.h>
+#include <dxgi.h>
+
+#include <algorithm>
+
+#include "modules/desktop_capture/win/desktop_capture_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+using Microsoft::WRL::ComPtr;
+
+namespace {
+
+bool IsValidRect(const RECT& rect) {
+  return rect.right > rect.left && rect.bottom > rect.top;
+}
+
+}  // namespace
+
+DxgiAdapterDuplicator::DxgiAdapterDuplicator(const D3dDevice& device)
+    : device_(device) {}
+DxgiAdapterDuplicator::DxgiAdapterDuplicator(DxgiAdapterDuplicator&&) = default;
+DxgiAdapterDuplicator::~DxgiAdapterDuplicator() = default;
+
+bool DxgiAdapterDuplicator::Initialize() {
+  if (DoInitialize()) {
+    return true;
+  }
+  duplicators_.clear();
+  return false;
+}
+
+bool DxgiAdapterDuplicator::DoInitialize() {
+  for (int i = 0;; i++) {
+    ComPtr<IDXGIOutput> output;
+    _com_error error =
+        device_.dxgi_adapter()->EnumOutputs(i, output.GetAddressOf());
+    if (error.Error() == DXGI_ERROR_NOT_FOUND) {
+      break;
+    }
+
+    if (error.Error() == DXGI_ERROR_NOT_CURRENTLY_AVAILABLE) {
+      RTC_LOG(LS_WARNING) << "IDXGIAdapter::EnumOutputs returned "
+                          << "NOT_CURRENTLY_AVAILABLE. This may happen when "
+                          << "running in session 0.";
+      break;
+    }
+
+    if (error.Error() != S_OK || !output) {
+      RTC_LOG(LS_WARNING) << "IDXGIAdapter::EnumOutputs returned an unexpected "
+                          << "result: "
+                          << desktop_capture::utils::ComErrorToString(error);
+      continue;
+    }
+
+    DXGI_OUTPUT_DESC desc;
+    error = output->GetDesc(&desc);
+    if (error.Error() == S_OK) {
+      if (desc.AttachedToDesktop && IsValidRect(desc.DesktopCoordinates)) {
+        ComPtr<IDXGIOutput1> output1;
+        error = output.As(&output1);
+        if (error.Error() != S_OK || !output1) {
+          RTC_LOG(LS_WARNING)
+              << "Failed to convert IDXGIOutput to IDXGIOutput1, this usually "
+              << "means the system does not support DirectX 11";
+          continue;
+        }
+        DxgiOutputDuplicator duplicator(device_, output1, desc);
+        if (!duplicator.Initialize()) {
+          RTC_LOG(LS_WARNING) << "Failed to initialize DxgiOutputDuplicator on "
+                              << "output " << i;
+          continue;
+        }
+
+        duplicators_.push_back(std::move(duplicator));
+        desktop_rect_.UnionWith(duplicators_.back().desktop_rect());
+      } else {
+        RTC_LOG(LS_ERROR) << (desc.AttachedToDesktop ? "Attached" : "Detached")
+                          << " output " << i << " ("
+                          << desc.DesktopCoordinates.top << ", "
+                          << desc.DesktopCoordinates.left << ") - ("
+                          << desc.DesktopCoordinates.bottom << ", "
+                          << desc.DesktopCoordinates.right << ") is ignored.";
+      }
+    } else {
+      RTC_LOG(LS_WARNING) << "Failed to get output description of device " << i
+                          << ", ignore.";
+    }
+  }
+
+  if (duplicators_.empty()) {
+    RTC_LOG(LS_WARNING)
+        << "Cannot initialize any DxgiOutputDuplicator instance.";
+  }
+
+  return !duplicators_.empty();
+}
+
+void DxgiAdapterDuplicator::Setup(Context* context) {
+  RTC_DCHECK(context->contexts.empty());
+  context->contexts.resize(duplicators_.size());
+  for (size_t i = 0; i < duplicators_.size(); i++) {
+    duplicators_[i].Setup(&context->contexts[i]);
+  }
+}
+
+void DxgiAdapterDuplicator::Unregister(const Context* const context) {
+  RTC_DCHECK_EQ(context->contexts.size(), duplicators_.size());
+  for (size_t i = 0; i < duplicators_.size(); i++) {
+    duplicators_[i].Unregister(&context->contexts[i]);
+  }
+}
+
+bool DxgiAdapterDuplicator::Duplicate(Context* context,
+                                      SharedDesktopFrame* target) {
+  RTC_DCHECK_EQ(context->contexts.size(), duplicators_.size());
+  for (size_t i = 0; i < duplicators_.size(); i++) {
+    if (!duplicators_[i].Duplicate(&context->contexts[i],
+                                   duplicators_[i].desktop_rect().top_left(),
+                                   target)) {
+      return false;
+    }
+  }
+  return true;
+}
+
+bool DxgiAdapterDuplicator::DuplicateMonitor(Context* context,
+                                             int monitor_id,
+                                             SharedDesktopFrame* target) {
+  RTC_DCHECK_GE(monitor_id, 0);
+  RTC_DCHECK_LT(monitor_id, duplicators_.size());
+  RTC_DCHECK_EQ(context->contexts.size(), duplicators_.size());
+  return duplicators_[monitor_id].Duplicate(&context->contexts[monitor_id],
+                                            DesktopVector(), target);
+}
+
+DesktopRect DxgiAdapterDuplicator::ScreenRect(int id) const {
+  RTC_DCHECK_GE(id, 0);
+  RTC_DCHECK_LT(id, duplicators_.size());
+  return duplicators_[id].desktop_rect();
+}
+
+const std::string& DxgiAdapterDuplicator::GetDeviceName(int id) const {
+  RTC_DCHECK_GE(id, 0);
+  RTC_DCHECK_LT(id, duplicators_.size());
+  return duplicators_[id].device_name();
+}
+
+int DxgiAdapterDuplicator::screen_count() const {
+  return static_cast<int>(duplicators_.size());
+}
+
+int64_t DxgiAdapterDuplicator::GetNumFramesCaptured() const {
+  int64_t min = INT64_MAX;
+  for (const auto& duplicator : duplicators_) {
+    min = std::min(min, duplicator.num_frames_captured());
+  }
+
+  return min;
+}
+
+void DxgiAdapterDuplicator::TranslateRect(const DesktopVector& position) {
+  desktop_rect_.Translate(position);
+  RTC_DCHECK_GE(desktop_rect_.left(), 0);
+  RTC_DCHECK_GE(desktop_rect_.top(), 0);
+  for (auto& duplicator : duplicators_) {
+    duplicator.TranslateRect(position);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.h b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.h
new file mode 100644
index 0000000000..5931b51f9e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_ADAPTER_DUPLICATOR_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DXGI_ADAPTER_DUPLICATOR_H_
+
+#include <wrl/client.h>
+
+#include <vector>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "modules/desktop_capture/win/d3d_device.h"
+#include "modules/desktop_capture/win/dxgi_context.h"
+#include "modules/desktop_capture/win/dxgi_output_duplicator.h"
+
+namespace webrtc {
+
+// A container of DxgiOutputDuplicators to duplicate monitors attached to a
+// single video card.
+class DxgiAdapterDuplicator {
+ public:
+  using Context = DxgiAdapterContext;
+
+  // Creates an instance of DxgiAdapterDuplicator from a D3dDevice. Only
+  // DxgiDuplicatorController can create an instance.
+  explicit DxgiAdapterDuplicator(const D3dDevice& device);
+
+  // Move constructor, to make it possible to store instances of
+  // DxgiAdapterDuplicator in std::vector<>.
+  DxgiAdapterDuplicator(DxgiAdapterDuplicator&& other);
+
+  ~DxgiAdapterDuplicator();
+
+  // Initializes the DxgiAdapterDuplicator from a D3dDevice.
+  bool Initialize();
+
+  // Sequentially calls Duplicate function of all the DxgiOutputDuplicator
+  // instances owned by this instance, and writes into `target`.
+  bool Duplicate(Context* context, SharedDesktopFrame* target);
+
+  // Captures one monitor and writes into `target`. `monitor_id` should be
+  // between [0, screen_count()).
+  bool DuplicateMonitor(Context* context,
+                        int monitor_id,
+                        SharedDesktopFrame* target);
+
+  // Returns desktop rect covered by this DxgiAdapterDuplicator.
+  DesktopRect desktop_rect() const { return desktop_rect_; }
+
+  // Returns the size of one screen owned by this DxgiAdapterDuplicator. `id`
+  // should be between [0, screen_count()).
+  DesktopRect ScreenRect(int id) const;
+
+  // Returns the device name of one screen owned by this DxgiAdapterDuplicator
+  // in utf8 encoding. `id` should be between [0, screen_count()).
+  const std::string& GetDeviceName(int id) const;
+
+  // Returns the count of screens owned by this DxgiAdapterDuplicator. These
+  // screens can be retrieved by an interger in the range of
+  // [0, screen_count()).
+  int screen_count() const;
+
+  void Setup(Context* context);
+
+  void Unregister(const Context* const context);
+
+  // The minimum num_frames_captured() returned by `duplicators_`.
+  int64_t GetNumFramesCaptured() const;
+
+  // Moves `desktop_rect_` and all underlying `duplicators_`. See
+  // DxgiDuplicatorController::TranslateRect().
+  void TranslateRect(const DesktopVector& position);
+
+ private:
+  bool DoInitialize();
+
+  const D3dDevice device_;
+  std::vector<DxgiOutputDuplicator> duplicators_;
+  DesktopRect desktop_rect_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DXGI_ADAPTER_DUPLICATOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_context.cc b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_context.cc
new file mode 100644
index 0000000000..c18b238f03
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_context.cc
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/dxgi_context.h"
+
+#include "modules/desktop_capture/win/dxgi_duplicator_controller.h"
+
+namespace webrtc {
+
+DxgiAdapterContext::DxgiAdapterContext() = default;
+DxgiAdapterContext::DxgiAdapterContext(const DxgiAdapterContext& context) =
+    default;
+DxgiAdapterContext::~DxgiAdapterContext() = default;
+
+DxgiFrameContext::DxgiFrameContext() = default;
+
+DxgiFrameContext::~DxgiFrameContext() {
+  Reset();
+}
+
+void DxgiFrameContext::Reset() {
+  DxgiDuplicatorController::Instance()->Unregister(this);
+  controller_id = 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_context.h b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_context.h
new file mode 100644
index 0000000000..59c2112db5
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_context.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_CONTEXT_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DXGI_CONTEXT_H_
+
+#include <vector>
+
+#include "modules/desktop_capture/desktop_region.h"
+
+namespace webrtc {
+
+// A DxgiOutputContext stores the status of a single DxgiFrame of
+// DxgiOutputDuplicator.
+struct DxgiOutputContext final {
+  // The updated region DxgiOutputDuplicator::DetectUpdatedRegion() output
+  // during last Duplicate() function call. It's always relative to the (0, 0).
+  DesktopRegion updated_region;
+};
+
+// A DxgiAdapterContext stores the status of a single DxgiFrame of
+// DxgiAdapterDuplicator.
+struct DxgiAdapterContext final {
+  DxgiAdapterContext();
+  DxgiAdapterContext(const DxgiAdapterContext& other);
+  ~DxgiAdapterContext();
+
+  // Child DxgiOutputContext belongs to this AdapterContext.
+  std::vector<DxgiOutputContext> contexts;
+};
+
+// A DxgiFrameContext stores the status of a single DxgiFrame of
+// DxgiDuplicatorController.
+struct DxgiFrameContext final {
+ public:
+  DxgiFrameContext();
+  // Unregister this Context instance from DxgiDuplicatorController during
+  // destructing.
+  ~DxgiFrameContext();
+
+  // Reset current Context, so it will be reinitialized next time.
+  void Reset();
+
+  // A Context will have an exactly same `controller_id` as
+  // DxgiDuplicatorController, to ensure it has been correctly setted up after
+  // each DxgiDuplicatorController::Initialize().
+  int controller_id = 0;
+
+  // Child DxgiAdapterContext belongs to this DxgiFrameContext.
+  std::vector<DxgiAdapterContext> contexts;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DXGI_CONTEXT_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.cc b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.cc
new file mode 100644
index 0000000000..a776896f6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.cc
@@ -0,0 +1,514 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/dxgi_duplicator_controller.h"
+
+#include <windows.h>
+
+#include <algorithm>
+#include <string>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/win/dxgi_frame.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/sleep.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr DWORD kInvalidSessionId = 0xFFFFFFFF;
+
+DWORD GetCurrentSessionId() {
+  DWORD session_id = kInvalidSessionId;
+  if (!::ProcessIdToSessionId(::GetCurrentProcessId(), &session_id)) {
+    RTC_LOG(LS_WARNING)
+        << "Failed to retrieve current session Id, current binary "
+           "may not have required priviledge.";
+  }
+  return session_id;
+}
+
+bool IsConsoleSession() {
+  return WTSGetActiveConsoleSessionId() == GetCurrentSessionId();
+}
+
+}  // namespace
+
+// static
+std::string DxgiDuplicatorController::ResultName(
+    DxgiDuplicatorController::Result result) {
+  switch (result) {
+    case Result::SUCCEEDED:
+      return "Succeeded";
+    case Result::UNSUPPORTED_SESSION:
+      return "Unsupported session";
+    case Result::FRAME_PREPARE_FAILED:
+      return "Frame preparation failed";
+    case Result::INITIALIZATION_FAILED:
+      return "Initialization failed";
+    case Result::DUPLICATION_FAILED:
+      return "Duplication failed";
+    case Result::INVALID_MONITOR_ID:
+      return "Invalid monitor id";
+    default:
+      return "Unknown error";
+  }
+}
+
+// static
+rtc::scoped_refptr<DxgiDuplicatorController>
+DxgiDuplicatorController::Instance() {
+  // The static instance won't be deleted to ensure it can be used by other
+  // threads even during program exiting.
+  static DxgiDuplicatorController* instance = new DxgiDuplicatorController();
+  return rtc::scoped_refptr<DxgiDuplicatorController>(instance);
+}
+
+// static
+bool DxgiDuplicatorController::IsCurrentSessionSupported() {
+  DWORD current_session_id = GetCurrentSessionId();
+  return current_session_id != kInvalidSessionId && current_session_id != 0;
+}
+
+DxgiDuplicatorController::DxgiDuplicatorController() : refcount_(0) {}
+
+void DxgiDuplicatorController::AddRef() {
+  int refcount = (++refcount_);
+  RTC_DCHECK(refcount > 0);
+}
+
+void DxgiDuplicatorController::Release() {
+  int refcount = (--refcount_);
+  RTC_DCHECK(refcount >= 0);
+  if (refcount == 0) {
+    RTC_LOG(LS_WARNING) << "Count of references reaches zero, "
+                           "DxgiDuplicatorController will be unloaded.";
+    Unload();
+  }
+}
+
+bool DxgiDuplicatorController::IsSupported() {
+  MutexLock lock(&mutex_);
+  return Initialize();
+}
+
+bool DxgiDuplicatorController::RetrieveD3dInfo(D3dInfo* info) {
+  bool result = false;
+  {
+    MutexLock lock(&mutex_);
+    result = Initialize();
+    *info = d3d_info_;
+  }
+  if (!result) {
+    RTC_LOG(LS_WARNING) << "Failed to initialize DXGI components, the D3dInfo "
+                           "retrieved may not accurate or out of date.";
+  }
+  return result;
+}
+
+DxgiDuplicatorController::Result DxgiDuplicatorController::Duplicate(
+    DxgiFrame* frame) {
+  return DoDuplicate(frame, -1);
+}
+
+DxgiDuplicatorController::Result DxgiDuplicatorController::DuplicateMonitor(
+    DxgiFrame* frame,
+    int monitor_id) {
+  RTC_DCHECK_GE(monitor_id, 0);
+  return DoDuplicate(frame, monitor_id);
+}
+
+DesktopVector DxgiDuplicatorController::dpi() {
+  MutexLock lock(&mutex_);
+  if (Initialize()) {
+    return dpi_;
+  }
+  return DesktopVector();
+}
+
+int DxgiDuplicatorController::ScreenCount() {
+  MutexLock lock(&mutex_);
+  if (Initialize()) {
+    return ScreenCountUnlocked();
+  }
+  return 0;
+}
+
+bool DxgiDuplicatorController::GetDeviceNames(
+    std::vector<std::string>* output) {
+  MutexLock lock(&mutex_);
+  if (Initialize()) {
+    GetDeviceNamesUnlocked(output);
+    return true;
+  }
+  return false;
+}
+
+DxgiDuplicatorController::Result DxgiDuplicatorController::DoDuplicate(
+    DxgiFrame* frame,
+    int monitor_id) {
+  RTC_DCHECK(frame);
+  MutexLock lock(&mutex_);
+
+  // The dxgi components and APIs do not update the screen resolution without
+  // a reinitialization. So we use the GetDC() function to retrieve the screen
+  // resolution to decide whether dxgi components need to be reinitialized.
+  // If the screen resolution changed, it's very likely the next Duplicate()
+  // function call will fail because of a missing monitor or the frame size is
+  // not enough to store the output. So we reinitialize dxgi components in-place
+  // to avoid a capture failure.
+  // But there is no guarantee GetDC() function returns the same resolution as
+  // dxgi APIs, we still rely on dxgi components to return the output frame
+  // size.
+  // TODO(zijiehe): Confirm whether IDXGIOutput::GetDesc() and
+  // IDXGIOutputDuplication::GetDesc() can detect the resolution change without
+  // reinitialization.
+  if (display_configuration_monitor_.IsChanged()) {
+    Deinitialize();
+  }
+
+  if (!Initialize()) {
+    if (succeeded_duplications_ == 0 && !IsCurrentSessionSupported()) {
+      RTC_LOG(LS_WARNING) << "Current binary is running in session 0. DXGI "
+                             "components cannot be initialized.";
+      return Result::UNSUPPORTED_SESSION;
+    }
+
+    // Cannot initialize COM components now, display mode may be changing.
+    return Result::INITIALIZATION_FAILED;
+  }
+
+  if (!frame->Prepare(SelectedDesktopSize(monitor_id), monitor_id)) {
+    return Result::FRAME_PREPARE_FAILED;
+  }
+
+  frame->frame()->mutable_updated_region()->Clear();
+
+  if (DoDuplicateUnlocked(frame->context(), monitor_id, frame->frame())) {
+    succeeded_duplications_++;
+    return Result::SUCCEEDED;
+  }
+  if (monitor_id >= ScreenCountUnlocked()) {
+    // It's a user error to provide a `monitor_id` larger than screen count. We
+    // do not need to deinitialize.
+    return Result::INVALID_MONITOR_ID;
+  }
+
+  // If the `monitor_id` is valid, but DoDuplicateUnlocked() failed, something
+  // must be wrong from capturer APIs. We should Deinitialize().
+  Deinitialize();
+  return Result::DUPLICATION_FAILED;
+}
+
+void DxgiDuplicatorController::Unload() {
+  MutexLock lock(&mutex_);
+  Deinitialize();
+}
+
+void DxgiDuplicatorController::Unregister(const Context* const context) {
+  MutexLock lock(&mutex_);
+  if (ContextExpired(context)) {
+    // The Context has not been setup after a recent initialization, so it
+    // should not been registered in duplicators.
+    return;
+  }
+  for (size_t i = 0; i < duplicators_.size(); i++) {
+    duplicators_[i].Unregister(&context->contexts[i]);
+  }
+}
+
+bool DxgiDuplicatorController::Initialize() {
+  if (!duplicators_.empty()) {
+    return true;
+  }
+
+  if (DoInitialize()) {
+    return true;
+  }
+  Deinitialize();
+  return false;
+}
+
+bool DxgiDuplicatorController::DoInitialize() {
+  RTC_DCHECK(desktop_rect_.is_empty());
+  RTC_DCHECK(duplicators_.empty());
+
+  d3d_info_.min_feature_level = static_cast<D3D_FEATURE_LEVEL>(0);
+  d3d_info_.max_feature_level = static_cast<D3D_FEATURE_LEVEL>(0);
+
+  std::vector<D3dDevice> devices = D3dDevice::EnumDevices();
+  if (devices.empty()) {
+    RTC_LOG(LS_WARNING) << "No D3dDevice found.";
+    return false;
+  }
+
+  for (size_t i = 0; i < devices.size(); i++) {
+    D3D_FEATURE_LEVEL feature_level =
+        devices[i].d3d_device()->GetFeatureLevel();
+    if (d3d_info_.max_feature_level == 0 ||
+        feature_level > d3d_info_.max_feature_level) {
+      d3d_info_.max_feature_level = feature_level;
+    }
+    if (d3d_info_.min_feature_level == 0 ||
+        feature_level < d3d_info_.min_feature_level) {
+      d3d_info_.min_feature_level = feature_level;
+    }
+
+    DxgiAdapterDuplicator duplicator(devices[i]);
+    // There may be several video cards on the system, some of them may not
+    // support IDXGOutputDuplication. But they should not impact others from
+    // taking effect, so we should continually try other adapters. This usually
+    // happens when a non-official virtual adapter is installed on the system.
+    if (!duplicator.Initialize()) {
+      RTC_LOG(LS_WARNING) << "Failed to initialize DxgiAdapterDuplicator on "
+                             "adapter "
+                          << i;
+      continue;
+    }
+    RTC_DCHECK(!duplicator.desktop_rect().is_empty());
+    duplicators_.push_back(std::move(duplicator));
+
+    desktop_rect_.UnionWith(duplicators_.back().desktop_rect());
+  }
+  TranslateRect();
+
+  HDC hdc = GetDC(nullptr);
+  // Use old DPI value if failed.
+  if (hdc) {
+    dpi_.set(GetDeviceCaps(hdc, LOGPIXELSX), GetDeviceCaps(hdc, LOGPIXELSY));
+    ReleaseDC(nullptr, hdc);
+  }
+
+  identity_++;
+
+  if (duplicators_.empty()) {
+    RTC_LOG(LS_WARNING)
+        << "Cannot initialize any DxgiAdapterDuplicator instance.";
+  }
+
+  return !duplicators_.empty();
+}
+
+void DxgiDuplicatorController::Deinitialize() {
+  desktop_rect_ = DesktopRect();
+  duplicators_.clear();
+  display_configuration_monitor_.Reset();
+}
+
+bool DxgiDuplicatorController::ContextExpired(
+    const Context* const context) const {
+  RTC_DCHECK(context);
+  return context->controller_id != identity_ ||
+         context->contexts.size() != duplicators_.size();
+}
+
+void DxgiDuplicatorController::Setup(Context* context) {
+  if (ContextExpired(context)) {
+    RTC_DCHECK(context);
+    context->contexts.clear();
+    context->contexts.resize(duplicators_.size());
+    for (size_t i = 0; i < duplicators_.size(); i++) {
+      duplicators_[i].Setup(&context->contexts[i]);
+    }
+    context->controller_id = identity_;
+  }
+}
+
+bool DxgiDuplicatorController::DoDuplicateUnlocked(Context* context,
+                                                   int monitor_id,
+                                                   SharedDesktopFrame* target) {
+  Setup(context);
+
+  if (!EnsureFrameCaptured(context, target)) {
+    return false;
+  }
+
+  bool result = false;
+  if (monitor_id < 0) {
+    // Capture entire screen.
+    result = DoDuplicateAll(context, target);
+  } else {
+    result = DoDuplicateOne(context, monitor_id, target);
+  }
+
+  if (result) {
+    target->set_dpi(dpi_);
+    return true;
+  }
+
+  return false;
+}
+
+bool DxgiDuplicatorController::DoDuplicateAll(Context* context,
+                                              SharedDesktopFrame* target) {
+  for (size_t i = 0; i < duplicators_.size(); i++) {
+    if (!duplicators_[i].Duplicate(&context->contexts[i], target)) {
+      return false;
+    }
+  }
+  return true;
+}
+
+bool DxgiDuplicatorController::DoDuplicateOne(Context* context,
+                                              int monitor_id,
+                                              SharedDesktopFrame* target) {
+  RTC_DCHECK(monitor_id >= 0);
+  for (size_t i = 0; i < duplicators_.size() && i < context->contexts.size();
+       i++) {
+    if (monitor_id >= duplicators_[i].screen_count()) {
+      monitor_id -= duplicators_[i].screen_count();
+    } else {
+      if (duplicators_[i].DuplicateMonitor(&context->contexts[i], monitor_id,
+                                           target)) {
+        target->set_top_left(duplicators_[i].ScreenRect(monitor_id).top_left());
+        return true;
+      }
+      return false;
+    }
+  }
+  return false;
+}
+
+int64_t DxgiDuplicatorController::GetNumFramesCaptured() const {
+  int64_t min = INT64_MAX;
+  for (const auto& duplicator : duplicators_) {
+    min = std::min(min, duplicator.GetNumFramesCaptured());
+  }
+
+  return min;
+}
+
+DesktopSize DxgiDuplicatorController::desktop_size() const {
+  return desktop_rect_.size();
+}
+
+DesktopRect DxgiDuplicatorController::ScreenRect(int id) const {
+  RTC_DCHECK(id >= 0);
+  for (size_t i = 0; i < duplicators_.size(); i++) {
+    if (id >= duplicators_[i].screen_count()) {
+      id -= duplicators_[i].screen_count();
+    } else {
+      return duplicators_[i].ScreenRect(id);
+    }
+  }
+  return DesktopRect();
+}
+
+int DxgiDuplicatorController::ScreenCountUnlocked() const {
+  int result = 0;
+  for (auto& duplicator : duplicators_) {
+    result += duplicator.screen_count();
+  }
+  return result;
+}
+
+void DxgiDuplicatorController::GetDeviceNamesUnlocked(
+    std::vector<std::string>* output) const {
+  RTC_DCHECK(output);
+  for (auto& duplicator : duplicators_) {
+    for (int i = 0; i < duplicator.screen_count(); i++) {
+      output->push_back(duplicator.GetDeviceName(i));
+    }
+  }
+}
+
+DesktopSize DxgiDuplicatorController::SelectedDesktopSize(
+    int monitor_id) const {
+  if (monitor_id < 0) {
+    return desktop_size();
+  }
+
+  return ScreenRect(monitor_id).size();
+}
+
+bool DxgiDuplicatorController::EnsureFrameCaptured(Context* context,
+                                                   SharedDesktopFrame* target) {
+  // On a modern system, the FPS / monitor refresh rate is usually larger than
+  // or equal to 60. So 17 milliseconds is enough to capture at least one frame.
+  const int64_t ms_per_frame = 17;
+  // Skip frames to ensure a full frame refresh has occurred and the DXGI
+  // machinery is producing frames before this function returns.
+  int64_t frames_to_skip = 1;
+  // The total time out milliseconds for this function. If we cannot get enough
+  // frames during this time interval, this function returns false, and cause
+  // the DXGI components to be reinitialized. This usually should not happen
+  // unless the system is switching display mode when this function is being
+  // called. 500 milliseconds should be enough for ~30 frames.
+  const int64_t timeout_ms = 500;
+
+  if (GetNumFramesCaptured() == 0 && !IsConsoleSession()) {
+    // When capturing a console session, waiting for a single frame is
+    // sufficient to ensure that DXGI output duplication is working. When the
+    // session is not attached to the console, it has been observed that DXGI
+    // may produce up to 4 frames (typically 1-2 though) before stopping. When
+    // this condition occurs, no errors are returned from the output duplication
+    // API, it simply appears that nothing is changing on the screen. Thus for
+    // detached sessions, we need to capture a few extra frames before we can be
+    // confident that output duplication was initialized properly.
+    frames_to_skip = 5;
+  }
+
+  if (GetNumFramesCaptured() >= frames_to_skip) {
+    return true;
+  }
+
+  std::unique_ptr<SharedDesktopFrame> fallback_frame;
+  SharedDesktopFrame* shared_frame = nullptr;
+  if (target->size().width() >= desktop_size().width() &&
+      target->size().height() >= desktop_size().height()) {
+    // `target` is large enough to cover entire screen, we do not need to use
+    // `fallback_frame`.
+    shared_frame = target;
+  } else {
+    fallback_frame = SharedDesktopFrame::Wrap(
+        std::unique_ptr<DesktopFrame>(new BasicDesktopFrame(desktop_size())));
+    shared_frame = fallback_frame.get();
+  }
+
+  const int64_t start_ms = rtc::TimeMillis();
+  while (GetNumFramesCaptured() < frames_to_skip) {
+    if (!DoDuplicateAll(context, shared_frame)) {
+      return false;
+    }
+
+    // Calling DoDuplicateAll() may change the number of frames captured.
+    if (GetNumFramesCaptured() >= frames_to_skip) {
+      break;
+    }
+
+    if (rtc::TimeMillis() - start_ms > timeout_ms) {
+      RTC_LOG(LS_ERROR) << "Failed to capture " << frames_to_skip
+                        << " frames "
+                           "within "
+                        << timeout_ms << " milliseconds.";
+      return false;
+    }
+
+    // Sleep `ms_per_frame` before attempting to capture the next frame to
+    // ensure the video adapter has time to update the screen.
+    webrtc::SleepMs(ms_per_frame);
+  }
+  return true;
+}
+
+void DxgiDuplicatorController::TranslateRect() {
+  const DesktopVector position =
+      DesktopVector().subtract(desktop_rect_.top_left());
+  desktop_rect_.Translate(position);
+  for (auto& duplicator : duplicators_) {
+    duplicator.TranslateRect(position);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.h b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.h
new file mode 100644
index 0000000000..88c2939187
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_duplicator_controller.h
@@ -0,0 +1,253 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_DUPLICATOR_CONTROLLER_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DXGI_DUPLICATOR_CONTROLLER_H_
+
+#include <d3dcommon.h>
+
+#include <atomic>
+#include <string>
+#include <vector>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "modules/desktop_capture/win/d3d_device.h"
+#include "modules/desktop_capture/win/display_configuration_monitor.h"
+#include "modules/desktop_capture/win/dxgi_adapter_duplicator.h"
+#include "modules/desktop_capture/win/dxgi_context.h"
+#include "modules/desktop_capture/win/dxgi_frame.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// A controller for all the objects we need to call Windows DirectX capture APIs
+// It's a singleton because only one IDXGIOutputDuplication instance per monitor
+// is allowed per application.
+//
+// Consumers should create a DxgiDuplicatorController::Context and keep it
+// throughout their lifetime, and pass it when calling Duplicate(). Consumers
+// can also call IsSupported() to determine whether the system supports DXGI
+// duplicator or not. If a previous IsSupported() function call returns true,
+// but a later Duplicate() returns false, this usually means the display mode is
+// changing. Consumers should retry after a while. (Typically 50 milliseconds,
+// but according to hardware performance, this time may vary.)
+// The underyling DxgiOutputDuplicators may take an additional reference on the
+// frame passed in to the Duplicate methods so that they can guarantee delivery
+// of new frames when requested; since if there have been no updates to the
+// surface, they may be unable to capture a frame.
+class RTC_EXPORT DxgiDuplicatorController {
+ public:
+  using Context = DxgiFrameContext;
+
+  // A collection of D3d information we are interested on, which may impact
+  // capturer performance or reliability.
+  struct D3dInfo {
+    // Each video adapter has its own D3D_FEATURE_LEVEL, so this structure
+    // contains the minimum and maximium D3D_FEATURE_LEVELs current system
+    // supports.
+    // Both fields can be 0, which is the default value to indicate no valid
+    // D3D_FEATURE_LEVEL has been retrieved from underlying OS APIs.
+    D3D_FEATURE_LEVEL min_feature_level;
+    D3D_FEATURE_LEVEL max_feature_level;
+
+    // TODO(zijiehe): Add more fields, such as manufacturer name, mode, driver
+    // version.
+  };
+
+  enum class Result {
+    SUCCEEDED,
+    UNSUPPORTED_SESSION,
+    FRAME_PREPARE_FAILED,
+    INITIALIZATION_FAILED,
+    DUPLICATION_FAILED,
+    INVALID_MONITOR_ID,
+  };
+
+  // Converts `result` into user-friendly string representation. The return
+  // value should not be used to identify error types.
+  static std::string ResultName(Result result);
+
+  // Returns the singleton instance of DxgiDuplicatorController.
+  static rtc::scoped_refptr<DxgiDuplicatorController> Instance();
+
+  // See ScreenCapturerWinDirectx::IsCurrentSessionSupported().
+  static bool IsCurrentSessionSupported();
+
+  // All the following public functions implicitly call Initialize() function.
+
+  // Detects whether the system supports DXGI based capturer.
+  bool IsSupported();
+
+  // Returns a copy of D3dInfo composed by last Initialize() function call. This
+  // function always copies the latest information into `info`. But once the
+  // function returns false, the information in `info` may not accurate.
+  bool RetrieveD3dInfo(D3dInfo* info);
+
+  // Captures current screen and writes into `frame`. May retain a reference to
+  // `frame`'s underlying |SharedDesktopFrame|.
+  // TODO(zijiehe): Windows cannot guarantee the frames returned by each
+  // IDXGIOutputDuplication are synchronized. But we are using a totally
+  // different threading model than the way Windows suggested, it's hard to
+  // synchronize them manually. We should find a way to do it.
+  Result Duplicate(DxgiFrame* frame);
+
+  // Captures one monitor and writes into target. `monitor_id` should >= 0. If
+  // `monitor_id` is greater than the total screen count of all the Duplicators,
+  // this function returns false. May retain a reference to `frame`'s underlying
+  // |SharedDesktopFrame|.
+  Result DuplicateMonitor(DxgiFrame* frame, int monitor_id);
+
+  // Returns dpi of current system. Returns an empty DesktopVector if system
+  // does not support DXGI based capturer.
+  DesktopVector dpi();
+
+  // Returns the count of screens on the system. These screens can be retrieved
+  // by an integer in the range of [0, ScreenCount()). If system does not
+  // support DXGI based capturer, this function returns 0.
+  int ScreenCount();
+
+  // Returns the device names of all screens on the system in utf8 encoding.
+  // These screens can be retrieved by an integer in the range of
+  // [0, output->size()). If system does not support DXGI based capturer, this
+  // function returns false.
+  bool GetDeviceNames(std::vector<std::string>* output);
+
+ private:
+  // DxgiFrameContext calls private Unregister(Context*) function in Reset().
+  friend void DxgiFrameContext::Reset();
+
+  // scoped_refptr<DxgiDuplicatorController> accesses private AddRef() and
+  // Release() functions.
+  friend class rtc::scoped_refptr<DxgiDuplicatorController>;
+
+  // A private constructor to ensure consumers to use
+  // DxgiDuplicatorController::Instance().
+  DxgiDuplicatorController();
+
+  // Not implemented: The singleton DxgiDuplicatorController instance should not
+  // be deleted.
+  ~DxgiDuplicatorController();
+
+  // RefCountedInterface implementations.
+  void AddRef();
+  void Release();
+
+  // Does the real duplication work. Setting `monitor_id` < 0 to capture entire
+  // screen. This function calls Initialize(). And if the duplication failed,
+  // this function calls Deinitialize() to ensure the Dxgi components can be
+  // reinitialized next time.
+  Result DoDuplicate(DxgiFrame* frame, int monitor_id);
+
+  // Unload all the DXGI components and releases the resources. This function
+  // wraps Deinitialize() with `mutex_`.
+  void Unload();
+
+  // Unregisters Context from this instance and all DxgiAdapterDuplicator(s)
+  // it owns.
+  void Unregister(const Context* const context);
+
+  // All functions below should be called in `mutex_` locked scope and should be
+  // after a successful Initialize().
+
+  // If current instance has not been initialized, executes DoInitialize()
+  // function, and returns initialize result. Otherwise directly returns true.
+  // This function may calls Deinitialize() if initialization failed.
+  bool Initialize() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Does the real initialization work, this function should only be called in
+  // Initialize().
+  bool DoInitialize() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Clears all COM components referred by this instance. So next Duplicate()
+  // call will eventually initialize this instance again.
+  void Deinitialize() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // A helper function to check whether a Context has been expired.
+  bool ContextExpired(const Context* const context) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Updates Context if needed.
+  void Setup(Context* context) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  bool DoDuplicateUnlocked(Context* context,
+                           int monitor_id,
+                           SharedDesktopFrame* target)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Captures all monitors.
+  bool DoDuplicateAll(Context* context, SharedDesktopFrame* target)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Captures one monitor.
+  bool DoDuplicateOne(Context* context,
+                      int monitor_id,
+                      SharedDesktopFrame* target)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // The minimum GetNumFramesCaptured() returned by `duplicators_`.
+  int64_t GetNumFramesCaptured() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Returns a DesktopSize to cover entire `desktop_rect_`.
+  DesktopSize desktop_size() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Returns the size of one screen. `id` should be >= 0. If system does not
+  // support DXGI based capturer, or `id` is greater than the total screen count
+  // of all the Duplicators, this function returns an empty DesktopRect.
+  DesktopRect ScreenRect(int id) const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  int ScreenCountUnlocked() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  void GetDeviceNamesUnlocked(std::vector<std::string>* output) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Returns the desktop size of the selected screen `monitor_id`. Setting
+  // `monitor_id` < 0 to return the entire screen size.
+  DesktopSize SelectedDesktopSize(int monitor_id) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Retries DoDuplicateAll() for several times until GetNumFramesCaptured() is
+  // large enough. Returns false if DoDuplicateAll() returns false, or
+  // GetNumFramesCaptured() has never reached the requirement.
+  // According to http://crbug.com/682112, dxgi capturer returns a black frame
+  // during first several capture attempts.
+  bool EnsureFrameCaptured(Context* context, SharedDesktopFrame* target)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Moves `desktop_rect_` and all underlying `duplicators_`, putting top left
+  // corner of the desktop at (0, 0). This is necessary because DXGI_OUTPUT_DESC
+  // may return negative coordinates. Called from DoInitialize() after all
+  // DxgiAdapterDuplicator and DxgiOutputDuplicator instances are initialized.
+  void TranslateRect() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // The count of references which are now "living".
+  std::atomic_int refcount_;
+
+  // This lock must be locked whenever accessing any of the following objects.
+  Mutex mutex_;
+
+  // A self-incremented integer to compare with the one in Context. It ensures
+  // a Context instance is always initialized after DxgiDuplicatorController.
+  int identity_ RTC_GUARDED_BY(mutex_) = 0;
+  DesktopRect desktop_rect_ RTC_GUARDED_BY(mutex_);
+  DesktopVector dpi_ RTC_GUARDED_BY(mutex_);
+  std::vector<DxgiAdapterDuplicator> duplicators_ RTC_GUARDED_BY(mutex_);
+  D3dInfo d3d_info_ RTC_GUARDED_BY(mutex_);
+  DisplayConfigurationMonitor display_configuration_monitor_
+      RTC_GUARDED_BY(mutex_);
+  // A number to indicate how many succeeded duplications have been performed.
+  uint32_t succeeded_duplications_ RTC_GUARDED_BY(mutex_) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DXGI_DUPLICATOR_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_frame.cc b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_frame.cc
new file mode 100644
index 0000000000..13d5b4b62e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_frame.cc
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/dxgi_frame.h"
+
+#include <string.h>
+
+#include <utility>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/win/dxgi_duplicator_controller.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+DxgiFrame::DxgiFrame(SharedMemoryFactory* factory) : factory_(factory) {}
+
+DxgiFrame::~DxgiFrame() = default;
+
+bool DxgiFrame::Prepare(DesktopSize size, DesktopCapturer::SourceId source_id) {
+  if (source_id != source_id_) {
+    // Once the source has been changed, the entire source should be copied.
+    source_id_ = source_id;
+    context_.Reset();
+  }
+
+  if (resolution_tracker_.SetResolution(size)) {
+    // Once the output size changed, recreate the SharedDesktopFrame.
+    frame_.reset();
+  }
+
+  if (!frame_) {
+    std::unique_ptr<DesktopFrame> frame;
+    if (factory_) {
+      frame = SharedMemoryDesktopFrame::Create(size, factory_);
+
+      if (!frame) {
+        RTC_LOG(LS_WARNING) << "DxgiFrame cannot create a new DesktopFrame.";
+        return false;
+      }
+
+      // DirectX capturer won't paint each pixel in the frame due to its one
+      // capturer per monitor design. So once the new frame is created, we
+      // should clear it to avoid the legacy image to be remained on it. See
+      // http://crbug.com/708766.
+      RTC_DCHECK_EQ(frame->stride(),
+                    frame->size().width() * DesktopFrame::kBytesPerPixel);
+      memset(frame->data(), 0, frame->stride() * frame->size().height());
+    } else {
+      frame.reset(new BasicDesktopFrame(size));
+    }
+
+    frame_ = SharedDesktopFrame::Wrap(std::move(frame));
+  }
+
+  return !!frame_;
+}
+
+SharedDesktopFrame* DxgiFrame::frame() const {
+  RTC_DCHECK(frame_);
+  return frame_.get();
+}
+
+DxgiFrame::Context* DxgiFrame::context() {
+  RTC_DCHECK(frame_);
+  return &context_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_frame.h b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_frame.h
new file mode 100644
index 0000000000..6a9ce868a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_frame.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_FRAME_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DXGI_FRAME_H_
+
+#include <memory>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/resolution_tracker.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "modules/desktop_capture/shared_memory.h"
+#include "modules/desktop_capture/win/dxgi_context.h"
+
+namespace webrtc {
+
+class DxgiDuplicatorController;
+
+// A pair of a SharedDesktopFrame and a DxgiDuplicatorController::Context for
+// the client of DxgiDuplicatorController.
+class DxgiFrame final {
+ public:
+  using Context = DxgiFrameContext;
+
+  // DxgiFrame does not take ownership of `factory`, consumers should ensure it
+  // outlives this instance. nullptr is acceptable.
+  explicit DxgiFrame(SharedMemoryFactory* factory);
+  ~DxgiFrame();
+
+  // Should not be called if Prepare() is not executed or returns false.
+  SharedDesktopFrame* frame() const;
+
+ private:
+  // Allows DxgiDuplicatorController to access Prepare() and context() function
+  // as well as Context class.
+  friend class DxgiDuplicatorController;
+
+  // Prepares current instance with desktop size and source id.
+  bool Prepare(DesktopSize size, DesktopCapturer::SourceId source_id);
+
+  // Should not be called if Prepare() is not executed or returns false.
+  Context* context();
+
+  SharedMemoryFactory* const factory_;
+  ResolutionTracker resolution_tracker_;
+  DesktopCapturer::SourceId source_id_ = kFullDesktopScreenId;
+  std::unique_ptr<SharedDesktopFrame> frame_;
+  Context context_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DXGI_FRAME_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_output_duplicator.cc b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_output_duplicator.cc
new file mode 100644
index 0000000000..caad6f43d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_output_duplicator.cc
@@ -0,0 +1,390 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/dxgi_output_duplicator.h"
+
+#include <dxgi.h>
+#include <dxgiformat.h>
+#include <string.h>
+#include <unknwn.h>
+#include <windows.h>
+
+#include <algorithm>
+
+#include "modules/desktop_capture/win/desktop_capture_utils.h"
+#include "modules/desktop_capture/win/dxgi_texture_mapping.h"
+#include "modules/desktop_capture/win/dxgi_texture_staging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/win32.h"
+
+namespace webrtc {
+
+using Microsoft::WRL::ComPtr;
+
+namespace {
+
+// Timeout for AcquireNextFrame() call.
+// DxgiDuplicatorController leverages external components to do the capture
+// scheduling. So here DxgiOutputDuplicator does not need to actively wait for a
+// new frame.
+const int kAcquireTimeoutMs = 0;
+
+DesktopRect RECTToDesktopRect(const RECT& rect) {
+  return DesktopRect::MakeLTRB(rect.left, rect.top, rect.right, rect.bottom);
+}
+
+Rotation DxgiRotationToRotation(DXGI_MODE_ROTATION rotation) {
+  switch (rotation) {
+    case DXGI_MODE_ROTATION_IDENTITY:
+    case DXGI_MODE_ROTATION_UNSPECIFIED:
+      return Rotation::CLOCK_WISE_0;
+    case DXGI_MODE_ROTATION_ROTATE90:
+      return Rotation::CLOCK_WISE_90;
+    case DXGI_MODE_ROTATION_ROTATE180:
+      return Rotation::CLOCK_WISE_180;
+    case DXGI_MODE_ROTATION_ROTATE270:
+      return Rotation::CLOCK_WISE_270;
+  }
+
+  RTC_DCHECK_NOTREACHED();
+  return Rotation::CLOCK_WISE_0;
+}
+
+}  // namespace
+
+DxgiOutputDuplicator::DxgiOutputDuplicator(const D3dDevice& device,
+                                           const ComPtr<IDXGIOutput1>& output,
+                                           const DXGI_OUTPUT_DESC& desc)
+    : device_(device),
+      output_(output),
+      device_name_(rtc::ToUtf8(desc.DeviceName)),
+      desktop_rect_(RECTToDesktopRect(desc.DesktopCoordinates)) {
+  RTC_DCHECK(output_);
+  RTC_DCHECK(!desktop_rect_.is_empty());
+  RTC_DCHECK_GT(desktop_rect_.width(), 0);
+  RTC_DCHECK_GT(desktop_rect_.height(), 0);
+}
+
+DxgiOutputDuplicator::DxgiOutputDuplicator(DxgiOutputDuplicator&& other) =
+    default;
+
+DxgiOutputDuplicator::~DxgiOutputDuplicator() {
+  if (duplication_) {
+    duplication_->ReleaseFrame();
+  }
+  texture_.reset();
+}
+
+bool DxgiOutputDuplicator::Initialize() {
+  if (DuplicateOutput()) {
+    if (desc_.DesktopImageInSystemMemory) {
+      texture_.reset(new DxgiTextureMapping(duplication_.Get()));
+    } else {
+      texture_.reset(new DxgiTextureStaging(device_));
+    }
+    return true;
+  } else {
+    duplication_.Reset();
+    return false;
+  }
+}
+
+bool DxgiOutputDuplicator::DuplicateOutput() {
+  RTC_DCHECK(!duplication_);
+  _com_error error =
+      output_->DuplicateOutput(static_cast<IUnknown*>(device_.d3d_device()),
+                               duplication_.GetAddressOf());
+  if (error.Error() != S_OK || !duplication_) {
+    RTC_LOG(LS_WARNING) << "Failed to duplicate output from IDXGIOutput1: "
+                        << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+
+  memset(&desc_, 0, sizeof(desc_));
+  duplication_->GetDesc(&desc_);
+  if (desc_.ModeDesc.Format != DXGI_FORMAT_B8G8R8A8_UNORM) {
+    RTC_LOG(LS_ERROR) << "IDXGIDuplicateOutput does not use RGBA (8 bit) "
+                      << "format, which is required by downstream components, "
+                      << "format is " << desc_.ModeDesc.Format;
+    return false;
+  }
+
+  if (static_cast<int>(desc_.ModeDesc.Width) != desktop_rect_.width() ||
+      static_cast<int>(desc_.ModeDesc.Height) != desktop_rect_.height()) {
+    RTC_LOG(LS_ERROR)
+        << "IDXGIDuplicateOutput does not return a same size as its "
+        << "IDXGIOutput1, size returned by IDXGIDuplicateOutput is "
+        << desc_.ModeDesc.Width << " x " << desc_.ModeDesc.Height
+        << ", size returned by IDXGIOutput1 is " << desktop_rect_.width()
+        << " x " << desktop_rect_.height();
+    return false;
+  }
+
+  rotation_ = DxgiRotationToRotation(desc_.Rotation);
+  unrotated_size_ = RotateSize(desktop_size(), ReverseRotation(rotation_));
+
+  return true;
+}
+
+bool DxgiOutputDuplicator::ReleaseFrame() {
+  RTC_DCHECK(duplication_);
+  _com_error error = duplication_->ReleaseFrame();
+  if (error.Error() != S_OK) {
+    RTC_LOG(LS_ERROR) << "Failed to release frame from IDXGIOutputDuplication: "
+                      << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+  return true;
+}
+
+bool DxgiOutputDuplicator::Duplicate(Context* context,
+                                     DesktopVector offset,
+                                     SharedDesktopFrame* target) {
+  RTC_DCHECK(duplication_);
+  RTC_DCHECK(texture_);
+  RTC_DCHECK(target);
+  if (!DesktopRect::MakeSize(target->size())
+           .ContainsRect(GetTranslatedDesktopRect(offset))) {
+    // target size is not large enough to cover current output region.
+    return false;
+  }
+
+  DXGI_OUTDUPL_FRAME_INFO frame_info;
+  memset(&frame_info, 0, sizeof(frame_info));
+  ComPtr<IDXGIResource> resource;
+  _com_error error = duplication_->AcquireNextFrame(
+      kAcquireTimeoutMs, &frame_info, resource.GetAddressOf());
+  if (error.Error() != S_OK && error.Error() != DXGI_ERROR_WAIT_TIMEOUT) {
+    RTC_LOG(LS_ERROR) << "Failed to capture frame: "
+                      << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+
+  // We need to merge updated region with the one from context, but only spread
+  // updated region from current frame. So keeps a copy of updated region from
+  // context here. The `updated_region` always starts from (0, 0).
+  DesktopRegion updated_region;
+  updated_region.Swap(&context->updated_region);
+  if (error.Error() == S_OK && frame_info.AccumulatedFrames > 0 && resource) {
+    DetectUpdatedRegion(frame_info, &context->updated_region);
+    SpreadContextChange(context);
+    if (!texture_->CopyFrom(frame_info, resource.Get())) {
+      return false;
+    }
+    updated_region.AddRegion(context->updated_region);
+    // TODO(zijiehe): Figure out why clearing context->updated_region() here
+    // triggers screen flickering?
+
+    const DesktopFrame& source = texture_->AsDesktopFrame();
+    if (rotation_ != Rotation::CLOCK_WISE_0) {
+      for (DesktopRegion::Iterator it(updated_region); !it.IsAtEnd();
+           it.Advance()) {
+        // The `updated_region` returned by Windows is rotated, but the `source`
+        // frame is not. So we need to rotate it reversely.
+        const DesktopRect source_rect =
+            RotateRect(it.rect(), desktop_size(), ReverseRotation(rotation_));
+        RotateDesktopFrame(source, source_rect, rotation_, offset, target);
+      }
+    } else {
+      for (DesktopRegion::Iterator it(updated_region); !it.IsAtEnd();
+           it.Advance()) {
+        // The DesktopRect in `target`, starts from offset.
+        DesktopRect dest_rect = it.rect();
+        dest_rect.Translate(offset);
+        target->CopyPixelsFrom(source, it.rect().top_left(), dest_rect);
+      }
+    }
+    last_frame_ = target->Share();
+    last_frame_offset_ = offset;
+    updated_region.Translate(offset.x(), offset.y());
+    target->mutable_updated_region()->AddRegion(updated_region);
+    num_frames_captured_++;
+    return texture_->Release() && ReleaseFrame();
+  }
+
+  if (last_frame_) {
+    // No change since last frame or AcquireNextFrame() timed out, we will
+    // export last frame to the target.
+    for (DesktopRegion::Iterator it(updated_region); !it.IsAtEnd();
+         it.Advance()) {
+      // The DesktopRect in `source`, starts from last_frame_offset_.
+      DesktopRect source_rect = it.rect();
+      // The DesktopRect in `target`, starts from offset.
+      DesktopRect target_rect = source_rect;
+      source_rect.Translate(last_frame_offset_);
+      target_rect.Translate(offset);
+      target->CopyPixelsFrom(*last_frame_, source_rect.top_left(), target_rect);
+    }
+    updated_region.Translate(offset.x(), offset.y());
+    target->mutable_updated_region()->AddRegion(updated_region);
+  } else {
+    // If we were at the very first frame, and capturing failed, the
+    // context->updated_region should be kept unchanged for next attempt.
+    context->updated_region.Swap(&updated_region);
+  }
+  // If AcquireNextFrame() failed with timeout error, we do not need to release
+  // the frame.
+  return error.Error() == DXGI_ERROR_WAIT_TIMEOUT || ReleaseFrame();
+}
+
+DesktopRect DxgiOutputDuplicator::GetTranslatedDesktopRect(
+    DesktopVector offset) const {
+  DesktopRect result(DesktopRect::MakeSize(desktop_size()));
+  result.Translate(offset);
+  return result;
+}
+
+DesktopRect DxgiOutputDuplicator::GetUntranslatedDesktopRect() const {
+  return DesktopRect::MakeSize(desktop_size());
+}
+
+void DxgiOutputDuplicator::DetectUpdatedRegion(
+    const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+    DesktopRegion* updated_region) {
+  if (DoDetectUpdatedRegion(frame_info, updated_region)) {
+    // Make sure even a region returned by Windows API is out of the scope of
+    // desktop_rect_, we still won't export it to the target DesktopFrame.
+    updated_region->IntersectWith(GetUntranslatedDesktopRect());
+  } else {
+    updated_region->SetRect(GetUntranslatedDesktopRect());
+  }
+}
+
+bool DxgiOutputDuplicator::DoDetectUpdatedRegion(
+    const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+    DesktopRegion* updated_region) {
+  RTC_DCHECK(updated_region);
+  updated_region->Clear();
+  if (frame_info.TotalMetadataBufferSize == 0) {
+    // This should not happen, since frame_info.AccumulatedFrames > 0.
+    RTC_LOG(LS_ERROR) << "frame_info.AccumulatedFrames > 0, "
+                      << "but TotalMetadataBufferSize == 0";
+    return false;
+  }
+
+  if (metadata_.size() < frame_info.TotalMetadataBufferSize) {
+    metadata_.clear();  // Avoid data copy
+    metadata_.resize(frame_info.TotalMetadataBufferSize);
+  }
+
+  UINT buff_size = 0;
+  DXGI_OUTDUPL_MOVE_RECT* move_rects =
+      reinterpret_cast<DXGI_OUTDUPL_MOVE_RECT*>(metadata_.data());
+  size_t move_rects_count = 0;
+  _com_error error = duplication_->GetFrameMoveRects(
+      static_cast<UINT>(metadata_.size()), move_rects, &buff_size);
+  if (error.Error() != S_OK) {
+    RTC_LOG(LS_ERROR) << "Failed to get move rectangles: "
+                      << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+  move_rects_count = buff_size / sizeof(DXGI_OUTDUPL_MOVE_RECT);
+
+  RECT* dirty_rects = reinterpret_cast<RECT*>(metadata_.data() + buff_size);
+  size_t dirty_rects_count = 0;
+  error = duplication_->GetFrameDirtyRects(
+      static_cast<UINT>(metadata_.size()) - buff_size, dirty_rects, &buff_size);
+  if (error.Error() != S_OK) {
+    RTC_LOG(LS_ERROR) << "Failed to get dirty rectangles: "
+                      << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+  dirty_rects_count = buff_size / sizeof(RECT);
+
+  while (move_rects_count > 0) {
+    // DirectX capturer API may randomly return unmoved move_rects, which should
+    // be skipped to avoid unnecessary wasting of differing and encoding
+    // resources.
+    // By using testing application it2me_standalone_host_main, this check
+    // reduces average capture time by 0.375% (4.07 -> 4.055), and average
+    // encode time by 0.313% (8.042 -> 8.016) without other impacts.
+    if (move_rects->SourcePoint.x != move_rects->DestinationRect.left ||
+        move_rects->SourcePoint.y != move_rects->DestinationRect.top) {
+      updated_region->AddRect(
+          RotateRect(DesktopRect::MakeXYWH(move_rects->SourcePoint.x,
+                                           move_rects->SourcePoint.y,
+                                           move_rects->DestinationRect.right -
+                                               move_rects->DestinationRect.left,
+                                           move_rects->DestinationRect.bottom -
+                                               move_rects->DestinationRect.top),
+                     unrotated_size_, rotation_));
+      updated_region->AddRect(
+          RotateRect(DesktopRect::MakeLTRB(move_rects->DestinationRect.left,
+                                           move_rects->DestinationRect.top,
+                                           move_rects->DestinationRect.right,
+                                           move_rects->DestinationRect.bottom),
+                     unrotated_size_, rotation_));
+    } else {
+      RTC_LOG(LS_INFO) << "Unmoved move_rect detected, ["
+                       << move_rects->DestinationRect.left << ", "
+                       << move_rects->DestinationRect.top << "] - ["
+                       << move_rects->DestinationRect.right << ", "
+                       << move_rects->DestinationRect.bottom << "].";
+    }
+    move_rects++;
+    move_rects_count--;
+  }
+
+  while (dirty_rects_count > 0) {
+    updated_region->AddRect(RotateRect(
+        DesktopRect::MakeLTRB(dirty_rects->left, dirty_rects->top,
+                              dirty_rects->right, dirty_rects->bottom),
+        unrotated_size_, rotation_));
+    dirty_rects++;
+    dirty_rects_count--;
+  }
+
+  return true;
+}
+
+void DxgiOutputDuplicator::Setup(Context* context) {
+  RTC_DCHECK(context->updated_region.is_empty());
+  // Always copy entire monitor during the first Duplicate() function call.
+  context->updated_region.AddRect(GetUntranslatedDesktopRect());
+  RTC_DCHECK(std::find(contexts_.begin(), contexts_.end(), context) ==
+             contexts_.end());
+  contexts_.push_back(context);
+}
+
+void DxgiOutputDuplicator::Unregister(const Context* const context) {
+  auto it = std::find(contexts_.begin(), contexts_.end(), context);
+  RTC_DCHECK(it != contexts_.end());
+  contexts_.erase(it);
+}
+
+void DxgiOutputDuplicator::SpreadContextChange(const Context* const source) {
+  for (Context* dest : contexts_) {
+    RTC_DCHECK(dest);
+    if (dest != source) {
+      dest->updated_region.AddRegion(source->updated_region);
+    }
+  }
+}
+
+DesktopSize DxgiOutputDuplicator::desktop_size() const {
+  return desktop_rect_.size();
+}
+
+int64_t DxgiOutputDuplicator::num_frames_captured() const {
+#if !defined(NDEBUG)
+  RTC_DCHECK_EQ(!!last_frame_, num_frames_captured_ > 0);
+#endif
+  return num_frames_captured_;
+}
+
+void DxgiOutputDuplicator::TranslateRect(const DesktopVector& position) {
+  desktop_rect_.Translate(position);
+  RTC_DCHECK_GE(desktop_rect_.left(), 0);
+  RTC_DCHECK_GE(desktop_rect_.top(), 0);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_output_duplicator.h b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_output_duplicator.h
new file mode 100644
index 0000000000..df15fe566e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_output_duplicator.h
@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_OUTPUT_DUPLICATOR_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DXGI_OUTPUT_DUPLICATOR_H_
+
+#include <comdef.h>
+#include <dxgi.h>
+#include <dxgi1_2.h>
+#include <wrl/client.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_frame_rotation.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "modules/desktop_capture/win/d3d_device.h"
+#include "modules/desktop_capture/win/dxgi_context.h"
+#include "modules/desktop_capture/win/dxgi_texture.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Duplicates the content on one IDXGIOutput, i.e. one monitor attached to one
+// video card. None of functions in this class is thread-safe.
+class DxgiOutputDuplicator {
+ public:
+  using Context = DxgiOutputContext;
+
+  // Creates an instance of DxgiOutputDuplicator from a D3dDevice and one of its
+  // IDXGIOutput1. Caller must maintain the lifetime of device, to make sure it
+  // outlives this instance. Only DxgiAdapterDuplicator can create an instance.
+  DxgiOutputDuplicator(const D3dDevice& device,
+                       const Microsoft::WRL::ComPtr<IDXGIOutput1>& output,
+                       const DXGI_OUTPUT_DESC& desc);
+
+  // To allow this class to work with vector.
+  DxgiOutputDuplicator(DxgiOutputDuplicator&& other);
+
+  // Destructs this instance. We need to make sure texture_ has been released
+  // before duplication_.
+  ~DxgiOutputDuplicator();
+
+  // Initializes duplication_ object.
+  bool Initialize();
+
+  // Copies the content of current IDXGIOutput to the `target`. To improve the
+  // performance, this function copies only regions merged from
+  // `context`->updated_region and DetectUpdatedRegion(). The `offset` decides
+  // the offset in the `target` where the content should be copied to. i.e. this
+  // function copies the content to the rectangle of (offset.x(), offset.y()) to
+  // (offset.x() + desktop_rect_.width(), offset.y() + desktop_rect_.height()).
+  // Returns false in case of a failure.
+  // May retain a reference to `target` so that a "captured" frame can be
+  // returned in the event that a new frame is not ready to be captured yet.
+  // (Or in other words, if the call to IDXGIOutputDuplication::AcquireNextFrame
+  // indicates that there is not yet a new frame, this is usually because no
+  // updates have occurred to the frame).
+  bool Duplicate(Context* context,
+                 DesktopVector offset,
+                 SharedDesktopFrame* target);
+
+  // Returns the desktop rect covered by this DxgiOutputDuplicator.
+  DesktopRect desktop_rect() const { return desktop_rect_; }
+
+  // Returns the device name from DXGI_OUTPUT_DESC in utf8 encoding.
+  const std::string& device_name() const { return device_name_; }
+
+  void Setup(Context* context);
+
+  void Unregister(const Context* const context);
+
+  // How many frames have been captured by this DxigOutputDuplicator.
+  int64_t num_frames_captured() const;
+
+  // Moves `desktop_rect_`. See DxgiDuplicatorController::TranslateRect().
+  void TranslateRect(const DesktopVector& position);
+
+ private:
+  // Calls DoDetectUpdatedRegion(). If it fails, this function sets the
+  // `updated_region` as entire UntranslatedDesktopRect().
+  void DetectUpdatedRegion(const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+                           DesktopRegion* updated_region);
+
+  // Returns untranslated updated region, which are directly returned by Windows
+  // APIs. Returns false in case of a failure.
+  bool DoDetectUpdatedRegion(const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+                             DesktopRegion* updated_region);
+
+  bool ReleaseFrame();
+
+  // Initializes duplication_ instance. Expects duplication_ is in empty status.
+  // Returns false if system does not support IDXGIOutputDuplication.
+  bool DuplicateOutput();
+
+  // Returns a DesktopRect with the same size of desktop_size(), but translated
+  // by offset.
+  DesktopRect GetTranslatedDesktopRect(DesktopVector offset) const;
+
+  // Returns a DesktopRect with the same size of desktop_size(), but starts from
+  // (0, 0).
+  DesktopRect GetUntranslatedDesktopRect() const;
+
+  // Spreads changes from `context` to other registered Context(s) in
+  // contexts_.
+  void SpreadContextChange(const Context* const context);
+
+  // Returns the size of desktop rectangle current instance representing.
+  DesktopSize desktop_size() const;
+
+  const D3dDevice device_;
+  const Microsoft::WRL::ComPtr<IDXGIOutput1> output_;
+  const std::string device_name_;
+  DesktopRect desktop_rect_;
+  Microsoft::WRL::ComPtr<IDXGIOutputDuplication> duplication_;
+  DXGI_OUTDUPL_DESC desc_;
+  std::vector<uint8_t> metadata_;
+  std::unique_ptr<DxgiTexture> texture_;
+  Rotation rotation_;
+  DesktopSize unrotated_size_;
+
+  // After each AcquireNextFrame() function call, updated_region_(s) of all
+  // active Context(s) need to be updated. Since they have missed the
+  // change this time. And during next Duplicate() function call, their
+  // updated_region_ will be merged and copied.
+  std::vector<Context*> contexts_;
+
+  // The last full frame of this output and its offset. If on AcquireNextFrame()
+  // failed because of timeout, i.e. no update, we can copy content from
+  // `last_frame_`.
+  std::unique_ptr<SharedDesktopFrame> last_frame_;
+  DesktopVector last_frame_offset_;
+
+  int64_t num_frames_captured_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DXGI_OUTPUT_DUPLICATOR_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture.cc b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture.cc
new file mode 100644
index 0000000000..b8f5b81f90
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/dxgi_texture.h"
+
+#include <comdef.h>
+#include <d3d11.h>
+#include <wrl/client.h>
+
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/win/desktop_capture_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+using Microsoft::WRL::ComPtr;
+
+namespace webrtc {
+
+namespace {
+
+class DxgiDesktopFrame : public DesktopFrame {
+ public:
+  explicit DxgiDesktopFrame(const DxgiTexture& texture)
+      : DesktopFrame(texture.desktop_size(),
+                     texture.pitch(),
+                     texture.bits(),
+                     nullptr) {}
+
+  ~DxgiDesktopFrame() override = default;
+};
+
+}  // namespace
+
+DxgiTexture::DxgiTexture() = default;
+DxgiTexture::~DxgiTexture() = default;
+
+bool DxgiTexture::CopyFrom(const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+                           IDXGIResource* resource) {
+  RTC_DCHECK_GT(frame_info.AccumulatedFrames, 0);
+  RTC_DCHECK(resource);
+  ComPtr<ID3D11Texture2D> texture;
+  _com_error error = resource->QueryInterface(
+      __uuidof(ID3D11Texture2D),
+      reinterpret_cast<void**>(texture.GetAddressOf()));
+  if (error.Error() != S_OK || !texture) {
+    RTC_LOG(LS_ERROR) << "Failed to convert IDXGIResource to ID3D11Texture2D: "
+                      << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+
+  D3D11_TEXTURE2D_DESC desc = {0};
+  texture->GetDesc(&desc);
+  desktop_size_.set(desc.Width, desc.Height);
+
+  return CopyFromTexture(frame_info, texture.Get());
+}
+
+const DesktopFrame& DxgiTexture::AsDesktopFrame() {
+  if (!frame_) {
+    frame_.reset(new DxgiDesktopFrame(*this));
+  }
+  return *frame_;
+}
+
+bool DxgiTexture::Release() {
+  frame_.reset();
+  return DoRelease();
+}
+
+DXGI_MAPPED_RECT* DxgiTexture::rect() {
+  return &rect_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture.h b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture.h
new file mode 100644
index 0000000000..a663b95a04
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_H_
+
+#include <d3d11.h>
+#include <dxgi1_2.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+
+namespace webrtc {
+
+class DesktopRegion;
+
+// A texture copied or mapped from a DXGI_OUTDUPL_FRAME_INFO and IDXGIResource.
+class DxgiTexture {
+ public:
+  // Creates a DxgiTexture instance, which represents the `desktop_size` area of
+  // entire screen -- usually a monitor on the system.
+  DxgiTexture();
+
+  virtual ~DxgiTexture();
+
+  // Copies selected regions of a frame represented by frame_info and resource.
+  // Returns false if anything wrong.
+  bool CopyFrom(const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+                IDXGIResource* resource);
+
+  const DesktopSize& desktop_size() const { return desktop_size_; }
+
+  uint8_t* bits() const { return static_cast<uint8_t*>(rect_.pBits); }
+
+  int pitch() const { return static_cast<int>(rect_.Pitch); }
+
+  // Releases the resource currently holds by this instance. Returns false if
+  // anything wrong, and this instance should be deprecated in this state. bits,
+  // pitch and AsDesktopFrame are only valid after a success CopyFrom() call,
+  // but before Release() call.
+  bool Release();
+
+  // Returns a DesktopFrame snapshot of a DxgiTexture instance. This
+  // DesktopFrame is used to copy a DxgiTexture content to another DesktopFrame
+  // only. And it should not outlive its DxgiTexture instance.
+  const DesktopFrame& AsDesktopFrame();
+
+ protected:
+  DXGI_MAPPED_RECT* rect();
+
+  virtual bool CopyFromTexture(const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+                               ID3D11Texture2D* texture) = 0;
+
+  virtual bool DoRelease() = 0;
+
+ private:
+  DXGI_MAPPED_RECT rect_ = {0};
+  DesktopSize desktop_size_;
+  std::unique_ptr<DesktopFrame> frame_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_mapping.cc b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_mapping.cc
new file mode 100644
index 0000000000..7ecf1adc61
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_mapping.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/dxgi_texture_mapping.h"
+
+#include <comdef.h>
+#include <dxgi.h>
+#include <dxgi1_2.h>
+
+#include "modules/desktop_capture/win/desktop_capture_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+DxgiTextureMapping::DxgiTextureMapping(IDXGIOutputDuplication* duplication)
+    : duplication_(duplication) {
+  RTC_DCHECK(duplication_);
+}
+
+DxgiTextureMapping::~DxgiTextureMapping() = default;
+
+bool DxgiTextureMapping::CopyFromTexture(
+    const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+    ID3D11Texture2D* texture) {
+  RTC_DCHECK_GT(frame_info.AccumulatedFrames, 0);
+  RTC_DCHECK(texture);
+  *rect() = {0};
+  _com_error error = duplication_->MapDesktopSurface(rect());
+  if (error.Error() != S_OK) {
+    *rect() = {0};
+    RTC_LOG(LS_ERROR)
+        << "Failed to map the IDXGIOutputDuplication to a bitmap: "
+        << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+
+  return true;
+}
+
+bool DxgiTextureMapping::DoRelease() {
+  _com_error error = duplication_->UnMapDesktopSurface();
+  if (error.Error() != S_OK) {
+    RTC_LOG(LS_ERROR) << "Failed to unmap the IDXGIOutputDuplication: "
+                      << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_mapping.h b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_mapping.h
new file mode 100644
index 0000000000..71f00b99ab
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_mapping.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_MAPPING_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_MAPPING_H_
+
+#include <d3d11.h>
+#include <dxgi1_2.h>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/win/dxgi_texture.h"
+
+namespace webrtc {
+
+// A DxgiTexture which directly maps bitmap from IDXGIResource. This class is
+// used when DXGI_OUTDUPL_DESC.DesktopImageInSystemMemory is true. (This usually
+// means the video card shares main memory with CPU, instead of having its own
+// individual memory.)
+class DxgiTextureMapping : public DxgiTexture {
+ public:
+  // Creates a DxgiTextureMapping instance. Caller must maintain the lifetime
+  // of input `duplication` to make sure it outlives this instance.
+  explicit DxgiTextureMapping(IDXGIOutputDuplication* duplication);
+
+  ~DxgiTextureMapping() override;
+
+ protected:
+  bool CopyFromTexture(const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+                       ID3D11Texture2D* texture) override;
+
+  bool DoRelease() override;
+
+ private:
+  IDXGIOutputDuplication* const duplication_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_MAPPING_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_staging.cc b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_staging.cc
new file mode 100644
index 0000000000..17e8518a7d
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_staging.cc
@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/dxgi_texture_staging.h"
+
+#include <comdef.h>
+#include <dxgi.h>
+#include <dxgi1_2.h>
+#include <unknwn.h>
+
+#include "modules/desktop_capture/win/desktop_capture_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+using Microsoft::WRL::ComPtr;
+
+namespace webrtc {
+
+DxgiTextureStaging::DxgiTextureStaging(const D3dDevice& device)
+    : device_(device) {}
+
+DxgiTextureStaging::~DxgiTextureStaging() = default;
+
+bool DxgiTextureStaging::InitializeStage(ID3D11Texture2D* texture) {
+  RTC_DCHECK(texture);
+  D3D11_TEXTURE2D_DESC desc = {0};
+  texture->GetDesc(&desc);
+
+  desc.ArraySize = 1;
+  desc.BindFlags = 0;
+  desc.CPUAccessFlags = D3D11_CPU_ACCESS_READ;
+  desc.MipLevels = 1;
+  desc.MiscFlags = 0;
+  desc.SampleDesc.Count = 1;
+  desc.SampleDesc.Quality = 0;
+  desc.Usage = D3D11_USAGE_STAGING;
+  if (stage_) {
+    AssertStageAndSurfaceAreSameObject();
+    D3D11_TEXTURE2D_DESC current_desc;
+    stage_->GetDesc(&current_desc);
+    const bool recreate_needed =
+        (memcmp(&desc, &current_desc, sizeof(D3D11_TEXTURE2D_DESC)) != 0);
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.DesktopCapture.StagingTextureRecreate",
+                          recreate_needed);
+    if (!recreate_needed) {
+      return true;
+    }
+
+    // The descriptions are not consistent, we need to create a new
+    // ID3D11Texture2D instance.
+    stage_.Reset();
+    surface_.Reset();
+  } else {
+    RTC_DCHECK(!surface_);
+  }
+
+  _com_error error = device_.d3d_device()->CreateTexture2D(
+      &desc, nullptr, stage_.GetAddressOf());
+  if (error.Error() != S_OK || !stage_) {
+    RTC_LOG(LS_ERROR) << "Failed to create a new ID3D11Texture2D as stage: "
+                      << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+
+  error = stage_.As(&surface_);
+  if (error.Error() != S_OK || !surface_) {
+    RTC_LOG(LS_ERROR) << "Failed to convert ID3D11Texture2D to IDXGISurface: "
+                      << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+
+  return true;
+}
+
+void DxgiTextureStaging::AssertStageAndSurfaceAreSameObject() {
+  ComPtr<IUnknown> left;
+  ComPtr<IUnknown> right;
+  bool left_result = SUCCEEDED(stage_.As(&left));
+  bool right_result = SUCCEEDED(surface_.As(&right));
+  RTC_DCHECK(left_result);
+  RTC_DCHECK(right_result);
+  RTC_DCHECK(left.Get() == right.Get());
+}
+
+bool DxgiTextureStaging::CopyFromTexture(
+    const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+    ID3D11Texture2D* texture) {
+  RTC_DCHECK_GT(frame_info.AccumulatedFrames, 0);
+  RTC_DCHECK(texture);
+
+  // AcquireNextFrame returns a CPU inaccessible IDXGIResource, so we need to
+  // copy it to a CPU accessible staging ID3D11Texture2D.
+  if (!InitializeStage(texture)) {
+    return false;
+  }
+
+  device_.context()->CopyResource(static_cast<ID3D11Resource*>(stage_.Get()),
+                                  static_cast<ID3D11Resource*>(texture));
+
+  *rect() = {0};
+  _com_error error = surface_->Map(rect(), DXGI_MAP_READ);
+  if (error.Error() != S_OK) {
+    *rect() = {0};
+    RTC_LOG(LS_ERROR) << "Failed to map the IDXGISurface to a bitmap: "
+                      << desktop_capture::utils::ComErrorToString(error);
+    return false;
+  }
+
+  return true;
+}
+
+bool DxgiTextureStaging::DoRelease() {
+  _com_error error = surface_->Unmap();
+  if (error.Error() != S_OK) {
+    stage_.Reset();
+    surface_.Reset();
+  }
+  // If using staging mode, we only need to recreate ID3D11Texture2D instance.
+  // This will happen during next CopyFrom call. So this function always returns
+  // true.
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_staging.h b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_staging.h
new file mode 100644
index 0000000000..e8c2af6662
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/dxgi_texture_staging.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_STAGING_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_STAGING_H_
+
+#include <d3d11.h>
+#include <dxgi1_2.h>
+#include <wrl/client.h>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/win/d3d_device.h"
+#include "modules/desktop_capture/win/dxgi_texture.h"
+
+namespace webrtc {
+
+// A pair of an ID3D11Texture2D and an IDXGISurface. We need an ID3D11Texture2D
+// instance to copy GPU texture to RAM, but an IDXGISurface instance to map the
+// texture into a bitmap buffer. These two instances are pointing to a same
+// object.
+//
+// An ID3D11Texture2D is created by an ID3D11Device, so a DxgiTexture cannot be
+// shared between two DxgiAdapterDuplicators.
+class DxgiTextureStaging : public DxgiTexture {
+ public:
+  // Creates a DxgiTextureStaging instance. Caller must maintain the lifetime
+  // of input device to make sure it outlives this instance.
+  explicit DxgiTextureStaging(const D3dDevice& device);
+
+  ~DxgiTextureStaging() override;
+
+ protected:
+  // Copies selected regions of a frame represented by frame_info and texture.
+  // Returns false if anything wrong.
+  bool CopyFromTexture(const DXGI_OUTDUPL_FRAME_INFO& frame_info,
+                       ID3D11Texture2D* texture) override;
+
+  bool DoRelease() override;
+
+ private:
+  // Initializes stage_ from a CPU inaccessible IDXGIResource. Returns false if
+  // it failed to execute Windows APIs, or the size of the texture is not
+  // consistent with desktop_rect.
+  bool InitializeStage(ID3D11Texture2D* texture);
+
+  // Makes sure stage_ and surface_ are always pointing to a same object.
+  // We need an ID3D11Texture2D instance for
+  // ID3D11DeviceContext::CopySubresourceRegion, but an IDXGISurface for
+  // IDXGISurface::Map.
+  void AssertStageAndSurfaceAreSameObject();
+
+  const DesktopRect desktop_rect_;
+  const D3dDevice device_;
+  Microsoft::WRL::ComPtr<ID3D11Texture2D> stage_;
+  Microsoft::WRL::ComPtr<IDXGISurface> surface_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_STAGING_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/full_screen_win_application_handler.cc b/third_party/libwebrtc/modules/desktop_capture/win/full_screen_win_application_handler.cc
new file mode 100644
index 0000000000..4222dfc01e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/full_screen_win_application_handler.cc
@@ -0,0 +1,294 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/full_screen_win_application_handler.h"
+
+#include <algorithm>
+#include <cwctype>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/ascii.h"
+#include "absl/strings/match.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "modules/desktop_capture/win/window_capture_utils.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/logging.h"  // For RTC_LOG_GLE
+#include "rtc_base/string_utils.h"
+
+namespace webrtc {
+namespace {
+
+// Utility function to verify that `window` has class name equal to `class_name`
+bool CheckWindowClassName(HWND window, const wchar_t* class_name) {
+  const size_t classNameLength = wcslen(class_name);
+
+  // https://docs.microsoft.com/en-us/windows/win32/api/winuser/ns-winuser-wndclassa
+  // says lpszClassName field in WNDCLASS is limited by 256 symbols, so we don't
+  // need to have a buffer bigger than that.
+  constexpr size_t kMaxClassNameLength = 256;
+  WCHAR buffer[kMaxClassNameLength];
+
+  const int length = ::GetClassNameW(window, buffer, kMaxClassNameLength);
+  if (length <= 0)
+    return false;
+
+  if (static_cast<size_t>(length) != classNameLength)
+    return false;
+  return wcsncmp(buffer, class_name, classNameLength) == 0;
+}
+
+std::string WindowText(HWND window) {
+  size_t len = ::GetWindowTextLength(window);
+  if (len == 0)
+    return std::string();
+
+  std::vector<wchar_t> buffer(len + 1, 0);
+  size_t copied = ::GetWindowTextW(window, buffer.data(), buffer.size());
+  if (copied == 0)
+    return std::string();
+  return rtc::ToUtf8(buffer.data(), copied);
+}
+
+DWORD WindowProcessId(HWND window) {
+  DWORD dwProcessId = 0;
+  ::GetWindowThreadProcessId(window, &dwProcessId);
+  return dwProcessId;
+}
+
+std::wstring FileNameFromPath(const std::wstring& path) {
+  auto found = path.rfind(L"\\");
+  if (found == std::string::npos)
+    return path;
+  return path.substr(found + 1);
+}
+
+// Returns windows which belong to given process id
+// `sources` is a full list of available windows
+// `processId` is a process identifier (window owner)
+// `window_to_exclude` is a window to be exluded from result
+DesktopCapturer::SourceList GetProcessWindows(
+    const DesktopCapturer::SourceList& sources,
+    DWORD processId,
+    HWND window_to_exclude) {
+  DesktopCapturer::SourceList result;
+  std::copy_if(sources.begin(), sources.end(), std::back_inserter(result),
+               [&](DesktopCapturer::Source source) {
+                 const HWND source_hwnd = reinterpret_cast<HWND>(source.id);
+                 return window_to_exclude != source_hwnd &&
+                        WindowProcessId(source_hwnd) == processId;
+               });
+  return result;
+}
+
+class FullScreenPowerPointHandler : public FullScreenApplicationHandler {
+ public:
+  explicit FullScreenPowerPointHandler(DesktopCapturer::SourceId sourceId)
+      : FullScreenApplicationHandler(sourceId) {}
+
+  ~FullScreenPowerPointHandler() override {}
+
+  DesktopCapturer::SourceId FindFullScreenWindow(
+      const DesktopCapturer::SourceList& window_list,
+      int64_t timestamp) const override {
+    if (window_list.empty())
+      return 0;
+
+    HWND original_window = reinterpret_cast<HWND>(GetSourceId());
+    DWORD process_id = WindowProcessId(original_window);
+
+    DesktopCapturer::SourceList powerpoint_windows =
+        GetProcessWindows(window_list, process_id, original_window);
+
+    if (powerpoint_windows.empty())
+      return 0;
+
+    if (GetWindowType(original_window) != WindowType::kEditor)
+      return 0;
+
+    const auto original_document = GetDocumentFromEditorTitle(original_window);
+
+    for (const auto& source : powerpoint_windows) {
+      HWND window = reinterpret_cast<HWND>(source.id);
+
+      // Looking for slide show window for the same document
+      if (GetWindowType(window) != WindowType::kSlideShow ||
+          GetDocumentFromSlideShowTitle(window) != original_document) {
+        continue;
+      }
+
+      return source.id;
+    }
+
+    return 0;
+  }
+
+ private:
+  enum class WindowType { kEditor, kSlideShow, kOther };
+
+  WindowType GetWindowType(HWND window) const {
+    if (IsEditorWindow(window))
+      return WindowType::kEditor;
+    else if (IsSlideShowWindow(window))
+      return WindowType::kSlideShow;
+    else
+      return WindowType::kOther;
+  }
+
+  constexpr static char kDocumentTitleSeparator[] = " - ";
+
+  std::string GetDocumentFromEditorTitle(HWND window) const {
+    std::string title = WindowText(window);
+    auto position = title.find(kDocumentTitleSeparator);
+    return std::string(absl::StripAsciiWhitespace(
+        absl::string_view(title).substr(0, position)));
+  }
+
+  std::string GetDocumentFromSlideShowTitle(HWND window) const {
+    std::string title = WindowText(window);
+    auto left_pos = title.find(kDocumentTitleSeparator);
+    auto right_pos = title.rfind(kDocumentTitleSeparator);
+    constexpr size_t kSeparatorLength = arraysize(kDocumentTitleSeparator) - 1;
+    if (left_pos == std::string::npos || right_pos == std::string::npos)
+      return title;
+
+    if (right_pos > left_pos + kSeparatorLength) {
+      auto result_len = right_pos - left_pos - kSeparatorLength;
+      auto document = absl::string_view(title).substr(
+          left_pos + kSeparatorLength, result_len);
+      return std::string(absl::StripAsciiWhitespace(document));
+    } else {
+      auto document = absl::string_view(title).substr(
+          left_pos + kSeparatorLength, std::wstring::npos);
+      return std::string(absl::StripAsciiWhitespace(document));
+    }
+  }
+
+  bool IsEditorWindow(HWND window) const {
+    return CheckWindowClassName(window, L"PPTFrameClass");
+  }
+
+  bool IsSlideShowWindow(HWND window) const {
+    const LONG style = ::GetWindowLong(window, GWL_STYLE);
+    const bool min_box = WS_MINIMIZEBOX & style;
+    const bool max_box = WS_MAXIMIZEBOX & style;
+    return !min_box && !max_box;
+  }
+};
+
+class OpenOfficeApplicationHandler : public FullScreenApplicationHandler {
+ public:
+  explicit OpenOfficeApplicationHandler(DesktopCapturer::SourceId sourceId)
+      : FullScreenApplicationHandler(sourceId) {}
+
+  DesktopCapturer::SourceId FindFullScreenWindow(
+      const DesktopCapturer::SourceList& window_list,
+      int64_t timestamp) const override {
+    if (window_list.empty())
+      return 0;
+
+    DWORD process_id = WindowProcessId(reinterpret_cast<HWND>(GetSourceId()));
+
+    DesktopCapturer::SourceList app_windows =
+        GetProcessWindows(window_list, process_id, nullptr);
+
+    DesktopCapturer::SourceList document_windows;
+    std::copy_if(
+        app_windows.begin(), app_windows.end(),
+        std::back_inserter(document_windows),
+        [this](const DesktopCapturer::Source& x) { return IsEditorWindow(x); });
+
+    // Check if we have only one document window, otherwise it's not possible
+    // to securely match a document window and a slide show window which has
+    // empty title.
+    if (document_windows.size() != 1) {
+      return 0;
+    }
+
+    // Check if document window has been selected as a source
+    if (document_windows.front().id != GetSourceId()) {
+      return 0;
+    }
+
+    // Check if we have a slide show window.
+    auto slide_show_window =
+        std::find_if(app_windows.begin(), app_windows.end(),
+                     [this](const DesktopCapturer::Source& x) {
+                       return IsSlideShowWindow(x);
+                     });
+
+    if (slide_show_window == app_windows.end())
+      return 0;
+
+    return slide_show_window->id;
+  }
+
+ private:
+  bool IsEditorWindow(const DesktopCapturer::Source& source) const {
+    if (source.title.empty()) {
+      return false;
+    }
+
+    return CheckWindowClassName(reinterpret_cast<HWND>(source.id), L"SALFRAME");
+  }
+
+  bool IsSlideShowWindow(const DesktopCapturer::Source& source) const {
+    // Check title size to filter out a Presenter Control window which shares
+    // window class with Slide Show window but has non empty title.
+    if (!source.title.empty()) {
+      return false;
+    }
+
+    return CheckWindowClassName(reinterpret_cast<HWND>(source.id),
+                                L"SALTMPSUBFRAME");
+  }
+};
+
+std::wstring GetPathByWindowId(HWND window_id) {
+  DWORD process_id = WindowProcessId(window_id);
+  HANDLE process =
+      ::OpenProcess(PROCESS_QUERY_LIMITED_INFORMATION, FALSE, process_id);
+  if (process == NULL)
+    return L"";
+  DWORD path_len = MAX_PATH;
+  WCHAR path[MAX_PATH];
+  std::wstring result;
+  if (::QueryFullProcessImageNameW(process, 0, path, &path_len))
+    result = std::wstring(path, path_len);
+  else
+    RTC_LOG_GLE(LS_ERROR) << "QueryFullProcessImageName failed.";
+
+  ::CloseHandle(process);
+  return result;
+}
+
+}  // namespace
+
+std::unique_ptr<FullScreenApplicationHandler>
+CreateFullScreenWinApplicationHandler(DesktopCapturer::SourceId source_id) {
+  std::unique_ptr<FullScreenApplicationHandler> result;
+  HWND hwnd = reinterpret_cast<HWND>(source_id);
+  std::wstring exe_path = GetPathByWindowId(hwnd);
+  std::wstring file_name = FileNameFromPath(exe_path);
+  std::transform(file_name.begin(), file_name.end(), file_name.begin(),
+                 std::towupper);
+
+  if (file_name == L"POWERPNT.EXE") {
+    result = std::make_unique<FullScreenPowerPointHandler>(source_id);
+  } else if (file_name == L"SOFFICE.BIN" &&
+             absl::EndsWith(WindowText(hwnd), "OpenOffice Impress")) {
+    result = std::make_unique<OpenOfficeApplicationHandler>(source_id);
+  }
+
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/full_screen_win_application_handler.h b/third_party/libwebrtc/modules/desktop_capture/win/full_screen_win_application_handler.h
new file mode 100644
index 0000000000..c97cbe252b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/full_screen_win_application_handler.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_FULL_SCREEN_WIN_APPLICATION_HANDLER_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_FULL_SCREEN_WIN_APPLICATION_HANDLER_H_
+
+#include <memory>
+#include "modules/desktop_capture/full_screen_application_handler.h"
+
+namespace webrtc {
+
+std::unique_ptr<FullScreenApplicationHandler>
+CreateFullScreenWinApplicationHandler(DesktopCapturer::SourceId sourceId);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_FULL_SCREEN_WIN_APPLICATION_HANDLER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/scoped_gdi_object.h b/third_party/libwebrtc/modules/desktop_capture/win/scoped_gdi_object.h
new file mode 100644
index 0000000000..2b01941e20
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/scoped_gdi_object.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_SCOPED_GDI_HANDLE_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_SCOPED_GDI_HANDLE_H_
+
+#include <windows.h>
+
+namespace webrtc {
+namespace win {
+
+// Scoper for GDI objects.
+template <class T, class Traits>
+class ScopedGDIObject {
+ public:
+  ScopedGDIObject() : handle_(NULL) {}
+  explicit ScopedGDIObject(T object) : handle_(object) {}
+
+  ~ScopedGDIObject() { Traits::Close(handle_); }
+
+  ScopedGDIObject(const ScopedGDIObject&) = delete;
+  ScopedGDIObject& operator=(const ScopedGDIObject&) = delete;
+
+  T Get() { return handle_; }
+
+  void Set(T object) {
+    if (handle_ && object != handle_)
+      Traits::Close(handle_);
+    handle_ = object;
+  }
+
+  ScopedGDIObject& operator=(T object) {
+    Set(object);
+    return *this;
+  }
+
+  T release() {
+    T object = handle_;
+    handle_ = NULL;
+    return object;
+  }
+
+  operator T() { return handle_; }
+
+ private:
+  T handle_;
+};
+
+// The traits class that uses DeleteObject() to close a handle.
+template <typename T>
+class DeleteObjectTraits {
+ public:
+  DeleteObjectTraits() = delete;
+  DeleteObjectTraits(const DeleteObjectTraits&) = delete;
+  DeleteObjectTraits& operator=(const DeleteObjectTraits&) = delete;
+
+  // Closes the handle.
+  static void Close(T handle) {
+    if (handle)
+      DeleteObject(handle);
+  }
+};
+
+// The traits class that uses DestroyCursor() to close a handle.
+class DestroyCursorTraits {
+ public:
+  DestroyCursorTraits() = delete;
+  DestroyCursorTraits(const DestroyCursorTraits&) = delete;
+  DestroyCursorTraits& operator=(const DestroyCursorTraits&) = delete;
+
+  // Closes the handle.
+  static void Close(HCURSOR handle) {
+    if (handle)
+      DestroyCursor(handle);
+  }
+};
+
+typedef ScopedGDIObject<HBITMAP, DeleteObjectTraits<HBITMAP> > ScopedBitmap;
+typedef ScopedGDIObject<HCURSOR, DestroyCursorTraits> ScopedCursor;
+
+}  // namespace win
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_SCOPED_GDI_HANDLE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/scoped_thread_desktop.cc b/third_party/libwebrtc/modules/desktop_capture/win/scoped_thread_desktop.cc
new file mode 100644
index 0000000000..22e8e7bc8f
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/scoped_thread_desktop.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/scoped_thread_desktop.h"
+
+#include "modules/desktop_capture/win/desktop.h"
+
+namespace webrtc {
+
+ScopedThreadDesktop::ScopedThreadDesktop()
+    : initial_(Desktop::GetThreadDesktop()) {}
+
+ScopedThreadDesktop::~ScopedThreadDesktop() {
+  Revert();
+}
+
+bool ScopedThreadDesktop::IsSame(const Desktop& desktop) {
+  if (assigned_.get() != NULL) {
+    return assigned_->IsSame(desktop);
+  } else {
+    return initial_->IsSame(desktop);
+  }
+}
+
+void ScopedThreadDesktop::Revert() {
+  if (assigned_.get() != NULL) {
+    initial_->SetThreadDesktop();
+    assigned_.reset();
+  }
+}
+
+bool ScopedThreadDesktop::SetThreadDesktop(Desktop* desktop) {
+  Revert();
+
+  std::unique_ptr<Desktop> scoped_desktop(desktop);
+
+  if (initial_->IsSame(*desktop))
+    return true;
+
+  if (!desktop->SetThreadDesktop())
+    return false;
+
+  assigned_.reset(scoped_desktop.release());
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/scoped_thread_desktop.h b/third_party/libwebrtc/modules/desktop_capture/win/scoped_thread_desktop.h
new file mode 100644
index 0000000000..98f151a46c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/scoped_thread_desktop.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_SCOPED_THREAD_DESKTOP_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_SCOPED_THREAD_DESKTOP_H_
+
+#include <windows.h>
+
+#include <memory>
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class Desktop;
+
+class RTC_EXPORT ScopedThreadDesktop {
+ public:
+  ScopedThreadDesktop();
+  ~ScopedThreadDesktop();
+
+  ScopedThreadDesktop(const ScopedThreadDesktop&) = delete;
+  ScopedThreadDesktop& operator=(const ScopedThreadDesktop&) = delete;
+
+  // Returns true if `desktop` has the same desktop name as the currently
+  // assigned desktop (if assigned) or as the initial desktop (if not assigned).
+  // Returns false in any other case including failing Win32 APIs and
+  // uninitialized desktop handles.
+  bool IsSame(const Desktop& desktop);
+
+  // Reverts the calling thread to use the initial desktop.
+  void Revert();
+
+  // Assigns `desktop` to be the calling thread. Returns true if the thread has
+  // been switched to `desktop` successfully. Takes ownership of `desktop`.
+  bool SetThreadDesktop(Desktop* desktop);
+
+ private:
+  // The desktop handle assigned to the calling thread by Set
+  std::unique_ptr<Desktop> assigned_;
+
+  // The desktop handle assigned to the calling thread at creation.
+  std::unique_ptr<Desktop> initial_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_SCOPED_THREAD_DESKTOP_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils.cc b/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils.cc
new file mode 100644
index 0000000000..1dc2918d08
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils.cc
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+
+#include <windows.h>
+
+#include <string>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/win32.h"
+
+namespace webrtc {
+
+bool HasActiveDisplay() {
+  DesktopCapturer::SourceList screens;
+
+  return GetScreenList(&screens) && !screens.empty();
+}
+
+bool GetScreenList(DesktopCapturer::SourceList* screens,
+                   std::vector<std::string>* device_names /* = nullptr */) {
+  RTC_DCHECK(screens->empty());
+  RTC_DCHECK(!device_names || device_names->empty());
+
+  BOOL enum_result = TRUE;
+  for (int device_index = 0;; ++device_index) {
+    DISPLAY_DEVICEW device;
+    device.cb = sizeof(device);
+    enum_result = EnumDisplayDevicesW(NULL, device_index, &device, 0);
+
+    // `enum_result` is 0 if we have enumerated all devices.
+    if (!enum_result) {
+      break;
+    }
+
+    // We only care about active displays.
+    if (!(device.StateFlags & DISPLAY_DEVICE_ACTIVE)) {
+      continue;
+    }
+
+    screens->push_back({device_index, 0, std::string()});
+    if (device_names) {
+      device_names->push_back(rtc::ToUtf8(device.DeviceName));
+    }
+  }
+  return true;
+}
+
+bool GetHmonitorFromDeviceIndex(const DesktopCapturer::SourceId device_index,
+                                HMONITOR* hmonitor) {
+  // A device index of `kFullDesktopScreenId` or -1 represents all screens, an
+  // HMONITOR of 0 indicates the same.
+  if (device_index == kFullDesktopScreenId) {
+    *hmonitor = 0;
+    return true;
+  }
+
+  std::wstring device_key;
+  if (!IsScreenValid(device_index, &device_key)) {
+    return false;
+  }
+
+  DesktopRect screen_rect = GetScreenRect(device_index, device_key);
+  if (screen_rect.is_empty()) {
+    return false;
+  }
+
+  RECT rect = {screen_rect.left(), screen_rect.top(), screen_rect.right(),
+               screen_rect.bottom()};
+
+  HMONITOR monitor = MonitorFromRect(&rect, MONITOR_DEFAULTTONULL);
+  if (monitor == NULL) {
+    RTC_LOG(LS_WARNING) << "No HMONITOR found for supplied device index.";
+    return false;
+  }
+
+  *hmonitor = monitor;
+  return true;
+}
+
+bool IsMonitorValid(const HMONITOR monitor) {
+  // An HMONITOR of 0 refers to a virtual monitor that spans all physical
+  // monitors.
+  if (monitor == 0) {
+    // There is a bug in a Windows OS API that causes a crash when capturing if
+    // there are no active displays. We must ensure there is an active display
+    // before returning true.
+    if (!HasActiveDisplay())
+      return false;
+
+    return true;
+  }
+
+  MONITORINFO monitor_info;
+  monitor_info.cbSize = sizeof(MONITORINFO);
+  return GetMonitorInfoA(monitor, &monitor_info);
+}
+
+DesktopRect GetMonitorRect(const HMONITOR monitor) {
+  MONITORINFO monitor_info;
+  monitor_info.cbSize = sizeof(MONITORINFO);
+  if (!GetMonitorInfoA(monitor, &monitor_info)) {
+    return DesktopRect();
+  }
+
+  return DesktopRect::MakeLTRB(
+      monitor_info.rcMonitor.left, monitor_info.rcMonitor.top,
+      monitor_info.rcMonitor.right, monitor_info.rcMonitor.bottom);
+}
+
+bool IsScreenValid(const DesktopCapturer::SourceId screen,
+                   std::wstring* device_key) {
+  if (screen == kFullDesktopScreenId) {
+    *device_key = L"";
+    return true;
+  }
+
+  DISPLAY_DEVICEW device;
+  device.cb = sizeof(device);
+  BOOL enum_result = EnumDisplayDevicesW(NULL, screen, &device, 0);
+  if (enum_result) {
+    *device_key = device.DeviceKey;
+  }
+
+  return !!enum_result;
+}
+
+DesktopRect GetFullscreenRect() {
+  return DesktopRect::MakeXYWH(GetSystemMetrics(SM_XVIRTUALSCREEN),
+                               GetSystemMetrics(SM_YVIRTUALSCREEN),
+                               GetSystemMetrics(SM_CXVIRTUALSCREEN),
+                               GetSystemMetrics(SM_CYVIRTUALSCREEN));
+}
+
+DesktopRect GetScreenRect(const DesktopCapturer::SourceId screen,
+                          const std::wstring& device_key) {
+  RTC_DCHECK(IsGUIThread(false));
+  if (screen == kFullDesktopScreenId) {
+    return GetFullscreenRect();
+  }
+
+  DISPLAY_DEVICEW device;
+  device.cb = sizeof(device);
+  BOOL result = EnumDisplayDevicesW(NULL, screen, &device, 0);
+  if (!result) {
+    return DesktopRect();
+  }
+
+  // Verifies the device index still maps to the same display device, to make
+  // sure we are capturing the same device when devices are added or removed.
+  // DeviceKey is documented as reserved, but it actually contains the registry
+  // key for the device and is unique for each monitor, while DeviceID is not.
+  if (device_key != device.DeviceKey) {
+    return DesktopRect();
+  }
+
+  DEVMODEW device_mode;
+  device_mode.dmSize = sizeof(device_mode);
+  device_mode.dmDriverExtra = 0;
+  result = EnumDisplaySettingsExW(device.DeviceName, ENUM_CURRENT_SETTINGS,
+                                  &device_mode, 0);
+  if (!result) {
+    return DesktopRect();
+  }
+
+  return DesktopRect::MakeXYWH(
+      device_mode.dmPosition.x, device_mode.dmPosition.y,
+      device_mode.dmPelsWidth, device_mode.dmPelsHeight);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils.h b/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils.h
new file mode 100644
index 0000000000..97bfe816d8
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURE_UTILS_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURE_UTILS_H_
+
+#if defined(WEBRTC_WIN)
+// Forward declare HMONITOR in a windows.h compatible way so that we can avoid
+// including windows.h.
+#define WEBRTC_DECLARE_HANDLE(name) \
+struct name##__;                  \
+typedef struct name##__* name
+WEBRTC_DECLARE_HANDLE(HMONITOR);
+#undef WEBRTC_DECLARE_HANDLE
+#endif
+
+#include <string>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Returns true if the system has at least one active display.
+bool HasActiveDisplay();
+
+// Output the list of active screens into `screens`. Returns true if succeeded,
+// or false if it fails to enumerate the display devices. If the `device_names`
+// is provided, it will be filled with the DISPLAY_DEVICE.DeviceName in UTF-8
+// encoding. If this function returns true, consumers can always assume that
+// `screens`[i] and `device_names`[i] indicate the same monitor on the system.
+bool GetScreenList(DesktopCapturer::SourceList* screens,
+                   std::vector<std::string>* device_names = nullptr);
+
+// Converts a device index (which are returned by `GetScreenList`) into an
+// HMONITOR.
+bool GetHmonitorFromDeviceIndex(DesktopCapturer::SourceId device_index,
+                                HMONITOR* hmonitor);
+
+// Returns true if `monitor` represents a valid display
+// monitor. Consumers should recheck the validity of HMONITORs before use if a
+// WM_DISPLAYCHANGE message has been received.
+bool IsMonitorValid(HMONITOR monitor);
+
+// Returns the rect of the monitor identified by `monitor`, relative to the
+// primary display's top-left. On failure, returns an empty rect.
+DesktopRect GetMonitorRect(HMONITOR monitor);
+
+// Returns true if `screen` is a valid screen. The screen device key is
+// returned through `device_key` if the screen is valid. The device key can be
+// used in GetScreenRect to verify the screen matches the previously obtained
+// id.
+bool IsScreenValid(DesktopCapturer::SourceId screen, std::wstring* device_key);
+
+// Get the rect of the entire system in system coordinate system. I.e. the
+// primary monitor always starts from (0, 0).
+DesktopRect GetFullscreenRect();
+
+// Get the rect of the screen identified by `screen`, relative to the primary
+// display's top-left. If the screen device key does not match `device_key`, or
+// the screen does not exist, or any error happens, an empty rect is returned.
+RTC_EXPORT DesktopRect GetScreenRect(DesktopCapturer::SourceId screen,
+                                     const std::wstring& device_key);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURE_UTILS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils_unittest.cc
new file mode 100644
index 0000000000..2e58c6b164
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capture_utils_unittest.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+
+#include <string>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "rtc_base/logging.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(ScreenCaptureUtilsTest, GetScreenList) {
+  DesktopCapturer::SourceList screens;
+  std::vector<std::string> device_names;
+
+  ASSERT_TRUE(GetScreenList(&screens));
+  screens.clear();
+  ASSERT_TRUE(GetScreenList(&screens, &device_names));
+
+  ASSERT_EQ(screens.size(), device_names.size());
+}
+
+TEST(ScreenCaptureUtilsTest, DeviceIndexToHmonitor) {
+  DesktopCapturer::SourceList screens;
+  ASSERT_TRUE(GetScreenList(&screens));
+  if (screens.empty()) {
+    RTC_LOG(LS_INFO)
+        << "Skip ScreenCaptureUtilsTest on systems with no monitors.";
+    GTEST_SKIP();
+  }
+
+  HMONITOR hmonitor;
+  ASSERT_TRUE(GetHmonitorFromDeviceIndex(screens[0].id, &hmonitor));
+  ASSERT_TRUE(IsMonitorValid(hmonitor));
+}
+
+TEST(ScreenCaptureUtilsTest, FullScreenDeviceIndexToHmonitor) {
+  if (!HasActiveDisplay()) {
+    RTC_LOG(LS_INFO)
+        << "Skip ScreenCaptureUtilsTest on systems with no monitors.";
+    GTEST_SKIP();
+  }
+
+  HMONITOR hmonitor;
+  ASSERT_TRUE(GetHmonitorFromDeviceIndex(kFullDesktopScreenId, &hmonitor));
+  ASSERT_EQ(hmonitor, static_cast<HMONITOR>(0));
+  ASSERT_TRUE(IsMonitorValid(hmonitor));
+}
+
+TEST(ScreenCaptureUtilsTest, NoMonitors) {
+  if (HasActiveDisplay()) {
+    RTC_LOG(LS_INFO) << "Skip ScreenCaptureUtilsTest designed specifically for "
+                        "systems with no monitors";
+    GTEST_SKIP();
+  }
+
+  HMONITOR hmonitor;
+  ASSERT_TRUE(GetHmonitorFromDeviceIndex(kFullDesktopScreenId, &hmonitor));
+  ASSERT_EQ(hmonitor, static_cast<HMONITOR>(0));
+
+  // The monitor should be invalid since the system has no attached displays.
+  ASSERT_FALSE(IsMonitorValid(hmonitor));
+}
+
+TEST(ScreenCaptureUtilsTest, InvalidDeviceIndexToHmonitor) {
+  HMONITOR hmonitor;
+  ASSERT_FALSE(GetHmonitorFromDeviceIndex(kInvalidScreenId, &hmonitor));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx.cc b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx.cc
new file mode 100644
index 0000000000..efa763993a
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx.cc
@@ -0,0 +1,230 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/screen_capturer_win_directx.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capture_metrics_helper.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+using Microsoft::WRL::ComPtr;
+
+// static
+bool ScreenCapturerWinDirectx::IsSupported() {
+  // Forwards IsSupported() function call to DxgiDuplicatorController.
+  return DxgiDuplicatorController::Instance()->IsSupported();
+}
+
+// static
+bool ScreenCapturerWinDirectx::RetrieveD3dInfo(D3dInfo* info) {
+  // Forwards SupportedFeatureLevels() function call to
+  // DxgiDuplicatorController.
+  return DxgiDuplicatorController::Instance()->RetrieveD3dInfo(info);
+}
+
+// static
+bool ScreenCapturerWinDirectx::IsCurrentSessionSupported() {
+  return DxgiDuplicatorController::IsCurrentSessionSupported();
+}
+
+// static
+bool ScreenCapturerWinDirectx::GetScreenListFromDeviceNames(
+    const std::vector<std::string>& device_names,
+    DesktopCapturer::SourceList* screens) {
+  RTC_DCHECK(screens->empty());
+
+  DesktopCapturer::SourceList gdi_screens;
+  std::vector<std::string> gdi_names;
+  if (!GetScreenList(&gdi_screens, &gdi_names)) {
+    return false;
+  }
+
+  RTC_DCHECK_EQ(gdi_screens.size(), gdi_names.size());
+
+  ScreenId max_screen_id = -1;
+  for (const DesktopCapturer::Source& screen : gdi_screens) {
+    max_screen_id = std::max(max_screen_id, screen.id);
+  }
+
+  for (const auto& device_name : device_names) {
+    const auto it = std::find(gdi_names.begin(), gdi_names.end(), device_name);
+    if (it == gdi_names.end()) {
+      // devices_names[i] has not been found in gdi_names, so use max_screen_id.
+      max_screen_id++;
+      screens->push_back({max_screen_id});
+    } else {
+      screens->push_back({gdi_screens[it - gdi_names.begin()]});
+    }
+  }
+
+  return true;
+}
+
+// static
+int ScreenCapturerWinDirectx::GetIndexFromScreenId(
+    ScreenId id,
+    const std::vector<std::string>& device_names) {
+  DesktopCapturer::SourceList screens;
+  if (!GetScreenListFromDeviceNames(device_names, &screens)) {
+    return -1;
+  }
+
+  RTC_DCHECK_EQ(device_names.size(), screens.size());
+
+  for (size_t i = 0; i < screens.size(); i++) {
+    if (screens[i].id == id) {
+      return static_cast<int>(i);
+    }
+  }
+
+  return -1;
+}
+
+ScreenCapturerWinDirectx::ScreenCapturerWinDirectx()
+    : controller_(DxgiDuplicatorController::Instance()) {}
+
+ScreenCapturerWinDirectx::~ScreenCapturerWinDirectx() = default;
+
+void ScreenCapturerWinDirectx::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+  RecordCapturerImpl(DesktopCapturerId::kScreenCapturerWinDirectx);
+
+  callback_ = callback;
+}
+
+void ScreenCapturerWinDirectx::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  shared_memory_factory_ = std::move(shared_memory_factory);
+}
+
+void ScreenCapturerWinDirectx::CaptureFrame() {
+  RTC_DCHECK(callback_);
+  TRACE_EVENT0("webrtc", "ScreenCapturerWinDirectx::CaptureFrame");
+
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+  // Note that the [] operator will create the ScreenCaptureFrameQueue if it
+  // doesn't exist, so this is safe.
+  ScreenCaptureFrameQueue<DxgiFrame>& frames =
+      frame_queue_map_[current_screen_id_];
+
+  frames.MoveToNextFrame();
+
+  if (!frames.current_frame()) {
+    frames.ReplaceCurrentFrame(
+        std::make_unique<DxgiFrame>(shared_memory_factory_.get()));
+  }
+
+  DxgiDuplicatorController::Result result;
+  if (current_screen_id_ == kFullDesktopScreenId) {
+    result = controller_->Duplicate(frames.current_frame());
+  } else {
+    result = controller_->DuplicateMonitor(frames.current_frame(),
+                                           current_screen_id_);
+  }
+
+  using DuplicateResult = DxgiDuplicatorController::Result;
+  if (result != DuplicateResult::SUCCEEDED) {
+    RTC_LOG(LS_ERROR) << "DxgiDuplicatorController failed to capture desktop, "
+                         "error code "
+                      << DxgiDuplicatorController::ResultName(result);
+  }
+  switch (result) {
+    case DuplicateResult::UNSUPPORTED_SESSION: {
+      RTC_LOG(LS_ERROR)
+          << "Current binary is running on a session not supported "
+             "by DirectX screen capturer.";
+      callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+      break;
+    }
+    case DuplicateResult::FRAME_PREPARE_FAILED: {
+      RTC_LOG(LS_ERROR) << "Failed to allocate a new DesktopFrame.";
+      // This usually means we do not have enough memory or SharedMemoryFactory
+      // cannot work correctly.
+      callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+      break;
+    }
+    case DuplicateResult::INVALID_MONITOR_ID: {
+      RTC_LOG(LS_ERROR) << "Invalid monitor id " << current_screen_id_;
+      callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+      break;
+    }
+    case DuplicateResult::INITIALIZATION_FAILED:
+    case DuplicateResult::DUPLICATION_FAILED: {
+      callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+      break;
+    }
+    case DuplicateResult::SUCCEEDED: {
+      std::unique_ptr<DesktopFrame> frame =
+          frames.current_frame()->frame()->Share();
+
+      int capture_time_ms = (rtc::TimeNanos() - capture_start_time_nanos) /
+                            rtc::kNumNanosecsPerMillisec;
+      RTC_HISTOGRAM_COUNTS_1000(
+          "WebRTC.DesktopCapture.Win.DirectXCapturerFrameTime",
+          capture_time_ms);
+      frame->set_capture_time_ms(capture_time_ms);
+      frame->set_capturer_id(DesktopCapturerId::kScreenCapturerWinDirectx);
+
+      // TODO(julien.isorce): http://crbug.com/945468. Set the icc profile on
+      // the frame, see WindowCapturerMac::CaptureFrame.
+
+      callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
+      break;
+    }
+  }
+}
+
+bool ScreenCapturerWinDirectx::GetSourceList(SourceList* sources) {
+  std::vector<std::string> device_names;
+  if (!controller_->GetDeviceNames(&device_names)) {
+    return false;
+  }
+
+  return GetScreenListFromDeviceNames(device_names, sources);
+}
+
+bool ScreenCapturerWinDirectx::SelectSource(SourceId id) {
+  if (id == kFullDesktopScreenId) {
+    current_screen_id_ = id;
+    return true;
+  }
+
+  std::vector<std::string> device_names;
+  if (!controller_->GetDeviceNames(&device_names)) {
+    return false;
+  }
+
+  int index;
+  index = GetIndexFromScreenId(id, device_names);
+  if (index == -1) {
+    return false;
+  }
+
+  current_screen_id_ = index;
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx.h b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx.h
new file mode 100644
index 0000000000..801a0632fc
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx.h
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_DIRECTX_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_DIRECTX_H_
+
+#include <d3dcommon.h>
+
+#include <memory>
+#include <unordered_map>
+#include <vector>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/screen_capture_frame_queue.h"
+#include "modules/desktop_capture/win/dxgi_duplicator_controller.h"
+#include "modules/desktop_capture/win/dxgi_frame.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// ScreenCapturerWinDirectx captures 32bit RGBA using DirectX.
+class RTC_EXPORT ScreenCapturerWinDirectx : public DesktopCapturer {
+ public:
+  using D3dInfo = DxgiDuplicatorController::D3dInfo;
+
+  // Whether the system supports DirectX based capturing.
+  static bool IsSupported();
+
+  // Returns a most recent D3dInfo composed by
+  // DxgiDuplicatorController::Initialize() function. This function implicitly
+  // calls DxgiDuplicatorController::Initialize() if it has not been
+  // initialized. This function returns false and output parameter is kept
+  // unchanged if DxgiDuplicatorController::Initialize() failed.
+  // The D3dInfo may change based on hardware configuration even without
+  // restarting the hardware and software. Refer to https://goo.gl/OOCppq. So
+  // consumers should not cache the result returned by this function.
+  static bool RetrieveD3dInfo(D3dInfo* info);
+
+  // Whether current process is running in a Windows session which is supported
+  // by ScreenCapturerWinDirectx.
+  // Usually using ScreenCapturerWinDirectx in unsupported sessions will fail.
+  // But this behavior may vary on different Windows version. So consumers can
+  // always try IsSupported() function.
+  static bool IsCurrentSessionSupported();
+
+  // Maps `device_names` with the result from GetScreenList() and creates a new
+  // SourceList to include only the ones in `device_names`. If this function
+  // returns true, consumers can always assume `device_names`.size() equals to
+  // `screens`->size(), meanwhile `device_names`[i] and `screens`[i] indicate
+  // the same monitor on the system.
+  // Public for test only.
+  static bool GetScreenListFromDeviceNames(
+      const std::vector<std::string>& device_names,
+      DesktopCapturer::SourceList* screens);
+
+  // Maps `id` with the result from GetScreenListFromDeviceNames() and returns
+  // the index of the entity in `device_names`. This function returns -1 if `id`
+  // cannot be found.
+  // Public for test only.
+  static int GetIndexFromScreenId(ScreenId id,
+                                  const std::vector<std::string>& device_names);
+
+  explicit ScreenCapturerWinDirectx();
+
+  ~ScreenCapturerWinDirectx() override;
+
+  ScreenCapturerWinDirectx(const ScreenCapturerWinDirectx&) = delete;
+  ScreenCapturerWinDirectx& operator=(const ScreenCapturerWinDirectx&) = delete;
+
+  // DesktopCapturer implementation.
+  void Start(Callback* callback) override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+
+ private:
+  const rtc::scoped_refptr<DxgiDuplicatorController> controller_;
+
+  // The underlying DxgiDuplicators may retain a reference to the frames that
+  // we ask them to duplicate so that they can continue returning valid frames
+  // in the event that the target has not been updated. Thus, we need to ensure
+  // that we have a separate frame queue for each source id, so that these held
+  // frames don't get overwritten with the data from another Duplicator/monitor.
+  std::unordered_map<SourceId, ScreenCaptureFrameQueue<DxgiFrame>>
+      frame_queue_map_;
+  std::unique_ptr<SharedMemoryFactory> shared_memory_factory_;
+  Callback* callback_ = nullptr;
+  SourceId current_screen_id_ = kFullDesktopScreenId;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_DIRECTX_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx_unittest.cc
new file mode 100644
index 0000000000..c9f46f782c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_directx_unittest.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/screen_capturer_win_directx.h"
+
+#include <string>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// This test cannot ensure GetScreenListFromDeviceNames() won't reorder the
+// devices in its output, since the device name is missing.
+TEST(ScreenCaptureUtilsTest, GetScreenListFromDeviceNamesAndGetIndex) {
+  const std::vector<std::string> device_names = {
+      "\\\\.\\DISPLAY0",
+      "\\\\.\\DISPLAY1",
+      "\\\\.\\DISPLAY2",
+  };
+  DesktopCapturer::SourceList screens;
+  ASSERT_TRUE(ScreenCapturerWinDirectx::GetScreenListFromDeviceNames(
+      device_names, &screens));
+  ASSERT_EQ(device_names.size(), screens.size());
+
+  for (size_t i = 0; i < screens.size(); i++) {
+    ASSERT_EQ(ScreenCapturerWinDirectx::GetIndexFromScreenId(screens[i].id,
+                                                             device_names),
+              static_cast<int>(i));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
new file mode 100644
index 0000000000..c6d4a75931
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
@@ -0,0 +1,241 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/screen_capturer_win_gdi.h"
+
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capture_metrics_helper.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_frame_win.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/win/cursor.h"
+#include "modules/desktop_capture/win/desktop.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+// Constants from dwmapi.h.
+const UINT DWM_EC_DISABLECOMPOSITION = 0;
+const UINT DWM_EC_ENABLECOMPOSITION = 1;
+
+const wchar_t kDwmapiLibraryName[] = L"dwmapi.dll";
+
+}  // namespace
+
+ScreenCapturerWinGdi::ScreenCapturerWinGdi(
+    const DesktopCaptureOptions& options) {
+  if (options.disable_effects()) {
+    // Load dwmapi.dll dynamically since it is not available on XP.
+    if (!dwmapi_library_)
+      dwmapi_library_ = LoadLibraryW(kDwmapiLibraryName);
+
+    if (dwmapi_library_) {
+      composition_func_ = reinterpret_cast<DwmEnableCompositionFunc>(
+          GetProcAddress(dwmapi_library_, "DwmEnableComposition"));
+    }
+  }
+}
+
+ScreenCapturerWinGdi::~ScreenCapturerWinGdi() {
+  if (desktop_dc_)
+    ReleaseDC(NULL, desktop_dc_);
+  if (memory_dc_)
+    DeleteDC(memory_dc_);
+
+  // Restore Aero.
+  if (composition_func_)
+    (*composition_func_)(DWM_EC_ENABLECOMPOSITION);
+
+  if (dwmapi_library_)
+    FreeLibrary(dwmapi_library_);
+}
+
+void ScreenCapturerWinGdi::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  shared_memory_factory_ = std::move(shared_memory_factory);
+}
+
+void ScreenCapturerWinGdi::CaptureFrame() {
+  TRACE_EVENT0("webrtc", "ScreenCapturerWinGdi::CaptureFrame");
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+  queue_.MoveToNextFrame();
+  if (queue_.current_frame() && queue_.current_frame()->IsShared()) {
+    RTC_DLOG(LS_WARNING) << "Overwriting frame that is still shared.";
+  }
+
+  // Make sure the GDI capture resources are up-to-date.
+  PrepareCaptureResources();
+
+  if (!CaptureImage()) {
+    RTC_LOG(LS_WARNING) << "Failed to capture screen by GDI.";
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+
+  // Emit the current frame.
+  std::unique_ptr<DesktopFrame> frame = queue_.current_frame()->Share();
+  frame->set_dpi(DesktopVector(GetDeviceCaps(desktop_dc_, LOGPIXELSX),
+                               GetDeviceCaps(desktop_dc_, LOGPIXELSY)));
+  frame->mutable_updated_region()->SetRect(
+      DesktopRect::MakeSize(frame->size()));
+
+  int capture_time_ms = (rtc::TimeNanos() - capture_start_time_nanos) /
+                        rtc::kNumNanosecsPerMillisec;
+  RTC_HISTOGRAM_COUNTS_1000(
+      "WebRTC.DesktopCapture.Win.ScreenGdiCapturerFrameTime", capture_time_ms);
+  frame->set_capture_time_ms(capture_time_ms);
+  frame->set_capturer_id(DesktopCapturerId::kScreenCapturerWinGdi);
+  callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
+}
+
+bool ScreenCapturerWinGdi::GetSourceList(SourceList* sources) {
+  return webrtc::GetScreenList(sources);
+}
+
+bool ScreenCapturerWinGdi::SelectSource(SourceId id) {
+  bool valid = IsScreenValid(id, &current_device_key_);
+  if (valid)
+    current_screen_id_ = id;
+  return valid;
+}
+
+void ScreenCapturerWinGdi::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+  RecordCapturerImpl(DesktopCapturerId::kScreenCapturerWinGdi);
+
+  callback_ = callback;
+
+  // Vote to disable Aero composited desktop effects while capturing. Windows
+  // will restore Aero automatically if the process exits. This has no effect
+  // under Windows 8 or higher.  See crbug.com/124018.
+  if (composition_func_)
+    (*composition_func_)(DWM_EC_DISABLECOMPOSITION);
+}
+
+void ScreenCapturerWinGdi::PrepareCaptureResources() {
+  // Switch to the desktop receiving user input if different from the current
+  // one.
+  std::unique_ptr<Desktop> input_desktop(Desktop::GetInputDesktop());
+  if (input_desktop && !desktop_.IsSame(*input_desktop)) {
+    // Release GDI resources otherwise SetThreadDesktop will fail.
+    if (desktop_dc_) {
+      ReleaseDC(NULL, desktop_dc_);
+      desktop_dc_ = nullptr;
+    }
+
+    if (memory_dc_) {
+      DeleteDC(memory_dc_);
+      memory_dc_ = nullptr;
+    }
+
+    // If SetThreadDesktop() fails, the thread is still assigned a desktop.
+    // So we can continue capture screen bits, just from the wrong desktop.
+    desktop_.SetThreadDesktop(input_desktop.release());
+
+    // Re-assert our vote to disable Aero.
+    // See crbug.com/124018 and crbug.com/129906.
+    if (composition_func_) {
+      (*composition_func_)(DWM_EC_DISABLECOMPOSITION);
+    }
+  }
+
+  // If the display configurations have changed then recreate GDI resources.
+  if (display_configuration_monitor_.IsChanged()) {
+    if (desktop_dc_) {
+      ReleaseDC(NULL, desktop_dc_);
+      desktop_dc_ = nullptr;
+    }
+    if (memory_dc_) {
+      DeleteDC(memory_dc_);
+      memory_dc_ = nullptr;
+    }
+  }
+
+  if (!desktop_dc_) {
+    RTC_DCHECK(!memory_dc_);
+
+    // Create GDI device contexts to capture from the desktop into memory.
+    desktop_dc_ = GetDC(nullptr);
+    RTC_CHECK(desktop_dc_);
+    memory_dc_ = CreateCompatibleDC(desktop_dc_);
+    RTC_CHECK(memory_dc_);
+
+    // Make sure the frame buffers will be reallocated.
+    queue_.Reset();
+  }
+}
+
+bool ScreenCapturerWinGdi::CaptureImage() {
+  RTC_DCHECK(IsGUIThread(false));
+  DesktopRect screen_rect =
+      GetScreenRect(current_screen_id_, current_device_key_);
+  if (screen_rect.is_empty()) {
+    RTC_LOG(LS_WARNING) << "Failed to get screen rect.";
+    return false;
+  }
+
+  DesktopSize size = screen_rect.size();
+  // If the current buffer is from an older generation then allocate a new one.
+  // Note that we can't reallocate other buffers at this point, since the caller
+  // may still be reading from them.
+  if (!queue_.current_frame() ||
+      !queue_.current_frame()->size().equals(screen_rect.size())) {
+    RTC_DCHECK(desktop_dc_);
+    RTC_DCHECK(memory_dc_);
+
+    std::unique_ptr<DesktopFrame> buffer = DesktopFrameWin::Create(
+        size, shared_memory_factory_.get(), desktop_dc_);
+    if (!buffer) {
+      RTC_LOG(LS_WARNING) << "Failed to create frame buffer.";
+      return false;
+    }
+    queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(std::move(buffer)));
+  }
+  queue_.current_frame()->set_top_left(
+      screen_rect.top_left().subtract(GetFullscreenRect().top_left()));
+
+  // Select the target bitmap into the memory dc and copy the rect from desktop
+  // to memory.
+  DesktopFrameWin* current = static_cast<DesktopFrameWin*>(
+      queue_.current_frame()->GetUnderlyingFrame());
+  HGDIOBJ previous_object = SelectObject(memory_dc_, current->bitmap());
+  if (!previous_object || previous_object == HGDI_ERROR) {
+    RTC_LOG(LS_WARNING) << "Failed to select current bitmap into memery dc.";
+    return false;
+  }
+
+  bool result = (BitBlt(memory_dc_, 0, 0, screen_rect.width(),
+                        screen_rect.height(), desktop_dc_, screen_rect.left(),
+                        screen_rect.top(), SRCCOPY | CAPTUREBLT) != FALSE);
+  if (!result) {
+    RTC_LOG_GLE(LS_WARNING) << "BitBlt failed";
+  }
+
+  // Select back the previously selected object to that the device contect
+  // could be destroyed independently of the bitmap if needed.
+  SelectObject(memory_dc_, previous_object);
+
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
new file mode 100644
index 0000000000..7c3977ed42
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_GDI_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_GDI_H_
+
+#include <windows.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/screen_capture_frame_queue.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "modules/desktop_capture/win/display_configuration_monitor.h"
+#include "modules/desktop_capture/win/scoped_thread_desktop.h"
+
+namespace webrtc {
+
+// ScreenCapturerWinGdi captures 32bit RGB using GDI.
+//
+// ScreenCapturerWinGdi is double-buffered as required by ScreenCapturer.
+// This class does not detect DesktopFrame::updated_region(), the field is
+// always set to the entire frame rectangle. ScreenCapturerDifferWrapper should
+// be used if that functionality is necessary.
+class ScreenCapturerWinGdi : public DesktopCapturer {
+ public:
+  explicit ScreenCapturerWinGdi(const DesktopCaptureOptions& options);
+  ~ScreenCapturerWinGdi() override;
+
+  ScreenCapturerWinGdi(const ScreenCapturerWinGdi&) = delete;
+  ScreenCapturerWinGdi& operator=(const ScreenCapturerWinGdi&) = delete;
+
+  // Overridden from ScreenCapturer:
+  void Start(Callback* callback) override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+
+ private:
+  typedef HRESULT(WINAPI* DwmEnableCompositionFunc)(UINT);
+
+  // Make sure that the device contexts match the screen configuration.
+  void PrepareCaptureResources();
+
+  // Captures the current screen contents into the current buffer. Returns true
+  // if succeeded.
+  bool CaptureImage();
+
+  // Capture the current cursor shape.
+  void CaptureCursor();
+
+  Callback* callback_ = nullptr;
+  std::unique_ptr<SharedMemoryFactory> shared_memory_factory_;
+  SourceId current_screen_id_ = kFullDesktopScreenId;
+  std::wstring current_device_key_;
+
+  ScopedThreadDesktop desktop_;
+
+  // GDI resources used for screen capture.
+  HDC desktop_dc_ = NULL;
+  HDC memory_dc_ = NULL;
+
+  // Queue of the frames buffers.
+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
+
+  DisplayConfigurationMonitor display_configuration_monitor_;
+
+  HMODULE dwmapi_library_ = NULL;
+  DwmEnableCompositionFunc composition_func_ = nullptr;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_GDI_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
new file mode 100644
index 0000000000..ce747e0141
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
@@ -0,0 +1,398 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/screen_capturer_win_magnifier.h"
+
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capture_metrics_helper.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_frame_win.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/mouse_cursor.h"
+#include "modules/desktop_capture/win/cursor.h"
+#include "modules/desktop_capture/win/desktop.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+DWORD GetTlsIndex() {
+  static const DWORD tls_index = TlsAlloc();
+  RTC_DCHECK(tls_index != TLS_OUT_OF_INDEXES);
+  return tls_index;
+}
+
+}  // namespace
+
+// kMagnifierWindowClass has to be "Magnifier" according to the Magnification
+// API. The other strings can be anything.
+static wchar_t kMagnifierHostClass[] = L"ScreenCapturerWinMagnifierHost";
+static wchar_t kHostWindowName[] = L"MagnifierHost";
+static wchar_t kMagnifierWindowClass[] = L"Magnifier";
+static wchar_t kMagnifierWindowName[] = L"MagnifierWindow";
+
+ScreenCapturerWinMagnifier::ScreenCapturerWinMagnifier() = default;
+ScreenCapturerWinMagnifier::~ScreenCapturerWinMagnifier() {
+  // DestroyWindow must be called before MagUninitialize. magnifier_window_ is
+  // destroyed automatically when host_window_ is destroyed.
+  if (host_window_) {
+    DestroyWindow(host_window_);
+    host_window_ = NULL;
+  }
+
+  if (magnifier_initialized_) {
+    mag_uninitialize_func_();
+    magnifier_initialized_ = false;
+  }
+
+  if (mag_lib_handle_) {
+    FreeLibrary(mag_lib_handle_);
+    mag_lib_handle_ = NULL;
+  }
+
+  if (desktop_dc_) {
+    ReleaseDC(NULL, desktop_dc_);
+    desktop_dc_ = NULL;
+  }
+}
+
+void ScreenCapturerWinMagnifier::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+  RecordCapturerImpl(DesktopCapturerId::kScreenCapturerWinMagnifier);
+
+  callback_ = callback;
+
+  if (!InitializeMagnifier()) {
+    RTC_LOG_F(LS_WARNING) << "Magnifier initialization failed.";
+  }
+}
+
+void ScreenCapturerWinMagnifier::SetSharedMemoryFactory(
+    std::unique_ptr<SharedMemoryFactory> shared_memory_factory) {
+  shared_memory_factory_ = std::move(shared_memory_factory);
+}
+
+void ScreenCapturerWinMagnifier::CaptureFrame() {
+  RTC_DCHECK(callback_);
+  if (!magnifier_initialized_) {
+    RTC_LOG_F(LS_WARNING) << "Magnifier initialization failed.";
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+  // Switch to the desktop receiving user input if different from the current
+  // one.
+  std::unique_ptr<Desktop> input_desktop(Desktop::GetInputDesktop());
+  if (input_desktop.get() != NULL && !desktop_.IsSame(*input_desktop)) {
+    // Release GDI resources otherwise SetThreadDesktop will fail.
+    if (desktop_dc_) {
+      ReleaseDC(NULL, desktop_dc_);
+      desktop_dc_ = NULL;
+    }
+    // If SetThreadDesktop() fails, the thread is still assigned a desktop.
+    // So we can continue capture screen bits, just from the wrong desktop.
+    desktop_.SetThreadDesktop(input_desktop.release());
+  }
+
+  DesktopRect rect = GetScreenRect(current_screen_id_, current_device_key_);
+  queue_.MoveToNextFrame();
+  CreateCurrentFrameIfNecessary(rect.size());
+  // CaptureImage may fail in some situations, e.g. windows8 metro mode. So
+  // defer to the fallback capturer if magnifier capturer did not work.
+  if (!CaptureImage(rect)) {
+    RTC_LOG_F(LS_WARNING) << "Magnifier capturer failed to capture a frame.";
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+
+  // Emit the current frame.
+  std::unique_ptr<DesktopFrame> frame = queue_.current_frame()->Share();
+  frame->set_dpi(DesktopVector(GetDeviceCaps(desktop_dc_, LOGPIXELSX),
+                               GetDeviceCaps(desktop_dc_, LOGPIXELSY)));
+  frame->mutable_updated_region()->SetRect(
+      DesktopRect::MakeSize(frame->size()));
+
+  int capture_time_ms = (rtc::TimeNanos() - capture_start_time_nanos) /
+                        rtc::kNumNanosecsPerMillisec;
+  RTC_HISTOGRAM_COUNTS_1000(
+      "WebRTC.DesktopCapture.Win.MagnifierCapturerFrameTime", capture_time_ms);
+  frame->set_capture_time_ms(capture_time_ms);
+  frame->set_capturer_id(DesktopCapturerId::kScreenCapturerWinMagnifier);
+  callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
+}
+
+bool ScreenCapturerWinMagnifier::GetSourceList(SourceList* sources) {
+  return webrtc::GetScreenList(sources);
+}
+
+bool ScreenCapturerWinMagnifier::SelectSource(SourceId id) {
+  if (IsScreenValid(id, &current_device_key_)) {
+    current_screen_id_ = id;
+    return true;
+  }
+
+  return false;
+}
+
+void ScreenCapturerWinMagnifier::SetExcludedWindow(WindowId excluded_window) {
+  excluded_window_ = (HWND)excluded_window;
+  if (excluded_window_ && magnifier_initialized_) {
+    set_window_filter_list_func_(magnifier_window_, MW_FILTERMODE_EXCLUDE, 1,
+                                 &excluded_window_);
+  }
+}
+
+bool ScreenCapturerWinMagnifier::CaptureImage(const DesktopRect& rect) {
+  RTC_DCHECK(magnifier_initialized_);
+
+  // Set the magnifier control to cover the captured rect. The content of the
+  // magnifier control will be the captured image.
+  BOOL result = SetWindowPos(magnifier_window_, NULL, rect.left(), rect.top(),
+                             rect.width(), rect.height(), 0);
+  if (!result) {
+    RTC_LOG_F(LS_WARNING) << "Failed to call SetWindowPos: " << GetLastError()
+                          << ". Rect = {" << rect.left() << ", " << rect.top()
+                          << ", " << rect.right() << ", " << rect.bottom()
+                          << "}";
+    return false;
+  }
+
+  magnifier_capture_succeeded_ = false;
+
+  RECT native_rect = {rect.left(), rect.top(), rect.right(), rect.bottom()};
+
+  TlsSetValue(GetTlsIndex(), this);
+  // OnCaptured will be called via OnMagImageScalingCallback and fill in the
+  // frame before set_window_source_func_ returns.
+  result = set_window_source_func_(magnifier_window_, native_rect);
+
+  if (!result) {
+    RTC_LOG_F(LS_WARNING) << "Failed to call MagSetWindowSource: "
+                          << GetLastError() << ". Rect = {" << rect.left()
+                          << ", " << rect.top() << ", " << rect.right() << ", "
+                          << rect.bottom() << "}";
+    return false;
+  }
+
+  return magnifier_capture_succeeded_;
+}
+
+BOOL ScreenCapturerWinMagnifier::OnMagImageScalingCallback(
+    HWND hwnd,
+    void* srcdata,
+    MAGIMAGEHEADER srcheader,
+    void* destdata,
+    MAGIMAGEHEADER destheader,
+    RECT unclipped,
+    RECT clipped,
+    HRGN dirty) {
+  ScreenCapturerWinMagnifier* owner =
+      reinterpret_cast<ScreenCapturerWinMagnifier*>(TlsGetValue(GetTlsIndex()));
+  TlsSetValue(GetTlsIndex(), nullptr);
+  owner->OnCaptured(srcdata, srcheader);
+
+  return TRUE;
+}
+
+// TODO(zijiehe): These functions are available on Windows Vista or upper, so we
+// do not need to use LoadLibrary and GetProcAddress anymore. Use regular
+// include and function calls instead of a dynamical loaded library.
+bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
+  RTC_DCHECK(!magnifier_initialized_);
+
+  if (GetSystemMetrics(SM_CMONITORS) != 1) {
+    // Do not try to use the magnifier in multi-screen setup (where the API
+    // crashes sometimes).
+    RTC_LOG_F(LS_WARNING) << "Magnifier capturer cannot work on multi-screen "
+                             "system.";
+    return false;
+  }
+
+  desktop_dc_ = GetDC(nullptr);
+
+  mag_lib_handle_ = LoadLibraryW(L"Magnification.dll");
+  if (!mag_lib_handle_)
+    return false;
+
+  // Initialize Magnification API function pointers.
+  mag_initialize_func_ = reinterpret_cast<MagInitializeFunc>(
+      GetProcAddress(mag_lib_handle_, "MagInitialize"));
+  mag_uninitialize_func_ = reinterpret_cast<MagUninitializeFunc>(
+      GetProcAddress(mag_lib_handle_, "MagUninitialize"));
+  set_window_source_func_ = reinterpret_cast<MagSetWindowSourceFunc>(
+      GetProcAddress(mag_lib_handle_, "MagSetWindowSource"));
+  set_window_filter_list_func_ = reinterpret_cast<MagSetWindowFilterListFunc>(
+      GetProcAddress(mag_lib_handle_, "MagSetWindowFilterList"));
+  set_image_scaling_callback_func_ =
+      reinterpret_cast<MagSetImageScalingCallbackFunc>(
+          GetProcAddress(mag_lib_handle_, "MagSetImageScalingCallback"));
+
+  if (!mag_initialize_func_ || !mag_uninitialize_func_ ||
+      !set_window_source_func_ || !set_window_filter_list_func_ ||
+      !set_image_scaling_callback_func_) {
+    RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
+                             "library functions missing.";
+    return false;
+  }
+
+  BOOL result = mag_initialize_func_();
+  if (!result) {
+    RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
+                             "error from MagInitialize "
+                          << GetLastError();
+    return false;
+  }
+
+  HMODULE hInstance = nullptr;
+  result =
+      GetModuleHandleExA(GET_MODULE_HANDLE_EX_FLAG_FROM_ADDRESS |
+                             GET_MODULE_HANDLE_EX_FLAG_UNCHANGED_REFCOUNT,
+                         reinterpret_cast<char*>(&DefWindowProc), &hInstance);
+  if (!result) {
+    mag_uninitialize_func_();
+    RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
+                             "error from GetModulehandleExA "
+                          << GetLastError();
+    return false;
+  }
+
+  // Register the host window class. See the MSDN documentation of the
+  // Magnification API for more infomation.
+  WNDCLASSEXW wcex = {};
+  wcex.cbSize = sizeof(WNDCLASSEX);
+  wcex.lpfnWndProc = &DefWindowProc;
+  wcex.hInstance = hInstance;
+  wcex.hCursor = LoadCursor(nullptr, IDC_ARROW);
+  wcex.lpszClassName = kMagnifierHostClass;
+
+  // Ignore the error which may happen when the class is already registered.
+  RegisterClassExW(&wcex);
+
+  // Create the host window.
+  host_window_ =
+      CreateWindowExW(WS_EX_LAYERED, kMagnifierHostClass, kHostWindowName, 0, 0,
+                      0, 0, 0, nullptr, nullptr, hInstance, nullptr);
+  if (!host_window_) {
+    mag_uninitialize_func_();
+    RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
+                             "error from creating host window "
+                          << GetLastError();
+    return false;
+  }
+
+  // Create the magnifier control.
+  magnifier_window_ = CreateWindowW(kMagnifierWindowClass, kMagnifierWindowName,
+                                    WS_CHILD | WS_VISIBLE, 0, 0, 0, 0,
+                                    host_window_, nullptr, hInstance, nullptr);
+  if (!magnifier_window_) {
+    mag_uninitialize_func_();
+    RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
+                             "error from creating magnifier window "
+                          << GetLastError();
+    return false;
+  }
+
+  // Hide the host window.
+  ShowWindow(host_window_, SW_HIDE);
+
+  // Set the scaling callback to receive captured image.
+  result = set_image_scaling_callback_func_(
+      magnifier_window_,
+      &ScreenCapturerWinMagnifier::OnMagImageScalingCallback);
+  if (!result) {
+    mag_uninitialize_func_();
+    RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
+                             "error from MagSetImageScalingCallback "
+                          << GetLastError();
+    return false;
+  }
+
+  if (excluded_window_) {
+    result = set_window_filter_list_func_(
+        magnifier_window_, MW_FILTERMODE_EXCLUDE, 1, &excluded_window_);
+    if (!result) {
+      mag_uninitialize_func_();
+      RTC_LOG_F(LS_WARNING)
+          << "Failed to initialize ScreenCapturerWinMagnifier: "
+             "error from MagSetWindowFilterList "
+          << GetLastError();
+      return false;
+    }
+  }
+
+  magnifier_initialized_ = true;
+  return true;
+}
+
+void ScreenCapturerWinMagnifier::OnCaptured(void* data,
+                                            const MAGIMAGEHEADER& header) {
+  DesktopFrame* current_frame = queue_.current_frame();
+
+  // Verify the format.
+  // TODO(jiayl): support capturing sources with pixel formats other than RGBA.
+  int captured_bytes_per_pixel = header.cbSize / header.width / header.height;
+  if (header.format != GUID_WICPixelFormat32bppRGBA ||
+      header.width != static_cast<UINT>(current_frame->size().width()) ||
+      header.height != static_cast<UINT>(current_frame->size().height()) ||
+      header.stride != static_cast<UINT>(current_frame->stride()) ||
+      captured_bytes_per_pixel != DesktopFrame::kBytesPerPixel) {
+    RTC_LOG_F(LS_WARNING)
+        << "Output format does not match the captured format: "
+           "width = "
+        << header.width
+        << ", "
+           "height = "
+        << header.height
+        << ", "
+           "stride = "
+        << header.stride
+        << ", "
+           "bpp = "
+        << captured_bytes_per_pixel
+        << ", "
+           "pixel format RGBA ? "
+        << (header.format == GUID_WICPixelFormat32bppRGBA) << ".";
+    return;
+  }
+
+  // Copy the data into the frame.
+  current_frame->CopyPixelsFrom(
+      reinterpret_cast<uint8_t*>(data), header.stride,
+      DesktopRect::MakeXYWH(0, 0, header.width, header.height));
+
+  magnifier_capture_succeeded_ = true;
+}
+
+void ScreenCapturerWinMagnifier::CreateCurrentFrameIfNecessary(
+    const DesktopSize& size) {
+  // If the current buffer is from an older generation then allocate a new one.
+  // Note that we can't reallocate other buffers at this point, since the caller
+  // may still be reading from them.
+  if (!queue_.current_frame() || !queue_.current_frame()->size().equals(size)) {
+    std::unique_ptr<DesktopFrame> frame =
+        shared_memory_factory_
+            ? SharedMemoryDesktopFrame::Create(size,
+                                               shared_memory_factory_.get())
+            : std::unique_ptr<DesktopFrame>(new BasicDesktopFrame(size));
+    queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(std::move(frame)));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
new file mode 100644
index 0000000000..07c5b1e9e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
@@ -0,0 +1,140 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_MAGNIFIER_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_MAGNIFIER_H_
+
+#include <magnification.h>
+#include <wincodec.h>
+#include <windows.h>
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/screen_capture_frame_queue.h"
+#include "modules/desktop_capture/screen_capturer_helper.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "modules/desktop_capture/win/scoped_thread_desktop.h"
+
+namespace webrtc {
+
+class DesktopFrame;
+class DesktopRect;
+
+// Captures the screen using the Magnification API to support window exclusion.
+// Each capturer must run on a dedicated thread because it uses thread local
+// storage for redirecting the library callback. Also the thread must have a UI
+// message loop to handle the window messages for the magnifier window.
+//
+// This class does not detect DesktopFrame::updated_region(), the field is
+// always set to the entire frame rectangle. ScreenCapturerDifferWrapper should
+// be used if that functionality is necessary.
+class ScreenCapturerWinMagnifier : public DesktopCapturer {
+ public:
+  ScreenCapturerWinMagnifier();
+  ~ScreenCapturerWinMagnifier() override;
+
+  ScreenCapturerWinMagnifier(const ScreenCapturerWinMagnifier&) = delete;
+  ScreenCapturerWinMagnifier& operator=(const ScreenCapturerWinMagnifier&) =
+      delete;
+
+  // Overridden from ScreenCapturer:
+  void Start(Callback* callback) override;
+  void SetSharedMemoryFactory(
+      std::unique_ptr<SharedMemoryFactory> shared_memory_factory) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* screens) override;
+  bool SelectSource(SourceId id) override;
+  void SetExcludedWindow(WindowId window) override;
+
+ private:
+  typedef BOOL(WINAPI* MagImageScalingCallback)(HWND hwnd,
+                                                void* srcdata,
+                                                MAGIMAGEHEADER srcheader,
+                                                void* destdata,
+                                                MAGIMAGEHEADER destheader,
+                                                RECT unclipped,
+                                                RECT clipped,
+                                                HRGN dirty);
+  typedef BOOL(WINAPI* MagInitializeFunc)(void);
+  typedef BOOL(WINAPI* MagUninitializeFunc)(void);
+  typedef BOOL(WINAPI* MagSetWindowSourceFunc)(HWND hwnd, RECT rect);
+  typedef BOOL(WINAPI* MagSetWindowFilterListFunc)(HWND hwnd,
+                                                   DWORD dwFilterMode,
+                                                   int count,
+                                                   HWND* pHWND);
+  typedef BOOL(WINAPI* MagSetImageScalingCallbackFunc)(
+      HWND hwnd,
+      MagImageScalingCallback callback);
+
+  static BOOL WINAPI OnMagImageScalingCallback(HWND hwnd,
+                                               void* srcdata,
+                                               MAGIMAGEHEADER srcheader,
+                                               void* destdata,
+                                               MAGIMAGEHEADER destheader,
+                                               RECT unclipped,
+                                               RECT clipped,
+                                               HRGN dirty);
+
+  // Captures the screen within `rect` in the desktop coordinates. Returns true
+  // if succeeded.
+  // It can only capture the primary screen for now. The magnification library
+  // crashes under some screen configurations (e.g. secondary screen on top of
+  // primary screen) if it tries to capture a non-primary screen. The caller
+  // must make sure not calling it on non-primary screens.
+  bool CaptureImage(const DesktopRect& rect);
+
+  // Helper method for setting up the magnifier control. Returns true if
+  // succeeded.
+  bool InitializeMagnifier();
+
+  // Called by OnMagImageScalingCallback to output captured data.
+  void OnCaptured(void* data, const MAGIMAGEHEADER& header);
+
+  // Makes sure the current frame exists and matches `size`.
+  void CreateCurrentFrameIfNecessary(const DesktopSize& size);
+
+  Callback* callback_ = nullptr;
+  std::unique_ptr<SharedMemoryFactory> shared_memory_factory_;
+  ScreenId current_screen_id_ = kFullDesktopScreenId;
+  std::wstring current_device_key_;
+  HWND excluded_window_ = NULL;
+
+  // Queue of the frames buffers.
+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
+
+  ScopedThreadDesktop desktop_;
+
+  // Used for getting the screen dpi.
+  HDC desktop_dc_ = NULL;
+
+  HMODULE mag_lib_handle_ = NULL;
+  MagInitializeFunc mag_initialize_func_ = nullptr;
+  MagUninitializeFunc mag_uninitialize_func_ = nullptr;
+  MagSetWindowSourceFunc set_window_source_func_ = nullptr;
+  MagSetWindowFilterListFunc set_window_filter_list_func_ = nullptr;
+  MagSetImageScalingCallbackFunc set_image_scaling_callback_func_ = nullptr;
+
+  // The hidden window hosting the magnifier control.
+  HWND host_window_ = NULL;
+  // The magnifier control that captures the screen.
+  HWND magnifier_window_ = NULL;
+
+  // True if the magnifier control has been successfully initialized.
+  bool magnifier_initialized_ = false;
+
+  // True if the last OnMagImageScalingCallback was called and handled
+  // successfully. Reset at the beginning of each CaptureImage call.
+  bool magnifier_capture_succeeded_ = true;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURER_WIN_MAGNIFIER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/selected_window_context.cc b/third_party/libwebrtc/modules/desktop_capture/win/selected_window_context.cc
new file mode 100644
index 0000000000..398ea1e53a
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/selected_window_context.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/selected_window_context.h"
+
+namespace webrtc {
+
+SelectedWindowContext::SelectedWindowContext(
+    HWND selected_window,
+    DesktopRect selected_window_rect,
+    WindowCaptureHelperWin* window_capture_helper)
+    : selected_window_(selected_window),
+      selected_window_rect_(selected_window_rect),
+      window_capture_helper_(window_capture_helper) {
+  selected_window_thread_id_ =
+      GetWindowThreadProcessId(selected_window, &selected_window_process_id_);
+}
+
+bool SelectedWindowContext::IsSelectedWindowValid() const {
+  return selected_window_thread_id_ != 0;
+}
+
+bool SelectedWindowContext::IsWindowOwnedBySelectedWindow(HWND hwnd) const {
+  // This check works for drop-down menus & dialog pop-up windows.
+  if (GetAncestor(hwnd, GA_ROOTOWNER) == selected_window_) {
+    return true;
+  }
+
+  // Assume that all other windows are unrelated to the selected window.
+  // This will cause some windows that are actually related to be missed,
+  // e.g. context menus and tool-tips, but avoids the risk of capturing
+  // unrelated windows. Using heuristics such as matching the thread and
+  // process Ids suffers from false-positives, e.g. in multi-document
+  // applications.
+
+  return false;
+}
+
+bool SelectedWindowContext::IsWindowOverlappingSelectedWindow(HWND hwnd) const {
+  return window_capture_helper_->AreWindowsOverlapping(hwnd, selected_window_,
+                                                       selected_window_rect_);
+}
+
+HWND SelectedWindowContext::selected_window() const {
+  return selected_window_;
+}
+
+WindowCaptureHelperWin* SelectedWindowContext::window_capture_helper() const {
+  return window_capture_helper_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/selected_window_context.h b/third_party/libwebrtc/modules/desktop_capture/win/selected_window_context.h
new file mode 100644
index 0000000000..99e38e3fa2
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/selected_window_context.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_SELECTED_WINDOW_CONTEXT_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_SELECTED_WINDOW_CONTEXT_H_
+
+#include <windows.h>
+
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/win/window_capture_utils.h"
+
+namespace webrtc {
+
+class SelectedWindowContext {
+ public:
+  SelectedWindowContext(HWND selected_window,
+                        DesktopRect selected_window_rect,
+                        WindowCaptureHelperWin* window_capture_helper);
+
+  bool IsSelectedWindowValid() const;
+
+  bool IsWindowOwnedBySelectedWindow(HWND hwnd) const;
+  bool IsWindowOverlappingSelectedWindow(HWND hwnd) const;
+
+  HWND selected_window() const;
+  WindowCaptureHelperWin* window_capture_helper() const;
+
+ private:
+  const HWND selected_window_;
+  const DesktopRect selected_window_rect_;
+  WindowCaptureHelperWin* const window_capture_helper_;
+  DWORD selected_window_thread_id_;
+  DWORD selected_window_process_id_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_SELECTED_WINDOW_CONTEXT_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/test_support/test_window.cc b/third_party/libwebrtc/modules/desktop_capture/win/test_support/test_window.cc
new file mode 100644
index 0000000000..c07ff74aa5
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/test_support/test_window.cc
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/test_support/test_window.h"
+
+namespace webrtc {
+namespace {
+
+const WCHAR kWindowClass[] = L"DesktopCaptureTestWindowClass";
+const int kWindowHeight = 200;
+const int kWindowWidth = 300;
+
+LRESULT CALLBACK WindowProc(HWND hwnd,
+                            UINT msg,
+                            WPARAM w_param,
+                            LPARAM l_param) {
+  switch (msg) {
+    case WM_PAINT:
+      PAINTSTRUCT paint_struct;
+      HDC hdc = BeginPaint(hwnd, &paint_struct);
+
+      // Paint the window so the color is consistent and we can inspect the
+      // pixels in tests and know what to expect.
+      FillRect(hdc, &paint_struct.rcPaint,
+               CreateSolidBrush(RGB(kTestWindowRValue, kTestWindowGValue,
+                                    kTestWindowBValue)));
+
+      EndPaint(hwnd, &paint_struct);
+  }
+  return DefWindowProc(hwnd, msg, w_param, l_param);
+}
+
+}  // namespace
+
+WindowInfo CreateTestWindow(const WCHAR* window_title,
+                            const int height,
+                            const int width,
+                            const LONG extended_styles) {
+  WindowInfo info;
+  ::GetModuleHandleExW(GET_MODULE_HANDLE_EX_FLAG_FROM_ADDRESS |
+                           GET_MODULE_HANDLE_EX_FLAG_UNCHANGED_REFCOUNT,
+                       reinterpret_cast<LPCWSTR>(&WindowProc),
+                       &info.window_instance);
+
+  WNDCLASSEXW wcex;
+  memset(&wcex, 0, sizeof(wcex));
+  wcex.cbSize = sizeof(wcex);
+  wcex.style = CS_HREDRAW | CS_VREDRAW;
+  wcex.hInstance = info.window_instance;
+  wcex.lpfnWndProc = &WindowProc;
+  wcex.lpszClassName = kWindowClass;
+  info.window_class = ::RegisterClassExW(&wcex);
+
+  // Use the default height and width if the caller did not supply the optional
+  // height and width parameters, or if they supplied invalid values.
+  int window_height = height <= 0 ? kWindowHeight : height;
+  int window_width = width <= 0 ? kWindowWidth : width;
+  info.hwnd =
+      ::CreateWindowExW(extended_styles, kWindowClass, window_title,
+                        WS_OVERLAPPEDWINDOW, CW_USEDEFAULT, CW_USEDEFAULT,
+                        window_width, window_height, /*parent_window=*/nullptr,
+                        /*menu_bar=*/nullptr, info.window_instance,
+                        /*additional_params=*/nullptr);
+
+  ::ShowWindow(info.hwnd, SW_SHOWNORMAL);
+  ::UpdateWindow(info.hwnd);
+  return info;
+}
+
+void ResizeTestWindow(const HWND hwnd, const int width, const int height) {
+  // SWP_NOMOVE results in the x and y params being ignored.
+  ::SetWindowPos(hwnd, HWND_TOP, /*x-coord=*/0, /*y-coord=*/0, width, height,
+                 SWP_SHOWWINDOW | SWP_NOMOVE);
+  ::UpdateWindow(hwnd);
+}
+
+void MoveTestWindow(const HWND hwnd, const int x, const int y) {
+  // SWP_NOSIZE results in the width and height params being ignored.
+  ::SetWindowPos(hwnd, HWND_TOP, x, y, /*width=*/0, /*height=*/0,
+                 SWP_SHOWWINDOW | SWP_NOSIZE);
+  ::UpdateWindow(hwnd);
+}
+
+void MinimizeTestWindow(const HWND hwnd) {
+  ::ShowWindow(hwnd, SW_MINIMIZE);
+}
+
+void UnminimizeTestWindow(const HWND hwnd) {
+  ::OpenIcon(hwnd);
+}
+
+void DestroyTestWindow(WindowInfo info) {
+  ::DestroyWindow(info.hwnd);
+  ::UnregisterClass(MAKEINTATOM(info.window_class), info.window_instance);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/test_support/test_window.h b/third_party/libwebrtc/modules/desktop_capture/win/test_support/test_window.h
new file mode 100644
index 0000000000..b055da7ccd
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/test_support/test_window.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_TEST_SUPPORT_TEST_WINDOW_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_TEST_SUPPORT_TEST_WINDOW_H_
+
+#ifndef WIN32_LEAN_AND_MEAN
+#define WIN32_LEAN_AND_MEAN
+#endif
+#include <windows.h>
+
+namespace webrtc {
+
+typedef unsigned char uint8_t;
+
+// Define an arbitrary color for the test window with unique R, G, and B values
+// so consumers can verify captured content in tests.
+const uint8_t kTestWindowRValue = 191;
+const uint8_t kTestWindowGValue = 99;
+const uint8_t kTestWindowBValue = 12;
+
+struct WindowInfo {
+  HWND hwnd;
+  HINSTANCE window_instance;
+  ATOM window_class;
+};
+
+WindowInfo CreateTestWindow(const WCHAR* window_title,
+                            int height = 0,
+                            int width = 0,
+                            LONG extended_styles = 0);
+
+void ResizeTestWindow(HWND hwnd, int width, int height);
+
+void MoveTestWindow(HWND hwnd, int x, int y);
+
+void MinimizeTestWindow(HWND hwnd);
+
+void UnminimizeTestWindow(HWND hwnd);
+
+void DestroyTestWindow(WindowInfo info);
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_TEST_SUPPORT_TEST_WINDOW_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_session.cc b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_session.cc
new file mode 100644
index 0000000000..ea5565c89c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_session.cc
@@ -0,0 +1,429 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/wgc_capture_session.h"
+
+#include <DispatcherQueue.h>
+#include <windows.graphics.capture.interop.h>
+#include <windows.graphics.directX.direct3d11.interop.h>
+#include <windows.graphics.h>
+#include <wrl/client.h>
+#include <wrl/event.h>
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "modules/desktop_capture/win/wgc_desktop_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/win/create_direct3d_device.h"
+#include "rtc_base/win/get_activation_factory.h"
+#include "system_wrappers/include/metrics.h"
+
+using Microsoft::WRL::ComPtr;
+namespace WGC = ABI::Windows::Graphics::Capture;
+
+namespace webrtc {
+namespace {
+
+// We must use a BGRA pixel format that has 4 bytes per pixel, as required by
+// the DesktopFrame interface.
+constexpr auto kPixelFormat = ABI::Windows::Graphics::DirectX::
+    DirectXPixelFormat::DirectXPixelFormat_B8G8R8A8UIntNormalized;
+
+// The maximum time `GetFrame` will wait for a frame to arrive, if we don't have
+// any in the pool.
+constexpr TimeDelta kMaxWaitForFrame = TimeDelta::Millis(50);
+constexpr TimeDelta kMaxWaitForFirstFrame = TimeDelta::Millis(500);
+
+// These values are persisted to logs. Entries should not be renumbered and
+// numeric values should never be reused.
+enum class StartCaptureResult {
+  kSuccess = 0,
+  kSourceClosed = 1,
+  kAddClosedFailed = 2,
+  kDxgiDeviceCastFailed = 3,
+  kD3dDelayLoadFailed = 4,
+  kD3dDeviceCreationFailed = 5,
+  kFramePoolActivationFailed = 6,
+  // kFramePoolCastFailed = 7, (deprecated)
+  // kGetItemSizeFailed = 8, (deprecated)
+  kCreateFramePoolFailed = 9,
+  kCreateCaptureSessionFailed = 10,
+  kStartCaptureFailed = 11,
+  kMaxValue = kStartCaptureFailed
+};
+
+// These values are persisted to logs. Entries should not be renumbered and
+// numeric values should never be reused.
+enum class GetFrameResult {
+  kSuccess = 0,
+  kItemClosed = 1,
+  kTryGetNextFrameFailed = 2,
+  kFrameDropped = 3,
+  kGetSurfaceFailed = 4,
+  kDxgiInterfaceAccessFailed = 5,
+  kTexture2dCastFailed = 6,
+  kCreateMappedTextureFailed = 7,
+  kMapFrameFailed = 8,
+  kGetContentSizeFailed = 9,
+  kResizeMappedTextureFailed = 10,
+  kRecreateFramePoolFailed = 11,
+  kMaxValue = kRecreateFramePoolFailed
+};
+
+void RecordStartCaptureResult(StartCaptureResult error) {
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.DesktopCapture.Win.WgcCaptureSessionStartResult",
+      static_cast<int>(error), static_cast<int>(StartCaptureResult::kMaxValue));
+}
+
+void RecordGetFrameResult(GetFrameResult error) {
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.DesktopCapture.Win.WgcCaptureSessionGetFrameResult",
+      static_cast<int>(error), static_cast<int>(GetFrameResult::kMaxValue));
+}
+
+}  // namespace
+
+WgcCaptureSession::WgcCaptureSession(ComPtr<ID3D11Device> d3d11_device,
+                                     ComPtr<WGC::IGraphicsCaptureItem> item,
+                                     ABI::Windows::Graphics::SizeInt32 size)
+    : d3d11_device_(std::move(d3d11_device)),
+      item_(std::move(item)),
+      size_(size) {}
+WgcCaptureSession::~WgcCaptureSession() {
+  RemoveEventHandlers();
+}
+
+HRESULT WgcCaptureSession::StartCapture(const DesktopCaptureOptions& options) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  RTC_DCHECK(!is_capture_started_);
+
+  if (item_closed_) {
+    RTC_LOG(LS_ERROR) << "The target source has been closed.";
+    RecordStartCaptureResult(StartCaptureResult::kSourceClosed);
+    return E_ABORT;
+  }
+
+  RTC_DCHECK(d3d11_device_);
+  RTC_DCHECK(item_);
+
+  // Listen for the Closed event, to detect if the source we are capturing is
+  // closed (e.g. application window is closed or monitor is disconnected). If
+  // it is, we should abort the capture.
+  item_closed_token_ = std::make_unique<EventRegistrationToken>();
+  auto closed_handler =
+      Microsoft::WRL::Callback<ABI::Windows::Foundation::ITypedEventHandler<
+          WGC::GraphicsCaptureItem*, IInspectable*>>(
+          this, &WgcCaptureSession::OnItemClosed);
+  HRESULT hr =
+      item_->add_Closed(closed_handler.Get(), item_closed_token_.get());
+  if (FAILED(hr)) {
+    RecordStartCaptureResult(StartCaptureResult::kAddClosedFailed);
+    return hr;
+  }
+
+  ComPtr<IDXGIDevice> dxgi_device;
+  hr = d3d11_device_->QueryInterface(IID_PPV_ARGS(&dxgi_device));
+  if (FAILED(hr)) {
+    RecordStartCaptureResult(StartCaptureResult::kDxgiDeviceCastFailed);
+    return hr;
+  }
+
+  if (!ResolveCoreWinRTDirect3DDelayload()) {
+    RecordStartCaptureResult(StartCaptureResult::kD3dDelayLoadFailed);
+    return E_FAIL;
+  }
+
+  hr = CreateDirect3DDeviceFromDXGIDevice(dxgi_device.Get(), &direct3d_device_);
+  if (FAILED(hr)) {
+    RecordStartCaptureResult(StartCaptureResult::kD3dDeviceCreationFailed);
+    return hr;
+  }
+
+  ComPtr<WGC::IDirect3D11CaptureFramePoolStatics> frame_pool_statics;
+  hr = GetActivationFactory<
+      WGC::IDirect3D11CaptureFramePoolStatics,
+      RuntimeClass_Windows_Graphics_Capture_Direct3D11CaptureFramePool>(
+      &frame_pool_statics);
+  if (FAILED(hr)) {
+    RecordStartCaptureResult(StartCaptureResult::kFramePoolActivationFailed);
+    return hr;
+  }
+
+  hr = frame_pool_statics->Create(direct3d_device_.Get(), kPixelFormat,
+                                  kNumBuffers, size_, &frame_pool_);
+  if (FAILED(hr)) {
+    RecordStartCaptureResult(StartCaptureResult::kCreateFramePoolFailed);
+    return hr;
+  }
+
+  frames_in_pool_ = 0;
+
+  // Because `WgcCapturerWin` created a `DispatcherQueue`, and we created
+  // `frame_pool_` via `Create`, the `FrameArrived` event will be delivered on
+  // the current thread.
+  frame_arrived_token_ = std::make_unique<EventRegistrationToken>();
+  auto frame_arrived_handler =
+      Microsoft::WRL::Callback<ABI::Windows::Foundation::ITypedEventHandler<
+          WGC::Direct3D11CaptureFramePool*, IInspectable*>>(
+          this, &WgcCaptureSession::OnFrameArrived);
+  hr = frame_pool_->add_FrameArrived(frame_arrived_handler.Get(),
+                                     frame_arrived_token_.get());
+
+  hr = frame_pool_->CreateCaptureSession(item_.Get(), &session_);
+  if (FAILED(hr)) {
+    RecordStartCaptureResult(StartCaptureResult::kCreateCaptureSessionFailed);
+    return hr;
+  }
+
+  if (!options.prefer_cursor_embedded()) {
+    ComPtr<ABI::Windows::Graphics::Capture::IGraphicsCaptureSession2> session2;
+    if (SUCCEEDED(session_->QueryInterface(
+            ABI::Windows::Graphics::Capture::IID_IGraphicsCaptureSession2,
+            &session2))) {
+      session2->put_IsCursorCaptureEnabled(false);
+    }
+  }
+
+  hr = session_->StartCapture();
+  if (FAILED(hr)) {
+    RTC_LOG(LS_ERROR) << "Failed to start CaptureSession: " << hr;
+    RecordStartCaptureResult(StartCaptureResult::kStartCaptureFailed);
+    return hr;
+  }
+
+  RecordStartCaptureResult(StartCaptureResult::kSuccess);
+
+  is_capture_started_ = true;
+  return hr;
+}
+
+HRESULT WgcCaptureSession::GetFrame(
+    std::unique_ptr<DesktopFrame>* output_frame) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  if (item_closed_) {
+    RTC_LOG(LS_ERROR) << "The target source has been closed.";
+    RecordGetFrameResult(GetFrameResult::kItemClosed);
+    return E_ABORT;
+  }
+
+  RTC_DCHECK(is_capture_started_);
+
+  if (frames_in_pool_ < 1)
+    wait_for_frame_event_.Wait(first_frame_ ? kMaxWaitForFirstFrame
+                                            : kMaxWaitForFrame);
+
+  ComPtr<WGC::IDirect3D11CaptureFrame> capture_frame;
+  HRESULT hr = frame_pool_->TryGetNextFrame(&capture_frame);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_ERROR) << "TryGetNextFrame failed: " << hr;
+    RecordGetFrameResult(GetFrameResult::kTryGetNextFrameFailed);
+    return hr;
+  }
+
+  if (!capture_frame) {
+    RecordGetFrameResult(GetFrameResult::kFrameDropped);
+    return hr;
+  }
+
+  first_frame_ = false;
+  --frames_in_pool_;
+
+  // We need to get `capture_frame` as an `ID3D11Texture2D` so that we can get
+  // the raw image data in the format required by the `DesktopFrame` interface.
+  ComPtr<ABI::Windows::Graphics::DirectX::Direct3D11::IDirect3DSurface>
+      d3d_surface;
+  hr = capture_frame->get_Surface(&d3d_surface);
+  if (FAILED(hr)) {
+    RecordGetFrameResult(GetFrameResult::kGetSurfaceFailed);
+    return hr;
+  }
+
+  ComPtr<Windows::Graphics::DirectX::Direct3D11::IDirect3DDxgiInterfaceAccess>
+      direct3DDxgiInterfaceAccess;
+  hr = d3d_surface->QueryInterface(IID_PPV_ARGS(&direct3DDxgiInterfaceAccess));
+  if (FAILED(hr)) {
+    RecordGetFrameResult(GetFrameResult::kDxgiInterfaceAccessFailed);
+    return hr;
+  }
+
+  ComPtr<ID3D11Texture2D> texture_2D;
+  hr = direct3DDxgiInterfaceAccess->GetInterface(IID_PPV_ARGS(&texture_2D));
+  if (FAILED(hr)) {
+    RecordGetFrameResult(GetFrameResult::kTexture2dCastFailed);
+    return hr;
+  }
+
+  if (!mapped_texture_) {
+    hr = CreateMappedTexture(texture_2D);
+    if (FAILED(hr)) {
+      RecordGetFrameResult(GetFrameResult::kCreateMappedTextureFailed);
+      return hr;
+    }
+  }
+
+  // We need to copy `texture_2D` into `mapped_texture_` as the latter has the
+  // D3D11_CPU_ACCESS_READ flag set, which lets us access the image data.
+  // Otherwise it would only be readable by the GPU.
+  ComPtr<ID3D11DeviceContext> d3d_context;
+  d3d11_device_->GetImmediateContext(&d3d_context);
+
+  ABI::Windows::Graphics::SizeInt32 new_size;
+  hr = capture_frame->get_ContentSize(&new_size);
+  if (FAILED(hr)) {
+    RecordGetFrameResult(GetFrameResult::kGetContentSizeFailed);
+    return hr;
+  }
+
+  // If the size changed, we must resize `mapped_texture_` and `frame_pool_` to
+  // fit the new size. This must be done before `CopySubresourceRegion` so that
+  // the textures are the same size.
+  if (size_.Height != new_size.Height || size_.Width != new_size.Width) {
+    hr = CreateMappedTexture(texture_2D, new_size.Width, new_size.Height);
+    if (FAILED(hr)) {
+      RecordGetFrameResult(GetFrameResult::kResizeMappedTextureFailed);
+      return hr;
+    }
+
+    hr = frame_pool_->Recreate(direct3d_device_.Get(), kPixelFormat,
+                               kNumBuffers, new_size);
+    if (FAILED(hr)) {
+      RecordGetFrameResult(GetFrameResult::kRecreateFramePoolFailed);
+      return hr;
+    }
+  }
+
+  // If the size has changed since the last capture, we must be sure to use
+  // the smaller dimensions. Otherwise we might overrun our buffer, or
+  // read stale data from the last frame.
+  int image_height = std::min(size_.Height, new_size.Height);
+  int image_width = std::min(size_.Width, new_size.Width);
+
+  D3D11_BOX copy_region;
+  copy_region.left = 0;
+  copy_region.top = 0;
+  copy_region.right = image_width;
+  copy_region.bottom = image_height;
+  // Our textures are 2D so we just want one "slice" of the box.
+  copy_region.front = 0;
+  copy_region.back = 1;
+  d3d_context->CopySubresourceRegion(mapped_texture_.Get(),
+                                     /*dst_subresource_index=*/0, /*dst_x=*/0,
+                                     /*dst_y=*/0, /*dst_z=*/0, texture_2D.Get(),
+                                     /*src_subresource_index=*/0, &copy_region);
+
+  D3D11_MAPPED_SUBRESOURCE map_info;
+  hr = d3d_context->Map(mapped_texture_.Get(), /*subresource_index=*/0,
+                        D3D11_MAP_READ, /*D3D11_MAP_FLAG_DO_NOT_WAIT=*/0,
+                        &map_info);
+  if (FAILED(hr)) {
+    RecordGetFrameResult(GetFrameResult::kMapFrameFailed);
+    return hr;
+  }
+
+  int row_data_length = image_width * DesktopFrame::kBytesPerPixel;
+
+  // Make a copy of the data pointed to by `map_info.pData` so we are free to
+  // unmap our texture.
+  uint8_t* src_data = static_cast<uint8_t*>(map_info.pData);
+  std::vector<uint8_t> image_data;
+  image_data.resize(image_height * row_data_length);
+  uint8_t* image_data_ptr = image_data.data();
+  for (int i = 0; i < image_height; i++) {
+    memcpy(image_data_ptr, src_data, row_data_length);
+    image_data_ptr += row_data_length;
+    src_data += map_info.RowPitch;
+  }
+
+  d3d_context->Unmap(mapped_texture_.Get(), 0);
+
+  // Transfer ownership of `image_data` to the output_frame.
+  DesktopSize size(image_width, image_height);
+  *output_frame = std::make_unique<WgcDesktopFrame>(size, row_data_length,
+                                                    std::move(image_data));
+
+  size_ = new_size;
+  RecordGetFrameResult(GetFrameResult::kSuccess);
+  return hr;
+}
+
+HRESULT WgcCaptureSession::CreateMappedTexture(
+    ComPtr<ID3D11Texture2D> src_texture,
+    UINT width,
+    UINT height) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  D3D11_TEXTURE2D_DESC src_desc;
+  src_texture->GetDesc(&src_desc);
+  D3D11_TEXTURE2D_DESC map_desc;
+  map_desc.Width = width == 0 ? src_desc.Width : width;
+  map_desc.Height = height == 0 ? src_desc.Height : height;
+  map_desc.MipLevels = src_desc.MipLevels;
+  map_desc.ArraySize = src_desc.ArraySize;
+  map_desc.Format = src_desc.Format;
+  map_desc.SampleDesc = src_desc.SampleDesc;
+  map_desc.Usage = D3D11_USAGE_STAGING;
+  map_desc.BindFlags = 0;
+  map_desc.CPUAccessFlags = D3D11_CPU_ACCESS_READ;
+  map_desc.MiscFlags = 0;
+  return d3d11_device_->CreateTexture2D(&map_desc, nullptr, &mapped_texture_);
+}
+
+HRESULT WgcCaptureSession::OnFrameArrived(
+    WGC::IDirect3D11CaptureFramePool* sender,
+    IInspectable* event_args) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  RTC_DCHECK_LT(frames_in_pool_, kNumBuffers);
+  ++frames_in_pool_;
+  wait_for_frame_event_.Set();
+  return S_OK;
+}
+
+HRESULT WgcCaptureSession::OnItemClosed(WGC::IGraphicsCaptureItem* sender,
+                                        IInspectable* event_args) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  RTC_LOG(LS_INFO) << "Capture target has been closed.";
+  item_closed_ = true;
+
+  RemoveEventHandlers();
+
+  // Do not attempt to free resources in the OnItemClosed handler, as this
+  // causes a race where we try to delete the item that is calling us. Removing
+  // the event handlers and setting `item_closed_` above is sufficient to ensure
+  // that the resources are no longer used, and the next time the capturer tries
+  // to get a frame, we will report a permanent failure and be destroyed.
+  return S_OK;
+}
+
+void WgcCaptureSession::RemoveEventHandlers() {
+  HRESULT hr;
+  if (frame_pool_ && frame_arrived_token_) {
+    hr = frame_pool_->remove_FrameArrived(*frame_arrived_token_);
+    frame_arrived_token_.reset();
+    if (FAILED(hr)) {
+      RTC_LOG(LS_WARNING) << "Failed to remove FrameArrived event handler: "
+                          << hr;
+    }
+  }
+  if (item_ && item_closed_token_) {
+    hr = item_->remove_Closed(*item_closed_token_);
+    item_closed_token_.reset();
+    if (FAILED(hr))
+      RTC_LOG(LS_WARNING) << "Failed to remove Closed event handler: " << hr;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_session.h b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_session.h
new file mode 100644
index 0000000000..dfe1fa60bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_session.h
@@ -0,0 +1,137 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_WGC_CAPTURE_SESSION_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_WGC_CAPTURE_SESSION_H_
+
+#include <d3d11.h>
+#include <windows.graphics.capture.h>
+#include <windows.graphics.h>
+#include <wrl/client.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/win/wgc_capture_source.h"
+#include "rtc_base/event.h"
+
+namespace webrtc {
+
+class WgcCaptureSession final {
+ public:
+  WgcCaptureSession(
+      Microsoft::WRL::ComPtr<ID3D11Device> d3d11_device,
+      Microsoft::WRL::ComPtr<
+          ABI::Windows::Graphics::Capture::IGraphicsCaptureItem> item,
+      ABI::Windows::Graphics::SizeInt32 size);
+
+  // Disallow copy and assign.
+  WgcCaptureSession(const WgcCaptureSession&) = delete;
+  WgcCaptureSession& operator=(const WgcCaptureSession&) = delete;
+
+  ~WgcCaptureSession();
+
+  HRESULT StartCapture(const DesktopCaptureOptions& options);
+
+  // Returns a frame from the frame pool, if any are present.
+  HRESULT GetFrame(std::unique_ptr<DesktopFrame>* output_frame);
+
+  bool IsCaptureStarted() const {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    return is_capture_started_;
+  }
+
+  // We keep 2 buffers in the frame pool to balance the staleness of the frame
+  // with having to wait for frames to arrive too frequently. Too many buffers
+  // will lead to a high latency, and too few will lead to poor performance.
+  // We make this public for tests.
+  static constexpr int kNumBuffers = 2;
+
+ private:
+  // Initializes `mapped_texture_` with the properties of the `src_texture`,
+  // overrides the values of some necessary properties like the
+  // D3D11_CPU_ACCESS_READ flag. Also has optional parameters for what size
+  // `mapped_texture_` should be, if they aren't provided we will use the size
+  // of `src_texture`.
+  HRESULT CreateMappedTexture(
+      Microsoft::WRL::ComPtr<ID3D11Texture2D> src_texture,
+      UINT width = 0,
+      UINT height = 0);
+
+  // Event handler for `item_`'s Closed event.
+  HRESULT OnItemClosed(
+      ABI::Windows::Graphics::Capture::IGraphicsCaptureItem* sender,
+      IInspectable* event_args);
+
+  // Event handler for `frame_pool_`'s FrameArrived event.
+  HRESULT OnFrameArrived(
+      ABI::Windows::Graphics::Capture::IDirect3D11CaptureFramePool* sender,
+      IInspectable* event_args);
+
+  void RemoveEventHandlers();
+
+  // We wait on this event in `GetFrame` if there are no frames in the pool.
+  // `OnFrameArrived` will set the event so we can proceed.
+  rtc::Event wait_for_frame_event_;
+  int frames_in_pool_;
+
+  // We're willing to wait for a frame a little longer if it's the first one.
+  bool first_frame_ = true;
+
+  std::unique_ptr<EventRegistrationToken> frame_arrived_token_;
+  std::unique_ptr<EventRegistrationToken> item_closed_token_;
+
+  // A Direct3D11 Device provided by the caller. We use this to create an
+  // IDirect3DDevice, and also to create textures that will hold the image data.
+  Microsoft::WRL::ComPtr<ID3D11Device> d3d11_device_;
+
+  // This item represents what we are capturing, we use it to create the
+  // capture session, and also to listen for the Closed event.
+  Microsoft::WRL::ComPtr<ABI::Windows::Graphics::Capture::IGraphicsCaptureItem>
+      item_;
+
+  // The IDirect3DDevice is necessary to instantiate the frame pool.
+  Microsoft::WRL::ComPtr<
+      ABI::Windows::Graphics::DirectX::Direct3D11::IDirect3DDevice>
+      direct3d_device_;
+
+  // The frame pool is where frames are deposited during capture, we retrieve
+  // them from here with TryGetNextFrame().
+  Microsoft::WRL::ComPtr<
+      ABI::Windows::Graphics::Capture::IDirect3D11CaptureFramePool>
+      frame_pool_;
+
+  // This texture holds the final image data. We made it a member so we can
+  // reuse it, instead of having to create a new texture every time we grab a
+  // frame.
+  Microsoft::WRL::ComPtr<ID3D11Texture2D> mapped_texture_;
+
+  // This is the size of `mapped_texture_` and the buffers in `frame_pool_`. We
+  // store this as a member so we can compare it to the size of incoming frames
+  // and resize if necessary.
+  ABI::Windows::Graphics::SizeInt32 size_;
+
+  // The capture session lets us set properties about the capture before it
+  // starts such as whether to capture the mouse cursor, and it lets us tell WGC
+  // to start capturing frames.
+  Microsoft::WRL::ComPtr<
+      ABI::Windows::Graphics::Capture::IGraphicsCaptureSession>
+      session_;
+
+  bool item_closed_ = false;
+  bool is_capture_started_ = false;
+
+  SequenceChecker sequence_checker_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_WGC_CAPTURE_SESSION_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_source.cc b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_source.cc
new file mode 100644
index 0000000000..24e6129ec7
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_source.cc
@@ -0,0 +1,218 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/wgc_capture_source.h"
+
+#include <dwmapi.h>
+#include <windows.graphics.capture.interop.h>
+#include <windows.h>
+
+#include <utility>
+
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "modules/desktop_capture/win/window_capture_utils.h"
+#include "rtc_base/win/get_activation_factory.h"
+
+using Microsoft::WRL::ComPtr;
+namespace WGC = ABI::Windows::Graphics::Capture;
+
+namespace webrtc {
+
+WgcCaptureSource::WgcCaptureSource(DesktopCapturer::SourceId source_id)
+    : source_id_(source_id) {}
+WgcCaptureSource::~WgcCaptureSource() = default;
+
+bool WgcCaptureSource::IsCapturable() {
+  // If we can create a capture item, then we can capture it. Unfortunately,
+  // we can't cache this item because it may be created in a different COM
+  // apartment than where capture will eventually start from.
+  ComPtr<WGC::IGraphicsCaptureItem> item;
+  return SUCCEEDED(CreateCaptureItem(&item));
+}
+
+bool WgcCaptureSource::FocusOnSource() {
+  return false;
+}
+
+ABI::Windows::Graphics::SizeInt32 WgcCaptureSource::GetSize() {
+  if (!item_)
+    return {0, 0};
+
+  ABI::Windows::Graphics::SizeInt32 item_size;
+  HRESULT hr = item_->get_Size(&item_size);
+  if (FAILED(hr))
+    return {0, 0};
+
+  return item_size;
+}
+
+HRESULT WgcCaptureSource::GetCaptureItem(
+    ComPtr<WGC::IGraphicsCaptureItem>* result) {
+  HRESULT hr = S_OK;
+  if (!item_)
+    hr = CreateCaptureItem(&item_);
+
+  *result = item_;
+  return hr;
+}
+
+WgcCaptureSourceFactory::~WgcCaptureSourceFactory() = default;
+
+WgcWindowSourceFactory::WgcWindowSourceFactory() = default;
+WgcWindowSourceFactory::~WgcWindowSourceFactory() = default;
+
+std::unique_ptr<WgcCaptureSource> WgcWindowSourceFactory::CreateCaptureSource(
+    DesktopCapturer::SourceId source_id) {
+  return std::make_unique<WgcWindowSource>(source_id);
+}
+
+WgcScreenSourceFactory::WgcScreenSourceFactory() = default;
+WgcScreenSourceFactory::~WgcScreenSourceFactory() = default;
+
+std::unique_ptr<WgcCaptureSource> WgcScreenSourceFactory::CreateCaptureSource(
+    DesktopCapturer::SourceId source_id) {
+  return std::make_unique<WgcScreenSource>(source_id);
+}
+
+WgcWindowSource::WgcWindowSource(DesktopCapturer::SourceId source_id)
+    : WgcCaptureSource(source_id) {}
+WgcWindowSource::~WgcWindowSource() = default;
+
+DesktopVector WgcWindowSource::GetTopLeft() {
+  DesktopRect window_rect;
+  if (!GetWindowRect(reinterpret_cast<HWND>(GetSourceId()), &window_rect))
+    return DesktopVector();
+
+  return window_rect.top_left();
+}
+
+ABI::Windows::Graphics::SizeInt32 WgcWindowSource::GetSize() {
+  RECT window_rect;
+  HRESULT hr = ::DwmGetWindowAttribute(
+      reinterpret_cast<HWND>(GetSourceId()), DWMWA_EXTENDED_FRAME_BOUNDS,
+      reinterpret_cast<void*>(&window_rect), sizeof(window_rect));
+  if (FAILED(hr))
+    return WgcCaptureSource::GetSize();
+
+  return {window_rect.right - window_rect.left,
+          window_rect.bottom - window_rect.top};
+}
+
+bool WgcWindowSource::IsCapturable() {
+  if (!IsWindowValidAndVisible(reinterpret_cast<HWND>(GetSourceId())))
+    return false;
+
+  return WgcCaptureSource::IsCapturable();
+}
+
+bool WgcWindowSource::FocusOnSource() {
+  if (!IsWindowValidAndVisible(reinterpret_cast<HWND>(GetSourceId())))
+    return false;
+
+  return ::BringWindowToTop(reinterpret_cast<HWND>(GetSourceId())) &&
+         ::SetForegroundWindow(reinterpret_cast<HWND>(GetSourceId()));
+}
+
+HRESULT WgcWindowSource::CreateCaptureItem(
+    ComPtr<WGC::IGraphicsCaptureItem>* result) {
+  if (!ResolveCoreWinRTDelayload())
+    return E_FAIL;
+
+  ComPtr<IGraphicsCaptureItemInterop> interop;
+  HRESULT hr = GetActivationFactory<
+      IGraphicsCaptureItemInterop,
+      RuntimeClass_Windows_Graphics_Capture_GraphicsCaptureItem>(&interop);
+  if (FAILED(hr))
+    return hr;
+
+  ComPtr<WGC::IGraphicsCaptureItem> item;
+  hr = interop->CreateForWindow(reinterpret_cast<HWND>(GetSourceId()),
+                                IID_PPV_ARGS(&item));
+  if (FAILED(hr))
+    return hr;
+
+  if (!item)
+    return E_HANDLE;
+
+  *result = std::move(item);
+  return hr;
+}
+
+WgcScreenSource::WgcScreenSource(DesktopCapturer::SourceId source_id)
+    : WgcCaptureSource(source_id) {
+  // Getting the HMONITOR could fail if the source_id is invalid. In that case,
+  // we leave hmonitor_ uninitialized and `IsCapturable()` will fail.
+  HMONITOR hmon;
+  if (GetHmonitorFromDeviceIndex(GetSourceId(), &hmon))
+    hmonitor_ = hmon;
+}
+
+WgcScreenSource::~WgcScreenSource() = default;
+
+DesktopVector WgcScreenSource::GetTopLeft() {
+  if (!hmonitor_)
+    return DesktopVector();
+
+  return GetMonitorRect(*hmonitor_).top_left();
+}
+
+ABI::Windows::Graphics::SizeInt32 WgcScreenSource::GetSize() {
+  ABI::Windows::Graphics::SizeInt32 size = WgcCaptureSource::GetSize();
+  if (!hmonitor_ || (size.Width != 0 && size.Height != 0))
+    return size;
+
+  DesktopRect rect = GetMonitorRect(*hmonitor_);
+  return {rect.width(), rect.height()};
+}
+
+bool WgcScreenSource::IsCapturable() {
+  if (!hmonitor_)
+    return false;
+
+  if (!IsMonitorValid(*hmonitor_))
+    return false;
+
+  return WgcCaptureSource::IsCapturable();
+}
+
+HRESULT WgcScreenSource::CreateCaptureItem(
+    ComPtr<WGC::IGraphicsCaptureItem>* result) {
+  if (!hmonitor_)
+    return E_ABORT;
+
+  if (!ResolveCoreWinRTDelayload())
+    return E_FAIL;
+
+  ComPtr<IGraphicsCaptureItemInterop> interop;
+  HRESULT hr = GetActivationFactory<
+      IGraphicsCaptureItemInterop,
+      RuntimeClass_Windows_Graphics_Capture_GraphicsCaptureItem>(&interop);
+  if (FAILED(hr))
+    return hr;
+
+  // Ensure the monitor is still valid (hasn't disconnected) before trying to
+  // create the item. On versions of Windows before Win11, `CreateForMonitor`
+  // will crash if no displays are connected.
+  if (!IsMonitorValid(hmonitor_.value()))
+    return E_ABORT;
+
+  ComPtr<WGC::IGraphicsCaptureItem> item;
+  hr = interop->CreateForMonitor(*hmonitor_, IID_PPV_ARGS(&item));
+  if (FAILED(hr))
+    return hr;
+
+  if (!item)
+    return E_HANDLE;
+
+  *result = std::move(item);
+  return hr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_source.h b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_source.h
new file mode 100644
index 0000000000..d1275b6168
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_source.h
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_WGC_CAPTURE_SOURCE_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_WGC_CAPTURE_SOURCE_H_
+
+#include <windows.graphics.capture.h>
+#include <windows.graphics.h>
+#include <wrl/client.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+
+namespace webrtc {
+
+// Abstract class to represent the source that WGC-based capturers capture
+// from. Could represent an application window or a screen. Consumers should use
+// the appropriate Wgc*SourceFactory class to create WgcCaptureSource objects
+// of the appropriate type.
+class WgcCaptureSource {
+ public:
+  explicit WgcCaptureSource(DesktopCapturer::SourceId source_id);
+  virtual ~WgcCaptureSource();
+
+  virtual DesktopVector GetTopLeft() = 0;
+  virtual bool IsCapturable();
+  virtual bool FocusOnSource();
+  virtual ABI::Windows::Graphics::SizeInt32 GetSize();
+  HRESULT GetCaptureItem(
+      Microsoft::WRL::ComPtr<
+          ABI::Windows::Graphics::Capture::IGraphicsCaptureItem>* result);
+  DesktopCapturer::SourceId GetSourceId() { return source_id_; }
+
+ protected:
+  virtual HRESULT CreateCaptureItem(
+      Microsoft::WRL::ComPtr<
+          ABI::Windows::Graphics::Capture::IGraphicsCaptureItem>* result) = 0;
+
+ private:
+  Microsoft::WRL::ComPtr<ABI::Windows::Graphics::Capture::IGraphicsCaptureItem>
+      item_;
+  const DesktopCapturer::SourceId source_id_;
+};
+
+class WgcCaptureSourceFactory {
+ public:
+  virtual ~WgcCaptureSourceFactory();
+
+  virtual std::unique_ptr<WgcCaptureSource> CreateCaptureSource(
+      DesktopCapturer::SourceId) = 0;
+};
+
+class WgcWindowSourceFactory final : public WgcCaptureSourceFactory {
+ public:
+  WgcWindowSourceFactory();
+
+  // Disallow copy and assign.
+  WgcWindowSourceFactory(const WgcWindowSourceFactory&) = delete;
+  WgcWindowSourceFactory& operator=(const WgcWindowSourceFactory&) = delete;
+
+  ~WgcWindowSourceFactory() override;
+
+  std::unique_ptr<WgcCaptureSource> CreateCaptureSource(
+      DesktopCapturer::SourceId) override;
+};
+
+class WgcScreenSourceFactory final : public WgcCaptureSourceFactory {
+ public:
+  WgcScreenSourceFactory();
+
+  WgcScreenSourceFactory(const WgcScreenSourceFactory&) = delete;
+  WgcScreenSourceFactory& operator=(const WgcScreenSourceFactory&) = delete;
+
+  ~WgcScreenSourceFactory() override;
+
+  std::unique_ptr<WgcCaptureSource> CreateCaptureSource(
+      DesktopCapturer::SourceId) override;
+};
+
+// Class for capturing application windows.
+class WgcWindowSource final : public WgcCaptureSource {
+ public:
+  explicit WgcWindowSource(DesktopCapturer::SourceId source_id);
+
+  WgcWindowSource(const WgcWindowSource&) = delete;
+  WgcWindowSource& operator=(const WgcWindowSource&) = delete;
+
+  ~WgcWindowSource() override;
+
+  DesktopVector GetTopLeft() override;
+  ABI::Windows::Graphics::SizeInt32 GetSize() override;
+  bool IsCapturable() override;
+  bool FocusOnSource() override;
+
+ private:
+  HRESULT CreateCaptureItem(
+      Microsoft::WRL::ComPtr<
+          ABI::Windows::Graphics::Capture::IGraphicsCaptureItem>* result)
+      override;
+};
+
+// Class for capturing screens/monitors/displays.
+class WgcScreenSource final : public WgcCaptureSource {
+ public:
+  explicit WgcScreenSource(DesktopCapturer::SourceId source_id);
+
+  WgcScreenSource(const WgcScreenSource&) = delete;
+  WgcScreenSource& operator=(const WgcScreenSource&) = delete;
+
+  ~WgcScreenSource() override;
+
+  DesktopVector GetTopLeft() override;
+  ABI::Windows::Graphics::SizeInt32 GetSize() override;
+  bool IsCapturable() override;
+
+ private:
+  HRESULT CreateCaptureItem(
+      Microsoft::WRL::ComPtr<
+          ABI::Windows::Graphics::Capture::IGraphicsCaptureItem>* result)
+      override;
+
+  // To maintain compatibility with other capturers, this class accepts a
+  // device index as it's SourceId. However, WGC requires we use an HMONITOR to
+  // describe which screen to capture. So, we internally convert the supplied
+  // device index into an HMONITOR when `IsCapturable()` is called.
+  absl::optional<HMONITOR> hmonitor_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_WGC_CAPTURE_SOURCE_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_source_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_source_unittest.cc
new file mode 100644
index 0000000000..dc37ec2e0d
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capture_source_unittest.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/wgc_capture_source.h"
+
+#include <windows.graphics.capture.h>
+#include <wrl/client.h>
+
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "modules/desktop_capture/win/test_support/test_window.h"
+#include "modules/desktop_capture/win/wgc_capturer_win.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/win/scoped_com_initializer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+const WCHAR kWindowTitle[] = L"WGC Capture Source Test Window";
+
+const int kFirstXCoord = 25;
+const int kFirstYCoord = 50;
+const int kSecondXCoord = 50;
+const int kSecondYCoord = 75;
+
+}  // namespace
+
+class WgcCaptureSourceTest : public ::testing::TestWithParam<CaptureType> {
+ public:
+  void SetUp() override {
+    com_initializer_ =
+        std::make_unique<ScopedCOMInitializer>(ScopedCOMInitializer::kMTA);
+    ASSERT_TRUE(com_initializer_->Succeeded());
+  }
+
+  void TearDown() override {
+    if (window_open_) {
+      DestroyTestWindow(window_info_);
+    }
+  }
+
+  void SetUpForWindowSource() {
+    window_info_ = CreateTestWindow(kWindowTitle);
+    window_open_ = true;
+    source_id_ = reinterpret_cast<DesktopCapturer::SourceId>(window_info_.hwnd);
+    source_factory_ = std::make_unique<WgcWindowSourceFactory>();
+  }
+
+  void SetUpForScreenSource() {
+    source_id_ = kFullDesktopScreenId;
+    source_factory_ = std::make_unique<WgcScreenSourceFactory>();
+  }
+
+ protected:
+  std::unique_ptr<ScopedCOMInitializer> com_initializer_;
+  std::unique_ptr<WgcCaptureSourceFactory> source_factory_;
+  std::unique_ptr<WgcCaptureSource> source_;
+  DesktopCapturer::SourceId source_id_;
+  WindowInfo window_info_;
+  bool window_open_ = false;
+};
+
+// Window specific test
+TEST_F(WgcCaptureSourceTest, WindowPosition) {
+  if (!IsWgcSupported(CaptureType::kWindow)) {
+    RTC_LOG(LS_INFO)
+        << "Skipping WgcCapturerWinTests on unsupported platforms.";
+    GTEST_SKIP();
+  }
+
+  SetUpForWindowSource();
+  source_ = source_factory_->CreateCaptureSource(source_id_);
+  ASSERT_TRUE(source_);
+  EXPECT_EQ(source_->GetSourceId(), source_id_);
+
+  MoveTestWindow(window_info_.hwnd, kFirstXCoord, kFirstYCoord);
+  DesktopVector source_vector = source_->GetTopLeft();
+  EXPECT_EQ(source_vector.x(), kFirstXCoord);
+  EXPECT_EQ(source_vector.y(), kFirstYCoord);
+
+  MoveTestWindow(window_info_.hwnd, kSecondXCoord, kSecondYCoord);
+  source_vector = source_->GetTopLeft();
+  EXPECT_EQ(source_vector.x(), kSecondXCoord);
+  EXPECT_EQ(source_vector.y(), kSecondYCoord);
+}
+
+// Screen specific test
+TEST_F(WgcCaptureSourceTest, ScreenPosition) {
+  if (!IsWgcSupported(CaptureType::kScreen)) {
+    RTC_LOG(LS_INFO)
+        << "Skipping WgcCapturerWinTests on unsupported platforms.";
+    GTEST_SKIP();
+  }
+
+  SetUpForScreenSource();
+  source_ = source_factory_->CreateCaptureSource(source_id_);
+  ASSERT_TRUE(source_);
+  EXPECT_EQ(source_id_, source_->GetSourceId());
+
+  DesktopRect screen_rect = GetFullscreenRect();
+  DesktopVector source_vector = source_->GetTopLeft();
+  EXPECT_EQ(source_vector.x(), screen_rect.left());
+  EXPECT_EQ(source_vector.y(), screen_rect.top());
+}
+
+// Source agnostic test
+TEST_P(WgcCaptureSourceTest, CreateSource) {
+  if (!IsWgcSupported(GetParam())) {
+    RTC_LOG(LS_INFO)
+        << "Skipping WgcCapturerWinTests on unsupported platforms.";
+    GTEST_SKIP();
+  }
+
+  if (GetParam() == CaptureType::kWindow) {
+    SetUpForWindowSource();
+  } else {
+    SetUpForScreenSource();
+  }
+
+  source_ = source_factory_->CreateCaptureSource(source_id_);
+  ASSERT_TRUE(source_);
+  EXPECT_EQ(source_id_, source_->GetSourceId());
+  EXPECT_TRUE(source_->IsCapturable());
+
+  Microsoft::WRL::ComPtr<ABI::Windows::Graphics::Capture::IGraphicsCaptureItem>
+      item;
+  EXPECT_TRUE(SUCCEEDED(source_->GetCaptureItem(&item)));
+  EXPECT_TRUE(item);
+}
+
+INSTANTIATE_TEST_SUITE_P(SourceAgnostic,
+                         WgcCaptureSourceTest,
+                         ::testing::Values(CaptureType::kWindow,
+                                           CaptureType::kScreen));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_capturer_win.cc b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capturer_win.cc
new file mode 100644
index 0000000000..8ec6a29f23
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capturer_win.cc
@@ -0,0 +1,365 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/wgc_capturer_win.h"
+
+#include <DispatcherQueue.h>
+#include <windows.foundation.metadata.h>
+#include <windows.graphics.capture.h>
+
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capture_metrics_helper.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/win/wgc_desktop_frame.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/win/get_activation_factory.h"
+#include "rtc_base/win/hstring.h"
+#include "rtc_base/win/windows_version.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace WGC = ABI::Windows::Graphics::Capture;
+using Microsoft::WRL::ComPtr;
+
+namespace webrtc {
+
+namespace {
+
+constexpr wchar_t kCoreMessagingDll[] = L"CoreMessaging.dll";
+
+constexpr wchar_t kWgcSessionType[] =
+    L"Windows.Graphics.Capture.GraphicsCaptureSession";
+constexpr wchar_t kApiContract[] = L"Windows.Foundation.UniversalApiContract";
+constexpr UINT16 kRequiredApiContractVersion = 8;
+
+enum class WgcCapturerResult {
+  kSuccess = 0,
+  kNoDirect3dDevice = 1,
+  kNoSourceSelected = 2,
+  kItemCreationFailure = 3,
+  kSessionStartFailure = 4,
+  kGetFrameFailure = 5,
+  kFrameDropped = 6,
+  kCreateDispatcherQueueFailure = 7,
+  kMaxValue = kCreateDispatcherQueueFailure
+};
+
+void RecordWgcCapturerResult(WgcCapturerResult error) {
+  RTC_HISTOGRAM_ENUMERATION("WebRTC.DesktopCapture.Win.WgcCapturerResult",
+                            static_cast<int>(error),
+                            static_cast<int>(WgcCapturerResult::kMaxValue));
+}
+
+}  // namespace
+
+bool IsWgcSupported(CaptureType capture_type) {
+  if (!HasActiveDisplay()) {
+    // There is a bug in `CreateForMonitor` that causes a crash if there are no
+    // active displays. The crash was fixed in Win11, but we are still unable
+    // to capture screens without an active display.
+    if (capture_type == CaptureType::kScreen)
+      return false;
+
+    // There is a bug in the DWM (Desktop Window Manager) that prevents it from
+    // providing image data if there are no displays attached. This was fixed in
+    // Windows 11.
+    if (rtc::rtc_win::GetVersion() < rtc::rtc_win::Version::VERSION_WIN11)
+      return false;
+  }
+
+  // A bug in the WGC API `CreateForMonitor` prevents capturing the entire
+  // virtual screen (all monitors simultaneously), this was fixed in 20H1. Since
+  // we can't assert that we won't be asked to capture the entire virtual
+  // screen, we report unsupported so we can fallback to another capturer.
+  if (capture_type == CaptureType::kScreen &&
+      rtc::rtc_win::GetVersion() < rtc::rtc_win::Version::VERSION_WIN10_20H1) {
+    return false;
+  }
+
+  if (!ResolveCoreWinRTDelayload())
+    return false;
+
+  // We need to check if the WGC APIs are presesnt on the system. Certain SKUs
+  // of Windows ship without these APIs.
+  ComPtr<ABI::Windows::Foundation::Metadata::IApiInformationStatics>
+      api_info_statics;
+  HRESULT hr = GetActivationFactory<
+      ABI::Windows::Foundation::Metadata::IApiInformationStatics,
+      RuntimeClass_Windows_Foundation_Metadata_ApiInformation>(
+      &api_info_statics);
+  if (FAILED(hr))
+    return false;
+
+  HSTRING api_contract;
+  hr = webrtc::CreateHstring(kApiContract, wcslen(kApiContract), &api_contract);
+  if (FAILED(hr))
+    return false;
+
+  boolean is_api_present;
+  hr = api_info_statics->IsApiContractPresentByMajor(
+      api_contract, kRequiredApiContractVersion, &is_api_present);
+  webrtc::DeleteHstring(api_contract);
+  if (FAILED(hr) || !is_api_present)
+    return false;
+
+  HSTRING wgc_session_type;
+  hr = webrtc::CreateHstring(kWgcSessionType, wcslen(kWgcSessionType),
+                             &wgc_session_type);
+  if (FAILED(hr))
+    return false;
+
+  boolean is_type_present;
+  hr = api_info_statics->IsTypePresent(wgc_session_type, &is_type_present);
+  webrtc::DeleteHstring(wgc_session_type);
+  if (FAILED(hr) || !is_type_present)
+    return false;
+
+  // If the APIs are present, we need to check that they are supported.
+  ComPtr<WGC::IGraphicsCaptureSessionStatics> capture_session_statics;
+  hr = GetActivationFactory<
+      WGC::IGraphicsCaptureSessionStatics,
+      RuntimeClass_Windows_Graphics_Capture_GraphicsCaptureSession>(
+      &capture_session_statics);
+  if (FAILED(hr))
+    return false;
+
+  boolean is_supported;
+  hr = capture_session_statics->IsSupported(&is_supported);
+  if (FAILED(hr) || !is_supported)
+    return false;
+
+  return true;
+}
+
+WgcCapturerWin::WgcCapturerWin(
+    const DesktopCaptureOptions& options,
+    std::unique_ptr<WgcCaptureSourceFactory> source_factory,
+    std::unique_ptr<SourceEnumerator> source_enumerator,
+    bool allow_delayed_capturable_check)
+    : options_(options),
+      source_factory_(std::move(source_factory)),
+      source_enumerator_(std::move(source_enumerator)),
+      allow_delayed_capturable_check_(allow_delayed_capturable_check) {
+  if (!core_messaging_library_)
+    core_messaging_library_ = LoadLibraryW(kCoreMessagingDll);
+
+  if (core_messaging_library_) {
+    create_dispatcher_queue_controller_func_ =
+        reinterpret_cast<CreateDispatcherQueueControllerFunc>(GetProcAddress(
+            core_messaging_library_, "CreateDispatcherQueueController"));
+  }
+}
+
+WgcCapturerWin::~WgcCapturerWin() {
+  if (core_messaging_library_)
+    FreeLibrary(core_messaging_library_);
+}
+
+// static
+std::unique_ptr<DesktopCapturer> WgcCapturerWin::CreateRawWindowCapturer(
+    const DesktopCaptureOptions& options,
+    bool allow_delayed_capturable_check) {
+  return std::make_unique<WgcCapturerWin>(
+      options, std::make_unique<WgcWindowSourceFactory>(),
+      std::make_unique<WindowEnumerator>(
+          options.enumerate_current_process_windows()),
+      allow_delayed_capturable_check);
+}
+
+// static
+std::unique_ptr<DesktopCapturer> WgcCapturerWin::CreateRawScreenCapturer(
+    const DesktopCaptureOptions& options) {
+  return std::make_unique<WgcCapturerWin>(
+      options, std::make_unique<WgcScreenSourceFactory>(),
+      std::make_unique<ScreenEnumerator>(), false);
+}
+
+bool WgcCapturerWin::GetSourceList(SourceList* sources) {
+  return source_enumerator_->FindAllSources(sources);
+}
+
+bool WgcCapturerWin::SelectSource(DesktopCapturer::SourceId id) {
+  capture_source_ = source_factory_->CreateCaptureSource(id);
+  if (allow_delayed_capturable_check_)
+    return true;
+
+  return capture_source_->IsCapturable();
+}
+
+bool WgcCapturerWin::FocusOnSelectedSource() {
+  if (!capture_source_)
+    return false;
+
+  return capture_source_->FocusOnSource();
+}
+
+void WgcCapturerWin::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+  RecordCapturerImpl(DesktopCapturerId::kWgcCapturerWin);
+
+  callback_ = callback;
+
+  // Create a Direct3D11 device to share amongst the WgcCaptureSessions. Many
+  // parameters are nullptr as the implemention uses defaults that work well for
+  // us.
+  HRESULT hr = D3D11CreateDevice(
+      /*adapter=*/nullptr, D3D_DRIVER_TYPE_HARDWARE,
+      /*software_rasterizer=*/nullptr, D3D11_CREATE_DEVICE_BGRA_SUPPORT,
+      /*feature_levels=*/nullptr, /*feature_levels_size=*/0, D3D11_SDK_VERSION,
+      &d3d11_device_, /*feature_level=*/nullptr, /*device_context=*/nullptr);
+  if (hr == DXGI_ERROR_UNSUPPORTED) {
+    // If a hardware device could not be created, use WARP which is a high speed
+    // software device.
+    hr = D3D11CreateDevice(
+        /*adapter=*/nullptr, D3D_DRIVER_TYPE_WARP,
+        /*software_rasterizer=*/nullptr, D3D11_CREATE_DEVICE_BGRA_SUPPORT,
+        /*feature_levels=*/nullptr, /*feature_levels_size=*/0,
+        D3D11_SDK_VERSION, &d3d11_device_, /*feature_level=*/nullptr,
+        /*device_context=*/nullptr);
+  }
+
+  if (FAILED(hr)) {
+    RTC_LOG(LS_ERROR) << "Failed to create D3D11Device: " << hr;
+  }
+}
+
+void WgcCapturerWin::CaptureFrame() {
+  RTC_DCHECK(callback_);
+
+  if (!capture_source_) {
+    RTC_LOG(LS_ERROR) << "Source hasn't been selected";
+    callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_PERMANENT,
+                               /*frame=*/nullptr);
+    RecordWgcCapturerResult(WgcCapturerResult::kNoSourceSelected);
+    return;
+  }
+
+  if (!d3d11_device_) {
+    RTC_LOG(LS_ERROR) << "No D3D11D3evice, cannot capture.";
+    callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_PERMANENT,
+                               /*frame=*/nullptr);
+    RecordWgcCapturerResult(WgcCapturerResult::kNoDirect3dDevice);
+    return;
+  }
+
+  if (allow_delayed_capturable_check_ && !capture_source_->IsCapturable()) {
+    RTC_LOG(LS_ERROR) << "Source is not capturable.";
+    callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_PERMANENT,
+                               /*frame=*/nullptr);
+    return;
+  }
+
+  HRESULT hr;
+  if (!dispatcher_queue_created_) {
+    // Set the apartment type to NONE because this thread should already be COM
+    // initialized.
+    DispatcherQueueOptions options{
+        sizeof(DispatcherQueueOptions),
+        DISPATCHERQUEUE_THREAD_TYPE::DQTYPE_THREAD_CURRENT,
+        DISPATCHERQUEUE_THREAD_APARTMENTTYPE::DQTAT_COM_NONE};
+    ComPtr<ABI::Windows::System::IDispatcherQueueController> queue_controller;
+    hr = create_dispatcher_queue_controller_func_(options, &queue_controller);
+
+    // If there is already a DispatcherQueue on this thread, that is fine. Its
+    // lifetime is tied to the thread's, and as long as the thread has one, even
+    // if we didn't create it, the capture session's events will be delivered on
+    // this thread.
+    if (FAILED(hr) && hr != RPC_E_WRONG_THREAD) {
+      RecordWgcCapturerResult(WgcCapturerResult::kCreateDispatcherQueueFailure);
+      callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_PERMANENT,
+                                 /*frame=*/nullptr);
+    } else {
+      dispatcher_queue_created_ = true;
+    }
+  }
+
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+  WgcCaptureSession* capture_session = nullptr;
+  std::map<SourceId, WgcCaptureSession>::iterator session_iter =
+      ongoing_captures_.find(capture_source_->GetSourceId());
+  if (session_iter == ongoing_captures_.end()) {
+    ComPtr<WGC::IGraphicsCaptureItem> item;
+    hr = capture_source_->GetCaptureItem(&item);
+    if (FAILED(hr)) {
+      RTC_LOG(LS_ERROR) << "Failed to create a GraphicsCaptureItem: " << hr;
+      callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_PERMANENT,
+                                 /*frame=*/nullptr);
+      RecordWgcCapturerResult(WgcCapturerResult::kItemCreationFailure);
+      return;
+    }
+
+    std::pair<std::map<SourceId, WgcCaptureSession>::iterator, bool>
+        iter_success_pair = ongoing_captures_.emplace(
+            std::piecewise_construct,
+            std::forward_as_tuple(capture_source_->GetSourceId()),
+            std::forward_as_tuple(d3d11_device_, item,
+                                  capture_source_->GetSize()));
+    RTC_DCHECK(iter_success_pair.second);
+    capture_session = &iter_success_pair.first->second;
+  } else {
+    capture_session = &session_iter->second;
+  }
+
+  if (!capture_session->IsCaptureStarted()) {
+    hr = capture_session->StartCapture(options_);
+    if (FAILED(hr)) {
+      RTC_LOG(LS_ERROR) << "Failed to start capture: " << hr;
+      ongoing_captures_.erase(capture_source_->GetSourceId());
+      callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_PERMANENT,
+                                 /*frame=*/nullptr);
+      RecordWgcCapturerResult(WgcCapturerResult::kSessionStartFailure);
+      return;
+    }
+  }
+
+  std::unique_ptr<DesktopFrame> frame;
+  hr = capture_session->GetFrame(&frame);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_ERROR) << "GetFrame failed: " << hr;
+    ongoing_captures_.erase(capture_source_->GetSourceId());
+    callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_PERMANENT,
+                               /*frame=*/nullptr);
+    RecordWgcCapturerResult(WgcCapturerResult::kGetFrameFailure);
+    return;
+  }
+
+  if (!frame) {
+    callback_->OnCaptureResult(DesktopCapturer::Result::ERROR_TEMPORARY,
+                               /*frame=*/nullptr);
+    RecordWgcCapturerResult(WgcCapturerResult::kFrameDropped);
+    return;
+  }
+
+  int capture_time_ms = (rtc::TimeNanos() - capture_start_time_nanos) /
+                        rtc::kNumNanosecsPerMillisec;
+  RTC_HISTOGRAM_COUNTS_1000("WebRTC.DesktopCapture.Win.WgcCapturerFrameTime",
+                            capture_time_ms);
+  frame->set_capture_time_ms(capture_time_ms);
+  frame->set_capturer_id(DesktopCapturerId::kWgcCapturerWin);
+  frame->set_may_contain_cursor(options_.prefer_cursor_embedded());
+  frame->set_top_left(capture_source_->GetTopLeft());
+  RecordWgcCapturerResult(WgcCapturerResult::kSuccess);
+  callback_->OnCaptureResult(DesktopCapturer::Result::SUCCESS,
+                             std::move(frame));
+}
+
+bool WgcCapturerWin::IsSourceBeingCaptured(DesktopCapturer::SourceId id) {
+  std::map<DesktopCapturer::SourceId, WgcCaptureSession>::iterator
+      session_iter = ongoing_captures_.find(id);
+  if (session_iter == ongoing_captures_.end())
+    return false;
+
+  return session_iter->second.IsCaptureStarted();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_capturer_win.h b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capturer_win.h
new file mode 100644
index 0000000000..30253d9db6
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capturer_win.h
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_WGC_CAPTURER_WIN_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_WGC_CAPTURER_WIN_H_
+
+#include <DispatcherQueue.h>
+#include <d3d11.h>
+#include <wrl/client.h>
+
+#include <map>
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "modules/desktop_capture/win/wgc_capture_session.h"
+#include "modules/desktop_capture/win/wgc_capture_source.h"
+#include "modules/desktop_capture/win/window_capture_utils.h"
+
+namespace webrtc {
+
+// Checks if the WGC API is present and supported on the system.
+bool IsWgcSupported(CaptureType capture_type);
+
+// WgcCapturerWin is initialized with an implementation of this base class,
+// which it uses to find capturable sources of a particular type. This way,
+// WgcCapturerWin can remain source-agnostic.
+class SourceEnumerator {
+ public:
+  virtual ~SourceEnumerator() = default;
+
+  virtual bool FindAllSources(DesktopCapturer::SourceList* sources) = 0;
+};
+
+class WindowEnumerator final : public SourceEnumerator {
+ public:
+  explicit WindowEnumerator(bool enumerate_current_process_windows)
+      : enumerate_current_process_windows_(enumerate_current_process_windows) {}
+
+  WindowEnumerator(const WindowEnumerator&) = delete;
+  WindowEnumerator& operator=(const WindowEnumerator&) = delete;
+
+  ~WindowEnumerator() override = default;
+
+  bool FindAllSources(DesktopCapturer::SourceList* sources) override {
+    // WGC fails to capture windows with the WS_EX_TOOLWINDOW style, so we
+    // provide it as a filter to ensure windows with the style are not returned.
+    return window_capture_helper_.EnumerateCapturableWindows(
+        sources, enumerate_current_process_windows_, WS_EX_TOOLWINDOW);
+  }
+
+ private:
+  WindowCaptureHelperWin window_capture_helper_;
+  bool enumerate_current_process_windows_;
+};
+
+class ScreenEnumerator final : public SourceEnumerator {
+ public:
+  ScreenEnumerator() = default;
+
+  ScreenEnumerator(const ScreenEnumerator&) = delete;
+  ScreenEnumerator& operator=(const ScreenEnumerator&) = delete;
+
+  ~ScreenEnumerator() override = default;
+
+  bool FindAllSources(DesktopCapturer::SourceList* sources) override {
+    return webrtc::GetScreenList(sources);
+  }
+};
+
+// A capturer that uses the Window.Graphics.Capture APIs. It is suitable for
+// both window and screen capture (but only one type per instance). Consumers
+// should not instantiate this class directly, instead they should use
+// `CreateRawWindowCapturer()` or `CreateRawScreenCapturer()` to receive a
+// capturer appropriate for the type of source they want to capture.
+class WgcCapturerWin : public DesktopCapturer {
+ public:
+  WgcCapturerWin(const DesktopCaptureOptions& options,
+                 std::unique_ptr<WgcCaptureSourceFactory> source_factory,
+                 std::unique_ptr<SourceEnumerator> source_enumerator,
+                 bool allow_delayed_capturable_check);
+
+  WgcCapturerWin(const WgcCapturerWin&) = delete;
+  WgcCapturerWin& operator=(const WgcCapturerWin&) = delete;
+
+  ~WgcCapturerWin() override;
+
+  static std::unique_ptr<DesktopCapturer> CreateRawWindowCapturer(
+      const DesktopCaptureOptions& options,
+      bool allow_delayed_capturable_check = false);
+
+  static std::unique_ptr<DesktopCapturer> CreateRawScreenCapturer(
+      const DesktopCaptureOptions& options);
+
+  // DesktopCapturer interface.
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  void Start(Callback* callback) override;
+  void CaptureFrame() override;
+
+  // Used in WgcCapturerTests.
+  bool IsSourceBeingCaptured(SourceId id);
+
+ private:
+  typedef HRESULT(WINAPI* CreateDispatcherQueueControllerFunc)(
+      DispatcherQueueOptions,
+      ABI::Windows::System::IDispatcherQueueController**);
+
+  DesktopCaptureOptions options_;
+
+  // We need to either create or ensure that someone else created a
+  // `DispatcherQueue` on the current thread so that events will be delivered
+  // on the current thread rather than an arbitrary thread. A
+  // `DispatcherQueue`'s lifetime is tied to the thread's, and we don't post
+  // any work to it, so we don't need to hold a reference.
+  bool dispatcher_queue_created_ = false;
+
+  // Statically linking to CoreMessaging.lib is disallowed in Chromium, so we
+  // load it at runtime.
+  HMODULE core_messaging_library_ = NULL;
+  CreateDispatcherQueueControllerFunc create_dispatcher_queue_controller_func_ =
+      nullptr;
+
+  // Factory to create a WgcCaptureSource for us whenever SelectSource is
+  // called. Initialized at construction with a source-specific implementation.
+  std::unique_ptr<WgcCaptureSourceFactory> source_factory_;
+
+  // The source enumerator helps us find capturable sources of the appropriate
+  // type. Initialized at construction with a source-specific implementation.
+  std::unique_ptr<SourceEnumerator> source_enumerator_;
+
+  // The WgcCaptureSource represents the source we are capturing. It tells us
+  // if the source is capturable and it creates the GraphicsCaptureItem for us.
+  std::unique_ptr<WgcCaptureSource> capture_source_;
+
+  // A map of all the sources we are capturing and the associated
+  // WgcCaptureSession. Frames for the current source (indicated via
+  // SelectSource) will be retrieved from the appropriate session when
+  // requested via CaptureFrame.
+  // This helps us efficiently capture multiple sources (e.g. when consumers
+  // are trying to display a list of available capture targets with thumbnails).
+  std::map<SourceId, WgcCaptureSession> ongoing_captures_;
+
+  // The callback that we deliver frames to, synchronously, before CaptureFrame
+  // returns.
+  Callback* callback_ = nullptr;
+
+  // WgcCaptureSource::IsCapturable is expensive to run. So, caller can
+  // delay capturable check till capture frame is called if the WgcCapturerWin
+  // is used as a fallback capturer.
+  bool allow_delayed_capturable_check_ = false;
+
+  // A Direct3D11 device that is shared amongst the WgcCaptureSessions, who
+  // require one to perform the capture.
+  Microsoft::WRL::ComPtr<::ID3D11Device> d3d11_device_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_WGC_CAPTURER_WIN_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_capturer_win_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capturer_win_unittest.cc
new file mode 100644
index 0000000000..a7b656fcfc
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_capturer_win_unittest.cc
@@ -0,0 +1,572 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/wgc_capturer_win.h"
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/win/test_support/test_window.h"
+#include "modules/desktop_capture/win/wgc_capture_session.h"
+#include "modules/desktop_capture/win/window_capture_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/win/scoped_com_initializer.h"
+#include "rtc_base/win/windows_version.h"
+#include "system_wrappers/include/metrics.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr char kWindowThreadName[] = "wgc_capturer_test_window_thread";
+constexpr WCHAR kWindowTitle[] = L"WGC Capturer Test Window";
+
+constexpr char kCapturerImplHistogram[] =
+    "WebRTC.DesktopCapture.Win.DesktopCapturerImpl";
+
+constexpr char kCapturerResultHistogram[] =
+    "WebRTC.DesktopCapture.Win.WgcCapturerResult";
+constexpr int kSuccess = 0;
+constexpr int kSessionStartFailure = 4;
+
+constexpr char kCaptureSessionResultHistogram[] =
+    "WebRTC.DesktopCapture.Win.WgcCaptureSessionStartResult";
+constexpr int kSourceClosed = 1;
+
+constexpr char kCaptureTimeHistogram[] =
+    "WebRTC.DesktopCapture.Win.WgcCapturerFrameTime";
+
+// The capturer keeps `kNumBuffers` in its frame pool, so we need to request
+// that many frames to clear those out. The next frame will have the new size
+// (if the size has changed) so we will resize the frame pool at this point.
+// Then, we need to clear any frames that may have delivered to the frame pool
+// before the resize. Finally, the next frame will be guaranteed to be the new
+// size.
+constexpr int kNumCapturesToFlushBuffers =
+    WgcCaptureSession::kNumBuffers * 2 + 1;
+
+constexpr int kSmallWindowWidth = 200;
+constexpr int kSmallWindowHeight = 100;
+constexpr int kMediumWindowWidth = 300;
+constexpr int kMediumWindowHeight = 200;
+constexpr int kLargeWindowWidth = 400;
+constexpr int kLargeWindowHeight = 500;
+
+// The size of the image we capture is slightly smaller than the actual size of
+// the window.
+constexpr int kWindowWidthSubtrahend = 14;
+constexpr int kWindowHeightSubtrahend = 7;
+
+// Custom message constants so we can direct our thread to close windows and
+// quit running.
+constexpr UINT kDestroyWindow = WM_APP;
+constexpr UINT kQuitRunning = WM_APP + 1;
+
+// When testing changes to real windows, sometimes the effects (close or resize)
+// don't happen immediately, we want to keep trying until we see the effect but
+// only for a reasonable amount of time.
+constexpr int kMaxTries = 50;
+
+}  // namespace
+
+class WgcCapturerWinTest : public ::testing::TestWithParam<CaptureType>,
+                           public DesktopCapturer::Callback {
+ public:
+  void SetUp() override {
+    com_initializer_ =
+        std::make_unique<ScopedCOMInitializer>(ScopedCOMInitializer::kMTA);
+    EXPECT_TRUE(com_initializer_->Succeeded());
+
+    if (!IsWgcSupported(GetParam())) {
+      RTC_LOG(LS_INFO)
+          << "Skipping WgcCapturerWinTests on unsupported platforms.";
+      GTEST_SKIP();
+    }
+  }
+
+  void SetUpForWindowCapture(int window_width = kMediumWindowWidth,
+                             int window_height = kMediumWindowHeight) {
+    capturer_ = WgcCapturerWin::CreateRawWindowCapturer(
+        DesktopCaptureOptions::CreateDefault());
+    CreateWindowOnSeparateThread(window_width, window_height);
+    StartWindowThreadMessageLoop();
+    source_id_ = GetTestWindowIdFromSourceList();
+  }
+
+  void SetUpForScreenCapture() {
+    capturer_ = WgcCapturerWin::CreateRawScreenCapturer(
+        DesktopCaptureOptions::CreateDefault());
+    source_id_ = GetScreenIdFromSourceList();
+  }
+
+  void TearDown() override {
+    if (window_open_) {
+      CloseTestWindow();
+    }
+  }
+
+  // The window must live on a separate thread so that we can run a message pump
+  // without blocking the test thread. This is necessary if we are interested in
+  // having GraphicsCaptureItem events (i.e. the Closed event) fire, and it more
+  // closely resembles how capture works in the wild.
+  void CreateWindowOnSeparateThread(int window_width, int window_height) {
+    window_thread_ = rtc::Thread::Create();
+    window_thread_->SetName(kWindowThreadName, nullptr);
+    window_thread_->Start();
+    SendTask(window_thread_.get(), [this, window_width, window_height]() {
+      window_thread_id_ = GetCurrentThreadId();
+      window_info_ =
+          CreateTestWindow(kWindowTitle, window_height, window_width);
+      window_open_ = true;
+
+      while (!IsWindowResponding(window_info_.hwnd)) {
+        RTC_LOG(LS_INFO) << "Waiting for test window to become responsive in "
+                            "WgcWindowCaptureTest.";
+      }
+
+      while (!IsWindowValidAndVisible(window_info_.hwnd)) {
+        RTC_LOG(LS_INFO) << "Waiting for test window to be visible in "
+                            "WgcWindowCaptureTest.";
+      }
+    });
+
+    ASSERT_TRUE(window_thread_->RunningForTest());
+    ASSERT_FALSE(window_thread_->IsCurrent());
+  }
+
+  void StartWindowThreadMessageLoop() {
+    window_thread_->PostTask([this]() {
+      MSG msg;
+      BOOL gm;
+      while ((gm = ::GetMessage(&msg, NULL, 0, 0)) != 0 && gm != -1) {
+        ::DispatchMessage(&msg);
+        if (msg.message == kDestroyWindow) {
+          DestroyTestWindow(window_info_);
+        }
+        if (msg.message == kQuitRunning) {
+          PostQuitMessage(0);
+        }
+      }
+    });
+  }
+
+  void CloseTestWindow() {
+    ::PostThreadMessage(window_thread_id_, kDestroyWindow, 0, 0);
+    ::PostThreadMessage(window_thread_id_, kQuitRunning, 0, 0);
+    window_thread_->Stop();
+    window_open_ = false;
+  }
+
+  DesktopCapturer::SourceId GetTestWindowIdFromSourceList() {
+    // Frequently, the test window will not show up in GetSourceList because it
+    // was created too recently. Since we are confident the window will be found
+    // eventually we loop here until we find it.
+    intptr_t src_id = 0;
+    do {
+      DesktopCapturer::SourceList sources;
+      EXPECT_TRUE(capturer_->GetSourceList(&sources));
+      auto it = std::find_if(
+          sources.begin(), sources.end(),
+          [&](const DesktopCapturer::Source& src) {
+            return src.id == reinterpret_cast<intptr_t>(window_info_.hwnd);
+          });
+
+      if (it != sources.end())
+        src_id = it->id;
+    } while (src_id != reinterpret_cast<intptr_t>(window_info_.hwnd));
+
+    return src_id;
+  }
+
+  DesktopCapturer::SourceId GetScreenIdFromSourceList() {
+    DesktopCapturer::SourceList sources;
+    EXPECT_TRUE(capturer_->GetSourceList(&sources));
+    EXPECT_GT(sources.size(), 0ULL);
+    return sources[0].id;
+  }
+
+  void DoCapture(int num_captures = 1) {
+    // Capture the requested number of frames. We expect the first capture to
+    // always succeed. If we're asked for multiple frames, we do expect to see a
+    // a couple dropped frames due to resizing the window.
+    const int max_tries = num_captures == 1 ? 1 : kMaxTries;
+    int success_count = 0;
+    for (int i = 0; success_count < num_captures && i < max_tries; i++) {
+      capturer_->CaptureFrame();
+      if (result_ == DesktopCapturer::Result::ERROR_PERMANENT)
+        break;
+      if (result_ == DesktopCapturer::Result::SUCCESS)
+        success_count++;
+    }
+
+    total_successful_captures_ += success_count;
+    EXPECT_EQ(success_count, num_captures);
+    EXPECT_EQ(result_, DesktopCapturer::Result::SUCCESS);
+    EXPECT_TRUE(frame_);
+    EXPECT_GE(metrics::NumEvents(kCapturerResultHistogram, kSuccess),
+              total_successful_captures_);
+  }
+
+  void ValidateFrame(int expected_width, int expected_height) {
+    EXPECT_EQ(frame_->size().width(), expected_width - kWindowWidthSubtrahend);
+    EXPECT_EQ(frame_->size().height(),
+              expected_height - kWindowHeightSubtrahend);
+
+    // Verify the buffer contains as much data as it should.
+    int data_length = frame_->stride() * frame_->size().height();
+
+    // The first and last pixel should have the same color because they will be
+    // from the border of the window.
+    // Pixels have 4 bytes of data so the whole pixel needs a uint32_t to fit.
+    uint32_t first_pixel = static_cast<uint32_t>(*frame_->data());
+    uint32_t last_pixel = static_cast<uint32_t>(
+        *(frame_->data() + data_length - DesktopFrame::kBytesPerPixel));
+    EXPECT_EQ(first_pixel, last_pixel);
+
+    // Let's also check a pixel from the middle of the content area, which the
+    // test window will paint a consistent color for us to verify.
+    uint8_t* middle_pixel = frame_->data() + (data_length / 2);
+
+    int sub_pixel_offset = DesktopFrame::kBytesPerPixel / 4;
+    EXPECT_EQ(*middle_pixel, kTestWindowBValue);
+    middle_pixel += sub_pixel_offset;
+    EXPECT_EQ(*middle_pixel, kTestWindowGValue);
+    middle_pixel += sub_pixel_offset;
+    EXPECT_EQ(*middle_pixel, kTestWindowRValue);
+    middle_pixel += sub_pixel_offset;
+
+    // The window is opaque so we expect 0xFF for the Alpha channel.
+    EXPECT_EQ(*middle_pixel, 0xFF);
+  }
+
+  // DesktopCapturer::Callback interface
+  // The capturer synchronously invokes this method before `CaptureFrame()`
+  // returns.
+  void OnCaptureResult(DesktopCapturer::Result result,
+                       std::unique_ptr<DesktopFrame> frame) override {
+    result_ = result;
+    frame_ = std::move(frame);
+  }
+
+ protected:
+  std::unique_ptr<ScopedCOMInitializer> com_initializer_;
+  DWORD window_thread_id_;
+  std::unique_ptr<rtc::Thread> window_thread_;
+  WindowInfo window_info_;
+  intptr_t source_id_;
+  bool window_open_ = false;
+  DesktopCapturer::Result result_;
+  int total_successful_captures_ = 0;
+  std::unique_ptr<DesktopFrame> frame_;
+  std::unique_ptr<DesktopCapturer> capturer_;
+};
+
+TEST_P(WgcCapturerWinTest, SelectValidSource) {
+  if (GetParam() == CaptureType::kWindow) {
+    SetUpForWindowCapture();
+  } else {
+    SetUpForScreenCapture();
+  }
+
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+}
+
+TEST_P(WgcCapturerWinTest, SelectInvalidSource) {
+  if (GetParam() == CaptureType::kWindow) {
+    capturer_ = WgcCapturerWin::CreateRawWindowCapturer(
+        DesktopCaptureOptions::CreateDefault());
+    source_id_ = kNullWindowId;
+  } else {
+    capturer_ = WgcCapturerWin::CreateRawScreenCapturer(
+        DesktopCaptureOptions::CreateDefault());
+    source_id_ = kInvalidScreenId;
+  }
+
+  EXPECT_FALSE(capturer_->SelectSource(source_id_));
+}
+
+TEST_P(WgcCapturerWinTest, Capture) {
+  if (GetParam() == CaptureType::kWindow) {
+    SetUpForWindowCapture();
+  } else {
+    SetUpForScreenCapture();
+  }
+
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+
+  capturer_->Start(this);
+  EXPECT_GE(metrics::NumEvents(kCapturerImplHistogram,
+                               DesktopCapturerId::kWgcCapturerWin),
+            1);
+
+  DoCapture();
+  EXPECT_GT(frame_->size().width(), 0);
+  EXPECT_GT(frame_->size().height(), 0);
+}
+
+TEST_P(WgcCapturerWinTest, CaptureTime) {
+  if (GetParam() == CaptureType::kWindow) {
+    SetUpForWindowCapture();
+  } else {
+    SetUpForScreenCapture();
+  }
+
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+  capturer_->Start(this);
+
+  int64_t start_time;
+  start_time = rtc::TimeNanos();
+  capturer_->CaptureFrame();
+
+  int capture_time_ms =
+      (rtc::TimeNanos() - start_time) / rtc::kNumNanosecsPerMillisec;
+  EXPECT_EQ(result_, DesktopCapturer::Result::SUCCESS);
+  EXPECT_TRUE(frame_);
+
+  // The test may measure the time slightly differently than the capturer. So we
+  // just check if it's within 5 ms.
+  EXPECT_NEAR(frame_->capture_time_ms(), capture_time_ms, 5);
+  EXPECT_GE(
+      metrics::NumEvents(kCaptureTimeHistogram, frame_->capture_time_ms()), 1);
+}
+
+INSTANTIATE_TEST_SUITE_P(SourceAgnostic,
+                         WgcCapturerWinTest,
+                         ::testing::Values(CaptureType::kWindow,
+                                           CaptureType::kScreen));
+
+TEST(WgcCapturerNoMonitorTest, NoMonitors) {
+  ScopedCOMInitializer com_initializer(ScopedCOMInitializer::kMTA);
+  EXPECT_TRUE(com_initializer.Succeeded());
+  if (HasActiveDisplay()) {
+    RTC_LOG(LS_INFO) << "Skip WgcCapturerWinTest designed specifically for "
+                        "systems with no monitors";
+    GTEST_SKIP();
+  }
+
+  // A bug in `CreateForMonitor` prevents screen capture when no displays are
+  // attached.
+  EXPECT_FALSE(IsWgcSupported(CaptureType::kScreen));
+
+  // A bug in the DWM (Desktop Window Manager) prevents it from providing image
+  // data if there are no displays attached. This was fixed in Windows 11.
+  if (rtc::rtc_win::GetVersion() < rtc::rtc_win::Version::VERSION_WIN11)
+    EXPECT_FALSE(IsWgcSupported(CaptureType::kWindow));
+  else
+    EXPECT_TRUE(IsWgcSupported(CaptureType::kWindow));
+}
+
+class WgcCapturerMonitorTest : public WgcCapturerWinTest {
+ public:
+  void SetUp() {
+    com_initializer_ =
+        std::make_unique<ScopedCOMInitializer>(ScopedCOMInitializer::kMTA);
+    EXPECT_TRUE(com_initializer_->Succeeded());
+
+    if (!IsWgcSupported(CaptureType::kScreen)) {
+      RTC_LOG(LS_INFO)
+          << "Skipping WgcCapturerWinTests on unsupported platforms.";
+      GTEST_SKIP();
+    }
+  }
+};
+
+TEST_F(WgcCapturerMonitorTest, FocusOnMonitor) {
+  SetUpForScreenCapture();
+  EXPECT_TRUE(capturer_->SelectSource(0));
+
+  // You can't set focus on a monitor.
+  EXPECT_FALSE(capturer_->FocusOnSelectedSource());
+}
+
+TEST_F(WgcCapturerMonitorTest, CaptureAllMonitors) {
+  SetUpForScreenCapture();
+  EXPECT_TRUE(capturer_->SelectSource(kFullDesktopScreenId));
+
+  capturer_->Start(this);
+  DoCapture();
+  EXPECT_GT(frame_->size().width(), 0);
+  EXPECT_GT(frame_->size().height(), 0);
+}
+
+class WgcCapturerWindowTest : public WgcCapturerWinTest {
+ public:
+  void SetUp() {
+    com_initializer_ =
+        std::make_unique<ScopedCOMInitializer>(ScopedCOMInitializer::kMTA);
+    EXPECT_TRUE(com_initializer_->Succeeded());
+
+    if (!IsWgcSupported(CaptureType::kWindow)) {
+      RTC_LOG(LS_INFO)
+          << "Skipping WgcCapturerWinTests on unsupported platforms.";
+      GTEST_SKIP();
+    }
+  }
+};
+
+TEST_F(WgcCapturerWindowTest, FocusOnWindow) {
+  capturer_ = WgcCapturerWin::CreateRawWindowCapturer(
+      DesktopCaptureOptions::CreateDefault());
+  window_info_ = CreateTestWindow(kWindowTitle);
+  source_id_ = GetScreenIdFromSourceList();
+
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+  EXPECT_TRUE(capturer_->FocusOnSelectedSource());
+
+  HWND hwnd = reinterpret_cast<HWND>(source_id_);
+  EXPECT_EQ(hwnd, ::GetActiveWindow());
+  EXPECT_EQ(hwnd, ::GetForegroundWindow());
+  EXPECT_EQ(hwnd, ::GetFocus());
+  DestroyTestWindow(window_info_);
+}
+
+TEST_F(WgcCapturerWindowTest, SelectMinimizedWindow) {
+  SetUpForWindowCapture();
+  MinimizeTestWindow(reinterpret_cast<HWND>(source_id_));
+  EXPECT_FALSE(capturer_->SelectSource(source_id_));
+
+  UnminimizeTestWindow(reinterpret_cast<HWND>(source_id_));
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+}
+
+TEST_F(WgcCapturerWindowTest, SelectClosedWindow) {
+  SetUpForWindowCapture();
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+
+  CloseTestWindow();
+  EXPECT_FALSE(capturer_->SelectSource(source_id_));
+}
+
+TEST_F(WgcCapturerWindowTest, UnsupportedWindowStyle) {
+  // Create a window with the WS_EX_TOOLWINDOW style, which WGC does not
+  // support.
+  window_info_ = CreateTestWindow(kWindowTitle, kMediumWindowWidth,
+                                  kMediumWindowHeight, WS_EX_TOOLWINDOW);
+  capturer_ = WgcCapturerWin::CreateRawWindowCapturer(
+      DesktopCaptureOptions::CreateDefault());
+  DesktopCapturer::SourceList sources;
+  EXPECT_TRUE(capturer_->GetSourceList(&sources));
+  auto it = std::find_if(
+      sources.begin(), sources.end(), [&](const DesktopCapturer::Source& src) {
+        return src.id == reinterpret_cast<intptr_t>(window_info_.hwnd);
+      });
+
+  // We should not find the window, since we filter for unsupported styles.
+  EXPECT_EQ(it, sources.end());
+  DestroyTestWindow(window_info_);
+}
+
+TEST_F(WgcCapturerWindowTest, IncreaseWindowSizeMidCapture) {
+  SetUpForWindowCapture(kSmallWindowWidth, kSmallWindowHeight);
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+
+  capturer_->Start(this);
+  DoCapture();
+  ValidateFrame(kSmallWindowWidth, kSmallWindowHeight);
+
+  ResizeTestWindow(window_info_.hwnd, kSmallWindowWidth, kMediumWindowHeight);
+  DoCapture(kNumCapturesToFlushBuffers);
+  ValidateFrame(kSmallWindowWidth, kMediumWindowHeight);
+
+  ResizeTestWindow(window_info_.hwnd, kLargeWindowWidth, kMediumWindowHeight);
+  DoCapture(kNumCapturesToFlushBuffers);
+  ValidateFrame(kLargeWindowWidth, kMediumWindowHeight);
+}
+
+TEST_F(WgcCapturerWindowTest, ReduceWindowSizeMidCapture) {
+  SetUpForWindowCapture(kLargeWindowWidth, kLargeWindowHeight);
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+
+  capturer_->Start(this);
+  DoCapture();
+  ValidateFrame(kLargeWindowWidth, kLargeWindowHeight);
+
+  ResizeTestWindow(window_info_.hwnd, kLargeWindowWidth, kMediumWindowHeight);
+  DoCapture(kNumCapturesToFlushBuffers);
+  ValidateFrame(kLargeWindowWidth, kMediumWindowHeight);
+
+  ResizeTestWindow(window_info_.hwnd, kSmallWindowWidth, kMediumWindowHeight);
+  DoCapture(kNumCapturesToFlushBuffers);
+  ValidateFrame(kSmallWindowWidth, kMediumWindowHeight);
+}
+
+TEST_F(WgcCapturerWindowTest, MinimizeWindowMidCapture) {
+  SetUpForWindowCapture();
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+
+  capturer_->Start(this);
+
+  // Minmize the window and capture should continue but return temporary errors.
+  MinimizeTestWindow(window_info_.hwnd);
+  for (int i = 0; i < 5; ++i) {
+    capturer_->CaptureFrame();
+    EXPECT_EQ(result_, DesktopCapturer::Result::ERROR_TEMPORARY);
+  }
+
+  // Reopen the window and the capture should continue normally.
+  UnminimizeTestWindow(window_info_.hwnd);
+  DoCapture();
+  // We can't verify the window size here because the test window does not
+  // repaint itself after it is unminimized, but capturing successfully is still
+  // a good test.
+}
+
+TEST_F(WgcCapturerWindowTest, CloseWindowMidCapture) {
+  SetUpForWindowCapture();
+  EXPECT_TRUE(capturer_->SelectSource(source_id_));
+
+  capturer_->Start(this);
+  DoCapture();
+  ValidateFrame(kMediumWindowWidth, kMediumWindowHeight);
+
+  CloseTestWindow();
+
+  // We need to pump our message queue so the Closed event will be delivered to
+  // the capturer's event handler. If we are too early and the Closed event
+  // hasn't arrived yet we should keep trying until the capturer receives it and
+  // stops.
+  auto* wgc_capturer = static_cast<WgcCapturerWin*>(capturer_.get());
+  MSG msg;
+  for (int i = 0;
+       wgc_capturer->IsSourceBeingCaptured(source_id_) && i < kMaxTries; ++i) {
+    // Unlike GetMessage, PeekMessage will not hang if there are no messages in
+    // the queue.
+    PeekMessage(&msg, 0, 0, 0, PM_REMOVE);
+    SleepMs(1);
+  }
+
+  EXPECT_FALSE(wgc_capturer->IsSourceBeingCaptured(source_id_));
+
+  // The frame pool can buffer `kNumBuffers` frames. We must consume these
+  // and then make one more call to CaptureFrame before we expect to see the
+  // failure.
+  int num_tries = 0;
+  do {
+    capturer_->CaptureFrame();
+  } while (result_ == DesktopCapturer::Result::SUCCESS &&
+           ++num_tries <= WgcCaptureSession::kNumBuffers);
+
+  EXPECT_GE(metrics::NumEvents(kCapturerResultHistogram, kSessionStartFailure),
+            1);
+  EXPECT_GE(metrics::NumEvents(kCaptureSessionResultHistogram, kSourceClosed),
+            1);
+  EXPECT_EQ(result_, DesktopCapturer::Result::ERROR_PERMANENT);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_desktop_frame.cc b/third_party/libwebrtc/modules/desktop_capture/win/wgc_desktop_frame.cc
new file mode 100644
index 0000000000..dd9009120b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_desktop_frame.cc
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/wgc_desktop_frame.h"
+
+#include <utility>
+
+namespace webrtc {
+
+WgcDesktopFrame::WgcDesktopFrame(DesktopSize size,
+                                 int stride,
+                                 std::vector<uint8_t>&& image_data)
+    : DesktopFrame(size, stride, image_data.data(), nullptr),
+      image_data_(std::move(image_data)) {}
+
+WgcDesktopFrame::~WgcDesktopFrame() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/wgc_desktop_frame.h b/third_party/libwebrtc/modules/desktop_capture/win/wgc_desktop_frame.h
new file mode 100644
index 0000000000..0a671cf2f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/wgc_desktop_frame.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_WGC_DESKTOP_FRAME_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_WGC_DESKTOP_FRAME_H_
+
+#include <d3d11.h>
+#include <wrl/client.h>
+
+#include <memory>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+
+namespace webrtc {
+
+// DesktopFrame implementation used by capturers that use the
+// Windows.Graphics.Capture API.
+class WgcDesktopFrame final : public DesktopFrame {
+ public:
+  // WgcDesktopFrame receives an rvalue reference to the `image_data` vector
+  // so that it can take ownership of it (and avoid a copy).
+  WgcDesktopFrame(DesktopSize size,
+                  int stride,
+                  std::vector<uint8_t>&& image_data);
+
+  WgcDesktopFrame(const WgcDesktopFrame&) = delete;
+  WgcDesktopFrame& operator=(const WgcDesktopFrame&) = delete;
+
+  ~WgcDesktopFrame() override;
+
+ private:
+  std::vector<uint8_t> image_data_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_WGC_DESKTOP_FRAME_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils.cc b/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils.cc
new file mode 100644
index 0000000000..d58c02e17c
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils.cc
@@ -0,0 +1,486 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/window_capture_utils.h"
+
+// Just for the DWMWINDOWATTRIBUTE enums (DWMWA_CLOAKED).
+#include <dwmapi.h>
+
+#include <algorithm>
+
+#include "modules/desktop_capture/win/scoped_gdi_object.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/win/windows_version.h"
+
+namespace webrtc {
+
+namespace {
+
+struct GetWindowListParams {
+  GetWindowListParams(int flags,
+                      LONG ex_style_filters,
+                      DesktopCapturer::SourceList* result)
+      : ignore_untitled(flags & GetWindowListFlags::kIgnoreUntitled),
+        ignore_unresponsive(flags & GetWindowListFlags::kIgnoreUnresponsive),
+        ignore_current_process_windows(
+            flags & GetWindowListFlags::kIgnoreCurrentProcessWindows),
+        ex_style_filters(ex_style_filters),
+        result(result) {}
+  const bool ignore_untitled;
+  const bool ignore_unresponsive;
+  const bool ignore_current_process_windows;
+  const LONG ex_style_filters;
+  DesktopCapturer::SourceList* const result;
+};
+
+bool IsWindowOwnedByCurrentProcess(HWND hwnd) {
+  DWORD process_id;
+  GetWindowThreadProcessId(hwnd, &process_id);
+  return process_id == GetCurrentProcessId();
+}
+
+BOOL CALLBACK GetWindowListHandler(HWND hwnd, LPARAM param) {
+  GetWindowListParams* params = reinterpret_cast<GetWindowListParams*>(param);
+  DesktopCapturer::SourceList* list = params->result;
+
+  // Skip invisible and minimized windows
+  if (!IsWindowVisible(hwnd) || IsIconic(hwnd)) {
+    return TRUE;
+  }
+
+  // Skip windows which are not presented in the taskbar,
+  // namely owned window if they don't have the app window style set
+  HWND owner = GetWindow(hwnd, GW_OWNER);
+  LONG exstyle = GetWindowLong(hwnd, GWL_EXSTYLE);
+  if (owner && !(exstyle & WS_EX_APPWINDOW)) {
+    return TRUE;
+  }
+
+  // Filter out windows that match the extended styles the caller has specified,
+  // e.g. WS_EX_TOOLWINDOW for capturers that don't support overlay windows.
+  if (exstyle & params->ex_style_filters) {
+    return TRUE;
+  }
+
+  if (params->ignore_unresponsive && !IsWindowResponding(hwnd)) {
+    return TRUE;
+  }
+
+  DesktopCapturer::Source window;
+  window.id = reinterpret_cast<WindowId>(hwnd);
+
+  DWORD pid;
+  GetWindowThreadProcessId(hwnd, &pid);
+  window.pid = static_cast<pid_t>(pid);
+
+  // GetWindowText* are potentially blocking operations if `hwnd` is
+  // owned by the current process. The APIs will send messages to the window's
+  // message loop, and if the message loop is waiting on this operation we will
+  // enter a deadlock.
+  // https://docs.microsoft.com/en-us/windows/win32/api/winuser/nf-winuser-getwindowtexta#remarks
+  //
+  // To help consumers avoid this, there is a DesktopCaptureOption to ignore
+  // windows owned by the current process. Consumers should either ensure that
+  // the thread running their message loop never waits on this operation, or use
+  // the option to exclude these windows from the source list.
+  bool owned_by_current_process = IsWindowOwnedByCurrentProcess(hwnd);
+  if (owned_by_current_process && params->ignore_current_process_windows) {
+    return TRUE;
+  }
+
+  // Even if consumers request to enumerate windows owned by the current
+  // process, we should not call GetWindowText* on unresponsive windows owned by
+  // the current process because we will hang. Unfortunately, we could still
+  // hang if the window becomes unresponsive after this check, hence the option
+  // to avoid these completely.
+  if (!owned_by_current_process || IsWindowResponding(hwnd)) {
+    const size_t kTitleLength = 500;
+    WCHAR window_title[kTitleLength] = L"";
+    if (GetWindowTextLength(hwnd) != 0 &&
+        GetWindowTextW(hwnd, window_title, kTitleLength) > 0) {
+      window.title = rtc::ToUtf8(window_title);
+    }
+  }
+
+  // Skip windows when we failed to convert the title or it is empty.
+  if (params->ignore_untitled && window.title.empty())
+    return TRUE;
+
+  // Capture the window class name, to allow specific window classes to be
+  // skipped.
+  //
+  // https://docs.microsoft.com/en-us/windows/win32/api/winuser/ns-winuser-wndclassa
+  // says lpszClassName field in WNDCLASS is limited by 256 symbols, so we don't
+  // need to have a buffer bigger than that.
+  const size_t kMaxClassNameLength = 256;
+  WCHAR class_name[kMaxClassNameLength] = L"";
+  const int class_name_length =
+      GetClassNameW(hwnd, class_name, kMaxClassNameLength);
+  if (class_name_length < 1)
+    return TRUE;
+
+  // Skip Program Manager window.
+  if (wcscmp(class_name, L"Progman") == 0)
+    return TRUE;
+
+  // Skip Start button window on Windows Vista, Windows 7.
+  // On Windows 8, Windows 8.1, Windows 10 Start button is not a top level
+  // window, so it will not be examined here.
+  if (wcscmp(class_name, L"Button") == 0)
+    return TRUE;
+
+  list->push_back(window);
+
+  return TRUE;
+}
+
+}  // namespace
+
+// Prefix used to match the window class for Chrome windows.
+const wchar_t kChromeWindowClassPrefix[] = L"Chrome_WidgetWin_";
+
+// The hiddgen taskbar will leave a 2 pixel margin on the screen.
+const int kHiddenTaskbarMarginOnScreen = 2;
+
+bool GetWindowRect(HWND window, DesktopRect* result) {
+  RECT rect;
+  if (!::GetWindowRect(window, &rect)) {
+    return false;
+  }
+  *result = DesktopRect::MakeLTRB(rect.left, rect.top, rect.right, rect.bottom);
+  return true;
+}
+
+bool GetCroppedWindowRect(HWND window,
+                          bool avoid_cropping_border,
+                          DesktopRect* cropped_rect,
+                          DesktopRect* original_rect) {
+  DesktopRect window_rect;
+  if (!GetWindowRect(window, &window_rect)) {
+    return false;
+  }
+
+  if (original_rect) {
+    *original_rect = window_rect;
+  }
+  *cropped_rect = window_rect;
+
+  bool is_maximized = false;
+  if (!IsWindowMaximized(window, &is_maximized)) {
+    return false;
+  }
+
+  // As of Windows8, transparent resize borders are added by the OS at
+  // left/bottom/right sides of a resizeable window. If the cropped window
+  // doesn't remove these borders, the background will be exposed a bit.
+  if (rtc::rtc_win::GetVersion() >= rtc::rtc_win::Version::VERSION_WIN8 ||
+      is_maximized) {
+    // Only apply this cropping to windows with a resize border (otherwise,
+    // it'd clip the edges of captured pop-up windows without this border).
+    LONG style = GetWindowLong(window, GWL_STYLE);
+    if (style & WS_THICKFRAME || style & DS_MODALFRAME) {
+      int width = GetSystemMetrics(SM_CXSIZEFRAME);
+      int bottom_height = GetSystemMetrics(SM_CYSIZEFRAME);
+      const int visible_border_height = GetSystemMetrics(SM_CYBORDER);
+      int top_height = visible_border_height;
+
+      // If requested, avoid cropping the visible window border. This is used
+      // for pop-up windows to include their border, but not for the outermost
+      // window (where a partially-transparent border may expose the
+      // background a bit).
+      if (avoid_cropping_border) {
+        width = std::max(0, width - GetSystemMetrics(SM_CXBORDER));
+        bottom_height = std::max(0, bottom_height - visible_border_height);
+        top_height = 0;
+      }
+      cropped_rect->Extend(-width, -top_height, -width, -bottom_height);
+    }
+  }
+
+  return true;
+}
+
+bool GetWindowContentRect(HWND window, DesktopRect* result) {
+  if (!GetWindowRect(window, result)) {
+    return false;
+  }
+
+  RECT rect;
+  if (!::GetClientRect(window, &rect)) {
+    return false;
+  }
+
+  const int width = rect.right - rect.left;
+  // The GetClientRect() is not expected to return a larger area than
+  // GetWindowRect().
+  if (width > 0 && width < result->width()) {
+    // - GetClientRect() always set the left / top of RECT to 0. So we need to
+    //   estimate the border width from GetClientRect() and GetWindowRect().
+    // - Border width of a window varies according to the window type.
+    // - GetClientRect() excludes the title bar, which should be considered as
+    //   part of the content and included in the captured frame. So we always
+    //   estimate the border width according to the window width.
+    // - We assume a window has same border width in each side.
+    // So we shrink half of the width difference from all four sides.
+    const int shrink = ((width - result->width()) / 2);
+    // When `shrink` is negative, DesktopRect::Extend() shrinks itself.
+    result->Extend(shrink, 0, shrink, 0);
+    // Usually this should not happen, just in case we have received a strange
+    // window, which has only left and right borders.
+    if (result->height() > shrink * 2) {
+      result->Extend(0, shrink, 0, shrink);
+    }
+    RTC_DCHECK(!result->is_empty());
+  }
+
+  return true;
+}
+
+int GetWindowRegionTypeWithBoundary(HWND window, DesktopRect* result) {
+  win::ScopedGDIObject<HRGN, win::DeleteObjectTraits<HRGN>> scoped_hrgn(
+      CreateRectRgn(0, 0, 0, 0));
+  const int region_type = GetWindowRgn(window, scoped_hrgn.Get());
+
+  if (region_type == SIMPLEREGION) {
+    RECT rect;
+    GetRgnBox(scoped_hrgn.Get(), &rect);
+    *result =
+        DesktopRect::MakeLTRB(rect.left, rect.top, rect.right, rect.bottom);
+  }
+  return region_type;
+}
+
+bool GetDcSize(HDC hdc, DesktopSize* size) {
+  win::ScopedGDIObject<HGDIOBJ, win::DeleteObjectTraits<HGDIOBJ>> scoped_hgdi(
+      GetCurrentObject(hdc, OBJ_BITMAP));
+  BITMAP bitmap;
+  memset(&bitmap, 0, sizeof(BITMAP));
+  if (GetObject(scoped_hgdi.Get(), sizeof(BITMAP), &bitmap) == 0) {
+    return false;
+  }
+  size->set(bitmap.bmWidth, bitmap.bmHeight);
+  return true;
+}
+
+bool IsWindowMaximized(HWND window, bool* result) {
+  WINDOWPLACEMENT placement;
+  memset(&placement, 0, sizeof(WINDOWPLACEMENT));
+  placement.length = sizeof(WINDOWPLACEMENT);
+  if (!::GetWindowPlacement(window, &placement)) {
+    return false;
+  }
+
+  *result = (placement.showCmd == SW_SHOWMAXIMIZED);
+  return true;
+}
+
+bool IsWindowValidAndVisible(HWND window) {
+  return IsWindow(window) && IsWindowVisible(window) && !IsIconic(window);
+}
+
+bool IsWindowResponding(HWND window) {
+  // 50ms is chosen in case the system is under heavy load, but it's also not
+  // too long to delay window enumeration considerably.
+  const UINT uTimeoutMs = 50;
+  return SendMessageTimeout(window, WM_NULL, 0, 0, SMTO_ABORTIFHUNG, uTimeoutMs,
+                            nullptr);
+}
+
+bool GetWindowList(int flags,
+                   DesktopCapturer::SourceList* windows,
+                   LONG ex_style_filters) {
+  GetWindowListParams params(flags, ex_style_filters, windows);
+  return ::EnumWindows(&GetWindowListHandler,
+                       reinterpret_cast<LPARAM>(&params)) != 0;
+}
+
+// WindowCaptureHelperWin implementation.
+WindowCaptureHelperWin::WindowCaptureHelperWin() {
+  // Try to load dwmapi.dll dynamically since it is not available on XP.
+  dwmapi_library_ = LoadLibraryW(L"dwmapi.dll");
+  if (dwmapi_library_) {
+    func_ = reinterpret_cast<DwmIsCompositionEnabledFunc>(
+        GetProcAddress(dwmapi_library_, "DwmIsCompositionEnabled"));
+    dwm_get_window_attribute_func_ =
+        reinterpret_cast<DwmGetWindowAttributeFunc>(
+            GetProcAddress(dwmapi_library_, "DwmGetWindowAttribute"));
+  }
+
+  if (rtc::rtc_win::GetVersion() >= rtc::rtc_win::Version::VERSION_WIN10) {
+    if (FAILED(::CoCreateInstance(__uuidof(VirtualDesktopManager), nullptr,
+                                  CLSCTX_ALL,
+                                  IID_PPV_ARGS(&virtual_desktop_manager_)))) {
+      RTC_LOG(LS_WARNING) << "Fail to create instance of VirtualDesktopManager";
+    }
+  }
+}
+
+WindowCaptureHelperWin::~WindowCaptureHelperWin() {
+  if (dwmapi_library_) {
+    FreeLibrary(dwmapi_library_);
+  }
+}
+
+bool WindowCaptureHelperWin::IsAeroEnabled() {
+  BOOL result = FALSE;
+  if (func_) {
+    func_(&result);
+  }
+  return result != FALSE;
+}
+
+// This is just a best guess of a notification window. Chrome uses the Windows
+// native framework for showing notifications. So far what we know about such a
+// window includes: no title, class name with prefix "Chrome_WidgetWin_" and
+// with certain extended styles.
+bool WindowCaptureHelperWin::IsWindowChromeNotification(HWND hwnd) {
+  const size_t kTitleLength = 32;
+  WCHAR window_title[kTitleLength];
+  GetWindowTextW(hwnd, window_title, kTitleLength);
+  if (wcsnlen_s(window_title, kTitleLength) != 0) {
+    return false;
+  }
+
+  const size_t kClassLength = 256;
+  WCHAR class_name[kClassLength];
+  const int class_name_length = GetClassNameW(hwnd, class_name, kClassLength);
+  if (class_name_length < 1 ||
+      wcsncmp(class_name, kChromeWindowClassPrefix,
+              wcsnlen_s(kChromeWindowClassPrefix, kClassLength)) != 0) {
+    return false;
+  }
+
+  const LONG exstyle = GetWindowLong(hwnd, GWL_EXSTYLE);
+  if ((exstyle & WS_EX_NOACTIVATE) && (exstyle & WS_EX_TOOLWINDOW) &&
+      (exstyle & WS_EX_TOPMOST)) {
+    return true;
+  }
+
+  return false;
+}
+
+// `content_rect` is preferred because,
+// 1. WindowCapturerWinGdi is using GDI capturer, which cannot capture DX
+// output.
+//    So ScreenCapturer should be used as much as possible to avoid
+//    uncapturable cases. Note: lots of new applications are using DX output
+//    (hardware acceleration) to improve the performance which cannot be
+//    captured by WindowCapturerWinGdi. See bug http://crbug.com/741770.
+// 2. WindowCapturerWinGdi is still useful because we do not want to expose the
+//    content on other windows if the target window is covered by them.
+// 3. Shadow and borders should not be considered as "content" on other
+//    windows because they do not expose any useful information.
+//
+// So we can bear the false-negative cases (target window is covered by the
+// borders or shadow of other windows, but we have not detected it) in favor
+// of using ScreenCapturer, rather than let the false-positive cases (target
+// windows is only covered by borders or shadow of other windows, but we treat
+// it as overlapping) impact the user experience.
+bool WindowCaptureHelperWin::AreWindowsOverlapping(
+    HWND hwnd,
+    HWND selected_hwnd,
+    const DesktopRect& selected_window_rect) {
+  DesktopRect content_rect;
+  if (!GetWindowContentRect(hwnd, &content_rect)) {
+    // Bail out if failed to get the window area.
+    return true;
+  }
+  content_rect.IntersectWith(selected_window_rect);
+
+  if (content_rect.is_empty()) {
+    return false;
+  }
+
+  // When the taskbar is automatically hidden, it will leave a 2 pixel margin on
+  // the screen which will overlap the maximized selected window that will use
+  // up the full screen area. Since there is no solid way to identify a hidden
+  // taskbar window, we have to make an exemption here if the overlapping is
+  // 2 x screen_width/height to a maximized window.
+  bool is_maximized = false;
+  IsWindowMaximized(selected_hwnd, &is_maximized);
+  bool overlaps_hidden_horizontal_taskbar =
+      selected_window_rect.width() == content_rect.width() &&
+      content_rect.height() == kHiddenTaskbarMarginOnScreen;
+  bool overlaps_hidden_vertical_taskbar =
+      selected_window_rect.height() == content_rect.height() &&
+      content_rect.width() == kHiddenTaskbarMarginOnScreen;
+  if (is_maximized && (overlaps_hidden_horizontal_taskbar ||
+                       overlaps_hidden_vertical_taskbar)) {
+    return false;
+  }
+
+  return true;
+}
+
+bool WindowCaptureHelperWin::IsWindowOnCurrentDesktop(HWND hwnd) {
+  // Make sure the window is on the current virtual desktop.
+  if (virtual_desktop_manager_) {
+    BOOL on_current_desktop;
+    if (SUCCEEDED(virtual_desktop_manager_->IsWindowOnCurrentVirtualDesktop(
+            hwnd, &on_current_desktop)) &&
+        !on_current_desktop) {
+      return false;
+    }
+  }
+  return true;
+}
+
+bool WindowCaptureHelperWin::IsWindowVisibleOnCurrentDesktop(HWND hwnd) {
+  return IsWindowValidAndVisible(hwnd) && IsWindowOnCurrentDesktop(hwnd) &&
+         !IsWindowCloaked(hwnd);
+}
+
+// A cloaked window is composited but not visible to the user.
+// Example: Cortana or the Action Center when collapsed.
+bool WindowCaptureHelperWin::IsWindowCloaked(HWND hwnd) {
+  if (!dwm_get_window_attribute_func_) {
+    // Does not apply.
+    return false;
+  }
+
+  int res = 0;
+  if (dwm_get_window_attribute_func_(hwnd, DWMWA_CLOAKED, &res, sizeof(res)) !=
+      S_OK) {
+    // Cannot tell so assume not cloaked for backward compatibility.
+    return false;
+  }
+
+  return res != 0;
+}
+
+bool WindowCaptureHelperWin::EnumerateCapturableWindows(
+    DesktopCapturer::SourceList* results,
+    bool enumerate_current_process_windows,
+    LONG ex_style_filters) {
+  int flags = (GetWindowListFlags::kIgnoreUntitled |
+               GetWindowListFlags::kIgnoreUnresponsive);
+  if (!enumerate_current_process_windows) {
+    flags |= GetWindowListFlags::kIgnoreCurrentProcessWindows;
+  }
+
+  if (!webrtc::GetWindowList(flags, results, ex_style_filters)) {
+    return false;
+  }
+
+  for (auto it = results->begin(); it != results->end();) {
+    if (!IsWindowVisibleOnCurrentDesktop(reinterpret_cast<HWND>(it->id))) {
+      it = results->erase(it);
+    } else {
+      ++it;
+    }
+  }
+
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils.h b/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils.h
new file mode 100644
index 0000000000..caea07958d
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils.h
@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_WINDOW_CAPTURE_UTILS_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_WINDOW_CAPTURE_UTILS_H_
+
+#include <shlobj.h>
+#include <windows.h>
+#include <wrl/client.h>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+
+namespace webrtc {
+
+// Outputs the window rect. The returned DesktopRect is in system coordinates,
+// i.e. the primary monitor on the system always starts from (0, 0). This
+// function returns false if native APIs fail.
+bool GetWindowRect(HWND window, DesktopRect* result);
+
+// Outputs the window rect, with the left/right/bottom frame border cropped if
+// the window is maximized or has a transparent resize border.
+// `avoid_cropping_border` may be set to true to avoid cropping the visible
+// border when cropping any resize border.
+// `cropped_rect` is the cropped rect relative to the
+// desktop. `original_rect` is the original rect returned from GetWindowRect.
+// Returns true if all API calls succeeded. The returned DesktopRect is in
+// system coordinates, i.e. the primary monitor on the system always starts from
+// (0, 0). `original_rect` can be nullptr.
+//
+// TODO(zijiehe): Move this function to CroppingWindowCapturerWin after it has
+// been removed from MouseCursorMonitorWin.
+// This function should only be used by CroppingWindowCapturerWin. Instead a
+// DesktopRect CropWindowRect(const DesktopRect& rect)
+// should be added as a utility function to help CroppingWindowCapturerWin and
+// WindowCapturerWinGdi to crop out the borders or shadow according to their
+// scenarios. But this function is too generic and easy to be misused.
+bool GetCroppedWindowRect(HWND window,
+                          bool avoid_cropping_border,
+                          DesktopRect* cropped_rect,
+                          DesktopRect* original_rect);
+
+// Retrieves the rectangle of the content area of `window`. Usually it contains
+// title bar and window client area, but borders or shadow are excluded. The
+// returned DesktopRect is in system coordinates, i.e. the primary monitor on
+// the system always starts from (0, 0). This function returns false if native
+// APIs fail.
+bool GetWindowContentRect(HWND window, DesktopRect* result);
+
+// Returns the region type of the `window` and fill `rect` with the region of
+// `window` if region type is SIMPLEREGION.
+int GetWindowRegionTypeWithBoundary(HWND window, DesktopRect* result);
+
+// Retrieves the size of the `hdc`. This function returns false if native APIs
+// fail.
+bool GetDcSize(HDC hdc, DesktopSize* size);
+
+// Retrieves whether the `window` is maximized and stores in `result`. This
+// function returns false if native APIs fail.
+bool IsWindowMaximized(HWND window, bool* result);
+
+// Checks that the HWND is for a valid window, that window's visibility state is
+// visible, and that it is not minimized.
+bool IsWindowValidAndVisible(HWND window);
+
+// Checks if a window responds to a message within 50ms.
+bool IsWindowResponding(HWND window);
+
+enum GetWindowListFlags {
+  kNone = 0x00,
+  kIgnoreUntitled = 1 << 0,
+  kIgnoreUnresponsive = 1 << 1,
+  kIgnoreCurrentProcessWindows = 1 << 2,
+};
+
+// Retrieves the list of top-level windows on the screen.
+// Some windows will be ignored:
+// - Those that are invisible or minimized.
+// - Program Manager & Start menu.
+// - [with kIgnoreUntitled] windows with no title.
+// - [with kIgnoreUnresponsive] windows that are unresponsive.
+// - [with kIgnoreCurrentProcessWindows] windows owned by the current process.
+// - Any windows with extended styles that match `ex_style_filters`.
+// Returns false if native APIs failed.
+bool GetWindowList(int flags,
+                   DesktopCapturer::SourceList* windows,
+                   LONG ex_style_filters = 0);
+
+typedef HRESULT(WINAPI* DwmIsCompositionEnabledFunc)(BOOL* enabled);
+typedef HRESULT(WINAPI* DwmGetWindowAttributeFunc)(HWND hwnd,
+                                                   DWORD flag,
+                                                   PVOID result_ptr,
+                                                   DWORD result_size);
+class WindowCaptureHelperWin {
+ public:
+  WindowCaptureHelperWin();
+  ~WindowCaptureHelperWin();
+
+  WindowCaptureHelperWin(const WindowCaptureHelperWin&) = delete;
+  WindowCaptureHelperWin& operator=(const WindowCaptureHelperWin&) = delete;
+
+  bool IsAeroEnabled();
+  bool IsWindowChromeNotification(HWND hwnd);
+  bool AreWindowsOverlapping(HWND hwnd,
+                             HWND selected_hwnd,
+                             const DesktopRect& selected_window_rect);
+  bool IsWindowOnCurrentDesktop(HWND hwnd);
+  bool IsWindowVisibleOnCurrentDesktop(HWND hwnd);
+  bool IsWindowCloaked(HWND hwnd);
+
+  // The optional `ex_style_filters` parameter allows callers to provide
+  // extended window styles (e.g. WS_EX_TOOLWINDOW) and prevent windows that
+  // match from being included in `results`.
+  bool EnumerateCapturableWindows(DesktopCapturer::SourceList* results,
+                                  bool enumerate_current_process_windows,
+                                  LONG ex_style_filters = 0);
+
+ private:
+  HMODULE dwmapi_library_ = nullptr;
+  DwmIsCompositionEnabledFunc func_ = nullptr;
+  DwmGetWindowAttributeFunc dwm_get_window_attribute_func_ = nullptr;
+
+  // Only used on Win10+.
+  Microsoft::WRL::ComPtr<IVirtualDesktopManager> virtual_desktop_manager_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_WINDOW_CAPTURE_UTILS_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils_unittest.cc
new file mode 100644
index 0000000000..137440b09e
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/window_capture_utils_unittest.cc
@@ -0,0 +1,153 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/window_capture_utils.h"
+
+#include <winuser.h>
+
+#include <algorithm>
+#include <memory>
+#include <mutex>
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/win/test_support/test_window.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/thread.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+const char kWindowThreadName[] = "window_capture_utils_test_thread";
+const WCHAR kWindowTitle[] = L"Window Capture Utils Test";
+
+std::unique_ptr<rtc::Thread> SetUpUnresponsiveWindow(std::mutex& mtx,
+                                                     WindowInfo& info) {
+  std::unique_ptr<rtc::Thread> window_thread;
+  window_thread = rtc::Thread::Create();
+  window_thread->SetName(kWindowThreadName, nullptr);
+  window_thread->Start();
+
+  SendTask(window_thread.get(), [&] { info = CreateTestWindow(kWindowTitle); });
+
+  // Intentionally create a deadlock to cause the window to become unresponsive.
+  mtx.lock();
+  window_thread->PostTask([&mtx]() {
+    mtx.lock();
+    mtx.unlock();
+  });
+
+  return window_thread;
+}
+
+}  // namespace
+
+TEST(WindowCaptureUtilsTest, GetWindowList) {
+  WindowInfo info = CreateTestWindow(kWindowTitle);
+  DesktopCapturer::SourceList window_list;
+  ASSERT_TRUE(GetWindowList(GetWindowListFlags::kNone, &window_list));
+  EXPECT_GT(window_list.size(), 0ULL);
+  EXPECT_NE(std::find_if(window_list.begin(), window_list.end(),
+                         [&info](DesktopCapturer::Source window) {
+                           return reinterpret_cast<HWND>(window.id) ==
+                                  info.hwnd;
+                         }),
+            window_list.end());
+  DestroyTestWindow(info);
+}
+
+TEST(WindowCaptureUtilsTest, IncludeUnresponsiveWindows) {
+  std::mutex mtx;
+  WindowInfo info;
+  std::unique_ptr<rtc::Thread> window_thread =
+      SetUpUnresponsiveWindow(mtx, info);
+
+  EXPECT_FALSE(IsWindowResponding(info.hwnd));
+
+  DesktopCapturer::SourceList window_list;
+  ASSERT_TRUE(GetWindowList(GetWindowListFlags::kNone, &window_list));
+  EXPECT_GT(window_list.size(), 0ULL);
+  EXPECT_NE(std::find_if(window_list.begin(), window_list.end(),
+                         [&info](DesktopCapturer::Source window) {
+                           return reinterpret_cast<HWND>(window.id) ==
+                                  info.hwnd;
+                         }),
+            window_list.end());
+
+  mtx.unlock();
+  SendTask(window_thread.get(), [&info]() { DestroyTestWindow(info); });
+  window_thread->Stop();
+}
+
+TEST(WindowCaptureUtilsTest, IgnoreUnresponsiveWindows) {
+  std::mutex mtx;
+  WindowInfo info;
+  std::unique_ptr<rtc::Thread> window_thread =
+      SetUpUnresponsiveWindow(mtx, info);
+
+  EXPECT_FALSE(IsWindowResponding(info.hwnd));
+
+  DesktopCapturer::SourceList window_list;
+  ASSERT_TRUE(
+      GetWindowList(GetWindowListFlags::kIgnoreUnresponsive, &window_list));
+  EXPECT_EQ(std::find_if(window_list.begin(), window_list.end(),
+                         [&info](DesktopCapturer::Source window) {
+                           return reinterpret_cast<HWND>(window.id) ==
+                                  info.hwnd;
+                         }),
+            window_list.end());
+
+  mtx.unlock();
+  SendTask(window_thread.get(), [&info]() { DestroyTestWindow(info); });
+  window_thread->Stop();
+}
+
+TEST(WindowCaptureUtilsTest, IncludeUntitledWindows) {
+  WindowInfo info = CreateTestWindow(L"");
+  DesktopCapturer::SourceList window_list;
+  ASSERT_TRUE(GetWindowList(GetWindowListFlags::kNone, &window_list));
+  EXPECT_GT(window_list.size(), 0ULL);
+  EXPECT_NE(std::find_if(window_list.begin(), window_list.end(),
+                         [&info](DesktopCapturer::Source window) {
+                           return reinterpret_cast<HWND>(window.id) ==
+                                  info.hwnd;
+                         }),
+            window_list.end());
+  DestroyTestWindow(info);
+}
+
+TEST(WindowCaptureUtilsTest, IgnoreUntitledWindows) {
+  WindowInfo info = CreateTestWindow(L"");
+  DesktopCapturer::SourceList window_list;
+  ASSERT_TRUE(GetWindowList(GetWindowListFlags::kIgnoreUntitled, &window_list));
+  EXPECT_EQ(std::find_if(window_list.begin(), window_list.end(),
+                         [&info](DesktopCapturer::Source window) {
+                           return reinterpret_cast<HWND>(window.id) ==
+                                  info.hwnd;
+                         }),
+            window_list.end());
+  DestroyTestWindow(info);
+}
+
+TEST(WindowCaptureUtilsTest, IgnoreCurrentProcessWindows) {
+  WindowInfo info = CreateTestWindow(kWindowTitle);
+  DesktopCapturer::SourceList window_list;
+  ASSERT_TRUE(GetWindowList(GetWindowListFlags::kIgnoreCurrentProcessWindows,
+                            &window_list));
+  EXPECT_EQ(std::find_if(window_list.begin(), window_list.end(),
+                         [&info](DesktopCapturer::Source window) {
+                           return reinterpret_cast<HWND>(window.id) ==
+                                  info.hwnd;
+                         }),
+            window_list.end());
+  DestroyTestWindow(info);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/window_capturer_win_gdi.cc b/third_party/libwebrtc/modules/desktop_capture/win/window_capturer_win_gdi.cc
new file mode 100644
index 0000000000..bc3a762264
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/window_capturer_win_gdi.cc
@@ -0,0 +1,403 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/win/window_capturer_win_gdi.h"
+
+#include <cmath>
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "modules/desktop_capture/cropped_desktop_frame.h"
+#include "modules/desktop_capture/desktop_capture_metrics_helper.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame_win.h"
+#include "modules/desktop_capture/win/screen_capture_utils.h"
+#include "modules/desktop_capture/win/selected_window_context.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "rtc_base/win/windows_version.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+// Used to pass input/output data during the EnumWindows call to collect
+// owned/pop-up windows that should be captured.
+struct OwnedWindowCollectorContext : public SelectedWindowContext {
+  OwnedWindowCollectorContext(HWND selected_window,
+                              DesktopRect selected_window_rect,
+                              WindowCaptureHelperWin* window_capture_helper,
+                              std::vector<HWND>* owned_windows)
+      : SelectedWindowContext(selected_window,
+                              selected_window_rect,
+                              window_capture_helper),
+        owned_windows(owned_windows) {}
+
+  std::vector<HWND>* owned_windows;
+};
+
+// Called via EnumWindows for each root window; adds owned/pop-up windows that
+// should be captured to a vector it's passed.
+BOOL CALLBACK OwnedWindowCollector(HWND hwnd, LPARAM param) {
+  OwnedWindowCollectorContext* context =
+      reinterpret_cast<OwnedWindowCollectorContext*>(param);
+  if (hwnd == context->selected_window()) {
+    // Windows are enumerated in top-down z-order, so we can stop enumerating
+    // upon reaching the selected window.
+    return FALSE;
+  }
+
+  // Skip windows that aren't visible pop-up windows.
+  if (!(GetWindowLong(hwnd, GWL_STYLE) & WS_POPUP) ||
+      !context->window_capture_helper()->IsWindowVisibleOnCurrentDesktop(
+          hwnd)) {
+    return TRUE;
+  }
+
+  // Owned windows that intersect the selected window should be captured.
+  if (context->IsWindowOwnedBySelectedWindow(hwnd) &&
+      context->IsWindowOverlappingSelectedWindow(hwnd)) {
+    // Skip windows that draw shadows around menus. These "SysShadow" windows
+    // would otherwise be captured as solid black bars with no transparency
+    // gradient (since this capturer doesn't detect / respect variations in the
+    // window alpha channel). Any other semi-transparent owned windows will be
+    // captured fully-opaque. This seems preferable to excluding them (at least
+    // when they have content aside from a solid fill color / visual adornment;
+    // e.g. some tooltips have the transparent style set).
+    if (GetWindowLong(hwnd, GWL_EXSTYLE) & WS_EX_TRANSPARENT) {
+      const WCHAR kSysShadow[] = L"SysShadow";
+      const size_t kClassLength = arraysize(kSysShadow);
+      WCHAR class_name[kClassLength];
+      const int class_name_length =
+          GetClassNameW(hwnd, class_name, kClassLength);
+      if (class_name_length == kClassLength - 1 &&
+          wcscmp(class_name, kSysShadow) == 0) {
+        return TRUE;
+      }
+    }
+
+    context->owned_windows->push_back(hwnd);
+  }
+
+  return TRUE;
+}
+
+WindowCapturerWinGdi::WindowCapturerWinGdi(
+    bool enumerate_current_process_windows)
+    : enumerate_current_process_windows_(enumerate_current_process_windows) {}
+WindowCapturerWinGdi::~WindowCapturerWinGdi() {}
+
+bool WindowCapturerWinGdi::GetSourceList(SourceList* sources) {
+  if (!window_capture_helper_.EnumerateCapturableWindows(
+          sources, enumerate_current_process_windows_))
+    return false;
+
+  std::map<HWND, DesktopSize> new_map;
+  for (const auto& item : *sources) {
+    HWND hwnd = reinterpret_cast<HWND>(item.id);
+    new_map[hwnd] = window_size_map_[hwnd];
+  }
+  window_size_map_.swap(new_map);
+
+  return true;
+}
+
+bool WindowCapturerWinGdi::SelectSource(SourceId id) {
+  HWND window = reinterpret_cast<HWND>(id);
+  if (!IsWindowValidAndVisible(window))
+    return false;
+
+  window_ = window;
+  // When a window is not in the map, window_size_map_[window] will create an
+  // item with DesktopSize (0, 0).
+  previous_size_ = window_size_map_[window];
+  return true;
+}
+
+bool WindowCapturerWinGdi::FocusOnSelectedSource() {
+  if (!window_)
+    return false;
+
+  if (!IsWindowValidAndVisible(window_))
+    return false;
+
+  return BringWindowToTop(window_) && SetForegroundWindow(window_);
+}
+
+bool WindowCapturerWinGdi::IsOccluded(const DesktopVector& pos) {
+  DesktopVector sys_pos = pos.add(GetFullscreenRect().top_left());
+  HWND hwnd =
+      reinterpret_cast<HWND>(window_finder_.GetWindowUnderPoint(sys_pos));
+
+  return hwnd != window_ &&
+         std::find(owned_windows_.begin(), owned_windows_.end(), hwnd) ==
+             owned_windows_.end();
+}
+
+void WindowCapturerWinGdi::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+  RecordCapturerImpl(DesktopCapturerId::kWindowCapturerWinGdi);
+
+  callback_ = callback;
+}
+
+void WindowCapturerWinGdi::CaptureFrame() {
+  RTC_DCHECK(callback_);
+  int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+  CaptureResults results = CaptureFrame(/*capture_owned_windows*/ true);
+  if (!results.frame) {
+    // Don't return success if we have no frame.
+    results.result = results.result == Result::SUCCESS ? Result::ERROR_TEMPORARY
+                                                       : results.result;
+    callback_->OnCaptureResult(results.result, nullptr);
+    return;
+  }
+
+  int capture_time_ms = (rtc::TimeNanos() - capture_start_time_nanos) /
+                        rtc::kNumNanosecsPerMillisec;
+  RTC_HISTOGRAM_COUNTS_1000(
+      "WebRTC.DesktopCapture.Win.WindowGdiCapturerFrameTime", capture_time_ms);
+  results.frame->set_capture_time_ms(capture_time_ms);
+  results.frame->set_capturer_id(DesktopCapturerId::kWindowCapturerWinGdi);
+  callback_->OnCaptureResult(results.result, std::move(results.frame));
+}
+
+WindowCapturerWinGdi::CaptureResults WindowCapturerWinGdi::CaptureFrame(
+    bool capture_owned_windows) {
+  TRACE_EVENT0("webrtc", "WindowCapturerWinGdi::CaptureFrame");
+
+  if (!window_) {
+    RTC_LOG(LS_ERROR) << "Window hasn't been selected: " << GetLastError();
+    return {Result::ERROR_PERMANENT, nullptr};
+  }
+
+  // Stop capturing if the window has been closed.
+  if (!IsWindow(window_)) {
+    RTC_LOG(LS_ERROR) << "Target window has been closed.";
+    return {Result::ERROR_PERMANENT, nullptr};
+  }
+
+  // Determine the window region excluding any resize border, and including
+  // any visible border if capturing an owned window / dialog. (Don't include
+  // any visible border for the selected window for consistency with
+  // CroppingWindowCapturerWin, which would expose a bit of the background
+  // through the partially-transparent border.)
+  const bool avoid_cropping_border = !capture_owned_windows;
+  DesktopRect cropped_rect;
+  DesktopRect original_rect;
+
+  if (!GetCroppedWindowRect(window_, avoid_cropping_border, &cropped_rect,
+                            &original_rect)) {
+    RTC_LOG(LS_WARNING) << "Failed to get drawable window area: "
+                        << GetLastError();
+    return {Result::ERROR_TEMPORARY, nullptr};
+  }
+
+  // Return a 1x1 black frame if the window is minimized or invisible on current
+  // desktop, to match behavior on mace. Window can be temporarily invisible
+  // during the transition of full screen mode on/off.
+  if (original_rect.is_empty() ||
+      !window_capture_helper_.IsWindowVisibleOnCurrentDesktop(window_)) {
+    std::unique_ptr<DesktopFrame> frame(
+        new BasicDesktopFrame(DesktopSize(1, 1)));
+
+    previous_size_ = frame->size();
+    window_size_map_[window_] = previous_size_;
+    return {Result::SUCCESS, std::move(frame)};
+  }
+
+  HDC window_dc = GetWindowDC(window_);
+  if (!window_dc) {
+    RTC_LOG(LS_WARNING) << "Failed to get window DC: " << GetLastError();
+    return {Result::ERROR_TEMPORARY, nullptr};
+  }
+
+  DesktopRect unscaled_cropped_rect = cropped_rect;
+  double horizontal_scale = 1.0;
+  double vertical_scale = 1.0;
+
+  DesktopSize window_dc_size;
+  if (GetDcSize(window_dc, &window_dc_size)) {
+    // The `window_dc_size` is used to detect the scaling of the original
+    // window. If the application does not support high-DPI settings, it will
+    // be scaled by Windows according to the scaling setting.
+    // https://www.google.com/search?q=windows+scaling+settings&ie=UTF-8
+    // So the size of the `window_dc`, i.e. the bitmap we can retrieve from
+    // PrintWindow() or BitBlt() function, will be smaller than
+    // `original_rect` and `cropped_rect`. Part of the captured desktop frame
+    // will be black. See
+    // bug https://bugs.chromium.org/p/webrtc/issues/detail?id=8112 for
+    // details.
+
+    // If `window_dc_size` is smaller than `window_rect`, let's resize both
+    // `original_rect` and `cropped_rect` according to the scaling factor.
+    // This will adjust the width and height of the two rects.
+    horizontal_scale =
+        static_cast<double>(window_dc_size.width()) / original_rect.width();
+    vertical_scale =
+        static_cast<double>(window_dc_size.height()) / original_rect.height();
+    original_rect.Scale(horizontal_scale, vertical_scale);
+    cropped_rect.Scale(horizontal_scale, vertical_scale);
+
+    // Translate `cropped_rect` to the left so that its position within
+    // `original_rect` remains accurate after scaling.
+    // See crbug.com/1083527 for more info.
+    int translate_left = static_cast<int>(std::round(
+        (cropped_rect.left() - original_rect.left()) * (horizontal_scale - 1)));
+    int translate_top = static_cast<int>(std::round(
+        (cropped_rect.top() - original_rect.top()) * (vertical_scale - 1)));
+    cropped_rect.Translate(translate_left, translate_top);
+  }
+
+  std::unique_ptr<DesktopFrameWin> frame(
+      DesktopFrameWin::Create(original_rect.size(), nullptr, window_dc));
+  if (!frame.get()) {
+    RTC_LOG(LS_WARNING) << "Failed to create frame.";
+    ReleaseDC(window_, window_dc);
+    return {Result::ERROR_TEMPORARY, nullptr};
+  }
+
+  HDC mem_dc = CreateCompatibleDC(window_dc);
+  HGDIOBJ previous_object = SelectObject(mem_dc, frame->bitmap());
+  BOOL result = FALSE;
+
+  // When desktop composition (Aero) is enabled each window is rendered to a
+  // private buffer allowing BitBlt() to get the window content even if the
+  // window is occluded. PrintWindow() is slower but lets rendering the window
+  // contents to an off-screen device context when Aero is not available.
+  // PrintWindow() is not supported by some applications.
+  //
+  // If Aero is enabled, we prefer BitBlt() because it's faster and avoids
+  // window flickering. Otherwise, we prefer PrintWindow() because BitBlt() may
+  // render occluding windows on top of the desired window.
+  //
+  // When composition is enabled the DC returned by GetWindowDC() doesn't always
+  // have window frame rendered correctly. Windows renders it only once and then
+  // caches the result between captures. We hack it around by calling
+  // PrintWindow() whenever window size changes, including the first time of
+  // capturing - it somehow affects what we get from BitBlt() on the subsequent
+  // captures.
+  //
+  // For Windows 8.1 and later, we want to always use PrintWindow when the
+  // cropping screen capturer falls back to the window capturer. I.e.
+  // on Windows 8.1 and later, PrintWindow is only used when the window is
+  // occluded. When the window is not occluded, it is much faster to capture
+  // the screen and to crop it to the window position and size.
+  if (rtc::rtc_win::GetVersion() >= rtc::rtc_win::Version::VERSION_WIN8) {
+    // Special flag that makes PrintWindow to work on Windows 8.1 and later.
+    // Indeed certain apps (e.g. those using DirectComposition rendering) can't
+    // be captured using BitBlt or PrintWindow without this flag. Note that on
+    // Windows 8.0 this flag is not supported so the block below will fallback
+    // to the other call to PrintWindow. It seems to be very tricky to detect
+    // Windows 8.0 vs 8.1 so a try/fallback is more approriate here.
+    const UINT flags = PW_RENDERFULLCONTENT;
+    result = PrintWindow(window_, mem_dc, flags);
+  }
+
+  if (!result && (!window_capture_helper_.IsAeroEnabled() ||
+                  !previous_size_.equals(frame->size()))) {
+    result = PrintWindow(window_, mem_dc, 0);
+  }
+
+  // Aero is enabled or PrintWindow() failed, use BitBlt.
+  if (!result) {
+    result = BitBlt(mem_dc, 0, 0, frame->size().width(), frame->size().height(),
+                    window_dc, 0, 0, SRCCOPY);
+  }
+
+  SelectObject(mem_dc, previous_object);
+  DeleteDC(mem_dc);
+  ReleaseDC(window_, window_dc);
+
+  previous_size_ = frame->size();
+  window_size_map_[window_] = previous_size_;
+
+  frame->mutable_updated_region()->SetRect(
+      DesktopRect::MakeSize(frame->size()));
+  frame->set_top_left(
+      original_rect.top_left().subtract(GetFullscreenRect().top_left()));
+
+  if (!result) {
+    RTC_LOG(LS_ERROR) << "Both PrintWindow() and BitBlt() failed.";
+    return {Result::ERROR_TEMPORARY, nullptr};
+  }
+
+  // Rect for the data is relative to the first pixel of the frame.
+  cropped_rect.Translate(-original_rect.left(), -original_rect.top());
+  std::unique_ptr<DesktopFrame> cropped_frame =
+      CreateCroppedDesktopFrame(std::move(frame), cropped_rect);
+  RTC_DCHECK(cropped_frame);
+
+  if (capture_owned_windows) {
+    // If any owned/pop-up windows overlap the selected window, capture them
+    // and copy/composite their contents into the frame.
+    owned_windows_.clear();
+    OwnedWindowCollectorContext context(window_, unscaled_cropped_rect,
+                                        &window_capture_helper_,
+                                        &owned_windows_);
+
+    if (context.IsSelectedWindowValid()) {
+      EnumWindows(OwnedWindowCollector, reinterpret_cast<LPARAM>(&context));
+
+      if (!owned_windows_.empty()) {
+        if (!owned_window_capturer_) {
+          owned_window_capturer_ = std::make_unique<WindowCapturerWinGdi>(
+              enumerate_current_process_windows_);
+        }
+
+        // Owned windows are stored in top-down z-order, so this iterates in
+        // reverse to capture / draw them in bottom-up z-order
+        for (auto it = owned_windows_.rbegin(); it != owned_windows_.rend();
+             it++) {
+          HWND hwnd = *it;
+          if (owned_window_capturer_->SelectSource(
+                  reinterpret_cast<SourceId>(hwnd))) {
+            CaptureResults results = owned_window_capturer_->CaptureFrame(
+                /*capture_owned_windows*/ false);
+
+            if (results.result != DesktopCapturer::Result::SUCCESS) {
+              // Simply log any error capturing an owned/pop-up window without
+              // bubbling it up to the caller (an expected error here is that
+              // the owned/pop-up window was closed; any unexpected errors won't
+              // fail the outer capture).
+              RTC_LOG(LS_INFO) << "Capturing owned window failed (previous "
+                                  "error/warning pertained to that)";
+            } else {
+              // Copy / composite the captured frame into the outer frame. This
+              // may no-op if they no longer intersect (if the owned window was
+              // moved outside the owner bounds since scheduled for capture.)
+              cropped_frame->CopyIntersectingPixelsFrom(
+                  *results.frame, horizontal_scale, vertical_scale);
+            }
+          }
+        }
+      }
+    }
+  }
+
+  return {Result::SUCCESS, std::move(cropped_frame)};
+}
+
+// static
+std::unique_ptr<DesktopCapturer> WindowCapturerWinGdi::CreateRawWindowCapturer(
+    const DesktopCaptureOptions& options) {
+  return std::unique_ptr<DesktopCapturer>(
+      new WindowCapturerWinGdi(options.enumerate_current_process_windows()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/win/window_capturer_win_gdi.h b/third_party/libwebrtc/modules/desktop_capture/win/window_capturer_win_gdi.h
new file mode 100644
index 0000000000..bf94dfe192
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/win/window_capturer_win_gdi.h
@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WIN_WINDOW_CAPTURER_WIN_GDI_H_
+#define MODULES_DESKTOP_CAPTURE_WIN_WINDOW_CAPTURER_WIN_GDI_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/win/window_capture_utils.h"
+#include "modules/desktop_capture/window_finder_win.h"
+
+namespace webrtc {
+
+class WindowCapturerWinGdi : public DesktopCapturer {
+ public:
+  explicit WindowCapturerWinGdi(bool enumerate_current_process_windows);
+
+  // Disallow copy and assign
+  WindowCapturerWinGdi(const WindowCapturerWinGdi&) = delete;
+  WindowCapturerWinGdi& operator=(const WindowCapturerWinGdi&) = delete;
+
+  ~WindowCapturerWinGdi() override;
+
+  static std::unique_ptr<DesktopCapturer> CreateRawWindowCapturer(
+      const DesktopCaptureOptions& options);
+
+  // DesktopCapturer interface.
+  void Start(Callback* callback) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  bool IsOccluded(const DesktopVector& pos) override;
+
+ private:
+  struct CaptureResults {
+    Result result;
+    std::unique_ptr<DesktopFrame> frame;
+  };
+
+  CaptureResults CaptureFrame(bool capture_owned_windows);
+
+  Callback* callback_ = nullptr;
+
+  // HWND and HDC for the currently selected window or nullptr if window is not
+  // selected.
+  HWND window_ = nullptr;
+
+  DesktopSize previous_size_;
+
+  WindowCaptureHelperWin window_capture_helper_;
+
+  bool enumerate_current_process_windows_;
+
+  // This map is used to avoid flickering for the case when SelectWindow() calls
+  // are interleaved with Capture() calls.
+  std::map<HWND, DesktopSize> window_size_map_;
+
+  WindowFinderWin window_finder_;
+
+  std::vector<HWND> owned_windows_;
+  std::unique_ptr<WindowCapturerWinGdi> owned_window_capturer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WIN_WINDOW_CAPTURER_WIN_GDI_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_capturer_linux.cc b/third_party/libwebrtc/modules/desktop_capture/window_capturer_linux.cc
new file mode 100644
index 0000000000..4205bf9bc0
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_capturer_linux.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+
+#if defined(WEBRTC_USE_PIPEWIRE)
+#include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
+#endif  // defined(WEBRTC_USE_PIPEWIRE)
+
+#if defined(WEBRTC_USE_X11)
+#include "modules/desktop_capture/linux/x11/window_capturer_x11.h"
+#endif  // defined(WEBRTC_USE_X11)
+
+namespace webrtc {
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
+    const DesktopCaptureOptions& options) {
+#if defined(WEBRTC_USE_PIPEWIRE)
+  if (options.allow_pipewire() && BaseCapturerPipeWire::IsSupported()) {
+    return std::make_unique<BaseCapturerPipeWire>(options,
+                                                  CaptureType::kWindow);
+  }
+#endif  // defined(WEBRTC_USE_PIPEWIRE)
+
+#if defined(WEBRTC_USE_X11)
+  if (!DesktopCapturer::IsRunningUnderWayland())
+    return WindowCapturerX11::CreateRawWindowCapturer(options);
+#endif  // defined(WEBRTC_USE_X11)
+
+  return nullptr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_capturer_mac.mm b/third_party/libwebrtc/modules/desktop_capture/window_capturer_mac.mm
new file mode 100644
index 0000000000..10f6a74650
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_capturer_mac.mm
@@ -0,0 +1,221 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <ApplicationServices/ApplicationServices.h>
+#include <Cocoa/Cocoa.h>
+#include <CoreFoundation/CoreFoundation.h>
+
+#include <utility>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
+#include "modules/desktop_capture/mac/desktop_frame_cgimage.h"
+#include "modules/desktop_capture/mac/window_list_utils.h"
+#include "modules/desktop_capture/window_finder_mac.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+namespace {
+
+// Returns true if the window exists.
+bool IsWindowValid(CGWindowID id) {
+  CFArrayRef window_id_array =
+      CFArrayCreate(nullptr, reinterpret_cast<const void**>(&id), 1, nullptr);
+  CFArrayRef window_array =
+      CGWindowListCreateDescriptionFromArray(window_id_array);
+  bool valid = window_array && CFArrayGetCount(window_array);
+  CFRelease(window_id_array);
+  CFRelease(window_array);
+
+  return valid;
+}
+
+class WindowCapturerMac : public DesktopCapturer {
+ public:
+  explicit WindowCapturerMac(
+      rtc::scoped_refptr<FullScreenWindowDetector> full_screen_window_detector,
+      rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor);
+  ~WindowCapturerMac() override;
+
+  WindowCapturerMac(const WindowCapturerMac&) = delete;
+  WindowCapturerMac& operator=(const WindowCapturerMac&) = delete;
+
+  // DesktopCapturer interface.
+  void Start(Callback* callback) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+  bool FocusOnSelectedSource() override;
+  bool IsOccluded(const DesktopVector& pos) override;
+
+ private:
+  Callback* callback_ = nullptr;
+
+  // The window being captured.
+  CGWindowID window_id_ = 0;
+
+  rtc::scoped_refptr<FullScreenWindowDetector> full_screen_window_detector_;
+
+  const rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_;
+
+  WindowFinderMac window_finder_;
+
+  // Used to make sure that we only log the usage of fullscreen detection once.
+  bool fullscreen_usage_logged_ = false;
+};
+
+WindowCapturerMac::WindowCapturerMac(
+    rtc::scoped_refptr<FullScreenWindowDetector> full_screen_window_detector,
+    rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor)
+    : full_screen_window_detector_(std::move(full_screen_window_detector)),
+      configuration_monitor_(std::move(configuration_monitor)),
+      window_finder_(configuration_monitor_) {}
+
+WindowCapturerMac::~WindowCapturerMac() {}
+
+bool WindowCapturerMac::GetSourceList(SourceList* sources) {
+  return webrtc::GetWindowList(sources, true, true);
+}
+
+bool WindowCapturerMac::SelectSource(SourceId id) {
+  if (!IsWindowValid(id))
+    return false;
+  window_id_ = id;
+  return true;
+}
+
+bool WindowCapturerMac::FocusOnSelectedSource() {
+  if (!window_id_)
+    return false;
+
+  CGWindowID ids[1];
+  ids[0] = window_id_;
+  CFArrayRef window_id_array =
+      CFArrayCreate(nullptr, reinterpret_cast<const void**>(&ids), 1, nullptr);
+
+  CFArrayRef window_array =
+      CGWindowListCreateDescriptionFromArray(window_id_array);
+  if (!window_array || 0 == CFArrayGetCount(window_array)) {
+    // Could not find the window. It might have been closed.
+    RTC_LOG(LS_INFO) << "Window not found";
+    CFRelease(window_id_array);
+    return false;
+  }
+
+  CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
+      CFArrayGetValueAtIndex(window_array, 0));
+  CFNumberRef pid_ref = reinterpret_cast<CFNumberRef>(
+      CFDictionaryGetValue(window, kCGWindowOwnerPID));
+
+  int pid;
+  CFNumberGetValue(pid_ref, kCFNumberIntType, &pid);
+
+  // TODO(jiayl): this will bring the process main window to the front. We
+  // should find a way to bring only the window to the front.
+  bool result =
+      [[NSRunningApplication runningApplicationWithProcessIdentifier: pid]
+          activateWithOptions: NSApplicationActivateIgnoringOtherApps];
+
+  CFRelease(window_id_array);
+  CFRelease(window_array);
+  return result;
+}
+
+bool WindowCapturerMac::IsOccluded(const DesktopVector& pos) {
+  DesktopVector sys_pos = pos;
+  if (configuration_monitor_) {
+    auto configuration = configuration_monitor_->desktop_configuration();
+    sys_pos = pos.add(configuration.bounds.top_left());
+  }
+  return window_finder_.GetWindowUnderPoint(sys_pos) != window_id_;
+}
+
+void WindowCapturerMac::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+
+  callback_ = callback;
+}
+
+void WindowCapturerMac::CaptureFrame() {
+  TRACE_EVENT0("webrtc", "WindowCapturerMac::CaptureFrame");
+
+  if (!IsWindowValid(window_id_)) {
+    RTC_LOG(LS_ERROR) << "The window is not valid any longer.";
+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+    return;
+  }
+
+  CGWindowID on_screen_window = window_id_;
+  if (full_screen_window_detector_) {
+    full_screen_window_detector_->UpdateWindowListIfNeeded(
+        window_id_, [](DesktopCapturer::SourceList* sources) {
+          // Not using webrtc::GetWindowList(sources, true, false)
+          // as it doesn't allow to have in the result window with
+          // empty title along with titled window owned by the same pid.
+          return webrtc::GetWindowList(
+              [sources](CFDictionaryRef window) {
+                WindowId window_id = GetWindowId(window);
+                int pid = GetWindowOwnerPid(window);
+                if (window_id != kNullWindowId) {
+                  sources->push_back(DesktopCapturer::Source{window_id, pid, GetWindowTitle(window)});
+                }
+                return true;
+              },
+              true,
+              false);
+        });
+
+    CGWindowID full_screen_window = full_screen_window_detector_->FindFullScreenWindow(window_id_);
+
+    if (full_screen_window != kCGNullWindowID) {
+      // If this is the first time this happens, report to UMA that the feature is active.
+      if (!fullscreen_usage_logged_) {
+        LogDesktopCapturerFullscreenDetectorUsage();
+        fullscreen_usage_logged_ = true;
+      }
+      on_screen_window = full_screen_window;
+    }
+  }
+
+  std::unique_ptr<DesktopFrame> frame = DesktopFrameCGImage::CreateForWindow(on_screen_window);
+  if (!frame) {
+    RTC_LOG(LS_WARNING) << "Temporarily failed to capture window.";
+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+    return;
+  }
+
+  frame->mutable_updated_region()->SetRect(
+      DesktopRect::MakeSize(frame->size()));
+  frame->set_top_left(GetWindowBounds(on_screen_window).top_left());
+
+  float scale_factor = GetWindowScaleFactor(window_id_, frame->size());
+  frame->set_dpi(DesktopVector(kStandardDPI * scale_factor, kStandardDPI * scale_factor));
+
+  callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
+}
+
+}  // namespace
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
+    const DesktopCaptureOptions& options) {
+  return std::unique_ptr<DesktopCapturer>(new WindowCapturerMac(
+      options.full_screen_window_detector(), options.configuration_monitor()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_capturer_null.cc b/third_party/libwebrtc/modules/desktop_capture/window_capturer_null.cc
new file mode 100644
index 0000000000..6da2a76691
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_capturer_null.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+class WindowCapturerNull : public DesktopCapturer {
+ public:
+  WindowCapturerNull();
+  ~WindowCapturerNull() override;
+
+  WindowCapturerNull(const WindowCapturerNull&) = delete;
+  WindowCapturerNull& operator=(const WindowCapturerNull&) = delete;
+
+  // DesktopCapturer interface.
+  void Start(Callback* callback) override;
+  void CaptureFrame() override;
+  bool GetSourceList(SourceList* sources) override;
+  bool SelectSource(SourceId id) override;
+
+ private:
+  Callback* callback_ = nullptr;
+};
+
+WindowCapturerNull::WindowCapturerNull() {}
+WindowCapturerNull::~WindowCapturerNull() {}
+
+bool WindowCapturerNull::GetSourceList(SourceList* sources) {
+  // Not implemented yet.
+  return false;
+}
+
+bool WindowCapturerNull::SelectSource(SourceId id) {
+  // Not implemented yet.
+  return false;
+}
+
+void WindowCapturerNull::Start(Callback* callback) {
+  RTC_DCHECK(!callback_);
+  RTC_DCHECK(callback);
+
+  callback_ = callback;
+}
+
+void WindowCapturerNull::CaptureFrame() {
+  // Not implemented yet.
+  callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
+}
+
+}  // namespace
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
+    const DesktopCaptureOptions& options) {
+  return std::unique_ptr<DesktopCapturer>(new WindowCapturerNull());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_capturer_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/window_capturer_unittest.cc
new file mode 100644
index 0000000000..519c04601b
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_capturer_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class WindowCapturerTest : public ::testing::Test,
+                           public DesktopCapturer::Callback {
+ public:
+  void SetUp() override {
+    capturer_ = DesktopCapturer::CreateWindowCapturer(
+        DesktopCaptureOptions::CreateDefault());
+    ASSERT_TRUE(capturer_);
+  }
+
+  void TearDown() override {}
+
+  // DesktopCapturer::Callback interface
+  void OnCaptureResult(DesktopCapturer::Result result,
+                       std::unique_ptr<DesktopFrame> frame) override {
+    frame_ = std::move(frame);
+  }
+
+ protected:
+  std::unique_ptr<DesktopCapturer> capturer_;
+  std::unique_ptr<DesktopFrame> frame_;
+};
+
+// Verify that we can enumerate windows.
+// TODO(bugs.webrtc.org/12950): Re-enable when libc++ issue is fixed
+#if defined(WEBRTC_LINUX) && defined(MEMORY_SANITIZER)
+#define MAYBE_Enumerate DISABLED_Enumerate
+#else
+#define MAYBE_Enumerate Enumerate
+#endif
+TEST_F(WindowCapturerTest, MAYBE_Enumerate) {
+  DesktopCapturer::SourceList sources;
+  EXPECT_TRUE(capturer_->GetSourceList(&sources));
+
+  // Verify that window titles are set.
+  for (auto it = sources.begin(); it != sources.end(); ++it) {
+    EXPECT_FALSE(it->title.empty());
+  }
+}
+
+// Flaky on Linux. See: crbug.com/webrtc/7830.
+// Failing on macOS 11: See bugs.webrtc.org/12801
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+#define MAYBE_Capture DISABLED_Capture
+#else
+#define MAYBE_Capture Capture
+#endif
+// Verify we can capture a window.
+//
+// TODO(sergeyu): Currently this test just looks at the windows that already
+// exist. Ideally it should create a test window and capture from it, but there
+// is no easy cross-platform way to create new windows (potentially we could
+// have a python script showing Tk dialog, but launching code will differ
+// between platforms).
+TEST_F(WindowCapturerTest, MAYBE_Capture) {
+  DesktopCapturer::SourceList sources;
+  capturer_->Start(this);
+  EXPECT_TRUE(capturer_->GetSourceList(&sources));
+
+  // Verify that we can select and capture each window.
+  for (auto it = sources.begin(); it != sources.end(); ++it) {
+    frame_.reset();
+    if (capturer_->SelectSource(it->id)) {
+      capturer_->CaptureFrame();
+    }
+
+    // If we failed to capture a window make sure it no longer exists.
+    if (!frame_.get()) {
+      DesktopCapturer::SourceList new_list;
+      EXPECT_TRUE(capturer_->GetSourceList(&new_list));
+      for (auto new_list_it = new_list.begin(); new_list_it != new_list.end();
+           ++new_list_it) {
+        EXPECT_FALSE(it->id == new_list_it->id);
+      }
+      continue;
+    }
+
+    EXPECT_GT(frame_->size().width(), 0);
+    EXPECT_GT(frame_->size().height(), 0);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_capturer_win.cc b/third_party/libwebrtc/modules/desktop_capture/window_capturer_win.cc
new file mode 100644
index 0000000000..f289746e30
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_capturer_win.cc
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/win/window_capturer_win_gdi.h"
+
+#if defined(RTC_ENABLE_WIN_WGC)
+#include "modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h"
+#include "modules/desktop_capture/fallback_desktop_capturer_wrapper.h"
+#include "modules/desktop_capture/win/wgc_capturer_win.h"
+#include "rtc_base/win/windows_version.h"
+#endif  // defined(RTC_ENABLE_WIN_WGC)
+
+namespace webrtc {
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
+    const DesktopCaptureOptions& options) {
+  std::unique_ptr<DesktopCapturer> capturer(
+      WindowCapturerWinGdi::CreateRawWindowCapturer(options));
+#if defined(RTC_ENABLE_WIN_WGC)
+  if (options.allow_wgc_capturer_fallback() &&
+      rtc::rtc_win::GetVersion() >= rtc::rtc_win::Version::VERSION_WIN11) {
+    // BlankDectector capturer will send an error when it detects a failed
+    // GDI rendering, then Fallback capturer will try to capture it again with
+    // WGC.
+    capturer = std::make_unique<BlankDetectorDesktopCapturerWrapper>(
+        std::move(capturer), RgbaColor(0, 0, 0, 0),
+        /*check_per_capture*/ true);
+
+    capturer = std::make_unique<FallbackDesktopCapturerWrapper>(
+        std::move(capturer),
+        WgcCapturerWin::CreateRawWindowCapturer(
+            options, /*allow_delayed_capturable_check*/ true));
+  }
+#endif  // defined(RTC_ENABLE_WIN_WGC)
+  return capturer;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_finder.cc b/third_party/libwebrtc/modules/desktop_capture/window_finder.cc
new file mode 100644
index 0000000000..86127d4c05
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_finder.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/window_finder.h"
+
+namespace webrtc {
+
+WindowFinder::Options::Options() = default;
+WindowFinder::Options::~Options() = default;
+WindowFinder::Options::Options(const WindowFinder::Options& other) = default;
+WindowFinder::Options::Options(WindowFinder::Options&& other) = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_finder.h b/third_party/libwebrtc/modules/desktop_capture/window_finder.h
new file mode 100644
index 0000000000..99e3cce559
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_finder.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_H_
+#define MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_H_
+
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
+#endif
+
+namespace webrtc {
+
+#if defined(WEBRTC_USE_X11)
+class XAtomCache;
+#endif
+
+// An interface to return the id of the visible window under a certain point.
+class WindowFinder {
+ public:
+  WindowFinder() = default;
+  virtual ~WindowFinder() = default;
+
+  // Returns the id of the visible window under `point`. This function returns
+  // kNullWindowId if no window is under `point` and the platform does not have
+  // "root window" concept, i.e. the visible area under `point` is the desktop.
+  // `point` is always in system coordinate, i.e. the primary monitor always
+  // starts from (0, 0).
+  virtual WindowId GetWindowUnderPoint(DesktopVector point) = 0;
+
+  struct Options final {
+    Options();
+    ~Options();
+    Options(const Options& other);
+    Options(Options&& other);
+
+#if defined(WEBRTC_USE_X11)
+    XAtomCache* cache = nullptr;
+#endif
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+    rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor;
+#endif
+  };
+
+  // Creates a platform-independent WindowFinder implementation. This function
+  // returns nullptr if `options` does not contain enough information or
+  // WindowFinder does not support current platform.
+  static std::unique_ptr<WindowFinder> Create(const Options& options);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_finder_mac.h b/third_party/libwebrtc/modules/desktop_capture/window_finder_mac.h
new file mode 100644
index 0000000000..988dd497dd
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_finder_mac.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_MAC_H_
+#define MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_MAC_H_
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/window_finder.h"
+
+namespace webrtc {
+
+class DesktopConfigurationMonitor;
+
+// The implementation of WindowFinder for Mac OSX.
+class WindowFinderMac final : public WindowFinder {
+ public:
+  explicit WindowFinderMac(
+      rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor);
+  ~WindowFinderMac() override;
+
+  // WindowFinder implementation.
+  WindowId GetWindowUnderPoint(DesktopVector point) override;
+
+ private:
+  const rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_MAC_H_
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_finder_mac.mm b/third_party/libwebrtc/modules/desktop_capture/window_finder_mac.mm
new file mode 100644
index 0000000000..e1d0316c79
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_finder_mac.mm
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/window_finder_mac.h"
+
+#include <CoreFoundation/CoreFoundation.h>
+
+#include <memory>
+#include <utility>
+
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
+#include "modules/desktop_capture/mac/window_list_utils.h"
+
+namespace webrtc {
+
+WindowFinderMac::WindowFinderMac(
+    rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor)
+    : configuration_monitor_(std::move(configuration_monitor)) {}
+WindowFinderMac::~WindowFinderMac() = default;
+
+WindowId WindowFinderMac::GetWindowUnderPoint(DesktopVector point) {
+  WindowId id = kNullWindowId;
+  GetWindowList(
+      [&id, point](CFDictionaryRef window) {
+        DesktopRect bounds;
+        bounds = GetWindowBounds(window);
+        if (bounds.Contains(point)) {
+          id = GetWindowId(window);
+          return false;
+        }
+        return true;
+      },
+      true,
+      true);
+  return id;
+}
+
+// static
+std::unique_ptr<WindowFinder> WindowFinder::Create(
+    const WindowFinder::Options& options) {
+  return std::make_unique<WindowFinderMac>(options.configuration_monitor);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_finder_unittest.cc b/third_party/libwebrtc/modules/desktop_capture/window_finder_unittest.cc
new file mode 100644
index 0000000000..ac13f124d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_finder_unittest.cc
@@ -0,0 +1,178 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/window_finder.h"
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/screen_drawer.h"
+#include "rtc_base/logging.h"
+#include "test/gtest.h"
+
+#if defined(WEBRTC_USE_X11)
+#include "modules/desktop_capture/linux/x11/shared_x_display.h"
+#include "modules/desktop_capture/linux/x11/x_atom_cache.h"
+#endif
+
+#if defined(WEBRTC_WIN)
+#include <windows.h>
+
+#include "modules/desktop_capture/win/window_capture_utils.h"
+#include "modules/desktop_capture/window_finder_win.h"
+#endif
+
+namespace webrtc {
+
+namespace {
+
+#if defined(WEBRTC_WIN)
+// ScreenDrawerWin does not have a message loop, so it's unresponsive to user
+// inputs. WindowFinderWin cannot detect this kind of unresponsive windows.
+// Instead, console window is used to test WindowFinderWin.
+TEST(WindowFinderTest, FindConsoleWindow) {
+  // Creates a ScreenDrawer to avoid this test from conflicting with
+  // ScreenCapturerIntegrationTest: both tests require its window to be in
+  // foreground.
+  //
+  // In ScreenCapturer related tests, this is controlled by
+  // ScreenDrawer, which has a global lock to ensure only one ScreenDrawer
+  // window is active. So even we do not use ScreenDrawer for Windows test,
+  // creating an instance can block ScreenCapturer related tests until this test
+  // finishes.
+  //
+  // Usually the test framework should take care of this "isolated test"
+  // requirement, but unfortunately WebRTC trybots do not support this.
+  std::unique_ptr<ScreenDrawer> drawer = ScreenDrawer::Create();
+  const int kMaxSize = 10000;
+  // Enlarges current console window.
+  system("mode 1000,1000");
+  const HWND console_window = GetConsoleWindow();
+  // Ensures that current console window is visible.
+  ShowWindow(console_window, SW_MAXIMIZE);
+  // Moves the window to the top-left of the display.
+  MoveWindow(console_window, 0, 0, kMaxSize, kMaxSize, true);
+
+  bool should_restore_notopmost =
+      (GetWindowLong(console_window, GWL_EXSTYLE) & WS_EX_TOPMOST) == 0;
+
+  // Brings console window to top.
+  SetWindowPos(console_window, HWND_TOPMOST, 0, 0, 0, 0,
+               SWP_NOMOVE | SWP_NOSIZE);
+  BringWindowToTop(console_window);
+
+  bool success = false;
+  WindowFinderWin finder;
+  for (int i = 0; i < kMaxSize; i++) {
+    const DesktopVector spot(i, i);
+    const HWND id = reinterpret_cast<HWND>(finder.GetWindowUnderPoint(spot));
+    if (id == console_window) {
+      success = true;
+      break;
+    }
+  }
+  if (should_restore_notopmost)
+    SetWindowPos(console_window, HWND_NOTOPMOST, 0, 0, 0, 0,
+                 SWP_NOMOVE | SWP_NOSIZE | SWP_NOACTIVATE);
+
+  if (!success)
+    FAIL();
+}
+
+#else
+TEST(WindowFinderTest, FindDrawerWindow) {
+  WindowFinder::Options options;
+#if defined(WEBRTC_USE_X11)
+  std::unique_ptr<XAtomCache> cache;
+  const auto shared_x_display = SharedXDisplay::CreateDefault();
+  if (shared_x_display) {
+    cache = std::make_unique<XAtomCache>(shared_x_display->display());
+    options.cache = cache.get();
+  }
+#endif
+  std::unique_ptr<WindowFinder> finder = WindowFinder::Create(options);
+  if (!finder) {
+    RTC_LOG(LS_WARNING)
+        << "No WindowFinder implementation for current platform.";
+    return;
+  }
+
+  std::unique_ptr<ScreenDrawer> drawer = ScreenDrawer::Create();
+  if (!drawer) {
+    RTC_LOG(LS_WARNING)
+        << "No ScreenDrawer implementation for current platform.";
+    return;
+  }
+
+  if (drawer->window_id() == kNullWindowId) {
+    // TODO(zijiehe): WindowFinderTest can use a dedicated window without
+    // relying on ScreenDrawer.
+    RTC_LOG(LS_WARNING)
+        << "ScreenDrawer implementation for current platform does "
+           "create a window.";
+    return;
+  }
+
+  // ScreenDrawer may not be able to bring the window to the top. So we test
+  // several spots, at least one of them should succeed.
+  const DesktopRect region = drawer->DrawableRegion();
+  if (region.is_empty()) {
+    RTC_LOG(LS_WARNING)
+        << "ScreenDrawer::DrawableRegion() is too small for the "
+           "WindowFinderTest.";
+    return;
+  }
+
+  for (int i = 0; i < region.width(); i++) {
+    const DesktopVector spot(
+        region.left() + i, region.top() + i * region.height() / region.width());
+    const WindowId id = finder->GetWindowUnderPoint(spot);
+    if (id == drawer->window_id()) {
+      return;
+    }
+  }
+
+  FAIL();
+}
+#endif
+
+TEST(WindowFinderTest, ShouldReturnNullWindowIfSpotIsOutOfScreen) {
+  WindowFinder::Options options;
+#if defined(WEBRTC_USE_X11)
+  std::unique_ptr<XAtomCache> cache;
+  const auto shared_x_display = SharedXDisplay::CreateDefault();
+  if (shared_x_display) {
+    cache = std::make_unique<XAtomCache>(shared_x_display->display());
+    options.cache = cache.get();
+  }
+#endif
+  std::unique_ptr<WindowFinder> finder = WindowFinder::Create(options);
+  if (!finder) {
+    RTC_LOG(LS_WARNING)
+        << "No WindowFinder implementation for current platform.";
+    return;
+  }
+
+  ASSERT_EQ(kNullWindowId,
+            finder->GetWindowUnderPoint(DesktopVector(INT16_MAX, INT16_MAX)));
+  ASSERT_EQ(kNullWindowId,
+            finder->GetWindowUnderPoint(DesktopVector(INT16_MAX, INT16_MIN)));
+  ASSERT_EQ(kNullWindowId,
+            finder->GetWindowUnderPoint(DesktopVector(INT16_MIN, INT16_MAX)));
+  ASSERT_EQ(kNullWindowId,
+            finder->GetWindowUnderPoint(DesktopVector(INT16_MIN, INT16_MIN)));
+}
+
+}  // namespace
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_finder_win.cc b/third_party/libwebrtc/modules/desktop_capture/window_finder_win.cc
new file mode 100644
index 0000000000..a8c3d39e19
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_finder_win.cc
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/desktop_capture/window_finder_win.h"
+
+#include <windows.h>
+
+#include <memory>
+
+namespace webrtc {
+
+WindowFinderWin::WindowFinderWin() = default;
+WindowFinderWin::~WindowFinderWin() = default;
+
+WindowId WindowFinderWin::GetWindowUnderPoint(DesktopVector point) {
+  HWND window = WindowFromPoint(POINT{point.x(), point.y()});
+  if (!window) {
+    return kNullWindowId;
+  }
+
+  // The difference between GA_ROOTOWNER and GA_ROOT can be found at
+  // https://groups.google.com/a/chromium.org/forum/#!topic/chromium-dev/Hirr_DkuZdw.
+  // In short, we should use GA_ROOT, since we only care about the root window
+  // but not the owner.
+  window = GetAncestor(window, GA_ROOT);
+  if (!window) {
+    return kNullWindowId;
+  }
+
+  return reinterpret_cast<WindowId>(window);
+}
+
+// static
+std::unique_ptr<WindowFinder> WindowFinder::Create(
+    const WindowFinder::Options& options) {
+  return std::make_unique<WindowFinderWin>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/desktop_capture/window_finder_win.h b/third_party/libwebrtc/modules/desktop_capture/window_finder_win.h
new file mode 100644
index 0000000000..a04e7e1aae
--- /dev/null
+++ b/third_party/libwebrtc/modules/desktop_capture/window_finder_win.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_WIN_H_
+#define MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_WIN_H_
+
+#include "modules/desktop_capture/window_finder.h"
+
+namespace webrtc {
+
+// The implementation of WindowFinder for Windows.
+class WindowFinderWin final : public WindowFinder {
+ public:
+  WindowFinderWin();
+  ~WindowFinderWin() override;
+
+  // WindowFinder implementation.
+  WindowId GetWindowUnderPoint(DesktopVector point) override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_WIN_H_
diff --git a/third_party/libwebrtc/modules/include/module_common_types.h b/third_party/libwebrtc/modules/include/module_common_types.h
new file mode 100644
index 0000000000..8e4e5465fa
--- /dev/null
+++ b/third_party/libwebrtc/modules/include/module_common_types.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+#define MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+namespace webrtc {
+
+// Interface used by the CallStats class to distribute call statistics.
+// Callbacks will be triggered as soon as the class has been registered to a
+// CallStats object using RegisterStatsObserver.
+class CallStatsObserver {
+ public:
+  virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) = 0;
+
+  virtual ~CallStatsObserver() {}
+};
+
+// Interface used by NackModule and JitterBuffer.
+class NackSender {
+ public:
+  // If `buffering_allowed`, other feedback messages (e.g. key frame requests)
+  // may be added to the same outgoing feedback message. In that case, it's up
+  // to the user of the interface to ensure that when all buffer-able messages
+  // have been added, the feedback message is triggered.
+  virtual void SendNack(const std::vector<uint16_t>& sequence_numbers,
+                        bool buffering_allowed) = 0;
+
+ protected:
+  virtual ~NackSender() {}
+};
+
+// Interface used by NackModule and JitterBuffer.
+class KeyFrameRequestSender {
+ public:
+  virtual void RequestKeyFrame() = 0;
+
+ protected:
+  virtual ~KeyFrameRequestSender() {}
+};
+
+// Interface used by LossNotificationController to communicate to RtpRtcp.
+class LossNotificationSender {
+ public:
+  virtual ~LossNotificationSender() {}
+
+  virtual void SendLossNotification(uint16_t last_decoded_seq_num,
+                                    uint16_t last_received_seq_num,
+                                    bool decodability_flag,
+                                    bool buffering_allowed) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
diff --git a/third_party/libwebrtc/modules/include/module_common_types_public.h b/third_party/libwebrtc/modules/include/module_common_types_public.h
new file mode 100644
index 0000000000..5a6f634df7
--- /dev/null
+++ b/third_party/libwebrtc/modules/include/module_common_types_public.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_INCLUDE_MODULE_COMMON_TYPES_PUBLIC_H_
+#define MODULES_INCLUDE_MODULE_COMMON_TYPES_PUBLIC_H_
+
+#include <limits>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+template <typename U>
+inline bool IsNewer(U value, U prev_value) {
+  static_assert(!std::numeric_limits<U>::is_signed, "U must be unsigned");
+  // kBreakpoint is the half-way mark for the type U. For instance, for a
+  // uint16_t it will be 0x8000, and for a uint32_t, it will be 0x8000000.
+  constexpr U kBreakpoint = (std::numeric_limits<U>::max() >> 1) + 1;
+  // Distinguish between elements that are exactly kBreakpoint apart.
+  // If t1>t2 and |t1-t2| = kBreakpoint: IsNewer(t1,t2)=true,
+  // IsNewer(t2,t1)=false
+  // rather than having IsNewer(t1,t2) = IsNewer(t2,t1) = false.
+  if (value - prev_value == kBreakpoint) {
+    return value > prev_value;
+  }
+  return value != prev_value &&
+         static_cast<U>(value - prev_value) < kBreakpoint;
+}
+
+// NB: Doesn't fulfill strict weak ordering requirements.
+//     Mustn't be used as std::map Compare function.
+inline bool IsNewerSequenceNumber(uint16_t sequence_number,
+                                  uint16_t prev_sequence_number) {
+  return IsNewer(sequence_number, prev_sequence_number);
+}
+
+// NB: Doesn't fulfill strict weak ordering requirements.
+//     Mustn't be used as std::map Compare function.
+inline bool IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) {
+  return IsNewer(timestamp, prev_timestamp);
+}
+
+inline uint16_t LatestSequenceNumber(uint16_t sequence_number1,
+                                     uint16_t sequence_number2) {
+  return IsNewerSequenceNumber(sequence_number1, sequence_number2)
+             ? sequence_number1
+             : sequence_number2;
+}
+
+inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) {
+  return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2;
+}
+
+}  // namespace webrtc
+#endif  // MODULES_INCLUDE_MODULE_COMMON_TYPES_PUBLIC_H_
diff --git a/third_party/libwebrtc/modules/include/module_fec_types.h b/third_party/libwebrtc/modules/include/module_fec_types.h
new file mode 100644
index 0000000000..fa35342764
--- /dev/null
+++ b/third_party/libwebrtc/modules/include/module_fec_types.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_INCLUDE_MODULE_FEC_TYPES_H_
+#define MODULES_INCLUDE_MODULE_FEC_TYPES_H_
+
+namespace webrtc {
+
+// Types for the FEC packet masks. The type `kFecMaskRandom` is based on a
+// random loss model. The type `kFecMaskBursty` is based on a bursty/consecutive
+// loss model. The packet masks are defined in
+// modules/rtp_rtcp/fec_private_tables_random(bursty).h
+enum FecMaskType {
+  kFecMaskRandom,
+  kFecMaskBursty,
+};
+
+// Struct containing forward error correction settings.
+struct FecProtectionParams {
+  int fec_rate = 0;
+  int max_fec_frames = 0;
+  FecMaskType fec_mask_type = FecMaskType::kFecMaskRandom;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_INCLUDE_MODULE_FEC_TYPES_H_
diff --git a/third_party/libwebrtc/modules/module_api_gn/moz.build b/third_party/libwebrtc/modules/module_api_gn/moz.build
new file mode 100644
index 0000000000..3b999fdfdf
--- /dev/null
+++ b/third_party/libwebrtc/modules/module_api_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("module_api_gn")
diff --git a/third_party/libwebrtc/modules/module_api_public_gn/moz.build b/third_party/libwebrtc/modules/module_api_public_gn/moz.build
new file mode 100644
index 0000000000..6d3952410c
--- /dev/null
+++ b/third_party/libwebrtc/modules/module_api_public_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("module_api_public_gn")
diff --git a/third_party/libwebrtc/modules/module_common_types_unittest.cc b/third_party/libwebrtc/modules/module_common_types_unittest.cc
new file mode 100644
index 0000000000..bec3b5da22
--- /dev/null
+++ b/third_party/libwebrtc/modules/module_common_types_unittest.cc
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/include/module_common_types.h"
+
+#include "modules/include/module_common_types_public.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(IsNewerSequenceNumber, Equal) {
+  EXPECT_FALSE(IsNewerSequenceNumber(0x0001, 0x0001));
+}
+
+TEST(IsNewerSequenceNumber, NoWrap) {
+  EXPECT_TRUE(IsNewerSequenceNumber(0xFFFF, 0xFFFE));
+  EXPECT_TRUE(IsNewerSequenceNumber(0x0001, 0x0000));
+  EXPECT_TRUE(IsNewerSequenceNumber(0x0100, 0x00FF));
+}
+
+TEST(IsNewerSequenceNumber, ForwardWrap) {
+  EXPECT_TRUE(IsNewerSequenceNumber(0x0000, 0xFFFF));
+  EXPECT_TRUE(IsNewerSequenceNumber(0x0000, 0xFF00));
+  EXPECT_TRUE(IsNewerSequenceNumber(0x00FF, 0xFFFF));
+  EXPECT_TRUE(IsNewerSequenceNumber(0x00FF, 0xFF00));
+}
+
+TEST(IsNewerSequenceNumber, BackwardWrap) {
+  EXPECT_FALSE(IsNewerSequenceNumber(0xFFFF, 0x0000));
+  EXPECT_FALSE(IsNewerSequenceNumber(0xFF00, 0x0000));
+  EXPECT_FALSE(IsNewerSequenceNumber(0xFFFF, 0x00FF));
+  EXPECT_FALSE(IsNewerSequenceNumber(0xFF00, 0x00FF));
+}
+
+TEST(IsNewerSequenceNumber, HalfWayApart) {
+  EXPECT_TRUE(IsNewerSequenceNumber(0x8000, 0x0000));
+  EXPECT_FALSE(IsNewerSequenceNumber(0x0000, 0x8000));
+}
+
+TEST(IsNewerTimestamp, Equal) {
+  EXPECT_FALSE(IsNewerTimestamp(0x00000001, 0x000000001));
+}
+
+TEST(IsNewerTimestamp, NoWrap) {
+  EXPECT_TRUE(IsNewerTimestamp(0xFFFFFFFF, 0xFFFFFFFE));
+  EXPECT_TRUE(IsNewerTimestamp(0x00000001, 0x00000000));
+  EXPECT_TRUE(IsNewerTimestamp(0x00010000, 0x0000FFFF));
+}
+
+TEST(IsNewerTimestamp, ForwardWrap) {
+  EXPECT_TRUE(IsNewerTimestamp(0x00000000, 0xFFFFFFFF));
+  EXPECT_TRUE(IsNewerTimestamp(0x00000000, 0xFFFF0000));
+  EXPECT_TRUE(IsNewerTimestamp(0x0000FFFF, 0xFFFFFFFF));
+  EXPECT_TRUE(IsNewerTimestamp(0x0000FFFF, 0xFFFF0000));
+}
+
+TEST(IsNewerTimestamp, BackwardWrap) {
+  EXPECT_FALSE(IsNewerTimestamp(0xFFFFFFFF, 0x00000000));
+  EXPECT_FALSE(IsNewerTimestamp(0xFFFF0000, 0x00000000));
+  EXPECT_FALSE(IsNewerTimestamp(0xFFFFFFFF, 0x0000FFFF));
+  EXPECT_FALSE(IsNewerTimestamp(0xFFFF0000, 0x0000FFFF));
+}
+
+TEST(IsNewerTimestamp, HalfWayApart) {
+  EXPECT_TRUE(IsNewerTimestamp(0x80000000, 0x00000000));
+  EXPECT_FALSE(IsNewerTimestamp(0x00000000, 0x80000000));
+}
+
+TEST(LatestSequenceNumber, NoWrap) {
+  EXPECT_EQ(0xFFFFu, LatestSequenceNumber(0xFFFF, 0xFFFE));
+  EXPECT_EQ(0x0001u, LatestSequenceNumber(0x0001, 0x0000));
+  EXPECT_EQ(0x0100u, LatestSequenceNumber(0x0100, 0x00FF));
+
+  EXPECT_EQ(0xFFFFu, LatestSequenceNumber(0xFFFE, 0xFFFF));
+  EXPECT_EQ(0x0001u, LatestSequenceNumber(0x0000, 0x0001));
+  EXPECT_EQ(0x0100u, LatestSequenceNumber(0x00FF, 0x0100));
+}
+
+TEST(LatestSequenceNumber, Wrap) {
+  EXPECT_EQ(0x0000u, LatestSequenceNumber(0x0000, 0xFFFF));
+  EXPECT_EQ(0x0000u, LatestSequenceNumber(0x0000, 0xFF00));
+  EXPECT_EQ(0x00FFu, LatestSequenceNumber(0x00FF, 0xFFFF));
+  EXPECT_EQ(0x00FFu, LatestSequenceNumber(0x00FF, 0xFF00));
+
+  EXPECT_EQ(0x0000u, LatestSequenceNumber(0xFFFF, 0x0000));
+  EXPECT_EQ(0x0000u, LatestSequenceNumber(0xFF00, 0x0000));
+  EXPECT_EQ(0x00FFu, LatestSequenceNumber(0xFFFF, 0x00FF));
+  EXPECT_EQ(0x00FFu, LatestSequenceNumber(0xFF00, 0x00FF));
+}
+
+TEST(LatestTimestamp, NoWrap) {
+  EXPECT_EQ(0xFFFFFFFFu, LatestTimestamp(0xFFFFFFFF, 0xFFFFFFFE));
+  EXPECT_EQ(0x00000001u, LatestTimestamp(0x00000001, 0x00000000));
+  EXPECT_EQ(0x00010000u, LatestTimestamp(0x00010000, 0x0000FFFF));
+}
+
+TEST(LatestTimestamp, Wrap) {
+  EXPECT_EQ(0x00000000u, LatestTimestamp(0x00000000, 0xFFFFFFFF));
+  EXPECT_EQ(0x00000000u, LatestTimestamp(0x00000000, 0xFFFF0000));
+  EXPECT_EQ(0x0000FFFFu, LatestTimestamp(0x0000FFFF, 0xFFFFFFFF));
+  EXPECT_EQ(0x0000FFFFu, LatestTimestamp(0x0000FFFF, 0xFFFF0000));
+
+  EXPECT_EQ(0x00000000u, LatestTimestamp(0xFFFFFFFF, 0x00000000));
+  EXPECT_EQ(0x00000000u, LatestTimestamp(0xFFFF0000, 0x00000000));
+  EXPECT_EQ(0x0000FFFFu, LatestTimestamp(0xFFFFFFFF, 0x0000FFFF));
+  EXPECT_EQ(0x0000FFFFu, LatestTimestamp(0xFFFF0000, 0x0000FFFF));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/module_fec_api_gn/moz.build b/third_party/libwebrtc/modules/module_fec_api_gn/moz.build
new file mode 100644
index 0000000000..3badc9ded1
--- /dev/null
+++ b/third_party/libwebrtc/modules/module_fec_api_gn/moz.build
@@ -0,0 +1,201 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("module_fec_api_gn")
diff --git a/third_party/libwebrtc/modules/pacing/BUILD.gn b/third_party/libwebrtc/modules/pacing/BUILD.gn
new file mode 100644
index 0000000000..3fdac71a0c
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/BUILD.gn
@@ -0,0 +1,117 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("pacing") {
+  # Client code SHOULD NOT USE THIS TARGET, but for now it needs to be public
+  # because there exists client code that uses it.
+  # TODO(bugs.webrtc.org/9808): Move to private visibility as soon as that
+  # client code gets updated.
+  visibility = [ "*" ]
+  sources = [
+    "bitrate_prober.cc",
+    "bitrate_prober.h",
+    "pacing_controller.cc",
+    "pacing_controller.h",
+    "packet_router.cc",
+    "packet_router.h",
+    "prioritized_packet_queue.cc",
+    "prioritized_packet_queue.h",
+    "rtp_packet_pacer.h",
+    "task_queue_paced_sender.cc",
+    "task_queue_paced_sender.h",
+  ]
+
+  deps = [
+    ":interval_budget",
+    "../../api:field_trials_view",
+    "../../api:field_trials_view",
+    "../../api:function_view",
+    "../../api:sequence_checker",
+    "../../api/rtc_event_log",
+    "../../api/task_queue:pending_task_safety_flag",
+    "../../api/task_queue:task_queue",
+    "../../api/transport:field_trial_based_config",
+    "../../api/transport:network_control",
+    "../../api/units:data_rate",
+    "../../api/units:data_size",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../logging:rtc_event_bwe",
+    "../../logging:rtc_event_pacing",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:rtc_task_queue",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:unused",
+    "../../system_wrappers",
+    "../../system_wrappers:metrics",
+    "../rtp_rtcp",
+    "../rtp_rtcp:rtp_rtcp_format",
+    "../utility:utility",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("interval_budget") {
+  sources = [
+    "interval_budget.cc",
+    "interval_budget.h",
+  ]
+
+  deps = [
+    "../../rtc_base:checks",
+    "../../rtc_base:safe_conversions",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("pacing_unittests") {
+    testonly = true
+
+    sources = [
+      "bitrate_prober_unittest.cc",
+      "interval_budget_unittest.cc",
+      "pacing_controller_unittest.cc",
+      "packet_router_unittest.cc",
+      "prioritized_packet_queue_unittest.cc",
+      "task_queue_paced_sender_unittest.cc",
+    ]
+    deps = [
+      ":interval_budget",
+      ":pacing",
+      "../../api/task_queue:task_queue",
+      "../../api/transport:network_control",
+      "../../api/units:data_rate",
+      "../../api/units:time_delta",
+      "../../api/units:timestamp",
+      "../../rtc_base:checks",
+      "../../rtc_base:rtc_base_tests_utils",
+      "../../rtc_base/experiments:alr_experiment",
+      "../../system_wrappers",
+      "../../test:explicit_key_value_config",
+      "../../test:scoped_key_value_config",
+      "../../test:test_support",
+      "../../test/time_controller:time_controller",
+      "../rtp_rtcp",
+      "../rtp_rtcp:mock_rtp_rtcp",
+      "../rtp_rtcp:rtp_rtcp_format",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/pacing/DEPS b/third_party/libwebrtc/modules/pacing/DEPS
new file mode 100644
index 0000000000..42f3dfcb14
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/DEPS
@@ -0,0 +1,6 @@
+include_rules = [
+  "+system_wrappers", 
+  # Avoid directly using field_trial. Instead use FieldTrialsView.
+  "-system_wrappers/include/field_trial.h",  
+  "+logging/rtc_event_log"
+]
diff --git a/third_party/libwebrtc/modules/pacing/OWNERS b/third_party/libwebrtc/modules/pacing/OWNERS
new file mode 100644
index 0000000000..7266156891
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/OWNERS
@@ -0,0 +1,5 @@
+stefan@webrtc.org
+mflodman@webrtc.org
+philipel@webrtc.org
+srte@webrtc.org
+sprang@webrtc.org
diff --git a/third_party/libwebrtc/modules/pacing/bitrate_prober.cc b/third_party/libwebrtc/modules/pacing/bitrate_prober.cc
new file mode 100644
index 0000000000..e8ebf54f32
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/bitrate_prober.cc
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/bitrate_prober.h"
+
+#include <algorithm>
+
+#include "absl/memory/memory.h"
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+constexpr TimeDelta kProbeClusterTimeout = TimeDelta::Seconds(5);
+constexpr size_t kMaxPendingProbeClusters = 5;
+
+}  // namespace
+
+BitrateProberConfig::BitrateProberConfig(
+    const FieldTrialsView* key_value_config)
+    : min_probe_delta("min_probe_delta", TimeDelta::Millis(2)),
+      max_probe_delay("max_probe_delay", TimeDelta::Millis(10)),
+      min_packet_size("min_packet_size", DataSize::Bytes(200)) {
+  ParseFieldTrial({&min_probe_delta, &max_probe_delay, &min_packet_size},
+                  key_value_config->Lookup("WebRTC-Bwe-ProbingBehavior"));
+}
+
+BitrateProber::BitrateProber(const FieldTrialsView& field_trials)
+    : probing_state_(ProbingState::kDisabled),
+      next_probe_time_(Timestamp::PlusInfinity()),
+      config_(&field_trials) {
+  SetEnabled(true);
+}
+
+void BitrateProber::SetEnabled(bool enable) {
+  if (enable) {
+    if (probing_state_ == ProbingState::kDisabled) {
+      probing_state_ = ProbingState::kInactive;
+      RTC_LOG(LS_INFO) << "Bandwidth probing enabled, set to inactive";
+    }
+  } else {
+    probing_state_ = ProbingState::kDisabled;
+    RTC_LOG(LS_INFO) << "Bandwidth probing disabled";
+  }
+}
+
+void BitrateProber::OnIncomingPacket(DataSize packet_size) {
+  // Don't initialize probing unless we have something large enough to start
+  // probing.
+  // Note that the pacer can send several packets at once when sending a probe,
+  // and thus, packets can be smaller than needed for a probe.
+  if (probing_state_ == ProbingState::kInactive && !clusters_.empty() &&
+      packet_size >=
+          std::min(RecommendedMinProbeSize(), config_.min_packet_size.Get())) {
+    // Send next probe right away.
+    next_probe_time_ = Timestamp::MinusInfinity();
+    probing_state_ = ProbingState::kActive;
+  }
+}
+
+void BitrateProber::CreateProbeCluster(
+    const ProbeClusterConfig& cluster_config) {
+  RTC_DCHECK(probing_state_ != ProbingState::kDisabled);
+
+  while (!clusters_.empty() &&
+         (cluster_config.at_time - clusters_.front().requested_at >
+              kProbeClusterTimeout ||
+          clusters_.size() > kMaxPendingProbeClusters)) {
+    clusters_.pop();
+  }
+
+  ProbeCluster cluster;
+  cluster.requested_at = cluster_config.at_time;
+  cluster.pace_info.probe_cluster_min_probes =
+      cluster_config.target_probe_count;
+  cluster.pace_info.probe_cluster_min_bytes =
+      (cluster_config.target_data_rate * cluster_config.target_duration)
+          .bytes();
+  RTC_DCHECK_GE(cluster.pace_info.probe_cluster_min_bytes, 0);
+  cluster.pace_info.send_bitrate_bps = cluster_config.target_data_rate.bps();
+  cluster.pace_info.probe_cluster_id = cluster_config.id;
+  clusters_.push(cluster);
+
+  RTC_LOG(LS_INFO) << "Probe cluster (bitrate:min bytes:min packets): ("
+                   << cluster.pace_info.send_bitrate_bps << ":"
+                   << cluster.pace_info.probe_cluster_min_bytes << ":"
+                   << cluster.pace_info.probe_cluster_min_probes << ")";
+
+  // If we are already probing, continue to do so. Otherwise set it to
+  // kInactive and wait for OnIncomingPacket to start the probing.
+  if (probing_state_ != ProbingState::kActive)
+    probing_state_ = ProbingState::kInactive;
+}
+
+Timestamp BitrateProber::NextProbeTime(Timestamp now) const {
+  // Probing is not active or probing is already complete.
+  if (probing_state_ != ProbingState::kActive || clusters_.empty()) {
+    return Timestamp::PlusInfinity();
+  }
+
+  return next_probe_time_;
+}
+
+absl::optional<PacedPacketInfo> BitrateProber::CurrentCluster(Timestamp now) {
+  if (clusters_.empty() || probing_state_ != ProbingState::kActive) {
+    return absl::nullopt;
+  }
+
+  if (next_probe_time_.IsFinite() &&
+      now - next_probe_time_ > config_.max_probe_delay.Get()) {
+    RTC_DLOG(LS_WARNING) << "Probe delay too high"
+                            " (next_ms:"
+                         << next_probe_time_.ms() << ", now_ms: " << now.ms()
+                         << "), discarding probe cluster.";
+    clusters_.pop();
+    if (clusters_.empty()) {
+      probing_state_ = ProbingState::kSuspended;
+      return absl::nullopt;
+    }
+  }
+
+  PacedPacketInfo info = clusters_.front().pace_info;
+  info.probe_cluster_bytes_sent = clusters_.front().sent_bytes;
+  return info;
+}
+
+DataSize BitrateProber::RecommendedMinProbeSize() const {
+  if (clusters_.empty()) {
+    return DataSize::Zero();
+  }
+  DataRate send_rate =
+      DataRate::BitsPerSec(clusters_.front().pace_info.send_bitrate_bps);
+  return send_rate * config_.min_probe_delta;
+}
+
+void BitrateProber::ProbeSent(Timestamp now, DataSize size) {
+  RTC_DCHECK(probing_state_ == ProbingState::kActive);
+  RTC_DCHECK(!size.IsZero());
+
+  if (!clusters_.empty()) {
+    ProbeCluster* cluster = &clusters_.front();
+    if (cluster->sent_probes == 0) {
+      RTC_DCHECK(cluster->started_at.IsInfinite());
+      cluster->started_at = now;
+    }
+    cluster->sent_bytes += size.bytes<int>();
+    cluster->sent_probes += 1;
+    next_probe_time_ = CalculateNextProbeTime(*cluster);
+    if (cluster->sent_bytes >= cluster->pace_info.probe_cluster_min_bytes &&
+        cluster->sent_probes >= cluster->pace_info.probe_cluster_min_probes) {
+      clusters_.pop();
+    }
+    if (clusters_.empty()) {
+      probing_state_ = ProbingState::kSuspended;
+    }
+  }
+}
+
+Timestamp BitrateProber::CalculateNextProbeTime(
+    const ProbeCluster& cluster) const {
+  RTC_CHECK_GT(cluster.pace_info.send_bitrate_bps, 0);
+  RTC_CHECK(cluster.started_at.IsFinite());
+
+  // Compute the time delta from the cluster start to ensure probe bitrate stays
+  // close to the target bitrate. Result is in milliseconds.
+  DataSize sent_bytes = DataSize::Bytes(cluster.sent_bytes);
+  DataRate send_bitrate =
+      DataRate::BitsPerSec(cluster.pace_info.send_bitrate_bps);
+
+  TimeDelta delta = sent_bytes / send_bitrate;
+  return cluster.started_at + delta;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/bitrate_prober.h b/third_party/libwebrtc/modules/pacing/bitrate_prober.h
new file mode 100644
index 0000000000..4d8ec68c4f
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/bitrate_prober.h
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PACING_BITRATE_PROBER_H_
+#define MODULES_PACING_BITRATE_PROBER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <queue>
+
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/network_types.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+class RtcEventLog;
+
+struct BitrateProberConfig {
+  explicit BitrateProberConfig(const FieldTrialsView* key_value_config);
+  BitrateProberConfig(const BitrateProberConfig&) = default;
+  BitrateProberConfig& operator=(const BitrateProberConfig&) = default;
+  ~BitrateProberConfig() = default;
+
+  // A minimum interval between probes to allow scheduling to be feasible.
+  FieldTrialParameter<TimeDelta> min_probe_delta;
+  // Maximum amount of time each probe can be delayed.
+  FieldTrialParameter<TimeDelta> max_probe_delay;
+  // This is used to start sending a probe after a large enough packet.
+  // The min packet size is scaled with the bitrate we're probing at.
+  // This defines the max min packet size, meaning that on high bitrates
+  // a packet of at least this size is needed to trigger sending a probe.
+  FieldTrialParameter<DataSize> min_packet_size;
+};
+
+// Note that this class isn't thread-safe by itself and therefore relies
+// on being protected by the caller.
+class BitrateProber {
+ public:
+  explicit BitrateProber(const FieldTrialsView& field_trials);
+  ~BitrateProber() = default;
+
+  void SetEnabled(bool enable);
+
+  // Returns true if the prober is in a probing session, i.e., it currently
+  // wants packets to be sent out according to the time returned by
+  // TimeUntilNextProbe().
+  bool is_probing() const { return probing_state_ == ProbingState::kActive; }
+
+  // Initializes a new probing session if the prober is allowed to probe. Does
+  // not initialize the prober unless the packet size is large enough to probe
+  // with.
+  void OnIncomingPacket(DataSize packet_size);
+
+  // Create a cluster used to probe.
+  void CreateProbeCluster(const ProbeClusterConfig& cluster_config);
+  // Returns the time at which the next probe should be sent to get accurate
+  // probing. If probing is not desired at this time, Timestamp::PlusInfinity()
+  // will be returned.
+  // TODO(bugs.webrtc.org/11780): Remove `now` argument when old mode is gone.
+  Timestamp NextProbeTime(Timestamp now) const;
+
+  // Information about the current probing cluster.
+  absl::optional<PacedPacketInfo> CurrentCluster(Timestamp now);
+
+  // Returns the minimum number of bytes that the prober recommends for
+  // the next probe, or zero if not probing. A probe can consist of multiple
+  // packets that are sent back to back.
+  DataSize RecommendedMinProbeSize() const;
+
+  // Called to report to the prober that a probe has been sent. In case of
+  // multiple packets per probe, this call would be made at the end of sending
+  // the last packet in probe. `size` is the total size of all packets in probe.
+  void ProbeSent(Timestamp now, DataSize size);
+
+ private:
+  enum class ProbingState {
+    // Probing will not be triggered in this state at all times.
+    kDisabled,
+    // Probing is enabled and ready to trigger on the first packet arrival.
+    kInactive,
+    // Probe cluster is filled with the set of data rates to be probed and
+    // probes are being sent.
+    kActive,
+    // Probing is enabled, but currently suspended until an explicit trigger
+    // to start probing again.
+    kSuspended,
+  };
+
+  // A probe cluster consists of a set of probes. Each probe in turn can be
+  // divided into a number of packets to accommodate the MTU on the network.
+  struct ProbeCluster {
+    PacedPacketInfo pace_info;
+
+    int sent_probes = 0;
+    int sent_bytes = 0;
+    Timestamp requested_at = Timestamp::MinusInfinity();
+    Timestamp started_at = Timestamp::MinusInfinity();
+    int retries = 0;
+  };
+
+  Timestamp CalculateNextProbeTime(const ProbeCluster& cluster) const;
+
+  ProbingState probing_state_;
+
+  // Probe bitrate per packet. These are used to compute the delta relative to
+  // the previous probe packet based on the size and time when that packet was
+  // sent.
+  std::queue<ProbeCluster> clusters_;
+
+  // Time the next probe should be sent when in kActive state.
+  Timestamp next_probe_time_;
+
+  BitrateProberConfig config_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_PACING_BITRATE_PROBER_H_
diff --git a/third_party/libwebrtc/modules/pacing/bitrate_prober_unittest.cc b/third_party/libwebrtc/modules/pacing/bitrate_prober_unittest.cc
new file mode 100644
index 0000000000..3be7d2d99e
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/bitrate_prober_unittest.cc
@@ -0,0 +1,371 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/bitrate_prober.h"
+
+#include <algorithm>
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(BitrateProberTest, VerifyStatesAndTimeBetweenProbes) {
+  const FieldTrialBasedConfig config;
+  BitrateProber prober(config);
+  EXPECT_FALSE(prober.is_probing());
+
+  Timestamp now = Timestamp::Zero();
+  const Timestamp start_time = now;
+  EXPECT_EQ(prober.NextProbeTime(now), Timestamp::PlusInfinity());
+
+  const DataRate kTestBitrate1 = DataRate::KilobitsPerSec(900);
+  const DataRate kTestBitrate2 = DataRate::KilobitsPerSec(1800);
+  const int kClusterSize = 5;
+  const DataSize kProbeSize = DataSize::Bytes(1000);
+  const TimeDelta kMinProbeDuration = TimeDelta::Millis(15);
+
+  prober.CreateProbeCluster({.at_time = now,
+                             .target_data_rate = kTestBitrate1,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  prober.CreateProbeCluster({.at_time = now,
+                             .target_data_rate = kTestBitrate2,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 1});
+  EXPECT_FALSE(prober.is_probing());
+
+  prober.OnIncomingPacket(kProbeSize);
+  EXPECT_TRUE(prober.is_probing());
+  EXPECT_EQ(0, prober.CurrentCluster(now)->probe_cluster_id);
+
+  // First packet should probe as soon as possible.
+  EXPECT_EQ(Timestamp::MinusInfinity(), prober.NextProbeTime(now));
+
+  for (int i = 0; i < kClusterSize; ++i) {
+    now = std::max(now, prober.NextProbeTime(now));
+    EXPECT_EQ(now, std::max(now, prober.NextProbeTime(now)));
+    EXPECT_EQ(0, prober.CurrentCluster(now)->probe_cluster_id);
+    prober.ProbeSent(now, kProbeSize);
+  }
+
+  EXPECT_GE(now - start_time, kMinProbeDuration);
+  // Verify that the actual bitrate is withing 10% of the target.
+  DataRate bitrate = kProbeSize * (kClusterSize - 1) / (now - start_time);
+  EXPECT_GT(bitrate, kTestBitrate1 * 0.9);
+  EXPECT_LT(bitrate, kTestBitrate1 * 1.1);
+
+  now = std::max(now, prober.NextProbeTime(now));
+  Timestamp probe2_started = now;
+
+  for (int i = 0; i < kClusterSize; ++i) {
+    now = std::max(now, prober.NextProbeTime(now));
+    EXPECT_EQ(now, std::max(now, prober.NextProbeTime(now)));
+    EXPECT_EQ(1, prober.CurrentCluster(now)->probe_cluster_id);
+    prober.ProbeSent(now, kProbeSize);
+  }
+
+  // Verify that the actual bitrate is withing 10% of the target.
+  TimeDelta duration = now - probe2_started;
+  EXPECT_GE(duration, kMinProbeDuration);
+  bitrate = (kProbeSize * (kClusterSize - 1)) / duration;
+  EXPECT_GT(bitrate, kTestBitrate2 * 0.9);
+  EXPECT_LT(bitrate, kTestBitrate2 * 1.1);
+
+  EXPECT_EQ(prober.NextProbeTime(now), Timestamp::PlusInfinity());
+  EXPECT_FALSE(prober.is_probing());
+}
+
+TEST(BitrateProberTest, DoesntProbeWithoutRecentPackets) {
+  const FieldTrialBasedConfig config;
+  BitrateProber prober(config);
+  const DataSize kProbeSize = DataSize::Bytes(1000);
+
+  Timestamp now = Timestamp::Zero();
+  EXPECT_EQ(prober.NextProbeTime(now), Timestamp::PlusInfinity());
+
+  prober.CreateProbeCluster({.at_time = now,
+                             .target_data_rate = DataRate::KilobitsPerSec(900),
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  EXPECT_FALSE(prober.is_probing());
+
+  prober.OnIncomingPacket(kProbeSize);
+  EXPECT_TRUE(prober.is_probing());
+  EXPECT_EQ(now, std::max(now, prober.NextProbeTime(now)));
+  prober.ProbeSent(now, kProbeSize);
+}
+
+TEST(BitrateProberTest, DiscardsDelayedProbes) {
+  const TimeDelta kMaxProbeDelay = TimeDelta::Millis(3);
+  const test::ExplicitKeyValueConfig trials(
+      "WebRTC-Bwe-ProbingBehavior/"
+      "abort_delayed_probes:1,"
+      "max_probe_delay:3ms/");
+  BitrateProber prober(trials);
+  const DataSize kProbeSize = DataSize::Bytes(1000);
+
+  Timestamp now = Timestamp::Zero();
+
+  // Add two probe clusters.
+  prober.CreateProbeCluster({.at_time = now,
+                             .target_data_rate = DataRate::KilobitsPerSec(900),
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+
+  prober.OnIncomingPacket(kProbeSize);
+  EXPECT_TRUE(prober.is_probing());
+  EXPECT_EQ(prober.CurrentCluster(now)->probe_cluster_id, 0);
+  // Advance to first probe time and indicate sent probe.
+  now = std::max(now, prober.NextProbeTime(now));
+  prober.ProbeSent(now, kProbeSize);
+
+  // Advance time 1ms past timeout for the next probe.
+  Timestamp next_probe_time = prober.NextProbeTime(now);
+  EXPECT_GT(next_probe_time, now);
+  now += next_probe_time - now + kMaxProbeDelay + TimeDelta::Millis(1);
+
+  // Still indicates the time we wanted to probe at.
+  EXPECT_EQ(prober.NextProbeTime(now), next_probe_time);
+  // First and only cluster removed due to timeout.
+  EXPECT_FALSE(prober.CurrentCluster(now).has_value());
+}
+
+TEST(BitrateProberTest, LimitsNumberOfPendingProbeClusters) {
+  const FieldTrialBasedConfig config;
+  BitrateProber prober(config);
+  const DataSize kProbeSize = DataSize::Bytes(1000);
+  Timestamp now = Timestamp::Zero();
+  prober.CreateProbeCluster({.at_time = now,
+                             .target_data_rate = DataRate::KilobitsPerSec(900),
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  prober.OnIncomingPacket(kProbeSize);
+  ASSERT_TRUE(prober.is_probing());
+  ASSERT_EQ(prober.CurrentCluster(now)->probe_cluster_id, 0);
+
+  for (int i = 1; i < 11; ++i) {
+    prober.CreateProbeCluster(
+        {.at_time = now,
+         .target_data_rate = DataRate::KilobitsPerSec(900),
+         .target_duration = TimeDelta::Millis(15),
+         .target_probe_count = 5,
+         .id = i});
+    prober.OnIncomingPacket(kProbeSize);
+  }
+  // Expect some clusters has been dropped.
+  EXPECT_TRUE(prober.is_probing());
+  EXPECT_GE(prober.CurrentCluster(now)->probe_cluster_id, 5);
+
+  Timestamp max_expected_probe_time = now + TimeDelta::Seconds(1);
+  while (prober.is_probing() && now < max_expected_probe_time) {
+    now = std::max(now, prober.NextProbeTime(now));
+    prober.ProbeSent(now, kProbeSize);
+  }
+  EXPECT_FALSE(prober.is_probing());
+}
+
+TEST(BitrateProberTest, DoesntInitializeProbingForSmallPackets) {
+  const FieldTrialBasedConfig config;
+  BitrateProber prober(config);
+  prober.SetEnabled(true);
+  ASSERT_FALSE(prober.is_probing());
+
+  prober.CreateProbeCluster({.at_time = Timestamp::Zero(),
+                             .target_data_rate = DataRate::KilobitsPerSec(1000),
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  prober.OnIncomingPacket(DataSize::Bytes(100));
+
+  EXPECT_FALSE(prober.is_probing());
+}
+
+TEST(BitrateProberTest, DoesInitializeProbingForSmallPacketsIfConfigured) {
+  const test::ExplicitKeyValueConfig config(
+      "WebRTC-Bwe-ProbingBehavior/"
+      "min_packet_size:0bytes/");
+  BitrateProber prober(config);
+  prober.SetEnabled(true);
+  ASSERT_FALSE(prober.is_probing());
+
+  prober.CreateProbeCluster({.at_time = Timestamp::Zero(),
+                             .target_data_rate = DataRate::KilobitsPerSec(1000),
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  prober.OnIncomingPacket(DataSize::Bytes(10));
+
+  EXPECT_TRUE(prober.is_probing());
+}
+
+TEST(BitrateProberTest, VerifyProbeSizeOnHighBitrate) {
+  const FieldTrialBasedConfig config;
+  BitrateProber prober(config);
+
+  const DataRate kHighBitrate = DataRate::KilobitsPerSec(10000);  // 10 Mbps
+
+  prober.CreateProbeCluster({.at_time = Timestamp::Zero(),
+                             .target_data_rate = kHighBitrate,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  // Probe size should ensure a minimum of 1 ms interval.
+  EXPECT_GT(prober.RecommendedMinProbeSize(),
+            kHighBitrate * TimeDelta::Millis(1));
+}
+
+TEST(BitrateProberTest, ProbeSizeCanBeSetWithFieldTrial) {
+  const test::ExplicitKeyValueConfig trials(
+      "WebRTC-Bwe-ProbingBehavior/min_probe_delta:20ms/");
+  BitrateProber prober(trials);
+  prober.SetEnabled(true);
+
+  const DataRate kHighBitrate = DataRate::KilobitsPerSec(10000);  // 10 Mbps
+
+  prober.CreateProbeCluster({.at_time = Timestamp::Zero(),
+                             .target_data_rate = kHighBitrate,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  EXPECT_EQ(prober.RecommendedMinProbeSize(),
+            kHighBitrate * TimeDelta::Millis(20));
+
+  prober.OnIncomingPacket(DataSize::Bytes(1000));
+  // Next time to send probe should be "min_probe_delta" if the recommended
+  // number of bytes has been sent.
+  prober.ProbeSent(Timestamp::Zero(), prober.RecommendedMinProbeSize());
+  EXPECT_EQ(prober.NextProbeTime(Timestamp::Zero()),
+            Timestamp::Zero() + TimeDelta::Millis(20));
+}
+
+TEST(BitrateProberTest, MinumumNumberOfProbingPackets) {
+  const FieldTrialBasedConfig config;
+  BitrateProber prober(config);
+  // Even when probing at a low bitrate we expect a minimum number
+  // of packets to be sent.
+  const DataRate kBitrate = DataRate::KilobitsPerSec(100);
+  const DataSize kPacketSize = DataSize::Bytes(1000);
+
+  Timestamp now = Timestamp::Zero();
+  prober.CreateProbeCluster({.at_time = Timestamp::Zero(),
+                             .target_data_rate = kBitrate,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+
+  prober.OnIncomingPacket(kPacketSize);
+  for (int i = 0; i < 5; ++i) {
+    EXPECT_TRUE(prober.is_probing());
+    prober.ProbeSent(now, kPacketSize);
+  }
+
+  EXPECT_FALSE(prober.is_probing());
+}
+
+TEST(BitrateProberTest, ScaleBytesUsedForProbing) {
+  const FieldTrialBasedConfig config;
+  BitrateProber prober(config);
+  const DataRate kBitrate = DataRate::KilobitsPerSec(10000);  // 10 Mbps.
+  const DataSize kPacketSize = DataSize::Bytes(1000);
+  const DataSize kExpectedDataSent = kBitrate * TimeDelta::Millis(15);
+
+  Timestamp now = Timestamp::Zero();
+  prober.CreateProbeCluster({.at_time = Timestamp::Zero(),
+                             .target_data_rate = kBitrate,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  prober.OnIncomingPacket(kPacketSize);
+  DataSize data_sent = DataSize::Zero();
+  while (data_sent < kExpectedDataSent) {
+    ASSERT_TRUE(prober.is_probing());
+    prober.ProbeSent(now, kPacketSize);
+    data_sent += kPacketSize;
+  }
+
+  EXPECT_FALSE(prober.is_probing());
+}
+
+TEST(BitrateProberTest, HighBitrateProbing) {
+  const FieldTrialBasedConfig config;
+  BitrateProber prober(config);
+  const DataRate kBitrate = DataRate::KilobitsPerSec(1000000);  // 1 Gbps.
+  const DataSize kPacketSize = DataSize::Bytes(1000);
+  const DataSize kExpectedDataSent = kBitrate * TimeDelta::Millis(15);
+
+  Timestamp now = Timestamp::Zero();
+  prober.CreateProbeCluster({.at_time = Timestamp::Zero(),
+                             .target_data_rate = kBitrate,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  prober.OnIncomingPacket(kPacketSize);
+  DataSize data_sent = DataSize::Zero();
+  while (data_sent < kExpectedDataSent) {
+    ASSERT_TRUE(prober.is_probing());
+    prober.ProbeSent(now, kPacketSize);
+    data_sent += kPacketSize;
+  }
+
+  EXPECT_FALSE(prober.is_probing());
+}
+
+TEST(BitrateProberTest, ProbeClusterTimeout) {
+  const FieldTrialBasedConfig config;
+  BitrateProber prober(config);
+  const DataRate kBitrate = DataRate::KilobitsPerSec(300);
+  const DataSize kSmallPacketSize = DataSize::Bytes(20);
+  // Expecting two probe clusters of 5 packets each.
+  const DataSize kExpectedDataSent = kSmallPacketSize * 2 * 5;
+  const TimeDelta kTimeout = TimeDelta::Millis(5000);
+
+  Timestamp now = Timestamp::Zero();
+  prober.CreateProbeCluster({.at_time = now,
+                             .target_data_rate = kBitrate,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 0});
+  prober.OnIncomingPacket(kSmallPacketSize);
+  EXPECT_FALSE(prober.is_probing());
+  now += kTimeout;
+  prober.CreateProbeCluster({.at_time = now,
+                             .target_data_rate = kBitrate / 10,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 1});
+  prober.OnIncomingPacket(kSmallPacketSize);
+  EXPECT_FALSE(prober.is_probing());
+  now += TimeDelta::Millis(1);
+  prober.CreateProbeCluster({.at_time = now,
+                             .target_data_rate = kBitrate / 10,
+                             .target_duration = TimeDelta::Millis(15),
+                             .target_probe_count = 5,
+                             .id = 2});
+  prober.OnIncomingPacket(kSmallPacketSize);
+  EXPECT_TRUE(prober.is_probing());
+  DataSize data_sent = DataSize::Zero();
+  while (data_sent < kExpectedDataSent) {
+    ASSERT_TRUE(prober.is_probing());
+    prober.ProbeSent(now, kSmallPacketSize);
+    data_sent += kSmallPacketSize;
+  }
+
+  EXPECT_FALSE(prober.is_probing());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/g3doc/index.md b/third_party/libwebrtc/modules/pacing/g3doc/index.md
new file mode 100644
index 0000000000..69f1e69513
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/g3doc/index.md
@@ -0,0 +1,164 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'sprang' reviewed: '2021-04-12'} *-->
+
+# Paced Sending
+
+The paced sender, often referred to as just the "pacer", is a part of the WebRTC
+RTP stack used primarily to smooth the flow of packets sent onto the network.
+
+## Background
+
+Consider a video stream at 5Mbps and 30fps. This would in an ideal world result
+in each frame being ~21kB large and packetized into 18 RTP packets. While the
+average bitrate over say a one second sliding window would be a correct 5Mbps,
+on a shorter time scale it can be seen as a burst of 167Mbps every 33ms, each
+followed by a 32ms silent period. Further, it is quite common that video
+encoders overshoot the target frame size in case of sudden movement especially
+dealing with screensharing. Frames being 10x or even 100x larger than the ideal
+size is an all too real scenario. These packet bursts can cause several issues,
+such as congesting networks and causing buffer bloat or even packet loss. Most
+sessions have more than one media stream, e.g. a video and an audio track. If
+you put a frame on the wire in one go, and those packets take 100ms to reach the
+other side - that means you have now blocked any audio packets from reaching the
+remote end in time as well.
+
+The paced sender solves this by having a buffer in which media is queued, and
+then using a _leaky bucket_ algorithm to pace them onto the network. The buffer
+contains separate fifo streams for all media tracks so that e.g. audio can be
+prioritized over video - and equal prio streams can be sent in a round-robin
+fashion to avoid any one stream blocking others.
+
+Since the pacer is in control of the bitrate sent on the wire, it is also used
+to generate padding in cases where a minimum send rate is required - and to
+generate packet trains if bitrate probing is used.
+
+## Life of a Packet
+
+The typical path for media packets when using the paced sender looks something
+like this:
+
+1.  `RTPSenderVideo` or `RTPSenderAudio` packetizes media into RTP packets.
+2.  The packets are sent to the [RTPSender] class for transmission.
+3.  The pacer is called via [RtpPacketSender] interface to enqueue the packet
+    batch.
+4.  The packets are put into a queue within the pacer awaiting opportune moments
+    to send them.
+5.  At a calculated time, the pacer calls the `PacingController::PacketSender()`
+    callback method, normally implemented by the [PacketRouter] class.
+6.  The router forwards the packet to the correct RTP module based on the
+    packet's SSRC, and in which the `RTPSenderEgress` class makes final time
+    stamping, potentially records it for retransmissions etc.
+7.  The packet is sent to the low-level `Transport` interface, after which it is
+    now out of scope.
+
+Asynchronously to this, the estimated available send bandwidth is determined -
+and the target send rate is set on the `RtpPacketPacer` via the `void
+SetPacingRates(DataRate pacing_rate, DataRate padding_rate)` method.
+
+## Packet Prioritization
+
+The pacer prioritized packets based on two criteria:
+
+*   Packet type, with most to least prioritized:
+    1.  Audio
+    2.  Retransmissions
+    3.  Video and FEC
+    4.  Padding
+*   Enqueue order
+
+The enqueue order is enforced on a per stream (SSRC) basis. Given equal
+priority, the [RoundRobinPacketQueue] alternates between media streams to ensure
+no stream needlessly blocks others.
+
+## Implementations
+
+The main class to use is called [TaskQueuePacedSender]. It uses a task queue to
+manage thread safety and schedule delayed tasks, but delegates most of the actual
+work to the `PacingController` class.
+This way, it's possible to develop a custom pacer with different scheduling
+mechanism - but ratain the same pacing logic.
+
+## The Packet Router
+
+An adjacent component called [PacketRouter] is used to route packets coming out
+of the pacer and into the correct RTP module. It has the following functions:
+
+*   The `SendPacket` method looks up an RTP module with an SSRC corresponding to
+    the packet for further routing to the network.
+*   If send-side bandwidth estimation is used, it populates the transport-wide
+    sequence number extension.
+*   Generate padding. Modules supporting payload-based padding are prioritized,
+    with the last module to have sent media always being the first choice.
+*   Returns any generated FEC after having sent media.
+*   Forwards REMB and/or TransportFeedback messages to suitable RTP modules.
+
+At present the FEC is generated on a per SSRC basis, so is always returned from
+an RTP module after sending media. Hopefully one day we will support covering
+multiple streams with a single FlexFEC stream - and the packet router is the
+likely place for that FEC generator to live. It may even be used for FEC padding
+as an alternative to RTX.
+
+## The API
+
+The section outlines the classes and methods relevant to a few different use
+cases of the pacer.
+
+### Packet sending
+
+For sending packets, use
+`RtpPacketSender::EnqueuePackets(std::vector<std::unique_ptr<RtpPacketToSend>>
+packets)` The pacer takes a `PacingController::PacketSender` as constructor
+argument, this callback is used when it's time to actually send packets.
+
+### Send rates
+
+To control the send rate, use `void SetPacingRates(DataRate pacing_rate,
+DataRate padding_rate)` If the packet queue becomes empty and the send rate
+drops below `padding_rate`, the pacer will request padding packets from the
+`PacketRouter`.
+
+In order to completely suspend/resume sending data (e.g. due to network
+availability), use the `Pause()` and `Resume()` methods.
+
+The specified pacing rate may be overriden in some cases, e.g. due to extreme
+encoder overshoot. Use `void SetQueueTimeLimit(TimeDelta limit)` to specify the
+longest time you want packets to spend waiting in the pacer queue (pausing
+excluded). The actual send rate may then be increased past the pacing_rate to
+try to make the _average_ queue time less than that requested limit. The
+rationale for this is that if the send queue is say longer than three seconds,
+it's better to risk packet loss and then try to recover using a key-frame rather
+than cause severe delays.
+
+### Bandwidth estimation
+
+If the bandwidth estimator supports bandwidth probing, it may request a cluster
+of packets to be sent at a specified rate in order to gauge if this causes
+increased delay/loss on the network. Use the `void CreateProbeCluster(...)`
+method - packets sent via this `PacketRouter` will be marked with the
+corresponding cluster_id in the attached `PacedPacketInfo` struct.
+
+If congestion window pushback is used, the state can be updated using
+`SetCongestionWindow()` and `UpdateOutstandingData()`.
+
+A few more methods control how we pace: * `SetAccountForAudioPackets()`
+determines if audio packets count into bandwidth consumed. *
+`SetIncludeOverhead()` determines if the entire RTP packet size counts into
+bandwidth used (otherwise just media payload). * `SetTransportOverhead()` sets
+an additional data size consumed per packet, representing e.g. UDP/IP headers.
+
+### Stats
+
+Several methods are used to gather statistics in pacer state:
+
+*   `OldestPacketWaitTime()` time since the oldest packet in the queue was
+    added.
+*   `QueueSizeData()` total bytes currently in the queue.
+*   `FirstSentPacketTime()` absolute time the first packet was sent.
+*   `ExpectedQueueTime()` total bytes in the queue divided by the send rate.
+
+[RTPSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.h;drc=77ee8542dd35d5143b5788ddf47fb7cdb96eb08e
+[RtpPacketSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/rtp_rtcp/include/rtp_packet_sender.h;drc=ea55b0872f14faab23a4e5dbcb6956369c8ed5dc
+[RtpPacketPacer]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/rtp_packet_pacer.h;drc=e7bc3a347760023dd4840cf6ebdd1e6c8592f4d7
+[PacketRouter]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/packet_router.h;drc=3d2210876e31d0bb5c7de88b27fd02ceb1f4e03e
+[TaskQueuePacedSender]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/task_queue_paced_sender.h;drc=5051693ada61bc7b78855c6fb3fa87a0394fa813
+[RoundRobinPacketQueue]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/pacing/round_robin_packet_queue.h;drc=b571ff48f8fe07678da5a854cd6c3f5dde02855f
diff --git a/third_party/libwebrtc/modules/pacing/interval_budget.cc b/third_party/libwebrtc/modules/pacing/interval_budget.cc
new file mode 100644
index 0000000000..321ca46be4
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/interval_budget.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/interval_budget.h"
+
+#include <algorithm>
+
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace {
+constexpr int64_t kWindowMs = 500;
+}
+
+IntervalBudget::IntervalBudget(int initial_target_rate_kbps)
+    : IntervalBudget(initial_target_rate_kbps, false) {}
+
+IntervalBudget::IntervalBudget(int initial_target_rate_kbps,
+                               bool can_build_up_underuse)
+    : bytes_remaining_(0), can_build_up_underuse_(can_build_up_underuse) {
+  set_target_rate_kbps(initial_target_rate_kbps);
+}
+
+void IntervalBudget::set_target_rate_kbps(int target_rate_kbps) {
+  target_rate_kbps_ = target_rate_kbps;
+  max_bytes_in_budget_ = (kWindowMs * target_rate_kbps_) / 8;
+  bytes_remaining_ = std::min(std::max(-max_bytes_in_budget_, bytes_remaining_),
+                              max_bytes_in_budget_);
+}
+
+void IntervalBudget::IncreaseBudget(int64_t delta_time_ms) {
+  int64_t bytes = target_rate_kbps_ * delta_time_ms / 8;
+  if (bytes_remaining_ < 0 || can_build_up_underuse_) {
+    // We overused last interval, compensate this interval.
+    bytes_remaining_ = std::min(bytes_remaining_ + bytes, max_bytes_in_budget_);
+  } else {
+    // If we underused last interval we can't use it this interval.
+    bytes_remaining_ = std::min(bytes, max_bytes_in_budget_);
+  }
+}
+
+void IntervalBudget::UseBudget(size_t bytes) {
+  bytes_remaining_ = std::max(bytes_remaining_ - static_cast<int>(bytes),
+                              -max_bytes_in_budget_);
+}
+
+size_t IntervalBudget::bytes_remaining() const {
+  return rtc::saturated_cast<size_t>(std::max<int64_t>(0, bytes_remaining_));
+}
+
+double IntervalBudget::budget_ratio() const {
+  if (max_bytes_in_budget_ == 0)
+    return 0.0;
+  return static_cast<double>(bytes_remaining_) / max_bytes_in_budget_;
+}
+
+int IntervalBudget::target_rate_kbps() const {
+  return target_rate_kbps_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/interval_budget.h b/third_party/libwebrtc/modules/pacing/interval_budget.h
new file mode 100644
index 0000000000..faeb1d8fc4
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/interval_budget.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PACING_INTERVAL_BUDGET_H_
+#define MODULES_PACING_INTERVAL_BUDGET_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// TODO(tschumim): Reflector IntervalBudget so that we can set a under- and
+// over-use budget in ms.
+class IntervalBudget {
+ public:
+  explicit IntervalBudget(int initial_target_rate_kbps);
+  IntervalBudget(int initial_target_rate_kbps, bool can_build_up_underuse);
+  void set_target_rate_kbps(int target_rate_kbps);
+
+  // TODO(tschumim): Unify IncreaseBudget and UseBudget to one function.
+  void IncreaseBudget(int64_t delta_time_ms);
+  void UseBudget(size_t bytes);
+
+  size_t bytes_remaining() const;
+  double budget_ratio() const;
+  int target_rate_kbps() const;
+
+ private:
+  int target_rate_kbps_;
+  int64_t max_bytes_in_budget_;
+  int64_t bytes_remaining_;
+  bool can_build_up_underuse_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_PACING_INTERVAL_BUDGET_H_
diff --git a/third_party/libwebrtc/modules/pacing/interval_budget_gn/moz.build b/third_party/libwebrtc/modules/pacing/interval_budget_gn/moz.build
new file mode 100644
index 0000000000..17d4371d72
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/interval_budget_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/pacing/interval_budget.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("interval_budget_gn")
diff --git a/third_party/libwebrtc/modules/pacing/interval_budget_unittest.cc b/third_party/libwebrtc/modules/pacing/interval_budget_unittest.cc
new file mode 100644
index 0000000000..e182d35510
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/interval_budget_unittest.cc
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/interval_budget.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kWindowMs = 500;
+constexpr int kBitrateKbps = 100;
+constexpr bool kCanBuildUpUnderuse = true;
+constexpr bool kCanNotBuildUpUnderuse = false;
+size_t TimeToBytes(int bitrate_kbps, int time_ms) {
+  return static_cast<size_t>(bitrate_kbps * time_ms / 8);
+}
+}  // namespace
+
+TEST(IntervalBudgetTest, InitailState) {
+  IntervalBudget interval_budget(kBitrateKbps);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(), 0.0);
+  EXPECT_EQ(interval_budget.bytes_remaining(), 0u);
+}
+
+TEST(IntervalBudgetTest, Underuse) {
+  IntervalBudget interval_budget(kBitrateKbps);
+  int delta_time_ms = 50;
+  interval_budget.IncreaseBudget(delta_time_ms);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(),
+                   kWindowMs / static_cast<double>(100 * delta_time_ms));
+  EXPECT_EQ(interval_budget.bytes_remaining(),
+            TimeToBytes(kBitrateKbps, delta_time_ms));
+}
+
+TEST(IntervalBudgetTest, DontUnderuseMoreThanMaxWindow) {
+  IntervalBudget interval_budget(kBitrateKbps);
+  int delta_time_ms = 1000;
+  interval_budget.IncreaseBudget(delta_time_ms);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(), 1.0);
+  EXPECT_EQ(interval_budget.bytes_remaining(),
+            TimeToBytes(kBitrateKbps, kWindowMs));
+}
+
+TEST(IntervalBudgetTest, DontUnderuseMoreThanMaxWindowWhenChangeBitrate) {
+  IntervalBudget interval_budget(kBitrateKbps);
+  int delta_time_ms = kWindowMs / 2;
+  interval_budget.IncreaseBudget(delta_time_ms);
+  interval_budget.set_target_rate_kbps(kBitrateKbps / 10);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(), 1.0);
+  EXPECT_EQ(interval_budget.bytes_remaining(),
+            TimeToBytes(kBitrateKbps / 10, kWindowMs));
+}
+
+TEST(IntervalBudgetTest, BalanceChangeOnBitrateChange) {
+  IntervalBudget interval_budget(kBitrateKbps);
+  int delta_time_ms = kWindowMs;
+  interval_budget.IncreaseBudget(delta_time_ms);
+  interval_budget.set_target_rate_kbps(kBitrateKbps * 2);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(), 0.5);
+  EXPECT_EQ(interval_budget.bytes_remaining(),
+            TimeToBytes(kBitrateKbps, kWindowMs));
+}
+
+TEST(IntervalBudgetTest, Overuse) {
+  IntervalBudget interval_budget(kBitrateKbps);
+  int overuse_time_ms = 50;
+  int used_bytes = TimeToBytes(kBitrateKbps, overuse_time_ms);
+  interval_budget.UseBudget(used_bytes);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(),
+                   -kWindowMs / static_cast<double>(100 * overuse_time_ms));
+  EXPECT_EQ(interval_budget.bytes_remaining(), 0u);
+}
+
+TEST(IntervalBudgetTest, DontOveruseMoreThanMaxWindow) {
+  IntervalBudget interval_budget(kBitrateKbps);
+  int overuse_time_ms = 1000;
+  int used_bytes = TimeToBytes(kBitrateKbps, overuse_time_ms);
+  interval_budget.UseBudget(used_bytes);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(), -1.0);
+  EXPECT_EQ(interval_budget.bytes_remaining(), 0u);
+}
+
+TEST(IntervalBudgetTest, CanBuildUpUnderuseWhenConfigured) {
+  IntervalBudget interval_budget(kBitrateKbps, kCanBuildUpUnderuse);
+  int delta_time_ms = 50;
+  interval_budget.IncreaseBudget(delta_time_ms);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(),
+                   kWindowMs / static_cast<double>(100 * delta_time_ms));
+  EXPECT_EQ(interval_budget.bytes_remaining(),
+            TimeToBytes(kBitrateKbps, delta_time_ms));
+
+  interval_budget.IncreaseBudget(delta_time_ms);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(),
+                   2 * kWindowMs / static_cast<double>(100 * delta_time_ms));
+  EXPECT_EQ(interval_budget.bytes_remaining(),
+            TimeToBytes(kBitrateKbps, 2 * delta_time_ms));
+}
+
+TEST(IntervalBudgetTest, CanNotBuildUpUnderuseWhenConfigured) {
+  IntervalBudget interval_budget(kBitrateKbps, kCanNotBuildUpUnderuse);
+  int delta_time_ms = 50;
+  interval_budget.IncreaseBudget(delta_time_ms);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(),
+                   kWindowMs / static_cast<double>(100 * delta_time_ms));
+  EXPECT_EQ(interval_budget.bytes_remaining(),
+            TimeToBytes(kBitrateKbps, delta_time_ms));
+
+  interval_budget.IncreaseBudget(delta_time_ms);
+  EXPECT_DOUBLE_EQ(interval_budget.budget_ratio(),
+                   kWindowMs / static_cast<double>(100 * delta_time_ms));
+  EXPECT_EQ(interval_budget.bytes_remaining(),
+            TimeToBytes(kBitrateKbps, delta_time_ms));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.cc b/third_party/libwebrtc/modules/pacing/pacing_controller.cc
new file mode 100644
index 0000000000..3ba7598cc2
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/pacing_controller.cc
@@ -0,0 +1,710 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/pacing_controller.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/pacing/bitrate_prober.h"
+#include "modules/pacing/interval_budget.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+// Time limit in milliseconds between packet bursts.
+constexpr TimeDelta kDefaultMinPacketLimit = TimeDelta::Millis(5);
+constexpr TimeDelta kCongestedPacketInterval = TimeDelta::Millis(500);
+// TODO(sprang): Consider dropping this limit.
+// The maximum debt level, in terms of time, capped when sending packets.
+constexpr TimeDelta kMaxDebtInTime = TimeDelta::Millis(500);
+constexpr TimeDelta kMaxElapsedTime = TimeDelta::Seconds(2);
+
+bool IsDisabled(const FieldTrialsView& field_trials, absl::string_view key) {
+  return absl::StartsWith(field_trials.Lookup(key), "Disabled");
+}
+
+bool IsEnabled(const FieldTrialsView& field_trials, absl::string_view key) {
+  return absl::StartsWith(field_trials.Lookup(key), "Enabled");
+}
+
+}  // namespace
+
+const TimeDelta PacingController::kMaxExpectedQueueLength =
+    TimeDelta::Millis(2000);
+const TimeDelta PacingController::kPausedProcessInterval =
+    kCongestedPacketInterval;
+const TimeDelta PacingController::kMinSleepTime = TimeDelta::Millis(1);
+const TimeDelta PacingController::kTargetPaddingDuration = TimeDelta::Millis(5);
+const TimeDelta PacingController::kMaxPaddingReplayDuration =
+    TimeDelta::Millis(50);
+const TimeDelta PacingController::kMaxEarlyProbeProcessing =
+    TimeDelta::Millis(1);
+
+PacingController::PacingController(Clock* clock,
+                                   PacketSender* packet_sender,
+                                   const FieldTrialsView& field_trials)
+    : clock_(clock),
+      packet_sender_(packet_sender),
+      field_trials_(field_trials),
+      drain_large_queues_(
+          !IsDisabled(field_trials_, "WebRTC-Pacer-DrainQueue")),
+      send_padding_if_silent_(
+          IsEnabled(field_trials_, "WebRTC-Pacer-PadInSilence")),
+      pace_audio_(IsEnabled(field_trials_, "WebRTC-Pacer-BlockAudio")),
+      ignore_transport_overhead_(
+          IsEnabled(field_trials_, "WebRTC-Pacer-IgnoreTransportOverhead")),
+      fast_retransmissions_(
+          IsEnabled(field_trials_, "WebRTC-Pacer-FastRetransmissions")),
+      min_packet_limit_(kDefaultMinPacketLimit),
+      transport_overhead_per_packet_(DataSize::Zero()),
+      send_burst_interval_(TimeDelta::Zero()),
+      last_timestamp_(clock_->CurrentTime()),
+      paused_(false),
+      media_debt_(DataSize::Zero()),
+      padding_debt_(DataSize::Zero()),
+      pacing_rate_(DataRate::Zero()),
+      adjusted_media_rate_(DataRate::Zero()),
+      padding_rate_(DataRate::Zero()),
+      prober_(field_trials_),
+      probing_send_failure_(false),
+      last_process_time_(clock->CurrentTime()),
+      last_send_time_(last_process_time_),
+      seen_first_packet_(false),
+      packet_queue_(/*creation_time=*/last_process_time_),
+      congested_(false),
+      queue_time_limit_(kMaxExpectedQueueLength),
+      account_for_audio_(false),
+      include_overhead_(false),
+      circuit_breaker_threshold_(1 << 16) {
+  if (!drain_large_queues_) {
+    RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
+                           "pushback experiment must be enabled.";
+  }
+  FieldTrialParameter<int> min_packet_limit_ms("", min_packet_limit_.ms());
+  ParseFieldTrial({&min_packet_limit_ms},
+                  field_trials_.Lookup("WebRTC-Pacer-MinPacketLimitMs"));
+  min_packet_limit_ = TimeDelta::Millis(min_packet_limit_ms.Get());
+  UpdateBudgetWithElapsedTime(min_packet_limit_);
+}
+
+PacingController::~PacingController() = default;
+
+void PacingController::CreateProbeClusters(
+    rtc::ArrayView<const ProbeClusterConfig> probe_cluster_configs) {
+  for (const ProbeClusterConfig probe_cluster_config : probe_cluster_configs) {
+    prober_.CreateProbeCluster(probe_cluster_config);
+  }
+}
+
+void PacingController::Pause() {
+  if (!paused_)
+    RTC_LOG(LS_INFO) << "PacedSender paused.";
+  paused_ = true;
+  packet_queue_.SetPauseState(true, CurrentTime());
+}
+
+void PacingController::Resume() {
+  if (paused_)
+    RTC_LOG(LS_INFO) << "PacedSender resumed.";
+  paused_ = false;
+  packet_queue_.SetPauseState(false, CurrentTime());
+}
+
+bool PacingController::IsPaused() const {
+  return paused_;
+}
+
+void PacingController::SetCongested(bool congested) {
+  if (congested_ && !congested) {
+    UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(CurrentTime()));
+  }
+  congested_ = congested;
+}
+
+void PacingController::SetCircuitBreakerThreshold(int num_iterations) {
+  circuit_breaker_threshold_ = num_iterations;
+}
+
+void PacingController::RemovePacketsForSsrc(uint32_t ssrc) {
+  packet_queue_.RemovePacketsForSsrc(ssrc);
+}
+
+bool PacingController::IsProbing() const {
+  return prober_.is_probing();
+}
+
+Timestamp PacingController::CurrentTime() const {
+  Timestamp time = clock_->CurrentTime();
+  if (time < last_timestamp_) {
+    RTC_LOG(LS_WARNING)
+        << "Non-monotonic clock behavior observed. Previous timestamp: "
+        << last_timestamp_.ms() << ", new timestamp: " << time.ms();
+    RTC_DCHECK_GE(time, last_timestamp_);
+    time = last_timestamp_;
+  }
+  last_timestamp_ = time;
+  return time;
+}
+
+void PacingController::SetProbingEnabled(bool enabled) {
+  RTC_CHECK(!seen_first_packet_);
+  prober_.SetEnabled(enabled);
+}
+
+void PacingController::SetPacingRates(DataRate pacing_rate,
+                                      DataRate padding_rate) {
+  static constexpr DataRate kMaxRate = DataRate::KilobitsPerSec(100'000);
+  RTC_CHECK_GT(pacing_rate, DataRate::Zero());
+  RTC_CHECK_GE(padding_rate, DataRate::Zero());
+  if (padding_rate > pacing_rate) {
+    RTC_LOG(LS_WARNING) << "Padding rate " << padding_rate.kbps()
+                        << "kbps is higher than the pacing rate "
+                        << pacing_rate.kbps() << "kbps, capping.";
+    padding_rate = pacing_rate;
+  }
+
+  if (pacing_rate > kMaxRate || padding_rate > kMaxRate) {
+    RTC_LOG(LS_WARNING) << "Very high pacing rates ( > " << kMaxRate.kbps()
+                        << " kbps) configured: pacing = " << pacing_rate.kbps()
+                        << " kbps, padding = " << padding_rate.kbps()
+                        << " kbps.";
+  }
+  pacing_rate_ = pacing_rate;
+  padding_rate_ = padding_rate;
+  MaybeUpdateMediaRateDueToLongQueue(CurrentTime());
+
+  RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps=" << pacing_rate_.kbps()
+                      << " padding_budget_kbps=" << padding_rate.kbps();
+}
+
+void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
+  RTC_DCHECK(pacing_rate_ > DataRate::Zero())
+      << "SetPacingRate must be called before InsertPacket.";
+  RTC_CHECK(packet->packet_type());
+
+  prober_.OnIncomingPacket(DataSize::Bytes(packet->payload_size()));
+
+  const Timestamp now = CurrentTime();
+  if (packet_queue_.Empty()) {
+    // If queue is empty, we need to "fast-forward" the last process time,
+    // so that we don't use passed time as budget for sending the first new
+    // packet.
+    Timestamp target_process_time = now;
+    Timestamp next_send_time = NextSendTime();
+    if (next_send_time.IsFinite()) {
+      // There was already a valid planned send time, such as a keep-alive.
+      // Use that as last process time only if it's prior to now.
+      target_process_time = std::min(now, next_send_time);
+    }
+    UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(target_process_time));
+  }
+  packet_queue_.Push(now, std::move(packet));
+  seen_first_packet_ = true;
+
+  // Queue length has increased, check if we need to change the pacing rate.
+  MaybeUpdateMediaRateDueToLongQueue(now);
+}
+
+void PacingController::SetAccountForAudioPackets(bool account_for_audio) {
+  account_for_audio_ = account_for_audio;
+}
+
+void PacingController::SetIncludeOverhead() {
+  include_overhead_ = true;
+}
+
+void PacingController::SetTransportOverhead(DataSize overhead_per_packet) {
+  if (ignore_transport_overhead_)
+    return;
+  transport_overhead_per_packet_ = overhead_per_packet;
+}
+
+void PacingController::SetSendBurstInterval(TimeDelta burst_interval) {
+  send_burst_interval_ = burst_interval;
+}
+
+TimeDelta PacingController::ExpectedQueueTime() const {
+  RTC_DCHECK_GT(adjusted_media_rate_, DataRate::Zero());
+  return QueueSizeData() / adjusted_media_rate_;
+}
+
+size_t PacingController::QueueSizePackets() const {
+  return rtc::checked_cast<size_t>(packet_queue_.SizeInPackets());
+}
+
+const std::array<int, kNumMediaTypes>&
+PacingController::SizeInPacketsPerRtpPacketMediaType() const {
+  return packet_queue_.SizeInPacketsPerRtpPacketMediaType();
+}
+
+DataSize PacingController::QueueSizeData() const {
+  DataSize size = packet_queue_.SizeInPayloadBytes();
+  if (include_overhead_) {
+    size += static_cast<int64_t>(packet_queue_.SizeInPackets()) *
+            transport_overhead_per_packet_;
+  }
+  return size;
+}
+
+DataSize PacingController::CurrentBufferLevel() const {
+  return std::max(media_debt_, padding_debt_);
+}
+
+absl::optional<Timestamp> PacingController::FirstSentPacketTime() const {
+  return first_sent_packet_time_;
+}
+
+Timestamp PacingController::OldestPacketEnqueueTime() const {
+  return packet_queue_.OldestEnqueueTime();
+}
+
+TimeDelta PacingController::UpdateTimeAndGetElapsed(Timestamp now) {
+  // If no previous processing, or last process was "in the future" because of
+  // early probe processing, then there is no elapsed time to add budget for.
+  if (last_process_time_.IsMinusInfinity() || now < last_process_time_) {
+    return TimeDelta::Zero();
+  }
+  TimeDelta elapsed_time = now - last_process_time_;
+  last_process_time_ = now;
+  if (elapsed_time > kMaxElapsedTime) {
+    RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time.ms()
+                        << " ms) longer than expected, limiting to "
+                        << kMaxElapsedTime.ms();
+    elapsed_time = kMaxElapsedTime;
+  }
+  return elapsed_time;
+}
+
+bool PacingController::ShouldSendKeepalive(Timestamp now) const {
+  if (send_padding_if_silent_ || paused_ || congested_ || !seen_first_packet_) {
+    // We send a padding packet every 500 ms to ensure we won't get stuck in
+    // congested state due to no feedback being received.
+    if (now - last_send_time_ >= kCongestedPacketInterval) {
+      return true;
+    }
+  }
+  return false;
+}
+
+Timestamp PacingController::NextSendTime() const {
+  const Timestamp now = CurrentTime();
+  Timestamp next_send_time = Timestamp::PlusInfinity();
+
+  if (paused_) {
+    return last_send_time_ + kPausedProcessInterval;
+  }
+
+  // If probing is active, that always takes priority.
+  if (prober_.is_probing() && !probing_send_failure_) {
+    Timestamp probe_time = prober_.NextProbeTime(now);
+    if (!probe_time.IsPlusInfinity()) {
+      return probe_time.IsMinusInfinity() ? now : probe_time;
+    }
+  }
+
+  // If queue contains a packet which should not be paced, its target send time
+  // is the time at which it was enqueued.
+  Timestamp unpaced_send_time = NextUnpacedSendTime();
+  if (unpaced_send_time.IsFinite()) {
+    return unpaced_send_time;
+  }
+
+  if (congested_ || !seen_first_packet_) {
+    // We need to at least send keep-alive packets with some interval.
+    return last_send_time_ + kCongestedPacketInterval;
+  }
+
+  if (adjusted_media_rate_ > DataRate::Zero() && !packet_queue_.Empty()) {
+    // If packets are allowed to be sent in a burst, the
+    // debt is allowed to grow up to one packet more than what can be sent
+    // during 'send_burst_period_'.
+    TimeDelta drain_time = media_debt_ / adjusted_media_rate_;
+    next_send_time =
+        last_process_time_ +
+        ((send_burst_interval_ > drain_time) ? TimeDelta::Zero() : drain_time);
+  } else if (padding_rate_ > DataRate::Zero() && packet_queue_.Empty()) {
+    // If we _don't_ have pending packets, check how long until we have
+    // bandwidth for padding packets. Both media and padding debts must
+    // have been drained to do this.
+    RTC_DCHECK_GT(adjusted_media_rate_, DataRate::Zero());
+    TimeDelta drain_time = std::max(media_debt_ / adjusted_media_rate_,
+                                    padding_debt_ / padding_rate_);
+
+    if (drain_time.IsZero() &&
+        (!media_debt_.IsZero() || !padding_debt_.IsZero())) {
+      // We have a non-zero debt, but drain time is smaller than tick size of
+      // TimeDelta, round it up to the smallest possible non-zero delta.
+      drain_time = TimeDelta::Micros(1);
+    }
+    next_send_time = last_process_time_ + drain_time;
+  } else {
+    // Nothing to do.
+    next_send_time = last_process_time_ + kPausedProcessInterval;
+  }
+
+  if (send_padding_if_silent_) {
+    next_send_time =
+        std::min(next_send_time, last_send_time_ + kPausedProcessInterval);
+  }
+
+  return next_send_time;
+}
+
+void PacingController::ProcessPackets() {
+  const Timestamp now = CurrentTime();
+  Timestamp target_send_time = now;
+
+  if (ShouldSendKeepalive(now)) {
+    DataSize keepalive_data_sent = DataSize::Zero();
+    // We can not send padding unless a normal packet has first been sent. If
+    // we do, timestamps get messed up.
+    if (seen_first_packet_) {
+      std::vector<std::unique_ptr<RtpPacketToSend>> keepalive_packets =
+          packet_sender_->GeneratePadding(DataSize::Bytes(1));
+      for (auto& packet : keepalive_packets) {
+        keepalive_data_sent +=
+            DataSize::Bytes(packet->payload_size() + packet->padding_size());
+        packet_sender_->SendPacket(std::move(packet), PacedPacketInfo());
+        for (auto& packet : packet_sender_->FetchFec()) {
+          EnqueuePacket(std::move(packet));
+        }
+      }
+    }
+    OnPacketSent(RtpPacketMediaType::kPadding, keepalive_data_sent, now);
+  }
+
+  if (paused_) {
+    return;
+  }
+
+  TimeDelta early_execute_margin =
+      prober_.is_probing() ? kMaxEarlyProbeProcessing : TimeDelta::Zero();
+
+  target_send_time = NextSendTime();
+  if (now + early_execute_margin < target_send_time) {
+    // We are too early, but if queue is empty still allow draining some debt.
+    // Probing is allowed to be sent up to kMinSleepTime early.
+    UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(now));
+    return;
+  }
+
+  TimeDelta elapsed_time = UpdateTimeAndGetElapsed(target_send_time);
+
+  if (elapsed_time > TimeDelta::Zero()) {
+    UpdateBudgetWithElapsedTime(elapsed_time);
+  }
+
+  PacedPacketInfo pacing_info;
+  DataSize recommended_probe_size = DataSize::Zero();
+  bool is_probing = prober_.is_probing();
+  if (is_probing) {
+    // Probe timing is sensitive, and handled explicitly by BitrateProber, so
+    // use actual send time rather than target.
+    pacing_info = prober_.CurrentCluster(now).value_or(PacedPacketInfo());
+    if (pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe) {
+      recommended_probe_size = prober_.RecommendedMinProbeSize();
+      RTC_DCHECK_GT(recommended_probe_size, DataSize::Zero());
+    } else {
+      // No valid probe cluster returned, probe might have timed out.
+      is_probing = false;
+    }
+  }
+
+  DataSize data_sent = DataSize::Zero();
+  int iteration = 0;
+  int packets_sent = 0;
+  int padding_packets_generated = 0;
+  for (; iteration < circuit_breaker_threshold_; ++iteration) {
+    // Fetch packet, so long as queue is not empty or budget is not
+    // exhausted.
+    std::unique_ptr<RtpPacketToSend> rtp_packet =
+        GetPendingPacket(pacing_info, target_send_time, now);
+    if (rtp_packet == nullptr) {
+      // No packet available to send, check if we should send padding.
+      if (now - target_send_time > kMaxPaddingReplayDuration) {
+        // The target send time is more than `kMaxPaddingReplayDuration` behind
+        // the real-time clock. This can happen if the clock is adjusted forward
+        // without `ProcessPackets()` having been called at the expected times.
+        target_send_time = now - kMaxPaddingReplayDuration;
+        last_process_time_ = std::max(last_process_time_, target_send_time);
+      }
+
+      DataSize padding_to_add = PaddingToAdd(recommended_probe_size, data_sent);
+      if (padding_to_add > DataSize::Zero()) {
+        std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
+            packet_sender_->GeneratePadding(padding_to_add);
+        if (!padding_packets.empty()) {
+          padding_packets_generated += padding_packets.size();
+          for (auto& packet : padding_packets) {
+            EnqueuePacket(std::move(packet));
+          }
+          // Continue loop to send the padding that was just added.
+          continue;
+        } else {
+          // Can't generate padding, still update padding budget for next send
+          // time.
+          UpdatePaddingBudgetWithSentData(padding_to_add);
+        }
+      }
+      // Can't fetch new packet and no padding to send, exit send loop.
+      break;
+    } else {
+      RTC_DCHECK(rtp_packet);
+      RTC_DCHECK(rtp_packet->packet_type().has_value());
+      const RtpPacketMediaType packet_type = *rtp_packet->packet_type();
+      DataSize packet_size = DataSize::Bytes(rtp_packet->payload_size() +
+                                             rtp_packet->padding_size());
+
+      if (include_overhead_) {
+        packet_size += DataSize::Bytes(rtp_packet->headers_size()) +
+                       transport_overhead_per_packet_;
+      }
+
+      packet_sender_->SendPacket(std::move(rtp_packet), pacing_info);
+      for (auto& packet : packet_sender_->FetchFec()) {
+        EnqueuePacket(std::move(packet));
+      }
+      data_sent += packet_size;
+      ++packets_sent;
+
+      // Send done, update send time.
+      OnPacketSent(packet_type, packet_size, now);
+
+      if (is_probing) {
+        pacing_info.probe_cluster_bytes_sent += packet_size.bytes();
+        // If we are currently probing, we need to stop the send loop when we
+        // have reached the send target.
+        if (data_sent >= recommended_probe_size) {
+          break;
+        }
+      }
+
+      // Update target send time in case that are more packets that we are late
+      // in processing.
+      target_send_time = NextSendTime();
+      if (target_send_time > now) {
+        // Exit loop if not probing.
+        if (!is_probing) {
+          break;
+        }
+        target_send_time = now;
+      }
+      UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(target_send_time));
+    }
+  }
+
+  if (iteration >= circuit_breaker_threshold_) {
+    // Circuit break activated. Log warning, adjust send time and return.
+    // TODO(sprang): Consider completely clearing state.
+    RTC_LOG(LS_ERROR)
+        << "PacingController exceeded max iterations in "
+           "send-loop. Debug info: "
+        << " packets sent = " << packets_sent
+        << ", padding packets generated = " << padding_packets_generated
+        << ", bytes sent = " << data_sent.bytes()
+        << ", probing = " << (is_probing ? "true" : "false")
+        << ", recommended_probe_size = " << recommended_probe_size.bytes()
+        << ", now = " << now.us()
+        << ", target_send_time = " << target_send_time.us()
+        << ", last_process_time = " << last_process_time_.us()
+        << ", last_send_time = " << last_send_time_.us()
+        << ", paused = " << (paused_ ? "true" : "false")
+        << ", media_debt = " << media_debt_.bytes()
+        << ", padding_debt = " << padding_debt_.bytes()
+        << ", pacing_rate = " << pacing_rate_.bps()
+        << ", adjusted_media_rate = " << adjusted_media_rate_.bps()
+        << ", padding_rate = " << padding_rate_.bps()
+        << ", queue size (packets) = " << packet_queue_.SizeInPackets()
+        << ", queue size (payload bytes) = "
+        << packet_queue_.SizeInPayloadBytes();
+    last_send_time_ = now;
+    last_process_time_ = now;
+    return;
+  }
+
+  if (is_probing) {
+    probing_send_failure_ = data_sent == DataSize::Zero();
+    if (!probing_send_failure_) {
+      prober_.ProbeSent(CurrentTime(), data_sent);
+    }
+  }
+
+  // Queue length has probably decreased, check if pacing rate needs to updated.
+  // Poll the time again, since we might have enqueued new fec/padding packets
+  // with a later timestamp than `now`.
+  MaybeUpdateMediaRateDueToLongQueue(CurrentTime());
+}
+
+DataSize PacingController::PaddingToAdd(DataSize recommended_probe_size,
+                                        DataSize data_sent) const {
+  if (!packet_queue_.Empty()) {
+    // Actual payload available, no need to add padding.
+    return DataSize::Zero();
+  }
+
+  if (congested_) {
+    // Don't add padding if congested, even if requested for probing.
+    return DataSize::Zero();
+  }
+
+  if (!seen_first_packet_) {
+    // We can not send padding unless a normal packet has first been sent. If
+    // we do, timestamps get messed up.
+    return DataSize::Zero();
+  }
+
+  if (!recommended_probe_size.IsZero()) {
+    if (recommended_probe_size > data_sent) {
+      return recommended_probe_size - data_sent;
+    }
+    return DataSize::Zero();
+  }
+
+  if (padding_rate_ > DataRate::Zero() && padding_debt_ == DataSize::Zero()) {
+    return kTargetPaddingDuration * padding_rate_;
+  }
+  return DataSize::Zero();
+}
+
+std::unique_ptr<RtpPacketToSend> PacingController::GetPendingPacket(
+    const PacedPacketInfo& pacing_info,
+    Timestamp target_send_time,
+    Timestamp now) {
+  const bool is_probe =
+      pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe;
+  // If first packet in probe, insert a small padding packet so we have a
+  // more reliable start window for the rate estimation.
+  if (is_probe && pacing_info.probe_cluster_bytes_sent == 0) {
+    auto padding = packet_sender_->GeneratePadding(DataSize::Bytes(1));
+    // If no RTP modules sending media are registered, we may not get a
+    // padding packet back.
+    if (!padding.empty()) {
+      // We should never get more than one padding packets with a requested
+      // size of 1 byte.
+      RTC_DCHECK_EQ(padding.size(), 1u);
+      return std::move(padding[0]);
+    }
+  }
+
+  if (packet_queue_.Empty()) {
+    return nullptr;
+  }
+
+  // First, check if there is any reason _not_ to send the next queued packet.
+  // Unpaced packets and probes are exempted from send checks.
+  if (NextUnpacedSendTime().IsInfinite() && !is_probe) {
+    if (congested_) {
+      // Don't send anything if congested.
+      return nullptr;
+    }
+
+    if (now <= target_send_time && send_burst_interval_.IsZero()) {
+      // We allow sending slightly early if we think that we would actually
+      // had been able to, had we been right on time - i.e. the current debt
+      // is not more than would be reduced to zero at the target sent time.
+      // If we allow packets to be sent in a burst, packet are allowed to be
+      // sent early.
+      TimeDelta flush_time = media_debt_ / adjusted_media_rate_;
+      if (now + flush_time > target_send_time) {
+        return nullptr;
+      }
+    }
+  }
+
+  return packet_queue_.Pop();
+}
+
+void PacingController::OnPacketSent(RtpPacketMediaType packet_type,
+                                    DataSize packet_size,
+                                    Timestamp send_time) {
+  if (!first_sent_packet_time_ && packet_type != RtpPacketMediaType::kPadding) {
+    first_sent_packet_time_ = send_time;
+  }
+
+  bool audio_packet = packet_type == RtpPacketMediaType::kAudio;
+  if ((!audio_packet || account_for_audio_) && packet_size > DataSize::Zero()) {
+    UpdateBudgetWithSentData(packet_size);
+  }
+
+  last_send_time_ = send_time;
+}
+
+void PacingController::UpdateBudgetWithElapsedTime(TimeDelta delta) {
+  media_debt_ -= std::min(media_debt_, adjusted_media_rate_ * delta);
+  padding_debt_ -= std::min(padding_debt_, padding_rate_ * delta);
+}
+
+void PacingController::UpdateBudgetWithSentData(DataSize size) {
+  media_debt_ += size;
+  media_debt_ = std::min(media_debt_, adjusted_media_rate_ * kMaxDebtInTime);
+  UpdatePaddingBudgetWithSentData(size);
+}
+
+void PacingController::UpdatePaddingBudgetWithSentData(DataSize size) {
+  padding_debt_ += size;
+  padding_debt_ = std::min(padding_debt_, padding_rate_ * kMaxDebtInTime);
+}
+
+void PacingController::SetQueueTimeLimit(TimeDelta limit) {
+  queue_time_limit_ = limit;
+}
+
+void PacingController::MaybeUpdateMediaRateDueToLongQueue(Timestamp now) {
+  adjusted_media_rate_ = pacing_rate_;
+  if (!drain_large_queues_) {
+    return;
+  }
+
+  DataSize queue_size_data = QueueSizeData();
+  if (queue_size_data > DataSize::Zero()) {
+    // Assuming equal size packets and input/output rate, the average packet
+    // has avg_time_left_ms left to get queue_size_bytes out of the queue, if
+    // time constraint shall be met. Determine bitrate needed for that.
+    packet_queue_.UpdateAverageQueueTime(now);
+    TimeDelta avg_time_left =
+        std::max(TimeDelta::Millis(1),
+                 queue_time_limit_ - packet_queue_.AverageQueueTime());
+    DataRate min_rate_needed = queue_size_data / avg_time_left;
+    if (min_rate_needed > pacing_rate_) {
+      adjusted_media_rate_ = min_rate_needed;
+      RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps="
+                          << pacing_rate_.kbps();
+    }
+  }
+}
+
+Timestamp PacingController::NextUnpacedSendTime() const {
+  if (!pace_audio_) {
+    Timestamp leading_audio_send_time =
+        packet_queue_.LeadingPacketEnqueueTime(RtpPacketMediaType::kAudio);
+    if (leading_audio_send_time.IsFinite()) {
+      return leading_audio_send_time;
+    }
+  }
+  if (fast_retransmissions_) {
+    Timestamp leading_retransmission_send_time =
+        packet_queue_.LeadingPacketEnqueueTime(
+            RtpPacketMediaType::kRetransmission);
+    if (leading_retransmission_send_time.IsFinite()) {
+      return leading_retransmission_send_time;
+    }
+  }
+  return Timestamp::MinusInfinity();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.h b/third_party/libwebrtc/modules/pacing/pacing_controller.h
new file mode 100644
index 0000000000..47dba179f5
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/pacing_controller.h
@@ -0,0 +1,251 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PACING_PACING_CONTROLLER_H_
+#define MODULES_PACING_PACING_CONTROLLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <array>
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/function_view.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/network_types.h"
+#include "modules/pacing/bitrate_prober.h"
+#include "modules/pacing/interval_budget.h"
+#include "modules/pacing/prioritized_packet_queue.h"
+#include "modules/pacing/rtp_packet_pacer.h"
+#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// This class implements a leaky-bucket packet pacing algorithm. It handles the
+// logic of determining which packets to send when, but the actual timing of
+// the processing is done externally (e.g. RtpPacketPacer). Furthermore, the
+// forwarding of packets when they are ready to be sent is also handled
+// externally, via the PacingController::PacketSender interface.
+class PacingController {
+ public:
+  class PacketSender {
+   public:
+    virtual ~PacketSender() = default;
+    virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
+                            const PacedPacketInfo& cluster_info) = 0;
+    // Should be called after each call to SendPacket().
+    virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() = 0;
+    virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+        DataSize size) = 0;
+
+    // TODO(bugs.webrtc.org/11340): Make pure virtual once downstream projects
+    // have been updated.
+    virtual void OnAbortedRetransmissions(
+        uint32_t ssrc,
+        rtc::ArrayView<const uint16_t> sequence_numbers) {}
+    virtual absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const {
+      return absl::nullopt;
+    }
+  };
+
+  // Expected max pacer delay. If ExpectedQueueTime() is higher than
+  // this value, the packet producers should wait (eg drop frames rather than
+  // encoding them). Bitrate sent may temporarily exceed target set by
+  // UpdateBitrate() so that this limit will be upheld.
+  static const TimeDelta kMaxExpectedQueueLength;
+  // If no media or paused, wake up at least every `kPausedProcessIntervalMs` in
+  // order to send a keep-alive packet so we don't get stuck in a bad state due
+  // to lack of feedback.
+  static const TimeDelta kPausedProcessInterval;
+  // The default minimum time that should elapse calls to `ProcessPackets()`.
+  static const TimeDelta kMinSleepTime;
+  // When padding should be generated, add packets to the buffer with a size
+  // corresponding to this duration times the current padding rate.
+  static const TimeDelta kTargetPaddingDuration;
+  // The maximum time that the pacer can use when "replaying" passed time where
+  // padding should have been generated.
+  static const TimeDelta kMaxPaddingReplayDuration;
+  // Allow probes to be processed slightly ahead of inteded send time. Currently
+  // set to 1ms as this is intended to allow times be rounded down to the
+  // nearest millisecond.
+  static const TimeDelta kMaxEarlyProbeProcessing;
+
+  PacingController(Clock* clock,
+                   PacketSender* packet_sender,
+                   const FieldTrialsView& field_trials);
+
+  ~PacingController();
+
+  // Adds the packet to the queue and calls PacketRouter::SendPacket() when
+  // it's time to send.
+  void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet);
+
+  void CreateProbeClusters(
+      rtc::ArrayView<const ProbeClusterConfig> probe_cluster_configs);
+
+  void Pause();   // Temporarily pause all sending.
+  void Resume();  // Resume sending packets.
+  bool IsPaused() const;
+
+  void SetCongested(bool congested);
+
+  // Sets the pacing rates. Must be called once before packets can be sent.
+  void SetPacingRates(DataRate pacing_rate, DataRate padding_rate);
+  DataRate pacing_rate() const { return adjusted_media_rate_; }
+
+  // Currently audio traffic is not accounted by pacer and passed through.
+  // With the introduction of audio BWE audio traffic will be accounted for
+  // the pacer budget calculation. The audio traffic still will be injected
+  // at high priority.
+  void SetAccountForAudioPackets(bool account_for_audio);
+  void SetIncludeOverhead();
+
+  void SetTransportOverhead(DataSize overhead_per_packet);
+  // The pacer is allowed to send enqued packets in bursts and can build up a
+  // packet "debt" that correspond to approximately the send rate during
+  // 'burst_interval'.
+  void SetSendBurstInterval(TimeDelta burst_interval);
+
+  // Returns the time when the oldest packet was queued.
+  Timestamp OldestPacketEnqueueTime() const;
+
+  // Number of packets in the pacer queue.
+  size_t QueueSizePackets() const;
+  // Number of packets in the pacer queue per media type (RtpPacketMediaType
+  // values are used as lookup index).
+  const std::array<int, kNumMediaTypes>& SizeInPacketsPerRtpPacketMediaType()
+      const;
+  // Totals size of packets in the pacer queue.
+  DataSize QueueSizeData() const;
+
+  // Current buffer level, i.e. max of media and padding debt.
+  DataSize CurrentBufferLevel() const;
+
+  // Returns the time when the first packet was sent.
+  absl::optional<Timestamp> FirstSentPacketTime() const;
+
+  // Returns the number of milliseconds it will take to send the current
+  // packets in the queue, given the current size and bitrate, ignoring prio.
+  TimeDelta ExpectedQueueTime() const;
+
+  void SetQueueTimeLimit(TimeDelta limit);
+
+  // Enable bitrate probing. Enabled by default, mostly here to simplify
+  // testing. Must be called before any packets are being sent to have an
+  // effect.
+  void SetProbingEnabled(bool enabled);
+
+  // Returns the next time we expect ProcessPackets() to be called.
+  Timestamp NextSendTime() const;
+
+  // Check queue of pending packets and send them or padding packets, if budget
+  // is available.
+  void ProcessPackets();
+
+  bool IsProbing() const;
+
+  // Note: Intended for debugging purposes only, will be removed.
+  // Sets the number of iterations of the main loop in `ProcessPackets()` that
+  // is considered erroneous to exceed.
+  void SetCircuitBreakerThreshold(int num_iterations);
+
+  // Remove any pending packets matching this SSRC from the packet queue.
+  void RemovePacketsForSsrc(uint32_t ssrc);
+
+ private:
+  TimeDelta UpdateTimeAndGetElapsed(Timestamp now);
+  bool ShouldSendKeepalive(Timestamp now) const;
+
+  // Updates the number of bytes that can be sent for the next time interval.
+  void UpdateBudgetWithElapsedTime(TimeDelta delta);
+  void UpdateBudgetWithSentData(DataSize size);
+  void UpdatePaddingBudgetWithSentData(DataSize size);
+
+  DataSize PaddingToAdd(DataSize recommended_probe_size,
+                        DataSize data_sent) const;
+
+  std::unique_ptr<RtpPacketToSend> GetPendingPacket(
+      const PacedPacketInfo& pacing_info,
+      Timestamp target_send_time,
+      Timestamp now);
+  void OnPacketSent(RtpPacketMediaType packet_type,
+                    DataSize packet_size,
+                    Timestamp send_time);
+  void MaybeUpdateMediaRateDueToLongQueue(Timestamp now);
+
+  Timestamp CurrentTime() const;
+
+  // Helper methods for packet that may not be paced. Returns a finite Timestamp
+  // if a packet type is configured to not be paced and the packet queue has at
+  // least one packet of that type. Otherwise returns
+  // Timestamp::MinusInfinity().
+  Timestamp NextUnpacedSendTime() const;
+
+  Clock* const clock_;
+  PacketSender* const packet_sender_;
+  const FieldTrialsView& field_trials_;
+
+  const bool drain_large_queues_;
+  const bool send_padding_if_silent_;
+  const bool pace_audio_;
+  const bool ignore_transport_overhead_;
+  const bool fast_retransmissions_;
+
+  TimeDelta min_packet_limit_;
+  DataSize transport_overhead_per_packet_;
+  TimeDelta send_burst_interval_;
+
+  // TODO(webrtc:9716): Remove this when we are certain clocks are monotonic.
+  // The last millisecond timestamp returned by `clock_`.
+  mutable Timestamp last_timestamp_;
+  bool paused_;
+
+  // Amount of outstanding data for media and padding.
+  DataSize media_debt_;
+  DataSize padding_debt_;
+
+  // The target pacing rate, signaled via SetPacingRates().
+  DataRate pacing_rate_;
+  // The media send rate, which might adjusted from pacing_rate_, e.g. if the
+  // pacing queue is growing too long.
+  DataRate adjusted_media_rate_;
+  // The padding target rate. We aim to fill up to this rate with padding what
+  // is not already used by media.
+  DataRate padding_rate_;
+
+  BitrateProber prober_;
+  bool probing_send_failure_;
+
+  Timestamp last_process_time_;
+  Timestamp last_send_time_;
+  absl::optional<Timestamp> first_sent_packet_time_;
+  bool seen_first_packet_;
+
+  PrioritizedPacketQueue packet_queue_;
+
+  bool congested_;
+
+  TimeDelta queue_time_limit_;
+  bool account_for_audio_;
+  bool include_overhead_;
+
+  int circuit_breaker_threshold_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_PACING_PACING_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc
new file mode 100644
index 0000000000..3b3c3eb761
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc
@@ -0,0 +1,2175 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/pacing_controller.h"
+
+#include <algorithm>
+#include <list>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "modules/pacing/packet_router.h"
+#include "system_wrappers/include/clock.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::Field;
+using ::testing::Pointee;
+using ::testing::Property;
+using ::testing::Return;
+using ::testing::WithoutArgs;
+
+using ::webrtc::test::ExplicitKeyValueConfig;
+
+namespace webrtc {
+namespace {
+constexpr DataRate kFirstClusterRate = DataRate::KilobitsPerSec(900);
+constexpr DataRate kSecondClusterRate = DataRate::KilobitsPerSec(1800);
+
+// The error stems from truncating the time interval of probe packets to integer
+// values. This results in probing slightly higher than the target bitrate.
+// For 1.8 Mbps, this comes to be about 120 kbps with 1200 probe packets.
+constexpr DataRate kProbingErrorMargin = DataRate::KilobitsPerSec(150);
+
+const float kPaceMultiplier = 2.5f;
+
+constexpr uint32_t kAudioSsrc = 12345;
+constexpr uint32_t kVideoSsrc = 234565;
+
+constexpr DataRate kTargetRate = DataRate::KilobitsPerSec(800);
+
+std::unique_ptr<RtpPacketToSend> BuildPacket(RtpPacketMediaType type,
+                                             uint32_t ssrc,
+                                             uint16_t sequence_number,
+                                             int64_t capture_time_ms,
+                                             size_t size) {
+  auto packet = std::make_unique<RtpPacketToSend>(nullptr);
+  packet->set_packet_type(type);
+  packet->SetSsrc(ssrc);
+  packet->SetSequenceNumber(sequence_number);
+  packet->set_capture_time(Timestamp::Millis(capture_time_ms));
+  packet->SetPayloadSize(size);
+  return packet;
+}
+
+class MediaStream {
+ public:
+  MediaStream(SimulatedClock& clock,
+              RtpPacketMediaType type,
+              uint32_t ssrc,
+              size_t packet_size)
+      : clock_(clock), type_(type), ssrc_(ssrc), packet_size_(packet_size) {}
+
+  std::unique_ptr<RtpPacketToSend> BuildNextPacket() {
+    return BuildPacket(type_, ssrc_, seq_num_++, clock_.TimeInMilliseconds(),
+                       packet_size_);
+  }
+  std::unique_ptr<RtpPacketToSend> BuildNextPacket(size_t size) {
+    return BuildPacket(type_, ssrc_, seq_num_++, clock_.TimeInMilliseconds(),
+                       size);
+  }
+
+ private:
+  SimulatedClock& clock_;
+  const RtpPacketMediaType type_;
+  const uint32_t ssrc_;
+  const size_t packet_size_;
+  uint16_t seq_num_ = 1000;
+};
+
+// Mock callback proxy, where both new and old api redirects to common mock
+// methods that focus on core aspects.
+class MockPacingControllerCallback : public PacingController::PacketSender {
+ public:
+  void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
+                  const PacedPacketInfo& cluster_info) override {
+    SendPacket(packet->Ssrc(), packet->SequenceNumber(),
+               packet->capture_time().ms(),
+               packet->packet_type() == RtpPacketMediaType::kRetransmission,
+               packet->packet_type() == RtpPacketMediaType::kPadding);
+  }
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      DataSize target_size) override {
+    std::vector<std::unique_ptr<RtpPacketToSend>> ret;
+    size_t padding_size = SendPadding(target_size.bytes());
+    if (padding_size > 0) {
+      auto packet = std::make_unique<RtpPacketToSend>(nullptr);
+      packet->SetPayloadSize(padding_size);
+      packet->set_packet_type(RtpPacketMediaType::kPadding);
+      ret.emplace_back(std::move(packet));
+    }
+    return ret;
+  }
+
+  MOCK_METHOD(void,
+              SendPacket,
+              (uint32_t ssrc,
+               uint16_t sequence_number,
+               int64_t capture_timestamp,
+               bool retransmission,
+               bool padding));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              FetchFec,
+              (),
+              (override));
+  MOCK_METHOD(size_t, SendPadding, (size_t target_size));
+  MOCK_METHOD(void,
+              OnAbortedRetransmissions,
+              (uint32_t, rtc::ArrayView<const uint16_t>),
+              (override));
+  MOCK_METHOD(absl::optional<uint32_t>,
+              GetRtxSsrcForMedia,
+              (uint32_t),
+              (const, override));
+};
+
+// Mock callback implementing the raw api.
+class MockPacketSender : public PacingController::PacketSender {
+ public:
+  MOCK_METHOD(void,
+              SendPacket,
+              (std::unique_ptr<RtpPacketToSend> packet,
+               const PacedPacketInfo& cluster_info),
+              (override));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              FetchFec,
+              (),
+              (override));
+
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              GeneratePadding,
+              (DataSize target_size),
+              (override));
+  MOCK_METHOD(void,
+              OnAbortedRetransmissions,
+              (uint32_t, rtc::ArrayView<const uint16_t>),
+              (override));
+  MOCK_METHOD(absl::optional<uint32_t>,
+              GetRtxSsrcForMedia,
+              (uint32_t),
+              (const, override));
+};
+
+class PacingControllerPadding : public PacingController::PacketSender {
+ public:
+  static const size_t kPaddingPacketSize = 224;
+
+  PacingControllerPadding() : padding_sent_(0), total_bytes_sent_(0) {}
+
+  void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
+                  const PacedPacketInfo& pacing_info) override {
+    total_bytes_sent_ += packet->payload_size();
+  }
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() override {
+    return {};
+  }
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      DataSize target_size) override {
+    size_t num_packets =
+        (target_size.bytes() + kPaddingPacketSize - 1) / kPaddingPacketSize;
+    std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+    for (size_t i = 0; i < num_packets; ++i) {
+      packets.emplace_back(std::make_unique<RtpPacketToSend>(nullptr));
+      packets.back()->SetPadding(kPaddingPacketSize);
+      packets.back()->set_packet_type(RtpPacketMediaType::kPadding);
+      padding_sent_ += kPaddingPacketSize;
+    }
+    return packets;
+  }
+
+  void OnAbortedRetransmissions(uint32_t,
+                                rtc::ArrayView<const uint16_t>) override {}
+  absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t) const override {
+    return absl::nullopt;
+  }
+
+  size_t padding_sent() { return padding_sent_; }
+  size_t total_bytes_sent() { return total_bytes_sent_; }
+
+ private:
+  size_t padding_sent_;
+  size_t total_bytes_sent_;
+};
+
+class PacingControllerProbing : public PacingController::PacketSender {
+ public:
+  PacingControllerProbing() : packets_sent_(0), padding_sent_(0) {}
+
+  void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
+                  const PacedPacketInfo& pacing_info) override {
+    if (packet->packet_type() != RtpPacketMediaType::kPadding) {
+      ++packets_sent_;
+    }
+    last_pacing_info_ = pacing_info;
+  }
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() override {
+    return {};
+  }
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      DataSize target_size) override {
+    // From RTPSender:
+    // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
+    const DataSize kMaxPadding = DataSize::Bytes(224);
+
+    std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+    while (target_size > DataSize::Zero()) {
+      DataSize padding_size = std::min(kMaxPadding, target_size);
+      packets.emplace_back(std::make_unique<RtpPacketToSend>(nullptr));
+      packets.back()->SetPadding(padding_size.bytes());
+      packets.back()->set_packet_type(RtpPacketMediaType::kPadding);
+      padding_sent_ += padding_size.bytes();
+      target_size -= padding_size;
+    }
+    return packets;
+  }
+
+  void OnAbortedRetransmissions(uint32_t,
+                                rtc::ArrayView<const uint16_t>) override {}
+  absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t) const override {
+    return absl::nullopt;
+  }
+
+  int packets_sent() const { return packets_sent_; }
+  int padding_sent() const { return padding_sent_; }
+  int total_packets_sent() const { return packets_sent_ + padding_sent_; }
+  PacedPacketInfo last_pacing_info() const { return last_pacing_info_; }
+
+ private:
+  int packets_sent_;
+  int padding_sent_;
+  PacedPacketInfo last_pacing_info_;
+};
+
+class PacingControllerTest : public ::testing::Test {
+ protected:
+  PacingControllerTest() : clock_(123456), trials_("") {}
+
+  void SendAndExpectPacket(PacingController* pacer,
+                           RtpPacketMediaType type,
+                           uint32_t ssrc,
+                           uint16_t sequence_number,
+                           int64_t capture_time_ms,
+                           size_t size) {
+    pacer->EnqueuePacket(
+        BuildPacket(type, ssrc, sequence_number, capture_time_ms, size));
+
+    EXPECT_CALL(callback_,
+                SendPacket(ssrc, sequence_number, capture_time_ms,
+                           type == RtpPacketMediaType::kRetransmission, false));
+  }
+
+  void AdvanceTimeUntil(webrtc::Timestamp time) {
+    Timestamp now = clock_.CurrentTime();
+    clock_.AdvanceTime(std::max(TimeDelta::Zero(), time - now));
+  }
+
+  void ConsumeInitialBudget(PacingController* pacer) {
+    const uint32_t kSsrc = 54321;
+    uint16_t sequence_number = 1234;
+    int64_t capture_time_ms = clock_.TimeInMilliseconds();
+    const size_t kPacketSize = 250;
+
+    EXPECT_TRUE(pacer->OldestPacketEnqueueTime().IsInfinite());
+
+    // Due to the multiplicative factor we can send 5 packets during a send
+    // interval. (network capacity * multiplier / (8 bits per byte *
+    // (packet size * #send intervals per second)
+    const size_t packets_to_send_per_interval =
+        kTargetRate.bps() * kPaceMultiplier / (8 * kPacketSize * 200);
+    for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+      SendAndExpectPacket(pacer, RtpPacketMediaType::kVideo, kSsrc,
+                          sequence_number++, capture_time_ms, kPacketSize);
+    }
+
+    while (pacer->QueueSizePackets() > 0) {
+      AdvanceTimeUntil(pacer->NextSendTime());
+      pacer->ProcessPackets();
+    }
+  }
+
+  SimulatedClock clock_;
+
+  MediaStream audio_ = MediaStream(clock_,
+                                   /*type*/ RtpPacketMediaType::kAudio,
+                                   /*ssrc*/ kAudioSsrc,
+                                   /*packet_size*/ 100);
+  MediaStream video_ = MediaStream(clock_,
+                                   /*type*/ RtpPacketMediaType::kVideo,
+                                   /*ssrc*/ kVideoSsrc,
+                                   /*packet_size*/ 1000);
+
+  ::testing::NiceMock<MockPacingControllerCallback> callback_;
+  ExplicitKeyValueConfig trials_;
+};
+
+TEST_F(PacingControllerTest, DefaultNoPaddingInSilence) {
+  const test::ExplicitKeyValueConfig trials("");
+  PacingController pacer(&clock_, &callback_, trials);
+  pacer.SetPacingRates(kTargetRate, DataRate::Zero());
+  // Video packet to reset last send time and provide padding data.
+  pacer.EnqueuePacket(video_.BuildNextPacket());
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  clock_.AdvanceTimeMilliseconds(5);
+  pacer.ProcessPackets();
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  // Waiting 500 ms should not trigger sending of padding.
+  clock_.AdvanceTimeMilliseconds(500);
+  pacer.ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, PaddingInSilenceWithTrial) {
+  const test::ExplicitKeyValueConfig trials(
+      "WebRTC-Pacer-PadInSilence/Enabled/");
+  PacingController pacer(&clock_, &callback_, trials);
+  pacer.SetPacingRates(kTargetRate, DataRate::Zero());
+  // Video packet to reset last send time and provide padding data.
+  pacer.EnqueuePacket(video_.BuildNextPacket());
+  EXPECT_CALL(callback_, SendPacket).Times(2);
+  clock_.AdvanceTimeMilliseconds(5);
+  pacer.ProcessPackets();
+  EXPECT_CALL(callback_, SendPadding).WillOnce(Return(1000));
+  // Waiting 500 ms should trigger sending of padding.
+  clock_.AdvanceTimeMilliseconds(500);
+  pacer.ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, CongestionWindowAffectsAudioInTrial) {
+  const test::ExplicitKeyValueConfig trials("WebRTC-Pacer-BlockAudio/Enabled/");
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  PacingController pacer(&clock_, &callback_, trials);
+  pacer.SetPacingRates(DataRate::KilobitsPerSec(10000), DataRate::Zero());
+  // Video packet fills congestion window.
+  pacer.EnqueuePacket(video_.BuildNextPacket());
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+  pacer.SetCongested(true);
+  // Audio packet blocked due to congestion.
+  pacer.EnqueuePacket(audio_.BuildNextPacket());
+  EXPECT_CALL(callback_, SendPacket).Times(0);
+  // Forward time to where we send keep-alive.
+  EXPECT_CALL(callback_, SendPadding(1)).Times(2);
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+  // Audio packet unblocked when congestion window clear.
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+  pacer.SetCongested(false);
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, DefaultCongestionWindowDoesNotAffectAudio) {
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  const test::ExplicitKeyValueConfig trials("");
+  PacingController pacer(&clock_, &callback_, trials);
+  pacer.SetPacingRates(DataRate::BitsPerSec(10000000), DataRate::Zero());
+  // Video packet fills congestion window.
+  pacer.EnqueuePacket(video_.BuildNextPacket());
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+  pacer.SetCongested(true);
+  // Audio not blocked due to congestion.
+  pacer.EnqueuePacket(audio_.BuildNextPacket());
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, BudgetAffectsAudioInTrial) {
+  ExplicitKeyValueConfig trials("WebRTC-Pacer-BlockAudio/Enabled/");
+  PacingController pacer(&clock_, &callback_, trials);
+  const size_t kPacketSize = 1000;
+  const int kProcessIntervalsPerSecond = 1000 / 5;
+  DataRate pacing_rate =
+      DataRate::BitsPerSec(kPacketSize / 3 * 8 * kProcessIntervalsPerSecond);
+  pacer.SetPacingRates(pacing_rate, DataRate::Zero());
+  // Video fills budget for following process periods.
+  pacer.EnqueuePacket(video_.BuildNextPacket(kPacketSize));
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+  // Audio packet blocked due to budget limit.
+  pacer.EnqueuePacket(audio_.BuildNextPacket());
+  Timestamp wait_start_time = clock_.CurrentTime();
+  Timestamp wait_end_time = Timestamp::MinusInfinity();
+  EXPECT_CALL(callback_, SendPacket).WillOnce(WithoutArgs([&]() {
+    wait_end_time = clock_.CurrentTime();
+  }));
+  while (!wait_end_time.IsFinite()) {
+    AdvanceTimeUntil(pacer.NextSendTime());
+    pacer.ProcessPackets();
+  }
+  const TimeDelta expected_wait_time =
+      DataSize::Bytes(kPacketSize) / pacing_rate;
+  // Verify delay is near expectation, within timing margin.
+  EXPECT_LT(((wait_end_time - wait_start_time) - expected_wait_time).Abs(),
+            PacingController::kMinSleepTime);
+}
+
+TEST_F(PacingControllerTest, DefaultBudgetDoesNotAffectAudio) {
+  const size_t kPacketSize = 1000;
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  const test::ExplicitKeyValueConfig trials("");
+  PacingController pacer(&clock_, &callback_, trials);
+  const int kProcessIntervalsPerSecond = 1000 / 5;
+  pacer.SetPacingRates(
+      DataRate::BitsPerSec(kPacketSize / 3 * 8 * kProcessIntervalsPerSecond),
+      DataRate::Zero());
+  // Video fills budget for following process periods.
+  pacer.EnqueuePacket(video_.BuildNextPacket(kPacketSize));
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+  // Audio packet not blocked due to budget limit.
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  pacer.EnqueuePacket(audio_.BuildNextPacket());
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, FirstSentPacketTimeIsSet) {
+  const Timestamp kStartTime = clock_.CurrentTime();
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  // No packet sent.
+  EXPECT_FALSE(pacer->FirstSentPacketTime().has_value());
+  pacer->EnqueuePacket(video_.BuildNextPacket());
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+  EXPECT_EQ(kStartTime, pacer->FirstSentPacketTime());
+}
+
+TEST_F(PacingControllerTest, QueueAndPacePackets) {
+  const uint32_t kSsrc = 12345;
+  uint16_t sequence_number = 1234;
+  const DataSize kPackeSize = DataSize::Bytes(250);
+  const TimeDelta kSendInterval = TimeDelta::Millis(5);
+
+  // Due to the multiplicative factor we can send 5 packets during a 5ms send
+  // interval. (send interval * network capacity * multiplier / packet size)
+  const size_t kPacketsToSend = (kSendInterval * kTargetRate).bytes() *
+                                kPaceMultiplier / kPackeSize.bytes();
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  for (size_t i = 0; i < kPacketsToSend; ++i) {
+    SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, kSsrc,
+                        sequence_number++, clock_.TimeInMilliseconds(),
+                        kPackeSize.bytes());
+  }
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+
+  // Enqueue one extra packet.
+  int64_t queued_packet_timestamp = clock_.TimeInMilliseconds();
+  pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kSsrc,
+                                   sequence_number, queued_packet_timestamp,
+                                   kPackeSize.bytes()));
+  EXPECT_EQ(kPacketsToSend + 1, pacer->QueueSizePackets());
+
+  // Send packets until the initial kPacketsToSend packets are done.
+  Timestamp start_time = clock_.CurrentTime();
+  while (pacer->QueueSizePackets() > 1) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+  EXPECT_LT(clock_.CurrentTime() - start_time, kSendInterval);
+
+  // Proceed till last packet can be sent.
+  EXPECT_CALL(callback_, SendPacket(kSsrc, sequence_number,
+                                    queued_packet_timestamp, false, false))
+      .Times(1);
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+  EXPECT_GE(clock_.CurrentTime() - start_time, kSendInterval);
+  EXPECT_EQ(pacer->QueueSizePackets(), 0u);
+}
+
+TEST_F(PacingControllerTest, PaceQueuedPackets) {
+  uint32_t ssrc = 12345;
+  uint16_t sequence_number = 1234;
+  const size_t kPacketSize = 250;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t packets_to_send_per_interval =
+      kTargetRate.bps() * kPaceMultiplier / (8 * kPacketSize * 200);
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                        sequence_number++, clock_.TimeInMilliseconds(),
+                        kPacketSize);
+  }
+
+  for (size_t j = 0; j < packets_to_send_per_interval * 10; ++j) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                     sequence_number++,
+                                     clock_.TimeInMilliseconds(), kPacketSize));
+  }
+  EXPECT_EQ(packets_to_send_per_interval + packets_to_send_per_interval * 10,
+            pacer->QueueSizePackets());
+
+  while (pacer->QueueSizePackets() > packets_to_send_per_interval * 10) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+  EXPECT_EQ(pacer->QueueSizePackets(), packets_to_send_per_interval * 10);
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+
+  EXPECT_CALL(callback_, SendPacket(ssrc, _, _, false, false))
+      .Times(pacer->QueueSizePackets());
+  const TimeDelta expected_pace_time =
+      DataSize::Bytes(pacer->QueueSizePackets() * kPacketSize) /
+      (kPaceMultiplier * kTargetRate);
+  Timestamp start_time = clock_.CurrentTime();
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+  const TimeDelta actual_pace_time = clock_.CurrentTime() - start_time;
+  EXPECT_LT((actual_pace_time - expected_pace_time).Abs(),
+            PacingController::kMinSleepTime);
+
+  EXPECT_EQ(0u, pacer->QueueSizePackets());
+  AdvanceTimeUntil(pacer->NextSendTime());
+  EXPECT_EQ(0u, pacer->QueueSizePackets());
+  pacer->ProcessPackets();
+
+  // Send some more packet, just show that we can..?
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                        sequence_number++, clock_.TimeInMilliseconds(), 250);
+  }
+  EXPECT_EQ(packets_to_send_per_interval, pacer->QueueSizePackets());
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+  EXPECT_EQ(0u, pacer->QueueSizePackets());
+}
+
+TEST_F(PacingControllerTest, RepeatedRetransmissionsAllowed) {
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  // Send one packet, then two retransmissions of that packet.
+  for (size_t i = 0; i < 3; i++) {
+    constexpr uint32_t ssrc = 333;
+    constexpr uint16_t sequence_number = 444;
+    constexpr size_t bytes = 250;
+    bool is_retransmission = (i != 0);  // Original followed by retransmissions.
+    SendAndExpectPacket(pacer.get(),
+                        is_retransmission ? RtpPacketMediaType::kRetransmission
+                                          : RtpPacketMediaType::kVideo,
+                        ssrc, sequence_number, clock_.TimeInMilliseconds(),
+                        bytes);
+    clock_.AdvanceTimeMilliseconds(5);
+  }
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+}
+
+TEST_F(PacingControllerTest,
+       CanQueuePacketsWithSameSequenceNumberOnDifferentSsrcs) {
+  uint32_t ssrc = 12345;
+  uint16_t sequence_number = 1234;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                      sequence_number, clock_.TimeInMilliseconds(), 250);
+
+  // Expect packet on second ssrc to be queued and sent as well.
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc + 1,
+                      sequence_number, clock_.TimeInMilliseconds(), 250);
+
+  clock_.AdvanceTimeMilliseconds(1000);
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+}
+
+TEST_F(PacingControllerTest, Padding) {
+  uint32_t ssrc = 12345;
+  uint16_t sequence_number = 1234;
+  const size_t kPacketSize = 250;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, kTargetRate);
+
+  const size_t kPacketsToSend = 20;
+  for (size_t i = 0; i < kPacketsToSend; ++i) {
+    SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                        sequence_number++, clock_.TimeInMilliseconds(),
+                        kPacketSize);
+  }
+  const TimeDelta expected_pace_time =
+      DataSize::Bytes(pacer->QueueSizePackets() * kPacketSize) /
+      (kPaceMultiplier * kTargetRate);
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  // Only the media packets should be sent.
+  Timestamp start_time = clock_.CurrentTime();
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+    const TimeDelta actual_pace_time = clock_.CurrentTime() - start_time;
+    EXPECT_LE((actual_pace_time - expected_pace_time).Abs(),
+              PacingController::kMinSleepTime);
+
+    // Pacing media happens at 2.5x, but padding was configured with 1.0x
+    // factor. We have to wait until the padding debt is gone before we start
+    // sending padding.
+    const TimeDelta time_to_padding_debt_free =
+        (expected_pace_time * kPaceMultiplier) - actual_pace_time;
+    clock_.AdvanceTime(time_to_padding_debt_free -
+                       PacingController::kMinSleepTime);
+    pacer->ProcessPackets();
+
+    // Send 10 padding packets.
+    const size_t kPaddingPacketsToSend = 10;
+    DataSize padding_sent = DataSize::Zero();
+    size_t packets_sent = 0;
+    Timestamp first_send_time = Timestamp::MinusInfinity();
+    Timestamp last_send_time = Timestamp::MinusInfinity();
+
+    EXPECT_CALL(callback_, SendPadding)
+        .Times(kPaddingPacketsToSend)
+        .WillRepeatedly([&](size_t target_size) {
+          ++packets_sent;
+          if (packets_sent < kPaddingPacketsToSend) {
+            // Don't count bytes of last packet, instead just
+            // use this as the time the last packet finished
+            // sending.
+            padding_sent += DataSize::Bytes(target_size);
+          }
+          if (first_send_time.IsInfinite()) {
+            first_send_time = clock_.CurrentTime();
+          } else {
+            last_send_time = clock_.CurrentTime();
+          }
+          return target_size;
+        });
+    EXPECT_CALL(callback_, SendPacket(_, _, _, false, true))
+        .Times(kPaddingPacketsToSend);
+
+    while (packets_sent < kPaddingPacketsToSend) {
+      AdvanceTimeUntil(pacer->NextSendTime());
+      pacer->ProcessPackets();
+    }
+
+    // Verify rate of sent padding.
+    TimeDelta padding_duration = last_send_time - first_send_time;
+    DataRate padding_rate = padding_sent / padding_duration;
+    EXPECT_EQ(padding_rate, kTargetRate);
+}
+
+TEST_F(PacingControllerTest, NoPaddingBeforeNormalPacket) {
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, kTargetRate);
+
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+
+  pacer->ProcessPackets();
+  AdvanceTimeUntil(pacer->NextSendTime());
+
+  pacer->ProcessPackets();
+  AdvanceTimeUntil(pacer->NextSendTime());
+
+  uint32_t ssrc = 12345;
+  uint16_t sequence_number = 1234;
+  int64_t capture_time_ms = 56789;
+
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                      sequence_number++, capture_time_ms, 250);
+  bool padding_sent = false;
+  EXPECT_CALL(callback_, SendPadding).WillOnce([&](size_t padding) {
+    padding_sent = true;
+    return padding;
+  });
+  EXPECT_CALL(callback_, SendPacket(_, _, _, _, true)).Times(1);
+  while (!padding_sent) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+}
+
+TEST_F(PacingControllerTest, VerifyAverageBitrateVaryingMediaPayload) {
+  uint32_t ssrc = 12345;
+  uint16_t sequence_number = 1234;
+  int64_t capture_time_ms = 56789;
+  const TimeDelta kAveragingWindowLength = TimeDelta::Seconds(10);
+  PacingControllerPadding callback;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback, trials_);
+  pacer->SetProbingEnabled(false);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, kTargetRate);
+
+  Timestamp start_time = clock_.CurrentTime();
+  size_t media_bytes = 0;
+  while (clock_.CurrentTime() - start_time < kAveragingWindowLength) {
+    // Maybe add some new media packets corresponding to expected send rate.
+    int rand_value = rand();  // NOLINT (rand_r instead of rand)
+    while (
+        media_bytes <
+        (kTargetRate * (clock_.CurrentTime() - start_time)).bytes<size_t>()) {
+      size_t media_payload = rand_value % 400 + 800;  // [400, 1200] bytes.
+      pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                       sequence_number++, capture_time_ms,
+                                       media_payload));
+      media_bytes += media_payload;
+    }
+
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+
+  EXPECT_NEAR(
+      kTargetRate.bps(),
+      (DataSize::Bytes(callback.total_bytes_sent()) / kAveragingWindowLength)
+          .bps(),
+      (kTargetRate * 0.01 /* 1% error marging */).bps());
+}
+
+TEST_F(PacingControllerTest, Priority) {
+  uint32_t ssrc_low_priority = 12345;
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  int64_t capture_time_ms = 56789;
+  int64_t capture_time_ms_low_priority = 1234567;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  ConsumeInitialBudget(pacer.get());
+
+  // Expect normal and low priority to be queued and high to pass through.
+  pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo,
+                                   ssrc_low_priority, sequence_number++,
+                                   capture_time_ms_low_priority, 250));
+
+  const size_t packets_to_send_per_interval =
+      kTargetRate.bps() * kPaceMultiplier / (8 * 250 * 200);
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kRetransmission, ssrc,
+                                     sequence_number++, capture_time_ms, 250));
+  }
+  pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kAudio, ssrc,
+                                   sequence_number++, capture_time_ms, 250));
+
+  // Expect all high and normal priority to be sent out first.
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  EXPECT_CALL(callback_, SendPacket(ssrc, _, capture_time_ms, _, _))
+      .Times(packets_to_send_per_interval + 1);
+
+  while (pacer->QueueSizePackets() > 1) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+
+  EXPECT_EQ(1u, pacer->QueueSizePackets());
+
+  EXPECT_CALL(callback_, SendPacket(ssrc_low_priority, _,
+                                    capture_time_ms_low_priority, _, _));
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, RetransmissionPriority) {
+  uint32_t ssrc = 12345;
+  uint16_t sequence_number = 1234;
+  int64_t capture_time_ms = 45678;
+  int64_t capture_time_ms_retransmission = 56789;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t packets_to_send_per_interval =
+      kTargetRate.bps() * kPaceMultiplier / (8 * 250 * 200);
+  pacer->ProcessPackets();
+  EXPECT_EQ(0u, pacer->QueueSizePackets());
+
+  // Alternate retransmissions and normal packets.
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                     sequence_number++, capture_time_ms, 250));
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kRetransmission, ssrc,
+                                     sequence_number++,
+                                     capture_time_ms_retransmission, 250));
+  }
+  EXPECT_EQ(2 * packets_to_send_per_interval, pacer->QueueSizePackets());
+
+  // Expect all retransmissions to be sent out first despite having a later
+  // capture time.
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  EXPECT_CALL(callback_, SendPacket(_, _, _, false, _)).Times(0);
+  EXPECT_CALL(callback_,
+              SendPacket(ssrc, _, capture_time_ms_retransmission, true, _))
+      .Times(packets_to_send_per_interval);
+
+  while (pacer->QueueSizePackets() > packets_to_send_per_interval) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+  EXPECT_EQ(packets_to_send_per_interval, pacer->QueueSizePackets());
+
+  // Expect the remaining (non-retransmission) packets to be sent.
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  EXPECT_CALL(callback_, SendPacket(_, _, _, true, _)).Times(0);
+  EXPECT_CALL(callback_, SendPacket(ssrc, _, capture_time_ms, false, _))
+      .Times(packets_to_send_per_interval);
+
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+  EXPECT_EQ(0u, pacer->QueueSizePackets());
+}
+
+TEST_F(PacingControllerTest, HighPrioDoesntAffectBudget) {
+  const size_t kPacketSize = 250;
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  int64_t capture_time_ms = 56789;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  // As high prio packets doesn't affect the budget, we should be able to send
+  // a high number of them at once.
+  const size_t kNumAudioPackets = 25;
+  for (size_t i = 0; i < kNumAudioPackets; ++i) {
+    SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kAudio, ssrc,
+                        sequence_number++, capture_time_ms, kPacketSize);
+  }
+  pacer->ProcessPackets();
+  // Low prio packets does affect the budget.
+  // Due to the multiplicative factor we can send 5 packets during a send
+  // interval. (network capacity * multiplier / (8 bits per byte *
+  // (packet size * #send intervals per second)
+  const size_t kPacketsToSendPerInterval =
+      kTargetRate.bps() * kPaceMultiplier / (8 * kPacketSize * 200);
+  for (size_t i = 0; i < kPacketsToSendPerInterval; ++i) {
+    SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                        sequence_number++, clock_.TimeInMilliseconds(),
+                        kPacketSize);
+  }
+
+  // Send all packets and measure pace time.
+  Timestamp start_time = clock_.CurrentTime();
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+
+  // Measure pacing time. Expect only low-prio packets to affect this.
+  TimeDelta pacing_time = clock_.CurrentTime() - start_time;
+  TimeDelta expected_pacing_time =
+      DataSize::Bytes(kPacketsToSendPerInterval * kPacketSize) /
+      (kTargetRate * kPaceMultiplier);
+  EXPECT_NEAR(pacing_time.us<double>(), expected_pacing_time.us<double>(),
+              PacingController::kMinSleepTime.us<double>());
+}
+
+TEST_F(PacingControllerTest, SendsOnlyPaddingWhenCongested) {
+  uint32_t ssrc = 202020;
+  uint16_t sequence_number = 1000;
+  int kPacketSize = 250;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  // Send an initial packet so we have a last send time.
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                      sequence_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize);
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+
+  // Set congested state, we should not send anything until the 500ms since
+  // last send time limit for keep-alives is triggered.
+  EXPECT_CALL(callback_, SendPacket).Times(0);
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  pacer->SetCongested(true);
+  size_t blocked_packets = 0;
+  int64_t expected_time_until_padding = 500;
+  while (expected_time_until_padding > 5) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                     sequence_number++,
+                                     clock_.TimeInMilliseconds(), kPacketSize));
+    blocked_packets++;
+    clock_.AdvanceTimeMilliseconds(5);
+    pacer->ProcessPackets();
+    expected_time_until_padding -= 5;
+  }
+
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+  EXPECT_CALL(callback_, SendPadding(1)).WillOnce(Return(1));
+  EXPECT_CALL(callback_, SendPacket(_, _, _, _, true)).Times(1);
+  clock_.AdvanceTimeMilliseconds(5);
+  pacer->ProcessPackets();
+  EXPECT_EQ(blocked_packets, pacer->QueueSizePackets());
+}
+
+TEST_F(PacingControllerTest, DoesNotAllowOveruseAfterCongestion) {
+  uint32_t ssrc = 202020;
+  uint16_t seq_num = 1000;
+  int size = 1000;
+  auto now_ms = [this] { return clock_.TimeInMilliseconds(); };
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  // The pacing rate is low enough that the budget should not allow two packets
+  // to be sent in a row.
+  pacer->SetPacingRates(DataRate::BitsPerSec(400 * 8 * 1000 / 5),
+                        DataRate::Zero());
+  // Not yet budget limited or congested, packet is sent.
+  pacer->EnqueuePacket(
+      BuildPacket(RtpPacketMediaType::kVideo, ssrc, seq_num++, now_ms(), size));
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  clock_.AdvanceTimeMilliseconds(5);
+  pacer->ProcessPackets();
+  // Packet blocked due to congestion.
+  pacer->SetCongested(true);
+  pacer->EnqueuePacket(
+      BuildPacket(RtpPacketMediaType::kVideo, ssrc, seq_num++, now_ms(), size));
+  EXPECT_CALL(callback_, SendPacket).Times(0);
+  clock_.AdvanceTimeMilliseconds(5);
+  pacer->ProcessPackets();
+  // Packet blocked due to congestion.
+  pacer->EnqueuePacket(
+      BuildPacket(RtpPacketMediaType::kVideo, ssrc, seq_num++, now_ms(), size));
+  EXPECT_CALL(callback_, SendPacket).Times(0);
+  clock_.AdvanceTimeMilliseconds(5);
+  pacer->ProcessPackets();
+  // Congestion removed and budget has recovered, packet is sent.
+  pacer->EnqueuePacket(
+      BuildPacket(RtpPacketMediaType::kVideo, ssrc, seq_num++, now_ms(), size));
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  clock_.AdvanceTimeMilliseconds(5);
+  pacer->SetCongested(false);
+  pacer->ProcessPackets();
+  // Should be blocked due to budget limitation as congestion has be removed.
+  pacer->EnqueuePacket(
+      BuildPacket(RtpPacketMediaType::kVideo, ssrc, seq_num++, now_ms(), size));
+  EXPECT_CALL(callback_, SendPacket).Times(0);
+  clock_.AdvanceTimeMilliseconds(5);
+  pacer->ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, Pause) {
+  uint32_t ssrc_low_priority = 12345;
+  uint32_t ssrc = 12346;
+  uint32_t ssrc_high_priority = 12347;
+  uint16_t sequence_number = 1234;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  EXPECT_TRUE(pacer->OldestPacketEnqueueTime().IsInfinite());
+
+  ConsumeInitialBudget(pacer.get());
+
+  pacer->Pause();
+
+  int64_t capture_time_ms = clock_.TimeInMilliseconds();
+  const size_t packets_to_send_per_interval =
+      kTargetRate.bps() * kPaceMultiplier / (8 * 250 * 200);
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo,
+                                     ssrc_low_priority, sequence_number++,
+                                     capture_time_ms, 250));
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kRetransmission, ssrc,
+                                     sequence_number++, capture_time_ms, 250));
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kAudio,
+                                     ssrc_high_priority, sequence_number++,
+                                     capture_time_ms, 250));
+  }
+  clock_.AdvanceTimeMilliseconds(10000);
+  int64_t second_capture_time_ms = clock_.TimeInMilliseconds();
+  for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo,
+                                     ssrc_low_priority, sequence_number++,
+                                     second_capture_time_ms, 250));
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kRetransmission, ssrc,
+                                     sequence_number++, second_capture_time_ms,
+                                     250));
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kAudio,
+                                     ssrc_high_priority, sequence_number++,
+                                     second_capture_time_ms, 250));
+  }
+
+  // Expect everything to be queued.
+  EXPECT_EQ(capture_time_ms, pacer->OldestPacketEnqueueTime().ms());
+
+  // Process triggers keep-alive packet.
+  EXPECT_CALL(callback_, SendPadding).WillOnce([](size_t padding) {
+    return padding;
+  });
+  EXPECT_CALL(callback_, SendPacket(_, _, _, _, true)).Times(1);
+  pacer->ProcessPackets();
+
+  // Verify no packets sent for the rest of the paused process interval.
+  const TimeDelta kProcessInterval = TimeDelta::Millis(5);
+  TimeDelta expected_time_until_send = PacingController::kPausedProcessInterval;
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  while (expected_time_until_send >= kProcessInterval) {
+    pacer->ProcessPackets();
+    clock_.AdvanceTime(kProcessInterval);
+    expected_time_until_send -= kProcessInterval;
+  }
+
+  // New keep-alive packet.
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+  EXPECT_CALL(callback_, SendPadding).WillOnce([](size_t padding) {
+    return padding;
+  });
+  EXPECT_CALL(callback_, SendPacket(_, _, _, _, true)).Times(1);
+  clock_.AdvanceTime(kProcessInterval);
+  pacer->ProcessPackets();
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+
+  // Expect high prio packets to come out first followed by normal
+  // prio packets and low prio packets (all in capture order).
+  {
+    ::testing::InSequence sequence;
+    EXPECT_CALL(callback_,
+                SendPacket(ssrc_high_priority, _, capture_time_ms, _, _))
+        .Times(packets_to_send_per_interval);
+    EXPECT_CALL(callback_,
+                SendPacket(ssrc_high_priority, _, second_capture_time_ms, _, _))
+        .Times(packets_to_send_per_interval);
+
+    for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+      EXPECT_CALL(callback_, SendPacket(ssrc, _, capture_time_ms, _, _))
+          .Times(1);
+    }
+    for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+      EXPECT_CALL(callback_, SendPacket(ssrc, _, second_capture_time_ms, _, _))
+          .Times(1);
+    }
+    for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+      EXPECT_CALL(callback_,
+                  SendPacket(ssrc_low_priority, _, capture_time_ms, _, _))
+          .Times(1);
+    }
+    for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
+      EXPECT_CALL(callback_, SendPacket(ssrc_low_priority, _,
+                                        second_capture_time_ms, _, _))
+          .Times(1);
+    }
+  }
+  pacer->Resume();
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+
+  EXPECT_TRUE(pacer->OldestPacketEnqueueTime().IsInfinite());
+}
+
+TEST_F(PacingControllerTest, InactiveFromStart) {
+  // Recreate the pacer without the inital time forwarding.
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetProbingEnabled(false);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, kTargetRate);
+
+  // No packets sent, there should be no keep-alives sent either.
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  EXPECT_CALL(callback_, SendPacket).Times(0);
+  pacer->ProcessPackets();
+
+  const Timestamp start_time = clock_.CurrentTime();
+
+  // Determine the margin need so we can advance to the last possible moment
+  // that will not cause a process event.
+  const TimeDelta time_margin =
+      PacingController::kMinSleepTime + TimeDelta::Micros(1);
+
+  EXPECT_EQ(pacer->NextSendTime() - start_time,
+            PacingController::kPausedProcessInterval);
+  clock_.AdvanceTime(PacingController::kPausedProcessInterval - time_margin);
+  pacer->ProcessPackets();
+  EXPECT_EQ(pacer->NextSendTime() - start_time,
+            PacingController::kPausedProcessInterval);
+
+  clock_.AdvanceTime(time_margin);
+  pacer->ProcessPackets();
+  EXPECT_EQ(pacer->NextSendTime() - start_time,
+            2 * PacingController::kPausedProcessInterval);
+}
+
+TEST_F(PacingControllerTest, QueueTimeGrowsOverTime) {
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+  EXPECT_TRUE(pacer->OldestPacketEnqueueTime().IsInfinite());
+
+  pacer->SetPacingRates(DataRate::BitsPerSec(30000 * kPaceMultiplier),
+                        DataRate::Zero());
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                      sequence_number, clock_.TimeInMilliseconds(), 1200);
+
+  clock_.AdvanceTimeMilliseconds(500);
+  EXPECT_EQ(clock_.TimeInMilliseconds() - 500,
+            pacer->OldestPacketEnqueueTime().ms());
+  pacer->ProcessPackets();
+  EXPECT_TRUE(pacer->OldestPacketEnqueueTime().IsInfinite());
+}
+
+TEST_F(PacingControllerTest, ProbingWithInsertedPackets) {
+  const size_t kPacketSize = 1200;
+  const int kInitialBitrateBps = 300000;
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+
+  PacingControllerProbing packet_sender;
+  auto pacer =
+      std::make_unique<PacingController>(&clock_, &packet_sender, trials_);
+  std::vector<ProbeClusterConfig> probe_clusters = {
+      {.at_time = clock_.CurrentTime(),
+       .target_data_rate = kFirstClusterRate,
+       .target_duration = TimeDelta::Millis(15),
+       .target_probe_count = 5,
+       .id = 0},
+      {.at_time = clock_.CurrentTime(),
+       .target_data_rate = kSecondClusterRate,
+       .target_duration = TimeDelta::Millis(15),
+       .target_probe_count = 5,
+       .id = 1}};
+  pacer->CreateProbeClusters(probe_clusters);
+  pacer->SetPacingRates(
+      DataRate::BitsPerSec(kInitialBitrateBps * kPaceMultiplier),
+      DataRate::Zero());
+
+  for (int i = 0; i < 10; ++i) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                     sequence_number++,
+                                     clock_.TimeInMilliseconds(), kPacketSize));
+  }
+
+  int64_t start = clock_.TimeInMilliseconds();
+  while (packet_sender.packets_sent() < 5) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+  int packets_sent = packet_sender.packets_sent();
+  // Validate first cluster bitrate. Note that we have to account for number
+  // of intervals and hence (packets_sent - 1) on the first cluster.
+  EXPECT_NEAR((packets_sent - 1) * kPacketSize * 8000 /
+                  (clock_.TimeInMilliseconds() - start),
+              kFirstClusterRate.bps(), kProbingErrorMargin.bps());
+  // Probing always starts with a small padding packet.
+  EXPECT_EQ(1, packet_sender.padding_sent());
+
+  AdvanceTimeUntil(pacer->NextSendTime());
+  start = clock_.TimeInMilliseconds();
+  while (packet_sender.packets_sent() < 10) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+  packets_sent = packet_sender.packets_sent() - packets_sent;
+  // Validate second cluster bitrate.
+  EXPECT_NEAR((packets_sent - 1) * kPacketSize * 8000 /
+                  (clock_.TimeInMilliseconds() - start),
+              kSecondClusterRate.bps(), kProbingErrorMargin.bps());
+}
+
+TEST_F(PacingControllerTest, SkipsProbesWhenProcessIntervalTooLarge) {
+  const size_t kPacketSize = 1200;
+  const int kInitialBitrateBps = 300000;
+  const uint32_t ssrc = 12346;
+  const int kProbeClusterId = 3;
+
+  uint16_t sequence_number = 1234;
+
+  PacingControllerProbing packet_sender;
+
+  const test::ExplicitKeyValueConfig trials(
+      "WebRTC-Bwe-ProbingBehavior/max_probe_delay:2ms/");
+  auto pacer =
+      std::make_unique<PacingController>(&clock_, &packet_sender, trials);
+  pacer->SetPacingRates(
+      DataRate::BitsPerSec(kInitialBitrateBps * kPaceMultiplier),
+      DataRate::BitsPerSec(kInitialBitrateBps));
+
+  for (int i = 0; i < 10; ++i) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                     sequence_number++,
+                                     clock_.TimeInMilliseconds(), kPacketSize));
+  }
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+
+    // Probe at a very high rate.
+    std::vector<ProbeClusterConfig> probe_clusters = {
+        {.at_time = clock_.CurrentTime(),
+         .target_data_rate = DataRate::KilobitsPerSec(10000),  // 10 Mbps,
+         .target_duration = TimeDelta::Millis(15),
+         .target_probe_count = 5,
+         .id = kProbeClusterId}};
+    pacer->CreateProbeClusters(probe_clusters);
+
+    // We need one packet to start the probe.
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                     sequence_number++,
+                                     clock_.TimeInMilliseconds(), kPacketSize));
+    const int packets_sent_before_probe = packet_sender.packets_sent();
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+    EXPECT_EQ(packet_sender.packets_sent(), packets_sent_before_probe + 1);
+
+    // Figure out how long between probe packets.
+    Timestamp start_time = clock_.CurrentTime();
+    AdvanceTimeUntil(pacer->NextSendTime());
+    TimeDelta time_between_probes = clock_.CurrentTime() - start_time;
+    // Advance that distance again + 1ms.
+    clock_.AdvanceTime(time_between_probes);
+
+    // Send second probe packet.
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                     sequence_number++,
+                                     clock_.TimeInMilliseconds(), kPacketSize));
+    pacer->ProcessPackets();
+    EXPECT_EQ(packet_sender.packets_sent(), packets_sent_before_probe + 2);
+    PacedPacketInfo last_pacing_info = packet_sender.last_pacing_info();
+    EXPECT_EQ(last_pacing_info.probe_cluster_id, kProbeClusterId);
+
+    // We're exactly where we should be for the next probe.
+    const Timestamp probe_time = clock_.CurrentTime();
+    EXPECT_EQ(pacer->NextSendTime(), clock_.CurrentTime());
+
+    BitrateProberConfig probing_config(&trials);
+    EXPECT_GT(probing_config.max_probe_delay.Get(), TimeDelta::Zero());
+    // Advance to within max probe delay, should still return same target.
+    clock_.AdvanceTime(probing_config.max_probe_delay.Get());
+    EXPECT_EQ(pacer->NextSendTime(), probe_time);
+
+    // Too high probe delay, drop it!
+    clock_.AdvanceTime(TimeDelta::Micros(1));
+
+    int packets_sent_before_timeout = packet_sender.total_packets_sent();
+    // Expected next process time is unchanged, but calling should not
+    // generate new packets.
+    EXPECT_EQ(pacer->NextSendTime(), probe_time);
+    pacer->ProcessPackets();
+    EXPECT_EQ(packet_sender.total_packets_sent(), packets_sent_before_timeout);
+
+    // Next packet sent is not part of probe.
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+    const int expected_probe_id = PacedPacketInfo::kNotAProbe;
+    EXPECT_EQ(packet_sender.last_pacing_info().probe_cluster_id,
+              expected_probe_id);
+}
+
+TEST_F(PacingControllerTest, ProbingWithPaddingSupport) {
+  const size_t kPacketSize = 1200;
+  const int kInitialBitrateBps = 300000;
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+
+  PacingControllerProbing packet_sender;
+  auto pacer =
+      std::make_unique<PacingController>(&clock_, &packet_sender, trials_);
+  std::vector<ProbeClusterConfig> probe_clusters = {
+      {.at_time = clock_.CurrentTime(),
+       .target_data_rate = kFirstClusterRate,
+       .target_duration = TimeDelta::Millis(15),
+       .target_probe_count = 5,
+       .id = 0}};
+  pacer->CreateProbeClusters(probe_clusters);
+
+  pacer->SetPacingRates(
+      DataRate::BitsPerSec(kInitialBitrateBps * kPaceMultiplier),
+      DataRate::Zero());
+
+  for (int i = 0; i < 3; ++i) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                     sequence_number++,
+                                     clock_.TimeInMilliseconds(), kPacketSize));
+  }
+
+  int64_t start = clock_.TimeInMilliseconds();
+  int process_count = 0;
+  while (process_count < 5) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+    ++process_count;
+  }
+  int packets_sent = packet_sender.packets_sent();
+  int padding_sent = packet_sender.padding_sent();
+  EXPECT_GT(packets_sent, 0);
+  EXPECT_GT(padding_sent, 0);
+  // Note that the number of intervals here for kPacketSize is
+  // packets_sent due to padding in the same cluster.
+  EXPECT_NEAR((packets_sent * kPacketSize * 8000 + padding_sent) /
+                  (clock_.TimeInMilliseconds() - start),
+              kFirstClusterRate.bps(), kProbingErrorMargin.bps());
+}
+
+TEST_F(PacingControllerTest, PaddingOveruse) {
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  const size_t kPacketSize = 1200;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  // Initially no padding rate.
+  pacer->ProcessPackets();
+  pacer->SetPacingRates(DataRate::BitsPerSec(60000 * kPaceMultiplier),
+                        DataRate::Zero());
+
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                      sequence_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize);
+  pacer->ProcessPackets();
+
+  // Add 30kbit padding. When increasing budget, media budget will increase from
+  // negative (overuse) while padding budget will increase from 0.
+  clock_.AdvanceTimeMilliseconds(5);
+  pacer->SetPacingRates(DataRate::BitsPerSec(60000 * kPaceMultiplier),
+                        DataRate::BitsPerSec(30000));
+
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                      sequence_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize);
+  EXPECT_LT(TimeDelta::Millis(5), pacer->ExpectedQueueTime());
+  // Don't send padding if queue is non-empty, even if padding budget > 0.
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, ProbeClusterId) {
+  MockPacketSender callback;
+  uint32_t ssrc = 12346;
+  uint16_t sequence_number = 1234;
+  const size_t kPacketSize = 1200;
+
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback, trials_);
+  pacer->CreateProbeClusters(std::vector<ProbeClusterConfig>(
+      {{.at_time = clock_.CurrentTime(),
+        .target_data_rate = kFirstClusterRate,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 5,
+        .id = 0},
+       {.at_time = clock_.CurrentTime(),
+        .target_data_rate = kSecondClusterRate,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 5,
+        .id = 1}}));
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, kTargetRate);
+  pacer->SetProbingEnabled(true);
+  for (int i = 0; i < 10; ++i) {
+    pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc,
+                                     sequence_number++,
+                                     clock_.TimeInMilliseconds(), kPacketSize));
+  }
+
+  // First probing cluster.
+  EXPECT_CALL(callback,
+              SendPacket(_, Field(&PacedPacketInfo::probe_cluster_id, 0)))
+      .Times(5);
+
+  for (int i = 0; i < 5; ++i) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+
+  // Second probing cluster.
+  EXPECT_CALL(callback,
+              SendPacket(_, Field(&PacedPacketInfo::probe_cluster_id, 1)))
+      .Times(5);
+
+  for (int i = 0; i < 5; ++i) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+
+  // Needed for the Field comparer below.
+  const int kNotAProbe = PacedPacketInfo::kNotAProbe;
+  // No more probing packets.
+  EXPECT_CALL(callback, GeneratePadding).WillOnce([&](DataSize padding_size) {
+    std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
+    padding_packets.emplace_back(
+        BuildPacket(RtpPacketMediaType::kPadding, ssrc, sequence_number++,
+                    clock_.TimeInMilliseconds(), padding_size.bytes()));
+    return padding_packets;
+  });
+  bool non_probe_packet_seen = false;
+  EXPECT_CALL(callback, SendPacket)
+      .WillOnce([&](std::unique_ptr<RtpPacketToSend> packet,
+                    const PacedPacketInfo& cluster_info) {
+        EXPECT_EQ(cluster_info.probe_cluster_id, kNotAProbe);
+        non_probe_packet_seen = true;
+      });
+  while (!non_probe_packet_seen) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+}
+
+TEST_F(PacingControllerTest, OwnedPacketPrioritizedOnType) {
+  MockPacketSender callback;
+  uint32_t ssrc = 123;
+
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback, trials_);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  // Insert a packet of each type, from low to high priority. Since priority
+  // is weighted higher than insert order, these should come out of the pacer
+  // in backwards order with the exception of FEC and Video.
+
+  for (RtpPacketMediaType type :
+       {RtpPacketMediaType::kPadding,
+        RtpPacketMediaType::kForwardErrorCorrection, RtpPacketMediaType::kVideo,
+        RtpPacketMediaType::kRetransmission, RtpPacketMediaType::kAudio}) {
+    pacer->EnqueuePacket(BuildPacket(type, ++ssrc, /*sequence_number=*/123,
+                                     clock_.TimeInMilliseconds(),
+                                     /*size=*/150));
+  }
+
+  ::testing::InSequence seq;
+  EXPECT_CALL(callback,
+              SendPacket(Pointee(Property(&RtpPacketToSend::packet_type,
+                                          RtpPacketMediaType::kAudio)),
+                         _));
+  EXPECT_CALL(callback,
+              SendPacket(Pointee(Property(&RtpPacketToSend::packet_type,
+                                          RtpPacketMediaType::kRetransmission)),
+                         _));
+
+  // FEC and video actually have the same priority, so will come out in
+  // insertion order.
+  EXPECT_CALL(
+      callback,
+      SendPacket(Pointee(Property(&RtpPacketToSend::packet_type,
+                                  RtpPacketMediaType::kForwardErrorCorrection)),
+                 _));
+  EXPECT_CALL(callback,
+              SendPacket(Pointee(Property(&RtpPacketToSend::packet_type,
+                                          RtpPacketMediaType::kVideo)),
+                         _));
+
+  EXPECT_CALL(callback,
+              SendPacket(Pointee(Property(&RtpPacketToSend::packet_type,
+                                          RtpPacketMediaType::kPadding)),
+                         _));
+
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+}
+
+TEST_F(PacingControllerTest, SmallFirstProbePacket) {
+  MockPacketSender callback;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback, trials_);
+  std::vector<ProbeClusterConfig> probe_clusters = {
+      {.at_time = clock_.CurrentTime(),
+       .target_data_rate = kFirstClusterRate,
+       .target_duration = TimeDelta::Millis(15),
+       .target_probe_count = 5,
+       .id = 0}};
+  pacer->CreateProbeClusters(probe_clusters);
+
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+
+  // Add high prio media.
+  pacer->EnqueuePacket(audio_.BuildNextPacket(234));
+
+  // Expect small padding packet to be requested.
+  EXPECT_CALL(callback, GeneratePadding(DataSize::Bytes(1)))
+      .WillOnce([&](DataSize padding_size) {
+        std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
+        padding_packets.emplace_back(
+            BuildPacket(RtpPacketMediaType::kPadding, kAudioSsrc, 1,
+                        clock_.TimeInMilliseconds(), 1));
+        return padding_packets;
+      });
+
+  size_t packets_sent = 0;
+  bool media_seen = false;
+  EXPECT_CALL(callback, SendPacket)
+      .Times(AnyNumber())
+      .WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet,
+                          const PacedPacketInfo& cluster_info) {
+        if (packets_sent == 0) {
+          EXPECT_EQ(packet->packet_type(), RtpPacketMediaType::kPadding);
+        } else {
+          if (packet->packet_type() == RtpPacketMediaType::kAudio) {
+            media_seen = true;
+          }
+        }
+        packets_sent++;
+      });
+  while (!media_seen) {
+    pacer->ProcessPackets();
+    clock_.AdvanceTimeMilliseconds(5);
+  }
+}
+
+TEST_F(PacingControllerTest, TaskLate) {
+  // Set a low send rate to more easily test timing issues.
+  DataRate kSendRate = DataRate::KilobitsPerSec(30);
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetPacingRates(kSendRate, DataRate::Zero());
+
+  // Add four packets of equal size and priority.
+  pacer->EnqueuePacket(video_.BuildNextPacket(1000));
+  pacer->EnqueuePacket(video_.BuildNextPacket(1000));
+  pacer->EnqueuePacket(video_.BuildNextPacket(1000));
+  pacer->EnqueuePacket(video_.BuildNextPacket(1000));
+
+  // Process packets, only first should be sent.
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+  pacer->ProcessPackets();
+
+  Timestamp next_send_time = pacer->NextSendTime();
+  // Determine time between packets (ca 62ms)
+  const TimeDelta time_between_packets = next_send_time - clock_.CurrentTime();
+
+  // Simulate a late process call, executed just before we allow sending the
+  // fourth packet.
+  const TimeDelta kOffset = TimeDelta::Millis(1);
+  clock_.AdvanceTime((time_between_packets * 3) - kOffset);
+
+  EXPECT_CALL(callback_, SendPacket).Times(2);
+  pacer->ProcessPackets();
+
+  // Check that next scheduled send time is in ca 1ms.
+  next_send_time = pacer->NextSendTime();
+  const TimeDelta time_left = next_send_time - clock_.CurrentTime();
+  EXPECT_EQ(time_left.RoundTo(TimeDelta::Millis(1)), kOffset);
+
+  clock_.AdvanceTime(time_left);
+  EXPECT_CALL(callback_, SendPacket);
+  pacer->ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, NoProbingWhilePaused) {
+  uint32_t ssrc = 12345;
+  uint16_t sequence_number = 1234;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  pacer->SetProbingEnabled(true);
+  pacer->SetPacingRates(kTargetRate * kPaceMultiplier, DataRate::Zero());
+  pacer->CreateProbeClusters(std::vector<ProbeClusterConfig>(
+      {{.at_time = clock_.CurrentTime(),
+        .target_data_rate = kFirstClusterRate,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 5,
+        .id = 0},
+       {.at_time = clock_.CurrentTime(),
+        .target_data_rate = kSecondClusterRate,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 5,
+        .id = 1}}));
+
+  // Send at least one packet so probing can initate.
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, ssrc,
+                      sequence_number, clock_.TimeInMilliseconds(), 250);
+  while (pacer->QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer->NextSendTime());
+    pacer->ProcessPackets();
+  }
+
+  // Trigger probing.
+  std::vector<ProbeClusterConfig> probe_clusters = {
+      {.at_time = clock_.CurrentTime(),
+       .target_data_rate = DataRate::KilobitsPerSec(10000),  // 10 Mbps.
+       .target_duration = TimeDelta::Millis(15),
+       .target_probe_count = 5,
+       .id = 3}};
+  pacer->CreateProbeClusters(probe_clusters);
+
+  // Time to next send time should be small.
+  EXPECT_LT(pacer->NextSendTime() - clock_.CurrentTime(),
+            PacingController::kPausedProcessInterval);
+
+  // Pause pacer, time to next send time should now be the pause process
+  // interval.
+  pacer->Pause();
+
+  EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(),
+            PacingController::kPausedProcessInterval);
+}
+
+TEST_F(PacingControllerTest, AudioNotPacedEvenWhenAccountedFor) {
+  const uint32_t kSsrc = 12345;
+  uint16_t sequence_number = 1234;
+  const size_t kPacketSize = 123;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+
+  // Account for audio - so that audio packets can cause pushback on other
+  // types such as video. Audio packet should still be immediated passed
+  // through though ("WebRTC-Pacer-BlockAudio" needs to be enabled in order
+  // to pace audio packets).
+  pacer->SetAccountForAudioPackets(true);
+
+  // Set pacing rate to 1 packet/s, no padding.
+  pacer->SetPacingRates(DataSize::Bytes(kPacketSize) / TimeDelta::Seconds(1),
+                        DataRate::Zero());
+
+  // Add and send an audio packet.
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kAudio, kSsrc,
+                      sequence_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize);
+  pacer->ProcessPackets();
+
+  // Advance time, add another audio packet and process. It should be sent
+  // immediately.
+  clock_.AdvanceTimeMilliseconds(5);
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kAudio, kSsrc,
+                      sequence_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize);
+  pacer->ProcessPackets();
+}
+
+TEST_F(PacingControllerTest,
+       PaddingResumesAfterSaturationEvenWithConcurrentAudio) {
+  const uint32_t kSsrc = 12345;
+  const DataRate kPacingDataRate = DataRate::KilobitsPerSec(125);
+  const DataRate kPaddingDataRate = DataRate::KilobitsPerSec(100);
+  const TimeDelta kMaxBufferInTime = TimeDelta::Millis(500);
+  const DataSize kPacketSize = DataSize::Bytes(130);
+  const TimeDelta kAudioPacketInterval = TimeDelta::Millis(20);
+
+  // In this test, we fist send a burst of video in order to saturate the
+  // padding debt level.
+  // We then proceed to send audio at a bitrate that is slightly lower than
+  // the padding rate, meaning there will be a period with audio but no
+  // padding sent while the debt is draining, then audio and padding will
+  // be interlieved.
+
+  // Verify both with and without accounting for audio.
+  for (bool account_for_audio : {false, true}) {
+    uint16_t sequence_number = 1234;
+    MockPacketSender callback;
+    EXPECT_CALL(callback, SendPacket).Times(AnyNumber());
+    auto pacer =
+        std::make_unique<PacingController>(&clock_, &callback, trials_);
+    pacer->SetAccountForAudioPackets(account_for_audio);
+
+    // First, saturate the padding budget.
+    pacer->SetPacingRates(kPacingDataRate, kPaddingDataRate);
+
+    const TimeDelta kPaddingSaturationTime =
+        kMaxBufferInTime * kPaddingDataRate /
+        (kPacingDataRate - kPaddingDataRate);
+    const DataSize kVideoToSend = kPaddingSaturationTime * kPacingDataRate;
+    const DataSize kVideoPacketSize = DataSize::Bytes(1200);
+    DataSize video_sent = DataSize::Zero();
+    while (video_sent < kVideoToSend) {
+      pacer->EnqueuePacket(
+          BuildPacket(RtpPacketMediaType::kVideo, kSsrc, sequence_number++,
+                      clock_.TimeInMilliseconds(), kVideoPacketSize.bytes()));
+      video_sent += kVideoPacketSize;
+    }
+    while (pacer->QueueSizePackets() > 0) {
+      AdvanceTimeUntil(pacer->NextSendTime());
+      pacer->ProcessPackets();
+    }
+
+    // Add a stream of audio packets at a rate slightly lower than the padding
+    // rate, once the padding debt is paid off we expect padding to be
+    // generated.
+    pacer->SetPacingRates(kPacingDataRate, kPaddingDataRate);
+    bool padding_seen = false;
+    EXPECT_CALL(callback, GeneratePadding).WillOnce([&](DataSize padding_size) {
+      padding_seen = true;
+      std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
+      padding_packets.emplace_back(
+          BuildPacket(RtpPacketMediaType::kPadding, kSsrc, sequence_number++,
+                      clock_.TimeInMilliseconds(), padding_size.bytes()));
+      return padding_packets;
+    });
+
+    Timestamp start_time = clock_.CurrentTime();
+    Timestamp last_audio_time = start_time;
+    while (!padding_seen) {
+      Timestamp now = clock_.CurrentTime();
+      Timestamp next_send_time = pacer->NextSendTime();
+      TimeDelta sleep_time =
+          std::min(next_send_time, last_audio_time + kAudioPacketInterval) -
+          now;
+      clock_.AdvanceTime(sleep_time);
+      while (clock_.CurrentTime() >= last_audio_time + kAudioPacketInterval) {
+        pacer->EnqueuePacket(
+            BuildPacket(RtpPacketMediaType::kAudio, kSsrc, sequence_number++,
+                        clock_.TimeInMilliseconds(), kPacketSize.bytes()));
+        last_audio_time += kAudioPacketInterval;
+      }
+      pacer->ProcessPackets();
+    }
+
+    // Verify how long it took to drain the padding debt. Allow 2% error margin.
+    const DataRate kAudioDataRate = kPacketSize / kAudioPacketInterval;
+    const TimeDelta expected_drain_time =
+        account_for_audio ? (kMaxBufferInTime * kPaddingDataRate /
+                             (kPaddingDataRate - kAudioDataRate))
+                          : kMaxBufferInTime;
+    const TimeDelta actual_drain_time = clock_.CurrentTime() - start_time;
+    EXPECT_NEAR(actual_drain_time.ms(), expected_drain_time.ms(),
+                expected_drain_time.ms() * 0.02)
+        << " where account_for_audio = "
+        << (account_for_audio ? "true" : "false");
+  }
+}
+
+TEST_F(PacingControllerTest, AccountsForAudioEnqueueTime) {
+  const uint32_t kSsrc = 12345;
+  const DataRate kPacingDataRate = DataRate::KilobitsPerSec(125);
+  const DataRate kPaddingDataRate = DataRate::Zero();
+  const DataSize kPacketSize = DataSize::Bytes(130);
+  const TimeDelta kPacketPacingTime = kPacketSize / kPacingDataRate;
+  uint32_t sequnce_number = 1;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+  // Audio not paced, but still accounted for in budget.
+  pacer->SetAccountForAudioPackets(true);
+  pacer->SetPacingRates(kPacingDataRate, kPaddingDataRate);
+
+  // Enqueue two audio packets, advance clock to where one packet
+  // should have drained the buffer already, has they been sent
+  // immediately.
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kAudio, kSsrc,
+                      sequnce_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize.bytes());
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kAudio, kSsrc,
+                      sequnce_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize.bytes());
+  clock_.AdvanceTime(kPacketPacingTime);
+  // Now process and make sure both packets were sent.
+  pacer->ProcessPackets();
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+
+  // Add a video packet. I can't be sent until debt from audio
+  // packets have been drained.
+  pacer->EnqueuePacket(
+      BuildPacket(RtpPacketMediaType::kVideo, kSsrc + 1, sequnce_number++,
+                  clock_.TimeInMilliseconds(), kPacketSize.bytes()));
+  EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(), kPacketPacingTime);
+}
+
+TEST_F(PacingControllerTest, NextSendTimeAccountsForPadding) {
+  const uint32_t kSsrc = 12345;
+  const DataRate kPacingDataRate = DataRate::KilobitsPerSec(125);
+  const DataSize kPacketSize = DataSize::Bytes(130);
+  const TimeDelta kPacketPacingTime = kPacketSize / kPacingDataRate;
+  uint32_t sequnce_number = 1;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+
+  // Start with no padding.
+  pacer->SetPacingRates(kPacingDataRate, DataRate::Zero());
+
+  // Send a single packet.
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, kSsrc,
+                      sequnce_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize.bytes());
+  pacer->ProcessPackets();
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+
+  // With current conditions, no need to wake until next keep-alive.
+  EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(),
+            PacingController::kPausedProcessInterval);
+
+  // Enqueue a new packet, that can't be sent until previous buffer has
+  // drained.
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, kSsrc,
+                      sequnce_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize.bytes());
+  EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(), kPacketPacingTime);
+  clock_.AdvanceTime(kPacketPacingTime);
+  pacer->ProcessPackets();
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+
+  // With current conditions, again no need to wake until next keep-alive.
+  EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(),
+            PacingController::kPausedProcessInterval);
+
+  // Set a non-zero padding rate. Padding also can't be sent until
+  // previous debt has cleared. Since padding was disabled before, there
+  // currently is no padding debt.
+  pacer->SetPacingRates(kPacingDataRate, kPacingDataRate / 2);
+  EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(), kPacketPacingTime);
+
+  // Advance time, expect padding.
+  EXPECT_CALL(callback_, SendPadding).WillOnce(Return(kPacketSize.bytes()));
+  clock_.AdvanceTime(kPacketPacingTime);
+  pacer->ProcessPackets();
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+
+  // Since padding rate is half of pacing rate, next time we can send
+  // padding is double the packet pacing time.
+  EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(),
+            kPacketPacingTime * 2);
+
+  // Insert a packet to be sent, this take precedence again.
+  pacer->EnqueuePacket(
+      BuildPacket(RtpPacketMediaType::kVideo, kSsrc, sequnce_number++,
+                  clock_.TimeInMilliseconds(), kPacketSize.bytes()));
+  EXPECT_EQ(pacer->NextSendTime() - clock_.CurrentTime(), kPacketPacingTime);
+}
+
+TEST_F(PacingControllerTest, PaddingTargetAccountsForPaddingRate) {
+  // Target size for a padding packet is 5ms * padding rate.
+  const TimeDelta kPaddingTarget = TimeDelta::Millis(5);
+  srand(0);
+  // Need to initialize PacingController after we initialize clock.
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+
+  const uint32_t kSsrc = 12345;
+  const DataRate kPacingDataRate = DataRate::KilobitsPerSec(125);
+  const DataSize kPacketSize = DataSize::Bytes(130);
+
+  uint32_t sequnce_number = 1;
+
+  // Start with pacing and padding rate equal.
+  pacer->SetPacingRates(kPacingDataRate, kPacingDataRate);
+
+  // Send a single packet.
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, kSsrc,
+                      sequnce_number++, clock_.TimeInMilliseconds(),
+                      kPacketSize.bytes());
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+
+  size_t expected_padding_target_bytes =
+      (kPaddingTarget * kPacingDataRate).bytes();
+  EXPECT_CALL(callback_, SendPadding(expected_padding_target_bytes))
+      .WillOnce(Return(expected_padding_target_bytes));
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+
+  // Half the padding rate - expect half the padding target.
+  pacer->SetPacingRates(kPacingDataRate, kPacingDataRate / 2);
+  EXPECT_CALL(callback_, SendPadding(expected_padding_target_bytes / 2))
+      .WillOnce(Return(expected_padding_target_bytes / 2));
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, SendsFecPackets) {
+  const uint32_t kSsrc = 12345;
+  const uint32_t kFlexSsrc = 54321;
+  uint16_t sequence_number = 1234;
+  uint16_t flexfec_sequence_number = 4321;
+  const size_t kPacketSize = 123;
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+
+  // Set pacing rate to 1000 packet/s, no padding.
+  pacer->SetPacingRates(
+      DataSize::Bytes(1000 * kPacketSize) / TimeDelta::Seconds(1),
+      DataRate::Zero());
+
+  int64_t now = clock_.TimeInMilliseconds();
+  pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kSsrc,
+                                   sequence_number, now, kPacketSize));
+  EXPECT_CALL(callback_, SendPacket(kSsrc, sequence_number, now, false, false));
+  EXPECT_CALL(callback_, FetchFec).WillOnce([&]() {
+    EXPECT_CALL(callback_, SendPacket(kFlexSsrc, flexfec_sequence_number, now,
+                                      false, false));
+    EXPECT_CALL(callback_, FetchFec);
+    std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets;
+    fec_packets.push_back(
+        BuildPacket(RtpPacketMediaType::kForwardErrorCorrection, kFlexSsrc,
+                    flexfec_sequence_number, now, kPacketSize));
+    return fec_packets;
+  });
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+  AdvanceTimeUntil(pacer->NextSendTime());
+  pacer->ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, GapInPacingDoesntAccumulateBudget) {
+  const uint32_t kSsrc = 12345;
+  uint16_t sequence_number = 1234;
+  const DataSize kPackeSize = DataSize::Bytes(250);
+  const TimeDelta kPacketSendTime = TimeDelta::Millis(15);
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+
+  pacer->SetPacingRates(kPackeSize / kPacketSendTime,
+                        /*padding_rate=*/DataRate::Zero());
+
+  // Send an initial packet.
+  SendAndExpectPacket(pacer.get(), RtpPacketMediaType::kVideo, kSsrc,
+                      sequence_number++, clock_.TimeInMilliseconds(),
+                      kPackeSize.bytes());
+  pacer->ProcessPackets();
+  ::testing::Mock::VerifyAndClearExpectations(&callback_);
+
+  // Advance time kPacketSendTime past where the media debt should be 0.
+  clock_.AdvanceTime(2 * kPacketSendTime);
+
+  // Enqueue two new packets. Expect only one to be sent one ProcessPackets().
+  pacer->EnqueuePacket(
+      BuildPacket(RtpPacketMediaType::kVideo, kSsrc, sequence_number + 1,
+                  clock_.TimeInMilliseconds(), kPackeSize.bytes()));
+  pacer->EnqueuePacket(
+      BuildPacket(RtpPacketMediaType::kVideo, kSsrc, sequence_number + 2,
+                  clock_.TimeInMilliseconds(), kPackeSize.bytes()));
+  EXPECT_CALL(callback_, SendPacket(kSsrc, sequence_number + 1,
+                                    clock_.TimeInMilliseconds(), false, false));
+  pacer->ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, HandlesSubMicrosecondSendIntervals) {
+  static constexpr DataSize kPacketSize = DataSize::Bytes(1);
+  static constexpr TimeDelta kPacketSendTime = TimeDelta::Micros(1);
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+
+  // Set pacing rate such that a packet is sent in 0.5us.
+  pacer->SetPacingRates(/*pacing_rate=*/2 * kPacketSize / kPacketSendTime,
+                        /*padding_rate=*/DataRate::Zero());
+
+  // Enqueue three packets, the first two should be sent immediately - the third
+  // should cause a non-zero delta to the next process time.
+  EXPECT_CALL(callback_, SendPacket).Times(2);
+  for (int i = 0; i < 3; ++i) {
+    pacer->EnqueuePacket(BuildPacket(
+        RtpPacketMediaType::kVideo, /*ssrc=*/12345, /*sequence_number=*/i,
+        clock_.TimeInMilliseconds(), kPacketSize.bytes()));
+  }
+  pacer->ProcessPackets();
+
+  EXPECT_GT(pacer->NextSendTime(), clock_.CurrentTime());
+}
+
+TEST_F(PacingControllerTest, HandlesSubMicrosecondPaddingInterval) {
+  static constexpr DataSize kPacketSize = DataSize::Bytes(1);
+  static constexpr TimeDelta kPacketSendTime = TimeDelta::Micros(1);
+  auto pacer = std::make_unique<PacingController>(&clock_, &callback_, trials_);
+
+  // Set both pacing and padding rates to 1 byte per 0.5us.
+  pacer->SetPacingRates(/*pacing_rate=*/2 * kPacketSize / kPacketSendTime,
+                        /*padding_rate=*/2 * kPacketSize / kPacketSendTime);
+
+  // Enqueue and send one packet.
+  EXPECT_CALL(callback_, SendPacket);
+  pacer->EnqueuePacket(BuildPacket(
+      RtpPacketMediaType::kVideo, /*ssrc=*/12345, /*sequence_number=*/1234,
+      clock_.TimeInMilliseconds(), kPacketSize.bytes()));
+  pacer->ProcessPackets();
+
+  // The padding debt is now 1 byte, and the pacing time for that is lower than
+  // the precision of a TimeStamp tick. Make sure the pacer still indicates a
+  // non-zero sleep time is needed until the next process.
+  EXPECT_GT(pacer->NextSendTime(), clock_.CurrentTime());
+}
+
+TEST_F(PacingControllerTest, SendsPacketsInBurstImmediately) {
+  constexpr TimeDelta kMaxDelay = TimeDelta::Millis(20);
+  PacingController pacer(&clock_, &callback_, trials_);
+  pacer.SetSendBurstInterval(kMaxDelay);
+  pacer.SetPacingRates(DataRate::BytesPerSec(10000), DataRate::Zero());
+
+  // Max allowed send burst size is 100000*20/1000) = 200byte
+  pacer.EnqueuePacket(video_.BuildNextPacket(100));
+  pacer.EnqueuePacket(video_.BuildNextPacket(100));
+  pacer.EnqueuePacket(video_.BuildNextPacket(100));
+  pacer.ProcessPackets();
+  EXPECT_EQ(pacer.QueueSizePackets(), 1u);
+  EXPECT_EQ(pacer.NextSendTime(), clock_.CurrentTime() + kMaxDelay);
+
+  AdvanceTimeUntil(pacer.NextSendTime());
+  pacer.ProcessPackets();
+  EXPECT_EQ(pacer.QueueSizePackets(), 0u);
+}
+
+TEST_F(PacingControllerTest, SendsPacketsInBurstEvenIfNotEnqueedAtSameTime) {
+  constexpr TimeDelta kMaxDelay = TimeDelta::Millis(20);
+  PacingController pacer(&clock_, &callback_, trials_);
+  pacer.SetSendBurstInterval(kMaxDelay);
+  pacer.SetPacingRates(DataRate::BytesPerSec(10000), DataRate::Zero());
+  pacer.EnqueuePacket(video_.BuildNextPacket(200));
+  EXPECT_EQ(pacer.NextSendTime(), clock_.CurrentTime());
+  pacer.ProcessPackets();
+  clock_.AdvanceTime(TimeDelta::Millis(1));
+  pacer.EnqueuePacket(video_.BuildNextPacket(200));
+  EXPECT_EQ(pacer.NextSendTime(), clock_.CurrentTime());
+  pacer.ProcessPackets();
+  EXPECT_EQ(pacer.QueueSizePackets(), 0u);
+}
+
+TEST_F(PacingControllerTest, RespectsTargetRateWhenSendingPacketsInBursts) {
+  PacingController pacer(&clock_, &callback_, trials_);
+  pacer.SetSendBurstInterval(TimeDelta::Millis(20));
+  pacer.SetAccountForAudioPackets(true);
+  pacer.SetPacingRates(DataRate::KilobitsPerSec(1000), DataRate::Zero());
+  Timestamp start_time = clock_.CurrentTime();
+  // Inject 100 packets, with size 1000bytes over 100ms.
+  // Expect only 1Mbps / (8*1000) / 10 =  12 packets to be sent.
+  // Packets are sent in burst. Each burst is then 3 packets * 1000bytes at
+  // 1Mbits = 24ms long. Thus, expect 4 bursts.
+  EXPECT_CALL(callback_, SendPacket).Times(12);
+  int number_of_bursts = 0;
+  while (clock_.CurrentTime() < start_time + TimeDelta::Millis(100)) {
+    pacer.EnqueuePacket(video_.BuildNextPacket(1000));
+    pacer.EnqueuePacket(video_.BuildNextPacket(1000));
+    pacer.EnqueuePacket(video_.BuildNextPacket(1000));
+    pacer.EnqueuePacket(video_.BuildNextPacket(1000));
+    pacer.EnqueuePacket(video_.BuildNextPacket(1000));
+    if (pacer.NextSendTime() <= clock_.CurrentTime()) {
+      pacer.ProcessPackets();
+      ++number_of_bursts;
+    }
+    clock_.AdvanceTime(TimeDelta::Millis(5));
+  }
+  EXPECT_EQ(pacer.QueueSizePackets(), 88u);
+  EXPECT_EQ(number_of_bursts, 4);
+}
+
+TEST_F(PacingControllerTest, RespectsQueueTimeLimit) {
+  static constexpr DataSize kPacketSize = DataSize::Bytes(100);
+  static constexpr DataRate kNominalPacingRate = DataRate::KilobitsPerSec(200);
+  static constexpr TimeDelta kPacketPacingTime =
+      kPacketSize / kNominalPacingRate;
+  static constexpr TimeDelta kQueueTimeLimit = TimeDelta::Millis(1000);
+
+  PacingController pacer(&clock_, &callback_, trials_);
+  pacer.SetPacingRates(kNominalPacingRate, /*padding_rate=*/DataRate::Zero());
+  pacer.SetQueueTimeLimit(kQueueTimeLimit);
+
+  // Fill pacer up to queue time limit.
+  static constexpr int kNumPackets = kQueueTimeLimit / kPacketPacingTime;
+  for (int i = 0; i < kNumPackets; ++i) {
+    pacer.EnqueuePacket(video_.BuildNextPacket(kPacketSize.bytes()));
+  }
+  EXPECT_EQ(pacer.ExpectedQueueTime(), kQueueTimeLimit);
+  EXPECT_EQ(pacer.pacing_rate(), kNominalPacingRate);
+
+  // Double the amount of packets in the queue, the queue time limit should
+  // effectively double the pacing rate in response.
+  for (int i = 0; i < kNumPackets; ++i) {
+    pacer.EnqueuePacket(video_.BuildNextPacket(kPacketSize.bytes()));
+  }
+  EXPECT_EQ(pacer.ExpectedQueueTime(), kQueueTimeLimit);
+  EXPECT_EQ(pacer.pacing_rate(), 2 * kNominalPacingRate);
+
+  // Send all the packets, should take as long as the queue time limit.
+  Timestamp start_time = clock_.CurrentTime();
+  while (pacer.QueueSizePackets() > 0) {
+    AdvanceTimeUntil(pacer.NextSendTime());
+    pacer.ProcessPackets();
+  }
+  EXPECT_EQ(clock_.CurrentTime() - start_time, kQueueTimeLimit);
+
+  // We're back in a normal state - pacing rate should be back to previous
+  // levels.
+  EXPECT_EQ(pacer.pacing_rate(), kNominalPacingRate);
+}
+
+TEST_F(PacingControllerTest, BudgetDoesNotAffectRetransmissionInsTrial) {
+  const DataSize kPacketSize = DataSize::Bytes(1000);
+
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  const test::ExplicitKeyValueConfig trials(
+      "WebRTC-Pacer-FastRetransmissions/Enabled/");
+  PacingController pacer(&clock_, &callback_, trials);
+  pacer.SetPacingRates(kTargetRate, /*padding_rate=*/DataRate::Zero());
+
+  // Send a video packet so that we have a bit debt.
+  pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kVideoSsrc,
+                                  /*sequence_number=*/1,
+                                  /*capture_time=*/1, kPacketSize.bytes()));
+  EXPECT_CALL(callback_, SendPacket);
+  pacer.ProcessPackets();
+  EXPECT_GT(pacer.NextSendTime(), clock_.CurrentTime());
+
+  // A retransmission packet should still be immediately processed.
+  EXPECT_CALL(callback_, SendPacket);
+  pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kRetransmission,
+                                  kVideoSsrc,
+                                  /*sequence_number=*/1,
+                                  /*capture_time=*/1, kPacketSize.bytes()));
+  pacer.ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, AbortsAfterReachingCircuitBreakLimit) {
+  const DataSize kPacketSize = DataSize::Bytes(1000);
+
+  EXPECT_CALL(callback_, SendPadding).Times(0);
+  PacingController pacer(&clock_, &callback_, trials_);
+  pacer.SetPacingRates(kTargetRate, /*padding_rate=*/DataRate::Zero());
+
+  // Set the circuit breaker to abort after one iteration of the main
+  // sending loop.
+  pacer.SetCircuitBreakerThreshold(1);
+  EXPECT_CALL(callback_, SendPacket).Times(1);
+
+  // Send two packets.
+  pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kVideoSsrc,
+                                  /*sequence_number=*/1,
+                                  /*capture_time=*/1, kPacketSize.bytes()));
+  pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kVideoSsrc,
+                                  /*sequence_number=*/2,
+                                  /*capture_time=*/2, kPacketSize.bytes()));
+
+  // Advance time to way past where both should be eligible for sending.
+  clock_.AdvanceTime(TimeDelta::Seconds(1));
+
+  pacer.ProcessPackets();
+}
+
+TEST_F(PacingControllerTest, DoesNotPadIfProcessThreadIsBorked) {
+  PacingControllerPadding callback;
+  PacingController pacer(&clock_, &callback, trials_);
+
+  // Set both pacing and padding rate to be non-zero.
+  pacer.SetPacingRates(kTargetRate, /*padding_rate=*/kTargetRate);
+
+  // Add one packet to the queue, but do not send it yet.
+  pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kVideoSsrc,
+                                  /*sequence_number=*/1,
+                                  /*capture_time=*/1,
+                                  /*size=*/1000));
+
+  // Advance time to waaay after the packet should have been sent.
+  clock_.AdvanceTime(TimeDelta::Seconds(42));
+
+  // `ProcessPackets()` should send the delayed packet, followed by a small
+  // amount of missed padding.
+  pacer.ProcessPackets();
+
+  // The max padding window is the max replay duration + the target padding
+  // duration.
+  const DataSize kMaxPadding = (PacingController::kMaxPaddingReplayDuration +
+                                PacingController::kTargetPaddingDuration) *
+                               kTargetRate;
+
+  EXPECT_LE(callback.padding_sent(), kMaxPadding.bytes<size_t>());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/pacing_gn/moz.build b/third_party/libwebrtc/modules/pacing/pacing_gn/moz.build
new file mode 100644
index 0000000000..61ece82430
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/pacing_gn/moz.build
@@ -0,0 +1,239 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/pacing/bitrate_prober.cc",
+    "/third_party/libwebrtc/modules/pacing/pacing_controller.cc",
+    "/third_party/libwebrtc/modules/pacing/packet_router.cc",
+    "/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("pacing_gn")
diff --git a/third_party/libwebrtc/modules/pacing/packet_router.cc b/third_party/libwebrtc/modules/pacing/packet_router.cc
new file mode 100644
index 0000000000..b28d9776dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/packet_router.cc
@@ -0,0 +1,356 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/packet_router.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/system/unused.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+PacketRouter::PacketRouter() : PacketRouter(0) {}
+
+PacketRouter::PacketRouter(uint16_t start_transport_seq)
+    : last_send_module_(nullptr),
+      active_remb_module_(nullptr),
+      transport_seq_(start_transport_seq) {}
+
+PacketRouter::~PacketRouter() {
+  RTC_DCHECK(send_modules_map_.empty());
+  RTC_DCHECK(send_modules_list_.empty());
+  RTC_DCHECK(rtcp_feedback_senders_.empty());
+  RTC_DCHECK(sender_remb_candidates_.empty());
+  RTC_DCHECK(receiver_remb_candidates_.empty());
+  RTC_DCHECK(active_remb_module_ == nullptr);
+}
+
+void PacketRouter::AddSendRtpModule(RtpRtcpInterface* rtp_module,
+                                    bool remb_candidate) {
+  MutexLock lock(&modules_mutex_);
+
+  AddSendRtpModuleToMap(rtp_module, rtp_module->SSRC());
+  if (absl::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
+    AddSendRtpModuleToMap(rtp_module, *rtx_ssrc);
+  }
+  if (absl::optional<uint32_t> flexfec_ssrc = rtp_module->FlexfecSsrc()) {
+    AddSendRtpModuleToMap(rtp_module, *flexfec_ssrc);
+  }
+
+  if (rtp_module->SupportsRtxPayloadPadding()) {
+    last_send_module_ = rtp_module;
+  }
+
+  if (remb_candidate) {
+    AddRembModuleCandidate(rtp_module, /* media_sender = */ true);
+  }
+}
+
+void PacketRouter::AddSendRtpModuleToMap(RtpRtcpInterface* rtp_module,
+                                         uint32_t ssrc) {
+  RTC_DCHECK(send_modules_map_.find(ssrc) == send_modules_map_.end());
+
+  // Signal to module that the pacer thread is attached and can send packets.
+  rtp_module->OnPacketSendingThreadSwitched();
+
+  // Always keep the audio modules at the back of the list, so that when we
+  // iterate over the modules in order to find one that can send padding we
+  // will prioritize video. This is important to make sure they are counted
+  // into the bandwidth estimate properly.
+  if (rtp_module->IsAudioConfigured()) {
+    send_modules_list_.push_back(rtp_module);
+  } else {
+    send_modules_list_.push_front(rtp_module);
+  }
+  send_modules_map_[ssrc] = rtp_module;
+}
+
+void PacketRouter::RemoveSendRtpModuleFromMap(uint32_t ssrc) {
+  auto it = send_modules_map_.find(ssrc);
+  RTC_DCHECK(it != send_modules_map_.end());
+  send_modules_list_.remove(it->second);
+  send_modules_map_.erase(it);
+}
+
+void PacketRouter::RemoveSendRtpModule(RtpRtcpInterface* rtp_module) {
+  MutexLock lock(&modules_mutex_);
+  MaybeRemoveRembModuleCandidate(rtp_module, /* media_sender = */ true);
+
+  RemoveSendRtpModuleFromMap(rtp_module->SSRC());
+  if (absl::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
+    RemoveSendRtpModuleFromMap(*rtx_ssrc);
+  }
+  if (absl::optional<uint32_t> flexfec_ssrc = rtp_module->FlexfecSsrc()) {
+    RemoveSendRtpModuleFromMap(*flexfec_ssrc);
+  }
+
+  if (last_send_module_ == rtp_module) {
+    last_send_module_ = nullptr;
+  }
+  rtp_module->OnPacketSendingThreadSwitched();
+}
+
+void PacketRouter::AddReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender,
+                                       bool remb_candidate) {
+  MutexLock lock(&modules_mutex_);
+  RTC_DCHECK(std::find(rtcp_feedback_senders_.begin(),
+                       rtcp_feedback_senders_.end(),
+                       rtcp_sender) == rtcp_feedback_senders_.end());
+
+  rtcp_feedback_senders_.push_back(rtcp_sender);
+
+  if (remb_candidate) {
+    AddRembModuleCandidate(rtcp_sender, /* media_sender = */ false);
+  }
+}
+
+void PacketRouter::RemoveReceiveRtpModule(
+    RtcpFeedbackSenderInterface* rtcp_sender) {
+  MutexLock lock(&modules_mutex_);
+  MaybeRemoveRembModuleCandidate(rtcp_sender, /* media_sender = */ false);
+  auto it = std::find(rtcp_feedback_senders_.begin(),
+                      rtcp_feedback_senders_.end(), rtcp_sender);
+  RTC_DCHECK(it != rtcp_feedback_senders_.end());
+  rtcp_feedback_senders_.erase(it);
+}
+
+void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
+                              const PacedPacketInfo& cluster_info) {
+  TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"), "PacketRouter::SendPacket",
+               "sequence_number", packet->SequenceNumber(), "rtp_timestamp",
+               packet->Timestamp());
+
+  MutexLock lock(&modules_mutex_);
+  // With the new pacer code path, transport sequence numbers are only set here,
+  // on the pacer thread. Therefore we don't need atomics/synchronization.
+  bool assign_transport_sequence_number =
+      packet->HasExtension<TransportSequenceNumber>();
+  if (assign_transport_sequence_number) {
+    packet->SetExtension<TransportSequenceNumber>((transport_seq_ + 1) &
+                                                  0xFFFF);
+  }
+
+  uint32_t ssrc = packet->Ssrc();
+  auto it = send_modules_map_.find(ssrc);
+  if (it == send_modules_map_.end()) {
+    RTC_LOG(LS_WARNING)
+        << "Failed to send packet, matching RTP module not found "
+           "or transport error. SSRC = "
+        << packet->Ssrc() << ", sequence number " << packet->SequenceNumber();
+    return;
+  }
+
+  RtpRtcpInterface* rtp_module = it->second;
+  if (!rtp_module->TrySendPacket(packet.get(), cluster_info)) {
+    RTC_LOG(LS_WARNING) << "Failed to send packet, rejected by RTP module.";
+    return;
+  }
+
+  // Sending succeeded.
+
+  if (assign_transport_sequence_number) {
+    ++transport_seq_;
+  }
+
+  if (rtp_module->SupportsRtxPayloadPadding()) {
+    // This is now the last module to send media, and has the desired
+    // properties needed for payload based padding. Cache it for later use.
+    last_send_module_ = rtp_module;
+  }
+
+  for (auto& packet : rtp_module->FetchFecPackets()) {
+    pending_fec_packets_.push_back(std::move(packet));
+  }
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::FetchFec() {
+  MutexLock lock(&modules_mutex_);
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+      std::move(pending_fec_packets_);
+  pending_fec_packets_.clear();
+  return fec_packets;
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding(
+    DataSize size) {
+  TRACE_EVENT1(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+               "PacketRouter::GeneratePadding", "bytes", size.bytes());
+
+  MutexLock lock(&modules_mutex_);
+  // First try on the last rtp module to have sent media. This increases the
+  // the chance that any payload based padding will be useful as it will be
+  // somewhat distributed over modules according the packet rate, even if it
+  // will be more skewed towards the highest bitrate stream. At the very least
+  // this prevents sending payload padding on a disabled stream where it's
+  // guaranteed not to be useful.
+  std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
+  if (last_send_module_ != nullptr &&
+      last_send_module_->SupportsRtxPayloadPadding()) {
+    padding_packets = last_send_module_->GeneratePadding(size.bytes());
+  }
+
+  if (padding_packets.empty()) {
+    // Iterate over all modules send module. Video modules will be at the front
+    // and so will be prioritized. This is important since audio packets may not
+    // be taken into account by the bandwidth estimator, e.g. in FF.
+    for (RtpRtcpInterface* rtp_module : send_modules_list_) {
+      if (rtp_module->SupportsPadding()) {
+        padding_packets = rtp_module->GeneratePadding(size.bytes());
+        if (!padding_packets.empty()) {
+          last_send_module_ = rtp_module;
+          break;
+        }
+      }
+    }
+  }
+
+  for (auto& packet : padding_packets) {
+    RTC_UNUSED(packet);
+    TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+                 "PacketRouter::GeneratePadding::Loop", "sequence_number",
+                 packet->SequenceNumber(), "rtp_timestamp",
+                 packet->Timestamp());
+  }
+
+  return padding_packets;
+}
+
+void PacketRouter::OnAbortedRetransmissions(
+    uint32_t ssrc,
+    rtc::ArrayView<const uint16_t> sequence_numbers) {
+  MutexLock lock(&modules_mutex_);
+  auto it = send_modules_map_.find(ssrc);
+  if (it != send_modules_map_.end()) {
+    it->second->OnAbortedRetransmissions(sequence_numbers);
+  }
+}
+
+absl::optional<uint32_t> PacketRouter::GetRtxSsrcForMedia(uint32_t ssrc) const {
+  MutexLock lock(&modules_mutex_);
+  auto it = send_modules_map_.find(ssrc);
+  if (it != send_modules_map_.end() && it->second->SSRC() == ssrc) {
+    // A module is registered with the given SSRC, and that SSRC is the main
+    // media SSRC for that RTP module.
+    return it->second->RtxSsrc();
+  }
+  return absl::nullopt;
+}
+
+uint16_t PacketRouter::CurrentTransportSequenceNumber() const {
+  MutexLock lock(&modules_mutex_);
+  return transport_seq_ & 0xFFFF;
+}
+
+void PacketRouter::SendRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) {
+  MutexLock lock(&modules_mutex_);
+
+  if (!active_remb_module_) {
+    return;
+  }
+
+  // The Add* and Remove* methods above ensure that REMB is disabled on all
+  // other modules, because otherwise, they will send REMB with stale info.
+  active_remb_module_->SetRemb(bitrate_bps, std::move(ssrcs));
+}
+
+void PacketRouter::SendCombinedRtcpPacket(
+    std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets) {
+  MutexLock lock(&modules_mutex_);
+
+  // Prefer send modules.
+  for (RtpRtcpInterface* rtp_module : send_modules_list_) {
+    if (rtp_module->RTCP() == RtcpMode::kOff) {
+      continue;
+    }
+    rtp_module->SendCombinedRtcpPacket(std::move(packets));
+    return;
+  }
+
+  if (rtcp_feedback_senders_.empty()) {
+    return;
+  }
+  auto* rtcp_sender = rtcp_feedback_senders_[0];
+  rtcp_sender->SendCombinedRtcpPacket(std::move(packets));
+}
+
+void PacketRouter::AddRembModuleCandidate(
+    RtcpFeedbackSenderInterface* candidate_module,
+    bool media_sender) {
+  RTC_DCHECK(candidate_module);
+  std::vector<RtcpFeedbackSenderInterface*>& candidates =
+      media_sender ? sender_remb_candidates_ : receiver_remb_candidates_;
+  RTC_DCHECK(std::find(candidates.cbegin(), candidates.cend(),
+                       candidate_module) == candidates.cend());
+  candidates.push_back(candidate_module);
+  DetermineActiveRembModule();
+}
+
+void PacketRouter::MaybeRemoveRembModuleCandidate(
+    RtcpFeedbackSenderInterface* candidate_module,
+    bool media_sender) {
+  RTC_DCHECK(candidate_module);
+  std::vector<RtcpFeedbackSenderInterface*>& candidates =
+      media_sender ? sender_remb_candidates_ : receiver_remb_candidates_;
+  auto it = std::find(candidates.begin(), candidates.end(), candidate_module);
+
+  if (it == candidates.end()) {
+    return;  // Function called due to removal of non-REMB-candidate module.
+  }
+
+  if (*it == active_remb_module_) {
+    UnsetActiveRembModule();
+  }
+  candidates.erase(it);
+  DetermineActiveRembModule();
+}
+
+void PacketRouter::UnsetActiveRembModule() {
+  RTC_CHECK(active_remb_module_);
+  active_remb_module_->UnsetRemb();
+  active_remb_module_ = nullptr;
+}
+
+void PacketRouter::DetermineActiveRembModule() {
+  // Sender modules take precedence over receiver modules, because SRs (sender
+  // reports) are sent more frequently than RR (receiver reports).
+  // When adding the first sender module, we should change the active REMB
+  // module to be that. Otherwise, we remain with the current active module.
+
+  RtcpFeedbackSenderInterface* new_active_remb_module;
+
+  if (!sender_remb_candidates_.empty()) {
+    new_active_remb_module = sender_remb_candidates_.front();
+  } else if (!receiver_remb_candidates_.empty()) {
+    new_active_remb_module = receiver_remb_candidates_.front();
+  } else {
+    new_active_remb_module = nullptr;
+  }
+
+  if (new_active_remb_module != active_remb_module_ && active_remb_module_) {
+    UnsetActiveRembModule();
+  }
+
+  active_remb_module_ = new_active_remb_module;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/packet_router.h b/third_party/libwebrtc/modules/pacing/packet_router.h
new file mode 100644
index 0000000000..68b82c6bd4
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/packet_router.h
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PACING_PACKET_ROUTER_H_
+#define MODULES_PACING_PACKET_ROUTER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+#include <memory>
+#include <unordered_map>
+#include <utility>
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "modules/pacing/pacing_controller.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class RtpRtcpInterface;
+
+// PacketRouter keeps track of rtp send modules to support the pacer.
+// In addition, it handles feedback messages, which are sent on a send
+// module if possible (sender report), otherwise on receive module
+// (receiver report). For the latter case, we also keep track of the
+// receive modules.
+class PacketRouter : public PacingController::PacketSender {
+ public:
+  PacketRouter();
+  explicit PacketRouter(uint16_t start_transport_seq);
+  ~PacketRouter() override;
+
+  PacketRouter(const PacketRouter&) = delete;
+  PacketRouter& operator=(const PacketRouter&) = delete;
+
+  void AddSendRtpModule(RtpRtcpInterface* rtp_module, bool remb_candidate);
+  void RemoveSendRtpModule(RtpRtcpInterface* rtp_module);
+
+  void AddReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender,
+                           bool remb_candidate);
+  void RemoveReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender);
+
+  void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
+                  const PacedPacketInfo& cluster_info) override;
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFec() override;
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      DataSize size) override;
+  void OnAbortedRetransmissions(
+      uint32_t ssrc,
+      rtc::ArrayView<const uint16_t> sequence_numbers) override;
+  absl::optional<uint32_t> GetRtxSsrcForMedia(uint32_t ssrc) const override;
+
+  uint16_t CurrentTransportSequenceNumber() const;
+
+  // Send REMB feedback.
+  void SendRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs);
+
+  // Sends `packets` in one or more IP packets.
+  void SendCombinedRtcpPacket(
+      std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets);
+
+ private:
+  void AddRembModuleCandidate(RtcpFeedbackSenderInterface* candidate_module,
+                              bool media_sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_mutex_);
+  void MaybeRemoveRembModuleCandidate(
+      RtcpFeedbackSenderInterface* candidate_module,
+      bool media_sender) RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_mutex_);
+  void UnsetActiveRembModule() RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_mutex_);
+  void DetermineActiveRembModule() RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_mutex_);
+  void AddSendRtpModuleToMap(RtpRtcpInterface* rtp_module, uint32_t ssrc)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_mutex_);
+  void RemoveSendRtpModuleFromMap(uint32_t ssrc)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_mutex_);
+
+  mutable Mutex modules_mutex_;
+  // Ssrc to RtpRtcpInterface module;
+  std::unordered_map<uint32_t, RtpRtcpInterface*> send_modules_map_
+      RTC_GUARDED_BY(modules_mutex_);
+  std::list<RtpRtcpInterface*> send_modules_list_
+      RTC_GUARDED_BY(modules_mutex_);
+  // The last module used to send media.
+  RtpRtcpInterface* last_send_module_ RTC_GUARDED_BY(modules_mutex_);
+  // Rtcp modules of the rtp receivers.
+  std::vector<RtcpFeedbackSenderInterface*> rtcp_feedback_senders_
+      RTC_GUARDED_BY(modules_mutex_);
+
+  // Candidates for the REMB module can be RTP sender/receiver modules, with
+  // the sender modules taking precedence.
+  std::vector<RtcpFeedbackSenderInterface*> sender_remb_candidates_
+      RTC_GUARDED_BY(modules_mutex_);
+  std::vector<RtcpFeedbackSenderInterface*> receiver_remb_candidates_
+      RTC_GUARDED_BY(modules_mutex_);
+  RtcpFeedbackSenderInterface* active_remb_module_
+      RTC_GUARDED_BY(modules_mutex_);
+
+  uint64_t transport_seq_ RTC_GUARDED_BY(modules_mutex_);
+
+  // TODO(bugs.webrtc.org/10809): Replace lock with a sequence checker once the
+  // process thread is gone.
+  std::vector<std::unique_ptr<RtpPacketToSend>> pending_fec_packets_
+      RTC_GUARDED_BY(modules_mutex_);
+};
+}  // namespace webrtc
+#endif  // MODULES_PACING_PACKET_ROUTER_H_
diff --git a/third_party/libwebrtc/modules/pacing/packet_router_unittest.cc b/third_party/libwebrtc/modules/pacing/packet_router_unittest.cc
new file mode 100644
index 0000000000..65b2ad24d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/packet_router_unittest.cc
@@ -0,0 +1,736 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/packet_router.h"
+
+#include <cstddef>
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fake_clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// TODO(eladalon): Restructure and/or replace the existing monolithic tests
+// (only some of the test are monolithic) according to the new
+// guidelines - small tests for one thing at a time.
+// (I'm not removing any tests during CL, so as to demonstrate no regressions.)
+
+namespace {
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::AtLeast;
+using ::testing::ElementsAreArray;
+using ::testing::Field;
+using ::testing::Gt;
+using ::testing::Le;
+using ::testing::NiceMock;
+using ::testing::Property;
+using ::testing::Return;
+using ::testing::SaveArg;
+
+constexpr int kProbeMinProbes = 5;
+constexpr int kProbeMinBytes = 1000;
+
+}  // namespace
+
+class PacketRouterTest : public ::testing::Test {
+ public:
+  PacketRouterTest() {
+    extension_manager.Register<TransportSequenceNumber>(/*id=*/1);
+  }
+
+ protected:
+  std::unique_ptr<RtpPacketToSend> BuildRtpPacket(uint32_t ssrc) {
+    std::unique_ptr<RtpPacketToSend> packet =
+        std::make_unique<RtpPacketToSend>(&extension_manager);
+    packet->SetSsrc(ssrc);
+    return packet;
+  }
+
+  PacketRouter packet_router_;
+  RtpHeaderExtensionMap extension_manager;
+};
+
+TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_GeneratePadding) {
+  constexpr DataSize bytes = DataSize::Bytes(300);
+  const PacedPacketInfo paced_info(1, kProbeMinProbes, kProbeMinBytes);
+
+  EXPECT_TRUE(packet_router_.GeneratePadding(bytes).empty());
+}
+
+
+TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_SendRemb) {
+  const std::vector<uint32_t> ssrcs = {1, 2, 3};
+  constexpr uint32_t bitrate_bps = 10000;
+  // Expect not to crash
+  packet_router_.SendRemb(bitrate_bps, ssrcs);
+}
+
+TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_SendTransportFeedback) {
+  std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback;
+  feedback.push_back(std::make_unique<rtcp::TransportFeedback>());
+  // Expect not to crash
+  packet_router_.SendCombinedRtcpPacket(std::move(feedback));
+}
+
+TEST_F(PacketRouterTest, GeneratePaddingPrioritizesRtx) {
+  // Two RTP modules. The first (prioritized due to rtx) isn't sending media so
+  // should not be called.
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 4567;
+
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  ON_CALL(rtp_1, RtxSendStatus()).WillByDefault(Return(kRtxRedundantPayloads));
+  ON_CALL(rtp_1, SSRC()).WillByDefault(Return(kSsrc1));
+  ON_CALL(rtp_1, SupportsPadding).WillByDefault(Return(false));
+
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+  ON_CALL(rtp_2, RtxSendStatus()).WillByDefault(Return(kRtxOff));
+  ON_CALL(rtp_2, SSRC()).WillByDefault(Return(kSsrc2));
+  ON_CALL(rtp_2, SupportsPadding).WillByDefault(Return(true));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddSendRtpModule(&rtp_2, false);
+
+  const size_t kPaddingSize = 123;
+  const size_t kExpectedPaddingPackets = 1;
+  EXPECT_CALL(rtp_1, GeneratePadding(_)).Times(0);
+  EXPECT_CALL(rtp_2, GeneratePadding(kPaddingSize))
+      .WillOnce([&](size_t padding_size) {
+        return std::vector<std::unique_ptr<RtpPacketToSend>>(
+            kExpectedPaddingPackets);
+      });
+  auto generated_padding =
+      packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+  EXPECT_EQ(generated_padding.size(), kExpectedPaddingPackets);
+
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+  packet_router_.RemoveSendRtpModule(&rtp_2);
+}
+
+TEST_F(PacketRouterTest, GeneratePaddingPrioritizesVideo) {
+  // Two RTP modules. Neither support RTX, both support padding,
+  // but the first one is for audio and second for video.
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 4567;
+  const size_t kPaddingSize = 123;
+  const size_t kExpectedPaddingPackets = 1;
+
+  auto generate_padding = [&](size_t padding_size) {
+    return std::vector<std::unique_ptr<RtpPacketToSend>>(
+        kExpectedPaddingPackets);
+  };
+
+  NiceMock<MockRtpRtcpInterface> audio_module;
+  ON_CALL(audio_module, RtxSendStatus()).WillByDefault(Return(kRtxOff));
+  ON_CALL(audio_module, SSRC()).WillByDefault(Return(kSsrc1));
+  ON_CALL(audio_module, SupportsPadding).WillByDefault(Return(true));
+  ON_CALL(audio_module, IsAudioConfigured).WillByDefault(Return(true));
+
+  NiceMock<MockRtpRtcpInterface> video_module;
+  ON_CALL(video_module, RtxSendStatus()).WillByDefault(Return(kRtxOff));
+  ON_CALL(video_module, SSRC()).WillByDefault(Return(kSsrc2));
+  ON_CALL(video_module, SupportsPadding).WillByDefault(Return(true));
+  ON_CALL(video_module, IsAudioConfigured).WillByDefault(Return(false));
+
+  // First add only the audio module. Since this is the only choice we have,
+  // padding should be sent on the audio ssrc.
+  packet_router_.AddSendRtpModule(&audio_module, false);
+  EXPECT_CALL(audio_module, GeneratePadding(kPaddingSize))
+      .WillOnce(generate_padding);
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+
+  // Add the video module, this should now be prioritized since we cannot
+  // guarantee that audio packets will be included in the BWE.
+  packet_router_.AddSendRtpModule(&video_module, false);
+  EXPECT_CALL(audio_module, GeneratePadding).Times(0);
+  EXPECT_CALL(video_module, GeneratePadding(kPaddingSize))
+      .WillOnce(generate_padding);
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+
+  // Remove and the add audio module again. Module order shouldn't matter;
+  // video should still be prioritized.
+  packet_router_.RemoveSendRtpModule(&audio_module);
+  packet_router_.AddSendRtpModule(&audio_module, false);
+  EXPECT_CALL(audio_module, GeneratePadding).Times(0);
+  EXPECT_CALL(video_module, GeneratePadding(kPaddingSize))
+      .WillOnce(generate_padding);
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+
+  // Remove and the video module, we should fall back to padding on the
+  // audio module again.
+  packet_router_.RemoveSendRtpModule(&video_module);
+  EXPECT_CALL(audio_module, GeneratePadding(kPaddingSize))
+      .WillOnce(generate_padding);
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+
+  packet_router_.RemoveSendRtpModule(&audio_module);
+}
+
+TEST_F(PacketRouterTest, PadsOnLastActiveMediaStream) {
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 4567;
+  const uint16_t kSsrc3 = 8901;
+
+  // First two rtp modules send media and have rtx.
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
+  EXPECT_CALL(rtp_1, SupportsPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_1, SupportsRtxPayloadPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_1, TrySendPacket).WillRepeatedly(Return(false));
+  EXPECT_CALL(
+      rtp_1,
+      TrySendPacket(
+          ::testing::Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc1)), _))
+      .WillRepeatedly(Return(true));
+
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+  EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
+  EXPECT_CALL(rtp_2, SupportsPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_2, SupportsRtxPayloadPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_2, TrySendPacket).WillRepeatedly(Return(false));
+  EXPECT_CALL(
+      rtp_2,
+      TrySendPacket(
+          ::testing::Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc2)), _))
+      .WillRepeatedly(Return(true));
+
+  // Third module is sending media, but does not support rtx.
+  NiceMock<MockRtpRtcpInterface> rtp_3;
+  EXPECT_CALL(rtp_3, SSRC()).WillRepeatedly(Return(kSsrc3));
+  EXPECT_CALL(rtp_3, SupportsPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_3, SupportsRtxPayloadPadding).WillRepeatedly(Return(false));
+  EXPECT_CALL(rtp_3, TrySendPacket).WillRepeatedly(Return(false));
+  EXPECT_CALL(
+      rtp_3,
+      TrySendPacket(
+          ::testing::Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc3)), _))
+      .WillRepeatedly(Return(true));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddSendRtpModule(&rtp_2, false);
+  packet_router_.AddSendRtpModule(&rtp_3, false);
+
+  const size_t kPaddingBytes = 100;
+
+  // Initially, padding will be sent on last added rtp module that sends media
+  // and supports rtx.
+  EXPECT_CALL(rtp_2, GeneratePadding(kPaddingBytes))
+      .Times(1)
+      .WillOnce([&](size_t target_size_bytes) {
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+        packets.push_back(BuildRtpPacket(kSsrc2));
+        return packets;
+      });
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingBytes));
+
+  // Send media on first module. Padding should be sent on that module.
+  packet_router_.SendPacket(BuildRtpPacket(kSsrc1), PacedPacketInfo());
+
+  EXPECT_CALL(rtp_1, GeneratePadding(kPaddingBytes))
+      .Times(1)
+      .WillOnce([&](size_t target_size_bytes) {
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+        packets.push_back(BuildRtpPacket(kSsrc1));
+        return packets;
+      });
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingBytes));
+
+  // Send media on second module. Padding should be sent there.
+  packet_router_.SendPacket(BuildRtpPacket(kSsrc2), PacedPacketInfo());
+
+  // If the last active module is removed, and no module sends media before
+  // the next padding request, and arbitrary module will be selected.
+  packet_router_.RemoveSendRtpModule(&rtp_2);
+
+  // Send on and then remove all remaining modules.
+  RtpRtcpInterface* last_send_module;
+  EXPECT_CALL(rtp_1, GeneratePadding(kPaddingBytes))
+      .Times(1)
+      .WillOnce([&](size_t target_size_bytes) {
+        last_send_module = &rtp_1;
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+        packets.push_back(BuildRtpPacket(kSsrc1));
+        return packets;
+      });
+  EXPECT_CALL(rtp_3, GeneratePadding(kPaddingBytes))
+      .Times(1)
+      .WillOnce([&](size_t target_size_bytes) {
+        last_send_module = &rtp_3;
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+        packets.push_back(BuildRtpPacket(kSsrc3));
+        return packets;
+      });
+
+  for (int i = 0; i < 2; ++i) {
+    last_send_module = nullptr;
+    packet_router_.GeneratePadding(DataSize::Bytes(kPaddingBytes));
+    EXPECT_NE(last_send_module, nullptr);
+    packet_router_.RemoveSendRtpModule(last_send_module);
+  }
+}
+
+TEST_F(PacketRouterTest, AllocatesTransportSequenceNumbers) {
+  const uint16_t kStartSeq = 0xFFF0;
+  const size_t kNumPackets = 32;
+  const uint16_t kSsrc1 = 1234;
+
+  PacketRouter packet_router(kStartSeq - 1);
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
+  EXPECT_CALL(rtp_1, TrySendPacket).WillRepeatedly(Return(true));
+  packet_router.AddSendRtpModule(&rtp_1, false);
+
+  for (size_t i = 0; i < kNumPackets; ++i) {
+    auto packet = BuildRtpPacket(kSsrc1);
+    EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+    packet_router.SendPacket(std::move(packet), PacedPacketInfo());
+    uint32_t expected_unwrapped_seq = static_cast<uint32_t>(kStartSeq) + i;
+    EXPECT_EQ(static_cast<uint16_t>(expected_unwrapped_seq & 0xFFFF),
+              packet_router.CurrentTransportSequenceNumber());
+  }
+
+  packet_router.RemoveSendRtpModule(&rtp_1);
+}
+
+TEST_F(PacketRouterTest, SendTransportFeedback) {
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+
+  ON_CALL(rtp_1, RTCP()).WillByDefault(Return(RtcpMode::kCompound));
+  ON_CALL(rtp_2, RTCP()).WillByDefault(Return(RtcpMode::kCompound));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddReceiveRtpModule(&rtp_2, false);
+
+  std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback;
+  feedback.push_back(std::make_unique<rtcp::TransportFeedback>());
+  EXPECT_CALL(rtp_1, SendCombinedRtcpPacket);
+  packet_router_.SendCombinedRtcpPacket(std::move(feedback));
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+  EXPECT_CALL(rtp_2, SendCombinedRtcpPacket);
+  std::vector<std::unique_ptr<rtcp::RtcpPacket>> new_feedback;
+  new_feedback.push_back(std::make_unique<rtcp::TransportFeedback>());
+  packet_router_.SendCombinedRtcpPacket(std::move(new_feedback));
+  packet_router_.RemoveReceiveRtpModule(&rtp_2);
+}
+
+TEST_F(PacketRouterTest, SendPacketWithoutTransportSequenceNumbers) {
+  const uint16_t kSsrc1 = 1234;
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(true));
+  ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+
+  // Send a packet without TransportSequenceNumber extension registered,
+  // packets sent should not have the extension set.
+  RtpHeaderExtensionMap extension_manager;
+  auto packet = std::make_unique<RtpPacketToSend>(&extension_manager);
+  packet->SetSsrc(kSsrc1);
+  EXPECT_CALL(
+      rtp_1,
+      TrySendPacket(
+          Property(&RtpPacketToSend::HasExtension<TransportSequenceNumber>,
+                   false),
+          _))
+      .WillOnce(Return(true));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+}
+
+TEST_F(PacketRouterTest, SendPacketAssignsTransportSequenceNumbers) {
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 2345;
+
+  ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
+  ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddSendRtpModule(&rtp_2, false);
+
+  // Transport sequence numbers start at 1, for historical reasons.
+  uint16_t transport_sequence_number = 1;
+
+  auto packet = BuildRtpPacket(kSsrc1);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+  EXPECT_CALL(
+      rtp_1,
+      TrySendPacket(
+          Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                   transport_sequence_number),
+          _))
+      .WillOnce(Return(true));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  ++transport_sequence_number;
+  packet = BuildRtpPacket(kSsrc2);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+
+  EXPECT_CALL(
+      rtp_2,
+      TrySendPacket(
+          Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                   transport_sequence_number),
+          _))
+      .WillOnce(Return(true));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+  packet_router_.RemoveSendRtpModule(&rtp_2);
+}
+
+TEST_F(PacketRouterTest, DoesNotIncrementTransportSequenceNumberOnSendFailure) {
+  NiceMock<MockRtpRtcpInterface> rtp;
+  constexpr uint32_t kSsrc = 1234;
+  ON_CALL(rtp, SSRC).WillByDefault(Return(kSsrc));
+  packet_router_.AddSendRtpModule(&rtp, false);
+
+  // Transport sequence numbers start at 1, for historical reasons.
+  const uint16_t kStartTransportSequenceNumber = 1;
+
+  // Build and send a packet - it should be assigned the start sequence number.
+  // Return failure status code to make sure sequence number is not incremented.
+  auto packet = BuildRtpPacket(kSsrc);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+  EXPECT_CALL(
+      rtp, TrySendPacket(
+               Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                        kStartTransportSequenceNumber),
+               _))
+      .WillOnce(Return(false));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  // Send another packet, verify transport sequence number is still at the
+  // start state.
+  packet = BuildRtpPacket(kSsrc);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+
+  EXPECT_CALL(
+      rtp, TrySendPacket(
+               Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                        kStartTransportSequenceNumber),
+               _))
+      .WillOnce(Return(true));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  packet_router_.RemoveSendRtpModule(&rtp);
+}
+
+TEST_F(PacketRouterTest, ForwardsAbortedRetransmissions) {
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+
+  const uint32_t kSsrc1 = 1234;
+  const uint32_t kSsrc2 = 2345;
+  const uint32_t kInvalidSsrc = 3456;
+
+  ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
+  ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddSendRtpModule(&rtp_2, false);
+
+  // Sets of retransmission sequence numbers we wish to abort, per ssrc.
+  const uint16_t kAbortedRetransmissionsOnSsrc1[] = {17, 42};
+  const uint16_t kAbortedRetransmissionsOnSsrc2[] = {1337, 4711};
+  const uint16_t kAbortedRetransmissionsOnSsrc3[] = {123};
+
+  EXPECT_CALL(rtp_1, OnAbortedRetransmissions(
+                         ElementsAreArray(kAbortedRetransmissionsOnSsrc1)));
+  EXPECT_CALL(rtp_2, OnAbortedRetransmissions(
+                         ElementsAreArray(kAbortedRetransmissionsOnSsrc2)));
+
+  packet_router_.OnAbortedRetransmissions(kSsrc1,
+                                          kAbortedRetransmissionsOnSsrc1);
+  packet_router_.OnAbortedRetransmissions(kSsrc2,
+                                          kAbortedRetransmissionsOnSsrc2);
+
+  // Should be noop and not cause any issues.
+  packet_router_.OnAbortedRetransmissions(kInvalidSsrc,
+                                          kAbortedRetransmissionsOnSsrc3);
+
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+  packet_router_.RemoveSendRtpModule(&rtp_2);
+}
+
+TEST_F(PacketRouterTest, ReportsRtxSsrc) {
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+
+  const uint32_t kSsrc1 = 1234;
+  const uint32_t kRtxSsrc1 = 1235;
+  const uint32_t kSsrc2 = 2345;
+  const uint32_t kInvalidSsrc = 3456;
+
+  ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
+  ON_CALL(rtp_1, RtxSsrc).WillByDefault(Return(kRtxSsrc1));
+  ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddSendRtpModule(&rtp_2, false);
+
+  EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kSsrc1), kRtxSsrc1);
+  EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kRtxSsrc1), absl::nullopt);
+  EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kSsrc2), absl::nullopt);
+  EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kInvalidSsrc), absl::nullopt);
+
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+  packet_router_.RemoveSendRtpModule(&rtp_2);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+using PacketRouterDeathTest = PacketRouterTest;
+TEST_F(PacketRouterDeathTest, DoubleRegistrationOfSendModuleDisallowed) {
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = false;  // Value irrelevant.
+  packet_router_.AddSendRtpModule(&module, remb_candidate);
+  EXPECT_DEATH(packet_router_.AddSendRtpModule(&module, remb_candidate), "");
+
+  // Test tear-down
+  packet_router_.RemoveSendRtpModule(&module);
+}
+
+TEST_F(PacketRouterDeathTest, DoubleRegistrationOfReceiveModuleDisallowed) {
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = false;  // Value irrelevant.
+  packet_router_.AddReceiveRtpModule(&module, remb_candidate);
+  EXPECT_DEATH(packet_router_.AddReceiveRtpModule(&module, remb_candidate), "");
+
+  // Test tear-down
+  packet_router_.RemoveReceiveRtpModule(&module);
+}
+
+TEST_F(PacketRouterDeathTest, RemovalOfNeverAddedSendModuleDisallowed) {
+  NiceMock<MockRtpRtcpInterface> module;
+
+  EXPECT_DEATH(packet_router_.RemoveSendRtpModule(&module), "");
+}
+
+TEST_F(PacketRouterDeathTest, RemovalOfNeverAddedReceiveModuleDisallowed) {
+  NiceMock<MockRtpRtcpInterface> module;
+
+  EXPECT_DEATH(packet_router_.RemoveReceiveRtpModule(&module), "");
+}
+#endif  // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+TEST(PacketRouterRembTest, ChangeSendRtpModuleChangeRembSender) {
+  rtc::ScopedFakeClock clock;
+  NiceMock<MockRtpRtcpInterface> rtp_send;
+  NiceMock<MockRtpRtcpInterface> rtp_recv;
+  PacketRouter packet_router;
+  packet_router.AddSendRtpModule(&rtp_send, true);
+  packet_router.AddReceiveRtpModule(&rtp_recv, true);
+
+  uint32_t bitrate_estimate = 456;
+  std::vector<uint32_t> ssrcs = {1234, 5678};
+
+  EXPECT_CALL(rtp_send, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Remove the sending module -> should get remb on the second module.
+  packet_router.RemoveSendRtpModule(&rtp_send);
+
+  EXPECT_CALL(rtp_recv, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  packet_router.RemoveReceiveRtpModule(&rtp_recv);
+}
+
+// Only register receiving modules and make sure we fallback to trigger a REMB
+// packet on this one.
+TEST(PacketRouterRembTest, NoSendingRtpModule) {
+  rtc::ScopedFakeClock clock;
+  NiceMock<MockRtpRtcpInterface> rtp;
+  PacketRouter packet_router;
+
+  packet_router.AddReceiveRtpModule(&rtp, true);
+
+  uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+
+  EXPECT_CALL(rtp, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Lower the estimate to trigger a new packet REMB packet.
+  EXPECT_CALL(rtp, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  EXPECT_CALL(rtp, UnsetRemb());
+  packet_router.RemoveReceiveRtpModule(&rtp);
+}
+
+TEST(PacketRouterRembTest, NonCandidateSendRtpModuleNotUsedForRemb) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = false;
+
+  packet_router.AddSendRtpModule(&module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(module, SetRemb(_, _)).Times(0);
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveSendRtpModule(&module);
+}
+
+TEST(PacketRouterRembTest, CandidateSendRtpModuleUsedForRemb) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = true;
+
+  packet_router.AddSendRtpModule(&module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(module, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveSendRtpModule(&module);
+}
+
+TEST(PacketRouterRembTest, NonCandidateReceiveRtpModuleNotUsedForRemb) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = false;
+
+  packet_router.AddReceiveRtpModule(&module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(module, SetRemb(_, _)).Times(0);
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&module);
+}
+
+TEST(PacketRouterRembTest, CandidateReceiveRtpModuleUsedForRemb) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = true;
+
+  packet_router.AddReceiveRtpModule(&module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(module, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&module);
+}
+
+TEST(PacketRouterRembTest,
+     SendCandidatePreferredOverReceiveCandidate_SendModuleAddedFirst) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> send_module;
+  NiceMock<MockRtpRtcpInterface> receive_module;
+
+  constexpr bool remb_candidate = true;
+
+  // Send module added - activated.
+  packet_router.AddSendRtpModule(&send_module, remb_candidate);
+
+  // Receive module added - the send module remains the active one.
+  packet_router.AddReceiveRtpModule(&receive_module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(send_module, SetRemb(bitrate_estimate, ssrcs));
+  EXPECT_CALL(receive_module, SetRemb(_, _)).Times(0);
+
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&receive_module);
+  packet_router.RemoveSendRtpModule(&send_module);
+}
+
+TEST(PacketRouterRembTest,
+     SendCandidatePreferredOverReceiveCandidate_ReceiveModuleAddedFirst) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> send_module;
+  NiceMock<MockRtpRtcpInterface> receive_module;
+
+  constexpr bool remb_candidate = true;
+
+  // Receive module added - activated.
+  packet_router.AddReceiveRtpModule(&receive_module, remb_candidate);
+
+  // Send module added - replaces receive module as active.
+  packet_router.AddSendRtpModule(&send_module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(send_module, SetRemb(bitrate_estimate, ssrcs));
+  EXPECT_CALL(receive_module, SetRemb(_, _)).Times(0);
+
+  clock.AdvanceTime(TimeDelta::Millis(1000));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&receive_module);
+  packet_router.RemoveSendRtpModule(&send_module);
+}
+
+TEST(PacketRouterRembTest, ReceiveModuleTakesOverWhenLastSendModuleRemoved) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> send_module;
+  NiceMock<MockRtpRtcpInterface> receive_module;
+
+  constexpr bool remb_candidate = true;
+
+  // Send module active, receive module inactive.
+  packet_router.AddSendRtpModule(&send_module, remb_candidate);
+  packet_router.AddReceiveRtpModule(&receive_module, remb_candidate);
+
+  // Send module removed - receive module becomes active.
+  packet_router.RemoveSendRtpModule(&send_module);
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(send_module, SetRemb(_, _)).Times(0);
+  EXPECT_CALL(receive_module, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&receive_module);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc
new file mode 100644
index 0000000000..0c285c463a
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc
@@ -0,0 +1,343 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/prioritized_packet_queue.h"
+
+#include <utility>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kAudioPrioLevel = 0;
+
+int GetPriorityForType(RtpPacketMediaType type) {
+  // Lower number takes priority over higher.
+  switch (type) {
+    case RtpPacketMediaType::kAudio:
+      // Audio is always prioritized over other packet types.
+      return kAudioPrioLevel;
+    case RtpPacketMediaType::kRetransmission:
+      // Send retransmissions before new media.
+      return kAudioPrioLevel + 1;
+    case RtpPacketMediaType::kVideo:
+    case RtpPacketMediaType::kForwardErrorCorrection:
+      // Video has "normal" priority, in the old speak.
+      // Send redundancy concurrently to video. If it is delayed it might have a
+      // lower chance of being useful.
+      return kAudioPrioLevel + 2;
+    case RtpPacketMediaType::kPadding:
+      // Packets that are in themselves likely useless, only sent to keep the
+      // BWE high.
+      return kAudioPrioLevel + 3;
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+}  // namespace
+
+DataSize PrioritizedPacketQueue::QueuedPacket::PacketSize() const {
+  return DataSize::Bytes(packet->payload_size() + packet->padding_size());
+}
+
+PrioritizedPacketQueue::StreamQueue::StreamQueue(Timestamp creation_time)
+    : last_enqueue_time_(creation_time) {}
+
+bool PrioritizedPacketQueue::StreamQueue::EnqueuePacket(QueuedPacket packet,
+                                                        int priority_level) {
+  bool first_packet_at_level = packets_[priority_level].empty();
+  packets_[priority_level].push_back(std::move(packet));
+  return first_packet_at_level;
+}
+
+PrioritizedPacketQueue::QueuedPacket
+PrioritizedPacketQueue::StreamQueue::DequeuePacket(int priority_level) {
+  RTC_DCHECK(!packets_[priority_level].empty());
+  QueuedPacket packet = std::move(packets_[priority_level].front());
+  packets_[priority_level].pop_front();
+  return packet;
+}
+
+bool PrioritizedPacketQueue::StreamQueue::HasPacketsAtPrio(
+    int priority_level) const {
+  return !packets_[priority_level].empty();
+}
+
+bool PrioritizedPacketQueue::StreamQueue::IsEmpty() const {
+  for (const std::deque<QueuedPacket>& queue : packets_) {
+    if (!queue.empty()) {
+      return false;
+    }
+  }
+  return true;
+}
+
+Timestamp PrioritizedPacketQueue::StreamQueue::LeadingPacketEnqueueTime(
+    int priority_level) const {
+  RTC_DCHECK(!packets_[priority_level].empty());
+  return packets_[priority_level].begin()->enqueue_time;
+}
+
+Timestamp PrioritizedPacketQueue::StreamQueue::LastEnqueueTime() const {
+  return last_enqueue_time_;
+}
+
+std::array<std::deque<PrioritizedPacketQueue::QueuedPacket>,
+           PrioritizedPacketQueue::kNumPriorityLevels>
+PrioritizedPacketQueue::StreamQueue::DequeueAll() {
+  std::array<std::deque<QueuedPacket>, kNumPriorityLevels> packets_by_prio;
+  for (int i = 0; i < kNumPriorityLevels; ++i) {
+    packets_by_prio[i].swap(packets_[i]);
+  }
+  return packets_by_prio;
+}
+
+PrioritizedPacketQueue::PrioritizedPacketQueue(Timestamp creation_time)
+    : queue_time_sum_(TimeDelta::Zero()),
+      pause_time_sum_(TimeDelta::Zero()),
+      size_packets_(0),
+      size_packets_per_media_type_({}),
+      size_payload_(DataSize::Zero()),
+      last_update_time_(creation_time),
+      paused_(false),
+      last_culling_time_(creation_time),
+      top_active_prio_level_(-1) {}
+
+void PrioritizedPacketQueue::Push(Timestamp enqueue_time,
+                                  std::unique_ptr<RtpPacketToSend> packet) {
+  StreamQueue* stream_queue;
+  auto [it, inserted] = streams_.emplace(packet->Ssrc(), nullptr);
+  if (inserted) {
+    it->second = std::make_unique<StreamQueue>(enqueue_time);
+  }
+  stream_queue = it->second.get();
+
+  auto enqueue_time_iterator =
+      enqueue_times_.insert(enqueue_times_.end(), enqueue_time);
+  RTC_DCHECK(packet->packet_type().has_value());
+  RtpPacketMediaType packet_type = packet->packet_type().value();
+  int prio_level = GetPriorityForType(packet_type);
+  RTC_DCHECK_GE(prio_level, 0);
+  RTC_DCHECK_LT(prio_level, kNumPriorityLevels);
+  QueuedPacket queued_packed = {.packet = std::move(packet),
+                                .enqueue_time = enqueue_time,
+                                .enqueue_time_iterator = enqueue_time_iterator};
+  // In order to figure out how much time a packet has spent in the queue
+  // while not in a paused state, we subtract the total amount of time the
+  // queue has been paused so far, and when the packet is popped we subtract
+  // the total amount of time the queue has been paused at that moment. This
+  // way we subtract the total amount of time the packet has spent in the
+  // queue while in a paused state.
+  UpdateAverageQueueTime(enqueue_time);
+  queued_packed.enqueue_time -= pause_time_sum_;
+  ++size_packets_;
+  ++size_packets_per_media_type_[static_cast<size_t>(packet_type)];
+  size_payload_ += queued_packed.PacketSize();
+
+  if (stream_queue->EnqueuePacket(std::move(queued_packed), prio_level)) {
+    // Number packets at `prio_level` for this steam is now non-zero.
+    streams_by_prio_[prio_level].push_back(stream_queue);
+  }
+  if (top_active_prio_level_ < 0 || prio_level < top_active_prio_level_) {
+    top_active_prio_level_ = prio_level;
+  }
+
+  static constexpr TimeDelta kTimeout = TimeDelta::Millis(500);
+  if (enqueue_time - last_culling_time_ > kTimeout) {
+    for (auto it = streams_.begin(); it != streams_.end();) {
+      if (it->second->IsEmpty() &&
+          it->second->LastEnqueueTime() + kTimeout < enqueue_time) {
+        streams_.erase(it++);
+      } else {
+        ++it;
+      }
+    }
+    last_culling_time_ = enqueue_time;
+  }
+}
+
+std::unique_ptr<RtpPacketToSend> PrioritizedPacketQueue::Pop() {
+  if (size_packets_ == 0) {
+    return nullptr;
+  }
+
+  RTC_DCHECK_GE(top_active_prio_level_, 0);
+  StreamQueue& stream_queue = *streams_by_prio_[top_active_prio_level_].front();
+  QueuedPacket packet = stream_queue.DequeuePacket(top_active_prio_level_);
+  DequeuePacketInternal(packet);
+
+  // Remove StreamQueue from head of fifo-queue for this prio level, and
+  // and add it to the end if it still has packets.
+  streams_by_prio_[top_active_prio_level_].pop_front();
+  if (stream_queue.HasPacketsAtPrio(top_active_prio_level_)) {
+    streams_by_prio_[top_active_prio_level_].push_back(&stream_queue);
+  } else {
+    MaybeUpdateTopPrioLevel();
+  }
+
+  return std::move(packet.packet);
+}
+
+int PrioritizedPacketQueue::SizeInPackets() const {
+  return size_packets_;
+}
+
+DataSize PrioritizedPacketQueue::SizeInPayloadBytes() const {
+  return size_payload_;
+}
+
+bool PrioritizedPacketQueue::Empty() const {
+  return size_packets_ == 0;
+}
+
+const std::array<int, kNumMediaTypes>&
+PrioritizedPacketQueue::SizeInPacketsPerRtpPacketMediaType() const {
+  return size_packets_per_media_type_;
+}
+
+Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTime(
+    RtpPacketMediaType type) const {
+  const int priority_level = GetPriorityForType(type);
+  if (streams_by_prio_[priority_level].empty()) {
+    return Timestamp::MinusInfinity();
+  }
+  return streams_by_prio_[priority_level].front()->LeadingPacketEnqueueTime(
+      priority_level);
+}
+
+Timestamp PrioritizedPacketQueue::OldestEnqueueTime() const {
+  return enqueue_times_.empty() ? Timestamp::MinusInfinity()
+                                : enqueue_times_.front();
+}
+
+TimeDelta PrioritizedPacketQueue::AverageQueueTime() const {
+  if (size_packets_ == 0) {
+    return TimeDelta::Zero();
+  }
+  return queue_time_sum_ / size_packets_;
+}
+
+void PrioritizedPacketQueue::UpdateAverageQueueTime(Timestamp now) {
+  RTC_CHECK_GE(now, last_update_time_);
+  if (now == last_update_time_) {
+    return;
+  }
+
+  TimeDelta delta = now - last_update_time_;
+
+  if (paused_) {
+    pause_time_sum_ += delta;
+  } else {
+    queue_time_sum_ += delta * size_packets_;
+  }
+
+  last_update_time_ = now;
+}
+
+void PrioritizedPacketQueue::SetPauseState(bool paused, Timestamp now) {
+  UpdateAverageQueueTime(now);
+  paused_ = paused;
+}
+
+void PrioritizedPacketQueue::RemovePacketsForSsrc(uint32_t ssrc) {
+  auto kv = streams_.find(ssrc);
+  if (kv != streams_.end()) {
+    // Dequeue all packets from the queue for this SSRC.
+    StreamQueue& queue = *kv->second;
+    std::array<std::deque<QueuedPacket>, kNumPriorityLevels> packets_by_prio =
+        queue.DequeueAll();
+    for (int i = 0; i < kNumPriorityLevels; ++i) {
+      std::deque<QueuedPacket>& packet_queue = packets_by_prio[i];
+      if (packet_queue.empty()) {
+        continue;
+      }
+
+      // First erase all packets at this prio level.
+      while (!packet_queue.empty()) {
+        QueuedPacket packet = std::move(packet_queue.front());
+        packet_queue.pop_front();
+        DequeuePacketInternal(packet);
+      }
+
+      // Next, deregister this `StreamQueue` from the round-robin tables.
+      RTC_DCHECK(!streams_by_prio_[i].empty());
+      if (streams_by_prio_[i].size() == 1) {
+        // This is the last and only queue that had packets for this prio level.
+        // Update the global top prio level if neccessary.
+        RTC_DCHECK(streams_by_prio_[i].front() == &queue);
+        streams_by_prio_[i].pop_front();
+        if (i == top_active_prio_level_) {
+          MaybeUpdateTopPrioLevel();
+        }
+      } else {
+        // More than stream had packets at this prio level, filter this one out.
+        std::deque<StreamQueue*> filtered_queue;
+        for (StreamQueue* queue_ptr : streams_by_prio_[i]) {
+          if (queue_ptr != &queue) {
+            filtered_queue.push_back(queue_ptr);
+          }
+        }
+        streams_by_prio_[i].swap(filtered_queue);
+      }
+    }
+  }
+}
+
+void PrioritizedPacketQueue::DequeuePacketInternal(QueuedPacket& packet) {
+  --size_packets_;
+  RTC_DCHECK(packet.packet->packet_type().has_value());
+  RtpPacketMediaType packet_type = packet.packet->packet_type().value();
+  --size_packets_per_media_type_[static_cast<size_t>(packet_type)];
+  RTC_DCHECK_GE(size_packets_per_media_type_[static_cast<size_t>(packet_type)],
+                0);
+  size_payload_ -= packet.PacketSize();
+
+  // Calculate the total amount of time spent by this packet in the queue
+  // while in a non-paused state. Note that the `pause_time_sum_ms_` was
+  // subtracted from `packet.enqueue_time_ms` when the packet was pushed, and
+  // by subtracting it now we effectively remove the time spent in in the
+  // queue while in a paused state.
+  TimeDelta time_in_non_paused_state =
+      last_update_time_ - packet.enqueue_time - pause_time_sum_;
+  queue_time_sum_ -= time_in_non_paused_state;
+
+  // Set the time spent in the send queue, which is the per-packet equivalent of
+  // totalPacketSendDelay. The notion of being paused is an implementation
+  // detail that we do not want to expose, so it makes sense to report the
+  // metric excluding the pause time. This also avoids spikes in the metric.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
+  packet.packet->set_time_in_send_queue(time_in_non_paused_state);
+
+  RTC_DCHECK(size_packets_ > 0 || queue_time_sum_ == TimeDelta::Zero());
+
+  RTC_CHECK(packet.enqueue_time_iterator != enqueue_times_.end());
+  enqueue_times_.erase(packet.enqueue_time_iterator);
+}
+
+void PrioritizedPacketQueue::MaybeUpdateTopPrioLevel() {
+  if (streams_by_prio_[top_active_prio_level_].empty()) {
+    // No stream queues have packets at this prio level, find top priority
+    // that is not empty.
+    if (size_packets_ == 0) {
+      top_active_prio_level_ = -1;
+    } else {
+      for (int i = 0; i < kNumPriorityLevels; ++i) {
+        if (!streams_by_prio_[i].empty()) {
+          top_active_prio_level_ = i;
+          break;
+        }
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h
new file mode 100644
index 0000000000..364b53af11
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h
@@ -0,0 +1,171 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PACING_PRIORITIZED_PACKET_QUEUE_H_
+#define MODULES_PACING_PRIORITIZED_PACKET_QUEUE_H_
+
+#include <stddef.h>
+
+#include <array>
+#include <deque>
+#include <list>
+#include <memory>
+#include <unordered_map>
+#include <vector>
+
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+class PrioritizedPacketQueue {
+ public:
+  explicit PrioritizedPacketQueue(Timestamp creation_time);
+  PrioritizedPacketQueue(const PrioritizedPacketQueue&) = delete;
+  PrioritizedPacketQueue& operator=(const PrioritizedPacketQueue&) = delete;
+
+  // Add a packet to the queue. The enqueue time is used for queue time stats
+  // and to report the leading packet enqueue time per packet type.
+  void Push(Timestamp enqueue_time, std::unique_ptr<RtpPacketToSend> packet);
+
+  // Remove the next packet from the queue. Packets a prioritized first
+  // according to packet type, in the following order:
+  // - audio, retransmissions, video / fec, padding
+  // For each packet type, we use one FIFO-queue per SSRC and emit from
+  // those queues in a round-robin fashion.
+  std::unique_ptr<RtpPacketToSend> Pop();
+
+  // Number of packets in the queue.
+  int SizeInPackets() const;
+
+  // Sum of all payload bytes in the queue, where the payload is calculated
+  // as `packet->payload_size() + packet->padding_size()`.
+  DataSize SizeInPayloadBytes() const;
+
+  // Convenience method for `SizeInPackets() == 0`.
+  bool Empty() const;
+
+  // Total packets in the queue per media type (RtpPacketMediaType values are
+  // used as lookup index).
+  const std::array<int, kNumMediaTypes>& SizeInPacketsPerRtpPacketMediaType()
+      const;
+
+  // The enqueue time of the next packet this queue will return via the Pop()
+  // method, for the given packet type. If queue has no packets, of that type,
+  // returns Timestamp::MinusInfinity().
+  Timestamp LeadingPacketEnqueueTime(RtpPacketMediaType type) const;
+
+  // Enqueue time of the oldest packet in the queue,
+  // Timestamp::MinusInfinity() if queue is empty.
+  Timestamp OldestEnqueueTime() const;
+
+  // Average queue time for the packets currently in the queue.
+  // The queuing time is calculated from Push() to the last UpdateQueueTime()
+  // call - with any time spent in a paused state subtracted.
+  // Returns TimeDelta::Zero() for an empty queue.
+  TimeDelta AverageQueueTime() const;
+
+  // Called during packet processing or when pause stats changes. Since the
+  // AverageQueueTime() method does not look at the wall time, this method
+  // needs to be called before querying queue time.
+  void UpdateAverageQueueTime(Timestamp now);
+
+  // Set the pause state, while `paused` is true queuing time is not counted.
+  void SetPauseState(bool paused, Timestamp now);
+
+  // Remove any packets matching the given SSRC.
+  void RemovePacketsForSsrc(uint32_t ssrc);
+
+ private:
+  static constexpr int kNumPriorityLevels = 4;
+
+  class QueuedPacket {
+   public:
+    DataSize PacketSize() const;
+
+    std::unique_ptr<RtpPacketToSend> packet;
+    Timestamp enqueue_time;
+    std::list<Timestamp>::iterator enqueue_time_iterator;
+  };
+
+  // Class containing packets for an RTP stream.
+  // For each priority level, packets are simply stored in a fifo queue.
+  class StreamQueue {
+   public:
+    explicit StreamQueue(Timestamp creation_time);
+    StreamQueue(StreamQueue&&) = default;
+    StreamQueue& operator=(StreamQueue&&) = default;
+
+    StreamQueue(const StreamQueue&) = delete;
+    StreamQueue& operator=(const StreamQueue&) = delete;
+
+    // Enqueue packet at the given priority level. Returns true if the packet
+    // count for that priority level went from zero to non-zero.
+    bool EnqueuePacket(QueuedPacket packet, int priority_level);
+
+    QueuedPacket DequeuePacket(int priority_level);
+
+    bool HasPacketsAtPrio(int priority_level) const;
+    bool IsEmpty() const;
+    Timestamp LeadingPacketEnqueueTime(int priority_level) const;
+    Timestamp LastEnqueueTime() const;
+
+    std::array<std::deque<QueuedPacket>, kNumPriorityLevels> DequeueAll();
+
+   private:
+    std::deque<QueuedPacket> packets_[kNumPriorityLevels];
+    Timestamp last_enqueue_time_;
+  };
+
+  // Remove the packet from the internal state, e.g. queue time / size etc.
+  void DequeuePacketInternal(QueuedPacket& packet);
+
+  // Check if the queue pointed to by `top_active_prio_level_` is empty and
+  // if so move it to the lowest non-empty index.
+  void MaybeUpdateTopPrioLevel();
+
+  // Cumulative sum, over all packets, of time spent in the queue.
+  TimeDelta queue_time_sum_;
+  // Cumulative sum of time the queue has spent in a paused state.
+  TimeDelta pause_time_sum_;
+  // Total number of packets stored in this queue.
+  int size_packets_;
+  // Total number of packets stored in this queue per RtpPacketMediaType.
+  std::array<int, kNumMediaTypes> size_packets_per_media_type_;
+  // Sum of payload sizes for all packts stored in this queue.
+  DataSize size_payload_;
+  // The last time queue/pause time sums were updated.
+  Timestamp last_update_time_;
+  bool paused_;
+
+  // Last time `streams_` was culled for inactive streams.
+  Timestamp last_culling_time_;
+
+  // Map from SSRC to packet queues for the associated RTP stream.
+  std::unordered_map<uint32_t, std::unique_ptr<StreamQueue>> streams_;
+
+  // For each priority level, a queue of StreamQueues which have at least one
+  // packet pending for that prio level.
+  std::deque<StreamQueue*> streams_by_prio_[kNumPriorityLevels];
+
+  // The first index into `stream_by_prio_` that is non-empty.
+  int top_active_prio_level_;
+
+  // Ordered list of enqueue times. Additions are always increasing and added to
+  // the end. QueuedPacket instances have a iterators into this list for fast
+  // removal.
+  std::list<Timestamp> enqueue_times_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_PACING_PRIORITIZED_PACKET_QUEUE_H_
diff --git a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc
new file mode 100644
index 0000000000..964051c0c7
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc
@@ -0,0 +1,363 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/prioritized_packet_queue.h"
+
+#include <utility>
+
+#include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr uint32_t kDefaultSsrc = 123;
+constexpr int kDefaultPayloadSize = 789;
+
+std::unique_ptr<RtpPacketToSend> CreatePacket(RtpPacketMediaType type,
+                                              uint16_t sequence_number,
+                                              uint32_t ssrc = kDefaultSsrc) {
+  auto packet = std::make_unique<RtpPacketToSend>(/*extensions=*/nullptr);
+  packet->set_packet_type(type);
+  packet->SetSsrc(ssrc);
+  packet->SetSequenceNumber(sequence_number);
+  packet->SetPayloadSize(kDefaultPayloadSize);
+  return packet;
+}
+
+}  // namespace
+
+TEST(PrioritizedPacketQueue, ReturnsPacketsInPrioritizedOrder) {
+  Timestamp now = Timestamp::Zero();
+  PrioritizedPacketQueue queue(now);
+
+  // Add packets in low to high packet order.
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/1));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kForwardErrorCorrection,
+                               /*seq=*/3));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/4));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/5));
+
+  // Packets should be returned in high to low order.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 5);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 4);
+  // Video and FEC prioritized equally - but video was enqueued first.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 2);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 3);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 1);
+}
+
+TEST(PrioritizedPacketQueue, ReturnsEqualPrioPacketsInRoundRobinOrder) {
+  Timestamp now = Timestamp::Zero();
+  PrioritizedPacketQueue queue(now);
+
+  // Insert video packets (prioritized equally), simulating a simulcast-type use
+  // case.
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/1, /*ssrc=*/100));
+
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2, /*ssrc=*/101));
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/3, /*ssrc=*/101));
+
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/4, /*ssrc=*/102));
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/5, /*ssrc=*/102));
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/6, /*ssrc=*/102));
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/7, /*ssrc=*/102));
+
+  // First packet from each SSRC.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 1);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 2);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 4);
+
+  // Second packets from streams that have packets left.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 3);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 5);
+
+  // Only packets from last stream remaining.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 6);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 7);
+}
+
+TEST(PrioritizedPacketQueue, ReportsSizeInPackets) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.SizeInPackets(), 0);
+
+  queue.Push(/*enqueue_time=*/Timestamp::Zero(),
+             CreatePacket(RtpPacketMediaType::kVideo,
+                          /*seq_no=*/1));
+  EXPECT_EQ(queue.SizeInPackets(), 1);
+
+  queue.Pop();
+  EXPECT_EQ(queue.SizeInPackets(), 0);
+}
+
+TEST(PrioritizedPacketQueue, ReportsPayloadSize) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Zero());
+
+  queue.Push(/*enqueue_time=*/Timestamp::Zero(),
+             CreatePacket(RtpPacketMediaType::kVideo,
+                          /*seq_no=*/1));
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Bytes(kDefaultPayloadSize));
+
+  queue.Pop();
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Zero());
+}
+
+TEST(PrioritizedPacketQueue, ReportsPaddingSize) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Zero());
+  static constexpr DataSize kPaddingSize = DataSize::Bytes(190);
+
+  auto packet = std::make_unique<RtpPacketToSend>(/*extensions=*/nullptr);
+  packet->set_packet_type(RtpPacketMediaType::kPadding);
+  packet->SetSsrc(kDefaultSsrc);
+  packet->SetSequenceNumber(/*seq=*/1);
+  packet->SetPadding(kPaddingSize.bytes());
+  queue.Push(/*enqueue_time=*/Timestamp::Zero(), std::move(packet));
+  EXPECT_EQ(queue.SizeInPayloadBytes(), kPaddingSize);
+
+  queue.Pop();
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Zero());
+}
+
+TEST(PrioritizedPacketQueue, ReportsOldestEnqueueTime) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::MinusInfinity());
+
+  // Add three packets, with the middle packet having higher prio.
+  queue.Push(Timestamp::Millis(10),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/1));
+  queue.Push(Timestamp::Millis(20),
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+  queue.Push(Timestamp::Millis(30),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/3));
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::Millis(10));
+
+  queue.Pop();  // Pop packet with enqueue time 20.
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::Millis(10));
+
+  queue.Pop();  // Pop packet with enqueue time 10.
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::Millis(30));
+
+  queue.Pop();  // Pop packet with enqueue time 30, queue empty again.
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::MinusInfinity());
+}
+
+TEST(PrioritizedPacketQueue, ReportsAverageQueueTime) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Zero());
+
+  // Add three packets, with the middle packet having higher prio.
+  queue.Push(Timestamp::Millis(10),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/1));
+  queue.Push(Timestamp::Millis(20),
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+  queue.Push(Timestamp::Millis(30),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/3));
+
+  queue.UpdateAverageQueueTime(Timestamp::Millis(40));
+  // Packets have waited 30, 20, 10 ms -> average = 20ms.
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(20));
+
+  queue.Pop();  // Pop packet with enqueue time 20.
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(20));
+
+  queue.Pop();  // Pop packet with enqueue time 10.
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(10));
+
+  queue.Pop();  // Pop packet with enqueue time 30, queue empty again.
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Zero());
+}
+
+TEST(PrioritizedPacketQueue, SubtractsPusedTimeFromAverageQueueTime) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Zero());
+
+  // Add a packet and then enable paused state.
+  queue.Push(Timestamp::Millis(100),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/1));
+  queue.SetPauseState(true, Timestamp::Millis(600));
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(500));
+
+  // Enqueue a packet 500ms into the paused state. Queue time of
+  // original packet is still seen as 500ms and new one has 0ms giving
+  // an average of 250ms.
+  queue.Push(Timestamp::Millis(1100),
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(250));
+
+  // Unpause some time later, queue time still unchanged.
+  queue.SetPauseState(false, Timestamp::Millis(1600));
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(250));
+
+  // Update queue time 500ms after pause state ended.
+  queue.UpdateAverageQueueTime(Timestamp::Millis(2100));
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(750));
+}
+
+TEST(PrioritizedPacketQueue, ReportsLeadingPacketEnqueueTime) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kAudio),
+            Timestamp::MinusInfinity());
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kVideo),
+            Timestamp::MinusInfinity());
+
+  queue.Push(Timestamp::Millis(10),
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/1));
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kAudio),
+            Timestamp::MinusInfinity());
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kVideo),
+            Timestamp::Millis(10));
+
+  queue.Push(Timestamp::Millis(20),
+             CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/2));
+
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kAudio),
+            Timestamp::Millis(20));
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kVideo),
+            Timestamp::Millis(10));
+
+  queue.Pop();  // Pop audio packet.
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kAudio),
+            Timestamp::MinusInfinity());
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kVideo),
+            Timestamp::Millis(10));
+
+  queue.Pop();  // Pop video packet.
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kAudio),
+            Timestamp::MinusInfinity());
+  EXPECT_EQ(queue.LeadingPacketEnqueueTime(RtpPacketMediaType::kVideo),
+            Timestamp::MinusInfinity());
+}
+
+TEST(PrioritizedPacketQueue,
+     PushAndPopUpdatesSizeInPacketsPerRtpPacketMediaType) {
+  Timestamp now = Timestamp::Zero();
+  PrioritizedPacketQueue queue(now);
+
+  // Initially all sizes are zero.
+  for (size_t i = 0; i < kNumMediaTypes; ++i) {
+    EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[i], 0);
+  }
+
+  // Push packets.
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, 1));
+  EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[static_cast<size_t>(
+                RtpPacketMediaType::kAudio)],
+            1);
+
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, 2));
+  EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[static_cast<size_t>(
+                RtpPacketMediaType::kVideo)],
+            1);
+
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, 3));
+  EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[static_cast<size_t>(
+                RtpPacketMediaType::kRetransmission)],
+            1);
+
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kForwardErrorCorrection, 4));
+  EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[static_cast<size_t>(
+                RtpPacketMediaType::kForwardErrorCorrection)],
+            1);
+
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kPadding, 5));
+  EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[static_cast<size_t>(
+                RtpPacketMediaType::kPadding)],
+            1);
+
+  // Now all sizes are 1.
+  for (size_t i = 0; i < kNumMediaTypes; ++i) {
+    EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[i], 1);
+  }
+
+  // Popping happens in a priority order based on media type. This test does not
+  // assert what this order is, only that the counter for the popped packet's
+  // media type is decremented.
+  for (size_t i = 0; i < kNumMediaTypes; ++i) {
+    auto popped_packet = queue.Pop();
+    EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[static_cast<size_t>(
+                  popped_packet->packet_type().value())],
+              0);
+  }
+
+  // We've popped all packets, so all sizes are zero.
+  for (size_t i = 0; i < kNumMediaTypes; ++i) {
+    EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[i], 0);
+  }
+}
+
+TEST(PrioritizedPacketQueue, ClearsPackets) {
+  Timestamp now = Timestamp::Zero();
+  PrioritizedPacketQueue queue(now);
+  const uint32_t kSsrc = 1;
+
+  // Add two packets of each type, all using the same SSRC.
+  int sequence_number = 0;
+  for (size_t i = 0; i < kNumMediaTypes; ++i) {
+    queue.Push(now, CreatePacket(static_cast<RtpPacketMediaType>(i),
+                                 sequence_number++, kSsrc));
+    queue.Push(now, CreatePacket(static_cast<RtpPacketMediaType>(i),
+                                 sequence_number++, kSsrc));
+  }
+  EXPECT_EQ(queue.SizeInPackets(), 2 * int{kNumMediaTypes});
+
+  // Remove all of them.
+  queue.RemovePacketsForSsrc(kSsrc);
+  EXPECT_TRUE(queue.Empty());
+}
+
+TEST(PrioritizedPacketQueue, ClearPacketsAffectsOnlySpecifiedSsrc) {
+  Timestamp now = Timestamp::Zero();
+  PrioritizedPacketQueue queue(now);
+  const uint32_t kRemovingSsrc = 1;
+  const uint32_t kStayingSsrc = 2;
+
+  // Add an audio packet and a retransmission for the SSRC we will remove,
+  // ensuring they are first in line.
+  queue.Push(
+      now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/1, kRemovingSsrc));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/2,
+                               kRemovingSsrc));
+
+  // Add a video packet and a retransmission for the SSRC that will remain.
+  // The retransmission packets now both have pointers to their respective qeues
+  // from the same prio level.
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/3, kStayingSsrc));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/4,
+                               kStayingSsrc));
+
+  EXPECT_EQ(queue.SizeInPackets(), 4);
+
+  // Clear the first two packets.
+  queue.RemovePacketsForSsrc(kRemovingSsrc);
+  EXPECT_EQ(queue.SizeInPackets(), 2);
+
+  // We should get the single remaining retransmission first, then the video
+  // packet.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 4);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 3);
+  EXPECT_TRUE(queue.Empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h b/third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h
new file mode 100644
index 0000000000..e2cf806385
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/rtp_packet_pacer.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PACING_RTP_PACKET_PACER_H_
+#define MODULES_PACING_RTP_PACKET_PACER_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
+
+namespace webrtc {
+
+class RtpPacketPacer {
+ public:
+  virtual ~RtpPacketPacer() = default;
+
+  virtual void CreateProbeClusters(
+      std::vector<ProbeClusterConfig> probe_cluster_configs) = 0;
+
+  // Temporarily pause all sending.
+  virtual void Pause() = 0;
+
+  // Resume sending packets.
+  virtual void Resume() = 0;
+
+  virtual void SetCongested(bool congested) = 0;
+
+  // Sets the pacing rates. Must be called once before packets can be sent.
+  virtual void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) = 0;
+
+  // Time since the oldest packet currently in the queue was added.
+  virtual TimeDelta OldestPacketWaitTime() const = 0;
+
+  // Sum of payload + padding bytes of all packets currently in the pacer queue.
+  virtual DataSize QueueSizeData() const = 0;
+
+  // Returns the time when the first packet was sent.
+  virtual absl::optional<Timestamp> FirstSentPacketTime() const = 0;
+
+  // Returns the expected number of milliseconds it will take to send the
+  // current packets in the queue, given the current size and bitrate, ignoring
+  // priority.
+  virtual TimeDelta ExpectedQueueTime() const = 0;
+
+  // Set the average upper bound on pacer queuing delay. The pacer may send at
+  // a higher rate than what was configured via SetPacingRates() in order to
+  // keep ExpectedQueueTimeMs() below `limit_ms` on average.
+  virtual void SetQueueTimeLimit(TimeDelta limit) = 0;
+
+  // Currently audio traffic is not accounted by pacer and passed through.
+  // With the introduction of audio BWE audio traffic will be accounted for
+  // the pacer budget calculation. The audio traffic still will be injected
+  // at high priority.
+  virtual void SetAccountForAudioPackets(bool account_for_audio) = 0;
+  virtual void SetIncludeOverhead() = 0;
+  virtual void SetTransportOverhead(DataSize overhead_per_packet) = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_PACING_RTP_PACKET_PACER_H_
diff --git a/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.cc b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.cc
new file mode 100644
index 0000000000..4ba249582c
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.cc
@@ -0,0 +1,372 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/task_queue_paced_sender.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/transport/network_types.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/field_trial_units.h"
+#include "rtc_base/system/unused.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr const char* kBurstyPacerFieldTrial = "WebRTC-BurstyPacer";
+
+constexpr const char* kSlackedTaskQueuePacedSenderFieldTrial =
+    "WebRTC-SlackedTaskQueuePacedSender";
+
+}  // namespace
+
+const int TaskQueuePacedSender::kNoPacketHoldback = -1;
+
+TaskQueuePacedSender::BurstyPacerFlags::BurstyPacerFlags(
+    const FieldTrialsView& field_trials)
+    : burst("burst") {
+  ParseFieldTrial({&burst}, field_trials.Lookup(kBurstyPacerFieldTrial));
+}
+
+TaskQueuePacedSender::SlackedPacerFlags::SlackedPacerFlags(
+    const FieldTrialsView& field_trials)
+    : allow_low_precision("Enabled"),
+      max_low_precision_expected_queue_time("max_queue_time"),
+      send_burst_interval("send_burst_interval") {
+  ParseFieldTrial({&allow_low_precision, &max_low_precision_expected_queue_time,
+                   &send_burst_interval},
+                  field_trials.Lookup(kSlackedTaskQueuePacedSenderFieldTrial));
+}
+
+TaskQueuePacedSender::TaskQueuePacedSender(
+    Clock* clock,
+    PacingController::PacketSender* packet_sender,
+    const FieldTrialsView& field_trials,
+    TaskQueueFactory* task_queue_factory,
+    TimeDelta max_hold_back_window,
+    int max_hold_back_window_in_packets,
+    absl::optional<TimeDelta> burst_interval)
+    : clock_(clock),
+      bursty_pacer_flags_(field_trials),
+      slacked_pacer_flags_(field_trials),
+      max_hold_back_window_(slacked_pacer_flags_.allow_low_precision
+                                ? PacingController::kMinSleepTime
+                                : max_hold_back_window),
+      max_hold_back_window_in_packets_(slacked_pacer_flags_.allow_low_precision
+                                           ? 0
+                                           : max_hold_back_window_in_packets),
+      pacing_controller_(clock, packet_sender, field_trials),
+      next_process_time_(Timestamp::MinusInfinity()),
+      is_started_(false),
+      is_shutdown_(false),
+      packet_size_(/*alpha=*/0.95),
+      include_overhead_(false),
+      task_queue_(field_trials, "TaskQueuePacedSender", task_queue_factory) {
+  RTC_DCHECK_GE(max_hold_back_window_, PacingController::kMinSleepTime);
+  // There are multiple field trials that can affect burst. If multiple bursts
+  // are specified we pick the largest of the values.
+  absl::optional<TimeDelta> burst = bursty_pacer_flags_.burst.GetOptional();
+  if (slacked_pacer_flags_.allow_low_precision &&
+      slacked_pacer_flags_.send_burst_interval) {
+    TimeDelta slacked_burst = slacked_pacer_flags_.send_burst_interval.Value();
+    if (!burst.has_value() || burst.value() < slacked_burst) {
+      burst = slacked_burst;
+    }
+  }
+  // If not overriden by an experiment, the burst is specified by the
+  // `burst_interval` argument.
+  if (!burst.has_value()) {
+    burst = burst_interval;
+  }
+  if (burst.has_value()) {
+    pacing_controller_.SetSendBurstInterval(burst.value());
+  }
+}
+
+TaskQueuePacedSender::~TaskQueuePacedSender() {
+  // Post an immediate task to mark the queue as shutting down.
+  // The rtc::TaskQueue destructor will wait for pending tasks to
+  // complete before continuing.
+  task_queue_.RunOrPost([&]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    is_shutdown_ = true;
+  });
+}
+
+void TaskQueuePacedSender::EnsureStarted() {
+  task_queue_.RunOrPost([this]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    is_started_ = true;
+    MaybeProcessPackets(Timestamp::MinusInfinity());
+  });
+}
+
+void TaskQueuePacedSender::CreateProbeClusters(
+    std::vector<ProbeClusterConfig> probe_cluster_configs) {
+  task_queue_.RunOrPost(
+      [this, probe_cluster_configs = std::move(probe_cluster_configs)]() {
+        RTC_DCHECK_RUN_ON(&task_queue_);
+        pacing_controller_.CreateProbeClusters(probe_cluster_configs);
+        MaybeProcessPackets(Timestamp::MinusInfinity());
+      });
+}
+
+void TaskQueuePacedSender::Pause() {
+  task_queue_.RunOrPost([this]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    pacing_controller_.Pause();
+  });
+}
+
+void TaskQueuePacedSender::Resume() {
+  task_queue_.RunOrPost([this]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    pacing_controller_.Resume();
+    MaybeProcessPackets(Timestamp::MinusInfinity());
+  });
+}
+
+void TaskQueuePacedSender::SetCongested(bool congested) {
+  task_queue_.RunOrPost([this, congested]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    pacing_controller_.SetCongested(congested);
+    MaybeProcessPackets(Timestamp::MinusInfinity());
+  });
+}
+
+void TaskQueuePacedSender::SetPacingRates(DataRate pacing_rate,
+                                          DataRate padding_rate) {
+  task_queue_.RunOrPost([this, pacing_rate, padding_rate]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    pacing_controller_.SetPacingRates(pacing_rate, padding_rate);
+    MaybeProcessPackets(Timestamp::MinusInfinity());
+  });
+}
+
+void TaskQueuePacedSender::EnqueuePackets(
+    std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+  task_queue_.TaskQueueForPost()->PostTask(task_queue_.MaybeSafeTask(
+      safety_.flag(), [this, packets = std::move(packets)]() mutable {
+        RTC_DCHECK_RUN_ON(&task_queue_);
+        TRACE_EVENT0(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+                     "TaskQueuePacedSender::EnqueuePackets");
+        for (auto& packet : packets) {
+          TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+                       "TaskQueuePacedSender::EnqueuePackets::Loop",
+                       "sequence_number", packet->SequenceNumber(),
+                       "rtp_timestamp", packet->Timestamp());
+
+          size_t packet_size = packet->payload_size() + packet->padding_size();
+          if (include_overhead_) {
+            packet_size += packet->headers_size();
+          }
+          packet_size_.Apply(1, packet_size);
+          RTC_DCHECK_GE(packet->capture_time(), Timestamp::Zero());
+          pacing_controller_.EnqueuePacket(std::move(packet));
+        }
+        MaybeProcessPackets(Timestamp::MinusInfinity());
+      }));
+}
+
+void TaskQueuePacedSender::RemovePacketsForSsrc(uint32_t ssrc) {
+  task_queue_.RunOrPost([this, ssrc]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    pacing_controller_.RemovePacketsForSsrc(ssrc);
+    MaybeProcessPackets(Timestamp::MinusInfinity());
+  });
+}
+
+void TaskQueuePacedSender::SetAccountForAudioPackets(bool account_for_audio) {
+  task_queue_.RunOrPost([this, account_for_audio]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    pacing_controller_.SetAccountForAudioPackets(account_for_audio);
+    MaybeProcessPackets(Timestamp::MinusInfinity());
+  });
+}
+
+void TaskQueuePacedSender::SetIncludeOverhead() {
+  task_queue_.RunOrPost([this]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    include_overhead_ = true;
+    pacing_controller_.SetIncludeOverhead();
+    MaybeProcessPackets(Timestamp::MinusInfinity());
+  });
+}
+
+void TaskQueuePacedSender::SetTransportOverhead(DataSize overhead_per_packet) {
+  task_queue_.RunOrPost([this, overhead_per_packet]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    pacing_controller_.SetTransportOverhead(overhead_per_packet);
+    MaybeProcessPackets(Timestamp::MinusInfinity());
+  });
+}
+
+void TaskQueuePacedSender::SetQueueTimeLimit(TimeDelta limit) {
+  task_queue_.RunOrPost([this, limit]() {
+    RTC_DCHECK_RUN_ON(&task_queue_);
+    pacing_controller_.SetQueueTimeLimit(limit);
+    MaybeProcessPackets(Timestamp::MinusInfinity());
+  });
+}
+
+TimeDelta TaskQueuePacedSender::ExpectedQueueTime() const {
+  return GetStats().expected_queue_time;
+}
+
+DataSize TaskQueuePacedSender::QueueSizeData() const {
+  return GetStats().queue_size;
+}
+
+absl::optional<Timestamp> TaskQueuePacedSender::FirstSentPacketTime() const {
+  return GetStats().first_sent_packet_time;
+}
+
+TimeDelta TaskQueuePacedSender::OldestPacketWaitTime() const {
+  Timestamp oldest_packet = GetStats().oldest_packet_enqueue_time;
+  if (oldest_packet.IsInfinite()) {
+    return TimeDelta::Zero();
+  }
+
+  // (webrtc:9716): The clock is not always monotonic.
+  Timestamp current = clock_->CurrentTime();
+  if (current < oldest_packet) {
+    return TimeDelta::Zero();
+  }
+
+  return current - oldest_packet;
+}
+
+void TaskQueuePacedSender::OnStatsUpdated(const Stats& stats) {
+  MutexLock lock(&stats_mutex_);
+  current_stats_ = stats;
+}
+
+void TaskQueuePacedSender::MaybeProcessPackets(
+    Timestamp scheduled_process_time) {
+  RTC_DCHECK_RUN_ON(&task_queue_);
+
+  TRACE_EVENT0(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+               "TaskQueuePacedSender::MaybeProcessPackets");
+
+  if (is_shutdown_ || !is_started_) {
+    return;
+  }
+
+  Timestamp next_send_time = pacing_controller_.NextSendTime();
+  RTC_DCHECK(next_send_time.IsFinite());
+  const Timestamp now = clock_->CurrentTime();
+  TimeDelta early_execute_margin =
+      pacing_controller_.IsProbing()
+          ? PacingController::kMaxEarlyProbeProcessing
+          : TimeDelta::Zero();
+
+  // Process packets and update stats.
+  while (next_send_time <= now + early_execute_margin) {
+    pacing_controller_.ProcessPackets();
+    next_send_time = pacing_controller_.NextSendTime();
+    RTC_DCHECK(next_send_time.IsFinite());
+
+    // Probing state could change. Get margin after process packets.
+    early_execute_margin = pacing_controller_.IsProbing()
+                               ? PacingController::kMaxEarlyProbeProcessing
+                               : TimeDelta::Zero();
+  }
+  UpdateStats();
+
+  // Ignore retired scheduled task, otherwise reset `next_process_time_`.
+  if (scheduled_process_time.IsFinite()) {
+    if (scheduled_process_time != next_process_time_) {
+      return;
+    }
+    next_process_time_ = Timestamp::MinusInfinity();
+  }
+
+  // Do not hold back in probing.
+  TimeDelta hold_back_window = TimeDelta::Zero();
+  if (!pacing_controller_.IsProbing()) {
+    hold_back_window = max_hold_back_window_;
+    DataRate pacing_rate = pacing_controller_.pacing_rate();
+    if (max_hold_back_window_in_packets_ != kNoPacketHoldback &&
+        !pacing_rate.IsZero() &&
+        packet_size_.filtered() != rtc::ExpFilter::kValueUndefined) {
+      TimeDelta avg_packet_send_time =
+          DataSize::Bytes(packet_size_.filtered()) / pacing_rate;
+      hold_back_window =
+          std::min(hold_back_window,
+                   avg_packet_send_time * max_hold_back_window_in_packets_);
+    }
+  }
+
+  // Calculate next process time.
+  TimeDelta time_to_next_process =
+      std::max(hold_back_window, next_send_time - now - early_execute_margin);
+  next_send_time = now + time_to_next_process;
+
+  // If no in flight task or in flight task is later than `next_send_time`,
+  // schedule a new one. Previous in flight task will be retired.
+  if (next_process_time_.IsMinusInfinity() ||
+      next_process_time_ > next_send_time) {
+    // Prefer low precision if allowed and not probing.
+    TaskQueueBase::DelayPrecision precision =
+        slacked_pacer_flags_.allow_low_precision &&
+                !pacing_controller_.IsProbing()
+            ? TaskQueueBase::DelayPrecision::kLow
+            : TaskQueueBase::DelayPrecision::kHigh;
+    // Check for cases where we need high precision.
+    if (precision == TaskQueueBase::DelayPrecision::kLow) {
+      auto& packets_per_type =
+          pacing_controller_.SizeInPacketsPerRtpPacketMediaType();
+      bool audio_or_retransmission_packets_in_queue =
+          packets_per_type[static_cast<size_t>(RtpPacketMediaType::kAudio)] >
+              0 ||
+          packets_per_type[static_cast<size_t>(
+              RtpPacketMediaType::kRetransmission)] > 0;
+      bool queue_time_too_large =
+          slacked_pacer_flags_.max_low_precision_expected_queue_time &&
+          pacing_controller_.ExpectedQueueTime() >=
+              slacked_pacer_flags_.max_low_precision_expected_queue_time
+                  .Value();
+      if (audio_or_retransmission_packets_in_queue || queue_time_too_large) {
+        precision = TaskQueueBase::DelayPrecision::kHigh;
+      }
+    }
+
+    task_queue_.TaskQueueForDelayedTasks()->PostDelayedTaskWithPrecision(
+        precision,
+        task_queue_.MaybeSafeTask(
+            safety_.flag(),
+            [this, next_send_time]() { MaybeProcessPackets(next_send_time); }),
+        time_to_next_process.RoundUpTo(TimeDelta::Millis(1)));
+    next_process_time_ = next_send_time;
+  }
+}
+
+void TaskQueuePacedSender::UpdateStats() {
+  Stats new_stats;
+  new_stats.expected_queue_time = pacing_controller_.ExpectedQueueTime();
+  new_stats.first_sent_packet_time = pacing_controller_.FirstSentPacketTime();
+  new_stats.oldest_packet_enqueue_time =
+      pacing_controller_.OldestPacketEnqueueTime();
+  new_stats.queue_size = pacing_controller_.QueueSizeData();
+  OnStatsUpdated(new_stats);
+}
+
+TaskQueuePacedSender::Stats TaskQueuePacedSender::GetStats() const {
+  MutexLock lock(&stats_mutex_);
+  return current_stats_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.h b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.h
new file mode 100644
index 0000000000..ea335fd8e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender.h
@@ -0,0 +1,209 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PACING_TASK_QUEUE_PACED_SENDER_H_
+#define MODULES_PACING_TASK_QUEUE_PACED_SENDER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/pacing/pacing_controller.h"
+#include "modules/pacing/rtp_packet_pacer.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/utility/maybe_worker_thread.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/numerics/exp_filter.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+class Clock;
+
+class TaskQueuePacedSender : public RtpPacketPacer, public RtpPacketSender {
+ public:
+  static const int kNoPacketHoldback;
+
+  // The pacer can be configured using `field_trials` or specified parameters.
+  //
+  // The `hold_back_window` parameter sets a lower bound on time to sleep if
+  // there is currently a pacer queue and packets can't immediately be
+  // processed. Increasing this reduces thread wakeups at the expense of higher
+  // latency.
+  //
+  // If the `burst_interval` parameter is set, the pacer is allowed to build up
+  // a packet "debt" that correspond to approximately the send rate during the
+  // specified interval. This greatly reduced wake ups by not pacing packets
+  // within the allowed burst budget.
+  TaskQueuePacedSender(
+      Clock* clock,
+      PacingController::PacketSender* packet_sender,
+      const FieldTrialsView& field_trials,
+      TaskQueueFactory* task_queue_factory,
+      TimeDelta max_hold_back_window,
+      int max_hold_back_window_in_packets,
+      absl::optional<TimeDelta> burst_interval = absl::nullopt);
+
+  ~TaskQueuePacedSender() override;
+
+  // Ensure that necessary delayed tasks are scheduled.
+  void EnsureStarted();
+
+  // Methods implementing RtpPacketSender.
+
+  // Adds the packet to the queue and calls
+  // PacingController::PacketSender::SendPacket() when it's time to send.
+  void EnqueuePackets(
+      std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;
+  // Remove any pending packets matching this SSRC from the packet queue.
+  void RemovePacketsForSsrc(uint32_t ssrc) override;
+
+  // Methods implementing RtpPacketPacer.
+
+  void CreateProbeClusters(
+      std::vector<ProbeClusterConfig> probe_cluster_configs) override;
+
+  // Temporarily pause all sending.
+  void Pause() override;
+
+  // Resume sending packets.
+  void Resume() override;
+
+  void SetCongested(bool congested) override;
+
+  // Sets the pacing rates. Must be called once before packets can be sent.
+  void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) override;
+
+  // Currently audio traffic is not accounted for by pacer and passed through.
+  // With the introduction of audio BWE, audio traffic will be accounted for
+  // in the pacer budget calculation. The audio traffic will still be injected
+  // at high priority.
+  void SetAccountForAudioPackets(bool account_for_audio) override;
+
+  void SetIncludeOverhead() override;
+  void SetTransportOverhead(DataSize overhead_per_packet) override;
+
+  // Returns the time since the oldest queued packet was enqueued.
+  TimeDelta OldestPacketWaitTime() const override;
+
+  // Returns total size of all packets in the pacer queue.
+  DataSize QueueSizeData() const override;
+
+  // Returns the time when the first packet was sent;
+  absl::optional<Timestamp> FirstSentPacketTime() const override;
+
+  // Returns the number of milliseconds it will take to send the current
+  // packets in the queue, given the current size and bitrate, ignoring prio.
+  TimeDelta ExpectedQueueTime() const override;
+
+  // Set the max desired queuing delay, pacer will override the pacing rate
+  // specified by SetPacingRates() if needed to achieve this goal.
+  void SetQueueTimeLimit(TimeDelta limit) override;
+
+ protected:
+  // Exposed as protected for test.
+  struct Stats {
+    Stats()
+        : oldest_packet_enqueue_time(Timestamp::MinusInfinity()),
+          queue_size(DataSize::Zero()),
+          expected_queue_time(TimeDelta::Zero()) {}
+    Timestamp oldest_packet_enqueue_time;
+    DataSize queue_size;
+    TimeDelta expected_queue_time;
+    absl::optional<Timestamp> first_sent_packet_time;
+  };
+  void OnStatsUpdated(const Stats& stats);
+
+ private:
+  // Check if it is time to send packets, or schedule a delayed task if not.
+  // Use Timestamp::MinusInfinity() to indicate that this call has _not_
+  // been scheduled by the pacing controller. If this is the case, check if
+  // can execute immediately otherwise schedule a delay task that calls this
+  // method again with desired (finite) scheduled process time.
+  void MaybeProcessPackets(Timestamp scheduled_process_time);
+
+  void UpdateStats() RTC_RUN_ON(task_queue_);
+  Stats GetStats() const;
+
+  Clock* const clock_;
+  struct BurstyPacerFlags {
+    // Parses `kBurstyPacerFieldTrial`. Example:
+    // --force-fieldtrials=WebRTC-BurstyPacer/burst:20ms/
+    explicit BurstyPacerFlags(const FieldTrialsView& field_trials);
+    // If set, the pacer is allowed to build up a packet "debt" that correspond
+    // to approximately the send rate during the specified interval.
+    FieldTrialOptional<TimeDelta> burst;
+  };
+  const BurstyPacerFlags bursty_pacer_flags_;
+  struct SlackedPacerFlags {
+    // Parses `kSlackedTaskQueuePacedSenderFieldTrial`. Example:
+    // --force-fieldtrials=WebRTC-SlackedTaskQueuePacedSender/Enabled,max_queue_time:75ms/
+    explicit SlackedPacerFlags(const FieldTrialsView& field_trials);
+    // When "Enabled", delayed tasks invoking MaybeProcessPackets() are
+    // scheduled using low precision instead of high precision, resulting in
+    // less idle wake ups and packets being sent in bursts if the `task_queue_`
+    // implementation supports slack. When probing, high precision is used
+    // regardless to ensure good bandwidth estimation.
+    FieldTrialFlag allow_low_precision;
+    // Controlled via the "max_queue_time" experiment argument. If set, uses
+    // high precision scheduling of MaybeProcessPackets() whenever the expected
+    // queue time is greater than or equal to this value.
+    FieldTrialOptional<TimeDelta> max_low_precision_expected_queue_time;
+    // Controlled via "send_burst_interval" experiment argument. If set, the
+    // pacer is allowed to build up a packet "debt" that correspond to
+    // approximately the send rate during the specified interval.
+    FieldTrialOptional<TimeDelta> send_burst_interval;
+  };
+  const SlackedPacerFlags slacked_pacer_flags_;
+  // The holdback window prevents too frequent delayed MaybeProcessPackets()
+  // calls. These are only applicable if `allow_low_precision` is false.
+  const TimeDelta max_hold_back_window_;
+  const int max_hold_back_window_in_packets_;
+
+  PacingController pacing_controller_ RTC_GUARDED_BY(task_queue_);
+
+  // We want only one (valid) delayed process task in flight at a time.
+  // If the value of `next_process_time_` is finite, it is an id for a
+  // delayed task that will call MaybeProcessPackets() with that time
+  // as parameter.
+  // Timestamp::MinusInfinity() indicates no valid pending task.
+  Timestamp next_process_time_ RTC_GUARDED_BY(task_queue_);
+
+  // Indicates if this task queue is started. If not, don't allow
+  // posting delayed tasks yet.
+  bool is_started_ RTC_GUARDED_BY(task_queue_);
+
+  // Indicates if this task queue is shutting down. If so, don't allow
+  // posting any more delayed tasks as that can cause the task queue to
+  // never drain.
+  bool is_shutdown_ RTC_GUARDED_BY(task_queue_);
+
+  // Filtered size of enqueued packets, in bytes.
+  rtc::ExpFilter packet_size_ RTC_GUARDED_BY(task_queue_);
+  bool include_overhead_ RTC_GUARDED_BY(task_queue_);
+
+  // TODO(webrtc:14502): Remove stats_mutex_ when pacer runs on the worker
+  // thread.
+  mutable Mutex stats_mutex_;
+  Stats current_stats_ RTC_GUARDED_BY(stats_mutex_);
+
+  ScopedTaskSafety safety_;
+  MaybeWorkerThread task_queue_;
+};
+}  // namespace webrtc
+#endif  // MODULES_PACING_TASK_QUEUE_PACED_SENDER_H_
diff --git a/third_party/libwebrtc/modules/pacing/task_queue_paced_sender_unittest.cc b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender_unittest.cc
new file mode 100644
index 0000000000..69c7b9b7ef
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/task_queue_paced_sender_unittest.cc
@@ -0,0 +1,913 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/pacing/task_queue_paced_sender.h"
+
+#include <algorithm>
+#include <atomic>
+#include <list>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "modules/pacing/packet_router.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Return;
+using ::testing::SaveArg;
+
+namespace webrtc {
+namespace {
+constexpr uint32_t kAudioSsrc = 12345;
+constexpr uint32_t kVideoSsrc = 234565;
+constexpr uint32_t kVideoRtxSsrc = 34567;
+constexpr uint32_t kFlexFecSsrc = 45678;
+constexpr size_t kDefaultPacketSize = 1234;
+
+class MockPacketRouter : public PacketRouter {
+ public:
+  MOCK_METHOD(void,
+              SendPacket,
+              (std::unique_ptr<RtpPacketToSend> packet,
+               const PacedPacketInfo& cluster_info),
+              (override));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              FetchFec,
+              (),
+              (override));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              GeneratePadding,
+              (DataSize target_size),
+              (override));
+};
+
+std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+    DataSize target_size) {
+  // 224 bytes is the max padding size for plain padding packets generated by
+  // RTPSender::GeneratePadding().
+  const DataSize kMaxPaddingPacketSize = DataSize::Bytes(224);
+  DataSize padding_generated = DataSize::Zero();
+  std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
+  while (padding_generated < target_size) {
+    DataSize packet_size =
+        std::min(target_size - padding_generated, kMaxPaddingPacketSize);
+    padding_generated += packet_size;
+    auto padding_packet =
+        std::make_unique<RtpPacketToSend>(/*extensions=*/nullptr);
+    padding_packet->set_packet_type(RtpPacketMediaType::kPadding);
+    padding_packet->SetPadding(packet_size.bytes());
+    padding_packets.push_back(std::move(padding_packet));
+  }
+  return padding_packets;
+}
+
+class TaskQueueWithFakePrecisionFactory : public TaskQueueFactory {
+ public:
+  explicit TaskQueueWithFakePrecisionFactory(
+      TaskQueueFactory* task_queue_factory)
+      : task_queue_factory_(task_queue_factory) {}
+
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
+      absl::string_view name,
+      Priority priority) const override {
+    return std::unique_ptr<TaskQueueBase, TaskQueueDeleter>(
+        new TaskQueueWithFakePrecision(
+            const_cast<TaskQueueWithFakePrecisionFactory*>(this),
+            task_queue_factory_));
+  }
+
+  int delayed_low_precision_count() const {
+    return delayed_low_precision_count_;
+  }
+  int delayed_high_precision_count() const {
+    return delayed_high_precision_count_;
+  }
+
+ private:
+  friend class TaskQueueWithFakePrecision;
+
+  class TaskQueueWithFakePrecision : public TaskQueueBase {
+   public:
+    TaskQueueWithFakePrecision(
+        TaskQueueWithFakePrecisionFactory* parent_factory,
+        TaskQueueFactory* task_queue_factory)
+        : parent_factory_(parent_factory),
+          task_queue_(task_queue_factory->CreateTaskQueue(
+              "TaskQueueWithFakePrecision",
+              TaskQueueFactory::Priority::NORMAL)) {}
+    ~TaskQueueWithFakePrecision() override {}
+
+    void Delete() override {
+      // `task_queue_->Delete()` is implicitly called in the destructor due to
+      // TaskQueueDeleter.
+      delete this;
+    }
+    void PostTask(absl::AnyInvocable<void() &&> task) override {
+      task_queue_->PostTask(WrapTask(std::move(task)));
+    }
+    void PostDelayedTask(absl::AnyInvocable<void() &&> task,
+                         TimeDelta delay) override {
+      ++parent_factory_->delayed_low_precision_count_;
+      task_queue_->PostDelayedTask(WrapTask(std::move(task)), delay);
+    }
+    void PostDelayedHighPrecisionTask(absl::AnyInvocable<void() &&> task,
+                                      TimeDelta delay) override {
+      ++parent_factory_->delayed_high_precision_count_;
+      task_queue_->PostDelayedHighPrecisionTask(WrapTask(std::move(task)),
+                                                delay);
+    }
+
+   private:
+    absl::AnyInvocable<void() &&> WrapTask(absl::AnyInvocable<void() &&> task) {
+      return [this, task = std::move(task)]() mutable {
+        CurrentTaskQueueSetter set_current(this);
+        std::move(task)();
+      };
+    }
+
+    TaskQueueWithFakePrecisionFactory* parent_factory_;
+    std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
+  };
+
+  TaskQueueFactory* task_queue_factory_;
+  std::atomic<int> delayed_low_precision_count_ = 0u;
+  std::atomic<int> delayed_high_precision_count_ = 0u;
+};
+
+}  // namespace
+
+namespace test {
+
+std::unique_ptr<RtpPacketToSend> BuildRtpPacket(RtpPacketMediaType type) {
+  auto packet = std::make_unique<RtpPacketToSend>(nullptr);
+  packet->set_packet_type(type);
+  switch (type) {
+    case RtpPacketMediaType::kAudio:
+      packet->SetSsrc(kAudioSsrc);
+      break;
+    case RtpPacketMediaType::kVideo:
+      packet->SetSsrc(kVideoSsrc);
+      break;
+    case RtpPacketMediaType::kRetransmission:
+    case RtpPacketMediaType::kPadding:
+      packet->SetSsrc(kVideoRtxSsrc);
+      break;
+    case RtpPacketMediaType::kForwardErrorCorrection:
+      packet->SetSsrc(kFlexFecSsrc);
+      break;
+  }
+
+  packet->SetPayloadSize(kDefaultPacketSize);
+  return packet;
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePackets(
+    RtpPacketMediaType type,
+    size_t num_packets) {
+  std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+  for (size_t i = 0; i < num_packets; ++i) {
+    packets.push_back(BuildRtpPacket(type));
+  }
+  return packets;
+}
+
+constexpr char kSendPacketOnWorkerThreadFieldTrial[] =
+    "WebRTC-SendPacketsOnWorkerThread/Enabled/";
+
+std::vector<std::string> ParameterizedFieldTrials() {
+  return {{""}, {kSendPacketOnWorkerThreadFieldTrial}};
+}
+
+bool UsingWorkerThread(absl::string_view field_trials) {
+  return field_trials.find(kSendPacketOnWorkerThreadFieldTrial) !=
+         std::string::npos;
+}
+
+class TaskQueuePacedSenderTest
+    : public ::testing::TestWithParam<std::string /*field_trials*/> {};
+
+INSTANTIATE_TEST_SUITE_P(TaskQueuePacedSenderTest,
+                         TaskQueuePacedSenderTest,
+                         testing::ValuesIn(ParameterizedFieldTrials()),
+                         [](const testing::TestParamInfo<std::string>& info) {
+                           return UsingWorkerThread(info.param) ? "UsingWt"
+                                                                : "OwnedTQ";
+                         });
+
+TEST_P(TaskQueuePacedSenderTest, PacesPackets) {
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  ScopedKeyValueConfig trials(GetParam());
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             PacingController::kMinSleepTime,
+                             TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Insert a number of packets, covering one second.
+  static constexpr size_t kPacketsToSend = 42;
+  SequenceChecker sequence_checker;
+  pacer.SetPacingRates(
+      DataRate::BitsPerSec(kDefaultPacketSize * 8 * kPacketsToSend),
+      DataRate::Zero());
+  pacer.EnsureStarted();
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kPacketsToSend));
+
+  // Expect all of them to be sent.
+  size_t packets_sent = 0;
+  Timestamp end_time = Timestamp::PlusInfinity();
+  EXPECT_CALL(packet_router, SendPacket)
+      .WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet,
+                          const PacedPacketInfo& cluster_info) {
+        ++packets_sent;
+        if (packets_sent == kPacketsToSend) {
+          end_time = time_controller.GetClock()->CurrentTime();
+        }
+        EXPECT_EQ(sequence_checker.IsCurrent(), UsingWorkerThread(GetParam()));
+      });
+
+  const Timestamp start_time = time_controller.GetClock()->CurrentTime();
+
+  // Packets should be sent over a period of close to 1s. Expect a little
+  // lower than this since initial probing is a bit quicker.
+  time_controller.AdvanceTime(TimeDelta::Seconds(1));
+  EXPECT_EQ(packets_sent, kPacketsToSend);
+  ASSERT_TRUE(end_time.IsFinite());
+  EXPECT_NEAR((end_time - start_time).ms<double>(), 1000.0, 50.0);
+}
+
+// Same test as above, but with 0.5s of burst applied.
+TEST_P(TaskQueuePacedSenderTest, PacesPacketsWithBurst) {
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  ScopedKeyValueConfig trials(GetParam());
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             PacingController::kMinSleepTime,
+                             TaskQueuePacedSender::kNoPacketHoldback,
+                             // Half a second of bursting.
+                             TimeDelta::Seconds(0.5));
+
+  // Insert a number of packets, covering one second.
+  static constexpr size_t kPacketsToSend = 42;
+  SequenceChecker sequence_checker;
+  pacer.SetPacingRates(
+      DataRate::BitsPerSec(kDefaultPacketSize * 8 * kPacketsToSend),
+      DataRate::Zero());
+  pacer.EnsureStarted();
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kPacketsToSend));
+
+  // Expect all of them to be sent.
+  size_t packets_sent = 0;
+  Timestamp end_time = Timestamp::PlusInfinity();
+  EXPECT_CALL(packet_router, SendPacket)
+      .WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet,
+                          const PacedPacketInfo& cluster_info) {
+        ++packets_sent;
+        if (packets_sent == kPacketsToSend) {
+          end_time = time_controller.GetClock()->CurrentTime();
+        }
+        EXPECT_EQ(sequence_checker.IsCurrent(), UsingWorkerThread(GetParam()));
+      });
+
+  const Timestamp start_time = time_controller.GetClock()->CurrentTime();
+
+  // Packets should be sent over a period of close to 1s. Expect a little
+  // lower than this since initial probing is a bit quicker.
+  time_controller.AdvanceTime(TimeDelta::Seconds(1));
+  EXPECT_EQ(packets_sent, kPacketsToSend);
+  ASSERT_TRUE(end_time.IsFinite());
+  // Because of half a second of burst, what would normally have been paced over
+  // ~1 second now takes ~0.5 seconds.
+  EXPECT_NEAR((end_time - start_time).ms<double>(), 500.0, 50.0);
+}
+
+TEST_P(TaskQueuePacedSenderTest, ReschedulesProcessOnRateChange) {
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  ScopedKeyValueConfig trials(GetParam());
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             PacingController::kMinSleepTime,
+                             TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Insert a number of packets to be sent 200ms apart.
+  const size_t kPacketsPerSecond = 5;
+  const DataRate kPacingRate =
+      DataRate::BitsPerSec(kDefaultPacketSize * 8 * kPacketsPerSecond);
+  pacer.SetPacingRates(kPacingRate, DataRate::Zero());
+  pacer.EnsureStarted();
+
+  // Send some initial packets to be rid of any probes.
+  EXPECT_CALL(packet_router, SendPacket).Times(kPacketsPerSecond);
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kPacketsPerSecond));
+  time_controller.AdvanceTime(TimeDelta::Seconds(1));
+
+  // Insert three packets, and record send time of each of them.
+  // After the second packet is sent, double the send rate so we can
+  // check the third packets is sent after half the wait time.
+  Timestamp first_packet_time = Timestamp::MinusInfinity();
+  Timestamp second_packet_time = Timestamp::MinusInfinity();
+  Timestamp third_packet_time = Timestamp::MinusInfinity();
+
+  EXPECT_CALL(packet_router, SendPacket)
+      .Times(3)
+      .WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet,
+                          const PacedPacketInfo& cluster_info) {
+        if (first_packet_time.IsInfinite()) {
+          first_packet_time = time_controller.GetClock()->CurrentTime();
+        } else if (second_packet_time.IsInfinite()) {
+          second_packet_time = time_controller.GetClock()->CurrentTime();
+          // Avoid invoke SetPacingRate in the context of sending a packet.
+          time_controller.GetMainThread()->PostTask(
+              [&] { pacer.SetPacingRates(2 * kPacingRate, DataRate::Zero()); });
+        } else {
+          third_packet_time = time_controller.GetClock()->CurrentTime();
+        }
+      });
+
+  pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 3));
+  time_controller.AdvanceTime(TimeDelta::Millis(500));
+  ASSERT_TRUE(third_packet_time.IsFinite());
+  EXPECT_NEAR((second_packet_time - first_packet_time).ms<double>(), 200.0,
+              1.0);
+  EXPECT_NEAR((third_packet_time - second_packet_time).ms<double>(), 100.0,
+              1.0);
+}
+
+TEST_P(TaskQueuePacedSenderTest, SendsAudioImmediately) {
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  ScopedKeyValueConfig trials(GetParam());
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             PacingController::kMinSleepTime,
+                             TaskQueuePacedSender::kNoPacketHoldback);
+
+  const DataRate kPacingDataRate = DataRate::KilobitsPerSec(125);
+  const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize);
+  const TimeDelta kPacketPacingTime = kPacketSize / kPacingDataRate;
+
+  pacer.SetPacingRates(kPacingDataRate, DataRate::Zero());
+  pacer.EnsureStarted();
+
+  // Add some initial video packets, only one should be sent.
+  EXPECT_CALL(packet_router, SendPacket);
+  pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 10));
+  time_controller.AdvanceTime(TimeDelta::Zero());
+  ::testing::Mock::VerifyAndClearExpectations(&packet_router);
+
+  // Advance time, but still before next packet should be sent.
+  time_controller.AdvanceTime(kPacketPacingTime / 2);
+
+  // Insert an audio packet, it should be sent immediately.
+  EXPECT_CALL(packet_router, SendPacket);
+  pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kAudio, 1));
+  time_controller.AdvanceTime(TimeDelta::Zero());
+  ::testing::Mock::VerifyAndClearExpectations(&packet_router);
+}
+
+TEST_P(TaskQueuePacedSenderTest, SleepsDuringCoalscingWindow) {
+  const TimeDelta kCoalescingWindow = TimeDelta::Millis(5);
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  ScopedKeyValueConfig trials(GetParam());
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             kCoalescingWindow,
+                             TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Set rates so one packet adds one ms of buffer level.
+  const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize);
+  const TimeDelta kPacketPacingTime = TimeDelta::Millis(1);
+  const DataRate kPacingDataRate = kPacketSize / kPacketPacingTime;
+
+  pacer.SetPacingRates(kPacingDataRate, DataRate::Zero());
+  pacer.EnsureStarted();
+
+  // Add 10 packets. The first should be sent immediately since the buffers
+  // are clear.
+  EXPECT_CALL(packet_router, SendPacket);
+  pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 10));
+  time_controller.AdvanceTime(TimeDelta::Zero());
+  ::testing::Mock::VerifyAndClearExpectations(&packet_router);
+
+  // Advance time to 1ms before the coalescing window ends. No packets should
+  // be sent.
+  EXPECT_CALL(packet_router, SendPacket).Times(0);
+  time_controller.AdvanceTime(kCoalescingWindow - TimeDelta::Millis(1));
+
+  // Advance time to where coalescing window ends. All packets that should
+  // have been sent up til now will be sent.
+  EXPECT_CALL(packet_router, SendPacket).Times(5);
+  time_controller.AdvanceTime(TimeDelta::Millis(1));
+  ::testing::Mock::VerifyAndClearExpectations(&packet_router);
+}
+
+TEST_P(TaskQueuePacedSenderTest, ProbingOverridesCoalescingWindow) {
+  const TimeDelta kCoalescingWindow = TimeDelta::Millis(5);
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  ScopedKeyValueConfig trials(GetParam());
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             kCoalescingWindow,
+                             TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Set rates so one packet adds one ms of buffer level.
+  const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize);
+  const TimeDelta kPacketPacingTime = TimeDelta::Millis(1);
+  const DataRate kPacingDataRate = kPacketSize / kPacketPacingTime;
+
+  pacer.SetPacingRates(kPacingDataRate, DataRate::Zero());
+  pacer.EnsureStarted();
+
+  // Add 10 packets. The first should be sent immediately since the buffers
+  // are clear. This will also trigger the probe to start.
+  EXPECT_CALL(packet_router, SendPacket).Times(AtLeast(1));
+  pacer.CreateProbeClusters(
+      {{.at_time = time_controller.GetClock()->CurrentTime(),
+        .target_data_rate = kPacingDataRate * 2,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 5,
+        .id = 17}});
+  pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 10));
+  time_controller.AdvanceTime(TimeDelta::Zero());
+  ::testing::Mock::VerifyAndClearExpectations(&packet_router);
+
+  // Advance time to 1ms before the coalescing window ends. Packets should be
+  // flying.
+  EXPECT_CALL(packet_router, SendPacket).Times(AtLeast(1));
+  time_controller.AdvanceTime(kCoalescingWindow - TimeDelta::Millis(1));
+}
+
+TEST_P(TaskQueuePacedSenderTest, SchedulesProbeAtSentTime) {
+  ScopedKeyValueConfig trials(
+      GetParam() + "WebRTC-Bwe-ProbingBehavior/min_probe_delta:1ms/");
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             PacingController::kMinSleepTime,
+                             TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Set rates so one packet adds 4ms of buffer level.
+  const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize);
+  const TimeDelta kPacketPacingTime = TimeDelta::Millis(4);
+  const DataRate kPacingDataRate = kPacketSize / kPacketPacingTime;
+  pacer.SetPacingRates(kPacingDataRate, /*padding_rate=*/DataRate::Zero());
+  pacer.EnsureStarted();
+  EXPECT_CALL(packet_router, FetchFec).WillRepeatedly([]() {
+    return std::vector<std::unique_ptr<RtpPacketToSend>>();
+  });
+  EXPECT_CALL(packet_router, GeneratePadding(_))
+      .WillRepeatedly(
+          [](DataSize target_size) { return GeneratePadding(target_size); });
+
+  // Enqueue two packets, only the first is sent immediately and the next
+  // will be scheduled for sending in 4ms.
+  pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 2));
+  const int kNotAProbe = PacedPacketInfo::kNotAProbe;
+  EXPECT_CALL(packet_router,
+              SendPacket(_, ::testing::Field(&PacedPacketInfo::probe_cluster_id,
+                                             kNotAProbe)));
+  // Advance to less than 3ms before next packet send time.
+  time_controller.AdvanceTime(TimeDelta::Micros(1001));
+
+  // Trigger a probe at 2x the current pacing rate and insert the number of
+  // packets the probe needs.
+  const DataRate kProbeRate = 2 * kPacingDataRate;
+  const int kProbeClusterId = 1;
+  pacer.CreateProbeClusters(
+      {{.at_time = time_controller.GetClock()->CurrentTime(),
+        .target_data_rate = kProbeRate,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 4,
+        .id = kProbeClusterId}});
+
+  // Expected size for each probe in a cluster is twice the expected bits sent
+  // during min_probe_delta.
+  // Expect one additional call since probe always starts with a small (1 byte)
+  // padding packet that's not counted into the probe rate here.
+  const TimeDelta kProbeTimeDelta = TimeDelta::Millis(2);
+  const DataSize kProbeSize = kProbeRate * kProbeTimeDelta;
+  const size_t kNumPacketsInProbe =
+      (kProbeSize + kPacketSize - DataSize::Bytes(1)) / kPacketSize;
+  EXPECT_CALL(packet_router,
+              SendPacket(_, ::testing::Field(&PacedPacketInfo::probe_cluster_id,
+                                             kProbeClusterId)))
+      .Times(kNumPacketsInProbe + 1);
+
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kNumPacketsInProbe));
+  time_controller.AdvanceTime(TimeDelta::Zero());
+
+  // The pacer should have scheduled the next probe to be sent in
+  // kProbeTimeDelta. That there was existing scheduled call less than
+  // PacingController::kMinSleepTime before this should not matter.
+  EXPECT_CALL(packet_router,
+              SendPacket(_, ::testing::Field(&PacedPacketInfo::probe_cluster_id,
+                                             kProbeClusterId)))
+      .Times(AtLeast(1));
+  time_controller.AdvanceTime(TimeDelta::Millis(2));
+}
+
+TEST_P(TaskQueuePacedSenderTest, NoMinSleepTimeWhenProbing) {
+  // Set min_probe_delta to be less than kMinSleepTime (1ms).
+  const TimeDelta kMinProbeDelta = TimeDelta::Micros(200);
+  ScopedKeyValueConfig trials(
+      GetParam() + "WebRTC-Bwe-ProbingBehavior/min_probe_delta:200us/");
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             PacingController::kMinSleepTime,
+                             TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Set rates so one packet adds 4ms of buffer level.
+  const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize);
+  const TimeDelta kPacketPacingTime = TimeDelta::Millis(4);
+  const DataRate kPacingDataRate = kPacketSize / kPacketPacingTime;
+  pacer.SetPacingRates(kPacingDataRate, /*padding_rate=*/DataRate::Zero());
+  pacer.EnsureStarted();
+  EXPECT_CALL(packet_router, FetchFec).WillRepeatedly([]() {
+    return std::vector<std::unique_ptr<RtpPacketToSend>>();
+  });
+  EXPECT_CALL(packet_router, GeneratePadding)
+      .WillRepeatedly(
+          [](DataSize target_size) { return GeneratePadding(target_size); });
+
+  // Set a high probe rate.
+  const int kProbeClusterId = 1;
+  DataRate kProbingRate = kPacingDataRate * 10;
+
+  pacer.CreateProbeClusters(
+      {{.at_time = time_controller.GetClock()->CurrentTime(),
+        .target_data_rate = kProbingRate,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 5,
+        .id = kProbeClusterId}});
+
+  // Advance time less than PacingController::kMinSleepTime, probing packets
+  // for the first millisecond should be sent immediately. Min delta between
+  // probes is 200us, meaning 4 times per ms we will get least one call to
+  // SendPacket().
+  DataSize data_sent = DataSize::Zero();
+  EXPECT_CALL(packet_router,
+              SendPacket(_, ::testing::Field(&PacedPacketInfo::probe_cluster_id,
+                                             kProbeClusterId)))
+      .Times(AtLeast(4))
+      .WillRepeatedly([&](std::unique_ptr<RtpPacketToSend> packet,
+                          const PacedPacketInfo&) {
+        data_sent +=
+            DataSize::Bytes(packet->payload_size() + packet->padding_size());
+      });
+
+  // Add one packet to kickstart probing, the rest will be padding packets.
+  pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 1));
+  time_controller.AdvanceTime(kMinProbeDelta);
+
+  // Verify the amount of probing data sent.
+  // Probe always starts with a small (1 byte) padding packet that's not
+  // counted into the probe rate here.
+  const DataSize kMinProbeSize = kMinProbeDelta * kProbingRate;
+  EXPECT_EQ(data_sent, DataSize::Bytes(1) + kPacketSize + 4 * kMinProbeSize);
+}
+
+TEST_P(TaskQueuePacedSenderTest, PacketBasedCoalescing) {
+  const TimeDelta kFixedCoalescingWindow = TimeDelta::Millis(10);
+  const int kPacketBasedHoldback = 5;
+
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  ScopedKeyValueConfig trials(GetParam());
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             kFixedCoalescingWindow, kPacketBasedHoldback);
+
+  // Set rates so one packet adds one ms of buffer level.
+  const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize);
+  const TimeDelta kPacketPacingTime = TimeDelta::Millis(1);
+  const DataRate kPacingDataRate = kPacketSize / kPacketPacingTime;
+  const TimeDelta kExpectedHoldbackWindow =
+      kPacketPacingTime * kPacketBasedHoldback;
+  // `kFixedCoalescingWindow` sets the upper bound for the window.
+  ASSERT_GE(kFixedCoalescingWindow, kExpectedHoldbackWindow);
+
+  pacer.SetPacingRates(kPacingDataRate, DataRate::Zero());
+  EXPECT_CALL(packet_router, FetchFec).WillRepeatedly([]() {
+    return std::vector<std::unique_ptr<RtpPacketToSend>>();
+  });
+  pacer.EnsureStarted();
+
+  // Add some packets and wait till all have been sent, so that the pacer
+  // has a valid estimate of packet size.
+  const int kNumWarmupPackets = 40;
+  EXPECT_CALL(packet_router, SendPacket).Times(kNumWarmupPackets);
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kNumWarmupPackets));
+  // Wait until all packes have been sent, with a 2x margin.
+  time_controller.AdvanceTime(kPacketPacingTime * (kNumWarmupPackets * 2));
+
+  // Enqueue packets. Expect only the first one to be sent immediately.
+  EXPECT_CALL(packet_router, SendPacket).Times(1);
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kPacketBasedHoldback));
+  time_controller.AdvanceTime(TimeDelta::Zero());
+
+  // Advance time to 1ms before the coalescing window ends.
+  EXPECT_CALL(packet_router, SendPacket).Times(0);
+  time_controller.AdvanceTime(kExpectedHoldbackWindow - TimeDelta::Millis(1));
+
+  // Advance past where the coalescing window should end.
+  EXPECT_CALL(packet_router, SendPacket).Times(kPacketBasedHoldback - 1);
+  time_controller.AdvanceTime(TimeDelta::Millis(1));
+}
+
+TEST_P(TaskQueuePacedSenderTest, FixedHoldBackHasPriorityOverPackets) {
+  const TimeDelta kFixedCoalescingWindow = TimeDelta::Millis(2);
+  const int kPacketBasedHoldback = 5;
+
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  ScopedKeyValueConfig trials(GetParam());
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             kFixedCoalescingWindow, kPacketBasedHoldback);
+
+  // Set rates so one packet adds one ms of buffer level.
+  const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize);
+  const TimeDelta kPacketPacingTime = TimeDelta::Millis(1);
+  const DataRate kPacingDataRate = kPacketSize / kPacketPacingTime;
+  const TimeDelta kExpectedPacketHoldbackWindow =
+      kPacketPacingTime * kPacketBasedHoldback;
+  // |kFixedCoalescingWindow| sets the upper bound for the window.
+  ASSERT_LT(kFixedCoalescingWindow, kExpectedPacketHoldbackWindow);
+
+  pacer.SetPacingRates(kPacingDataRate, DataRate::Zero());
+  EXPECT_CALL(packet_router, FetchFec).WillRepeatedly([]() {
+    return std::vector<std::unique_ptr<RtpPacketToSend>>();
+  });
+  pacer.EnsureStarted();
+
+  // Add some packets and wait till all have been sent, so that the pacer
+  // has a valid estimate of packet size.
+  const int kNumWarmupPackets = 40;
+  EXPECT_CALL(packet_router, SendPacket).Times(kNumWarmupPackets);
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kNumWarmupPackets));
+  // Wait until all packes have been sent, with a 2x margin.
+  time_controller.AdvanceTime(kPacketPacingTime * (kNumWarmupPackets * 2));
+
+  // Enqueue packets. Expect onlt the first one to be sent immediately.
+  EXPECT_CALL(packet_router, SendPacket).Times(1);
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kPacketBasedHoldback));
+  time_controller.AdvanceTime(TimeDelta::Zero());
+
+  // Advance time to the fixed coalescing window, that should take presedence so
+  // at least some of the packets should be sent.
+  EXPECT_CALL(packet_router, SendPacket).Times(AtLeast(1));
+  time_controller.AdvanceTime(kFixedCoalescingWindow);
+}
+
+TEST_P(TaskQueuePacedSenderTest, ProbingStopDuringSendLoop) {
+  // Set a low `min_probe_delta` to let probing finish during send loop.
+  ScopedKeyValueConfig trials(
+      GetParam() + "WebRTC-Bwe-ProbingBehavior/min_probe_delta:100us/");
+
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  MockPacketRouter packet_router;
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             PacingController::kMinSleepTime,
+                             TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Set rates so 2 packets adds 1ms of buffer level.
+  const DataSize kPacketSize = DataSize::Bytes(kDefaultPacketSize);
+  const TimeDelta kPacketPacingTime = TimeDelta::Millis(1);
+  const DataRate kPacingDataRate = 2 * kPacketSize / kPacketPacingTime;
+
+  pacer.SetPacingRates(kPacingDataRate, DataRate::Zero());
+  pacer.EnsureStarted();
+
+  EXPECT_CALL(packet_router, FetchFec).WillRepeatedly([]() {
+    return std::vector<std::unique_ptr<RtpPacketToSend>>();
+  });
+  EXPECT_CALL(packet_router, GeneratePadding(_))
+      .WillRepeatedly(
+          [](DataSize target_size) { return GeneratePadding(target_size); });
+
+  // Set probe rate.
+  const int kProbeClusterId = 1;
+  const DataRate kProbingRate = kPacingDataRate;
+
+  pacer.CreateProbeClusters(
+      {{.at_time = time_controller.GetClock()->CurrentTime(),
+        .target_data_rate = kProbingRate,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 4,
+        .id = kProbeClusterId}});
+
+  const int kPacketsToSend = 100;
+  const TimeDelta kPacketsPacedTime =
+      std::max(kPacketsToSend * kPacketSize / kPacingDataRate,
+               kPacketsToSend * kPacketSize / kProbingRate);
+
+  // Expect all packets and one padding packet sent.
+  EXPECT_CALL(packet_router, SendPacket).Times(kPacketsToSend + 1);
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kPacketsToSend));
+  time_controller.AdvanceTime(kPacketsPacedTime + TimeDelta::Millis(1));
+}
+
+TEST_P(TaskQueuePacedSenderTest, Stats) {
+  static constexpr Timestamp kStartTime = Timestamp::Millis(1234);
+  GlobalSimulatedTimeController time_controller(kStartTime);
+  MockPacketRouter packet_router;
+  ScopedKeyValueConfig trials(GetParam());
+  TaskQueuePacedSender pacer(time_controller.GetClock(), &packet_router, trials,
+                             time_controller.GetTaskQueueFactory(),
+                             PacingController::kMinSleepTime,
+                             TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Simulate ~2mbps video stream, covering one second.
+  static constexpr size_t kPacketsToSend = 200;
+  static constexpr DataRate kPacingRate =
+      DataRate::BytesPerSec(kDefaultPacketSize * kPacketsToSend);
+  pacer.SetPacingRates(kPacingRate, DataRate::Zero());
+  pacer.EnsureStarted();
+
+  // Allowed `QueueSizeData` and `ExpectedQueueTime` deviation.
+  static constexpr size_t kAllowedPacketsDeviation = 1;
+  static constexpr DataSize kAllowedQueueSizeDeviation =
+      DataSize::Bytes(kDefaultPacketSize * kAllowedPacketsDeviation);
+  static constexpr TimeDelta kAllowedQueueTimeDeviation =
+      kAllowedQueueSizeDeviation / kPacingRate;
+
+  DataSize expected_queue_size = DataSize::MinusInfinity();
+  TimeDelta expected_queue_time = TimeDelta::MinusInfinity();
+
+  EXPECT_CALL(packet_router, SendPacket).Times(kPacketsToSend);
+
+  // Stats before insert any packets.
+  EXPECT_TRUE(pacer.OldestPacketWaitTime().IsZero());
+  EXPECT_FALSE(pacer.FirstSentPacketTime().has_value());
+  EXPECT_TRUE(pacer.QueueSizeData().IsZero());
+  EXPECT_TRUE(pacer.ExpectedQueueTime().IsZero());
+
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kPacketsToSend));
+
+  // Advance to 200ms.
+  time_controller.AdvanceTime(TimeDelta::Millis(200));
+  EXPECT_EQ(pacer.OldestPacketWaitTime(), TimeDelta::Millis(200));
+  EXPECT_EQ(pacer.FirstSentPacketTime(), kStartTime);
+
+  expected_queue_size = kPacingRate * TimeDelta::Millis(800);
+  expected_queue_time = expected_queue_size / kPacingRate;
+  EXPECT_NEAR(pacer.QueueSizeData().bytes(), expected_queue_size.bytes(),
+              kAllowedQueueSizeDeviation.bytes());
+  EXPECT_NEAR(pacer.ExpectedQueueTime().ms(), expected_queue_time.ms(),
+              kAllowedQueueTimeDeviation.ms());
+
+  // Advance to 500ms.
+  time_controller.AdvanceTime(TimeDelta::Millis(300));
+  EXPECT_EQ(pacer.OldestPacketWaitTime(), TimeDelta::Millis(500));
+  EXPECT_EQ(pacer.FirstSentPacketTime(), kStartTime);
+
+  expected_queue_size = kPacingRate * TimeDelta::Millis(500);
+  expected_queue_time = expected_queue_size / kPacingRate;
+  EXPECT_NEAR(pacer.QueueSizeData().bytes(), expected_queue_size.bytes(),
+              kAllowedQueueSizeDeviation.bytes());
+  EXPECT_NEAR(pacer.ExpectedQueueTime().ms(), expected_queue_time.ms(),
+              kAllowedQueueTimeDeviation.ms());
+
+  // Advance to 1000ms+, expect all packets to be sent.
+  time_controller.AdvanceTime(TimeDelta::Millis(500) +
+                              kAllowedQueueTimeDeviation);
+  EXPECT_TRUE(pacer.OldestPacketWaitTime().IsZero());
+  EXPECT_EQ(pacer.FirstSentPacketTime(), kStartTime);
+  EXPECT_TRUE(pacer.QueueSizeData().IsZero());
+  EXPECT_TRUE(pacer.ExpectedQueueTime().IsZero());
+}
+
+// TODO(webrtc:14502): Rewrite these tests if the functionality is needed if
+// pacing is done on the worker thread.
+TEST(TaskQueuePacedSenderTest, HighPrecisionPacingWhenSlackIsDisabled) {
+  ScopedKeyValueConfig trials("WebRTC-SlackedTaskQueuePacedSender/Disabled/");
+
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  TaskQueueWithFakePrecisionFactory task_queue_factory(
+      time_controller.GetTaskQueueFactory());
+
+  MockPacketRouter packet_router;
+  TaskQueuePacedSender pacer(
+      time_controller.GetClock(), &packet_router, trials, &task_queue_factory,
+      PacingController::kMinSleepTime, TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Send enough packets (covering one second) that pacing is triggered, i.e.
+  // delayed tasks being scheduled.
+  static constexpr size_t kPacketsToSend = 42;
+  static constexpr DataRate kPacingRate =
+      DataRate::BitsPerSec(kDefaultPacketSize * 8 * kPacketsToSend);
+  pacer.SetPacingRates(kPacingRate, DataRate::Zero());
+  pacer.EnsureStarted();
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kPacketsToSend));
+  // Expect all of them to be sent.
+  size_t packets_sent = 0;
+  EXPECT_CALL(packet_router, SendPacket)
+      .WillRepeatedly(
+          [&](std::unique_ptr<RtpPacketToSend> packet,
+              const PacedPacketInfo& cluster_info) { ++packets_sent; });
+  time_controller.AdvanceTime(TimeDelta::Seconds(1));
+  EXPECT_EQ(packets_sent, kPacketsToSend);
+
+  // Expect pacing to make use of high precision.
+  EXPECT_EQ(task_queue_factory.delayed_low_precision_count(), 0);
+  EXPECT_GT(task_queue_factory.delayed_high_precision_count(), 0);
+
+  // Create probe cluster which is also high precision.
+  pacer.CreateProbeClusters(
+      {{.at_time = time_controller.GetClock()->CurrentTime(),
+        .target_data_rate = kPacingRate,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 4,
+        .id = 123}});
+  pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 1));
+  time_controller.AdvanceTime(TimeDelta::Seconds(1));
+  EXPECT_EQ(task_queue_factory.delayed_low_precision_count(), 0);
+  EXPECT_GT(task_queue_factory.delayed_high_precision_count(), 0);
+}
+
+// TODO(webrtc:14502): Rewrite these tests if the functionality is needed if
+// pacing is done on the worker thread.
+TEST(TaskQueuePacedSenderTest, LowPrecisionPacingWhenSlackIsEnabled) {
+  ScopedKeyValueConfig trials("WebRTC-SlackedTaskQueuePacedSender/Enabled/");
+
+  GlobalSimulatedTimeController time_controller(Timestamp::Millis(1234));
+  TaskQueueWithFakePrecisionFactory task_queue_factory(
+      time_controller.GetTaskQueueFactory());
+
+  MockPacketRouter packet_router;
+  TaskQueuePacedSender pacer(
+      time_controller.GetClock(), &packet_router, trials, &task_queue_factory,
+      PacingController::kMinSleepTime, TaskQueuePacedSender::kNoPacketHoldback);
+
+  // Send enough packets (covering one second) that pacing is triggered, i.e.
+  // delayed tasks being scheduled.
+  static constexpr size_t kPacketsToSend = 42;
+  static constexpr DataRate kPacingRate =
+      DataRate::BitsPerSec(kDefaultPacketSize * 8 * kPacketsToSend);
+  pacer.SetPacingRates(kPacingRate, DataRate::Zero());
+  pacer.EnsureStarted();
+  pacer.EnqueuePackets(
+      GeneratePackets(RtpPacketMediaType::kVideo, kPacketsToSend));
+  // Expect all of them to be sent.
+  size_t packets_sent = 0;
+  EXPECT_CALL(packet_router, SendPacket)
+      .WillRepeatedly(
+          [&](std::unique_ptr<RtpPacketToSend> packet,
+              const PacedPacketInfo& cluster_info) { ++packets_sent; });
+  time_controller.AdvanceTime(TimeDelta::Seconds(1));
+  EXPECT_EQ(packets_sent, kPacketsToSend);
+
+  // Expect pacing to make use of low precision.
+  EXPECT_GT(task_queue_factory.delayed_low_precision_count(), 0);
+  EXPECT_EQ(task_queue_factory.delayed_high_precision_count(), 0);
+
+  // Create probe cluster, which uses high precision despite regular pacing
+  // being low precision.
+  pacer.CreateProbeClusters(
+      {{.at_time = time_controller.GetClock()->CurrentTime(),
+        .target_data_rate = kPacingRate,
+        .target_duration = TimeDelta::Millis(15),
+        .target_probe_count = 4,
+        .id = 123}});
+  pacer.EnqueuePackets(GeneratePackets(RtpPacketMediaType::kVideo, 1));
+  time_controller.AdvanceTime(TimeDelta::Seconds(1));
+  EXPECT_GT(task_queue_factory.delayed_high_precision_count(), 0);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/portal/BUILD.gn b/third_party/libwebrtc/modules/portal/BUILD.gn
new file mode 100644
index 0000000000..d7768b2323
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/BUILD.gn
@@ -0,0 +1,148 @@
+# Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//build/config/linux/pkg_config.gni")
+import("//tools/generate_stubs/rules.gni")
+import("../../webrtc.gni")
+
+if ((is_linux || is_chromeos) && rtc_use_pipewire) {
+if (!build_with_mozilla) {
+  pkg_config("gio") {
+    packages = [
+      "gio-2.0",
+      "gio-unix-2.0",
+    ]
+  }
+
+  pkg_config("pipewire") {
+    packages = [ "libpipewire-0.3" ]
+    if (!rtc_link_pipewire) {
+      ignore_libs = true
+    }
+  }
+
+  pkg_config("gbm") {
+    packages = [ "gbm" ]
+  }
+  pkg_config("egl") {
+    packages = [ "egl" ]
+  }
+  pkg_config("epoxy") {
+    packages = [ "epoxy" ]
+    ignore_libs = true
+  }
+  pkg_config("libdrm") {
+    packages = [ "libdrm" ]
+  }
+
+  if (!rtc_link_pipewire) {
+    # When libpipewire is not directly linked, use stubs to allow for dlopening of
+    # the binary.
+    generate_stubs("pipewire_stubs") {
+      configs = [
+        "../../:common_config",
+        ":pipewire",
+      ]
+      deps = [ "../../rtc_base:logging" ]
+      extra_header = "pipewire_stub_header.fragment"
+      logging_function = "RTC_LOG(LS_VERBOSE)"
+      logging_include = "rtc_base/logging.h"
+      output_name = "pipewire_stubs"
+      path_from_source = "modules/portal"
+      sigs = [ "pipewire.sigs" ]
+      if (!build_with_chromium) {
+        macro_include = "rtc_base/system/no_cfi_icall.h"
+        macro_deps = [ "../../rtc_base/system:no_cfi_icall" ]
+      }
+    }
+  }
+
+  config("pipewire_base") {
+    configs = [
+      ":gio",
+      ":pipewire",
+    ]
+  }
+
+  config("pipewire_all") {
+    configs = [
+      ":pipewire_base",
+      ":gbm",
+      ":egl",
+      ":epoxy",
+      ":libdrm",
+    ]
+  }
+
+  config("pipewire_config") {
+    defines = [ "WEBRTC_USE_PIPEWIRE" ]
+
+    # Chromecast build config overrides `WEBRTC_USE_PIPEWIRE` even when
+    # `rtc_use_pipewire` is not set, which causes pipewire_config to not be
+    # included in targets. More details in: webrtc:13898
+    if (is_linux && !is_castos) {
+      defines += [ "WEBRTC_USE_GIO" ]
+    }
+  }
+} else {
+  config("pipewire_all") {
+  }
+  config("pipewire_config") {
+  }
+}
+
+  rtc_library("portal") {
+    sources = [
+      "pipewire_utils.cc",
+      "pipewire_utils.h",
+      "portal_request_response.h",
+      "scoped_glib.cc",
+      "scoped_glib.h",
+      "xdg_desktop_portal_utils.cc",
+      "xdg_desktop_portal_utils.h",
+      "xdg_session_details.h",
+    ]
+
+    configs += [
+      ":gio",
+      ":pipewire",
+      ":pipewire_config",
+    ]
+
+    deps = [
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:sanitizer",
+      "../../rtc_base/system:rtc_export",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+
+    if (!rtc_link_pipewire) {
+      defines = [ "WEBRTC_DLOPEN_PIPEWIRE" ]
+
+      deps += [ ":pipewire_stubs" ]
+    }
+
+    if (build_with_mozilla) {
+      configs -= [
+        ":gio",
+        ":pipewire",
+        ":pipewire_config",
+      ]
+      deps -= [ ":pipewire_stubs" ]
+      defines -= [ "WEBRTC_DLOPEN_PIPEWIRE" ]
+      public_deps = [
+        "//third_party/pipewire",
+        "//third_party/drm",
+        "//third_party/gbm",
+        "//third_party/libepoxy"
+      ]
+    }
+  }
+}
+
diff --git a/third_party/libwebrtc/modules/portal/OWNERS b/third_party/libwebrtc/modules/portal/OWNERS
new file mode 100644
index 0000000000..e3bc32ee5c
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/OWNERS
@@ -0,0 +1,2 @@
+alcooper@chromium.org
+mfoltz@chromium.org
diff --git a/third_party/libwebrtc/modules/portal/pipewire.sigs b/third_party/libwebrtc/modules/portal/pipewire.sigs
new file mode 100644
index 0000000000..139a8c37a5
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/pipewire.sigs
@@ -0,0 +1,52 @@
+// Copyright 2018 The WebRTC project authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+//------------------------------------------------
+// Functions from PipeWire used in capturer code.
+//------------------------------------------------
+
+// core.h
+int pw_core_disconnect(pw_core *core);
+
+// loop.h
+void pw_loop_destroy(pw_loop *loop);
+pw_loop * pw_loop_new(const spa_dict *props);
+
+
+// pipewire.h
+void pw_init(int *argc, char **argv[]);
+const char* pw_get_library_version();
+
+// properties.h
+pw_properties * pw_properties_new_string(const char *args);
+
+// stream.h
+void pw_stream_add_listener(pw_stream *stream, spa_hook *listener, const pw_stream_events *events, void *data);
+int pw_stream_connect(pw_stream *stream, enum pw_direction direction, uint32_t target_id, enum pw_stream_flags flags, const spa_pod **params, uint32_t n_params);
+int pw_stream_disconnect(pw_stream *stream);
+pw_buffer *pw_stream_dequeue_buffer(pw_stream *stream);
+void pw_stream_destroy(pw_stream *stream);
+pw_stream * pw_stream_new(pw_core *core, const char *name, pw_properties *props);
+int pw_stream_queue_buffer(pw_stream *stream, pw_buffer *buffer);
+int pw_stream_set_active(pw_stream *stream, bool active);
+int pw_stream_update_params(pw_stream *stream, const spa_pod **params, uint32_t n_params);
+uint32_t pw_stream_get_node_id(pw_stream *stream);
+pw_stream_state pw_stream_get_state(pw_stream *stream, const char **error);
+
+// thread-loop.h
+void pw_thread_loop_destroy(pw_thread_loop *loop);
+pw_thread_loop * pw_thread_loop_new(const char *name, const spa_dict *props);
+int pw_thread_loop_start(pw_thread_loop *loop);
+void pw_thread_loop_stop(pw_thread_loop *loop);
+void pw_thread_loop_lock(pw_thread_loop *loop);
+void pw_thread_loop_unlock(pw_thread_loop *loop);
+pw_loop * pw_thread_loop_get_loop(pw_thread_loop *loop);
+void pw_thread_loop_signal(pw_thread_loop *loop, bool wait_for_accept);
+void pw_thread_loop_wait(pw_thread_loop *loop);
+
+// context.h
+void pw_context_destroy(pw_context *context);
+pw_context *pw_context_new(pw_loop *main_loop, pw_properties *props, size_t user_data_size);
+pw_core * pw_context_connect(pw_context *context, pw_properties *properties, size_t user_data_size);
+pw_core * pw_context_connect_fd(pw_context *context, int fd, pw_properties *properties, size_t user_data_size);
diff --git a/third_party/libwebrtc/modules/portal/pipewire_stub_header.fragment b/third_party/libwebrtc/modules/portal/pipewire_stub_header.fragment
new file mode 100644
index 0000000000..9d7dbd27c5
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/pipewire_stub_header.fragment
@@ -0,0 +1,8 @@
+// The extra include header needed in the generated stub file for defining
+// various PipeWire types.
+
+extern "C" {
+
+#include <pipewire/pipewire.h>
+
+}
diff --git a/third_party/libwebrtc/modules/portal/pipewire_utils.cc b/third_party/libwebrtc/modules/portal/pipewire_utils.cc
new file mode 100644
index 0000000000..fd96b4a4db
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/pipewire_utils.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/portal/pipewire_utils.h"
+
+#include <pipewire/pipewire.h>
+
+#include "rtc_base/sanitizer.h"
+
+#if defined(WEBRTC_DLOPEN_PIPEWIRE)
+#include "modules/portal/pipewire_stubs.h"
+#endif  // defined(WEBRTC_DLOPEN_PIPEWIRE)
+
+namespace webrtc {
+
+RTC_NO_SANITIZE("cfi-icall")
+bool InitializePipeWire() {
+#if defined(WEBRTC_DLOPEN_PIPEWIRE)
+  static constexpr char kPipeWireLib[] = "libpipewire-0.3.so.0";
+
+  using modules_portal::InitializeStubs;
+  using modules_portal::kModulePipewire;
+
+  modules_portal::StubPathMap paths;
+
+  // Check if the PipeWire library is available.
+  paths[kModulePipewire].push_back(kPipeWireLib);
+
+  static bool result = InitializeStubs(paths);
+
+  return result;
+#else
+  return true;
+#endif  // defined(WEBRTC_DLOPEN_PIPEWIRE)
+}
+
+PipeWireThreadLoopLock::PipeWireThreadLoopLock(pw_thread_loop* loop)
+    : loop_(loop) {
+  pw_thread_loop_lock(loop_);
+}
+
+PipeWireThreadLoopLock::~PipeWireThreadLoopLock() {
+  pw_thread_loop_unlock(loop_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/portal/pipewire_utils.h b/third_party/libwebrtc/modules/portal/pipewire_utils.h
new file mode 100644
index 0000000000..0f5ccf3292
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/pipewire_utils.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PORTAL_PIPEWIRE_UTILS_H_
+#define MODULES_PORTAL_PIPEWIRE_UTILS_H_
+
+struct pw_thread_loop;
+
+namespace webrtc {
+
+// Prepare PipeWire so that it is ready to be used. If it needs to be dlopen'd
+// this will do so. Note that this does not guarantee a PipeWire server is
+// running nor does it establish a connection to one.
+bool InitializePipeWire();
+
+// Locks pw_thread_loop in the current scope
+class PipeWireThreadLoopLock {
+ public:
+  explicit PipeWireThreadLoopLock(pw_thread_loop* loop);
+  ~PipeWireThreadLoopLock();
+
+ private:
+  pw_thread_loop* const loop_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_PORTAL_PIPEWIRE_UTILS_H_
diff --git a/third_party/libwebrtc/modules/portal/portal_gn/moz.build b/third_party/libwebrtc/modules/portal/portal_gn/moz.build
new file mode 100644
index 0000000000..77603c780b
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/portal_gn/moz.build
@@ -0,0 +1,109 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+if CONFIG["MOZ_WIDGET_TOOLKIT"] == "gtk":
+    CXXFLAGS += CONFIG["MOZ_GTK3_CFLAGS"]
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["USE_AURA"] = "1"
+DEFINES["USE_GLIB"] = "1"
+DEFINES["USE_NSS_CERTS"] = "1"
+DEFINES["USE_OZONE"] = "1"
+DEFINES["USE_UDEV"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_LINUX"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_POSIX"] = True
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+DEFINES["WEBRTC_USE_PIPEWIRE"] = True
+DEFINES["_FILE_OFFSET_BITS"] = "64"
+DEFINES["_GNU_SOURCE"] = True
+DEFINES["_LARGEFILE64_SOURCE"] = True
+DEFINES["_LARGEFILE_SOURCE"] = True
+DEFINES["__STDC_CONSTANT_MACROS"] = True
+DEFINES["__STDC_FORMAT_MACROS"] = True
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/gfx/angle/checkout/include/",
+    "/ipc/chromium/src",
+    "/third_party/drm/drm/",
+    "/third_party/drm/drm/include/",
+    "/third_party/drm/drm/include/libdrm/",
+    "/third_party/gbm/gbm/",
+    "/third_party/libepoxy/libepoxy/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/third_party/pipewire/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/portal/pipewire_utils.cc",
+    "/third_party/libwebrtc/modules/portal/scoped_glib.cc",
+    "/third_party/libwebrtc/modules/portal/xdg_desktop_portal_utils.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+Library("portal_gn")
diff --git a/third_party/libwebrtc/modules/portal/portal_request_response.h b/third_party/libwebrtc/modules/portal/portal_request_response.h
new file mode 100644
index 0000000000..5fac4eb137
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/portal_request_response.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PORTAL_PORTAL_REQUEST_RESPONSE_H_
+#define MODULES_PORTAL_PORTAL_REQUEST_RESPONSE_H_
+
+namespace webrtc {
+namespace xdg_portal {
+
+// Contains type of responses that can be observed when making a request to
+// a desktop portal interface.
+enum class RequestResponse {
+  // Unknown, the initialized status.
+  kUnknown,
+  // Success, the request is carried out.
+  kSuccess,
+  // The user cancelled the interaction.
+  kUserCancelled,
+  // The user interaction was ended in some other way.
+  kError,
+
+  kMaxValue = kError,
+};
+
+}  // namespace xdg_portal
+}  // namespace webrtc
+#endif  // MODULES_PORTAL_PORTAL_REQUEST_RESPONSE_H_
diff --git a/third_party/libwebrtc/modules/portal/scoped_glib.cc b/third_party/libwebrtc/modules/portal/scoped_glib.cc
new file mode 100644
index 0000000000..31ba8cb77a
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/scoped_glib.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/portal/scoped_glib.h"
+
+namespace webrtc {
+
+template class RTC_EXPORT_TEMPLATE_DEFINE(RTC_EXPORT) Scoped<GError>;
+template class RTC_EXPORT_TEMPLATE_DEFINE(RTC_EXPORT) Scoped<char>;
+template class RTC_EXPORT_TEMPLATE_DEFINE(RTC_EXPORT) Scoped<GVariant>;
+template class RTC_EXPORT_TEMPLATE_DEFINE(RTC_EXPORT) Scoped<GVariantIter>;
+template class RTC_EXPORT_TEMPLATE_DEFINE(RTC_EXPORT) Scoped<GDBusMessage>;
+template class RTC_EXPORT_TEMPLATE_DEFINE(RTC_EXPORT) Scoped<GUnixFDList>;
+
+template <>
+Scoped<GError>::~Scoped() {
+  if (ptr_) {
+    g_error_free(ptr_);
+  }
+}
+
+template <>
+Scoped<char>::~Scoped() {
+  if (ptr_) {
+    g_free(ptr_);
+  }
+}
+
+template <>
+Scoped<GVariant>::~Scoped() {
+  if (ptr_) {
+    g_variant_unref(ptr_);
+  }
+}
+
+template <>
+Scoped<GVariantIter>::~Scoped() {
+  if (ptr_) {
+    g_variant_iter_free(ptr_);
+  }
+}
+
+template <>
+Scoped<GDBusMessage>::~Scoped() {
+  if (ptr_) {
+    g_object_unref(ptr_);
+  }
+}
+
+template <>
+Scoped<GUnixFDList>::~Scoped() {
+  if (ptr_) {
+    g_object_unref(ptr_);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/portal/scoped_glib.h b/third_party/libwebrtc/modules/portal/scoped_glib.h
new file mode 100644
index 0000000000..d97e47ae1f
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/scoped_glib.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PORTAL_SCOPED_GLIB_H_
+#define MODULES_PORTAL_SCOPED_GLIB_H_
+
+#include <gio/gio.h>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/system/rtc_export_template.h"
+
+namespace webrtc {
+
+template <class T>
+class Scoped {
+ public:
+  Scoped() {}
+  explicit Scoped(T* val) { ptr_ = val; }
+  ~Scoped() { RTC_DCHECK_NOTREACHED(); }
+
+  T* operator->() const { return ptr_; }
+
+  explicit operator bool() const { return ptr_ != nullptr; }
+
+  bool operator!() const { return ptr_ == nullptr; }
+
+  T* get() const { return ptr_; }
+
+  T** receive() {
+    RTC_CHECK(!ptr_);
+    return &ptr_;
+  }
+
+  Scoped& operator=(T* val) {
+    RTC_DCHECK(val);
+    ptr_ = val;
+    return *this;
+  }
+
+ protected:
+  T* ptr_ = nullptr;
+};
+
+template <>
+Scoped<GError>::~Scoped();
+template <>
+Scoped<char>::~Scoped();
+template <>
+Scoped<GVariant>::~Scoped();
+template <>
+Scoped<GVariantIter>::~Scoped();
+template <>
+Scoped<GDBusMessage>::~Scoped();
+template <>
+Scoped<GUnixFDList>::~Scoped();
+
+extern template class RTC_EXPORT_TEMPLATE_DECLARE(RTC_EXPORT) Scoped<GError>;
+extern template class RTC_EXPORT_TEMPLATE_DECLARE(RTC_EXPORT) Scoped<char>;
+extern template class RTC_EXPORT_TEMPLATE_DECLARE(RTC_EXPORT) Scoped<GVariant>;
+extern template class RTC_EXPORT_TEMPLATE_DECLARE(
+    RTC_EXPORT) Scoped<GVariantIter>;
+extern template class RTC_EXPORT_TEMPLATE_DECLARE(
+    RTC_EXPORT) Scoped<GDBusMessage>;
+extern template class RTC_EXPORT_TEMPLATE_DECLARE(
+    RTC_EXPORT) Scoped<GUnixFDList>;
+
+}  // namespace webrtc
+
+#endif  // MODULES_PORTAL_SCOPED_GLIB_H_
diff --git a/third_party/libwebrtc/modules/portal/xdg_desktop_portal_utils.cc b/third_party/libwebrtc/modules/portal/xdg_desktop_portal_utils.cc
new file mode 100644
index 0000000000..271e084463
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/xdg_desktop_portal_utils.cc
@@ -0,0 +1,195 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/portal/xdg_desktop_portal_utils.h"
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/portal/scoped_glib.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace xdg_portal {
+
+std::string RequestResponseToString(RequestResponse request) {
+  switch (request) {
+    case RequestResponse::kUnknown:
+      return "kUnknown";
+    case RequestResponse::kSuccess:
+      return "kSuccess";
+    case RequestResponse::kUserCancelled:
+      return "kUserCancelled";
+    case RequestResponse::kError:
+      return "kError";
+    default:
+      return "Uknown";
+  }
+}
+
+RequestResponse RequestResponseFromPortalResponse(uint32_t portal_response) {
+  // See:
+  //  https://docs.flatpak.org/en/latest/portal-api-reference.html#gdbus-signal-org-freedesktop-portal-Request.Response
+  switch (portal_response) {
+    case 0:
+      return RequestResponse::kSuccess;
+    case 1:
+      return RequestResponse::kUserCancelled;
+    case 2:
+      return RequestResponse::kError;
+    default:
+      return RequestResponse::kUnknown;
+  }
+}
+
+std::string PrepareSignalHandle(absl::string_view token,
+                                GDBusConnection* connection) {
+  Scoped<char> sender(
+      g_strdup(g_dbus_connection_get_unique_name(connection) + 1));
+  for (int i = 0; sender.get()[i]; ++i) {
+    if (sender.get()[i] == '.') {
+      sender.get()[i] = '_';
+    }
+  }
+  const char* handle =
+      g_strconcat(kDesktopRequestObjectPath, "/", sender.get(), "/",
+                  std::string(token).c_str(), /*end of varargs*/ nullptr);
+  return handle;
+}
+
+uint32_t SetupRequestResponseSignal(absl::string_view object_path,
+                                    const GDBusSignalCallback callback,
+                                    gpointer user_data,
+                                    GDBusConnection* connection) {
+  return g_dbus_connection_signal_subscribe(
+      connection, kDesktopBusName, kRequestInterfaceName, "Response",
+      std::string(object_path).c_str(), /*arg0=*/nullptr,
+      G_DBUS_SIGNAL_FLAGS_NO_MATCH_RULE, callback, user_data,
+      /*user_data_free_func=*/nullptr);
+}
+
+void RequestSessionProxy(absl::string_view interface_name,
+                         const ProxyRequestCallback proxy_request_callback,
+                         GCancellable* cancellable,
+                         gpointer user_data) {
+  g_dbus_proxy_new_for_bus(
+      G_BUS_TYPE_SESSION, G_DBUS_PROXY_FLAGS_NONE, /*info=*/nullptr,
+      kDesktopBusName, kDesktopObjectPath, std::string(interface_name).c_str(),
+      cancellable,
+      reinterpret_cast<GAsyncReadyCallback>(proxy_request_callback), user_data);
+}
+
+void SetupSessionRequestHandlers(
+    absl::string_view portal_prefix,
+    const SessionRequestCallback session_request_callback,
+    const SessionRequestResponseSignalHandler request_response_signale_handler,
+    GDBusConnection* connection,
+    GDBusProxy* proxy,
+    GCancellable* cancellable,
+    std::string& portal_handle,
+    guint& session_request_signal_id,
+    gpointer user_data) {
+  GVariantBuilder builder;
+  Scoped<char> variant_string;
+
+  g_variant_builder_init(&builder, G_VARIANT_TYPE_VARDICT);
+  variant_string =
+      g_strdup_printf("%.*s_session%d", static_cast<int>(portal_prefix.size()),
+                      portal_prefix.data(), g_random_int_range(0, G_MAXINT));
+  g_variant_builder_add(&builder, "{sv}", "session_handle_token",
+                        g_variant_new_string(variant_string.get()));
+
+  variant_string =
+      g_strdup_printf("%.*s_%d", static_cast<int>(portal_prefix.size()),
+                      portal_prefix.data(), g_random_int_range(0, G_MAXINT));
+  g_variant_builder_add(&builder, "{sv}", "handle_token",
+                        g_variant_new_string(variant_string.get()));
+
+  portal_handle = PrepareSignalHandle(variant_string.get(), connection);
+  session_request_signal_id = SetupRequestResponseSignal(
+      portal_handle.c_str(), request_response_signale_handler, user_data,
+      connection);
+
+  RTC_LOG(LS_INFO) << "Desktop session requested.";
+  g_dbus_proxy_call(
+      proxy, "CreateSession", g_variant_new("(a{sv})", &builder),
+      G_DBUS_CALL_FLAGS_NONE, /*timeout=*/-1, cancellable,
+      reinterpret_cast<GAsyncReadyCallback>(session_request_callback),
+      user_data);
+}
+
+void StartSessionRequest(
+    absl::string_view prefix,
+    absl::string_view session_handle,
+    const StartRequestResponseSignalHandler signal_handler,
+    const SessionStartRequestedHandler session_started_handler,
+    GDBusProxy* proxy,
+    GDBusConnection* connection,
+    GCancellable* cancellable,
+    guint& start_request_signal_id,
+    std::string& start_handle,
+    gpointer user_data) {
+  GVariantBuilder builder;
+  Scoped<char> variant_string;
+
+  g_variant_builder_init(&builder, G_VARIANT_TYPE_VARDICT);
+  variant_string =
+      g_strdup_printf("%.*s%d", static_cast<int>(prefix.size()), prefix.data(),
+                      g_random_int_range(0, G_MAXINT));
+  g_variant_builder_add(&builder, "{sv}", "handle_token",
+                        g_variant_new_string(variant_string.get()));
+
+  start_handle = PrepareSignalHandle(variant_string.get(), connection);
+  start_request_signal_id = SetupRequestResponseSignal(
+      start_handle.c_str(), signal_handler, user_data, connection);
+
+  // "Identifier for the application window", this is Wayland, so not "x11:...".
+  const char parent_window[] = "";
+
+  RTC_LOG(LS_INFO) << "Starting the portal session.";
+  g_dbus_proxy_call(
+      proxy, "Start",
+      g_variant_new("(osa{sv})", std::string(session_handle).c_str(),
+                    parent_window, &builder),
+      G_DBUS_CALL_FLAGS_NONE, /*timeout=*/-1, cancellable,
+      reinterpret_cast<GAsyncReadyCallback>(session_started_handler),
+      user_data);
+}
+
+void TearDownSession(absl::string_view session_handle,
+                     GDBusProxy* proxy,
+                     GCancellable* cancellable,
+                     GDBusConnection* connection) {
+  if (!session_handle.empty()) {
+    Scoped<GDBusMessage> message(g_dbus_message_new_method_call(
+        kDesktopBusName, std::string(session_handle).c_str(),
+        kSessionInterfaceName, "Close"));
+    if (message.get()) {
+      Scoped<GError> error;
+      g_dbus_connection_send_message(connection, message.get(),
+                                     G_DBUS_SEND_MESSAGE_FLAGS_NONE,
+                                     /*out_serial=*/nullptr, error.receive());
+      if (error.get()) {
+        RTC_LOG(LS_ERROR) << "Failed to close the session: " << error->message;
+      }
+    }
+  }
+
+  if (cancellable) {
+    g_cancellable_cancel(cancellable);
+    g_object_unref(cancellable);
+  }
+
+  if (proxy) {
+    g_object_unref(proxy);
+  }
+}
+
+}  // namespace xdg_portal
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/portal/xdg_desktop_portal_utils.h b/third_party/libwebrtc/modules/portal/xdg_desktop_portal_utils.h
new file mode 100644
index 0000000000..eff46a01ab
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/xdg_desktop_portal_utils.h
@@ -0,0 +1,113 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PORTAL_XDG_DESKTOP_PORTAL_UTILS_H_
+#define MODULES_PORTAL_XDG_DESKTOP_PORTAL_UTILS_H_
+
+#include <gio/gio.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "modules/portal/portal_request_response.h"
+#include "modules/portal/scoped_glib.h"
+#include "modules/portal/xdg_session_details.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace xdg_portal {
+
+constexpr char kDesktopBusName[] = "org.freedesktop.portal.Desktop";
+constexpr char kDesktopObjectPath[] = "/org/freedesktop/portal/desktop";
+constexpr char kDesktopRequestObjectPath[] =
+    "/org/freedesktop/portal/desktop/request";
+constexpr char kSessionInterfaceName[] = "org.freedesktop.portal.Session";
+constexpr char kRequestInterfaceName[] = "org.freedesktop.portal.Request";
+constexpr char kScreenCastInterfaceName[] = "org.freedesktop.portal.ScreenCast";
+
+using ProxyRequestCallback = void (*)(GObject*, GAsyncResult*, gpointer);
+using SessionRequestCallback = void (*)(GDBusProxy*, GAsyncResult*, gpointer);
+using SessionRequestResponseSignalHandler = void (*)(GDBusConnection*,
+                                                     const char*,
+                                                     const char*,
+                                                     const char*,
+                                                     const char*,
+                                                     GVariant*,
+                                                     gpointer);
+using StartRequestResponseSignalHandler = void (*)(GDBusConnection*,
+                                                   const char*,
+                                                   const char*,
+                                                   const char*,
+                                                   const char*,
+                                                   GVariant*,
+                                                   gpointer);
+using SessionStartRequestedHandler = void (*)(GDBusProxy*,
+                                              GAsyncResult*,
+                                              gpointer);
+
+RTC_EXPORT std::string RequestResponseToString(RequestResponse request);
+
+RequestResponse RequestResponseFromPortalResponse(uint32_t portal_response);
+
+// Returns a string path for signal handle based on the provided connection and
+// token.
+RTC_EXPORT std::string PrepareSignalHandle(absl::string_view token,
+                                           GDBusConnection* connection);
+
+// Sets up the callback to execute when a response signal is received for the
+// given object.
+RTC_EXPORT uint32_t
+SetupRequestResponseSignal(absl::string_view object_path,
+                           const GDBusSignalCallback callback,
+                           gpointer user_data,
+                           GDBusConnection* connection);
+
+RTC_EXPORT void RequestSessionProxy(
+    absl::string_view interface_name,
+    const ProxyRequestCallback proxy_request_callback,
+    GCancellable* cancellable,
+    gpointer user_data);
+
+RTC_EXPORT void SetupSessionRequestHandlers(
+    absl::string_view portal_prefix,
+    const SessionRequestCallback session_request_callback,
+    const SessionRequestResponseSignalHandler request_response_signale_handler,
+    GDBusConnection* connection,
+    GDBusProxy* proxy,
+    GCancellable* cancellable,
+    std::string& portal_handle,
+    guint& session_request_signal_id,
+    gpointer user_data);
+
+RTC_EXPORT void StartSessionRequest(
+    absl::string_view prefix,
+    absl::string_view session_handle,
+    const StartRequestResponseSignalHandler signal_handler,
+    const SessionStartRequestedHandler session_started_handler,
+    GDBusProxy* proxy,
+    GDBusConnection* connection,
+    GCancellable* cancellable,
+    guint& start_request_signal_id,
+    std::string& start_handle,
+    gpointer user_data);
+
+// Tears down the portal session and cleans up related objects.
+RTC_EXPORT void TearDownSession(absl::string_view session_handle,
+                                GDBusProxy* proxy,
+                                GCancellable* cancellable,
+                                GDBusConnection* connection);
+
+}  // namespace xdg_portal
+}  // namespace webrtc
+
+#endif  // MODULES_PORTAL_XDG_DESKTOP_PORTAL_UTILS_H_
diff --git a/third_party/libwebrtc/modules/portal/xdg_session_details.h b/third_party/libwebrtc/modules/portal/xdg_session_details.h
new file mode 100644
index 0000000000..ab52508c2f
--- /dev/null
+++ b/third_party/libwebrtc/modules/portal/xdg_session_details.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_PORTAL_XDG_SESSION_DETAILS_H_
+#define MODULES_PORTAL_XDG_SESSION_DETAILS_H_
+
+#include <gio/gio.h>
+
+#include <string>
+
+namespace webrtc {
+namespace xdg_portal {
+
+// Details of the session associated with XDG desktop portal session. Portal API
+// calls can be invoked by utilizing the information here.
+struct SessionDetails {
+  GDBusProxy* proxy = nullptr;
+  GCancellable* cancellable = nullptr;
+  std::string session_handle;
+  uint32_t pipewire_stream_node_id = 0;
+};
+
+}  // namespace xdg_portal
+}  // namespace webrtc
+
+#endif  // MODULES_PORTAL_XDG_SESSION_DETAILS_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/BUILD.gn b/third_party/libwebrtc/modules/remote_bitrate_estimator/BUILD.gn
new file mode 100644
index 0000000000..463530b973
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/BUILD.gn
@@ -0,0 +1,143 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("remote_bitrate_estimator") {
+  sources = [
+    "aimd_rate_control.cc",
+    "aimd_rate_control.h",
+    "bwe_defines.cc",
+    "include/bwe_defines.h",
+    "include/remote_bitrate_estimator.h",
+    "inter_arrival.cc",
+    "inter_arrival.h",
+    "overuse_detector.cc",
+    "overuse_detector.h",
+    "overuse_estimator.cc",
+    "overuse_estimator.h",
+    "packet_arrival_map.cc",
+    "packet_arrival_map.h",
+    "remote_bitrate_estimator_abs_send_time.cc",
+    "remote_bitrate_estimator_abs_send_time.h",
+    "remote_bitrate_estimator_single_stream.cc",
+    "remote_bitrate_estimator_single_stream.h",
+    "remote_estimator_proxy.cc",
+    "remote_estimator_proxy.h",
+    "test/bwe_test_logging.h",
+  ]
+
+  if (rtc_enable_bwe_test_logging) {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
+
+    sources += [ "test/bwe_test_logging.cc" ]
+  } else {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
+  }
+
+  deps = [
+    "../../api:field_trials_view",
+    "../../api:network_state_predictor_api",
+    "../../api:rtp_headers",
+    "../../api/transport:field_trial_based_config",
+    "../../api/transport:network_control",
+    "../../api/units:data_rate",
+    "../../api/units:data_size",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../modules:module_api",
+    "../../modules:module_api_public",
+    "../../modules/congestion_controller/goog_cc:link_capacity_estimator",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:platform_thread",
+    "../../rtc_base:race_checker",
+    "../../rtc_base:rate_statistics",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base:stringutils",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
+    "../../system_wrappers:field_trial",
+    "../../system_wrappers:metrics",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (!build_with_chromium) {
+  rtc_library("bwe_rtp") {
+    testonly = true
+    sources = [
+      "tools/bwe_rtp.cc",
+      "tools/bwe_rtp.h",
+    ]
+    deps = [
+      "../../test:rtp_test_utils",
+      "../rtp_rtcp:rtp_rtcp_format",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+    ]
+  }
+
+  rtc_executable("rtp_to_text") {
+    testonly = true
+    sources = [ "tools/rtp_to_text.cc" ]
+    deps = [
+      ":bwe_rtp",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:stringutils",
+      "../../test:rtp_test_utils",
+      "../rtp_rtcp:rtp_rtcp_format",
+    ]
+  }
+}
+
+if (rtc_include_tests) {
+  rtc_library("remote_bitrate_estimator_unittests") {
+    testonly = true
+
+    sources = [
+      "aimd_rate_control_unittest.cc",
+      "inter_arrival_unittest.cc",
+      "overuse_detector_unittest.cc",
+      "packet_arrival_map_test.cc",
+      "remote_bitrate_estimator_abs_send_time_unittest.cc",
+      "remote_bitrate_estimator_single_stream_unittest.cc",
+      "remote_bitrate_estimator_unittest_helper.cc",
+      "remote_bitrate_estimator_unittest_helper.h",
+      "remote_estimator_proxy_unittest.cc",
+    ]
+    deps = [
+      ":remote_bitrate_estimator",
+      "..:module_api_public",
+      "../../api/transport:field_trial_based_config",
+      "../../api/transport:mock_network_control",
+      "../../api/transport:network_control",
+      "../../api/units:data_rate",
+      "../../api/units:data_size",
+      "../../api/units:time_delta",
+      "../../api/units:timestamp",
+      "../../rtc_base:checks",
+      "../../rtc_base:random",
+      "../../system_wrappers",
+      "../../test:field_trial",
+      "../../test:fileutils",
+      "../../test:test_support",
+      "../pacing",
+      "../rtp_rtcp:rtp_rtcp_format",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/DEPS b/third_party/libwebrtc/modules/remote_bitrate_estimator/DEPS
new file mode 100644
index 0000000000..35a62119e5
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/DEPS
@@ -0,0 +1,6 @@
+include_rules = [
+  "+logging/rtc_event_log",
+  "+system_wrappers",
+  # Avoid directly using field_trial. Instead use FieldTrialsView.
+  "-system_wrappers/include/field_trial.h",
+]
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/OWNERS b/third_party/libwebrtc/modules/remote_bitrate_estimator/OWNERS
new file mode 100644
index 0000000000..993e2fd77b
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/OWNERS
@@ -0,0 +1,5 @@
+danilchap@webrtc.org
+stefan@webrtc.org
+terelius@webrtc.org
+mflodman@webrtc.org
+perkj@webrtc.org
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
new file mode 100644
index 0000000000..6c3638b59f
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
@@ -0,0 +1,431 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+
+#include <inttypes.h>
+
+#include <algorithm>
+#include <cmath>
+#include <cstdio>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/overuse_detector.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+
+constexpr TimeDelta kDefaultRtt = TimeDelta::Millis(200);
+constexpr double kDefaultBackoffFactor = 0.85;
+
+constexpr char kBweBackOffFactorExperiment[] = "WebRTC-BweBackOffFactor";
+
+bool IsEnabled(const FieldTrialsView& field_trials, absl::string_view key) {
+  return absl::StartsWith(field_trials.Lookup(key), "Enabled");
+}
+
+double ReadBackoffFactor(const FieldTrialsView& key_value_config) {
+  std::string experiment_string =
+      key_value_config.Lookup(kBweBackOffFactorExperiment);
+  double backoff_factor;
+  int parsed_values =
+      sscanf(experiment_string.c_str(), "Enabled-%lf", &backoff_factor);
+  if (parsed_values == 1) {
+    if (backoff_factor >= 1.0) {
+      RTC_LOG(LS_WARNING) << "Back-off factor must be less than 1.";
+    } else if (backoff_factor <= 0.0) {
+      RTC_LOG(LS_WARNING) << "Back-off factor must be greater than 0.";
+    } else {
+      return backoff_factor;
+    }
+  }
+  RTC_LOG(LS_WARNING) << "Failed to parse parameters for AimdRateControl "
+                         "experiment from field trial string. Using default.";
+  return kDefaultBackoffFactor;
+}
+
+}  // namespace
+
+AimdRateControl::AimdRateControl(const FieldTrialsView* key_value_config)
+    : AimdRateControl(key_value_config, /* send_side =*/false) {}
+
+AimdRateControl::AimdRateControl(const FieldTrialsView* key_value_config,
+                                 bool send_side)
+    : min_configured_bitrate_(kCongestionControllerMinBitrate),
+      max_configured_bitrate_(DataRate::KilobitsPerSec(30000)),
+      current_bitrate_(max_configured_bitrate_),
+      latest_estimated_throughput_(current_bitrate_),
+      link_capacity_(),
+      rate_control_state_(RateControlState::kRcHold),
+      time_last_bitrate_change_(Timestamp::MinusInfinity()),
+      time_last_bitrate_decrease_(Timestamp::MinusInfinity()),
+      time_first_throughput_estimate_(Timestamp::MinusInfinity()),
+      bitrate_is_initialized_(false),
+      beta_(IsEnabled(*key_value_config, kBweBackOffFactorExperiment)
+                ? ReadBackoffFactor(*key_value_config)
+                : kDefaultBackoffFactor),
+      in_alr_(false),
+      rtt_(kDefaultRtt),
+      send_side_(send_side),
+      no_bitrate_increase_in_alr_(
+          IsEnabled(*key_value_config,
+                    "WebRTC-DontIncreaseDelayBasedBweInAlr")),
+      initial_backoff_interval_("initial_backoff_interval"),
+      link_capacity_fix_("link_capacity_fix") {
+  ParseFieldTrial(
+      {&disable_estimate_bounded_increase_, &estimate_bounded_increase_ratio_,
+       &ignore_throughput_limit_if_network_estimate_,
+       &ignore_network_estimate_decrease_, &increase_to_network_estimate_},
+      key_value_config->Lookup("WebRTC-Bwe-EstimateBoundedIncrease"));
+  // E.g
+  // WebRTC-BweAimdRateControlConfig/initial_backoff_interval:100ms/
+  ParseFieldTrial({&initial_backoff_interval_, &link_capacity_fix_},
+                  key_value_config->Lookup("WebRTC-BweAimdRateControlConfig"));
+  if (initial_backoff_interval_) {
+    RTC_LOG(LS_INFO) << "Using aimd rate control with initial back-off interval"
+                        " "
+                     << ToString(*initial_backoff_interval_) << ".";
+  }
+  RTC_LOG(LS_INFO) << "Using aimd rate control with back off factor " << beta_;
+}
+
+AimdRateControl::~AimdRateControl() {}
+
+void AimdRateControl::SetStartBitrate(DataRate start_bitrate) {
+  current_bitrate_ = start_bitrate;
+  latest_estimated_throughput_ = current_bitrate_;
+  bitrate_is_initialized_ = true;
+}
+
+void AimdRateControl::SetMinBitrate(DataRate min_bitrate) {
+  min_configured_bitrate_ = min_bitrate;
+  current_bitrate_ = std::max(min_bitrate, current_bitrate_);
+}
+
+bool AimdRateControl::ValidEstimate() const {
+  return bitrate_is_initialized_;
+}
+
+TimeDelta AimdRateControl::GetFeedbackInterval() const {
+  // Estimate how often we can send RTCP if we allocate up to 5% of bandwidth
+  // to feedback.
+  const DataSize kRtcpSize = DataSize::Bytes(80);
+  const DataRate rtcp_bitrate = current_bitrate_ * 0.05;
+  const TimeDelta interval = kRtcpSize / rtcp_bitrate;
+  const TimeDelta kMinFeedbackInterval = TimeDelta::Millis(200);
+  const TimeDelta kMaxFeedbackInterval = TimeDelta::Millis(1000);
+  return interval.Clamped(kMinFeedbackInterval, kMaxFeedbackInterval);
+}
+
+bool AimdRateControl::TimeToReduceFurther(Timestamp at_time,
+                                          DataRate estimated_throughput) const {
+  const TimeDelta bitrate_reduction_interval =
+      rtt_.Clamped(TimeDelta::Millis(10), TimeDelta::Millis(200));
+  if (at_time - time_last_bitrate_change_ >= bitrate_reduction_interval) {
+    return true;
+  }
+  if (ValidEstimate()) {
+    // TODO(terelius/holmer): Investigate consequences of increasing
+    // the threshold to 0.95 * LatestEstimate().
+    const DataRate threshold = 0.5 * LatestEstimate();
+    return estimated_throughput < threshold;
+  }
+  return false;
+}
+
+bool AimdRateControl::InitialTimeToReduceFurther(Timestamp at_time) const {
+  if (!initial_backoff_interval_) {
+    return ValidEstimate() &&
+           TimeToReduceFurther(at_time,
+                               LatestEstimate() / 2 - DataRate::BitsPerSec(1));
+  }
+  // TODO(terelius): We could use the RTT (clamped to suitable limits) instead
+  // of a fixed bitrate_reduction_interval.
+  if (time_last_bitrate_decrease_.IsInfinite() ||
+      at_time - time_last_bitrate_decrease_ >= *initial_backoff_interval_) {
+    return true;
+  }
+  return false;
+}
+
+DataRate AimdRateControl::LatestEstimate() const {
+  return current_bitrate_;
+}
+
+void AimdRateControl::SetRtt(TimeDelta rtt) {
+  rtt_ = rtt;
+}
+
+DataRate AimdRateControl::Update(const RateControlInput* input,
+                                 Timestamp at_time) {
+  RTC_CHECK(input);
+
+  // Set the initial bit rate value to what we're receiving the first half
+  // second.
+  // TODO(bugs.webrtc.org/9379): The comment above doesn't match to the code.
+  if (!bitrate_is_initialized_) {
+    const TimeDelta kInitializationTime = TimeDelta::Seconds(5);
+    RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTime.ms());
+    if (time_first_throughput_estimate_.IsInfinite()) {
+      if (input->estimated_throughput)
+        time_first_throughput_estimate_ = at_time;
+    } else if (at_time - time_first_throughput_estimate_ >
+                   kInitializationTime &&
+               input->estimated_throughput) {
+      current_bitrate_ = *input->estimated_throughput;
+      bitrate_is_initialized_ = true;
+    }
+  }
+
+  ChangeBitrate(*input, at_time);
+  return current_bitrate_;
+}
+
+void AimdRateControl::SetInApplicationLimitedRegion(bool in_alr) {
+  in_alr_ = in_alr;
+}
+
+void AimdRateControl::SetEstimate(DataRate bitrate, Timestamp at_time) {
+  bitrate_is_initialized_ = true;
+  DataRate prev_bitrate = current_bitrate_;
+  current_bitrate_ = ClampBitrate(bitrate);
+  time_last_bitrate_change_ = at_time;
+  if (current_bitrate_ < prev_bitrate) {
+    time_last_bitrate_decrease_ = at_time;
+  }
+}
+
+void AimdRateControl::SetNetworkStateEstimate(
+    const absl::optional<NetworkStateEstimate>& estimate) {
+  network_estimate_ = estimate;
+}
+
+double AimdRateControl::GetNearMaxIncreaseRateBpsPerSecond() const {
+  RTC_DCHECK(!current_bitrate_.IsZero());
+  const TimeDelta kFrameInterval = TimeDelta::Seconds(1) / 30;
+  DataSize frame_size = current_bitrate_ * kFrameInterval;
+  const DataSize kPacketSize = DataSize::Bytes(1200);
+  double packets_per_frame = std::ceil(frame_size / kPacketSize);
+  DataSize avg_packet_size = frame_size / packets_per_frame;
+
+  // Approximate the over-use estimator delay to 100 ms.
+  TimeDelta response_time = rtt_ + TimeDelta::Millis(100);
+
+    response_time = response_time * 2;
+  double increase_rate_bps_per_second =
+      (avg_packet_size / response_time).bps<double>();
+  double kMinIncreaseRateBpsPerSecond = 4000;
+  return std::max(kMinIncreaseRateBpsPerSecond, increase_rate_bps_per_second);
+}
+
+TimeDelta AimdRateControl::GetExpectedBandwidthPeriod() const {
+  const TimeDelta kMinPeriod = TimeDelta::Seconds(2);
+  const TimeDelta kDefaultPeriod = TimeDelta::Seconds(3);
+  const TimeDelta kMaxPeriod = TimeDelta::Seconds(50);
+
+  double increase_rate_bps_per_second = GetNearMaxIncreaseRateBpsPerSecond();
+  if (!last_decrease_)
+    return kDefaultPeriod;
+  double time_to_recover_decrease_seconds =
+      last_decrease_->bps() / increase_rate_bps_per_second;
+  TimeDelta period = TimeDelta::Seconds(time_to_recover_decrease_seconds);
+  return period.Clamped(kMinPeriod, kMaxPeriod);
+}
+
+void AimdRateControl::ChangeBitrate(const RateControlInput& input,
+                                    Timestamp at_time) {
+  absl::optional<DataRate> new_bitrate;
+  DataRate estimated_throughput =
+      input.estimated_throughput.value_or(latest_estimated_throughput_);
+  if (input.estimated_throughput)
+    latest_estimated_throughput_ = *input.estimated_throughput;
+
+  // An over-use should always trigger us to reduce the bitrate, even though
+  // we have not yet established our first estimate. By acting on the over-use,
+  // we will end up with a valid estimate.
+  if (!bitrate_is_initialized_ &&
+      input.bw_state != BandwidthUsage::kBwOverusing)
+    return;
+
+  ChangeState(input, at_time);
+
+  switch (rate_control_state_) {
+    case RateControlState::kRcHold:
+      break;
+
+    case RateControlState::kRcIncrease: {
+      if (estimated_throughput > link_capacity_.UpperBound())
+        link_capacity_.Reset();
+
+      // We limit the new bitrate based on the troughput to avoid unlimited
+      // bitrate increases. We allow a bit more lag at very low rates to not too
+      // easily get stuck if the encoder produces uneven outputs.
+      DataRate increase_limit =
+          1.5 * estimated_throughput + DataRate::KilobitsPerSec(10);
+      if (ignore_throughput_limit_if_network_estimate_ && network_estimate_ &&
+          network_estimate_->link_capacity_upper.IsFinite()) {
+        // If we have a Network estimate, we do allow the estimate to increase.
+        increase_limit = network_estimate_->link_capacity_upper *
+                         estimate_bounded_increase_ratio_.Get();
+      } else if (send_side_ && in_alr_ && no_bitrate_increase_in_alr_) {
+        // Do not increase the delay based estimate in alr since the estimator
+        // will not be able to get transport feedback necessary to detect if
+        // the new estimate is correct.
+        // If we have previously increased above the limit (for instance due to
+        // probing), we don't allow further changes.
+        increase_limit = current_bitrate_;
+      }
+
+      if (current_bitrate_ < increase_limit) {
+        DataRate increased_bitrate = DataRate::MinusInfinity();
+        if (increase_to_network_estimate_ && network_estimate_ &&
+            network_estimate_->link_capacity_upper.IsFinite()) {
+          increased_bitrate = increase_limit;
+        } else if (link_capacity_.has_estimate()) {
+          // The link_capacity estimate is reset if the measured throughput
+          // is too far from the estimate. We can therefore assume that our
+          // target rate is reasonably close to link capacity and use additive
+          // increase.
+          DataRate additive_increase =
+              AdditiveRateIncrease(at_time, time_last_bitrate_change_);
+          increased_bitrate = current_bitrate_ + additive_increase;
+        } else {
+          // If we don't have an estimate of the link capacity, use faster ramp
+          // up to discover the capacity.
+          DataRate multiplicative_increase = MultiplicativeRateIncrease(
+              at_time, time_last_bitrate_change_, current_bitrate_);
+          increased_bitrate = current_bitrate_ + multiplicative_increase;
+        }
+        new_bitrate = std::min(increased_bitrate, increase_limit);
+      }
+      time_last_bitrate_change_ = at_time;
+      break;
+    }
+
+    case RateControlState::kRcDecrease: {
+      DataRate decreased_bitrate = DataRate::PlusInfinity();
+
+      // Set bit rate to something slightly lower than the measured throughput
+      // to get rid of any self-induced delay.
+      decreased_bitrate = estimated_throughput * beta_;
+      if (decreased_bitrate > current_bitrate_ && !link_capacity_fix_) {
+        // TODO(terelius): The link_capacity estimate may be based on old
+        // throughput measurements. Relying on them may lead to unnecessary
+        // BWE drops.
+        if (link_capacity_.has_estimate()) {
+          decreased_bitrate = beta_ * link_capacity_.estimate();
+        }
+      }
+      // Avoid increasing the rate when over-using.
+      if (decreased_bitrate < current_bitrate_) {
+        new_bitrate = decreased_bitrate;
+      }
+
+      if (bitrate_is_initialized_ && estimated_throughput < current_bitrate_) {
+        if (!new_bitrate.has_value()) {
+          last_decrease_ = DataRate::Zero();
+        } else {
+          last_decrease_ = current_bitrate_ - *new_bitrate;
+        }
+      }
+      if (estimated_throughput < link_capacity_.LowerBound()) {
+        // The current throughput is far from the estimated link capacity. Clear
+        // the estimate to allow an immediate update in OnOveruseDetected.
+        link_capacity_.Reset();
+      }
+
+      bitrate_is_initialized_ = true;
+      link_capacity_.OnOveruseDetected(estimated_throughput);
+      // Stay on hold until the pipes are cleared.
+      rate_control_state_ = RateControlState::kRcHold;
+      time_last_bitrate_change_ = at_time;
+      time_last_bitrate_decrease_ = at_time;
+      break;
+    }
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+
+  current_bitrate_ = ClampBitrate(new_bitrate.value_or(current_bitrate_));
+}
+
+DataRate AimdRateControl::ClampBitrate(DataRate new_bitrate) const {
+  if (!disable_estimate_bounded_increase_ && network_estimate_ &&
+      network_estimate_->link_capacity_upper.IsFinite()) {
+    DataRate upper_bound = network_estimate_->link_capacity_upper *
+                           estimate_bounded_increase_ratio_.Get();
+    if (ignore_network_estimate_decrease_) {
+      upper_bound = std::max(upper_bound, current_bitrate_);
+    }
+    new_bitrate = std::min(upper_bound, new_bitrate);
+  }
+  if (network_estimate_ && network_estimate_->link_capacity_lower.IsFinite() &&
+      new_bitrate < current_bitrate_) {
+    new_bitrate = std::min(
+        current_bitrate_,
+        std::max(new_bitrate, network_estimate_->link_capacity_lower * beta_));
+  }
+  new_bitrate = std::max(new_bitrate, min_configured_bitrate_);
+  return new_bitrate;
+}
+
+DataRate AimdRateControl::MultiplicativeRateIncrease(
+    Timestamp at_time,
+    Timestamp last_time,
+    DataRate current_bitrate) const {
+  double alpha = 1.08;
+  if (last_time.IsFinite()) {
+    auto time_since_last_update = at_time - last_time;
+    alpha = pow(alpha, std::min(time_since_last_update.seconds<double>(), 1.0));
+  }
+  DataRate multiplicative_increase =
+      std::max(current_bitrate * (alpha - 1.0), DataRate::BitsPerSec(1000));
+  return multiplicative_increase;
+}
+
+DataRate AimdRateControl::AdditiveRateIncrease(Timestamp at_time,
+                                               Timestamp last_time) const {
+  double time_period_seconds = (at_time - last_time).seconds<double>();
+  double data_rate_increase_bps =
+      GetNearMaxIncreaseRateBpsPerSecond() * time_period_seconds;
+  return DataRate::BitsPerSec(data_rate_increase_bps);
+}
+
+void AimdRateControl::ChangeState(const RateControlInput& input,
+                                  Timestamp at_time) {
+  switch (input.bw_state) {
+    case BandwidthUsage::kBwNormal:
+      if (rate_control_state_ == RateControlState::kRcHold) {
+        time_last_bitrate_change_ = at_time;
+        rate_control_state_ = RateControlState::kRcIncrease;
+      }
+      break;
+    case BandwidthUsage::kBwOverusing:
+      if (rate_control_state_ != RateControlState::kRcDecrease) {
+        rate_control_state_ = RateControlState::kRcDecrease;
+      }
+      break;
+    case BandwidthUsage::kBwUnderusing:
+      rate_control_state_ = RateControlState::kRcHold;
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
new file mode 100644
index 0000000000..8321fd5239
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/link_capacity_estimator.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+// A rate control implementation based on additive increases of
+// bitrate when no over-use is detected and multiplicative decreases when
+// over-uses are detected. When we think the available bandwidth has changes or
+// is unknown, we will switch to a "slow-start mode" where we increase
+// multiplicatively.
+class AimdRateControl {
+ public:
+  explicit AimdRateControl(const FieldTrialsView* key_value_config);
+  AimdRateControl(const FieldTrialsView* key_value_config, bool send_side);
+  ~AimdRateControl();
+
+  // Returns true if the target bitrate has been initialized. This happens
+  // either if it has been explicitly set via SetStartBitrate/SetEstimate, or if
+  // we have measured a throughput.
+  bool ValidEstimate() const;
+  void SetStartBitrate(DataRate start_bitrate);
+  void SetMinBitrate(DataRate min_bitrate);
+  TimeDelta GetFeedbackInterval() const;
+
+  // Returns true if the bitrate estimate hasn't been changed for more than
+  // an RTT, or if the estimated_throughput is less than half of the current
+  // estimate. Should be used to decide if we should reduce the rate further
+  // when over-using.
+  bool TimeToReduceFurther(Timestamp at_time,
+                           DataRate estimated_throughput) const;
+  // As above. To be used if overusing before we have measured a throughput.
+  bool InitialTimeToReduceFurther(Timestamp at_time) const;
+
+  DataRate LatestEstimate() const;
+  void SetRtt(TimeDelta rtt);
+  DataRate Update(const RateControlInput* input, Timestamp at_time);
+  void SetInApplicationLimitedRegion(bool in_alr);
+  void SetEstimate(DataRate bitrate, Timestamp at_time);
+  void SetNetworkStateEstimate(
+      const absl::optional<NetworkStateEstimate>& estimate);
+
+  // Returns the increase rate when used bandwidth is near the link capacity.
+  double GetNearMaxIncreaseRateBpsPerSecond() const;
+  // Returns the expected time between overuse signals (assuming steady state).
+  TimeDelta GetExpectedBandwidthPeriod() const;
+
+ private:
+  enum class RateControlState { kRcHold, kRcIncrease, kRcDecrease };
+
+  friend class GoogCcStatePrinter;
+  // Update the target bitrate based on, among other things, the current rate
+  // control state, the current target bitrate and the estimated throughput.
+  // When in the "increase" state the bitrate will be increased either
+  // additively or multiplicatively depending on the rate control region. When
+  // in the "decrease" state the bitrate will be decreased to slightly below the
+  // current throughput. When in the "hold" state the bitrate will be kept
+  // constant to allow built up queues to drain.
+  void ChangeBitrate(const RateControlInput& input, Timestamp at_time);
+
+  DataRate ClampBitrate(DataRate new_bitrate) const;
+  DataRate MultiplicativeRateIncrease(Timestamp at_time,
+                                      Timestamp last_ms,
+                                      DataRate current_bitrate) const;
+  DataRate AdditiveRateIncrease(Timestamp at_time, Timestamp last_time) const;
+  void UpdateChangePeriod(Timestamp at_time);
+  void ChangeState(const RateControlInput& input, Timestamp at_time);
+
+  DataRate min_configured_bitrate_;
+  DataRate max_configured_bitrate_;
+  DataRate current_bitrate_;
+  DataRate latest_estimated_throughput_;
+  LinkCapacityEstimator link_capacity_;
+  absl::optional<NetworkStateEstimate> network_estimate_;
+  RateControlState rate_control_state_;
+  Timestamp time_last_bitrate_change_;
+  Timestamp time_last_bitrate_decrease_;
+  Timestamp time_first_throughput_estimate_;
+  bool bitrate_is_initialized_;
+  double beta_;
+  bool in_alr_;
+  TimeDelta rtt_;
+  const bool send_side_;
+  // Allow the delay based estimate to only increase as long as application
+  // limited region (alr) is not detected.
+  const bool no_bitrate_increase_in_alr_;
+  // If false, uses estimated link capacity upper bound *
+  // `estimate_bounded_increase_ratio_` as upper limit for the estimate.
+  FieldTrialFlag disable_estimate_bounded_increase_{"Disabled"};
+  FieldTrialParameter<double> estimate_bounded_increase_ratio_{"ratio", 1.0};
+  FieldTrialParameter<bool> ignore_throughput_limit_if_network_estimate_{
+      "ignore_acked", false};
+  FieldTrialParameter<bool> increase_to_network_estimate_{"immediate_incr",
+                                                          false};
+  FieldTrialParameter<bool> ignore_network_estimate_decrease_{"ignore_decr",
+                                                              false};
+  absl::optional<DataRate> last_decrease_;
+  FieldTrialOptional<TimeDelta> initial_backoff_interval_;
+  FieldTrialFlag link_capacity_fix_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
new file mode 100644
index 0000000000..aa80dae55b
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
@@ -0,0 +1,484 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+
+#include <memory>
+
+#include "api/transport/field_trial_based_config.h"
+#include "api/units/data_rate.h"
+#include "system_wrappers/include/clock.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int64_t kClockInitialTime = 123456;
+
+constexpr int kMinBwePeriodMs = 2000;
+constexpr int kDefaultPeriodMs = 3000;
+constexpr int kMaxBwePeriodMs = 50000;
+
+// After an overuse, we back off to 85% to the received bitrate.
+constexpr double kFractionAfterOveruse = 0.85;
+
+struct AimdRateControlStates {
+  std::unique_ptr<AimdRateControl> aimd_rate_control;
+  std::unique_ptr<SimulatedClock> simulated_clock;
+  FieldTrialBasedConfig field_trials;
+};
+
+AimdRateControlStates CreateAimdRateControlStates(bool send_side = false) {
+  AimdRateControlStates states;
+  states.aimd_rate_control.reset(
+      new AimdRateControl(&states.field_trials, send_side));
+  states.simulated_clock.reset(new SimulatedClock(kClockInitialTime));
+  return states;
+}
+absl::optional<DataRate> OptionalRateFromOptionalBps(
+    absl::optional<int> bitrate_bps) {
+  if (bitrate_bps) {
+    return DataRate::BitsPerSec(*bitrate_bps);
+  } else {
+    return absl::nullopt;
+  }
+}
+void UpdateRateControl(const AimdRateControlStates& states,
+                       const BandwidthUsage& bandwidth_usage,
+                       absl::optional<uint32_t> throughput_estimate,
+                       int64_t now_ms) {
+  RateControlInput input(bandwidth_usage,
+                         OptionalRateFromOptionalBps(throughput_estimate));
+  states.aimd_rate_control->Update(&input, Timestamp::Millis(now_ms));
+}
+void SetEstimate(const AimdRateControlStates& states, int bitrate_bps) {
+  states.aimd_rate_control->SetEstimate(DataRate::BitsPerSec(bitrate_bps),
+                                        states.simulated_clock->CurrentTime());
+}
+
+}  // namespace
+
+TEST(AimdRateControlTest, MinNearMaxIncreaseRateOnLowBandwith) {
+  auto states = CreateAimdRateControlStates();
+  constexpr int kBitrate = 30000;
+  SetEstimate(states, kBitrate);
+  EXPECT_EQ(4000,
+            states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond());
+}
+
+TEST(AimdRateControlTest, NearMaxIncreaseRateIs5kbpsOn90kbpsAnd200msRtt) {
+  auto states = CreateAimdRateControlStates();
+  constexpr int kBitrate = 90000;
+  SetEstimate(states, kBitrate);
+  EXPECT_EQ(5000,
+            states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond());
+}
+
+TEST(AimdRateControlTest, NearMaxIncreaseRateIs5kbpsOn60kbpsAnd100msRtt) {
+  auto states = CreateAimdRateControlStates();
+  constexpr int kBitrate = 60000;
+  SetEstimate(states, kBitrate);
+  states.aimd_rate_control->SetRtt(TimeDelta::Millis(100));
+  EXPECT_EQ(5000,
+            states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond());
+}
+
+TEST(AimdRateControlTest, GetIncreaseRateAndBandwidthPeriod) {
+  // Smoothing experiment disabled
+  auto states = CreateAimdRateControlStates();
+  constexpr int kBitrate = 300000;
+  SetEstimate(states, kBitrate);
+  UpdateRateControl(states, BandwidthUsage::kBwOverusing, kBitrate,
+                    states.simulated_clock->TimeInMilliseconds());
+  EXPECT_NEAR(14000,
+              states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond(),
+              1000);
+  EXPECT_EQ(kDefaultPeriodMs,
+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
+}
+
+TEST(AimdRateControlTest, BweLimitedByAckedBitrate) {
+  auto states = CreateAimdRateControlStates();
+  constexpr int kAckedBitrate = 10000;
+  SetEstimate(states, kAckedBitrate);
+  while (states.simulated_clock->TimeInMilliseconds() - kClockInitialTime <
+         20000) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, kAckedBitrate,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+  }
+  ASSERT_TRUE(states.aimd_rate_control->ValidEstimate());
+  EXPECT_EQ(static_cast<uint32_t>(1.5 * kAckedBitrate + 10000),
+            states.aimd_rate_control->LatestEstimate().bps());
+}
+
+TEST(AimdRateControlTest, BweNotLimitedByDecreasingAckedBitrate) {
+  auto states = CreateAimdRateControlStates();
+  constexpr int kAckedBitrate = 100000;
+  SetEstimate(states, kAckedBitrate);
+  while (states.simulated_clock->TimeInMilliseconds() - kClockInitialTime <
+         20000) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, kAckedBitrate,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+  }
+  ASSERT_TRUE(states.aimd_rate_control->ValidEstimate());
+  // If the acked bitrate decreases the BWE shouldn't be reduced to 1.5x
+  // what's being acked, but also shouldn't get to increase more.
+  uint32_t prev_estimate = states.aimd_rate_control->LatestEstimate().bps();
+  UpdateRateControl(states, BandwidthUsage::kBwNormal, kAckedBitrate / 2,
+                    states.simulated_clock->TimeInMilliseconds());
+  uint32_t new_estimate = states.aimd_rate_control->LatestEstimate().bps();
+  EXPECT_NEAR(new_estimate, static_cast<uint32_t>(1.5 * kAckedBitrate + 10000),
+              2000);
+  EXPECT_EQ(new_estimate, prev_estimate);
+}
+
+TEST(AimdRateControlTest, DefaultPeriodUntilFirstOveruse) {
+  // Smoothing experiment disabled
+  auto states = CreateAimdRateControlStates();
+  states.aimd_rate_control->SetStartBitrate(DataRate::KilobitsPerSec(300));
+  EXPECT_EQ(kDefaultPeriodMs,
+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
+  states.simulated_clock->AdvanceTimeMilliseconds(100);
+  UpdateRateControl(states, BandwidthUsage::kBwOverusing, 280000,
+                    states.simulated_clock->TimeInMilliseconds());
+  EXPECT_NE(kDefaultPeriodMs,
+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
+}
+
+TEST(AimdRateControlTest, ExpectedPeriodAfter20kbpsDropAnd5kbpsIncrease) {
+  auto states = CreateAimdRateControlStates();
+  constexpr int kInitialBitrate = 110000;
+  SetEstimate(states, kInitialBitrate);
+  states.simulated_clock->AdvanceTimeMilliseconds(100);
+  // Make the bitrate drop by 20 kbps to get to 90 kbps.
+  // The rate increase at 90 kbps should be 5 kbps, so the period should be 4 s.
+  constexpr int kAckedBitrate =
+      (kInitialBitrate - 20000) / kFractionAfterOveruse;
+  UpdateRateControl(states, BandwidthUsage::kBwOverusing, kAckedBitrate,
+                    states.simulated_clock->TimeInMilliseconds());
+  EXPECT_EQ(5000,
+            states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond());
+  EXPECT_EQ(4000, states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
+}
+
+TEST(AimdRateControlTest, BandwidthPeriodIsNotBelowMin) {
+  auto states = CreateAimdRateControlStates();
+  constexpr int kInitialBitrate = 10000;
+  SetEstimate(states, kInitialBitrate);
+  states.simulated_clock->AdvanceTimeMilliseconds(100);
+  // Make a small (1.5 kbps) bitrate drop to 8.5 kbps.
+  UpdateRateControl(states, BandwidthUsage::kBwOverusing, kInitialBitrate - 1,
+                    states.simulated_clock->TimeInMilliseconds());
+  EXPECT_EQ(kMinBwePeriodMs,
+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
+}
+
+TEST(AimdRateControlTest, BandwidthPeriodIsNotAboveMaxNoSmoothingExp) {
+  auto states = CreateAimdRateControlStates();
+  constexpr int kInitialBitrate = 10010000;
+  SetEstimate(states, kInitialBitrate);
+  states.simulated_clock->AdvanceTimeMilliseconds(100);
+  // Make a large (10 Mbps) bitrate drop to 10 kbps.
+  constexpr int kAckedBitrate = 10000 / kFractionAfterOveruse;
+  UpdateRateControl(states, BandwidthUsage::kBwOverusing, kAckedBitrate,
+                    states.simulated_clock->TimeInMilliseconds());
+  EXPECT_EQ(kMaxBwePeriodMs,
+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
+}
+
+TEST(AimdRateControlTest, SendingRateBoundedWhenThroughputNotEstimated) {
+  auto states = CreateAimdRateControlStates();
+  constexpr int kInitialBitrateBps = 123000;
+  UpdateRateControl(states, BandwidthUsage::kBwNormal, kInitialBitrateBps,
+                    states.simulated_clock->TimeInMilliseconds());
+  // AimdRateControl sets the initial bit rate to what it receives after
+  // five seconds has passed.
+  // TODO(bugs.webrtc.org/9379): The comment in the AimdRateControl does not
+  // match the constant.
+  constexpr int kInitializationTimeMs = 5000;
+  states.simulated_clock->AdvanceTimeMilliseconds(kInitializationTimeMs + 1);
+  UpdateRateControl(states, BandwidthUsage::kBwNormal, kInitialBitrateBps,
+                    states.simulated_clock->TimeInMilliseconds());
+  for (int i = 0; i < 100; ++i) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+  }
+  EXPECT_LE(states.aimd_rate_control->LatestEstimate().bps(),
+            kInitialBitrateBps * 1.5 + 10000);
+}
+
+TEST(AimdRateControlTest, EstimateDoesNotIncreaseInAlr) {
+  // When alr is detected, the delay based estimator is not allowed to increase
+  // bwe since there will be no feedback from the network if the new estimate
+  // is correct.
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-DontIncreaseDelayBasedBweInAlr/Enabled/");
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  constexpr int kInitialBitrateBps = 123000;
+  SetEstimate(states, kInitialBitrateBps);
+  states.aimd_rate_control->SetInApplicationLimitedRegion(true);
+  UpdateRateControl(states, BandwidthUsage::kBwNormal, kInitialBitrateBps,
+                    states.simulated_clock->TimeInMilliseconds());
+  ASSERT_EQ(states.aimd_rate_control->LatestEstimate().bps(),
+            kInitialBitrateBps);
+
+  for (int i = 0; i < 100; ++i) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+  }
+  EXPECT_EQ(states.aimd_rate_control->LatestEstimate().bps(),
+            kInitialBitrateBps);
+}
+
+TEST(AimdRateControlTest, SetEstimateIncreaseBweInAlr) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-DontIncreaseDelayBasedBweInAlr/Enabled/");
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  constexpr int kInitialBitrateBps = 123000;
+  SetEstimate(states, kInitialBitrateBps);
+  states.aimd_rate_control->SetInApplicationLimitedRegion(true);
+  ASSERT_EQ(states.aimd_rate_control->LatestEstimate().bps(),
+            kInitialBitrateBps);
+  SetEstimate(states, 2 * kInitialBitrateBps);
+  EXPECT_EQ(states.aimd_rate_control->LatestEstimate().bps(),
+            2 * kInitialBitrateBps);
+}
+
+TEST(AimdRateControlTest, SetEstimateUpperLimitedByNetworkEstimate) {
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  NetworkStateEstimate network_estimate;
+  network_estimate.link_capacity_upper = DataRate::KilobitsPerSec(400);
+  states.aimd_rate_control->SetNetworkStateEstimate(network_estimate);
+  SetEstimate(states, 500'000);
+  EXPECT_EQ(states.aimd_rate_control->LatestEstimate(),
+            network_estimate.link_capacity_upper);
+}
+
+TEST(AimdRateControlTest, SetEstimateLowerLimitedByNetworkEstimate) {
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  NetworkStateEstimate network_estimate;
+  network_estimate.link_capacity_lower = DataRate::KilobitsPerSec(400);
+  states.aimd_rate_control->SetNetworkStateEstimate(network_estimate);
+  SetEstimate(states, 100'000);
+  // 0.85 is default backoff factor. (`beta_`)
+  EXPECT_EQ(states.aimd_rate_control->LatestEstimate(),
+            network_estimate.link_capacity_lower * 0.85);
+}
+
+TEST(AimdRateControlTest,
+     SetEstimateIgnoredIfLowerThanNetworkEstimateAndCurrent) {
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  SetEstimate(states, 200'000);
+  ASSERT_EQ(states.aimd_rate_control->LatestEstimate().kbps(), 200);
+  NetworkStateEstimate network_estimate;
+  network_estimate.link_capacity_lower = DataRate::KilobitsPerSec(400);
+  states.aimd_rate_control->SetNetworkStateEstimate(network_estimate);
+  // Ignore the next SetEstimate, since the estimate is lower than 85% of
+  // the network estimate.
+  SetEstimate(states, 100'000);
+  EXPECT_EQ(states.aimd_rate_control->LatestEstimate().kbps(), 200);
+}
+
+TEST(AimdRateControlTest, SetEstimateIgnoresNetworkEstimatesLowerThanCurrent) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Bwe-EstimateBoundedIncrease/"
+      "ratio:0.85,ignore_acked:true,ignore_decr:true/");
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  states.aimd_rate_control->SetStartBitrate(DataRate::KilobitsPerSec(30));
+  NetworkStateEstimate network_estimate;
+  network_estimate.link_capacity_upper = DataRate::KilobitsPerSec(400);
+  states.aimd_rate_control->SetNetworkStateEstimate(network_estimate);
+  SetEstimate(states, 500'000);
+  ASSERT_EQ(states.aimd_rate_control->LatestEstimate(),
+            network_estimate.link_capacity_upper * 0.85);
+
+  NetworkStateEstimate lower_network_estimate;
+  lower_network_estimate.link_capacity_upper = DataRate::KilobitsPerSec(300);
+  states.aimd_rate_control->SetNetworkStateEstimate(lower_network_estimate);
+  SetEstimate(states, 500'000);
+  EXPECT_EQ(states.aimd_rate_control->LatestEstimate(),
+            network_estimate.link_capacity_upper * 0.85);
+}
+
+TEST(AimdRateControlTest, EstimateIncreaseWhileNotInAlr) {
+  // Allow the estimate to increase as long as alr is not detected to ensure
+  // tha BWE can not get stuck at a certain bitrate.
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-DontIncreaseDelayBasedBweInAlr/Enabled/");
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  constexpr int kInitialBitrateBps = 123000;
+  SetEstimate(states, kInitialBitrateBps);
+  states.aimd_rate_control->SetInApplicationLimitedRegion(false);
+  UpdateRateControl(states, BandwidthUsage::kBwNormal, kInitialBitrateBps,
+                    states.simulated_clock->TimeInMilliseconds());
+  for (int i = 0; i < 100; ++i) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+  }
+  EXPECT_GT(states.aimd_rate_control->LatestEstimate().bps(),
+            kInitialBitrateBps);
+}
+
+TEST(AimdRateControlTest, EstimateNotLimitedByNetworkEstimateIfDisabled) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Bwe-EstimateBoundedIncrease/Disabled/");
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  constexpr int kInitialBitrateBps = 123000;
+  SetEstimate(states, kInitialBitrateBps);
+  states.aimd_rate_control->SetInApplicationLimitedRegion(false);
+  NetworkStateEstimate network_estimate;
+  network_estimate.link_capacity_upper = DataRate::KilobitsPerSec(150);
+  states.aimd_rate_control->SetNetworkStateEstimate(network_estimate);
+
+  for (int i = 0; i < 100; ++i) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+  }
+  EXPECT_GT(states.aimd_rate_control->LatestEstimate(),
+            network_estimate.link_capacity_upper);
+}
+
+TEST(AimdRateControlTest,
+     EstimateSlowlyIncreaseToUpperLinkCapacityEstimateIfConfigured) {
+  // Even if alr is detected, the delay based estimator is allowed to increase
+  // up to a percentage of upper link capacity.
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Bwe-EstimateBoundedIncrease/"
+      "ratio:0.85,ignore_acked:true,immediate_incr:false/"
+      "WebRTC-DontIncreaseDelayBasedBweInAlr/Enabled/");
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  constexpr int kInitialBitrateBps = 123000;
+  SetEstimate(states, kInitialBitrateBps);
+  states.aimd_rate_control->SetInApplicationLimitedRegion(true);
+
+  NetworkStateEstimate network_estimate;
+  network_estimate.link_capacity_upper = DataRate::KilobitsPerSec(200);
+  states.aimd_rate_control->SetNetworkStateEstimate(network_estimate);
+  for (int i = 0; i < 10; ++i) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+    EXPECT_LT(states.aimd_rate_control->LatestEstimate(),
+              network_estimate.link_capacity_upper * 0.85);
+  }
+  for (int i = 0; i < 50; ++i) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+  }
+  EXPECT_EQ(states.aimd_rate_control->LatestEstimate(),
+            network_estimate.link_capacity_upper * 0.85);
+}
+
+TEST(AimdRateControlTest,
+     EstimateImmediatelyIncreaseToUpperLinkCapacityEstimateIfConfigured) {
+  // Even if alr is detected, the delay based estimator is allowed to increase
+  // up to a percentage of upper link capacity.
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Bwe-EstimateBoundedIncrease/"
+      "ratio:0.85,ignore_acked:true,immediate_incr:true/"
+      "WebRTC-DontIncreaseDelayBasedBweInAlr/Enabled/");
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  constexpr int kInitialBitrateBps = 123000;
+  SetEstimate(states, kInitialBitrateBps);
+  states.aimd_rate_control->SetInApplicationLimitedRegion(true);
+
+  NetworkStateEstimate network_estimate;
+  network_estimate.link_capacity_upper = DataRate::KilobitsPerSec(200);
+  states.aimd_rate_control->SetNetworkStateEstimate(network_estimate);
+  UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
+                    states.simulated_clock->TimeInMilliseconds());
+  EXPECT_EQ(states.aimd_rate_control->LatestEstimate(),
+            network_estimate.link_capacity_upper * 0.85);
+}
+
+TEST(AimdRateControlTest, EstimateNotLoweredByNetworkEstimate) {
+  // The delay based estimator is allowed to increase up to a percentage of
+  // upper link capacity but does not decrease unless the delay detector
+  // discover an overuse.
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Bwe-EstimateBoundedIncrease/"
+      "ratio:0.85,ignore_acked:true,ignore_decr:true/"
+      "WebRTC-DontIncreaseDelayBasedBweInAlr/Enabled/");
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  constexpr int kInitialBitrateBps = 123000;
+  constexpr int kEstimatedThroughputBps = 30'000;
+  SetEstimate(states, kInitialBitrateBps);
+
+  NetworkStateEstimate network_estimate;
+  network_estimate.link_capacity_upper = DataRate::KilobitsPerSec(200);
+  states.aimd_rate_control->SetNetworkStateEstimate(network_estimate);
+  for (int i = 0; i < 100; ++i) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal,
+                      kEstimatedThroughputBps,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+  }
+  DataRate estimate_after_increase = states.aimd_rate_control->LatestEstimate();
+  ASSERT_EQ(estimate_after_increase,
+            network_estimate.link_capacity_upper * 0.85);
+
+  // A lower network estimate does not decrease the estimate immediately,
+  // but the estimate is not allowed to increase.
+  network_estimate.link_capacity_upper = DataRate::KilobitsPerSec(100);
+  network_estimate.link_capacity_lower = DataRate::KilobitsPerSec(80);
+  states.aimd_rate_control->SetNetworkStateEstimate(network_estimate);
+  for (int i = 0; i < 10; ++i) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal,
+                      kEstimatedThroughputBps,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+    EXPECT_EQ(states.aimd_rate_control->LatestEstimate(),
+              estimate_after_increase);
+  }
+
+  // If the detector detects and overuse, BWE drops to a value relative the
+  // network estimate.
+  UpdateRateControl(states, BandwidthUsage::kBwOverusing,
+                    kEstimatedThroughputBps,
+                    states.simulated_clock->TimeInMilliseconds());
+  EXPECT_LT(states.aimd_rate_control->LatestEstimate(),
+            network_estimate.link_capacity_lower);
+  EXPECT_GT(states.aimd_rate_control->LatestEstimate().bps(),
+            kEstimatedThroughputBps);
+}
+
+TEST(AimdRateControlTest, EstimateDoesNotIncreaseInAlrIfNetworkEstimateNotSet) {
+  // When alr is detected, the delay based estimator is not allowed to increase
+  // bwe since there will be no feedback from the network if the new estimate
+  // is correct.
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Bwe-EstimateBoundedIncrease/ratio:0.85,ignore_acked:true/"
+      "WebRTC-DontIncreaseDelayBasedBweInAlr/Enabled/");
+  auto states = CreateAimdRateControlStates(/*send_side=*/true);
+  constexpr int kInitialBitrateBps = 123000;
+  SetEstimate(states, kInitialBitrateBps);
+  states.aimd_rate_control->SetInApplicationLimitedRegion(true);
+  UpdateRateControl(states, BandwidthUsage::kBwNormal, kInitialBitrateBps,
+                    states.simulated_clock->TimeInMilliseconds());
+  ASSERT_EQ(states.aimd_rate_control->LatestEstimate().bps(),
+            kInitialBitrateBps);
+
+  for (int i = 0; i < 100; ++i) {
+    UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
+                      states.simulated_clock->TimeInMilliseconds());
+    states.simulated_clock->AdvanceTimeMilliseconds(100);
+  }
+  EXPECT_EQ(states.aimd_rate_control->LatestEstimate().bps(),
+            kInitialBitrateBps);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/bwe_defines.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/bwe_defines.cc
new file mode 100644
index 0000000000..db92f46717
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/bwe_defines.cc
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+
+namespace webrtc {
+
+const char kBweTypeHistogram[] = "WebRTC.BWE.Types";
+
+RateControlInput::RateControlInput(
+    BandwidthUsage bw_state,
+    const absl::optional<DataRate>& estimated_throughput)
+    : bw_state(bw_state), estimated_throughput(estimated_throughput) {}
+
+RateControlInput::~RateControlInput() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/include/bwe_defines.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
new file mode 100644
index 0000000000..d3dd96be75
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/network_state_predictor.h"
+#include "api/units/data_rate.h"
+
+namespace webrtc {
+
+constexpr DataRate kCongestionControllerMinBitrate = DataRate::BitsPerSec(5000);
+
+static const int64_t kBitrateWindowMs = 1000;
+
+extern const char kBweTypeHistogram[];
+
+enum BweNames {
+  kReceiverNoExtension = 0,
+  kReceiverTOffset = 1,
+  kReceiverAbsSendTime = 2,
+  kSendSideTransportSeqNum = 3,
+  kBweNamesMax = 4
+};
+
+struct RateControlInput {
+  RateControlInput(BandwidthUsage bw_state,
+                   const absl::optional<DataRate>& estimated_throughput);
+  ~RateControlInput();
+
+  BandwidthUsage bw_state;
+  absl::optional<DataRate> estimated_throughput;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
new file mode 100644
index 0000000000..0d4e15e9e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This class estimates the incoming available bandwidth.
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "modules/include/module_common_types.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+
+class Clock;
+
+// RemoteBitrateObserver is used to signal changes in bitrate estimates for
+// the incoming streams.
+class RemoteBitrateObserver {
+ public:
+  // Called when a receive channel group has a new bitrate estimate for the
+  // incoming streams.
+  virtual void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                                       uint32_t bitrate) = 0;
+
+  virtual ~RemoteBitrateObserver() {}
+};
+
+class RemoteBitrateEstimator : public CallStatsObserver {
+ public:
+  ~RemoteBitrateEstimator() override {}
+
+  // Called for each incoming packet. Updates the incoming payload bitrate
+  // estimate and the over-use detector. If an over-use is detected the
+  // remote bitrate estimate will be updated. Note that `payload_size` is the
+  // packet size excluding headers.
+  // Note that `arrival_time_ms` can be of an arbitrary time base.
+  virtual void IncomingPacket(int64_t arrival_time_ms,
+                              size_t payload_size,
+                              const RTPHeader& header) = 0;
+
+  // Removes all data for `ssrc`.
+  virtual void RemoveStream(uint32_t ssrc) = 0;
+
+  // Returns latest estimate or DataRate::Zero() if estimation is unavailable.
+  virtual DataRate LatestEstimate() const = 0;
+
+  virtual TimeDelta Process() = 0;
+
+ protected:
+  static const int64_t kProcessIntervalMs = 500;
+  static const int64_t kStreamTimeOutMs = 2000;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival.cc
new file mode 100644
index 0000000000..41c092bf91
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival.cc
@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/inter_arrival.h"
+
+#include "modules/include/module_common_types_public.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+static const int kBurstDeltaThresholdMs = 5;
+static const int kMaxBurstDurationMs = 100;
+
+InterArrival::InterArrival(uint32_t timestamp_group_length_ticks,
+                           double timestamp_to_ms_coeff,
+                           bool enable_burst_grouping)
+    : kTimestampGroupLengthTicks(timestamp_group_length_ticks),
+      current_timestamp_group_(),
+      prev_timestamp_group_(),
+      timestamp_to_ms_coeff_(timestamp_to_ms_coeff),
+      burst_grouping_(enable_burst_grouping),
+      num_consecutive_reordered_packets_(0) {}
+
+bool InterArrival::ComputeDeltas(uint32_t timestamp,
+                                 int64_t arrival_time_ms,
+                                 int64_t system_time_ms,
+                                 size_t packet_size,
+                                 uint32_t* timestamp_delta,
+                                 int64_t* arrival_time_delta_ms,
+                                 int* packet_size_delta) {
+  RTC_DCHECK(timestamp_delta);
+  RTC_DCHECK(arrival_time_delta_ms);
+  RTC_DCHECK(packet_size_delta);
+  bool calculated_deltas = false;
+  if (current_timestamp_group_.IsFirstPacket()) {
+    // We don't have enough data to update the filter, so we store it until we
+    // have two frames of data to process.
+    current_timestamp_group_.timestamp = timestamp;
+    current_timestamp_group_.first_timestamp = timestamp;
+    current_timestamp_group_.first_arrival_ms = arrival_time_ms;
+  } else if (!PacketInOrder(timestamp)) {
+    return false;
+  } else if (NewTimestampGroup(arrival_time_ms, timestamp)) {
+    // First packet of a later frame, the previous frame sample is ready.
+    if (prev_timestamp_group_.complete_time_ms >= 0) {
+      *timestamp_delta =
+          current_timestamp_group_.timestamp - prev_timestamp_group_.timestamp;
+      *arrival_time_delta_ms = current_timestamp_group_.complete_time_ms -
+                               prev_timestamp_group_.complete_time_ms;
+      // Check system time differences to see if we have an unproportional jump
+      // in arrival time. In that case reset the inter-arrival computations.
+      int64_t system_time_delta_ms =
+          current_timestamp_group_.last_system_time_ms -
+          prev_timestamp_group_.last_system_time_ms;
+      if (*arrival_time_delta_ms - system_time_delta_ms >=
+          kArrivalTimeOffsetThresholdMs) {
+        RTC_LOG(LS_WARNING)
+            << "The arrival time clock offset has changed (diff = "
+            << *arrival_time_delta_ms - system_time_delta_ms
+            << " ms), resetting.";
+        Reset();
+        return false;
+      }
+      if (*arrival_time_delta_ms < 0) {
+        // The group of packets has been reordered since receiving its local
+        // arrival timestamp.
+        ++num_consecutive_reordered_packets_;
+        if (num_consecutive_reordered_packets_ >= kReorderedResetThreshold) {
+          RTC_LOG(LS_WARNING)
+              << "Packets are being reordered on the path from the "
+                 "socket to the bandwidth estimator. Ignoring this "
+                 "packet for bandwidth estimation, resetting.";
+          Reset();
+        }
+        return false;
+      } else {
+        num_consecutive_reordered_packets_ = 0;
+      }
+      RTC_DCHECK_GE(*arrival_time_delta_ms, 0);
+      *packet_size_delta = static_cast<int>(current_timestamp_group_.size) -
+                           static_cast<int>(prev_timestamp_group_.size);
+      calculated_deltas = true;
+    }
+    prev_timestamp_group_ = current_timestamp_group_;
+    // The new timestamp is now the current frame.
+    current_timestamp_group_.first_timestamp = timestamp;
+    current_timestamp_group_.timestamp = timestamp;
+    current_timestamp_group_.first_arrival_ms = arrival_time_ms;
+    current_timestamp_group_.size = 0;
+  } else {
+    current_timestamp_group_.timestamp =
+        LatestTimestamp(current_timestamp_group_.timestamp, timestamp);
+  }
+  // Accumulate the frame size.
+  current_timestamp_group_.size += packet_size;
+  current_timestamp_group_.complete_time_ms = arrival_time_ms;
+  current_timestamp_group_.last_system_time_ms = system_time_ms;
+
+  return calculated_deltas;
+}
+
+bool InterArrival::PacketInOrder(uint32_t timestamp) {
+  if (current_timestamp_group_.IsFirstPacket()) {
+    return true;
+  } else {
+    // Assume that a diff which is bigger than half the timestamp interval
+    // (32 bits) must be due to reordering. This code is almost identical to
+    // that in IsNewerTimestamp() in module_common_types.h.
+    uint32_t timestamp_diff =
+        timestamp - current_timestamp_group_.first_timestamp;
+    return timestamp_diff < 0x80000000;
+  }
+}
+
+// Assumes that `timestamp` is not reordered compared to
+// `current_timestamp_group_`.
+bool InterArrival::NewTimestampGroup(int64_t arrival_time_ms,
+                                     uint32_t timestamp) const {
+  if (current_timestamp_group_.IsFirstPacket()) {
+    return false;
+  } else if (BelongsToBurst(arrival_time_ms, timestamp)) {
+    return false;
+  } else {
+    uint32_t timestamp_diff =
+        timestamp - current_timestamp_group_.first_timestamp;
+    return timestamp_diff > kTimestampGroupLengthTicks;
+  }
+}
+
+bool InterArrival::BelongsToBurst(int64_t arrival_time_ms,
+                                  uint32_t timestamp) const {
+  if (!burst_grouping_) {
+    return false;
+  }
+  RTC_DCHECK_GE(current_timestamp_group_.complete_time_ms, 0);
+  int64_t arrival_time_delta_ms =
+      arrival_time_ms - current_timestamp_group_.complete_time_ms;
+  uint32_t timestamp_diff = timestamp - current_timestamp_group_.timestamp;
+  int64_t ts_delta_ms = timestamp_to_ms_coeff_ * timestamp_diff + 0.5;
+  if (ts_delta_ms == 0)
+    return true;
+  int propagation_delta_ms = arrival_time_delta_ms - ts_delta_ms;
+  if (propagation_delta_ms < 0 &&
+      arrival_time_delta_ms <= kBurstDeltaThresholdMs &&
+      arrival_time_ms - current_timestamp_group_.first_arrival_ms <
+          kMaxBurstDurationMs)
+    return true;
+  return false;
+}
+
+void InterArrival::Reset() {
+  num_consecutive_reordered_packets_ = 0;
+  current_timestamp_group_ = TimestampGroup();
+  prev_timestamp_group_ = TimestampGroup();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival.h
new file mode 100644
index 0000000000..d31a8b63c3
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_INTER_ARRIVAL_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_INTER_ARRIVAL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// Helper class to compute the inter-arrival time delta and the size delta
+// between two timestamp groups. A timestamp is a 32 bit unsigned number with
+// a client defined rate.
+class InterArrival {
+ public:
+  // After this many packet groups received out of order InterArrival will
+  // reset, assuming that clocks have made a jump.
+  static constexpr int kReorderedResetThreshold = 3;
+  static constexpr int64_t kArrivalTimeOffsetThresholdMs = 3000;
+
+  // A timestamp group is defined as all packets with a timestamp which are at
+  // most timestamp_group_length_ticks older than the first timestamp in that
+  // group.
+  InterArrival(uint32_t timestamp_group_length_ticks,
+               double timestamp_to_ms_coeff,
+               bool enable_burst_grouping);
+
+  InterArrival() = delete;
+  InterArrival(const InterArrival&) = delete;
+  InterArrival& operator=(const InterArrival&) = delete;
+
+  // This function returns true if a delta was computed, or false if the current
+  // group is still incomplete or if only one group has been completed.
+  // `timestamp` is the timestamp.
+  // `arrival_time_ms` is the local time at which the packet arrived.
+  // `packet_size` is the size of the packet.
+  // `timestamp_delta` (output) is the computed timestamp delta.
+  // `arrival_time_delta_ms` (output) is the computed arrival-time delta.
+  // `packet_size_delta` (output) is the computed size delta.
+  bool ComputeDeltas(uint32_t timestamp,
+                     int64_t arrival_time_ms,
+                     int64_t system_time_ms,
+                     size_t packet_size,
+                     uint32_t* timestamp_delta,
+                     int64_t* arrival_time_delta_ms,
+                     int* packet_size_delta);
+
+ private:
+  struct TimestampGroup {
+    TimestampGroup()
+        : size(0),
+          first_timestamp(0),
+          timestamp(0),
+          first_arrival_ms(-1),
+          complete_time_ms(-1) {}
+
+    bool IsFirstPacket() const { return complete_time_ms == -1; }
+
+    size_t size;
+    uint32_t first_timestamp;
+    uint32_t timestamp;
+    int64_t first_arrival_ms;
+    int64_t complete_time_ms;
+    int64_t last_system_time_ms;
+  };
+
+  // Returns true if the packet with timestamp `timestamp` arrived in order.
+  bool PacketInOrder(uint32_t timestamp);
+
+  // Returns true if the last packet was the end of the current batch and the
+  // packet with `timestamp` is the first of a new batch.
+  bool NewTimestampGroup(int64_t arrival_time_ms, uint32_t timestamp) const;
+
+  bool BelongsToBurst(int64_t arrival_time_ms, uint32_t timestamp) const;
+
+  void Reset();
+
+  const uint32_t kTimestampGroupLengthTicks;
+  TimestampGroup current_timestamp_group_;
+  TimestampGroup prev_timestamp_group_;
+  double timestamp_to_ms_coeff_;
+  bool burst_grouping_;
+  int num_consecutive_reordered_packets_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_INTER_ARRIVAL_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival_unittest.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival_unittest.cc
new file mode 100644
index 0000000000..72a772ed21
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival_unittest.cc
@@ -0,0 +1,531 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/inter_arrival.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace testing {
+
+enum {
+  kTimestampGroupLengthUs = 5000,
+  kMinStep = 20,
+  kTriggerNewGroupUs = kTimestampGroupLengthUs + kMinStep,
+  kBurstThresholdMs = 5,
+  kAbsSendTimeFraction = 18,
+  kAbsSendTimeInterArrivalUpshift = 8,
+  kInterArrivalShift = kAbsSendTimeFraction + kAbsSendTimeInterArrivalUpshift,
+};
+
+const double kRtpTimestampToMs = 1.0 / 90.0;
+const double kAstToMs = 1000.0 / static_cast<double>(1 << kInterArrivalShift);
+
+class InterArrivalTest : public ::testing::Test {
+ protected:
+  virtual void SetUp() {
+    inter_arrival_.reset(
+        new InterArrival(kTimestampGroupLengthUs / 1000, 1.0, true));
+    inter_arrival_rtp_.reset(new InterArrival(
+        MakeRtpTimestamp(kTimestampGroupLengthUs), kRtpTimestampToMs, true));
+    inter_arrival_ast_.reset(new InterArrival(
+        MakeAbsSendTime(kTimestampGroupLengthUs), kAstToMs, true));
+  }
+
+  // Test that neither inter_arrival instance complete the timestamp group from
+  // the given data.
+  void ExpectFalse(int64_t timestamp_us,
+                   int64_t arrival_time_ms,
+                   size_t packet_size) {
+    InternalExpectFalse(inter_arrival_rtp_.get(),
+                        MakeRtpTimestamp(timestamp_us), arrival_time_ms,
+                        packet_size);
+    InternalExpectFalse(inter_arrival_ast_.get(), MakeAbsSendTime(timestamp_us),
+                        arrival_time_ms, packet_size);
+  }
+
+  // Test that both inter_arrival instances complete the timestamp group from
+  // the given data and that all returned deltas are as expected (except
+  // timestamp delta, which is rounded from us to different ranges and must
+  // match within an interval, given in |timestamp_near].
+  void ExpectTrue(int64_t timestamp_us,
+                  int64_t arrival_time_ms,
+                  size_t packet_size,
+                  int64_t expected_timestamp_delta_us,
+                  int64_t expected_arrival_time_delta_ms,
+                  int expected_packet_size_delta,
+                  uint32_t timestamp_near) {
+    InternalExpectTrue(inter_arrival_rtp_.get(), MakeRtpTimestamp(timestamp_us),
+                       arrival_time_ms, packet_size,
+                       MakeRtpTimestamp(expected_timestamp_delta_us),
+                       expected_arrival_time_delta_ms,
+                       expected_packet_size_delta, timestamp_near);
+    InternalExpectTrue(inter_arrival_ast_.get(), MakeAbsSendTime(timestamp_us),
+                       arrival_time_ms, packet_size,
+                       MakeAbsSendTime(expected_timestamp_delta_us),
+                       expected_arrival_time_delta_ms,
+                       expected_packet_size_delta, timestamp_near << 8);
+  }
+
+  void WrapTestHelper(int64_t wrap_start_us,
+                      uint32_t timestamp_near,
+                      bool unorderly_within_group) {
+    // Step through the range of a 32 bit int, 1/4 at a time to not cause
+    // packets close to wraparound to be judged as out of order.
+
+    // G1
+    int64_t arrival_time = 17;
+    ExpectFalse(0, arrival_time, 1);
+
+    // G2
+    arrival_time += kBurstThresholdMs + 1;
+    ExpectFalse(wrap_start_us / 4, arrival_time, 1);
+
+    // G3
+    arrival_time += kBurstThresholdMs + 1;
+    ExpectTrue(wrap_start_us / 2, arrival_time, 1, wrap_start_us / 4, 6,
+               0,  // Delta G2-G1
+               0);
+
+    // G4
+    arrival_time += kBurstThresholdMs + 1;
+    int64_t g4_arrival_time = arrival_time;
+    ExpectTrue(wrap_start_us / 2 + wrap_start_us / 4, arrival_time, 1,
+               wrap_start_us / 4, 6, 0,  // Delta G3-G2
+               timestamp_near);
+
+    // G5
+    arrival_time += kBurstThresholdMs + 1;
+    ExpectTrue(wrap_start_us, arrival_time, 2, wrap_start_us / 4, 6,
+               0,  // Delta G4-G3
+               timestamp_near);
+    for (int i = 0; i < 10; ++i) {
+      // Slowly step across the wrap point.
+      arrival_time += kBurstThresholdMs + 1;
+      if (unorderly_within_group) {
+        // These packets arrive with timestamps in decreasing order but are
+        // nevertheless accumulated to group because their timestamps are higher
+        // than the initial timestamp of the group.
+        ExpectFalse(wrap_start_us + kMinStep * (9 - i), arrival_time, 1);
+      } else {
+        ExpectFalse(wrap_start_us + kMinStep * i, arrival_time, 1);
+      }
+    }
+    int64_t g5_arrival_time = arrival_time;
+
+    // This packet is out of order and should be dropped.
+    arrival_time += kBurstThresholdMs + 1;
+    ExpectFalse(wrap_start_us - 100, arrival_time, 100);
+
+    // G6
+    arrival_time += kBurstThresholdMs + 1;
+    int64_t g6_arrival_time = arrival_time;
+    ExpectTrue(wrap_start_us + kTriggerNewGroupUs, arrival_time, 10,
+               wrap_start_us / 4 + 9 * kMinStep,
+               g5_arrival_time - g4_arrival_time,
+               (2 + 10) - 1,  // Delta G5-G4
+               timestamp_near);
+
+    // This packet is out of order and should be dropped.
+    arrival_time += kBurstThresholdMs + 1;
+    ExpectFalse(wrap_start_us + kTimestampGroupLengthUs, arrival_time, 100);
+
+    // G7
+    arrival_time += kBurstThresholdMs + 1;
+    ExpectTrue(wrap_start_us + 2 * kTriggerNewGroupUs, arrival_time, 100,
+               // Delta G6-G5
+               kTriggerNewGroupUs - 9 * kMinStep,
+               g6_arrival_time - g5_arrival_time, 10 - (2 + 10),
+               timestamp_near);
+  }
+
+  std::unique_ptr<InterArrival> inter_arrival_;
+
+ private:
+  static uint32_t MakeRtpTimestamp(int64_t us) {
+    return static_cast<uint32_t>(static_cast<uint64_t>(us * 90 + 500) / 1000);
+  }
+
+  static uint32_t MakeAbsSendTime(int64_t us) {
+    uint32_t absolute_send_time =
+        static_cast<uint32_t>(((static_cast<uint64_t>(us) << 18) + 500000) /
+                              1000000) &
+        0x00FFFFFFul;
+    return absolute_send_time << 8;
+  }
+
+  static void InternalExpectFalse(InterArrival* inter_arrival,
+                                  uint32_t timestamp,
+                                  int64_t arrival_time_ms,
+                                  size_t packet_size) {
+    uint32_t dummy_timestamp = 101;
+    int64_t dummy_arrival_time_ms = 303;
+    int dummy_packet_size = 909;
+    bool computed = inter_arrival->ComputeDeltas(
+        timestamp, arrival_time_ms, arrival_time_ms, packet_size,
+        &dummy_timestamp, &dummy_arrival_time_ms, &dummy_packet_size);
+    EXPECT_EQ(computed, false);
+    EXPECT_EQ(101ul, dummy_timestamp);
+    EXPECT_EQ(303, dummy_arrival_time_ms);
+    EXPECT_EQ(909, dummy_packet_size);
+  }
+
+  static void InternalExpectTrue(InterArrival* inter_arrival,
+                                 uint32_t timestamp,
+                                 int64_t arrival_time_ms,
+                                 size_t packet_size,
+                                 uint32_t expected_timestamp_delta,
+                                 int64_t expected_arrival_time_delta_ms,
+                                 int expected_packet_size_delta,
+                                 uint32_t timestamp_near) {
+    uint32_t delta_timestamp = 101;
+    int64_t delta_arrival_time_ms = 303;
+    int delta_packet_size = 909;
+    bool computed = inter_arrival->ComputeDeltas(
+        timestamp, arrival_time_ms, arrival_time_ms, packet_size,
+        &delta_timestamp, &delta_arrival_time_ms, &delta_packet_size);
+    EXPECT_EQ(true, computed);
+    EXPECT_NEAR(expected_timestamp_delta, delta_timestamp, timestamp_near);
+    EXPECT_EQ(expected_arrival_time_delta_ms, delta_arrival_time_ms);
+    EXPECT_EQ(expected_packet_size_delta, delta_packet_size);
+  }
+
+  std::unique_ptr<InterArrival> inter_arrival_rtp_;
+  std::unique_ptr<InterArrival> inter_arrival_ast_;
+};
+
+TEST_F(InterArrivalTest, FirstPacket) {
+  ExpectFalse(0, 17, 1);
+}
+
+TEST_F(InterArrivalTest, FirstGroup) {
+  // G1
+  int64_t arrival_time = 17;
+  int64_t g1_arrival_time = arrival_time;
+  ExpectFalse(0, arrival_time, 1);
+
+  // G2
+  arrival_time += kBurstThresholdMs + 1;
+  int64_t g2_arrival_time = arrival_time;
+  ExpectFalse(kTriggerNewGroupUs, arrival_time, 2);
+
+  // G3
+  // Only once the first packet of the third group arrives, do we see the deltas
+  // between the first two.
+  arrival_time += kBurstThresholdMs + 1;
+  ExpectTrue(2 * kTriggerNewGroupUs, arrival_time, 1,
+             // Delta G2-G1
+             kTriggerNewGroupUs, g2_arrival_time - g1_arrival_time, 1, 0);
+}
+
+TEST_F(InterArrivalTest, SecondGroup) {
+  // G1
+  int64_t arrival_time = 17;
+  int64_t g1_arrival_time = arrival_time;
+  ExpectFalse(0, arrival_time, 1);
+
+  // G2
+  arrival_time += kBurstThresholdMs + 1;
+  int64_t g2_arrival_time = arrival_time;
+  ExpectFalse(kTriggerNewGroupUs, arrival_time, 2);
+
+  // G3
+  arrival_time += kBurstThresholdMs + 1;
+  int64_t g3_arrival_time = arrival_time;
+  ExpectTrue(2 * kTriggerNewGroupUs, arrival_time, 1,
+             // Delta G2-G1
+             kTriggerNewGroupUs, g2_arrival_time - g1_arrival_time, 1, 0);
+
+  // G4
+  // First packet of 4th group yields deltas between group 2 and 3.
+  arrival_time += kBurstThresholdMs + 1;
+  ExpectTrue(3 * kTriggerNewGroupUs, arrival_time, 2,
+             // Delta G3-G2
+             kTriggerNewGroupUs, g3_arrival_time - g2_arrival_time, -1, 0);
+}
+
+TEST_F(InterArrivalTest, AccumulatedGroup) {
+  // G1
+  int64_t arrival_time = 17;
+  int64_t g1_arrival_time = arrival_time;
+  ExpectFalse(0, arrival_time, 1);
+
+  // G2
+  arrival_time += kBurstThresholdMs + 1;
+  ExpectFalse(kTriggerNewGroupUs, 28, 2);
+  int64_t timestamp = kTriggerNewGroupUs;
+  for (int i = 0; i < 10; ++i) {
+    // A bunch of packets arriving within the same group.
+    arrival_time += kBurstThresholdMs + 1;
+    timestamp += kMinStep;
+    ExpectFalse(timestamp, arrival_time, 1);
+  }
+  int64_t g2_arrival_time = arrival_time;
+  int64_t g2_timestamp = timestamp;
+
+  // G3
+  arrival_time = 500;
+  ExpectTrue(2 * kTriggerNewGroupUs, arrival_time, 100, g2_timestamp,
+             g2_arrival_time - g1_arrival_time,
+             (2 + 10) - 1,  // Delta G2-G1
+             0);
+}
+
+TEST_F(InterArrivalTest, OutOfOrderPacket) {
+  // G1
+  int64_t arrival_time = 17;
+  int64_t timestamp = 0;
+  ExpectFalse(timestamp, arrival_time, 1);
+  int64_t g1_timestamp = timestamp;
+  int64_t g1_arrival_time = arrival_time;
+
+  // G2
+  arrival_time += 11;
+  timestamp += kTriggerNewGroupUs;
+  ExpectFalse(timestamp, 28, 2);
+  for (int i = 0; i < 10; ++i) {
+    arrival_time += kBurstThresholdMs + 1;
+    timestamp += kMinStep;
+    ExpectFalse(timestamp, arrival_time, 1);
+  }
+  int64_t g2_timestamp = timestamp;
+  int64_t g2_arrival_time = arrival_time;
+
+  // This packet is out of order and should be dropped.
+  arrival_time = 281;
+  ExpectFalse(g1_timestamp, arrival_time, 100);
+
+  // G3
+  arrival_time = 500;
+  timestamp = 2 * kTriggerNewGroupUs;
+  ExpectTrue(timestamp, arrival_time, 100,
+             // Delta G2-G1
+             g2_timestamp - g1_timestamp, g2_arrival_time - g1_arrival_time,
+             (2 + 10) - 1, 0);
+}
+
+TEST_F(InterArrivalTest, OutOfOrderWithinGroup) {
+  // G1
+  int64_t arrival_time = 17;
+  int64_t timestamp = 0;
+  ExpectFalse(timestamp, arrival_time, 1);
+  int64_t g1_timestamp = timestamp;
+  int64_t g1_arrival_time = arrival_time;
+
+  // G2
+  timestamp += kTriggerNewGroupUs;
+  arrival_time += 11;
+  ExpectFalse(kTriggerNewGroupUs, 28, 2);
+  timestamp += 10 * kMinStep;
+  int64_t g2_timestamp = timestamp;
+  for (int i = 0; i < 10; ++i) {
+    // These packets arrive with timestamps in decreasing order but are
+    // nevertheless accumulated to group because their timestamps are higher
+    // than the initial timestamp of the group.
+    arrival_time += kBurstThresholdMs + 1;
+    ExpectFalse(timestamp, arrival_time, 1);
+    timestamp -= kMinStep;
+  }
+  int64_t g2_arrival_time = arrival_time;
+
+  // However, this packet is deemed out of order and should be dropped.
+  arrival_time = 281;
+  timestamp = g1_timestamp;
+  ExpectFalse(timestamp, arrival_time, 100);
+
+  // G3
+  timestamp = 2 * kTriggerNewGroupUs;
+  arrival_time = 500;
+  ExpectTrue(timestamp, arrival_time, 100, g2_timestamp - g1_timestamp,
+             g2_arrival_time - g1_arrival_time, (2 + 10) - 1, 0);
+}
+
+TEST_F(InterArrivalTest, TwoBursts) {
+  // G1
+  int64_t g1_arrival_time = 17;
+  ExpectFalse(0, g1_arrival_time, 1);
+
+  // G2
+  int64_t timestamp = kTriggerNewGroupUs;
+  int64_t arrival_time = 100;  // Simulate no packets arriving for 100 ms.
+  for (int i = 0; i < 10; ++i) {
+    // A bunch of packets arriving in one burst (within 5 ms apart).
+    timestamp += 30000;
+    arrival_time += kBurstThresholdMs;
+    ExpectFalse(timestamp, arrival_time, 1);
+  }
+  int64_t g2_arrival_time = arrival_time;
+  int64_t g2_timestamp = timestamp;
+
+  // G3
+  timestamp += 30000;
+  arrival_time += kBurstThresholdMs + 1;
+  ExpectTrue(timestamp, arrival_time, 100, g2_timestamp,
+             g2_arrival_time - g1_arrival_time,
+             10 - 1,  // Delta G2-G1
+             0);
+}
+
+TEST_F(InterArrivalTest, NoBursts) {
+  // G1
+  ExpectFalse(0, 17, 1);
+
+  // G2
+  int64_t timestamp = kTriggerNewGroupUs;
+  int64_t arrival_time = 28;
+  ExpectFalse(timestamp, arrival_time, 2);
+
+  // G3
+  ExpectTrue(kTriggerNewGroupUs + 30000, arrival_time + kBurstThresholdMs + 1,
+             100, timestamp - 0, arrival_time - 17,
+             2 - 1,  // Delta G2-G1
+             0);
+}
+
+// Yields 0xfffffffe when converted to internal representation in
+// inter_arrival_rtp_ and inter_arrival_ast_ respectively.
+static const int64_t kStartRtpTimestampWrapUs = 47721858827;
+static const int64_t kStartAbsSendTimeWrapUs = 63999995;
+
+TEST_F(InterArrivalTest, RtpTimestampWrap) {
+  WrapTestHelper(kStartRtpTimestampWrapUs, 1, false);
+}
+
+TEST_F(InterArrivalTest, AbsSendTimeWrap) {
+  WrapTestHelper(kStartAbsSendTimeWrapUs, 1, false);
+}
+
+TEST_F(InterArrivalTest, RtpTimestampWrapOutOfOrderWithinGroup) {
+  WrapTestHelper(kStartRtpTimestampWrapUs, 1, true);
+}
+
+TEST_F(InterArrivalTest, AbsSendTimeWrapOutOfOrderWithinGroup) {
+  WrapTestHelper(kStartAbsSendTimeWrapUs, 1, true);
+}
+
+TEST_F(InterArrivalTest, PositiveArrivalTimeJump) {
+  const size_t kPacketSize = 1000;
+  uint32_t send_time_ms = 10000;
+  int64_t arrival_time_ms = 20000;
+  int64_t system_time_ms = 30000;
+
+  uint32_t send_delta;
+  int64_t arrival_delta;
+  int size_delta;
+  EXPECT_FALSE(inter_arrival_->ComputeDeltas(
+      send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+      &arrival_delta, &size_delta));
+
+  const int kTimeDeltaMs = 30;
+  send_time_ms += kTimeDeltaMs;
+  arrival_time_ms += kTimeDeltaMs;
+  system_time_ms += kTimeDeltaMs;
+  EXPECT_FALSE(inter_arrival_->ComputeDeltas(
+      send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+      &arrival_delta, &size_delta));
+
+  send_time_ms += kTimeDeltaMs;
+  arrival_time_ms += kTimeDeltaMs + InterArrival::kArrivalTimeOffsetThresholdMs;
+  system_time_ms += kTimeDeltaMs;
+  EXPECT_TRUE(inter_arrival_->ComputeDeltas(
+      send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+      &arrival_delta, &size_delta));
+  EXPECT_EQ(kTimeDeltaMs, static_cast<int>(send_delta));
+  EXPECT_EQ(kTimeDeltaMs, arrival_delta);
+  EXPECT_EQ(size_delta, 0);
+
+  send_time_ms += kTimeDeltaMs;
+  arrival_time_ms += kTimeDeltaMs;
+  system_time_ms += kTimeDeltaMs;
+  // The previous arrival time jump should now be detected and cause a reset.
+  EXPECT_FALSE(inter_arrival_->ComputeDeltas(
+      send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+      &arrival_delta, &size_delta));
+
+  // The two next packets will not give a valid delta since we're in the initial
+  // state.
+  for (int i = 0; i < 2; ++i) {
+    send_time_ms += kTimeDeltaMs;
+    arrival_time_ms += kTimeDeltaMs;
+    system_time_ms += kTimeDeltaMs;
+    EXPECT_FALSE(inter_arrival_->ComputeDeltas(
+        send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+        &arrival_delta, &size_delta));
+  }
+
+  send_time_ms += kTimeDeltaMs;
+  arrival_time_ms += kTimeDeltaMs;
+  system_time_ms += kTimeDeltaMs;
+  EXPECT_TRUE(inter_arrival_->ComputeDeltas(
+      send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+      &arrival_delta, &size_delta));
+  EXPECT_EQ(kTimeDeltaMs, static_cast<int>(send_delta));
+  EXPECT_EQ(kTimeDeltaMs, arrival_delta);
+  EXPECT_EQ(size_delta, 0);
+}
+
+TEST_F(InterArrivalTest, NegativeArrivalTimeJump) {
+  const size_t kPacketSize = 1000;
+  uint32_t send_time_ms = 10000;
+  int64_t arrival_time_ms = 20000;
+  int64_t system_time_ms = 30000;
+
+  uint32_t send_delta;
+  int64_t arrival_delta;
+  int size_delta;
+  EXPECT_FALSE(inter_arrival_->ComputeDeltas(
+      send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+      &arrival_delta, &size_delta));
+
+  const int kTimeDeltaMs = 30;
+  send_time_ms += kTimeDeltaMs;
+  arrival_time_ms += kTimeDeltaMs;
+  system_time_ms += kTimeDeltaMs;
+  EXPECT_FALSE(inter_arrival_->ComputeDeltas(
+      send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+      &arrival_delta, &size_delta));
+
+  send_time_ms += kTimeDeltaMs;
+  arrival_time_ms += kTimeDeltaMs;
+  system_time_ms += kTimeDeltaMs;
+  EXPECT_TRUE(inter_arrival_->ComputeDeltas(
+      send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+      &arrival_delta, &size_delta));
+  EXPECT_EQ(kTimeDeltaMs, static_cast<int>(send_delta));
+  EXPECT_EQ(kTimeDeltaMs, arrival_delta);
+  EXPECT_EQ(size_delta, 0);
+
+  // Three out of order will fail, after that we will be reset and two more will
+  // fail before we get our first valid delta after the reset.
+  arrival_time_ms -= 1000;
+  for (int i = 0; i < InterArrival::kReorderedResetThreshold + 3; ++i) {
+    send_time_ms += kTimeDeltaMs;
+    arrival_time_ms += kTimeDeltaMs;
+    system_time_ms += kTimeDeltaMs;
+    // The previous arrival time jump should now be detected and cause a reset.
+    EXPECT_FALSE(inter_arrival_->ComputeDeltas(
+        send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+        &arrival_delta, &size_delta));
+  }
+
+  send_time_ms += kTimeDeltaMs;
+  arrival_time_ms += kTimeDeltaMs;
+  system_time_ms += kTimeDeltaMs;
+  EXPECT_TRUE(inter_arrival_->ComputeDeltas(
+      send_time_ms, arrival_time_ms, system_time_ms, kPacketSize, &send_delta,
+      &arrival_delta, &size_delta));
+  EXPECT_EQ(kTimeDeltaMs, static_cast<int>(send_delta));
+  EXPECT_EQ(kTimeDeltaMs, arrival_delta);
+  EXPECT_EQ(size_delta, 0);
+}
+}  // namespace testing
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector.cc
new file mode 100644
index 0000000000..bd2d756876
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/overuse_detector.h"
+
+#include <math.h>
+#include <stdio.h>
+
+#include <algorithm>
+#include <string>
+
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+const double kMaxAdaptOffsetMs = 15.0;
+const double kOverUsingTimeThreshold = 10;
+const int kMaxNumDeltas = 60;
+
+OveruseDetector::OveruseDetector(const FieldTrialsView* key_value_config)
+    // Experiment is on by default, but can be disabled with finch by setting
+    // the field trial string to "WebRTC-AdaptiveBweThreshold/Disabled/".
+    : k_up_(0.0087),
+      k_down_(0.039),
+      overusing_time_threshold_(kOverUsingTimeThreshold),
+      threshold_(12.5),
+      last_update_ms_(-1),
+      prev_offset_(0.0),
+      time_over_using_(-1),
+      overuse_counter_(0),
+      hypothesis_(BandwidthUsage::kBwNormal) {}
+
+OveruseDetector::~OveruseDetector() {}
+
+BandwidthUsage OveruseDetector::State() const {
+  return hypothesis_;
+}
+
+BandwidthUsage OveruseDetector::Detect(double offset,
+                                       double ts_delta,
+                                       int num_of_deltas,
+                                       int64_t now_ms) {
+  if (num_of_deltas < 2) {
+    return BandwidthUsage::kBwNormal;
+  }
+  const double T = std::min(num_of_deltas, kMaxNumDeltas) * offset;
+  BWE_TEST_LOGGING_PLOT(1, "T", now_ms, T);
+  BWE_TEST_LOGGING_PLOT(1, "threshold", now_ms, threshold_);
+  if (T > threshold_) {
+    if (time_over_using_ == -1) {
+      // Initialize the timer. Assume that we've been
+      // over-using half of the time since the previous
+      // sample.
+      time_over_using_ = ts_delta / 2;
+    } else {
+      // Increment timer
+      time_over_using_ += ts_delta;
+    }
+    overuse_counter_++;
+    if (time_over_using_ > overusing_time_threshold_ && overuse_counter_ > 1) {
+      if (offset >= prev_offset_) {
+        time_over_using_ = 0;
+        overuse_counter_ = 0;
+        hypothesis_ = BandwidthUsage::kBwOverusing;
+      }
+    }
+  } else if (T < -threshold_) {
+    time_over_using_ = -1;
+    overuse_counter_ = 0;
+    hypothesis_ = BandwidthUsage::kBwUnderusing;
+  } else {
+    time_over_using_ = -1;
+    overuse_counter_ = 0;
+    hypothesis_ = BandwidthUsage::kBwNormal;
+  }
+  prev_offset_ = offset;
+
+  UpdateThreshold(T, now_ms);
+
+  return hypothesis_;
+}
+
+void OveruseDetector::UpdateThreshold(double modified_offset, int64_t now_ms) {
+  if (last_update_ms_ == -1)
+    last_update_ms_ = now_ms;
+
+  if (fabs(modified_offset) > threshold_ + kMaxAdaptOffsetMs) {
+    // Avoid adapting the threshold to big latency spikes, caused e.g.,
+    // by a sudden capacity drop.
+    last_update_ms_ = now_ms;
+    return;
+  }
+
+  const double k = fabs(modified_offset) < threshold_ ? k_down_ : k_up_;
+  const int64_t kMaxTimeDeltaMs = 100;
+  int64_t time_delta_ms = std::min(now_ms - last_update_ms_, kMaxTimeDeltaMs);
+  threshold_ += k * (fabs(modified_offset) - threshold_) * time_delta_ms;
+  threshold_ = rtc::SafeClamp(threshold_, 6.f, 600.f);
+  last_update_ms_ = now_ms;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector.h
new file mode 100644
index 0000000000..07ae8734c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_DETECTOR_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_DETECTOR_H_
+
+#include <stdint.h>
+
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+
+namespace webrtc {
+
+class OveruseDetector {
+ public:
+  explicit OveruseDetector(const FieldTrialsView* key_value_config);
+  virtual ~OveruseDetector();
+
+  OveruseDetector(const OveruseDetector&) = delete;
+  OveruseDetector& operator=(const OveruseDetector&) = delete;
+
+  // Update the detection state based on the estimated inter-arrival time delta
+  // offset. `timestamp_delta` is the delta between the last timestamp which the
+  // estimated offset is based on and the last timestamp on which the last
+  // offset was based on, representing the time between detector updates.
+  // `num_of_deltas` is the number of deltas the offset estimate is based on.
+  // Returns the state after the detection update.
+  BandwidthUsage Detect(double offset,
+                        double timestamp_delta,
+                        int num_of_deltas,
+                        int64_t now_ms);
+
+  // Returns the current detector state.
+  BandwidthUsage State() const;
+
+ private:
+  void UpdateThreshold(double modified_offset, int64_t now_ms);
+  void InitializeExperiment(const FieldTrialsView& key_value_config);
+
+  const double k_up_;
+  const double k_down_;
+  const double overusing_time_threshold_;
+  double threshold_;
+  int64_t last_update_ms_;
+  double prev_offset_;
+  double time_over_using_;
+  int overuse_counter_;
+  BandwidthUsage hypothesis_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
new file mode 100644
index 0000000000..e91d4f0d22
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
@@ -0,0 +1,682 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/overuse_detector.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include <algorithm>
+#include <cstdlib>
+#include <memory>
+
+#include "api/transport/field_trial_based_config.h"
+#include "modules/remote_bitrate_estimator/inter_arrival.h"
+#include "modules/remote_bitrate_estimator/overuse_estimator.h"
+#include "rtc_base/random.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace testing {
+
+const double kRtpTimestampToMs = 1.0 / 90.0;
+
+class OveruseDetectorTest : public ::testing::Test {
+ public:
+  OveruseDetectorTest()
+      : now_ms_(0),
+        receive_time_ms_(0),
+        rtp_timestamp_(10 * 90),
+        overuse_detector_(),
+        overuse_estimator_(new OveruseEstimator(options_)),
+        inter_arrival_(new InterArrival(5 * 90, kRtpTimestampToMs, true)),
+        random_(123456789) {}
+
+ protected:
+  void SetUp() override {
+    overuse_detector_.reset(new OveruseDetector(&field_trials_));
+  }
+
+  int Run100000Samples(int packets_per_frame,
+                       size_t packet_size,
+                       int mean_ms,
+                       int standard_deviation_ms) {
+    int unique_overuse = 0;
+    int last_overuse = -1;
+    for (int i = 0; i < 100000; ++i) {
+      for (int j = 0; j < packets_per_frame; ++j) {
+        UpdateDetector(rtp_timestamp_, receive_time_ms_, packet_size);
+      }
+      rtp_timestamp_ += mean_ms * 90;
+      now_ms_ += mean_ms;
+      receive_time_ms_ = std::max<int64_t>(
+          receive_time_ms_,
+          now_ms_ + static_cast<int64_t>(
+                        random_.Gaussian(0, standard_deviation_ms) + 0.5));
+      if (BandwidthUsage::kBwOverusing == overuse_detector_->State()) {
+        if (last_overuse + 1 != i) {
+          unique_overuse++;
+        }
+        last_overuse = i;
+      }
+    }
+    return unique_overuse;
+  }
+
+  int RunUntilOveruse(int packets_per_frame,
+                      size_t packet_size,
+                      int mean_ms,
+                      int standard_deviation_ms,
+                      int drift_per_frame_ms) {
+    // Simulate a higher send pace, that is too high.
+    for (int i = 0; i < 1000; ++i) {
+      for (int j = 0; j < packets_per_frame; ++j) {
+        UpdateDetector(rtp_timestamp_, receive_time_ms_, packet_size);
+      }
+      rtp_timestamp_ += mean_ms * 90;
+      now_ms_ += mean_ms + drift_per_frame_ms;
+      receive_time_ms_ = std::max<int64_t>(
+          receive_time_ms_,
+          now_ms_ + static_cast<int64_t>(
+                        random_.Gaussian(0, standard_deviation_ms) + 0.5));
+      if (BandwidthUsage::kBwOverusing == overuse_detector_->State()) {
+        return i + 1;
+      }
+    }
+    return -1;
+  }
+
+  void UpdateDetector(uint32_t rtp_timestamp,
+                      int64_t receive_time_ms,
+                      size_t packet_size) {
+    uint32_t timestamp_delta;
+    int64_t time_delta;
+    int size_delta;
+    if (inter_arrival_->ComputeDeltas(
+            rtp_timestamp, receive_time_ms, receive_time_ms, packet_size,
+            &timestamp_delta, &time_delta, &size_delta)) {
+      double timestamp_delta_ms = timestamp_delta / 90.0;
+      overuse_estimator_->Update(time_delta, timestamp_delta_ms, size_delta,
+                                 overuse_detector_->State(), receive_time_ms);
+      overuse_detector_->Detect(
+          overuse_estimator_->offset(), timestamp_delta_ms,
+          overuse_estimator_->num_of_deltas(), receive_time_ms);
+    }
+  }
+
+  const FieldTrialBasedConfig field_trials_;
+  int64_t now_ms_;
+  int64_t receive_time_ms_;
+  uint32_t rtp_timestamp_;
+  OverUseDetectorOptions options_;
+  std::unique_ptr<OveruseDetector> overuse_detector_;
+  std::unique_ptr<OveruseEstimator> overuse_estimator_;
+  std::unique_ptr<InterArrival> inter_arrival_;
+  Random random_;
+};
+
+TEST_F(OveruseDetectorTest, GaussianRandom) {
+  int buckets[100];
+  memset(buckets, 0, sizeof(buckets));
+  for (int i = 0; i < 100000; ++i) {
+    int index = random_.Gaussian(49, 10);
+    if (index >= 0 && index < 100)
+      buckets[index]++;
+  }
+  for (int n = 0; n < 100; ++n) {
+    printf("Bucket n:%d, %d\n", n, buckets[n]);
+  }
+}
+
+TEST_F(OveruseDetectorTest, SimpleNonOveruse30fps) {
+  size_t packet_size = 1200;
+  uint32_t frame_duration_ms = 33;
+  uint32_t rtp_timestamp = 10 * 90;
+
+  // No variance.
+  for (int i = 0; i < 1000; ++i) {
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    now_ms_ += frame_duration_ms;
+    rtp_timestamp += frame_duration_ms * 90;
+    EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
+  }
+}
+
+// Roughly 1 Mbit/s
+TEST_F(OveruseDetectorTest, SimpleNonOveruseWithReceiveVariance) {
+  uint32_t frame_duration_ms = 10;
+  uint32_t rtp_timestamp = 10 * 90;
+  size_t packet_size = 1200;
+
+  for (int i = 0; i < 1000; ++i) {
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    rtp_timestamp += frame_duration_ms * 90;
+    if (i % 2) {
+      now_ms_ += frame_duration_ms - 5;
+    } else {
+      now_ms_ += frame_duration_ms + 5;
+    }
+    EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
+  }
+}
+
+TEST_F(OveruseDetectorTest, SimpleNonOveruseWithRtpTimestampVariance) {
+  // Roughly 1 Mbit/s.
+  uint32_t frame_duration_ms = 10;
+  uint32_t rtp_timestamp = 10 * 90;
+  size_t packet_size = 1200;
+
+  for (int i = 0; i < 1000; ++i) {
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    now_ms_ += frame_duration_ms;
+    if (i % 2) {
+      rtp_timestamp += (frame_duration_ms - 5) * 90;
+    } else {
+      rtp_timestamp += (frame_duration_ms + 5) * 90;
+    }
+    EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
+  }
+}
+
+TEST_F(OveruseDetectorTest, SimpleOveruse2000Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 6;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 0;  // No variance.
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(7, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, SimpleOveruse100kbit10fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 100;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 0;  // No variance.
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(7, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, OveruseWithLowVariance2000Kbit30fps) {
+  uint32_t frame_duration_ms = 33;
+  uint32_t drift_per_frame_ms = 1;
+  uint32_t rtp_timestamp = frame_duration_ms * 90;
+  size_t packet_size = 1200;
+  int offset = 0;
+
+  // Run 1000 samples to reach steady state.
+  for (int i = 0; i < 1000; ++i) {
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    rtp_timestamp += frame_duration_ms * 90;
+    if (i % 2) {
+      offset = random_.Rand(0, 1);
+      now_ms_ += frame_duration_ms - offset;
+    } else {
+      now_ms_ += frame_duration_ms + offset;
+    }
+    EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
+  }
+  // Simulate a higher send pace, that is too high.
+  // Total build up of 30 ms.
+  for (int j = 0; j < 3; ++j) {
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+    now_ms_ += frame_duration_ms + drift_per_frame_ms * 6;
+    rtp_timestamp += frame_duration_ms * 90;
+    EXPECT_EQ(BandwidthUsage::kBwNormal, overuse_detector_->State());
+  }
+  UpdateDetector(rtp_timestamp, now_ms_, packet_size);
+  EXPECT_EQ(BandwidthUsage::kBwOverusing, overuse_detector_->State());
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVariance30Kbit3fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 333;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(20, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift30Kbit3fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 333;
+  int drift_per_frame_ms = 100;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(4, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVariance30Kbit3fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 333;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(44, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift30Kbit3fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 333;
+  int drift_per_frame_ms = 100;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(4, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVariance100Kbit5fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 2;
+  int frame_duration_ms = 200;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(20, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVariance100Kbit5fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 2;
+  int frame_duration_ms = 200;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(44, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVariance100Kbit10fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 100;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(20, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVariance100Kbit10fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 100;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(44, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVariance300Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(19, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift300Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 10;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(5, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVariance300Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(44, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift300Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 1;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 10;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(10, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVariance1000Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 3;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(19, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift1000Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 3;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 10;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(5, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVariance1000Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 3;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(44, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift1000Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 3;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 10;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(10, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVariance2000Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 6;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(19, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift2000Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 6;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 10;
+  int sigma_ms = 3;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(5, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVariance2000Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 6;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 1;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(44, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift2000Kbit30fps) {
+  size_t packet_size = 1200;
+  int packets_per_frame = 6;
+  int frame_duration_ms = 33;
+  int drift_per_frame_ms = 10;
+  int sigma_ms = 10;
+  int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
+                                        frame_duration_ms, sigma_ms);
+  EXPECT_EQ(0, unique_overuse);
+  int frames_until_overuse =
+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
+                      sigma_ms, drift_per_frame_ms);
+  EXPECT_EQ(10, frames_until_overuse);
+}
+
+TEST_F(OveruseDetectorTest, ThresholdAdapts) {
+  const double kOffset = 0.21;
+  double kTsDelta = 3000.0;
+  int64_t now_ms = 0;
+  int num_deltas = 60;
+  const int kBatchLength = 10;
+
+  // Pass in a positive offset and verify it triggers overuse.
+  bool overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == BandwidthUsage::kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
+
+  // Force the threshold to increase by passing in a higher offset.
+  overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(1.1 * kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == BandwidthUsage::kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
+
+  // Verify that the same offset as before no longer triggers overuse.
+  overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == BandwidthUsage::kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_FALSE(overuse_detected);
+
+  // Pass in a low offset to make the threshold adapt down.
+  for (int i = 0; i < 15 * kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(0.7 * kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == BandwidthUsage::kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_FALSE(overuse_detected);
+
+  // Make sure the original offset now again triggers overuse.
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == BandwidthUsage::kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
+}
+
+TEST_F(OveruseDetectorTest, DoesntAdaptToSpikes) {
+  const double kOffset = 1.0;
+  const double kLargeOffset = 20.0;
+  double kTsDelta = 3000.0;
+  int64_t now_ms = 0;
+  int num_deltas = 60;
+  const int kBatchLength = 10;
+  const int kShortBatchLength = 3;
+
+  // Pass in a positive offset and verify it triggers overuse.
+  bool overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == BandwidthUsage::kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+
+  // Pass in a large offset. This shouldn't have a too big impact on the
+  // threshold, but still trigger an overuse.
+  now_ms += 100;
+  overuse_detected = false;
+  for (int i = 0; i < kShortBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kLargeOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == BandwidthUsage::kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
+
+  // Pass in a positive normal offset and verify it still triggers.
+  overuse_detected = false;
+  for (int i = 0; i < kBatchLength; ++i) {
+    BandwidthUsage overuse_state =
+        overuse_detector_->Detect(kOffset, kTsDelta, num_deltas, now_ms);
+    if (overuse_state == BandwidthUsage::kBwOverusing) {
+      overuse_detected = true;
+    }
+    ++num_deltas;
+    now_ms += 5;
+  }
+  EXPECT_TRUE(overuse_detected);
+}
+}  // namespace testing
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_estimator.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
new file mode 100644
index 0000000000..684143fcce
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
@@ -0,0 +1,164 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/overuse_estimator.h"
+
+#include <math.h>
+#include <string.h>
+
+#include <algorithm>
+
+#include "api/network_state_predictor.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+enum { kMinFramePeriodHistoryLength = 60 };
+enum { kDeltaCounterMax = 1000 };
+
+OveruseEstimator::OveruseEstimator(const OverUseDetectorOptions& options)
+    : options_(options),
+      num_of_deltas_(0),
+      slope_(options_.initial_slope),
+      offset_(options_.initial_offset),
+      prev_offset_(options_.initial_offset),
+      E_(),
+      process_noise_(),
+      avg_noise_(options_.initial_avg_noise),
+      var_noise_(options_.initial_var_noise),
+      ts_delta_hist_() {
+  memcpy(E_, options_.initial_e, sizeof(E_));
+  memcpy(process_noise_, options_.initial_process_noise,
+         sizeof(process_noise_));
+}
+
+OveruseEstimator::~OveruseEstimator() {
+  ts_delta_hist_.clear();
+}
+
+void OveruseEstimator::Update(int64_t t_delta,
+                              double ts_delta,
+                              int size_delta,
+                              BandwidthUsage current_hypothesis,
+                              int64_t now_ms) {
+  const double min_frame_period = UpdateMinFramePeriod(ts_delta);
+  const double t_ts_delta = t_delta - ts_delta;
+  BWE_TEST_LOGGING_PLOT(1, "dm_ms", now_ms, t_ts_delta);
+  double fs_delta = size_delta;
+
+  ++num_of_deltas_;
+  if (num_of_deltas_ > kDeltaCounterMax) {
+    num_of_deltas_ = kDeltaCounterMax;
+  }
+
+  // Update the Kalman filter.
+  E_[0][0] += process_noise_[0];
+  E_[1][1] += process_noise_[1];
+
+  if ((current_hypothesis == BandwidthUsage::kBwOverusing &&
+       offset_ < prev_offset_) ||
+      (current_hypothesis == BandwidthUsage::kBwUnderusing &&
+       offset_ > prev_offset_)) {
+    E_[1][1] += 10 * process_noise_[1];
+  }
+
+  const double h[2] = {fs_delta, 1.0};
+  const double Eh[2] = {E_[0][0] * h[0] + E_[0][1] * h[1],
+                        E_[1][0] * h[0] + E_[1][1] * h[1]};
+
+  BWE_TEST_LOGGING_PLOT(1, "d_ms", now_ms, slope_ * h[0] - offset_);
+
+  const double residual = t_ts_delta - slope_ * h[0] - offset_;
+
+  const bool in_stable_state =
+      (current_hypothesis == BandwidthUsage::kBwNormal);
+  const double max_residual = 3.0 * sqrt(var_noise_);
+  // We try to filter out very late frames. For instance periodic key
+  // frames doesn't fit the Gaussian model well.
+  if (fabs(residual) < max_residual) {
+    UpdateNoiseEstimate(residual, min_frame_period, in_stable_state);
+  } else {
+    UpdateNoiseEstimate(residual < 0 ? -max_residual : max_residual,
+                        min_frame_period, in_stable_state);
+  }
+
+  const double denom = var_noise_ + h[0] * Eh[0] + h[1] * Eh[1];
+
+  const double K[2] = {Eh[0] / denom, Eh[1] / denom};
+
+  const double IKh[2][2] = {{1.0 - K[0] * h[0], -K[0] * h[1]},
+                            {-K[1] * h[0], 1.0 - K[1] * h[1]}};
+  const double e00 = E_[0][0];
+  const double e01 = E_[0][1];
+
+  // Update state.
+  E_[0][0] = e00 * IKh[0][0] + E_[1][0] * IKh[0][1];
+  E_[0][1] = e01 * IKh[0][0] + E_[1][1] * IKh[0][1];
+  E_[1][0] = e00 * IKh[1][0] + E_[1][0] * IKh[1][1];
+  E_[1][1] = e01 * IKh[1][0] + E_[1][1] * IKh[1][1];
+
+  // The covariance matrix must be positive semi-definite.
+  bool positive_semi_definite =
+      E_[0][0] + E_[1][1] >= 0 &&
+      E_[0][0] * E_[1][1] - E_[0][1] * E_[1][0] >= 0 && E_[0][0] >= 0;
+  RTC_DCHECK(positive_semi_definite);
+  if (!positive_semi_definite) {
+    RTC_LOG(LS_ERROR)
+        << "The over-use estimator's covariance matrix is no longer "
+           "semi-definite.";
+  }
+
+  slope_ = slope_ + K[0] * residual;
+  prev_offset_ = offset_;
+  offset_ = offset_ + K[1] * residual;
+
+  BWE_TEST_LOGGING_PLOT(1, "kc", now_ms, K[0]);
+  BWE_TEST_LOGGING_PLOT(1, "km", now_ms, K[1]);
+  BWE_TEST_LOGGING_PLOT(1, "slope_1/bps", now_ms, slope_);
+  BWE_TEST_LOGGING_PLOT(1, "var_noise", now_ms, var_noise_);
+}
+
+double OveruseEstimator::UpdateMinFramePeriod(double ts_delta) {
+  double min_frame_period = ts_delta;
+  if (ts_delta_hist_.size() >= kMinFramePeriodHistoryLength) {
+    ts_delta_hist_.pop_front();
+  }
+  for (const double old_ts_delta : ts_delta_hist_) {
+    min_frame_period = std::min(old_ts_delta, min_frame_period);
+  }
+  ts_delta_hist_.push_back(ts_delta);
+  return min_frame_period;
+}
+
+void OveruseEstimator::UpdateNoiseEstimate(double residual,
+                                           double ts_delta,
+                                           bool stable_state) {
+  if (!stable_state) {
+    return;
+  }
+  // Faster filter during startup to faster adapt to the jitter level
+  // of the network. `alpha` is tuned for 30 frames per second, but is scaled
+  // according to `ts_delta`.
+  double alpha = 0.01;
+  if (num_of_deltas_ > 10 * 30) {
+    alpha = 0.002;
+  }
+  // Only update the noise estimate if we're not over-using. `beta` is a
+  // function of alpha and the time delta since the previous update.
+  const double beta = pow(1 - alpha, ts_delta * 30.0 / 1000.0);
+  avg_noise_ = beta * avg_noise_ + (1 - beta) * residual;
+  var_noise_ = beta * var_noise_ +
+               (1 - beta) * (avg_noise_ - residual) * (avg_noise_ - residual);
+  if (var_noise_ < 1) {
+    var_noise_ = 1;
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_estimator.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_estimator.h
new file mode 100644
index 0000000000..c021f00da7
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_estimator.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_ESTIMATOR_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_ESTIMATOR_H_
+
+#include <stdint.h>
+
+#include <deque>
+
+#include "api/network_state_predictor.h"
+
+namespace webrtc {
+
+// Bandwidth over-use detector options.  These are used to drive
+// experimentation with bandwidth estimation parameters.
+// TODO(terelius): This is only used in overuse_estimator.cc, and only in the
+// default constructed state. Can we move the relevant variables into that
+// class and delete this?
+struct OverUseDetectorOptions {
+  OverUseDetectorOptions() = default;
+  double initial_slope = 8.0 / 512.0;
+  double initial_offset = 0;
+  double initial_e[2][2] = {{100.0, 0.0}, {0.0, 1e-1}};
+  double initial_process_noise[2] = {1e-13, 1e-3};
+  double initial_avg_noise = 0.0;
+  double initial_var_noise = 50.0;
+};
+
+class OveruseEstimator {
+ public:
+  explicit OveruseEstimator(const OverUseDetectorOptions& options);
+  ~OveruseEstimator();
+
+  OveruseEstimator(const OveruseEstimator&) = delete;
+  OveruseEstimator& operator=(const OveruseEstimator&) = delete;
+
+  // Update the estimator with a new sample. The deltas should represent deltas
+  // between timestamp groups as defined by the InterArrival class.
+  // `current_hypothesis` should be the hypothesis of the over-use detector at
+  // this time.
+  void Update(int64_t t_delta,
+              double ts_delta,
+              int size_delta,
+              BandwidthUsage current_hypothesis,
+              int64_t now_ms);
+
+  // Returns the estimated noise/jitter variance in ms^2.
+  double var_noise() const { return var_noise_; }
+
+  // Returns the estimated inter-arrival time delta offset in ms.
+  double offset() const { return offset_; }
+
+  // Returns the number of deltas which the current over-use estimator state is
+  // based on.
+  unsigned int num_of_deltas() const { return num_of_deltas_; }
+
+ private:
+  double UpdateMinFramePeriod(double ts_delta);
+  void UpdateNoiseEstimate(double residual, double ts_delta, bool stable_state);
+
+  // Must be first member variable. Cannot be const because we need to be
+  // copyable.
+  OverUseDetectorOptions options_;
+  uint16_t num_of_deltas_;
+  double slope_;
+  double offset_;
+  double prev_offset_;
+  double E_[2][2];
+  double process_noise_[2];
+  double avg_noise_;
+  double var_noise_;
+  std::deque<double> ts_delta_hist_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map.cc
new file mode 100644
index 0000000000..182297d303
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map.cc
@@ -0,0 +1,164 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/remote_bitrate_estimator/packet_arrival_map.h"
+
+#include <algorithm>
+#include <cstdint>
+
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void PacketArrivalTimeMap::AddPacket(int64_t sequence_number,
+                                     Timestamp arrival_time) {
+  RTC_DCHECK_GE(arrival_time, Timestamp::Zero());
+  if (!has_seen_packet()) {
+    // First packet.
+    Reallocate(kMinCapacity);
+    begin_sequence_number_ = sequence_number;
+    end_sequence_number_ = sequence_number + 1;
+    arrival_times_[Index(sequence_number)] = arrival_time;
+    return;
+  }
+
+  if (sequence_number >= begin_sequence_number() &&
+      sequence_number < end_sequence_number()) {
+    // The packet is within the buffer - no need to expand it.
+    arrival_times_[Index(sequence_number)] = arrival_time;
+    return;
+  }
+
+  if (sequence_number < begin_sequence_number()) {
+    // The packet goes before the current buffer. Expand to add packet, but only
+    // if it fits within kMaxNumberOfPackets.
+    int64_t new_size = end_sequence_number() - sequence_number;
+    if (new_size > kMaxNumberOfPackets) {
+      // Don't expand the buffer further, as that would remove newly received
+      // packets.
+      return;
+    }
+    AdjustToSize(new_size);
+
+    arrival_times_[Index(sequence_number)] = arrival_time;
+    SetNotReceived(sequence_number + 1, begin_sequence_number_);
+    begin_sequence_number_ = sequence_number;
+    return;
+  }
+
+  // The packet goes after the buffer.
+  RTC_DCHECK_GE(sequence_number, end_sequence_number_);
+  int64_t new_end_sequence_number = sequence_number + 1;
+
+  if (new_end_sequence_number >= end_sequence_number_ + kMaxNumberOfPackets) {
+    // All old packets have to be removed.
+    begin_sequence_number_ = sequence_number;
+    end_sequence_number_ = new_end_sequence_number;
+    arrival_times_[Index(sequence_number)] = arrival_time;
+    return;
+  }
+
+  if (begin_sequence_number_ < new_end_sequence_number - kMaxNumberOfPackets) {
+    // Remove oldest entries
+    begin_sequence_number_ = new_end_sequence_number - kMaxNumberOfPackets;
+    RTC_DCHECK_GT(end_sequence_number_, begin_sequence_number_);
+  }
+
+  AdjustToSize(new_end_sequence_number - begin_sequence_number_);
+
+  // Packets can be received out-of-order. If this isn't the next expected
+  // packet, add enough placeholders to fill the gap.
+  SetNotReceived(end_sequence_number_, sequence_number);
+  end_sequence_number_ = new_end_sequence_number;
+  arrival_times_[Index(sequence_number)] = arrival_time;
+}
+
+void PacketArrivalTimeMap::SetNotReceived(
+    int64_t begin_sequence_number_inclusive,
+    int64_t end_sequence_number_exclusive) {
+  static constexpr Timestamp value = Timestamp::MinusInfinity();
+
+  int begin_index = Index(begin_sequence_number_inclusive);
+  int end_index = Index(end_sequence_number_exclusive);
+
+  if (begin_index <= end_index) {
+    // Entries to clear are in single block:
+    // [......{-----}....]
+    std::fill(arrival_times_.get() + begin_index,
+              arrival_times_.get() + end_index, value);
+  } else {
+    // Entries to clear span across arrival_times_ border:
+    // [--}..........{---]
+    std::fill(arrival_times_.get() + begin_index,
+              arrival_times_.get() + capacity(), value);
+    std::fill(arrival_times_.get(), arrival_times_.get() + end_index, value);
+  }
+}
+
+void PacketArrivalTimeMap::RemoveOldPackets(int64_t sequence_number,
+                                            Timestamp arrival_time_limit) {
+  int64_t check_to = std::min(sequence_number, end_sequence_number_);
+  while (begin_sequence_number_ < check_to &&
+         arrival_times_[Index(begin_sequence_number_)] <= arrival_time_limit) {
+    ++begin_sequence_number_;
+  }
+  AdjustToSize(end_sequence_number_ - begin_sequence_number_);
+}
+
+void PacketArrivalTimeMap::EraseTo(int64_t sequence_number) {
+  if (sequence_number < begin_sequence_number_) {
+    return;
+  }
+  if (sequence_number >= end_sequence_number_) {
+    // Erase all.
+    begin_sequence_number_ = end_sequence_number_;
+    return;
+  }
+  // Remove some.
+  begin_sequence_number_ = sequence_number;
+  AdjustToSize(end_sequence_number_ - begin_sequence_number_);
+}
+
+void PacketArrivalTimeMap::AdjustToSize(int new_size) {
+  if (new_size > capacity()) {
+    int new_capacity = capacity();
+    while (new_capacity < new_size)
+      new_capacity *= 2;
+    Reallocate(new_capacity);
+  }
+  if (capacity() > std::max(kMinCapacity, 4 * new_size)) {
+    int new_capacity = capacity();
+    while (new_capacity > 2 * std::max(new_size, kMinCapacity)) {
+      new_capacity /= 2;
+    }
+    Reallocate(new_capacity);
+  }
+  RTC_DCHECK_LE(new_size, capacity());
+}
+
+void PacketArrivalTimeMap::Reallocate(int new_capacity) {
+  int new_capacity_minus_1 = new_capacity - 1;
+  // Check capacity is a power of 2.
+  RTC_DCHECK_EQ(new_capacity & new_capacity_minus_1, 0);
+  // Create uninitialized memory.
+  // All valid entries should be set by `AddPacket` before use.
+  void* raw = operator new[](new_capacity * sizeof(Timestamp));
+  Timestamp* new_buffer = static_cast<Timestamp*>(raw);
+
+  for (int64_t sequence_number = begin_sequence_number_;
+       sequence_number < end_sequence_number_; ++sequence_number) {
+    new_buffer[sequence_number & new_capacity_minus_1] =
+        arrival_times_[sequence_number & capacity_minus_1_];
+  }
+  arrival_times_.reset(new_buffer);
+  capacity_minus_1_ = new_capacity_minus_1;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map.h
new file mode 100644
index 0000000000..4ae47172bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map.h
@@ -0,0 +1,144 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_PACKET_ARRIVAL_MAP_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_PACKET_ARRIVAL_MAP_H_
+
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+#include <memory>
+
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// PacketArrivalTimeMap is an optimized map of packet sequence number to arrival
+// time, limited in size to never exceed `kMaxNumberOfPackets`. It will grow as
+// needed, and remove old packets, and will expand to allow earlier packets to
+// be added (out-of-order).
+//
+// Not yet received packets have the arrival time zero. The queue will not span
+// larger than necessary and the last packet should always be received. The
+// first packet in the queue doesn't have to be received in case of receiving
+// packets out-of-order.
+class PacketArrivalTimeMap {
+ public:
+  struct PacketArrivalTime {
+    Timestamp arrival_time;
+    int64_t sequence_number;
+  };
+  // Impossible to request feedback older than what can be represented by 15
+  // bits.
+  static constexpr int kMaxNumberOfPackets = (1 << 15);
+
+  PacketArrivalTimeMap() = default;
+  PacketArrivalTimeMap(const PacketArrivalTimeMap&) = delete;
+  PacketArrivalTimeMap& operator=(const PacketArrivalTimeMap&) = delete;
+  ~PacketArrivalTimeMap() = default;
+
+  // Indicates if the packet with `sequence_number` has already been received.
+  bool has_received(int64_t sequence_number) const {
+    return sequence_number >= begin_sequence_number() &&
+           sequence_number < end_sequence_number() &&
+           arrival_times_[Index(sequence_number)] >= Timestamp::Zero();
+  }
+
+  // Returns the sequence number of the first entry in the map, i.e. the
+  // sequence number that a `begin()` iterator would represent.
+  int64_t begin_sequence_number() const { return begin_sequence_number_; }
+
+  // Returns the sequence number of the element just after the map, i.e. the
+  // sequence number that an `end()` iterator would represent.
+  int64_t end_sequence_number() const { return end_sequence_number_; }
+
+  // Returns an element by `sequence_number`, which must be valid, i.e.
+  // between [begin_sequence_number, end_sequence_number).
+  Timestamp get(int64_t sequence_number) {
+    RTC_DCHECK_GE(sequence_number, begin_sequence_number());
+    RTC_DCHECK_LT(sequence_number, end_sequence_number());
+    return arrival_times_[Index(sequence_number)];
+  }
+
+  // Returns timestamp and sequence number of the received packet with sequence
+  // number equal or larger than `sequence_number`. `sequence_number` must be in
+  // range [begin_sequence_number, end_sequence_number).
+  PacketArrivalTime FindNextAtOrAfter(int64_t sequence_number) const {
+    RTC_DCHECK_GE(sequence_number, begin_sequence_number());
+    RTC_DCHECK_LT(sequence_number, end_sequence_number());
+    while (true) {
+      Timestamp t = arrival_times_[Index(sequence_number)];
+      if (t >= Timestamp::Zero()) {
+        return {.arrival_time = t, .sequence_number = sequence_number};
+      }
+      ++sequence_number;
+    }
+  }
+
+  // Clamps `sequence_number` between [begin_sequence_number,
+  // end_sequence_number].
+  int64_t clamp(int64_t sequence_number) const {
+    return std::clamp(sequence_number, begin_sequence_number(),
+                      end_sequence_number());
+  }
+
+  // Erases all elements from the beginning of the map until `sequence_number`.
+  void EraseTo(int64_t sequence_number);
+
+  // Records the fact that a packet with `sequence_number` arrived at
+  // `arrival_time_ms`.
+  void AddPacket(int64_t sequence_number, Timestamp arrival_time);
+
+  // Removes packets from the beginning of the map as long as they are received
+  // before `sequence_number` and with an age older than `arrival_time_limit`
+  void RemoveOldPackets(int64_t sequence_number, Timestamp arrival_time_limit);
+
+ private:
+  static constexpr int kMinCapacity = 128;
+
+  // Returns index in the `arrival_times_` for value for `sequence_number`.
+  int Index(int64_t sequence_number) const {
+    // Note that sequence_number might be negative, thus taking '%' requires
+    // extra handling and can be slow. Because capacity is a power of two, it
+    // is much faster to use '&' operator.
+    return sequence_number & capacity_minus_1_;
+  }
+
+  void SetNotReceived(int64_t begin_sequence_number_inclusive,
+                      int64_t end_sequence_number_exclusive);
+
+  // Adjust capacity to match new_size, may reduce capacity.
+  // On return guarantees capacity >= new_size.
+  void AdjustToSize(int new_size);
+  void Reallocate(int new_capacity);
+
+  int capacity() const { return capacity_minus_1_ + 1; }
+  bool has_seen_packet() const { return arrival_times_ != nullptr; }
+
+  // Circular buffer. Packet with sequence number `sequence_number`
+  // is stored in the slot `sequence_number % capacity_`
+  std::unique_ptr<Timestamp[]> arrival_times_ = nullptr;
+
+  // Allocated size of the `arrival_times_`
+  // capacity_ is a power of 2 in range [kMinCapacity, kMaxNumberOfPackets]
+  // `capacity - 1` is used much more often than `capacity`, thus that value is
+  // stored.
+  int capacity_minus_1_ = -1;
+
+  // The unwrapped sequence number for valid range of sequence numbers.
+  // arrival_times_ entries only valid for sequence numbers in range
+  // `begin_sequence_number_ <= sequence_number < end_sequence_number_`
+  int64_t begin_sequence_number_ = 0;
+  int64_t end_sequence_number_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_PACKET_ARRIVAL_MAP_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map_test.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map_test.cc
new file mode 100644
index 0000000000..d86f0397e7
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map_test.cc
@@ -0,0 +1,272 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/remote_bitrate_estimator/packet_arrival_map.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+TEST(PacketArrivalMapTest, IsConsistentWhenEmpty) {
+  PacketArrivalTimeMap map;
+
+  EXPECT_EQ(map.begin_sequence_number(), map.end_sequence_number());
+  EXPECT_FALSE(map.has_received(0));
+  EXPECT_EQ(map.clamp(-5), 0);
+  EXPECT_EQ(map.clamp(5), 0);
+}
+
+TEST(PacketArrivalMapTest, InsertsFirstItemIntoMap) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(42, Timestamp::Millis(10));
+  EXPECT_EQ(map.begin_sequence_number(), 42);
+  EXPECT_EQ(map.end_sequence_number(), 43);
+
+  EXPECT_FALSE(map.has_received(41));
+  EXPECT_TRUE(map.has_received(42));
+  EXPECT_FALSE(map.has_received(44));
+
+  EXPECT_EQ(map.clamp(-100), 42);
+  EXPECT_EQ(map.clamp(42), 42);
+  EXPECT_EQ(map.clamp(100), 43);
+}
+
+TEST(PacketArrivalMapTest, InsertsWithGaps) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(42, Timestamp::Zero());
+  map.AddPacket(45, Timestamp::Millis(11));
+  EXPECT_EQ(map.begin_sequence_number(), 42);
+  EXPECT_EQ(map.end_sequence_number(), 46);
+
+  EXPECT_FALSE(map.has_received(41));
+  EXPECT_TRUE(map.has_received(42));
+  EXPECT_FALSE(map.has_received(43));
+  EXPECT_FALSE(map.has_received(44));
+  EXPECT_TRUE(map.has_received(45));
+  EXPECT_FALSE(map.has_received(46));
+
+  EXPECT_EQ(map.get(42), Timestamp::Zero());
+  EXPECT_LT(map.get(43), Timestamp::Zero());
+  EXPECT_LT(map.get(44), Timestamp::Zero());
+  EXPECT_EQ(map.get(45), Timestamp::Millis(11));
+
+  EXPECT_EQ(map.clamp(-100), 42);
+  EXPECT_EQ(map.clamp(44), 44);
+  EXPECT_EQ(map.clamp(100), 46);
+}
+
+TEST(PacketArrivalMapTest, FindNextAtOrAfterWithGaps) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(42, Timestamp::Zero());
+  map.AddPacket(45, Timestamp::Millis(11));
+  EXPECT_EQ(map.begin_sequence_number(), 42);
+  EXPECT_EQ(map.end_sequence_number(), 46);
+
+  PacketArrivalTimeMap::PacketArrivalTime packet = map.FindNextAtOrAfter(42);
+  EXPECT_EQ(packet.arrival_time, Timestamp::Zero());
+  EXPECT_EQ(packet.sequence_number, 42);
+
+  packet = map.FindNextAtOrAfter(43);
+  EXPECT_EQ(packet.arrival_time, Timestamp::Millis(11));
+  EXPECT_EQ(packet.sequence_number, 45);
+}
+
+TEST(PacketArrivalMapTest, InsertsWithinBuffer) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(42, Timestamp::Millis(10));
+  map.AddPacket(45, Timestamp::Millis(11));
+
+  map.AddPacket(43, Timestamp::Millis(12));
+  map.AddPacket(44, Timestamp::Millis(13));
+
+  EXPECT_EQ(map.begin_sequence_number(), 42);
+  EXPECT_EQ(map.end_sequence_number(), 46);
+
+  EXPECT_FALSE(map.has_received(41));
+  EXPECT_TRUE(map.has_received(42));
+  EXPECT_TRUE(map.has_received(43));
+  EXPECT_TRUE(map.has_received(44));
+  EXPECT_TRUE(map.has_received(45));
+  EXPECT_FALSE(map.has_received(46));
+
+  EXPECT_EQ(map.get(42), Timestamp::Millis(10));
+  EXPECT_EQ(map.get(43), Timestamp::Millis(12));
+  EXPECT_EQ(map.get(44), Timestamp::Millis(13));
+  EXPECT_EQ(map.get(45), Timestamp::Millis(11));
+}
+
+TEST(PacketArrivalMapTest, GrowsBufferAndRemoveOld) {
+  PacketArrivalTimeMap map;
+
+  constexpr int64_t kLargeSeq = 42 + PacketArrivalTimeMap::kMaxNumberOfPackets;
+  map.AddPacket(42, Timestamp::Millis(10));
+  map.AddPacket(43, Timestamp::Millis(11));
+  map.AddPacket(44, Timestamp::Millis(12));
+  map.AddPacket(45, Timestamp::Millis(13));
+  map.AddPacket(kLargeSeq, Timestamp::Millis(12));
+
+  EXPECT_EQ(map.begin_sequence_number(), 43);
+  EXPECT_EQ(map.end_sequence_number(), kLargeSeq + 1);
+  EXPECT_EQ(map.end_sequence_number() - map.begin_sequence_number(),
+            PacketArrivalTimeMap::kMaxNumberOfPackets);
+
+  EXPECT_FALSE(map.has_received(41));
+  EXPECT_FALSE(map.has_received(42));
+  EXPECT_TRUE(map.has_received(43));
+  EXPECT_TRUE(map.has_received(44));
+  EXPECT_TRUE(map.has_received(45));
+  EXPECT_FALSE(map.has_received(46));
+  EXPECT_TRUE(map.has_received(kLargeSeq));
+  EXPECT_FALSE(map.has_received(kLargeSeq + 1));
+}
+
+TEST(PacketArrivalMapTest, SequenceNumberJumpsDeletesAll) {
+  PacketArrivalTimeMap map;
+
+  constexpr int64_t kLargeSeq =
+      42 + 2 * PacketArrivalTimeMap::kMaxNumberOfPackets;
+  map.AddPacket(42, Timestamp::Millis(10));
+  map.AddPacket(kLargeSeq, Timestamp::Millis(12));
+
+  EXPECT_EQ(map.begin_sequence_number(), kLargeSeq);
+  EXPECT_EQ(map.end_sequence_number(), kLargeSeq + 1);
+
+  EXPECT_FALSE(map.has_received(42));
+  EXPECT_TRUE(map.has_received(kLargeSeq));
+  EXPECT_FALSE(map.has_received(kLargeSeq + 1));
+}
+
+TEST(PacketArrivalMapTest, ExpandsBeforeBeginning) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(42, Timestamp::Millis(10));
+  map.AddPacket(-1000, Timestamp::Millis(13));
+
+  EXPECT_EQ(map.begin_sequence_number(), -1000);
+  EXPECT_EQ(map.end_sequence_number(), 43);
+
+  EXPECT_FALSE(map.has_received(-1001));
+  EXPECT_TRUE(map.has_received(-1000));
+  EXPECT_FALSE(map.has_received(-999));
+  EXPECT_TRUE(map.has_received(42));
+  EXPECT_FALSE(map.has_received(43));
+}
+
+TEST(PacketArrivalMapTest, ExpandingBeforeBeginningKeepsReceived) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(42, Timestamp::Millis(10));
+  constexpr int64_t kSmallSeq =
+      static_cast<int64_t>(42) - 2 * PacketArrivalTimeMap::kMaxNumberOfPackets;
+  map.AddPacket(kSmallSeq, Timestamp::Millis(13));
+
+  EXPECT_EQ(map.begin_sequence_number(), 42);
+  EXPECT_EQ(map.end_sequence_number(), 43);
+}
+
+TEST(PacketArrivalMapTest, ErasesToRemoveElements) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(42, Timestamp::Millis(10));
+  map.AddPacket(43, Timestamp::Millis(11));
+  map.AddPacket(44, Timestamp::Millis(12));
+  map.AddPacket(45, Timestamp::Millis(13));
+
+  map.EraseTo(44);
+
+  EXPECT_EQ(map.begin_sequence_number(), 44);
+  EXPECT_EQ(map.end_sequence_number(), 46);
+
+  EXPECT_FALSE(map.has_received(43));
+  EXPECT_TRUE(map.has_received(44));
+  EXPECT_TRUE(map.has_received(45));
+  EXPECT_FALSE(map.has_received(46));
+}
+
+TEST(PacketArrivalMapTest, ErasesInEmptyMap) {
+  PacketArrivalTimeMap map;
+
+  EXPECT_EQ(map.begin_sequence_number(), map.end_sequence_number());
+
+  map.EraseTo(map.end_sequence_number());
+  EXPECT_EQ(map.begin_sequence_number(), map.end_sequence_number());
+}
+
+TEST(PacketArrivalMapTest, IsTolerantToWrongArgumentsForErase) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(42, Timestamp::Millis(10));
+  map.AddPacket(43, Timestamp::Millis(11));
+
+  map.EraseTo(1);
+
+  EXPECT_EQ(map.begin_sequence_number(), 42);
+  EXPECT_EQ(map.end_sequence_number(), 44);
+
+  map.EraseTo(100);
+
+  EXPECT_EQ(map.begin_sequence_number(), 44);
+  EXPECT_EQ(map.end_sequence_number(), 44);
+}
+
+TEST(PacketArrivalMapTest, EraseAllRemembersBeginningSeqNbr) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(42, Timestamp::Millis(10));
+  map.AddPacket(43, Timestamp::Millis(11));
+  map.AddPacket(44, Timestamp::Millis(12));
+  map.AddPacket(45, Timestamp::Millis(13));
+
+  map.EraseTo(46);
+
+  map.AddPacket(50, Timestamp::Millis(10));
+
+  EXPECT_EQ(map.begin_sequence_number(), 46);
+  EXPECT_EQ(map.end_sequence_number(), 51);
+
+  EXPECT_FALSE(map.has_received(45));
+  EXPECT_FALSE(map.has_received(46));
+  EXPECT_FALSE(map.has_received(47));
+  EXPECT_FALSE(map.has_received(48));
+  EXPECT_FALSE(map.has_received(49));
+  EXPECT_TRUE(map.has_received(50));
+  EXPECT_FALSE(map.has_received(51));
+}
+
+TEST(PacketArrivalMapTest, EraseToMissingSequenceNumber) {
+  PacketArrivalTimeMap map;
+
+  map.AddPacket(37, Timestamp::Millis(10));
+  map.AddPacket(39, Timestamp::Millis(11));
+  map.AddPacket(40, Timestamp::Millis(12));
+  map.AddPacket(41, Timestamp::Millis(13));
+
+  map.EraseTo(38);
+
+  map.AddPacket(42, Timestamp::Millis(40));
+
+  EXPECT_EQ(map.begin_sequence_number(), 38);
+  EXPECT_EQ(map.end_sequence_number(), 43);
+
+  EXPECT_FALSE(map.has_received(37));
+  EXPECT_FALSE(map.has_received(38));
+  EXPECT_TRUE(map.has_received(39));
+  EXPECT_TRUE(map.has_received(40));
+  EXPECT_TRUE(map.has_received(41));
+  EXPECT_TRUE(map.has_received(42));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
new file mode 100644
index 0000000000..e9fb1b99f6
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -0,0 +1,399 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "api/transport/field_trial_based_config.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+
+constexpr TimeDelta kMinClusterDelta = TimeDelta::Millis(1);
+constexpr TimeDelta kInitialProbingInterval = TimeDelta::Seconds(2);
+constexpr int kTimestampGroupLengthMs = 5;
+constexpr int kAbsSendTimeInterArrivalUpshift = 8;
+constexpr int kInterArrivalShift =
+    RTPHeaderExtension::kAbsSendTimeFraction + kAbsSendTimeInterArrivalUpshift;
+constexpr int kMinClusterSize = 4;
+constexpr int kMaxProbePackets = 15;
+constexpr int kExpectedNumberOfProbes = 3;
+constexpr double kTimestampToMs =
+    1000.0 / static_cast<double>(1 << kInterArrivalShift);
+
+absl::optional<DataRate> OptionalRateFromOptionalBps(
+    absl::optional<int> bitrate_bps) {
+  if (bitrate_bps) {
+    return DataRate::BitsPerSec(*bitrate_bps);
+  } else {
+    return absl::nullopt;
+  }
+}
+
+template <typename K, typename V>
+std::vector<K> Keys(const std::map<K, V>& map) {
+  std::vector<K> keys;
+  keys.reserve(map.size());
+  for (const auto& kv_pair : map) {
+    keys.push_back(kv_pair.first);
+  }
+  return keys;
+}
+
+}  // namespace
+
+RemoteBitrateEstimatorAbsSendTime::~RemoteBitrateEstimatorAbsSendTime() =
+    default;
+
+bool RemoteBitrateEstimatorAbsSendTime::IsWithinClusterBounds(
+    TimeDelta send_delta,
+    const Cluster& cluster_aggregate) {
+  if (cluster_aggregate.count == 0)
+    return true;
+  TimeDelta cluster_mean =
+      cluster_aggregate.send_mean / cluster_aggregate.count;
+  return (send_delta - cluster_mean).Abs() < TimeDelta::Micros(2'500);
+}
+
+void RemoteBitrateEstimatorAbsSendTime::MaybeAddCluster(
+    const Cluster& cluster_aggregate,
+    std::list<Cluster>& clusters) {
+  if (cluster_aggregate.count < kMinClusterSize ||
+      cluster_aggregate.send_mean <= TimeDelta::Zero() ||
+      cluster_aggregate.recv_mean <= TimeDelta::Zero()) {
+    return;
+  }
+
+  Cluster cluster;
+  cluster.send_mean = cluster_aggregate.send_mean / cluster_aggregate.count;
+  cluster.recv_mean = cluster_aggregate.recv_mean / cluster_aggregate.count;
+  cluster.mean_size = cluster_aggregate.mean_size / cluster_aggregate.count;
+  cluster.count = cluster_aggregate.count;
+  cluster.num_above_min_delta = cluster_aggregate.num_above_min_delta;
+  clusters.push_back(cluster);
+}
+
+RemoteBitrateEstimatorAbsSendTime::RemoteBitrateEstimatorAbsSendTime(
+    RemoteBitrateObserver* observer,
+    Clock* clock)
+    : clock_(clock),
+      observer_(observer),
+      detector_(&field_trials_),
+      remote_rate_(&field_trials_) {
+  RTC_DCHECK(clock_);
+  RTC_DCHECK(observer_);
+  RTC_LOG(LS_INFO) << "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
+}
+
+std::list<RemoteBitrateEstimatorAbsSendTime::Cluster>
+RemoteBitrateEstimatorAbsSendTime::ComputeClusters() const {
+  std::list<Cluster> clusters;
+  Cluster cluster_aggregate;
+  Timestamp prev_send_time = Timestamp::MinusInfinity();
+  Timestamp prev_recv_time = Timestamp::MinusInfinity();
+  for (const Probe& probe : probes_) {
+    if (prev_send_time.IsFinite()) {
+      TimeDelta send_delta = probe.send_time - prev_send_time;
+      TimeDelta recv_delta = probe.recv_time - prev_recv_time;
+      if (send_delta >= kMinClusterDelta && recv_delta >= kMinClusterDelta) {
+        ++cluster_aggregate.num_above_min_delta;
+      }
+      if (!IsWithinClusterBounds(send_delta, cluster_aggregate)) {
+        MaybeAddCluster(cluster_aggregate, clusters);
+        cluster_aggregate = Cluster();
+      }
+      cluster_aggregate.send_mean += send_delta;
+      cluster_aggregate.recv_mean += recv_delta;
+      cluster_aggregate.mean_size += probe.payload_size;
+      ++cluster_aggregate.count;
+    }
+    prev_send_time = probe.send_time;
+    prev_recv_time = probe.recv_time;
+  }
+  MaybeAddCluster(cluster_aggregate, clusters);
+  return clusters;
+}
+
+const RemoteBitrateEstimatorAbsSendTime::Cluster*
+RemoteBitrateEstimatorAbsSendTime::FindBestProbe(
+    const std::list<Cluster>& clusters) const {
+  DataRate highest_probe_bitrate = DataRate::Zero();
+  const Cluster* best = nullptr;
+  for (const auto& cluster : clusters) {
+    if (cluster.send_mean == TimeDelta::Zero() ||
+        cluster.recv_mean == TimeDelta::Zero()) {
+      continue;
+    }
+    if (cluster.num_above_min_delta > cluster.count / 2 &&
+        (cluster.recv_mean - cluster.send_mean <= TimeDelta::Millis(2) &&
+         cluster.send_mean - cluster.recv_mean <= TimeDelta::Millis(5))) {
+      DataRate probe_bitrate =
+          std::min(cluster.SendBitrate(), cluster.RecvBitrate());
+      if (probe_bitrate > highest_probe_bitrate) {
+        highest_probe_bitrate = probe_bitrate;
+        best = &cluster;
+      }
+    } else {
+      RTC_LOG(LS_INFO) << "Probe failed, sent at "
+                       << cluster.SendBitrate().bps() << " bps, received at "
+                       << cluster.RecvBitrate().bps()
+                       << " bps. Mean send delta: " << cluster.send_mean.ms()
+                       << " ms, mean recv delta: " << cluster.recv_mean.ms()
+                       << " ms, num probes: " << cluster.count;
+      break;
+    }
+  }
+  return best;
+}
+
+RemoteBitrateEstimatorAbsSendTime::ProbeResult
+RemoteBitrateEstimatorAbsSendTime::ProcessClusters(Timestamp now) {
+  std::list<Cluster> clusters = ComputeClusters();
+  if (clusters.empty()) {
+    // If we reach the max number of probe packets and still have no clusters,
+    // we will remove the oldest one.
+    if (probes_.size() >= kMaxProbePackets)
+      probes_.pop_front();
+    return ProbeResult::kNoUpdate;
+  }
+
+  if (const Cluster* best = FindBestProbe(clusters)) {
+    DataRate probe_bitrate = std::min(best->SendBitrate(), best->RecvBitrate());
+    // Make sure that a probe sent on a lower bitrate than our estimate can't
+    // reduce the estimate.
+    if (IsBitrateImproving(probe_bitrate)) {
+      RTC_LOG(LS_INFO) << "Probe successful, sent at "
+                       << best->SendBitrate().bps() << " bps, received at "
+                       << best->RecvBitrate().bps()
+                       << " bps. Mean send delta: " << best->send_mean.ms()
+                       << " ms, mean recv delta: " << best->recv_mean.ms()
+                       << " ms, num probes: " << best->count;
+      remote_rate_.SetEstimate(probe_bitrate, now);
+      return ProbeResult::kBitrateUpdated;
+    }
+  }
+
+  // Not probing and received non-probe packet, or finished with current set
+  // of probes.
+  if (clusters.size() >= kExpectedNumberOfProbes)
+    probes_.clear();
+  return ProbeResult::kNoUpdate;
+}
+
+bool RemoteBitrateEstimatorAbsSendTime::IsBitrateImproving(
+    DataRate probe_bitrate) const {
+  bool initial_probe =
+      !remote_rate_.ValidEstimate() && probe_bitrate > DataRate::Zero();
+  bool bitrate_above_estimate = remote_rate_.ValidEstimate() &&
+                                probe_bitrate > remote_rate_.LatestEstimate();
+  return initial_probe || bitrate_above_estimate;
+}
+
+void RemoteBitrateEstimatorAbsSendTime::IncomingPacket(
+    int64_t arrival_time_ms,
+    size_t payload_size,
+    const RTPHeader& header) {
+  RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
+  if (!header.extension.hasAbsoluteSendTime) {
+    RTC_LOG(LS_WARNING)
+        << "RemoteBitrateEstimatorAbsSendTimeImpl: Incoming packet "
+           "is missing absolute send time extension!";
+    return;
+  }
+  IncomingPacketInfo(Timestamp::Millis(arrival_time_ms),
+                     header.extension.absoluteSendTime,
+                     DataSize::Bytes(payload_size), header.ssrc);
+}
+
+void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
+    Timestamp arrival_time,
+    uint32_t send_time_24bits,
+    DataSize payload_size,
+    uint32_t ssrc) {
+  RTC_CHECK(send_time_24bits < (1ul << 24));
+  if (!uma_recorded_) {
+    RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram, BweNames::kReceiverAbsSendTime,
+                              BweNames::kBweNamesMax);
+    uma_recorded_ = true;
+  }
+  // Shift up send time to use the full 32 bits that inter_arrival works with,
+  // so wrapping works properly.
+  uint32_t timestamp = send_time_24bits << kAbsSendTimeInterArrivalUpshift;
+  Timestamp send_time =
+      Timestamp::Millis(static_cast<int64_t>(timestamp) * kTimestampToMs);
+
+  Timestamp now = clock_->CurrentTime();
+  // TODO(holmer): SSRCs are only needed for REMB, should be broken out from
+  // here.
+
+  // Check if incoming bitrate estimate is valid, and if it needs to be reset.
+  absl::optional<uint32_t> incoming_bitrate =
+      incoming_bitrate_.Rate(arrival_time.ms());
+  if (incoming_bitrate) {
+    incoming_bitrate_initialized_ = true;
+  } else if (incoming_bitrate_initialized_) {
+    // Incoming bitrate had a previous valid value, but now not enough data
+    // point are left within the current window. Reset incoming bitrate
+    // estimator so that the window size will only contain new data points.
+    incoming_bitrate_.Reset();
+    incoming_bitrate_initialized_ = false;
+  }
+  incoming_bitrate_.Update(payload_size.bytes(), arrival_time.ms());
+
+  if (first_packet_time_.IsInfinite()) {
+    first_packet_time_ = now;
+  }
+
+  uint32_t ts_delta = 0;
+  int64_t t_delta = 0;
+  int size_delta = 0;
+  bool update_estimate = false;
+  DataRate target_bitrate = DataRate::Zero();
+  std::vector<uint32_t> ssrcs;
+  {
+    MutexLock lock(&mutex_);
+
+    TimeoutStreams(now);
+    RTC_DCHECK(inter_arrival_);
+    RTC_DCHECK(estimator_);
+    ssrcs_.insert_or_assign(ssrc, now);
+
+    // For now only try to detect probes while we don't have a valid estimate.
+    // We currently assume that only packets larger than 200 bytes are paced by
+    // the sender.
+    static constexpr DataSize kMinProbePacketSize = DataSize::Bytes(200);
+    if (payload_size > kMinProbePacketSize &&
+        (!remote_rate_.ValidEstimate() ||
+         now - first_packet_time_ < kInitialProbingInterval)) {
+      // TODO(holmer): Use a map instead to get correct order?
+      if (total_probes_received_ < kMaxProbePackets) {
+        TimeDelta send_delta = TimeDelta::Millis(-1);
+        TimeDelta recv_delta = TimeDelta::Millis(-1);
+        if (!probes_.empty()) {
+          send_delta = send_time - probes_.back().send_time;
+          recv_delta = arrival_time - probes_.back().recv_time;
+        }
+        RTC_LOG(LS_INFO) << "Probe packet received: send time="
+                         << send_time.ms()
+                         << " ms, recv time=" << arrival_time.ms()
+                         << " ms, send delta=" << send_delta.ms()
+                         << " ms, recv delta=" << recv_delta.ms() << " ms.";
+      }
+      probes_.emplace_back(send_time, arrival_time, payload_size);
+      ++total_probes_received_;
+      // Make sure that a probe which updated the bitrate immediately has an
+      // effect by calling the OnReceiveBitrateChanged callback.
+      if (ProcessClusters(now) == ProbeResult::kBitrateUpdated)
+        update_estimate = true;
+    }
+    if (inter_arrival_->ComputeDeltas(timestamp, arrival_time.ms(), now.ms(),
+                                      payload_size.bytes(), &ts_delta, &t_delta,
+                                      &size_delta)) {
+      double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift);
+      estimator_->Update(t_delta, ts_delta_ms, size_delta, detector_.State(),
+                         arrival_time.ms());
+      detector_.Detect(estimator_->offset(), ts_delta_ms,
+                       estimator_->num_of_deltas(), arrival_time.ms());
+    }
+
+    if (!update_estimate) {
+      // Check if it's time for a periodic update or if we should update because
+      // of an over-use.
+      if (last_update_.IsInfinite() ||
+          now.ms() - last_update_.ms() >
+              remote_rate_.GetFeedbackInterval().ms()) {
+        update_estimate = true;
+      } else if (detector_.State() == BandwidthUsage::kBwOverusing) {
+        absl::optional<uint32_t> incoming_rate =
+            incoming_bitrate_.Rate(arrival_time.ms());
+        if (incoming_rate && remote_rate_.TimeToReduceFurther(
+                                 now, DataRate::BitsPerSec(*incoming_rate))) {
+          update_estimate = true;
+        }
+      }
+    }
+
+    if (update_estimate) {
+      // The first overuse should immediately trigger a new estimate.
+      // We also have to update the estimate immediately if we are overusing
+      // and the target bitrate is too high compared to what we are receiving.
+      const RateControlInput input(
+          detector_.State(), OptionalRateFromOptionalBps(
+                                 incoming_bitrate_.Rate(arrival_time.ms())));
+      target_bitrate = remote_rate_.Update(&input, now);
+      update_estimate = remote_rate_.ValidEstimate();
+      ssrcs = Keys(ssrcs_);
+    }
+  }
+  if (update_estimate) {
+    last_update_ = now;
+    observer_->OnReceiveBitrateChanged(ssrcs, target_bitrate.bps<uint32_t>());
+  }
+}
+
+TimeDelta RemoteBitrateEstimatorAbsSendTime::Process() {
+  return TimeDelta::PlusInfinity();
+}
+
+void RemoteBitrateEstimatorAbsSendTime::TimeoutStreams(Timestamp now) {
+  for (auto it = ssrcs_.begin(); it != ssrcs_.end();) {
+    if (now - it->second > TimeDelta::Millis(kStreamTimeOutMs)) {
+      ssrcs_.erase(it++);
+    } else {
+      ++it;
+    }
+  }
+  if (ssrcs_.empty()) {
+    // We can't update the estimate if we don't have any active streams.
+    inter_arrival_ = std::make_unique<InterArrival>(
+        (kTimestampGroupLengthMs << kInterArrivalShift) / 1000, kTimestampToMs,
+        true);
+    estimator_ = std::make_unique<OveruseEstimator>(OverUseDetectorOptions());
+    // We deliberately don't reset the first_packet_time_ms_ here for now since
+    // we only probe for bandwidth in the beginning of a call right now.
+  }
+}
+
+void RemoteBitrateEstimatorAbsSendTime::OnRttUpdate(int64_t avg_rtt_ms,
+                                                    int64_t /*max_rtt_ms*/) {
+  MutexLock lock(&mutex_);
+  remote_rate_.SetRtt(TimeDelta::Millis(avg_rtt_ms));
+}
+
+void RemoteBitrateEstimatorAbsSendTime::RemoveStream(uint32_t ssrc) {
+  MutexLock lock(&mutex_);
+  ssrcs_.erase(ssrc);
+}
+
+DataRate RemoteBitrateEstimatorAbsSendTime::LatestEstimate() const {
+  // Currently accessed only from the worker thread (see Call::GetStats()).
+  MutexLock lock(&mutex_);
+  if (!remote_rate_.ValidEstimate() || ssrcs_.empty()) {
+    return DataRate::Zero();
+  }
+  return remote_rate_.LatestEstimate();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
new file mode 100644
index 0000000000..fd33c84b04
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_ABS_SEND_TIME_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_ABS_SEND_TIME_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "api/rtp_headers.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "modules/remote_bitrate_estimator/inter_arrival.h"
+#include "modules/remote_bitrate_estimator/overuse_detector.h"
+#include "modules/remote_bitrate_estimator/overuse_estimator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
+ public:
+  RemoteBitrateEstimatorAbsSendTime(RemoteBitrateObserver* observer,
+                                    Clock* clock);
+
+  RemoteBitrateEstimatorAbsSendTime() = delete;
+  RemoteBitrateEstimatorAbsSendTime(const RemoteBitrateEstimatorAbsSendTime&) =
+      delete;
+  RemoteBitrateEstimatorAbsSendTime& operator=(
+      const RemoteBitrateEstimatorAbsSendTime&) = delete;
+
+  ~RemoteBitrateEstimatorAbsSendTime() override;
+
+  void IncomingPacket(int64_t arrival_time_ms,
+                      size_t payload_size,
+                      const RTPHeader& header) override;
+  TimeDelta Process() override;
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+  void RemoveStream(uint32_t ssrc) override;
+  DataRate LatestEstimate() const override;
+
+ private:
+  struct Probe {
+    Probe(Timestamp send_time, Timestamp recv_time, DataSize payload_size)
+        : send_time(send_time),
+          recv_time(recv_time),
+          payload_size(payload_size) {}
+
+    Timestamp send_time;
+    Timestamp recv_time;
+    DataSize payload_size;
+  };
+
+  struct Cluster {
+    DataRate SendBitrate() const { return mean_size / send_mean; }
+    DataRate RecvBitrate() const { return mean_size / recv_mean; }
+
+    TimeDelta send_mean = TimeDelta::Zero();
+    TimeDelta recv_mean = TimeDelta::Zero();
+    // TODO(holmer): Add some variance metric as well?
+    DataSize mean_size = DataSize::Zero();
+    int count = 0;
+    int num_above_min_delta = 0;
+  };
+
+  enum class ProbeResult { kBitrateUpdated, kNoUpdate };
+
+  static bool IsWithinClusterBounds(TimeDelta send_delta,
+                                    const Cluster& cluster_aggregate);
+
+  static void MaybeAddCluster(const Cluster& cluster_aggregate,
+                              std::list<Cluster>& clusters);
+
+  void IncomingPacketInfo(Timestamp arrival_time,
+                          uint32_t send_time_24bits,
+                          DataSize payload_size,
+                          uint32_t ssrc);
+
+  std::list<Cluster> ComputeClusters() const;
+
+  const Cluster* FindBestProbe(const std::list<Cluster>& clusters) const;
+
+  // Returns true if a probe which changed the estimate was detected.
+  ProbeResult ProcessClusters(Timestamp now)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(&mutex_);
+
+  bool IsBitrateImproving(DataRate probe_bitrate) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(&mutex_);
+
+  void TimeoutStreams(Timestamp now) RTC_EXCLUSIVE_LOCKS_REQUIRED(&mutex_);
+
+  rtc::RaceChecker network_race_;
+  Clock* const clock_;
+  const FieldTrialBasedConfig field_trials_;
+  RemoteBitrateObserver* const observer_;
+  std::unique_ptr<InterArrival> inter_arrival_;
+  std::unique_ptr<OveruseEstimator> estimator_;
+  OveruseDetector detector_;
+  RateStatistics incoming_bitrate_{kBitrateWindowMs, 8000};
+  bool incoming_bitrate_initialized_ = false;
+  std::list<Probe> probes_;
+  size_t total_probes_received_ = 0;
+  Timestamp first_packet_time_ = Timestamp::MinusInfinity();
+  Timestamp last_update_ = Timestamp::MinusInfinity();
+  bool uma_recorded_ = false;
+
+  mutable Mutex mutex_;
+  std::map<uint32_t, Timestamp> ssrcs_ RTC_GUARDED_BY(&mutex_);
+  AimdRateControl remote_rate_ RTC_GUARDED_BY(&mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_ABS_SEND_TIME_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
new file mode 100644
index 0000000000..d8ef23cc92
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
@@ -0,0 +1,296 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class RemoteBitrateEstimatorAbsSendTimeTest
+    : public RemoteBitrateEstimatorTest {
+ public:
+  RemoteBitrateEstimatorAbsSendTimeTest() {}
+
+  RemoteBitrateEstimatorAbsSendTimeTest(
+      const RemoteBitrateEstimatorAbsSendTimeTest&) = delete;
+  RemoteBitrateEstimatorAbsSendTimeTest& operator=(
+      const RemoteBitrateEstimatorAbsSendTimeTest&) = delete;
+
+  virtual void SetUp() {
+    bitrate_estimator_.reset(new RemoteBitrateEstimatorAbsSendTime(
+        bitrate_observer_.get(), &clock_));
+  }
+
+ protected:
+};
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, InitialBehavior) {
+  InitialBehaviorTestHelper(674840);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, RateIncreaseReordering) {
+  RateIncreaseReorderingTestHelper(674840);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, RateIncreaseRtpTimestamps) {
+  RateIncreaseRtpTimestampsTestHelper(1237);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropOneStream) {
+  CapacityDropTestHelper(1, false, 633, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropPosOffsetChange) {
+  CapacityDropTestHelper(1, false, 267, 30000);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropNegOffsetChange) {
+  CapacityDropTestHelper(1, false, 267, -30000);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropOneStreamWrap) {
+  CapacityDropTestHelper(1, true, 633, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropTwoStreamsWrap) {
+  CapacityDropTestHelper(2, true, 700, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropThreeStreamsWrap) {
+  CapacityDropTestHelper(3, true, 633, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropThirteenStreamsWrap) {
+  CapacityDropTestHelper(13, true, 667, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropNineteenStreamsWrap) {
+  CapacityDropTestHelper(19, true, 667, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, CapacityDropThirtyStreamsWrap) {
+  CapacityDropTestHelper(30, true, 667, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestTimestampGrouping) {
+  TestTimestampGroupingTestHelper();
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestShortTimeoutAndWrap) {
+  // Simulate a client leaving and rejoining the call after 35 seconds. This
+  // will make abs send time wrap, so if streams aren't timed out properly
+  // the next 30 seconds of packets will be out of order.
+  TestWrappingHelper(35);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestLongTimeoutAndWrap) {
+  // Simulate a client leaving and rejoining the call after some multiple of
+  // 64 seconds later. This will cause a zero difference in abs send times due
+  // to the wrap, but a big difference in arrival time, if streams aren't
+  // properly timed out.
+  TestWrappingHelper(10 * 64);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProcessAfterTimeout) {
+  // This time constant must be equal to the ones defined for the
+  // RemoteBitrateEstimator.
+  const int64_t kStreamTimeOutMs = 2000;
+  const int64_t kProcessIntervalMs = 1000;
+  IncomingPacket(0, 1000, clock_.TimeInMilliseconds(), 0, 0);
+  clock_.AdvanceTimeMilliseconds(kStreamTimeOutMs + 1);
+  // Trigger timeout.
+  bitrate_estimator_->Process();
+  clock_.AdvanceTimeMilliseconds(kProcessIntervalMs);
+  // This shouldn't crash.
+  bitrate_estimator_->Process();
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetection) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(10);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
+  }
+
+  // Second burst sent at 8 * 1000 / 5 = 1600 kbps.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
+  }
+
+  bitrate_estimator_->Process();
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_GT(bitrate_observer_->latest_bitrate(), 1500000u);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
+       TestProbeDetectionNonPacedPackets) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps, but with every other packet
+  // not being paced which could mess things up.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
+    // Non-paced packet, arriving 5 ms after.
+    clock_.AdvanceTimeMilliseconds(5);
+    IncomingPacket(0, 100, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
+  }
+
+  bitrate_estimator_->Process();
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_GT(bitrate_observer_->latest_bitrate(), 800000u);
+}
+
+// Packets will require 5 ms to be transmitted to the receiver, causing packets
+// of the second probe to be dispersed.
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
+       TestProbeDetectionTooHighBitrate) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  int64_t send_time_ms = 0;
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(10);
+    now_ms = clock_.TimeInMilliseconds();
+    send_time_ms += 10;
+    IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
+                   AbsSendTime(send_time_ms, 1000));
+  }
+
+  // Second burst sent at 8 * 1000 / 5 = 1600 kbps, arriving at 8 * 1000 / 8 =
+  // 1000 kbps.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(8);
+    now_ms = clock_.TimeInMilliseconds();
+    send_time_ms += 5;
+    IncomingPacket(0, 1000, now_ms, send_time_ms,
+                   AbsSendTime(send_time_ms, 1000));
+  }
+
+  bitrate_estimator_->Process();
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_NEAR(bitrate_observer_->latest_bitrate(), 800000u, 10000);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
+       TestProbeDetectionSlightlyFasterArrival) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  // Arriving at 8 * 1000 / 5 = 1600 kbps.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    send_time_ms += 10;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
+                   AbsSendTime(send_time_ms, 1000));
+  }
+
+  bitrate_estimator_->Process();
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_GT(bitrate_observer_->latest_bitrate(), 800000u);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetectionFasterArrival) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  // Arriving at 8 * 1000 / 5 = 1600 kbps.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(1);
+    send_time_ms += 10;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
+                   AbsSendTime(send_time_ms, 1000));
+  }
+
+  bitrate_estimator_->Process();
+  EXPECT_FALSE(bitrate_observer_->updated());
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, TestProbeDetectionSlowerArrival) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 5 = 1600 kbps.
+  // Arriving at 8 * 1000 / 7 = 1142 kbps.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(7);
+    send_time_ms += 5;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
+                   AbsSendTime(send_time_ms, 1000));
+  }
+
+  bitrate_estimator_->Process();
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_NEAR(bitrate_observer_->latest_bitrate(), 1140000, 10000);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest,
+       TestProbeDetectionSlowerArrivalHighBitrate) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // Burst sent at 8 * 1000 / 1 = 8000 kbps.
+  // Arriving at 8 * 1000 / 2 = 4000 kbps.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(2);
+    send_time_ms += 1;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * send_time_ms,
+                   AbsSendTime(send_time_ms, 1000));
+  }
+
+  bitrate_estimator_->Process();
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_NEAR(bitrate_observer_->latest_bitrate(), 4000000u, 10000);
+}
+
+TEST_F(RemoteBitrateEstimatorAbsSendTimeTest, ProbingIgnoresSmallPackets) {
+  const int kProbeLength = 5;
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // Probing with 200 bytes every 10 ms, should be ignored by the probe
+  // detection.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(10);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 200, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
+  }
+
+  bitrate_estimator_->Process();
+  EXPECT_FALSE(bitrate_observer_->updated());
+
+  // Followed by a probe with 1000 bytes packets, should be detected as a
+  // probe.
+  for (int i = 0; i < kProbeLength; ++i) {
+    clock_.AdvanceTimeMilliseconds(10);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000));
+  }
+
+  // Wait long enough so that we can call Process again.
+  clock_.AdvanceTimeMilliseconds(1000);
+
+  bitrate_estimator_->Process();
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_NEAR(bitrate_observer_->latest_bitrate(), 800000u, 10000);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_gn/moz.build b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_gn/moz.build
new file mode 100644
index 0000000000..e572e90c31
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_gn/moz.build
@@ -0,0 +1,244 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc",
+    "/third_party/libwebrtc/modules/remote_bitrate_estimator/bwe_defines.cc",
+    "/third_party/libwebrtc/modules/remote_bitrate_estimator/inter_arrival.cc",
+    "/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_detector.cc",
+    "/third_party/libwebrtc/modules/remote_bitrate_estimator/overuse_estimator.cc",
+    "/third_party/libwebrtc/modules/remote_bitrate_estimator/packet_arrival_map.cc",
+    "/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc",
+    "/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("remote_bitrate_estimator_gn")
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
new file mode 100644
index 0000000000..8f15912a49
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
@@ -0,0 +1,245 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
+
+
+#include <cstdint>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/inter_arrival.h"
+#include "modules/remote_bitrate_estimator/overuse_detector.h"
+#include "modules/remote_bitrate_estimator/overuse_estimator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+absl::optional<DataRate> OptionalRateFromOptionalBps(
+    absl::optional<int> bitrate_bps) {
+  if (bitrate_bps) {
+    return DataRate::BitsPerSec(*bitrate_bps);
+  } else {
+    return absl::nullopt;
+  }
+}
+}  // namespace
+
+enum { kTimestampGroupLengthMs = 5 };
+static const double kTimestampToMs = 1.0 / 90.0;
+
+struct RemoteBitrateEstimatorSingleStream::Detector {
+  explicit Detector(int64_t last_packet_time_ms,
+                    const OverUseDetectorOptions& options,
+                    bool enable_burst_grouping,
+                    const FieldTrialsView* key_value_config)
+      : last_packet_time_ms(last_packet_time_ms),
+        inter_arrival(90 * kTimestampGroupLengthMs,
+                      kTimestampToMs,
+                      enable_burst_grouping),
+        estimator(options),
+        detector(key_value_config) {}
+  int64_t last_packet_time_ms;
+  InterArrival inter_arrival;
+  OveruseEstimator estimator;
+  OveruseDetector detector;
+};
+
+RemoteBitrateEstimatorSingleStream::RemoteBitrateEstimatorSingleStream(
+    RemoteBitrateObserver* observer,
+    Clock* clock)
+    : clock_(clock),
+      incoming_bitrate_(kBitrateWindowMs, 8000),
+      last_valid_incoming_bitrate_(0),
+      remote_rate_(&field_trials_),
+      observer_(observer),
+      last_process_time_(-1),
+      process_interval_ms_(kProcessIntervalMs),
+      uma_recorded_(false) {
+  RTC_LOG(LS_INFO) << "RemoteBitrateEstimatorSingleStream: Instantiating.";
+}
+
+RemoteBitrateEstimatorSingleStream::~RemoteBitrateEstimatorSingleStream() {
+  while (!overuse_detectors_.empty()) {
+    SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.begin();
+    delete it->second;
+    overuse_detectors_.erase(it);
+  }
+}
+
+void RemoteBitrateEstimatorSingleStream::IncomingPacket(
+    int64_t arrival_time_ms,
+    size_t payload_size,
+    const RTPHeader& header) {
+  if (!uma_recorded_) {
+    BweNames type = BweNames::kReceiverTOffset;
+    if (!header.extension.hasTransmissionTimeOffset)
+      type = BweNames::kReceiverNoExtension;
+    RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram, type, BweNames::kBweNamesMax);
+    uma_recorded_ = true;
+  }
+  uint32_t ssrc = header.ssrc;
+  uint32_t rtp_timestamp =
+      header.timestamp + header.extension.transmissionTimeOffset;
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  MutexLock lock(&mutex_);
+  SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.find(ssrc);
+  if (it == overuse_detectors_.end()) {
+    // This is a new SSRC. Adding to map.
+    // TODO(holmer): If the channel changes SSRC the old SSRC will still be
+    // around in this map until the channel is deleted. This is OK since the
+    // callback will no longer be called for the old SSRC. This will be
+    // automatically cleaned up when we have one RemoteBitrateEstimator per REMB
+    // group.
+    std::pair<SsrcOveruseEstimatorMap::iterator, bool> insert_result =
+        overuse_detectors_.insert(
+            std::make_pair(ssrc, new Detector(now_ms, OverUseDetectorOptions(),
+                                              true, &field_trials_)));
+    it = insert_result.first;
+  }
+  Detector* estimator = it->second;
+  estimator->last_packet_time_ms = now_ms;
+
+  // Check if incoming bitrate estimate is valid, and if it needs to be reset.
+  absl::optional<uint32_t> incoming_bitrate = incoming_bitrate_.Rate(now_ms);
+  if (incoming_bitrate) {
+    last_valid_incoming_bitrate_ = *incoming_bitrate;
+  } else if (last_valid_incoming_bitrate_ > 0) {
+    // Incoming bitrate had a previous valid value, but now not enough data
+    // point are left within the current window. Reset incoming bitrate
+    // estimator so that the window size will only contain new data points.
+    incoming_bitrate_.Reset();
+    last_valid_incoming_bitrate_ = 0;
+  }
+  incoming_bitrate_.Update(payload_size, now_ms);
+
+  const BandwidthUsage prior_state = estimator->detector.State();
+  uint32_t timestamp_delta = 0;
+  int64_t time_delta = 0;
+  int size_delta = 0;
+  if (estimator->inter_arrival.ComputeDeltas(
+          rtp_timestamp, arrival_time_ms, now_ms, payload_size,
+          &timestamp_delta, &time_delta, &size_delta)) {
+    double timestamp_delta_ms = timestamp_delta * kTimestampToMs;
+    estimator->estimator.Update(time_delta, timestamp_delta_ms, size_delta,
+                                estimator->detector.State(), now_ms);
+    estimator->detector.Detect(estimator->estimator.offset(),
+                               timestamp_delta_ms,
+                               estimator->estimator.num_of_deltas(), now_ms);
+  }
+  if (estimator->detector.State() == BandwidthUsage::kBwOverusing) {
+    absl::optional<uint32_t> incoming_bitrate_bps =
+        incoming_bitrate_.Rate(now_ms);
+    if (incoming_bitrate_bps &&
+        (prior_state != BandwidthUsage::kBwOverusing ||
+         remote_rate_.TimeToReduceFurther(
+             Timestamp::Millis(now_ms),
+             DataRate::BitsPerSec(*incoming_bitrate_bps)))) {
+      // The first overuse should immediately trigger a new estimate.
+      // We also have to update the estimate immediately if we are overusing
+      // and the target bitrate is too high compared to what we are receiving.
+      UpdateEstimate(now_ms);
+    }
+  }
+}
+
+TimeDelta RemoteBitrateEstimatorSingleStream::Process() {
+  MutexLock lock(&mutex_);
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  int64_t next_process_time_ms = last_process_time_ + process_interval_ms_;
+  if (last_process_time_ == -1 || now_ms >= next_process_time_ms) {
+    UpdateEstimate(now_ms);
+    last_process_time_ = now_ms;
+    return TimeDelta::Millis(process_interval_ms_);
+  }
+
+  return TimeDelta::Millis(next_process_time_ms - now_ms);
+}
+
+void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) {
+  BandwidthUsage bw_state = BandwidthUsage::kBwNormal;
+  SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.begin();
+  while (it != overuse_detectors_.end()) {
+    const int64_t time_of_last_received_packet =
+        it->second->last_packet_time_ms;
+    if (time_of_last_received_packet >= 0 &&
+        now_ms - time_of_last_received_packet > kStreamTimeOutMs) {
+      // This over-use detector hasn't received packets for `kStreamTimeOutMs`
+      // milliseconds and is considered stale.
+      delete it->second;
+      overuse_detectors_.erase(it++);
+    } else {
+      // Make sure that we trigger an over-use if any of the over-use detectors
+      // is detecting over-use.
+      if (it->second->detector.State() > bw_state) {
+        bw_state = it->second->detector.State();
+      }
+      ++it;
+    }
+  }
+  // We can't update the estimate if we don't have any active streams.
+  if (overuse_detectors_.empty()) {
+    return;
+  }
+
+  const RateControlInput input(
+      bw_state, OptionalRateFromOptionalBps(incoming_bitrate_.Rate(now_ms)));
+  uint32_t target_bitrate =
+      remote_rate_.Update(&input, Timestamp::Millis(now_ms)).bps<uint32_t>();
+  if (remote_rate_.ValidEstimate()) {
+    process_interval_ms_ = remote_rate_.GetFeedbackInterval().ms();
+    RTC_DCHECK_GT(process_interval_ms_, 0);
+    std::vector<uint32_t> ssrcs;
+    GetSsrcs(&ssrcs);
+    if (observer_)
+      observer_->OnReceiveBitrateChanged(ssrcs, target_bitrate);
+  }
+}
+
+void RemoteBitrateEstimatorSingleStream::OnRttUpdate(int64_t avg_rtt_ms,
+                                                     int64_t max_rtt_ms) {
+  MutexLock lock(&mutex_);
+  remote_rate_.SetRtt(TimeDelta::Millis(avg_rtt_ms));
+}
+
+void RemoteBitrateEstimatorSingleStream::RemoveStream(unsigned int ssrc) {
+  MutexLock lock(&mutex_);
+  SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.find(ssrc);
+  if (it != overuse_detectors_.end()) {
+    delete it->second;
+    overuse_detectors_.erase(it);
+  }
+}
+
+DataRate RemoteBitrateEstimatorSingleStream::LatestEstimate() const {
+  MutexLock lock(&mutex_);
+  if (!remote_rate_.ValidEstimate() || overuse_detectors_.empty()) {
+    return DataRate::Zero();
+  }
+  return remote_rate_.LatestEstimate();
+}
+
+void RemoteBitrateEstimatorSingleStream::GetSsrcs(
+    std::vector<uint32_t>* ssrcs) const {
+  RTC_DCHECK(ssrcs);
+  ssrcs->resize(overuse_detectors_.size());
+  int i = 0;
+  for (SsrcOveruseEstimatorMap::const_iterator it = overuse_detectors_.begin();
+       it != overuse_detectors_.end(); ++it, ++i) {
+    (*ssrcs)[i] = it->first;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
new file mode 100644
index 0000000000..699f259d48
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_SINGLE_STREAM_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_SINGLE_STREAM_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+#include <vector>
+
+#include "api/transport/field_trial_based_config.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class Clock;
+struct RTPHeader;
+
+class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator {
+ public:
+  RemoteBitrateEstimatorSingleStream(RemoteBitrateObserver* observer,
+                                     Clock* clock);
+
+  RemoteBitrateEstimatorSingleStream() = delete;
+  RemoteBitrateEstimatorSingleStream(
+      const RemoteBitrateEstimatorSingleStream&) = delete;
+  RemoteBitrateEstimatorSingleStream& operator=(
+      const RemoteBitrateEstimatorSingleStream&) = delete;
+
+  ~RemoteBitrateEstimatorSingleStream() override;
+
+  void IncomingPacket(int64_t arrival_time_ms,
+                      size_t payload_size,
+                      const RTPHeader& header) override;
+  TimeDelta Process() override;
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+  void RemoveStream(uint32_t ssrc) override;
+  DataRate LatestEstimate() const override;
+
+ private:
+  struct Detector;
+
+  typedef std::map<uint32_t, Detector*> SsrcOveruseEstimatorMap;
+
+  // Triggers a new estimate calculation.
+  void UpdateEstimate(int64_t time_now) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  void GetSsrcs(std::vector<uint32_t>* ssrcs) const
+      RTC_SHARED_LOCKS_REQUIRED(mutex_);
+
+  Clock* const clock_;
+  const FieldTrialBasedConfig field_trials_;
+  SsrcOveruseEstimatorMap overuse_detectors_ RTC_GUARDED_BY(mutex_);
+  RateStatistics incoming_bitrate_ RTC_GUARDED_BY(mutex_);
+  uint32_t last_valid_incoming_bitrate_ RTC_GUARDED_BY(mutex_);
+  AimdRateControl remote_rate_ RTC_GUARDED_BY(mutex_);
+  RemoteBitrateObserver* const observer_ RTC_GUARDED_BY(mutex_);
+  mutable Mutex mutex_;
+  int64_t last_process_time_;
+  int64_t process_interval_ms_ RTC_GUARDED_BY(mutex_);
+  bool uma_recorded_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_SINGLE_STREAM_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
new file mode 100644
index 0000000000..64ef39d935
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
+
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class RemoteBitrateEstimatorSingleTest : public RemoteBitrateEstimatorTest {
+ public:
+  RemoteBitrateEstimatorSingleTest() {}
+
+  RemoteBitrateEstimatorSingleTest(const RemoteBitrateEstimatorSingleTest&) =
+      delete;
+  RemoteBitrateEstimatorSingleTest& operator=(
+      const RemoteBitrateEstimatorSingleTest&) = delete;
+
+  virtual void SetUp() {
+    bitrate_estimator_.reset(new RemoteBitrateEstimatorSingleStream(
+        bitrate_observer_.get(), &clock_));
+  }
+
+ protected:
+};
+
+TEST_F(RemoteBitrateEstimatorSingleTest, InitialBehavior) {
+  InitialBehaviorTestHelper(508017);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, RateIncreaseReordering) {
+  RateIncreaseReorderingTestHelper(506422);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, RateIncreaseRtpTimestamps) {
+  RateIncreaseRtpTimestampsTestHelper(1267);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropOneStream) {
+  CapacityDropTestHelper(1, false, 633, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropOneStreamWrap) {
+  CapacityDropTestHelper(1, true, 633, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropTwoStreamsWrap) {
+  CapacityDropTestHelper(2, true, 767, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThreeStreamsWrap) {
+  CapacityDropTestHelper(3, true, 567, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThirteenStreamsWrap) {
+  CapacityDropTestHelper(13, true, 567, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropNineteenStreamsWrap) {
+  CapacityDropTestHelper(19, true, 700, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThirtyStreamsWrap) {
+  CapacityDropTestHelper(30, true, 733, 0);
+}
+
+TEST_F(RemoteBitrateEstimatorSingleTest, TestTimestampGrouping) {
+  TestTimestampGroupingTestHelper();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
new file mode 100644
index 0000000000..899037f5a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
@@ -0,0 +1,594 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
+
+#include <algorithm>
+#include <limits>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+const size_t kMtu = 1200;
+const uint32_t kAcceptedBitrateErrorBps = 50000;
+
+// Number of packets needed before we have a valid estimate.
+const int kNumInitialPackets = 2;
+
+namespace testing {
+
+void TestBitrateObserver::OnReceiveBitrateChanged(
+    const std::vector<uint32_t>& ssrcs,
+    uint32_t bitrate) {
+  latest_bitrate_ = bitrate;
+  updated_ = true;
+}
+
+RtpStream::RtpStream(int fps,
+                     int bitrate_bps,
+                     uint32_t ssrc,
+                     uint32_t frequency,
+                     uint32_t timestamp_offset,
+                     int64_t rtcp_receive_time)
+    : fps_(fps),
+      bitrate_bps_(bitrate_bps),
+      ssrc_(ssrc),
+      frequency_(frequency),
+      next_rtp_time_(0),
+      next_rtcp_time_(rtcp_receive_time),
+      rtp_timestamp_offset_(timestamp_offset),
+      kNtpFracPerMs(4.294967296E6) {
+  RTC_DCHECK_GT(fps_, 0);
+}
+
+void RtpStream::set_rtp_timestamp_offset(uint32_t offset) {
+  rtp_timestamp_offset_ = offset;
+}
+
+// Generates a new frame for this stream. If called too soon after the
+// previous frame, no frame will be generated. The frame is split into
+// packets.
+int64_t RtpStream::GenerateFrame(int64_t time_now_us, PacketList* packets) {
+  if (time_now_us < next_rtp_time_) {
+    return next_rtp_time_;
+  }
+  RTC_DCHECK(packets);
+  size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
+  size_t n_packets =
+      std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
+  size_t packet_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
+  for (size_t i = 0; i < n_packets; ++i) {
+    RtpPacket* packet = new RtpPacket;
+    packet->send_time = time_now_us + kSendSideOffsetUs;
+    packet->size = packet_size;
+    packet->rtp_timestamp =
+        rtp_timestamp_offset_ +
+        static_cast<uint32_t>(((frequency_ / 1000) * packet->send_time + 500) /
+                              1000);
+    packet->ssrc = ssrc_;
+    packets->push_back(packet);
+  }
+  next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
+  return next_rtp_time_;
+}
+
+// The send-side time when the next frame can be generated.
+int64_t RtpStream::next_rtp_time() const {
+  return next_rtp_time_;
+}
+
+// Generates an RTCP packet.
+RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) {
+  if (time_now_us < next_rtcp_time_) {
+    return NULL;
+  }
+  RtcpPacket* rtcp = new RtcpPacket;
+  int64_t send_time_us = time_now_us + kSendSideOffsetUs;
+  rtcp->timestamp =
+      rtp_timestamp_offset_ +
+      static_cast<uint32_t>(((frequency_ / 1000) * send_time_us + 500) / 1000);
+  rtcp->ntp_secs = send_time_us / 1000000;
+  rtcp->ntp_frac =
+      static_cast<int64_t>((send_time_us % 1000000) * kNtpFracPerMs);
+  rtcp->ssrc = ssrc_;
+  next_rtcp_time_ = time_now_us + kRtcpIntervalUs;
+  return rtcp;
+}
+
+void RtpStream::set_bitrate_bps(int bitrate_bps) {
+  ASSERT_GE(bitrate_bps, 0);
+  bitrate_bps_ = bitrate_bps;
+}
+
+int RtpStream::bitrate_bps() const {
+  return bitrate_bps_;
+}
+
+uint32_t RtpStream::ssrc() const {
+  return ssrc_;
+}
+
+bool RtpStream::Compare(const std::pair<uint32_t, RtpStream*>& left,
+                        const std::pair<uint32_t, RtpStream*>& right) {
+  return left.second->next_rtp_time_ < right.second->next_rtp_time_;
+}
+
+StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
+    : capacity_(capacity), prev_arrival_time_us_(time_now) {}
+
+StreamGenerator::~StreamGenerator() {
+  for (StreamMap::iterator it = streams_.begin(); it != streams_.end(); ++it) {
+    delete it->second;
+  }
+  streams_.clear();
+}
+
+// Add a new stream.
+void StreamGenerator::AddStream(RtpStream* stream) {
+  streams_[stream->ssrc()] = stream;
+}
+
+// Set the link capacity.
+void StreamGenerator::set_capacity_bps(int capacity_bps) {
+  ASSERT_GT(capacity_bps, 0);
+  capacity_ = capacity_bps;
+}
+
+// Divides `bitrate_bps` among all streams. The allocated bitrate per stream
+// is decided by the current allocation ratios.
+void StreamGenerator::SetBitrateBps(int bitrate_bps) {
+  ASSERT_GE(streams_.size(), 0u);
+  int total_bitrate_before = 0;
+  for (StreamMap::iterator it = streams_.begin(); it != streams_.end(); ++it) {
+    total_bitrate_before += it->second->bitrate_bps();
+  }
+  int64_t bitrate_before = 0;
+  int total_bitrate_after = 0;
+  for (StreamMap::iterator it = streams_.begin(); it != streams_.end(); ++it) {
+    bitrate_before += it->second->bitrate_bps();
+    int64_t bitrate_after =
+        (bitrate_before * bitrate_bps + total_bitrate_before / 2) /
+        total_bitrate_before;
+    it->second->set_bitrate_bps(bitrate_after - total_bitrate_after);
+    total_bitrate_after += it->second->bitrate_bps();
+  }
+  ASSERT_EQ(bitrate_before, total_bitrate_before);
+  EXPECT_EQ(total_bitrate_after, bitrate_bps);
+}
+
+// Set the RTP timestamp offset for the stream identified by `ssrc`.
+void StreamGenerator::set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset) {
+  streams_[ssrc]->set_rtp_timestamp_offset(offset);
+}
+
+// TODO(holmer): Break out the channel simulation part from this class to make
+// it possible to simulate different types of channels.
+int64_t StreamGenerator::GenerateFrame(RtpStream::PacketList* packets,
+                                       int64_t time_now_us) {
+  RTC_DCHECK(packets);
+  RTC_DCHECK(packets->empty());
+  RTC_DCHECK_GT(capacity_, 0);
+  StreamMap::iterator it =
+      std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+  (*it).second->GenerateFrame(time_now_us, packets);
+  for (RtpStream::PacketList::iterator packet_it = packets->begin();
+       packet_it != packets->end(); ++packet_it) {
+    int capacity_bpus = capacity_ / 1000;
+    int64_t required_network_time_us =
+        (8 * 1000 * (*packet_it)->size + capacity_bpus / 2) / capacity_bpus;
+    prev_arrival_time_us_ =
+        std::max(time_now_us + required_network_time_us,
+                 prev_arrival_time_us_ + required_network_time_us);
+    (*packet_it)->arrival_time = prev_arrival_time_us_;
+  }
+  it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+  return std::max((*it).second->next_rtp_time(), time_now_us);
+}
+}  // namespace testing
+
+RemoteBitrateEstimatorTest::RemoteBitrateEstimatorTest()
+    : clock_(100000000),
+      bitrate_observer_(new testing::TestBitrateObserver),
+      stream_generator_(
+          new testing::StreamGenerator(1e6,  // Capacity.
+                                       clock_.TimeInMicroseconds())),
+      arrival_time_offset_ms_(0) {}
+
+RemoteBitrateEstimatorTest::~RemoteBitrateEstimatorTest() {}
+
+void RemoteBitrateEstimatorTest::AddDefaultStream() {
+  stream_generator_->AddStream(
+      new testing::RtpStream(30,          // Frames per second.
+                             3e5,         // Bitrate.
+                             1,           // SSRC.
+                             90000,       // RTP frequency.
+                             0xFFFFF000,  // Timestamp offset.
+                             0));         // RTCP receive time.
+}
+
+uint32_t RemoteBitrateEstimatorTest::AbsSendTime(int64_t t, int64_t denom) {
+  return (((t << 18) + (denom >> 1)) / denom) & 0x00fffffful;
+}
+
+uint32_t RemoteBitrateEstimatorTest::AddAbsSendTime(uint32_t t1, uint32_t t2) {
+  return (t1 + t2) & 0x00fffffful;
+}
+
+const uint32_t RemoteBitrateEstimatorTest::kDefaultSsrc = 1;
+
+void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc,
+                                                size_t payload_size,
+                                                int64_t arrival_time,
+                                                uint32_t rtp_timestamp,
+                                                uint32_t absolute_send_time) {
+  RTPHeader header;
+  memset(&header, 0, sizeof(header));
+  header.ssrc = ssrc;
+  header.timestamp = rtp_timestamp;
+  header.extension.hasAbsoluteSendTime = true;
+  header.extension.absoluteSendTime = absolute_send_time;
+  RTC_CHECK_GE(arrival_time + arrival_time_offset_ms_, 0);
+  bitrate_estimator_->IncomingPacket(arrival_time + arrival_time_offset_ms_,
+                                     payload_size, header);
+}
+
+// Generates a frame of packets belonging to a stream at a given bitrate and
+// with a given ssrc. The stream is pushed through a very simple simulated
+// network, and is then given to the receive-side bandwidth estimator.
+// Returns true if an over-use was seen, false otherwise.
+// The StreamGenerator::updated() should be used to check for any changes in
+// target bitrate after the call to this function.
+bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(uint32_t ssrc,
+                                                         uint32_t bitrate_bps) {
+  RTC_DCHECK_GT(bitrate_bps, 0);
+  stream_generator_->SetBitrateBps(bitrate_bps);
+  testing::RtpStream::PacketList packets;
+  int64_t next_time_us =
+      stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
+  bool overuse = false;
+  while (!packets.empty()) {
+    testing::RtpStream::RtpPacket* packet = packets.front();
+    bitrate_observer_->Reset();
+    // The simulated clock should match the time of packet->arrival_time
+    // since both are used in IncomingPacket().
+    clock_.AdvanceTimeMicroseconds(packet->arrival_time -
+                                   clock_.TimeInMicroseconds());
+    IncomingPacket(packet->ssrc, packet->size,
+                   (packet->arrival_time + 500) / 1000, packet->rtp_timestamp,
+                   AbsSendTime(packet->send_time, 1000000));
+    if (bitrate_observer_->updated()) {
+      if (bitrate_observer_->latest_bitrate() < bitrate_bps)
+        overuse = true;
+    }
+    delete packet;
+    packets.pop_front();
+  }
+  bitrate_estimator_->Process();
+  clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
+  return overuse;
+}
+
+// Run the bandwidth estimator with a stream of `number_of_frames` frames, or
+// until it reaches `target_bitrate`.
+// Can for instance be used to run the estimator for some time to get it
+// into a steady state.
+uint32_t RemoteBitrateEstimatorTest::SteadyStateRun(uint32_t ssrc,
+                                                    int max_number_of_frames,
+                                                    uint32_t start_bitrate,
+                                                    uint32_t min_bitrate,
+                                                    uint32_t max_bitrate,
+                                                    uint32_t target_bitrate) {
+  uint32_t bitrate_bps = start_bitrate;
+  bool bitrate_update_seen = false;
+  // Produce `number_of_frames` frames and give them to the estimator.
+  for (int i = 0; i < max_number_of_frames; ++i) {
+    bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
+    if (overuse) {
+      EXPECT_LT(bitrate_observer_->latest_bitrate(), max_bitrate);
+      EXPECT_GT(bitrate_observer_->latest_bitrate(), min_bitrate);
+      bitrate_bps = bitrate_observer_->latest_bitrate();
+      bitrate_update_seen = true;
+    } else if (bitrate_observer_->updated()) {
+      bitrate_bps = bitrate_observer_->latest_bitrate();
+      bitrate_observer_->Reset();
+    }
+    if (bitrate_update_seen && bitrate_bps > target_bitrate) {
+      break;
+    }
+  }
+  EXPECT_TRUE(bitrate_update_seen);
+  return bitrate_bps;
+}
+
+void RemoteBitrateEstimatorTest::InitialBehaviorTestHelper(
+    uint32_t expected_converge_bitrate) {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate);
+  uint32_t timestamp = 0;
+  uint32_t absolute_send_time = 0;
+  EXPECT_EQ(bitrate_estimator_->LatestEstimate(), DataRate::Zero());
+  clock_.AdvanceTimeMilliseconds(1000);
+  bitrate_estimator_->Process();
+  EXPECT_EQ(bitrate_estimator_->LatestEstimate(), DataRate::Zero());
+  EXPECT_FALSE(bitrate_observer_->updated());
+  bitrate_observer_->Reset();
+  clock_.AdvanceTimeMilliseconds(1000);
+  // Inserting packets for 5 seconds to get a valid estimate.
+  for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+    if (i == kNumInitialPackets) {
+      bitrate_estimator_->Process();
+      EXPECT_EQ(bitrate_estimator_->LatestEstimate(), DataRate::Zero());
+      EXPECT_FALSE(bitrate_observer_->updated());
+      bitrate_observer_->Reset();
+    }
+
+    IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
+                   absolute_send_time);
+    clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
+    timestamp += 90 * kFrameIntervalMs;
+    absolute_send_time =
+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
+  }
+  bitrate_estimator_->Process();
+  uint32_t bitrate_bps = bitrate_estimator_->LatestEstimate().bps<uint32_t>();
+  EXPECT_NEAR(expected_converge_bitrate, bitrate_bps, kAcceptedBitrateErrorBps);
+  EXPECT_TRUE(bitrate_observer_->updated());
+  bitrate_observer_->Reset();
+  EXPECT_EQ(bitrate_observer_->latest_bitrate(), bitrate_bps);
+  bitrate_estimator_->RemoveStream(kDefaultSsrc);
+  EXPECT_EQ(bitrate_estimator_->LatestEstimate(), DataRate::Zero());
+}
+
+void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
+    uint32_t expected_bitrate_bps) {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate);
+  uint32_t timestamp = 0;
+  uint32_t absolute_send_time = 0;
+  // Inserting packets for five seconds to get a valid estimate.
+  for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+    // TODO(sprang): Remove this hack once the single stream estimator is gone,
+    // as it doesn't do anything in Process().
+    if (i == kNumInitialPackets) {
+      // Process after we have enough frames to get a valid input rate estimate.
+      bitrate_estimator_->Process();
+      EXPECT_FALSE(bitrate_observer_->updated());  // No valid estimate.
+    }
+
+    IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
+                   absolute_send_time);
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    timestamp += 90 * kFrameIntervalMs;
+    absolute_send_time =
+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
+  }
+  bitrate_estimator_->Process();
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_->latest_bitrate(),
+              kAcceptedBitrateErrorBps);
+  for (int i = 0; i < 10; ++i) {
+    clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+    timestamp += 2 * 90 * kFrameIntervalMs;
+    absolute_send_time =
+        AddAbsSendTime(absolute_send_time, 2 * kFrameIntervalAbsSendTime);
+    IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
+                   absolute_send_time);
+    IncomingPacket(
+        kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
+        timestamp - 90 * kFrameIntervalMs,
+        AddAbsSendTime(absolute_send_time,
+                       -static_cast<int>(kFrameIntervalAbsSendTime)));
+  }
+  bitrate_estimator_->Process();
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_->latest_bitrate(),
+              kAcceptedBitrateErrorBps);
+}
+
+// Make sure we initially increase the bitrate as expected.
+void RemoteBitrateEstimatorTest::RateIncreaseRtpTimestampsTestHelper(
+    int expected_iterations) {
+  // This threshold corresponds approximately to increasing linearly with
+  // bitrate(i) = 1.04 * bitrate(i-1) + 1000
+  // until bitrate(i) > 500000, with bitrate(1) ~= 30000.
+  uint32_t bitrate_bps = 30000;
+  int iterations = 0;
+  AddDefaultStream();
+  // Feed the estimator with a stream of packets and verify that it reaches
+  // 500 kbps at the expected time.
+  while (bitrate_bps < 5e5) {
+    bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+    if (overuse) {
+      EXPECT_GT(bitrate_observer_->latest_bitrate(), bitrate_bps);
+      bitrate_bps = bitrate_observer_->latest_bitrate();
+      bitrate_observer_->Reset();
+    } else if (bitrate_observer_->updated()) {
+      bitrate_bps = bitrate_observer_->latest_bitrate();
+      bitrate_observer_->Reset();
+    }
+    ++iterations;
+    ASSERT_LE(iterations, expected_iterations);
+  }
+  ASSERT_EQ(expected_iterations, iterations);
+}
+
+void RemoteBitrateEstimatorTest::CapacityDropTestHelper(
+    int number_of_streams,
+    bool wrap_time_stamp,
+    uint32_t expected_bitrate_drop_delta,
+    int64_t receiver_clock_offset_change_ms) {
+  const int kFramerate = 30;
+  const int kStartBitrate = 900e3;
+  const int kMinExpectedBitrate = 800e3;
+  const int kMaxExpectedBitrate = 1100e3;
+  const uint32_t kInitialCapacityBps = 1000e3;
+  const uint32_t kReducedCapacityBps = 500e3;
+
+  int steady_state_time = 0;
+  if (number_of_streams <= 1) {
+    steady_state_time = 10;
+    AddDefaultStream();
+  } else {
+    steady_state_time = 10 * number_of_streams;
+    int bitrate_sum = 0;
+    int kBitrateDenom = number_of_streams * (number_of_streams - 1);
+    for (int i = 0; i < number_of_streams; i++) {
+      // First stream gets half available bitrate, while the rest share the
+      // remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
+      int bitrate = kStartBitrate / 2;
+      if (i > 0) {
+        bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
+      }
+      uint32_t mask = ~0ull << (32 - i);
+      stream_generator_->AddStream(
+          new testing::RtpStream(kFramerate,          // Frames per second.
+                                 bitrate,             // Bitrate.
+                                 kDefaultSsrc + i,    // SSRC.
+                                 90000,               // RTP frequency.
+                                 0xFFFFF000u ^ mask,  // Timestamp offset.
+                                 0));                 // RTCP receive time.
+      bitrate_sum += bitrate;
+    }
+    ASSERT_EQ(bitrate_sum, kStartBitrate);
+  }
+  if (wrap_time_stamp) {
+    stream_generator_->set_rtp_timestamp_offset(
+        kDefaultSsrc,
+        std::numeric_limits<uint32_t>::max() - steady_state_time * 90000);
+  }
+
+  // Run in steady state to make the estimator converge.
+  stream_generator_->set_capacity_bps(kInitialCapacityBps);
+  uint32_t bitrate_bps = SteadyStateRun(
+      kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
+      kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
+  EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 130000u);
+  bitrate_observer_->Reset();
+
+  // Add an offset to make sure the BWE can handle it.
+  arrival_time_offset_ms_ += receiver_clock_offset_change_ms;
+
+  // Reduce the capacity and verify the decrease time.
+  stream_generator_->set_capacity_bps(kReducedCapacityBps);
+  int64_t overuse_start_time = clock_.TimeInMilliseconds();
+  int64_t bitrate_drop_time = -1;
+  for (int i = 0; i < 100 * number_of_streams; ++i) {
+    GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+    if (bitrate_drop_time == -1 &&
+        bitrate_observer_->latest_bitrate() <= kReducedCapacityBps) {
+      bitrate_drop_time = clock_.TimeInMilliseconds();
+    }
+    if (bitrate_observer_->updated())
+      bitrate_bps = bitrate_observer_->latest_bitrate();
+  }
+
+  EXPECT_NEAR(expected_bitrate_drop_delta,
+              bitrate_drop_time - overuse_start_time, 33);
+
+  // Remove stream one by one.
+  for (int i = 0; i < number_of_streams; i++) {
+    EXPECT_EQ(bitrate_estimator_->LatestEstimate().bps(), bitrate_bps);
+    bitrate_estimator_->RemoveStream(kDefaultSsrc + i);
+  }
+  EXPECT_EQ(bitrate_estimator_->LatestEstimate(), DataRate::Zero());
+}
+
+void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate);
+  uint32_t timestamp = 0;
+  // Initialize absolute_send_time (24 bits) so that it will definitely wrap
+  // during the test.
+  uint32_t absolute_send_time = AddAbsSendTime(
+      (1 << 24), -static_cast<int>(50 * kFrameIntervalAbsSendTime));
+  // Initial set of frames to increase the bitrate. 6 seconds to have enough
+  // time for the first estimate to be generated and for Process() to be called.
+  for (int i = 0; i <= 6 * kFramerate; ++i) {
+    IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
+                   absolute_send_time);
+    bitrate_estimator_->Process();
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    timestamp += 90 * kFrameIntervalMs;
+    absolute_send_time =
+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
+  }
+  EXPECT_TRUE(bitrate_observer_->updated());
+  EXPECT_GE(bitrate_observer_->latest_bitrate(), 400000u);
+
+  // Insert batches of frames which were sent very close in time. Also simulate
+  // capacity over-use to see that we back off correctly.
+  const int kTimestampGroupLength = 15;
+  const uint32_t kTimestampGroupLengthAbsSendTime =
+      AbsSendTime(kTimestampGroupLength, 90000);
+  const uint32_t kSingleRtpTickAbsSendTime = AbsSendTime(1, 90000);
+  for (int i = 0; i < 100; ++i) {
+    for (int j = 0; j < kTimestampGroupLength; ++j) {
+      // Insert `kTimestampGroupLength` frames with just 1 timestamp ticks in
+      // between. Should be treated as part of the same group by the estimator.
+      IncomingPacket(kDefaultSsrc, 100, clock_.TimeInMilliseconds(), timestamp,
+                     absolute_send_time);
+      clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
+      timestamp += 1;
+      absolute_send_time =
+          AddAbsSendTime(absolute_send_time, kSingleRtpTickAbsSendTime);
+    }
+    // Increase time until next batch to simulate over-use.
+    clock_.AdvanceTimeMilliseconds(10);
+    timestamp += 90 * kFrameIntervalMs - kTimestampGroupLength;
+    absolute_send_time = AddAbsSendTime(
+        absolute_send_time,
+        AddAbsSendTime(kFrameIntervalAbsSendTime,
+                       -static_cast<int>(kTimestampGroupLengthAbsSendTime)));
+    bitrate_estimator_->Process();
+  }
+  EXPECT_TRUE(bitrate_observer_->updated());
+  // Should have reduced the estimate.
+  EXPECT_LT(bitrate_observer_->latest_bitrate(), 400000u);
+}
+
+void RemoteBitrateEstimatorTest::TestWrappingHelper(int silence_time_s) {
+  const int kFramerate = 100;
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate);
+  uint32_t absolute_send_time = 0;
+  uint32_t timestamp = 0;
+
+  for (size_t i = 0; i < 3000; ++i) {
+    IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
+                   absolute_send_time);
+    timestamp += kFrameIntervalMs;
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    absolute_send_time =
+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
+    bitrate_estimator_->Process();
+  }
+  DataRate bitrate_before = bitrate_estimator_->LatestEstimate();
+
+  clock_.AdvanceTimeMilliseconds(silence_time_s * 1000);
+  absolute_send_time =
+      AddAbsSendTime(absolute_send_time, AbsSendTime(silence_time_s, 1));
+  bitrate_estimator_->Process();
+  for (size_t i = 0; i < 21; ++i) {
+    IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
+                   absolute_send_time);
+    timestamp += kFrameIntervalMs;
+    clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+    absolute_send_time =
+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
+    bitrate_estimator_->Process();
+  }
+  DataRate bitrate_after = bitrate_estimator_->LatestEstimate();
+  EXPECT_LT(bitrate_after, bitrate_before);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
new file mode 100644
index 0000000000..a3b1cfdb34
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_UNITTEST_HELPER_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_UNITTEST_HELPER_H_
+
+#include <list>
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace testing {
+
+class TestBitrateObserver : public RemoteBitrateObserver {
+ public:
+  TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
+  virtual ~TestBitrateObserver() {}
+
+  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                               uint32_t bitrate) override;
+
+  void Reset() { updated_ = false; }
+
+  bool updated() const { return updated_; }
+
+  uint32_t latest_bitrate() const { return latest_bitrate_; }
+
+ private:
+  bool updated_;
+  uint32_t latest_bitrate_;
+};
+
+class RtpStream {
+ public:
+  struct RtpPacket {
+    int64_t send_time;
+    int64_t arrival_time;
+    uint32_t rtp_timestamp;
+    size_t size;
+    uint32_t ssrc;
+  };
+
+  struct RtcpPacket {
+    uint32_t ntp_secs;
+    uint32_t ntp_frac;
+    uint32_t timestamp;
+    uint32_t ssrc;
+  };
+
+  typedef std::list<RtpPacket*> PacketList;
+
+  enum { kSendSideOffsetUs = 1000000 };
+
+  RtpStream(int fps,
+            int bitrate_bps,
+            uint32_t ssrc,
+            uint32_t frequency,
+            uint32_t timestamp_offset,
+            int64_t rtcp_receive_time);
+
+  RtpStream(const RtpStream&) = delete;
+  RtpStream& operator=(const RtpStream&) = delete;
+
+  void set_rtp_timestamp_offset(uint32_t offset);
+
+  // Generates a new frame for this stream. If called too soon after the
+  // previous frame, no frame will be generated. The frame is split into
+  // packets.
+  int64_t GenerateFrame(int64_t time_now_us, PacketList* packets);
+
+  // The send-side time when the next frame can be generated.
+  int64_t next_rtp_time() const;
+
+  // Generates an RTCP packet.
+  RtcpPacket* Rtcp(int64_t time_now_us);
+
+  void set_bitrate_bps(int bitrate_bps);
+
+  int bitrate_bps() const;
+
+  uint32_t ssrc() const;
+
+  static bool Compare(const std::pair<uint32_t, RtpStream*>& left,
+                      const std::pair<uint32_t, RtpStream*>& right);
+
+ private:
+  enum { kRtcpIntervalUs = 1000000 };
+
+  int fps_;
+  int bitrate_bps_;
+  uint32_t ssrc_;
+  uint32_t frequency_;
+  int64_t next_rtp_time_;
+  int64_t next_rtcp_time_;
+  uint32_t rtp_timestamp_offset_;
+  const double kNtpFracPerMs;
+};
+
+class StreamGenerator {
+ public:
+  typedef std::list<RtpStream::RtcpPacket*> RtcpList;
+
+  StreamGenerator(int capacity, int64_t time_now);
+
+  ~StreamGenerator();
+
+  StreamGenerator(const StreamGenerator&) = delete;
+  StreamGenerator& operator=(const StreamGenerator&) = delete;
+
+  // Add a new stream.
+  void AddStream(RtpStream* stream);
+
+  // Set the link capacity.
+  void set_capacity_bps(int capacity_bps);
+
+  // Divides `bitrate_bps` among all streams. The allocated bitrate per stream
+  // is decided by the initial allocation ratios.
+  void SetBitrateBps(int bitrate_bps);
+
+  // Set the RTP timestamp offset for the stream identified by `ssrc`.
+  void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset);
+
+  // TODO(holmer): Break out the channel simulation part from this class to make
+  // it possible to simulate different types of channels.
+  int64_t GenerateFrame(RtpStream::PacketList* packets, int64_t time_now_us);
+
+ private:
+  typedef std::map<uint32_t, RtpStream*> StreamMap;
+
+  // Capacity of the simulated channel in bits per second.
+  int capacity_;
+  // The time when the last packet arrived.
+  int64_t prev_arrival_time_us_;
+  // All streams being transmitted on this simulated channel.
+  StreamMap streams_;
+};
+}  // namespace testing
+
+class RemoteBitrateEstimatorTest : public ::testing::Test {
+ public:
+  RemoteBitrateEstimatorTest();
+  virtual ~RemoteBitrateEstimatorTest();
+
+  RemoteBitrateEstimatorTest(const RemoteBitrateEstimatorTest&) = delete;
+  RemoteBitrateEstimatorTest& operator=(const RemoteBitrateEstimatorTest&) =
+      delete;
+
+ protected:
+  virtual void SetUp() = 0;
+
+  void AddDefaultStream();
+
+  // Helper to convert some time format to resolution used in absolute send time
+  // header extension, rounded upwards. `t` is the time to convert, in some
+  // resolution. `denom` is the value to divide `t` by to get whole seconds,
+  // e.g. `denom` = 1000 if `t` is in milliseconds.
+  static uint32_t AbsSendTime(int64_t t, int64_t denom);
+
+  // Helper to add two absolute send time values and keep it less than 1<<24.
+  static uint32_t AddAbsSendTime(uint32_t t1, uint32_t t2);
+
+  // Helper to create an RTPHeader containing the relevant data for the
+  // estimator (all other fields are cleared) and call IncomingPacket on the
+  // estimator.
+  void IncomingPacket(uint32_t ssrc,
+                      size_t payload_size,
+                      int64_t arrival_time,
+                      uint32_t rtp_timestamp,
+                      uint32_t absolute_send_time);
+
+  // Generates a frame of packets belonging to a stream at a given bitrate and
+  // with a given ssrc. The stream is pushed through a very simple simulated
+  // network, and is then given to the receive-side bandwidth estimator.
+  // Returns true if an over-use was seen, false otherwise.
+  // The StreamGenerator::updated() should be used to check for any changes in
+  // target bitrate after the call to this function.
+  bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps);
+
+  // Run the bandwidth estimator with a stream of `number_of_frames` frames, or
+  // until it reaches `target_bitrate`.
+  // Can for instance be used to run the estimator for some time to get it
+  // into a steady state.
+  uint32_t SteadyStateRun(uint32_t ssrc,
+                          int number_of_frames,
+                          uint32_t start_bitrate,
+                          uint32_t min_bitrate,
+                          uint32_t max_bitrate,
+                          uint32_t target_bitrate);
+
+  void TestTimestampGroupingTestHelper();
+
+  void TestWrappingHelper(int silence_time_s);
+
+  void InitialBehaviorTestHelper(uint32_t expected_converge_bitrate);
+  void RateIncreaseReorderingTestHelper(uint32_t expected_bitrate);
+  void RateIncreaseRtpTimestampsTestHelper(int expected_iterations);
+  void CapacityDropTestHelper(int number_of_streams,
+                              bool wrap_time_stamp,
+                              uint32_t expected_bitrate_drop_delta,
+                              int64_t receiver_clock_offset_change_ms);
+
+  static const uint32_t kDefaultSsrc;
+
+  SimulatedClock clock_;  // Time at the receiver.
+  std::unique_ptr<testing::TestBitrateObserver> bitrate_observer_;
+  std::unique_ptr<RemoteBitrateEstimator> bitrate_estimator_;
+  std::unique_ptr<testing::StreamGenerator> stream_generator_;
+  int64_t arrival_time_offset_ms_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_UNITTEST_HELPER_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
new file mode 100644
index 0000000000..598279e0af
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
@@ -0,0 +1,328 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "api/units/data_size.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+// The maximum allowed value for a timestamp in milliseconds. This is lower
+// than the numerical limit since we often convert to microseconds.
+constexpr int64_t kMaxTimeMs = std::numeric_limits<int64_t>::max() / 1000;
+constexpr TimeDelta kBackWindow = TimeDelta::Millis(500);
+constexpr TimeDelta kMinInterval = TimeDelta::Millis(50);
+constexpr TimeDelta kMaxInterval = TimeDelta::Millis(250);
+constexpr TimeDelta kDefaultInterval = TimeDelta::Millis(100);
+
+TimeDelta GetAbsoluteSendTimeDelta(uint32_t new_sendtime,
+                                   uint32_t previous_sendtime) {
+  static constexpr uint32_t kWrapAroundPeriod = 0x0100'0000;
+  RTC_DCHECK_LT(new_sendtime, kWrapAroundPeriod);
+  RTC_DCHECK_LT(previous_sendtime, kWrapAroundPeriod);
+  uint32_t delta = (new_sendtime - previous_sendtime) % kWrapAroundPeriod;
+  if (delta >= kWrapAroundPeriod / 2) {
+    // absolute send time wraps around, thus treat deltas larger than half of
+    // the wrap around period as negative. Ignore reordering of packets and
+    // treat them as they have approximately the same send time.
+    return TimeDelta::Zero();
+  }
+  return TimeDelta::Micros(int64_t{delta} * 1'000'000 / (1 << 18));
+}
+}  // namespace
+
+RemoteEstimatorProxy::RemoteEstimatorProxy(
+    TransportFeedbackSender feedback_sender,
+    NetworkStateEstimator* network_state_estimator)
+    : feedback_sender_(std::move(feedback_sender)),
+      last_process_time_(Timestamp::MinusInfinity()),
+      network_state_estimator_(network_state_estimator),
+      media_ssrc_(0),
+      feedback_packet_count_(0),
+      packet_overhead_(DataSize::Zero()),
+      send_interval_(kDefaultInterval),
+      send_periodic_feedback_(true),
+      previous_abs_send_time_(0),
+      abs_send_timestamp_(Timestamp::Zero()) {
+  RTC_LOG(LS_INFO)
+      << "Maximum interval between transport feedback RTCP messages: "
+      << kMaxInterval;
+}
+
+RemoteEstimatorProxy::~RemoteEstimatorProxy() {}
+
+void RemoteEstimatorProxy::MaybeCullOldPackets(int64_t sequence_number,
+                                               Timestamp arrival_time) {
+  if (periodic_window_start_seq_ >=
+          packet_arrival_times_.end_sequence_number() &&
+      arrival_time - Timestamp::Zero() >= kBackWindow) {
+    // Start new feedback packet, cull old packets.
+    packet_arrival_times_.RemoveOldPackets(sequence_number,
+                                           arrival_time - kBackWindow);
+  }
+}
+
+void RemoteEstimatorProxy::IncomingPacket(int64_t arrival_time_ms,
+                                          size_t payload_size,
+                                          const RTPHeader& header) {
+  if (arrival_time_ms < 0 || arrival_time_ms >= kMaxTimeMs) {
+    RTC_LOG(LS_WARNING) << "Arrival time out of bounds: " << arrival_time_ms;
+    return;
+  }
+  Packet packet = {.arrival_time = Timestamp::Millis(arrival_time_ms),
+                   .size = DataSize::Bytes(header.headerLength + payload_size),
+                   .ssrc = header.ssrc};
+  if (header.extension.hasTransportSequenceNumber) {
+    packet.transport_sequence_number = header.extension.transportSequenceNumber;
+  }
+  if (header.extension.hasAbsoluteSendTime) {
+    packet.absolute_send_time_24bits = header.extension.absoluteSendTime;
+  }
+  packet.feedback_request = header.extension.feedback_request;
+
+  IncomingPacket(packet);
+}
+
+void RemoteEstimatorProxy::IncomingPacket(Packet packet) {
+  MutexLock lock(&lock_);
+  media_ssrc_ = packet.ssrc;
+  int64_t seq = 0;
+
+  if (packet.transport_sequence_number.has_value()) {
+    seq = unwrapper_.Unwrap(*packet.transport_sequence_number);
+
+    if (send_periodic_feedback_) {
+      MaybeCullOldPackets(seq, packet.arrival_time);
+
+      if (!periodic_window_start_seq_ || seq < *periodic_window_start_seq_) {
+        periodic_window_start_seq_ = seq;
+      }
+    }
+
+    // We are only interested in the first time a packet is received.
+    if (packet_arrival_times_.has_received(seq)) {
+      return;
+    }
+
+    packet_arrival_times_.AddPacket(seq, packet.arrival_time);
+
+    // Limit the range of sequence numbers to send feedback for.
+    if (!periodic_window_start_seq_.has_value() ||
+        periodic_window_start_seq_.value() <
+            packet_arrival_times_.begin_sequence_number()) {
+      periodic_window_start_seq_ =
+          packet_arrival_times_.begin_sequence_number();
+    }
+
+    if (packet.feedback_request) {
+      // Send feedback packet immediately.
+      SendFeedbackOnRequest(seq, *packet.feedback_request);
+    }
+  }
+
+  if (network_state_estimator_ && packet.absolute_send_time_24bits) {
+    PacketResult packet_result;
+    packet_result.receive_time = packet.arrival_time;
+    abs_send_timestamp_ += GetAbsoluteSendTimeDelta(
+        *packet.absolute_send_time_24bits, previous_abs_send_time_);
+    previous_abs_send_time_ = *packet.absolute_send_time_24bits;
+    packet_result.sent_packet.send_time = abs_send_timestamp_;
+    packet_result.sent_packet.size = packet.size + packet_overhead_;
+    packet_result.sent_packet.sequence_number = seq;
+    network_state_estimator_->OnReceivedPacket(packet_result);
+  }
+}
+
+TimeDelta RemoteEstimatorProxy::Process(Timestamp now) {
+  MutexLock lock(&lock_);
+  if (!send_periodic_feedback_) {
+    return TimeDelta::PlusInfinity();
+  }
+  Timestamp next_process_time = last_process_time_ + send_interval_;
+  if (now >= next_process_time) {
+    last_process_time_ = now;
+    SendPeriodicFeedbacks();
+    return send_interval_;
+  }
+
+  return next_process_time - now;
+}
+
+void RemoteEstimatorProxy::OnBitrateChanged(int bitrate_bps) {
+  // TwccReportSize = Ipv4(20B) + UDP(8B) + SRTP(10B) +
+  // AverageTwccReport(30B)
+  // TwccReport size at 50ms interval is 24 byte.
+  // TwccReport size at 250ms interval is 36 byte.
+  // AverageTwccReport = (TwccReport(50ms) + TwccReport(250ms)) / 2
+  constexpr DataSize kTwccReportSize = DataSize::Bytes(20 + 8 + 10 + 30);
+  constexpr DataRate kMinTwccRate = kTwccReportSize / kMaxInterval;
+
+  // Let TWCC reports occupy 5% of total bandwidth.
+  DataRate twcc_bitrate = DataRate::BitsPerSec(0.05 * bitrate_bps);
+
+  // Check upper send_interval bound by checking bitrate to avoid overflow when
+  // dividing by small bitrate, in particular avoid dividing by zero bitrate.
+  TimeDelta send_interval =
+      twcc_bitrate <= kMinTwccRate
+          ? kMaxInterval
+          : std::max(kTwccReportSize / twcc_bitrate, kMinInterval);
+
+  MutexLock lock(&lock_);
+  send_interval_ = send_interval;
+}
+
+void RemoteEstimatorProxy::SetSendPeriodicFeedback(
+    bool send_periodic_feedback) {
+  MutexLock lock(&lock_);
+  send_periodic_feedback_ = send_periodic_feedback;
+}
+
+void RemoteEstimatorProxy::SetTransportOverhead(DataSize overhead_per_packet) {
+  MutexLock lock(&lock_);
+  packet_overhead_ = overhead_per_packet;
+}
+
+void RemoteEstimatorProxy::SendPeriodicFeedbacks() {
+  // `periodic_window_start_seq_` is the first sequence number to include in
+  // the current feedback packet. Some older may still be in the map, in case
+  // a reordering happens and we need to retransmit them.
+  if (!periodic_window_start_seq_)
+    return;
+
+  std::unique_ptr<rtcp::RemoteEstimate> remote_estimate;
+  if (network_state_estimator_) {
+    absl::optional<NetworkStateEstimate> state_estimate =
+        network_state_estimator_->GetCurrentEstimate();
+    if (state_estimate) {
+      remote_estimate = std::make_unique<rtcp::RemoteEstimate>();
+      remote_estimate->SetEstimate(state_estimate.value());
+    }
+  }
+
+  int64_t packet_arrival_times_end_seq =
+      packet_arrival_times_.end_sequence_number();
+  while (periodic_window_start_seq_ < packet_arrival_times_end_seq) {
+    auto feedback_packet = MaybeBuildFeedbackPacket(
+        /*include_timestamps=*/true, periodic_window_start_seq_.value(),
+        packet_arrival_times_end_seq,
+        /*is_periodic_update=*/true);
+
+    if (feedback_packet == nullptr) {
+      break;
+    }
+
+    RTC_DCHECK(feedback_sender_ != nullptr);
+
+    std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets;
+    if (remote_estimate) {
+      packets.push_back(std::move(remote_estimate));
+    }
+    packets.push_back(std::move(feedback_packet));
+
+    feedback_sender_(std::move(packets));
+    // Note: Don't erase items from packet_arrival_times_ after sending, in
+    // case they need to be re-sent after a reordering. Removal will be
+    // handled by OnPacketArrival once packets are too old.
+  }
+}
+
+void RemoteEstimatorProxy::SendFeedbackOnRequest(
+    int64_t sequence_number,
+    const FeedbackRequest& feedback_request) {
+  if (feedback_request.sequence_count == 0) {
+    return;
+  }
+
+  int64_t first_sequence_number =
+      sequence_number - feedback_request.sequence_count + 1;
+
+  auto feedback_packet = MaybeBuildFeedbackPacket(
+      feedback_request.include_timestamps, first_sequence_number,
+      sequence_number + 1, /*is_periodic_update=*/false);
+
+  // This is called when a packet has just been added.
+  RTC_DCHECK(feedback_packet != nullptr);
+
+  // Clear up to the first packet that is included in this feedback packet.
+  packet_arrival_times_.EraseTo(first_sequence_number);
+
+  RTC_DCHECK(feedback_sender_ != nullptr);
+  std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets;
+  packets.push_back(std::move(feedback_packet));
+  feedback_sender_(std::move(packets));
+}
+
+std::unique_ptr<rtcp::TransportFeedback>
+RemoteEstimatorProxy::MaybeBuildFeedbackPacket(
+    bool include_timestamps,
+    int64_t begin_sequence_number_inclusive,
+    int64_t end_sequence_number_exclusive,
+    bool is_periodic_update) {
+  RTC_DCHECK_LT(begin_sequence_number_inclusive, end_sequence_number_exclusive);
+
+  int64_t start_seq =
+      packet_arrival_times_.clamp(begin_sequence_number_inclusive);
+
+  int64_t end_seq = packet_arrival_times_.clamp(end_sequence_number_exclusive);
+
+  // Create the packet on demand, as it's not certain that there are packets
+  // in the range that have been received.
+  std::unique_ptr<rtcp::TransportFeedback> feedback_packet = nullptr;
+
+  int64_t next_sequence_number = begin_sequence_number_inclusive;
+
+  for (int64_t seq = start_seq; seq < end_seq; ++seq) {
+    PacketArrivalTimeMap::PacketArrivalTime packet =
+        packet_arrival_times_.FindNextAtOrAfter(seq);
+    seq = packet.sequence_number;
+    if (seq >= end_seq) {
+      break;
+    }
+
+    if (feedback_packet == nullptr) {
+      feedback_packet =
+          std::make_unique<rtcp::TransportFeedback>(include_timestamps);
+      feedback_packet->SetMediaSsrc(media_ssrc_);
+      // Base sequence number is the expected first sequence number. This is
+      // known, but we might not have actually received it, so the base time
+      // shall be the time of the first received packet in the feedback.
+      feedback_packet->SetBase(
+          static_cast<uint16_t>(begin_sequence_number_inclusive & 0xFFFF),
+          packet.arrival_time);
+      feedback_packet->SetFeedbackSequenceNumber(feedback_packet_count_++);
+    }
+
+    if (!feedback_packet->AddReceivedPacket(static_cast<uint16_t>(seq & 0xFFFF),
+                                            packet.arrival_time)) {
+      // Could not add timestamp, feedback packet might be full. Return and
+      // try again with a fresh packet.
+      break;
+    }
+
+    next_sequence_number = seq + 1;
+  }
+  if (is_periodic_update) {
+    periodic_window_start_seq_ = next_sequence_number;
+  }
+  return feedback_packet;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
new file mode 100644
index 0000000000..54257ea6f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_
+
+#include <deque>
+#include <functional>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/rtp_headers.h"
+#include "api/transport/network_control.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/packet_arrival_map.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// Class used when send-side BWE is enabled: This proxy is instantiated on the
+// receive side. It buffers a number of receive timestamps and then sends
+// transport feedback messages back too the send side.
+class RemoteEstimatorProxy {
+ public:
+  // Used for sending transport feedback messages when send side
+  // BWE is used.
+  using TransportFeedbackSender = std::function<void(
+      std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets)>;
+  RemoteEstimatorProxy(TransportFeedbackSender feedback_sender,
+                       NetworkStateEstimator* network_state_estimator);
+  ~RemoteEstimatorProxy();
+
+  struct Packet {
+    Timestamp arrival_time;
+    DataSize size;
+    uint32_t ssrc;
+    absl::optional<uint32_t> absolute_send_time_24bits;
+    absl::optional<uint16_t> transport_sequence_number;
+    absl::optional<FeedbackRequest> feedback_request;
+  };
+  void IncomingPacket(Packet packet);
+
+  void IncomingPacket(int64_t arrival_time_ms,
+                      size_t payload_size,
+                      const RTPHeader& header);
+
+  // Sends periodic feedback if it is time to send it.
+  // Returns time until next call to Process should be made.
+  TimeDelta Process(Timestamp now);
+
+  void OnBitrateChanged(int bitrate);
+  void SetSendPeriodicFeedback(bool send_periodic_feedback);
+  void SetTransportOverhead(DataSize overhead_per_packet);
+
+ private:
+  void MaybeCullOldPackets(int64_t sequence_number, Timestamp arrival_time)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
+  void SendPeriodicFeedbacks() RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
+  void SendFeedbackOnRequest(int64_t sequence_number,
+                             const FeedbackRequest& feedback_request)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
+
+  // Returns a Transport Feedback packet with information about as many packets
+  // that has been received between [`begin_sequence_number_incl`,
+  // `end_sequence_number_excl`) that can fit in it. If `is_periodic_update`,
+  // this represents sending a periodic feedback message, which will make it
+  // update the `periodic_window_start_seq_` variable with the first packet that
+  // was not included in the feedback packet, so that the next update can
+  // continue from that sequence number.
+  //
+  // If no incoming packets were added, nullptr is returned.
+  //
+  // `include_timestamps` decide if the returned TransportFeedback should
+  // include timestamps.
+  std::unique_ptr<rtcp::TransportFeedback> MaybeBuildFeedbackPacket(
+      bool include_timestamps,
+      int64_t begin_sequence_number_inclusive,
+      int64_t end_sequence_number_exclusive,
+      bool is_periodic_update) RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
+
+  const TransportFeedbackSender feedback_sender_;
+  Timestamp last_process_time_;
+
+  Mutex lock_;
+  //  `network_state_estimator_` may be null.
+  NetworkStateEstimator* const network_state_estimator_
+      RTC_PT_GUARDED_BY(&lock_);
+  uint32_t media_ssrc_ RTC_GUARDED_BY(&lock_);
+  uint8_t feedback_packet_count_ RTC_GUARDED_BY(&lock_);
+  SeqNumUnwrapper<uint16_t> unwrapper_ RTC_GUARDED_BY(&lock_);
+  DataSize packet_overhead_ RTC_GUARDED_BY(&lock_);
+
+  // The next sequence number that should be the start sequence number during
+  // periodic reporting. Will be absl::nullopt before the first seen packet.
+  absl::optional<int64_t> periodic_window_start_seq_ RTC_GUARDED_BY(&lock_);
+
+  // Packet arrival times, by sequence number.
+  PacketArrivalTimeMap packet_arrival_times_ RTC_GUARDED_BY(&lock_);
+
+  TimeDelta send_interval_ RTC_GUARDED_BY(&lock_);
+  bool send_periodic_feedback_ RTC_GUARDED_BY(&lock_);
+
+  // Unwraps absolute send times.
+  uint32_t previous_abs_send_time_ RTC_GUARDED_BY(&lock_);
+  Timestamp abs_send_timestamp_ RTC_GUARDED_BY(&lock_);
+};
+
+}  // namespace webrtc
+
+#endif  //  MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_ESTIMATOR_PROXY_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
new file mode 100644
index 0000000000..437fd6f1e2
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
@@ -0,0 +1,644 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/transport/network_types.h"
+#include "api/transport/test/mock_network_control.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::Invoke;
+using ::testing::MockFunction;
+using ::testing::Return;
+using ::testing::SizeIs;
+
+constexpr DataSize kDefaultPacketSize = DataSize::Bytes(100);
+constexpr uint32_t kMediaSsrc = 456;
+constexpr uint16_t kBaseSeq = 10;
+constexpr Timestamp kBaseTime = Timestamp::Millis(123);
+constexpr TimeDelta kBaseTimeWrapAround =
+    rtcp::TransportFeedback::kDeltaTick * (int64_t{1} << 32);
+constexpr TimeDelta kMaxSmallDelta = rtcp::TransportFeedback::kDeltaTick * 0xFF;
+
+constexpr TimeDelta kBackWindow = TimeDelta::Millis(500);
+constexpr TimeDelta kMinSendInterval = TimeDelta::Millis(50);
+constexpr TimeDelta kMaxSendInterval = TimeDelta::Millis(250);
+constexpr TimeDelta kDefaultSendInterval = TimeDelta::Millis(100);
+
+std::vector<uint16_t> SequenceNumbers(
+    const rtcp::TransportFeedback& feedback_packet) {
+  std::vector<uint16_t> sequence_numbers;
+  for (const auto& rtp_packet_received : feedback_packet.GetReceivedPackets()) {
+    sequence_numbers.push_back(rtp_packet_received.sequence_number());
+  }
+  return sequence_numbers;
+}
+
+std::vector<Timestamp> Timestamps(
+    const rtcp::TransportFeedback& feedback_packet) {
+  std::vector<Timestamp> timestamps;
+  Timestamp timestamp = feedback_packet.BaseTime();
+  // rtcp::TransportFeedback makes no promises about epoch of the base time,
+  // It may add several kBaseTimeWrapAround periods to make it large enough and
+  // thus to support negative deltas. Align it close to the kBaseTime to make
+  // tests expectations simpler.
+  if (timestamp > kBaseTime) {
+    timestamp -= (timestamp - kBaseTime).RoundTo(kBaseTimeWrapAround);
+  }
+  for (const auto& rtp_packet_received : feedback_packet.GetReceivedPackets()) {
+    timestamp += rtp_packet_received.delta();
+    timestamps.push_back(timestamp);
+  }
+  return timestamps;
+}
+
+class RemoteEstimatorProxyTest : public ::testing::Test {
+ public:
+  RemoteEstimatorProxyTest()
+      : clock_(0),
+        proxy_(feedback_sender_.AsStdFunction(), &network_state_estimator_) {}
+
+ protected:
+  void IncomingPacket(
+      uint16_t seq,
+      Timestamp arrival_time,
+      absl::optional<FeedbackRequest> feedback_request = absl::nullopt) {
+    proxy_.IncomingPacket({.arrival_time = arrival_time,
+                           .size = DataSize::Bytes(100),
+                           .ssrc = kMediaSsrc,
+                           .transport_sequence_number = seq,
+                           .feedback_request = feedback_request});
+  }
+
+  void Process() {
+    clock_.AdvanceTime(kDefaultSendInterval);
+    proxy_.Process(clock_.CurrentTime());
+  }
+
+  SimulatedClock clock_;
+  MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
+      feedback_sender_;
+  ::testing::NiceMock<MockNetworkStateEstimator> network_state_estimator_;
+  RemoteEstimatorProxy proxy_;
+};
+
+TEST_F(RemoteEstimatorProxyTest, SendsSinglePacketFeedback) {
+  IncomingPacket(kBaseSeq, kBaseTime);
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq));
+            EXPECT_THAT(Timestamps(*feedback_packet), ElementsAre(kBaseTime));
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, DuplicatedPackets) {
+  IncomingPacket(kBaseSeq, kBaseTime);
+  IncomingPacket(kBaseSeq, kBaseTime + TimeDelta::Seconds(1));
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq));
+            EXPECT_THAT(Timestamps(*feedback_packet), ElementsAre(kBaseTime));
+            return true;
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, FeedbackWithMissingStart) {
+  // First feedback.
+  IncomingPacket(kBaseSeq, kBaseTime);
+  IncomingPacket(kBaseSeq + 1, kBaseTime + TimeDelta::Seconds(1));
+  EXPECT_CALL(feedback_sender_, Call);
+  Process();
+
+  // Second feedback starts with a missing packet (DROP kBaseSeq + 2).
+  IncomingPacket(kBaseSeq + 3, kBaseTime + TimeDelta::Seconds(3));
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq + 2, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq + 3));
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kBaseTime + TimeDelta::Seconds(3)));
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, SendsFeedbackWithVaryingDeltas) {
+  IncomingPacket(kBaseSeq, kBaseTime);
+  IncomingPacket(kBaseSeq + 1, kBaseTime + kMaxSmallDelta);
+  IncomingPacket(kBaseSeq + 2,
+                 kBaseTime + (2 * kMaxSmallDelta) + TimeDelta::Millis(1));
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq, kBaseSeq + 1, kBaseSeq + 2));
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kBaseTime, kBaseTime + kMaxSmallDelta,
+                                    kBaseTime + (2 * kMaxSmallDelta) +
+                                        TimeDelta::Millis(1)));
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, SendsFragmentedFeedback) {
+  static constexpr TimeDelta kTooLargeDelta =
+      rtcp::TransportFeedback::kDeltaTick * (1 << 16);
+
+  IncomingPacket(kBaseSeq, kBaseTime);
+  IncomingPacket(kBaseSeq + 1, kBaseTime + kTooLargeDelta);
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq));
+            EXPECT_THAT(Timestamps(*feedback_packet), ElementsAre(kBaseTime));
+          }))
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq + 1, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq + 1));
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kBaseTime + kTooLargeDelta));
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, HandlesReorderingAndWrap) {
+  const TimeDelta kDelta = TimeDelta::Seconds(1);
+  const uint16_t kLargeSeq = 62762;
+  IncomingPacket(kBaseSeq, kBaseTime);
+  IncomingPacket(kLargeSeq, kBaseTime + kDelta);
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [&](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kLargeSeq, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kBaseTime + kDelta, kBaseTime));
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, HandlesMalformedSequenceNumbers) {
+  // This test generates incoming packets with large jumps in sequence numbers.
+  // When unwrapped, the sequeunce numbers of these 30 incoming packets, will
+  // span a range of roughly 650k packets. Test that we only send feedback for
+  // the last packets. Test for regression found in chromium:949020.
+  const TimeDelta kDelta = TimeDelta::Seconds(1);
+  for (int i = 0; i < 10; ++i) {
+    IncomingPacket(kBaseSeq + i, kBaseTime + 3 * i * kDelta);
+    IncomingPacket(kBaseSeq + 20000 + i, kBaseTime + (3 * i + 1) * kDelta);
+    IncomingPacket(kBaseSeq + 40000 + i, kBaseTime + (3 * i + 2) * kDelta);
+  }
+
+  // Only expect feedback for the last two packets.
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [&](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq + 20009, kBaseSeq + 40009));
+            EXPECT_THAT(
+                Timestamps(*feedback_packet),
+                ElementsAre(kBaseTime + 28 * kDelta, kBaseTime + 29 * kDelta));
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, HandlesBackwardsWrappingSequenceNumbers) {
+  // This test is like HandlesMalformedSequenceNumbers but for negative wrap
+  // arounds. Test that we only send feedback for the packets with highest
+  // sequence numbers.  Test for regression found in chromium:949020.
+  const TimeDelta kDelta = TimeDelta::Seconds(1);
+  for (int i = 0; i < 10; ++i) {
+    IncomingPacket(kBaseSeq + i, kBaseTime + 3 * i * kDelta);
+    IncomingPacket(kBaseSeq + 40000 + i, kBaseTime + (3 * i + 1) * kDelta);
+    IncomingPacket(kBaseSeq + 20000 + i, kBaseTime + (3 * i + 2) * kDelta);
+  }
+
+  // Only expect feedback for the first two packets.
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [&](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq + 40000, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq + 40000, kBaseSeq));
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kBaseTime + kDelta, kBaseTime));
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, ResendsTimestampsOnReordering) {
+  IncomingPacket(kBaseSeq, kBaseTime);
+  IncomingPacket(kBaseSeq + 2, kBaseTime + TimeDelta::Millis(2));
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq, kBaseSeq + 2));
+            EXPECT_THAT(
+                Timestamps(*feedback_packet),
+                ElementsAre(kBaseTime, kBaseTime + TimeDelta::Millis(2)));
+          }));
+
+  Process();
+
+  IncomingPacket(kBaseSeq + 1, kBaseTime + TimeDelta::Millis(1));
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq + 1, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq + 1, kBaseSeq + 2));
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kBaseTime + TimeDelta::Millis(1),
+                                    kBaseTime + TimeDelta::Millis(2)));
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, RemovesTimestampsOutOfScope) {
+  const Timestamp kTimeoutTime = kBaseTime + kBackWindow;
+
+  IncomingPacket(kBaseSeq + 2, kBaseTime);
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq + 2, feedback_packet->GetBaseSequence());
+
+            EXPECT_THAT(Timestamps(*feedback_packet), ElementsAre(kBaseTime));
+          }));
+
+  Process();
+
+  IncomingPacket(kBaseSeq + 3, kTimeoutTime);  // kBaseSeq + 2 times out here.
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [&](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq + 3, feedback_packet->GetBaseSequence());
+
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kTimeoutTime));
+          }));
+
+  Process();
+
+  // New group, with sequence starting below the first so that they may be
+  // retransmitted.
+  IncomingPacket(kBaseSeq, kBaseTime - TimeDelta::Millis(1));
+  IncomingPacket(kBaseSeq + 1, kTimeoutTime - TimeDelta::Millis(1));
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [&](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq, feedback_packet->GetBaseSequence());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq, kBaseSeq + 1, kBaseSeq + 3));
+            EXPECT_THAT(
+                Timestamps(*feedback_packet),
+                ElementsAre(kBaseTime - TimeDelta::Millis(1),
+                            kTimeoutTime - TimeDelta::Millis(1), kTimeoutTime));
+          }));
+
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyTest, TimeUntilNextProcessIsDefaultOnUnkownBitrate) {
+  EXPECT_EQ(proxy_.Process(clock_.CurrentTime()), kDefaultSendInterval);
+}
+
+TEST_F(RemoteEstimatorProxyTest, TimeUntilNextProcessIsMinIntervalOn300kbps) {
+  proxy_.OnBitrateChanged(300'000);
+  EXPECT_EQ(proxy_.Process(clock_.CurrentTime()), kMinSendInterval);
+}
+
+TEST_F(RemoteEstimatorProxyTest, TimeUntilNextProcessIsMaxIntervalOn0kbps) {
+  // TimeUntilNextProcess should be limited by `kMaxSendIntervalMs` when
+  // bitrate is small. We choose 0 bps as a special case, which also tests
+  // erroneous behaviors like division-by-zero.
+  proxy_.OnBitrateChanged(0);
+  EXPECT_EQ(proxy_.Process(clock_.CurrentTime()), kMaxSendInterval);
+}
+
+TEST_F(RemoteEstimatorProxyTest, TimeUntilNextProcessIsMaxIntervalOn20kbps) {
+  proxy_.OnBitrateChanged(20'000);
+  EXPECT_EQ(proxy_.Process(clock_.CurrentTime()), kMaxSendInterval);
+}
+
+TEST_F(RemoteEstimatorProxyTest, TwccReportsUse5PercentOfAvailableBandwidth) {
+  proxy_.OnBitrateChanged(80'000);
+  // 80kbps * 0.05 = TwccReportSize(68B * 8b/B) * 1000ms / SendInterval(136ms)
+  EXPECT_EQ(proxy_.Process(clock_.CurrentTime()), TimeDelta::Millis(136));
+}
+
+//////////////////////////////////////////////////////////////////////////////
+// Tests for the extended protocol where the feedback is explicitly requested
+// by the sender.
+//////////////////////////////////////////////////////////////////////////////
+typedef RemoteEstimatorProxyTest RemoteEstimatorProxyOnRequestTest;
+TEST_F(RemoteEstimatorProxyOnRequestTest, DisablesPeriodicProcess) {
+  proxy_.SetSendPeriodicFeedback(false);
+  EXPECT_EQ(proxy_.Process(clock_.CurrentTime()), TimeDelta::PlusInfinity());
+}
+
+TEST_F(RemoteEstimatorProxyOnRequestTest, ProcessDoesNotSendFeedback) {
+  proxy_.SetSendPeriodicFeedback(false);
+  IncomingPacket(kBaseSeq, kBaseTime);
+  EXPECT_CALL(feedback_sender_, Call).Times(0);
+  Process();
+}
+
+TEST_F(RemoteEstimatorProxyOnRequestTest, RequestSinglePacketFeedback) {
+  proxy_.SetSendPeriodicFeedback(false);
+  IncomingPacket(kBaseSeq, kBaseTime);
+  IncomingPacket(kBaseSeq + 1, kBaseTime + kMaxSmallDelta);
+  IncomingPacket(kBaseSeq + 2, kBaseTime + 2 * kMaxSmallDelta);
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq + 3, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq + 3));
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kBaseTime + 3 * kMaxSmallDelta));
+          }));
+
+  constexpr FeedbackRequest kSinglePacketFeedbackRequest = {
+      /*include_timestamps=*/true, /*sequence_count=*/1};
+  IncomingPacket(kBaseSeq + 3, kBaseTime + 3 * kMaxSmallDelta,
+                 kSinglePacketFeedbackRequest);
+}
+
+TEST_F(RemoteEstimatorProxyOnRequestTest, RequestLastFivePacketFeedback) {
+  proxy_.SetSendPeriodicFeedback(false);
+  int i = 0;
+  for (; i < 10; ++i) {
+    IncomingPacket(kBaseSeq + i, kBaseTime + i * kMaxSmallDelta);
+  }
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq + 6, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq + 6, kBaseSeq + 7, kBaseSeq + 8,
+                                    kBaseSeq + 9, kBaseSeq + 10));
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kBaseTime + 6 * kMaxSmallDelta,
+                                    kBaseTime + 7 * kMaxSmallDelta,
+                                    kBaseTime + 8 * kMaxSmallDelta,
+                                    kBaseTime + 9 * kMaxSmallDelta,
+                                    kBaseTime + 10 * kMaxSmallDelta));
+          }));
+
+  constexpr FeedbackRequest kFivePacketsFeedbackRequest = {
+      /*include_timestamps=*/true, /*sequence_count=*/5};
+  IncomingPacket(kBaseSeq + i, kBaseTime + i * kMaxSmallDelta,
+                 kFivePacketsFeedbackRequest);
+}
+
+TEST_F(RemoteEstimatorProxyOnRequestTest,
+       RequestLastFivePacketFeedbackMissingPackets) {
+  proxy_.SetSendPeriodicFeedback(false);
+  int i = 0;
+  for (; i < 10; ++i) {
+    if (i != 7 && i != 9)
+      IncomingPacket(kBaseSeq + i, kBaseTime + i * kMaxSmallDelta);
+  }
+
+  EXPECT_CALL(feedback_sender_, Call)
+      .WillOnce(Invoke(
+          [](std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback_packets) {
+            rtcp::TransportFeedback* feedback_packet =
+                static_cast<rtcp::TransportFeedback*>(
+                    feedback_packets[0].get());
+            EXPECT_EQ(kBaseSeq + 6, feedback_packet->GetBaseSequence());
+            EXPECT_EQ(kMediaSsrc, feedback_packet->media_ssrc());
+
+            EXPECT_THAT(SequenceNumbers(*feedback_packet),
+                        ElementsAre(kBaseSeq + 6, kBaseSeq + 8, kBaseSeq + 10));
+            EXPECT_THAT(Timestamps(*feedback_packet),
+                        ElementsAre(kBaseTime + 6 * kMaxSmallDelta,
+                                    kBaseTime + 8 * kMaxSmallDelta,
+                                    kBaseTime + 10 * kMaxSmallDelta));
+          }));
+
+  constexpr FeedbackRequest kFivePacketsFeedbackRequest = {
+      /*include_timestamps=*/true, /*sequence_count=*/5};
+  IncomingPacket(kBaseSeq + i, kBaseTime + i * kMaxSmallDelta,
+                 kFivePacketsFeedbackRequest);
+}
+
+TEST_F(RemoteEstimatorProxyTest, ReportsIncomingPacketToNetworkStateEstimator) {
+  Timestamp first_send_timestamp = Timestamp::Zero();
+  const DataSize kPacketOverhead = DataSize::Bytes(38);
+  proxy_.SetTransportOverhead(kPacketOverhead);
+
+  EXPECT_CALL(network_state_estimator_, OnReceivedPacket(_))
+      .WillOnce(Invoke([&](const PacketResult& packet) {
+        EXPECT_EQ(packet.receive_time, kBaseTime);
+        EXPECT_EQ(packet.sent_packet.size,
+                  kDefaultPacketSize + kPacketOverhead);
+        first_send_timestamp = packet.sent_packet.send_time;
+      }));
+  // Incoming packet with abs sendtime but without transport sequence number.
+  proxy_.IncomingPacket(
+      {.arrival_time = kBaseTime,
+       .size = kDefaultPacketSize,
+       .ssrc = kMediaSsrc,
+       .absolute_send_time_24bits = AbsoluteSendTime::To24Bits(kBaseTime)});
+
+  // Expect packet with older abs send time to be treated as sent at the same
+  // time as the previous packet due to reordering.
+  EXPECT_CALL(network_state_estimator_, OnReceivedPacket(_))
+      .WillOnce(Invoke([&first_send_timestamp](const PacketResult& packet) {
+        EXPECT_EQ(packet.receive_time, kBaseTime);
+        EXPECT_EQ(packet.sent_packet.send_time, first_send_timestamp);
+      }));
+
+  proxy_.IncomingPacket(
+      {.arrival_time = kBaseTime,
+       .size = kDefaultPacketSize,
+       .ssrc = kMediaSsrc,
+       .absolute_send_time_24bits =
+           AbsoluteSendTime::To24Bits(kBaseTime - TimeDelta::Millis(12))});
+}
+
+TEST_F(RemoteEstimatorProxyTest, IncomingPacketHandlesWrapInAbsSendTime) {
+  // abs send time use 24bit precision.
+  const uint32_t kFirstAbsSendTime =
+      AbsoluteSendTime::To24Bits(Timestamp::Millis((1 << 24) - 30));
+  // Second abs send time has wrapped.
+  const uint32_t kSecondAbsSendTime =
+      AbsoluteSendTime::To24Bits(Timestamp::Millis(1 << 24));
+  const TimeDelta kExpectedAbsSendTimeDelta = TimeDelta::Millis(30);
+
+  Timestamp first_send_timestamp = Timestamp::Zero();
+  EXPECT_CALL(network_state_estimator_, OnReceivedPacket(_))
+      .WillOnce(Invoke([&first_send_timestamp](const PacketResult& packet) {
+        EXPECT_EQ(packet.receive_time, kBaseTime);
+        first_send_timestamp = packet.sent_packet.send_time;
+      }));
+  proxy_.IncomingPacket({.arrival_time = kBaseTime,
+                         .size = kDefaultPacketSize,
+                         .ssrc = kMediaSsrc,
+                         .absolute_send_time_24bits = kFirstAbsSendTime,
+                         .transport_sequence_number = kBaseSeq});
+
+  EXPECT_CALL(network_state_estimator_, OnReceivedPacket(_))
+      .WillOnce(Invoke([first_send_timestamp,
+                        kExpectedAbsSendTimeDelta](const PacketResult& packet) {
+        EXPECT_EQ(packet.receive_time, kBaseTime + TimeDelta::Millis(123));
+        EXPECT_EQ(packet.sent_packet.send_time.ms(),
+                  (first_send_timestamp + kExpectedAbsSendTimeDelta).ms());
+      }));
+  proxy_.IncomingPacket({.arrival_time = kBaseTime + TimeDelta::Millis(123),
+                         .size = kDefaultPacketSize,
+                         .ssrc = kMediaSsrc,
+                         .absolute_send_time_24bits = kSecondAbsSendTime,
+                         .transport_sequence_number = kBaseSeq + 1});
+}
+
+TEST_F(RemoteEstimatorProxyTest, SendTransportFeedbackAndNetworkStateUpdate) {
+  proxy_.IncomingPacket(
+      {.arrival_time = kBaseTime,
+       .size = kDefaultPacketSize,
+       .ssrc = kMediaSsrc,
+       .absolute_send_time_24bits =
+           AbsoluteSendTime::To24Bits(kBaseTime - TimeDelta::Millis(1)),
+       .transport_sequence_number = kBaseSeq});
+  EXPECT_CALL(network_state_estimator_, GetCurrentEstimate())
+      .WillOnce(Return(NetworkStateEstimate()));
+  EXPECT_CALL(feedback_sender_, Call(SizeIs(2)));
+  Process();
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
new file mode 100644
index 0000000000..c8f6faa127
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
@@ -0,0 +1,262 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+
+#include <inttypes.h>
+#include <stdarg.h>
+#include <stdio.h>
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+
+static std::string ToString(uint32_t v) {
+  rtc::StringBuilder ss;
+  ss << v;
+  return ss.Release();
+}
+
+Logging::ThreadState::ThreadState() = default;
+Logging::ThreadState::~ThreadState() = default;
+
+Logging::Context::Context(uint32_t name, int64_t timestamp_ms, bool enabled) {
+  Logging::GetInstance()->PushState(ToString(name), timestamp_ms, enabled);
+}
+
+Logging::Context::Context(const std::string& name,
+                          int64_t timestamp_ms,
+                          bool enabled) {
+  Logging::GetInstance()->PushState(name, timestamp_ms, enabled);
+}
+
+Logging::Context::Context(const char* name,
+                          int64_t timestamp_ms,
+                          bool enabled) {
+  Logging::GetInstance()->PushState(name, timestamp_ms, enabled);
+}
+
+Logging::Context::~Context() {
+  Logging::GetInstance()->PopState();
+}
+
+Logging* Logging::GetInstance() {
+  static Logging* logging = new Logging();
+  return logging;
+}
+
+void Logging::SetGlobalContext(uint32_t name) {
+  MutexLock lock(&mutex_);
+  thread_map_[rtc::CurrentThreadId()].global_state.tag = ToString(name);
+}
+
+void Logging::SetGlobalContext(const std::string& name) {
+  MutexLock lock(&mutex_);
+  thread_map_[rtc::CurrentThreadId()].global_state.tag = name;
+}
+
+void Logging::SetGlobalContext(const char* name) {
+  MutexLock lock(&mutex_);
+  thread_map_[rtc::CurrentThreadId()].global_state.tag = name;
+}
+
+void Logging::SetGlobalEnable(bool enabled) {
+  MutexLock lock(&mutex_);
+  thread_map_[rtc::CurrentThreadId()].global_state.enabled = enabled;
+}
+
+void Logging::Log(const char format[], ...) {
+  MutexLock lock(&mutex_);
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  RTC_DCHECK(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("%s\t", state.tag.c_str());
+    va_list args;
+    va_start(args, format);
+    vprintf(format, args);
+    va_end(args);
+    printf("\n");
+  }
+}
+
+void Logging::Plot(int figure, const std::string& name, double value) {
+  Plot(figure, name, value, 0, "-");
+}
+
+void Logging::Plot(int figure,
+                   const std::string& name,
+                   double value,
+                   uint32_t ssrc) {
+  Plot(figure, name, value, ssrc, "-");
+}
+
+void Logging::Plot(int figure,
+                   const std::string& name,
+                   double value,
+                   const std::string& alg_name) {
+  Plot(figure, name, value, 0, alg_name);
+}
+
+void Logging::Plot(int figure,
+                   const std::string& name,
+                   double value,
+                   uint32_t ssrc,
+                   const std::string& alg_name) {
+  MutexLock lock(&mutex_);
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  RTC_DCHECK(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("PLOT\t%d\t%s:%" PRIu32 "@%s\t%f\t%f\n", figure, name.c_str(), ssrc,
+           alg_name.c_str(), state.timestamp_ms * 0.001, value);
+  }
+}
+
+void Logging::PlotBar(int figure,
+                      const std::string& name,
+                      double value,
+                      int flow_id) {
+  MutexLock lock(&mutex_);
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  RTC_DCHECK(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("BAR\t%d\t%s_%d\t%f\n", figure, name.c_str(), flow_id, value);
+  }
+}
+
+void Logging::PlotBaselineBar(int figure,
+                              const std::string& name,
+                              double value,
+                              int flow_id) {
+  MutexLock lock(&mutex_);
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  RTC_DCHECK(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("BASELINE\t%d\t%s_%d\t%f\n", figure, name.c_str(), flow_id, value);
+  }
+}
+
+void Logging::PlotErrorBar(int figure,
+                           const std::string& name,
+                           double value,
+                           double ylow,
+                           double yhigh,
+                           const std::string& error_title,
+                           int flow_id) {
+  MutexLock lock(&mutex_);
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  RTC_DCHECK(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("ERRORBAR\t%d\t%s_%d\t%f\t%f\t%f\t%s\n", figure, name.c_str(),
+           flow_id, value, ylow, yhigh, error_title.c_str());
+  }
+}
+
+void Logging::PlotLimitErrorBar(int figure,
+                                const std::string& name,
+                                double value,
+                                double ylow,
+                                double yhigh,
+                                const std::string& error_title,
+                                double ymax,
+                                const std::string& limit_title,
+                                int flow_id) {
+  MutexLock lock(&mutex_);
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  RTC_DCHECK(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("LIMITERRORBAR\t%d\t%s_%d\t%f\t%f\t%f\t%s\t%f\t%s\n", figure,
+           name.c_str(), flow_id, value, ylow, yhigh, error_title.c_str(), ymax,
+           limit_title.c_str());
+  }
+}
+
+void Logging::PlotLabel(int figure,
+                        const std::string& title,
+                        const std::string& y_label,
+                        int num_flows) {
+  MutexLock lock(&mutex_);
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  RTC_DCHECK(it != thread_map_.end());
+  const State& state = it->second.stack.top();
+  if (state.enabled) {
+    printf("LABEL\t%d\t%s\t%s\t%d\n", figure, title.c_str(), y_label.c_str(),
+           num_flows);
+  }
+}
+
+Logging::Logging() : thread_map_() {}
+
+Logging::~Logging() = default;
+
+Logging::State::State() : tag(""), timestamp_ms(0), enabled(true) {}
+
+Logging::State::State(const std::string& tag,
+                      int64_t timestamp_ms,
+                      bool enabled)
+    : tag(tag), timestamp_ms(timestamp_ms), enabled(enabled) {}
+
+void Logging::State::MergePrevious(const State& previous) {
+  if (tag.empty()) {
+    tag = previous.tag;
+  } else if (!previous.tag.empty()) {
+    tag = previous.tag + "_" + tag;
+  }
+  timestamp_ms = std::max(previous.timestamp_ms, timestamp_ms);
+  enabled = previous.enabled && enabled;
+}
+
+void Logging::PushState(const std::string& append_to_tag,
+                        int64_t timestamp_ms,
+                        bool enabled) {
+  MutexLock lock(&mutex_);
+  State new_state(append_to_tag, timestamp_ms, enabled);
+  ThreadState* thread_state = &thread_map_[rtc::CurrentThreadId()];
+  std::stack<State>* stack = &thread_state->stack;
+  if (stack->empty()) {
+    new_state.MergePrevious(thread_state->global_state);
+  } else {
+    new_state.MergePrevious(stack->top());
+  }
+  stack->push(new_state);
+}
+
+void Logging::PopState() {
+  MutexLock lock(&mutex_);
+  ThreadMap::iterator it = thread_map_.find(rtc::CurrentThreadId());
+  RTC_DCHECK(it != thread_map_.end());
+  std::stack<State>* stack = &it->second.stack;
+  int64_t newest_timestamp_ms = stack->top().timestamp_ms;
+  stack->pop();
+  if (!stack->empty()) {
+    State* state = &stack->top();
+    // Update time so that next log/plot will use the latest time seen so far
+    // in this call tree.
+    state->timestamp_ms = std::max(state->timestamp_ms, newest_timestamp_ms);
+  }
+}
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
+
+#endif  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
new file mode 100644
index 0000000000..49e1e716b2
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
@@ -0,0 +1,360 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_TEST_LOGGING_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_TEST_LOGGING_H_
+
+// To enable BWE logging, run this command from trunk/ :
+// build/gyp_chromium --depth=. webrtc/modules/modules.gyp
+//   -Denable_bwe_test_logging=1
+#ifndef BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+#define BWE_TEST_LOGGING_COMPILE_TIME_ENABLE 0
+#endif  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+
+// BWE logging allows you to insert dynamically named log/plot points in the
+// call tree. E.g. the function:
+//  void f1() {
+//    BWE_TEST_LOGGING_TIME(clock_->TimeInMilliseconds());
+//    BWE_TEST_LOGGING_CONTEXT("stream");
+//    for (uint32_t i=0; i<4; ++i) {
+//      BWE_TEST_LOGGING_ENABLE(i & 1);
+//      BWE_TEST_LOGGING_CONTEXT(i);
+//      BWE_TEST_LOGGING_LOG1("weight", "%f tonnes", weights_[i]);
+//      for (float j=0.0f; j<1.0; j+=0.4f) {
+//        BWE_TEST_LOGGING_PLOT(0, "bps", -1, j);
+//      }
+//    }
+//  }
+//
+// Might produce the output:
+//   stream_00000001_weight 13.000000 tonnes
+//   PLOT  stream_00000001_bps  1.000000  0.000000
+//   PLOT  stream_00000001_bps  1.000000  0.400000
+//   PLOT  stream_00000001_bps  1.000000  0.800000
+//   stream_00000003_weight 39.000000 tonnes
+//   PLOT  stream_00000003_bps  1.000000  0.000000
+//   PLOT  stream_00000003_bps  1.000000  0.400000
+//   PLOT  stream_00000003_bps  1.000000  0.800000
+//
+// Log *contexts* are names concatenated with '_' between them, with the name
+// of the logged/plotted string/value last. Plot *time* is inherited down the
+// tree. A branch is enabled by default but can be *disabled* to reduce output.
+// The difference between the RTC_LOG and PLOT macros is that PLOT prefixes the
+// line so it can be easily filtered, plus it outputs the current time.
+
+#if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
+
+// Set a thread-global base logging context. This name will be prepended to all
+// hierarchical contexts.
+// `name` is a char*, std::string or uint32_t to name the context.
+#define BWE_TEST_LOGGING_GLOBAL_CONTEXT(name)
+
+// Thread-globally allow/disallow logging.
+// `enable` is expected to be a bool.
+#define BWE_TEST_LOGGING_GLOBAL_ENABLE(enabled)
+
+// Insert a (hierarchical) logging context.
+// `name` is a char*, std::string or uint32_t to name the context.
+#define BWE_TEST_LOGGING_CONTEXT(name)
+
+// Allow/disallow logging down the call tree from this point. Logging must be
+// enabled all the way to the root of the call tree to take place.
+// `enable` is expected to be a bool.
+#define BWE_TEST_LOGGING_ENABLE(enabled)
+
+// Set current time (only affects PLOT output). Down the call tree, the latest
+// time set always takes precedence.
+// `time` is an int64_t time in ms, or -1 to inherit time from previous context.
+#define BWE_TEST_LOGGING_TIME(time)
+
+// Print to stdout, e.g.:
+//   Context1_Context2_Name  printf-formated-string
+// `name` is a char*, std::string or uint32_t to name the log line.
+// `format` is a printf format string.
+// |_1...| are arguments for printf.
+#define BWE_TEST_LOGGING_LOG1(name, format, _1)
+#define BWE_TEST_LOGGING_LOG2(name, format, _1, _2)
+#define BWE_TEST_LOGGING_LOG3(name, format, _1, _2, _3)
+#define BWE_TEST_LOGGING_LOG4(name, format, _1, _2, _3, _4)
+#define BWE_TEST_LOGGING_LOG5(name, format, _1, _2, _3, _4, _5)
+
+// Print to stdout in tab-separated format suitable for plotting, e.g.:
+//   PLOT figure Context1_Context2_Name  time  value
+// `figure` is a figure id. Different figures are plotted in different windows.
+// `name` is a char*, std::string or uint32_t to name the plotted value.
+// `time` is an int64_t time in ms, or -1 to inherit time from previous context.
+// `value` is a double precision float to be plotted.
+// `ssrc` identifies the source of a stream
+// `alg_name` is an optional argument, a string
+#define BWE_TEST_LOGGING_PLOT(figure, name, time, value)
+#define BWE_TEST_LOGGING_PLOT_WITH_NAME(figure, name, time, value, alg_name)
+#define BWE_TEST_LOGGING_PLOT_WITH_SSRC(figure, name, time, value, ssrc)
+#define BWE_TEST_LOGGING_PLOT_WITH_NAME_AND_SSRC(figure, name, time, value, \
+                                                 ssrc, alg_name)
+
+// Print to stdout in tab-separated format suitable for plotting, e.g.:
+//   BAR figure Context1_Context2_Name  x_left  width  value
+// `figure` is a figure id. Different figures are plotted in different windows.
+// `name` is a char*, std::string or uint32_t to name the plotted value.
+// `value` is a double precision float to be plotted.
+// `ylow` and `yhigh` are double precision float for the error line.
+// `title` is a string and refers to the error label.
+// `ymax` is a double precision float for the limit horizontal line.
+// `limit_title` is a string and refers to the limit label.
+#define BWE_TEST_LOGGING_BAR(figure, name, value, flow_id)
+#define BWE_TEST_LOGGING_ERRORBAR(figure, name, value, ylow, yhigh, \
+                                  error_title, flow_id)
+#define BWE_TEST_LOGGING_LIMITERRORBAR( \
+    figure, name, value, ylow, yhigh, error_title, ymax, limit_title, flow_id)
+
+#define BWE_TEST_LOGGING_BASELINEBAR(figure, name, value, flow_id)
+
+// `num_flows` is an integer refering to the number of RMCAT flows in the
+// scenario.
+// Define `x_label` and `y_label` for plots.
+#define BWE_TEST_LOGGING_LABEL(figure, x_label, y_label, num_flows)
+
+#else  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+
+#include <map>
+#include <memory>
+#include <stack>
+#include <string>
+
+#include "rtc_base/synchronization/mutex.h"
+
+#define BWE_TEST_LOGGING_GLOBAL_CONTEXT(name)                             \
+  do {                                                                    \
+    webrtc::testing::bwe::Logging::GetInstance()->SetGlobalContext(name); \
+  } while (0)
+
+#define BWE_TEST_LOGGING_GLOBAL_ENABLE(enabled)                             \
+  do {                                                                      \
+    webrtc::testing::bwe::Logging::GetInstance()->SetGlobalEnable(enabled); \
+  } while (0)
+
+#define __BWE_TEST_LOGGING_CONTEXT_NAME(ctx, line) ctx##line
+#define __BWE_TEST_LOGGING_CONTEXT_DECLARE(ctx, line, name, time, enabled) \
+  webrtc::testing::bwe::Logging::Context __BWE_TEST_LOGGING_CONTEXT_NAME(  \
+      ctx, line)(name, time, enabled)
+
+#define BWE_TEST_LOGGING_CONTEXT(name) \
+  __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, name, -1, true)
+#define BWE_TEST_LOGGING_ENABLE(enabled)                           \
+  __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, "", -1, \
+                                     static_cast<bool>(enabled))
+#define BWE_TEST_LOGGING_TIME(time)                            \
+  __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, "", \
+                                     static_cast<int64_t>(time), true)
+
+#define BWE_TEST_LOGGING_LOG1(name, format, _1)                    \
+  do {                                                             \
+    BWE_TEST_LOGGING_CONTEXT(name);                                \
+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1); \
+  } while (0)
+#define BWE_TEST_LOGGING_LOG2(name, format, _1, _2)                    \
+  do {                                                                 \
+    BWE_TEST_LOGGING_CONTEXT(name);                                    \
+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2); \
+  } while (0)
+#define BWE_TEST_LOGGING_LOG3(name, format, _1, _2, _3)                    \
+  do {                                                                     \
+    BWE_TEST_LOGGING_CONTEXT(name);                                        \
+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3); \
+  } while (0)
+#define BWE_TEST_LOGGING_LOG4(name, format, _1, _2, _3, _4)                    \
+  do {                                                                         \
+    BWE_TEST_LOGGING_CONTEXT(name);                                            \
+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, _4); \
+  } while (0)
+#define BWE_TEST_LOGGING_LOG5(name, format, _1, _2, _3, _4, _5)               \
+  do {                                                                        \
+    BWE_TEST_LOGGING_CONTEXT(name);                                           \
+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, _4, \
+                                                      _5);                    \
+  } while (0)
+
+#define BWE_TEST_LOGGING_PLOT(figure, name, time, value)                     \
+  do {                                                                       \
+    __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __PLOT__, name,           \
+                                       static_cast<int64_t>(time), true);    \
+    webrtc::testing::bwe::Logging::GetInstance()->Plot(figure, name, value); \
+  } while (0)
+
+#define BWE_TEST_LOGGING_PLOT_WITH_NAME(figure, name, time, value, alg_name) \
+  do {                                                                       \
+    __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __PLOT__, name,           \
+                                       static_cast<int64_t>(time), true);    \
+    webrtc::testing::bwe::Logging::GetInstance()->Plot(figure, name, value,  \
+                                                       alg_name);            \
+  } while (0)
+
+#define BWE_TEST_LOGGING_PLOT_WITH_SSRC(figure, name, time, value, ssrc)    \
+  do {                                                                      \
+    __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __PLOT__, name,          \
+                                       static_cast<int64_t>(time), true);   \
+    webrtc::testing::bwe::Logging::GetInstance()->Plot(figure, name, value, \
+                                                       ssrc);               \
+  } while (0)
+
+#define BWE_TEST_LOGGING_PLOT_WITH_NAME_AND_SSRC(figure, name, time, value, \
+                                                 ssrc, alg_name)            \
+  do {                                                                      \
+    __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __PLOT__, name,          \
+                                       static_cast<int64_t>(time), true);   \
+    webrtc::testing::bwe::Logging::GetInstance()->Plot(figure, name, value, \
+                                                       ssrc, alg_name);     \
+  } while (0)
+
+#define BWE_TEST_LOGGING_BAR(figure, name, value, flow_id)                     \
+  do {                                                                         \
+    BWE_TEST_LOGGING_CONTEXT(name);                                            \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotBar(figure, name, value, \
+                                                          flow_id);            \
+  } while (0)
+
+#define BWE_TEST_LOGGING_BASELINEBAR(figure, name, value, flow_id) \
+  do {                                                             \
+    BWE_TEST_LOGGING_CONTEXT(name);                                \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotBaselineBar( \
+        figure, name, value, flow_id);                             \
+  } while (0)
+
+#define BWE_TEST_LOGGING_ERRORBAR(figure, name, value, ylow, yhigh, title, \
+                                  flow_id)                                 \
+  do {                                                                     \
+    BWE_TEST_LOGGING_CONTEXT(name);                                        \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotErrorBar(            \
+        figure, name, value, ylow, yhigh, title, flow_id);                 \
+  } while (0)
+
+#define BWE_TEST_LOGGING_LIMITERRORBAR(                                        \
+    figure, name, value, ylow, yhigh, error_title, ymax, limit_title, flow_id) \
+  do {                                                                         \
+    BWE_TEST_LOGGING_CONTEXT(name);                                            \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotLimitErrorBar(           \
+        figure, name, value, ylow, yhigh, error_title, ymax, limit_title,      \
+        flow_id);                                                              \
+  } while (0)
+
+#define BWE_TEST_LOGGING_LABEL(figure, title, y_label, num_flows) \
+  do {                                                            \
+    BWE_TEST_LOGGING_CONTEXT(title);                              \
+    webrtc::testing::bwe::Logging::GetInstance()->PlotLabel(      \
+        figure, title, y_label, num_flows);                       \
+  } while (0)
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+
+class Logging {
+ public:
+  class Context {
+   public:
+    Context(uint32_t name, int64_t timestamp_ms, bool enabled);
+    Context(const std::string& name, int64_t timestamp_ms, bool enabled);
+    Context(const char* name, int64_t timestamp_ms, bool enabled);
+
+    Context() = delete;
+    Context(const Context&) = delete;
+    Context& operator=(const Context&) = delete;
+    ~Context();
+  };
+
+  static Logging* GetInstance();
+
+  void SetGlobalContext(uint32_t name);
+  void SetGlobalContext(const std::string& name);
+  void SetGlobalContext(const char* name);
+  void SetGlobalEnable(bool enabled);
+
+#if defined(__GNUC__)
+  // Note: Implicit `this` argument counts as the first argument.
+  __attribute__((__format__(__printf__, 2, 3)))
+#endif
+  void
+  Log(const char format[], ...);
+  void Plot(int figure, const std::string& name, double value);
+  void Plot(int figure,
+            const std::string& name,
+            double value,
+            const std::string& alg_name);
+  void Plot(int figure, const std::string& name, double value, uint32_t ssrc);
+  void Plot(int figure,
+            const std::string& name,
+            double value,
+            uint32_t ssrc,
+            const std::string& alg_name);
+  void PlotBar(int figure, const std::string& name, double value, int flow_id);
+  void PlotBaselineBar(int figure,
+                       const std::string& name,
+                       double value,
+                       int flow_id);
+  void PlotErrorBar(int figure,
+                    const std::string& name,
+                    double value,
+                    double ylow,
+                    double yhigh,
+                    const std::string& error_title,
+                    int flow_id);
+
+  void PlotLimitErrorBar(int figure,
+                         const std::string& name,
+                         double value,
+                         double ylow,
+                         double yhigh,
+                         const std::string& error_title,
+                         double ymax,
+                         const std::string& limit_title,
+                         int flow_id);
+  void PlotLabel(int figure,
+                 const std::string& title,
+                 const std::string& y_label,
+                 int num_flows);
+
+ private:
+  struct State {
+    State();
+    State(const std::string& new_tag, int64_t timestamp_ms, bool enabled);
+    void MergePrevious(const State& previous);
+
+    std::string tag;
+    int64_t timestamp_ms;
+    bool enabled;
+  };
+  struct ThreadState {
+    ThreadState();
+    ~ThreadState();
+    State global_state;
+    std::stack<State> stack;
+  };
+  typedef std::map<uint32_t, ThreadState> ThreadMap;
+
+  Logging();
+  ~Logging();
+
+  Logging(const Logging&) = delete;
+  Logging& operator=(const Logging&) = delete;
+
+  void PushState(const std::string& append_to_tag,
+                 int64_t timestamp_ms,
+                 bool enabled);
+  void PopState();
+
+  Mutex mutex_;
+  ThreadMap thread_map_;
+};
+}  // namespace bwe
+}  // namespace testing
+}  // namespace webrtc
+
+#endif  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_TEST_LOGGING_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
new file mode 100644
index 0000000000..403f81fd03
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/remote_bitrate_estimator/tools/bwe_rtp.h"
+
+#include <stdio.h>
+
+#include <set>
+#include <sstream>
+#include <string>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "test/rtp_file_reader.h"
+
+ABSL_FLAG(std::string,
+          extension_type,
+          "abs",
+          "Extension type, either abs for absolute send time or tsoffset "
+          "for timestamp offset.");
+std::string ExtensionType() {
+  return absl::GetFlag(FLAGS_extension_type);
+}
+
+ABSL_FLAG(int, extension_id, 3, "Extension id.");
+int ExtensionId() {
+  return absl::GetFlag(FLAGS_extension_id);
+}
+
+ABSL_FLAG(std::string, input_file, "", "Input file.");
+std::string InputFile() {
+  return absl::GetFlag(FLAGS_input_file);
+}
+
+ABSL_FLAG(std::string,
+          ssrc_filter,
+          "",
+          "Comma-separated list of SSRCs in hexadecimal which are to be "
+          "used as input to the BWE (only applicable to pcap files).");
+std::set<uint32_t> SsrcFilter() {
+  std::string ssrc_filter_string = absl::GetFlag(FLAGS_ssrc_filter);
+  if (ssrc_filter_string.empty())
+    return std::set<uint32_t>();
+  std::stringstream ss;
+  std::string ssrc_filter = ssrc_filter_string;
+  std::set<uint32_t> ssrcs;
+
+  // Parse the ssrcs in hexadecimal format.
+  ss << std::hex << ssrc_filter;
+  uint32_t ssrc;
+  while (ss >> ssrc) {
+    ssrcs.insert(ssrc);
+    ss.ignore(1, ',');
+  }
+  return ssrcs;
+}
+
+bool ParseArgsAndSetupRtpReader(
+    int argc,
+    char** argv,
+    std::unique_ptr<webrtc::test::RtpFileReader>& rtp_reader,
+    webrtc::RtpHeaderExtensionMap& rtp_header_extensions) {
+  absl::ParseCommandLine(argc, argv);
+  std::string filename = InputFile();
+
+  std::set<uint32_t> ssrc_filter = SsrcFilter();
+  fprintf(stderr, "Filter on SSRC: ");
+  for (auto& s : ssrc_filter) {
+    fprintf(stderr, "0x%08x, ", s);
+  }
+  fprintf(stderr, "\n");
+  if (filename.substr(filename.find_last_of('.')) == ".pcap") {
+    fprintf(stderr, "Opening as pcap\n");
+    rtp_reader.reset(webrtc::test::RtpFileReader::Create(
+        webrtc::test::RtpFileReader::kPcap, filename.c_str(), SsrcFilter()));
+  } else {
+    fprintf(stderr, "Opening as rtp\n");
+    rtp_reader.reset(webrtc::test::RtpFileReader::Create(
+        webrtc::test::RtpFileReader::kRtpDump, filename.c_str()));
+  }
+  if (!rtp_reader) {
+    fprintf(stderr, "Cannot open input file %s\n", filename.c_str());
+    return false;
+  }
+  fprintf(stderr, "Input file: %s\n\n", filename.c_str());
+
+  webrtc::RTPExtensionType extension = webrtc::kRtpExtensionAbsoluteSendTime;
+  if (ExtensionType() == "tsoffset") {
+    extension = webrtc::kRtpExtensionTransmissionTimeOffset;
+    fprintf(stderr, "Extension: toffset\n");
+  } else if (ExtensionType() == "abs") {
+    fprintf(stderr, "Extension: abs\n");
+  } else {
+    fprintf(stderr, "Unknown extension type\n");
+    return false;
+  }
+
+  rtp_header_extensions.RegisterByType(ExtensionId(), extension);
+
+  return true;
+}
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h
new file mode 100644
index 0000000000..3b161db37b
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_
+
+#include <memory>
+
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "test/rtp_file_reader.h"
+
+bool ParseArgsAndSetupRtpReader(
+    int argc,
+    char** argv,
+    std::unique_ptr<webrtc::test::RtpFileReader>& rtp_reader,
+    webrtc::RtpHeaderExtensionMap& rtp_header_extensions);
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
new file mode 100644
index 0000000000..e8dc59f740
--- /dev/null
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "modules/remote_bitrate_estimator/tools/bwe_rtp.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/rtp_file_reader.h"
+
+int main(int argc, char* argv[]) {
+  std::unique_ptr<webrtc::test::RtpFileReader> reader;
+  webrtc::RtpHeaderExtensionMap rtp_header_extensions;
+  if (!ParseArgsAndSetupRtpReader(argc, argv, reader, rtp_header_extensions)) {
+    return -1;
+  }
+
+  bool arrival_time_only = (argc >= 5 && strncmp(argv[4], "-t", 2) == 0);
+
+  fprintf(stdout,
+          "seqnum timestamp ts_offset abs_sendtime recvtime "
+          "markerbit ssrc size original_size\n");
+  int packet_counter = 0;
+  int non_zero_abs_send_time = 0;
+  int non_zero_ts_offsets = 0;
+  webrtc::test::RtpPacket packet;
+  while (reader->NextPacket(&packet)) {
+    webrtc::RtpPacket header(&rtp_header_extensions);
+    header.Parse(packet.data, packet.length);
+    uint32_t abs_send_time = 0;
+    if (header.GetExtension<webrtc::AbsoluteSendTime>(&abs_send_time) &&
+        abs_send_time != 0)
+      ++non_zero_abs_send_time;
+    int32_t toffset = 0;
+    if (header.GetExtension<webrtc::TransmissionOffset>(&toffset) &&
+        toffset != 0)
+      ++non_zero_ts_offsets;
+    if (arrival_time_only) {
+      rtc::StringBuilder ss;
+      ss << static_cast<int64_t>(packet.time_ms) * 1000000;
+      fprintf(stdout, "%s\n", ss.str().c_str());
+    } else {
+      fprintf(stdout, "%u %u %d %u %u %d %u %zu %zu\n", header.SequenceNumber(),
+              header.Timestamp(), toffset, abs_send_time, packet.time_ms,
+              header.Marker(), header.Ssrc(), packet.length,
+              packet.original_length);
+    }
+    ++packet_counter;
+  }
+  fprintf(stderr, "Parsed %d packets\n", packet_counter);
+  fprintf(stderr, "Packets with non-zero absolute send time: %d\n",
+          non_zero_abs_send_time);
+  fprintf(stderr, "Packets with non-zero timestamp offset: %d\n",
+          non_zero_ts_offsets);
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/BUILD.gn b/third_party/libwebrtc/modules/rtp_rtcp/BUILD.gn
new file mode 100644
index 0000000000..6014ae420b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/BUILD.gn
@@ -0,0 +1,713 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("leb128") {
+  public = [ "source/leb128.h" ]
+  sources = [ "source/leb128.cc" ]
+}
+
+rtc_library("rtp_rtcp_format") {
+  visibility = [ "*" ]
+  public = [
+    "include/recovered_packet_receiver.h",
+    "include/report_block_data.h",
+    "include/rtcp_statistics.h",
+    "include/rtp_cvo.h",
+    "include/rtp_header_extension_map.h",
+    "include/rtp_packet_sender.h",
+    "include/rtp_rtcp_defines.h",
+    "source/byte_io.h",
+    "source/rtcp_packet.h",
+    "source/rtcp_packet/app.h",
+    "source/rtcp_packet/bye.h",
+    "source/rtcp_packet/common_header.h",
+    "source/rtcp_packet/compound_packet.h",
+    "source/rtcp_packet/dlrr.h",
+    "source/rtcp_packet/extended_reports.h",
+    "source/rtcp_packet/fir.h",
+    "source/rtcp_packet/loss_notification.h",
+    "source/rtcp_packet/nack.h",
+    "source/rtcp_packet/pli.h",
+    "source/rtcp_packet/psfb.h",
+    "source/rtcp_packet/rapid_resync_request.h",
+    "source/rtcp_packet/receiver_report.h",
+    "source/rtcp_packet/remb.h",
+    "source/rtcp_packet/remote_estimate.h",
+    "source/rtcp_packet/report_block.h",
+    "source/rtcp_packet/rrtr.h",
+    "source/rtcp_packet/rtpfb.h",
+    "source/rtcp_packet/sdes.h",
+    "source/rtcp_packet/sender_report.h",
+    "source/rtcp_packet/target_bitrate.h",
+    "source/rtcp_packet/tmmb_item.h",
+    "source/rtcp_packet/tmmbn.h",
+    "source/rtcp_packet/tmmbr.h",
+    "source/rtcp_packet/transport_feedback.h",
+    "source/rtp_dependency_descriptor_extension.h",
+    "source/rtp_generic_frame_descriptor.h",
+    "source/rtp_generic_frame_descriptor_extension.h",
+    "source/rtp_header_extensions.h",
+    "source/rtp_packet.h",
+    "source/rtp_packet_received.h",
+    "source/rtp_packet_to_send.h",
+    "source/rtp_util.h",
+    "source/rtp_video_layers_allocation_extension.h",
+  ]
+  sources = [
+    "include/report_block_data.cc",
+    "include/rtp_rtcp_defines.cc",
+    "source/rtcp_packet.cc",
+    "source/rtcp_packet/app.cc",
+    "source/rtcp_packet/bye.cc",
+    "source/rtcp_packet/common_header.cc",
+    "source/rtcp_packet/compound_packet.cc",
+    "source/rtcp_packet/dlrr.cc",
+    "source/rtcp_packet/extended_reports.cc",
+    "source/rtcp_packet/fir.cc",
+    "source/rtcp_packet/loss_notification.cc",
+    "source/rtcp_packet/nack.cc",
+    "source/rtcp_packet/pli.cc",
+    "source/rtcp_packet/psfb.cc",
+    "source/rtcp_packet/rapid_resync_request.cc",
+    "source/rtcp_packet/receiver_report.cc",
+    "source/rtcp_packet/remb.cc",
+    "source/rtcp_packet/remote_estimate.cc",
+    "source/rtcp_packet/report_block.cc",
+    "source/rtcp_packet/rrtr.cc",
+    "source/rtcp_packet/rtpfb.cc",
+    "source/rtcp_packet/sdes.cc",
+    "source/rtcp_packet/sender_report.cc",
+    "source/rtcp_packet/target_bitrate.cc",
+    "source/rtcp_packet/tmmb_item.cc",
+    "source/rtcp_packet/tmmbn.cc",
+    "source/rtcp_packet/tmmbr.cc",
+    "source/rtcp_packet/transport_feedback.cc",
+    "source/rtp_dependency_descriptor_extension.cc",
+    "source/rtp_dependency_descriptor_reader.cc",
+    "source/rtp_dependency_descriptor_reader.h",
+    "source/rtp_dependency_descriptor_writer.cc",
+    "source/rtp_dependency_descriptor_writer.h",
+    "source/rtp_generic_frame_descriptor.cc",
+    "source/rtp_generic_frame_descriptor_extension.cc",
+    "source/rtp_header_extension_map.cc",
+    "source/rtp_header_extensions.cc",
+    "source/rtp_packet.cc",
+    "source/rtp_packet_received.cc",
+    "source/rtp_packet_to_send.cc",
+    "source/rtp_util.cc",
+    "source/rtp_video_layers_allocation_extension.cc",
+  ]
+
+  deps = [
+    ":leb128",
+    "..:module_api_public",
+    "../../api:array_view",
+    "../../api:function_view",
+    "../../api:refcountedbase",
+    "../../api:rtp_headers",
+    "../../api:rtp_parameters",
+    "../../api:scoped_refptr",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/transport:network_control",
+    "../../api/transport/rtp:dependency_descriptor",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../api/video:video_frame",
+    "../../api/video:video_layers_allocation",
+    "../../api/video:video_rtp_headers",
+    "../../common_video",
+    "../../rtc_base:bit_buffer",
+    "../../rtc_base:bitstream_reader",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:copy_on_write_buffer",
+    "../../rtc_base:divide_round",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:stringutils",
+    "../../system_wrappers",
+    "../video_coding:codec_globals_headers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+    "//third_party/abseil-cpp/absl/types:variant",
+  ]
+}
+
+rtc_library("rtp_rtcp") {
+  visibility = [ "*" ]
+  sources = [
+    "include/flexfec_receiver.h",
+    "include/flexfec_sender.h",
+    "include/receive_statistics.h",
+    "include/remote_ntp_time_estimator.h",
+    "source/absolute_capture_time_interpolator.cc",
+    "source/absolute_capture_time_interpolator.h",
+    "source/absolute_capture_time_sender.cc",
+    "source/absolute_capture_time_sender.h",
+    "source/active_decode_targets_helper.cc",
+    "source/active_decode_targets_helper.h",
+    "source/capture_clock_offset_updater.cc",
+    "source/capture_clock_offset_updater.h",
+    "source/create_video_rtp_depacketizer.cc",
+    "source/create_video_rtp_depacketizer.h",
+    "source/dtmf_queue.cc",
+    "source/dtmf_queue.h",
+    "source/fec_private_tables_bursty.cc",
+    "source/fec_private_tables_bursty.h",
+    "source/fec_private_tables_random.cc",
+    "source/fec_private_tables_random.h",
+    "source/flexfec_header_reader_writer.cc",
+    "source/flexfec_header_reader_writer.h",
+    "source/flexfec_receiver.cc",
+    "source/flexfec_sender.cc",
+    "source/forward_error_correction.cc",
+    "source/forward_error_correction.h",
+    "source/forward_error_correction_internal.cc",
+    "source/forward_error_correction_internal.h",
+    "source/packet_loss_stats.cc",
+    "source/packet_loss_stats.h",
+    "source/packet_sequencer.cc",
+    "source/packet_sequencer.h",
+    "source/receive_statistics_impl.cc",
+    "source/receive_statistics_impl.h",
+    "source/remote_ntp_time_estimator.cc",
+    "source/rtcp_nack_stats.cc",
+    "source/rtcp_nack_stats.h",
+    "source/rtcp_receiver.cc",
+    "source/rtcp_receiver.h",
+    "source/rtcp_sender.cc",
+    "source/rtcp_sender.h",
+    "source/rtp_descriptor_authentication.cc",
+    "source/rtp_descriptor_authentication.h",
+    "source/rtp_format.cc",
+    "source/rtp_format.h",
+    "source/rtp_format_h264.cc",
+    "source/rtp_format_h264.h",
+    "source/rtp_format_video_generic.cc",
+    "source/rtp_format_video_generic.h",
+    "source/rtp_format_vp8.cc",
+    "source/rtp_format_vp8.h",
+    "source/rtp_format_vp9.cc",
+    "source/rtp_format_vp9.h",
+    "source/rtp_header_extension_size.cc",
+    "source/rtp_header_extension_size.h",
+    "source/rtp_packet_history.cc",
+    "source/rtp_packet_history.h",
+    "source/rtp_packetizer_av1.cc",
+    "source/rtp_packetizer_av1.h",
+    "source/rtp_rtcp_config.h",
+    "source/rtp_rtcp_impl2.cc",
+    "source/rtp_rtcp_impl2.h",
+    "source/rtp_rtcp_interface.h",
+    "source/rtp_sender.cc",
+    "source/rtp_sender.h",
+    "source/rtp_sender_audio.cc",
+    "source/rtp_sender_audio.h",
+    "source/rtp_sender_egress.cc",
+    "source/rtp_sender_egress.h",
+    "source/rtp_sender_video.cc",
+    "source/rtp_sender_video.h",
+    "source/rtp_sender_video_frame_transformer_delegate.cc",
+    "source/rtp_sender_video_frame_transformer_delegate.h",
+    "source/rtp_sequence_number_map.cc",
+    "source/rtp_sequence_number_map.h",
+    "source/source_tracker.cc",
+    "source/source_tracker.h",
+    "source/time_util.cc",
+    "source/time_util.h",
+    "source/tmmbr_help.cc",
+    "source/tmmbr_help.h",
+    "source/ulpfec_generator.cc",
+    "source/ulpfec_generator.h",
+    "source/ulpfec_header_reader_writer.cc",
+    "source/ulpfec_header_reader_writer.h",
+    "source/ulpfec_receiver.cc",
+    "source/ulpfec_receiver.h",
+    "source/video_fec_generator.h",
+    "source/video_rtp_depacketizer.cc",
+    "source/video_rtp_depacketizer.h",
+    "source/video_rtp_depacketizer_av1.cc",
+    "source/video_rtp_depacketizer_av1.h",
+    "source/video_rtp_depacketizer_generic.cc",
+    "source/video_rtp_depacketizer_generic.h",
+    "source/video_rtp_depacketizer_h264.cc",
+    "source/video_rtp_depacketizer_h264.h",
+    "source/video_rtp_depacketizer_raw.cc",
+    "source/video_rtp_depacketizer_raw.h",
+    "source/video_rtp_depacketizer_vp8.cc",
+    "source/video_rtp_depacketizer_vp8.h",
+    "source/video_rtp_depacketizer_vp9.cc",
+    "source/video_rtp_depacketizer_vp9.h",
+  ]
+
+  if (rtc_enable_bwe_test_logging) {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
+  } else {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
+  }
+
+  deps = [
+    ":leb128",
+    ":rtp_rtcp_format",
+    ":rtp_video_header",
+    "..:module_api_public",
+    "..:module_fec_api",
+    "../../api:array_view",
+    "../../api:field_trials_view",
+    "../../api:frame_transformer_interface",
+    "../../api:function_view",
+    "../../api:libjingle_peerconnection_api",
+    "../../api:rtp_headers",
+    "../../api:rtp_packet_info",
+    "../../api:rtp_parameters",
+    "../../api:scoped_refptr",
+    "../../api:sequence_checker",
+    "../../api:transport_api",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/crypto:frame_encryptor_interface",
+    "../../api/rtc_event_log",
+    "../../api/task_queue:pending_task_safety_flag",
+    "../../api/task_queue:task_queue",
+    "../../api/transport:field_trial_based_config",
+    "../../api/transport/rtp:dependency_descriptor",
+    "../../api/transport/rtp:rtp_source",
+    "../../api/units:data_rate",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../api/video:encoded_frame",
+    "../../api/video:encoded_image",
+    "../../api/video:video_bitrate_allocation",
+    "../../api/video:video_bitrate_allocator",
+    "../../api/video:video_codec_constants",
+    "../../api/video:video_frame",
+    "../../api/video:video_frame_type",
+    "../../api/video:video_layers_allocation",
+    "../../api/video:video_rtp_headers",
+    "../../api/video_codecs:video_codecs_api",
+    "../../call:rtp_interfaces",
+    "../../common_video",
+    "../../logging:rtc_event_audio",
+    "../../logging:rtc_event_rtp_rtcp",
+    "../../modules/audio_coding:audio_coding_module_typedefs",
+    "../../rtc_base:bit_buffer",
+    "../../rtc_base:bitstream_reader",
+    "../../rtc_base:buffer",
+    "../../rtc_base:byte_buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:copy_on_write_buffer",
+    "../../rtc_base:divide_round",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:gtest_prod",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:mod_ops",
+    "../../rtc_base:one_time_event",
+    "../../rtc_base:race_checker",
+    "../../rtc_base:random",
+    "../../rtc_base:rate_limiter",
+    "../../rtc_base:rate_statistics",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/containers:flat_map",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:no_unique_address",
+    "../../rtc_base/task_utils:repeating_task",
+    "../../system_wrappers",
+    "../../system_wrappers:metrics",
+    "../remote_bitrate_estimator",
+    "../video_coding:codec_globals_headers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+    "//third_party/abseil-cpp/absl/types:variant",
+  ]
+}
+
+rtc_source_set("rtp_rtcp_legacy") {
+  sources = [
+    "include/rtp_rtcp.h",
+    "source/deprecated/deprecated_rtp_sender_egress.cc",
+    "source/deprecated/deprecated_rtp_sender_egress.h",
+    "source/rtp_rtcp_impl.cc",
+    "source/rtp_rtcp_impl.h",
+  ]
+  deps = [
+    ":rtp_rtcp",
+    ":rtp_rtcp_format",
+    "..:module_fec_api",
+    "../../api:rtp_headers",
+    "../../api:transport_api",
+    "../../api/rtc_event_log",
+    "../../api/transport:field_trial_based_config",
+    "../../api/units:data_rate",
+    "../../api/units:timestamp",
+    "../../api/video:video_bitrate_allocation",
+    "../../logging:rtc_event_rtp_rtcp",
+    "../../rtc_base:checks",
+    "../../rtc_base:gtest_prod",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rate_statistics",
+    "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
+    "../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("rtcp_transceiver") {
+  visibility = [ "*" ]
+  public = [
+    "source/rtcp_transceiver.h",
+    "source/rtcp_transceiver_config.h",
+    "source/rtcp_transceiver_impl.h",
+  ]
+  sources = [
+    "source/rtcp_transceiver.cc",
+    "source/rtcp_transceiver_config.cc",
+    "source/rtcp_transceiver_impl.cc",
+  ]
+  deps = [
+    ":rtp_rtcp",
+    ":rtp_rtcp_format",
+    "../../api:array_view",
+    "../../api:rtp_headers",
+    "../../api:transport_api",
+    "../../api/task_queue",
+    "../../api/units:data_rate",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../api/video:video_bitrate_allocation",
+    "../../rtc_base:checks",
+    "../../rtc_base:copy_on_write_buffer",
+    "../../rtc_base:divide_round",
+    "../../rtc_base:logging",
+    "../../rtc_base:rtc_event",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/containers:flat_map",
+    "../../rtc_base/task_utils:repeating_task",
+    "../../system_wrappers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/cleanup",
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("rtp_video_header") {
+  visibility = [ "*" ]
+  sources = [
+    "source/rtp_video_header.cc",
+    "source/rtp_video_header.h",
+  ]
+  deps = [
+    "../../api:rtp_headers",
+    "../../api/transport/rtp:dependency_descriptor",
+    "../../api/video:video_frame",
+    "../../api/video:video_frame_metadata",
+    "../../api/video:video_frame_type",
+    "../../api/video:video_rtp_headers",
+    "../../modules/video_coding:codec_globals_headers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
+    "//third_party/abseil-cpp/absl/types:optional",
+    "//third_party/abseil-cpp/absl/types:variant",
+  ]
+}
+
+rtc_source_set("rtp_video_header_unittest") {
+  testonly = true
+  sources = [ "source/rtp_video_header_unittest.cc" ]
+  deps = [
+    ":rtp_video_header",
+    "../../api/video:video_frame_metadata",
+    "../../api/video:video_frame_type",
+    "../../test:test_support",
+  ]
+}
+
+rtc_library("fec_test_helper") {
+  testonly = true
+  sources = [
+    "source/fec_test_helper.cc",
+    "source/fec_test_helper.h",
+  ]
+  deps = [
+    ":rtp_rtcp",
+    ":rtp_rtcp_format",
+    "../../rtc_base:checks",
+    "../../rtc_base:random",
+  ]
+}
+
+rtc_library("mock_rtp_rtcp") {
+  testonly = true
+  public = [
+    "mocks/mock_recovered_packet_receiver.h",
+    "mocks/mock_rtcp_bandwidth_observer.h",
+    "mocks/mock_rtcp_rtt_stats.h",
+    "mocks/mock_rtp_rtcp.h",
+  ]
+  deps = [
+    ":rtp_rtcp",
+    ":rtp_rtcp_format",
+    "../../api/video:video_bitrate_allocation",
+    "../../rtc_base:checks",
+    "../../test:test_support",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("rtp_packetizer_av1_test_helper") {
+  testonly = true
+  sources = [
+    "source/rtp_packetizer_av1_test_helper.cc",
+    "source/rtp_packetizer_av1_test_helper.h",
+  ]
+}
+
+if (rtc_include_tests) {
+  if (!build_with_chromium) {
+    rtc_executable("test_packet_masks_metrics") {
+      testonly = true
+
+      sources = [
+        "test/testFec/average_residual_loss_xor_codes.h",
+        "test/testFec/test_packet_masks_metrics.cc",
+      ]
+
+      deps = [
+        ":rtp_rtcp",
+        "../../test:fileutils",
+        "../../test:test_main",
+        "../../test:test_support",
+        "//testing/gtest",
+      ]
+    }  # test_packet_masks_metrics
+  }
+
+  rtc_library("rtp_rtcp_modules_tests") {
+    testonly = true
+
+    sources = [ "test/testFec/test_fec.cc" ]
+    deps = [
+      ":rtp_rtcp",
+      ":rtp_rtcp_format",
+      "../../rtc_base:random",
+      "../../test:fileutils",
+      "../../test:test_support",
+    ]
+  }
+
+  rtc_library("rtp_rtcp_unittests") {
+    testonly = true
+
+    sources = [
+      "source/absolute_capture_time_interpolator_unittest.cc",
+      "source/absolute_capture_time_sender_unittest.cc",
+      "source/active_decode_targets_helper_unittest.cc",
+      "source/byte_io_unittest.cc",
+      "source/capture_clock_offset_updater_unittest.cc",
+      "source/fec_private_tables_bursty_unittest.cc",
+      "source/flexfec_header_reader_writer_unittest.cc",
+      "source/flexfec_receiver_unittest.cc",
+      "source/flexfec_sender_unittest.cc",
+      "source/leb128_unittest.cc",
+      "source/nack_rtx_unittest.cc",
+      "source/packet_loss_stats_unittest.cc",
+      "source/packet_sequencer_unittest.cc",
+      "source/receive_statistics_unittest.cc",
+      "source/remote_ntp_time_estimator_unittest.cc",
+      "source/rtcp_nack_stats_unittest.cc",
+      "source/rtcp_packet/app_unittest.cc",
+      "source/rtcp_packet/bye_unittest.cc",
+      "source/rtcp_packet/common_header_unittest.cc",
+      "source/rtcp_packet/compound_packet_unittest.cc",
+      "source/rtcp_packet/dlrr_unittest.cc",
+      "source/rtcp_packet/extended_reports_unittest.cc",
+      "source/rtcp_packet/fir_unittest.cc",
+      "source/rtcp_packet/loss_notification_unittest.cc",
+      "source/rtcp_packet/nack_unittest.cc",
+      "source/rtcp_packet/pli_unittest.cc",
+      "source/rtcp_packet/rapid_resync_request_unittest.cc",
+      "source/rtcp_packet/receiver_report_unittest.cc",
+      "source/rtcp_packet/remb_unittest.cc",
+      "source/rtcp_packet/remote_estimate_unittest.cc",
+      "source/rtcp_packet/report_block_unittest.cc",
+      "source/rtcp_packet/rrtr_unittest.cc",
+      "source/rtcp_packet/sdes_unittest.cc",
+      "source/rtcp_packet/sender_report_unittest.cc",
+      "source/rtcp_packet/target_bitrate_unittest.cc",
+      "source/rtcp_packet/tmmbn_unittest.cc",
+      "source/rtcp_packet/tmmbr_unittest.cc",
+      "source/rtcp_packet/transport_feedback_unittest.cc",
+      "source/rtcp_packet_unittest.cc",
+      "source/rtcp_receiver_unittest.cc",
+      "source/rtcp_sender_unittest.cc",
+      "source/rtcp_transceiver_impl_unittest.cc",
+      "source/rtcp_transceiver_unittest.cc",
+      "source/rtp_dependency_descriptor_extension_unittest.cc",
+      "source/rtp_fec_unittest.cc",
+      "source/rtp_format_h264_unittest.cc",
+      "source/rtp_format_unittest.cc",
+      "source/rtp_format_video_generic_unittest.cc",
+      "source/rtp_format_vp8_test_helper.cc",
+      "source/rtp_format_vp8_test_helper.h",
+      "source/rtp_format_vp8_unittest.cc",
+      "source/rtp_format_vp9_unittest.cc",
+      "source/rtp_generic_frame_descriptor_extension_unittest.cc",
+      "source/rtp_header_extension_map_unittest.cc",
+      "source/rtp_header_extension_size_unittest.cc",
+      "source/rtp_packet_history_unittest.cc",
+      "source/rtp_packet_unittest.cc",
+      "source/rtp_packetizer_av1_unittest.cc",
+      "source/rtp_rtcp_impl2_unittest.cc",
+      "source/rtp_rtcp_impl_unittest.cc",
+      "source/rtp_sender_audio_unittest.cc",
+      "source/rtp_sender_egress_unittest.cc",
+      "source/rtp_sender_unittest.cc",
+      "source/rtp_sender_video_unittest.cc",
+      "source/rtp_sequence_number_map_unittest.cc",
+      "source/rtp_util_unittest.cc",
+      "source/rtp_video_layers_allocation_extension_unittest.cc",
+      "source/source_tracker_unittest.cc",
+      "source/time_util_unittest.cc",
+      "source/ulpfec_generator_unittest.cc",
+      "source/ulpfec_header_reader_writer_unittest.cc",
+      "source/ulpfec_receiver_unittest.cc",
+      "source/video_rtp_depacketizer_av1_unittest.cc",
+      "source/video_rtp_depacketizer_generic_unittest.cc",
+      "source/video_rtp_depacketizer_h264_unittest.cc",
+      "source/video_rtp_depacketizer_raw_unittest.cc",
+      "source/video_rtp_depacketizer_vp8_unittest.cc",
+      "source/video_rtp_depacketizer_vp9_unittest.cc",
+    ]
+    deps = [
+      ":fec_test_helper",
+      ":frame_transformer_factory_unittest",
+      ":leb128",
+      ":mock_rtp_rtcp",
+      ":rtcp_transceiver",
+      ":rtp_packetizer_av1_test_helper",
+      ":rtp_rtcp",
+      ":rtp_rtcp_format",
+      ":rtp_rtcp_legacy",
+      ":rtp_video_header_unittest",
+      "../../api:array_view",
+      "../../api:create_time_controller",
+      "../../api:field_trials_registry",
+      "../../api:frame_transformer_factory",
+      "../../api:libjingle_peerconnection_api",
+      "../../api:mock_frame_encryptor",
+      "../../api:rtp_headers",
+      "../../api:rtp_packet_info",
+      "../../api:rtp_parameters",
+      "../../api:scoped_refptr",
+      "../../api:time_controller",
+      "../../api:transport_api",
+      "../../api/rtc_event_log",
+      "../../api/task_queue",
+      "../../api/transport:field_trial_based_config",
+      "../../api/transport/rtp:dependency_descriptor",
+      "../../api/units:data_rate",
+      "../../api/units:data_size",
+      "../../api/units:time_delta",
+      "../../api/units:timestamp",
+      "../../api/video:encoded_image",
+      "../../api/video:video_bitrate_allocation",
+      "../../api/video:video_bitrate_allocator",
+      "../../api/video:video_codec_constants",
+      "../../api/video:video_frame",
+      "../../api/video:video_layers_allocation",
+      "../../api/video:video_rtp_headers",
+      "../../api/video_codecs:video_codecs_api",
+      "../../call:rtp_receiver",
+      "../../common_video",
+      "../../common_video/generic_frame_descriptor",
+      "../../common_video/test:utilities",
+      "../../logging:mocks",
+      "../../rtc_base:bit_buffer",
+      "../../rtc_base:buffer",
+      "../../rtc_base:checks",
+      "../../rtc_base:copy_on_write_buffer",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:random",
+      "../../rtc_base:rate_limiter",
+      "../../rtc_base:rtc_base_tests_utils",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:rtc_numerics",
+      "../../rtc_base:stringutils",
+      "../../rtc_base:task_queue_for_test",
+      "../../rtc_base:threading",
+      "../../rtc_base:timeutils",
+      "../../system_wrappers",
+      "../../test:explicit_key_value_config",
+      "../../test:field_trial",
+      "../../test:mock_frame_transformer",
+      "../../test:mock_transport",
+      "../../test:rtp_test_utils",
+      "../../test:run_loop",
+      "../../test:scoped_key_value_config",
+      "../../test:test_support",
+      "../../test/time_controller:time_controller",
+      "../video_coding:codec_globals_headers",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/algorithm:container",
+      "//third_party/abseil-cpp/absl/base:core_headers",
+      "//third_party/abseil-cpp/absl/memory",
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+}
+
+rtc_source_set("frame_transformer_factory_unittest") {
+  testonly = true
+  sources = [ "source/frame_transformer_factory_unittest.cc" ]
+  deps = [
+    "../../api:frame_transformer_factory",
+    "../../api:transport_api",
+    "../../call:video_stream_api",
+    "../../modules/rtp_rtcp",
+    "../../rtc_base:rtc_event",
+    "../../test:mock_frame_transformer",
+    "../../test:test_support",
+    "../../video",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+}
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/DEPS b/third_party/libwebrtc/modules/rtp_rtcp/DEPS
new file mode 100644
index 0000000000..3eec1ca90d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/DEPS
@@ -0,0 +1,8 @@
+include_rules = [
+  "+call",
+  "+common_video",
+  "+logging/rtc_event_log",
+  "+system_wrappers",
+  # Avoid directly using field_trial. Instead use FieldTrialsView.
+  "-system_wrappers/include/field_trial.h",
+]
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/OWNERS b/third_party/libwebrtc/modules/rtp_rtcp/OWNERS
new file mode 100644
index 0000000000..47d12c401f
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/OWNERS
@@ -0,0 +1,6 @@
+stefan@webrtc.org
+henrik.lundin@webrtc.org
+mflodman@webrtc.org
+asapersson@webrtc.org
+danilchap@webrtc.org
+sprang@webrtc.org
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/flexfec_receiver.h b/third_party/libwebrtc/modules/rtp_rtcp/include/flexfec_receiver.h
new file mode 100644
index 0000000000..29d9e72786
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/flexfec_receiver.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_FLEXFEC_RECEIVER_H_
+#define MODULES_RTP_RTCP_INCLUDE_FLEXFEC_RECEIVER_H_
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/rtp_rtcp/include/recovered_packet_receiver.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/ulpfec_receiver.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class Clock;
+
+class FlexfecReceiver {
+ public:
+  /* Mozilla: Avoid this since it could use GetRealTimeClock().
+  FlexfecReceiver(uint32_t ssrc,
+                  uint32_t protected_media_ssrc,
+                  RecoveredPacketReceiver* recovered_packet_receiver);
+   */
+  FlexfecReceiver(Clock* clock,
+                  uint32_t ssrc,
+                  uint32_t protected_media_ssrc,
+                  RecoveredPacketReceiver* recovered_packet_receiver);
+  ~FlexfecReceiver();
+
+  // Inserts a received packet (can be either media or FlexFEC) into the
+  // internal buffer, and sends the received packets to the erasure code.
+  // All newly recovered packets are sent back through the callback.
+  void OnRtpPacket(const RtpPacketReceived& packet);
+
+  // Returns a counter describing the added and recovered packets.
+  FecPacketCounter GetPacketCounter() const;
+
+  // Protected to aid testing.
+ protected:
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> AddReceivedPacket(
+      const RtpPacketReceived& packet);
+  void ProcessReceivedPacket(
+      const ForwardErrorCorrection::ReceivedPacket& received_packet);
+
+ private:
+  // Config.
+  const uint32_t ssrc_;
+  const uint32_t protected_media_ssrc_;
+
+  // Erasure code interfacing and callback.
+  std::unique_ptr<ForwardErrorCorrection> erasure_code_
+      RTC_GUARDED_BY(sequence_checker_);
+  ForwardErrorCorrection::RecoveredPacketList recovered_packets_
+      RTC_GUARDED_BY(sequence_checker_);
+  RecoveredPacketReceiver* const recovered_packet_receiver_;
+
+  // Logging and stats.
+  Clock* const clock_;
+  int64_t last_recovered_packet_ms_ RTC_GUARDED_BY(sequence_checker_);
+  FecPacketCounter packet_counter_ RTC_GUARDED_BY(sequence_checker_);
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_FLEXFEC_RECEIVER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/flexfec_sender.h b/third_party/libwebrtc/modules/rtp_rtcp/include/flexfec_sender.h
new file mode 100644
index 0000000000..f0acfe6c3d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/flexfec_sender.h
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_FLEXFEC_SENDER_H_
+#define MODULES_RTP_RTCP_INCLUDE_FLEXFEC_SENDER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/rtp_parameters.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
+#include "modules/rtp_rtcp/source/ulpfec_generator.h"
+#include "modules/rtp_rtcp/source/video_fec_generator.h"
+#include "rtc_base/random.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class Clock;
+class RtpPacketToSend;
+
+// Note that this class is not thread safe, and thus requires external
+// synchronization. Currently, this is done using the lock in PayloadRouter.
+
+class FlexfecSender : public VideoFecGenerator {
+ public:
+  FlexfecSender(int payload_type,
+                uint32_t ssrc,
+                uint32_t protected_media_ssrc,
+                absl::string_view mid,
+                const std::vector<RtpExtension>& rtp_header_extensions,
+                rtc::ArrayView<const RtpExtensionSize> extension_sizes,
+                const RtpState* rtp_state,
+                Clock* clock);
+  ~FlexfecSender();
+
+  FecType GetFecType() const override {
+    return VideoFecGenerator::FecType::kFlexFec;
+  }
+  absl::optional<uint32_t> FecSsrc() override { return ssrc_; }
+
+  // Sets the FEC rate, max frames sent before FEC packets are sent,
+  // and what type of generator matrices are used.
+  void SetProtectionParameters(const FecProtectionParams& delta_params,
+                               const FecProtectionParams& key_params) override;
+
+  // Adds a media packet to the internal buffer. When enough media packets
+  // have been added, the FEC packets are generated and stored internally.
+  // These FEC packets are then obtained by calling GetFecPackets().
+  void AddPacketAndGenerateFec(const RtpPacketToSend& packet) override;
+
+  // Returns generated FlexFEC packets.
+  std::vector<std::unique_ptr<RtpPacketToSend>> GetFecPackets() override;
+
+  // Returns the overhead, per packet, for FlexFEC.
+  size_t MaxPacketOverhead() const override;
+
+  DataRate CurrentFecRate() const override;
+
+  // Only called on the VideoSendStream queue, after operation has shut down.
+  absl::optional<RtpState> GetRtpState() override;
+
+ private:
+  // Utility.
+  Clock* const clock_;
+  Random random_;
+  int64_t last_generated_packet_ms_;
+
+  // Config.
+  const int payload_type_;
+  const uint32_t timestamp_offset_;
+  const uint32_t ssrc_;
+  const uint32_t protected_media_ssrc_;
+  // MID value to send in the MID header extension.
+  const std::string mid_;
+  // Sequence number of next packet to generate.
+  uint16_t seq_num_;
+
+  // Implementation.
+  UlpfecGenerator ulpfec_generator_;
+  const RtpHeaderExtensionMap rtp_header_extension_map_;
+  const size_t header_extensions_size_;
+
+  mutable Mutex mutex_;
+  RateStatistics fec_bitrate_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_FLEXFEC_SENDER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/receive_statistics.h b/third_party/libwebrtc/modules/rtp_rtcp/include/receive_statistics.h
new file mode 100644
index 0000000000..827fd3a7a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/receive_statistics.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+#define MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+namespace webrtc {
+
+class Clock;
+
+class ReceiveStatisticsProvider {
+ public:
+  virtual ~ReceiveStatisticsProvider() = default;
+  // Collects receive statistic in a form of rtcp report blocks.
+  // Returns at most `max_blocks` report blocks.
+  virtual std::vector<rtcp::ReportBlock> RtcpReportBlocks(
+      size_t max_blocks) = 0;
+};
+
+class StreamStatistician {
+ public:
+  virtual ~StreamStatistician();
+
+  virtual RtpReceiveStats GetStats() const = 0;
+
+  // Returns average over the stream life time.
+  virtual absl::optional<int> GetFractionLostInPercent() const = 0;
+
+  // TODO(bugs.webrtc.org/10679): Delete, migrate users to the above GetStats
+  // method (and extend RtpReceiveStats if needed).
+  // Gets receive stream data counters.
+  virtual StreamDataCounters GetReceiveStreamDataCounters() const = 0;
+
+  virtual uint32_t BitrateReceived() const = 0;
+};
+
+class ReceiveStatistics : public ReceiveStatisticsProvider,
+                          public RtpPacketSinkInterface {
+ public:
+  ~ReceiveStatistics() override = default;
+
+  // Returns a thread-safe instance of ReceiveStatistics.
+  // https://chromium.googlesource.com/chromium/src/+/lkgr/docs/threading_and_tasks.md#threading-lexicon
+  static std::unique_ptr<ReceiveStatistics> Create(Clock* clock);
+  // Returns a thread-compatible instance of ReceiveStatistics.
+  static std::unique_ptr<ReceiveStatistics> CreateThreadCompatible(
+      Clock* clock);
+
+  // Returns a pointer to the statistician of an ssrc.
+  virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
+
+  // TODO(bugs.webrtc.org/10669): Deprecated, delete as soon as downstream
+  // projects are updated. This method sets the max reordering threshold of all
+  // current and future streams.
+  virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
+
+  // Sets the max reordering threshold in number of packets.
+  virtual void SetMaxReorderingThreshold(uint32_t ssrc,
+                                         int max_reordering_threshold) = 0;
+  // Detect retransmissions, enabling updates of the retransmitted counters. The
+  // default is false.
+  virtual void EnableRetransmitDetection(uint32_t ssrc, bool enable) = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/recovered_packet_receiver.h b/third_party/libwebrtc/modules/rtp_rtcp/include/recovered_packet_receiver.h
new file mode 100644
index 0000000000..4e92c486e2
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/recovered_packet_receiver.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_INCLUDE_RECOVERED_PACKET_RECEIVER_H_
+#define MODULES_RTP_RTCP_INCLUDE_RECOVERED_PACKET_RECEIVER_H_
+
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Callback interface for packets recovered by FlexFEC or ULPFEC. In
+// the FlexFEC case, the implementation should be able to demultiplex
+// the recovered RTP packets based on SSRC.
+class RecoveredPacketReceiver {
+ public:
+  virtual void OnRecoveredPacket(const RtpPacketReceived& packet) = 0;
+
+ protected:
+  virtual ~RecoveredPacketReceiver() = default;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RECOVERED_PACKET_RECEIVER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/third_party/libwebrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
new file mode 100644
index 0000000000..01d0c85f94
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#define MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/numerics/moving_percentile_filter.h"
+#include "system_wrappers/include/rtp_to_ntp_estimator.h"
+
+namespace webrtc {
+
+class Clock;
+
+// RemoteNtpTimeEstimator can be used to estimate a given RTP timestamp's NTP
+// time in local timebase.
+// Note that it needs to be trained with at least 2 RTCP SR (by calling
+// `UpdateRtcpTimestamp`) before it can be used.
+class RemoteNtpTimeEstimator {
+ public:
+  explicit RemoteNtpTimeEstimator(Clock* clock);
+  RemoteNtpTimeEstimator(const RemoteNtpTimeEstimator&) = delete;
+  RemoteNtpTimeEstimator& operator=(const RemoteNtpTimeEstimator&) = delete;
+  ~RemoteNtpTimeEstimator() = default;
+
+  // Updates the estimator with round trip time `rtt` and
+  // new NTP time <-> RTP timestamp mapping from an RTCP sender report.
+  bool UpdateRtcpTimestamp(TimeDelta rtt,
+                           NtpTime sender_send_time,
+                           uint32_t rtp_timestamp);
+
+  // Estimates the NTP timestamp in local timebase from `rtp_timestamp`.
+  // Returns the NTP timestamp in ms when success. -1 if failed.
+  int64_t Estimate(uint32_t rtp_timestamp) {
+    NtpTime ntp_time = EstimateNtp(rtp_timestamp);
+    if (!ntp_time.Valid()) {
+      return -1;
+    }
+    return ntp_time.ToMs();
+  }
+
+  // Estimates the NTP timestamp in local timebase from `rtp_timestamp`.
+  // Returns invalid NtpTime (i.e. NtpTime(0)) on failure.
+  NtpTime EstimateNtp(uint32_t rtp_timestamp);
+
+  // Estimates the offset between the remote clock and the
+  // local one. This is equal to local NTP clock - remote NTP clock.
+  // The offset is returned in ntp time resolution, i.e. 1/2^32 sec ~= 0.2 ns.
+  // Returns nullopt on failure.
+  absl::optional<int64_t> EstimateRemoteToLocalClockOffset();
+
+ private:
+  Clock* clock_;
+  // Offset is measured with the same precision as NtpTime: in 1/2^32 seconds ~=
+  // 0.2 ns.
+  MovingMedianFilter<int64_t> ntp_clocks_offset_estimator_;
+  RtpToNtpEstimator rtp_to_ntp_;
+  Timestamp last_timing_log_ = Timestamp::MinusInfinity();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.cc b/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.cc
new file mode 100644
index 0000000000..ec4d9d82e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/report_block_data.h"
+
+namespace webrtc {
+
+ReportBlockData::ReportBlockData()
+    : report_block_(),
+      report_block_timestamp_utc_us_(0),
+      last_rtt_ms_(0),
+      min_rtt_ms_(0),
+      max_rtt_ms_(0),
+      sum_rtt_ms_(0),
+      num_rtts_(0) {}
+
+double ReportBlockData::AvgRttMs() const {
+  return num_rtts_ ? static_cast<double>(sum_rtt_ms_) / num_rtts_ : 0.0;
+}
+
+void ReportBlockData::SetReportBlock(RTCPReportBlock report_block,
+                                     int64_t report_block_timestamp_utc_us) {
+  report_block_ = report_block;
+  report_block_timestamp_utc_us_ = report_block_timestamp_utc_us;
+}
+
+void ReportBlockData::AddRoundTripTimeSample(int64_t rtt_ms) {
+  if (rtt_ms > max_rtt_ms_)
+    max_rtt_ms_ = rtt_ms;
+  if (num_rtts_ == 0 || rtt_ms < min_rtt_ms_)
+    min_rtt_ms_ = rtt_ms;
+  last_rtt_ms_ = rtt_ms;
+  sum_rtt_ms_ += rtt_ms;
+  ++num_rtts_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.h b/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.h
new file mode 100644
index 0000000000..2c4533ada8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_
+#define MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class ReportBlockData {
+ public:
+  ReportBlockData();
+
+  const RTCPReportBlock& report_block() const { return report_block_; }
+  int64_t report_block_timestamp_utc_us() const {
+    return report_block_timestamp_utc_us_;
+  }
+  int64_t last_rtt_ms() const { return last_rtt_ms_; }
+  int64_t min_rtt_ms() const { return min_rtt_ms_; }
+  int64_t max_rtt_ms() const { return max_rtt_ms_; }
+  int64_t sum_rtt_ms() const { return sum_rtt_ms_; }
+  size_t num_rtts() const { return num_rtts_; }
+  bool has_rtt() const { return num_rtts_ != 0; }
+
+  double AvgRttMs() const;
+
+  void SetReportBlock(RTCPReportBlock report_block,
+                      int64_t report_block_timestamp_utc_us);
+  void AddRoundTripTimeSample(int64_t rtt_ms);
+
+ private:
+  RTCPReportBlock report_block_;
+  int64_t report_block_timestamp_utc_us_;
+
+  int64_t last_rtt_ms_;
+  int64_t min_rtt_ms_;
+  int64_t max_rtt_ms_;
+  int64_t sum_rtt_ms_;
+  size_t num_rtts_;
+};
+
+class ReportBlockDataObserver {
+ public:
+  virtual ~ReportBlockDataObserver() = default;
+
+  virtual void OnReportBlockDataUpdated(ReportBlockData report_block_data) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/rtcp_statistics.h b/third_party/libwebrtc/modules/rtp_rtcp/include/rtcp_statistics.h
new file mode 100644
index 0000000000..6d6246d8a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/rtcp_statistics.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
+
+#include <stdint.h>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+// Statistics for RTCP packet types.
+struct RtcpPacketTypeCounter {
+  RtcpPacketTypeCounter()
+      : nack_packets(0),
+        fir_packets(0),
+        pli_packets(0),
+        nack_requests(0),
+        unique_nack_requests(0) {}
+
+  void Add(const RtcpPacketTypeCounter& other) {
+    nack_packets += other.nack_packets;
+    fir_packets += other.fir_packets;
+    pli_packets += other.pli_packets;
+    nack_requests += other.nack_requests;
+    unique_nack_requests += other.unique_nack_requests;
+  }
+
+  void Subtract(const RtcpPacketTypeCounter& other) {
+    nack_packets -= other.nack_packets;
+    fir_packets -= other.fir_packets;
+    pli_packets -= other.pli_packets;
+    nack_requests -= other.nack_requests;
+    unique_nack_requests -= other.unique_nack_requests;
+  }
+
+  int UniqueNackRequestsInPercent() const {
+    if (nack_requests == 0) {
+      return 0;
+    }
+    return static_cast<int>((unique_nack_requests * 100.0f / nack_requests) +
+                            0.5f);
+  }
+
+  uint32_t nack_packets;          // Number of RTCP NACK packets.
+  uint32_t fir_packets;           // Number of RTCP FIR packets.
+  uint32_t pli_packets;           // Number of RTCP PLI packets.
+  uint32_t nack_requests;         // Number of NACKed RTP packets.
+  uint32_t unique_nack_requests;  // Number of unique NACKed RTP packets.
+};
+
+class RtcpPacketTypeCounterObserver {
+ public:
+  virtual ~RtcpPacketTypeCounterObserver() {}
+  virtual void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) = 0;
+};
+
+// Invoked for each cname passed in RTCP SDES blocks.
+class RtcpCnameCallback {
+ public:
+  virtual ~RtcpCnameCallback() = default;
+
+  virtual void OnCname(uint32_t ssrc, absl::string_view cname) = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_cvo.h b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_cvo.h
new file mode 100644
index 0000000000..497946d6a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_cvo.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+
+#include "api/video/video_rotation.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Please refer to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/
+// 12.07.00_60/ts_126114v120700p.pdf Section 7.4.5. The rotation of a frame is
+// the clockwise angle the frames must be rotated in order to display the frames
+// correctly if the display is rotated in its natural orientation.
+inline uint8_t ConvertVideoRotationToCVOByte(VideoRotation rotation) {
+  switch (rotation) {
+    case kVideoRotation_0:
+      return 0;
+    case kVideoRotation_90:
+      return 1;
+    case kVideoRotation_180:
+      return 2;
+    case kVideoRotation_270:
+      return 3;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return 0;
+}
+
+inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t cvo_byte) {
+  // CVO byte: |0 0 0 0 C F R R|.
+  const uint8_t rotation_bits = cvo_byte & 0x3;
+  switch (rotation_bits) {
+    case 0:
+      return kVideoRotation_0;
+    case 1:
+      return kVideoRotation_90;
+    case 2:
+      return kVideoRotation_180;
+    case 3:
+      return kVideoRotation_270;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return kVideoRotation_0;
+  }
+}
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
new file mode 100644
index 0000000000..ff1ea61f52
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_EXTENSION_MAP_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_EXTENSION_MAP_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/rtp_parameters.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class RtpHeaderExtensionMap {
+ public:
+  static constexpr RTPExtensionType kInvalidType = kRtpExtensionNone;
+  static constexpr int kInvalidId = 0;
+
+  RtpHeaderExtensionMap();
+  explicit RtpHeaderExtensionMap(bool extmap_allow_mixed);
+  explicit RtpHeaderExtensionMap(rtc::ArrayView<const RtpExtension> extensions);
+
+  void Reset(rtc::ArrayView<const RtpExtension> extensions);
+
+  template <typename Extension>
+  bool Register(int id) {
+    return Register(id, Extension::kId, Extension::Uri());
+  }
+  bool RegisterByType(int id, RTPExtensionType type);
+  bool RegisterByUri(int id, absl::string_view uri);
+
+  bool IsRegistered(RTPExtensionType type) const {
+    return GetId(type) != kInvalidId;
+  }
+  // Return kInvalidType if not found.
+  RTPExtensionType GetType(int id) const;
+  // Return kInvalidId if not found.
+  uint8_t GetId(RTPExtensionType type) const {
+    RTC_DCHECK_GT(type, kRtpExtensionNone);
+    RTC_DCHECK_LT(type, kRtpExtensionNumberOfExtensions);
+    return ids_[type];
+  }
+
+  void Deregister(absl::string_view uri);
+
+  // Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
+  // Set to true if it's allowed to mix one- and two-byte RTP header extensions
+  // in the same stream.
+  bool ExtmapAllowMixed() const { return extmap_allow_mixed_; }
+  void SetExtmapAllowMixed(bool extmap_allow_mixed) {
+    extmap_allow_mixed_ = extmap_allow_mixed;
+  }
+
+ private:
+  bool Register(int id, RTPExtensionType type, absl::string_view uri);
+
+  uint8_t ids_[kRtpExtensionNumberOfExtensions];
+  bool extmap_allow_mixed_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_EXTENSION_MAP_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_packet_sender.h b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_packet_sender.h
new file mode 100644
index 0000000000..ebc65298a5
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_packet_sender.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
+
+#include <memory>
+#include <vector>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+class RtpPacketSender {
+ public:
+  virtual ~RtpPacketSender() = default;
+
+  // Insert a set of packets into queue, for eventual transmission. Based on the
+  // type of packets, they will be prioritized and scheduled relative to other
+  // packets and the current target send rate.
+  virtual void EnqueuePackets(
+      std::vector<std::unique_ptr<RtpPacketToSend>> packets) = 0;
+
+  // Clear any pending packets with the given SSRC from the queue.
+  // TODO(crbug.com/1395081): Make pure virtual when downstream code has been
+  // updated.
+  virtual void RemovePacketsForSsrc(uint32_t ssrc) {}
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_PACKET_SENDER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp.h
new file mode 100644
index 0000000000..e56d5ef637
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+
+#include <memory>
+
+#include "absl/base/attributes.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+
+namespace webrtc {
+
+class ABSL_DEPRECATED("") RtpRtcp : public RtpRtcpInterface {
+ public:
+  // Instantiates a deprecated version of the RtpRtcp module.
+  static std::unique_ptr<RtpRtcp> ABSL_DEPRECATED("")
+      Create(const Configuration& configuration) {
+    return DEPRECATED_Create(configuration);
+  }
+
+  static std::unique_ptr<RtpRtcp> DEPRECATED_Create(
+      const Configuration& configuration);
+
+  // Process any pending tasks such as timeouts.
+  virtual void Process() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
new file mode 100644
index 0000000000..e4aec93696
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+#include <string.h>
+
+#include <type_traits>
+
+#include "absl/algorithm/container.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+namespace {
+constexpr size_t kMidRsidMaxSize = 16;
+
+// Check if passed character is a "token-char" from RFC 4566.
+// https://datatracker.ietf.org/doc/html/rfc4566#section-9
+//    token-char =          %x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
+//                         / %x41-5A / %x5E-7E
+bool IsTokenChar(char ch) {
+  return ch == 0x21 || (ch >= 0x23 && ch <= 0x27) || ch == 0x2a || ch == 0x2b ||
+         ch == 0x2d || ch == 0x2e || (ch >= 0x30 && ch <= 0x39) ||
+         (ch >= 0x41 && ch <= 0x5a) || (ch >= 0x5e && ch <= 0x7e);
+}
+}  // namespace
+
+bool IsLegalMidName(absl::string_view name) {
+  return (name.size() <= kMidRsidMaxSize && !name.empty() &&
+          absl::c_all_of(name, IsTokenChar));
+}
+
+bool IsLegalRsidName(absl::string_view name) {
+  return (name.size() <= kMidRsidMaxSize && !name.empty() &&
+          absl::c_all_of(name, isalnum));
+}
+
+StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {}
+
+RtpPacketCounter::RtpPacketCounter(const RtpPacket& packet)
+    : header_bytes(packet.headers_size()),
+      payload_bytes(packet.payload_size()),
+      padding_bytes(packet.padding_size()),
+      packets(1) {}
+
+RtpPacketCounter::RtpPacketCounter(const RtpPacketToSend& packet_to_send)
+    : RtpPacketCounter(static_cast<const RtpPacket&>(packet_to_send)) {
+  total_packet_delay =
+      packet_to_send.time_in_send_queue().value_or(TimeDelta::Zero());
+}
+
+void RtpPacketCounter::AddPacket(const RtpPacket& packet) {
+  ++packets;
+  header_bytes += packet.headers_size();
+  padding_bytes += packet.padding_size();
+  payload_bytes += packet.payload_size();
+}
+
+void RtpPacketCounter::AddPacket(const RtpPacketToSend& packet_to_send) {
+  AddPacket(static_cast<const RtpPacket&>(packet_to_send));
+  total_packet_delay +=
+      packet_to_send.time_in_send_queue().value_or(TimeDelta::Zero());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
new file mode 100644
index 0000000000..882f861d0b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -0,0 +1,495 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+
+#include <stddef.h>
+
+#include <list>
+#include <memory>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "absl/types/variant.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/rtp_headers.h"
+#include "api/transport/network_types.h"
+#include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
+#include "system_wrappers/include/clock.h"
+
+#define RTCP_CNAME_SIZE 256  // RFC 3550 page 44, including null termination
+#define IP_PACKET_SIZE 1500  // we assume ethernet
+
+namespace webrtc {
+class RtpPacket;
+class RtpPacketToSend;
+namespace rtcp {
+class TransportFeedback;
+}
+
+const int kVideoPayloadTypeFrequency = 90000;
+
+// TODO(bugs.webrtc.org/6458): Remove this when all the depending projects are
+// updated to correctly set rtp rate for RtcpSender.
+const int kBogusRtpRateForAudioRtcp = 8000;
+
+// Minimum RTP header size in bytes.
+const uint8_t kRtpHeaderSize = 12;
+
+bool IsLegalMidName(absl::string_view name);
+bool IsLegalRsidName(absl::string_view name);
+
+// This enum must not have any gaps, i.e., all integers between
+// kRtpExtensionNone and kRtpExtensionNumberOfExtensions must be valid enum
+// entries.
+enum RTPExtensionType : int {
+  kRtpExtensionNone,
+  kRtpExtensionTransmissionTimeOffset,
+  kRtpExtensionAudioLevel,
+  kRtpExtensionCsrcAudioLevel,
+  kRtpExtensionInbandComfortNoise,
+  kRtpExtensionAbsoluteSendTime,
+  kRtpExtensionAbsoluteCaptureTime,
+  kRtpExtensionVideoRotation,
+  kRtpExtensionTransportSequenceNumber,
+  kRtpExtensionTransportSequenceNumber02,
+  kRtpExtensionPlayoutDelay,
+  kRtpExtensionVideoContentType,
+  kRtpExtensionVideoLayersAllocation,
+  kRtpExtensionVideoTiming,
+  kRtpExtensionRtpStreamId,
+  kRtpExtensionRepairedRtpStreamId,
+  kRtpExtensionMid,
+  kRtpExtensionGenericFrameDescriptor,
+  kRtpExtensionGenericFrameDescriptor00 [[deprecated]] =
+      kRtpExtensionGenericFrameDescriptor,
+  kRtpExtensionDependencyDescriptor,
+  kRtpExtensionGenericFrameDescriptor02 [[deprecated]] =
+      kRtpExtensionDependencyDescriptor,
+  kRtpExtensionColorSpace,
+  kRtpExtensionVideoFrameTrackingId,
+  kRtpExtensionNumberOfExtensions  // Must be the last entity in the enum.
+};
+
+enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };
+
+// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
+enum RTCPPacketType : uint32_t {
+  kRtcpReport = 0x0001,
+  kRtcpSr = 0x0002,
+  kRtcpRr = 0x0004,
+  kRtcpSdes = 0x0008,
+  kRtcpBye = 0x0010,
+  kRtcpPli = 0x0020,
+  kRtcpNack = 0x0040,
+  kRtcpFir = 0x0080,
+  kRtcpTmmbr = 0x0100,
+  kRtcpTmmbn = 0x0200,
+  kRtcpSrReq = 0x0400,
+  kRtcpLossNotification = 0x2000,
+  kRtcpRemb = 0x10000,
+  kRtcpTransmissionTimeOffset = 0x20000,
+  kRtcpXrReceiverReferenceTime = 0x40000,
+  kRtcpXrDlrrReportBlock = 0x80000,
+  kRtcpTransportFeedback = 0x100000,
+  kRtcpXrTargetBitrate = 0x200000
+};
+
+enum class KeyFrameReqMethod : uint8_t {
+  kNone,     // Don't request keyframes.
+  kPliRtcp,  // Request keyframes through Picture Loss Indication.
+  kFirRtcp   // Request keyframes through Full Intra-frame Request.
+};
+
+enum RtxMode {
+  kRtxOff = 0x0,
+  kRtxRetransmitted = 0x1,     // Only send retransmissions over RTX.
+  kRtxRedundantPayloads = 0x2  // Preventively send redundant payloads
+                               // instead of padding.
+};
+
+const size_t kRtxHeaderSize = 2;
+
+struct RTCPReportBlock {
+  RTCPReportBlock()
+      : sender_ssrc(0),
+        source_ssrc(0),
+        fraction_lost(0),
+        packets_lost(0),
+        extended_highest_sequence_number(0),
+        jitter(0),
+        last_sender_report_timestamp(0),
+        delay_since_last_sender_report(0) {}
+
+  RTCPReportBlock(uint32_t sender_ssrc,
+                  uint32_t source_ssrc,
+                  uint8_t fraction_lost,
+                  int32_t packets_lost,
+                  uint32_t extended_highest_sequence_number,
+                  uint32_t jitter,
+                  uint32_t last_sender_report_timestamp,
+                  uint32_t delay_since_last_sender_report)
+      : sender_ssrc(sender_ssrc),
+        source_ssrc(source_ssrc),
+        fraction_lost(fraction_lost),
+        packets_lost(packets_lost),
+        extended_highest_sequence_number(extended_highest_sequence_number),
+        jitter(jitter),
+        last_sender_report_timestamp(last_sender_report_timestamp),
+        delay_since_last_sender_report(delay_since_last_sender_report) {}
+
+  // Fields as described by RFC 3550 6.4.2.
+  uint32_t sender_ssrc;  // SSRC of sender of this report.
+  uint32_t source_ssrc;  // SSRC of the RTP packet sender.
+  uint8_t fraction_lost;
+  int32_t packets_lost;  // 24 bits valid.
+  uint32_t extended_highest_sequence_number;
+  uint32_t jitter;
+  uint32_t last_sender_report_timestamp;
+  uint32_t delay_since_last_sender_report;
+};
+
+typedef std::list<RTCPReportBlock> ReportBlockList;
+
+struct RtpState {
+  RtpState()
+      : sequence_number(0),
+        start_timestamp(0),
+        timestamp(0),
+        capture_time_ms(-1),
+        last_timestamp_time_ms(-1),
+        ssrc_has_acked(false) {}
+  uint16_t sequence_number;
+  uint32_t start_timestamp;
+  uint32_t timestamp;
+  int64_t capture_time_ms;
+  int64_t last_timestamp_time_ms;
+  bool ssrc_has_acked;
+};
+
+class RtcpIntraFrameObserver {
+ public:
+  virtual ~RtcpIntraFrameObserver() {}
+
+  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
+};
+
+// Observer for incoming LossNotification RTCP messages.
+// See the documentation of LossNotification for details.
+class RtcpLossNotificationObserver {
+ public:
+  virtual ~RtcpLossNotificationObserver() = default;
+
+  virtual void OnReceivedLossNotification(uint32_t ssrc,
+                                          uint16_t seq_num_of_last_decodable,
+                                          uint16_t seq_num_of_last_received,
+                                          bool decodability_flag) = 0;
+};
+
+class RtcpBandwidthObserver {
+ public:
+  // REMB or TMMBR
+  virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
+
+  virtual void OnReceivedRtcpReceiverReport(
+      const ReportBlockList& report_blocks,
+      int64_t rtt,
+      int64_t now_ms) = 0;
+
+  virtual ~RtcpBandwidthObserver() {}
+};
+
+class RtcpEventObserver {
+ public:
+  virtual void OnRtcpBye() = 0;
+  virtual void OnRtcpTimeout() = 0;
+
+  virtual ~RtcpEventObserver() {}
+};
+
+// NOTE! `kNumMediaTypes` must be kept in sync with RtpPacketMediaType!
+static constexpr size_t kNumMediaTypes = 5;
+enum class RtpPacketMediaType : size_t {
+  kAudio,                         // Audio media packets.
+  kVideo,                         // Video media packets.
+  kRetransmission,                // Retransmisions, sent as response to NACK.
+  kForwardErrorCorrection,        // FEC packets.
+  kPadding = kNumMediaTypes - 1,  // RTX or plain padding sent to maintain BWE.
+  // Again, don't forget to udate `kNumMediaTypes` if you add another value!
+};
+
+struct RtpPacketSendInfo {
+  uint16_t transport_sequence_number = 0;
+  absl::optional<uint32_t> media_ssrc;
+  uint16_t rtp_sequence_number = 0;  // Only valid if `media_ssrc` is set.
+  uint32_t rtp_timestamp = 0;
+  size_t length = 0;
+  absl::optional<RtpPacketMediaType> packet_type;
+  PacedPacketInfo pacing_info;
+};
+
+class NetworkStateEstimateObserver {
+ public:
+  virtual void OnRemoteNetworkEstimate(NetworkStateEstimate estimate) = 0;
+  virtual ~NetworkStateEstimateObserver() = default;
+};
+
+class TransportFeedbackObserver {
+ public:
+  TransportFeedbackObserver() {}
+  virtual ~TransportFeedbackObserver() {}
+
+  virtual void OnAddPacket(const RtpPacketSendInfo& packet_info) = 0;
+  virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
+};
+
+// Interface for PacketRouter to send rtcp feedback on behalf of
+// congestion controller.
+// TODO(bugs.webrtc.org/8239): Remove and use RtcpTransceiver directly
+// when RtcpTransceiver always present in rtp transport.
+class RtcpFeedbackSenderInterface {
+ public:
+  virtual ~RtcpFeedbackSenderInterface() = default;
+  virtual void SendCombinedRtcpPacket(
+      std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) = 0;
+  virtual void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) = 0;
+  virtual void UnsetRemb() = 0;
+};
+
+class StreamFeedbackObserver {
+ public:
+  struct StreamPacketInfo {
+    bool received;
+
+    // `rtp_sequence_number` and `is_retransmission` are only valid if `ssrc`
+    // is populated.
+    absl::optional<uint32_t> ssrc;
+    uint16_t rtp_sequence_number;
+    bool is_retransmission;
+  };
+  virtual ~StreamFeedbackObserver() = default;
+
+  virtual void OnPacketFeedbackVector(
+      std::vector<StreamPacketInfo> packet_feedback_vector) = 0;
+};
+
+class StreamFeedbackProvider {
+ public:
+  virtual void RegisterStreamFeedbackObserver(
+      std::vector<uint32_t> ssrcs,
+      StreamFeedbackObserver* observer) = 0;
+  virtual void DeRegisterStreamFeedbackObserver(
+      StreamFeedbackObserver* observer) = 0;
+  virtual ~StreamFeedbackProvider() = default;
+};
+
+class RtcpRttStats {
+ public:
+  virtual void OnRttUpdate(int64_t rtt) = 0;
+
+  virtual int64_t LastProcessedRtt() const = 0;
+
+  virtual ~RtcpRttStats() {}
+};
+
+struct RtpPacketCounter {
+  RtpPacketCounter()
+      : header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}
+
+  explicit RtpPacketCounter(const RtpPacket& packet);
+  explicit RtpPacketCounter(const RtpPacketToSend& packet_to_send);
+
+  void Add(const RtpPacketCounter& other) {
+    header_bytes += other.header_bytes;
+    payload_bytes += other.payload_bytes;
+    padding_bytes += other.padding_bytes;
+    packets += other.packets;
+    total_packet_delay += other.total_packet_delay;
+  }
+
+  void Subtract(const RtpPacketCounter& other) {
+    RTC_DCHECK_GE(header_bytes, other.header_bytes);
+    header_bytes -= other.header_bytes;
+    RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
+    payload_bytes -= other.payload_bytes;
+    RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
+    padding_bytes -= other.padding_bytes;
+    RTC_DCHECK_GE(packets, other.packets);
+    packets -= other.packets;
+    RTC_DCHECK_GE(total_packet_delay, other.total_packet_delay);
+    total_packet_delay -= other.total_packet_delay;
+  }
+
+  bool operator==(const RtpPacketCounter& other) const {
+    return header_bytes == other.header_bytes &&
+           payload_bytes == other.payload_bytes &&
+           padding_bytes == other.padding_bytes && packets == other.packets &&
+           total_packet_delay == other.total_packet_delay;
+  }
+
+  // Not inlined, since use of RtpPacket would result in circular includes.
+  void AddPacket(const RtpPacket& packet);
+  void AddPacket(const RtpPacketToSend& packet_to_send);
+
+  size_t TotalBytes() const {
+    return header_bytes + payload_bytes + padding_bytes;
+  }
+
+  size_t header_bytes;   // Number of bytes used by RTP headers.
+  size_t payload_bytes;  // Payload bytes, excluding RTP headers and padding.
+  size_t padding_bytes;  // Number of padding bytes.
+  uint32_t packets;      // Number of packets.
+  // The total delay of all `packets`. For RtpPacketToSend packets, this is
+  // `time_in_send_queue()`. For receive packets, this is zero.
+  webrtc::TimeDelta total_packet_delay = webrtc::TimeDelta::Zero();
+};
+
+// Data usage statistics for a (rtp) stream.
+struct StreamDataCounters {
+  StreamDataCounters();
+
+  void Add(const StreamDataCounters& other) {
+    transmitted.Add(other.transmitted);
+    retransmitted.Add(other.retransmitted);
+    fec.Add(other.fec);
+    if (other.first_packet_time_ms != -1 &&
+        (other.first_packet_time_ms < first_packet_time_ms ||
+         first_packet_time_ms == -1)) {
+      // Use oldest time.
+      first_packet_time_ms = other.first_packet_time_ms;
+    }
+  }
+
+  void Subtract(const StreamDataCounters& other) {
+    transmitted.Subtract(other.transmitted);
+    retransmitted.Subtract(other.retransmitted);
+    fec.Subtract(other.fec);
+    if (other.first_packet_time_ms != -1 &&
+        (other.first_packet_time_ms > first_packet_time_ms ||
+         first_packet_time_ms == -1)) {
+      // Use youngest time.
+      first_packet_time_ms = other.first_packet_time_ms;
+    }
+  }
+
+  int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
+    return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
+  }
+
+  // Returns the number of bytes corresponding to the actual media payload (i.e.
+  // RTP headers, padding, retransmissions and fec packets are excluded).
+  // Note this function does not have meaning for an RTX stream.
+  size_t MediaPayloadBytes() const {
+    return transmitted.payload_bytes - retransmitted.payload_bytes -
+           fec.payload_bytes;
+  }
+
+  int64_t first_packet_time_ms;  // Time when first packet is sent/received.
+  // The timestamp at which the last packet was received, i.e. the time of the
+  // local clock when it was received - not the RTP timestamp of that packet.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
+  absl::optional<int64_t> last_packet_received_timestamp_ms;
+  RtpPacketCounter transmitted;    // Number of transmitted packets/bytes.
+  RtpPacketCounter retransmitted;  // Number of retransmitted packets/bytes.
+  RtpPacketCounter fec;            // Number of redundancy packets/bytes.
+};
+
+class RtpSendRates {
+  template <std::size_t... Is>
+  constexpr std::array<DataRate, sizeof...(Is)> make_zero_array(
+      std::index_sequence<Is...>) {
+    return {{(static_cast<void>(Is), DataRate::Zero())...}};
+  }
+
+ public:
+  RtpSendRates()
+      : send_rates_(
+            make_zero_array(std::make_index_sequence<kNumMediaTypes>())) {}
+  RtpSendRates(const RtpSendRates& rhs) = default;
+  RtpSendRates& operator=(const RtpSendRates&) = default;
+
+  DataRate& operator[](RtpPacketMediaType type) {
+    return send_rates_[static_cast<size_t>(type)];
+  }
+  const DataRate& operator[](RtpPacketMediaType type) const {
+    return send_rates_[static_cast<size_t>(type)];
+  }
+  DataRate Sum() const {
+    return absl::c_accumulate(send_rates_, DataRate::Zero());
+  }
+
+ private:
+  std::array<DataRate, kNumMediaTypes> send_rates_;
+};
+
+// Callback, called whenever byte/packet counts have been updated.
+class StreamDataCountersCallback {
+ public:
+  virtual ~StreamDataCountersCallback() {}
+
+  virtual void DataCountersUpdated(const StreamDataCounters& counters,
+                                   uint32_t ssrc) = 0;
+};
+
+// Information exposed through the GetStats api.
+struct RtpReceiveStats {
+  // `packets_lost` and `jitter` are defined by RFC 3550, and exposed in the
+  // RTCReceivedRtpStreamStats dictionary, see
+  // https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*
+  int32_t packets_lost = 0;
+  // Interarrival jitter in samples.
+  uint32_t jitter = 0;
+  // Interarrival jitter in time.
+  webrtc::TimeDelta interarrival_jitter = webrtc::TimeDelta::Zero();
+
+  // Timestamp and counters exposed in RTCInboundRtpStreamStats, see
+  // https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
+  absl::optional<int64_t> last_packet_received_timestamp_ms;
+  RtpPacketCounter packet_counter;
+};
+
+// Callback, used to notify an observer whenever new rates have been estimated.
+class BitrateStatisticsObserver {
+ public:
+  virtual ~BitrateStatisticsObserver() {}
+
+  virtual void Notify(uint32_t total_bitrate_bps,
+                      uint32_t retransmit_bitrate_bps,
+                      uint32_t ssrc) = 0;
+};
+
+// Callback, used to notify an observer whenever the send-side delay is updated.
+class SendSideDelayObserver {
+ public:
+  virtual ~SendSideDelayObserver() {}
+  virtual void SendSideDelayUpdated(int avg_delay_ms,
+                                    int max_delay_ms,
+                                    uint32_t ssrc) = 0;
+};
+
+// Callback, used to notify an observer whenever a packet is sent to the
+// transport.
+// TODO(asapersson): This class will remove the need for SendSideDelayObserver.
+// Remove SendSideDelayObserver once possible.
+class SendPacketObserver {
+ public:
+  virtual ~SendPacketObserver() {}
+  virtual void OnSendPacket(uint16_t packet_id,
+                            int64_t capture_time_ms,
+                            uint32_t ssrc) = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/leb128_gn/moz.build b/third_party/libwebrtc/modules/rtp_rtcp/leb128_gn/moz.build
new file mode 100644
index 0000000000..f03986e482
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/leb128_gn/moz.build
@@ -0,0 +1,217 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/leb128.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("leb128_gn")
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h b/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h
new file mode 100644
index 0000000000..d308b2cfa8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_MOCKS_MOCK_RECOVERED_PACKET_RECEIVER_H_
+#define MODULES_RTP_RTCP_MOCKS_MOCK_RECOVERED_PACKET_RECEIVER_H_
+
+#include "modules/rtp_rtcp/include/flexfec_receiver.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRecoveredPacketReceiver : public RecoveredPacketReceiver {
+ public:
+  MOCK_METHOD(void,
+              OnRecoveredPacket,
+              (const RtpPacketReceived& packet),
+              (override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_MOCKS_MOCK_RECOVERED_PACKET_RECEIVER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h b/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h
new file mode 100644
index 0000000000..12f143ae8b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_BANDWIDTH_OBSERVER_H_
+#define MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_BANDWIDTH_OBSERVER_H_
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRtcpBandwidthObserver : public RtcpBandwidthObserver {
+ public:
+  MOCK_METHOD(void, OnReceivedEstimatedBitrate, (uint32_t), (override));
+  MOCK_METHOD(void,
+              OnReceivedRtcpReceiverReport,
+              (const ReportBlockList&, int64_t, int64_t),
+              (override));
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_BANDWIDTH_OBSERVER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h b/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
new file mode 100644
index 0000000000..e9a7d52691
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
+#define MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRtcpRttStats : public RtcpRttStats {
+ public:
+  MOCK_METHOD(void, OnRttUpdate, (int64_t rtt), (override));
+  MOCK_METHOD(int64_t, LastProcessedRtt, (), (const, override));
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_RTT_STATS_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
new file mode 100644
index 0000000000..75c30742df
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -0,0 +1,192 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
+#define MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
+
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRtpRtcpInterface : public RtpRtcpInterface {
+ public:
+  MOCK_METHOD(void,
+              IncomingRtcpPacket,
+              (const uint8_t* incoming_packet, size_t packet_length),
+              (override));
+  MOCK_METHOD(void, SetRemoteSSRC, (uint32_t ssrc), (override));
+  MOCK_METHOD(void, SetLocalSsrc, (uint32_t ssrc), (override));
+  MOCK_METHOD(void, SetMaxRtpPacketSize, (size_t size), (override));
+  MOCK_METHOD(size_t, MaxRtpPacketSize, (), (const, override));
+  MOCK_METHOD(void,
+              RegisterSendPayloadFrequency,
+              (int payload_type, int frequency),
+              (override));
+  MOCK_METHOD(int32_t,
+              DeRegisterSendPayload,
+              (int8_t payload_type),
+              (override));
+  MOCK_METHOD(void, SetExtmapAllowMixed, (bool extmap_allow_mixed), (override));
+  MOCK_METHOD(void,
+              RegisterRtpHeaderExtension,
+              (absl::string_view uri, int id),
+              (override));
+  MOCK_METHOD(void,
+              DeregisterSendRtpHeaderExtension,
+              (absl::string_view uri),
+              (override));
+  MOCK_METHOD(bool, SupportsPadding, (), (const, override));
+  MOCK_METHOD(bool, SupportsRtxPayloadPadding, (), (const, override));
+  MOCK_METHOD(uint32_t, StartTimestamp, (), (const, override));
+  MOCK_METHOD(void, SetStartTimestamp, (uint32_t timestamp), (override));
+  MOCK_METHOD(uint16_t, SequenceNumber, (), (const, override));
+  MOCK_METHOD(void, SetSequenceNumber, (uint16_t seq), (override));
+  MOCK_METHOD(void, SetRtpState, (const RtpState& rtp_state), (override));
+  MOCK_METHOD(void, SetRtxState, (const RtpState& rtp_state), (override));
+  MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override));
+  MOCK_METHOD(RtpState, GetRtpState, (), (const, override));
+  MOCK_METHOD(RtpState, GetRtxState, (), (const, override));
+  MOCK_METHOD(uint32_t, SSRC, (), (const, override));
+  MOCK_METHOD(void, SetMid, (absl::string_view mid), (override));
+  MOCK_METHOD(void, SetRtxSendStatus, (int modes), (override));
+  MOCK_METHOD(int, RtxSendStatus, (), (const, override));
+  MOCK_METHOD(absl::optional<uint32_t>, RtxSsrc, (), (const, override));
+  MOCK_METHOD(void, SetRtxSendPayloadType, (int, int), (override));
+  MOCK_METHOD(absl::optional<uint32_t>, FlexfecSsrc, (), (const, override));
+  MOCK_METHOD(int32_t, SetSendingStatus, (bool sending), (override));
+  MOCK_METHOD(bool, Sending, (), (const, override));
+  MOCK_METHOD(void, SetSendingMediaStatus, (bool sending), (override));
+  MOCK_METHOD(bool, SendingMedia, (), (const, override));
+  MOCK_METHOD(bool, IsAudioConfigured, (), (const, override));
+  MOCK_METHOD(void, SetAsPartOfAllocation, (bool), (override));
+  MOCK_METHOD(RtpSendRates, GetSendRates, (), (const, override));
+  MOCK_METHOD(bool,
+              OnSendingRtpFrame,
+              (uint32_t, int64_t, int, bool),
+              (override));
+  MOCK_METHOD(bool,
+              TrySendPacket,
+              (RtpPacketToSend * packet, const PacedPacketInfo& pacing_info),
+              (override));
+  MOCK_METHOD(void,
+              SetFecProtectionParams,
+              (const FecProtectionParams& delta_params,
+               const FecProtectionParams& key_params),
+              (override));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              FetchFecPackets,
+              (),
+              (override));
+  MOCK_METHOD(void,
+              OnAbortedRetransmissions,
+              (rtc::ArrayView<const uint16_t>),
+              (override));
+  MOCK_METHOD(void,
+              OnPacketsAcknowledged,
+              (rtc::ArrayView<const uint16_t>),
+              (override));
+  MOCK_METHOD(std::vector<std::unique_ptr<RtpPacketToSend>>,
+              GeneratePadding,
+              (size_t target_size_bytes),
+              (override));
+  MOCK_METHOD(std::vector<RtpSequenceNumberMap::Info>,
+              GetSentRtpPacketInfos,
+              (rtc::ArrayView<const uint16_t> sequence_numbers),
+              (const, override));
+  MOCK_METHOD(size_t, ExpectedPerPacketOverhead, (), (const, override));
+  MOCK_METHOD(void, OnPacketSendingThreadSwitched, (), (override));
+  MOCK_METHOD(RtcpMode, RTCP, (), (const, override));
+  MOCK_METHOD(void, SetRTCPStatus, (RtcpMode method), (override));
+  MOCK_METHOD(int32_t, SetCNAME, (absl::string_view cname), (override));
+  MOCK_METHOD(int32_t,
+              RemoteNTP,
+              (uint32_t * received_ntp_secs,
+               uint32_t* received_ntp_frac,
+               uint32_t* rtcp_arrival_time_secs,
+               uint32_t* rtcp_arrival_time_frac,
+               uint32_t* rtcp_timestamp),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              RTT,
+              (uint32_t remote_ssrc,
+               int64_t* rtt,
+               int64_t* avg_rtt,
+               int64_t* min_rtt,
+               int64_t* max_rtt),
+              (const, override));
+  MOCK_METHOD(int64_t, ExpectedRetransmissionTimeMs, (), (const, override));
+  MOCK_METHOD(int32_t, SendRTCP, (RTCPPacketType packet_type), (override));
+  MOCK_METHOD(void,
+              GetSendStreamDataCounters,
+              (StreamDataCounters*, StreamDataCounters*),
+              (const, override));
+  MOCK_METHOD(std::vector<ReportBlockData>,
+              GetLatestReportBlockData,
+              (),
+              (const, override));
+  MOCK_METHOD(absl::optional<SenderReportStats>,
+              GetSenderReportStats,
+              (),
+              (const, override));
+  MOCK_METHOD(absl::optional<NonSenderRttStats>,
+              GetNonSenderRttStats,
+              (),
+              (const, override));
+  MOCK_METHOD(void,
+              SetRemb,
+              (int64_t bitrate, std::vector<uint32_t> ssrcs),
+              (override));
+  MOCK_METHOD(void, UnsetRemb, (), (override));
+  MOCK_METHOD(int32_t,
+              SendNACK,
+              (const uint16_t* nack_list, uint16_t size),
+              (override));
+  MOCK_METHOD(void,
+              SendNack,
+              (const std::vector<uint16_t>& sequence_numbers),
+              (override));
+  MOCK_METHOD(void,
+              SetStorePacketsStatus,
+              (bool enable, uint16_t number_to_store),
+              (override));
+  MOCK_METHOD(void,
+              SendCombinedRtcpPacket,
+              (std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets),
+              (override));
+  MOCK_METHOD(int32_t,
+              SendLossNotification,
+              (uint16_t last_decoded_seq_num,
+               uint16_t last_received_seq_num,
+               bool decodability_flag,
+               bool buffering_allowed),
+              (override));
+  MOCK_METHOD(void,
+              SetVideoBitrateAllocation,
+              (const VideoBitrateAllocation&),
+              (override));
+  MOCK_METHOD(RTPSender*, RtpSender, (), (override));
+  MOCK_METHOD(const RTPSender*, RtpSender, (), (const, override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_format_gn/moz.build b/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_format_gn/moz.build
new file mode 100644
index 0000000000..da69b456d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_format_gn/moz.build
@@ -0,0 +1,274 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_received.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rtp_rtcp_format_gn")
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_gn/moz.build b/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_gn/moz.build
new file mode 100644
index 0000000000..5683b643c0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/rtp_rtcp_gn/moz.build
@@ -0,0 +1,284 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/dtmf_queue.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_descriptor_authentication.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/time_util.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/tmmbr_help.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc",
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rtp_rtcp_gn")
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/rtp_video_header_gn/moz.build b/third_party/libwebrtc/modules/rtp_rtcp/rtp_video_header_gn/moz.build
new file mode 100644
index 0000000000..99447dd149
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/rtp_video_header_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rtp_video_header_gn")
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
new file mode 100644
index 0000000000..99fc030aca
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
@@ -0,0 +1,125 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
+
+#include <limits>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr Timestamp kInvalidLastReceiveTime = Timestamp::MinusInfinity();
+}  // namespace
+
+constexpr TimeDelta AbsoluteCaptureTimeInterpolator::kInterpolationMaxInterval;
+
+AbsoluteCaptureTimeInterpolator::AbsoluteCaptureTimeInterpolator(Clock* clock)
+    : clock_(clock), last_receive_time_(kInvalidLastReceiveTime) {}
+
+uint32_t AbsoluteCaptureTimeInterpolator::GetSource(
+    uint32_t ssrc,
+    rtc::ArrayView<const uint32_t> csrcs) {
+  if (csrcs.empty()) {
+    return ssrc;
+  }
+
+  return csrcs[0];
+}
+
+absl::optional<AbsoluteCaptureTime>
+AbsoluteCaptureTimeInterpolator::OnReceivePacket(
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    uint32_t rtp_clock_frequency,
+    const absl::optional<AbsoluteCaptureTime>& received_extension) {
+  const Timestamp receive_time = clock_->CurrentTime();
+
+  MutexLock lock(&mutex_);
+
+  AbsoluteCaptureTime extension;
+  if (received_extension == absl::nullopt) {
+    if (!ShouldInterpolateExtension(receive_time, source, rtp_timestamp,
+                                    rtp_clock_frequency)) {
+      last_receive_time_ = kInvalidLastReceiveTime;
+      return absl::nullopt;
+    }
+
+    extension.absolute_capture_timestamp = InterpolateAbsoluteCaptureTimestamp(
+        rtp_timestamp, rtp_clock_frequency, last_rtp_timestamp_,
+        last_absolute_capture_timestamp_);
+    extension.estimated_capture_clock_offset =
+        last_estimated_capture_clock_offset_;
+  } else {
+    last_source_ = source;
+    last_rtp_timestamp_ = rtp_timestamp;
+    last_rtp_clock_frequency_ = rtp_clock_frequency;
+    last_absolute_capture_timestamp_ =
+        received_extension->absolute_capture_timestamp;
+    last_estimated_capture_clock_offset_ =
+        received_extension->estimated_capture_clock_offset;
+
+    last_receive_time_ = receive_time;
+
+    extension = *received_extension;
+  }
+
+  return extension;
+}
+
+uint64_t AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
+    uint32_t rtp_timestamp,
+    uint32_t rtp_clock_frequency,
+    uint32_t last_rtp_timestamp,
+    uint64_t last_absolute_capture_timestamp) {
+  RTC_DCHECK_GT(rtp_clock_frequency, 0);
+
+  return last_absolute_capture_timestamp +
+         static_cast<int64_t>(
+             rtc::dchecked_cast<uint64_t>(rtp_timestamp - last_rtp_timestamp)
+             << 32) /
+             rtp_clock_frequency;
+}
+
+bool AbsoluteCaptureTimeInterpolator::ShouldInterpolateExtension(
+    Timestamp receive_time,
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    uint32_t rtp_clock_frequency) const {
+  // Shouldn't if we don't have a previously received extension stored.
+  if (last_receive_time_ == kInvalidLastReceiveTime) {
+    return false;
+  }
+
+  // Shouldn't if the last received extension is too old.
+  if ((receive_time - last_receive_time_) > kInterpolationMaxInterval) {
+    return false;
+  }
+
+  // Shouldn't if the source has changed.
+  if (last_source_ != source) {
+    return false;
+  }
+
+  // Shouldn't if the RTP clock frequency has changed.
+  if (last_rtp_clock_frequency_ != rtp_clock_frequency) {
+    return false;
+  }
+
+  // Shouldn't if the RTP clock frequency is invalid.
+  if (rtp_clock_frequency <= 0) {
+    return false;
+  }
+
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
new file mode 100644
index 0000000000..f5ec820dd5
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
+#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
+
+#include "api/array_view.h"
+#include "api/rtp_headers.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+//
+// Helper class for interpolating the `AbsoluteCaptureTime` header extension.
+//
+// Supports the "timestamp interpolation" optimization:
+//   A receiver SHOULD memorize the capture system (i.e. CSRC/SSRC), capture
+//   timestamp, and RTP timestamp of the most recently received abs-capture-time
+//   packet on each received stream. It can then use that information, in
+//   combination with RTP timestamps of packets without abs-capture-time, to
+//   extrapolate missing capture timestamps.
+//
+// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
+//
+class AbsoluteCaptureTimeInterpolator {
+ public:
+  static constexpr TimeDelta kInterpolationMaxInterval =
+      TimeDelta::Millis(5000);
+
+  explicit AbsoluteCaptureTimeInterpolator(Clock* clock);
+
+  // Returns the source (i.e. SSRC or CSRC) of the capture system.
+  static uint32_t GetSource(uint32_t ssrc,
+                            rtc::ArrayView<const uint32_t> csrcs);
+
+  // Returns a received header extension, an interpolated header extension, or
+  // `absl::nullopt` if it's not possible to interpolate a header extension.
+  absl::optional<AbsoluteCaptureTime> OnReceivePacket(
+      uint32_t source,
+      uint32_t rtp_timestamp,
+      uint32_t rtp_clock_frequency,
+      const absl::optional<AbsoluteCaptureTime>& received_extension);
+
+ private:
+  friend class AbsoluteCaptureTimeSender;
+
+  static uint64_t InterpolateAbsoluteCaptureTimestamp(
+      uint32_t rtp_timestamp,
+      uint32_t rtp_clock_frequency,
+      uint32_t last_rtp_timestamp,
+      uint64_t last_absolute_capture_timestamp);
+
+  bool ShouldInterpolateExtension(Timestamp receive_time,
+                                  uint32_t source,
+                                  uint32_t rtp_timestamp,
+                                  uint32_t rtp_clock_frequency) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock* const clock_;
+
+  Mutex mutex_;
+
+  Timestamp last_receive_time_ RTC_GUARDED_BY(mutex_);
+
+  uint32_t last_source_ RTC_GUARDED_BY(mutex_);
+  uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_);
+  uint32_t last_rtp_clock_frequency_ RTC_GUARDED_BY(mutex_);
+  uint64_t last_absolute_capture_timestamp_ RTC_GUARDED_BY(mutex_);
+  absl::optional<int64_t> last_estimated_capture_clock_offset_
+      RTC_GUARDED_BY(mutex_);
+};  // AbsoluteCaptureTimeInterpolator
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc
new file mode 100644
index 0000000000..6a312f9b43
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator_unittest.cc
@@ -0,0 +1,353 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
+
+#include "system_wrappers/include/ntp_time.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(AbsoluteCaptureTimeInterpolatorTest, GetSourceWithoutCsrcs) {
+  constexpr uint32_t kSsrc = 12;
+
+  EXPECT_EQ(AbsoluteCaptureTimeInterpolator::GetSource(kSsrc, nullptr), kSsrc);
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest, GetSourceWithCsrcs) {
+  constexpr uint32_t kSsrc = 12;
+  constexpr uint32_t kCsrcs[] = {34, 56, 78, 90};
+
+  EXPECT_EQ(AbsoluteCaptureTimeInterpolator::GetSource(kSsrc, kCsrcs),
+            kCsrcs[0]);
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest, ReceiveExtensionReturnsExtension) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9020), absl::nullopt};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            kExtension0);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp1,
+                                         kRtpClockFrequency, kExtension1),
+            kExtension1);
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest,
+     ReceiveNoExtensionReturnsNoExtension) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 = absl::nullopt;
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            absl::nullopt);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp1,
+                                         kRtpClockFrequency, kExtension1),
+            absl::nullopt);
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest, InterpolateLaterPacketArrivingLater) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 + 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            kExtension0);
+
+  absl::optional<AbsoluteCaptureTime> extension = interpolator.OnReceivePacket(
+      kSource, kRtpTimestamp1, kRtpClockFrequency, kExtension1);
+  EXPECT_TRUE(extension.has_value());
+  EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
+            UQ32x32ToInt64Ms(kExtension0->absolute_capture_timestamp) + 20);
+  EXPECT_EQ(extension->estimated_capture_clock_offset,
+            kExtension0->estimated_capture_clock_offset);
+
+  extension = interpolator.OnReceivePacket(kSource, kRtpTimestamp2,
+                                           kRtpClockFrequency, kExtension2);
+  EXPECT_TRUE(extension.has_value());
+  EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
+            UQ32x32ToInt64Ms(kExtension0->absolute_capture_timestamp) + 40);
+  EXPECT_EQ(extension->estimated_capture_clock_offset,
+            kExtension0->estimated_capture_clock_offset);
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest,
+     InterpolateEarlierPacketArrivingLater) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 - 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 - 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            kExtension0);
+
+  absl::optional<AbsoluteCaptureTime> extension = interpolator.OnReceivePacket(
+      kSource, kRtpTimestamp1, kRtpClockFrequency, kExtension1);
+  EXPECT_TRUE(extension.has_value());
+  EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
+            UQ32x32ToInt64Ms(kExtension0->absolute_capture_timestamp) - 20);
+  EXPECT_EQ(extension->estimated_capture_clock_offset,
+            kExtension0->estimated_capture_clock_offset);
+
+  extension = interpolator.OnReceivePacket(kSource, kRtpTimestamp2,
+                                           kRtpClockFrequency, kExtension2);
+  EXPECT_TRUE(extension.has_value());
+  EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
+            UQ32x32ToInt64Ms(kExtension0->absolute_capture_timestamp) - 40);
+  EXPECT_EQ(extension->estimated_capture_clock_offset,
+            kExtension0->estimated_capture_clock_offset);
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest,
+     InterpolateLaterPacketArrivingLaterWithRtpTimestampWrapAround) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = ~uint32_t{0} - 79;
+  constexpr uint32_t kRtpTimestamp1 = 1280 - 80;
+  constexpr uint32_t kRtpTimestamp2 = 2560 - 80;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            kExtension0);
+
+  absl::optional<AbsoluteCaptureTime> extension = interpolator.OnReceivePacket(
+      kSource, kRtpTimestamp1, kRtpClockFrequency, kExtension1);
+  EXPECT_TRUE(extension.has_value());
+  EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
+            UQ32x32ToInt64Ms(kExtension0->absolute_capture_timestamp) + 20);
+  EXPECT_EQ(extension->estimated_capture_clock_offset,
+            kExtension0->estimated_capture_clock_offset);
+
+  extension = interpolator.OnReceivePacket(kSource, kRtpTimestamp2,
+                                           kRtpClockFrequency, kExtension2);
+  EXPECT_TRUE(extension.has_value());
+  EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
+            UQ32x32ToInt64Ms(kExtension0->absolute_capture_timestamp) + 40);
+  EXPECT_EQ(extension->estimated_capture_clock_offset,
+            kExtension0->estimated_capture_clock_offset);
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest,
+     InterpolateEarlierPacketArrivingLaterWithRtpTimestampWrapAround) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 799;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 - 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 - 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            kExtension0);
+
+  absl::optional<AbsoluteCaptureTime> extension = interpolator.OnReceivePacket(
+      kSource, kRtpTimestamp1, kRtpClockFrequency, kExtension1);
+  EXPECT_TRUE(extension.has_value());
+  EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
+            UQ32x32ToInt64Ms(kExtension0->absolute_capture_timestamp) - 20);
+  EXPECT_EQ(extension->estimated_capture_clock_offset,
+            kExtension0->estimated_capture_clock_offset);
+
+  extension = interpolator.OnReceivePacket(kSource, kRtpTimestamp2,
+                                           kRtpClockFrequency, kExtension2);
+  EXPECT_TRUE(extension.has_value());
+  EXPECT_EQ(UQ32x32ToInt64Ms(extension->absolute_capture_timestamp),
+            UQ32x32ToInt64Ms(kExtension0->absolute_capture_timestamp) - 40);
+  EXPECT_EQ(extension->estimated_capture_clock_offset,
+            kExtension0->estimated_capture_clock_offset);
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest, SkipInterpolateIfTooLate) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp1 + 1280;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            kExtension0);
+
+  clock.AdvanceTime(AbsoluteCaptureTimeInterpolator::kInterpolationMaxInterval);
+
+  EXPECT_TRUE(interpolator
+                  .OnReceivePacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
+                                   kExtension1)
+                  .has_value());
+
+  clock.AdvanceTimeMilliseconds(1);
+
+  EXPECT_FALSE(interpolator
+                   .OnReceivePacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
+                                    kExtension2)
+                   .has_value());
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest, SkipInterpolateIfSourceChanged) {
+  constexpr uint32_t kSource0 = 1337;
+  constexpr uint32_t kSource1 = 1338;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource0, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            kExtension0);
+
+  EXPECT_FALSE(interpolator
+                   .OnReceivePacket(kSource1, kRtpTimestamp1,
+                                    kRtpClockFrequency, kExtension1)
+                   .has_value());
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest,
+     SkipInterpolateIfRtpClockFrequencyChanged) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency0 = 64000;
+  constexpr uint32_t kRtpClockFrequency1 = 32000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 640;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp0,
+                                         kRtpClockFrequency0, kExtension0),
+            kExtension0);
+
+  EXPECT_FALSE(interpolator
+                   .OnReceivePacket(kSource, kRtpTimestamp1,
+                                    kRtpClockFrequency1, kExtension1)
+                   .has_value());
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest,
+     SkipInterpolateIfRtpClockFrequencyIsInvalid) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 0;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 640;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            kExtension0);
+
+  EXPECT_FALSE(interpolator
+                   .OnReceivePacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
+                                    kExtension1)
+                   .has_value());
+}
+
+TEST(AbsoluteCaptureTimeInterpolatorTest, SkipInterpolateIsSticky) {
+  constexpr uint32_t kSource0 = 1337;
+  constexpr uint32_t kSource1 = 1338;
+  constexpr uint32_t kSource2 = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp1 + 1280;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 = absl::nullopt;
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 = absl::nullopt;
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeInterpolator interpolator(&clock);
+
+  EXPECT_EQ(interpolator.OnReceivePacket(kSource0, kRtpTimestamp0,
+                                         kRtpClockFrequency, kExtension0),
+            kExtension0);
+
+  EXPECT_FALSE(interpolator
+                   .OnReceivePacket(kSource1, kRtpTimestamp1,
+                                    kRtpClockFrequency, kExtension1)
+                   .has_value());
+
+  EXPECT_FALSE(interpolator
+                   .OnReceivePacket(kSource2, kRtpTimestamp2,
+                                    kRtpClockFrequency, kExtension2)
+                   .has_value());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender.cc
new file mode 100644
index 0000000000..28266769ff
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender.cc
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
+
+#include <limits>
+
+#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace {
+
+constexpr Timestamp kInvalidLastSendTime = Timestamp::MinusInfinity();
+
+}  // namespace
+
+constexpr TimeDelta AbsoluteCaptureTimeSender::kInterpolationMaxInterval;
+constexpr TimeDelta AbsoluteCaptureTimeSender::kInterpolationMaxError;
+
+static_assert(
+    AbsoluteCaptureTimeInterpolator::kInterpolationMaxInterval >=
+        AbsoluteCaptureTimeSender::kInterpolationMaxInterval,
+    "Receivers should be as willing to interpolate timestamps as senders.");
+
+AbsoluteCaptureTimeSender::AbsoluteCaptureTimeSender(Clock* clock)
+    : clock_(clock), last_send_time_(kInvalidLastSendTime) {}
+
+uint32_t AbsoluteCaptureTimeSender::GetSource(
+    uint32_t ssrc,
+    rtc::ArrayView<const uint32_t> csrcs) {
+  return AbsoluteCaptureTimeInterpolator::GetSource(ssrc, csrcs);
+}
+
+absl::optional<AbsoluteCaptureTime> AbsoluteCaptureTimeSender::OnSendPacket(
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    uint32_t rtp_clock_frequency,
+    uint64_t absolute_capture_timestamp,
+    absl::optional<int64_t> estimated_capture_clock_offset) {
+  const Timestamp send_time = clock_->CurrentTime();
+
+  MutexLock lock(&mutex_);
+
+  if (!ShouldSendExtension(send_time, source, rtp_timestamp,
+                           rtp_clock_frequency, absolute_capture_timestamp,
+                           estimated_capture_clock_offset)) {
+    return absl::nullopt;
+  }
+
+  last_source_ = source;
+  last_rtp_timestamp_ = rtp_timestamp;
+  last_rtp_clock_frequency_ = rtp_clock_frequency;
+  last_absolute_capture_timestamp_ = absolute_capture_timestamp;
+  last_estimated_capture_clock_offset_ = estimated_capture_clock_offset;
+
+  last_send_time_ = send_time;
+
+  AbsoluteCaptureTime extension;
+  extension.absolute_capture_timestamp = absolute_capture_timestamp;
+  extension.estimated_capture_clock_offset = estimated_capture_clock_offset;
+  return extension;
+}
+
+bool AbsoluteCaptureTimeSender::ShouldSendExtension(
+    Timestamp send_time,
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    uint32_t rtp_clock_frequency,
+    uint64_t absolute_capture_timestamp,
+    absl::optional<int64_t> estimated_capture_clock_offset) const {
+  // Should if we've never sent anything before.
+  if (last_send_time_ == kInvalidLastSendTime) {
+    return true;
+  }
+
+  // Should if the last sent extension is too old.
+  if ((send_time - last_send_time_) > kInterpolationMaxInterval) {
+    return true;
+  }
+
+  // Should if the source has changed.
+  if (last_source_ != source) {
+    return true;
+  }
+
+  // Should if the RTP clock frequency has changed.
+  if (last_rtp_clock_frequency_ != rtp_clock_frequency) {
+    return true;
+  }
+
+  // Should if the RTP clock frequency is invalid.
+  if (rtp_clock_frequency <= 0) {
+    return true;
+  }
+
+  // Should if the estimated capture clock offset has changed.
+  if (last_estimated_capture_clock_offset_ != estimated_capture_clock_offset) {
+    return true;
+  }
+
+  // Should if interpolation would introduce too much error.
+  const uint64_t interpolated_absolute_capture_timestamp =
+      AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
+          rtp_timestamp, rtp_clock_frequency, last_rtp_timestamp_,
+          last_absolute_capture_timestamp_);
+  const int64_t interpolation_error_ms = UQ32x32ToInt64Ms(std::min(
+      interpolated_absolute_capture_timestamp - absolute_capture_timestamp,
+      absolute_capture_timestamp - interpolated_absolute_capture_timestamp));
+  if (interpolation_error_ms > kInterpolationMaxError.ms()) {
+    return true;
+  }
+
+  return false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender.h b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender.h
new file mode 100644
index 0000000000..be5a77d5e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender.h
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
+#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
+
+#include "api/array_view.h"
+#include "api/rtp_headers.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+//
+// Helper class for sending the `AbsoluteCaptureTime` header extension.
+//
+// Supports the "timestamp interpolation" optimization:
+//   A sender SHOULD save bandwidth by not sending abs-capture-time with every
+//   RTP packet. It SHOULD still send them at regular intervals (e.g. every
+//   second) to help mitigate the impact of clock drift and packet loss. Mixers
+//   SHOULD always send abs-capture-time with the first RTP packet after
+//   changing capture system.
+//
+//   Timestamp interpolation works fine as long as there’s reasonably low
+//   NTP/RTP clock drift. This is not always true. Senders that detect “jumps”
+//   between its NTP and RTP clock mappings SHOULD send abs-capture-time with
+//   the first RTP packet after such a thing happening.
+//
+// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
+//
+class AbsoluteCaptureTimeSender {
+ public:
+  static constexpr TimeDelta kInterpolationMaxInterval =
+      TimeDelta::Millis(1000);
+  static constexpr TimeDelta kInterpolationMaxError = TimeDelta::Millis(1);
+
+  explicit AbsoluteCaptureTimeSender(Clock* clock);
+
+  // Returns the source (i.e. SSRC or CSRC) of the capture system.
+  static uint32_t GetSource(uint32_t ssrc,
+                            rtc::ArrayView<const uint32_t> csrcs);
+
+  // Returns a header extension to be sent, or `absl::nullopt` if the header
+  // extension shouldn't be sent.
+  absl::optional<AbsoluteCaptureTime> OnSendPacket(
+      uint32_t source,
+      uint32_t rtp_timestamp,
+      uint32_t rtp_clock_frequency,
+      uint64_t absolute_capture_timestamp,
+      absl::optional<int64_t> estimated_capture_clock_offset);
+
+ private:
+  bool ShouldSendExtension(
+      Timestamp send_time,
+      uint32_t source,
+      uint32_t rtp_timestamp,
+      uint32_t rtp_clock_frequency,
+      uint64_t absolute_capture_timestamp,
+      absl::optional<int64_t> estimated_capture_clock_offset) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock* const clock_;
+
+  Mutex mutex_;
+
+  Timestamp last_send_time_ RTC_GUARDED_BY(mutex_);
+
+  uint32_t last_source_ RTC_GUARDED_BY(mutex_);
+  uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_);
+  uint32_t last_rtp_clock_frequency_ RTC_GUARDED_BY(mutex_);
+  uint64_t last_absolute_capture_timestamp_ RTC_GUARDED_BY(mutex_);
+  absl::optional<int64_t> last_estimated_capture_clock_offset_
+      RTC_GUARDED_BY(mutex_);
+};  // AbsoluteCaptureTimeSender
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender_unittest.cc
new file mode 100644
index 0000000000..db3fc75100
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_sender_unittest.cc
@@ -0,0 +1,374 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
+
+#include "system_wrappers/include/ntp_time.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(AbsoluteCaptureTimeSenderTest, GetSourceWithoutCsrcs) {
+  constexpr uint32_t kSsrc = 12;
+
+  EXPECT_EQ(AbsoluteCaptureTimeSender::GetSource(kSsrc, nullptr), kSsrc);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest, GetSourceWithCsrcs) {
+  constexpr uint32_t kSsrc = 12;
+  constexpr uint32_t kCsrcs[] = {34, 56, 78, 90};
+
+  EXPECT_EQ(AbsoluteCaptureTimeSender::GetSource(kSsrc, kCsrcs), kCsrcs[0]);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest, InterpolateLaterPacketSentLater) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 + 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 20), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 40), Int64MsToQ32x32(-350)};
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp0, kRtpClockFrequency,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            absl::nullopt);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            absl::nullopt);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest, InterpolateEarlierPacketSentLater) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 - 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 - 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 - 20), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 - 40), Int64MsToQ32x32(-350)};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp0, kRtpClockFrequency,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            absl::nullopt);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            absl::nullopt);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest,
+     InterpolateLaterPacketSentLaterWithRtpTimestampWrapAround) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = ~uint32_t{0} - 79;
+  constexpr uint32_t kRtpTimestamp1 = 1280 - 80;
+  constexpr uint32_t kRtpTimestamp2 = 2560 - 80;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 20), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 40), Int64MsToQ32x32(-350)};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp0, kRtpClockFrequency,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            absl::nullopt);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            absl::nullopt);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest,
+     InterpolateEarlierPacketSentLaterWithRtpTimestampWrapAround) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 799;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 - 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 - 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 - 20), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 - 40), Int64MsToQ32x32(-350)};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp0, kRtpClockFrequency,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            absl::nullopt);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            absl::nullopt);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest, SkipInterpolateIfTooLate) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 + 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 20), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 40), Int64MsToQ32x32(-350)};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp0, kRtpClockFrequency,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  clock.AdvanceTime(AbsoluteCaptureTimeSender::kInterpolationMaxInterval);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            absl::nullopt);
+
+  clock.AdvanceTimeMicroseconds(1);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            kExtension2);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest, SkipInterpolateIfSourceChanged) {
+  constexpr uint32_t kSource0 = 1337;
+  constexpr uint32_t kSource1 = 1338;
+  constexpr uint32_t kSource2 = 1338;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 + 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 20), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 40), Int64MsToQ32x32(-350)};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource0, kRtpTimestamp0, kRtpClockFrequency,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource1, kRtpTimestamp1, kRtpClockFrequency,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            kExtension1);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource2, kRtpTimestamp2, kRtpClockFrequency,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            absl::nullopt);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest, SkipInterpolateIfRtpClockFrequencyChanged) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency0 = 64000;
+  constexpr uint32_t kRtpClockFrequency1 = 32000;
+  constexpr uint32_t kRtpClockFrequency2 = 32000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 640;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 + 1280;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 20), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 40), Int64MsToQ32x32(-350)};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp0, kRtpClockFrequency0,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency1,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            kExtension1);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency2,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            absl::nullopt);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest,
+     SkipInterpolateIfRtpClockFrequencyIsInvalid) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency0 = 0;
+  constexpr uint32_t kRtpClockFrequency1 = 0;
+  constexpr uint32_t kRtpClockFrequency2 = 0;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 20), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 40), Int64MsToQ32x32(-350)};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp0, kRtpClockFrequency0,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency1,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            kExtension1);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency2,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            kExtension2);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest,
+     SkipInterpolateIfEstimatedCaptureClockOffsetChanged) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 + 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 20), Int64MsToQ32x32(370)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000 + 40), absl::nullopt};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp0, kRtpClockFrequency,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            kExtension1);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            kExtension2);
+}
+
+TEST(AbsoluteCaptureTimeSenderTest,
+     SkipInterpolateIfTooMuchInterpolationError) {
+  constexpr uint32_t kSource = 1337;
+  constexpr uint32_t kRtpClockFrequency = 64000;
+  constexpr uint32_t kRtpTimestamp0 = 1020300000;
+  constexpr uint32_t kRtpTimestamp1 = kRtpTimestamp0 + 1280;
+  constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp0 + 2560;
+  static const absl::optional<AbsoluteCaptureTime> kExtension0 =
+      AbsoluteCaptureTime{Int64MsToUQ32x32(9000), Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension1 =
+      AbsoluteCaptureTime{
+          Int64MsToUQ32x32(
+              9000 + 20 +
+              AbsoluteCaptureTimeSender::kInterpolationMaxError.ms()),
+          Int64MsToQ32x32(-350)};
+  static const absl::optional<AbsoluteCaptureTime> kExtension2 =
+      AbsoluteCaptureTime{
+          Int64MsToUQ32x32(
+              9000 + 40 +
+              AbsoluteCaptureTimeSender::kInterpolationMaxError.ms() + 1),
+          Int64MsToQ32x32(-350)};
+
+  SimulatedClock clock(0);
+  AbsoluteCaptureTimeSender sender(&clock);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp0, kRtpClockFrequency,
+                                kExtension0->absolute_capture_timestamp,
+                                kExtension0->estimated_capture_clock_offset),
+            kExtension0);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp1, kRtpClockFrequency,
+                                kExtension1->absolute_capture_timestamp,
+                                kExtension1->estimated_capture_clock_offset),
+            absl::nullopt);
+
+  EXPECT_EQ(sender.OnSendPacket(kSource, kRtpTimestamp2, kRtpClockFrequency,
+                                kExtension2->absolute_capture_timestamp,
+                                kExtension2->estimated_capture_clock_offset),
+            kExtension2);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper.cc
new file mode 100644
index 0000000000..71e7e8cf78
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper.cc
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/active_decode_targets_helper.h"
+
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+// Returns mask of ids of chains previous frame is part of.
+// Assumes for each chain frames are seen in order and no frame on any chain is
+// missing. That assumptions allows a simple detection when previous frame is
+// part of a chain.
+std::bitset<32> LastSendOnChain(int frame_diff,
+                                rtc::ArrayView<const int> chain_diffs) {
+  std::bitset<32> bitmask = 0;
+  for (size_t i = 0; i < chain_diffs.size(); ++i) {
+    if (frame_diff == chain_diffs[i]) {
+      bitmask.set(i);
+    }
+  }
+  return bitmask;
+}
+
+// Returns bitmask with first `num` bits set to 1.
+std::bitset<32> AllActive(size_t num) {
+  RTC_DCHECK_LE(num, 32);
+  return (~uint32_t{0}) >> (32 - num);
+}
+
+// Returns bitmask of chains that protect at least one active decode target.
+std::bitset<32> ActiveChains(
+    rtc::ArrayView<const int> decode_target_protected_by_chain,
+    int num_chains,
+    std::bitset<32> active_decode_targets) {
+  std::bitset<32> active_chains = 0;
+  for (size_t dt = 0; dt < decode_target_protected_by_chain.size(); ++dt) {
+    if (dt < active_decode_targets.size() && !active_decode_targets[dt]) {
+      continue;
+    }
+    int chain_idx = decode_target_protected_by_chain[dt];
+    RTC_DCHECK_LT(chain_idx, num_chains);
+    active_chains.set(chain_idx);
+  }
+  return active_chains;
+}
+
+}  // namespace
+
+void ActiveDecodeTargetsHelper::OnFrame(
+    rtc::ArrayView<const int> decode_target_protected_by_chain,
+    std::bitset<32> active_decode_targets,
+    bool is_keyframe,
+    int64_t frame_id,
+    rtc::ArrayView<const int> chain_diffs) {
+  const int num_chains = chain_diffs.size();
+  if (num_chains == 0) {
+    // Avoid printing the warning
+    // when already printed the warning for the same active decode targets, or
+    // when active_decode_targets are not changed from it's default value of
+    // all are active, including non-existent decode targets.
+    if (last_active_decode_targets_ != active_decode_targets &&
+        !active_decode_targets.all()) {
+      RTC_LOG(LS_WARNING) << "No chains are configured, but some decode "
+                             "targets might be inactive. Unsupported.";
+    }
+    last_active_decode_targets_ = active_decode_targets;
+    return;
+  }
+  const size_t num_decode_targets = decode_target_protected_by_chain.size();
+  RTC_DCHECK_GT(num_decode_targets, 0);
+  std::bitset<32> all_decode_targets = AllActive(num_decode_targets);
+  // Default value for active_decode_targets is 'all are active', i.e. all bits
+  // are set. Default value is set before number of decode targets is known.
+  // It is up to this helper to make the value cleaner and unset unused bits.
+  active_decode_targets &= all_decode_targets;
+
+  if (is_keyframe) {
+    // Key frame resets the state.
+    last_active_decode_targets_ = all_decode_targets;
+    last_active_chains_ = AllActive(num_chains);
+    unsent_on_chain_.reset();
+  } else {
+    // Update state assuming previous frame was sent.
+    unsent_on_chain_ &=
+        ~LastSendOnChain(frame_id - last_frame_id_, chain_diffs);
+  }
+  // Save for the next call to OnFrame.
+  // Though usually `frame_id == last_frame_id_ + 1`, it might not be so when
+  // frame id space is shared by several simulcast rtp streams.
+  last_frame_id_ = frame_id;
+
+  if (active_decode_targets == last_active_decode_targets_) {
+    return;
+  }
+  last_active_decode_targets_ = active_decode_targets;
+
+  if (active_decode_targets.none()) {
+    RTC_LOG(LS_ERROR) << "It is invalid to produce a frame (" << frame_id
+                      << ") while there are no active decode targets";
+    return;
+  }
+  last_active_chains_ = ActiveChains(decode_target_protected_by_chain,
+                                     num_chains, active_decode_targets);
+  // Frames that are part of inactive chains might not be produced by the
+  // encoder. Thus stop sending `active_decode_target` bitmask when it is sent
+  // on all active chains rather than on all chains.
+  unsent_on_chain_ = last_active_chains_;
+  RTC_DCHECK(!unsent_on_chain_.none());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper.h b/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper.h
new file mode 100644
index 0000000000..13755e8d80
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_ACTIVE_DECODE_TARGETS_HELPER_H_
+#define MODULES_RTP_RTCP_SOURCE_ACTIVE_DECODE_TARGETS_HELPER_H_
+
+#include <stdint.h>
+
+#include <bitset>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Helper class that decides when active_decode_target_bitmask should be written
+// into the dependency descriptor rtp header extension.
+// See: https://aomediacodec.github.io/av1-rtp-spec/#a44-switching
+// This class is thread-compatible
+class ActiveDecodeTargetsHelper {
+ public:
+  ActiveDecodeTargetsHelper() = default;
+  ActiveDecodeTargetsHelper(const ActiveDecodeTargetsHelper&) = delete;
+  ActiveDecodeTargetsHelper& operator=(const ActiveDecodeTargetsHelper&) =
+      delete;
+  ~ActiveDecodeTargetsHelper() = default;
+
+  // Decides if active decode target bitmask should be attached to the frame
+  // that is about to be sent.
+  void OnFrame(rtc::ArrayView<const int> decode_target_protected_by_chain,
+               std::bitset<32> active_decode_targets,
+               bool is_keyframe,
+               int64_t frame_id,
+               rtc::ArrayView<const int> chain_diffs);
+
+  // Returns active decode target to attach to the dependency descriptor.
+  absl::optional<uint32_t> ActiveDecodeTargetsBitmask() const {
+    if (unsent_on_chain_.none())
+      return absl::nullopt;
+    return last_active_decode_targets_.to_ulong();
+  }
+
+  std::bitset<32> ActiveChainsBitmask() const { return last_active_chains_; }
+
+ private:
+  // `unsent_on_chain_[i]` indicates last active decode
+  // target bitmask wasn't attached to a packet on the chain with id `i`.
+  std::bitset<32> unsent_on_chain_ = 0;
+  std::bitset<32> last_active_decode_targets_ = 0;
+  std::bitset<32> last_active_chains_ = 0;
+  int64_t last_frame_id_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ACTIVE_DECODE_TARGETS_HELPER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc
new file mode 100644
index 0000000000..6f64fd1418
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/active_decode_targets_helper_unittest.cc
@@ -0,0 +1,272 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/active_decode_targets_helper.h"
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+constexpr std::bitset<32> kAll = ~uint32_t{0};
+}  // namespace
+
+TEST(ActiveDecodeTargetsHelperTest,
+     ReturnsNulloptOnKeyFrameWhenAllDecodeTargetsAreActive) {
+  constexpr int kDecodeTargetProtectedByChain[] = {0, 0};
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs[] = {0};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b11,
+                 /*is_keyframe=*/true, /*frame_id=*/1, chain_diffs);
+
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+}
+
+TEST(ActiveDecodeTargetsHelperTest,
+     ReturnsNulloptOnKeyFrameWhenAllDecodeTargetsAreActiveAfterDeltaFrame) {
+  constexpr int kDecodeTargetProtectedByChain[] = {0, 0};
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs_key[] = {0};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b11,
+                 /*is_keyframe=*/true, /*frame_id=*/1, chain_diffs_key);
+  int chain_diffs_delta[] = {1};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/false, /*frame_id=*/2, chain_diffs_delta);
+
+  ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b01u);
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b11,
+                 /*is_keyframe=*/true, /*frame_id=*/3, chain_diffs_key);
+
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+}
+
+TEST(ActiveDecodeTargetsHelperTest,
+     ReturnsBitmaskOnKeyFrameWhenSomeDecodeTargetsAreInactive) {
+  constexpr int kDecodeTargetProtectedByChain[] = {0, 0};
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs[] = {0};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/true, /*frame_id=*/1, chain_diffs);
+
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b01u);
+}
+
+TEST(ActiveDecodeTargetsHelperTest,
+     ReturnsBitmaskOnKeyFrameWhenSomeDecodeTargetsAreInactiveAfterDeltaFrame) {
+  constexpr int kDecodeTargetProtectedByChain[] = {0, 0};
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs_key[] = {0};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/true, /*frame_id=*/1, chain_diffs_key);
+  int chain_diffs_delta[] = {1};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/false, /*frame_id=*/2, chain_diffs_delta);
+
+  ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/true, /*frame_id=*/3, chain_diffs_key);
+
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b01u);
+}
+
+TEST(ActiveDecodeTargetsHelperTest,
+     ReturnsNulloptWhenActiveDecodeTargetsAreUnused) {
+  constexpr int kDecodeTargetProtectedByChain[] = {0, 0};
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs[] = {0};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/kAll,
+                 /*is_keyframe=*/true, /*frame_id=*/1, chain_diffs);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/kAll,
+                 /*is_keyframe=*/false, /*frame_id=*/2, chain_diffs);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+}
+
+TEST(ActiveDecodeTargetsHelperTest,
+     ReturnsNulloptOnDeltaFrameAfterSentOnKeyFrame) {
+  constexpr int kDecodeTargetProtectedByChain[] = {0, 0};
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs_key[] = {0};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/true, /*frame_id=*/1, chain_diffs_key);
+  int chain_diffs_delta[] = {1};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/false, /*frame_id=*/2, chain_diffs_delta);
+
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+}
+
+TEST(ActiveDecodeTargetsHelperTest, ReturnsNewBitmaskOnDeltaFrame) {
+  constexpr int kDecodeTargetProtectedByChain[] = {0, 0};
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs_key[] = {0};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b11,
+                 /*is_keyframe=*/true, /*frame_id=*/1, chain_diffs_key);
+  ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+  int chain_diffs_delta[] = {1};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/false, /*frame_id=*/2, chain_diffs_delta);
+
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b01u);
+}
+
+TEST(ActiveDecodeTargetsHelperTest,
+     ReturnsBitmaskWhenAllDecodeTargetsReactivatedOnDeltaFrame) {
+  constexpr int kDecodeTargetProtectedByChain[] = {0, 0};
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs_key[] = {0};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/true, /*frame_id=*/1, chain_diffs_key);
+  ASSERT_NE(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+  int chain_diffs_delta[] = {1};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b01,
+                 /*is_keyframe=*/false, /*frame_id=*/2, chain_diffs_delta);
+  ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+
+  // Reactive all the decode targets
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/kAll,
+                 /*is_keyframe=*/false, /*frame_id=*/3, chain_diffs_delta);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b11u);
+}
+
+TEST(ActiveDecodeTargetsHelperTest, ReturnsNulloptAfterSentOnAllActiveChains) {
+  // Active decode targets (0 and 1) are protected by chains 1 and 2.
+  const std::bitset<32> kSome = 0b011;
+  constexpr int kDecodeTargetProtectedByChain[] = {2, 1, 0};
+
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs_key[] = {0, 0, 0};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b111,
+                 /*is_keyframe=*/true,
+                 /*frame_id=*/0, chain_diffs_key);
+  ASSERT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+
+  int chain_diffs_delta1[] = {1, 1, 1};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/kSome,
+                 /*is_keyframe=*/false,
+                 /*frame_id=*/1, chain_diffs_delta1);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b011u);
+
+  int chain_diffs_delta2[] = {2, 2, 1};  // Previous frame was part of chain#2
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/kSome,
+                 /*is_keyframe=*/false,
+                 /*frame_id=*/2, chain_diffs_delta2);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b011u);
+
+  // active_decode_targets_bitmask was send on chains 1 and 2. It was never sent
+  // on chain 0, but chain 0 only protects inactive decode target#2
+  int chain_diffs_delta3[] = {3, 1, 2};  // Previous frame was part of chain#1
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/kSome,
+                 /*is_keyframe=*/false,
+                 /*frame_id=*/3, chain_diffs_delta3);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+}
+
+TEST(ActiveDecodeTargetsHelperTest, ReturnsBitmaskWhenChanged) {
+  constexpr int kDecodeTargetProtectedByChain[] = {0, 1, 1};
+
+  ActiveDecodeTargetsHelper helper;
+  int chain_diffs_key[] = {0, 0};
+  helper.OnFrame(kDecodeTargetProtectedByChain, /*active_decode_targets=*/0b111,
+                 /*is_keyframe=*/true,
+                 /*frame_id=*/0, chain_diffs_key);
+  int chain_diffs_delta1[] = {1, 1};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b011,
+                 /*is_keyframe=*/false,
+                 /*frame_id=*/1, chain_diffs_delta1);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b011u);
+
+  int chain_diffs_delta2[] = {1, 2};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b101,
+                 /*is_keyframe=*/false,
+                 /*frame_id=*/2, chain_diffs_delta2);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b101u);
+
+  // active_decode_target_bitmask was send on chain0, but it was an old one.
+  int chain_diffs_delta3[] = {2, 1};
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b101,
+                 /*is_keyframe=*/false,
+                 /*frame_id=*/3, chain_diffs_delta3);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), 0b101u);
+}
+
+TEST(ActiveDecodeTargetsHelperTest, ReturnsNulloptWhenChainsAreNotUsed) {
+  const rtc::ArrayView<const int> kDecodeTargetProtectedByChain;
+  const rtc::ArrayView<const int> kNoChainDiffs;
+
+  ActiveDecodeTargetsHelper helper;
+  helper.OnFrame(kDecodeTargetProtectedByChain, /*active_decode_targets=*/kAll,
+                 /*is_keyframe=*/true,
+                 /*frame_id=*/0, kNoChainDiffs);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+
+  helper.OnFrame(kDecodeTargetProtectedByChain,
+                 /*active_decode_targets=*/0b101,
+                 /*is_keyframe=*/false,
+                 /*frame_id=*/1, kNoChainDiffs);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+}
+
+TEST(ActiveDecodeTargetsHelperTest, Supports32DecodeTargets) {
+  std::bitset<32> some;
+  std::vector<int> decode_target_protected_by_chain(32);
+  for (int i = 0; i < 32; ++i) {
+    decode_target_protected_by_chain[i] = i;
+    some[i] = i % 2 == 0;
+  }
+
+  ActiveDecodeTargetsHelper helper;
+  std::vector<int> chain_diffs_key(32, 0);
+  helper.OnFrame(decode_target_protected_by_chain,
+                 /*active_decode_targets=*/some,
+                 /*is_keyframe=*/true,
+                 /*frame_id=*/1, chain_diffs_key);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), some.to_ulong());
+  std::vector<int> chain_diffs_delta(32, 1);
+  helper.OnFrame(decode_target_protected_by_chain,
+                 /*active_decode_targets=*/some,
+                 /*is_keyframe=*/false,
+                 /*frame_id=*/2, chain_diffs_delta);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), absl::nullopt);
+  helper.OnFrame(decode_target_protected_by_chain,
+                 /*active_decode_targets=*/kAll,
+                 /*is_keyframe=*/false,
+                 /*frame_id=*/2, chain_diffs_delta);
+  EXPECT_EQ(helper.ActiveDecodeTargetsBitmask(), kAll.to_ulong());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/byte_io.h b/third_party/libwebrtc/modules/rtp_rtcp/source/byte_io.h
new file mode 100644
index 0000000000..a98eb3073b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/byte_io.h
@@ -0,0 +1,402 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_BYTE_IO_H_
+#define MODULES_RTP_RTCP_SOURCE_BYTE_IO_H_
+
+// This file contains classes for reading and writing integer types from/to
+// byte array representations. Signed/unsigned, partial (whole byte) sizes,
+// and big/little endian byte order is all supported.
+//
+// Usage examples:
+//
+// uint8_t* buffer = ...;
+//
+// // Read an unsigned 4 byte integer in big endian format
+// uint32_t val = ByteReader<uint32_t>::ReadBigEndian(buffer);
+//
+// // Read a signed 24-bit (3 byte) integer in little endian format
+// int32_t val = ByteReader<int32_t, 3>::ReadLittle(buffer);
+//
+// // Write an unsigned 8 byte integer in little endian format
+// ByteWriter<uint64_t>::WriteLittleEndian(buffer, val);
+//
+// Write an unsigned 40-bit (5 byte) integer in big endian format
+// ByteWriter<uint64_t, 5>::WriteBigEndian(buffer, val);
+//
+// These classes are implemented as recursive templetizations, inteded to make
+// it easy for the compiler to completely inline the reading/writing.
+
+#include <stdint.h>
+
+#include <limits>
+
+namespace webrtc {
+
+// According to ISO C standard ISO/IEC 9899, section 6.2.6.2 (2), the three
+// representations of signed integers allowed are two's complement, one's
+// complement and sign/magnitude. We can detect which is used by looking at
+// the two last bits of -1, which will be 11 in two's complement, 10 in one's
+// complement and 01 in sign/magnitude.
+// TODO(sprang): In the unlikely event that we actually need to support a
+// platform that doesn't use two's complement, implement conversion to/from
+// wire format.
+
+// Assume the if any one signed integer type is two's complement, then all
+// other will be too.
+static_assert(
+    (-1 & 0x03) == 0x03,
+    "Only two's complement representation of signed integers supported.");
+
+// Plain const char* won't work for static_assert, use #define instead.
+#define kSizeErrorMsg "Byte size must be less than or equal to data type size."
+
+// Utility class for getting the unsigned equivalent of a signed type.
+template <typename T>
+struct UnsignedOf;
+
+// Class for reading integers from a sequence of bytes.
+// T = type of integer, B = bytes to read, is_signed = true if signed integer.
+// If is_signed is true and B < sizeof(T), sign extension might be needed.
+template <typename T,
+          unsigned int B = sizeof(T),
+          bool is_signed = std::numeric_limits<T>::is_signed>
+class ByteReader;
+
+// Specialization of ByteReader for unsigned types.
+template <typename T, unsigned int B>
+class ByteReader<T, B, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
+    return InternalReadBigEndian(data);
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
+    return InternalReadLittleEndian(data);
+  }
+
+ private:
+  static T InternalReadBigEndian(const uint8_t* data) {
+    T val(0);
+    for (unsigned int i = 0; i < B; ++i)
+      val |= static_cast<T>(data[i]) << ((B - 1 - i) * 8);
+    return val;
+  }
+
+  static T InternalReadLittleEndian(const uint8_t* data) {
+    T val(0);
+    for (unsigned int i = 0; i < B; ++i)
+      val |= static_cast<T>(data[i]) << (i * 8);
+    return val;
+  }
+};
+
+// Specialization of ByteReader for signed types.
+template <typename T, unsigned int B>
+class ByteReader<T, B, true> {
+ public:
+  typedef typename UnsignedOf<T>::Type U;
+
+  static T ReadBigEndian(const uint8_t* data) {
+    U unsigned_val = ByteReader<T, B, false>::ReadBigEndian(data);
+    if (B < sizeof(T))
+      unsigned_val = SignExtend(unsigned_val);
+    return ReinterpretAsSigned(unsigned_val);
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    U unsigned_val = ByteReader<T, B, false>::ReadLittleEndian(data);
+    if (B < sizeof(T))
+      unsigned_val = SignExtend(unsigned_val);
+    return ReinterpretAsSigned(unsigned_val);
+  }
+
+ private:
+  // As a hack to avoid implementation-specific or undefined behavior when
+  // bit-shifting or casting signed integers, read as a signed equivalent
+  // instead and convert to signed. This is safe since we have asserted that
+  // two's complement for is used.
+  static T ReinterpretAsSigned(U unsigned_val) {
+    // An unsigned value with only the highest order bit set (ex 0x80).
+    const U kUnsignedHighestBitMask = static_cast<U>(1)
+                                      << ((sizeof(U) * 8) - 1);
+    // A signed value with only the highest bit set. Since this is two's
+    // complement form, we can use the min value from std::numeric_limits.
+    const T kSignedHighestBitMask = std::numeric_limits<T>::min();
+
+    T val;
+    if ((unsigned_val & kUnsignedHighestBitMask) != 0) {
+      // Casting is only safe when unsigned value can be represented in the
+      // signed target type, so mask out highest bit and mask it back manually.
+      val = static_cast<T>(unsigned_val & ~kUnsignedHighestBitMask);
+      val |= kSignedHighestBitMask;
+    } else {
+      val = static_cast<T>(unsigned_val);
+    }
+    return val;
+  }
+
+  // If number of bytes is less than native data type (eg 24 bit, in int32_t),
+  // and the most significant bit of the actual data is set, we must sign
+  // extend the remaining byte(s) with ones so that the correct negative
+  // number is retained.
+  // Ex: 0x810A0B -> 0xFF810A0B, but 0x710A0B -> 0x00710A0B
+  static U SignExtend(const U val) {
+    const uint8_t kMsb = static_cast<uint8_t>(val >> ((B - 1) * 8));
+    if ((kMsb & 0x80) != 0) {
+      // Create a mask where all bits used by the B bytes are set to one,
+      // for instance 0x00FFFFFF for B = 3. Bit-wise invert that mask (to
+      // (0xFF000000 in the example above) and add it to the input value.
+      // The "B % sizeof(T)" is a workaround to undefined values warnings for
+      // B == sizeof(T), in which case this code won't be called anyway.
+      const U kUsedBitsMask = (1 << ((B % sizeof(T)) * 8)) - 1;
+      return ~kUsedBitsMask | val;
+    }
+    return val;
+  }
+};
+
+// Class for writing integers to a sequence of bytes
+// T = type of integer, B = bytes to write
+template <typename T,
+          unsigned int B = sizeof(T),
+          bool is_signed = std::numeric_limits<T>::is_signed>
+class ByteWriter;
+
+// Specialization of ByteWriter for unsigned types.
+template <typename T, unsigned int B>
+class ByteWriter<T, B, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
+    for (unsigned int i = 0; i < B; ++i) {
+      data[i] = val >> ((B - 1 - i) * 8);
+    }
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(B <= sizeof(T), kSizeErrorMsg);
+    for (unsigned int i = 0; i < B; ++i) {
+      data[i] = val >> (i * 8);
+    }
+  }
+};
+
+// Specialization of ByteWriter for signed types.
+template <typename T, unsigned int B>
+class ByteWriter<T, B, true> {
+ public:
+  typedef typename UnsignedOf<T>::Type U;
+
+  static void WriteBigEndian(uint8_t* data, T val) {
+    ByteWriter<U, B, false>::WriteBigEndian(data, ReinterpretAsUnsigned(val));
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    ByteWriter<U, B, false>::WriteLittleEndian(data,
+                                               ReinterpretAsUnsigned(val));
+  }
+
+ private:
+  static U ReinterpretAsUnsigned(T val) {
+    // According to ISO C standard ISO/IEC 9899, section 6.3.1.3 (1, 2) a
+    // conversion from signed to unsigned keeps the value if the new type can
+    // represent it, and otherwise adds one more than the max value of T until
+    // the value is in range. For two's complement, this fortunately means
+    // that the bit-wise value will be intact. Thus, since we have asserted that
+    // two's complement form is actually used, a simple cast is sufficient.
+    return static_cast<U>(val);
+  }
+};
+
+// ----- Below follows specializations of UnsignedOf utility class -----
+
+template <>
+struct UnsignedOf<int8_t> {
+  typedef uint8_t Type;
+};
+template <>
+struct UnsignedOf<int16_t> {
+  typedef uint16_t Type;
+};
+template <>
+struct UnsignedOf<int32_t> {
+  typedef uint32_t Type;
+};
+template <>
+struct UnsignedOf<int64_t> {
+  typedef uint64_t Type;
+};
+
+// ----- Below follows specializations for unsigned, B in { 1, 2, 4, 8 } -----
+
+// TODO(sprang): Check if these actually help or if generic cases will be
+// unrolled to and optimized to similar performance.
+
+// Specializations for single bytes
+template <typename T>
+class ByteReader<T, 1, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    return data[0];
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    return data[0];
+  }
+};
+
+template <typename T>
+class ByteWriter<T, 1, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    data[0] = val;
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) == 1, kSizeErrorMsg);
+    data[0] = val;
+  }
+};
+
+// Specializations for two byte words
+template <typename T>
+class ByteReader<T, 2, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
+    return (data[0] << 8) | data[1];
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
+    return data[0] | (data[1] << 8);
+  }
+};
+
+template <typename T>
+class ByteWriter<T, 2, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
+    data[0] = val >> 8;
+    data[1] = val;
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 2, kSizeErrorMsg);
+    data[0] = val;
+    data[1] = val >> 8;
+  }
+};
+
+// Specializations for four byte words.
+template <typename T>
+class ByteReader<T, 4, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
+    return (Get(data, 0) << 24) | (Get(data, 1) << 16) | (Get(data, 2) << 8) |
+           Get(data, 3);
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
+    return Get(data, 0) | (Get(data, 1) << 8) | (Get(data, 2) << 16) |
+           (Get(data, 3) << 24);
+  }
+
+ private:
+  inline static T Get(const uint8_t* data, unsigned int index) {
+    return static_cast<T>(data[index]);
+  }
+};
+
+// Specializations for four byte words.
+template <typename T>
+class ByteWriter<T, 4, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
+    data[0] = val >> 24;
+    data[1] = val >> 16;
+    data[2] = val >> 8;
+    data[3] = val;
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 4, kSizeErrorMsg);
+    data[0] = val;
+    data[1] = val >> 8;
+    data[2] = val >> 16;
+    data[3] = val >> 24;
+  }
+};
+
+// Specializations for eight byte words.
+template <typename T>
+class ByteReader<T, 8, false> {
+ public:
+  static T ReadBigEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
+    return (Get(data, 0) << 56) | (Get(data, 1) << 48) | (Get(data, 2) << 40) |
+           (Get(data, 3) << 32) | (Get(data, 4) << 24) | (Get(data, 5) << 16) |
+           (Get(data, 6) << 8) | Get(data, 7);
+  }
+
+  static T ReadLittleEndian(const uint8_t* data) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
+    return Get(data, 0) | (Get(data, 1) << 8) | (Get(data, 2) << 16) |
+           (Get(data, 3) << 24) | (Get(data, 4) << 32) | (Get(data, 5) << 40) |
+           (Get(data, 6) << 48) | (Get(data, 7) << 56);
+  }
+
+ private:
+  inline static T Get(const uint8_t* data, unsigned int index) {
+    return static_cast<T>(data[index]);
+  }
+};
+
+template <typename T>
+class ByteWriter<T, 8, false> {
+ public:
+  static void WriteBigEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
+    data[0] = val >> 56;
+    data[1] = val >> 48;
+    data[2] = val >> 40;
+    data[3] = val >> 32;
+    data[4] = val >> 24;
+    data[5] = val >> 16;
+    data[6] = val >> 8;
+    data[7] = val;
+  }
+
+  static void WriteLittleEndian(uint8_t* data, T val) {
+    static_assert(sizeof(T) >= 8, kSizeErrorMsg);
+    data[0] = val;
+    data[1] = val >> 8;
+    data[2] = val >> 16;
+    data[3] = val >> 24;
+    data[4] = val >> 32;
+    data[5] = val >> 40;
+    data[6] = val >> 48;
+    data[7] = val >> 56;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_BYTE_IO_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/byte_io_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/byte_io_unittest.cc
new file mode 100644
index 0000000000..e4dea813b8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/byte_io_unittest.cc
@@ -0,0 +1,270 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+
+#include <limits>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+class ByteIoTest : public ::testing::Test {
+ protected:
+  ByteIoTest() = default;
+  ~ByteIoTest() override = default;
+
+  enum { kAlignments = sizeof(uint64_t) - 1 };
+
+  // Method to create a test value that is not the same when byte reversed.
+  template <typename T>
+  T CreateTestValue(bool negative, uint8_t num_bytes) {
+    // Examples of output:
+    // T = int32_t, negative = false, num_bytes = 4: 0x00010203
+    // T = int32_t, negative = true, num_bytes = 4: 0xFFFEFDFC
+    // T = int32_t, negative = false, num_bytes = 3: 0x000102
+    // * T = int32_t, negative = true, num_bytes = 3: 0xFFFEFD
+
+    T val = 0;
+    for (uint8_t i = 0; i != num_bytes; ++i) {
+      val = (val << 8) + (negative ? (0xFF - i) : (i + 1));
+    }
+
+    // This loop will create a sign extend mask if num_bytes if necessary.
+    // For the last example (marked * above), the number needs to be sign
+    // extended to be a valid int32_t. The sign extend mask is 0xFF000000.
+    // Comments for each step with this example below.
+    if (std::numeric_limits<T>::is_signed && negative &&
+        num_bytes < sizeof(T)) {
+      // Start with mask = 0xFFFFFFFF.
+      T mask = static_cast<T>(-1);
+      // Create a temporary for the lowest byte (0x000000FF).
+      const T neg_byte = static_cast<T>(0xFF);
+      for (int i = 0; i < num_bytes; ++i) {
+        // And the inverse of the temporary and the mask:
+        // 0xFFFFFFFF & 0xFFFFFF00 = 0xFFFFFF00.
+        // 0xFFFFFF00 & 0xFFFF00FF = 0xFFFF0000.
+        // 0xFFFF0000 & 0xFF00FFFF = 0xFF000000.
+        mask &= ~(neg_byte << (i * 8));
+      }
+      // Add the sign extension mask to the actual value.
+      val |= mask;
+    }
+    return val;
+  }
+
+  // Populate byte buffer with value, in big endian format.
+  template <typename T>
+  void PopulateTestData(uint8_t* data, T value, int num_bytes, bool bigendian) {
+    if (bigendian) {
+      for (int i = 0; i < num_bytes; ++i) {
+        data[i] = (value >> ((num_bytes - i - 1) * 8)) & 0xFF;
+      }
+    } else {
+      for (int i = 0; i < num_bytes; ++i) {
+        data[i] = (value >> (i * 8)) & 0xFF;
+      }
+    }
+  }
+
+  // Test reading big endian numbers.
+  // Template arguments: Type T, read method RM(buffer), B bytes of data.
+  template <typename T, T (*RM)(const uint8_t*), int B>
+  void TestRead(bool big_endian) {
+    // Test both for values that are positive and negative (if signed)
+    for (int neg = 0; neg < 2; ++neg) {
+      bool negative = neg > 0;
+
+      // Write test value to byte buffer, in big endian format.
+      T test_value = CreateTestValue<T>(negative, B);
+      uint8_t bytes[B + kAlignments];
+
+      // Make one test for each alignment.
+      for (int i = 0; i < kAlignments; ++i) {
+        PopulateTestData(bytes + i, test_value, B, big_endian);
+
+        // Check that test value is retrieved from buffer when used read method.
+        EXPECT_EQ(test_value, RM(bytes + i));
+      }
+    }
+  }
+
+  // Test writing big endian numbers.
+  // Template arguments: Type T, write method WM(buffer, value), B bytes of data
+  template <typename T, void (*WM)(uint8_t*, T), int B>
+  void TestWrite(bool big_endian) {
+    // Test both for values that are positive and negative (if signed).
+    for (int neg = 0; neg < 2; ++neg) {
+      bool negative = neg > 0;
+
+      // Write test value to byte buffer, in big endian format.
+      T test_value = CreateTestValue<T>(negative, B);
+      uint8_t expected_bytes[B + kAlignments];
+      uint8_t bytes[B + kAlignments];
+
+      // Make one test for each alignment.
+      for (int i = 0; i < kAlignments; ++i) {
+        PopulateTestData(expected_bytes + i, test_value, B, big_endian);
+
+        // Zero initialize buffer and let WM populate it.
+        memset(bytes, 0, B + kAlignments);
+        WM(bytes + i, test_value);
+
+        // Check that data produced by WM is big endian as expected.
+        for (int j = 0; j < B; ++j) {
+          EXPECT_EQ(expected_bytes[i + j], bytes[i + j]);
+        }
+      }
+    }
+  }
+};
+
+TEST_F(ByteIoTest, Test16UBitBigEndian) {
+  TestRead<uint16_t, ByteReader<uint16_t>::ReadBigEndian, sizeof(uint16_t)>(
+      true);
+  TestWrite<uint16_t, ByteWriter<uint16_t>::WriteBigEndian, sizeof(uint16_t)>(
+      true);
+}
+
+TEST_F(ByteIoTest, Test24UBitBigEndian) {
+  TestRead<uint32_t, ByteReader<uint32_t, 3>::ReadBigEndian, 3>(true);
+  TestWrite<uint32_t, ByteWriter<uint32_t, 3>::WriteBigEndian, 3>(true);
+}
+
+TEST_F(ByteIoTest, Test32UBitBigEndian) {
+  TestRead<uint32_t, ByteReader<uint32_t>::ReadBigEndian, sizeof(uint32_t)>(
+      true);
+  TestWrite<uint32_t, ByteWriter<uint32_t>::WriteBigEndian, sizeof(uint32_t)>(
+      true);
+}
+
+TEST_F(ByteIoTest, Test64UBitBigEndian) {
+  TestRead<uint64_t, ByteReader<uint64_t>::ReadBigEndian, sizeof(uint64_t)>(
+      true);
+  TestWrite<uint64_t, ByteWriter<uint64_t>::WriteBigEndian, sizeof(uint64_t)>(
+      true);
+}
+
+TEST_F(ByteIoTest, Test16SBitBigEndian) {
+  TestRead<int16_t, ByteReader<int16_t>::ReadBigEndian, sizeof(int16_t)>(true);
+  TestWrite<int16_t, ByteWriter<int16_t>::WriteBigEndian, sizeof(int16_t)>(
+      true);
+}
+
+TEST_F(ByteIoTest, Test24SBitBigEndian) {
+  TestRead<int32_t, ByteReader<int32_t, 3>::ReadBigEndian, 3>(true);
+  TestWrite<int32_t, ByteWriter<int32_t, 3>::WriteBigEndian, 3>(true);
+}
+
+TEST_F(ByteIoTest, Test32SBitBigEndian) {
+  TestRead<int32_t, ByteReader<int32_t>::ReadBigEndian, sizeof(int32_t)>(true);
+  TestWrite<int32_t, ByteWriter<int32_t>::WriteBigEndian, sizeof(int32_t)>(
+      true);
+}
+
+TEST_F(ByteIoTest, Test64SBitBigEndian) {
+  TestRead<int64_t, ByteReader<int64_t>::ReadBigEndian, sizeof(int64_t)>(true);
+  TestWrite<int64_t, ByteWriter<int64_t>::WriteBigEndian, sizeof(int64_t)>(
+      true);
+}
+
+TEST_F(ByteIoTest, Test16UBitLittleEndian) {
+  TestRead<uint16_t, ByteReader<uint16_t>::ReadLittleEndian, sizeof(uint16_t)>(
+      false);
+  TestWrite<uint16_t, ByteWriter<uint16_t>::WriteLittleEndian,
+            sizeof(uint16_t)>(false);
+}
+
+TEST_F(ByteIoTest, Test24UBitLittleEndian) {
+  TestRead<uint32_t, ByteReader<uint32_t, 3>::ReadLittleEndian, 3>(false);
+  TestWrite<uint32_t, ByteWriter<uint32_t, 3>::WriteLittleEndian, 3>(false);
+}
+
+TEST_F(ByteIoTest, Test32UBitLittleEndian) {
+  TestRead<uint32_t, ByteReader<uint32_t>::ReadLittleEndian, sizeof(uint32_t)>(
+      false);
+  TestWrite<uint32_t, ByteWriter<uint32_t>::WriteLittleEndian,
+            sizeof(uint32_t)>(false);
+}
+
+TEST_F(ByteIoTest, Test64UBitLittleEndian) {
+  TestRead<uint64_t, ByteReader<uint64_t>::ReadLittleEndian, sizeof(uint64_t)>(
+      false);
+  TestWrite<uint64_t, ByteWriter<uint64_t>::WriteLittleEndian,
+            sizeof(uint64_t)>(false);
+}
+
+TEST_F(ByteIoTest, Test16SBitLittleEndian) {
+  TestRead<int16_t, ByteReader<int16_t>::ReadLittleEndian, sizeof(int16_t)>(
+      false);
+  TestWrite<int16_t, ByteWriter<int16_t>::WriteLittleEndian, sizeof(int16_t)>(
+      false);
+}
+
+TEST_F(ByteIoTest, Test24SBitLittleEndian) {
+  TestRead<int32_t, ByteReader<int32_t, 3>::ReadLittleEndian, 3>(false);
+  TestWrite<int32_t, ByteWriter<int32_t, 3>::WriteLittleEndian, 3>(false);
+}
+
+TEST_F(ByteIoTest, Test32SBitLittleEndian) {
+  TestRead<int32_t, ByteReader<int32_t>::ReadLittleEndian, sizeof(int32_t)>(
+      false);
+  TestWrite<int32_t, ByteWriter<int32_t>::WriteLittleEndian, sizeof(int32_t)>(
+      false);
+}
+
+TEST_F(ByteIoTest, Test64SBitLittleEndian) {
+  TestRead<int64_t, ByteReader<int64_t>::ReadLittleEndian, sizeof(int64_t)>(
+      false);
+  TestWrite<int64_t, ByteWriter<int64_t>::WriteLittleEndian, sizeof(int64_t)>(
+      false);
+}
+
+// Sets up a fixed byte array and converts N bytes from the array into a
+// uint64_t. Verifies the value with hard-coded reference.
+TEST(ByteIo, SanityCheckFixedByteArrayUnsignedReadBigEndian) {
+  uint8_t data[8] = {0xFF, 0xEE, 0xDD, 0xCC, 0xBB, 0xAA, 0x99, 0x88};
+  uint64_t value = ByteReader<uint64_t, 2>::ReadBigEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xFFEE), value);
+  value = ByteReader<uint64_t, 3>::ReadBigEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xFFEEDD), value);
+  value = ByteReader<uint64_t, 4>::ReadBigEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xFFEEDDCC), value);
+  value = ByteReader<uint64_t, 5>::ReadBigEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xFFEEDDCCBB), value);
+  value = ByteReader<uint64_t, 6>::ReadBigEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xFFEEDDCCBBAA), value);
+  value = ByteReader<uint64_t, 7>::ReadBigEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xFFEEDDCCBBAA99), value);
+  value = ByteReader<uint64_t, 8>::ReadBigEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xFFEEDDCCBBAA9988), value);
+}
+
+// Same as above, but for little-endian reading.
+TEST(ByteIo, SanityCheckFixedByteArrayUnsignedReadLittleEndian) {
+  uint8_t data[8] = {0xFF, 0xEE, 0xDD, 0xCC, 0xBB, 0xAA, 0x99, 0x88};
+  uint64_t value = ByteReader<uint64_t, 2>::ReadLittleEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xEEFF), value);
+  value = ByteReader<uint64_t, 3>::ReadLittleEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xDDEEFF), value);
+  value = ByteReader<uint64_t, 4>::ReadLittleEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xCCDDEEFF), value);
+  value = ByteReader<uint64_t, 5>::ReadLittleEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xBBCCDDEEFF), value);
+  value = ByteReader<uint64_t, 6>::ReadLittleEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0xAABBCCDDEEFF), value);
+  value = ByteReader<uint64_t, 7>::ReadLittleEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0x99AABBCCDDEEFF), value);
+  value = ByteReader<uint64_t, 8>::ReadLittleEndian(data);
+  EXPECT_EQ(static_cast<uint64_t>(0x8899AABBCCDDEEFF), value);
+}
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater.cc
new file mode 100644
index 0000000000..a5b12cb422
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater.cc
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/capture_clock_offset_updater.h"
+
+namespace webrtc {
+
+absl::optional<int64_t>
+CaptureClockOffsetUpdater::AdjustEstimatedCaptureClockOffset(
+    absl::optional<int64_t> remote_capture_clock_offset) const {
+  if (remote_capture_clock_offset == absl::nullopt ||
+      remote_to_local_clock_offset_ == absl::nullopt) {
+    return absl::nullopt;
+  }
+
+  // Do calculations as "unsigned" to make overflows deterministic.
+  return static_cast<uint64_t>(*remote_capture_clock_offset) +
+         static_cast<uint64_t>(*remote_to_local_clock_offset_);
+}
+
+void CaptureClockOffsetUpdater::SetRemoteToLocalClockOffset(
+    absl::optional<int64_t> offset_q32x32) {
+  remote_to_local_clock_offset_ = offset_q32x32;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater.h b/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater.h
new file mode 100644
index 0000000000..71d3eb4831
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_CAPTURE_CLOCK_OFFSET_UPDATER_H_
+#define MODULES_RTP_RTCP_SOURCE_CAPTURE_CLOCK_OFFSET_UPDATER_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+//
+// Helper class for calculating the clock offset against the capturer's clock.
+//
+// This is achieved by adjusting the estimated capture clock offset in received
+// Absolute Capture Time RTP header extension (see
+// https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/), which
+// represents the clock offset between a remote sender and the capturer, by
+// adding local-to-remote clock offset.
+
+class CaptureClockOffsetUpdater {
+ public:
+  // Adjusts remote_capture_clock_offset, which originates from Absolute Capture
+  // Time RTP header extension, to get the local clock offset against the
+  // capturer's clock.
+  absl::optional<int64_t> AdjustEstimatedCaptureClockOffset(
+      absl::optional<int64_t> remote_capture_clock_offset) const;
+
+  // Sets the NTP clock offset between the sender system (which may be different
+  // from the capture system) and the local system. This information is normally
+  // provided by passing half the value of the Round-Trip Time estimation given
+  // by RTCP sender reports (see DLSR/DLRR).
+  //
+  // Note that the value must be in Q32.32-formatted fixed-point seconds.
+  void SetRemoteToLocalClockOffset(absl::optional<int64_t> offset_q32x32);
+
+ private:
+  absl::optional<int64_t> remote_to_local_clock_offset_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_CAPTURE_CLOCK_OFFSET_UPDATER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater_unittest.cc
new file mode 100644
index 0000000000..43e1dd1379
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/capture_clock_offset_updater_unittest.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/capture_clock_offset_updater.h"
+
+#include "system_wrappers/include/ntp_time.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(AbsoluteCaptureTimeReceiverTest,
+     SkipEstimatedCaptureClockOffsetIfRemoteToLocalClockOffsetIsUnknown) {
+  static const absl::optional<int64_t> kRemoteCaptureClockOffset =
+      Int64MsToQ32x32(-350);
+  CaptureClockOffsetUpdater updater;
+  updater.SetRemoteToLocalClockOffset(absl::nullopt);
+  EXPECT_EQ(
+      updater.AdjustEstimatedCaptureClockOffset(kRemoteCaptureClockOffset),
+      absl::nullopt);
+}
+
+TEST(AbsoluteCaptureTimeReceiverTest,
+     SkipEstimatedCaptureClockOffsetIfRemoteCaptureClockOffsetIsUnknown) {
+  static const absl::optional<int64_t> kCaptureClockOffsetNull = absl::nullopt;
+  CaptureClockOffsetUpdater updater;
+  updater.SetRemoteToLocalClockOffset(0);
+  EXPECT_EQ(updater.AdjustEstimatedCaptureClockOffset(kCaptureClockOffsetNull),
+            kCaptureClockOffsetNull);
+
+  static const absl::optional<int64_t> kRemoteCaptureClockOffset =
+      Int64MsToQ32x32(-350);
+  EXPECT_EQ(
+      updater.AdjustEstimatedCaptureClockOffset(kRemoteCaptureClockOffset),
+      kRemoteCaptureClockOffset);
+}
+
+TEST(AbsoluteCaptureTimeReceiverTest, EstimatedCaptureClockOffsetArithmetic) {
+  static const absl::optional<int64_t> kRemoteCaptureClockOffset =
+      Int64MsToQ32x32(-350);
+  static const absl::optional<int64_t> kRemoteToLocalClockOffset =
+      Int64MsToQ32x32(-7000007);
+  CaptureClockOffsetUpdater updater;
+  updater.SetRemoteToLocalClockOffset(kRemoteToLocalClockOffset);
+  EXPECT_THAT(
+      updater.AdjustEstimatedCaptureClockOffset(kRemoteCaptureClockOffset),
+      ::testing::Optional(::testing::Eq(*kRemoteCaptureClockOffset +
+                                        *kRemoteToLocalClockOffset)));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc
new file mode 100644
index 0000000000..f1e4eddb4b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
+
+#include <memory>
+
+#include "api/video/video_codec_type.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
+
+namespace webrtc {
+
+std::unique_ptr<VideoRtpDepacketizer> CreateVideoRtpDepacketizer(
+    VideoCodecType codec) {
+  switch (codec) {
+    case kVideoCodecH264:
+      return std::make_unique<VideoRtpDepacketizerH264>();
+    case kVideoCodecVP8:
+      return std::make_unique<VideoRtpDepacketizerVp8>();
+    case kVideoCodecVP9:
+      return std::make_unique<VideoRtpDepacketizerVp9>();
+    case kVideoCodecAV1:
+      return std::make_unique<VideoRtpDepacketizerAv1>();
+    case kVideoCodecGeneric:
+    case kVideoCodecMultiplex:
+      return std::make_unique<VideoRtpDepacketizerGeneric>();
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.h b/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.h
new file mode 100644
index 0000000000..102cacf598
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_CREATE_VIDEO_RTP_DEPACKETIZER_H_
+#define MODULES_RTP_RTCP_SOURCE_CREATE_VIDEO_RTP_DEPACKETIZER_H_
+
+#include <memory>
+
+#include "api/video/video_codec_type.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+
+namespace webrtc {
+
+std::unique_ptr<VideoRtpDepacketizer> CreateVideoRtpDepacketizer(
+    VideoCodecType codec);
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_CREATE_VIDEO_RTP_DEPACKETIZER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
new file mode 100644
index 0000000000..2e7e219f94
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
@@ -0,0 +1,465 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h"
+
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+constexpr uint32_t kTimestampTicksPerMs = 90;
+constexpr int kSendSideDelayWindowMs = 1000;
+constexpr int kBitrateStatisticsWindowMs = 1000;
+constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
+
+}  // namespace
+
+DEPRECATED_RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
+    DEPRECATED_RtpSenderEgress* sender,
+    PacketSequencer* sequence_number_assigner)
+    : transport_sequence_number_(0),
+      sender_(sender),
+      sequence_number_assigner_(sequence_number_assigner) {
+  RTC_DCHECK(sequence_number_assigner_);
+}
+DEPRECATED_RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() =
+    default;
+
+void DEPRECATED_RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
+    std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+  for (auto& packet : packets) {
+    // Assign sequence numbers, but not for flexfec which is already running on
+    // an internally maintained sequence number series.
+    if (packet->Ssrc() != sender_->FlexFecSsrc()) {
+      sequence_number_assigner_->Sequence(*packet);
+    }
+    if (!packet->SetExtension<TransportSequenceNumber>(
+            ++transport_sequence_number_)) {
+      --transport_sequence_number_;
+    }
+    packet->ReserveExtension<TransmissionOffset>();
+    packet->ReserveExtension<AbsoluteSendTime>();
+    sender_->SendPacket(packet.get(), PacedPacketInfo());
+  }
+}
+
+DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
+    const RtpRtcpInterface::Configuration& config,
+    RtpPacketHistory* packet_history)
+    : ssrc_(config.local_media_ssrc),
+      rtx_ssrc_(config.rtx_send_ssrc),
+      flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc()
+                                         : absl::nullopt),
+      populate_network2_timestamp_(config.populate_network2_timestamp),
+      clock_(config.clock),
+      packet_history_(packet_history),
+      transport_(config.outgoing_transport),
+      event_log_(config.event_log),
+      is_audio_(config.audio),
+      need_rtp_packet_infos_(config.need_rtp_packet_infos),
+      transport_feedback_observer_(config.transport_feedback_callback),
+      send_side_delay_observer_(config.send_side_delay_observer),
+      send_packet_observer_(config.send_packet_observer),
+      rtp_stats_callback_(config.rtp_stats_callback),
+      bitrate_callback_(config.send_bitrate_observer),
+      media_has_been_sent_(false),
+      force_part_of_allocation_(false),
+      timestamp_offset_(0),
+      max_delay_it_(send_delays_.end()),
+      sum_delays_ms_(0),
+      send_rates_(kNumMediaTypes,
+                  {kBitrateStatisticsWindowMs, RateStatistics::kBpsScale}),
+      rtp_sequence_number_map_(need_rtp_packet_infos_
+                                   ? std::make_unique<RtpSequenceNumberMap>(
+                                         kRtpSequenceNumberMapMaxEntries)
+                                   : nullptr) {}
+
+void DEPRECATED_RtpSenderEgress::SendPacket(
+    RtpPacketToSend* packet,
+    const PacedPacketInfo& pacing_info) {
+  RTC_DCHECK(packet);
+
+  const uint32_t packet_ssrc = packet->Ssrc();
+  RTC_DCHECK(packet->packet_type().has_value());
+  RTC_DCHECK(HasCorrectSsrc(*packet));
+  Timestamp now = clock_->CurrentTime();
+  int64_t now_ms = now.ms();
+
+  if (is_audio_) {
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
+                                    GetSendRates().Sum().kbps(), packet_ssrc);
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
+        1, "AudioNackBitrate_kbps", now_ms,
+        GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
+        packet_ssrc);
+#endif
+  } else {
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
+                                    GetSendRates().Sum().kbps(), packet_ssrc);
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
+        1, "VideoNackBitrate_kbps", now_ms,
+        GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
+        packet_ssrc);
+#endif
+  }
+
+  PacketOptions options;
+  {
+    MutexLock lock(&lock_);
+    options.included_in_allocation = force_part_of_allocation_;
+
+    if (need_rtp_packet_infos_ &&
+        packet->packet_type() == RtpPacketToSend::Type::kVideo) {
+      RTC_DCHECK(rtp_sequence_number_map_);
+      // Last packet of a frame, add it to sequence number info map.
+      const uint32_t timestamp = packet->Timestamp() - timestamp_offset_;
+      bool is_first_packet_of_frame = packet->is_first_packet_of_frame();
+      bool is_last_packet_of_frame = packet->Marker();
+
+      rtp_sequence_number_map_->InsertPacket(
+          packet->SequenceNumber(),
+          RtpSequenceNumberMap::Info(timestamp, is_first_packet_of_frame,
+                                     is_last_packet_of_frame));
+    }
+  }
+
+  // Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
+  // the pacer, these modifications of the header below are happening after the
+  // FEC protection packets are calculated. This will corrupt recovered packets
+  // at the same place. It's not an issue for extensions, which are present in
+  // all the packets (their content just may be incorrect on recovered packets).
+  // In case of VideoTimingExtension, since it's present not in every packet,
+  // data after rtp header may be corrupted if these packets are protected by
+  // the FEC.
+  int64_t diff_ms = now_ms - packet->capture_time().ms();
+  if (packet->HasExtension<TransmissionOffset>()) {
+    packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff_ms);
+  }
+  if (packet->HasExtension<AbsoluteSendTime>()) {
+    packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
+  }
+
+  if (packet->HasExtension<VideoTimingExtension>()) {
+    if (populate_network2_timestamp_) {
+      packet->set_network2_time(now);
+    } else {
+      packet->set_pacer_exit_time(now);
+    }
+  }
+
+  const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio ||
+                        packet->packet_type() == RtpPacketMediaType::kVideo;
+
+  // Downstream code actually uses this flag to distinguish between media and
+  // everything else.
+  options.is_retransmit = !is_media;
+  if (auto packet_id = packet->GetExtension<TransportSequenceNumber>()) {
+    options.packet_id = *packet_id;
+    options.included_in_feedback = true;
+    options.included_in_allocation = true;
+    AddPacketToTransportFeedback(*packet_id, *packet, pacing_info);
+  }
+
+  options.additional_data = packet->additional_data();
+
+  if (packet->packet_type() != RtpPacketMediaType::kPadding &&
+      packet->packet_type() != RtpPacketMediaType::kRetransmission) {
+    UpdateDelayStatistics(packet->capture_time().ms(), now_ms, packet_ssrc);
+    UpdateOnSendPacket(options.packet_id, packet->capture_time().ms(),
+                       packet_ssrc);
+  }
+
+  const bool send_success = SendPacketToNetwork(*packet, options, pacing_info);
+
+  // Put packet in retransmission history or update pending status even if
+  // actual sending fails.
+  if (is_media && packet->allow_retransmission()) {
+    packet_history_->PutRtpPacket(std::make_unique<RtpPacketToSend>(*packet),
+                                  now);
+  } else if (packet->retransmitted_sequence_number()) {
+    packet_history_->MarkPacketAsSent(*packet->retransmitted_sequence_number());
+  }
+
+  if (send_success) {
+    MutexLock lock(&lock_);
+    UpdateRtpStats(*packet);
+    media_has_been_sent_ = true;
+  }
+}
+
+void DEPRECATED_RtpSenderEgress::ProcessBitrateAndNotifyObservers() {
+  if (!bitrate_callback_)
+    return;
+
+  MutexLock lock(&lock_);
+  RtpSendRates send_rates = GetSendRatesLocked();
+  bitrate_callback_->Notify(
+      send_rates.Sum().bps(),
+      send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
+}
+
+RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRates() const {
+  MutexLock lock(&lock_);
+  return GetSendRatesLocked();
+}
+
+RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRatesLocked() const {
+  const int64_t now_ms = clock_->TimeInMilliseconds();
+  RtpSendRates current_rates;
+  for (size_t i = 0; i < kNumMediaTypes; ++i) {
+    RtpPacketMediaType type = static_cast<RtpPacketMediaType>(i);
+    current_rates[type] =
+        DataRate::BitsPerSec(send_rates_[i].Rate(now_ms).value_or(0));
+  }
+  return current_rates;
+}
+
+void DEPRECATED_RtpSenderEgress::GetDataCounters(
+    StreamDataCounters* rtp_stats,
+    StreamDataCounters* rtx_stats) const {
+  MutexLock lock(&lock_);
+  *rtp_stats = rtp_stats_;
+  *rtx_stats = rtx_rtp_stats_;
+}
+
+void DEPRECATED_RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
+    bool part_of_allocation) {
+  MutexLock lock(&lock_);
+  force_part_of_allocation_ = part_of_allocation;
+}
+
+bool DEPRECATED_RtpSenderEgress::MediaHasBeenSent() const {
+  MutexLock lock(&lock_);
+  return media_has_been_sent_;
+}
+
+void DEPRECATED_RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) {
+  MutexLock lock(&lock_);
+  media_has_been_sent_ = media_sent;
+}
+
+void DEPRECATED_RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) {
+  MutexLock lock(&lock_);
+  timestamp_offset_ = timestamp;
+}
+
+std::vector<RtpSequenceNumberMap::Info>
+DEPRECATED_RtpSenderEgress::GetSentRtpPacketInfos(
+    rtc::ArrayView<const uint16_t> sequence_numbers) const {
+  RTC_DCHECK(!sequence_numbers.empty());
+  if (!need_rtp_packet_infos_) {
+    return std::vector<RtpSequenceNumberMap::Info>();
+  }
+
+  std::vector<RtpSequenceNumberMap::Info> results;
+  results.reserve(sequence_numbers.size());
+
+  MutexLock lock(&lock_);
+  for (uint16_t sequence_number : sequence_numbers) {
+    const auto& info = rtp_sequence_number_map_->Get(sequence_number);
+    if (!info) {
+      // The empty vector will be returned. We can delay the clearing
+      // of the vector until after we exit the critical section.
+      return std::vector<RtpSequenceNumberMap::Info>();
+    }
+    results.push_back(*info);
+  }
+
+  return results;
+}
+
+bool DEPRECATED_RtpSenderEgress::HasCorrectSsrc(
+    const RtpPacketToSend& packet) const {
+  switch (*packet.packet_type()) {
+    case RtpPacketMediaType::kAudio:
+    case RtpPacketMediaType::kVideo:
+      return packet.Ssrc() == ssrc_;
+    case RtpPacketMediaType::kRetransmission:
+    case RtpPacketMediaType::kPadding:
+      // Both padding and retransmission must be on either the media or the
+      // RTX stream.
+      return packet.Ssrc() == rtx_ssrc_ || packet.Ssrc() == ssrc_;
+    case RtpPacketMediaType::kForwardErrorCorrection:
+      // FlexFEC is on separate SSRC, ULPFEC uses media SSRC.
+      return packet.Ssrc() == ssrc_ || packet.Ssrc() == flexfec_ssrc_;
+  }
+  return false;
+}
+
+void DEPRECATED_RtpSenderEgress::AddPacketToTransportFeedback(
+    uint16_t packet_id,
+    const RtpPacketToSend& packet,
+    const PacedPacketInfo& pacing_info) {
+  if (transport_feedback_observer_) {
+    RtpPacketSendInfo packet_info;
+    packet_info.media_ssrc = ssrc_;
+    packet_info.transport_sequence_number = packet_id;
+    packet_info.rtp_sequence_number = packet.SequenceNumber();
+    packet_info.length = packet.size();
+    packet_info.pacing_info = pacing_info;
+    packet_info.packet_type = packet.packet_type();
+    transport_feedback_observer_->OnAddPacket(packet_info);
+  }
+}
+
+void DEPRECATED_RtpSenderEgress::UpdateDelayStatistics(int64_t capture_time_ms,
+                                                       int64_t now_ms,
+                                                       uint32_t ssrc) {
+  if (!send_side_delay_observer_ || capture_time_ms <= 0)
+    return;
+
+  int avg_delay_ms = 0;
+  int max_delay_ms = 0;
+  {
+    MutexLock lock(&lock_);
+    // Compute the max and average of the recent capture-to-send delays.
+    // The time complexity of the current approach depends on the distribution
+    // of the delay values. This could be done more efficiently.
+
+    // Remove elements older than kSendSideDelayWindowMs.
+    auto lower_bound =
+        send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs);
+    for (auto it = send_delays_.begin(); it != lower_bound; ++it) {
+      if (max_delay_it_ == it) {
+        max_delay_it_ = send_delays_.end();
+      }
+      sum_delays_ms_ -= it->second;
+    }
+    send_delays_.erase(send_delays_.begin(), lower_bound);
+    if (max_delay_it_ == send_delays_.end()) {
+      // Removed the previous max. Need to recompute.
+      RecomputeMaxSendDelay();
+    }
+
+    // Add the new element.
+    RTC_DCHECK_GE(now_ms, 0);
+    RTC_DCHECK_LE(now_ms, std::numeric_limits<int64_t>::max() / 2);
+    RTC_DCHECK_GE(capture_time_ms, 0);
+    RTC_DCHECK_LE(capture_time_ms, std::numeric_limits<int64_t>::max() / 2);
+    int64_t diff_ms = now_ms - capture_time_ms;
+    RTC_DCHECK_GE(diff_ms, static_cast<int64_t>(0));
+    RTC_DCHECK_LE(diff_ms, std::numeric_limits<int>::max());
+    int new_send_delay = rtc::dchecked_cast<int>(now_ms - capture_time_ms);
+    SendDelayMap::iterator it;
+    bool inserted;
+    std::tie(it, inserted) =
+        send_delays_.insert(std::make_pair(now_ms, new_send_delay));
+    if (!inserted) {
+      // TODO(terelius): If we have multiple delay measurements during the same
+      // millisecond then we keep the most recent one. It is not clear that this
+      // is the right decision, but it preserves an earlier behavior.
+      int previous_send_delay = it->second;
+      sum_delays_ms_ -= previous_send_delay;
+      it->second = new_send_delay;
+      if (max_delay_it_ == it && new_send_delay < previous_send_delay) {
+        RecomputeMaxSendDelay();
+      }
+    }
+    if (max_delay_it_ == send_delays_.end() ||
+        it->second >= max_delay_it_->second) {
+      max_delay_it_ = it;
+    }
+    sum_delays_ms_ += new_send_delay;
+
+    size_t num_delays = send_delays_.size();
+    RTC_DCHECK(max_delay_it_ != send_delays_.end());
+    max_delay_ms = rtc::dchecked_cast<int>(max_delay_it_->second);
+    int64_t avg_ms = (sum_delays_ms_ + num_delays / 2) / num_delays;
+    RTC_DCHECK_GE(avg_ms, static_cast<int64_t>(0));
+    RTC_DCHECK_LE(avg_ms,
+                  static_cast<int64_t>(std::numeric_limits<int>::max()));
+    avg_delay_ms =
+        rtc::dchecked_cast<int>((sum_delays_ms_ + num_delays / 2) / num_delays);
+  }
+  send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms,
+                                                  ssrc);
+}
+
+void DEPRECATED_RtpSenderEgress::RecomputeMaxSendDelay() {
+  max_delay_it_ = send_delays_.begin();
+  for (auto it = send_delays_.begin(); it != send_delays_.end(); ++it) {
+    if (it->second >= max_delay_it_->second) {
+      max_delay_it_ = it;
+    }
+  }
+}
+
+void DEPRECATED_RtpSenderEgress::UpdateOnSendPacket(int packet_id,
+                                                    int64_t capture_time_ms,
+                                                    uint32_t ssrc) {
+  if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1) {
+    return;
+  }
+
+  send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
+}
+
+bool DEPRECATED_RtpSenderEgress::SendPacketToNetwork(
+    const RtpPacketToSend& packet,
+    const PacketOptions& options,
+    const PacedPacketInfo& pacing_info) {
+  int bytes_sent = -1;
+  if (transport_) {
+    bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
+                     ? static_cast<int>(packet.size())
+                     : -1;
+    if (event_log_ && bytes_sent > 0) {
+      event_log_->Log(std::make_unique<RtcEventRtpPacketOutgoing>(
+          packet, pacing_info.probe_cluster_id));
+    }
+  }
+
+  if (bytes_sent <= 0) {
+    RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
+    return false;
+  }
+  return true;
+}
+
+void DEPRECATED_RtpSenderEgress::UpdateRtpStats(const RtpPacketToSend& packet) {
+  int64_t now_ms = clock_->TimeInMilliseconds();
+
+  StreamDataCounters* counters =
+      packet.Ssrc() == rtx_ssrc_ ? &rtx_rtp_stats_ : &rtp_stats_;
+
+  if (counters->first_packet_time_ms == -1) {
+    counters->first_packet_time_ms = now_ms;
+  }
+
+  if (packet.packet_type() == RtpPacketMediaType::kForwardErrorCorrection) {
+    counters->fec.AddPacket(packet);
+  }
+
+  if (packet.packet_type() == RtpPacketMediaType::kRetransmission) {
+    counters->retransmitted.AddPacket(packet);
+  }
+  counters->transmitted.AddPacket(packet);
+
+  RTC_DCHECK(packet.packet_type().has_value());
+  send_rates_[static_cast<size_t>(*packet.packet_type())].Update(packet.size(),
+                                                                 now_ms);
+
+  if (rtp_stats_callback_) {
+    rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h b/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
new file mode 100644
index 0000000000..609a90d4fe
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_
+#define MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/call/transport.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/units/data_rate.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/packet_sequencer.h"
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class DEPRECATED_RtpSenderEgress {
+ public:
+  // Helper class that redirects packets directly to the send part of this class
+  // without passing through an actual paced sender.
+  class NonPacedPacketSender : public RtpPacketSender {
+   public:
+    NonPacedPacketSender(DEPRECATED_RtpSenderEgress* sender,
+                         PacketSequencer* sequence_number_assigner);
+    virtual ~NonPacedPacketSender();
+
+    void EnqueuePackets(
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;
+    void RemovePacketsForSsrc(uint32_t ssrc) override {}
+
+   private:
+    uint16_t transport_sequence_number_;
+    DEPRECATED_RtpSenderEgress* const sender_;
+    PacketSequencer* sequence_number_assigner_;
+  };
+
+  DEPRECATED_RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
+                             RtpPacketHistory* packet_history);
+  ~DEPRECATED_RtpSenderEgress() = default;
+
+  void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info)
+      RTC_LOCKS_EXCLUDED(lock_);
+  uint32_t Ssrc() const { return ssrc_; }
+  absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
+  absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
+
+  void ProcessBitrateAndNotifyObservers() RTC_LOCKS_EXCLUDED(lock_);
+  RtpSendRates GetSendRates() const RTC_LOCKS_EXCLUDED(lock_);
+  void GetDataCounters(StreamDataCounters* rtp_stats,
+                       StreamDataCounters* rtx_stats) const
+      RTC_LOCKS_EXCLUDED(lock_);
+
+  void ForceIncludeSendPacketsInAllocation(bool part_of_allocation)
+      RTC_LOCKS_EXCLUDED(lock_);
+  bool MediaHasBeenSent() const RTC_LOCKS_EXCLUDED(lock_);
+  void SetMediaHasBeenSent(bool media_sent) RTC_LOCKS_EXCLUDED(lock_);
+  void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(lock_);
+
+  // For each sequence number in `sequence_number`, recall the last RTP packet
+  // which bore it - its timestamp and whether it was the first and/or last
+  // packet in that frame. If all of the given sequence numbers could be
+  // recalled, return a vector with all of them (in corresponding order).
+  // If any could not be recalled, return an empty vector.
+  std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+      rtc::ArrayView<const uint16_t> sequence_numbers) const
+      RTC_LOCKS_EXCLUDED(lock_);
+
+ private:
+  // Maps capture time in milliseconds to send-side delay in milliseconds.
+  // Send-side delay is the difference between transmission time and capture
+  // time.
+  typedef std::map<int64_t, int> SendDelayMap;
+
+  RtpSendRates GetSendRatesLocked() const RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  bool HasCorrectSsrc(const RtpPacketToSend& packet) const;
+  void AddPacketToTransportFeedback(uint16_t packet_id,
+                                    const RtpPacketToSend& packet,
+                                    const PacedPacketInfo& pacing_info);
+  void UpdateDelayStatistics(int64_t capture_time_ms,
+                             int64_t now_ms,
+                             uint32_t ssrc);
+  void RecomputeMaxSendDelay() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  void UpdateOnSendPacket(int packet_id,
+                          int64_t capture_time_ms,
+                          uint32_t ssrc);
+  // Sends packet on to `transport_`, leaving the RTP module.
+  bool SendPacketToNetwork(const RtpPacketToSend& packet,
+                           const PacketOptions& options,
+                           const PacedPacketInfo& pacing_info);
+  void UpdateRtpStats(const RtpPacketToSend& packet)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+
+  const uint32_t ssrc_;
+  const absl::optional<uint32_t> rtx_ssrc_;
+  const absl::optional<uint32_t> flexfec_ssrc_;
+  const bool populate_network2_timestamp_;
+  Clock* const clock_;
+  RtpPacketHistory* const packet_history_;
+  Transport* const transport_;
+  RtcEventLog* const event_log_;
+  const bool is_audio_;
+  const bool need_rtp_packet_infos_;
+
+  TransportFeedbackObserver* const transport_feedback_observer_;
+  SendSideDelayObserver* const send_side_delay_observer_;
+  SendPacketObserver* const send_packet_observer_;
+  StreamDataCountersCallback* const rtp_stats_callback_;
+  BitrateStatisticsObserver* const bitrate_callback_;
+
+  mutable Mutex lock_;
+  bool media_has_been_sent_ RTC_GUARDED_BY(lock_);
+  bool force_part_of_allocation_ RTC_GUARDED_BY(lock_);
+  uint32_t timestamp_offset_ RTC_GUARDED_BY(lock_);
+
+  SendDelayMap send_delays_ RTC_GUARDED_BY(lock_);
+  SendDelayMap::const_iterator max_delay_it_ RTC_GUARDED_BY(lock_);
+  // The sum of delays over a kSendSideDelayWindowMs sliding window.
+  int64_t sum_delays_ms_ RTC_GUARDED_BY(lock_);
+  StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_);
+  StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
+  // One element per value in RtpPacketMediaType, with index matching value.
+  std::vector<RateStatistics> send_rates_ RTC_GUARDED_BY(lock_);
+
+  // Maps sent packets' sequence numbers to a tuple consisting of:
+  // 1. The timestamp, without the randomizing offset mandated by the RFC.
+  // 2. Whether the packet was the first in its frame.
+  // 3. Whether the packet was the last in its frame.
+  const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_
+      RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/dtmf_queue.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/dtmf_queue.cc
new file mode 100644
index 0000000000..df06d2a2f3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/dtmf_queue.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/dtmf_queue.h"
+
+#include <stddef.h>
+
+#include "rtc_base/checks.h"
+
+namespace {
+constexpr size_t kDtmfOutbandMax = 20;
+}
+
+namespace webrtc {
+DtmfQueue::DtmfQueue() {}
+
+DtmfQueue::~DtmfQueue() {}
+
+bool DtmfQueue::AddDtmf(const Event& event) {
+  MutexLock lock(&dtmf_mutex_);
+  if (queue_.size() >= kDtmfOutbandMax) {
+    return false;
+  }
+  queue_.push_back(event);
+  return true;
+}
+
+bool DtmfQueue::NextDtmf(Event* event) {
+  RTC_DCHECK(event);
+  MutexLock lock(&dtmf_mutex_);
+  if (queue_.empty()) {
+    return false;
+  }
+
+  *event = queue_.front();
+  queue_.pop_front();
+  return true;
+}
+
+bool DtmfQueue::PendingDtmf() const {
+  MutexLock lock(&dtmf_mutex_);
+  return !queue_.empty();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/dtmf_queue.h b/third_party/libwebrtc/modules/rtp_rtcp/source/dtmf_queue.h
new file mode 100644
index 0000000000..1d1867fd27
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/dtmf_queue.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
+#define MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
+
+#include <stdint.h>
+
+#include <list>
+
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+class DtmfQueue {
+ public:
+  struct Event {
+    uint16_t duration_ms = 0;
+    uint8_t payload_type = 0;
+    uint8_t key = 0;
+    uint8_t level = 0;
+  };
+
+  DtmfQueue();
+  ~DtmfQueue();
+
+  bool AddDtmf(const Event& event);
+  bool NextDtmf(Event* event);
+  bool PendingDtmf() const;
+
+ private:
+  mutable Mutex dtmf_mutex_;
+  std::list<Event> queue_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc
new file mode 100644
index 0000000000..9dbc012368
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc
@@ -0,0 +1,660 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
+
+namespace {
+// clang-format off
+#define kMaskBursty1_1 \
+  0x80, 0x00
+
+#define kMaskBursty2_1 \
+  0xc0, 0x00
+
+#define kMaskBursty2_2 \
+  0x80, 0x00, \
+  0xc0, 0x00
+
+#define kMaskBursty3_1 \
+  0xe0, 0x00
+
+#define kMaskBursty3_2 \
+  0xc0, 0x00, \
+  0xa0, 0x00
+
+#define kMaskBursty3_3 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00
+
+#define kMaskBursty4_1 \
+  0xf0, 0x00
+
+#define kMaskBursty4_2 \
+  0xa0, 0x00, \
+  0xd0, 0x00
+
+#define kMaskBursty4_3 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x90, 0x00
+
+#define kMaskBursty4_4 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00
+
+#define kMaskBursty5_1 \
+  0xf8, 0x00
+
+#define kMaskBursty5_2 \
+  0xd0, 0x00, \
+  0xa8, 0x00
+
+#define kMaskBursty5_3 \
+  0x70, 0x00, \
+  0x90, 0x00, \
+  0xc8, 0x00
+
+#define kMaskBursty5_4 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x88, 0x00
+
+#define kMaskBursty5_5 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00
+
+#define kMaskBursty6_1 \
+  0xfc, 0x00
+
+#define kMaskBursty6_2 \
+  0xa8, 0x00, \
+  0xd4, 0x00
+
+#define kMaskBursty6_3 \
+  0x94, 0x00, \
+  0xc8, 0x00, \
+  0x64, 0x00
+
+#define kMaskBursty6_4 \
+  0x60, 0x00, \
+  0x38, 0x00, \
+  0x88, 0x00, \
+  0xc4, 0x00
+
+#define kMaskBursty6_5 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x84, 0x00
+
+#define kMaskBursty6_6 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00
+
+#define kMaskBursty7_1 \
+  0xfe, 0x00
+
+#define kMaskBursty7_2 \
+  0xd4, 0x00, \
+  0xaa, 0x00
+
+#define kMaskBursty7_3 \
+  0xc8, 0x00, \
+  0x74, 0x00, \
+  0x92, 0x00
+
+#define kMaskBursty7_4 \
+  0x38, 0x00, \
+  0x8a, 0x00, \
+  0xc4, 0x00, \
+  0x62, 0x00
+
+#define kMaskBursty7_5 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x1c, 0x00, \
+  0x84, 0x00, \
+  0xc2, 0x00
+
+#define kMaskBursty7_6 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x82, 0x00
+
+#define kMaskBursty7_7 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00
+
+#define kMaskBursty8_1 \
+  0xff, 0x00
+
+#define kMaskBursty8_2 \
+  0xaa, 0x00, \
+  0xd5, 0x00
+
+#define kMaskBursty8_3 \
+  0x74, 0x00, \
+  0x92, 0x00, \
+  0xc9, 0x00
+
+#define kMaskBursty8_4 \
+  0x8a, 0x00, \
+  0xc5, 0x00, \
+  0x62, 0x00, \
+  0x31, 0x00
+
+#define kMaskBursty8_5 \
+  0x30, 0x00, \
+  0x1c, 0x00, \
+  0x85, 0x00, \
+  0xc2, 0x00, \
+  0x61, 0x00
+
+#define kMaskBursty8_6 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0e, 0x00, \
+  0x82, 0x00, \
+  0xc1, 0x00
+
+#define kMaskBursty8_7 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x81, 0x00
+
+#define kMaskBursty8_8 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00
+
+#define kMaskBursty9_1 \
+  0xff, 0x80
+
+#define kMaskBursty9_2 \
+  0xd5, 0x00, \
+  0xaa, 0x80
+
+#define kMaskBursty9_3 \
+  0x92, 0x00, \
+  0xc9, 0x00, \
+  0x74, 0x80
+
+#define kMaskBursty9_4 \
+  0xc5, 0x00, \
+  0x62, 0x00, \
+  0x39, 0x00, \
+  0x8a, 0x80
+
+#define kMaskBursty9_5 \
+  0x1c, 0x00, \
+  0x85, 0x00, \
+  0xc2, 0x80, \
+  0x61, 0x00, \
+  0x30, 0x80
+
+#define kMaskBursty9_6 \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0e, 0x00, \
+  0x82, 0x80, \
+  0xc1, 0x00, \
+  0x60, 0x80
+
+#define kMaskBursty9_7 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x07, 0x00, \
+  0x81, 0x00, \
+  0xc0, 0x80
+
+#define kMaskBursty9_8 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x80, 0x80
+
+#define kMaskBursty9_9 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80
+
+#define kMaskBursty10_1 \
+  0xff, 0xc0
+
+#define kMaskBursty10_2 \
+  0xaa, 0x80, \
+  0xd5, 0x40
+
+#define kMaskBursty10_3 \
+  0xc9, 0x00, \
+  0x74, 0x80, \
+  0x92, 0x40
+
+#define kMaskBursty10_4 \
+  0x62, 0x00, \
+  0x39, 0x00, \
+  0x8a, 0x80, \
+  0xc5, 0x40
+
+#define kMaskBursty10_5 \
+  0x85, 0x00, \
+  0xc2, 0x80, \
+  0x61, 0x40, \
+  0x30, 0x80, \
+  0x18, 0x40
+
+#define kMaskBursty10_6 \
+  0x18, 0x00, \
+  0x0e, 0x00, \
+  0x82, 0x80, \
+  0xc1, 0x40, \
+  0x60, 0x80, \
+  0x30, 0x40
+
+#define kMaskBursty10_7 \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x07, 0x00, \
+  0x81, 0x40, \
+  0xc0, 0x80, \
+  0x60, 0x40
+
+#define kMaskBursty10_8 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x80, 0x80, \
+  0xc0, 0x40
+
+#define kMaskBursty10_9 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x80, 0x40
+
+#define kMaskBursty10_10 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0
+
+#define kMaskBursty11_1 \
+  0xff, 0xe0
+
+#define kMaskBursty11_2 \
+  0xd5, 0x40, \
+  0xaa, 0xa0
+
+#define kMaskBursty11_3 \
+  0x74, 0x80, \
+  0x92, 0x40, \
+  0xc9, 0x20
+
+#define kMaskBursty11_4 \
+  0x39, 0x00, \
+  0x8a, 0x80, \
+  0xc5, 0x40, \
+  0x62, 0x20
+
+#define kMaskBursty11_5 \
+  0xc2, 0xc0, \
+  0x61, 0x00, \
+  0x30, 0xa0, \
+  0x1c, 0x40, \
+  0x85, 0x20
+
+#define kMaskBursty11_6 \
+  0x0e, 0x00, \
+  0x82, 0x80, \
+  0xc1, 0x40, \
+  0x60, 0xa0, \
+  0x30, 0x40, \
+  0x18, 0x20
+
+#define kMaskBursty11_7 \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x07, 0x00, \
+  0x81, 0x40, \
+  0xc0, 0xa0, \
+  0x60, 0x40, \
+  0x30, 0x20
+
+#define kMaskBursty11_8 \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x40, \
+  0x80, 0xa0, \
+  0xc0, 0x40, \
+  0x60, 0x20
+
+#define kMaskBursty11_9 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x80, 0x40, \
+  0xc0, 0x20
+
+#define kMaskBursty11_10 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x80, 0x20
+
+#define kMaskBursty11_11 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x00, 0x60
+
+#define kMaskBursty12_1 \
+  0xff, 0xf0
+
+#define kMaskBursty12_2 \
+  0xaa, 0xa0, \
+  0xd5, 0x50
+
+#define kMaskBursty12_3 \
+  0x92, 0x40, \
+  0xc9, 0x20, \
+  0x74, 0x90
+
+#define kMaskBursty12_4 \
+  0x8a, 0x80, \
+  0xc5, 0x40, \
+  0x62, 0x20, \
+  0x39, 0x10
+
+#define kMaskBursty12_5 \
+  0x61, 0x00, \
+  0x30, 0xa0, \
+  0x1c, 0x50, \
+  0x85, 0x20, \
+  0xc2, 0x90
+
+#define kMaskBursty12_6 \
+  0x82, 0x90, \
+  0xc1, 0x40, \
+  0x60, 0xa0, \
+  0x30, 0x50, \
+  0x18, 0x20, \
+  0x0c, 0x10
+
+#define kMaskBursty12_7 \
+  0x0c, 0x00, \
+  0x07, 0x00, \
+  0x81, 0x40, \
+  0xc0, 0xa0, \
+  0x60, 0x50, \
+  0x30, 0x20, \
+  0x18, 0x10
+
+#define kMaskBursty12_8 \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x80, 0xa0, \
+  0xc0, 0x50, \
+  0x60, 0x20, \
+  0x30, 0x10
+
+#define kMaskBursty12_9 \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x80, 0x50, \
+  0xc0, 0x20, \
+  0x60, 0x10
+
+#define kMaskBursty12_10 \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x80, 0x20, \
+  0xc0, 0x10
+
+#define kMaskBursty12_11 \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x00, 0x60, \
+  0x80, 0x10
+
+#define kMaskBursty12_12 \
+  0x80, 0x00, \
+  0xc0, 0x00, \
+  0x60, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0x0c, 0x00, \
+  0x06, 0x00, \
+  0x03, 0x00, \
+  0x01, 0x80, \
+  0x00, 0xc0, \
+  0x00, 0x60, \
+  0x00, 0x30
+
+#define kPacketMaskBursty1 1, \
+  kMaskBursty1_1
+
+#define kPacketMaskBursty2 2, \
+  kMaskBursty2_1, \
+  kMaskBursty2_2
+
+#define kPacketMaskBursty3 3, \
+  kMaskBursty3_1, \
+  kMaskBursty3_2, \
+  kMaskBursty3_3
+
+#define kPacketMaskBursty4 4, \
+  kMaskBursty4_1, \
+  kMaskBursty4_2, \
+  kMaskBursty4_3, \
+  kMaskBursty4_4
+
+#define kPacketMaskBursty5 5, \
+  kMaskBursty5_1, \
+  kMaskBursty5_2, \
+  kMaskBursty5_3, \
+  kMaskBursty5_4, \
+  kMaskBursty5_5
+
+#define kPacketMaskBursty6 6, \
+  kMaskBursty6_1, \
+  kMaskBursty6_2, \
+  kMaskBursty6_3, \
+  kMaskBursty6_4, \
+  kMaskBursty6_5, \
+  kMaskBursty6_6
+
+#define kPacketMaskBursty7 7, \
+  kMaskBursty7_1, \
+  kMaskBursty7_2, \
+  kMaskBursty7_3, \
+  kMaskBursty7_4, \
+  kMaskBursty7_5, \
+  kMaskBursty7_6, \
+  kMaskBursty7_7
+
+#define kPacketMaskBursty8 8, \
+  kMaskBursty8_1, \
+  kMaskBursty8_2, \
+  kMaskBursty8_3, \
+  kMaskBursty8_4, \
+  kMaskBursty8_5, \
+  kMaskBursty8_6, \
+  kMaskBursty8_7, \
+  kMaskBursty8_8
+
+#define kPacketMaskBursty9 9, \
+  kMaskBursty9_1, \
+  kMaskBursty9_2, \
+  kMaskBursty9_3, \
+  kMaskBursty9_4, \
+  kMaskBursty9_5, \
+  kMaskBursty9_6, \
+  kMaskBursty9_7, \
+  kMaskBursty9_8, \
+  kMaskBursty9_9
+
+#define kPacketMaskBursty10 10, \
+  kMaskBursty10_1, \
+  kMaskBursty10_2, \
+  kMaskBursty10_3, \
+  kMaskBursty10_4, \
+  kMaskBursty10_5, \
+  kMaskBursty10_6, \
+  kMaskBursty10_7, \
+  kMaskBursty10_8, \
+  kMaskBursty10_9, \
+  kMaskBursty10_10
+
+#define kPacketMaskBursty11 11, \
+  kMaskBursty11_1, \
+  kMaskBursty11_2, \
+  kMaskBursty11_3, \
+  kMaskBursty11_4, \
+  kMaskBursty11_5, \
+  kMaskBursty11_6, \
+  kMaskBursty11_7, \
+  kMaskBursty11_8, \
+  kMaskBursty11_9, \
+  kMaskBursty11_10, \
+  kMaskBursty11_11
+
+#define kPacketMaskBursty12 12, \
+  kMaskBursty12_1, \
+  kMaskBursty12_2, \
+  kMaskBursty12_3, \
+  kMaskBursty12_4, \
+  kMaskBursty12_5, \
+  kMaskBursty12_6, \
+  kMaskBursty12_7, \
+  kMaskBursty12_8, \
+  kMaskBursty12_9, \
+  kMaskBursty12_10, \
+  kMaskBursty12_11, \
+  kMaskBursty12_12
+
+// clang-format on
+}  // namespace
+
+namespace webrtc {
+namespace fec_private_tables {
+
+const uint8_t kPacketMaskBurstyTbl[] = {
+    12,
+    kPacketMaskBursty1,
+    kPacketMaskBursty2,
+    kPacketMaskBursty3,
+    kPacketMaskBursty4,
+    kPacketMaskBursty5,
+    kPacketMaskBursty6,
+    kPacketMaskBursty7,
+    kPacketMaskBursty8,
+    kPacketMaskBursty9,
+    kPacketMaskBursty10,
+    kPacketMaskBursty11,
+    kPacketMaskBursty12,
+};
+
+}  // namespace fec_private_tables
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
new file mode 100644
index 0000000000..217d9505e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_
+#define MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_
+
+// This file contains a set of packets masks for the FEC code. The masks in
+// this table are specifically designed to favor recovery of bursty/consecutive
+// loss network conditions. The tradeoff is worse recovery for random losses.
+// These packet masks are currently defined to protect up to 12 media packets.
+// They have the following property: for any packet mask defined by the
+// parameters (k,m), where k = number of media packets, m = number of FEC
+// packets, all "consecutive" losses of size <= m are completely recoverable.
+// By consecutive losses we mean consecutive with respect to the sequence
+// number ordering of the list (media and FEC) of packets. The difference
+// between these masks (`kFecMaskBursty`) and `kFecMaskRandom` type, defined
+// in fec_private_tables.h, is more significant for longer codes
+// (i.e., more packets/symbols in the code, so larger (k,m), i.e.,  k > 4,
+// m > 3).
+
+#include <stdint.h>
+
+namespace webrtc {
+namespace fec_private_tables {
+
+extern const uint8_t kPacketMaskBurstyTbl[];
+
+}  // namespace fec_private_tables
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_BURSTY_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc
new file mode 100644
index 0000000000..c62f7d5606
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
+
+#include "modules/rtp_rtcp/source/fec_private_tables_random.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+constexpr uint8_t kMaskRandom15_6[] = {0x82, 0x08, 0x41, 0x04, 0x20, 0x82,
+                                       0x10, 0x40, 0x08, 0x20, 0x04, 0x10};
+}
+
+namespace fec_private_tables {
+
+using internal::LookUpInFecTable;
+
+TEST(FecTable, TestBurstyLookup) {
+  rtc::ArrayView<const uint8_t> result;
+  result = LookUpInFecTable(&kPacketMaskBurstyTbl[0], 0, 0);
+  // Should match kMaskBursty1_1.
+  EXPECT_EQ(2u, result.size());
+  EXPECT_EQ(0x80u, result[0]);
+
+  result = LookUpInFecTable(&kPacketMaskBurstyTbl[0], 3, 0);
+  // Should match kMaskBursty4_1.
+  EXPECT_EQ(2u, result.size());
+  EXPECT_EQ(0xf0u, result[0]);
+  EXPECT_EQ(0x00u, result[1]);
+
+  result = LookUpInFecTable(&kPacketMaskBurstyTbl[0], 1, 1);
+  // Should match kMaskBursty2_2.
+  EXPECT_EQ(4u, result.size());
+  EXPECT_EQ(0x80u, result[0]);
+  EXPECT_EQ(0xc0u, result[2]);
+
+  result = LookUpInFecTable(&kPacketMaskBurstyTbl[0], 11, 11);
+  // Should match kMaskBursty12_12.
+  EXPECT_EQ(24u, result.size());
+  EXPECT_EQ(0x80u, result[0]);
+  EXPECT_EQ(0x30u, result[23]);
+}
+
+TEST(FecTable, TestRandomLookup) {
+  rtc::ArrayView<const uint8_t> result;
+  result = LookUpInFecTable(&kPacketMaskRandomTbl[0], 0, 0);
+  EXPECT_EQ(2u, result.size());
+  EXPECT_EQ(0x80u, result[0]);
+  EXPECT_EQ(0x00u, result[1]);
+
+  result = LookUpInFecTable(&kPacketMaskRandomTbl[0], 4, 1);
+  // kMaskRandom5_2.
+  EXPECT_EQ(4u, result.size());
+  EXPECT_EQ(0xa8u, result[0]);
+  EXPECT_EQ(0xd0u, result[2]);
+}
+
+TEST(FecTable, TestRandomGenerated) {
+  FecMaskType fec_mask_type = kFecMaskRandom;
+  int num_media_packets = 15;
+  int num_fec_packets = 6;
+  size_t mask_size = sizeof(kMaskRandom15_6) / sizeof(uint8_t);
+  internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
+  rtc::ArrayView<const uint8_t> mask =
+      mask_table.LookUp(num_media_packets, num_fec_packets);
+  EXPECT_EQ(mask.size(), mask_size);
+  for (size_t i = 0; i < mask_size; ++i) {
+    EXPECT_EQ(mask[i], kMaskRandom15_6[i]);
+  }
+}
+
+}  // namespace fec_private_tables
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc
new file mode 100644
index 0000000000..3cac5db17b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc
@@ -0,0 +1,660 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/fec_private_tables_random.h"
+
+namespace {
+// clang-format off
+#define kMaskRandom1_1 \
+  0x80, 0x00
+
+#define kMaskRandom2_1 \
+  0xc0, 0x00
+
+#define kMaskRandom2_2 \
+  0xc0, 0x00, \
+  0x80, 0x00
+
+#define kMaskRandom3_1 \
+  0xe0, 0x00
+
+#define kMaskRandom3_2 \
+  0xc0, 0x00, \
+  0xa0, 0x00
+
+#define kMaskRandom3_3 \
+  0xc0, 0x00, \
+  0xa0, 0x00, \
+  0x60, 0x00
+
+#define kMaskRandom4_1 \
+  0xf0, 0x00
+
+#define kMaskRandom4_2 \
+  0xc0, 0x00, \
+  0xb0, 0x00
+
+#define kMaskRandom4_3 \
+  0xc0, 0x00, \
+  0xb0, 0x00, \
+  0x60, 0x00
+
+#define kMaskRandom4_4 \
+  0xc0, 0x00, \
+  0xa0, 0x00, \
+  0x30, 0x00, \
+  0x50, 0x00
+
+#define kMaskRandom5_1 \
+  0xf8, 0x00
+
+#define kMaskRandom5_2 \
+  0xa8, 0x00, \
+  0xd0, 0x00
+
+#define kMaskRandom5_3 \
+  0xb0, 0x00, \
+  0xc8, 0x00, \
+  0x50, 0x00
+
+#define kMaskRandom5_4 \
+  0xc8, 0x00, \
+  0xb0, 0x00, \
+  0x50, 0x00, \
+  0x28, 0x00
+
+#define kMaskRandom5_5 \
+  0xc0, 0x00, \
+  0x30, 0x00, \
+  0x18, 0x00, \
+  0xa0, 0x00, \
+  0x48, 0x00
+
+#define kMaskRandom6_1 \
+  0xfc, 0x00
+
+#define kMaskRandom6_2 \
+  0xa8, 0x00, \
+  0xd4, 0x00
+
+#define kMaskRandom6_3 \
+  0xd0, 0x00, \
+  0x68, 0x00, \
+  0xa4, 0x00
+
+#define kMaskRandom6_4 \
+  0xa8, 0x00, \
+  0x58, 0x00, \
+  0x64, 0x00, \
+  0x94, 0x00
+
+#define kMaskRandom6_5 \
+  0xa8, 0x00, \
+  0x84, 0x00, \
+  0x64, 0x00, \
+  0x90, 0x00, \
+  0x58, 0x00
+
+#define kMaskRandom6_6 \
+  0x98, 0x00, \
+  0x64, 0x00, \
+  0x50, 0x00, \
+  0x14, 0x00, \
+  0xa8, 0x00, \
+  0xe0, 0x00
+
+#define kMaskRandom7_1 \
+  0xfe, 0x00
+
+#define kMaskRandom7_2 \
+  0xd4, 0x00, \
+  0xaa, 0x00
+
+#define kMaskRandom7_3 \
+  0xd0, 0x00, \
+  0xaa, 0x00, \
+  0x64, 0x00
+
+#define kMaskRandom7_4 \
+  0xd0, 0x00, \
+  0xaa, 0x00, \
+  0x64, 0x00, \
+  0x1c, 0x00
+
+#define kMaskRandom7_5 \
+  0x0c, 0x00, \
+  0xb0, 0x00, \
+  0x1a, 0x00, \
+  0xc4, 0x00, \
+  0x62, 0x00
+
+#define kMaskRandom7_6 \
+  0x8c, 0x00, \
+  0x4a, 0x00, \
+  0x64, 0x00, \
+  0xd0, 0x00, \
+  0xa0, 0x00, \
+  0x32, 0x00
+
+#define kMaskRandom7_7 \
+  0x4a, 0x00, \
+  0x94, 0x00, \
+  0x1a, 0x00, \
+  0xc4, 0x00, \
+  0x28, 0x00, \
+  0xc2, 0x00, \
+  0x34, 0x00
+
+#define kMaskRandom8_1 \
+  0xff, 0x00
+
+#define kMaskRandom8_2 \
+  0xaa, 0x00, \
+  0xd5, 0x00
+
+#define kMaskRandom8_3 \
+  0xc5, 0x00, \
+  0x92, 0x00, \
+  0x6a, 0x00
+
+#define kMaskRandom8_4 \
+  0x45, 0x00, \
+  0xb4, 0x00, \
+  0x6a, 0x00, \
+  0x89, 0x00
+
+#define kMaskRandom8_5 \
+  0x8c, 0x00, \
+  0x92, 0x00, \
+  0x2b, 0x00, \
+  0x51, 0x00, \
+  0x64, 0x00
+
+#define kMaskRandom8_6 \
+  0xa1, 0x00, \
+  0x52, 0x00, \
+  0x91, 0x00, \
+  0x2a, 0x00, \
+  0xc4, 0x00, \
+  0x4c, 0x00
+
+#define kMaskRandom8_7 \
+  0x15, 0x00, \
+  0xc2, 0x00, \
+  0x25, 0x00, \
+  0x62, 0x00, \
+  0x58, 0x00, \
+  0x8c, 0x00, \
+  0xa3, 0x00
+
+#define kMaskRandom8_8 \
+  0x25, 0x00, \
+  0x8a, 0x00, \
+  0x91, 0x00, \
+  0x68, 0x00, \
+  0x32, 0x00, \
+  0x43, 0x00, \
+  0xc4, 0x00, \
+  0x1c, 0x00
+
+#define kMaskRandom9_1 \
+  0xff, 0x80
+
+#define kMaskRandom9_2 \
+  0xaa, 0x80, \
+  0xd5, 0x00
+
+#define kMaskRandom9_3 \
+  0xa5, 0x00, \
+  0xc8, 0x00, \
+  0x52, 0x80
+
+#define kMaskRandom9_4 \
+  0xa2, 0x00, \
+  0xc9, 0x00, \
+  0x52, 0x80, \
+  0x24, 0x80
+
+#define kMaskRandom9_5 \
+  0x8c, 0x00, \
+  0x25, 0x00, \
+  0x92, 0x80, \
+  0x41, 0x80, \
+  0x58, 0x00
+
+#define kMaskRandom9_6 \
+  0x84, 0x80, \
+  0x27, 0x00, \
+  0x51, 0x80, \
+  0x1a, 0x00, \
+  0x68, 0x00, \
+  0x89, 0x00
+
+#define kMaskRandom9_7 \
+  0x8c, 0x00, \
+  0x47, 0x00, \
+  0x81, 0x80, \
+  0x12, 0x80, \
+  0x58, 0x00, \
+  0x28, 0x80, \
+  0xb4, 0x00
+
+#define kMaskRandom9_8 \
+  0x2c, 0x00, \
+  0x91, 0x00, \
+  0x40, 0x80, \
+  0x06, 0x80, \
+  0xc8, 0x00, \
+  0x45, 0x00, \
+  0x30, 0x80, \
+  0xa2, 0x00
+
+#define kMaskRandom9_9 \
+  0x4c, 0x00, \
+  0x62, 0x00, \
+  0x91, 0x00, \
+  0x42, 0x80, \
+  0xa4, 0x00, \
+  0x13, 0x00, \
+  0x30, 0x80, \
+  0x88, 0x80, \
+  0x09, 0x00
+
+#define kMaskRandom10_1 \
+  0xff, 0xc0
+
+#define kMaskRandom10_10 \
+  0x4c, 0x00, \
+  0x51, 0x00, \
+  0xa0, 0x40, \
+  0x04, 0xc0, \
+  0x03, 0x80, \
+  0x86, 0x00, \
+  0x29, 0x00, \
+  0x42, 0x40, \
+  0x98, 0x00, \
+  0x30, 0x80
+
+#define kMaskRandom10_2 \
+  0xaa, 0x80, \
+  0xd5, 0x40
+
+#define kMaskRandom10_3 \
+  0xa4, 0x40, \
+  0xc9, 0x00, \
+  0x52, 0x80
+
+#define kMaskRandom10_4 \
+  0xca, 0x00, \
+  0x32, 0x80, \
+  0xa1, 0x40, \
+  0x55, 0x00
+
+#define kMaskRandom10_5 \
+  0xca, 0x00, \
+  0x32, 0x80, \
+  0xa1, 0x40, \
+  0x55, 0x00, \
+  0x08, 0xc0
+
+#define kMaskRandom10_6 \
+  0x0e, 0x00, \
+  0x33, 0x00, \
+  0x10, 0xc0, \
+  0x45, 0x40, \
+  0x88, 0x80, \
+  0xe0, 0x00
+
+#define kMaskRandom10_7 \
+  0x46, 0x00, \
+  0x33, 0x00, \
+  0x80, 0xc0, \
+  0x0c, 0x40, \
+  0x28, 0x80, \
+  0x94, 0x00, \
+  0xc1, 0x00
+
+#define kMaskRandom10_8 \
+  0x2c, 0x00, \
+  0x81, 0x80, \
+  0xa0, 0x40, \
+  0x05, 0x40, \
+  0x18, 0x80, \
+  0xc2, 0x00, \
+  0x22, 0x80, \
+  0x50, 0x40
+
+#define kMaskRandom10_9 \
+  0x4c, 0x00, \
+  0x23, 0x00, \
+  0x88, 0xc0, \
+  0x21, 0x40, \
+  0x52, 0x80, \
+  0x94, 0x00, \
+  0x26, 0x00, \
+  0x48, 0x40, \
+  0x91, 0x80
+
+#define kMaskRandom11_1 \
+  0xff, 0xe0
+
+#define kMaskRandom11_10 \
+  0x64, 0x40, \
+  0x51, 0x40, \
+  0xa9, 0x00, \
+  0x04, 0xc0, \
+  0xd0, 0x00, \
+  0x82, 0x40, \
+  0x21, 0x20, \
+  0x0c, 0x20, \
+  0x4a, 0x00, \
+  0x12, 0xa0
+
+#define kMaskRandom11_11 \
+  0x46, 0x40, \
+  0x33, 0x20, \
+  0x99, 0x00, \
+  0x05, 0x80, \
+  0x80, 0xa0, \
+  0x84, 0x40, \
+  0x40, 0x60, \
+  0x0a, 0x80, \
+  0x68, 0x00, \
+  0x10, 0x20, \
+  0x30, 0x40
+
+#define kMaskRandom11_2 \
+  0xec, 0xc0, \
+  0x9b, 0xa0
+
+#define kMaskRandom11_3 \
+  0xca, 0xc0, \
+  0xf1, 0x40, \
+  0xb6, 0x20
+
+#define kMaskRandom11_4 \
+  0xc4, 0xc0, \
+  0x31, 0x60, \
+  0x4b, 0x20, \
+  0x2c, 0xa0
+
+#define kMaskRandom11_5 \
+  0x86, 0x80, \
+  0x23, 0x20, \
+  0x16, 0x20, \
+  0x4c, 0x20, \
+  0x41, 0xc0
+
+#define kMaskRandom11_6 \
+  0x64, 0x40, \
+  0x51, 0x40, \
+  0x0c, 0xa0, \
+  0xa1, 0x20, \
+  0x12, 0xa0, \
+  0x8a, 0x40
+
+#define kMaskRandom11_7 \
+  0x46, 0x40, \
+  0x33, 0x20, \
+  0x91, 0x80, \
+  0xa4, 0x20, \
+  0x50, 0xa0, \
+  0x84, 0xc0, \
+  0x09, 0x60
+
+#define kMaskRandom11_8 \
+  0x0c, 0x80, \
+  0x80, 0x60, \
+  0xa0, 0x80, \
+  0x05, 0x40, \
+  0x43, 0x00, \
+  0x1a, 0x00, \
+  0x60, 0x20, \
+  0x14, 0x20
+
+#define kMaskRandom11_9 \
+  0x46, 0x40, \
+  0x62, 0x60, \
+  0x8c, 0x00, \
+  0x01, 0x60, \
+  0x07, 0x80, \
+  0xa0, 0x80, \
+  0x18, 0xa0, \
+  0x91, 0x00, \
+  0x78, 0x00
+
+#define kMaskRandom12_1 \
+  0xff, 0xf0
+
+#define kMaskRandom12_10 \
+  0x51, 0x40, \
+  0x45, 0x10, \
+  0x80, 0xd0, \
+  0x24, 0x20, \
+  0x0a, 0x20, \
+  0x00, 0xe0, \
+  0xb8, 0x00, \
+  0x09, 0x10, \
+  0x56, 0x00, \
+  0xa2, 0x80
+
+#define kMaskRandom12_11 \
+  0x53, 0x60, \
+  0x21, 0x30, \
+  0x10, 0x90, \
+  0x00, 0x70, \
+  0x0c, 0x10, \
+  0x40, 0xc0, \
+  0x6a, 0x00, \
+  0x86, 0x00, \
+  0x24, 0x80, \
+  0x89, 0x00, \
+  0xc0, 0x20
+
+#define kMaskRandom12_12 \
+  0x10, 0x60, \
+  0x02, 0x30, \
+  0x40, 0x50, \
+  0x21, 0x80, \
+  0x81, 0x10, \
+  0x14, 0x80, \
+  0x98, 0x00, \
+  0x08, 0x90, \
+  0x62, 0x00, \
+  0x24, 0x20, \
+  0x8a, 0x00, \
+  0x84, 0x40
+
+#define kMaskRandom12_2 \
+  0xec, 0xc0, \
+  0x93, 0xb0
+
+#define kMaskRandom12_3 \
+  0x9b, 0x80, \
+  0x4f, 0x10, \
+  0x3c, 0x60
+
+#define kMaskRandom12_4 \
+  0x8b, 0x20, \
+  0x14, 0xb0, \
+  0x22, 0xd0, \
+  0x45, 0x50
+
+#define kMaskRandom12_5 \
+  0x53, 0x60, \
+  0x64, 0x20, \
+  0x0c, 0xc0, \
+  0x82, 0xa0, \
+  0x09, 0x30
+
+#define kMaskRandom12_6 \
+  0x51, 0x40, \
+  0xc5, 0x10, \
+  0x21, 0x80, \
+  0x12, 0x30, \
+  0x08, 0xe0, \
+  0x2e, 0x00
+
+#define kMaskRandom12_7 \
+  0x53, 0x60, \
+  0x21, 0x30, \
+  0x90, 0x90, \
+  0x02, 0x50, \
+  0x06, 0xa0, \
+  0x2c, 0x00, \
+  0x88, 0x60
+
+#define kMaskRandom12_8 \
+  0x20, 0x60, \
+  0x80, 0x30, \
+  0x42, 0x40, \
+  0x01, 0x90, \
+  0x14, 0x10, \
+  0x0a, 0x80, \
+  0x38, 0x00, \
+  0xc5, 0x00
+
+#define kMaskRandom12_9 \
+  0x53, 0x60, \
+  0xe4, 0x20, \
+  0x24, 0x40, \
+  0xa1, 0x10, \
+  0x18, 0x30, \
+  0x03, 0x90, \
+  0x8a, 0x10, \
+  0x04, 0x90, \
+  0x00, 0xe0
+
+#define kPacketMaskRandom1 1, \
+  kMaskRandom1_1
+
+#define kPacketMaskRandom2 2, \
+  kMaskRandom2_1, \
+  kMaskRandom2_2
+
+#define kPacketMaskRandom3 3, \
+  kMaskRandom3_1, \
+  kMaskRandom3_2, \
+  kMaskRandom3_3
+
+#define kPacketMaskRandom4 4, \
+  kMaskRandom4_1, \
+  kMaskRandom4_2, \
+  kMaskRandom4_3, \
+  kMaskRandom4_4
+
+#define kPacketMaskRandom5 5, \
+  kMaskRandom5_1, \
+  kMaskRandom5_2, \
+  kMaskRandom5_3, \
+  kMaskRandom5_4, \
+  kMaskRandom5_5
+
+#define kPacketMaskRandom6 6, \
+  kMaskRandom6_1, \
+  kMaskRandom6_2, \
+  kMaskRandom6_3, \
+  kMaskRandom6_4, \
+  kMaskRandom6_5, \
+  kMaskRandom6_6
+
+#define kPacketMaskRandom7 7, \
+  kMaskRandom7_1, \
+  kMaskRandom7_2, \
+  kMaskRandom7_3, \
+  kMaskRandom7_4, \
+  kMaskRandom7_5, \
+  kMaskRandom7_6, \
+  kMaskRandom7_7
+
+#define kPacketMaskRandom8 8, \
+  kMaskRandom8_1, \
+  kMaskRandom8_2, \
+  kMaskRandom8_3, \
+  kMaskRandom8_4, \
+  kMaskRandom8_5, \
+  kMaskRandom8_6, \
+  kMaskRandom8_7, \
+  kMaskRandom8_8
+
+#define kPacketMaskRandom9 9, \
+  kMaskRandom9_1, \
+  kMaskRandom9_2, \
+  kMaskRandom9_3, \
+  kMaskRandom9_4, \
+  kMaskRandom9_5, \
+  kMaskRandom9_6, \
+  kMaskRandom9_7, \
+  kMaskRandom9_8, \
+  kMaskRandom9_9
+
+#define kPacketMaskRandom10 10, \
+  kMaskRandom10_1, \
+  kMaskRandom10_2, \
+  kMaskRandom10_3, \
+  kMaskRandom10_4, \
+  kMaskRandom10_5, \
+  kMaskRandom10_6, \
+  kMaskRandom10_7, \
+  kMaskRandom10_8, \
+  kMaskRandom10_9, \
+  kMaskRandom10_10
+
+#define kPacketMaskRandom11 11, \
+  kMaskRandom11_1, \
+  kMaskRandom11_2, \
+  kMaskRandom11_3, \
+  kMaskRandom11_4, \
+  kMaskRandom11_5, \
+  kMaskRandom11_6, \
+  kMaskRandom11_7, \
+  kMaskRandom11_8, \
+  kMaskRandom11_9, \
+  kMaskRandom11_10, \
+  kMaskRandom11_11
+
+#define kPacketMaskRandom12 12, \
+  kMaskRandom12_1, \
+  kMaskRandom12_2, \
+  kMaskRandom12_3, \
+  kMaskRandom12_4, \
+  kMaskRandom12_5, \
+  kMaskRandom12_6, \
+  kMaskRandom12_7, \
+  kMaskRandom12_8, \
+  kMaskRandom12_9, \
+  kMaskRandom12_10, \
+  kMaskRandom12_11, \
+  kMaskRandom12_12
+
+// clang-format on
+}  // namespace
+
+namespace webrtc {
+namespace fec_private_tables {
+
+const uint8_t kPacketMaskRandomTbl[] = {
+    12,
+    kPacketMaskRandom1,  // 2 byte entries.
+    kPacketMaskRandom2,
+    kPacketMaskRandom3,
+    kPacketMaskRandom4,
+    kPacketMaskRandom5,
+    kPacketMaskRandom6,
+    kPacketMaskRandom7,
+    kPacketMaskRandom8,
+    kPacketMaskRandom9,
+    kPacketMaskRandom10,
+    kPacketMaskRandom11,
+    kPacketMaskRandom12,
+};
+
+}  // namespace fec_private_tables
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_random.h b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
new file mode 100644
index 0000000000..cc7b92984c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
+#define MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
+
+// This file contains a set of packets masks for the FEC code. The masks in
+// this table are specifically designed to favor recovery to random loss.
+// These packet masks are defined to protect up to maximum of 48 media packets.
+
+#include <stdint.h>
+
+namespace webrtc {
+namespace fec_private_tables {
+
+extern const uint8_t kPacketMaskRandomTbl[];
+
+}  // namespace fec_private_tables
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_RANDOM_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/fec_test_helper.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_test_helper.cc
new file mode 100644
index 0000000000..23e66c23bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_test_helper.cc
@@ -0,0 +1,230 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/fec_test_helper.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+namespace fec {
+
+namespace {
+
+constexpr uint8_t kRtpMarkerBitMask = 0x80;
+
+constexpr uint8_t kFecPayloadType = 96;
+constexpr uint8_t kRedPayloadType = 97;
+constexpr uint8_t kVp8PayloadType = 120;
+
+constexpr int kPacketTimestampIncrement = 3000;
+}  // namespace
+
+MediaPacketGenerator::MediaPacketGenerator(uint32_t min_packet_size,
+                                           uint32_t max_packet_size,
+                                           uint32_t ssrc,
+                                           Random* random)
+    : min_packet_size_(min_packet_size),
+      max_packet_size_(max_packet_size),
+      ssrc_(ssrc),
+      random_(random) {}
+
+MediaPacketGenerator::~MediaPacketGenerator() = default;
+
+ForwardErrorCorrection::PacketList MediaPacketGenerator::ConstructMediaPackets(
+    int num_media_packets,
+    uint16_t start_seq_num) {
+  RTC_DCHECK_GT(num_media_packets, 0);
+  uint16_t seq_num = start_seq_num;
+  int time_stamp = random_->Rand<int>();
+
+  ForwardErrorCorrection::PacketList media_packets;
+
+  for (int i = 0; i < num_media_packets; ++i) {
+    std::unique_ptr<ForwardErrorCorrection::Packet> media_packet(
+        new ForwardErrorCorrection::Packet());
+    media_packet->data.SetSize(
+        random_->Rand(min_packet_size_, max_packet_size_));
+
+    uint8_t* data = media_packet->data.MutableData();
+    // Generate random values for the first 2 bytes
+    data[0] = random_->Rand<uint8_t>();
+    data[1] = random_->Rand<uint8_t>();
+
+    // The first two bits are assumed to be 10 by the FEC encoder.
+    // In fact the FEC decoder will set the two first bits to 10 regardless of
+    // what they actually were. Set the first two bits to 10 so that a memcmp
+    // can be performed for the whole restored packet.
+    data[0] |= 0x80;
+    data[0] &= 0xbf;
+
+    // FEC is applied to a whole frame.
+    // A frame is signaled by multiple packets without the marker bit set
+    // followed by the last packet of the frame for which the marker bit is set.
+    // Only push one (fake) frame to the FEC.
+    data[1] &= 0x7f;
+
+    webrtc::ByteWriter<uint16_t>::WriteBigEndian(&data[2], seq_num);
+    webrtc::ByteWriter<uint32_t>::WriteBigEndian(&data[4], time_stamp);
+    webrtc::ByteWriter<uint32_t>::WriteBigEndian(&data[8], ssrc_);
+
+    // Generate random values for payload.
+    for (size_t j = 12; j < media_packet->data.size(); ++j)
+      data[j] = random_->Rand<uint8_t>();
+    seq_num++;
+    media_packets.push_back(std::move(media_packet));
+  }
+  // Last packet, set marker bit.
+  ForwardErrorCorrection::Packet* media_packet = media_packets.back().get();
+  RTC_DCHECK(media_packet);
+  media_packet->data.MutableData()[1] |= 0x80;
+
+  next_seq_num_ = seq_num;
+
+  return media_packets;
+}
+
+ForwardErrorCorrection::PacketList MediaPacketGenerator::ConstructMediaPackets(
+    int num_media_packets) {
+  return ConstructMediaPackets(num_media_packets, random_->Rand<uint16_t>());
+}
+
+uint16_t MediaPacketGenerator::GetNextSeqNum() {
+  return next_seq_num_;
+}
+
+AugmentedPacketGenerator::AugmentedPacketGenerator(uint32_t ssrc)
+    : num_packets_(0), ssrc_(ssrc), seq_num_(0), timestamp_(0) {}
+
+void AugmentedPacketGenerator::NewFrame(size_t num_packets) {
+  num_packets_ = num_packets;
+  timestamp_ += kPacketTimestampIncrement;
+}
+
+uint16_t AugmentedPacketGenerator::NextPacketSeqNum() {
+  return ++seq_num_;
+}
+
+std::unique_ptr<AugmentedPacket> AugmentedPacketGenerator::NextPacket(
+    size_t offset,
+    size_t length) {
+  std::unique_ptr<AugmentedPacket> packet(new AugmentedPacket());
+
+  packet->data.SetSize(length + kRtpHeaderSize);
+  uint8_t* data = packet->data.MutableData();
+  for (size_t i = 0; i < length; ++i)
+    data[i + kRtpHeaderSize] = offset + i;
+  packet->data.SetSize(length + kRtpHeaderSize);
+  packet->header.headerLength = kRtpHeaderSize;
+  packet->header.markerBit = (num_packets_ == 1);
+  packet->header.payloadType = kVp8PayloadType;
+  packet->header.sequenceNumber = seq_num_;
+  packet->header.timestamp = timestamp_;
+  packet->header.ssrc = ssrc_;
+  WriteRtpHeader(packet->header, data);
+  ++seq_num_;
+  --num_packets_;
+
+  return packet;
+}
+
+void AugmentedPacketGenerator::WriteRtpHeader(const RTPHeader& header,
+                                              uint8_t* data) {
+  data[0] = 0x80;  // Version 2.
+  data[1] = header.payloadType;
+  data[1] |= (header.markerBit ? kRtpMarkerBitMask : 0);
+  ByteWriter<uint16_t>::WriteBigEndian(data + 2, header.sequenceNumber);
+  ByteWriter<uint32_t>::WriteBigEndian(data + 4, header.timestamp);
+  ByteWriter<uint32_t>::WriteBigEndian(data + 8, header.ssrc);
+}
+
+FlexfecPacketGenerator::FlexfecPacketGenerator(uint32_t media_ssrc,
+                                               uint32_t flexfec_ssrc)
+    : AugmentedPacketGenerator(media_ssrc),
+      flexfec_ssrc_(flexfec_ssrc),
+      flexfec_seq_num_(0),
+      flexfec_timestamp_(0) {}
+
+std::unique_ptr<AugmentedPacket> FlexfecPacketGenerator::BuildFlexfecPacket(
+    const ForwardErrorCorrection::Packet& packet) {
+  RTC_DCHECK_LE(packet.data.size(),
+                static_cast<size_t>(IP_PACKET_SIZE - kRtpHeaderSize));
+
+  RTPHeader header;
+  header.sequenceNumber = flexfec_seq_num_;
+  ++flexfec_seq_num_;
+  header.timestamp = flexfec_timestamp_;
+  flexfec_timestamp_ += kPacketTimestampIncrement;
+  header.ssrc = flexfec_ssrc_;
+
+  std::unique_ptr<AugmentedPacket> packet_with_rtp_header(
+      new AugmentedPacket());
+  packet_with_rtp_header->data.SetSize(kRtpHeaderSize + packet.data.size());
+  WriteRtpHeader(header, packet_with_rtp_header->data.MutableData());
+  memcpy(packet_with_rtp_header->data.MutableData() + kRtpHeaderSize,
+         packet.data.cdata(), packet.data.size());
+
+  return packet_with_rtp_header;
+}
+
+UlpfecPacketGenerator::UlpfecPacketGenerator(uint32_t ssrc)
+    : AugmentedPacketGenerator(ssrc) {}
+
+RtpPacketReceived UlpfecPacketGenerator::BuildMediaRedPacket(
+    const AugmentedPacket& packet,
+    bool is_recovered) {
+  // Create a temporary buffer used to wrap the media packet in RED.
+  rtc::CopyOnWriteBuffer red_buffer;
+  const size_t kHeaderLength = packet.header.headerLength;
+  // Append header.
+  red_buffer.SetData(packet.data.data(), kHeaderLength);
+  // Find payload type and add it as RED header.
+  uint8_t media_payload_type = red_buffer[1] & 0x7F;
+  red_buffer.AppendData({media_payload_type});
+  // Append rest of payload/padding.
+  red_buffer.AppendData(
+      packet.data.Slice(kHeaderLength, packet.data.size() - kHeaderLength));
+
+  RtpPacketReceived red_packet;
+  RTC_CHECK(red_packet.Parse(std::move(red_buffer)));
+  red_packet.SetPayloadType(kRedPayloadType);
+  red_packet.set_recovered(is_recovered);
+
+  return red_packet;
+}
+
+RtpPacketReceived UlpfecPacketGenerator::BuildUlpfecRedPacket(
+    const ForwardErrorCorrection::Packet& packet) {
+  // Create a fake media packet to get a correct header. 1 byte RED header.
+  ++num_packets_;
+  std::unique_ptr<AugmentedPacket> fake_packet =
+      NextPacket(0, packet.data.size() + 1);
+
+  RtpPacketReceived red_packet;
+  red_packet.Parse(fake_packet->data);
+  red_packet.SetMarker(false);
+  uint8_t* rtp_payload = red_packet.AllocatePayload(packet.data.size() + 1);
+  rtp_payload[0] = kFecPayloadType;
+  red_packet.SetPayloadType(kRedPayloadType);
+  memcpy(rtp_payload + 1, packet.data.cdata(), packet.data.size());
+  red_packet.set_recovered(false);
+
+  return red_packet;
+}
+
+}  // namespace fec
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/fec_test_helper.h b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_test_helper.h
new file mode 100644
index 0000000000..92e09fd44f
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/fec_test_helper.h
@@ -0,0 +1,125 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
+#define MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
+
+#include <memory>
+
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+namespace test {
+namespace fec {
+
+struct AugmentedPacket : public ForwardErrorCorrection::Packet {
+  RTPHeader header;
+};
+
+// TODO(brandtr): Consider merging MediaPacketGenerator and
+// AugmentedPacketGenerator into a single class, since their functionality is
+// similar.
+
+// This class generates media packets corresponding to a single frame.
+class MediaPacketGenerator {
+ public:
+  MediaPacketGenerator(uint32_t min_packet_size,
+                       uint32_t max_packet_size,
+                       uint32_t ssrc,
+                       Random* random);
+  ~MediaPacketGenerator();
+
+  // Construct the media packets, up to `num_media_packets` packets.
+  ForwardErrorCorrection::PacketList ConstructMediaPackets(
+      int num_media_packets,
+      uint16_t start_seq_num);
+  ForwardErrorCorrection::PacketList ConstructMediaPackets(
+      int num_media_packets);
+
+  uint16_t GetNextSeqNum();
+
+ private:
+  uint32_t min_packet_size_;
+  uint32_t max_packet_size_;
+  uint32_t ssrc_;
+  Random* random_;
+
+  ForwardErrorCorrection::PacketList media_packets_;
+  uint16_t next_seq_num_;
+};
+
+// This class generates media packets with a certain structure of the payload.
+class AugmentedPacketGenerator {
+ public:
+  explicit AugmentedPacketGenerator(uint32_t ssrc);
+
+  // Prepare for generating a new set of packets, corresponding to a frame.
+  void NewFrame(size_t num_packets);
+
+  // Increment and return the newly incremented sequence number.
+  uint16_t NextPacketSeqNum();
+
+  // Return the next packet in the current frame.
+  std::unique_ptr<AugmentedPacket> NextPacket(size_t offset, size_t length);
+
+ protected:
+  // Given `header`, writes the appropriate RTP header fields in `data`.
+  static void WriteRtpHeader(const RTPHeader& header, uint8_t* data);
+
+  // Number of packets left to generate, in the current frame.
+  size_t num_packets_;
+
+ private:
+  uint32_t ssrc_;
+  uint16_t seq_num_;
+  uint32_t timestamp_;
+};
+
+// This class generates media and FlexFEC packets for a single frame.
+class FlexfecPacketGenerator : public AugmentedPacketGenerator {
+ public:
+  FlexfecPacketGenerator(uint32_t media_ssrc, uint32_t flexfec_ssrc);
+
+  // Creates a new AugmentedPacket (with RTP headers) from a
+  // FlexFEC packet (without RTP headers).
+  std::unique_ptr<AugmentedPacket> BuildFlexfecPacket(
+      const ForwardErrorCorrection::Packet& packet);
+
+ private:
+  uint32_t flexfec_ssrc_;
+  uint16_t flexfec_seq_num_;
+  uint32_t flexfec_timestamp_;
+};
+
+// This class generates media and ULPFEC packets (both encapsulated in RED)
+// for a single frame.
+class UlpfecPacketGenerator : public AugmentedPacketGenerator {
+ public:
+  explicit UlpfecPacketGenerator(uint32_t ssrc);
+
+  // Creates a new RtpPacket with the RED header added to the packet.
+  static RtpPacketReceived BuildMediaRedPacket(const AugmentedPacket& packet,
+                                               bool is_recovered);
+
+  // Creates a new RtpPacket with FEC payload and RED header. Does this by
+  // creating a new fake media AugmentedPacket, clears the marker bit and adds a
+  // RED header. Finally replaces the payload with the content of
+  // `packet->data`.
+  RtpPacketReceived BuildUlpfecRedPacket(
+      const ForwardErrorCorrection::Packet& packet);
+};
+
+}  // namespace fec
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
new file mode 100644
index 0000000000..2fbb3caae8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
@@ -0,0 +1,320 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
+
+#include <string.h>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// Maximum number of media packets that can be protected in one batch.
+constexpr size_t kMaxMediaPackets = 48;  // Since we are reusing ULPFEC masks.
+
+// Maximum number of media packets tracked by FEC decoder.
+// Maintain a sufficiently larger tracking window than `kMaxMediaPackets`
+// to account for packet reordering in pacer/ network.
+constexpr size_t kMaxTrackedMediaPackets = 4 * kMaxMediaPackets;
+
+// Maximum number of FEC packets stored inside ForwardErrorCorrection.
+constexpr size_t kMaxFecPackets = kMaxMediaPackets;
+
+// Size (in bytes) of packet masks, given number of K bits set.
+constexpr size_t kFlexfecPacketMaskSizes[] = {2, 6, 14};
+
+// Size (in bytes) of part of header which is not packet mask specific.
+constexpr size_t kBaseHeaderSize = 12;
+
+// Size (in bytes) of part of header which is stream specific.
+constexpr size_t kStreamSpecificHeaderSize = 6;
+
+// Size (in bytes) of header, given the single stream packet mask size, i.e.
+// the number of K-bits set.
+constexpr size_t kHeaderSizes[] = {
+    kBaseHeaderSize + kStreamSpecificHeaderSize + kFlexfecPacketMaskSizes[0],
+    kBaseHeaderSize + kStreamSpecificHeaderSize + kFlexfecPacketMaskSizes[1],
+    kBaseHeaderSize + kStreamSpecificHeaderSize + kFlexfecPacketMaskSizes[2]};
+
+// We currently only support single-stream protection.
+// TODO(brandtr): Update this when we support multistream protection.
+constexpr uint8_t kSsrcCount = 1;
+
+// There are three reserved bytes that MUST be set to zero in the header.
+constexpr uint32_t kReservedBits = 0;
+
+// TODO(brandtr): Update this when we support multistream protection.
+constexpr size_t kPacketMaskOffset =
+    kBaseHeaderSize + kStreamSpecificHeaderSize;
+
+// Here we count the K-bits as belonging to the packet mask.
+// This can be used in conjunction with FlexfecHeaderWriter::MinPacketMaskSize,
+// which calculates a bound on the needed packet mask size including K-bits,
+// given a packet mask without K-bits.
+size_t FlexfecHeaderSize(size_t packet_mask_size) {
+  RTC_DCHECK_LE(packet_mask_size, kFlexfecPacketMaskSizes[2]);
+  if (packet_mask_size <= kFlexfecPacketMaskSizes[0]) {
+    return kHeaderSizes[0];
+  } else if (packet_mask_size <= kFlexfecPacketMaskSizes[1]) {
+    return kHeaderSizes[1];
+  }
+  return kHeaderSizes[2];
+}
+
+}  // namespace
+
+FlexfecHeaderReader::FlexfecHeaderReader()
+    : FecHeaderReader(kMaxTrackedMediaPackets, kMaxFecPackets) {}
+
+FlexfecHeaderReader::~FlexfecHeaderReader() = default;
+
+// TODO(brandtr): Update this function when we support flexible masks,
+// retransmissions, and/or several protected SSRCs.
+bool FlexfecHeaderReader::ReadFecHeader(
+    ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const {
+  if (fec_packet->pkt->data.size() <=
+      kBaseHeaderSize + kStreamSpecificHeaderSize) {
+    RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC packet.";
+    return false;
+  }
+  uint8_t* const data = fec_packet->pkt->data.MutableData();
+  bool r_bit = (data[0] & 0x80) != 0;
+  if (r_bit) {
+    RTC_LOG(LS_INFO)
+        << "FlexFEC packet with retransmission bit set. We do not yet "
+           "support this, thus discarding the packet.";
+    return false;
+  }
+  bool f_bit = (data[0] & 0x40) != 0;
+  if (f_bit) {
+    RTC_LOG(LS_INFO)
+        << "FlexFEC packet with inflexible generator matrix. We do "
+           "not yet support this, thus discarding packet.";
+    return false;
+  }
+  uint8_t ssrc_count = ByteReader<uint8_t>::ReadBigEndian(&data[8]);
+  if (ssrc_count != 1) {
+    RTC_LOG(LS_INFO)
+        << "FlexFEC packet protecting multiple media SSRCs. We do not "
+           "yet support this, thus discarding packet.";
+    return false;
+  }
+  uint32_t protected_ssrc = ByteReader<uint32_t>::ReadBigEndian(&data[12]);
+  uint16_t seq_num_base = ByteReader<uint16_t>::ReadBigEndian(&data[16]);
+
+  // Parse the FlexFEC packet mask and remove the interleaved K-bits.
+  // (See FEC header schematic in flexfec_header_reader_writer.h.)
+  // We store the packed packet mask in-band, which "destroys" the standards
+  // compliance of the header. That is fine though, since the code that
+  // reads from the header (from this point and onwards) is aware of this.
+  // TODO(brandtr): When the FEC packet classes have been refactored, store
+  // the packed packet masks out-of-band, thus leaving the FlexFEC header as is.
+  //
+  // We treat the mask parts as unsigned integers with host order endianness
+  // in order to simplify the bit shifting between bytes.
+  if (fec_packet->pkt->data.size() < kHeaderSizes[0]) {
+    RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC packet.";
+    return false;
+  }
+  uint8_t* const packet_mask = data + kPacketMaskOffset;
+  bool k_bit0 = (packet_mask[0] & 0x80) != 0;
+  uint16_t mask_part0 = ByteReader<uint16_t>::ReadBigEndian(&packet_mask[0]);
+  // Shift away K-bit 0, implicitly clearing the last bit.
+  mask_part0 <<= 1;
+  ByteWriter<uint16_t>::WriteBigEndian(&packet_mask[0], mask_part0);
+  size_t packet_mask_size;
+  if (k_bit0) {
+    // The first K-bit is set, and the packet mask is thus only 2 bytes long.
+    // We have now read the entire FEC header, and the rest of the packet
+    // is payload.
+    packet_mask_size = kFlexfecPacketMaskSizes[0];
+  } else {
+    if (fec_packet->pkt->data.size() < kHeaderSizes[1]) {
+      return false;
+    }
+    bool k_bit1 = (packet_mask[2] & 0x80) != 0;
+    // We have already shifted the first two bytes of the packet mask one step
+    // to the left, thus removing K-bit 0. We will now shift the next four bytes
+    // of the packet mask two steps to the left. (One step for the removed
+    // K-bit 0, and one step for the to be removed K-bit 1).
+    uint8_t bit15 = (packet_mask[2] >> 6) & 0x01;
+    packet_mask[1] |= bit15;
+    uint32_t mask_part1 = ByteReader<uint32_t>::ReadBigEndian(&packet_mask[2]);
+    // Shift away K-bit 1 and bit 15, implicitly clearing the last two bits.
+    mask_part1 <<= 2;
+    ByteWriter<uint32_t>::WriteBigEndian(&packet_mask[2], mask_part1);
+    if (k_bit1) {
+      // The first K-bit is clear, but the second K-bit is set. The packet
+      // mask is thus 6 bytes long.  We have now read the entire FEC header,
+      // and the rest of the packet is payload.
+      packet_mask_size = kFlexfecPacketMaskSizes[1];
+    } else {
+      if (fec_packet->pkt->data.size() < kHeaderSizes[2]) {
+        RTC_LOG(LS_WARNING) << "Discarding truncated FlexFEC packet.";
+        return false;
+      }
+      bool k_bit2 = (packet_mask[6] & 0x80) != 0;
+      if (k_bit2) {
+        // The first and second K-bits are clear, but the third K-bit is set.
+        // The packet mask is thus 14 bytes long. We have now read the entire
+        // FEC header, and the rest of the packet is payload.
+        packet_mask_size = kFlexfecPacketMaskSizes[2];
+      } else {
+        RTC_LOG(LS_WARNING)
+            << "Discarding FlexFEC packet with malformed header.";
+        return false;
+      }
+      // At this point, K-bits 0 and 1 have been removed, and the front-most
+      // part of the FlexFEC packet mask has been packed accordingly. We will
+      // now shift the remaning part of the packet mask three steps to the left.
+      // This corresponds to the (in total) three K-bits, which have been
+      // removed.
+      uint8_t tail_bits = (packet_mask[6] >> 5) & 0x03;
+      packet_mask[5] |= tail_bits;
+      uint64_t mask_part2 =
+          ByteReader<uint64_t>::ReadBigEndian(&packet_mask[6]);
+      // Shift away K-bit 2, bit 46, and bit 47, implicitly clearing the last
+      // three bits.
+      mask_part2 <<= 3;
+      ByteWriter<uint64_t>::WriteBigEndian(&packet_mask[6], mask_part2);
+    }
+  }
+
+  // Store "ULPFECized" packet mask info.
+  fec_packet->fec_header_size = FlexfecHeaderSize(packet_mask_size);
+  fec_packet->protected_ssrc = protected_ssrc;
+  fec_packet->seq_num_base = seq_num_base;
+  fec_packet->packet_mask_offset = kPacketMaskOffset;
+  fec_packet->packet_mask_size = packet_mask_size;
+
+  // In FlexFEC, all media packets are protected in their entirety.
+  fec_packet->protection_length =
+      fec_packet->pkt->data.size() - fec_packet->fec_header_size;
+
+  return true;
+}
+
+FlexfecHeaderWriter::FlexfecHeaderWriter()
+    : FecHeaderWriter(kMaxMediaPackets, kMaxFecPackets, kHeaderSizes[2]) {}
+
+FlexfecHeaderWriter::~FlexfecHeaderWriter() = default;
+
+size_t FlexfecHeaderWriter::MinPacketMaskSize(const uint8_t* packet_mask,
+                                              size_t packet_mask_size) const {
+  if (packet_mask_size == kUlpfecPacketMaskSizeLBitClear &&
+      (packet_mask[1] & 0x01) == 0) {
+    // Packet mask is 16 bits long, with bit 15 clear.
+    // It can be used as is.
+    return kFlexfecPacketMaskSizes[0];
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitClear) {
+    // Packet mask is 16 bits long, with bit 15 set.
+    // We must expand the packet mask with zeros in the FlexFEC header.
+    return kFlexfecPacketMaskSizes[1];
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitSet &&
+             (packet_mask[5] & 0x03) == 0) {
+    // Packet mask is 48 bits long, with bits 46 and 47 clear.
+    // It can be used as is.
+    return kFlexfecPacketMaskSizes[1];
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitSet) {
+    // Packet mask is 48 bits long, with at least one of bits 46 and 47 set.
+    // We must expand it with zeros.
+    return kFlexfecPacketMaskSizes[2];
+  }
+  RTC_DCHECK_NOTREACHED() << "Incorrect packet mask size: " << packet_mask_size
+                          << ".";
+  return kFlexfecPacketMaskSizes[2];
+}
+
+size_t FlexfecHeaderWriter::FecHeaderSize(size_t packet_mask_size) const {
+  return FlexfecHeaderSize(packet_mask_size);
+}
+
+// This function adapts the precomputed ULPFEC packet masks to the
+// FlexFEC header standard. Note that the header size is computed by
+// FecHeaderSize(), so in this function we can be sure that we are
+// writing in space that is intended for the header.
+//
+// TODO(brandtr): Update this function when we support offset-based masks,
+// retransmissions, and protecting multiple SSRCs.
+void FlexfecHeaderWriter::FinalizeFecHeader(
+    uint32_t media_ssrc,
+    uint16_t seq_num_base,
+    const uint8_t* packet_mask,
+    size_t packet_mask_size,
+    ForwardErrorCorrection::Packet* fec_packet) const {
+  uint8_t* data = fec_packet->data.MutableData();
+  data[0] &= 0x7f;  // Clear R bit.
+  data[0] &= 0xbf;  // Clear F bit.
+  ByteWriter<uint8_t>::WriteBigEndian(&data[8], kSsrcCount);
+  ByteWriter<uint32_t, 3>::WriteBigEndian(&data[9], kReservedBits);
+  ByteWriter<uint32_t>::WriteBigEndian(&data[12], media_ssrc);
+  ByteWriter<uint16_t>::WriteBigEndian(&data[16], seq_num_base);
+  // Adapt ULPFEC packet mask to FlexFEC header.
+  //
+  // We treat the mask parts as unsigned integers with host order endianness
+  // in order to simplify the bit shifting between bytes.
+  uint8_t* const written_packet_mask = data + kPacketMaskOffset;
+  if (packet_mask_size == kUlpfecPacketMaskSizeLBitSet) {
+    // The packet mask is 48 bits long.
+    uint16_t tmp_mask_part0 =
+        ByteReader<uint16_t>::ReadBigEndian(&packet_mask[0]);
+    uint32_t tmp_mask_part1 =
+        ByteReader<uint32_t>::ReadBigEndian(&packet_mask[2]);
+
+    tmp_mask_part0 >>= 1;  // Shift, thus clearing K-bit 0.
+    ByteWriter<uint16_t>::WriteBigEndian(&written_packet_mask[0],
+                                         tmp_mask_part0);
+    tmp_mask_part1 >>= 2;  // Shift, thus clearing K-bit 1 and bit 15.
+    ByteWriter<uint32_t>::WriteBigEndian(&written_packet_mask[2],
+                                         tmp_mask_part1);
+    bool bit15 = (packet_mask[1] & 0x01) != 0;
+    if (bit15)
+      written_packet_mask[2] |= 0x40;  // Set bit 15.
+    bool bit46 = (packet_mask[5] & 0x02) != 0;
+    bool bit47 = (packet_mask[5] & 0x01) != 0;
+    if (!bit46 && !bit47) {
+      written_packet_mask[2] |= 0x80;  // Set K-bit 1.
+    } else {
+      memset(&written_packet_mask[6], 0, 8);  // Clear all trailing bits.
+      written_packet_mask[6] |= 0x80;         // Set K-bit 2.
+      if (bit46)
+        written_packet_mask[6] |= 0x40;  // Set bit 46.
+      if (bit47)
+        written_packet_mask[6] |= 0x20;  // Set bit 47.
+    }
+  } else if (packet_mask_size == kUlpfecPacketMaskSizeLBitClear) {
+    // The packet mask is 16 bits long.
+    uint16_t tmp_mask_part0 =
+        ByteReader<uint16_t>::ReadBigEndian(&packet_mask[0]);
+
+    tmp_mask_part0 >>= 1;  // Shift, thus clearing K-bit 0.
+    ByteWriter<uint16_t>::WriteBigEndian(&written_packet_mask[0],
+                                         tmp_mask_part0);
+    bool bit15 = (packet_mask[1] & 0x01) != 0;
+    if (!bit15) {
+      written_packet_mask[0] |= 0x80;  // Set K-bit 0.
+    } else {
+      memset(&written_packet_mask[2], 0U, 4);  // Clear all trailing bits.
+      written_packet_mask[2] |= 0x80;          // Set K-bit 1.
+      written_packet_mask[2] |= 0x40;          // Set bit 15.
+    }
+  } else {
+    RTC_DCHECK_NOTREACHED()
+        << "Incorrect packet mask size: " << packet_mask_size << ".";
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
new file mode 100644
index 0000000000..d305c4c288
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_FLEXFEC_HEADER_READER_WRITER_H_
+#define MODULES_RTP_RTCP_SOURCE_FLEXFEC_HEADER_READER_WRITER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+
+namespace webrtc {
+
+// FlexFEC header, minimum 20 bytes.
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |R|F|P|X|  CC   |M| PT recovery |        length recovery        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                          TS recovery                          |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |   SSRCCount   |                    reserved                   |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// 12 |                             SSRC_i                            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |           SN base_i           |k|          Mask [0-14]        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 |k|                   Mask [15-45] (optional)                   |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 24 |k|                                                             |
+//    +-+                   Mask [46-108] (optional)                  |
+// 28 |                                                               |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//    :                     ... next in SSRC_i ...                    :
+//
+//
+// FlexFEC header in 'inflexible' mode (F = 1), 20 bytes.
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |0|1|P|X|  CC   |M| PT recovery |        length recovery        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                          TS recovery                          |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |   SSRCCount   |                    reserved                   |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |                             SSRC_i                            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |           SN base_i           |  M (columns)  |    N (rows)   |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+class FlexfecHeaderReader : public FecHeaderReader {
+ public:
+  FlexfecHeaderReader();
+  ~FlexfecHeaderReader() override;
+
+  bool ReadFecHeader(
+      ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const override;
+};
+
+class FlexfecHeaderWriter : public FecHeaderWriter {
+ public:
+  FlexfecHeaderWriter();
+  ~FlexfecHeaderWriter() override;
+
+  size_t MinPacketMaskSize(const uint8_t* packet_mask,
+                           size_t packet_mask_size) const override;
+
+  size_t FecHeaderSize(size_t packet_mask_row_size) const override;
+
+  void FinalizeFecHeader(
+      uint32_t media_ssrc,
+      uint16_t seq_num_base,
+      const uint8_t* packet_mask,
+      size_t packet_mask_size,
+      ForwardErrorCorrection::Packet* fec_packet) const override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FLEXFEC_HEADER_READER_WRITER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc
new file mode 100644
index 0000000000..4a24e90ec3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc
@@ -0,0 +1,560 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
+
+#include <string.h>
+
+#include <memory>
+#include <utility>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/random.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using Packet = ForwardErrorCorrection::Packet;
+using ReceivedFecPacket = ForwardErrorCorrection::ReceivedFecPacket;
+
+// General. Assume single-stream protection.
+constexpr uint32_t kMediaSsrc = 1254983;
+constexpr uint16_t kMediaStartSeqNum = 825;
+constexpr size_t kMediaPacketLength = 1234;
+constexpr uint32_t kFlexfecSsrc = 52142;
+
+constexpr size_t kFlexfecHeaderSizes[] = {20, 24, 32};
+constexpr size_t kFlexfecPacketMaskOffset = 18;
+constexpr size_t kFlexfecPacketMaskSizes[] = {2, 6, 14};
+constexpr size_t kFlexfecMaxPacketSize = kFlexfecPacketMaskSizes[2];
+
+// Reader tests.
+constexpr uint8_t kNoRBit = 0 << 7;
+constexpr uint8_t kNoFBit = 0 << 6;
+constexpr uint8_t kPtRecovery = 123;
+constexpr uint8_t kLengthRecov[] = {0xab, 0xcd};
+constexpr uint8_t kTsRecovery[] = {0x01, 0x23, 0x45, 0x67};
+constexpr uint8_t kSsrcCount = 1;
+constexpr uint8_t kReservedBits = 0x00;
+constexpr uint8_t kProtSsrc[] = {0x11, 0x22, 0x33, 0x44};
+constexpr uint8_t kSnBase[] = {0xaa, 0xbb};
+constexpr uint8_t kPayloadBits = 0x00;
+
+std::unique_ptr<uint8_t[]> GeneratePacketMask(size_t packet_mask_size,
+                                              uint64_t seed) {
+  Random random(seed);
+  std::unique_ptr<uint8_t[]> packet_mask(new uint8_t[kFlexfecMaxPacketSize]);
+  memset(packet_mask.get(), 0, kFlexfecMaxPacketSize);
+  for (size_t i = 0; i < packet_mask_size; ++i) {
+    packet_mask[i] = random.Rand<uint8_t>();
+  }
+  return packet_mask;
+}
+
+void ClearBit(size_t index, uint8_t* packet_mask) {
+  packet_mask[index / 8] &= ~(1 << (7 - index % 8));
+}
+
+void SetBit(size_t index, uint8_t* packet_mask) {
+  packet_mask[index / 8] |= (1 << (7 - index % 8));
+}
+
+rtc::scoped_refptr<Packet> WriteHeader(const uint8_t* packet_mask,
+                                       size_t packet_mask_size) {
+  FlexfecHeaderWriter writer;
+  rtc::scoped_refptr<Packet> written_packet(new Packet());
+  written_packet->data.SetSize(kMediaPacketLength);
+  uint8_t* data = written_packet->data.MutableData();
+  for (size_t i = 0; i < written_packet->data.size(); ++i) {
+    data[i] = i;  // Actual content doesn't matter.
+  }
+  writer.FinalizeFecHeader(kMediaSsrc, kMediaStartSeqNum, packet_mask,
+                           packet_mask_size, written_packet.get());
+  return written_packet;
+}
+
+std::unique_ptr<ReceivedFecPacket> ReadHeader(const Packet& written_packet) {
+  FlexfecHeaderReader reader;
+  std::unique_ptr<ReceivedFecPacket> read_packet(new ReceivedFecPacket());
+  read_packet->ssrc = kFlexfecSsrc;
+  read_packet->pkt = rtc::scoped_refptr<Packet>(new Packet());
+  read_packet->pkt->data = written_packet.data;
+  EXPECT_TRUE(reader.ReadFecHeader(read_packet.get()));
+  return read_packet;
+}
+
+void VerifyReadHeaders(size_t expected_fec_header_size,
+                       const uint8_t* expected_packet_mask,
+                       size_t expected_packet_mask_size,
+                       const ReceivedFecPacket& read_packet) {
+  EXPECT_EQ(expected_fec_header_size, read_packet.fec_header_size);
+  EXPECT_EQ(ByteReader<uint32_t>::ReadBigEndian(kProtSsrc),
+            read_packet.protected_ssrc);
+  EXPECT_EQ(ByteReader<uint16_t>::ReadBigEndian(kSnBase),
+            read_packet.seq_num_base);
+  const size_t packet_mask_offset = read_packet.packet_mask_offset;
+  EXPECT_EQ(kFlexfecPacketMaskOffset, packet_mask_offset);
+  EXPECT_EQ(expected_packet_mask_size, read_packet.packet_mask_size);
+  EXPECT_EQ(read_packet.pkt->data.size() - expected_fec_header_size,
+            read_packet.protection_length);
+  // Ensure that the K-bits are removed and the packet mask has been packed.
+  EXPECT_THAT(
+      ::testing::make_tuple(read_packet.pkt->data.cdata() + packet_mask_offset,
+                            read_packet.packet_mask_size),
+      ::testing::ElementsAreArray(expected_packet_mask,
+                                  expected_packet_mask_size));
+}
+
+void VerifyFinalizedHeaders(const uint8_t* expected_packet_mask,
+                            size_t expected_packet_mask_size,
+                            const Packet& written_packet) {
+  const uint8_t* packet = written_packet.data.cdata();
+  EXPECT_EQ(0x00, packet[0] & 0x80);  // F bit clear.
+  EXPECT_EQ(0x00, packet[0] & 0x40);  // R bit clear.
+  EXPECT_EQ(0x01, packet[8]);         // SSRCCount = 1.
+  EXPECT_EQ(kMediaSsrc, ByteReader<uint32_t>::ReadBigEndian(packet + 12));
+  EXPECT_EQ(kMediaStartSeqNum,
+            ByteReader<uint16_t>::ReadBigEndian(packet + 16));
+  EXPECT_THAT(::testing::make_tuple(packet + kFlexfecPacketMaskOffset,
+                                    expected_packet_mask_size),
+              ::testing::ElementsAreArray(expected_packet_mask,
+                                          expected_packet_mask_size));
+}
+
+void VerifyWrittenAndReadHeaders(size_t expected_fec_header_size,
+                                 const uint8_t* expected_packet_mask,
+                                 size_t expected_packet_mask_size,
+                                 const Packet& written_packet,
+                                 const ReceivedFecPacket& read_packet) {
+  EXPECT_EQ(kFlexfecSsrc, read_packet.ssrc);
+  EXPECT_EQ(expected_fec_header_size, read_packet.fec_header_size);
+  EXPECT_EQ(kMediaSsrc, read_packet.protected_ssrc);
+  EXPECT_EQ(kMediaStartSeqNum, read_packet.seq_num_base);
+  EXPECT_EQ(kFlexfecPacketMaskOffset, read_packet.packet_mask_offset);
+  ASSERT_EQ(expected_packet_mask_size, read_packet.packet_mask_size);
+  EXPECT_EQ(written_packet.data.size() - expected_fec_header_size,
+            read_packet.protection_length);
+  // Verify that the call to ReadFecHeader did normalize the packet masks.
+  EXPECT_THAT(::testing::make_tuple(
+                  read_packet.pkt->data.cdata() + kFlexfecPacketMaskOffset,
+                  read_packet.packet_mask_size),
+              ::testing::ElementsAreArray(expected_packet_mask,
+                                          expected_packet_mask_size));
+  // Verify that the call to ReadFecHeader did not tamper with the payload.
+  EXPECT_THAT(::testing::make_tuple(
+                  read_packet.pkt->data.cdata() + read_packet.fec_header_size,
+                  read_packet.pkt->data.size() - read_packet.fec_header_size),
+              ::testing::ElementsAreArray(
+                  written_packet.data.cdata() + expected_fec_header_size,
+                  written_packet.data.size() - expected_fec_header_size));
+}
+
+}  // namespace
+
+TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit0Set) {
+  constexpr uint8_t kKBit0 = 1 << 7;
+  constexpr size_t kExpectedPacketMaskSize = 2;
+  constexpr size_t kExpectedFecHeaderSize = 20;
+  // clang-format off
+  constexpr uint8_t kFlexfecPktMask[] = {kKBit0 | 0x08, 0x81};
+  constexpr uint8_t kUlpfecPacketMask[] =        {0x11, 0x02};
+  // clang-format on
+  constexpr uint8_t kPacketData[] = {
+      kNoRBit | kNoFBit, kPtRecovery,    kLengthRecov[0],    kLengthRecov[1],
+      kTsRecovery[0],    kTsRecovery[1], kTsRecovery[2],     kTsRecovery[3],
+      kSsrcCount,        kReservedBits,  kReservedBits,      kReservedBits,
+      kProtSsrc[0],      kProtSsrc[1],   kProtSsrc[2],       kProtSsrc[3],
+      kSnBase[0],        kSnBase[1],     kFlexfecPktMask[0], kFlexfecPktMask[1],
+      kPayloadBits,      kPayloadBits,   kPayloadBits,       kPayloadBits};
+  const size_t packet_length = sizeof(kPacketData);
+  ReceivedFecPacket read_packet;
+  read_packet.pkt = rtc::scoped_refptr<Packet>(new Packet());
+  read_packet.pkt->data.SetData(kPacketData, packet_length);
+
+  FlexfecHeaderReader reader;
+  EXPECT_TRUE(reader.ReadFecHeader(&read_packet));
+
+  VerifyReadHeaders(kExpectedFecHeaderSize, kUlpfecPacketMask,
+                    kExpectedPacketMaskSize, read_packet);
+}
+
+TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit1Set) {
+  constexpr uint8_t kKBit0 = 0 << 7;
+  constexpr uint8_t kKBit1 = 1 << 7;
+  constexpr size_t kExpectedPacketMaskSize = 6;
+  constexpr size_t kExpectedFecHeaderSize = 24;
+  // clang-format off
+  constexpr uint8_t kFlxfecPktMsk[] = {kKBit0 | 0x48, 0x81,
+                                       kKBit1 | 0x02, 0x11, 0x00, 0x21};
+  constexpr uint8_t kUlpfecPacketMask[] =      {0x91, 0x02,
+                                                0x08, 0x44, 0x00, 0x84};
+  // clang-format on
+  constexpr uint8_t kPacketData[] = {
+      kNoRBit | kNoFBit, kPtRecovery,      kLengthRecov[0],  kLengthRecov[1],
+      kTsRecovery[0],    kTsRecovery[1],   kTsRecovery[2],   kTsRecovery[3],
+      kSsrcCount,        kReservedBits,    kReservedBits,    kReservedBits,
+      kProtSsrc[0],      kProtSsrc[1],     kProtSsrc[2],     kProtSsrc[3],
+      kSnBase[0],        kSnBase[1],       kFlxfecPktMsk[0], kFlxfecPktMsk[1],
+      kFlxfecPktMsk[2],  kFlxfecPktMsk[3], kFlxfecPktMsk[4], kFlxfecPktMsk[5],
+      kPayloadBits,      kPayloadBits,     kPayloadBits,     kPayloadBits};
+  const size_t packet_length = sizeof(kPacketData);
+  ReceivedFecPacket read_packet;
+  read_packet.pkt = rtc::scoped_refptr<Packet>(new Packet());
+  read_packet.pkt->data.SetData(kPacketData, packet_length);
+
+  FlexfecHeaderReader reader;
+  EXPECT_TRUE(reader.ReadFecHeader(&read_packet));
+
+  VerifyReadHeaders(kExpectedFecHeaderSize, kUlpfecPacketMask,
+                    kExpectedPacketMaskSize, read_packet);
+}
+
+TEST(FlexfecHeaderReaderTest, ReadsHeaderWithKBit2Set) {
+  constexpr uint8_t kKBit0 = 0 << 7;
+  constexpr uint8_t kKBit1 = 0 << 7;
+  constexpr uint8_t kKBit2 = 1 << 7;
+  constexpr size_t kExpectedPacketMaskSize = 14;
+  constexpr size_t kExpectedFecHeaderSize = 32;
+  // clang-format off
+  constexpr uint8_t kFlxfcPktMsk[] = {kKBit0 | 0x48, 0x81,
+                                      kKBit1 | 0x02, 0x11, 0x00, 0x21,
+                                      kKBit2 | 0x01, 0x11, 0x11, 0x11,
+                                               0x11, 0x11, 0x11, 0x11};
+  constexpr uint8_t kUlpfecPacketMask[] =     {0x91, 0x02,
+                                               0x08, 0x44, 0x00, 0x84,
+                                               0x08, 0x88, 0x88, 0x88,
+                                               0x88, 0x88, 0x88, 0x88};
+  // clang-format on
+  constexpr uint8_t kPacketData[] = {
+      kNoRBit | kNoFBit, kPtRecovery,      kLengthRecov[0],  kLengthRecov[1],
+      kTsRecovery[0],    kTsRecovery[1],   kTsRecovery[2],   kTsRecovery[3],
+      kSsrcCount,        kReservedBits,    kReservedBits,    kReservedBits,
+      kProtSsrc[0],      kProtSsrc[1],     kProtSsrc[2],     kProtSsrc[3],
+      kSnBase[0],        kSnBase[1],       kFlxfcPktMsk[0],  kFlxfcPktMsk[1],
+      kFlxfcPktMsk[2],   kFlxfcPktMsk[3],  kFlxfcPktMsk[4],  kFlxfcPktMsk[5],
+      kFlxfcPktMsk[6],   kFlxfcPktMsk[7],  kFlxfcPktMsk[8],  kFlxfcPktMsk[9],
+      kFlxfcPktMsk[10],  kFlxfcPktMsk[11], kFlxfcPktMsk[12], kFlxfcPktMsk[13],
+      kPayloadBits,      kPayloadBits,     kPayloadBits,     kPayloadBits};
+  const size_t packet_length = sizeof(kPacketData);
+  ReceivedFecPacket read_packet;
+  read_packet.pkt = rtc::scoped_refptr<Packet>(new Packet());
+  read_packet.pkt->data.SetData(kPacketData, packet_length);
+
+  FlexfecHeaderReader reader;
+  EXPECT_TRUE(reader.ReadFecHeader(&read_packet));
+
+  VerifyReadHeaders(kExpectedFecHeaderSize, kUlpfecPacketMask,
+                    kExpectedPacketMaskSize, read_packet);
+}
+
+TEST(FlexfecHeaderReaderTest, ReadPacketWithoutStreamSpecificHeaderShouldFail) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+
+  // Simulate short received packet.
+  ReceivedFecPacket read_packet;
+  read_packet.ssrc = kFlexfecSsrc;
+  read_packet.pkt = std::move(written_packet);
+  read_packet.pkt->data.SetSize(12);
+
+  FlexfecHeaderReader reader;
+  EXPECT_FALSE(reader.ReadFecHeader(&read_packet));
+}
+
+TEST(FlexfecHeaderReaderTest, ReadShortPacketWithKBit0SetShouldFail) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+
+  // Simulate short received packet.
+  ReceivedFecPacket read_packet;
+  read_packet.ssrc = kFlexfecSsrc;
+  read_packet.pkt = std::move(written_packet);
+  read_packet.pkt->data.SetSize(18);
+
+  FlexfecHeaderReader reader;
+  EXPECT_FALSE(reader.ReadFecHeader(&read_packet));
+}
+
+TEST(FlexfecHeaderReaderTest, ReadShortPacketWithKBit1SetShouldFail) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  SetBit(15, packet_mask.get());  // This expands the packet mask "once".
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+
+  // Simulate short received packet.
+  ReceivedFecPacket read_packet;
+  read_packet.ssrc = kFlexfecSsrc;
+  read_packet.pkt = std::move(written_packet);
+  read_packet.pkt->data.SetSize(20);
+
+  FlexfecHeaderReader reader;
+  EXPECT_FALSE(reader.ReadFecHeader(&read_packet));
+}
+
+TEST(FlexfecHeaderReaderTest, ReadShortPacketWithKBit2SetShouldFail) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  SetBit(47, packet_mask.get());  // This expands the packet mask "twice".
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+
+  // Simulate short received packet.
+  ReceivedFecPacket read_packet;
+  read_packet.ssrc = kFlexfecSsrc;
+  read_packet.pkt = std::move(written_packet);
+  read_packet.pkt->data.SetSize(24);
+
+  FlexfecHeaderReader reader;
+  EXPECT_FALSE(reader.ReadFecHeader(&read_packet));
+}
+
+TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithKBit0Set) {
+  constexpr size_t kExpectedPacketMaskSize = 2;
+  constexpr uint8_t kFlexfecPacketMask[] = {0x88, 0x81};
+  constexpr uint8_t kUlpfecPacketMask[] = {0x11, 0x02};
+  Packet written_packet;
+  written_packet.data.SetSize(kMediaPacketLength);
+  uint8_t* data = written_packet.data.MutableData();
+  for (size_t i = 0; i < written_packet.data.size(); ++i) {
+    data[i] = i;
+  }
+
+  FlexfecHeaderWriter writer;
+  writer.FinalizeFecHeader(kMediaSsrc, kMediaStartSeqNum, kUlpfecPacketMask,
+                           sizeof(kUlpfecPacketMask), &written_packet);
+
+  VerifyFinalizedHeaders(kFlexfecPacketMask, kExpectedPacketMaskSize,
+                         written_packet);
+}
+
+TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithKBit1Set) {
+  constexpr size_t kExpectedPacketMaskSize = 6;
+  constexpr uint8_t kFlexfecPacketMask[] = {0x48, 0x81, 0x82, 0x11, 0x00, 0x21};
+  constexpr uint8_t kUlpfecPacketMask[] = {0x91, 0x02, 0x08, 0x44, 0x00, 0x84};
+  Packet written_packet;
+  written_packet.data.SetSize(kMediaPacketLength);
+  uint8_t* data = written_packet.data.MutableData();
+  for (size_t i = 0; i < written_packet.data.size(); ++i) {
+    data[i] = i;
+  }
+
+  FlexfecHeaderWriter writer;
+  writer.FinalizeFecHeader(kMediaSsrc, kMediaStartSeqNum, kUlpfecPacketMask,
+                           sizeof(kUlpfecPacketMask), &written_packet);
+
+  VerifyFinalizedHeaders(kFlexfecPacketMask, kExpectedPacketMaskSize,
+                         written_packet);
+}
+
+TEST(FlexfecHeaderWriterTest, FinalizesHeaderWithKBit2Set) {
+  constexpr size_t kExpectedPacketMaskSize = 14;
+  constexpr uint8_t kFlexfecPacketMask[] = {
+      0x11, 0x11,                                     // K-bit 0 clear.
+      0x11, 0x11, 0x11, 0x10,                         // K-bit 1 clear.
+      0xa0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00  // K-bit 2 set.
+  };
+  constexpr uint8_t kUlpfecPacketMask[] = {0x22, 0x22, 0x44, 0x44, 0x44, 0x41};
+  Packet written_packet;
+  written_packet.data.SetSize(kMediaPacketLength);
+  uint8_t* data = written_packet.data.MutableData();
+  for (size_t i = 0; i < written_packet.data.size(); ++i) {
+    data[i] = i;
+  }
+
+  FlexfecHeaderWriter writer;
+  writer.FinalizeFecHeader(kMediaSsrc, kMediaStartSeqNum, kUlpfecPacketMask,
+                           sizeof(kUlpfecPacketMask), &written_packet);
+
+  VerifyFinalizedHeaders(kFlexfecPacketMask, kExpectedPacketMaskSize,
+                         written_packet);
+}
+
+TEST(FlexfecHeaderWriterTest, ContractsShortUlpfecPacketMaskWithBit15Clear) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  ClearBit(15, packet_mask.get());
+
+  FlexfecHeaderWriter writer;
+  size_t min_packet_mask_size =
+      writer.MinPacketMaskSize(packet_mask.get(), packet_mask_size);
+
+  EXPECT_EQ(kFlexfecPacketMaskSizes[0], min_packet_mask_size);
+  EXPECT_EQ(kFlexfecHeaderSizes[0], writer.FecHeaderSize(min_packet_mask_size));
+}
+
+TEST(FlexfecHeaderWriterTest, ExpandsShortUlpfecPacketMaskWithBit15Set) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  SetBit(15, packet_mask.get());
+
+  FlexfecHeaderWriter writer;
+  size_t min_packet_mask_size =
+      writer.MinPacketMaskSize(packet_mask.get(), packet_mask_size);
+
+  EXPECT_EQ(kFlexfecPacketMaskSizes[1], min_packet_mask_size);
+  EXPECT_EQ(kFlexfecHeaderSizes[1], writer.FecHeaderSize(min_packet_mask_size));
+}
+
+TEST(FlexfecHeaderWriterTest,
+     ContractsLongUlpfecPacketMaskWithBit46ClearBit47Clear) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  ClearBit(46, packet_mask.get());
+  ClearBit(47, packet_mask.get());
+
+  FlexfecHeaderWriter writer;
+  size_t min_packet_mask_size =
+      writer.MinPacketMaskSize(packet_mask.get(), packet_mask_size);
+
+  EXPECT_EQ(kFlexfecPacketMaskSizes[1], min_packet_mask_size);
+  EXPECT_EQ(kFlexfecHeaderSizes[1], writer.FecHeaderSize(min_packet_mask_size));
+}
+
+TEST(FlexfecHeaderWriterTest,
+     ExpandsLongUlpfecPacketMaskWithBit46SetBit47Clear) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  SetBit(46, packet_mask.get());
+  ClearBit(47, packet_mask.get());
+
+  FlexfecHeaderWriter writer;
+  size_t min_packet_mask_size =
+      writer.MinPacketMaskSize(packet_mask.get(), packet_mask_size);
+
+  EXPECT_EQ(kFlexfecPacketMaskSizes[2], min_packet_mask_size);
+  EXPECT_EQ(kFlexfecHeaderSizes[2], writer.FecHeaderSize(min_packet_mask_size));
+}
+
+TEST(FlexfecHeaderWriterTest,
+     ExpandsLongUlpfecPacketMaskWithBit46ClearBit47Set) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  ClearBit(46, packet_mask.get());
+  SetBit(47, packet_mask.get());
+
+  FlexfecHeaderWriter writer;
+  size_t min_packet_mask_size =
+      writer.MinPacketMaskSize(packet_mask.get(), packet_mask_size);
+
+  EXPECT_EQ(kFlexfecPacketMaskSizes[2], min_packet_mask_size);
+  EXPECT_EQ(kFlexfecHeaderSizes[2], writer.FecHeaderSize(min_packet_mask_size));
+}
+
+TEST(FlexfecHeaderWriterTest, ExpandsLongUlpfecPacketMaskWithBit46SetBit47Set) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  SetBit(46, packet_mask.get());
+  SetBit(47, packet_mask.get());
+
+  FlexfecHeaderWriter writer;
+  size_t min_packet_mask_size =
+      writer.MinPacketMaskSize(packet_mask.get(), packet_mask_size);
+
+  EXPECT_EQ(kFlexfecPacketMaskSizes[2], min_packet_mask_size);
+  EXPECT_EQ(kFlexfecHeaderSizes[2], writer.FecHeaderSize(min_packet_mask_size));
+}
+
+TEST(FlexfecHeaderReaderWriterTest,
+     WriteAndReadSmallUlpfecPacketHeaderWithMaskBit15Clear) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  ClearBit(15, packet_mask.get());
+
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+  auto read_packet = ReadHeader(*written_packet);
+
+  VerifyWrittenAndReadHeaders(kFlexfecHeaderSizes[0], packet_mask.get(),
+                              kFlexfecPacketMaskSizes[0], *written_packet,
+                              *read_packet);
+}
+
+TEST(FlexfecHeaderReaderWriterTest,
+     WriteAndReadSmallUlpfecPacketHeaderWithMaskBit15Set) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  SetBit(15, packet_mask.get());
+
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+  auto read_packet = ReadHeader(*written_packet);
+
+  VerifyWrittenAndReadHeaders(kFlexfecHeaderSizes[1], packet_mask.get(),
+                              kFlexfecPacketMaskSizes[1], *written_packet,
+                              *read_packet);
+}
+
+TEST(FlexfecHeaderReaderWriterTest,
+     WriteAndReadLargeUlpfecPacketHeaderWithMaskBits46And47Clear) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  ClearBit(46, packet_mask.get());
+  ClearBit(47, packet_mask.get());
+
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+  auto read_packet = ReadHeader(*written_packet);
+
+  VerifyWrittenAndReadHeaders(kFlexfecHeaderSizes[1], packet_mask.get(),
+                              kFlexfecPacketMaskSizes[1], *written_packet,
+                              *read_packet);
+}
+
+TEST(FlexfecHeaderReaderWriterTest,
+     WriteAndReadLargeUlpfecPacketHeaderWithMaskBit46SetBit47Clear) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  SetBit(46, packet_mask.get());
+  ClearBit(47, packet_mask.get());
+
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+  auto read_packet = ReadHeader(*written_packet);
+
+  VerifyWrittenAndReadHeaders(kFlexfecHeaderSizes[2], packet_mask.get(),
+                              kFlexfecPacketMaskSizes[2], *written_packet,
+                              *read_packet);
+}
+
+TEST(FlexfecHeaderReaderWriterTest,
+     WriteAndReadLargeUlpfecPacketHeaderMaskWithBit46ClearBit47Set) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  ClearBit(46, packet_mask.get());
+  SetBit(47, packet_mask.get());
+
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+  auto read_packet = ReadHeader(*written_packet);
+
+  VerifyWrittenAndReadHeaders(kFlexfecHeaderSizes[2], packet_mask.get(),
+                              kFlexfecPacketMaskSizes[2], *written_packet,
+                              *read_packet);
+}
+
+TEST(FlexfecHeaderReaderWriterTest,
+     WriteAndReadLargeUlpfecPacketHeaderWithMaskBits46And47Set) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  SetBit(46, packet_mask.get());
+  SetBit(47, packet_mask.get());
+
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+  auto read_packet = ReadHeader(*written_packet);
+
+  VerifyWrittenAndReadHeaders(kFlexfecHeaderSizes[2], packet_mask.get(),
+                              kFlexfecPacketMaskSizes[2], *written_packet,
+                              *read_packet);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
new file mode 100644
index 0000000000..bd67de1f83
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/flexfec_receiver.h"
+
+#include <string.h>
+
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// Minimum header size (in bytes) of a well-formed non-singular FlexFEC packet.
+constexpr size_t kMinFlexfecHeaderSize = 20;
+
+// How often to log the recovered packets to the text log.
+constexpr int kPacketLogIntervalMs = 10000;
+
+}  // namespace
+
+/* Mozilla: Avoid this since it could use GetRealTimeClock().
+FlexfecReceiver::FlexfecReceiver(
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc,
+    RecoveredPacketReceiver* recovered_packet_receiver)
+    : FlexfecReceiver(Clock::GetRealTimeClock(),
+                      ssrc,
+                      protected_media_ssrc,
+                      recovered_packet_receiver) {}
+ */
+
+FlexfecReceiver::FlexfecReceiver(
+    Clock* clock,
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc,
+    RecoveredPacketReceiver* recovered_packet_receiver)
+    : ssrc_(ssrc),
+      protected_media_ssrc_(protected_media_ssrc),
+      erasure_code_(
+          ForwardErrorCorrection::CreateFlexfec(ssrc, protected_media_ssrc)),
+      recovered_packet_receiver_(recovered_packet_receiver),
+      clock_(clock),
+      last_recovered_packet_ms_(-1) {
+  // It's OK to create this object on a different thread/task queue than
+  // the one used during main operation.
+  sequence_checker_.Detach();
+}
+
+FlexfecReceiver::~FlexfecReceiver() = default;
+
+void FlexfecReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // If this packet was recovered, it might be originating from
+  // ProcessReceivedPacket in this object. To avoid lifetime issues with
+  // `recovered_packets_`, we therefore break the cycle here.
+  // This might reduce decoding efficiency a bit, since we can't disambiguate
+  // recovered packets by RTX from recovered packets by FlexFEC.
+  if (packet.recovered())
+    return;
+
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet =
+      AddReceivedPacket(packet);
+  if (!received_packet)
+    return;
+
+  ProcessReceivedPacket(*received_packet);
+}
+
+FecPacketCounter FlexfecReceiver::GetPacketCounter() const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  return packet_counter_;
+}
+
+// TODO(eladalon): Consider using packet.recovered() to avoid processing
+// recovered packets here.
+std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>
+FlexfecReceiver::AddReceivedPacket(const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // RTP packets with a full base header (12 bytes), but without payload,
+  // could conceivably be useful in the decoding. Therefore we check
+  // with a non-strict inequality here.
+  RTC_DCHECK_GE(packet.size(), kRtpHeaderSize);
+
+  // Demultiplex based on SSRC, and insert into erasure code decoder.
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet(
+      new ForwardErrorCorrection::ReceivedPacket());
+  received_packet->seq_num = packet.SequenceNumber();
+  received_packet->ssrc = packet.Ssrc();
+  received_packet->extensions = packet.extension_manager();
+  if (received_packet->ssrc == ssrc_) {
+    // This is a FlexFEC packet.
+    if (packet.payload_size() < kMinFlexfecHeaderSize) {
+      RTC_LOG(LS_WARNING) << "Truncated FlexFEC packet, discarding.";
+      return nullptr;
+    }
+    received_packet->is_fec = true;
+    ++packet_counter_.num_fec_packets;
+
+    // Insert packet payload into erasure code.
+    received_packet->pkt = rtc::scoped_refptr<ForwardErrorCorrection::Packet>(
+        new ForwardErrorCorrection::Packet());
+    received_packet->pkt->data =
+        packet.Buffer().Slice(packet.headers_size(), packet.payload_size());
+  } else {
+    // This is a media packet, or a FlexFEC packet belonging to some
+    // other FlexFEC stream.
+    if (received_packet->ssrc != protected_media_ssrc_) {
+      return nullptr;
+    }
+    received_packet->is_fec = false;
+
+    // Insert entire packet into erasure code.
+    // Create a copy and fill with zeros all mutable extensions.
+    received_packet->pkt = rtc::scoped_refptr<ForwardErrorCorrection::Packet>(
+        new ForwardErrorCorrection::Packet());
+    RtpPacketReceived packet_copy(packet);
+    packet_copy.ZeroMutableExtensions();
+    received_packet->pkt->data = packet_copy.Buffer();
+  }
+
+  ++packet_counter_.num_packets;
+
+  return received_packet;
+}
+
+// Note that the implementation of this member function and the implementation
+// in UlpfecReceiver::ProcessReceivedFec() are slightly different.
+// This implementation only returns _recovered_ media packets through the
+// callback, whereas the implementation in UlpfecReceiver returns _all inserted_
+// media packets through the callback. The latter behaviour makes sense
+// for ULPFEC, since the ULPFEC receiver is owned by the RtpVideoStreamReceiver.
+// Here, however, the received media pipeline is more decoupled from the
+// FlexFEC decoder, and we therefore do not interfere with the reception
+// of non-recovered media packets.
+void FlexfecReceiver::ProcessReceivedPacket(
+    const ForwardErrorCorrection::ReceivedPacket& received_packet) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // Decode.
+  erasure_code_->DecodeFec(received_packet, &recovered_packets_);
+
+  // Return recovered packets through callback.
+  for (const auto& recovered_packet : recovered_packets_) {
+    RTC_CHECK(recovered_packet);
+    if (recovered_packet->returned) {
+      continue;
+    }
+    ++packet_counter_.num_recovered_packets;
+    // Set this flag first, since OnRecoveredPacket may end up here
+    // again, with the same packet.
+    recovered_packet->returned = true;
+    RTC_CHECK_GE(recovered_packet->pkt->data.size(), kRtpHeaderSize);
+
+    RtpPacketReceived parsed_packet(&received_packet.extensions);
+    if (!parsed_packet.Parse(recovered_packet->pkt->data)) {
+      continue;
+    }
+    parsed_packet.set_recovered(true);
+
+    // TODO(brandtr): Update here when we support protecting audio packets too.
+    parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+    recovered_packet_receiver_->OnRecoveredPacket(parsed_packet);
+
+    // Periodically log the incoming packets at LS_INFO.
+    int64_t now_ms = clock_->TimeInMilliseconds();
+    bool should_log_periodically =
+        now_ms - last_recovered_packet_ms_ > kPacketLogIntervalMs;
+    if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE) || should_log_periodically) {
+      rtc::LoggingSeverity level =
+          should_log_periodically ? rtc::LS_INFO : rtc::LS_VERBOSE;
+      RTC_LOG_V(level) << "Recovered media packet with SSRC: "
+                       << parsed_packet.Ssrc() << " seq "
+                       << parsed_packet.SequenceNumber() << " recovered length "
+                       << recovered_packet->pkt->data.size()
+                       << " from FlexFEC stream with SSRC: " << ssrc_;
+      if (should_log_periodically) {
+        last_recovered_packet_ms_ = now_ms;
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
new file mode 100644
index 0000000000..1243858b6b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
@@ -0,0 +1,691 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/flexfec_receiver.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h"
+#include "modules/rtp_rtcp/source/fec_test_helper.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using ::testing::_;
+using ::testing::Eq;
+using ::testing::Property;
+
+using test::fec::FlexfecPacketGenerator;
+using Packet = ForwardErrorCorrection::Packet;
+using PacketList = ForwardErrorCorrection::PacketList;
+
+constexpr size_t kPayloadLength = 500;
+constexpr uint32_t kFlexfecSsrc = 42984;
+constexpr uint32_t kMediaSsrc = 8353;
+
+RtpPacketReceived ParsePacket(const Packet& packet) {
+  RtpPacketReceived parsed_packet;
+  EXPECT_TRUE(parsed_packet.Parse(packet.data));
+  return parsed_packet;
+}
+
+}  // namespace
+
+class FlexfecReceiverForTest : public FlexfecReceiver {
+ public:
+  FlexfecReceiverForTest(uint32_t ssrc,
+                         uint32_t protected_media_ssrc,
+                         RecoveredPacketReceiver* recovered_packet_receiver)
+      : FlexfecReceiver(Clock::GetRealTimeClock(),
+                        ssrc,
+                        protected_media_ssrc,
+                        recovered_packet_receiver) {}
+  // Expose methods for tests.
+  using FlexfecReceiver::AddReceivedPacket;
+  using FlexfecReceiver::ProcessReceivedPacket;
+};
+
+class FlexfecReceiverTest : public ::testing::Test {
+ protected:
+  FlexfecReceiverTest()
+      : receiver_(kFlexfecSsrc, kMediaSsrc, &recovered_packet_receiver_),
+        erasure_code_(
+            ForwardErrorCorrection::CreateFlexfec(kFlexfecSsrc, kMediaSsrc)),
+        packet_generator_(kMediaSsrc, kFlexfecSsrc) {}
+
+  // Generates `num_media_packets` corresponding to a single frame.
+  void PacketizeFrame(size_t num_media_packets,
+                      size_t frame_offset,
+                      PacketList* media_packets);
+
+  // Generates `num_fec_packets` FEC packets, given `media_packets`.
+  std::list<Packet*> EncodeFec(const PacketList& media_packets,
+                               size_t num_fec_packets);
+
+  FlexfecReceiverForTest receiver_;
+  std::unique_ptr<ForwardErrorCorrection> erasure_code_;
+
+  FlexfecPacketGenerator packet_generator_;
+  ::testing::StrictMock<MockRecoveredPacketReceiver> recovered_packet_receiver_;
+};
+
+void FlexfecReceiverTest::PacketizeFrame(size_t num_media_packets,
+                                         size_t frame_offset,
+                                         PacketList* media_packets) {
+  packet_generator_.NewFrame(num_media_packets);
+  for (size_t i = 0; i < num_media_packets; ++i) {
+    std::unique_ptr<Packet> next_packet(
+        packet_generator_.NextPacket(frame_offset + i, kPayloadLength));
+    media_packets->push_back(std::move(next_packet));
+  }
+}
+
+std::list<Packet*> FlexfecReceiverTest::EncodeFec(
+    const PacketList& media_packets,
+    size_t num_fec_packets) {
+  const uint8_t protection_factor =
+      num_fec_packets * 255 / media_packets.size();
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr FecMaskType kFecMaskType = kFecMaskRandom;
+  std::list<Packet*> fec_packets;
+  EXPECT_EQ(0, erasure_code_->EncodeFec(
+                   media_packets, protection_factor, kNumImportantPackets,
+                   kUseUnequalProtection, kFecMaskType, &fec_packets));
+  EXPECT_EQ(num_fec_packets, fec_packets.size());
+  return fec_packets;
+}
+
+TEST_F(FlexfecReceiverTest, ReceivesMediaPacket) {
+  packet_generator_.NewFrame(1);
+  std::unique_ptr<Packet> media_packet(
+      packet_generator_.NextPacket(0, kPayloadLength));
+
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet =
+      receiver_.AddReceivedPacket(ParsePacket(*media_packet));
+  ASSERT_TRUE(received_packet);
+  receiver_.ProcessReceivedPacket(*received_packet);
+}
+
+TEST_F(FlexfecReceiverTest, ReceivesMediaAndFecPackets) {
+  const size_t kNumMediaPackets = 1;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+  const auto& media_packet = media_packets.front();
+  auto fec_packet = packet_generator_.BuildFlexfecPacket(*fec_packets.front());
+
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet =
+      receiver_.AddReceivedPacket(ParsePacket(*media_packet));
+  ASSERT_TRUE(received_packet);
+  receiver_.ProcessReceivedPacket(*received_packet);
+  received_packet = receiver_.AddReceivedPacket(ParsePacket(*fec_packet));
+  ASSERT_TRUE(received_packet);
+  receiver_.ProcessReceivedPacket(*received_packet);
+}
+
+TEST_F(FlexfecReceiverTest, FailsOnTruncatedFecPacket) {
+  const size_t kNumMediaPackets = 1;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+  const auto& media_packet = media_packets.front();
+  // Simulate truncated FlexFEC payload.
+  fec_packets.front()->data.SetSize(1);
+  auto fec_packet = packet_generator_.BuildFlexfecPacket(*fec_packets.front());
+
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet =
+      receiver_.AddReceivedPacket(ParsePacket(*media_packet));
+  ASSERT_TRUE(received_packet);
+  receiver_.ProcessReceivedPacket(*received_packet);
+  EXPECT_FALSE(receiver_.AddReceivedPacket(ParsePacket(*fec_packet)));
+}
+
+TEST_F(FlexfecReceiverTest, FailsOnUnknownMediaSsrc) {
+  const size_t kNumMediaPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  auto& media_packet = media_packets.front();
+  // Corrupt the SSRC.
+  media_packet->data.MutableData()[8] = 0;
+  media_packet->data.MutableData()[9] = 1;
+  media_packet->data.MutableData()[10] = 2;
+  media_packet->data.MutableData()[11] = 3;
+
+  EXPECT_FALSE(receiver_.AddReceivedPacket(ParsePacket(*media_packet)));
+}
+
+TEST_F(FlexfecReceiverTest, FailsOnUnknownFecSsrc) {
+  const size_t kNumMediaPackets = 1;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+  const auto& media_packet = media_packets.front();
+  auto fec_packet = packet_generator_.BuildFlexfecPacket(*fec_packets.front());
+  // Corrupt the SSRC.
+  fec_packet->data.MutableData()[8] = 4;
+  fec_packet->data.MutableData()[9] = 5;
+  fec_packet->data.MutableData()[10] = 6;
+  fec_packet->data.MutableData()[11] = 7;
+
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet =
+      receiver_.AddReceivedPacket(ParsePacket(*media_packet));
+  ASSERT_TRUE(received_packet);
+  receiver_.ProcessReceivedPacket(*received_packet);
+  EXPECT_FALSE(receiver_.AddReceivedPacket(ParsePacket(*fec_packet)));
+}
+
+TEST_F(FlexfecReceiverTest, ReceivesMultiplePackets) {
+  const size_t kNumMediaPackets = 2;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+
+  // Receive all media packets.
+  for (const auto& media_packet : media_packets) {
+    std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet =
+        receiver_.AddReceivedPacket(ParsePacket(*media_packet));
+    ASSERT_TRUE(received_packet);
+    receiver_.ProcessReceivedPacket(*received_packet);
+  }
+
+  // Receive FEC packet.
+  auto* fec_packet = fec_packets.front();
+  std::unique_ptr<Packet> packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(*fec_packet);
+  std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet =
+      receiver_.AddReceivedPacket(ParsePacket(*packet_with_rtp_header));
+  ASSERT_TRUE(received_packet);
+  receiver_.ProcessReceivedPacket(*received_packet);
+}
+
+TEST_F(FlexfecReceiverTest, RecoversFromSingleMediaLoss) {
+  const size_t kNumMediaPackets = 2;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+
+  // Receive first media packet but drop second.
+  auto media_it = media_packets.begin();
+  receiver_.OnRtpPacket(ParsePacket(**media_it));
+
+  // Receive FEC packet and ensure recovery of lost media packet.
+  auto fec_it = fec_packets.begin();
+  std::unique_ptr<Packet> packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(**fec_it);
+  media_it++;
+  EXPECT_CALL(recovered_packet_receiver_,
+              OnRecoveredPacket(
+                  Property(&RtpPacketReceived::Buffer, Eq((*media_it)->data))));
+  receiver_.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+}
+
+TEST_F(FlexfecReceiverTest, RecoversFromDoubleMediaLoss) {
+  const size_t kNumMediaPackets = 2;
+  const size_t kNumFecPackets = 2;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+
+  // Drop both media packets.
+
+  // Receive first FEC packet and recover first lost media packet.
+  auto fec_it = fec_packets.begin();
+  std::unique_ptr<Packet> packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(**fec_it);
+  auto media_it = media_packets.begin();
+  EXPECT_CALL(recovered_packet_receiver_,
+              OnRecoveredPacket(
+                  Property(&RtpPacketReceived::Buffer, Eq((*media_it)->data))));
+  receiver_.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+
+  // Receive second FEC packet and recover second lost media packet.
+  fec_it++;
+  packet_with_rtp_header = packet_generator_.BuildFlexfecPacket(**fec_it);
+  media_it++;
+  EXPECT_CALL(recovered_packet_receiver_,
+              OnRecoveredPacket(
+                  Property(&RtpPacketReceived::Buffer, Eq((*media_it)->data))));
+
+  receiver_.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+}
+
+TEST_F(FlexfecReceiverTest, DoesNotRecoverFromMediaAndFecLoss) {
+  const size_t kNumMediaPackets = 2;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+
+  // Receive first media packet.
+  auto media_it = media_packets.begin();
+  receiver_.OnRtpPacket(ParsePacket(**media_it));
+
+  // Drop second media packet and FEC packet. Do not expect call back.
+}
+
+TEST_F(FlexfecReceiverTest, DoesNotCallbackTwice) {
+  const size_t kNumMediaPackets = 2;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+
+  // Receive first media packet but drop second.
+  auto media_it = media_packets.begin();
+  receiver_.OnRtpPacket(ParsePacket(**media_it));
+
+  // Receive FEC packet and ensure recovery of lost media packet.
+  auto fec_it = fec_packets.begin();
+  std::unique_ptr<Packet> packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(**fec_it);
+  media_it++;
+  EXPECT_CALL(recovered_packet_receiver_,
+              OnRecoveredPacket(
+                  Property(&RtpPacketReceived::Buffer, Eq((*media_it)->data))));
+  receiver_.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+
+  // Receive the FEC packet again, but do not call back.
+  receiver_.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+
+  // Receive the first media packet again, but do not call back.
+  media_it = media_packets.begin();
+  receiver_.OnRtpPacket(ParsePacket(**media_it));
+
+  // Receive the second media packet again (the one recovered above),
+  // but do not call back again.
+  media_it++;
+  receiver_.OnRtpPacket(ParsePacket(**media_it));
+}
+
+// Here we are implicitly assuming packet masks that are suitable for
+// this type of 50% correlated loss. If we are changing our precomputed
+// packet masks, this test might need to be updated.
+TEST_F(FlexfecReceiverTest, RecoversFrom50PercentLoss) {
+  const size_t kNumFecPackets = 5;
+  const size_t kNumFrames = 2 * kNumFecPackets;
+  const size_t kNumMediaPacketsPerFrame = 1;
+
+  PacketList media_packets;
+  for (size_t i = 0; i < kNumFrames; ++i) {
+    PacketizeFrame(kNumMediaPacketsPerFrame, i, &media_packets);
+  }
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+
+  // Drop every second media packet.
+  auto media_it = media_packets.begin();
+  while (media_it != media_packets.end()) {
+    receiver_.OnRtpPacket(ParsePacket(**media_it));
+    ++media_it;
+    if (media_it == media_packets.end()) {
+      break;
+    }
+    ++media_it;
+  }
+
+  // Receive all FEC packets.
+  media_it = media_packets.begin();
+  for (const auto* fec_packet : fec_packets) {
+    std::unique_ptr<Packet> fec_packet_with_rtp_header =
+        packet_generator_.BuildFlexfecPacket(*fec_packet);
+    ++media_it;
+    if (media_it == media_packets.end()) {
+      break;
+    }
+    EXPECT_CALL(recovered_packet_receiver_,
+                OnRecoveredPacket(Property(&RtpPacketReceived::Buffer,
+                                           Eq((*media_it)->data))));
+    receiver_.OnRtpPacket(ParsePacket(*fec_packet_with_rtp_header));
+    ++media_it;
+  }
+}
+
+TEST_F(FlexfecReceiverTest, DelayedFecPacketDoesHelp) {
+  // These values need to be updated if the underlying erasure code
+  // implementation changes.
+  // Delay FEC packet by maximum number of media packets tracked by receiver.
+  const size_t kNumFrames = 192;
+  const size_t kNumMediaPacketsPerFrame = 1;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPacketsPerFrame, 0, &media_packets);
+  PacketizeFrame(kNumMediaPacketsPerFrame, 1, &media_packets);
+  // Protect two first frames.
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+  for (size_t i = 2; i < kNumFrames; ++i) {
+    PacketizeFrame(kNumMediaPacketsPerFrame, i, &media_packets);
+  }
+
+  // Drop first media packet and delay FEC packet.
+  auto media_it = media_packets.begin();
+  ++media_it;
+
+  // Receive all other media packets.
+  while (media_it != media_packets.end()) {
+    receiver_.OnRtpPacket(ParsePacket(**media_it));
+    ++media_it;
+  }
+
+  // Receive FEC packet and recover first media packet.
+  auto fec_it = fec_packets.begin();
+  std::unique_ptr<Packet> packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(**fec_it);
+  media_it = media_packets.begin();
+  EXPECT_CALL(recovered_packet_receiver_,
+              OnRecoveredPacket(
+                  Property(&RtpPacketReceived::Buffer, Eq((*media_it)->data))));
+  receiver_.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+}
+
+TEST_F(FlexfecReceiverTest, TooDelayedFecPacketDoesNotHelp) {
+  // These values need to be updated if the underlying erasure code
+  // implementation changes.
+  // Delay FEC packet by one more than maximum number of media packets
+  // tracked by receiver.
+  const size_t kNumFrames = 193;
+  const size_t kNumMediaPacketsPerFrame = 1;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPacketsPerFrame, 0, &media_packets);
+  PacketizeFrame(kNumMediaPacketsPerFrame, 1, &media_packets);
+  // Protect first two frames.
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+  for (size_t i = 2; i < kNumFrames; ++i) {
+    PacketizeFrame(kNumMediaPacketsPerFrame, i, &media_packets);
+  }
+
+  // Drop first media packet and delay FEC packet.
+  auto media_it = media_packets.begin();
+  ++media_it;
+
+  // Receive all other media packets.
+  while (media_it != media_packets.end()) {
+    receiver_.OnRtpPacket(ParsePacket(**media_it));
+    ++media_it;
+  }
+
+  // Receive FEC packet.
+  auto fec_it = fec_packets.begin();
+  std::unique_ptr<Packet> packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(**fec_it);
+  receiver_.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+
+  // Do not expect a call back.
+}
+
+TEST_F(FlexfecReceiverTest, SurvivesOldRecoveredPacketBeingReinserted) {
+  // Simulates the behaviour of the
+  // Call->FlexfecReceiveStream->FlexfecReceiver->Call loop in production code.
+  class LoopbackRecoveredPacketReceiver : public RecoveredPacketReceiver {
+   public:
+    LoopbackRecoveredPacketReceiver() : receiver_(nullptr) {}
+
+    void SetReceiver(FlexfecReceiver* receiver) { receiver_ = receiver; }
+
+    // Implements RecoveredPacketReceiver.
+    void OnRecoveredPacket(const RtpPacketReceived& packet) override {
+      EXPECT_TRUE(packet.recovered());
+      RTC_DCHECK(receiver_);
+      receiver_->OnRtpPacket(packet);
+    }
+
+   private:
+    FlexfecReceiver* receiver_;
+  } loopback_recovered_packet_receiver;
+
+  // Feed recovered packets back into `receiver`.
+  FlexfecReceiver receiver(Clock::GetRealTimeClock(), kFlexfecSsrc, kMediaSsrc,
+                           &loopback_recovered_packet_receiver);
+  loopback_recovered_packet_receiver.SetReceiver(&receiver);
+
+  // Receive first set of packets.
+  PacketList first_media_packets;
+  for (int i = 0; i < 46; ++i) {
+    PacketizeFrame(1, 0, &first_media_packets);
+  }
+  for (const auto& media_packet : first_media_packets) {
+    receiver.OnRtpPacket(ParsePacket(*media_packet));
+  }
+
+  // Protect one media packet. Lose the media packet,
+  // but do not receive FEC packet yet.
+  PacketList protected_media_packet;
+  PacketizeFrame(1, 0, &protected_media_packet);
+  const std::list<Packet*> fec_packets = EncodeFec(protected_media_packet, 1);
+  EXPECT_EQ(1u, fec_packets.size());
+  std::unique_ptr<Packet> fec_packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(*fec_packets.front());
+
+  // Lose some packets, thus introducing a sequence number gap.
+  PacketList lost_packets;
+  for (int i = 0; i < 100; ++i) {
+    PacketizeFrame(1, 0, &lost_packets);
+  }
+
+  // Receive one more packet.
+  PacketList second_media_packets;
+  PacketizeFrame(1, 0, &second_media_packets);
+  for (const auto& media_packet : second_media_packets) {
+    receiver.OnRtpPacket(ParsePacket(*media_packet));
+  }
+
+  // Receive delayed FEC packet.
+  receiver.OnRtpPacket(ParsePacket(*fec_packet_with_rtp_header));
+
+  // Expect no crash.
+}
+
+TEST_F(FlexfecReceiverTest, RecoversWithMediaPacketsOutOfOrder) {
+  const size_t kNumMediaPackets = 6;
+  const size_t kNumFecPackets = 2;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+
+  // Lose two media packets, and receive the others out of order.
+  auto media_it = media_packets.begin();
+  auto media_packet0 = media_it++;
+  auto media_packet1 = media_it++;
+  auto media_packet2 = media_it++;
+  auto media_packet3 = media_it++;
+  auto media_packet4 = media_it++;
+  auto media_packet5 = media_it++;
+  receiver_.OnRtpPacket(ParsePacket(**media_packet5));
+  receiver_.OnRtpPacket(ParsePacket(**media_packet2));
+  receiver_.OnRtpPacket(ParsePacket(**media_packet3));
+  receiver_.OnRtpPacket(ParsePacket(**media_packet0));
+
+  // Expect to recover lost media packets.
+  EXPECT_CALL(recovered_packet_receiver_,
+              OnRecoveredPacket(Property(&RtpPacketReceived::Buffer,
+                                         Eq((*media_packet1)->data))));
+  EXPECT_CALL(recovered_packet_receiver_,
+              OnRecoveredPacket(Property(&RtpPacketReceived::Buffer,
+                                         Eq((*media_packet4)->data))));
+  // Add FEC packets.
+  auto fec_it = fec_packets.begin();
+  std::unique_ptr<Packet> packet_with_rtp_header;
+  while (fec_it != fec_packets.end()) {
+    packet_with_rtp_header = packet_generator_.BuildFlexfecPacket(**fec_it);
+    receiver_.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+    ++fec_it;
+  }
+}
+
+// Recovered media packets may be fed back into the FlexfecReceiver by the
+// callback. This test ensures the idempotency of such a situation.
+TEST_F(FlexfecReceiverTest, RecoveryCallbackDoesNotLoopInfinitely) {
+  class LoopbackRecoveredPacketReceiver : public RecoveredPacketReceiver {
+   public:
+    const int kMaxRecursionDepth = 10;
+
+    LoopbackRecoveredPacketReceiver()
+        : receiver_(nullptr),
+          did_receive_call_back_(false),
+          recursion_depth_(0),
+          deep_recursion_(false) {}
+
+    void SetReceiver(FlexfecReceiver* receiver) { receiver_ = receiver; }
+    bool DidReceiveCallback() const { return did_receive_call_back_; }
+    bool DeepRecursion() const { return deep_recursion_; }
+
+    // Implements RecoveredPacketReceiver.
+    void OnRecoveredPacket(const RtpPacketReceived& packet) override {
+      did_receive_call_back_ = true;
+
+      if (recursion_depth_ > kMaxRecursionDepth) {
+        deep_recursion_ = true;
+        return;
+      }
+      ++recursion_depth_;
+      RTC_DCHECK(receiver_);
+      receiver_->OnRtpPacket(packet);
+      --recursion_depth_;
+    }
+
+   private:
+    FlexfecReceiver* receiver_;
+    bool did_receive_call_back_;
+    int recursion_depth_;
+    bool deep_recursion_;
+  } loopback_recovered_packet_receiver;
+
+  // Feed recovered packets back into `receiver`.
+  FlexfecReceiver receiver(Clock::GetRealTimeClock(), kFlexfecSsrc, kMediaSsrc,
+                           &loopback_recovered_packet_receiver);
+  loopback_recovered_packet_receiver.SetReceiver(&receiver);
+
+  const size_t kNumMediaPackets = 2;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+
+  // Receive first media packet but drop second.
+  auto media_it = media_packets.begin();
+  receiver.OnRtpPacket(ParsePacket(**media_it));
+
+  // Receive FEC packet and verify that a packet was recovered.
+  auto fec_it = fec_packets.begin();
+  std::unique_ptr<Packet> packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(**fec_it);
+  receiver.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+  EXPECT_TRUE(loopback_recovered_packet_receiver.DidReceiveCallback());
+  EXPECT_FALSE(loopback_recovered_packet_receiver.DeepRecursion());
+}
+
+TEST_F(FlexfecReceiverTest, CalculatesNumberOfPackets) {
+  const size_t kNumMediaPackets = 2;
+  const size_t kNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kNumMediaPackets, 0, &media_packets);
+  std::list<Packet*> fec_packets = EncodeFec(media_packets, kNumFecPackets);
+
+  // Receive first media packet but drop second.
+  auto media_it = media_packets.begin();
+  receiver_.OnRtpPacket(ParsePacket(**media_it));
+
+  // Receive FEC packet and ensure recovery of lost media packet.
+  auto fec_it = fec_packets.begin();
+  std::unique_ptr<Packet> packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(**fec_it);
+  media_it++;
+  EXPECT_CALL(recovered_packet_receiver_,
+              OnRecoveredPacket(
+                  Property(&RtpPacketReceived::Buffer, Eq((*media_it)->data))));
+  receiver_.OnRtpPacket(ParsePacket(*packet_with_rtp_header));
+
+  // Check stats calculations.
+  FecPacketCounter packet_counter = receiver_.GetPacketCounter();
+  EXPECT_EQ(2U, packet_counter.num_packets);
+  EXPECT_EQ(1U, packet_counter.num_fec_packets);
+  EXPECT_EQ(1U, packet_counter.num_recovered_packets);
+}
+
+TEST_F(FlexfecReceiverTest, DoesNotDecodeWrappedMediaSequenceUsingOldFec) {
+  const size_t kFirstFrameNumMediaPackets = 2;
+  const size_t kFirstFrameNumFecPackets = 1;
+
+  PacketList media_packets;
+  PacketizeFrame(kFirstFrameNumMediaPackets, 0, &media_packets);
+
+  // Protect first frame (sequences 0 and 1) with 1 FEC packet.
+  std::list<Packet*> fec_packets =
+      EncodeFec(media_packets, kFirstFrameNumFecPackets);
+
+  // Generate enough media packets to simulate media sequence number wraparound.
+  // Use no FEC for these frames to make sure old FEC is not purged due to age.
+  const size_t kNumFramesSequenceWrapAround =
+      std::numeric_limits<uint16_t>::max();
+  const size_t kNumMediaPacketsPerFrame = 1;
+
+  for (size_t i = 1; i <= kNumFramesSequenceWrapAround; ++i) {
+    PacketizeFrame(kNumMediaPacketsPerFrame, i, &media_packets);
+  }
+
+  // Receive first (`kFirstFrameNumMediaPackets` + 192) media packets.
+  // Simulate an old FEC packet by separating it from its encoded media
+  // packets by at least 192 packets.
+  auto media_it = media_packets.begin();
+  for (size_t i = 0; i < (kFirstFrameNumMediaPackets + 192); i++) {
+    if (i == 1) {
+      // Drop the second packet of the first frame.
+      media_it++;
+    } else {
+      receiver_.OnRtpPacket(ParsePacket(**media_it++));
+    }
+  }
+
+  // Receive FEC packet. Although a protected packet was dropped,
+  // expect no recovery callback since it is delayed from first frame
+  // by more than 192 packets.
+  auto fec_it = fec_packets.begin();
+  std::unique_ptr<Packet> fec_packet_with_rtp_header =
+      packet_generator_.BuildFlexfecPacket(**fec_it);
+  receiver_.OnRtpPacket(ParsePacket(*fec_packet_with_rtp_header));
+
+  // Receive remaining media packets.
+  // NOTE: Because we sent enough to simulate wrap around, sequence 0 is
+  // received again, but is a different packet than the original first
+  // packet of first frame.
+  while (media_it != media_packets.end()) {
+    receiver_.OnRtpPacket(ParsePacket(**media_it++));
+  }
+
+  // Do not expect a recovery callback, the FEC packet is old
+  // and should not decode wrapped around media sequences.
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender.cc
new file mode 100644
index 0000000000..292fe4a8dd
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender.cc
@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/flexfec_sender.h"
+
+#include <string.h>
+
+#include <list>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// Let first sequence number be in the first half of the interval.
+constexpr uint16_t kMaxInitRtpSeqNumber = 0x7fff;
+
+// See breakdown in flexfec_header_reader_writer.cc.
+constexpr size_t kFlexfecMaxHeaderSize = 32;
+
+// Since we will mainly use FlexFEC to protect video streams, we use a 90 kHz
+// clock for the RTP timestamps. (This is according to the RFC, which states
+// that it is RECOMMENDED to use the same clock frequency for FlexFEC as for
+// the protected media stream.)
+// The constant converts from clock millisecond timestamps to the 90 kHz
+// RTP timestamp.
+const int kMsToRtpTimestamp = kVideoPayloadTypeFrequency / 1000;
+
+// How often to log the generated FEC packets to the text log.
+constexpr int64_t kPacketLogIntervalMs = 10000;
+
+RtpHeaderExtensionMap RegisterSupportedExtensions(
+    const std::vector<RtpExtension>& rtp_header_extensions) {
+  RtpHeaderExtensionMap map;
+  for (const auto& extension : rtp_header_extensions) {
+    if (extension.uri == TransportSequenceNumber::Uri()) {
+      map.Register<TransportSequenceNumber>(extension.id);
+    } else if (extension.uri == AbsoluteSendTime::Uri()) {
+      map.Register<AbsoluteSendTime>(extension.id);
+    } else if (extension.uri == TransmissionOffset::Uri()) {
+      map.Register<TransmissionOffset>(extension.id);
+    } else if (extension.uri == RtpMid::Uri()) {
+      map.Register<RtpMid>(extension.id);
+    } else {
+      RTC_LOG(LS_INFO)
+          << "FlexfecSender only supports RTP header extensions for "
+             "BWE and MID, so the extension "
+          << extension.ToString() << " will not be used.";
+    }
+  }
+  return map;
+}
+
+}  // namespace
+
+FlexfecSender::FlexfecSender(
+    int payload_type,
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc,
+    absl::string_view mid,
+    const std::vector<RtpExtension>& rtp_header_extensions,
+    rtc::ArrayView<const RtpExtensionSize> extension_sizes,
+    const RtpState* rtp_state,
+    Clock* clock)
+    : clock_(clock),
+      random_(clock_->TimeInMicroseconds()),
+      last_generated_packet_ms_(-1),
+      payload_type_(payload_type),
+      // Reset RTP state if this is not the first time we are operating.
+      // Otherwise, randomize the initial timestamp offset and RTP sequence
+      // numbers. (This is not intended to be cryptographically strong.)
+      timestamp_offset_(rtp_state ? rtp_state->start_timestamp
+                                  : random_.Rand<uint32_t>()),
+      ssrc_(ssrc),
+      protected_media_ssrc_(protected_media_ssrc),
+      mid_(mid),
+      seq_num_(rtp_state ? rtp_state->sequence_number
+                         : random_.Rand(1, kMaxInitRtpSeqNumber)),
+      ulpfec_generator_(
+          ForwardErrorCorrection::CreateFlexfec(ssrc, protected_media_ssrc),
+          clock_),
+      rtp_header_extension_map_(
+          RegisterSupportedExtensions(rtp_header_extensions)),
+      header_extensions_size_(
+          RtpHeaderExtensionSize(extension_sizes, rtp_header_extension_map_)),
+      fec_bitrate_(/*max_window_size_ms=*/1000, RateStatistics::kBpsScale) {
+  // This object should not have been instantiated if FlexFEC is disabled.
+  RTC_DCHECK_GE(payload_type, 0);
+  RTC_DCHECK_LE(payload_type, 127);
+}
+
+FlexfecSender::~FlexfecSender() = default;
+
+// We are reusing the implementation from UlpfecGenerator for SetFecParameters,
+// AddRtpPacketAndGenerateFec, and FecAvailable.
+void FlexfecSender::SetProtectionParameters(
+    const FecProtectionParams& delta_params,
+    const FecProtectionParams& key_params) {
+  ulpfec_generator_.SetProtectionParameters(delta_params, key_params);
+}
+
+void FlexfecSender::AddPacketAndGenerateFec(const RtpPacketToSend& packet) {
+  // TODO(brandtr): Generalize this SSRC check when we support multistream
+  // protection.
+  RTC_DCHECK_EQ(packet.Ssrc(), protected_media_ssrc_);
+  ulpfec_generator_.AddPacketAndGenerateFec(packet);
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> FlexfecSender::GetFecPackets() {
+  RTC_CHECK_RUNS_SERIALIZED(&ulpfec_generator_.race_checker_);
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets_to_send;
+  fec_packets_to_send.reserve(ulpfec_generator_.generated_fec_packets_.size());
+  size_t total_fec_data_bytes = 0;
+  for (const auto* fec_packet : ulpfec_generator_.generated_fec_packets_) {
+    std::unique_ptr<RtpPacketToSend> fec_packet_to_send(
+        new RtpPacketToSend(&rtp_header_extension_map_));
+    fec_packet_to_send->set_packet_type(
+        RtpPacketMediaType::kForwardErrorCorrection);
+    fec_packet_to_send->set_allow_retransmission(false);
+
+    // RTP header.
+    fec_packet_to_send->SetMarker(false);
+    fec_packet_to_send->SetPayloadType(payload_type_);
+    fec_packet_to_send->SetSequenceNumber(seq_num_++);
+    fec_packet_to_send->SetTimestamp(
+        timestamp_offset_ +
+        static_cast<uint32_t>(kMsToRtpTimestamp *
+                              clock_->TimeInMilliseconds()));
+    // Set "capture time" so that the TransmissionOffset header extension
+    // can be set by the RTPSender.
+    fec_packet_to_send->set_capture_time(clock_->CurrentTime());
+    fec_packet_to_send->SetSsrc(ssrc_);
+    // Reserve extensions, if registered. These will be set by the RTPSender.
+    fec_packet_to_send->ReserveExtension<AbsoluteSendTime>();
+    fec_packet_to_send->ReserveExtension<TransmissionOffset>();
+    fec_packet_to_send->ReserveExtension<TransportSequenceNumber>();
+    // Possibly include the MID header extension.
+    if (!mid_.empty()) {
+      // This is a no-op if the MID header extension is not registered.
+      fec_packet_to_send->SetExtension<RtpMid>(mid_);
+    }
+
+    // RTP payload.
+    uint8_t* payload =
+        fec_packet_to_send->AllocatePayload(fec_packet->data.size());
+    memcpy(payload, fec_packet->data.cdata(), fec_packet->data.size());
+
+    total_fec_data_bytes += fec_packet_to_send->size();
+    fec_packets_to_send.push_back(std::move(fec_packet_to_send));
+  }
+
+  if (!fec_packets_to_send.empty()) {
+    ulpfec_generator_.ResetState();
+  }
+
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (!fec_packets_to_send.empty() &&
+      now_ms - last_generated_packet_ms_ > kPacketLogIntervalMs) {
+    RTC_LOG(LS_VERBOSE) << "Generated " << fec_packets_to_send.size()
+                        << " FlexFEC packets with payload type: "
+                        << payload_type_ << " and SSRC: " << ssrc_ << ".";
+    last_generated_packet_ms_ = now_ms;
+  }
+
+  MutexLock lock(&mutex_);
+  fec_bitrate_.Update(total_fec_data_bytes, now_ms);
+
+  return fec_packets_to_send;
+}
+
+// The overhead is BWE RTP header extensions and FlexFEC header.
+size_t FlexfecSender::MaxPacketOverhead() const {
+  return header_extensions_size_ + kFlexfecMaxHeaderSize;
+}
+
+DataRate FlexfecSender::CurrentFecRate() const {
+  MutexLock lock(&mutex_);
+  return DataRate::BitsPerSec(
+      fec_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0));
+}
+
+absl::optional<RtpState> FlexfecSender::GetRtpState() {
+  RtpState rtp_state;
+  rtp_state.sequence_number = seq_num_;
+  rtp_state.start_timestamp = timestamp_offset_;
+  return rtp_state;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender_unittest.cc
new file mode 100644
index 0000000000..19614d2bbd
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_sender_unittest.cc
@@ -0,0 +1,342 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/flexfec_sender.h"
+
+#include <vector>
+
+#include "api/rtp_parameters.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/fec_test_helper.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_sender.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using test::fec::AugmentedPacket;
+using test::fec::AugmentedPacketGenerator;
+
+constexpr int kFlexfecPayloadType = 123;
+constexpr uint32_t kMediaSsrc = 1234;
+constexpr uint32_t kFlexfecSsrc = 5678;
+const char kNoMid[] = "";
+const std::vector<RtpExtension> kNoRtpHeaderExtensions;
+const std::vector<RtpExtensionSize> kNoRtpHeaderExtensionSizes;
+// Assume a single protected media SSRC.
+constexpr size_t kFlexfecMaxHeaderSize = 32;
+constexpr size_t kPayloadLength = 50;
+
+constexpr int64_t kInitialSimulatedClockTime = 1;
+// These values are deterministically given by the PRNG, due to our fixed seed.
+// They should be updated if the PRNG implementation changes.
+constexpr uint16_t kDeterministicSequenceNumber = 28732;
+constexpr uint32_t kDeterministicTimestamp = 2305613085;
+
+// Round up to the nearest size that is a multiple of 4.
+size_t Word32Align(size_t size) {
+  uint32_t remainder = size % 4;
+  if (remainder != 0)
+    return size + 4 - remainder;
+  return size;
+}
+
+std::unique_ptr<RtpPacketToSend> GenerateSingleFlexfecPacket(
+    FlexfecSender* sender) {
+  // Parameters selected to generate a single FEC packet.
+  FecProtectionParams params;
+  params.fec_rate = 15;
+  params.max_fec_frames = 1;
+  params.fec_mask_type = kFecMaskRandom;
+  constexpr size_t kNumPackets = 4;
+
+  sender->SetProtectionParameters(params, params);
+  AugmentedPacketGenerator packet_generator(kMediaSsrc);
+  packet_generator.NewFrame(kNumPackets);
+  for (size_t i = 0; i < kNumPackets; ++i) {
+    std::unique_ptr<AugmentedPacket> packet =
+        packet_generator.NextPacket(i, kPayloadLength);
+    RtpPacketToSend rtp_packet(nullptr);  // No header extensions.
+    rtp_packet.Parse(packet->data);
+    sender->AddPacketAndGenerateFec(rtp_packet);
+  }
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+      sender->GetFecPackets();
+  EXPECT_EQ(1U, fec_packets.size());
+  EXPECT_TRUE(sender->GetFecPackets().empty());
+
+  return std::move(fec_packets.front());
+}
+
+}  // namespace
+
+TEST(FlexfecSenderTest, Ssrc) {
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+
+  EXPECT_EQ(kFlexfecSsrc, sender.FecSsrc());
+}
+
+TEST(FlexfecSenderTest, NoFecAvailableBeforeMediaAdded) {
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+
+  EXPECT_TRUE(sender.GetFecPackets().empty());
+}
+
+TEST(FlexfecSenderTest, ProtectOneFrameWithOneFecPacket) {
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+  auto fec_packet = GenerateSingleFlexfecPacket(&sender);
+
+  EXPECT_EQ(kRtpHeaderSize, fec_packet->headers_size());
+  EXPECT_FALSE(fec_packet->Marker());
+  EXPECT_EQ(kFlexfecPayloadType, fec_packet->PayloadType());
+  EXPECT_EQ(kDeterministicSequenceNumber, fec_packet->SequenceNumber());
+  EXPECT_EQ(kDeterministicTimestamp, fec_packet->Timestamp());
+  EXPECT_EQ(kFlexfecSsrc, fec_packet->Ssrc());
+  EXPECT_LE(kPayloadLength, fec_packet->payload_size());
+}
+
+TEST(FlexfecSenderTest, ProtectTwoFramesWithOneFecPacket) {
+  // FEC parameters selected to generate a single FEC packet per frame.
+  FecProtectionParams params;
+  params.fec_rate = 15;
+  params.max_fec_frames = 2;
+  params.fec_mask_type = kFecMaskRandom;
+  constexpr size_t kNumFrames = 2;
+  constexpr size_t kNumPacketsPerFrame = 2;
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+  sender.SetProtectionParameters(params, params);
+
+  AugmentedPacketGenerator packet_generator(kMediaSsrc);
+  for (size_t i = 0; i < kNumFrames; ++i) {
+    packet_generator.NewFrame(kNumPacketsPerFrame);
+    for (size_t j = 0; j < kNumPacketsPerFrame; ++j) {
+      std::unique_ptr<AugmentedPacket> packet =
+          packet_generator.NextPacket(i, kPayloadLength);
+      RtpPacketToSend rtp_packet(nullptr);
+      rtp_packet.Parse(packet->data);
+      sender.AddPacketAndGenerateFec(rtp_packet);
+    }
+  }
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+      sender.GetFecPackets();
+  ASSERT_EQ(1U, fec_packets.size());
+  EXPECT_TRUE(sender.GetFecPackets().empty());
+
+  RtpPacketToSend* fec_packet = fec_packets.front().get();
+  EXPECT_EQ(kRtpHeaderSize, fec_packet->headers_size());
+  EXPECT_FALSE(fec_packet->Marker());
+  EXPECT_EQ(kFlexfecPayloadType, fec_packet->PayloadType());
+  EXPECT_EQ(kDeterministicSequenceNumber, fec_packet->SequenceNumber());
+  EXPECT_EQ(kDeterministicTimestamp, fec_packet->Timestamp());
+  EXPECT_EQ(kFlexfecSsrc, fec_packet->Ssrc());
+}
+
+TEST(FlexfecSenderTest, ProtectTwoFramesWithTwoFecPackets) {
+  // FEC parameters selected to generate a single FEC packet per frame.
+  FecProtectionParams params;
+  params.fec_rate = 30;
+  params.max_fec_frames = 1;
+  params.fec_mask_type = kFecMaskRandom;
+  constexpr size_t kNumFrames = 2;
+  constexpr size_t kNumPacketsPerFrame = 2;
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+  sender.SetProtectionParameters(params, params);
+
+  AugmentedPacketGenerator packet_generator(kMediaSsrc);
+  for (size_t i = 0; i < kNumFrames; ++i) {
+    packet_generator.NewFrame(kNumPacketsPerFrame);
+    for (size_t j = 0; j < kNumPacketsPerFrame; ++j) {
+      std::unique_ptr<AugmentedPacket> packet =
+          packet_generator.NextPacket(i, kPayloadLength);
+      RtpPacketToSend rtp_packet(nullptr);
+      rtp_packet.Parse(packet->data);
+      sender.AddPacketAndGenerateFec(rtp_packet);
+    }
+    std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+        sender.GetFecPackets();
+    ASSERT_EQ(1U, fec_packets.size());
+    EXPECT_TRUE(sender.GetFecPackets().empty());
+
+    RtpPacketToSend* fec_packet = fec_packets.front().get();
+    EXPECT_EQ(kRtpHeaderSize, fec_packet->headers_size());
+    EXPECT_FALSE(fec_packet->Marker());
+    EXPECT_EQ(kFlexfecPayloadType, fec_packet->PayloadType());
+    EXPECT_EQ(static_cast<uint16_t>(kDeterministicSequenceNumber + i),
+              fec_packet->SequenceNumber());
+    EXPECT_EQ(kDeterministicTimestamp, fec_packet->Timestamp());
+    EXPECT_EQ(kFlexfecSsrc, fec_packet->Ssrc());
+  }
+}
+
+// In the tests, we only consider RTP header extensions that are useful for BWE.
+TEST(FlexfecSenderTest, NoRtpHeaderExtensionsForBweByDefault) {
+  const std::vector<RtpExtension> kRtpHeaderExtensions{};
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+  auto fec_packet = GenerateSingleFlexfecPacket(&sender);
+
+  EXPECT_FALSE(fec_packet->HasExtension<AbsoluteSendTime>());
+  EXPECT_FALSE(fec_packet->HasExtension<TransmissionOffset>());
+  EXPECT_FALSE(fec_packet->HasExtension<TransportSequenceNumber>());
+}
+
+TEST(FlexfecSenderTest, RegisterAbsoluteSendTimeRtpHeaderExtension) {
+  const std::vector<RtpExtension> kRtpHeaderExtensions{
+      {RtpExtension::kAbsSendTimeUri, 1}};
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+  auto fec_packet = GenerateSingleFlexfecPacket(&sender);
+
+  EXPECT_TRUE(fec_packet->HasExtension<AbsoluteSendTime>());
+  EXPECT_FALSE(fec_packet->HasExtension<TransmissionOffset>());
+  EXPECT_FALSE(fec_packet->HasExtension<TransportSequenceNumber>());
+}
+
+TEST(FlexfecSenderTest, RegisterTransmissionOffsetRtpHeaderExtension) {
+  const std::vector<RtpExtension> kRtpHeaderExtensions{
+      {RtpExtension::kTimestampOffsetUri, 1}};
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+  auto fec_packet = GenerateSingleFlexfecPacket(&sender);
+
+  EXPECT_FALSE(fec_packet->HasExtension<AbsoluteSendTime>());
+  EXPECT_TRUE(fec_packet->HasExtension<TransmissionOffset>());
+  EXPECT_FALSE(fec_packet->HasExtension<TransportSequenceNumber>());
+}
+
+TEST(FlexfecSenderTest, RegisterTransportSequenceNumberRtpHeaderExtension) {
+  const std::vector<RtpExtension> kRtpHeaderExtensions{
+      {RtpExtension::kTransportSequenceNumberUri, 1}};
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+  auto fec_packet = GenerateSingleFlexfecPacket(&sender);
+
+  EXPECT_FALSE(fec_packet->HasExtension<AbsoluteSendTime>());
+  EXPECT_FALSE(fec_packet->HasExtension<TransmissionOffset>());
+  EXPECT_TRUE(fec_packet->HasExtension<TransportSequenceNumber>());
+}
+
+TEST(FlexfecSenderTest, RegisterAllRtpHeaderExtensionsForBwe) {
+  const std::vector<RtpExtension> kRtpHeaderExtensions{
+      {RtpExtension::kAbsSendTimeUri, 1},
+      {RtpExtension::kTimestampOffsetUri, 2},
+      {RtpExtension::kTransportSequenceNumberUri, 3}};
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+  auto fec_packet = GenerateSingleFlexfecPacket(&sender);
+
+  EXPECT_TRUE(fec_packet->HasExtension<AbsoluteSendTime>());
+  EXPECT_TRUE(fec_packet->HasExtension<TransmissionOffset>());
+  EXPECT_TRUE(fec_packet->HasExtension<TransportSequenceNumber>());
+}
+
+TEST(FlexfecSenderTest, MaxPacketOverhead) {
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+
+  EXPECT_EQ(kFlexfecMaxHeaderSize, sender.MaxPacketOverhead());
+}
+
+TEST(FlexfecSenderTest, MaxPacketOverheadWithExtensions) {
+  const std::vector<RtpExtension> kRtpHeaderExtensions{
+      {RtpExtension::kAbsSendTimeUri, 1},
+      {RtpExtension::kTimestampOffsetUri, 2},
+      {RtpExtension::kTransportSequenceNumberUri, 3}};
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  const size_t kExtensionHeaderLength = 1;
+  const size_t kRtpOneByteHeaderLength = 4;
+  const size_t kExtensionsTotalSize =
+      Word32Align(kRtpOneByteHeaderLength + kExtensionHeaderLength +
+                  AbsoluteSendTime::kValueSizeBytes + kExtensionHeaderLength +
+                  TransmissionOffset::kValueSizeBytes + kExtensionHeaderLength +
+                  TransportSequenceNumber::kValueSizeBytes);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kRtpHeaderExtensions, RTPSender::FecExtensionSizes(),
+                       nullptr /* rtp_state */, &clock);
+
+  EXPECT_EQ(kExtensionsTotalSize + kFlexfecMaxHeaderSize,
+            sender.MaxPacketOverhead());
+}
+
+TEST(FlexfecSenderTest, MidIncludedInPacketsWhenSet) {
+  const std::vector<RtpExtension> kRtpHeaderExtensions{
+      {RtpExtension::kMidUri, 1}};
+  const char kMid[] = "mid";
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kMid,
+                       kRtpHeaderExtensions, RTPSender::FecExtensionSizes(),
+                       nullptr /* rtp_state */, &clock);
+
+  auto fec_packet = GenerateSingleFlexfecPacket(&sender);
+
+  std::string mid;
+  ASSERT_TRUE(fec_packet->GetExtension<RtpMid>(&mid));
+  EXPECT_EQ(kMid, mid);
+}
+
+TEST(FlexfecSenderTest, SetsAndGetsRtpState) {
+  RtpState initial_rtp_state;
+  initial_rtp_state.sequence_number = 100;
+  initial_rtp_state.start_timestamp = 200;
+  SimulatedClock clock(kInitialSimulatedClockTime);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       &initial_rtp_state, &clock);
+
+  auto fec_packet = GenerateSingleFlexfecPacket(&sender);
+  EXPECT_EQ(initial_rtp_state.sequence_number, fec_packet->SequenceNumber());
+  EXPECT_EQ(initial_rtp_state.start_timestamp, fec_packet->Timestamp());
+
+  clock.AdvanceTimeMilliseconds(1000);
+  fec_packet = GenerateSingleFlexfecPacket(&sender);
+  EXPECT_EQ(initial_rtp_state.sequence_number + 1,
+            fec_packet->SequenceNumber());
+  EXPECT_EQ(initial_rtp_state.start_timestamp + 1 * kVideoPayloadTypeFrequency,
+            fec_packet->Timestamp());
+
+  RtpState updated_rtp_state = sender.GetRtpState().value();
+  EXPECT_EQ(initial_rtp_state.sequence_number + 2,
+            updated_rtp_state.sequence_number);
+  EXPECT_EQ(initial_rtp_state.start_timestamp,
+            updated_rtp_state.start_timestamp);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction.cc
new file mode 100644
index 0000000000..1462c2f481
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -0,0 +1,816 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/mod_ops.h"
+
+namespace webrtc {
+
+namespace {
+// Transport header size in bytes. Assume UDP/IPv4 as a reasonable minimum.
+constexpr size_t kTransportOverhead = 28;
+
+constexpr uint16_t kOldSequenceThreshold = 0x3fff;
+}  // namespace
+
+ForwardErrorCorrection::Packet::Packet() : data(0), ref_count_(0) {}
+ForwardErrorCorrection::Packet::~Packet() = default;
+
+int32_t ForwardErrorCorrection::Packet::AddRef() {
+  return ++ref_count_;
+}
+
+int32_t ForwardErrorCorrection::Packet::Release() {
+  int32_t ref_count;
+  ref_count = --ref_count_;
+  if (ref_count == 0)
+    delete this;
+  return ref_count;
+}
+
+// This comparator is used to compare std::unique_ptr's pointing to
+// subclasses of SortablePackets. It needs to be parametric since
+// the std::unique_ptr's are not covariant w.r.t. the types that
+// they are pointing to.
+template <typename S, typename T>
+bool ForwardErrorCorrection::SortablePacket::LessThan::operator()(
+    const S& first,
+    const T& second) {
+  RTC_DCHECK_EQ(first->ssrc, second->ssrc);
+  return IsNewerSequenceNumber(second->seq_num, first->seq_num);
+}
+
+ForwardErrorCorrection::ReceivedPacket::ReceivedPacket() = default;
+ForwardErrorCorrection::ReceivedPacket::~ReceivedPacket() = default;
+
+ForwardErrorCorrection::RecoveredPacket::RecoveredPacket() = default;
+ForwardErrorCorrection::RecoveredPacket::~RecoveredPacket() = default;
+
+ForwardErrorCorrection::ProtectedPacket::ProtectedPacket() = default;
+ForwardErrorCorrection::ProtectedPacket::~ProtectedPacket() = default;
+
+ForwardErrorCorrection::ReceivedFecPacket::ReceivedFecPacket() = default;
+ForwardErrorCorrection::ReceivedFecPacket::~ReceivedFecPacket() = default;
+
+ForwardErrorCorrection::ForwardErrorCorrection(
+    std::unique_ptr<FecHeaderReader> fec_header_reader,
+    std::unique_ptr<FecHeaderWriter> fec_header_writer,
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc)
+    : ssrc_(ssrc),
+      protected_media_ssrc_(protected_media_ssrc),
+      fec_header_reader_(std::move(fec_header_reader)),
+      fec_header_writer_(std::move(fec_header_writer)),
+      generated_fec_packets_(fec_header_writer_->MaxFecPackets()),
+      packet_mask_size_(0) {}
+
+ForwardErrorCorrection::~ForwardErrorCorrection() = default;
+
+std::unique_ptr<ForwardErrorCorrection> ForwardErrorCorrection::CreateUlpfec(
+    uint32_t ssrc) {
+  std::unique_ptr<FecHeaderReader> fec_header_reader(new UlpfecHeaderReader());
+  std::unique_ptr<FecHeaderWriter> fec_header_writer(new UlpfecHeaderWriter());
+  return std::unique_ptr<ForwardErrorCorrection>(new ForwardErrorCorrection(
+      std::move(fec_header_reader), std::move(fec_header_writer), ssrc, ssrc));
+}
+
+std::unique_ptr<ForwardErrorCorrection> ForwardErrorCorrection::CreateFlexfec(
+    uint32_t ssrc,
+    uint32_t protected_media_ssrc) {
+  std::unique_ptr<FecHeaderReader> fec_header_reader(new FlexfecHeaderReader());
+  std::unique_ptr<FecHeaderWriter> fec_header_writer(new FlexfecHeaderWriter());
+  return std::unique_ptr<ForwardErrorCorrection>(new ForwardErrorCorrection(
+      std::move(fec_header_reader), std::move(fec_header_writer), ssrc,
+      protected_media_ssrc));
+}
+
+int ForwardErrorCorrection::EncodeFec(const PacketList& media_packets,
+                                      uint8_t protection_factor,
+                                      int num_important_packets,
+                                      bool use_unequal_protection,
+                                      FecMaskType fec_mask_type,
+                                      std::list<Packet*>* fec_packets) {
+  const size_t num_media_packets = media_packets.size();
+
+  // Sanity check arguments.
+  RTC_DCHECK_GT(num_media_packets, 0);
+  RTC_DCHECK_GE(num_important_packets, 0);
+  RTC_DCHECK_LE(num_important_packets, num_media_packets);
+  RTC_DCHECK(fec_packets->empty());
+  const size_t max_media_packets = fec_header_writer_->MaxMediaPackets();
+  if (num_media_packets > max_media_packets) {
+    RTC_LOG(LS_WARNING) << "Can't protect " << num_media_packets
+                        << " media packets per frame. Max is "
+                        << max_media_packets << ".";
+    return -1;
+  }
+
+  // Error check the media packets.
+  for (const auto& media_packet : media_packets) {
+    RTC_DCHECK(media_packet);
+    if (media_packet->data.size() < kRtpHeaderSize) {
+      RTC_LOG(LS_WARNING) << "Media packet " << media_packet->data.size()
+                          << " bytes "
+                             "is smaller than RTP header.";
+      return -1;
+    }
+    // Ensure the FEC packets will fit in a typical MTU.
+    if (media_packet->data.size() + MaxPacketOverhead() + kTransportOverhead >
+        IP_PACKET_SIZE) {
+      RTC_LOG(LS_WARNING) << "Media packet " << media_packet->data.size()
+                          << " bytes "
+                             "with overhead is larger than "
+                          << IP_PACKET_SIZE << " bytes.";
+    }
+  }
+
+  // Prepare generated FEC packets.
+  int num_fec_packets = NumFecPackets(num_media_packets, protection_factor);
+  if (num_fec_packets == 0) {
+    return 0;
+  }
+  for (int i = 0; i < num_fec_packets; ++i) {
+    generated_fec_packets_[i].data.EnsureCapacity(IP_PACKET_SIZE);
+    memset(generated_fec_packets_[i].data.MutableData(), 0, IP_PACKET_SIZE);
+    // Use this as a marker for untouched packets.
+    generated_fec_packets_[i].data.SetSize(0);
+    fec_packets->push_back(&generated_fec_packets_[i]);
+  }
+
+  internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
+  packet_mask_size_ = internal::PacketMaskSize(num_media_packets);
+  memset(packet_masks_, 0, num_fec_packets * packet_mask_size_);
+  internal::GeneratePacketMasks(num_media_packets, num_fec_packets,
+                                num_important_packets, use_unequal_protection,
+                                &mask_table, packet_masks_);
+
+  // Adapt packet masks to missing media packets.
+  int num_mask_bits = InsertZerosInPacketMasks(media_packets, num_fec_packets);
+  if (num_mask_bits < 0) {
+    RTC_LOG(LS_INFO) << "Due to sequence number gaps, cannot protect media "
+                        "packets with a single block of FEC packets.";
+    fec_packets->clear();
+    return -1;
+  }
+  packet_mask_size_ = internal::PacketMaskSize(num_mask_bits);
+
+  // Write FEC packets to `generated_fec_packets_`.
+  GenerateFecPayloads(media_packets, num_fec_packets);
+  // TODO(brandtr): Generalize this when multistream protection support is
+  // added.
+  const uint32_t media_ssrc = ParseSsrc(media_packets.front()->data.data());
+  const uint16_t seq_num_base =
+      ParseSequenceNumber(media_packets.front()->data.data());
+  FinalizeFecHeaders(num_fec_packets, media_ssrc, seq_num_base);
+
+  return 0;
+}
+
+int ForwardErrorCorrection::NumFecPackets(int num_media_packets,
+                                          int protection_factor) {
+  // Result in Q0 with an unsigned round.
+  int num_fec_packets = (num_media_packets * protection_factor + (1 << 7)) >> 8;
+  // Generate at least one FEC packet if we need protection.
+  if (protection_factor > 0 && num_fec_packets == 0) {
+    num_fec_packets = 1;
+  }
+  RTC_DCHECK_LE(num_fec_packets, num_media_packets);
+  return num_fec_packets;
+}
+
+void ForwardErrorCorrection::GenerateFecPayloads(
+    const PacketList& media_packets,
+    size_t num_fec_packets) {
+  RTC_DCHECK(!media_packets.empty());
+  for (size_t i = 0; i < num_fec_packets; ++i) {
+    Packet* const fec_packet = &generated_fec_packets_[i];
+    size_t pkt_mask_idx = i * packet_mask_size_;
+    const size_t min_packet_mask_size = fec_header_writer_->MinPacketMaskSize(
+        &packet_masks_[pkt_mask_idx], packet_mask_size_);
+    const size_t fec_header_size =
+        fec_header_writer_->FecHeaderSize(min_packet_mask_size);
+
+    size_t media_pkt_idx = 0;
+    auto media_packets_it = media_packets.cbegin();
+    uint16_t prev_seq_num =
+        ParseSequenceNumber((*media_packets_it)->data.data());
+    while (media_packets_it != media_packets.end()) {
+      Packet* const media_packet = media_packets_it->get();
+      // Should `media_packet` be protected by `fec_packet`?
+      if (packet_masks_[pkt_mask_idx] & (1 << (7 - media_pkt_idx))) {
+        size_t media_payload_length =
+            media_packet->data.size() - kRtpHeaderSize;
+
+        size_t fec_packet_length = fec_header_size + media_payload_length;
+        if (fec_packet_length > fec_packet->data.size()) {
+          size_t old_size = fec_packet->data.size();
+          fec_packet->data.SetSize(fec_packet_length);
+          memset(fec_packet->data.MutableData() + old_size, 0,
+                 fec_packet_length - old_size);
+        }
+        XorHeaders(*media_packet, fec_packet);
+        XorPayloads(*media_packet, media_payload_length, fec_header_size,
+                    fec_packet);
+      }
+      media_packets_it++;
+      if (media_packets_it != media_packets.end()) {
+        uint16_t seq_num =
+            ParseSequenceNumber((*media_packets_it)->data.data());
+        media_pkt_idx += static_cast<uint16_t>(seq_num - prev_seq_num);
+        prev_seq_num = seq_num;
+      }
+      pkt_mask_idx += media_pkt_idx / 8;
+      media_pkt_idx %= 8;
+    }
+    RTC_DCHECK_GT(fec_packet->data.size(), 0)
+        << "Packet mask is wrong or poorly designed.";
+  }
+}
+
+int ForwardErrorCorrection::InsertZerosInPacketMasks(
+    const PacketList& media_packets,
+    size_t num_fec_packets) {
+  size_t num_media_packets = media_packets.size();
+  if (num_media_packets <= 1) {
+    return num_media_packets;
+  }
+  uint16_t last_seq_num =
+      ParseSequenceNumber(media_packets.back()->data.data());
+  uint16_t first_seq_num =
+      ParseSequenceNumber(media_packets.front()->data.data());
+  size_t total_missing_seq_nums =
+      static_cast<uint16_t>(last_seq_num - first_seq_num) - num_media_packets +
+      1;
+  if (total_missing_seq_nums == 0) {
+    // All sequence numbers are covered by the packet mask.
+    // No zero insertion required.
+    return num_media_packets;
+  }
+  const size_t max_media_packets = fec_header_writer_->MaxMediaPackets();
+  if (total_missing_seq_nums + num_media_packets > max_media_packets) {
+    return -1;
+  }
+  // Allocate the new mask.
+  size_t tmp_packet_mask_size =
+      internal::PacketMaskSize(total_missing_seq_nums + num_media_packets);
+  memset(tmp_packet_masks_, 0, num_fec_packets * tmp_packet_mask_size);
+
+  auto media_packets_it = media_packets.cbegin();
+  uint16_t prev_seq_num = first_seq_num;
+  ++media_packets_it;
+
+  // Insert the first column.
+  internal::CopyColumn(tmp_packet_masks_, tmp_packet_mask_size, packet_masks_,
+                       packet_mask_size_, num_fec_packets, 0, 0);
+  size_t new_bit_index = 1;
+  size_t old_bit_index = 1;
+  // Insert zeros in the bit mask for every hole in the sequence.
+  while (media_packets_it != media_packets.end()) {
+    if (new_bit_index == max_media_packets) {
+      // We can only cover up to 48 packets.
+      break;
+    }
+    uint16_t seq_num = ParseSequenceNumber((*media_packets_it)->data.data());
+    const int num_zeros_to_insert =
+        static_cast<uint16_t>(seq_num - prev_seq_num - 1);
+    if (num_zeros_to_insert > 0) {
+      internal::InsertZeroColumns(num_zeros_to_insert, tmp_packet_masks_,
+                                  tmp_packet_mask_size, num_fec_packets,
+                                  new_bit_index);
+    }
+    new_bit_index += num_zeros_to_insert;
+    internal::CopyColumn(tmp_packet_masks_, tmp_packet_mask_size, packet_masks_,
+                         packet_mask_size_, num_fec_packets, new_bit_index,
+                         old_bit_index);
+    ++new_bit_index;
+    ++old_bit_index;
+    prev_seq_num = seq_num;
+    ++media_packets_it;
+  }
+  if (new_bit_index % 8 != 0) {
+    // We didn't fill the last byte. Shift bits to correct position.
+    for (uint16_t row = 0; row < num_fec_packets; ++row) {
+      int new_byte_index = row * tmp_packet_mask_size + new_bit_index / 8;
+      tmp_packet_masks_[new_byte_index] <<= (7 - (new_bit_index % 8));
+    }
+  }
+  // Replace the old mask with the new.
+  memcpy(packet_masks_, tmp_packet_masks_,
+         num_fec_packets * tmp_packet_mask_size);
+  return new_bit_index;
+}
+
+void ForwardErrorCorrection::FinalizeFecHeaders(size_t num_fec_packets,
+                                                uint32_t media_ssrc,
+                                                uint16_t seq_num_base) {
+  for (size_t i = 0; i < num_fec_packets; ++i) {
+    fec_header_writer_->FinalizeFecHeader(
+        media_ssrc, seq_num_base, &packet_masks_[i * packet_mask_size_],
+        packet_mask_size_, &generated_fec_packets_[i]);
+  }
+}
+
+void ForwardErrorCorrection::ResetState(
+    RecoveredPacketList* recovered_packets) {
+  // Free the memory for any existing recovered packets, if the caller hasn't.
+  recovered_packets->clear();
+  received_fec_packets_.clear();
+}
+
+void ForwardErrorCorrection::InsertMediaPacket(
+    RecoveredPacketList* recovered_packets,
+    const ReceivedPacket& received_packet) {
+  RTC_DCHECK_EQ(received_packet.ssrc, protected_media_ssrc_);
+
+  // Search for duplicate packets.
+  for (const auto& recovered_packet : *recovered_packets) {
+    RTC_DCHECK_EQ(recovered_packet->ssrc, received_packet.ssrc);
+    if (recovered_packet->seq_num == received_packet.seq_num) {
+      // Duplicate packet, no need to add to list.
+      return;
+    }
+  }
+
+  std::unique_ptr<RecoveredPacket> recovered_packet(new RecoveredPacket());
+  // This "recovered packet" was not recovered using parity packets.
+  recovered_packet->was_recovered = false;
+  // This media packet has already been passed on.
+  recovered_packet->returned = true;
+  recovered_packet->ssrc = received_packet.ssrc;
+  recovered_packet->seq_num = received_packet.seq_num;
+  recovered_packet->pkt = received_packet.pkt;
+  // TODO(holmer): Consider replacing this with a binary search for the right
+  // position, and then just insert the new packet. Would get rid of the sort.
+  RecoveredPacket* recovered_packet_ptr = recovered_packet.get();
+  recovered_packets->push_back(std::move(recovered_packet));
+  recovered_packets->sort(SortablePacket::LessThan());
+  UpdateCoveringFecPackets(*recovered_packet_ptr);
+}
+
+void ForwardErrorCorrection::UpdateCoveringFecPackets(
+    const RecoveredPacket& packet) {
+  for (auto& fec_packet : received_fec_packets_) {
+    // Is this FEC packet protecting the media packet `packet`?
+    auto protected_it = absl::c_lower_bound(
+        fec_packet->protected_packets, &packet, SortablePacket::LessThan());
+    if (protected_it != fec_packet->protected_packets.end() &&
+        (*protected_it)->seq_num == packet.seq_num) {
+      // Found an FEC packet which is protecting `packet`.
+      (*protected_it)->pkt = packet.pkt;
+    }
+  }
+}
+
+void ForwardErrorCorrection::InsertFecPacket(
+    const RecoveredPacketList& recovered_packets,
+    const ReceivedPacket& received_packet) {
+  RTC_DCHECK_EQ(received_packet.ssrc, ssrc_);
+
+  // Check for duplicate.
+  for (const auto& existing_fec_packet : received_fec_packets_) {
+    RTC_DCHECK_EQ(existing_fec_packet->ssrc, received_packet.ssrc);
+    if (existing_fec_packet->seq_num == received_packet.seq_num) {
+      // Drop duplicate FEC packet data.
+      return;
+    }
+  }
+
+  std::unique_ptr<ReceivedFecPacket> fec_packet(new ReceivedFecPacket());
+  fec_packet->pkt = received_packet.pkt;
+  fec_packet->ssrc = received_packet.ssrc;
+  fec_packet->seq_num = received_packet.seq_num;
+  // Parse ULPFEC/FlexFEC header specific info.
+  bool ret = fec_header_reader_->ReadFecHeader(fec_packet.get());
+  if (!ret) {
+    return;
+  }
+
+  // TODO(brandtr): Update here when we support multistream protection.
+  if (fec_packet->protected_ssrc != protected_media_ssrc_) {
+    RTC_LOG(LS_INFO)
+        << "Received FEC packet is protecting an unknown media SSRC; dropping.";
+    return;
+  }
+
+  if (fec_packet->packet_mask_offset + fec_packet->packet_mask_size >
+      fec_packet->pkt->data.size()) {
+    RTC_LOG(LS_INFO) << "Received corrupted FEC packet; dropping.";
+    return;
+  }
+
+  // Parse packet mask from header and represent as protected packets.
+  for (uint16_t byte_idx = 0; byte_idx < fec_packet->packet_mask_size;
+       ++byte_idx) {
+    uint8_t packet_mask =
+        fec_packet->pkt->data[fec_packet->packet_mask_offset + byte_idx];
+    for (uint16_t bit_idx = 0; bit_idx < 8; ++bit_idx) {
+      if (packet_mask & (1 << (7 - bit_idx))) {
+        std::unique_ptr<ProtectedPacket> protected_packet(
+            new ProtectedPacket());
+        // This wraps naturally with the sequence number.
+        protected_packet->ssrc = protected_media_ssrc_;
+        protected_packet->seq_num = static_cast<uint16_t>(
+            fec_packet->seq_num_base + (byte_idx << 3) + bit_idx);
+        protected_packet->pkt = nullptr;
+        fec_packet->protected_packets.push_back(std::move(protected_packet));
+      }
+    }
+  }
+
+  if (fec_packet->protected_packets.empty()) {
+    // All-zero packet mask; we can discard this FEC packet.
+    RTC_LOG(LS_WARNING) << "Received FEC packet has an all-zero packet mask.";
+  } else {
+    AssignRecoveredPackets(recovered_packets, fec_packet.get());
+    // TODO(holmer): Consider replacing this with a binary search for the right
+    // position, and then just insert the new packet. Would get rid of the sort.
+    received_fec_packets_.push_back(std::move(fec_packet));
+    received_fec_packets_.sort(SortablePacket::LessThan());
+    const size_t max_fec_packets = fec_header_reader_->MaxFecPackets();
+    if (received_fec_packets_.size() > max_fec_packets) {
+      received_fec_packets_.pop_front();
+    }
+    RTC_DCHECK_LE(received_fec_packets_.size(), max_fec_packets);
+  }
+}
+
+void ForwardErrorCorrection::AssignRecoveredPackets(
+    const RecoveredPacketList& recovered_packets,
+    ReceivedFecPacket* fec_packet) {
+  ProtectedPacketList* protected_packets = &fec_packet->protected_packets;
+  std::vector<RecoveredPacket*> recovered_protected_packets;
+
+  // Find intersection between the (sorted) containers `protected_packets`
+  // and `recovered_packets`, i.e. all protected packets that have already
+  // been recovered. Update the corresponding protected packets to point to
+  // the recovered packets.
+  auto it_p = protected_packets->cbegin();
+  auto it_r = recovered_packets.cbegin();
+  SortablePacket::LessThan less_than;
+  while (it_p != protected_packets->end() && it_r != recovered_packets.end()) {
+    if (less_than(*it_p, *it_r)) {
+      ++it_p;
+    } else if (less_than(*it_r, *it_p)) {
+      ++it_r;
+    } else {  // *it_p == *it_r.
+      // This protected packet has already been recovered.
+      (*it_p)->pkt = (*it_r)->pkt;
+      ++it_p;
+      ++it_r;
+    }
+  }
+}
+
+void ForwardErrorCorrection::InsertPacket(
+    const ReceivedPacket& received_packet,
+    RecoveredPacketList* recovered_packets) {
+  // Discard old FEC packets such that the sequence numbers in
+  // `received_fec_packets_` span at most 1/2 of the sequence number space.
+  // This is important for keeping `received_fec_packets_` sorted, and may
+  // also reduce the possibility of incorrect decoding due to sequence number
+  // wrap-around.
+  if (!received_fec_packets_.empty() &&
+      received_packet.ssrc == received_fec_packets_.front()->ssrc) {
+    // It only makes sense to detect wrap-around when `received_packet`
+    // and `front_received_fec_packet` belong to the same sequence number
+    // space, i.e., the same SSRC. This happens when `received_packet`
+    // is a FEC packet, or if `received_packet` is a media packet and
+    // RED+ULPFEC is used.
+    auto it = received_fec_packets_.begin();
+    while (it != received_fec_packets_.end()) {
+      uint16_t seq_num_diff = MinDiff(received_packet.seq_num, (*it)->seq_num);
+      if (seq_num_diff > kOldSequenceThreshold) {
+        it = received_fec_packets_.erase(it);
+      } else {
+        // No need to keep iterating, since `received_fec_packets_` is sorted.
+        break;
+      }
+    }
+  }
+
+  if (received_packet.is_fec) {
+    InsertFecPacket(*recovered_packets, received_packet);
+  } else {
+    InsertMediaPacket(recovered_packets, received_packet);
+  }
+
+  DiscardOldRecoveredPackets(recovered_packets);
+}
+
+bool ForwardErrorCorrection::StartPacketRecovery(
+    const ReceivedFecPacket& fec_packet,
+    RecoveredPacket* recovered_packet) {
+  // Ensure pkt is initialized.
+  recovered_packet->pkt = new Packet();
+  // Sanity check packet length.
+  if (fec_packet.pkt->data.size() <
+      fec_packet.fec_header_size + fec_packet.protection_length) {
+    RTC_LOG(LS_WARNING)
+        << "The FEC packet is truncated: it does not contain enough room "
+           "for its own header.";
+    return false;
+  }
+  if (fec_packet.protection_length >
+      std::min(size_t{IP_PACKET_SIZE - kRtpHeaderSize},
+               IP_PACKET_SIZE - fec_packet.fec_header_size)) {
+    RTC_LOG(LS_WARNING) << "Incorrect protection length, dropping FEC packet.";
+    return false;
+  }
+  // Initialize recovered packet data.
+  recovered_packet->pkt->data.EnsureCapacity(IP_PACKET_SIZE);
+  recovered_packet->pkt->data.SetSize(fec_packet.protection_length +
+                                      kRtpHeaderSize);
+  recovered_packet->returned = false;
+  recovered_packet->was_recovered = true;
+  // Copy bytes corresponding to minimum RTP header size.
+  // Note that the sequence number and SSRC fields will be overwritten
+  // at the end of packet recovery.
+  memcpy(recovered_packet->pkt->data.MutableData(),
+         fec_packet.pkt->data.cdata(), kRtpHeaderSize);
+  // Copy remaining FEC payload.
+  if (fec_packet.protection_length > 0) {
+    memcpy(recovered_packet->pkt->data.MutableData() + kRtpHeaderSize,
+           fec_packet.pkt->data.cdata() + fec_packet.fec_header_size,
+           fec_packet.protection_length);
+  }
+  return true;
+}
+
+bool ForwardErrorCorrection::FinishPacketRecovery(
+    const ReceivedFecPacket& fec_packet,
+    RecoveredPacket* recovered_packet) {
+  uint8_t* data = recovered_packet->pkt->data.MutableData();
+  // Set the RTP version to 2.
+  data[0] |= 0x80;  // Set the 1st bit.
+  data[0] &= 0xbf;  // Clear the 2nd bit.
+  // Recover the packet length, from temporary location.
+  const size_t new_size =
+      ByteReader<uint16_t>::ReadBigEndian(&data[2]) + kRtpHeaderSize;
+  if (new_size > size_t{IP_PACKET_SIZE - kRtpHeaderSize}) {
+    RTC_LOG(LS_WARNING) << "The recovered packet had a length larger than a "
+                           "typical IP packet, and is thus dropped.";
+    return false;
+  }
+  size_t old_size = recovered_packet->pkt->data.size();
+  recovered_packet->pkt->data.SetSize(new_size);
+  data = recovered_packet->pkt->data.MutableData();
+  if (new_size > old_size) {
+    memset(data + old_size, 0, new_size - old_size);
+  }
+
+  // Set the SN field.
+  ByteWriter<uint16_t>::WriteBigEndian(&data[2], recovered_packet->seq_num);
+  // Set the SSRC field.
+  ByteWriter<uint32_t>::WriteBigEndian(&data[8], fec_packet.protected_ssrc);
+  recovered_packet->ssrc = fec_packet.protected_ssrc;
+  return true;
+}
+
+void ForwardErrorCorrection::XorHeaders(const Packet& src, Packet* dst) {
+  uint8_t* dst_data = dst->data.MutableData();
+  const uint8_t* src_data = src.data.cdata();
+  // XOR the first 2 bytes of the header: V, P, X, CC, M, PT fields.
+  dst_data[0] ^= src_data[0];
+  dst_data[1] ^= src_data[1];
+
+  // XOR the length recovery field.
+  uint8_t src_payload_length_network_order[2];
+  ByteWriter<uint16_t>::WriteBigEndian(src_payload_length_network_order,
+                                       src.data.size() - kRtpHeaderSize);
+  dst_data[2] ^= src_payload_length_network_order[0];
+  dst_data[3] ^= src_payload_length_network_order[1];
+
+  // XOR the 5th to 8th bytes of the header: the timestamp field.
+  dst_data[4] ^= src_data[4];
+  dst_data[5] ^= src_data[5];
+  dst_data[6] ^= src_data[6];
+  dst_data[7] ^= src_data[7];
+
+  // Skip the 9th to 12th bytes of the header.
+}
+
+void ForwardErrorCorrection::XorPayloads(const Packet& src,
+                                         size_t payload_length,
+                                         size_t dst_offset,
+                                         Packet* dst) {
+  // XOR the payload.
+  RTC_DCHECK_LE(kRtpHeaderSize + payload_length, src.data.size());
+  RTC_DCHECK_LE(dst_offset + payload_length, dst->data.capacity());
+  if (dst_offset + payload_length > dst->data.size()) {
+    size_t old_size = dst->data.size();
+    size_t new_size = dst_offset + payload_length;
+    dst->data.SetSize(new_size);
+    memset(dst->data.MutableData() + old_size, 0, new_size - old_size);
+  }
+  uint8_t* dst_data = dst->data.MutableData();
+  const uint8_t* src_data = src.data.cdata();
+  for (size_t i = 0; i < payload_length; ++i) {
+    dst_data[dst_offset + i] ^= src_data[kRtpHeaderSize + i];
+  }
+}
+
+bool ForwardErrorCorrection::RecoverPacket(const ReceivedFecPacket& fec_packet,
+                                           RecoveredPacket* recovered_packet) {
+  if (!StartPacketRecovery(fec_packet, recovered_packet)) {
+    return false;
+  }
+  for (const auto& protected_packet : fec_packet.protected_packets) {
+    if (protected_packet->pkt == nullptr) {
+      // This is the packet we're recovering.
+      recovered_packet->seq_num = protected_packet->seq_num;
+    } else {
+      XorHeaders(*protected_packet->pkt, recovered_packet->pkt.get());
+      XorPayloads(*protected_packet->pkt,
+                  protected_packet->pkt->data.size() - kRtpHeaderSize,
+                  kRtpHeaderSize, recovered_packet->pkt.get());
+    }
+  }
+  if (!FinishPacketRecovery(fec_packet, recovered_packet)) {
+    return false;
+  }
+  return true;
+}
+
+void ForwardErrorCorrection::AttemptRecovery(
+    RecoveredPacketList* recovered_packets) {
+  auto fec_packet_it = received_fec_packets_.begin();
+  while (fec_packet_it != received_fec_packets_.end()) {
+    // Search for each FEC packet's protected media packets.
+    int packets_missing = NumCoveredPacketsMissing(**fec_packet_it);
+
+    // We can only recover one packet with an FEC packet.
+    if (packets_missing == 1) {
+      // Recovery possible.
+      std::unique_ptr<RecoveredPacket> recovered_packet(new RecoveredPacket());
+      recovered_packet->pkt = nullptr;
+      if (!RecoverPacket(**fec_packet_it, recovered_packet.get())) {
+        // Can't recover using this packet, drop it.
+        fec_packet_it = received_fec_packets_.erase(fec_packet_it);
+        continue;
+      }
+
+      auto* recovered_packet_ptr = recovered_packet.get();
+      // Add recovered packet to the list of recovered packets and update any
+      // FEC packets covering this packet with a pointer to the data.
+      // TODO(holmer): Consider replacing this with a binary search for the
+      // right position, and then just insert the new packet. Would get rid of
+      // the sort.
+      recovered_packets->push_back(std::move(recovered_packet));
+      recovered_packets->sort(SortablePacket::LessThan());
+      UpdateCoveringFecPackets(*recovered_packet_ptr);
+      DiscardOldRecoveredPackets(recovered_packets);
+      fec_packet_it = received_fec_packets_.erase(fec_packet_it);
+
+      // A packet has been recovered. We need to check the FEC list again, as
+      // this may allow additional packets to be recovered.
+      // Restart for first FEC packet.
+      fec_packet_it = received_fec_packets_.begin();
+    } else if (packets_missing == 0 ||
+               IsOldFecPacket(**fec_packet_it, recovered_packets)) {
+      // Either all protected packets arrived or have been recovered, or the FEC
+      // packet is old. We can discard this FEC packet.
+      fec_packet_it = received_fec_packets_.erase(fec_packet_it);
+    } else {
+      fec_packet_it++;
+    }
+  }
+}
+
+int ForwardErrorCorrection::NumCoveredPacketsMissing(
+    const ReceivedFecPacket& fec_packet) {
+  int packets_missing = 0;
+  for (const auto& protected_packet : fec_packet.protected_packets) {
+    if (protected_packet->pkt == nullptr) {
+      ++packets_missing;
+      if (packets_missing > 1) {
+        break;  // We can't recover more than one packet.
+      }
+    }
+  }
+  return packets_missing;
+}
+
+void ForwardErrorCorrection::DiscardOldRecoveredPackets(
+    RecoveredPacketList* recovered_packets) {
+  const size_t max_media_packets = fec_header_reader_->MaxMediaPackets();
+  while (recovered_packets->size() > max_media_packets) {
+    recovered_packets->pop_front();
+  }
+  RTC_DCHECK_LE(recovered_packets->size(), max_media_packets);
+}
+
+bool ForwardErrorCorrection::IsOldFecPacket(
+    const ReceivedFecPacket& fec_packet,
+    const RecoveredPacketList* recovered_packets) {
+  if (recovered_packets->empty()) {
+    return false;
+  }
+
+  const uint16_t back_recovered_seq_num = recovered_packets->back()->seq_num;
+  const uint16_t last_protected_seq_num =
+      fec_packet.protected_packets.back()->seq_num;
+
+  // FEC packet is old if its last protected sequence number is much
+  // older than the latest protected sequence number received.
+  return (MinDiff(back_recovered_seq_num, last_protected_seq_num) >
+          kOldSequenceThreshold);
+}
+
+uint16_t ForwardErrorCorrection::ParseSequenceNumber(const uint8_t* packet) {
+  return (packet[2] << 8) + packet[3];
+}
+
+uint32_t ForwardErrorCorrection::ParseSsrc(const uint8_t* packet) {
+  return (packet[8] << 24) + (packet[9] << 16) + (packet[10] << 8) + packet[11];
+}
+
+void ForwardErrorCorrection::DecodeFec(const ReceivedPacket& received_packet,
+                                       RecoveredPacketList* recovered_packets) {
+  RTC_DCHECK(recovered_packets);
+
+  const size_t max_media_packets = fec_header_reader_->MaxMediaPackets();
+  if (recovered_packets->size() == max_media_packets) {
+    const RecoveredPacket* back_recovered_packet =
+        recovered_packets->back().get();
+
+    if (received_packet.ssrc == back_recovered_packet->ssrc) {
+      const unsigned int seq_num_diff =
+          MinDiff(received_packet.seq_num, back_recovered_packet->seq_num);
+      if (seq_num_diff > max_media_packets) {
+        // A big gap in sequence numbers. The old recovered packets
+        // are now useless, so it's safe to do a reset.
+        RTC_LOG(LS_INFO) << "Big gap in media/ULPFEC sequence numbers. No need "
+                            "to keep the old packets in the FEC buffers, thus "
+                            "resetting them.";
+        ResetState(recovered_packets);
+      }
+    }
+  }
+
+  InsertPacket(received_packet, recovered_packets);
+  AttemptRecovery(recovered_packets);
+}
+
+size_t ForwardErrorCorrection::MaxPacketOverhead() const {
+  return fec_header_writer_->MaxPacketOverhead();
+}
+
+FecHeaderReader::FecHeaderReader(size_t max_media_packets,
+                                 size_t max_fec_packets)
+    : max_media_packets_(max_media_packets),
+      max_fec_packets_(max_fec_packets) {}
+
+FecHeaderReader::~FecHeaderReader() = default;
+
+size_t FecHeaderReader::MaxMediaPackets() const {
+  return max_media_packets_;
+}
+
+size_t FecHeaderReader::MaxFecPackets() const {
+  return max_fec_packets_;
+}
+
+FecHeaderWriter::FecHeaderWriter(size_t max_media_packets,
+                                 size_t max_fec_packets,
+                                 size_t max_packet_overhead)
+    : max_media_packets_(max_media_packets),
+      max_fec_packets_(max_fec_packets),
+      max_packet_overhead_(max_packet_overhead) {}
+
+FecHeaderWriter::~FecHeaderWriter() = default;
+
+size_t FecHeaderWriter::MaxMediaPackets() const {
+  return max_media_packets_;
+}
+
+size_t FecHeaderWriter::MaxFecPackets() const {
+  return max_fec_packets_;
+}
+
+size_t FecHeaderWriter::MaxPacketOverhead() const {
+  return max_packet_overhead_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction.h b/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction.h
new file mode 100644
index 0000000000..ac8f5361e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction.h
@@ -0,0 +1,424 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
+#define MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+#include <memory>
+#include <vector>
+
+#include "api/scoped_refptr.h"
+#include "api/units/timestamp.h"
+#include "modules/include/module_fec_types.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class FecHeaderReader;
+class FecHeaderWriter;
+
+// Performs codec-independent forward error correction (FEC), based on RFC 5109.
+// Option exists to enable unequal protection (UEP) across packets.
+// This is not to be confused with protection within packets
+// (referred to as uneven level protection (ULP) in RFC 5109).
+// TODO(brandtr): Split this class into a separate encoder
+// and a separate decoder.
+class ForwardErrorCorrection {
+ public:
+  // TODO(holmer): As a next step all these struct-like packet classes should be
+  // refactored into proper classes, and their members should be made private.
+  // This will require parts of the functionality in forward_error_correction.cc
+  // and receiver_fec.cc to be refactored into the packet classes.
+  class Packet {
+   public:
+    Packet();
+    virtual ~Packet();
+
+    // Add a reference.
+    virtual int32_t AddRef();
+
+    // Release a reference. Will delete the object if the reference count
+    // reaches zero.
+    virtual int32_t Release();
+
+    rtc::CopyOnWriteBuffer data;  // Packet data.
+
+   private:
+    int32_t ref_count_;  // Counts the number of references to a packet.
+  };
+
+  // TODO(holmer): Refactor into a proper class.
+  class SortablePacket {
+   public:
+    // Functor which returns true if the sequence number of `first`
+    // is < the sequence number of `second`. Should only ever be called for
+    // packets belonging to the same SSRC.
+    struct LessThan {
+      template <typename S, typename T>
+      bool operator()(const S& first, const T& second);
+    };
+
+    uint32_t ssrc;
+    uint16_t seq_num;
+  };
+
+  // Used for the input to DecodeFec().
+  class ReceivedPacket : public SortablePacket {
+   public:
+    ReceivedPacket();
+    ~ReceivedPacket();
+
+    bool is_fec;  // Set to true if this is an FEC packet and false
+                  // otherwise.
+    bool is_recovered;
+    RtpHeaderExtensionMap extensions;
+    rtc::scoped_refptr<Packet> pkt;  // Pointer to the packet storage.
+  };
+
+  // The recovered list parameter of DecodeFec() references structs of
+  // this type.
+  // TODO(holmer): Refactor into a proper class.
+  class RecoveredPacket : public SortablePacket {
+   public:
+    RecoveredPacket();
+    ~RecoveredPacket();
+
+    bool was_recovered;  // Will be true if this packet was recovered by
+                         // the FEC. Otherwise it was a media packet passed in
+                         // through the received packet list.
+    bool returned;  // True when the packet already has been returned to the
+                    // caller through the callback.
+    rtc::scoped_refptr<Packet> pkt;  // Pointer to the packet storage.
+  };
+
+  // Used to link media packets to their protecting FEC packets.
+  //
+  // TODO(holmer): Refactor into a proper class.
+  class ProtectedPacket : public SortablePacket {
+   public:
+    ProtectedPacket();
+    ~ProtectedPacket();
+
+    rtc::scoped_refptr<ForwardErrorCorrection::Packet> pkt;
+  };
+
+  using ProtectedPacketList = std::list<std::unique_ptr<ProtectedPacket>>;
+
+  // Used for internal storage of received FEC packets in a list.
+  //
+  // TODO(holmer): Refactor into a proper class.
+  class ReceivedFecPacket : public SortablePacket {
+   public:
+    ReceivedFecPacket();
+    ~ReceivedFecPacket();
+
+    // List of media packets that this FEC packet protects.
+    ProtectedPacketList protected_packets;
+    // RTP header fields.
+    uint32_t ssrc;
+    // FEC header fields.
+    size_t fec_header_size;
+    uint32_t protected_ssrc;
+    uint16_t seq_num_base;
+    size_t packet_mask_offset;  // Relative start of FEC header.
+    size_t packet_mask_size;
+    size_t protection_length;
+    // Raw data.
+    rtc::scoped_refptr<ForwardErrorCorrection::Packet> pkt;
+  };
+
+  using PacketList = std::list<std::unique_ptr<Packet>>;
+  using RecoveredPacketList = std::list<std::unique_ptr<RecoveredPacket>>;
+  using ReceivedFecPacketList = std::list<std::unique_ptr<ReceivedFecPacket>>;
+
+  ~ForwardErrorCorrection();
+
+  // Creates a ForwardErrorCorrection tailored for a specific FEC scheme.
+  static std::unique_ptr<ForwardErrorCorrection> CreateUlpfec(uint32_t ssrc);
+  static std::unique_ptr<ForwardErrorCorrection> CreateFlexfec(
+      uint32_t ssrc,
+      uint32_t protected_media_ssrc);
+
+  // Generates a list of FEC packets from supplied media packets.
+  //
+  // Input:  media_packets          List of media packets to protect, of type
+  //                                Packet. All packets must belong to the
+  //                                same frame and the list must not be empty.
+  // Input:  protection_factor      FEC protection overhead in the [0, 255]
+  //                                domain. To obtain 100% overhead, or an
+  //                                equal number of FEC packets as
+  //                                media packets, use 255.
+  // Input:  num_important_packets  The number of "important" packets in the
+  //                                frame. These packets may receive greater
+  //                                protection than the remaining packets.
+  //                                The important packets must be located at the
+  //                                start of the media packet list. For codecs
+  //                                with data partitioning, the important
+  //                                packets may correspond to first partition
+  //                                packets.
+  // Input:  use_unequal_protection Parameter to enable/disable unequal
+  //                                protection (UEP) across packets. Enabling
+  //                                UEP will allocate more protection to the
+  //                                num_important_packets from the start of the
+  //                                media_packets.
+  // Input:  fec_mask_type          The type of packet mask used in the FEC.
+  //                                Random or bursty type may be selected. The
+  //                                bursty type is only defined up to 12 media
+  //                                packets. If the number of media packets is
+  //                                above 12, the packet masks from the random
+  //                                table will be selected.
+  // Output: fec_packets            List of pointers to generated FEC packets,
+  //                                of type Packet. Must be empty on entry.
+  //                                The memory available through the list will
+  //                                be valid until the next call to
+  //                                EncodeFec().
+  //
+  // Returns 0 on success, -1 on failure.
+  //
+  int EncodeFec(const PacketList& media_packets,
+                uint8_t protection_factor,
+                int num_important_packets,
+                bool use_unequal_protection,
+                FecMaskType fec_mask_type,
+                std::list<Packet*>* fec_packets);
+
+  // Decodes a list of received media and FEC packets. It will parse the
+  // `received_packets`, storing FEC packets internally, and move
+  // media packets to `recovered_packets`. The recovered list will be
+  // sorted by ascending sequence number and have duplicates removed.
+  // The function should be called as new packets arrive, and
+  // `recovered_packets` will be progressively assembled with each call.
+  // When the function returns, `received_packets` will be empty.
+  //
+  // The caller will allocate packets submitted through `received_packets`.
+  // The function will handle allocation of recovered packets.
+  //
+  // Input:  received_packets   List of new received packets, of type
+  //                            ReceivedPacket, belonging to a single
+  //                            frame. At output the list will be empty,
+  //                            with packets either stored internally,
+  //                            or accessible through the recovered list.
+  // Output: recovered_packets  List of recovered media packets, of type
+  //                            RecoveredPacket, belonging to a single
+  //                            frame. The memory available through the
+  //                            list will be valid until the next call to
+  //                            DecodeFec().
+  //
+  void DecodeFec(const ReceivedPacket& received_packet,
+                 RecoveredPacketList* recovered_packets);
+
+  // Get the number of generated FEC packets, given the number of media packets
+  // and the protection factor.
+  static int NumFecPackets(int num_media_packets, int protection_factor);
+
+  // Gets the maximum size of the FEC headers in bytes, which must be
+  // accounted for as packet overhead.
+  size_t MaxPacketOverhead() const;
+
+  // Reset internal states from last frame and clear `recovered_packets`.
+  // Frees all memory allocated by this class.
+  void ResetState(RecoveredPacketList* recovered_packets);
+
+  // TODO(brandtr): Remove these functions when the Packet classes
+  // have been refactored.
+  static uint16_t ParseSequenceNumber(const uint8_t* packet);
+  static uint32_t ParseSsrc(const uint8_t* packet);
+
+ protected:
+  ForwardErrorCorrection(std::unique_ptr<FecHeaderReader> fec_header_reader,
+                         std::unique_ptr<FecHeaderWriter> fec_header_writer,
+                         uint32_t ssrc,
+                         uint32_t protected_media_ssrc);
+
+ private:
+  // Analyzes `media_packets` for holes in the sequence and inserts zero columns
+  // into the `packet_mask` where those holes are found. Zero columns means that
+  // those packets will have no protection.
+  // Returns the number of bits used for one row of the new packet mask.
+  // Requires that `packet_mask` has at least 6 * `num_fec_packets` bytes
+  // allocated.
+  int InsertZerosInPacketMasks(const PacketList& media_packets,
+                               size_t num_fec_packets);
+
+  // Writes FEC payloads and some recovery fields in the FEC headers.
+  void GenerateFecPayloads(const PacketList& media_packets,
+                           size_t num_fec_packets);
+
+  // Writes the FEC header fields that are not written by GenerateFecPayloads.
+  // This includes writing the packet masks.
+  void FinalizeFecHeaders(size_t num_fec_packets,
+                          uint32_t media_ssrc,
+                          uint16_t seq_num_base);
+
+  // Inserts the `received_packet` into the internal received FEC packet list
+  // or into `recovered_packets`.
+  void InsertPacket(const ReceivedPacket& received_packet,
+                    RecoveredPacketList* recovered_packets);
+
+  // Inserts the `received_packet` into `recovered_packets`. Deletes duplicates.
+  void InsertMediaPacket(RecoveredPacketList* recovered_packets,
+                         const ReceivedPacket& received_packet);
+
+  // Assigns pointers to the recovered packet from all FEC packets which cover
+  // it.
+  // Note: This reduces the complexity when we want to try to recover a packet
+  // since we don't have to find the intersection between recovered packets and
+  // packets covered by the FEC packet.
+  void UpdateCoveringFecPackets(const RecoveredPacket& packet);
+
+  // Insert `received_packet` into internal FEC list. Deletes duplicates.
+  void InsertFecPacket(const RecoveredPacketList& recovered_packets,
+                       const ReceivedPacket& received_packet);
+
+  // Assigns pointers to already recovered packets covered by `fec_packet`.
+  static void AssignRecoveredPackets(
+      const RecoveredPacketList& recovered_packets,
+      ReceivedFecPacket* fec_packet);
+
+  // Attempt to recover missing packets, using the internally stored
+  // received FEC packets.
+  void AttemptRecovery(RecoveredPacketList* recovered_packets);
+
+  // Initializes headers and payload before the XOR operation
+  // that recovers a packet.
+  static bool StartPacketRecovery(const ReceivedFecPacket& fec_packet,
+                                  RecoveredPacket* recovered_packet);
+
+  // Performs XOR between the first 8 bytes of `src` and `dst` and stores
+  // the result in `dst`. The 3rd and 4th bytes are used for storing
+  // the length recovery field.
+  static void XorHeaders(const Packet& src, Packet* dst);
+
+  // Performs XOR between the payloads of `src` and `dst` and stores the result
+  // in `dst`. The parameter `dst_offset` determines at  what byte the
+  // XOR operation starts in `dst`. In total, `payload_length` bytes are XORed.
+  static void XorPayloads(const Packet& src,
+                          size_t payload_length,
+                          size_t dst_offset,
+                          Packet* dst);
+
+  // Finalizes recovery of packet by setting RTP header fields.
+  // This is not specific to the FEC scheme used.
+  static bool FinishPacketRecovery(const ReceivedFecPacket& fec_packet,
+                                   RecoveredPacket* recovered_packet);
+
+  // Recover a missing packet.
+  static bool RecoverPacket(const ReceivedFecPacket& fec_packet,
+                            RecoveredPacket* recovered_packet);
+
+  // Get the number of missing media packets which are covered by `fec_packet`.
+  // An FEC packet can recover at most one packet, and if zero packets are
+  // missing the FEC packet can be discarded. This function returns 2 when two
+  // or more packets are missing.
+  static int NumCoveredPacketsMissing(const ReceivedFecPacket& fec_packet);
+
+  // Discards old packets in `recovered_packets`, which are no longer relevant
+  // for recovering lost packets.
+  void DiscardOldRecoveredPackets(RecoveredPacketList* recovered_packets);
+
+  // Checks if the FEC packet is old enough and no longer relevant for
+  // recovering lost media packets.
+  bool IsOldFecPacket(const ReceivedFecPacket& fec_packet,
+                      const RecoveredPacketList* recovered_packets);
+
+  // These SSRCs are only used by the decoder.
+  const uint32_t ssrc_;
+  const uint32_t protected_media_ssrc_;
+
+  std::unique_ptr<FecHeaderReader> fec_header_reader_;
+  std::unique_ptr<FecHeaderWriter> fec_header_writer_;
+
+  std::vector<Packet> generated_fec_packets_;
+  ReceivedFecPacketList received_fec_packets_;
+
+  // Arrays used to avoid dynamically allocating memory when generating
+  // the packet masks.
+  // (There are never more than `kUlpfecMaxMediaPackets` FEC packets generated.)
+  uint8_t packet_masks_[kUlpfecMaxMediaPackets * kUlpfecMaxPacketMaskSize];
+  uint8_t tmp_packet_masks_[kUlpfecMaxMediaPackets * kUlpfecMaxPacketMaskSize];
+  size_t packet_mask_size_;
+};
+
+// Classes derived from FecHeader{Reader,Writer} encapsulate the
+// specifics of reading and writing FEC header for, e.g., ULPFEC
+// and FlexFEC.
+class FecHeaderReader {
+ public:
+  virtual ~FecHeaderReader();
+
+  // The maximum number of media packets that can be covered by one FEC packet.
+  size_t MaxMediaPackets() const;
+
+  // The maximum number of FEC packets that is supported, per call
+  // to ForwardErrorCorrection::EncodeFec().
+  size_t MaxFecPackets() const;
+
+  // Parses FEC header and stores information in ReceivedFecPacket members.
+  virtual bool ReadFecHeader(
+      ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const = 0;
+
+ protected:
+  FecHeaderReader(size_t max_media_packets, size_t max_fec_packets);
+
+  const size_t max_media_packets_;
+  const size_t max_fec_packets_;
+};
+
+class FecHeaderWriter {
+ public:
+  virtual ~FecHeaderWriter();
+
+  // The maximum number of media packets that can be covered by one FEC packet.
+  size_t MaxMediaPackets() const;
+
+  // The maximum number of FEC packets that is supported, per call
+  // to ForwardErrorCorrection::EncodeFec().
+  size_t MaxFecPackets() const;
+
+  // The maximum overhead (in bytes) per packet, due to FEC headers.
+  size_t MaxPacketOverhead() const;
+
+  // Calculates the minimum packet mask size needed (in bytes),
+  // given the discrete options of the ULPFEC masks and the bits
+  // set in the current packet mask.
+  virtual size_t MinPacketMaskSize(const uint8_t* packet_mask,
+                                   size_t packet_mask_size) const = 0;
+
+  // The header size (in bytes), given the packet mask size.
+  virtual size_t FecHeaderSize(size_t packet_mask_size) const = 0;
+
+  // Writes FEC header.
+  virtual void FinalizeFecHeader(
+      uint32_t media_ssrc,
+      uint16_t seq_num_base,
+      const uint8_t* packet_mask,
+      size_t packet_mask_size,
+      ForwardErrorCorrection::Packet* fec_packet) const = 0;
+
+ protected:
+  FecHeaderWriter(size_t max_media_packets,
+                  size_t max_fec_packets,
+                  size_t max_packet_overhead);
+
+  const size_t max_media_packets_;
+  const size_t max_fec_packets_;
+  const size_t max_packet_overhead_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
new file mode 100644
index 0000000000..ac68162d26
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
@@ -0,0 +1,519 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+
+#include <string.h>
+
+#include <algorithm>
+
+#include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
+#include "modules/rtp_rtcp/source/fec_private_tables_random.h"
+#include "rtc_base/checks.h"
+
+namespace {
+// Allow for different modes of protection for packets in UEP case.
+enum ProtectionMode {
+  kModeNoOverlap,
+  kModeOverlap,
+  kModeBiasFirstPacket,
+};
+
+// Fits an input mask (sub_mask) to an output mask.
+// The mask is a matrix where the rows are the FEC packets,
+// and the columns are the source packets the FEC is applied to.
+// Each row of the mask is represented by a number of mask bytes.
+//
+// \param[in]  num_mask_bytes     The number of mask bytes of output mask.
+// \param[in]  num_sub_mask_bytes The number of mask bytes of input mask.
+// \param[in]  num_rows           The number of rows of the input mask.
+// \param[in]  sub_mask           A pointer to hold the input mask, of size
+//                                [0, num_rows * num_sub_mask_bytes]
+// \param[out] packet_mask        A pointer to hold the output mask, of size
+//                                [0, x * num_mask_bytes], where x >= num_rows.
+void FitSubMask(int num_mask_bytes,
+                int num_sub_mask_bytes,
+                int num_rows,
+                const uint8_t* sub_mask,
+                uint8_t* packet_mask) {
+  if (num_mask_bytes == num_sub_mask_bytes) {
+    memcpy(packet_mask, sub_mask, num_rows * num_sub_mask_bytes);
+  } else {
+    for (int i = 0; i < num_rows; ++i) {
+      int pkt_mask_idx = i * num_mask_bytes;
+      int pkt_mask_idx2 = i * num_sub_mask_bytes;
+      for (int j = 0; j < num_sub_mask_bytes; ++j) {
+        packet_mask[pkt_mask_idx] = sub_mask[pkt_mask_idx2];
+        pkt_mask_idx++;
+        pkt_mask_idx2++;
+      }
+    }
+  }
+}
+
+// Shifts a mask by number of columns (bits), and fits it to an output mask.
+// The mask is a matrix where the rows are the FEC packets,
+// and the columns are the source packets the FEC is applied to.
+// Each row of the mask is represented by a number of mask bytes.
+//
+// \param[in]  num_mask_bytes     The number of mask bytes of output mask.
+// \param[in]  num_sub_mask_bytes The number of mask bytes of input mask.
+// \param[in]  num_column_shift   The number columns to be shifted, and
+//                                the starting row for the output mask.
+// \param[in]  end_row            The ending row for the output mask.
+// \param[in]  sub_mask           A pointer to hold the input mask, of size
+//                                [0, (end_row_fec - start_row_fec) *
+//                                    num_sub_mask_bytes]
+// \param[out] packet_mask        A pointer to hold the output mask, of size
+//                                [0, x * num_mask_bytes],
+//                                where x >= end_row_fec.
+// TODO(marpan): This function is doing three things at the same time:
+// shift within a byte, byte shift and resizing.
+// Split up into subroutines.
+void ShiftFitSubMask(int num_mask_bytes,
+                     int res_mask_bytes,
+                     int num_column_shift,
+                     int end_row,
+                     const uint8_t* sub_mask,
+                     uint8_t* packet_mask) {
+  // Number of bit shifts within a byte
+  const int num_bit_shifts = (num_column_shift % 8);
+  const int num_byte_shifts = num_column_shift >> 3;
+
+  // Modify new mask with sub-mask21.
+
+  // Loop over the remaining FEC packets.
+  for (int i = num_column_shift; i < end_row; ++i) {
+    // Byte index of new mask, for row i and column res_mask_bytes,
+    // offset by the number of bytes shifts
+    int pkt_mask_idx =
+        i * num_mask_bytes + res_mask_bytes - 1 + num_byte_shifts;
+    // Byte index of sub_mask, for row i and column res_mask_bytes
+    int pkt_mask_idx2 =
+        (i - num_column_shift) * res_mask_bytes + res_mask_bytes - 1;
+
+    uint8_t shift_right_curr_byte = 0;
+    uint8_t shift_left_prev_byte = 0;
+    uint8_t comb_new_byte = 0;
+
+    // Handle case of num_mask_bytes > res_mask_bytes:
+    // For a given row, copy the rightmost "numBitShifts" bits
+    // of the last byte of sub_mask into output mask.
+    if (num_mask_bytes > res_mask_bytes) {
+      shift_left_prev_byte = (sub_mask[pkt_mask_idx2] << (8 - num_bit_shifts));
+      packet_mask[pkt_mask_idx + 1] = shift_left_prev_byte;
+    }
+
+    // For each row i (FEC packet), shift the bit-mask of the sub_mask.
+    // Each row of the mask contains "resMaskBytes" of bytes.
+    // We start from the last byte of the sub_mask and move to first one.
+    for (int j = res_mask_bytes - 1; j > 0; j--) {
+      // Shift current byte of sub21 to the right by "numBitShifts".
+      shift_right_curr_byte = sub_mask[pkt_mask_idx2] >> num_bit_shifts;
+
+      // Fill in shifted bits with bits from the previous (left) byte:
+      // First shift the previous byte to the left by "8-numBitShifts".
+      shift_left_prev_byte =
+          (sub_mask[pkt_mask_idx2 - 1] << (8 - num_bit_shifts));
+
+      // Then combine both shifted bytes into new mask byte.
+      comb_new_byte = shift_right_curr_byte | shift_left_prev_byte;
+
+      // Assign to new mask.
+      packet_mask[pkt_mask_idx] = comb_new_byte;
+      pkt_mask_idx--;
+      pkt_mask_idx2--;
+    }
+    // For the first byte in the row (j=0 case).
+    shift_right_curr_byte = sub_mask[pkt_mask_idx2] >> num_bit_shifts;
+    packet_mask[pkt_mask_idx] = shift_right_curr_byte;
+  }
+}
+
+}  // namespace
+
+namespace webrtc {
+namespace internal {
+
+PacketMaskTable::PacketMaskTable(FecMaskType fec_mask_type,
+                                 int num_media_packets)
+    : table_(PickTable(fec_mask_type, num_media_packets)) {}
+
+PacketMaskTable::~PacketMaskTable() = default;
+
+rtc::ArrayView<const uint8_t> PacketMaskTable::LookUp(int num_media_packets,
+                                                      int num_fec_packets) {
+  RTC_DCHECK_GT(num_media_packets, 0);
+  RTC_DCHECK_GT(num_fec_packets, 0);
+  RTC_DCHECK_LE(num_media_packets, kUlpfecMaxMediaPackets);
+  RTC_DCHECK_LE(num_fec_packets, num_media_packets);
+
+  if (num_media_packets <= 12) {
+    return LookUpInFecTable(table_, num_media_packets - 1, num_fec_packets - 1);
+  }
+  int mask_length =
+      static_cast<int>(PacketMaskSize(static_cast<size_t>(num_media_packets)));
+
+  // Generate FEC code mask for {num_media_packets(M), num_fec_packets(N)} (use
+  // N FEC packets to protect M media packets) In the mask, each FEC packet
+  // occupies one row, each bit / coloumn represent one media packet. E.g. Row
+  // A, Col/Bit B is set to 1, means FEC packet A will have protection for media
+  // packet B.
+
+  // Loop through each fec packet.
+  for (int row = 0; row < num_fec_packets; row++) {
+    // Loop through each fec code in a row, one code has 8 bits.
+    // Bit X will be set to 1 if media packet X shall be protected by current
+    // FEC packet. In this implementation, the protection is interleaved, thus
+    // media packet X will be protected by FEC packet (X % N)
+    for (int col = 0; col < mask_length; col++) {
+      fec_packet_mask_[row * mask_length + col] =
+          ((col * 8) % num_fec_packets == row && (col * 8) < num_media_packets
+               ? 0x80
+               : 0x00) |
+          ((col * 8 + 1) % num_fec_packets == row &&
+                   (col * 8 + 1) < num_media_packets
+               ? 0x40
+               : 0x00) |
+          ((col * 8 + 2) % num_fec_packets == row &&
+                   (col * 8 + 2) < num_media_packets
+               ? 0x20
+               : 0x00) |
+          ((col * 8 + 3) % num_fec_packets == row &&
+                   (col * 8 + 3) < num_media_packets
+               ? 0x10
+               : 0x00) |
+          ((col * 8 + 4) % num_fec_packets == row &&
+                   (col * 8 + 4) < num_media_packets
+               ? 0x08
+               : 0x00) |
+          ((col * 8 + 5) % num_fec_packets == row &&
+                   (col * 8 + 5) < num_media_packets
+               ? 0x04
+               : 0x00) |
+          ((col * 8 + 6) % num_fec_packets == row &&
+                   (col * 8 + 6) < num_media_packets
+               ? 0x02
+               : 0x00) |
+          ((col * 8 + 7) % num_fec_packets == row &&
+                   (col * 8 + 7) < num_media_packets
+               ? 0x01
+               : 0x00);
+    }
+  }
+  return {&fec_packet_mask_[0],
+          static_cast<size_t>(num_fec_packets * mask_length)};
+}
+
+// If `num_media_packets` is larger than the maximum allowed by `fec_mask_type`
+// for the bursty type, or the random table is explicitly asked for, then the
+// random type is selected. Otherwise the bursty table callback is returned.
+const uint8_t* PacketMaskTable::PickTable(FecMaskType fec_mask_type,
+                                          int num_media_packets) {
+  RTC_DCHECK_GE(num_media_packets, 0);
+  RTC_DCHECK_LE(static_cast<size_t>(num_media_packets), kUlpfecMaxMediaPackets);
+
+  if (fec_mask_type != kFecMaskRandom &&
+      num_media_packets <=
+          static_cast<int>(fec_private_tables::kPacketMaskBurstyTbl[0])) {
+    return &fec_private_tables::kPacketMaskBurstyTbl[0];
+  }
+
+  return &fec_private_tables::kPacketMaskRandomTbl[0];
+}
+
+// Remaining protection after important (first partition) packet protection
+void RemainingPacketProtection(int num_media_packets,
+                               int num_fec_remaining,
+                               int num_fec_for_imp_packets,
+                               int num_mask_bytes,
+                               ProtectionMode mode,
+                               uint8_t* packet_mask,
+                               PacketMaskTable* mask_table) {
+  if (mode == kModeNoOverlap) {
+    // sub_mask21
+
+    const int res_mask_bytes =
+        PacketMaskSize(num_media_packets - num_fec_for_imp_packets);
+
+    auto end_row = (num_fec_for_imp_packets + num_fec_remaining);
+    rtc::ArrayView<const uint8_t> packet_mask_sub_21 = mask_table->LookUp(
+        num_media_packets - num_fec_for_imp_packets, num_fec_remaining);
+
+    ShiftFitSubMask(num_mask_bytes, res_mask_bytes, num_fec_for_imp_packets,
+                    end_row, &packet_mask_sub_21[0], packet_mask);
+
+  } else if (mode == kModeOverlap || mode == kModeBiasFirstPacket) {
+    // sub_mask22
+    rtc::ArrayView<const uint8_t> packet_mask_sub_22 =
+        mask_table->LookUp(num_media_packets, num_fec_remaining);
+
+    FitSubMask(num_mask_bytes, num_mask_bytes, num_fec_remaining,
+               &packet_mask_sub_22[0],
+               &packet_mask[num_fec_for_imp_packets * num_mask_bytes]);
+
+    if (mode == kModeBiasFirstPacket) {
+      for (int i = 0; i < num_fec_remaining; ++i) {
+        int pkt_mask_idx = i * num_mask_bytes;
+        packet_mask[pkt_mask_idx] = packet_mask[pkt_mask_idx] | (1 << 7);
+      }
+    }
+  } else {
+    RTC_DCHECK_NOTREACHED();
+  }
+}
+
+// Protection for important (first partition) packets
+void ImportantPacketProtection(int num_fec_for_imp_packets,
+                               int num_imp_packets,
+                               int num_mask_bytes,
+                               uint8_t* packet_mask,
+                               PacketMaskTable* mask_table) {
+  const int num_imp_mask_bytes = PacketMaskSize(num_imp_packets);
+
+  // Get sub_mask1 from table
+  rtc::ArrayView<const uint8_t> packet_mask_sub_1 =
+      mask_table->LookUp(num_imp_packets, num_fec_for_imp_packets);
+
+  FitSubMask(num_mask_bytes, num_imp_mask_bytes, num_fec_for_imp_packets,
+             &packet_mask_sub_1[0], packet_mask);
+}
+
+// This function sets the protection allocation: i.e., how many FEC packets
+// to use for num_imp (1st partition) packets, given the: number of media
+// packets, number of FEC packets, and number of 1st partition packets.
+int SetProtectionAllocation(int num_media_packets,
+                            int num_fec_packets,
+                            int num_imp_packets) {
+  // TODO(marpan): test different cases for protection allocation:
+
+  // Use at most (alloc_par * num_fec_packets) for important packets.
+  float alloc_par = 0.5;
+  int max_num_fec_for_imp = alloc_par * num_fec_packets;
+
+  int num_fec_for_imp_packets = (num_imp_packets < max_num_fec_for_imp)
+                                    ? num_imp_packets
+                                    : max_num_fec_for_imp;
+
+  // Fall back to equal protection in this case
+  if (num_fec_packets == 1 && (num_media_packets > 2 * num_imp_packets)) {
+    num_fec_for_imp_packets = 0;
+  }
+
+  return num_fec_for_imp_packets;
+}
+
+// Modification for UEP: reuse the off-line tables for the packet masks.
+// Note: these masks were designed for equal packet protection case,
+// assuming random packet loss.
+
+// Current version has 3 modes (options) to build UEP mask from existing ones.
+// Various other combinations may be added in future versions.
+// Longer-term, we may add another set of tables specifically for UEP cases.
+// TODO(marpan): also consider modification of masks for bursty loss cases.
+
+// Mask is characterized as (#packets_to_protect, #fec_for_protection).
+// Protection factor defined as: (#fec_for_protection / #packets_to_protect).
+
+// Let k=num_media_packets, n=total#packets, (n-k)=num_fec_packets,
+// m=num_imp_packets.
+
+// For ProtectionMode 0 and 1:
+// one mask (sub_mask1) is used for 1st partition packets,
+// the other mask (sub_mask21/22, for 0/1) is for the remaining FEC packets.
+
+// In both mode 0 and 1, the packets of 1st partition (num_imp_packets) are
+// treated equally important, and are afforded more protection than the
+// residual partition packets.
+
+// For num_imp_packets:
+// sub_mask1 = (m, t): protection = t/(m), where t=F(k,n-k,m).
+// t=F(k,n-k,m) is the number of packets used to protect first partition in
+// sub_mask1. This is determined from the function SetProtectionAllocation().
+
+// For the left-over protection:
+// Mode 0: sub_mask21 = (k-m,n-k-t): protection = (n-k-t)/(k-m)
+// mode 0 has no protection overlap between the two partitions.
+// For mode 0, we would typically set t = min(m, n-k).
+
+// Mode 1: sub_mask22 = (k, n-k-t), with protection (n-k-t)/(k)
+// mode 1 has protection overlap between the two partitions (preferred).
+
+// For ProtectionMode 2:
+// This gives 1st packet of list (which is 1st packet of 1st partition) more
+// protection. In mode 2, the equal protection mask (which is obtained from
+// mode 1 for t=0) is modified (more "1s" added in 1st column of packet mask)
+// to bias higher protection for the 1st source packet.
+
+// Protection Mode 2 may be extended for a sort of sliding protection
+// (i.e., vary the number/density of "1s" across columns) across packets.
+
+void UnequalProtectionMask(int num_media_packets,
+                           int num_fec_packets,
+                           int num_imp_packets,
+                           int num_mask_bytes,
+                           uint8_t* packet_mask,
+                           PacketMaskTable* mask_table) {
+  // Set Protection type and allocation
+  // TODO(marpan): test/update for best mode and some combinations thereof.
+
+  ProtectionMode mode = kModeOverlap;
+  int num_fec_for_imp_packets = 0;
+
+  if (mode != kModeBiasFirstPacket) {
+    num_fec_for_imp_packets = SetProtectionAllocation(
+        num_media_packets, num_fec_packets, num_imp_packets);
+  }
+
+  int num_fec_remaining = num_fec_packets - num_fec_for_imp_packets;
+  // Done with setting protection type and allocation
+
+  //
+  // Generate sub_mask1
+  //
+  if (num_fec_for_imp_packets > 0) {
+    ImportantPacketProtection(num_fec_for_imp_packets, num_imp_packets,
+                              num_mask_bytes, packet_mask, mask_table);
+  }
+
+  //
+  // Generate sub_mask2
+  //
+  if (num_fec_remaining > 0) {
+    RemainingPacketProtection(num_media_packets, num_fec_remaining,
+                              num_fec_for_imp_packets, num_mask_bytes, mode,
+                              packet_mask, mask_table);
+  }
+}
+
+// This algorithm is tailored to look up data in the `kPacketMaskRandomTbl` and
+// `kPacketMaskBurstyTbl` tables. These tables only cover fec code for up to 12
+// media packets. Starting from 13 media packets, the fec code will be generated
+// at runtime. The format of those arrays is that they're essentially a 3
+// dimensional array with the following dimensions: * media packet
+//   * Size for kPacketMaskRandomTbl: 12
+//   * Size for kPacketMaskBurstyTbl: 12
+// * fec index
+//   * Size for both random and bursty table increases from 1 to number of rows.
+//     (i.e. 1-48, or 1-12 respectively).
+// * Fec data (what actually gets returned)
+//   * Size for kPacketMaskRandomTbl: 2 bytes.
+//     * For all entries: 2 * fec index (1 based)
+//   * Size for kPacketMaskBurstyTbl: 2 bytes.
+//     * For all entries: 2 * fec index (1 based)
+rtc::ArrayView<const uint8_t> LookUpInFecTable(const uint8_t* table,
+                                               int media_packet_index,
+                                               int fec_index) {
+  RTC_DCHECK_LT(media_packet_index, table[0]);
+
+  // Skip over the table size.
+  const uint8_t* entry = &table[1];
+
+  uint8_t entry_size_increment = 2;  // 0-16 are 2 byte wide, then changes to 6.
+
+  // Hop over un-interesting array entries.
+  for (int i = 0; i < media_packet_index; ++i) {
+    if (i == 16)
+      entry_size_increment = 6;
+    uint8_t count = entry[0];
+    ++entry;  // skip over the count.
+    for (int j = 0; j < count; ++j) {
+      entry += entry_size_increment * (j + 1);  // skip over the data.
+    }
+  }
+
+  if (media_packet_index == 16)
+    entry_size_increment = 6;
+
+  RTC_DCHECK_LT(fec_index, entry[0]);
+  ++entry;  // Skip over the size.
+
+  // Find the appropriate data in the second dimension.
+
+  // Find the specific data we're looking for.
+  for (int i = 0; i < fec_index; ++i)
+    entry += entry_size_increment * (i + 1);  // skip over the data.
+
+  size_t size = entry_size_increment * (fec_index + 1);
+  return {&entry[0], size};
+}
+
+void GeneratePacketMasks(int num_media_packets,
+                         int num_fec_packets,
+                         int num_imp_packets,
+                         bool use_unequal_protection,
+                         PacketMaskTable* mask_table,
+                         uint8_t* packet_mask) {
+  RTC_DCHECK_GT(num_media_packets, 0);
+  RTC_DCHECK_GT(num_fec_packets, 0);
+  RTC_DCHECK_LE(num_fec_packets, num_media_packets);
+  RTC_DCHECK_LE(num_imp_packets, num_media_packets);
+  RTC_DCHECK_GE(num_imp_packets, 0);
+
+  const int num_mask_bytes = PacketMaskSize(num_media_packets);
+
+  // Equal-protection for these cases.
+  if (!use_unequal_protection || num_imp_packets == 0) {
+    // Retrieve corresponding mask table directly:for equal-protection case.
+    // Mask = (k,n-k), with protection factor = (n-k)/k,
+    // where k = num_media_packets, n=total#packets, (n-k)=num_fec_packets.
+    rtc::ArrayView<const uint8_t> mask =
+        mask_table->LookUp(num_media_packets, num_fec_packets);
+    memcpy(packet_mask, &mask[0], mask.size());
+  } else {  // UEP case
+    UnequalProtectionMask(num_media_packets, num_fec_packets, num_imp_packets,
+                          num_mask_bytes, packet_mask, mask_table);
+  }  // End of UEP modification
+}  // End of GetPacketMasks
+
+size_t PacketMaskSize(size_t num_sequence_numbers) {
+  RTC_DCHECK_LE(num_sequence_numbers, 8 * kUlpfecPacketMaskSizeLBitSet);
+  if (num_sequence_numbers > 8 * kUlpfecPacketMaskSizeLBitClear) {
+    return kUlpfecPacketMaskSizeLBitSet;
+  }
+  return kUlpfecPacketMaskSizeLBitClear;
+}
+
+void InsertZeroColumns(int num_zeros,
+                       uint8_t* new_mask,
+                       int new_mask_bytes,
+                       int num_fec_packets,
+                       int new_bit_index) {
+  for (uint16_t row = 0; row < num_fec_packets; ++row) {
+    const int new_byte_index = row * new_mask_bytes + new_bit_index / 8;
+    const int max_shifts = (7 - (new_bit_index % 8));
+    new_mask[new_byte_index] <<= std::min(num_zeros, max_shifts);
+  }
+}
+
+void CopyColumn(uint8_t* new_mask,
+                int new_mask_bytes,
+                uint8_t* old_mask,
+                int old_mask_bytes,
+                int num_fec_packets,
+                int new_bit_index,
+                int old_bit_index) {
+  RTC_CHECK_LT(new_bit_index, 8 * new_mask_bytes);
+
+  // Copy column from the old mask to the beginning of the new mask and shift it
+  // out from the old mask.
+  for (uint16_t row = 0; row < num_fec_packets; ++row) {
+    int new_byte_index = row * new_mask_bytes + new_bit_index / 8;
+    int old_byte_index = row * old_mask_bytes + old_bit_index / 8;
+    new_mask[new_byte_index] |= ((old_mask[old_byte_index] & 0x80) >> 7);
+    if (new_bit_index % 8 != 7) {
+      new_mask[new_byte_index] <<= 1;
+    }
+    old_mask[old_byte_index] <<= 1;
+  }
+}
+
+}  // namespace internal
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h b/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h
new file mode 100644
index 0000000000..31acf73e3e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_
+#define MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "modules/include/module_fec_types.h"
+
+namespace webrtc {
+
+// Maximum number of media packets that can be protected
+// by these packet masks.
+constexpr size_t kUlpfecMaxMediaPackets = 48;
+
+// Packet mask size in bytes (given L bit).
+constexpr size_t kUlpfecPacketMaskSizeLBitClear = 2;
+constexpr size_t kUlpfecPacketMaskSizeLBitSet = 6;
+
+// Packet code mask maximum length. kFECPacketMaskMaxSize = MaxNumFECPackets *
+// (kUlpfecMaxMediaPackets / 8), and MaxNumFECPackets is equal to maximum number
+// of media packets (kUlpfecMaxMediaPackets)
+constexpr size_t kFECPacketMaskMaxSize = 288;
+
+// Convenience constants.
+constexpr size_t kUlpfecMinPacketMaskSize = kUlpfecPacketMaskSizeLBitClear;
+constexpr size_t kUlpfecMaxPacketMaskSize = kUlpfecPacketMaskSizeLBitSet;
+
+namespace internal {
+
+class PacketMaskTable {
+ public:
+  PacketMaskTable(FecMaskType fec_mask_type, int num_media_packets);
+  ~PacketMaskTable();
+
+  rtc::ArrayView<const uint8_t> LookUp(int num_media_packets,
+                                       int num_fec_packets);
+
+ private:
+  static const uint8_t* PickTable(FecMaskType fec_mask_type,
+                                  int num_media_packets);
+  const uint8_t* table_;
+  uint8_t fec_packet_mask_[kFECPacketMaskMaxSize];
+};
+
+rtc::ArrayView<const uint8_t> LookUpInFecTable(const uint8_t* table,
+                                               int media_packet_index,
+                                               int fec_index);
+
+// Returns an array of packet masks. The mask of a single FEC packet
+// corresponds to a number of mask bytes. The mask indicates which
+// media packets should be protected by the FEC packet.
+
+// \param[in]  num_media_packets       The number of media packets to protect.
+//                                     [1, max_media_packets].
+// \param[in]  num_fec_packets         The number of FEC packets which will
+//                                     be generated. [1, num_media_packets].
+// \param[in]  num_imp_packets         The number of important packets.
+//                                     [0, num_media_packets].
+//                                     num_imp_packets = 0 is the equal
+//                                     protection scenario.
+// \param[in]  use_unequal_protection  Enables unequal protection: allocates
+//                                     more protection to the num_imp_packets.
+// \param[in]  mask_table              An instance of the `PacketMaskTable`
+//                                     class, which contains the type of FEC
+//                                     packet mask used, and a pointer to the
+//                                     corresponding packet masks.
+// \param[out] packet_mask             A pointer to hold the packet mask array,
+//                                     of size: num_fec_packets *
+//                                     "number of mask bytes".
+void GeneratePacketMasks(int num_media_packets,
+                         int num_fec_packets,
+                         int num_imp_packets,
+                         bool use_unequal_protection,
+                         PacketMaskTable* mask_table,
+                         uint8_t* packet_mask);
+
+// Returns the required packet mask size, given the number of sequence numbers
+// that will be covered.
+size_t PacketMaskSize(size_t num_sequence_numbers);
+
+// Inserts `num_zeros` zero columns into `new_mask` at position
+// `new_bit_index`. If the current byte of `new_mask` can't fit all zeros, the
+// byte will be filled with zeros from `new_bit_index`, but the next byte will
+// be untouched.
+void InsertZeroColumns(int num_zeros,
+                       uint8_t* new_mask,
+                       int new_mask_bytes,
+                       int num_fec_packets,
+                       int new_bit_index);
+
+// Copies the left most bit column from the byte pointed to by
+// `old_bit_index` in `old_mask` to the right most column of the byte pointed
+// to by `new_bit_index` in `new_mask`. `old_mask_bytes` and `new_mask_bytes`
+// represent the number of bytes used per row for each mask. `num_fec_packets`
+// represent the number of rows of the masks.
+// The copied bit is shifted out from `old_mask` and is shifted one step to
+// the left in `new_mask`. `new_mask` will contain "xxxx xxn0" after this
+// operation, where x are previously inserted bits and n is the new bit.
+void CopyColumn(uint8_t* new_mask,
+                int new_mask_bytes,
+                uint8_t* old_mask,
+                int old_mask_bytes,
+                int num_fec_packets,
+                int new_bit_index,
+                int old_bit_index);
+
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc
new file mode 100644
index 0000000000..65a0e4cbb4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/frame_transformer_factory_unittest.cc
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/frame_transformer_factory.h"
+
+#include <cstdio>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "api/call/transport.h"
+#include "call/video_receive_stream.h"
+#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
+#include "rtc_base/event.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_frame_transformer.h"
+
+namespace webrtc {
+namespace {
+
+using testing::NiceMock;
+using testing::Return;
+
+class MockTransformableVideoFrame
+    : public webrtc::TransformableVideoFrameInterface {
+ public:
+  MOCK_METHOD(rtc::ArrayView<const uint8_t>, GetData, (), (const override));
+  MOCK_METHOD(void, SetData, (rtc::ArrayView<const uint8_t> data), (override));
+  MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override));
+  MOCK_METHOD(uint32_t, GetSsrc, (), (const, override));
+  MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override));
+  MOCK_METHOD(TransformableFrameInterface::Direction,
+              GetDirection,
+              (),
+              (const, override));
+  MOCK_METHOD(bool, IsKeyFrame, (), (const, override));
+  MOCK_METHOD(std::vector<uint8_t>, GetAdditionalData, (), (const, override));
+  MOCK_METHOD(const webrtc::VideoFrameMetadata&,
+              GetMetadata,
+              (),
+              (const, override));
+  MOCK_METHOD(void,
+              SetMetadata,
+              (const webrtc::VideoFrameMetadata&),
+              (override));
+};
+
+TEST(FrameTransformerFactory, CloneVideoFrame) {
+  NiceMock<MockTransformableVideoFrame> original_frame;
+  uint8_t data[10];
+  std::fill_n(data, 10, 5);
+  rtc::ArrayView<uint8_t> data_view(data);
+  EXPECT_CALL(original_frame, GetData()).WillRepeatedly(Return(data_view));
+  auto cloned_frame = CloneVideoFrame(&original_frame);
+  EXPECT_EQ(cloned_frame->GetData().size(), 10u);
+  EXPECT_THAT(cloned_frame->GetData(), testing::Each(5u));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/leb128.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/leb128.cc
new file mode 100644
index 0000000000..131ec4b523
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/leb128.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2023 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/leb128.h"
+
+#include <cstdint>
+
+namespace webrtc {
+
+int Leb128Size(uint64_t value) {
+  int size = 0;
+  while (value >= 0x80) {
+    ++size;
+    value >>= 7;
+  }
+  return size + 1;
+}
+
+uint64_t ReadLeb128(const uint8_t*& read_at, const uint8_t* end) {
+  uint64_t value = 0;
+  int fill_bits = 0;
+  while (read_at != end && fill_bits < 64 - 7) {
+    uint8_t leb128_byte = *read_at;
+    value |= uint64_t{leb128_byte & 0x7Fu} << fill_bits;
+    ++read_at;
+    fill_bits += 7;
+    if ((leb128_byte & 0x80) == 0) {
+      return value;
+    }
+  }
+  // Read 9 bytes and didn't find the terminator byte. Check if 10th byte
+  // is that terminator, however to fit result into uint64_t it may carry only
+  // single bit.
+  if (read_at != end && *read_at <= 1) {
+    value |= uint64_t{*read_at} << fill_bits;
+    ++read_at;
+    return value;
+  }
+  // Failed to find terminator leb128 byte.
+  read_at = nullptr;
+  return 0;
+}
+
+int WriteLeb128(uint64_t value, uint8_t* buffer) {
+  int size = 0;
+  while (value >= 0x80) {
+    buffer[size] = 0x80 | (value & 0x7F);
+    ++size;
+    value >>= 7;
+  }
+  buffer[size] = value;
+  ++size;
+  return size;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/leb128.h b/third_party/libwebrtc/modules/rtp_rtcp/source/leb128.h
new file mode 100644
index 0000000000..6a793f23ed
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/leb128.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2023 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_LEB128_H_
+#define MODULES_RTP_RTCP_SOURCE_LEB128_H_
+
+#include <cstdint>
+
+namespace webrtc {
+
+// Returns number of bytes needed to store `value` in leb128 format.
+int Leb128Size(uint64_t value);
+
+// Reads leb128 encoded value and advance read_at by number of bytes consumed.
+// Sets read_at to nullptr on error.
+uint64_t ReadLeb128(const uint8_t*& read_at, const uint8_t* end);
+
+// Encodes `value` in leb128 format. Assumes buffer has size of at least
+// Leb128Size(value). Returns number of bytes consumed.
+int WriteLeb128(uint64_t value, uint8_t* buffer);
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_LEB128_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/leb128_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/leb128_unittest.cc
new file mode 100644
index 0000000000..dbabcb36f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/leb128_unittest.cc
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2023 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/leb128.h"
+
+#include <cstdint>
+#include <iterator>
+#include <limits>
+
+#include "api/array_view.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+
+TEST(Leb128Test, Size) {
+  EXPECT_EQ(Leb128Size(0), 1);
+  EXPECT_EQ(Leb128Size(0b0111'1111), 1);
+  EXPECT_EQ(Leb128Size(0b1000'0000), 2);
+  EXPECT_EQ(Leb128Size(std::numeric_limits<uint64_t>::max()), 10);
+}
+
+TEST(Leb128Test, ReadZero) {
+  const uint8_t one_byte[] = {0};
+  const uint8_t* read_at = one_byte;
+  EXPECT_EQ(ReadLeb128(read_at, std::end(one_byte)), uint64_t{0});
+  EXPECT_EQ(std::distance(read_at, std::end(one_byte)), 0);
+}
+
+TEST(Leb128Test, ReadOneByte) {
+  const uint8_t buffer[] = {0b0010'1100};
+  const uint8_t* read_at = buffer;
+  EXPECT_EQ(ReadLeb128(read_at, std::end(buffer)), uint64_t{0b0010'1100});
+  EXPECT_EQ(std::distance(read_at, std::end(buffer)), 0);
+}
+
+TEST(Leb128Test, ReadTwoByte) {
+  const uint8_t buffer[] = {0b1010'1100, 0b0111'0000};
+  const uint8_t* read_at = buffer;
+  EXPECT_EQ(ReadLeb128(read_at, std::end(buffer)),
+            uint64_t{0b111'0000'010'1100});
+  EXPECT_EQ(std::distance(read_at, std::end(buffer)), 0);
+}
+
+TEST(Leb128Test, ReadNearlyMaxValue1) {
+  const uint8_t buffer[] = {0xff, 0xff, 0xff, 0xff, 0xff,
+                            0xff, 0xff, 0xff, 0x7f};
+  const uint8_t* read_at = buffer;
+  EXPECT_EQ(ReadLeb128(read_at, std::end(buffer)),
+            uint64_t{0x7fff'ffff'ffff'ffff});
+  EXPECT_EQ(std::distance(read_at, std::end(buffer)), 0);
+}
+
+TEST(Leb128Test, ReadNearlyMaxValue2) {
+  // This is valid, though not optimal way to store 63 bits of the value.
+  const uint8_t buffer[] = {0xff, 0xff, 0xff, 0xff, 0xff,
+                            0xff, 0xff, 0xff, 0xff, 0x0};
+  const uint8_t* read_at = buffer;
+  EXPECT_EQ(ReadLeb128(read_at, std::end(buffer)),
+            uint64_t{0x7fff'ffff'ffff'ffff});
+  EXPECT_EQ(std::distance(read_at, std::end(buffer)), 0);
+}
+
+TEST(Leb128Test, ReadMaxValue) {
+  // This is valid, though not optimal way to store 63 bits of the value.
+  const uint8_t buffer[] = {0xff, 0xff, 0xff, 0xff, 0xff,
+                            0xff, 0xff, 0xff, 0xff, 0x1};
+  const uint8_t* read_at = buffer;
+  EXPECT_EQ(ReadLeb128(read_at, std::end(buffer)), 0xffff'ffff'ffff'ffff);
+  EXPECT_EQ(std::distance(read_at, std::end(buffer)), 0);
+}
+
+TEST(Leb128Test, FailsToReadMoreThanMaxValue) {
+  const uint8_t buffer[] = {0xff, 0xff, 0xff, 0xff, 0xff,
+                            0xff, 0xff, 0xff, 0xff, 0x2};
+  const uint8_t* read_at = buffer;
+  ReadLeb128(read_at, std::end(buffer));
+  EXPECT_EQ(read_at, nullptr);
+}
+
+TEST(Leb128Test, DoesntReadMoreThan10Bytes) {
+  // Though this array represent leb128 encoded value that can fit in uint64_t,
+  // ReadLeb128 function discards it to avoid reading too many bytes from the
+  // buffer.
+  const uint8_t buffer[] = {0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
+                            0xff, 0xff, 0xff, 0x80, 0x00};
+  const uint8_t* read_at = buffer;
+  ReadLeb128(read_at, std::end(buffer));
+  EXPECT_EQ(read_at, nullptr);
+}
+
+TEST(Leb128Test, WriteZero) {
+  uint8_t buffer[16];
+  EXPECT_EQ(WriteLeb128(0, buffer), 1);
+  EXPECT_EQ(buffer[0], 0);
+}
+
+TEST(Leb128Test, WriteOneByteValue) {
+  uint8_t buffer[16];
+  EXPECT_EQ(WriteLeb128(0b0010'1100, buffer), 1);
+  EXPECT_EQ(buffer[0], 0b0010'1100);
+}
+
+TEST(Leb128Test, WriteTwoByteValue) {
+  uint8_t buffer[16];
+  EXPECT_EQ(WriteLeb128(0b11'1111'010'1100, buffer), 2);
+  EXPECT_EQ(buffer[0], 0b1010'1100);
+  EXPECT_EQ(buffer[1], 0b0011'1111);
+}
+
+TEST(Leb128Test, WriteNearlyMaxValue) {
+  uint8_t buffer[16];
+  EXPECT_EQ(WriteLeb128(0x7fff'ffff'ffff'ffff, buffer), 9);
+  EXPECT_THAT(
+      rtc::MakeArrayView(buffer, 9),
+      ElementsAre(0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0x7f));
+}
+
+TEST(Leb128Test, WriteMaxValue) {
+  uint8_t buffer[16];
+  EXPECT_EQ(WriteLeb128(0xffff'ffff'ffff'ffff, buffer), 10);
+  EXPECT_THAT(
+      rtc::MakeArrayView(buffer, 10),
+      ElementsAre(0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0x01));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
new file mode 100644
index 0000000000..d0617f3804
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -0,0 +1,293 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <iterator>
+#include <list>
+#include <memory>
+#include <set>
+
+#include "absl/algorithm/container.h"
+#include "api/call/transport.h"
+#include "api/transport/field_trial_based_config.h"
+#include "call/rtp_stream_receiver_controller.h"
+#include "call/rtx_receive_stream.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
+#include "rtc_base/rate_limiter.h"
+#include "rtc_base/thread.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+const int kVideoNackListSize = 30;
+const uint32_t kTestSsrc = 3456;
+const uint32_t kTestRtxSsrc = kTestSsrc + 1;
+const uint16_t kTestSequenceNumber = 2345;
+const uint32_t kTestNumberOfPackets = 1350;
+const int kTestNumberOfRtxPackets = 149;
+const int kNumFrames = 30;
+const int kPayloadType = 123;
+const int kRtxPayloadType = 98;
+const int64_t kMaxRttMs = 1000;
+
+class VerifyingMediaStream : public RtpPacketSinkInterface {
+ public:
+  VerifyingMediaStream() {}
+
+  void OnRtpPacket(const RtpPacketReceived& packet) override {
+    if (!sequence_numbers_.empty())
+      EXPECT_EQ(kTestSsrc, packet.Ssrc());
+
+    sequence_numbers_.push_back(packet.SequenceNumber());
+  }
+  std::list<uint16_t> sequence_numbers_;
+};
+
+class RtxLoopBackTransport : public webrtc::Transport {
+ public:
+  explicit RtxLoopBackTransport(uint32_t rtx_ssrc)
+      : count_(0),
+        packet_loss_(0),
+        consecutive_drop_start_(0),
+        consecutive_drop_end_(0),
+        rtx_ssrc_(rtx_ssrc),
+        count_rtx_ssrc_(0),
+        module_(NULL) {}
+
+  void SetSendModule(RtpRtcpInterface* rtpRtcpModule) {
+    module_ = rtpRtcpModule;
+  }
+
+  void DropEveryNthPacket(int n) { packet_loss_ = n; }
+
+  void DropConsecutivePackets(int start, int total) {
+    consecutive_drop_start_ = start;
+    consecutive_drop_end_ = start + total;
+    packet_loss_ = 0;
+  }
+
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override {
+    count_++;
+    RtpPacketReceived packet;
+    if (!packet.Parse(data, len))
+      return false;
+    if (packet.Ssrc() == rtx_ssrc_) {
+      count_rtx_ssrc_++;
+    } else {
+      // For non-RTX packets only.
+      expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
+                                        packet.SequenceNumber());
+    }
+    if (packet_loss_ > 0) {
+      if ((count_ % packet_loss_) == 0) {
+        return true;
+      }
+    } else if (count_ >= consecutive_drop_start_ &&
+               count_ < consecutive_drop_end_) {
+      return true;
+    }
+    EXPECT_TRUE(stream_receiver_controller_.OnRtpPacket(packet));
+    return true;
+  }
+
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    module_->IncomingRtcpPacket((const uint8_t*)data, len);
+    return true;
+  }
+  int count_;
+  int packet_loss_;
+  int consecutive_drop_start_;
+  int consecutive_drop_end_;
+  uint32_t rtx_ssrc_;
+  int count_rtx_ssrc_;
+  RtpRtcpInterface* module_;
+  RtpStreamReceiverController stream_receiver_controller_;
+  std::set<uint16_t> expected_sequence_numbers_;
+};
+
+class RtpRtcpRtxNackTest : public ::testing::Test {
+ protected:
+  RtpRtcpRtxNackTest()
+      : rtp_rtcp_module_(nullptr),
+        transport_(kTestRtxSsrc),
+        rtx_stream_(&media_stream_, rtx_associated_payload_types_, kTestSsrc),
+        fake_clock(123456),
+        retransmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
+  ~RtpRtcpRtxNackTest() override {}
+
+  void SetUp() override {
+    RtpRtcpInterface::Configuration configuration;
+    configuration.audio = false;
+    configuration.clock = &fake_clock;
+    receive_statistics_ = ReceiveStatistics::Create(&fake_clock);
+    configuration.receive_statistics = receive_statistics_.get();
+    configuration.outgoing_transport = &transport_;
+    configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
+    configuration.local_media_ssrc = kTestSsrc;
+    configuration.rtx_send_ssrc = kTestRtxSsrc;
+    rtp_rtcp_module_ = ModuleRtpRtcpImpl2::Create(configuration);
+    FieldTrialBasedConfig field_trials;
+    RTPSenderVideo::Config video_config;
+    video_config.clock = &fake_clock;
+    video_config.rtp_sender = rtp_rtcp_module_->RtpSender();
+    video_config.field_trials = &field_trials;
+    rtp_sender_video_ = std::make_unique<RTPSenderVideo>(video_config);
+    rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
+    rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
+    EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));
+    rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber);
+    rtp_rtcp_module_->SetStartTimestamp(111111);
+
+    // Used for NACK processing.
+    rtp_rtcp_module_->SetRemoteSSRC(kTestSsrc);
+
+    rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
+    transport_.SetSendModule(rtp_rtcp_module_.get());
+    media_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
+        kTestSsrc, &media_stream_);
+
+    for (size_t n = 0; n < sizeof(payload_data); n++) {
+      payload_data[n] = n % 10;
+    }
+  }
+
+  int BuildNackList(uint16_t* nack_list) {
+    media_stream_.sequence_numbers_.sort();
+    std::list<uint16_t> missing_sequence_numbers;
+    std::list<uint16_t>::iterator it = media_stream_.sequence_numbers_.begin();
+
+    while (it != media_stream_.sequence_numbers_.end()) {
+      uint16_t sequence_number_1 = *it;
+      ++it;
+      if (it != media_stream_.sequence_numbers_.end()) {
+        uint16_t sequence_number_2 = *it;
+        // Add all missing sequence numbers to list
+        for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
+          missing_sequence_numbers.push_back(i);
+        }
+      }
+    }
+    int n = 0;
+    for (it = missing_sequence_numbers.begin();
+         it != missing_sequence_numbers.end(); ++it) {
+      nack_list[n++] = (*it);
+    }
+    return n;
+  }
+
+  bool ExpectedPacketsReceived() {
+    std::list<uint16_t> received_sorted;
+    absl::c_copy(media_stream_.sequence_numbers_,
+                 std::back_inserter(received_sorted));
+    received_sorted.sort();
+    return absl::c_equal(received_sorted,
+                         transport_.expected_sequence_numbers_);
+  }
+
+  void RunRtxTest(RtxMode rtx_method, int loss) {
+    rtx_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
+        kTestRtxSsrc, &rtx_stream_);
+    rtp_rtcp_module_->SetRtxSendStatus(rtx_method);
+    transport_.DropEveryNthPacket(loss);
+    uint32_t timestamp = 3000;
+    uint16_t nack_list[kVideoNackListSize];
+    for (int frame = 0; frame < kNumFrames; ++frame) {
+      RTPVideoHeader video_header;
+      EXPECT_TRUE(rtp_rtcp_module_->OnSendingRtpFrame(timestamp, timestamp / 90,
+                                                      kPayloadType, false));
+      video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+      EXPECT_TRUE(rtp_sender_video_->SendVideo(
+          kPayloadType, VideoCodecType::kVideoCodecGeneric, timestamp,
+          timestamp / 90, payload_data, video_header, 0));
+      // Min required delay until retransmit = 5 + RTT ms (RTT = 0).
+      fake_clock.AdvanceTimeMilliseconds(5);
+      int length = BuildNackList(nack_list);
+      if (length > 0)
+        rtp_rtcp_module_->SendNACK(nack_list, length);
+      fake_clock.AdvanceTimeMilliseconds(28);  //  33ms - 5ms delay.
+      // Prepare next frame.
+      timestamp += 3000;
+    }
+    media_stream_.sequence_numbers_.sort();
+  }
+
+  rtc::AutoThread main_thread_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_module_;
+  std::unique_ptr<RTPSenderVideo> rtp_sender_video_;
+  RtxLoopBackTransport transport_;
+  const std::map<int, int> rtx_associated_payload_types_ = {
+      {kRtxPayloadType, kPayloadType}};
+  VerifyingMediaStream media_stream_;
+  RtxReceiveStream rtx_stream_;
+  uint8_t payload_data[65000];
+  SimulatedClock fake_clock;
+  RateLimiter retransmission_rate_limiter_;
+  std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
+  std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
+};
+
+TEST_F(RtpRtcpRtxNackTest, LongNackList) {
+  const int kNumPacketsToDrop = 900;
+  const int kNumRequiredRtcp = 4;
+  uint32_t timestamp = 3000;
+  uint16_t nack_list[kNumPacketsToDrop];
+  // Disable StorePackets to be able to set a larger packet history.
+  rtp_rtcp_module_->SetStorePacketsStatus(false, 0);
+  // Enable StorePackets with a packet history of 2000 packets.
+  rtp_rtcp_module_->SetStorePacketsStatus(true, 2000);
+  // Drop 900 packets from the second one so that we get a NACK list which is
+  // big enough to require 4 RTCP packets to be fully transmitted to the sender.
+  transport_.DropConsecutivePackets(2, kNumPacketsToDrop);
+  // Send 30 frames which at the default size is roughly what we need to get
+  // enough packets.
+  for (int frame = 0; frame < kNumFrames; ++frame) {
+    RTPVideoHeader video_header;
+    EXPECT_TRUE(rtp_rtcp_module_->OnSendingRtpFrame(timestamp, timestamp / 90,
+                                                    kPayloadType, false));
+    video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+    EXPECT_TRUE(rtp_sender_video_->SendVideo(
+        kPayloadType, VideoCodecType::kVideoCodecGeneric, timestamp,
+        timestamp / 90, payload_data, video_header, 0));
+    // Prepare next frame.
+    timestamp += 3000;
+    fake_clock.AdvanceTimeMilliseconds(33);
+  }
+  EXPECT_FALSE(transport_.expected_sequence_numbers_.empty());
+  EXPECT_FALSE(media_stream_.sequence_numbers_.empty());
+  size_t last_receive_count = media_stream_.sequence_numbers_.size();
+  int length = BuildNackList(nack_list);
+  for (int i = 0; i < kNumRequiredRtcp - 1; ++i) {
+    rtp_rtcp_module_->SendNACK(nack_list, length);
+    EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
+    last_receive_count = media_stream_.sequence_numbers_.size();
+    EXPECT_FALSE(ExpectedPacketsReceived());
+  }
+  rtp_rtcp_module_->SendNACK(nack_list, length);
+  EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
+  EXPECT_TRUE(ExpectedPacketsReceived());
+}
+
+TEST_F(RtpRtcpRtxNackTest, RtxNack) {
+  RunRtxTest(kRtxRetransmitted, 10);
+  EXPECT_EQ(kTestSequenceNumber, *(media_stream_.sequence_numbers_.begin()));
+  EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
+            *(media_stream_.sequence_numbers_.rbegin()));
+  EXPECT_EQ(kTestNumberOfPackets, media_stream_.sequence_numbers_.size());
+  EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
+  EXPECT_TRUE(ExpectedPacketsReceived());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
new file mode 100644
index 0000000000..36f0a63d59
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/packet_loss_stats.h"
+
+#include <cstdint>
+#include <iterator>
+#include <vector>
+
+#include "rtc_base/checks.h"
+
+// After this many packets are added, adding additional packets will cause the
+// oldest packets to be pruned from the buffer.
+static const int kBufferSize = 100;
+
+namespace webrtc {
+
+PacketLossStats::PacketLossStats()
+    : single_loss_historic_count_(0),
+      multiple_loss_historic_event_count_(0),
+      multiple_loss_historic_packet_count_(0) {}
+
+PacketLossStats::~PacketLossStats() = default;
+
+void PacketLossStats::AddLostPacket(uint16_t sequence_number) {
+  // Detect sequence number wrap around.
+  if (!lost_packets_buffer_.empty() &&
+      static_cast<int>(*(lost_packets_buffer_.rbegin())) - sequence_number >
+          0x8000) {
+    // The buffer contains large numbers and this is a small number.
+    lost_packets_wrapped_buffer_.insert(sequence_number);
+  } else {
+    lost_packets_buffer_.insert(sequence_number);
+  }
+  if (lost_packets_wrapped_buffer_.size() + lost_packets_buffer_.size() >
+          kBufferSize ||
+      (!lost_packets_wrapped_buffer_.empty() &&
+       *(lost_packets_wrapped_buffer_.rbegin()) > 0x4000)) {
+    PruneBuffer();
+  }
+}
+
+int PacketLossStats::GetSingleLossCount() const {
+  int single_loss_count, unused1, unused2;
+  ComputeLossCounts(&single_loss_count, &unused1, &unused2);
+  return single_loss_count;
+}
+
+int PacketLossStats::GetMultipleLossEventCount() const {
+  int event_count, unused1, unused2;
+  ComputeLossCounts(&unused1, &event_count, &unused2);
+  return event_count;
+}
+
+int PacketLossStats::GetMultipleLossPacketCount() const {
+  int packet_count, unused1, unused2;
+  ComputeLossCounts(&unused1, &unused2, &packet_count);
+  return packet_count;
+}
+
+void PacketLossStats::ComputeLossCounts(
+    int* out_single_loss_count,
+    int* out_multiple_loss_event_count,
+    int* out_multiple_loss_packet_count) const {
+  *out_single_loss_count = single_loss_historic_count_;
+  *out_multiple_loss_event_count = multiple_loss_historic_event_count_;
+  *out_multiple_loss_packet_count = multiple_loss_historic_packet_count_;
+  if (lost_packets_buffer_.empty()) {
+    RTC_DCHECK(lost_packets_wrapped_buffer_.empty());
+    return;
+  }
+  uint16_t last_num = 0;
+  int sequential_count = 0;
+  std::vector<const std::set<uint16_t>*> buffers;
+  buffers.push_back(&lost_packets_buffer_);
+  buffers.push_back(&lost_packets_wrapped_buffer_);
+  for (const auto* buffer : buffers) {
+    for (auto it = buffer->begin(); it != buffer->end(); ++it) {
+      uint16_t current_num = *it;
+      if (sequential_count > 0 && current_num != ((last_num + 1) & 0xFFFF)) {
+        if (sequential_count == 1) {
+          (*out_single_loss_count)++;
+        } else {
+          (*out_multiple_loss_event_count)++;
+          *out_multiple_loss_packet_count += sequential_count;
+        }
+        sequential_count = 0;
+      }
+      sequential_count++;
+      last_num = current_num;
+    }
+  }
+  if (sequential_count == 1) {
+    (*out_single_loss_count)++;
+  } else if (sequential_count > 1) {
+    (*out_multiple_loss_event_count)++;
+    *out_multiple_loss_packet_count += sequential_count;
+  }
+}
+
+void PacketLossStats::PruneBuffer() {
+  // Remove the oldest lost packet and any contiguous packets and move them
+  // into the historic counts.
+  auto it = lost_packets_buffer_.begin();
+  uint16_t last_removed = 0;
+  int remove_count = 0;
+  // Count adjacent packets and continue counting if it is wrap around by
+  // swapping in the wrapped buffer and letting our value wrap as well.
+  while (remove_count == 0 || (!lost_packets_buffer_.empty() &&
+                               *it == ((last_removed + 1) & 0xFFFF))) {
+    last_removed = *it;
+    remove_count++;
+    auto to_erase = it++;
+    lost_packets_buffer_.erase(to_erase);
+    if (lost_packets_buffer_.empty()) {
+      lost_packets_buffer_.swap(lost_packets_wrapped_buffer_);
+      it = lost_packets_buffer_.begin();
+    }
+  }
+  if (remove_count > 1) {
+    multiple_loss_historic_event_count_++;
+    multiple_loss_historic_packet_count_ += remove_count;
+  } else {
+    single_loss_historic_count_++;
+  }
+  // Continue pruning if the wrapped buffer is beyond a threshold and there are
+  // things left in the pre-wrapped buffer.
+  if (!lost_packets_wrapped_buffer_.empty() &&
+      *(lost_packets_wrapped_buffer_.rbegin()) > 0x4000) {
+    PruneBuffer();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats.h b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats.h
new file mode 100644
index 0000000000..60d20294a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
+#define MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
+
+#include <stdint.h>
+
+#include <set>
+
+namespace webrtc {
+
+// Keeps track of statistics of packet loss including whether losses are a
+// single packet or multiple packets in a row.
+class PacketLossStats {
+ public:
+  PacketLossStats();
+  ~PacketLossStats();
+
+  // Adds a lost packet to the stats by sequence number.
+  void AddLostPacket(uint16_t sequence_number);
+
+  // Queries the number of packets that were lost by themselves, no neighboring
+  // packets were lost.
+  int GetSingleLossCount() const;
+
+  // Queries the number of times that multiple packets with sequential numbers
+  // were lost. This is the number of events with more than one packet lost,
+  // regardless of the size of the event;
+  int GetMultipleLossEventCount() const;
+
+  // Queries the number of packets lost in multiple packet loss events. Combined
+  // with the event count, this can be used to determine the average event size.
+  int GetMultipleLossPacketCount() const;
+
+ private:
+  std::set<uint16_t> lost_packets_buffer_;
+  std::set<uint16_t> lost_packets_wrapped_buffer_;
+  int single_loss_historic_count_;
+  int multiple_loss_historic_event_count_;
+  int multiple_loss_historic_packet_count_;
+
+  void ComputeLossCounts(int* out_single_loss_count,
+                         int* out_multiple_loss_event_count,
+                         int* out_multiple_loss_packet_count) const;
+  void PruneBuffer();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_PACKET_LOSS_STATS_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats_unittest.cc
new file mode 100644
index 0000000000..673b223867
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_loss_stats_unittest.cc
@@ -0,0 +1,198 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/packet_loss_stats.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class PacketLossStatsTest : public ::testing::Test {
+ protected:
+  PacketLossStats stats_;
+};
+
+// Add a lost packet as every other packet, they should all count as single
+// losses.
+TEST_F(PacketLossStatsTest, EveryOtherPacket) {
+  for (int i = 0; i < 1000; i += 2) {
+    stats_.AddLostPacket(i);
+  }
+  EXPECT_EQ(500, stats_.GetSingleLossCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as every other packet, but such that the sequence numbers
+// will wrap around while they are being added.
+TEST_F(PacketLossStatsTest, EveryOtherPacketWrapped) {
+  for (int i = 65500; i < 66500; i += 2) {
+    stats_.AddLostPacket(i & 0xFFFF);
+  }
+  EXPECT_EQ(500, stats_.GetSingleLossCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as every other packet, but such that the sequence numbers
+// will wrap around close to the very end, such that the buffer contains packets
+// on either side of the wrapping.
+TEST_F(PacketLossStatsTest, EveryOtherPacketWrappedAtEnd) {
+  for (int i = 64600; i < 65600; i += 2) {
+    stats_.AddLostPacket(i & 0xFFFF);
+  }
+  EXPECT_EQ(500, stats_.GetSingleLossCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(0, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as the first three of every eight packets. Each set of
+// three should count as a multiple loss event and three multiple loss packets.
+TEST_F(PacketLossStatsTest, FirstThreeOfEight) {
+  for (int i = 0; i < 1000; ++i) {
+    if ((i & 7) < 3) {
+      stats_.AddLostPacket(i);
+    }
+  }
+  EXPECT_EQ(0, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as the first three of every eight packets such that the
+// sequence numbers wrap in the middle of adding them.
+TEST_F(PacketLossStatsTest, FirstThreeOfEightWrapped) {
+  for (int i = 65500; i < 66500; ++i) {
+    if ((i & 7) < 3) {
+      stats_.AddLostPacket(i & 0xFFFF);
+    }
+  }
+  EXPECT_EQ(0, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add a lost packet as the first three of every eight packets such that the
+// sequence numbers wrap near the end of adding them and there are still numbers
+// in the buffer from before the wrapping.
+TEST_F(PacketLossStatsTest, FirstThreeOfEightWrappedAtEnd) {
+  for (int i = 64600; i < 65600; ++i) {
+    if ((i & 7) < 3) {
+      stats_.AddLostPacket(i & 0xFFFF);
+    }
+  }
+  EXPECT_EQ(0, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets as the first three and the fifth of every eight packets. The
+// set of three should be multiple loss and the fifth should be single loss.
+TEST_F(PacketLossStatsTest, FirstThreeAndFifthOfEight) {
+  for (int i = 0; i < 1000; ++i) {
+    if ((i & 7) < 3 || (i & 7) == 4) {
+      stats_.AddLostPacket(i);
+    }
+  }
+  EXPECT_EQ(125, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets as the first three and the fifth of every eight packets such
+// that the sequence numbers wrap in the middle of adding them.
+TEST_F(PacketLossStatsTest, FirstThreeAndFifthOfEightWrapped) {
+  for (int i = 65500; i < 66500; ++i) {
+    if ((i & 7) < 3 || (i & 7) == 4) {
+      stats_.AddLostPacket(i & 0xFFFF);
+    }
+  }
+  EXPECT_EQ(125, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets as the first three and the fifth of every eight packets such
+// that the sequence numbers wrap near the end of adding them and there are
+// packets from before the wrapping still in the buffer.
+TEST_F(PacketLossStatsTest, FirstThreeAndFifthOfEightWrappedAtEnd) {
+  for (int i = 64600; i < 65600; ++i) {
+    if ((i & 7) < 3 || (i & 7) == 4) {
+      stats_.AddLostPacket(i & 0xFFFF);
+    }
+  }
+  EXPECT_EQ(125, stats_.GetSingleLossCount());
+  EXPECT_EQ(125, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(375, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets such that there is a multiple loss event that continues
+// around the wrapping of sequence numbers.
+TEST_F(PacketLossStatsTest, MultipleLossEventWrapped) {
+  for (int i = 60000; i < 60500; i += 2) {
+    stats_.AddLostPacket(i);
+  }
+  for (int i = 65530; i < 65540; ++i) {
+    stats_.AddLostPacket(i & 0xFFFF);
+  }
+  EXPECT_EQ(250, stats_.GetSingleLossCount());
+  EXPECT_EQ(1, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(10, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets such that there is a multiple loss event that continues
+// around the wrapping of sequence numbers and then is pushed out of the buffer.
+TEST_F(PacketLossStatsTest, MultipleLossEventWrappedPushedOut) {
+  for (int i = 60000; i < 60500; i += 2) {
+    stats_.AddLostPacket(i);
+  }
+  for (int i = 65530; i < 65540; ++i) {
+    stats_.AddLostPacket(i & 0xFFFF);
+  }
+  for (int i = 1000; i < 1500; i += 2) {
+    stats_.AddLostPacket(i);
+  }
+  EXPECT_EQ(500, stats_.GetSingleLossCount());
+  EXPECT_EQ(1, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(10, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets out of order and ensure that they still get counted
+// correctly as single or multiple loss events.
+TEST_F(PacketLossStatsTest, OutOfOrder) {
+  for (int i = 0; i < 1000; i += 10) {
+    stats_.AddLostPacket(i + 5);
+    stats_.AddLostPacket(i + 7);
+    stats_.AddLostPacket(i + 4);
+    stats_.AddLostPacket(i + 1);
+    stats_.AddLostPacket(i + 2);
+  }
+  EXPECT_EQ(100, stats_.GetSingleLossCount());
+  EXPECT_EQ(200, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(400, stats_.GetMultipleLossPacketCount());
+}
+
+// Add loss packets out of order and ensure that they still get counted
+// correctly as single or multiple loss events, and wrap in the middle of
+// adding.
+TEST_F(PacketLossStatsTest, OutOfOrderWrapped) {
+  for (int i = 65000; i < 66000; i += 10) {
+    stats_.AddLostPacket((i + 5) & 0xFFFF);
+    stats_.AddLostPacket((i + 7) & 0xFFFF);
+    stats_.AddLostPacket((i + 4) & 0xFFFF);
+    stats_.AddLostPacket((i + 1) & 0xFFFF);
+    stats_.AddLostPacket((i + 2) & 0xFFFF);
+  }
+  EXPECT_EQ(100, stats_.GetSingleLossCount());
+  EXPECT_EQ(200, stats_.GetMultipleLossEventCount());
+  EXPECT_EQ(400, stats_.GetMultipleLossPacketCount());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer.cc
new file mode 100644
index 0000000000..55edd768a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/packet_sequencer.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+
+namespace {
+// RED header is first byte of payload, if present.
+constexpr size_t kRedForFecHeaderLength = 1;
+
+// Timestamps use a 90kHz clock.
+constexpr uint32_t kTimestampTicksPerMs = 90;
+}  // namespace
+
+PacketSequencer::PacketSequencer(uint32_t media_ssrc,
+                                 absl::optional<uint32_t> rtx_ssrc,
+                                 bool require_marker_before_media_padding,
+                                 Clock* clock)
+    : media_ssrc_(media_ssrc),
+      rtx_ssrc_(rtx_ssrc),
+      require_marker_before_media_padding_(require_marker_before_media_padding),
+      clock_(clock),
+      media_sequence_number_(0),
+      rtx_sequence_number_(0),
+      last_payload_type_(-1),
+      last_rtp_timestamp_(0),
+      last_capture_time_ms_(0),
+      last_timestamp_time_ms_(0),
+      last_packet_marker_bit_(false) {
+  Random random(clock_->TimeInMicroseconds());
+  // Random start, 16 bits. Upper half of range is avoided in order to prevent
+  // wraparound issues during startup. Sequence number 0 is avoided for
+  // historical reasons, presumably to avoid debugability or test usage
+  // conflicts.
+  constexpr uint16_t kMaxInitRtpSeqNumber = 0x7fff;  // 2^15 - 1.
+  media_sequence_number_ = random.Rand(1, kMaxInitRtpSeqNumber);
+  rtx_sequence_number_ = random.Rand(1, kMaxInitRtpSeqNumber);
+}
+
+void PacketSequencer::Sequence(RtpPacketToSend& packet) {
+  if (packet.Ssrc() == media_ssrc_) {
+    if (packet.packet_type() == RtpPacketMediaType::kRetransmission) {
+      // Retransmission of an already sequenced packet, ignore.
+      return;
+    } else if (packet.packet_type() == RtpPacketMediaType::kPadding) {
+      PopulatePaddingFields(packet);
+    }
+    packet.SetSequenceNumber(media_sequence_number_++);
+    if (packet.packet_type() != RtpPacketMediaType::kPadding) {
+      UpdateLastPacketState(packet);
+    }
+  } else if (packet.Ssrc() == rtx_ssrc_) {
+    if (packet.packet_type() == RtpPacketMediaType::kPadding) {
+      PopulatePaddingFields(packet);
+    }
+    packet.SetSequenceNumber(rtx_sequence_number_++);
+  } else {
+    RTC_DCHECK_NOTREACHED() << "Unexpected ssrc " << packet.Ssrc();
+  }
+}
+
+void PacketSequencer::SetRtpState(const RtpState& state) {
+  media_sequence_number_ = state.sequence_number;
+  last_rtp_timestamp_ = state.timestamp;
+  last_capture_time_ms_ = state.capture_time_ms;
+  last_timestamp_time_ms_ = state.last_timestamp_time_ms;
+}
+
+void PacketSequencer::PopulateRtpState(RtpState& state) const {
+  state.sequence_number = media_sequence_number_;
+  state.timestamp = last_rtp_timestamp_;
+  state.capture_time_ms = last_capture_time_ms_;
+  state.last_timestamp_time_ms = last_timestamp_time_ms_;
+}
+
+void PacketSequencer::UpdateLastPacketState(const RtpPacketToSend& packet) {
+  // Remember marker bit to determine if padding can be inserted with
+  // sequence number following `packet`.
+  last_packet_marker_bit_ = packet.Marker();
+  // Remember media payload type to use in the padding packet if rtx is
+  // disabled.
+  if (packet.is_red()) {
+    RTC_DCHECK_GE(packet.payload_size(), kRedForFecHeaderLength);
+    last_payload_type_ = packet.PayloadBuffer()[0];
+  } else {
+    last_payload_type_ = packet.PayloadType();
+  }
+  // Save timestamps to generate timestamp field and extensions for the padding.
+  last_rtp_timestamp_ = packet.Timestamp();
+  last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
+  last_capture_time_ms_ = packet.capture_time().ms();
+}
+
+void PacketSequencer::PopulatePaddingFields(RtpPacketToSend& packet) {
+  if (packet.Ssrc() == media_ssrc_) {
+    RTC_DCHECK(CanSendPaddingOnMediaSsrc());
+
+    packet.SetTimestamp(last_rtp_timestamp_);
+    packet.set_capture_time(Timestamp::Millis(last_capture_time_ms_));
+    packet.SetPayloadType(last_payload_type_);
+    return;
+  }
+
+  RTC_DCHECK(packet.Ssrc() == rtx_ssrc_);
+  if (packet.payload_size() > 0) {
+    // This is payload padding packet, don't update timestamp fields.
+    return;
+  }
+
+  packet.SetTimestamp(last_rtp_timestamp_);
+  packet.set_capture_time(Timestamp::Millis(last_capture_time_ms_));
+
+  // Only change the timestamp of padding packets sent over RTX.
+  // Padding only packets over RTP has to be sent as part of a media
+  // frame (and therefore the same timestamp).
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (last_timestamp_time_ms_ > 0) {
+    packet.SetTimestamp(packet.Timestamp() +
+                        (now_ms - last_timestamp_time_ms_) *
+                            kTimestampTicksPerMs);
+    if (packet.capture_time() > Timestamp::Zero()) {
+      packet.set_capture_time(
+          packet.capture_time() +
+          TimeDelta::Millis(now_ms - last_timestamp_time_ms_));
+    }
+  }
+}
+
+bool PacketSequencer::CanSendPaddingOnMediaSsrc() const {
+  if (last_payload_type_ == -1) {
+    return false;
+  }
+
+  // Without RTX we can't send padding in the middle of frames.
+  // For audio marker bits doesn't mark the end of a frame and frames
+  // are usually a single packet, so for now we don't apply this rule
+  // for audio.
+  if (require_marker_before_media_padding_ && !last_packet_marker_bit_) {
+    return false;
+  }
+
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer.h b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer.h
new file mode 100644
index 0000000000..7c0dee7a5b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
+#define MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+// Helper class used to assign RTP sequence numbers and populate some fields for
+// padding packets based on the last sequenced packets.
+// This class is not thread safe, the caller must provide that.
+class PacketSequencer {
+ public:
+  // If `require_marker_before_media_padding_` is true, padding packets on the
+  // media ssrc is not allowed unless the last sequenced media packet had the
+  // marker bit set (i.e. don't insert padding packets between the first and
+  // last packets of a video frame).
+  // Packets with unknown SSRCs will be ignored.
+  PacketSequencer(uint32_t media_ssrc,
+                  absl::optional<uint32_t> rtx_ssrc,
+                  bool require_marker_before_media_padding,
+                  Clock* clock);
+
+  // Assigns sequence number, and in the case of non-RTX padding also timestamps
+  // and payload type.
+  void Sequence(RtpPacketToSend& packet);
+
+  void set_media_sequence_number(uint16_t sequence_number) {
+    media_sequence_number_ = sequence_number;
+  }
+  void set_rtx_sequence_number(uint16_t sequence_number) {
+    rtx_sequence_number_ = sequence_number;
+  }
+
+  void SetRtpState(const RtpState& state);
+  void PopulateRtpState(RtpState& state) const;
+
+  uint16_t media_sequence_number() const { return media_sequence_number_; }
+  uint16_t rtx_sequence_number() const { return rtx_sequence_number_; }
+
+  // Checks whether it is allowed to send padding on the media SSRC at this
+  // time, e.g. that we don't send padding in the middle of a video frame.
+  bool CanSendPaddingOnMediaSsrc() const;
+
+ private:
+  void UpdateLastPacketState(const RtpPacketToSend& packet);
+  void PopulatePaddingFields(RtpPacketToSend& packet);
+
+  const uint32_t media_ssrc_;
+  const absl::optional<uint32_t> rtx_ssrc_;
+  const bool require_marker_before_media_padding_;
+  Clock* const clock_;
+
+  uint16_t media_sequence_number_;
+  uint16_t rtx_sequence_number_;
+
+  int8_t last_payload_type_;
+  uint32_t last_rtp_timestamp_;
+  int64_t last_capture_time_ms_;
+  int64_t last_timestamp_time_ms_;
+  bool last_packet_marker_bit_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer_unittest.cc
new file mode 100644
index 0000000000..d892863768
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/packet_sequencer_unittest.cc
@@ -0,0 +1,250 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/packet_sequencer.h"
+
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+constexpr Timestamp kStartTime = Timestamp::Millis(98765);
+constexpr uint32_t kMediaSsrc = 123456;
+constexpr uint32_t kRtxSsrc = 123457;
+constexpr uint8_t kMediaPayloadType = 42;
+constexpr uint16_t kMediaStartSequenceNumber = 123;
+constexpr uint16_t kRtxStartSequenceNumber = 234;
+constexpr uint16_t kDefaultSequenceNumber = 0x1234;
+constexpr uint32_t kStartRtpTimestamp = 798;
+
+class PacketSequencerTest : public ::testing::Test {
+ public:
+  PacketSequencerTest()
+      : clock_(kStartTime),
+        sequencer_(kMediaSsrc,
+                   kRtxSsrc,
+                   /*require_marker_before_media_padding=*/true,
+                   &clock_) {}
+
+  RtpPacketToSend CreatePacket(RtpPacketMediaType type, uint32_t ssrc) {
+    RtpPacketToSend packet(/*extension_manager=*/nullptr);
+    packet.set_packet_type(type);
+    packet.SetSsrc(ssrc);
+    packet.SetSequenceNumber(kDefaultSequenceNumber);
+    packet.set_capture_time(clock_.CurrentTime());
+    packet.SetTimestamp(
+        kStartRtpTimestamp +
+        static_cast<uint32_t>(packet.capture_time().ms() - kStartTime.ms()));
+    return packet;
+  }
+
+ protected:
+  SimulatedClock clock_;
+  PacketSequencer sequencer_;
+};
+
+TEST_F(PacketSequencerTest, IgnoresMediaSsrcRetransmissions) {
+  RtpPacketToSend packet =
+      CreatePacket(RtpPacketMediaType::kRetransmission, kMediaSsrc);
+  sequencer_.set_media_sequence_number(kMediaStartSequenceNumber);
+  sequencer_.Sequence(packet);
+  EXPECT_EQ(packet.SequenceNumber(), kDefaultSequenceNumber);
+  EXPECT_EQ(sequencer_.media_sequence_number(), kMediaStartSequenceNumber);
+}
+
+TEST_F(PacketSequencerTest, SequencesAudio) {
+  RtpPacketToSend packet = CreatePacket(RtpPacketMediaType::kAudio, kMediaSsrc);
+  sequencer_.set_media_sequence_number(kMediaStartSequenceNumber);
+  sequencer_.Sequence(packet);
+  EXPECT_EQ(packet.SequenceNumber(), kMediaStartSequenceNumber);
+  EXPECT_EQ(sequencer_.media_sequence_number(), kMediaStartSequenceNumber + 1);
+}
+
+TEST_F(PacketSequencerTest, SequencesVideo) {
+  RtpPacketToSend packet = CreatePacket(RtpPacketMediaType::kVideo, kMediaSsrc);
+  sequencer_.set_media_sequence_number(kMediaStartSequenceNumber);
+  sequencer_.Sequence(packet);
+  EXPECT_EQ(packet.SequenceNumber(), kMediaStartSequenceNumber);
+  EXPECT_EQ(sequencer_.media_sequence_number(), kMediaStartSequenceNumber + 1);
+}
+
+TEST_F(PacketSequencerTest, SequencesUlpFec) {
+  RtpPacketToSend packet =
+      CreatePacket(RtpPacketMediaType::kForwardErrorCorrection, kMediaSsrc);
+  sequencer_.set_media_sequence_number(kMediaStartSequenceNumber);
+  sequencer_.Sequence(packet);
+  EXPECT_EQ(packet.SequenceNumber(), kMediaStartSequenceNumber);
+  EXPECT_EQ(sequencer_.media_sequence_number(), kMediaStartSequenceNumber + 1);
+}
+
+TEST_F(PacketSequencerTest, SequencesRtxRetransmissions) {
+  RtpPacketToSend packet =
+      CreatePacket(RtpPacketMediaType::kRetransmission, kRtxSsrc);
+  sequencer_.set_rtx_sequence_number(kRtxStartSequenceNumber);
+  sequencer_.Sequence(packet);
+  EXPECT_EQ(packet.SequenceNumber(), kRtxStartSequenceNumber);
+  EXPECT_EQ(sequencer_.rtx_sequence_number(), kRtxStartSequenceNumber + 1);
+}
+
+TEST_F(PacketSequencerTest, ProhibitsPaddingWithinVideoFrame) {
+  // Send a video packet with the marker bit set to false (indicating it is not
+  // the last packet of a frame).
+  RtpPacketToSend media_packet =
+      CreatePacket(RtpPacketMediaType::kVideo, kMediaSsrc);
+  media_packet.SetPayloadType(kMediaPayloadType);
+  media_packet.SetMarker(false);
+  sequencer_.Sequence(media_packet);
+
+  // Sending padding on the media SSRC should not be allowed at this point.
+  EXPECT_FALSE(sequencer_.CanSendPaddingOnMediaSsrc());
+
+  // Send a video packet with marker set to true, padding should be allowed
+  // again.
+  media_packet.SetMarker(true);
+  sequencer_.Sequence(media_packet);
+  EXPECT_TRUE(sequencer_.CanSendPaddingOnMediaSsrc());
+}
+
+TEST_F(PacketSequencerTest, AllowsPaddingAtAnyTimeIfSoConfigured) {
+  PacketSequencer packet_sequencer(
+      kMediaSsrc, kRtxSsrc,
+      /*require_marker_before_media_padding=*/false, &clock_);
+
+  // Send an audio packet with the marker bit set to false.
+  RtpPacketToSend media_packet =
+      CreatePacket(RtpPacketMediaType::kAudio, kMediaSsrc);
+  media_packet.SetPayloadType(kMediaPayloadType);
+  media_packet.SetMarker(false);
+  packet_sequencer.Sequence(media_packet);
+
+  // Sending padding on the media SSRC should be allowed despite no marker bit.
+  EXPECT_TRUE(packet_sequencer.CanSendPaddingOnMediaSsrc());
+}
+
+TEST_F(PacketSequencerTest, UpdatesPaddingBasedOnLastMediaPacket) {
+  // First send a media packet.
+  RtpPacketToSend media_packet =
+      CreatePacket(RtpPacketMediaType::kVideo, kMediaSsrc);
+  media_packet.SetPayloadType(kMediaPayloadType);
+  media_packet.SetMarker(true);
+  // Advance time so current time doesn't exactly match timestamp.
+  clock_.AdvanceTime(TimeDelta::Millis(5));
+  sequencer_.set_media_sequence_number(kMediaStartSequenceNumber);
+  sequencer_.Sequence(media_packet);
+
+  // Next send a padding packet and verify the media packet's timestamps and
+  // payload type is transferred to the padding packet.
+  RtpPacketToSend padding_packet =
+      CreatePacket(RtpPacketMediaType::kPadding, kMediaSsrc);
+  padding_packet.SetPadding(/*padding_size=*/100);
+  sequencer_.Sequence(padding_packet);
+
+  EXPECT_EQ(padding_packet.SequenceNumber(), kMediaStartSequenceNumber + 1);
+  EXPECT_EQ(padding_packet.PayloadType(), kMediaPayloadType);
+  EXPECT_EQ(padding_packet.Timestamp(), media_packet.Timestamp());
+  EXPECT_EQ(padding_packet.capture_time(), media_packet.capture_time());
+}
+
+TEST_F(PacketSequencerTest, UpdatesPaddingBasedOnLastRedPacket) {
+  // First send a media packet.
+  RtpPacketToSend media_packet =
+      CreatePacket(RtpPacketMediaType::kVideo, kMediaSsrc);
+  media_packet.SetPayloadType(kMediaPayloadType);
+  // Simulate a packet with RED encapsulation;
+  media_packet.set_is_red(true);
+  uint8_t* payload_buffer = media_packet.SetPayloadSize(1);
+  payload_buffer[0] = kMediaPayloadType + 1;
+
+  media_packet.SetMarker(true);
+  // Advance time so current time doesn't exactly match timestamp.
+  clock_.AdvanceTime(TimeDelta::Millis(5));
+  sequencer_.set_media_sequence_number(kMediaStartSequenceNumber);
+  sequencer_.Sequence(media_packet);
+
+  // Next send a padding packet and verify the media packet's timestamps and
+  // payload type is transferred to the padding packet.
+  RtpPacketToSend padding_packet =
+      CreatePacket(RtpPacketMediaType::kPadding, kMediaSsrc);
+  padding_packet.SetPadding(100);
+  sequencer_.Sequence(padding_packet);
+
+  EXPECT_EQ(padding_packet.SequenceNumber(), kMediaStartSequenceNumber + 1);
+  EXPECT_EQ(padding_packet.PayloadType(), kMediaPayloadType + 1);
+  EXPECT_EQ(padding_packet.Timestamp(), media_packet.Timestamp());
+  EXPECT_EQ(padding_packet.capture_time(), media_packet.capture_time());
+}
+
+TEST_F(PacketSequencerTest, DoesNotUpdateFieldsOnPayloadPadding) {
+  // First send a media packet.
+  RtpPacketToSend media_packet =
+      CreatePacket(RtpPacketMediaType::kVideo, kMediaSsrc);
+  media_packet.SetPayloadType(kMediaPayloadType);
+  media_packet.SetMarker(true);
+  // Advance time so current time doesn't exactly match timestamp.
+  clock_.AdvanceTime(TimeDelta::Millis(5));
+  sequencer_.set_media_sequence_number(kMediaStartSequenceNumber);
+  sequencer_.Sequence(media_packet);
+
+  // Simulate a payload padding packet on the RTX SSRC.
+  RtpPacketToSend padding_packet =
+      CreatePacket(RtpPacketMediaType::kPadding, kRtxSsrc);
+  padding_packet.SetPayloadSize(100);
+  padding_packet.SetPayloadType(kMediaPayloadType + 1);
+  padding_packet.SetTimestamp(kStartRtpTimestamp + 1);
+  padding_packet.set_capture_time(kStartTime + TimeDelta::Millis(1));
+  sequencer_.set_rtx_sequence_number(kRtxStartSequenceNumber);
+  sequencer_.Sequence(padding_packet);
+
+  // The sequence number should be updated, but timestamps kept.
+  EXPECT_EQ(padding_packet.SequenceNumber(), kRtxStartSequenceNumber);
+  EXPECT_EQ(padding_packet.PayloadType(), kMediaPayloadType + 1);
+  EXPECT_EQ(padding_packet.Timestamp(), kStartRtpTimestamp + 1);
+  EXPECT_EQ(padding_packet.capture_time(), kStartTime + TimeDelta::Millis(1));
+}
+
+TEST_F(PacketSequencerTest, UpdatesRtxPaddingBasedOnLastMediaPacket) {
+  constexpr uint32_t kTimestampTicksPerMs = 90;
+
+  // First send a media packet.
+  RtpPacketToSend media_packet =
+      CreatePacket(RtpPacketMediaType::kVideo, kMediaSsrc);
+  media_packet.SetPayloadType(kMediaPayloadType);
+  media_packet.SetMarker(true);
+  sequencer_.set_media_sequence_number(kMediaStartSequenceNumber);
+  sequencer_.Sequence(media_packet);
+
+  // Advance time, this time delta will be used to interpolate padding
+  // timestamps.
+  constexpr TimeDelta kTimeDelta = TimeDelta::Millis(10);
+  clock_.AdvanceTime(kTimeDelta);
+
+  RtpPacketToSend padding_packet =
+      CreatePacket(RtpPacketMediaType::kPadding, kRtxSsrc);
+  padding_packet.SetPadding(100);
+  padding_packet.SetPayloadType(kMediaPayloadType + 1);
+  sequencer_.set_rtx_sequence_number(kRtxStartSequenceNumber);
+  sequencer_.Sequence(padding_packet);
+
+  // Assigned RTX sequence number, but payload type unchanged in this case.
+  EXPECT_EQ(padding_packet.SequenceNumber(), kRtxStartSequenceNumber);
+  EXPECT_EQ(padding_packet.PayloadType(), kMediaPayloadType + 1);
+  // Timestamps are offset realtive to last media packet.
+  EXPECT_EQ(
+      padding_packet.Timestamp(),
+      media_packet.Timestamp() + (kTimeDelta.ms() * kTimestampTicksPerMs));
+  EXPECT_EQ(padding_packet.capture_time(),
+            media_packet.capture_time() + kTimeDelta);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
new file mode 100644
index 0000000000..1dc756d876
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -0,0 +1,431 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/receive_statistics_impl.h"
+
+#include <cmath>
+#include <cstdlib>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/units/time_delta.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+constexpr int64_t kStatisticsTimeoutMs = 8000;
+constexpr int64_t kStatisticsProcessIntervalMs = 1000;
+}  // namespace
+
+StreamStatistician::~StreamStatistician() {}
+
+StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc, Clock* clock,
+                                               int max_reordering_threshold)
+    : ssrc_(ssrc),
+      clock_(clock),
+      delta_internal_unix_epoch_ms_(clock_->CurrentNtpInMilliseconds() -
+                                    clock_->TimeInMilliseconds() -
+                                    rtc::kNtpJan1970Millisecs),
+      incoming_bitrate_(kStatisticsProcessIntervalMs,
+                        RateStatistics::kBpsScale),
+      max_reordering_threshold_(max_reordering_threshold),
+      enable_retransmit_detection_(false),
+      cumulative_loss_is_capped_(false),
+      jitter_q4_(0),
+      cumulative_loss_(0),
+      cumulative_loss_rtcp_offset_(0),
+      last_receive_time_ms_(0),
+      last_received_timestamp_(0),
+      received_seq_first_(-1),
+      received_seq_max_(-1),
+      last_report_cumulative_loss_(0),
+      last_report_seq_max_(-1),
+      last_payload_type_frequency_(0) {}
+
+StreamStatisticianImpl::~StreamStatisticianImpl() = default;
+
+bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
+                                              int64_t sequence_number,
+                                              int64_t now_ms) {
+  // Check if `packet` is second packet of a stream restart.
+  if (received_seq_out_of_order_) {
+    // Count the previous packet as a received; it was postponed below.
+    --cumulative_loss_;
+
+    uint16_t expected_sequence_number = *received_seq_out_of_order_ + 1;
+    received_seq_out_of_order_ = absl::nullopt;
+    if (packet.SequenceNumber() == expected_sequence_number) {
+      // Ignore sequence number gap caused by stream restart for packet loss
+      // calculation, by setting received_seq_max_ to the sequence number just
+      // before the out-of-order seqno. This gives a net zero change of
+      // `cumulative_loss_`, for the two packets interpreted as a stream reset.
+      //
+      // Fraction loss for the next report may get a bit off, since we don't
+      // update last_report_seq_max_ and last_report_cumulative_loss_ in a
+      // consistent way.
+      last_report_seq_max_ = sequence_number - 2;
+      received_seq_max_ = sequence_number - 2;
+      return false;
+    }
+  }
+
+  if (std::abs(sequence_number - received_seq_max_) >
+      max_reordering_threshold_) {
+    // Sequence number gap looks too large, wait until next packet to check
+    // for a stream restart.
+    received_seq_out_of_order_ = packet.SequenceNumber();
+    // Postpone counting this as a received packet until we know how to update
+    // `received_seq_max_`, otherwise we temporarily decrement
+    // `cumulative_loss_`. The
+    // ReceiveStatisticsTest.StreamRestartDoesntCountAsLoss test expects
+    // `cumulative_loss_` to be unchanged by the reception of the first packet
+    // after stream reset.
+    ++cumulative_loss_;
+    return true;
+  }
+
+  if (sequence_number > received_seq_max_)
+    return false;
+
+  // Old out of order packet, may be retransmit.
+  if (enable_retransmit_detection_ && IsRetransmitOfOldPacket(packet, now_ms))
+    receive_counters_.retransmitted.AddPacket(packet);
+  return true;
+}
+
+void StreamStatisticianImpl::UpdateCounters(const RtpPacketReceived& packet) {
+  RTC_DCHECK_EQ(ssrc_, packet.Ssrc());
+  int64_t now_ms = clock_->TimeInMilliseconds();
+
+  incoming_bitrate_.Update(packet.size(), now_ms);
+  receive_counters_.last_packet_received_timestamp_ms = now_ms;
+  receive_counters_.transmitted.AddPacket(packet);
+  --cumulative_loss_;
+
+  // Use PeekUnwrap and later update the state to avoid updating the state for
+  // out of order packets.
+  int64_t sequence_number = seq_unwrapper_.PeekUnwrap(packet.SequenceNumber());
+
+  if (!ReceivedRtpPacket()) {
+    received_seq_first_ = sequence_number;
+    last_report_seq_max_ = sequence_number - 1;
+    received_seq_max_ = sequence_number - 1;
+    receive_counters_.first_packet_time_ms = now_ms;
+  } else if (UpdateOutOfOrder(packet, sequence_number, now_ms)) {
+    return;
+  }
+  // In order packet.
+  cumulative_loss_ += sequence_number - received_seq_max_;
+  received_seq_max_ = sequence_number;
+  // Update the internal state of `seq_unwrapper_`.
+  seq_unwrapper_.Unwrap(packet.SequenceNumber());
+
+  // If new time stamp and more than one in-order packet received, calculate
+  // new jitter statistics.
+  if (packet.Timestamp() != last_received_timestamp_ &&
+      (receive_counters_.transmitted.packets -
+       receive_counters_.retransmitted.packets) > 1) {
+    UpdateJitter(packet, now_ms);
+  }
+  last_received_timestamp_ = packet.Timestamp();
+  last_receive_time_ms_ = now_ms;
+}
+
+void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet,
+                                          int64_t receive_time_ms) {
+  int64_t receive_diff_ms = receive_time_ms - last_receive_time_ms_;
+  RTC_DCHECK_GE(receive_diff_ms, 0);
+  uint32_t receive_diff_rtp = static_cast<uint32_t>(
+      (receive_diff_ms * packet.payload_type_frequency()) / 1000);
+  int32_t time_diff_samples =
+      receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_);
+
+  time_diff_samples = std::abs(time_diff_samples);
+
+  ReviseFrequencyAndJitter(packet.payload_type_frequency());
+
+  // lib_jingle sometimes deliver crazy jumps in TS for the same stream.
+  // If this happens, don't update jitter value. Use 5 secs video frequency
+  // as the threshold.
+  if (time_diff_samples < 450000) {
+    // Note we calculate in Q4 to avoid using float.
+    int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;
+    jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
+  }
+}
+
+void StreamStatisticianImpl::ReviseFrequencyAndJitter(
+    int payload_type_frequency) {
+  if (payload_type_frequency == last_payload_type_frequency_) {
+    return;
+  }
+
+  if (payload_type_frequency != 0) {
+    if (last_payload_type_frequency_ != 0) {
+      // Value in "jitter_q4_" variable is a number of samples.
+      // I.e. jitter = timestamp (ms) * frequency (kHz).
+      // Since the frequency has changed we have to update the number of samples
+      // accordingly. The new value should rely on a new frequency.
+
+      // If we don't do such procedure we end up with the number of samples that
+      // cannot be converted into milliseconds correctly
+      // (i.e. jitter_ms = jitter_q4_ >> 4 / (payload_type_frequency / 1000)).
+      // In such case, the number of samples has a "mix".
+
+      // Doing so we pretend that everything prior and including the current
+      // packet were computed on packet's frequency.
+      jitter_q4_ = static_cast<int>(static_cast<uint64_t>(jitter_q4_) *
+                                    payload_type_frequency /
+                                    last_payload_type_frequency_);
+    }
+    // If last_payload_type_frequency_ is not present, the jitter_q4_
+    // variable has its initial value.
+
+    // Keep last_payload_type_frequency_ up to date and non-zero (set).
+    last_payload_type_frequency_ = payload_type_frequency;
+  }
+}
+
+void StreamStatisticianImpl::SetMaxReorderingThreshold(
+    int max_reordering_threshold) {
+  max_reordering_threshold_ = max_reordering_threshold;
+}
+
+void StreamStatisticianImpl::EnableRetransmitDetection(bool enable) {
+  enable_retransmit_detection_ = enable;
+}
+
+RtpReceiveStats StreamStatisticianImpl::GetStats() const {
+  RtpReceiveStats stats;
+  stats.packets_lost = cumulative_loss_;
+  // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
+  stats.jitter = jitter_q4_ >> 4;
+  if (last_payload_type_frequency_ > 0) {
+    // Divide value in fractional seconds by frequency to get jitter in
+    // fractional seconds.
+    stats.interarrival_jitter =
+        webrtc::TimeDelta::Seconds(stats.jitter) / last_payload_type_frequency_;
+  }
+  if (receive_counters_.last_packet_received_timestamp_ms.has_value()) {
+    stats.last_packet_received_timestamp_ms =
+        *receive_counters_.last_packet_received_timestamp_ms +
+        delta_internal_unix_epoch_ms_;
+  }
+  stats.packet_counter = receive_counters_.transmitted;
+  return stats;
+}
+
+void StreamStatisticianImpl::MaybeAppendReportBlockAndReset(
+    std::vector<rtcp::ReportBlock>& report_blocks) {
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (now_ms - last_receive_time_ms_ >= kStatisticsTimeoutMs) {
+    // Not active.
+    return;
+  }
+  if (!ReceivedRtpPacket()) {
+    return;
+  }
+
+  report_blocks.emplace_back();
+  rtcp::ReportBlock& stats = report_blocks.back();
+  stats.SetMediaSsrc(ssrc_);
+  // Calculate fraction lost.
+  int64_t exp_since_last = received_seq_max_ - last_report_seq_max_;
+  RTC_DCHECK_GE(exp_since_last, 0);
+
+  int32_t lost_since_last = cumulative_loss_ - last_report_cumulative_loss_;
+  if (exp_since_last > 0 && lost_since_last > 0) {
+    // Scale 0 to 255, where 255 is 100% loss.
+    stats.SetFractionLost(255 * lost_since_last / exp_since_last);
+  }
+
+  int packets_lost = cumulative_loss_ + cumulative_loss_rtcp_offset_;
+  if (packets_lost < 0) {
+    // Clamp to zero. Work around to accomodate for senders that misbehave with
+    // negative cumulative loss.
+    packets_lost = 0;
+    cumulative_loss_rtcp_offset_ = -cumulative_loss_;
+  }
+  if (packets_lost > 0x7fffff) {
+    // Packets lost is a 24 bit signed field, and thus should be clamped, as
+    // described in https://datatracker.ietf.org/doc/html/rfc3550#appendix-A.3
+    if (!cumulative_loss_is_capped_) {
+      cumulative_loss_is_capped_ = true;
+      RTC_LOG(LS_WARNING) << "Cumulative loss reached maximum value for ssrc "
+                          << ssrc_;
+    }
+    packets_lost = 0x7fffff;
+  }
+  stats.SetCumulativeLost(packets_lost);
+  stats.SetExtHighestSeqNum(received_seq_max_);
+  // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
+  stats.SetJitter(jitter_q4_ >> 4);
+
+  // Only for report blocks in RTCP SR and RR.
+  last_report_cumulative_loss_ = cumulative_loss_;
+  last_report_seq_max_ = received_seq_max_;
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts", now_ms,
+                                  cumulative_loss_, ssrc_);
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "received_seq_max_pkts", now_ms,
+                                  (received_seq_max_ - received_seq_first_),
+                                  ssrc_);
+}
+
+absl::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const {
+  if (!ReceivedRtpPacket()) {
+    return absl::nullopt;
+  }
+  int64_t expected_packets = 1 + received_seq_max_ - received_seq_first_;
+  if (expected_packets <= 0) {
+    return absl::nullopt;
+  }
+  if (cumulative_loss_ <= 0) {
+    return 0;
+  }
+  return 100 * static_cast<int64_t>(cumulative_loss_) / expected_packets;
+}
+
+StreamDataCounters StreamStatisticianImpl::GetReceiveStreamDataCounters()
+    const {
+  return receive_counters_;
+}
+
+uint32_t StreamStatisticianImpl::BitrateReceived() const {
+  return incoming_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0);
+}
+
+bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
+    const RtpPacketReceived& packet,
+    int64_t now_ms) const {
+  uint32_t frequency_khz = packet.payload_type_frequency() / 1000;
+  RTC_DCHECK_GT(frequency_khz, 0);
+
+  int64_t time_diff_ms = now_ms - last_receive_time_ms_;
+
+  // Diff in time stamp since last received in order.
+  uint32_t timestamp_diff = packet.Timestamp() - last_received_timestamp_;
+  uint32_t rtp_time_stamp_diff_ms = timestamp_diff / frequency_khz;
+
+  int64_t max_delay_ms = 0;
+
+  // Jitter standard deviation in samples.
+  float jitter_std = std::sqrt(static_cast<float>(jitter_q4_ >> 4));
+
+  // 2 times the standard deviation => 95% confidence.
+  // And transform to milliseconds by dividing by the frequency in kHz.
+  max_delay_ms = static_cast<int64_t>((2 * jitter_std) / frequency_khz);
+
+  // Min max_delay_ms is 1.
+  if (max_delay_ms == 0) {
+    max_delay_ms = 1;
+  }
+  return time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms;
+}
+
+std::unique_ptr<ReceiveStatistics> ReceiveStatistics::Create(Clock* clock) {
+  return std::make_unique<ReceiveStatisticsLocked>(
+      clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) {
+        return std::make_unique<StreamStatisticianLocked>(
+            ssrc, clock, max_reordering_threshold);
+      });
+}
+
+std::unique_ptr<ReceiveStatistics> ReceiveStatistics::CreateThreadCompatible(
+    Clock* clock) {
+  return std::make_unique<ReceiveStatisticsImpl>(
+      clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) {
+        return std::make_unique<StreamStatisticianImpl>(
+            ssrc, clock, max_reordering_threshold);
+      });
+}
+
+ReceiveStatisticsImpl::ReceiveStatisticsImpl(
+    Clock* clock,
+    std::function<std::unique_ptr<StreamStatisticianImplInterface>(
+        uint32_t ssrc,
+        Clock* clock,
+        int max_reordering_threshold)> stream_statistician_factory)
+    : clock_(clock),
+      stream_statistician_factory_(std::move(stream_statistician_factory)),
+      last_returned_ssrc_idx_(0),
+      max_reordering_threshold_(kDefaultMaxReorderingThreshold) {}
+
+void ReceiveStatisticsImpl::OnRtpPacket(const RtpPacketReceived& packet) {
+  // StreamStatisticianImpl instance is created once and only destroyed when
+  // this whole ReceiveStatisticsImpl is destroyed. StreamStatisticianImpl has
+  // it's own locking so don't hold receive_statistics_lock_ (potential
+  // deadlock).
+  GetOrCreateStatistician(packet.Ssrc())->UpdateCounters(packet);
+}
+
+StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
+    uint32_t ssrc) const {
+  const auto& it = statisticians_.find(ssrc);
+  if (it == statisticians_.end())
+    return nullptr;
+  return it->second.get();
+}
+
+StreamStatisticianImplInterface* ReceiveStatisticsImpl::GetOrCreateStatistician(
+    uint32_t ssrc) {
+  std::unique_ptr<StreamStatisticianImplInterface>& impl = statisticians_[ssrc];
+  if (impl == nullptr) {  // new element
+    impl =
+        stream_statistician_factory_(ssrc, clock_, max_reordering_threshold_);
+    all_ssrcs_.push_back(ssrc);
+  }
+  return impl.get();
+}
+
+void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
+    int max_reordering_threshold) {
+  max_reordering_threshold_ = max_reordering_threshold;
+  for (auto& statistician : statisticians_) {
+    statistician.second->SetMaxReorderingThreshold(max_reordering_threshold);
+  }
+}
+
+void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
+    uint32_t ssrc,
+    int max_reordering_threshold) {
+  GetOrCreateStatistician(ssrc)->SetMaxReorderingThreshold(
+      max_reordering_threshold);
+}
+
+void ReceiveStatisticsImpl::EnableRetransmitDetection(uint32_t ssrc,
+                                                      bool enable) {
+  GetOrCreateStatistician(ssrc)->EnableRetransmitDetection(enable);
+}
+
+std::vector<rtcp::ReportBlock> ReceiveStatisticsImpl::RtcpReportBlocks(
+    size_t max_blocks) {
+  std::vector<rtcp::ReportBlock> result;
+  result.reserve(std::min(max_blocks, all_ssrcs_.size()));
+
+  size_t ssrc_idx = 0;
+  for (size_t i = 0; i < all_ssrcs_.size() && result.size() < max_blocks; ++i) {
+    ssrc_idx = (last_returned_ssrc_idx_ + i + 1) % all_ssrcs_.size();
+    const uint32_t media_ssrc = all_ssrcs_[ssrc_idx];
+    auto statistician_it = statisticians_.find(media_ssrc);
+    RTC_DCHECK(statistician_it != statisticians_.end());
+    statistician_it->second->MaybeAppendReportBlockAndReset(result);
+  }
+  last_returned_ssrc_idx_ = ssrc_idx;
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
new file mode 100644
index 0000000000..11853b3244
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -0,0 +1,250 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
+#define MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
+
+#include <algorithm>
+#include <functional>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "rtc_base/containers/flat_map.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Extends StreamStatistician with methods needed by the implementation.
+class StreamStatisticianImplInterface : public StreamStatistician {
+ public:
+  virtual ~StreamStatisticianImplInterface() = default;
+  virtual void MaybeAppendReportBlockAndReset(
+      std::vector<rtcp::ReportBlock>& report_blocks) = 0;
+  virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
+  virtual void EnableRetransmitDetection(bool enable) = 0;
+  virtual void UpdateCounters(const RtpPacketReceived& packet) = 0;
+};
+
+// Thread-compatible implementation of StreamStatisticianImplInterface.
+class StreamStatisticianImpl : public StreamStatisticianImplInterface {
+ public:
+  StreamStatisticianImpl(uint32_t ssrc,
+                         Clock* clock,
+                         int max_reordering_threshold);
+  ~StreamStatisticianImpl() override;
+
+  // Implements StreamStatistician
+  RtpReceiveStats GetStats() const override;
+  absl::optional<int> GetFractionLostInPercent() const override;
+  StreamDataCounters GetReceiveStreamDataCounters() const override;
+  uint32_t BitrateReceived() const override;
+
+  // Implements StreamStatisticianImplInterface
+  void MaybeAppendReportBlockAndReset(
+      std::vector<rtcp::ReportBlock>& report_blocks) override;
+  void SetMaxReorderingThreshold(int max_reordering_threshold) override;
+  void EnableRetransmitDetection(bool enable) override;
+  // Updates StreamStatistician for incoming packets.
+  void UpdateCounters(const RtpPacketReceived& packet) override;
+
+ private:
+  bool IsRetransmitOfOldPacket(const RtpPacketReceived& packet,
+                               int64_t now_ms) const;
+  void UpdateJitter(const RtpPacketReceived& packet, int64_t receive_time_ms);
+  void ReviseFrequencyAndJitter(int payload_type_frequency);
+  // Updates StreamStatistician for out of order packets.
+  // Returns true if packet considered to be out of order.
+  bool UpdateOutOfOrder(const RtpPacketReceived& packet,
+                        int64_t sequence_number,
+                        int64_t now_ms);
+  // Checks if this StreamStatistician received any rtp packets.
+  bool ReceivedRtpPacket() const { return received_seq_first_ >= 0; }
+
+  const uint32_t ssrc_;
+  Clock* const clock_;
+  // Delta used to map internal timestamps to Unix epoch ones.
+  const int64_t delta_internal_unix_epoch_ms_;
+  RateStatistics incoming_bitrate_;
+  // In number of packets or sequence numbers.
+  int max_reordering_threshold_;
+  bool enable_retransmit_detection_;
+  bool cumulative_loss_is_capped_;
+
+  // Stats on received RTP packets.
+  uint32_t jitter_q4_;
+  // Cumulative loss according to RFC 3550, which may be negative (and often is,
+  // if packets are reordered and there are non-RTX retransmissions).
+  int32_t cumulative_loss_;
+  // Offset added to outgoing rtcp reports, to make ensure that the reported
+  // cumulative loss is non-negative. Reports with negative values confuse some
+  // senders, in particular, our own loss-based bandwidth estimator.
+  int32_t cumulative_loss_rtcp_offset_;
+
+  int64_t last_receive_time_ms_;
+  uint32_t last_received_timestamp_;
+  RtpSequenceNumberUnwrapper seq_unwrapper_;
+  int64_t received_seq_first_;
+  int64_t received_seq_max_;
+  // Assume that the other side restarted when there are two sequential packets
+  // with large jump from received_seq_max_.
+  absl::optional<uint16_t> received_seq_out_of_order_;
+
+  // Current counter values.
+  StreamDataCounters receive_counters_;
+
+  // Counter values when we sent the last report.
+  int32_t last_report_cumulative_loss_;
+  int64_t last_report_seq_max_;
+
+  // The sample frequency of the last received packet.
+  int last_payload_type_frequency_;
+};
+
+// Thread-safe implementation of StreamStatisticianImplInterface.
+class StreamStatisticianLocked : public StreamStatisticianImplInterface {
+ public:
+  StreamStatisticianLocked(uint32_t ssrc,
+                           Clock* clock,
+                           int max_reordering_threshold)
+      : impl_(ssrc, clock, max_reordering_threshold) {}
+  ~StreamStatisticianLocked() override = default;
+
+  RtpReceiveStats GetStats() const override {
+    MutexLock lock(&stream_lock_);
+    return impl_.GetStats();
+  }
+  absl::optional<int> GetFractionLostInPercent() const override {
+    MutexLock lock(&stream_lock_);
+    return impl_.GetFractionLostInPercent();
+  }
+  StreamDataCounters GetReceiveStreamDataCounters() const override {
+    MutexLock lock(&stream_lock_);
+    return impl_.GetReceiveStreamDataCounters();
+  }
+  uint32_t BitrateReceived() const override {
+    MutexLock lock(&stream_lock_);
+    return impl_.BitrateReceived();
+  }
+  void MaybeAppendReportBlockAndReset(
+      std::vector<rtcp::ReportBlock>& report_blocks) override {
+    MutexLock lock(&stream_lock_);
+    impl_.MaybeAppendReportBlockAndReset(report_blocks);
+  }
+  void SetMaxReorderingThreshold(int max_reordering_threshold) override {
+    MutexLock lock(&stream_lock_);
+    return impl_.SetMaxReorderingThreshold(max_reordering_threshold);
+  }
+  void EnableRetransmitDetection(bool enable) override {
+    MutexLock lock(&stream_lock_);
+    return impl_.EnableRetransmitDetection(enable);
+  }
+  void UpdateCounters(const RtpPacketReceived& packet) override {
+    MutexLock lock(&stream_lock_);
+    return impl_.UpdateCounters(packet);
+  }
+
+ private:
+  mutable Mutex stream_lock_;
+  StreamStatisticianImpl impl_ RTC_GUARDED_BY(&stream_lock_);
+};
+
+// Thread-compatible implementation.
+class ReceiveStatisticsImpl : public ReceiveStatistics {
+ public:
+  ReceiveStatisticsImpl(
+      Clock* clock,
+      std::function<std::unique_ptr<StreamStatisticianImplInterface>(
+          uint32_t ssrc,
+          Clock* clock,
+          int max_reordering_threshold)> stream_statistician_factory);
+  ~ReceiveStatisticsImpl() override = default;
+
+  // Implements ReceiveStatisticsProvider.
+  std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override;
+
+  // Implements RtpPacketSinkInterface
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+
+  // Implements ReceiveStatistics.
+  StreamStatistician* GetStatistician(uint32_t ssrc) const override;
+  void SetMaxReorderingThreshold(int max_reordering_threshold) override;
+  void SetMaxReorderingThreshold(uint32_t ssrc,
+                                 int max_reordering_threshold) override;
+  void EnableRetransmitDetection(uint32_t ssrc, bool enable) override;
+
+ private:
+  StreamStatisticianImplInterface* GetOrCreateStatistician(uint32_t ssrc);
+
+  Clock* const clock_;
+  std::function<std::unique_ptr<StreamStatisticianImplInterface>(
+      uint32_t ssrc,
+      Clock* clock,
+      int max_reordering_threshold)>
+      stream_statistician_factory_;
+  // The index within `all_ssrcs_` that was last returned.
+  size_t last_returned_ssrc_idx_;
+  std::vector<uint32_t> all_ssrcs_;
+  int max_reordering_threshold_;
+  flat_map<uint32_t /*ssrc*/, std::unique_ptr<StreamStatisticianImplInterface>>
+      statisticians_;
+};
+
+// Thread-safe implementation wrapping access to ReceiveStatisticsImpl with a
+// mutex.
+class ReceiveStatisticsLocked : public ReceiveStatistics {
+ public:
+  explicit ReceiveStatisticsLocked(
+      Clock* clock,
+      std::function<std::unique_ptr<StreamStatisticianImplInterface>(
+          uint32_t ssrc,
+          Clock* clock,
+          int max_reordering_threshold)> stream_statitician_factory)
+      : impl_(clock, std::move(stream_statitician_factory)) {}
+  ~ReceiveStatisticsLocked() override = default;
+  std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.RtcpReportBlocks(max_blocks);
+  }
+  void OnRtpPacket(const RtpPacketReceived& packet) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.OnRtpPacket(packet);
+  }
+  StreamStatistician* GetStatistician(uint32_t ssrc) const override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.GetStatistician(ssrc);
+  }
+  void SetMaxReorderingThreshold(int max_reordering_threshold) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.SetMaxReorderingThreshold(max_reordering_threshold);
+  }
+  void SetMaxReorderingThreshold(uint32_t ssrc,
+                                 int max_reordering_threshold) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.SetMaxReorderingThreshold(ssrc, max_reordering_threshold);
+  }
+  void EnableRetransmitDetection(uint32_t ssrc, bool enable) override {
+    MutexLock lock(&receive_statistics_lock_);
+    return impl_.EnableRetransmitDetection(ssrc, enable);
+  }
+
+ private:
+  mutable Mutex receive_statistics_lock_;
+  ReceiveStatisticsImpl impl_ RTC_GUARDED_BY(&receive_statistics_lock_);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
new file mode 100644
index 0000000000..92c8f34196
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -0,0 +1,901 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+
+#include <cstdint>
+#include <memory>
+#include <vector>
+
+#include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/random.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAre;
+
+const size_t kPacketSize1 = 100;
+const size_t kPacketSize2 = 300;
+const uint32_t kSsrc1 = 101;
+const uint32_t kSsrc2 = 202;
+const uint32_t kSsrc3 = 203;
+const uint32_t kSsrc4 = 304;
+
+RtpPacketReceived CreateRtpPacket(uint32_t ssrc,
+                                  size_t header_size,
+                                  size_t payload_size,
+                                  size_t padding_size) {
+  RtpPacketReceived packet;
+  packet.SetSsrc(ssrc);
+  packet.SetSequenceNumber(100);
+  packet.set_payload_type_frequency(90000);
+  RTC_CHECK_GE(header_size, 12);
+  RTC_CHECK_EQ(header_size % 4, 0);
+  if (header_size > 12) {
+    // Insert csrcs to increase header size.
+    const int num_csrcs = (header_size - 12) / 4;
+    std::vector<uint32_t> csrcs(num_csrcs);
+    packet.SetCsrcs(csrcs);
+  }
+  packet.SetPayloadSize(payload_size);
+  packet.SetPadding(padding_size);
+  return packet;
+}
+
+RtpPacketReceived MakeRtpPacket(int payload_type_frequency,
+                                uint32_t timestamp) {
+  RtpPacketReceived packet =
+      CreateRtpPacket(kSsrc1,
+                      /*header_size=*/12, kPacketSize1 - 12,
+                      /*padding_size=*/0);
+  packet.SetTimestamp(timestamp);
+  packet.set_payload_type_frequency(payload_type_frequency);
+  return packet;
+}
+
+RtpPacketReceived MakeNextRtpPacket(const RtpPacketReceived& previous_packet,
+                                    int payload_type_frequency,
+                                    uint32_t timestamp) {
+  RtpPacketReceived packet = MakeRtpPacket(payload_type_frequency, timestamp);
+  packet.SetSequenceNumber(previous_packet.SequenceNumber() + 1);
+  return packet;
+}
+
+RtpPacketReceived CreateRtpPacket(uint32_t ssrc, size_t packet_size) {
+  return CreateRtpPacket(ssrc, 12, packet_size - 12, 0);
+}
+
+void IncrementSequenceNumber(RtpPacketReceived* packet, uint16_t incr) {
+  packet->SetSequenceNumber(packet->SequenceNumber() + incr);
+}
+
+void IncrementSequenceNumber(RtpPacketReceived* packet) {
+  IncrementSequenceNumber(packet, 1);
+}
+
+uint32_t GetJitter(const ReceiveStatistics& stats) {
+  return stats.GetStatistician(kSsrc1)->GetStats().jitter;
+}
+
+class ReceiveStatisticsTest : public ::testing::TestWithParam<bool> {
+ public:
+  ReceiveStatisticsTest()
+      : clock_(0),
+        receive_statistics_(
+            GetParam() ? ReceiveStatistics::Create(&clock_)
+                       : ReceiveStatistics::CreateThreadCompatible(&clock_)) {
+    packet1_ = CreateRtpPacket(kSsrc1, kPacketSize1);
+    packet2_ = CreateRtpPacket(kSsrc2, kPacketSize2);
+  }
+
+ protected:
+  SimulatedClock clock_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  RtpPacketReceived packet1_;
+  RtpPacketReceived packet2_;
+};
+
+INSTANTIATE_TEST_SUITE_P(All,
+                         ReceiveStatisticsTest,
+                         ::testing::Bool(),
+                         [](::testing::TestParamInfo<bool> info) {
+                           return info.param ? "WithMutex" : "WithoutMutex";
+                         });
+
+TEST_P(ReceiveStatisticsTest, TwoIncomingSsrcs) {
+  receive_statistics_->OnRtpPacket(packet1_);
+  IncrementSequenceNumber(&packet1_);
+  receive_statistics_->OnRtpPacket(packet2_);
+  IncrementSequenceNumber(&packet2_);
+  clock_.AdvanceTimeMilliseconds(100);
+  receive_statistics_->OnRtpPacket(packet1_);
+  IncrementSequenceNumber(&packet1_);
+  receive_statistics_->OnRtpPacket(packet2_);
+  IncrementSequenceNumber(&packet2_);
+
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  ASSERT_TRUE(statistician != NULL);
+  EXPECT_GT(statistician->BitrateReceived(), 0u);
+  StreamDataCounters counters = statistician->GetReceiveStreamDataCounters();
+  EXPECT_EQ(176u, counters.transmitted.payload_bytes);
+  EXPECT_EQ(24u, counters.transmitted.header_bytes);
+  EXPECT_EQ(0u, counters.transmitted.padding_bytes);
+  EXPECT_EQ(2u, counters.transmitted.packets);
+
+  statistician = receive_statistics_->GetStatistician(kSsrc2);
+  ASSERT_TRUE(statistician != NULL);
+  EXPECT_GT(statistician->BitrateReceived(), 0u);
+  counters = statistician->GetReceiveStreamDataCounters();
+  EXPECT_EQ(576u, counters.transmitted.payload_bytes);
+  EXPECT_EQ(24u, counters.transmitted.header_bytes);
+  EXPECT_EQ(0u, counters.transmitted.padding_bytes);
+  EXPECT_EQ(2u, counters.transmitted.packets);
+
+  EXPECT_EQ(2u, receive_statistics_->RtcpReportBlocks(3).size());
+  // Add more incoming packets and verify that they are registered in both
+  // access methods.
+  receive_statistics_->OnRtpPacket(packet1_);
+  IncrementSequenceNumber(&packet1_);
+  receive_statistics_->OnRtpPacket(packet2_);
+  IncrementSequenceNumber(&packet2_);
+
+  counters = receive_statistics_->GetStatistician(kSsrc1)
+                 ->GetReceiveStreamDataCounters();
+  EXPECT_EQ(264u, counters.transmitted.payload_bytes);
+  EXPECT_EQ(36u, counters.transmitted.header_bytes);
+  EXPECT_EQ(0u, counters.transmitted.padding_bytes);
+  EXPECT_EQ(3u, counters.transmitted.packets);
+
+  counters = receive_statistics_->GetStatistician(kSsrc2)
+                 ->GetReceiveStreamDataCounters();
+  EXPECT_EQ(864u, counters.transmitted.payload_bytes);
+  EXPECT_EQ(36u, counters.transmitted.header_bytes);
+  EXPECT_EQ(0u, counters.transmitted.padding_bytes);
+  EXPECT_EQ(3u, counters.transmitted.packets);
+}
+
+TEST_P(ReceiveStatisticsTest,
+       RtcpReportBlocksReturnsMaxBlocksWhenThereAreMoreStatisticians) {
+  RtpPacketReceived packet1 = CreateRtpPacket(kSsrc1, kPacketSize1);
+  RtpPacketReceived packet2 = CreateRtpPacket(kSsrc2, kPacketSize1);
+  RtpPacketReceived packet3 = CreateRtpPacket(kSsrc3, kPacketSize1);
+  receive_statistics_->OnRtpPacket(packet1);
+  receive_statistics_->OnRtpPacket(packet2);
+  receive_statistics_->OnRtpPacket(packet3);
+
+  EXPECT_THAT(receive_statistics_->RtcpReportBlocks(2), SizeIs(2));
+  EXPECT_THAT(receive_statistics_->RtcpReportBlocks(2), SizeIs(2));
+  EXPECT_THAT(receive_statistics_->RtcpReportBlocks(2), SizeIs(2));
+}
+
+TEST_P(ReceiveStatisticsTest,
+       RtcpReportBlocksReturnsAllObservedSsrcsWithMultipleCalls) {
+  RtpPacketReceived packet1 = CreateRtpPacket(kSsrc1, kPacketSize1);
+  RtpPacketReceived packet2 = CreateRtpPacket(kSsrc2, kPacketSize1);
+  RtpPacketReceived packet3 = CreateRtpPacket(kSsrc3, kPacketSize1);
+  RtpPacketReceived packet4 = CreateRtpPacket(kSsrc4, kPacketSize1);
+  receive_statistics_->OnRtpPacket(packet1);
+  receive_statistics_->OnRtpPacket(packet2);
+  receive_statistics_->OnRtpPacket(packet3);
+  receive_statistics_->OnRtpPacket(packet4);
+
+  std::vector<uint32_t> observed_ssrcs;
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(2);
+  ASSERT_THAT(report_blocks, SizeIs(2));
+  observed_ssrcs.push_back(report_blocks[0].source_ssrc());
+  observed_ssrcs.push_back(report_blocks[1].source_ssrc());
+
+  report_blocks = receive_statistics_->RtcpReportBlocks(2);
+  ASSERT_THAT(report_blocks, SizeIs(2));
+  observed_ssrcs.push_back(report_blocks[0].source_ssrc());
+  observed_ssrcs.push_back(report_blocks[1].source_ssrc());
+
+  EXPECT_THAT(observed_ssrcs,
+              UnorderedElementsAre(kSsrc1, kSsrc2, kSsrc3, kSsrc4));
+}
+
+TEST_P(ReceiveStatisticsTest, ActiveStatisticians) {
+  receive_statistics_->OnRtpPacket(packet1_);
+  IncrementSequenceNumber(&packet1_);
+  clock_.AdvanceTimeMilliseconds(1000);
+  receive_statistics_->OnRtpPacket(packet2_);
+  IncrementSequenceNumber(&packet2_);
+  // Nothing should time out since only 1000 ms has passed since the first
+  // packet came in.
+  EXPECT_EQ(2u, receive_statistics_->RtcpReportBlocks(3).size());
+
+  clock_.AdvanceTimeMilliseconds(7000);
+  // kSsrc1 should have timed out.
+  EXPECT_EQ(1u, receive_statistics_->RtcpReportBlocks(3).size());
+
+  clock_.AdvanceTimeMilliseconds(1000);
+  // kSsrc2 should have timed out.
+  EXPECT_EQ(0u, receive_statistics_->RtcpReportBlocks(3).size());
+
+  receive_statistics_->OnRtpPacket(packet1_);
+  IncrementSequenceNumber(&packet1_);
+  // kSsrc1 should be active again and the data counters should have survived.
+  EXPECT_EQ(1u, receive_statistics_->RtcpReportBlocks(3).size());
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  ASSERT_TRUE(statistician != NULL);
+  StreamDataCounters counters = statistician->GetReceiveStreamDataCounters();
+  EXPECT_EQ(176u, counters.transmitted.payload_bytes);
+  EXPECT_EQ(24u, counters.transmitted.header_bytes);
+  EXPECT_EQ(0u, counters.transmitted.padding_bytes);
+  EXPECT_EQ(2u, counters.transmitted.packets);
+}
+
+TEST_P(ReceiveStatisticsTest,
+       DoesntCreateRtcpReportBlockUntilFirstReceivedPacketForSsrc) {
+  // Creates a statistician object for the ssrc.
+  receive_statistics_->EnableRetransmitDetection(kSsrc1, true);
+  EXPECT_TRUE(receive_statistics_->GetStatistician(kSsrc1) != nullptr);
+  EXPECT_EQ(0u, receive_statistics_->RtcpReportBlocks(3).size());
+  // Receive first packet
+  receive_statistics_->OnRtpPacket(packet1_);
+  EXPECT_EQ(1u, receive_statistics_->RtcpReportBlocks(3).size());
+}
+
+TEST_P(ReceiveStatisticsTest, GetReceiveStreamDataCounters) {
+  receive_statistics_->OnRtpPacket(packet1_);
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  ASSERT_TRUE(statistician != NULL);
+
+  StreamDataCounters counters = statistician->GetReceiveStreamDataCounters();
+  EXPECT_GT(counters.first_packet_time_ms, -1);
+  EXPECT_EQ(1u, counters.transmitted.packets);
+
+  receive_statistics_->OnRtpPacket(packet1_);
+  counters = statistician->GetReceiveStreamDataCounters();
+  EXPECT_GT(counters.first_packet_time_ms, -1);
+  EXPECT_EQ(2u, counters.transmitted.packets);
+}
+
+TEST_P(ReceiveStatisticsTest, SimpleLossComputation) {
+  packet1_.SetSequenceNumber(1);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(3);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(4);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(5);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  // 20% = 51/255.
+  EXPECT_EQ(51u, report_blocks[0].fraction_lost());
+  EXPECT_EQ(1, report_blocks[0].cumulative_lost_signed());
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  EXPECT_EQ(20, statistician->GetFractionLostInPercent());
+}
+
+TEST_P(ReceiveStatisticsTest, LossComputationWithReordering) {
+  packet1_.SetSequenceNumber(1);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(3);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(2);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(5);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  // 20% = 51/255.
+  EXPECT_EQ(51u, report_blocks[0].fraction_lost());
+  EXPECT_EQ(1, report_blocks[0].cumulative_lost_signed());
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  EXPECT_EQ(20, statistician->GetFractionLostInPercent());
+}
+
+TEST_P(ReceiveStatisticsTest, LossComputationWithDuplicates) {
+  // Lose 2 packets, but also receive 1 duplicate. Should actually count as
+  // only 1 packet being lost.
+  packet1_.SetSequenceNumber(1);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(4);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(4);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(5);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  // 20% = 51/255.
+  EXPECT_EQ(51u, report_blocks[0].fraction_lost());
+  EXPECT_EQ(1, report_blocks[0].cumulative_lost_signed());
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  EXPECT_EQ(20, statistician->GetFractionLostInPercent());
+}
+
+TEST_P(ReceiveStatisticsTest, LossComputationWithSequenceNumberWrapping) {
+  // First, test loss computation over a period that included a sequence number
+  // rollover.
+  packet1_.SetSequenceNumber(0xfffd);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(0);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(0xfffe);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(1);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  // Only one packet was actually lost, 0xffff.
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  // 20% = 51/255.
+  EXPECT_EQ(51u, report_blocks[0].fraction_lost());
+  EXPECT_EQ(1, report_blocks[0].cumulative_lost_signed());
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  EXPECT_EQ(20, statistician->GetFractionLostInPercent());
+
+  // Now test losing one packet *after* the rollover.
+  packet1_.SetSequenceNumber(3);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  report_blocks = receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  // 50% = 127/255.
+  EXPECT_EQ(127u, report_blocks[0].fraction_lost());
+  EXPECT_EQ(2, report_blocks[0].cumulative_lost_signed());
+  // 2 packets lost, 7 expected
+  EXPECT_EQ(28, statistician->GetFractionLostInPercent());
+}
+
+TEST_P(ReceiveStatisticsTest, StreamRestartDoesntCountAsLoss) {
+  receive_statistics_->SetMaxReorderingThreshold(kSsrc1, 200);
+
+  packet1_.SetSequenceNumber(0);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(1);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  packet1_.SetSequenceNumber(400);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(0, report_blocks[0].fraction_lost());
+  EXPECT_EQ(0, report_blocks[0].cumulative_lost_signed());
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  EXPECT_EQ(0, statistician->GetFractionLostInPercent());
+
+  packet1_.SetSequenceNumber(401);
+  receive_statistics_->OnRtpPacket(packet1_);
+  report_blocks = receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(0, report_blocks[0].fraction_lost());
+  EXPECT_EQ(0, report_blocks[0].cumulative_lost_signed());
+  EXPECT_EQ(0, statistician->GetFractionLostInPercent());
+}
+
+TEST_P(ReceiveStatisticsTest, CountsLossAfterStreamRestart) {
+  receive_statistics_->SetMaxReorderingThreshold(kSsrc1, 200);
+
+  packet1_.SetSequenceNumber(0);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(1);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  packet1_.SetSequenceNumber(400);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(401);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(403);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(1, report_blocks[0].cumulative_lost_signed());
+
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  // Is this reasonable? */
+  EXPECT_EQ(0, statistician->GetFractionLostInPercent());
+}
+
+TEST_P(ReceiveStatisticsTest, StreamCanRestartAtSequenceNumberWrapAround) {
+  receive_statistics_->SetMaxReorderingThreshold(kSsrc1, 200);
+
+  packet1_.SetSequenceNumber(0xffff - 401);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(0xffff - 400);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  packet1_.SetSequenceNumber(0xffff);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(0);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(2);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(1, report_blocks[0].cumulative_lost_signed());
+}
+
+TEST_P(ReceiveStatisticsTest, StreamRestartNeedsTwoConsecutivePackets) {
+  receive_statistics_->SetMaxReorderingThreshold(kSsrc1, 200);
+
+  packet1_.SetSequenceNumber(400);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(401);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  packet1_.SetSequenceNumber(1);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(3);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(401u, report_blocks[0].extended_high_seq_num());
+
+  packet1_.SetSequenceNumber(4);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  report_blocks = receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(4u, report_blocks[0].extended_high_seq_num());
+}
+
+TEST_P(ReceiveStatisticsTest, WrapsAroundExtendedHighestSequenceNumber) {
+  packet1_.SetSequenceNumber(0xffff);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  std::vector<rtcp::ReportBlock> report_blocks =
+      receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(0xffffu, report_blocks[0].extended_high_seq_num());
+
+  // Wrap around.
+  packet1_.SetSequenceNumber(1);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  report_blocks = receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(0x10001u, report_blocks[0].extended_high_seq_num());
+
+  // Should be treated as out of order; shouldn't increment highest extended
+  // sequence number.
+  packet1_.SetSequenceNumber(0x10000 - 6);
+  report_blocks = receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(0x10001u, report_blocks[0].extended_high_seq_num());
+
+  // Receive a couple packets then wrap around again.
+  receive_statistics_->SetMaxReorderingThreshold(kSsrc1, 200);
+  for (int i = 10; i < 0xffff; i += 150) {
+    packet1_.SetSequenceNumber(i);
+    receive_statistics_->OnRtpPacket(packet1_);
+  }
+  packet1_.SetSequenceNumber(1);
+  receive_statistics_->OnRtpPacket(packet1_);
+  report_blocks = receive_statistics_->RtcpReportBlocks(1);
+  ASSERT_THAT(report_blocks, SizeIs(1));
+  EXPECT_EQ(kSsrc1, report_blocks[0].source_ssrc());
+
+  EXPECT_EQ(0x20001u, report_blocks[0].extended_high_seq_num());
+}
+
+TEST_P(ReceiveStatisticsTest, StreamDataCounters) {
+  receive_statistics_->EnableRetransmitDetection(kSsrc1, true);
+
+  const size_t kHeaderLength = 20;
+  const size_t kPaddingLength = 9;
+
+  // One packet with payload size kPacketSize1.
+  RtpPacketReceived packet1 =
+      CreateRtpPacket(kSsrc1, kHeaderLength, kPacketSize1, 0);
+  receive_statistics_->OnRtpPacket(packet1);
+  StreamDataCounters counters = receive_statistics_->GetStatistician(kSsrc1)
+                                    ->GetReceiveStreamDataCounters();
+  EXPECT_EQ(counters.transmitted.payload_bytes, kPacketSize1);
+  EXPECT_EQ(counters.transmitted.header_bytes, kHeaderLength);
+  EXPECT_EQ(counters.transmitted.padding_bytes, 0u);
+  EXPECT_EQ(counters.transmitted.packets, 1u);
+  EXPECT_EQ(counters.retransmitted.payload_bytes, 0u);
+  EXPECT_EQ(counters.retransmitted.header_bytes, 0u);
+  EXPECT_EQ(counters.retransmitted.padding_bytes, 0u);
+  EXPECT_EQ(counters.retransmitted.packets, 0u);
+  EXPECT_EQ(counters.fec.packets, 0u);
+
+  // Another packet of size kPacketSize1 with 9 bytes padding.
+  RtpPacketReceived packet2 =
+      CreateRtpPacket(kSsrc1, kHeaderLength, kPacketSize1, 9);
+  packet2.SetSequenceNumber(packet1.SequenceNumber() + 1);
+  clock_.AdvanceTimeMilliseconds(5);
+  receive_statistics_->OnRtpPacket(packet2);
+  counters = receive_statistics_->GetStatistician(kSsrc1)
+                 ->GetReceiveStreamDataCounters();
+  EXPECT_EQ(counters.transmitted.payload_bytes, kPacketSize1 * 2);
+  EXPECT_EQ(counters.transmitted.header_bytes, kHeaderLength * 2);
+  EXPECT_EQ(counters.transmitted.padding_bytes, kPaddingLength);
+  EXPECT_EQ(counters.transmitted.packets, 2u);
+
+  clock_.AdvanceTimeMilliseconds(5);
+  // Retransmit last packet.
+  receive_statistics_->OnRtpPacket(packet2);
+  counters = receive_statistics_->GetStatistician(kSsrc1)
+                 ->GetReceiveStreamDataCounters();
+  EXPECT_EQ(counters.transmitted.payload_bytes, kPacketSize1 * 3);
+  EXPECT_EQ(counters.transmitted.header_bytes, kHeaderLength * 3);
+  EXPECT_EQ(counters.transmitted.padding_bytes, kPaddingLength * 2);
+  EXPECT_EQ(counters.transmitted.packets, 3u);
+  EXPECT_EQ(counters.retransmitted.payload_bytes, kPacketSize1);
+  EXPECT_EQ(counters.retransmitted.header_bytes, kHeaderLength);
+  EXPECT_EQ(counters.retransmitted.padding_bytes, kPaddingLength);
+  EXPECT_EQ(counters.retransmitted.packets, 1u);
+}
+
+TEST_P(ReceiveStatisticsTest, LastPacketReceivedTimestamp) {
+  clock_.AdvanceTimeMilliseconds(42);
+  receive_statistics_->OnRtpPacket(packet1_);
+  StreamDataCounters counters = receive_statistics_->GetStatistician(kSsrc1)
+                                    ->GetReceiveStreamDataCounters();
+
+  EXPECT_EQ(42, counters.last_packet_received_timestamp_ms);
+
+  clock_.AdvanceTimeMilliseconds(3);
+  receive_statistics_->OnRtpPacket(packet1_);
+  counters = receive_statistics_->GetStatistician(kSsrc1)
+                 ->GetReceiveStreamDataCounters();
+  EXPECT_EQ(45, counters.last_packet_received_timestamp_ms);
+}
+
+TEST_P(ReceiveStatisticsTest, SimpleJitterComputation) {
+  const int kMsPerPacket = 20;
+  const int kCodecSampleRate = 48'000;
+  const int kSamplesPerPacket = kMsPerPacket * kCodecSampleRate / 1'000;
+  const int kLateArrivalDeltaMs = 100;
+  const int kLateArrivalDeltaSamples =
+      kLateArrivalDeltaMs * kCodecSampleRate / 1'000;
+
+  packet1_.set_payload_type_frequency(kCodecSampleRate);
+  packet1_.SetSequenceNumber(1);
+  packet1_.SetTimestamp(0);
+  receive_statistics_->OnRtpPacket(packet1_);
+  packet1_.SetSequenceNumber(2);
+  packet1_.SetTimestamp(kSamplesPerPacket);
+  // Arrives 100 ms late.
+  clock_.AdvanceTimeMilliseconds(kMsPerPacket + kLateArrivalDeltaMs);
+  receive_statistics_->OnRtpPacket(packet1_);
+
+  StreamStatistician* statistician =
+      receive_statistics_->GetStatistician(kSsrc1);
+  // See jitter caluculation in https://www.rfc-editor.org/rfc/rfc3550 6.4.1.
+  const uint32_t expected_jitter = (kLateArrivalDeltaSamples) / 16;
+  EXPECT_EQ(expected_jitter, statistician->GetStats().jitter);
+  EXPECT_EQ(webrtc::TimeDelta::Seconds(expected_jitter) / kCodecSampleRate,
+            statistician->GetStats().interarrival_jitter);
+}
+
+TEST(ReviseJitterTest, AllPacketsHaveSamePayloadTypeFrequency) {
+  SimulatedClock clock(0);
+  std::unique_ptr<ReceiveStatistics> statistics =
+      ReceiveStatistics::Create(&clock);
+  RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/8'000,
+                                            /*timestamp=*/1);
+  RtpPacketReceived packet2 = MakeNextRtpPacket(
+      packet1, /*payload_type_frequency=*/8'000, /*timestamp=*/1 + 160);
+
+  RtpPacketReceived packet3 = MakeNextRtpPacket(
+      packet2, /*payload_type_frequency=*/8'000, /*timestamp=*/1 + 2 * 160);
+
+  statistics->OnRtpPacket(packet1);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet2);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet3);
+
+  // packet1: no jitter calculation
+  // packet2: jitter = 0[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 0[jitter] + 8)
+  //          / 16 = 240
+  // packet3: jitter = 240[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 240[jitter] + 8)
+  //          / 16 = 465
+  // final jitter: 465 / 16 = 29
+  EXPECT_EQ(GetJitter(*statistics), 29U);
+}
+
+TEST(ReviseJitterTest, AllPacketsHaveDifferentPayloadTypeFrequency) {
+  SimulatedClock clock(0);
+  std::unique_ptr<ReceiveStatistics> statistics =
+      ReceiveStatistics::Create(&clock);
+  RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/8'000,
+                                            /*timestamp=*/1);
+  RtpPacketReceived packet2 = MakeNextRtpPacket(
+      packet1, /*payload_type_frequency=*/8'000, /*timestamp=*/1 + 160);
+  RtpPacketReceived packet3 = MakeNextRtpPacket(
+      packet2, /*payload_type_frequency=*/48'000, /*timestamp=*/1 + 160 + 960);
+
+  statistics->OnRtpPacket(packet1);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet2);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet3);
+
+  // packet1: no jitter calculation
+  // packet2: jitter = 0[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 0[jitter] + 8)
+  //          / 16 = 240
+  // packet3: revised jitter: 240 * 48[frequency KHz] / 8[frequency KHz] = 1'440
+  //          jitter = 1'440[jitter] + (abs(50[receive time ms] *
+  //          48[frequency KHz] - 960[timestamp diff]) * 16 - 1'440[jitter] + 8)
+  //          / 16 = 2'790
+  // final jitter: 2'790 / 16 = 174
+  EXPECT_EQ(GetJitter(*statistics), 174U);
+}
+
+TEST(ReviseJitterTest,
+     FirstPacketPayloadTypeFrequencyIsZeroAndFrequencyChanged) {
+  SimulatedClock clock(0);
+  std::unique_ptr<ReceiveStatistics> statistics =
+      ReceiveStatistics::Create(&clock);
+  RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/0,
+                                            /*timestamp=*/1);
+  RtpPacketReceived packet2 = MakeNextRtpPacket(
+      packet1, /*payload_type_frequency=*/8'000, /*timestamp=*/1 + 160);
+  RtpPacketReceived packet3 = MakeNextRtpPacket(
+      packet2, /*payload_type_frequency=*/48'000, /*timestamp=*/1 + 160 + 960);
+
+  statistics->OnRtpPacket(packet1);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet2);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet3);
+
+  // packet1: no jitter calculation
+  // packet2: jitter = 0[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 0[jitter] + 8)
+  //          / 16 = 240
+  // packet3: revised jitter: 240 * 48[frequency KHz] / 8[frequency KHz] = 1'440
+  //          jitter = 1'440[jitter] + (abs(50[receive time ms] *
+  //          48[frequency KHz] - 960[timestamp diff]) * 16 - 1'440[jitter] + 8)
+  //          / 16 = 2'790
+  // final jitter: 2'790 / 16 = 174
+  EXPECT_EQ(GetJitter(*statistics), 174U);
+}
+
+TEST(ReviseJitterTest,
+     FirstPacketPayloadTypeFrequencyIsZeroAndFrequencyNotChanged) {
+  SimulatedClock clock(0);
+  std::unique_ptr<ReceiveStatistics> statistics =
+      ReceiveStatistics::Create(&clock);
+  RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/0,
+                                            /*timestamp=*/1);
+  RtpPacketReceived packet2 = MakeNextRtpPacket(
+      packet1, /*payload_type_frequency=*/8'000, /*timestamp=*/1 + 160);
+  RtpPacketReceived packet3 = MakeNextRtpPacket(
+      packet2, /*payload_type_frequency=*/8'000, /*timestamp=*/1 + 160 + 160);
+
+  statistics->OnRtpPacket(packet1);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet2);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet3);
+
+  // packet1: no jitter calculation
+  // packet2: jitter = 0[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 0[jitter] + 8)
+  //          / 16 = 240
+  // packet3: jitter = 240[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 240[jitter] + 8)
+  //          / 16 = 465
+  // final jitter: 465 / 16 = 29
+  EXPECT_EQ(GetJitter(*statistics), 29U);
+}
+
+TEST(ReviseJitterTest,
+     TwoFirstPacketPayloadTypeFrequencyIsZeroAndFrequencyChanged) {
+  SimulatedClock clock(0);
+  std::unique_ptr<ReceiveStatistics> statistics =
+      ReceiveStatistics::Create(&clock);
+  RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/0,
+                                            /*timestamp=*/1);
+  RtpPacketReceived packet2 = MakeNextRtpPacket(
+      packet1, /*payload_type_frequency=*/0, /*timestamp=*/1 + 160);
+  RtpPacketReceived packet3 = MakeNextRtpPacket(
+      packet2, /*payload_type_frequency=*/48'000, /*timestamp=*/1 + 160 + 960);
+  RtpPacketReceived packet4 =
+      MakeNextRtpPacket(packet3, /*payload_type_frequency=*/8'000,
+                        /*timestamp=*/1 + 160 + 960 + 160);
+
+  statistics->OnRtpPacket(packet1);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet2);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet3);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet4);
+
+  // packet1: no jitter calculation
+  // packet2: jitter = 0[jitter] + (abs(50[receive time ms] *
+  //          0[frequency KHz] - 160[timestamp diff]) * 16 - 0[jitter] + 8)
+  //          / 16 = 160
+  // packet3: jitter = 160[jitter] + (abs(50[receive time ms] *
+  //          48[frequency KHz] - 960[timestamp diff]) * 16 - 160[jitter] + 8)
+  //          / 16 = 1'590
+  // packet4: revised jitter: 1'590 * 8[frequency KHz] / 48[frequency KHz] = 265
+  // packet4: jitter = 265[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 265[jitter] + 8)
+  //          / 16 = 488
+  // final jitter: 488 / 16 = 30
+  EXPECT_EQ(GetJitter(*statistics), 30U);
+}
+
+TEST(ReviseJitterTest,
+     TwoFirstPacketPayloadTypeFrequencyIsZeroAndFrequencyNotChanged) {
+  SimulatedClock clock(0);
+  std::unique_ptr<ReceiveStatistics> statistics =
+      ReceiveStatistics::Create(&clock);
+  RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/0,
+                                            /*timestamp=*/1);
+  RtpPacketReceived packet2 = MakeNextRtpPacket(
+      packet1, /*payload_type_frequency=*/0, /*timestamp=*/1 + 160);
+  RtpPacketReceived packet3 = MakeNextRtpPacket(
+      packet2, /*payload_type_frequency=*/8'000, /*timestamp=*/1 + 160 + 160);
+  RtpPacketReceived packet4 =
+      MakeNextRtpPacket(packet3, /*payload_type_frequency=*/8'000,
+                        /*timestamp=*/1 + 160 + 160 + 160);
+
+  statistics->OnRtpPacket(packet1);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet2);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet3);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet4);
+
+  // packet1: no jitter calculation
+  // packet2: jitter = 0[jitter] + (abs(50[receive time ms] *
+  //          0[frequency KHz] - 160[timestamp diff]) * 16 - 0[jitter] + 8)
+  //          / 16 = 160
+  // packet3: jitter = 160[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 160[jitter] + 8)
+  //          / 16 = 390
+  // packet4: jitter = 390[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 390[jitter] + 8)
+  //          / 16 = 606
+  // final jitter: 606 / 16 = 37
+  EXPECT_EQ(GetJitter(*statistics), 37U);
+}
+
+TEST(ReviseJitterTest,
+     MiddlePacketPayloadTypeFrequencyIsZeroAndFrequencyChanged) {
+  SimulatedClock clock(0);
+  std::unique_ptr<ReceiveStatistics> statistics =
+      ReceiveStatistics::Create(&clock);
+  RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/48'000,
+                                            /*timestamp=*/1);
+  RtpPacketReceived packet2 = MakeNextRtpPacket(
+      packet1, /*payload_type_frequency=*/48'000, /*timestamp=*/1 + 960);
+  RtpPacketReceived packet3 = MakeNextRtpPacket(
+      packet2, /*payload_type_frequency=*/0, /*timestamp=*/1 + 960 + 55);
+  RtpPacketReceived packet4 =
+      MakeNextRtpPacket(packet3, /*payload_type_frequency=*/8'000,
+                        /*timestamp=*/1 + 960 + 55 + 160);
+
+  statistics->OnRtpPacket(packet1);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet2);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet3);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet4);
+
+  // packet1: no jitter calculation
+  // packet2: jitter = 0[jitter] + (abs(50[receive time ms] *
+  //          48[frequency KHz] - 960[timestamp diff]) * 16 - 0[jitter] + 8)
+  //          / 16 = 1'440
+  // packet3: jitter = 1'440[jitter] + (abs(50[receive time ms] *
+  //          0[frequency KHz] - 55[timestamp diff]) * 16 - 1'440[jitter] + 8)
+  //          / 16 = 1'405
+  // packet4: revised jitter: 1'405 * 8[frequency KHz] / 48[frequency KHz] = 234
+  //          jitter = 234[jitter] + (abs(50[receive time ms] *
+  //          8[frequency KHz] - 160[timestamp diff]) * 16 - 234[jitter] + 8)
+  //          / 16 = 459
+  // final jitter: 459 / 16 = 28
+  EXPECT_EQ(GetJitter(*statistics), 28U);
+}
+
+TEST(ReviseJitterTest,
+     MiddlePacketPayloadTypeFrequencyIsZeroAndFrequencyNotChanged) {
+  SimulatedClock clock(0);
+  std::unique_ptr<ReceiveStatistics> statistics =
+      ReceiveStatistics::Create(&clock);
+  RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/48'000,
+                                            /*timestamp=*/1);
+  RtpPacketReceived packet2 = MakeNextRtpPacket(
+      packet1, /*payload_type_frequency=*/48'000, /*timestamp=*/1 + 960);
+  RtpPacketReceived packet3 = MakeNextRtpPacket(
+      packet2, /*payload_type_frequency=*/0, /*timestamp=*/1 + 960 + 55);
+  RtpPacketReceived packet4 =
+      MakeNextRtpPacket(packet3, /*payload_type_frequency=*/48'000,
+                        /*timestamp=*/1 + 960 + 55 + 960);
+
+  statistics->OnRtpPacket(packet1);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet2);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet3);
+  clock.AdvanceTimeMilliseconds(50);
+  statistics->OnRtpPacket(packet4);
+
+  // packet1: no jitter calculation
+  // packet2: jitter = 0[jitter] + (abs(50[receive time ms] *
+  //          48[frequency KHz] - 960[timestamp diff]) * 16 - 0[jitter] + 8)
+  //          / 16 = 1'440
+  // packet3: jitter = 1'440[jitter] + (abs(50[receive time ms] *
+  //          0[frequency KHz] - 55[timestamp diff]) * 16 - 1'440[jitter] + 8)
+  //          / 16 = 1'405
+  // packet4: jitter = 1'405[jitter] + (abs(50[receive time ms] *
+  //          48[frequency KHz] - 960[timestamp diff]) * 16 - 1'405[jitter] + 8)
+  //          / 16 = 2'757
+  // final jitter: 2'757 / 16 = 172
+  EXPECT_EQ(GetJitter(*statistics), 172U);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
new file mode 100644
index 0000000000..6f90cd175c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+
+#include <cstdint>
+
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kMinimumNumberOfSamples = 2;
+constexpr TimeDelta kTimingLogInterval = TimeDelta::Seconds(10);
+constexpr int kClocksOffsetSmoothingWindow = 100;
+
+// Subtracts two NtpTime values keeping maximum precision.
+int64_t Subtract(NtpTime minuend, NtpTime subtrahend) {
+  uint64_t a = static_cast<uint64_t>(minuend);
+  uint64_t b = static_cast<uint64_t>(subtrahend);
+  return a >= b ? static_cast<int64_t>(a - b) : -static_cast<int64_t>(b - a);
+}
+
+NtpTime Add(NtpTime lhs, int64_t rhs) {
+  uint64_t result = static_cast<uint64_t>(lhs);
+  if (rhs >= 0) {
+    result += static_cast<uint64_t>(rhs);
+  } else {
+    result -= static_cast<uint64_t>(-rhs);
+  }
+  return NtpTime(result);
+}
+
+}  // namespace
+
+// TODO(wu): Refactor this class so that it can be shared with
+// vie_sync_module.cc.
+RemoteNtpTimeEstimator::RemoteNtpTimeEstimator(Clock* clock)
+    : clock_(clock),
+      ntp_clocks_offset_estimator_(kClocksOffsetSmoothingWindow) {}
+
+bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(TimeDelta rtt,
+                                                 NtpTime sender_send_time,
+                                                 uint32_t rtp_timestamp) {
+  switch (rtp_to_ntp_.UpdateMeasurements(sender_send_time, rtp_timestamp)) {
+    case RtpToNtpEstimator::kInvalidMeasurement:
+      return false;
+    case RtpToNtpEstimator::kSameMeasurement:
+      // No new RTCP SR since last time this function was called.
+      return true;
+    case RtpToNtpEstimator::kNewMeasurement:
+      break;
+  }
+
+  // Assume connection is symmetric and thus time to deliver the packet is half
+  // the round trip time.
+  int64_t deliver_time_ntp = ToNtpUnits(rtt) / 2;
+
+  // Update extrapolator with the new arrival time.
+  NtpTime receiver_arrival_time = clock_->CurrentNtpTime();
+  int64_t remote_to_local_clocks_offset =
+      Subtract(receiver_arrival_time, sender_send_time) - deliver_time_ntp;
+  ntp_clocks_offset_estimator_.Insert(remote_to_local_clocks_offset);
+  return true;
+}
+
+NtpTime RemoteNtpTimeEstimator::EstimateNtp(uint32_t rtp_timestamp) {
+  NtpTime sender_capture = rtp_to_ntp_.Estimate(rtp_timestamp);
+  if (!sender_capture.Valid()) {
+    return sender_capture;
+  }
+
+  int64_t remote_to_local_clocks_offset =
+      ntp_clocks_offset_estimator_.GetFilteredValue();
+  NtpTime receiver_capture = Add(sender_capture, remote_to_local_clocks_offset);
+
+  Timestamp now = clock_->CurrentTime();
+  if (now - last_timing_log_ > kTimingLogInterval) {
+    RTC_LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp
+                     << " in NTP clock: " << sender_capture.ToMs()
+                     << " estimated time in receiver NTP clock: "
+                     << receiver_capture.ToMs();
+    last_timing_log_ = now;
+  }
+
+  return receiver_capture;
+}
+
+absl::optional<int64_t>
+RemoteNtpTimeEstimator::EstimateRemoteToLocalClockOffset() {
+  if (ntp_clocks_offset_estimator_.GetNumberOfSamplesStored() <
+      kMinimumNumberOfSamples) {
+    return absl::nullopt;
+  }
+  return ntp_clocks_offset_estimator_.GetFilteredValue();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
new file mode 100644
index 0000000000..8dbfaec940
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/ntp_time.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr TimeDelta kTestRtt = TimeDelta::Millis(10);
+constexpr Timestamp kLocalClockInitialTime = Timestamp::Millis(123);
+constexpr Timestamp kRemoteClockInitialTime = Timestamp::Millis(373);
+constexpr uint32_t kTimestampOffset = 567;
+constexpr int64_t kRemoteToLocalClockOffsetNtp =
+    ToNtpUnits(kLocalClockInitialTime - kRemoteClockInitialTime);
+
+class RemoteNtpTimeEstimatorTest : public ::testing::Test {
+ protected:
+  void AdvanceTime(TimeDelta delta) {
+    local_clock_.AdvanceTime(delta);
+    remote_clock_.AdvanceTime(delta);
+  }
+
+  uint32_t GetRemoteTimestamp() {
+    return static_cast<uint32_t>(remote_clock_.TimeInMilliseconds()) * 90 +
+           kTimestampOffset;
+  }
+
+  void SendRtcpSr() {
+    uint32_t rtcp_timestamp = GetRemoteTimestamp();
+    NtpTime ntp = remote_clock_.CurrentNtpTime();
+
+    AdvanceTime(kTestRtt / 2);
+    RTC_DCHECK(estimator_.UpdateRtcpTimestamp(kTestRtt, ntp, rtcp_timestamp));
+  }
+
+  void SendRtcpSrInaccurately(TimeDelta ntp_error, TimeDelta networking_delay) {
+    uint32_t rtcp_timestamp = GetRemoteTimestamp();
+    int64_t ntp_error_fractions = ToNtpUnits(ntp_error);
+    NtpTime ntp(static_cast<uint64_t>(remote_clock_.CurrentNtpTime()) +
+                ntp_error_fractions);
+    AdvanceTime(kTestRtt / 2 + networking_delay);
+    RTC_DCHECK(estimator_.UpdateRtcpTimestamp(kTestRtt, ntp, rtcp_timestamp));
+  }
+
+  SimulatedClock local_clock_{kLocalClockInitialTime};
+  SimulatedClock remote_clock_{kRemoteClockInitialTime};
+  RemoteNtpTimeEstimator estimator_{&local_clock_};
+};
+
+TEST_F(RemoteNtpTimeEstimatorTest, FailsWithoutValidNtpTime) {
+  EXPECT_FALSE(
+      estimator_.UpdateRtcpTimestamp(kTestRtt, NtpTime(), /*rtp_timestamp=*/0));
+}
+
+TEST_F(RemoteNtpTimeEstimatorTest, Estimate) {
+  // Remote peer sends first RTCP SR.
+  SendRtcpSr();
+
+  // Remote sends a RTP packet.
+  AdvanceTime(TimeDelta::Millis(15));
+  uint32_t rtp_timestamp = GetRemoteTimestamp();
+  int64_t capture_ntp_time_ms = local_clock_.CurrentNtpInMilliseconds();
+
+  // Local peer needs at least 2 RTCP SR to calculate the capture time.
+  const int64_t kNotEnoughRtcpSr = -1;
+  EXPECT_EQ(kNotEnoughRtcpSr, estimator_.Estimate(rtp_timestamp));
+  EXPECT_EQ(estimator_.EstimateRemoteToLocalClockOffset(), absl::nullopt);
+
+  AdvanceTime(TimeDelta::Millis(800));
+  // Remote sends second RTCP SR.
+  SendRtcpSr();
+
+  // Local peer gets enough RTCP SR to calculate the capture time.
+  EXPECT_EQ(capture_ntp_time_ms, estimator_.Estimate(rtp_timestamp));
+  EXPECT_EQ(estimator_.EstimateRemoteToLocalClockOffset(),
+            kRemoteToLocalClockOffsetNtp);
+}
+
+TEST_F(RemoteNtpTimeEstimatorTest, AveragesErrorsOut) {
+  // Remote peer sends first 10 RTCP SR without errors.
+  for (int i = 0; i < 10; ++i) {
+    AdvanceTime(TimeDelta::Seconds(1));
+    SendRtcpSr();
+  }
+
+  AdvanceTime(TimeDelta::Millis(150));
+  uint32_t rtp_timestamp = GetRemoteTimestamp();
+  int64_t capture_ntp_time_ms = local_clock_.CurrentNtpInMilliseconds();
+  // Local peer gets enough RTCP SR to calculate the capture time.
+  EXPECT_EQ(capture_ntp_time_ms, estimator_.Estimate(rtp_timestamp));
+  EXPECT_EQ(kRemoteToLocalClockOffsetNtp,
+            estimator_.EstimateRemoteToLocalClockOffset());
+
+  // Remote sends corrupted RTCP SRs
+  AdvanceTime(TimeDelta::Seconds(1));
+  SendRtcpSrInaccurately(/*ntp_error=*/TimeDelta::Millis(2),
+                         /*networking_delay=*/TimeDelta::Millis(-1));
+  AdvanceTime(TimeDelta::Seconds(1));
+  SendRtcpSrInaccurately(/*ntp_error=*/TimeDelta::Millis(-2),
+                         /*networking_delay=*/TimeDelta::Millis(1));
+
+  // New RTP packet to estimate timestamp.
+  AdvanceTime(TimeDelta::Millis(150));
+  rtp_timestamp = GetRemoteTimestamp();
+  capture_ntp_time_ms = local_clock_.CurrentNtpInMilliseconds();
+
+  // Errors should be averaged out.
+  EXPECT_EQ(capture_ntp_time_ms, estimator_.Estimate(rtp_timestamp));
+  EXPECT_EQ(kRemoteToLocalClockOffsetNtp,
+            estimator_.EstimateRemoteToLocalClockOffset());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
new file mode 100644
index 0000000000..1d652d0b5b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_nack_stats.h"
+
+#include "modules/include/module_common_types_public.h"
+
+namespace webrtc {
+
+RtcpNackStats::RtcpNackStats()
+    : max_sequence_number_(0), requests_(0), unique_requests_(0) {}
+
+void RtcpNackStats::ReportRequest(uint16_t sequence_number) {
+  if (requests_ == 0 ||
+      IsNewerSequenceNumber(sequence_number, max_sequence_number_)) {
+    max_sequence_number_ = sequence_number;
+    ++unique_requests_;
+  }
+  ++requests_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h
new file mode 100644
index 0000000000..9da4351a59
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_NACK_STATS_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_NACK_STATS_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+class RtcpNackStats {
+ public:
+  RtcpNackStats();
+
+  // Updates stats with requested sequence number.
+  // This function should be called for each NACK request to calculate the
+  // number of unique NACKed RTP packets.
+  void ReportRequest(uint16_t sequence_number);
+
+  // Gets the number of NACKed RTP packets.
+  uint32_t requests() const { return requests_; }
+
+  // Gets the number of unique NACKed RTP packets.
+  uint32_t unique_requests() const { return unique_requests_; }
+
+ private:
+  uint16_t max_sequence_number_;
+  uint32_t requests_;
+  uint32_t unique_requests_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_NACK_STATS_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats_unittest.cc
new file mode 100644
index 0000000000..60858e197e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_nack_stats_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_nack_stats.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(RtcpNackStatsTest, Requests) {
+  RtcpNackStats stats;
+  EXPECT_EQ(0U, stats.unique_requests());
+  EXPECT_EQ(0U, stats.requests());
+  stats.ReportRequest(10);
+  EXPECT_EQ(1U, stats.unique_requests());
+  EXPECT_EQ(1U, stats.requests());
+
+  stats.ReportRequest(10);
+  EXPECT_EQ(1U, stats.unique_requests());
+  stats.ReportRequest(11);
+  EXPECT_EQ(2U, stats.unique_requests());
+
+  stats.ReportRequest(11);
+  EXPECT_EQ(2U, stats.unique_requests());
+  stats.ReportRequest(13);
+  EXPECT_EQ(3U, stats.unique_requests());
+
+  stats.ReportRequest(11);
+  EXPECT_EQ(3U, stats.unique_requests());
+  EXPECT_EQ(6U, stats.requests());
+}
+
+TEST(RtcpNackStatsTest, RequestsWithWrap) {
+  RtcpNackStats stats;
+  stats.ReportRequest(65534);
+  EXPECT_EQ(1U, stats.unique_requests());
+
+  stats.ReportRequest(65534);
+  EXPECT_EQ(1U, stats.unique_requests());
+  stats.ReportRequest(65535);
+  EXPECT_EQ(2U, stats.unique_requests());
+
+  stats.ReportRequest(65535);
+  EXPECT_EQ(2U, stats.unique_requests());
+  stats.ReportRequest(0);
+  EXPECT_EQ(3U, stats.unique_requests());
+
+  stats.ReportRequest(65535);
+  EXPECT_EQ(3U, stats.unique_requests());
+  stats.ReportRequest(0);
+  EXPECT_EQ(3U, stats.unique_requests());
+  stats.ReportRequest(1);
+  EXPECT_EQ(4U, stats.unique_requests());
+  EXPECT_EQ(8U, stats.requests());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.cc
new file mode 100644
index 0000000000..bac03e73d2
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.cc
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr size_t RtcpPacket::kHeaderLength;
+
+rtc::Buffer RtcpPacket::Build() const {
+  rtc::Buffer packet(BlockLength());
+
+  size_t length = 0;
+  bool created = Create(packet.data(), &length, packet.capacity(), nullptr);
+  RTC_DCHECK(created) << "Invalid packet is not supported.";
+  RTC_DCHECK_EQ(length, packet.size())
+      << "BlockLength mispredicted size used by Create";
+
+  return packet;
+}
+
+bool RtcpPacket::Build(size_t max_length, PacketReadyCallback callback) const {
+  RTC_CHECK_LE(max_length, IP_PACKET_SIZE);
+  uint8_t buffer[IP_PACKET_SIZE];
+  size_t index = 0;
+  if (!Create(buffer, &index, max_length, callback))
+    return false;
+  return OnBufferFull(buffer, &index, callback);
+}
+
+bool RtcpPacket::OnBufferFull(uint8_t* packet,
+                              size_t* index,
+                              PacketReadyCallback callback) const {
+  if (*index == 0)
+    return false;
+  RTC_DCHECK(callback) << "Fragmentation not supported.";
+  callback(rtc::ArrayView<const uint8_t>(packet, *index));
+  *index = 0;
+  return true;
+}
+
+size_t RtcpPacket::HeaderLength() const {
+  size_t length_in_bytes = BlockLength();
+  RTC_DCHECK_GT(length_in_bytes, 0);
+  RTC_DCHECK_EQ(length_in_bytes % 4, 0)
+      << "Padding must be handled by each subclass.";
+  // Length in 32-bit words without common header.
+  return (length_in_bytes - kHeaderLength) / 4;
+}
+
+// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
+//
+// RTP header format.
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P| RC/FMT  |      PT       |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+void RtcpPacket::CreateHeader(
+    size_t count_or_format,  // Depends on packet type.
+    uint8_t packet_type,
+    size_t length,
+    uint8_t* buffer,
+    size_t* pos) {
+  CreateHeader(count_or_format, packet_type, length, /*padding=*/false, buffer,
+               pos);
+}
+
+void RtcpPacket::CreateHeader(
+    size_t count_or_format,  // Depends on packet type.
+    uint8_t packet_type,
+    size_t length,
+    bool padding,
+    uint8_t* buffer,
+    size_t* pos) {
+  RTC_DCHECK_LE(length, 0xffffU);
+  RTC_DCHECK_LE(count_or_format, 0x1f);
+  constexpr uint8_t kVersionBits = 2 << 6;
+  uint8_t padding_bit = padding ? 1 << 5 : 0;
+  buffer[*pos + 0] =
+      kVersionBits | padding_bit | static_cast<uint8_t>(count_or_format);
+  buffer[*pos + 1] = packet_type;
+  buffer[*pos + 2] = (length >> 8) & 0xff;
+  buffer[*pos + 3] = length & 0xff;
+  *pos += kHeaderLength;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.h
new file mode 100644
index 0000000000..07deb0f9bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "api/function_view.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace rtcp {
+// Class for building RTCP packets.
+//
+//  Example:
+//  ReportBlock report_block;
+//  report_block.SetMediaSsrc(234);
+//  report_block.SetFractionLost(10);
+//
+//  ReceiverReport rr;
+//  rr.SetSenderSsrc(123);
+//  rr.AddReportBlock(report_block);
+//
+//  Fir fir;
+//  fir.SetSenderSsrc(123);
+//  fir.AddRequestTo(234, 56);
+//
+//  size_t length = 0;                     // Builds an intra frame request
+//  uint8_t packet[kPacketSize];           // with sequence number 56.
+//  fir.Build(packet, &length, kPacketSize);
+//
+//  rtc::Buffer packet = fir.Build();      // Returns a RawPacket holding
+//                                         // the built rtcp packet.
+//
+//  CompoundPacket compound;               // Builds a compound RTCP packet with
+//  compound.Append(&rr);                  // a receiver report, report block
+//  compound.Append(&fir);                 // and fir message.
+//  rtc::Buffer packet = compound.Build();
+
+class RtcpPacket {
+ public:
+  // Callback used to signal that an RTCP packet is ready. Note that this may
+  // not contain all data in this RtcpPacket; if a packet cannot fit in
+  // max_length bytes, it will be fragmented and multiple calls to this
+  // callback will be made.
+  using PacketReadyCallback =
+      rtc::FunctionView<void(rtc::ArrayView<const uint8_t> packet)>;
+
+  virtual ~RtcpPacket() = default;
+
+  void SetSenderSsrc(uint32_t ssrc) { sender_ssrc_ = ssrc; }
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+
+  // Convenience method mostly used for test. Creates packet without
+  // fragmentation using BlockLength() to allocate big enough buffer.
+  rtc::Buffer Build() const;
+
+  // Returns true if call to Create succeeded.
+  bool Build(size_t max_length, PacketReadyCallback callback) const;
+
+  // Size of this packet in bytes (including headers).
+  virtual size_t BlockLength() const = 0;
+
+  // Creates packet in the given buffer at the given position.
+  // Calls PacketReadyCallback::OnPacketReady if remaining buffer is too small
+  // and assume buffer can be reused after OnPacketReady returns.
+  virtual bool Create(uint8_t* packet,
+                      size_t* index,
+                      size_t max_length,
+                      PacketReadyCallback callback) const = 0;
+
+ protected:
+  // Size of the rtcp common header.
+  static constexpr size_t kHeaderLength = 4;
+  RtcpPacket() {}
+
+  static void CreateHeader(size_t count_or_format,
+                           uint8_t packet_type,
+                           size_t block_length,  // Payload size in 32bit words.
+                           uint8_t* buffer,
+                           size_t* pos);
+
+  static void CreateHeader(size_t count_or_format,
+                           uint8_t packet_type,
+                           size_t block_length,  // Payload size in 32bit words.
+                           bool padding,  // True if there are padding bytes.
+                           uint8_t* buffer,
+                           size_t* pos);
+
+  bool OnBufferFull(uint8_t* packet,
+                    size_t* index,
+                    PacketReadyCallback callback) const;
+  // Size of the rtcp packet as written in header.
+  size_t HeaderLength() const;
+
+ private:
+  uint32_t sender_ssrc_ = 0;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
new file mode 100644
index 0000000000..d5734c6dd5
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+
+#include <string.h>
+
+#include <cstdint>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t App::kPacketType;
+constexpr size_t App::kMaxDataSize;
+// Application-Defined packet (APP) (RFC 3550).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P| subtype |   PT=APP=204  |             length            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |                           SSRC/CSRC                           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                          name (ASCII)                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |                   application-dependent data                ...
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+App::App() : sub_type_(0), name_(0) {}
+
+App::~App() = default;
+
+bool App::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  if (packet.payload_size_bytes() < kAppBaseLength) {
+    RTC_LOG(LS_WARNING) << "Packet is too small to be a valid APP packet";
+    return false;
+  }
+  if (packet.payload_size_bytes() % 4 != 0) {
+    RTC_LOG(LS_WARNING)
+        << "Packet payload must be 32 bits aligned to make a valid APP packet";
+    return false;
+  }
+  sub_type_ = packet.fmt();
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(&packet.payload()[0]));
+  name_ = ByteReader<uint32_t>::ReadBigEndian(&packet.payload()[4]);
+  data_.SetData(packet.payload() + kAppBaseLength,
+                packet.payload_size_bytes() - kAppBaseLength);
+  return true;
+}
+
+void App::SetSubType(uint8_t subtype) {
+  RTC_DCHECK_LE(subtype, 0x1f);
+  sub_type_ = subtype;
+}
+
+void App::SetData(const uint8_t* data, size_t data_length) {
+  RTC_DCHECK(data);
+  RTC_DCHECK_EQ(data_length % 4, 0) << "Data must be 32 bits aligned.";
+  RTC_DCHECK_LE(data_length, kMaxDataSize)
+      << "App data size " << data_length << " exceed maximum of "
+      << kMaxDataSize << " bytes.";
+  data_.SetData(data, data_length);
+}
+
+size_t App::BlockLength() const {
+  return kHeaderLength + kAppBaseLength + data_.size();
+}
+
+bool App::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+  CreateHeader(sub_type_, kPacketType, HeaderLength(), packet, index);
+
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], sender_ssrc());
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], name_);
+  if (!data_.empty()) {
+    memcpy(&packet[*index + 8], data_.data(), data_.size());
+  }
+  *index += (8 + data_.size());
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
new file mode 100644
index 0000000000..4518792e5a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class App : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 204;
+  App();
+  App(App&&) = default;
+  ~App() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void SetSubType(uint8_t subtype);
+  void SetName(uint32_t name) { name_ = name; }
+  void SetData(const uint8_t* data, size_t data_length);
+
+  uint8_t sub_type() const { return sub_type_; }
+  uint32_t name() const { return name_; }
+  size_t data_size() const { return data_.size(); }
+  const uint8_t* data() const { return data_.data(); }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+  static inline constexpr uint32_t NameToInt(const char name[5]) {
+    return static_cast<uint32_t>(name[0]) << 24 |
+           static_cast<uint32_t>(name[1]) << 16 |
+           static_cast<uint32_t>(name[2]) << 8 | static_cast<uint32_t>(name[3]);
+  }
+
+ private:
+  static constexpr size_t kAppBaseLength = 8;  // Ssrc and Name.
+  static constexpr size_t kMaxDataSize = 0xffff * 4 - kAppBaseLength;
+
+  uint8_t sub_type_;
+  uint32_t name_;
+  rtc::Buffer data_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
new file mode 100644
index 0000000000..8690e8e5a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAreArray;
+using ::testing::make_tuple;
+using ::webrtc::rtcp::App;
+
+constexpr uint32_t kName = ((uint32_t)'n' << 24) | ((uint32_t)'a' << 16) |
+                           ((uint32_t)'m' << 8) | (uint32_t)'e';
+constexpr uint8_t kSubtype = 0x1e;
+constexpr uint32_t kSenderSsrc = 0x12345678;
+constexpr uint8_t kData[] = {'t', 'e', 's', 't', 'd', 'a', 't', 'a'};
+constexpr uint8_t kVersionBits = 2 << 6;
+constexpr uint8_t kPaddingBit = 1 << 5;
+// clang-format off
+constexpr uint8_t kPacketWithoutData[] = {
+    kVersionBits | kSubtype, App::kPacketType, 0x00, 0x02,
+    0x12, 0x34, 0x56, 0x78,
+    'n',  'a',  'm',  'e'};
+constexpr uint8_t kPacketWithData[] = {
+    kVersionBits | kSubtype, App::kPacketType, 0x00, 0x04,
+    0x12, 0x34, 0x56, 0x78,
+    'n',  'a',  'm',  'e',
+    't',  'e',  's',  't',
+    'd',  'a',  't',  'a'};
+constexpr uint8_t kTooSmallPacket[] = {
+    kVersionBits | kSubtype, App::kPacketType, 0x00, 0x01,
+    0x12, 0x34, 0x56, 0x78};
+constexpr uint8_t kPaddingSize = 1;
+constexpr uint8_t kPacketWithUnalignedPayload[] = {
+    kVersionBits | kPaddingBit | kSubtype, App::kPacketType, 0x00, 0x03,
+    0x12, 0x34, 0x56, 0x78,
+     'n',  'a',  'm',  'e',
+     'd',  'a',  't', kPaddingSize};
+// clang-format on
+}  // namespace
+
+TEST(RtcpPacketAppTest, CreateWithoutData) {
+  App app;
+  app.SetSenderSsrc(kSenderSsrc);
+  app.SetSubType(kSubtype);
+  app.SetName(kName);
+
+  rtc::Buffer raw = app.Build();
+
+  EXPECT_THAT(make_tuple(raw.data(), raw.size()),
+              ElementsAreArray(kPacketWithoutData));
+}
+
+TEST(RtcpPacketAppTest, ParseWithoutData) {
+  App parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacketWithoutData, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kSubtype, parsed.sub_type());
+  EXPECT_EQ(kName, parsed.name());
+  EXPECT_EQ(0u, parsed.data_size());
+}
+
+TEST(RtcpPacketAppTest, CreateWithData) {
+  App app;
+  app.SetSenderSsrc(kSenderSsrc);
+  app.SetSubType(kSubtype);
+  app.SetName(kName);
+  app.SetData(kData, sizeof(kData));
+
+  rtc::Buffer raw = app.Build();
+
+  EXPECT_THAT(make_tuple(raw.data(), raw.size()),
+              ElementsAreArray(kPacketWithData));
+}
+
+TEST(RtcpPacketAppTest, ParseWithData) {
+  App parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacketWithData, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kSubtype, parsed.sub_type());
+  EXPECT_EQ(kName, parsed.name());
+  EXPECT_THAT(make_tuple(parsed.data(), parsed.data_size()),
+              ElementsAreArray(kData));
+}
+
+TEST(RtcpPacketAppTest, ParseFailsOnTooSmallPacket) {
+  App parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kTooSmallPacket, &parsed));
+}
+
+TEST(RtcpPacketAppTest, ParseFailsOnUnalignedPayload) {
+  App parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kPacketWithUnalignedPayload, &parsed));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
new file mode 100644
index 0000000000..a6471772b1
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+
+#include <string.h>
+
+#include <cstdint>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Bye::kPacketType;
+// Bye packet (BYE) (RFC 3550).
+//
+//        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       |V=2|P|    SC   |   PT=BYE=203  |             length            |
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       |                           SSRC/CSRC                           |
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//       :                              ...                              :
+//       +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// (opt) |     length    |               reason for leaving            ...
+//       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+Bye::Bye() = default;
+
+Bye::~Bye() = default;
+
+bool Bye::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+
+  const uint8_t src_count = packet.count();
+  // Validate packet.
+  if (packet.payload_size_bytes() < 4u * src_count) {
+    RTC_LOG(LS_WARNING)
+        << "Packet is too small to contain CSRCs it promise to have.";
+    return false;
+  }
+  const uint8_t* const payload = packet.payload();
+  bool has_reason = packet.payload_size_bytes() > 4u * src_count;
+  uint8_t reason_length = 0;
+  if (has_reason) {
+    reason_length = payload[4u * src_count];
+    if (packet.payload_size_bytes() - 4u * src_count < 1u + reason_length) {
+      RTC_LOG(LS_WARNING) << "Invalid reason length: " << reason_length;
+      return false;
+    }
+  }
+  // Once sure packet is valid, copy values.
+  if (src_count == 0) {  // A count value of zero is valid, but useless.
+    SetSenderSsrc(0);
+    csrcs_.clear();
+  } else {
+    SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(payload));
+    csrcs_.resize(src_count - 1);
+    for (size_t i = 1; i < src_count; ++i)
+      csrcs_[i - 1] = ByteReader<uint32_t>::ReadBigEndian(&payload[4 * i]);
+  }
+
+  if (has_reason) {
+    reason_.assign(reinterpret_cast<const char*>(&payload[4u * src_count + 1]),
+                   reason_length);
+  } else {
+    reason_.clear();
+  }
+
+  return true;
+}
+
+bool Bye::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+
+  CreateHeader(1 + csrcs_.size(), kPacketType, HeaderLength(), packet, index);
+  // Store srcs of the leaving clients.
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], sender_ssrc());
+  *index += sizeof(uint32_t);
+  for (uint32_t csrc : csrcs_) {
+    ByteWriter<uint32_t>::WriteBigEndian(&packet[*index], csrc);
+    *index += sizeof(uint32_t);
+  }
+  // Store the reason to leave.
+  if (!reason_.empty()) {
+    uint8_t reason_length = static_cast<uint8_t>(reason_.size());
+    packet[(*index)++] = reason_length;
+    memcpy(&packet[*index], reason_.data(), reason_length);
+    *index += reason_length;
+    // Add padding bytes if needed.
+    size_t bytes_to_pad = index_end - *index;
+    RTC_DCHECK_LE(bytes_to_pad, 3);
+    if (bytes_to_pad > 0) {
+      memset(&packet[*index], 0, bytes_to_pad);
+      *index += bytes_to_pad;
+    }
+  }
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+bool Bye::SetCsrcs(std::vector<uint32_t> csrcs) {
+  if (csrcs.size() > kMaxNumberOfCsrcs) {
+    RTC_LOG(LS_WARNING) << "Too many CSRCs for Bye packet.";
+    return false;
+  }
+  csrcs_ = std::move(csrcs);
+  return true;
+}
+
+void Bye::SetReason(absl::string_view reason) {
+  RTC_DCHECK_LE(reason.size(), 0xffu);
+  reason_ = std::string(reason);
+}
+
+size_t Bye::BlockLength() const {
+  size_t src_count = (1 + csrcs_.size());
+  size_t reason_size_in_32bits = reason_.empty() ? 0 : (reason_.size() / 4 + 1);
+  return kHeaderLength + 4 * (src_count + reason_size_in_32bits);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
new file mode 100644
index 0000000000..d31205793a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class Bye : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 203;
+
+  Bye();
+  ~Bye() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  bool SetCsrcs(std::vector<uint32_t> csrcs);
+  void SetReason(absl::string_view reason);
+
+  const std::vector<uint32_t>& csrcs() const { return csrcs_; }
+  const std::string& reason() const { return reason_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static const int kMaxNumberOfCsrcs = 0x1f - 1;  // First item is sender SSRC.
+
+  std::vector<uint32_t> csrcs_;
+  std::string reason_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_BYE_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc
new file mode 100644
index 0000000000..448c2d4194
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAre;
+using webrtc::rtcp::Bye;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kCsrc1 = 0x22232425;
+const uint32_t kCsrc2 = 0x33343536;
+}  // namespace
+
+TEST(RtcpPacketByeTest, CreateAndParseWithoutReason) {
+  Bye bye;
+  bye.SetSenderSsrc(kSenderSsrc);
+
+  rtc::Buffer raw = bye.Build();
+  Bye parsed_bye;
+  EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed_bye));
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_TRUE(parsed_bye.csrcs().empty());
+  EXPECT_TRUE(parsed_bye.reason().empty());
+}
+
+TEST(RtcpPacketByeTest, CreateAndParseWithCsrcs) {
+  Bye bye;
+  bye.SetSenderSsrc(kSenderSsrc);
+  EXPECT_TRUE(bye.SetCsrcs({kCsrc1, kCsrc2}));
+  EXPECT_TRUE(bye.reason().empty());
+
+  rtc::Buffer raw = bye.Build();
+  Bye parsed_bye;
+  EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed_bye));
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_THAT(parsed_bye.csrcs(), ElementsAre(kCsrc1, kCsrc2));
+  EXPECT_TRUE(parsed_bye.reason().empty());
+}
+
+TEST(RtcpPacketByeTest, CreateAndParseWithCsrcsAndAReason) {
+  Bye bye;
+  const std::string kReason = "Some Reason";
+
+  bye.SetSenderSsrc(kSenderSsrc);
+  EXPECT_TRUE(bye.SetCsrcs({kCsrc1, kCsrc2}));
+  bye.SetReason(kReason);
+
+  rtc::Buffer raw = bye.Build();
+  Bye parsed_bye;
+  EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed_bye));
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_THAT(parsed_bye.csrcs(), ElementsAre(kCsrc1, kCsrc2));
+  EXPECT_EQ(kReason, parsed_bye.reason());
+}
+
+TEST(RtcpPacketByeTest, CreateWithTooManyCsrcs) {
+  Bye bye;
+  bye.SetSenderSsrc(kSenderSsrc);
+  const int kMaxCsrcs = (1 << 5) - 2;  // 5 bit len, first item is sender SSRC.
+  EXPECT_TRUE(bye.SetCsrcs(std::vector<uint32_t>(kMaxCsrcs, kCsrc1)));
+  EXPECT_FALSE(bye.SetCsrcs(std::vector<uint32_t>(kMaxCsrcs + 1, kCsrc1)));
+}
+
+TEST(RtcpPacketByeTest, CreateAndParseWithAReason) {
+  Bye bye;
+  const std::string kReason = "Some Random Reason";
+
+  bye.SetSenderSsrc(kSenderSsrc);
+  bye.SetReason(kReason);
+
+  rtc::Buffer raw = bye.Build();
+  Bye parsed_bye;
+  EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed_bye));
+
+  EXPECT_EQ(kSenderSsrc, parsed_bye.sender_ssrc());
+  EXPECT_TRUE(parsed_bye.csrcs().empty());
+  EXPECT_EQ(kReason, parsed_bye.reason());
+}
+
+TEST(RtcpPacketByeTest, CreateAndParseWithReasons) {
+  // Test that packet creation/parsing behave with reasons of different length
+  // both when it require padding and when it does not.
+  for (size_t reminder = 0; reminder < 4; ++reminder) {
+    const std::string kReason(4 + reminder, 'a' + reminder);
+    Bye bye;
+    bye.SetSenderSsrc(kSenderSsrc);
+    bye.SetReason(kReason);
+
+    rtc::Buffer raw = bye.Build();
+    Bye parsed_bye;
+    EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed_bye));
+
+    EXPECT_EQ(kReason, parsed_bye.reason());
+  }
+}
+
+TEST(RtcpPacketByeTest, ParseEmptyPacket) {
+  uint8_t kEmptyPacket[] = {0x80, Bye::kPacketType, 0, 0};
+  Bye parsed_bye;
+  EXPECT_TRUE(test::ParseSinglePacket(kEmptyPacket, &parsed_bye));
+  EXPECT_EQ(0u, parsed_bye.sender_ssrc());
+  EXPECT_TRUE(parsed_bye.csrcs().empty());
+  EXPECT_TRUE(parsed_bye.reason().empty());
+}
+
+TEST(RtcpPacketByeTest, ParseFailOnInvalidSrcCount) {
+  Bye bye;
+  bye.SetSenderSsrc(kSenderSsrc);
+
+  rtc::Buffer raw = bye.Build();
+  raw[0]++;  // Damage the packet: increase ssrc count by one.
+
+  Bye parsed_bye;
+  EXPECT_FALSE(test::ParseSinglePacket(raw, &parsed_bye));
+}
+
+TEST(RtcpPacketByeTest, ParseFailOnInvalidReasonLength) {
+  Bye bye;
+  bye.SetSenderSsrc(kSenderSsrc);
+  bye.SetReason("18 characters long");
+
+  rtc::Buffer raw = bye.Build();
+  // Damage the packet: decrease payload size by 4 bytes
+  raw[3]--;
+  raw.SetSize(raw.size() - 4);
+
+  Bye parsed_bye;
+  EXPECT_FALSE(test::ParseSinglePacket(raw, &parsed_bye));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.cc
new file mode 100644
index 0000000000..5b54982220
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr size_t CommonHeader::kHeaderSizeBytes;
+//    0                   1           1       2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |V=2|P|   C/F   |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 1                 |  Packet Type  |
+//   ----------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 2                                 |             length            |
+//   --------------------------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Common header for all RTCP packets, 4 octets.
+bool CommonHeader::Parse(const uint8_t* buffer, size_t size_bytes) {
+  const uint8_t kVersion = 2;
+
+  if (size_bytes < kHeaderSizeBytes) {
+    RTC_LOG(LS_WARNING)
+        << "Too little data (" << size_bytes << " byte"
+        << (size_bytes != 1 ? "s" : "")
+        << ") remaining in buffer to parse RTCP header (4 bytes).";
+    return false;
+  }
+
+  uint8_t version = buffer[0] >> 6;
+  if (version != kVersion) {
+    RTC_LOG(LS_WARNING) << "Invalid RTCP header: Version must be "
+                        << static_cast<int>(kVersion) << " but was "
+                        << static_cast<int>(version);
+    return false;
+  }
+
+  bool has_padding = (buffer[0] & 0x20) != 0;
+  count_or_format_ = buffer[0] & 0x1F;
+  packet_type_ = buffer[1];
+  payload_size_ = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) * 4;
+  payload_ = buffer + kHeaderSizeBytes;
+  padding_size_ = 0;
+
+  if (size_bytes < kHeaderSizeBytes + payload_size_) {
+    RTC_LOG(LS_WARNING) << "Buffer too small (" << size_bytes
+                        << " bytes) to fit an RtcpPacket with a header and "
+                        << payload_size_ << " bytes.";
+    return false;
+  }
+
+  if (has_padding) {
+    if (payload_size_ == 0) {
+      RTC_LOG(LS_WARNING)
+          << "Invalid RTCP header: Padding bit set but 0 payload "
+             "size specified.";
+      return false;
+    }
+
+    padding_size_ = payload_[payload_size_ - 1];
+    if (padding_size_ == 0) {
+      RTC_LOG(LS_WARNING)
+          << "Invalid RTCP header: Padding bit set but 0 padding "
+             "size specified.";
+      return false;
+    }
+    if (padding_size_ > payload_size_) {
+      RTC_LOG(LS_WARNING) << "Invalid RTCP header: Too many padding bytes ("
+                          << padding_size_ << ") for a packet payload size of "
+                          << payload_size_ << " bytes.";
+      return false;
+    }
+    payload_size_ -= padding_size_;
+  }
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h
new file mode 100644
index 0000000000..5416406091
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader {
+ public:
+  static constexpr size_t kHeaderSizeBytes = 4;
+
+  CommonHeader() {}
+  CommonHeader(const CommonHeader&) = default;
+  CommonHeader& operator=(const CommonHeader&) = default;
+
+  bool Parse(const uint8_t* buffer, size_t size_bytes);
+
+  uint8_t type() const { return packet_type_; }
+  // Depending on packet type same header field can be used either as count or
+  // as feedback message type (fmt). Caller expected to know how it is used.
+  uint8_t fmt() const { return count_or_format_; }
+  uint8_t count() const { return count_or_format_; }
+  size_t payload_size_bytes() const { return payload_size_; }
+  const uint8_t* payload() const { return payload_; }
+  size_t packet_size() const {
+    return kHeaderSizeBytes + payload_size_ + padding_size_;
+  }
+  // Returns pointer to the next RTCP packet in compound packet.
+  const uint8_t* NextPacket() const {
+    return payload_ + payload_size_ + padding_size_;
+  }
+
+ private:
+  uint8_t packet_type_ = 0;
+  uint8_t count_or_format_ = 0;
+  uint8_t padding_size_ = 0;
+  uint32_t payload_size_ = 0;
+  const uint8_t* payload_ = nullptr;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header_unittest.cc
new file mode 100644
index 0000000000..e8b4c52c68
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/common_header_unittest.cc
@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+
+#include "test/gtest.h"
+
+using webrtc::rtcp::CommonHeader;
+
+namespace webrtc {
+
+TEST(RtcpCommonHeaderTest, TooSmallBuffer) {
+  uint8_t buffer[] = {0x80, 0x00, 0x00, 0x00};
+  CommonHeader header;
+  // Buffer needs to be able to hold the header.
+  EXPECT_FALSE(header.Parse(buffer, 0));
+  EXPECT_FALSE(header.Parse(buffer, 1));
+  EXPECT_FALSE(header.Parse(buffer, 2));
+  EXPECT_FALSE(header.Parse(buffer, 3));
+  EXPECT_TRUE(header.Parse(buffer, 4));
+}
+
+TEST(RtcpCommonHeaderTest, Version) {
+  uint8_t buffer[] = {0x00, 0x00, 0x00, 0x00};
+  CommonHeader header;
+  // Version 2 is the only allowed.
+  buffer[0] = 0 << 6;
+  EXPECT_FALSE(header.Parse(buffer, sizeof(buffer)));
+  buffer[0] = 1 << 6;
+  EXPECT_FALSE(header.Parse(buffer, sizeof(buffer)));
+  buffer[0] = 2 << 6;
+  EXPECT_TRUE(header.Parse(buffer, sizeof(buffer)));
+  buffer[0] = 3 << 6;
+  EXPECT_FALSE(header.Parse(buffer, sizeof(buffer)));
+}
+
+TEST(RtcpCommonHeaderTest, PacketSize) {
+  uint8_t buffer[] = {0x80, 0x00, 0x00, 0x02, 0x00, 0x00,
+                      0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+  CommonHeader header;
+  EXPECT_FALSE(header.Parse(buffer, sizeof(buffer) - 1));
+  EXPECT_TRUE(header.Parse(buffer, sizeof(buffer)));
+  EXPECT_EQ(8u, header.payload_size_bytes());
+  EXPECT_EQ(buffer + sizeof(buffer), header.NextPacket());
+  EXPECT_EQ(sizeof(buffer), header.packet_size());
+}
+
+TEST(RtcpCommonHeaderTest, PaddingAndPayloadSize) {
+  // Set v = 2, p = 1, but leave fmt, pt as 0.
+  uint8_t buffer[] = {0xa0, 0x00, 0x00, 0x00, 0x00, 0x00,
+                      0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+  CommonHeader header;
+  // Padding bit set, but no byte for padding (can't specify padding length).
+  EXPECT_FALSE(header.Parse(buffer, 4));
+
+  buffer[3] = 2;  //  Set payload size to 2x32bit.
+  const size_t kPayloadSizeBytes = buffer[3] * 4;
+  const size_t kPaddingAddress =
+      CommonHeader::kHeaderSizeBytes + kPayloadSizeBytes - 1;
+
+  // Padding one byte larger than possible.
+  buffer[kPaddingAddress] = kPayloadSizeBytes + 1;
+  EXPECT_FALSE(header.Parse(buffer, sizeof(buffer)));
+
+  // Invalid zero padding size.
+  buffer[kPaddingAddress] = 0;
+  EXPECT_FALSE(header.Parse(buffer, sizeof(buffer)));
+
+  // Pure padding packet.
+  buffer[kPaddingAddress] = kPayloadSizeBytes;
+  EXPECT_TRUE(header.Parse(buffer, sizeof(buffer)));
+  EXPECT_EQ(0u, header.payload_size_bytes());
+  EXPECT_EQ(buffer + sizeof(buffer), header.NextPacket());
+  EXPECT_EQ(header.payload(), buffer + CommonHeader::kHeaderSizeBytes);
+  EXPECT_EQ(header.packet_size(), sizeof(buffer));
+
+  // Single byte of actual data.
+  buffer[kPaddingAddress] = kPayloadSizeBytes - 1;
+  EXPECT_TRUE(header.Parse(buffer, sizeof(buffer)));
+  EXPECT_EQ(1u, header.payload_size_bytes());
+  EXPECT_EQ(buffer + sizeof(buffer), header.NextPacket());
+  EXPECT_EQ(header.packet_size(), sizeof(buffer));
+}
+
+TEST(RtcpCommonHeaderTest, FormatAndPayloadType) {
+  uint8_t buffer[] = {0x9e, 0xab, 0x00, 0x00};
+  CommonHeader header;
+  EXPECT_TRUE(header.Parse(buffer, sizeof(buffer)));
+
+  EXPECT_EQ(header.count(), 0x1e);
+  EXPECT_EQ(header.fmt(), 0x1e);
+  EXPECT_EQ(header.type(), 0xab);
+  EXPECT_EQ(header.payload_size_bytes(), 0u);
+  EXPECT_EQ(header.payload(), buffer + CommonHeader::kHeaderSizeBytes);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc
new file mode 100644
index 0000000000..54f3555fc6
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rtcp {
+
+CompoundPacket::CompoundPacket() = default;
+
+CompoundPacket::~CompoundPacket() = default;
+
+void CompoundPacket::Append(std::unique_ptr<RtcpPacket> packet) {
+  RTC_CHECK(packet);
+  appended_packets_.push_back(std::move(packet));
+}
+
+bool CompoundPacket::Create(uint8_t* packet,
+                            size_t* index,
+                            size_t max_length,
+                            PacketReadyCallback callback) const {
+  for (const auto& appended : appended_packets_) {
+    if (!appended->Create(packet, index, max_length, callback))
+      return false;
+  }
+  return true;
+}
+
+size_t CompoundPacket::BlockLength() const {
+  size_t block_length = 0;
+  for (const auto& appended : appended_packets_) {
+    block_length += appended->BlockLength();
+  }
+  return block_length;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
new file mode 100644
index 0000000000..d98dbd088d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
+
+#include <memory>
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class CompoundPacket : public RtcpPacket {
+ public:
+  CompoundPacket();
+  ~CompoundPacket() override;
+
+  CompoundPacket(const CompoundPacket&) = delete;
+  CompoundPacket& operator=(const CompoundPacket&) = delete;
+
+  void Append(std::unique_ptr<RtcpPacket> packet);
+
+  // Size of this packet in bytes (i.e. total size of nested packets).
+  size_t BlockLength() const override;
+  // Returns true if all calls to Create succeeded.
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ protected:
+  std::vector<std::unique_ptr<RtcpPacket>> appended_packets_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMPOUND_PACKET_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc
new file mode 100644
index 0000000000..9348aee7e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet_unittest.cc
@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+using ::testing::MockFunction;
+using webrtc::rtcp::Bye;
+using webrtc::rtcp::CompoundPacket;
+using webrtc::rtcp::Fir;
+using webrtc::rtcp::ReceiverReport;
+using webrtc::rtcp::ReportBlock;
+using webrtc::rtcp::SenderReport;
+using webrtc::test::RtcpPacketParser;
+
+namespace webrtc {
+
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kSeqNo = 13;
+
+TEST(RtcpCompoundPacketTest, AppendPacket) {
+  CompoundPacket compound;
+  auto fir = std::make_unique<Fir>();
+  fir->AddRequestTo(kRemoteSsrc, kSeqNo);
+  ReportBlock rb;
+  auto rr = std::make_unique<ReceiverReport>();
+  rr->SetSenderSsrc(kSenderSsrc);
+  EXPECT_TRUE(rr->AddReportBlock(rb));
+  compound.Append(std::move(rr));
+  compound.Append(std::move(fir));
+
+  rtc::Buffer packet = compound.Build();
+  RtcpPacketParser parser;
+  parser.Parse(packet.data(), packet.size());
+  EXPECT_EQ(1, parser.receiver_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser.receiver_report()->sender_ssrc());
+  EXPECT_EQ(1u, parser.receiver_report()->report_blocks().size());
+  EXPECT_EQ(1, parser.fir()->num_packets());
+}
+
+TEST(RtcpCompoundPacketTest, AppendPacketWithOwnAppendedPacket) {
+  CompoundPacket root;
+  auto leaf = std::make_unique<CompoundPacket>();
+
+  auto fir = std::make_unique<Fir>();
+  fir->AddRequestTo(kRemoteSsrc, kSeqNo);
+  auto bye = std::make_unique<Bye>();
+  ReportBlock rb;
+
+  auto rr = std::make_unique<ReceiverReport>();
+  EXPECT_TRUE(rr->AddReportBlock(rb));
+  leaf->Append(std::move(rr));
+  leaf->Append(std::move(fir));
+
+  auto sr = std::make_unique<SenderReport>();
+  root.Append(std::move(sr));
+  root.Append(std::move(bye));
+  root.Append(std::move(leaf));
+
+  rtc::Buffer packet = root.Build();
+  RtcpPacketParser parser;
+  parser.Parse(packet.data(), packet.size());
+  EXPECT_EQ(1, parser.sender_report()->num_packets());
+  EXPECT_EQ(1, parser.receiver_report()->num_packets());
+  EXPECT_EQ(1u, parser.receiver_report()->report_blocks().size());
+  EXPECT_EQ(1, parser.bye()->num_packets());
+  EXPECT_EQ(1, parser.fir()->num_packets());
+}
+
+TEST(RtcpCompoundPacketTest, BuildWithInputBuffer) {
+  CompoundPacket compound;
+  auto fir = std::make_unique<Fir>();
+  fir->AddRequestTo(kRemoteSsrc, kSeqNo);
+  ReportBlock rb;
+  auto rr = std::make_unique<ReceiverReport>();
+  rr->SetSenderSsrc(kSenderSsrc);
+  EXPECT_TRUE(rr->AddReportBlock(rb));
+  compound.Append(std::move(rr));
+  compound.Append(std::move(fir));
+
+  const size_t kRrLength = 8;
+  const size_t kReportBlockLength = 24;
+  const size_t kFirLength = 20;
+
+  const size_t kBufferSize = kRrLength + kReportBlockLength + kFirLength;
+  MockFunction<void(rtc::ArrayView<const uint8_t>)> callback;
+  EXPECT_CALL(callback, Call(_))
+      .WillOnce(Invoke([&](rtc::ArrayView<const uint8_t> packet) {
+        RtcpPacketParser parser;
+        parser.Parse(packet.data(), packet.size());
+        EXPECT_EQ(1, parser.receiver_report()->num_packets());
+        EXPECT_EQ(1u, parser.receiver_report()->report_blocks().size());
+        EXPECT_EQ(1, parser.fir()->num_packets());
+      }));
+
+  EXPECT_TRUE(compound.Build(kBufferSize, callback.AsStdFunction()));
+}
+
+TEST(RtcpCompoundPacketTest, BuildWithTooSmallBuffer_FragmentedSend) {
+  CompoundPacket compound;
+  auto fir = std::make_unique<Fir>();
+  fir->AddRequestTo(kRemoteSsrc, kSeqNo);
+  ReportBlock rb;
+  auto rr = std::make_unique<ReceiverReport>();
+  rr->SetSenderSsrc(kSenderSsrc);
+  EXPECT_TRUE(rr->AddReportBlock(rb));
+  compound.Append(std::move(rr));
+  compound.Append(std::move(fir));
+
+  const size_t kRrLength = 8;
+  const size_t kReportBlockLength = 24;
+
+  const size_t kBufferSize = kRrLength + kReportBlockLength;
+  MockFunction<void(rtc::ArrayView<const uint8_t>)> callback;
+  EXPECT_CALL(callback, Call(_))
+      .WillOnce(Invoke([&](rtc::ArrayView<const uint8_t> packet) {
+        RtcpPacketParser parser;
+        parser.Parse(packet.data(), packet.size());
+        EXPECT_EQ(1, parser.receiver_report()->num_packets());
+        EXPECT_EQ(1U, parser.receiver_report()->report_blocks().size());
+        EXPECT_EQ(0, parser.fir()->num_packets());
+      }))
+      .WillOnce(Invoke([&](rtc::ArrayView<const uint8_t> packet) {
+        RtcpPacketParser parser;
+        parser.Parse(packet.data(), packet.size());
+        EXPECT_EQ(0, parser.receiver_report()->num_packets());
+        EXPECT_EQ(0U, parser.receiver_report()->report_blocks().size());
+        EXPECT_EQ(1, parser.fir()->num_packets());
+      }));
+
+  EXPECT_TRUE(compound.Build(kBufferSize, callback.AsStdFunction()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc
new file mode 100644
index 0000000000..6863def2fe
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.cc
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace rtcp {
+// DLRR Report Block (RFC 3611).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     BT=5      |   reserved    |         block length          |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                 SSRC_1 (SSRC of first receiver)               | sub-
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
+//  |                         last RR (LRR)                         |   1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                   delay since last RR (DLRR)                  |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                 SSRC_2 (SSRC of second receiver)              | sub-
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
+//  :                               ...                             :   2
+
+Dlrr::Dlrr() = default;
+
+Dlrr::Dlrr(const Dlrr& other) = default;
+
+Dlrr::~Dlrr() = default;
+
+bool Dlrr::Parse(const uint8_t* buffer, uint16_t block_length_32bits) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // kReserved = buffer[1];
+  RTC_DCHECK_EQ(block_length_32bits,
+                ByteReader<uint16_t>::ReadBigEndian(&buffer[2]));
+  if (block_length_32bits % 3 != 0) {
+    RTC_LOG(LS_WARNING) << "Invalid size for dlrr block.";
+    return false;
+  }
+
+  size_t blocks_count = block_length_32bits / 3;
+  const uint8_t* read_at = buffer + kBlockHeaderLength;
+  sub_blocks_.resize(blocks_count);
+  for (ReceiveTimeInfo& sub_block : sub_blocks_) {
+    sub_block.ssrc = ByteReader<uint32_t>::ReadBigEndian(&read_at[0]);
+    sub_block.last_rr = ByteReader<uint32_t>::ReadBigEndian(&read_at[4]);
+    sub_block.delay_since_last_rr =
+        ByteReader<uint32_t>::ReadBigEndian(&read_at[8]);
+    read_at += kSubBlockLength;
+  }
+  return true;
+}
+
+size_t Dlrr::BlockLength() const {
+  if (sub_blocks_.empty())
+    return 0;
+  return kBlockHeaderLength + kSubBlockLength * sub_blocks_.size();
+}
+
+void Dlrr::Create(uint8_t* buffer) const {
+  if (sub_blocks_.empty())  // No subblocks, no need to write header either.
+    return;
+  // Create block header.
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(
+      &buffer[2], rtc::dchecked_cast<uint16_t>(3 * sub_blocks_.size()));
+  // Create sub blocks.
+  uint8_t* write_at = buffer + kBlockHeaderLength;
+  for (const ReceiveTimeInfo& sub_block : sub_blocks_) {
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[0], sub_block.ssrc);
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[4], sub_block.last_rr);
+    ByteWriter<uint32_t>::WriteBigEndian(&write_at[8],
+                                         sub_block.delay_since_last_rr);
+    write_at += kSubBlockLength;
+  }
+  RTC_DCHECK_EQ(buffer + BlockLength(), write_at);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h
new file mode 100644
index 0000000000..ad91dfdcc6
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+namespace webrtc {
+namespace rtcp {
+struct ReceiveTimeInfo {
+  // RFC 3611 4.5
+  ReceiveTimeInfo() : ssrc(0), last_rr(0), delay_since_last_rr(0) {}
+  ReceiveTimeInfo(uint32_t ssrc, uint32_t last_rr, uint32_t delay)
+      : ssrc(ssrc), last_rr(last_rr), delay_since_last_rr(delay) {}
+
+  uint32_t ssrc;
+  uint32_t last_rr;
+  uint32_t delay_since_last_rr;
+};
+
+inline bool operator==(const ReceiveTimeInfo& lhs, const ReceiveTimeInfo& rhs) {
+  return lhs.ssrc == rhs.ssrc && lhs.last_rr == rhs.last_rr &&
+         lhs.delay_since_last_rr == rhs.delay_since_last_rr;
+}
+
+inline bool operator!=(const ReceiveTimeInfo& lhs, const ReceiveTimeInfo& rhs) {
+  return !(lhs == rhs);
+}
+
+// DLRR Report Block: Delay since the Last Receiver Report (RFC 3611).
+class Dlrr {
+ public:
+  static const uint8_t kBlockType = 5;
+
+  Dlrr();
+  Dlrr(const Dlrr& other);
+  ~Dlrr();
+
+  Dlrr& operator=(const Dlrr& other) = default;
+
+  // Dlrr without items treated same as no dlrr block.
+  explicit operator bool() const { return !sub_blocks_.empty(); }
+
+  // Second parameter is value read from block header,
+  // i.e. size of block in 32bits excluding block header itself.
+  bool Parse(const uint8_t* buffer, uint16_t block_length_32bits);
+
+  size_t BlockLength() const;
+  // Fills buffer with the Dlrr.
+  // Consumes BlockLength() bytes.
+  void Create(uint8_t* buffer) const;
+
+  void ClearItems() { sub_blocks_.clear(); }
+  void AddDlrrItem(const ReceiveTimeInfo& time_info) {
+    sub_blocks_.push_back(time_info);
+  }
+
+  const std::vector<ReceiveTimeInfo>& sub_blocks() const { return sub_blocks_; }
+
+ private:
+  static const size_t kBlockHeaderLength = 4;
+  static const size_t kSubBlockLength = 12;
+
+  std::vector<ReceiveTimeInfo> sub_blocks_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc
new file mode 100644
index 0000000000..408d0011b8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr_unittest.cc
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "test/gtest.h"
+
+using webrtc::rtcp::Dlrr;
+using webrtc::rtcp::ReceiveTimeInfo;
+
+namespace webrtc {
+namespace {
+const uint32_t kSsrc = 0x12345678;
+const uint32_t kLastRR = 0x23344556;
+const uint32_t kDelay = 0x33343536;
+const uint8_t kBlock[] = {0x05, 0x00, 0x00, 0x03, 0x12, 0x34, 0x56, 0x78,
+                          0x23, 0x34, 0x45, 0x56, 0x33, 0x34, 0x35, 0x36};
+const size_t kBlockSizeBytes = sizeof(kBlock);
+}  // namespace
+
+TEST(RtcpPacketDlrrTest, Empty) {
+  Dlrr dlrr;
+
+  EXPECT_EQ(0u, dlrr.BlockLength());
+}
+
+TEST(RtcpPacketDlrrTest, Create) {
+  Dlrr dlrr;
+  dlrr.AddDlrrItem(ReceiveTimeInfo(kSsrc, kLastRR, kDelay));
+
+  ASSERT_EQ(kBlockSizeBytes, dlrr.BlockLength());
+  uint8_t buffer[kBlockSizeBytes];
+
+  dlrr.Create(buffer);
+  EXPECT_EQ(0, memcmp(buffer, kBlock, kBlockSizeBytes));
+}
+
+TEST(RtcpPacketDlrrTest, Parse) {
+  Dlrr dlrr;
+  uint16_t block_length = ByteReader<uint16_t>::ReadBigEndian(&kBlock[2]);
+  EXPECT_TRUE(dlrr.Parse(kBlock, block_length));
+
+  EXPECT_EQ(1u, dlrr.sub_blocks().size());
+  const ReceiveTimeInfo& block = dlrr.sub_blocks().front();
+  EXPECT_EQ(kSsrc, block.ssrc);
+  EXPECT_EQ(kLastRR, block.last_rr);
+  EXPECT_EQ(kDelay, block.delay_since_last_rr);
+}
+
+TEST(RtcpPacketDlrrTest, ParseFailsOnBadSize) {
+  const size_t kBigBufferSize = 0x100;  // More than enough.
+  uint8_t buffer[kBigBufferSize];
+  buffer[0] = Dlrr::kBlockType;
+  buffer[1] = 0;  // Reserved.
+  buffer[2] = 0;  // Most significant size byte.
+  for (uint8_t size = 3; size < 6; ++size) {
+    buffer[3] = size;
+    Dlrr dlrr;
+    // Parse should be successful only when size is multiple of 3.
+    EXPECT_EQ(size % 3 == 0, dlrr.Parse(buffer, static_cast<uint16_t>(size)));
+  }
+}
+
+TEST(RtcpPacketDlrrTest, CreateAndParseManySubBlocks) {
+  const size_t kBufferSize = 0x1000;  // More than enough.
+  const size_t kManyDlrrItems = 50;
+  uint8_t buffer[kBufferSize];
+
+  // Create.
+  Dlrr dlrr;
+  for (size_t i = 1; i <= kManyDlrrItems; ++i)
+    dlrr.AddDlrrItem(ReceiveTimeInfo(kSsrc + i, kLastRR + i, kDelay + i));
+  size_t used_buffer_size = dlrr.BlockLength();
+  ASSERT_LE(used_buffer_size, kBufferSize);
+  dlrr.Create(buffer);
+
+  // Parse.
+  Dlrr parsed;
+  uint16_t block_length = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]);
+  EXPECT_EQ(used_buffer_size, (block_length + 1) * 4u);
+  EXPECT_TRUE(parsed.Parse(buffer, block_length));
+  EXPECT_EQ(kManyDlrrItems, parsed.sub_blocks().size());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
new file mode 100644
index 0000000000..ce57bd5a88
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
@@ -0,0 +1,195 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t ExtendedReports::kPacketType;
+constexpr size_t ExtendedReports::kMaxNumberOfDlrrItems;
+// From RFC 3611: RTP Control Protocol Extended Reports (RTCP XR).
+//
+// Format for XR packets:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|reserved |   PT=XR=207   |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                              SSRC                             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :                         report blocks                         :
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Extended report block:
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  Block Type   |   reserved    |         block length          |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :             type-specific block contents                      :
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ExtendedReports::ExtendedReports() = default;
+ExtendedReports::ExtendedReports(const ExtendedReports& xr) = default;
+ExtendedReports::~ExtendedReports() = default;
+
+bool ExtendedReports::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+
+  if (packet.payload_size_bytes() < kXrBaseLength) {
+    RTC_LOG(LS_WARNING)
+        << "Packet is too small to be an ExtendedReports packet.";
+    return false;
+  }
+
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(packet.payload()));
+  rrtr_block_.reset();
+  dlrr_block_.ClearItems();
+  target_bitrate_ = absl::nullopt;
+
+  const uint8_t* current_block = packet.payload() + kXrBaseLength;
+  const uint8_t* const packet_end =
+      packet.payload() + packet.payload_size_bytes();
+  constexpr size_t kBlockHeaderSizeBytes = 4;
+  while (current_block + kBlockHeaderSizeBytes <= packet_end) {
+    uint8_t block_type = ByteReader<uint8_t>::ReadBigEndian(current_block);
+    uint16_t block_length =
+        ByteReader<uint16_t>::ReadBigEndian(current_block + 2);
+    const uint8_t* next_block =
+        current_block + kBlockHeaderSizeBytes + block_length * 4;
+    if (next_block > packet_end) {
+      RTC_LOG(LS_WARNING)
+          << "Report block in extended report packet is too big.";
+      return false;
+    }
+    switch (block_type) {
+      case Rrtr::kBlockType:
+        ParseRrtrBlock(current_block, block_length);
+        break;
+      case Dlrr::kBlockType:
+        ParseDlrrBlock(current_block, block_length);
+        break;
+      case TargetBitrate::kBlockType:
+        ParseTargetBitrateBlock(current_block, block_length);
+        break;
+      default:
+        // Unknown block, ignore.
+        RTC_LOG(LS_WARNING)
+            << "Unknown extended report block type " << block_type;
+        break;
+    }
+    current_block = next_block;
+  }
+
+  return true;
+}
+
+void ExtendedReports::SetRrtr(const Rrtr& rrtr) {
+  if (rrtr_block_)
+    RTC_LOG(LS_WARNING) << "Rrtr already set, overwriting.";
+  rrtr_block_.emplace(rrtr);
+}
+
+bool ExtendedReports::AddDlrrItem(const ReceiveTimeInfo& time_info) {
+  if (dlrr_block_.sub_blocks().size() >= kMaxNumberOfDlrrItems) {
+    RTC_LOG(LS_WARNING) << "Reached maximum number of DLRR items.";
+    return false;
+  }
+  dlrr_block_.AddDlrrItem(time_info);
+  return true;
+}
+
+void ExtendedReports::SetTargetBitrate(const TargetBitrate& bitrate) {
+  if (target_bitrate_)
+    RTC_LOG(LS_WARNING) << "TargetBitrate already set, overwriting.";
+
+  target_bitrate_ = bitrate;
+}
+
+size_t ExtendedReports::BlockLength() const {
+  return kHeaderLength + kXrBaseLength + RrtrLength() + DlrrLength() +
+         TargetBitrateLength();
+}
+
+bool ExtendedReports::Create(uint8_t* packet,
+                             size_t* index,
+                             size_t max_length,
+                             PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  size_t index_end = *index + BlockLength();
+  const uint8_t kReserved = 0;
+  CreateHeader(kReserved, kPacketType, HeaderLength(), packet, index);
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, sender_ssrc());
+  *index += sizeof(uint32_t);
+  if (rrtr_block_) {
+    rrtr_block_->Create(packet + *index);
+    *index += Rrtr::kLength;
+  }
+  if (dlrr_block_) {
+    dlrr_block_.Create(packet + *index);
+    *index += dlrr_block_.BlockLength();
+  }
+  if (target_bitrate_) {
+    target_bitrate_->Create(packet + *index);
+    *index += target_bitrate_->BlockLength();
+  }
+  RTC_CHECK_EQ(*index, index_end);
+  return true;
+}
+
+size_t ExtendedReports::TargetBitrateLength() const {
+  if (target_bitrate_)
+    return target_bitrate_->BlockLength();
+  return 0;
+}
+
+void ExtendedReports::ParseRrtrBlock(const uint8_t* block,
+                                     uint16_t block_length) {
+  if (block_length != Rrtr::kBlockLength) {
+    RTC_LOG(LS_WARNING) << "Incorrect rrtr block size " << block_length
+                        << " Should be " << Rrtr::kBlockLength;
+    return;
+  }
+  if (rrtr_block_) {
+    RTC_LOG(LS_WARNING)
+        << "Two rrtr blocks found in same Extended Report packet";
+    return;
+  }
+  rrtr_block_.emplace();
+  rrtr_block_->Parse(block);
+}
+
+void ExtendedReports::ParseDlrrBlock(const uint8_t* block,
+                                     uint16_t block_length) {
+  if (dlrr_block_) {
+    RTC_LOG(LS_WARNING)
+        << "Two Dlrr blocks found in same Extended Report packet";
+    return;
+  }
+  dlrr_block_.Parse(block, block_length);
+}
+
+void ExtendedReports::ParseTargetBitrateBlock(const uint8_t* block,
+                                              uint16_t block_length) {
+  target_bitrate_.emplace();
+  target_bitrate_->Parse(block, block_length);
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
new file mode 100644
index 0000000000..6c804bbc7b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+// From RFC 3611: RTP Control Protocol Extended Reports (RTCP XR).
+class ExtendedReports : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 207;
+  static constexpr size_t kMaxNumberOfDlrrItems = 50;
+
+  ExtendedReports();
+  ExtendedReports(const ExtendedReports& xr);
+  ~ExtendedReports() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void SetRrtr(const Rrtr& rrtr);
+  bool AddDlrrItem(const ReceiveTimeInfo& time_info);
+  void SetTargetBitrate(const TargetBitrate& target_bitrate);
+
+  const absl::optional<Rrtr>& rrtr() const { return rrtr_block_; }
+  const Dlrr& dlrr() const { return dlrr_block_; }
+  const absl::optional<TargetBitrate>& target_bitrate() const {
+    return target_bitrate_;
+  }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static constexpr size_t kXrBaseLength = 4;
+
+  size_t RrtrLength() const { return rrtr_block_ ? Rrtr::kLength : 0; }
+  size_t DlrrLength() const { return dlrr_block_.BlockLength(); }
+  size_t TargetBitrateLength() const;
+
+  void ParseRrtrBlock(const uint8_t* block, uint16_t block_length);
+  void ParseDlrrBlock(const uint8_t* block, uint16_t block_length);
+  void ParseTargetBitrateBlock(const uint8_t* block, uint16_t block_length);
+
+  absl::optional<Rrtr> rrtr_block_;
+  Dlrr dlrr_block_;  // Dlrr without items treated same as no dlrr block.
+  absl::optional<TargetBitrate> target_bitrate_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc
new file mode 100644
index 0000000000..3d9a2a3408
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+
+#include "rtc_base/random.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::make_tuple;
+using ::testing::SizeIs;
+using webrtc::rtcp::ExtendedReports;
+using webrtc::rtcp::ReceiveTimeInfo;
+using webrtc::rtcp::Rrtr;
+
+namespace webrtc {
+namespace {
+constexpr uint32_t kSenderSsrc = 0x12345678;
+constexpr uint8_t kEmptyPacket[] = {0x80, 207,  0x00, 0x01,
+                                    0x12, 0x34, 0x56, 0x78};
+}  // namespace
+
+class RtcpPacketExtendedReportsTest : public ::testing::Test {
+ public:
+  RtcpPacketExtendedReportsTest() : random_(0x123456789) {}
+
+ protected:
+  template <typename T>
+  T Rand() {
+    return random_.Rand<T>();
+  }
+
+ private:
+  Random random_;
+};
+
+template <>
+ReceiveTimeInfo RtcpPacketExtendedReportsTest::Rand<ReceiveTimeInfo>() {
+  uint32_t ssrc = Rand<uint32_t>();
+  uint32_t last_rr = Rand<uint32_t>();
+  uint32_t delay_since_last_rr = Rand<uint32_t>();
+  return ReceiveTimeInfo(ssrc, last_rr, delay_since_last_rr);
+}
+
+template <>
+NtpTime RtcpPacketExtendedReportsTest::Rand<NtpTime>() {
+  uint32_t secs = Rand<uint32_t>();
+  uint32_t frac = Rand<uint32_t>();
+  return NtpTime(secs, frac);
+}
+
+template <>
+Rrtr RtcpPacketExtendedReportsTest::Rand<Rrtr>() {
+  Rrtr rrtr;
+  rrtr.SetNtp(Rand<NtpTime>());
+  return rrtr;
+}
+
+TEST_F(RtcpPacketExtendedReportsTest, CreateWithoutReportBlocks) {
+  ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+
+  rtc::Buffer packet = xr.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kEmptyPacket));
+}
+
+TEST_F(RtcpPacketExtendedReportsTest, ParseWithoutReportBlocks) {
+  ExtendedReports parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kEmptyPacket, &parsed));
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_FALSE(parsed.rrtr());
+  EXPECT_FALSE(parsed.dlrr());
+}
+
+TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithRrtrBlock) {
+  const Rrtr kRrtr = Rand<Rrtr>();
+  ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.SetRrtr(kRrtr);
+  rtc::Buffer packet = xr.Build();
+
+  ExtendedReports mparsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &mparsed));
+  const ExtendedReports& parsed = mparsed;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRrtr, parsed.rrtr());
+}
+
+TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithDlrrWithOneSubBlock) {
+  const ReceiveTimeInfo kTimeInfo = Rand<ReceiveTimeInfo>();
+  ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(kTimeInfo);
+
+  rtc::Buffer packet = xr.Build();
+
+  ExtendedReports mparsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &mparsed));
+  const ExtendedReports& parsed = mparsed;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_THAT(parsed.dlrr().sub_blocks(), ElementsAre(kTimeInfo));
+}
+
+TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithDlrrWithTwoSubBlocks) {
+  const ReceiveTimeInfo kTimeInfo1 = Rand<ReceiveTimeInfo>();
+  const ReceiveTimeInfo kTimeInfo2 = Rand<ReceiveTimeInfo>();
+  ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(kTimeInfo1);
+  xr.AddDlrrItem(kTimeInfo2);
+
+  rtc::Buffer packet = xr.Build();
+
+  ExtendedReports mparsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &mparsed));
+  const ExtendedReports& parsed = mparsed;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_THAT(parsed.dlrr().sub_blocks(), ElementsAre(kTimeInfo1, kTimeInfo2));
+}
+
+TEST_F(RtcpPacketExtendedReportsTest, CreateLimitsTheNumberOfDlrrSubBlocks) {
+  const ReceiveTimeInfo kTimeInfo = Rand<ReceiveTimeInfo>();
+  ExtendedReports xr;
+
+  for (size_t i = 0; i < ExtendedReports::kMaxNumberOfDlrrItems; ++i)
+    EXPECT_TRUE(xr.AddDlrrItem(kTimeInfo));
+  EXPECT_FALSE(xr.AddDlrrItem(kTimeInfo));
+
+  EXPECT_THAT(xr.dlrr().sub_blocks(),
+              SizeIs(ExtendedReports::kMaxNumberOfDlrrItems));
+}
+
+TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithMaximumReportBlocks) {
+  const Rrtr kRrtr = Rand<Rrtr>();
+
+  ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.SetRrtr(kRrtr);
+  for (size_t i = 0; i < ExtendedReports::kMaxNumberOfDlrrItems; ++i)
+    xr.AddDlrrItem(Rand<ReceiveTimeInfo>());
+
+  rtc::Buffer packet = xr.Build();
+
+  ExtendedReports mparsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &mparsed));
+  const ExtendedReports& parsed = mparsed;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRrtr, parsed.rrtr());
+  EXPECT_THAT(parsed.dlrr().sub_blocks(),
+              ElementsAreArray(xr.dlrr().sub_blocks()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc
new file mode 100644
index 0000000000..fd4a4c947a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Fir::kFeedbackMessageType;
+// RFC 4585: Feedback format.
+// Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|   FMT   |       PT      |          length               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |             SSRC of media source (unused) = 0                 |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :            Feedback Control Information (FCI)                 :
+//  :                                                               :
+// Full intra request (FIR) (RFC 5104).
+// The Feedback Control Information (FCI) for the Full Intra Request
+// consists of one or more FCI entries.
+// FCI:
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                              SSRC                             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  | Seq nr.       |    Reserved = 0                               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+Fir::Fir() = default;
+
+Fir::Fir(const Fir& fir) = default;
+
+Fir::~Fir() = default;
+
+bool Fir::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  // The FCI field MUST contain one or more FIR entries.
+  if (packet.payload_size_bytes() < kCommonFeedbackLength + kFciLength) {
+    RTC_LOG(LS_WARNING) << "Packet is too small to be a valid FIR packet.";
+    return false;
+  }
+
+  if ((packet.payload_size_bytes() - kCommonFeedbackLength) % kFciLength != 0) {
+    RTC_LOG(LS_WARNING) << "Invalid size for a valid FIR packet.";
+    return false;
+  }
+
+  ParseCommonFeedback(packet.payload());
+
+  size_t number_of_fci_items =
+      (packet.payload_size_bytes() - kCommonFeedbackLength) / kFciLength;
+  const uint8_t* next_fci = packet.payload() + kCommonFeedbackLength;
+  items_.resize(number_of_fci_items);
+  for (Request& request : items_) {
+    request.ssrc = ByteReader<uint32_t>::ReadBigEndian(next_fci);
+    request.seq_nr = ByteReader<uint8_t>::ReadBigEndian(next_fci + 4);
+    next_fci += kFciLength;
+  }
+  return true;
+}
+
+size_t Fir::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength + kFciLength * items_.size();
+}
+
+bool Fir::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 PacketReadyCallback callback) const {
+  RTC_DCHECK(!items_.empty());
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  size_t index_end = *index + BlockLength();
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  RTC_DCHECK_EQ(Psfb::media_ssrc(), 0);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+
+  constexpr uint32_t kReserved = 0;
+  for (const Request& request : items_) {
+    ByteWriter<uint32_t>::WriteBigEndian(packet + *index, request.ssrc);
+    ByteWriter<uint8_t>::WriteBigEndian(packet + *index + 4, request.seq_nr);
+    ByteWriter<uint32_t, 3>::WriteBigEndian(packet + *index + 5, kReserved);
+    *index += kFciLength;
+  }
+  RTC_CHECK_EQ(*index, index_end);
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h
new file mode 100644
index 0000000000..383dc96114
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_FIR_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_FIR_H_
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+// Full intra request (FIR) (RFC 5104).
+class Fir : public Psfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 4;
+  struct Request {
+    Request() : ssrc(0), seq_nr(0) {}
+    Request(uint32_t ssrc, uint8_t seq_nr) : ssrc(ssrc), seq_nr(seq_nr) {}
+    uint32_t ssrc;
+    uint8_t seq_nr;
+  };
+
+  Fir();
+  Fir(const Fir& fir);
+  ~Fir() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void AddRequestTo(uint32_t ssrc, uint8_t seq_num) {
+    items_.emplace_back(ssrc, seq_num);
+  }
+  const std::vector<Request>& requests() const { return items_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static constexpr size_t kFciLength = 8;
+
+  // SSRC of media source is not used in FIR packet. Shadow base functions.
+  void SetMediaSsrc(uint32_t ssrc);
+  uint32_t media_ssrc() const;
+
+  std::vector<Request> items_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_FIR_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir_unittest.cc
new file mode 100644
index 0000000000..01593e12ba
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/fir_unittest.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::AllOf;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::make_tuple;
+using webrtc::rtcp::Fir;
+
+namespace webrtc {
+namespace {
+
+constexpr uint32_t kSenderSsrc = 0x12345678;
+constexpr uint32_t kRemoteSsrc = 0x23456789;
+constexpr uint8_t kSeqNr = 13;
+// Manually created Fir packet matching constants above.
+constexpr uint8_t kPacket[] = {0x84, 206,  0x00, 0x04, 0x12, 0x34, 0x56,
+                               0x78, 0x00, 0x00, 0x00, 0x00, 0x23, 0x45,
+                               0x67, 0x89, 0x0d, 0x00, 0x00, 0x00};
+}  // namespace
+
+TEST(RtcpPacketFirTest, Parse) {
+  Fir mutable_parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &mutable_parsed));
+  const Fir& parsed = mutable_parsed;  // Read values from constant object.
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_THAT(parsed.requests(),
+              ElementsAre(AllOf(Field(&Fir::Request::ssrc, Eq(kRemoteSsrc)),
+                                Field(&Fir::Request::seq_nr, Eq(kSeqNr)))));
+}
+
+TEST(RtcpPacketFirTest, Create) {
+  Fir fir;
+  fir.SetSenderSsrc(kSenderSsrc);
+  fir.AddRequestTo(kRemoteSsrc, kSeqNr);
+
+  rtc::Buffer packet = fir.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketFirTest, TwoFciEntries) {
+  Fir fir;
+  fir.SetSenderSsrc(kSenderSsrc);
+  fir.AddRequestTo(kRemoteSsrc, kSeqNr);
+  fir.AddRequestTo(kRemoteSsrc + 1, kSeqNr + 1);
+
+  rtc::Buffer packet = fir.Build();
+  Fir parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_THAT(parsed.requests(),
+              ElementsAre(AllOf(Field(&Fir::Request::ssrc, Eq(kRemoteSsrc)),
+                                Field(&Fir::Request::seq_nr, Eq(kSeqNr))),
+                          AllOf(Field(&Fir::Request::ssrc, Eq(kRemoteSsrc + 1)),
+                                Field(&Fir::Request::seq_nr, Eq(kSeqNr + 1)))));
+}
+
+TEST(RtcpPacketFirTest, ParseFailsOnZeroFciEntries) {
+  constexpr uint8_t kPacketWithoutFci[] = {0x84, 206,  0x00, 0x02, 0x12, 0x34,
+                                           0x56, 0x78, 0x00, 0x00, 0x00, 0x00};
+  Fir parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kPacketWithoutFci, &parsed));
+}
+
+TEST(RtcpPacketFirTest, ParseFailsOnFractionalFciEntries) {
+  constexpr uint8_t kPacketWithOneAndHalfFci[] = {
+      0x84, 206,  0x00, 0x05, 0x12, 0x34, 0x56, 0x78, 0x00, 0x00, 0x00, 0x00,
+      0x23, 0x45, 0x67, 0x89, 0x0d, 0x00, 0x00, 0x00, 'h',  'a',  'l',  'f'};
+
+  Fir parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kPacketWithOneAndHalfFci, &parsed));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc
new file mode 100644
index 0000000000..0817846f95
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification.cc
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// Loss Notification
+// -----------------
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P| FMT=15  |   PT=206      |             length            |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  0 |                  SSRC of packet sender                        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                  SSRC of media source                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |  Unique identifier 'L' 'N' 'T' 'F'                            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 | Last Decoded Sequence Number  | Last Received SeqNum Delta  |D|
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+LossNotification::LossNotification()
+    : last_decoded_(0), last_received_(0), decodability_flag_(false) {}
+
+LossNotification::LossNotification(uint16_t last_decoded,
+                                   uint16_t last_received,
+                                   bool decodability_flag)
+    : last_decoded_(last_decoded),
+      last_received_(last_received),
+      decodability_flag_(decodability_flag) {}
+
+LossNotification::LossNotification(const LossNotification& rhs) = default;
+
+LossNotification::~LossNotification() = default;
+
+size_t LossNotification::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength + kLossNotificationPayloadLength;
+}
+
+bool LossNotification::Create(uint8_t* packet,
+                              size_t* index,
+                              size_t max_length,
+                              PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+
+  const size_t index_end = *index + BlockLength();
+
+  // Note: `index` updated by the function below.
+  CreateHeader(Psfb::kAfbMessageType, kPacketType, HeaderLength(), packet,
+               index);
+
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, kUniqueIdentifier);
+  *index += sizeof(uint32_t);
+
+  ByteWriter<uint16_t>::WriteBigEndian(packet + *index, last_decoded_);
+  *index += sizeof(uint16_t);
+
+  const uint16_t last_received_delta = last_received_ - last_decoded_;
+  RTC_DCHECK_LE(last_received_delta, 0x7fff);
+  const uint16_t last_received_delta_and_decodability =
+      (last_received_delta << 1) | (decodability_flag_ ? 0x0001 : 0x0000);
+
+  ByteWriter<uint16_t>::WriteBigEndian(packet + *index,
+                                       last_received_delta_and_decodability);
+  *index += sizeof(uint16_t);
+
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+bool LossNotification::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), Psfb::kAfbMessageType);
+
+  if (packet.payload_size_bytes() <
+      kCommonFeedbackLength + kLossNotificationPayloadLength) {
+    return false;
+  }
+
+  const uint8_t* const payload = packet.payload();
+
+  if (ByteReader<uint32_t>::ReadBigEndian(&payload[8]) != kUniqueIdentifier) {
+    return false;
+  }
+
+  ParseCommonFeedback(payload);
+
+  last_decoded_ = ByteReader<uint16_t>::ReadBigEndian(&payload[12]);
+
+  const uint16_t last_received_delta_and_decodability =
+      ByteReader<uint16_t>::ReadBigEndian(&payload[14]);
+  last_received_ = last_decoded_ + (last_received_delta_and_decodability >> 1);
+  decodability_flag_ = (last_received_delta_and_decodability & 0x0001);
+
+  return true;
+}
+
+bool LossNotification::Set(uint16_t last_decoded,
+                           uint16_t last_received,
+                           bool decodability_flag) {
+  const uint16_t delta = last_received - last_decoded;
+  if (delta > 0x7fff) {
+    return false;
+  }
+  last_received_ = last_received;
+  last_decoded_ = last_decoded;
+  decodability_flag_ = decodability_flag;
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h
new file mode 100644
index 0000000000..0f70cf75c3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_LOSS_NOTIFICATION_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_LOSS_NOTIFICATION_H_
+
+#include "absl/base/attributes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class LossNotification : public Psfb {
+ public:
+  LossNotification();
+  LossNotification(uint16_t last_decoded,
+                   uint16_t last_received,
+                   bool decodability_flag);
+  LossNotification(const LossNotification& other);
+  ~LossNotification() override;
+
+  size_t BlockLength() const override;
+
+  ABSL_MUST_USE_RESULT
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+  // Parse assumes header is already parsed and validated.
+  ABSL_MUST_USE_RESULT
+  bool Parse(const CommonHeader& packet);
+
+  // Set all of the values transmitted by the loss notification message.
+  // If the values may not be represented by a loss notification message,
+  // false is returned, and no change is made to the object; this happens
+  // when `last_received` is ahead of `last_decoded` by more than 0x7fff.
+  // This is because `last_received` is represented on the wire as a delta,
+  // and only 15 bits are available for that delta.
+  ABSL_MUST_USE_RESULT
+  bool Set(uint16_t last_decoded,
+           uint16_t last_received,
+           bool decodability_flag);
+
+  // RTP sequence number of the first packet belong to the last decoded
+  // non-discardable frame.
+  uint16_t last_decoded() const { return last_decoded_; }
+
+  // RTP sequence number of the last received packet.
+  uint16_t last_received() const { return last_received_; }
+
+  // A decodability flag, whose specific meaning depends on the last-received
+  // RTP sequence number. The decodability flag is true if and only if all of
+  // the frame's dependencies are known to be decodable, and the frame itself
+  // is not yet known to be unassemblable.
+  // * Clarification #1: In a multi-packet frame, the first packet's
+  //   dependencies are known, but it is not yet known whether all parts
+  //   of the current frame will be received.
+  // * Clarification #2: In a multi-packet frame, the dependencies would be
+  //   unknown if the first packet was not received. Then, the packet will
+  //   be known-unassemblable.
+  bool decodability_flag() const { return decodability_flag_; }
+
+ private:
+  static constexpr uint32_t kUniqueIdentifier = 0x4C4E5446;  // 'L' 'N' 'T' 'F'.
+  static constexpr size_t kLossNotificationPayloadLength = 8;
+
+  uint16_t last_decoded_;
+  uint16_t last_received_;
+  bool decodability_flag_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_LOSS_NOTIFICATION_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc
new file mode 100644
index 0000000000..c38e7f4438
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/loss_notification_unittest.cc
@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+namespace webrtc {
+
+using ::testing::ElementsAreArray;
+using ::testing::make_tuple;
+using ::webrtc::rtcp::LossNotification;
+
+TEST(RtcpPacketLossNotificationTest, SetWithIllegalValuesFails) {
+  constexpr uint16_t kLastDecoded = 0x3c7b;
+  constexpr uint16_t kLastReceived = kLastDecoded + 0x7fff + 1;
+  constexpr bool kDecodabilityFlag = true;
+  LossNotification loss_notification;
+  EXPECT_FALSE(
+      loss_notification.Set(kLastDecoded, kLastReceived, kDecodabilityFlag));
+}
+
+TEST(RtcpPacketLossNotificationTest, SetWithLegalValuesSucceeds) {
+  constexpr uint16_t kLastDecoded = 0x3c7b;
+  constexpr uint16_t kLastReceived = kLastDecoded + 0x7fff;
+  constexpr bool kDecodabilityFlag = true;
+  LossNotification loss_notification;
+  EXPECT_TRUE(
+      loss_notification.Set(kLastDecoded, kLastReceived, kDecodabilityFlag));
+}
+
+TEST(RtcpPacketLossNotificationTest, CreateProducesExpectedWireFormat) {
+  // Note that (0x6542 >> 1) is used just to make the pattern in kPacket
+  // more apparent; there's nothing truly special about the value,
+  // it's only an implementation detail that last-received is represented
+  // as a delta from last-decoded, and that this delta is shifted before
+  // it's put on the wire.
+  constexpr uint16_t kLastDecoded = 0x3c7b;
+  constexpr uint16_t kLastReceived = kLastDecoded + (0x6542 >> 1);
+  constexpr bool kDecodabilityFlag = true;
+
+  const uint8_t kPacket[] = {0x8f, 206,  0x00, 0x04, 0x12, 0x34, 0x56, 0x78,  //
+                             0xab, 0xcd, 0xef, 0x01, 'L',  'N',  'T',  'F',   //
+                             0x3c, 0x7b, 0x65, 0x43};
+
+  LossNotification loss_notification;
+  loss_notification.SetSenderSsrc(0x12345678);
+  loss_notification.SetMediaSsrc(0xabcdef01);
+  ASSERT_TRUE(
+      loss_notification.Set(kLastDecoded, kLastReceived, kDecodabilityFlag));
+
+  rtc::Buffer packet = loss_notification.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketLossNotificationTest,
+     ParseFailsOnTooSmallPacketToBeLossNotification) {
+  uint8_t packet[] = {0x8f, 206,  0x00, 0x04, 0x12, 0x34, 0x56, 0x78,  //
+                      0xab, 0xcd, 0xef, 0x01, 'L',  'N',  'T',  'F',   //
+                      0x3c, 0x7b, 0x65, 0x43};
+  size_t packet_length_bytes = sizeof(packet);
+
+  LossNotification loss_notification;
+
+  // First, prove that the failure we're expecting to see happens because of
+  // the length, by showing that before the modification to the length,
+  // the packet was correctly parsed.
+  ASSERT_TRUE(
+      test::ParseSinglePacket(packet, packet_length_bytes, &loss_notification));
+
+  // Show that after shaving off a word, the packet is no longer parsable.
+  packet[3] -= 1;            // Change the `length` field of the RTCP packet.
+  packet_length_bytes -= 4;  // Effectively forget the last 32-bit word.
+  EXPECT_FALSE(
+      test::ParseSinglePacket(packet, packet_length_bytes, &loss_notification));
+}
+
+TEST(RtcpPacketLossNotificationTest,
+     ParseFailsWhenUniqueIdentifierIsNotLossNotification) {
+  uint8_t packet[] = {0x8f, 206,  0x00, 0x04, 0x12, 0x34, 0x56, 0x78,  //
+                      0xab, 0xcd, 0xef, 0x01, 'L',  'N',  'T',  'F',   //
+                      0x3c, 0x7b, 0x65, 0x43};
+
+  LossNotification loss_notification;
+
+  // First, prove that the failure we're expecting to see happens because of
+  // the identifier, by showing that before the modification to the identifier,
+  // the packet was correctly parsed.
+  ASSERT_TRUE(test::ParseSinglePacket(packet, &loss_notification));
+
+  // Show that after changing the identifier, the packet is no longer parsable.
+  RTC_DCHECK_EQ(packet[12], 'L');
+  RTC_DCHECK_EQ(packet[13], 'N');
+  RTC_DCHECK_EQ(packet[14], 'T');
+  RTC_DCHECK_EQ(packet[15], 'F');
+  packet[14] = 'x';
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &loss_notification));
+}
+
+TEST(RtcpPacketLossNotificationTest,
+     ParseLegalLossNotificationMessagesCorrectly) {
+  // Note that (0x6542 >> 1) is used just to make the pattern in kPacket
+  // more apparent; there's nothing truly special about the value,
+  // it's only an implementation detail that last-received is represented
+  // as a delta from last-decoded, and that this delta is shifted before
+  // it's put on the wire.
+  constexpr uint16_t kLastDecoded = 0x3c7b;
+  constexpr uint16_t kLastReceived = kLastDecoded + (0x6542 >> 1);
+  constexpr bool kDecodabilityFlag = true;
+
+  const uint8_t kPacket[] = {0x8f, 206,  0x00, 0x04, 0x12, 0x34, 0x56, 0x78,  //
+                             0xab, 0xcd, 0xef, 0x01, 'L',  'N',  'T',  'F',   //
+                             0x3c, 0x7b, 0x65, 0x43};
+
+  LossNotification loss_notification;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &loss_notification));
+
+  EXPECT_EQ(loss_notification.sender_ssrc(), 0x12345678u);
+  EXPECT_EQ(loss_notification.media_ssrc(), 0xabcdef01u);
+  EXPECT_EQ(loss_notification.last_decoded(), kLastDecoded);
+  EXPECT_EQ(loss_notification.last_received(), kLastReceived);
+  EXPECT_EQ(loss_notification.decodability_flag(), kDecodabilityFlag);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc
new file mode 100644
index 0000000000..6fe7eade62
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc
@@ -0,0 +1,176 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <utility>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Nack::kFeedbackMessageType;
+constexpr size_t Nack::kNackItemLength;
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+//
+// Generic NACK (RFC 4585).
+//
+// FCI:
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |            PID                |             BLP               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+Nack::Nack() = default;
+Nack::Nack(const Nack& rhs) = default;
+Nack::~Nack() = default;
+
+bool Nack::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  if (packet.payload_size_bytes() < kCommonFeedbackLength + kNackItemLength) {
+    RTC_LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
+                        << " is too small for a Nack.";
+    return false;
+  }
+  size_t nack_items =
+      (packet.payload_size_bytes() - kCommonFeedbackLength) / kNackItemLength;
+
+  ParseCommonFeedback(packet.payload());
+  const uint8_t* next_nack = packet.payload() + kCommonFeedbackLength;
+
+  packet_ids_.clear();
+  packed_.resize(nack_items);
+  for (size_t index = 0; index < nack_items; ++index) {
+    packed_[index].first_pid = ByteReader<uint16_t>::ReadBigEndian(next_nack);
+    packed_[index].bitmask = ByteReader<uint16_t>::ReadBigEndian(next_nack + 2);
+    next_nack += kNackItemLength;
+  }
+  Unpack();
+
+  return true;
+}
+
+size_t Nack::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength +
+         packed_.size() * kNackItemLength;
+}
+
+bool Nack::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  PacketReadyCallback callback) const {
+  RTC_DCHECK(!packed_.empty());
+  // If nack list can't fit in packet, try to fragment.
+  constexpr size_t kNackHeaderLength = kHeaderLength + kCommonFeedbackLength;
+  for (size_t nack_index = 0; nack_index < packed_.size();) {
+    size_t bytes_left_in_buffer = max_length - *index;
+    if (bytes_left_in_buffer < kNackHeaderLength + kNackItemLength) {
+      if (!OnBufferFull(packet, index, callback))
+        return false;
+      continue;
+    }
+    size_t num_nack_fields =
+        std::min((bytes_left_in_buffer - kNackHeaderLength) / kNackItemLength,
+                 packed_.size() - nack_index);
+
+    size_t payload_size_bytes =
+        kCommonFeedbackLength + (num_nack_fields * kNackItemLength);
+    size_t payload_size_32bits =
+        rtc::CheckedDivExact<size_t>(payload_size_bytes, 4);
+    CreateHeader(kFeedbackMessageType, kPacketType, payload_size_32bits, packet,
+                 index);
+
+    CreateCommonFeedback(packet + *index);
+    *index += kCommonFeedbackLength;
+
+    size_t nack_end_index = nack_index + num_nack_fields;
+    for (; nack_index < nack_end_index; ++nack_index) {
+      const PackedNack& item = packed_[nack_index];
+      ByteWriter<uint16_t>::WriteBigEndian(packet + *index + 0, item.first_pid);
+      ByteWriter<uint16_t>::WriteBigEndian(packet + *index + 2, item.bitmask);
+      *index += kNackItemLength;
+    }
+    RTC_DCHECK_LE(*index, max_length);
+  }
+
+  return true;
+}
+
+void Nack::SetPacketIds(const uint16_t* nack_list, size_t length) {
+  RTC_DCHECK(nack_list);
+  SetPacketIds(std::vector<uint16_t>(nack_list, nack_list + length));
+}
+
+void Nack::SetPacketIds(std::vector<uint16_t> nack_list) {
+  RTC_DCHECK(packet_ids_.empty());
+  RTC_DCHECK(packed_.empty());
+  packet_ids_ = std::move(nack_list);
+  Pack();
+}
+
+void Nack::Pack() {
+  RTC_DCHECK(!packet_ids_.empty());
+  RTC_DCHECK(packed_.empty());
+  auto it = packet_ids_.begin();
+  const auto end = packet_ids_.end();
+  while (it != end) {
+    PackedNack item;
+    item.first_pid = *it++;
+    // Bitmask specifies losses in any of the 16 packets following the pid.
+    item.bitmask = 0;
+    while (it != end) {
+      uint16_t shift = static_cast<uint16_t>(*it - item.first_pid - 1);
+      if (shift <= 15) {
+        item.bitmask |= (1 << shift);
+        ++it;
+      } else {
+        break;
+      }
+    }
+    packed_.push_back(item);
+  }
+}
+
+void Nack::Unpack() {
+  RTC_DCHECK(packet_ids_.empty());
+  RTC_DCHECK(!packed_.empty());
+  for (const PackedNack& item : packed_) {
+    packet_ids_.push_back(item.first_pid);
+    uint16_t pid = item.first_pid + 1;
+    for (uint16_t bitmask = item.bitmask; bitmask != 0; bitmask >>= 1, ++pid) {
+      if (bitmask & 1)
+        packet_ids_.push_back(pid);
+    }
+  }
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
new file mode 100644
index 0000000000..9153733fb9
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class Nack : public Rtpfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 1;
+  Nack();
+  Nack(const Nack&);
+  ~Nack() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void SetPacketIds(const uint16_t* nack_list, size_t length);
+  void SetPacketIds(std::vector<uint16_t> nack_list);
+  const std::vector<uint16_t>& packet_ids() const { return packet_ids_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static constexpr size_t kNackItemLength = 4;
+  struct PackedNack {
+    uint16_t first_pid;
+    uint16_t bitmask;
+  };
+
+  void Pack();    // Fills packed_ using packed_ids_. (used in SetPacketIds).
+  void Unpack();  // Fills packet_ids_ using packed_. (used in Parse).
+
+  std::vector<PackedNack> packed_;
+  std::vector<uint16_t> packet_ids_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_NACK_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
new file mode 100644
index 0000000000..aabae0dc48
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/nack_unittest.cc
@@ -0,0 +1,171 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::Invoke;
+using ::testing::make_tuple;
+using ::testing::MockFunction;
+using ::testing::UnorderedElementsAreArray;
+using ::webrtc::rtcp::Nack;
+
+constexpr uint32_t kSenderSsrc = 0x12345678;
+constexpr uint32_t kRemoteSsrc = 0x23456789;
+
+constexpr uint16_t kList[] = {0, 1, 3, 8, 16};
+constexpr size_t kListLength = sizeof(kList) / sizeof(kList[0]);
+constexpr uint8_t kVersionBits = 2 << 6;
+// clang-format off
+constexpr uint8_t kPacket[] = {
+    kVersionBits | Nack::kFeedbackMessageType, Nack::kPacketType, 0, 3,
+    0x12, 0x34, 0x56, 0x78,
+    0x23, 0x45, 0x67, 0x89,
+    0x00, 0x00, 0x80, 0x85};
+
+constexpr uint16_t kWrapList[] = {0xffdc, 0xffec, 0xfffe, 0xffff, 0x0000,
+                                  0x0001, 0x0003, 0x0014, 0x0064};
+constexpr size_t kWrapListLength = sizeof(kWrapList) / sizeof(kWrapList[0]);
+constexpr uint8_t kWrapPacket[] = {
+    kVersionBits | Nack::kFeedbackMessageType, Nack::kPacketType, 0, 6,
+    0x12, 0x34, 0x56, 0x78,
+    0x23, 0x45, 0x67, 0x89,
+    0xff, 0xdc, 0x80, 0x00,
+    0xff, 0xfe, 0x00, 0x17,
+    0x00, 0x14, 0x00, 0x00,
+    0x00, 0x64, 0x00, 0x00};
+constexpr uint8_t kTooSmallPacket[] = {
+    kVersionBits | Nack::kFeedbackMessageType, Nack::kPacketType, 0, 2,
+    0x12, 0x34, 0x56, 0x78,
+    0x23, 0x45, 0x67, 0x89};
+// clang-format on
+}  // namespace
+
+TEST(RtcpPacketNackTest, Create) {
+  Nack nack;
+  nack.SetSenderSsrc(kSenderSsrc);
+  nack.SetMediaSsrc(kRemoteSsrc);
+  nack.SetPacketIds(kList, kListLength);
+
+  rtc::Buffer packet = nack.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketNackTest, Parse) {
+  Nack parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &parsed));
+  const Nack& const_parsed = parsed;
+
+  EXPECT_EQ(kSenderSsrc, const_parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, const_parsed.media_ssrc());
+  EXPECT_THAT(const_parsed.packet_ids(), ElementsAreArray(kList));
+}
+
+TEST(RtcpPacketNackTest, CreateWrap) {
+  Nack nack;
+  nack.SetSenderSsrc(kSenderSsrc);
+  nack.SetMediaSsrc(kRemoteSsrc);
+  nack.SetPacketIds(kWrapList, kWrapListLength);
+
+  rtc::Buffer packet = nack.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kWrapPacket));
+}
+
+TEST(RtcpPacketNackTest, ParseWrap) {
+  Nack parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kWrapPacket, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+  EXPECT_THAT(parsed.packet_ids(), ElementsAreArray(kWrapList));
+}
+
+TEST(RtcpPacketNackTest, BadOrder) {
+  // Does not guarantee optimal packing, but should guarantee correctness.
+  const uint16_t kUnorderedList[] = {1, 25, 13, 12, 9, 27, 29};
+  const size_t kUnorderedListLength =
+      sizeof(kUnorderedList) / sizeof(kUnorderedList[0]);
+  Nack nack;
+  nack.SetSenderSsrc(kSenderSsrc);
+  nack.SetMediaSsrc(kRemoteSsrc);
+  nack.SetPacketIds(kUnorderedList, kUnorderedListLength);
+
+  rtc::Buffer packet = nack.Build();
+
+  Nack parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+  EXPECT_THAT(parsed.packet_ids(), UnorderedElementsAreArray(kUnorderedList));
+}
+
+TEST(RtcpPacketNackTest, CreateFragmented) {
+  Nack nack;
+  const uint16_t kList[] = {1, 100, 200, 300, 400};
+  const uint16_t kListLength = sizeof(kList) / sizeof(kList[0]);
+  nack.SetSenderSsrc(kSenderSsrc);
+  nack.SetMediaSsrc(kRemoteSsrc);
+  nack.SetPacketIds(kList, kListLength);
+
+  const size_t kBufferSize = 12 + (3 * 4);  // Fits common header + 3 nack items
+
+  MockFunction<void(rtc::ArrayView<const uint8_t>)> callback;
+  EXPECT_CALL(callback, Call(_))
+      .WillOnce(Invoke([&](rtc::ArrayView<const uint8_t> packet) {
+        Nack nack;
+        EXPECT_TRUE(test::ParseSinglePacket(packet, &nack));
+        EXPECT_EQ(kSenderSsrc, nack.sender_ssrc());
+        EXPECT_EQ(kRemoteSsrc, nack.media_ssrc());
+        EXPECT_THAT(nack.packet_ids(), ElementsAre(1, 100, 200));
+      }))
+      .WillOnce(Invoke([&](rtc::ArrayView<const uint8_t> packet) {
+        Nack nack;
+        EXPECT_TRUE(test::ParseSinglePacket(packet, &nack));
+        EXPECT_EQ(kSenderSsrc, nack.sender_ssrc());
+        EXPECT_EQ(kRemoteSsrc, nack.media_ssrc());
+        EXPECT_THAT(nack.packet_ids(), ElementsAre(300, 400));
+      }));
+
+  EXPECT_TRUE(nack.Build(kBufferSize, callback.AsStdFunction()));
+}
+
+TEST(RtcpPacketNackTest, CreateFailsWithTooSmallBuffer) {
+  const uint16_t kList[] = {1};
+  const size_t kMinNackBlockSize = 16;
+  Nack nack;
+  nack.SetSenderSsrc(kSenderSsrc);
+  nack.SetMediaSsrc(kRemoteSsrc);
+  nack.SetPacketIds(kList, 1);
+
+  MockFunction<void(rtc::ArrayView<const uint8_t>)> callback;
+  EXPECT_CALL(callback, Call(_)).Times(0);
+  EXPECT_FALSE(nack.Build(kMinNackBlockSize - 1, callback.AsStdFunction()));
+}
+
+TEST(RtcpPacketNackTest, ParseFailsWithTooSmallBuffer) {
+  Nack parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kTooSmallPacket, &parsed));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc
new file mode 100644
index 0000000000..5b41aa5c2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Pli::kFeedbackMessageType;
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|   FMT   |       PT      |          length               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of media source                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :            Feedback Control Information (FCI)                 :
+//  :                                                               :
+
+Pli::Pli() = default;
+
+Pli::Pli(const Pli& pli) = default;
+
+Pli::~Pli() = default;
+
+//
+// Picture loss indication (PLI) (RFC 4585).
+// FCI: no feedback control information.
+bool Pli::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  if (packet.payload_size_bytes() < kCommonFeedbackLength) {
+    RTC_LOG(LS_WARNING) << "Packet is too small to be a valid PLI packet";
+    return false;
+  }
+
+  ParseCommonFeedback(packet.payload());
+  return true;
+}
+
+size_t Pli::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength;
+}
+
+bool Pli::Create(uint8_t* packet,
+                 size_t* index,
+                 size_t max_length,
+                 PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
new file mode 100644
index 0000000000..b9b9c45a9c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
+
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+// Picture loss indication (PLI) (RFC 4585).
+class Pli : public Psfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 1;
+
+  Pli();
+  Pli(const Pli& pli);
+  ~Pli() override;
+
+  bool Parse(const CommonHeader& packet);
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc
new file mode 100644
index 0000000000..c971e22bc1
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAreArray;
+using ::testing::make_tuple;
+using webrtc::rtcp::Pli;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+// Manually created Pli packet matching constants above.
+const uint8_t kPacket[] = {0x81, 206,  0x00, 0x02, 0x12, 0x34,
+                           0x56, 0x78, 0x23, 0x45, 0x67, 0x89};
+}  // namespace
+
+TEST(RtcpPacketPliTest, Parse) {
+  Pli mutable_parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &mutable_parsed));
+  const Pli& parsed = mutable_parsed;  // Read values from constant object.
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+}
+
+TEST(RtcpPacketPliTest, Create) {
+  Pli pli;
+  pli.SetSenderSsrc(kSenderSsrc);
+  pli.SetMediaSsrc(kRemoteSsrc);
+
+  rtc::Buffer packet = pli.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketPliTest, ParseFailsOnTooSmallPacket) {
+  const uint8_t kTooSmallPacket[] = {0x81, 206,  0x00, 0x01,
+                                     0x12, 0x34, 0x56, 0x78};
+
+  Pli parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kTooSmallPacket, &parsed));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc
new file mode 100644
index 0000000000..384d8ba811
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.cc
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Psfb::kPacketType;
+constexpr uint8_t Psfb::kAfbMessageType;
+constexpr size_t Psfb::kCommonFeedbackLength;
+// RFC 4585: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+
+void Psfb::ParseCommonFeedback(const uint8_t* payload) {
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(&payload[0]));
+  SetMediaSsrc(ByteReader<uint32_t>::ReadBigEndian(&payload[4]));
+}
+
+void Psfb::CreateCommonFeedback(uint8_t* payload) const {
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[0], sender_ssrc());
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[4], media_ssrc());
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
new file mode 100644
index 0000000000..d6b8bca7c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// PSFB: Payload-specific feedback message.
+// RFC 4585, Section 6.3.
+class Psfb : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 206;
+  static constexpr uint8_t kAfbMessageType = 15;
+
+  Psfb() = default;
+  ~Psfb() override = default;
+
+  void SetMediaSsrc(uint32_t ssrc) { media_ssrc_ = ssrc; }
+
+  uint32_t media_ssrc() const { return media_ssrc_; }
+
+ protected:
+  static constexpr size_t kCommonFeedbackLength = 8;
+  void ParseCommonFeedback(const uint8_t* payload);
+  void CreateCommonFeedback(uint8_t* payload) const;
+
+ private:
+  uint32_t media_ssrc_ = 0;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.cc
new file mode 100644
index 0000000000..8563c28373
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t RapidResyncRequest::kFeedbackMessageType;
+// RFC 4585: Feedback format.
+// Rapid Resynchronisation Request (draft-perkins-avt-rapid-rtp-sync-03).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|  FMT=5  |     PT=205    |         length=2              |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of media source                         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+bool RapidResyncRequest::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  if (packet.payload_size_bytes() != kCommonFeedbackLength) {
+    RTC_LOG(LS_WARNING) << "Packet payload size should be "
+                        << kCommonFeedbackLength << " instead of "
+                        << packet.payload_size_bytes()
+                        << " to be a valid Rapid Resynchronisation Request";
+    return false;
+  }
+
+  ParseCommonFeedback(packet.payload());
+  return true;
+}
+
+size_t RapidResyncRequest::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength;
+}
+
+bool RapidResyncRequest::Create(uint8_t* packet,
+                                size_t* index,
+                                size_t max_length,
+                                PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
new file mode 100644
index 0000000000..1955b98f5c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+// draft-perkins-avt-rapid-rtp-sync-03
+class RapidResyncRequest : public Rtpfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 5;
+
+  RapidResyncRequest() {}
+  ~RapidResyncRequest() override {}
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& header);
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
new file mode 100644
index 0000000000..d0e40fd83d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAreArray;
+using ::testing::make_tuple;
+using webrtc::rtcp::RapidResyncRequest;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+// Manually created packet matching constants above.
+const uint8_t kPacket[] = {0x85, 205,  0x00, 0x02, 0x12, 0x34,
+                           0x56, 0x78, 0x23, 0x45, 0x67, 0x89};
+}  // namespace
+
+TEST(RtcpPacketRapidResyncRequestTest, Parse) {
+  RapidResyncRequest mutable_parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &mutable_parsed));
+  const RapidResyncRequest& parsed = mutable_parsed;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parsed.media_ssrc());
+}
+
+TEST(RtcpPacketRapidResyncRequestTest, Create) {
+  RapidResyncRequest rrr;
+  rrr.SetSenderSsrc(kSenderSsrc);
+  rrr.SetMediaSsrc(kRemoteSsrc);
+
+  rtc::Buffer packet = rrr.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketRapidResyncRequestTest, ParseFailsOnTooSmallPacket) {
+  const uint8_t kTooSmallPacket[] = {0x85, 205,  0x00, 0x01,
+                                     0x12, 0x34, 0x56, 0x78};
+  RapidResyncRequest parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kTooSmallPacket, &parsed));
+}
+
+TEST(RtcpPacketRapidResyncRequestTest, ParseFailsOnTooLargePacket) {
+  const uint8_t kTooLargePacket[] = {0x85, 205,  0x00, 0x03, 0x12, 0x34,
+                                     0x56, 0x78, 0x32, 0x21, 0x65, 0x87,
+                                     0x23, 0x45, 0x67, 0x89};
+  RapidResyncRequest parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kTooLargePacket, &parsed));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc
new file mode 100644
index 0000000000..185011dff1
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.cc
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+
+#include <utility>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t ReceiverReport::kPacketType;
+constexpr size_t ReceiverReport::kMaxNumberOfReportBlocks;
+// RTCP receiver report (RFC 3550).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|    RC   |   PT=RR=201   |             length            |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                     SSRC of packet sender                     |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  |                         report block(s)                       |
+//  |                            ....                               |
+
+ReceiverReport::ReceiverReport() = default;
+
+ReceiverReport::ReceiverReport(const ReceiverReport& rhs) = default;
+
+ReceiverReport::~ReceiverReport() = default;
+
+bool ReceiverReport::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+
+  const uint8_t report_blocks_count = packet.count();
+
+  if (packet.payload_size_bytes() <
+      kRrBaseLength + report_blocks_count * ReportBlock::kLength) {
+    RTC_LOG(LS_WARNING) << "Packet is too small to contain all the data.";
+    return false;
+  }
+
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(packet.payload()));
+
+  const uint8_t* next_report_block = packet.payload() + kRrBaseLength;
+
+  report_blocks_.resize(report_blocks_count);
+  for (ReportBlock& block : report_blocks_) {
+    block.Parse(next_report_block, ReportBlock::kLength);
+    next_report_block += ReportBlock::kLength;
+  }
+
+  RTC_DCHECK_LE(next_report_block - packet.payload(),
+                static_cast<ptrdiff_t>(packet.payload_size_bytes()));
+  return true;
+}
+
+size_t ReceiverReport::BlockLength() const {
+  return kHeaderLength + kRrBaseLength +
+         report_blocks_.size() * ReportBlock::kLength;
+}
+
+bool ReceiverReport::Create(uint8_t* packet,
+                            size_t* index,
+                            size_t max_length,
+                            PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  CreateHeader(report_blocks_.size(), kPacketType, HeaderLength(), packet,
+               index);
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, sender_ssrc());
+  *index += kRrBaseLength;
+  for (const ReportBlock& block : report_blocks_) {
+    block.Create(packet + *index);
+    *index += ReportBlock::kLength;
+  }
+  return true;
+}
+
+bool ReceiverReport::AddReportBlock(const ReportBlock& block) {
+  if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
+    RTC_LOG(LS_WARNING) << "Max report blocks reached.";
+    return false;
+  }
+  report_blocks_.push_back(block);
+  return true;
+}
+
+bool ReceiverReport::SetReportBlocks(std::vector<ReportBlock> blocks) {
+  if (blocks.size() > kMaxNumberOfReportBlocks) {
+    RTC_LOG(LS_WARNING) << "Too many report blocks (" << blocks.size()
+                        << ") for receiver report.";
+    return false;
+  }
+  report_blocks_ = std::move(blocks);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
new file mode 100644
index 0000000000..b9c1c466c7
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class ReceiverReport : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 201;
+  static constexpr size_t kMaxNumberOfReportBlocks = 0x1f;
+
+  ReceiverReport();
+  ReceiverReport(const ReceiverReport&);
+  ~ReceiverReport() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  bool AddReportBlock(const ReportBlock& block);
+  bool SetReportBlocks(std::vector<ReportBlock> blocks);
+
+  const std::vector<ReportBlock>& report_blocks() const {
+    return report_blocks_;
+  }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static const size_t kRrBaseLength = 4;
+
+  std::vector<ReportBlock> report_blocks_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
new file mode 100644
index 0000000000..23ea49622b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+
+#include <utility>
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+using ::testing::make_tuple;
+using webrtc::rtcp::ReceiverReport;
+using webrtc::rtcp::ReportBlock;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kFractionLost = 55;
+const int32_t kCumulativeLost = 0x111213;
+const uint32_t kExtHighestSeqNum = 0x22232425;
+const uint32_t kJitter = 0x33343536;
+const uint32_t kLastSr = 0x44454647;
+const uint32_t kDelayLastSr = 0x55565758;
+// Manually created ReceiverReport with one ReportBlock matching constants
+// above.
+// Having this block allows to test Create and Parse separately.
+const uint8_t kPacket[] = {0x81, 201,  0x00, 0x07, 0x12, 0x34, 0x56, 0x78,
+                           0x23, 0x45, 0x67, 0x89, 55,   0x11, 0x12, 0x13,
+                           0x22, 0x23, 0x24, 0x25, 0x33, 0x34, 0x35, 0x36,
+                           0x44, 0x45, 0x46, 0x47, 0x55, 0x56, 0x57, 0x58};
+}  // namespace
+
+TEST(RtcpPacketReceiverReportTest, ParseWithOneReportBlock) {
+  ReceiverReport rr;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &rr));
+  const ReceiverReport& parsed = rr;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(1u, parsed.report_blocks().size());
+  const ReportBlock& rb = parsed.report_blocks().front();
+  EXPECT_EQ(kRemoteSsrc, rb.source_ssrc());
+  EXPECT_EQ(kFractionLost, rb.fraction_lost());
+  EXPECT_EQ(kCumulativeLost, rb.cumulative_lost_signed());
+  EXPECT_EQ(kExtHighestSeqNum, rb.extended_high_seq_num());
+  EXPECT_EQ(kJitter, rb.jitter());
+  EXPECT_EQ(kLastSr, rb.last_sr());
+  EXPECT_EQ(kDelayLastSr, rb.delay_since_last_sr());
+}
+
+TEST(RtcpPacketReceiverReportTest, ParseFailsOnIncorrectSize) {
+  rtc::Buffer damaged_packet(kPacket);
+  damaged_packet[0]++;  // Damage the packet: increase count field.
+  ReceiverReport rr;
+  EXPECT_FALSE(test::ParseSinglePacket(damaged_packet, &rr));
+}
+
+TEST(RtcpPacketReceiverReportTest, CreateWithOneReportBlock) {
+  ReceiverReport rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+  ReportBlock rb;
+  rb.SetMediaSsrc(kRemoteSsrc);
+  rb.SetFractionLost(kFractionLost);
+  rb.SetCumulativeLost(kCumulativeLost);
+  rb.SetExtHighestSeqNum(kExtHighestSeqNum);
+  rb.SetJitter(kJitter);
+  rb.SetLastSr(kLastSr);
+  rb.SetDelayLastSr(kDelayLastSr);
+  rr.AddReportBlock(rb);
+
+  rtc::Buffer raw = rr.Build();
+
+  EXPECT_THAT(make_tuple(raw.data(), raw.size()), ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketReceiverReportTest, CreateAndParseWithoutReportBlocks) {
+  ReceiverReport rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+
+  rtc::Buffer raw = rr.Build();
+  ReceiverReport parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_THAT(parsed.report_blocks(), IsEmpty());
+}
+
+TEST(RtcpPacketReceiverReportTest, CreateAndParseWithTwoReportBlocks) {
+  ReceiverReport rr;
+  ReportBlock rb1;
+  rb1.SetMediaSsrc(kRemoteSsrc);
+  ReportBlock rb2;
+  rb2.SetMediaSsrc(kRemoteSsrc + 1);
+
+  rr.SetSenderSsrc(kSenderSsrc);
+  EXPECT_TRUE(rr.AddReportBlock(rb1));
+  EXPECT_TRUE(rr.AddReportBlock(rb2));
+
+  rtc::Buffer raw = rr.Build();
+  ReceiverReport parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(2u, parsed.report_blocks().size());
+  EXPECT_EQ(kRemoteSsrc, parsed.report_blocks()[0].source_ssrc());
+  EXPECT_EQ(kRemoteSsrc + 1, parsed.report_blocks()[1].source_ssrc());
+}
+
+TEST(RtcpPacketReceiverReportTest, CreateWithTooManyReportBlocks) {
+  ReceiverReport rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+  ReportBlock rb;
+  for (size_t i = 0; i < ReceiverReport::kMaxNumberOfReportBlocks; ++i) {
+    rb.SetMediaSsrc(kRemoteSsrc + i);
+    EXPECT_TRUE(rr.AddReportBlock(rb));
+  }
+  rb.SetMediaSsrc(kRemoteSsrc + ReceiverReport::kMaxNumberOfReportBlocks);
+  EXPECT_FALSE(rr.AddReportBlock(rb));
+}
+
+TEST(RtcpPacketReceiverReportTest, SetReportBlocksOverwritesOldBlocks) {
+  ReceiverReport rr;
+  ReportBlock report_block;
+  // Use jitter field of the report blocks to distinguish them.
+  report_block.SetJitter(1001u);
+  rr.AddReportBlock(report_block);
+  ASSERT_EQ(rr.report_blocks().size(), 1u);
+  ASSERT_EQ(rr.report_blocks()[0].jitter(), 1001u);
+
+  std::vector<ReportBlock> blocks(3u);
+  blocks[0].SetJitter(2001u);
+  blocks[1].SetJitter(3001u);
+  blocks[2].SetJitter(4001u);
+  EXPECT_TRUE(rr.SetReportBlocks(blocks));
+  ASSERT_EQ(rr.report_blocks().size(), 3u);
+  EXPECT_EQ(rr.report_blocks()[0].jitter(), 2001u);
+  EXPECT_EQ(rr.report_blocks()[1].jitter(), 3001u);
+  EXPECT_EQ(rr.report_blocks()[2].jitter(), 4001u);
+}
+
+TEST(RtcpPacketReceiverReportTest, SetReportBlocksMaxLimit) {
+  ReceiverReport rr;
+  std::vector<ReportBlock> max_blocks(ReceiverReport::kMaxNumberOfReportBlocks);
+  EXPECT_TRUE(rr.SetReportBlocks(std::move(max_blocks)));
+
+  std::vector<ReportBlock> one_too_many_blocks(
+      ReceiverReport::kMaxNumberOfReportBlocks + 1);
+  EXPECT_FALSE(rr.SetReportBlocks(std::move(one_too_many_blocks)));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
new file mode 100644
index 0000000000..1389ca7836
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
@@ -0,0 +1,143 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+
+#include <cstdint>
+#include <utility>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+// Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P| FMT=15  |   PT=206      |             length            |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  0 |                  SSRC of packet sender                        |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                       Unused = 0                              |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |  Unique identifier 'R' 'E' 'M' 'B'                            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |  Num SSRC     | BR Exp    |  BR Mantissa                      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |   SSRC feedback                                               |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    :  ...                                                          :
+
+Remb::Remb() : bitrate_bps_(0) {}
+
+Remb::Remb(const Remb& rhs) = default;
+
+Remb::~Remb() = default;
+
+bool Remb::Parse(const CommonHeader& packet) {
+  RTC_DCHECK(packet.type() == kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), Psfb::kAfbMessageType);
+
+  if (packet.payload_size_bytes() < 16) {
+    RTC_LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
+                        << " is too small for Remb packet.";
+    return false;
+  }
+  const uint8_t* const payload = packet.payload();
+  if (kUniqueIdentifier != ByteReader<uint32_t>::ReadBigEndian(&payload[8])) {
+    return false;
+  }
+  uint8_t number_of_ssrcs = payload[12];
+  if (packet.payload_size_bytes() !=
+      kCommonFeedbackLength + (2 + number_of_ssrcs) * 4) {
+    RTC_LOG(LS_WARNING) << "Payload size " << packet.payload_size_bytes()
+                        << " does not match " << number_of_ssrcs << " ssrcs.";
+    return false;
+  }
+
+  ParseCommonFeedback(payload);
+  uint8_t exponent = payload[13] >> 2;
+  uint64_t mantissa = (static_cast<uint32_t>(payload[13] & 0x03) << 16) |
+                      ByteReader<uint16_t>::ReadBigEndian(&payload[14]);
+  bitrate_bps_ = (mantissa << exponent);
+  bool shift_overflow =
+      (static_cast<uint64_t>(bitrate_bps_) >> exponent) != mantissa;
+  if (bitrate_bps_ < 0 || shift_overflow) {
+    RTC_LOG(LS_ERROR) << "Invalid remb bitrate value : " << mantissa << "*2^"
+                      << static_cast<int>(exponent);
+    return false;
+  }
+
+  const uint8_t* next_ssrc = payload + 16;
+  ssrcs_.clear();
+  ssrcs_.reserve(number_of_ssrcs);
+  for (uint8_t i = 0; i < number_of_ssrcs; ++i) {
+    ssrcs_.push_back(ByteReader<uint32_t>::ReadBigEndian(next_ssrc));
+    next_ssrc += sizeof(uint32_t);
+  }
+
+  return true;
+}
+
+bool Remb::SetSsrcs(std::vector<uint32_t> ssrcs) {
+  if (ssrcs.size() > kMaxNumberOfSsrcs) {
+    RTC_LOG(LS_WARNING) << "Not enough space for all given SSRCs.";
+    return false;
+  }
+  ssrcs_ = std::move(ssrcs);
+  return true;
+}
+
+size_t Remb::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength + (2 + ssrcs_.size()) * 4;
+}
+
+bool Remb::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  size_t index_end = *index + BlockLength();
+  CreateHeader(Psfb::kAfbMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  RTC_DCHECK_EQ(0, Psfb::media_ssrc());
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+
+  ByteWriter<uint32_t>::WriteBigEndian(packet + *index, kUniqueIdentifier);
+  *index += sizeof(uint32_t);
+  const uint32_t kMaxMantissa = 0x3ffff;  // 18 bits.
+  uint64_t mantissa = bitrate_bps_;
+  uint8_t exponenta = 0;
+  while (mantissa > kMaxMantissa) {
+    mantissa >>= 1;
+    ++exponenta;
+  }
+  packet[(*index)++] = static_cast<uint8_t>(ssrcs_.size());
+  packet[(*index)++] = (exponenta << 2) | (mantissa >> 16);
+  ByteWriter<uint16_t>::WriteBigEndian(packet + *index, mantissa & 0xffff);
+  *index += sizeof(uint16_t);
+
+  for (uint32_t ssrc : ssrcs_) {
+    ByteWriter<uint32_t>::WriteBigEndian(packet + *index, ssrc);
+    *index += sizeof(uint32_t);
+  }
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h
new file mode 100644
index 0000000000..b7075c0f23
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMB_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMB_H_
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+// Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb).
+class Remb : public Psfb {
+ public:
+  static constexpr size_t kMaxNumberOfSsrcs = 0xff;
+
+  Remb();
+  Remb(const Remb&);
+  ~Remb() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  bool SetSsrcs(std::vector<uint32_t> ssrcs);
+  void SetBitrateBps(int64_t bitrate_bps) { bitrate_bps_ = bitrate_bps; }
+
+  int64_t bitrate_bps() const { return bitrate_bps_; }
+  const std::vector<uint32_t>& ssrcs() const { return ssrcs_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static constexpr uint32_t kUniqueIdentifier = 0x52454D42;  // 'R' 'E' 'M' 'B'.
+
+  // Media ssrc is unused, shadow base class setter and getter.
+  void SetMediaSsrc(uint32_t);
+  uint32_t media_ssrc() const;
+
+  int64_t bitrate_bps_;
+  std::vector<uint32_t> ssrcs_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMB_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb_unittest.cc
new file mode 100644
index 0000000000..c439d9c5f6
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remb_unittest.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+using ::testing::make_tuple;
+using webrtc::rtcp::Remb;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrcs[] = {0x23456789, 0x2345678a, 0x2345678b};
+const uint32_t kBitrateBps = 0x3fb93 * 2;  // 522022;
+const int64_t kBitrateBps64bit = int64_t{0x3fb93} << 30;
+const uint8_t kPacket[] = {0x8f, 206,  0x00, 0x07, 0x12, 0x34, 0x56, 0x78,
+                           0x00, 0x00, 0x00, 0x00, 'R',  'E',  'M',  'B',
+                           0x03, 0x07, 0xfb, 0x93, 0x23, 0x45, 0x67, 0x89,
+                           0x23, 0x45, 0x67, 0x8a, 0x23, 0x45, 0x67, 0x8b};
+const size_t kPacketLength = sizeof(kPacket);
+}  // namespace
+
+TEST(RtcpPacketRembTest, Create) {
+  Remb remb;
+  remb.SetSenderSsrc(kSenderSsrc);
+  remb.SetSsrcs(
+      std::vector<uint32_t>(std::begin(kRemoteSsrcs), std::end(kRemoteSsrcs)));
+  remb.SetBitrateBps(kBitrateBps);
+
+  rtc::Buffer packet = remb.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketRembTest, Parse) {
+  Remb remb;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &remb));
+  const Remb& parsed = remb;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kBitrateBps, parsed.bitrate_bps());
+  EXPECT_THAT(parsed.ssrcs(), ElementsAreArray(kRemoteSsrcs));
+}
+
+TEST(RtcpPacketRembTest, CreateAndParseWithoutSsrcs) {
+  Remb remb;
+  remb.SetSenderSsrc(kSenderSsrc);
+  remb.SetBitrateBps(kBitrateBps);
+  rtc::Buffer packet = remb.Build();
+
+  Remb parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kBitrateBps, parsed.bitrate_bps());
+  EXPECT_THAT(parsed.ssrcs(), IsEmpty());
+}
+
+TEST(RtcpPacketRembTest, CreateAndParse64bitBitrate) {
+  Remb remb;
+  remb.SetBitrateBps(kBitrateBps64bit);
+  rtc::Buffer packet = remb.Build();
+
+  Remb parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+  EXPECT_EQ(kBitrateBps64bit, parsed.bitrate_bps());
+}
+
+TEST(RtcpPacketRembTest, ParseFailsOnTooSmallPacketToBeRemb) {
+  // Make it too small.
+  constexpr size_t kTooSmallSize = (1 + 3) * 4;
+  uint8_t packet[kTooSmallSize];
+  memcpy(packet, kPacket, kTooSmallSize);
+  packet[3] = 3;
+
+  Remb remb;
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &remb));
+}
+
+TEST(RtcpPacketRembTest, ParseFailsWhenUniqueIdentifierIsNotRemb) {
+  uint8_t packet[kPacketLength];
+  memcpy(packet, kPacket, kPacketLength);
+  packet[12] = 'N';  // Swap 'R' -> 'N' in the 'REMB' unique identifier.
+
+  Remb remb;
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &remb));
+}
+
+TEST(RtcpPacketRembTest, ParseFailsWhenBitrateDoNotFitIn64bits) {
+  uint8_t packet[kPacketLength];
+  memcpy(packet, kPacket, kPacketLength);
+  packet[17] |= 0xfc;  // Set exponent component to maximum of 63.
+  packet[19] |= 0x02;  // Ensure mantissa is at least 2.
+
+  Remb remb;
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &remb));
+}
+
+TEST(RtcpPacketRembTest, ParseFailsWhenBitrateDoNotFitIn63bits) {
+  uint8_t packet[kPacketLength];
+  memcpy(packet, kPacket, kPacketLength);
+  packet[17] = 56 << 2;  // Set exponent component to 56.
+  packet[18] = 0;        // Set mantissa to 200 > 128
+  packet[19] = 200;
+
+  // Result value 200 * 2^56 can't be represented with int64_t and thus should
+  // be rejected.
+  Remb remb;
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &remb));
+}
+
+TEST(RtcpPacketRembTest, ParseFailsWhenSsrcCountMismatchLength) {
+  uint8_t packet[kPacketLength];
+  memcpy(packet, kPacket, kPacketLength);
+  packet[16]++;  // Swap 3 -> 4 in the ssrcs count.
+
+  Remb remb;
+  EXPECT_FALSE(test::ParseSinglePacket(packet, &remb));
+}
+
+TEST(RtcpPacketRembTest, TooManySsrcs) {
+  Remb remb;
+  EXPECT_FALSE(remb.SetSsrcs(
+      std::vector<uint32_t>(Remb::kMaxNumberOfSsrcs + 1, kRemoteSsrcs[0])));
+  EXPECT_TRUE(remb.SetSsrcs(
+      std::vector<uint32_t>(Remb::kMaxNumberOfSsrcs, kRemoteSsrcs[0])));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc
new file mode 100644
index 0000000000..ca59791248
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
+
+#include <algorithm>
+#include <cmath>
+#include <type_traits>
+#include <utility>
+#include <vector>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+namespace {
+
+static constexpr int kFieldValueSize = 3;
+static constexpr int kFieldSize = 1 + kFieldValueSize;
+static constexpr DataRate kDataRateResolution = DataRate::KilobitsPerSec(1);
+constexpr int64_t kMaxEncoded = (1 << (kFieldValueSize * 8)) - 1;
+
+class DataRateSerializer {
+ public:
+  DataRateSerializer(
+      uint8_t id,
+      std::function<DataRate*(NetworkStateEstimate*)> field_getter)
+      : id_(id), field_getter_(field_getter) {}
+
+  uint8_t id() const { return id_; }
+
+  void Read(const uint8_t* src, NetworkStateEstimate* target) const {
+    int64_t scaled = ByteReader<uint32_t, kFieldValueSize>::ReadBigEndian(src);
+    if (scaled == kMaxEncoded) {
+      *field_getter_(target) = DataRate::PlusInfinity();
+    } else {
+      *field_getter_(target) = kDataRateResolution * scaled;
+    }
+  }
+
+  bool Write(const NetworkStateEstimate& src, uint8_t* target) const {
+    auto value = *field_getter_(const_cast<NetworkStateEstimate*>(&src));
+    if (value.IsMinusInfinity()) {
+      RTC_LOG(LS_WARNING) << "Trying to serialize MinusInfinity";
+      return false;
+    }
+    ByteWriter<uint8_t>::WriteBigEndian(target++, id_);
+    int64_t scaled;
+    if (value.IsPlusInfinity()) {
+      scaled = kMaxEncoded;
+    } else {
+      scaled = value / kDataRateResolution;
+      if (scaled >= kMaxEncoded) {
+        scaled = kMaxEncoded;
+        RTC_LOG(LS_WARNING) << ToString(value) << " is larger than max ("
+                            << ToString(kMaxEncoded * kDataRateResolution)
+                            << "), encoded as PlusInfinity.";
+      }
+    }
+    ByteWriter<uint32_t, kFieldValueSize>::WriteBigEndian(target, scaled);
+    return true;
+  }
+
+ private:
+  const uint8_t id_;
+  const std::function<DataRate*(NetworkStateEstimate*)> field_getter_;
+};
+
+class RemoteEstimateSerializerImpl : public RemoteEstimateSerializer {
+ public:
+  explicit RemoteEstimateSerializerImpl(std::vector<DataRateSerializer> fields)
+      : fields_(fields) {}
+
+  rtc::Buffer Serialize(const NetworkStateEstimate& src) const override {
+    size_t max_size = fields_.size() * kFieldSize;
+    size_t size = 0;
+    rtc::Buffer buf(max_size);
+    for (const auto& field : fields_) {
+      if (field.Write(src, buf.data() + size)) {
+        size += kFieldSize;
+      }
+    }
+    buf.SetSize(size);
+    return buf;
+  }
+
+  bool Parse(rtc::ArrayView<const uint8_t> src,
+             NetworkStateEstimate* target) const override {
+    if (src.size() % kFieldSize != 0)
+      return false;
+    RTC_DCHECK_EQ(src.size() % kFieldSize, 0);
+    for (const uint8_t* data_ptr = src.data(); data_ptr < src.end();
+         data_ptr += kFieldSize) {
+      uint8_t field_id = ByteReader<uint8_t>::ReadBigEndian(data_ptr);
+      for (const auto& field : fields_) {
+        if (field.id() == field_id) {
+          field.Read(data_ptr + 1, target);
+          break;
+        }
+      }
+    }
+    return true;
+  }
+
+ private:
+  const std::vector<DataRateSerializer> fields_;
+};
+
+}  // namespace
+
+const RemoteEstimateSerializer* GetRemoteEstimateSerializer() {
+  using E = NetworkStateEstimate;
+  static auto* serializer = new RemoteEstimateSerializerImpl({
+      {1, [](E* e) { return &e->link_capacity_lower; }},
+      {2, [](E* e) { return &e->link_capacity_upper; }},
+  });
+  return serializer;
+}
+
+RemoteEstimate::RemoteEstimate() : serializer_(GetRemoteEstimateSerializer()) {
+  SetSubType(kSubType);
+  SetName(kName);
+  SetSenderSsrc(0);
+}
+
+RemoteEstimate::RemoteEstimate(App&& app)
+    : App(std::move(app)), serializer_(GetRemoteEstimateSerializer()) {}
+
+bool RemoteEstimate::ParseData() {
+  return serializer_->Parse({data(), data_size()}, &estimate_);
+}
+
+void RemoteEstimate::SetEstimate(NetworkStateEstimate estimate) {
+  estimate_ = estimate;
+  auto buf = serializer_->Serialize(estimate);
+  SetData(buf.data(), buf.size());
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h
new file mode 100644
index 0000000000..3400274568
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMOTE_ESTIMATE_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMOTE_ESTIMATE_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class CommonHeader;
+class RemoteEstimateSerializer {
+ public:
+  virtual bool Parse(rtc::ArrayView<const uint8_t> src,
+                     NetworkStateEstimate* target) const = 0;
+  virtual rtc::Buffer Serialize(const NetworkStateEstimate& src) const = 0;
+  virtual ~RemoteEstimateSerializer() = default;
+};
+
+// Using a static global implementation to avoid incurring initialization
+// overhead of the serializer every time RemoteEstimate is created.
+const RemoteEstimateSerializer* GetRemoteEstimateSerializer();
+
+// The RemoteEstimate packet provides network estimation results from the
+// receive side. This functionality is experimental and subject to change
+// without notice.
+class RemoteEstimate : public App {
+ public:
+  RemoteEstimate();
+  explicit RemoteEstimate(App&& app);
+  // Note, sub type must be unique among all app messages with "goog" name.
+  static constexpr uint8_t kSubType = 13;
+  static constexpr uint32_t kName = NameToInt("goog");
+  static TimeDelta GetTimestampPeriod();
+
+  bool ParseData();
+  void SetEstimate(NetworkStateEstimate estimate);
+  NetworkStateEstimate estimate() const { return estimate_; }
+
+ private:
+  NetworkStateEstimate estimate_;
+  const RemoteEstimateSerializer* const serializer_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMOTE_ESTIMATE_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate_unittest.cc
new file mode 100644
index 0000000000..bf0e0e2610
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/remote_estimate_unittest.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace rtcp {
+TEST(RemoteEstimateTest, EncodesCapacityBounds) {
+  NetworkStateEstimate src;
+  src.link_capacity_lower = DataRate::KilobitsPerSec(10);
+  src.link_capacity_upper = DataRate::KilobitsPerSec(1000000);
+  rtc::Buffer data = GetRemoteEstimateSerializer()->Serialize(src);
+  NetworkStateEstimate dst;
+  EXPECT_TRUE(GetRemoteEstimateSerializer()->Parse(data, &dst));
+  EXPECT_EQ(src.link_capacity_lower, dst.link_capacity_lower);
+  EXPECT_EQ(src.link_capacity_upper, dst.link_capacity_upper);
+}
+
+TEST(RemoteEstimateTest, ExpandsToPlusInfinity) {
+  NetworkStateEstimate src;
+  // White box testing: We know that the value is stored in an unsigned 24 int
+  // with kbps resolution. We expected it be represented as plus infinity.
+  src.link_capacity_lower = DataRate::KilobitsPerSec(2 << 24);
+  src.link_capacity_upper = DataRate::PlusInfinity();
+  rtc::Buffer data = GetRemoteEstimateSerializer()->Serialize(src);
+
+  NetworkStateEstimate dst;
+  EXPECT_TRUE(GetRemoteEstimateSerializer()->Parse(data, &dst));
+  EXPECT_TRUE(dst.link_capacity_lower.IsPlusInfinity());
+  EXPECT_TRUE(dst.link_capacity_upper.IsPlusInfinity());
+}
+
+TEST(RemoteEstimateTest, DoesNotEncodeNegative) {
+  NetworkStateEstimate src;
+  src.link_capacity_lower = DataRate::MinusInfinity();
+  src.link_capacity_upper = DataRate::MinusInfinity();
+  rtc::Buffer data = GetRemoteEstimateSerializer()->Serialize(src);
+  // Since MinusInfinity can't be represented, the buffer should be empty.
+  EXPECT_EQ(data.size(), 0u);
+  NetworkStateEstimate dst;
+  dst.link_capacity_lower = DataRate::KilobitsPerSec(300);
+  EXPECT_TRUE(GetRemoteEstimateSerializer()->Parse(data, &dst));
+  // The fields will be left unchanged by the parser as they were not encoded.
+  EXPECT_EQ(dst.link_capacity_lower, DataRate::KilobitsPerSec(300));
+  EXPECT_TRUE(dst.link_capacity_upper.IsMinusInfinity());
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
new file mode 100644
index 0000000000..d4579fc8d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
+//
+// RTCP report block (RFC 3550).
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  0 |                 SSRC_1 (SSRC of first source)                 |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 | fraction lost |       cumulative number of packets lost       |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |           extended highest sequence number received           |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |                      interarrival jitter                      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |                         last SR (LSR)                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 |                   delay since last SR (DLSR)                  |
+// 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+ReportBlock::ReportBlock()
+    : source_ssrc_(0),
+      fraction_lost_(0),
+      cumulative_lost_(0),
+      extended_high_seq_num_(0),
+      jitter_(0),
+      last_sr_(0),
+      delay_since_last_sr_(0) {}
+
+bool ReportBlock::Parse(const uint8_t* buffer, size_t length) {
+  RTC_DCHECK(buffer != nullptr);
+  if (length < ReportBlock::kLength) {
+    RTC_LOG(LS_ERROR) << "Report Block should be 24 bytes long";
+    return false;
+  }
+
+  source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]);
+  fraction_lost_ = buffer[4];
+  cumulative_lost_ = ByteReader<int32_t, 3>::ReadBigEndian(&buffer[5]);
+  extended_high_seq_num_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  jitter_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[12]);
+  last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[16]);
+  delay_since_last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[20]);
+
+  return true;
+}
+
+void ReportBlock::Create(uint8_t* buffer) const {
+  // Runtime check should be done while setting cumulative_lost.
+  RTC_DCHECK_LT(cumulative_lost_signed(),
+                (1 << 23));  // Have only 3 bytes for it.
+
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[0], source_ssrc());
+  ByteWriter<uint8_t>::WriteBigEndian(&buffer[4], fraction_lost());
+  ByteWriter<int32_t, 3>::WriteBigEndian(&buffer[5], cumulative_lost_signed());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], extended_high_seq_num());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[12], jitter());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[16], last_sr());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[20], delay_since_last_sr());
+}
+
+bool ReportBlock::SetCumulativeLost(int32_t cumulative_lost) {
+  // We have only 3 bytes to store it, and it's a signed value.
+  if (cumulative_lost >= (1 << 23) || cumulative_lost < -(1 << 23)) {
+    RTC_LOG(LS_WARNING)
+        << "Cumulative lost is too big to fit into Report Block";
+    return false;
+  }
+  cumulative_lost_ = cumulative_lost;
+  return true;
+}
+
+uint32_t ReportBlock::cumulative_lost() const {
+  if (cumulative_lost_ < 0) {
+    RTC_LOG(LS_VERBOSE) << "Ignoring negative value of cumulative_lost";
+    return 0;
+  }
+  return cumulative_lost_;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h
new file mode 100644
index 0000000000..eb16640ae2
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+namespace rtcp {
+
+// A ReportBlock represents the Sender Report packet from
+// RFC 3550 section 6.4.1.
+class ReportBlock {
+ public:
+  static const size_t kLength = 24;
+
+  ReportBlock();
+  ~ReportBlock() {}
+
+  bool Parse(const uint8_t* buffer, size_t length);
+
+  // Fills buffer with the ReportBlock.
+  // Consumes ReportBlock::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void SetMediaSsrc(uint32_t ssrc) { source_ssrc_ = ssrc; }
+  void SetFractionLost(uint8_t fraction_lost) {
+    fraction_lost_ = fraction_lost;
+  }
+  bool SetCumulativeLost(int32_t cumulative_lost);
+  void SetExtHighestSeqNum(uint32_t ext_highest_seq_num) {
+    extended_high_seq_num_ = ext_highest_seq_num;
+  }
+  void SetJitter(uint32_t jitter) { jitter_ = jitter; }
+  void SetLastSr(uint32_t last_sr) { last_sr_ = last_sr; }
+  void SetDelayLastSr(uint32_t delay_last_sr) {
+    delay_since_last_sr_ = delay_last_sr;
+  }
+
+  uint32_t source_ssrc() const { return source_ssrc_; }
+  uint8_t fraction_lost() const { return fraction_lost_; }
+  int32_t cumulative_lost_signed() const { return cumulative_lost_; }
+  // Deprecated - returns max(0, cumulative_lost_), not negative values.
+  uint32_t cumulative_lost() const;
+  uint32_t extended_high_seq_num() const { return extended_high_seq_num_; }
+  uint32_t jitter() const { return jitter_; }
+  uint32_t last_sr() const { return last_sr_; }
+  uint32_t delay_since_last_sr() const { return delay_since_last_sr_; }
+
+ private:
+  uint32_t source_ssrc_;     // 32 bits
+  uint8_t fraction_lost_;    // 8 bits representing a fixed point value 0..1
+  int32_t cumulative_lost_;  // Signed 24-bit value
+  uint32_t extended_high_seq_num_;  // 32 bits
+  uint32_t jitter_;                 // 32 bits
+  uint32_t last_sr_;                // 32 bits
+  uint32_t delay_since_last_sr_;    // 32 bits, units of 1/65536 seconds
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc
new file mode 100644
index 0000000000..5cc102fed0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+
+#include <limits>
+
+#include "rtc_base/random.h"
+#include "test/gtest.h"
+
+using webrtc::rtcp::ReportBlock;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint8_t kFractionLost = 55;
+// Use values that are streamed differently LE and BE.
+const int32_t kCumulativeLost = 0x111213;
+const uint32_t kExtHighestSeqNum = 0x22232425;
+const uint32_t kJitter = 0x33343536;
+const uint32_t kLastSr = 0x44454647;
+const uint32_t kDelayLastSr = 0x55565758;
+const size_t kBufferLength = ReportBlock::kLength;
+
+TEST(RtcpPacketReportBlockTest, ParseChecksLength) {
+  uint8_t buffer[kBufferLength];
+  memset(buffer, 0, sizeof(buffer));
+
+  ReportBlock rb;
+  EXPECT_FALSE(rb.Parse(buffer, kBufferLength - 1));
+  EXPECT_TRUE(rb.Parse(buffer, kBufferLength));
+}
+
+TEST(RtcpPacketReportBlockTest, ParseAnyData) {
+  uint8_t buffer[kBufferLength];
+  // Fill buffer with semi-random data.
+  Random generator(0x256F8A285EC829ull);
+  for (size_t i = 0; i < kBufferLength; ++i)
+    buffer[i] = static_cast<uint8_t>(generator.Rand(0, 0xff));
+
+  ReportBlock rb;
+  EXPECT_TRUE(rb.Parse(buffer, kBufferLength));
+}
+
+TEST(RtcpPacketReportBlockTest, ParseMatchCreate) {
+  ReportBlock rb;
+  rb.SetMediaSsrc(kRemoteSsrc);
+  rb.SetFractionLost(kFractionLost);
+  rb.SetCumulativeLost(kCumulativeLost);
+  rb.SetExtHighestSeqNum(kExtHighestSeqNum);
+  rb.SetJitter(kJitter);
+  rb.SetLastSr(kLastSr);
+  rb.SetDelayLastSr(kDelayLastSr);
+
+  uint8_t buffer[kBufferLength];
+  rb.Create(buffer);
+
+  ReportBlock parsed;
+  EXPECT_TRUE(parsed.Parse(buffer, kBufferLength));
+
+  EXPECT_EQ(kRemoteSsrc, parsed.source_ssrc());
+  EXPECT_EQ(kFractionLost, parsed.fraction_lost());
+  EXPECT_EQ(kCumulativeLost, parsed.cumulative_lost_signed());
+  EXPECT_EQ(kExtHighestSeqNum, parsed.extended_high_seq_num());
+  EXPECT_EQ(kJitter, parsed.jitter());
+  EXPECT_EQ(kLastSr, parsed.last_sr());
+  EXPECT_EQ(kDelayLastSr, parsed.delay_since_last_sr());
+}
+
+TEST(RtcpPacketReportBlockTest, ValidateCumulativeLost) {
+  // CumulativeLost is a signed 24-bit integer.
+  // However, existing code expects it to be an unsigned integer.
+  // The case of negative values should be unusual; we return 0
+  // when caller wants an unsigned integer.
+  const int32_t kMaxCumulativeLost = 0x7fffff;
+  const int32_t kMinCumulativeLost = -0x800000;
+  ReportBlock rb;
+  EXPECT_FALSE(rb.SetCumulativeLost(kMaxCumulativeLost + 1));
+  EXPECT_TRUE(rb.SetCumulativeLost(kMaxCumulativeLost));
+  EXPECT_FALSE(rb.SetCumulativeLost(kMinCumulativeLost - 1));
+  EXPECT_TRUE(rb.SetCumulativeLost(kMinCumulativeLost));
+  EXPECT_EQ(kMinCumulativeLost, rb.cumulative_lost_signed());
+  EXPECT_EQ(0u, rb.cumulative_lost());
+}
+
+TEST(RtcpPacketReportBlockTest, ParseNegativeCumulativeLost) {
+  // CumulativeLost is a signed 24-bit integer.
+  const int32_t kNegativeCumulativeLost = -123;
+  ReportBlock rb;
+  EXPECT_TRUE(rb.SetCumulativeLost(kNegativeCumulativeLost));
+
+  uint8_t buffer[kBufferLength];
+  rb.Create(buffer);
+
+  ReportBlock parsed;
+  EXPECT_TRUE(parsed.Parse(buffer, kBufferLength));
+
+  EXPECT_EQ(kNegativeCumulativeLost, parsed.cumulative_lost_signed());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc
new file mode 100644
index 0000000000..95fc890b19
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rtcp {
+// Receiver Reference Time Report Block (RFC 3611).
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     BT=4      |   reserved    |       block length = 2        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |              NTP timestamp, most significant word             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |             NTP timestamp, least significant word             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+void Rrtr::Parse(const uint8_t* buffer) {
+  RTC_DCHECK(buffer[0] == kBlockType);
+  // reserved = buffer[1];
+  RTC_DCHECK(ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) == kBlockLength);
+  uint32_t seconds = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  uint32_t fraction = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  ntp_.Set(seconds, fraction);
+}
+
+void Rrtr::Create(uint8_t* buffer) const {
+  const uint8_t kReserved = 0;
+  buffer[0] = kBlockType;
+  buffer[1] = kReserved;
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockLength);
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], ntp_.seconds());
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], ntp_.fractions());
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
new file mode 100644
index 0000000000..827bd74399
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace rtcp {
+
+class Rrtr {
+ public:
+  static const uint8_t kBlockType = 4;
+  static const uint16_t kBlockLength = 2;
+  static const size_t kLength = 4 * (kBlockLength + 1);  // 12
+
+  Rrtr() {}
+  Rrtr(const Rrtr&) = default;
+  ~Rrtr() {}
+
+  Rrtr& operator=(const Rrtr&) = default;
+
+  void Parse(const uint8_t* buffer);
+
+  // Fills buffer with the Rrtr.
+  // Consumes Rrtr::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void SetNtp(NtpTime ntp) { ntp_ = ntp; }
+
+  NtpTime ntp() const { return ntp_; }
+
+ private:
+  NtpTime ntp_;
+};
+
+inline bool operator==(const Rrtr& rrtr1, const Rrtr& rrtr2) {
+  return rrtr1.ntp() == rrtr2.ntp();
+}
+
+inline bool operator!=(const Rrtr& rrtr1, const Rrtr& rrtr2) {
+  return !(rrtr1 == rrtr2);
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc
new file mode 100644
index 0000000000..56622ea81a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+
+#include "test/gtest.h"
+
+using webrtc::rtcp::Rrtr;
+
+namespace webrtc {
+namespace {
+
+const uint32_t kNtpSec = 0x12345678;
+const uint32_t kNtpFrac = 0x23456789;
+const uint8_t kBlock[] = {0x04, 0x00, 0x00, 0x02, 0x12, 0x34,
+                          0x56, 0x78, 0x23, 0x45, 0x67, 0x89};
+const size_t kBlockSizeBytes = sizeof(kBlock);
+static_assert(
+    kBlockSizeBytes == Rrtr::kLength,
+    "Size of manually created Rrtr block should match class constant");
+
+TEST(RtcpPacketRrtrTest, Create) {
+  uint8_t buffer[Rrtr::kLength];
+  Rrtr rrtr;
+  rrtr.SetNtp(NtpTime(kNtpSec, kNtpFrac));
+
+  rrtr.Create(buffer);
+  EXPECT_EQ(0, memcmp(buffer, kBlock, kBlockSizeBytes));
+}
+
+TEST(RtcpPacketRrtrTest, Parse) {
+  Rrtr read_rrtr;
+  read_rrtr.Parse(kBlock);
+
+  // Run checks on const object to ensure all accessors have const modifier.
+  const Rrtr& parsed = read_rrtr;
+
+  EXPECT_EQ(kNtpSec, parsed.ntp().seconds());
+  EXPECT_EQ(kNtpFrac, parsed.ntp().fractions());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc
new file mode 100644
index 0000000000..18097de330
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Rtpfb::kPacketType;
+// RFC 4585, Section 6.1: Feedback format.
+//
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 |                  SSRC of media source                         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+
+void Rtpfb::ParseCommonFeedback(const uint8_t* payload) {
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(&payload[0]));
+  SetMediaSsrc(ByteReader<uint32_t>::ReadBigEndian(&payload[4]));
+}
+
+void Rtpfb::CreateCommonFeedback(uint8_t* payload) const {
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[0], sender_ssrc());
+  ByteWriter<uint32_t>::WriteBigEndian(&payload[4], media_ssrc());
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
new file mode 100644
index 0000000000..973b429a2d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// RTPFB: Transport layer feedback message.
+// RFC4585, Section 6.2
+class Rtpfb : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 205;
+
+  Rtpfb() = default;
+  ~Rtpfb() override = default;
+
+  void SetMediaSsrc(uint32_t ssrc) { media_ssrc_ = ssrc; }
+
+  uint32_t media_ssrc() const { return media_ssrc_; }
+
+ protected:
+  static constexpr size_t kCommonFeedbackLength = 8;
+  void ParseCommonFeedback(const uint8_t* payload);
+  void CreateCommonFeedback(uint8_t* payload) const;
+
+ private:
+  uint32_t media_ssrc_ = 0;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.cc
new file mode 100644
index 0000000000..f244ec5f37
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.cc
@@ -0,0 +1,199 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+
+#include <string.h>
+
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Sdes::kPacketType;
+constexpr size_t Sdes::kMaxNumberOfChunks;
+// Source Description (SDES) (RFC 3550).
+//
+//         0                   1                   2                   3
+//         0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//        +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// header |V=2|P|    SC   |  PT=SDES=202  |             length            |
+//        +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// chunk  |                          SSRC/CSRC_1                          |
+//   1    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//        |                           SDES items                          |
+//        |                              ...                              |
+//        +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// chunk  |                          SSRC/CSRC_2                          |
+//   2    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//        |                           SDES items                          |
+//        |                              ...                              |
+//        +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//
+// Canonical End-Point Identifier SDES Item (CNAME)
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |    CNAME=1    |     length    | user and domain name        ...
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+namespace {
+const uint8_t kTerminatorTag = 0;
+const uint8_t kCnameTag = 1;
+
+size_t ChunkSize(const Sdes::Chunk& chunk) {
+  // Chunk:
+  // SSRC/CSRC (4 bytes) | CNAME=1 (1 byte) | length (1 byte) | cname | padding.
+  size_t chunk_payload_size = 4 + 1 + 1 + chunk.cname.size();
+  size_t padding_size = 4 - (chunk_payload_size % 4);  // Minimum 1.
+  return chunk_payload_size + padding_size;
+}
+}  // namespace
+
+Sdes::Sdes() : block_length_(RtcpPacket::kHeaderLength) {}
+
+Sdes::~Sdes() {}
+
+bool Sdes::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+
+  uint8_t number_of_chunks = packet.count();
+  std::vector<Chunk> chunks;  // Read chunk into temporary array, so that in
+                              // case of an error original array would stay
+                              // unchanged.
+  size_t block_length = kHeaderLength;
+
+  if (packet.payload_size_bytes() % 4 != 0) {
+    RTC_LOG(LS_WARNING) << "Invalid payload size "
+                        << packet.payload_size_bytes()
+                        << " bytes for a valid Sdes packet. Size should be"
+                           " multiple of 4 bytes";
+  }
+  const uint8_t* const payload_end =
+      packet.payload() + packet.payload_size_bytes();
+  const uint8_t* looking_at = packet.payload();
+  chunks.resize(number_of_chunks);
+  for (size_t i = 0; i < number_of_chunks;) {
+    // Each chunk consumes at least 8 bytes.
+    if (payload_end - looking_at < 8) {
+      RTC_LOG(LS_WARNING) << "Not enough space left for chunk #" << (i + 1);
+      return false;
+    }
+    chunks[i].ssrc = ByteReader<uint32_t>::ReadBigEndian(looking_at);
+    looking_at += sizeof(uint32_t);
+    bool cname_found = false;
+
+    uint8_t item_type;
+    while ((item_type = *(looking_at++)) != kTerminatorTag) {
+      if (looking_at >= payload_end) {
+        RTC_LOG(LS_WARNING)
+            << "Unexpected end of packet while reading chunk #" << (i + 1)
+            << ". Expected to find size of the text.";
+        return false;
+      }
+      uint8_t item_length = *(looking_at++);
+      const size_t kTerminatorSize = 1;
+      if (looking_at + item_length + kTerminatorSize > payload_end) {
+        RTC_LOG(LS_WARNING)
+            << "Unexpected end of packet while reading chunk #" << (i + 1)
+            << ". Expected to find text of size " << item_length;
+        return false;
+      }
+      if (item_type == kCnameTag) {
+        if (cname_found) {
+          RTC_LOG(LS_WARNING)
+              << "Found extra CNAME for same ssrc in chunk #" << (i + 1);
+          return false;
+        }
+        cname_found = true;
+        chunks[i].cname.assign(reinterpret_cast<const char*>(looking_at),
+                               item_length);
+      }
+      looking_at += item_length;
+    }
+    if (cname_found) {
+      // block_length calculates length of the packet that would be generated by
+      // Build/Create functions. Adjust it same way WithCName function does.
+      block_length += ChunkSize(chunks[i]);
+      ++i;
+    } else {
+      // RFC states CNAME item is mandatory.
+      // But same time it allows chunk without items.
+      // So while parsing, ignore all chunks without cname,
+      // but do not fail the parse.
+      RTC_LOG(LS_WARNING) << "CNAME not found for ssrc " << chunks[i].ssrc;
+      --number_of_chunks;
+      chunks.resize(number_of_chunks);
+    }
+    // Adjust to 32bit boundary.
+    looking_at += (payload_end - looking_at) % 4;
+  }
+
+  chunks_ = std::move(chunks);
+  block_length_ = block_length;
+  return true;
+}
+
+bool Sdes::AddCName(uint32_t ssrc, absl::string_view cname) {
+  RTC_DCHECK_LE(cname.length(), 0xffu);
+  if (chunks_.size() >= kMaxNumberOfChunks) {
+    RTC_LOG(LS_WARNING) << "Max SDES chunks reached.";
+    return false;
+  }
+  Chunk chunk;
+  chunk.ssrc = ssrc;
+  chunk.cname = std::string(cname);
+  chunks_.push_back(chunk);
+  block_length_ += ChunkSize(chunk);
+  return true;
+}
+
+size_t Sdes::BlockLength() const {
+  return block_length_;
+}
+
+bool Sdes::Create(uint8_t* packet,
+                  size_t* index,
+                  size_t max_length,
+                  PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+  CreateHeader(chunks_.size(), kPacketType, HeaderLength(), packet, index);
+
+  for (const Sdes::Chunk& chunk : chunks_) {
+    ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], chunk.ssrc);
+    ByteWriter<uint8_t>::WriteBigEndian(&packet[*index + 4], kCnameTag);
+    ByteWriter<uint8_t>::WriteBigEndian(
+        &packet[*index + 5], static_cast<uint8_t>(chunk.cname.size()));
+    memcpy(&packet[*index + 6], chunk.cname.data(), chunk.cname.size());
+    *index += (6 + chunk.cname.size());
+
+    // In each chunk, the list of items must be terminated by one or more null
+    // octets. The next chunk must start on a 32-bit boundary.
+    // CNAME (1 byte) | length (1 byte) | name | padding.
+    size_t padding_size = 4 - ((6 + chunk.cname.size()) % 4);
+    const int kPadding = 0;
+    memset(packet + *index, kPadding, padding_size);
+    *index += padding_size;
+  }
+
+  RTC_CHECK_EQ(*index, index_end);
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h
new file mode 100644
index 0000000000..36b63ba29f
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SDES_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SDES_H_
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+// Source Description (SDES) (RFC 3550).
+class Sdes : public RtcpPacket {
+ public:
+  struct Chunk {
+    uint32_t ssrc;
+    std::string cname;
+  };
+  static constexpr uint8_t kPacketType = 202;
+  static constexpr size_t kMaxNumberOfChunks = 0x1f;
+
+  Sdes();
+  ~Sdes() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  bool AddCName(uint32_t ssrc, absl::string_view cname);
+
+  const std::vector<Chunk>& chunks() const { return chunks_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  std::vector<Chunk> chunks_;
+  size_t block_length_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SDES_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes_unittest.cc
new file mode 100644
index 0000000000..15a39efe87
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sdes_unittest.cc
@@ -0,0 +1,244 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using webrtc::rtcp::Sdes;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint8_t kPadding = 0;
+const uint8_t kTerminatorTag = 0;
+const uint8_t kCnameTag = 1;
+const uint8_t kNameTag = 2;
+const uint8_t kEmailTag = 3;
+}  // namespace
+
+TEST(RtcpPacketSdesTest, CreateAndParseWithoutChunks) {
+  Sdes sdes;
+
+  rtc::Buffer packet = sdes.Build();
+  Sdes parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+
+  EXPECT_EQ(0u, parsed.chunks().size());
+}
+
+TEST(RtcpPacketSdesTest, CreateAndParseWithOneChunk) {
+  const std::string kCname = "alice@host";
+
+  Sdes sdes;
+  EXPECT_TRUE(sdes.AddCName(kSenderSsrc, kCname));
+
+  rtc::Buffer packet = sdes.Build();
+  Sdes sdes_parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &sdes_parsed));
+  const Sdes& parsed = sdes_parsed;  // Ensure accessors are const.
+
+  EXPECT_EQ(1u, parsed.chunks().size());
+  EXPECT_EQ(kSenderSsrc, parsed.chunks()[0].ssrc);
+  EXPECT_EQ(kCname, parsed.chunks()[0].cname);
+}
+
+TEST(RtcpPacketSdesTest, CreateAndParseWithMultipleChunks) {
+  Sdes sdes;
+  EXPECT_TRUE(sdes.AddCName(kSenderSsrc + 0, "a"));
+  EXPECT_TRUE(sdes.AddCName(kSenderSsrc + 1, "ab"));
+  EXPECT_TRUE(sdes.AddCName(kSenderSsrc + 2, "abc"));
+  EXPECT_TRUE(sdes.AddCName(kSenderSsrc + 3, "abcd"));
+  EXPECT_TRUE(sdes.AddCName(kSenderSsrc + 4, "abcde"));
+  EXPECT_TRUE(sdes.AddCName(kSenderSsrc + 5, "abcdef"));
+
+  rtc::Buffer packet = sdes.Build();
+  Sdes parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+
+  EXPECT_EQ(6u, parsed.chunks().size());
+  EXPECT_EQ(kSenderSsrc + 5, parsed.chunks()[5].ssrc);
+  EXPECT_EQ("abcdef", parsed.chunks()[5].cname);
+}
+
+TEST(RtcpPacketSdesTest, CreateWithTooManyChunks) {
+  const size_t kMaxChunks = (1 << 5) - 1;
+  Sdes sdes;
+  for (size_t i = 0; i < kMaxChunks; ++i) {
+    uint32_t ssrc = kSenderSsrc + i;
+    rtc::StringBuilder oss;
+    oss << "cname" << i;
+    EXPECT_TRUE(sdes.AddCName(ssrc, oss.str()));
+  }
+  EXPECT_FALSE(sdes.AddCName(kSenderSsrc + kMaxChunks, "foo"));
+}
+
+TEST(RtcpPacketSdesTest, CreateAndParseCnameItemWithEmptyString) {
+  Sdes sdes;
+  EXPECT_TRUE(sdes.AddCName(kSenderSsrc, ""));
+
+  rtc::Buffer packet = sdes.Build();
+  Sdes parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+
+  EXPECT_EQ(1u, parsed.chunks().size());
+  EXPECT_EQ(kSenderSsrc, parsed.chunks()[0].ssrc);
+  EXPECT_EQ("", parsed.chunks()[0].cname);
+}
+
+TEST(RtcpPacketSdesTest, ParseSkipsNonCNameField) {
+  const uint8_t kName[] = "abc";
+  const uint8_t kCname[] = "de";
+  const uint8_t kValidPacket[] = {
+      0x81, 202,       0x00,      0x04,           0x12,     0x34,     0x56,
+      0x78, kNameTag,  3,         kName[0],       kName[1], kName[2], kCnameTag,
+      2,    kCname[0], kCname[1], kTerminatorTag, kPadding, kPadding};
+  // Sanity checks packet was assembled correctly.
+  ASSERT_EQ(0u, sizeof(kValidPacket) % 4);
+  ASSERT_EQ(kValidPacket[3] + 1u, sizeof(kValidPacket) / 4);
+
+  Sdes parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kValidPacket, &parsed));
+
+  EXPECT_EQ(1u, parsed.chunks().size());
+  EXPECT_EQ(kSenderSsrc, parsed.chunks()[0].ssrc);
+  EXPECT_EQ("de", parsed.chunks()[0].cname);
+}
+
+TEST(RtcpPacketSdesTest, ParseSkipsChunksWithoutCName) {
+  const uint8_t kName[] = "ab";
+  const uint8_t kEmail[] = "de";
+  const uint8_t kCname[] = "def";
+  const uint8_t kPacket[] = {
+      0x82,           202,      0x00,      0x07,      0x12,
+      0x34,           0x56,     0x78,  // 1st chunk.
+      kNameTag,       3,        kName[0],  kName[1],  kName[2],
+      kEmailTag,      2,        kEmail[0], kEmail[1], kTerminatorTag,
+      kPadding,       kPadding, 0x23,      0x45,      0x67,
+      0x89,  // 2nd chunk.
+      kCnameTag,      3,        kCname[0], kCname[1], kCname[2],
+      kTerminatorTag, kPadding, kPadding};
+  // Sanity checks packet was assembled correctly.
+  ASSERT_EQ(0u, sizeof(kPacket) % 4);
+  ASSERT_EQ(kPacket[3] + 1u, sizeof(kPacket) / 4);
+
+  Sdes parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &parsed));
+  ASSERT_EQ(1u, parsed.chunks().size());
+  EXPECT_EQ(0x23456789u, parsed.chunks()[0].ssrc);
+  EXPECT_EQ("def", parsed.chunks()[0].cname);
+}
+
+TEST(RtcpPacketSdesTest, ParseFailsWithoutChunkItemTerminator) {
+  const uint8_t kName[] = "abc";
+  const uint8_t kCname[] = "d";
+  // No place for next chunk item.
+  const uint8_t kInvalidPacket[] = {
+      0x81,     202, 0x00,     0x03,     0x12,     0x34,      0x56, 0x78,
+      kNameTag, 3,   kName[0], kName[1], kName[2], kCnameTag, 1,    kCname[0]};
+  // Sanity checks packet was assembled correctly.
+  ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
+  ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
+
+  Sdes parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kInvalidPacket, &parsed));
+}
+
+TEST(RtcpPacketSdesTest, ParseFailsWithDamagedChunkItem) {
+  const uint8_t kName[] = "ab";
+  const uint8_t kCname[] = "d";
+  // Next chunk item has non-terminator type, but not the size.
+  const uint8_t kInvalidPacket[] = {
+      0x81,     202, 0x00,     0x03,     0x12,      0x34, 0x56,      0x78,
+      kNameTag, 2,   kName[0], kName[1], kCnameTag, 1,    kCname[0], kEmailTag};
+  // Sanity checks packet was assembled correctly.
+  ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
+  ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
+
+  Sdes parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kInvalidPacket, &parsed));
+}
+
+TEST(RtcpPacketSdesTest, ParseFailsWithTooLongChunkItem) {
+  const uint8_t kName[] = "abc";
+  const uint8_t kCname[] = "d";
+  // Last chunk item has length that goes beyond the buffer end.
+  const uint8_t kInvalidPacket[] = {
+      0x81,     202, 0x00,     0x03,     0x12,     0x34,      0x56, 0x78,
+      kNameTag, 3,   kName[0], kName[1], kName[2], kCnameTag, 2,    kCname[0]};
+  // Sanity checks packet was assembled correctly.
+  ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
+  ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
+
+  Sdes parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kInvalidPacket, &parsed));
+}
+
+TEST(RtcpPacketSdesTest, ParseFailsWithTwoCNames) {
+  const uint8_t kCname1[] = "a";
+  const uint8_t kCname2[] = "de";
+  const uint8_t kInvalidPacket[] = {
+      0x81,       202,           0x00, 0x03,       0x12,      0x34, 0x56,
+      0x78,       kCnameTag,     1,    kCname1[0], kCnameTag, 2,    kCname2[0],
+      kCname2[1], kTerminatorTag};
+  // Sanity checks packet was assembled correctly.
+  ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
+  ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
+
+  Sdes parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kInvalidPacket, &parsed));
+}
+
+TEST(RtcpPacketSdesTest, ParseFailsWithTooLittleSpaceForNextChunk) {
+  const uint8_t kCname[] = "a";
+  const uint8_t kEmail[] = "de";
+  // Two chunks are promised in the header, but no place for the second chunk.
+  const uint8_t kInvalidPacket[] = {
+      0x82,           202,  0x00,      0x04,      0x12, 0x34,      0x56,
+      0x78,  // 1st chunk.
+      kCnameTag,      1,    kCname[0], kEmailTag, 2,    kEmail[0], kEmail[1],
+      kTerminatorTag, 0x23, 0x45,      0x67,      0x89};  // 2nd chunk.
+  // Sanity checks packet was assembled correctly.
+  ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
+  ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
+
+  Sdes parsed;
+  EXPECT_FALSE(test::ParseSinglePacket(kInvalidPacket, &parsed));
+}
+
+TEST(RtcpPacketSdesTest, ParsedSdesCanBeReusedForBuilding) {
+  Sdes source;
+  const std::string kAlice = "alice@host";
+  const std::string kBob = "bob@host";
+  source.AddCName(kSenderSsrc, kAlice);
+
+  rtc::Buffer packet1 = source.Build();
+  Sdes middle;
+  test::ParseSinglePacket(packet1, &middle);
+
+  EXPECT_EQ(source.BlockLength(), middle.BlockLength());
+
+  middle.AddCName(kSenderSsrc + 1, kBob);
+
+  rtc::Buffer packet2 = middle.Build();
+  Sdes destination;
+  test::ParseSinglePacket(packet2, &destination);
+
+  EXPECT_EQ(middle.BlockLength(), destination.BlockLength());
+
+  EXPECT_EQ(2u, destination.chunks().size());
+  EXPECT_EQ(kSenderSsrc, destination.chunks()[0].ssrc);
+  EXPECT_EQ(kAlice, destination.chunks()[0].cname);
+  EXPECT_EQ(kSenderSsrc + 1, destination.chunks()[1].ssrc);
+  EXPECT_EQ(kBob, destination.chunks()[1].cname);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc
new file mode 100644
index 0000000000..73738376c3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+
+#include <utility>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t SenderReport::kPacketType;
+constexpr size_t SenderReport::kMaxNumberOfReportBlocks;
+constexpr size_t SenderReport::kSenderBaseLength;
+//    Sender report (SR) (RFC 3550).
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P|    RC   |   PT=SR=200   |             length            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |                         SSRC of sender                        |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  4 |              NTP timestamp, most significant word             |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |             NTP timestamp, least significant word             |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |                         RTP timestamp                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |                     sender's packet count                     |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 |                      sender's octet count                     |
+// 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+
+SenderReport::SenderReport()
+    : rtp_timestamp_(0), sender_packet_count_(0), sender_octet_count_(0) {}
+
+SenderReport::SenderReport(const SenderReport&) = default;
+SenderReport::SenderReport(SenderReport&&) = default;
+SenderReport& SenderReport::operator=(const SenderReport&) = default;
+SenderReport& SenderReport::operator=(SenderReport&&) = default;
+SenderReport::~SenderReport() = default;
+
+bool SenderReport::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+
+  const uint8_t report_block_count = packet.count();
+  if (packet.payload_size_bytes() <
+      kSenderBaseLength + report_block_count * ReportBlock::kLength) {
+    RTC_LOG(LS_WARNING) << "Packet is too small to contain all the data.";
+    return false;
+  }
+  // Read SenderReport header.
+  const uint8_t* const payload = packet.payload();
+  SetSenderSsrc(ByteReader<uint32_t>::ReadBigEndian(&payload[0]));
+  uint32_t secs = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
+  uint32_t frac = ByteReader<uint32_t>::ReadBigEndian(&payload[8]);
+  ntp_.Set(secs, frac);
+  rtp_timestamp_ = ByteReader<uint32_t>::ReadBigEndian(&payload[12]);
+  sender_packet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[16]);
+  sender_octet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[20]);
+  report_blocks_.resize(report_block_count);
+  const uint8_t* next_block = payload + kSenderBaseLength;
+  for (ReportBlock& block : report_blocks_) {
+    bool block_parsed = block.Parse(next_block, ReportBlock::kLength);
+    RTC_DCHECK(block_parsed);
+    next_block += ReportBlock::kLength;
+  }
+  // Double check we didn't read beyond provided buffer.
+  RTC_DCHECK_LE(next_block - payload,
+                static_cast<ptrdiff_t>(packet.payload_size_bytes()));
+  return true;
+}
+
+size_t SenderReport::BlockLength() const {
+  return kHeaderLength + kSenderBaseLength +
+         report_blocks_.size() * ReportBlock::kLength;
+}
+
+bool SenderReport::Create(uint8_t* packet,
+                          size_t* index,
+                          size_t max_length,
+                          PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+
+  CreateHeader(report_blocks_.size(), kPacketType, HeaderLength(), packet,
+               index);
+  // Write SenderReport header.
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], sender_ssrc());
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], ntp_.seconds());
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 8], ntp_.fractions());
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 12], rtp_timestamp_);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 16],
+                                       sender_packet_count_);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 20],
+                                       sender_octet_count_);
+  *index += kSenderBaseLength;
+  // Write report blocks.
+  for (const ReportBlock& block : report_blocks_) {
+    block.Create(packet + *index);
+    *index += ReportBlock::kLength;
+  }
+  // Ensure bytes written match expected.
+  RTC_DCHECK_EQ(*index, index_end);
+  return true;
+}
+
+bool SenderReport::AddReportBlock(const ReportBlock& block) {
+  if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
+    RTC_LOG(LS_WARNING) << "Max report blocks reached.";
+    return false;
+  }
+  report_blocks_.push_back(block);
+  return true;
+}
+
+bool SenderReport::SetReportBlocks(std::vector<ReportBlock> blocks) {
+  if (blocks.size() > kMaxNumberOfReportBlocks) {
+    RTC_LOG(LS_WARNING) << "Too many report blocks (" << blocks.size()
+                        << ") for sender report.";
+    return false;
+  }
+  report_blocks_ = std::move(blocks);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h
new file mode 100644
index 0000000000..66ced31721
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SENDER_REPORT_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SENDER_REPORT_H_
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class SenderReport : public RtcpPacket {
+ public:
+  static constexpr uint8_t kPacketType = 200;
+  static constexpr size_t kMaxNumberOfReportBlocks = 0x1f;
+
+  SenderReport();
+  SenderReport(const SenderReport&);
+  SenderReport(SenderReport&&);
+  SenderReport& operator=(const SenderReport&);
+  SenderReport& operator=(SenderReport&&);
+  ~SenderReport() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void SetNtp(NtpTime ntp) { ntp_ = ntp; }
+  void SetRtpTimestamp(uint32_t rtp_timestamp) {
+    rtp_timestamp_ = rtp_timestamp;
+  }
+  void SetPacketCount(uint32_t packet_count) {
+    sender_packet_count_ = packet_count;
+  }
+  void SetOctetCount(uint32_t octet_count) {
+    sender_octet_count_ = octet_count;
+  }
+  bool AddReportBlock(const ReportBlock& block);
+  bool SetReportBlocks(std::vector<ReportBlock> blocks);
+  void ClearReportBlocks() { report_blocks_.clear(); }
+
+  NtpTime ntp() const { return ntp_; }
+  uint32_t rtp_timestamp() const { return rtp_timestamp_; }
+  uint32_t sender_packet_count() const { return sender_packet_count_; }
+  uint32_t sender_octet_count() const { return sender_octet_count_; }
+
+  const std::vector<ReportBlock>& report_blocks() const {
+    return report_blocks_;
+  }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  static constexpr size_t kSenderBaseLength = 24;
+
+  NtpTime ntp_;
+  uint32_t rtp_timestamp_;
+  uint32_t sender_packet_count_;
+  uint32_t sender_octet_count_;
+  std::vector<ReportBlock> report_blocks_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SENDER_REPORT_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report_unittest.cc
new file mode 100644
index 0000000000..37f268e6b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report_unittest.cc
@@ -0,0 +1,142 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+
+#include <utility>
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAreArray;
+using ::testing::make_tuple;
+using webrtc::rtcp::ReportBlock;
+using webrtc::rtcp::SenderReport;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+const NtpTime kNtp(0x11121418, 0x22242628);
+const uint32_t kRtpTimestamp = 0x33343536;
+const uint32_t kPacketCount = 0x44454647;
+const uint32_t kOctetCount = 0x55565758;
+const uint8_t kPacket[] = {0x80, 200,  0x00, 0x06, 0x12, 0x34, 0x56,
+                           0x78, 0x11, 0x12, 0x14, 0x18, 0x22, 0x24,
+                           0x26, 0x28, 0x33, 0x34, 0x35, 0x36, 0x44,
+                           0x45, 0x46, 0x47, 0x55, 0x56, 0x57, 0x58};
+}  // namespace
+
+TEST(RtcpPacketSenderReportTest, CreateWithoutReportBlocks) {
+  SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  sr.SetNtp(kNtp);
+  sr.SetRtpTimestamp(kRtpTimestamp);
+  sr.SetPacketCount(kPacketCount);
+  sr.SetOctetCount(kOctetCount);
+
+  rtc::Buffer raw = sr.Build();
+  EXPECT_THAT(make_tuple(raw.data(), raw.size()), ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketSenderReportTest, ParseWithoutReportBlocks) {
+  SenderReport parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(kNtp, parsed.ntp());
+  EXPECT_EQ(kRtpTimestamp, parsed.rtp_timestamp());
+  EXPECT_EQ(kPacketCount, parsed.sender_packet_count());
+  EXPECT_EQ(kOctetCount, parsed.sender_octet_count());
+  EXPECT_TRUE(parsed.report_blocks().empty());
+}
+
+TEST(RtcpPacketSenderReportTest, CreateAndParseWithOneReportBlock) {
+  ReportBlock rb;
+  rb.SetMediaSsrc(kRemoteSsrc);
+
+  SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  EXPECT_TRUE(sr.AddReportBlock(rb));
+
+  rtc::Buffer raw = sr.Build();
+  SenderReport parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(1u, parsed.report_blocks().size());
+  EXPECT_EQ(kRemoteSsrc, parsed.report_blocks()[0].source_ssrc());
+}
+
+TEST(RtcpPacketSenderReportTest, CreateAndParseWithTwoReportBlocks) {
+  ReportBlock rb1;
+  rb1.SetMediaSsrc(kRemoteSsrc);
+  ReportBlock rb2;
+  rb2.SetMediaSsrc(kRemoteSsrc + 1);
+
+  SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  EXPECT_TRUE(sr.AddReportBlock(rb1));
+  EXPECT_TRUE(sr.AddReportBlock(rb2));
+
+  rtc::Buffer raw = sr.Build();
+  SenderReport parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(2u, parsed.report_blocks().size());
+  EXPECT_EQ(kRemoteSsrc, parsed.report_blocks()[0].source_ssrc());
+  EXPECT_EQ(kRemoteSsrc + 1, parsed.report_blocks()[1].source_ssrc());
+}
+
+TEST(RtcpPacketSenderReportTest, CreateWithTooManyReportBlocks) {
+  SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  ReportBlock rb;
+  for (size_t i = 0; i < SenderReport::kMaxNumberOfReportBlocks; ++i) {
+    rb.SetMediaSsrc(kRemoteSsrc + i);
+    EXPECT_TRUE(sr.AddReportBlock(rb));
+  }
+  rb.SetMediaSsrc(kRemoteSsrc + SenderReport::kMaxNumberOfReportBlocks);
+  EXPECT_FALSE(sr.AddReportBlock(rb));
+}
+
+TEST(RtcpPacketSenderReportTest, SetReportBlocksOverwritesOldBlocks) {
+  SenderReport sr;
+  ReportBlock report_block;
+  // Use jitter field of the report blocks to distinguish them.
+  report_block.SetJitter(1001u);
+  sr.AddReportBlock(report_block);
+  ASSERT_EQ(sr.report_blocks().size(), 1u);
+  ASSERT_EQ(sr.report_blocks()[0].jitter(), 1001u);
+
+  std::vector<ReportBlock> blocks(3u);
+  blocks[0].SetJitter(2001u);
+  blocks[1].SetJitter(3001u);
+  blocks[2].SetJitter(4001u);
+  EXPECT_TRUE(sr.SetReportBlocks(blocks));
+  ASSERT_EQ(sr.report_blocks().size(), 3u);
+  EXPECT_EQ(sr.report_blocks()[0].jitter(), 2001u);
+  EXPECT_EQ(sr.report_blocks()[1].jitter(), 3001u);
+  EXPECT_EQ(sr.report_blocks()[2].jitter(), 4001u);
+}
+
+TEST(RtcpPacketSenderReportTest, SetReportBlocksMaxLimit) {
+  SenderReport sr;
+  std::vector<ReportBlock> max_blocks(SenderReport::kMaxNumberOfReportBlocks);
+  EXPECT_TRUE(sr.SetReportBlocks(std::move(max_blocks)));
+
+  std::vector<ReportBlock> one_too_many_blocks(
+      SenderReport::kMaxNumberOfReportBlocks + 1);
+  EXPECT_FALSE(sr.SetReportBlocks(std::move(one_too_many_blocks)));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.cc
new file mode 100644
index 0000000000..601b24fe94
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.cc
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr size_t kTargetBitrateHeaderSizeBytes = 4;
+constexpr uint8_t TargetBitrate::kBlockType;
+const size_t TargetBitrate::kBitrateItemSizeBytes = 4;
+
+TargetBitrate::BitrateItem::BitrateItem()
+    : spatial_layer(0), temporal_layer(0), target_bitrate_kbps(0) {}
+
+TargetBitrate::BitrateItem::BitrateItem(uint8_t spatial_layer,
+                                        uint8_t temporal_layer,
+                                        uint32_t target_bitrate_kbps)
+    : spatial_layer(spatial_layer),
+      temporal_layer(temporal_layer),
+      target_bitrate_kbps(target_bitrate_kbps) {}
+
+//  RFC 4585: Feedback format.
+//
+//  Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |     BT=42     |   reserved    |         block length          |
+//  +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//
+//  Target bitrate item (repeat as many times as necessary).
+//
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |   S   |   T   |                Target Bitrate                 |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :  ...                                                          :
+//
+//  Spatial Layer (S): 4 bits
+//    Indicates which temporal layer this bitrate concerns.
+//
+//  Temporal Layer (T): 4 bits
+//    Indicates which temporal layer this bitrate concerns.
+//
+//  Target Bitrate: 24 bits
+//    The encoder target bitrate for this layer, in kbps.
+//
+//  As an example of how S and T are intended to be used, VP8 simulcast will
+//  use a separate TargetBitrate message per stream, since they are transmitted
+//  on separate SSRCs, with temporal layers grouped by stream.
+//  If VP9 SVC is used, there will be only one SSRC, so each spatial and
+//  temporal layer combo used shall be specified in the TargetBitrate packet.
+
+TargetBitrate::TargetBitrate() = default;
+TargetBitrate::TargetBitrate(const TargetBitrate&) = default;
+TargetBitrate& TargetBitrate::operator=(const TargetBitrate&) = default;
+TargetBitrate::~TargetBitrate() = default;
+
+void TargetBitrate::Parse(const uint8_t* block, uint16_t block_length) {
+  // Validate block header (should already have been parsed and checked).
+  RTC_DCHECK_EQ(block[0], kBlockType);
+  RTC_DCHECK_EQ(block_length, ByteReader<uint16_t>::ReadBigEndian(&block[2]));
+
+  // Header specifies block length - 1, but since we ignore the header, which
+  // occupies exactly on block, we can just treat this as payload length.
+  const size_t payload_bytes = block_length * 4;
+  const size_t num_items = payload_bytes / kBitrateItemSizeBytes;
+  size_t index = kTargetBitrateHeaderSizeBytes;
+  bitrates_.clear();
+  for (size_t i = 0; i < num_items; ++i) {
+    uint8_t layers = block[index];
+    uint32_t bitrate_kbps =
+        ByteReader<uint32_t, 3>::ReadBigEndian(&block[index + 1]);
+    index += kBitrateItemSizeBytes;
+    AddTargetBitrate((layers >> 4) & 0x0F, layers & 0x0F, bitrate_kbps);
+  }
+}
+
+void TargetBitrate::AddTargetBitrate(uint8_t spatial_layer,
+                                     uint8_t temporal_layer,
+                                     uint32_t target_bitrate_kbps) {
+  RTC_DCHECK_LE(spatial_layer, 0x0F);
+  RTC_DCHECK_LE(temporal_layer, 0x0F);
+  RTC_DCHECK_LE(target_bitrate_kbps, 0x00FFFFFFU);
+  bitrates_.push_back(
+      BitrateItem(spatial_layer, temporal_layer, target_bitrate_kbps));
+}
+
+const std::vector<TargetBitrate::BitrateItem>&
+TargetBitrate::GetTargetBitrates() const {
+  return bitrates_;
+}
+
+size_t TargetBitrate::BlockLength() const {
+  return kTargetBitrateHeaderSizeBytes +
+         bitrates_.size() * kBitrateItemSizeBytes;
+}
+
+void TargetBitrate::Create(uint8_t* buffer) const {
+  buffer[0] = kBlockType;
+  buffer[1] = 0;  // Reserved.
+  uint16_t block_length_words =
+      rtc::dchecked_cast<uint16_t>((BlockLength() / 4) - 1);
+  ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], block_length_words);
+
+  size_t index = kTargetBitrateHeaderSizeBytes;
+  for (const BitrateItem& item : bitrates_) {
+    buffer[index] = (item.spatial_layer << 4) | item.temporal_layer;
+    ByteWriter<uint32_t, 3>::WriteBigEndian(&buffer[index + 1],
+                                            item.target_bitrate_kbps);
+    index += kBitrateItemSizeBytes;
+  }
+}
+
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h
new file mode 100644
index 0000000000..07e5da1a49
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TARGET_BITRATE_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TARGET_BITRATE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+namespace webrtc {
+namespace rtcp {
+
+class TargetBitrate {
+ public:
+  // TODO(sprang): This block type is just a place holder. We need to get an
+  //               id assigned by IANA.
+  static constexpr uint8_t kBlockType = 42;
+  static const size_t kBitrateItemSizeBytes;
+
+  struct BitrateItem {
+    BitrateItem();
+    BitrateItem(uint8_t spatial_layer,
+                uint8_t temporal_layer,
+                uint32_t target_bitrate_kbps);
+
+    uint8_t spatial_layer;
+    uint8_t temporal_layer;
+    uint32_t target_bitrate_kbps;
+  };
+
+  TargetBitrate();
+  TargetBitrate(const TargetBitrate&);
+  TargetBitrate& operator=(const TargetBitrate&);
+  ~TargetBitrate();
+
+  void AddTargetBitrate(uint8_t spatial_layer,
+                        uint8_t temporal_layer,
+                        uint32_t target_bitrate_kbps);
+
+  const std::vector<BitrateItem>& GetTargetBitrates() const;
+
+  void Parse(const uint8_t* block, uint16_t block_length);
+
+  size_t BlockLength() const;
+
+  void Create(uint8_t* buffer) const;
+
+ private:
+  std::vector<BitrateItem> bitrates_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TARGET_BITRATE_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc
new file mode 100644
index 0000000000..b16bb5beaa
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "rtc_base/buffer.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+namespace webrtc {
+namespace {
+using BitrateItem = rtcp::TargetBitrate::BitrateItem;
+using rtcp::TargetBitrate;
+using test::ParseSinglePacket;
+
+constexpr uint32_t kSsrc = 0x12345678;
+
+// clang-format off
+const uint8_t kPacket[] = { TargetBitrate::kBlockType,  // Block ID.
+                                  0x00,                 // Reserved.
+                                        0x00, 0x04,     // Length = 4 words.
+                            0x00, 0x01, 0x02, 0x03,     // S0T0 0x010203 kbps.
+                            0x01, 0x02, 0x03, 0x04,     // S0T1 0x020304 kbps.
+                            0x10, 0x03, 0x04, 0x05,     // S1T0 0x030405 kbps.
+                            0x11, 0x04, 0x05, 0x06 };   // S1T1 0x040506 kbps.
+constexpr size_t kPacketLengthBlocks = ((sizeof(kPacket) + 3) / 4) - 1;
+// clang-format on
+
+void ExpectBirateItemEquals(const BitrateItem& expected,
+                            const BitrateItem& actual) {
+  EXPECT_EQ(expected.spatial_layer, actual.spatial_layer);
+  EXPECT_EQ(expected.temporal_layer, actual.temporal_layer);
+  EXPECT_EQ(expected.target_bitrate_kbps, actual.target_bitrate_kbps);
+}
+
+void CheckBitrateItems(const std::vector<BitrateItem>& bitrates) {
+  EXPECT_EQ(4U, bitrates.size());
+  ExpectBirateItemEquals(BitrateItem(0, 0, 0x010203), bitrates[0]);
+  ExpectBirateItemEquals(BitrateItem(0, 1, 0x020304), bitrates[1]);
+  ExpectBirateItemEquals(BitrateItem(1, 0, 0x030405), bitrates[2]);
+  ExpectBirateItemEquals(BitrateItem(1, 1, 0x040506), bitrates[3]);
+}
+
+}  // namespace
+
+TEST(TargetBitrateTest, Parse) {
+  TargetBitrate target_bitrate;
+  target_bitrate.Parse(kPacket, kPacketLengthBlocks);
+  CheckBitrateItems(target_bitrate.GetTargetBitrates());
+}
+
+TEST(TargetBitrateTest, FullPacket) {
+  const size_t kXRHeaderSize = 8;  // RTCP header (4) + SSRC (4).
+  const size_t kTotalSize = kXRHeaderSize + sizeof(kPacket);
+  uint8_t kRtcpPacket[kTotalSize] = {2 << 6, 207,  0x00, (kTotalSize / 4) - 1,
+                                     0x12,   0x34, 0x56, 0x78};  // SSRC.
+  memcpy(&kRtcpPacket[kXRHeaderSize], kPacket, sizeof(kPacket));
+  rtcp::ExtendedReports xr;
+  EXPECT_TRUE(ParseSinglePacket(kRtcpPacket, &xr));
+  EXPECT_EQ(kSsrc, xr.sender_ssrc());
+  const absl::optional<TargetBitrate>& target_bitrate = xr.target_bitrate();
+  ASSERT_TRUE(static_cast<bool>(target_bitrate));
+  CheckBitrateItems(target_bitrate->GetTargetBitrates());
+}
+
+TEST(TargetBitrateTest, Create) {
+  TargetBitrate target_bitrate;
+  target_bitrate.AddTargetBitrate(0, 0, 0x010203);
+  target_bitrate.AddTargetBitrate(0, 1, 0x020304);
+  target_bitrate.AddTargetBitrate(1, 0, 0x030405);
+  target_bitrate.AddTargetBitrate(1, 1, 0x040506);
+
+  uint8_t buffer[sizeof(kPacket)] = {};
+  ASSERT_EQ(sizeof(kPacket), target_bitrate.BlockLength());
+  target_bitrate.Create(buffer);
+
+  EXPECT_EQ(0, memcmp(kPacket, buffer, sizeof(kPacket)));
+}
+
+TEST(TargetBitrateTest, ParseNullBitratePacket) {
+  const uint8_t kNullPacket[] = {TargetBitrate::kBlockType, 0x00, 0x00, 0x00};
+  TargetBitrate target_bitrate;
+  target_bitrate.Parse(kNullPacket, 0);
+  EXPECT_TRUE(target_bitrate.GetTargetBitrates().empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.cc
new file mode 100644
index 0000000000..810e1e267a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+TmmbItem::TmmbItem(uint32_t ssrc, uint64_t bitrate_bps, uint16_t overhead)
+    : ssrc_(ssrc), bitrate_bps_(bitrate_bps), packet_overhead_(overhead) {
+  RTC_DCHECK_LE(overhead, 0x1ffu);
+}
+
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 0 |                              SSRC                             |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 4 | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+bool TmmbItem::Parse(const uint8_t* buffer) {
+  ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]);
+  // Read 4 bytes into 1 block.
+  uint32_t compact = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  // Split 1 block into 3 components.
+  uint8_t exponent = compact >> 26;              // 6 bits.
+  uint64_t mantissa = (compact >> 9) & 0x1ffff;  // 17 bits.
+  uint16_t overhead = compact & 0x1ff;           // 9 bits.
+  // Combine 3 components into 2 values.
+  bitrate_bps_ = (mantissa << exponent);
+
+  bool shift_overflow = (bitrate_bps_ >> exponent) != mantissa;
+  if (shift_overflow) {
+    RTC_LOG(LS_ERROR) << "Invalid tmmb bitrate value : " << mantissa << "*2^"
+                      << static_cast<int>(exponent);
+    return false;
+  }
+  packet_overhead_ = overhead;
+  return true;
+}
+
+void TmmbItem::Create(uint8_t* buffer) const {
+  constexpr uint64_t kMaxMantissa = 0x1ffff;  // 17 bits.
+  uint64_t mantissa = bitrate_bps_;
+  uint32_t exponent = 0;
+  while (mantissa > kMaxMantissa) {
+    mantissa >>= 1;
+    ++exponent;
+  }
+
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[0], ssrc_);
+  uint32_t compact = (exponent << 26) | (mantissa << 9) | packet_overhead_;
+  ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], compact);
+}
+
+void TmmbItem::set_packet_overhead(uint16_t overhead) {
+  RTC_DCHECK_LE(overhead, 0x1ffu);
+  packet_overhead_ = overhead;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h
new file mode 100644
index 0000000000..dc5d1b2c2d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMB_ITEM_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMB_ITEM_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+namespace rtcp {
+// RFC5104, Section 3.5.4
+// Temporary Maximum Media Stream Bitrate Request/Notification.
+// Used both by TMMBR and TMMBN rtcp packets.
+class TmmbItem {
+ public:
+  static const size_t kLength = 8;
+
+  TmmbItem() : ssrc_(0), bitrate_bps_(0), packet_overhead_(0) {}
+  TmmbItem(uint32_t ssrc, uint64_t bitrate_bps, uint16_t overhead);
+
+  bool Parse(const uint8_t* buffer);
+  void Create(uint8_t* buffer) const;
+
+  void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; }
+  void set_bitrate_bps(uint64_t bitrate_bps) { bitrate_bps_ = bitrate_bps; }
+  void set_packet_overhead(uint16_t overhead);
+
+  uint32_t ssrc() const { return ssrc_; }
+  uint64_t bitrate_bps() const { return bitrate_bps_; }
+  uint16_t packet_overhead() const { return packet_overhead_; }
+
+ private:
+  // Media stream id.
+  uint32_t ssrc_;
+  // Maximum total media bit rate that the media receiver is
+  // currently prepared to accept for this media stream.
+  uint64_t bitrate_bps_;
+  // Per-packet overhead that the media receiver has observed
+  // for this media stream at its chosen reference protocol layer.
+  uint16_t packet_overhead_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMB_ITEM_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
new file mode 100644
index 0000000000..f57e5749c2
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Tmmbn::kFeedbackMessageType;
+// RFC 4585: Feedback format.
+// Common packet format:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |V=2|P|   FMT   |       PT      |          length               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |                  SSRC of packet sender                        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |             SSRC of media source (unused) = 0                 |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   :            Feedback Control Information (FCI)                 :
+//   :                                                               :
+// Temporary Maximum Media Stream Bit Rate Notification (TMMBN) (RFC 5104).
+// The Feedback Control Information (FCI) consists of zero, one, or more
+// TMMBN FCI entries.
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |                              SSRC                             |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+Tmmbn::Tmmbn() = default;
+
+Tmmbn::~Tmmbn() = default;
+
+bool Tmmbn::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  if (packet.payload_size_bytes() < kCommonFeedbackLength) {
+    RTC_LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
+                        << " is too small for TMMBN.";
+    return false;
+  }
+  size_t items_size_bytes = packet.payload_size_bytes() - kCommonFeedbackLength;
+  if (items_size_bytes % TmmbItem::kLength != 0) {
+    RTC_LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
+                        << " is not valid for TMMBN.";
+    return false;
+  }
+  ParseCommonFeedback(packet.payload());
+  const uint8_t* next_item = packet.payload() + kCommonFeedbackLength;
+
+  size_t number_of_items = items_size_bytes / TmmbItem::kLength;
+  items_.resize(number_of_items);
+  for (TmmbItem& item : items_) {
+    if (!item.Parse(next_item))
+      return false;
+    next_item += TmmbItem::kLength;
+  }
+  return true;
+}
+
+void Tmmbn::AddTmmbr(const TmmbItem& item) {
+  items_.push_back(item);
+}
+
+size_t Tmmbn::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength +
+         TmmbItem::kLength * items_.size();
+}
+
+bool Tmmbn::Create(uint8_t* packet,
+                   size_t* index,
+                   size_t max_length,
+                   PacketReadyCallback callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  RTC_DCHECK_EQ(0, Rtpfb::media_ssrc());
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  for (const TmmbItem& item : items_) {
+    item.Create(packet + *index);
+    *index += TmmbItem::kLength;
+  }
+  RTC_CHECK_EQ(index_end, *index);
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
new file mode 100644
index 0000000000..ff7779d8ac
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBN_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBN_H_
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+// Temporary Maximum Media Stream Bit Rate Notification (TMMBN).
+// RFC 5104, Section 4.2.2.
+class Tmmbn : public Rtpfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 4;
+
+  Tmmbn();
+  ~Tmmbn() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void AddTmmbr(const TmmbItem& item);
+
+  const std::vector<TmmbItem>& items() const { return items_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  // Media ssrc is unused, shadow base class setter and getter.
+  void SetMediaSsrc(uint32_t ssrc);
+  uint32_t media_ssrc() const;
+
+  std::vector<TmmbItem> items_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBN_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc
new file mode 100644
index 0000000000..3a37bb1c0e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+using ::testing::make_tuple;
+using webrtc::rtcp::TmmbItem;
+using webrtc::rtcp::Tmmbn;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint32_t kBitrateBps = 312000;
+const uint16_t kOverhead = 0x1fe;
+const uint8_t kPacket[] = {0x84, 205,  0x00, 0x04, 0x12, 0x34, 0x56,
+                           0x78, 0x00, 0x00, 0x00, 0x00, 0x23, 0x45,
+                           0x67, 0x89, 0x0a, 0x61, 0x61, 0xfe};
+}  // namespace
+
+TEST(RtcpPacketTmmbnTest, Create) {
+  Tmmbn tmmbn;
+  tmmbn.SetSenderSsrc(kSenderSsrc);
+  tmmbn.AddTmmbr(TmmbItem(kRemoteSsrc, kBitrateBps, kOverhead));
+
+  rtc::Buffer packet = tmmbn.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketTmmbnTest, Parse) {
+  Tmmbn tmmbn;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &tmmbn));
+
+  const Tmmbn& parsed = tmmbn;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  ASSERT_EQ(1u, parsed.items().size());
+  EXPECT_EQ(kRemoteSsrc, parsed.items().front().ssrc());
+  EXPECT_EQ(kBitrateBps, parsed.items().front().bitrate_bps());
+  EXPECT_EQ(kOverhead, parsed.items().front().packet_overhead());
+}
+
+TEST(RtcpPacketTmmbnTest, CreateAndParseWithoutItems) {
+  Tmmbn tmmbn;
+  tmmbn.SetSenderSsrc(kSenderSsrc);
+
+  rtc::Buffer packet = tmmbn.Build();
+  Tmmbn parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_THAT(parsed.items(), IsEmpty());
+}
+
+TEST(RtcpPacketTmmbnTest, CreateAndParseWithTwoItems) {
+  Tmmbn tmmbn;
+  tmmbn.SetSenderSsrc(kSenderSsrc);
+  tmmbn.AddTmmbr(TmmbItem(kRemoteSsrc, kBitrateBps, kOverhead));
+  tmmbn.AddTmmbr(TmmbItem(kRemoteSsrc + 1, 4 * kBitrateBps, 40));
+
+  rtc::Buffer packet = tmmbn.Build();
+  Tmmbn parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(2u, parsed.items().size());
+  EXPECT_EQ(kRemoteSsrc, parsed.items()[0].ssrc());
+  EXPECT_EQ(kBitrateBps, parsed.items()[0].bitrate_bps());
+  EXPECT_EQ(kOverhead, parsed.items()[0].packet_overhead());
+  EXPECT_EQ(kRemoteSsrc + 1, parsed.items()[1].ssrc());
+  EXPECT_EQ(4 * kBitrateBps, parsed.items()[1].bitrate_bps());
+  EXPECT_EQ(40U, parsed.items()[1].packet_overhead());
+}
+
+TEST(RtcpPacketTmmbnTest, ParseFailsOnTooSmallPacket) {
+  const uint8_t kSmallPacket[] = {0x84, 205,  0x00, 0x01,
+                                  0x12, 0x34, 0x56, 0x78};
+  Tmmbn tmmbn;
+  EXPECT_FALSE(test::ParseSinglePacket(kSmallPacket, &tmmbn));
+}
+
+TEST(RtcpPacketTmmbnTest, ParseFailsOnUnAlignedPacket) {
+  const uint8_t kUnalignedPacket[] = {0x84, 205,  0x00, 0x03, 0x12, 0x34,
+                                      0x56, 0x78, 0x00, 0x00, 0x00, 0x00,
+                                      0x23, 0x45, 0x67, 0x89};
+
+  Tmmbn tmmbn;
+  EXPECT_FALSE(test::ParseSinglePacket(kUnalignedPacket, &tmmbn));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
new file mode 100644
index 0000000000..9dc745e509
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace rtcp {
+constexpr uint8_t Tmmbr::kFeedbackMessageType;
+// RFC 4585: Feedback format.
+// Common packet format:
+//
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |V=2|P|   FMT   |       PT      |          length               |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                  SSRC of packet sender                        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |             SSRC of media source (unused) = 0                 |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  :            Feedback Control Information (FCI)                 :
+//  :                                                               :
+// Temporary Maximum Media Stream Bit Rate Request (TMMBR) (RFC 5104).
+// The Feedback Control Information (FCI) for the TMMBR
+// consists of one or more FCI entries.
+// FCI:
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |                              SSRC                             |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  | MxTBR Exp |  MxTBR Mantissa                 |Measured Overhead|
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+Tmmbr::Tmmbr() = default;
+
+Tmmbr::~Tmmbr() = default;
+
+bool Tmmbr::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+
+  if (packet.payload_size_bytes() < kCommonFeedbackLength + TmmbItem::kLength) {
+    RTC_LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
+                        << " is too small for a TMMBR.";
+    return false;
+  }
+  size_t items_size_bytes = packet.payload_size_bytes() - kCommonFeedbackLength;
+  if (items_size_bytes % TmmbItem::kLength != 0) {
+    RTC_LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
+                        << " is not valid for a TMMBR.";
+    return false;
+  }
+  ParseCommonFeedback(packet.payload());
+
+  const uint8_t* next_item = packet.payload() + kCommonFeedbackLength;
+  size_t number_of_items = items_size_bytes / TmmbItem::kLength;
+  items_.resize(number_of_items);
+  for (TmmbItem& item : items_) {
+    if (!item.Parse(next_item))
+      return false;
+    next_item += TmmbItem::kLength;
+  }
+  return true;
+}
+
+void Tmmbr::AddTmmbr(const TmmbItem& item) {
+  items_.push_back(item);
+}
+
+size_t Tmmbr::BlockLength() const {
+  return kHeaderLength + kCommonFeedbackLength +
+         TmmbItem::kLength * items_.size();
+}
+
+bool Tmmbr::Create(uint8_t* packet,
+                   size_t* index,
+                   size_t max_length,
+                   PacketReadyCallback callback) const {
+  RTC_DCHECK(!items_.empty());
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
+               index);
+  RTC_DCHECK_EQ(0, Rtpfb::media_ssrc());
+  CreateCommonFeedback(packet + *index);
+  *index += kCommonFeedbackLength;
+  for (const TmmbItem& item : items_) {
+    item.Create(packet + *index);
+    *index += TmmbItem::kLength;
+  }
+  RTC_CHECK_EQ(index_end, *index);
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
new file mode 100644
index 0000000000..7482cb75cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBR_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBR_H_
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+// Temporary Maximum Media Stream Bit Rate Request (TMMBR).
+// RFC 5104, Section 4.2.1.
+class Tmmbr : public Rtpfb {
+ public:
+  static constexpr uint8_t kFeedbackMessageType = 3;
+
+  Tmmbr();
+  ~Tmmbr() override;
+
+  // Parse assumes header is already parsed and validated.
+  bool Parse(const CommonHeader& packet);
+
+  void AddTmmbr(const TmmbItem& item);
+
+  const std::vector<TmmbItem>& requests() const { return items_; }
+
+  size_t BlockLength() const override;
+
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  // Media ssrc is unused, shadow base class setter.
+  void SetMediaSsrc(uint32_t ssrc);
+
+  std::vector<TmmbItem> items_;
+};
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMBR_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
new file mode 100644
index 0000000000..1bac808aa9
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAreArray;
+using ::testing::make_tuple;
+using webrtc::rtcp::TmmbItem;
+using webrtc::rtcp::Tmmbr;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345678;
+const uint32_t kRemoteSsrc = 0x23456789;
+const uint32_t kBitrateBps = 312000;
+const uint16_t kOverhead = 0x1fe;
+const uint8_t kPacket[] = {0x83, 205,  0x00, 0x04, 0x12, 0x34, 0x56,
+                           0x78, 0x00, 0x00, 0x00, 0x00, 0x23, 0x45,
+                           0x67, 0x89, 0x0a, 0x61, 0x61, 0xfe};
+}  // namespace
+
+TEST(RtcpPacketTmmbrTest, Create) {
+  Tmmbr tmmbr;
+  tmmbr.SetSenderSsrc(kSenderSsrc);
+  tmmbr.AddTmmbr(TmmbItem(kRemoteSsrc, kBitrateBps, kOverhead));
+
+  rtc::Buffer packet = tmmbr.Build();
+
+  EXPECT_THAT(make_tuple(packet.data(), packet.size()),
+              ElementsAreArray(kPacket));
+}
+
+TEST(RtcpPacketTmmbrTest, Parse) {
+  Tmmbr tmmbr;
+  EXPECT_TRUE(test::ParseSinglePacket(kPacket, &tmmbr));
+  const Tmmbr& parsed = tmmbr;
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  ASSERT_EQ(1u, parsed.requests().size());
+  EXPECT_EQ(kRemoteSsrc, parsed.requests().front().ssrc());
+  EXPECT_EQ(kBitrateBps, parsed.requests().front().bitrate_bps());
+  EXPECT_EQ(kOverhead, parsed.requests().front().packet_overhead());
+}
+
+TEST(RtcpPacketTmmbrTest, CreateAndParseWithTwoEntries) {
+  Tmmbr tmmbr;
+  tmmbr.SetSenderSsrc(kSenderSsrc);
+  tmmbr.AddTmmbr(TmmbItem(kRemoteSsrc, kBitrateBps, kOverhead));
+  tmmbr.AddTmmbr(TmmbItem(kRemoteSsrc + 1, 4 * kBitrateBps, kOverhead + 1));
+
+  rtc::Buffer packet = tmmbr.Build();
+
+  Tmmbr parsed;
+  EXPECT_TRUE(test::ParseSinglePacket(packet, &parsed));
+
+  EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
+  EXPECT_EQ(2u, parsed.requests().size());
+  EXPECT_EQ(kRemoteSsrc, parsed.requests()[0].ssrc());
+  EXPECT_EQ(kRemoteSsrc + 1, parsed.requests()[1].ssrc());
+}
+
+TEST(RtcpPacketTmmbrTest, ParseFailsWithoutItems) {
+  const uint8_t kZeroItemsPacket[] = {0x83, 205,  0x00, 0x02, 0x12, 0x34,
+                                      0x56, 0x78, 0x00, 0x00, 0x00, 0x00};
+
+  Tmmbr tmmbr;
+  EXPECT_FALSE(test::ParseSinglePacket(kZeroItemsPacket, &tmmbr));
+}
+
+TEST(RtcpPacketTmmbrTest, ParseFailsOnUnAlignedPacket) {
+  const uint8_t kUnalignedPacket[] = {
+      0x83, 205,  0x00, 0x05, 0x12, 0x34, 0x56, 0x78, 0x00, 0x00, 0x00, 0x00,
+      0x23, 0x45, 0x67, 0x89, 0x0a, 0x61, 0x61, 0xfe, 0x34, 0x56, 0x78, 0x9a};
+
+  Tmmbr tmmbr;
+  EXPECT_FALSE(test::ParseSinglePacket(kUnalignedPacket, &tmmbr));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
new file mode 100644
index 0000000000..003effad29
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
@@ -0,0 +1,737 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <numeric>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+namespace rtcp {
+namespace {
+// Header size:
+// * 4 bytes Common RTCP Packet Header
+// * 8 bytes Common Packet Format for RTCP Feedback Messages
+// * 8 bytes FeedbackPacket header
+constexpr size_t kTransportFeedbackHeaderSizeBytes = 4 + 8 + 8;
+constexpr size_t kChunkSizeBytes = 2;
+// TODO(sprang): Add support for dynamic max size for easier fragmentation,
+// eg. set it to what's left in the buffer or IP_PACKET_SIZE.
+// Size constraint imposed by RTCP common header: 16bit size field interpreted
+// as number of four byte words minus the first header word.
+constexpr size_t kMaxSizeBytes = (1 << 16) * 4;
+// Payload size:
+// * 8 bytes Common Packet Format for RTCP Feedback Messages
+// * 8 bytes FeedbackPacket header.
+// * 2 bytes for one chunk.
+constexpr size_t kMinPayloadSizeBytes = 8 + 8 + 2;
+constexpr TimeDelta kBaseTimeTick = TransportFeedback::kDeltaTick * (1 << 8);
+constexpr TimeDelta kTimeWrapPeriod = kBaseTimeTick * (1 << 24);
+
+//    Message format
+//
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P|  FMT=15 |    PT=205     |           length              |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |                     SSRC of packet sender                     |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  4 |                      SSRC of media source                     |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |      base sequence number     |      packet status count      |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |                 reference time                | fb pkt. count |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |          packet chunk         |         packet chunk          |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    .                                                               .
+//    .                                                               .
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |         packet chunk          |  recv delta   |  recv delta   |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    .                                                               .
+//    .                                                               .
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |           recv delta          |  recv delta   | zero padding  |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+}  // namespace
+constexpr uint8_t TransportFeedback::kFeedbackMessageType;
+constexpr size_t TransportFeedback::kMaxReportedPackets;
+
+constexpr size_t TransportFeedback::LastChunk::kMaxRunLengthCapacity;
+constexpr size_t TransportFeedback::LastChunk::kMaxOneBitCapacity;
+constexpr size_t TransportFeedback::LastChunk::kMaxTwoBitCapacity;
+constexpr size_t TransportFeedback::LastChunk::kMaxVectorCapacity;
+
+TransportFeedback::LastChunk::LastChunk() {
+  Clear();
+}
+
+bool TransportFeedback::LastChunk::Empty() const {
+  return size_ == 0;
+}
+
+void TransportFeedback::LastChunk::Clear() {
+  size_ = 0;
+  all_same_ = true;
+  has_large_delta_ = false;
+}
+
+bool TransportFeedback::LastChunk::CanAdd(DeltaSize delta_size) const {
+  RTC_DCHECK_LE(delta_size, 2);
+  if (size_ < kMaxTwoBitCapacity)
+    return true;
+  if (size_ < kMaxOneBitCapacity && !has_large_delta_ && delta_size != kLarge)
+    return true;
+  if (size_ < kMaxRunLengthCapacity && all_same_ &&
+      delta_sizes_[0] == delta_size)
+    return true;
+  return false;
+}
+
+void TransportFeedback::LastChunk::Add(DeltaSize delta_size) {
+  RTC_DCHECK(CanAdd(delta_size));
+  if (size_ < kMaxVectorCapacity)
+    delta_sizes_[size_] = delta_size;
+  size_++;
+  all_same_ = all_same_ && delta_size == delta_sizes_[0];
+  has_large_delta_ = has_large_delta_ || delta_size == kLarge;
+}
+
+void TransportFeedback::LastChunk::AddMissingPackets(size_t num_missing) {
+  RTC_DCHECK_EQ(size_, 0);
+  RTC_DCHECK(all_same_);
+  RTC_DCHECK(!has_large_delta_);
+  RTC_DCHECK_LT(num_missing, kMaxRunLengthCapacity);
+  absl::c_fill(delta_sizes_, DeltaSize(0));
+  size_ = num_missing;
+}
+
+uint16_t TransportFeedback::LastChunk::Emit() {
+  RTC_DCHECK(!CanAdd(0) || !CanAdd(1) || !CanAdd(2));
+  if (all_same_) {
+    uint16_t chunk = EncodeRunLength();
+    Clear();
+    return chunk;
+  }
+  if (size_ == kMaxOneBitCapacity) {
+    uint16_t chunk = EncodeOneBit();
+    Clear();
+    return chunk;
+  }
+  RTC_DCHECK_GE(size_, kMaxTwoBitCapacity);
+  uint16_t chunk = EncodeTwoBit(kMaxTwoBitCapacity);
+  // Remove `kMaxTwoBitCapacity` encoded delta sizes:
+  // Shift remaining delta sizes and recalculate all_same_ && has_large_delta_.
+  size_ -= kMaxTwoBitCapacity;
+  all_same_ = true;
+  has_large_delta_ = false;
+  for (size_t i = 0; i < size_; ++i) {
+    DeltaSize delta_size = delta_sizes_[kMaxTwoBitCapacity + i];
+    delta_sizes_[i] = delta_size;
+    all_same_ = all_same_ && delta_size == delta_sizes_[0];
+    has_large_delta_ = has_large_delta_ || delta_size == kLarge;
+  }
+
+  return chunk;
+}
+
+uint16_t TransportFeedback::LastChunk::EncodeLast() const {
+  RTC_DCHECK_GT(size_, 0);
+  if (all_same_)
+    return EncodeRunLength();
+  if (size_ <= kMaxTwoBitCapacity)
+    return EncodeTwoBit(size_);
+  return EncodeOneBit();
+}
+
+// Appends content of the Lastchunk to `deltas`.
+void TransportFeedback::LastChunk::AppendTo(
+    std::vector<DeltaSize>* deltas) const {
+  if (all_same_) {
+    deltas->insert(deltas->end(), size_, delta_sizes_[0]);
+  } else {
+    deltas->insert(deltas->end(), delta_sizes_.begin(),
+                   delta_sizes_.begin() + size_);
+  }
+}
+
+void TransportFeedback::LastChunk::Decode(uint16_t chunk, size_t max_size) {
+  if ((chunk & 0x8000) == 0) {
+    DecodeRunLength(chunk, max_size);
+  } else if ((chunk & 0x4000) == 0) {
+    DecodeOneBit(chunk, max_size);
+  } else {
+    DecodeTwoBit(chunk, max_size);
+  }
+}
+
+//  One Bit Status Vector Chunk
+//
+//  0                   1
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |T|S|       symbol list         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+//  T = 1
+//  S = 0
+//  Symbol list = 14 entries where 0 = not received, 1 = received 1-byte delta.
+uint16_t TransportFeedback::LastChunk::EncodeOneBit() const {
+  RTC_DCHECK(!has_large_delta_);
+  RTC_DCHECK_LE(size_, kMaxOneBitCapacity);
+  uint16_t chunk = 0x8000;
+  for (size_t i = 0; i < size_; ++i)
+    chunk |= delta_sizes_[i] << (kMaxOneBitCapacity - 1 - i);
+  return chunk;
+}
+
+void TransportFeedback::LastChunk::DecodeOneBit(uint16_t chunk,
+                                                size_t max_size) {
+  RTC_DCHECK_EQ(chunk & 0xc000, 0x8000);
+  size_ = std::min(kMaxOneBitCapacity, max_size);
+  has_large_delta_ = false;
+  all_same_ = false;
+  for (size_t i = 0; i < size_; ++i)
+    delta_sizes_[i] = (chunk >> (kMaxOneBitCapacity - 1 - i)) & 0x01;
+}
+
+//  Two Bit Status Vector Chunk
+//
+//  0                   1
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |T|S|       symbol list         |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+//  T = 1
+//  S = 1
+//  symbol list = 7 entries of two bits each.
+uint16_t TransportFeedback::LastChunk::EncodeTwoBit(size_t size) const {
+  RTC_DCHECK_LE(size, size_);
+  uint16_t chunk = 0xc000;
+  for (size_t i = 0; i < size; ++i)
+    chunk |= delta_sizes_[i] << 2 * (kMaxTwoBitCapacity - 1 - i);
+  return chunk;
+}
+
+void TransportFeedback::LastChunk::DecodeTwoBit(uint16_t chunk,
+                                                size_t max_size) {
+  RTC_DCHECK_EQ(chunk & 0xc000, 0xc000);
+  size_ = std::min(kMaxTwoBitCapacity, max_size);
+  has_large_delta_ = true;
+  all_same_ = false;
+  for (size_t i = 0; i < size_; ++i)
+    delta_sizes_[i] = (chunk >> 2 * (kMaxTwoBitCapacity - 1 - i)) & 0x03;
+}
+
+//  Run Length Status Vector Chunk
+//
+//  0                   1
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |T| S |       Run Length        |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+//  T = 0
+//  S = symbol
+//  Run Length = Unsigned integer denoting the run length of the symbol
+uint16_t TransportFeedback::LastChunk::EncodeRunLength() const {
+  RTC_DCHECK(all_same_);
+  RTC_DCHECK_LE(size_, kMaxRunLengthCapacity);
+  return (delta_sizes_[0] << 13) | static_cast<uint16_t>(size_);
+}
+
+void TransportFeedback::LastChunk::DecodeRunLength(uint16_t chunk,
+                                                   size_t max_count) {
+  RTC_DCHECK_EQ(chunk & 0x8000, 0);
+  size_ = std::min<size_t>(chunk & 0x1fff, max_count);
+  DeltaSize delta_size = (chunk >> 13) & 0x03;
+  has_large_delta_ = delta_size >= kLarge;
+  all_same_ = true;
+  // To make it consistent with Add function, populate delta_sizes_ beyond 1st.
+  for (size_t i = 0; i < std::min<size_t>(size_, kMaxVectorCapacity); ++i)
+    delta_sizes_[i] = delta_size;
+}
+
+TransportFeedback::TransportFeedback()
+    : TransportFeedback(/*include_timestamps=*/true) {}
+
+TransportFeedback::TransportFeedback(bool include_timestamps)
+    : base_seq_no_(0),
+      num_seq_no_(0),
+      base_time_ticks_(0),
+      feedback_seq_(0),
+      include_timestamps_(include_timestamps),
+      last_timestamp_(Timestamp::Zero()),
+      size_bytes_(kTransportFeedbackHeaderSizeBytes) {}
+
+TransportFeedback::TransportFeedback(const TransportFeedback&) = default;
+
+TransportFeedback::TransportFeedback(TransportFeedback&& other)
+    : base_seq_no_(other.base_seq_no_),
+      num_seq_no_(other.num_seq_no_),
+      base_time_ticks_(other.base_time_ticks_),
+      feedback_seq_(other.feedback_seq_),
+      include_timestamps_(other.include_timestamps_),
+      last_timestamp_(other.last_timestamp_),
+      received_packets_(std::move(other.received_packets_)),
+      all_packets_(std::move(other.all_packets_)),
+      encoded_chunks_(std::move(other.encoded_chunks_)),
+      last_chunk_(other.last_chunk_),
+      size_bytes_(other.size_bytes_) {
+  other.Clear();
+}
+
+TransportFeedback::~TransportFeedback() {}
+
+void TransportFeedback::SetBase(uint16_t base_sequence,
+                                Timestamp ref_timestamp) {
+  RTC_DCHECK_EQ(num_seq_no_, 0);
+  base_seq_no_ = base_sequence;
+  base_time_ticks_ =
+      (ref_timestamp.us() % kTimeWrapPeriod.us()) / kBaseTimeTick.us();
+  last_timestamp_ = BaseTime();
+}
+
+void TransportFeedback::SetFeedbackSequenceNumber(uint8_t feedback_sequence) {
+  feedback_seq_ = feedback_sequence;
+}
+
+bool TransportFeedback::AddReceivedPacket(uint16_t sequence_number,
+                                          Timestamp timestamp) {
+  // Set delta to zero if timestamps are not included, this will simplify the
+  // encoding process.
+  int16_t delta = 0;
+  if (include_timestamps_) {
+    // Convert to ticks and round.
+    if (last_timestamp_ > timestamp) {
+      timestamp += (last_timestamp_ - timestamp).RoundUpTo(kTimeWrapPeriod);
+    }
+    RTC_DCHECK_GE(timestamp, last_timestamp_);
+    int64_t delta_full =
+        (timestamp - last_timestamp_).us() % kTimeWrapPeriod.us();
+    if (delta_full > kTimeWrapPeriod.us() / 2) {
+      delta_full -= kTimeWrapPeriod.us();
+      delta_full -= kDeltaTick.us() / 2;
+    } else {
+      delta_full += kDeltaTick.us() / 2;
+    }
+    delta_full /= kDeltaTick.us();
+
+    delta = static_cast<int16_t>(delta_full);
+    // If larger than 16bit signed, we can't represent it - need new fb packet.
+    if (delta != delta_full) {
+      RTC_LOG(LS_WARNING) << "Delta value too large ( >= 2^16 ticks )";
+      return false;
+    }
+  }
+
+  uint16_t next_seq_no = base_seq_no_ + num_seq_no_;
+  if (sequence_number != next_seq_no) {
+    uint16_t last_seq_no = next_seq_no - 1;
+    if (!IsNewerSequenceNumber(sequence_number, last_seq_no))
+      return false;
+    uint16_t num_missing_packets = sequence_number - next_seq_no;
+    if (!AddMissingPackets(num_missing_packets))
+      return false;
+  }
+
+  DeltaSize delta_size = (delta >= 0 && delta <= 0xff) ? 1 : 2;
+  if (!AddDeltaSize(delta_size))
+    return false;
+
+  received_packets_.emplace_back(sequence_number, delta);
+  last_timestamp_ += delta * kDeltaTick;
+  if (include_timestamps_) {
+    size_bytes_ += delta_size;
+  }
+  return true;
+}
+
+const std::vector<TransportFeedback::ReceivedPacket>&
+TransportFeedback::GetReceivedPackets() const {
+  return received_packets_;
+}
+
+void TransportFeedback::ForAllPackets(
+    rtc::FunctionView<void(uint16_t, TimeDelta)> handler) const {
+  TimeDelta delta_since_base = TimeDelta::Zero();
+  auto received_it = received_packets_.begin();
+  const uint16_t last_seq_num = base_seq_no_ + num_seq_no_;
+  for (uint16_t seq_num = base_seq_no_; seq_num != last_seq_num; ++seq_num) {
+    if (received_it != received_packets_.end() &&
+        received_it->sequence_number() == seq_num) {
+      delta_since_base += received_it->delta();
+      handler(seq_num, delta_since_base);
+      ++received_it;
+    } else {
+      handler(seq_num, TimeDelta::PlusInfinity());
+    }
+  }
+  RTC_DCHECK(received_it == received_packets_.end());
+}
+
+uint16_t TransportFeedback::GetBaseSequence() const {
+  return base_seq_no_;
+}
+
+Timestamp TransportFeedback::BaseTime() const {
+  // Add an extra kTimeWrapPeriod to allow add received packets arrived earlier
+  // than the first added packet (and thus allow to record negative deltas)
+  // even when base_time_ticks_ == 0.
+  return Timestamp::Zero() + kTimeWrapPeriod +
+         int64_t{base_time_ticks_} * kBaseTimeTick;
+}
+
+TimeDelta TransportFeedback::GetBaseDelta(Timestamp prev_timestamp) const {
+  TimeDelta delta = BaseTime() - prev_timestamp;
+  // Compensate for wrap around.
+  if ((delta - kTimeWrapPeriod).Abs() < delta.Abs()) {
+    delta -= kTimeWrapPeriod;  // Wrap backwards.
+  } else if ((delta + kTimeWrapPeriod).Abs() < delta.Abs()) {
+    delta += kTimeWrapPeriod;  // Wrap forwards.
+  }
+  return delta;
+}
+
+// De-serialize packet.
+bool TransportFeedback::Parse(const CommonHeader& packet) {
+  RTC_DCHECK_EQ(packet.type(), kPacketType);
+  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);
+  TRACE_EVENT0("webrtc", "TransportFeedback::Parse");
+
+  if (packet.payload_size_bytes() < kMinPayloadSizeBytes) {
+    RTC_LOG(LS_WARNING) << "Buffer too small (" << packet.payload_size_bytes()
+                        << " bytes) to fit a "
+                           "FeedbackPacket. Minimum size = "
+                        << kMinPayloadSizeBytes;
+    return false;
+  }
+
+  const uint8_t* const payload = packet.payload();
+  ParseCommonFeedback(payload);
+
+  base_seq_no_ = ByteReader<uint16_t>::ReadBigEndian(&payload[8]);
+  uint16_t status_count = ByteReader<uint16_t>::ReadBigEndian(&payload[10]);
+  base_time_ticks_ = ByteReader<uint32_t, 3>::ReadBigEndian(&payload[12]);
+  feedback_seq_ = payload[15];
+  Clear();
+  size_t index = 16;
+  const size_t end_index = packet.payload_size_bytes();
+
+  if (status_count == 0) {
+    RTC_LOG(LS_WARNING) << "Empty feedback messages not allowed.";
+    return false;
+  }
+
+  std::vector<uint8_t> delta_sizes;
+  delta_sizes.reserve(status_count);
+  while (delta_sizes.size() < status_count) {
+    if (index + kChunkSizeBytes > end_index) {
+      RTC_LOG(LS_WARNING) << "Buffer overflow while parsing packet.";
+      Clear();
+      return false;
+    }
+
+    uint16_t chunk = ByteReader<uint16_t>::ReadBigEndian(&payload[index]);
+    index += kChunkSizeBytes;
+    encoded_chunks_.push_back(chunk);
+    last_chunk_.Decode(chunk, status_count - delta_sizes.size());
+    last_chunk_.AppendTo(&delta_sizes);
+  }
+  // Last chunk is stored in the `last_chunk_`.
+  encoded_chunks_.pop_back();
+  RTC_DCHECK_EQ(delta_sizes.size(), status_count);
+  num_seq_no_ = status_count;
+
+  uint16_t seq_no = base_seq_no_;
+  size_t recv_delta_size = absl::c_accumulate(delta_sizes, 0);
+
+  // Determine if timestamps, that is, recv_delta are included in the packet.
+  if (end_index >= index + recv_delta_size) {
+    for (size_t delta_size : delta_sizes) {
+      RTC_DCHECK_LE(index + delta_size, end_index);
+      switch (delta_size) {
+        case 0:
+          break;
+        case 1: {
+          int16_t delta = payload[index];
+          received_packets_.emplace_back(seq_no, delta);
+          last_timestamp_ += delta * kDeltaTick;
+          index += delta_size;
+          break;
+        }
+        case 2: {
+          int16_t delta = ByteReader<int16_t>::ReadBigEndian(&payload[index]);
+          received_packets_.emplace_back(seq_no, delta);
+          last_timestamp_ += delta * kDeltaTick;
+          index += delta_size;
+          break;
+        }
+        case 3:
+          Clear();
+          RTC_LOG(LS_WARNING) << "Invalid delta_size for seq_no " << seq_no;
+
+          return false;
+        default:
+          RTC_DCHECK_NOTREACHED();
+          break;
+      }
+      ++seq_no;
+    }
+  } else {
+    // The packet does not contain receive deltas.
+    include_timestamps_ = false;
+    for (size_t delta_size : delta_sizes) {
+      // Use delta sizes to detect if packet was received.
+      if (delta_size > 0) {
+        received_packets_.emplace_back(seq_no, 0);
+      }
+      ++seq_no;
+    }
+  }
+  size_bytes_ = RtcpPacket::kHeaderLength + index;
+  RTC_DCHECK_LE(index, end_index);
+  return true;
+}
+
+std::unique_ptr<TransportFeedback> TransportFeedback::ParseFrom(
+    const uint8_t* buffer,
+    size_t length) {
+  CommonHeader header;
+  if (!header.Parse(buffer, length))
+    return nullptr;
+  if (header.type() != kPacketType || header.fmt() != kFeedbackMessageType)
+    return nullptr;
+  std::unique_ptr<TransportFeedback> parsed(new TransportFeedback);
+  if (!parsed->Parse(header))
+    return nullptr;
+  return parsed;
+}
+
+bool TransportFeedback::IsConsistent() const {
+  size_t packet_size = kTransportFeedbackHeaderSizeBytes;
+  std::vector<DeltaSize> delta_sizes;
+  LastChunk chunk_decoder;
+  for (uint16_t chunk : encoded_chunks_) {
+    chunk_decoder.Decode(chunk, kMaxReportedPackets);
+    chunk_decoder.AppendTo(&delta_sizes);
+    packet_size += kChunkSizeBytes;
+  }
+  if (!last_chunk_.Empty()) {
+    last_chunk_.AppendTo(&delta_sizes);
+    packet_size += kChunkSizeBytes;
+  }
+  if (num_seq_no_ != delta_sizes.size()) {
+    RTC_LOG(LS_ERROR) << delta_sizes.size() << " packets encoded. Expected "
+                      << num_seq_no_;
+    return false;
+  }
+  Timestamp timestamp = BaseTime();
+  auto packet_it = received_packets_.begin();
+  uint16_t seq_no = base_seq_no_;
+  for (DeltaSize delta_size : delta_sizes) {
+    if (delta_size > 0) {
+      if (packet_it == received_packets_.end()) {
+        RTC_LOG(LS_ERROR) << "Failed to find delta for seq_no " << seq_no;
+        return false;
+      }
+      if (packet_it->sequence_number() != seq_no) {
+        RTC_LOG(LS_ERROR) << "Expected to find delta for seq_no " << seq_no
+                          << ". Next delta is for "
+                          << packet_it->sequence_number();
+        return false;
+      }
+      if (delta_size == 1 &&
+          (packet_it->delta_ticks() < 0 || packet_it->delta_ticks() > 0xff)) {
+        RTC_LOG(LS_ERROR) << "Delta " << packet_it->delta_ticks()
+                          << " for seq_no " << seq_no
+                          << " doesn't fit into one byte";
+        return false;
+      }
+      timestamp += packet_it->delta();
+      ++packet_it;
+    }
+    if (include_timestamps_) {
+      packet_size += delta_size;
+    }
+    ++seq_no;
+  }
+  if (packet_it != received_packets_.end()) {
+    RTC_LOG(LS_ERROR) << "Unencoded delta for seq_no "
+                      << packet_it->sequence_number();
+    return false;
+  }
+  if (timestamp != last_timestamp_) {
+    RTC_LOG(LS_ERROR) << "Last timestamp mismatch. Calculated: "
+                      << ToLogString(timestamp)
+                      << ". Saved: " << ToLogString(last_timestamp_);
+    return false;
+  }
+  if (size_bytes_ != packet_size) {
+    RTC_LOG(LS_ERROR) << "Rtcp packet size mismatch. Calculated: "
+                      << packet_size << ". Saved: " << size_bytes_;
+    return false;
+  }
+  return true;
+}
+
+size_t TransportFeedback::BlockLength() const {
+  // Round size_bytes_ up to multiple of 32bits.
+  return (size_bytes_ + 3) & (~static_cast<size_t>(3));
+}
+
+size_t TransportFeedback::PaddingLength() const {
+  return BlockLength() - size_bytes_;
+}
+
+// Serialize packet.
+bool TransportFeedback::Create(uint8_t* packet,
+                               size_t* position,
+                               size_t max_length,
+                               PacketReadyCallback callback) const {
+  if (num_seq_no_ == 0)
+    return false;
+
+  while (*position + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, position, callback))
+      return false;
+  }
+  const size_t position_end = *position + BlockLength();
+  const size_t padding_length = PaddingLength();
+  bool has_padding = padding_length > 0;
+  CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), has_padding,
+               packet, position);
+  CreateCommonFeedback(packet + *position);
+  *position += kCommonFeedbackLength;
+
+  ByteWriter<uint16_t>::WriteBigEndian(&packet[*position], base_seq_no_);
+  *position += 2;
+
+  ByteWriter<uint16_t>::WriteBigEndian(&packet[*position], num_seq_no_);
+  *position += 2;
+
+  ByteWriter<uint32_t, 3>::WriteBigEndian(&packet[*position], base_time_ticks_);
+  *position += 3;
+
+  packet[(*position)++] = feedback_seq_;
+
+  for (uint16_t chunk : encoded_chunks_) {
+    ByteWriter<uint16_t>::WriteBigEndian(&packet[*position], chunk);
+    *position += 2;
+  }
+  if (!last_chunk_.Empty()) {
+    uint16_t chunk = last_chunk_.EncodeLast();
+    ByteWriter<uint16_t>::WriteBigEndian(&packet[*position], chunk);
+    *position += 2;
+  }
+
+  if (include_timestamps_) {
+    for (const auto& received_packet : received_packets_) {
+      int16_t delta = received_packet.delta_ticks();
+      if (delta >= 0 && delta <= 0xFF) {
+        packet[(*position)++] = delta;
+      } else {
+        ByteWriter<int16_t>::WriteBigEndian(&packet[*position], delta);
+        *position += 2;
+      }
+    }
+  }
+
+  if (padding_length > 0) {
+    for (size_t i = 0; i < padding_length - 1; ++i) {
+      packet[(*position)++] = 0;
+    }
+    packet[(*position)++] = padding_length;
+  }
+  RTC_DCHECK_EQ(*position, position_end);
+  return true;
+}
+
+void TransportFeedback::Clear() {
+  num_seq_no_ = 0;
+  last_timestamp_ = BaseTime();
+  received_packets_.clear();
+  all_packets_.clear();
+  encoded_chunks_.clear();
+  last_chunk_.Clear();
+  size_bytes_ = kTransportFeedbackHeaderSizeBytes;
+}
+
+bool TransportFeedback::AddDeltaSize(DeltaSize delta_size) {
+  if (num_seq_no_ == kMaxReportedPackets)
+    return false;
+  size_t add_chunk_size = last_chunk_.Empty() ? kChunkSizeBytes : 0;
+  if (size_bytes_ + delta_size + add_chunk_size > kMaxSizeBytes)
+    return false;
+
+  if (last_chunk_.CanAdd(delta_size)) {
+    size_bytes_ += add_chunk_size;
+    last_chunk_.Add(delta_size);
+    ++num_seq_no_;
+    return true;
+  }
+  if (size_bytes_ + delta_size + kChunkSizeBytes > kMaxSizeBytes)
+    return false;
+
+  encoded_chunks_.push_back(last_chunk_.Emit());
+  size_bytes_ += kChunkSizeBytes;
+  last_chunk_.Add(delta_size);
+  ++num_seq_no_;
+  return true;
+}
+
+bool TransportFeedback::AddMissingPackets(size_t num_missing_packets) {
+  size_t new_num_seq_no = num_seq_no_ + num_missing_packets;
+  if (new_num_seq_no > kMaxReportedPackets) {
+    return false;
+  }
+
+  if (!last_chunk_.Empty()) {
+    while (num_missing_packets > 0 && last_chunk_.CanAdd(0)) {
+      last_chunk_.Add(0);
+      --num_missing_packets;
+    }
+    if (num_missing_packets == 0) {
+      num_seq_no_ = new_num_seq_no;
+      return true;
+    }
+    encoded_chunks_.push_back(last_chunk_.Emit());
+  }
+  RTC_DCHECK(last_chunk_.Empty());
+  size_t full_chunks = num_missing_packets / LastChunk::kMaxRunLengthCapacity;
+  size_t partial_chunk = num_missing_packets % LastChunk::kMaxRunLengthCapacity;
+  size_t num_chunks = full_chunks + (partial_chunk > 0 ? 1 : 0);
+  if (size_bytes_ + kChunkSizeBytes * num_chunks > kMaxSizeBytes) {
+    num_seq_no_ = (new_num_seq_no - num_missing_packets);
+    return false;
+  }
+  size_bytes_ += kChunkSizeBytes * num_chunks;
+  // T = 0, S = 0, run length = kMaxRunLengthCapacity, see EncodeRunLength().
+  encoded_chunks_.insert(encoded_chunks_.end(), full_chunks,
+                         LastChunk::kMaxRunLengthCapacity);
+  last_chunk_.AddMissingPackets(partial_chunk);
+  num_seq_no_ = new_num_seq_no;
+  return true;
+}
+}  // namespace rtcp
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
new file mode 100644
index 0000000000..4d17b54c3a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TRANSPORT_FEEDBACK_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TRANSPORT_FEEDBACK_H_
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "api/function_view.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+
+namespace webrtc {
+namespace rtcp {
+class CommonHeader;
+
+class TransportFeedback : public Rtpfb {
+ public:
+  class ReceivedPacket {
+   public:
+    ReceivedPacket(uint16_t sequence_number, int16_t delta_ticks)
+        : sequence_number_(sequence_number), delta_ticks_(delta_ticks) {}
+    ReceivedPacket(const ReceivedPacket&) = default;
+    ReceivedPacket& operator=(const ReceivedPacket&) = default;
+
+    uint16_t sequence_number() const { return sequence_number_; }
+    int16_t delta_ticks() const { return delta_ticks_; }
+    TimeDelta delta() const { return delta_ticks_ * kDeltaTick; }
+
+   private:
+    uint16_t sequence_number_;
+    int16_t delta_ticks_;
+  };
+  // TODO(sprang): IANA reg?
+  static constexpr uint8_t kFeedbackMessageType = 15;
+  // Convert to multiples of 0.25ms.
+  static constexpr TimeDelta kDeltaTick = TimeDelta::Micros(250);
+  // Maximum number of packets (including missing) TransportFeedback can report.
+  static constexpr size_t kMaxReportedPackets = 0xffff;
+
+  TransportFeedback();
+
+  // If `include_timestamps` is set to false, the created packet will not
+  // contain the receive delta block.
+  explicit TransportFeedback(bool include_timestamps);
+  TransportFeedback(const TransportFeedback&);
+  TransportFeedback(TransportFeedback&&);
+
+  ~TransportFeedback() override;
+
+  void SetBase(uint16_t base_sequence,    // Seq# of first packet in this msg.
+               Timestamp ref_timestamp);  // Reference timestamp for this msg.
+
+  void SetFeedbackSequenceNumber(uint8_t feedback_sequence);
+  // NOTE: This method requires increasing sequence numbers (excepting wraps).
+  bool AddReceivedPacket(uint16_t sequence_number, Timestamp timestamp);
+  const std::vector<ReceivedPacket>& GetReceivedPackets() const;
+
+  // Calls `handler` for all packets this feedback describes.
+  // For received packets pass receieve time as `delta_since_base` since the
+  // `BaseTime()`. For missed packets calls `handler` with `delta_since_base =
+  // PlusInfinity()`.
+  void ForAllPackets(
+      rtc::FunctionView<void(uint16_t sequence_number,
+                             TimeDelta delta_since_base)> handler) const;
+
+  uint16_t GetBaseSequence() const;
+
+  // Returns number of packets (including missing) this feedback describes.
+  size_t GetPacketStatusCount() const { return num_seq_no_; }
+
+  // Get the reference time including any precision loss.
+  Timestamp BaseTime() const;
+
+  // Get the unwrapped delta between current base time and `prev_timestamp`.
+  TimeDelta GetBaseDelta(Timestamp prev_timestamp) const;
+
+  // Does the feedback packet contain timestamp information?
+  bool IncludeTimestamps() const { return include_timestamps_; }
+
+  bool Parse(const CommonHeader& packet);
+  static std::unique_ptr<TransportFeedback> ParseFrom(const uint8_t* buffer,
+                                                      size_t length);
+  // Pre and postcondition for all public methods. Should always return true.
+  // This function is for tests.
+  bool IsConsistent() const;
+
+  size_t BlockLength() const override;
+  size_t PaddingLength() const;
+
+  bool Create(uint8_t* packet,
+              size_t* position,
+              size_t max_length,
+              PacketReadyCallback callback) const override;
+
+ private:
+  // Size in bytes of a delta time in rtcp packet.
+  // Valid values are 0 (packet wasn't received), 1 or 2.
+  using DeltaSize = uint8_t;
+  // Keeps DeltaSizes that can be encoded into single chunk if it is last chunk.
+  class LastChunk {
+   public:
+    using DeltaSize = TransportFeedback::DeltaSize;
+    static constexpr size_t kMaxRunLengthCapacity = 0x1fff;
+
+    LastChunk();
+
+    bool Empty() const;
+    void Clear();
+    // Return if delta sizes still can be encoded into single chunk with added
+    // `delta_size`.
+    bool CanAdd(DeltaSize delta_size) const;
+    // Add `delta_size`, assumes `CanAdd(delta_size)`,
+    void Add(DeltaSize delta_size);
+    // Equivalent to calling Add(0) `num_missing` times. Assumes `Empty()`.
+    void AddMissingPackets(size_t num_missing);
+
+    // Encode chunk as large as possible removing encoded delta sizes.
+    // Assume CanAdd() == false for some valid delta_size.
+    uint16_t Emit();
+    // Encode all stored delta_sizes into single chunk, pad with 0s if needed.
+    uint16_t EncodeLast() const;
+
+    // Decode up to `max_size` delta sizes from `chunk`.
+    void Decode(uint16_t chunk, size_t max_size);
+    // Appends content of the Lastchunk to `deltas`.
+    void AppendTo(std::vector<DeltaSize>* deltas) const;
+
+   private:
+    static constexpr size_t kMaxOneBitCapacity = 14;
+    static constexpr size_t kMaxTwoBitCapacity = 7;
+    static constexpr size_t kMaxVectorCapacity = kMaxOneBitCapacity;
+    static constexpr DeltaSize kLarge = 2;
+
+    uint16_t EncodeOneBit() const;
+    void DecodeOneBit(uint16_t chunk, size_t max_size);
+
+    uint16_t EncodeTwoBit(size_t size) const;
+    void DecodeTwoBit(uint16_t chunk, size_t max_size);
+
+    uint16_t EncodeRunLength() const;
+    void DecodeRunLength(uint16_t chunk, size_t max_size);
+
+    std::array<DeltaSize, kMaxVectorCapacity> delta_sizes_;
+    size_t size_;
+    bool all_same_;
+    bool has_large_delta_;
+  };
+
+  // Reset packet to consistent empty state.
+  void Clear();
+
+  bool AddDeltaSize(DeltaSize delta_size);
+  // Adds `num_missing_packets` deltas of size 0.
+  bool AddMissingPackets(size_t num_missing_packets);
+
+  uint16_t base_seq_no_;
+  uint16_t num_seq_no_;
+  uint32_t base_time_ticks_;
+  uint8_t feedback_seq_;
+  bool include_timestamps_;
+
+  Timestamp last_timestamp_;
+  std::vector<ReceivedPacket> received_packets_;
+  std::vector<ReceivedPacket> all_packets_;
+  // All but last encoded packet chunks.
+  std::vector<uint16_t> encoded_chunks_;
+  LastChunk last_chunk_;
+  size_t size_bytes_;
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TRANSPORT_FEEDBACK_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
new file mode 100644
index 0000000000..4248a4d3ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
@@ -0,0 +1,667 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using rtcp::TransportFeedback;
+using ::testing::AllOf;
+using ::testing::Each;
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::InSequence;
+using ::testing::MockFunction;
+using ::testing::Ne;
+using ::testing::Property;
+using ::testing::SizeIs;
+
+constexpr int kHeaderSize = 20;
+constexpr int kStatusChunkSize = 2;
+constexpr int kSmallDeltaSize = 1;
+constexpr int kLargeDeltaSize = 2;
+
+constexpr TimeDelta kDeltaLimit = 0xFF * TransportFeedback::kDeltaTick;
+constexpr TimeDelta kBaseTimeTick = TransportFeedback::kDeltaTick * (1 << 8);
+constexpr TimeDelta kBaseTimeWrapPeriod = kBaseTimeTick * (1 << 24);
+
+MATCHER_P2(Near, value, max_abs_error, "") {
+  return value - max_abs_error <= arg && arg <= value + max_abs_error;
+}
+
+MATCHER(IsValidFeedback, "") {
+  rtcp::CommonHeader rtcp_header;
+  TransportFeedback feedback;
+  return rtcp_header.Parse(std::data(arg), std::size(arg)) &&
+         rtcp_header.type() == TransportFeedback::kPacketType &&
+         rtcp_header.fmt() == TransportFeedback::kFeedbackMessageType &&
+         feedback.Parse(rtcp_header);
+}
+
+TransportFeedback Parse(rtc::ArrayView<const uint8_t> buffer) {
+  rtcp::CommonHeader header;
+  RTC_DCHECK(header.Parse(buffer.data(), buffer.size()));
+  RTC_DCHECK_EQ(header.type(), TransportFeedback::kPacketType);
+  RTC_DCHECK_EQ(header.fmt(), TransportFeedback::kFeedbackMessageType);
+  TransportFeedback feedback;
+  RTC_DCHECK(feedback.Parse(header));
+  return feedback;
+}
+
+class FeedbackTester {
+ public:
+  FeedbackTester() : FeedbackTester(true) {}
+  explicit FeedbackTester(bool include_timestamps)
+      : expected_size_(kAnySize),
+        default_delta_(TransportFeedback::kDeltaTick * 4),
+        include_timestamps_(include_timestamps) {}
+
+  void WithExpectedSize(size_t expected_size) {
+    expected_size_ = expected_size;
+  }
+
+  void WithDefaultDelta(TimeDelta delta) { default_delta_ = delta; }
+
+  void WithInput(rtc::ArrayView<const uint16_t> received_seq,
+                 rtc::ArrayView<const Timestamp> received_ts = {}) {
+    std::vector<Timestamp> temp_timestamps;
+    if (received_ts.empty()) {
+      temp_timestamps = GenerateReceiveTimestamps(received_seq);
+      received_ts = temp_timestamps;
+    }
+    RTC_DCHECK_EQ(received_seq.size(), received_ts.size());
+
+    expected_deltas_.clear();
+    feedback_.emplace(include_timestamps_);
+    feedback_->SetBase(received_seq[0], received_ts[0]);
+    ASSERT_TRUE(feedback_->IsConsistent());
+    // First delta is special: it doesn't represent the delta between two times,
+    // but a compensation for the reduced precision of the base time.
+    EXPECT_TRUE(feedback_->AddReceivedPacket(received_seq[0], received_ts[0]));
+    // GetBaseDelta suppose to return balanced diff between base time of the new
+    // feedback message (stored internally) and base time of the old feedback
+    // message (passed as parameter), but first delta is the difference between
+    // 1st timestamp (passed as parameter) and base time (stored internally),
+    // thus to get the first delta need to negate whatever GetBaseDelta returns.
+    expected_deltas_.push_back(-feedback_->GetBaseDelta(received_ts[0]));
+
+    for (size_t i = 1; i < received_ts.size(); ++i) {
+      EXPECT_TRUE(
+          feedback_->AddReceivedPacket(received_seq[i], received_ts[i]));
+      expected_deltas_.push_back(received_ts[i] - received_ts[i - 1]);
+    }
+    ASSERT_TRUE(feedback_->IsConsistent());
+    expected_seq_.assign(received_seq.begin(), received_seq.end());
+  }
+
+  void VerifyPacket() {
+    ASSERT_TRUE(feedback_->IsConsistent());
+    serialized_ = feedback_->Build();
+    VerifyInternal();
+
+    feedback_.emplace(Parse(serialized_));
+    ASSERT_TRUE(feedback_->IsConsistent());
+    EXPECT_EQ(include_timestamps_, feedback_->IncludeTimestamps());
+    VerifyInternal();
+  }
+
+  static constexpr size_t kAnySize = static_cast<size_t>(0) - 1;
+
+ private:
+  void VerifyInternal() {
+    if (expected_size_ != kAnySize) {
+      // Round up to whole 32-bit words.
+      size_t expected_size_words = (expected_size_ + 3) / 4;
+      size_t expected_size_bytes = expected_size_words * 4;
+      EXPECT_EQ(expected_size_bytes, serialized_.size());
+    }
+
+    std::vector<uint16_t> actual_seq_nos;
+    std::vector<TimeDelta> actual_deltas;
+    for (const auto& packet : feedback_->GetReceivedPackets()) {
+      actual_seq_nos.push_back(packet.sequence_number());
+      actual_deltas.push_back(packet.delta());
+    }
+    EXPECT_THAT(actual_seq_nos, ElementsAreArray(expected_seq_));
+    if (include_timestamps_) {
+      EXPECT_THAT(actual_deltas, ElementsAreArray(expected_deltas_));
+    }
+  }
+
+  std::vector<Timestamp> GenerateReceiveTimestamps(
+      rtc::ArrayView<const uint16_t> seq_nums) {
+    RTC_DCHECK(!seq_nums.empty());
+    uint16_t last_seq = seq_nums[0];
+    Timestamp time = Timestamp::Zero();
+    std::vector<Timestamp> result;
+
+    for (uint16_t seq : seq_nums) {
+      if (seq < last_seq)
+        time += 0x10000 * default_delta_;
+      last_seq = seq;
+
+      result.push_back(time + last_seq * default_delta_);
+    }
+    return result;
+  }
+
+  std::vector<uint16_t> expected_seq_;
+  std::vector<TimeDelta> expected_deltas_;
+  size_t expected_size_;
+  TimeDelta default_delta_;
+  absl::optional<TransportFeedback> feedback_;
+  rtc::Buffer serialized_;
+  bool include_timestamps_;
+};
+
+// The following tests use FeedbackTester that simulates received packets as
+// specified by the parameters `received_seq[]` and `received_ts[]` (optional).
+// The following is verified in these tests:
+// - Expected size of serialized packet.
+// - Expected sequence numbers and receive deltas.
+// - Sequence numbers and receive deltas are persistent after serialization
+//   followed by parsing.
+// - The internal state of a feedback packet is consistent.
+TEST(RtcpPacketTest, TransportFeedbackOneBitVector) {
+  const uint16_t kReceived[] = {1, 2, 7, 8, 9, 10, 13};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackOneBitVectorNoRecvDelta) {
+  const uint16_t kReceived[] = {1, 2, 7, 8, 9, 10, 13};
+  const size_t kExpectedSizeBytes = kHeaderSize + kStatusChunkSize;
+
+  FeedbackTester test(/*include_timestamps=*/false);
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackFullOneBitVector) {
+  const uint16_t kReceived[] = {1, 2, 7, 8, 9, 10, 13, 14};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackOneBitVectorWrapReceived) {
+  const uint16_t kMax = 0xFFFF;
+  const uint16_t kReceived[] = {kMax - 2, kMax - 1, kMax, 0, 1, 2};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackOneBitVectorWrapMissing) {
+  const uint16_t kMax = 0xFFFF;
+  const uint16_t kReceived[] = {kMax - 2, kMax - 1, 1, 2};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackTwoBitVector) {
+  const uint16_t kReceived[] = {1, 2, 6, 7};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (kLength * kLargeDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithDefaultDelta(kDeltaLimit + TransportFeedback::kDeltaTick);
+  test.WithInput(kReceived);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackTwoBitVectorFull) {
+  const uint16_t kReceived[] = {1, 2, 6, 7, 8};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + (2 * kStatusChunkSize) + (kLength * kLargeDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithDefaultDelta(kDeltaLimit + TransportFeedback::kDeltaTick);
+  test.WithInput(kReceived);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackWithLargeBaseTimeIsConsistent) {
+  TransportFeedback tb;
+  constexpr Timestamp kTimestamp =
+      Timestamp::Zero() + int64_t{0x7fff'ffff} * TransportFeedback::kDeltaTick;
+  tb.SetBase(/*base_sequence=*/0, /*ref_timestamp=*/kTimestamp);
+  tb.AddReceivedPacket(/*base_sequence=*/0, /*ref_timestamp=*/kTimestamp);
+  EXPECT_TRUE(tb.IsConsistent());
+}
+
+TEST(RtcpPacketTest, TransportFeedbackLargeAndNegativeDeltas) {
+  const uint16_t kReceived[] = {1, 2, 6, 7, 8};
+  const Timestamp kReceiveTimes[] = {
+      Timestamp::Millis(2), Timestamp::Millis(1), Timestamp::Millis(4),
+      Timestamp::Millis(3),
+      Timestamp::Millis(3) + TransportFeedback::kDeltaTick * (1 << 8)};
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + (3 * kLargeDeltaSize) + kSmallDeltaSize;
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackMaxRle) {
+  // Expected chunks created:
+  // * 1-bit vector chunk (1xreceived + 13xdropped)
+  // * RLE chunk of max length for dropped symbol
+  // * 1-bit vector chunk (1xreceived + 13xdropped)
+
+  const size_t kPacketCount = (1 << 13) - 1 + 14;
+  const uint16_t kReceived[] = {0, kPacketCount};
+  const Timestamp kReceiveTimes[] = {Timestamp::Millis(1),
+                                     Timestamp::Millis(2)};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + (3 * kStatusChunkSize) + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackMinRle) {
+  // Expected chunks created:
+  // * 1-bit vector chunk (1xreceived + 13xdropped)
+  // * RLE chunk of length 15 for dropped symbol
+  // * 1-bit vector chunk (1xreceived + 13xdropped)
+
+  const uint16_t kReceived[] = {0, (14 * 2) + 1};
+  const Timestamp kReceiveTimes[] = {Timestamp::Millis(1),
+                                     Timestamp::Millis(2)};
+  const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + (3 * kStatusChunkSize) + (kLength * kSmallDeltaSize);
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackOneToTwoBitVector) {
+  const size_t kTwoBitVectorCapacity = 7;
+  const uint16_t kReceived[] = {0, kTwoBitVectorCapacity - 1};
+  const Timestamp kReceiveTimes[] = {
+      Timestamp::Zero(),
+      Timestamp::Zero() + kDeltaLimit + TransportFeedback::kDeltaTick};
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + kSmallDeltaSize + kLargeDeltaSize;
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackOneToTwoBitVectorSimpleSplit) {
+  const size_t kTwoBitVectorCapacity = 7;
+  const uint16_t kReceived[] = {0, kTwoBitVectorCapacity};
+  const Timestamp kReceiveTimes[] = {
+      Timestamp::Zero(),
+      Timestamp::Zero() + kDeltaLimit + TransportFeedback::kDeltaTick};
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + (kStatusChunkSize * 2) + kSmallDeltaSize + kLargeDeltaSize;
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, kReceiveTimes);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackOneToTwoBitVectorSplit) {
+  // With received small delta = S, received large delta = L, use input
+  // SSSSSSSSLSSSSSSSSSSSS. This will cause a 1:2 split at the L.
+  // After split there will be two symbols in symbol_vec: SL.
+
+  const TimeDelta kLargeDelta = TransportFeedback::kDeltaTick * (1 << 8);
+  const size_t kNumPackets = (3 * 7) + 1;
+  const size_t kExpectedSizeBytes = kHeaderSize + (kStatusChunkSize * 3) +
+                                    (kSmallDeltaSize * (kNumPackets - 1)) +
+                                    (kLargeDeltaSize * 1);
+
+  uint16_t kReceived[kNumPackets];
+  for (size_t i = 0; i < kNumPackets; ++i)
+    kReceived[i] = i;
+
+  std::vector<Timestamp> receive_times;
+  receive_times.reserve(kNumPackets);
+  receive_times.push_back(Timestamp::Millis(1));
+  for (size_t i = 1; i < kNumPackets; ++i) {
+    TimeDelta delta = (i == 8) ? kLargeDelta : TimeDelta::Millis(1);
+    receive_times.push_back(receive_times.back() + delta);
+  }
+
+  FeedbackTester test;
+  test.WithExpectedSize(kExpectedSizeBytes);
+  test.WithInput(kReceived, receive_times);
+  test.VerifyPacket();
+}
+
+TEST(RtcpPacketTest, TransportFeedbackAliasing) {
+  TransportFeedback feedback;
+  feedback.SetBase(0, Timestamp::Zero());
+
+  const int kSamples = 100;
+  const TimeDelta kTooSmallDelta = TransportFeedback::kDeltaTick / 3;
+
+  for (int i = 0; i < kSamples; ++i)
+    feedback.AddReceivedPacket(i, Timestamp::Zero() + i * kTooSmallDelta);
+
+  feedback.Build();
+
+  TimeDelta accumulated_delta = TimeDelta::Zero();
+  int num_samples = 0;
+  for (const auto& packet : feedback.GetReceivedPackets()) {
+    accumulated_delta += packet.delta();
+    TimeDelta expected_time = num_samples * kTooSmallDelta;
+    ++num_samples;
+
+    EXPECT_THAT(accumulated_delta,
+                Near(expected_time, TransportFeedback::kDeltaTick / 2));
+  }
+}
+
+TEST(RtcpPacketTest, TransportFeedbackLimits) {
+  // Sequence number wrap above 0x8000.
+  std::unique_ptr<TransportFeedback> packet(new TransportFeedback());
+  packet->SetBase(0, Timestamp::Zero());
+  EXPECT_TRUE(packet->AddReceivedPacket(0x0, Timestamp::Zero()));
+  EXPECT_TRUE(packet->AddReceivedPacket(0x8000, Timestamp::Millis(1)));
+
+  packet.reset(new TransportFeedback());
+  packet->SetBase(0, Timestamp::Zero());
+  EXPECT_TRUE(packet->AddReceivedPacket(0x0, Timestamp::Zero()));
+  EXPECT_FALSE(packet->AddReceivedPacket(0x8000 + 1, Timestamp::Millis(1)));
+
+  // Packet status count max 0xFFFF.
+  packet.reset(new TransportFeedback());
+  packet->SetBase(0, Timestamp::Zero());
+  EXPECT_TRUE(packet->AddReceivedPacket(0x0, Timestamp::Zero()));
+  EXPECT_TRUE(packet->AddReceivedPacket(0x8000, Timestamp::Millis(1)));
+  EXPECT_TRUE(packet->AddReceivedPacket(0xFFFE, Timestamp::Millis(2)));
+  EXPECT_FALSE(packet->AddReceivedPacket(0xFFFF, Timestamp::Millis(3)));
+
+  // Too large delta.
+  packet.reset(new TransportFeedback());
+  packet->SetBase(0, Timestamp::Zero());
+  TimeDelta kMaxPositiveTimeDelta =
+      std::numeric_limits<int16_t>::max() * TransportFeedback::kDeltaTick;
+  EXPECT_FALSE(packet->AddReceivedPacket(1, Timestamp::Zero() +
+                                                kMaxPositiveTimeDelta +
+                                                TransportFeedback::kDeltaTick));
+  EXPECT_TRUE(
+      packet->AddReceivedPacket(1, Timestamp::Zero() + kMaxPositiveTimeDelta));
+
+  // Too large negative delta.
+  packet.reset(new TransportFeedback());
+  TimeDelta kMaxNegativeTimeDelta =
+      std::numeric_limits<int16_t>::min() * TransportFeedback::kDeltaTick;
+  // Use larger base time to avoid kBaseTime + kNegativeDelta to be negative.
+  Timestamp kBaseTime = Timestamp::Seconds(1'000'000);
+  packet->SetBase(0, kBaseTime);
+  EXPECT_FALSE(packet->AddReceivedPacket(
+      1, kBaseTime + kMaxNegativeTimeDelta - TransportFeedback::kDeltaTick));
+  EXPECT_TRUE(packet->AddReceivedPacket(1, kBaseTime + kMaxNegativeTimeDelta));
+
+  // TODO(sprang): Once we support max length lower than RTCP length limit,
+  // add back test for max size in bytes.
+}
+
+TEST(RtcpPacketTest, BaseTimeIsConsistentAcrossMultiplePackets) {
+  constexpr Timestamp kMaxBaseTime =
+      Timestamp::Zero() + kBaseTimeWrapPeriod - kBaseTimeTick;
+
+  TransportFeedback packet1;
+  packet1.SetBase(0, kMaxBaseTime);
+  packet1.AddReceivedPacket(0, kMaxBaseTime);
+  // Build and parse packet to simulate sending it over the wire.
+  TransportFeedback parsed_packet1 = Parse(packet1.Build());
+
+  TransportFeedback packet2;
+  packet2.SetBase(1, kMaxBaseTime + kBaseTimeTick);
+  packet2.AddReceivedPacket(1, kMaxBaseTime + kBaseTimeTick);
+  TransportFeedback parsed_packet2 = Parse(packet2.Build());
+
+  EXPECT_EQ(parsed_packet2.GetBaseDelta(parsed_packet1.BaseTime()),
+            kBaseTimeTick);
+}
+
+TEST(RtcpPacketTest, SupportsMaximumNumberOfNegativeDeltas) {
+  TransportFeedback feedback;
+  // Use large base time to avoid hitting zero limit while filling the feedback,
+  // but use multiple of kBaseTimeWrapPeriod to hit edge case where base time
+  // is recorded as zero in the raw rtcp packet.
+  Timestamp time = Timestamp::Zero() + 1'000 * kBaseTimeWrapPeriod;
+  feedback.SetBase(0, time);
+  static constexpr TimeDelta kMinDelta =
+      TransportFeedback::kDeltaTick * std::numeric_limits<int16_t>::min();
+  uint16_t num_received_rtp_packets = 0;
+  time += kMinDelta;
+  while (feedback.AddReceivedPacket(++num_received_rtp_packets, time)) {
+    ASSERT_GE(time, Timestamp::Zero() - kMinDelta);
+    time += kMinDelta;
+  }
+  // Subtract one last packet that failed to add.
+  --num_received_rtp_packets;
+  EXPECT_TRUE(feedback.IsConsistent());
+
+  TransportFeedback parsed = Parse(feedback.Build());
+  EXPECT_EQ(parsed.GetReceivedPackets().size(), num_received_rtp_packets);
+  EXPECT_THAT(parsed.GetReceivedPackets(),
+              AllOf(SizeIs(num_received_rtp_packets),
+                    Each(Property(&TransportFeedback::ReceivedPacket::delta,
+                                  Eq(kMinDelta)))));
+  EXPECT_GE(parsed.BaseTime(),
+            Timestamp::Zero() - kMinDelta * num_received_rtp_packets);
+}
+
+TEST(RtcpPacketTest, TransportFeedbackPadding) {
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + kSmallDeltaSize;
+  const size_t kExpectedSizeWords = (kExpectedSizeBytes + 3) / 4;
+  const size_t kExpectedPaddingSizeBytes =
+      4 * kExpectedSizeWords - kExpectedSizeBytes;
+
+  TransportFeedback feedback;
+  feedback.SetBase(0, Timestamp::Zero());
+  EXPECT_TRUE(feedback.AddReceivedPacket(0, Timestamp::Zero()));
+
+  rtc::Buffer packet = feedback.Build();
+  EXPECT_EQ(kExpectedSizeWords * 4, packet.size());
+  ASSERT_GT(kExpectedSizeWords * 4, kExpectedSizeBytes);
+  for (size_t i = kExpectedSizeBytes; i < (kExpectedSizeWords * 4 - 1); ++i)
+    EXPECT_EQ(0u, packet[i]);
+
+  EXPECT_EQ(kExpectedPaddingSizeBytes, packet[kExpectedSizeWords * 4 - 1]);
+
+  // Modify packet by adding 4 bytes of padding at the end. Not currently used
+  // when we're sending, but need to be able to handle it when receiving.
+
+  const int kPaddingBytes = 4;
+  const size_t kExpectedSizeWithPadding =
+      (kExpectedSizeWords * 4) + kPaddingBytes;
+  uint8_t mod_buffer[kExpectedSizeWithPadding];
+  memcpy(mod_buffer, packet.data(), kExpectedSizeWords * 4);
+  memset(&mod_buffer[kExpectedSizeWords * 4], 0, kPaddingBytes - 1);
+  mod_buffer[kExpectedSizeWithPadding - 1] =
+      kPaddingBytes + kExpectedPaddingSizeBytes;
+  const uint8_t padding_flag = 1 << 5;
+  mod_buffer[0] |= padding_flag;
+  ByteWriter<uint16_t>::WriteBigEndian(
+      &mod_buffer[2], ByteReader<uint16_t>::ReadBigEndian(&mod_buffer[2]) +
+                          ((kPaddingBytes + 3) / 4));
+
+  EXPECT_THAT(mod_buffer, IsValidFeedback());
+}
+
+TEST(RtcpPacketTest, TransportFeedbackPaddingBackwardsCompatibility) {
+  const size_t kExpectedSizeBytes =
+      kHeaderSize + kStatusChunkSize + kSmallDeltaSize;
+  const size_t kExpectedSizeWords = (kExpectedSizeBytes + 3) / 4;
+  const size_t kExpectedPaddingSizeBytes =
+      4 * kExpectedSizeWords - kExpectedSizeBytes;
+
+  TransportFeedback feedback;
+  feedback.SetBase(0, Timestamp::Zero());
+  EXPECT_TRUE(feedback.AddReceivedPacket(0, Timestamp::Zero()));
+
+  rtc::Buffer packet = feedback.Build();
+  EXPECT_EQ(kExpectedSizeWords * 4, packet.size());
+  ASSERT_GT(kExpectedSizeWords * 4, kExpectedSizeBytes);
+  for (size_t i = kExpectedSizeBytes; i < (kExpectedSizeWords * 4 - 1); ++i)
+    EXPECT_EQ(0u, packet[i]);
+
+  EXPECT_GT(kExpectedPaddingSizeBytes, 0u);
+  EXPECT_EQ(kExpectedPaddingSizeBytes, packet[kExpectedSizeWords * 4 - 1]);
+
+  // Modify packet by removing padding bit and writing zero at the last padding
+  // byte to verify that we can parse packets from old clients, where zero
+  // padding of up to three bytes was used without the padding bit being set.
+  uint8_t mod_buffer[kExpectedSizeWords * 4];
+  memcpy(mod_buffer, packet.data(), kExpectedSizeWords * 4);
+  mod_buffer[kExpectedSizeWords * 4 - 1] = 0;
+  const uint8_t padding_flag = 1 << 5;
+  mod_buffer[0] &= ~padding_flag;  // Unset padding flag.
+
+  EXPECT_THAT(mod_buffer, IsValidFeedback());
+}
+
+TEST(RtcpPacketTest, TransportFeedbackCorrectlySplitsVectorChunks) {
+  const int kOneBitVectorCapacity = 14;
+  const TimeDelta kLargeTimeDelta = TransportFeedback::kDeltaTick * (1 << 8);
+
+  // Test that a number of small deltas followed by a large delta results in a
+  // correct split into multiple chunks, as needed.
+
+  for (int deltas = 0; deltas <= kOneBitVectorCapacity + 1; ++deltas) {
+    TransportFeedback feedback;
+    feedback.SetBase(0, Timestamp::Zero());
+    for (int i = 0; i < deltas; ++i)
+      feedback.AddReceivedPacket(i, Timestamp::Millis(i));
+    feedback.AddReceivedPacket(deltas,
+                               Timestamp::Millis(deltas) + kLargeTimeDelta);
+
+    EXPECT_THAT(feedback.Build(), IsValidFeedback());
+  }
+}
+
+TEST(RtcpPacketTest, TransportFeedbackMoveConstructor) {
+  const int kSamples = 100;
+  const uint16_t kBaseSeqNo = 7531;
+  const Timestamp kBaseTimestamp = Timestamp::Micros(123'456'789);
+  const uint8_t kFeedbackSeqNo = 90;
+
+  TransportFeedback feedback;
+  feedback.SetBase(kBaseSeqNo, kBaseTimestamp);
+  feedback.SetFeedbackSequenceNumber(kFeedbackSeqNo);
+  for (int i = 0; i < kSamples; ++i) {
+    feedback.AddReceivedPacket(
+        kBaseSeqNo + i, kBaseTimestamp + i * TransportFeedback::kDeltaTick);
+  }
+  EXPECT_TRUE(feedback.IsConsistent());
+
+  TransportFeedback feedback_copy(feedback);
+  EXPECT_TRUE(feedback_copy.IsConsistent());
+  EXPECT_TRUE(feedback.IsConsistent());
+  EXPECT_EQ(feedback_copy.Build(), feedback.Build());
+
+  TransportFeedback moved(std::move(feedback));
+  EXPECT_TRUE(moved.IsConsistent());
+  EXPECT_TRUE(feedback.IsConsistent());
+  EXPECT_EQ(moved.Build(), feedback_copy.Build());
+}
+
+TEST(TransportFeedbackTest, ReportsMissingPackets) {
+  const uint16_t kBaseSeqNo = 1000;
+  const Timestamp kBaseTimestamp = Timestamp::Millis(10);
+  const uint8_t kFeedbackSeqNo = 90;
+  TransportFeedback feedback_builder(/*include_timestamps*/ true);
+  feedback_builder.SetBase(kBaseSeqNo, kBaseTimestamp);
+  feedback_builder.SetFeedbackSequenceNumber(kFeedbackSeqNo);
+  feedback_builder.AddReceivedPacket(kBaseSeqNo + 0, kBaseTimestamp);
+  // Packet losses indicated by jump in sequence number.
+  feedback_builder.AddReceivedPacket(kBaseSeqNo + 3,
+                                     kBaseTimestamp + TimeDelta::Millis(2));
+
+  MockFunction<void(uint16_t, TimeDelta)> handler;
+  InSequence s;
+  EXPECT_CALL(handler, Call(kBaseSeqNo + 0, Ne(TimeDelta::PlusInfinity())));
+  EXPECT_CALL(handler, Call(kBaseSeqNo + 1, TimeDelta::PlusInfinity()));
+  EXPECT_CALL(handler, Call(kBaseSeqNo + 2, TimeDelta::PlusInfinity()));
+  EXPECT_CALL(handler, Call(kBaseSeqNo + 3, Ne(TimeDelta::PlusInfinity())));
+  Parse(feedback_builder.Build()).ForAllPackets(handler.AsStdFunction());
+}
+
+TEST(TransportFeedbackTest, ReportsMissingPacketsWithoutTimestamps) {
+  const uint16_t kBaseSeqNo = 1000;
+  const uint8_t kFeedbackSeqNo = 90;
+  TransportFeedback feedback_builder(/*include_timestamps*/ false);
+  feedback_builder.SetBase(kBaseSeqNo, Timestamp::Millis(10));
+  feedback_builder.SetFeedbackSequenceNumber(kFeedbackSeqNo);
+  feedback_builder.AddReceivedPacket(kBaseSeqNo + 0, Timestamp::Zero());
+  // Packet losses indicated by jump in sequence number.
+  feedback_builder.AddReceivedPacket(kBaseSeqNo + 3, Timestamp::Zero());
+
+  MockFunction<void(uint16_t, TimeDelta)> handler;
+  InSequence s;
+  EXPECT_CALL(handler, Call(kBaseSeqNo + 0, Ne(TimeDelta::PlusInfinity())));
+  EXPECT_CALL(handler, Call(kBaseSeqNo + 1, TimeDelta::PlusInfinity()));
+  EXPECT_CALL(handler, Call(kBaseSeqNo + 2, TimeDelta::PlusInfinity()));
+  EXPECT_CALL(handler, Call(kBaseSeqNo + 3, Ne(TimeDelta::PlusInfinity())));
+  Parse(feedback_builder.Build()).ForAllPackets(handler.AsStdFunction());
+}
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
new file mode 100644
index 0000000000..dccd1354a9
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace {
+
+using ::testing::_;
+using ::testing::MockFunction;
+using ::webrtc::rtcp::ReceiverReport;
+using ::webrtc::rtcp::ReportBlock;
+
+const uint32_t kSenderSsrc = 0x12345678;
+
+TEST(RtcpPacketTest, BuildWithTooSmallBuffer) {
+  ReportBlock rb;
+  ReceiverReport rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+  EXPECT_TRUE(rr.AddReportBlock(rb));
+
+  const size_t kRrLength = 8;
+  const size_t kReportBlockLength = 24;
+
+  // No packet.
+  MockFunction<void(rtc::ArrayView<const uint8_t>)> callback;
+  EXPECT_CALL(callback, Call(_)).Times(0);
+  const size_t kBufferSize = kRrLength + kReportBlockLength - 1;
+  EXPECT_FALSE(rr.Build(kBufferSize, callback.AsStdFunction()));
+}
+
+}  // namespace
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
new file mode 100644
index 0000000000..936750c263
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -0,0 +1,1292 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_receiver.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <limits>
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_bitrate_allocator.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "modules/rtp_rtcp/source/tmmbr_help.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace {
+
+using rtcp::CommonHeader;
+using rtcp::ReportBlock;
+
+// The number of RTCP time intervals needed to trigger a timeout.
+const int kRrTimeoutIntervals = 3;
+
+const int64_t kTmmbrTimeoutIntervalMs = 5 * 5000;
+
+const int64_t kMaxWarningLogIntervalMs = 10000;
+const int64_t kRtcpMinFrameLengthMs = 17;
+
+// Maximum number of received RRTRs that will be stored.
+const size_t kMaxNumberOfStoredRrtrs = 300;
+
+constexpr TimeDelta kDefaultVideoReportInterval = TimeDelta::Seconds(1);
+constexpr TimeDelta kDefaultAudioReportInterval = TimeDelta::Seconds(5);
+
+// Returns true if the `timestamp` has exceeded the |interval *
+// kRrTimeoutIntervals| period and was reset (set to PlusInfinity()). Returns
+// false if the timer was either already reset or if it has not expired.
+bool ResetTimestampIfExpired(const Timestamp now,
+                             Timestamp& timestamp,
+                             TimeDelta interval) {
+  if (timestamp.IsInfinite() ||
+      now <= timestamp + interval * kRrTimeoutIntervals) {
+    return false;
+  }
+
+  timestamp = Timestamp::PlusInfinity();
+  return true;
+}
+
+}  // namespace
+
+constexpr size_t RTCPReceiver::RegisteredSsrcs::kMediaSsrcIndex;
+constexpr size_t RTCPReceiver::RegisteredSsrcs::kMaxSsrcs;
+
+RTCPReceiver::RegisteredSsrcs::RegisteredSsrcs(
+    bool disable_sequence_checker,
+    const RtpRtcpInterface::Configuration& config)
+    : packet_sequence_checker_(disable_sequence_checker) {
+  packet_sequence_checker_.Detach();
+  ssrcs_.push_back(config.local_media_ssrc);
+  if (config.rtx_send_ssrc) {
+    ssrcs_.push_back(*config.rtx_send_ssrc);
+  }
+  if (config.fec_generator) {
+    absl::optional<uint32_t> flexfec_ssrc = config.fec_generator->FecSsrc();
+    if (flexfec_ssrc) {
+      ssrcs_.push_back(*flexfec_ssrc);
+    }
+  }
+  // Ensure that the RegisteredSsrcs can inline the SSRCs.
+  RTC_DCHECK_LE(ssrcs_.size(), RTCPReceiver::RegisteredSsrcs::kMaxSsrcs);
+}
+
+bool RTCPReceiver::RegisteredSsrcs::contains(uint32_t ssrc) const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return absl::c_linear_search(ssrcs_, ssrc);
+}
+
+uint32_t RTCPReceiver::RegisteredSsrcs::media_ssrc() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return ssrcs_[kMediaSsrcIndex];
+}
+
+void RTCPReceiver::RegisteredSsrcs::set_media_ssrc(uint32_t ssrc) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  ssrcs_[kMediaSsrcIndex] = ssrc;
+}
+
+struct RTCPReceiver::PacketInformation {
+  uint32_t packet_type_flags = 0;  // RTCPPacketTypeFlags bit field.
+
+  uint32_t remote_ssrc = 0;
+  std::vector<uint16_t> nack_sequence_numbers;
+  // TODO(hbos): Remove `report_blocks` in favor of `report_block_datas`.
+  ReportBlockList report_blocks;
+  std::vector<ReportBlockData> report_block_datas;
+  int64_t rtt_ms = 0;
+  uint32_t receiver_estimated_max_bitrate_bps = 0;
+  std::unique_ptr<rtcp::TransportFeedback> transport_feedback;
+  absl::optional<VideoBitrateAllocation> target_bitrate_allocation;
+  absl::optional<NetworkStateEstimate> network_state_estimate;
+  std::unique_ptr<rtcp::LossNotification> loss_notification;
+};
+
+RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
+                           ModuleRtpRtcpImpl2* owner)
+    : clock_(config.clock),
+      receiver_only_(config.receiver_only),
+      rtp_rtcp_(owner),
+      registered_ssrcs_(false, config),
+      rtcp_bandwidth_observer_(config.bandwidth_callback),
+      rtcp_event_observer_(config.rtcp_event_observer),
+      rtcp_intra_frame_observer_(config.intra_frame_callback),
+      rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
+      network_state_estimate_observer_(config.network_state_estimate_observer),
+      transport_feedback_observer_(config.transport_feedback_callback),
+      bitrate_allocation_observer_(config.bitrate_allocation_observer),
+      report_interval_(config.rtcp_report_interval_ms > 0
+                           ? TimeDelta::Millis(config.rtcp_report_interval_ms)
+                           : (config.audio ? kDefaultAudioReportInterval
+                                           : kDefaultVideoReportInterval)),
+      // TODO(bugs.webrtc.org/10774): Remove fallback.
+      remote_ssrc_(0),
+      remote_sender_rtp_time_(0),
+      remote_sender_packet_count_(0),
+      remote_sender_octet_count_(0),
+      remote_sender_reports_count_(0),
+      xr_rrtr_status_(config.non_sender_rtt_measurement),
+      xr_rr_rtt_ms_(0),
+      oldest_tmmbr_info_ms_(0),
+      cname_callback_(config.rtcp_cname_callback),
+      report_block_data_observer_(config.report_block_data_observer),
+      packet_type_counter_observer_(config.rtcp_packet_type_counter_observer),
+      num_skipped_packets_(0),
+      last_skipped_packets_warning_ms_(clock_->TimeInMilliseconds()) {
+  RTC_DCHECK(owner);
+}
+
+RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
+                           ModuleRtpRtcp* owner)
+    : clock_(config.clock),
+      receiver_only_(config.receiver_only),
+      rtp_rtcp_(owner),
+      registered_ssrcs_(true, config),
+      rtcp_bandwidth_observer_(config.bandwidth_callback),
+      rtcp_event_observer_(config.rtcp_event_observer),
+      rtcp_intra_frame_observer_(config.intra_frame_callback),
+      rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
+      network_state_estimate_observer_(config.network_state_estimate_observer),
+      transport_feedback_observer_(config.transport_feedback_callback),
+      bitrate_allocation_observer_(config.bitrate_allocation_observer),
+      report_interval_(config.rtcp_report_interval_ms > 0
+                           ? TimeDelta::Millis(config.rtcp_report_interval_ms)
+                           : (config.audio ? kDefaultAudioReportInterval
+                                           : kDefaultVideoReportInterval)),
+      // TODO(bugs.webrtc.org/10774): Remove fallback.
+      remote_ssrc_(0),
+      remote_sender_rtp_time_(0),
+      remote_sender_packet_count_(0),
+      remote_sender_octet_count_(0),
+      remote_sender_reports_count_(0),
+      xr_rrtr_status_(config.non_sender_rtt_measurement),
+      xr_rr_rtt_ms_(0),
+      oldest_tmmbr_info_ms_(0),
+      cname_callback_(config.rtcp_cname_callback),
+      report_block_data_observer_(config.report_block_data_observer),
+      packet_type_counter_observer_(config.rtcp_packet_type_counter_observer),
+      num_skipped_packets_(0),
+      last_skipped_packets_warning_ms_(clock_->TimeInMilliseconds()) {
+  RTC_DCHECK(owner);
+  // Dear reader - if you're here because of this log statement and are
+  // wondering what this is about, chances are that you are using an instance
+  // of RTCPReceiver without using the webrtc APIs. This creates a bit of a
+  // problem for WebRTC because this class is a part of an internal
+  // implementation that is constantly changing and being improved.
+  // The intention of this log statement is to give a heads up that changes
+  // are coming and encourage you to use the public APIs or be prepared that
+  // things might break down the line as more changes land. A thing you could
+  // try out for now is to replace the `CustomSequenceChecker` in the header
+  // with a regular `SequenceChecker` and see if that triggers an
+  // error in your code. If it does, chances are you have your own threading
+  // model that is not the same as WebRTC internally has.
+  RTC_LOG(LS_INFO) << "************** !!!DEPRECATION WARNING!! **************";
+}
+
+RTCPReceiver::~RTCPReceiver() {}
+
+void RTCPReceiver::IncomingPacket(rtc::ArrayView<const uint8_t> packet) {
+  if (packet.empty()) {
+    RTC_LOG(LS_WARNING) << "Incoming empty RTCP packet";
+    return;
+  }
+
+  PacketInformation packet_information;
+  if (!ParseCompoundPacket(packet, &packet_information))
+    return;
+  TriggerCallbacksFromRtcpPacket(packet_information);
+}
+
+// This method is only used by test and legacy code, so we should be able to
+// remove it soon.
+int64_t RTCPReceiver::LastReceivedReportBlockMs() const {
+  MutexLock lock(&rtcp_receiver_lock_);
+  return last_received_rb_.IsFinite() ? last_received_rb_.ms() : 0;
+}
+
+void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
+  MutexLock lock(&rtcp_receiver_lock_);
+  // New SSRC reset old reports.
+  last_received_sr_ntp_.Reset();
+  remote_ssrc_ = ssrc;
+}
+
+void RTCPReceiver::set_local_media_ssrc(uint32_t ssrc) {
+  registered_ssrcs_.set_media_ssrc(ssrc);
+}
+
+uint32_t RTCPReceiver::local_media_ssrc() const {
+  return registered_ssrcs_.media_ssrc();
+}
+
+uint32_t RTCPReceiver::RemoteSSRC() const {
+  MutexLock lock(&rtcp_receiver_lock_);
+  return remote_ssrc_;
+}
+
+void RTCPReceiver::RttStats::AddRtt(TimeDelta rtt) {
+  last_rtt_ = rtt;
+  if (rtt < min_rtt_) {
+    min_rtt_ = rtt;
+  }
+  if (rtt > max_rtt_) {
+    max_rtt_ = rtt;
+  }
+  sum_rtt_ += rtt;
+  ++num_rtts_;
+}
+
+int32_t RTCPReceiver::RTT(uint32_t remote_ssrc,
+                          int64_t* last_rtt_ms,
+                          int64_t* avg_rtt_ms,
+                          int64_t* min_rtt_ms,
+                          int64_t* max_rtt_ms) const {
+  MutexLock lock(&rtcp_receiver_lock_);
+
+  auto it = rtts_.find(remote_ssrc);
+  if (it == rtts_.end()) {
+    return -1;
+  }
+
+  if (last_rtt_ms) {
+    *last_rtt_ms = it->second.last_rtt().ms();
+  }
+
+  if (avg_rtt_ms) {
+    *avg_rtt_ms = it->second.average_rtt().ms();
+  }
+
+  if (min_rtt_ms) {
+    *min_rtt_ms = it->second.min_rtt().ms();
+  }
+
+  if (max_rtt_ms) {
+    *max_rtt_ms = it->second.max_rtt().ms();
+  }
+
+  return 0;
+}
+
+RTCPReceiver::NonSenderRttStats RTCPReceiver::GetNonSenderRTT() const {
+  MutexLock lock(&rtcp_receiver_lock_);
+  auto it = non_sender_rtts_.find(remote_ssrc_);
+  if (it == non_sender_rtts_.end()) {
+    return {};
+  }
+  return it->second;
+}
+
+void RTCPReceiver::SetNonSenderRttMeasurement(bool enabled) {
+  MutexLock lock(&rtcp_receiver_lock_);
+  xr_rrtr_status_ = enabled;
+}
+
+bool RTCPReceiver::GetAndResetXrRrRtt(int64_t* rtt_ms) {
+  RTC_DCHECK(rtt_ms);
+  MutexLock lock(&rtcp_receiver_lock_);
+  if (xr_rr_rtt_ms_ == 0) {
+    return false;
+  }
+  *rtt_ms = xr_rr_rtt_ms_;
+  xr_rr_rtt_ms_ = 0;
+  return true;
+}
+
+// Called regularly (1/sec) on the worker thread to do rtt  calculations.
+absl::optional<TimeDelta> RTCPReceiver::OnPeriodicRttUpdate(
+    Timestamp newer_than,
+    bool sending) {
+  // Running on the worker thread (same as construction thread).
+  absl::optional<TimeDelta> rtt;
+
+  if (sending) {
+    // Check if we've received a report block within the last kRttUpdateInterval
+    // amount of time.
+    MutexLock lock(&rtcp_receiver_lock_);
+    if (last_received_rb_.IsInfinite() || last_received_rb_ > newer_than) {
+      TimeDelta max_rtt = TimeDelta::MinusInfinity();
+      for (const auto& rtt_stats : rtts_) {
+        if (rtt_stats.second.last_rtt() > max_rtt) {
+          max_rtt = rtt_stats.second.last_rtt();
+        }
+      }
+      if (max_rtt.IsFinite()) {
+        rtt = max_rtt;
+      }
+    }
+
+    // Check for expired timers and if so, log and reset.
+    auto now = clock_->CurrentTime();
+    if (RtcpRrTimeoutLocked(now)) {
+      RTC_LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
+    } else if (RtcpRrSequenceNumberTimeoutLocked(now)) {
+      RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended "
+                               "highest sequence number.";
+    }
+  } else {
+    // Report rtt from receiver.
+    int64_t rtt_ms;
+    if (GetAndResetXrRrRtt(&rtt_ms)) {
+      rtt.emplace(TimeDelta::Millis(rtt_ms));
+    }
+  }
+
+  return rtt;
+}
+
+bool RTCPReceiver::NTP(uint32_t* received_ntp_secs,
+                       uint32_t* received_ntp_frac,
+                       uint32_t* rtcp_arrival_time_secs,
+                       uint32_t* rtcp_arrival_time_frac,
+                       uint32_t* rtcp_timestamp,
+                       uint32_t* remote_sender_packet_count,
+                       uint64_t* remote_sender_octet_count,
+                       uint64_t* remote_sender_reports_count) const {
+  MutexLock lock(&rtcp_receiver_lock_);
+  if (!last_received_sr_ntp_.Valid())
+    return false;
+
+  // NTP from incoming SenderReport.
+  if (received_ntp_secs)
+    *received_ntp_secs = remote_sender_ntp_time_.seconds();
+  if (received_ntp_frac)
+    *received_ntp_frac = remote_sender_ntp_time_.fractions();
+  // Rtp time from incoming SenderReport.
+  if (rtcp_timestamp)
+    *rtcp_timestamp = remote_sender_rtp_time_;
+
+  // Local NTP time when we received a RTCP packet with a send block.
+  if (rtcp_arrival_time_secs)
+    *rtcp_arrival_time_secs = last_received_sr_ntp_.seconds();
+  if (rtcp_arrival_time_frac)
+    *rtcp_arrival_time_frac = last_received_sr_ntp_.fractions();
+
+  // Counters.
+  if (remote_sender_packet_count)
+    *remote_sender_packet_count = remote_sender_packet_count_;
+  if (remote_sender_octet_count)
+    *remote_sender_octet_count = remote_sender_octet_count_;
+  if (remote_sender_reports_count)
+    *remote_sender_reports_count = remote_sender_reports_count_;
+
+  return true;
+}
+
+std::vector<rtcp::ReceiveTimeInfo>
+RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
+  MutexLock lock(&rtcp_receiver_lock_);
+
+  const size_t last_xr_rtis_size = std::min(
+      received_rrtrs_.size(), rtcp::ExtendedReports::kMaxNumberOfDlrrItems);
+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis;
+  last_xr_rtis.reserve(last_xr_rtis_size);
+
+  const uint32_t now_ntp = CompactNtp(clock_->CurrentNtpTime());
+
+  for (size_t i = 0; i < last_xr_rtis_size; ++i) {
+    RrtrInformation& rrtr = received_rrtrs_.front();
+    last_xr_rtis.emplace_back(rrtr.ssrc, rrtr.received_remote_mid_ntp_time,
+                              now_ntp - rrtr.local_receive_mid_ntp_time);
+    received_rrtrs_ssrc_it_.erase(rrtr.ssrc);
+    received_rrtrs_.pop_front();
+  }
+
+  return last_xr_rtis;
+}
+
+void RTCPReceiver::RemoteRTCPSenderInfo(uint32_t* packet_count,
+                                        uint32_t* octet_count,
+                                        int64_t* ntp_timestamp_ms,
+                                        int64_t* remote_ntp_timestamp_ms) const {
+  MutexLock lock(&rtcp_receiver_lock_);
+  *packet_count = remote_sender_packet_count_;
+  *octet_count = remote_sender_octet_count_;
+  *ntp_timestamp_ms = last_received_sr_ntp_.ToMs();
+  *remote_ntp_timestamp_ms = remote_sender_ntp_time_.ToMs();
+}
+
+std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
+  std::vector<ReportBlockData> result;
+  MutexLock lock(&rtcp_receiver_lock_);
+  for (const auto& report : received_report_blocks_) {
+    result.push_back(report.second);
+  }
+  return result;
+}
+
+bool RTCPReceiver::ParseCompoundPacket(rtc::ArrayView<const uint8_t> packet,
+                                       PacketInformation* packet_information) {
+  MutexLock lock(&rtcp_receiver_lock_);
+
+  CommonHeader rtcp_block;
+  // If a sender report is received but no DLRR, we need to reset the
+  // roundTripTime stat according to the standard, see
+  // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
+  struct RtcpReceivedBlock {
+    bool sender_report = false;
+    bool dlrr = false;
+  };
+  // For each remote SSRC we store if we've received a sender report or a DLRR
+  // block.
+  flat_map<uint32_t, RtcpReceivedBlock> received_blocks;
+  for (const uint8_t* next_block = packet.begin(); next_block != packet.end();
+       next_block = rtcp_block.NextPacket()) {
+    ptrdiff_t remaining_blocks_size = packet.end() - next_block;
+    RTC_DCHECK_GT(remaining_blocks_size, 0);
+    if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
+      if (next_block == packet.begin()) {
+        // Failed to parse 1st header, nothing was extracted from this packet.
+        RTC_LOG(LS_WARNING) << "Incoming invalid RTCP packet";
+        return false;
+      }
+      ++num_skipped_packets_;
+      break;
+    }
+
+    switch (rtcp_block.type()) {
+      case rtcp::SenderReport::kPacketType:
+        HandleSenderReport(rtcp_block, packet_information);
+        received_blocks[packet_information->remote_ssrc].sender_report = true;
+        break;
+      case rtcp::ReceiverReport::kPacketType:
+        HandleReceiverReport(rtcp_block, packet_information);
+        break;
+      case rtcp::Sdes::kPacketType:
+        HandleSdes(rtcp_block, packet_information);
+        break;
+      case rtcp::ExtendedReports::kPacketType: {
+        bool contains_dlrr = false;
+        uint32_t ssrc = 0;
+        HandleXr(rtcp_block, packet_information, contains_dlrr, ssrc);
+        if (contains_dlrr) {
+          received_blocks[ssrc].dlrr = true;
+        }
+        break;
+      }
+      case rtcp::Bye::kPacketType:
+        HandleBye(rtcp_block);
+        break;
+      case rtcp::App::kPacketType:
+        HandleApp(rtcp_block, packet_information);
+        break;
+      case rtcp::Rtpfb::kPacketType:
+        switch (rtcp_block.fmt()) {
+          case rtcp::Nack::kFeedbackMessageType:
+            HandleNack(rtcp_block, packet_information);
+            break;
+          case rtcp::Tmmbr::kFeedbackMessageType:
+            HandleTmmbr(rtcp_block, packet_information);
+            break;
+          case rtcp::Tmmbn::kFeedbackMessageType:
+            HandleTmmbn(rtcp_block, packet_information);
+            break;
+          case rtcp::RapidResyncRequest::kFeedbackMessageType:
+            HandleSrReq(rtcp_block, packet_information);
+            break;
+          case rtcp::TransportFeedback::kFeedbackMessageType:
+            HandleTransportFeedback(rtcp_block, packet_information);
+            break;
+          default:
+            ++num_skipped_packets_;
+            break;
+        }
+        break;
+      case rtcp::Psfb::kPacketType:
+        switch (rtcp_block.fmt()) {
+          case rtcp::Pli::kFeedbackMessageType:
+            HandlePli(rtcp_block, packet_information);
+            break;
+          case rtcp::Fir::kFeedbackMessageType:
+            HandleFir(rtcp_block, packet_information);
+            break;
+          case rtcp::Psfb::kAfbMessageType:
+            HandlePsfbApp(rtcp_block, packet_information);
+            break;
+          default:
+            ++num_skipped_packets_;
+            break;
+        }
+        break;
+      default:
+        ++num_skipped_packets_;
+        break;
+    }
+  }
+
+  for (const auto& rb : received_blocks) {
+    if (rb.second.sender_report && !rb.second.dlrr) {
+      auto rtt_stats = non_sender_rtts_.find(rb.first);
+      if (rtt_stats != non_sender_rtts_.end()) {
+        rtt_stats->second.Invalidate();
+      }
+    }
+  }
+
+  if (packet_type_counter_observer_) {
+    packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
+        local_media_ssrc(), packet_type_counter_);
+  }
+
+  if (num_skipped_packets_ > 0) {
+    const int64_t now_ms = clock_->TimeInMilliseconds();
+    if (now_ms - last_skipped_packets_warning_ms_ >= kMaxWarningLogIntervalMs) {
+      last_skipped_packets_warning_ms_ = now_ms;
+      RTC_LOG(LS_WARNING)
+          << num_skipped_packets_
+          << " RTCP blocks were skipped due to being malformed or of "
+             "unrecognized/unsupported type, during the past "
+          << (kMaxWarningLogIntervalMs / 1000) << " second period.";
+    }
+  }
+
+  return true;
+}
+
+void RTCPReceiver::HandleSenderReport(const CommonHeader& rtcp_block,
+                                      PacketInformation* packet_information) {
+  rtcp::SenderReport sender_report;
+  if (!sender_report.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  const uint32_t remote_ssrc = sender_report.sender_ssrc();
+
+  packet_information->remote_ssrc = remote_ssrc;
+
+  UpdateTmmbrRemoteIsAlive(remote_ssrc);
+
+  // Have I received RTP packets from this party?
+  if (remote_ssrc_ == remote_ssrc) {
+    // Only signal that we have received a SR when we accept one.
+    packet_information->packet_type_flags |= kRtcpSr;
+
+    remote_sender_ntp_time_ = sender_report.ntp();
+    remote_sender_rtp_time_ = sender_report.rtp_timestamp();
+    last_received_sr_ntp_ = clock_->CurrentNtpTime();
+    remote_sender_packet_count_ = sender_report.sender_packet_count();
+    remote_sender_octet_count_ = sender_report.sender_octet_count();
+    remote_sender_reports_count_++;
+  } else {
+    // We will only store the send report from one source, but
+    // we will store all the receive blocks.
+    packet_information->packet_type_flags |= kRtcpRr;
+  }
+
+  for (const rtcp::ReportBlock& report_block : sender_report.report_blocks())
+    HandleReportBlock(report_block, packet_information, remote_ssrc);
+}
+
+void RTCPReceiver::HandleReceiverReport(const CommonHeader& rtcp_block,
+                                        PacketInformation* packet_information) {
+  rtcp::ReceiverReport receiver_report;
+  if (!receiver_report.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  const uint32_t remote_ssrc = receiver_report.sender_ssrc();
+
+  packet_information->remote_ssrc = remote_ssrc;
+
+  UpdateTmmbrRemoteIsAlive(remote_ssrc);
+
+  packet_information->packet_type_flags |= kRtcpRr;
+
+  for (const ReportBlock& report_block : receiver_report.report_blocks())
+    HandleReportBlock(report_block, packet_information, remote_ssrc);
+}
+
+void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block,
+                                     PacketInformation* packet_information,
+                                     uint32_t remote_ssrc) {
+  // This will be called once per report block in the RTCP packet.
+  // We filter out all report blocks that are not for us.
+  // Each packet has max 31 RR blocks.
+  //
+  // We can calc RTT if we send a send report and get a report block back.
+
+  // `report_block.source_ssrc()` is the SSRC identifier of the source to
+  // which the information in this reception report block pertains.
+
+  // Filter out all report blocks that are not for us.
+  if (!registered_ssrcs_.contains(report_block.source_ssrc()))
+    return;
+
+  last_received_rb_ = clock_->CurrentTime();
+
+  ReportBlockData* report_block_data =
+      &received_report_blocks_[report_block.source_ssrc()];
+  RTCPReportBlock rtcp_report_block;
+  rtcp_report_block.sender_ssrc = remote_ssrc;
+  rtcp_report_block.source_ssrc = report_block.source_ssrc();
+  rtcp_report_block.fraction_lost = report_block.fraction_lost();
+  rtcp_report_block.packets_lost = report_block.cumulative_lost_signed();
+  if (report_block.extended_high_seq_num() >
+      report_block_data->report_block().extended_highest_sequence_number) {
+    // We have successfully delivered new RTP packets to the remote side after
+    // the last RR was sent from the remote side.
+    last_increased_sequence_number_ = last_received_rb_;
+  }
+  rtcp_report_block.extended_highest_sequence_number =
+      report_block.extended_high_seq_num();
+  rtcp_report_block.jitter = report_block.jitter();
+  rtcp_report_block.delay_since_last_sender_report =
+      report_block.delay_since_last_sr();
+  rtcp_report_block.last_sender_report_timestamp = report_block.last_sr();
+  // Number of seconds since 1900 January 1 00:00 GMT (see
+  // https://tools.ietf.org/html/rfc868).
+  report_block_data->SetReportBlock(
+      rtcp_report_block,
+      (clock_->CurrentNtpInMilliseconds() - rtc::kNtpJan1970Millisecs) *
+          rtc::kNumMicrosecsPerMillisec);
+
+  uint32_t send_time_ntp = report_block.last_sr();
+  // RFC3550, section 6.4.1, LSR field discription states:
+  // If no SR has been received yet, the field is set to zero.
+  // Receiver rtp_rtcp module is not expected to calculate rtt using
+  // Sender Reports even if it accidentally can.
+  if (send_time_ntp != 0) {
+    uint32_t delay_ntp = report_block.delay_since_last_sr();
+    // Local NTP time.
+    uint32_t receive_time_ntp =
+        CompactNtp(clock_->ConvertTimestampToNtpTime(last_received_rb_));
+
+    // RTT in 1/(2^16) seconds.
+    uint32_t rtt_ntp = receive_time_ntp - delay_ntp - send_time_ntp;
+    // Convert to 1/1000 seconds (milliseconds).
+    TimeDelta rtt = CompactNtpRttToTimeDelta(rtt_ntp);
+    report_block_data->AddRoundTripTimeSample(rtt.ms());
+    if (report_block.source_ssrc() == local_media_ssrc()) {
+      rtts_[remote_ssrc].AddRtt(rtt);
+    }
+
+    packet_information->rtt_ms = rtt.ms();
+  }
+
+  packet_information->report_blocks.push_back(
+      report_block_data->report_block());
+  packet_information->report_block_datas.push_back(*report_block_data);
+}
+
+RTCPReceiver::TmmbrInformation* RTCPReceiver::FindOrCreateTmmbrInfo(
+    uint32_t remote_ssrc) {
+  // Create or find receive information.
+  TmmbrInformation* tmmbr_info = &tmmbr_infos_[remote_ssrc];
+  // Update that this remote is alive.
+  tmmbr_info->last_time_received_ms = clock_->TimeInMilliseconds();
+  return tmmbr_info;
+}
+
+void RTCPReceiver::UpdateTmmbrRemoteIsAlive(uint32_t remote_ssrc) {
+  auto tmmbr_it = tmmbr_infos_.find(remote_ssrc);
+  if (tmmbr_it != tmmbr_infos_.end())
+    tmmbr_it->second.last_time_received_ms = clock_->TimeInMilliseconds();
+}
+
+RTCPReceiver::TmmbrInformation* RTCPReceiver::GetTmmbrInformation(
+    uint32_t remote_ssrc) {
+  auto it = tmmbr_infos_.find(remote_ssrc);
+  if (it == tmmbr_infos_.end())
+    return nullptr;
+  return &it->second;
+}
+
+// These two methods (RtcpRrTimeout and RtcpRrSequenceNumberTimeout) only exist
+// for tests and legacy code (rtp_rtcp_impl.cc). We should be able to to delete
+// the methods and require that access to the locked variables only happens on
+// the worker thread and thus no locking is needed.
+bool RTCPReceiver::RtcpRrTimeout() {
+  MutexLock lock(&rtcp_receiver_lock_);
+  return RtcpRrTimeoutLocked(clock_->CurrentTime());
+}
+
+bool RTCPReceiver::RtcpRrSequenceNumberTimeout() {
+  MutexLock lock(&rtcp_receiver_lock_);
+  return RtcpRrSequenceNumberTimeoutLocked(clock_->CurrentTime());
+}
+
+bool RTCPReceiver::UpdateTmmbrTimers() {
+  MutexLock lock(&rtcp_receiver_lock_);
+
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  int64_t timeout_ms = now_ms - kTmmbrTimeoutIntervalMs;
+
+  if (oldest_tmmbr_info_ms_ >= timeout_ms)
+    return false;
+
+  bool update_bounding_set = false;
+  oldest_tmmbr_info_ms_ = -1;
+  for (auto tmmbr_it = tmmbr_infos_.begin(); tmmbr_it != tmmbr_infos_.end();) {
+    TmmbrInformation* tmmbr_info = &tmmbr_it->second;
+    if (tmmbr_info->last_time_received_ms > 0) {
+      if (tmmbr_info->last_time_received_ms < timeout_ms) {
+        // No rtcp packet for the last 5 regular intervals, reset limitations.
+        tmmbr_info->tmmbr.clear();
+        // Prevent that we call this over and over again.
+        tmmbr_info->last_time_received_ms = 0;
+        // Send new TMMBN to all channels using the default codec.
+        update_bounding_set = true;
+      } else if (oldest_tmmbr_info_ms_ == -1 ||
+                 tmmbr_info->last_time_received_ms < oldest_tmmbr_info_ms_) {
+        oldest_tmmbr_info_ms_ = tmmbr_info->last_time_received_ms;
+      }
+      ++tmmbr_it;
+    } else if (tmmbr_info->ready_for_delete) {
+      // When we dont have a last_time_received_ms and the object is marked
+      // ready_for_delete it's removed from the map.
+      tmmbr_it = tmmbr_infos_.erase(tmmbr_it);
+    } else {
+      ++tmmbr_it;
+    }
+  }
+  return update_bounding_set;
+}
+
+std::vector<rtcp::TmmbItem> RTCPReceiver::BoundingSet(bool* tmmbr_owner) {
+  MutexLock lock(&rtcp_receiver_lock_);
+  TmmbrInformation* tmmbr_info = GetTmmbrInformation(remote_ssrc_);
+  if (!tmmbr_info)
+    return std::vector<rtcp::TmmbItem>();
+
+  *tmmbr_owner = TMMBRHelp::IsOwner(tmmbr_info->tmmbn, local_media_ssrc());
+  return tmmbr_info->tmmbn;
+}
+
+void RTCPReceiver::HandleSdes(const CommonHeader& rtcp_block,
+                              PacketInformation* packet_information) {
+  rtcp::Sdes sdes;
+  if (!sdes.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  for (const rtcp::Sdes::Chunk& chunk : sdes.chunks()) {
+    if (cname_callback_)
+      cname_callback_->OnCname(chunk.ssrc, chunk.cname);
+  }
+  packet_information->packet_type_flags |= kRtcpSdes;
+}
+
+void RTCPReceiver::HandleNack(const CommonHeader& rtcp_block,
+                              PacketInformation* packet_information) {
+  rtcp::Nack nack;
+  if (!nack.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  if (receiver_only_ || local_media_ssrc() != nack.media_ssrc())  // Not to us.
+    return;
+
+  packet_information->nack_sequence_numbers.insert(
+      packet_information->nack_sequence_numbers.end(),
+      nack.packet_ids().begin(), nack.packet_ids().end());
+  for (uint16_t packet_id : nack.packet_ids())
+    nack_stats_.ReportRequest(packet_id);
+
+  if (!nack.packet_ids().empty()) {
+    packet_information->packet_type_flags |= kRtcpNack;
+    ++packet_type_counter_.nack_packets;
+    packet_type_counter_.nack_requests = nack_stats_.requests();
+    packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
+  }
+}
+
+void RTCPReceiver::HandleApp(const rtcp::CommonHeader& rtcp_block,
+                             PacketInformation* packet_information) {
+  rtcp::App app;
+  if (app.Parse(rtcp_block)) {
+    if (app.name() == rtcp::RemoteEstimate::kName &&
+        app.sub_type() == rtcp::RemoteEstimate::kSubType) {
+      rtcp::RemoteEstimate estimate(std::move(app));
+      if (estimate.ParseData()) {
+        packet_information->network_state_estimate = estimate.estimate();
+        return;
+      }
+    }
+  }
+  ++num_skipped_packets_;
+}
+
+void RTCPReceiver::HandleBye(const CommonHeader& rtcp_block) {
+  rtcp::Bye bye;
+  if (!bye.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  if (rtcp_event_observer_) {
+    rtcp_event_observer_->OnRtcpBye();
+  }
+
+  // Clear our lists.
+  rtts_.erase(bye.sender_ssrc());
+  EraseIf(received_report_blocks_, [&](const auto& elem) {
+    return elem.second.report_block().sender_ssrc == bye.sender_ssrc();
+  });
+
+  TmmbrInformation* tmmbr_info = GetTmmbrInformation(bye.sender_ssrc());
+  if (tmmbr_info)
+    tmmbr_info->ready_for_delete = true;
+
+  last_fir_.erase(bye.sender_ssrc());
+  auto it = received_rrtrs_ssrc_it_.find(bye.sender_ssrc());
+  if (it != received_rrtrs_ssrc_it_.end()) {
+    received_rrtrs_.erase(it->second);
+    received_rrtrs_ssrc_it_.erase(it);
+  }
+  xr_rr_rtt_ms_ = 0;
+}
+
+void RTCPReceiver::HandleXr(const CommonHeader& rtcp_block,
+                            PacketInformation* packet_information,
+                            bool& contains_dlrr,
+                            uint32_t& ssrc) {
+  rtcp::ExtendedReports xr;
+  if (!xr.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+  ssrc = xr.sender_ssrc();
+  contains_dlrr = !xr.dlrr().sub_blocks().empty();
+
+  if (xr.rrtr())
+    HandleXrReceiveReferenceTime(xr.sender_ssrc(), *xr.rrtr());
+
+  for (const rtcp::ReceiveTimeInfo& time_info : xr.dlrr().sub_blocks())
+    HandleXrDlrrReportBlock(xr.sender_ssrc(), time_info);
+
+  if (xr.target_bitrate()) {
+    HandleXrTargetBitrate(xr.sender_ssrc(), *xr.target_bitrate(),
+                          packet_information);
+  }
+}
+
+void RTCPReceiver::HandleXrReceiveReferenceTime(uint32_t sender_ssrc,
+                                                const rtcp::Rrtr& rrtr) {
+  uint32_t received_remote_mid_ntp_time = CompactNtp(rrtr.ntp());
+  uint32_t local_receive_mid_ntp_time = CompactNtp(clock_->CurrentNtpTime());
+
+  auto it = received_rrtrs_ssrc_it_.find(sender_ssrc);
+  if (it != received_rrtrs_ssrc_it_.end()) {
+    it->second->received_remote_mid_ntp_time = received_remote_mid_ntp_time;
+    it->second->local_receive_mid_ntp_time = local_receive_mid_ntp_time;
+  } else {
+    if (received_rrtrs_.size() < kMaxNumberOfStoredRrtrs) {
+      received_rrtrs_.emplace_back(sender_ssrc, received_remote_mid_ntp_time,
+                                   local_receive_mid_ntp_time);
+      received_rrtrs_ssrc_it_[sender_ssrc] = std::prev(received_rrtrs_.end());
+    } else {
+      RTC_LOG(LS_WARNING) << "Discarding received RRTR for ssrc " << sender_ssrc
+                          << ", reached maximum number of stored RRTRs.";
+    }
+  }
+}
+
+void RTCPReceiver::HandleXrDlrrReportBlock(uint32_t sender_ssrc,
+                                           const rtcp::ReceiveTimeInfo& rti) {
+  if (!registered_ssrcs_.contains(rti.ssrc))  // Not to us.
+    return;
+
+  // Caller should explicitly enable rtt calculation using extended reports.
+  if (!xr_rrtr_status_)
+    return;
+
+  // The send_time and delay_rr fields are in units of 1/2^16 sec.
+  uint32_t send_time_ntp = rti.last_rr;
+  // RFC3611, section 4.5, LRR field discription states:
+  // If no such block has been received, the field is set to zero.
+  if (send_time_ntp == 0) {
+    auto rtt_stats = non_sender_rtts_.find(sender_ssrc);
+    if (rtt_stats != non_sender_rtts_.end()) {
+      rtt_stats->second.Invalidate();
+    }
+    return;
+  }
+
+  uint32_t delay_ntp = rti.delay_since_last_rr;
+  uint32_t now_ntp = CompactNtp(clock_->CurrentNtpTime());
+
+  uint32_t rtt_ntp = now_ntp - delay_ntp - send_time_ntp;
+  TimeDelta rtt = CompactNtpRttToTimeDelta(rtt_ntp);
+  xr_rr_rtt_ms_ = rtt.ms();
+
+  non_sender_rtts_[sender_ssrc].Update(rtt);
+}
+
+void RTCPReceiver::HandleXrTargetBitrate(
+    uint32_t ssrc,
+    const rtcp::TargetBitrate& target_bitrate,
+    PacketInformation* packet_information) {
+  if (ssrc != remote_ssrc_) {
+    return;  // Not for us.
+  }
+
+  VideoBitrateAllocation bitrate_allocation;
+  for (const auto& item : target_bitrate.GetTargetBitrates()) {
+    if (item.spatial_layer >= kMaxSpatialLayers ||
+        item.temporal_layer >= kMaxTemporalStreams) {
+      RTC_LOG(LS_WARNING)
+          << "Invalid layer in XR target bitrate pack: spatial index "
+          << item.spatial_layer << ", temporal index " << item.temporal_layer
+          << ", dropping.";
+    } else {
+      bitrate_allocation.SetBitrate(item.spatial_layer, item.temporal_layer,
+                                    item.target_bitrate_kbps * 1000);
+    }
+  }
+  packet_information->target_bitrate_allocation.emplace(bitrate_allocation);
+}
+
+void RTCPReceiver::HandlePli(const CommonHeader& rtcp_block,
+                             PacketInformation* packet_information) {
+  rtcp::Pli pli;
+  if (!pli.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  if (local_media_ssrc() == pli.media_ssrc()) {
+    ++packet_type_counter_.pli_packets;
+    // Received a signal that we need to send a new key frame.
+    packet_information->packet_type_flags |= kRtcpPli;
+  }
+}
+
+void RTCPReceiver::HandleTmmbr(const CommonHeader& rtcp_block,
+                               PacketInformation* packet_information) {
+  rtcp::Tmmbr tmmbr;
+  if (!tmmbr.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  uint32_t sender_ssrc = tmmbr.sender_ssrc();
+  if (tmmbr.media_ssrc()) {
+    // media_ssrc() SHOULD be 0 if same as SenderSSRC.
+    // In relay mode this is a valid number.
+    sender_ssrc = tmmbr.media_ssrc();
+  }
+
+  for (const rtcp::TmmbItem& request : tmmbr.requests()) {
+    if (local_media_ssrc() != request.ssrc() || request.bitrate_bps() == 0)
+      continue;
+
+    TmmbrInformation* tmmbr_info = FindOrCreateTmmbrInfo(tmmbr.sender_ssrc());
+    auto* entry = &tmmbr_info->tmmbr[sender_ssrc];
+    entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc, request.bitrate_bps(),
+                                       request.packet_overhead());
+    // FindOrCreateTmmbrInfo always sets `last_time_received_ms` to
+    // `clock_->TimeInMilliseconds()`.
+    entry->last_updated_ms = tmmbr_info->last_time_received_ms;
+
+    packet_information->packet_type_flags |= kRtcpTmmbr;
+    break;
+  }
+}
+
+void RTCPReceiver::HandleTmmbn(const CommonHeader& rtcp_block,
+                               PacketInformation* packet_information) {
+  rtcp::Tmmbn tmmbn;
+  if (!tmmbn.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  TmmbrInformation* tmmbr_info = FindOrCreateTmmbrInfo(tmmbn.sender_ssrc());
+
+  packet_information->packet_type_flags |= kRtcpTmmbn;
+
+  tmmbr_info->tmmbn = tmmbn.items();
+}
+
+void RTCPReceiver::HandleSrReq(const CommonHeader& rtcp_block,
+                               PacketInformation* packet_information) {
+  rtcp::RapidResyncRequest sr_req;
+  if (!sr_req.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  packet_information->packet_type_flags |= kRtcpSrReq;
+}
+
+void RTCPReceiver::HandlePsfbApp(const CommonHeader& rtcp_block,
+                                 PacketInformation* packet_information) {
+  {
+    rtcp::Remb remb;
+    if (remb.Parse(rtcp_block)) {
+      packet_information->packet_type_flags |= kRtcpRemb;
+      packet_information->receiver_estimated_max_bitrate_bps =
+          remb.bitrate_bps();
+      return;
+    }
+  }
+
+  {
+    auto loss_notification = std::make_unique<rtcp::LossNotification>();
+    if (loss_notification->Parse(rtcp_block)) {
+      packet_information->packet_type_flags |= kRtcpLossNotification;
+      packet_information->loss_notification = std::move(loss_notification);
+      return;
+    }
+  }
+
+  RTC_LOG(LS_WARNING) << "Unknown PSFB-APP packet.";
+
+  ++num_skipped_packets_;
+}
+
+void RTCPReceiver::HandleFir(const CommonHeader& rtcp_block,
+                             PacketInformation* packet_information) {
+  rtcp::Fir fir;
+  if (!fir.Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  if (fir.requests().empty())
+    return;
+
+  const int64_t now_ms = clock_->TimeInMilliseconds();
+  for (const rtcp::Fir::Request& fir_request : fir.requests()) {
+    // Is it our sender that is requested to generate a new keyframe.
+    if (local_media_ssrc() != fir_request.ssrc)
+      continue;
+
+    ++packet_type_counter_.fir_packets;
+
+    auto inserted = last_fir_.insert(std::make_pair(
+        fir.sender_ssrc(), LastFirStatus(now_ms, fir_request.seq_nr)));
+    if (!inserted.second) {  // There was already an entry.
+      LastFirStatus* last_fir = &inserted.first->second;
+
+      // Check if we have reported this FIRSequenceNumber before.
+      if (fir_request.seq_nr == last_fir->sequence_number)
+        continue;
+
+      // Sanity: don't go crazy with the callbacks.
+      if (now_ms - last_fir->request_ms < kRtcpMinFrameLengthMs)
+        continue;
+
+      last_fir->request_ms = now_ms;
+      last_fir->sequence_number = fir_request.seq_nr;
+    }
+    // Received signal that we need to send a new key frame.
+    packet_information->packet_type_flags |= kRtcpFir;
+  }
+}
+
+void RTCPReceiver::HandleTransportFeedback(
+    const CommonHeader& rtcp_block,
+    PacketInformation* packet_information) {
+  std::unique_ptr<rtcp::TransportFeedback> transport_feedback(
+      new rtcp::TransportFeedback());
+  if (!transport_feedback->Parse(rtcp_block)) {
+    ++num_skipped_packets_;
+    return;
+  }
+
+  packet_information->packet_type_flags |= kRtcpTransportFeedback;
+  packet_information->transport_feedback = std::move(transport_feedback);
+}
+
+void RTCPReceiver::NotifyTmmbrUpdated() {
+  // Find bounding set.
+  std::vector<rtcp::TmmbItem> bounding =
+      TMMBRHelp::FindBoundingSet(TmmbrReceived());
+
+  if (!bounding.empty() && rtcp_bandwidth_observer_) {
+    // We have a new bandwidth estimate on this channel.
+    uint64_t bitrate_bps = TMMBRHelp::CalcMinBitrateBps(bounding);
+    if (bitrate_bps <= std::numeric_limits<uint32_t>::max())
+      rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate_bps);
+  }
+
+  // Send tmmbn to inform remote clients about the new bandwidth.
+  rtp_rtcp_->SetTmmbn(std::move(bounding));
+}
+
+// Holding no Critical section.
+void RTCPReceiver::TriggerCallbacksFromRtcpPacket(
+    const PacketInformation& packet_information) {
+  // Process TMMBR and REMB first to avoid multiple callbacks
+  // to OnNetworkChanged.
+  if (packet_information.packet_type_flags & kRtcpTmmbr) {
+    // Might trigger a OnReceivedBandwidthEstimateUpdate.
+    NotifyTmmbrUpdated();
+  }
+
+  if (!receiver_only_ && (packet_information.packet_type_flags & kRtcpSrReq)) {
+    rtp_rtcp_->OnRequestSendReport();
+  }
+  if (!receiver_only_ && (packet_information.packet_type_flags & kRtcpNack)) {
+    if (!packet_information.nack_sequence_numbers.empty()) {
+      RTC_LOG(LS_VERBOSE) << "Incoming NACK length: "
+                          << packet_information.nack_sequence_numbers.size();
+      rtp_rtcp_->OnReceivedNack(packet_information.nack_sequence_numbers);
+    }
+  }
+
+  // We need feedback that we have received a report block(s) so that we
+  // can generate a new packet in a conference relay scenario, one received
+  // report can generate several RTCP packets, based on number relayed/mixed
+  // a send report block should go out to all receivers.
+  if (rtcp_intra_frame_observer_) {
+    RTC_DCHECK(!receiver_only_);
+    if ((packet_information.packet_type_flags & kRtcpPli) ||
+        (packet_information.packet_type_flags & kRtcpFir)) {
+      if (packet_information.packet_type_flags & kRtcpPli) {
+        RTC_LOG(LS_VERBOSE)
+            << "Incoming PLI from SSRC " << packet_information.remote_ssrc;
+      } else {
+        RTC_LOG(LS_VERBOSE)
+            << "Incoming FIR from SSRC " << packet_information.remote_ssrc;
+      }
+      rtcp_intra_frame_observer_->OnReceivedIntraFrameRequest(
+          local_media_ssrc());
+    }
+  }
+  if (rtcp_loss_notification_observer_ &&
+      (packet_information.packet_type_flags & kRtcpLossNotification)) {
+    rtcp::LossNotification* loss_notification =
+        packet_information.loss_notification.get();
+    RTC_DCHECK(loss_notification);
+    if (loss_notification->media_ssrc() == local_media_ssrc()) {
+      rtcp_loss_notification_observer_->OnReceivedLossNotification(
+          loss_notification->media_ssrc(), loss_notification->last_decoded(),
+          loss_notification->last_received(),
+          loss_notification->decodability_flag());
+    }
+  }
+  if (rtcp_bandwidth_observer_) {
+    RTC_DCHECK(!receiver_only_);
+    if (packet_information.packet_type_flags & kRtcpRemb) {
+      RTC_LOG(LS_VERBOSE)
+          << "Incoming REMB: "
+          << packet_information.receiver_estimated_max_bitrate_bps;
+      rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(
+          packet_information.receiver_estimated_max_bitrate_bps);
+    }
+    if ((packet_information.packet_type_flags & kRtcpSr) ||
+        (packet_information.packet_type_flags & kRtcpRr)) {
+      int64_t now_ms = clock_->TimeInMilliseconds();
+      rtcp_bandwidth_observer_->OnReceivedRtcpReceiverReport(
+          packet_information.report_blocks, packet_information.rtt_ms, now_ms);
+    }
+  }
+  if ((packet_information.packet_type_flags & kRtcpSr) ||
+      (packet_information.packet_type_flags & kRtcpRr)) {
+    rtp_rtcp_->OnReceivedRtcpReportBlocks(packet_information.report_blocks);
+  }
+
+  if (transport_feedback_observer_ &&
+      (packet_information.packet_type_flags & kRtcpTransportFeedback)) {
+    uint32_t media_source_ssrc =
+        packet_information.transport_feedback->media_ssrc();
+    if (media_source_ssrc == local_media_ssrc() ||
+        registered_ssrcs_.contains(media_source_ssrc)) {
+      transport_feedback_observer_->OnTransportFeedback(
+          *packet_information.transport_feedback);
+    }
+  }
+
+  if (network_state_estimate_observer_ &&
+      packet_information.network_state_estimate) {
+    network_state_estimate_observer_->OnRemoteNetworkEstimate(
+        *packet_information.network_state_estimate);
+  }
+
+  if (bitrate_allocation_observer_ &&
+      packet_information.target_bitrate_allocation) {
+    bitrate_allocation_observer_->OnBitrateAllocationUpdated(
+        *packet_information.target_bitrate_allocation);
+  }
+
+  if (!receiver_only_) {
+    if (report_block_data_observer_) {
+      for (const auto& report_block_data :
+           packet_information.report_block_datas) {
+        report_block_data_observer_->OnReportBlockDataUpdated(
+            report_block_data);
+      }
+    }
+  }
+}
+
+std::vector<rtcp::TmmbItem> RTCPReceiver::TmmbrReceived() {
+  MutexLock lock(&rtcp_receiver_lock_);
+  std::vector<rtcp::TmmbItem> candidates;
+
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  int64_t timeout_ms = now_ms - kTmmbrTimeoutIntervalMs;
+
+  for (auto& kv : tmmbr_infos_) {
+    for (auto it = kv.second.tmmbr.begin(); it != kv.second.tmmbr.end();) {
+      if (it->second.last_updated_ms < timeout_ms) {
+        // Erase timeout entries.
+        it = kv.second.tmmbr.erase(it);
+      } else {
+        candidates.push_back(it->second.tmmbr_item);
+        ++it;
+      }
+    }
+  }
+  return candidates;
+}
+
+bool RTCPReceiver::RtcpRrTimeoutLocked(Timestamp now) {
+  bool result = ResetTimestampIfExpired(now, last_received_rb_, report_interval_);
+  if (result && rtcp_event_observer_) {
+    rtcp_event_observer_->OnRtcpTimeout();
+  }
+  return result;
+}
+
+bool RTCPReceiver::RtcpRrSequenceNumberTimeoutLocked(Timestamp now) {
+  bool result = ResetTimestampIfExpired(now, last_increased_sequence_number_,
+                                 report_interval_);
+  if (result && rtcp_event_observer_) {
+    rtcp_event_observer_->OnRtcpTimeout();
+  }
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.h
new file mode 100644
index 0000000000..e3f5bc765c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -0,0 +1,457 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
+
+#include <list>
+#include <map>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/sequence_checker.h"
+#include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_nack_stats.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/containers/flat_map.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+class ModuleRtpRtcpImpl2;
+class VideoBitrateAllocationObserver;
+
+namespace rtcp {
+class CommonHeader;
+class ReportBlock;
+class Rrtr;
+class TargetBitrate;
+class TmmbItem;
+}  // namespace rtcp
+
+class RTCPReceiver final {
+ public:
+  class ModuleRtpRtcp {
+   public:
+    virtual void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) = 0;
+    virtual void OnRequestSendReport() = 0;
+    virtual void OnReceivedNack(
+        const std::vector<uint16_t>& nack_sequence_numbers) = 0;
+    virtual void OnReceivedRtcpReportBlocks(
+        const ReportBlockList& report_blocks) = 0;
+
+   protected:
+    virtual ~ModuleRtpRtcp() = default;
+  };
+  // Standardized stats derived from the non-sender RTT.
+  class NonSenderRttStats {
+   public:
+    NonSenderRttStats() = default;
+    NonSenderRttStats(const NonSenderRttStats&) = default;
+    NonSenderRttStats& operator=(const NonSenderRttStats&) = default;
+    ~NonSenderRttStats() = default;
+    void Update(TimeDelta non_sender_rtt_seconds) {
+      round_trip_time_ = non_sender_rtt_seconds;
+      total_round_trip_time_ += non_sender_rtt_seconds;
+      round_trip_time_measurements_++;
+    }
+    void Invalidate() { round_trip_time_.reset(); }
+    // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
+    absl::optional<TimeDelta> round_trip_time() const {
+      return round_trip_time_;
+    }
+    // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
+    TimeDelta total_round_trip_time() const { return total_round_trip_time_; }
+    // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
+    int round_trip_time_measurements() const {
+      return round_trip_time_measurements_;
+    }
+
+   private:
+    absl::optional<TimeDelta> round_trip_time_;
+    TimeDelta total_round_trip_time_ = TimeDelta::Zero();
+    int round_trip_time_measurements_ = 0;
+  };
+
+  RTCPReceiver(const RtpRtcpInterface::Configuration& config,
+               ModuleRtpRtcp* owner);
+
+  RTCPReceiver(const RtpRtcpInterface::Configuration& config,
+               ModuleRtpRtcpImpl2* owner);
+
+  ~RTCPReceiver();
+
+  void IncomingPacket(const uint8_t* packet, size_t packet_size) {
+    IncomingPacket(rtc::MakeArrayView(packet, packet_size));
+  }
+  void IncomingPacket(rtc::ArrayView<const uint8_t> packet);
+
+  int64_t LastReceivedReportBlockMs() const;
+
+  void set_local_media_ssrc(uint32_t ssrc);
+  uint32_t local_media_ssrc() const;
+
+  void SetRemoteSSRC(uint32_t ssrc);
+  uint32_t RemoteSSRC() const;
+
+  bool receiver_only() const { return receiver_only_; }
+
+  // Get received NTP.
+  // The types for the arguments below derive from the specification:
+  // - `remote_sender_packet_count`: `RTCSentRtpStreamStats.packetsSent` [1]
+  // - `remote_sender_octet_count`: `RTCSentRtpStreamStats.bytesSent` [1]
+  // - `remote_sender_reports_count`:
+  //   `RTCRemoteOutboundRtpStreamStats.reportsSent` [2]
+  // [1] https://www.w3.org/TR/webrtc-stats/#remoteoutboundrtpstats-dict*
+  // [2] https://www.w3.org/TR/webrtc-stats/#dom-rtcsentrtpstreamstats
+  bool NTP(uint32_t* received_ntp_secs,
+           uint32_t* received_ntp_frac,
+           uint32_t* rtcp_arrival_time_secs,
+           uint32_t* rtcp_arrival_time_frac,
+           uint32_t* rtcp_timestamp,
+           uint32_t* remote_sender_packet_count,
+           uint64_t* remote_sender_octet_count,
+           uint64_t* remote_sender_reports_count) const;
+
+  std::vector<rtcp::ReceiveTimeInfo> ConsumeReceivedXrReferenceTimeInfo();
+
+  // Get received sender packet and octet counts
+  void RemoteRTCPSenderInfo(uint32_t* packet_count,
+                            uint32_t* octet_count,
+                            int64_t* ntp_timestamp_ms,
+                            int64_t* remote_ntp_timestamp_ms) const;
+
+  // Get rtt.
+  int32_t RTT(uint32_t remote_ssrc,
+              int64_t* last_rtt_ms,
+              int64_t* avg_rtt_ms,
+              int64_t* min_rtt_ms,
+              int64_t* max_rtt_ms) const;
+
+  // Returns non-sender RTT metrics for the remote SSRC.
+  NonSenderRttStats GetNonSenderRTT() const;
+
+  void SetNonSenderRttMeasurement(bool enabled);
+  bool GetAndResetXrRrRtt(int64_t* rtt_ms);
+
+  // Called once per second on the worker thread to do rtt calculations.
+  // Returns an optional rtt value if one is available.
+  absl::optional<TimeDelta> OnPeriodicRttUpdate(Timestamp newer_than,
+                                                bool sending);
+
+  // A snapshot of Report Blocks with additional data of interest to statistics.
+  // Within this list, the source SSRC is unique and ReportBlockData represents
+  // the latest Report Block that was received for that SSRC.
+  std::vector<ReportBlockData> GetLatestReportBlockData() const;
+
+  // Returns true if we haven't received an RTCP RR for several RTCP
+  // intervals, but only triggers true once.
+  bool RtcpRrTimeout();
+
+  // Returns true if we haven't received an RTCP RR telling the receive side
+  // has not received RTP packets for too long, i.e. extended highest sequence
+  // number hasn't increased for several RTCP intervals. The function only
+  // returns true once until a new RR is received.
+  bool RtcpRrSequenceNumberTimeout();
+
+  std::vector<rtcp::TmmbItem> TmmbrReceived();
+  // Return true if new bandwidth should be set.
+  bool UpdateTmmbrTimers();
+  std::vector<rtcp::TmmbItem> BoundingSet(bool* tmmbr_owner);
+  // Set new bandwidth and notify remote clients about it.
+  void NotifyTmmbrUpdated();
+
+ private:
+#if RTC_DCHECK_IS_ON
+  class CustomSequenceChecker : public SequenceChecker {
+   public:
+    explicit CustomSequenceChecker(bool disable_checks)
+        : disable_checks_(disable_checks) {}
+    bool IsCurrent() const {
+      if (disable_checks_)
+        return true;
+      return SequenceChecker::IsCurrent();
+    }
+
+   private:
+    const bool disable_checks_;
+  };
+#else
+  class CustomSequenceChecker : public SequenceChecker {
+   public:
+    explicit CustomSequenceChecker(bool) {}
+  };
+#endif
+
+  // A lightweight inlined set of local SSRCs.
+  class RegisteredSsrcs {
+   public:
+    static constexpr size_t kMediaSsrcIndex = 0;
+    static constexpr size_t kMaxSsrcs = 3;
+    // Initializes the set of registered local SSRCS by extracting them from the
+    // provided `config`. The `disable_sequence_checker` flag is a workaround
+    // to be able to use a sequence checker without breaking downstream
+    // code that currently doesn't follow the same threading rules as webrtc.
+    RegisteredSsrcs(bool disable_sequence_checker,
+                    const RtpRtcpInterface::Configuration& config);
+
+    // Indicates if `ssrc` is in the set of registered local SSRCs.
+    bool contains(uint32_t ssrc) const;
+    uint32_t media_ssrc() const;
+    void set_media_ssrc(uint32_t ssrc);
+
+   private:
+    RTC_NO_UNIQUE_ADDRESS CustomSequenceChecker packet_sequence_checker_;
+    absl::InlinedVector<uint32_t, kMaxSsrcs> ssrcs_
+        RTC_GUARDED_BY(packet_sequence_checker_);
+  };
+
+  struct PacketInformation;
+
+  // Structure for handing TMMBR and TMMBN rtcp messages (RFC5104,
+  // section 3.5.4).
+  struct TmmbrInformation {
+    struct TimedTmmbrItem {
+      rtcp::TmmbItem tmmbr_item;
+      int64_t last_updated_ms;
+    };
+
+    int64_t last_time_received_ms = 0;
+
+    bool ready_for_delete = false;
+
+    std::vector<rtcp::TmmbItem> tmmbn;
+    std::map<uint32_t, TimedTmmbrItem> tmmbr;
+  };
+
+  // Structure for storing received RRTR RTCP messages (RFC3611, section 4.4).
+  struct RrtrInformation {
+    RrtrInformation(uint32_t ssrc,
+                    uint32_t received_remote_mid_ntp_time,
+                    uint32_t local_receive_mid_ntp_time)
+        : ssrc(ssrc),
+          received_remote_mid_ntp_time(received_remote_mid_ntp_time),
+          local_receive_mid_ntp_time(local_receive_mid_ntp_time) {}
+
+    uint32_t ssrc;
+    // Received NTP timestamp in compact representation.
+    uint32_t received_remote_mid_ntp_time;
+    // NTP time when the report was received in compact representation.
+    uint32_t local_receive_mid_ntp_time;
+  };
+
+  struct LastFirStatus {
+    LastFirStatus(int64_t now_ms, uint8_t sequence_number)
+        : request_ms(now_ms), sequence_number(sequence_number) {}
+    int64_t request_ms;
+    uint8_t sequence_number;
+  };
+
+  class RttStats {
+   public:
+    RttStats() = default;
+    RttStats(const RttStats&) = default;
+    RttStats& operator=(const RttStats&) = default;
+
+    void AddRtt(TimeDelta rtt);
+
+    TimeDelta last_rtt() const { return last_rtt_; }
+    TimeDelta min_rtt() const { return min_rtt_; }
+    TimeDelta max_rtt() const { return max_rtt_; }
+    TimeDelta average_rtt() const { return sum_rtt_ / num_rtts_; }
+
+   private:
+    TimeDelta last_rtt_ = TimeDelta::Zero();
+    TimeDelta min_rtt_ = TimeDelta::PlusInfinity();
+    TimeDelta max_rtt_ = TimeDelta::MinusInfinity();
+    TimeDelta sum_rtt_ = TimeDelta::Zero();
+    size_t num_rtts_ = 0;
+  };
+
+  bool ParseCompoundPacket(rtc::ArrayView<const uint8_t> packet,
+                           PacketInformation* packet_information);
+
+  void TriggerCallbacksFromRtcpPacket(
+      const PacketInformation& packet_information);
+
+  TmmbrInformation* FindOrCreateTmmbrInfo(uint32_t remote_ssrc)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+  // Update TmmbrInformation (if present) is alive.
+  void UpdateTmmbrRemoteIsAlive(uint32_t remote_ssrc)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+  TmmbrInformation* GetTmmbrInformation(uint32_t remote_ssrc)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleSenderReport(const rtcp::CommonHeader& rtcp_block,
+                          PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleReceiverReport(const rtcp::CommonHeader& rtcp_block,
+                            PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleReportBlock(const rtcp::ReportBlock& report_block,
+                         PacketInformation* packet_information,
+                         uint32_t remote_ssrc)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleSdes(const rtcp::CommonHeader& rtcp_block,
+                  PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleXr(const rtcp::CommonHeader& rtcp_block,
+                PacketInformation* packet_information,
+                bool& contains_dlrr,
+                uint32_t& ssrc)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleXrReceiveReferenceTime(uint32_t sender_ssrc,
+                                    const rtcp::Rrtr& rrtr)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleXrDlrrReportBlock(uint32_t ssrc, const rtcp::ReceiveTimeInfo& rti)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleXrTargetBitrate(uint32_t ssrc,
+                             const rtcp::TargetBitrate& target_bitrate,
+                             PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleNack(const rtcp::CommonHeader& rtcp_block,
+                  PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleApp(const rtcp::CommonHeader& rtcp_block,
+                 PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleBye(const rtcp::CommonHeader& rtcp_block)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandlePli(const rtcp::CommonHeader& rtcp_block,
+                 PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandlePsfbApp(const rtcp::CommonHeader& rtcp_block,
+                     PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleTmmbr(const rtcp::CommonHeader& rtcp_block,
+                   PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleTmmbn(const rtcp::CommonHeader& rtcp_block,
+                   PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleSrReq(const rtcp::CommonHeader& rtcp_block,
+                   PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleFir(const rtcp::CommonHeader& rtcp_block,
+                 PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  void HandleTransportFeedback(const rtcp::CommonHeader& rtcp_block,
+                               PacketInformation* packet_information)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  bool RtcpRrTimeoutLocked(Timestamp now)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  bool RtcpRrSequenceNumberTimeoutLocked(Timestamp now)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
+
+  Clock* const clock_;
+  const bool receiver_only_;
+  ModuleRtpRtcp* const rtp_rtcp_;
+  // The set of registered local SSRCs.
+  RegisteredSsrcs registered_ssrcs_;
+
+  RtcpBandwidthObserver* const rtcp_bandwidth_observer_;
+  RtcpEventObserver* const rtcp_event_observer_;
+  RtcpIntraFrameObserver* const rtcp_intra_frame_observer_;
+  RtcpLossNotificationObserver* const rtcp_loss_notification_observer_;
+  NetworkStateEstimateObserver* const network_state_estimate_observer_;
+  TransportFeedbackObserver* const transport_feedback_observer_;
+  VideoBitrateAllocationObserver* const bitrate_allocation_observer_;
+  const TimeDelta report_interval_;
+
+  mutable Mutex rtcp_receiver_lock_;
+  uint32_t remote_ssrc_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+
+  // Received sender report.
+  NtpTime remote_sender_ntp_time_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+  uint32_t remote_sender_rtp_time_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+  // When did we receive the last send report.
+  NtpTime last_received_sr_ntp_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+  uint32_t remote_sender_packet_count_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+  uint64_t remote_sender_octet_count_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+  uint64_t remote_sender_reports_count_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+
+  // Received RRTR information in ascending receive time order.
+  std::list<RrtrInformation> received_rrtrs_
+      RTC_GUARDED_BY(rtcp_receiver_lock_);
+  // Received RRTR information mapped by remote ssrc.
+  flat_map<uint32_t, std::list<RrtrInformation>::iterator>
+      received_rrtrs_ssrc_it_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+
+  // Estimated rtt, zero when there is no valid estimate.
+  bool xr_rrtr_status_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+  int64_t xr_rr_rtt_ms_;
+
+  int64_t oldest_tmmbr_info_ms_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+  // Mapped by remote ssrc.
+  flat_map<uint32_t, TmmbrInformation> tmmbr_infos_
+      RTC_GUARDED_BY(rtcp_receiver_lock_);
+
+  // Round-Trip Time per remote sender ssrc.
+  flat_map<uint32_t, RttStats> rtts_ RTC_GUARDED_BY(rtcp_receiver_lock_);
+  // Non-sender Round-trip time per remote ssrc.
+  flat_map<uint32_t, NonSenderRttStats> non_sender_rtts_
+      RTC_GUARDED_BY(rtcp_receiver_lock_);
+
+  // Report blocks per local source ssrc.
+  flat_map<uint32_t, ReportBlockData> received_report_blocks_
+      RTC_GUARDED_BY(rtcp_receiver_lock_);
+  flat_map<uint32_t, LastFirStatus> last_fir_
+      RTC_GUARDED_BY(rtcp_receiver_lock_);
+
+  // The last time we received an RTCP Report block for this module.
+  Timestamp last_received_rb_ RTC_GUARDED_BY(rtcp_receiver_lock_) =
+      Timestamp::PlusInfinity();
+
+  // The time we last received an RTCP RR telling we have successfully
+  // delivered RTP packet to the remote side.
+  Timestamp last_increased_sequence_number_ = Timestamp::PlusInfinity();
+
+  RtcpCnameCallback* const cname_callback_;
+  ReportBlockDataObserver* const report_block_data_observer_;
+
+  RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
+  RtcpPacketTypeCounter packet_type_counter_;
+
+  RtcpNackStats nack_stats_;
+
+  size_t num_skipped_packets_;
+  int64_t last_skipped_packets_warning_ms_;
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
new file mode 100644
index 0000000000..ff39ccca9c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -0,0 +1,2012 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_receiver.h"
+
+#include <memory>
+#include <set>
+#include <utility>
+
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_bitrate_allocator.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/random.h"
+#include "system_wrappers/include/ntp_time.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using rtcp::ReceiveTimeInfo;
+using ::testing::_;
+using ::testing::AllOf;
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::InSequence;
+using ::testing::IsEmpty;
+using ::testing::NiceMock;
+using ::testing::Property;
+using ::testing::SizeIs;
+using ::testing::StrEq;
+using ::testing::StrictMock;
+using ::testing::UnorderedElementsAre;
+
+class MockRtcpPacketTypeCounterObserver : public RtcpPacketTypeCounterObserver {
+ public:
+  MOCK_METHOD(void,
+              RtcpPacketTypesCounterUpdated,
+              (uint32_t, const RtcpPacketTypeCounter&),
+              (override));
+};
+
+class MockRtcpIntraFrameObserver : public RtcpIntraFrameObserver {
+ public:
+  MOCK_METHOD(void, OnReceivedIntraFrameRequest, (uint32_t), (override));
+};
+
+class MockRtcpLossNotificationObserver : public RtcpLossNotificationObserver {
+ public:
+  ~MockRtcpLossNotificationObserver() override = default;
+  MOCK_METHOD(void,
+              OnReceivedLossNotification,
+              (uint32_t ssrc,
+               uint16_t seq_num_of_last_decodable,
+               uint16_t seq_num_of_last_received,
+               bool decodability_flag),
+              (override));
+};
+
+class MockCnameCallbackImpl : public RtcpCnameCallback {
+ public:
+  MOCK_METHOD(void, OnCname, (uint32_t, absl::string_view), (override));
+};
+
+class MockReportBlockDataObserverImpl : public ReportBlockDataObserver {
+ public:
+  MOCK_METHOD(void, OnReportBlockDataUpdated, (ReportBlockData), (override));
+};
+
+class MockTransportFeedbackObserver : public TransportFeedbackObserver {
+ public:
+  MOCK_METHOD(void, OnAddPacket, (const RtpPacketSendInfo&), (override));
+  MOCK_METHOD(void,
+              OnTransportFeedback,
+              (const rtcp::TransportFeedback&),
+              (override));
+};
+
+class MockModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
+ public:
+  MOCK_METHOD(void, SetTmmbn, (std::vector<rtcp::TmmbItem>), (override));
+  MOCK_METHOD(void, OnRequestSendReport, (), (override));
+  MOCK_METHOD(void, OnReceivedNack, (const std::vector<uint16_t>&), (override));
+  MOCK_METHOD(void,
+              OnReceivedRtcpReportBlocks,
+              (const ReportBlockList&),
+              (override));
+};
+
+class MockVideoBitrateAllocationObserver
+    : public VideoBitrateAllocationObserver {
+ public:
+  MOCK_METHOD(void,
+              OnBitrateAllocationUpdated,
+              (const VideoBitrateAllocation& allocation),
+              (override));
+};
+
+// SSRC of remote peer, that sends rtcp packet to the rtcp receiver under test.
+constexpr uint32_t kSenderSsrc = 0x10203;
+// SSRCs of local peer, that rtcp packet addressed to.
+constexpr uint32_t kReceiverMainSsrc = 0x123456;
+// RtcpReceiver can accept several ssrc, e.g. regular and rtx streams.
+constexpr uint32_t kReceiverExtraSsrc = 0x1234567;
+// SSRCs to ignore (i.e. not configured in RtcpReceiver).
+constexpr uint32_t kNotToUsSsrc = 0x654321;
+constexpr uint32_t kUnknownSenderSsrc = 0x54321;
+
+constexpr int64_t kRtcpIntervalMs = 1000;
+
+}  // namespace
+
+struct ReceiverMocks {
+  ReceiverMocks() : clock(1335900000) {}
+
+  SimulatedClock clock;
+  // Callbacks to packet_type_counter_observer are frequent but most of the time
+  // are not interesting.
+  NiceMock<MockRtcpPacketTypeCounterObserver> packet_type_counter_observer;
+  StrictMock<MockRtcpBandwidthObserver> bandwidth_observer;
+  StrictMock<MockRtcpIntraFrameObserver> intra_frame_observer;
+  StrictMock<MockRtcpLossNotificationObserver> rtcp_loss_notification_observer;
+  StrictMock<MockTransportFeedbackObserver> transport_feedback_observer;
+  StrictMock<MockVideoBitrateAllocationObserver> bitrate_allocation_observer;
+  StrictMock<MockModuleRtpRtcp> rtp_rtcp_impl;
+};
+
+RtpRtcpInterface::Configuration DefaultConfiguration(ReceiverMocks* mocks) {
+  RtpRtcpInterface::Configuration config;
+  config.clock = &mocks->clock;
+  config.receiver_only = false;
+  config.rtcp_packet_type_counter_observer =
+      &mocks->packet_type_counter_observer;
+  config.bandwidth_callback = &mocks->bandwidth_observer;
+  config.intra_frame_callback = &mocks->intra_frame_observer;
+  config.rtcp_loss_notification_observer =
+      &mocks->rtcp_loss_notification_observer;
+  config.transport_feedback_callback = &mocks->transport_feedback_observer;
+  config.bitrate_allocation_observer = &mocks->bitrate_allocation_observer;
+  config.rtcp_report_interval_ms = kRtcpIntervalMs;
+  config.local_media_ssrc = kReceiverMainSsrc;
+  config.rtx_send_ssrc = kReceiverExtraSsrc;
+  return config;
+}
+
+TEST(RtcpReceiverTest, BrokenPacketIsIgnored) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+
+  const uint8_t bad_packet[] = {0, 0, 0, 0};
+  EXPECT_CALL(mocks.packet_type_counter_observer, RtcpPacketTypesCounterUpdated)
+      .Times(0);
+  receiver.IncomingPacket(bad_packet);
+}
+
+TEST(RtcpReceiverTest, InvalidFeedbackPacketIsIgnored) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+
+  // Too short feedback packet.
+  const uint8_t bad_packet[] = {0x81, rtcp::Rtpfb::kPacketType, 0, 0};
+
+  // TODO(danilchap): Add expectation RtcpPacketTypesCounterUpdated
+  // is not called once parser would be adjusted to avoid that callback on
+  // semi-valid packets.
+  receiver.IncomingPacket(bad_packet);
+}
+
+TEST(RtcpReceiverTest, InjectSrPacket) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  EXPECT_FALSE(receiver.NTP(nullptr, nullptr, nullptr, nullptr, nullptr,
+                            nullptr, nullptr, nullptr));
+
+  int64_t now = mocks.clock.TimeInMilliseconds();
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(IsEmpty()));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(IsEmpty(), _, now));
+  receiver.IncomingPacket(sr.Build());
+
+  EXPECT_TRUE(receiver.NTP(nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
+                           nullptr, nullptr));
+}
+
+TEST(RtcpReceiverTest, InjectSrPacketFromUnknownSender) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  int64_t now = mocks.clock.TimeInMilliseconds();
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kUnknownSenderSsrc);
+
+  // The parser will handle report blocks in Sender Report from other than their
+  // expected peer.
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(_, _, now));
+  receiver.IncomingPacket(sr.Build());
+
+  // But will not flag that he's gotten sender information.
+  EXPECT_FALSE(receiver.NTP(nullptr, nullptr, nullptr, nullptr, nullptr,
+                            nullptr, nullptr, nullptr));
+}
+
+TEST(RtcpReceiverTest, InjectSrPacketCalculatesRTT) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const TimeDelta kRtt = TimeDelta::Millis(123);
+  const uint32_t kDelayNtp = 0x4321;
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+
+  int64_t rtt_ms = 0;
+  EXPECT_EQ(-1, receiver.RTT(kSenderSsrc, &rtt_ms, nullptr, nullptr, nullptr));
+
+  uint32_t sent_ntp = CompactNtp(mocks.clock.CurrentNtpTime());
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  rtcp::ReportBlock block;
+  block.SetMediaSsrc(kReceiverMainSsrc);
+  block.SetLastSr(sent_ntp);
+  block.SetDelayLastSr(kDelayNtp);
+  sr.AddReportBlock(block);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(sr.Build());
+
+  EXPECT_EQ(0, receiver.RTT(kSenderSsrc, &rtt_ms, nullptr, nullptr, nullptr));
+  EXPECT_NEAR(rtt_ms, kRtt.ms(), 1);
+}
+
+TEST(RtcpReceiverTest, InjectSrPacketCalculatesNegativeRTTAsOne) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const TimeDelta kRtt = TimeDelta::Millis(-13);
+  const uint32_t kDelayNtp = 0x4321;
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+
+  int64_t rtt_ms = 0;
+  EXPECT_EQ(-1, receiver.RTT(kSenderSsrc, &rtt_ms, nullptr, nullptr, nullptr));
+
+  uint32_t sent_ntp = CompactNtp(mocks.clock.CurrentNtpTime());
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  rtcp::ReportBlock block;
+  block.SetMediaSsrc(kReceiverMainSsrc);
+  block.SetLastSr(sent_ntp);
+  block.SetDelayLastSr(kDelayNtp);
+  sr.AddReportBlock(block);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(SizeIs(1)));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(SizeIs(1), _, _));
+  receiver.IncomingPacket(sr.Build());
+
+  EXPECT_EQ(0, receiver.RTT(kSenderSsrc, &rtt_ms, nullptr, nullptr, nullptr));
+  EXPECT_EQ(1, rtt_ms);
+}
+
+TEST(RtcpReceiverTest,
+     TwoReportBlocksWithLastOneWithoutLastSrCalculatesRttForBandwidthObserver) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const TimeDelta kRtt = TimeDelta::Millis(120);
+  const uint32_t kDelayNtp = 123000;
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+
+  uint32_t sent_ntp = CompactNtp(mocks.clock.CurrentNtpTime());
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  rtcp::ReportBlock block;
+  block.SetMediaSsrc(kReceiverMainSsrc);
+  block.SetLastSr(sent_ntp);
+  block.SetDelayLastSr(kDelayNtp);
+  sr.AddReportBlock(block);
+  block.SetMediaSsrc(kReceiverExtraSsrc);
+  block.SetLastSr(0);
+  sr.AddReportBlock(block);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(SizeIs(2)));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(SizeIs(2), kRtt.ms(), _));
+  receiver.IncomingPacket(sr.Build());
+}
+
+TEST(RtcpReceiverTest, InjectRrPacket) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  int64_t now = mocks.clock.TimeInMilliseconds();
+  rtcp::ReceiverReport rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(IsEmpty()));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(IsEmpty(), _, now));
+  receiver.IncomingPacket(rr.Build());
+
+  EXPECT_THAT(receiver.GetLatestReportBlockData(), IsEmpty());
+}
+
+TEST(RtcpReceiverTest, InjectRrPacketWithReportBlockNotToUsIgnored) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  int64_t now = mocks.clock.TimeInMilliseconds();
+  rtcp::ReportBlock rb;
+  rb.SetMediaSsrc(kNotToUsSsrc);
+  rtcp::ReceiverReport rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+  rr.AddReportBlock(rb);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(IsEmpty()));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(IsEmpty(), _, now));
+  receiver.IncomingPacket(rr.Build());
+
+  EXPECT_EQ(0, receiver.LastReceivedReportBlockMs());
+  EXPECT_THAT(receiver.GetLatestReportBlockData(), IsEmpty());
+}
+
+TEST(RtcpReceiverTest, InjectRrPacketWithOneReportBlock) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  int64_t now = mocks.clock.TimeInMilliseconds();
+
+  rtcp::ReportBlock rb;
+  rb.SetMediaSsrc(kReceiverMainSsrc);
+  rtcp::ReceiverReport rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+  rr.AddReportBlock(rb);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(SizeIs(1)));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(SizeIs(1), _, now));
+  receiver.IncomingPacket(rr.Build());
+
+  EXPECT_EQ(now, receiver.LastReceivedReportBlockMs());
+  EXPECT_THAT(receiver.GetLatestReportBlockData(), SizeIs(1));
+}
+
+TEST(RtcpReceiverTest, InjectSrPacketWithOneReportBlock) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  int64_t now = mocks.clock.TimeInMilliseconds();
+
+  rtcp::ReportBlock rb;
+  rb.SetMediaSsrc(kReceiverMainSsrc);
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  sr.AddReportBlock(rb);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(SizeIs(1)));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(SizeIs(1), _, now));
+  receiver.IncomingPacket(sr.Build());
+
+  EXPECT_EQ(now, receiver.LastReceivedReportBlockMs());
+  EXPECT_THAT(receiver.GetLatestReportBlockData(), SizeIs(1));
+}
+
+TEST(RtcpReceiverTest, InjectRrPacketWithTwoReportBlocks) {
+  const uint16_t kSequenceNumbers[] = {10, 12423};
+  const uint32_t kCumLost[] = {13, 555};
+  const uint8_t kFracLost[] = {20, 11};
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  int64_t now = mocks.clock.TimeInMilliseconds();
+
+  rtcp::ReportBlock rb1;
+  rb1.SetMediaSsrc(kReceiverMainSsrc);
+  rb1.SetExtHighestSeqNum(kSequenceNumbers[0]);
+  rb1.SetFractionLost(10);
+
+  rtcp::ReportBlock rb2;
+  rb2.SetMediaSsrc(kReceiverExtraSsrc);
+  rb2.SetExtHighestSeqNum(kSequenceNumbers[1]);
+  rb2.SetFractionLost(0);
+
+  rtcp::ReceiverReport rr1;
+  rr1.SetSenderSsrc(kSenderSsrc);
+  rr1.AddReportBlock(rb1);
+  rr1.AddReportBlock(rb2);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(SizeIs(2)));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(SizeIs(2), _, now));
+  receiver.IncomingPacket(rr1.Build());
+
+  EXPECT_EQ(now, receiver.LastReceivedReportBlockMs());
+  EXPECT_THAT(receiver.GetLatestReportBlockData(),
+              UnorderedElementsAre(
+                  Property(&ReportBlockData::report_block,
+                           Field(&RTCPReportBlock::fraction_lost, 0)),
+                  Property(&ReportBlockData::report_block,
+                           Field(&RTCPReportBlock::fraction_lost, 10))));
+
+  // Insert next receiver report with same ssrc but new values.
+  rtcp::ReportBlock rb3;
+  rb3.SetMediaSsrc(kReceiverMainSsrc);
+  rb3.SetExtHighestSeqNum(kSequenceNumbers[0]);
+  rb3.SetFractionLost(kFracLost[0]);
+  rb3.SetCumulativeLost(kCumLost[0]);
+
+  rtcp::ReportBlock rb4;
+  rb4.SetMediaSsrc(kReceiverExtraSsrc);
+  rb4.SetExtHighestSeqNum(kSequenceNumbers[1]);
+  rb4.SetFractionLost(kFracLost[1]);
+  rb4.SetCumulativeLost(kCumLost[1]);
+
+  rtcp::ReceiverReport rr2;
+  rr2.SetSenderSsrc(kSenderSsrc);
+  rr2.AddReportBlock(rb3);
+  rr2.AddReportBlock(rb4);
+
+  // Advance time to make 1st sent time and 2nd sent time different.
+  mocks.clock.AdvanceTimeMilliseconds(500);
+  now = mocks.clock.TimeInMilliseconds();
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(SizeIs(2)));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(SizeIs(2), _, now));
+  receiver.IncomingPacket(rr2.Build());
+
+  EXPECT_THAT(
+      receiver.GetLatestReportBlockData(),
+      UnorderedElementsAre(
+          Property(
+              &ReportBlockData::report_block,
+              AllOf(Field(&RTCPReportBlock::source_ssrc, kReceiverMainSsrc),
+                    Field(&RTCPReportBlock::fraction_lost, kFracLost[0]),
+                    Field(&RTCPReportBlock::packets_lost, kCumLost[0]),
+                    Field(&RTCPReportBlock::extended_highest_sequence_number,
+                          kSequenceNumbers[0]))),
+          Property(
+              &ReportBlockData::report_block,
+              AllOf(Field(&RTCPReportBlock::source_ssrc, kReceiverExtraSsrc),
+                    Field(&RTCPReportBlock::fraction_lost, kFracLost[1]),
+                    Field(&RTCPReportBlock::packets_lost, kCumLost[1]),
+                    Field(&RTCPReportBlock::extended_highest_sequence_number,
+                          kSequenceNumbers[1])))));
+}
+
+TEST(RtcpReceiverTest,
+     InjectRrPacketsFromTwoRemoteSsrcsReturnsLatestReportBlock) {
+  const uint32_t kSenderSsrc2 = 0x20304;
+  const uint16_t kSequenceNumbers[] = {10, 12423};
+  const int32_t kCumLost[] = {13, 555};
+  const uint8_t kFracLost[] = {20, 11};
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::ReportBlock rb1;
+  rb1.SetMediaSsrc(kReceiverMainSsrc);
+  rb1.SetExtHighestSeqNum(kSequenceNumbers[0]);
+  rb1.SetFractionLost(kFracLost[0]);
+  rb1.SetCumulativeLost(kCumLost[0]);
+  rtcp::ReceiverReport rr1;
+  rr1.SetSenderSsrc(kSenderSsrc);
+  rr1.AddReportBlock(rb1);
+
+  int64_t now = mocks.clock.TimeInMilliseconds();
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(SizeIs(1)));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(SizeIs(1), _, now));
+  receiver.IncomingPacket(rr1.Build());
+
+  EXPECT_EQ(now, receiver.LastReceivedReportBlockMs());
+
+  EXPECT_THAT(
+      receiver.GetLatestReportBlockData(),
+      ElementsAre(Property(
+          &ReportBlockData::report_block,
+          AllOf(Field(&RTCPReportBlock::source_ssrc, kReceiverMainSsrc),
+                Field(&RTCPReportBlock::sender_ssrc, kSenderSsrc),
+                Field(&RTCPReportBlock::fraction_lost, kFracLost[0]),
+                Field(&RTCPReportBlock::packets_lost, kCumLost[0]),
+                Field(&RTCPReportBlock::extended_highest_sequence_number,
+                      kSequenceNumbers[0])))));
+
+  rtcp::ReportBlock rb2;
+  rb2.SetMediaSsrc(kReceiverMainSsrc);
+  rb2.SetExtHighestSeqNum(kSequenceNumbers[1]);
+  rb2.SetFractionLost(kFracLost[1]);
+  rb2.SetCumulativeLost(kCumLost[1]);
+  rtcp::ReceiverReport rr2;
+  rr2.SetSenderSsrc(kSenderSsrc2);
+  rr2.AddReportBlock(rb2);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks(SizeIs(1)));
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedRtcpReceiverReport(SizeIs(1), _, now));
+  receiver.IncomingPacket(rr2.Build());
+
+  EXPECT_THAT(
+      receiver.GetLatestReportBlockData(),
+      UnorderedElementsAre(
+          Property(
+              &ReportBlockData::report_block,
+              AllOf(Field(&RTCPReportBlock::source_ssrc, kReceiverMainSsrc),
+                    Field(&RTCPReportBlock::sender_ssrc, kSenderSsrc2),
+                    Field(&RTCPReportBlock::fraction_lost, kFracLost[1]),
+                    Field(&RTCPReportBlock::packets_lost, kCumLost[1]),
+                    Field(&RTCPReportBlock::extended_highest_sequence_number,
+                          kSequenceNumbers[1])))));
+}
+
+TEST(RtcpReceiverTest, GetRtt) {
+  const uint32_t kSentCompactNtp = 0x1234;
+  const uint32_t kDelayCompactNtp = 0x222;
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  // No report block received.
+  EXPECT_EQ(-1, receiver.RTT(kSenderSsrc, nullptr, nullptr, nullptr, nullptr));
+
+  rtcp::ReportBlock rb;
+  rb.SetMediaSsrc(kReceiverMainSsrc);
+  rb.SetLastSr(kSentCompactNtp);
+  rb.SetDelayLastSr(kDelayCompactNtp);
+
+  rtcp::ReceiverReport rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+  rr.AddReportBlock(rb);
+  int64_t now = mocks.clock.TimeInMilliseconds();
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rr.Build());
+
+  EXPECT_EQ(now, receiver.LastReceivedReportBlockMs());
+  EXPECT_EQ(0, receiver.RTT(kSenderSsrc, nullptr, nullptr, nullptr, nullptr));
+}
+
+// App packets are ignored.
+TEST(RtcpReceiverTest, InjectApp) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::App app;
+  app.SetSubType(30);
+  app.SetName(0x17a177e);
+  const uint8_t kData[] = {'t', 'e', 's', 't', 'd', 'a', 't', 'a'};
+  app.SetData(kData, sizeof(kData));
+
+  receiver.IncomingPacket(app.Build());
+}
+
+TEST(RtcpReceiverTest, InjectSdesWithOneChunk) {
+  ReceiverMocks mocks;
+  MockCnameCallbackImpl callback;
+  RtpRtcpInterface::Configuration config = DefaultConfiguration(&mocks);
+  config.rtcp_cname_callback = &callback;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const char kCname[] = "alice@host";
+  rtcp::Sdes sdes;
+  sdes.AddCName(kSenderSsrc, kCname);
+
+  EXPECT_CALL(callback, OnCname(kSenderSsrc, StrEq(kCname)));
+  receiver.IncomingPacket(sdes.Build());
+}
+
+TEST(RtcpReceiverTest, InjectByePacket_RemovesReportBlocks) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::ReportBlock rb1;
+  rb1.SetMediaSsrc(kReceiverMainSsrc);
+  rtcp::ReportBlock rb2;
+  rb2.SetMediaSsrc(kReceiverExtraSsrc);
+  rtcp::ReceiverReport rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+  rr.AddReportBlock(rb1);
+  rr.AddReportBlock(rb2);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rr.Build());
+
+  EXPECT_THAT(receiver.GetLatestReportBlockData(), SizeIs(2));
+
+  // Verify that BYE removes the report blocks.
+  rtcp::Bye bye;
+  bye.SetSenderSsrc(kSenderSsrc);
+
+  receiver.IncomingPacket(bye.Build());
+
+  EXPECT_THAT(receiver.GetLatestReportBlockData(), IsEmpty());
+
+  // Inject packet again.
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rr.Build());
+
+  EXPECT_THAT(receiver.GetLatestReportBlockData(), SizeIs(2));
+}
+
+TEST(RtcpReceiverTest, InjectByePacketRemovesReferenceTimeInfo) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  rtcp::Rrtr rrtr;
+  rrtr.SetNtp(NtpTime(0x10203, 0x40506));
+  xr.SetRrtr(rrtr);
+  receiver.IncomingPacket(xr.Build());
+
+  rtcp::Bye bye;
+  bye.SetSenderSsrc(kSenderSsrc);
+  receiver.IncomingPacket(bye.Build());
+
+  EXPECT_THAT(receiver.ConsumeReceivedXrReferenceTimeInfo(), IsEmpty());
+}
+
+TEST(RtcpReceiverTest, InjectPliPacket) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::Pli pli;
+  pli.SetMediaSsrc(kReceiverMainSsrc);
+
+  EXPECT_CALL(
+      mocks.packet_type_counter_observer,
+      RtcpPacketTypesCounterUpdated(
+          kReceiverMainSsrc, Field(&RtcpPacketTypeCounter::pli_packets, 1)));
+  EXPECT_CALL(mocks.intra_frame_observer,
+              OnReceivedIntraFrameRequest(kReceiverMainSsrc));
+  receiver.IncomingPacket(pli.Build());
+}
+
+TEST(RtcpReceiverTest, PliPacketNotToUsIgnored) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::Pli pli;
+  pli.SetMediaSsrc(kNotToUsSsrc);
+
+  EXPECT_CALL(
+      mocks.packet_type_counter_observer,
+      RtcpPacketTypesCounterUpdated(
+          kReceiverMainSsrc, Field(&RtcpPacketTypeCounter::pli_packets, 0)));
+  EXPECT_CALL(mocks.intra_frame_observer, OnReceivedIntraFrameRequest).Times(0);
+  receiver.IncomingPacket(pli.Build());
+}
+
+TEST(RtcpReceiverTest, InjectFirPacket) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::Fir fir;
+  fir.AddRequestTo(kReceiverMainSsrc, 13);
+
+  EXPECT_CALL(
+      mocks.packet_type_counter_observer,
+      RtcpPacketTypesCounterUpdated(
+          kReceiverMainSsrc, Field(&RtcpPacketTypeCounter::fir_packets, 1)));
+  EXPECT_CALL(mocks.intra_frame_observer,
+              OnReceivedIntraFrameRequest(kReceiverMainSsrc));
+  receiver.IncomingPacket(fir.Build());
+}
+
+TEST(RtcpReceiverTest, FirPacketNotToUsIgnored) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::Fir fir;
+  fir.AddRequestTo(kNotToUsSsrc, 13);
+
+  EXPECT_CALL(mocks.intra_frame_observer, OnReceivedIntraFrameRequest).Times(0);
+  receiver.IncomingPacket(fir.Build());
+}
+
+TEST(RtcpReceiverTest, ExtendedReportsPacketWithZeroReportBlocksIgnored) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+
+  receiver.IncomingPacket(xr.Build());
+}
+
+TEST(RtcpReceiverTest, InjectExtendedReportsReceiverReferenceTimePacket) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const NtpTime kNtp(0x10203, 0x40506);
+  rtcp::Rrtr rrtr;
+  rrtr.SetNtp(kNtp);
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.SetRrtr(rrtr);
+
+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
+      receiver.ConsumeReceivedXrReferenceTimeInfo();
+  EXPECT_THAT(last_xr_rtis, IsEmpty());
+
+  receiver.IncomingPacket(xr.Build());
+
+  last_xr_rtis = receiver.ConsumeReceivedXrReferenceTimeInfo();
+  ASSERT_THAT(last_xr_rtis, SizeIs(1));
+  EXPECT_EQ(kSenderSsrc, last_xr_rtis[0].ssrc);
+  EXPECT_EQ(CompactNtp(kNtp), last_xr_rtis[0].last_rr);
+  EXPECT_EQ(0U, last_xr_rtis[0].delay_since_last_rr);
+}
+
+TEST(RtcpReceiverTest, ExtendedReportsDlrrPacketNotToUsIgnored) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  // Allow calculate rtt using dlrr/rrtr, simulating media receiver side.
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kNotToUsSsrc, 0x12345, 0x67890));
+
+  receiver.IncomingPacket(xr.Build());
+
+  int64_t rtt_ms = 0;
+  EXPECT_FALSE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_FALSE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_TRUE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_EQ(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+}
+
+TEST(RtcpReceiverTest, InjectExtendedReportsDlrrPacketWithSubBlock) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint32_t kLastRR = 0x12345;
+  const uint32_t kDelay = 0x23456;
+  int64_t rtt_ms = 0;
+  EXPECT_FALSE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, kLastRR, kDelay));
+
+  receiver.IncomingPacket(xr.Build());
+
+  uint32_t compact_ntp_now = CompactNtp(mocks.clock.CurrentNtpTime());
+  EXPECT_TRUE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+  uint32_t rtt_ntp = compact_ntp_now - kDelay - kLastRR;
+  EXPECT_NEAR(CompactNtpRttToTimeDelta(rtt_ntp).ms(), rtt_ms, 1);
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_GT(non_sender_rtt_stats.round_trip_time(), TimeDelta::Zero());
+  EXPECT_FALSE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_GT(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+}
+
+TEST(RtcpReceiverTest, InjectExtendedReportsDlrrPacketWithMultipleSubBlocks) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint32_t kLastRR = 0x12345;
+  const uint32_t kDelay = 0x56789;
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, kLastRR, kDelay));
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc + 1, 0x12345, 0x67890));
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc + 2, 0x12345, 0x67890));
+
+  receiver.IncomingPacket(xr.Build());
+
+  uint32_t compact_ntp_now = CompactNtp(mocks.clock.CurrentNtpTime());
+  int64_t rtt_ms = 0;
+  EXPECT_TRUE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+  uint32_t rtt_ntp = compact_ntp_now - kDelay - kLastRR;
+  EXPECT_NEAR(CompactNtpRttToTimeDelta(rtt_ntp).ms(), rtt_ms, 1);
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_GT(non_sender_rtt_stats.round_trip_time(), TimeDelta::Zero());
+  EXPECT_FALSE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_GT(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+}
+
+TEST(RtcpReceiverTest, InjectExtendedReportsPacketWithMultipleReportBlocks) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::Rrtr rrtr;
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.SetRrtr(rrtr);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, 0x12345, 0x67890));
+
+  receiver.IncomingPacket(xr.Build());
+
+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
+      receiver.ConsumeReceivedXrReferenceTimeInfo();
+  EXPECT_THAT(last_xr_rtis, SizeIs(1));
+  int64_t rtt_ms = 0;
+  EXPECT_TRUE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+}
+
+TEST(RtcpReceiverTest, InjectExtendedReportsPacketWithUnknownReportBlock) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::Rrtr rrtr;
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.SetRrtr(rrtr);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, 0x12345, 0x67890));
+
+  rtc::Buffer packet = xr.Build();
+  // Modify the DLRR block to have an unsupported block type, from 5 to 6.
+  ASSERT_EQ(5, packet.data()[20]);
+  packet.data()[20] = 6;
+  receiver.IncomingPacket(packet);
+
+  // Validate Rrtr was received and processed.
+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
+      receiver.ConsumeReceivedXrReferenceTimeInfo();
+  EXPECT_THAT(last_xr_rtis, SizeIs(1));
+  // Validate Dlrr report wasn't processed.
+  int64_t rtt_ms = 0;
+  EXPECT_FALSE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_FALSE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_TRUE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_EQ(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+}
+
+TEST(RtcpReceiverTest, TestExtendedReportsRrRttInitiallyFalse) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  int64_t rtt_ms;
+  EXPECT_FALSE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_FALSE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_TRUE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_EQ(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+}
+
+TEST(RtcpReceiverTest, RttCalculatedAfterExtendedReportsDlrr) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  Random rand(0x0123456789abcdef);
+  const TimeDelta kRtt = TimeDelta::Millis(rand.Rand(1, 9 * 3600 * 1000));
+  const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+  NtpTime now = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp = CompactNtp(now);
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp, kDelayNtp));
+
+  receiver.IncomingPacket(xr.Build());
+
+  int64_t rtt_ms = 0;
+  EXPECT_TRUE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+  EXPECT_NEAR(kRtt.ms(), rtt_ms, 1);
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_TRUE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_FALSE(non_sender_rtt_stats.round_trip_time().value().IsZero());
+  EXPECT_FALSE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_GT(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+}
+
+// Same test as above but enables receive-side RTT using the setter instead of
+// the config struct.
+TEST(RtcpReceiverTest, SetterEnablesReceiverRtt) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = false;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+  receiver.SetNonSenderRttMeasurement(true);
+
+  Random rand(0x0123456789abcdef);
+  const TimeDelta kRtt = TimeDelta::Millis(rand.Rand(1, 9 * 3600 * 1000));
+  const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+  NtpTime now = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp = CompactNtp(now);
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp, kDelayNtp));
+
+  receiver.IncomingPacket(xr.Build());
+
+  int64_t rtt_ms = 0;
+  EXPECT_TRUE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+  EXPECT_NEAR(rtt_ms, kRtt.ms(), 1);
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_TRUE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_FALSE(non_sender_rtt_stats.round_trip_time().value().IsZero());
+  EXPECT_FALSE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_GT(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+}
+
+// Same test as above but disables receive-side RTT using the setter instead of
+// the config struct.
+TEST(RtcpReceiverTest, DoesntCalculateRttOnReceivedDlrr) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+  receiver.SetNonSenderRttMeasurement(false);
+
+  Random rand(0x0123456789abcdef);
+  const TimeDelta kRtt = TimeDelta::Millis(rand.Rand(1, 9 * 3600 * 1000));
+  const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+  NtpTime now = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp = CompactNtp(now);
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp, kDelayNtp));
+
+  receiver.IncomingPacket(xr.Build());
+
+  // We expect that no RTT is available (because receive-side RTT was disabled).
+  int64_t rtt_ms = 0;
+  EXPECT_FALSE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_FALSE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_TRUE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_EQ(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+}
+
+TEST(RtcpReceiverTest, XrDlrrCalculatesNegativeRttAsOne) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  Random rand(0x0123456789abcdef);
+  const TimeDelta kRtt = TimeDelta::Millis(rand.Rand(-3600 * 1000, -1));
+  const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+  NtpTime now = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp = CompactNtp(now);
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp, kDelayNtp));
+
+  receiver.IncomingPacket(xr.Build());
+
+  int64_t rtt_ms = 0;
+  EXPECT_TRUE(receiver.GetAndResetXrRrRtt(&rtt_ms));
+  EXPECT_EQ(1, rtt_ms);
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_TRUE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_FALSE(non_sender_rtt_stats.round_trip_time().value().IsZero());
+  EXPECT_FALSE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_GT(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+}
+
+// Test receiver RTT stats with multiple measurements.
+TEST(RtcpReceiverTest, ReceiverRttWithMultipleMeasurements) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  Random rand(0x0123456789abcdef);
+  const TimeDelta kRtt = TimeDelta::Millis(rand.Rand(1, 9 * 3600 * 1000));
+  const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+  NtpTime now = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp = CompactNtp(now);
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp, kDelayNtp));
+
+  receiver.IncomingPacket(xr.Build());
+
+  // Check that the non-sender RTT stats are valid and based on a single
+  // measurement.
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_TRUE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_NEAR(non_sender_rtt_stats.round_trip_time()->ms(), kRtt.ms(), 1);
+  EXPECT_EQ(non_sender_rtt_stats.round_trip_time_measurements(), 1);
+  EXPECT_EQ(non_sender_rtt_stats.total_round_trip_time().ms(),
+            non_sender_rtt_stats.round_trip_time()->ms());
+
+  // Generate another XR report with the same RTT and delay.
+  NtpTime now2 = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp2 = CompactNtp(now2);
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::ExtendedReports xr2;
+  xr2.SetSenderSsrc(kSenderSsrc);
+  xr2.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp2, kDelayNtp));
+
+  receiver.IncomingPacket(xr2.Build());
+
+  // Check that the non-sender RTT stats are based on 2 measurements, and that
+  // the values are as expected.
+  non_sender_rtt_stats = receiver.GetNonSenderRTT();
+  EXPECT_TRUE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_NEAR(non_sender_rtt_stats.round_trip_time()->ms(), kRtt.ms(), 1);
+  EXPECT_EQ(non_sender_rtt_stats.round_trip_time_measurements(), 2);
+  EXPECT_NEAR(non_sender_rtt_stats.total_round_trip_time().ms(), 2 * kRtt.ms(),
+              2);
+}
+
+// Test that the receiver RTT stat resets when receiving a SR without XR. This
+// behavior is described in the standard, see
+// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime.
+TEST(RtcpReceiverTest, ReceiverRttResetOnSrWithoutXr) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  Random rand(0x0123456789abcdef);
+  const TimeDelta kRtt = TimeDelta::Millis(rand.Rand(1, 9 * 3600 * 1000));
+  const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+  NtpTime now = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp = CompactNtp(now);
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp, kDelayNtp));
+
+  receiver.IncomingPacket(xr.Build());
+
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_TRUE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_NEAR(non_sender_rtt_stats.round_trip_time()->ms(), kRtt.ms(), 1);
+
+  // Generate a SR without XR.
+  rtcp::ReportBlock rb;
+  rb.SetMediaSsrc(kReceiverMainSsrc);
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  sr.AddReportBlock(rb);
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+
+  receiver.IncomingPacket(sr.Build());
+
+  // Check that the non-sender RTT stat is not set.
+  non_sender_rtt_stats = receiver.GetNonSenderRTT();
+  EXPECT_FALSE(non_sender_rtt_stats.round_trip_time().has_value());
+}
+
+// Test that the receiver RTT stat resets when receiving a DLRR with a timestamp
+// of zero. This behavior is described in the standard, see
+// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime.
+TEST(RtcpReceiverTest, ReceiverRttResetOnDlrrWithZeroTimestamp) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = true;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  Random rand(0x0123456789abcdef);
+  const TimeDelta kRtt = TimeDelta::Millis(rand.Rand(1, 9 * 3600 * 1000));
+  const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+  NtpTime now = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp = CompactNtp(now);
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp, kDelayNtp));
+
+  receiver.IncomingPacket(xr.Build());
+
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_TRUE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_NEAR(non_sender_rtt_stats.round_trip_time()->ms(), kRtt.ms(), 1);
+
+  // Generate an XR+DLRR with zero timestamp.
+  rtcp::ExtendedReports xr2;
+  xr2.SetSenderSsrc(kSenderSsrc);
+  xr2.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, 0, kDelayNtp));
+
+  receiver.IncomingPacket(xr2.Build());
+
+  // Check that the non-sender RTT stat is not set.
+  non_sender_rtt_stats = receiver.GetNonSenderRTT();
+  EXPECT_FALSE(non_sender_rtt_stats.round_trip_time().has_value());
+}
+
+// Check that the receiver RTT works correctly when the remote SSRC changes.
+TEST(RtcpReceiverTest, ReceiverRttWithMultipleRemoteSsrcs) {
+  ReceiverMocks mocks;
+  auto config = DefaultConfiguration(&mocks);
+  config.non_sender_rtt_measurement = false;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+  receiver.SetNonSenderRttMeasurement(true);
+
+  Random rand(0x0123456789abcdef);
+  const TimeDelta kRtt = TimeDelta::Millis(rand.Rand(1, 9 * 3600 * 1000));
+  const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+  NtpTime now = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp = CompactNtp(now);
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp, kDelayNtp));
+
+  receiver.IncomingPacket(xr.Build());
+
+  // Generate an XR report for another SSRC.
+  const TimeDelta kRtt2 = TimeDelta::Millis(rand.Rand(1, 9 * 3600 * 1000));
+  const uint32_t kDelayNtp2 = rand.Rand(0, 0x7fffffff);
+  const TimeDelta kDelay2 = CompactNtpRttToTimeDelta(kDelayNtp2);
+  NtpTime now2 = mocks.clock.CurrentNtpTime();
+  uint32_t sent_ntp2 = CompactNtp(now2);
+  mocks.clock.AdvanceTime(kRtt2 + kDelay2);
+
+  rtcp::ExtendedReports xr2;
+  xr2.SetSenderSsrc(kSenderSsrc + 1);
+  xr2.AddDlrrItem(ReceiveTimeInfo(kReceiverMainSsrc, sent_ntp2, kDelayNtp2));
+
+  receiver.IncomingPacket(xr2.Build());
+
+  // Check that the non-sender RTT stats match the first XR.
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      receiver.GetNonSenderRTT();
+  EXPECT_TRUE(non_sender_rtt_stats.round_trip_time().has_value());
+  EXPECT_NEAR(non_sender_rtt_stats.round_trip_time()->ms(), kRtt.ms(), 1);
+  EXPECT_FALSE(non_sender_rtt_stats.total_round_trip_time().IsZero());
+  EXPECT_GT(non_sender_rtt_stats.round_trip_time_measurements(), 0);
+
+  // Change the remote SSRC and check that the stats match the second XR.
+  receiver.SetRemoteSSRC(kSenderSsrc + 1);
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats2 =
+      receiver.GetNonSenderRTT();
+  EXPECT_TRUE(non_sender_rtt_stats2.round_trip_time().has_value());
+  EXPECT_NEAR(non_sender_rtt_stats2.round_trip_time()->ms(), kRtt2.ms(), 1);
+  EXPECT_FALSE(non_sender_rtt_stats2.total_round_trip_time().IsZero());
+  EXPECT_GT(non_sender_rtt_stats2.round_trip_time_measurements(), 0);
+}
+
+TEST(RtcpReceiverTest, ConsumeReceivedXrReferenceTimeInfoInitiallyEmpty) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  EXPECT_THAT(receiver.ConsumeReceivedXrReferenceTimeInfo(), IsEmpty());
+}
+
+TEST(RtcpReceiverTest, ConsumeReceivedXrReferenceTimeInfo) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const NtpTime kNtp(0x10203, 0x40506);
+  const uint32_t kNtpMid = CompactNtp(kNtp);
+
+  rtcp::Rrtr rrtr;
+  rrtr.SetNtp(kNtp);
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  xr.SetRrtr(rrtr);
+
+  receiver.IncomingPacket(xr.Build());
+
+  mocks.clock.AdvanceTimeMilliseconds(1000);
+
+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
+      receiver.ConsumeReceivedXrReferenceTimeInfo();
+  ASSERT_THAT(last_xr_rtis, SizeIs(1));
+  EXPECT_EQ(kSenderSsrc, last_xr_rtis[0].ssrc);
+  EXPECT_EQ(kNtpMid, last_xr_rtis[0].last_rr);
+  EXPECT_EQ(65536U, last_xr_rtis[0].delay_since_last_rr);
+}
+
+TEST(RtcpReceiverTest,
+     ReceivedRrtrFromSameSsrcUpdatesReceivedReferenceTimeInfo) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const NtpTime kNtp1(0x10203, 0x40506);
+  const NtpTime kNtp2(0x11223, 0x44556);
+  const int64_t kDelayMs = 2000;
+
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kSenderSsrc);
+  rtcp::Rrtr rrtr1;
+  rrtr1.SetNtp(kNtp1);
+  xr.SetRrtr(rrtr1);
+  receiver.IncomingPacket(xr.Build());
+  mocks.clock.AdvanceTimeMilliseconds(kDelayMs);
+  rtcp::Rrtr rrtr2;
+  rrtr2.SetNtp(kNtp2);
+  xr.SetRrtr(rrtr2);
+  receiver.IncomingPacket(xr.Build());
+  mocks.clock.AdvanceTimeMilliseconds(kDelayMs);
+
+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
+      receiver.ConsumeReceivedXrReferenceTimeInfo();
+  ASSERT_THAT(last_xr_rtis, SizeIs(1));
+  EXPECT_EQ(kSenderSsrc, last_xr_rtis[0].ssrc);
+  EXPECT_EQ(CompactNtp(kNtp2), last_xr_rtis[0].last_rr);
+  EXPECT_EQ(kDelayMs * 65536 / 1000, last_xr_rtis[0].delay_since_last_rr);
+}
+
+TEST(RtcpReceiverTest, StoresLastReceivedRrtrPerSsrc) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const size_t kNumBufferedReports = 1;
+  const size_t kNumReports =
+      rtcp::ExtendedReports::kMaxNumberOfDlrrItems + kNumBufferedReports;
+  for (size_t i = 0; i < kNumReports; ++i) {
+    rtcp::ExtendedReports xr;
+    xr.SetSenderSsrc(i * 100);
+    rtcp::Rrtr rrtr;
+    rrtr.SetNtp(NtpTime(i * 200, i * 300));
+    xr.SetRrtr(rrtr);
+    receiver.IncomingPacket(xr.Build());
+    mocks.clock.AdvanceTimeMilliseconds(1000);
+  }
+
+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
+      receiver.ConsumeReceivedXrReferenceTimeInfo();
+  ASSERT_THAT(last_xr_rtis,
+              SizeIs(rtcp::ExtendedReports::kMaxNumberOfDlrrItems));
+  for (size_t i = 0; i < rtcp::ExtendedReports::kMaxNumberOfDlrrItems; ++i) {
+    EXPECT_EQ(i * 100, last_xr_rtis[i].ssrc);
+    EXPECT_EQ(CompactNtp(NtpTime(i * 200, i * 300)), last_xr_rtis[i].last_rr);
+    EXPECT_EQ(65536U * (kNumReports - i), last_xr_rtis[i].delay_since_last_rr);
+  }
+
+  last_xr_rtis = receiver.ConsumeReceivedXrReferenceTimeInfo();
+  ASSERT_THAT(last_xr_rtis, SizeIs(kNumBufferedReports));
+}
+
+TEST(RtcpReceiverTest, ReceiveReportTimeout) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint16_t kSequenceNumber = 1234;
+  mocks.clock.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
+
+  // No RR received, shouldn't trigger a timeout.
+  EXPECT_FALSE(receiver.RtcpRrTimeout());
+  EXPECT_FALSE(receiver.RtcpRrSequenceNumberTimeout());
+
+  // Add a RR and advance the clock just enough to not trigger a timeout.
+  rtcp::ReportBlock rb1;
+  rb1.SetMediaSsrc(kReceiverMainSsrc);
+  rb1.SetExtHighestSeqNum(kSequenceNumber);
+  rtcp::ReceiverReport rr1;
+  rr1.SetSenderSsrc(kSenderSsrc);
+  rr1.AddReportBlock(rb1);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rr1.Build());
+
+  mocks.clock.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs - 1);
+  EXPECT_FALSE(receiver.RtcpRrTimeout());
+  EXPECT_FALSE(receiver.RtcpRrSequenceNumberTimeout());
+
+  // Add a RR with the same extended max as the previous RR to trigger a
+  // sequence number timeout, but not a RR timeout.
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rr1.Build());
+
+  mocks.clock.AdvanceTimeMilliseconds(2);
+  EXPECT_FALSE(receiver.RtcpRrTimeout());
+  EXPECT_TRUE(receiver.RtcpRrSequenceNumberTimeout());
+
+  // Advance clock enough to trigger an RR timeout too.
+  mocks.clock.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
+  EXPECT_TRUE(receiver.RtcpRrTimeout());
+
+  // We should only get one timeout even though we still haven't received a new
+  // RR.
+  EXPECT_FALSE(receiver.RtcpRrTimeout());
+  EXPECT_FALSE(receiver.RtcpRrSequenceNumberTimeout());
+
+  // Add a new RR with increase sequence number to reset timers.
+  rtcp::ReportBlock rb2;
+  rb2.SetMediaSsrc(kReceiverMainSsrc);
+  rb2.SetExtHighestSeqNum(kSequenceNumber + 1);
+  rtcp::ReceiverReport rr2;
+  rr2.SetSenderSsrc(kSenderSsrc);
+  rr2.AddReportBlock(rb2);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rr2.Build());
+
+  EXPECT_FALSE(receiver.RtcpRrTimeout());
+  EXPECT_FALSE(receiver.RtcpRrSequenceNumberTimeout());
+
+  // Verify we can get a timeout again once we've received new RR.
+  mocks.clock.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rr2.Build());
+
+  mocks.clock.AdvanceTimeMilliseconds(kRtcpIntervalMs + 1);
+  EXPECT_FALSE(receiver.RtcpRrTimeout());
+  EXPECT_TRUE(receiver.RtcpRrSequenceNumberTimeout());
+
+  mocks.clock.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
+  EXPECT_TRUE(receiver.RtcpRrTimeout());
+}
+
+TEST(RtcpReceiverTest, TmmbrReceivedWithNoIncomingPacket) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  EXPECT_THAT(receiver.TmmbrReceived(), IsEmpty());
+}
+
+TEST(RtcpReceiverTest, TmmbrPacketAccepted) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint32_t kBitrateBps = 30000;
+  auto tmmbr = std::make_unique<rtcp::Tmmbr>();
+  tmmbr->SetSenderSsrc(kSenderSsrc);
+  tmmbr->AddTmmbr(rtcp::TmmbItem(kReceiverMainSsrc, kBitrateBps, 0));
+  auto sr = std::make_unique<rtcp::SenderReport>();
+  sr->SetSenderSsrc(kSenderSsrc);
+  rtcp::CompoundPacket compound;
+  compound.Append(std::move(sr));
+  compound.Append(std::move(tmmbr));
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.rtp_rtcp_impl, SetTmmbn(SizeIs(1)));
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedEstimatedBitrate(kBitrateBps));
+  receiver.IncomingPacket(compound.Build());
+
+  std::vector<rtcp::TmmbItem> tmmbr_received = receiver.TmmbrReceived();
+  ASSERT_EQ(1u, tmmbr_received.size());
+  EXPECT_EQ(kBitrateBps, tmmbr_received[0].bitrate_bps());
+  EXPECT_EQ(kSenderSsrc, tmmbr_received[0].ssrc());
+}
+
+TEST(RtcpReceiverTest, TmmbrPacketNotForUsIgnored) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint32_t kBitrateBps = 30000;
+  auto tmmbr = std::make_unique<rtcp::Tmmbr>();
+  tmmbr->SetSenderSsrc(kSenderSsrc);
+  tmmbr->AddTmmbr(rtcp::TmmbItem(kNotToUsSsrc, kBitrateBps, 0));
+
+  auto sr = std::make_unique<rtcp::SenderReport>();
+  sr->SetSenderSsrc(kSenderSsrc);
+  rtcp::CompoundPacket compound;
+  compound.Append(std::move(sr));
+  compound.Append(std::move(tmmbr));
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedEstimatedBitrate).Times(0);
+  receiver.IncomingPacket(compound.Build());
+
+  EXPECT_EQ(0u, receiver.TmmbrReceived().size());
+}
+
+TEST(RtcpReceiverTest, TmmbrPacketZeroRateIgnored) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  auto tmmbr = std::make_unique<rtcp::Tmmbr>();
+  tmmbr->SetSenderSsrc(kSenderSsrc);
+  tmmbr->AddTmmbr(rtcp::TmmbItem(kReceiverMainSsrc, 0, 0));
+  auto sr = std::make_unique<rtcp::SenderReport>();
+  sr->SetSenderSsrc(kSenderSsrc);
+  rtcp::CompoundPacket compound;
+  compound.Append(std::move(sr));
+  compound.Append(std::move(tmmbr));
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedEstimatedBitrate).Times(0);
+  receiver.IncomingPacket(compound.Build());
+
+  EXPECT_EQ(0u, receiver.TmmbrReceived().size());
+}
+
+TEST(RtcpReceiverTest, TmmbrThreeConstraintsTimeOut) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  // Inject 3 packets "from" kSenderSsrc, kSenderSsrc+1, kSenderSsrc+2.
+  // The times of arrival are starttime + 0, starttime + 5 and starttime + 10.
+  for (uint32_t ssrc = kSenderSsrc; ssrc < kSenderSsrc + 3; ++ssrc) {
+    auto tmmbr = std::make_unique<rtcp::Tmmbr>();
+    tmmbr->SetSenderSsrc(ssrc);
+    tmmbr->AddTmmbr(rtcp::TmmbItem(kReceiverMainSsrc, 30000, 0));
+    auto sr = std::make_unique<rtcp::SenderReport>();
+    sr->SetSenderSsrc(ssrc);
+    rtcp::CompoundPacket compound;
+    compound.Append(std::move(sr));
+    compound.Append(std::move(tmmbr));
+
+    EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+    EXPECT_CALL(mocks.rtp_rtcp_impl, SetTmmbn);
+    EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+    EXPECT_CALL(mocks.bandwidth_observer, OnReceivedEstimatedBitrate);
+    receiver.IncomingPacket(compound.Build());
+
+    // 5 seconds between each packet.
+    mocks.clock.AdvanceTimeMilliseconds(5000);
+  }
+  // It is now starttime + 15.
+  EXPECT_THAT(receiver.TmmbrReceived(),
+              AllOf(SizeIs(3),
+                    Each(Property(&rtcp::TmmbItem::bitrate_bps, Eq(30'000U)))));
+
+  // We expect the timeout to be 25 seconds. Advance the clock by 12
+  // seconds, timing out the first packet.
+  mocks.clock.AdvanceTimeMilliseconds(12000);
+  EXPECT_THAT(receiver.TmmbrReceived(),
+              UnorderedElementsAre(
+                  Property(&rtcp::TmmbItem::ssrc, Eq(kSenderSsrc + 1)),
+                  Property(&rtcp::TmmbItem::ssrc, Eq(kSenderSsrc + 2))));
+}
+
+TEST(RtcpReceiverTest,
+     VerifyBlockAndTimestampObtainedFromReportBlockDataObserver) {
+  ReceiverMocks mocks;
+  MockReportBlockDataObserverImpl observer;
+  RtpRtcpInterface::Configuration config = DefaultConfiguration(&mocks);
+  config.report_block_data_observer = &observer;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint8_t kFractionLoss = 3;
+  const uint32_t kCumulativeLoss = 7;
+  const uint32_t kJitter = 9;
+  const uint16_t kSequenceNumber = 1234;
+  const int64_t kNtpNowMs =
+      mocks.clock.CurrentNtpInMilliseconds() - rtc::kNtpJan1970Millisecs;
+
+  rtcp::ReportBlock rtcp_block;
+  rtcp_block.SetMediaSsrc(kReceiverMainSsrc);
+  rtcp_block.SetExtHighestSeqNum(kSequenceNumber);
+  rtcp_block.SetFractionLost(kFractionLoss);
+  rtcp_block.SetCumulativeLost(kCumulativeLoss);
+  rtcp_block.SetJitter(kJitter);
+
+  rtcp::ReceiverReport rtcp_report;
+  rtcp_report.SetSenderSsrc(kSenderSsrc);
+  rtcp_report.AddReportBlock(rtcp_block);
+  EXPECT_CALL(observer, OnReportBlockDataUpdated)
+      .WillOnce([&](ReportBlockData report_block_data) {
+        const auto& report_block = report_block_data.report_block();
+        EXPECT_EQ(rtcp_block.source_ssrc(), report_block.source_ssrc);
+        EXPECT_EQ(kSenderSsrc, report_block.sender_ssrc);
+        EXPECT_EQ(rtcp_block.fraction_lost(), report_block.fraction_lost);
+        EXPECT_EQ(rtcp_block.cumulative_lost_signed(),
+                  report_block.packets_lost);
+        EXPECT_EQ(rtcp_block.extended_high_seq_num(),
+                  report_block.extended_highest_sequence_number);
+        EXPECT_EQ(rtcp_block.jitter(), report_block.jitter);
+        EXPECT_EQ(kNtpNowMs * rtc::kNumMicrosecsPerMillisec,
+                  report_block_data.report_block_timestamp_utc_us());
+        // No RTT is calculated in this test.
+        EXPECT_EQ(0u, report_block_data.num_rtts());
+      });
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rtcp_report.Build());
+}
+
+TEST(RtcpReceiverTest, VerifyRttObtainedFromReportBlockDataObserver) {
+  ReceiverMocks mocks;
+  MockReportBlockDataObserverImpl observer;
+  RtpRtcpInterface::Configuration config = DefaultConfiguration(&mocks);
+  config.report_block_data_observer = &observer;
+  RTCPReceiver receiver(config, &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const TimeDelta kRtt = TimeDelta::Millis(120);
+  const uint32_t kDelayNtp = 123000;
+  const TimeDelta kDelay = CompactNtpRttToTimeDelta(kDelayNtp);
+
+  uint32_t sent_ntp = CompactNtp(mocks.clock.CurrentNtpTime());
+  mocks.clock.AdvanceTime(kRtt + kDelay);
+
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  rtcp::ReportBlock block;
+  block.SetMediaSsrc(kReceiverMainSsrc);
+  block.SetLastSr(sent_ntp);
+  block.SetDelayLastSr(kDelayNtp);
+  sr.AddReportBlock(block);
+  block.SetMediaSsrc(kReceiverExtraSsrc);
+  block.SetLastSr(0);
+  sr.AddReportBlock(block);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  InSequence sequence;
+  EXPECT_CALL(observer, OnReportBlockDataUpdated)
+      .WillOnce([&](ReportBlockData report_block_data) {
+        EXPECT_EQ(kReceiverMainSsrc,
+                  report_block_data.report_block().source_ssrc);
+        EXPECT_EQ(1u, report_block_data.num_rtts());
+        EXPECT_EQ(kRtt.ms(), report_block_data.min_rtt_ms());
+        EXPECT_EQ(kRtt.ms(), report_block_data.max_rtt_ms());
+        EXPECT_EQ(kRtt.ms(), report_block_data.sum_rtt_ms());
+        EXPECT_EQ(kRtt.ms(), report_block_data.last_rtt_ms());
+      });
+  EXPECT_CALL(observer, OnReportBlockDataUpdated)
+      .WillOnce([](ReportBlockData report_block_data) {
+        EXPECT_EQ(kReceiverExtraSsrc,
+                  report_block_data.report_block().source_ssrc);
+        EXPECT_EQ(0u, report_block_data.num_rtts());
+      });
+  receiver.IncomingPacket(sr.Build());
+}
+
+TEST(RtcpReceiverTest, GetReportBlockDataAfterOneReportBlock) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint16_t kSequenceNumber = 1234;
+
+  rtcp::ReportBlock rtcp_block;
+  rtcp_block.SetMediaSsrc(kReceiverMainSsrc);
+  rtcp_block.SetExtHighestSeqNum(kSequenceNumber);
+
+  rtcp::ReceiverReport rtcp_report;
+  rtcp_report.SetSenderSsrc(kSenderSsrc);
+  rtcp_report.AddReportBlock(rtcp_block);
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rtcp_report.Build());
+
+  auto report_block_datas = receiver.GetLatestReportBlockData();
+  ASSERT_THAT(report_block_datas, SizeIs(1));
+  EXPECT_EQ(kReceiverMainSsrc,
+            report_block_datas[0].report_block().source_ssrc);
+  EXPECT_EQ(
+      kSequenceNumber,
+      report_block_datas[0].report_block().extended_highest_sequence_number);
+}
+
+TEST(RtcpReceiverTest, GetReportBlockDataAfterTwoReportBlocksOfSameSsrc) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint16_t kSequenceNumber1 = 1234;
+  const uint16_t kSequenceNumber2 = 1235;
+
+  rtcp::ReportBlock rtcp_block1;
+  rtcp_block1.SetMediaSsrc(kReceiverMainSsrc);
+  rtcp_block1.SetExtHighestSeqNum(kSequenceNumber1);
+
+  rtcp::ReceiverReport rtcp_report1;
+  rtcp_report1.SetSenderSsrc(kSenderSsrc);
+  rtcp_report1.AddReportBlock(rtcp_block1);
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rtcp_report1.Build());
+
+  // Inject a report block with an increased the sequence number for the same
+  // source SSRC.
+  rtcp::ReportBlock rtcp_block2;
+  rtcp_block2.SetMediaSsrc(kReceiverMainSsrc);
+  rtcp_block2.SetExtHighestSeqNum(kSequenceNumber2);
+
+  rtcp::ReceiverReport rtcp_report2;
+  rtcp_report2.SetSenderSsrc(kSenderSsrc);
+  rtcp_report2.AddReportBlock(rtcp_block2);
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rtcp_report2.Build());
+
+  // Only the latest block should be returned.
+  auto report_block_datas = receiver.GetLatestReportBlockData();
+  ASSERT_THAT(report_block_datas, SizeIs(1));
+  EXPECT_EQ(kReceiverMainSsrc,
+            report_block_datas[0].report_block().source_ssrc);
+  EXPECT_EQ(
+      kSequenceNumber2,
+      report_block_datas[0].report_block().extended_highest_sequence_number);
+}
+
+TEST(RtcpReceiverTest, GetReportBlockDataAfterTwoReportBlocksOfDifferentSsrcs) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint16_t kSequenceNumber1 = 1234;
+  const uint16_t kSequenceNumber2 = 42;
+
+  rtcp::ReportBlock rtcp_block1;
+  rtcp_block1.SetMediaSsrc(kReceiverMainSsrc);
+  rtcp_block1.SetExtHighestSeqNum(kSequenceNumber1);
+
+  rtcp::ReceiverReport rtcp_report1;
+  rtcp_report1.SetSenderSsrc(kSenderSsrc);
+  rtcp_report1.AddReportBlock(rtcp_block1);
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rtcp_report1.Build());
+
+  // Inject a report block for a different source SSRC.
+  rtcp::ReportBlock rtcp_block2;
+  rtcp_block2.SetMediaSsrc(kReceiverExtraSsrc);
+  rtcp_block2.SetExtHighestSeqNum(kSequenceNumber2);
+
+  rtcp::ReceiverReport rtcp_report2;
+  rtcp_report2.SetSenderSsrc(kSenderSsrc);
+  rtcp_report2.AddReportBlock(rtcp_block2);
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedRtcpReportBlocks);
+  EXPECT_CALL(mocks.bandwidth_observer, OnReceivedRtcpReceiverReport);
+  receiver.IncomingPacket(rtcp_report2.Build());
+
+  // Both report blocks should be returned.
+  auto report_block_datas = receiver.GetLatestReportBlockData();
+  ASSERT_THAT(report_block_datas, SizeIs(2));
+  EXPECT_EQ(kReceiverMainSsrc,
+            report_block_datas[0].report_block().source_ssrc);
+  EXPECT_EQ(
+      kSequenceNumber1,
+      report_block_datas[0].report_block().extended_highest_sequence_number);
+  EXPECT_EQ(kReceiverExtraSsrc,
+            report_block_datas[1].report_block().source_ssrc);
+  EXPECT_EQ(
+      kSequenceNumber2,
+      report_block_datas[1].report_block().extended_highest_sequence_number);
+}
+
+TEST(RtcpReceiverTest, ReceivesTransportFeedback) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::TransportFeedback packet;
+  packet.SetMediaSsrc(kReceiverMainSsrc);
+  packet.SetSenderSsrc(kSenderSsrc);
+  packet.SetBase(1, Timestamp::Millis(1));
+  packet.AddReceivedPacket(1, Timestamp::Millis(1));
+
+  EXPECT_CALL(
+      mocks.transport_feedback_observer,
+      OnTransportFeedback(AllOf(
+          Property(&rtcp::TransportFeedback::media_ssrc, kReceiverMainSsrc),
+          Property(&rtcp::TransportFeedback::sender_ssrc, kSenderSsrc))));
+  receiver.IncomingPacket(packet.Build());
+}
+
+TEST(RtcpReceiverTest, ReceivesRemb) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint32_t kBitrateBps = 500000;
+  rtcp::Remb remb;
+  remb.SetSenderSsrc(kSenderSsrc);
+  remb.SetBitrateBps(kBitrateBps);
+
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedEstimatedBitrate(kBitrateBps));
+  receiver.IncomingPacket(remb.Build());
+}
+
+TEST(RtcpReceiverTest, HandlesInvalidTransportFeedback) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  // Send a compound packet with a TransportFeedback followed by something else.
+  auto packet = std::make_unique<rtcp::TransportFeedback>();
+  packet->SetMediaSsrc(kReceiverMainSsrc);
+  packet->SetSenderSsrc(kSenderSsrc);
+  packet->SetBase(1, Timestamp::Millis(1));
+  packet->AddReceivedPacket(1, Timestamp::Millis(1));
+
+  static uint32_t kBitrateBps = 50000;
+  auto remb = std::make_unique<rtcp::Remb>();
+  remb->SetSenderSsrc(kSenderSsrc);
+  remb->SetBitrateBps(kBitrateBps);
+  rtcp::CompoundPacket compound;
+  compound.Append(std::move(packet));
+  compound.Append(std::move(remb));
+  rtc::Buffer built_packet = compound.Build();
+
+  // Modify the TransportFeedback packet so that it is invalid.
+  const size_t kStatusCountOffset = 14;
+  ByteWriter<uint16_t>::WriteBigEndian(&built_packet.data()[kStatusCountOffset],
+                                       42);
+
+  // Stress no transport feedback is expected.
+  EXPECT_CALL(mocks.transport_feedback_observer, OnTransportFeedback).Times(0);
+  // But remb should be processed and cause a callback
+  EXPECT_CALL(mocks.bandwidth_observer,
+              OnReceivedEstimatedBitrate(kBitrateBps));
+  receiver.IncomingPacket(built_packet);
+}
+
+TEST(RtcpReceiverTest, Nack) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint16_t kNackList1[] = {1, 2, 3, 5};
+  const uint16_t kNackList23[] = {5, 7, 30, 40, 41, 58, 59, 61, 63};
+  const size_t kNackListLength2 = 4;
+  const size_t kNackListLength3 = arraysize(kNackList23) - kNackListLength2;
+  std::set<uint16_t> nack_set;
+  nack_set.insert(std::begin(kNackList1), std::end(kNackList1));
+  nack_set.insert(std::begin(kNackList23), std::end(kNackList23));
+
+  auto nack1 = std::make_unique<rtcp::Nack>();
+  nack1->SetSenderSsrc(kSenderSsrc);
+  nack1->SetMediaSsrc(kReceiverMainSsrc);
+  nack1->SetPacketIds(kNackList1, arraysize(kNackList1));
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl,
+              OnReceivedNack(ElementsAreArray(kNackList1)));
+  EXPECT_CALL(mocks.packet_type_counter_observer,
+              RtcpPacketTypesCounterUpdated(
+                  kReceiverMainSsrc,
+                  AllOf(Field(&RtcpPacketTypeCounter::nack_requests,
+                              arraysize(kNackList1)),
+                        Field(&RtcpPacketTypeCounter::unique_nack_requests,
+                              arraysize(kNackList1)))));
+  receiver.IncomingPacket(nack1->Build());
+
+  auto nack2 = std::make_unique<rtcp::Nack>();
+  nack2->SetSenderSsrc(kSenderSsrc);
+  nack2->SetMediaSsrc(kReceiverMainSsrc);
+  nack2->SetPacketIds(kNackList23, kNackListLength2);
+
+  auto nack3 = std::make_unique<rtcp::Nack>();
+  nack3->SetSenderSsrc(kSenderSsrc);
+  nack3->SetMediaSsrc(kReceiverMainSsrc);
+  nack3->SetPacketIds(kNackList23 + kNackListLength2, kNackListLength3);
+
+  rtcp::CompoundPacket two_nacks;
+  two_nacks.Append(std::move(nack2));
+  two_nacks.Append(std::move(nack3));
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl,
+              OnReceivedNack(ElementsAreArray(kNackList23)));
+  EXPECT_CALL(mocks.packet_type_counter_observer,
+              RtcpPacketTypesCounterUpdated(
+                  kReceiverMainSsrc,
+                  AllOf(Field(&RtcpPacketTypeCounter::nack_requests,
+                              arraysize(kNackList1) + arraysize(kNackList23)),
+                        Field(&RtcpPacketTypeCounter::unique_nack_requests,
+                              nack_set.size()))));
+  receiver.IncomingPacket(two_nacks.Build());
+}
+
+TEST(RtcpReceiverTest, NackNotForUsIgnored) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  const uint16_t kNackList1[] = {1, 2, 3, 5};
+  const size_t kNackListLength1 = std::end(kNackList1) - std::begin(kNackList1);
+
+  rtcp::Nack nack;
+  nack.SetSenderSsrc(kSenderSsrc);
+  nack.SetMediaSsrc(kNotToUsSsrc);
+  nack.SetPacketIds(kNackList1, kNackListLength1);
+
+  EXPECT_CALL(mocks.packet_type_counter_observer,
+              RtcpPacketTypesCounterUpdated(
+                  _, Field(&RtcpPacketTypeCounter::nack_requests, 0)));
+  receiver.IncomingPacket(nack.Build());
+}
+
+TEST(RtcpReceiverTest, ForceSenderReport) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  rtcp::RapidResyncRequest rr;
+  rr.SetSenderSsrc(kSenderSsrc);
+  rr.SetMediaSsrc(kReceiverMainSsrc);
+
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnRequestSendReport());
+  receiver.IncomingPacket(rr.Build());
+}
+
+TEST(RtcpReceiverTest, ReceivesTargetBitrate) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  VideoBitrateAllocation expected_allocation;
+  expected_allocation.SetBitrate(0, 0, 10000);
+  expected_allocation.SetBitrate(0, 1, 20000);
+  expected_allocation.SetBitrate(1, 0, 40000);
+  expected_allocation.SetBitrate(1, 1, 80000);
+
+  rtcp::TargetBitrate bitrate;
+  bitrate.AddTargetBitrate(0, 0, expected_allocation.GetBitrate(0, 0) / 1000);
+  bitrate.AddTargetBitrate(0, 1, expected_allocation.GetBitrate(0, 1) / 1000);
+  bitrate.AddTargetBitrate(1, 0, expected_allocation.GetBitrate(1, 0) / 1000);
+  bitrate.AddTargetBitrate(1, 1, expected_allocation.GetBitrate(1, 1) / 1000);
+
+  rtcp::ExtendedReports xr;
+  xr.SetTargetBitrate(bitrate);
+
+  // Wrong sender ssrc, target bitrate should be discarded.
+  xr.SetSenderSsrc(kSenderSsrc + 1);
+  EXPECT_CALL(mocks.bitrate_allocation_observer,
+              OnBitrateAllocationUpdated(expected_allocation))
+      .Times(0);
+  receiver.IncomingPacket(xr.Build());
+
+  // Set correct ssrc, callback should be called once.
+  xr.SetSenderSsrc(kSenderSsrc);
+  EXPECT_CALL(mocks.bitrate_allocation_observer,
+              OnBitrateAllocationUpdated(expected_allocation));
+  receiver.IncomingPacket(xr.Build());
+}
+
+TEST(RtcpReceiverTest, HandlesIncorrectTargetBitrate) {
+  ReceiverMocks mocks;
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  VideoBitrateAllocation expected_allocation;
+  expected_allocation.SetBitrate(0, 0, 10000);
+
+  rtcp::TargetBitrate bitrate;
+  bitrate.AddTargetBitrate(0, 0, expected_allocation.GetBitrate(0, 0) / 1000);
+  bitrate.AddTargetBitrate(0, kMaxTemporalStreams, 20000);
+  bitrate.AddTargetBitrate(kMaxSpatialLayers, 0, 40000);
+
+  rtcp::ExtendedReports xr;
+  xr.SetTargetBitrate(bitrate);
+  xr.SetSenderSsrc(kSenderSsrc);
+
+  EXPECT_CALL(mocks.bitrate_allocation_observer,
+              OnBitrateAllocationUpdated(expected_allocation));
+  receiver.IncomingPacket(xr.Build());
+}
+
+TEST(RtcpReceiverTest, ChangeLocalMediaSsrc) {
+  ReceiverMocks mocks;
+  // Construct a receiver with `kReceiverMainSsrc` (default) local media ssrc.
+  RTCPReceiver receiver(DefaultConfiguration(&mocks), &mocks.rtp_rtcp_impl);
+  receiver.SetRemoteSSRC(kSenderSsrc);
+
+  constexpr uint32_t kSecondarySsrc = kReceiverMainSsrc + 1;
+
+  // Expect to only get the `OnReceivedNack()` callback once since we'll
+  // configure it for the `kReceiverMainSsrc` media ssrc.
+  EXPECT_CALL(mocks.rtp_rtcp_impl, OnReceivedNack);
+
+  // We'll get two callbacks to RtcpPacketTypesCounterUpdated, one for each
+  // call to `IncomingPacket`, differentiated by the local media ssrc.
+  EXPECT_CALL(mocks.packet_type_counter_observer,
+              RtcpPacketTypesCounterUpdated(kReceiverMainSsrc, _));
+  EXPECT_CALL(mocks.packet_type_counter_observer,
+              RtcpPacketTypesCounterUpdated(kSecondarySsrc, _));
+
+  // Construct a test nack packet with media ssrc set to `kReceiverMainSsrc`.
+  rtcp::Nack nack;
+  nack.SetSenderSsrc(kSenderSsrc);
+  nack.SetMediaSsrc(kReceiverMainSsrc);
+  const uint16_t kNackList[] = {1, 2, 3, 5};
+  nack.SetPacketIds(kNackList, std::size(kNackList));
+
+  // Deliver the first callback.
+  receiver.IncomingPacket(nack.Build());
+
+  // Change the set local media ssrc.
+  receiver.set_local_media_ssrc(kSecondarySsrc);
+
+  // Deliver another packet - this time there will be no callback to
+  // OnReceivedNack due to the ssrc not matching.
+  receiver.IncomingPacket(nack.Build());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender.cc
new file mode 100644
index 0000000000..983851a55b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -0,0 +1,974 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_sender.h"
+
+#include <string.h>  // memcpy
+
+#include <algorithm>  // std::min
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/rtp_headers.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "modules/rtp_rtcp/source/tmmbr_help.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+namespace {
+const uint32_t kRtcpAnyExtendedReports = kRtcpXrReceiverReferenceTime |
+                                         kRtcpXrDlrrReportBlock |
+                                         kRtcpXrTargetBitrate;
+constexpr int32_t kDefaultVideoReportInterval = 1000;
+constexpr int32_t kDefaultAudioReportInterval = 5000;
+}  // namespace
+
+// Helper to put several RTCP packets into lower layer datagram RTCP packet.
+class RTCPSender::PacketSender {
+ public:
+  PacketSender(rtcp::RtcpPacket::PacketReadyCallback callback,
+               size_t max_packet_size)
+      : callback_(callback), max_packet_size_(max_packet_size) {
+    RTC_CHECK_LE(max_packet_size, IP_PACKET_SIZE);
+  }
+  ~PacketSender() { RTC_DCHECK_EQ(index_, 0) << "Unsent rtcp packet."; }
+
+  // Appends a packet to pending compound packet.
+  // Sends rtcp packet if buffer is full and resets the buffer.
+  void AppendPacket(const rtcp::RtcpPacket& packet) {
+    packet.Create(buffer_, &index_, max_packet_size_, callback_);
+  }
+
+  // Sends pending rtcp packet.
+  void Send() {
+    if (index_ > 0) {
+      callback_(rtc::ArrayView<const uint8_t>(buffer_, index_));
+      index_ = 0;
+    }
+  }
+
+ private:
+  const rtcp::RtcpPacket::PacketReadyCallback callback_;
+  const size_t max_packet_size_;
+  size_t index_ = 0;
+  uint8_t buffer_[IP_PACKET_SIZE];
+};
+
+RTCPSender::FeedbackState::FeedbackState()
+    : packets_sent(0),
+      media_bytes_sent(0),
+      send_bitrate(0),
+      last_rr_ntp_secs(0),
+      last_rr_ntp_frac(0),
+      remote_sr(0),
+      receiver(nullptr) {}
+
+RTCPSender::FeedbackState::FeedbackState(const FeedbackState&) = default;
+
+RTCPSender::FeedbackState::FeedbackState(FeedbackState&&) = default;
+
+RTCPSender::FeedbackState::~FeedbackState() = default;
+
+class RTCPSender::RtcpContext {
+ public:
+  RtcpContext(const FeedbackState& feedback_state,
+              int32_t nack_size,
+              const uint16_t* nack_list,
+              Timestamp now)
+      : feedback_state_(feedback_state),
+        nack_size_(nack_size),
+        nack_list_(nack_list),
+        now_(now) {}
+
+  const FeedbackState& feedback_state_;
+  const int32_t nack_size_;
+  const uint16_t* nack_list_;
+  const Timestamp now_;
+};
+
+RTCPSender::Configuration RTCPSender::Configuration::FromRtpRtcpConfiguration(
+    const RtpRtcpInterface::Configuration& configuration) {
+  RTCPSender::Configuration result;
+  result.audio = configuration.audio;
+  result.local_media_ssrc = configuration.local_media_ssrc;
+  result.clock = configuration.clock;
+  result.outgoing_transport = configuration.outgoing_transport;
+  result.non_sender_rtt_measurement = configuration.non_sender_rtt_measurement;
+  result.event_log = configuration.event_log;
+  if (configuration.rtcp_report_interval_ms) {
+    result.rtcp_report_interval =
+        TimeDelta::Millis(configuration.rtcp_report_interval_ms);
+  }
+  result.receive_statistics = configuration.receive_statistics;
+  result.rtcp_packet_type_counter_observer =
+      configuration.rtcp_packet_type_counter_observer;
+  return result;
+}
+
+RTCPSender::RTCPSender(Configuration config)
+    : audio_(config.audio),
+      ssrc_(config.local_media_ssrc),
+      clock_(config.clock),
+      random_(clock_->TimeInMicroseconds()),
+      method_(RtcpMode::kOff),
+      event_log_(config.event_log),
+      transport_(config.outgoing_transport),
+      report_interval_(config.rtcp_report_interval.value_or(
+          TimeDelta::Millis(config.audio ? kDefaultAudioReportInterval
+                                         : kDefaultVideoReportInterval))),
+      schedule_next_rtcp_send_evaluation_function_(
+          std::move(config.schedule_next_rtcp_send_evaluation_function)),
+      sending_(false),
+      timestamp_offset_(0),
+      last_rtp_timestamp_(0),
+      remote_ssrc_(0),
+      receive_statistics_(config.receive_statistics),
+
+      sequence_number_fir_(0),
+
+      remb_bitrate_(0),
+
+      tmmbr_send_bps_(0),
+      packet_oh_send_(0),
+      max_packet_size_(IP_PACKET_SIZE - 28),  // IPv4 + UDP by default.
+
+      xr_send_receiver_reference_time_enabled_(
+          config.non_sender_rtt_measurement),
+      packet_type_counter_observer_(config.rtcp_packet_type_counter_observer),
+      send_video_bitrate_allocation_(false),
+      last_payload_type_(-1) {
+  RTC_DCHECK(transport_ != nullptr);
+
+  builders_[kRtcpSr] = &RTCPSender::BuildSR;
+  builders_[kRtcpRr] = &RTCPSender::BuildRR;
+  builders_[kRtcpSdes] = &RTCPSender::BuildSDES;
+  builders_[kRtcpPli] = &RTCPSender::BuildPLI;
+  builders_[kRtcpFir] = &RTCPSender::BuildFIR;
+  builders_[kRtcpRemb] = &RTCPSender::BuildREMB;
+  builders_[kRtcpBye] = &RTCPSender::BuildBYE;
+  builders_[kRtcpLossNotification] = &RTCPSender::BuildLossNotification;
+  builders_[kRtcpTmmbr] = &RTCPSender::BuildTMMBR;
+  builders_[kRtcpTmmbn] = &RTCPSender::BuildTMMBN;
+  builders_[kRtcpNack] = &RTCPSender::BuildNACK;
+  builders_[kRtcpAnyExtendedReports] = &RTCPSender::BuildExtendedReports;
+}
+
+RTCPSender::~RTCPSender() {}
+
+RtcpMode RTCPSender::Status() const {
+  MutexLock lock(&mutex_rtcp_sender_);
+  return method_;
+}
+
+void RTCPSender::SetRTCPStatus(RtcpMode new_method) {
+  MutexLock lock(&mutex_rtcp_sender_);
+
+  if (new_method == RtcpMode::kOff) {
+    next_time_to_send_rtcp_ = absl::nullopt;
+  } else if (method_ == RtcpMode::kOff) {
+    // When switching on, reschedule the next packet
+    SetNextRtcpSendEvaluationDuration(RTCP_INTERVAL_RAPID_SYNC_MS / 2);
+  }
+  method_ = new_method;
+}
+
+bool RTCPSender::Sending() const {
+  MutexLock lock(&mutex_rtcp_sender_);
+  return sending_;
+}
+
+void RTCPSender::SetSendingStatus(const FeedbackState& feedback_state,
+                                  bool sending) {
+  bool sendRTCPBye = false;
+  {
+    MutexLock lock(&mutex_rtcp_sender_);
+
+    if (method_ != RtcpMode::kOff) {
+      if (sending == false && sending_ == true) {
+        // Trigger RTCP bye
+        sendRTCPBye = true;
+      }
+    }
+    sending_ = sending;
+  }
+  if (sendRTCPBye) {
+    if (SendRTCP(feedback_state, kRtcpBye) != 0) {
+      RTC_LOG(LS_WARNING) << "Failed to send RTCP BYE";
+    }
+  }
+}
+
+void RTCPSender::SetNonSenderRttMeasurement(bool enabled) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  xr_send_receiver_reference_time_enabled_ = enabled;
+}
+
+int32_t RTCPSender::SendLossNotification(const FeedbackState& feedback_state,
+                                         uint16_t last_decoded_seq_num,
+                                         uint16_t last_received_seq_num,
+                                         bool decodability_flag,
+                                         bool buffering_allowed) {
+  int32_t error_code = -1;
+  auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
+    transport_->SendRtcp(packet.data(), packet.size());
+    error_code = 0;
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
+    }
+  };
+  absl::optional<PacketSender> sender;
+  {
+    MutexLock lock(&mutex_rtcp_sender_);
+
+    if (!loss_notification_.Set(last_decoded_seq_num, last_received_seq_num,
+                                decodability_flag)) {
+      return -1;
+    }
+
+    SetFlag(kRtcpLossNotification, /*is_volatile=*/true);
+
+    if (buffering_allowed) {
+      // The loss notification will be batched with additional feedback
+      // messages.
+      return 0;
+    }
+
+    sender.emplace(callback, max_packet_size_);
+    auto result = ComputeCompoundRTCPPacket(
+        feedback_state, RTCPPacketType::kRtcpLossNotification, 0, nullptr,
+        *sender);
+    if (result) {
+      return *result;
+    }
+  }
+  sender->Send();
+
+  return error_code;
+}
+
+void RTCPSender::SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) {
+  RTC_CHECK_GE(bitrate_bps, 0);
+  MutexLock lock(&mutex_rtcp_sender_);
+  if (method_ == RtcpMode::kOff) {
+    RTC_LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
+    return;
+  }
+  remb_bitrate_ = bitrate_bps;
+  remb_ssrcs_ = std::move(ssrcs);
+
+  SetFlag(kRtcpRemb, /*is_volatile=*/false);
+  // Send a REMB immediately if we have a new REMB. The frequency of REMBs is
+  // throttled by the caller.
+  SetNextRtcpSendEvaluationDuration(TimeDelta::Zero());
+}
+
+void RTCPSender::UnsetRemb() {
+  MutexLock lock(&mutex_rtcp_sender_);
+  // Stop sending REMB each report until it is reenabled and REMB data set.
+  ConsumeFlag(kRtcpRemb, /*forced=*/true);
+}
+
+bool RTCPSender::TMMBR() const {
+  MutexLock lock(&mutex_rtcp_sender_);
+  return IsFlagPresent(RTCPPacketType::kRtcpTmmbr);
+}
+
+void RTCPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  max_packet_size_ = max_packet_size;
+}
+
+void RTCPSender::SetTimestampOffset(uint32_t timestamp_offset) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  timestamp_offset_ = timestamp_offset;
+}
+
+void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
+                                absl::optional<Timestamp> capture_time,
+                                absl::optional<int8_t> payload_type) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  // For compatibility with clients who don't set payload type correctly on all
+  // calls.
+  if (payload_type.has_value()) {
+    last_payload_type_ = *payload_type;
+  }
+  last_rtp_timestamp_ = rtp_timestamp;
+  if (!capture_time.has_value()) {
+    // We don't currently get a capture time from VoiceEngine.
+    last_frame_capture_time_ = clock_->CurrentTime();
+  } else {
+    last_frame_capture_time_ = *capture_time;
+  }
+}
+
+void RTCPSender::SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  rtp_clock_rates_khz_[payload_type] = rtp_clock_rate_hz / 1000;
+}
+
+uint32_t RTCPSender::SSRC() const {
+  MutexLock lock(&mutex_rtcp_sender_);
+  return ssrc_;
+}
+
+void RTCPSender::SetSsrc(uint32_t ssrc) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  ssrc_ = ssrc;
+}
+
+void RTCPSender::SetRemoteSSRC(uint32_t ssrc) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  remote_ssrc_ = ssrc;
+}
+
+int32_t RTCPSender::SetCNAME(absl::string_view c_name) {
+  RTC_DCHECK_LT(c_name.size(), RTCP_CNAME_SIZE);
+  MutexLock lock(&mutex_rtcp_sender_);
+  cname_ = std::string(c_name);
+  return 0;
+}
+
+bool RTCPSender::TimeToSendRTCPReport(bool sendKeyframeBeforeRTP) const {
+  /*
+      For audio we use a configurable interval (default: 5 seconds)
+
+      For video we use a configurable interval (default: 1 second) for a BW
+          smaller than 360 kbit/s, technicaly we break the max 5% RTCP BW for
+          video below 10 kbit/s but that should be extremely rare
+
+
+  From RFC 3550
+
+      MAX RTCP BW is 5% if the session BW
+          A send report is approximately 65 bytes inc CNAME
+          A receiver report is approximately 28 bytes
+
+      The RECOMMENDED value for the reduced minimum in seconds is 360
+        divided by the session bandwidth in kilobits/second.  This minimum
+        is smaller than 5 seconds for bandwidths greater than 72 kb/s.
+
+      If the participant has not yet sent an RTCP packet (the variable
+        initial is true), the constant Tmin is set to half of the configured
+        interval.
+
+      The interval between RTCP packets is varied randomly over the
+        range [0.5,1.5] times the calculated interval to avoid unintended
+        synchronization of all participants
+
+      if we send
+      If the participant is a sender (we_sent true), the constant C is
+        set to the average RTCP packet size (avg_rtcp_size) divided by 25%
+        of the RTCP bandwidth (rtcp_bw), and the constant n is set to the
+        number of senders.
+
+      if we receive only
+        If we_sent is not true, the constant C is set
+        to the average RTCP packet size divided by 75% of the RTCP
+        bandwidth.  The constant n is set to the number of receivers
+        (members - senders).  If the number of senders is greater than
+        25%, senders and receivers are treated together.
+
+      reconsideration NOT required for peer-to-peer
+        "timer reconsideration" is
+        employed.  This algorithm implements a simple back-off mechanism
+        which causes users to hold back RTCP packet transmission if the
+        group sizes are increasing.
+
+        n = number of members
+        C = avg_size/(rtcpBW/4)
+
+     3. The deterministic calculated interval Td is set to max(Tmin, n*C).
+
+     4. The calculated interval T is set to a number uniformly distributed
+        between 0.5 and 1.5 times the deterministic calculated interval.
+
+     5. The resulting value of T is divided by e-3/2=1.21828 to compensate
+        for the fact that the timer reconsideration algorithm converges to
+        a value of the RTCP bandwidth below the intended average
+  */
+
+  Timestamp now = clock_->CurrentTime();
+
+  MutexLock lock(&mutex_rtcp_sender_);
+  RTC_DCHECK(
+      (method_ == RtcpMode::kOff && !next_time_to_send_rtcp_.has_value()) ||
+      (method_ != RtcpMode::kOff && next_time_to_send_rtcp_.has_value()));
+  if (method_ == RtcpMode::kOff)
+    return false;
+
+  if (!audio_ && sendKeyframeBeforeRTP) {
+    // for video key-frames we want to send the RTCP before the large key-frame
+    // if we have a 100 ms margin
+    now += RTCP_SEND_BEFORE_KEY_FRAME;
+  }
+
+  return now >= *next_time_to_send_rtcp_;
+}
+
+void RTCPSender::BuildSR(const RtcpContext& ctx, PacketSender& sender) {
+  // Timestamp shouldn't be estimated before first media frame.
+  RTC_DCHECK(last_frame_capture_time_.has_value());
+  // The timestamp of this RTCP packet should be estimated as the timestamp of
+  // the frame being captured at this moment. We are calculating that
+  // timestamp as the last frame's timestamp + the time since the last frame
+  // was captured.
+  int rtp_rate = rtp_clock_rates_khz_[last_payload_type_];
+  if (rtp_rate <= 0) {
+    rtp_rate =
+        (audio_ ? kBogusRtpRateForAudioRtcp : kVideoPayloadTypeFrequency) /
+        1000;
+  }
+  // Round now_us_ to the closest millisecond, because Ntp time is rounded
+  // when converted to milliseconds,
+  uint32_t rtp_timestamp =
+      timestamp_offset_ + last_rtp_timestamp_ +
+      ((ctx.now_.us() + 500) / 1000 - last_frame_capture_time_->ms()) *
+          rtp_rate;
+
+  rtcp::SenderReport report;
+  report.SetSenderSsrc(ssrc_);
+  report.SetNtp(clock_->ConvertTimestampToNtpTime(ctx.now_));
+  report.SetRtpTimestamp(rtp_timestamp);
+  report.SetPacketCount(ctx.feedback_state_.packets_sent);
+  report.SetOctetCount(ctx.feedback_state_.media_bytes_sent);
+  report.SetReportBlocks(CreateReportBlocks(ctx.feedback_state_));
+  sender.AppendPacket(report);
+}
+
+void RTCPSender::BuildSDES(const RtcpContext& ctx, PacketSender& sender) {
+  size_t length_cname = cname_.length();
+  RTC_CHECK_LT(length_cname, RTCP_CNAME_SIZE);
+
+  rtcp::Sdes sdes;
+  sdes.AddCName(ssrc_, cname_);
+  sender.AppendPacket(sdes);
+}
+
+void RTCPSender::BuildRR(const RtcpContext& ctx, PacketSender& sender) {
+  rtcp::ReceiverReport report;
+  report.SetSenderSsrc(ssrc_);
+  report.SetReportBlocks(CreateReportBlocks(ctx.feedback_state_));
+  if (method_ == RtcpMode::kCompound || !report.report_blocks().empty()) {
+    sender.AppendPacket(report);
+  }
+}
+
+void RTCPSender::BuildPLI(const RtcpContext& ctx, PacketSender& sender) {
+  rtcp::Pli pli;
+  pli.SetSenderSsrc(ssrc_);
+  pli.SetMediaSsrc(remote_ssrc_);
+
+  ++packet_type_counter_.pli_packets;
+  sender.AppendPacket(pli);
+}
+
+void RTCPSender::BuildFIR(const RtcpContext& ctx, PacketSender& sender) {
+  ++sequence_number_fir_;
+
+  rtcp::Fir fir;
+  fir.SetSenderSsrc(ssrc_);
+  fir.AddRequestTo(remote_ssrc_, sequence_number_fir_);
+
+  ++packet_type_counter_.fir_packets;
+  sender.AppendPacket(fir);
+}
+
+void RTCPSender::BuildREMB(const RtcpContext& ctx, PacketSender& sender) {
+  rtcp::Remb remb;
+  remb.SetSenderSsrc(ssrc_);
+  remb.SetBitrateBps(remb_bitrate_);
+  remb.SetSsrcs(remb_ssrcs_);
+  sender.AppendPacket(remb);
+}
+
+void RTCPSender::SetTargetBitrate(unsigned int target_bitrate) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  tmmbr_send_bps_ = target_bitrate;
+}
+
+void RTCPSender::BuildTMMBR(const RtcpContext& ctx, PacketSender& sender) {
+  if (ctx.feedback_state_.receiver == nullptr)
+    return;
+  // Before sending the TMMBR check the received TMMBN, only an owner is
+  // allowed to raise the bitrate:
+  // * If the sender is an owner of the TMMBN -> send TMMBR
+  // * If not an owner but the TMMBR would enter the TMMBN -> send TMMBR
+
+  // get current bounding set from RTCP receiver
+  bool tmmbr_owner = false;
+
+  // holding mutex_rtcp_sender_ while calling RTCPreceiver which
+  // will accuire criticalSectionRTCPReceiver_ is a potental deadlock but
+  // since RTCPreceiver is not doing the reverse we should be fine
+  std::vector<rtcp::TmmbItem> candidates =
+      ctx.feedback_state_.receiver->BoundingSet(&tmmbr_owner);
+
+  if (!candidates.empty()) {
+    for (const auto& candidate : candidates) {
+      if (candidate.bitrate_bps() == tmmbr_send_bps_ &&
+          candidate.packet_overhead() == packet_oh_send_) {
+        // Do not send the same tuple.
+        return;
+      }
+    }
+    if (!tmmbr_owner) {
+      // Use received bounding set as candidate set.
+      // Add current tuple.
+      candidates.emplace_back(ssrc_, tmmbr_send_bps_, packet_oh_send_);
+
+      // Find bounding set.
+      std::vector<rtcp::TmmbItem> bounding =
+          TMMBRHelp::FindBoundingSet(std::move(candidates));
+      tmmbr_owner = TMMBRHelp::IsOwner(bounding, ssrc_);
+      if (!tmmbr_owner) {
+        // Did not enter bounding set, no meaning to send this request.
+        return;
+      }
+    }
+  }
+
+  if (!tmmbr_send_bps_)
+    return;
+
+  rtcp::Tmmbr tmmbr;
+  tmmbr.SetSenderSsrc(ssrc_);
+  rtcp::TmmbItem request;
+  request.set_ssrc(remote_ssrc_);
+  request.set_bitrate_bps(tmmbr_send_bps_);
+  request.set_packet_overhead(packet_oh_send_);
+  tmmbr.AddTmmbr(request);
+  sender.AppendPacket(tmmbr);
+}
+
+void RTCPSender::BuildTMMBN(const RtcpContext& ctx, PacketSender& sender) {
+  rtcp::Tmmbn tmmbn;
+  tmmbn.SetSenderSsrc(ssrc_);
+  for (const rtcp::TmmbItem& tmmbr : tmmbn_to_send_) {
+    if (tmmbr.bitrate_bps() > 0) {
+      tmmbn.AddTmmbr(tmmbr);
+    }
+  }
+  sender.AppendPacket(tmmbn);
+}
+
+void RTCPSender::BuildAPP(const RtcpContext& ctx, PacketSender& sender) {
+  rtcp::App app;
+  app.SetSenderSsrc(ssrc_);
+  sender.AppendPacket(app);
+}
+
+void RTCPSender::BuildLossNotification(const RtcpContext& ctx,
+                                       PacketSender& sender) {
+  loss_notification_.SetSenderSsrc(ssrc_);
+  loss_notification_.SetMediaSsrc(remote_ssrc_);
+  sender.AppendPacket(loss_notification_);
+}
+
+void RTCPSender::BuildNACK(const RtcpContext& ctx, PacketSender& sender) {
+  rtcp::Nack nack;
+  nack.SetSenderSsrc(ssrc_);
+  nack.SetMediaSsrc(remote_ssrc_);
+  nack.SetPacketIds(ctx.nack_list_, ctx.nack_size_);
+
+  // Report stats.
+  for (int idx = 0; idx < ctx.nack_size_; ++idx) {
+    nack_stats_.ReportRequest(ctx.nack_list_[idx]);
+  }
+  packet_type_counter_.nack_requests = nack_stats_.requests();
+  packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
+
+  ++packet_type_counter_.nack_packets;
+  sender.AppendPacket(nack);
+}
+
+void RTCPSender::BuildBYE(const RtcpContext& ctx, PacketSender& sender) {
+  rtcp::Bye bye;
+  bye.SetSenderSsrc(ssrc_);
+  bye.SetCsrcs(csrcs_);
+  sender.AppendPacket(bye);
+}
+
+void RTCPSender::BuildExtendedReports(const RtcpContext& ctx,
+                                      PacketSender& sender) {
+  rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(ssrc_);
+
+  if (!sending_ && xr_send_receiver_reference_time_enabled_) {
+    rtcp::Rrtr rrtr;
+    rrtr.SetNtp(clock_->ConvertTimestampToNtpTime(ctx.now_));
+    xr.SetRrtr(rrtr);
+  }
+
+  for (const rtcp::ReceiveTimeInfo& rti : ctx.feedback_state_.last_xr_rtis) {
+    xr.AddDlrrItem(rti);
+  }
+
+  if (send_video_bitrate_allocation_) {
+    rtcp::TargetBitrate target_bitrate;
+
+    for (int sl = 0; sl < kMaxSpatialLayers; ++sl) {
+      for (int tl = 0; tl < kMaxTemporalStreams; ++tl) {
+        if (video_bitrate_allocation_.HasBitrate(sl, tl)) {
+          target_bitrate.AddTargetBitrate(
+              sl, tl, video_bitrate_allocation_.GetBitrate(sl, tl) / 1000);
+        }
+      }
+    }
+
+    xr.SetTargetBitrate(target_bitrate);
+    send_video_bitrate_allocation_ = false;
+  }
+  sender.AppendPacket(xr);
+}
+
+int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
+                             RTCPPacketType packet_type,
+                             int32_t nack_size,
+                             const uint16_t* nack_list) {
+  int32_t error_code = -1;
+  auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
+    if (transport_->SendRtcp(packet.data(), packet.size())) {
+      error_code = 0;
+      if (event_log_) {
+        event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
+      }
+    }
+  };
+  absl::optional<PacketSender> sender;
+  {
+    MutexLock lock(&mutex_rtcp_sender_);
+    sender.emplace(callback, max_packet_size_);
+    auto result = ComputeCompoundRTCPPacket(feedback_state, packet_type,
+                                            nack_size, nack_list, *sender);
+    if (result) {
+      return *result;
+    }
+  }
+  sender->Send();
+
+  return error_code;
+}
+
+absl::optional<int32_t> RTCPSender::ComputeCompoundRTCPPacket(
+    const FeedbackState& feedback_state,
+    RTCPPacketType packet_type,
+    int32_t nack_size,
+    const uint16_t* nack_list,
+    PacketSender& sender) {
+  if (method_ == RtcpMode::kOff) {
+    RTC_LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
+    return -1;
+  }
+  // Add the flag as volatile. Non volatile entries will not be overwritten.
+  // The new volatile flag will be consumed by the end of this call.
+  SetFlag(packet_type, true);
+
+  // Prevent sending streams to send SR before any media has been sent.
+  const bool can_calculate_rtp_timestamp = last_frame_capture_time_.has_value();
+  if (!can_calculate_rtp_timestamp) {
+    bool consumed_sr_flag = ConsumeFlag(kRtcpSr);
+    bool consumed_report_flag = sending_ && ConsumeFlag(kRtcpReport);
+    bool sender_report = consumed_report_flag || consumed_sr_flag;
+    if (sender_report && AllVolatileFlagsConsumed()) {
+      // This call was for Sender Report and nothing else.
+      return 0;
+    }
+    if (sending_ && method_ == RtcpMode::kCompound) {
+      // Not allowed to send any RTCP packet without sender report.
+      return -1;
+    }
+  }
+
+  // We need to send our NTP even if we haven't received any reports.
+  RtcpContext context(feedback_state, nack_size, nack_list,
+                      clock_->CurrentTime());
+
+  PrepareReport(feedback_state);
+
+  bool create_bye = false;
+
+  auto it = report_flags_.begin();
+  while (it != report_flags_.end()) {
+    uint32_t rtcp_packet_type = it->type;
+
+    if (it->is_volatile) {
+      report_flags_.erase(it++);
+    } else {
+      ++it;
+    }
+
+    // If there is a BYE, don't append now - save it and append it
+    // at the end later.
+    if (rtcp_packet_type == kRtcpBye) {
+      create_bye = true;
+      continue;
+    }
+    auto builder_it = builders_.find(rtcp_packet_type);
+    if (builder_it == builders_.end()) {
+      RTC_DCHECK_NOTREACHED()
+          << "Could not find builder for packet type " << rtcp_packet_type;
+    } else {
+      BuilderFunc func = builder_it->second;
+      (this->*func)(context, sender);
+    }
+  }
+
+  // Append the BYE now at the end
+  if (create_bye) {
+    BuildBYE(context, sender);
+  }
+
+  if (packet_type_counter_observer_ != nullptr) {
+    packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
+        remote_ssrc_, packet_type_counter_);
+  }
+
+  RTC_DCHECK(AllVolatileFlagsConsumed());
+  return absl::nullopt;
+}
+
+void RTCPSender::PrepareReport(const FeedbackState& feedback_state) {
+  bool generate_report;
+  if (IsFlagPresent(kRtcpSr) || IsFlagPresent(kRtcpRr)) {
+    // Report type already explicitly set, don't automatically populate.
+    generate_report = true;
+    RTC_DCHECK(ConsumeFlag(kRtcpReport) == false);
+  } else {
+    generate_report =
+        (ConsumeFlag(kRtcpReport) && method_ == RtcpMode::kReducedSize) ||
+        method_ == RtcpMode::kCompound;
+    if (generate_report)
+      SetFlag(sending_ ? kRtcpSr : kRtcpRr, true);
+  }
+
+  if (IsFlagPresent(kRtcpSr) || (IsFlagPresent(kRtcpRr) && !cname_.empty()))
+    SetFlag(kRtcpSdes, true);
+
+  if (generate_report) {
+    if ((!sending_ && xr_send_receiver_reference_time_enabled_) ||
+        !feedback_state.last_xr_rtis.empty() ||
+        send_video_bitrate_allocation_) {
+      SetFlag(kRtcpAnyExtendedReports, true);
+    }
+
+    // generate next time to send an RTCP report
+    TimeDelta min_interval = report_interval_;
+
+    if (!audio_ && sending_) {
+      // Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
+      int send_bitrate_kbit = feedback_state.send_bitrate / 1000;
+      if (send_bitrate_kbit != 0) {
+        min_interval = std::min(TimeDelta::Millis(360000 / send_bitrate_kbit),
+                                report_interval_);
+      }
+    }
+
+    // The interval between RTCP packets is varied randomly over the
+    // range [1/2,3/2] times the calculated interval.
+    int min_interval_int = rtc::dchecked_cast<int>(min_interval.ms());
+    TimeDelta time_to_next = TimeDelta::Millis(
+        random_.Rand(min_interval_int * 1 / 2, min_interval_int * 3 / 2));
+
+    RTC_DCHECK(!time_to_next.IsZero());
+    SetNextRtcpSendEvaluationDuration(time_to_next);
+
+    // RtcpSender expected to be used for sending either just sender reports
+    // or just receiver reports.
+    RTC_DCHECK(!(IsFlagPresent(kRtcpSr) && IsFlagPresent(kRtcpRr)));
+  }
+}
+
+std::vector<rtcp::ReportBlock> RTCPSender::CreateReportBlocks(
+    const FeedbackState& feedback_state) {
+  std::vector<rtcp::ReportBlock> result;
+  if (!receive_statistics_)
+    return result;
+
+  // TODO(danilchap): Support sending more than `RTCP_MAX_REPORT_BLOCKS` per
+  // compound rtcp packet when single rtcp module is used for multiple media
+  // streams.
+  result = receive_statistics_->RtcpReportBlocks(RTCP_MAX_REPORT_BLOCKS);
+
+  if (!result.empty() && ((feedback_state.last_rr_ntp_secs != 0) ||
+                          (feedback_state.last_rr_ntp_frac != 0))) {
+    // Get our NTP as late as possible to avoid a race.
+    uint32_t now = CompactNtp(clock_->CurrentNtpTime());
+
+    uint32_t receive_time = feedback_state.last_rr_ntp_secs & 0x0000FFFF;
+    receive_time <<= 16;
+    receive_time += (feedback_state.last_rr_ntp_frac & 0xffff0000) >> 16;
+
+    uint32_t delay_since_last_sr = now - receive_time;
+    // TODO(danilchap): Instead of setting same value on all report blocks,
+    // set only when media_ssrc match sender ssrc of the sender report
+    // remote times were taken from.
+    for (auto& report_block : result) {
+      report_block.SetLastSr(feedback_state.remote_sr);
+      report_block.SetDelayLastSr(delay_since_last_sr);
+    }
+  }
+  return result;
+}
+
+void RTCPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
+  RTC_DCHECK_LE(csrcs.size(), kRtpCsrcSize);
+  MutexLock lock(&mutex_rtcp_sender_);
+  csrcs_ = csrcs;
+}
+
+void RTCPSender::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  tmmbn_to_send_ = std::move(bounding_set);
+  SetFlag(kRtcpTmmbn, true);
+}
+
+void RTCPSender::SetFlag(uint32_t type, bool is_volatile) {
+  if (type & kRtcpAnyExtendedReports) {
+    report_flags_.insert(ReportFlag(kRtcpAnyExtendedReports, is_volatile));
+  } else {
+    report_flags_.insert(ReportFlag(type, is_volatile));
+  }
+}
+
+bool RTCPSender::IsFlagPresent(uint32_t type) const {
+  return report_flags_.find(ReportFlag(type, false)) != report_flags_.end();
+}
+
+bool RTCPSender::ConsumeFlag(uint32_t type, bool forced) {
+  auto it = report_flags_.find(ReportFlag(type, false));
+  if (it == report_flags_.end())
+    return false;
+  if (it->is_volatile || forced)
+    report_flags_.erase((it));
+  return true;
+}
+
+bool RTCPSender::AllVolatileFlagsConsumed() const {
+  for (const ReportFlag& flag : report_flags_) {
+    if (flag.is_volatile)
+      return false;
+  }
+  return true;
+}
+
+void RTCPSender::SetVideoBitrateAllocation(
+    const VideoBitrateAllocation& bitrate) {
+  MutexLock lock(&mutex_rtcp_sender_);
+  if (method_ == RtcpMode::kOff) {
+    RTC_LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
+    return;
+  }
+  // Check if this allocation is first ever, or has a different set of
+  // spatial/temporal layers signaled and enabled, if so trigger an rtcp report
+  // as soon as possible.
+  absl::optional<VideoBitrateAllocation> new_bitrate =
+      CheckAndUpdateLayerStructure(bitrate);
+  if (new_bitrate) {
+    video_bitrate_allocation_ = *new_bitrate;
+    RTC_LOG(LS_INFO) << "Emitting TargetBitrate XR for SSRC " << ssrc_
+                     << " with new layers enabled/disabled: "
+                     << video_bitrate_allocation_.ToString();
+    SetNextRtcpSendEvaluationDuration(TimeDelta::Zero());
+  } else {
+    video_bitrate_allocation_ = bitrate;
+  }
+
+  send_video_bitrate_allocation_ = true;
+  SetFlag(kRtcpAnyExtendedReports, true);
+}
+
+absl::optional<VideoBitrateAllocation> RTCPSender::CheckAndUpdateLayerStructure(
+    const VideoBitrateAllocation& bitrate) const {
+  absl::optional<VideoBitrateAllocation> updated_bitrate;
+  for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
+    for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
+      if (!updated_bitrate &&
+          (bitrate.HasBitrate(si, ti) !=
+               video_bitrate_allocation_.HasBitrate(si, ti) ||
+           (bitrate.GetBitrate(si, ti) == 0) !=
+               (video_bitrate_allocation_.GetBitrate(si, ti) == 0))) {
+        updated_bitrate = bitrate;
+      }
+      if (video_bitrate_allocation_.GetBitrate(si, ti) > 0 &&
+          bitrate.GetBitrate(si, ti) == 0) {
+        // Make sure this stream disabling is explicitly signaled.
+        updated_bitrate->SetBitrate(si, ti, 0);
+      }
+    }
+  }
+
+  return updated_bitrate;
+}
+
+void RTCPSender::SendCombinedRtcpPacket(
+    std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
+  size_t max_packet_size;
+  uint32_t ssrc;
+  {
+    MutexLock lock(&mutex_rtcp_sender_);
+    if (method_ == RtcpMode::kOff) {
+      RTC_LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
+      return;
+    }
+
+    max_packet_size = max_packet_size_;
+    ssrc = ssrc_;
+  }
+  RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
+  auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
+    if (transport_->SendRtcp(packet.data(), packet.size())) {
+      if (event_log_)
+        event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
+    }
+  };
+  PacketSender sender(callback, max_packet_size);
+  for (auto& rtcp_packet : rtcp_packets) {
+    rtcp_packet->SetSenderSsrc(ssrc);
+    sender.AppendPacket(*rtcp_packet);
+  }
+  sender.Send();
+}
+
+void RTCPSender::SetNextRtcpSendEvaluationDuration(TimeDelta duration) {
+  next_time_to_send_rtcp_ = clock_->CurrentTime() + duration;
+  // TODO(bugs.webrtc.org/11581): make unconditional once downstream consumers
+  // are using the callback method.
+  if (schedule_next_rtcp_send_evaluation_function_)
+    schedule_next_rtcp_send_evaluation_function_(duration);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender.h
new file mode 100644
index 0000000000..8f51e7947d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -0,0 +1,332 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
+
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/call/transport.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_nack_stats.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/random.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class RTCPReceiver;
+class RtcEventLog;
+
+class RTCPSender final {
+ public:
+  struct Configuration {
+    // TODO(bugs.webrtc.org/11581): Remove this temporary conversion utility
+    // once rtc_rtcp_impl.cc/h are gone.
+    static Configuration FromRtpRtcpConfiguration(
+        const RtpRtcpInterface::Configuration& config);
+
+    // True for a audio version of the RTP/RTCP module object false will create
+    // a video version.
+    bool audio = false;
+    // SSRCs for media and retransmission, respectively.
+    // FlexFec SSRC is fetched from `flexfec_sender`.
+    uint32_t local_media_ssrc = 0;
+    // The clock to use to read time. If nullptr then system clock will be used.
+    Clock* clock = nullptr;
+    // Transport object that will be called when packets are ready to be sent
+    // out on the network.
+    Transport* outgoing_transport = nullptr;
+    // Estimate RTT as non-sender as described in
+    // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
+    bool non_sender_rtt_measurement = false;
+    // Optional callback which, if specified, is used by RTCPSender to schedule
+    // the next time to evaluate if RTCP should be sent by means of
+    // TimeToSendRTCPReport/SendRTCP.
+    // The RTCPSender client still needs to call TimeToSendRTCPReport/SendRTCP
+    // to actually get RTCP sent.
+    //
+    // Note: It's recommended to use the callback to ensure program design that
+    // doesn't use polling.
+    // TODO(bugs.webrtc.org/11581): Make mandatory once downstream consumers
+    // have migrated to the callback solution.
+    std::function<void(TimeDelta)> schedule_next_rtcp_send_evaluation_function;
+
+    RtcEventLog* event_log = nullptr;
+    absl::optional<TimeDelta> rtcp_report_interval;
+    ReceiveStatisticsProvider* receive_statistics = nullptr;
+    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr;
+  };
+  struct FeedbackState {
+    FeedbackState();
+    FeedbackState(const FeedbackState&);
+    FeedbackState(FeedbackState&&);
+
+    ~FeedbackState();
+
+    uint32_t packets_sent;
+    size_t media_bytes_sent;
+    uint32_t send_bitrate;
+
+    uint32_t last_rr_ntp_secs;
+    uint32_t last_rr_ntp_frac;
+    uint32_t remote_sr;
+
+    std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis;
+
+    // Used when generating TMMBR.
+    RTCPReceiver* receiver;
+  };
+
+  explicit RTCPSender(Configuration config);
+
+  RTCPSender() = delete;
+  RTCPSender(const RTCPSender&) = delete;
+  RTCPSender& operator=(const RTCPSender&) = delete;
+
+  virtual ~RTCPSender();
+
+  RtcpMode Status() const RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+  void SetRTCPStatus(RtcpMode method) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  bool Sending() const RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+  void SetSendingStatus(const FeedbackState& feedback_state,
+                        bool enabled)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);  // combine the functions
+
+  void SetNonSenderRttMeasurement(bool enabled)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  void SetTimestampOffset(uint32_t timestamp_offset)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  void SetLastRtpTime(uint32_t rtp_timestamp,
+                      absl::optional<Timestamp> capture_time,
+                      absl::optional<int8_t> payload_type)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  void SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  uint32_t SSRC() const;
+  void SetSsrc(uint32_t ssrc);
+
+  void SetRemoteSSRC(uint32_t ssrc) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  int32_t SetCNAME(absl::string_view cName)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  int32_t SendRTCP(const FeedbackState& feedback_state,
+                   RTCPPacketType packetType,
+                   int32_t nackSize = 0,
+                   const uint16_t* nackList = 0)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  int32_t SendLossNotification(const FeedbackState& feedback_state,
+                               uint16_t last_decoded_seq_num,
+                               uint16_t last_received_seq_num,
+                               bool decodability_flag,
+                               bool buffering_allowed)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  void UnsetRemb() RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  bool TMMBR() const RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  void SetMaxRtpPacketSize(size_t max_packet_size)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  void SetCsrcs(const std::vector<uint32_t>& csrcs)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+  void SetTargetBitrate(unsigned int target_bitrate)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+  void SetVideoBitrateAllocation(const VideoBitrateAllocation& bitrate)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+  void SendCombinedRtcpPacket(
+      std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets)
+      RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
+
+ private:
+  class RtcpContext;
+  class PacketSender;
+
+  absl::optional<int32_t> ComputeCompoundRTCPPacket(
+      const FeedbackState& feedback_state,
+      RTCPPacketType packet_type,
+      int32_t nack_size,
+      const uint16_t* nack_list,
+      PacketSender& sender) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+
+  // Determine which RTCP messages should be sent and setup flags.
+  void PrepareReport(const FeedbackState& feedback_state)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+
+  std::vector<rtcp::ReportBlock> CreateReportBlocks(
+      const FeedbackState& feedback_state)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+
+  void BuildSR(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildRR(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildSDES(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildPLI(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildREMB(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildTMMBR(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildTMMBN(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildAPP(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildLossNotification(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildExtendedReports(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildBYE(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildFIR(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  void BuildNACK(const RtcpContext& context, PacketSender& sender)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+
+  // `duration` being TimeDelta::Zero() means schedule immediately.
+  void SetNextRtcpSendEvaluationDuration(TimeDelta duration)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+
+  const bool audio_;
+  // TODO(bugs.webrtc.org/11581): `mutex_rtcp_sender_` shouldn't be required if
+  // we consistently run network related operations on the network thread.
+  // This is currently not possible due to callbacks from the process thread in
+  // ModuleRtpRtcpImpl2.
+  uint32_t ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  Clock* const clock_;
+  Random random_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  RtcpMode method_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  RtcEventLog* const event_log_;
+  Transport* const transport_;
+
+  const TimeDelta report_interval_;
+  // Set from
+  // RTCPSender::Configuration::schedule_next_rtcp_send_evaluation_function.
+  const std::function<void(TimeDelta)>
+      schedule_next_rtcp_send_evaluation_function_;
+
+  mutable Mutex mutex_rtcp_sender_;
+  bool sending_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  absl::optional<Timestamp> next_time_to_send_rtcp_
+      RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  uint32_t timestamp_offset_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  absl::optional<Timestamp> last_frame_capture_time_
+      RTC_GUARDED_BY(mutex_rtcp_sender_);
+  // SSRC that we receive on our RTP channel
+  uint32_t remote_ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  std::string cname_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  ReceiveStatisticsProvider* receive_statistics_
+      RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  // send CSRCs
+  std::vector<uint32_t> csrcs_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  // Full intra request
+  uint8_t sequence_number_fir_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  rtcp::LossNotification loss_notification_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  // REMB
+  int64_t remb_bitrate_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  std::vector<uint32_t> remb_ssrcs_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  std::vector<rtcp::TmmbItem> tmmbn_to_send_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  uint32_t tmmbr_send_bps_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  uint32_t packet_oh_send_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  size_t max_packet_size_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  // True if sending of XR Receiver reference time report is enabled.
+  bool xr_send_receiver_reference_time_enabled_
+      RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
+  RtcpPacketTypeCounter packet_type_counter_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  RtcpNackStats nack_stats_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  VideoBitrateAllocation video_bitrate_allocation_
+      RTC_GUARDED_BY(mutex_rtcp_sender_);
+  bool send_video_bitrate_allocation_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  std::map<int8_t, int> rtp_clock_rates_khz_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+  int8_t last_payload_type_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  absl::optional<VideoBitrateAllocation> CheckAndUpdateLayerStructure(
+      const VideoBitrateAllocation& bitrate) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+
+  void SetFlag(uint32_t type, bool is_volatile)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  bool IsFlagPresent(uint32_t type) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  bool ConsumeFlag(uint32_t type, bool forced = false)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  bool AllVolatileFlagsConsumed() const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
+  struct ReportFlag {
+    ReportFlag(uint32_t type, bool is_volatile)
+        : type(type), is_volatile(is_volatile) {}
+    bool operator<(const ReportFlag& flag) const { return type < flag.type; }
+    bool operator==(const ReportFlag& flag) const { return type == flag.type; }
+    const uint32_t type;
+    const bool is_volatile;
+  };
+
+  std::set<ReportFlag> report_flags_ RTC_GUARDED_BY(mutex_rtcp_sender_);
+
+  typedef void (RTCPSender::*BuilderFunc)(const RtcpContext&, PacketSender&);
+  // Map from RTCPPacketType to builder.
+  std::map<uint32_t, BuilderFunc> builders_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
new file mode 100644
index 0000000000..ae59dc5b0c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -0,0 +1,844 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_sender.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/base/macros.h"
+#include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/rate_limiter.h"
+#include "rtc_base/thread.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::Eq;
+using ::testing::Invoke;
+using ::testing::Property;
+using ::testing::SizeIs;
+
+namespace webrtc {
+
+class RtcpPacketTypeCounterObserverImpl : public RtcpPacketTypeCounterObserver {
+ public:
+  RtcpPacketTypeCounterObserverImpl() : ssrc_(0) {}
+  ~RtcpPacketTypeCounterObserverImpl() override = default;
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) override {
+    ssrc_ = ssrc;
+    counter_ = packet_counter;
+  }
+  uint32_t ssrc_;
+  RtcpPacketTypeCounter counter_;
+};
+
+class TestTransport : public Transport {
+ public:
+  TestTransport() {}
+
+  bool SendRtp(const uint8_t* /*data*/,
+               size_t /*len*/,
+               const PacketOptions& options) override {
+    return false;
+  }
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    parser_.Parse(data, len);
+    return true;
+  }
+  test::RtcpPacketParser parser_;
+};
+
+namespace {
+static const uint32_t kSenderSsrc = 0x11111111;
+static const uint32_t kRemoteSsrc = 0x22222222;
+static const uint32_t kStartRtpTimestamp = 0x34567;
+static const uint32_t kRtpTimestamp = 0x45678;
+
+std::unique_ptr<RTCPSender> CreateRtcpSender(
+    const RTCPSender::Configuration& config,
+    bool init_timestamps = true) {
+  auto rtcp_sender = std::make_unique<RTCPSender>(config);
+  rtcp_sender->SetRemoteSSRC(kRemoteSsrc);
+  if (init_timestamps) {
+    rtcp_sender->SetTimestampOffset(kStartRtpTimestamp);
+    rtcp_sender->SetLastRtpTime(kRtpTimestamp, config.clock->CurrentTime(),
+                                /*payload_type=*/0);
+  }
+  return rtcp_sender;
+}
+}  // namespace
+
+class RtcpSenderTest : public ::testing::Test {
+ protected:
+  RtcpSenderTest()
+      : clock_(1335900000),
+        receive_statistics_(ReceiveStatistics::Create(&clock_)) {
+    rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl2(GetDefaultRtpRtcpConfig()));
+  }
+
+  RTCPSender::Configuration GetDefaultConfig() {
+    RTCPSender::Configuration configuration;
+    configuration.audio = false;
+    configuration.clock = &clock_;
+    configuration.outgoing_transport = &test_transport_;
+    configuration.rtcp_report_interval = TimeDelta::Millis(1000);
+    configuration.receive_statistics = receive_statistics_.get();
+    configuration.local_media_ssrc = kSenderSsrc;
+    return configuration;
+  }
+
+  RtpRtcpInterface::Configuration GetDefaultRtpRtcpConfig() {
+    RTCPSender::Configuration config = GetDefaultConfig();
+    RtpRtcpInterface::Configuration result;
+    result.audio = config.audio;
+    result.clock = config.clock;
+    result.outgoing_transport = config.outgoing_transport;
+    result.rtcp_report_interval_ms = config.rtcp_report_interval->ms();
+    result.receive_statistics = config.receive_statistics;
+    result.local_media_ssrc = config.local_media_ssrc;
+    return result;
+  }
+
+  void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
+    RtpPacketReceived packet;
+    packet.SetSsrc(remote_ssrc);
+    packet.SetSequenceNumber(seq_num);
+    packet.SetTimestamp(12345);
+    packet.SetPayloadSize(100 - 12);
+    receive_statistics_->OnRtpPacket(packet);
+  }
+
+  test::RtcpPacketParser* parser() { return &test_transport_.parser_; }
+
+  RTCPSender::FeedbackState feedback_state() {
+    return rtp_rtcp_impl_->GetFeedbackState();
+  }
+
+  rtc::AutoThread main_thread_;
+  SimulatedClock clock_;
+  TestTransport test_transport_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_impl_;
+};
+
+TEST_F(RtcpSenderTest, SetRtcpStatus) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  EXPECT_EQ(RtcpMode::kOff, rtcp_sender->Status());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(RtcpMode::kReducedSize, rtcp_sender->Status());
+}
+
+TEST_F(RtcpSenderTest, SetSendingStatus) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  EXPECT_FALSE(rtcp_sender->Sending());
+  rtcp_sender->SetSendingStatus(feedback_state(), true);
+  EXPECT_TRUE(rtcp_sender->Sending());
+}
+
+TEST_F(RtcpSenderTest, NoPacketSentIfOff) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kOff);
+  EXPECT_EQ(-1, rtcp_sender->SendRTCP(feedback_state(), kRtcpSr));
+}
+
+TEST_F(RtcpSenderTest, SendSr) {
+  const uint32_t kPacketCount = 0x12345;
+  const uint32_t kOctetCount = 0x23456;
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
+  rtcp_sender->SetSendingStatus(feedback_state, true);
+  feedback_state.packets_sent = kPacketCount;
+  feedback_state.media_bytes_sent = kOctetCount;
+  NtpTime ntp = clock_.CurrentNtpTime();
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state, kRtcpSr));
+  EXPECT_EQ(1, parser()->sender_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->sender_report()->sender_ssrc());
+  EXPECT_EQ(ntp, parser()->sender_report()->ntp());
+  EXPECT_EQ(kPacketCount, parser()->sender_report()->sender_packet_count());
+  EXPECT_EQ(kOctetCount, parser()->sender_report()->sender_octet_count());
+  EXPECT_EQ(kStartRtpTimestamp + kRtpTimestamp,
+            parser()->sender_report()->rtp_timestamp());
+  EXPECT_EQ(0U, parser()->sender_report()->report_blocks().size());
+}
+
+TEST_F(RtcpSenderTest, SendConsecutiveSrWithExactSlope) {
+  const uint32_t kPacketCount = 0x12345;
+  const uint32_t kOctetCount = 0x23456;
+  const int kTimeBetweenSRsUs = 10043;  // Not exact value in milliseconds.
+  const int kExtraPackets = 30;
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  // Make sure clock is not exactly at some milliseconds point.
+  clock_.AdvanceTimeMicroseconds(kTimeBetweenSRsUs);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
+  rtcp_sender->SetSendingStatus(feedback_state, true);
+  feedback_state.packets_sent = kPacketCount;
+  feedback_state.media_bytes_sent = kOctetCount;
+
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state, kRtcpSr));
+  EXPECT_EQ(1, parser()->sender_report()->num_packets());
+  NtpTime ntp1 = parser()->sender_report()->ntp();
+  uint32_t rtp1 = parser()->sender_report()->rtp_timestamp();
+
+  // Send more SRs to ensure slope is always exact for different offsets
+  for (int packets = 1; packets <= kExtraPackets; ++packets) {
+    clock_.AdvanceTimeMicroseconds(kTimeBetweenSRsUs);
+    EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state, kRtcpSr));
+    EXPECT_EQ(packets + 1, parser()->sender_report()->num_packets());
+
+    NtpTime ntp2 = parser()->sender_report()->ntp();
+    uint32_t rtp2 = parser()->sender_report()->rtp_timestamp();
+
+    uint32_t ntp_diff_in_rtp_units =
+        (ntp2.ToMs() - ntp1.ToMs()) * (kVideoPayloadTypeFrequency / 1000);
+    EXPECT_EQ(rtp2 - rtp1, ntp_diff_in_rtp_units);
+  }
+}
+
+TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
+  RTCPSender::Configuration config;
+  config.clock = &clock_;
+  config.receive_statistics = receive_statistics_.get();
+  config.outgoing_transport = &test_transport_;
+  config.rtcp_report_interval = TimeDelta::Millis(1000);
+  config.local_media_ssrc = kSenderSsrc;
+  auto rtcp_sender = CreateRtcpSender(config, /*init_timestamps=*/false);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  rtcp_sender->SetSendingStatus(feedback_state(), true);
+
+  // Sender Report shouldn't be send as an SR nor as a Report.
+  rtcp_sender->SendRTCP(feedback_state(), kRtcpSr);
+  EXPECT_EQ(0, parser()->sender_report()->num_packets());
+  rtcp_sender->SendRTCP(feedback_state(), kRtcpReport);
+  EXPECT_EQ(0, parser()->sender_report()->num_packets());
+  // Other packets (e.g. Pli) are allowed, even if useless.
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpPli));
+  EXPECT_EQ(1, parser()->pli()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, DoNotSendCompundBeforeRtp) {
+  RTCPSender::Configuration config;
+  config.clock = &clock_;
+  config.receive_statistics = receive_statistics_.get();
+  config.outgoing_transport = &test_transport_;
+  config.rtcp_report_interval = TimeDelta::Millis(1000);
+  config.local_media_ssrc = kSenderSsrc;
+  auto rtcp_sender = CreateRtcpSender(config, /*init_timestamps=*/false);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetSendingStatus(feedback_state(), true);
+
+  // In compound mode no packets are allowed (e.g. Pli) because compound mode
+  // should start with Sender Report.
+  EXPECT_EQ(-1, rtcp_sender->SendRTCP(feedback_state(), kRtcpPli));
+  EXPECT_EQ(0, parser()->pli()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, SendRr) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpRr));
+  EXPECT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->receiver_report()->sender_ssrc());
+  EXPECT_EQ(0U, parser()->receiver_report()->report_blocks().size());
+}
+
+TEST_F(RtcpSenderTest, DoesntSendEmptyRrInReducedSizeMode) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  rtcp_sender->SendRTCP(feedback_state(), kRtcpRr);
+  EXPECT_EQ(parser()->receiver_report()->num_packets(), 0);
+}
+
+TEST_F(RtcpSenderTest, SendRrWithOneReportBlock) {
+  const uint16_t kSeqNum = 11111;
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  InsertIncomingPacket(kRemoteSsrc, kSeqNum);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpRr));
+  EXPECT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->receiver_report()->sender_ssrc());
+  ASSERT_EQ(1U, parser()->receiver_report()->report_blocks().size());
+  const rtcp::ReportBlock& rb = parser()->receiver_report()->report_blocks()[0];
+  EXPECT_EQ(kRemoteSsrc, rb.source_ssrc());
+  EXPECT_EQ(0U, rb.fraction_lost());
+  EXPECT_EQ(0, rb.cumulative_lost_signed());
+  EXPECT_EQ(kSeqNum, rb.extended_high_seq_num());
+}
+
+TEST_F(RtcpSenderTest, SendRrWithTwoReportBlocks) {
+  const uint16_t kSeqNum = 11111;
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  InsertIncomingPacket(kRemoteSsrc, kSeqNum);
+  InsertIncomingPacket(kRemoteSsrc + 1, kSeqNum + 1);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpRr));
+  EXPECT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->receiver_report()->sender_ssrc());
+  EXPECT_THAT(
+      parser()->receiver_report()->report_blocks(),
+      UnorderedElementsAre(
+          Property(&rtcp::ReportBlock::source_ssrc, Eq(kRemoteSsrc)),
+          Property(&rtcp::ReportBlock::source_ssrc, Eq(kRemoteSsrc + 1))));
+}
+
+TEST_F(RtcpSenderTest, SendSdes) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender->SetCNAME("alice@host"));
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpSdes));
+  EXPECT_EQ(1, parser()->sdes()->num_packets());
+  EXPECT_EQ(1U, parser()->sdes()->chunks().size());
+  EXPECT_EQ(kSenderSsrc, parser()->sdes()->chunks()[0].ssrc);
+  EXPECT_EQ("alice@host", parser()->sdes()->chunks()[0].cname);
+}
+
+TEST_F(RtcpSenderTest, SdesIncludedInCompoundPacket) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender->SetCNAME("alice@host"));
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(1, parser()->sdes()->num_packets());
+  EXPECT_EQ(1U, parser()->sdes()->chunks().size());
+}
+
+TEST_F(RtcpSenderTest, SendBye) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpBye));
+  EXPECT_EQ(1, parser()->bye()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->bye()->sender_ssrc());
+}
+
+TEST_F(RtcpSenderTest, StopSendingTriggersBye) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  rtcp_sender->SetSendingStatus(feedback_state(), true);
+  rtcp_sender->SetSendingStatus(feedback_state(), false);
+  EXPECT_EQ(1, parser()->bye()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->bye()->sender_ssrc());
+}
+
+TEST_F(RtcpSenderTest, SendFir) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpFir));
+  EXPECT_EQ(1, parser()->fir()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->fir()->sender_ssrc());
+  EXPECT_EQ(1U, parser()->fir()->requests().size());
+  EXPECT_EQ(kRemoteSsrc, parser()->fir()->requests()[0].ssrc);
+  uint8_t seq = parser()->fir()->requests()[0].seq_nr;
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpFir));
+  EXPECT_EQ(2, parser()->fir()->num_packets());
+  EXPECT_EQ(seq + 1, parser()->fir()->requests()[0].seq_nr);
+}
+
+TEST_F(RtcpSenderTest, SendPli) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpPli));
+  EXPECT_EQ(1, parser()->pli()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->pli()->sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parser()->pli()->media_ssrc());
+}
+
+TEST_F(RtcpSenderTest, SendNack) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  const uint16_t kList[] = {0, 1, 16};
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpNack,
+                                     ABSL_ARRAYSIZE(kList), kList));
+  EXPECT_EQ(1, parser()->nack()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->nack()->sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parser()->nack()->media_ssrc());
+  EXPECT_THAT(parser()->nack()->packet_ids(), ElementsAre(0, 1, 16));
+}
+
+TEST_F(RtcpSenderTest, SendLossNotificationBufferingNotAllowed) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  constexpr uint16_t kLastDecoded = 0x1234;
+  constexpr uint16_t kLastReceived = 0x4321;
+  constexpr bool kDecodabilityFlag = true;
+  constexpr bool kBufferingAllowed = false;
+  EXPECT_EQ(rtcp_sender->SendLossNotification(feedback_state(), kLastDecoded,
+                                              kLastReceived, kDecodabilityFlag,
+                                              kBufferingAllowed),
+            0);
+  EXPECT_EQ(parser()->processed_rtcp_packets(), 1u);
+  EXPECT_EQ(parser()->loss_notification()->num_packets(), 1);
+  EXPECT_EQ(kSenderSsrc, parser()->loss_notification()->sender_ssrc());
+  EXPECT_EQ(kRemoteSsrc, parser()->loss_notification()->media_ssrc());
+}
+
+TEST_F(RtcpSenderTest, SendLossNotificationBufferingAllowed) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  constexpr uint16_t kLastDecoded = 0x1234;
+  constexpr uint16_t kLastReceived = 0x4321;
+  constexpr bool kDecodabilityFlag = true;
+  constexpr bool kBufferingAllowed = true;
+  EXPECT_EQ(rtcp_sender->SendLossNotification(feedback_state(), kLastDecoded,
+                                              kLastReceived, kDecodabilityFlag,
+                                              kBufferingAllowed),
+            0);
+
+  // No RTCP messages sent yet.
+  ASSERT_EQ(parser()->processed_rtcp_packets(), 0u);
+
+  // Sending another messages triggers sending the LNTF messages as well.
+  const uint16_t kList[] = {0, 1, 16};
+  EXPECT_EQ(rtcp_sender->SendRTCP(feedback_state(), kRtcpNack,
+                                  ABSL_ARRAYSIZE(kList), kList),
+            0);
+
+  // Exactly one packet was produced, and it contained both the buffered LNTF
+  // as well as the message that had triggered the packet.
+  EXPECT_EQ(parser()->processed_rtcp_packets(), 1u);
+  EXPECT_EQ(parser()->loss_notification()->num_packets(), 1);
+  EXPECT_EQ(parser()->loss_notification()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(parser()->loss_notification()->media_ssrc(), kRemoteSsrc);
+  EXPECT_EQ(parser()->nack()->num_packets(), 1);
+  EXPECT_EQ(parser()->nack()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(parser()->nack()->media_ssrc(), kRemoteSsrc);
+}
+
+TEST_F(RtcpSenderTest, RembNotIncludedBeforeSet) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+
+  rtcp_sender->SendRTCP(feedback_state(), kRtcpRr);
+
+  ASSERT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(0, parser()->remb()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, RembNotIncludedAfterUnset) {
+  const int64_t kBitrate = 261011;
+  const std::vector<uint32_t> kSsrcs = {kRemoteSsrc, kRemoteSsrc + 1};
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetRemb(kBitrate, kSsrcs);
+  rtcp_sender->SendRTCP(feedback_state(), kRtcpRr);
+  ASSERT_EQ(1, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(1, parser()->remb()->num_packets());
+
+  // Turn off REMB. rtcp_sender no longer should send it.
+  rtcp_sender->UnsetRemb();
+  rtcp_sender->SendRTCP(feedback_state(), kRtcpRr);
+  ASSERT_EQ(2, parser()->receiver_report()->num_packets());
+  EXPECT_EQ(1, parser()->remb()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, SendRemb) {
+  const int64_t kBitrate = 261011;
+  const std::vector<uint32_t> kSsrcs = {kRemoteSsrc, kRemoteSsrc + 1};
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  rtcp_sender->SetRemb(kBitrate, kSsrcs);
+
+  rtcp_sender->SendRTCP(feedback_state(), kRtcpRemb);
+
+  EXPECT_EQ(1, parser()->remb()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->remb()->sender_ssrc());
+  EXPECT_EQ(kBitrate, parser()->remb()->bitrate_bps());
+  EXPECT_THAT(parser()->remb()->ssrcs(),
+              ElementsAre(kRemoteSsrc, kRemoteSsrc + 1));
+}
+
+TEST_F(RtcpSenderTest, RembIncludedInEachCompoundPacketAfterSet) {
+  const int kBitrate = 261011;
+  const std::vector<uint32_t> kSsrcs = {kRemoteSsrc, kRemoteSsrc + 1};
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetRemb(kBitrate, kSsrcs);
+
+  rtcp_sender->SendRTCP(feedback_state(), kRtcpReport);
+  EXPECT_EQ(1, parser()->remb()->num_packets());
+  // REMB should be included in each compound packet.
+  rtcp_sender->SendRTCP(feedback_state(), kRtcpReport);
+  EXPECT_EQ(2, parser()->remb()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, SendXrWithDlrr) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
+  rtcp::ReceiveTimeInfo last_xr_rr;
+  last_xr_rr.ssrc = 0x11111111;
+  last_xr_rr.last_rr = 0x22222222;
+  last_xr_rr.delay_since_last_rr = 0x33333333;
+  feedback_state.last_xr_rtis.push_back(last_xr_rr);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state, kRtcpReport));
+  EXPECT_EQ(1, parser()->xr()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
+  ASSERT_THAT(parser()->xr()->dlrr().sub_blocks(), SizeIs(1));
+  EXPECT_EQ(last_xr_rr.ssrc, parser()->xr()->dlrr().sub_blocks()[0].ssrc);
+  EXPECT_EQ(last_xr_rr.last_rr, parser()->xr()->dlrr().sub_blocks()[0].last_rr);
+  EXPECT_EQ(last_xr_rr.delay_since_last_rr,
+            parser()->xr()->dlrr().sub_blocks()[0].delay_since_last_rr);
+}
+
+TEST_F(RtcpSenderTest, SendXrWithMultipleDlrrSubBlocks) {
+  const size_t kNumReceivers = 2;
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
+  for (size_t i = 0; i < kNumReceivers; ++i) {
+    rtcp::ReceiveTimeInfo last_xr_rr;
+    last_xr_rr.ssrc = i;
+    last_xr_rr.last_rr = (i + 1) * 100;
+    last_xr_rr.delay_since_last_rr = (i + 2) * 200;
+    feedback_state.last_xr_rtis.push_back(last_xr_rr);
+  }
+
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state, kRtcpReport));
+  EXPECT_EQ(1, parser()->xr()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
+  ASSERT_THAT(parser()->xr()->dlrr().sub_blocks(), SizeIs(kNumReceivers));
+  for (size_t i = 0; i < kNumReceivers; ++i) {
+    EXPECT_EQ(feedback_state.last_xr_rtis[i].ssrc,
+              parser()->xr()->dlrr().sub_blocks()[i].ssrc);
+    EXPECT_EQ(feedback_state.last_xr_rtis[i].last_rr,
+              parser()->xr()->dlrr().sub_blocks()[i].last_rr);
+    EXPECT_EQ(feedback_state.last_xr_rtis[i].delay_since_last_rr,
+              parser()->xr()->dlrr().sub_blocks()[i].delay_since_last_rr);
+  }
+}
+
+TEST_F(RtcpSenderTest, SendXrWithRrtr) {
+  RTCPSender::Configuration config = GetDefaultConfig();
+  config.non_sender_rtt_measurement = true;
+  auto rtcp_sender = CreateRtcpSender(config);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetSendingStatus(feedback_state(), false);
+  NtpTime ntp = clock_.CurrentNtpTime();
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(1, parser()->xr()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
+  EXPECT_FALSE(parser()->xr()->dlrr());
+  ASSERT_TRUE(parser()->xr()->rrtr());
+  EXPECT_EQ(ntp, parser()->xr()->rrtr()->ntp());
+}
+
+// Same test as above, but enable Rrtr with the setter.
+TEST_F(RtcpSenderTest, SendXrWithRrtrUsingSetter) {
+  RTCPSender::Configuration config = GetDefaultConfig();
+  config.non_sender_rtt_measurement = false;
+  auto rtcp_sender = CreateRtcpSender(config);
+  rtcp_sender->SetNonSenderRttMeasurement(true);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetSendingStatus(feedback_state(), false);
+  NtpTime ntp = clock_.CurrentNtpTime();
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(1, parser()->xr()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
+  EXPECT_FALSE(parser()->xr()->dlrr());
+  ASSERT_TRUE(parser()->xr()->rrtr());
+  EXPECT_EQ(ntp, parser()->xr()->rrtr()->ntp());
+}
+
+// Same test as above, but disable Rrtr with the setter.
+TEST_F(RtcpSenderTest, SendsNoRrtrUsingSetter) {
+  RTCPSender::Configuration config = GetDefaultConfig();
+  config.non_sender_rtt_measurement = true;
+  auto rtcp_sender = CreateRtcpSender(config);
+  rtcp_sender->SetNonSenderRttMeasurement(false);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetSendingStatus(feedback_state(), false);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(0, parser()->xr()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfSending) {
+  RTCPSender::Configuration config = GetDefaultConfig();
+  config.non_sender_rtt_measurement = true;
+  auto rtcp_sender = CreateRtcpSender(config);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetSendingStatus(feedback_state(), true);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(0, parser()->xr()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfNotEnabled) {
+  RTCPSender::Configuration config = GetDefaultConfig();
+  config.non_sender_rtt_measurement = false;
+  auto rtcp_sender = CreateRtcpSender(config);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetSendingStatus(feedback_state(), false);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(0, parser()->xr()->num_packets());
+}
+
+TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
+  RtcpPacketTypeCounterObserverImpl observer;
+  RTCPSender::Configuration config;
+  config.clock = &clock_;
+  config.receive_statistics = receive_statistics_.get();
+  config.outgoing_transport = &test_transport_;
+  config.rtcp_packet_type_counter_observer = &observer;
+  config.rtcp_report_interval = TimeDelta::Millis(1000);
+  auto rtcp_sender = CreateRtcpSender(config);
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpPli));
+  EXPECT_EQ(1, parser()->pli()->num_packets());
+  EXPECT_EQ(kRemoteSsrc, observer.ssrc_);
+  EXPECT_EQ(1U, observer.counter_.pli_packets);
+}
+
+TEST_F(RtcpSenderTest, SendTmmbr) {
+  const unsigned int kBitrateBps = 312000;
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  rtcp_sender->SetTargetBitrate(kBitrateBps);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpTmmbr));
+  EXPECT_EQ(1, parser()->tmmbr()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->tmmbr()->sender_ssrc());
+  EXPECT_EQ(1U, parser()->tmmbr()->requests().size());
+  EXPECT_EQ(kBitrateBps, parser()->tmmbr()->requests()[0].bitrate_bps());
+  // TODO(asapersson): tmmbr_item()->Overhead() looks broken, always zero.
+}
+
+TEST_F(RtcpSenderTest, SendTmmbn) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetSendingStatus(feedback_state(), true);
+  std::vector<rtcp::TmmbItem> bounding_set;
+  const uint32_t kBitrateBps = 32768000;
+  const uint32_t kPacketOh = 40;
+  const uint32_t kSourceSsrc = 12345;
+  const rtcp::TmmbItem tmmbn(kSourceSsrc, kBitrateBps, kPacketOh);
+  bounding_set.push_back(tmmbn);
+  rtcp_sender->SetTmmbn(bounding_set);
+
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpSr));
+  EXPECT_EQ(1, parser()->sender_report()->num_packets());
+  EXPECT_EQ(1, parser()->tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->tmmbn()->sender_ssrc());
+  EXPECT_EQ(1U, parser()->tmmbn()->items().size());
+  EXPECT_EQ(kBitrateBps, parser()->tmmbn()->items()[0].bitrate_bps());
+  EXPECT_EQ(kPacketOh, parser()->tmmbn()->items()[0].packet_overhead());
+  EXPECT_EQ(kSourceSsrc, parser()->tmmbn()->items()[0].ssrc());
+}
+
+// This test is written to verify actual behaviour. It does not seem
+// to make much sense to send an empty TMMBN, since there is no place
+// to put an actual limit here. It's just information that no limit
+// is set, which is kind of the starting assumption.
+// See http://code.google.com/p/webrtc/issues/detail?id=468 for one
+// situation where this caused confusion.
+TEST_F(RtcpSenderTest, SendsTmmbnIfSetAndEmpty) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetSendingStatus(feedback_state(), true);
+  std::vector<rtcp::TmmbItem> bounding_set;
+  rtcp_sender->SetTmmbn(bounding_set);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpSr));
+  EXPECT_EQ(1, parser()->sender_report()->num_packets());
+  EXPECT_EQ(1, parser()->tmmbn()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->tmmbn()->sender_ssrc());
+  EXPECT_EQ(0U, parser()->tmmbn()->items().size());
+}
+
+// This test is written to verify that BYE is always the last packet
+// type in a RTCP compoud packet.  The rtcp_sender is recreated with
+// mock_transport, which is used to check for whether BYE at the end
+// of a RTCP compound packet.
+TEST_F(RtcpSenderTest, ByeMustBeLast) {
+  MockTransport mock_transport;
+  EXPECT_CALL(mock_transport, SendRtcp(_, _))
+      .WillOnce(Invoke([](const uint8_t* data, size_t len) {
+        const uint8_t* next_packet = data;
+        const uint8_t* const packet_end = data + len;
+        rtcp::CommonHeader packet;
+        while (next_packet < packet_end) {
+          EXPECT_TRUE(packet.Parse(next_packet, packet_end - next_packet));
+          next_packet = packet.NextPacket();
+          if (packet.type() ==
+              rtcp::Bye::kPacketType)  // Main test expectation.
+            EXPECT_EQ(0, packet_end - next_packet)
+                << "Bye packet should be last in a compound RTCP packet.";
+          if (next_packet == packet_end)  // Validate test was set correctly.
+            EXPECT_EQ(packet.type(), rtcp::Bye::kPacketType)
+                << "Last packet in this test expected to be Bye.";
+        }
+
+        return true;
+      }));
+
+  // Re-configure rtcp_sender with mock_transport_
+  RTCPSender::Configuration config;
+  config.clock = &clock_;
+  config.receive_statistics = receive_statistics_.get();
+  config.outgoing_transport = &mock_transport;
+  config.rtcp_report_interval = TimeDelta::Millis(1000);
+  config.local_media_ssrc = kSenderSsrc;
+  auto rtcp_sender = CreateRtcpSender(config);
+
+  rtcp_sender->SetTimestampOffset(kStartRtpTimestamp);
+  rtcp_sender->SetLastRtpTime(kRtpTimestamp, clock_.CurrentTime(),
+                              /*payload_type=*/0);
+
+  // Set up REMB info to be included with BYE.
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  rtcp_sender->SetRemb(1234, {});
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpBye));
+}
+
+TEST_F(RtcpSenderTest, SendXrWithTargetBitrate) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  const size_t kNumSpatialLayers = 2;
+  const size_t kNumTemporalLayers = 2;
+  VideoBitrateAllocation allocation;
+  for (size_t sl = 0; sl < kNumSpatialLayers; ++sl) {
+    uint32_t start_bitrate_bps = (sl + 1) * 100000;
+    for (size_t tl = 0; tl < kNumTemporalLayers; ++tl)
+      allocation.SetBitrate(sl, tl, start_bitrate_bps + (tl * 20000));
+  }
+  rtcp_sender->SetVideoBitrateAllocation(allocation);
+
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  EXPECT_EQ(1, parser()->xr()->num_packets());
+  EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
+  const absl::optional<rtcp::TargetBitrate>& target_bitrate =
+      parser()->xr()->target_bitrate();
+  ASSERT_TRUE(target_bitrate);
+  const std::vector<rtcp::TargetBitrate::BitrateItem>& bitrates =
+      target_bitrate->GetTargetBitrates();
+  EXPECT_EQ(kNumSpatialLayers * kNumTemporalLayers, bitrates.size());
+
+  for (size_t sl = 0; sl < kNumSpatialLayers; ++sl) {
+    uint32_t start_bitrate_bps = (sl + 1) * 100000;
+    for (size_t tl = 0; tl < kNumTemporalLayers; ++tl) {
+      size_t index = (sl * kNumSpatialLayers) + tl;
+      const rtcp::TargetBitrate::BitrateItem& item = bitrates[index];
+      EXPECT_EQ(sl, item.spatial_layer);
+      EXPECT_EQ(tl, item.temporal_layer);
+      EXPECT_EQ(start_bitrate_bps + (tl * 20000),
+                item.target_bitrate_kbps * 1000);
+    }
+  }
+}
+
+TEST_F(RtcpSenderTest, SendImmediateXrWithTargetBitrate) {
+  // Initialize. Send a first report right away.
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  clock_.AdvanceTimeMilliseconds(5);
+
+  // Video bitrate allocation generated, save until next time we send a report.
+  VideoBitrateAllocation allocation;
+  allocation.SetBitrate(0, 0, 100000);
+  rtcp_sender->SetVideoBitrateAllocation(allocation);
+  // First seen instance will be sent immediately.
+  EXPECT_TRUE(rtcp_sender->TimeToSendRTCPReport(false));
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  clock_.AdvanceTimeMilliseconds(5);
+
+  // Update bitrate of existing layer, does not quality for immediate sending.
+  allocation.SetBitrate(0, 0, 150000);
+  rtcp_sender->SetVideoBitrateAllocation(allocation);
+  EXPECT_FALSE(rtcp_sender->TimeToSendRTCPReport(false));
+
+  // A new spatial layer enabled, signal this as soon as possible.
+  allocation.SetBitrate(1, 0, 200000);
+  rtcp_sender->SetVideoBitrateAllocation(allocation);
+  EXPECT_TRUE(rtcp_sender->TimeToSendRTCPReport(false));
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  clock_.AdvanceTimeMilliseconds(5);
+
+  // Explicitly disable top layer. The same set of layers now has a bitrate
+  // defined, but the explicit 0 indicates shutdown. Signal immediately.
+  allocation.SetBitrate(1, 0, 0);
+  EXPECT_FALSE(rtcp_sender->TimeToSendRTCPReport(false));
+  rtcp_sender->SetVideoBitrateAllocation(allocation);
+  EXPECT_TRUE(rtcp_sender->TimeToSendRTCPReport(false));
+}
+
+TEST_F(RtcpSenderTest, SendTargetBitrateExplicitZeroOnStreamRemoval) {
+  // Set up and send a bitrate allocation with two layers.
+
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kCompound);
+  VideoBitrateAllocation allocation;
+  allocation.SetBitrate(0, 0, 100000);
+  allocation.SetBitrate(1, 0, 200000);
+  rtcp_sender->SetVideoBitrateAllocation(allocation);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  absl::optional<rtcp::TargetBitrate> target_bitrate =
+      parser()->xr()->target_bitrate();
+  ASSERT_TRUE(target_bitrate);
+  std::vector<rtcp::TargetBitrate::BitrateItem> bitrates =
+      target_bitrate->GetTargetBitrates();
+  ASSERT_EQ(2u, bitrates.size());
+  EXPECT_EQ(bitrates[0].target_bitrate_kbps,
+            allocation.GetBitrate(0, 0) / 1000);
+  EXPECT_EQ(bitrates[1].target_bitrate_kbps,
+            allocation.GetBitrate(1, 0) / 1000);
+
+  // Create a new allocation, where the second stream is no longer available.
+  VideoBitrateAllocation new_allocation;
+  new_allocation.SetBitrate(0, 0, 150000);
+  rtcp_sender->SetVideoBitrateAllocation(new_allocation);
+  EXPECT_EQ(0, rtcp_sender->SendRTCP(feedback_state(), kRtcpReport));
+  target_bitrate = parser()->xr()->target_bitrate();
+  ASSERT_TRUE(target_bitrate);
+  bitrates = target_bitrate->GetTargetBitrates();
+
+  // Two bitrates should still be set, with an explicit entry indicating the
+  // removed stream is gone.
+  ASSERT_EQ(2u, bitrates.size());
+  EXPECT_EQ(bitrates[0].target_bitrate_kbps,
+            new_allocation.GetBitrate(0, 0) / 1000);
+  EXPECT_EQ(bitrates[1].target_bitrate_kbps, 0u);
+}
+
+TEST_F(RtcpSenderTest, DoesntSchedulesInitialReportWhenSsrcSetOnConstruction) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+  rtcp_sender->SetRemoteSSRC(kRemoteSsrc);
+  // New report should not have been scheduled yet.
+  clock_.AdvanceTimeMilliseconds(100);
+  EXPECT_FALSE(rtcp_sender->TimeToSendRTCPReport(false));
+}
+
+TEST_F(RtcpSenderTest, SendsCombinedRtcpPacket) {
+  auto rtcp_sender = CreateRtcpSender(GetDefaultConfig());
+  rtcp_sender->SetRTCPStatus(RtcpMode::kReducedSize);
+
+  std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets;
+  auto transport_feedback = std::make_unique<rtcp::TransportFeedback>();
+  transport_feedback->AddReceivedPacket(321, Timestamp::Millis(10));
+  packets.push_back(std::move(transport_feedback));
+  auto remote_estimate = std::make_unique<rtcp::RemoteEstimate>();
+  packets.push_back(std::move(remote_estimate));
+  rtcp_sender->SendCombinedRtcpPacket(std::move(packets));
+
+  EXPECT_EQ(parser()->transport_feedback()->num_packets(), 1);
+  EXPECT_EQ(parser()->transport_feedback()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(parser()->app()->num_packets(), 1);
+  EXPECT_EQ(parser()->app()->sender_ssrc(), kSenderSsrc);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc
new file mode 100644
index 0000000000..f265bd5825
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc
@@ -0,0 +1,150 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_transceiver.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/cleanup/cleanup.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+RtcpTransceiver::RtcpTransceiver(const RtcpTransceiverConfig& config)
+    : clock_(config.clock),
+      task_queue_(config.task_queue),
+      rtcp_transceiver_(std::make_unique<RtcpTransceiverImpl>(config)) {
+  RTC_DCHECK(task_queue_);
+}
+
+RtcpTransceiver::~RtcpTransceiver() {
+  if (!rtcp_transceiver_)
+    return;
+  auto rtcp_transceiver = std::move(rtcp_transceiver_);
+  task_queue_->PostTask([rtcp_transceiver = std::move(rtcp_transceiver)] {
+    rtcp_transceiver->StopPeriodicTask();
+  });
+  RTC_DCHECK(!rtcp_transceiver_);
+}
+
+void RtcpTransceiver::Stop(absl::AnyInvocable<void() &&> on_destroyed) {
+  RTC_DCHECK(rtcp_transceiver_);
+  auto rtcp_transceiver = std::move(rtcp_transceiver_);
+  absl::Cleanup cleanup = std::move(on_destroyed);
+  task_queue_->PostTask(
+      [rtcp_transceiver = std::move(rtcp_transceiver),
+       cleanup = std::move(cleanup)] { rtcp_transceiver->StopPeriodicTask(); });
+  RTC_DCHECK(!rtcp_transceiver_);
+}
+
+void RtcpTransceiver::AddMediaReceiverRtcpObserver(
+    uint32_t remote_ssrc,
+    MediaReceiverRtcpObserver* observer) {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  task_queue_->PostTask([ptr, remote_ssrc, observer] {
+    ptr->AddMediaReceiverRtcpObserver(remote_ssrc, observer);
+  });
+}
+
+void RtcpTransceiver::RemoveMediaReceiverRtcpObserver(
+    uint32_t remote_ssrc,
+    MediaReceiverRtcpObserver* observer,
+    absl::AnyInvocable<void() &&> on_removed) {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  absl::Cleanup cleanup = std::move(on_removed);
+  task_queue_->PostTask(
+      [ptr, remote_ssrc, observer, cleanup = std::move(cleanup)] {
+        ptr->RemoveMediaReceiverRtcpObserver(remote_ssrc, observer);
+      });
+}
+
+void RtcpTransceiver::SetReadyToSend(bool ready) {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  task_queue_->PostTask([ptr, ready] { ptr->SetReadyToSend(ready); });
+}
+
+void RtcpTransceiver::ReceivePacket(rtc::CopyOnWriteBuffer packet) {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  Timestamp now = clock_->CurrentTime();
+  task_queue_->PostTask(
+      [ptr, packet, now] { ptr->ReceivePacket(packet, now); });
+}
+
+void RtcpTransceiver::SendCompoundPacket() {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  task_queue_->PostTask([ptr] { ptr->SendCompoundPacket(); });
+}
+
+void RtcpTransceiver::SetRemb(int64_t bitrate_bps,
+                              std::vector<uint32_t> ssrcs) {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  task_queue_->PostTask([ptr, bitrate_bps, ssrcs = std::move(ssrcs)]() mutable {
+    ptr->SetRemb(bitrate_bps, std::move(ssrcs));
+  });
+}
+
+void RtcpTransceiver::UnsetRemb() {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  task_queue_->PostTask([ptr] { ptr->UnsetRemb(); });
+}
+
+void RtcpTransceiver::SendCombinedRtcpPacket(
+    std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  task_queue_->PostTask(
+      [ptr, rtcp_packets = std::move(rtcp_packets)]() mutable {
+        ptr->SendCombinedRtcpPacket(std::move(rtcp_packets));
+      });
+}
+
+void RtcpTransceiver::SendNack(uint32_t ssrc,
+                               std::vector<uint16_t> sequence_numbers) {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  task_queue_->PostTask(
+      [ptr, ssrc, sequence_numbers = std::move(sequence_numbers)]() mutable {
+        ptr->SendNack(ssrc, std::move(sequence_numbers));
+      });
+}
+
+void RtcpTransceiver::SendPictureLossIndication(uint32_t ssrc) {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  task_queue_->PostTask([ptr, ssrc] { ptr->SendPictureLossIndication(ssrc); });
+}
+
+void RtcpTransceiver::SendFullIntraRequest(std::vector<uint32_t> ssrcs) {
+  return SendFullIntraRequest(std::move(ssrcs), true);
+}
+
+void RtcpTransceiver::SendFullIntraRequest(std::vector<uint32_t> ssrcs,
+                                           bool new_request) {
+  RTC_CHECK(rtcp_transceiver_);
+  RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
+  task_queue_->PostTask([ptr, ssrcs = std::move(ssrcs), new_request] {
+    ptr->SendFullIntraRequest(ssrcs, new_request);
+  });
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver.h
new file mode 100644
index 0000000000..22fcc73337
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver.h
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/rtp_rtcp/source/rtcp_transceiver_config.h"
+#include "modules/rtp_rtcp/source/rtcp_transceiver_impl.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+//
+// Manage incoming and outgoing rtcp messages for multiple BUNDLED streams.
+//
+// This class is thread-safe wrapper of RtcpTransceiverImpl
+class RtcpTransceiver : public RtcpFeedbackSenderInterface {
+ public:
+  explicit RtcpTransceiver(const RtcpTransceiverConfig& config);
+  RtcpTransceiver(const RtcpTransceiver&) = delete;
+  RtcpTransceiver& operator=(const RtcpTransceiver&) = delete;
+  // Note that interfaces provided in constructor still might be used after the
+  // destructor. However they can only be used on the confic.task_queue.
+  // Use Stop function to get notified when they are no longer used or
+  // ensure those objects outlive the task queue.
+  ~RtcpTransceiver() override;
+
+  // Start asynchronious destruction of the RtcpTransceiver.
+  // It is safe to call destructor right after Stop exits.
+  // No other methods can be called.
+  // Note that interfaces provided in constructor or registered with AddObserver
+  // still might be used by the transceiver on the task queue
+  // until `on_destroyed` runs.
+  void Stop(absl::AnyInvocable<void() &&> on_destroyed);
+
+  // Registers observer to be notified about incoming rtcp packets.
+  // Calls to observer will be done on the `config.task_queue`.
+  void AddMediaReceiverRtcpObserver(uint32_t remote_ssrc,
+                                    MediaReceiverRtcpObserver* observer);
+  // Deregisters the observer. Might return before observer is deregistered.
+  // Runs `on_removed` when observer is deregistered.
+  void RemoveMediaReceiverRtcpObserver(
+      uint32_t remote_ssrc,
+      MediaReceiverRtcpObserver* observer,
+      absl::AnyInvocable<void() &&> on_removed);
+
+  // Enables/disables sending rtcp packets eventually.
+  // Packets may be sent after the SetReadyToSend(false) returns, but no new
+  // packets will be scheduled.
+  void SetReadyToSend(bool ready);
+
+  // Handles incoming rtcp packets.
+  void ReceivePacket(rtc::CopyOnWriteBuffer packet);
+
+  // Sends RTCP packets starting with a sender or receiver report.
+  void SendCompoundPacket();
+
+  // (REMB) Receiver Estimated Max Bitrate.
+  // Includes REMB in following compound packets and sends a REMB message
+  // immediately if 'RtcpTransceiverConfig::send_remb_on_change' is set.
+  void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
+  // Stops sending REMB in following compound packets.
+  void UnsetRemb() override;
+
+  // TODO(bugs.webrtc.org/8239): Remove SendCombinedRtcpPacket
+  // and move generating of the TransportFeedback message inside
+  // RtcpTransceiverImpl when there is one RtcpTransceiver per rtp transport.
+  void SendCombinedRtcpPacket(
+      std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) override;
+
+  // Reports missing packets, https://tools.ietf.org/html/rfc4585#section-6.2.1
+  void SendNack(uint32_t ssrc, std::vector<uint16_t> sequence_numbers);
+
+  // Requests new key frame.
+  // using PLI, https://tools.ietf.org/html/rfc4585#section-6.3.1.1
+  void SendPictureLossIndication(uint32_t ssrc);
+  // using FIR, https://tools.ietf.org/html/rfc5104#section-4.3.1.2
+  // Use the SendFullIntraRequest(ssrcs, true) instead.
+  void SendFullIntraRequest(std::vector<uint32_t> ssrcs);
+  // If new_request is true then requested sequence no. will increase for each
+  // requested ssrc.
+  void SendFullIntraRequest(std::vector<uint32_t> ssrcs, bool new_request);
+
+ private:
+  Clock* const clock_;
+  TaskQueueBase* const task_queue_;
+  std::unique_ptr<RtcpTransceiverImpl> rtcp_transceiver_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.cc
new file mode 100644
index 0000000000..7acaa0f600
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.cc
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_transceiver_config.h"
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+RtcpTransceiverConfig::RtcpTransceiverConfig() = default;
+RtcpTransceiverConfig::RtcpTransceiverConfig(const RtcpTransceiverConfig&) =
+    default;
+RtcpTransceiverConfig& RtcpTransceiverConfig::operator=(
+    const RtcpTransceiverConfig&) = default;
+RtcpTransceiverConfig::~RtcpTransceiverConfig() = default;
+
+bool RtcpTransceiverConfig::Validate() const {
+  if (feedback_ssrc == 0) {
+    RTC_LOG(LS_WARNING)
+        << debug_id
+        << "Ssrc 0 may be treated by some implementation as invalid.";
+  }
+  if (cname.size() > 255) {
+    RTC_LOG(LS_ERROR) << debug_id << "cname can be maximum 255 characters.";
+    return false;
+  }
+  if (max_packet_size < 100) {
+    RTC_LOG(LS_ERROR) << debug_id << "max packet size " << max_packet_size
+                      << " is too small.";
+    return false;
+  }
+  if (max_packet_size > IP_PACKET_SIZE) {
+    RTC_LOG(LS_ERROR) << debug_id << "max packet size " << max_packet_size
+                      << " more than " << IP_PACKET_SIZE << " is unsupported.";
+    return false;
+  }
+  if (clock == nullptr) {
+    RTC_LOG(LS_ERROR) << debug_id << "clock must be set";
+    return false;
+  }
+  if (!outgoing_transport) {
+    RTC_LOG(LS_ERROR) << debug_id << "outgoing transport must be set";
+    return false;
+  }
+  if (initial_report_delay < TimeDelta::Zero()) {
+    RTC_LOG(LS_ERROR) << debug_id << "delay " << initial_report_delay.ms()
+                      << "ms before first report shouldn't be negative.";
+    return false;
+  }
+  if (report_period <= TimeDelta::Zero()) {
+    RTC_LOG(LS_ERROR) << debug_id << "period " << report_period.ms()
+                      << "ms between reports should be positive.";
+    return false;
+  }
+  if (schedule_periodic_compound_packets && task_queue == nullptr) {
+    RTC_LOG(LS_ERROR) << debug_id
+                      << "missing task queue for periodic compound packets";
+    return false;
+  }
+  if (rtcp_mode != RtcpMode::kCompound && rtcp_mode != RtcpMode::kReducedSize) {
+    RTC_LOG(LS_ERROR) << debug_id << "unsupported rtcp mode";
+    return false;
+  }
+  if (non_sender_rtt_measurement && !network_link_observer) {
+    RTC_LOG(LS_WARNING) << debug_id
+                        << "Enabled special feature to calculate rtt, but no "
+                           "rtt observer is provided.";
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h
new file mode 100644
index 0000000000..3122ad5c36
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h
@@ -0,0 +1,188 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_
+
+#include <string>
+
+#include "api/array_view.h"
+#include "api/rtp_headers.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+class ReceiveStatisticsProvider;
+class Transport;
+
+// Interface to watch incoming rtcp packets related to the link in general.
+// All message handlers have default empty implementation. This way users only
+// need to implement the ones they are interested in.
+// All message handles pass `receive_time` parameter, which is receive time
+// of the rtcp packet that triggered the update.
+class NetworkLinkRtcpObserver {
+ public:
+  virtual ~NetworkLinkRtcpObserver() = default;
+
+  virtual void OnTransportFeedback(Timestamp receive_time,
+                                   const rtcp::TransportFeedback& feedback) {}
+  virtual void OnReceiverEstimatedMaxBitrate(Timestamp receive_time,
+                                             DataRate bitrate) {}
+  virtual void OnReportBlocks(
+      Timestamp receive_time,
+      rtc::ArrayView<const rtcp::ReportBlock> report_blocks) {}
+  virtual void OnRttUpdate(Timestamp receive_time, TimeDelta rtt) {}
+};
+
+// Interface to watch incoming rtcp packets by media (rtp) receiver.
+// All message handlers have default empty implementation. This way users only
+// need to implement the ones they are interested in.
+class MediaReceiverRtcpObserver {
+ public:
+  virtual ~MediaReceiverRtcpObserver() = default;
+
+  virtual void OnSenderReport(uint32_t sender_ssrc,
+                              NtpTime ntp_time,
+                              uint32_t rtp_time) {}
+  virtual void OnBye(uint32_t sender_ssrc) {}
+  virtual void OnBitrateAllocation(uint32_t sender_ssrc,
+                                   const VideoBitrateAllocation& allocation) {}
+};
+
+// Handles RTCP related messages for a single RTP stream (i.e. single SSRC)
+class RtpStreamRtcpHandler {
+ public:
+  virtual ~RtpStreamRtcpHandler() = default;
+
+  // Statistic about sent RTP packets to propagate to RTCP sender report.
+  class RtpStats {
+   public:
+    RtpStats() = default;
+    RtpStats(const RtpStats&) = default;
+    RtpStats& operator=(const RtpStats&) = default;
+    ~RtpStats() = default;
+
+    size_t num_sent_packets() const { return num_sent_packets_; }
+    size_t num_sent_bytes() const { return num_sent_bytes_; }
+    Timestamp last_capture_time() const { return last_capture_time_; }
+    uint32_t last_rtp_timestamp() const { return last_rtp_timestamp_; }
+    int last_clock_rate() const { return last_clock_rate_; }
+
+    void set_num_sent_packets(size_t v) { num_sent_packets_ = v; }
+    void set_num_sent_bytes(size_t v) { num_sent_bytes_ = v; }
+    void set_last_capture_time(Timestamp v) { last_capture_time_ = v; }
+    void set_last_rtp_timestamp(uint32_t v) { last_rtp_timestamp_ = v; }
+    void set_last_clock_rate(int v) { last_clock_rate_ = v; }
+
+   private:
+    size_t num_sent_packets_ = 0;
+    size_t num_sent_bytes_ = 0;
+    Timestamp last_capture_time_ = Timestamp::Zero();
+    uint32_t last_rtp_timestamp_ = 0;
+    int last_clock_rate_ = 90'000;
+  };
+  virtual RtpStats SentStats() = 0;
+
+  virtual void OnNack(uint32_t sender_ssrc,
+                      rtc::ArrayView<const uint16_t> sequence_numbers) {}
+  virtual void OnFir(uint32_t sender_ssrc) {}
+  virtual void OnPli(uint32_t sender_ssrc) {}
+  virtual void OnReportBlock(uint32_t sender_ssrc,
+                             const rtcp::ReportBlock& report_block) {}
+};
+
+struct RtcpTransceiverConfig {
+  RtcpTransceiverConfig();
+  RtcpTransceiverConfig(const RtcpTransceiverConfig&);
+  RtcpTransceiverConfig& operator=(const RtcpTransceiverConfig&);
+  ~RtcpTransceiverConfig();
+
+  // Logs the error and returns false if configuration miss key objects or
+  // is inconsistant. May log warnings.
+  bool Validate() const;
+
+  // Used to prepend all log messages. Can be empty.
+  std::string debug_id;
+
+  // Ssrc to use as default sender ssrc, e.g. for transport-wide feedbacks.
+  uint32_t feedback_ssrc = 1;
+
+  // Canonical End-Point Identifier of the local particiapnt.
+  // Defined in rfc3550 section 6 note 2 and section 6.5.1.
+  std::string cname;
+
+  // Maximum packet size outgoing transport accepts.
+  size_t max_packet_size = 1200;
+
+  // The clock to use when querying for the NTP time. Should be set.
+  Clock* clock = nullptr;
+
+  // Transport to send rtcp packets to. Should be set.
+  Transport* outgoing_transport = nullptr;
+
+  // Queue for scheduling delayed tasks, e.g. sending periodic compound packets.
+  TaskQueueBase* task_queue = nullptr;
+
+  // Rtcp report block generator for outgoing receiver reports.
+  ReceiveStatisticsProvider* receive_statistics = nullptr;
+
+  // Should outlive RtcpTransceiver.
+  // Callbacks will be invoked on the `task_queue`.
+  NetworkLinkRtcpObserver* network_link_observer = nullptr;
+
+  // Configures if sending should
+  //  enforce compound packets: https://tools.ietf.org/html/rfc4585#section-3.1
+  //  or allow reduced size packets: https://tools.ietf.org/html/rfc5506
+  // Receiving accepts both compound and reduced-size packets.
+  RtcpMode rtcp_mode = RtcpMode::kCompound;
+  //
+  // Tuning parameters.
+  //
+  // Initial state if `outgoing_transport` ready to accept packets.
+  bool initial_ready_to_send = true;
+  // Delay before 1st periodic compound packet.
+  TimeDelta initial_report_delay = TimeDelta::Millis(500);
+
+  // Period between periodic compound packets.
+  TimeDelta report_period = TimeDelta::Seconds(1);
+
+  //
+  // Flags for features and experiments.
+  //
+  bool schedule_periodic_compound_packets = true;
+  // Estimate RTT as non-sender as described in
+  // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
+  bool non_sender_rtt_measurement = false;
+
+  // Reply to incoming RRTR messages so that remote endpoint may estimate RTT as
+  // non-sender as described in https://tools.ietf.org/html/rfc3611#section-4.4
+  // and #section-4.5
+  bool reply_to_non_sender_rtt_measurement = true;
+
+  // Reply to incoming RRTR messages multiple times, one per sender SSRC, to
+  // support clients that calculate and process RTT per sender SSRC.
+  bool reply_to_non_sender_rtt_mesaurments_on_all_ssrcs = true;
+
+  // Allows a REMB message to be sent immediately when SetRemb is called without
+  // having to wait for the next compount message to be sent.
+  bool send_remb_on_change = false;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
new file mode 100644
index 0000000000..bb4f96b970
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
@@ -0,0 +1,857 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_transceiver_impl.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/memory/memory.h"
+#include "api/call/transport.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/containers/flat_map.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/divide_round.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+
+struct SenderReportTimes {
+  Timestamp local_received_time;
+  NtpTime remote_sent_time;
+};
+
+}  // namespace
+
+struct RtcpTransceiverImpl::RemoteSenderState {
+  uint8_t fir_sequence_number = 0;
+  absl::optional<SenderReportTimes> last_received_sender_report;
+  std::vector<MediaReceiverRtcpObserver*> observers;
+};
+
+struct RtcpTransceiverImpl::LocalSenderState {
+  uint32_t ssrc;
+  size_t last_num_sent_bytes = 0;
+  // Sequence number of the last FIR message per sender SSRC.
+  flat_map<uint32_t, uint8_t> last_fir;
+  RtpStreamRtcpHandler* handler = nullptr;
+};
+
+// Helper to put several RTCP packets into lower layer datagram composing
+// Compound or Reduced-Size RTCP packet, as defined by RFC 5506 section 2.
+// TODO(danilchap): When in compound mode and packets are so many that several
+// compound RTCP packets need to be generated, ensure each packet is compound.
+class RtcpTransceiverImpl::PacketSender {
+ public:
+  PacketSender(rtcp::RtcpPacket::PacketReadyCallback callback,
+               size_t max_packet_size)
+      : callback_(callback), max_packet_size_(max_packet_size) {
+    RTC_CHECK_LE(max_packet_size, IP_PACKET_SIZE);
+  }
+  ~PacketSender() { RTC_DCHECK_EQ(index_, 0) << "Unsent rtcp packet."; }
+
+  // Appends a packet to pending compound packet.
+  // Sends rtcp compound packet if buffer was already full and resets buffer.
+  void AppendPacket(const rtcp::RtcpPacket& packet) {
+    packet.Create(buffer_, &index_, max_packet_size_, callback_);
+  }
+
+  // Sends pending rtcp compound packet.
+  void Send() {
+    if (index_ > 0) {
+      callback_(rtc::ArrayView<const uint8_t>(buffer_, index_));
+      index_ = 0;
+    }
+  }
+
+  bool IsEmpty() const { return index_ == 0; }
+
+ private:
+  const rtcp::RtcpPacket::PacketReadyCallback callback_;
+  const size_t max_packet_size_;
+  size_t index_ = 0;
+  uint8_t buffer_[IP_PACKET_SIZE];
+};
+
+RtcpTransceiverImpl::RtcpTransceiverImpl(const RtcpTransceiverConfig& config)
+    : config_(config), ready_to_send_(config.initial_ready_to_send) {
+  RTC_CHECK(config_.Validate());
+  if (ready_to_send_ && config_.schedule_periodic_compound_packets) {
+    SchedulePeriodicCompoundPackets(config_.initial_report_delay);
+  }
+}
+
+RtcpTransceiverImpl::~RtcpTransceiverImpl() = default;
+
+void RtcpTransceiverImpl::AddMediaReceiverRtcpObserver(
+    uint32_t remote_ssrc,
+    MediaReceiverRtcpObserver* observer) {
+  if (config_.receive_statistics == nullptr && remote_senders_.empty()) {
+    RTC_LOG(LS_WARNING) << config_.debug_id
+                        << "receive statistic is not set. RTCP report blocks "
+                           "will not be generated.";
+  }
+  auto& stored = remote_senders_[remote_ssrc].observers;
+  RTC_DCHECK(!absl::c_linear_search(stored, observer));
+  stored.push_back(observer);
+}
+
+void RtcpTransceiverImpl::RemoveMediaReceiverRtcpObserver(
+    uint32_t remote_ssrc,
+    MediaReceiverRtcpObserver* observer) {
+  auto remote_sender_it = remote_senders_.find(remote_ssrc);
+  if (remote_sender_it == remote_senders_.end())
+    return;
+  auto& stored = remote_sender_it->second.observers;
+  auto it = absl::c_find(stored, observer);
+  if (it == stored.end())
+    return;
+  stored.erase(it);
+}
+
+bool RtcpTransceiverImpl::AddMediaSender(uint32_t local_ssrc,
+                                         RtpStreamRtcpHandler* handler) {
+  RTC_DCHECK(handler != nullptr);
+  LocalSenderState state;
+  state.ssrc = local_ssrc;
+  state.handler = handler;
+  local_senders_.push_back(state);
+  auto it = std::prev(local_senders_.end());
+  auto [unused, inserted] = local_senders_by_ssrc_.emplace(local_ssrc, it);
+  if (!inserted) {
+    local_senders_.pop_back();
+    return false;
+  }
+  return true;
+}
+
+bool RtcpTransceiverImpl::RemoveMediaSender(uint32_t local_ssrc) {
+  auto index_it = local_senders_by_ssrc_.find(local_ssrc);
+  if (index_it == local_senders_by_ssrc_.end()) {
+    return false;
+  }
+  local_senders_.erase(index_it->second);
+  local_senders_by_ssrc_.erase(index_it);
+  return true;
+}
+
+void RtcpTransceiverImpl::SetReadyToSend(bool ready) {
+  if (config_.schedule_periodic_compound_packets) {
+    if (ready_to_send_ && !ready)
+      periodic_task_handle_.Stop();
+
+    if (!ready_to_send_ && ready)  // Restart periodic sending.
+      SchedulePeriodicCompoundPackets(config_.report_period / 2);
+  }
+  ready_to_send_ = ready;
+}
+
+void RtcpTransceiverImpl::ReceivePacket(rtc::ArrayView<const uint8_t> packet,
+                                        Timestamp now) {
+  // Report blocks may be spread across multiple sender and receiver reports.
+  std::vector<rtcp::ReportBlock> report_blocks;
+
+  while (!packet.empty()) {
+    rtcp::CommonHeader rtcp_block;
+    if (!rtcp_block.Parse(packet.data(), packet.size()))
+      break;
+
+    HandleReceivedPacket(rtcp_block, now, report_blocks);
+
+    packet = packet.subview(rtcp_block.packet_size());
+  }
+
+  if (!report_blocks.empty()) {
+    ProcessReportBlocks(now, report_blocks);
+  }
+}
+
+void RtcpTransceiverImpl::SendCompoundPacket() {
+  if (!ready_to_send_)
+    return;
+  SendPeriodicCompoundPacket();
+  ReschedulePeriodicCompoundPackets();
+}
+
+void RtcpTransceiverImpl::SetRemb(int64_t bitrate_bps,
+                                  std::vector<uint32_t> ssrcs) {
+  RTC_DCHECK_GE(bitrate_bps, 0);
+
+  bool send_now = config_.send_remb_on_change &&
+                  (!remb_.has_value() || bitrate_bps != remb_->bitrate_bps());
+  remb_.emplace();
+  remb_->SetSsrcs(std::move(ssrcs));
+  remb_->SetBitrateBps(bitrate_bps);
+  remb_->SetSenderSsrc(config_.feedback_ssrc);
+  // TODO(bugs.webrtc.org/8239): Move logic from PacketRouter for sending remb
+  // immideately on large bitrate change when there is one RtcpTransceiver per
+  // rtp transport.
+  if (send_now) {
+    absl::optional<rtcp::Remb> remb;
+    remb.swap(remb_);
+    SendImmediateFeedback(*remb);
+    remb.swap(remb_);
+  }
+}
+
+void RtcpTransceiverImpl::UnsetRemb() {
+  remb_.reset();
+}
+
+void RtcpTransceiverImpl::SendRawPacket(rtc::ArrayView<const uint8_t> packet) {
+  if (!ready_to_send_)
+    return;
+  // Unlike other senders, this functions just tries to send packet away and
+  // disregard rtcp_mode, max_packet_size or anything else.
+  // TODO(bugs.webrtc.org/8239): respect config_ by creating the
+  // TransportFeedback inside this class when there is one per rtp transport.
+  config_.outgoing_transport->SendRtcp(packet.data(), packet.size());
+}
+
+void RtcpTransceiverImpl::SendNack(uint32_t ssrc,
+                                   std::vector<uint16_t> sequence_numbers) {
+  RTC_DCHECK(!sequence_numbers.empty());
+  if (!ready_to_send_)
+    return;
+  rtcp::Nack nack;
+  nack.SetSenderSsrc(config_.feedback_ssrc);
+  nack.SetMediaSsrc(ssrc);
+  nack.SetPacketIds(std::move(sequence_numbers));
+  SendImmediateFeedback(nack);
+}
+
+void RtcpTransceiverImpl::SendPictureLossIndication(uint32_t ssrc) {
+  if (!ready_to_send_)
+    return;
+  rtcp::Pli pli;
+  pli.SetSenderSsrc(config_.feedback_ssrc);
+  pli.SetMediaSsrc(ssrc);
+  SendImmediateFeedback(pli);
+}
+
+void RtcpTransceiverImpl::SendFullIntraRequest(
+    rtc::ArrayView<const uint32_t> ssrcs,
+    bool new_request) {
+  RTC_DCHECK(!ssrcs.empty());
+  if (!ready_to_send_)
+    return;
+  rtcp::Fir fir;
+  fir.SetSenderSsrc(config_.feedback_ssrc);
+  for (uint32_t media_ssrc : ssrcs) {
+    uint8_t& command_seq_num = remote_senders_[media_ssrc].fir_sequence_number;
+    if (new_request)
+      command_seq_num += 1;
+    fir.AddRequestTo(media_ssrc, command_seq_num);
+  }
+  SendImmediateFeedback(fir);
+}
+
+void RtcpTransceiverImpl::HandleReceivedPacket(
+    const rtcp::CommonHeader& rtcp_packet_header,
+    Timestamp now,
+    std::vector<rtcp::ReportBlock>& report_blocks) {
+  switch (rtcp_packet_header.type()) {
+    case rtcp::Bye::kPacketType:
+      HandleBye(rtcp_packet_header);
+      break;
+    case rtcp::SenderReport::kPacketType:
+      HandleSenderReport(rtcp_packet_header, now, report_blocks);
+      break;
+    case rtcp::ReceiverReport::kPacketType:
+      HandleReceiverReport(rtcp_packet_header, report_blocks);
+      break;
+    case rtcp::ExtendedReports::kPacketType:
+      HandleExtendedReports(rtcp_packet_header, now);
+      break;
+    case rtcp::Psfb::kPacketType:
+      HandlePayloadSpecificFeedback(rtcp_packet_header, now);
+      break;
+    case rtcp::Rtpfb::kPacketType:
+      HandleRtpFeedback(rtcp_packet_header, now);
+      break;
+  }
+}
+
+void RtcpTransceiverImpl::HandleBye(
+    const rtcp::CommonHeader& rtcp_packet_header) {
+  rtcp::Bye bye;
+  if (!bye.Parse(rtcp_packet_header))
+    return;
+  auto remote_sender_it = remote_senders_.find(bye.sender_ssrc());
+  if (remote_sender_it == remote_senders_.end())
+    return;
+  for (MediaReceiverRtcpObserver* observer : remote_sender_it->second.observers)
+    observer->OnBye(bye.sender_ssrc());
+}
+
+void RtcpTransceiverImpl::HandleSenderReport(
+    const rtcp::CommonHeader& rtcp_packet_header,
+    Timestamp now,
+    std::vector<rtcp::ReportBlock>& report_blocks) {
+  rtcp::SenderReport sender_report;
+  if (!sender_report.Parse(rtcp_packet_header))
+    return;
+  RemoteSenderState& remote_sender =
+      remote_senders_[sender_report.sender_ssrc()];
+  remote_sender.last_received_sender_report = {{now, sender_report.ntp()}};
+  const auto& received_report_blocks = sender_report.report_blocks();
+  CallbackOnReportBlocks(sender_report.sender_ssrc(), received_report_blocks);
+  report_blocks.insert(report_blocks.end(), received_report_blocks.begin(),
+                       received_report_blocks.end());
+
+  for (MediaReceiverRtcpObserver* observer : remote_sender.observers)
+    observer->OnSenderReport(sender_report.sender_ssrc(), sender_report.ntp(),
+                             sender_report.rtp_timestamp());
+}
+
+void RtcpTransceiverImpl::HandleReceiverReport(
+    const rtcp::CommonHeader& rtcp_packet_header,
+    std::vector<rtcp::ReportBlock>& report_blocks) {
+  rtcp::ReceiverReport receiver_report;
+  if (!receiver_report.Parse(rtcp_packet_header)) {
+    return;
+  }
+  const auto& received_report_blocks = receiver_report.report_blocks();
+  CallbackOnReportBlocks(receiver_report.sender_ssrc(), received_report_blocks);
+  report_blocks.insert(report_blocks.end(), received_report_blocks.begin(),
+                       received_report_blocks.end());
+}
+
+void RtcpTransceiverImpl::CallbackOnReportBlocks(
+    uint32_t sender_ssrc,
+    rtc::ArrayView<const rtcp::ReportBlock> report_blocks) {
+  if (local_senders_.empty()) {
+    return;
+  }
+  for (const rtcp::ReportBlock& block : report_blocks) {
+    auto sender_it = local_senders_by_ssrc_.find(block.source_ssrc());
+    if (sender_it != local_senders_by_ssrc_.end()) {
+      sender_it->second->handler->OnReportBlock(sender_ssrc, block);
+    }
+  }
+}
+
+void RtcpTransceiverImpl::HandlePayloadSpecificFeedback(
+    const rtcp::CommonHeader& rtcp_packet_header,
+    Timestamp now) {
+  switch (rtcp_packet_header.fmt()) {
+    case rtcp::Fir::kFeedbackMessageType:
+      HandleFir(rtcp_packet_header);
+      break;
+    case rtcp::Pli::kFeedbackMessageType:
+      HandlePli(rtcp_packet_header);
+      break;
+    case rtcp::Psfb::kAfbMessageType:
+      HandleRemb(rtcp_packet_header, now);
+      break;
+  }
+}
+
+void RtcpTransceiverImpl::HandleFir(
+    const rtcp::CommonHeader& rtcp_packet_header) {
+  rtcp::Fir fir;
+  if (local_senders_.empty() || !fir.Parse(rtcp_packet_header)) {
+    return;
+  }
+  for (const rtcp::Fir::Request& r : fir.requests()) {
+    auto it = local_senders_by_ssrc_.find(r.ssrc);
+    if (it == local_senders_by_ssrc_.end()) {
+      continue;
+    }
+    auto [fir_it, is_new] =
+        it->second->last_fir.emplace(fir.sender_ssrc(), r.seq_nr);
+    if (is_new || fir_it->second != r.seq_nr) {
+      it->second->handler->OnFir(fir.sender_ssrc());
+      fir_it->second = r.seq_nr;
+    }
+  }
+}
+
+void RtcpTransceiverImpl::HandlePli(
+    const rtcp::CommonHeader& rtcp_packet_header) {
+  rtcp::Pli pli;
+  if (local_senders_.empty() || !pli.Parse(rtcp_packet_header)) {
+    return;
+  }
+  auto it = local_senders_by_ssrc_.find(pli.media_ssrc());
+  if (it != local_senders_by_ssrc_.end()) {
+    it->second->handler->OnPli(pli.sender_ssrc());
+  }
+}
+
+void RtcpTransceiverImpl::HandleRemb(
+    const rtcp::CommonHeader& rtcp_packet_header,
+    Timestamp now) {
+  rtcp::Remb remb;
+  if (config_.network_link_observer == nullptr ||
+      !remb.Parse(rtcp_packet_header)) {
+    return;
+  }
+  config_.network_link_observer->OnReceiverEstimatedMaxBitrate(
+      now, DataRate::BitsPerSec(remb.bitrate_bps()));
+}
+
+void RtcpTransceiverImpl::HandleRtpFeedback(
+    const rtcp::CommonHeader& rtcp_packet_header,
+    Timestamp now) {
+  switch (rtcp_packet_header.fmt()) {
+    case rtcp::Nack::kFeedbackMessageType:
+      HandleNack(rtcp_packet_header);
+      break;
+    case rtcp::TransportFeedback::kFeedbackMessageType:
+      HandleTransportFeedback(rtcp_packet_header, now);
+      break;
+  }
+}
+
+void RtcpTransceiverImpl::HandleNack(
+    const rtcp::CommonHeader& rtcp_packet_header) {
+  rtcp::Nack nack;
+  if (local_senders_.empty() || !nack.Parse(rtcp_packet_header)) {
+    return;
+  }
+  auto it = local_senders_by_ssrc_.find(nack.media_ssrc());
+  if (it != local_senders_by_ssrc_.end()) {
+    it->second->handler->OnNack(nack.sender_ssrc(), nack.packet_ids());
+  }
+}
+
+void RtcpTransceiverImpl::HandleTransportFeedback(
+    const rtcp::CommonHeader& rtcp_packet_header,
+    Timestamp now) {
+  RTC_DCHECK_EQ(rtcp_packet_header.fmt(),
+                rtcp::TransportFeedback::kFeedbackMessageType);
+  if (config_.network_link_observer == nullptr) {
+    return;
+  }
+  rtcp::TransportFeedback feedback;
+  if (feedback.Parse(rtcp_packet_header)) {
+    config_.network_link_observer->OnTransportFeedback(now, feedback);
+  }
+}
+
+void RtcpTransceiverImpl::HandleExtendedReports(
+    const rtcp::CommonHeader& rtcp_packet_header,
+    Timestamp now) {
+  rtcp::ExtendedReports extended_reports;
+  if (!extended_reports.Parse(rtcp_packet_header))
+    return;
+
+  if (config_.reply_to_non_sender_rtt_measurement && extended_reports.rrtr()) {
+    RrtrTimes& rrtr = received_rrtrs_[extended_reports.sender_ssrc()];
+    rrtr.received_remote_mid_ntp_time =
+        CompactNtp(extended_reports.rrtr()->ntp());
+    rrtr.local_receive_mid_ntp_time =
+        CompactNtp(config_.clock->ConvertTimestampToNtpTime(now));
+  }
+
+  if (extended_reports.dlrr())
+    HandleDlrr(extended_reports.dlrr(), now);
+
+  if (extended_reports.target_bitrate())
+    HandleTargetBitrate(*extended_reports.target_bitrate(),
+                        extended_reports.sender_ssrc());
+}
+
+void RtcpTransceiverImpl::HandleDlrr(const rtcp::Dlrr& dlrr, Timestamp now) {
+  if (!config_.non_sender_rtt_measurement ||
+      config_.network_link_observer == nullptr) {
+    return;
+  }
+
+  // Delay and last_rr are transferred using 32bit compact ntp resolution.
+  // Convert packet arrival time to same format through 64bit ntp format.
+  uint32_t receive_time_ntp =
+      CompactNtp(config_.clock->ConvertTimestampToNtpTime(now));
+  for (const rtcp::ReceiveTimeInfo& rti : dlrr.sub_blocks()) {
+    if (rti.ssrc != config_.feedback_ssrc)
+      continue;
+    uint32_t rtt_ntp = receive_time_ntp - rti.delay_since_last_rr - rti.last_rr;
+    TimeDelta rtt = CompactNtpRttToTimeDelta(rtt_ntp);
+    config_.network_link_observer->OnRttUpdate(now, rtt);
+  }
+}
+
+void RtcpTransceiverImpl::ProcessReportBlocks(
+    Timestamp now,
+    rtc::ArrayView<const rtcp::ReportBlock> report_blocks) {
+  RTC_DCHECK(!report_blocks.empty());
+  if (config_.network_link_observer == nullptr) {
+    return;
+  }
+  // Round trip time calculated from different report blocks suppose to be about
+  // the same, as those blocks should be generated by the same remote sender.
+  // To avoid too many callbacks, this code accumulate multiple rtts into one.
+  TimeDelta rtt_sum = TimeDelta::Zero();
+  size_t num_rtts = 0;
+  uint32_t receive_time_ntp =
+      CompactNtp(config_.clock->ConvertTimestampToNtpTime(now));
+  for (const rtcp::ReportBlock& report_block : report_blocks) {
+    if (report_block.last_sr() == 0) {
+      continue;
+    }
+
+    uint32_t rtt_ntp = receive_time_ntp - report_block.delay_since_last_sr() -
+                       report_block.last_sr();
+    rtt_sum += CompactNtpRttToTimeDelta(rtt_ntp);
+    ++num_rtts;
+  }
+  // For backward compatibility, do not report rtt based on report blocks to the
+  // `config_.rtt_observer`
+  if (num_rtts > 0) {
+    config_.network_link_observer->OnRttUpdate(now, rtt_sum / num_rtts);
+  }
+  config_.network_link_observer->OnReportBlocks(now, report_blocks);
+}
+
+void RtcpTransceiverImpl::HandleTargetBitrate(
+    const rtcp::TargetBitrate& target_bitrate,
+    uint32_t remote_ssrc) {
+  auto remote_sender_it = remote_senders_.find(remote_ssrc);
+  if (remote_sender_it == remote_senders_.end() ||
+      remote_sender_it->second.observers.empty())
+    return;
+
+  // Convert rtcp::TargetBitrate to VideoBitrateAllocation.
+  VideoBitrateAllocation bitrate_allocation;
+  for (const rtcp::TargetBitrate::BitrateItem& item :
+       target_bitrate.GetTargetBitrates()) {
+    if (item.spatial_layer >= kMaxSpatialLayers ||
+        item.temporal_layer >= kMaxTemporalStreams) {
+      RTC_DLOG(LS_WARNING)
+          << config_.debug_id
+          << "Invalid incoming TargetBitrate with spatial layer "
+          << item.spatial_layer << ", temporal layer " << item.temporal_layer;
+      continue;
+    }
+    bitrate_allocation.SetBitrate(item.spatial_layer, item.temporal_layer,
+                                  item.target_bitrate_kbps * 1000);
+  }
+
+  for (MediaReceiverRtcpObserver* observer : remote_sender_it->second.observers)
+    observer->OnBitrateAllocation(remote_ssrc, bitrate_allocation);
+}
+
+void RtcpTransceiverImpl::ReschedulePeriodicCompoundPackets() {
+  if (!config_.schedule_periodic_compound_packets)
+    return;
+  periodic_task_handle_.Stop();
+  RTC_DCHECK(ready_to_send_);
+  SchedulePeriodicCompoundPackets(config_.report_period);
+}
+
+void RtcpTransceiverImpl::SchedulePeriodicCompoundPackets(TimeDelta delay) {
+  periodic_task_handle_ = RepeatingTaskHandle::DelayedStart(
+      config_.task_queue, delay,
+      [this] {
+        RTC_DCHECK(config_.schedule_periodic_compound_packets);
+        RTC_DCHECK(ready_to_send_);
+        SendPeriodicCompoundPacket();
+        return config_.report_period;
+      },
+      TaskQueueBase::DelayPrecision::kLow, config_.clock);
+}
+
+std::vector<uint32_t> RtcpTransceiverImpl::FillReports(
+    Timestamp now,
+    ReservedBytes reserved,
+    PacketSender& rtcp_sender) {
+  // Sender/receiver reports should be first in the RTCP packet.
+  RTC_DCHECK(rtcp_sender.IsEmpty());
+
+  size_t available_bytes = config_.max_packet_size;
+  if (reserved.per_packet > available_bytes) {
+    // Because reserved.per_packet is unsigned, substracting would underflow and
+    // will not produce desired result.
+    available_bytes = 0;
+  } else {
+    available_bytes -= reserved.per_packet;
+  }
+
+  const size_t sender_report_size_bytes = 28 + reserved.per_sender;
+  const size_t full_sender_report_size_bytes =
+      sender_report_size_bytes +
+      rtcp::SenderReport::kMaxNumberOfReportBlocks * rtcp::ReportBlock::kLength;
+  size_t max_full_sender_reports =
+      available_bytes / full_sender_report_size_bytes;
+  size_t max_report_blocks =
+      max_full_sender_reports * rtcp::SenderReport::kMaxNumberOfReportBlocks;
+  size_t available_bytes_for_last_sender_report =
+      available_bytes - max_full_sender_reports * full_sender_report_size_bytes;
+  if (available_bytes_for_last_sender_report >= sender_report_size_bytes) {
+    max_report_blocks +=
+        (available_bytes_for_last_sender_report - sender_report_size_bytes) /
+        rtcp::ReportBlock::kLength;
+  }
+
+  std::vector<rtcp::ReportBlock> report_blocks =
+      CreateReportBlocks(now, max_report_blocks);
+  // Previous calculation of max number of sender report made space for max
+  // number of report blocks per sender report, but if number of report blocks
+  // is low, more sender reports may fit in.
+  size_t max_sender_reports =
+      (available_bytes - report_blocks.size() * rtcp::ReportBlock::kLength) /
+      sender_report_size_bytes;
+
+  auto last_handled_sender_it = local_senders_.end();
+  auto report_block_it = report_blocks.begin();
+  std::vector<uint32_t> sender_ssrcs;
+  for (auto it = local_senders_.begin();
+       it != local_senders_.end() && sender_ssrcs.size() < max_sender_reports;
+       ++it) {
+    LocalSenderState& rtp_sender = *it;
+    RtpStreamRtcpHandler::RtpStats stats = rtp_sender.handler->SentStats();
+
+    if (stats.num_sent_bytes() < rtp_sender.last_num_sent_bytes) {
+      RTC_LOG(LS_ERROR) << "Inconsistent SR for SSRC " << rtp_sender.ssrc
+                        << ". Number of total sent bytes decreased.";
+      rtp_sender.last_num_sent_bytes = 0;
+    }
+    if (stats.num_sent_bytes() == rtp_sender.last_num_sent_bytes) {
+      // Skip because no RTP packet was send for this SSRC since last report.
+      continue;
+    }
+    rtp_sender.last_num_sent_bytes = stats.num_sent_bytes();
+
+    last_handled_sender_it = it;
+    rtcp::SenderReport sender_report;
+    sender_report.SetSenderSsrc(rtp_sender.ssrc);
+    sender_report.SetPacketCount(stats.num_sent_packets());
+    sender_report.SetOctetCount(stats.num_sent_bytes());
+    sender_report.SetNtp(config_.clock->ConvertTimestampToNtpTime(now));
+    RTC_DCHECK_GE(now, stats.last_capture_time());
+    sender_report.SetRtpTimestamp(
+        stats.last_rtp_timestamp() +
+        ((now - stats.last_capture_time()) * stats.last_clock_rate())
+            .seconds());
+    if (report_block_it != report_blocks.end()) {
+      size_t num_blocks =
+          std::min<size_t>(rtcp::SenderReport::kMaxNumberOfReportBlocks,
+                           report_blocks.end() - report_block_it);
+      std::vector<rtcp::ReportBlock> sub_blocks(report_block_it,
+                                                report_block_it + num_blocks);
+      sender_report.SetReportBlocks(std::move(sub_blocks));
+      report_block_it += num_blocks;
+    }
+    rtcp_sender.AppendPacket(sender_report);
+    sender_ssrcs.push_back(rtp_sender.ssrc);
+  }
+  if (last_handled_sender_it != local_senders_.end()) {
+    // Rotate `local_senders_` so that the 1st unhandled sender become first in
+    // the list, and thus will be first to generate rtcp sender report for on
+    // the next call to `FillReports`.
+    local_senders_.splice(local_senders_.end(), local_senders_,
+                          local_senders_.begin(),
+                          std::next(last_handled_sender_it));
+  }
+
+  // Calculcate number of receiver reports to attach remaining report blocks to.
+  size_t num_receiver_reports =
+      DivideRoundUp(report_blocks.end() - report_block_it,
+                    rtcp::ReceiverReport::kMaxNumberOfReportBlocks);
+
+  // In compound mode each RTCP packet has to start with a sender or receiver
+  // report.
+  if (config_.rtcp_mode == RtcpMode::kCompound && sender_ssrcs.empty() &&
+      num_receiver_reports == 0) {
+    num_receiver_reports = 1;
+  }
+
+  uint32_t sender_ssrc =
+      sender_ssrcs.empty() ? config_.feedback_ssrc : sender_ssrcs.front();
+  for (size_t i = 0; i < num_receiver_reports; ++i) {
+    rtcp::ReceiverReport receiver_report;
+    receiver_report.SetSenderSsrc(sender_ssrc);
+    size_t num_blocks =
+        std::min<size_t>(rtcp::ReceiverReport::kMaxNumberOfReportBlocks,
+                         report_blocks.end() - report_block_it);
+    std::vector<rtcp::ReportBlock> sub_blocks(report_block_it,
+                                              report_block_it + num_blocks);
+    receiver_report.SetReportBlocks(std::move(sub_blocks));
+    report_block_it += num_blocks;
+    rtcp_sender.AppendPacket(receiver_report);
+  }
+  // All report blocks should be attached at this point.
+  RTC_DCHECK_EQ(report_blocks.end() - report_block_it, 0);
+  return sender_ssrcs;
+}
+
+void RtcpTransceiverImpl::CreateCompoundPacket(Timestamp now,
+                                               size_t reserved_bytes,
+                                               PacketSender& sender) {
+  RTC_DCHECK(sender.IsEmpty());
+  ReservedBytes reserved = {.per_packet = reserved_bytes};
+  absl::optional<rtcp::Sdes> sdes;
+  if (!config_.cname.empty()) {
+    sdes.emplace();
+    bool added = sdes->AddCName(config_.feedback_ssrc, config_.cname);
+    RTC_DCHECK(added) << "Failed to add CNAME " << config_.cname
+                      << " to RTCP SDES packet.";
+    reserved.per_packet += sdes->BlockLength();
+  }
+  if (remb_.has_value()) {
+    reserved.per_packet += remb_->BlockLength();
+  }
+  absl::optional<rtcp::ExtendedReports> xr_with_dlrr;
+  if (!received_rrtrs_.empty()) {
+    RTC_DCHECK(config_.reply_to_non_sender_rtt_measurement);
+    xr_with_dlrr.emplace();
+    uint32_t now_ntp =
+        CompactNtp(config_.clock->ConvertTimestampToNtpTime(now));
+    for (const auto& [ssrc, rrtr_info] : received_rrtrs_) {
+      rtcp::ReceiveTimeInfo reply;
+      reply.ssrc = ssrc;
+      reply.last_rr = rrtr_info.received_remote_mid_ntp_time;
+      reply.delay_since_last_rr =
+          now_ntp - rrtr_info.local_receive_mid_ntp_time;
+      xr_with_dlrr->AddDlrrItem(reply);
+    }
+    if (config_.reply_to_non_sender_rtt_mesaurments_on_all_ssrcs) {
+      reserved.per_sender += xr_with_dlrr->BlockLength();
+    } else {
+      reserved.per_packet += xr_with_dlrr->BlockLength();
+    }
+  }
+  if (config_.non_sender_rtt_measurement) {
+    // It looks like bytes for ExtendedReport header are reserved twice, but in
+    // practice the same RtcpTransceiver won't both produce RRTR (i.e. it is a
+    // receiver-only) and reply to RRTR (i.e. remote participant is a receiver
+    // only). If that happen, then `reserved_bytes` would be slightly larger
+    // than it should, which is not an issue.
+
+    // 4 bytes for common RTCP header + 4 bytes for the ExtenedReports header.
+    reserved.per_packet += (4 + 4 + rtcp::Rrtr::kLength);
+  }
+
+  std::vector<uint32_t> sender_ssrcs = FillReports(now, reserved, sender);
+  bool has_sender_report = !sender_ssrcs.empty();
+  uint32_t sender_ssrc =
+      has_sender_report ? sender_ssrcs.front() : config_.feedback_ssrc;
+
+  if (sdes.has_value() && !sender.IsEmpty()) {
+    sender.AppendPacket(*sdes);
+  }
+  if (remb_.has_value()) {
+    remb_->SetSenderSsrc(sender_ssrc);
+    sender.AppendPacket(*remb_);
+  }
+  if (!has_sender_report && config_.non_sender_rtt_measurement) {
+    rtcp::ExtendedReports xr_with_rrtr;
+    xr_with_rrtr.SetSenderSsrc(config_.feedback_ssrc);
+    rtcp::Rrtr rrtr;
+    rrtr.SetNtp(config_.clock->ConvertTimestampToNtpTime(now));
+    xr_with_rrtr.SetRrtr(rrtr);
+    sender.AppendPacket(xr_with_rrtr);
+  }
+  if (xr_with_dlrr.has_value()) {
+    rtc::ArrayView<const uint32_t> ssrcs(&sender_ssrc, 1);
+    if (config_.reply_to_non_sender_rtt_mesaurments_on_all_ssrcs &&
+        !sender_ssrcs.empty()) {
+      ssrcs = sender_ssrcs;
+    }
+    RTC_DCHECK(!ssrcs.empty());
+    for (uint32_t ssrc : ssrcs) {
+      xr_with_dlrr->SetSenderSsrc(ssrc);
+      sender.AppendPacket(*xr_with_dlrr);
+    }
+  }
+}
+
+void RtcpTransceiverImpl::SendPeriodicCompoundPacket() {
+  auto send_packet = [this](rtc::ArrayView<const uint8_t> packet) {
+    config_.outgoing_transport->SendRtcp(packet.data(), packet.size());
+  };
+  Timestamp now = config_.clock->CurrentTime();
+  PacketSender sender(send_packet, config_.max_packet_size);
+  CreateCompoundPacket(now, /*reserved_bytes=*/0, sender);
+  sender.Send();
+}
+
+void RtcpTransceiverImpl::SendCombinedRtcpPacket(
+    std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
+  auto send_packet = [this](rtc::ArrayView<const uint8_t> packet) {
+    config_.outgoing_transport->SendRtcp(packet.data(), packet.size());
+  };
+  PacketSender sender(send_packet, config_.max_packet_size);
+
+  for (auto& rtcp_packet : rtcp_packets) {
+    rtcp_packet->SetSenderSsrc(config_.feedback_ssrc);
+    sender.AppendPacket(*rtcp_packet);
+  }
+  sender.Send();
+}
+
+void RtcpTransceiverImpl::SendImmediateFeedback(
+    const rtcp::RtcpPacket& rtcp_packet) {
+  auto send_packet = [this](rtc::ArrayView<const uint8_t> packet) {
+    config_.outgoing_transport->SendRtcp(packet.data(), packet.size());
+  };
+  PacketSender sender(send_packet, config_.max_packet_size);
+  // Compound mode requires every sent rtcp packet to be compound, i.e. start
+  // with a sender or receiver report.
+  if (config_.rtcp_mode == RtcpMode::kCompound) {
+    Timestamp now = config_.clock->CurrentTime();
+    CreateCompoundPacket(now, /*reserved_bytes=*/rtcp_packet.BlockLength(),
+                         sender);
+  }
+
+  sender.AppendPacket(rtcp_packet);
+  sender.Send();
+
+  // If compound packet was sent, delay (reschedule) the periodic one.
+  if (config_.rtcp_mode == RtcpMode::kCompound)
+    ReschedulePeriodicCompoundPackets();
+}
+
+std::vector<rtcp::ReportBlock> RtcpTransceiverImpl::CreateReportBlocks(
+    Timestamp now,
+    size_t num_max_blocks) {
+  if (!config_.receive_statistics)
+    return {};
+  std::vector<rtcp::ReportBlock> report_blocks =
+      config_.receive_statistics->RtcpReportBlocks(num_max_blocks);
+  uint32_t last_sr = 0;
+  uint32_t last_delay = 0;
+  for (rtcp::ReportBlock& report_block : report_blocks) {
+    auto it = remote_senders_.find(report_block.source_ssrc());
+    if (it == remote_senders_.end() ||
+        !it->second.last_received_sender_report) {
+      continue;
+    }
+    const SenderReportTimes& last_sender_report =
+        *it->second.last_received_sender_report;
+    last_sr = CompactNtp(last_sender_report.remote_sent_time);
+    last_delay =
+        SaturatedToCompactNtp(now - last_sender_report.local_received_time);
+    report_block.SetLastSr(last_sr);
+    report_block.SetDelayLastSr(last_delay);
+  }
+  return report_blocks;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
new file mode 100644
index 0000000000..8a3333d45c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
@@ -0,0 +1,170 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_IMPL_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_IMPL_H_
+
+#include <list>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
+#include "modules/rtp_rtcp/source/rtcp_transceiver_config.h"
+#include "rtc_base/containers/flat_map.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+//
+// Manage incoming and outgoing rtcp messages for multiple BUNDLED streams.
+//
+// This class is not thread-safe.
+class RtcpTransceiverImpl {
+ public:
+  explicit RtcpTransceiverImpl(const RtcpTransceiverConfig& config);
+  RtcpTransceiverImpl(const RtcpTransceiverImpl&) = delete;
+  RtcpTransceiverImpl& operator=(const RtcpTransceiverImpl&) = delete;
+  ~RtcpTransceiverImpl();
+
+  void StopPeriodicTask() { periodic_task_handle_.Stop(); }
+
+  void AddMediaReceiverRtcpObserver(uint32_t remote_ssrc,
+                                    MediaReceiverRtcpObserver* observer);
+  void RemoveMediaReceiverRtcpObserver(uint32_t remote_ssrc,
+                                       MediaReceiverRtcpObserver* observer);
+
+  // Returns false on failure, e.g. when there is already an handler for the
+  // `local_ssrc`.
+  bool AddMediaSender(uint32_t local_ssrc, RtpStreamRtcpHandler* handler);
+  bool RemoveMediaSender(uint32_t local_ssrc);
+
+  void SetReadyToSend(bool ready);
+
+  void ReceivePacket(rtc::ArrayView<const uint8_t> packet, Timestamp now);
+
+  void SendCompoundPacket();
+
+  void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs);
+  void UnsetRemb();
+  // Temporary helpers to send pre-built TransportFeedback rtcp packet.
+  uint32_t sender_ssrc() const { return config_.feedback_ssrc; }
+  void SendRawPacket(rtc::ArrayView<const uint8_t> packet);
+
+  void SendNack(uint32_t ssrc, std::vector<uint16_t> sequence_numbers);
+
+  void SendPictureLossIndication(uint32_t ssrc);
+  // If new_request is true then requested sequence no. will increase for each
+  // requested ssrc.
+  void SendFullIntraRequest(rtc::ArrayView<const uint32_t> ssrcs,
+                            bool new_request);
+
+  // SendCombinedRtcpPacket ignores rtcp mode and does not send a compound
+  // message. https://tools.ietf.org/html/rfc4585#section-3.1
+  void SendCombinedRtcpPacket(
+      std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets);
+
+ private:
+  class PacketSender;
+  struct RemoteSenderState;
+  struct LocalSenderState;
+  struct RrtrTimes {
+    // Received remote NTP timestamp in compact representation.
+    uint32_t received_remote_mid_ntp_time;
+
+    // Local NTP time when the report was received in compact representation.
+    uint32_t local_receive_mid_ntp_time;
+  };
+
+  void HandleReceivedPacket(const rtcp::CommonHeader& rtcp_packet_header,
+                            Timestamp now,
+                            std::vector<rtcp::ReportBlock>& report_blocks);
+  // Individual rtcp packet handlers.
+  void HandleBye(const rtcp::CommonHeader& rtcp_packet_header);
+  void HandleSenderReport(const rtcp::CommonHeader& rtcp_packet_header,
+                          Timestamp now,
+                          std::vector<rtcp::ReportBlock>& report_blocks);
+  void HandleReceiverReport(const rtcp::CommonHeader& rtcp_packet_header,
+                            std::vector<rtcp::ReportBlock>& report_blocks);
+  void CallbackOnReportBlocks(
+      uint32_t sender_ssrc,
+      rtc::ArrayView<const rtcp::ReportBlock> report_blocks);
+  void HandlePayloadSpecificFeedback(
+      const rtcp::CommonHeader& rtcp_packet_header,
+      Timestamp now);
+  void HandleRtpFeedback(const rtcp::CommonHeader& rtcp_packet_header,
+                         Timestamp now);
+  void HandleFir(const rtcp::CommonHeader& rtcp_packet_header);
+  void HandlePli(const rtcp::CommonHeader& rtcp_packet_header);
+  void HandleRemb(const rtcp::CommonHeader& rtcp_packet_header, Timestamp now);
+  void HandleNack(const rtcp::CommonHeader& rtcp_packet_header);
+  void HandleTransportFeedback(const rtcp::CommonHeader& rtcp_packet_header,
+                               Timestamp now);
+  void HandleExtendedReports(const rtcp::CommonHeader& rtcp_packet_header,
+                             Timestamp now);
+  // Extended Reports blocks handlers.
+  void HandleDlrr(const rtcp::Dlrr& dlrr, Timestamp now);
+  void HandleTargetBitrate(const rtcp::TargetBitrate& target_bitrate,
+                           uint32_t remote_ssrc);
+  void ProcessReportBlocks(
+      Timestamp now,
+      rtc::ArrayView<const rtcp::ReportBlock> report_blocks);
+
+  void ReschedulePeriodicCompoundPackets();
+  void SchedulePeriodicCompoundPackets(TimeDelta delay);
+  // Appends RTCP sender and receiver reports to the `sender`.
+  // Both sender and receiver reports may have attached report blocks.
+  // Uses up to `config_.max_packet_size - reserved_bytes.per_packet`
+  // Returns list of sender ssrc in sender reports.
+  struct ReservedBytes {
+    size_t per_packet = 0;
+    size_t per_sender = 0;
+  };
+  std::vector<uint32_t> FillReports(Timestamp now,
+                                    ReservedBytes reserved_bytes,
+                                    PacketSender& rtcp_sender);
+
+  // Creates compound RTCP packet, as defined in
+  // https://tools.ietf.org/html/rfc5506#section-2
+  void CreateCompoundPacket(Timestamp now,
+                            size_t reserved_bytes,
+                            PacketSender& rtcp_sender);
+
+  // Sends RTCP packets.
+  void SendPeriodicCompoundPacket();
+  void SendImmediateFeedback(const rtcp::RtcpPacket& rtcp_packet);
+  // Generate Report Blocks to be send in Sender or Receiver Reports.
+  std::vector<rtcp::ReportBlock> CreateReportBlocks(Timestamp now,
+                                                    size_t num_max_blocks);
+
+  const RtcpTransceiverConfig config_;
+
+  bool ready_to_send_;
+  absl::optional<rtcp::Remb> remb_;
+  // TODO(danilchap): Remove entries from remote_senders_ that are no longer
+  // needed.
+  flat_map<uint32_t, RemoteSenderState> remote_senders_;
+  std::list<LocalSenderState> local_senders_;
+  flat_map<uint32_t, std::list<LocalSenderState>::iterator>
+      local_senders_by_ssrc_;
+  flat_map<uint32_t, RrtrTimes> received_rrtrs_;
+  RepeatingTaskHandle periodic_task_handle_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_IMPL_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
new file mode 100644
index 0000000000..ad9561f66b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
@@ -0,0 +1,1741 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_transceiver_impl.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "api/rtp_headers.h"
+#include "api/test/create_time_controller.h"
+#include "api/test/time_controller.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
+#include "test/rtcp_packet_parser.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::Ge;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::SizeIs;
+using ::testing::StrictMock;
+using ::testing::UnorderedElementsAre;
+using ::testing::WithArg;
+using ::webrtc::rtcp::Bye;
+using ::webrtc::rtcp::CompoundPacket;
+using ::webrtc::rtcp::ReportBlock;
+using ::webrtc::rtcp::SenderReport;
+using ::webrtc::test::RtcpPacketParser;
+
+class MockReceiveStatisticsProvider : public ReceiveStatisticsProvider {
+ public:
+  MOCK_METHOD(std::vector<ReportBlock>, RtcpReportBlocks, (size_t), (override));
+};
+
+class MockMediaReceiverRtcpObserver : public MediaReceiverRtcpObserver {
+ public:
+  MOCK_METHOD(void, OnSenderReport, (uint32_t, NtpTime, uint32_t), (override));
+  MOCK_METHOD(void, OnBye, (uint32_t), (override));
+  MOCK_METHOD(void,
+              OnBitrateAllocation,
+              (uint32_t, const VideoBitrateAllocation&),
+              (override));
+};
+
+class MockRtpStreamRtcpHandler : public RtpStreamRtcpHandler {
+ public:
+  MockRtpStreamRtcpHandler() {
+    // With each next call increase number of sent packets and bytes to simulate
+    // active RTP sender.
+    ON_CALL(*this, SentStats).WillByDefault([this] {
+      RtpStats stats;
+      stats.set_num_sent_packets(++num_calls_);
+      stats.set_num_sent_bytes(1'000 * num_calls_);
+      return stats;
+    });
+  }
+
+  MOCK_METHOD(RtpStats, SentStats, (), (override));
+  MOCK_METHOD(void,
+              OnNack,
+              (uint32_t, rtc::ArrayView<const uint16_t>),
+              (override));
+  MOCK_METHOD(void, OnFir, (uint32_t), (override));
+  MOCK_METHOD(void, OnPli, (uint32_t), (override));
+  MOCK_METHOD(void,
+              OnReportBlock,
+              (uint32_t, const rtcp::ReportBlock&),
+              (override));
+
+ private:
+  int num_calls_ = 0;
+};
+
+class MockNetworkLinkRtcpObserver : public NetworkLinkRtcpObserver {
+ public:
+  MOCK_METHOD(void,
+              OnRttUpdate,
+              (Timestamp receive_time, TimeDelta rtt),
+              (override));
+  MOCK_METHOD(void,
+              OnTransportFeedback,
+              (Timestamp receive_time, const rtcp::TransportFeedback& feedback),
+              (override));
+  MOCK_METHOD(void,
+              OnReceiverEstimatedMaxBitrate,
+              (Timestamp receive_time, DataRate bitrate),
+              (override));
+  MOCK_METHOD(void,
+              OnReportBlocks,
+              (Timestamp receive_time,
+               rtc::ArrayView<const rtcp::ReportBlock> report_blocks),
+              (override));
+};
+
+constexpr TimeDelta kReportPeriod = TimeDelta::Seconds(1);
+constexpr TimeDelta kAlmostForever = TimeDelta::Seconds(2);
+constexpr TimeDelta kTimePrecision = TimeDelta::Millis(1);
+
+MATCHER_P(Near, value, "") {
+  return arg > value - kTimePrecision && arg < value + kTimePrecision;
+}
+
+// Helper to wait for an rtcp packet produced on a different thread/task queue.
+class FakeRtcpTransport : public webrtc::Transport {
+ public:
+  explicit FakeRtcpTransport(TimeController& time) : time_(time) {}
+
+  bool SendRtcp(const uint8_t* data, size_t size) override {
+    sent_rtcp_ = true;
+    return true;
+  }
+  bool SendRtp(const uint8_t*, size_t, const webrtc::PacketOptions&) override {
+    ADD_FAILURE() << "RtcpTransciver shouldn't send rtp packets.";
+    return true;
+  }
+
+  // Returns true when packet was received by the transport.
+  bool WaitPacket() {
+    bool got_packet = time_.Wait([this] { return sent_rtcp_; }, kAlmostForever);
+    // Clear the 'event' to allow waiting for multiple packets.
+    sent_rtcp_ = false;
+    return got_packet;
+  }
+
+ private:
+  TimeController& time_;
+  bool sent_rtcp_ = false;
+};
+
+class RtcpParserTransport : public webrtc::Transport {
+ public:
+  explicit RtcpParserTransport(RtcpPacketParser* parser) : parser_(parser) {}
+  // Returns total number of rtcp packet received.
+  int num_packets() const { return num_packets_; }
+
+ private:
+  bool SendRtcp(const uint8_t* data, size_t size) override {
+    ++num_packets_;
+    parser_->Parse(data, size);
+    return true;
+  }
+
+  bool SendRtp(const uint8_t*, size_t, const webrtc::PacketOptions&) override {
+    ADD_FAILURE() << "RtcpTransciver shouldn't send rtp packets.";
+    return true;
+  }
+
+  RtcpPacketParser* const parser_;
+  int num_packets_ = 0;
+};
+
+class RtcpTransceiverImplTest : public ::testing::Test {
+ public:
+  RtcpTransceiverConfig DefaultTestConfig() {
+    // RtcpTransceiverConfig default constructor sets default values for prod.
+    // Test doesn't need to support all key features: Default test config
+    // returns valid config with all features turned off.
+    RtcpTransceiverConfig config;
+    config.clock = time_->GetClock();
+    config.outgoing_transport = &null_transport_;
+    config.schedule_periodic_compound_packets = false;
+    config.initial_report_delay = kReportPeriod / 2;
+    config.report_period = kReportPeriod;
+    return config;
+  }
+
+  TimeController& time_controller() { return *time_; }
+  Timestamp CurrentTime() { return time_->GetClock()->CurrentTime(); }
+  void AdvanceTime(TimeDelta time) { time_->AdvanceTime(time); }
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue() {
+    return time_->GetTaskQueueFactory()->CreateTaskQueue(
+        "rtcp", TaskQueueFactory::Priority::NORMAL);
+  }
+
+ private:
+  MockTransport null_transport_;
+  std::unique_ptr<TimeController> time_ = CreateSimulatedTimeController();
+};
+
+TEST_F(RtcpTransceiverImplTest, NeedToStopPeriodicTaskToDestroyOnTaskQueue) {
+  FakeRtcpTransport transport(time_controller());
+  auto queue = CreateTaskQueue();
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.task_queue = queue.get();
+  config.schedule_periodic_compound_packets = true;
+  config.outgoing_transport = &transport;
+  auto* rtcp_transceiver = new RtcpTransceiverImpl(config);
+  // Wait for a periodic packet.
+  EXPECT_TRUE(transport.WaitPacket());
+
+  bool done = false;
+  queue->PostTask([rtcp_transceiver, &done] {
+    rtcp_transceiver->StopPeriodicTask();
+    delete rtcp_transceiver;
+    done = true;
+  });
+  ASSERT_TRUE(time_controller().Wait([&] { return done; }, kAlmostForever));
+}
+
+TEST_F(RtcpTransceiverImplTest, CanBeDestroyedRightAfterCreation) {
+  FakeRtcpTransport transport(time_controller());
+  auto queue = CreateTaskQueue();
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.task_queue = queue.get();
+  config.schedule_periodic_compound_packets = true;
+  config.outgoing_transport = &transport;
+
+  bool done = false;
+  queue->PostTask([&] {
+    RtcpTransceiverImpl rtcp_transceiver(config);
+    rtcp_transceiver.StopPeriodicTask();
+    done = true;
+  });
+  ASSERT_TRUE(time_controller().Wait([&] { return done; }, kAlmostForever));
+}
+
+TEST_F(RtcpTransceiverImplTest, CanDestroyAfterTaskQueue) {
+  FakeRtcpTransport transport(time_controller());
+  auto queue = CreateTaskQueue();
+
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.task_queue = queue.get();
+  config.schedule_periodic_compound_packets = true;
+  config.outgoing_transport = &transport;
+  auto* rtcp_transceiver = new RtcpTransceiverImpl(config);
+  // Wait for a periodic packet.
+  EXPECT_TRUE(transport.WaitPacket());
+
+  queue = nullptr;
+  delete rtcp_transceiver;
+}
+
+TEST_F(RtcpTransceiverImplTest, DelaysSendingFirstCompondPacket) {
+  auto queue = CreateTaskQueue();
+  FakeRtcpTransport transport(time_controller());
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.schedule_periodic_compound_packets = true;
+  config.outgoing_transport = &transport;
+  config.initial_report_delay = TimeDelta::Millis(10);
+  config.task_queue = queue.get();
+  absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
+
+  Timestamp started = CurrentTime();
+  queue->PostTask([&] { rtcp_transceiver.emplace(config); });
+  EXPECT_TRUE(transport.WaitPacket());
+
+  EXPECT_GE(CurrentTime() - started, config.initial_report_delay);
+
+  // Cleanup.
+  bool done = false;
+  queue->PostTask([&] {
+    rtcp_transceiver->StopPeriodicTask();
+    rtcp_transceiver.reset();
+    done = true;
+  });
+  ASSERT_TRUE(time_controller().Wait([&] { return done; }, kAlmostForever));
+}
+
+TEST_F(RtcpTransceiverImplTest, PeriodicallySendsPackets) {
+  auto queue = CreateTaskQueue();
+  FakeRtcpTransport transport(time_controller());
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.schedule_periodic_compound_packets = true;
+  config.outgoing_transport = &transport;
+  config.initial_report_delay = TimeDelta::Zero();
+  config.report_period = kReportPeriod;
+  config.task_queue = queue.get();
+  absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
+  Timestamp time_just_before_1st_packet = Timestamp::MinusInfinity();
+  queue->PostTask([&] {
+    // Because initial_report_delay_ms is set to 0, time_just_before_the_packet
+    // should be very close to the time_of_the_packet.
+    time_just_before_1st_packet = CurrentTime();
+    rtcp_transceiver.emplace(config);
+  });
+
+  EXPECT_TRUE(transport.WaitPacket());
+  EXPECT_TRUE(transport.WaitPacket());
+  Timestamp time_just_after_2nd_packet = CurrentTime();
+
+  EXPECT_GE(time_just_after_2nd_packet - time_just_before_1st_packet,
+            config.report_period);
+
+  // Cleanup.
+  bool done = false;
+  queue->PostTask([&] {
+    rtcp_transceiver->StopPeriodicTask();
+    rtcp_transceiver.reset();
+    done = true;
+  });
+  ASSERT_TRUE(time_controller().Wait([&] { return done; }, kAlmostForever));
+}
+
+TEST_F(RtcpTransceiverImplTest, SendCompoundPacketDelaysPeriodicSendPackets) {
+  auto queue = CreateTaskQueue();
+  FakeRtcpTransport transport(time_controller());
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.schedule_periodic_compound_packets = true;
+  config.outgoing_transport = &transport;
+  config.initial_report_delay = TimeDelta::Zero();
+  config.report_period = kReportPeriod;
+  config.task_queue = queue.get();
+  absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
+  queue->PostTask([&] { rtcp_transceiver.emplace(config); });
+
+  // Wait for the first packet.
+  EXPECT_TRUE(transport.WaitPacket());
+  // Send non periodic one after half period.
+  bool non_periodic = false;
+  Timestamp time_of_non_periodic_packet = Timestamp::MinusInfinity();
+  queue->PostDelayedTask(
+      [&] {
+        time_of_non_periodic_packet = CurrentTime();
+        rtcp_transceiver->SendCompoundPacket();
+        non_periodic = true;
+      },
+      config.report_period / 2);
+  // Though non-periodic packet is scheduled just in between periodic, due to
+  // small period and task queue flakiness it migth end-up 1ms after next
+  // periodic packet. To be sure duration after non-periodic packet is tested
+  // wait for transport after ensuring non-periodic packet was sent.
+  EXPECT_TRUE(
+      time_controller().Wait([&] { return non_periodic; }, kAlmostForever));
+  EXPECT_TRUE(transport.WaitPacket());
+  // Wait for next periodic packet.
+  EXPECT_TRUE(transport.WaitPacket());
+  Timestamp time_of_last_periodic_packet = CurrentTime();
+  EXPECT_GE(time_of_last_periodic_packet - time_of_non_periodic_packet,
+            config.report_period);
+
+  // Cleanup.
+  bool done = false;
+  queue->PostTask([&] {
+    rtcp_transceiver->StopPeriodicTask();
+    rtcp_transceiver.reset();
+    done = true;
+  });
+  ASSERT_TRUE(time_controller().Wait([&] { return done; }, kAlmostForever));
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsNoRtcpWhenNetworkStateIsDown) {
+  MockTransport mock_transport;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.initial_ready_to_send = false;
+  config.outgoing_transport = &mock_transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  EXPECT_CALL(mock_transport, SendRtcp(_, _)).Times(0);
+
+  const uint8_t raw[] = {1, 2, 3, 4};
+  const std::vector<uint16_t> sequence_numbers = {45, 57};
+  const uint32_t ssrcs[] = {123};
+  rtcp_transceiver.SendCompoundPacket();
+  rtcp_transceiver.SendRawPacket(raw);
+  rtcp_transceiver.SendNack(ssrcs[0], sequence_numbers);
+  rtcp_transceiver.SendPictureLossIndication(ssrcs[0]);
+  rtcp_transceiver.SendFullIntraRequest(ssrcs, true);
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsRtcpWhenNetworkStateIsUp) {
+  MockTransport mock_transport;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.initial_ready_to_send = false;
+  config.outgoing_transport = &mock_transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SetReadyToSend(true);
+
+  EXPECT_CALL(mock_transport, SendRtcp(_, _)).Times(5);
+
+  const uint8_t raw[] = {1, 2, 3, 4};
+  const std::vector<uint16_t> sequence_numbers = {45, 57};
+  const uint32_t ssrcs[] = {123};
+  rtcp_transceiver.SendCompoundPacket();
+  rtcp_transceiver.SendRawPacket(raw);
+  rtcp_transceiver.SendNack(ssrcs[0], sequence_numbers);
+  rtcp_transceiver.SendPictureLossIndication(ssrcs[0]);
+  rtcp_transceiver.SendFullIntraRequest(ssrcs, true);
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsPeriodicRtcpWhenNetworkStateIsUp) {
+  auto queue = CreateTaskQueue();
+  FakeRtcpTransport transport(time_controller());
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.schedule_periodic_compound_packets = true;
+  config.initial_ready_to_send = false;
+  config.outgoing_transport = &transport;
+  config.task_queue = queue.get();
+  absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
+  rtcp_transceiver.emplace(config);
+
+  queue->PostTask([&] { rtcp_transceiver->SetReadyToSend(true); });
+
+  EXPECT_TRUE(transport.WaitPacket());
+
+  // Cleanup.
+  bool done = false;
+  queue->PostTask([&] {
+    rtcp_transceiver->StopPeriodicTask();
+    rtcp_transceiver.reset();
+    done = true;
+  });
+  ASSERT_TRUE(time_controller().Wait([&] { return done; }, kAlmostForever));
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsMinimalCompoundPacket) {
+  const uint32_t kSenderSsrc = 12345;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  config.cname = "cname";
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SendCompoundPacket();
+
+  // Minimal compound RTCP packet contains sender or receiver report and sdes
+  // with cname.
+  ASSERT_GT(rtcp_parser.receiver_report()->num_packets(), 0);
+  EXPECT_EQ(rtcp_parser.receiver_report()->sender_ssrc(), kSenderSsrc);
+  ASSERT_GT(rtcp_parser.sdes()->num_packets(), 0);
+  ASSERT_EQ(rtcp_parser.sdes()->chunks().size(), 1u);
+  EXPECT_EQ(rtcp_parser.sdes()->chunks()[0].ssrc, kSenderSsrc);
+  EXPECT_EQ(rtcp_parser.sdes()->chunks()[0].cname, config.cname);
+}
+
+TEST_F(RtcpTransceiverImplTest, AvoidsEmptyPacketsInReducedMode) {
+  MockTransport transport;
+  EXPECT_CALL(transport, SendRtcp).Times(0);
+  NiceMock<MockReceiveStatisticsProvider> receive_statistics;
+
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.outgoing_transport = &transport;
+  config.rtcp_mode = webrtc::RtcpMode::kReducedSize;
+  config.receive_statistics = &receive_statistics;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SendCompoundPacket();
+}
+
+TEST_F(RtcpTransceiverImplTest, AvoidsEmptyReceiverReportsInReducedMode) {
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  NiceMock<MockReceiveStatisticsProvider> receive_statistics;
+
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.outgoing_transport = &transport;
+  config.rtcp_mode = webrtc::RtcpMode::kReducedSize;
+  config.receive_statistics = &receive_statistics;
+  // Set it to produce something (RRTR) in the "periodic" rtcp packets.
+  config.non_sender_rtt_measurement = true;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  // Rather than waiting for the right time to produce the periodic packet,
+  // trigger it manually.
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(rtcp_parser.receiver_report()->num_packets(), 0);
+  EXPECT_GT(rtcp_parser.xr()->num_packets(), 0);
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsNoRembInitially) {
+  const uint32_t kSenderSsrc = 12345;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(transport.num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 0);
+}
+
+TEST_F(RtcpTransceiverImplTest, SetRembIncludesRembInNextCompoundPacket) {
+  const uint32_t kSenderSsrc = 12345;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SetRemb(/*bitrate_bps=*/10000, /*ssrcs=*/{54321, 64321});
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.remb()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(rtcp_parser.remb()->bitrate_bps(), 10000);
+  EXPECT_THAT(rtcp_parser.remb()->ssrcs(), ElementsAre(54321, 64321));
+}
+
+TEST_F(RtcpTransceiverImplTest, SetRembUpdatesValuesToSend) {
+  const uint32_t kSenderSsrc = 12345;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SetRemb(/*bitrate_bps=*/10000, /*ssrcs=*/{54321, 64321});
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.remb()->bitrate_bps(), 10000);
+  EXPECT_THAT(rtcp_parser.remb()->ssrcs(), ElementsAre(54321, 64321));
+
+  rtcp_transceiver.SetRemb(/*bitrate_bps=*/70000, /*ssrcs=*/{67321});
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 2);
+  EXPECT_EQ(rtcp_parser.remb()->bitrate_bps(), 70000);
+  EXPECT_THAT(rtcp_parser.remb()->ssrcs(), ElementsAre(67321));
+}
+
+TEST_F(RtcpTransceiverImplTest, SetRembSendsImmediatelyIfSendRembOnChange) {
+  const uint32_t kSenderSsrc = 12345;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.send_remb_on_change = true;
+  config.feedback_ssrc = kSenderSsrc;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SetRemb(/*bitrate_bps=*/10000, /*ssrcs=*/{});
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.remb()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(rtcp_parser.remb()->bitrate_bps(), 10000);
+
+  // If there is no change, the packet is not sent immediately.
+  rtcp_transceiver.SetRemb(/*bitrate_bps=*/10000, /*ssrcs=*/{});
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 1);
+
+  rtcp_transceiver.SetRemb(/*bitrate_bps=*/20000, /*ssrcs=*/{});
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 2);
+  EXPECT_EQ(rtcp_parser.remb()->bitrate_bps(), 20000);
+}
+
+TEST_F(RtcpTransceiverImplTest,
+       SetRembSendsImmediatelyIfSendRembOnChangeReducedSize) {
+  const uint32_t kSenderSsrc = 12345;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.send_remb_on_change = true;
+  config.rtcp_mode = webrtc::RtcpMode::kReducedSize;
+  config.feedback_ssrc = kSenderSsrc;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SetRemb(/*bitrate_bps=*/10000, /*ssrcs=*/{});
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.remb()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(rtcp_parser.remb()->bitrate_bps(), 10000);
+}
+
+TEST_F(RtcpTransceiverImplTest, SetRembIncludesRembInAllCompoundPackets) {
+  const uint32_t kSenderSsrc = 12345;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SetRemb(/*bitrate_bps=*/10000, /*ssrcs=*/{54321, 64321});
+  rtcp_transceiver.SendCompoundPacket();
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(transport.num_packets(), 2);
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 2);
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsNoRembAfterUnset) {
+  const uint32_t kSenderSsrc = 12345;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SetRemb(/*bitrate_bps=*/10000, /*ssrcs=*/{54321, 64321});
+  rtcp_transceiver.SendCompoundPacket();
+  EXPECT_EQ(transport.num_packets(), 1);
+  ASSERT_EQ(rtcp_parser.remb()->num_packets(), 1);
+
+  rtcp_transceiver.UnsetRemb();
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(transport.num_packets(), 2);
+  EXPECT_EQ(rtcp_parser.remb()->num_packets(), 1);
+}
+
+TEST_F(RtcpTransceiverImplTest, ReceiverReportUsesReceiveStatistics) {
+  const uint32_t kSenderSsrc = 12345;
+  const uint32_t kMediaSsrc = 54321;
+  MockReceiveStatisticsProvider receive_statistics;
+  std::vector<ReportBlock> report_blocks(1);
+  report_blocks[0].SetMediaSsrc(kMediaSsrc);
+  EXPECT_CALL(receive_statistics, RtcpReportBlocks(_))
+      .WillRepeatedly(Return(report_blocks));
+
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.receive_statistics = &receive_statistics;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SendCompoundPacket();
+
+  ASSERT_GT(rtcp_parser.receiver_report()->num_packets(), 0);
+  EXPECT_EQ(rtcp_parser.receiver_report()->sender_ssrc(), kSenderSsrc);
+  ASSERT_THAT(rtcp_parser.receiver_report()->report_blocks(),
+              SizeIs(report_blocks.size()));
+  EXPECT_EQ(rtcp_parser.receiver_report()->report_blocks()[0].source_ssrc(),
+            kMediaSsrc);
+}
+
+TEST_F(RtcpTransceiverImplTest, MultipleObserversOnSameSsrc) {
+  const uint32_t kRemoteSsrc = 12345;
+  StrictMock<MockMediaReceiverRtcpObserver> observer1;
+  StrictMock<MockMediaReceiverRtcpObserver> observer2;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, &observer1);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, &observer2);
+
+  const NtpTime kRemoteNtp(0x9876543211);
+  const uint32_t kRemoteRtp = 0x444555;
+  SenderReport sr;
+  sr.SetSenderSsrc(kRemoteSsrc);
+  sr.SetNtp(kRemoteNtp);
+  sr.SetRtpTimestamp(kRemoteRtp);
+  auto raw_packet = sr.Build();
+
+  EXPECT_CALL(observer1, OnSenderReport(kRemoteSsrc, kRemoteNtp, kRemoteRtp));
+  EXPECT_CALL(observer2, OnSenderReport(kRemoteSsrc, kRemoteNtp, kRemoteRtp));
+  rtcp_transceiver.ReceivePacket(raw_packet, Timestamp::Micros(0));
+}
+
+TEST_F(RtcpTransceiverImplTest, DoesntCallsObserverAfterRemoved) {
+  const uint32_t kRemoteSsrc = 12345;
+  StrictMock<MockMediaReceiverRtcpObserver> observer1;
+  StrictMock<MockMediaReceiverRtcpObserver> observer2;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, &observer1);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, &observer2);
+
+  SenderReport sr;
+  sr.SetSenderSsrc(kRemoteSsrc);
+  auto raw_packet = sr.Build();
+
+  rtcp_transceiver.RemoveMediaReceiverRtcpObserver(kRemoteSsrc, &observer1);
+
+  EXPECT_CALL(observer1, OnSenderReport(_, _, _)).Times(0);
+  EXPECT_CALL(observer2, OnSenderReport(_, _, _));
+  rtcp_transceiver.ReceivePacket(raw_packet, Timestamp::Micros(0));
+}
+
+TEST_F(RtcpTransceiverImplTest, CallsObserverOnSenderReportBySenderSsrc) {
+  const uint32_t kRemoteSsrc1 = 12345;
+  const uint32_t kRemoteSsrc2 = 22345;
+  StrictMock<MockMediaReceiverRtcpObserver> observer1;
+  StrictMock<MockMediaReceiverRtcpObserver> observer2;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc1, &observer1);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc2, &observer2);
+
+  const NtpTime kRemoteNtp(0x9876543211);
+  const uint32_t kRemoteRtp = 0x444555;
+  SenderReport sr;
+  sr.SetSenderSsrc(kRemoteSsrc1);
+  sr.SetNtp(kRemoteNtp);
+  sr.SetRtpTimestamp(kRemoteRtp);
+  auto raw_packet = sr.Build();
+
+  EXPECT_CALL(observer1, OnSenderReport(kRemoteSsrc1, kRemoteNtp, kRemoteRtp));
+  EXPECT_CALL(observer2, OnSenderReport).Times(0);
+  rtcp_transceiver.ReceivePacket(raw_packet, Timestamp::Micros(0));
+}
+
+TEST_F(RtcpTransceiverImplTest, CallsObserverOnByeBySenderSsrc) {
+  const uint32_t kRemoteSsrc1 = 12345;
+  const uint32_t kRemoteSsrc2 = 22345;
+  StrictMock<MockMediaReceiverRtcpObserver> observer1;
+  StrictMock<MockMediaReceiverRtcpObserver> observer2;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc1, &observer1);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc2, &observer2);
+
+  Bye bye;
+  bye.SetSenderSsrc(kRemoteSsrc1);
+  auto raw_packet = bye.Build();
+
+  EXPECT_CALL(observer1, OnBye(kRemoteSsrc1));
+  EXPECT_CALL(observer2, OnBye(_)).Times(0);
+  rtcp_transceiver.ReceivePacket(raw_packet, Timestamp::Micros(0));
+}
+
+TEST_F(RtcpTransceiverImplTest, CallsObserverOnTargetBitrateBySenderSsrc) {
+  const uint32_t kRemoteSsrc1 = 12345;
+  const uint32_t kRemoteSsrc2 = 22345;
+  StrictMock<MockMediaReceiverRtcpObserver> observer1;
+  StrictMock<MockMediaReceiverRtcpObserver> observer2;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc1, &observer1);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc2, &observer2);
+
+  webrtc::rtcp::TargetBitrate target_bitrate;
+  target_bitrate.AddTargetBitrate(0, 0, /*target_bitrate_kbps=*/10);
+  target_bitrate.AddTargetBitrate(0, 1, /*target_bitrate_kbps=*/20);
+  target_bitrate.AddTargetBitrate(1, 0, /*target_bitrate_kbps=*/40);
+  target_bitrate.AddTargetBitrate(1, 1, /*target_bitrate_kbps=*/80);
+  webrtc::rtcp::ExtendedReports xr;
+  xr.SetSenderSsrc(kRemoteSsrc1);
+  xr.SetTargetBitrate(target_bitrate);
+  auto raw_packet = xr.Build();
+
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(0, 0, /*bitrate_bps=*/10000);
+  bitrate_allocation.SetBitrate(0, 1, /*bitrate_bps=*/20000);
+  bitrate_allocation.SetBitrate(1, 0, /*bitrate_bps=*/40000);
+  bitrate_allocation.SetBitrate(1, 1, /*bitrate_bps=*/80000);
+  EXPECT_CALL(observer1, OnBitrateAllocation(kRemoteSsrc1, bitrate_allocation));
+  EXPECT_CALL(observer2, OnBitrateAllocation(_, _)).Times(0);
+  rtcp_transceiver.ReceivePacket(raw_packet, Timestamp::Micros(0));
+}
+
+TEST_F(RtcpTransceiverImplTest, SkipsIncorrectTargetBitrateEntries) {
+  const uint32_t kRemoteSsrc = 12345;
+  MockMediaReceiverRtcpObserver observer;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, &observer);
+
+  webrtc::rtcp::TargetBitrate target_bitrate;
+  target_bitrate.AddTargetBitrate(0, 0, /*target_bitrate_kbps=*/10);
+  target_bitrate.AddTargetBitrate(0, webrtc::kMaxTemporalStreams, 20);
+  target_bitrate.AddTargetBitrate(webrtc::kMaxSpatialLayers, 0, 40);
+
+  webrtc::rtcp::ExtendedReports xr;
+  xr.SetTargetBitrate(target_bitrate);
+  xr.SetSenderSsrc(kRemoteSsrc);
+  auto raw_packet = xr.Build();
+
+  VideoBitrateAllocation expected_allocation;
+  expected_allocation.SetBitrate(0, 0, /*bitrate_bps=*/10000);
+  EXPECT_CALL(observer, OnBitrateAllocation(kRemoteSsrc, expected_allocation));
+  rtcp_transceiver.ReceivePacket(raw_packet, Timestamp::Micros(0));
+}
+
+TEST_F(RtcpTransceiverImplTest, CallsObserverOnByeBehindSenderReport) {
+  const uint32_t kRemoteSsrc = 12345;
+  MockMediaReceiverRtcpObserver observer;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, &observer);
+
+  CompoundPacket compound;
+  auto sr = std::make_unique<SenderReport>();
+  sr->SetSenderSsrc(kRemoteSsrc);
+  compound.Append(std::move(sr));
+  auto bye = std::make_unique<Bye>();
+  bye->SetSenderSsrc(kRemoteSsrc);
+  compound.Append(std::move(bye));
+  auto raw_packet = compound.Build();
+
+  EXPECT_CALL(observer, OnBye(kRemoteSsrc));
+  EXPECT_CALL(observer, OnSenderReport(kRemoteSsrc, _, _));
+  rtcp_transceiver.ReceivePacket(raw_packet, Timestamp::Micros(0));
+}
+
+TEST_F(RtcpTransceiverImplTest, CallsObserverOnByeBehindUnknownRtcpPacket) {
+  const uint32_t kRemoteSsrc = 12345;
+  MockMediaReceiverRtcpObserver observer;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, &observer);
+
+  CompoundPacket compound;
+  // Use Application-Defined rtcp packet as unknown.
+  auto app = std::make_unique<webrtc::rtcp::App>();
+  compound.Append(std::move(app));
+  auto bye = std::make_unique<Bye>();
+  bye->SetSenderSsrc(kRemoteSsrc);
+  compound.Append(std::move(bye));
+  auto raw_packet = compound.Build();
+
+  EXPECT_CALL(observer, OnBye(kRemoteSsrc));
+  rtcp_transceiver.ReceivePacket(raw_packet, Timestamp::Micros(0));
+}
+
+TEST_F(RtcpTransceiverImplTest,
+       WhenSendsReceiverReportSetsLastSenderReportTimestampPerRemoteSsrc) {
+  const uint32_t kRemoteSsrc1 = 4321;
+  const uint32_t kRemoteSsrc2 = 5321;
+  std::vector<ReportBlock> statistics_report_blocks(2);
+  statistics_report_blocks[0].SetMediaSsrc(kRemoteSsrc1);
+  statistics_report_blocks[1].SetMediaSsrc(kRemoteSsrc2);
+  MockReceiveStatisticsProvider receive_statistics;
+  EXPECT_CALL(receive_statistics, RtcpReportBlocks(_))
+      .WillOnce(Return(statistics_report_blocks));
+
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.receive_statistics = &receive_statistics;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  const NtpTime kRemoteNtp(0x9876543211);
+  // Receive SenderReport for RemoteSsrc1, but no report for RemoteSsrc2.
+  SenderReport sr;
+  sr.SetSenderSsrc(kRemoteSsrc1);
+  sr.SetNtp(kRemoteNtp);
+  auto raw_packet = sr.Build();
+  rtcp_transceiver.ReceivePacket(raw_packet, Timestamp::Micros(0));
+
+  // Trigger sending ReceiverReport.
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_GT(rtcp_parser.receiver_report()->num_packets(), 0);
+  const auto& report_blocks = rtcp_parser.receiver_report()->report_blocks();
+  ASSERT_EQ(report_blocks.size(), 2u);
+  // RtcpTransceiverImpl doesn't guarantee order of the report blocks
+  // match result of ReceiveStatisticsProvider::RtcpReportBlocks callback,
+  // but for simplicity of the test asume it is the same.
+  ASSERT_EQ(report_blocks[0].source_ssrc(), kRemoteSsrc1);
+  EXPECT_EQ(report_blocks[0].last_sr(), CompactNtp(kRemoteNtp));
+
+  ASSERT_EQ(report_blocks[1].source_ssrc(), kRemoteSsrc2);
+  // No matching Sender Report for kRemoteSsrc2, LastSR fields has to be 0.
+  EXPECT_EQ(report_blocks[1].last_sr(), 0u);
+}
+
+TEST_F(RtcpTransceiverImplTest,
+       WhenSendsReceiverReportCalculatesDelaySinceLastSenderReport) {
+  const uint32_t kRemoteSsrc1 = 4321;
+  const uint32_t kRemoteSsrc2 = 5321;
+
+  std::vector<ReportBlock> statistics_report_blocks(2);
+  statistics_report_blocks[0].SetMediaSsrc(kRemoteSsrc1);
+  statistics_report_blocks[1].SetMediaSsrc(kRemoteSsrc2);
+  MockReceiveStatisticsProvider receive_statistics;
+  EXPECT_CALL(receive_statistics, RtcpReportBlocks(_))
+      .WillOnce(Return(statistics_report_blocks));
+
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.receive_statistics = &receive_statistics;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  auto receive_sender_report = [&](uint32_t remote_ssrc) {
+    SenderReport sr;
+    sr.SetSenderSsrc(remote_ssrc);
+    rtcp_transceiver.ReceivePacket(sr.Build(), CurrentTime());
+  };
+
+  receive_sender_report(kRemoteSsrc1);
+  time_controller().AdvanceTime(TimeDelta::Millis(100));
+
+  receive_sender_report(kRemoteSsrc2);
+  time_controller().AdvanceTime(TimeDelta::Millis(100));
+
+  // Trigger ReceiverReport back.
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_GT(rtcp_parser.receiver_report()->num_packets(), 0);
+  const auto& report_blocks = rtcp_parser.receiver_report()->report_blocks();
+  ASSERT_EQ(report_blocks.size(), 2u);
+  // RtcpTransceiverImpl doesn't guarantee order of the report blocks
+  // match result of ReceiveStatisticsProvider::RtcpReportBlocks callback,
+  // but for simplicity of the test asume it is the same.
+  ASSERT_EQ(report_blocks[0].source_ssrc(), kRemoteSsrc1);
+  EXPECT_THAT(CompactNtpRttToTimeDelta(report_blocks[0].delay_since_last_sr()),
+              Near(TimeDelta::Millis(200)));
+
+  ASSERT_EQ(report_blocks[1].source_ssrc(), kRemoteSsrc2);
+  EXPECT_THAT(CompactNtpRttToTimeDelta(report_blocks[1].delay_since_last_sr()),
+              Near(TimeDelta::Millis(100)));
+}
+
+TEST_F(RtcpTransceiverImplTest, MaySendMultipleReceiverReportInSinglePacket) {
+  std::vector<ReportBlock> statistics_report_blocks(40);
+  MockReceiveStatisticsProvider receive_statistics;
+  EXPECT_CALL(receive_statistics, RtcpReportBlocks(/*max_blocks=*/Ge(40u)))
+      .WillOnce(Return(statistics_report_blocks));
+
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.receive_statistics = &receive_statistics;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  // Trigger ReceiverReports.
+  rtcp_transceiver.SendCompoundPacket();
+
+  // Expect a single RTCP packet with multiple receiver reports in it.
+  EXPECT_EQ(transport.num_packets(), 1);
+  // Receiver report may contain up to 31 report blocks, thus 2 reports are
+  // needed to carry 40 blocks: 31 in the first, 9 in the last.
+  EXPECT_EQ(rtcp_parser.receiver_report()->num_packets(), 2);
+  // RtcpParser remembers just the last receiver report, thus can't check number
+  // of blocks in the first receiver report.
+  EXPECT_THAT(rtcp_parser.receiver_report()->report_blocks(), SizeIs(9));
+}
+
+TEST_F(RtcpTransceiverImplTest, AttachMaxNumberOfReportBlocksToCompoundPacket) {
+  MockReceiveStatisticsProvider receive_statistics;
+  EXPECT_CALL(receive_statistics, RtcpReportBlocks)
+      .WillOnce([](size_t max_blocks) {
+        return std::vector<ReportBlock>(max_blocks);
+      });
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.rtcp_mode = RtcpMode::kCompound;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.receive_statistics = &receive_statistics;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  EXPECT_EQ(transport.num_packets(), 0);
+  // Send some fast feedback message. Because of compound mode, report blocks
+  // should be attached.
+  rtcp_transceiver.SendPictureLossIndication(/*ssrc=*/123);
+
+  // Expect single RTCP packet with multiple receiver reports and a PLI.
+  EXPECT_EQ(transport.num_packets(), 1);
+  EXPECT_GT(rtcp_parser.receiver_report()->num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.pli()->num_packets(), 1);
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsNack) {
+  const uint32_t kSenderSsrc = 1234;
+  const uint32_t kRemoteSsrc = 4321;
+  std::vector<uint16_t> kMissingSequenceNumbers = {34, 37, 38};
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SendNack(kRemoteSsrc, kMissingSequenceNumbers);
+
+  EXPECT_EQ(rtcp_parser.nack()->num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.nack()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(rtcp_parser.nack()->media_ssrc(), kRemoteSsrc);
+  EXPECT_EQ(rtcp_parser.nack()->packet_ids(), kMissingSequenceNumbers);
+}
+
+TEST_F(RtcpTransceiverImplTest, ReceivesNack) {
+  static constexpr uint32_t kRemoteSsrc = 4321;
+  static constexpr uint32_t kMediaSsrc1 = 1234;
+  static constexpr uint32_t kMediaSsrc2 = 1235;
+  std::vector<uint16_t> kMissingSequenceNumbers = {34, 37, 38};
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  MockRtpStreamRtcpHandler local_stream1;
+  MockRtpStreamRtcpHandler local_stream2;
+  EXPECT_CALL(local_stream1,
+              OnNack(kRemoteSsrc, ElementsAreArray(kMissingSequenceNumbers)));
+  EXPECT_CALL(local_stream2, OnNack).Times(0);
+
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc1, &local_stream1));
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc2, &local_stream2));
+
+  rtcp::Nack nack;
+  nack.SetSenderSsrc(kRemoteSsrc);
+  nack.SetMediaSsrc(kMediaSsrc1);
+  nack.SetPacketIds(kMissingSequenceNumbers);
+  rtcp_transceiver.ReceivePacket(nack.Build(), config.clock->CurrentTime());
+}
+
+TEST_F(RtcpTransceiverImplTest, RequestKeyFrameWithPictureLossIndication) {
+  const uint32_t kSenderSsrc = 1234;
+  const uint32_t kRemoteSsrc = 4321;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SendPictureLossIndication(kRemoteSsrc);
+
+  EXPECT_EQ(transport.num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.pli()->num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.pli()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(rtcp_parser.pli()->media_ssrc(), kRemoteSsrc);
+}
+
+TEST_F(RtcpTransceiverImplTest, ReceivesPictureLossIndication) {
+  static constexpr uint32_t kRemoteSsrc = 4321;
+  static constexpr uint32_t kMediaSsrc1 = 1234;
+  static constexpr uint32_t kMediaSsrc2 = 1235;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  MockRtpStreamRtcpHandler local_stream1;
+  MockRtpStreamRtcpHandler local_stream2;
+  EXPECT_CALL(local_stream1, OnPli(kRemoteSsrc));
+  EXPECT_CALL(local_stream2, OnPli).Times(0);
+
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc1, &local_stream1));
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc2, &local_stream2));
+
+  rtcp::Pli pli;
+  pli.SetSenderSsrc(kRemoteSsrc);
+  pli.SetMediaSsrc(kMediaSsrc1);
+  rtcp_transceiver.ReceivePacket(pli.Build(), config.clock->CurrentTime());
+}
+
+TEST_F(RtcpTransceiverImplTest, RequestKeyFrameWithFullIntraRequest) {
+  const uint32_t kSenderSsrc = 1234;
+  const uint32_t kRemoteSsrcs[] = {4321, 5321};
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SendFullIntraRequest(kRemoteSsrcs, true);
+
+  EXPECT_EQ(rtcp_parser.fir()->num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.fir()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(rtcp_parser.fir()->requests()[0].ssrc, kRemoteSsrcs[0]);
+  EXPECT_EQ(rtcp_parser.fir()->requests()[1].ssrc, kRemoteSsrcs[1]);
+}
+
+TEST_F(RtcpTransceiverImplTest, RequestKeyFrameWithFirIncreaseSeqNoPerSsrc) {
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  const uint32_t kBothRemoteSsrcs[] = {4321, 5321};
+  const uint32_t kOneRemoteSsrc[] = {4321};
+
+  rtcp_transceiver.SendFullIntraRequest(kBothRemoteSsrcs, true);
+  ASSERT_EQ(rtcp_parser.fir()->requests()[0].ssrc, kBothRemoteSsrcs[0]);
+  uint8_t fir_sequence_number0 = rtcp_parser.fir()->requests()[0].seq_nr;
+  ASSERT_EQ(rtcp_parser.fir()->requests()[1].ssrc, kBothRemoteSsrcs[1]);
+  uint8_t fir_sequence_number1 = rtcp_parser.fir()->requests()[1].seq_nr;
+
+  rtcp_transceiver.SendFullIntraRequest(kOneRemoteSsrc, true);
+  ASSERT_EQ(rtcp_parser.fir()->requests().size(), 1u);
+  ASSERT_EQ(rtcp_parser.fir()->requests()[0].ssrc, kBothRemoteSsrcs[0]);
+  EXPECT_EQ(rtcp_parser.fir()->requests()[0].seq_nr, fir_sequence_number0 + 1);
+
+  rtcp_transceiver.SendFullIntraRequest(kBothRemoteSsrcs, true);
+  ASSERT_EQ(rtcp_parser.fir()->requests().size(), 2u);
+  ASSERT_EQ(rtcp_parser.fir()->requests()[0].ssrc, kBothRemoteSsrcs[0]);
+  EXPECT_EQ(rtcp_parser.fir()->requests()[0].seq_nr, fir_sequence_number0 + 2);
+  ASSERT_EQ(rtcp_parser.fir()->requests()[1].ssrc, kBothRemoteSsrcs[1]);
+  EXPECT_EQ(rtcp_parser.fir()->requests()[1].seq_nr, fir_sequence_number1 + 1);
+}
+
+TEST_F(RtcpTransceiverImplTest, SendFirDoesNotIncreaseSeqNoIfOldRequest) {
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  const uint32_t kBothRemoteSsrcs[] = {4321, 5321};
+
+  rtcp_transceiver.SendFullIntraRequest(kBothRemoteSsrcs, true);
+  ASSERT_EQ(rtcp_parser.fir()->requests().size(), 2u);
+  ASSERT_EQ(rtcp_parser.fir()->requests()[0].ssrc, kBothRemoteSsrcs[0]);
+  uint8_t fir_sequence_number0 = rtcp_parser.fir()->requests()[0].seq_nr;
+  ASSERT_EQ(rtcp_parser.fir()->requests()[1].ssrc, kBothRemoteSsrcs[1]);
+  uint8_t fir_sequence_number1 = rtcp_parser.fir()->requests()[1].seq_nr;
+
+  rtcp_transceiver.SendFullIntraRequest(kBothRemoteSsrcs, false);
+  ASSERT_EQ(rtcp_parser.fir()->requests().size(), 2u);
+  ASSERT_EQ(rtcp_parser.fir()->requests()[0].ssrc, kBothRemoteSsrcs[0]);
+  EXPECT_EQ(rtcp_parser.fir()->requests()[0].seq_nr, fir_sequence_number0);
+  ASSERT_EQ(rtcp_parser.fir()->requests()[1].ssrc, kBothRemoteSsrcs[1]);
+  EXPECT_EQ(rtcp_parser.fir()->requests()[1].seq_nr, fir_sequence_number1);
+}
+
+TEST_F(RtcpTransceiverImplTest, ReceivesFir) {
+  static constexpr uint32_t kRemoteSsrc = 4321;
+  static constexpr uint32_t kMediaSsrc1 = 1234;
+  static constexpr uint32_t kMediaSsrc2 = 1235;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  MockRtpStreamRtcpHandler local_stream1;
+  MockRtpStreamRtcpHandler local_stream2;
+  EXPECT_CALL(local_stream1, OnFir(kRemoteSsrc));
+  EXPECT_CALL(local_stream2, OnFir).Times(0);
+
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc1, &local_stream1));
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc2, &local_stream2));
+
+  rtcp::Fir fir;
+  fir.SetSenderSsrc(kRemoteSsrc);
+  fir.AddRequestTo(kMediaSsrc1, /*seq_num=*/13);
+
+  rtcp_transceiver.ReceivePacket(fir.Build(), config.clock->CurrentTime());
+}
+
+TEST_F(RtcpTransceiverImplTest, IgnoresReceivedFirWithRepeatedSequenceNumber) {
+  static constexpr uint32_t kRemoteSsrc = 4321;
+  static constexpr uint32_t kMediaSsrc1 = 1234;
+  static constexpr uint32_t kMediaSsrc2 = 1235;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  MockRtpStreamRtcpHandler local_stream1;
+  MockRtpStreamRtcpHandler local_stream2;
+  EXPECT_CALL(local_stream1, OnFir(kRemoteSsrc)).Times(1);
+  EXPECT_CALL(local_stream2, OnFir(kRemoteSsrc)).Times(2);
+
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc1, &local_stream1));
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc2, &local_stream2));
+
+  rtcp::Fir fir1;
+  fir1.SetSenderSsrc(kRemoteSsrc);
+  fir1.AddRequestTo(kMediaSsrc1, /*seq_num=*/132);
+  fir1.AddRequestTo(kMediaSsrc2, /*seq_num=*/10);
+  rtcp_transceiver.ReceivePacket(fir1.Build(), config.clock->CurrentTime());
+
+  // Repeat request for MediaSsrc1 - expect it to be ignored,
+  // Change FIR sequence number for MediaSsrc2 - expect a 2nd callback.
+  rtcp::Fir fir2;
+  fir2.SetSenderSsrc(kRemoteSsrc);
+  fir2.AddRequestTo(kMediaSsrc1, /*seq_num=*/132);
+  fir2.AddRequestTo(kMediaSsrc2, /*seq_num=*/13);
+  rtcp_transceiver.ReceivePacket(fir2.Build(), config.clock->CurrentTime());
+}
+
+TEST_F(RtcpTransceiverImplTest, ReceivedFirTracksSequenceNumberPerRemoteSsrc) {
+  static constexpr uint32_t kRemoteSsrc1 = 4321;
+  static constexpr uint32_t kRemoteSsrc2 = 4323;
+  static constexpr uint32_t kMediaSsrc = 1234;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  MockRtpStreamRtcpHandler local_stream;
+  EXPECT_CALL(local_stream, OnFir(kRemoteSsrc1));
+  EXPECT_CALL(local_stream, OnFir(kRemoteSsrc2));
+
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc, &local_stream));
+
+  rtcp::Fir fir1;
+  fir1.SetSenderSsrc(kRemoteSsrc1);
+  fir1.AddRequestTo(kMediaSsrc, /*seq_num=*/13);
+  rtcp_transceiver.ReceivePacket(fir1.Build(), config.clock->CurrentTime());
+
+  // Use the same FIR sequence number, but different sender SSRC.
+  rtcp::Fir fir2;
+  fir2.SetSenderSsrc(kRemoteSsrc2);
+  fir2.AddRequestTo(kMediaSsrc, /*seq_num=*/13);
+  rtcp_transceiver.ReceivePacket(fir2.Build(), config.clock->CurrentTime());
+}
+
+TEST_F(RtcpTransceiverImplTest, KeyFrameRequestCreatesCompoundPacket) {
+  const uint32_t kRemoteSsrcs[] = {4321};
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  // Turn periodic off to ensure sent rtcp packet is explicitly requested.
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+
+  config.rtcp_mode = webrtc::RtcpMode::kCompound;
+
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.SendFullIntraRequest(kRemoteSsrcs, true);
+
+  // Test sent packet is compound by expecting presense of receiver report.
+  EXPECT_EQ(transport.num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.receiver_report()->num_packets(), 1);
+}
+
+TEST_F(RtcpTransceiverImplTest, KeyFrameRequestCreatesReducedSizePacket) {
+  const uint32_t kRemoteSsrcs[] = {4321};
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  // Turn periodic off to ensure sent rtcp packet is explicitly requested.
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+
+  config.rtcp_mode = webrtc::RtcpMode::kReducedSize;
+
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  rtcp_transceiver.SendFullIntraRequest(kRemoteSsrcs, true);
+
+  // Test sent packet is reduced size by expecting absense of receiver report.
+  EXPECT_EQ(transport.num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.receiver_report()->num_packets(), 0);
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsXrRrtrWhenEnabled) {
+  const uint32_t kSenderSsrc = 4321;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.non_sender_rtt_measurement = true;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SendCompoundPacket();
+  NtpTime ntp_time_now = config.clock->CurrentNtpTime();
+
+  EXPECT_EQ(rtcp_parser.xr()->num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.xr()->sender_ssrc(), kSenderSsrc);
+  ASSERT_TRUE(rtcp_parser.xr()->rrtr());
+  EXPECT_EQ(rtcp_parser.xr()->rrtr()->ntp(), ntp_time_now);
+}
+
+TEST_F(RtcpTransceiverImplTest, RepliesToRrtrWhenEnabled) {
+  static constexpr uint32_t kSenderSsrc[] = {4321, 9876};
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.reply_to_non_sender_rtt_measurement = true;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp::ExtendedReports xr;
+  rtcp::Rrtr rrtr;
+  rrtr.SetNtp(NtpTime(uint64_t{0x1111'2222'3333'4444}));
+  xr.SetRrtr(rrtr);
+  xr.SetSenderSsrc(kSenderSsrc[0]);
+  rtcp_transceiver.ReceivePacket(xr.Build(), CurrentTime());
+  AdvanceTime(TimeDelta::Millis(1'500));
+
+  rrtr.SetNtp(NtpTime(uint64_t{0x4444'5555'6666'7777}));
+  xr.SetRrtr(rrtr);
+  xr.SetSenderSsrc(kSenderSsrc[1]);
+  rtcp_transceiver.ReceivePacket(xr.Build(), CurrentTime());
+  AdvanceTime(TimeDelta::Millis(500));
+
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(rtcp_parser.xr()->num_packets(), 1);
+  static constexpr uint32_t kComactNtpOneSecond = 0x0001'0000;
+  EXPECT_THAT(rtcp_parser.xr()->dlrr().sub_blocks(),
+              UnorderedElementsAre(
+                  rtcp::ReceiveTimeInfo(kSenderSsrc[0], 0x2222'3333,
+                                        /*delay=*/2 * kComactNtpOneSecond),
+                  rtcp::ReceiveTimeInfo(kSenderSsrc[1], 0x5555'6666,
+                                        /*delay=*/kComactNtpOneSecond / 2)));
+}
+
+TEST_F(RtcpTransceiverImplTest, CanReplyToRrtrOnceForAllLocalSsrcs) {
+  static constexpr uint32_t kRemoteSsrc = 4321;
+  static constexpr uint32_t kLocalSsrcs[] = {1234, 5678};
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.reply_to_non_sender_rtt_measurement = true;
+  config.reply_to_non_sender_rtt_mesaurments_on_all_ssrcs = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  MockRtpStreamRtcpHandler local_sender0;
+  MockRtpStreamRtcpHandler local_sender1;
+  rtcp_transceiver.AddMediaSender(kLocalSsrcs[0], &local_sender0);
+  rtcp_transceiver.AddMediaSender(kLocalSsrcs[1], &local_sender1);
+
+  rtcp::ExtendedReports xr;
+  rtcp::Rrtr rrtr;
+  rrtr.SetNtp(NtpTime(uint64_t{0x1111'2222'3333'4444}));
+  xr.SetRrtr(rrtr);
+  xr.SetSenderSsrc(kRemoteSsrc);
+  rtcp_transceiver.ReceivePacket(xr.Build(), CurrentTime());
+  AdvanceTime(TimeDelta::Millis(1'500));
+
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(rtcp_parser.xr()->num_packets(), 1);
+}
+
+TEST_F(RtcpTransceiverImplTest, CanReplyToRrtrForEachLocalSsrc) {
+  static constexpr uint32_t kRemoteSsrc = 4321;
+  static constexpr uint32_t kLocalSsrc[] = {1234, 5678};
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.reply_to_non_sender_rtt_measurement = true;
+  config.reply_to_non_sender_rtt_mesaurments_on_all_ssrcs = true;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  MockRtpStreamRtcpHandler local_sender0;
+  MockRtpStreamRtcpHandler local_sender1;
+  rtcp_transceiver.AddMediaSender(kLocalSsrc[0], &local_sender0);
+  rtcp_transceiver.AddMediaSender(kLocalSsrc[1], &local_sender1);
+
+  rtcp::ExtendedReports xr;
+  rtcp::Rrtr rrtr;
+  rrtr.SetNtp(NtpTime(uint64_t{0x1111'2222'3333'4444}));
+  xr.SetRrtr(rrtr);
+  xr.SetSenderSsrc(kRemoteSsrc);
+  rtcp_transceiver.ReceivePacket(xr.Build(), CurrentTime());
+  AdvanceTime(TimeDelta::Millis(1'500));
+
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(rtcp_parser.xr()->num_packets(), 2);
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsNoXrRrtrWhenDisabled) {
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.schedule_periodic_compound_packets = false;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.non_sender_rtt_measurement = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  rtcp_transceiver.SendCompoundPacket();
+
+  EXPECT_EQ(transport.num_packets(), 1);
+  // Extended reports rtcp packet might be included for another reason,
+  // but it shouldn't contain rrtr block.
+  EXPECT_FALSE(rtcp_parser.xr()->rrtr());
+}
+
+TEST_F(RtcpTransceiverImplTest, PassRttFromDlrrToLinkObserver) {
+  const uint32_t kSenderSsrc = 4321;
+  MockNetworkLinkRtcpObserver link_observer;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  config.network_link_observer = &link_observer;
+  config.non_sender_rtt_measurement = true;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  Timestamp send_time = Timestamp::Seconds(5678);
+  Timestamp receive_time = send_time + TimeDelta::Millis(110);
+  rtcp::ReceiveTimeInfo rti;
+  rti.ssrc = kSenderSsrc;
+  rti.last_rr = CompactNtp(config.clock->ConvertTimestampToNtpTime(send_time));
+  rti.delay_since_last_rr = SaturatedToCompactNtp(TimeDelta::Millis(10));
+  rtcp::ExtendedReports xr;
+  xr.AddDlrrItem(rti);
+
+  EXPECT_CALL(link_observer,
+              OnRttUpdate(receive_time, Near(TimeDelta::Millis(100))));
+  rtcp_transceiver.ReceivePacket(xr.Build(), receive_time);
+}
+
+TEST_F(RtcpTransceiverImplTest, CalculatesRoundTripTimeFromReportBlocks) {
+  MockNetworkLinkRtcpObserver link_observer;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.network_link_observer = &link_observer;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  TimeDelta rtt = TimeDelta::Millis(100);
+  Timestamp send_time = Timestamp::Seconds(5678);
+  Timestamp receive_time = send_time + TimeDelta::Millis(110);
+  rtcp::ReceiverReport rr;
+  rtcp::ReportBlock rb1;
+  rb1.SetLastSr(CompactNtp(config.clock->ConvertTimestampToNtpTime(
+      receive_time - rtt - TimeDelta::Millis(10))));
+  rb1.SetDelayLastSr(SaturatedToCompactNtp(TimeDelta::Millis(10)));
+  rr.AddReportBlock(rb1);
+  rtcp::ReportBlock rb2;
+  rb2.SetLastSr(CompactNtp(config.clock->ConvertTimestampToNtpTime(
+      receive_time - rtt - TimeDelta::Millis(20))));
+  rb2.SetDelayLastSr(SaturatedToCompactNtp(TimeDelta::Millis(20)));
+  rr.AddReportBlock(rb2);
+
+  EXPECT_CALL(link_observer, OnRttUpdate(receive_time, Near(rtt)));
+  rtcp_transceiver.ReceivePacket(rr.Build(), receive_time);
+}
+
+TEST_F(RtcpTransceiverImplTest, IgnoresUnknownSsrcInDlrr) {
+  const uint32_t kSenderSsrc = 4321;
+  const uint32_t kUnknownSsrc = 4322;
+  MockNetworkLinkRtcpObserver link_observer;
+  MockTransport null_transport;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kSenderSsrc;
+  config.schedule_periodic_compound_packets = false;
+  config.outgoing_transport = &null_transport;
+  config.non_sender_rtt_measurement = true;
+  config.network_link_observer = &link_observer;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  Timestamp time = Timestamp::Micros(12345678);
+  webrtc::rtcp::ReceiveTimeInfo rti;
+  rti.ssrc = kUnknownSsrc;
+  rti.last_rr = CompactNtp(config.clock->ConvertTimestampToNtpTime(time));
+  webrtc::rtcp::ExtendedReports xr;
+  xr.AddDlrrItem(rti);
+  auto raw_packet = xr.Build();
+
+  EXPECT_CALL(link_observer, OnRttUpdate).Times(0);
+  rtcp_transceiver.ReceivePacket(raw_packet, time + TimeDelta::Millis(100));
+}
+
+TEST_F(RtcpTransceiverImplTest, ParsesTransportFeedback) {
+  MockNetworkLinkRtcpObserver link_observer;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.network_link_observer = &link_observer;
+  Timestamp receive_time = Timestamp::Seconds(5678);
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  EXPECT_CALL(link_observer, OnTransportFeedback(receive_time, _))
+      .WillOnce(WithArg<1>([](const rtcp::TransportFeedback& message) {
+        EXPECT_EQ(message.GetBaseSequence(), 321);
+        EXPECT_THAT(message.GetReceivedPackets(), SizeIs(2));
+      }));
+
+  rtcp::TransportFeedback tb;
+  tb.SetBase(/*base_sequence=*/321, Timestamp::Micros(15));
+  tb.AddReceivedPacket(/*base_sequence=*/321, Timestamp::Micros(15));
+  tb.AddReceivedPacket(/*base_sequence=*/322, Timestamp::Micros(17));
+  rtcp_transceiver.ReceivePacket(tb.Build(), receive_time);
+}
+
+TEST_F(RtcpTransceiverImplTest, ParsesRemb) {
+  MockNetworkLinkRtcpObserver link_observer;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.network_link_observer = &link_observer;
+  Timestamp receive_time = Timestamp::Seconds(5678);
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  EXPECT_CALL(link_observer,
+              OnReceiverEstimatedMaxBitrate(receive_time,
+                                            DataRate::BitsPerSec(1'234'000)));
+
+  rtcp::Remb remb;
+  remb.SetBitrateBps(1'234'000);
+  rtcp_transceiver.ReceivePacket(remb.Build(), receive_time);
+}
+
+TEST_F(RtcpTransceiverImplTest,
+       CombinesReportBlocksFromSenderAndRecieverReports) {
+  MockNetworkLinkRtcpObserver link_observer;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.network_link_observer = &link_observer;
+  Timestamp receive_time = Timestamp::Seconds(5678);
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  // Assemble compound packet with multiple rtcp packets in it.
+  rtcp::CompoundPacket packet;
+  auto sr = std::make_unique<rtcp::SenderReport>();
+  sr->SetSenderSsrc(1234);
+  sr->SetReportBlocks(std::vector<ReportBlock>(31));
+  packet.Append(std::move(sr));
+  auto rr1 = std::make_unique<rtcp::ReceiverReport>();
+  rr1->SetReportBlocks(std::vector<ReportBlock>(31));
+  packet.Append(std::move(rr1));
+  auto rr2 = std::make_unique<rtcp::ReceiverReport>();
+  rr2->SetReportBlocks(std::vector<ReportBlock>(2));
+  packet.Append(std::move(rr2));
+
+  EXPECT_CALL(link_observer, OnReportBlocks(receive_time, SizeIs(64)));
+
+  rtcp_transceiver.ReceivePacket(packet.Build(), receive_time);
+}
+
+TEST_F(RtcpTransceiverImplTest,
+       CallbackOnReportBlocksFromSenderAndReceiverReports) {
+  static constexpr uint32_t kRemoteSsrc = 5678;
+  // Has registered sender, report block attached to sender report.
+  static constexpr uint32_t kMediaSsrc1 = 1234;
+  // No registered sender, report block attached to receiver report.
+  // Such report block shouldn't prevent handling following report block.
+  static constexpr uint32_t kMediaSsrc2 = 1235;
+  // Has registered sender, no report block attached.
+  static constexpr uint32_t kMediaSsrc3 = 1236;
+  // Has registered sender, report block attached to receiver report.
+  static constexpr uint32_t kMediaSsrc4 = 1237;
+
+  MockNetworkLinkRtcpObserver link_observer;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  Timestamp receive_time = Timestamp::Seconds(5678);
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  MockRtpStreamRtcpHandler local_stream1;
+  MockRtpStreamRtcpHandler local_stream3;
+  MockRtpStreamRtcpHandler local_stream4;
+  EXPECT_CALL(local_stream1, OnReportBlock(kRemoteSsrc, _));
+  EXPECT_CALL(local_stream3, OnReportBlock).Times(0);
+  EXPECT_CALL(local_stream4, OnReportBlock(kRemoteSsrc, _));
+
+  ASSERT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc1, &local_stream1));
+  ASSERT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc3, &local_stream3));
+  ASSERT_TRUE(rtcp_transceiver.AddMediaSender(kMediaSsrc4, &local_stream4));
+
+  // Assemble compound packet with multiple RTCP packets in it.
+  rtcp::CompoundPacket packet;
+  auto sr = std::make_unique<rtcp::SenderReport>();
+  sr->SetSenderSsrc(kRemoteSsrc);
+  std::vector<ReportBlock> rb(1);
+  rb[0].SetMediaSsrc(kMediaSsrc1);
+  sr->SetReportBlocks(std::move(rb));
+  packet.Append(std::move(sr));
+  auto rr = std::make_unique<rtcp::ReceiverReport>();
+  rr->SetSenderSsrc(kRemoteSsrc);
+  rb = std::vector<ReportBlock>(2);
+  rb[0].SetMediaSsrc(kMediaSsrc2);
+  rb[1].SetMediaSsrc(kMediaSsrc4);
+  rr->SetReportBlocks(std::move(rb));
+  packet.Append(std::move(rr));
+
+  rtcp_transceiver.ReceivePacket(packet.Build(), receive_time);
+}
+
+TEST_F(RtcpTransceiverImplTest, FailsToRegisterTwoSendersWithTheSameSsrc) {
+  RtcpTransceiverImpl rtcp_transceiver(DefaultTestConfig());
+  MockRtpStreamRtcpHandler sender1;
+  MockRtpStreamRtcpHandler sender2;
+
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(/*local_ssrc=*/10001, &sender1));
+  EXPECT_FALSE(rtcp_transceiver.AddMediaSender(/*local_ssrc=*/10001, &sender2));
+  EXPECT_TRUE(rtcp_transceiver.AddMediaSender(/*local_ssrc=*/10002, &sender2));
+
+  EXPECT_TRUE(rtcp_transceiver.RemoveMediaSender(/*local_ssrc=*/10001));
+  EXPECT_FALSE(rtcp_transceiver.RemoveMediaSender(/*local_ssrc=*/10001));
+}
+
+TEST_F(RtcpTransceiverImplTest, SendsSenderReport) {
+  static constexpr uint32_t kFeedbackSsrc = 123;
+  static constexpr uint32_t kSenderSsrc = 12345;
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.feedback_ssrc = kFeedbackSsrc;
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  RtpStreamRtcpHandler::RtpStats sender_stats;
+  sender_stats.set_num_sent_packets(10);
+  sender_stats.set_num_sent_bytes(1000);
+  sender_stats.set_last_rtp_timestamp(0x3333);
+  sender_stats.set_last_capture_time(CurrentTime() - TimeDelta::Seconds(2));
+  sender_stats.set_last_clock_rate(0x1000);
+  MockRtpStreamRtcpHandler sender;
+  ON_CALL(sender, SentStats).WillByDefault(Return(sender_stats));
+  rtcp_transceiver.AddMediaSender(kSenderSsrc, &sender);
+
+  rtcp_transceiver.SendCompoundPacket();
+
+  ASSERT_GT(rtcp_parser.sender_report()->num_packets(), 0);
+  EXPECT_EQ(rtcp_parser.sender_report()->sender_ssrc(), kSenderSsrc);
+  EXPECT_EQ(rtcp_parser.sender_report()->ntp(),
+            time_controller().GetClock()->CurrentNtpTime());
+  EXPECT_EQ(rtcp_parser.sender_report()->rtp_timestamp(), 0x3333u + 0x2000u);
+  EXPECT_EQ(rtcp_parser.sender_report()->sender_packet_count(), 10u);
+  EXPECT_EQ(rtcp_parser.sender_report()->sender_octet_count(), 1000u);
+}
+
+TEST_F(RtcpTransceiverImplTest,
+       MaySendBothSenderReportAndReceiverReportInTheSamePacket) {
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  std::vector<ReportBlock> statistics_report_blocks(40);
+  MockReceiveStatisticsProvider receive_statistics;
+  EXPECT_CALL(receive_statistics, RtcpReportBlocks(/*max_blocks=*/Ge(40u)))
+      .WillOnce(Return(statistics_report_blocks));
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.outgoing_transport = &transport;
+  config.receive_statistics = &receive_statistics;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  MockRtpStreamRtcpHandler sender;
+  rtcp_transceiver.AddMediaSender(/*ssrc=*/12345, &sender);
+
+  rtcp_transceiver.SendCompoundPacket();
+
+  // Expect a single RTCP packet with a sender and a receiver reports in it.
+  EXPECT_EQ(transport.num_packets(), 1);
+  ASSERT_EQ(rtcp_parser.sender_report()->num_packets(), 1);
+  ASSERT_EQ(rtcp_parser.receiver_report()->num_packets(), 1);
+  // Sender report may contain up to 31 report blocks, thus remaining 9 report
+  // block should be attached to the receiver report.
+  EXPECT_THAT(rtcp_parser.sender_report()->report_blocks(), SizeIs(31));
+  EXPECT_THAT(rtcp_parser.receiver_report()->report_blocks(), SizeIs(9));
+}
+
+TEST_F(RtcpTransceiverImplTest, RotatesSendersWhenAllSenderReportDoNotFit) {
+  // Send 6 compound packet, each should contain 5 sender reports,
+  // each of 6 senders should be mentioned 5 times.
+  static constexpr int kNumSenders = 6;
+  static constexpr uint32_t kSenderSsrc[kNumSenders] = {10, 20, 30, 40, 50, 60};
+  static constexpr int kSendersPerPacket = 5;
+  // RtcpPacketParser remembers only latest block for each type, but this test
+  // is about sending multiple sender reports in the same packet, thus need
+  // a more advance parser: RtcpTranceiver
+  RtcpTransceiverConfig receiver_config = DefaultTestConfig();
+  RtcpTransceiverImpl rtcp_receiver(receiver_config);
+  // Main expectatation: all senders are spread equally across multiple packets.
+  NiceMock<MockMediaReceiverRtcpObserver> receiver[kNumSenders];
+  for (int i = 0; i < kNumSenders; ++i) {
+    SCOPED_TRACE(i);
+    EXPECT_CALL(receiver[i], OnSenderReport(kSenderSsrc[i], _, _))
+        .Times(kSendersPerPacket);
+    rtcp_receiver.AddMediaReceiverRtcpObserver(kSenderSsrc[i], &receiver[i]);
+  }
+
+  MockTransport transport;
+  EXPECT_CALL(transport, SendRtcp)
+      .Times(kNumSenders)
+      .WillRepeatedly([&](const uint8_t* data, size_t size) {
+        rtcp_receiver.ReceivePacket(rtc::MakeArrayView(data, size),
+                                    CurrentTime());
+        return true;
+      });
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  config.outgoing_transport = &transport;
+  // Limit packet to have space just for kSendersPerPacket sender reports.
+  // Sender report without report blocks require 28 bytes.
+  config.max_packet_size = kSendersPerPacket * 28;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+  NiceMock<MockRtpStreamRtcpHandler> sender[kNumSenders];
+  for (int i = 0; i < kNumSenders; ++i) {
+    rtcp_transceiver.AddMediaSender(kSenderSsrc[i], &sender[i]);
+  }
+
+  for (int i = 1; i <= kNumSenders; ++i) {
+    SCOPED_TRACE(i);
+    rtcp_transceiver.SendCompoundPacket();
+  }
+}
+
+TEST_F(RtcpTransceiverImplTest, SkipsSenderReportForInactiveSender) {
+  static constexpr uint32_t kSenderSsrc[] = {12345, 23456};
+  RtcpTransceiverConfig config = DefaultTestConfig();
+  RtcpPacketParser rtcp_parser;
+  RtcpParserTransport transport(&rtcp_parser);
+  config.outgoing_transport = &transport;
+  RtcpTransceiverImpl rtcp_transceiver(config);
+
+  RtpStreamRtcpHandler::RtpStats sender_stats[2];
+  NiceMock<MockRtpStreamRtcpHandler> sender[2];
+  ON_CALL(sender[0], SentStats).WillByDefault([&] { return sender_stats[0]; });
+  ON_CALL(sender[1], SentStats).WillByDefault([&] { return sender_stats[1]; });
+  rtcp_transceiver.AddMediaSender(kSenderSsrc[0], &sender[0]);
+  rtcp_transceiver.AddMediaSender(kSenderSsrc[1], &sender[1]);
+
+  // Start with both senders beeing active.
+  sender_stats[0].set_num_sent_packets(10);
+  sender_stats[0].set_num_sent_bytes(1'000);
+  sender_stats[1].set_num_sent_packets(5);
+  sender_stats[1].set_num_sent_bytes(2'000);
+  rtcp_transceiver.SendCompoundPacket();
+  EXPECT_EQ(transport.num_packets(), 1);
+  EXPECT_EQ(rtcp_parser.sender_report()->num_packets(), 2);
+
+  // Keep 1st sender active, but make 2nd second look inactive by returning the
+  // same RtpStats.
+  sender_stats[0].set_num_sent_packets(15);
+  sender_stats[0].set_num_sent_bytes(2'000);
+  rtcp_transceiver.SendCompoundPacket();
+  EXPECT_EQ(transport.num_packets(), 2);
+  EXPECT_EQ(rtcp_parser.sender_report()->num_packets(), 3);
+  EXPECT_EQ(rtcp_parser.sender_report()->sender_ssrc(), kSenderSsrc[0]);
+
+  // Swap active sender.
+  sender_stats[1].set_num_sent_packets(20);
+  sender_stats[1].set_num_sent_bytes(3'000);
+  rtcp_transceiver.SendCompoundPacket();
+  EXPECT_EQ(transport.num_packets(), 3);
+  EXPECT_EQ(rtcp_parser.sender_report()->num_packets(), 4);
+  EXPECT_EQ(rtcp_parser.sender_report()->sender_ssrc(), kSenderSsrc[1]);
+
+  // Activate both senders again.
+  sender_stats[0].set_num_sent_packets(20);
+  sender_stats[0].set_num_sent_bytes(3'000);
+  sender_stats[1].set_num_sent_packets(25);
+  sender_stats[1].set_num_sent_bytes(3'500);
+  rtcp_transceiver.SendCompoundPacket();
+  EXPECT_EQ(transport.num_packets(), 4);
+  EXPECT_EQ(rtcp_parser.sender_report()->num_packets(), 6);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
new file mode 100644
index 0000000000..611ba9d75d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
@@ -0,0 +1,366 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtcp_transceiver.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/event.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
+#include "test/rtcp_packet_parser.h"
+
+namespace {
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::Invoke;
+using ::testing::InvokeWithoutArgs;
+using ::testing::IsNull;
+using ::testing::NiceMock;
+using ::webrtc::MockTransport;
+using ::webrtc::RtcpTransceiver;
+using ::webrtc::RtcpTransceiverConfig;
+using ::webrtc::SimulatedClock;
+using ::webrtc::TaskQueueForTest;
+using ::webrtc::Timestamp;
+using ::webrtc::rtcp::RemoteEstimate;
+using ::webrtc::rtcp::RtcpPacket;
+using ::webrtc::rtcp::TransportFeedback;
+using ::webrtc::test::RtcpPacketParser;
+
+class MockMediaReceiverRtcpObserver : public webrtc::MediaReceiverRtcpObserver {
+ public:
+  MOCK_METHOD(void,
+              OnSenderReport,
+              (uint32_t, webrtc::NtpTime, uint32_t),
+              (override));
+};
+
+constexpr webrtc::TimeDelta kTimeout = webrtc::TimeDelta::Seconds(1);
+
+void WaitPostedTasks(TaskQueueForTest* queue) {
+  rtc::Event done;
+  queue->PostTask([&done] { done.Set(); });
+  ASSERT_TRUE(done.Wait(kTimeout));
+}
+
+TEST(RtcpTransceiverTest, SendsRtcpOnTaskQueueWhenCreatedOffTaskQueue) {
+  SimulatedClock clock(0);
+  MockTransport outgoing_transport;
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.outgoing_transport = &outgoing_transport;
+  config.task_queue = queue.Get();
+  EXPECT_CALL(outgoing_transport, SendRtcp(_, _))
+      .WillRepeatedly(InvokeWithoutArgs([&] {
+        EXPECT_TRUE(queue.IsCurrent());
+        return true;
+      }));
+
+  RtcpTransceiver rtcp_transceiver(config);
+  rtcp_transceiver.SendCompoundPacket();
+  WaitPostedTasks(&queue);
+}
+
+TEST(RtcpTransceiverTest, SendsRtcpOnTaskQueueWhenCreatedOnTaskQueue) {
+  SimulatedClock clock(0);
+  MockTransport outgoing_transport;
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.outgoing_transport = &outgoing_transport;
+  config.task_queue = queue.Get();
+  EXPECT_CALL(outgoing_transport, SendRtcp(_, _))
+      .WillRepeatedly(InvokeWithoutArgs([&] {
+        EXPECT_TRUE(queue.IsCurrent());
+        return true;
+      }));
+
+  std::unique_ptr<RtcpTransceiver> rtcp_transceiver;
+  queue.PostTask([&] {
+    rtcp_transceiver = std::make_unique<RtcpTransceiver>(config);
+    rtcp_transceiver->SendCompoundPacket();
+  });
+  WaitPostedTasks(&queue);
+}
+
+TEST(RtcpTransceiverTest, CanBeDestroyedOnTaskQueue) {
+  SimulatedClock clock(0);
+  NiceMock<MockTransport> outgoing_transport;
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.outgoing_transport = &outgoing_transport;
+  config.task_queue = queue.Get();
+  auto rtcp_transceiver = std::make_unique<RtcpTransceiver>(config);
+
+  queue.PostTask([&] {
+    // Insert a packet just before destruction to test for races.
+    rtcp_transceiver->SendCompoundPacket();
+    rtcp_transceiver.reset();
+  });
+  WaitPostedTasks(&queue);
+}
+
+TEST(RtcpTransceiverTest, CanBeDestroyedWithoutBlocking) {
+  SimulatedClock clock(0);
+  TaskQueueForTest queue("rtcp");
+  NiceMock<MockTransport> outgoing_transport;
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.outgoing_transport = &outgoing_transport;
+  config.task_queue = queue.Get();
+  auto* rtcp_transceiver = new RtcpTransceiver(config);
+  rtcp_transceiver->SendCompoundPacket();
+
+  rtc::Event done;
+  rtc::Event heavy_task;
+  queue.PostTask([&] {
+    EXPECT_TRUE(heavy_task.Wait(kTimeout));
+    done.Set();
+  });
+  delete rtcp_transceiver;
+
+  heavy_task.Set();
+  EXPECT_TRUE(done.Wait(kTimeout));
+}
+
+TEST(RtcpTransceiverTest, MaySendPacketsAfterDestructor) {  // i.e. Be careful!
+  SimulatedClock clock(0);
+  NiceMock<MockTransport> outgoing_transport;  // Must outlive queue below.
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.outgoing_transport = &outgoing_transport;
+  config.task_queue = queue.Get();
+  auto* rtcp_transceiver = new RtcpTransceiver(config);
+
+  rtc::Event heavy_task;
+  queue.PostTask([&] { EXPECT_TRUE(heavy_task.Wait(kTimeout)); });
+  rtcp_transceiver->SendCompoundPacket();
+  delete rtcp_transceiver;
+
+  EXPECT_CALL(outgoing_transport, SendRtcp);
+  heavy_task.Set();
+
+  WaitPostedTasks(&queue);
+}
+
+// Use rtp timestamp to distinguish different incoming sender reports.
+rtc::CopyOnWriteBuffer CreateSenderReport(uint32_t ssrc, uint32_t rtp_time) {
+  webrtc::rtcp::SenderReport sr;
+  sr.SetSenderSsrc(ssrc);
+  sr.SetRtpTimestamp(rtp_time);
+  rtc::Buffer buffer = sr.Build();
+  // Switch to an efficient way creating CopyOnWriteBuffer from RtcpPacket when
+  // there is one. Until then do not worry about extra memcpy in test.
+  return rtc::CopyOnWriteBuffer(buffer.data(), buffer.size());
+}
+
+TEST(RtcpTransceiverTest, DoesntPostToRtcpObserverAfterCallToRemove) {
+  const uint32_t kRemoteSsrc = 1234;
+  SimulatedClock clock(0);
+  MockTransport null_transport;
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.outgoing_transport = &null_transport;
+  config.task_queue = queue.Get();
+  RtcpTransceiver rtcp_transceiver(config);
+  rtc::Event observer_deleted;
+
+  auto observer = std::make_unique<MockMediaReceiverRtcpObserver>();
+  EXPECT_CALL(*observer, OnSenderReport(kRemoteSsrc, _, 1));
+  EXPECT_CALL(*observer, OnSenderReport(kRemoteSsrc, _, 2)).Times(0);
+
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, observer.get());
+  rtcp_transceiver.ReceivePacket(CreateSenderReport(kRemoteSsrc, 1));
+  rtcp_transceiver.RemoveMediaReceiverRtcpObserver(kRemoteSsrc, observer.get(),
+                                                   /*on_removed=*/[&] {
+                                                     observer.reset();
+                                                     observer_deleted.Set();
+                                                   });
+  rtcp_transceiver.ReceivePacket(CreateSenderReport(kRemoteSsrc, 2));
+
+  EXPECT_TRUE(observer_deleted.Wait(kTimeout));
+  WaitPostedTasks(&queue);
+}
+
+TEST(RtcpTransceiverTest, RemoveMediaReceiverRtcpObserverIsNonBlocking) {
+  const uint32_t kRemoteSsrc = 1234;
+  SimulatedClock clock(0);
+  MockTransport null_transport;
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.outgoing_transport = &null_transport;
+  config.task_queue = queue.Get();
+  RtcpTransceiver rtcp_transceiver(config);
+  auto observer = std::make_unique<MockMediaReceiverRtcpObserver>();
+  rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, observer.get());
+
+  rtc::Event queue_blocker;
+  rtc::Event observer_deleted;
+  queue.PostTask([&] { EXPECT_TRUE(queue_blocker.Wait(kTimeout)); });
+  rtcp_transceiver.RemoveMediaReceiverRtcpObserver(kRemoteSsrc, observer.get(),
+                                                   /*on_removed=*/[&] {
+                                                     observer.reset();
+                                                     observer_deleted.Set();
+                                                   });
+
+  EXPECT_THAT(observer, Not(IsNull()));
+  queue_blocker.Set();
+  EXPECT_TRUE(observer_deleted.Wait(kTimeout));
+}
+
+TEST(RtcpTransceiverTest, CanCallSendCompoundPacketFromAnyThread) {
+  SimulatedClock clock(0);
+  MockTransport outgoing_transport;
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.outgoing_transport = &outgoing_transport;
+  config.task_queue = queue.Get();
+
+  EXPECT_CALL(outgoing_transport, SendRtcp(_, _))
+      // If test is slow, a periodic task may send an extra packet.
+      .Times(AtLeast(3))
+      .WillRepeatedly(InvokeWithoutArgs([&] {
+        EXPECT_TRUE(queue.IsCurrent());
+        return true;
+      }));
+
+  RtcpTransceiver rtcp_transceiver(config);
+
+  // Call from the construction thread.
+  rtcp_transceiver.SendCompoundPacket();
+  // Call from the same queue transceiver use for processing.
+  queue.PostTask([&] { rtcp_transceiver.SendCompoundPacket(); });
+  // Call from unrelated task queue.
+  TaskQueueForTest queue_send("send_packet");
+  queue_send.PostTask([&] { rtcp_transceiver.SendCompoundPacket(); });
+
+  WaitPostedTasks(&queue_send);
+  WaitPostedTasks(&queue);
+}
+
+TEST(RtcpTransceiverTest, DoesntSendPacketsAfterStopCallback) {
+  SimulatedClock clock(0);
+  NiceMock<MockTransport> outgoing_transport;
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.outgoing_transport = &outgoing_transport;
+  config.task_queue = queue.Get();
+  config.schedule_periodic_compound_packets = true;
+
+  auto rtcp_transceiver = std::make_unique<RtcpTransceiver>(config);
+  rtc::Event done;
+  rtcp_transceiver->SendCompoundPacket();
+  rtcp_transceiver->Stop([&] {
+    EXPECT_CALL(outgoing_transport, SendRtcp).Times(0);
+    done.Set();
+  });
+  rtcp_transceiver = nullptr;
+  EXPECT_TRUE(done.Wait(kTimeout));
+}
+
+TEST(RtcpTransceiverTest, SendsCombinedRtcpPacketOnTaskQueue) {
+  static constexpr uint32_t kSenderSsrc = 12345;
+
+  SimulatedClock clock(0);
+  MockTransport outgoing_transport;
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.feedback_ssrc = kSenderSsrc;
+  config.outgoing_transport = &outgoing_transport;
+  config.task_queue = queue.Get();
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiver rtcp_transceiver(config);
+
+  EXPECT_CALL(outgoing_transport, SendRtcp)
+      .WillOnce([&](const uint8_t* buffer, size_t size) {
+        EXPECT_TRUE(queue.IsCurrent());
+        RtcpPacketParser rtcp_parser;
+        rtcp_parser.Parse(buffer, size);
+        EXPECT_EQ(rtcp_parser.transport_feedback()->num_packets(), 1);
+        EXPECT_EQ(rtcp_parser.transport_feedback()->sender_ssrc(), kSenderSsrc);
+        EXPECT_EQ(rtcp_parser.app()->num_packets(), 1);
+        EXPECT_EQ(rtcp_parser.app()->sender_ssrc(), kSenderSsrc);
+        return true;
+      });
+
+  // Create minimalistic transport feedback packet.
+  std::vector<std::unique_ptr<RtcpPacket>> packets;
+  auto transport_feedback = std::make_unique<TransportFeedback>();
+  transport_feedback->AddReceivedPacket(321, Timestamp::Millis(10));
+  packets.push_back(std::move(transport_feedback));
+
+  auto remote_estimate = std::make_unique<RemoteEstimate>();
+  packets.push_back(std::move(remote_estimate));
+
+  rtcp_transceiver.SendCombinedRtcpPacket(std::move(packets));
+  WaitPostedTasks(&queue);
+}
+
+TEST(RtcpTransceiverTest, SendFrameIntraRequestDefaultsToNewRequest) {
+  static constexpr uint32_t kSenderSsrc = 12345;
+
+  SimulatedClock clock(0);
+  MockTransport outgoing_transport;
+  TaskQueueForTest queue("rtcp");
+  RtcpTransceiverConfig config;
+  config.clock = &clock;
+  config.feedback_ssrc = kSenderSsrc;
+  config.outgoing_transport = &outgoing_transport;
+  config.task_queue = queue.Get();
+  config.schedule_periodic_compound_packets = false;
+  RtcpTransceiver rtcp_transceiver(config);
+
+  uint8_t first_seq_nr;
+  EXPECT_CALL(outgoing_transport, SendRtcp)
+      .WillOnce([&](const uint8_t* buffer, size_t size) {
+        EXPECT_TRUE(queue.IsCurrent());
+        RtcpPacketParser rtcp_parser;
+        rtcp_parser.Parse(buffer, size);
+        EXPECT_EQ(rtcp_parser.fir()->requests()[0].ssrc, kSenderSsrc);
+        first_seq_nr = rtcp_parser.fir()->requests()[0].seq_nr;
+        return true;
+      })
+      .WillOnce([&](const uint8_t* buffer, size_t size) {
+        EXPECT_TRUE(queue.IsCurrent());
+        RtcpPacketParser rtcp_parser;
+        rtcp_parser.Parse(buffer, size);
+        EXPECT_EQ(rtcp_parser.fir()->requests()[0].ssrc, kSenderSsrc);
+        EXPECT_EQ(rtcp_parser.fir()->requests()[0].seq_nr, first_seq_nr + 1);
+        return true;
+      });
+
+  // Send 2 FIR packets because the sequence numbers are incremented after,
+  // sending. One wouldn't be able to differentiate the new_request.
+  rtcp_transceiver.SendFullIntraRequest({kSenderSsrc});
+  rtcp_transceiver.SendFullIntraRequest({kSenderSsrc});
+
+  WaitPostedTasks(&queue);
+}
+
+}  // namespace
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc
new file mode 100644
index 0000000000..fd42b798d4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+
+#include <bitset>
+#include <cstdint>
+
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h"
+#include "rtc_base/numerics/divide_round.h"
+
+namespace webrtc {
+
+constexpr RTPExtensionType RtpDependencyDescriptorExtension::kId;
+constexpr std::bitset<32> RtpDependencyDescriptorExtension::kAllChainsAreActive;
+
+bool RtpDependencyDescriptorExtension::Parse(
+    rtc::ArrayView<const uint8_t> data,
+    const FrameDependencyStructure* structure,
+    DependencyDescriptor* descriptor) {
+  RtpDependencyDescriptorReader reader(data, structure, descriptor);
+  return reader.ParseSuccessful();
+}
+
+size_t RtpDependencyDescriptorExtension::ValueSize(
+    const FrameDependencyStructure& structure,
+    std::bitset<32> active_chains,
+    const DependencyDescriptor& descriptor) {
+  RtpDependencyDescriptorWriter writer(/*data=*/{}, structure, active_chains,
+                                       descriptor);
+  return DivideRoundUp(writer.ValueSizeBits(), 8);
+}
+
+bool RtpDependencyDescriptorExtension::Write(
+    rtc::ArrayView<uint8_t> data,
+    const FrameDependencyStructure& structure,
+    std::bitset<32> active_chains,
+    const DependencyDescriptor& descriptor) {
+  RtpDependencyDescriptorWriter writer(data, structure, active_chains,
+                                       descriptor);
+  return writer.Write();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
new file mode 100644
index 0000000000..8d6e4b8d37
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_EXTENSION_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_EXTENSION_H_
+
+#include <bitset>
+#include <cstdint>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/rtp_parameters.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+// Trait to read/write the dependency descriptor extension as described in
+// https://aomediacodec.github.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension
+class RtpDependencyDescriptorExtension {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionDependencyDescriptor;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kDependencyDescriptorUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    const FrameDependencyStructure* structure,
+                    DependencyDescriptor* descriptor);
+
+  static size_t ValueSize(const FrameDependencyStructure& structure,
+                          const DependencyDescriptor& descriptor) {
+    return ValueSize(structure, kAllChainsAreActive, descriptor);
+  }
+  static size_t ValueSize(const FrameDependencyStructure& structure,
+                          std::bitset<32> active_chains,
+                          const DependencyDescriptor& descriptor);
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    const FrameDependencyStructure& structure,
+                    const DependencyDescriptor& descriptor) {
+    return Write(data, structure, kAllChainsAreActive, descriptor);
+  }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    const FrameDependencyStructure& structure,
+                    std::bitset<32> active_chains,
+                    const DependencyDescriptor& descriptor);
+
+ private:
+  static constexpr std::bitset<32> kAllChainsAreActive = ~uint32_t{0};
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_EXTENSION_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc
new file mode 100644
index 0000000000..974557ce6e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc
@@ -0,0 +1,137 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Each;
+
+TEST(RtpDependencyDescriptorExtensionTest, Writer3BytesForPerfectTemplate) {
+  uint8_t buffer[3];
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 2;
+  structure.num_chains = 2;
+  structure.templates = {
+      FrameDependencyTemplate().Dtis("SR").FrameDiffs({1}).ChainDiffs({2, 2})};
+  DependencyDescriptor descriptor;
+  descriptor.frame_dependencies = structure.templates[0];
+
+  EXPECT_EQ(RtpDependencyDescriptorExtension::ValueSize(structure, descriptor),
+            3u);
+  EXPECT_TRUE(
+      RtpDependencyDescriptorExtension::Write(buffer, structure, descriptor));
+}
+
+TEST(RtpDependencyDescriptorExtensionTest, WriteZeroInUnusedBits) {
+  uint8_t buffer[32];
+  std::memset(buffer, 0xff, sizeof(buffer));
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 2;
+  structure.num_chains = 2;
+  structure.templates = {
+      FrameDependencyTemplate().Dtis("SR").FrameDiffs({1}).ChainDiffs({1, 1})};
+  DependencyDescriptor descriptor;
+  descriptor.frame_dependencies = structure.templates[0];
+  descriptor.frame_dependencies.frame_diffs = {2};
+
+  // To test unused bytes are zeroed, need a buffer large enough.
+  size_t value_size =
+      RtpDependencyDescriptorExtension::ValueSize(structure, descriptor);
+  ASSERT_LT(value_size, sizeof(buffer));
+
+  ASSERT_TRUE(
+      RtpDependencyDescriptorExtension::Write(buffer, structure, descriptor));
+
+  const uint8_t* unused_bytes = buffer + value_size;
+  size_t num_unused_bytes = buffer + sizeof(buffer) - unused_bytes;
+  // Check remaining bytes are zeroed.
+  EXPECT_THAT(rtc::MakeArrayView(unused_bytes, num_unused_bytes), Each(0));
+}
+
+// In practice chain diff for inactive chain will grow uboundly because no
+// frames are produced for it, that shouldn't block writing the extension.
+TEST(RtpDependencyDescriptorExtensionTest,
+     TemplateMatchingSkipsInactiveChains) {
+  uint8_t buffer[3];
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 2;
+  structure.num_chains = 2;
+  structure.templates = {
+      FrameDependencyTemplate().Dtis("SR").ChainDiffs({2, 2})};
+  DependencyDescriptor descriptor;
+  descriptor.frame_dependencies = structure.templates[0];
+
+  // Set only 1st chain as active.
+  std::bitset<32> active_chains = 0b01;
+  descriptor.frame_dependencies.chain_diffs[1] = 1000;
+
+  // Expect perfect template match since the only difference is for an inactive
+  // chain. Pefect template match consumes 3 bytes.
+  EXPECT_EQ(RtpDependencyDescriptorExtension::ValueSize(
+                structure, active_chains, descriptor),
+            3u);
+  EXPECT_TRUE(RtpDependencyDescriptorExtension::Write(
+      buffer, structure, active_chains, descriptor));
+}
+
+TEST(RtpDependencyDescriptorExtensionTest,
+     AcceptsInvalidChainDiffForInactiveChainWhenChainsAreCustom) {
+  uint8_t buffer[256];
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 2;
+  structure.num_chains = 2;
+  structure.templates = {
+      FrameDependencyTemplate().Dtis("SR").ChainDiffs({2, 2})};
+  DependencyDescriptor descriptor;
+  descriptor.frame_dependencies = structure.templates[0];
+
+  // Set only 1st chain as active.
+  std::bitset<32> active_chains = 0b01;
+  // Set chain_diff different to the template to make it custom.
+  descriptor.frame_dependencies.chain_diffs[0] = 1;
+  // Set chain diff for inactive chain beyound limit of 255 max chain diff.
+  descriptor.frame_dependencies.chain_diffs[1] = 1000;
+
+  // Because chains are custom, should use more than base 3 bytes.
+  EXPECT_GT(RtpDependencyDescriptorExtension::ValueSize(
+                structure, active_chains, descriptor),
+            3u);
+  EXPECT_TRUE(RtpDependencyDescriptorExtension::Write(
+      buffer, structure, active_chains, descriptor));
+}
+
+TEST(RtpDependencyDescriptorExtensionTest, FailsToWriteInvalidDescriptor) {
+  uint8_t buffer[256];
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 2;
+  structure.num_chains = 2;
+  structure.templates = {
+      FrameDependencyTemplate().T(0).Dtis("SR").ChainDiffs({2, 2})};
+  DependencyDescriptor descriptor;
+  descriptor.frame_dependencies = structure.templates[0];
+  descriptor.frame_dependencies.temporal_id = 1;
+
+  EXPECT_EQ(
+      RtpDependencyDescriptorExtension::ValueSize(structure, 0b11, descriptor),
+      0u);
+  EXPECT_FALSE(RtpDependencyDescriptorExtension::Write(buffer, structure, 0b11,
+                                                       descriptor));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc
new file mode 100644
index 0000000000..1a56efd9b3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.cc
@@ -0,0 +1,239 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "rtc_base/bitstream_reader.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+RtpDependencyDescriptorReader::RtpDependencyDescriptorReader(
+    rtc::ArrayView<const uint8_t> raw_data,
+    const FrameDependencyStructure* structure,
+    DependencyDescriptor* descriptor)
+    : descriptor_(descriptor), buffer_(raw_data) {
+  RTC_DCHECK(descriptor);
+
+  ReadMandatoryFields();
+  if (raw_data.size() > 3)
+    ReadExtendedFields();
+
+  structure_ = descriptor->attached_structure
+                   ? descriptor->attached_structure.get()
+                   : structure;
+  if (structure_ == nullptr) {
+    buffer_.Invalidate();
+    return;
+  }
+  if (active_decode_targets_present_flag_) {
+    descriptor->active_decode_targets_bitmask =
+        buffer_.ReadBits(structure_->num_decode_targets);
+  }
+
+  ReadFrameDependencyDefinition();
+}
+
+void RtpDependencyDescriptorReader::ReadTemplateDependencyStructure() {
+  descriptor_->attached_structure =
+      std::make_unique<FrameDependencyStructure>();
+  descriptor_->attached_structure->structure_id = buffer_.ReadBits(6);
+  descriptor_->attached_structure->num_decode_targets = buffer_.ReadBits(5) + 1;
+
+  ReadTemplateLayers();
+  ReadTemplateDtis();
+  ReadTemplateFdiffs();
+  ReadTemplateChains();
+
+  if (buffer_.Read<bool>())
+    ReadResolutions();
+}
+
+void RtpDependencyDescriptorReader::ReadTemplateLayers() {
+  enum NextLayerIdc {
+    kSameLayer = 0,
+    kNextTemporalLayer = 1,
+    kNextSpatialLayer = 2,
+    kNoMoreTemplates = 3,
+  };
+  std::vector<FrameDependencyTemplate> templates;
+
+  int temporal_id = 0;
+  int spatial_id = 0;
+  NextLayerIdc next_layer_idc;
+  do {
+    if (templates.size() == DependencyDescriptor::kMaxTemplates) {
+      buffer_.Invalidate();
+      break;
+    }
+    templates.emplace_back();
+    FrameDependencyTemplate& last_template = templates.back();
+    last_template.temporal_id = temporal_id;
+    last_template.spatial_id = spatial_id;
+
+    next_layer_idc = static_cast<NextLayerIdc>(buffer_.ReadBits(2));
+    if (next_layer_idc == kNextTemporalLayer) {
+      temporal_id++;
+      if (temporal_id >= DependencyDescriptor::kMaxTemporalIds) {
+        buffer_.Invalidate();
+        break;
+      }
+    } else if (next_layer_idc == kNextSpatialLayer) {
+      temporal_id = 0;
+      spatial_id++;
+      if (spatial_id >= DependencyDescriptor::kMaxSpatialIds) {
+        buffer_.Invalidate();
+        break;
+      }
+    }
+  } while (next_layer_idc != kNoMoreTemplates && buffer_.Ok());
+
+  descriptor_->attached_structure->templates = std::move(templates);
+}
+
+void RtpDependencyDescriptorReader::ReadTemplateDtis() {
+  FrameDependencyStructure* structure = descriptor_->attached_structure.get();
+  for (FrameDependencyTemplate& current_template : structure->templates) {
+    current_template.decode_target_indications.resize(
+        structure->num_decode_targets);
+    for (int i = 0; i < structure->num_decode_targets; ++i) {
+      current_template.decode_target_indications[i] =
+          static_cast<DecodeTargetIndication>(buffer_.ReadBits(2));
+    }
+  }
+}
+
+void RtpDependencyDescriptorReader::ReadTemplateFdiffs() {
+  for (FrameDependencyTemplate& current_template :
+       descriptor_->attached_structure->templates) {
+    for (bool fdiff_follows = buffer_.Read<bool>(); fdiff_follows;
+         fdiff_follows = buffer_.Read<bool>()) {
+      uint64_t fdiff_minus_one = buffer_.ReadBits(4);
+      current_template.frame_diffs.push_back(fdiff_minus_one + 1);
+    }
+  }
+}
+
+void RtpDependencyDescriptorReader::ReadTemplateChains() {
+  FrameDependencyStructure* structure = descriptor_->attached_structure.get();
+  structure->num_chains =
+      buffer_.ReadNonSymmetric(structure->num_decode_targets + 1);
+  if (structure->num_chains == 0)
+    return;
+  for (int i = 0; i < structure->num_decode_targets; ++i) {
+    uint32_t protected_by_chain =
+        buffer_.ReadNonSymmetric(structure->num_chains);
+    structure->decode_target_protected_by_chain.push_back(protected_by_chain);
+  }
+  for (FrameDependencyTemplate& frame_template : structure->templates) {
+    for (int chain_id = 0; chain_id < structure->num_chains; ++chain_id) {
+      frame_template.chain_diffs.push_back(buffer_.ReadBits(4));
+    }
+  }
+}
+
+void RtpDependencyDescriptorReader::ReadResolutions() {
+  FrameDependencyStructure* structure = descriptor_->attached_structure.get();
+  // The way templates are bitpacked, they are always ordered by spatial_id.
+  int spatial_layers = structure->templates.back().spatial_id + 1;
+  structure->resolutions.reserve(spatial_layers);
+  for (int sid = 0; sid < spatial_layers; ++sid) {
+    uint16_t width_minus_1 = buffer_.Read<uint16_t>();
+    uint16_t height_minus_1 = buffer_.Read<uint16_t>();
+    structure->resolutions.emplace_back(width_minus_1 + 1, height_minus_1 + 1);
+  }
+}
+
+void RtpDependencyDescriptorReader::ReadMandatoryFields() {
+  descriptor_->first_packet_in_frame = buffer_.Read<bool>();
+  descriptor_->last_packet_in_frame = buffer_.Read<bool>();
+  frame_dependency_template_id_ = buffer_.ReadBits(6);
+  descriptor_->frame_number = buffer_.Read<uint16_t>();
+}
+
+void RtpDependencyDescriptorReader::ReadExtendedFields() {
+  bool template_dependency_structure_present_flag = buffer_.Read<bool>();
+  active_decode_targets_present_flag_ = buffer_.Read<bool>();
+  custom_dtis_flag_ = buffer_.Read<bool>();
+  custom_fdiffs_flag_ = buffer_.Read<bool>();
+  custom_chains_flag_ = buffer_.Read<bool>();
+  if (template_dependency_structure_present_flag) {
+    ReadTemplateDependencyStructure();
+    RTC_DCHECK(descriptor_->attached_structure);
+    descriptor_->active_decode_targets_bitmask =
+        (uint64_t{1} << descriptor_->attached_structure->num_decode_targets) -
+        1;
+  }
+}
+
+void RtpDependencyDescriptorReader::ReadFrameDependencyDefinition() {
+  size_t template_index =
+      (frame_dependency_template_id_ + DependencyDescriptor::kMaxTemplates -
+       structure_->structure_id) %
+      DependencyDescriptor::kMaxTemplates;
+
+  if (template_index >= structure_->templates.size()) {
+    buffer_.Invalidate();
+    return;
+  }
+
+  // Copy all the fields from the matching template
+  descriptor_->frame_dependencies = structure_->templates[template_index];
+
+  if (custom_dtis_flag_)
+    ReadFrameDtis();
+  if (custom_fdiffs_flag_)
+    ReadFrameFdiffs();
+  if (custom_chains_flag_)
+    ReadFrameChains();
+
+  if (structure_->resolutions.empty()) {
+    descriptor_->resolution = absl::nullopt;
+  } else {
+    // Format guarantees that if there were resolutions in the last structure,
+    // then each spatial layer got one.
+    RTC_DCHECK_LE(descriptor_->frame_dependencies.spatial_id,
+                  structure_->resolutions.size());
+    descriptor_->resolution =
+        structure_->resolutions[descriptor_->frame_dependencies.spatial_id];
+  }
+}
+
+void RtpDependencyDescriptorReader::ReadFrameDtis() {
+  RTC_DCHECK_EQ(
+      descriptor_->frame_dependencies.decode_target_indications.size(),
+      structure_->num_decode_targets);
+  for (auto& dti : descriptor_->frame_dependencies.decode_target_indications) {
+    dti = static_cast<DecodeTargetIndication>(buffer_.ReadBits(2));
+  }
+}
+
+void RtpDependencyDescriptorReader::ReadFrameFdiffs() {
+  descriptor_->frame_dependencies.frame_diffs.clear();
+  for (uint64_t next_fdiff_size = buffer_.ReadBits(2); next_fdiff_size > 0;
+       next_fdiff_size = buffer_.ReadBits(2)) {
+    uint64_t fdiff_minus_one = buffer_.ReadBits(4 * next_fdiff_size);
+    descriptor_->frame_dependencies.frame_diffs.push_back(fdiff_minus_one + 1);
+  }
+}
+
+void RtpDependencyDescriptorReader::ReadFrameChains() {
+  RTC_DCHECK_EQ(descriptor_->frame_dependencies.chain_diffs.size(),
+                structure_->num_chains);
+  for (auto& chain_diff : descriptor_->frame_dependencies.chain_diffs) {
+    chain_diff = buffer_.Read<uint8_t>();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h
new file mode 100644
index 0000000000..f79d3d1d07
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_reader.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_READER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_READER_H_
+
+#include <cstdint>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "rtc_base/bitstream_reader.h"
+
+namespace webrtc {
+// Deserializes DependencyDescriptor rtp header extension.
+class RtpDependencyDescriptorReader {
+ public:
+  // Parses the dependency descriptor.
+  RtpDependencyDescriptorReader(rtc::ArrayView<const uint8_t> raw_data,
+                                const FrameDependencyStructure* structure,
+                                DependencyDescriptor* descriptor);
+  RtpDependencyDescriptorReader(const RtpDependencyDescriptorReader&) = delete;
+  RtpDependencyDescriptorReader& operator=(
+      const RtpDependencyDescriptorReader&) = delete;
+
+  // Returns true if parse was successful.
+  bool ParseSuccessful() { return buffer_.Ok(); }
+
+ private:
+  // Functions to read template dependency structure.
+  void ReadTemplateDependencyStructure();
+  void ReadTemplateLayers();
+  void ReadTemplateDtis();
+  void ReadTemplateFdiffs();
+  void ReadTemplateChains();
+  void ReadResolutions();
+
+  // Function to read details for the current frame.
+  void ReadMandatoryFields();
+  void ReadExtendedFields();
+  void ReadFrameDependencyDefinition();
+
+  void ReadFrameDtis();
+  void ReadFrameFdiffs();
+  void ReadFrameChains();
+
+  // Output.
+  DependencyDescriptor* const descriptor_;
+  // Values that are needed while reading the descriptor, but can be discarded
+  // when reading is complete.
+  BitstreamReader buffer_;
+  int frame_dependency_template_id_ = 0;
+  bool active_decode_targets_present_flag_ = false;
+  bool custom_dtis_flag_ = false;
+  bool custom_fdiffs_flag_ = false;
+  bool custom_chains_flag_ = false;
+  const FrameDependencyStructure* structure_ = nullptr;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_READER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.cc
new file mode 100644
index 0000000000..31df783064
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.cc
@@ -0,0 +1,396 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h"
+
+#include <bitset>
+#include <cstddef>
+#include <cstdint>
+#include <iterator>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "rtc_base/bit_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+enum class NextLayerIdc : uint64_t {
+  kSameLayer = 0,
+  kNextTemporal = 1,
+  kNewSpatial = 2,
+  kNoMoreLayers = 3,
+  kInvalid = 4
+};
+
+NextLayerIdc GetNextLayerIdc(const FrameDependencyTemplate& previous,
+                             const FrameDependencyTemplate& next) {
+  RTC_DCHECK_LT(next.spatial_id, DependencyDescriptor::kMaxSpatialIds);
+  RTC_DCHECK_LT(next.temporal_id, DependencyDescriptor::kMaxTemporalIds);
+
+  if (next.spatial_id == previous.spatial_id &&
+      next.temporal_id == previous.temporal_id) {
+    return NextLayerIdc::kSameLayer;
+  } else if (next.spatial_id == previous.spatial_id &&
+             next.temporal_id == previous.temporal_id + 1) {
+    return NextLayerIdc::kNextTemporal;
+  } else if (next.spatial_id == previous.spatial_id + 1 &&
+             next.temporal_id == 0) {
+    return NextLayerIdc::kNewSpatial;
+  }
+  // Everything else is unsupported.
+  return NextLayerIdc::kInvalid;
+}
+
+}  // namespace
+
+RtpDependencyDescriptorWriter::RtpDependencyDescriptorWriter(
+    rtc::ArrayView<uint8_t> data,
+    const FrameDependencyStructure& structure,
+    std::bitset<32> active_chains,
+    const DependencyDescriptor& descriptor)
+    : descriptor_(descriptor),
+      structure_(structure),
+      active_chains_(active_chains),
+      bit_writer_(data.data(), data.size()) {
+  FindBestTemplate();
+}
+
+bool RtpDependencyDescriptorWriter::Write() {
+  if (build_failed_) {
+    return false;
+  }
+  WriteMandatoryFields();
+  if (HasExtendedFields()) {
+    WriteExtendedFields();
+    WriteFrameDependencyDefinition();
+  }
+  size_t remaining_bits = bit_writer_.RemainingBitCount();
+  // Zero remaining memory to avoid leaving it uninitialized.
+  if (remaining_bits % 64 != 0) {
+    WriteBits(/*val=*/0, remaining_bits % 64);
+  }
+  for (size_t i = 0; i < remaining_bits / 64; ++i) {
+    WriteBits(/*val=*/0, 64);
+  }
+  return !build_failed_;
+}
+
+int RtpDependencyDescriptorWriter::ValueSizeBits() const {
+  if (build_failed_) {
+    return 0;
+  }
+  static constexpr int kMandatoryFields = 1 + 1 + 6 + 16;
+  int value_size_bits = kMandatoryFields + best_template_.extra_size_bits;
+  if (HasExtendedFields()) {
+    value_size_bits += 5;
+    if (descriptor_.attached_structure)
+      value_size_bits += StructureSizeBits();
+    if (ShouldWriteActiveDecodeTargetsBitmask())
+      value_size_bits += structure_.num_decode_targets;
+  }
+  return value_size_bits;
+}
+
+int RtpDependencyDescriptorWriter::StructureSizeBits() const {
+  // template_id offset (6 bits) and number of decode targets (5 bits)
+  int bits = 11;
+  // template layers.
+  bits += 2 * structure_.templates.size();
+  // dtis.
+  bits += 2 * structure_.templates.size() * structure_.num_decode_targets;
+  // fdiffs. each templates uses 1 + 5 * sizeof(fdiff) bits.
+  bits += structure_.templates.size();
+  for (const FrameDependencyTemplate& frame_template : structure_.templates) {
+    bits += 5 * frame_template.frame_diffs.size();
+  }
+  bits += rtc::BitBufferWriter::SizeNonSymmetricBits(
+      structure_.num_chains, structure_.num_decode_targets + 1);
+  if (structure_.num_chains > 0) {
+    for (int protected_by : structure_.decode_target_protected_by_chain) {
+      bits += rtc::BitBufferWriter::SizeNonSymmetricBits(protected_by,
+                                                         structure_.num_chains);
+    }
+    bits += 4 * structure_.templates.size() * structure_.num_chains;
+  }
+  // Resolutions.
+  bits += 1 + 32 * structure_.resolutions.size();
+  return bits;
+}
+
+RtpDependencyDescriptorWriter::TemplateMatch
+RtpDependencyDescriptorWriter::CalculateMatch(
+    TemplateIterator frame_template) const {
+  TemplateMatch result;
+  result.template_position = frame_template;
+  result.need_custom_fdiffs =
+      descriptor_.frame_dependencies.frame_diffs != frame_template->frame_diffs;
+  result.need_custom_dtis =
+      descriptor_.frame_dependencies.decode_target_indications !=
+      frame_template->decode_target_indications;
+  result.need_custom_chains = false;
+  for (int i = 0; i < structure_.num_chains; ++i) {
+    if (active_chains_[i] && descriptor_.frame_dependencies.chain_diffs[i] !=
+                                 frame_template->chain_diffs[i]) {
+      result.need_custom_chains = true;
+      break;
+    }
+  }
+
+  result.extra_size_bits = 0;
+  if (result.need_custom_fdiffs) {
+    result.extra_size_bits +=
+        2 * (1 + descriptor_.frame_dependencies.frame_diffs.size());
+    for (int fdiff : descriptor_.frame_dependencies.frame_diffs) {
+      if (fdiff <= (1 << 4))
+        result.extra_size_bits += 4;
+      else if (fdiff <= (1 << 8))
+        result.extra_size_bits += 8;
+      else
+        result.extra_size_bits += 12;
+    }
+  }
+  if (result.need_custom_dtis) {
+    result.extra_size_bits +=
+        2 * descriptor_.frame_dependencies.decode_target_indications.size();
+  }
+  if (result.need_custom_chains)
+    result.extra_size_bits += 8 * structure_.num_chains;
+  return result;
+}
+
+void RtpDependencyDescriptorWriter::FindBestTemplate() {
+  const std::vector<FrameDependencyTemplate>& templates = structure_.templates;
+  // Find range of templates with matching spatial/temporal id.
+  auto same_layer = [&](const FrameDependencyTemplate& frame_template) {
+    return descriptor_.frame_dependencies.spatial_id ==
+               frame_template.spatial_id &&
+           descriptor_.frame_dependencies.temporal_id ==
+               frame_template.temporal_id;
+  };
+  auto first = absl::c_find_if(templates, same_layer);
+  if (first == templates.end()) {
+    build_failed_ = true;
+    return;
+  }
+  auto last = std::find_if_not(first, templates.end(), same_layer);
+
+  best_template_ = CalculateMatch(first);
+  // Search if there any better template than the first one.
+  for (auto next = std::next(first); next != last; ++next) {
+    TemplateMatch match = CalculateMatch(next);
+    if (match.extra_size_bits < best_template_.extra_size_bits)
+      best_template_ = match;
+  }
+}
+
+bool RtpDependencyDescriptorWriter::ShouldWriteActiveDecodeTargetsBitmask()
+    const {
+  if (!descriptor_.active_decode_targets_bitmask)
+    return false;
+  const uint64_t all_decode_targets_bitmask =
+      (uint64_t{1} << structure_.num_decode_targets) - 1;
+  if (descriptor_.attached_structure &&
+      descriptor_.active_decode_targets_bitmask == all_decode_targets_bitmask)
+    return false;
+  return true;
+}
+
+bool RtpDependencyDescriptorWriter::HasExtendedFields() const {
+  return best_template_.extra_size_bits > 0 || descriptor_.attached_structure ||
+         descriptor_.active_decode_targets_bitmask;
+}
+
+uint64_t RtpDependencyDescriptorWriter::TemplateId() const {
+  return (best_template_.template_position - structure_.templates.begin() +
+          structure_.structure_id) %
+         DependencyDescriptor::kMaxTemplates;
+}
+
+void RtpDependencyDescriptorWriter::WriteBits(uint64_t val, size_t bit_count) {
+  if (!bit_writer_.WriteBits(val, bit_count))
+    build_failed_ = true;
+}
+
+void RtpDependencyDescriptorWriter::WriteNonSymmetric(uint32_t value,
+                                                      uint32_t num_values) {
+  if (!bit_writer_.WriteNonSymmetric(value, num_values))
+    build_failed_ = true;
+}
+
+void RtpDependencyDescriptorWriter::WriteTemplateDependencyStructure() {
+  RTC_DCHECK_GE(structure_.structure_id, 0);
+  RTC_DCHECK_LT(structure_.structure_id, DependencyDescriptor::kMaxTemplates);
+  RTC_DCHECK_GT(structure_.num_decode_targets, 0);
+  RTC_DCHECK_LE(structure_.num_decode_targets,
+                DependencyDescriptor::kMaxDecodeTargets);
+
+  WriteBits(structure_.structure_id, 6);
+  WriteBits(structure_.num_decode_targets - 1, 5);
+  WriteTemplateLayers();
+  WriteTemplateDtis();
+  WriteTemplateFdiffs();
+  WriteTemplateChains();
+  uint64_t has_resolutions = structure_.resolutions.empty() ? 0 : 1;
+  WriteBits(has_resolutions, 1);
+  if (has_resolutions)
+    WriteResolutions();
+}
+
+void RtpDependencyDescriptorWriter::WriteTemplateLayers() {
+  const auto& templates = structure_.templates;
+  RTC_DCHECK(!templates.empty());
+  RTC_DCHECK_LE(templates.size(), DependencyDescriptor::kMaxTemplates);
+  RTC_DCHECK_EQ(templates[0].spatial_id, 0);
+  RTC_DCHECK_EQ(templates[0].temporal_id, 0);
+
+  for (size_t i = 1; i < templates.size(); ++i) {
+    uint64_t next_layer_idc =
+        static_cast<uint64_t>(GetNextLayerIdc(templates[i - 1], templates[i]));
+    RTC_DCHECK_LE(next_layer_idc, 3);
+    WriteBits(next_layer_idc, 2);
+  }
+  WriteBits(static_cast<uint64_t>(NextLayerIdc::kNoMoreLayers), 2);
+}
+
+void RtpDependencyDescriptorWriter::WriteTemplateDtis() {
+  for (const FrameDependencyTemplate& current_template : structure_.templates) {
+    RTC_DCHECK_EQ(current_template.decode_target_indications.size(),
+                  structure_.num_decode_targets);
+    for (DecodeTargetIndication dti :
+         current_template.decode_target_indications) {
+      WriteBits(static_cast<uint32_t>(dti), 2);
+    }
+  }
+}
+
+void RtpDependencyDescriptorWriter::WriteTemplateFdiffs() {
+  for (const FrameDependencyTemplate& current_template : structure_.templates) {
+    for (int fdiff : current_template.frame_diffs) {
+      RTC_DCHECK_GE(fdiff - 1, 0);
+      RTC_DCHECK_LT(fdiff - 1, 1 << 4);
+      WriteBits((1u << 4) | (fdiff - 1), 1 + 4);
+    }
+    // No more diffs for current template.
+    WriteBits(/*val=*/0, /*bit_count=*/1);
+  }
+}
+
+void RtpDependencyDescriptorWriter::WriteTemplateChains() {
+  RTC_DCHECK_GE(structure_.num_chains, 0);
+  RTC_DCHECK_LE(structure_.num_chains, structure_.num_decode_targets);
+
+  WriteNonSymmetric(structure_.num_chains, structure_.num_decode_targets + 1);
+  if (structure_.num_chains == 0)
+    return;
+
+  RTC_DCHECK_EQ(structure_.decode_target_protected_by_chain.size(),
+                structure_.num_decode_targets);
+  for (int protected_by : structure_.decode_target_protected_by_chain) {
+    RTC_DCHECK_GE(protected_by, 0);
+    RTC_DCHECK_LT(protected_by, structure_.num_chains);
+    WriteNonSymmetric(protected_by, structure_.num_chains);
+  }
+  for (const auto& frame_template : structure_.templates) {
+    RTC_DCHECK_EQ(frame_template.chain_diffs.size(), structure_.num_chains);
+    for (int chain_diff : frame_template.chain_diffs) {
+      RTC_DCHECK_GE(chain_diff, 0);
+      RTC_DCHECK_LT(chain_diff, 1 << 4);
+      WriteBits(chain_diff, 4);
+    }
+  }
+}
+
+void RtpDependencyDescriptorWriter::WriteResolutions() {
+  int max_spatial_id = structure_.templates.back().spatial_id;
+  RTC_DCHECK_EQ(structure_.resolutions.size(), max_spatial_id + 1);
+  for (const RenderResolution& resolution : structure_.resolutions) {
+    RTC_DCHECK_GT(resolution.Width(), 0);
+    RTC_DCHECK_LE(resolution.Width(), 1 << 16);
+    RTC_DCHECK_GT(resolution.Height(), 0);
+    RTC_DCHECK_LE(resolution.Height(), 1 << 16);
+
+    WriteBits(resolution.Width() - 1, 16);
+    WriteBits(resolution.Height() - 1, 16);
+  }
+}
+
+void RtpDependencyDescriptorWriter::WriteMandatoryFields() {
+  WriteBits(descriptor_.first_packet_in_frame, 1);
+  WriteBits(descriptor_.last_packet_in_frame, 1);
+  WriteBits(TemplateId(), 6);
+  WriteBits(descriptor_.frame_number, 16);
+}
+
+void RtpDependencyDescriptorWriter::WriteExtendedFields() {
+  uint64_t template_dependency_structure_present_flag =
+      descriptor_.attached_structure ? 1u : 0u;
+  WriteBits(template_dependency_structure_present_flag, 1);
+  uint64_t active_decode_targets_present_flag =
+      ShouldWriteActiveDecodeTargetsBitmask() ? 1u : 0u;
+  WriteBits(active_decode_targets_present_flag, 1);
+  WriteBits(best_template_.need_custom_dtis, 1);
+  WriteBits(best_template_.need_custom_fdiffs, 1);
+  WriteBits(best_template_.need_custom_chains, 1);
+  if (template_dependency_structure_present_flag)
+    WriteTemplateDependencyStructure();
+  if (active_decode_targets_present_flag)
+    WriteBits(*descriptor_.active_decode_targets_bitmask,
+              structure_.num_decode_targets);
+}
+
+void RtpDependencyDescriptorWriter::WriteFrameDependencyDefinition() {
+  if (best_template_.need_custom_dtis)
+    WriteFrameDtis();
+  if (best_template_.need_custom_fdiffs)
+    WriteFrameFdiffs();
+  if (best_template_.need_custom_chains)
+    WriteFrameChains();
+}
+
+void RtpDependencyDescriptorWriter::WriteFrameDtis() {
+  RTC_DCHECK_EQ(descriptor_.frame_dependencies.decode_target_indications.size(),
+                structure_.num_decode_targets);
+  for (DecodeTargetIndication dti :
+       descriptor_.frame_dependencies.decode_target_indications) {
+    WriteBits(static_cast<uint32_t>(dti), 2);
+  }
+}
+
+void RtpDependencyDescriptorWriter::WriteFrameFdiffs() {
+  for (int fdiff : descriptor_.frame_dependencies.frame_diffs) {
+    RTC_DCHECK_GT(fdiff, 0);
+    RTC_DCHECK_LE(fdiff, 1 << 12);
+    if (fdiff <= (1 << 4))
+      WriteBits((1u << 4) | (fdiff - 1), 2 + 4);
+    else if (fdiff <= (1 << 8))
+      WriteBits((2u << 8) | (fdiff - 1), 2 + 8);
+    else  // fdiff <= (1 << 12)
+      WriteBits((3u << 12) | (fdiff - 1), 2 + 12);
+  }
+  // No more diffs.
+  WriteBits(/*val=*/0, /*bit_count=*/2);
+}
+
+void RtpDependencyDescriptorWriter::WriteFrameChains() {
+  RTC_DCHECK_EQ(descriptor_.frame_dependencies.chain_diffs.size(),
+                structure_.num_chains);
+  for (int i = 0; i < structure_.num_chains; ++i) {
+    int chain_diff =
+        active_chains_[i] ? descriptor_.frame_dependencies.chain_diffs[i] : 0;
+    RTC_DCHECK_GE(chain_diff, 0);
+    RTC_DCHECK_LT(chain_diff, 1 << 8);
+    WriteBits(chain_diff, 8);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h
new file mode 100644
index 0000000000..568e0a8aab
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_writer.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_WRITER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_WRITER_H_
+
+#include <bitset>
+#include <cstddef>
+#include <cstdint>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "rtc_base/bit_buffer.h"
+
+namespace webrtc {
+class RtpDependencyDescriptorWriter {
+ public:
+  // Assumes `structure` and `descriptor` are valid and
+  // `descriptor` matches the `structure`.
+  RtpDependencyDescriptorWriter(rtc::ArrayView<uint8_t> data,
+                                const FrameDependencyStructure& structure,
+                                std::bitset<32> active_chains,
+                                const DependencyDescriptor& descriptor);
+
+  // Serializes DependencyDescriptor rtp header extension.
+  // Returns false if `data` is too small to serialize the `descriptor`.
+  bool Write();
+
+  // Returns minimum number of bits needed to serialize descriptor with respect
+  // to the `structure`. Returns 0 if `descriptor` can't be serialized.
+  int ValueSizeBits() const;
+
+ private:
+  // Used both as pointer to the template and as index in the templates vector.
+  using TemplateIterator = std::vector<FrameDependencyTemplate>::const_iterator;
+  struct TemplateMatch {
+    TemplateIterator template_position;
+    bool need_custom_dtis;
+    bool need_custom_fdiffs;
+    bool need_custom_chains;
+    // Size in bits to store frame-specific details, i.e.
+    // excluding mandatory fields and template dependency structure.
+    int extra_size_bits;
+  };
+  int StructureSizeBits() const;
+  TemplateMatch CalculateMatch(TemplateIterator frame_template) const;
+  void FindBestTemplate();
+  bool ShouldWriteActiveDecodeTargetsBitmask() const;
+  bool HasExtendedFields() const;
+  uint64_t TemplateId() const;
+
+  void WriteBits(uint64_t val, size_t bit_count);
+  void WriteNonSymmetric(uint32_t value, uint32_t num_values);
+
+  // Functions to read template dependency structure.
+  void WriteTemplateDependencyStructure();
+  void WriteTemplateLayers();
+  void WriteTemplateDtis();
+  void WriteTemplateFdiffs();
+  void WriteTemplateChains();
+  void WriteResolutions();
+
+  // Function to read details for the current frame.
+  void WriteMandatoryFields();
+  void WriteExtendedFields();
+  void WriteFrameDependencyDefinition();
+
+  void WriteFrameDtis();
+  void WriteFrameFdiffs();
+  void WriteFrameChains();
+
+  bool build_failed_ = false;
+  const DependencyDescriptor& descriptor_;
+  const FrameDependencyStructure& structure_;
+  std::bitset<32> active_chains_;
+  rtc::BitBufferWriter bit_writer_;
+  TemplateMatch best_template_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_WRITER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_descriptor_authentication.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_descriptor_authentication.cc
new file mode 100644
index 0000000000..f4525f0db1
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_descriptor_authentication.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
+
+#include <cstdint>
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+
+namespace webrtc {
+
+std::vector<uint8_t> RtpDescriptorAuthentication(
+    const RTPVideoHeader& rtp_video_header) {
+  if (!rtp_video_header.generic) {
+    return {};
+  }
+  const RTPVideoHeader::GenericDescriptorInfo& descriptor =
+      *rtp_video_header.generic;
+  // Default way of creating additional data for an encrypted frame.
+  if (descriptor.spatial_index < 0 || descriptor.temporal_index < 0 ||
+      descriptor.spatial_index >=
+          RtpGenericFrameDescriptor::kMaxSpatialLayers ||
+      descriptor.temporal_index >=
+          RtpGenericFrameDescriptor::kMaxTemporalLayers ||
+      descriptor.dependencies.size() >
+          RtpGenericFrameDescriptor::kMaxNumFrameDependencies) {
+    return {};
+  }
+  RtpGenericFrameDescriptor frame_descriptor;
+  frame_descriptor.SetFirstPacketInSubFrame(true);
+  frame_descriptor.SetLastPacketInSubFrame(false);
+  frame_descriptor.SetTemporalLayer(descriptor.temporal_index);
+  frame_descriptor.SetSpatialLayersBitmask(1 << descriptor.spatial_index);
+  frame_descriptor.SetFrameId(descriptor.frame_id & 0xFFFF);
+  for (int64_t dependency : descriptor.dependencies) {
+    frame_descriptor.AddFrameDependencyDiff(descriptor.frame_id - dependency);
+  }
+  if (descriptor.dependencies.empty()) {
+    frame_descriptor.SetResolution(rtp_video_header.width,
+                                   rtp_video_header.height);
+  }
+  std::vector<uint8_t> result(
+      RtpGenericFrameDescriptorExtension00::ValueSize(frame_descriptor));
+  RtpGenericFrameDescriptorExtension00::Write(result, frame_descriptor);
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_descriptor_authentication.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_descriptor_authentication.h
new file mode 100644
index 0000000000..1791abecd8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_descriptor_authentication.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_DESCRIPTOR_AUTHENTICATION_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_DESCRIPTOR_AUTHENTICATION_H_
+
+#include <cstdint>
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+
+namespace webrtc {
+
+// Converts frame dependencies into array of bytes for authentication.
+std::vector<uint8_t> RtpDescriptorAuthentication(
+    const RTPVideoHeader& rtp_video_header);
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_DESCRIPTOR_AUTHENTICATION_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
new file mode 100644
index 0000000000..2c01a0d40a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
@@ -0,0 +1,1129 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <list>
+#include <memory>
+
+#include "absl/algorithm/container.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/fec_test_helper.h"
+#include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
+#include "rtc_base/random.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// Transport header size in bytes. Assume UDP/IPv4 as a reasonable minimum.
+constexpr size_t kTransportOverhead = 28;
+
+constexpr uint32_t kMediaSsrc = 83542;
+constexpr uint32_t kFlexfecSsrc = 43245;
+
+constexpr size_t kMaxMediaPackets = 48;
+
+// Deep copies `src` to `dst`, but only keeps every Nth packet.
+void DeepCopyEveryNthPacket(const ForwardErrorCorrection::PacketList& src,
+                            int n,
+                            ForwardErrorCorrection::PacketList* dst) {
+  RTC_DCHECK_GT(n, 0);
+  int i = 0;
+  for (const auto& packet : src) {
+    if (i % n == 0) {
+      dst->emplace_back(new ForwardErrorCorrection::Packet(*packet));
+    }
+    ++i;
+  }
+}
+
+}  // namespace
+
+using ::testing::Types;
+
+template <typename ForwardErrorCorrectionType>
+class RtpFecTest : public ::testing::Test {
+ protected:
+  RtpFecTest()
+      : random_(0xabcdef123456),
+        media_packet_generator_(
+            kRtpHeaderSize,  // Minimum packet size.
+            IP_PACKET_SIZE - kRtpHeaderSize - kTransportOverhead -
+                fec_.MaxPacketOverhead(),  // Maximum packet size.
+            kMediaSsrc,
+            &random_) {}
+
+  // Construct `received_packets_`: a subset of the media and FEC packets.
+  //
+  // Media packet "i" is lost if media_loss_mask_[i] = 1, received if
+  // media_loss_mask_[i] = 0.
+  // FEC packet "i" is lost if fec_loss_mask_[i] = 1, received if
+  // fec_loss_mask_[i] = 0.
+  void NetworkReceivedPackets(int* media_loss_mask, int* fec_loss_mask);
+
+  // Add packet from `packet_list` to list of received packets, using the
+  // `loss_mask`.
+  // The `packet_list` may be a media packet list (is_fec = false), or a
+  // FEC packet list (is_fec = true).
+  template <typename T>
+  void ReceivedPackets(const T& packet_list, int* loss_mask, bool is_fec);
+
+  // Check for complete recovery after FEC decoding.
+  bool IsRecoveryComplete();
+
+  ForwardErrorCorrectionType fec_;
+
+  Random random_;
+  test::fec::MediaPacketGenerator media_packet_generator_;
+
+  ForwardErrorCorrection::PacketList media_packets_;
+  std::list<ForwardErrorCorrection::Packet*> generated_fec_packets_;
+  std::vector<std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>>
+      received_packets_;
+  ForwardErrorCorrection::RecoveredPacketList recovered_packets_;
+
+  int media_loss_mask_[kUlpfecMaxMediaPackets];
+  int fec_loss_mask_[kUlpfecMaxMediaPackets];
+};
+
+template <typename ForwardErrorCorrectionType>
+void RtpFecTest<ForwardErrorCorrectionType>::NetworkReceivedPackets(
+    int* media_loss_mask,
+    int* fec_loss_mask) {
+  constexpr bool kFecPacket = true;
+  this->received_packets_.clear();
+  ReceivedPackets(media_packets_, media_loss_mask, !kFecPacket);
+  ReceivedPackets(generated_fec_packets_, fec_loss_mask, kFecPacket);
+}
+
+template <typename ForwardErrorCorrectionType>
+template <typename PacketListType>
+void RtpFecTest<ForwardErrorCorrectionType>::ReceivedPackets(
+    const PacketListType& packet_list,
+    int* loss_mask,
+    bool is_fec) {
+  uint16_t fec_seq_num = ForwardErrorCorrectionType::GetFirstFecSeqNum(
+      media_packet_generator_.GetNextSeqNum());
+  int packet_idx = 0;
+
+  for (const auto& packet : packet_list) {
+    if (loss_mask[packet_idx] == 0) {
+      std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet(
+          new ForwardErrorCorrection::ReceivedPacket());
+      received_packet->pkt = new ForwardErrorCorrection::Packet();
+      received_packet->pkt->data = packet->data;
+      received_packet->is_fec = is_fec;
+      if (!is_fec) {
+        received_packet->ssrc = kMediaSsrc;
+        // For media packets, the sequence number is obtained from the
+        // RTP header as written by MediaPacketGenerator::ConstructMediaPackets.
+        received_packet->seq_num =
+            ByteReader<uint16_t>::ReadBigEndian(packet->data.data() + 2);
+      } else {
+        received_packet->ssrc = ForwardErrorCorrectionType::kFecSsrc;
+        // For FEC packets, we simulate the sequence numbers differently
+        // depending on if ULPFEC or FlexFEC is used. See the definition of
+        // ForwardErrorCorrectionType::GetFirstFecSeqNum.
+        received_packet->seq_num = fec_seq_num;
+      }
+      received_packets_.push_back(std::move(received_packet));
+    }
+    packet_idx++;
+    // Sequence number of FEC packets are defined as increment by 1 from
+    // last media packet in frame.
+    if (is_fec)
+      fec_seq_num++;
+  }
+}
+
+template <typename ForwardErrorCorrectionType>
+bool RtpFecTest<ForwardErrorCorrectionType>::IsRecoveryComplete() {
+  // We must have equally many recovered packets as original packets and all
+  // recovered packets must be identical to the corresponding original packets.
+  return absl::c_equal(
+      media_packets_, recovered_packets_,
+      [](const std::unique_ptr<ForwardErrorCorrection::Packet>& media_packet,
+         const std::unique_ptr<ForwardErrorCorrection::RecoveredPacket>&
+             recovered_packet) {
+        if (media_packet->data.size() != recovered_packet->pkt->data.size()) {
+          return false;
+        }
+        if (memcmp(media_packet->data.cdata(),
+                   recovered_packet->pkt->data.cdata(),
+                   media_packet->data.size()) != 0) {
+          return false;
+        }
+        return true;
+      });
+}
+
+// Define gTest typed test to loop over both ULPFEC and FlexFEC.
+// Since the tests now are parameterized, we need to access
+// member variables using `this`, thereby enforcing runtime
+// resolution.
+
+class FlexfecForwardErrorCorrection : public ForwardErrorCorrection {
+ public:
+  static const uint32_t kFecSsrc = kFlexfecSsrc;
+
+  FlexfecForwardErrorCorrection()
+      : ForwardErrorCorrection(
+            std::unique_ptr<FecHeaderReader>(new FlexfecHeaderReader()),
+            std::unique_ptr<FecHeaderWriter>(new FlexfecHeaderWriter()),
+            kFecSsrc,
+            kMediaSsrc) {}
+
+  // For FlexFEC we let the FEC packet sequence numbers be independent of
+  // the media packet sequence numbers.
+  static uint16_t GetFirstFecSeqNum(uint16_t next_media_seq_num) {
+    Random random(0xbe110);
+    return random.Rand<uint16_t>();
+  }
+};
+
+class UlpfecForwardErrorCorrection : public ForwardErrorCorrection {
+ public:
+  static const uint32_t kFecSsrc = kMediaSsrc;
+
+  UlpfecForwardErrorCorrection()
+      : ForwardErrorCorrection(
+            std::unique_ptr<FecHeaderReader>(new UlpfecHeaderReader()),
+            std::unique_ptr<FecHeaderWriter>(new UlpfecHeaderWriter()),
+            kFecSsrc,
+            kMediaSsrc) {}
+
+  // For ULPFEC we assume that the FEC packets are subsequent to the media
+  // packets in terms of sequence number.
+  static uint16_t GetFirstFecSeqNum(uint16_t next_media_seq_num) {
+    return next_media_seq_num;
+  }
+};
+
+using FecTypes =
+    Types<FlexfecForwardErrorCorrection, UlpfecForwardErrorCorrection>;
+TYPED_TEST_SUITE(RtpFecTest, FecTypes);
+
+TYPED_TEST(RtpFecTest, WillProtectMediaPacketsWithLargeSequenceNumberGap) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr int kNumMediaPackets = 2;
+  constexpr uint8_t kProtectionFactor = 127;
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  // Create |kMaxMediaPackets - 1| sequence number difference.
+  ByteWriter<uint16_t>::WriteBigEndian(
+      this->media_packets_.front()->data.MutableData() + 2, 1);
+  ByteWriter<uint16_t>::WriteBigEndian(
+      this->media_packets_.back()->data.MutableData() + 2, kMaxMediaPackets);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+}
+
+TYPED_TEST(RtpFecTest,
+           WillNotProtectMediaPacketsWithTooLargeSequenceNumberGap) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr int kNumMediaPackets = 2;
+  constexpr uint8_t kProtectionFactor = 127;
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  // Create `kMaxMediaPackets` sequence number difference.
+  ByteWriter<uint16_t>::WriteBigEndian(
+      this->media_packets_.front()->data.MutableData() + 2, 1);
+  ByteWriter<uint16_t>::WriteBigEndian(
+      this->media_packets_.back()->data.MutableData() + 2,
+      kMaxMediaPackets + 1);
+
+  EXPECT_EQ(
+      -1, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                               kNumImportantPackets, kUseUnequalProtection,
+                               kFecMaskBursty, &this->generated_fec_packets_));
+  EXPECT_TRUE(this->generated_fec_packets_.empty());
+}
+
+TYPED_TEST(RtpFecTest, FecRecoveryNoLoss) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr int kNumMediaPackets = 4;
+  constexpr uint8_t kProtectionFactor = 60;
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 1 FEC packet.
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+
+  // No packets lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // No packets lost, expect complete recovery.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+}
+
+TYPED_TEST(RtpFecTest, FecRecoveryWithLoss) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr int kNumMediaPackets = 4;
+  constexpr uint8_t kProtectionFactor = 60;
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 1 FEC packet.
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+
+  // 1 media packet lost
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // One packet lost, one FEC packet, expect complete recovery.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // 2 media packets lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[1] = 1;
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // 2 packets lost, one FEC packet, cannot get complete recovery.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+}
+
+// Verify that we don't use an old FEC packet for FEC decoding.
+TYPED_TEST(RtpFecTest, NoFecRecoveryWithOldFecPacket) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr uint8_t kProtectionFactor = 20;
+
+  // Two frames: first frame (old) with two media packets and 1 FEC packet.
+  // Third frame (new) with 3 media packets, and no FEC packets.
+  //
+  //  #0(media) #1(media) #2(FEC)              ----Frame 1-----
+  //  #32767(media) 32768(media) 32769(media)  ----Frame 2-----
+  //  #65535(media) #0(media) #1(media).       ----Frame 3-----
+  // If we lose either packet 0 or 1 of third frame, FEC decoding should not
+  // try to decode using "old" FEC packet #2.
+
+  // Construct media packets for first frame, starting at sequence number 0.
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(2, 0);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+  // Expect 1 FEC packet.
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+  // Add FEC packet (seq#2) of this first frame to received list (i.e., assume
+  // the two media packet were lost).
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->ReceivedPackets(this->generated_fec_packets_, this->fec_loss_mask_,
+                        true);
+
+  // Construct media packets for second frame, with sequence number wrap.
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(3, 32767);
+
+  // Expect 3 media packets for this frame.
+  EXPECT_EQ(3u, this->media_packets_.size());
+
+  // No packets lost
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  this->ReceivedPackets(this->media_packets_, this->media_loss_mask_, false);
+
+  // Construct media packets for third frame, with sequence number wrap.
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(3, 65535);
+
+  // Expect 3 media packets for this frame.
+  EXPECT_EQ(3u, this->media_packets_.size());
+
+  // Second media packet lost (seq#0).
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  this->media_loss_mask_[1] = 1;
+  // Add packets #65535, and #1 to received list.
+  this->ReceivedPackets(this->media_packets_, this->media_loss_mask_, false);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Expect that no decoding is done to get missing packet (seq#0) of third
+  // frame, using old FEC packet (seq#2) from first (old) frame. So number of
+  // recovered packets is 5 (0 from first frame, three from second frame, and 2
+  // for the third frame, with no packets recovered via FEC).
+  EXPECT_EQ(5u, this->recovered_packets_.size());
+  EXPECT_TRUE(this->recovered_packets_.size() != this->media_packets_.size());
+}
+
+// Verify we can still recover frame if sequence number wrap occurs within
+// the frame and FEC packet following wrap is received after media packets.
+TYPED_TEST(RtpFecTest, FecRecoveryWithSeqNumGapOneFrameRecovery) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr uint8_t kProtectionFactor = 20;
+
+  // One frame, with sequence number wrap in media packets.
+  //         -----Frame 1----
+  //  #65534(media) #65535(media) #0(media) #1(FEC).
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(3, 65534);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 1 FEC packet.
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+
+  // Lose one media packet (seq# 65535).
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[1] = 1;
+  this->ReceivedPackets(this->media_packets_, this->media_loss_mask_, false);
+  // Add FEC packet to received list following the media packets.
+  this->ReceivedPackets(this->generated_fec_packets_, this->fec_loss_mask_,
+                        true);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Expect 3 media packets in recovered list, and complete recovery.
+  // Wrap-around won't remove FEC packet, as it follows the wrap.
+  EXPECT_EQ(3u, this->recovered_packets_.size());
+  EXPECT_TRUE(this->IsRecoveryComplete());
+}
+
+// Sequence number wrap occurs within the ULPFEC packets for the frame.
+// Same problem will occur if wrap is within media packets but ULPFEC packet is
+// received before the media packets. This may be improved if timing information
+// is used to detect old ULPFEC packets.
+
+// TODO(nisse): There's some logic to discard ULPFEC packets at wrap-around,
+// however, that is not actually exercised by this test: When the first FEC
+// packet is processed, it results in full recovery of one media packet and the
+// FEC packet is forgotten. And then the wraparound isn't noticed when the next
+// FEC packet is received. We should fix wraparound handling, which currently
+// appears broken, and then figure out how to test it properly.
+using RtpFecTestUlpfecOnly = RtpFecTest<UlpfecForwardErrorCorrection>;
+TEST_F(RtpFecTestUlpfecOnly, FecRecoveryWithSeqNumGapOneFrameRecovery) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr uint8_t kProtectionFactor = 200;
+
+  // 1 frame: 3 media packets and 2 FEC packets.
+  // Sequence number wrap in FEC packets.
+  //           -----Frame 1----
+  // #65532(media) #65533(media) #65534(media) #65535(FEC) #0(FEC).
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(3, 65532);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 2 FEC packets.
+  EXPECT_EQ(2u, this->generated_fec_packets_.size());
+
+  // Lose the last two media packets (seq# 65533, 65534).
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[1] = 1;
+  this->media_loss_mask_[2] = 1;
+  this->ReceivedPackets(this->media_packets_, this->media_loss_mask_, false);
+  this->ReceivedPackets(this->generated_fec_packets_, this->fec_loss_mask_,
+                        true);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // The two FEC packets are received and should allow for complete recovery,
+  // but because of the wrap the first FEC packet will be discarded, and only
+  // one media packet is recoverable. So expect 2 media packets on recovered
+  // list and no complete recovery.
+  EXPECT_EQ(3u, this->recovered_packets_.size());
+  EXPECT_EQ(this->recovered_packets_.size(), this->media_packets_.size());
+  EXPECT_TRUE(this->IsRecoveryComplete());
+}
+
+// TODO(brandtr): This test mimics the one above, ensuring that the recovery
+// strategy of FlexFEC matches the recovery strategy of ULPFEC. Since FlexFEC
+// does not share the sequence number space with the media, however, having a
+// matching recovery strategy may be suboptimal. Study this further.
+// TODO(nisse): In this test, recovery based on the first FEC packet fails with
+// the log message "The recovered packet had a length larger than a typical IP
+// packet, and is thus dropped." This is probably not intended, and needs
+// investigation.
+using RtpFecTestFlexfecOnly = RtpFecTest<FlexfecForwardErrorCorrection>;
+TEST_F(RtpFecTestFlexfecOnly, FecRecoveryWithSeqNumGapOneFrameNoRecovery) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr uint8_t kProtectionFactor = 200;
+
+  // 1 frame: 3 media packets and 2 FEC packets.
+  // Sequence number wrap in FEC packets.
+  //           -----Frame 1----
+  // #65532(media) #65533(media) #65534(media) #65535(FEC) #0(FEC).
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(3, 65532);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 2 FEC packets.
+  EXPECT_EQ(2u, this->generated_fec_packets_.size());
+
+  // Overwrite the sequence numbers generated by ConstructMediaPackets,
+  // to make sure that we do have a wrap.
+  auto it = this->generated_fec_packets_.begin();
+  ByteWriter<uint16_t>::WriteBigEndian(&(*it)->data.MutableData()[2], 65535);
+  ++it;
+  ByteWriter<uint16_t>::WriteBigEndian(&(*it)->data.MutableData()[2], 0);
+
+  // Lose the last two media packets (seq# 65533, 65534).
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[1] = 1;
+  this->media_loss_mask_[2] = 1;
+  this->ReceivedPackets(this->media_packets_, this->media_loss_mask_, false);
+  this->ReceivedPackets(this->generated_fec_packets_, this->fec_loss_mask_,
+                        true);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // The two FEC packets are received and should allow for complete recovery,
+  // but because of the wrap the first FEC packet will be discarded, and only
+  // one media packet is recoverable. So expect 2 media packets on recovered
+  // list and no complete recovery.
+  EXPECT_EQ(2u, this->recovered_packets_.size());
+  EXPECT_TRUE(this->recovered_packets_.size() != this->media_packets_.size());
+  EXPECT_FALSE(this->IsRecoveryComplete());
+}
+
+// Verify we can still recover frame if media packets are reordered.
+TYPED_TEST(RtpFecTest, FecRecoveryWithMediaOutOfOrder) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr uint8_t kProtectionFactor = 20;
+
+  // One frame: 3 media packets, 1 FEC packet.
+  //         -----Frame 1----
+  //  #0(media) #1(media) #2(media) #3(FEC).
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(3, 0);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 1 FEC packet.
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+
+  // Lose one media packet (seq# 1).
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[1] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  // Reorder received media packets.
+  auto it0 = this->received_packets_.begin();
+  auto it1 = this->received_packets_.begin();
+  it1++;
+  std::swap(*it0, *it1);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Expect 3 media packets in recovered list, and complete recovery.
+  EXPECT_EQ(3u, this->recovered_packets_.size());
+  EXPECT_TRUE(this->IsRecoveryComplete());
+}
+
+// Verify we can still recover frame if FEC is received before media packets.
+TYPED_TEST(RtpFecTest, FecRecoveryWithFecOutOfOrder) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr uint8_t kProtectionFactor = 20;
+
+  // One frame: 3 media packets, 1 FEC packet.
+  //         -----Frame 1----
+  //  #0(media) #1(media) #2(media) #3(FEC).
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(3, 0);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 1 FEC packet.
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+
+  // Lose one media packet (seq# 1).
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[1] = 1;
+  // Add FEC packet to received list before the media packets.
+  this->ReceivedPackets(this->generated_fec_packets_, this->fec_loss_mask_,
+                        true);
+  // Add media packets to received list.
+  this->ReceivedPackets(this->media_packets_, this->media_loss_mask_, false);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Expect 3 media packets in recovered list, and complete recovery.
+  EXPECT_EQ(3u, this->recovered_packets_.size());
+  EXPECT_TRUE(this->IsRecoveryComplete());
+}
+
+// Test 50% protection with random mask type: Two cases are considered:
+// a 50% non-consecutive loss which can be fully recovered, and a 50%
+// consecutive loss which cannot be fully recovered.
+TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percRandomMask) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr int kNumMediaPackets = 4;
+  constexpr uint8_t kProtectionFactor = 255;
+
+  // Packet Mask for (4,4,0) code, from random mask table.
+  // (kNumMediaPackets = 4; num_fec_packets = 4, kNumImportantPackets = 0)
+
+  //         media#0   media#1  media#2    media#3
+  // fec#0:    1          1        0          0
+  // fec#1:    1          0        1          0
+  // fec#2:    0          0        1          1
+  // fec#3:    0          1        0          1
+  //
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskRandom, &this->generated_fec_packets_));
+
+  // Expect 4 FEC packets.
+  EXPECT_EQ(4u, this->generated_fec_packets_.size());
+
+  // 4 packets lost: 3 media packets (0, 2, 3), and one FEC packet (0) lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->fec_loss_mask_[0] = 1;
+  this->media_loss_mask_[0] = 1;
+  this->media_loss_mask_[2] = 1;
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // With media packet#1 and FEC packets #1, #2, #3, expect complete recovery.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // 4 consecutive packets lost: media packets 0, 1, 2, 3.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[0] = 1;
+  this->media_loss_mask_[1] = 1;
+  this->media_loss_mask_[2] = 1;
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Cannot get complete recovery for this loss configuration with random mask.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+}
+
+// Test 50% protection with bursty type: Three cases are considered:
+// two 50% consecutive losses which can be fully recovered, and one
+// non-consecutive which cannot be fully recovered.
+TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percBurstyMask) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr int kNumMediaPackets = 4;
+  constexpr uint8_t kProtectionFactor = 255;
+
+  // Packet Mask for (4,4,0) code, from bursty mask table.
+  // (kNumMediaPackets = 4; num_fec_packets = 4, kNumImportantPackets = 0)
+
+  //         media#0   media#1  media#2    media#3
+  // fec#0:    1          0        0          0
+  // fec#1:    1          1        0          0
+  // fec#2:    0          1        1          0
+  // fec#3:    0          0        1          1
+  //
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 4 FEC packets.
+  EXPECT_EQ(4u, this->generated_fec_packets_.size());
+
+  // 4 consecutive packets lost: media packets 0,1,2,3.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[0] = 1;
+  this->media_loss_mask_[1] = 1;
+  this->media_loss_mask_[2] = 1;
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Expect complete recovery for consecutive packet loss <= 50%.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // 4 consecutive packets lost: media packets 1,2, 3, and FEC packet 0.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->fec_loss_mask_[0] = 1;
+  this->media_loss_mask_[1] = 1;
+  this->media_loss_mask_[2] = 1;
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Expect complete recovery for consecutive packet loss <= 50%.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // 4 packets lost (non-consecutive loss): media packets 0, 3, and FEC# 0, 3.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->fec_loss_mask_[0] = 1;
+  this->fec_loss_mask_[3] = 1;
+  this->media_loss_mask_[0] = 1;
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Cannot get complete recovery for this loss configuration.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+}
+
+TYPED_TEST(RtpFecTest, FecRecoveryNoLossUep) {
+  constexpr int kNumImportantPackets = 2;
+  constexpr bool kUseUnequalProtection = true;
+  constexpr int kNumMediaPackets = 4;
+  constexpr uint8_t kProtectionFactor = 60;
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 1 FEC packet.
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+
+  // No packets lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // No packets lost, expect complete recovery.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+}
+
+TYPED_TEST(RtpFecTest, FecRecoveryWithLossUep) {
+  constexpr int kNumImportantPackets = 2;
+  constexpr bool kUseUnequalProtection = true;
+  constexpr int kNumMediaPackets = 4;
+  constexpr uint8_t kProtectionFactor = 60;
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 1 FEC packet.
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+
+  // 1 media packet lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // One packet lost, one FEC packet, expect complete recovery.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // 2 media packets lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[1] = 1;
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // 2 packets lost, one FEC packet, cannot get complete recovery.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+}
+
+// Test 50% protection with random mask type for UEP on.
+TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percUepRandomMask) {
+  constexpr int kNumImportantPackets = 1;
+  constexpr bool kUseUnequalProtection = true;
+  constexpr int kNumMediaPackets = 4;
+  constexpr uint8_t kProtectionFactor = 255;
+
+  // Packet Mask for (4,4,1) code, from random mask table.
+  // (kNumMediaPackets = 4; num_fec_packets = 4, kNumImportantPackets = 1)
+
+  //         media#0   media#1  media#2    media#3
+  // fec#0:    1          0        0          0
+  // fec#1:    1          1        0          0
+  // fec#2:    1          0        1          1
+  // fec#3:    0          1        1          0
+  //
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(this->media_packets_, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskRandom, &this->generated_fec_packets_));
+
+  // Expect 4 FEC packets.
+  EXPECT_EQ(4u, this->generated_fec_packets_.size());
+
+  // 4 packets lost: 3 media packets and FEC packet#1 lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->fec_loss_mask_[1] = 1;
+  this->media_loss_mask_[0] = 1;
+  this->media_loss_mask_[2] = 1;
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // With media packet#3 and FEC packets #0, #1, #3, expect complete recovery.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // 5 packets lost: 4 media packets and one FEC packet#2 lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->fec_loss_mask_[2] = 1;
+  this->media_loss_mask_[0] = 1;
+  this->media_loss_mask_[1] = 1;
+  this->media_loss_mask_[2] = 1;
+  this->media_loss_mask_[3] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Cannot get complete recovery for this loss configuration.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+}
+
+TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePackets) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr int kNumMediaPackets = 5;
+  constexpr uint8_t kProtectionFactor = 60;
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  // Create a new temporary packet list for generating FEC packets.
+  // This list should have every other packet removed.
+  ForwardErrorCorrection::PacketList protected_media_packets;
+  DeepCopyEveryNthPacket(this->media_packets_, 2, &protected_media_packets);
+
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(protected_media_packets, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 1 FEC packet.
+  EXPECT_EQ(1u, this->generated_fec_packets_.size());
+
+  // 1 protected media packet lost
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[2] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // One packet lost, one FEC packet, expect complete recovery.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // Unprotected packet lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[1] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Unprotected packet lost. Recovery not possible.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // 2 media packets lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[0] = 1;
+  this->media_loss_mask_[2] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // 2 protected packets lost, one FEC packet, cannot get complete recovery.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+}
+
+TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePacketsExtension) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr int kNumMediaPackets = 21;
+  uint8_t kProtectionFactor = 127;
+
+  this->media_packets_ =
+      this->media_packet_generator_.ConstructMediaPackets(kNumMediaPackets);
+
+  // Create a new temporary packet list for generating FEC packets.
+  // This list should have every other packet removed.
+  ForwardErrorCorrection::PacketList protected_media_packets;
+  DeepCopyEveryNthPacket(this->media_packets_, 2, &protected_media_packets);
+
+  // Zero column insertion will have to extend the size of the packet
+  // mask since the number of actual packets are 21, while the number
+  // of protected packets are 11.
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(protected_media_packets, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 5 FEC packet.
+  EXPECT_EQ(5u, this->generated_fec_packets_.size());
+
+  // Last protected media packet lost
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[kNumMediaPackets - 1] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // One packet lost, one FEC packet, expect complete recovery.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // Last unprotected packet lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[kNumMediaPackets - 2] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Unprotected packet lost. Recovery not possible.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // 6 media packets lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[kNumMediaPackets - 11] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 9] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 7] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 5] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 3] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 1] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // 5 protected packets lost, one FEC packet, cannot get complete recovery.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+}
+
+TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePacketsWrap) {
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr int kNumMediaPackets = 21;
+  uint8_t kProtectionFactor = 127;
+
+  this->media_packets_ = this->media_packet_generator_.ConstructMediaPackets(
+      kNumMediaPackets, 0xFFFF - 5);
+
+  // Create a new temporary packet list for generating FEC packets.
+  // This list should have every other packet removed.
+  ForwardErrorCorrection::PacketList protected_media_packets;
+  DeepCopyEveryNthPacket(this->media_packets_, 2, &protected_media_packets);
+
+  // Zero column insertion will have to extend the size of the packet
+  // mask since the number of actual packets are 21, while the number
+  // of protected packets are 11.
+  EXPECT_EQ(
+      0, this->fec_.EncodeFec(protected_media_packets, kProtectionFactor,
+                              kNumImportantPackets, kUseUnequalProtection,
+                              kFecMaskBursty, &this->generated_fec_packets_));
+
+  // Expect 5 FEC packet.
+  EXPECT_EQ(5u, this->generated_fec_packets_.size());
+
+  // Last protected media packet lost
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[kNumMediaPackets - 1] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // One packet lost, one FEC packet, expect complete recovery.
+  EXPECT_TRUE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // Last unprotected packet lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[kNumMediaPackets - 2] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // Unprotected packet lost. Recovery not possible.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+  this->recovered_packets_.clear();
+
+  // 6 media packets lost.
+  memset(this->media_loss_mask_, 0, sizeof(this->media_loss_mask_));
+  memset(this->fec_loss_mask_, 0, sizeof(this->fec_loss_mask_));
+  this->media_loss_mask_[kNumMediaPackets - 11] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 9] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 7] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 5] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 3] = 1;
+  this->media_loss_mask_[kNumMediaPackets - 1] = 1;
+  this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
+
+  for (const auto& received_packet : this->received_packets_) {
+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
+  }
+
+  // 5 protected packets lost, one FEC packet, cannot get complete recovery.
+  EXPECT_FALSE(this->IsRecoveryComplete());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.cc
new file mode 100644
index 0000000000..7550b70f69
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.cc
@@ -0,0 +1,144 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format.h"
+
+#include <memory>
+
+#include "absl/types/variant.h"
+#include "modules/rtp_rtcp/source/rtp_format_h264.h"
+#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
+#include "modules/rtp_rtcp/source/rtp_packetizer_av1.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
+    absl::optional<VideoCodecType> type,
+    rtc::ArrayView<const uint8_t> payload,
+    PayloadSizeLimits limits,
+    // Codec-specific details.
+    const RTPVideoHeader& rtp_video_header) {
+  if (!type) {
+    // Use raw packetizer.
+    return std::make_unique<RtpPacketizerGeneric>(payload, limits);
+  }
+
+  switch (*type) {
+    case kVideoCodecH264: {
+      const auto& h264 =
+          absl::get<RTPVideoHeaderH264>(rtp_video_header.video_type_header);
+      return std::make_unique<RtpPacketizerH264>(payload, limits,
+                                                 h264.packetization_mode);
+    }
+    case kVideoCodecVP8: {
+      const auto& vp8 =
+          absl::get<RTPVideoHeaderVP8>(rtp_video_header.video_type_header);
+      return std::make_unique<RtpPacketizerVp8>(payload, limits, vp8);
+    }
+    case kVideoCodecVP9: {
+      const auto& vp9 =
+          absl::get<RTPVideoHeaderVP9>(rtp_video_header.video_type_header);
+      return std::make_unique<RtpPacketizerVp9>(payload, limits, vp9);
+    }
+    case kVideoCodecAV1:
+      return std::make_unique<RtpPacketizerAv1>(
+          payload, limits, rtp_video_header.frame_type,
+          rtp_video_header.is_last_frame_in_picture);
+    default: {
+      return std::make_unique<RtpPacketizerGeneric>(payload, limits,
+                                                    rtp_video_header);
+    }
+  }
+}
+
+std::vector<int> RtpPacketizer::SplitAboutEqually(
+    int payload_len,
+    const PayloadSizeLimits& limits) {
+  RTC_DCHECK_GT(payload_len, 0);
+  // First or last packet larger than normal are unsupported.
+  RTC_DCHECK_GE(limits.first_packet_reduction_len, 0);
+  RTC_DCHECK_GE(limits.last_packet_reduction_len, 0);
+
+  std::vector<int> result;
+  if (limits.max_payload_len >=
+      limits.single_packet_reduction_len + payload_len) {
+    result.push_back(payload_len);
+    return result;
+  }
+  if (limits.max_payload_len - limits.first_packet_reduction_len < 1 ||
+      limits.max_payload_len - limits.last_packet_reduction_len < 1) {
+    // Capacity is not enough to put a single byte into one of the packets.
+    return result;
+  }
+  // First and last packet of the frame can be smaller. Pretend that it's
+  // the same size, but we must write more payload to it.
+  // Assume frame fits in single packet if packet has extra space for sum
+  // of first and last packets reductions.
+  int total_bytes = payload_len + limits.first_packet_reduction_len +
+                    limits.last_packet_reduction_len;
+  // Integer divisions with rounding up.
+  int num_packets_left =
+      (total_bytes + limits.max_payload_len - 1) / limits.max_payload_len;
+  if (num_packets_left == 1) {
+    // Single packet is a special case handled above.
+    num_packets_left = 2;
+  }
+
+  if (payload_len < num_packets_left) {
+    // Edge case where limits force to have more packets than there are payload
+    // bytes. This may happen when there is single byte of payload that can't be
+    // put into single packet if
+    // first_packet_reduction + last_packet_reduction >= max_payload_len.
+    return result;
+  }
+
+  int bytes_per_packet = total_bytes / num_packets_left;
+  int num_larger_packets = total_bytes % num_packets_left;
+  int remaining_data = payload_len;
+
+  result.reserve(num_packets_left);
+  bool first_packet = true;
+  while (remaining_data > 0) {
+    // Last num_larger_packets are 1 byte wider than the rest. Increase
+    // per-packet payload size when needed.
+    if (num_packets_left == num_larger_packets)
+      ++bytes_per_packet;
+    int current_packet_bytes = bytes_per_packet;
+    if (first_packet) {
+      if (current_packet_bytes > limits.first_packet_reduction_len + 1)
+        current_packet_bytes -= limits.first_packet_reduction_len;
+      else
+        current_packet_bytes = 1;
+    }
+    if (current_packet_bytes > remaining_data) {
+      current_packet_bytes = remaining_data;
+    }
+    // This is not the last packet in the whole payload, but there's no data
+    // left for the last packet. Leave at least one byte for the last packet.
+    if (num_packets_left == 2 && current_packet_bytes == remaining_data) {
+      --current_packet_bytes;
+    }
+    result.push_back(current_packet_bytes);
+
+    remaining_data -= current_packet_bytes;
+    --num_packets_left;
+    first_packet = false;
+  }
+
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h
new file mode 100644
index 0000000000..19abd3feb2
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+
+namespace webrtc {
+
+class RtpPacketToSend;
+
+class RtpPacketizer {
+ public:
+  struct PayloadSizeLimits {
+    int max_payload_len = 1200;
+    int first_packet_reduction_len = 0;
+    int last_packet_reduction_len = 0;
+    // Reduction len for packet that is first & last at the same time.
+    int single_packet_reduction_len = 0;
+  };
+
+  // If type is not set, returns a raw packetizer.
+  static std::unique_ptr<RtpPacketizer> Create(
+      absl::optional<VideoCodecType> type,
+      rtc::ArrayView<const uint8_t> payload,
+      PayloadSizeLimits limits,
+      // Codec-specific details.
+      const RTPVideoHeader& rtp_video_header);
+
+  virtual ~RtpPacketizer() = default;
+
+  // Returns number of remaining packets to produce by the packetizer.
+  virtual size_t NumPackets() const = 0;
+
+  // Get the next payload with payload header.
+  // Write payload and set marker bit of the `packet`.
+  // Returns true on success, false otherwise.
+  virtual bool NextPacket(RtpPacketToSend* packet) = 0;
+
+  // Split payload_len into sum of integers with respect to `limits`.
+  // Returns empty vector on failure.
+  static std::vector<int> SplitAboutEqually(int payload_len,
+                                            const PayloadSizeLimits& limits);
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
new file mode 100644
index 0000000000..cc8d1bff34
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -0,0 +1,313 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_h264.h"
+
+#include <string.h>
+
+#include <cstddef>
+#include <cstdint>
+#include <iterator>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "absl/types/variant.h"
+#include "common_video/h264/h264_common.h"
+#include "common_video/h264/pps_parser.h"
+#include "common_video/h264/sps_parser.h"
+#include "common_video/h264/sps_vui_rewriter.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+static const size_t kNalHeaderSize = 1;
+static const size_t kFuAHeaderSize = 2;
+static const size_t kLengthFieldSize = 2;
+
+}  // namespace
+
+RtpPacketizerH264::RtpPacketizerH264(rtc::ArrayView<const uint8_t> payload,
+                                     PayloadSizeLimits limits,
+                                     H264PacketizationMode packetization_mode)
+    : limits_(limits), num_packets_left_(0) {
+  // Guard against uninitialized memory in packetization_mode.
+  RTC_CHECK(packetization_mode == H264PacketizationMode::NonInterleaved ||
+            packetization_mode == H264PacketizationMode::SingleNalUnit);
+
+  for (const auto& nalu :
+       H264::FindNaluIndices(payload.data(), payload.size())) {
+    input_fragments_.push_back(
+        payload.subview(nalu.payload_start_offset, nalu.payload_size));
+  }
+
+  if (!GeneratePackets(packetization_mode)) {
+    // If failed to generate all the packets, discard already generated
+    // packets in case the caller would ignore return value and still try to
+    // call NextPacket().
+    num_packets_left_ = 0;
+    while (!packets_.empty()) {
+      packets_.pop();
+    }
+  }
+}
+
+RtpPacketizerH264::~RtpPacketizerH264() = default;
+
+size_t RtpPacketizerH264::NumPackets() const {
+  return num_packets_left_;
+}
+
+bool RtpPacketizerH264::GeneratePackets(
+    H264PacketizationMode packetization_mode) {
+  for (size_t i = 0; i < input_fragments_.size();) {
+    switch (packetization_mode) {
+      case H264PacketizationMode::SingleNalUnit:
+        if (!PacketizeSingleNalu(i))
+          return false;
+        ++i;
+        break;
+      case H264PacketizationMode::NonInterleaved:
+        int fragment_len = input_fragments_[i].size();
+        int single_packet_capacity = limits_.max_payload_len;
+        if (input_fragments_.size() == 1)
+          single_packet_capacity -= limits_.single_packet_reduction_len;
+        else if (i == 0)
+          single_packet_capacity -= limits_.first_packet_reduction_len;
+        else if (i + 1 == input_fragments_.size())
+          single_packet_capacity -= limits_.last_packet_reduction_len;
+
+        if (fragment_len > single_packet_capacity) {
+          if (!PacketizeFuA(i))
+            return false;
+          ++i;
+        } else {
+          i = PacketizeStapA(i);
+        }
+        break;
+    }
+  }
+  return true;
+}
+
+bool RtpPacketizerH264::PacketizeFuA(size_t fragment_index) {
+  // Fragment payload into packets (FU-A).
+  rtc::ArrayView<const uint8_t> fragment = input_fragments_[fragment_index];
+
+  PayloadSizeLimits limits = limits_;
+  // Leave room for the FU-A header.
+  limits.max_payload_len -= kFuAHeaderSize;
+  // Update single/first/last packet reductions unless it is single/first/last
+  // fragment.
+  if (input_fragments_.size() != 1) {
+    // if this fragment is put into a single packet, it might still be the
+    // first or the last packet in the whole sequence of packets.
+    if (fragment_index == input_fragments_.size() - 1) {
+      limits.single_packet_reduction_len = limits_.last_packet_reduction_len;
+    } else if (fragment_index == 0) {
+      limits.single_packet_reduction_len = limits_.first_packet_reduction_len;
+    } else {
+      limits.single_packet_reduction_len = 0;
+    }
+  }
+  if (fragment_index != 0)
+    limits.first_packet_reduction_len = 0;
+  if (fragment_index != input_fragments_.size() - 1)
+    limits.last_packet_reduction_len = 0;
+
+  // Strip out the original header.
+  size_t payload_left = fragment.size() - kNalHeaderSize;
+  int offset = kNalHeaderSize;
+
+  std::vector<int> payload_sizes = SplitAboutEqually(payload_left, limits);
+  if (payload_sizes.empty())
+    return false;
+
+  for (size_t i = 0; i < payload_sizes.size(); ++i) {
+    int packet_length = payload_sizes[i];
+    RTC_CHECK_GT(packet_length, 0);
+    packets_.push(PacketUnit(fragment.subview(offset, packet_length),
+                             /*first_fragment=*/i == 0,
+                             /*last_fragment=*/i == payload_sizes.size() - 1,
+                             false, fragment[0]));
+    offset += packet_length;
+    payload_left -= packet_length;
+  }
+  num_packets_left_ += payload_sizes.size();
+  RTC_CHECK_EQ(0, payload_left);
+  return true;
+}
+
+size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
+  // Aggregate fragments into one packet (STAP-A).
+  size_t payload_size_left = limits_.max_payload_len;
+  if (input_fragments_.size() == 1)
+    payload_size_left -= limits_.single_packet_reduction_len;
+  else if (fragment_index == 0)
+    payload_size_left -= limits_.first_packet_reduction_len;
+  int aggregated_fragments = 0;
+  size_t fragment_headers_length = 0;
+  rtc::ArrayView<const uint8_t> fragment = input_fragments_[fragment_index];
+  RTC_CHECK_GE(payload_size_left, fragment.size());
+  ++num_packets_left_;
+
+  auto payload_size_needed = [&] {
+    size_t fragment_size = fragment.size() + fragment_headers_length;
+    if (input_fragments_.size() == 1) {
+      // Single fragment, single packet, payload_size_left already adjusted
+      // with limits_.single_packet_reduction_len.
+      return fragment_size;
+    }
+    if (fragment_index == input_fragments_.size() - 1) {
+      // Last fragment, so STAP-A might be the last packet.
+      return fragment_size + limits_.last_packet_reduction_len;
+    }
+    return fragment_size;
+  };
+
+  while (payload_size_left >= payload_size_needed()) {
+    RTC_CHECK_GT(fragment.size(), 0);
+    packets_.push(PacketUnit(fragment, aggregated_fragments == 0, false, true,
+                             fragment[0]));
+    payload_size_left -= fragment.size();
+    payload_size_left -= fragment_headers_length;
+
+    fragment_headers_length = kLengthFieldSize;
+    // If we are going to try to aggregate more fragments into this packet
+    // we need to add the STAP-A NALU header and a length field for the first
+    // NALU of this packet.
+    if (aggregated_fragments == 0)
+      fragment_headers_length += kNalHeaderSize + kLengthFieldSize;
+    ++aggregated_fragments;
+
+    // Next fragment.
+    ++fragment_index;
+    if (fragment_index == input_fragments_.size())
+      break;
+    fragment = input_fragments_[fragment_index];
+  }
+  RTC_CHECK_GT(aggregated_fragments, 0);
+  packets_.back().last_fragment = true;
+  return fragment_index;
+}
+
+bool RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
+  // Add a single NALU to the queue, no aggregation.
+  size_t payload_size_left = limits_.max_payload_len;
+  if (input_fragments_.size() == 1)
+    payload_size_left -= limits_.single_packet_reduction_len;
+  else if (fragment_index == 0)
+    payload_size_left -= limits_.first_packet_reduction_len;
+  else if (fragment_index + 1 == input_fragments_.size())
+    payload_size_left -= limits_.last_packet_reduction_len;
+  rtc::ArrayView<const uint8_t> fragment = input_fragments_[fragment_index];
+  if (payload_size_left < fragment.size()) {
+    RTC_LOG(LS_ERROR) << "Failed to fit a fragment to packet in SingleNalu "
+                         "packetization mode. Payload size left "
+                      << payload_size_left << ", fragment length "
+                      << fragment.size() << ", packet capacity "
+                      << limits_.max_payload_len;
+    return false;
+  }
+  RTC_CHECK_GT(fragment.size(), 0u);
+  packets_.push(PacketUnit(fragment, true /* first */, true /* last */,
+                           false /* aggregated */, fragment[0]));
+  ++num_packets_left_;
+  return true;
+}
+
+bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet) {
+  RTC_DCHECK(rtp_packet);
+  if (packets_.empty()) {
+    return false;
+  }
+
+  PacketUnit packet = packets_.front();
+  if (packet.first_fragment && packet.last_fragment) {
+    // Single NAL unit packet.
+    size_t bytes_to_send = packet.source_fragment.size();
+    uint8_t* buffer = rtp_packet->AllocatePayload(bytes_to_send);
+    memcpy(buffer, packet.source_fragment.data(), bytes_to_send);
+    packets_.pop();
+    input_fragments_.pop_front();
+  } else if (packet.aggregated) {
+    NextAggregatePacket(rtp_packet);
+  } else {
+    NextFragmentPacket(rtp_packet);
+  }
+  rtp_packet->SetMarker(packets_.empty());
+  --num_packets_left_;
+  return true;
+}
+
+void RtpPacketizerH264::NextAggregatePacket(RtpPacketToSend* rtp_packet) {
+  // Reserve maximum available payload, set actual payload size later.
+  size_t payload_capacity = rtp_packet->FreeCapacity();
+  RTC_CHECK_GE(payload_capacity, kNalHeaderSize);
+  uint8_t* buffer = rtp_packet->AllocatePayload(payload_capacity);
+  RTC_DCHECK(buffer);
+  PacketUnit* packet = &packets_.front();
+  RTC_CHECK(packet->first_fragment);
+  // STAP-A NALU header.
+  buffer[0] =
+      (packet->header & (kH264FBit | kH264NriMask)) | H264::NaluType::kStapA;
+  size_t index = kNalHeaderSize;
+  bool is_last_fragment = packet->last_fragment;
+  while (packet->aggregated) {
+    rtc::ArrayView<const uint8_t> fragment = packet->source_fragment;
+    RTC_CHECK_LE(index + kLengthFieldSize + fragment.size(), payload_capacity);
+    // Add NAL unit length field.
+    ByteWriter<uint16_t>::WriteBigEndian(&buffer[index], fragment.size());
+    index += kLengthFieldSize;
+    // Add NAL unit.
+    memcpy(&buffer[index], fragment.data(), fragment.size());
+    index += fragment.size();
+    packets_.pop();
+    input_fragments_.pop_front();
+    if (is_last_fragment)
+      break;
+    packet = &packets_.front();
+    is_last_fragment = packet->last_fragment;
+  }
+  RTC_CHECK(is_last_fragment);
+  rtp_packet->SetPayloadSize(index);
+}
+
+void RtpPacketizerH264::NextFragmentPacket(RtpPacketToSend* rtp_packet) {
+  PacketUnit* packet = &packets_.front();
+  // NAL unit fragmented over multiple packets (FU-A).
+  // We do not send original NALU header, so it will be replaced by the
+  // FU indicator header of the first packet.
+  uint8_t fu_indicator =
+      (packet->header & (kH264FBit | kH264NriMask)) | H264::NaluType::kFuA;
+  uint8_t fu_header = 0;
+
+  // S | E | R | 5 bit type.
+  fu_header |= (packet->first_fragment ? kH264SBit : 0);
+  fu_header |= (packet->last_fragment ? kH264EBit : 0);
+  uint8_t type = packet->header & kH264TypeMask;
+  fu_header |= type;
+  rtc::ArrayView<const uint8_t> fragment = packet->source_fragment;
+  uint8_t* buffer =
+      rtp_packet->AllocatePayload(kFuAHeaderSize + fragment.size());
+  buffer[0] = fu_indicator;
+  buffer[1] = fu_header;
+  memcpy(buffer + kFuAHeaderSize, fragment.data(), fragment.size());
+  if (packet->last_fragment)
+    input_fragments_.pop_front();
+  packets_.pop();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.h
new file mode 100644
index 0000000000..f95c3b6c6b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+#include <queue>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+// Bit masks for NAL (F, NRI, Type) indicators.
+constexpr uint8_t kH264FBit = 0x80;
+constexpr uint8_t kH264NriMask = 0x60;
+constexpr uint8_t kH264TypeMask = 0x1F;
+
+// Bit masks for FU (A and B) headers.
+constexpr uint8_t kH264SBit = 0x80;
+constexpr uint8_t kH264EBit = 0x40;
+constexpr uint8_t kH264RBit = 0x20;
+
+class RtpPacketizerH264 : public RtpPacketizer {
+ public:
+  // Initialize with payload from encoder.
+  // The payload_data must be exactly one encoded H264 frame.
+  RtpPacketizerH264(rtc::ArrayView<const uint8_t> payload,
+                    PayloadSizeLimits limits,
+                    H264PacketizationMode packetization_mode);
+
+  ~RtpPacketizerH264() override;
+
+  RtpPacketizerH264(const RtpPacketizerH264&) = delete;
+  RtpPacketizerH264& operator=(const RtpPacketizerH264&) = delete;
+
+  size_t NumPackets() const override;
+
+  // Get the next payload with H264 payload header.
+  // Write payload and set marker bit of the `packet`.
+  // Returns true on success, false otherwise.
+  bool NextPacket(RtpPacketToSend* rtp_packet) override;
+
+ private:
+  // A packet unit (H264 packet), to be put into an RTP packet:
+  // If a NAL unit is too large for an RTP packet, this packet unit will
+  // represent a FU-A packet of a single fragment of the NAL unit.
+  // If a NAL unit is small enough to fit within a single RTP packet, this
+  // packet unit may represent a single NAL unit or a STAP-A packet, of which
+  // there may be multiple in a single RTP packet (if so, aggregated = true).
+  struct PacketUnit {
+    PacketUnit(rtc::ArrayView<const uint8_t> source_fragment,
+               bool first_fragment,
+               bool last_fragment,
+               bool aggregated,
+               uint8_t header)
+        : source_fragment(source_fragment),
+          first_fragment(first_fragment),
+          last_fragment(last_fragment),
+          aggregated(aggregated),
+          header(header) {}
+
+    rtc::ArrayView<const uint8_t> source_fragment;
+    bool first_fragment;
+    bool last_fragment;
+    bool aggregated;
+    uint8_t header;
+  };
+
+  bool GeneratePackets(H264PacketizationMode packetization_mode);
+  bool PacketizeFuA(size_t fragment_index);
+  size_t PacketizeStapA(size_t fragment_index);
+  bool PacketizeSingleNalu(size_t fragment_index);
+
+  void NextAggregatePacket(RtpPacketToSend* rtp_packet);
+  void NextFragmentPacket(RtpPacketToSend* rtp_packet);
+
+  const PayloadSizeLimits limits_;
+  size_t num_packets_left_;
+  std::deque<rtc::ArrayView<const uint8_t>> input_fragments_;
+  std::queue<PacketUnit> packets_;
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
new file mode 100644
index 0000000000..80d8801437
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -0,0 +1,496 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_h264.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "api/array_view.h"
+#include "common_video/h264/h264_common.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Each;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::IsEmpty;
+using ::testing::SizeIs;
+
+constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr;
+constexpr size_t kMaxPayloadSize = 1200;
+constexpr size_t kLengthFieldLength = 2;
+constexpr RtpPacketizer::PayloadSizeLimits kNoLimits;
+
+enum Nalu {
+  kSlice = 1,
+  kIdr = 5,
+  kSei = 6,
+  kSps = 7,
+  kPps = 8,
+  kStapA = 24,
+  kFuA = 28
+};
+
+static const size_t kNalHeaderSize = 1;
+static const size_t kFuAHeaderSize = 2;
+
+// Creates Buffer that looks like nal unit of given size.
+rtc::Buffer GenerateNalUnit(size_t size) {
+  RTC_CHECK_GT(size, 0);
+  rtc::Buffer buffer(size);
+  // Set some valid header.
+  buffer[0] = kSlice;
+  for (size_t i = 1; i < size; ++i) {
+    buffer[i] = static_cast<uint8_t>(i);
+  }
+  // Last byte shouldn't be 0, or it may be counted as part of next 4-byte start
+  // sequence.
+  buffer[size - 1] |= 0x10;
+  return buffer;
+}
+
+// Create frame consisting of nalus of given size.
+rtc::Buffer CreateFrame(std::initializer_list<size_t> nalu_sizes) {
+  static constexpr int kStartCodeSize = 3;
+  rtc::Buffer frame(absl::c_accumulate(nalu_sizes, size_t{0}) +
+                    kStartCodeSize * nalu_sizes.size());
+  size_t offset = 0;
+  for (size_t nalu_size : nalu_sizes) {
+    EXPECT_GE(nalu_size, 1u);
+    // Insert nalu start code
+    frame[offset] = 0;
+    frame[offset + 1] = 0;
+    frame[offset + 2] = 1;
+    // Set some valid header.
+    frame[offset + 3] = 1;
+    // Fill payload avoiding accidental start codes
+    if (nalu_size > 1) {
+      memset(frame.data() + offset + 4, 0x3f, nalu_size - 1);
+    }
+    offset += (kStartCodeSize + nalu_size);
+  }
+  return frame;
+}
+
+// Create frame consisting of given nalus.
+rtc::Buffer CreateFrame(rtc::ArrayView<const rtc::Buffer> nalus) {
+  static constexpr int kStartCodeSize = 3;
+  int frame_size = 0;
+  for (const rtc::Buffer& nalu : nalus) {
+    frame_size += (kStartCodeSize + nalu.size());
+  }
+  rtc::Buffer frame(frame_size);
+  size_t offset = 0;
+  for (const rtc::Buffer& nalu : nalus) {
+    // Insert nalu start code
+    frame[offset] = 0;
+    frame[offset + 1] = 0;
+    frame[offset + 2] = 1;
+    // Copy the nalu unit.
+    memcpy(frame.data() + offset + 3, nalu.data(), nalu.size());
+    offset += (kStartCodeSize + nalu.size());
+  }
+  return frame;
+}
+
+std::vector<RtpPacketToSend> FetchAllPackets(RtpPacketizerH264* packetizer) {
+  std::vector<RtpPacketToSend> result;
+  size_t num_packets = packetizer->NumPackets();
+  result.reserve(num_packets);
+  RtpPacketToSend packet(kNoExtensions);
+  while (packetizer->NextPacket(&packet)) {
+    result.push_back(packet);
+  }
+  EXPECT_THAT(result, SizeIs(num_packets));
+  return result;
+}
+
+// Tests that should work with both packetization mode 0 and
+// packetization mode 1.
+class RtpPacketizerH264ModeTest
+    : public ::testing::TestWithParam<H264PacketizationMode> {};
+
+TEST_P(RtpPacketizerH264ModeTest, SingleNalu) {
+  const uint8_t frame[] = {0, 0, 1, kIdr, 0xFF};
+
+  RtpPacketizerH264 packetizer(frame, kNoLimits, GetParam());
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  ASSERT_THAT(packets, SizeIs(1));
+  EXPECT_THAT(packets[0].payload(), ElementsAre(kIdr, 0xFF));
+}
+
+TEST_P(RtpPacketizerH264ModeTest, SingleNaluTwoPackets) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = kMaxPayloadSize;
+  rtc::Buffer nalus[] = {GenerateNalUnit(kMaxPayloadSize),
+                         GenerateNalUnit(100)};
+  rtc::Buffer frame = CreateFrame(nalus);
+
+  RtpPacketizerH264 packetizer(frame, limits, GetParam());
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  ASSERT_THAT(packets, SizeIs(2));
+  EXPECT_THAT(packets[0].payload(), ElementsAreArray(nalus[0]));
+  EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[1]));
+}
+
+TEST_P(RtpPacketizerH264ModeTest,
+       SingleNaluFirstPacketReductionAppliesOnlyToFirstFragment) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 200;
+  limits.first_packet_reduction_len = 5;
+  rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/195),
+                         GenerateNalUnit(/*size=*/200),
+                         GenerateNalUnit(/*size=*/200)};
+  rtc::Buffer frame = CreateFrame(nalus);
+
+  RtpPacketizerH264 packetizer(frame, limits, GetParam());
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  ASSERT_THAT(packets, SizeIs(3));
+  EXPECT_THAT(packets[0].payload(), ElementsAreArray(nalus[0]));
+  EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[1]));
+  EXPECT_THAT(packets[2].payload(), ElementsAreArray(nalus[2]));
+}
+
+TEST_P(RtpPacketizerH264ModeTest,
+       SingleNaluLastPacketReductionAppliesOnlyToLastFragment) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 200;
+  limits.last_packet_reduction_len = 5;
+  rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/200),
+                         GenerateNalUnit(/*size=*/200),
+                         GenerateNalUnit(/*size=*/195)};
+  rtc::Buffer frame = CreateFrame(nalus);
+
+  RtpPacketizerH264 packetizer(frame, limits, GetParam());
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  ASSERT_THAT(packets, SizeIs(3));
+  EXPECT_THAT(packets[0].payload(), ElementsAreArray(nalus[0]));
+  EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[1]));
+  EXPECT_THAT(packets[2].payload(), ElementsAreArray(nalus[2]));
+}
+
+TEST_P(RtpPacketizerH264ModeTest,
+       SingleNaluFirstAndLastPacketReductionSumsForSinglePacket) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 200;
+  limits.first_packet_reduction_len = 20;
+  limits.last_packet_reduction_len = 30;
+  rtc::Buffer frame = CreateFrame({150});
+
+  RtpPacketizerH264 packetizer(frame, limits, GetParam());
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  EXPECT_THAT(packets, SizeIs(1));
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    PacketMode,
+    RtpPacketizerH264ModeTest,
+    ::testing::Values(H264PacketizationMode::SingleNalUnit,
+                      H264PacketizationMode::NonInterleaved));
+
+// Aggregation tests.
+TEST(RtpPacketizerH264Test, StapA) {
+  rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/2),
+                         GenerateNalUnit(/*size=*/2),
+                         GenerateNalUnit(/*size=*/0x123)};
+  rtc::Buffer frame = CreateFrame(nalus);
+
+  RtpPacketizerH264 packetizer(frame, kNoLimits,
+                               H264PacketizationMode::NonInterleaved);
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  ASSERT_THAT(packets, SizeIs(1));
+  auto payload = packets[0].payload();
+  EXPECT_EQ(payload.size(),
+            kNalHeaderSize + 3 * kLengthFieldLength + 2 + 2 + 0x123);
+
+  EXPECT_EQ(payload[0], kStapA);
+  payload = payload.subview(kNalHeaderSize);
+  // 1st fragment.
+  EXPECT_THAT(payload.subview(0, kLengthFieldLength),
+              ElementsAre(0, 2));  // Size.
+  EXPECT_THAT(payload.subview(kLengthFieldLength, 2),
+              ElementsAreArray(nalus[0]));
+  payload = payload.subview(kLengthFieldLength + 2);
+  // 2nd fragment.
+  EXPECT_THAT(payload.subview(0, kLengthFieldLength),
+              ElementsAre(0, 2));  // Size.
+  EXPECT_THAT(payload.subview(kLengthFieldLength, 2),
+              ElementsAreArray(nalus[1]));
+  payload = payload.subview(kLengthFieldLength + 2);
+  // 3rd fragment.
+  EXPECT_THAT(payload.subview(0, kLengthFieldLength),
+              ElementsAre(0x1, 0x23));  // Size.
+  EXPECT_THAT(payload.subview(kLengthFieldLength), ElementsAreArray(nalus[2]));
+}
+
+TEST(RtpPacketizerH264Test, SingleNalUnitModeHasNoStapA) {
+  // This is the same setup as for the StapA test.
+  rtc::Buffer frame = CreateFrame({2, 2, 0x123});
+
+  RtpPacketizerH264 packetizer(frame, kNoLimits,
+                               H264PacketizationMode::SingleNalUnit);
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  // The three fragments should be returned as three packets.
+  ASSERT_THAT(packets, SizeIs(3));
+  EXPECT_EQ(packets[0].payload_size(), 2u);
+  EXPECT_EQ(packets[1].payload_size(), 2u);
+  EXPECT_EQ(packets[2].payload_size(), 0x123u);
+}
+
+TEST(RtpPacketizerH264Test, StapARespectsFirstPacketReduction) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1000;
+  limits.first_packet_reduction_len = 100;
+  const size_t kFirstFragmentSize =
+      limits.max_payload_len - limits.first_packet_reduction_len;
+  rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/kFirstFragmentSize),
+                         GenerateNalUnit(/*size=*/2),
+                         GenerateNalUnit(/*size=*/2)};
+  rtc::Buffer frame = CreateFrame(nalus);
+
+  RtpPacketizerH264 packetizer(frame, limits,
+                               H264PacketizationMode::NonInterleaved);
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  ASSERT_THAT(packets, SizeIs(2));
+  // Expect 1st packet is single nalu.
+  EXPECT_THAT(packets[0].payload(), ElementsAreArray(nalus[0]));
+  // Expect 2nd packet is aggregate of last two fragments.
+  EXPECT_THAT(packets[1].payload(),
+              ElementsAre(kStapA,                          //
+                          0, 2, nalus[1][0], nalus[1][1],  //
+                          0, 2, nalus[2][0], nalus[2][1]));
+}
+
+TEST(RtpPacketizerH264Test, StapARespectsLastPacketReduction) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1000;
+  limits.last_packet_reduction_len = 100;
+  const size_t kLastFragmentSize =
+      limits.max_payload_len - limits.last_packet_reduction_len;
+  rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/2),
+                         GenerateNalUnit(/*size=*/2),
+                         GenerateNalUnit(/*size=*/kLastFragmentSize)};
+  rtc::Buffer frame = CreateFrame(nalus);
+
+  RtpPacketizerH264 packetizer(frame, limits,
+                               H264PacketizationMode::NonInterleaved);
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  ASSERT_THAT(packets, SizeIs(2));
+  // Expect 1st packet is aggregate of 1st two fragments.
+  EXPECT_THAT(packets[0].payload(),
+              ElementsAre(kStapA,                          //
+                          0, 2, nalus[0][0], nalus[0][1],  //
+                          0, 2, nalus[1][0], nalus[1][1]));
+  // Expect 2nd packet is single nalu.
+  EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[2]));
+}
+
+TEST(RtpPacketizerH264Test, TooSmallForStapAHeaders) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1000;
+  const size_t kLastFragmentSize =
+      limits.max_payload_len - 3 * kLengthFieldLength - 4;
+  rtc::Buffer nalus[] = {GenerateNalUnit(/*size=*/2),
+                         GenerateNalUnit(/*size=*/2),
+                         GenerateNalUnit(/*size=*/kLastFragmentSize)};
+  rtc::Buffer frame = CreateFrame(nalus);
+
+  RtpPacketizerH264 packetizer(frame, limits,
+                               H264PacketizationMode::NonInterleaved);
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  ASSERT_THAT(packets, SizeIs(2));
+  // Expect 1st packet is aggregate of 1st two fragments.
+  EXPECT_THAT(packets[0].payload(),
+              ElementsAre(kStapA,                          //
+                          0, 2, nalus[0][0], nalus[0][1],  //
+                          0, 2, nalus[1][0], nalus[1][1]));
+  // Expect 2nd packet is single nalu.
+  EXPECT_THAT(packets[1].payload(), ElementsAreArray(nalus[2]));
+}
+
+// Fragmentation + aggregation.
+TEST(RtpPacketizerH264Test, MixedStapAFUA) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 100;
+  const size_t kFuaPayloadSize = 70;
+  const size_t kFuaNaluSize = kNalHeaderSize + 2 * kFuaPayloadSize;
+  const size_t kStapANaluSize = 20;
+  rtc::Buffer nalus[] = {GenerateNalUnit(kFuaNaluSize),
+                         GenerateNalUnit(kStapANaluSize),
+                         GenerateNalUnit(kStapANaluSize)};
+  rtc::Buffer frame = CreateFrame(nalus);
+
+  RtpPacketizerH264 packetizer(frame, limits,
+                               H264PacketizationMode::NonInterleaved);
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  ASSERT_THAT(packets, SizeIs(3));
+  // First expect two FU-A packets.
+  EXPECT_THAT(packets[0].payload().subview(0, kFuAHeaderSize),
+              ElementsAre(kFuA, kH264SBit | nalus[0][0]));
+  EXPECT_THAT(
+      packets[0].payload().subview(kFuAHeaderSize),
+      ElementsAreArray(nalus[0].data() + kNalHeaderSize, kFuaPayloadSize));
+
+  EXPECT_THAT(packets[1].payload().subview(0, kFuAHeaderSize),
+              ElementsAre(kFuA, kH264EBit | nalus[0][0]));
+  EXPECT_THAT(
+      packets[1].payload().subview(kFuAHeaderSize),
+      ElementsAreArray(nalus[0].data() + kNalHeaderSize + kFuaPayloadSize,
+                       kFuaPayloadSize));
+
+  // Then expect one STAP-A packet with two nal units.
+  EXPECT_THAT(packets[2].payload()[0], kStapA);
+  auto payload = packets[2].payload().subview(kNalHeaderSize);
+  EXPECT_THAT(payload.subview(0, kLengthFieldLength),
+              ElementsAre(0, kStapANaluSize));
+  EXPECT_THAT(payload.subview(kLengthFieldLength, kStapANaluSize),
+              ElementsAreArray(nalus[1]));
+  payload = payload.subview(kLengthFieldLength + kStapANaluSize);
+  EXPECT_THAT(payload.subview(0, kLengthFieldLength),
+              ElementsAre(0, kStapANaluSize));
+  EXPECT_THAT(payload.subview(kLengthFieldLength), ElementsAreArray(nalus[2]));
+}
+
+TEST(RtpPacketizerH264Test, LastFragmentFitsInSingleButNotLastPacket) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1178;
+  limits.first_packet_reduction_len = 0;
+  limits.last_packet_reduction_len = 20;
+  limits.single_packet_reduction_len = 20;
+  // Actual sizes, which triggered this bug.
+  rtc::Buffer frame = CreateFrame({20, 8, 18, 1161});
+
+  RtpPacketizerH264 packetizer(frame, limits,
+                               H264PacketizationMode::NonInterleaved);
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  // Last packet has to be of correct size.
+  // Incorrect implementation might miss this constraint and not split the last
+  // fragment in two packets.
+  EXPECT_LE(static_cast<int>(packets.back().payload_size()),
+            limits.max_payload_len - limits.last_packet_reduction_len);
+}
+
+// Splits frame with payload size `frame_payload_size` without fragmentation,
+// Returns sizes of the payloads excluding fua headers.
+std::vector<int> TestFua(size_t frame_payload_size,
+                         const RtpPacketizer::PayloadSizeLimits& limits) {
+  rtc::Buffer nalu[] = {GenerateNalUnit(kNalHeaderSize + frame_payload_size)};
+  rtc::Buffer frame = CreateFrame(nalu);
+
+  RtpPacketizerH264 packetizer(frame, limits,
+                               H264PacketizationMode::NonInterleaved);
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  EXPECT_GE(packets.size(), 2u);  // Single packet indicates it is not FuA.
+  std::vector<uint16_t> fua_header;
+  std::vector<int> payload_sizes;
+
+  for (const RtpPacketToSend& packet : packets) {
+    auto payload = packet.payload();
+    EXPECT_GT(payload.size(), kFuAHeaderSize);
+    fua_header.push_back((payload[0] << 8) | payload[1]);
+    payload_sizes.push_back(payload.size() - kFuAHeaderSize);
+  }
+
+  EXPECT_TRUE(fua_header.front() & kH264SBit);
+  EXPECT_TRUE(fua_header.back() & kH264EBit);
+  // Clear S and E bits before testing all are duplicating same original header.
+  fua_header.front() &= ~kH264SBit;
+  fua_header.back() &= ~kH264EBit;
+  EXPECT_THAT(fua_header, Each(Eq((kFuA << 8) | nalu[0][0])));
+
+  return payload_sizes;
+}
+
+// Fragmentation tests.
+TEST(RtpPacketizerH264Test, FUAOddSize) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  EXPECT_THAT(TestFua(1200, limits), ElementsAre(600, 600));
+}
+
+TEST(RtpPacketizerH264Test, FUAWithFirstPacketReduction) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  limits.first_packet_reduction_len = 4;
+  limits.single_packet_reduction_len = 4;
+  EXPECT_THAT(TestFua(1198, limits), ElementsAre(597, 601));
+}
+
+TEST(RtpPacketizerH264Test, FUAWithLastPacketReduction) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  limits.last_packet_reduction_len = 4;
+  limits.single_packet_reduction_len = 4;
+  EXPECT_THAT(TestFua(1198, limits), ElementsAre(601, 597));
+}
+
+TEST(RtpPacketizerH264Test, FUAWithSinglePacketReduction) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1199;
+  limits.single_packet_reduction_len = 200;
+  EXPECT_THAT(TestFua(1000, limits), ElementsAre(500, 500));
+}
+
+TEST(RtpPacketizerH264Test, FUAEvenSize) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  EXPECT_THAT(TestFua(1201, limits), ElementsAre(600, 601));
+}
+
+TEST(RtpPacketizerH264Test, FUARounding) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1448;
+  EXPECT_THAT(TestFua(10123, limits),
+              ElementsAre(1265, 1265, 1265, 1265, 1265, 1266, 1266, 1266));
+}
+
+TEST(RtpPacketizerH264Test, FUABig) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  // Generate 10 full sized packets, leave room for FU-A headers.
+  EXPECT_THAT(
+      TestFua(10 * (1200 - kFuAHeaderSize), limits),
+      ElementsAre(1198, 1198, 1198, 1198, 1198, 1198, 1198, 1198, 1198, 1198));
+}
+
+TEST(RtpPacketizerH264Test, RejectsOverlongDataInPacketizationMode0) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  rtc::Buffer frame = CreateFrame({kMaxPayloadSize + 1});
+
+  RtpPacketizerH264 packetizer(frame, limits,
+                               H264PacketizationMode::SingleNalUnit);
+  std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
+
+  EXPECT_THAT(packets, IsEmpty());
+}
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_unittest.cc
new file mode 100644
index 0000000000..53264c6609
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_unittest.cc
@@ -0,0 +1,283 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format.h"
+
+#include <memory>
+#include <numeric>
+
+#include "absl/algorithm/container.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Each;
+using ::testing::ElementsAre;
+using ::testing::Gt;
+using ::testing::IsEmpty;
+using ::testing::Le;
+using ::testing::Not;
+using ::testing::SizeIs;
+
+// Calculate difference between largest and smallest packets respecting sizes
+// adjustement provided by limits,
+// i.e. last packet expected to be smaller than 'average' by reduction_len.
+int EffectivePacketsSizeDifference(
+    std::vector<int> sizes,
+    const RtpPacketizer::PayloadSizeLimits& limits) {
+  // Account for larger last packet header.
+  sizes.back() += limits.last_packet_reduction_len;
+
+  auto minmax = absl::c_minmax_element(sizes);
+  // MAX-MIN
+  return *minmax.second - *minmax.first;
+}
+
+int Sum(const std::vector<int>& sizes) {
+  return absl::c_accumulate(sizes, 0);
+}
+
+TEST(RtpPacketizerSplitAboutEqually, AllPacketsAreEqualSumToPayloadLen) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 5;
+  limits.last_packet_reduction_len = 2;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(13, limits);
+
+  EXPECT_THAT(Sum(payload_sizes), 13);
+}
+
+TEST(RtpPacketizerSplitAboutEqually, AllPacketsAreEqualRespectsMaxPayloadSize) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 5;
+  limits.last_packet_reduction_len = 2;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(13, limits);
+
+  EXPECT_THAT(payload_sizes, Each(Le(limits.max_payload_len)));
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     AllPacketsAreEqualRespectsFirstPacketReduction) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 5;
+  limits.first_packet_reduction_len = 2;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(13, limits);
+
+  ASSERT_THAT(payload_sizes, Not(IsEmpty()));
+  EXPECT_EQ(payload_sizes.front() + limits.first_packet_reduction_len,
+            limits.max_payload_len);
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     AllPacketsAreEqualRespectsLastPacketReductionLength) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 5;
+  limits.last_packet_reduction_len = 2;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(13, limits);
+
+  ASSERT_THAT(payload_sizes, Not(IsEmpty()));
+  EXPECT_LE(payload_sizes.back() + limits.last_packet_reduction_len,
+            limits.max_payload_len);
+}
+
+TEST(RtpPacketizerSplitAboutEqually, AllPacketsAreEqualInSize) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 5;
+  limits.last_packet_reduction_len = 2;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(13, limits);
+
+  EXPECT_EQ(EffectivePacketsSizeDifference(payload_sizes, limits), 0);
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     AllPacketsAreEqualGeneratesMinimumNumberOfPackets) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 5;
+  limits.last_packet_reduction_len = 2;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(13, limits);
+  // Computed by hand. 3 packets would have exactly capacity 3*5-2=13
+  // (max length - for each packet minus last packet reduction).
+  EXPECT_THAT(payload_sizes, SizeIs(3));
+}
+
+TEST(RtpPacketizerSplitAboutEqually, SomePacketsAreSmallerSumToPayloadLen) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 7;
+  limits.last_packet_reduction_len = 5;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(28, limits);
+
+  EXPECT_THAT(Sum(payload_sizes), 28);
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     SomePacketsAreSmallerRespectsMaxPayloadSize) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 7;
+  limits.last_packet_reduction_len = 5;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(28, limits);
+
+  EXPECT_THAT(payload_sizes, Each(Le(limits.max_payload_len)));
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     SomePacketsAreSmallerRespectsFirstPacketReduction) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 7;
+  limits.first_packet_reduction_len = 5;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(28, limits);
+
+  EXPECT_LE(payload_sizes.front() + limits.first_packet_reduction_len,
+            limits.max_payload_len);
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     SomePacketsAreSmallerRespectsLastPacketReductionLength) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 7;
+  limits.last_packet_reduction_len = 5;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(28, limits);
+
+  EXPECT_LE(payload_sizes.back(),
+            limits.max_payload_len - limits.last_packet_reduction_len);
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     SomePacketsAreSmallerPacketsAlmostEqualInSize) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 7;
+  limits.last_packet_reduction_len = 5;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(28, limits);
+
+  EXPECT_LE(EffectivePacketsSizeDifference(payload_sizes, limits), 1);
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     SomePacketsAreSmallerGeneratesMinimumNumberOfPackets) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 7;
+  limits.last_packet_reduction_len = 5;
+
+  std::vector<int> payload_sizes = RtpPacketizer::SplitAboutEqually(24, limits);
+  // Computed by hand. 4 packets would have capacity 4*7-5=23 (max length -
+  // for each packet minus last packet reduction).
+  // 5 packets is enough for kPayloadSize.
+  EXPECT_THAT(payload_sizes, SizeIs(5));
+}
+
+TEST(RtpPacketizerSplitAboutEqually, GivesNonZeroPayloadLengthEachPacket) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 600;
+  limits.first_packet_reduction_len = 500;
+  limits.last_packet_reduction_len = 550;
+
+  // Naive implementation would split 1450 payload + 1050 reduction bytes into 5
+  // packets 500 bytes each, thus leaving first packet zero bytes and even less
+  // to last packet.
+  std::vector<int> payload_sizes =
+      RtpPacketizer::SplitAboutEqually(1450, limits);
+
+  EXPECT_EQ(Sum(payload_sizes), 1450);
+  EXPECT_THAT(payload_sizes, Each(Gt(0)));
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     IgnoresFirstAndLastPacketReductionWhenPayloadFitsIntoSinglePacket) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 30;
+  limits.first_packet_reduction_len = 29;
+  limits.last_packet_reduction_len = 29;
+  limits.single_packet_reduction_len = 10;
+
+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(20, limits), ElementsAre(20));
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     OnePacketWhenExtraSpaceIsEnoughForSinglePacketReduction) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 30;
+  limits.single_packet_reduction_len = 10;
+
+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(20, limits), ElementsAre(20));
+}
+
+TEST(RtpPacketizerSplitAboutEqually,
+     TwoPacketsWhenExtraSpaceIsTooSmallForSinglePacketReduction) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 29;
+  limits.first_packet_reduction_len = 3;
+  limits.last_packet_reduction_len = 1;
+  limits.single_packet_reduction_len = 10;
+
+  // First packet needs two more extra bytes compared to last one,
+  // so should have two less payload bytes.
+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(20, limits), ElementsAre(9, 11));
+}
+
+TEST(RtpPacketizerSplitAboutEqually, RejectsZeroMaxPayloadLen) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 0;
+
+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(20, limits), IsEmpty());
+}
+
+TEST(RtpPacketizerSplitAboutEqually, RejectsZeroFirstPacketLen) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 5;
+  limits.first_packet_reduction_len = 5;
+
+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(20, limits), IsEmpty());
+}
+
+TEST(RtpPacketizerSplitAboutEqually, RejectsZeroLastPacketLen) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 5;
+  limits.last_packet_reduction_len = 5;
+
+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(20, limits), IsEmpty());
+}
+
+TEST(RtpPacketizerSplitAboutEqually, CantPutSinglePayloadByteInTwoPackets) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 10;
+  limits.single_packet_reduction_len = 10;
+
+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(1, limits), IsEmpty());
+}
+
+TEST(RtpPacketizerSplitAboutEqually, CanPutTwoPayloadBytesInTwoPackets) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 10;
+  limits.single_packet_reduction_len = 10;
+
+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(2, limits), ElementsAre(1, 1));
+}
+
+TEST(RtpPacketizerSplitAboutEqually, CanPutSinglePayloadByteInOnePacket) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 11;
+  limits.single_packet_reduction_len = 10;
+
+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(1, limits), ElementsAre(1));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
new file mode 100644
index 0000000000..f5c7f2ee29
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
+
+#include <string.h>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+static const size_t kGenericHeaderLength = 1;
+static const size_t kExtendedHeaderLength = 2;
+
+RtpPacketizerGeneric::RtpPacketizerGeneric(
+    rtc::ArrayView<const uint8_t> payload,
+    PayloadSizeLimits limits,
+    const RTPVideoHeader& rtp_video_header)
+    : remaining_payload_(payload) {
+  BuildHeader(rtp_video_header);
+
+  limits.max_payload_len -= header_size_;
+  payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  current_packet_ = payload_sizes_.begin();
+}
+
+RtpPacketizerGeneric::RtpPacketizerGeneric(
+    rtc::ArrayView<const uint8_t> payload,
+    PayloadSizeLimits limits)
+    : header_size_(0), remaining_payload_(payload) {
+  payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  current_packet_ = payload_sizes_.begin();
+}
+
+RtpPacketizerGeneric::~RtpPacketizerGeneric() = default;
+
+size_t RtpPacketizerGeneric::NumPackets() const {
+  return payload_sizes_.end() - current_packet_;
+}
+
+bool RtpPacketizerGeneric::NextPacket(RtpPacketToSend* packet) {
+  RTC_DCHECK(packet);
+  if (current_packet_ == payload_sizes_.end())
+    return false;
+
+  size_t next_packet_payload_len = *current_packet_;
+
+  uint8_t* out_ptr =
+      packet->AllocatePayload(header_size_ + next_packet_payload_len);
+  RTC_CHECK(out_ptr);
+
+  if (header_size_ > 0) {
+    memcpy(out_ptr, header_, header_size_);
+    // Remove first-packet bit, following packets are intermediate.
+    header_[0] &= ~RtpFormatVideoGeneric::kFirstPacketBit;
+  }
+
+  memcpy(out_ptr + header_size_, remaining_payload_.data(),
+         next_packet_payload_len);
+
+  remaining_payload_ = remaining_payload_.subview(next_packet_payload_len);
+
+  ++current_packet_;
+
+  // Packets left to produce and data left to split should end at the same time.
+  RTC_DCHECK_EQ(current_packet_ == payload_sizes_.end(),
+                remaining_payload_.empty());
+
+  packet->SetMarker(remaining_payload_.empty());
+  return true;
+}
+
+void RtpPacketizerGeneric::BuildHeader(const RTPVideoHeader& rtp_video_header) {
+  header_size_ = kGenericHeaderLength;
+  header_[0] = RtpFormatVideoGeneric::kFirstPacketBit;
+  if (rtp_video_header.frame_type == VideoFrameType::kVideoFrameKey) {
+    header_[0] |= RtpFormatVideoGeneric::kKeyFrameBit;
+  }
+  if (const auto* generic_header = absl::get_if<RTPVideoHeaderLegacyGeneric>(
+          &rtp_video_header.video_type_header)) {
+    // Store bottom 15 bits of the picture id. Only 15 bits are used for
+    // compatibility with other packetizer implemenetations.
+    uint16_t picture_id = generic_header->picture_id;
+    header_[0] |= RtpFormatVideoGeneric::kExtendedHeaderBit;
+    header_[1] = (picture_id >> 8) & 0x7F;
+    header_[2] = picture_id & 0xFF;
+    header_size_ += kExtendedHeaderLength;
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
new file mode 100644
index 0000000000..fd44bd1980
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+
+namespace webrtc {
+
+class RtpPacketToSend;
+struct RTPVideoHeader;
+
+namespace RtpFormatVideoGeneric {
+static const uint8_t kKeyFrameBit = 0x01;
+static const uint8_t kFirstPacketBit = 0x02;
+// If this bit is set, there will be an extended header contained in this
+// packet. This was added later so old clients will not send this.
+static const uint8_t kExtendedHeaderBit = 0x04;
+}  // namespace RtpFormatVideoGeneric
+
+class RtpPacketizerGeneric : public RtpPacketizer {
+ public:
+  // Initialize with payload from encoder.
+  // The payload_data must be exactly one encoded generic frame.
+  // Packets returned by `NextPacket` will contain the generic payload header.
+  RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload,
+                       PayloadSizeLimits limits,
+                       const RTPVideoHeader& rtp_video_header);
+  // Initialize with payload from encoder.
+  // The payload_data must be exactly one encoded generic frame.
+  // Packets returned by `NextPacket` will contain raw payload without the
+  // generic payload header.
+  RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload,
+                       PayloadSizeLimits limits);
+
+  ~RtpPacketizerGeneric() override;
+
+  RtpPacketizerGeneric(const RtpPacketizerGeneric&) = delete;
+  RtpPacketizerGeneric& operator=(const RtpPacketizerGeneric&) = delete;
+
+  size_t NumPackets() const override;
+
+  // Get the next payload.
+  // Write payload and set marker bit of the `packet`.
+  // Returns true on success, false otherwise.
+  bool NextPacket(RtpPacketToSend* packet) override;
+
+ private:
+  // Fills header_ and header_size_ members.
+  void BuildHeader(const RTPVideoHeader& rtp_video_header);
+
+  uint8_t header_[3];
+  size_t header_size_;
+  rtc::ArrayView<const uint8_t> remaining_payload_;
+  std::vector<int> payload_sizes_;
+  std::vector<int>::const_iterator current_packet_;
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
new file mode 100644
index 0000000000..d83c3b03c9
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Contains;
+using ::testing::Each;
+using ::testing::ElementsAreArray;
+using ::testing::Le;
+
+constexpr RtpPacketizer::PayloadSizeLimits kNoSizeLimits;
+
+std::vector<int> NextPacketFillPayloadSizes(RtpPacketizerGeneric* packetizer) {
+  RtpPacketToSend packet(nullptr);
+  std::vector<int> result;
+  while (packetizer->NextPacket(&packet)) {
+    result.push_back(packet.payload_size());
+  }
+  return result;
+}
+
+TEST(RtpPacketizerVideoGeneric, RespectsMaxPayloadSize) {
+  const size_t kPayloadSize = 50;
+  const uint8_t kPayload[kPayloadSize] = {};
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 6;
+  RtpPacketizerGeneric packetizer(kPayload, limits, RTPVideoHeader());
+
+  std::vector<int> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
+
+  EXPECT_THAT(payload_sizes, Each(Le(limits.max_payload_len)));
+}
+
+TEST(RtpPacketizerVideoGeneric, UsesMaxPayloadSize) {
+  const size_t kPayloadSize = 50;
+  const uint8_t kPayload[kPayloadSize] = {};
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 6;
+  RtpPacketizerGeneric packetizer(kPayload, limits, RTPVideoHeader());
+
+  std::vector<int> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
+
+  // With kPayloadSize > max_payload_len^2, there should be packets that use
+  // all the payload, otherwise it is possible to use less packets.
+  EXPECT_THAT(payload_sizes, Contains(limits.max_payload_len));
+}
+
+TEST(RtpPacketizerVideoGeneric, WritesExtendedHeaderWhenPictureIdIsSet) {
+  const size_t kPayloadSize = 13;
+  const uint8_t kPayload[kPayloadSize] = {};
+
+  RTPVideoHeader rtp_video_header;
+  rtp_video_header.video_type_header.emplace<RTPVideoHeaderLegacyGeneric>()
+      .picture_id = 37;
+  rtp_video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  RtpPacketizerGeneric packetizer(kPayload, kNoSizeLimits, rtp_video_header);
+
+  RtpPacketToSend packet(nullptr);
+  ASSERT_TRUE(packetizer.NextPacket(&packet));
+
+  rtc::ArrayView<const uint8_t> payload = packet.payload();
+  EXPECT_EQ(payload.size(), 3 + kPayloadSize);
+  EXPECT_TRUE(payload[0] & 0x04);  // Extended header bit is set.
+  // Frame id is 37.
+  EXPECT_EQ(0u, payload[1]);
+  EXPECT_EQ(37u, payload[2]);
+}
+
+TEST(RtpPacketizerVideoGeneric, RespectsMaxPayloadSizeWithExtendedHeader) {
+  const int kPayloadSize = 50;
+  const uint8_t kPayload[kPayloadSize] = {};
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 6;
+  RTPVideoHeader rtp_video_header;
+  rtp_video_header.video_type_header.emplace<RTPVideoHeaderLegacyGeneric>()
+      .picture_id = 37;
+  RtpPacketizerGeneric packetizer(kPayload, limits, rtp_video_header);
+
+  std::vector<int> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
+
+  EXPECT_THAT(payload_sizes, Each(Le(limits.max_payload_len)));
+}
+
+TEST(RtpPacketizerVideoGeneric, UsesMaxPayloadSizeWithExtendedHeader) {
+  const int kPayloadSize = 50;
+  const uint8_t kPayload[kPayloadSize] = {};
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 6;
+  RTPVideoHeader rtp_video_header;
+  rtp_video_header.video_type_header.emplace<RTPVideoHeaderLegacyGeneric>()
+      .picture_id = 37;
+  RtpPacketizerGeneric packetizer(kPayload, limits, rtp_video_header);
+  std::vector<int> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
+
+  // With kPayloadSize > max_payload_len^2, there should be packets that use
+  // all the payload, otherwise it is possible to use less packets.
+  EXPECT_THAT(payload_sizes, Contains(limits.max_payload_len));
+}
+
+TEST(RtpPacketizerVideoGeneric, FrameIdOver15bitsWrapsAround) {
+  const int kPayloadSize = 13;
+  const uint8_t kPayload[kPayloadSize] = {};
+
+  RTPVideoHeader rtp_video_header;
+  rtp_video_header.video_type_header.emplace<RTPVideoHeaderLegacyGeneric>()
+      .picture_id = 0x8137;
+  rtp_video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  RtpPacketizerGeneric packetizer(kPayload, kNoSizeLimits, rtp_video_header);
+
+  RtpPacketToSend packet(nullptr);
+  ASSERT_TRUE(packetizer.NextPacket(&packet));
+
+  rtc::ArrayView<const uint8_t> payload = packet.payload();
+  EXPECT_TRUE(payload[0] & 0x04);  // Extended header bit is set.
+  // Frame id is 0x137.
+  EXPECT_EQ(0x01u, payload[1]);
+  EXPECT_EQ(0x37u, payload[2]);
+}
+
+TEST(RtpPacketizerVideoGeneric, NoFrameIdDoesNotWriteExtendedHeader) {
+  const int kPayloadSize = 13;
+  const uint8_t kPayload[kPayloadSize] = {};
+
+  RtpPacketizerGeneric packetizer(kPayload, kNoSizeLimits, RTPVideoHeader());
+
+  RtpPacketToSend packet(nullptr);
+  ASSERT_TRUE(packetizer.NextPacket(&packet));
+
+  rtc::ArrayView<const uint8_t> payload = packet.payload();
+  EXPECT_FALSE(payload[0] & 0x04);
+}
+
+TEST(RtpPacketizerVideoGeneric, DoesNotWriteHeaderForRawPayload) {
+  const uint8_t kPayload[] = {0x05, 0x25, 0x52};
+
+  RtpPacketizerGeneric packetizer(kPayload, kNoSizeLimits);
+
+  RtpPacketToSend packet(nullptr);
+  ASSERT_TRUE(packetizer.NextPacket(&packet));
+
+  rtc::ArrayView<const uint8_t> payload = packet.payload();
+  EXPECT_THAT(payload, ElementsAreArray(kPayload));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
new file mode 100644
index 0000000000..ae5f4e50a4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+
+#include <stdint.h>
+#include <string.h>  // memcpy
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kXBit = 0x80;
+constexpr int kNBit = 0x20;
+constexpr int kSBit = 0x10;
+constexpr int kKeyIdxField = 0x1F;
+constexpr int kIBit = 0x80;
+constexpr int kLBit = 0x40;
+constexpr int kTBit = 0x20;
+constexpr int kKBit = 0x10;
+constexpr int kYBit = 0x20;
+
+bool ValidateHeader(const RTPVideoHeaderVP8& hdr_info) {
+  if (hdr_info.pictureId != kNoPictureId) {
+    RTC_DCHECK_GE(hdr_info.pictureId, 0);
+    RTC_DCHECK_LE(hdr_info.pictureId, 0x7FFF);
+  }
+  if (hdr_info.tl0PicIdx != kNoTl0PicIdx) {
+    RTC_DCHECK_GE(hdr_info.tl0PicIdx, 0);
+    RTC_DCHECK_LE(hdr_info.tl0PicIdx, 0xFF);
+  }
+  if (hdr_info.temporalIdx != kNoTemporalIdx) {
+    RTC_DCHECK_GE(hdr_info.temporalIdx, 0);
+    RTC_DCHECK_LE(hdr_info.temporalIdx, 3);
+  } else {
+    RTC_DCHECK(!hdr_info.layerSync);
+  }
+  if (hdr_info.keyIdx != kNoKeyIdx) {
+    RTC_DCHECK_GE(hdr_info.keyIdx, 0);
+    RTC_DCHECK_LE(hdr_info.keyIdx, 0x1F);
+  }
+  return true;
+}
+
+}  // namespace
+
+RtpPacketizerVp8::RtpPacketizerVp8(rtc::ArrayView<const uint8_t> payload,
+                                   PayloadSizeLimits limits,
+                                   const RTPVideoHeaderVP8& hdr_info)
+    : hdr_(BuildHeader(hdr_info)), remaining_payload_(payload) {
+  limits.max_payload_len -= hdr_.size();
+  payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  current_packet_ = payload_sizes_.begin();
+}
+
+RtpPacketizerVp8::~RtpPacketizerVp8() = default;
+
+size_t RtpPacketizerVp8::NumPackets() const {
+  return payload_sizes_.end() - current_packet_;
+}
+
+bool RtpPacketizerVp8::NextPacket(RtpPacketToSend* packet) {
+  RTC_DCHECK(packet);
+  if (current_packet_ == payload_sizes_.end()) {
+    return false;
+  }
+
+  size_t packet_payload_len = *current_packet_;
+  ++current_packet_;
+
+  uint8_t* buffer = packet->AllocatePayload(hdr_.size() + packet_payload_len);
+  RTC_CHECK(buffer);
+
+  memcpy(buffer, hdr_.data(), hdr_.size());
+  memcpy(buffer + hdr_.size(), remaining_payload_.data(), packet_payload_len);
+
+  remaining_payload_ = remaining_payload_.subview(packet_payload_len);
+  hdr_[0] &= (~kSBit);  //  Clear 'Start of partition' bit.
+  packet->SetMarker(current_packet_ == payload_sizes_.end());
+  return true;
+}
+
+RtpPacketizerVp8::RawHeader RtpPacketizerVp8::BuildHeader(
+    const RTPVideoHeaderVP8& header) {
+  // VP8 payload descriptor
+  // https://datatracker.ietf.org/doc/html/rfc7741#section-4.2
+  //
+  //       0 1 2 3 4 5 6 7
+  //      +-+-+-+-+-+-+-+-+
+  //      |X|R|N|S|R| PID | (REQUIRED)
+  //      +-+-+-+-+-+-+-+-+
+  // X:   |I|L|T|K| RSV   | (OPTIONAL)
+  //      +-+-+-+-+-+-+-+-+
+  // I:   |M| PictureID   | (OPTIONAL)
+  //      +-+-+-+-+-+-+-+-+
+  //      |   PictureID   |
+  //      +-+-+-+-+-+-+-+-+
+  // L:   |   TL0PICIDX   | (OPTIONAL)
+  //      +-+-+-+-+-+-+-+-+
+  // T/K: |TID|Y| KEYIDX  | (OPTIONAL)
+  //      +-+-+-+-+-+-+-+-+
+  RTC_DCHECK(ValidateHeader(header));
+
+  RawHeader result;
+  bool tid_present = header.temporalIdx != kNoTemporalIdx;
+  bool keyid_present = header.keyIdx != kNoKeyIdx;
+  bool tl0_pid_present = header.tl0PicIdx != kNoTl0PicIdx;
+  bool pid_present = header.pictureId != kNoPictureId;
+  uint8_t x_field = 0;
+  if (pid_present)
+    x_field |= kIBit;
+  if (tl0_pid_present)
+    x_field |= kLBit;
+  if (tid_present)
+    x_field |= kTBit;
+  if (keyid_present)
+    x_field |= kKBit;
+
+  uint8_t flags = 0;
+  if (x_field != 0)
+    flags |= kXBit;
+  if (header.nonReference)
+    flags |= kNBit;
+  // Create header as first packet in the frame. NextPacket() will clear it
+  // after first use.
+  flags |= kSBit;
+  result.push_back(flags);
+  if (x_field == 0) {
+    return result;
+  }
+  result.push_back(x_field);
+  if (pid_present) {
+    const uint16_t pic_id = static_cast<uint16_t>(header.pictureId);
+    result.push_back(0x80 | ((pic_id >> 8) & 0x7F));
+    result.push_back(pic_id & 0xFF);
+  }
+  if (tl0_pid_present) {
+    result.push_back(header.tl0PicIdx);
+  }
+  if (tid_present || keyid_present) {
+    uint8_t data_field = 0;
+    if (tid_present) {
+      data_field |= header.temporalIdx << 6;
+      if (header.layerSync)
+        data_field |= kYBit;
+    }
+    if (keyid_present) {
+      data_field |= (header.keyIdx & kKeyIdxField);
+    }
+    result.push_back(data_field);
+  }
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
new file mode 100644
index 0000000000..d1f569a946
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains the declaration of the VP8 packetizer class.
+ * A packetizer object is created for each encoded video frame. The
+ * constructor is called with the payload data and size,
+ * together with the fragmentation information and a packetizer mode
+ * of choice. Alternatively, if no fragmentation info is available, the
+ * second constructor can be used with only payload data and size; in that
+ * case the mode kEqualSize is used.
+ *
+ * After creating the packetizer, the method NextPacket is called
+ * repeatedly to get all packets for the frame. The method returns
+ * false as long as there are more packets left to fetch.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
+
+namespace webrtc {
+
+// Packetizer for VP8.
+class RtpPacketizerVp8 : public RtpPacketizer {
+ public:
+  // Initialize with payload from encoder.
+  // The payload_data must be exactly one encoded VP8 frame.
+  RtpPacketizerVp8(rtc::ArrayView<const uint8_t> payload,
+                   PayloadSizeLimits limits,
+                   const RTPVideoHeaderVP8& hdr_info);
+
+  ~RtpPacketizerVp8() override;
+
+  RtpPacketizerVp8(const RtpPacketizerVp8&) = delete;
+  RtpPacketizerVp8& operator=(const RtpPacketizerVp8&) = delete;
+
+  size_t NumPackets() const override;
+
+  // Get the next payload with VP8 payload header.
+  // Write payload and set marker bit of the `packet`.
+  // Returns true on success, false otherwise.
+  bool NextPacket(RtpPacketToSend* packet) override;
+
+ private:
+  // VP8 header can use up to 6 bytes.
+  using RawHeader = absl::InlinedVector<uint8_t, 6>;
+  static RawHeader BuildHeader(const RTPVideoHeaderVP8& header);
+
+  RawHeader hdr_;
+  rtc::ArrayView<const uint8_t> remaining_payload_;
+  std::vector<int> payload_sizes_;
+  std::vector<int>::const_iterator current_packet_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
new file mode 100644
index 0000000000..0088ff8f31
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
+
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+// VP8 payload descriptor
+// https://datatracker.ietf.org/doc/html/rfc7741#section-4.2
+//
+//       0 1 2 3 4 5 6 7
+//      +-+-+-+-+-+-+-+-+
+//      |X|R|N|S|R| PID | (REQUIRED)
+//      +-+-+-+-+-+-+-+-+
+// X:   |I|L|T|K| RSV   | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+// I:   |M| PictureID   | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+//      |   PictureID   |
+//      +-+-+-+-+-+-+-+-+
+// L:   |   TL0PICIDX   | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+// T/K: |TID|Y| KEYIDX  | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAreArray;
+
+constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr;
+
+int Bit(uint8_t byte, int position) {
+  return (byte >> position) & 0x01;
+}
+
+}  // namespace
+
+RtpFormatVp8TestHelper::RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr,
+                                               size_t payload_len)
+    : hdr_info_(hdr), payload_(payload_len) {
+  for (size_t i = 0; i < payload_.size(); ++i) {
+    payload_[i] = i;
+  }
+}
+
+RtpFormatVp8TestHelper::~RtpFormatVp8TestHelper() = default;
+
+void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
+    RtpPacketizerVp8* packetizer,
+    rtc::ArrayView<const size_t> expected_sizes) {
+  EXPECT_EQ(packetizer->NumPackets(), expected_sizes.size());
+  const uint8_t* data_ptr = payload_.begin();
+  RtpPacketToSend packet(kNoExtensions);
+  for (size_t i = 0; i < expected_sizes.size(); ++i) {
+    EXPECT_TRUE(packetizer->NextPacket(&packet));
+    auto rtp_payload = packet.payload();
+    EXPECT_EQ(rtp_payload.size(), expected_sizes[i]);
+
+    int payload_offset = CheckHeader(rtp_payload, /*first=*/i == 0);
+    // Verify that the payload (i.e., after the headers) of the packet is
+    // identical to the expected (as found in data_ptr).
+    auto vp8_payload = rtp_payload.subview(payload_offset);
+    ASSERT_GE(payload_.end() - data_ptr, static_cast<int>(vp8_payload.size()));
+    EXPECT_THAT(vp8_payload, ElementsAreArray(data_ptr, vp8_payload.size()));
+    data_ptr += vp8_payload.size();
+  }
+  EXPECT_EQ(payload_.end() - data_ptr, 0);
+}
+
+int RtpFormatVp8TestHelper::CheckHeader(rtc::ArrayView<const uint8_t> buffer,
+                                        bool first) {
+  int x_bit = Bit(buffer[0], 7);
+  EXPECT_EQ(Bit(buffer[0], 6), 0);  // Reserved.
+  EXPECT_EQ(Bit(buffer[0], 5), hdr_info_->nonReference ? 1 : 0);
+  EXPECT_EQ(Bit(buffer[0], 4), first ? 1 : 0);
+  EXPECT_EQ(buffer[0] & 0x0f, 0);  // RtpPacketizerVp8 always uses partition 0.
+
+  int payload_offset = 1;
+  if (hdr_info_->pictureId != kNoPictureId ||
+      hdr_info_->temporalIdx != kNoTemporalIdx ||
+      hdr_info_->tl0PicIdx != kNoTl0PicIdx || hdr_info_->keyIdx != kNoKeyIdx) {
+    EXPECT_EQ(x_bit, 1);
+    ++payload_offset;
+    CheckPictureID(buffer, &payload_offset);
+    CheckTl0PicIdx(buffer, &payload_offset);
+    CheckTIDAndKeyIdx(buffer, &payload_offset);
+    EXPECT_EQ(buffer[1] & 0x07, 0);  // Reserved.
+  } else {
+    EXPECT_EQ(x_bit, 0);
+  }
+
+  return payload_offset;
+}
+
+// Verify that the I bit and the PictureID field are both set in accordance
+// with the information in hdr_info_->pictureId.
+void RtpFormatVp8TestHelper::CheckPictureID(
+    rtc::ArrayView<const uint8_t> buffer,
+    int* offset) {
+  int i_bit = Bit(buffer[1], 7);
+  if (hdr_info_->pictureId != kNoPictureId) {
+    EXPECT_EQ(i_bit, 1);
+    int two_byte_picture_id = Bit(buffer[*offset], 7);
+    EXPECT_EQ(two_byte_picture_id, 1);
+    EXPECT_EQ(buffer[*offset] & 0x7F, (hdr_info_->pictureId >> 8) & 0x7F);
+    EXPECT_EQ(buffer[(*offset) + 1], hdr_info_->pictureId & 0xFF);
+    (*offset) += 2;
+  } else {
+    EXPECT_EQ(i_bit, 0);
+  }
+}
+
+// Verify that the L bit and the TL0PICIDX field are both set in accordance
+// with the information in hdr_info_->tl0PicIdx.
+void RtpFormatVp8TestHelper::CheckTl0PicIdx(
+    rtc::ArrayView<const uint8_t> buffer,
+    int* offset) {
+  int l_bit = Bit(buffer[1], 6);
+  if (hdr_info_->tl0PicIdx != kNoTl0PicIdx) {
+    EXPECT_EQ(l_bit, 1);
+    EXPECT_EQ(buffer[*offset], hdr_info_->tl0PicIdx);
+    ++*offset;
+  } else {
+    EXPECT_EQ(l_bit, 0);
+  }
+}
+
+// Verify that the T bit and the TL0PICIDX field, and the K bit and KEYIDX
+// field are all set in accordance with the information in
+// hdr_info_->temporalIdx and hdr_info_->keyIdx, respectively.
+void RtpFormatVp8TestHelper::CheckTIDAndKeyIdx(
+    rtc::ArrayView<const uint8_t> buffer,
+    int* offset) {
+  int t_bit = Bit(buffer[1], 5);
+  int k_bit = Bit(buffer[1], 4);
+  if (hdr_info_->temporalIdx == kNoTemporalIdx &&
+      hdr_info_->keyIdx == kNoKeyIdx) {
+    EXPECT_EQ(t_bit, 0);
+    EXPECT_EQ(k_bit, 0);
+    return;
+  }
+  int temporal_id = (buffer[*offset] & 0xC0) >> 6;
+  int y_bit = Bit(buffer[*offset], 5);
+  int key_idx = buffer[*offset] & 0x1f;
+  if (hdr_info_->temporalIdx != kNoTemporalIdx) {
+    EXPECT_EQ(t_bit, 1);
+    EXPECT_EQ(temporal_id, hdr_info_->temporalIdx);
+    EXPECT_EQ(y_bit, hdr_info_->layerSync ? 1 : 0);
+  } else {
+    EXPECT_EQ(t_bit, 0);
+    EXPECT_EQ(temporal_id, 0);
+    EXPECT_EQ(y_bit, 0);
+  }
+  if (hdr_info_->keyIdx != kNoKeyIdx) {
+    EXPECT_EQ(k_bit, 1);
+    EXPECT_EQ(key_idx, hdr_info_->keyIdx);
+  } else {
+    EXPECT_EQ(k_bit, 0);
+    EXPECT_EQ(key_idx, 0);
+  }
+  ++*offset;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
new file mode 100644
index 0000000000..3ecaa476da
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains the class RtpFormatVp8TestHelper. The class is
+// responsible for setting up a fake VP8 bitstream according to the
+// RTPVideoHeaderVP8 header. The packetizer can then be provided to this helper
+// class, which will then extract all packets and compare to the expected
+// outcome.
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class RtpFormatVp8TestHelper {
+ public:
+  RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr, size_t payload_len);
+  ~RtpFormatVp8TestHelper();
+
+  RtpFormatVp8TestHelper(const RtpFormatVp8TestHelper&) = delete;
+  RtpFormatVp8TestHelper& operator=(const RtpFormatVp8TestHelper&) = delete;
+
+  void GetAllPacketsAndCheck(RtpPacketizerVp8* packetizer,
+                             rtc::ArrayView<const size_t> expected_sizes);
+
+  rtc::ArrayView<const uint8_t> payload() const { return payload_; }
+  size_t payload_size() const { return payload_.size(); }
+
+ private:
+  // Returns header size, i.e. payload offset.
+  int CheckHeader(rtc::ArrayView<const uint8_t> rtp_payload, bool first);
+  void CheckPictureID(rtc::ArrayView<const uint8_t> rtp_payload, int* offset);
+  void CheckTl0PicIdx(rtc::ArrayView<const uint8_t> rtp_payload, int* offset);
+  void CheckTIDAndKeyIdx(rtc::ArrayView<const uint8_t> rtp_payload,
+                         int* offset);
+  void CheckPayload(const uint8_t* data_ptr);
+
+  const RTPVideoHeaderVP8* const hdr_info_;
+  rtc::Buffer payload_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
new file mode 100644
index 0000000000..7934ff8ea9
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+
+#include <memory>
+
+#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr RtpPacketizer::PayloadSizeLimits kNoSizeLimits;
+
+TEST(RtpPacketizerVp8Test, ResultPacketsAreAlmostEqualSize) {
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  hdr_info.pictureId = 200;
+  RtpFormatVp8TestHelper helper(&hdr_info, /*payload_len=*/30);
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 12;  // Small enough to produce 4 packets.
+  RtpPacketizerVp8 packetizer(helper.payload(), limits, hdr_info);
+
+  const size_t kExpectedSizes[] = {11, 11, 12, 12};
+  helper.GetAllPacketsAndCheck(&packetizer, kExpectedSizes);
+}
+
+TEST(RtpPacketizerVp8Test, EqualSizeWithLastPacketReduction) {
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  hdr_info.pictureId = 200;
+  RtpFormatVp8TestHelper helper(&hdr_info, /*payload_len=*/43);
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 15;  // Small enough to produce 5 packets.
+  limits.last_packet_reduction_len = 5;
+  RtpPacketizerVp8 packetizer(helper.payload(), limits, hdr_info);
+
+  // Calculated by hand. VP8 payload descriptors are 4 byte each. 5 packets is
+  // minimum possible to fit 43 payload bytes into packets with capacity of
+  // 15 - 4 = 11 and leave 5 free bytes in the last packet. All packets are
+  // almost equal in size, even last packet if counted with free space (which
+  // will be filled up the stack by extra long RTP header).
+  const size_t kExpectedSizes[] = {13, 13, 14, 14, 9};
+  helper.GetAllPacketsAndCheck(&packetizer, kExpectedSizes);
+}
+
+// Verify that non-reference bit is set.
+TEST(RtpPacketizerVp8Test, NonReferenceBit) {
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  hdr_info.nonReference = true;
+  RtpFormatVp8TestHelper helper(&hdr_info, /*payload_len=*/30);
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 25;  // Small enough to produce two packets.
+  RtpPacketizerVp8 packetizer(helper.payload(), limits, hdr_info);
+
+  const size_t kExpectedSizes[] = {16, 16};
+  helper.GetAllPacketsAndCheck(&packetizer, kExpectedSizes);
+}
+
+// Verify Tl0PicIdx and TID fields, and layerSync bit.
+TEST(RtpPacketizerVp8Test, Tl0PicIdxAndTID) {
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  hdr_info.tl0PicIdx = 117;
+  hdr_info.temporalIdx = 2;
+  hdr_info.layerSync = true;
+  RtpFormatVp8TestHelper helper(&hdr_info, /*payload_len=*/30);
+
+  RtpPacketizerVp8 packetizer(helper.payload(), kNoSizeLimits, hdr_info);
+
+  const size_t kExpectedSizes[1] = {helper.payload_size() + 4};
+  helper.GetAllPacketsAndCheck(&packetizer, kExpectedSizes);
+}
+
+TEST(RtpPacketizerVp8Test, KeyIdx) {
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  hdr_info.keyIdx = 17;
+  RtpFormatVp8TestHelper helper(&hdr_info, /*payload_len=*/30);
+
+  RtpPacketizerVp8 packetizer(helper.payload(), kNoSizeLimits, hdr_info);
+
+  const size_t kExpectedSizes[1] = {helper.payload_size() + 3};
+  helper.GetAllPacketsAndCheck(&packetizer, kExpectedSizes);
+}
+
+// Verify TID field and KeyIdx field in combination.
+TEST(RtpPacketizerVp8Test, TIDAndKeyIdx) {
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  hdr_info.temporalIdx = 1;
+  hdr_info.keyIdx = 5;
+  RtpFormatVp8TestHelper helper(&hdr_info, /*payload_len=*/30);
+
+  RtpPacketizerVp8 packetizer(helper.payload(), kNoSizeLimits, hdr_info);
+
+  const size_t kExpectedSizes[1] = {helper.payload_size() + 3};
+  helper.GetAllPacketsAndCheck(&packetizer, kExpectedSizes);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
new file mode 100644
index 0000000000..15e059e85c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
@@ -0,0 +1,451 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
+
+#include <string.h>
+
+#include "api/video/video_codec_constants.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "rtc_base/bit_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+#define RETURN_FALSE_ON_ERROR(x) \
+  if (!(x)) {                    \
+    return false;                \
+  }
+
+namespace webrtc {
+namespace {
+// Length of VP9 payload descriptors' fixed part.
+const size_t kFixedPayloadDescriptorBytes = 1;
+
+const uint32_t kReservedBitValue0 = 0;
+
+uint8_t TemporalIdxField(const RTPVideoHeaderVP9& hdr, uint8_t def) {
+  return (hdr.temporal_idx == kNoTemporalIdx) ? def : hdr.temporal_idx;
+}
+
+uint8_t SpatialIdxField(const RTPVideoHeaderVP9& hdr, uint8_t def) {
+  return (hdr.spatial_idx == kNoSpatialIdx) ? def : hdr.spatial_idx;
+}
+
+int16_t Tl0PicIdxField(const RTPVideoHeaderVP9& hdr, uint8_t def) {
+  return (hdr.tl0_pic_idx == kNoTl0PicIdx) ? def : hdr.tl0_pic_idx;
+}
+
+// Picture ID:
+//
+//      +-+-+-+-+-+-+-+-+
+// I:   |M| PICTURE ID  |   M:0 => picture id is 7 bits.
+//      +-+-+-+-+-+-+-+-+   M:1 => picture id is 15 bits.
+// M:   | EXTENDED PID  |
+//      +-+-+-+-+-+-+-+-+
+//
+size_t PictureIdLength(const RTPVideoHeaderVP9& hdr) {
+  if (hdr.picture_id == kNoPictureId)
+    return 0;
+  return (hdr.max_picture_id == kMaxOneBytePictureId) ? 1 : 2;
+}
+
+bool PictureIdPresent(const RTPVideoHeaderVP9& hdr) {
+  return PictureIdLength(hdr) > 0;
+}
+
+// Layer indices:
+//
+// Flexible mode (F=1):     Non-flexible mode (F=0):
+//
+//      +-+-+-+-+-+-+-+-+   +-+-+-+-+-+-+-+-+
+// L:   |  T  |U|  S  |D|   |  T  |U|  S  |D|
+//      +-+-+-+-+-+-+-+-+   +-+-+-+-+-+-+-+-+
+//                          |   TL0PICIDX   |
+//                          +-+-+-+-+-+-+-+-+
+//
+size_t LayerInfoLength(const RTPVideoHeaderVP9& hdr) {
+  if (hdr.temporal_idx == kNoTemporalIdx && hdr.spatial_idx == kNoSpatialIdx) {
+    return 0;
+  }
+  return hdr.flexible_mode ? 1 : 2;
+}
+
+bool LayerInfoPresent(const RTPVideoHeaderVP9& hdr) {
+  return LayerInfoLength(hdr) > 0;
+}
+
+// Reference indices:
+//
+//      +-+-+-+-+-+-+-+-+                P=1,F=1: At least one reference index
+// P,F: | P_DIFF      |N|  up to 3 times          has to be specified.
+//      +-+-+-+-+-+-+-+-+                    N=1: An additional P_DIFF follows
+//                                                current P_DIFF.
+//
+size_t RefIndicesLength(const RTPVideoHeaderVP9& hdr) {
+  if (!hdr.inter_pic_predicted || !hdr.flexible_mode)
+    return 0;
+
+  RTC_DCHECK_GT(hdr.num_ref_pics, 0U);
+  RTC_DCHECK_LE(hdr.num_ref_pics, kMaxVp9RefPics);
+  return hdr.num_ref_pics;
+}
+
+// Scalability structure (SS).
+//
+//      +-+-+-+-+-+-+-+-+
+// V:   | N_S |Y|G|-|-|-|
+//      +-+-+-+-+-+-+-+-+              -|
+// Y:   |     WIDTH     | (OPTIONAL)    .
+//      +               +               .
+//      |               | (OPTIONAL)    .
+//      +-+-+-+-+-+-+-+-+               . N_S + 1 times
+//      |     HEIGHT    | (OPTIONAL)    .
+//      +               +               .
+//      |               | (OPTIONAL)    .
+//      +-+-+-+-+-+-+-+-+              -|
+// G:   |      N_G      | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+                           -|
+// N_G: |  T  |U| R |-|-| (OPTIONAL)                 .
+//      +-+-+-+-+-+-+-+-+              -|            . N_G times
+//      |    P_DIFF     | (OPTIONAL)    . R times    .
+//      +-+-+-+-+-+-+-+-+              -|           -|
+//
+size_t SsDataLength(const RTPVideoHeaderVP9& hdr) {
+  if (!hdr.ss_data_available)
+    return 0;
+
+  RTC_DCHECK_GT(hdr.num_spatial_layers, 0U);
+  RTC_DCHECK_LE(hdr.num_spatial_layers, kMaxVp9NumberOfSpatialLayers);
+  RTC_DCHECK_LE(hdr.gof.num_frames_in_gof, kMaxVp9FramesInGof);
+  size_t length = 1;  // V
+  if (hdr.spatial_layer_resolution_present) {
+    length += 4 * hdr.num_spatial_layers;  // Y
+  }
+  if (hdr.gof.num_frames_in_gof > 0) {
+    ++length;  // G
+  }
+  // N_G
+  length += hdr.gof.num_frames_in_gof;  // T, U, R
+  for (size_t i = 0; i < hdr.gof.num_frames_in_gof; ++i) {
+    RTC_DCHECK_LE(hdr.gof.num_ref_pics[i], kMaxVp9RefPics);
+    length += hdr.gof.num_ref_pics[i];  // R times
+  }
+  return length;
+}
+
+size_t PayloadDescriptorLengthMinusSsData(const RTPVideoHeaderVP9& hdr) {
+  return kFixedPayloadDescriptorBytes + PictureIdLength(hdr) +
+         LayerInfoLength(hdr) + RefIndicesLength(hdr);
+}
+
+// Picture ID:
+//
+//      +-+-+-+-+-+-+-+-+
+// I:   |M| PICTURE ID  |   M:0 => picture id is 7 bits.
+//      +-+-+-+-+-+-+-+-+   M:1 => picture id is 15 bits.
+// M:   | EXTENDED PID  |
+//      +-+-+-+-+-+-+-+-+
+//
+bool WritePictureId(const RTPVideoHeaderVP9& vp9,
+                    rtc::BitBufferWriter* writer) {
+  bool m_bit = (PictureIdLength(vp9) == 2);
+  RETURN_FALSE_ON_ERROR(writer->WriteBits(m_bit ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.picture_id, m_bit ? 15 : 7));
+  return true;
+}
+
+// Layer indices:
+//
+// Flexible mode (F=1):
+//
+//      +-+-+-+-+-+-+-+-+
+// L:   |  T  |U|  S  |D|
+//      +-+-+-+-+-+-+-+-+
+//
+bool WriteLayerInfoCommon(const RTPVideoHeaderVP9& vp9,
+                          rtc::BitBufferWriter* writer) {
+  RETURN_FALSE_ON_ERROR(writer->WriteBits(TemporalIdxField(vp9, 0), 3));
+  RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.temporal_up_switch ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer->WriteBits(SpatialIdxField(vp9, 0), 3));
+  RETURN_FALSE_ON_ERROR(
+      writer->WriteBits(vp9.inter_layer_predicted ? 1 : 0, 1));
+  return true;
+}
+
+// Non-flexible mode (F=0):
+//
+//      +-+-+-+-+-+-+-+-+
+// L:   |  T  |U|  S  |D|
+//      +-+-+-+-+-+-+-+-+
+//      |   TL0PICIDX   |
+//      +-+-+-+-+-+-+-+-+
+//
+bool WriteLayerInfoNonFlexibleMode(const RTPVideoHeaderVP9& vp9,
+                                   rtc::BitBufferWriter* writer) {
+  RETURN_FALSE_ON_ERROR(writer->WriteUInt8(Tl0PicIdxField(vp9, 0)));
+  return true;
+}
+
+bool WriteLayerInfo(const RTPVideoHeaderVP9& vp9,
+                    rtc::BitBufferWriter* writer) {
+  if (!WriteLayerInfoCommon(vp9, writer))
+    return false;
+
+  if (vp9.flexible_mode)
+    return true;
+
+  return WriteLayerInfoNonFlexibleMode(vp9, writer);
+}
+
+// Reference indices:
+//
+//      +-+-+-+-+-+-+-+-+                P=1,F=1: At least one reference index
+// P,F: | P_DIFF      |N|  up to 3 times          has to be specified.
+//      +-+-+-+-+-+-+-+-+                    N=1: An additional P_DIFF follows
+//                                                current P_DIFF.
+//
+bool WriteRefIndices(const RTPVideoHeaderVP9& vp9,
+                     rtc::BitBufferWriter* writer) {
+  if (!PictureIdPresent(vp9) || vp9.num_ref_pics == 0 ||
+      vp9.num_ref_pics > kMaxVp9RefPics) {
+    return false;
+  }
+  for (uint8_t i = 0; i < vp9.num_ref_pics; ++i) {
+    bool n_bit = !(i == vp9.num_ref_pics - 1);
+    RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.pid_diff[i], 7));
+    RETURN_FALSE_ON_ERROR(writer->WriteBits(n_bit ? 1 : 0, 1));
+  }
+  return true;
+}
+
+// Scalability structure (SS).
+//
+//      +-+-+-+-+-+-+-+-+
+// V:   | N_S |Y|G|-|-|-|
+//      +-+-+-+-+-+-+-+-+              -|
+// Y:   |     WIDTH     | (OPTIONAL)    .
+//      +               +               .
+//      |               | (OPTIONAL)    .
+//      +-+-+-+-+-+-+-+-+               . N_S + 1 times
+//      |     HEIGHT    | (OPTIONAL)    .
+//      +               +               .
+//      |               | (OPTIONAL)    .
+//      +-+-+-+-+-+-+-+-+              -|
+// G:   |      N_G      | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+                           -|
+// N_G: |  T  |U| R |-|-| (OPTIONAL)                 .
+//      +-+-+-+-+-+-+-+-+              -|            . N_G times
+//      |    P_DIFF     | (OPTIONAL)    . R times    .
+//      +-+-+-+-+-+-+-+-+              -|           -|
+//
+bool WriteSsData(const RTPVideoHeaderVP9& vp9, rtc::BitBufferWriter* writer) {
+  RTC_DCHECK_GT(vp9.num_spatial_layers, 0U);
+  RTC_DCHECK_LE(vp9.num_spatial_layers, kMaxVp9NumberOfSpatialLayers);
+  RTC_DCHECK_LE(vp9.gof.num_frames_in_gof, kMaxVp9FramesInGof);
+  bool g_bit = vp9.gof.num_frames_in_gof > 0;
+
+  RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.num_spatial_layers - 1, 3));
+  RETURN_FALSE_ON_ERROR(
+      writer->WriteBits(vp9.spatial_layer_resolution_present ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer->WriteBits(g_bit ? 1 : 0, 1));  // G
+  RETURN_FALSE_ON_ERROR(writer->WriteBits(kReservedBitValue0, 3));
+
+  if (vp9.spatial_layer_resolution_present) {
+    for (size_t i = 0; i < vp9.num_spatial_layers; ++i) {
+      RETURN_FALSE_ON_ERROR(writer->WriteUInt16(vp9.width[i]));
+      RETURN_FALSE_ON_ERROR(writer->WriteUInt16(vp9.height[i]));
+    }
+  }
+  if (g_bit) {
+    RETURN_FALSE_ON_ERROR(writer->WriteUInt8(vp9.gof.num_frames_in_gof));
+  }
+  for (size_t i = 0; i < vp9.gof.num_frames_in_gof; ++i) {
+    RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.gof.temporal_idx[i], 3));
+    RETURN_FALSE_ON_ERROR(
+        writer->WriteBits(vp9.gof.temporal_up_switch[i] ? 1 : 0, 1));
+    RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.gof.num_ref_pics[i], 2));
+    RETURN_FALSE_ON_ERROR(writer->WriteBits(kReservedBitValue0, 2));
+    for (uint8_t r = 0; r < vp9.gof.num_ref_pics[i]; ++r) {
+      RETURN_FALSE_ON_ERROR(writer->WriteUInt8(vp9.gof.pid_diff[i][r]));
+    }
+  }
+  return true;
+}
+
+// TODO(https://bugs.webrtc.org/11319):
+// Workaround for switching off spatial layers on the fly.
+// Sent layers must start from SL0 on RTP layer, but can start from any
+// spatial layer because WebRTC-SVC api isn't implemented yet and
+// current API to invoke SVC is not flexible enough.
+RTPVideoHeaderVP9 RemoveInactiveSpatialLayers(
+    const RTPVideoHeaderVP9& original_header) {
+  RTPVideoHeaderVP9 hdr(original_header);
+  if (original_header.first_active_layer == 0)
+    return hdr;
+  for (size_t i = hdr.first_active_layer; i < hdr.num_spatial_layers; ++i) {
+    hdr.width[i - hdr.first_active_layer] = hdr.width[i];
+    hdr.height[i - hdr.first_active_layer] = hdr.height[i];
+  }
+  for (size_t i = hdr.num_spatial_layers - hdr.first_active_layer;
+       i < hdr.num_spatial_layers; ++i) {
+    hdr.width[i] = 0;
+    hdr.height[i] = 0;
+  }
+  hdr.num_spatial_layers -= hdr.first_active_layer;
+  hdr.spatial_idx -= hdr.first_active_layer;
+  hdr.first_active_layer = 0;
+  return hdr;
+}
+}  // namespace
+
+RtpPacketizerVp9::RtpPacketizerVp9(rtc::ArrayView<const uint8_t> payload,
+                                   PayloadSizeLimits limits,
+                                   const RTPVideoHeaderVP9& hdr)
+    : hdr_(RemoveInactiveSpatialLayers(hdr)),
+      header_size_(PayloadDescriptorLengthMinusSsData(hdr_)),
+      first_packet_extra_header_size_(SsDataLength(hdr_)),
+      remaining_payload_(payload) {
+  RTC_DCHECK_EQ(hdr_.first_active_layer, 0);
+
+  limits.max_payload_len -= header_size_;
+  limits.first_packet_reduction_len += first_packet_extra_header_size_;
+  limits.single_packet_reduction_len += first_packet_extra_header_size_;
+
+  payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  current_packet_ = payload_sizes_.begin();
+}
+
+RtpPacketizerVp9::~RtpPacketizerVp9() = default;
+
+size_t RtpPacketizerVp9::NumPackets() const {
+  return payload_sizes_.end() - current_packet_;
+}
+
+bool RtpPacketizerVp9::NextPacket(RtpPacketToSend* packet) {
+  RTC_DCHECK(packet);
+  if (current_packet_ == payload_sizes_.end()) {
+    return false;
+  }
+
+  bool layer_begin = current_packet_ == payload_sizes_.begin();
+  int packet_payload_len = *current_packet_;
+  ++current_packet_;
+  bool layer_end = current_packet_ == payload_sizes_.end();
+
+  int header_size = header_size_;
+  if (layer_begin)
+    header_size += first_packet_extra_header_size_;
+
+  uint8_t* buffer = packet->AllocatePayload(header_size + packet_payload_len);
+  RTC_CHECK(buffer);
+
+  if (!WriteHeader(layer_begin, layer_end,
+                   rtc::MakeArrayView(buffer, header_size)))
+    return false;
+
+  memcpy(buffer + header_size, remaining_payload_.data(), packet_payload_len);
+  remaining_payload_ = remaining_payload_.subview(packet_payload_len);
+
+  // Ensure end_of_picture is always set on top spatial layer when it is not
+  // dropped.
+  RTC_DCHECK(hdr_.spatial_idx < hdr_.num_spatial_layers - 1 ||
+             hdr_.end_of_picture);
+
+  packet->SetMarker(layer_end && hdr_.end_of_picture);
+  return true;
+}
+
+// VP9 format:
+//
+// Payload descriptor for F = 1 (flexible mode)
+//       0 1 2 3 4 5 6 7
+//      +-+-+-+-+-+-+-+-+
+//      |I|P|L|F|B|E|V|Z| (REQUIRED)
+//      +-+-+-+-+-+-+-+-+
+// I:   |M| PICTURE ID  | (RECOMMENDED)
+//      +-+-+-+-+-+-+-+-+
+// M:   | EXTENDED PID  | (RECOMMENDED)
+//      +-+-+-+-+-+-+-+-+
+// L:   |  T  |U|  S  |D| (CONDITIONALLY RECOMMENDED)
+//      +-+-+-+-+-+-+-+-+                             -|
+// P,F: | P_DIFF      |N| (CONDITIONALLY RECOMMENDED)  . up to 3 times
+//      +-+-+-+-+-+-+-+-+                             -|
+// V:   | SS            |
+//      | ..            |
+//      +-+-+-+-+-+-+-+-+
+//
+// Payload descriptor for F = 0 (non-flexible mode)
+//       0 1 2 3 4 5 6 7
+//      +-+-+-+-+-+-+-+-+
+//      |I|P|L|F|B|E|V|Z| (REQUIRED)
+//      +-+-+-+-+-+-+-+-+
+// I:   |M| PICTURE ID  | (RECOMMENDED)
+//      +-+-+-+-+-+-+-+-+
+// M:   | EXTENDED PID  | (RECOMMENDED)
+//      +-+-+-+-+-+-+-+-+
+// L:   |  T  |U|  S  |D| (CONDITIONALLY RECOMMENDED)
+//      +-+-+-+-+-+-+-+-+
+//      |   TL0PICIDX   | (CONDITIONALLY REQUIRED)
+//      +-+-+-+-+-+-+-+-+
+// V:   | SS            |
+//      | ..            |
+//      +-+-+-+-+-+-+-+-+
+bool RtpPacketizerVp9::WriteHeader(bool layer_begin,
+                                   bool layer_end,
+                                   rtc::ArrayView<uint8_t> buffer) const {
+  // Required payload descriptor byte.
+  bool i_bit = PictureIdPresent(hdr_);
+  bool p_bit = hdr_.inter_pic_predicted;
+  bool l_bit = LayerInfoPresent(hdr_);
+  bool f_bit = hdr_.flexible_mode;
+  bool b_bit = layer_begin;
+  bool e_bit = layer_end;
+  bool v_bit = hdr_.ss_data_available && b_bit;
+  bool z_bit = hdr_.non_ref_for_inter_layer_pred;
+
+  rtc::BitBufferWriter writer(buffer.data(), buffer.size());
+  RETURN_FALSE_ON_ERROR(writer.WriteBits(i_bit ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer.WriteBits(p_bit ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer.WriteBits(l_bit ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer.WriteBits(f_bit ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer.WriteBits(b_bit ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer.WriteBits(e_bit ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer.WriteBits(v_bit ? 1 : 0, 1));
+  RETURN_FALSE_ON_ERROR(writer.WriteBits(z_bit ? 1 : 0, 1));
+
+  // Add fields that are present.
+  if (i_bit && !WritePictureId(hdr_, &writer)) {
+    RTC_LOG(LS_ERROR) << "Failed writing VP9 picture id.";
+    return false;
+  }
+  if (l_bit && !WriteLayerInfo(hdr_, &writer)) {
+    RTC_LOG(LS_ERROR) << "Failed writing VP9 layer info.";
+    return false;
+  }
+  if (p_bit && f_bit && !WriteRefIndices(hdr_, &writer)) {
+    RTC_LOG(LS_ERROR) << "Failed writing VP9 ref indices.";
+    return false;
+  }
+  if (v_bit && !WriteSsData(hdr_, &writer)) {
+    RTC_LOG(LS_ERROR) << "Failed writing VP9 SS data.";
+    return false;
+  }
+
+  size_t offset_bytes = 0;
+  size_t offset_bits = 0;
+  writer.GetCurrentOffset(&offset_bytes, &offset_bits);
+  RTC_DCHECK_EQ(offset_bits, 0);
+  RTC_DCHECK_EQ(offset_bytes, buffer.size());
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
new file mode 100644
index 0000000000..3cf4dd56e5
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+//
+// This file contains the declaration of the VP9 packetizer class.
+// A packetizer object is created for each encoded video frame. The
+// constructor is called with the payload data and size.
+//
+// After creating the packetizer, the method NextPacket is called
+// repeatedly to get all packets for the frame. The method returns
+// false as long as there are more packets left to fetch.
+//
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+
+namespace webrtc {
+
+class RtpPacketizerVp9 : public RtpPacketizer {
+ public:
+  // The `payload` must be one encoded VP9 layer frame.
+  RtpPacketizerVp9(rtc::ArrayView<const uint8_t> payload,
+                   PayloadSizeLimits limits,
+                   const RTPVideoHeaderVP9& hdr);
+
+  ~RtpPacketizerVp9() override;
+
+  RtpPacketizerVp9(const RtpPacketizerVp9&) = delete;
+  RtpPacketizerVp9& operator=(const RtpPacketizerVp9&) = delete;
+
+  size_t NumPackets() const override;
+
+  // Gets the next payload with VP9 payload header.
+  // Write payload and set marker bit of the `packet`.
+  // Returns true on success, false otherwise.
+  bool NextPacket(RtpPacketToSend* packet) override;
+
+ private:
+  // Writes the payload descriptor header.
+  // `layer_begin` and `layer_end` indicates the postision of the packet in
+  // the layer frame. Returns false on failure.
+  bool WriteHeader(bool layer_begin,
+                   bool layer_end,
+                   rtc::ArrayView<uint8_t> rtp_payload) const;
+
+  const RTPVideoHeaderVP9 hdr_;
+  const int header_size_;
+  const int first_packet_extra_header_size_;
+  rtc::ArrayView<const uint8_t> remaining_payload_;
+  std::vector<int> payload_sizes_;
+  std::vector<int>::const_iterator current_packet_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
new file mode 100644
index 0000000000..e18b8a803f
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
@@ -0,0 +1,608 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+void VerifyHeader(const RTPVideoHeaderVP9& expected,
+                  const RTPVideoHeaderVP9& actual) {
+  EXPECT_EQ(expected.inter_layer_predicted, actual.inter_layer_predicted);
+  EXPECT_EQ(expected.inter_pic_predicted, actual.inter_pic_predicted);
+  EXPECT_EQ(expected.flexible_mode, actual.flexible_mode);
+  EXPECT_EQ(expected.beginning_of_frame, actual.beginning_of_frame);
+  EXPECT_EQ(expected.end_of_frame, actual.end_of_frame);
+  EXPECT_EQ(expected.ss_data_available, actual.ss_data_available);
+  EXPECT_EQ(expected.non_ref_for_inter_layer_pred,
+            actual.non_ref_for_inter_layer_pred);
+  EXPECT_EQ(expected.picture_id, actual.picture_id);
+  EXPECT_EQ(expected.max_picture_id, actual.max_picture_id);
+  EXPECT_EQ(expected.temporal_idx, actual.temporal_idx);
+  EXPECT_EQ(expected.spatial_idx, actual.spatial_idx);
+  EXPECT_EQ(expected.gof_idx, actual.gof_idx);
+  EXPECT_EQ(expected.tl0_pic_idx, actual.tl0_pic_idx);
+  EXPECT_EQ(expected.temporal_up_switch, actual.temporal_up_switch);
+
+  EXPECT_EQ(expected.num_ref_pics, actual.num_ref_pics);
+  for (uint8_t i = 0; i < expected.num_ref_pics; ++i) {
+    EXPECT_EQ(expected.pid_diff[i], actual.pid_diff[i]);
+    EXPECT_EQ(expected.ref_picture_id[i], actual.ref_picture_id[i]);
+  }
+  if (expected.ss_data_available) {
+    EXPECT_EQ(expected.spatial_layer_resolution_present,
+              actual.spatial_layer_resolution_present);
+    EXPECT_EQ(expected.num_spatial_layers, actual.num_spatial_layers);
+    if (expected.spatial_layer_resolution_present) {
+      for (size_t i = 0; i < expected.num_spatial_layers; i++) {
+        EXPECT_EQ(expected.width[i], actual.width[i]);
+        EXPECT_EQ(expected.height[i], actual.height[i]);
+      }
+    }
+    EXPECT_EQ(expected.gof.num_frames_in_gof, actual.gof.num_frames_in_gof);
+    for (size_t i = 0; i < expected.gof.num_frames_in_gof; i++) {
+      EXPECT_EQ(expected.gof.temporal_up_switch[i],
+                actual.gof.temporal_up_switch[i]);
+      EXPECT_EQ(expected.gof.temporal_idx[i], actual.gof.temporal_idx[i]);
+      EXPECT_EQ(expected.gof.num_ref_pics[i], actual.gof.num_ref_pics[i]);
+      for (uint8_t j = 0; j < expected.gof.num_ref_pics[i]; j++) {
+        EXPECT_EQ(expected.gof.pid_diff[i][j], actual.gof.pid_diff[i][j]);
+      }
+    }
+  }
+}
+
+void ParseAndCheckPacket(const uint8_t* packet,
+                         const RTPVideoHeaderVP9& expected,
+                         int expected_hdr_length,
+                         size_t expected_length) {
+  RTPVideoHeader video_header;
+  EXPECT_EQ(VideoRtpDepacketizerVp9::ParseRtpPayload(
+                rtc::MakeArrayView(packet, expected_length), &video_header),
+            expected_hdr_length);
+  EXPECT_EQ(kVideoCodecVP9, video_header.codec);
+  auto& vp9_header =
+      absl::get<RTPVideoHeaderVP9>(video_header.video_type_header);
+  VerifyHeader(expected, vp9_header);
+}
+
+// Payload descriptor for flexible mode
+//        0 1 2 3 4 5 6 7
+//        +-+-+-+-+-+-+-+-+
+//        |I|P|L|F|B|E|V|Z| (REQUIRED)
+//        +-+-+-+-+-+-+-+-+
+//   I:   |M| PICTURE ID  | (RECOMMENDED)
+//        +-+-+-+-+-+-+-+-+
+//   M:   | EXTENDED PID  | (RECOMMENDED)
+//        +-+-+-+-+-+-+-+-+
+//   L:   |  T  |U|  S  |D| (CONDITIONALLY RECOMMENDED)
+//        +-+-+-+-+-+-+-+-+                             -|
+//   P,F: | P_DIFF      |N| (CONDITIONALLY RECOMMENDED)  . up to 3 times
+//        +-+-+-+-+-+-+-+-+                             -|
+//   V:   | SS            |
+//        | ..            |
+//        +-+-+-+-+-+-+-+-+
+//
+// Payload descriptor for non-flexible mode
+//        0 1 2 3 4 5 6 7
+//        +-+-+-+-+-+-+-+-+
+//        |I|P|L|F|B|E|V|Z| (REQUIRED)
+//        +-+-+-+-+-+-+-+-+
+//   I:   |M| PICTURE ID  | (RECOMMENDED)
+//        +-+-+-+-+-+-+-+-+
+//   M:   | EXTENDED PID  | (RECOMMENDED)
+//        +-+-+-+-+-+-+-+-+
+//   L:   |  T  |U|  S  |D| (CONDITIONALLY RECOMMENDED)
+//        +-+-+-+-+-+-+-+-+
+//        |   TL0PICIDX   | (CONDITIONALLY REQUIRED)
+//        +-+-+-+-+-+-+-+-+
+//   V:   | SS            |
+//        | ..            |
+//        +-+-+-+-+-+-+-+-+
+
+class RtpPacketizerVp9Test : public ::testing::Test {
+ protected:
+  static constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr;
+  static constexpr size_t kMaxPacketSize = 1200;
+
+  RtpPacketizerVp9Test() : packet_(kNoExtensions, kMaxPacketSize) {}
+  void SetUp() override { expected_.InitRTPVideoHeaderVP9(); }
+
+  RtpPacketToSend packet_;
+  std::vector<uint8_t> payload_;
+  size_t payload_pos_;
+  RTPVideoHeaderVP9 expected_;
+  std::unique_ptr<RtpPacketizerVp9> packetizer_;
+  size_t num_packets_;
+
+  void Init(size_t payload_size, size_t packet_size) {
+    payload_.assign(payload_size, 7);
+    payload_pos_ = 0;
+    RtpPacketizer::PayloadSizeLimits limits;
+    limits.max_payload_len = packet_size;
+    packetizer_.reset(new RtpPacketizerVp9(payload_, limits, expected_));
+    num_packets_ = packetizer_->NumPackets();
+  }
+
+  void CheckPayload(const uint8_t* packet,
+                    size_t start_pos,
+                    size_t end_pos,
+                    bool last) {
+    for (size_t i = start_pos; i < end_pos; ++i) {
+      EXPECT_EQ(packet[i], payload_[payload_pos_++]);
+    }
+    EXPECT_EQ(last, payload_pos_ == payload_.size());
+  }
+
+  void CreateParseAndCheckPackets(
+      rtc::ArrayView<const size_t> expected_hdr_sizes,
+      rtc::ArrayView<const size_t> expected_sizes) {
+    ASSERT_EQ(expected_hdr_sizes.size(), expected_sizes.size());
+    ASSERT_TRUE(packetizer_ != nullptr);
+    EXPECT_EQ(expected_sizes.size(), num_packets_);
+    for (size_t i = 0; i < expected_sizes.size(); ++i) {
+      EXPECT_TRUE(packetizer_->NextPacket(&packet_));
+      auto rtp_payload = packet_.payload();
+      EXPECT_EQ(expected_sizes[i], rtp_payload.size());
+      RTPVideoHeaderVP9 hdr = expected_;
+      hdr.beginning_of_frame = (i == 0);
+      hdr.end_of_frame = (i + 1) == expected_sizes.size();
+      ParseAndCheckPacket(rtp_payload.data(), hdr, expected_hdr_sizes[i],
+                          rtp_payload.size());
+      CheckPayload(rtp_payload.data(), expected_hdr_sizes[i],
+                   rtp_payload.size(), (i + 1) == expected_sizes.size());
+      expected_.ss_data_available = false;
+    }
+  }
+
+  void CreateParseAndCheckPacketsLayers(size_t num_spatial_layers,
+                                        size_t expected_layer) {
+    ASSERT_TRUE(packetizer_ != nullptr);
+    for (size_t i = 0; i < num_packets_; ++i) {
+      EXPECT_TRUE(packetizer_->NextPacket(&packet_));
+      RTPVideoHeader video_header;
+      VideoRtpDepacketizerVp9::ParseRtpPayload(packet_.payload(),
+                                               &video_header);
+      const auto& vp9_header =
+          absl::get<RTPVideoHeaderVP9>(video_header.video_type_header);
+      EXPECT_EQ(vp9_header.spatial_idx, expected_layer);
+      EXPECT_EQ(vp9_header.num_spatial_layers, num_spatial_layers);
+    }
+  }
+};
+
+TEST_F(RtpPacketizerVp9Test, TestEqualSizedMode_OnePacket) {
+  const size_t kFrameSize = 25;
+  const size_t kPacketSize = 26;
+  Init(kFrameSize, kPacketSize);
+
+  // One packet:
+  // I:0, P:0, L:0, F:0, B:1, E:1, V:0, Z:0  (1hdr + 25 payload)
+  const size_t kExpectedHdrSizes[] = {1};
+  const size_t kExpectedSizes[] = {26};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestEqualSizedMode_TwoPackets) {
+  const size_t kFrameSize = 27;
+  const size_t kPacketSize = 27;
+  Init(kFrameSize, kPacketSize);
+
+  // Two packets:
+  // I:0, P:0, L:0, F:0, B:1, E:0, V:0, Z:0  (1hdr + 14 payload)
+  // I:0, P:0, L:0, F:0, B:0, E:1, V:0, Z:0  (1hdr + 13 payload)
+  const size_t kExpectedHdrSizes[] = {1, 1};
+  const size_t kExpectedSizes[] = {14, 15};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestTooShortBufferToFitPayload) {
+  const size_t kFrameSize = 1;
+  const size_t kPacketSize = 1;
+  Init(kFrameSize, kPacketSize);  // 1hdr + 1 payload
+
+  EXPECT_FALSE(packetizer_->NextPacket(&packet_));
+}
+
+TEST_F(RtpPacketizerVp9Test, TestOneBytePictureId) {
+  const size_t kFrameSize = 30;
+  const size_t kPacketSize = 12;
+
+  expected_.picture_id = kMaxOneBytePictureId;  // 2 byte payload descriptor
+  expected_.max_picture_id = kMaxOneBytePictureId;
+  Init(kFrameSize, kPacketSize);
+
+  // Three packets:
+  // I:1, P:0, L:0, F:0, B:1, E:0, V:0, Z:0 (2hdr + 10 payload)
+  // I:1, P:0, L:0, F:0, B:0, E:0, V:0, Z:0 (2hdr + 10 payload)
+  // I:1, P:0, L:0, F:0, B:0, E:1, V:0, Z:0 (2hdr + 10 payload)
+  const size_t kExpectedHdrSizes[] = {2, 2, 2};
+  const size_t kExpectedSizes[] = {12, 12, 12};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestTwoBytePictureId) {
+  const size_t kFrameSize = 31;
+  const size_t kPacketSize = 13;
+
+  expected_.picture_id = kMaxTwoBytePictureId;  // 3 byte payload descriptor
+  Init(kFrameSize, kPacketSize);
+
+  // Four packets:
+  // I:1, P:0, L:0, F:0, B:1, E:0, V:0, Z:0 (3hdr + 8 payload)
+  // I:1, P:0, L:0, F:0, B:0, E:0, V:0, Z:0 (3hdr + 8 payload)
+  // I:1, P:0, L:0, F:0, B:0, E:0, V:0, Z:0 (3hdr + 8 payload)
+  // I:1, P:0, L:0, F:0, B:0, E:1, V:0, Z:0 (3hdr + 7 payload)
+  const size_t kExpectedHdrSizes[] = {3, 3, 3, 3};
+  const size_t kExpectedSizes[] = {10, 11, 11, 11};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestLayerInfoWithNonFlexibleMode) {
+  const size_t kFrameSize = 30;
+  const size_t kPacketSize = 25;
+
+  expected_.temporal_idx = 3;
+  expected_.temporal_up_switch = true;  // U
+  expected_.num_spatial_layers = 3;
+  expected_.spatial_idx = 2;
+  expected_.inter_layer_predicted = true;  // D
+  expected_.tl0_pic_idx = 117;
+  Init(kFrameSize, kPacketSize);
+
+  // Two packets:
+  //    | I:0, P:0, L:1, F:0, B:1, E:0, V:0 Z:0 | (3hdr + 15 payload)
+  // L: | T:3, U:1, S:2, D:1 | TL0PICIDX:117 |
+  //    | I:0, P:0, L:1, F:0, B:0, E:1, V:0 Z:0 | (3hdr + 15 payload)
+  // L: | T:3, U:1, S:2, D:1 | TL0PICIDX:117 |
+  const size_t kExpectedHdrSizes[] = {3, 3};
+  const size_t kExpectedSizes[] = {18, 18};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestLayerInfoWithFlexibleMode) {
+  const size_t kFrameSize = 21;
+  const size_t kPacketSize = 23;
+
+  expected_.flexible_mode = true;
+  expected_.temporal_idx = 3;
+  expected_.temporal_up_switch = true;  // U
+  expected_.num_spatial_layers = 3;
+  expected_.spatial_idx = 2;
+  expected_.inter_layer_predicted = false;  // D
+  Init(kFrameSize, kPacketSize);
+
+  // One packet:
+  // I:0, P:0, L:1, F:1, B:1, E:1, V:0 (2hdr + 21 payload)
+  // L:   T:3, U:1, S:2, D:0
+  const size_t kExpectedHdrSizes[] = {2};
+  const size_t kExpectedSizes[] = {23};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestRefIdx) {
+  const size_t kFrameSize = 16;
+  const size_t kPacketSize = 21;
+
+  expected_.inter_pic_predicted = true;  // P
+  expected_.flexible_mode = true;        // F
+  expected_.picture_id = 2;
+  expected_.max_picture_id = kMaxOneBytePictureId;
+
+  expected_.num_ref_pics = 3;
+  expected_.pid_diff[0] = 1;
+  expected_.pid_diff[1] = 3;
+  expected_.pid_diff[2] = 127;
+  expected_.ref_picture_id[0] = 1;    // 2 - 1 = 1
+  expected_.ref_picture_id[1] = 127;  // (kMaxPictureId + 1) + 2 - 3 = 127
+  expected_.ref_picture_id[2] = 3;    // (kMaxPictureId + 1) + 2 - 127 = 3
+  Init(kFrameSize, kPacketSize);
+
+  // Two packets:
+  // I:1, P:1, L:0, F:1, B:1, E:1, V:0, Z:0 (5hdr + 16 payload)
+  // I:   2
+  // P,F: P_DIFF:1,   N:1
+  //      P_DIFF:3,   N:1
+  //      P_DIFF:127, N:0
+  const size_t kExpectedHdrSizes[] = {5};
+  const size_t kExpectedSizes[] = {21};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestRefIdxFailsWithoutPictureId) {
+  const size_t kFrameSize = 16;
+  const size_t kPacketSize = 21;
+
+  expected_.inter_pic_predicted = true;
+  expected_.flexible_mode = true;
+  expected_.num_ref_pics = 1;
+  expected_.pid_diff[0] = 3;
+  Init(kFrameSize, kPacketSize);
+
+  EXPECT_FALSE(packetizer_->NextPacket(&packet_));
+}
+
+TEST_F(RtpPacketizerVp9Test, TestSsDataWithoutSpatialResolutionPresent) {
+  const size_t kFrameSize = 21;
+  const size_t kPacketSize = 26;
+
+  expected_.ss_data_available = true;
+  expected_.num_spatial_layers = 1;
+  expected_.spatial_layer_resolution_present = false;
+  expected_.gof.num_frames_in_gof = 1;
+  expected_.gof.temporal_idx[0] = 0;
+  expected_.gof.temporal_up_switch[0] = true;
+  expected_.gof.num_ref_pics[0] = 1;
+  expected_.gof.pid_diff[0][0] = 4;
+  Init(kFrameSize, kPacketSize);
+
+  // One packet:
+  // I:0, P:0, L:0, F:0, B:1, E:1, V:1, Z:0 (5hdr + 21 payload)
+  // N_S:0, Y:0, G:1
+  // N_G:1
+  // T:0, U:1, R:1 | P_DIFF[0][0]:4
+  const size_t kExpectedHdrSizes[] = {5};
+  const size_t kExpectedSizes[] = {26};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestSsDataWithoutGbitPresent) {
+  const size_t kFrameSize = 21;
+  const size_t kPacketSize = 23;
+
+  expected_.ss_data_available = true;
+  expected_.num_spatial_layers = 1;
+  expected_.spatial_layer_resolution_present = false;
+  expected_.gof.num_frames_in_gof = 0;
+  Init(kFrameSize, kPacketSize);
+
+  // One packet:
+  // I:0, P:0, L:0, F:0, B:1, E:1, V:1, Z:0 (2hdr + 21 payload)
+  // N_S:0, Y:0, G:0
+  const size_t kExpectedHdrSizes[] = {2};
+  const size_t kExpectedSizes[] = {23};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestSsData) {
+  const size_t kFrameSize = 21;
+  const size_t kPacketSize = 40;
+
+  expected_.ss_data_available = true;
+  expected_.num_spatial_layers = 2;
+  expected_.spatial_layer_resolution_present = true;
+  expected_.width[0] = 640;
+  expected_.width[1] = 1280;
+  expected_.height[0] = 360;
+  expected_.height[1] = 720;
+  expected_.gof.num_frames_in_gof = 3;
+  expected_.gof.temporal_idx[0] = 0;
+  expected_.gof.temporal_idx[1] = 1;
+  expected_.gof.temporal_idx[2] = 2;
+  expected_.gof.temporal_up_switch[0] = true;
+  expected_.gof.temporal_up_switch[1] = true;
+  expected_.gof.temporal_up_switch[2] = false;
+  expected_.gof.num_ref_pics[0] = 0;
+  expected_.gof.num_ref_pics[1] = 3;
+  expected_.gof.num_ref_pics[2] = 2;
+  expected_.gof.pid_diff[1][0] = 5;
+  expected_.gof.pid_diff[1][1] = 6;
+  expected_.gof.pid_diff[1][2] = 7;
+  expected_.gof.pid_diff[2][0] = 8;
+  expected_.gof.pid_diff[2][1] = 9;
+  Init(kFrameSize, kPacketSize);
+
+  // One packet:
+  // I:0, P:0, L:0, F:0, B:1, E:1, V:1, Z:0 (19hdr + 21 payload)
+  // N_S:1, Y:1, G:1
+  // WIDTH:640   // 2 bytes
+  // HEIGHT:360  // 2 bytes
+  // WIDTH:1280  // 2 bytes
+  // HEIGHT:720  // 2 bytes
+  // N_G:3
+  // T:0, U:1, R:0
+  // T:1, U:1, R:3 | P_DIFF[1][0]:5 | P_DIFF[1][1]:6 | P_DIFF[1][2]:7
+  // T:2, U:0, R:2 | P_DIFF[2][0]:8 | P_DIFF[2][0]:9
+  const size_t kExpectedHdrSizes[] = {19};
+  const size_t kExpectedSizes[] = {40};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, TestSsDataDoesNotFitInAveragePacket) {
+  const size_t kFrameSize = 24;
+  const size_t kPacketSize = 20;
+
+  expected_.ss_data_available = true;
+  expected_.num_spatial_layers = 2;
+  expected_.spatial_layer_resolution_present = true;
+  expected_.width[0] = 640;
+  expected_.width[1] = 1280;
+  expected_.height[0] = 360;
+  expected_.height[1] = 720;
+  expected_.gof.num_frames_in_gof = 3;
+  expected_.gof.temporal_idx[0] = 0;
+  expected_.gof.temporal_idx[1] = 1;
+  expected_.gof.temporal_idx[2] = 2;
+  expected_.gof.temporal_up_switch[0] = true;
+  expected_.gof.temporal_up_switch[1] = true;
+  expected_.gof.temporal_up_switch[2] = false;
+  expected_.gof.num_ref_pics[0] = 0;
+  expected_.gof.num_ref_pics[1] = 3;
+  expected_.gof.num_ref_pics[2] = 2;
+  expected_.gof.pid_diff[1][0] = 5;
+  expected_.gof.pid_diff[1][1] = 6;
+  expected_.gof.pid_diff[1][2] = 7;
+  expected_.gof.pid_diff[2][0] = 8;
+  expected_.gof.pid_diff[2][1] = 9;
+  Init(kFrameSize, kPacketSize);
+
+  // Three packets:
+  // I:0, P:0, L:0, F:0, B:1, E:1, V:1, Z:0 (19hdr + 1 payload)
+  // N_S:1, Y:1, G:1
+  // WIDTH:640   // 2 bytes
+  // HEIGHT:360  // 2 bytes
+  // WIDTH:1280  // 2 bytes
+  // HEIGHT:720  // 2 bytes
+  // N_G:3
+  // T:0, U:1, R:0
+  // T:1, U:1, R:3 | P_DIFF[1][0]:5 | P_DIFF[1][1]:6 | P_DIFF[1][2]:7
+  // T:2, U:0, R:2 | P_DIFF[2][0]:8 | P_DIFF[2][0]:9
+  // Last two packets 1 bytes vp9 hdrs and the rest of payload 14 and 9 bytes.
+  const size_t kExpectedHdrSizes[] = {19, 1, 1};
+  const size_t kExpectedSizes[] = {20, 15, 10};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test, EndOfPictureSetsSetMarker) {
+  const size_t kFrameSize = 10;
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 8;
+  const uint8_t kFrame[kFrameSize] = {7};
+
+  RTPVideoHeaderVP9 vp9_header;
+  vp9_header.InitRTPVideoHeaderVP9();
+  vp9_header.flexible_mode = true;
+  vp9_header.num_spatial_layers = 3;
+
+  RtpPacketToSend packet(kNoExtensions);
+
+  // Drop top layer and ensure that marker bit is set on last encoded layer.
+  for (size_t spatial_idx = 0; spatial_idx < vp9_header.num_spatial_layers - 1;
+       ++spatial_idx) {
+    const bool end_of_picture =
+        spatial_idx + 1 == vp9_header.num_spatial_layers - 1;
+    vp9_header.spatial_idx = spatial_idx;
+    vp9_header.end_of_picture = end_of_picture;
+    RtpPacketizerVp9 packetizer(kFrame, limits, vp9_header);
+    ASSERT_TRUE(packetizer.NextPacket(&packet));
+    EXPECT_FALSE(packet.Marker());
+    ASSERT_TRUE(packetizer.NextPacket(&packet));
+    EXPECT_EQ(packet.Marker(), end_of_picture);
+  }
+}
+
+TEST_F(RtpPacketizerVp9Test, TestGeneratesMinimumNumberOfPackets) {
+  const size_t kFrameSize = 10;
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 8;
+  // Calculated by hand. One packet can contain
+  // `kPacketSize` - `kVp9MinDiscriptorSize` = 6 bytes of the frame payload,
+  // thus to fit 10 bytes two packets are required.
+  const size_t kMinNumberOfPackets = 2;
+  const uint8_t kFrame[kFrameSize] = {7};
+
+  RTPVideoHeaderVP9 vp9_header;
+  vp9_header.InitRTPVideoHeaderVP9();
+
+  RtpPacketToSend packet(kNoExtensions);
+
+  RtpPacketizerVp9 packetizer(kFrame, limits, vp9_header);
+  EXPECT_EQ(packetizer.NumPackets(), kMinNumberOfPackets);
+  ASSERT_TRUE(packetizer.NextPacket(&packet));
+  EXPECT_FALSE(packet.Marker());
+  ASSERT_TRUE(packetizer.NextPacket(&packet));
+  EXPECT_TRUE(packet.Marker());
+}
+
+TEST_F(RtpPacketizerVp9Test, TestRespectsLastPacketReductionLen) {
+  const size_t kFrameSize = 10;
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 8;
+  limits.last_packet_reduction_len = 5;
+  // Calculated by hand. VP9 payload descriptor is 2 bytes. Like in the test
+  // above, 1 packet is not enough. 2 packets can contain
+  // 2*(`kPacketSize` - `kVp9MinDiscriptorSize`) - `kLastPacketReductionLen` = 7
+  // But three packets are enough, since they have capacity of 3*(8-2)-5=13
+  // bytes.
+  const size_t kMinNumberOfPackets = 3;
+  const uint8_t kFrame[kFrameSize] = {7};
+
+  RTPVideoHeaderVP9 vp9_header;
+  vp9_header.InitRTPVideoHeaderVP9();
+  vp9_header.flexible_mode = true;
+
+  RtpPacketToSend packet(kNoExtensions);
+
+  RtpPacketizerVp9 packetizer0(kFrame, limits, vp9_header);
+  EXPECT_EQ(packetizer0.NumPackets(), kMinNumberOfPackets);
+  ASSERT_TRUE(packetizer0.NextPacket(&packet));
+  EXPECT_FALSE(packet.Marker());
+  ASSERT_TRUE(packetizer0.NextPacket(&packet));
+  EXPECT_FALSE(packet.Marker());
+  ASSERT_TRUE(packetizer0.NextPacket(&packet));
+  EXPECT_TRUE(packet.Marker());
+}
+
+TEST_F(RtpPacketizerVp9Test, TestNonRefForInterLayerPred) {
+  const size_t kFrameSize = 25;
+  const size_t kPacketSize = 26;
+
+  expected_.non_ref_for_inter_layer_pred = true;
+  Init(kFrameSize, kPacketSize);
+
+  // I:0, P:0, L:0, F:0, B:1, E:1, V:0, Z:1  (1hdr + 25 payload)
+  const size_t kExpectedHdrSizes[] = {1};
+  const size_t kExpectedSizes[] = {26};
+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes);
+}
+
+TEST_F(RtpPacketizerVp9Test,
+       ShiftsSpatialLayersTowardZeroWhenFirstLayersAreDisabled) {
+  const size_t kFrameSize = 25;
+  const size_t kPacketSize = 1024;
+
+  expected_.width[0] = 0;
+  expected_.height[0] = 0;
+  expected_.width[1] = 640;
+  expected_.height[1] = 360;
+  expected_.width[2] = 1280;
+  expected_.height[2] = 720;
+  expected_.num_spatial_layers = 3;
+  expected_.first_active_layer = 1;
+  expected_.ss_data_available = true;
+  expected_.spatial_layer_resolution_present = true;
+  expected_.gof.num_frames_in_gof = 3;
+  expected_.gof.temporal_idx[0] = 0;
+  expected_.gof.temporal_idx[1] = 1;
+  expected_.gof.temporal_idx[2] = 2;
+  expected_.gof.temporal_up_switch[0] = true;
+  expected_.gof.temporal_up_switch[1] = true;
+  expected_.gof.temporal_up_switch[2] = false;
+  expected_.gof.num_ref_pics[0] = 0;
+  expected_.gof.num_ref_pics[1] = 3;
+  expected_.gof.num_ref_pics[2] = 2;
+  expected_.gof.pid_diff[1][0] = 5;
+  expected_.gof.pid_diff[1][1] = 6;
+  expected_.gof.pid_diff[1][2] = 7;
+  expected_.gof.pid_diff[2][0] = 8;
+  expected_.gof.pid_diff[2][1] = 9;
+
+  expected_.spatial_idx = 1;
+  Init(kFrameSize, kPacketSize);
+  CreateParseAndCheckPacketsLayers(/*num_spatial_layers=*/2,
+                                   /*expected_layer=*/0);
+
+  // Now check for SL 2;
+  expected_.spatial_idx = 2;
+  Init(kFrameSize, kPacketSize);
+  CreateParseAndCheckPacketsLayers(/*num_spatial_layers=*/2,
+                                   /*expected_layer=*/1);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
new file mode 100644
index 0000000000..465308ec45
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
+
+#include <cstdint>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+constexpr int RtpGenericFrameDescriptor::kMaxNumFrameDependencies;
+constexpr int RtpGenericFrameDescriptor::kMaxTemporalLayers;
+constexpr int RtpGenericFrameDescriptor::kMaxSpatialLayers;
+
+RtpGenericFrameDescriptor::RtpGenericFrameDescriptor() = default;
+RtpGenericFrameDescriptor::RtpGenericFrameDescriptor(
+    const RtpGenericFrameDescriptor&) = default;
+RtpGenericFrameDescriptor::~RtpGenericFrameDescriptor() = default;
+
+int RtpGenericFrameDescriptor::TemporalLayer() const {
+  RTC_DCHECK(FirstPacketInSubFrame());
+  return temporal_layer_;
+}
+
+void RtpGenericFrameDescriptor::SetTemporalLayer(int temporal_layer) {
+  RTC_DCHECK_GE(temporal_layer, 0);
+  RTC_DCHECK_LT(temporal_layer, kMaxTemporalLayers);
+  temporal_layer_ = temporal_layer;
+}
+
+int RtpGenericFrameDescriptor::SpatialLayer() const {
+  RTC_DCHECK(FirstPacketInSubFrame());
+  int layer = 0;
+  uint8_t spatial_layers = spatial_layers_;
+  while (spatial_layers_ != 0 && !(spatial_layers & 1)) {
+    spatial_layers >>= 1;
+    layer++;
+  }
+  return layer;
+}
+
+uint8_t RtpGenericFrameDescriptor::SpatialLayersBitmask() const {
+  RTC_DCHECK(FirstPacketInSubFrame());
+  return spatial_layers_;
+}
+
+void RtpGenericFrameDescriptor::SetSpatialLayersBitmask(
+    uint8_t spatial_layers) {
+  RTC_DCHECK(FirstPacketInSubFrame());
+  spatial_layers_ = spatial_layers;
+}
+
+void RtpGenericFrameDescriptor::SetResolution(int width, int height) {
+  RTC_DCHECK(FirstPacketInSubFrame());
+  RTC_DCHECK_GE(width, 0);
+  RTC_DCHECK_LE(width, 0xFFFF);
+  RTC_DCHECK_GE(height, 0);
+  RTC_DCHECK_LE(height, 0xFFFF);
+  width_ = width;
+  height_ = height;
+}
+
+uint16_t RtpGenericFrameDescriptor::FrameId() const {
+  RTC_DCHECK(FirstPacketInSubFrame());
+  return frame_id_;
+}
+
+void RtpGenericFrameDescriptor::SetFrameId(uint16_t frame_id) {
+  RTC_DCHECK(FirstPacketInSubFrame());
+  frame_id_ = frame_id;
+}
+
+rtc::ArrayView<const uint16_t>
+RtpGenericFrameDescriptor::FrameDependenciesDiffs() const {
+  RTC_DCHECK(FirstPacketInSubFrame());
+  return rtc::MakeArrayView(frame_deps_id_diffs_, num_frame_deps_);
+}
+
+bool RtpGenericFrameDescriptor::AddFrameDependencyDiff(uint16_t fdiff) {
+  RTC_DCHECK(FirstPacketInSubFrame());
+  if (num_frame_deps_ == kMaxNumFrameDependencies)
+    return false;
+  if (fdiff == 0)
+    return false;
+  RTC_DCHECK_LT(fdiff, 1 << 14);
+  RTC_DCHECK_GT(fdiff, 0);
+  frame_deps_id_diffs_[num_frame_deps_] = fdiff;
+  num_frame_deps_++;
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
new file mode 100644
index 0000000000..8760acca2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+
+class RtpGenericFrameDescriptorExtension;
+
+// Data to put on the wire for FrameDescriptor rtp header extension.
+class RtpGenericFrameDescriptor {
+ public:
+  static constexpr int kMaxNumFrameDependencies = 8;
+  static constexpr int kMaxTemporalLayers = 8;
+  static constexpr int kMaxSpatialLayers = 8;
+
+  RtpGenericFrameDescriptor();
+  RtpGenericFrameDescriptor(const RtpGenericFrameDescriptor&);
+  ~RtpGenericFrameDescriptor();
+
+  bool FirstPacketInSubFrame() const { return beginning_of_subframe_; }
+  void SetFirstPacketInSubFrame(bool first) { beginning_of_subframe_ = first; }
+  bool LastPacketInSubFrame() const { return end_of_subframe_; }
+  void SetLastPacketInSubFrame(bool last) { end_of_subframe_ = last; }
+
+  // Properties below undefined if !FirstPacketInSubFrame()
+  // Valid range for temporal layer: [0, 7]
+  int TemporalLayer() const;
+  void SetTemporalLayer(int temporal_layer);
+
+  // Frame might by used, possible indirectly, for spatial layer sid iff
+  // (bitmask & (1 << sid)) != 0
+  int SpatialLayer() const;
+  uint8_t SpatialLayersBitmask() const;
+  void SetSpatialLayersBitmask(uint8_t spatial_layers);
+
+  int Width() const { return width_; }
+  int Height() const { return height_; }
+  void SetResolution(int width, int height);
+
+  uint16_t FrameId() const;
+  void SetFrameId(uint16_t frame_id);
+
+  rtc::ArrayView<const uint16_t> FrameDependenciesDiffs() const;
+  void ClearFrameDependencies() { num_frame_deps_ = 0; }
+  // Returns false on failure, i.e. number of dependencies is too large.
+  bool AddFrameDependencyDiff(uint16_t fdiff);
+
+ private:
+  bool beginning_of_subframe_ = false;
+  bool end_of_subframe_ = false;
+
+  uint16_t frame_id_ = 0;
+  uint8_t spatial_layers_ = 1;
+  uint8_t temporal_layer_ = 0;
+  size_t num_frame_deps_ = 0;
+  uint16_t frame_deps_id_diffs_[kMaxNumFrameDependencies];
+  int width_ = 0;
+  int height_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
new file mode 100644
index 0000000000..8a0810f445
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr uint8_t kFlagBeginOfSubframe = 0x80;
+constexpr uint8_t kFlagEndOfSubframe = 0x40;
+
+// In version 00, the flags F and L in the first byte correspond to
+// kFlagFirstSubframeV00 and kFlagLastSubframeV00. In practice, they were
+// always set to `true`.
+constexpr uint8_t kFlagFirstSubframeV00 = 0x20;
+constexpr uint8_t kFlagLastSubframeV00 = 0x10;
+
+constexpr uint8_t kFlagDependencies = 0x08;
+constexpr uint8_t kMaskTemporalLayer = 0x07;
+
+constexpr uint8_t kFlagMoreDependencies = 0x01;
+constexpr uint8_t kFlageXtendedOffset = 0x02;
+}  // namespace
+//       0 1 2 3 4 5 6 7
+//      +-+-+-+-+-+-+-+-+
+//      |B|E|F|L|D|  T  |
+//      +-+-+-+-+-+-+-+-+
+// B:   |       S       |
+//      +-+-+-+-+-+-+-+-+
+//      |               |
+// B:   +      FID      +
+//      |               |
+//      +-+-+-+-+-+-+-+-+
+//      |               |
+//      +     Width     +
+// B=1  |               |
+// and  +-+-+-+-+-+-+-+-+
+// D=0  |               |
+//      +     Height    +
+//      |               |
+//      +-+-+-+-+-+-+-+-+
+// D:   |    FDIFF  |X|M|
+//      +---------------+
+// X:   |      ...      |
+//      +-+-+-+-+-+-+-+-+
+// M:   |    FDIFF  |X|M|
+//      +---------------+
+//      |      ...      |
+//      +-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType RtpGenericFrameDescriptorExtension00::kId;
+
+bool RtpGenericFrameDescriptorExtension00::Parse(
+    rtc::ArrayView<const uint8_t> data,
+    RtpGenericFrameDescriptor* descriptor) {
+  if (data.empty()) {
+    return false;
+  }
+
+  bool begins_subframe = (data[0] & kFlagBeginOfSubframe) != 0;
+  descriptor->SetFirstPacketInSubFrame(begins_subframe);
+  descriptor->SetLastPacketInSubFrame((data[0] & kFlagEndOfSubframe) != 0);
+
+  // Parse Subframe details provided in 1st packet of subframe.
+  if (!begins_subframe) {
+    return data.size() == 1;
+  }
+  if (data.size() < 4) {
+    return false;
+  }
+  descriptor->SetTemporalLayer(data[0] & kMaskTemporalLayer);
+  descriptor->SetSpatialLayersBitmask(data[1]);
+  descriptor->SetFrameId(data[2] | (data[3] << 8));
+
+  // Parse dependencies.
+  descriptor->ClearFrameDependencies();
+  size_t offset = 4;
+  bool has_more_dependencies = (data[0] & kFlagDependencies) != 0;
+  if (!has_more_dependencies && data.size() >= offset + 4) {
+    uint16_t width = (data[offset] << 8) | data[offset + 1];
+    uint16_t height = (data[offset + 2] << 8) | data[offset + 3];
+    descriptor->SetResolution(width, height);
+    offset += 4;
+  }
+  while (has_more_dependencies) {
+    if (data.size() == offset)
+      return false;
+    has_more_dependencies = (data[offset] & kFlagMoreDependencies) != 0;
+    bool extended = (data[offset] & kFlageXtendedOffset) != 0;
+    uint16_t fdiff = data[offset] >> 2;
+    offset++;
+    if (extended) {
+      if (data.size() == offset)
+        return false;
+      fdiff |= (data[offset] << 6);
+      offset++;
+    }
+    if (!descriptor->AddFrameDependencyDiff(fdiff))
+      return false;
+  }
+  return true;
+}
+
+size_t RtpGenericFrameDescriptorExtension00::ValueSize(
+    const RtpGenericFrameDescriptor& descriptor) {
+  if (!descriptor.FirstPacketInSubFrame())
+    return 1;
+
+  size_t size = 4;
+  for (uint16_t fdiff : descriptor.FrameDependenciesDiffs()) {
+    size += (fdiff >= (1 << 6)) ? 2 : 1;
+  }
+  if (descriptor.FirstPacketInSubFrame() &&
+      descriptor.FrameDependenciesDiffs().empty() && descriptor.Width() > 0 &&
+      descriptor.Height() > 0) {
+    size += 4;
+  }
+  return size;
+}
+
+bool RtpGenericFrameDescriptorExtension00::Write(
+    rtc::ArrayView<uint8_t> data,
+    const RtpGenericFrameDescriptor& descriptor) {
+  RTC_CHECK_EQ(data.size(), ValueSize(descriptor));
+  uint8_t base_header =
+      (descriptor.FirstPacketInSubFrame() ? kFlagBeginOfSubframe : 0) |
+      (descriptor.LastPacketInSubFrame() ? kFlagEndOfSubframe : 0);
+  base_header |= kFlagFirstSubframeV00;
+  base_header |= kFlagLastSubframeV00;
+
+  if (!descriptor.FirstPacketInSubFrame()) {
+    data[0] = base_header;
+    return true;
+  }
+  data[0] =
+      base_header |
+      (descriptor.FrameDependenciesDiffs().empty() ? 0 : kFlagDependencies) |
+      descriptor.TemporalLayer();
+  data[1] = descriptor.SpatialLayersBitmask();
+  uint16_t frame_id = descriptor.FrameId();
+  data[2] = frame_id & 0xff;
+  data[3] = frame_id >> 8;
+  rtc::ArrayView<const uint16_t> fdiffs = descriptor.FrameDependenciesDiffs();
+  size_t offset = 4;
+  if (descriptor.FirstPacketInSubFrame() && fdiffs.empty() &&
+      descriptor.Width() > 0 && descriptor.Height() > 0) {
+    data[offset++] = (descriptor.Width() >> 8);
+    data[offset++] = (descriptor.Width() & 0xFF);
+    data[offset++] = (descriptor.Height() >> 8);
+    data[offset++] = (descriptor.Height() & 0xFF);
+  }
+  for (size_t i = 0; i < fdiffs.size(); i++) {
+    bool extended = fdiffs[i] >= (1 << 6);
+    bool more = i < fdiffs.size() - 1;
+    data[offset++] = ((fdiffs[i] & 0x3f) << 2) |
+                     (extended ? kFlageXtendedOffset : 0) |
+                     (more ? kFlagMoreDependencies : 0);
+    if (extended) {
+      data[offset++] = fdiffs[i] >> 6;
+    }
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
new file mode 100644
index 0000000000..b4f686565f
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_EXTENSION_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_EXTENSION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/rtp_parameters.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
+
+namespace webrtc {
+
+// Trait to read/write the generic frame descriptor, the early version of the
+// dependency descriptor extension. Usage of this rtp header extension is
+// discouraged in favor of the dependency descriptor.
+class RtpGenericFrameDescriptorExtension00 {
+ public:
+  using value_type = RtpGenericFrameDescriptor;
+  static constexpr RTPExtensionType kId = kRtpExtensionGenericFrameDescriptor;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kGenericFrameDescriptorUri00;
+  }
+  static constexpr int kMaxSizeBytes = 16;
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    RtpGenericFrameDescriptor* descriptor);
+  static size_t ValueSize(const RtpGenericFrameDescriptor& descriptor);
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    const RtpGenericFrameDescriptor& descriptor);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_EXTENSION_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc
new file mode 100644
index 0000000000..7c27326f75
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc
@@ -0,0 +1,264 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+
+constexpr uint8_t kDeprecatedFlags = 0x30;
+
+TEST(RtpGenericFrameDescriptorExtensionTest,
+     ParseFirstPacketOfIndependenSubFrame) {
+  const int kTemporalLayer = 5;
+  constexpr uint8_t kRaw[] = {0x80 | kTemporalLayer, 0x49, 0x12, 0x34};
+  RtpGenericFrameDescriptor descriptor;
+
+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension00::Parse(kRaw, &descriptor));
+
+  EXPECT_TRUE(descriptor.FirstPacketInSubFrame());
+  EXPECT_FALSE(descriptor.LastPacketInSubFrame());
+
+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), IsEmpty());
+  EXPECT_EQ(descriptor.TemporalLayer(), kTemporalLayer);
+  EXPECT_EQ(descriptor.SpatialLayersBitmask(), 0x49);
+  EXPECT_EQ(descriptor.FrameId(), 0x3412);
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest,
+     WriteFirstPacketOfIndependenSubFrame) {
+  const int kTemporalLayer = 5;
+  uint8_t kRaw[] = {0x80 | kTemporalLayer | kDeprecatedFlags, 0x49, 0x12, 0x34};
+  RtpGenericFrameDescriptor descriptor;
+
+  descriptor.SetFirstPacketInSubFrame(true);
+  descriptor.SetTemporalLayer(kTemporalLayer);
+  descriptor.SetSpatialLayersBitmask(0x49);
+  descriptor.SetFrameId(0x3412);
+
+  ASSERT_EQ(RtpGenericFrameDescriptorExtension00::ValueSize(descriptor),
+            sizeof(kRaw));
+  uint8_t buffer[sizeof(kRaw)];
+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension00::Write(buffer, descriptor));
+
+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, ParseLastPacketOfSubFrame) {
+  constexpr uint8_t kRaw[] = {0x40};
+  RtpGenericFrameDescriptor descriptor;
+
+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension00::Parse(kRaw, &descriptor));
+
+  EXPECT_FALSE(descriptor.FirstPacketInSubFrame());
+  EXPECT_TRUE(descriptor.LastPacketInSubFrame());
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, WriteLastPacketOfSubFrame) {
+  uint8_t kRaw[] = {0x40 | kDeprecatedFlags};
+
+  RtpGenericFrameDescriptor descriptor;
+  descriptor.SetLastPacketInSubFrame(true);
+
+  ASSERT_EQ(RtpGenericFrameDescriptorExtension00::ValueSize(descriptor),
+            sizeof(kRaw));
+  uint8_t buffer[sizeof(kRaw)];
+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension00::Write(buffer, descriptor));
+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, ParseMinShortFrameDependencies) {
+  constexpr uint16_t kDiff = 1;
+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0x04};
+  RtpGenericFrameDescriptor descriptor;
+
+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension00::Parse(kRaw, &descriptor));
+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, WriteMinShortFrameDependencies) {
+  constexpr uint16_t kDiff = 1;
+  uint8_t kRaw[] = {0x88 | kDeprecatedFlags, 0x01, 0x00, 0x00, 0x04};
+  RtpGenericFrameDescriptor descriptor;
+  descriptor.SetFirstPacketInSubFrame(true);
+  descriptor.AddFrameDependencyDiff(kDiff);
+
+  ASSERT_EQ(RtpGenericFrameDescriptorExtension00::ValueSize(descriptor),
+            sizeof(kRaw));
+  uint8_t buffer[sizeof(kRaw)];
+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension00::Write(buffer, descriptor));
+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, ParseMaxShortFrameDependencies) {
+  constexpr uint16_t kDiff = 0x3f;
+  constexpr uint8_t kRaw[] = {0xb8, 0x01, 0x00, 0x00, 0xfc};
+  RtpGenericFrameDescriptor descriptor;
+
+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension00::Parse(kRaw, &descriptor));
+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, WriteMaxShortFrameDependencies) {
+  constexpr uint16_t kDiff = 0x3f;
+  uint8_t kRaw[] = {0x88 | kDeprecatedFlags, 0x01, 0x00, 0x00, 0xfc};
+  RtpGenericFrameDescriptor descriptor;
+  descriptor.SetFirstPacketInSubFrame(true);
+  descriptor.AddFrameDependencyDiff(kDiff);
+
+  ASSERT_EQ(RtpGenericFrameDescriptorExtension00::ValueSize(descriptor),
+            sizeof(kRaw));
+  uint8_t buffer[sizeof(kRaw)];
+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension00::Write(buffer, descriptor));
+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, ParseMinLongFrameDependencies) {
+  constexpr uint16_t kDiff = 0x40;
+  constexpr uint8_t kRaw[] = {0xb8, 0x01, 0x00, 0x00, 0x02, 0x01};
+  RtpGenericFrameDescriptor descriptor;
+
+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension00::Parse(kRaw, &descriptor));
+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, WriteMinLongFrameDependencies) {
+  constexpr uint16_t kDiff = 0x40;
+  uint8_t kRaw[] = {0x88 | kDeprecatedFlags, 0x01, 0x00, 0x00, 0x02, 0x01};
+  RtpGenericFrameDescriptor descriptor;
+  descriptor.SetFirstPacketInSubFrame(true);
+  descriptor.AddFrameDependencyDiff(kDiff);
+
+  ASSERT_EQ(RtpGenericFrameDescriptorExtension00::ValueSize(descriptor),
+            sizeof(kRaw));
+  uint8_t buffer[sizeof(kRaw)];
+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension00::Write(buffer, descriptor));
+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest,
+     ParseLongFrameDependenciesAsBigEndian) {
+  constexpr uint16_t kDiff = 0x7654 >> 2;
+  constexpr uint8_t kRaw[] = {0xb8, 0x01, 0x00, 0x00, 0x54 | 0x02, 0x76};
+  RtpGenericFrameDescriptor descriptor;
+
+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension00::Parse(kRaw, &descriptor));
+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest,
+     WriteLongFrameDependenciesAsBigEndian) {
+  constexpr uint16_t kDiff = 0x7654 >> 2;
+  uint8_t kRaw[] = {
+      0x88 | kDeprecatedFlags, 0x01, 0x00, 0x00, 0x54 | 0x02, 0x76};
+  RtpGenericFrameDescriptor descriptor;
+  descriptor.SetFirstPacketInSubFrame(true);
+  descriptor.AddFrameDependencyDiff(kDiff);
+
+  ASSERT_EQ(RtpGenericFrameDescriptorExtension00::ValueSize(descriptor),
+            sizeof(kRaw));
+  uint8_t buffer[sizeof(kRaw)];
+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension00::Write(buffer, descriptor));
+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, ParseMaxLongFrameDependencies) {
+  constexpr uint16_t kDiff = 0x3fff;
+  constexpr uint8_t kRaw[] = {0xb8, 0x01, 0x00, 0x00, 0xfe, 0xff};
+  RtpGenericFrameDescriptor descriptor;
+
+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension00::Parse(kRaw, &descriptor));
+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, WriteMaxLongFrameDependencies) {
+  constexpr uint16_t kDiff = 0x3fff;
+  uint8_t kRaw[] = {0x88 | kDeprecatedFlags, 0x01, 0x00, 0x00, 0xfe, 0xff};
+  RtpGenericFrameDescriptor descriptor;
+  descriptor.SetFirstPacketInSubFrame(true);
+  descriptor.AddFrameDependencyDiff(kDiff);
+
+  ASSERT_EQ(RtpGenericFrameDescriptorExtension00::ValueSize(descriptor),
+            sizeof(kRaw));
+  uint8_t buffer[sizeof(kRaw)];
+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension00::Write(buffer, descriptor));
+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, ParseTwoFrameDependencies) {
+  constexpr uint16_t kDiff1 = 9;
+  constexpr uint16_t kDiff2 = 15;
+  constexpr uint8_t kRaw[] = {
+      0xb8, 0x01, 0x00, 0x00, (kDiff1 << 2) | 0x01, kDiff2 << 2};
+  RtpGenericFrameDescriptor descriptor;
+
+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension00::Parse(kRaw, &descriptor));
+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff1, kDiff2));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest, WriteTwoFrameDependencies) {
+  constexpr uint16_t kDiff1 = 9;
+  constexpr uint16_t kDiff2 = 15;
+  uint8_t kRaw[] = {0x88 | kDeprecatedFlags, 0x01,       0x00, 0x00,
+                    (kDiff1 << 2) | 0x01,    kDiff2 << 2};
+  RtpGenericFrameDescriptor descriptor;
+  descriptor.SetFirstPacketInSubFrame(true);
+  descriptor.AddFrameDependencyDiff(kDiff1);
+  descriptor.AddFrameDependencyDiff(kDiff2);
+
+  ASSERT_EQ(RtpGenericFrameDescriptorExtension00::ValueSize(descriptor),
+            sizeof(kRaw));
+  uint8_t buffer[sizeof(kRaw)];
+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension00::Write(buffer, descriptor));
+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest,
+     ParseResolutionOnIndependentFrame) {
+  constexpr int kWidth = 0x2468;
+  constexpr int kHeight = 0x6543;
+  constexpr uint8_t kRaw[] = {0xb0, 0x01, 0x00, 0x00, 0x24, 0x68, 0x65, 0x43};
+  RtpGenericFrameDescriptor descriptor;
+
+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension00::Parse(kRaw, &descriptor));
+  EXPECT_EQ(descriptor.Width(), kWidth);
+  EXPECT_EQ(descriptor.Height(), kHeight);
+}
+
+TEST(RtpGenericFrameDescriptorExtensionTest,
+     WriteResolutionOnIndependentFrame) {
+  constexpr int kWidth = 0x2468;
+  constexpr int kHeight = 0x6543;
+  uint8_t kRaw[] = {
+      0x80 | kDeprecatedFlags, 0x01, 0x00, 0x00, 0x24, 0x68, 0x65, 0x43};
+  RtpGenericFrameDescriptor descriptor;
+  descriptor.SetFirstPacketInSubFrame(true);
+  descriptor.SetResolution(kWidth, kHeight);
+
+  ASSERT_EQ(RtpGenericFrameDescriptorExtension00::ValueSize(descriptor),
+            sizeof(kRaw));
+  uint8_t buffer[sizeof(kRaw)];
+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension00::Write(buffer, descriptor));
+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
+}
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
new file mode 100644
index 0000000000..4b8c7b5385
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+struct ExtensionInfo {
+  RTPExtensionType type;
+  absl::string_view uri;
+};
+
+template <typename Extension>
+constexpr ExtensionInfo CreateExtensionInfo() {
+  return {Extension::kId, Extension::Uri()};
+}
+
+constexpr ExtensionInfo kExtensions[] = {
+    CreateExtensionInfo<TransmissionOffset>(),
+    CreateExtensionInfo<AudioLevel>(),
+    CreateExtensionInfo<CsrcAudioLevel>(),
+    CreateExtensionInfo<AbsoluteSendTime>(),
+    CreateExtensionInfo<AbsoluteCaptureTimeExtension>(),
+    CreateExtensionInfo<VideoOrientation>(),
+    CreateExtensionInfo<TransportSequenceNumber>(),
+    CreateExtensionInfo<TransportSequenceNumberV2>(),
+    CreateExtensionInfo<PlayoutDelayLimits>(),
+    CreateExtensionInfo<VideoContentTypeExtension>(),
+    CreateExtensionInfo<RtpVideoLayersAllocationExtension>(),
+    CreateExtensionInfo<VideoTimingExtension>(),
+    CreateExtensionInfo<RtpStreamId>(),
+    CreateExtensionInfo<RepairedRtpStreamId>(),
+    CreateExtensionInfo<RtpMid>(),
+    CreateExtensionInfo<RtpGenericFrameDescriptorExtension00>(),
+    CreateExtensionInfo<RtpDependencyDescriptorExtension>(),
+    CreateExtensionInfo<ColorSpaceExtension>(),
+    CreateExtensionInfo<InbandComfortNoiseExtension>(),
+    CreateExtensionInfo<VideoFrameTrackingIdExtension>(),
+};
+
+// Because of kRtpExtensionNone, NumberOfExtension is 1 bigger than the actual
+// number of known extensions.
+static_assert(arraysize(kExtensions) ==
+                  static_cast<int>(kRtpExtensionNumberOfExtensions) - 1,
+              "kExtensions expect to list all known extensions");
+
+}  // namespace
+
+constexpr RTPExtensionType RtpHeaderExtensionMap::kInvalidType;
+constexpr int RtpHeaderExtensionMap::kInvalidId;
+
+RtpHeaderExtensionMap::RtpHeaderExtensionMap() : RtpHeaderExtensionMap(false) {}
+
+RtpHeaderExtensionMap::RtpHeaderExtensionMap(bool extmap_allow_mixed)
+    : extmap_allow_mixed_(extmap_allow_mixed) {
+  for (auto& id : ids_)
+    id = kInvalidId;
+}
+
+RtpHeaderExtensionMap::RtpHeaderExtensionMap(
+    rtc::ArrayView<const RtpExtension> extensions)
+    : RtpHeaderExtensionMap(false) {
+  for (const RtpExtension& extension : extensions)
+    RegisterByUri(extension.id, extension.uri);
+}
+
+void RtpHeaderExtensionMap::Reset(
+    rtc::ArrayView<const RtpExtension> extensions) {
+  for (auto& id : ids_)
+    id = kInvalidId;
+  for (const RtpExtension& extension : extensions)
+    RegisterByUri(extension.id, extension.uri);
+}
+
+bool RtpHeaderExtensionMap::RegisterByType(int id, RTPExtensionType type) {
+  for (const ExtensionInfo& extension : kExtensions)
+    if (type == extension.type)
+      return Register(id, extension.type, extension.uri);
+  RTC_DCHECK_NOTREACHED();
+  return false;
+}
+
+bool RtpHeaderExtensionMap::RegisterByUri(int id, absl::string_view uri) {
+  for (const ExtensionInfo& extension : kExtensions)
+    if (uri == extension.uri)
+      return Register(id, extension.type, extension.uri);
+  RTC_LOG(LS_WARNING) << "Unknown extension uri:'" << uri << "', id: " << id
+                      << '.';
+  return false;
+}
+
+RTPExtensionType RtpHeaderExtensionMap::GetType(int id) const {
+  RTC_DCHECK_GE(id, RtpExtension::kMinId);
+  RTC_DCHECK_LE(id, RtpExtension::kMaxId);
+  for (int type = kRtpExtensionNone + 1; type < kRtpExtensionNumberOfExtensions;
+       ++type) {
+    if (ids_[type] == id) {
+      return static_cast<RTPExtensionType>(type);
+    }
+  }
+  return kInvalidType;
+}
+
+void RtpHeaderExtensionMap::Deregister(absl::string_view uri) {
+  for (const ExtensionInfo& extension : kExtensions) {
+    if (extension.uri == uri) {
+      ids_[extension.type] = kInvalidId;
+      break;
+    }
+  }
+}
+
+bool RtpHeaderExtensionMap::Register(int id,
+                                     RTPExtensionType type,
+                                     absl::string_view uri) {
+  RTC_DCHECK_GT(type, kRtpExtensionNone);
+  RTC_DCHECK_LT(type, kRtpExtensionNumberOfExtensions);
+
+  if (id < RtpExtension::kMinId || id > RtpExtension::kMaxId) {
+    RTC_LOG(LS_WARNING) << "Failed to register extension uri:'" << uri
+                        << "' with invalid id:" << id << ".";
+    return false;
+  }
+
+  RTPExtensionType registered_type = GetType(id);
+  if (registered_type == type) {  // Same type/id pair already registered.
+    RTC_LOG(LS_VERBOSE) << "Reregistering extension uri:'" << uri
+                        << "', id:" << id;
+    return true;
+  }
+
+  if (registered_type !=
+      kInvalidType) {  // `id` used by another extension type.
+    RTC_LOG(LS_WARNING) << "Failed to register extension uri:'" << uri
+                        << "', id:" << id
+                        << ". Id already in use by extension type "
+                        << static_cast<int>(registered_type);
+    return false;
+  }
+  if (IsRegistered(type)) {
+    RTC_LOG(LS_WARNING) << "Illegal reregistration for uri: " << uri
+                        << " is previously registered with id " << GetId(type)
+                        << " and cannot be reregistered with id " << id;
+    return false;
+  }
+
+  // There is a run-time check above id fits into uint8_t.
+  ids_[type] = static_cast<uint8_t>(id);
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc
new file mode 100644
index 0000000000..42842cc876
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+
+#include <vector>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(RtpHeaderExtensionTest, RegisterByType) {
+  RtpHeaderExtensionMap map;
+  EXPECT_FALSE(map.IsRegistered(TransmissionOffset::kId));
+
+  EXPECT_TRUE(map.RegisterByType(3, TransmissionOffset::kId));
+
+  EXPECT_TRUE(map.IsRegistered(TransmissionOffset::kId));
+  EXPECT_EQ(3, map.GetId(TransmissionOffset::kId));
+  EXPECT_EQ(TransmissionOffset::kId, map.GetType(3));
+}
+
+TEST(RtpHeaderExtensionTest, RegisterByUri) {
+  RtpHeaderExtensionMap map;
+
+  EXPECT_TRUE(map.RegisterByUri(3, TransmissionOffset::Uri()));
+
+  EXPECT_TRUE(map.IsRegistered(TransmissionOffset::kId));
+  EXPECT_EQ(3, map.GetId(TransmissionOffset::kId));
+  EXPECT_EQ(TransmissionOffset::kId, map.GetType(3));
+}
+
+TEST(RtpHeaderExtensionTest, RegisterWithTrait) {
+  RtpHeaderExtensionMap map;
+
+  EXPECT_TRUE(map.Register<TransmissionOffset>(3));
+
+  EXPECT_TRUE(map.IsRegistered(TransmissionOffset::kId));
+  EXPECT_EQ(3, map.GetId(TransmissionOffset::kId));
+  EXPECT_EQ(TransmissionOffset::kId, map.GetType(3));
+}
+
+TEST(RtpHeaderExtensionTest, RegisterDuringContruction) {
+  const std::vector<RtpExtension> config = {{TransmissionOffset::Uri(), 1},
+                                            {AbsoluteSendTime::Uri(), 3}};
+  const RtpHeaderExtensionMap map(config);
+
+  EXPECT_EQ(1, map.GetId(TransmissionOffset::kId));
+  EXPECT_EQ(3, map.GetId(AbsoluteSendTime::kId));
+}
+
+TEST(RtpHeaderExtensionTest, RegisterTwoByteHeaderExtensions) {
+  RtpHeaderExtensionMap map;
+  // Two-byte header extension needed for id: [15-255].
+  EXPECT_TRUE(map.Register<TransmissionOffset>(18));
+  EXPECT_TRUE(map.Register<AbsoluteSendTime>(255));
+}
+
+TEST(RtpHeaderExtensionTest, RegisterIllegalArg) {
+  RtpHeaderExtensionMap map;
+  // Valid range for id: [1-255].
+  EXPECT_FALSE(map.Register<TransmissionOffset>(0));
+  EXPECT_FALSE(map.Register<TransmissionOffset>(256));
+}
+
+TEST(RtpHeaderExtensionTest, Idempotent) {
+  RtpHeaderExtensionMap map;
+
+  EXPECT_TRUE(map.Register<TransmissionOffset>(3));
+  EXPECT_TRUE(map.Register<TransmissionOffset>(3));
+
+  map.Deregister(TransmissionOffset::Uri());
+  map.Deregister(TransmissionOffset::Uri());
+}
+
+TEST(RtpHeaderExtensionTest, NonUniqueId) {
+  RtpHeaderExtensionMap map;
+  EXPECT_TRUE(map.Register<TransmissionOffset>(3));
+
+  EXPECT_FALSE(map.Register<AudioLevel>(3));
+  EXPECT_TRUE(map.Register<AudioLevel>(4));
+}
+
+TEST(RtpHeaderExtensionTest, GetType) {
+  RtpHeaderExtensionMap map;
+  EXPECT_EQ(RtpHeaderExtensionMap::kInvalidType, map.GetType(3));
+  EXPECT_TRUE(map.Register<TransmissionOffset>(3));
+
+  EXPECT_EQ(TransmissionOffset::kId, map.GetType(3));
+}
+
+TEST(RtpHeaderExtensionTest, GetId) {
+  RtpHeaderExtensionMap map;
+  EXPECT_EQ(RtpHeaderExtensionMap::kInvalidId,
+            map.GetId(TransmissionOffset::kId));
+  EXPECT_TRUE(map.Register<TransmissionOffset>(3));
+
+  EXPECT_EQ(3, map.GetId(TransmissionOffset::kId));
+}
+
+TEST(RtpHeaderExtensionTest, RemapFails) {
+  RtpHeaderExtensionMap map;
+  EXPECT_TRUE(map.Register<TransmissionOffset>(3));
+  EXPECT_FALSE(map.Register<TransmissionOffset>(4));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size.cc
new file mode 100644
index 0000000000..4acbcf4e6b
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size.cc
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
+
+#include "api/rtp_parameters.h"
+
+namespace webrtc {
+
+int RtpHeaderExtensionSize(rtc::ArrayView<const RtpExtensionSize> extensions,
+                           const RtpHeaderExtensionMap& registered_extensions) {
+  // RFC3550 Section 5.3.1
+  static constexpr int kExtensionBlockHeaderSize = 4;
+
+  int values_size = 0;
+  int num_extensions = 0;
+  int each_extension_header_size = 1;
+  for (const RtpExtensionSize& extension : extensions) {
+    int id = registered_extensions.GetId(extension.type);
+    if (id == RtpHeaderExtensionMap::kInvalidId)
+      continue;
+    // All extensions should use same size header. Check if the `extension`
+    // forces to switch to two byte header that allows larger id and value size.
+    if (id > RtpExtension::kOneByteHeaderExtensionMaxId ||
+        extension.value_size >
+            RtpExtension::kOneByteHeaderExtensionMaxValueSize) {
+      each_extension_header_size = 2;
+    }
+    values_size += extension.value_size;
+    num_extensions++;
+  }
+  if (values_size == 0)
+    return 0;
+  int size = kExtensionBlockHeaderSize +
+             each_extension_header_size * num_extensions + values_size;
+  // Extension size specified in 32bit words,
+  // so result must be multiple of 4 bytes. Round up.
+  return size + 3 - (size + 3) % 4;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size.h
new file mode 100644
index 0000000000..1fb2eb2a1e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_SIZE_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_SIZE_H_
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+struct RtpExtensionSize {
+  RTPExtensionType type;
+  int value_size;
+};
+
+// Calculates rtp header extension size in bytes assuming packet contain
+// all `extensions` with provided `value_size`.
+// Counts only extensions present among `registered_extensions`.
+int RtpHeaderExtensionSize(rtc::ArrayView<const RtpExtensionSize> extensions,
+                           const RtpHeaderExtensionMap& registered_extensions);
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_SIZE_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size_unittest.cc
new file mode 100644
index 0000000000..5cc26bc652
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extension_size_unittest.cc
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
+
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "test/gtest.h"
+
+namespace {
+
+using ::webrtc::RtpExtensionSize;
+using ::webrtc::RtpHeaderExtensionMap;
+using ::webrtc::RtpHeaderExtensionSize;
+using ::webrtc::RtpMid;
+using ::webrtc::RtpStreamId;
+
+// id for 1-byte header extension. actual value is irrelevant for these tests.
+constexpr int kId = 1;
+// id that forces to use 2-byte header extension.
+constexpr int kIdForceTwoByteHeader = 15;
+
+TEST(RtpHeaderExtensionSizeTest, ReturnsZeroIfNoExtensionsAreRegistered) {
+  constexpr RtpExtensionSize kExtensionSizes[] = {{RtpMid::kId, 3}};
+  // Register different extension than ask size for.
+  RtpHeaderExtensionMap registered;
+  registered.Register<RtpStreamId>(kId);
+
+  EXPECT_EQ(RtpHeaderExtensionSize(kExtensionSizes, registered), 0);
+}
+
+TEST(RtpHeaderExtensionSizeTest, IncludesSizeOfExtensionHeaders) {
+  constexpr RtpExtensionSize kExtensionSizes[] = {{RtpMid::kId, 3}};
+  RtpHeaderExtensionMap registered;
+  registered.Register<RtpMid>(kId);
+
+  // 4 bytes for extension block header + 1 byte for individual extension header
+  // + 3 bytes for the value.
+  EXPECT_EQ(RtpHeaderExtensionSize(kExtensionSizes, registered), 8);
+}
+
+TEST(RtpHeaderExtensionSizeTest, RoundsUpTo32bitAlignmant) {
+  constexpr RtpExtensionSize kExtensionSizes[] = {{RtpMid::kId, 5}};
+  RtpHeaderExtensionMap registered;
+  registered.Register<RtpMid>(kId);
+
+  // 10 bytes of data including headers + 2 bytes of padding.
+  EXPECT_EQ(RtpHeaderExtensionSize(kExtensionSizes, registered), 12);
+}
+
+TEST(RtpHeaderExtensionSizeTest, SumsSeveralExtensions) {
+  constexpr RtpExtensionSize kExtensionSizes[] = {{RtpMid::kId, 16},
+                                                  {RtpStreamId::kId, 2}};
+  RtpHeaderExtensionMap registered;
+  registered.Register<RtpMid>(kId);
+  registered.Register<RtpStreamId>(14);
+
+  // 4 bytes for extension block header + 18 bytes of value +
+  // 2 bytes for two headers
+  EXPECT_EQ(RtpHeaderExtensionSize(kExtensionSizes, registered), 24);
+}
+
+TEST(RtpHeaderExtensionSizeTest, LargeIdForce2BytesHeader) {
+  constexpr RtpExtensionSize kExtensionSizes[] = {{RtpMid::kId, 3},
+                                                  {RtpStreamId::kId, 2}};
+  RtpHeaderExtensionMap registered;
+  registered.Register<RtpMid>(kId);
+  registered.Register<RtpStreamId>(kIdForceTwoByteHeader);
+
+  // 4 bytes for extension block header + 5 bytes of value +
+  // 2*2 bytes for two headers + 3 bytes of padding.
+  EXPECT_EQ(RtpHeaderExtensionSize(kExtensionSizes, registered), 16);
+}
+
+TEST(RtpHeaderExtensionSizeTest, LargeValueForce2BytesHeader) {
+  constexpr RtpExtensionSize kExtensionSizes[] = {{RtpMid::kId, 17},
+                                                  {RtpStreamId::kId, 4}};
+  RtpHeaderExtensionMap registered;
+  registered.Register<RtpMid>(1);
+  registered.Register<RtpStreamId>(2);
+
+  // 4 bytes for extension block header + 21 bytes of value +
+  // 2*2 bytes for two headers + 3 byte of padding.
+  EXPECT_EQ(RtpHeaderExtensionSize(kExtensionSizes, registered), 32);
+}
+
+}  // namespace
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
new file mode 100644
index 0000000000..de29fd2075
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -0,0 +1,934 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+
+#include <string.h>
+
+#include <cmath>
+#include <cstdint>
+#include <limits>
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/include/rtp_cvo.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+// TODO(bug:9855) Move kNoSpatialIdx from vp9_globals.h to common_constants
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+// Absolute send time in RTP streams.
+//
+// The absolute send time is signaled to the receiver in-band using the
+// general mechanism for RTP header extensions [RFC8285]. The payload
+// of this extension (the transmitted value) is a 24-bit unsigned integer
+// containing the sender's current time in seconds as a fixed point number
+// with 18 bits fractional part.
+//
+// The form of the absolute send time extension block:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=2 |              absolute send time               |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType AbsoluteSendTime::kId;
+constexpr uint8_t AbsoluteSendTime::kValueSizeBytes;
+
+bool AbsoluteSendTime::Parse(rtc::ArrayView<const uint8_t> data,
+                             uint32_t* time_24bits) {
+  if (data.size() != 3)
+    return false;
+  *time_24bits = ByteReader<uint32_t, 3>::ReadBigEndian(data.data());
+  return true;
+}
+
+bool AbsoluteSendTime::Write(rtc::ArrayView<uint8_t> data,
+                             uint32_t time_24bits) {
+  RTC_DCHECK_EQ(data.size(), 3);
+  RTC_DCHECK_LE(time_24bits, 0x00FFFFFF);
+  ByteWriter<uint32_t, 3>::WriteBigEndian(data.data(), time_24bits);
+  return true;
+}
+
+// Absolute Capture Time
+//
+// The Absolute Capture Time extension is used to stamp RTP packets with a NTP
+// timestamp showing when the first audio or video frame in a packet was
+// originally captured. The intent of this extension is to provide a way to
+// accomplish audio-to-video synchronization when RTCP-terminating intermediate
+// systems (e.g. mixers) are involved.
+//
+// Data layout of the shortened version of abs-capture-time:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=7 |     absolute capture timestamp (bit 0-23)     |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |             absolute capture timestamp (bit 24-55)            |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ... (56-63)  |
+//   +-+-+-+-+-+-+-+-+
+//
+// Data layout of the extended version of abs-capture-time:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=15|     absolute capture timestamp (bit 0-23)     |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |             absolute capture timestamp (bit 24-55)            |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ... (56-63)  |   estimated capture clock offset (bit 0-23)   |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |           estimated capture clock offset (bit 24-55)          |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ... (56-63)  |
+//   +-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType AbsoluteCaptureTimeExtension::kId;
+constexpr uint8_t AbsoluteCaptureTimeExtension::kValueSizeBytes;
+constexpr uint8_t AbsoluteCaptureTimeExtension::
+    kValueSizeBytesWithoutEstimatedCaptureClockOffset;
+
+bool AbsoluteCaptureTimeExtension::Parse(rtc::ArrayView<const uint8_t> data,
+                                         AbsoluteCaptureTime* extension) {
+  if (data.size() != kValueSizeBytes &&
+      data.size() != kValueSizeBytesWithoutEstimatedCaptureClockOffset) {
+    return false;
+  }
+
+  extension->absolute_capture_timestamp =
+      ByteReader<uint64_t>::ReadBigEndian(data.data());
+
+  if (data.size() != kValueSizeBytesWithoutEstimatedCaptureClockOffset) {
+    extension->estimated_capture_clock_offset =
+        ByteReader<int64_t>::ReadBigEndian(data.data() + 8);
+  }
+
+  return true;
+}
+
+size_t AbsoluteCaptureTimeExtension::ValueSize(
+    const AbsoluteCaptureTime& extension) {
+  if (extension.estimated_capture_clock_offset != absl::nullopt) {
+    return kValueSizeBytes;
+  } else {
+    return kValueSizeBytesWithoutEstimatedCaptureClockOffset;
+  }
+}
+
+bool AbsoluteCaptureTimeExtension::Write(rtc::ArrayView<uint8_t> data,
+                                         const AbsoluteCaptureTime& extension) {
+  RTC_DCHECK_EQ(data.size(), ValueSize(extension));
+
+  ByteWriter<uint64_t>::WriteBigEndian(data.data(),
+                                       extension.absolute_capture_timestamp);
+
+  if (data.size() != kValueSizeBytesWithoutEstimatedCaptureClockOffset) {
+    ByteWriter<int64_t>::WriteBigEndian(
+        data.data() + 8, extension.estimated_capture_clock_offset.value());
+  }
+
+  return true;
+}
+
+// An RTP Header Extension for Client-to-Mixer Audio Level Indication
+//
+// https://tools.ietf.org/html/rfc6464
+//
+// The form of the audio level extension block:
+//
+//  0                   1
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |  ID   | len=0 |V| level       |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// Sample Audio Level Encoding Using the One-Byte Header Format
+//
+//  0                   1                   2
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |      ID       |     len=1     |V|    level    |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// Sample Audio Level Encoding Using the Two-Byte Header Format
+
+constexpr RTPExtensionType AudioLevel::kId;
+constexpr uint8_t AudioLevel::kValueSizeBytes;
+
+bool AudioLevel::Parse(rtc::ArrayView<const uint8_t> data,
+                       bool* voice_activity,
+                       uint8_t* audio_level) {
+  // One-byte and two-byte format share the same data definition.
+  if (data.size() != 1)
+    return false;
+  *voice_activity = (data[0] & 0x80) != 0;
+  *audio_level = data[0] & 0x7F;
+  return true;
+}
+
+bool AudioLevel::Write(rtc::ArrayView<uint8_t> data,
+                       bool voice_activity,
+                       uint8_t audio_level) {
+  // One-byte and two-byte format share the same data definition.
+  RTC_DCHECK_EQ(data.size(), 1);
+  RTC_CHECK_LE(audio_level, 0x7f);
+  data[0] = (voice_activity ? 0x80 : 0x00) | audio_level;
+  return true;
+}
+
+#if !defined(WEBRTC_MOZILLA_BUILD)
+// An RTP Header Extension for Mixer-to-Client Audio Level Indication
+//
+// https://tools.ietf.org/html/rfc6465
+//
+// The form of the audio level extension block:
+//
+//  0                   1                   2                   3
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |  ID   | len=2 |0|   level 1   |0|   level 2   |0|   level 3   |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// Sample Audio Level Encoding Using the One-Byte Header Format
+//
+//  0                   1                   2                   3
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |      ID       |     len=3     |0|   level 1   |0|   level 2   |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |0|   level 3   |    0 (pad)    |               ...             |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// Sample Audio Level Encoding Using the Two-Byte Header Format
+constexpr RTPExtensionType CsrcAudioLevel::kId;
+constexpr uint8_t CsrcAudioLevel::kMaxValueSizeBytes;
+
+bool CsrcAudioLevel::Parse(rtc::ArrayView<const uint8_t> data,
+                           std::vector<uint8_t>* csrc_audio_levels) {
+  if (data.size() > kRtpCsrcSize) {
+    return false;
+  }
+  csrc_audio_levels->resize(data.size());
+  for (size_t i = 0; i < data.size(); i++) {
+    (*csrc_audio_levels)[i] = data[i] & 0x7F;
+  }
+  return true;
+}
+
+size_t CsrcAudioLevel::ValueSize(
+    rtc::ArrayView<const uint8_t> csrc_audio_levels) {
+  return csrc_audio_levels.size();
+}
+
+bool CsrcAudioLevel::Write(rtc::ArrayView<uint8_t> data,
+                           rtc::ArrayView<const uint8_t> csrc_audio_levels) {
+  RTC_CHECK_LE(csrc_audio_levels.size(), kRtpCsrcSize);
+  if (csrc_audio_levels.size() != data.size()) {
+    return false;
+  }
+  for (size_t i = 0; i < csrc_audio_levels.size(); i++) {
+    data[i] = csrc_audio_levels[i] & 0x7F;
+  }
+  return true;
+}
+#endif
+
+// From RFC 5450: Transmission Time Offsets in RTP Streams.
+//
+// The transmission time is signaled to the receiver in-band using the
+// general mechanism for RTP header extensions [RFC8285]. The payload
+// of this extension (the transmitted value) is a 24-bit signed integer.
+// When added to the RTP timestamp of the packet, it represents the
+// "effective" RTP transmission time of the packet, on the RTP
+// timescale.
+//
+// The form of the transmission offset extension block:
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=2 |              transmission offset              |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType TransmissionOffset::kId;
+constexpr uint8_t TransmissionOffset::kValueSizeBytes;
+
+bool TransmissionOffset::Parse(rtc::ArrayView<const uint8_t> data,
+                               int32_t* rtp_time) {
+  if (data.size() != 3)
+    return false;
+  *rtp_time = ByteReader<int32_t, 3>::ReadBigEndian(data.data());
+  return true;
+}
+
+bool TransmissionOffset::Write(rtc::ArrayView<uint8_t> data, int32_t rtp_time) {
+  RTC_DCHECK_EQ(data.size(), 3);
+  RTC_DCHECK_LE(rtp_time, 0x00ffffff);
+  ByteWriter<int32_t, 3>::WriteBigEndian(data.data(), rtp_time);
+  return true;
+}
+
+// TransportSequenceNumber
+//
+//   0                   1                   2
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  ID   | L=1   |transport-wide sequence number |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType TransportSequenceNumber::kId;
+constexpr uint8_t TransportSequenceNumber::kValueSizeBytes;
+
+bool TransportSequenceNumber::Parse(rtc::ArrayView<const uint8_t> data,
+                                    uint16_t* transport_sequence_number) {
+  if (data.size() != kValueSizeBytes)
+    return false;
+  *transport_sequence_number = ByteReader<uint16_t>::ReadBigEndian(data.data());
+  return true;
+}
+
+bool TransportSequenceNumber::Write(rtc::ArrayView<uint8_t> data,
+                                    uint16_t transport_sequence_number) {
+  RTC_DCHECK_EQ(data.size(), ValueSize(transport_sequence_number));
+  ByteWriter<uint16_t>::WriteBigEndian(data.data(), transport_sequence_number);
+  return true;
+}
+
+// TransportSequenceNumberV2
+//
+// In addition to the format used for TransportSequencNumber, V2 also supports
+// the following packet format where two extra bytes are used to specify that
+// the sender requests immediate feedback.
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  ID   | L=3   |transport-wide sequence number |T|  seq count  |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |seq count cont.|
+//  +-+-+-+-+-+-+-+-+
+//
+// The bit `T` determines whether the feedback should include timing information
+// or not and `seq_count` determines how many packets the feedback packet should
+// cover including the current packet. If `seq_count` is zero no feedback is
+// requested.
+constexpr RTPExtensionType TransportSequenceNumberV2::kId;
+constexpr uint8_t TransportSequenceNumberV2::kValueSizeBytes;
+constexpr uint8_t
+    TransportSequenceNumberV2::kValueSizeBytesWithoutFeedbackRequest;
+constexpr uint16_t TransportSequenceNumberV2::kIncludeTimestampsBit;
+
+bool TransportSequenceNumberV2::Parse(
+    rtc::ArrayView<const uint8_t> data,
+    uint16_t* transport_sequence_number,
+    absl::optional<FeedbackRequest>* feedback_request) {
+  if (data.size() != kValueSizeBytes &&
+      data.size() != kValueSizeBytesWithoutFeedbackRequest)
+    return false;
+
+  *transport_sequence_number = ByteReader<uint16_t>::ReadBigEndian(data.data());
+
+  *feedback_request = absl::nullopt;
+  if (data.size() == kValueSizeBytes) {
+    uint16_t feedback_request_raw =
+        ByteReader<uint16_t>::ReadBigEndian(data.data() + 2);
+    bool include_timestamps =
+        (feedback_request_raw & kIncludeTimestampsBit) != 0;
+    uint16_t sequence_count = feedback_request_raw & ~kIncludeTimestampsBit;
+
+    // If `sequence_count` is zero no feedback is requested.
+    if (sequence_count != 0) {
+      *feedback_request = {include_timestamps, sequence_count};
+    }
+  }
+  return true;
+}
+
+bool TransportSequenceNumberV2::Write(
+    rtc::ArrayView<uint8_t> data,
+    uint16_t transport_sequence_number,
+    const absl::optional<FeedbackRequest>& feedback_request) {
+  RTC_DCHECK_EQ(data.size(),
+                ValueSize(transport_sequence_number, feedback_request));
+
+  ByteWriter<uint16_t>::WriteBigEndian(data.data(), transport_sequence_number);
+
+  if (feedback_request) {
+    RTC_DCHECK_GE(feedback_request->sequence_count, 0);
+    RTC_DCHECK_LT(feedback_request->sequence_count, kIncludeTimestampsBit);
+    uint16_t feedback_request_raw =
+        feedback_request->sequence_count |
+        (feedback_request->include_timestamps ? kIncludeTimestampsBit : 0);
+    ByteWriter<uint16_t>::WriteBigEndian(data.data() + 2, feedback_request_raw);
+  }
+  return true;
+}
+
+// Coordination of Video Orientation in RTP streams.
+//
+// Coordination of Video Orientation consists in signaling of the current
+// orientation of the image captured on the sender side to the receiver for
+// appropriate rendering and displaying.
+//
+//    0                   1
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=0 |0 0 0 0 C F R R|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType VideoOrientation::kId;
+constexpr uint8_t VideoOrientation::kValueSizeBytes;
+
+bool VideoOrientation::Parse(rtc::ArrayView<const uint8_t> data,
+                             VideoRotation* rotation) {
+  if (data.size() != 1)
+    return false;
+  *rotation = ConvertCVOByteToVideoRotation(data[0]);
+  return true;
+}
+
+bool VideoOrientation::Write(rtc::ArrayView<uint8_t> data,
+                             VideoRotation rotation) {
+  RTC_DCHECK_EQ(data.size(), 1);
+  data[0] = ConvertVideoRotationToCVOByte(rotation);
+  return true;
+}
+
+bool VideoOrientation::Parse(rtc::ArrayView<const uint8_t> data,
+                             uint8_t* value) {
+  if (data.size() != 1)
+    return false;
+  *value = data[0];
+  return true;
+}
+
+bool VideoOrientation::Write(rtc::ArrayView<uint8_t> data, uint8_t value) {
+  RTC_DCHECK_EQ(data.size(), 1);
+  data[0] = value;
+  return true;
+}
+
+//   0                   1                   2                   3
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  ID   | len=2 |   MIN delay           |   MAX delay           |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType PlayoutDelayLimits::kId;
+constexpr uint8_t PlayoutDelayLimits::kValueSizeBytes;
+
+bool PlayoutDelayLimits::Parse(rtc::ArrayView<const uint8_t> data,
+                               VideoPlayoutDelay* playout_delay) {
+  RTC_DCHECK(playout_delay);
+  if (data.size() != 3)
+    return false;
+  uint32_t raw = ByteReader<uint32_t, 3>::ReadBigEndian(data.data());
+  uint16_t min_raw = (raw >> 12);
+  uint16_t max_raw = (raw & 0xfff);
+  if (min_raw > max_raw)
+    return false;
+  playout_delay->min_ms = min_raw * kGranularityMs;
+  playout_delay->max_ms = max_raw * kGranularityMs;
+  return true;
+}
+
+bool PlayoutDelayLimits::Write(rtc::ArrayView<uint8_t> data,
+                               const VideoPlayoutDelay& playout_delay) {
+  RTC_DCHECK_EQ(data.size(), 3);
+  RTC_DCHECK_LE(0, playout_delay.min_ms);
+  RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms);
+  RTC_DCHECK_LE(playout_delay.max_ms, kMaxMs);
+  // Convert MS to value to be sent on extension header.
+  uint32_t min_delay = playout_delay.min_ms / kGranularityMs;
+  uint32_t max_delay = playout_delay.max_ms / kGranularityMs;
+  ByteWriter<uint32_t, 3>::WriteBigEndian(data.data(),
+                                          (min_delay << 12) | max_delay);
+  return true;
+}
+
+#if defined(WEBRTC_MOZILLA_BUILD)
+// CSRCAudioLevel
+//  Sample Audio Level Encoding Using the One-Byte Header Format
+//  Note that the range of len is 1 to 15 which is encoded as 0 to 14
+//  0                   1                   2                   3
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |  ID   | len=2 |0|   level 1   |0|   level 2   |0|   level 3   |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+
+constexpr RTPExtensionType CsrcAudioLevel::kId;
+constexpr const char* CsrcAudioLevel::kUri;
+
+bool CsrcAudioLevel::Parse(rtc::ArrayView<const uint8_t> data,
+                           CsrcAudioLevelList* csrcAudioLevels) {
+  if (data.size() < 1 || data.size() > kRtpCsrcSize)
+    return false;
+  csrcAudioLevels->numAudioLevels = data.size();
+  for(uint8_t i = 0; i < csrcAudioLevels->numAudioLevels; i++) {
+    // Ensure range is 0 to 127 inclusive
+    csrcAudioLevels->arrOfAudioLevels[i] = 0x7f & data[i];
+  }
+  return true;
+}
+
+size_t CsrcAudioLevel::ValueSize(const CsrcAudioLevelList& csrcAudioLevels) {
+  return csrcAudioLevels.numAudioLevels;
+}
+
+bool CsrcAudioLevel::Write(rtc::ArrayView<uint8_t> data,
+                           const CsrcAudioLevelList& csrcAudioLevels) {
+  RTC_DCHECK_GE(csrcAudioLevels.numAudioLevels, 0);
+  for(uint8_t i = 0; i < csrcAudioLevels.numAudioLevels; i++) {
+    data[i] = csrcAudioLevels.arrOfAudioLevels[i] & 0x7f;
+  }
+  // This extension if used must have at least one audio level
+  return csrcAudioLevels.numAudioLevels;
+}
+#endif
+
+// Video Content Type.
+//
+// E.g. default video or screenshare.
+//
+//    0                   1
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=0 | Content type  |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType VideoContentTypeExtension::kId;
+constexpr uint8_t VideoContentTypeExtension::kValueSizeBytes;
+
+bool VideoContentTypeExtension::Parse(rtc::ArrayView<const uint8_t> data,
+                                      VideoContentType* content_type) {
+  if (data.size() == 1 &&
+      videocontenttypehelpers::IsValidContentType(data[0])) {
+    *content_type = static_cast<VideoContentType>(data[0]);
+    return true;
+  }
+  return false;
+}
+
+bool VideoContentTypeExtension::Write(rtc::ArrayView<uint8_t> data,
+                                      VideoContentType content_type) {
+  RTC_DCHECK_EQ(data.size(), 1);
+  data[0] = static_cast<uint8_t>(content_type);
+  return true;
+}
+
+// Video Timing.
+// 6 timestamps in milliseconds counted from capture time stored in rtp header:
+// encode start/finish, packetization complete, pacer exit and reserved for
+// modification by the network modification. `flags` is a bitmask and has the
+// following allowed values:
+// 0 = Valid data, but no flags available (backwards compatibility)
+// 1 = Frame marked as timing frame due to cyclic timer.
+// 2 = Frame marked as timing frame due to size being outside limit.
+// 255 = Invalid. The whole timing frame extension should be ignored.
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=12|     flags     |     encode start ms delta     |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |    encode finish ms delta     |  packetizer finish ms delta   |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |     pacer exit ms delta       |  network timestamp ms delta   |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  network2 timestamp ms delta  |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+constexpr RTPExtensionType VideoTimingExtension::kId;
+constexpr uint8_t VideoTimingExtension::kValueSizeBytes;
+constexpr uint8_t VideoTimingExtension::kFlagsOffset;
+constexpr uint8_t VideoTimingExtension::kEncodeStartDeltaOffset;
+constexpr uint8_t VideoTimingExtension::kEncodeFinishDeltaOffset;
+constexpr uint8_t VideoTimingExtension::kPacketizationFinishDeltaOffset;
+constexpr uint8_t VideoTimingExtension::kPacerExitDeltaOffset;
+constexpr uint8_t VideoTimingExtension::kNetworkTimestampDeltaOffset;
+constexpr uint8_t VideoTimingExtension::kNetwork2TimestampDeltaOffset;
+
+bool VideoTimingExtension::Parse(rtc::ArrayView<const uint8_t> data,
+                                 VideoSendTiming* timing) {
+  RTC_DCHECK(timing);
+  // TODO(sprang): Deprecate support for old wire format.
+  ptrdiff_t off = 0;
+  switch (data.size()) {
+    case kValueSizeBytes - 1:
+      timing->flags = 0;
+      off = 1;  // Old wire format without the flags field.
+      break;
+    case kValueSizeBytes:
+      timing->flags = ByteReader<uint8_t>::ReadBigEndian(data.data());
+      break;
+    default:
+      return false;
+  }
+
+  timing->encode_start_delta_ms = ByteReader<uint16_t>::ReadBigEndian(
+      data.data() + kEncodeStartDeltaOffset - off);
+  timing->encode_finish_delta_ms = ByteReader<uint16_t>::ReadBigEndian(
+      data.data() + kEncodeFinishDeltaOffset - off);
+  timing->packetization_finish_delta_ms = ByteReader<uint16_t>::ReadBigEndian(
+      data.data() + kPacketizationFinishDeltaOffset - off);
+  timing->pacer_exit_delta_ms = ByteReader<uint16_t>::ReadBigEndian(
+      data.data() + kPacerExitDeltaOffset - off);
+  timing->network_timestamp_delta_ms = ByteReader<uint16_t>::ReadBigEndian(
+      data.data() + kNetworkTimestampDeltaOffset - off);
+  timing->network2_timestamp_delta_ms = ByteReader<uint16_t>::ReadBigEndian(
+      data.data() + kNetwork2TimestampDeltaOffset - off);
+  return true;
+}
+
+bool VideoTimingExtension::Write(rtc::ArrayView<uint8_t> data,
+                                 const VideoSendTiming& timing) {
+  RTC_DCHECK_EQ(data.size(), 1 + 2 * 6);
+  ByteWriter<uint8_t>::WriteBigEndian(data.data() + kFlagsOffset, timing.flags);
+  ByteWriter<uint16_t>::WriteBigEndian(data.data() + kEncodeStartDeltaOffset,
+                                       timing.encode_start_delta_ms);
+  ByteWriter<uint16_t>::WriteBigEndian(data.data() + kEncodeFinishDeltaOffset,
+                                       timing.encode_finish_delta_ms);
+  ByteWriter<uint16_t>::WriteBigEndian(
+      data.data() + kPacketizationFinishDeltaOffset,
+      timing.packetization_finish_delta_ms);
+  ByteWriter<uint16_t>::WriteBigEndian(data.data() + kPacerExitDeltaOffset,
+                                       timing.pacer_exit_delta_ms);
+  ByteWriter<uint16_t>::WriteBigEndian(
+      data.data() + kNetworkTimestampDeltaOffset,
+      timing.network_timestamp_delta_ms);
+  ByteWriter<uint16_t>::WriteBigEndian(
+      data.data() + kNetwork2TimestampDeltaOffset,
+      timing.network2_timestamp_delta_ms);
+  return true;
+}
+
+bool VideoTimingExtension::Write(rtc::ArrayView<uint8_t> data,
+                                 uint16_t time_delta_ms,
+                                 uint8_t offset) {
+  RTC_DCHECK_GE(data.size(), offset + 2);
+  RTC_DCHECK_LE(offset, kValueSizeBytes - sizeof(uint16_t));
+  ByteWriter<uint16_t>::WriteBigEndian(data.data() + offset, time_delta_ms);
+  return true;
+}
+
+// Color space including HDR metadata as an optional field.
+//
+// RTP header extension to carry color space information and optionally HDR
+// metadata. The float values in the HDR metadata struct are upscaled by a
+// static factor and transmitted as unsigned integers.
+//
+// Data layout of color space with HDR metadata (two-byte RTP header extension)
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |      ID       |   length=28   |   primaries   |   transfer    |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |    matrix     |range+chr.sit. |         luminance_max         |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |         luminance_min         |            mastering_metadata.|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |primary_r.x and .y             |            mastering_metadata.|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |primary_g.x and .y             |            mastering_metadata.|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |primary_b.x and .y             |            mastering_metadata.|
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |white.x and .y                 |    max_content_light_level    |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   | max_frame_average_light_level |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Data layout of color space w/o HDR metadata (one-byte RTP header extension)
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | L = 3 |   primaries   |   transfer    |    matrix     |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |range+chr.sit. |
+//   +-+-+-+-+-+-+-+-+
+
+constexpr RTPExtensionType ColorSpaceExtension::kId;
+constexpr uint8_t ColorSpaceExtension::kValueSizeBytes;
+
+bool ColorSpaceExtension::Parse(rtc::ArrayView<const uint8_t> data,
+                                ColorSpace* color_space) {
+  RTC_DCHECK(color_space);
+  if (data.size() != kValueSizeBytes &&
+      data.size() != kValueSizeBytesWithoutHdrMetadata)
+    return false;
+
+  size_t offset = 0;
+  // Read color space information.
+  if (!color_space->set_primaries_from_uint8(data[offset++]))
+    return false;
+  if (!color_space->set_transfer_from_uint8(data[offset++]))
+    return false;
+  if (!color_space->set_matrix_from_uint8(data[offset++]))
+    return false;
+
+  uint8_t range_and_chroma_siting = data[offset++];
+  if (!color_space->set_range_from_uint8((range_and_chroma_siting >> 4) & 0x03))
+    return false;
+  if (!color_space->set_chroma_siting_horizontal_from_uint8(
+          (range_and_chroma_siting >> 2) & 0x03))
+    return false;
+  if (!color_space->set_chroma_siting_vertical_from_uint8(
+          range_and_chroma_siting & 0x03))
+    return false;
+
+  // Read HDR metadata if it exists, otherwise clear it.
+  if (data.size() == kValueSizeBytesWithoutHdrMetadata) {
+    color_space->set_hdr_metadata(nullptr);
+  } else {
+    HdrMetadata hdr_metadata;
+    offset += ParseHdrMetadata(data.subview(offset), &hdr_metadata);
+    if (!hdr_metadata.Validate())
+      return false;
+    color_space->set_hdr_metadata(&hdr_metadata);
+  }
+  RTC_DCHECK_EQ(ValueSize(*color_space), offset);
+  return true;
+}
+
+bool ColorSpaceExtension::Write(rtc::ArrayView<uint8_t> data,
+                                const ColorSpace& color_space) {
+  RTC_DCHECK_EQ(data.size(), ValueSize(color_space));
+  size_t offset = 0;
+  // Write color space information.
+  data[offset++] = static_cast<uint8_t>(color_space.primaries());
+  data[offset++] = static_cast<uint8_t>(color_space.transfer());
+  data[offset++] = static_cast<uint8_t>(color_space.matrix());
+  data[offset++] = CombineRangeAndChromaSiting(
+      color_space.range(), color_space.chroma_siting_horizontal(),
+      color_space.chroma_siting_vertical());
+
+  // Write HDR metadata if it exists.
+  if (color_space.hdr_metadata()) {
+    offset +=
+        WriteHdrMetadata(data.subview(offset), *color_space.hdr_metadata());
+  }
+  RTC_DCHECK_EQ(ValueSize(color_space), offset);
+  return true;
+}
+
+// Combines range and chroma siting into one byte with the following bit layout:
+// bits 0-1 Chroma siting vertical.
+//      2-3 Chroma siting horizontal.
+//      4-5 Range.
+//      6-7 Unused.
+uint8_t ColorSpaceExtension::CombineRangeAndChromaSiting(
+    ColorSpace::RangeID range,
+    ColorSpace::ChromaSiting chroma_siting_horizontal,
+    ColorSpace::ChromaSiting chroma_siting_vertical) {
+  RTC_DCHECK_LE(static_cast<uint8_t>(range), 3);
+  RTC_DCHECK_LE(static_cast<uint8_t>(chroma_siting_horizontal), 3);
+  RTC_DCHECK_LE(static_cast<uint8_t>(chroma_siting_vertical), 3);
+  return (static_cast<uint8_t>(range) << 4) |
+         (static_cast<uint8_t>(chroma_siting_horizontal) << 2) |
+         static_cast<uint8_t>(chroma_siting_vertical);
+}
+
+size_t ColorSpaceExtension::ParseHdrMetadata(rtc::ArrayView<const uint8_t> data,
+                                             HdrMetadata* hdr_metadata) {
+  RTC_DCHECK_EQ(data.size(),
+                kValueSizeBytes - kValueSizeBytesWithoutHdrMetadata);
+  size_t offset = 0;
+  offset += ParseLuminance(data.data() + offset,
+                           &hdr_metadata->mastering_metadata.luminance_max,
+                           kLuminanceMaxDenominator);
+  offset += ParseLuminance(data.data() + offset,
+                           &hdr_metadata->mastering_metadata.luminance_min,
+                           kLuminanceMinDenominator);
+  offset += ParseChromaticity(data.data() + offset,
+                              &hdr_metadata->mastering_metadata.primary_r);
+  offset += ParseChromaticity(data.data() + offset,
+                              &hdr_metadata->mastering_metadata.primary_g);
+  offset += ParseChromaticity(data.data() + offset,
+                              &hdr_metadata->mastering_metadata.primary_b);
+  offset += ParseChromaticity(data.data() + offset,
+                              &hdr_metadata->mastering_metadata.white_point);
+  hdr_metadata->max_content_light_level =
+      ByteReader<uint16_t>::ReadBigEndian(data.data() + offset);
+  offset += 2;
+  hdr_metadata->max_frame_average_light_level =
+      ByteReader<uint16_t>::ReadBigEndian(data.data() + offset);
+  offset += 2;
+  return offset;
+}
+
+size_t ColorSpaceExtension::ParseChromaticity(
+    const uint8_t* data,
+    HdrMasteringMetadata::Chromaticity* p) {
+  uint16_t chromaticity_x_scaled = ByteReader<uint16_t>::ReadBigEndian(data);
+  uint16_t chromaticity_y_scaled =
+      ByteReader<uint16_t>::ReadBigEndian(data + 2);
+  p->x = static_cast<float>(chromaticity_x_scaled) / kChromaticityDenominator;
+  p->y = static_cast<float>(chromaticity_y_scaled) / kChromaticityDenominator;
+  return 4;  // Return number of bytes read.
+}
+
+size_t ColorSpaceExtension::ParseLuminance(const uint8_t* data,
+                                           float* f,
+                                           int denominator) {
+  uint16_t luminance_scaled = ByteReader<uint16_t>::ReadBigEndian(data);
+  *f = static_cast<float>(luminance_scaled) / denominator;
+  return 2;  // Return number of bytes read.
+}
+
+size_t ColorSpaceExtension::WriteHdrMetadata(rtc::ArrayView<uint8_t> data,
+                                             const HdrMetadata& hdr_metadata) {
+  RTC_DCHECK_EQ(data.size(),
+                kValueSizeBytes - kValueSizeBytesWithoutHdrMetadata);
+  RTC_DCHECK(hdr_metadata.Validate());
+  size_t offset = 0;
+  offset += WriteLuminance(data.data() + offset,
+                           hdr_metadata.mastering_metadata.luminance_max,
+                           kLuminanceMaxDenominator);
+  offset += WriteLuminance(data.data() + offset,
+                           hdr_metadata.mastering_metadata.luminance_min,
+                           kLuminanceMinDenominator);
+  offset += WriteChromaticity(data.data() + offset,
+                              hdr_metadata.mastering_metadata.primary_r);
+  offset += WriteChromaticity(data.data() + offset,
+                              hdr_metadata.mastering_metadata.primary_g);
+  offset += WriteChromaticity(data.data() + offset,
+                              hdr_metadata.mastering_metadata.primary_b);
+  offset += WriteChromaticity(data.data() + offset,
+                              hdr_metadata.mastering_metadata.white_point);
+
+  ByteWriter<uint16_t>::WriteBigEndian(data.data() + offset,
+                                       hdr_metadata.max_content_light_level);
+  offset += 2;
+  ByteWriter<uint16_t>::WriteBigEndian(
+      data.data() + offset, hdr_metadata.max_frame_average_light_level);
+  offset += 2;
+  return offset;
+}
+
+size_t ColorSpaceExtension::WriteChromaticity(
+    uint8_t* data,
+    const HdrMasteringMetadata::Chromaticity& p) {
+  RTC_DCHECK_GE(p.x, 0.0f);
+  RTC_DCHECK_LE(p.x, 1.0f);
+  RTC_DCHECK_GE(p.y, 0.0f);
+  RTC_DCHECK_LE(p.y, 1.0f);
+  ByteWriter<uint16_t>::WriteBigEndian(
+      data, std::round(p.x * kChromaticityDenominator));
+  ByteWriter<uint16_t>::WriteBigEndian(
+      data + 2, std::round(p.y * kChromaticityDenominator));
+  return 4;  // Return number of bytes written.
+}
+
+size_t ColorSpaceExtension::WriteLuminance(uint8_t* data,
+                                           float f,
+                                           int denominator) {
+  RTC_DCHECK_GE(f, 0.0f);
+  float upscaled_value = f * denominator;
+  RTC_DCHECK_LE(upscaled_value, std::numeric_limits<uint16_t>::max());
+  ByteWriter<uint16_t>::WriteBigEndian(data, std::round(upscaled_value));
+  return 2;  // Return number of bytes written.
+}
+
+bool BaseRtpStringExtension::Parse(rtc::ArrayView<const uint8_t> data,
+                                   std::string* str) {
+  if (data.empty() || data[0] == 0)  // Valid string extension can't be empty.
+    return false;
+  const char* cstr = reinterpret_cast<const char*>(data.data());
+  // If there is a \0 character in the middle of the `data`, treat it as end
+  // of the string. Well-formed string extensions shouldn't contain it.
+  str->assign(cstr, strnlen(cstr, data.size()));
+  RTC_DCHECK(!str->empty());
+  return true;
+}
+
+bool BaseRtpStringExtension::Write(rtc::ArrayView<uint8_t> data,
+                                   absl::string_view str) {
+  if (str.size() > kMaxValueSizeBytes) {
+    return false;
+  }
+  RTC_DCHECK_EQ(data.size(), str.size());
+  RTC_DCHECK_GE(str.size(), 1);
+  memcpy(data.data(), str.data(), str.size());
+  return true;
+}
+
+// Constant declarations for RTP header extension types.
+constexpr RTPExtensionType RtpStreamId::kId;
+constexpr RTPExtensionType RepairedRtpStreamId::kId;
+constexpr RTPExtensionType RtpMid::kId;
+
+// An RTP Header Extension for Inband Comfort Noise
+//
+// The form of the audio level extension block:
+//
+//  0                   1
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |  ID   | len=0 |N| level       |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// Sample Audio Level Encoding Using the One-Byte Header Format
+//
+//  0                   1                   2
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |      ID       |     len=1     |N|    level    |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// Sample Audio Level Encoding Using the Two-Byte Header Format
+
+constexpr RTPExtensionType InbandComfortNoiseExtension::kId;
+constexpr uint8_t InbandComfortNoiseExtension::kValueSizeBytes;
+constexpr const char InbandComfortNoiseExtension::kUri[];
+
+bool InbandComfortNoiseExtension::Parse(rtc::ArrayView<const uint8_t> data,
+                                        absl::optional<uint8_t>* level) {
+  if (data.size() != kValueSizeBytes)
+    return false;
+  *level = (data[0] & 0b1000'0000) != 0
+               ? absl::nullopt
+               : absl::make_optional(data[0] & 0b0111'1111);
+  return true;
+}
+
+bool InbandComfortNoiseExtension::Write(rtc::ArrayView<uint8_t> data,
+                                        absl::optional<uint8_t> level) {
+  RTC_DCHECK_EQ(data.size(), kValueSizeBytes);
+  data[0] = 0b0000'0000;
+  if (level) {
+    if (*level > 127) {
+      return false;
+    }
+    data[0] = 0b1000'0000 | *level;
+  }
+  return true;
+}
+
+// VideoFrameTrackingIdExtension
+//
+//   0                   1                   2
+//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  |  ID   | L=1   |    video-frame-tracking-id    |
+//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+constexpr RTPExtensionType VideoFrameTrackingIdExtension::kId;
+constexpr uint8_t VideoFrameTrackingIdExtension::kValueSizeBytes;
+
+bool VideoFrameTrackingIdExtension::Parse(rtc::ArrayView<const uint8_t> data,
+                                          uint16_t* video_frame_tracking_id) {
+  if (data.size() != kValueSizeBytes) {
+    return false;
+  }
+  *video_frame_tracking_id = ByteReader<uint16_t>::ReadBigEndian(data.data());
+  return true;
+}
+
+bool VideoFrameTrackingIdExtension::Write(rtc::ArrayView<uint8_t> data,
+                                          uint16_t video_frame_tracking_id) {
+  RTC_DCHECK_EQ(data.size(), kValueSizeBytes);
+  ByteWriter<uint16_t>::WriteBigEndian(data.data(), video_frame_tracking_id);
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
new file mode 100644
index 0000000000..4b4984bf6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -0,0 +1,386 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/rtp_headers.h"
+#include "api/rtp_parameters.h"
+#include "api/units/timestamp.h"
+#include "api/video/color_space.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_timing.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class AbsoluteSendTime {
+ public:
+  using value_type = uint32_t;
+  static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteSendTime;
+  static constexpr uint8_t kValueSizeBytes = 3;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kAbsSendTimeUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data, uint32_t* time_24bits);
+  static size_t ValueSize(uint32_t time_24bits) { return kValueSizeBytes; }
+  static bool Write(rtc::ArrayView<uint8_t> data, uint32_t time_24bits);
+
+  static constexpr uint32_t To24Bits(Timestamp time) {
+    int64_t time_us = time.us() % (int64_t{1 << 6} * 1'000'000);
+    int64_t time6x18 = (time_us << 18) / 1'000'000;
+    RTC_DCHECK_GE(time6x18, 0);
+    RTC_DCHECK_LT(time6x18, 1 << 24);
+    return static_cast<uint32_t>(time6x18);
+  }
+};
+
+class AbsoluteCaptureTimeExtension {
+ public:
+  using value_type = AbsoluteCaptureTime;
+  static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteCaptureTime;
+  static constexpr uint8_t kValueSizeBytes = 16;
+  static constexpr uint8_t kValueSizeBytesWithoutEstimatedCaptureClockOffset =
+      8;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kAbsoluteCaptureTimeUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    AbsoluteCaptureTime* extension);
+  static size_t ValueSize(const AbsoluteCaptureTime& extension);
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    const AbsoluteCaptureTime& extension);
+};
+
+class AudioLevel {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionAudioLevel;
+  static constexpr uint8_t kValueSizeBytes = 1;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kAudioLevelUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    bool* voice_activity,
+                    uint8_t* audio_level);
+  static size_t ValueSize(bool voice_activity, uint8_t audio_level) {
+    return kValueSizeBytes;
+  }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    bool voice_activity,
+                    uint8_t audio_level);
+};
+
+#if !defined(WEBRTC_MOZILLA_BUILD)
+class CsrcAudioLevel {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
+  static constexpr uint8_t kMaxValueSizeBytes = 15;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kCsrcAudioLevelsUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    std::vector<uint8_t>* csrc_audio_levels);
+  static size_t ValueSize(rtc::ArrayView<const uint8_t> csrc_audio_levels);
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    rtc::ArrayView<const uint8_t> csrc_audio_levels);
+};
+#endif
+
+class TransmissionOffset {
+ public:
+  using value_type = int32_t;
+  static constexpr RTPExtensionType kId = kRtpExtensionTransmissionTimeOffset;
+  static constexpr uint8_t kValueSizeBytes = 3;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kTimestampOffsetUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data, int32_t* rtp_time);
+  static size_t ValueSize(int32_t rtp_time) { return kValueSizeBytes; }
+  static bool Write(rtc::ArrayView<uint8_t> data, int32_t rtp_time);
+};
+
+class TransportSequenceNumber {
+ public:
+  using value_type = uint16_t;
+  static constexpr RTPExtensionType kId = kRtpExtensionTransportSequenceNumber;
+  static constexpr uint8_t kValueSizeBytes = 2;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kTransportSequenceNumberUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    uint16_t* transport_sequence_number);
+  static size_t ValueSize(uint16_t /*transport_sequence_number*/) {
+    return kValueSizeBytes;
+  }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    uint16_t transport_sequence_number);
+};
+
+class TransportSequenceNumberV2 {
+ public:
+  static constexpr RTPExtensionType kId =
+      kRtpExtensionTransportSequenceNumber02;
+  static constexpr uint8_t kValueSizeBytes = 4;
+  static constexpr uint8_t kValueSizeBytesWithoutFeedbackRequest = 2;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kTransportSequenceNumberV2Uri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    uint16_t* transport_sequence_number,
+                    absl::optional<FeedbackRequest>* feedback_request);
+  static size_t ValueSize(
+      uint16_t /*transport_sequence_number*/,
+      const absl::optional<FeedbackRequest>& feedback_request) {
+    return feedback_request ? kValueSizeBytes
+                            : kValueSizeBytesWithoutFeedbackRequest;
+  }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    uint16_t transport_sequence_number,
+                    const absl::optional<FeedbackRequest>& feedback_request);
+
+ private:
+  static constexpr uint16_t kIncludeTimestampsBit = 1 << 15;
+};
+
+class VideoOrientation {
+ public:
+  using value_type = VideoRotation;
+  static constexpr RTPExtensionType kId = kRtpExtensionVideoRotation;
+  static constexpr uint8_t kValueSizeBytes = 1;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoRotationUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data, VideoRotation* value);
+  static size_t ValueSize(VideoRotation) { return kValueSizeBytes; }
+  static bool Write(rtc::ArrayView<uint8_t> data, VideoRotation value);
+  static bool Parse(rtc::ArrayView<const uint8_t> data, uint8_t* value);
+  static size_t ValueSize(uint8_t value) { return kValueSizeBytes; }
+  static bool Write(rtc::ArrayView<uint8_t> data, uint8_t value);
+};
+
+class PlayoutDelayLimits {
+ public:
+  using value_type = VideoPlayoutDelay;
+  static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
+  static constexpr uint8_t kValueSizeBytes = 3;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kPlayoutDelayUri;
+  }
+
+  // Playout delay in milliseconds. A playout delay limit (min or max)
+  // has 12 bits allocated. This allows a range of 0-4095 values which
+  // translates to a range of 0-40950 in milliseconds.
+  static constexpr int kGranularityMs = 10;
+  // Maximum playout delay value in milliseconds.
+  static constexpr int kMaxMs = 0xfff * kGranularityMs;  // 40950.
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    VideoPlayoutDelay* playout_delay);
+  static size_t ValueSize(const VideoPlayoutDelay&) { return kValueSizeBytes; }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    const VideoPlayoutDelay& playout_delay);
+};
+
+class VideoContentTypeExtension {
+ public:
+  using value_type = VideoContentType;
+  static constexpr RTPExtensionType kId = kRtpExtensionVideoContentType;
+  static constexpr uint8_t kValueSizeBytes = 1;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoContentTypeUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    VideoContentType* content_type);
+  static size_t ValueSize(VideoContentType) { return kValueSizeBytes; }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    VideoContentType content_type);
+};
+
+class VideoTimingExtension {
+ public:
+  using value_type = VideoSendTiming;
+  static constexpr RTPExtensionType kId = kRtpExtensionVideoTiming;
+  static constexpr uint8_t kValueSizeBytes = 13;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoTimingUri;
+  }
+
+  // Offsets of the fields in the RTP header extension, counting from the first
+  // byte after the one-byte header.
+  static constexpr uint8_t kFlagsOffset = 0;
+  static constexpr uint8_t kEncodeStartDeltaOffset = 1;
+  static constexpr uint8_t kEncodeFinishDeltaOffset = 3;
+  static constexpr uint8_t kPacketizationFinishDeltaOffset = 5;
+  static constexpr uint8_t kPacerExitDeltaOffset = 7;
+  static constexpr uint8_t kNetworkTimestampDeltaOffset = 9;
+  static constexpr uint8_t kNetwork2TimestampDeltaOffset = 11;
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    VideoSendTiming* timing);
+  static size_t ValueSize(const VideoSendTiming&) { return kValueSizeBytes; }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    const VideoSendTiming& timing);
+
+  static size_t ValueSize(uint16_t time_delta_ms, uint8_t idx) {
+    return kValueSizeBytes;
+  }
+  // Writes only single time delta to position idx.
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    uint16_t time_delta_ms,
+                    uint8_t offset);
+};
+
+class ColorSpaceExtension {
+ public:
+  using value_type = ColorSpace;
+  static constexpr RTPExtensionType kId = kRtpExtensionColorSpace;
+  static constexpr uint8_t kValueSizeBytes = 28;
+  static constexpr uint8_t kValueSizeBytesWithoutHdrMetadata = 4;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kColorSpaceUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    ColorSpace* color_space);
+  static size_t ValueSize(const ColorSpace& color_space) {
+    return color_space.hdr_metadata() ? kValueSizeBytes
+                                      : kValueSizeBytesWithoutHdrMetadata;
+  }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    const ColorSpace& color_space);
+
+ private:
+  static constexpr int kChromaticityDenominator = 50000;  // 0.00002 resolution.
+  static constexpr int kLuminanceMaxDenominator = 1;      // 1 resolution.
+  static constexpr int kLuminanceMinDenominator = 10000;  // 0.0001 resolution.
+
+  static uint8_t CombineRangeAndChromaSiting(
+      ColorSpace::RangeID range,
+      ColorSpace::ChromaSiting chroma_siting_horizontal,
+      ColorSpace::ChromaSiting chroma_siting_vertical);
+  static size_t ParseHdrMetadata(rtc::ArrayView<const uint8_t> data,
+                                 HdrMetadata* hdr_metadata);
+  static size_t ParseChromaticity(const uint8_t* data,
+                                  HdrMasteringMetadata::Chromaticity* p);
+  static size_t ParseLuminance(const uint8_t* data, float* f, int denominator);
+  static size_t WriteHdrMetadata(rtc::ArrayView<uint8_t> data,
+                                 const HdrMetadata& hdr_metadata);
+  static size_t WriteChromaticity(uint8_t* data,
+                                  const HdrMasteringMetadata::Chromaticity& p);
+  static size_t WriteLuminance(uint8_t* data, float f, int denominator);
+};
+
+#if defined(WEBRTC_MOZILLA_BUILD)
+class CsrcAudioLevel {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kCsrcAudioLevelsUri;
+  }
+  static constexpr const char* kUri =
+      "urn:ietf:params:rtp-hdrext:csrc-audio-level";
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    CsrcAudioLevelList* csrcAudioLevels);
+  static size_t ValueSize(const CsrcAudioLevelList& csrcAudioLevels);
+  static bool Write(rtc::ArrayView<uint8_t> data, const CsrcAudioLevelList& csrcAudioLevels);
+};
+#endif
+
+// Base extension class for RTP header extensions which are strings.
+// Subclasses must defined kId and kUri static constexpr members.
+class BaseRtpStringExtension {
+ public:
+  using value_type = std::string;
+  // String RTP header extensions are limited to 16 bytes because it is the
+  // maximum length that can be encoded with one-byte header extensions.
+  static constexpr uint8_t kMaxValueSizeBytes = 16;
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data, std::string* str);
+  static size_t ValueSize(absl::string_view str) { return str.size(); }
+  static bool Write(rtc::ArrayView<uint8_t> data, absl::string_view str);
+};
+
+class RtpStreamId : public BaseRtpStringExtension {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
+  static constexpr absl::string_view Uri() { return RtpExtension::kRidUri; }
+};
+
+class RepairedRtpStreamId : public BaseRtpStringExtension {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionRepairedRtpStreamId;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kRepairedRidUri;
+  }
+};
+
+class RtpMid : public BaseRtpStringExtension {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionMid;
+  static constexpr absl::string_view Uri() { return RtpExtension::kMidUri; }
+};
+
+class InbandComfortNoiseExtension {
+ public:
+  using value_type = absl::optional<uint8_t>;
+
+  static constexpr RTPExtensionType kId = kRtpExtensionInbandComfortNoise;
+  static constexpr uint8_t kValueSizeBytes = 1;
+  static constexpr const char kUri[] =
+      "http://www.webrtc.org/experiments/rtp-hdrext/inband-cn";
+  static constexpr absl::string_view Uri() { return kUri; }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    absl::optional<uint8_t>* level);
+  static size_t ValueSize(absl::optional<uint8_t> level) {
+    return kValueSizeBytes;
+  }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    absl::optional<uint8_t> level);
+};
+
+class VideoFrameTrackingIdExtension {
+ public:
+  using value_type = uint16_t;
+  static constexpr RTPExtensionType kId = kRtpExtensionVideoFrameTrackingId;
+  static constexpr uint8_t kValueSizeBytes = 2;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoFrameTrackingIdUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    uint16_t* video_frame_tracking_id);
+  static size_t ValueSize(uint16_t /*video_frame_tracking_id*/) {
+    return kValueSizeBytes;
+  }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    uint16_t video_frame_tracking_id);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.cc
new file mode 100644
index 0000000000..fd2f5c5ae8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.cc
@@ -0,0 +1,712 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+
+#include <cstdint>
+#include <cstring>
+#include <utility>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace {
+constexpr size_t kFixedHeaderSize = 12;
+constexpr uint8_t kRtpVersion = 2;
+constexpr uint16_t kOneByteExtensionProfileId = 0xBEDE;
+constexpr uint16_t kTwoByteExtensionProfileId = 0x1000;
+constexpr uint16_t kTwobyteExtensionProfileIdAppBitsFilter = 0xfff0;
+constexpr size_t kOneByteExtensionHeaderLength = 1;
+constexpr size_t kTwoByteExtensionHeaderLength = 2;
+constexpr size_t kDefaultPacketSize = 1500;
+}  // namespace
+
+//  0                   1                   2                   3
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |V=2|P|X|  CC   |M|     PT      |       sequence number         |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |                           timestamp                           |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |           synchronization source (SSRC) identifier            |
+// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// |            Contributing source (CSRC) identifiers             |
+// |                             ....                              |
+// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// |  header eXtension profile id  |       length in 32bits        |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |                          Extensions                           |
+// |                             ....                              |
+// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+// |                           Payload                             |
+// |             ....              :  padding...                   |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |               padding         | Padding size  |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+RtpPacket::RtpPacket() : RtpPacket(nullptr, kDefaultPacketSize) {}
+
+RtpPacket::RtpPacket(const ExtensionManager* extensions)
+    : RtpPacket(extensions, kDefaultPacketSize) {}
+
+RtpPacket::RtpPacket(const ExtensionManager* extensions, size_t capacity)
+    : extensions_(extensions ? *extensions : ExtensionManager()),
+      buffer_(capacity) {
+  RTC_DCHECK_GE(capacity, kFixedHeaderSize);
+  Clear();
+}
+
+RtpPacket::RtpPacket(const RtpPacket&) = default;
+RtpPacket::RtpPacket(RtpPacket&&) = default;
+RtpPacket& RtpPacket::operator=(const RtpPacket&) = default;
+RtpPacket& RtpPacket::operator=(RtpPacket&&) = default;
+RtpPacket::~RtpPacket() = default;
+
+void RtpPacket::IdentifyExtensions(ExtensionManager extensions) {
+  extensions_ = std::move(extensions);
+}
+
+bool RtpPacket::Parse(const uint8_t* buffer, size_t buffer_size) {
+  if (!ParseBuffer(buffer, buffer_size)) {
+    Clear();
+    return false;
+  }
+  buffer_.SetData(buffer, buffer_size);
+  RTC_DCHECK_EQ(size(), buffer_size);
+  return true;
+}
+
+bool RtpPacket::Parse(rtc::ArrayView<const uint8_t> packet) {
+  return Parse(packet.data(), packet.size());
+}
+
+bool RtpPacket::Parse(rtc::CopyOnWriteBuffer buffer) {
+  if (!ParseBuffer(buffer.cdata(), buffer.size())) {
+    Clear();
+    return false;
+  }
+  size_t buffer_size = buffer.size();
+  buffer_ = std::move(buffer);
+  RTC_DCHECK_EQ(size(), buffer_size);
+  return true;
+}
+
+std::vector<uint32_t> RtpPacket::Csrcs() const {
+  size_t num_csrc = data()[0] & 0x0F;
+  RTC_DCHECK_GE(capacity(), kFixedHeaderSize + num_csrc * 4);
+  std::vector<uint32_t> csrcs(num_csrc);
+  for (size_t i = 0; i < num_csrc; ++i) {
+    csrcs[i] =
+        ByteReader<uint32_t>::ReadBigEndian(&data()[kFixedHeaderSize + i * 4]);
+  }
+  return csrcs;
+}
+
+void RtpPacket::CopyHeaderFrom(const RtpPacket& packet) {
+  marker_ = packet.marker_;
+  payload_type_ = packet.payload_type_;
+  sequence_number_ = packet.sequence_number_;
+  timestamp_ = packet.timestamp_;
+  ssrc_ = packet.ssrc_;
+  payload_offset_ = packet.payload_offset_;
+  extensions_ = packet.extensions_;
+  extension_entries_ = packet.extension_entries_;
+  extensions_size_ = packet.extensions_size_;
+  buffer_ = packet.buffer_.Slice(0, packet.headers_size());
+  // Reset payload and padding.
+  payload_size_ = 0;
+  padding_size_ = 0;
+}
+
+void RtpPacket::SetMarker(bool marker_bit) {
+  marker_ = marker_bit;
+  if (marker_) {
+    WriteAt(1, data()[1] | 0x80);
+  } else {
+    WriteAt(1, data()[1] & 0x7F);
+  }
+}
+
+void RtpPacket::SetPayloadType(uint8_t payload_type) {
+  RTC_DCHECK_LE(payload_type, 0x7Fu);
+  payload_type_ = payload_type;
+  WriteAt(1, (data()[1] & 0x80) | payload_type);
+}
+
+void RtpPacket::SetSequenceNumber(uint16_t seq_no) {
+  sequence_number_ = seq_no;
+  ByteWriter<uint16_t>::WriteBigEndian(WriteAt(2), seq_no);
+}
+
+void RtpPacket::SetTimestamp(uint32_t timestamp) {
+  timestamp_ = timestamp;
+  ByteWriter<uint32_t>::WriteBigEndian(WriteAt(4), timestamp);
+}
+
+void RtpPacket::SetSsrc(uint32_t ssrc) {
+  ssrc_ = ssrc;
+  ByteWriter<uint32_t>::WriteBigEndian(WriteAt(8), ssrc);
+}
+
+void RtpPacket::ZeroMutableExtensions() {
+  for (const ExtensionInfo& extension : extension_entries_) {
+    switch (extensions_.GetType(extension.id)) {
+      case RTPExtensionType::kRtpExtensionNone: {
+        RTC_LOG(LS_WARNING) << "Unidentified extension in the packet.";
+        break;
+      }
+      case RTPExtensionType::kRtpExtensionVideoTiming: {
+        // Nullify last entries, starting at pacer delay.
+        // These are set by pacer and SFUs
+        if (VideoTimingExtension::kPacerExitDeltaOffset < extension.length) {
+          memset(
+              WriteAt(extension.offset +
+                      VideoTimingExtension::kPacerExitDeltaOffset),
+              0,
+              extension.length - VideoTimingExtension::kPacerExitDeltaOffset);
+        }
+        break;
+      }
+      case RTPExtensionType::kRtpExtensionTransportSequenceNumber:
+      case RTPExtensionType::kRtpExtensionTransportSequenceNumber02:
+      case RTPExtensionType::kRtpExtensionTransmissionTimeOffset:
+      case RTPExtensionType::kRtpExtensionAbsoluteSendTime: {
+        // Nullify whole extension, as it's filled in the pacer.
+        memset(WriteAt(extension.offset), 0, extension.length);
+        break;
+      }
+      case RTPExtensionType::kRtpExtensionAudioLevel:
+#if !defined(WEBRTC_MOZILLA_BUILD)
+      case RTPExtensionType::kRtpExtensionCsrcAudioLevel:
+#endif
+      case RTPExtensionType::kRtpExtensionAbsoluteCaptureTime:
+      case RTPExtensionType::kRtpExtensionColorSpace:
+      case RTPExtensionType::kRtpExtensionGenericFrameDescriptor:
+      case RTPExtensionType::kRtpExtensionDependencyDescriptor:
+      case RTPExtensionType::kRtpExtensionMid:
+      case RTPExtensionType::kRtpExtensionNumberOfExtensions:
+      case RTPExtensionType::kRtpExtensionPlayoutDelay:
+      case RTPExtensionType::kRtpExtensionRepairedRtpStreamId:
+      case RTPExtensionType::kRtpExtensionRtpStreamId:
+      case RTPExtensionType::kRtpExtensionVideoContentType:
+      case RTPExtensionType::kRtpExtensionVideoLayersAllocation:
+      case RTPExtensionType::kRtpExtensionVideoRotation:
+      case RTPExtensionType::kRtpExtensionInbandComfortNoise:
+      case RTPExtensionType::kRtpExtensionVideoFrameTrackingId: {
+        // Non-mutable extension. Don't change it.
+        break;
+      }
+#if defined(WEBRTC_MOZILLA_BUILD)
+      case RTPExtensionType::kRtpExtensionCsrcAudioLevel: {
+        // TODO: This is a Mozilla addition, we need to add a handler for this.
+        RTC_CHECK(false);
+      }
+#endif
+    }
+  }
+}
+
+void RtpPacket::SetCsrcs(rtc::ArrayView<const uint32_t> csrcs) {
+  RTC_DCHECK_EQ(extensions_size_, 0);
+  RTC_DCHECK_EQ(payload_size_, 0);
+  RTC_DCHECK_EQ(padding_size_, 0);
+  RTC_DCHECK_LE(csrcs.size(), 0x0fu);
+  RTC_DCHECK_LE(kFixedHeaderSize + 4 * csrcs.size(), capacity());
+  payload_offset_ = kFixedHeaderSize + 4 * csrcs.size();
+  WriteAt(0, (data()[0] & 0xF0) | rtc::dchecked_cast<uint8_t>(csrcs.size()));
+  size_t offset = kFixedHeaderSize;
+  for (uint32_t csrc : csrcs) {
+    ByteWriter<uint32_t>::WriteBigEndian(WriteAt(offset), csrc);
+    offset += 4;
+  }
+  buffer_.SetSize(payload_offset_);
+}
+
+rtc::ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
+  RTC_DCHECK_GE(id, RtpExtension::kMinId);
+  RTC_DCHECK_LE(id, RtpExtension::kMaxId);
+  RTC_DCHECK_GE(length, 1);
+  RTC_DCHECK_LE(length, RtpExtension::kMaxValueSize);
+  const ExtensionInfo* extension_entry = FindExtensionInfo(id);
+  if (extension_entry != nullptr) {
+    // Extension already reserved. Check if same length is used.
+    if (extension_entry->length == length)
+      return rtc::MakeArrayView(WriteAt(extension_entry->offset), length);
+
+    RTC_LOG(LS_ERROR) << "Length mismatch for extension id " << id
+                      << ": expected "
+                      << static_cast<int>(extension_entry->length)
+                      << ". received " << length;
+    return nullptr;
+  }
+  if (payload_size_ > 0) {
+    RTC_LOG(LS_ERROR) << "Can't add new extension id " << id
+                      << " after payload was set.";
+    return nullptr;
+  }
+  if (padding_size_ > 0) {
+    RTC_LOG(LS_ERROR) << "Can't add new extension id " << id
+                      << " after padding was set.";
+    return nullptr;
+  }
+
+  const size_t num_csrc = data()[0] & 0x0F;
+  const size_t extensions_offset = kFixedHeaderSize + (num_csrc * 4) + 4;
+  // Determine if two-byte header is required for the extension based on id and
+  // length. Please note that a length of 0 also requires two-byte header
+  // extension. See RFC8285 Section 4.2-4.3.
+  const bool two_byte_header_required =
+      id > RtpExtension::kOneByteHeaderExtensionMaxId ||
+      length > RtpExtension::kOneByteHeaderExtensionMaxValueSize || length == 0;
+  RTC_CHECK(!two_byte_header_required || extensions_.ExtmapAllowMixed());
+
+  uint16_t profile_id;
+  if (extensions_size_ > 0) {
+    profile_id =
+        ByteReader<uint16_t>::ReadBigEndian(data() + extensions_offset - 4);
+    if (profile_id == kOneByteExtensionProfileId && two_byte_header_required) {
+      // Is buffer size big enough to fit promotion and new data field?
+      // The header extension will grow with one byte per already allocated
+      // extension + the size of the extension that is about to be allocated.
+      size_t expected_new_extensions_size =
+          extensions_size_ + extension_entries_.size() +
+          kTwoByteExtensionHeaderLength + length;
+      if (extensions_offset + expected_new_extensions_size > capacity()) {
+        RTC_LOG(LS_ERROR)
+            << "Extension cannot be registered: Not enough space left in "
+               "buffer to change to two-byte header extension and add new "
+               "extension.";
+        return nullptr;
+      }
+      // Promote already written data to two-byte header format.
+      PromoteToTwoByteHeaderExtension();
+      profile_id = kTwoByteExtensionProfileId;
+    }
+  } else {
+    // Profile specific ID, set to OneByteExtensionHeader unless
+    // TwoByteExtensionHeader is required.
+    profile_id = two_byte_header_required ? kTwoByteExtensionProfileId
+                                          : kOneByteExtensionProfileId;
+  }
+
+  const size_t extension_header_size = profile_id == kOneByteExtensionProfileId
+                                           ? kOneByteExtensionHeaderLength
+                                           : kTwoByteExtensionHeaderLength;
+  size_t new_extensions_size =
+      extensions_size_ + extension_header_size + length;
+  if (extensions_offset + new_extensions_size > capacity()) {
+    RTC_LOG(LS_ERROR)
+        << "Extension cannot be registered: Not enough space left in buffer.";
+    return nullptr;
+  }
+
+  // All checks passed, write down the extension headers.
+  if (extensions_size_ == 0) {
+    RTC_DCHECK_EQ(payload_offset_, kFixedHeaderSize + (num_csrc * 4));
+    WriteAt(0, data()[0] | 0x10);  // Set extension bit.
+    ByteWriter<uint16_t>::WriteBigEndian(WriteAt(extensions_offset - 4),
+                                         profile_id);
+  }
+
+  if (profile_id == kOneByteExtensionProfileId) {
+    uint8_t one_byte_header = rtc::dchecked_cast<uint8_t>(id) << 4;
+    one_byte_header |= rtc::dchecked_cast<uint8_t>(length - 1);
+    WriteAt(extensions_offset + extensions_size_, one_byte_header);
+  } else {
+    // TwoByteHeaderExtension.
+    uint8_t extension_id = rtc::dchecked_cast<uint8_t>(id);
+    WriteAt(extensions_offset + extensions_size_, extension_id);
+    uint8_t extension_length = rtc::dchecked_cast<uint8_t>(length);
+    WriteAt(extensions_offset + extensions_size_ + 1, extension_length);
+  }
+
+  const uint16_t extension_info_offset = rtc::dchecked_cast<uint16_t>(
+      extensions_offset + extensions_size_ + extension_header_size);
+  const uint8_t extension_info_length = rtc::dchecked_cast<uint8_t>(length);
+  extension_entries_.emplace_back(id, extension_info_length,
+                                  extension_info_offset);
+
+  extensions_size_ = new_extensions_size;
+
+  uint16_t extensions_size_padded =
+      SetExtensionLengthMaybeAddZeroPadding(extensions_offset);
+  payload_offset_ = extensions_offset + extensions_size_padded;
+  buffer_.SetSize(payload_offset_);
+  return rtc::MakeArrayView(WriteAt(extension_info_offset),
+                            extension_info_length);
+}
+
+void RtpPacket::PromoteToTwoByteHeaderExtension() {
+  size_t num_csrc = data()[0] & 0x0F;
+  size_t extensions_offset = kFixedHeaderSize + (num_csrc * 4) + 4;
+
+  RTC_CHECK_GT(extension_entries_.size(), 0);
+  RTC_CHECK_EQ(payload_size_, 0);
+  RTC_CHECK_EQ(kOneByteExtensionProfileId, ByteReader<uint16_t>::ReadBigEndian(
+                                               data() + extensions_offset - 4));
+  // Rewrite data.
+  // Each extension adds one to the offset. The write-read delta for the last
+  // extension is therefore the same as the number of extension entries.
+  size_t write_read_delta = extension_entries_.size();
+  for (auto extension_entry = extension_entries_.rbegin();
+       extension_entry != extension_entries_.rend(); ++extension_entry) {
+    size_t read_index = extension_entry->offset;
+    size_t write_index = read_index + write_read_delta;
+    // Update offset.
+    extension_entry->offset = rtc::dchecked_cast<uint16_t>(write_index);
+    // Copy data. Use memmove since read/write regions may overlap.
+    memmove(WriteAt(write_index), data() + read_index, extension_entry->length);
+    // Rewrite id and length.
+    WriteAt(--write_index, extension_entry->length);
+    WriteAt(--write_index, extension_entry->id);
+    --write_read_delta;
+  }
+
+  // Update profile header, extensions length, and zero padding.
+  ByteWriter<uint16_t>::WriteBigEndian(WriteAt(extensions_offset - 4),
+                                       kTwoByteExtensionProfileId);
+  extensions_size_ += extension_entries_.size();
+  uint16_t extensions_size_padded =
+      SetExtensionLengthMaybeAddZeroPadding(extensions_offset);
+  payload_offset_ = extensions_offset + extensions_size_padded;
+  buffer_.SetSize(payload_offset_);
+}
+
+uint16_t RtpPacket::SetExtensionLengthMaybeAddZeroPadding(
+    size_t extensions_offset) {
+  // Update header length field.
+  uint16_t extensions_words = rtc::dchecked_cast<uint16_t>(
+      (extensions_size_ + 3) / 4);  // Wrap up to 32bit.
+  ByteWriter<uint16_t>::WriteBigEndian(WriteAt(extensions_offset - 2),
+                                       extensions_words);
+  // Fill extension padding place with zeroes.
+  size_t extension_padding_size = 4 * extensions_words - extensions_size_;
+  memset(WriteAt(extensions_offset + extensions_size_), 0,
+         extension_padding_size);
+  return 4 * extensions_words;
+}
+
+uint8_t* RtpPacket::AllocatePayload(size_t size_bytes) {
+  // Reset payload size to 0. If CopyOnWrite buffer_ was shared, this will cause
+  // reallocation and memcpy. Keeping just header reduces memcpy size.
+  SetPayloadSize(0);
+  return SetPayloadSize(size_bytes);
+}
+
+uint8_t* RtpPacket::SetPayloadSize(size_t size_bytes) {
+  RTC_DCHECK_EQ(padding_size_, 0);
+  if (payload_offset_ + size_bytes > capacity()) {
+    RTC_LOG(LS_WARNING) << "Cannot set payload, not enough space in buffer.";
+    return nullptr;
+  }
+  payload_size_ = size_bytes;
+  buffer_.SetSize(payload_offset_ + payload_size_);
+  return WriteAt(payload_offset_);
+}
+
+bool RtpPacket::SetPadding(size_t padding_bytes) {
+  if (payload_offset_ + payload_size_ + padding_bytes > capacity()) {
+    RTC_LOG(LS_WARNING) << "Cannot set padding size " << padding_bytes
+                        << ", only "
+                        << (capacity() - payload_offset_ - payload_size_)
+                        << " bytes left in buffer.";
+    return false;
+  }
+  padding_size_ = rtc::dchecked_cast<uint8_t>(padding_bytes);
+  buffer_.SetSize(payload_offset_ + payload_size_ + padding_size_);
+  if (padding_size_ > 0) {
+    size_t padding_offset = payload_offset_ + payload_size_;
+    size_t padding_end = padding_offset + padding_size_;
+    memset(WriteAt(padding_offset), 0, padding_size_ - 1);
+    WriteAt(padding_end - 1, padding_size_);
+    WriteAt(0, data()[0] | 0x20);  // Set padding bit.
+  } else {
+    WriteAt(0, data()[0] & ~0x20);  // Clear padding bit.
+  }
+  return true;
+}
+
+void RtpPacket::Clear() {
+  marker_ = false;
+  payload_type_ = 0;
+  sequence_number_ = 0;
+  timestamp_ = 0;
+  ssrc_ = 0;
+  payload_offset_ = kFixedHeaderSize;
+  payload_size_ = 0;
+  padding_size_ = 0;
+  extensions_size_ = 0;
+  extension_entries_.clear();
+
+  memset(WriteAt(0), 0, kFixedHeaderSize);
+  buffer_.SetSize(kFixedHeaderSize);
+  WriteAt(0, kRtpVersion << 6);
+}
+
+bool RtpPacket::ParseBuffer(const uint8_t* buffer, size_t size) {
+  if (size < kFixedHeaderSize) {
+    return false;
+  }
+  const uint8_t version = buffer[0] >> 6;
+  if (version != kRtpVersion) {
+    return false;
+  }
+  const bool has_padding = (buffer[0] & 0x20) != 0;
+  const bool has_extension = (buffer[0] & 0x10) != 0;
+  const uint8_t number_of_crcs = buffer[0] & 0x0f;
+  marker_ = (buffer[1] & 0x80) != 0;
+  payload_type_ = buffer[1] & 0x7f;
+
+  sequence_number_ = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]);
+  timestamp_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
+  ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
+  if (size < kFixedHeaderSize + number_of_crcs * 4) {
+    return false;
+  }
+  payload_offset_ = kFixedHeaderSize + number_of_crcs * 4;
+
+  extensions_size_ = 0;
+  extension_entries_.clear();
+  if (has_extension) {
+    /* RTP header extension, RFC 3550.
+     0                   1                   2                   3
+     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    |      defined by profile       |           length              |
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    |                        header extension                       |
+    |                             ....                              |
+    */
+    size_t extension_offset = payload_offset_ + 4;
+    if (extension_offset > size) {
+      return false;
+    }
+    uint16_t profile =
+        ByteReader<uint16_t>::ReadBigEndian(&buffer[payload_offset_]);
+    size_t extensions_capacity =
+        ByteReader<uint16_t>::ReadBigEndian(&buffer[payload_offset_ + 2]);
+    extensions_capacity *= 4;
+    if (extension_offset + extensions_capacity > size) {
+      return false;
+    }
+    if (profile != kOneByteExtensionProfileId &&
+        (profile & kTwobyteExtensionProfileIdAppBitsFilter) !=
+            kTwoByteExtensionProfileId) {
+      RTC_LOG(LS_WARNING) << "Unsupported rtp extension " << profile;
+    } else {
+      size_t extension_header_length = profile == kOneByteExtensionProfileId
+                                           ? kOneByteExtensionHeaderLength
+                                           : kTwoByteExtensionHeaderLength;
+      constexpr uint8_t kPaddingByte = 0;
+      constexpr uint8_t kPaddingId = 0;
+      constexpr uint8_t kOneByteHeaderExtensionReservedId = 15;
+      while (extensions_size_ + extension_header_length < extensions_capacity) {
+        if (buffer[extension_offset + extensions_size_] == kPaddingByte) {
+          extensions_size_++;
+          continue;
+        }
+        int id;
+        uint8_t length;
+        if (profile == kOneByteExtensionProfileId) {
+          id = buffer[extension_offset + extensions_size_] >> 4;
+          length = 1 + (buffer[extension_offset + extensions_size_] & 0xf);
+          if (id == kOneByteHeaderExtensionReservedId ||
+              (id == kPaddingId && length != 1)) {
+            break;
+          }
+        } else {
+          id = buffer[extension_offset + extensions_size_];
+          length = buffer[extension_offset + extensions_size_ + 1];
+        }
+
+        if (extensions_size_ + extension_header_length + length >
+            extensions_capacity) {
+          RTC_LOG(LS_WARNING) << "Oversized rtp header extension.";
+          break;
+        }
+
+        ExtensionInfo& extension_info = FindOrCreateExtensionInfo(id);
+        if (extension_info.length != 0) {
+          RTC_LOG(LS_VERBOSE)
+              << "Duplicate rtp header extension id " << id << ". Overwriting.";
+        }
+
+        size_t offset =
+            extension_offset + extensions_size_ + extension_header_length;
+        if (!rtc::IsValueInRangeForNumericType<uint16_t>(offset)) {
+          RTC_DLOG(LS_WARNING) << "Oversized rtp header extension.";
+          break;
+        }
+        extension_info.offset = static_cast<uint16_t>(offset);
+        extension_info.length = length;
+        extensions_size_ += extension_header_length + length;
+      }
+    }
+    payload_offset_ = extension_offset + extensions_capacity;
+  }
+
+  if (has_padding && payload_offset_ < size) {
+    padding_size_ = buffer[size - 1];
+    if (padding_size_ == 0) {
+      RTC_LOG(LS_WARNING) << "Padding was set, but padding size is zero";
+      return false;
+    }
+  } else {
+    padding_size_ = 0;
+  }
+
+  if (payload_offset_ + padding_size_ > size) {
+    return false;
+  }
+  payload_size_ = size - payload_offset_ - padding_size_;
+  return true;
+}
+
+const RtpPacket::ExtensionInfo* RtpPacket::FindExtensionInfo(int id) const {
+  for (const ExtensionInfo& extension : extension_entries_) {
+    if (extension.id == id) {
+      return &extension;
+    }
+  }
+  return nullptr;
+}
+
+RtpPacket::ExtensionInfo& RtpPacket::FindOrCreateExtensionInfo(int id) {
+  for (ExtensionInfo& extension : extension_entries_) {
+    if (extension.id == id) {
+      return extension;
+    }
+  }
+  extension_entries_.emplace_back(id);
+  return extension_entries_.back();
+}
+
+rtc::ArrayView<const uint8_t> RtpPacket::FindExtension(
+    ExtensionType type) const {
+  uint8_t id = extensions_.GetId(type);
+  if (id == ExtensionManager::kInvalidId) {
+    // Extension not registered.
+    return nullptr;
+  }
+  ExtensionInfo const* extension_info = FindExtensionInfo(id);
+  if (extension_info == nullptr) {
+    return nullptr;
+  }
+  return rtc::MakeArrayView(data() + extension_info->offset,
+                            extension_info->length);
+}
+
+rtc::ArrayView<uint8_t> RtpPacket::AllocateExtension(ExtensionType type,
+                                                     size_t length) {
+  // TODO(webrtc:7990): Add support for empty extensions (length==0).
+  if (length == 0 || length > RtpExtension::kMaxValueSize ||
+      (!extensions_.ExtmapAllowMixed() &&
+       length > RtpExtension::kOneByteHeaderExtensionMaxValueSize)) {
+    return nullptr;
+  }
+
+  uint8_t id = extensions_.GetId(type);
+  if (id == ExtensionManager::kInvalidId) {
+    // Extension not registered.
+    return nullptr;
+  }
+  if (!extensions_.ExtmapAllowMixed() &&
+      id > RtpExtension::kOneByteHeaderExtensionMaxId) {
+    return nullptr;
+  }
+  return AllocateRawExtension(id, length);
+}
+
+bool RtpPacket::HasExtension(ExtensionType type) const {
+  uint8_t id = extensions_.GetId(type);
+  if (id == ExtensionManager::kInvalidId) {
+    // Extension not registered.
+    return false;
+  }
+  return FindExtensionInfo(id) != nullptr;
+}
+
+bool RtpPacket::RemoveExtension(ExtensionType type) {
+  uint8_t id_to_remove = extensions_.GetId(type);
+  if (id_to_remove == ExtensionManager::kInvalidId) {
+    // Extension not registered.
+    RTC_LOG(LS_ERROR) << "Extension not registered, type=" << type
+                      << ", packet=" << ToString();
+    return false;
+  }
+
+  // Rebuild new packet from scratch.
+  RtpPacket new_packet;
+
+  new_packet.SetMarker(Marker());
+  new_packet.SetPayloadType(PayloadType());
+  new_packet.SetSequenceNumber(SequenceNumber());
+  new_packet.SetTimestamp(Timestamp());
+  new_packet.SetSsrc(Ssrc());
+  new_packet.IdentifyExtensions(extensions_);
+
+  // Copy all extensions, except the one we are removing.
+  bool found_extension = false;
+  for (const ExtensionInfo& ext : extension_entries_) {
+    if (ext.id == id_to_remove) {
+      found_extension = true;
+    } else {
+      auto extension_data = new_packet.AllocateRawExtension(ext.id, ext.length);
+      if (extension_data.size() != ext.length) {
+        RTC_LOG(LS_ERROR) << "Failed to allocate extension id=" << ext.id
+                          << ", length=" << ext.length
+                          << ", packet=" << ToString();
+        return false;
+      }
+
+      // Copy extension data to new packet.
+      memcpy(extension_data.data(), ReadAt(ext.offset), ext.length);
+    }
+  }
+
+  if (!found_extension) {
+    RTC_LOG(LS_WARNING) << "Extension not present in RTP packet, type=" << type
+                        << ", packet=" << ToString();
+    return false;
+  }
+
+  // Copy payload data to new packet.
+  if (payload_size() > 0) {
+    memcpy(new_packet.AllocatePayload(payload_size()), payload().data(),
+           payload_size());
+  } else {
+    new_packet.SetPayloadSize(0);
+  }
+
+  // Allocate padding -- must be last!
+  new_packet.SetPadding(padding_size());
+
+  // Success, replace current packet with newly built packet.
+  *this = new_packet;
+  return true;
+}
+
+std::string RtpPacket::ToString() const {
+  rtc::StringBuilder result;
+  result << "{payload_type=" << payload_type_ << ", marker=" << marker_
+         << ", sequence_number=" << sequence_number_
+         << ", padding_size=" << padding_size_ << ", timestamp=" << timestamp_
+         << ", ssrc=" << ssrc_ << ", payload_offset=" << payload_offset_
+         << ", payload_size=" << payload_size_ << ", total_size=" << size()
+         << "}";
+
+  return result.Release();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h
new file mode 100644
index 0000000000..1db4a13017
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h
@@ -0,0 +1,283 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
+
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class RtpPacket {
+ public:
+  using ExtensionType = RTPExtensionType;
+  using ExtensionManager = RtpHeaderExtensionMap;
+
+  // `extensions` required for SetExtension/ReserveExtension functions during
+  // packet creating and used if available in Parse function.
+  // Adding and getting extensions will fail until `extensions` is
+  // provided via constructor or IdentifyExtensions function.
+  // |*extensions| is only accessed during construction; the pointer is not
+  // stored.
+  RtpPacket();
+  explicit RtpPacket(const ExtensionManager* extensions);
+  RtpPacket(const ExtensionManager* extensions, size_t capacity);
+
+  RtpPacket(const RtpPacket&);
+  RtpPacket(RtpPacket&&);
+  RtpPacket& operator=(const RtpPacket&);
+  RtpPacket& operator=(RtpPacket&&);
+
+  ~RtpPacket();
+
+  // Parse and copy given buffer into Packet.
+  // Does not require extension map to be registered (map is only required to
+  // read or allocate extensions in methods GetExtension, AllocateExtension,
+  // etc.)
+  bool Parse(const uint8_t* buffer, size_t size);
+  bool Parse(rtc::ArrayView<const uint8_t> packet);
+
+  // Parse and move given buffer into Packet.
+  bool Parse(rtc::CopyOnWriteBuffer packet);
+
+  // Maps extensions id to their types.
+  void IdentifyExtensions(ExtensionManager extensions);
+
+  // Returns the extension map used for identifying extensions in this packet.
+  const ExtensionManager& extension_manager() const { return extensions_; }
+
+  // Header.
+  bool Marker() const { return marker_; }
+  uint8_t PayloadType() const { return payload_type_; }
+  uint16_t SequenceNumber() const { return sequence_number_; }
+  uint32_t Timestamp() const { return timestamp_; }
+  uint32_t Ssrc() const { return ssrc_; }
+  std::vector<uint32_t> Csrcs() const;
+
+  size_t headers_size() const { return payload_offset_; }
+
+  // Payload.
+  size_t payload_size() const { return payload_size_; }
+  bool has_padding() const { return buffer_[0] & 0x20; }
+  size_t padding_size() const { return padding_size_; }
+  rtc::ArrayView<const uint8_t> payload() const {
+    return rtc::MakeArrayView(data() + payload_offset_, payload_size_);
+  }
+  rtc::CopyOnWriteBuffer PayloadBuffer() const {
+    return buffer_.Slice(payload_offset_, payload_size_);
+  }
+
+  // Buffer.
+  rtc::CopyOnWriteBuffer Buffer() const { return buffer_; }
+  size_t capacity() const { return buffer_.capacity(); }
+  size_t size() const {
+    return payload_offset_ + payload_size_ + padding_size_;
+  }
+  const uint8_t* data() const { return buffer_.cdata(); }
+  size_t FreeCapacity() const { return capacity() - size(); }
+  size_t MaxPayloadSize() const { return capacity() - headers_size(); }
+
+  // Reset fields and buffer.
+  void Clear();
+
+  // Header setters.
+  void CopyHeaderFrom(const RtpPacket& packet);
+  void SetMarker(bool marker_bit);
+  void SetPayloadType(uint8_t payload_type);
+  void SetSequenceNumber(uint16_t seq_no);
+  void SetTimestamp(uint32_t timestamp);
+  void SetSsrc(uint32_t ssrc);
+
+  // Fills with zeroes mutable extensions,
+  // which are modified after FEC protection is generated.
+  void ZeroMutableExtensions();
+
+  // Removes extension of given `type`, returns false is extension was not
+  // registered in packet's extension map or not present in the packet. Only
+  // extension that should be removed must be registered, other extensions may
+  // not be registered and will be preserved as is.
+  bool RemoveExtension(ExtensionType type);
+
+  // Writes csrc list. Assumes:
+  // a) There is enough room left in buffer.
+  // b) Extension headers, payload or padding data has not already been added.
+  void SetCsrcs(rtc::ArrayView<const uint32_t> csrcs);
+
+  // Header extensions.
+  template <typename Extension>
+  bool HasExtension() const;
+  bool HasExtension(ExtensionType type) const;
+
+  // Returns whether there is an associated id for the extension and thus it is
+  // possible to set the extension.
+  template <typename Extension>
+  bool IsRegistered() const;
+
+  template <typename Extension, typename FirstValue, typename... Values>
+  bool GetExtension(FirstValue, Values...) const;
+
+  template <typename Extension>
+  absl::optional<typename Extension::value_type> GetExtension() const;
+
+  // Returns view of the raw extension or empty view on failure.
+  template <typename Extension>
+  rtc::ArrayView<const uint8_t> GetRawExtension() const;
+
+  template <typename Extension, typename... Values>
+  bool SetExtension(const Values&...);
+
+  template <typename Extension>
+  bool SetRawExtension(rtc::ArrayView<const uint8_t> data);
+
+  template <typename Extension>
+  bool ReserveExtension();
+
+  // Find or allocate an extension `type`. Returns view of size `length`
+  // to write raw extension to or an empty view on failure.
+  rtc::ArrayView<uint8_t> AllocateExtension(ExtensionType type, size_t length);
+
+  // Find an extension `type`.
+  // Returns view of the raw extension or empty view on failure.
+  rtc::ArrayView<const uint8_t> FindExtension(ExtensionType type) const;
+
+  // Reserve size_bytes for payload. Returns nullptr on failure.
+  uint8_t* SetPayloadSize(size_t size_bytes);
+  // Same as SetPayloadSize but doesn't guarantee to keep current payload.
+  uint8_t* AllocatePayload(size_t size_bytes);
+
+  bool SetPadding(size_t padding_size);
+
+  // Returns debug string of RTP packet (without detailed extension info).
+  std::string ToString() const;
+
+ private:
+  struct ExtensionInfo {
+    explicit ExtensionInfo(uint8_t id) : ExtensionInfo(id, 0, 0) {}
+    ExtensionInfo(uint8_t id, uint8_t length, uint16_t offset)
+        : id(id), length(length), offset(offset) {}
+    uint8_t id;
+    uint8_t length;
+    uint16_t offset;
+  };
+
+  // Helper function for Parse. Fill header fields using data in given buffer,
+  // but does not touch packet own buffer, leaving packet in invalid state.
+  bool ParseBuffer(const uint8_t* buffer, size_t size);
+
+  // Returns pointer to extension info for a given id. Returns nullptr if not
+  // found.
+  const ExtensionInfo* FindExtensionInfo(int id) const;
+
+  // Returns reference to extension info for a given id. Creates a new entry
+  // with the specified id if not found.
+  ExtensionInfo& FindOrCreateExtensionInfo(int id);
+
+  // Allocates and returns place to store rtp header extension.
+  // Returns empty arrayview on failure.
+  rtc::ArrayView<uint8_t> AllocateRawExtension(int id, size_t length);
+
+  // Promotes existing one-byte header extensions to two-byte header extensions
+  // by rewriting the data and updates the corresponding extension offsets.
+  void PromoteToTwoByteHeaderExtension();
+
+  uint16_t SetExtensionLengthMaybeAddZeroPadding(size_t extensions_offset);
+
+  uint8_t* WriteAt(size_t offset) { return buffer_.MutableData() + offset; }
+  void WriteAt(size_t offset, uint8_t byte) {
+    buffer_.MutableData()[offset] = byte;
+  }
+  const uint8_t* ReadAt(size_t offset) const { return buffer_.data() + offset; }
+
+  // Header.
+  bool marker_;
+  uint8_t payload_type_;
+  uint8_t padding_size_;
+  uint16_t sequence_number_;
+  uint32_t timestamp_;
+  uint32_t ssrc_;
+  size_t payload_offset_;  // Match header size with csrcs and extensions.
+  size_t payload_size_;
+
+  ExtensionManager extensions_;
+  std::vector<ExtensionInfo> extension_entries_;
+  size_t extensions_size_ = 0;  // Unaligned.
+  rtc::CopyOnWriteBuffer buffer_;
+};
+
+template <typename Extension>
+bool RtpPacket::HasExtension() const {
+  return HasExtension(Extension::kId);
+}
+
+template <typename Extension>
+bool RtpPacket::IsRegistered() const {
+  return extensions_.IsRegistered(Extension::kId);
+}
+
+template <typename Extension, typename FirstValue, typename... Values>
+bool RtpPacket::GetExtension(FirstValue first, Values... values) const {
+  auto raw = FindExtension(Extension::kId);
+  if (raw.empty())
+    return false;
+  return Extension::Parse(raw, first, values...);
+}
+
+template <typename Extension>
+absl::optional<typename Extension::value_type> RtpPacket::GetExtension() const {
+  absl::optional<typename Extension::value_type> result;
+  auto raw = FindExtension(Extension::kId);
+  if (raw.empty() || !Extension::Parse(raw, &result.emplace()))
+    result = absl::nullopt;
+  return result;
+}
+
+template <typename Extension>
+rtc::ArrayView<const uint8_t> RtpPacket::GetRawExtension() const {
+  return FindExtension(Extension::kId);
+}
+
+template <typename Extension, typename... Values>
+bool RtpPacket::SetExtension(const Values&... values) {
+  const size_t value_size = Extension::ValueSize(values...);
+  auto buffer = AllocateExtension(Extension::kId, value_size);
+  if (buffer.empty())
+    return false;
+  return Extension::Write(buffer, values...);
+}
+
+template <typename Extension>
+bool RtpPacket::SetRawExtension(rtc::ArrayView<const uint8_t> data) {
+  rtc::ArrayView<uint8_t> buffer =
+      AllocateExtension(Extension::kId, data.size());
+  if (buffer.empty()) {
+    return false;
+  }
+  std::memcpy(buffer.data(), data.data(), data.size());
+  return true;
+}
+
+template <typename Extension>
+bool RtpPacket::ReserveExtension() {
+  auto buffer = AllocateExtension(Extension::kId, Extension::kValueSizeBytes);
+  if (buffer.empty())
+    return false;
+  memset(buffer.data(), 0, Extension::kValueSizeBytes);
+  return true;
+}
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
new file mode 100644
index 0000000000..c8d400a985
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -0,0 +1,428 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+RtpPacketHistory::StoredPacket::StoredPacket(
+    std::unique_ptr<RtpPacketToSend> packet,
+    Timestamp send_time,
+    uint64_t insert_order)
+    : packet_(std::move(packet)),
+      pending_transmission_(false),
+      send_time_(send_time),
+      insert_order_(insert_order),
+      times_retransmitted_(0) {}
+
+RtpPacketHistory::StoredPacket::StoredPacket(StoredPacket&&) = default;
+RtpPacketHistory::StoredPacket& RtpPacketHistory::StoredPacket::operator=(
+    RtpPacketHistory::StoredPacket&&) = default;
+RtpPacketHistory::StoredPacket::~StoredPacket() = default;
+
+void RtpPacketHistory::StoredPacket::IncrementTimesRetransmitted(
+    PacketPrioritySet* priority_set) {
+  // Check if this StoredPacket is in the priority set. If so, we need to remove
+  // it before updating `times_retransmitted_` since that is used in sorting,
+  // and then add it back.
+  const bool in_priority_set = priority_set && priority_set->erase(this) > 0;
+  ++times_retransmitted_;
+  if (in_priority_set) {
+    auto it = priority_set->insert(this);
+    RTC_DCHECK(it.second)
+        << "ERROR: Priority set already contains matching packet! In set: "
+           "insert order = "
+        << (*it.first)->insert_order_
+        << ", times retransmitted = " << (*it.first)->times_retransmitted_
+        << ". Trying to add: insert order = " << insert_order_
+        << ", times retransmitted = " << times_retransmitted_;
+  }
+}
+
+bool RtpPacketHistory::MoreUseful::operator()(StoredPacket* lhs,
+                                              StoredPacket* rhs) const {
+  // Prefer to send packets we haven't already sent as padding.
+  if (lhs->times_retransmitted() != rhs->times_retransmitted()) {
+    return lhs->times_retransmitted() < rhs->times_retransmitted();
+  }
+  // All else being equal, prefer newer packets.
+  return lhs->insert_order() > rhs->insert_order();
+}
+
+RtpPacketHistory::RtpPacketHistory(Clock* clock, bool enable_padding_prio)
+    : clock_(clock),
+      enable_padding_prio_(enable_padding_prio),
+      number_to_store_(0),
+      mode_(StorageMode::kDisabled),
+      rtt_(TimeDelta::MinusInfinity()),
+      packets_inserted_(0) {}
+
+RtpPacketHistory::~RtpPacketHistory() {}
+
+void RtpPacketHistory::SetStorePacketsStatus(StorageMode mode,
+                                             size_t number_to_store) {
+  RTC_DCHECK_LE(number_to_store, kMaxCapacity);
+  MutexLock lock(&lock_);
+  if (mode != StorageMode::kDisabled && mode_ != StorageMode::kDisabled) {
+    RTC_LOG(LS_WARNING) << "Purging packet history in order to re-set status.";
+  }
+  Reset();
+  mode_ = mode;
+  number_to_store_ = std::min(kMaxCapacity, number_to_store);
+}
+
+RtpPacketHistory::StorageMode RtpPacketHistory::GetStorageMode() const {
+  MutexLock lock(&lock_);
+  return mode_;
+}
+
+void RtpPacketHistory::SetRtt(TimeDelta rtt) {
+  MutexLock lock(&lock_);
+  RTC_DCHECK_GE(rtt, TimeDelta::Zero());
+  rtt_ = rtt;
+  // If storage is not disabled,  packets will be removed after a timeout
+  // that depends on the RTT. Changing the RTT may thus cause some packets
+  // become "old" and subject to removal.
+  if (mode_ != StorageMode::kDisabled) {
+    CullOldPackets();
+  }
+}
+
+void RtpPacketHistory::PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
+                                    Timestamp send_time) {
+  RTC_DCHECK(packet);
+  MutexLock lock(&lock_);
+  if (mode_ == StorageMode::kDisabled) {
+    return;
+  }
+
+  RTC_DCHECK(packet->allow_retransmission());
+  CullOldPackets();
+
+  // Store packet.
+  const uint16_t rtp_seq_no = packet->SequenceNumber();
+  int packet_index = GetPacketIndex(rtp_seq_no);
+  if (packet_index >= 0 &&
+      static_cast<size_t>(packet_index) < packet_history_.size() &&
+      packet_history_[packet_index].packet_ != nullptr) {
+    RTC_LOG(LS_WARNING) << "Duplicate packet inserted: " << rtp_seq_no;
+    // Remove previous packet to avoid inconsistent state.
+    RemovePacket(packet_index);
+    packet_index = GetPacketIndex(rtp_seq_no);
+  }
+
+  // Packet to be inserted ahead of first packet, expand front.
+  for (; packet_index < 0; ++packet_index) {
+    packet_history_.emplace_front();
+  }
+  // Packet to be inserted behind last packet, expand back.
+  while (static_cast<int>(packet_history_.size()) <= packet_index) {
+    packet_history_.emplace_back();
+  }
+
+  RTC_DCHECK_GE(packet_index, 0);
+  RTC_DCHECK_LT(packet_index, packet_history_.size());
+  RTC_DCHECK(packet_history_[packet_index].packet_ == nullptr);
+
+  packet_history_[packet_index] =
+      StoredPacket(std::move(packet), send_time, packets_inserted_++);
+
+  if (enable_padding_prio_) {
+    if (padding_priority_.size() >= kMaxPaddingHistory - 1) {
+      padding_priority_.erase(std::prev(padding_priority_.end()));
+    }
+    auto prio_it = padding_priority_.insert(&packet_history_[packet_index]);
+    RTC_DCHECK(prio_it.second) << "Failed to insert packet into prio set.";
+  }
+}
+
+std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacketAndMarkAsPending(
+    uint16_t sequence_number) {
+  return GetPacketAndMarkAsPending(
+      sequence_number, [](const RtpPacketToSend& packet) {
+        return std::make_unique<RtpPacketToSend>(packet);
+      });
+}
+
+std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacketAndMarkAsPending(
+    uint16_t sequence_number,
+    rtc::FunctionView<std::unique_ptr<RtpPacketToSend>(const RtpPacketToSend&)>
+        encapsulate) {
+  MutexLock lock(&lock_);
+  if (mode_ == StorageMode::kDisabled) {
+    return nullptr;
+  }
+
+  StoredPacket* packet = GetStoredPacket(sequence_number);
+  if (packet == nullptr) {
+    return nullptr;
+  }
+
+  if (packet->pending_transmission_) {
+    // Packet already in pacer queue, ignore this request.
+    return nullptr;
+  }
+
+  if (!VerifyRtt(*packet)) {
+    // Packet already resent within too short a time window, ignore.
+    return nullptr;
+  }
+
+  // Copy and/or encapsulate packet.
+  std::unique_ptr<RtpPacketToSend> encapsulated_packet =
+      encapsulate(*packet->packet_);
+  if (encapsulated_packet) {
+    packet->pending_transmission_ = true;
+  }
+
+  return encapsulated_packet;
+}
+
+void RtpPacketHistory::MarkPacketAsSent(uint16_t sequence_number) {
+  MutexLock lock(&lock_);
+  if (mode_ == StorageMode::kDisabled) {
+    return;
+  }
+
+  StoredPacket* packet = GetStoredPacket(sequence_number);
+  if (packet == nullptr) {
+    return;
+  }
+
+  // Update send-time, mark as no longer in pacer queue, and increment
+  // transmission count.
+  packet->set_send_time(clock_->CurrentTime());
+  packet->pending_transmission_ = false;
+  packet->IncrementTimesRetransmitted(enable_padding_prio_ ? &padding_priority_
+                                                           : nullptr);
+}
+
+bool RtpPacketHistory::GetPacketState(uint16_t sequence_number) const {
+  MutexLock lock(&lock_);
+  if (mode_ == StorageMode::kDisabled) {
+    return false;
+  }
+
+  int packet_index = GetPacketIndex(sequence_number);
+  if (packet_index < 0 ||
+      static_cast<size_t>(packet_index) >= packet_history_.size()) {
+    return false;
+  }
+  const StoredPacket& packet = packet_history_[packet_index];
+  if (packet.packet_ == nullptr) {
+    return false;
+  }
+
+  if (!VerifyRtt(packet)) {
+    return false;
+  }
+
+  return true;
+}
+
+bool RtpPacketHistory::VerifyRtt(
+    const RtpPacketHistory::StoredPacket& packet) const {
+  if (packet.times_retransmitted() > 0 &&
+      clock_->CurrentTime() - packet.send_time() < rtt_) {
+    // This packet has already been retransmitted once, and the time since
+    // that even is lower than on RTT. Ignore request as this packet is
+    // likely already in the network pipe.
+    return false;
+  }
+
+  return true;
+}
+
+std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPayloadPaddingPacket() {
+  // Default implementation always just returns a copy of the packet.
+  return GetPayloadPaddingPacket([](const RtpPacketToSend& packet) {
+    return std::make_unique<RtpPacketToSend>(packet);
+  });
+}
+
+std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPayloadPaddingPacket(
+    rtc::FunctionView<std::unique_ptr<RtpPacketToSend>(const RtpPacketToSend&)>
+        encapsulate) {
+  MutexLock lock(&lock_);
+  if (mode_ == StorageMode::kDisabled) {
+    return nullptr;
+  }
+
+  StoredPacket* best_packet = nullptr;
+  if (enable_padding_prio_ && !padding_priority_.empty()) {
+    auto best_packet_it = padding_priority_.begin();
+    best_packet = *best_packet_it;
+  } else if (!enable_padding_prio_ && !packet_history_.empty()) {
+    // Prioritization not available, pick the last packet.
+    for (auto it = packet_history_.rbegin(); it != packet_history_.rend();
+         ++it) {
+      if (it->packet_ != nullptr) {
+        best_packet = &(*it);
+        break;
+      }
+    }
+  }
+  if (best_packet == nullptr) {
+    return nullptr;
+  }
+
+  if (best_packet->pending_transmission_) {
+    // Because PacedSender releases it's lock when it calls
+    // GeneratePadding() there is the potential for a race where a new
+    // packet ends up here instead of the regular transmit path. In such a
+    // case, just return empty and it will be picked up on the next
+    // Process() call.
+    return nullptr;
+  }
+
+  auto padding_packet = encapsulate(*best_packet->packet_);
+  if (!padding_packet) {
+    return nullptr;
+  }
+
+  best_packet->set_send_time(clock_->CurrentTime());
+  best_packet->IncrementTimesRetransmitted(
+      enable_padding_prio_ ? &padding_priority_ : nullptr);
+
+  return padding_packet;
+}
+
+void RtpPacketHistory::CullAcknowledgedPackets(
+    rtc::ArrayView<const uint16_t> sequence_numbers) {
+  MutexLock lock(&lock_);
+  for (uint16_t sequence_number : sequence_numbers) {
+    int packet_index = GetPacketIndex(sequence_number);
+    if (packet_index < 0 ||
+        static_cast<size_t>(packet_index) >= packet_history_.size()) {
+      continue;
+    }
+    RemovePacket(packet_index);
+  }
+}
+
+void RtpPacketHistory::Clear() {
+  MutexLock lock(&lock_);
+  Reset();
+}
+
+void RtpPacketHistory::Reset() {
+  packet_history_.clear();
+  padding_priority_.clear();
+}
+
+void RtpPacketHistory::CullOldPackets() {
+  Timestamp now = clock_->CurrentTime();
+  TimeDelta packet_duration =
+      rtt_.IsFinite()
+          ? std::max(kMinPacketDurationRtt * rtt_, kMinPacketDuration)
+          : kMinPacketDuration;
+  while (!packet_history_.empty()) {
+    if (packet_history_.size() >= kMaxCapacity) {
+      // We have reached the absolute max capacity, remove one packet
+      // unconditionally.
+      RemovePacket(0);
+      continue;
+    }
+
+    const StoredPacket& stored_packet = packet_history_.front();
+    if (stored_packet.pending_transmission_) {
+      // Don't remove packets in the pacer queue, pending tranmission.
+      return;
+    }
+
+    if (stored_packet.send_time() + packet_duration > now) {
+      // Don't cull packets too early to avoid failed retransmission requests.
+      return;
+    }
+
+    if (packet_history_.size() >= number_to_store_ ||
+        stored_packet.send_time() +
+                (packet_duration * kPacketCullingDelayFactor) <=
+            now) {
+      // Too many packets in history, or this packet has timed out. Remove it
+      // and continue.
+      RemovePacket(0);
+    } else {
+      // No more packets can be removed right now.
+      return;
+    }
+  }
+}
+
+std::unique_ptr<RtpPacketToSend> RtpPacketHistory::RemovePacket(
+    int packet_index) {
+  // Move the packet out from the StoredPacket container.
+  std::unique_ptr<RtpPacketToSend> rtp_packet =
+      std::move(packet_history_[packet_index].packet_);
+
+  // Erase from padding priority set, if eligible.
+  if (enable_padding_prio_) {
+    padding_priority_.erase(&packet_history_[packet_index]);
+  }
+
+  if (packet_index == 0) {
+    while (!packet_history_.empty() &&
+           packet_history_.front().packet_ == nullptr) {
+      packet_history_.pop_front();
+    }
+  }
+
+  return rtp_packet;
+}
+
+int RtpPacketHistory::GetPacketIndex(uint16_t sequence_number) const {
+  if (packet_history_.empty()) {
+    return 0;
+  }
+
+  RTC_DCHECK(packet_history_.front().packet_ != nullptr);
+  int first_seq = packet_history_.front().packet_->SequenceNumber();
+  if (first_seq == sequence_number) {
+    return 0;
+  }
+
+  int packet_index = sequence_number - first_seq;
+  constexpr int kSeqNumSpan = std::numeric_limits<uint16_t>::max() + 1;
+
+  if (IsNewerSequenceNumber(sequence_number, first_seq)) {
+    if (sequence_number < first_seq) {
+      // Forward wrap.
+      packet_index += kSeqNumSpan;
+    }
+  } else if (sequence_number > first_seq) {
+    // Backwards wrap.
+    packet_index -= kSeqNumSpan;
+  }
+
+  return packet_index;
+}
+
+RtpPacketHistory::StoredPacket* RtpPacketHistory::GetStoredPacket(
+    uint16_t sequence_number) {
+  int index = GetPacketIndex(sequence_number);
+  if (index < 0 || static_cast<size_t>(index) >= packet_history_.size() ||
+      packet_history_[index].packet_ == nullptr) {
+    return nullptr;
+  }
+  return &packet_history_[index];
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history.h
new file mode 100644
index 0000000000..7475a35be3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
+
+#include <deque>
+#include <map>
+#include <memory>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "api/function_view.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class Clock;
+class RtpPacketToSend;
+
+class RtpPacketHistory {
+ public:
+  enum class StorageMode {
+    kDisabled,     // Don't store any packets.
+    kStoreAndCull  // Store up to `number_to_store` packets, but try to remove
+                   // packets as they time out or as signaled as received.
+  };
+
+  // Maximum number of packets we ever allow in the history.
+  static constexpr size_t kMaxCapacity = 9600;
+  // Maximum number of entries in prioritized queue of padding packets.
+  static constexpr size_t kMaxPaddingHistory = 63;
+  // Don't remove packets within max(1 second, 3x RTT).
+  static constexpr TimeDelta kMinPacketDuration = TimeDelta::Seconds(1);
+  static constexpr int kMinPacketDurationRtt = 3;
+  // With kStoreAndCull, always remove packets after 3x max(1000ms, 3x rtt).
+  static constexpr int kPacketCullingDelayFactor = 3;
+
+  RtpPacketHistory(Clock* clock, bool enable_padding_prio);
+
+  RtpPacketHistory() = delete;
+  RtpPacketHistory(const RtpPacketHistory&) = delete;
+  RtpPacketHistory& operator=(const RtpPacketHistory&) = delete;
+
+  ~RtpPacketHistory();
+
+  // Set/get storage mode. Note that setting the state will clear the history,
+  // even if setting the same state as is currently used.
+  void SetStorePacketsStatus(StorageMode mode, size_t number_to_store);
+  StorageMode GetStorageMode() const;
+
+  // Set RTT, used to avoid premature retransmission and to prevent over-writing
+  // a packet in the history before we are reasonably sure it has been received.
+  void SetRtt(TimeDelta rtt);
+
+  void PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
+                    Timestamp send_time);
+
+  // Gets stored RTP packet corresponding to the input |sequence number|.
+  // Returns nullptr if packet is not found or was (re)sent too recently.
+  // If a packet copy is returned, it will be marked as pending transmission but
+  // does not update send time, that must be done by MarkPacketAsSent().
+  std::unique_ptr<RtpPacketToSend> GetPacketAndMarkAsPending(
+      uint16_t sequence_number);
+
+  // In addition to getting packet and marking as sent, this method takes an
+  // encapsulator function that takes a reference to the packet and outputs a
+  // copy that may be wrapped in a container, eg RTX.
+  // If the the encapsulator returns nullptr, the retransmit is aborted and the
+  // packet will not be marked as pending.
+  std::unique_ptr<RtpPacketToSend> GetPacketAndMarkAsPending(
+      uint16_t sequence_number,
+      rtc::FunctionView<std::unique_ptr<RtpPacketToSend>(
+          const RtpPacketToSend&)> encapsulate);
+
+  // Updates the send time for the given packet and increments the transmission
+  // counter. Marks the packet as no longer being in the pacer queue.
+  void MarkPacketAsSent(uint16_t sequence_number);
+
+  // Returns true if history contains packet with `sequence_number` and it can
+  // be retransmitted.
+  bool GetPacketState(uint16_t sequence_number) const;
+
+  // Get the packet (if any) from the history, that is deemed most likely to
+  // the remote side. This is calculated from heuristics such as packet age
+  // and times retransmitted. Updated the send time of the packet, so is not
+  // a const method.
+  std::unique_ptr<RtpPacketToSend> GetPayloadPaddingPacket();
+
+  // Same as GetPayloadPaddingPacket(void), but adds an encapsulation
+  // that can be used for instance to encapsulate the packet in an RTX
+  // container, or to abort getting the packet if the function returns
+  // nullptr.
+  std::unique_ptr<RtpPacketToSend> GetPayloadPaddingPacket(
+      rtc::FunctionView<std::unique_ptr<RtpPacketToSend>(
+          const RtpPacketToSend&)> encapsulate);
+
+  // Cull packets that have been acknowledged as received by the remote end.
+  void CullAcknowledgedPackets(rtc::ArrayView<const uint16_t> sequence_numbers);
+
+  // Remove all pending packets from the history, but keep storage mode and
+  // capacity.
+  void Clear();
+
+ private:
+  struct MoreUseful;
+  class StoredPacket;
+  using PacketPrioritySet = std::set<StoredPacket*, MoreUseful>;
+
+  class StoredPacket {
+   public:
+    StoredPacket() = default;
+    StoredPacket(std::unique_ptr<RtpPacketToSend> packet,
+                 Timestamp send_time,
+                 uint64_t insert_order);
+    StoredPacket(StoredPacket&&);
+    StoredPacket& operator=(StoredPacket&&);
+    ~StoredPacket();
+
+    uint64_t insert_order() const { return insert_order_; }
+    size_t times_retransmitted() const { return times_retransmitted_; }
+    void IncrementTimesRetransmitted(PacketPrioritySet* priority_set);
+
+    // The time of last transmission, including retransmissions.
+    Timestamp send_time() const { return send_time_; }
+    void set_send_time(Timestamp value) { send_time_ = value; }
+
+    // The actual packet.
+    std::unique_ptr<RtpPacketToSend> packet_;
+
+    // True if the packet is currently in the pacer queue pending transmission.
+    bool pending_transmission_;
+
+   private:
+    Timestamp send_time_ = Timestamp::Zero();
+
+    // Unique number per StoredPacket, incremented by one for each added
+    // packet. Used to sort on insert order.
+    uint64_t insert_order_;
+
+    // Number of times RE-transmitted, ie excluding the first transmission.
+    size_t times_retransmitted_;
+  };
+  struct MoreUseful {
+    bool operator()(StoredPacket* lhs, StoredPacket* rhs) const;
+  };
+
+  // Helper method to check if packet has too recently been sent.
+  bool VerifyRtt(const StoredPacket& packet) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  void Reset() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  void CullOldPackets() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  // Removes the packet from the history, and context/mapping that has been
+  // stored. Returns the RTP packet instance contained within the StoredPacket.
+  std::unique_ptr<RtpPacketToSend> RemovePacket(int packet_index)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  int GetPacketIndex(uint16_t sequence_number) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  StoredPacket* GetStoredPacket(uint16_t sequence_number)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+
+  Clock* const clock_;
+  const bool enable_padding_prio_;
+  mutable Mutex lock_;
+  size_t number_to_store_ RTC_GUARDED_BY(lock_);
+  StorageMode mode_ RTC_GUARDED_BY(lock_);
+  TimeDelta rtt_ RTC_GUARDED_BY(lock_);
+
+  // Queue of stored packets, ordered by sequence number, with older packets in
+  // the front and new packets being added to the back. Note that there may be
+  // wrap-arounds so the back may have a lower sequence number.
+  // Packets may also be removed out-of-order, in which case there will be
+  // instances of StoredPacket with `packet_` set to nullptr. The first and last
+  // entry in the queue will however always be populated.
+  std::deque<StoredPacket> packet_history_ RTC_GUARDED_BY(lock_);
+
+  // Total number of packets with inserted.
+  uint64_t packets_inserted_ RTC_GUARDED_BY(lock_);
+  // Objects from `packet_history_` ordered by "most likely to be useful", used
+  // in GetPayloadPaddingPacket().
+  PacketPrioritySet padding_priority_ RTC_GUARDED_BY(lock_);
+};
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
new file mode 100644
index 0000000000..f50541849e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -0,0 +1,681 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+// Set a high sequence number so we'll suffer a wrap-around.
+constexpr uint16_t kStartSeqNum = 65534u;
+
+// Utility method for truncating sequence numbers to uint16.
+uint16_t To16u(size_t sequence_number) {
+  return static_cast<uint16_t>(sequence_number & 0xFFFF);
+}
+}  // namespace
+
+using StorageMode = RtpPacketHistory::StorageMode;
+
+class RtpPacketHistoryTest : public ::testing::TestWithParam<bool> {
+ protected:
+  RtpPacketHistoryTest()
+      : fake_clock_(123456),
+        hist_(&fake_clock_, /*enable_padding_prio=*/GetParam()) {}
+
+  SimulatedClock fake_clock_;
+  RtpPacketHistory hist_;
+
+  std::unique_ptr<RtpPacketToSend> CreateRtpPacket(uint16_t seq_num) {
+    // Payload, ssrc, timestamp and extensions are irrelevant for this tests.
+    std::unique_ptr<RtpPacketToSend> packet(new RtpPacketToSend(nullptr));
+    packet->SetSequenceNumber(seq_num);
+    packet->set_capture_time(fake_clock_.CurrentTime());
+    packet->set_allow_retransmission(true);
+    return packet;
+  }
+};
+
+TEST_P(RtpPacketHistoryTest, SetStoreStatus) {
+  EXPECT_EQ(StorageMode::kDisabled, hist_.GetStorageMode());
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  EXPECT_EQ(StorageMode::kStoreAndCull, hist_.GetStorageMode());
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  EXPECT_EQ(StorageMode::kStoreAndCull, hist_.GetStorageMode());
+  hist_.SetStorePacketsStatus(StorageMode::kDisabled, 0);
+  EXPECT_EQ(StorageMode::kDisabled, hist_.GetStorageMode());
+}
+
+TEST_P(RtpPacketHistoryTest, ClearsHistoryAfterSetStoreStatus) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum),
+                     /*send_time=*/fake_clock_.CurrentTime());
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+
+  // Changing store status, even to the current one, will clear the history.
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+}
+
+TEST_P(RtpPacketHistoryTest, StartSeqResetAfterReset) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum),
+                     /*send_time=*/fake_clock_.CurrentTime());
+  // Mark packet as pending so it won't be removed.
+  EXPECT_TRUE(hist_.GetPacketAndMarkAsPending(kStartSeqNum));
+
+  // Changing store status, to clear the history.
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+
+  // Add a new packet.
+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 1)),
+                     /*send_time=*/fake_clock_.CurrentTime());
+  EXPECT_TRUE(hist_.GetPacketAndMarkAsPending(To16u(kStartSeqNum + 1)));
+
+  // Advance time past where packet expires.
+  fake_clock_.AdvanceTime(RtpPacketHistory::kPacketCullingDelayFactor *
+                          RtpPacketHistory::kMinPacketDuration);
+
+  // Add one more packet and verify no state left from packet before reset.
+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 2)),
+                     /*send_time=*/fake_clock_.CurrentTime());
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 1)));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2)));
+}
+
+TEST_P(RtpPacketHistoryTest, NoStoreStatus) {
+  EXPECT_EQ(StorageMode::kDisabled, hist_.GetStorageMode());
+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
+  hist_.PutRtpPacket(std::move(packet),
+                     /*send_time=*/fake_clock_.CurrentTime());
+  // Packet should not be stored.
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+}
+
+TEST_P(RtpPacketHistoryTest, GetRtpPacket_NotStored) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  EXPECT_FALSE(hist_.GetPacketState(0));
+}
+
+TEST_P(RtpPacketHistoryTest, PutRtpPacket) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
+
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+  hist_.PutRtpPacket(std::move(packet),
+                     /*send_time=*/fake_clock_.CurrentTime());
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+}
+
+TEST_P(RtpPacketHistoryTest, GetRtpPacket) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  Timestamp capture_time = Timestamp::Millis(1);
+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
+  packet->set_capture_time(capture_time);
+  rtc::CopyOnWriteBuffer buffer = packet->Buffer();
+  hist_.PutRtpPacket(std::move(packet),
+                     /*send_time=*/fake_clock_.CurrentTime());
+
+  std::unique_ptr<RtpPacketToSend> packet_out =
+      hist_.GetPacketAndMarkAsPending(kStartSeqNum);
+  ASSERT_TRUE(packet_out);
+  EXPECT_EQ(buffer, packet_out->Buffer());
+  EXPECT_EQ(capture_time, packet_out->capture_time());
+}
+
+TEST_P(RtpPacketHistoryTest, MinResendTime) {
+  static const TimeDelta kMinRetransmitInterval = TimeDelta::Millis(100);
+
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+  hist_.SetRtt(kMinRetransmitInterval);
+  Timestamp capture_time = fake_clock_.CurrentTime();
+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
+  size_t len = packet->size();
+  hist_.PutRtpPacket(std::move(packet), fake_clock_.CurrentTime());
+
+  // First retransmission - allow early retransmission.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  packet = hist_.GetPacketAndMarkAsPending(kStartSeqNum);
+  ASSERT_TRUE(packet);
+  EXPECT_EQ(len, packet->size());
+  EXPECT_EQ(packet->capture_time(), capture_time);
+  hist_.MarkPacketAsSent(kStartSeqNum);
+
+  // Second retransmission - advance time to just before retransmission OK.
+  fake_clock_.AdvanceTime(kMinRetransmitInterval - TimeDelta::Millis(1));
+  EXPECT_FALSE(hist_.GetPacketAndMarkAsPending(kStartSeqNum));
+
+  // Advance time to just after retransmission OK.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  EXPECT_TRUE(hist_.GetPacketAndMarkAsPending(kStartSeqNum));
+}
+
+TEST_P(RtpPacketHistoryTest, RemovesOldestSentPacketWhenAtMaxSize) {
+  const size_t kMaxNumPackets = 10;
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, kMaxNumPackets);
+
+  // History does not allow removing packets within kMinPacketDuration,
+  // so in order to test capacity, make sure insertion spans this time.
+  const TimeDelta kPacketInterval =
+      RtpPacketHistory::kMinPacketDuration / kMaxNumPackets;
+
+  // Add packets until the buffer is full.
+  for (size_t i = 0; i < kMaxNumPackets; ++i) {
+    std::unique_ptr<RtpPacketToSend> packet =
+        CreateRtpPacket(To16u(kStartSeqNum + i));
+    // Immediate mark packet as sent.
+    hist_.PutRtpPacket(std::move(packet), fake_clock_.CurrentTime());
+    fake_clock_.AdvanceTime(kPacketInterval);
+  }
+
+  // First packet should still be there.
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+
+  // History is full, oldest one should be overwritten.
+  std::unique_ptr<RtpPacketToSend> packet =
+      CreateRtpPacket(To16u(kStartSeqNum + kMaxNumPackets));
+  hist_.PutRtpPacket(std::move(packet), fake_clock_.CurrentTime());
+
+  // Oldest packet should be gone, but packet after than one still present.
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 1)));
+}
+
+TEST_P(RtpPacketHistoryTest, RemovesOldestPacketWhenAtMaxCapacity) {
+  // Tests the absolute upper bound on number of stored packets. Don't allow
+  // storing more than this, even if packets have not yet been sent.
+  const size_t kMaxNumPackets = RtpPacketHistory::kMaxCapacity;
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull,
+                              RtpPacketHistory::kMaxCapacity);
+
+  // Add packets until the buffer is full.
+  for (size_t i = 0; i < kMaxNumPackets; ++i) {
+    std::unique_ptr<RtpPacketToSend> packet =
+        CreateRtpPacket(To16u(kStartSeqNum + i));
+    hist_.PutRtpPacket(std::move(packet),
+                       /*send_time=*/fake_clock_.CurrentTime());
+    // Mark packets as pending, preventing it from being removed.
+    hist_.GetPacketAndMarkAsPending(To16u(kStartSeqNum + i));
+  }
+
+  // First packet should still be there.
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+
+  // History is full, oldest one should be overwritten.
+  std::unique_ptr<RtpPacketToSend> packet =
+      CreateRtpPacket(To16u(kStartSeqNum + kMaxNumPackets));
+  hist_.PutRtpPacket(std::move(packet), fake_clock_.CurrentTime());
+
+  // Oldest packet should be gone, but packet after than one still present.
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 1)));
+}
+
+TEST_P(RtpPacketHistoryTest, RemovesLowestPrioPaddingWhenAtMaxCapacity) {
+  if (!GetParam()) {
+    // Padding prioritization is off, ignore this test.
+    return;
+  }
+
+  // Tests the absolute upper bound on number of packets in the prioritized
+  // set of potential padding packets.
+  const size_t kMaxNumPackets = RtpPacketHistory::kMaxPaddingHistory;
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, kMaxNumPackets * 2);
+  hist_.SetRtt(TimeDelta::Millis(1));
+
+  // Add packets until the max is reached, and then yet another one.
+  for (size_t i = 0; i < kMaxNumPackets + 1; ++i) {
+    std::unique_ptr<RtpPacketToSend> packet =
+        CreateRtpPacket(To16u(kStartSeqNum + i));
+    // Don't mark packets as sent, preventing them from being removed.
+    hist_.PutRtpPacket(std::move(packet), fake_clock_.CurrentTime());
+  }
+
+  // Advance time to allow retransmission/padding.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+
+  // The oldest packet will be least prioritized and has fallen out of the
+  // priority set.
+  for (size_t i = kMaxNumPackets - 1; i > 0; --i) {
+    auto packet = hist_.GetPayloadPaddingPacket();
+    ASSERT_TRUE(packet);
+    EXPECT_EQ(packet->SequenceNumber(), To16u(kStartSeqNum + i + 1));
+  }
+
+  // Wrap around to newest padding packet again.
+  auto packet = hist_.GetPayloadPaddingPacket();
+  ASSERT_TRUE(packet);
+  EXPECT_EQ(packet->SequenceNumber(), To16u(kStartSeqNum + kMaxNumPackets));
+}
+
+TEST_P(RtpPacketHistoryTest, DontRemoveTooRecentlyTransmittedPackets) {
+  // Set size to remove old packets as soon as possible.
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 1);
+
+  // Add a packet, marked as send, and advance time to just before removal time.
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+  fake_clock_.AdvanceTime(RtpPacketHistory::kMinPacketDuration -
+                          TimeDelta::Millis(1));
+
+  // Add a new packet to trigger culling.
+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 1)),
+                     fake_clock_.CurrentTime());
+  // First packet should still be there.
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+
+  // Advance time to where packet will be eligible for removal and try again.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 2)),
+                     fake_clock_.CurrentTime());
+  // First packet should no be gone, but next one still there.
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 1)));
+}
+
+TEST_P(RtpPacketHistoryTest, DontRemoveTooRecentlyTransmittedPacketsHighRtt) {
+  const TimeDelta kRtt = RtpPacketHistory::kMinPacketDuration * 2;
+  const TimeDelta kPacketTimeout =
+      kRtt * RtpPacketHistory::kMinPacketDurationRtt;
+
+  // Set size to remove old packets as soon as possible.
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 1);
+  hist_.SetRtt(kRtt);
+
+  // Add a packet, marked as send, and advance time to just before removal time.
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+  fake_clock_.AdvanceTime(kPacketTimeout - TimeDelta::Millis(1));
+
+  // Add a new packet to trigger culling.
+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 1)),
+                     fake_clock_.CurrentTime());
+  // First packet should still be there.
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+
+  // Advance time to where packet will be eligible for removal and try again.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 2)),
+                     fake_clock_.CurrentTime());
+  // First packet should no be gone, but next one still there.
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 1)));
+}
+
+TEST_P(RtpPacketHistoryTest, RemovesOldWithCulling) {
+  const size_t kMaxNumPackets = 10;
+  // Enable culling. Even without feedback, this can trigger early removal.
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, kMaxNumPackets);
+
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+
+  TimeDelta kMaxPacketDuration = RtpPacketHistory::kMinPacketDuration *
+                                 RtpPacketHistory::kPacketCullingDelayFactor;
+  fake_clock_.AdvanceTime(kMaxPacketDuration - TimeDelta::Millis(1));
+
+  // First packet should still be there.
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+
+  // Advance to where packet can be culled, even if buffer is not full.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 1)),
+                     fake_clock_.CurrentTime());
+
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+}
+
+TEST_P(RtpPacketHistoryTest, RemovesOldWithCullingHighRtt) {
+  const size_t kMaxNumPackets = 10;
+  const TimeDelta kRtt = RtpPacketHistory::kMinPacketDuration * 2;
+  // Enable culling. Even without feedback, this can trigger early removal.
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, kMaxNumPackets);
+  hist_.SetRtt(kRtt);
+
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+
+  TimeDelta kMaxPacketDuration = kRtt *
+                                 RtpPacketHistory::kMinPacketDurationRtt *
+                                 RtpPacketHistory::kPacketCullingDelayFactor;
+  fake_clock_.AdvanceTime(kMaxPacketDuration - TimeDelta::Millis(1));
+
+  // First packet should still be there.
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+
+  // Advance to where packet can be culled, even if buffer is not full.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 1)),
+                     fake_clock_.CurrentTime());
+
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+}
+
+TEST_P(RtpPacketHistoryTest, CullWithAcks) {
+  const TimeDelta kPacketLifetime = RtpPacketHistory::kMinPacketDuration *
+                                    RtpPacketHistory::kPacketCullingDelayFactor;
+
+  const Timestamp start_time = fake_clock_.CurrentTime();
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+
+  // Insert three packets 33ms apart, immediately mark them as sent.
+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
+  packet->SetPayloadSize(50);
+  hist_.PutRtpPacket(std::move(packet),
+                     /*send_time=*/fake_clock_.CurrentTime());
+  fake_clock_.AdvanceTimeMilliseconds(33);
+  packet = CreateRtpPacket(To16u(kStartSeqNum + 1));
+  packet->SetPayloadSize(50);
+  hist_.PutRtpPacket(std::move(packet),
+                     /*send_time=*/fake_clock_.CurrentTime());
+  fake_clock_.AdvanceTimeMilliseconds(33);
+  packet = CreateRtpPacket(To16u(kStartSeqNum + 2));
+  packet->SetPayloadSize(50);
+  hist_.PutRtpPacket(std::move(packet),
+                     /*send_time=*/fake_clock_.CurrentTime());
+
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 1)));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2)));
+
+  // Remove middle one using ack, check that only that one is gone.
+  std::vector<uint16_t> acked_sequence_numbers = {To16u(kStartSeqNum + 1)};
+  hist_.CullAcknowledgedPackets(acked_sequence_numbers);
+
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+  EXPECT_FALSE(hist_.GetPacketState(To16u(kStartSeqNum + 1)));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2)));
+
+  // Advance time to where second packet would have expired, verify first packet
+  // is removed.
+  Timestamp second_packet_expiry_time =
+      start_time + kPacketLifetime + TimeDelta::Millis(33 + 1);
+  fake_clock_.AdvanceTime(second_packet_expiry_time -
+                          fake_clock_.CurrentTime());
+  hist_.SetRtt(TimeDelta::Millis(1));  // Trigger culling of old packets.
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+  EXPECT_FALSE(hist_.GetPacketState(To16u(kStartSeqNum + 1)));
+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2)));
+
+  // Advance to where last packet expires, verify all gone.
+  fake_clock_.AdvanceTimeMilliseconds(33);
+  hist_.SetRtt(TimeDelta::Millis(1));  // Trigger culling of old packets.
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+  EXPECT_FALSE(hist_.GetPacketState(To16u(kStartSeqNum + 1)));
+  EXPECT_FALSE(hist_.GetPacketState(To16u(kStartSeqNum + 2)));
+}
+
+TEST_P(RtpPacketHistoryTest, GetPacketAndSetSent) {
+  const TimeDelta kRtt = RtpPacketHistory::kMinPacketDuration * 2;
+  hist_.SetRtt(kRtt);
+
+  // Set size to remove old packets as soon as possible.
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 1);
+
+  // Add a sent packet to the history.
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+
+  // Retransmission request, first retransmission is allowed immediately.
+  EXPECT_TRUE(hist_.GetPacketAndMarkAsPending(kStartSeqNum));
+
+  // Packet not yet sent, new retransmission not allowed.
+  fake_clock_.AdvanceTime(kRtt);
+  EXPECT_FALSE(hist_.GetPacketAndMarkAsPending(kStartSeqNum));
+
+  // Mark as sent, but too early for retransmission.
+  hist_.MarkPacketAsSent(kStartSeqNum);
+  EXPECT_FALSE(hist_.GetPacketAndMarkAsPending(kStartSeqNum));
+
+  // Enough time has passed, retransmission is allowed again.
+  fake_clock_.AdvanceTime(kRtt);
+  EXPECT_TRUE(hist_.GetPacketAndMarkAsPending(kStartSeqNum));
+}
+
+TEST_P(RtpPacketHistoryTest, GetPacketWithEncapsulation) {
+  const uint32_t kSsrc = 92384762;
+  const TimeDelta kRtt = RtpPacketHistory::kMinPacketDuration * 2;
+  hist_.SetRtt(kRtt);
+
+  // Set size to remove old packets as soon as possible.
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 1);
+
+  // Add a sent packet to the history, with a set SSRC.
+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
+  packet->SetSsrc(kSsrc);
+  hist_.PutRtpPacket(std::move(packet), fake_clock_.CurrentTime());
+
+  // Retransmission request, simulate an RTX-like encapsulation, were the packet
+  // is sent on a different SSRC.
+  std::unique_ptr<RtpPacketToSend> retransmit_packet =
+      hist_.GetPacketAndMarkAsPending(
+          kStartSeqNum, [](const RtpPacketToSend& packet) {
+            auto encapsulated_packet =
+                std::make_unique<RtpPacketToSend>(packet);
+            encapsulated_packet->SetSsrc(packet.Ssrc() + 1);
+            return encapsulated_packet;
+          });
+  ASSERT_TRUE(retransmit_packet);
+  EXPECT_EQ(retransmit_packet->Ssrc(), kSsrc + 1);
+}
+
+TEST_P(RtpPacketHistoryTest, GetPacketWithEncapsulationAbortOnNullptr) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 1);
+
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+
+  // Retransmission request, but the encapsulator determines that this packet is
+  // not suitable for retransmission (bandwidth exhausted?) so the retransmit is
+  // aborted and the packet is not marked as pending.
+  EXPECT_FALSE(hist_.GetPacketAndMarkAsPending(
+      kStartSeqNum, [](const RtpPacketToSend&) { return nullptr; }));
+
+  // New try, this time getting the packet should work, and it should not be
+  // blocked due to any pending status.
+  EXPECT_TRUE(hist_.GetPacketAndMarkAsPending(kStartSeqNum));
+}
+
+TEST_P(RtpPacketHistoryTest, DontRemovePendingTransmissions) {
+  const TimeDelta kRtt = RtpPacketHistory::kMinPacketDuration * 2;
+  const TimeDelta kPacketTimeout =
+      kRtt * RtpPacketHistory::kMinPacketDurationRtt;
+
+  // Set size to remove old packets as soon as possible.
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 1);
+  hist_.SetRtt(kRtt);
+
+  // Add a sent packet.
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+
+  // Advance clock to just before packet timeout.
+  fake_clock_.AdvanceTime(kPacketTimeout - TimeDelta::Millis(1));
+  // Mark as enqueued in pacer.
+  EXPECT_TRUE(hist_.GetPacketAndMarkAsPending(kStartSeqNum));
+
+  // Advance clock to where packet would have timed out. It should still
+  // be there and pending.
+  fake_clock_.AdvanceTimeMilliseconds(1);
+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
+
+  // Packet sent. Now it can be removed.
+  hist_.MarkPacketAsSent(kStartSeqNum);
+  hist_.SetRtt(kRtt);  // Force culling of old packets.
+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
+}
+
+TEST_P(RtpPacketHistoryTest, PrioritizedPayloadPadding) {
+  if (!GetParam()) {
+    // Padding prioritization is off, ignore this test.
+    return;
+  }
+
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 1);
+
+  // Add two sent packets, one millisecond apart.
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+  fake_clock_.AdvanceTimeMilliseconds(1);
+
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1),
+                     fake_clock_.CurrentTime());
+  fake_clock_.AdvanceTimeMilliseconds(1);
+
+  // Latest packet given equal retransmission count.
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(),
+            kStartSeqNum + 1);
+
+  // Older packet has lower retransmission count.
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(), kStartSeqNum);
+
+  // Equal retransmission count again, use newest packet.
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(),
+            kStartSeqNum + 1);
+
+  // Older packet has lower retransmission count.
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(), kStartSeqNum);
+
+  // Remove newest packet.
+  hist_.CullAcknowledgedPackets(std::vector<uint16_t>{kStartSeqNum + 1});
+
+  // Only older packet left.
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(), kStartSeqNum);
+
+  hist_.CullAcknowledgedPackets(std::vector<uint16_t>{kStartSeqNum});
+
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket(), nullptr);
+}
+
+TEST_P(RtpPacketHistoryTest, NoPendingPacketAsPadding) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 1);
+
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+  fake_clock_.AdvanceTimeMilliseconds(1);
+
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(), kStartSeqNum);
+
+  // If packet is pending retransmission, don't try to use it as padding.
+  hist_.GetPacketAndMarkAsPending(kStartSeqNum);
+  EXPECT_EQ(nullptr, hist_.GetPayloadPaddingPacket());
+
+  // Market it as no longer pending, should be usable as padding again.
+  hist_.MarkPacketAsSent(kStartSeqNum);
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(), kStartSeqNum);
+}
+
+TEST_P(RtpPacketHistoryTest, PayloadPaddingWithEncapsulation) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 1);
+
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+  fake_clock_.AdvanceTimeMilliseconds(1);
+
+  // Aborted padding.
+  EXPECT_EQ(nullptr, hist_.GetPayloadPaddingPacket(
+                         [](const RtpPacketToSend&) { return nullptr; }));
+
+  // Get copy of packet, but with sequence number modified.
+  auto padding_packet =
+      hist_.GetPayloadPaddingPacket([&](const RtpPacketToSend& packet) {
+        auto encapsulated_packet = std::make_unique<RtpPacketToSend>(packet);
+        encapsulated_packet->SetSequenceNumber(kStartSeqNum + 1);
+        return encapsulated_packet;
+      });
+  ASSERT_TRUE(padding_packet);
+  EXPECT_EQ(padding_packet->SequenceNumber(), kStartSeqNum + 1);
+}
+
+TEST_P(RtpPacketHistoryTest, NackAfterAckIsNoop) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 2);
+  // Add two sent packets.
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), fake_clock_.CurrentTime());
+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1),
+                     fake_clock_.CurrentTime());
+  // Remove newest one.
+  hist_.CullAcknowledgedPackets(std::vector<uint16_t>{kStartSeqNum + 1});
+  // Retransmission request for already acked packet, should be noop.
+  auto packet = hist_.GetPacketAndMarkAsPending(kStartSeqNum + 1);
+  EXPECT_EQ(packet.get(), nullptr);
+}
+
+TEST_P(RtpPacketHistoryTest, OutOfOrderInsertRemoval) {
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
+
+  // Insert packets, out of order, including both forwards and backwards
+  // sequence number wraps.
+  const int seq_offsets[] = {0, 1, -1, 2, -2, 3, -3};
+
+  for (int offset : seq_offsets) {
+    uint16_t seq_no = To16u(kStartSeqNum + offset);
+    std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(seq_no);
+    packet->SetPayloadSize(50);
+    hist_.PutRtpPacket(std::move(packet), fake_clock_.CurrentTime());
+    fake_clock_.AdvanceTimeMilliseconds(33);
+  }
+
+  // Check packet are there and remove them in the same out-of-order fashion.
+  for (int offset : seq_offsets) {
+    uint16_t seq_no = To16u(kStartSeqNum + offset);
+    EXPECT_TRUE(hist_.GetPacketState(seq_no));
+    std::vector<uint16_t> acked_sequence_numbers = {seq_no};
+    hist_.CullAcknowledgedPackets(acked_sequence_numbers);
+    EXPECT_FALSE(hist_.GetPacketState(seq_no));
+  }
+}
+
+TEST_P(RtpPacketHistoryTest, UsesLastPacketAsPaddingWithPrioOff) {
+  if (GetParam()) {
+    // Padding prioritization is enabled, ignore this test.
+    return;
+  }
+
+  const size_t kHistorySize = 10;
+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, kHistorySize);
+
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket(), nullptr);
+
+  for (size_t i = 0; i < kHistorySize; ++i) {
+    hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + i)),
+                       fake_clock_.CurrentTime());
+    hist_.MarkPacketAsSent(To16u(kStartSeqNum + i));
+    fake_clock_.AdvanceTimeMilliseconds(1);
+
+    // Last packet always returned.
+    EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(),
+              To16u(kStartSeqNum + i));
+    EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(),
+              To16u(kStartSeqNum + i));
+    EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(),
+              To16u(kStartSeqNum + i));
+  }
+
+  // Remove packets from the end, last in the list should be returned.
+  for (size_t i = kHistorySize - 1; i > 0; --i) {
+    hist_.CullAcknowledgedPackets(
+        std::vector<uint16_t>{To16u(kStartSeqNum + i)});
+
+    EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(),
+              To16u(kStartSeqNum + i - 1));
+    EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(),
+              To16u(kStartSeqNum + i - 1));
+    EXPECT_EQ(hist_.GetPayloadPaddingPacket()->SequenceNumber(),
+              To16u(kStartSeqNum + i - 1));
+  }
+
+  hist_.CullAcknowledgedPackets(std::vector<uint16_t>{kStartSeqNum});
+  EXPECT_EQ(hist_.GetPayloadPaddingPacket(), nullptr);
+}
+
+INSTANTIATE_TEST_SUITE_P(WithAndWithoutPaddingPrio,
+                         RtpPacketHistoryTest,
+                         ::testing::Bool());
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_received.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
new file mode 100644
index 0000000000..6b2cc76981
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+RtpPacketReceived::RtpPacketReceived() = default;
+RtpPacketReceived::RtpPacketReceived(
+    const ExtensionManager* extensions,
+    webrtc::Timestamp arrival_time /*= webrtc::Timestamp::MinusInfinity()*/)
+    : RtpPacket(extensions), arrival_time_(arrival_time) {}
+RtpPacketReceived::RtpPacketReceived(const RtpPacketReceived& packet) = default;
+RtpPacketReceived::RtpPacketReceived(RtpPacketReceived&& packet) = default;
+
+RtpPacketReceived& RtpPacketReceived::operator=(
+    const RtpPacketReceived& packet) = default;
+RtpPacketReceived& RtpPacketReceived::operator=(RtpPacketReceived&& packet) =
+    default;
+
+RtpPacketReceived::~RtpPacketReceived() {}
+
+void RtpPacketReceived::GetHeader(RTPHeader* header) const {
+  header->markerBit = Marker();
+  header->payloadType = PayloadType();
+  header->sequenceNumber = SequenceNumber();
+  header->timestamp = Timestamp();
+  header->ssrc = Ssrc();
+  std::vector<uint32_t> csrcs = Csrcs();
+  header->numCSRCs = rtc::dchecked_cast<uint8_t>(csrcs.size());
+  for (size_t i = 0; i < csrcs.size(); ++i) {
+    header->arrOfCSRCs[i] = csrcs[i];
+  }
+  header->paddingLength = padding_size();
+  header->headerLength = headers_size();
+  header->payload_type_frequency = payload_type_frequency();
+  header->extension.hasTransmissionTimeOffset =
+      GetExtension<TransmissionOffset>(
+          &header->extension.transmissionTimeOffset);
+  header->extension.hasAbsoluteSendTime =
+      GetExtension<AbsoluteSendTime>(&header->extension.absoluteSendTime);
+  header->extension.absolute_capture_time =
+      GetExtension<AbsoluteCaptureTimeExtension>();
+  header->extension.hasTransportSequenceNumber =
+      GetExtension<TransportSequenceNumberV2>(
+          &header->extension.transportSequenceNumber,
+          &header->extension.feedback_request) ||
+      GetExtension<TransportSequenceNumber>(
+          &header->extension.transportSequenceNumber);
+  header->extension.hasAudioLevel = GetExtension<AudioLevel>(
+      &header->extension.voiceActivity, &header->extension.audioLevel);
+  header->extension.hasVideoRotation =
+      GetExtension<VideoOrientation>(&header->extension.videoRotation);
+  header->extension.hasVideoContentType =
+      GetExtension<VideoContentTypeExtension>(
+          &header->extension.videoContentType);
+  header->extension.has_video_timing =
+      GetExtension<VideoTimingExtension>(&header->extension.video_timing);
+  GetExtension<RtpStreamId>(&header->extension.stream_id);
+  GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
+  GetExtension<RtpMid>(&header->extension.mid);
+  GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
+  header->extension.color_space = GetExtension<ColorSpaceExtension>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_received.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_received.h
new file mode 100644
index 0000000000..f290a643a4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_received.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
+
+#include <stdint.h>
+
+#include <utility>
+
+#include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/rtp_headers.h"
+#include "api/scoped_refptr.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+
+namespace webrtc {
+// Class to hold rtp packet with metadata for receiver side.
+// The metadata is not parsed from the rtp packet, but may be derived from the
+// data that is parsed from the rtp packet.
+class RtpPacketReceived : public RtpPacket {
+ public:
+  RtpPacketReceived();
+  explicit RtpPacketReceived(
+      const ExtensionManager* extensions,
+      webrtc::Timestamp arrival_time = webrtc::Timestamp::MinusInfinity());
+  RtpPacketReceived(const RtpPacketReceived& packet);
+  RtpPacketReceived(RtpPacketReceived&& packet);
+
+  RtpPacketReceived& operator=(const RtpPacketReceived& packet);
+  RtpPacketReceived& operator=(RtpPacketReceived&& packet);
+
+  ~RtpPacketReceived();
+
+  // TODO(danilchap): Remove this function when all code update to use RtpPacket
+  // directly. Function is there just for easier backward compatibilty.
+  void GetHeader(RTPHeader* header) const;
+
+  // Time in local time base as close as it can to packet arrived on the
+  // network.
+  webrtc::Timestamp arrival_time() const { return arrival_time_; }
+  void set_arrival_time(webrtc::Timestamp time) { arrival_time_ = time; }
+
+  // Flag if packet was recovered via RTX or FEC.
+  bool recovered() const { return recovered_; }
+  void set_recovered(bool value) { recovered_ = value; }
+
+  int payload_type_frequency() const { return payload_type_frequency_; }
+  void set_payload_type_frequency(int value) {
+    payload_type_frequency_ = value;
+  }
+
+  // An application can attach arbitrary data to an RTP packet using
+  // `additional_data`. The additional data does not affect WebRTC processing.
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data() const {
+    return additional_data_;
+  }
+  void set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data) {
+    additional_data_ = std::move(data);
+  }
+
+ private:
+  webrtc::Timestamp arrival_time_ = Timestamp::MinusInfinity();
+  int payload_type_frequency_ = 0;
+  bool recovered_ = false;
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
new file mode 100644
index 0000000000..b55e74aaf0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+#include <cstdint>
+
+namespace webrtc {
+
+RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions)
+    : RtpPacket(extensions) {}
+RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions,
+                                 size_t capacity)
+    : RtpPacket(extensions, capacity) {}
+RtpPacketToSend::RtpPacketToSend(const RtpPacketToSend& packet) = default;
+RtpPacketToSend::RtpPacketToSend(RtpPacketToSend&& packet) = default;
+
+RtpPacketToSend& RtpPacketToSend::operator=(const RtpPacketToSend& packet) =
+    default;
+RtpPacketToSend& RtpPacketToSend::operator=(RtpPacketToSend&& packet) = default;
+
+RtpPacketToSend::~RtpPacketToSend() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
new file mode 100644
index 0000000000..438ca354ed
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_timing.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+
+namespace webrtc {
+// Class to hold rtp packet with metadata for sender side.
+// The metadata is not send over the wire, but packet sender may use it to
+// create rtp header extensions or other data that is sent over the wire.
+class RtpPacketToSend : public RtpPacket {
+ public:
+  // RtpPacketToSend::Type is deprecated. Use RtpPacketMediaType directly.
+  using Type = RtpPacketMediaType;
+
+  explicit RtpPacketToSend(const ExtensionManager* extensions);
+  RtpPacketToSend(const ExtensionManager* extensions, size_t capacity);
+  RtpPacketToSend(const RtpPacketToSend& packet);
+  RtpPacketToSend(RtpPacketToSend&& packet);
+
+  RtpPacketToSend& operator=(const RtpPacketToSend& packet);
+  RtpPacketToSend& operator=(RtpPacketToSend&& packet);
+
+  ~RtpPacketToSend();
+
+  // Time in local time base as close as it can to frame capture time.
+  webrtc::Timestamp capture_time() const { return capture_time_; }
+  void set_capture_time(webrtc::Timestamp time) { capture_time_ = time; }
+
+  void set_packet_type(RtpPacketMediaType type) { packet_type_ = type; }
+  absl::optional<RtpPacketMediaType> packet_type() const {
+    return packet_type_;
+  }
+
+  // If this is a retransmission, indicates the sequence number of the original
+  // media packet that this packet represents. If RTX is used this will likely
+  // be different from SequenceNumber().
+  void set_retransmitted_sequence_number(uint16_t sequence_number) {
+    retransmitted_sequence_number_ = sequence_number;
+  }
+  absl::optional<uint16_t> retransmitted_sequence_number() const {
+    return retransmitted_sequence_number_;
+  }
+
+  void set_allow_retransmission(bool allow_retransmission) {
+    allow_retransmission_ = allow_retransmission;
+  }
+  bool allow_retransmission() const { return allow_retransmission_; }
+
+  // An application can attach arbitrary data to an RTP packet using
+  // `additional_data`. The additional data does not affect WebRTC processing.
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data() const {
+    return additional_data_;
+  }
+  void set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data) {
+    additional_data_ = std::move(data);
+  }
+
+  void set_packetization_finish_time(webrtc::Timestamp time) {
+    SetExtension<VideoTimingExtension>(
+        VideoSendTiming::GetDeltaCappedMs(time - capture_time_),
+        VideoTimingExtension::kPacketizationFinishDeltaOffset);
+  }
+
+  void set_pacer_exit_time(webrtc::Timestamp time) {
+    SetExtension<VideoTimingExtension>(
+        VideoSendTiming::GetDeltaCappedMs(time - capture_time_),
+        VideoTimingExtension::kPacerExitDeltaOffset);
+  }
+
+  void set_network_time(webrtc::Timestamp time) {
+    SetExtension<VideoTimingExtension>(
+        VideoSendTiming::GetDeltaCappedMs(time - capture_time_),
+        VideoTimingExtension::kNetworkTimestampDeltaOffset);
+  }
+
+  void set_network2_time(webrtc::Timestamp time) {
+    SetExtension<VideoTimingExtension>(
+        VideoSendTiming::GetDeltaCappedMs(time - capture_time_),
+        VideoTimingExtension::kNetwork2TimestampDeltaOffset);
+  }
+
+  // Indicates if packet is the first packet of a video frame.
+  void set_first_packet_of_frame(bool is_first_packet) {
+    is_first_packet_of_frame_ = is_first_packet;
+  }
+  bool is_first_packet_of_frame() const { return is_first_packet_of_frame_; }
+
+  // Indicates if packet contains payload for a video key-frame.
+  void set_is_key_frame(bool is_key_frame) { is_key_frame_ = is_key_frame; }
+  bool is_key_frame() const { return is_key_frame_; }
+
+  // Indicates if packets should be protected by FEC (Forward Error Correction).
+  void set_fec_protect_packet(bool protect) { fec_protect_packet_ = protect; }
+  bool fec_protect_packet() const { return fec_protect_packet_; }
+
+  // Indicates if packet is using RED encapsulation, in accordance with
+  // https://tools.ietf.org/html/rfc2198
+  void set_is_red(bool is_red) { is_red_ = is_red; }
+  bool is_red() const { return is_red_; }
+
+  // The amount of time spent in the send queue, used for totalPacketSendDelay.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
+  void set_time_in_send_queue(TimeDelta time_in_send_queue) {
+    time_in_send_queue_ = time_in_send_queue;
+  }
+  absl::optional<TimeDelta> time_in_send_queue() const {
+    return time_in_send_queue_;
+  }
+
+ private:
+  webrtc::Timestamp capture_time_ = webrtc::Timestamp::Zero();
+  absl::optional<RtpPacketMediaType> packet_type_;
+  bool allow_retransmission_ = false;
+  absl::optional<uint16_t> retransmitted_sequence_number_;
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
+  bool is_first_packet_of_frame_ = false;
+  bool is_key_frame_ = false;
+  bool fec_protect_packet_ = false;
+  bool is_red_ = false;
+  absl::optional<TimeDelta> time_in_send_queue_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
new file mode 100644
index 0000000000..41bc114efb
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
@@ -0,0 +1,1285 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "common_video/test/utilities.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/random.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Each;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+
+constexpr int8_t kPayloadType = 100;
+constexpr uint32_t kSsrc = 0x12345678;
+constexpr uint16_t kSeqNum = 0x1234;
+constexpr uint8_t kSeqNumFirstByte = kSeqNum >> 8;
+constexpr uint8_t kSeqNumSecondByte = kSeqNum & 0xff;
+constexpr uint32_t kTimestamp = 0x65431278;
+constexpr uint8_t kTransmissionOffsetExtensionId = 1;
+constexpr uint8_t kDependencyDescriptorExtensionId = 2;
+constexpr uint8_t kAudioLevelExtensionId = 9;
+constexpr uint8_t kRtpStreamIdExtensionId = 0xa;
+constexpr uint8_t kRtpMidExtensionId = 0xb;
+constexpr uint8_t kVideoTimingExtensionId = 0xc;
+constexpr uint8_t kTwoByteExtensionId = 0xf0;
+constexpr int32_t kTimeOffset = 0x56ce;
+constexpr bool kVoiceActive = true;
+constexpr uint8_t kAudioLevel = 0x5a;
+constexpr char kStreamId[] = "streamid";
+constexpr char kMid[] = "mid";
+constexpr char kLongMid[] = "extra-long string to test two-byte header";
+constexpr size_t kMaxPaddingSize = 224u;
+// clang-format off
+constexpr uint8_t kMinimumPacket[] = {
+    0x80, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78};
+
+constexpr uint8_t kPacketWithTO[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x01,
+    0x12, 0x00, 0x56, 0xce};
+
+constexpr uint8_t kPacketWithTOAndAL[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x02,
+    0x12, 0x00, 0x56, 0xce,
+    0x90, 0x80|kAudioLevel, 0x00, 0x00};
+
+constexpr uint8_t kPacketWithTwoByteExtensionIdLast[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0x10, 0x00, 0x00, 0x04,
+    0x01, 0x03, 0x00, 0x56,
+    0xce, 0x09, 0x01, 0x80|kAudioLevel,
+    kTwoByteExtensionId, 0x03, 0x00, 0x30,  // => 0x00 0x30 0x22
+    0x22, 0x00, 0x00, 0x00};                // => Playout delay.min_ms = 3*10
+                                            // => Playout delay.max_ms = 34*10
+
+constexpr uint8_t kPacketWithTwoByteExtensionIdFirst[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0x10, 0x00, 0x00, 0x04,
+    kTwoByteExtensionId, 0x03, 0x00, 0x30,  // => 0x00 0x30 0x22
+    0x22, 0x01, 0x03, 0x00,                 // => Playout delay.min_ms = 3*10
+    0x56, 0xce, 0x09, 0x01,                 // => Playout delay.max_ms = 34*10
+    0x80|kAudioLevel, 0x00, 0x00, 0x00};
+
+constexpr uint8_t kPacketWithTOAndALInvalidPadding[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x03,
+    0x12, 0x00, 0x56, 0xce,
+    0x00, 0x02, 0x00, 0x00,  // 0x02 is invalid padding, parsing should stop.
+    0x90, 0x80|kAudioLevel, 0x00, 0x00};
+
+constexpr uint8_t kPacketWithTOAndALReservedExtensionId[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x03,
+    0x12, 0x00, 0x56, 0xce,
+    0x00, 0xF0, 0x00, 0x00,  // F is a reserved id, parsing should stop.
+    0x90, 0x80|kAudioLevel, 0x00, 0x00};
+
+constexpr uint8_t kPacketWithRsid[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x03,
+    0xa7, 's',  't',  'r',
+    'e',  'a',  'm',  'i',
+    'd' , 0x00, 0x00, 0x00};
+
+constexpr uint8_t kPacketWithMid[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x01,
+    0xb2, 'm', 'i', 'd'};
+
+constexpr uint8_t kCsrcAudioLevelExtensionId = 0xc;
+constexpr uint8_t kCsrcAudioLevelsSize = 4;
+constexpr uint8_t kCsrcAudioLevels[] = {0x7f, 0x00, 0x10, 0x08};
+constexpr uint8_t kPacketWithCsrcAudioLevels[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x02,
+    (kCsrcAudioLevelExtensionId << 4) | (kCsrcAudioLevelsSize - 1),
+          0x7f, 0x00, 0x10,
+    0x08, 0x00, 0x00, 0x00};
+
+constexpr uint32_t kCsrcs[] = {0x34567890, 0x32435465};
+constexpr uint8_t kPayload[] = {'p', 'a', 'y', 'l', 'o', 'a', 'd'};
+constexpr uint8_t kPacketPaddingSize = 8;
+constexpr uint8_t kPacket[] = {
+    0xb2, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0x34, 0x56, 0x78, 0x90,
+    0x32, 0x43, 0x54, 0x65,
+    0xbe, 0xde, 0x00, 0x01,
+    0x12, 0x00, 0x56, 0xce,
+    'p', 'a', 'y', 'l', 'o', 'a', 'd',
+    'p', 'a', 'd', 'd', 'i', 'n', 'g', kPacketPaddingSize};
+
+constexpr uint8_t kPacketWithTwoByteHeaderExtension[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0x10, 0x00, 0x00, 0x02,  // Two-byte header extension profile id + length.
+    kTwoByteExtensionId, 0x03, 0x00, 0x56,
+    0xce, 0x00, 0x00, 0x00};
+
+constexpr uint8_t kPacketWithLongTwoByteHeaderExtension[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0x10, 0x00, 0x00, 0x0B,  // Two-byte header extension profile id + length.
+    kTwoByteExtensionId, 0x29, 'e', 'x',
+    't', 'r', 'a', '-', 'l', 'o', 'n', 'g',
+    ' ', 's', 't', 'r', 'i', 'n', 'g', ' ',
+    't', 'o', ' ', 't', 'e', 's', 't', ' ',
+    't', 'w', 'o', '-', 'b', 'y', 't', 'e',
+    ' ', 'h', 'e', 'a', 'd', 'e', 'r', 0x00};
+
+constexpr uint8_t kPacketWithTwoByteHeaderExtensionWithPadding[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0x10, 0x00, 0x00, 0x03,  // Two-byte header extension profile id + length.
+    kTwoByteExtensionId, 0x03, 0x00, 0x56,
+    0xce, 0x00, 0x00, 0x00,  // Three padding bytes.
+    kAudioLevelExtensionId, 0x01, 0x80|kAudioLevel, 0x00};
+
+constexpr uint8_t kPacketWithInvalidExtension[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,  // kTimestamp.
+    0x12, 0x34, 0x56, 0x78,  // kSSrc.
+    0xbe, 0xde, 0x00, 0x02,  // Extension block of size 2 x 32bit words.
+    (kTransmissionOffsetExtensionId << 4) | 6,  // (6+1)-byte extension, but
+           'e',  'x',  't',                     // Transmission Offset
+     'd',  'a',  't',  'a',                     // expected to be 3-bytes.
+     'p',  'a',  'y',  'l',  'o',  'a',  'd'};
+
+constexpr uint8_t kPacketWithLegacyTimingExtension[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,  // kTimestamp.
+    0x12, 0x34, 0x56, 0x78,  // kSSrc.
+    0xbe, 0xde, 0x00, 0x04,    // Extension block of size 4 x 32bit words.
+    (kVideoTimingExtensionId << 4)
+      | VideoTimingExtension::kValueSizeBytes - 2,  // Old format without flags.
+          0x00, 0x01, 0x00,
+    0x02, 0x00, 0x03, 0x00,
+    0x04, 0x00, 0x00, 0x00,
+    0x00, 0x00, 0x00, 0x00};
+// clang-format on
+
+void TestCreateAndParseColorSpaceExtension(bool with_hdr_metadata) {
+  // Create packet with extension.
+  RtpPacket::ExtensionManager extensions(/*extmap-allow-mixed=*/true);
+  extensions.Register<ColorSpaceExtension>(1);
+  RtpPacket packet(&extensions);
+  const ColorSpace kColorSpace = CreateTestColorSpace(with_hdr_metadata);
+  EXPECT_TRUE(packet.SetExtension<ColorSpaceExtension>(kColorSpace));
+  packet.SetPayloadSize(42);
+
+  // Read packet with the extension.
+  RtpPacketReceived parsed(&extensions);
+  EXPECT_TRUE(parsed.Parse(packet.Buffer()));
+  ColorSpace parsed_color_space;
+  EXPECT_TRUE(parsed.GetExtension<ColorSpaceExtension>(&parsed_color_space));
+  EXPECT_EQ(kColorSpace, parsed_color_space);
+}
+
+TEST(RtpPacketTest, CreateMinimum) {
+  RtpPacketToSend packet(nullptr);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  EXPECT_THAT(kMinimumPacket, ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, CreateWithExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  EXPECT_THAT(kPacketWithTO, ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, CreateWith2Extensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel);
+  EXPECT_THAT(kPacketWithTOAndAL,
+              ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, CreateWithTwoByteHeaderExtensionFirst) {
+  RtpPacketToSend::ExtensionManager extensions(true);
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  extensions.Register<PlayoutDelayLimits>(kTwoByteExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  // Set extension that requires two-byte header.
+  VideoPlayoutDelay playoutDelay = {30, 340};
+  ASSERT_TRUE(packet.SetExtension<PlayoutDelayLimits>(playoutDelay));
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel);
+  EXPECT_THAT(kPacketWithTwoByteExtensionIdFirst,
+              ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, CreateWithTwoByteHeaderExtensionLast) {
+  // This test will trigger RtpPacket::PromoteToTwoByteHeaderExtension().
+  RtpPacketToSend::ExtensionManager extensions(true);
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  extensions.Register<PlayoutDelayLimits>(kTwoByteExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel);
+  EXPECT_THAT(kPacketWithTOAndAL,
+              ElementsAreArray(packet.data(), packet.size()));
+  // Set extension that requires two-byte header.
+  VideoPlayoutDelay playoutDelay = {30, 340};
+  ASSERT_TRUE(packet.SetExtension<PlayoutDelayLimits>(playoutDelay));
+  EXPECT_THAT(kPacketWithTwoByteExtensionIdLast,
+              ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, CreateWithDynamicSizedExtensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<RtpStreamId>(kRtpStreamIdExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<RtpStreamId>(kStreamId);
+  EXPECT_THAT(kPacketWithRsid, ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, TryToCreateWithEmptyRsid) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<RtpStreamId>(kRtpStreamIdExtensionId);
+  RtpPacketToSend packet(&extensions);
+  EXPECT_FALSE(packet.SetExtension<RtpStreamId>(""));
+}
+
+TEST(RtpPacketTest, TryToCreateWithLongRsid) {
+  RtpPacketToSend::ExtensionManager extensions;
+  constexpr char kLongStreamId[] = "LoooooooooongRsid";
+  ASSERT_EQ(strlen(kLongStreamId), 17u);
+  extensions.Register<RtpStreamId>(kRtpStreamIdExtensionId);
+  RtpPacketToSend packet(&extensions);
+  EXPECT_FALSE(packet.SetExtension<RtpStreamId>(kLongStreamId));
+}
+
+TEST(RtpPacketTest, TryToCreateWithEmptyMid) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<RtpMid>(kRtpMidExtensionId);
+  RtpPacketToSend packet(&extensions);
+  EXPECT_FALSE(packet.SetExtension<RtpMid>(""));
+}
+
+TEST(RtpPacketTest, TryToCreateWithLongMid) {
+  RtpPacketToSend::ExtensionManager extensions;
+  constexpr char kLongMid[] = "LoooooooooonogMid";
+  ASSERT_EQ(strlen(kLongMid), 17u);
+  extensions.Register<RtpMid>(kRtpMidExtensionId);
+  RtpPacketToSend packet(&extensions);
+  EXPECT_FALSE(packet.SetExtension<RtpMid>(kLongMid));
+}
+
+TEST(RtpPacketTest, TryToCreateTwoByteHeaderNotSupported) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<AudioLevel>(kTwoByteExtensionId);
+  RtpPacketToSend packet(&extensions);
+  // Set extension that requires two-byte header.
+  EXPECT_FALSE(packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel));
+}
+
+TEST(RtpPacketTest, CreateWithMaxSizeHeaderExtension) {
+  const std::string kValue = "123456789abcdef";
+  RtpPacket::ExtensionManager extensions;
+  extensions.Register<RtpMid>(1);
+  extensions.Register<RtpStreamId>(2);
+
+  RtpPacket packet(&extensions);
+  EXPECT_TRUE(packet.SetExtension<RtpMid>(kValue));
+
+  packet.SetPayloadSize(42);
+  // Rewriting allocated extension is allowed.
+  EXPECT_TRUE(packet.SetExtension<RtpMid>(kValue));
+  // Adding another extension after payload is set is not allowed.
+  EXPECT_FALSE(packet.SetExtension<RtpStreamId>(kValue));
+
+  // Read packet with the extension.
+  RtpPacketReceived parsed(&extensions);
+  EXPECT_TRUE(parsed.Parse(packet.Buffer()));
+  std::string read;
+  EXPECT_TRUE(parsed.GetExtension<RtpMid>(&read));
+  EXPECT_EQ(read, kValue);
+}
+
+TEST(RtpPacketTest, SetsRegisteredExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  RtpPacketToSend packet(&extensions);
+
+  EXPECT_TRUE(packet.IsRegistered<TransmissionOffset>());
+  EXPECT_FALSE(packet.HasExtension<TransmissionOffset>());
+
+  // Try to set the extensions.
+  EXPECT_TRUE(packet.SetExtension<TransmissionOffset>(kTimeOffset));
+
+  EXPECT_TRUE(packet.HasExtension<TransmissionOffset>());
+  EXPECT_EQ(packet.GetExtension<TransmissionOffset>(), kTimeOffset);
+}
+
+TEST(RtpPacketTest, FailsToSetUnregisteredExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  RtpPacketToSend packet(&extensions);
+
+  EXPECT_FALSE(packet.IsRegistered<TransportSequenceNumber>());
+  EXPECT_FALSE(packet.HasExtension<TransportSequenceNumber>());
+
+  EXPECT_FALSE(packet.SetExtension<TransportSequenceNumber>(42));
+
+  EXPECT_FALSE(packet.HasExtension<TransportSequenceNumber>());
+  EXPECT_EQ(packet.GetExtension<TransportSequenceNumber>(), absl::nullopt);
+}
+
+TEST(RtpPacketTest, CreateWithDynamicSizedExtensionCsrcAudioLevel) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<CsrcAudioLevel>(kCsrcAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  CsrcAudioLevelList levels;
+  levels.numAudioLevels = kCsrcAudioLevelsSize;
+  for (uint8_t i = 0; i < kCsrcAudioLevelsSize; i++) {
+    levels.arrOfAudioLevels[i] = kCsrcAudioLevels[i];
+  }
+  packet.SetExtension<CsrcAudioLevel>(levels);
+  EXPECT_THAT(kPacketWithCsrcAudioLevels,
+    ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, SetReservedExtensionsAfterPayload) {
+  const size_t kPayloadSize = 4;
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+
+  EXPECT_TRUE(packet.ReserveExtension<TransmissionOffset>());
+  packet.SetPayloadSize(kPayloadSize);
+  // Can't set extension after payload.
+  EXPECT_FALSE(packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel));
+  // Unless reserved.
+  EXPECT_TRUE(packet.SetExtension<TransmissionOffset>(kTimeOffset));
+}
+
+TEST(RtpPacketTest, CreatePurePadding) {
+  const size_t kPaddingSize = kMaxPaddingSize - 1;
+  RtpPacketToSend packet(nullptr, 12 + kPaddingSize);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+
+  EXPECT_LT(packet.size(), packet.capacity());
+  EXPECT_FALSE(packet.SetPadding(kPaddingSize + 1));
+  EXPECT_TRUE(packet.SetPadding(kPaddingSize));
+  EXPECT_EQ(packet.size(), packet.capacity());
+}
+
+TEST(RtpPacketTest, CreateUnalignedPadding) {
+  const size_t kPayloadSize = 3;  // Make padding start at unaligned address.
+  RtpPacketToSend packet(nullptr, 12 + kPayloadSize + kMaxPaddingSize);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetPayloadSize(kPayloadSize);
+
+  EXPECT_LT(packet.size(), packet.capacity());
+  EXPECT_TRUE(packet.SetPadding(kMaxPaddingSize));
+  EXPECT_EQ(packet.size(), packet.capacity());
+}
+
+TEST(RtpPacketTest, WritesPaddingSizeToLastByte) {
+  const size_t kPaddingSize = 5;
+  RtpPacket packet;
+
+  EXPECT_TRUE(packet.SetPadding(kPaddingSize));
+  EXPECT_EQ(packet.data()[packet.size() - 1], kPaddingSize);
+}
+
+TEST(RtpPacketTest, UsesZerosForPadding) {
+  const size_t kPaddingSize = 5;
+  RtpPacket packet;
+
+  EXPECT_TRUE(packet.SetPadding(kPaddingSize));
+  EXPECT_THAT(rtc::MakeArrayView(packet.data() + 12, kPaddingSize - 1),
+              Each(0));
+}
+
+TEST(RtpPacketTest, CreateOneBytePadding) {
+  size_t kPayloadSize = 123;
+  RtpPacket packet(nullptr, 12 + kPayloadSize + 1);
+  packet.SetPayloadSize(kPayloadSize);
+
+  EXPECT_TRUE(packet.SetPadding(1));
+
+  EXPECT_EQ(packet.size(), 12 + kPayloadSize + 1);
+  EXPECT_EQ(packet.padding_size(), 1u);
+}
+
+TEST(RtpPacketTest, FailsToAddPaddingWithoutCapacity) {
+  size_t kPayloadSize = 123;
+  RtpPacket packet(nullptr, 12 + kPayloadSize);
+  packet.SetPayloadSize(kPayloadSize);
+
+  EXPECT_FALSE(packet.SetPadding(1));
+}
+
+TEST(RtpPacketTest, ParseMinimum) {
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(kMinimumPacket, sizeof(kMinimumPacket)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  EXPECT_EQ(0u, packet.padding_size());
+  EXPECT_EQ(0u, packet.payload_size());
+}
+
+TEST(RtpPacketTest, ParseBuffer) {
+  rtc::CopyOnWriteBuffer unparsed(kMinimumPacket);
+  const uint8_t* raw = unparsed.data();
+
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(std::move(unparsed)));
+  EXPECT_EQ(raw, packet.data());  // Expect packet take the buffer without copy.
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  EXPECT_EQ(0u, packet.padding_size());
+  EXPECT_EQ(0u, packet.payload_size());
+}
+
+TEST(RtpPacketTest, ParseWithExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  EXPECT_EQ(0u, packet.payload_size());
+  EXPECT_EQ(0u, packet.padding_size());
+}
+
+TEST(RtpPacketTest, ParseHeaderOnly) {
+  // clang-format off
+  constexpr uint8_t kPaddingHeader[] = {
+      0x80, 0x62, 0x35, 0x79,
+      0x65, 0x43, 0x12, 0x78,
+      0x12, 0x34, 0x56, 0x78};
+  // clang-format on
+
+  RtpPacket packet;
+  EXPECT_TRUE(packet.Parse(rtc::CopyOnWriteBuffer(kPaddingHeader)));
+  EXPECT_EQ(packet.PayloadType(), 0x62u);
+  EXPECT_EQ(packet.SequenceNumber(), 0x3579u);
+  EXPECT_EQ(packet.Timestamp(), 0x65431278u);
+  EXPECT_EQ(packet.Ssrc(), 0x12345678u);
+
+  EXPECT_FALSE(packet.has_padding());
+  EXPECT_EQ(packet.padding_size(), 0u);
+  EXPECT_EQ(packet.payload_size(), 0u);
+}
+
+TEST(RtpPacketTest, ParseHeaderOnlyWithPadding) {
+  // clang-format off
+  constexpr uint8_t kPaddingHeader[] = {
+      0xa0, 0x62, 0x35, 0x79,
+      0x65, 0x43, 0x12, 0x78,
+      0x12, 0x34, 0x56, 0x78};
+  // clang-format on
+
+  RtpPacket packet;
+  EXPECT_TRUE(packet.Parse(rtc::CopyOnWriteBuffer(kPaddingHeader)));
+
+  EXPECT_TRUE(packet.has_padding());
+  EXPECT_EQ(packet.padding_size(), 0u);
+  EXPECT_EQ(packet.payload_size(), 0u);
+}
+
+TEST(RtpPacketTest, ParseHeaderOnlyWithExtensionAndPadding) {
+  // clang-format off
+  constexpr uint8_t kPaddingHeader[] = {
+      0xb0, 0x62, 0x35, 0x79,
+      0x65, 0x43, 0x12, 0x78,
+      0x12, 0x34, 0x56, 0x78,
+      0xbe, 0xde, 0x00, 0x01,
+      0x11, 0x00, 0x00, 0x00};
+  // clang-format on
+
+  RtpHeaderExtensionMap extensions;
+  extensions.Register<TransmissionOffset>(1);
+  RtpPacket packet(&extensions);
+  EXPECT_TRUE(packet.Parse(rtc::CopyOnWriteBuffer(kPaddingHeader)));
+  EXPECT_TRUE(packet.has_padding());
+  EXPECT_TRUE(packet.HasExtension<TransmissionOffset>());
+  EXPECT_EQ(packet.padding_size(), 0u);
+}
+
+TEST(RtpPacketTest, ParsePaddingOnlyPacket) {
+  // clang-format off
+  constexpr uint8_t kPaddingHeader[] = {
+      0xa0, 0x62, 0x35, 0x79,
+      0x65, 0x43, 0x12, 0x78,
+      0x12, 0x34, 0x56, 0x78,
+      0, 0, 3};
+  // clang-format on
+
+  RtpPacket packet;
+  EXPECT_TRUE(packet.Parse(rtc::CopyOnWriteBuffer(kPaddingHeader)));
+  EXPECT_TRUE(packet.has_padding());
+  EXPECT_EQ(packet.padding_size(), 3u);
+}
+
+TEST(RtpPacketTest, GetExtensionWithoutParametersReturnsOptionalValue) {
+  RtpPacket::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<RtpStreamId>(kRtpStreamIdExtensionId);
+
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
+
+  auto time_offset = packet.GetExtension<TransmissionOffset>();
+  static_assert(
+      std::is_same<decltype(time_offset),
+                   absl::optional<TransmissionOffset::value_type>>::value,
+      "");
+  EXPECT_EQ(time_offset, kTimeOffset);
+  EXPECT_FALSE(packet.GetExtension<RtpStreamId>().has_value());
+}
+
+TEST(RtpPacketTest, GetRawExtensionWhenPresent) {
+  constexpr uint8_t kRawPacket[] = {
+      // comment for clang-format to align kRawPacket nicer.
+      0x90, 100,  0x5e, 0x04,  //
+      0x65, 0x43, 0x12, 0x78,  // Timestamp.
+      0x12, 0x34, 0x56, 0x78,  // Ssrc
+      0xbe, 0xde, 0x00, 0x01,  // Extension header
+      0x12, 'm',  'i',  'd',   // 3-byte extension with id=1.
+      'p',  'a',  'y',  'l',  'o', 'a', 'd'};
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<RtpMid>(1);
+  RtpPacket packet(&extensions);
+  ASSERT_TRUE(packet.Parse(kRawPacket, sizeof(kRawPacket)));
+  EXPECT_THAT(packet.GetRawExtension<RtpMid>(), ElementsAre('m', 'i', 'd'));
+}
+
+TEST(RtpPacketTest, GetRawExtensionWhenAbsent) {
+  constexpr uint8_t kRawPacket[] = {
+      // comment for clang-format to align kRawPacket nicer.
+      0x90, 100,  0x5e, 0x04,  //
+      0x65, 0x43, 0x12, 0x78,  // Timestamp.
+      0x12, 0x34, 0x56, 0x78,  // Ssrc
+      0xbe, 0xde, 0x00, 0x01,  // Extension header
+      0x12, 'm',  'i',  'd',   // 3-byte extension with id=1.
+      'p',  'a',  'y',  'l',  'o', 'a', 'd'};
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<RtpMid>(2);
+  RtpPacket packet(&extensions);
+  ASSERT_TRUE(packet.Parse(kRawPacket, sizeof(kRawPacket)));
+  EXPECT_THAT(packet.GetRawExtension<RtpMid>(), IsEmpty());
+}
+
+TEST(RtpPacketTest, ParseWithInvalidSizedExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithInvalidExtension,
+                           sizeof(kPacketWithInvalidExtension)));
+
+  // Extension should be ignored.
+  int32_t time_offset;
+  EXPECT_FALSE(packet.GetExtension<TransmissionOffset>(&time_offset));
+
+  // But shouldn't prevent reading payload.
+  EXPECT_THAT(packet.payload(), ElementsAreArray(kPayload));
+}
+
+TEST(RtpPacketTest, ParseWithOverSizedExtension) {
+  // clang-format off
+  const uint8_t bad_packet[] = {
+      0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+      0x65, 0x43, 0x12, 0x78,  // kTimestamp.
+      0x12, 0x34, 0x56, 0x78,  // kSsrc.
+      0xbe, 0xde, 0x00, 0x01,  // Extension of size 1x32bit word.
+      0x00,  // Add a byte of padding.
+            0x12,  // Extension id 1 size (2+1).
+                  0xda, 0x1a  // Only 2 bytes of extension payload.
+  };
+  // clang-format on
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(1);
+  RtpPacketReceived packet(&extensions);
+
+  // Parse should ignore bad extension and proceed.
+  EXPECT_TRUE(packet.Parse(bad_packet, sizeof(bad_packet)));
+  int32_t time_offset;
+  // But extracting extension should fail.
+  EXPECT_FALSE(packet.GetExtension<TransmissionOffset>(&time_offset));
+}
+
+TEST(RtpPacketTest, ParseWith2Extensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTOAndAL, sizeof(kPacketWithTOAndAL)));
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  bool voice_active;
+  uint8_t audio_level;
+  EXPECT_TRUE(packet.GetExtension<AudioLevel>(&voice_active, &audio_level));
+  EXPECT_EQ(kVoiceActive, voice_active);
+  EXPECT_EQ(kAudioLevel, audio_level);
+}
+
+TEST(RtpPacketTest, ParseSecondPacketWithFewerExtensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTOAndAL, sizeof(kPacketWithTOAndAL)));
+  EXPECT_TRUE(packet.HasExtension<TransmissionOffset>());
+  EXPECT_TRUE(packet.HasExtension<AudioLevel>());
+
+  // Second packet without audio level.
+  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
+  EXPECT_TRUE(packet.HasExtension<TransmissionOffset>());
+  EXPECT_FALSE(packet.HasExtension<AudioLevel>());
+}
+
+TEST(RtpPacketTest, ParseWith2ExtensionsInvalidPadding) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTOAndALInvalidPadding,
+                           sizeof(kPacketWithTOAndALInvalidPadding)));
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  bool voice_active;
+  uint8_t audio_level;
+  EXPECT_FALSE(packet.GetExtension<AudioLevel>(&voice_active, &audio_level));
+}
+
+TEST(RtpPacketTest, ParseWith2ExtensionsReservedExtensionId) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTOAndALReservedExtensionId,
+                           sizeof(kPacketWithTOAndALReservedExtensionId)));
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  bool voice_active;
+  uint8_t audio_level;
+  EXPECT_FALSE(packet.GetExtension<AudioLevel>(&voice_active, &audio_level));
+}
+
+TEST(RtpPacketTest, ParseWithAllFeatures) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacket, sizeof(kPacket)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  EXPECT_THAT(packet.Csrcs(), ElementsAreArray(kCsrcs));
+  EXPECT_THAT(packet.payload(), ElementsAreArray(kPayload));
+  EXPECT_EQ(kPacketPaddingSize, packet.padding_size());
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+}
+
+TEST(RtpPacketTest, ParseTwoByteHeaderExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTwoByteExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTwoByteHeaderExtension,
+                           sizeof(kPacketWithTwoByteHeaderExtension)));
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+}
+
+TEST(RtpPacketTest, ParseLongTwoByteHeaderExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<RtpMid>(kTwoByteExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithLongTwoByteHeaderExtension,
+                           sizeof(kPacketWithLongTwoByteHeaderExtension)));
+  std::string long_rtp_mid;
+  EXPECT_TRUE(packet.GetExtension<RtpMid>(&long_rtp_mid));
+  EXPECT_EQ(kLongMid, long_rtp_mid);
+}
+
+TEST(RtpPacketTest, ParseTwoByteHeaderExtensionWithPadding) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTwoByteExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(
+      packet.Parse(kPacketWithTwoByteHeaderExtensionWithPadding,
+                   sizeof(kPacketWithTwoByteHeaderExtensionWithPadding)));
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  bool voice_active;
+  uint8_t audio_level;
+  EXPECT_TRUE(packet.GetExtension<AudioLevel>(&voice_active, &audio_level));
+  EXPECT_EQ(kVoiceActive, voice_active);
+  EXPECT_EQ(kAudioLevel, audio_level);
+}
+
+TEST(RtpPacketTest, ParseWithExtensionDelayed) {
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+
+  int32_t time_offset;
+  EXPECT_FALSE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  packet.IdentifyExtensions(extensions);
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  EXPECT_EQ(0u, packet.payload_size());
+  EXPECT_EQ(0u, packet.padding_size());
+}
+
+TEST(RtpPacketTest, ParseDynamicSizeExtension) {
+  // clang-format off
+  const uint8_t kPacket1[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,  // Timestamp.
+    0x12, 0x34, 0x56, 0x78,  // Ssrc.
+    0xbe, 0xde, 0x00, 0x02,  // Extensions block of size 2x32bit words.
+    0x21, 'H', 'D',          // Extension with id = 2, size = (1+1).
+    0x12, 'r', 't', 'x',     // Extension with id = 1, size = (2+1).
+    0x00};  // Extension padding.
+  const uint8_t kPacket2[] = {
+    0x90, kPayloadType, kSeqNumFirstByte, kSeqNumSecondByte,
+    0x65, 0x43, 0x12, 0x78,  // Timestamp.
+    0x12, 0x34, 0x56, 0x79,  // Ssrc.
+    0xbe, 0xde, 0x00, 0x01,  // Extensions block of size 1x32bit words.
+    0x11, 'H', 'D',          // Extension with id = 1, size = (1+1).
+    0x00};  // Extension padding.
+  // clang-format on
+  RtpPacketReceived::ExtensionManager extensions;
+  extensions.Register<RtpStreamId>(1);
+  extensions.Register<RepairedRtpStreamId>(2);
+  RtpPacketReceived packet(&extensions);
+  ASSERT_TRUE(packet.Parse(kPacket1, sizeof(kPacket1)));
+
+  std::string rsid;
+  EXPECT_TRUE(packet.GetExtension<RtpStreamId>(&rsid));
+  EXPECT_EQ(rsid, "rtx");
+
+  std::string repaired_rsid;
+  EXPECT_TRUE(packet.GetExtension<RepairedRtpStreamId>(&repaired_rsid));
+  EXPECT_EQ(repaired_rsid, "HD");
+
+  // Parse another packet with RtpStreamId extension of different size.
+  ASSERT_TRUE(packet.Parse(kPacket2, sizeof(kPacket2)));
+  EXPECT_TRUE(packet.GetExtension<RtpStreamId>(&rsid));
+  EXPECT_EQ(rsid, "HD");
+  EXPECT_FALSE(packet.GetExtension<RepairedRtpStreamId>(&repaired_rsid));
+}
+
+TEST(RtpPacketTest, ParseWithMid) {
+  RtpPacketReceived::ExtensionManager extensions;
+  extensions.Register<RtpMid>(kRtpMidExtensionId);
+  RtpPacketReceived packet(&extensions);
+  ASSERT_TRUE(packet.Parse(kPacketWithMid, sizeof(kPacketWithMid)));
+
+  std::string mid;
+  EXPECT_TRUE(packet.GetExtension<RtpMid>(&mid));
+  EXPECT_EQ(mid, kMid);
+}
+
+struct UncopyableValue {
+  UncopyableValue() = default;
+  UncopyableValue(const UncopyableValue&) = delete;
+  UncopyableValue& operator=(const UncopyableValue&) = delete;
+};
+struct UncopyableExtension {
+  static constexpr RTPExtensionType kId = kRtpExtensionDependencyDescriptor;
+  static constexpr absl::string_view Uri() { return "uri"; }
+
+  static size_t ValueSize(const UncopyableValue& value) { return 1; }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    const UncopyableValue& value) {
+    return true;
+  }
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    UncopyableValue* value) {
+    return true;
+  }
+};
+
+TEST(RtpPacketTest, SetUncopyableExtension) {
+  RtpPacket::ExtensionManager extensions;
+  extensions.Register<UncopyableExtension>(1);
+  RtpPacket rtp_packet(&extensions);
+
+  UncopyableValue value;
+  EXPECT_TRUE(rtp_packet.SetExtension<UncopyableExtension>(value));
+}
+
+TEST(RtpPacketTest, GetUncopyableExtension) {
+  RtpPacket::ExtensionManager extensions;
+  extensions.Register<UncopyableExtension>(1);
+  RtpPacket rtp_packet(&extensions);
+  UncopyableValue value;
+  rtp_packet.SetExtension<UncopyableExtension>(value);
+
+  UncopyableValue value2;
+  EXPECT_TRUE(rtp_packet.GetExtension<UncopyableExtension>(&value2));
+}
+
+TEST(RtpPacketTest, CreateAndParseTimingFrameExtension) {
+  // Create a packet with video frame timing extension populated.
+  RtpPacketToSend::ExtensionManager send_extensions;
+  send_extensions.Register<VideoTimingExtension>(kVideoTimingExtensionId);
+  RtpPacketToSend send_packet(&send_extensions);
+  send_packet.SetPayloadType(kPayloadType);
+  send_packet.SetSequenceNumber(kSeqNum);
+  send_packet.SetTimestamp(kTimestamp);
+  send_packet.SetSsrc(kSsrc);
+
+  VideoSendTiming timing;
+  timing.encode_start_delta_ms = 1;
+  timing.encode_finish_delta_ms = 2;
+  timing.packetization_finish_delta_ms = 3;
+  timing.pacer_exit_delta_ms = 4;
+  timing.flags =
+      VideoSendTiming::kTriggeredByTimer | VideoSendTiming::kTriggeredBySize;
+
+  send_packet.SetExtension<VideoTimingExtension>(timing);
+
+  // Serialize the packet and then parse it again.
+  RtpPacketReceived::ExtensionManager extensions;
+  extensions.Register<VideoTimingExtension>(kVideoTimingExtensionId);
+  RtpPacketReceived receive_packet(&extensions);
+  EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
+
+  VideoSendTiming receivied_timing;
+  EXPECT_TRUE(
+      receive_packet.GetExtension<VideoTimingExtension>(&receivied_timing));
+
+  // Only check first and last timestamp (covered by other tests) plus flags.
+  EXPECT_EQ(receivied_timing.encode_start_delta_ms,
+            timing.encode_start_delta_ms);
+  EXPECT_EQ(receivied_timing.pacer_exit_delta_ms, timing.pacer_exit_delta_ms);
+  EXPECT_EQ(receivied_timing.flags, timing.flags);
+}
+
+TEST(RtpPacketTest, ParseLegacyTimingFrameExtension) {
+  // Parse the modified packet.
+  RtpPacketReceived::ExtensionManager extensions;
+  extensions.Register<VideoTimingExtension>(kVideoTimingExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithLegacyTimingExtension,
+                           sizeof(kPacketWithLegacyTimingExtension)));
+  VideoSendTiming receivied_timing;
+  EXPECT_TRUE(packet.GetExtension<VideoTimingExtension>(&receivied_timing));
+
+  // Check first and last timestamp are still OK. Flags should now be 0.
+  EXPECT_EQ(receivied_timing.encode_start_delta_ms, 1);
+  EXPECT_EQ(receivied_timing.pacer_exit_delta_ms, 4);
+  EXPECT_EQ(receivied_timing.flags, 0);
+}
+
+TEST(RtpPacketTest, CreateAndParseColorSpaceExtension) {
+  TestCreateAndParseColorSpaceExtension(/*with_hdr_metadata=*/true);
+}
+
+TEST(RtpPacketTest, CreateAndParseColorSpaceExtensionWithoutHdrMetadata) {
+  TestCreateAndParseColorSpaceExtension(/*with_hdr_metadata=*/false);
+}
+
+TEST(RtpPacketTest, CreateAndParseAbsoluteCaptureTime) {
+  // Create a packet with absolute capture time extension populated.
+  RtpPacketToSend::ExtensionManager extensions;
+  constexpr int kExtensionId = 1;
+  extensions.Register<AbsoluteCaptureTimeExtension>(kExtensionId);
+  RtpPacketToSend send_packet(&extensions);
+  send_packet.SetPayloadType(kPayloadType);
+  send_packet.SetSequenceNumber(kSeqNum);
+  send_packet.SetTimestamp(kTimestamp);
+  send_packet.SetSsrc(kSsrc);
+
+  constexpr AbsoluteCaptureTime kAbsoluteCaptureTime{
+      /*absolute_capture_timestamp=*/9876543210123456789ULL,
+      /*estimated_capture_clock_offset=*/-1234567890987654321LL};
+  send_packet.SetExtension<AbsoluteCaptureTimeExtension>(kAbsoluteCaptureTime);
+
+  // Serialize the packet and then parse it again.
+  RtpPacketReceived receive_packet(&extensions);
+  EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
+
+  AbsoluteCaptureTime received_absolute_capture_time;
+  EXPECT_TRUE(receive_packet.GetExtension<AbsoluteCaptureTimeExtension>(
+      &received_absolute_capture_time));
+  EXPECT_EQ(kAbsoluteCaptureTime.absolute_capture_timestamp,
+            received_absolute_capture_time.absolute_capture_timestamp);
+  EXPECT_EQ(kAbsoluteCaptureTime.estimated_capture_clock_offset,
+            received_absolute_capture_time.estimated_capture_clock_offset);
+}
+
+TEST(RtpPacketTest,
+     CreateAndParseAbsoluteCaptureTimeWithoutEstimatedCaptureClockOffset) {
+  // Create a packet with absolute capture time extension populated.
+  RtpPacketToSend::ExtensionManager extensions;
+  constexpr int kExtensionId = 1;
+  extensions.Register<AbsoluteCaptureTimeExtension>(kExtensionId);
+  RtpPacketToSend send_packet(&extensions);
+  send_packet.SetPayloadType(kPayloadType);
+  send_packet.SetSequenceNumber(kSeqNum);
+  send_packet.SetTimestamp(kTimestamp);
+  send_packet.SetSsrc(kSsrc);
+
+  constexpr AbsoluteCaptureTime kAbsoluteCaptureTime{
+      /*absolute_capture_timestamp=*/9876543210123456789ULL,
+      /*estimated_capture_clock_offset=*/absl::nullopt};
+  send_packet.SetExtension<AbsoluteCaptureTimeExtension>(kAbsoluteCaptureTime);
+
+  // Serialize the packet and then parse it again.
+  RtpPacketReceived receive_packet(&extensions);
+  EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
+
+  AbsoluteCaptureTime received_absolute_capture_time;
+  EXPECT_TRUE(receive_packet.GetExtension<AbsoluteCaptureTimeExtension>(
+      &received_absolute_capture_time));
+  EXPECT_EQ(kAbsoluteCaptureTime.absolute_capture_timestamp,
+            received_absolute_capture_time.absolute_capture_timestamp);
+  EXPECT_EQ(kAbsoluteCaptureTime.estimated_capture_clock_offset,
+            received_absolute_capture_time.estimated_capture_clock_offset);
+}
+
+TEST(RtpPacketTest, CreateAndParseTransportSequenceNumber) {
+  // Create a packet with transport sequence number extension populated.
+  RtpPacketToSend::ExtensionManager extensions;
+  constexpr int kExtensionId = 1;
+  extensions.Register<TransportSequenceNumber>(kExtensionId);
+  RtpPacketToSend send_packet(&extensions);
+  send_packet.SetPayloadType(kPayloadType);
+  send_packet.SetSequenceNumber(kSeqNum);
+  send_packet.SetTimestamp(kTimestamp);
+  send_packet.SetSsrc(kSsrc);
+
+  constexpr int kTransportSequenceNumber = 12345;
+  send_packet.SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+
+  // Serialize the packet and then parse it again.
+  RtpPacketReceived receive_packet(&extensions);
+  EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
+
+  uint16_t received_transport_sequeunce_number;
+  EXPECT_TRUE(receive_packet.GetExtension<TransportSequenceNumber>(
+      &received_transport_sequeunce_number));
+  EXPECT_EQ(received_transport_sequeunce_number, kTransportSequenceNumber);
+}
+
+TEST(RtpPacketTest, CreateAndParseTransportSequenceNumberV2) {
+  // Create a packet with transport sequence number V2 extension populated.
+  // No feedback request means that the extension will be two bytes unless it's
+  // pre-allocated.
+  RtpPacketToSend::ExtensionManager extensions;
+  constexpr int kExtensionId = 1;
+  extensions.Register<TransportSequenceNumberV2>(kExtensionId);
+  RtpPacketToSend send_packet(&extensions);
+  send_packet.SetPayloadType(kPayloadType);
+  send_packet.SetSequenceNumber(kSeqNum);
+  send_packet.SetTimestamp(kTimestamp);
+  send_packet.SetSsrc(kSsrc);
+
+  constexpr int kTransportSequenceNumber = 12345;
+  send_packet.SetExtension<TransportSequenceNumberV2>(kTransportSequenceNumber,
+                                                      absl::nullopt);
+  EXPECT_EQ(send_packet.GetRawExtension<TransportSequenceNumberV2>().size(),
+            2u);
+
+  // Serialize the packet and then parse it again.
+  RtpPacketReceived receive_packet(&extensions);
+  EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
+
+  uint16_t received_transport_sequeunce_number;
+  absl::optional<FeedbackRequest> received_feedback_request;
+  EXPECT_TRUE(receive_packet.GetExtension<TransportSequenceNumberV2>(
+      &received_transport_sequeunce_number, &received_feedback_request));
+  EXPECT_EQ(received_transport_sequeunce_number, kTransportSequenceNumber);
+  EXPECT_FALSE(received_feedback_request);
+}
+
+TEST(RtpPacketTest, CreateAndParseTransportSequenceNumberV2Preallocated) {
+  // Create a packet with transport sequence number V2 extension populated.
+  // No feedback request means that the extension could be two bytes, but since
+  // it's pre-allocated we don't know if it is with or without feedback request
+  // therefore the size is four bytes.
+  RtpPacketToSend::ExtensionManager extensions;
+  constexpr int kExtensionId = 1;
+  extensions.Register<TransportSequenceNumberV2>(kExtensionId);
+  RtpPacketToSend send_packet(&extensions);
+  send_packet.SetPayloadType(kPayloadType);
+  send_packet.SetSequenceNumber(kSeqNum);
+  send_packet.SetTimestamp(kTimestamp);
+  send_packet.SetSsrc(kSsrc);
+
+  constexpr int kTransportSequenceNumber = 12345;
+  constexpr absl::optional<FeedbackRequest> kNoFeedbackRequest =
+      FeedbackRequest{/*include_timestamps=*/false, /*sequence_count=*/0};
+  send_packet.ReserveExtension<TransportSequenceNumberV2>();
+  send_packet.SetExtension<TransportSequenceNumberV2>(kTransportSequenceNumber,
+                                                      kNoFeedbackRequest);
+  EXPECT_EQ(send_packet.GetRawExtension<TransportSequenceNumberV2>().size(),
+            4u);
+
+  // Serialize the packet and then parse it again.
+  RtpPacketReceived receive_packet(&extensions);
+  EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
+
+  uint16_t received_transport_sequeunce_number;
+  absl::optional<FeedbackRequest> received_feedback_request;
+  EXPECT_TRUE(receive_packet.GetExtension<TransportSequenceNumberV2>(
+      &received_transport_sequeunce_number, &received_feedback_request));
+  EXPECT_EQ(received_transport_sequeunce_number, kTransportSequenceNumber);
+  EXPECT_FALSE(received_feedback_request);
+}
+
+TEST(RtpPacketTest,
+     CreateAndParseTransportSequenceNumberV2WithFeedbackRequest) {
+  // Create a packet with TransportSequenceNumberV2 extension populated.
+  RtpPacketToSend::ExtensionManager extensions;
+  constexpr int kExtensionId = 1;
+  extensions.Register<TransportSequenceNumberV2>(kExtensionId);
+  RtpPacketToSend send_packet(&extensions);
+  send_packet.SetPayloadType(kPayloadType);
+  send_packet.SetSequenceNumber(kSeqNum);
+  send_packet.SetTimestamp(kTimestamp);
+  send_packet.SetSsrc(kSsrc);
+
+  constexpr int kTransportSequenceNumber = 12345;
+  constexpr absl::optional<FeedbackRequest> kFeedbackRequest =
+      FeedbackRequest{/*include_timestamps=*/true, /*sequence_count=*/3};
+  send_packet.SetExtension<TransportSequenceNumberV2>(kTransportSequenceNumber,
+                                                      kFeedbackRequest);
+
+  // Serialize the packet and then parse it again.
+  RtpPacketReceived receive_packet(&extensions);
+  EXPECT_TRUE(receive_packet.Parse(send_packet.Buffer()));
+
+  // Parse transport sequence number and feedback request.
+  uint16_t received_transport_sequeunce_number;
+  absl::optional<FeedbackRequest> received_feedback_request;
+  EXPECT_TRUE(receive_packet.GetExtension<TransportSequenceNumberV2>(
+      &received_transport_sequeunce_number, &received_feedback_request));
+  EXPECT_EQ(received_transport_sequeunce_number, kTransportSequenceNumber);
+  ASSERT_TRUE(received_feedback_request);
+  EXPECT_EQ(received_feedback_request->include_timestamps,
+            kFeedbackRequest->include_timestamps);
+  EXPECT_EQ(received_feedback_request->sequence_count,
+            kFeedbackRequest->sequence_count);
+}
+
+TEST(RtpPacketTest, ReservedExtensionsCountedAsSetExtension) {
+  // Register two extensions.
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+
+  RtpPacketReceived packet(&extensions);
+
+  // Reserve slot for only one of them.
+  EXPECT_TRUE(packet.ReserveExtension<TransmissionOffset>());
+  // Non-registered extension cannot be reserved.
+  EXPECT_FALSE(packet.ReserveExtension<VideoContentTypeExtension>());
+
+  // Only the extension that is both registered and reserved matches
+  // IsExtensionReserved().
+  EXPECT_FALSE(packet.HasExtension<VideoContentTypeExtension>());
+  EXPECT_FALSE(packet.HasExtension<AudioLevel>());
+  EXPECT_TRUE(packet.HasExtension<TransmissionOffset>());
+}
+
+// Tests that RtpPacket::RemoveExtension can successfully remove extensions.
+TEST(RtpPacketTest, RemoveMultipleExtensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel);
+
+  EXPECT_THAT(kPacketWithTOAndAL,
+              ElementsAreArray(packet.data(), packet.size()));
+
+  // Remove one of two extensions.
+  EXPECT_TRUE(packet.RemoveExtension(kRtpExtensionAudioLevel));
+
+  EXPECT_THAT(kPacketWithTO, ElementsAreArray(packet.data(), packet.size()));
+
+  // Remove remaining extension.
+  EXPECT_TRUE(packet.RemoveExtension(kRtpExtensionTransmissionTimeOffset));
+
+  EXPECT_THAT(kMinimumPacket, ElementsAreArray(packet.data(), packet.size()));
+}
+
+// Tests that RtpPacket::RemoveExtension can successfully remove extension when
+// other extensions are present but not registered.
+TEST(RtpPacketTest, RemoveExtensionPreservesOtherUnregisteredExtensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel);
+
+  EXPECT_THAT(kPacketWithTOAndAL,
+              ElementsAreArray(packet.data(), packet.size()));
+
+  // "Unregister" kRtpExtensionTransmissionTimeOffset.
+  RtpPacketToSend::ExtensionManager extensions1;
+  extensions1.Register<AudioLevel>(kAudioLevelExtensionId);
+  packet.IdentifyExtensions(extensions1);
+
+  // Make sure we can not delete extension which is set but not registered.
+  EXPECT_FALSE(packet.RemoveExtension(kRtpExtensionTransmissionTimeOffset));
+
+  // Remove registered extension.
+  EXPECT_TRUE(packet.RemoveExtension(kRtpExtensionAudioLevel));
+
+  EXPECT_THAT(kPacketWithTO, ElementsAreArray(packet.data(), packet.size()));
+}
+
+// Tests that RtpPacket::RemoveExtension fails if extension is not present or
+// not registered and does not modify packet.
+TEST(RtpPacketTest, RemoveExtensionFailure) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
+  extensions.Register<AudioLevel>(kAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+
+  EXPECT_THAT(kPacketWithTO, ElementsAreArray(packet.data(), packet.size()));
+
+  // Try to remove extension, which was registered, but not set.
+  EXPECT_FALSE(packet.RemoveExtension(kRtpExtensionAudioLevel));
+
+  EXPECT_THAT(kPacketWithTO, ElementsAreArray(packet.data(), packet.size()));
+
+  // Try to remove extension, which was not registered.
+  EXPECT_FALSE(packet.RemoveExtension(kRtpExtensionPlayoutDelay));
+
+  EXPECT_THAT(kPacketWithTO, ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, SetExtensionWithArray) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register<RtpDependencyDescriptorExtension>(
+      kDependencyDescriptorExtensionId);
+  RtpPacketToSend packet(&extensions);
+  const uint8_t extension_data[] = {1, 2, 3, 4, 5};
+  packet.SetRawExtension<RtpDependencyDescriptorExtension>(extension_data);
+  EXPECT_THAT(packet.GetRawExtension<RtpDependencyDescriptorExtension>(),
+              ElementsAreArray(extension_data));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
new file mode 100644
index 0000000000..95dbaf364c
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
@@ -0,0 +1,402 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/rtp_packetizer_av1.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <algorithm>
+
+#include "api/array_view.h"
+#include "api/video/video_frame_type.h"
+#include "modules/rtp_rtcp/source/leb128.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/byte_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+constexpr int kAggregationHeaderSize = 1;
+// when there are 3 or less OBU (fragments) in a packet, size of the last one
+// can be omited.
+constexpr int kMaxNumObusToOmitSize = 3;
+constexpr uint8_t kObuSizePresentBit = 0b0'0000'010;
+constexpr int kObuTypeSequenceHeader = 1;
+constexpr int kObuTypeTemporalDelimiter = 2;
+constexpr int kObuTypeTileList = 8;
+constexpr int kObuTypePadding = 15;
+
+bool ObuHasExtension(uint8_t obu_header) {
+  return obu_header & 0b0'0000'100;
+}
+
+bool ObuHasSize(uint8_t obu_header) {
+  return obu_header & kObuSizePresentBit;
+}
+
+int ObuType(uint8_t obu_header) {
+  return (obu_header & 0b0'1111'000) >> 3;
+}
+
+// Given `remaining_bytes` free bytes left in a packet, returns max size of an
+// OBU fragment that can fit into the packet.
+// i.e. MaxFragmentSize + Leb128Size(MaxFragmentSize) <= remaining_bytes.
+int MaxFragmentSize(int remaining_bytes) {
+  if (remaining_bytes <= 1) {
+    return 0;
+  }
+  for (int i = 1;; ++i) {
+    if (remaining_bytes < (1 << 7 * i) + i) {
+      return remaining_bytes - i;
+    }
+  }
+}
+
+}  // namespace
+
+RtpPacketizerAv1::RtpPacketizerAv1(rtc::ArrayView<const uint8_t> payload,
+                                   RtpPacketizer::PayloadSizeLimits limits,
+                                   VideoFrameType frame_type,
+                                   bool is_last_frame_in_picture)
+    : frame_type_(frame_type),
+      obus_(ParseObus(payload)),
+      packets_(Packetize(obus_, limits)),
+      is_last_frame_in_picture_(is_last_frame_in_picture) {}
+
+std::vector<RtpPacketizerAv1::Obu> RtpPacketizerAv1::ParseObus(
+    rtc::ArrayView<const uint8_t> payload) {
+  std::vector<Obu> result;
+  rtc::ByteBufferReader payload_reader(
+      reinterpret_cast<const char*>(payload.data()), payload.size());
+  while (payload_reader.Length() > 0) {
+    Obu obu;
+    payload_reader.ReadUInt8(&obu.header);
+    obu.size = 1;
+    if (ObuHasExtension(obu.header)) {
+      if (payload_reader.Length() == 0) {
+        RTC_DLOG(LS_ERROR) << "Malformed AV1 input: expected extension_header, "
+                              "no more bytes in the buffer. Offset: "
+                           << (payload.size() - payload_reader.Length());
+        return {};
+      }
+      payload_reader.ReadUInt8(&obu.extension_header);
+      ++obu.size;
+    }
+    if (!ObuHasSize(obu.header)) {
+      obu.payload = rtc::MakeArrayView(
+          reinterpret_cast<const uint8_t*>(payload_reader.Data()),
+          payload_reader.Length());
+      payload_reader.Consume(payload_reader.Length());
+    } else {
+      uint64_t size = 0;
+      if (!payload_reader.ReadUVarint(&size) ||
+          size > payload_reader.Length()) {
+        RTC_DLOG(LS_ERROR) << "Malformed AV1 input: declared size " << size
+                           << " is larger than remaining buffer size "
+                           << payload_reader.Length();
+        return {};
+      }
+      obu.payload = rtc::MakeArrayView(
+          reinterpret_cast<const uint8_t*>(payload_reader.Data()), size);
+      payload_reader.Consume(size);
+    }
+    obu.size += obu.payload.size();
+    // Skip obus that shouldn't be transfered over rtp.
+    int obu_type = ObuType(obu.header);
+    if (obu_type != kObuTypeTemporalDelimiter &&  //
+        obu_type != kObuTypeTileList &&           //
+        obu_type != kObuTypePadding) {
+      result.push_back(obu);
+    }
+  }
+  return result;
+}
+
+int RtpPacketizerAv1::AdditionalBytesForPreviousObuElement(
+    const Packet& packet) {
+  if (packet.packet_size == 0) {
+    // Packet is still empty => no last OBU element, no need to reserve space
+    // for it.
+    return 0;
+  }
+  if (packet.num_obu_elements > kMaxNumObusToOmitSize) {
+    // There is so many obu elements in the packet, all of them must be
+    // prepended with the length field. That imply space for the length of the
+    // last obu element is already reserved.
+    return 0;
+  }
+  // No space was reserved for length field of the last OBU element, but that
+  // element becoming non-last, so it now requires explicit length field.
+  // Calculate how many bytes are needed to store the length in leb128 format.
+  return Leb128Size(packet.last_obu_size);
+}
+
+std::vector<RtpPacketizerAv1::Packet> RtpPacketizerAv1::Packetize(
+    rtc::ArrayView<const Obu> obus,
+    PayloadSizeLimits limits) {
+  std::vector<Packet> packets;
+  if (obus.empty()) {
+    return packets;
+  }
+  // Ignore certian edge cases where packets should be very small. They are
+  // inpractical but adds complexity to handle.
+  if (limits.max_payload_len - limits.last_packet_reduction_len < 3 ||
+      limits.max_payload_len - limits.first_packet_reduction_len < 3) {
+    RTC_DLOG(LS_ERROR) << "Failed to packetize AV1 frame: requested packet "
+                          "size is unreasonable small.";
+    return packets;
+  }
+  // Aggregation header is present in all packets.
+  limits.max_payload_len -= kAggregationHeaderSize;
+
+  // Assemble packets. Push to current packet as much as it can hold before
+  // considering next one. That would normally cause uneven distribution across
+  // packets, specifically last one would be generally smaller.
+  packets.emplace_back(/*first_obu_index=*/0);
+  int packet_remaining_bytes =
+      limits.max_payload_len - limits.first_packet_reduction_len;
+  for (size_t obu_index = 0; obu_index < obus.size(); ++obu_index) {
+    const bool is_last_obu = obu_index == obus.size() - 1;
+    const Obu& obu = obus[obu_index];
+
+    // Putting `obu` into the last packet would make last obu element stored in
+    // that packet not last. All not last OBU elements must be prepend with the
+    // element length. AdditionalBytesForPreviousObuElement calculates how many
+    // bytes are needed to store that length.
+    int previous_obu_extra_size =
+        AdditionalBytesForPreviousObuElement(packets.back());
+    int min_required_size =
+        packets.back().num_obu_elements >= kMaxNumObusToOmitSize ? 2 : 1;
+    if (packet_remaining_bytes < previous_obu_extra_size + min_required_size) {
+      // Start a new packet.
+      packets.emplace_back(/*first_obu_index=*/obu_index);
+      packet_remaining_bytes = limits.max_payload_len;
+      previous_obu_extra_size = 0;
+    }
+    Packet& packet = packets.back();
+    // Start inserting current obu into the packet.
+    packet.packet_size += previous_obu_extra_size;
+    packet_remaining_bytes -= previous_obu_extra_size;
+    packet.num_obu_elements++;
+
+    bool must_write_obu_element_size =
+        packet.num_obu_elements > kMaxNumObusToOmitSize;
+    // Can fit all of the obu into the packet?
+    int required_bytes = obu.size;
+    if (must_write_obu_element_size) {
+      required_bytes += Leb128Size(obu.size);
+    }
+    int available_bytes = packet_remaining_bytes;
+    if (is_last_obu) {
+      // If this packet would be the last packet, available size is smaller.
+      if (packets.size() == 1) {
+        available_bytes += limits.first_packet_reduction_len;
+        available_bytes -= limits.single_packet_reduction_len;
+      } else {
+        available_bytes -= limits.last_packet_reduction_len;
+      }
+    }
+    if (required_bytes <= available_bytes) {
+      // Insert the obu into the packet unfragmented.
+      packet.last_obu_size = obu.size;
+      packet.packet_size += required_bytes;
+      packet_remaining_bytes -= required_bytes;
+      continue;
+    }
+
+    // Fragment the obu.
+    int max_first_fragment_size = must_write_obu_element_size
+                                      ? MaxFragmentSize(packet_remaining_bytes)
+                                      : packet_remaining_bytes;
+    // Because available_bytes might be different than
+    // packet_remaining_bytes it might happen that max_first_fragment_size >=
+    // obu.size. Also, since checks above verified `obu` should not be put
+    // completely into the `packet`, leave at least 1 byte for later packet.
+    int first_fragment_size = std::min(obu.size - 1, max_first_fragment_size);
+    if (first_fragment_size == 0) {
+      // Rather than writing 0-size element at the tail of the packet,
+      // 'uninsert' the `obu` from the `packet`.
+      packet.num_obu_elements--;
+      packet.packet_size -= previous_obu_extra_size;
+    } else {
+      packet.packet_size += first_fragment_size;
+      if (must_write_obu_element_size) {
+        packet.packet_size += Leb128Size(first_fragment_size);
+      }
+      packet.last_obu_size = first_fragment_size;
+    }
+
+    // Add middle fragments that occupy all of the packet.
+    // These are easy because
+    // - one obu per packet imply no need to store the size of the obu.
+    // - this packets are nor the first nor the last packets of the frame, so
+    // packet capacity is always limits.max_payload_len.
+    int obu_offset;
+    for (obu_offset = first_fragment_size;
+         obu_offset + limits.max_payload_len < obu.size;
+         obu_offset += limits.max_payload_len) {
+      packets.emplace_back(/*first_obu_index=*/obu_index);
+      Packet& packet = packets.back();
+      packet.num_obu_elements = 1;
+      packet.first_obu_offset = obu_offset;
+      int middle_fragment_size = limits.max_payload_len;
+      packet.last_obu_size = middle_fragment_size;
+      packet.packet_size = middle_fragment_size;
+    }
+
+    // Add the last fragment of the obu.
+    int last_fragment_size = obu.size - obu_offset;
+    // Check for corner case where last fragment of the last obu is too large
+    // to fit into last packet, but may fully fit into semi-last packet.
+    if (is_last_obu &&
+        last_fragment_size >
+            limits.max_payload_len - limits.last_packet_reduction_len) {
+      // Split last fragments into two.
+      RTC_DCHECK_GE(last_fragment_size, 2);
+      // Try to even packet sizes rather than payload sizes across the last
+      // two packets.
+      int semi_last_fragment_size =
+          (last_fragment_size + limits.last_packet_reduction_len) / 2;
+      // But leave at least one payload byte for the last packet to avoid
+      // weird scenarios where size of the fragment is zero and rtp payload has
+      // nothing except for an aggregation header.
+      if (semi_last_fragment_size >= last_fragment_size) {
+        semi_last_fragment_size = last_fragment_size - 1;
+      }
+      last_fragment_size -= semi_last_fragment_size;
+
+      packets.emplace_back(/*first_obu_index=*/obu_index);
+      Packet& packet = packets.back();
+      packet.num_obu_elements = 1;
+      packet.first_obu_offset = obu_offset;
+      packet.last_obu_size = semi_last_fragment_size;
+      packet.packet_size = semi_last_fragment_size;
+      obu_offset += semi_last_fragment_size;
+    }
+    packets.emplace_back(/*first_obu_index=*/obu_index);
+    Packet& last_packet = packets.back();
+    last_packet.num_obu_elements = 1;
+    last_packet.first_obu_offset = obu_offset;
+    last_packet.last_obu_size = last_fragment_size;
+    last_packet.packet_size = last_fragment_size;
+    packet_remaining_bytes = limits.max_payload_len - last_fragment_size;
+  }
+  return packets;
+}
+
+uint8_t RtpPacketizerAv1::AggregationHeader() const {
+  const Packet& packet = packets_[packet_index_];
+  uint8_t aggregation_header = 0;
+
+  // Set Z flag: first obu element is continuation of the previous OBU.
+  bool first_obu_element_is_fragment = packet.first_obu_offset > 0;
+  if (first_obu_element_is_fragment)
+    aggregation_header |= (1 << 7);
+
+  // Set Y flag: last obu element will be continuated in the next packet.
+  int last_obu_offset =
+      packet.num_obu_elements == 1 ? packet.first_obu_offset : 0;
+  bool last_obu_is_fragment =
+      last_obu_offset + packet.last_obu_size <
+      obus_[packet.first_obu + packet.num_obu_elements - 1].size;
+  if (last_obu_is_fragment)
+    aggregation_header |= (1 << 6);
+
+  // Set W field: number of obu elements in the packet (when not too large).
+  if (packet.num_obu_elements <= kMaxNumObusToOmitSize)
+    aggregation_header |= packet.num_obu_elements << 4;
+
+  // Set N flag: beginning of a new coded video sequence.
+  // Encoder may produce key frame without a sequence header, thus double check
+  // incoming frame includes the sequence header. Since Temporal delimiter is
+  // already filtered out, sequence header should be the first obu when present.
+  if (frame_type_ == VideoFrameType::kVideoFrameKey && packet_index_ == 0 &&
+      ObuType(obus_.front().header) == kObuTypeSequenceHeader) {
+    aggregation_header |= (1 << 3);
+  }
+  return aggregation_header;
+}
+
+bool RtpPacketizerAv1::NextPacket(RtpPacketToSend* packet) {
+  if (packet_index_ >= packets_.size()) {
+    return false;
+  }
+  const Packet& next_packet = packets_[packet_index_];
+
+  RTC_DCHECK_GT(next_packet.num_obu_elements, 0);
+  RTC_DCHECK_LT(next_packet.first_obu_offset,
+                obus_[next_packet.first_obu].size);
+  RTC_DCHECK_LE(
+      next_packet.last_obu_size,
+      obus_[next_packet.first_obu + next_packet.num_obu_elements - 1].size);
+
+  uint8_t* const rtp_payload =
+      packet->AllocatePayload(kAggregationHeaderSize + next_packet.packet_size);
+  uint8_t* write_at = rtp_payload;
+
+  *write_at++ = AggregationHeader();
+
+  int obu_offset = next_packet.first_obu_offset;
+  // Store all OBU elements except the last one.
+  for (int i = 0; i < next_packet.num_obu_elements - 1; ++i) {
+    const Obu& obu = obus_[next_packet.first_obu + i];
+    size_t fragment_size = obu.size - obu_offset;
+    write_at += WriteLeb128(fragment_size, write_at);
+    if (obu_offset == 0) {
+      *write_at++ = obu.header & ~kObuSizePresentBit;
+    }
+    if (obu_offset <= 1 && ObuHasExtension(obu.header)) {
+      *write_at++ = obu.extension_header;
+    }
+    int payload_offset =
+        std::max(0, obu_offset - (ObuHasExtension(obu.header) ? 2 : 1));
+    size_t payload_size = obu.payload.size() - payload_offset;
+    if (!obu.payload.empty() && payload_size > 0) {
+      memcpy(write_at, obu.payload.data() + payload_offset, payload_size);
+    }
+    write_at += payload_size;
+    // All obus are stored from the beginning, except, may be, the first one.
+    obu_offset = 0;
+  }
+  // Store the last OBU element.
+  const Obu& last_obu =
+      obus_[next_packet.first_obu + next_packet.num_obu_elements - 1];
+  int fragment_size = next_packet.last_obu_size;
+  RTC_DCHECK_GT(fragment_size, 0);
+  if (next_packet.num_obu_elements > kMaxNumObusToOmitSize) {
+    write_at += WriteLeb128(fragment_size, write_at);
+  }
+  if (obu_offset == 0 && fragment_size > 0) {
+    *write_at++ = last_obu.header & ~kObuSizePresentBit;
+    --fragment_size;
+  }
+  if (obu_offset <= 1 && ObuHasExtension(last_obu.header) &&
+      fragment_size > 0) {
+    *write_at++ = last_obu.extension_header;
+    --fragment_size;
+  }
+  RTC_DCHECK_EQ(write_at - rtp_payload + fragment_size,
+                kAggregationHeaderSize + next_packet.packet_size);
+  int payload_offset =
+      std::max(0, obu_offset - (ObuHasExtension(last_obu.header) ? 2 : 1));
+  memcpy(write_at, last_obu.payload.data() + payload_offset, fragment_size);
+  write_at += fragment_size;
+
+  RTC_DCHECK_EQ(write_at - rtp_payload,
+                kAggregationHeaderSize + next_packet.packet_size);
+
+  ++packet_index_;
+  bool is_last_packet_in_frame = packet_index_ == packets_.size();
+  packet->SetMarker(is_last_packet_in_frame && is_last_frame_in_picture_);
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.h
new file mode 100644
index 0000000000..520e746eac
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_AV1_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_AV1_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/video/video_frame_type.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+
+namespace webrtc {
+
+class RtpPacketizerAv1 : public RtpPacketizer {
+ public:
+  RtpPacketizerAv1(rtc::ArrayView<const uint8_t> payload,
+                   PayloadSizeLimits limits,
+                   VideoFrameType frame_type,
+                   bool is_last_frame_in_picture);
+  ~RtpPacketizerAv1() override = default;
+
+  size_t NumPackets() const override { return packets_.size() - packet_index_; }
+  bool NextPacket(RtpPacketToSend* packet) override;
+
+ private:
+  struct Obu {
+    uint8_t header;
+    uint8_t extension_header;  // undefined if (header & kXbit) == 0
+    rtc::ArrayView<const uint8_t> payload;
+    int size;  // size of the header and payload combined.
+  };
+  struct Packet {
+    explicit Packet(int first_obu_index) : first_obu(first_obu_index) {}
+    // Indexes into obus_ vector of the first and last obus that should put into
+    // the packet.
+    int first_obu;
+    int num_obu_elements = 0;
+    int first_obu_offset = 0;
+    int last_obu_size;
+    // Total size consumed by the packet.
+    int packet_size = 0;
+  };
+
+  // Parses the payload into serie of OBUs.
+  static std::vector<Obu> ParseObus(rtc::ArrayView<const uint8_t> payload);
+  // Returns the number of additional bytes needed to store the previous OBU
+  // element if an additonal OBU element is added to the packet.
+  static int AdditionalBytesForPreviousObuElement(const Packet& packet);
+  static std::vector<Packet> Packetize(rtc::ArrayView<const Obu> obus,
+                                       PayloadSizeLimits limits);
+  uint8_t AggregationHeader() const;
+
+  const VideoFrameType frame_type_;
+  const std::vector<Obu> obus_;
+  const std::vector<Packet> packets_;
+  const bool is_last_frame_in_picture_;
+  size_t packet_index_ = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_AV1_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_test_helper.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_test_helper.cc
new file mode 100644
index 0000000000..3d62bcef44
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_test_helper.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_packetizer_av1_test_helper.h"
+
+#include <stdint.h>
+
+#include <initializer_list>
+#include <vector>
+
+namespace webrtc {
+
+Av1Obu::Av1Obu(uint8_t obu_type) : header_(obu_type | kAv1ObuSizePresentBit) {}
+
+Av1Obu& Av1Obu::WithExtension(uint8_t extension) {
+  extension_ = extension;
+  header_ |= kAv1ObuExtensionPresentBit;
+  return *this;
+}
+Av1Obu& Av1Obu::WithoutSize() {
+  header_ &= ~kAv1ObuSizePresentBit;
+  return *this;
+}
+Av1Obu& Av1Obu::WithPayload(std::vector<uint8_t> payload) {
+  payload_ = std::move(payload);
+  return *this;
+}
+
+std::vector<uint8_t> BuildAv1Frame(std::initializer_list<Av1Obu> obus) {
+  std::vector<uint8_t> raw;
+  for (const Av1Obu& obu : obus) {
+    raw.push_back(obu.header_);
+    if (obu.header_ & kAv1ObuExtensionPresentBit) {
+      raw.push_back(obu.extension_);
+    }
+    if (obu.header_ & kAv1ObuSizePresentBit) {
+      // write size in leb128 format.
+      size_t payload_size = obu.payload_.size();
+      while (payload_size >= 0x80) {
+        raw.push_back(0x80 | (payload_size & 0x7F));
+        payload_size >>= 7;
+      }
+      raw.push_back(payload_size);
+    }
+    raw.insert(raw.end(), obu.payload_.begin(), obu.payload_.end());
+  }
+  return raw;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_test_helper.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_test_helper.h
new file mode 100644
index 0000000000..04a902fe56
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_test_helper.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_AV1_TEST_HELPER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_AV1_TEST_HELPER_H_
+
+#include <stdint.h>
+
+#include <initializer_list>
+#include <utility>
+#include <vector>
+
+namespace webrtc {
+// All obu types offset by 3 to take correct position in the obu_header.
+constexpr uint8_t kAv1ObuTypeSequenceHeader = 1 << 3;
+constexpr uint8_t kAv1ObuTypeTemporalDelimiter = 2 << 3;
+constexpr uint8_t kAv1ObuTypeFrameHeader = 3 << 3;
+constexpr uint8_t kAv1ObuTypeTileGroup = 4 << 3;
+constexpr uint8_t kAv1ObuTypeMetadata = 5 << 3;
+constexpr uint8_t kAv1ObuTypeFrame = 6 << 3;
+constexpr uint8_t kAv1ObuTypeTileList = 8 << 3;
+constexpr uint8_t kAv1ObuExtensionPresentBit = 0b0'0000'100;
+constexpr uint8_t kAv1ObuSizePresentBit = 0b0'0000'010;
+constexpr uint8_t kAv1ObuExtensionS1T1 = 0b001'01'000;
+
+class Av1Obu {
+ public:
+  explicit Av1Obu(uint8_t obu_type);
+
+  Av1Obu& WithExtension(uint8_t extension);
+  Av1Obu& WithoutSize();
+  Av1Obu& WithPayload(std::vector<uint8_t> payload);
+
+ private:
+  friend std::vector<uint8_t> BuildAv1Frame(std::initializer_list<Av1Obu> obus);
+  uint8_t header_;
+  uint8_t extension_ = 0;
+  std::vector<uint8_t> payload_;
+};
+
+std::vector<uint8_t> BuildAv1Frame(std::initializer_list<Av1Obu> obus);
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_AV1_TEST_HELPER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
new file mode 100644
index 0000000000..2151a59295
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
@@ -0,0 +1,342 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_packetizer_av1.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <initializer_list>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/video/encoded_image.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_packetizer_av1_test_helper.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Each;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::Le;
+using ::testing::SizeIs;
+
+constexpr uint8_t kNewCodedVideoSequenceBit = 0b00'00'1000;
+
+// Wrapper around rtp_packet to make it look like container of payload bytes.
+struct RtpPayload {
+  using value_type = rtc::ArrayView<const uint8_t>::value_type;
+  using const_iterator = rtc::ArrayView<const uint8_t>::const_iterator;
+
+  RtpPayload() : rtp_packet(/*extensions=*/nullptr) {}
+  RtpPayload& operator=(RtpPayload&&) = default;
+  RtpPayload(RtpPayload&&) = default;
+
+  const_iterator begin() const { return rtp_packet.payload().begin(); }
+  const_iterator end() const { return rtp_packet.payload().end(); }
+  const uint8_t* data() const { return rtp_packet.payload().data(); }
+  size_t size() const { return rtp_packet.payload().size(); }
+
+  uint8_t aggregation_header() const { return rtp_packet.payload()[0]; }
+
+  RtpPacketToSend rtp_packet;
+};
+
+// Wrapper around frame pointer to make it look like container of bytes with
+// nullptr frame look like empty container.
+class Av1Frame {
+ public:
+  using value_type = uint8_t;
+  using const_iterator = const uint8_t*;
+
+  explicit Av1Frame(rtc::scoped_refptr<EncodedImageBuffer> frame)
+      : frame_(std::move(frame)) {}
+
+  const_iterator begin() const { return frame_ ? frame_->data() : nullptr; }
+  const_iterator end() const {
+    return frame_ ? (frame_->data() + frame_->size()) : nullptr;
+  }
+
+ private:
+  rtc::scoped_refptr<EncodedImageBuffer> frame_;
+};
+
+std::vector<RtpPayload> Packetize(
+    rtc::ArrayView<const uint8_t> payload,
+    RtpPacketizer::PayloadSizeLimits limits,
+    VideoFrameType frame_type = VideoFrameType::kVideoFrameDelta,
+    bool is_last_frame_in_picture = true) {
+  // Run code under test.
+  RtpPacketizerAv1 packetizer(payload, limits, frame_type,
+                              is_last_frame_in_picture);
+  // Convert result into structure that is easier to run expectation against.
+  std::vector<RtpPayload> result(packetizer.NumPackets());
+  for (RtpPayload& rtp_payload : result) {
+    EXPECT_TRUE(packetizer.NextPacket(&rtp_payload.rtp_packet));
+  }
+  return result;
+}
+
+Av1Frame ReassembleFrame(rtc::ArrayView<const RtpPayload> rtp_payloads) {
+  std::vector<rtc::ArrayView<const uint8_t>> payloads(rtp_payloads.size());
+  for (size_t i = 0; i < rtp_payloads.size(); ++i) {
+    payloads[i] = rtp_payloads[i];
+  }
+  return Av1Frame(VideoRtpDepacketizerAv1().AssembleFrame(payloads));
+}
+
+TEST(RtpPacketizerAv1Test, PacketizeOneObuWithoutSizeAndExtension) {
+  auto kFrame = BuildAv1Frame({Av1Obu(kAv1ObuTypeFrame)
+                                   .WithoutSize()
+                                   .WithPayload({1, 2, 3, 4, 5, 6, 7})});
+  EXPECT_THAT(Packetize(kFrame, {}),
+              ElementsAre(ElementsAre(0b00'01'0000,  // aggregation header
+                                      kAv1ObuTypeFrame, 1, 2, 3, 4, 5, 6, 7)));
+}
+
+TEST(RtpPacketizerAv1Test, PacketizeOneObuWithoutSizeWithExtension) {
+  auto kFrame = BuildAv1Frame({Av1Obu(kAv1ObuTypeFrame)
+                                   .WithoutSize()
+                                   .WithExtension(kAv1ObuExtensionS1T1)
+                                   .WithPayload({2, 3, 4, 5, 6, 7})});
+  EXPECT_THAT(
+      Packetize(kFrame, {}),
+      ElementsAre(ElementsAre(0b00'01'0000,  // aggregation header
+                              kAv1ObuTypeFrame | kAv1ObuExtensionPresentBit,
+                              kAv1ObuExtensionS1T1, 2, 3, 4, 5, 6, 7)));
+}
+
+TEST(RtpPacketizerAv1Test, RemovesObuSizeFieldWithoutExtension) {
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeFrame).WithPayload({11, 12, 13, 14, 15, 16, 17})});
+  EXPECT_THAT(
+      Packetize(kFrame, {}),
+      ElementsAre(ElementsAre(0b00'01'0000,  // aggregation header
+                              kAv1ObuTypeFrame, 11, 12, 13, 14, 15, 16, 17)));
+}
+
+TEST(RtpPacketizerAv1Test, RemovesObuSizeFieldWithExtension) {
+  auto kFrame = BuildAv1Frame({Av1Obu(kAv1ObuTypeFrame)
+                                   .WithExtension(kAv1ObuExtensionS1T1)
+                                   .WithPayload({1, 2, 3, 4, 5, 6, 7})});
+  EXPECT_THAT(
+      Packetize(kFrame, {}),
+      ElementsAre(ElementsAre(0b00'01'0000,  // aggregation header
+                              kAv1ObuTypeFrame | kAv1ObuExtensionPresentBit,
+                              kAv1ObuExtensionS1T1, 1, 2, 3, 4, 5, 6, 7)));
+}
+
+TEST(RtpPacketizerAv1Test, OmitsSizeForLastObuWhenThreeObusFitsIntoThePacket) {
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeSequenceHeader).WithPayload({1, 2, 3, 4, 5, 6}),
+       Av1Obu(kAv1ObuTypeMetadata).WithPayload({11, 12, 13, 14}),
+       Av1Obu(kAv1ObuTypeFrame).WithPayload({21, 22, 23, 24, 25, 26})});
+  EXPECT_THAT(Packetize(kFrame, {}),
+              ElementsAre(ElementsAre(
+                  0b00'11'0000,  // aggregation header
+                  7, kAv1ObuTypeSequenceHeader, 1, 2, 3, 4, 5, 6,  //
+                  5, kAv1ObuTypeMetadata, 11, 12, 13, 14,          //
+                  kAv1ObuTypeFrame, 21, 22, 23, 24, 25, 26)));
+}
+
+TEST(RtpPacketizerAv1Test, UseSizeForAllObusWhenFourObusFitsIntoThePacket) {
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeSequenceHeader).WithPayload({1, 2, 3, 4, 5, 6}),
+       Av1Obu(kAv1ObuTypeMetadata).WithPayload({11, 12, 13, 14}),
+       Av1Obu(kAv1ObuTypeFrameHeader).WithPayload({21, 22, 23}),
+       Av1Obu(kAv1ObuTypeTileGroup).WithPayload({31, 32, 33, 34, 35, 36})});
+  EXPECT_THAT(Packetize(kFrame, {}),
+              ElementsAre(ElementsAre(
+                  0b00'00'0000,  // aggregation header
+                  7, kAv1ObuTypeSequenceHeader, 1, 2, 3, 4, 5, 6,  //
+                  5, kAv1ObuTypeMetadata, 11, 12, 13, 14,          //
+                  4, kAv1ObuTypeFrameHeader, 21, 22, 23,           //
+                  7, kAv1ObuTypeTileGroup, 31, 32, 33, 34, 35, 36)));
+}
+
+TEST(RtpPacketizerAv1Test, DiscardsTemporalDelimiterAndTileListObu) {
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeTemporalDelimiter), Av1Obu(kAv1ObuTypeMetadata),
+       Av1Obu(kAv1ObuTypeTileList).WithPayload({1, 2, 3, 4, 5, 6}),
+       Av1Obu(kAv1ObuTypeFrameHeader).WithPayload({21, 22, 23}),
+       Av1Obu(kAv1ObuTypeTileGroup).WithPayload({31, 32, 33, 34, 35, 36})});
+
+  EXPECT_THAT(
+      Packetize(kFrame, {}),
+      ElementsAre(ElementsAre(0b00'11'0000,  // aggregation header
+                              1,
+                              kAv1ObuTypeMetadata,  //
+                              4, kAv1ObuTypeFrameHeader, 21, 22,
+                              23,  //
+                              kAv1ObuTypeTileGroup, 31, 32, 33, 34, 35, 36)));
+}
+
+TEST(RtpPacketizerAv1Test, SplitTwoObusIntoTwoPacketForceSplitObuHeader) {
+  // Craft expected payloads so that there is only one way to split original
+  // frame into two packets.
+  const uint8_t kExpectPayload1[6] = {
+      0b01'10'0000,  // aggregation_header
+      3,
+      kAv1ObuTypeFrameHeader | kAv1ObuExtensionPresentBit,
+      kAv1ObuExtensionS1T1,
+      21,  //
+      kAv1ObuTypeTileGroup | kAv1ObuExtensionPresentBit};
+  const uint8_t kExpectPayload2[6] = {0b10'01'0000,  // aggregation_header
+                                      kAv1ObuExtensionS1T1, 11, 12, 13, 14};
+  auto kFrame = BuildAv1Frame({Av1Obu(kAv1ObuTypeFrameHeader)
+                                   .WithExtension(kAv1ObuExtensionS1T1)
+                                   .WithPayload({21}),
+                               Av1Obu(kAv1ObuTypeTileGroup)
+                                   .WithExtension(kAv1ObuExtensionS1T1)
+                                   .WithPayload({11, 12, 13, 14})});
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 6;
+  auto payloads = Packetize(kFrame, limits);
+  EXPECT_THAT(payloads, ElementsAre(ElementsAreArray(kExpectPayload1),
+                                    ElementsAreArray(kExpectPayload2)));
+}
+
+TEST(RtpPacketizerAv1Test,
+     SetsNbitAtTheFirstPacketOfAKeyFrameWithSequenceHeader) {
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeSequenceHeader).WithPayload({1, 2, 3, 4, 5, 6, 7})});
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 6;
+  auto packets = Packetize(kFrame, limits, VideoFrameType::kVideoFrameKey);
+  ASSERT_THAT(packets, SizeIs(2));
+  EXPECT_TRUE(packets[0].aggregation_header() & kNewCodedVideoSequenceBit);
+  EXPECT_FALSE(packets[1].aggregation_header() & kNewCodedVideoSequenceBit);
+}
+
+TEST(RtpPacketizerAv1Test,
+     DoesntSetNbitAtThePacketsOfAKeyFrameWithoutSequenceHeader) {
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeFrame).WithPayload({1, 2, 3, 4, 5, 6, 7})});
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 6;
+  auto packets = Packetize(kFrame, limits, VideoFrameType::kVideoFrameKey);
+  ASSERT_THAT(packets, SizeIs(2));
+  EXPECT_FALSE(packets[0].aggregation_header() & kNewCodedVideoSequenceBit);
+  EXPECT_FALSE(packets[1].aggregation_header() & kNewCodedVideoSequenceBit);
+}
+
+TEST(RtpPacketizerAv1Test, DoesntSetNbitAtThePacketsOfADeltaFrame) {
+  // Even when that delta frame starts with a (redundant) sequence header.
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeSequenceHeader).WithPayload({1, 2, 3, 4, 5, 6, 7})});
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 6;
+  auto packets = Packetize(kFrame, limits, VideoFrameType::kVideoFrameDelta);
+  ASSERT_THAT(packets, SizeIs(2));
+  EXPECT_FALSE(packets[0].aggregation_header() & kNewCodedVideoSequenceBit);
+  EXPECT_FALSE(packets[1].aggregation_header() & kNewCodedVideoSequenceBit);
+}
+
+// There are multiple valid reasonable ways to split payload into multiple
+// packets, do not validate current choice, instead use RtpDepacketizer
+// to validate frame is reconstracted to the same one. Note: since
+// RtpDepacketizer always inserts obu_size fields in the output, use frame where
+// each obu has obu_size fields for more streight forward validation.
+TEST(RtpPacketizerAv1Test, SplitSingleObuIntoTwoPackets) {
+  auto kFrame =
+      BuildAv1Frame({Av1Obu(kAv1ObuTypeFrame)
+                         .WithPayload({11, 12, 13, 14, 15, 16, 17, 18, 19})});
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 8;
+  auto payloads = Packetize(kFrame, limits);
+  EXPECT_THAT(payloads, ElementsAre(SizeIs(Le(8u)), SizeIs(Le(8u))));
+
+  // Use RtpDepacketizer to validate the split.
+  EXPECT_THAT(ReassembleFrame(payloads), ElementsAreArray(kFrame));
+}
+
+TEST(RtpPacketizerAv1Test, SplitSingleObuIntoManyPackets) {
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeFrame).WithPayload(std::vector<uint8_t>(1200, 27))});
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 100;
+  auto payloads = Packetize(kFrame, limits);
+  EXPECT_THAT(payloads, SizeIs(13u));
+  EXPECT_THAT(payloads, Each(SizeIs(Le(100u))));
+
+  // Use RtpDepacketizer to validate the split.
+  EXPECT_THAT(ReassembleFrame(payloads), ElementsAreArray(kFrame));
+}
+
+TEST(RtpPacketizerAv1Test, SetMarkerBitForLastPacketInEndOfPictureFrame) {
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeFrame).WithPayload(std::vector<uint8_t>(200, 27))});
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 100;
+  auto payloads = Packetize(kFrame, limits, VideoFrameType::kVideoFrameDelta,
+                            /*is_last_frame_in_picture=*/true);
+  ASSERT_THAT(payloads, SizeIs(3u));
+  EXPECT_FALSE(payloads[0].rtp_packet.Marker());
+  EXPECT_FALSE(payloads[1].rtp_packet.Marker());
+  EXPECT_TRUE(payloads[2].rtp_packet.Marker());
+}
+
+TEST(RtpPacketizerAv1Test, DoesntSetMarkerBitForPacketsNotInEndOfPictureFrame) {
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeFrame).WithPayload(std::vector<uint8_t>(200, 27))});
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 100;
+  auto payloads = Packetize(kFrame, limits, VideoFrameType::kVideoFrameDelta,
+                            /*is_last_frame_in_picture=*/false);
+  ASSERT_THAT(payloads, SizeIs(3u));
+  EXPECT_FALSE(payloads[0].rtp_packet.Marker());
+  EXPECT_FALSE(payloads[1].rtp_packet.Marker());
+  EXPECT_FALSE(payloads[2].rtp_packet.Marker());
+}
+
+TEST(RtpPacketizerAv1Test, SplitTwoObusIntoTwoPackets) {
+  // 2nd OBU is too large to fit into one packet, so its head would be in the
+  // same packet as the 1st OBU.
+  auto kFrame = BuildAv1Frame(
+      {Av1Obu(kAv1ObuTypeSequenceHeader).WithPayload({11, 12}),
+       Av1Obu(kAv1ObuTypeFrame).WithPayload({1, 2, 3, 4, 5, 6, 7, 8, 9})});
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 8;
+  auto payloads = Packetize(kFrame, limits);
+  EXPECT_THAT(payloads, ElementsAre(SizeIs(Le(8u)), SizeIs(Le(8u))));
+
+  // Use RtpDepacketizer to validate the split.
+  EXPECT_THAT(ReassembleFrame(payloads), ElementsAreArray(kFrame));
+}
+
+TEST(RtpPacketizerAv1Test,
+     SplitSingleObuIntoTwoPacketsBecauseOfSinglePacketLimit) {
+  auto kFrame =
+      BuildAv1Frame({Av1Obu(kAv1ObuTypeFrame)
+                         .WithPayload({11, 12, 13, 14, 15, 16, 17, 18, 19})});
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 10;
+  limits.single_packet_reduction_len = 8;
+  auto payloads = Packetize(kFrame, limits);
+  EXPECT_THAT(payloads, ElementsAre(SizeIs(Le(10u)), SizeIs(Le(10u))));
+
+  EXPECT_THAT(ReassembleFrame(payloads), ElementsAreArray(kFrame));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
new file mode 100644
index 0000000000..9ac7696ce9
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
+
+#include "api/units/time_delta.h"
+
+// Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
+namespace webrtc {
+constexpr int kDefaultMaxReorderingThreshold = 50;  // In sequence numbers.
+constexpr int kRtcpMaxNackFields = 253;
+
+constexpr TimeDelta RTCP_INTERVAL_RAPID_SYNC_MS =
+    TimeDelta::Millis(100); // RFX 6051
+constexpr TimeDelta RTCP_SEND_BEFORE_KEY_FRAME = TimeDelta::Millis(100);
+constexpr int RTCP_MAX_REPORT_BLOCKS = 31;  // RFC 3550 page 37
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
new file mode 100644
index 0000000000..1c31611409
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -0,0 +1,768 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/transport/field_trial_based_config.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "modules/rtp_rtcp/source/rtcp_sender.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/ntp_time.h"
+
+#ifdef _WIN32
+// Disable warning C4355: 'this' : used in base member initializer list.
+#pragma warning(disable : 4355)
+#endif
+
+namespace webrtc {
+namespace {
+const int64_t kRtpRtcpRttProcessTimeMs = 1000;
+const int64_t kRtpRtcpBitrateProcessTimeMs = 10;
+const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
+}  // namespace
+
+ModuleRtpRtcpImpl::RtpSenderContext::RtpSenderContext(
+    const RtpRtcpInterface::Configuration& config)
+    : packet_history(config.clock, config.enable_rtx_padding_prioritization),
+      sequencer_(config.local_media_ssrc,
+                 config.rtx_send_ssrc,
+                 /*require_marker_before_media_padding=*/!config.audio,
+                 config.clock),
+      packet_sender(config, &packet_history),
+      non_paced_sender(&packet_sender, &sequencer_),
+      packet_generator(
+          config,
+          &packet_history,
+          config.paced_sender ? config.paced_sender : &non_paced_sender) {}
+
+std::unique_ptr<RtpRtcp> RtpRtcp::DEPRECATED_Create(
+    const Configuration& configuration) {
+  RTC_DCHECK(configuration.clock);
+  return std::make_unique<ModuleRtpRtcpImpl>(configuration);
+}
+
+ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
+    : rtcp_sender_(
+          RTCPSender::Configuration::FromRtpRtcpConfiguration(configuration)),
+      rtcp_receiver_(configuration, this),
+      clock_(configuration.clock),
+      last_bitrate_process_time_(clock_->TimeInMilliseconds()),
+      last_rtt_process_time_(clock_->TimeInMilliseconds()),
+      packet_overhead_(28),  // IPV4 UDP.
+      nack_last_time_sent_full_ms_(0),
+      nack_last_seq_number_sent_(0),
+      rtt_stats_(configuration.rtt_stats),
+      rtt_ms_(0) {
+  if (!configuration.receiver_only) {
+    rtp_sender_ = std::make_unique<RtpSenderContext>(configuration);
+    // Make sure rtcp sender use same timestamp offset as rtp sender.
+    rtcp_sender_.SetTimestampOffset(
+        rtp_sender_->packet_generator.TimestampOffset());
+  }
+
+  // Set default packet size limit.
+  // TODO(nisse): Kind-of duplicates
+  // webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize.
+  const size_t kTcpOverIpv4HeaderSize = 40;
+  SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize);
+}
+
+ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() = default;
+
+// Process any pending tasks such as timeouts (non time critical events).
+void ModuleRtpRtcpImpl::Process() {
+  const int64_t now = clock_->TimeInMilliseconds();
+
+  if (rtp_sender_) {
+    if (now >= last_bitrate_process_time_ + kRtpRtcpBitrateProcessTimeMs) {
+      rtp_sender_->packet_sender.ProcessBitrateAndNotifyObservers();
+      last_bitrate_process_time_ = now;
+    }
+  }
+
+  // TODO(bugs.webrtc.org/11581): We update the RTT once a second, whereas other
+  // things that run in this method are updated much more frequently. Move the
+  // RTT checking over to the worker thread, which matches better with where the
+  // stats are maintained.
+  bool process_rtt = now >= last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs;
+  if (rtcp_sender_.Sending()) {
+    // Process RTT if we have received a report block and we haven't
+    // processed RTT for at least `kRtpRtcpRttProcessTimeMs` milliseconds.
+    // Note that LastReceivedReportBlockMs() grabs a lock, so check
+    // `process_rtt` first.
+    if (process_rtt && rtt_stats_ != nullptr &&
+        rtcp_receiver_.LastReceivedReportBlockMs() > last_rtt_process_time_) {
+      int64_t max_rtt_ms = 0;
+      for (const auto& block : rtcp_receiver_.GetLatestReportBlockData()) {
+        if (block.last_rtt_ms() > max_rtt_ms) {
+          max_rtt_ms = block.last_rtt_ms();
+        }
+      }
+      // Report the rtt.
+      if (max_rtt_ms > 0) {
+        rtt_stats_->OnRttUpdate(max_rtt_ms);
+      }
+    }
+
+    // Verify receiver reports are delivered and the reported sequence number
+    // is increasing.
+    // TODO(bugs.webrtc.org/11581): The timeout value needs to be checked every
+    // few seconds (see internals of RtcpRrTimeout). Here, we may be polling it
+    // a couple of hundred times a second, which isn't great since it grabs a
+    // lock. Note also that LastReceivedReportBlockMs() (called above) and
+    // RtcpRrTimeout() both grab the same lock and check the same timer, so
+    // it should be possible to consolidate that work somehow.
+    if (rtcp_receiver_.RtcpRrTimeout()) {
+      RTC_LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
+    } else if (rtcp_receiver_.RtcpRrSequenceNumberTimeout()) {
+      RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended "
+                               "highest sequence number.";
+    }
+  } else {
+    // Report rtt from receiver.
+    if (process_rtt) {
+      int64_t rtt_ms;
+      if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
+        rtt_stats_->OnRttUpdate(rtt_ms);
+      }
+    }
+  }
+
+  // Get processed rtt.
+  if (process_rtt) {
+    last_rtt_process_time_ = now;
+    if (rtt_stats_) {
+      // Make sure we have a valid RTT before setting.
+      int64_t last_rtt = rtt_stats_->LastProcessedRtt();
+      if (last_rtt >= 0)
+        set_rtt_ms(last_rtt);
+    }
+  }
+
+  if (rtcp_sender_.TimeToSendRTCPReport())
+    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+
+  if (rtcp_sender_.TMMBR() && rtcp_receiver_.UpdateTmmbrTimers()) {
+    rtcp_receiver_.NotifyTmmbrUpdated();
+  }
+}
+
+void ModuleRtpRtcpImpl::SetRtxSendStatus(int mode) {
+  rtp_sender_->packet_generator.SetRtxStatus(mode);
+}
+
+int ModuleRtpRtcpImpl::RtxSendStatus() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.RtxStatus() : kRtxOff;
+}
+
+void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
+                                              int associated_payload_type) {
+  rtp_sender_->packet_generator.SetRtxPayloadType(payload_type,
+                                                  associated_payload_type);
+}
+
+absl::optional<uint32_t> ModuleRtpRtcpImpl::RtxSsrc() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : absl::nullopt;
+}
+
+absl::optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
+  if (rtp_sender_) {
+    return rtp_sender_->packet_generator.FlexfecSsrc();
+  }
+  return absl::nullopt;
+}
+
+void ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet,
+                                           const size_t length) {
+  rtcp_receiver_.IncomingPacket(rtcp_packet, length);
+}
+
+void ModuleRtpRtcpImpl::RegisterSendPayloadFrequency(int payload_type,
+                                                     int payload_frequency) {
+  rtcp_sender_.SetRtpClockRate(payload_type, payload_frequency);
+}
+
+int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
+  return 0;
+}
+
+uint32_t ModuleRtpRtcpImpl::StartTimestamp() const {
+  return rtp_sender_->packet_generator.TimestampOffset();
+}
+
+// Configure start timestamp, default is a random number.
+void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) {
+  rtcp_sender_.SetTimestampOffset(timestamp);
+  rtp_sender_->packet_generator.SetTimestampOffset(timestamp);
+  rtp_sender_->packet_sender.SetTimestampOffset(timestamp);
+}
+
+uint16_t ModuleRtpRtcpImpl::SequenceNumber() const {
+  MutexLock lock(&rtp_sender_->sequencer_mutex);
+  return rtp_sender_->sequencer_.media_sequence_number();
+}
+
+// Set SequenceNumber, default is a random number.
+void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
+  MutexLock lock(&rtp_sender_->sequencer_mutex);
+  rtp_sender_->sequencer_.set_media_sequence_number(seq_num);
+}
+
+void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) {
+  MutexLock lock(&rtp_sender_->sequencer_mutex);
+  rtp_sender_->packet_generator.SetRtpState(rtp_state);
+  rtp_sender_->sequencer_.SetRtpState(rtp_state);
+  rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
+}
+
+void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) {
+  MutexLock lock(&rtp_sender_->sequencer_mutex);
+  rtp_sender_->packet_generator.SetRtxRtpState(rtp_state);
+  rtp_sender_->sequencer_.set_rtx_sequence_number(rtp_state.sequence_number);
+}
+
+RtpState ModuleRtpRtcpImpl::GetRtpState() const {
+  MutexLock lock(&rtp_sender_->sequencer_mutex);
+  RtpState state = rtp_sender_->packet_generator.GetRtpState();
+  rtp_sender_->sequencer_.PopulateRtpState(state);
+  return state;
+}
+
+RtpState ModuleRtpRtcpImpl::GetRtxState() const {
+  MutexLock lock(&rtp_sender_->sequencer_mutex);
+  RtpState state = rtp_sender_->packet_generator.GetRtxRtpState();
+  state.sequence_number = rtp_sender_->sequencer_.rtx_sequence_number();
+  return state;
+}
+
+void ModuleRtpRtcpImpl::SetMid(absl::string_view mid) {
+  if (rtp_sender_) {
+    rtp_sender_->packet_generator.SetMid(mid);
+  }
+  // TODO(bugs.webrtc.org/4050): If we end up supporting the MID SDES item for
+  // RTCP, this will need to be passed down to the RTCPSender also.
+}
+
+// TODO(pbos): Handle media and RTX streams separately (separate RTCP
+// feedbacks).
+RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
+  RTCPSender::FeedbackState state;
+  // This is called also when receiver_only is true. Hence below
+  // checks that rtp_sender_ exists.
+  if (rtp_sender_) {
+    StreamDataCounters rtp_stats;
+    StreamDataCounters rtx_stats;
+    rtp_sender_->packet_sender.GetDataCounters(&rtp_stats, &rtx_stats);
+    state.packets_sent =
+        rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
+    state.media_bytes_sent = rtp_stats.transmitted.payload_bytes +
+                             rtx_stats.transmitted.payload_bytes;
+    state.send_bitrate =
+        rtp_sender_->packet_sender.GetSendRates().Sum().bps<uint32_t>();
+  }
+  state.receiver = &rtcp_receiver_;
+
+  uint32_t received_ntp_secs = 0;
+  uint32_t received_ntp_frac = 0;
+  state.remote_sr = 0;
+  if (rtcp_receiver_.NTP(&received_ntp_secs, &received_ntp_frac,
+                         /*rtcp_arrival_time_secs=*/&state.last_rr_ntp_secs,
+                         /*rtcp_arrival_time_frac=*/&state.last_rr_ntp_frac,
+                         /*rtcp_timestamp=*/nullptr,
+                         /*remote_sender_packet_count=*/nullptr,
+                         /*remote_sender_octet_count=*/nullptr,
+                         /*remote_sender_reports_count=*/nullptr)) {
+    state.remote_sr = ((received_ntp_secs & 0x0000ffff) << 16) +
+                      ((received_ntp_frac & 0xffff0000) >> 16);
+  }
+
+  state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
+
+  return state;
+}
+
+int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
+  if (rtcp_sender_.Sending() != sending) {
+    // Sends RTCP BYE when going from true to false
+    rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending);
+  }
+  return 0;
+}
+
+bool ModuleRtpRtcpImpl::Sending() const {
+  return rtcp_sender_.Sending();
+}
+
+void ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) {
+  rtp_sender_->packet_generator.SetSendingMediaStatus(sending);
+}
+
+bool ModuleRtpRtcpImpl::SendingMedia() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.SendingMedia() : false;
+}
+
+bool ModuleRtpRtcpImpl::IsAudioConfigured() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.IsAudioConfigured()
+                     : false;
+}
+
+void ModuleRtpRtcpImpl::SetAsPartOfAllocation(bool part_of_allocation) {
+  RTC_CHECK(rtp_sender_);
+  rtp_sender_->packet_sender.ForceIncludeSendPacketsInAllocation(
+      part_of_allocation);
+}
+
+bool ModuleRtpRtcpImpl::OnSendingRtpFrame(uint32_t timestamp,
+                                          int64_t capture_time_ms,
+                                          int payload_type,
+                                          bool force_sender_report) {
+  if (!Sending())
+    return false;
+
+  // TODO(bugs.webrtc.org/12873): Migrate this method and it's users to use
+  // optional Timestamps.
+  absl::optional<Timestamp> capture_time;
+  if (capture_time_ms > 0) {
+    capture_time = Timestamp::Millis(capture_time_ms);
+  }
+  absl::optional<int> payload_type_optional;
+  if (payload_type >= 0)
+    payload_type_optional = payload_type;
+  rtcp_sender_.SetLastRtpTime(timestamp, capture_time, payload_type_optional);
+  // Make sure an RTCP report isn't queued behind a key frame.
+  if (rtcp_sender_.TimeToSendRTCPReport(force_sender_report))
+    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+
+  return true;
+}
+
+bool ModuleRtpRtcpImpl::TrySendPacket(RtpPacketToSend* packet,
+                                      const PacedPacketInfo& pacing_info) {
+  RTC_DCHECK(rtp_sender_);
+  // TODO(sprang): Consider if we can remove this check.
+  if (!rtp_sender_->packet_generator.SendingMedia()) {
+    return false;
+  }
+  {
+    MutexLock lock(&rtp_sender_->sequencer_mutex);
+    if (packet->packet_type() == RtpPacketMediaType::kPadding &&
+        packet->Ssrc() == rtp_sender_->packet_generator.SSRC() &&
+        !rtp_sender_->sequencer_.CanSendPaddingOnMediaSsrc()) {
+      // New media packet preempted this generated padding packet, discard it.
+      return false;
+    }
+    bool is_flexfec =
+        packet->packet_type() == RtpPacketMediaType::kForwardErrorCorrection &&
+        packet->Ssrc() == rtp_sender_->packet_generator.FlexfecSsrc();
+    if (!is_flexfec) {
+      rtp_sender_->sequencer_.Sequence(*packet);
+    }
+  }
+  rtp_sender_->packet_sender.SendPacket(packet, pacing_info);
+  return true;
+}
+
+void ModuleRtpRtcpImpl::SetFecProtectionParams(const FecProtectionParams&,
+                                               const FecProtectionParams&) {
+  // Deferred FEC not supported in deprecated RTP module.
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+ModuleRtpRtcpImpl::FetchFecPackets() {
+  // Deferred FEC not supported in deprecated RTP module.
+  return {};
+}
+
+void ModuleRtpRtcpImpl::OnAbortedRetransmissions(
+    rtc::ArrayView<const uint16_t> sequence_numbers) {
+  RTC_DCHECK_NOTREACHED()
+      << "Stream flushing not supported with legacy rtp modules.";
+}
+
+void ModuleRtpRtcpImpl::OnPacketsAcknowledged(
+    rtc::ArrayView<const uint16_t> sequence_numbers) {
+  RTC_DCHECK(rtp_sender_);
+  rtp_sender_->packet_history.CullAcknowledgedPackets(sequence_numbers);
+}
+
+bool ModuleRtpRtcpImpl::SupportsPadding() const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_generator.SupportsPadding();
+}
+
+bool ModuleRtpRtcpImpl::SupportsRtxPayloadPadding() const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_generator.SupportsRtxPayloadPadding();
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+ModuleRtpRtcpImpl::GeneratePadding(size_t target_size_bytes) {
+  RTC_DCHECK(rtp_sender_);
+  MutexLock lock(&rtp_sender_->sequencer_mutex);
+  return rtp_sender_->packet_generator.GeneratePadding(
+      target_size_bytes, rtp_sender_->packet_sender.MediaHasBeenSent(),
+      rtp_sender_->sequencer_.CanSendPaddingOnMediaSsrc());
+}
+
+std::vector<RtpSequenceNumberMap::Info>
+ModuleRtpRtcpImpl::GetSentRtpPacketInfos(
+    rtc::ArrayView<const uint16_t> sequence_numbers) const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_sender.GetSentRtpPacketInfos(sequence_numbers);
+}
+
+size_t ModuleRtpRtcpImpl::ExpectedPerPacketOverhead() const {
+  if (!rtp_sender_) {
+    return 0;
+  }
+  return rtp_sender_->packet_generator.ExpectedPerPacketOverhead();
+}
+
+void ModuleRtpRtcpImpl::OnPacketSendingThreadSwitched() {}
+
+size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_generator.MaxRtpPacketSize();
+}
+
+void ModuleRtpRtcpImpl::SetMaxRtpPacketSize(size_t rtp_packet_size) {
+  RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE)
+      << "rtp packet size too large: " << rtp_packet_size;
+  RTC_DCHECK_GT(rtp_packet_size, packet_overhead_)
+      << "rtp packet size too small: " << rtp_packet_size;
+
+  rtcp_sender_.SetMaxRtpPacketSize(rtp_packet_size);
+  if (rtp_sender_) {
+    rtp_sender_->packet_generator.SetMaxRtpPacketSize(rtp_packet_size);
+  }
+}
+
+RtcpMode ModuleRtpRtcpImpl::RTCP() const {
+  return rtcp_sender_.Status();
+}
+
+// Configure RTCP status i.e on/off.
+void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) {
+  rtcp_sender_.SetRTCPStatus(method);
+}
+
+int32_t ModuleRtpRtcpImpl::SetCNAME(absl::string_view c_name) {
+  return rtcp_sender_.SetCNAME(c_name);
+}
+
+int32_t ModuleRtpRtcpImpl::RemoteNTP(uint32_t* received_ntpsecs,
+                                     uint32_t* received_ntpfrac,
+                                     uint32_t* rtcp_arrival_time_secs,
+                                     uint32_t* rtcp_arrival_time_frac,
+                                     uint32_t* rtcp_timestamp) const {
+  return rtcp_receiver_.NTP(received_ntpsecs, received_ntpfrac,
+                            rtcp_arrival_time_secs, rtcp_arrival_time_frac,
+                            rtcp_timestamp,
+                            /*remote_sender_packet_count=*/nullptr,
+                            /*remote_sender_octet_count=*/nullptr,
+                            /*remote_sender_reports_count=*/nullptr)
+             ? 0
+             : -1;
+}
+
+// Get RoundTripTime.
+int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc,
+                               int64_t* rtt,
+                               int64_t* avg_rtt,
+                               int64_t* min_rtt,
+                               int64_t* max_rtt) const {
+  int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
+  if (rtt && *rtt == 0) {
+    // Try to get RTT from RtcpRttStats class.
+    *rtt = rtt_ms();
+  }
+  return ret;
+}
+
+int64_t ModuleRtpRtcpImpl::ExpectedRetransmissionTimeMs() const {
+  int64_t expected_retransmission_time_ms = rtt_ms();
+  if (expected_retransmission_time_ms > 0) {
+    return expected_retransmission_time_ms;
+  }
+  // No rtt available (`kRtpRtcpRttProcessTimeMs` not yet passed?), so try to
+  // poll avg_rtt_ms directly from rtcp receiver.
+  if (rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), nullptr,
+                         &expected_retransmission_time_ms, nullptr,
+                         nullptr) == 0) {
+    return expected_retransmission_time_ms;
+  }
+  return kDefaultExpectedRetransmissionTimeMs;
+}
+
+// Force a send of an RTCP packet.
+// Normal SR and RR are triggered via the process function.
+int32_t ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) {
+  return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type);
+}
+
+void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
+    StreamDataCounters* rtp_counters,
+    StreamDataCounters* rtx_counters) const {
+  rtp_sender_->packet_sender.GetDataCounters(rtp_counters, rtx_counters);
+}
+
+// Received RTCP report.
+void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo(
+    uint32_t* packet_count, uint32_t* octet_count, int64_t* ntp_timestamp_ms,
+    int64_t* remote_ntp_timestamp_ms) const {
+  return rtcp_receiver_.RemoteRTCPSenderInfo(
+      packet_count, octet_count, ntp_timestamp_ms, remote_ntp_timestamp_ms);
+}
+
+std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
+    const {
+  return rtcp_receiver_.GetLatestReportBlockData();
+}
+
+absl::optional<RtpRtcpInterface::SenderReportStats>
+ModuleRtpRtcpImpl::GetSenderReportStats() const {
+  SenderReportStats stats;
+  uint32_t remote_timestamp_secs;
+  uint32_t remote_timestamp_frac;
+  uint32_t arrival_timestamp_secs;
+  uint32_t arrival_timestamp_frac;
+  if (rtcp_receiver_.NTP(&remote_timestamp_secs, &remote_timestamp_frac,
+                         &arrival_timestamp_secs, &arrival_timestamp_frac,
+                         /*rtcp_timestamp=*/nullptr, &stats.packets_sent,
+                         &stats.bytes_sent, &stats.reports_count)) {
+    stats.last_remote_timestamp.Set(remote_timestamp_secs,
+                                    remote_timestamp_frac);
+    stats.last_arrival_timestamp.Set(arrival_timestamp_secs,
+                                     arrival_timestamp_frac);
+    return stats;
+  }
+  return absl::nullopt;
+}
+
+absl::optional<RtpRtcpInterface::NonSenderRttStats>
+ModuleRtpRtcpImpl::GetNonSenderRttStats() const {
+  // This is not implemented for this legacy class.
+  return absl::nullopt;
+}
+
+// (REMB) Receiver Estimated Max Bitrate.
+void ModuleRtpRtcpImpl::SetRemb(int64_t bitrate_bps,
+                                std::vector<uint32_t> ssrcs) {
+  rtcp_sender_.SetRemb(bitrate_bps, std::move(ssrcs));
+}
+
+void ModuleRtpRtcpImpl::UnsetRemb() {
+  rtcp_sender_.UnsetRemb();
+}
+
+void ModuleRtpRtcpImpl::SetExtmapAllowMixed(bool extmap_allow_mixed) {
+  rtp_sender_->packet_generator.SetExtmapAllowMixed(extmap_allow_mixed);
+}
+
+void ModuleRtpRtcpImpl::RegisterRtpHeaderExtension(absl::string_view uri,
+                                                   int id) {
+  bool registered =
+      rtp_sender_->packet_generator.RegisterRtpHeaderExtension(uri, id);
+  RTC_CHECK(registered);
+}
+
+void ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
+    absl::string_view uri) {
+  rtp_sender_->packet_generator.DeregisterRtpHeaderExtension(uri);
+}
+
+void ModuleRtpRtcpImpl::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
+  rtcp_sender_.SetTmmbn(std::move(bounding_set));
+}
+
+// Send a Negative acknowledgment packet.
+int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
+                                    const uint16_t size) {
+  uint16_t nack_length = size;
+  uint16_t start_id = 0;
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (TimeToSendFullNackList(now_ms)) {
+    nack_last_time_sent_full_ms_ = now_ms;
+  } else {
+    // Only send extended list.
+    if (nack_last_seq_number_sent_ == nack_list[size - 1]) {
+      // Last sequence number is the same, do not send list.
+      return 0;
+    }
+    // Send new sequence numbers.
+    for (int i = 0; i < size; ++i) {
+      if (nack_last_seq_number_sent_ == nack_list[i]) {
+        start_id = i + 1;
+        break;
+      }
+    }
+    nack_length = size - start_id;
+  }
+
+  // Our RTCP NACK implementation is limited to kRtcpMaxNackFields sequence
+  // numbers per RTCP packet.
+  if (nack_length > kRtcpMaxNackFields) {
+    nack_length = kRtcpMaxNackFields;
+  }
+  nack_last_seq_number_sent_ = nack_list[start_id + nack_length - 1];
+
+  return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, nack_length,
+                               &nack_list[start_id]);
+}
+
+void ModuleRtpRtcpImpl::SendNack(
+    const std::vector<uint16_t>& sequence_numbers) {
+  rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, sequence_numbers.size(),
+                        sequence_numbers.data());
+}
+
+bool ModuleRtpRtcpImpl::TimeToSendFullNackList(int64_t now) const {
+  // Use RTT from RtcpRttStats class if provided.
+  int64_t rtt = rtt_ms();
+  if (rtt == 0) {
+    rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
+  }
+
+  const int64_t kStartUpRttMs = 100;
+  int64_t wait_time = 5 + ((rtt * 3) >> 1);  // 5 + RTT * 1.5.
+  if (rtt == 0) {
+    wait_time = kStartUpRttMs;
+  }
+
+  // Send a full NACK list once within every `wait_time`.
+  return now - nack_last_time_sent_full_ms_ > wait_time;
+}
+
+// Store the sent packets, needed to answer to Negative acknowledgment requests.
+void ModuleRtpRtcpImpl::SetStorePacketsStatus(const bool enable,
+                                              const uint16_t number_to_store) {
+  rtp_sender_->packet_history.SetStorePacketsStatus(
+      enable ? RtpPacketHistory::StorageMode::kStoreAndCull
+             : RtpPacketHistory::StorageMode::kDisabled,
+      number_to_store);
+}
+
+bool ModuleRtpRtcpImpl::StorePackets() const {
+  return rtp_sender_->packet_history.GetStorageMode() !=
+         RtpPacketHistory::StorageMode::kDisabled;
+}
+
+void ModuleRtpRtcpImpl::SendCombinedRtcpPacket(
+    std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
+  rtcp_sender_.SendCombinedRtcpPacket(std::move(rtcp_packets));
+}
+
+int32_t ModuleRtpRtcpImpl::SendLossNotification(uint16_t last_decoded_seq_num,
+                                                uint16_t last_received_seq_num,
+                                                bool decodability_flag,
+                                                bool buffering_allowed) {
+  return rtcp_sender_.SendLossNotification(
+      GetFeedbackState(), last_decoded_seq_num, last_received_seq_num,
+      decodability_flag, buffering_allowed);
+}
+
+void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
+  // Inform about the incoming SSRC.
+  rtcp_sender_.SetRemoteSSRC(ssrc);
+  rtcp_receiver_.SetRemoteSSRC(ssrc);
+}
+
+void ModuleRtpRtcpImpl::SetLocalSsrc(uint32_t local_ssrc) {
+  rtcp_receiver_.set_local_media_ssrc(local_ssrc);
+  rtcp_sender_.SetSsrc(local_ssrc);
+}
+
+RtpSendRates ModuleRtpRtcpImpl::GetSendRates() const {
+  return rtp_sender_->packet_sender.GetSendRates();
+}
+
+void ModuleRtpRtcpImpl::OnRequestSendReport() {
+  SendRTCP(kRtcpSr);
+}
+
+void ModuleRtpRtcpImpl::OnReceivedNack(
+    const std::vector<uint16_t>& nack_sequence_numbers) {
+  if (!rtp_sender_)
+    return;
+
+  if (!StorePackets() || nack_sequence_numbers.empty()) {
+    return;
+  }
+  // Use RTT from RtcpRttStats class if provided.
+  int64_t rtt = rtt_ms();
+  if (rtt == 0) {
+    rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
+  }
+  rtp_sender_->packet_generator.OnReceivedNack(nack_sequence_numbers, rtt);
+}
+
+void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
+    const ReportBlockList& report_blocks) {
+  if (rtp_sender_) {
+    uint32_t ssrc = SSRC();
+    absl::optional<uint32_t> rtx_ssrc;
+    if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) {
+      rtx_ssrc = rtp_sender_->packet_generator.RtxSsrc();
+    }
+
+    for (const RTCPReportBlock& report_block : report_blocks) {
+      if (ssrc == report_block.source_ssrc) {
+        rtp_sender_->packet_generator.OnReceivedAckOnSsrc(
+            report_block.extended_highest_sequence_number);
+      } else if (rtx_ssrc && *rtx_ssrc == report_block.source_ssrc) {
+        rtp_sender_->packet_generator.OnReceivedAckOnRtxSsrc(
+            report_block.extended_highest_sequence_number);
+      }
+    }
+  }
+}
+
+void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
+  {
+    MutexLock lock(&mutex_rtt_);
+    rtt_ms_ = rtt_ms;
+  }
+  if (rtp_sender_) {
+    rtp_sender_->packet_history.SetRtt(TimeDelta::Millis(rtt_ms));
+  }
+}
+
+int64_t ModuleRtpRtcpImpl::rtt_ms() const {
+  MutexLock lock(&mutex_rtt_);
+  return rtt_ms_;
+}
+
+void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
+    const VideoBitrateAllocation& bitrate) {
+  rtcp_sender_.SetVideoBitrateAllocation(bitrate);
+}
+
+RTPSender* ModuleRtpRtcpImpl::RtpSender() {
+  return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
+}
+
+const RTPSender* ModuleRtpRtcpImpl::RtpSender() const {
+  return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
new file mode 100644
index 0000000000..6070b67d44
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -0,0 +1,332 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/rtp_headers.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/include/module_fec_types.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // RTCPPacketType
+#include "modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h"
+#include "modules/rtp_rtcp/source/packet_sequencer.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+#include "modules/rtp_rtcp/source/rtcp_receiver.h"
+#include "modules/rtp_rtcp/source/rtcp_sender.h"
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_sender.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class Clock;
+struct PacedPacketInfo;
+struct RTPVideoHeader;
+
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+class ABSL_DEPRECATED("") ModuleRtpRtcpImpl
+    : public RtpRtcp,
+      public RTCPReceiver::ModuleRtpRtcp {
+#pragma clang diagnostic pop
+ public:
+  explicit ModuleRtpRtcpImpl(
+      const RtpRtcpInterface::Configuration& configuration);
+  ~ModuleRtpRtcpImpl() override;
+
+  // Process any pending tasks such as timeouts.
+  void Process() override;
+
+  // Receiver part.
+
+  // Called when we receive an RTCP packet.
+  void IncomingRtcpPacket(const uint8_t* incoming_packet,
+                          size_t incoming_packet_length) override;
+
+  void SetRemoteSSRC(uint32_t ssrc) override;
+  void SetLocalSsrc(uint32_t ssrc) override;
+
+  // Sender part.
+  void RegisterSendPayloadFrequency(int payload_type,
+                                    int payload_frequency) override;
+
+  int32_t DeRegisterSendPayload(int8_t payload_type) override;
+
+  void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
+
+  // Register RTP header extension.
+  void RegisterRtpHeaderExtension(absl::string_view uri, int id) override;
+  void DeregisterSendRtpHeaderExtension(absl::string_view uri) override;
+
+  bool SupportsPadding() const override;
+  bool SupportsRtxPayloadPadding() const override;
+
+  // Get start timestamp.
+  uint32_t StartTimestamp() const override;
+
+  // Configure start timestamp, default is a random number.
+  void SetStartTimestamp(uint32_t timestamp) override;
+
+  uint16_t SequenceNumber() const override;
+
+  // Set SequenceNumber, default is a random number.
+  void SetSequenceNumber(uint16_t seq) override;
+
+  void SetRtpState(const RtpState& rtp_state) override;
+  void SetRtxState(const RtpState& rtp_state) override;
+  RtpState GetRtpState() const override;
+  RtpState GetRtxState() const override;
+
+  void SetNonSenderRttMeasurement(bool enabled) override {}
+
+  uint32_t SSRC() const override { return rtcp_sender_.SSRC(); }
+
+  void SetMid(absl::string_view mid) override;
+
+  RTCPSender::FeedbackState GetFeedbackState();
+
+  void SetRtxSendStatus(int mode) override;
+  int RtxSendStatus() const override;
+  absl::optional<uint32_t> RtxSsrc() const override;
+
+  void SetRtxSendPayloadType(int payload_type,
+                             int associated_payload_type) override;
+
+  absl::optional<uint32_t> FlexfecSsrc() const override;
+
+  // Sends kRtcpByeCode when going from true to false.
+  int32_t SetSendingStatus(bool sending) override;
+
+  bool Sending() const override;
+
+  // Drops or relays media packets.
+  void SetSendingMediaStatus(bool sending) override;
+
+  bool SendingMedia() const override;
+
+  bool IsAudioConfigured() const override;
+
+  void SetAsPartOfAllocation(bool part_of_allocation) override;
+
+  bool OnSendingRtpFrame(uint32_t timestamp,
+                         int64_t capture_time_ms,
+                         int payload_type,
+                         bool force_sender_report) override;
+
+  bool TrySendPacket(RtpPacketToSend* packet,
+                     const PacedPacketInfo& pacing_info) override;
+
+  void SetFecProtectionParams(const FecProtectionParams& delta_params,
+                              const FecProtectionParams& key_params) override;
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() override;
+
+  void OnAbortedRetransmissions(
+      rtc::ArrayView<const uint16_t> sequence_numbers) override;
+
+  void OnPacketsAcknowledged(
+      rtc::ArrayView<const uint16_t> sequence_numbers) override;
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      size_t target_size_bytes) override;
+
+  std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+      rtc::ArrayView<const uint16_t> sequence_numbers) const override;
+
+  size_t ExpectedPerPacketOverhead() const override;
+
+  void OnPacketSendingThreadSwitched() override;
+
+  // RTCP part.
+
+  // Get RTCP status.
+  RtcpMode RTCP() const override;
+
+  // Configure RTCP status i.e on/off.
+  void SetRTCPStatus(RtcpMode method) override;
+
+  // Set RTCP CName.
+  int32_t SetCNAME(absl::string_view c_name) override;
+
+  // Get remote NTP.
+  int32_t RemoteNTP(uint32_t* received_ntp_secs,
+                    uint32_t* received_ntp_frac,
+                    uint32_t* rtcp_arrival_time_secs,
+                    uint32_t* rtcp_arrival_time_frac,
+                    uint32_t* rtcp_timestamp) const override;
+
+  // Get RoundTripTime.
+  int32_t RTT(uint32_t remote_ssrc,
+              int64_t* rtt,
+              int64_t* avg_rtt,
+              int64_t* min_rtt,
+              int64_t* max_rtt) const override;
+
+  int64_t ExpectedRetransmissionTimeMs() const override;
+
+  // Force a send of an RTCP packet.
+  // Normal SR and RR are triggered via the process function.
+  int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;
+
+  void GetSendStreamDataCounters(
+      StreamDataCounters* rtp_counters,
+      StreamDataCounters* rtx_counters) const override;
+
+  void RemoteRTCPSenderInfo(uint32_t* packet_count,
+                            uint32_t* octet_count,
+                            int64_t* ntp_timestamp_ms,
+                            int64_t* remote_ntp_timestamp_ms) const override;
+
+  // A snapshot of the most recent Report Block with additional data of
+  // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
+  // Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
+  // which is the SSRC of the corresponding outbound RTP stream, is unique.
+  std::vector<ReportBlockData> GetLatestReportBlockData() const override;
+  absl::optional<SenderReportStats> GetSenderReportStats() const override;
+  // Round trip time statistics computed from the XR block contained in the last
+  // report.
+  absl::optional<NonSenderRttStats> GetNonSenderRttStats() const override;
+
+  // (REMB) Receiver Estimated Max Bitrate.
+  void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
+  void UnsetRemb() override;
+
+  void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override;
+
+  size_t MaxRtpPacketSize() const override;
+
+  void SetMaxRtpPacketSize(size_t max_packet_size) override;
+
+  // (NACK) Negative acknowledgment part.
+
+  // Send a Negative acknowledgment packet.
+  // TODO(philipel): Deprecate SendNACK and use SendNack instead.
+  int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override;
+
+  void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
+
+  // Store the sent packets, needed to answer to a negative acknowledgment
+  // requests.
+  void SetStorePacketsStatus(bool enable, uint16_t number_to_store) override;
+
+  void SendCombinedRtcpPacket(
+      std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) override;
+
+  // Video part.
+  int32_t SendLossNotification(uint16_t last_decoded_seq_num,
+                               uint16_t last_received_seq_num,
+                               bool decodability_flag,
+                               bool buffering_allowed) override;
+
+  RtpSendRates GetSendRates() const override;
+
+  void OnReceivedNack(
+      const std::vector<uint16_t>& nack_sequence_numbers) override;
+  void OnReceivedRtcpReportBlocks(
+      const ReportBlockList& report_blocks) override;
+  void OnRequestSendReport() override;
+
+  void SetVideoBitrateAllocation(
+      const VideoBitrateAllocation& bitrate) override;
+
+  RTPSender* RtpSender() override;
+  const RTPSender* RtpSender() const override;
+
+ protected:
+  bool UpdateRTCPReceiveInformationTimers();
+
+  RTPSender* rtp_sender() {
+    return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
+  }
+  const RTPSender* rtp_sender() const {
+    return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
+  }
+
+  RTCPSender* rtcp_sender() { return &rtcp_sender_; }
+  const RTCPSender* rtcp_sender() const { return &rtcp_sender_; }
+
+  RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; }
+  const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; }
+
+  void SetMediaHasBeenSent(bool media_has_been_sent) {
+    rtp_sender_->packet_sender.SetMediaHasBeenSent(media_has_been_sent);
+  }
+
+  Clock* clock() const { return clock_; }
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
+  FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
+
+  struct RtpSenderContext {
+    explicit RtpSenderContext(const RtpRtcpInterface::Configuration& config);
+    // Storage of packets, for retransmissions and padding, if applicable.
+    RtpPacketHistory packet_history;
+    // Handles sequence number assignment and padding timestamp generation.
+    mutable Mutex sequencer_mutex;
+    PacketSequencer sequencer_ RTC_GUARDED_BY(sequencer_mutex);
+    // Handles final time timestamping/stats/etc and handover to Transport.
+    DEPRECATED_RtpSenderEgress packet_sender;
+    // If no paced sender configured, this class will be used to pass packets
+    // from `packet_generator_` to `packet_sender_`.
+    DEPRECATED_RtpSenderEgress::NonPacedPacketSender non_paced_sender;
+    // Handles creation of RTP packets to be sent.
+    RTPSender packet_generator;
+  };
+
+  void set_rtt_ms(int64_t rtt_ms);
+  int64_t rtt_ms() const;
+
+  bool TimeToSendFullNackList(int64_t now) const;
+
+  // Returns true if the module is configured to store packets.
+  bool StorePackets() const;
+
+  // Returns current Receiver Reference Time Report (RTTR) status.
+  bool RtcpXrRrtrStatus() const;
+
+  std::unique_ptr<RtpSenderContext> rtp_sender_;
+
+  RTCPSender rtcp_sender_;
+  RTCPReceiver rtcp_receiver_;
+
+  Clock* const clock_;
+
+  int64_t last_bitrate_process_time_;
+  int64_t last_rtt_process_time_;
+  uint16_t packet_overhead_;
+
+  // Send side
+  int64_t nack_last_time_sent_full_ms_;
+  uint16_t nack_last_seq_number_sent_;
+
+  RtcpRttStats* const rtt_stats_;
+
+  // The processed RTT from RtcpRttStats.
+  mutable Mutex mutex_rtt_;
+  int64_t rtt_ms_ RTC_GUARDED_BY(mutex_rtt_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
new file mode 100644
index 0000000000..66d2e7a44e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -0,0 +1,832 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/ntp_time.h"
+
+#ifdef _WIN32
+// Disable warning C4355: 'this' : used in base member initializer list.
+#pragma warning(disable : 4355)
+#endif
+
+namespace webrtc {
+namespace {
+const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
+
+constexpr TimeDelta kRttUpdateInterval = TimeDelta::Millis(1000);
+
+RTCPSender::Configuration AddRtcpSendEvaluationCallback(
+    RTCPSender::Configuration config,
+    std::function<void(TimeDelta)> send_evaluation_callback) {
+  config.schedule_next_rtcp_send_evaluation_function =
+      std::move(send_evaluation_callback);
+  return config;
+}
+
+}  // namespace
+
+ModuleRtpRtcpImpl2::RtpSenderContext::RtpSenderContext(
+    const RtpRtcpInterface::Configuration& config)
+    : packet_history(config.clock, config.enable_rtx_padding_prioritization),
+      sequencer(config.local_media_ssrc,
+                config.rtx_send_ssrc,
+                /*require_marker_before_media_padding=*/!config.audio,
+                config.clock),
+      packet_sender(config, &packet_history),
+      non_paced_sender(&packet_sender, &sequencer),
+      packet_generator(
+          config,
+          &packet_history,
+          config.paced_sender ? config.paced_sender : &non_paced_sender) {}
+
+ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
+    : worker_queue_(TaskQueueBase::Current()),
+      rtcp_sender_(AddRtcpSendEvaluationCallback(
+          RTCPSender::Configuration::FromRtpRtcpConfiguration(configuration),
+          [this](TimeDelta duration) {
+            ScheduleRtcpSendEvaluation(duration);
+          })),
+      rtcp_receiver_(configuration, this),
+      clock_(configuration.clock),
+      packet_overhead_(28),  // IPV4 UDP.
+      nack_last_time_sent_full_ms_(0),
+      nack_last_seq_number_sent_(0),
+      rtt_stats_(configuration.rtt_stats),
+      rtt_ms_(0) {
+  RTC_DCHECK(worker_queue_);
+  rtcp_thread_checker_.Detach();
+  if (!configuration.receiver_only) {
+    rtp_sender_ = std::make_unique<RtpSenderContext>(configuration);
+    rtp_sender_->sequencing_checker.Detach();
+    // Make sure rtcp sender use same timestamp offset as rtp sender.
+    rtcp_sender_.SetTimestampOffset(
+        rtp_sender_->packet_generator.TimestampOffset());
+    rtp_sender_->packet_sender.SetTimestampOffset(
+        rtp_sender_->packet_generator.TimestampOffset());
+  }
+
+  // Set default packet size limit.
+  // TODO(nisse): Kind-of duplicates
+  // webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize.
+  const size_t kTcpOverIpv4HeaderSize = 40;
+  SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize);
+  rtt_update_task_ = RepeatingTaskHandle::DelayedStart(
+      worker_queue_, kRttUpdateInterval, [this]() {
+        PeriodicUpdate();
+        return kRttUpdateInterval;
+      });
+}
+
+ModuleRtpRtcpImpl2::~ModuleRtpRtcpImpl2() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  rtt_update_task_.Stop();
+}
+
+// static
+std::unique_ptr<ModuleRtpRtcpImpl2> ModuleRtpRtcpImpl2::Create(
+    const Configuration& configuration) {
+  RTC_DCHECK(configuration.clock);
+  RTC_DCHECK(TaskQueueBase::Current());
+  return std::make_unique<ModuleRtpRtcpImpl2>(configuration);
+}
+
+void ModuleRtpRtcpImpl2::SetRtxSendStatus(int mode) {
+  rtp_sender_->packet_generator.SetRtxStatus(mode);
+}
+
+int ModuleRtpRtcpImpl2::RtxSendStatus() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.RtxStatus() : kRtxOff;
+}
+
+void ModuleRtpRtcpImpl2::SetRtxSendPayloadType(int payload_type,
+                                               int associated_payload_type) {
+  rtp_sender_->packet_generator.SetRtxPayloadType(payload_type,
+                                                  associated_payload_type);
+}
+
+absl::optional<uint32_t> ModuleRtpRtcpImpl2::RtxSsrc() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : absl::nullopt;
+}
+
+absl::optional<uint32_t> ModuleRtpRtcpImpl2::FlexfecSsrc() const {
+  if (rtp_sender_) {
+    return rtp_sender_->packet_generator.FlexfecSsrc();
+  }
+  return absl::nullopt;
+}
+
+void ModuleRtpRtcpImpl2::IncomingRtcpPacket(const uint8_t* rtcp_packet,
+                                            const size_t length) {
+  RTC_DCHECK_RUN_ON(&rtcp_thread_checker_);
+  rtcp_receiver_.IncomingPacket(rtcp_packet, length);
+}
+
+void ModuleRtpRtcpImpl2::RegisterSendPayloadFrequency(int payload_type,
+                                                      int payload_frequency) {
+  rtcp_sender_.SetRtpClockRate(payload_type, payload_frequency);
+}
+
+int32_t ModuleRtpRtcpImpl2::DeRegisterSendPayload(const int8_t payload_type) {
+  return 0;
+}
+
+uint32_t ModuleRtpRtcpImpl2::StartTimestamp() const {
+  return rtp_sender_->packet_generator.TimestampOffset();
+}
+
+// Configure start timestamp, default is a random number.
+void ModuleRtpRtcpImpl2::SetStartTimestamp(const uint32_t timestamp) {
+  rtcp_sender_.SetTimestampOffset(timestamp);
+  rtp_sender_->packet_generator.SetTimestampOffset(timestamp);
+  rtp_sender_->packet_sender.SetTimestampOffset(timestamp);
+}
+
+uint16_t ModuleRtpRtcpImpl2::SequenceNumber() const {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  return rtp_sender_->sequencer.media_sequence_number();
+}
+
+// Set SequenceNumber, default is a random number.
+void ModuleRtpRtcpImpl2::SetSequenceNumber(const uint16_t seq_num) {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  if (rtp_sender_->sequencer.media_sequence_number() != seq_num) {
+    rtp_sender_->sequencer.set_media_sequence_number(seq_num);
+    rtp_sender_->packet_history.Clear();
+  }
+}
+
+void ModuleRtpRtcpImpl2::SetRtpState(const RtpState& rtp_state) {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  rtp_sender_->packet_generator.SetRtpState(rtp_state);
+  rtp_sender_->sequencer.SetRtpState(rtp_state);
+  rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
+  rtp_sender_->packet_sender.SetTimestampOffset(rtp_state.start_timestamp);
+}
+
+void ModuleRtpRtcpImpl2::SetRtxState(const RtpState& rtp_state) {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  rtp_sender_->packet_generator.SetRtxRtpState(rtp_state);
+  rtp_sender_->sequencer.set_rtx_sequence_number(rtp_state.sequence_number);
+}
+
+RtpState ModuleRtpRtcpImpl2::GetRtpState() const {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  RtpState state = rtp_sender_->packet_generator.GetRtpState();
+  rtp_sender_->sequencer.PopulateRtpState(state);
+  return state;
+}
+
+RtpState ModuleRtpRtcpImpl2::GetRtxState() const {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  RtpState state = rtp_sender_->packet_generator.GetRtxRtpState();
+  state.sequence_number = rtp_sender_->sequencer.rtx_sequence_number();
+  return state;
+}
+
+void ModuleRtpRtcpImpl2::SetNonSenderRttMeasurement(bool enabled) {
+  rtcp_sender_.SetNonSenderRttMeasurement(enabled);
+  rtcp_receiver_.SetNonSenderRttMeasurement(enabled);
+}
+
+uint32_t ModuleRtpRtcpImpl2::local_media_ssrc() const {
+  RTC_DCHECK_RUN_ON(&rtcp_thread_checker_);
+  RTC_DCHECK_EQ(rtcp_receiver_.local_media_ssrc(), rtcp_sender_.SSRC());
+  return rtcp_receiver_.local_media_ssrc();
+}
+
+void ModuleRtpRtcpImpl2::SetMid(absl::string_view mid) {
+  if (rtp_sender_) {
+    rtp_sender_->packet_generator.SetMid(mid);
+  }
+  // TODO(bugs.webrtc.org/4050): If we end up supporting the MID SDES item for
+  // RTCP, this will need to be passed down to the RTCPSender also.
+}
+
+// TODO(pbos): Handle media and RTX streams separately (separate RTCP
+// feedbacks).
+RTCPSender::FeedbackState ModuleRtpRtcpImpl2::GetFeedbackState() {
+  // TODO(bugs.webrtc.org/11581): Called by potentially multiple threads.
+  // Mostly "Send*" methods. Make sure it's only called on the
+  // construction thread.
+
+  RTCPSender::FeedbackState state;
+  // This is called also when receiver_only is true. Hence below
+  // checks that rtp_sender_ exists.
+  if (rtp_sender_) {
+    StreamDataCounters rtp_stats;
+    StreamDataCounters rtx_stats;
+    rtp_sender_->packet_sender.GetDataCounters(&rtp_stats, &rtx_stats);
+    state.packets_sent =
+        rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
+    state.media_bytes_sent = rtp_stats.transmitted.payload_bytes +
+                             rtx_stats.transmitted.payload_bytes;
+    state.send_bitrate =
+        rtp_sender_->packet_sender.GetSendRates().Sum().bps<uint32_t>();
+  }
+  state.receiver = &rtcp_receiver_;
+
+  uint32_t received_ntp_secs = 0;
+  uint32_t received_ntp_frac = 0;
+  state.remote_sr = 0;
+  if (rtcp_receiver_.NTP(&received_ntp_secs, &received_ntp_frac,
+                         /*rtcp_arrival_time_secs=*/&state.last_rr_ntp_secs,
+                         /*rtcp_arrival_time_frac=*/&state.last_rr_ntp_frac,
+                         /*rtcp_timestamp=*/nullptr,
+                         /*remote_sender_packet_count=*/nullptr,
+                         /*remote_sender_octet_count=*/nullptr,
+                         /*remote_sender_reports_count=*/nullptr)) {
+    state.remote_sr = ((received_ntp_secs & 0x0000ffff) << 16) +
+                      ((received_ntp_frac & 0xffff0000) >> 16);
+  }
+
+  state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
+
+  return state;
+}
+
+int32_t ModuleRtpRtcpImpl2::SetSendingStatus(const bool sending) {
+  if (rtcp_sender_.Sending() != sending) {
+    // Sends RTCP BYE when going from true to false
+    rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending);
+  }
+  return 0;
+}
+
+bool ModuleRtpRtcpImpl2::Sending() const {
+  return rtcp_sender_.Sending();
+}
+
+void ModuleRtpRtcpImpl2::SetSendingMediaStatus(const bool sending) {
+  rtp_sender_->packet_generator.SetSendingMediaStatus(sending);
+}
+
+bool ModuleRtpRtcpImpl2::SendingMedia() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.SendingMedia() : false;
+}
+
+bool ModuleRtpRtcpImpl2::IsAudioConfigured() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.IsAudioConfigured()
+                     : false;
+}
+
+void ModuleRtpRtcpImpl2::SetAsPartOfAllocation(bool part_of_allocation) {
+  RTC_CHECK(rtp_sender_);
+  rtp_sender_->packet_sender.ForceIncludeSendPacketsInAllocation(
+      part_of_allocation);
+}
+
+bool ModuleRtpRtcpImpl2::OnSendingRtpFrame(uint32_t timestamp,
+                                           int64_t capture_time_ms,
+                                           int payload_type,
+                                           bool force_sender_report) {
+  if (!Sending())
+    return false;
+
+  // TODO(bugs.webrtc.org/12873): Migrate this method and it's users to use
+  // optional Timestamps.
+  absl::optional<Timestamp> capture_time;
+  if (capture_time_ms > 0) {
+    capture_time = Timestamp::Millis(capture_time_ms);
+  }
+  absl::optional<int> payload_type_optional;
+  if (payload_type >= 0)
+    payload_type_optional = payload_type;
+  rtcp_sender_.SetLastRtpTime(timestamp, capture_time, payload_type_optional);
+  // Make sure an RTCP report isn't queued behind a key frame.
+  if (rtcp_sender_.TimeToSendRTCPReport(force_sender_report))
+    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+
+  return true;
+}
+
+bool ModuleRtpRtcpImpl2::TrySendPacket(RtpPacketToSend* packet,
+                                       const PacedPacketInfo& pacing_info) {
+  RTC_DCHECK(rtp_sender_);
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  if (!rtp_sender_->packet_generator.SendingMedia()) {
+    return false;
+  }
+  if (packet->packet_type() == RtpPacketMediaType::kPadding &&
+      packet->Ssrc() == rtp_sender_->packet_generator.SSRC() &&
+      !rtp_sender_->sequencer.CanSendPaddingOnMediaSsrc()) {
+    // New media packet preempted this generated padding packet, discard it.
+    return false;
+  }
+  bool is_flexfec =
+      packet->packet_type() == RtpPacketMediaType::kForwardErrorCorrection &&
+      packet->Ssrc() == rtp_sender_->packet_generator.FlexfecSsrc();
+  if (!is_flexfec) {
+    rtp_sender_->sequencer.Sequence(*packet);
+  }
+
+  rtp_sender_->packet_sender.SendPacket(packet, pacing_info);
+  return true;
+}
+
+void ModuleRtpRtcpImpl2::SetFecProtectionParams(
+    const FecProtectionParams& delta_params,
+    const FecProtectionParams& key_params) {
+  RTC_DCHECK(rtp_sender_);
+  rtp_sender_->packet_sender.SetFecProtectionParameters(delta_params,
+                                                        key_params);
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+ModuleRtpRtcpImpl2::FetchFecPackets() {
+  RTC_DCHECK(rtp_sender_);
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  return rtp_sender_->packet_sender.FetchFecPackets();
+}
+
+void ModuleRtpRtcpImpl2::OnAbortedRetransmissions(
+    rtc::ArrayView<const uint16_t> sequence_numbers) {
+  RTC_DCHECK(rtp_sender_);
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  rtp_sender_->packet_sender.OnAbortedRetransmissions(sequence_numbers);
+}
+
+void ModuleRtpRtcpImpl2::OnPacketsAcknowledged(
+    rtc::ArrayView<const uint16_t> sequence_numbers) {
+  RTC_DCHECK(rtp_sender_);
+  rtp_sender_->packet_history.CullAcknowledgedPackets(sequence_numbers);
+}
+
+bool ModuleRtpRtcpImpl2::SupportsPadding() const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_generator.SupportsPadding();
+}
+
+bool ModuleRtpRtcpImpl2::SupportsRtxPayloadPadding() const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_generator.SupportsRtxPayloadPadding();
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+ModuleRtpRtcpImpl2::GeneratePadding(size_t target_size_bytes) {
+  RTC_DCHECK(rtp_sender_);
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+
+  return rtp_sender_->packet_generator.GeneratePadding(
+      target_size_bytes, rtp_sender_->packet_sender.MediaHasBeenSent(),
+      rtp_sender_->sequencer.CanSendPaddingOnMediaSsrc());
+}
+
+std::vector<RtpSequenceNumberMap::Info>
+ModuleRtpRtcpImpl2::GetSentRtpPacketInfos(
+    rtc::ArrayView<const uint16_t> sequence_numbers) const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_sender.GetSentRtpPacketInfos(sequence_numbers);
+}
+
+size_t ModuleRtpRtcpImpl2::ExpectedPerPacketOverhead() const {
+  if (!rtp_sender_) {
+    return 0;
+  }
+  return rtp_sender_->packet_generator.ExpectedPerPacketOverhead();
+}
+
+void ModuleRtpRtcpImpl2::OnPacketSendingThreadSwitched() {
+  // Ownership of sequencing is being transferred to another thread.
+  rtp_sender_->sequencing_checker.Detach();
+}
+
+size_t ModuleRtpRtcpImpl2::MaxRtpPacketSize() const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_generator.MaxRtpPacketSize();
+}
+
+void ModuleRtpRtcpImpl2::SetMaxRtpPacketSize(size_t rtp_packet_size) {
+  RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE)
+      << "rtp packet size too large: " << rtp_packet_size;
+  RTC_DCHECK_GT(rtp_packet_size, packet_overhead_)
+      << "rtp packet size too small: " << rtp_packet_size;
+
+  rtcp_sender_.SetMaxRtpPacketSize(rtp_packet_size);
+  if (rtp_sender_) {
+    rtp_sender_->packet_generator.SetMaxRtpPacketSize(rtp_packet_size);
+  }
+}
+
+RtcpMode ModuleRtpRtcpImpl2::RTCP() const {
+  return rtcp_sender_.Status();
+}
+
+// Configure RTCP status i.e on/off.
+void ModuleRtpRtcpImpl2::SetRTCPStatus(const RtcpMode method) {
+  rtcp_sender_.SetRTCPStatus(method);
+}
+
+int32_t ModuleRtpRtcpImpl2::SetCNAME(absl::string_view c_name) {
+  return rtcp_sender_.SetCNAME(c_name);
+}
+
+int32_t ModuleRtpRtcpImpl2::RemoteNTP(uint32_t* received_ntpsecs,
+                                      uint32_t* received_ntpfrac,
+                                      uint32_t* rtcp_arrival_time_secs,
+                                      uint32_t* rtcp_arrival_time_frac,
+                                      uint32_t* rtcp_timestamp) const {
+  return rtcp_receiver_.NTP(received_ntpsecs, received_ntpfrac,
+                            rtcp_arrival_time_secs, rtcp_arrival_time_frac,
+                            rtcp_timestamp,
+                            /*remote_sender_packet_count=*/nullptr,
+                            /*remote_sender_octet_count=*/nullptr,
+                            /*remote_sender_reports_count=*/nullptr)
+             ? 0
+             : -1;
+}
+
+// TODO(tommi): Check if `avg_rtt_ms`, `min_rtt_ms`, `max_rtt_ms` params are
+// actually used in practice (some callers ask for it but don't use it). It
+// could be that only `rtt` is needed and if so, then the fast path could be to
+// just call rtt_ms() and rely on the calculation being done periodically.
+int32_t ModuleRtpRtcpImpl2::RTT(const uint32_t remote_ssrc,
+                                int64_t* rtt,
+                                int64_t* avg_rtt,
+                                int64_t* min_rtt,
+                                int64_t* max_rtt) const {
+  int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
+  if (rtt && *rtt == 0) {
+    // Try to get RTT from RtcpRttStats class.
+    *rtt = rtt_ms();
+  }
+  return ret;
+}
+
+int64_t ModuleRtpRtcpImpl2::ExpectedRetransmissionTimeMs() const {
+  int64_t expected_retransmission_time_ms = rtt_ms();
+  if (expected_retransmission_time_ms > 0) {
+    return expected_retransmission_time_ms;
+  }
+  // No rtt available (`kRttUpdateInterval` not yet passed?), so try to
+  // poll avg_rtt_ms directly from rtcp receiver.
+  if (rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), nullptr,
+                         &expected_retransmission_time_ms, nullptr,
+                         nullptr) == 0) {
+    return expected_retransmission_time_ms;
+  }
+  return kDefaultExpectedRetransmissionTimeMs;
+}
+
+// Force a send of an RTCP packet.
+// Normal SR and RR are triggered via the process function.
+int32_t ModuleRtpRtcpImpl2::SendRTCP(RTCPPacketType packet_type) {
+  return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type);
+}
+
+void ModuleRtpRtcpImpl2::GetSendStreamDataCounters(
+    StreamDataCounters* rtp_counters,
+    StreamDataCounters* rtx_counters) const {
+  rtp_sender_->packet_sender.GetDataCounters(rtp_counters, rtx_counters);
+}
+
+// Received RTCP report.
+void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(
+    uint32_t* packet_count, uint32_t* octet_count, int64_t* ntp_timestamp_ms,
+    int64_t* remote_ntp_timestamp_ms) const {
+  return rtcp_receiver_.RemoteRTCPSenderInfo(
+      packet_count, octet_count, ntp_timestamp_ms, remote_ntp_timestamp_ms);
+}
+
+std::vector<ReportBlockData> ModuleRtpRtcpImpl2::GetLatestReportBlockData()
+    const {
+  return rtcp_receiver_.GetLatestReportBlockData();
+}
+
+absl::optional<RtpRtcpInterface::SenderReportStats>
+ModuleRtpRtcpImpl2::GetSenderReportStats() const {
+  SenderReportStats stats;
+  uint32_t remote_timestamp_secs;
+  uint32_t remote_timestamp_frac;
+  uint32_t arrival_timestamp_secs;
+  uint32_t arrival_timestamp_frac;
+  if (rtcp_receiver_.NTP(&remote_timestamp_secs, &remote_timestamp_frac,
+                         &arrival_timestamp_secs, &arrival_timestamp_frac,
+                         /*rtcp_timestamp=*/nullptr, &stats.packets_sent,
+                         &stats.bytes_sent, &stats.reports_count)) {
+    stats.last_remote_timestamp.Set(remote_timestamp_secs,
+                                    remote_timestamp_frac);
+    stats.last_arrival_timestamp.Set(arrival_timestamp_secs,
+                                     arrival_timestamp_frac);
+    return stats;
+  }
+  return absl::nullopt;
+}
+
+absl::optional<RtpRtcpInterface::NonSenderRttStats>
+ModuleRtpRtcpImpl2::GetNonSenderRttStats() const {
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      rtcp_receiver_.GetNonSenderRTT();
+  return {{
+      non_sender_rtt_stats.round_trip_time(),
+      non_sender_rtt_stats.total_round_trip_time(),
+      non_sender_rtt_stats.round_trip_time_measurements(),
+  }};
+}
+
+// (REMB) Receiver Estimated Max Bitrate.
+void ModuleRtpRtcpImpl2::SetRemb(int64_t bitrate_bps,
+                                 std::vector<uint32_t> ssrcs) {
+  rtcp_sender_.SetRemb(bitrate_bps, std::move(ssrcs));
+}
+
+void ModuleRtpRtcpImpl2::UnsetRemb() {
+  rtcp_sender_.UnsetRemb();
+}
+
+void ModuleRtpRtcpImpl2::SetExtmapAllowMixed(bool extmap_allow_mixed) {
+  rtp_sender_->packet_generator.SetExtmapAllowMixed(extmap_allow_mixed);
+}
+
+void ModuleRtpRtcpImpl2::RegisterRtpHeaderExtension(absl::string_view uri,
+                                                    int id) {
+  bool registered =
+      rtp_sender_->packet_generator.RegisterRtpHeaderExtension(uri, id);
+  RTC_CHECK(registered);
+}
+
+void ModuleRtpRtcpImpl2::DeregisterSendRtpHeaderExtension(
+    absl::string_view uri) {
+  rtp_sender_->packet_generator.DeregisterRtpHeaderExtension(uri);
+}
+
+void ModuleRtpRtcpImpl2::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
+  rtcp_sender_.SetTmmbn(std::move(bounding_set));
+}
+
+// Send a Negative acknowledgment packet.
+int32_t ModuleRtpRtcpImpl2::SendNACK(const uint16_t* nack_list,
+                                     const uint16_t size) {
+  uint16_t nack_length = size;
+  uint16_t start_id = 0;
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (TimeToSendFullNackList(now_ms)) {
+    nack_last_time_sent_full_ms_ = now_ms;
+  } else {
+    // Only send extended list.
+    if (nack_last_seq_number_sent_ == nack_list[size - 1]) {
+      // Last sequence number is the same, do not send list.
+      return 0;
+    }
+    // Send new sequence numbers.
+    for (int i = 0; i < size; ++i) {
+      if (nack_last_seq_number_sent_ == nack_list[i]) {
+        start_id = i + 1;
+        break;
+      }
+    }
+    nack_length = size - start_id;
+  }
+
+  // Our RTCP NACK implementation is limited to kRtcpMaxNackFields sequence
+  // numbers per RTCP packet.
+  if (nack_length > kRtcpMaxNackFields) {
+    nack_length = kRtcpMaxNackFields;
+  }
+  nack_last_seq_number_sent_ = nack_list[start_id + nack_length - 1];
+
+  return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, nack_length,
+                               &nack_list[start_id]);
+}
+
+void ModuleRtpRtcpImpl2::SendNack(
+    const std::vector<uint16_t>& sequence_numbers) {
+  rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, sequence_numbers.size(),
+                        sequence_numbers.data());
+}
+
+bool ModuleRtpRtcpImpl2::TimeToSendFullNackList(int64_t now) const {
+  // Use RTT from RtcpRttStats class if provided.
+  int64_t rtt = rtt_ms();
+  if (rtt == 0) {
+    rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
+  }
+
+  const int64_t kStartUpRttMs = 100;
+  int64_t wait_time = 5 + ((rtt * 3) >> 1);  // 5 + RTT * 1.5.
+  if (rtt == 0) {
+    wait_time = kStartUpRttMs;
+  }
+
+  // Send a full NACK list once within every `wait_time`.
+  return now - nack_last_time_sent_full_ms_ > wait_time;
+}
+
+// Store the sent packets, needed to answer to Negative acknowledgment requests.
+void ModuleRtpRtcpImpl2::SetStorePacketsStatus(const bool enable,
+                                               const uint16_t number_to_store) {
+  rtp_sender_->packet_history.SetStorePacketsStatus(
+      enable ? RtpPacketHistory::StorageMode::kStoreAndCull
+             : RtpPacketHistory::StorageMode::kDisabled,
+      number_to_store);
+}
+
+bool ModuleRtpRtcpImpl2::StorePackets() const {
+  return rtp_sender_->packet_history.GetStorageMode() !=
+         RtpPacketHistory::StorageMode::kDisabled;
+}
+
+void ModuleRtpRtcpImpl2::SendCombinedRtcpPacket(
+    std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
+  rtcp_sender_.SendCombinedRtcpPacket(std::move(rtcp_packets));
+}
+
+int32_t ModuleRtpRtcpImpl2::SendLossNotification(uint16_t last_decoded_seq_num,
+                                                 uint16_t last_received_seq_num,
+                                                 bool decodability_flag,
+                                                 bool buffering_allowed) {
+  return rtcp_sender_.SendLossNotification(
+      GetFeedbackState(), last_decoded_seq_num, last_received_seq_num,
+      decodability_flag, buffering_allowed);
+}
+
+void ModuleRtpRtcpImpl2::SetRemoteSSRC(const uint32_t ssrc) {
+  // Inform about the incoming SSRC.
+  rtcp_sender_.SetRemoteSSRC(ssrc);
+  rtcp_receiver_.SetRemoteSSRC(ssrc);
+}
+
+void ModuleRtpRtcpImpl2::SetLocalSsrc(uint32_t local_ssrc) {
+  RTC_DCHECK_RUN_ON(&rtcp_thread_checker_);
+  rtcp_receiver_.set_local_media_ssrc(local_ssrc);
+  rtcp_sender_.SetSsrc(local_ssrc);
+}
+
+RtpSendRates ModuleRtpRtcpImpl2::GetSendRates() const {
+  // Typically called on the `rtp_transport_queue_` owned by an
+  // RtpTransportControllerSendInterface instance.
+  return rtp_sender_->packet_sender.GetSendRates();
+}
+
+void ModuleRtpRtcpImpl2::OnRequestSendReport() {
+  SendRTCP(kRtcpSr);
+}
+
+void ModuleRtpRtcpImpl2::OnReceivedNack(
+    const std::vector<uint16_t>& nack_sequence_numbers) {
+  if (!rtp_sender_)
+    return;
+
+  if (!StorePackets() || nack_sequence_numbers.empty()) {
+    return;
+  }
+  // Use RTT from RtcpRttStats class if provided.
+  int64_t rtt = rtt_ms();
+  if (rtt == 0) {
+    rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
+  }
+  rtp_sender_->packet_generator.OnReceivedNack(nack_sequence_numbers, rtt);
+}
+
+void ModuleRtpRtcpImpl2::OnReceivedRtcpReportBlocks(
+    const ReportBlockList& report_blocks) {
+  if (rtp_sender_) {
+    uint32_t ssrc = SSRC();
+    absl::optional<uint32_t> rtx_ssrc;
+    if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) {
+      rtx_ssrc = rtp_sender_->packet_generator.RtxSsrc();
+    }
+
+    for (const RTCPReportBlock& report_block : report_blocks) {
+      if (ssrc == report_block.source_ssrc) {
+        rtp_sender_->packet_generator.OnReceivedAckOnSsrc(
+            report_block.extended_highest_sequence_number);
+      } else if (rtx_ssrc && *rtx_ssrc == report_block.source_ssrc) {
+        rtp_sender_->packet_generator.OnReceivedAckOnRtxSsrc(
+            report_block.extended_highest_sequence_number);
+      }
+    }
+  }
+}
+
+void ModuleRtpRtcpImpl2::set_rtt_ms(int64_t rtt_ms) {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  {
+    MutexLock lock(&mutex_rtt_);
+    rtt_ms_ = rtt_ms;
+  }
+  if (rtp_sender_) {
+    rtp_sender_->packet_history.SetRtt(TimeDelta::Millis(rtt_ms));
+  }
+}
+
+int64_t ModuleRtpRtcpImpl2::rtt_ms() const {
+  MutexLock lock(&mutex_rtt_);
+  return rtt_ms_;
+}
+
+void ModuleRtpRtcpImpl2::SetVideoBitrateAllocation(
+    const VideoBitrateAllocation& bitrate) {
+  rtcp_sender_.SetVideoBitrateAllocation(bitrate);
+}
+
+RTPSender* ModuleRtpRtcpImpl2::RtpSender() {
+  return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
+}
+
+const RTPSender* ModuleRtpRtcpImpl2::RtpSender() const {
+  return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
+}
+
+void ModuleRtpRtcpImpl2::PeriodicUpdate() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+
+  Timestamp check_since = clock_->CurrentTime() - kRttUpdateInterval;
+  absl::optional<TimeDelta> rtt =
+      rtcp_receiver_.OnPeriodicRttUpdate(check_since, rtcp_sender_.Sending());
+  if (rtt) {
+    if (rtt_stats_) {
+      rtt_stats_->OnRttUpdate(rtt->ms());
+    }
+    set_rtt_ms(rtt->ms());
+  }
+}
+
+void ModuleRtpRtcpImpl2::MaybeSendRtcp() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  if (rtcp_sender_.TimeToSendRTCPReport())
+    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+}
+
+// TODO(bugs.webrtc.org/12889): Consider removing this function when the issue
+// is resolved.
+void ModuleRtpRtcpImpl2::MaybeSendRtcpAtOrAfterTimestamp(
+    Timestamp execution_time) {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  Timestamp now = clock_->CurrentTime();
+  if (now >= execution_time) {
+    MaybeSendRtcp();
+    return;
+  }
+
+  TimeDelta delta = execution_time - now;
+  // TaskQueue may run task 1ms earlier, so don't print warning if in this case.
+  if (delta > TimeDelta::Millis(1)) {
+    RTC_DLOG(LS_WARNING) << "BUGBUG: Task queue scheduled delayed call "
+                         << delta << " too early.";
+  }
+
+  ScheduleMaybeSendRtcpAtOrAfterTimestamp(execution_time, delta);
+}
+
+void ModuleRtpRtcpImpl2::ScheduleRtcpSendEvaluation(TimeDelta duration) {
+  // We end up here under various sequences including the worker queue, and
+  // the RTCPSender lock is held.
+  // We're assuming that the fact that RTCPSender executes under other sequences
+  // than the worker queue on which it's created on implies that external
+  // synchronization is present and removes this activity before destruction.
+  if (duration.IsZero()) {
+    worker_queue_->PostTask(SafeTask(task_safety_.flag(), [this] {
+      RTC_DCHECK_RUN_ON(worker_queue_);
+      MaybeSendRtcp();
+    }));
+  } else {
+    Timestamp execution_time = clock_->CurrentTime() + duration;
+    ScheduleMaybeSendRtcpAtOrAfterTimestamp(execution_time, duration);
+  }
+}
+
+void ModuleRtpRtcpImpl2::ScheduleMaybeSendRtcpAtOrAfterTimestamp(
+    Timestamp execution_time,
+    TimeDelta duration) {
+  // We end up here under various sequences including the worker queue, and
+  // the RTCPSender lock is held.
+  // See note in ScheduleRtcpSendEvaluation about why `worker_queue_` can be
+  // accessed.
+  worker_queue_->PostDelayedTask(
+      SafeTask(task_safety_.flag(),
+               [this, execution_time] {
+                 RTC_DCHECK_RUN_ON(worker_queue_);
+                 MaybeSendRtcpAtOrAfterTimestamp(execution_time);
+               }),
+      duration.RoundUpTo(TimeDelta::Millis(1)));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
new file mode 100644
index 0000000000..c43d0c34ba
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -0,0 +1,341 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL2_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL2_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/rtp_headers.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/include/module_fec_types.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // RTCPPacketType
+#include "modules/rtp_rtcp/source/packet_sequencer.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+#include "modules/rtp_rtcp/source/rtcp_receiver.h"
+#include "modules/rtp_rtcp/source/rtcp_sender.h"
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_sender.h"
+#include "modules/rtp_rtcp/source/rtp_sender_egress.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class Clock;
+struct PacedPacketInfo;
+struct RTPVideoHeader;
+
+class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
+                                 public RTCPReceiver::ModuleRtpRtcp {
+ public:
+  explicit ModuleRtpRtcpImpl2(
+      const RtpRtcpInterface::Configuration& configuration);
+  ~ModuleRtpRtcpImpl2() override;
+
+  // This method is provided to easy with migrating away from the
+  // RtpRtcp::Create factory method. Since this is an internal implementation
+  // detail though, creating an instance of ModuleRtpRtcpImpl2 directly should
+  // be fine.
+  static std::unique_ptr<ModuleRtpRtcpImpl2> Create(
+      const Configuration& configuration);
+
+  // Receiver part.
+
+  // Called when we receive an RTCP packet.
+  void IncomingRtcpPacket(const uint8_t* incoming_packet,
+                          size_t incoming_packet_length) override;
+
+  void SetRemoteSSRC(uint32_t ssrc) override;
+
+  void SetLocalSsrc(uint32_t local_ssrc) override;
+
+  // Sender part.
+  void RegisterSendPayloadFrequency(int payload_type,
+                                    int payload_frequency) override;
+
+  int32_t DeRegisterSendPayload(int8_t payload_type) override;
+
+  void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
+
+  void RegisterRtpHeaderExtension(absl::string_view uri, int id) override;
+  void DeregisterSendRtpHeaderExtension(absl::string_view uri) override;
+
+  bool SupportsPadding() const override;
+  bool SupportsRtxPayloadPadding() const override;
+
+  // Get start timestamp.
+  uint32_t StartTimestamp() const override;
+
+  // Configure start timestamp, default is a random number.
+  void SetStartTimestamp(uint32_t timestamp) override;
+
+  uint16_t SequenceNumber() const override;
+
+  // Set SequenceNumber, default is a random number.
+  void SetSequenceNumber(uint16_t seq) override;
+
+  void SetRtpState(const RtpState& rtp_state) override;
+  void SetRtxState(const RtpState& rtp_state) override;
+  RtpState GetRtpState() const override;
+  RtpState GetRtxState() const override;
+
+  void SetNonSenderRttMeasurement(bool enabled) override;
+
+  uint32_t SSRC() const override { return rtcp_sender_.SSRC(); }
+
+  // Semantically identical to `SSRC()` but must be called on the packet
+  // delivery thread/tq and returns the ssrc that maps to
+  // RtpRtcpInterface::Configuration::local_media_ssrc.
+  uint32_t local_media_ssrc() const;
+
+  void SetMid(absl::string_view mid) override;
+
+  RTCPSender::FeedbackState GetFeedbackState();
+
+  void SetRtxSendStatus(int mode) override;
+  int RtxSendStatus() const override;
+  absl::optional<uint32_t> RtxSsrc() const override;
+
+  void SetRtxSendPayloadType(int payload_type,
+                             int associated_payload_type) override;
+
+  absl::optional<uint32_t> FlexfecSsrc() const override;
+
+  // Sends kRtcpByeCode when going from true to false.
+  int32_t SetSendingStatus(bool sending) override;
+
+  bool Sending() const override;
+
+  // Drops or relays media packets.
+  void SetSendingMediaStatus(bool sending) override;
+
+  bool SendingMedia() const override;
+
+  bool IsAudioConfigured() const override;
+
+  void SetAsPartOfAllocation(bool part_of_allocation) override;
+
+  bool OnSendingRtpFrame(uint32_t timestamp,
+                         int64_t capture_time_ms,
+                         int payload_type,
+                         bool force_sender_report) override;
+
+  bool TrySendPacket(RtpPacketToSend* packet,
+                     const PacedPacketInfo& pacing_info) override;
+
+  void SetFecProtectionParams(const FecProtectionParams& delta_params,
+                              const FecProtectionParams& key_params) override;
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() override;
+
+  void OnAbortedRetransmissions(
+      rtc::ArrayView<const uint16_t> sequence_numbers) override;
+
+  void OnPacketsAcknowledged(
+      rtc::ArrayView<const uint16_t> sequence_numbers) override;
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      size_t target_size_bytes) override;
+
+  std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+      rtc::ArrayView<const uint16_t> sequence_numbers) const override;
+
+  size_t ExpectedPerPacketOverhead() const override;
+
+  void OnPacketSendingThreadSwitched() override;
+
+  // RTCP part.
+
+  // Get RTCP status.
+  RtcpMode RTCP() const override;
+
+  // Configure RTCP status i.e on/off.
+  void SetRTCPStatus(RtcpMode method) override;
+
+  // Set RTCP CName.
+  int32_t SetCNAME(absl::string_view c_name) override;
+
+  // Get remote NTP.
+  int32_t RemoteNTP(uint32_t* received_ntp_secs,
+                    uint32_t* received_ntp_frac,
+                    uint32_t* rtcp_arrival_time_secs,
+                    uint32_t* rtcp_arrival_time_frac,
+                    uint32_t* rtcp_timestamp) const override;
+
+  // Get RoundTripTime.
+  int32_t RTT(uint32_t remote_ssrc,
+              int64_t* rtt,
+              int64_t* avg_rtt,
+              int64_t* min_rtt,
+              int64_t* max_rtt) const override;
+
+  int64_t ExpectedRetransmissionTimeMs() const override;
+
+  // Force a send of an RTCP packet.
+  // Normal SR and RR are triggered via the task queue that's current when this
+  // object is created.
+  int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;
+
+  void GetSendStreamDataCounters(
+      StreamDataCounters* rtp_counters,
+      StreamDataCounters* rtx_counters) const override;
+
+  void RemoteRTCPSenderInfo(uint32_t* packet_count,
+                            uint32_t* octet_count,
+                            int64_t* ntp_timestamp_ms,
+                            int64_t* remote_ntp_timestamp_ms) const override;
+
+  // A snapshot of the most recent Report Block with additional data of
+  // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
+  // Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
+  // which is the SSRC of the corresponding outbound RTP stream, is unique.
+  std::vector<ReportBlockData> GetLatestReportBlockData() const override;
+  absl::optional<SenderReportStats> GetSenderReportStats() const override;
+  absl::optional<NonSenderRttStats> GetNonSenderRttStats() const override;
+
+  // (REMB) Receiver Estimated Max Bitrate.
+  void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
+  void UnsetRemb() override;
+
+  void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override;
+
+  size_t MaxRtpPacketSize() const override;
+
+  void SetMaxRtpPacketSize(size_t max_packet_size) override;
+
+  // (NACK) Negative acknowledgment part.
+
+  // Send a Negative acknowledgment packet.
+  // TODO(philipel): Deprecate SendNACK and use SendNack instead.
+  int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override;
+
+  void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
+
+  // Store the sent packets, needed to answer to a negative acknowledgment
+  // requests.
+  void SetStorePacketsStatus(bool enable, uint16_t number_to_store) override;
+
+  void SendCombinedRtcpPacket(
+      std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) override;
+
+  // Video part.
+  int32_t SendLossNotification(uint16_t last_decoded_seq_num,
+                               uint16_t last_received_seq_num,
+                               bool decodability_flag,
+                               bool buffering_allowed) override;
+
+  RtpSendRates GetSendRates() const override;
+
+  void OnReceivedNack(
+      const std::vector<uint16_t>& nack_sequence_numbers) override;
+  void OnReceivedRtcpReportBlocks(
+      const ReportBlockList& report_blocks) override;
+  void OnRequestSendReport() override;
+
+  void SetVideoBitrateAllocation(
+      const VideoBitrateAllocation& bitrate) override;
+
+  RTPSender* RtpSender() override;
+  const RTPSender* RtpSender() const override;
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(RtpRtcpImpl2Test, Rtt);
+  FRIEND_TEST_ALL_PREFIXES(RtpRtcpImpl2Test, RttForReceiverOnly);
+
+  struct RtpSenderContext {
+    explicit RtpSenderContext(const RtpRtcpInterface::Configuration& config);
+    // Storage of packets, for retransmissions and padding, if applicable.
+    RtpPacketHistory packet_history;
+    SequenceChecker sequencing_checker;
+    // Handles sequence number assignment and padding timestamp generation.
+    PacketSequencer sequencer RTC_GUARDED_BY(sequencing_checker);
+    // Handles final time timestamping/stats/etc and handover to Transport.
+    RtpSenderEgress packet_sender;
+    // If no paced sender configured, this class will be used to pass packets
+    // from `packet_generator_` to `packet_sender_`.
+    RtpSenderEgress::NonPacedPacketSender non_paced_sender;
+    // Handles creation of RTP packets to be sent.
+    RTPSender packet_generator;
+  };
+
+  void set_rtt_ms(int64_t rtt_ms);
+  int64_t rtt_ms() const;
+
+  bool TimeToSendFullNackList(int64_t now) const;
+
+  // Called on a timer, once a second, on the worker_queue_, to update the RTT,
+  // check if we need to send RTCP report, send TMMBR updates and fire events.
+  void PeriodicUpdate();
+
+  // Returns true if the module is configured to store packets.
+  bool StorePackets() const;
+
+  // Used from RtcpSenderMediator to maybe send rtcp.
+  void MaybeSendRtcp() RTC_RUN_ON(worker_queue_);
+
+  // Called when `rtcp_sender_` informs of the next RTCP instant. The method may
+  // be called on various sequences, and is called under a RTCPSenderLock.
+  void ScheduleRtcpSendEvaluation(TimeDelta duration);
+
+  // Helper method combating too early delayed calls from task queues.
+  // TODO(bugs.webrtc.org/12889): Consider removing this function when the issue
+  // is resolved.
+  void MaybeSendRtcpAtOrAfterTimestamp(Timestamp execution_time)
+      RTC_RUN_ON(worker_queue_);
+
+  // Schedules a call to MaybeSendRtcpAtOrAfterTimestamp delayed by `duration`.
+  void ScheduleMaybeSendRtcpAtOrAfterTimestamp(Timestamp execution_time,
+                                               TimeDelta duration);
+
+  TaskQueueBase* const worker_queue_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker rtcp_thread_checker_;
+
+  std::unique_ptr<RtpSenderContext> rtp_sender_;
+  RTCPSender rtcp_sender_;
+  RTCPReceiver rtcp_receiver_;
+
+  Clock* const clock_;
+
+  uint16_t packet_overhead_;
+
+  // Send side
+  int64_t nack_last_time_sent_full_ms_;
+  uint16_t nack_last_seq_number_sent_;
+
+  RtcpRttStats* const rtt_stats_;
+  RepeatingTaskHandle rtt_update_task_ RTC_GUARDED_BY(worker_queue_);
+
+  // The processed RTT from RtcpRttStats.
+  mutable Mutex mutex_rtt_;
+  int64_t rtt_ms_ RTC_GUARDED_BY(mutex_rtt_);
+
+  RTC_NO_UNIQUE_ADDRESS ScopedTaskSafety task_safety_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL2_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
new file mode 100644
index 0000000000..918e075be8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
@@ -0,0 +1,1157 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+
+#include <deque>
+#include <map>
+#include <memory>
+#include <set>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_registry.h"
+#include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/rate_limiter.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+#include "test/run_loop.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+using ::testing::AllOf;
+using ::testing::ElementsAre;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::Gt;
+using ::testing::Not;
+using ::testing::Optional;
+using ::testing::SizeIs;
+
+using webrtc::test::ExplicitKeyValueConfig;
+
+namespace webrtc {
+namespace {
+constexpr uint32_t kSenderSsrc = 0x12345;
+constexpr uint32_t kReceiverSsrc = 0x23456;
+constexpr uint32_t kRtxSenderSsrc = 0x12346;
+constexpr TimeDelta kOneWayNetworkDelay = TimeDelta::Millis(100);
+constexpr uint8_t kBaseLayerTid = 0;
+constexpr uint8_t kHigherLayerTid = 1;
+constexpr uint16_t kSequenceNumber = 100;
+constexpr uint8_t kPayloadType = 100;
+constexpr uint8_t kRtxPayloadType = 98;
+constexpr int kWidth = 320;
+constexpr int kHeight = 100;
+constexpr int kCaptureTimeMsToRtpTimestamp = 90;  // 90 kHz clock.
+constexpr TimeDelta kDefaultReportInterval = TimeDelta::Millis(1000);
+
+// RTP header extension ids.
+enum : int {
+  kAbsoluteSendTimeExtensionId = 1,
+  kTransportSequenceNumberExtensionId,
+  kTransmissionOffsetExtensionId,
+};
+
+class RtcpRttStatsTestImpl : public RtcpRttStats {
+ public:
+  RtcpRttStatsTestImpl() : rtt_ms_(0) {}
+  ~RtcpRttStatsTestImpl() override = default;
+
+  void OnRttUpdate(int64_t rtt_ms) override { rtt_ms_ = rtt_ms; }
+  int64_t LastProcessedRtt() const override { return rtt_ms_; }
+  int64_t rtt_ms_;
+};
+
+// TODO(bugs.webrtc.org/11581): remove inheritance once the ModuleRtpRtcpImpl2
+// Module/ProcessThread dependency is gone.
+class SendTransport : public Transport,
+                      public sim_time_impl::SimulatedSequenceRunner {
+ public:
+  SendTransport(TimeDelta delay, GlobalSimulatedTimeController* time_controller)
+      : receiver_(nullptr),
+        time_controller_(time_controller),
+        delay_(delay),
+        rtp_packets_sent_(0),
+        rtcp_packets_sent_(0),
+        last_packet_(&header_extensions_) {
+    time_controller_->Register(this);
+  }
+
+  ~SendTransport() { time_controller_->Unregister(this); }
+
+  void SetRtpRtcpModule(ModuleRtpRtcpImpl2* receiver) { receiver_ = receiver; }
+  void SimulateNetworkDelay(TimeDelta delay) { delay_ = delay; }
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override {
+    EXPECT_TRUE(last_packet_.Parse(data, len));
+    ++rtp_packets_sent_;
+    return true;
+  }
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    test::RtcpPacketParser parser;
+    parser.Parse(data, len);
+    last_nack_list_ = parser.nack()->packet_ids();
+    Timestamp current_time = time_controller_->GetClock()->CurrentTime();
+    Timestamp delivery_time = current_time + delay_;
+    rtcp_packets_.push_back(
+        Packet{delivery_time, std::vector<uint8_t>(data, data + len)});
+    ++rtcp_packets_sent_;
+    RunReady(current_time);
+    return true;
+  }
+
+  // sim_time_impl::SimulatedSequenceRunner
+  Timestamp GetNextRunTime() const override {
+    if (!rtcp_packets_.empty())
+      return rtcp_packets_.front().send_time;
+    return Timestamp::PlusInfinity();
+  }
+  void RunReady(Timestamp at_time) override {
+    while (!rtcp_packets_.empty() &&
+           rtcp_packets_.front().send_time <= at_time) {
+      Packet packet = std::move(rtcp_packets_.front());
+      rtcp_packets_.pop_front();
+      EXPECT_TRUE(receiver_);
+      receiver_->IncomingRtcpPacket(packet.data.data(), packet.data.size());
+    }
+  }
+  TaskQueueBase* GetAsTaskQueue() override {
+    return reinterpret_cast<TaskQueueBase*>(this);
+  }
+
+  size_t NumRtcpSent() { return rtcp_packets_sent_; }
+
+  ModuleRtpRtcpImpl2* receiver_;
+  GlobalSimulatedTimeController* const time_controller_;
+  TimeDelta delay_;
+  int rtp_packets_sent_;
+  size_t rtcp_packets_sent_;
+  std::vector<uint16_t> last_nack_list_;
+  RtpHeaderExtensionMap header_extensions_;
+  RtpPacketReceived last_packet_;
+  struct Packet {
+    Timestamp send_time;
+    std::vector<uint8_t> data;
+  };
+  std::deque<Packet> rtcp_packets_;
+};
+
+class RtpRtcpModule : public RtcpPacketTypeCounterObserver,
+                      public SendPacketObserver {
+ public:
+  struct SentPacket {
+    SentPacket(uint16_t packet_id, int64_t capture_time_ms, uint32_t ssrc)
+        : packet_id(packet_id), capture_time_ms(capture_time_ms), ssrc(ssrc) {}
+    uint16_t packet_id;
+    int64_t capture_time_ms;
+    uint32_t ssrc;
+  };
+
+  RtpRtcpModule(GlobalSimulatedTimeController* time_controller,
+                bool is_sender,
+                const FieldTrialsRegistry& trials)
+      : time_controller_(time_controller),
+        is_sender_(is_sender),
+        trials_(trials),
+        receive_statistics_(
+            ReceiveStatistics::Create(time_controller->GetClock())),
+        transport_(kOneWayNetworkDelay, time_controller) {
+    CreateModuleImpl();
+  }
+
+  TimeController* const time_controller_;
+  const bool is_sender_;
+  const FieldTrialsRegistry& trials_;
+  RtcpPacketTypeCounter packets_sent_;
+  RtcpPacketTypeCounter packets_received_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  SendTransport transport_;
+  RtcpRttStatsTestImpl rtt_stats_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> impl_;
+
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) override {
+    counter_map_[ssrc] = packet_counter;
+  }
+
+  void OnSendPacket(uint16_t packet_id,
+                    int64_t capture_time_ms,
+                    uint32_t ssrc) override {
+    last_sent_packet_.emplace(packet_id, capture_time_ms, ssrc);
+  }
+
+  absl::optional<SentPacket> last_sent_packet() const {
+    return last_sent_packet_;
+  }
+
+  RtcpPacketTypeCounter RtcpSent() {
+    // RTCP counters for remote SSRC.
+    return counter_map_[is_sender_ ? kReceiverSsrc : kSenderSsrc];
+  }
+
+  RtcpPacketTypeCounter RtcpReceived() {
+    // Received RTCP stats for (own) local SSRC.
+    return counter_map_[impl_->SSRC()];
+  }
+  int RtpSent() { return transport_.rtp_packets_sent_; }
+  uint16_t LastRtpSequenceNumber() { return last_packet().SequenceNumber(); }
+  std::vector<uint16_t> LastNackListSent() {
+    return transport_.last_nack_list_;
+  }
+  void SetRtcpReportIntervalAndReset(TimeDelta rtcp_report_interval) {
+    rtcp_report_interval_ = rtcp_report_interval;
+    CreateModuleImpl();
+  }
+  const RtpPacketReceived& last_packet() { return transport_.last_packet_; }
+  void RegisterHeaderExtension(absl::string_view uri, int id) {
+    impl_->RegisterRtpHeaderExtension(uri, id);
+    transport_.header_extensions_.RegisterByUri(id, uri);
+    transport_.last_packet_.IdentifyExtensions(transport_.header_extensions_);
+  }
+  void ReinintWithFec(VideoFecGenerator* fec_generator) {
+    fec_generator_ = fec_generator;
+    CreateModuleImpl();
+  }
+
+  void CreateModuleImpl() {
+    RtpRtcpInterface::Configuration config;
+    config.audio = false;
+    config.clock = time_controller_->GetClock();
+    config.outgoing_transport = &transport_;
+    config.receive_statistics = receive_statistics_.get();
+    config.rtcp_packet_type_counter_observer = this;
+    config.rtt_stats = &rtt_stats_;
+    config.rtcp_report_interval_ms = rtcp_report_interval_.ms();
+    config.local_media_ssrc = is_sender_ ? kSenderSsrc : kReceiverSsrc;
+    config.rtx_send_ssrc =
+        is_sender_ ? absl::make_optional(kRtxSenderSsrc) : absl::nullopt;
+    config.need_rtp_packet_infos = true;
+    config.non_sender_rtt_measurement = true;
+    config.field_trials = &trials_;
+    config.send_packet_observer = this;
+    config.fec_generator = fec_generator_;
+    impl_.reset(new ModuleRtpRtcpImpl2(config));
+    impl_->SetRemoteSSRC(is_sender_ ? kReceiverSsrc : kSenderSsrc);
+    impl_->SetRTCPStatus(RtcpMode::kCompound);
+  }
+
+ private:
+  std::map<uint32_t, RtcpPacketTypeCounter> counter_map_;
+  absl::optional<SentPacket> last_sent_packet_;
+  VideoFecGenerator* fec_generator_ = nullptr;
+  TimeDelta rtcp_report_interval_ = kDefaultReportInterval;
+};
+}  // namespace
+
+class RtpRtcpImpl2Test : public ::testing::Test {
+ protected:
+  RtpRtcpImpl2Test()
+      : time_controller_(Timestamp::Micros(133590000000000)),
+        field_trials_(""),
+        sender_(&time_controller_,
+                /*is_sender=*/true,
+                field_trials_),
+        receiver_(&time_controller_,
+                  /*is_sender=*/false,
+                  field_trials_) {}
+
+  void SetUp() override {
+    // Send module.
+    EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true));
+    sender_.impl_->SetSendingMediaStatus(true);
+    sender_.impl_->SetSequenceNumber(kSequenceNumber);
+    sender_.impl_->SetStorePacketsStatus(true, 100);
+
+    RTPSenderVideo::Config video_config;
+    video_config.clock = time_controller_.GetClock();
+    video_config.rtp_sender = sender_.impl_->RtpSender();
+    video_config.field_trials = &field_trials_;
+    sender_video_ = std::make_unique<RTPSenderVideo>(video_config);
+
+    // Receive module.
+    EXPECT_EQ(0, receiver_.impl_->SetSendingStatus(false));
+    receiver_.impl_->SetSendingMediaStatus(false);
+    // Transport settings.
+    sender_.transport_.SetRtpRtcpModule(receiver_.impl_.get());
+    receiver_.transport_.SetRtpRtcpModule(sender_.impl_.get());
+  }
+
+  void AdvanceTime(TimeDelta duration) {
+    time_controller_.AdvanceTime(duration);
+  }
+
+  void ReinitWithFec(VideoFecGenerator* fec_generator,
+                     absl::optional<int> red_payload_type) {
+    sender_.ReinintWithFec(fec_generator);
+    EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true));
+    sender_.impl_->SetSendingMediaStatus(true);
+    sender_.impl_->SetSequenceNumber(kSequenceNumber);
+    sender_.impl_->SetStorePacketsStatus(true, 100);
+    receiver_.transport_.SetRtpRtcpModule(sender_.impl_.get());
+
+    RTPSenderVideo::Config video_config;
+    video_config.clock = time_controller_.GetClock();
+    video_config.rtp_sender = sender_.impl_->RtpSender();
+    video_config.field_trials = &field_trials_;
+    video_config.fec_overhead_bytes = fec_generator->MaxPacketOverhead();
+    video_config.fec_type = fec_generator->GetFecType();
+    video_config.red_payload_type = red_payload_type;
+    sender_video_ = std::make_unique<RTPSenderVideo>(video_config);
+  }
+
+  GlobalSimulatedTimeController time_controller_;
+  test::ExplicitKeyValueConfig field_trials_;
+  RtpRtcpModule sender_;
+  std::unique_ptr<RTPSenderVideo> sender_video_;
+  RtpRtcpModule receiver_;
+
+  bool SendFrame(const RtpRtcpModule* module,
+                 RTPSenderVideo* sender,
+                 uint8_t tid) {
+    int64_t now_ms = time_controller_.GetClock()->TimeInMilliseconds();
+    return SendFrame(
+        module, sender, tid,
+        static_cast<uint32_t>(now_ms * kCaptureTimeMsToRtpTimestamp), now_ms);
+  }
+
+  bool SendFrame(const RtpRtcpModule* module,
+                 RTPSenderVideo* sender,
+                 uint8_t tid,
+                 uint32_t rtp_timestamp,
+                 int64_t capture_time_ms) {
+    RTPVideoHeaderVP8 vp8_header = {};
+    vp8_header.temporalIdx = tid;
+    RTPVideoHeader rtp_video_header;
+    rtp_video_header.frame_type = VideoFrameType::kVideoFrameKey;
+    rtp_video_header.width = kWidth;
+    rtp_video_header.height = kHeight;
+    rtp_video_header.rotation = kVideoRotation_0;
+    rtp_video_header.content_type = VideoContentType::UNSPECIFIED;
+    rtp_video_header.playout_delay = {-1, -1};
+    rtp_video_header.is_first_packet_in_frame = true;
+    rtp_video_header.simulcastIdx = 0;
+    rtp_video_header.codec = kVideoCodecVP8;
+    rtp_video_header.video_type_header = vp8_header;
+    rtp_video_header.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false};
+
+    const uint8_t payload[100] = {0};
+    bool success = module->impl_->OnSendingRtpFrame(0, 0, kPayloadType, true);
+
+    success &= sender->SendVideo(kPayloadType, VideoCodecType::kVideoCodecVP8,
+                                 rtp_timestamp, capture_time_ms, payload,
+                                 rtp_video_header, 0);
+    return success;
+  }
+
+  void IncomingRtcpNack(const RtpRtcpModule* module, uint16_t sequence_number) {
+    bool sender = module->impl_->SSRC() == kSenderSsrc;
+    rtcp::Nack nack;
+    uint16_t list[1];
+    list[0] = sequence_number;
+    const uint16_t kListLength = sizeof(list) / sizeof(list[0]);
+    nack.SetSenderSsrc(sender ? kReceiverSsrc : kSenderSsrc);
+    nack.SetMediaSsrc(sender ? kSenderSsrc : kReceiverSsrc);
+    nack.SetPacketIds(list, kListLength);
+    rtc::Buffer packet = nack.Build();
+    module->impl_->IncomingRtcpPacket(packet.data(), packet.size());
+  }
+};
+
+TEST_F(RtpRtcpImpl2Test, RetransmitsAllLayers) {
+  // Send frames.
+  EXPECT_EQ(0, sender_.RtpSent());
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(),
+                        kBaseLayerTid));  // kSequenceNumber
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(),
+                        kHigherLayerTid));  // kSequenceNumber + 1
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(),
+                        kNoTemporalIdx));  // kSequenceNumber + 2
+  EXPECT_EQ(3, sender_.RtpSent());
+  EXPECT_EQ(kSequenceNumber + 2, sender_.LastRtpSequenceNumber());
+
+  // Min required delay until retransmit = 5 + RTT ms (RTT = 0).
+  AdvanceTime(TimeDelta::Millis(5));
+
+  // Frame with kBaseLayerTid re-sent.
+  IncomingRtcpNack(&sender_, kSequenceNumber);
+  EXPECT_EQ(4, sender_.RtpSent());
+  EXPECT_EQ(kSequenceNumber, sender_.LastRtpSequenceNumber());
+  // Frame with kHigherLayerTid re-sent.
+  IncomingRtcpNack(&sender_, kSequenceNumber + 1);
+  EXPECT_EQ(5, sender_.RtpSent());
+  EXPECT_EQ(kSequenceNumber + 1, sender_.LastRtpSequenceNumber());
+  // Frame with kNoTemporalIdx re-sent.
+  IncomingRtcpNack(&sender_, kSequenceNumber + 2);
+  EXPECT_EQ(6, sender_.RtpSent());
+  EXPECT_EQ(kSequenceNumber + 2, sender_.LastRtpSequenceNumber());
+}
+
+TEST_F(RtpRtcpImpl2Test, Rtt) {
+  RtpPacketReceived packet;
+  packet.SetTimestamp(1);
+  packet.SetSequenceNumber(123);
+  packet.SetSsrc(kSenderSsrc);
+  packet.AllocatePayload(100 - 12);
+  receiver_.receive_statistics_->OnRtpPacket(packet);
+
+  // Send Frame before sending an SR.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  // Sender module should send an SR.
+  EXPECT_EQ(0, sender_.impl_->SendRTCP(kRtcpReport));
+  AdvanceTime(kOneWayNetworkDelay);
+
+  // Receiver module should send a RR with a response to the last received SR.
+  EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport));
+  AdvanceTime(kOneWayNetworkDelay);
+
+  // Verify RTT.
+  int64_t rtt;
+  int64_t avg_rtt;
+  int64_t min_rtt;
+  int64_t max_rtt;
+  EXPECT_EQ(
+      0, sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
+  EXPECT_NEAR(2 * kOneWayNetworkDelay.ms(), rtt, 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelay.ms(), avg_rtt, 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelay.ms(), min_rtt, 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelay.ms(), max_rtt, 1);
+
+  // No RTT from other ssrc.
+  EXPECT_EQ(-1, sender_.impl_->RTT(kReceiverSsrc + 1, &rtt, &avg_rtt, &min_rtt,
+                                   &max_rtt));
+
+  // Verify RTT from rtt_stats config.
+  EXPECT_EQ(0, sender_.rtt_stats_.LastProcessedRtt());
+  EXPECT_EQ(0, sender_.impl_->rtt_ms());
+  AdvanceTime(TimeDelta::Millis(1000));
+
+  EXPECT_NEAR(2 * kOneWayNetworkDelay.ms(),
+              sender_.rtt_stats_.LastProcessedRtt(), 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelay.ms(), sender_.impl_->rtt_ms(), 1);
+}
+
+TEST_F(RtpRtcpImpl2Test, RttForReceiverOnly) {
+  // Receiver module should send a Receiver time reference report (RTRR).
+  EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport));
+
+  // Sender module should send a response to the last received RTRR (DLRR).
+  AdvanceTime(TimeDelta::Millis(1000));
+  // Send Frame before sending a SR.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  EXPECT_EQ(0, sender_.impl_->SendRTCP(kRtcpReport));
+
+  // Verify RTT.
+  EXPECT_EQ(0, receiver_.rtt_stats_.LastProcessedRtt());
+  EXPECT_EQ(0, receiver_.impl_->rtt_ms());
+  AdvanceTime(TimeDelta::Millis(1000));
+  EXPECT_NEAR(2 * kOneWayNetworkDelay.ms(),
+              receiver_.rtt_stats_.LastProcessedRtt(), 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelay.ms(), receiver_.impl_->rtt_ms(), 1);
+}
+
+TEST_F(RtpRtcpImpl2Test, NoSrBeforeMedia) {
+  // Ignore fake transport delays in this test.
+  sender_.transport_.SimulateNetworkDelay(TimeDelta::Zero());
+  receiver_.transport_.SimulateNetworkDelay(TimeDelta::Zero());
+
+  // Move ahead to the instant a rtcp is expected.
+  // Verify no SR is sent before media has been sent, RR should still be sent
+  // from the receiving module though.
+  AdvanceTime(kDefaultReportInterval / 2);
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 0u);
+  EXPECT_EQ(receiver_.transport_.NumRtcpSent(), 1u);
+
+  // RTCP should be triggered by the RTP send.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u);
+}
+
+TEST_F(RtpRtcpImpl2Test, RtcpPacketTypeCounter_Nack) {
+  EXPECT_EQ(0U, sender_.RtcpReceived().nack_packets);
+  EXPECT_EQ(0U, receiver_.RtcpSent().nack_packets);
+
+  // Receive module sends a NACK.
+  const uint16_t kNackLength = 1;
+  uint16_t nack_list[kNackLength] = {123};
+  EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list, kNackLength));
+  AdvanceTime(kOneWayNetworkDelay);
+  EXPECT_EQ(1U, receiver_.RtcpSent().nack_packets);
+
+  // Send module receives the NACK.
+  EXPECT_EQ(1U, sender_.RtcpReceived().nack_packets);
+}
+
+TEST_F(RtpRtcpImpl2Test, AddStreamDataCounters) {
+  StreamDataCounters rtp;
+  const int64_t kStartTimeMs = 1;
+  rtp.first_packet_time_ms = kStartTimeMs;
+  rtp.transmitted.packets = 1;
+  rtp.transmitted.payload_bytes = 1;
+  rtp.transmitted.header_bytes = 2;
+  rtp.transmitted.padding_bytes = 3;
+  EXPECT_EQ(rtp.transmitted.TotalBytes(), rtp.transmitted.payload_bytes +
+                                              rtp.transmitted.header_bytes +
+                                              rtp.transmitted.padding_bytes);
+
+  StreamDataCounters rtp2;
+  rtp2.first_packet_time_ms = -1;
+  rtp2.transmitted.packets = 10;
+  rtp2.transmitted.payload_bytes = 10;
+  rtp2.retransmitted.header_bytes = 4;
+  rtp2.retransmitted.payload_bytes = 5;
+  rtp2.retransmitted.padding_bytes = 6;
+  rtp2.retransmitted.packets = 7;
+  rtp2.fec.packets = 8;
+
+  StreamDataCounters sum = rtp;
+  sum.Add(rtp2);
+  EXPECT_EQ(kStartTimeMs, sum.first_packet_time_ms);
+  EXPECT_EQ(11U, sum.transmitted.packets);
+  EXPECT_EQ(11U, sum.transmitted.payload_bytes);
+  EXPECT_EQ(2U, sum.transmitted.header_bytes);
+  EXPECT_EQ(3U, sum.transmitted.padding_bytes);
+  EXPECT_EQ(4U, sum.retransmitted.header_bytes);
+  EXPECT_EQ(5U, sum.retransmitted.payload_bytes);
+  EXPECT_EQ(6U, sum.retransmitted.padding_bytes);
+  EXPECT_EQ(7U, sum.retransmitted.packets);
+  EXPECT_EQ(8U, sum.fec.packets);
+  EXPECT_EQ(sum.transmitted.TotalBytes(),
+            rtp.transmitted.TotalBytes() + rtp2.transmitted.TotalBytes());
+
+  StreamDataCounters rtp3;
+  rtp3.first_packet_time_ms = kStartTimeMs + 10;
+  sum.Add(rtp3);
+  EXPECT_EQ(kStartTimeMs, sum.first_packet_time_ms);  // Holds oldest time.
+}
+
+TEST_F(RtpRtcpImpl2Test, SendsInitialNackList) {
+  // Send module sends a NACK.
+  const uint16_t kNackLength = 1;
+  uint16_t nack_list[kNackLength] = {123};
+  EXPECT_EQ(0U, sender_.RtcpSent().nack_packets);
+  // Send Frame before sending a compound RTCP that starts with SR.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123));
+}
+
+TEST_F(RtpRtcpImpl2Test, SendsExtendedNackList) {
+  // Send module sends a NACK.
+  const uint16_t kNackLength = 1;
+  uint16_t nack_list[kNackLength] = {123};
+  EXPECT_EQ(0U, sender_.RtcpSent().nack_packets);
+  // Send Frame before sending a compound RTCP that starts with SR.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123));
+
+  // Same list not re-send.
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123));
+
+  // Only extended list sent.
+  const uint16_t kNackExtLength = 2;
+  uint16_t nack_list_ext[kNackExtLength] = {123, 124};
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list_ext, kNackExtLength));
+  EXPECT_EQ(2U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(124));
+}
+
+TEST_F(RtpRtcpImpl2Test, ReSendsNackListAfterRttMs) {
+  sender_.transport_.SimulateNetworkDelay(TimeDelta::Zero());
+  // Send module sends a NACK.
+  const uint16_t kNackLength = 2;
+  uint16_t nack_list[kNackLength] = {123, 125};
+  EXPECT_EQ(0U, sender_.RtcpSent().nack_packets);
+  // Send Frame before sending a compound RTCP that starts with SR.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123, 125));
+
+  // Same list not re-send, rtt interval has not passed.
+  const TimeDelta kStartupRtt = TimeDelta::Millis(100);
+  AdvanceTime(kStartupRtt);
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+
+  // Rtt interval passed, full list sent.
+  AdvanceTime(TimeDelta::Millis(1));
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(2U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123, 125));
+}
+
+TEST_F(RtpRtcpImpl2Test, UniqueNackRequests) {
+  receiver_.transport_.SimulateNetworkDelay(TimeDelta::Zero());
+  EXPECT_EQ(0U, receiver_.RtcpSent().nack_packets);
+  EXPECT_EQ(0U, receiver_.RtcpSent().nack_requests);
+  EXPECT_EQ(0U, receiver_.RtcpSent().unique_nack_requests);
+  EXPECT_EQ(0, receiver_.RtcpSent().UniqueNackRequestsInPercent());
+
+  // Receive module sends NACK request.
+  const uint16_t kNackLength = 4;
+  uint16_t nack_list[kNackLength] = {10, 11, 13, 18};
+  EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, receiver_.RtcpSent().nack_packets);
+  EXPECT_EQ(4U, receiver_.RtcpSent().nack_requests);
+  EXPECT_EQ(4U, receiver_.RtcpSent().unique_nack_requests);
+  EXPECT_THAT(receiver_.LastNackListSent(), ElementsAre(10, 11, 13, 18));
+
+  // Send module receives the request.
+  EXPECT_EQ(1U, sender_.RtcpReceived().nack_packets);
+  EXPECT_EQ(4U, sender_.RtcpReceived().nack_requests);
+  EXPECT_EQ(4U, sender_.RtcpReceived().unique_nack_requests);
+  EXPECT_EQ(100, sender_.RtcpReceived().UniqueNackRequestsInPercent());
+
+  // Receive module sends new request with duplicated packets.
+  const TimeDelta kStartupRtt = TimeDelta::Millis(100);
+  AdvanceTime(kStartupRtt + TimeDelta::Millis(1));
+  const uint16_t kNackLength2 = 4;
+  uint16_t nack_list2[kNackLength2] = {11, 18, 20, 21};
+  EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list2, kNackLength2));
+  EXPECT_EQ(2U, receiver_.RtcpSent().nack_packets);
+  EXPECT_EQ(8U, receiver_.RtcpSent().nack_requests);
+  EXPECT_EQ(6U, receiver_.RtcpSent().unique_nack_requests);
+  EXPECT_THAT(receiver_.LastNackListSent(), ElementsAre(11, 18, 20, 21));
+
+  // Send module receives the request.
+  EXPECT_EQ(2U, sender_.RtcpReceived().nack_packets);
+  EXPECT_EQ(8U, sender_.RtcpReceived().nack_requests);
+  EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests);
+  EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent());
+}
+
+TEST_F(RtpRtcpImpl2Test, ConfigurableRtcpReportInterval) {
+  const TimeDelta kVideoReportInterval = TimeDelta::Millis(3000);
+
+  // Recreate sender impl with new configuration, and redo setup.
+  sender_.SetRtcpReportIntervalAndReset(kVideoReportInterval);
+  SetUp();
+
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+
+  // Initial state
+  EXPECT_EQ(0u, sender_.transport_.NumRtcpSent());
+
+  // Move ahead to the last ms before a rtcp is expected, no action.
+  AdvanceTime(kVideoReportInterval / 2 - TimeDelta::Millis(1));
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 0u);
+
+  // Move ahead to the first rtcp. Send RTCP.
+  AdvanceTime(TimeDelta::Millis(1));
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u);
+
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+
+  // Move ahead to the last possible second before second rtcp is expected.
+  AdvanceTime(kVideoReportInterval * 1 / 2 - TimeDelta::Millis(1));
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u);
+
+  // Move ahead into the range of second rtcp, the second rtcp may be sent.
+  AdvanceTime(TimeDelta::Millis(1));
+  EXPECT_GE(sender_.transport_.NumRtcpSent(), 1u);
+
+  AdvanceTime(kVideoReportInterval / 2);
+  EXPECT_GE(sender_.transport_.NumRtcpSent(), 1u);
+
+  // Move out the range of second rtcp, the second rtcp must have been sent.
+  AdvanceTime(kVideoReportInterval / 2);
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 2u);
+}
+
+TEST_F(RtpRtcpImpl2Test, RtpSenderEgressTimestampOffset) {
+  // RTP timestamp offset not explicitly set, default to random value.
+  uint16_t seqno = sender_.impl_->GetRtpState().sequence_number;
+  uint32_t media_rtp_ts = 1001;
+  uint32_t rtp_ts = media_rtp_ts + sender_.impl_->StartTimestamp();
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, rtp_ts,
+                        /*capture_time_ms=*/0));
+  AdvanceTime(kOneWayNetworkDelay);
+  EXPECT_THAT(
+      sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{seqno}),
+      ElementsAre(Field(&RtpSequenceNumberMap::Info::timestamp, media_rtp_ts)));
+
+  RtpState saved_rtp_state = sender_.impl_->GetRtpState();
+
+  // Change RTP timestamp offset.
+  sender_.impl_->SetStartTimestamp(2000);
+
+  // Restores RtpState and make sure the old timestamp offset is in place.
+  sender_.impl_->SetRtpState(saved_rtp_state);
+  seqno = sender_.impl_->GetRtpState().sequence_number;
+  media_rtp_ts = 1031;
+  rtp_ts = media_rtp_ts + sender_.impl_->StartTimestamp();
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, rtp_ts,
+                        /*capture_time_ms=*/0));
+  AdvanceTime(kOneWayNetworkDelay);
+  EXPECT_THAT(
+      sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{seqno}),
+      ElementsAre(Field(&RtpSequenceNumberMap::Info::timestamp, media_rtp_ts)));
+}
+
+TEST_F(RtpRtcpImpl2Test, StoresPacketInfoForSentPackets) {
+  const uint32_t kStartTimestamp = 1u;
+  SetUp();
+  sender_.impl_->SetStartTimestamp(kStartTimestamp);
+
+  sender_.impl_->SetSequenceNumber(1);
+
+  PacedPacketInfo pacing_info;
+  RtpPacketToSend packet(nullptr);
+  packet.set_packet_type(RtpPacketToSend::Type::kVideo);
+  packet.SetSsrc(kSenderSsrc);
+
+  // Single-packet frame.
+  packet.SetTimestamp(1);
+  packet.set_first_packet_of_frame(true);
+  packet.SetMarker(true);
+  sender_.impl_->TrySendPacket(&packet, pacing_info);
+  AdvanceTime(TimeDelta::Millis(1));
+
+  std::vector<RtpSequenceNumberMap::Info> seqno_info =
+      sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{1});
+
+  EXPECT_THAT(seqno_info, ElementsAre(RtpSequenceNumberMap::Info(
+                              /*timestamp=*/1 - kStartTimestamp,
+                              /*is_first=*/1,
+                              /*is_last=*/1)));
+
+  // Three-packet frame.
+  packet.SetTimestamp(2);
+  packet.set_first_packet_of_frame(true);
+  packet.SetMarker(false);
+  sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+  packet.set_first_packet_of_frame(false);
+  sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+  packet.SetMarker(true);
+  sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+  AdvanceTime(TimeDelta::Millis(1));
+
+  seqno_info =
+      sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{2, 3, 4});
+
+  EXPECT_THAT(seqno_info, ElementsAre(RtpSequenceNumberMap::Info(
+                                          /*timestamp=*/2 - kStartTimestamp,
+                                          /*is_first=*/1,
+                                          /*is_last=*/0),
+                                      RtpSequenceNumberMap::Info(
+                                          /*timestamp=*/2 - kStartTimestamp,
+                                          /*is_first=*/0,
+                                          /*is_last=*/0),
+                                      RtpSequenceNumberMap::Info(
+                                          /*timestamp=*/2 - kStartTimestamp,
+                                          /*is_first=*/0,
+                                          /*is_last=*/1)));
+}
+
+// Checks that the sender report stats are not available if no RTCP SR was sent.
+TEST_F(RtpRtcpImpl2Test, SenderReportStatsNotAvailable) {
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt));
+}
+
+// Checks that the sender report stats are available if an RTCP SR was sent.
+TEST_F(RtpRtcpImpl2Test, SenderReportStatsAvailable) {
+  // Send a frame in order to send an SR.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  // Send an SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  AdvanceTime(kOneWayNetworkDelay);
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Not(Eq(absl::nullopt)));
+}
+
+// Checks that the sender report stats are not available if an RTCP SR with an
+// unexpected SSRC is received.
+TEST_F(RtpRtcpImpl2Test, SenderReportStatsNotUpdatedWithUnexpectedSsrc) {
+  constexpr uint32_t kUnexpectedSenderSsrc = 0x87654321;
+  static_assert(kUnexpectedSenderSsrc != kSenderSsrc, "");
+  // Forge a sender report and pass it to the receiver as if an RTCP SR were
+  // sent by an unexpected sender.
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kUnexpectedSenderSsrc);
+  sr.SetNtp({/*seconds=*/1u, /*fractions=*/1u << 31});
+  sr.SetPacketCount(123u);
+  sr.SetOctetCount(456u);
+  auto raw_packet = sr.Build();
+  receiver_.impl_->IncomingRtcpPacket(raw_packet.data(), raw_packet.size());
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt));
+}
+
+// Checks the stats derived from the last received RTCP SR are set correctly.
+TEST_F(RtpRtcpImpl2Test, SenderReportStatsCheckStatsFromLastReport) {
+  using SenderReportStats = RtpRtcpInterface::SenderReportStats;
+  const NtpTime ntp(/*seconds=*/1u, /*fractions=*/1u << 31);
+  constexpr uint32_t kPacketCount = 123u;
+  constexpr uint32_t kOctetCount = 456u;
+  // Forge a sender report and pass it to the receiver as if an RTCP SR were
+  // sent by the sender.
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  sr.SetNtp(ntp);
+  sr.SetPacketCount(kPacketCount);
+  sr.SetOctetCount(kOctetCount);
+  auto raw_packet = sr.Build();
+  receiver_.impl_->IncomingRtcpPacket(raw_packet.data(), raw_packet.size());
+
+  EXPECT_THAT(
+      receiver_.impl_->GetSenderReportStats(),
+      Optional(AllOf(Field(&SenderReportStats::last_remote_timestamp, Eq(ntp)),
+                     Field(&SenderReportStats::packets_sent, Eq(kPacketCount)),
+                     Field(&SenderReportStats::bytes_sent, Eq(kOctetCount)))));
+}
+
+// Checks that the sender report stats count equals the number of sent RTCP SRs.
+TEST_F(RtpRtcpImpl2Test, SenderReportStatsCount) {
+  using SenderReportStats = RtpRtcpInterface::SenderReportStats;
+  // Send a frame in order to send an SR.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  // Send the first SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  AdvanceTime(kOneWayNetworkDelay);
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(),
+              Optional(Field(&SenderReportStats::reports_count, Eq(1u))));
+  // Send the second SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  AdvanceTime(kOneWayNetworkDelay);
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(),
+              Optional(Field(&SenderReportStats::reports_count, Eq(2u))));
+}
+
+// Checks that the sender report stats include a valid arrival time if an RTCP
+// SR was sent.
+TEST_F(RtpRtcpImpl2Test, SenderReportStatsArrivalTimestampSet) {
+  // Send a frame in order to send an SR.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  // Send an SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  AdvanceTime(kOneWayNetworkDelay);
+  auto stats = receiver_.impl_->GetSenderReportStats();
+  ASSERT_THAT(stats, Not(Eq(absl::nullopt)));
+  EXPECT_TRUE(stats->last_arrival_timestamp.Valid());
+}
+
+// Checks that the packet and byte counters from an RTCP SR are not zero once
+// a frame is sent.
+TEST_F(RtpRtcpImpl2Test, SenderReportStatsPacketByteCounters) {
+  using SenderReportStats = RtpRtcpInterface::SenderReportStats;
+  // Send a frame in order to send an SR.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  ASSERT_THAT(sender_.transport_.rtp_packets_sent_, Gt(0));
+  // Advance time otherwise the RTCP SR report will not include any packets
+  // generated by `SendFrame()`.
+  AdvanceTime(TimeDelta::Millis(1));
+  // Send an SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  AdvanceTime(kOneWayNetworkDelay);
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(),
+              Optional(AllOf(Field(&SenderReportStats::packets_sent, Gt(0u)),
+                             Field(&SenderReportStats::bytes_sent, Gt(0u)))));
+}
+
+TEST_F(RtpRtcpImpl2Test, SendingVideoAdvancesSequenceNumber) {
+  const uint16_t sequence_number = sender_.impl_->SequenceNumber();
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  ASSERT_THAT(sender_.transport_.rtp_packets_sent_, Gt(0));
+  EXPECT_EQ(sequence_number + 1, sender_.impl_->SequenceNumber());
+}
+
+TEST_F(RtpRtcpImpl2Test, SequenceNumberNotAdvancedWhenNotSending) {
+  const uint16_t sequence_number = sender_.impl_->SequenceNumber();
+  sender_.impl_->SetSendingMediaStatus(false);
+  EXPECT_FALSE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  ASSERT_THAT(sender_.transport_.rtp_packets_sent_, Eq(0));
+  EXPECT_EQ(sequence_number, sender_.impl_->SequenceNumber());
+}
+
+TEST_F(RtpRtcpImpl2Test, PaddingNotAllowedInMiddleOfFrame) {
+  constexpr size_t kPaddingSize = 100;
+
+  // Can't send padding before media.
+  EXPECT_THAT(sender_.impl_->GeneratePadding(kPaddingSize), SizeIs(0u));
+
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+
+  // Padding is now ok.
+  EXPECT_THAT(sender_.impl_->GeneratePadding(kPaddingSize), SizeIs(Gt(0u)));
+
+  // Send half a video frame.
+  PacedPacketInfo pacing_info;
+  std::unique_ptr<RtpPacketToSend> packet =
+      sender_.impl_->RtpSender()->AllocatePacket();
+  packet->set_packet_type(RtpPacketToSend::Type::kVideo);
+  packet->set_first_packet_of_frame(true);
+  packet->SetMarker(false);  // Marker false - not last packet of frame.
+
+  EXPECT_TRUE(sender_.impl_->TrySendPacket(packet.get(), pacing_info));
+
+  // Padding not allowed in middle of frame.
+  EXPECT_THAT(sender_.impl_->GeneratePadding(kPaddingSize), SizeIs(0u));
+
+  packet = sender_.impl_->RtpSender()->AllocatePacket();
+  packet->set_packet_type(RtpPacketToSend::Type::kVideo);
+  packet->set_first_packet_of_frame(true);
+  packet->SetMarker(true);
+
+  EXPECT_TRUE(sender_.impl_->TrySendPacket(packet.get(), pacing_info));
+
+  // Padding is OK again.
+  EXPECT_THAT(sender_.impl_->GeneratePadding(kPaddingSize), SizeIs(Gt(0u)));
+}
+
+TEST_F(RtpRtcpImpl2Test, PaddingTimestampMatchesMedia) {
+  constexpr size_t kPaddingSize = 100;
+  const uint32_t kTimestamp = 123;
+
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid,
+                        kTimestamp, /*capture_time_ms=*/0));
+  EXPECT_EQ(sender_.last_packet().Timestamp(), kTimestamp);
+  uint16_t media_seq = sender_.last_packet().SequenceNumber();
+
+  // Generate and send padding.
+  auto padding = sender_.impl_->GeneratePadding(kPaddingSize);
+  ASSERT_FALSE(padding.empty());
+  for (auto& packet : padding) {
+    sender_.impl_->TrySendPacket(packet.get(), PacedPacketInfo());
+  }
+
+  // Verify we sent a new packet, but with the same timestamp.
+  EXPECT_NE(sender_.last_packet().SequenceNumber(), media_seq);
+  EXPECT_EQ(sender_.last_packet().Timestamp(), kTimestamp);
+}
+
+TEST_F(RtpRtcpImpl2Test, AssignsTransportSequenceNumber) {
+  sender_.RegisterHeaderExtension(TransportSequenceNumber::Uri(),
+                                  kTransportSequenceNumberExtensionId);
+
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  uint16_t first_transport_seq = 0;
+  EXPECT_TRUE(sender_.last_packet().GetExtension<TransportSequenceNumber>(
+      &first_transport_seq));
+
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  uint16_t second_transport_seq = 0;
+  EXPECT_TRUE(sender_.last_packet().GetExtension<TransportSequenceNumber>(
+      &second_transport_seq));
+
+  EXPECT_EQ(first_transport_seq + 1, second_transport_seq);
+}
+
+TEST_F(RtpRtcpImpl2Test, AssignsAbsoluteSendTime) {
+  sender_.RegisterHeaderExtension(AbsoluteSendTime::Uri(),
+                                  kAbsoluteSendTimeExtensionId);
+
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  EXPECT_NE(sender_.last_packet().GetExtension<AbsoluteSendTime>(), 0u);
+}
+
+TEST_F(RtpRtcpImpl2Test, AssignsTransmissionTimeOffset) {
+  sender_.RegisterHeaderExtension(TransmissionOffset::Uri(),
+                                  kTransmissionOffsetExtensionId);
+
+  constexpr TimeDelta kOffset = TimeDelta::Millis(100);
+  // Transmission offset is calculated from difference between capture time
+  // and send time.
+  int64_t capture_time_ms = time_controller_.GetClock()->TimeInMilliseconds();
+  time_controller_.AdvanceTime(kOffset);
+
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid,
+                        /*timestamp=*/0, capture_time_ms));
+  EXPECT_EQ(sender_.last_packet().GetExtension<TransmissionOffset>(),
+            kOffset.ms() * kCaptureTimeMsToRtpTimestamp);
+}
+
+TEST_F(RtpRtcpImpl2Test, PropagatesSentPacketInfo) {
+  sender_.RegisterHeaderExtension(TransportSequenceNumber::Uri(),
+                                  kTransportSequenceNumberExtensionId);
+  int64_t now_ms = time_controller_.GetClock()->TimeInMilliseconds();
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  EXPECT_THAT(
+      sender_.last_sent_packet(),
+      Optional(
+          AllOf(Field(&RtpRtcpModule::SentPacket::packet_id,
+                      Eq(sender_.last_packet()
+                             .GetExtension<TransportSequenceNumber>())),
+                Field(&RtpRtcpModule::SentPacket::capture_time_ms, Eq(now_ms)),
+                Field(&RtpRtcpModule::SentPacket::ssrc, Eq(kSenderSsrc)))));
+}
+
+TEST_F(RtpRtcpImpl2Test, GeneratesFlexfec) {
+  constexpr int kFlexfecPayloadType = 118;
+  constexpr uint32_t kFlexfecSsrc = 17;
+  const char kNoMid[] = "";
+  const std::vector<RtpExtension> kNoRtpExtensions;
+  const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
+
+  // Make sure FlexFec sequence numbers start at a different point than media.
+  const uint16_t fec_start_seq = sender_.impl_->SequenceNumber() + 100;
+  RtpState start_state;
+  start_state.sequence_number = fec_start_seq;
+  FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kSenderSsrc,
+                               kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes,
+                               &start_state, time_controller_.GetClock());
+  ReinitWithFec(&flexfec_sender, /*red_payload_type=*/absl::nullopt);
+
+  // Parameters selected to generate a single FEC packet per media packet.
+  FecProtectionParams params;
+  params.fec_rate = 15;
+  params.max_fec_frames = 1;
+  params.fec_mask_type = kFecMaskRandom;
+  sender_.impl_->SetFecProtectionParams(params, params);
+
+  // Send a one packet frame, expect one media packet and one FEC packet.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  ASSERT_THAT(sender_.transport_.rtp_packets_sent_, Eq(2));
+
+  const RtpPacketReceived& fec_packet = sender_.last_packet();
+  EXPECT_EQ(fec_packet.SequenceNumber(), fec_start_seq);
+  EXPECT_EQ(fec_packet.Ssrc(), kFlexfecSsrc);
+  EXPECT_EQ(fec_packet.PayloadType(), kFlexfecPayloadType);
+}
+
+TEST_F(RtpRtcpImpl2Test, GeneratesUlpfec) {
+  constexpr int kUlpfecPayloadType = 118;
+  constexpr int kRedPayloadType = 119;
+  UlpfecGenerator ulpfec_sender(kRedPayloadType, kUlpfecPayloadType,
+                                time_controller_.GetClock());
+  ReinitWithFec(&ulpfec_sender, kRedPayloadType);
+
+  // Parameters selected to generate a single FEC packet per media packet.
+  FecProtectionParams params;
+  params.fec_rate = 15;
+  params.max_fec_frames = 1;
+  params.fec_mask_type = kFecMaskRandom;
+  sender_.impl_->SetFecProtectionParams(params, params);
+
+  // Send a one packet frame, expect one media packet and one FEC packet.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  ASSERT_THAT(sender_.transport_.rtp_packets_sent_, Eq(2));
+
+  // Ulpfec is sent on the media ssrc, sharing the sequene number series.
+  const RtpPacketReceived& fec_packet = sender_.last_packet();
+  EXPECT_EQ(fec_packet.SequenceNumber(), kSequenceNumber + 1);
+  EXPECT_EQ(fec_packet.Ssrc(), kSenderSsrc);
+  // The packets are encapsulated in RED packets, check that and that the RED
+  // header (first byte of payload) indicates the desired FEC payload type.
+  EXPECT_EQ(fec_packet.PayloadType(), kRedPayloadType);
+  EXPECT_EQ(fec_packet.payload()[0], kUlpfecPayloadType);
+}
+
+TEST_F(RtpRtcpImpl2Test, RtpStateReflectsCurrentState) {
+  // Verify that that each of the field of GetRtpState actually reflects
+  // the current state.
+
+  // Current time will be used for `timestamp`, `capture_time_ms` and
+  // `last_timestamp_time_ms`.
+  const int64_t time_ms = time_controller_.GetClock()->TimeInMilliseconds();
+
+  // Use different than default sequence number to test `sequence_number`.
+  const uint16_t kSeq = kSequenceNumber + 123;
+  // Hard-coded value for `start_timestamp`.
+  const uint32_t kStartTimestamp = 3456;
+  const int64_t capture_time_ms = time_ms;
+  const uint32_t timestamp = capture_time_ms * kCaptureTimeMsToRtpTimestamp;
+
+  sender_.impl_->SetSequenceNumber(kSeq - 1);
+  sender_.impl_->SetStartTimestamp(kStartTimestamp);
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+
+  // Simulate an RTCP receiver report in order to populate `ssrc_has_acked`.
+  RTCPReportBlock ack;
+  ack.source_ssrc = kSenderSsrc;
+  ack.extended_highest_sequence_number = kSeq;
+  sender_.impl_->OnReceivedRtcpReportBlocks({ack});
+
+  RtpState state = sender_.impl_->GetRtpState();
+  EXPECT_EQ(state.sequence_number, kSeq);
+  EXPECT_EQ(state.start_timestamp, kStartTimestamp);
+  EXPECT_EQ(state.timestamp, timestamp);
+  EXPECT_EQ(state.capture_time_ms, capture_time_ms);
+  EXPECT_EQ(state.last_timestamp_time_ms, time_ms);
+  EXPECT_EQ(state.ssrc_has_acked, true);
+
+  // Reset sender, advance time, restore state. Directly observing state
+  // is not feasible, so just verify returned state matches what we set.
+  sender_.CreateModuleImpl();
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  sender_.impl_->SetRtpState(state);
+
+  state = sender_.impl_->GetRtpState();
+  EXPECT_EQ(state.sequence_number, kSeq);
+  EXPECT_EQ(state.start_timestamp, kStartTimestamp);
+  EXPECT_EQ(state.timestamp, timestamp);
+  EXPECT_EQ(state.capture_time_ms, capture_time_ms);
+  EXPECT_EQ(state.last_timestamp_time_ms, time_ms);
+  EXPECT_EQ(state.ssrc_has_acked, true);
+}
+
+TEST_F(RtpRtcpImpl2Test, RtxRtpStateReflectsCurrentState) {
+  // Enable RTX.
+  sender_.impl_->SetStorePacketsStatus(/*enable=*/true, /*number_to_store=*/10);
+  sender_.impl_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
+  sender_.impl_->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+
+  // `start_timestamp` is the only timestamp populate in the RTX state.
+  const uint32_t kStartTimestamp = 3456;
+  sender_.impl_->SetStartTimestamp(kStartTimestamp);
+
+  // Send a frame and ask for a retransmit of the last packet. Capture the RTX
+  // packet in order to verify RTX sequence number.
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid));
+  time_controller_.AdvanceTime(TimeDelta::Millis(5));
+  sender_.impl_->OnReceivedNack(
+      std::vector<uint16_t>{sender_.transport_.last_packet_.SequenceNumber()});
+  RtpPacketReceived& rtx_packet = sender_.transport_.last_packet_;
+  EXPECT_EQ(rtx_packet.Ssrc(), kRtxSenderSsrc);
+
+  // Simulate an RTCP receiver report in order to populate `ssrc_has_acked`.
+  RTCPReportBlock ack;
+  ack.source_ssrc = kRtxSenderSsrc;
+  ack.extended_highest_sequence_number = rtx_packet.SequenceNumber();
+  sender_.impl_->OnReceivedRtcpReportBlocks({ack});
+
+  RtpState rtp_state = sender_.impl_->GetRtpState();
+  RtpState rtx_state = sender_.impl_->GetRtxState();
+  EXPECT_EQ(rtx_state.start_timestamp, kStartTimestamp);
+  EXPECT_EQ(rtx_state.ssrc_has_acked, true);
+  EXPECT_EQ(rtx_state.sequence_number, rtx_packet.SequenceNumber() + 1);
+
+  // Reset sender, advance time, restore state. Directly observing state
+  // is not feasible, so just verify returned state matches what we set.
+  // Needs SetRtpState() too in order to propagate start timestamp.
+  sender_.CreateModuleImpl();
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  sender_.impl_->SetRtpState(rtp_state);
+  sender_.impl_->SetRtxState(rtx_state);
+
+  rtx_state = sender_.impl_->GetRtxState();
+  EXPECT_EQ(rtx_state.start_timestamp, kStartTimestamp);
+  EXPECT_EQ(rtx_state.ssrc_has_acked, true);
+  EXPECT_EQ(rtx_state.sequence_number, rtx_packet.SequenceNumber() + 1);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
new file mode 100644
index 0000000000..96bef23a3a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -0,0 +1,705 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
+
+#include <map>
+#include <memory>
+#include <set>
+
+#include "api/transport/field_trial_based_config.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
+#include "rtc_base/rate_limiter.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+
+using ::testing::ElementsAre;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::Gt;
+using ::testing::Not;
+using ::testing::Optional;
+
+namespace webrtc {
+namespace {
+const uint32_t kSenderSsrc = 0x12345;
+const uint32_t kReceiverSsrc = 0x23456;
+const int64_t kOneWayNetworkDelayMs = 100;
+const uint8_t kBaseLayerTid = 0;
+const uint8_t kHigherLayerTid = 1;
+const uint16_t kSequenceNumber = 100;
+const uint8_t kPayloadType = 100;
+const int kWidth = 320;
+const int kHeight = 100;
+
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+
+class RtcpRttStatsTestImpl : public RtcpRttStats {
+ public:
+  RtcpRttStatsTestImpl() : rtt_ms_(0) {}
+  ~RtcpRttStatsTestImpl() override = default;
+
+  void OnRttUpdate(int64_t rtt_ms) override { rtt_ms_ = rtt_ms; }
+  int64_t LastProcessedRtt() const override { return rtt_ms_; }
+  int64_t rtt_ms_;
+};
+
+class SendTransport : public Transport {
+ public:
+  SendTransport()
+      : receiver_(nullptr),
+        clock_(nullptr),
+        delay_ms_(0),
+        rtp_packets_sent_(0),
+        rtcp_packets_sent_(0) {}
+
+  void SetRtpRtcpModule(ModuleRtpRtcpImpl* receiver) { receiver_ = receiver; }
+  void SimulateNetworkDelay(int64_t delay_ms, SimulatedClock* clock) {
+    clock_ = clock;
+    delay_ms_ = delay_ms;
+  }
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override {
+    RtpPacket packet;
+    EXPECT_TRUE(packet.Parse(data, len));
+    ++rtp_packets_sent_;
+    last_rtp_sequence_number_ = packet.SequenceNumber();
+    return true;
+  }
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    test::RtcpPacketParser parser;
+    parser.Parse(data, len);
+    last_nack_list_ = parser.nack()->packet_ids();
+
+    if (clock_) {
+      clock_->AdvanceTimeMilliseconds(delay_ms_);
+    }
+    EXPECT_TRUE(receiver_);
+    receiver_->IncomingRtcpPacket(data, len);
+    ++rtcp_packets_sent_;
+    return true;
+  }
+  size_t NumRtcpSent() { return rtcp_packets_sent_; }
+  ModuleRtpRtcpImpl* receiver_;
+  SimulatedClock* clock_;
+  int64_t delay_ms_;
+  int rtp_packets_sent_;
+  size_t rtcp_packets_sent_;
+  uint16_t last_rtp_sequence_number_;
+  std::vector<uint16_t> last_nack_list_;
+};
+
+class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
+ public:
+  RtpRtcpModule(SimulatedClock* clock, bool is_sender)
+      : is_sender_(is_sender),
+        receive_statistics_(ReceiveStatistics::Create(clock)),
+        clock_(clock) {
+    CreateModuleImpl();
+    transport_.SimulateNetworkDelay(kOneWayNetworkDelayMs, clock);
+  }
+
+  const bool is_sender_;
+  RtcpPacketTypeCounter packets_sent_;
+  RtcpPacketTypeCounter packets_received_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  SendTransport transport_;
+  RtcpRttStatsTestImpl rtt_stats_;
+  std::unique_ptr<ModuleRtpRtcpImpl> impl_;
+  int rtcp_report_interval_ms_ = 0;
+
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) override {
+    counter_map_[ssrc] = packet_counter;
+  }
+
+  RtcpPacketTypeCounter RtcpSent() {
+    // RTCP counters for remote SSRC.
+    return counter_map_[is_sender_ ? kReceiverSsrc : kSenderSsrc];
+  }
+
+  RtcpPacketTypeCounter RtcpReceived() {
+    // Received RTCP stats for (own) local SSRC.
+    return counter_map_[impl_->SSRC()];
+  }
+  int RtpSent() { return transport_.rtp_packets_sent_; }
+  uint16_t LastRtpSequenceNumber() {
+    return transport_.last_rtp_sequence_number_;
+  }
+  std::vector<uint16_t> LastNackListSent() {
+    return transport_.last_nack_list_;
+  }
+  void SetRtcpReportIntervalAndReset(int rtcp_report_interval_ms) {
+    rtcp_report_interval_ms_ = rtcp_report_interval_ms;
+    CreateModuleImpl();
+  }
+
+ private:
+  void CreateModuleImpl() {
+    RtpRtcpInterface::Configuration config;
+    config.audio = false;
+    config.clock = clock_;
+    config.outgoing_transport = &transport_;
+    config.receive_statistics = receive_statistics_.get();
+    config.rtcp_packet_type_counter_observer = this;
+    config.rtt_stats = &rtt_stats_;
+    config.rtcp_report_interval_ms = rtcp_report_interval_ms_;
+    config.local_media_ssrc = is_sender_ ? kSenderSsrc : kReceiverSsrc;
+    config.need_rtp_packet_infos = true;
+    config.non_sender_rtt_measurement = true;
+
+    impl_.reset(new ModuleRtpRtcpImpl(config));
+    impl_->SetRemoteSSRC(is_sender_ ? kReceiverSsrc : kSenderSsrc);
+    impl_->SetRTCPStatus(RtcpMode::kCompound);
+  }
+
+  SimulatedClock* const clock_;
+  std::map<uint32_t, RtcpPacketTypeCounter> counter_map_;
+};
+}  // namespace
+
+class RtpRtcpImplTest : public ::testing::Test {
+ protected:
+  RtpRtcpImplTest()
+      : clock_(133590000000000),
+        sender_(&clock_, /*is_sender=*/true),
+        receiver_(&clock_, /*is_sender=*/false) {}
+
+  void SetUp() override {
+    // Send module.
+    EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true));
+    sender_.impl_->SetSendingMediaStatus(true);
+    sender_.impl_->SetSequenceNumber(kSequenceNumber);
+    sender_.impl_->SetStorePacketsStatus(true, 100);
+
+    FieldTrialBasedConfig field_trials;
+    RTPSenderVideo::Config video_config;
+    video_config.clock = &clock_;
+    video_config.rtp_sender = sender_.impl_->RtpSender();
+    video_config.field_trials = &field_trials;
+    sender_video_ = std::make_unique<RTPSenderVideo>(video_config);
+
+    // Receive module.
+    EXPECT_EQ(0, receiver_.impl_->SetSendingStatus(false));
+    receiver_.impl_->SetSendingMediaStatus(false);
+    // Transport settings.
+    sender_.transport_.SetRtpRtcpModule(receiver_.impl_.get());
+    receiver_.transport_.SetRtpRtcpModule(sender_.impl_.get());
+  }
+
+  SimulatedClock clock_;
+  RtpRtcpModule sender_;
+  std::unique_ptr<RTPSenderVideo> sender_video_;
+  RtpRtcpModule receiver_;
+
+  void SendFrame(const RtpRtcpModule* module,
+                 RTPSenderVideo* sender,
+                 uint8_t tid) {
+    RTPVideoHeaderVP8 vp8_header = {};
+    vp8_header.temporalIdx = tid;
+    RTPVideoHeader rtp_video_header;
+    rtp_video_header.frame_type = VideoFrameType::kVideoFrameKey;
+    rtp_video_header.width = kWidth;
+    rtp_video_header.height = kHeight;
+    rtp_video_header.rotation = kVideoRotation_0;
+    rtp_video_header.content_type = VideoContentType::UNSPECIFIED;
+    rtp_video_header.playout_delay = {-1, -1};
+    rtp_video_header.is_first_packet_in_frame = true;
+    rtp_video_header.simulcastIdx = 0;
+    rtp_video_header.codec = kVideoCodecVP8;
+    rtp_video_header.video_type_header = vp8_header;
+    rtp_video_header.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false};
+
+    const uint8_t payload[100] = {0};
+    EXPECT_TRUE(module->impl_->OnSendingRtpFrame(0, 0, kPayloadType, true));
+    EXPECT_TRUE(sender->SendVideo(kPayloadType, VideoCodecType::kVideoCodecVP8,
+                                  0, 0, payload, rtp_video_header, 0));
+  }
+
+  void IncomingRtcpNack(const RtpRtcpModule* module, uint16_t sequence_number) {
+    bool sender = module->impl_->SSRC() == kSenderSsrc;
+    rtcp::Nack nack;
+    uint16_t list[1];
+    list[0] = sequence_number;
+    const uint16_t kListLength = sizeof(list) / sizeof(list[0]);
+    nack.SetSenderSsrc(sender ? kReceiverSsrc : kSenderSsrc);
+    nack.SetMediaSsrc(sender ? kSenderSsrc : kReceiverSsrc);
+    nack.SetPacketIds(list, kListLength);
+    rtc::Buffer packet = nack.Build();
+    module->impl_->IncomingRtcpPacket(packet.data(), packet.size());
+  }
+};
+
+TEST_F(RtpRtcpImplTest, RetransmitsAllLayers) {
+  // Send frames.
+  EXPECT_EQ(0, sender_.RtpSent());
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);  // kSequenceNumber
+  SendFrame(&sender_, sender_video_.get(),
+            kHigherLayerTid);  // kSequenceNumber + 1
+  SendFrame(&sender_, sender_video_.get(),
+            kNoTemporalIdx);  // kSequenceNumber + 2
+  EXPECT_EQ(3, sender_.RtpSent());
+  EXPECT_EQ(kSequenceNumber + 2, sender_.LastRtpSequenceNumber());
+
+  // Min required delay until retransmit = 5 + RTT ms (RTT = 0).
+  clock_.AdvanceTimeMilliseconds(5);
+
+  // Frame with kBaseLayerTid re-sent.
+  IncomingRtcpNack(&sender_, kSequenceNumber);
+  EXPECT_EQ(4, sender_.RtpSent());
+  EXPECT_EQ(kSequenceNumber, sender_.LastRtpSequenceNumber());
+  // Frame with kHigherLayerTid re-sent.
+  IncomingRtcpNack(&sender_, kSequenceNumber + 1);
+  EXPECT_EQ(5, sender_.RtpSent());
+  EXPECT_EQ(kSequenceNumber + 1, sender_.LastRtpSequenceNumber());
+  // Frame with kNoTemporalIdx re-sent.
+  IncomingRtcpNack(&sender_, kSequenceNumber + 2);
+  EXPECT_EQ(6, sender_.RtpSent());
+  EXPECT_EQ(kSequenceNumber + 2, sender_.LastRtpSequenceNumber());
+}
+
+TEST_F(RtpRtcpImplTest, Rtt) {
+  RtpPacketReceived packet;
+  packet.SetTimestamp(1);
+  packet.SetSequenceNumber(123);
+  packet.SetSsrc(kSenderSsrc);
+  packet.AllocatePayload(100 - 12);
+  receiver_.receive_statistics_->OnRtpPacket(packet);
+
+  // Send Frame before sending an SR.
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  // Sender module should send an SR.
+  EXPECT_EQ(0, sender_.impl_->SendRTCP(kRtcpReport));
+
+  // Receiver module should send a RR with a response to the last received SR.
+  clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport));
+
+  // Verify RTT.
+  int64_t rtt;
+  int64_t avg_rtt;
+  int64_t min_rtt;
+  int64_t max_rtt;
+  EXPECT_EQ(
+      0, sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, rtt, 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, avg_rtt, 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, min_rtt, 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, max_rtt, 1);
+
+  // No RTT from other ssrc.
+  EXPECT_EQ(-1, sender_.impl_->RTT(kReceiverSsrc + 1, &rtt, &avg_rtt, &min_rtt,
+                                   &max_rtt));
+
+  // Verify RTT from rtt_stats config.
+  EXPECT_EQ(0, sender_.rtt_stats_.LastProcessedRtt());
+  EXPECT_EQ(0, sender_.impl_->rtt_ms());
+  sender_.impl_->Process();
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, sender_.rtt_stats_.LastProcessedRtt(),
+              1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms(), 1);
+}
+
+TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
+  // Receiver module should send a Receiver time reference report (RTRR).
+  EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport));
+
+  // Sender module should send a response to the last received RTRR (DLRR).
+  clock_.AdvanceTimeMilliseconds(1000);
+  // Send Frame before sending a SR.
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  EXPECT_EQ(0, sender_.impl_->SendRTCP(kRtcpReport));
+
+  // Verify RTT.
+  EXPECT_EQ(0, receiver_.rtt_stats_.LastProcessedRtt());
+  EXPECT_EQ(0, receiver_.impl_->rtt_ms());
+  receiver_.impl_->Process();
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs,
+              receiver_.rtt_stats_.LastProcessedRtt(), 1);
+  EXPECT_NEAR(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms(), 1);
+}
+
+TEST_F(RtpRtcpImplTest, NoSrBeforeMedia) {
+  // Ignore fake transport delays in this test.
+  sender_.transport_.SimulateNetworkDelay(0, &clock_);
+  receiver_.transport_.SimulateNetworkDelay(0, &clock_);
+
+  sender_.impl_->Process();
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 0u);
+
+  // Verify no SR is sent before media has been sent, RR should still be sent
+  // from the receiving module though.
+  clock_.AdvanceTimeMilliseconds(2000);
+  sender_.impl_->Process();
+  receiver_.impl_->Process();
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 0u);
+  EXPECT_EQ(receiver_.transport_.NumRtcpSent(), 1u);
+
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u);
+}
+
+TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_Nack) {
+  EXPECT_EQ(0U, sender_.RtcpReceived().nack_packets);
+  EXPECT_EQ(0U, receiver_.RtcpSent().nack_packets);
+
+  // Receive module sends a NACK.
+  const uint16_t kNackLength = 1;
+  uint16_t nack_list[kNackLength] = {123};
+  EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, receiver_.RtcpSent().nack_packets);
+
+  // Send module receives the NACK.
+  EXPECT_EQ(1U, sender_.RtcpReceived().nack_packets);
+}
+
+TEST_F(RtpRtcpImplTest, AddStreamDataCounters) {
+  StreamDataCounters rtp;
+  rtp.transmitted.packets = 1;
+  rtp.transmitted.payload_bytes = 1;
+  rtp.transmitted.header_bytes = 2;
+  rtp.transmitted.padding_bytes = 3;
+  EXPECT_EQ(rtp.transmitted.TotalBytes(), rtp.transmitted.payload_bytes +
+                                              rtp.transmitted.header_bytes +
+                                              rtp.transmitted.padding_bytes);
+
+  StreamDataCounters rtp2;
+  rtp2.transmitted.packets = 10;
+  rtp2.transmitted.payload_bytes = 10;
+  rtp2.retransmitted.header_bytes = 4;
+  rtp2.retransmitted.payload_bytes = 5;
+  rtp2.retransmitted.padding_bytes = 6;
+  rtp2.retransmitted.packets = 7;
+  rtp2.fec.packets = 8;
+
+  StreamDataCounters sum = rtp;
+  sum.Add(rtp2);
+  EXPECT_EQ(11U, sum.transmitted.packets);
+  EXPECT_EQ(11U, sum.transmitted.payload_bytes);
+  EXPECT_EQ(2U, sum.transmitted.header_bytes);
+  EXPECT_EQ(3U, sum.transmitted.padding_bytes);
+  EXPECT_EQ(4U, sum.retransmitted.header_bytes);
+  EXPECT_EQ(5U, sum.retransmitted.payload_bytes);
+  EXPECT_EQ(6U, sum.retransmitted.padding_bytes);
+  EXPECT_EQ(7U, sum.retransmitted.packets);
+  EXPECT_EQ(8U, sum.fec.packets);
+  EXPECT_EQ(sum.transmitted.TotalBytes(),
+            rtp.transmitted.TotalBytes() + rtp2.transmitted.TotalBytes());
+}
+
+TEST_F(RtpRtcpImplTest, SendsInitialNackList) {
+  // Send module sends a NACK.
+  const uint16_t kNackLength = 1;
+  uint16_t nack_list[kNackLength] = {123};
+  EXPECT_EQ(0U, sender_.RtcpSent().nack_packets);
+  // Send Frame before sending a compound RTCP that starts with SR.
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123));
+}
+
+TEST_F(RtpRtcpImplTest, SendsExtendedNackList) {
+  // Send module sends a NACK.
+  const uint16_t kNackLength = 1;
+  uint16_t nack_list[kNackLength] = {123};
+  EXPECT_EQ(0U, sender_.RtcpSent().nack_packets);
+  // Send Frame before sending a compound RTCP that starts with SR.
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123));
+
+  // Same list not re-send.
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123));
+
+  // Only extended list sent.
+  const uint16_t kNackExtLength = 2;
+  uint16_t nack_list_ext[kNackExtLength] = {123, 124};
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list_ext, kNackExtLength));
+  EXPECT_EQ(2U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(124));
+}
+
+TEST_F(RtpRtcpImplTest, ReSendsNackListAfterRttMs) {
+  sender_.transport_.SimulateNetworkDelay(0, &clock_);
+  // Send module sends a NACK.
+  const uint16_t kNackLength = 2;
+  uint16_t nack_list[kNackLength] = {123, 125};
+  EXPECT_EQ(0U, sender_.RtcpSent().nack_packets);
+  // Send Frame before sending a compound RTCP that starts with SR.
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123, 125));
+
+  // Same list not re-send, rtt interval has not passed.
+  const int kStartupRttMs = 100;
+  clock_.AdvanceTimeMilliseconds(kStartupRttMs);
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, sender_.RtcpSent().nack_packets);
+
+  // Rtt interval passed, full list sent.
+  clock_.AdvanceTimeMilliseconds(1);
+  EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(2U, sender_.RtcpSent().nack_packets);
+  EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123, 125));
+}
+
+TEST_F(RtpRtcpImplTest, UniqueNackRequests) {
+  receiver_.transport_.SimulateNetworkDelay(0, &clock_);
+  EXPECT_EQ(0U, receiver_.RtcpSent().nack_packets);
+  EXPECT_EQ(0U, receiver_.RtcpSent().nack_requests);
+  EXPECT_EQ(0U, receiver_.RtcpSent().unique_nack_requests);
+  EXPECT_EQ(0, receiver_.RtcpSent().UniqueNackRequestsInPercent());
+
+  // Receive module sends NACK request.
+  const uint16_t kNackLength = 4;
+  uint16_t nack_list[kNackLength] = {10, 11, 13, 18};
+  EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list, kNackLength));
+  EXPECT_EQ(1U, receiver_.RtcpSent().nack_packets);
+  EXPECT_EQ(4U, receiver_.RtcpSent().nack_requests);
+  EXPECT_EQ(4U, receiver_.RtcpSent().unique_nack_requests);
+  EXPECT_THAT(receiver_.LastNackListSent(), ElementsAre(10, 11, 13, 18));
+
+  // Send module receives the request.
+  EXPECT_EQ(1U, sender_.RtcpReceived().nack_packets);
+  EXPECT_EQ(4U, sender_.RtcpReceived().nack_requests);
+  EXPECT_EQ(4U, sender_.RtcpReceived().unique_nack_requests);
+  EXPECT_EQ(100, sender_.RtcpReceived().UniqueNackRequestsInPercent());
+
+  // Receive module sends new request with duplicated packets.
+  const int kStartupRttMs = 100;
+  clock_.AdvanceTimeMilliseconds(kStartupRttMs + 1);
+  const uint16_t kNackLength2 = 4;
+  uint16_t nack_list2[kNackLength2] = {11, 18, 20, 21};
+  EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list2, kNackLength2));
+  EXPECT_EQ(2U, receiver_.RtcpSent().nack_packets);
+  EXPECT_EQ(8U, receiver_.RtcpSent().nack_requests);
+  EXPECT_EQ(6U, receiver_.RtcpSent().unique_nack_requests);
+  EXPECT_THAT(receiver_.LastNackListSent(), ElementsAre(11, 18, 20, 21));
+
+  // Send module receives the request.
+  EXPECT_EQ(2U, sender_.RtcpReceived().nack_packets);
+  EXPECT_EQ(8U, sender_.RtcpReceived().nack_requests);
+  EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests);
+  EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent());
+}
+
+TEST_F(RtpRtcpImplTest, ConfigurableRtcpReportInterval) {
+  const int kVideoReportInterval = 3000;
+
+  // Recreate sender impl with new configuration, and redo setup.
+  sender_.SetRtcpReportIntervalAndReset(kVideoReportInterval);
+  SetUp();
+
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+
+  // Initial state
+  sender_.impl_->Process();
+  EXPECT_EQ(0u, sender_.transport_.NumRtcpSent());
+
+  // Move ahead to the last ms before a rtcp is expected, no action.
+  clock_.AdvanceTimeMilliseconds(kVideoReportInterval / 2 - 1);
+  sender_.impl_->Process();
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 0u);
+
+  // Move ahead to the first rtcp. Send RTCP.
+  clock_.AdvanceTimeMilliseconds(1);
+  sender_.impl_->Process();
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u);
+
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+
+  // Move ahead to the last possible second before second rtcp is expected.
+  clock_.AdvanceTimeMilliseconds(kVideoReportInterval * 1 / 2 - 1);
+  sender_.impl_->Process();
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u);
+
+  // Move ahead into the range of second rtcp, the second rtcp may be sent.
+  clock_.AdvanceTimeMilliseconds(1);
+  sender_.impl_->Process();
+  EXPECT_GE(sender_.transport_.NumRtcpSent(), 1u);
+
+  clock_.AdvanceTimeMilliseconds(kVideoReportInterval / 2);
+  sender_.impl_->Process();
+  EXPECT_GE(sender_.transport_.NumRtcpSent(), 1u);
+
+  // Move out the range of second rtcp, the second rtcp must have been sent.
+  clock_.AdvanceTimeMilliseconds(kVideoReportInterval / 2);
+  sender_.impl_->Process();
+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 2u);
+}
+
+TEST_F(RtpRtcpImplTest, StoresPacketInfoForSentPackets) {
+  const uint32_t kStartTimestamp = 1u;
+  SetUp();
+  sender_.impl_->SetStartTimestamp(kStartTimestamp);
+  sender_.impl_->SetSequenceNumber(1);
+
+  PacedPacketInfo pacing_info;
+  RtpPacketToSend packet(nullptr);
+  packet.set_packet_type(RtpPacketToSend::Type::kVideo);
+  packet.SetSsrc(kSenderSsrc);
+
+  // Single-packet frame.
+  packet.SetTimestamp(1);
+  packet.set_first_packet_of_frame(true);
+  packet.SetMarker(true);
+  sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+  std::vector<RtpSequenceNumberMap::Info> seqno_info =
+      sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{1});
+
+  EXPECT_THAT(seqno_info, ElementsAre(RtpSequenceNumberMap::Info(
+                              /*timestamp=*/1 - kStartTimestamp,
+                              /*is_first=*/1,
+                              /*is_last=*/1)));
+
+  // Three-packet frame.
+  packet.SetTimestamp(2);
+  packet.set_first_packet_of_frame(true);
+  packet.SetMarker(false);
+  sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+  packet.set_first_packet_of_frame(false);
+  sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+  packet.SetMarker(true);
+  sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+  seqno_info =
+      sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{2, 3, 4});
+
+  EXPECT_THAT(seqno_info, ElementsAre(RtpSequenceNumberMap::Info(
+                                          /*timestamp=*/2 - kStartTimestamp,
+                                          /*is_first=*/1,
+                                          /*is_last=*/0),
+                                      RtpSequenceNumberMap::Info(
+                                          /*timestamp=*/2 - kStartTimestamp,
+                                          /*is_first=*/0,
+                                          /*is_last=*/0),
+                                      RtpSequenceNumberMap::Info(
+                                          /*timestamp=*/2 - kStartTimestamp,
+                                          /*is_first=*/0,
+                                          /*is_last=*/1)));
+}
+
+// Checks that the remote sender stats are not available if no RTCP SR was sent.
+TEST_F(RtpRtcpImplTest, SenderReportStatsNotAvailable) {
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt));
+}
+
+// Checks that the remote sender stats are available if an RTCP SR was sent.
+TEST_F(RtpRtcpImplTest, SenderReportStatsAvailable) {
+  // Send a frame in order to send an SR.
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  // Send an SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Not(Eq(absl::nullopt)));
+}
+
+// Checks that the remote sender stats are not available if an RTCP SR with an
+// unexpected SSRC is received.
+TEST_F(RtpRtcpImplTest, SenderReportStatsNotUpdatedWithUnexpectedSsrc) {
+  constexpr uint32_t kUnexpectedSenderSsrc = 0x87654321;
+  static_assert(kUnexpectedSenderSsrc != kSenderSsrc, "");
+  // Forge a sender report and pass it to the receiver as if an RTCP SR were
+  // sent by an unexpected sender.
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kUnexpectedSenderSsrc);
+  sr.SetNtp({/*seconds=*/1u, /*fractions=*/1u << 31});
+  sr.SetPacketCount(123u);
+  sr.SetOctetCount(456u);
+  auto raw_packet = sr.Build();
+  receiver_.impl_->IncomingRtcpPacket(raw_packet.data(), raw_packet.size());
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt));
+}
+
+// Checks the stats derived from the last received RTCP SR are set correctly.
+TEST_F(RtpRtcpImplTest, SenderReportStatsCheckStatsFromLastReport) {
+  using SenderReportStats = RtpRtcpInterface::SenderReportStats;
+  const NtpTime ntp(/*seconds=*/1u, /*fractions=*/1u << 31);
+  constexpr uint32_t kPacketCount = 123u;
+  constexpr uint32_t kOctetCount = 456u;
+  // Forge a sender report and pass it to the receiver as if an RTCP SR were
+  // sent by the sender.
+  rtcp::SenderReport sr;
+  sr.SetSenderSsrc(kSenderSsrc);
+  sr.SetNtp(ntp);
+  sr.SetPacketCount(kPacketCount);
+  sr.SetOctetCount(kOctetCount);
+  auto raw_packet = sr.Build();
+  receiver_.impl_->IncomingRtcpPacket(raw_packet.data(), raw_packet.size());
+
+  EXPECT_THAT(
+      receiver_.impl_->GetSenderReportStats(),
+      Optional(AllOf(Field(&SenderReportStats::last_remote_timestamp, Eq(ntp)),
+                     Field(&SenderReportStats::packets_sent, Eq(kPacketCount)),
+                     Field(&SenderReportStats::bytes_sent, Eq(kOctetCount)))));
+}
+
+// Checks that the remote sender stats count equals the number of sent RTCP SRs.
+TEST_F(RtpRtcpImplTest, SenderReportStatsCount) {
+  using SenderReportStats = RtpRtcpInterface::SenderReportStats;
+  // Send a frame in order to send an SR.
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  // Send the first SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(),
+              Optional(Field(&SenderReportStats::reports_count, Eq(1u))));
+  // Send the second SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(),
+              Optional(Field(&SenderReportStats::reports_count, Eq(2u))));
+}
+
+// Checks that the remote sender stats include a valid arrival time if an RTCP
+// SR was sent.
+TEST_F(RtpRtcpImplTest, SenderReportStatsArrivalTimestampSet) {
+  // Send a frame in order to send an SR.
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  // Send an SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  auto stats = receiver_.impl_->GetSenderReportStats();
+  ASSERT_THAT(stats, Not(Eq(absl::nullopt)));
+  EXPECT_TRUE(stats->last_arrival_timestamp.Valid());
+}
+
+// Checks that the packet and byte counters from an RTCP SR are not zero once
+// a frame is sent.
+TEST_F(RtpRtcpImplTest, SenderReportStatsPacketByteCounters) {
+  using SenderReportStats = RtpRtcpInterface::SenderReportStats;
+  // Send a frame in order to send an SR.
+  SendFrame(&sender_, sender_video_.get(), kBaseLayerTid);
+  ASSERT_THAT(sender_.transport_.rtp_packets_sent_, Gt(0));
+  // Advance time otherwise the RTCP SR report will not include any packets
+  // generated by `SendFrame()`.
+  clock_.AdvanceTimeMilliseconds(1);
+  // Send an SR.
+  ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0));
+  EXPECT_THAT(receiver_.impl_->GetSenderReportStats(),
+              Optional(AllOf(Field(&SenderReportStats::packets_sent, Gt(0u)),
+                             Field(&SenderReportStats::bytes_sent, Gt(0u)))));
+}
+
+#pragma clang diagnostic pop
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_interface.h
new file mode 100644
index 0000000000..cb4a0a427f
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -0,0 +1,468 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
+#include "modules/rtp_rtcp/source/video_fec_generator.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class FrameEncryptorInterface;
+class RateLimiter;
+class RtcEventLog;
+class RTPSender;
+class Transport;
+class VideoBitrateAllocationObserver;
+
+class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
+ public:
+  struct Configuration {
+    Configuration() = default;
+    Configuration(Configuration&& rhs) = default;
+
+    Configuration(const Configuration&) = delete;
+    Configuration& operator=(const Configuration&) = delete;
+
+    // True for a audio version of the RTP/RTCP module object false will create
+    // a video version.
+    bool audio = false;
+    bool receiver_only = false;
+
+    // The clock to use to read time. If nullptr then system clock will be used.
+    Clock* clock = nullptr;
+
+    ReceiveStatisticsProvider* receive_statistics = nullptr;
+
+    // Transport object that will be called when packets are ready to be sent
+    // out on the network.
+    Transport* outgoing_transport = nullptr;
+
+    // Called when the receiver requests an intra frame.
+    RtcpIntraFrameObserver* intra_frame_callback = nullptr;
+
+    // Called when the receiver sends a loss notification.
+    RtcpLossNotificationObserver* rtcp_loss_notification_observer = nullptr;
+
+    // Called when we receive a changed estimate from the receiver of out
+    // stream.
+    RtcpBandwidthObserver* bandwidth_callback = nullptr;
+
+    // Called when we receive a RTCP bye or timeout
+    RtcpEventObserver* rtcp_event_observer = nullptr;
+
+    NetworkStateEstimateObserver* network_state_estimate_observer = nullptr;
+    TransportFeedbackObserver* transport_feedback_callback = nullptr;
+    VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr;
+    RtcpRttStats* rtt_stats = nullptr;
+    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr;
+    // Called on receipt of RTCP report block from remote side.
+    // TODO(bugs.webrtc.org/10679): Consider whether we want to use
+    // only getters or only callbacks. If we decide on getters, the
+    // ReportBlockDataObserver should also be removed in favor of
+    // GetLatestReportBlockData().
+    RtcpCnameCallback* rtcp_cname_callback = nullptr;
+    ReportBlockDataObserver* report_block_data_observer = nullptr;
+
+    // Spread any bursts of packets into smaller bursts to minimize packet loss.
+    RtpPacketSender* paced_sender = nullptr;
+
+    // Generates FEC packets.
+    // TODO(sprang): Wire up to RtpSenderEgress.
+    VideoFecGenerator* fec_generator = nullptr;
+
+    BitrateStatisticsObserver* send_bitrate_observer = nullptr;
+    SendSideDelayObserver* send_side_delay_observer = nullptr;
+    RtcEventLog* event_log = nullptr;
+    SendPacketObserver* send_packet_observer = nullptr;
+    RateLimiter* retransmission_rate_limiter = nullptr;
+    StreamDataCountersCallback* rtp_stats_callback = nullptr;
+
+    int rtcp_report_interval_ms = 0;
+
+    // Update network2 instead of pacer_exit field of video timing extension.
+    bool populate_network2_timestamp = false;
+
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer;
+
+    // E2EE Custom Video Frame Encryption
+    FrameEncryptorInterface* frame_encryptor = nullptr;
+    // Require all outgoing frames to be encrypted with a FrameEncryptor.
+    bool require_frame_encryption = false;
+
+    // Corresponds to extmap-allow-mixed in SDP negotiation.
+    bool extmap_allow_mixed = false;
+
+    // If true, the RTP sender will always annotate outgoing packets with
+    // MID and RID header extensions, if provided and negotiated.
+    // If false, the RTP sender will stop sending MID and RID header extensions,
+    // when it knows that the receiver is ready to demux based on SSRC. This is
+    // done by RTCP RR acking.
+    bool always_send_mid_and_rid = false;
+
+    // If set, field trials are read from `field_trials`, otherwise
+    // defaults to  webrtc::FieldTrialBasedConfig.
+    const FieldTrialsView* field_trials = nullptr;
+
+    // SSRCs for media and retransmission, respectively.
+    // FlexFec SSRC is fetched from `flexfec_sender`.
+    uint32_t local_media_ssrc = 0;
+    absl::optional<uint32_t> rtx_send_ssrc;
+
+    bool need_rtp_packet_infos = false;
+
+    // If true, the RTP packet history will select RTX packets based on
+    // heuristics such as send time, retransmission count etc, in order to
+    // make padding potentially more useful.
+    // If false, the last packet will always be picked. This may reduce CPU
+    // overhead.
+    bool enable_rtx_padding_prioritization = true;
+
+    // Estimate RTT as non-sender as described in
+    // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
+    bool non_sender_rtt_measurement = false;
+
+    // If non-empty, sets the value for sending in the RID (and Repaired) RTP
+    // header extension. RIDs are used to identify an RTP stream if SSRCs are
+    // not negotiated. If the RID and Repaired RID extensions are not
+    // registered, the RID will not be sent.
+    std::string rid;
+  };
+
+  // Stats for RTCP sender reports (SR) for a specific SSRC.
+  // Refer to https://tools.ietf.org/html/rfc3550#section-6.4.1.
+  struct SenderReportStats {
+    // Arrival NTP timestamp for the last received RTCP SR.
+    NtpTime last_arrival_timestamp;
+    // Received (a.k.a., remote) NTP timestamp for the last received RTCP SR.
+    NtpTime last_remote_timestamp;
+    // Total number of RTP data packets transmitted by the sender since starting
+    // transmission up until the time this SR packet was generated. The count
+    // should be reset if the sender changes its SSRC identifier.
+    uint32_t packets_sent;
+    // Total number of payload octets (i.e., not including header or padding)
+    // transmitted in RTP data packets by the sender since starting transmission
+    // up until the time this SR packet was generated. The count should be reset
+    // if the sender changes its SSRC identifier.
+    uint64_t bytes_sent;
+    // Total number of RTCP SR blocks received.
+    // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-reportssent.
+    uint64_t reports_count;
+  };
+  // Stats about the non-sender SSRC, based on RTCP extended reports (XR).
+  // Refer to https://datatracker.ietf.org/doc/html/rfc3611#section-2.
+  struct NonSenderRttStats {
+    // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
+    absl::optional<TimeDelta> round_trip_time;
+    // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
+    TimeDelta total_round_trip_time = TimeDelta::Zero();
+    // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
+    int round_trip_time_measurements = 0;
+  };
+
+  // **************************************************************************
+  // Receiver functions
+  // **************************************************************************
+
+  virtual void IncomingRtcpPacket(const uint8_t* incoming_packet,
+                                  size_t incoming_packet_length) = 0;
+
+  virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
+
+  // Called when the local ssrc changes (post initialization) for receive
+  // streams to match with send. Called on the packet receive thread/tq.
+  virtual void SetLocalSsrc(uint32_t ssrc) = 0;
+
+  // **************************************************************************
+  // Sender
+  // **************************************************************************
+
+  // Sets the maximum size of an RTP packet, including RTP headers.
+  virtual void SetMaxRtpPacketSize(size_t size) = 0;
+
+  // Returns max RTP packet size. Takes into account RTP headers and
+  // FEC/ULP/RED overhead (when FEC is enabled).
+  virtual size_t MaxRtpPacketSize() const = 0;
+
+  virtual void RegisterSendPayloadFrequency(int payload_type,
+                                            int payload_frequency) = 0;
+
+  // Unregisters a send payload.
+  // `payload_type` - payload type of codec
+  // Returns -1 on failure else 0.
+  virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0;
+
+  virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0;
+
+  // Register extension by uri, triggers CHECK on falure.
+  virtual void RegisterRtpHeaderExtension(absl::string_view uri, int id) = 0;
+
+  virtual void DeregisterSendRtpHeaderExtension(absl::string_view uri) = 0;
+
+  // Returns true if RTP module is send media, and any of the extensions
+  // required for bandwidth estimation is registered.
+  virtual bool SupportsPadding() const = 0;
+  // Same as SupportsPadding(), but additionally requires that
+  // SetRtxSendStatus() has been called with the kRtxRedundantPayloads option
+  // enabled.
+  virtual bool SupportsRtxPayloadPadding() const = 0;
+
+  // Returns start timestamp.
+  virtual uint32_t StartTimestamp() const = 0;
+
+  // Sets start timestamp. Start timestamp is set to a random value if this
+  // function is never called.
+  virtual void SetStartTimestamp(uint32_t timestamp) = 0;
+
+  // Returns SequenceNumber.
+  virtual uint16_t SequenceNumber() const = 0;
+
+  // Sets SequenceNumber, default is a random number.
+  virtual void SetSequenceNumber(uint16_t seq) = 0;
+
+  virtual void SetRtpState(const RtpState& rtp_state) = 0;
+  virtual void SetRtxState(const RtpState& rtp_state) = 0;
+  virtual RtpState GetRtpState() const = 0;
+  virtual RtpState GetRtxState() const = 0;
+
+  // This can be used to enable/disable receive-side RTT.
+  virtual void SetNonSenderRttMeasurement(bool enabled) = 0;
+
+  // Returns SSRC.
+  virtual uint32_t SSRC() const = 0;
+
+  // Sets the value for sending in the MID RTP header extension.
+  // The MID RTP header extension should be registered for this to do anything.
+  // Once set, this value can not be changed or removed.
+  virtual void SetMid(absl::string_view mid) = 0;
+
+  // Turns on/off sending RTX (RFC 4588). The modes can be set as a combination
+  // of values of the enumerator RtxMode.
+  virtual void SetRtxSendStatus(int modes) = 0;
+
+  // Returns status of sending RTX (RFC 4588). The returned value can be
+  // a combination of values of the enumerator RtxMode.
+  virtual int RtxSendStatus() const = 0;
+
+  // Returns the SSRC used for RTX if set, otherwise a nullopt.
+  virtual absl::optional<uint32_t> RtxSsrc() const = 0;
+
+  // Sets the payload type to use when sending RTX packets. Note that this
+  // doesn't enable RTX, only the payload type is set.
+  virtual void SetRtxSendPayloadType(int payload_type,
+                                     int associated_payload_type) = 0;
+
+  // Returns the FlexFEC SSRC, if there is one.
+  virtual absl::optional<uint32_t> FlexfecSsrc() const = 0;
+
+  // Sets sending status. Sends kRtcpByeCode when going from true to false.
+  // Returns -1 on failure else 0.
+  virtual int32_t SetSendingStatus(bool sending) = 0;
+
+  // Returns current sending status.
+  virtual bool Sending() const = 0;
+
+  // Starts/Stops media packets. On by default.
+  virtual void SetSendingMediaStatus(bool sending) = 0;
+
+  // Returns current media sending status.
+  virtual bool SendingMedia() const = 0;
+
+  // Returns whether audio is configured (i.e. Configuration::audio = true).
+  virtual bool IsAudioConfigured() const = 0;
+
+  // Indicate that the packets sent by this module should be counted towards the
+  // bitrate estimate since the stream participates in the bitrate allocation.
+  virtual void SetAsPartOfAllocation(bool part_of_allocation) = 0;
+
+  // Returns bitrate sent (post-pacing) per packet type.
+  virtual RtpSendRates GetSendRates() const = 0;
+
+  virtual RTPSender* RtpSender() = 0;
+  virtual const RTPSender* RtpSender() const = 0;
+
+  // Record that a frame is about to be sent. Returns true on success, and false
+  // if the module isn't ready to send.
+  virtual bool OnSendingRtpFrame(uint32_t timestamp,
+                                 int64_t capture_time_ms,
+                                 int payload_type,
+                                 bool force_sender_report) = 0;
+
+  // Try to send the provided packet. Returns true iff packet matches any of
+  // the SSRCs for this module (media/rtx/fec etc) and was forwarded to the
+  // transport.
+  virtual bool TrySendPacket(RtpPacketToSend* packet,
+                             const PacedPacketInfo& pacing_info) = 0;
+
+  // Update the FEC protection parameters to use for delta- and key-frames.
+  // Only used when deferred FEC is active.
+  virtual void SetFecProtectionParams(
+      const FecProtectionParams& delta_params,
+      const FecProtectionParams& key_params) = 0;
+
+  // If deferred FEC generation is enabled, this method should be called after
+  // calling TrySendPacket(). Any generated FEC packets will be removed and
+  // returned from the FEC generator.
+  virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() = 0;
+
+  virtual void OnAbortedRetransmissions(
+      rtc::ArrayView<const uint16_t> sequence_numbers) = 0;
+
+  virtual void OnPacketsAcknowledged(
+      rtc::ArrayView<const uint16_t> sequence_numbers) = 0;
+
+  virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      size_t target_size_bytes) = 0;
+
+  virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+      rtc::ArrayView<const uint16_t> sequence_numbers) const = 0;
+
+  // Returns an expected per packet overhead representing the main RTP header,
+  // any CSRCs, and the registered header extensions that are expected on all
+  // packets (i.e. disregarding things like abs capture time which is only
+  // populated on a subset of packets, but counting MID/RID type extensions
+  // when we expect to send them).
+  virtual size_t ExpectedPerPacketOverhead() const = 0;
+
+  // Access to packet state (e.g. sequence numbering) must only be access by
+  // one thread at a time. It may be only one thread, or a construction thread
+  // that calls SetRtpState() - handing over to a pacer thread that calls
+  // TrySendPacket() - and at teardown ownership is handed to a destruciton
+  // thread that calls GetRtpState().
+  // This method is used to signal that "ownership" of the rtp state is being
+  // transferred to another thread.
+  virtual void OnPacketSendingThreadSwitched() = 0;
+
+  // **************************************************************************
+  // RTCP
+  // **************************************************************************
+
+  // Returns RTCP status.
+  virtual RtcpMode RTCP() const = 0;
+
+  // Sets RTCP status i.e on(compound or non-compound)/off.
+  // `method` - RTCP method to use.
+  virtual void SetRTCPStatus(RtcpMode method) = 0;
+
+  // Sets RTCP CName (i.e unique identifier).
+  // Returns -1 on failure else 0.
+  virtual int32_t SetCNAME(absl::string_view cname) = 0;
+
+  // Returns remote NTP.
+  // Returns -1 on failure else 0.
+  virtual int32_t RemoteNTP(uint32_t* received_ntp_secs,
+                            uint32_t* received_ntp_frac,
+                            uint32_t* rtcp_arrival_time_secs,
+                            uint32_t* rtcp_arrival_time_frac,
+                            uint32_t* rtcp_timestamp) const = 0;
+
+  // Returns current RTT (round-trip time) estimate.
+  // Returns -1 on failure else 0.
+  virtual int32_t RTT(uint32_t remote_ssrc,
+                      int64_t* rtt,
+                      int64_t* avg_rtt,
+                      int64_t* min_rtt,
+                      int64_t* max_rtt) const = 0;
+
+  // Returns the estimated RTT, with fallback to a default value.
+  virtual int64_t ExpectedRetransmissionTimeMs() const = 0;
+
+  // Forces a send of a RTCP packet. Periodic SR and RR are triggered via the
+  // process function.
+  // Returns -1 on failure else 0.
+  virtual int32_t SendRTCP(RTCPPacketType rtcp_packet_type) = 0;
+
+  // Returns send statistics for the RTP and RTX stream.
+  virtual void GetSendStreamDataCounters(
+      StreamDataCounters* rtp_counters,
+      StreamDataCounters* rtx_counters) const = 0;
+
+
+  // Returns packet count, octet count, and timestamps from RTCP sender report.
+  virtual void RemoteRTCPSenderInfo(uint32_t* packet_count,
+                                    uint32_t* octet_count,
+                                    int64_t* ntp_timestamp_ms,
+                                    int64_t* remote_ntp_timestamp_ms) const = 0;
+  // A snapshot of Report Blocks with additional data of interest to statistics.
+  // Within this list, the sender-source SSRC pair is unique and per-pair the
+  // ReportBlockData represents the latest Report Block that was received for
+  // that pair.
+  virtual std::vector<ReportBlockData> GetLatestReportBlockData() const = 0;
+  // Returns stats based on the received RTCP SRs.
+  virtual absl::optional<SenderReportStats> GetSenderReportStats() const = 0;
+  // Returns non-sender RTT stats, based on DLRR.
+  virtual absl::optional<NonSenderRttStats> GetNonSenderRttStats() const = 0;
+
+  // (REMB) Receiver Estimated Max Bitrate.
+  // Schedules sending REMB on next and following sender/receiver reports.
+  void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override = 0;
+  // Stops sending REMB on next and following sender/receiver reports.
+  void UnsetRemb() override = 0;
+
+  // (NACK)
+
+  // Sends a Negative acknowledgement packet.
+  // Returns -1 on failure else 0.
+  // TODO(philipel): Deprecate this and start using SendNack instead, mostly
+  // because we want a function that actually send NACK for the specified
+  // packets.
+  virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) = 0;
+
+  // Sends NACK for the packets specified.
+  // Note: This assumes the caller keeps track of timing and doesn't rely on
+  // the RTP module to do this.
+  virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0;
+
+  // Store the sent packets, needed to answer to a Negative acknowledgment
+  // requests.
+  virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
+
+  virtual void SetVideoBitrateAllocation(
+      const VideoBitrateAllocation& bitrate) = 0;
+
+  // **************************************************************************
+  // Video
+  // **************************************************************************
+
+  // Requests new key frame.
+  // using PLI, https://tools.ietf.org/html/rfc4585#section-6.3.1.1
+  void SendPictureLossIndication() { SendRTCP(kRtcpPli); }
+  // using FIR, https://tools.ietf.org/html/rfc5104#section-4.3.1.2
+  void SendFullIntraRequest() { SendRTCP(kRtcpFir); }
+
+  // Sends a LossNotification RTCP message.
+  // Returns -1 on failure else 0.
+  virtual int32_t SendLossNotification(uint16_t last_decoded_seq_num,
+                                       uint16_t last_received_seq_num,
+                                       bool decodability_flag,
+                                       bool buffering_allowed) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc
new file mode 100644
index 0000000000..d5e8bdcccb
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -0,0 +1,834 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sender.h"
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
+#include "modules/rtp_rtcp/include/rtp_cvo.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/rate_limiter.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
+constexpr size_t kMaxPaddingLength = 224;
+constexpr size_t kMinAudioPaddingLength = 50;
+constexpr size_t kRtpHeaderLength = 12;
+
+// Min size needed to get payload padding from packet history.
+constexpr int kMinPayloadPaddingBytes = 50;
+
+// Determines how much larger a payload padding packet may be, compared to the
+// requested padding size.
+constexpr double kMaxPaddingSizeFactor = 3.0;
+
+template <typename Extension>
+constexpr RtpExtensionSize CreateExtensionSize() {
+  return {Extension::kId, Extension::kValueSizeBytes};
+}
+
+template <typename Extension>
+constexpr RtpExtensionSize CreateMaxExtensionSize() {
+  return {Extension::kId, Extension::kMaxValueSizeBytes};
+}
+
+// Size info for header extensions that might be used in padding or FEC packets.
+constexpr RtpExtensionSize kFecOrPaddingExtensionSizes[] = {
+    CreateExtensionSize<AbsoluteSendTime>(),
+    CreateExtensionSize<TransmissionOffset>(),
+    CreateExtensionSize<TransportSequenceNumber>(),
+    CreateExtensionSize<PlayoutDelayLimits>(),
+    CreateMaxExtensionSize<RtpMid>(),
+    CreateExtensionSize<VideoTimingExtension>(),
+};
+
+// Size info for header extensions that might be used in video packets.
+constexpr RtpExtensionSize kVideoExtensionSizes[] = {
+    CreateExtensionSize<AbsoluteSendTime>(),
+    CreateExtensionSize<AbsoluteCaptureTimeExtension>(),
+    CreateExtensionSize<TransmissionOffset>(),
+    CreateExtensionSize<TransportSequenceNumber>(),
+    CreateExtensionSize<PlayoutDelayLimits>(),
+    CreateExtensionSize<VideoOrientation>(),
+    CreateExtensionSize<VideoContentTypeExtension>(),
+    CreateExtensionSize<VideoTimingExtension>(),
+    CreateMaxExtensionSize<RtpStreamId>(),
+    CreateMaxExtensionSize<RepairedRtpStreamId>(),
+    CreateMaxExtensionSize<RtpMid>(),
+    {RtpGenericFrameDescriptorExtension00::kId,
+     RtpGenericFrameDescriptorExtension00::kMaxSizeBytes},
+};
+
+// Size info for header extensions that might be used in audio packets.
+constexpr RtpExtensionSize kAudioExtensionSizes[] = {
+    CreateExtensionSize<AbsoluteSendTime>(),
+    CreateExtensionSize<AbsoluteCaptureTimeExtension>(),
+    CreateExtensionSize<AudioLevel>(),
+    CreateExtensionSize<InbandComfortNoiseExtension>(),
+    CreateExtensionSize<TransmissionOffset>(),
+    CreateExtensionSize<TransportSequenceNumber>(),
+    CreateMaxExtensionSize<RtpStreamId>(),
+    CreateMaxExtensionSize<RepairedRtpStreamId>(),
+    CreateMaxExtensionSize<RtpMid>(),
+};
+
+// Non-volatile extensions can be expected on all packets, if registered.
+// Volatile ones, such as VideoContentTypeExtension which is only set on
+// key-frames, are removed to simplify overhead calculations at the expense of
+// some accuracy.
+bool IsNonVolatile(RTPExtensionType type) {
+  switch (type) {
+    case kRtpExtensionTransmissionTimeOffset:
+    case kRtpExtensionAudioLevel:
+#if !defined(WEBRTC_MOZILLA_BUILD)
+    case kRtpExtensionCsrcAudioLevel:
+#endif
+    case kRtpExtensionAbsoluteSendTime:
+    case kRtpExtensionTransportSequenceNumber:
+    case kRtpExtensionTransportSequenceNumber02:
+    case kRtpExtensionRtpStreamId:
+    case kRtpExtensionRepairedRtpStreamId:
+    case kRtpExtensionMid:
+    case kRtpExtensionGenericFrameDescriptor:
+    case kRtpExtensionDependencyDescriptor:
+      return true;
+    case kRtpExtensionInbandComfortNoise:
+    case kRtpExtensionAbsoluteCaptureTime:
+    case kRtpExtensionVideoRotation:
+    case kRtpExtensionPlayoutDelay:
+    case kRtpExtensionVideoContentType:
+    case kRtpExtensionVideoLayersAllocation:
+    case kRtpExtensionVideoTiming:
+    case kRtpExtensionColorSpace:
+    case kRtpExtensionVideoFrameTrackingId:
+      return false;
+    case kRtpExtensionNone:
+    case kRtpExtensionNumberOfExtensions:
+      RTC_DCHECK_NOTREACHED();
+      return false;
+#if defined(WEBRTC_MOZILLA_BUILD)
+    case kRtpExtensionCsrcAudioLevel:
+      // TODO: Mozilla implement for CsrcAudioLevel
+      RTC_CHECK(false);
+      return false;
+#endif
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+bool HasBweExtension(const RtpHeaderExtensionMap& extensions_map) {
+  return extensions_map.IsRegistered(kRtpExtensionTransportSequenceNumber) ||
+         extensions_map.IsRegistered(kRtpExtensionTransportSequenceNumber02) ||
+         extensions_map.IsRegistered(kRtpExtensionAbsoluteSendTime) ||
+         extensions_map.IsRegistered(kRtpExtensionTransmissionTimeOffset);
+}
+
+}  // namespace
+
+RTPSender::RTPSender(const RtpRtcpInterface::Configuration& config,
+                     RtpPacketHistory* packet_history,
+                     RtpPacketSender* packet_sender)
+    : clock_(config.clock),
+      random_(clock_->TimeInMicroseconds()),
+      audio_configured_(config.audio),
+      ssrc_(config.local_media_ssrc),
+      rtx_ssrc_(config.rtx_send_ssrc),
+      flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc()
+                                         : absl::nullopt),
+      packet_history_(packet_history),
+      paced_sender_(packet_sender),
+      sending_media_(true),                   // Default to sending media.
+      max_packet_size_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
+      rtp_header_extension_map_(config.extmap_allow_mixed),
+      // RTP variables
+      rid_(config.rid),
+      always_send_mid_and_rid_(config.always_send_mid_and_rid),
+      ssrc_has_acked_(false),
+      rtx_ssrc_has_acked_(false),
+      csrcs_(),
+      rtx_(kRtxOff),
+      supports_bwe_extension_(false),
+      retransmission_rate_limiter_(config.retransmission_rate_limiter) {
+  // This random initialization is not intended to be cryptographic strong.
+  timestamp_offset_ = random_.Rand<uint32_t>();
+
+  RTC_DCHECK(paced_sender_);
+  RTC_DCHECK(packet_history_);
+  RTC_DCHECK_LE(rid_.size(), RtpStreamId::kMaxValueSizeBytes);
+
+  UpdateHeaderSizes();
+}
+
+RTPSender::~RTPSender() {
+  // TODO(tommi): Use a thread checker to ensure the object is created and
+  // deleted on the same thread.  At the moment this isn't possible due to
+  // voe::ChannelOwner in voice engine.  To reproduce, run:
+  // voe_auto_test --automated --gtest_filter=*MixManyChannelsForStressOpus
+
+  // TODO(tommi,holmer): We don't grab locks in the dtor before accessing member
+  // variables but we grab them in all other methods. (what's the design?)
+  // Start documenting what thread we're on in what method so that it's easier
+  // to understand performance attributes and possibly remove locks.
+}
+
+rtc::ArrayView<const RtpExtensionSize> RTPSender::FecExtensionSizes() {
+  return rtc::MakeArrayView(kFecOrPaddingExtensionSizes,
+                            arraysize(kFecOrPaddingExtensionSizes));
+}
+
+rtc::ArrayView<const RtpExtensionSize> RTPSender::VideoExtensionSizes() {
+  return rtc::MakeArrayView(kVideoExtensionSizes,
+                            arraysize(kVideoExtensionSizes));
+}
+
+rtc::ArrayView<const RtpExtensionSize> RTPSender::AudioExtensionSizes() {
+  return rtc::MakeArrayView(kAudioExtensionSizes,
+                            arraysize(kAudioExtensionSizes));
+}
+
+void RTPSender::SetExtmapAllowMixed(bool extmap_allow_mixed) {
+  MutexLock lock(&send_mutex_);
+  rtp_header_extension_map_.SetExtmapAllowMixed(extmap_allow_mixed);
+}
+
+bool RTPSender::RegisterRtpHeaderExtension(absl::string_view uri, int id) {
+  MutexLock lock(&send_mutex_);
+  bool registered = rtp_header_extension_map_.RegisterByUri(id, uri);
+  supports_bwe_extension_ = HasBweExtension(rtp_header_extension_map_);
+  UpdateHeaderSizes();
+  return registered;
+}
+
+bool RTPSender::IsRtpHeaderExtensionRegistered(RTPExtensionType type) const {
+  MutexLock lock(&send_mutex_);
+  return rtp_header_extension_map_.IsRegistered(type);
+}
+
+void RTPSender::DeregisterRtpHeaderExtension(absl::string_view uri) {
+  MutexLock lock(&send_mutex_);
+  rtp_header_extension_map_.Deregister(uri);
+  supports_bwe_extension_ = HasBweExtension(rtp_header_extension_map_);
+  UpdateHeaderSizes();
+}
+
+void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
+  RTC_DCHECK_GE(max_packet_size, 100);
+  RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
+  MutexLock lock(&send_mutex_);
+  max_packet_size_ = max_packet_size;
+}
+
+size_t RTPSender::MaxRtpPacketSize() const {
+  return max_packet_size_;
+}
+
+void RTPSender::SetRtxStatus(int mode) {
+  MutexLock lock(&send_mutex_);
+  if (mode != kRtxOff &&
+      (!rtx_ssrc_.has_value() || rtx_payload_type_map_.empty())) {
+    RTC_LOG(LS_ERROR)
+        << "Failed to enable RTX without RTX SSRC or payload types.";
+    return;
+  }
+  rtx_ = mode;
+}
+
+int RTPSender::RtxStatus() const {
+  MutexLock lock(&send_mutex_);
+  return rtx_;
+}
+
+void RTPSender::SetRtxPayloadType(int payload_type,
+                                  int associated_payload_type) {
+  MutexLock lock(&send_mutex_);
+  RTC_DCHECK_LE(payload_type, 127);
+  RTC_DCHECK_LE(associated_payload_type, 127);
+  if (payload_type < 0) {
+    RTC_LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type << ".";
+    return;
+  }
+
+  rtx_payload_type_map_[associated_payload_type] = payload_type;
+}
+
+int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
+  int32_t packet_size = 0;
+  const bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
+
+  std::unique_ptr<RtpPacketToSend> packet =
+      packet_history_->GetPacketAndMarkAsPending(
+          packet_id, [&](const RtpPacketToSend& stored_packet) {
+            // Check if we're overusing retransmission bitrate.
+            // TODO(sprang): Add histograms for nack success or failure
+            // reasons.
+            packet_size = stored_packet.size();
+            std::unique_ptr<RtpPacketToSend> retransmit_packet;
+            if (retransmission_rate_limiter_ &&
+                !retransmission_rate_limiter_->TryUseRate(packet_size)) {
+              return retransmit_packet;
+            }
+            if (rtx) {
+              retransmit_packet = BuildRtxPacket(stored_packet);
+            } else {
+              retransmit_packet =
+                  std::make_unique<RtpPacketToSend>(stored_packet);
+            }
+            if (retransmit_packet) {
+              retransmit_packet->set_retransmitted_sequence_number(
+                  stored_packet.SequenceNumber());
+            }
+            return retransmit_packet;
+          });
+  if (packet_size == 0) {
+    // Packet not found or already queued for retransmission, ignore.
+    RTC_DCHECK(!packet);
+    return 0;
+  }
+  if (!packet) {
+    // Packet was found, but lambda helper above chose not to create
+    // `retransmit_packet` out of it.
+    return -1;
+  }
+  packet->set_packet_type(RtpPacketMediaType::kRetransmission);
+  packet->set_fec_protect_packet(false);
+  std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+  packets.emplace_back(std::move(packet));
+  paced_sender_->EnqueuePackets(std::move(packets));
+
+  return packet_size;
+}
+
+void RTPSender::OnReceivedAckOnSsrc(int64_t extended_highest_sequence_number) {
+  MutexLock lock(&send_mutex_);
+  bool update_required = !ssrc_has_acked_;
+  ssrc_has_acked_ = true;
+  if (update_required) {
+    UpdateHeaderSizes();
+  }
+}
+
+void RTPSender::OnReceivedAckOnRtxSsrc(
+    int64_t extended_highest_sequence_number) {
+  MutexLock lock(&send_mutex_);
+  bool update_required = !rtx_ssrc_has_acked_;
+  rtx_ssrc_has_acked_ = true;
+  if (update_required) {
+    UpdateHeaderSizes();
+  }
+}
+
+void RTPSender::OnReceivedNack(
+    const std::vector<uint16_t>& nack_sequence_numbers,
+    int64_t avg_rtt) {
+  packet_history_->SetRtt(TimeDelta::Millis(5 + avg_rtt));
+  for (uint16_t seq_no : nack_sequence_numbers) {
+    const int32_t bytes_sent = ReSendPacket(seq_no);
+    if (bytes_sent < 0) {
+      // Failed to send one Sequence number. Give up the rest in this nack.
+      RTC_LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
+                          << ", Discard rest of packets.";
+      break;
+    }
+  }
+}
+
+bool RTPSender::SupportsPadding() const {
+  MutexLock lock(&send_mutex_);
+  return sending_media_ && supports_bwe_extension_;
+}
+
+bool RTPSender::SupportsRtxPayloadPadding() const {
+  MutexLock lock(&send_mutex_);
+  return sending_media_ && supports_bwe_extension_ &&
+         (rtx_ & kRtxRedundantPayloads);
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> RTPSender::GeneratePadding(
+    size_t target_size_bytes,
+    bool media_has_been_sent,
+    bool can_send_padding_on_media_ssrc) {
+  // This method does not actually send packets, it just generates
+  // them and puts them in the pacer queue. Since this should incur
+  // low overhead, keep the lock for the scope of the method in order
+  // to make the code more readable.
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
+  size_t bytes_left = target_size_bytes;
+  if (SupportsRtxPayloadPadding()) {
+    while (bytes_left >= kMinPayloadPaddingBytes) {
+      std::unique_ptr<RtpPacketToSend> packet =
+          packet_history_->GetPayloadPaddingPacket(
+              [&](const RtpPacketToSend& packet)
+                  -> std::unique_ptr<RtpPacketToSend> {
+                // Limit overshoot, generate <= `kMaxPaddingSizeFactor` *
+                // `target_size_bytes`.
+                const size_t max_overshoot_bytes = static_cast<size_t>(
+                    ((kMaxPaddingSizeFactor - 1.0) * target_size_bytes) + 0.5);
+                if (packet.payload_size() + kRtxHeaderSize >
+                    max_overshoot_bytes + bytes_left) {
+                  return nullptr;
+                }
+                return BuildRtxPacket(packet);
+              });
+      if (!packet) {
+        break;
+      }
+
+      bytes_left -= std::min(bytes_left, packet->payload_size());
+      packet->set_packet_type(RtpPacketMediaType::kPadding);
+      padding_packets.push_back(std::move(packet));
+    }
+  }
+
+  MutexLock lock(&send_mutex_);
+  if (!sending_media_) {
+    return {};
+  }
+
+  size_t padding_bytes_in_packet;
+  const size_t max_payload_size =
+      max_packet_size_ - max_padding_fec_packet_header_;
+  if (audio_configured_) {
+    // Allow smaller padding packets for audio.
+    padding_bytes_in_packet = rtc::SafeClamp<size_t>(
+        bytes_left, kMinAudioPaddingLength,
+        rtc::SafeMin(max_payload_size, kMaxPaddingLength));
+  } else {
+    // Always send full padding packets. This is accounted for by the
+    // RtpPacketSender, which will make sure we don't send too much padding even
+    // if a single packet is larger than requested.
+    // We do this to avoid frequently sending small packets on higher bitrates.
+    padding_bytes_in_packet = rtc::SafeMin(max_payload_size, kMaxPaddingLength);
+  }
+
+  while (bytes_left > 0) {
+    auto padding_packet =
+        std::make_unique<RtpPacketToSend>(&rtp_header_extension_map_);
+    padding_packet->set_packet_type(RtpPacketMediaType::kPadding);
+    padding_packet->SetMarker(false);
+    if (rtx_ == kRtxOff) {
+      if (!can_send_padding_on_media_ssrc) {
+        break;
+      }
+      padding_packet->SetSsrc(ssrc_);
+    } else {
+      // Without abs-send-time or transport sequence number a media packet
+      // must be sent before padding so that the timestamps used for
+      // estimation are correct.
+      if (!media_has_been_sent &&
+          !(rtp_header_extension_map_.IsRegistered(AbsoluteSendTime::kId) ||
+            rtp_header_extension_map_.IsRegistered(
+                TransportSequenceNumber::kId))) {
+        break;
+      }
+
+      RTC_DCHECK(rtx_ssrc_);
+      RTC_DCHECK(!rtx_payload_type_map_.empty());
+      padding_packet->SetSsrc(*rtx_ssrc_);
+      padding_packet->SetPayloadType(rtx_payload_type_map_.begin()->second);
+    }
+
+    if (rtp_header_extension_map_.IsRegistered(TransportSequenceNumber::kId)) {
+      padding_packet->ReserveExtension<TransportSequenceNumber>();
+    }
+    if (rtp_header_extension_map_.IsRegistered(TransmissionOffset::kId)) {
+      padding_packet->ReserveExtension<TransmissionOffset>();
+    }
+    if (rtp_header_extension_map_.IsRegistered(AbsoluteSendTime::kId)) {
+      padding_packet->ReserveExtension<AbsoluteSendTime>();
+    }
+
+    padding_packet->SetPadding(padding_bytes_in_packet);
+    bytes_left -= std::min(bytes_left, padding_bytes_in_packet);
+    padding_packets.push_back(std::move(padding_packet));
+  }
+
+  return padding_packets;
+}
+
+bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet) {
+  RTC_DCHECK(packet);
+  auto packet_type = packet->packet_type();
+  RTC_CHECK(packet_type) << "Packet type must be set before sending.";
+
+  if (packet->capture_time() <= Timestamp::Zero()) {
+    packet->set_capture_time(clock_->CurrentTime());
+  }
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+  packets.emplace_back(std::move(packet));
+  paced_sender_->EnqueuePackets(std::move(packets));
+
+  return true;
+}
+
+void RTPSender::EnqueuePackets(
+    std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+  RTC_DCHECK(!packets.empty());
+  Timestamp now = clock_->CurrentTime();
+  for (auto& packet : packets) {
+    RTC_DCHECK(packet);
+    RTC_CHECK(packet->packet_type().has_value())
+        << "Packet type must be set before sending.";
+    if (packet->capture_time() <= Timestamp::Zero()) {
+      packet->set_capture_time(now);
+    }
+  }
+
+  paced_sender_->EnqueuePackets(std::move(packets));
+}
+
+size_t RTPSender::FecOrPaddingPacketMaxRtpHeaderLength() const {
+  MutexLock lock(&send_mutex_);
+  return max_padding_fec_packet_header_;
+}
+
+size_t RTPSender::ExpectedPerPacketOverhead() const {
+  MutexLock lock(&send_mutex_);
+  return max_media_packet_header_;
+}
+
+std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
+  MutexLock lock(&send_mutex_);
+  // TODO(danilchap): Find better motivator and value for extra capacity.
+  // RtpPacketizer might slightly miscalulate needed size,
+  // SRTP may benefit from extra space in the buffer and do encryption in place
+  // saving reallocation.
+  // While sending slightly oversized packet increase chance of dropped packet,
+  // it is better than crash on drop packet without trying to send it.
+  static constexpr int kExtraCapacity = 16;
+  auto packet = std::make_unique<RtpPacketToSend>(
+      &rtp_header_extension_map_, max_packet_size_ + kExtraCapacity);
+  packet->SetSsrc(ssrc_);
+  packet->SetCsrcs(csrcs_);
+  // Reserve extensions, if registered, RtpSender set in SendToNetwork.
+  packet->ReserveExtension<AbsoluteSendTime>();
+  packet->ReserveExtension<TransmissionOffset>();
+  packet->ReserveExtension<TransportSequenceNumber>();
+
+  // BUNDLE requires that the receiver "bind" the received SSRC to the values
+  // in the MID and/or (R)RID header extensions if present. Therefore, the
+  // sender can reduce overhead by omitting these header extensions once it
+  // knows that the receiver has "bound" the SSRC.
+  // This optimization can be configured by setting
+  // `always_send_mid_and_rid_` appropriately.
+  //
+  // The algorithm here is fairly simple: Always attach a MID and/or RID (if
+  // configured) to the outgoing packets until an RTCP receiver report comes
+  // back for this SSRC. That feedback indicates the receiver must have
+  // received a packet with the SSRC and header extension(s), so the sender
+  // then stops attaching the MID and RID.
+  if (always_send_mid_and_rid_ || !ssrc_has_acked_) {
+    // These are no-ops if the corresponding header extension is not registered.
+    if (!mid_.empty()) {
+      packet->SetExtension<RtpMid>(mid_);
+    }
+    if (!rid_.empty()) {
+      packet->SetExtension<RtpStreamId>(rid_);
+    }
+  }
+  return packet;
+}
+
+size_t RTPSender::RtxPacketOverhead() const {
+  MutexLock lock(&send_mutex_);
+  if (rtx_ == kRtxOff) {
+    return 0;
+  }
+  size_t overhead = 0;
+
+  // Count space for the RTP header extensions that might need to be added to
+  // the RTX packet.
+  if (!always_send_mid_and_rid_ && (!rtx_ssrc_has_acked_ && ssrc_has_acked_)) {
+    // Prefer to reserve extra byte in case two byte header rtp header
+    // extensions are used.
+    static constexpr int kRtpExtensionHeaderSize = 2;
+
+    // Rtx packets hasn't been acked and would need to have mid and rrsid rtp
+    // header extensions, while media packets no longer needs to include mid and
+    // rsid extensions.
+    if (!mid_.empty()) {
+      overhead += (kRtpExtensionHeaderSize + mid_.size());
+    }
+    if (!rid_.empty()) {
+      overhead += (kRtpExtensionHeaderSize + rid_.size());
+    }
+    // RTP header extensions are rounded up to 4 bytes. Depending on already
+    // present extensions adding mid & rrsid may add up to 3 bytes of padding.
+    overhead += 3;
+  }
+
+  // Add two bytes for the original sequence number in the RTP payload.
+  overhead += kRtxHeaderSize;
+  return overhead;
+}
+
+void RTPSender::SetSendingMediaStatus(bool enabled) {
+  MutexLock lock(&send_mutex_);
+  sending_media_ = enabled;
+}
+
+bool RTPSender::SendingMedia() const {
+  MutexLock lock(&send_mutex_);
+  return sending_media_;
+}
+
+bool RTPSender::IsAudioConfigured() const {
+  return audio_configured_;
+}
+
+void RTPSender::SetTimestampOffset(uint32_t timestamp) {
+  MutexLock lock(&send_mutex_);
+  timestamp_offset_ = timestamp;
+}
+
+uint32_t RTPSender::TimestampOffset() const {
+  MutexLock lock(&send_mutex_);
+  return timestamp_offset_;
+}
+
+void RTPSender::SetMid(absl::string_view mid) {
+  // This is configured via the API.
+  MutexLock lock(&send_mutex_);
+  RTC_DCHECK_LE(mid.length(), RtpMid::kMaxValueSizeBytes);
+  mid_ = std::string(mid);
+  UpdateHeaderSizes();
+}
+
+void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
+  RTC_DCHECK_LE(csrcs.size(), kRtpCsrcSize);
+  MutexLock lock(&send_mutex_);
+  csrcs_ = csrcs;
+  UpdateHeaderSizes();
+}
+
+static void CopyHeaderAndExtensionsToRtxPacket(const RtpPacketToSend& packet,
+                                               RtpPacketToSend* rtx_packet) {
+  // Set the relevant fixed packet headers. The following are not set:
+  // * Payload type - it is replaced in rtx packets.
+  // * Sequence number - RTX has a separate sequence numbering.
+  // * SSRC - RTX stream has its own SSRC.
+  rtx_packet->SetMarker(packet.Marker());
+  rtx_packet->SetTimestamp(packet.Timestamp());
+
+  // Set the variable fields in the packet header:
+  // * CSRCs - must be set before header extensions.
+  // * Header extensions - replace Rid header with RepairedRid header.
+  const std::vector<uint32_t> csrcs = packet.Csrcs();
+  rtx_packet->SetCsrcs(csrcs);
+  for (int extension_num = kRtpExtensionNone + 1;
+       extension_num < kRtpExtensionNumberOfExtensions; ++extension_num) {
+    auto extension = static_cast<RTPExtensionType>(extension_num);
+
+    // Stream ID header extensions (MID, RSID) are sent per-SSRC. Since RTX
+    // operates on a different SSRC, the presence and values of these header
+    // extensions should be determined separately and not blindly copied.
+    if (extension == kRtpExtensionMid ||
+        extension == kRtpExtensionRtpStreamId) {
+      continue;
+    }
+
+    // Empty extensions should be supported, so not checking `source.empty()`.
+    if (!packet.HasExtension(extension)) {
+      continue;
+    }
+
+    rtc::ArrayView<const uint8_t> source = packet.FindExtension(extension);
+
+    rtc::ArrayView<uint8_t> destination =
+        rtx_packet->AllocateExtension(extension, source.size());
+
+    // Could happen if any:
+    // 1. Extension has 0 length.
+    // 2. Extension is not registered in destination.
+    // 3. Allocating extension in destination failed.
+    if (destination.empty() || source.size() != destination.size()) {
+      continue;
+    }
+
+    std::memcpy(destination.begin(), source.begin(), destination.size());
+  }
+}
+
+std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
+    const RtpPacketToSend& packet) {
+  std::unique_ptr<RtpPacketToSend> rtx_packet;
+
+  // Add original RTP header.
+  {
+    MutexLock lock(&send_mutex_);
+    if (!sending_media_)
+      return nullptr;
+
+    RTC_DCHECK(rtx_ssrc_);
+
+    // Replace payload type.
+    auto kv = rtx_payload_type_map_.find(packet.PayloadType());
+    if (kv == rtx_payload_type_map_.end())
+      return nullptr;
+
+    rtx_packet = std::make_unique<RtpPacketToSend>(&rtp_header_extension_map_,
+                                                   max_packet_size_);
+
+    rtx_packet->SetPayloadType(kv->second);
+
+    // Replace SSRC.
+    rtx_packet->SetSsrc(*rtx_ssrc_);
+
+    CopyHeaderAndExtensionsToRtxPacket(packet, rtx_packet.get());
+
+    // RTX packets are sent on an SSRC different from the main media, so the
+    // decision to attach MID and/or RRID header extensions is completely
+    // separate from that of the main media SSRC.
+    //
+    // Note that RTX packets must used the RepairedRtpStreamId (RRID) header
+    // extension instead of the RtpStreamId (RID) header extension even though
+    // the payload is identical.
+    if (always_send_mid_and_rid_ || !rtx_ssrc_has_acked_) {
+      // These are no-ops if the corresponding header extension is not
+      // registered.
+      if (!mid_.empty()) {
+        rtx_packet->SetExtension<RtpMid>(mid_);
+      }
+      if (!rid_.empty()) {
+        rtx_packet->SetExtension<RepairedRtpStreamId>(rid_);
+      }
+    }
+  }
+  RTC_DCHECK(rtx_packet);
+
+  uint8_t* rtx_payload =
+      rtx_packet->AllocatePayload(packet.payload_size() + kRtxHeaderSize);
+  if (rtx_payload == nullptr)
+    return nullptr;
+
+  // Add OSN (original sequence number).
+  ByteWriter<uint16_t>::WriteBigEndian(rtx_payload, packet.SequenceNumber());
+
+  // Add original payload data.
+  auto payload = packet.payload();
+  if (!payload.empty()) {
+    memcpy(rtx_payload + kRtxHeaderSize, payload.data(), payload.size());
+  }
+
+  // Add original additional data.
+  rtx_packet->set_additional_data(packet.additional_data());
+
+  // Copy capture time so e.g. TransmissionOffset is correctly set.
+  rtx_packet->set_capture_time(packet.capture_time());
+
+  return rtx_packet;
+}
+
+void RTPSender::SetRtpState(const RtpState& rtp_state) {
+  MutexLock lock(&send_mutex_);
+
+  timestamp_offset_ = rtp_state.start_timestamp;
+  ssrc_has_acked_ = rtp_state.ssrc_has_acked;
+  UpdateHeaderSizes();
+}
+
+RtpState RTPSender::GetRtpState() const {
+  MutexLock lock(&send_mutex_);
+
+  RtpState state;
+  state.start_timestamp = timestamp_offset_;
+  state.ssrc_has_acked = ssrc_has_acked_;
+  return state;
+}
+
+void RTPSender::SetRtxRtpState(const RtpState& rtp_state) {
+  MutexLock lock(&send_mutex_);
+  rtx_ssrc_has_acked_ = rtp_state.ssrc_has_acked;
+}
+
+RtpState RTPSender::GetRtxRtpState() const {
+  MutexLock lock(&send_mutex_);
+
+  RtpState state;
+  state.start_timestamp = timestamp_offset_;
+  state.ssrc_has_acked = rtx_ssrc_has_acked_;
+
+  return state;
+}
+
+void RTPSender::UpdateHeaderSizes() {
+  const size_t rtp_header_length =
+      kRtpHeaderLength + sizeof(uint32_t) * csrcs_.size();
+
+  max_padding_fec_packet_header_ =
+      rtp_header_length + RtpHeaderExtensionSize(kFecOrPaddingExtensionSizes,
+                                                 rtp_header_extension_map_);
+
+  // RtpStreamId, Mid and RepairedRtpStreamId are treated specially in that
+  // we check if they currently are being sent. RepairedRtpStreamId can be
+  // sent instead of RtpStreamID on RTX packets and may share the same space.
+  // When the primary SSRC has already been acked but the RTX SSRC has not
+  // yet been acked, RepairedRtpStreamId needs to be taken into account
+  // separately.
+  const bool send_mid_rid_on_rtx =
+      rtx_ssrc_.has_value() &&
+      (always_send_mid_and_rid_ || !rtx_ssrc_has_acked_);
+  const bool send_mid_rid = always_send_mid_and_rid_ || !ssrc_has_acked_;
+  std::vector<RtpExtensionSize> non_volatile_extensions;
+  for (auto& extension :
+       audio_configured_ ? AudioExtensionSizes() : VideoExtensionSizes()) {
+    if (IsNonVolatile(extension.type)) {
+      switch (extension.type) {
+        case RTPExtensionType::kRtpExtensionMid:
+          if ((send_mid_rid || send_mid_rid_on_rtx) && !mid_.empty()) {
+            non_volatile_extensions.push_back(extension);
+          }
+          break;
+        case RTPExtensionType::kRtpExtensionRtpStreamId:
+          if (send_mid_rid && !rid_.empty()) {
+            non_volatile_extensions.push_back(extension);
+          }
+          break;
+        case RTPExtensionType::kRtpExtensionRepairedRtpStreamId:
+          if (send_mid_rid_on_rtx && !send_mid_rid && !rid_.empty()) {
+            non_volatile_extensions.push_back(extension);
+          }
+          break;
+        default:
+          non_volatile_extensions.push_back(extension);
+      }
+    }
+  }
+  max_media_packet_header_ =
+      rtp_header_length + RtpHeaderExtensionSize(non_volatile_extensions,
+                                                 rtp_header_extension_map_);
+  // Reserve extra bytes if packet might be resent in an rtx packet.
+  if (rtx_ssrc_.has_value()) {
+    max_media_packet_header_ += kRtxHeaderSize;
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.h
new file mode 100644
index 0000000000..b49afe0dec
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -0,0 +1,217 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/call/transport.h"
+#include "api/field_trials_view.h"
+#include "modules/rtp_rtcp/include/flexfec_sender.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/random.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class FrameEncryptorInterface;
+class RateLimiter;
+class RtcEventLog;
+class RtpPacketToSend;
+
+class RTPSender {
+ public:
+  RTPSender(const RtpRtcpInterface::Configuration& config,
+            RtpPacketHistory* packet_history,
+            RtpPacketSender* packet_sender);
+  RTPSender(const RTPSender&) = delete;
+  RTPSender& operator=(const RTPSender&) = delete;
+
+  ~RTPSender();
+
+  void SetSendingMediaStatus(bool enabled) RTC_LOCKS_EXCLUDED(send_mutex_);
+  bool SendingMedia() const RTC_LOCKS_EXCLUDED(send_mutex_);
+  bool IsAudioConfigured() const RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  uint32_t TimestampOffset() const RTC_LOCKS_EXCLUDED(send_mutex_);
+  void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  void SetMid(absl::string_view mid) RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  uint16_t SequenceNumber() const RTC_LOCKS_EXCLUDED(send_mutex_);
+  void SetSequenceNumber(uint16_t seq) RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  void SetCsrcs(const std::vector<uint32_t>& csrcs)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  void SetMaxRtpPacketSize(size_t max_packet_size)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  void SetExtmapAllowMixed(bool extmap_allow_mixed)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  // RTP header extension
+  bool RegisterRtpHeaderExtension(absl::string_view uri, int id)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+  bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) const
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+  void DeregisterRtpHeaderExtension(absl::string_view uri)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  bool SupportsPadding() const RTC_LOCKS_EXCLUDED(send_mutex_);
+  bool SupportsRtxPayloadPadding() const RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      size_t target_size_bytes,
+      bool media_has_been_sent,
+      bool can_send_padding_on_media_ssrc) RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  // NACK.
+  void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
+                      int64_t avg_rtt) RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  int32_t ReSendPacket(uint16_t packet_id) RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  // ACK.
+  void OnReceivedAckOnSsrc(int64_t extended_highest_sequence_number)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+  void OnReceivedAckOnRtxSsrc(int64_t extended_highest_sequence_number)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  // RTX.
+  void SetRtxStatus(int mode) RTC_LOCKS_EXCLUDED(send_mutex_);
+  int RtxStatus() const RTC_LOCKS_EXCLUDED(send_mutex_);
+  absl::optional<uint32_t> RtxSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
+    return rtx_ssrc_;
+  }
+  // Returns expected size difference between an RTX packet and media packet
+  // that RTX packet is created from. Returns 0 if RTX is disabled.
+  size_t RtxPacketOverhead() const;
+
+  void SetRtxPayloadType(int payload_type, int associated_payload_type)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  // Size info for header extensions used by FEC packets.
+  static rtc::ArrayView<const RtpExtensionSize> FecExtensionSizes()
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  // Size info for header extensions used by video packets.
+  static rtc::ArrayView<const RtpExtensionSize> VideoExtensionSizes()
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  // Size info for header extensions used by audio packets.
+  static rtc::ArrayView<const RtpExtensionSize> AudioExtensionSizes()
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  // Create empty packet, fills ssrc, csrcs and reserve place for header
+  // extensions RtpSender updates before sending.
+  std::unique_ptr<RtpPacketToSend> AllocatePacket() const
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+  // Maximum header overhead per fec/padding packet.
+  size_t FecOrPaddingPacketMaxRtpHeaderLength() const
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+  // Expected header overhead per media packet.
+  size_t ExpectedPerPacketOverhead() const RTC_LOCKS_EXCLUDED(send_mutex_);
+  // Including RTP headers.
+  size_t MaxRtpPacketSize() const RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  uint32_t SSRC() const RTC_LOCKS_EXCLUDED(send_mutex_) { return ssrc_; }
+
+  absl::optional<uint32_t> FlexfecSsrc() const RTC_LOCKS_EXCLUDED(send_mutex_) {
+    return flexfec_ssrc_;
+  }
+
+  // Sends packet to `transport_` or to the pacer, depending on configuration.
+  // TODO(bugs.webrtc.org/XXX): Remove in favor of EnqueuePackets().
+  bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  // Pass a set of packets to RtpPacketSender instance, for paced or immediate
+  // sending to the network.
+  void EnqueuePackets(std::vector<std::unique_ptr<RtpPacketToSend>> packets)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+
+  void SetRtpState(const RtpState& rtp_state) RTC_LOCKS_EXCLUDED(send_mutex_);
+  RtpState GetRtpState() const RTC_LOCKS_EXCLUDED(send_mutex_);
+  void SetRtxRtpState(const RtpState& rtp_state)
+      RTC_LOCKS_EXCLUDED(send_mutex_);
+  RtpState GetRtxRtpState() const RTC_LOCKS_EXCLUDED(send_mutex_);
+
+ private:
+  std::unique_ptr<RtpPacketToSend> BuildRtxPacket(
+      const RtpPacketToSend& packet);
+
+  bool IsFecPacket(const RtpPacketToSend& packet) const;
+
+  void UpdateHeaderSizes() RTC_EXCLUSIVE_LOCKS_REQUIRED(send_mutex_);
+
+  void UpdateLastPacketState(const RtpPacketToSend& packet)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(send_mutex_);
+
+  Clock* const clock_;
+  Random random_ RTC_GUARDED_BY(send_mutex_);
+
+  const bool audio_configured_;
+
+  const uint32_t ssrc_;
+  const absl::optional<uint32_t> rtx_ssrc_;
+  const absl::optional<uint32_t> flexfec_ssrc_;
+
+  RtpPacketHistory* const packet_history_;
+  RtpPacketSender* const paced_sender_;
+
+  mutable Mutex send_mutex_;
+
+  bool sending_media_ RTC_GUARDED_BY(send_mutex_);
+  size_t max_packet_size_;
+
+  RtpHeaderExtensionMap rtp_header_extension_map_ RTC_GUARDED_BY(send_mutex_);
+  size_t max_media_packet_header_ RTC_GUARDED_BY(send_mutex_);
+  size_t max_padding_fec_packet_header_ RTC_GUARDED_BY(send_mutex_);
+
+  // RTP variables
+  uint32_t timestamp_offset_ RTC_GUARDED_BY(send_mutex_);
+  // RID value to send in the RID or RepairedRID header extension.
+  const std::string rid_;
+  // MID value to send in the MID header extension.
+  std::string mid_ RTC_GUARDED_BY(send_mutex_);
+  // Should we send MID/RID even when ACKed? (see below).
+  const bool always_send_mid_and_rid_;
+  // Track if any ACK has been received on the SSRC and RTX SSRC to indicate
+  // when to stop sending the MID and RID header extensions.
+  bool ssrc_has_acked_ RTC_GUARDED_BY(send_mutex_);
+  bool rtx_ssrc_has_acked_ RTC_GUARDED_BY(send_mutex_);
+  std::vector<uint32_t> csrcs_ RTC_GUARDED_BY(send_mutex_);
+  int rtx_ RTC_GUARDED_BY(send_mutex_);
+  // Mapping rtx_payload_type_map_[associated] = rtx.
+  std::map<int8_t, int8_t> rtx_payload_type_map_ RTC_GUARDED_BY(send_mutex_);
+  bool supports_bwe_extension_ RTC_GUARDED_BY(send_mutex_);
+
+  RateLimiter* const retransmission_rate_limiter_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
new file mode 100644
index 0000000000..244f644bd1
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -0,0 +1,400 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+
+#include <string.h>
+
+#include <memory>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+namespace {
+[[maybe_unused]] const char* FrameTypeToString(AudioFrameType frame_type) {
+  switch (frame_type) {
+    case AudioFrameType::kEmptyFrame:
+      return "empty";
+    case AudioFrameType::kAudioFrameSpeech:
+      return "audio_speech";
+    case AudioFrameType::kAudioFrameCN:
+      return "audio_cn";
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+constexpr char kIncludeCaptureClockOffset[] =
+    "WebRTC-IncludeCaptureClockOffset";
+
+}  // namespace
+
+RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
+    : clock_(clock),
+      rtp_sender_(rtp_sender),
+      absolute_capture_time_sender_(clock),
+      include_capture_clock_offset_(
+          !absl::StartsWith(field_trials_.Lookup(kIncludeCaptureClockOffset),
+                            "Disabled")) {
+  RTC_DCHECK(clock_);
+}
+
+RTPSenderAudio::~RTPSenderAudio() {}
+
+int32_t RTPSenderAudio::RegisterAudioPayload(absl::string_view payload_name,
+                                             const int8_t payload_type,
+                                             const uint32_t frequency,
+                                             const size_t channels,
+                                             const uint32_t rate) {
+  if (absl::EqualsIgnoreCase(payload_name, "cn")) {
+    MutexLock lock(&send_audio_mutex_);
+    //  we can have multiple CNG payload types
+    switch (frequency) {
+      case 8000:
+        cngnb_payload_type_ = payload_type;
+        break;
+      case 16000:
+        cngwb_payload_type_ = payload_type;
+        break;
+      case 32000:
+        cngswb_payload_type_ = payload_type;
+        break;
+      case 48000:
+        cngfb_payload_type_ = payload_type;
+        break;
+      default:
+        return -1;
+    }
+  } else if (absl::EqualsIgnoreCase(payload_name, "telephone-event")) {
+    MutexLock lock(&send_audio_mutex_);
+    // Don't add it to the list
+    // we dont want to allow send with a DTMF payloadtype
+    dtmf_payload_type_ = payload_type;
+    dtmf_payload_freq_ = frequency;
+    return 0;
+  } else if (payload_name == "audio") {
+    MutexLock lock(&send_audio_mutex_);
+    encoder_rtp_timestamp_frequency_ = frequency;
+    return 0;
+  }
+  return 0;
+}
+
+bool RTPSenderAudio::MarkerBit(AudioFrameType frame_type, int8_t payload_type) {
+  MutexLock lock(&send_audio_mutex_);
+  // for audio true for first packet in a speech burst
+  bool marker_bit = false;
+  if (last_payload_type_ != payload_type) {
+    if (payload_type != -1 && (cngnb_payload_type_ == payload_type ||
+                               cngwb_payload_type_ == payload_type ||
+                               cngswb_payload_type_ == payload_type ||
+                               cngfb_payload_type_ == payload_type)) {
+      // Only set a marker bit when we change payload type to a non CNG
+      return false;
+    }
+
+    // payload_type differ
+    if (last_payload_type_ == -1) {
+      if (frame_type != AudioFrameType::kAudioFrameCN) {
+        // first packet and NOT CNG
+        return true;
+      } else {
+        // first packet and CNG
+        inband_vad_active_ = true;
+        return false;
+      }
+    }
+
+    // not first packet AND
+    // not CNG AND
+    // payload_type changed
+
+    // set a marker bit when we change payload type
+    marker_bit = true;
+  }
+
+  // For G.723 G.729, AMR etc we can have inband VAD
+  if (frame_type == AudioFrameType::kAudioFrameCN) {
+    inband_vad_active_ = true;
+  } else if (inband_vad_active_) {
+    inband_vad_active_ = false;
+    marker_bit = true;
+  }
+  return marker_bit;
+}
+
+bool RTPSenderAudio::SendAudio(AudioFrameType frame_type,
+                               int8_t payload_type,
+                               uint32_t rtp_timestamp,
+                               const uint8_t* payload_data,
+                               size_t payload_size) {
+  return SendAudio(frame_type, payload_type, rtp_timestamp, payload_data,
+                   payload_size,
+                   // TODO(bugs.webrtc.org/10739) replace once plumbed.
+                   /*absolute_capture_timestamp_ms=*/-1);
+}
+
+bool RTPSenderAudio::SendAudio(AudioFrameType frame_type,
+                               int8_t payload_type,
+                               uint32_t rtp_timestamp,
+                               const uint8_t* payload_data,
+                               size_t payload_size,
+                               int64_t absolute_capture_timestamp_ms) {
+  TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
+                          FrameTypeToString(frame_type));
+
+  // From RFC 4733:
+  // A source has wide latitude as to how often it sends event updates. A
+  // natural interval is the spacing between non-event audio packets. [...]
+  // Alternatively, a source MAY decide to use a different spacing for event
+  // updates, with a value of 50 ms RECOMMENDED.
+  constexpr int kDtmfIntervalTimeMs = 50;
+  uint8_t audio_level_dbov = 0;
+  uint32_t dtmf_payload_freq = 0;
+  absl::optional<uint32_t> encoder_rtp_timestamp_frequency;
+  {
+    MutexLock lock(&send_audio_mutex_);
+    audio_level_dbov = audio_level_dbov_;
+    dtmf_payload_freq = dtmf_payload_freq_;
+    encoder_rtp_timestamp_frequency = encoder_rtp_timestamp_frequency_;
+  }
+
+  // Check if we have pending DTMFs to send
+  if (!dtmf_event_is_on_ && dtmf_queue_.PendingDtmf()) {
+    if ((clock_->TimeInMilliseconds() - dtmf_time_last_sent_) >
+        kDtmfIntervalTimeMs) {
+      // New tone to play
+      dtmf_timestamp_ = rtp_timestamp;
+      if (dtmf_queue_.NextDtmf(&dtmf_current_event_)) {
+        dtmf_event_first_packet_sent_ = false;
+        dtmf_length_samples_ =
+            dtmf_current_event_.duration_ms * (dtmf_payload_freq / 1000);
+        dtmf_event_is_on_ = true;
+      }
+    }
+  }
+
+  // A source MAY send events and coded audio packets for the same time
+  // but we don't support it
+  if (dtmf_event_is_on_) {
+    if (frame_type == AudioFrameType::kEmptyFrame) {
+      // kEmptyFrame is used to drive the DTMF when in CN mode
+      // it can be triggered more frequently than we want to send the
+      // DTMF packets.
+      const unsigned int dtmf_interval_time_rtp =
+          dtmf_payload_freq * kDtmfIntervalTimeMs / 1000;
+      if ((rtp_timestamp - dtmf_timestamp_last_sent_) <
+          dtmf_interval_time_rtp) {
+        // not time to send yet
+        return true;
+      }
+    }
+    dtmf_timestamp_last_sent_ = rtp_timestamp;
+    uint32_t dtmf_duration_samples = rtp_timestamp - dtmf_timestamp_;
+    bool ended = false;
+    bool send = true;
+
+    if (dtmf_length_samples_ > dtmf_duration_samples) {
+      if (dtmf_duration_samples <= 0) {
+        // Skip send packet at start, since we shouldn't use duration 0
+        send = false;
+      }
+    } else {
+      ended = true;
+      dtmf_event_is_on_ = false;
+      dtmf_time_last_sent_ = clock_->TimeInMilliseconds();
+    }
+    if (send) {
+      if (dtmf_duration_samples > 0xffff) {
+        // RFC 4733 2.5.2.3 Long-Duration Events
+        SendTelephoneEventPacket(ended, dtmf_timestamp_,
+                                 static_cast<uint16_t>(0xffff), false);
+
+        // set new timestap for this segment
+        dtmf_timestamp_ = rtp_timestamp;
+        dtmf_duration_samples -= 0xffff;
+        dtmf_length_samples_ -= 0xffff;
+
+        return SendTelephoneEventPacket(
+            ended, dtmf_timestamp_,
+            static_cast<uint16_t>(dtmf_duration_samples), false);
+      } else {
+        if (!SendTelephoneEventPacket(ended, dtmf_timestamp_,
+                                      dtmf_duration_samples,
+                                      !dtmf_event_first_packet_sent_)) {
+          return false;
+        }
+        dtmf_event_first_packet_sent_ = true;
+        return true;
+      }
+    }
+    return true;
+  }
+  if (payload_size == 0 || payload_data == NULL) {
+    if (frame_type == AudioFrameType::kEmptyFrame) {
+      // we don't send empty audio RTP packets
+      // no error since we use it to either drive DTMF when we use VAD, or
+      // enter DTX.
+      return true;
+    }
+    return false;
+  }
+
+  std::unique_ptr<RtpPacketToSend> packet = rtp_sender_->AllocatePacket();
+  packet->SetMarker(MarkerBit(frame_type, payload_type));
+  packet->SetPayloadType(payload_type);
+  packet->SetTimestamp(rtp_timestamp);
+  packet->set_capture_time(clock_->CurrentTime());
+  // Update audio level extension, if included.
+  packet->SetExtension<AudioLevel>(
+      frame_type == AudioFrameType::kAudioFrameSpeech, audio_level_dbov);
+
+  if (absolute_capture_timestamp_ms > 0) {
+    // Send absolute capture time periodically in order to optimize and save
+    // network traffic. Missing absolute capture times can be interpolated on
+    // the receiving end if sending intervals are small enough.
+    auto absolute_capture_time = absolute_capture_time_sender_.OnSendPacket(
+        AbsoluteCaptureTimeSender::GetSource(packet->Ssrc(), packet->Csrcs()),
+        packet->Timestamp(),
+        // Replace missing value with 0 (invalid frequency), this will trigger
+        // absolute capture time sending.
+        encoder_rtp_timestamp_frequency.value_or(0),
+        Int64MsToUQ32x32(clock_->ConvertTimestampToNtpTimeInMilliseconds(
+            absolute_capture_timestamp_ms)),
+        /*estimated_capture_clock_offset=*/
+        include_capture_clock_offset_ ? absl::make_optional(0) : absl::nullopt);
+    if (absolute_capture_time) {
+      // It also checks that extension was registered during SDP negotiation. If
+      // not then setter won't do anything.
+      packet->SetExtension<AbsoluteCaptureTimeExtension>(
+          *absolute_capture_time);
+    }
+  }
+
+  uint8_t* payload = packet->AllocatePayload(payload_size);
+  if (!payload)  // Too large payload buffer.
+    return false;
+  memcpy(payload, payload_data, payload_size);
+
+  {
+    MutexLock lock(&send_audio_mutex_);
+    last_payload_type_ = payload_type;
+  }
+  TRACE_EVENT_ASYNC_END2("webrtc", "Audio", rtp_timestamp, "timestamp",
+                         packet->Timestamp(), "seqnum",
+                         packet->SequenceNumber());
+  packet->set_packet_type(RtpPacketMediaType::kAudio);
+  packet->set_allow_retransmission(true);
+  bool send_result = rtp_sender_->SendToNetwork(std::move(packet));
+  if (first_packet_sent_()) {
+    RTC_LOG(LS_INFO) << "First audio RTP packet sent to pacer";
+  }
+  return send_result;
+}
+
+// Audio level magnitude and voice activity flag are set for each RTP packet
+int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dbov) {
+  if (level_dbov > 127) {
+    return -1;
+  }
+  MutexLock lock(&send_audio_mutex_);
+  audio_level_dbov_ = level_dbov;
+  return 0;
+}
+
+// Send a TelephoneEvent tone using RFC 2833 (4733)
+int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
+                                           uint16_t time_ms,
+                                           uint8_t level) {
+  DtmfQueue::Event event;
+  {
+    MutexLock lock(&send_audio_mutex_);
+    if (dtmf_payload_type_ < 0) {
+      // TelephoneEvent payloadtype not configured
+      return -1;
+    }
+    event.payload_type = dtmf_payload_type_;
+  }
+  event.key = key;
+  event.duration_ms = time_ms;
+  event.level = level;
+  return dtmf_queue_.AddDtmf(event) ? 0 : -1;
+}
+
+bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
+                                              uint32_t dtmf_timestamp,
+                                              uint16_t duration,
+                                              bool marker_bit) {
+  uint8_t send_count = 1;
+  bool result = true;
+
+  if (ended) {
+    // resend last packet in an event 3 times
+    send_count = 3;
+  }
+  do {
+    // Send DTMF data.
+    constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr;
+    constexpr size_t kDtmfSize = 4;
+    std::unique_ptr<RtpPacketToSend> packet(
+        new RtpPacketToSend(kNoExtensions, kRtpHeaderSize + kDtmfSize));
+    packet->SetPayloadType(dtmf_current_event_.payload_type);
+    packet->SetMarker(marker_bit);
+    packet->SetSsrc(rtp_sender_->SSRC());
+    packet->SetTimestamp(dtmf_timestamp);
+    packet->set_capture_time(clock_->CurrentTime());
+
+    // Create DTMF data.
+    uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize);
+    RTC_DCHECK(dtmfbuffer);
+    /*    From RFC 2833:
+     0                   1                   2                   3
+     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    |     event     |E|R| volume    |          duration             |
+    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    */
+    // R bit always cleared
+    uint8_t R = 0x00;
+    uint8_t volume = dtmf_current_event_.level;
+
+    // First packet un-ended
+    uint8_t E = ended ? 0x80 : 0x00;
+
+    // First byte is Event number, equals key number
+    dtmfbuffer[0] = dtmf_current_event_.key;
+    dtmfbuffer[1] = E | R | volume;
+    ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration);
+
+    packet->set_packet_type(RtpPacketMediaType::kAudio);
+    packet->set_allow_retransmission(true);
+    result = rtp_sender_->SendToNetwork(std::move(packet));
+    send_count--;
+  } while (send_count > 0 && result);
+
+  return result;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
new file mode 100644
index 0000000000..d40fee6386
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/transport/field_trial_based_config.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
+#include "modules/rtp_rtcp/source/dtmf_queue.h"
+#include "modules/rtp_rtcp/source/rtp_sender.h"
+#include "rtc_base/one_time_event.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class RTPSenderAudio {
+ public:
+  RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
+
+  RTPSenderAudio() = delete;
+  RTPSenderAudio(const RTPSenderAudio&) = delete;
+  RTPSenderAudio& operator=(const RTPSenderAudio&) = delete;
+
+  ~RTPSenderAudio();
+
+  int32_t RegisterAudioPayload(absl::string_view payload_name,
+                               int8_t payload_type,
+                               uint32_t frequency,
+                               size_t channels,
+                               uint32_t rate);
+
+  bool SendAudio(AudioFrameType frame_type,
+                 int8_t payload_type,
+                 uint32_t rtp_timestamp,
+                 const uint8_t* payload_data,
+                 size_t payload_size);
+
+  // `absolute_capture_timestamp_ms` and `Clock::CurrentTime`
+  // should be using the same epoch.
+  bool SendAudio(AudioFrameType frame_type,
+                 int8_t payload_type,
+                 uint32_t rtp_timestamp,
+                 const uint8_t* payload_data,
+                 size_t payload_size,
+                 int64_t absolute_capture_timestamp_ms);
+
+  // Store the audio level in dBov for
+  // header-extension-for-audio-level-indication.
+  // Valid range is [0,127]. Actual value is negative.
+  int32_t SetAudioLevel(uint8_t level_dbov);
+
+  // Send a DTMF tone using RFC 2833 (4733)
+  int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
+
+ protected:
+  bool SendTelephoneEventPacket(
+      bool ended,
+      uint32_t dtmf_timestamp,
+      uint16_t duration,
+      bool marker_bit);  // set on first packet in talk burst
+
+  bool MarkerBit(AudioFrameType frame_type, int8_t payload_type);
+
+ private:
+  Clock* const clock_ = nullptr;
+  RTPSender* const rtp_sender_ = nullptr;
+
+  Mutex send_audio_mutex_;
+
+  // DTMF.
+  bool dtmf_event_is_on_ = false;
+  bool dtmf_event_first_packet_sent_ = false;
+  int8_t dtmf_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  uint32_t dtmf_payload_freq_ RTC_GUARDED_BY(send_audio_mutex_) = 8000;
+  uint32_t dtmf_timestamp_ = 0;
+  uint32_t dtmf_length_samples_ = 0;
+  int64_t dtmf_time_last_sent_ = 0;
+  uint32_t dtmf_timestamp_last_sent_ = 0;
+  DtmfQueue::Event dtmf_current_event_;
+  DtmfQueue dtmf_queue_;
+
+  // VAD detection, used for marker bit.
+  bool inband_vad_active_ RTC_GUARDED_BY(send_audio_mutex_) = false;
+  int8_t cngnb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  int8_t cngwb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  int8_t cngswb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  int8_t cngfb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  int8_t last_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+
+  // Audio level indication.
+  // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
+  uint8_t audio_level_dbov_ RTC_GUARDED_BY(send_audio_mutex_) = 127;
+  OneTimeEvent first_packet_sent_;
+
+  absl::optional<uint32_t> encoder_rtp_timestamp_frequency_
+      RTC_GUARDED_BY(send_audio_mutex_);
+
+  AbsoluteCaptureTimeSender absolute_capture_time_sender_;
+
+  const FieldTrialBasedConfig field_trials_;
+  const bool include_capture_clock_offset_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
new file mode 100644
index 0000000000..7397a3ac4e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -0,0 +1,264 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/transport/field_trial_based_config.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/thread.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+enum : int {  // The first valid value is 1.
+  kAudioLevelExtensionId = 1,
+  kAbsoluteCaptureTimeExtensionId = 2,
+};
+
+const uint16_t kSeqNum = 33;
+const uint32_t kSsrc = 725242;
+const uint8_t kAudioLevel = 0x5a;
+const uint64_t kStartTime = 123456789;
+
+using ::testing::ElementsAreArray;
+
+class LoopbackTransportTest : public webrtc::Transport {
+ public:
+  LoopbackTransportTest() {
+    receivers_extensions_.Register<AudioLevel>(kAudioLevelExtensionId);
+    receivers_extensions_.Register<AbsoluteCaptureTimeExtension>(
+        kAbsoluteCaptureTimeExtensionId);
+  }
+
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& /*options*/) override {
+    sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
+    EXPECT_TRUE(sent_packets_.back().Parse(data, len));
+    return true;
+  }
+  bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
+  const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); }
+  int packets_sent() { return sent_packets_.size(); }
+
+ private:
+  RtpHeaderExtensionMap receivers_extensions_;
+  std::vector<RtpPacketReceived> sent_packets_;
+};
+
+}  // namespace
+
+class RtpSenderAudioTest : public ::testing::Test {
+ public:
+  RtpSenderAudioTest()
+      : fake_clock_(kStartTime),
+        rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
+          RtpRtcpInterface::Configuration config;
+          config.audio = true;
+          config.clock = &fake_clock_;
+          config.outgoing_transport = &transport_;
+          config.local_media_ssrc = kSsrc;
+          return config;
+        }())),
+        rtp_sender_audio_(
+            std::make_unique<RTPSenderAudio>(&fake_clock_,
+                                             rtp_module_->RtpSender())) {
+    rtp_module_->SetSequenceNumber(kSeqNum);
+  }
+
+  rtc::AutoThread main_thread_;
+  SimulatedClock fake_clock_;
+  LoopbackTransportTest transport_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_module_;
+  std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
+};
+
+TEST_F(RtpSenderAudioTest, SendAudio) {
+  const char payload_name[] = "PAYLOAD_NAME";
+  const uint8_t payload_type = 127;
+  ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
+                   payload_name, payload_type, 48000, 0, 1500));
+  uint8_t payload[] = {47, 11, 32, 93, 89};
+
+  ASSERT_TRUE(
+      rtp_sender_audio_->SendAudio(AudioFrameType::kAudioFrameCN, payload_type,
+                                   4321, payload, sizeof(payload),
+                                   /*absolute_capture_timestamp_ms=*/0));
+
+  auto sent_payload = transport_.last_sent_packet().payload();
+  EXPECT_THAT(sent_payload, ElementsAreArray(payload));
+}
+
+TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
+  EXPECT_EQ(0, rtp_sender_audio_->SetAudioLevel(kAudioLevel));
+  rtp_module_->RegisterRtpHeaderExtension(AudioLevel::Uri(),
+                                          kAudioLevelExtensionId);
+
+  const char payload_name[] = "PAYLOAD_NAME";
+  const uint8_t payload_type = 127;
+  ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
+                   payload_name, payload_type, 48000, 0, 1500));
+
+  uint8_t payload[] = {47, 11, 32, 93, 89};
+
+  ASSERT_TRUE(
+      rtp_sender_audio_->SendAudio(AudioFrameType::kAudioFrameCN, payload_type,
+                                   4321, payload, sizeof(payload),
+                                   /*absolute_capture_timestamp_ms=*/0));
+
+  auto sent_payload = transport_.last_sent_packet().payload();
+  EXPECT_THAT(sent_payload, ElementsAreArray(payload));
+  // Verify AudioLevel extension.
+  bool voice_activity;
+  uint8_t audio_level;
+  EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>(
+      &voice_activity, &audio_level));
+  EXPECT_EQ(kAudioLevel, audio_level);
+  EXPECT_FALSE(voice_activity);
+}
+
+TEST_F(RtpSenderAudioTest, SendAudioWithoutAbsoluteCaptureTime) {
+  constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
+  const char payload_name[] = "audio";
+  const uint8_t payload_type = 127;
+  ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
+                   payload_name, payload_type, 48000, 0, 1500));
+  uint8_t payload[] = {47, 11, 32, 93, 89};
+
+  ASSERT_TRUE(rtp_sender_audio_->SendAudio(
+      AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
+      sizeof(payload), kAbsoluteCaptureTimestampMs));
+
+  EXPECT_FALSE(transport_.last_sent_packet()
+                   .HasExtension<AbsoluteCaptureTimeExtension>());
+}
+
+// Essentially the same test as
+// SendAudioWithAbsoluteCaptureTimeWithCaptureClockOffset but with a field
+// trial. We will remove this test eventually.
+TEST_F(RtpSenderAudioTest, SendAudioWithAbsoluteCaptureTime) {
+  // Recreate rtp_sender_audio_ with new field trial.
+  test::ScopedFieldTrials field_trial(
+      "WebRTC-IncludeCaptureClockOffset/Disabled/");
+  rtp_sender_audio_ =
+      std::make_unique<RTPSenderAudio>(&fake_clock_, rtp_module_->RtpSender());
+
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
+                                          kAbsoluteCaptureTimeExtensionId);
+  constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
+  const char payload_name[] = "audio";
+  const uint8_t payload_type = 127;
+  ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
+                   payload_name, payload_type, 48000, 0, 1500));
+  uint8_t payload[] = {47, 11, 32, 93, 89};
+
+  ASSERT_TRUE(rtp_sender_audio_->SendAudio(
+      AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
+      sizeof(payload), kAbsoluteCaptureTimestampMs));
+
+  auto absolute_capture_time =
+      transport_.last_sent_packet()
+          .GetExtension<AbsoluteCaptureTimeExtension>();
+  EXPECT_TRUE(absolute_capture_time);
+  EXPECT_EQ(
+      absolute_capture_time->absolute_capture_timestamp,
+      Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
+          kAbsoluteCaptureTimestampMs)));
+  EXPECT_FALSE(
+      absolute_capture_time->estimated_capture_clock_offset.has_value());
+}
+
+TEST_F(RtpSenderAudioTest,
+       SendAudioWithAbsoluteCaptureTimeWithCaptureClockOffset) {
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
+                                          kAbsoluteCaptureTimeExtensionId);
+  constexpr uint32_t kAbsoluteCaptureTimestampMs = 521;
+  const char payload_name[] = "audio";
+  const uint8_t payload_type = 127;
+  ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
+                   payload_name, payload_type, 48000, 0, 1500));
+  uint8_t payload[] = {47, 11, 32, 93, 89};
+
+  ASSERT_TRUE(rtp_sender_audio_->SendAudio(
+      AudioFrameType::kAudioFrameCN, payload_type, 4321, payload,
+      sizeof(payload), kAbsoluteCaptureTimestampMs));
+
+  auto absolute_capture_time =
+      transport_.last_sent_packet()
+          .GetExtension<AbsoluteCaptureTimeExtension>();
+  EXPECT_TRUE(absolute_capture_time);
+  EXPECT_EQ(
+      absolute_capture_time->absolute_capture_timestamp,
+      Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
+          kAbsoluteCaptureTimestampMs)));
+  EXPECT_TRUE(
+      absolute_capture_time->estimated_capture_clock_offset.has_value());
+  EXPECT_EQ(0, *absolute_capture_time->estimated_capture_clock_offset);
+}
+
+// As RFC4733, named telephone events are carried as part of the audio stream
+// and must use the same sequence number and timestamp base as the regular
+// audio channel.
+// This test checks the marker bit for the first packet and the consequent
+// packets of the same telephone event. Since it is specifically for DTMF
+// events, ignoring audio packets and sending kEmptyFrame instead of those.
+TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
+  const char* kDtmfPayloadName = "telephone-event";
+  const uint32_t kPayloadFrequency = 8000;
+  const uint8_t kPayloadType = 126;
+  ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
+                   kDtmfPayloadName, kPayloadType, kPayloadFrequency, 0, 0));
+  // For Telephone events, payload is not added to the registered payload list,
+  // it will register only the payload used for audio stream.
+  // Registering the payload again for audio stream with different payload name.
+  const char* kPayloadName = "payload_name";
+  ASSERT_EQ(0, rtp_sender_audio_->RegisterAudioPayload(
+                   kPayloadName, kPayloadType, kPayloadFrequency, 1, 0));
+  // Start time is arbitrary.
+  uint32_t capture_timestamp = fake_clock_.TimeInMilliseconds();
+  // DTMF event key=9, duration=500 and attenuationdB=10
+  rtp_sender_audio_->SendTelephoneEvent(9, 500, 10);
+  // During start, it takes the starting timestamp as last sent timestamp.
+  // The duration is calculated as the difference of current and last sent
+  // timestamp. So for first call it will skip since the duration is zero.
+  ASSERT_TRUE(rtp_sender_audio_->SendAudio(
+      AudioFrameType::kEmptyFrame, kPayloadType, capture_timestamp, nullptr, 0,
+      /*absolute_capture_time_ms=0*/ 0));
+
+  // DTMF Sample Length is (Frequency/1000) * Duration.
+  // So in this case, it is (8000/1000) * 500 = 4000.
+  // Sending it as two packets.
+  ASSERT_TRUE(rtp_sender_audio_->SendAudio(AudioFrameType::kEmptyFrame,
+                                           kPayloadType,
+                                           capture_timestamp + 2000, nullptr, 0,
+                                           /*absolute_capture_time_ms=0*/ 0));
+
+  // Marker Bit should be set to 1 for first packet.
+  EXPECT_TRUE(transport_.last_sent_packet().Marker());
+
+  ASSERT_TRUE(rtp_sender_audio_->SendAudio(AudioFrameType::kEmptyFrame,
+                                           kPayloadType,
+                                           capture_timestamp + 4000, nullptr, 0,
+                                           /*absolute_capture_time_ms=0*/ 0));
+  // Marker Bit should be set to 0 for rest of the packets.
+  EXPECT_FALSE(transport_.last_sent_packet().Marker());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc
new file mode 100644
index 0000000000..e4e0993d71
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -0,0 +1,617 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sender_egress.h"
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "api/transport/field_trial_based_config.h"
+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+constexpr uint32_t kTimestampTicksPerMs = 90;
+constexpr TimeDelta kSendSideDelayWindow = TimeDelta::Seconds(1);
+constexpr int kBitrateStatisticsWindowMs = 1000;
+constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
+constexpr TimeDelta kUpdateInterval =
+    TimeDelta::Millis(kBitrateStatisticsWindowMs);
+}  // namespace
+
+RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
+    RtpSenderEgress* sender,
+    PacketSequencer* sequencer)
+    : transport_sequence_number_(0), sender_(sender), sequencer_(sequencer) {
+  RTC_DCHECK(sequencer);
+}
+RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() = default;
+
+void RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
+    std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+  for (auto& packet : packets) {
+    PrepareForSend(packet.get());
+    sender_->SendPacket(packet.get(), PacedPacketInfo());
+  }
+  auto fec_packets = sender_->FetchFecPackets();
+  if (!fec_packets.empty()) {
+    EnqueuePackets(std::move(fec_packets));
+  }
+}
+
+void RtpSenderEgress::NonPacedPacketSender::PrepareForSend(
+    RtpPacketToSend* packet) {
+  // Assign sequence numbers, but not for flexfec which is already running on
+  // an internally maintained sequence number series.
+  if (packet->Ssrc() != sender_->FlexFecSsrc()) {
+    sequencer_->Sequence(*packet);
+  }
+  if (!packet->SetExtension<TransportSequenceNumber>(
+          ++transport_sequence_number_)) {
+    --transport_sequence_number_;
+  }
+  packet->ReserveExtension<TransmissionOffset>();
+  packet->ReserveExtension<AbsoluteSendTime>();
+}
+
+RtpSenderEgress::RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
+                                 RtpPacketHistory* packet_history)
+    : worker_queue_(TaskQueueBase::Current()),
+      ssrc_(config.local_media_ssrc),
+      rtx_ssrc_(config.rtx_send_ssrc),
+      flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc()
+                                         : absl::nullopt),
+      populate_network2_timestamp_(config.populate_network2_timestamp),
+      clock_(config.clock),
+      packet_history_(packet_history),
+      transport_(config.outgoing_transport),
+      event_log_(config.event_log),
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+      is_audio_(config.audio),
+#endif
+      need_rtp_packet_infos_(config.need_rtp_packet_infos),
+      fec_generator_(config.fec_generator),
+      transport_feedback_observer_(config.transport_feedback_callback),
+      send_side_delay_observer_(config.send_side_delay_observer),
+      send_packet_observer_(config.send_packet_observer),
+      rtp_stats_callback_(config.rtp_stats_callback),
+      bitrate_callback_(config.send_bitrate_observer),
+      media_has_been_sent_(false),
+      force_part_of_allocation_(false),
+      timestamp_offset_(0),
+      max_delay_it_(send_delays_.end()),
+      sum_delays_(TimeDelta::Zero()),
+      send_rates_(kNumMediaTypes,
+                  {kBitrateStatisticsWindowMs, RateStatistics::kBpsScale}),
+      rtp_sequence_number_map_(need_rtp_packet_infos_
+                                   ? std::make_unique<RtpSequenceNumberMap>(
+                                         kRtpSequenceNumberMapMaxEntries)
+                                   : nullptr) {
+  RTC_DCHECK(worker_queue_);
+  pacer_checker_.Detach();
+  if (bitrate_callback_) {
+    update_task_ = RepeatingTaskHandle::DelayedStart(worker_queue_,
+                                                     kUpdateInterval, [this]() {
+                                                       PeriodicUpdate();
+                                                       return kUpdateInterval;
+                                                     });
+  }
+}
+
+RtpSenderEgress::~RtpSenderEgress() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  update_task_.Stop();
+}
+
+void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
+                                 const PacedPacketInfo& pacing_info) {
+  RTC_DCHECK_RUN_ON(&pacer_checker_);
+  RTC_DCHECK(packet);
+
+  if (packet->Ssrc() == ssrc_ &&
+      packet->packet_type() != RtpPacketMediaType::kRetransmission) {
+    if (last_sent_seq_.has_value()) {
+      RTC_DCHECK_EQ(static_cast<uint16_t>(*last_sent_seq_ + 1),
+                    packet->SequenceNumber());
+    }
+    last_sent_seq_ = packet->SequenceNumber();
+  } else if (packet->Ssrc() == rtx_ssrc_) {
+    if (last_sent_rtx_seq_.has_value()) {
+      RTC_DCHECK_EQ(static_cast<uint16_t>(*last_sent_rtx_seq_ + 1),
+                    packet->SequenceNumber());
+    }
+    last_sent_rtx_seq_ = packet->SequenceNumber();
+  }
+
+  RTC_DCHECK(packet->packet_type().has_value());
+  RTC_DCHECK(HasCorrectSsrc(*packet));
+  if (packet->packet_type() == RtpPacketMediaType::kRetransmission) {
+    RTC_DCHECK(packet->retransmitted_sequence_number().has_value());
+  }
+
+  const uint32_t packet_ssrc = packet->Ssrc();
+  const Timestamp now = clock_->CurrentTime();
+
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+  worker_queue_->PostTask(SafeTask(
+      task_safety_.flag(),
+      [this, now, packet_ssrc]() { BweTestLoggingPlot(now, packet_ssrc); }));
+#endif
+
+  if (need_rtp_packet_infos_ &&
+      packet->packet_type() == RtpPacketToSend::Type::kVideo) {
+    worker_queue_->PostTask(SafeTask(
+        task_safety_.flag(),
+        [this, packet_timestamp = packet->Timestamp(),
+         is_first_packet_of_frame = packet->is_first_packet_of_frame(),
+         is_last_packet_of_frame = packet->Marker(),
+         sequence_number = packet->SequenceNumber()]() {
+          RTC_DCHECK_RUN_ON(worker_queue_);
+          // Last packet of a frame, add it to sequence number info map.
+          const uint32_t timestamp = packet_timestamp - timestamp_offset_;
+          rtp_sequence_number_map_->InsertPacket(
+              sequence_number,
+              RtpSequenceNumberMap::Info(timestamp, is_first_packet_of_frame,
+                                         is_last_packet_of_frame));
+        }));
+  }
+
+  if (fec_generator_ && packet->fec_protect_packet()) {
+    // This packet should be protected by FEC, add it to packet generator.
+    RTC_DCHECK(fec_generator_);
+    RTC_DCHECK(packet->packet_type() == RtpPacketMediaType::kVideo);
+    absl::optional<std::pair<FecProtectionParams, FecProtectionParams>>
+        new_fec_params;
+    {
+      MutexLock lock(&lock_);
+      new_fec_params.swap(pending_fec_params_);
+    }
+    if (new_fec_params) {
+      fec_generator_->SetProtectionParameters(new_fec_params->first,
+                                              new_fec_params->second);
+    }
+    if (packet->is_red()) {
+      RtpPacketToSend unpacked_packet(*packet);
+
+      const rtc::CopyOnWriteBuffer buffer = packet->Buffer();
+      // Grab media payload type from RED header.
+      const size_t headers_size = packet->headers_size();
+      unpacked_packet.SetPayloadType(buffer[headers_size]);
+
+      // Copy the media payload into the unpacked buffer.
+      uint8_t* payload_buffer =
+          unpacked_packet.SetPayloadSize(packet->payload_size() - 1);
+      std::copy(&packet->payload()[0] + 1,
+                &packet->payload()[0] + packet->payload_size(), payload_buffer);
+
+      fec_generator_->AddPacketAndGenerateFec(unpacked_packet);
+    } else {
+      // If not RED encapsulated - we can just insert packet directly.
+      fec_generator_->AddPacketAndGenerateFec(*packet);
+    }
+  }
+
+  // Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
+  // the pacer, these modifications of the header below are happening after the
+  // FEC protection packets are calculated. This will corrupt recovered packets
+  // at the same place. It's not an issue for extensions, which are present in
+  // all the packets (their content just may be incorrect on recovered packets).
+  // In case of VideoTimingExtension, since it's present not in every packet,
+  // data after rtp header may be corrupted if these packets are protected by
+  // the FEC.
+  if (packet->HasExtension<TransmissionOffset>() &&
+      packet->capture_time() > Timestamp::Zero()) {
+    TimeDelta diff = now - packet->capture_time();
+    packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff.ms());
+  }
+  if (packet->HasExtension<AbsoluteSendTime>()) {
+    packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
+  }
+
+  if (packet->HasExtension<VideoTimingExtension>()) {
+    if (populate_network2_timestamp_) {
+      packet->set_network2_time(now);
+    } else {
+      packet->set_pacer_exit_time(now);
+    }
+  }
+
+  const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio ||
+                        packet->packet_type() == RtpPacketMediaType::kVideo;
+
+  PacketOptions options;
+  {
+    MutexLock lock(&lock_);
+    options.included_in_allocation = force_part_of_allocation_;
+  }
+
+  // Downstream code actually uses this flag to distinguish between media and
+  // everything else.
+  options.is_retransmit = !is_media;
+  if (auto packet_id = packet->GetExtension<TransportSequenceNumber>()) {
+    options.packet_id = *packet_id;
+    options.included_in_feedback = true;
+    options.included_in_allocation = true;
+    AddPacketToTransportFeedback(*packet_id, *packet, pacing_info);
+  }
+
+  options.additional_data = packet->additional_data();
+
+  if (packet->packet_type() != RtpPacketMediaType::kPadding &&
+      packet->packet_type() != RtpPacketMediaType::kRetransmission) {
+    UpdateDelayStatistics(packet->capture_time(), now, packet_ssrc);
+    UpdateOnSendPacket(options.packet_id, packet->capture_time(), packet_ssrc);
+  }
+
+  const bool send_success = SendPacketToNetwork(*packet, options, pacing_info);
+
+  // Put packet in retransmission history or update pending status even if
+  // actual sending fails.
+  if (is_media && packet->allow_retransmission()) {
+    packet_history_->PutRtpPacket(std::make_unique<RtpPacketToSend>(*packet),
+                                  now);
+  } else if (packet->retransmitted_sequence_number()) {
+    packet_history_->MarkPacketAsSent(*packet->retransmitted_sequence_number());
+  }
+
+  if (send_success) {
+    // `media_has_been_sent_` is used by RTPSender to figure out if it can send
+    // padding in the absence of transport-cc or abs-send-time.
+    // In those cases media must be sent first to set a reference timestamp.
+    media_has_been_sent_ = true;
+
+    // TODO(sprang): Add support for FEC protecting all header extensions, add
+    // media packet to generator here instead.
+
+    RTC_DCHECK(packet->packet_type().has_value());
+    RtpPacketMediaType packet_type = *packet->packet_type();
+    RtpPacketCounter counter(*packet);
+    size_t size = packet->size();
+    worker_queue_->PostTask(
+        SafeTask(task_safety_.flag(), [this, now, packet_ssrc, packet_type,
+                                       counter = std::move(counter), size]() {
+          RTC_DCHECK_RUN_ON(worker_queue_);
+          UpdateRtpStats(now, packet_ssrc, packet_type, std::move(counter),
+                         size);
+        }));
+  }
+}
+
+RtpSendRates RtpSenderEgress::GetSendRates() const {
+  MutexLock lock(&lock_);
+  return GetSendRatesLocked(clock_->CurrentTime());
+}
+
+RtpSendRates RtpSenderEgress::GetSendRatesLocked(Timestamp now) const {
+  RtpSendRates current_rates;
+  for (size_t i = 0; i < kNumMediaTypes; ++i) {
+    RtpPacketMediaType type = static_cast<RtpPacketMediaType>(i);
+    current_rates[type] =
+        DataRate::BitsPerSec(send_rates_[i].Rate(now.ms()).value_or(0));
+  }
+  return current_rates;
+}
+
+void RtpSenderEgress::GetDataCounters(StreamDataCounters* rtp_stats,
+                                      StreamDataCounters* rtx_stats) const {
+  // TODO(bugs.webrtc.org/11581): make sure rtx_rtp_stats_ and rtp_stats_ are
+  // only touched on the worker thread.
+  MutexLock lock(&lock_);
+  *rtp_stats = rtp_stats_;
+  *rtx_stats = rtx_rtp_stats_;
+}
+
+void RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
+    bool part_of_allocation) {
+  MutexLock lock(&lock_);
+  force_part_of_allocation_ = part_of_allocation;
+}
+
+bool RtpSenderEgress::MediaHasBeenSent() const {
+  RTC_DCHECK_RUN_ON(&pacer_checker_);
+  return media_has_been_sent_;
+}
+
+void RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) {
+  RTC_DCHECK_RUN_ON(&pacer_checker_);
+  media_has_been_sent_ = media_sent;
+}
+
+void RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  timestamp_offset_ = timestamp;
+}
+
+std::vector<RtpSequenceNumberMap::Info> RtpSenderEgress::GetSentRtpPacketInfos(
+    rtc::ArrayView<const uint16_t> sequence_numbers) const {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  RTC_DCHECK(!sequence_numbers.empty());
+  if (!need_rtp_packet_infos_) {
+    return std::vector<RtpSequenceNumberMap::Info>();
+  }
+
+  std::vector<RtpSequenceNumberMap::Info> results;
+  results.reserve(sequence_numbers.size());
+
+  for (uint16_t sequence_number : sequence_numbers) {
+    const auto& info = rtp_sequence_number_map_->Get(sequence_number);
+    if (!info) {
+      // The empty vector will be returned. We can delay the clearing
+      // of the vector until after we exit the critical section.
+      return std::vector<RtpSequenceNumberMap::Info>();
+    }
+    results.push_back(*info);
+  }
+
+  return results;
+}
+
+void RtpSenderEgress::SetFecProtectionParameters(
+    const FecProtectionParams& delta_params,
+    const FecProtectionParams& key_params) {
+  // TODO(sprang): Post task to pacer queue instead, one pacer is fully
+  // migrated to a task queue.
+  MutexLock lock(&lock_);
+  pending_fec_params_.emplace(delta_params, key_params);
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+RtpSenderEgress::FetchFecPackets() {
+  RTC_DCHECK_RUN_ON(&pacer_checker_);
+  if (fec_generator_) {
+    return fec_generator_->GetFecPackets();
+  }
+  return {};
+}
+
+void RtpSenderEgress::OnAbortedRetransmissions(
+    rtc::ArrayView<const uint16_t> sequence_numbers) {
+  RTC_DCHECK_RUN_ON(&pacer_checker_);
+  // Mark aborted retransmissions as sent, rather than leaving them in
+  // a 'pending' state - otherwise they can not be requested again and
+  // will not be cleared until the history has reached its max size.
+  for (uint16_t seq_no : sequence_numbers) {
+    packet_history_->MarkPacketAsSent(seq_no);
+  }
+}
+
+bool RtpSenderEgress::HasCorrectSsrc(const RtpPacketToSend& packet) const {
+  switch (*packet.packet_type()) {
+    case RtpPacketMediaType::kAudio:
+    case RtpPacketMediaType::kVideo:
+      return packet.Ssrc() == ssrc_;
+    case RtpPacketMediaType::kRetransmission:
+    case RtpPacketMediaType::kPadding:
+      // Both padding and retransmission must be on either the media or the
+      // RTX stream.
+      return packet.Ssrc() == rtx_ssrc_ || packet.Ssrc() == ssrc_;
+    case RtpPacketMediaType::kForwardErrorCorrection:
+      // FlexFEC is on separate SSRC, ULPFEC uses media SSRC.
+      return packet.Ssrc() == ssrc_ || packet.Ssrc() == flexfec_ssrc_;
+  }
+  return false;
+}
+
+void RtpSenderEgress::AddPacketToTransportFeedback(
+    uint16_t packet_id,
+    const RtpPacketToSend& packet,
+    const PacedPacketInfo& pacing_info) {
+  if (transport_feedback_observer_) {
+    RtpPacketSendInfo packet_info;
+    packet_info.transport_sequence_number = packet_id;
+    packet_info.rtp_timestamp = packet.Timestamp();
+    packet_info.length = packet.size();
+    packet_info.pacing_info = pacing_info;
+    packet_info.packet_type = packet.packet_type();
+
+    switch (*packet_info.packet_type) {
+      case RtpPacketMediaType::kAudio:
+      case RtpPacketMediaType::kVideo:
+        packet_info.media_ssrc = ssrc_;
+        packet_info.rtp_sequence_number = packet.SequenceNumber();
+        break;
+      case RtpPacketMediaType::kRetransmission:
+        // For retransmissions, we're want to remove the original media packet
+        // if the retransmit arrives - so populate that in the packet info.
+        packet_info.media_ssrc = ssrc_;
+        packet_info.rtp_sequence_number =
+            *packet.retransmitted_sequence_number();
+        break;
+      case RtpPacketMediaType::kPadding:
+      case RtpPacketMediaType::kForwardErrorCorrection:
+        // We're not interested in feedback about these packets being received
+        // or lost.
+        break;
+    }
+
+    transport_feedback_observer_->OnAddPacket(packet_info);
+  }
+}
+
+void RtpSenderEgress::UpdateDelayStatistics(Timestamp capture_time,
+                                            Timestamp now,
+                                            uint32_t ssrc) {
+  if (!send_side_delay_observer_ || capture_time.IsInfinite())
+    return;
+
+  TimeDelta avg_delay = TimeDelta::Zero();
+  TimeDelta max_delay = TimeDelta::Zero();
+  {
+    MutexLock lock(&lock_);
+    // Compute the max and average of the recent capture-to-send delays.
+    // The time complexity of the current approach depends on the distribution
+    // of the delay values. This could be done more efficiently.
+
+    // Remove elements older than kSendSideDelayWindowMs.
+    auto lower_bound = send_delays_.lower_bound(now - kSendSideDelayWindow);
+    for (auto it = send_delays_.begin(); it != lower_bound; ++it) {
+      if (max_delay_it_ == it) {
+        max_delay_it_ = send_delays_.end();
+      }
+      sum_delays_ -= it->second;
+    }
+    send_delays_.erase(send_delays_.begin(), lower_bound);
+    if (max_delay_it_ == send_delays_.end()) {
+      // Removed the previous max. Need to recompute.
+      RecomputeMaxSendDelay();
+    }
+
+    // Add the new element.
+    TimeDelta new_send_delay = now - capture_time;
+    auto [it, inserted] = send_delays_.emplace(now, new_send_delay);
+    if (!inserted) {
+      // TODO(terelius): If we have multiple delay measurements during the same
+      // millisecond then we keep the most recent one. It is not clear that this
+      // is the right decision, but it preserves an earlier behavior.
+      TimeDelta previous_send_delay = it->second;
+      sum_delays_ -= previous_send_delay;
+      it->second = new_send_delay;
+      if (max_delay_it_ == it && new_send_delay < previous_send_delay) {
+        RecomputeMaxSendDelay();
+      }
+    }
+    if (max_delay_it_ == send_delays_.end() ||
+        it->second >= max_delay_it_->second) {
+      max_delay_it_ = it;
+    }
+    sum_delays_ += new_send_delay;
+
+    size_t num_delays = send_delays_.size();
+    RTC_DCHECK(max_delay_it_ != send_delays_.end());
+    max_delay = max_delay_it_->second;
+    avg_delay = sum_delays_ / num_delays;
+  }
+  send_side_delay_observer_->SendSideDelayUpdated(avg_delay.ms(),
+                                                  max_delay.ms(), ssrc);
+}
+
+void RtpSenderEgress::RecomputeMaxSendDelay() {
+  max_delay_it_ = send_delays_.begin();
+  for (auto it = send_delays_.begin(); it != send_delays_.end(); ++it) {
+    if (it->second >= max_delay_it_->second) {
+      max_delay_it_ = it;
+    }
+  }
+}
+
+void RtpSenderEgress::UpdateOnSendPacket(int packet_id,
+                                         Timestamp capture_time,
+                                         uint32_t ssrc) {
+  if (!send_packet_observer_ || capture_time.IsInfinite() || packet_id == -1) {
+    return;
+  }
+
+  send_packet_observer_->OnSendPacket(packet_id, capture_time.ms(), ssrc);
+}
+
+bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
+                                          const PacketOptions& options,
+                                          const PacedPacketInfo& pacing_info) {
+  int bytes_sent = -1;
+  if (transport_) {
+    bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
+                     ? static_cast<int>(packet.size())
+                     : -1;
+    if (event_log_ && bytes_sent > 0) {
+      event_log_->Log(std::make_unique<RtcEventRtpPacketOutgoing>(
+          packet, pacing_info.probe_cluster_id));
+    }
+  }
+
+  if (bytes_sent <= 0) {
+    RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
+    return false;
+  }
+  return true;
+}
+
+void RtpSenderEgress::UpdateRtpStats(Timestamp now,
+                                     uint32_t packet_ssrc,
+                                     RtpPacketMediaType packet_type,
+                                     RtpPacketCounter counter,
+                                     size_t packet_size) {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+
+  // TODO(bugs.webrtc.org/11581): send_rates_ should be touched only on the
+  // worker thread.
+  RtpSendRates send_rates;
+  {
+    MutexLock lock(&lock_);
+
+    // TODO(bugs.webrtc.org/11581): make sure rtx_rtp_stats_ and rtp_stats_ are
+    // only touched on the worker thread.
+    StreamDataCounters* counters =
+        packet_ssrc == rtx_ssrc_ ? &rtx_rtp_stats_ : &rtp_stats_;
+
+    if (counters->first_packet_time_ms == -1) {
+      counters->first_packet_time_ms = now.ms();
+    }
+
+    if (packet_type == RtpPacketMediaType::kForwardErrorCorrection) {
+      counters->fec.Add(counter);
+    } else if (packet_type == RtpPacketMediaType::kRetransmission) {
+      counters->retransmitted.Add(counter);
+    }
+    counters->transmitted.Add(counter);
+
+    send_rates_[static_cast<size_t>(packet_type)].Update(packet_size, now.ms());
+    if (bitrate_callback_) {
+      send_rates = GetSendRatesLocked(now);
+    }
+
+    if (rtp_stats_callback_) {
+      rtp_stats_callback_->DataCountersUpdated(*counters, packet_ssrc);
+    }
+  }
+
+  // The bitrate_callback_ and rtp_stats_callback_ pointers in practice point
+  // to the same object, so these callbacks could be consolidated into one.
+  if (bitrate_callback_) {
+    bitrate_callback_->Notify(
+        send_rates.Sum().bps(),
+        send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
+  }
+}
+
+void RtpSenderEgress::PeriodicUpdate() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  RTC_DCHECK(bitrate_callback_);
+  RtpSendRates send_rates = GetSendRates();
+  bitrate_callback_->Notify(
+      send_rates.Sum().bps(),
+      send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
+}
+
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+void RtpSenderEgress::BweTestLoggingPlot(Timestamp now, uint32_t packet_ssrc) {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+
+  const auto rates = GetSendRates();
+  if (is_audio_) {
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now.ms(),
+                                    rates.Sum().kbps(), packet_ssrc);
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
+        1, "AudioNackBitrate_kbps", now.ms(),
+        rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
+  } else {
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now.ms(),
+                                    rates.Sum().kbps(), packet_ssrc);
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
+        1, "VideoNackBitrate_kbps", now.ms(),
+        rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
+  }
+}
+#endif  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress.h
new file mode 100644
index 0000000000..f76c42d4d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_
+
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/call/transport.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/packet_sequencer.h"
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class RtpSenderEgress {
+ public:
+  // Helper class that redirects packets directly to the send part of this class
+  // without passing through an actual paced sender.
+  class NonPacedPacketSender : public RtpPacketSender {
+   public:
+    NonPacedPacketSender(RtpSenderEgress* sender, PacketSequencer* sequencer);
+    virtual ~NonPacedPacketSender();
+
+    void EnqueuePackets(
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets) override;
+    // Since we don't pace packets, there's no pending packets to remove.
+    void RemovePacketsForSsrc(uint32_t ssrc) override {}
+
+   private:
+    void PrepareForSend(RtpPacketToSend* packet);
+    uint16_t transport_sequence_number_;
+    RtpSenderEgress* const sender_;
+    PacketSequencer* sequencer_;
+  };
+
+  RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
+                  RtpPacketHistory* packet_history);
+  ~RtpSenderEgress();
+
+  void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info)
+      RTC_LOCKS_EXCLUDED(lock_);
+  uint32_t Ssrc() const { return ssrc_; }
+  absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; }
+  absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; }
+
+  RtpSendRates GetSendRates() const RTC_LOCKS_EXCLUDED(lock_);
+  void GetDataCounters(StreamDataCounters* rtp_stats,
+                       StreamDataCounters* rtx_stats) const
+      RTC_LOCKS_EXCLUDED(lock_);
+
+  void ForceIncludeSendPacketsInAllocation(bool part_of_allocation)
+      RTC_LOCKS_EXCLUDED(lock_);
+  bool MediaHasBeenSent() const RTC_LOCKS_EXCLUDED(lock_);
+  void SetMediaHasBeenSent(bool media_sent) RTC_LOCKS_EXCLUDED(lock_);
+  void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(lock_);
+
+  // For each sequence number in `sequence_number`, recall the last RTP packet
+  // which bore it - its timestamp and whether it was the first and/or last
+  // packet in that frame. If all of the given sequence numbers could be
+  // recalled, return a vector with all of them (in corresponding order).
+  // If any could not be recalled, return an empty vector.
+  std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+      rtc::ArrayView<const uint16_t> sequence_numbers) const
+      RTC_LOCKS_EXCLUDED(lock_);
+
+  void SetFecProtectionParameters(const FecProtectionParams& delta_params,
+                                  const FecProtectionParams& key_params);
+  std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets();
+
+  // Clears pending status for these sequence numbers in the packet history.
+  void OnAbortedRetransmissions(
+      rtc::ArrayView<const uint16_t> sequence_numbers);
+
+ private:
+  RtpSendRates GetSendRatesLocked(Timestamp now) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  bool HasCorrectSsrc(const RtpPacketToSend& packet) const;
+  void AddPacketToTransportFeedback(uint16_t packet_id,
+                                    const RtpPacketToSend& packet,
+                                    const PacedPacketInfo& pacing_info);
+  void UpdateDelayStatistics(Timestamp capture_time,
+                             Timestamp now,
+                             uint32_t ssrc);
+  void RecomputeMaxSendDelay() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  void UpdateOnSendPacket(int packet_id, Timestamp capture_time, uint32_t ssrc);
+  // Sends packet on to `transport_`, leaving the RTP module.
+  bool SendPacketToNetwork(const RtpPacketToSend& packet,
+                           const PacketOptions& options,
+                           const PacedPacketInfo& pacing_info);
+
+  void UpdateRtpStats(Timestamp now,
+                      uint32_t packet_ssrc,
+                      RtpPacketMediaType packet_type,
+                      RtpPacketCounter counter,
+                      size_t packet_size);
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+  void BweTestLoggingPlot(Timestamp now, uint32_t packet_ssrc);
+#endif
+
+  // Called on a timer, once a second, on the worker_queue_.
+  void PeriodicUpdate();
+
+  TaskQueueBase* const worker_queue_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker pacer_checker_;
+  const uint32_t ssrc_;
+  const absl::optional<uint32_t> rtx_ssrc_;
+  const absl::optional<uint32_t> flexfec_ssrc_;
+  const bool populate_network2_timestamp_;
+  Clock* const clock_;
+  RtpPacketHistory* const packet_history_;
+  Transport* const transport_;
+  RtcEventLog* const event_log_;
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+  const bool is_audio_;
+#endif
+  const bool need_rtp_packet_infos_;
+  VideoFecGenerator* const fec_generator_ RTC_GUARDED_BY(pacer_checker_);
+  absl::optional<uint16_t> last_sent_seq_ RTC_GUARDED_BY(pacer_checker_);
+  absl::optional<uint16_t> last_sent_rtx_seq_ RTC_GUARDED_BY(pacer_checker_);
+
+  TransportFeedbackObserver* const transport_feedback_observer_;
+  SendSideDelayObserver* const send_side_delay_observer_;
+  SendPacketObserver* const send_packet_observer_;
+  StreamDataCountersCallback* const rtp_stats_callback_;
+  BitrateStatisticsObserver* const bitrate_callback_;
+
+  mutable Mutex lock_;
+  bool media_has_been_sent_ RTC_GUARDED_BY(pacer_checker_);
+  bool force_part_of_allocation_ RTC_GUARDED_BY(lock_);
+  uint32_t timestamp_offset_ RTC_GUARDED_BY(worker_queue_);
+
+  // Maps capture time to send-side delay. Send-side delay is the difference
+  // between transmission time and capture time.
+  std::map<Timestamp, TimeDelta> send_delays_ RTC_GUARDED_BY(lock_);
+  std::map<Timestamp, TimeDelta>::const_iterator max_delay_it_
+      RTC_GUARDED_BY(lock_);
+  // The sum of delays over a kSendSideDelayWindowMs sliding window.
+  TimeDelta sum_delays_ RTC_GUARDED_BY(lock_);
+  StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_);
+  StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
+  // One element per value in RtpPacketMediaType, with index matching value.
+  std::vector<RateStatistics> send_rates_ RTC_GUARDED_BY(lock_);
+  absl::optional<std::pair<FecProtectionParams, FecProtectionParams>>
+      pending_fec_params_ RTC_GUARDED_BY(lock_);
+
+  // Maps sent packets' sequence numbers to a tuple consisting of:
+  // 1. The timestamp, without the randomizing offset mandated by the RFC.
+  // 2. Whether the packet was the first in its frame.
+  // 3. Whether the packet was the last in its frame.
+  const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_
+      RTC_GUARDED_BY(worker_queue_);
+  RepeatingTaskHandle update_task_ RTC_GUARDED_BY(worker_queue_);
+  ScopedTaskSafety task_safety_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_EGRESS_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
new file mode 100644
index 0000000000..bfaffa1bdb
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
@@ -0,0 +1,990 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sender_egress.h"
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/call/transport.h"
+#include "api/field_trials_registry.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/rtp_rtcp/include/flexfec_sender.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::Field;
+using ::testing::NiceMock;
+using ::testing::Optional;
+using ::testing::StrictMock;
+
+constexpr Timestamp kStartTime = Timestamp::Millis(123456789);
+constexpr int kDefaultPayloadType = 100;
+constexpr int kFlexfectPayloadType = 110;
+constexpr uint16_t kStartSequenceNumber = 33;
+constexpr uint32_t kSsrc = 725242;
+constexpr uint32_t kRtxSsrc = 12345;
+constexpr uint32_t kFlexFecSsrc = 23456;
+enum : int {
+  kTransportSequenceNumberExtensionId = 1,
+  kAbsoluteSendTimeExtensionId,
+  kTransmissionOffsetExtensionId,
+  kVideoTimingExtensionId,
+};
+
+class MockSendPacketObserver : public SendPacketObserver {
+ public:
+  MOCK_METHOD(void, OnSendPacket, (uint16_t, int64_t, uint32_t), (override));
+};
+
+class MockTransportFeedbackObserver : public TransportFeedbackObserver {
+ public:
+  MOCK_METHOD(void, OnAddPacket, (const RtpPacketSendInfo&), (override));
+  MOCK_METHOD(void,
+              OnTransportFeedback,
+              (const rtcp::TransportFeedback&),
+              (override));
+};
+
+class MockStreamDataCountersCallback : public StreamDataCountersCallback {
+ public:
+  MOCK_METHOD(void,
+              DataCountersUpdated,
+              (const StreamDataCounters& counters, uint32_t ssrc),
+              (override));
+};
+
+class MockSendSideDelayObserver : public SendSideDelayObserver {
+ public:
+  MOCK_METHOD(void, SendSideDelayUpdated, (int, int, uint32_t), (override));
+};
+
+struct TransmittedPacket {
+  TransmittedPacket(rtc::ArrayView<const uint8_t> data,
+                    const PacketOptions& packet_options,
+                    RtpHeaderExtensionMap* extensions)
+      : packet(extensions), options(packet_options) {
+    EXPECT_TRUE(packet.Parse(data));
+  }
+  RtpPacketReceived packet;
+  PacketOptions options;
+};
+
+class TestTransport : public Transport {
+ public:
+  explicit TestTransport(RtpHeaderExtensionMap* extensions)
+      : total_data_sent_(DataSize::Zero()), extensions_(extensions) {}
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) override {
+    total_data_sent_ += DataSize::Bytes(length);
+    last_packet_.emplace(rtc::MakeArrayView(packet, length), options,
+                         extensions_);
+    return true;
+  }
+
+  bool SendRtcp(const uint8_t*, size_t) override { RTC_CHECK_NOTREACHED(); }
+
+  absl::optional<TransmittedPacket> last_packet() { return last_packet_; }
+
+ private:
+  DataSize total_data_sent_;
+  absl::optional<TransmittedPacket> last_packet_;
+  RtpHeaderExtensionMap* const extensions_;
+};
+
+}  // namespace
+
+class RtpSenderEgressTest : public ::testing::Test {
+ protected:
+  RtpSenderEgressTest()
+      : time_controller_(kStartTime),
+        clock_(time_controller_.GetClock()),
+        transport_(&header_extensions_),
+        packet_history_(clock_, /*enable_rtx_padding_prioritization=*/true),
+        trials_(""),
+        sequence_number_(kStartSequenceNumber) {}
+
+  std::unique_ptr<RtpSenderEgress> CreateRtpSenderEgress() {
+    return std::make_unique<RtpSenderEgress>(DefaultConfig(), &packet_history_);
+  }
+
+  RtpRtcpInterface::Configuration DefaultConfig() {
+    RtpRtcpInterface::Configuration config;
+    config.clock = clock_;
+    config.outgoing_transport = &transport_;
+    config.local_media_ssrc = kSsrc;
+    config.rtx_send_ssrc = kRtxSsrc;
+    config.fec_generator = nullptr;
+    config.event_log = &mock_rtc_event_log_;
+    config.send_packet_observer = &send_packet_observer_;
+    config.rtp_stats_callback = &mock_rtp_stats_callback_;
+    config.transport_feedback_callback = &feedback_observer_;
+    config.populate_network2_timestamp = false;
+    config.field_trials = &trials_;
+    return config;
+  }
+
+  std::unique_ptr<RtpPacketToSend> BuildRtpPacket(bool marker_bit,
+                                                  int64_t capture_time_ms) {
+    auto packet = std::make_unique<RtpPacketToSend>(&header_extensions_);
+    packet->SetSsrc(kSsrc);
+    packet->ReserveExtension<AbsoluteSendTime>();
+    packet->ReserveExtension<TransmissionOffset>();
+    packet->ReserveExtension<TransportSequenceNumber>();
+
+    packet->SetPayloadType(kDefaultPayloadType);
+    packet->set_packet_type(RtpPacketMediaType::kVideo);
+    packet->SetMarker(marker_bit);
+    packet->SetTimestamp(capture_time_ms * 90);
+    packet->set_capture_time(Timestamp::Millis(capture_time_ms));
+    packet->SetSequenceNumber(sequence_number_++);
+    return packet;
+  }
+
+  std::unique_ptr<RtpPacketToSend> BuildRtpPacket() {
+    return BuildRtpPacket(/*marker_bit=*/true, clock_->CurrentTime().ms());
+  }
+
+  GlobalSimulatedTimeController time_controller_;
+  Clock* const clock_;
+  NiceMock<MockRtcEventLog> mock_rtc_event_log_;
+  NiceMock<MockStreamDataCountersCallback> mock_rtp_stats_callback_;
+  NiceMock<MockSendPacketObserver> send_packet_observer_;
+  NiceMock<MockTransportFeedbackObserver> feedback_observer_;
+  RtpHeaderExtensionMap header_extensions_;
+  TestTransport transport_;
+  RtpPacketHistory packet_history_;
+  test::ExplicitKeyValueConfig trials_;
+  uint16_t sequence_number_;
+};
+
+TEST_F(RtpSenderEgressTest, TransportFeedbackObserverGetsCorrectByteCount) {
+  constexpr size_t kRtpOverheadBytesPerPacket = 12 + 8;
+  constexpr size_t kPayloadSize = 1400;
+  const uint16_t kTransportSequenceNumber = 17;
+
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+
+  const size_t expected_bytes = kPayloadSize + kRtpOverheadBytesPerPacket;
+
+  EXPECT_CALL(
+      feedback_observer_,
+      OnAddPacket(AllOf(
+          Field(&RtpPacketSendInfo::media_ssrc, kSsrc),
+          Field(&RtpPacketSendInfo::transport_sequence_number,
+                kTransportSequenceNumber),
+          Field(&RtpPacketSendInfo::rtp_sequence_number, kStartSequenceNumber),
+          Field(&RtpPacketSendInfo::length, expected_bytes),
+          Field(&RtpPacketSendInfo::pacing_info, PacedPacketInfo()))));
+
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+  packet->AllocatePayload(kPayloadSize);
+
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+}
+
+TEST_F(RtpSenderEgressTest, PacketOptionsIsRetransmitSetByPacketType) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+
+  std::unique_ptr<RtpPacketToSend> media_packet = BuildRtpPacket();
+  media_packet->set_packet_type(RtpPacketMediaType::kVideo);
+  sender->SendPacket(media_packet.get(), PacedPacketInfo());
+  EXPECT_FALSE(transport_.last_packet()->options.is_retransmit);
+
+  std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
+  retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
+  retransmission->set_retransmitted_sequence_number(
+      media_packet->SequenceNumber());
+  sender->SendPacket(retransmission.get(), PacedPacketInfo());
+  EXPECT_TRUE(transport_.last_packet()->options.is_retransmit);
+}
+
+TEST_F(RtpSenderEgressTest, DoesnSetIncludedInAllocationByDefault) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+  EXPECT_FALSE(transport_.last_packet()->options.included_in_feedback);
+  EXPECT_FALSE(transport_.last_packet()->options.included_in_allocation);
+}
+
+TEST_F(RtpSenderEgressTest,
+       SetsIncludedInFeedbackWhenTransportSequenceNumberExtensionIsRegistered) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+  EXPECT_TRUE(transport_.last_packet()->options.included_in_feedback);
+}
+
+TEST_F(
+    RtpSenderEgressTest,
+    SetsIncludedInAllocationWhenTransportSequenceNumberExtensionIsRegistered) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+  EXPECT_TRUE(transport_.last_packet()->options.included_in_allocation);
+}
+
+TEST_F(RtpSenderEgressTest,
+       SetsIncludedInAllocationWhenForcedAsPartOfAllocation) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  sender->ForceIncludeSendPacketsInAllocation(true);
+
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+  EXPECT_FALSE(transport_.last_packet()->options.included_in_feedback);
+  EXPECT_TRUE(transport_.last_packet()->options.included_in_allocation);
+}
+
+TEST_F(RtpSenderEgressTest,
+       DoesntWriteTransmissionOffsetOnRtxPaddingBeforeMedia) {
+  header_extensions_.RegisterByUri(kTransmissionOffsetExtensionId,
+                                   TransmissionOffset::Uri());
+
+  // Prior to sending media, timestamps are 0.
+  std::unique_ptr<RtpPacketToSend> padding =
+      BuildRtpPacket(/*marker_bit=*/true, /*capture_time_ms=*/0);
+  padding->set_packet_type(RtpPacketMediaType::kPadding);
+  padding->SetSsrc(kRtxSsrc);
+  EXPECT_TRUE(padding->HasExtension<TransmissionOffset>());
+
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  sender->SendPacket(std::move(padding), PacedPacketInfo());
+
+  absl::optional<int32_t> offset =
+      transport_.last_packet()->packet.GetExtension<TransmissionOffset>();
+  EXPECT_EQ(offset, 0);
+}
+
+TEST_F(RtpSenderEgressTest, OnSendSideDelayUpdated) {
+  StrictMock<MockSendSideDelayObserver> send_side_delay_observer;
+  RtpRtcpInterface::Configuration config = DefaultConfig();
+  config.send_side_delay_observer = &send_side_delay_observer;
+  auto sender = std::make_unique<RtpSenderEgress>(config, &packet_history_);
+
+  // Send packet with 10 ms send-side delay. The average, max and total should
+  // be 10 ms.
+  EXPECT_CALL(send_side_delay_observer, SendSideDelayUpdated(10, 10, kSsrc));
+  int64_t capture_time_ms = clock_->TimeInMilliseconds();
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  sender->SendPacket(BuildRtpPacket(/*marker=*/true, capture_time_ms).get(),
+                     PacedPacketInfo());
+
+  // Send another packet with 20 ms delay. The average, max and total should be
+  // 15, 20 and 30 ms respectively.
+  EXPECT_CALL(send_side_delay_observer, SendSideDelayUpdated(15, 20, kSsrc));
+  capture_time_ms = clock_->TimeInMilliseconds();
+  time_controller_.AdvanceTime(TimeDelta::Millis(20));
+  sender->SendPacket(BuildRtpPacket(/*marker=*/true, capture_time_ms).get(),
+                     PacedPacketInfo());
+
+  // Send another packet at the same time, which replaces the last packet.
+  // Since this packet has 0 ms delay, the average is now 5 ms and max is 10 ms.
+  // The total counter stays the same though.
+  // TODO(terelius): Is is not clear that this is the right behavior.
+  EXPECT_CALL(send_side_delay_observer, SendSideDelayUpdated(5, 10, kSsrc));
+  capture_time_ms = clock_->TimeInMilliseconds();
+  sender->SendPacket(BuildRtpPacket(/*marker=*/true, capture_time_ms).get(),
+                     PacedPacketInfo());
+
+  // Send a packet 1 second later. The earlier packets should have timed
+  // out, so both max and average should be the delay of this packet. The total
+  // keeps increasing.
+  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
+  EXPECT_CALL(send_side_delay_observer, SendSideDelayUpdated(1, 1, kSsrc));
+  capture_time_ms = clock_->TimeInMilliseconds();
+  time_controller_.AdvanceTime(TimeDelta::Millis(1));
+  sender->SendPacket(BuildRtpPacket(/*marker=*/true, capture_time_ms).get(),
+                     PacedPacketInfo());
+}
+
+TEST_F(RtpSenderEgressTest, WritesPacerExitToTimingExtension) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  header_extensions_.RegisterByUri(kVideoTimingExtensionId,
+                                   VideoTimingExtension::Uri());
+
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  packet->SetExtension<VideoTimingExtension>(VideoSendTiming{});
+
+  const int kStoredTimeInMs = 100;
+  time_controller_.AdvanceTime(TimeDelta::Millis(kStoredTimeInMs));
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+  ASSERT_TRUE(transport_.last_packet().has_value());
+
+  VideoSendTiming video_timing;
+  EXPECT_TRUE(
+      transport_.last_packet()->packet.GetExtension<VideoTimingExtension>(
+          &video_timing));
+  EXPECT_EQ(video_timing.pacer_exit_delta_ms, kStoredTimeInMs);
+}
+
+TEST_F(RtpSenderEgressTest, WritesNetwork2ToTimingExtension) {
+  RtpRtcpInterface::Configuration rtp_config = DefaultConfig();
+  rtp_config.populate_network2_timestamp = true;
+  auto sender = std::make_unique<RtpSenderEgress>(rtp_config, &packet_history_);
+  header_extensions_.RegisterByUri(kVideoTimingExtensionId,
+                                   VideoTimingExtension::Uri());
+
+  const uint16_t kPacerExitMs = 1234u;
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  VideoSendTiming send_timing = {};
+  send_timing.pacer_exit_delta_ms = kPacerExitMs;
+  packet->SetExtension<VideoTimingExtension>(send_timing);
+
+  const int kStoredTimeInMs = 100;
+  time_controller_.AdvanceTime(TimeDelta::Millis(kStoredTimeInMs));
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+  ASSERT_TRUE(transport_.last_packet().has_value());
+
+  VideoSendTiming video_timing;
+  EXPECT_TRUE(
+      transport_.last_packet()->packet.GetExtension<VideoTimingExtension>(
+          &video_timing));
+  EXPECT_EQ(video_timing.network2_timestamp_delta_ms, kStoredTimeInMs);
+  EXPECT_EQ(video_timing.pacer_exit_delta_ms, kPacerExitMs);
+}
+
+TEST_F(RtpSenderEgressTest, OnSendPacketUpdated) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+
+  const uint16_t kTransportSequenceNumber = 1;
+  EXPECT_CALL(send_packet_observer_,
+              OnSendPacket(kTransportSequenceNumber,
+                           clock_->TimeInMilliseconds(), kSsrc));
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+}
+
+TEST_F(RtpSenderEgressTest, OnSendPacketNotUpdatedForRetransmits) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+
+  const uint16_t kTransportSequenceNumber = 1;
+  EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(0);
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+  packet->set_packet_type(RtpPacketMediaType::kRetransmission);
+  packet->set_retransmitted_sequence_number(packet->SequenceNumber());
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+}
+
+TEST_F(RtpSenderEgressTest, ReportsFecRate) {
+  constexpr int kNumPackets = 10;
+  constexpr TimeDelta kTimeBetweenPackets = TimeDelta::Millis(33);
+
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  DataSize total_fec_data_sent = DataSize::Zero();
+  // Send some packets, alternating between media and FEC.
+  for (size_t i = 0; i < kNumPackets; ++i) {
+    std::unique_ptr<RtpPacketToSend> media_packet = BuildRtpPacket();
+    media_packet->set_packet_type(RtpPacketMediaType::kVideo);
+    media_packet->SetPayloadSize(500);
+    sender->SendPacket(media_packet.get(), PacedPacketInfo());
+
+    std::unique_ptr<RtpPacketToSend> fec_packet = BuildRtpPacket();
+    fec_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
+    fec_packet->SetPayloadSize(123);
+    sender->SendPacket(fec_packet.get(), PacedPacketInfo());
+    total_fec_data_sent += DataSize::Bytes(fec_packet->size());
+
+    time_controller_.AdvanceTime(kTimeBetweenPackets);
+  }
+
+  EXPECT_NEAR(
+      (sender->GetSendRates()[RtpPacketMediaType::kForwardErrorCorrection])
+          .bps(),
+      (total_fec_data_sent / (kTimeBetweenPackets * kNumPackets)).bps(), 500);
+}
+
+TEST_F(RtpSenderEgressTest, BitrateCallbacks) {
+  class MockBitrateStaticsObserver : public BitrateStatisticsObserver {
+   public:
+    MOCK_METHOD(void, Notify, (uint32_t, uint32_t, uint32_t), (override));
+  } observer;
+
+  RtpRtcpInterface::Configuration config = DefaultConfig();
+  config.send_bitrate_observer = &observer;
+  auto sender = std::make_unique<RtpSenderEgress>(config, &packet_history_);
+
+  // Simulate kNumPackets sent with kPacketInterval intervals, with the
+  // number of packets selected so that we fill (but don't overflow) the one
+  // second averaging window.
+  const TimeDelta kWindowSize = TimeDelta::Seconds(1);
+  const TimeDelta kPacketInterval = TimeDelta::Millis(20);
+  const int kNumPackets = (kWindowSize - kPacketInterval) / kPacketInterval;
+
+  DataSize total_data_sent = DataSize::Zero();
+
+  // Send all but on of the packets, expect a call for each packet but don't
+  // verify bitrate yet (noisy measurements in the beginning).
+  for (int i = 0; i < kNumPackets; ++i) {
+    std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+    packet->SetPayloadSize(500);
+    // Mark all packets as retransmissions - will cause total and retransmission
+    // rates to be equal.
+    packet->set_packet_type(RtpPacketMediaType::kRetransmission);
+    packet->set_retransmitted_sequence_number(packet->SequenceNumber());
+    total_data_sent += DataSize::Bytes(packet->size());
+
+    EXPECT_CALL(observer, Notify(_, _, kSsrc))
+        .WillOnce([&](uint32_t total_bitrate_bps,
+                      uint32_t retransmission_bitrate_bps, uint32_t /*ssrc*/) {
+          TimeDelta window_size = i * kPacketInterval + TimeDelta::Millis(1);
+          // If there is just a single data point, there is no well defined
+          // averaging window so a bitrate of zero will be reported.
+          const double expected_bitrate_bps =
+              i == 0 ? 0.0 : (total_data_sent / window_size).bps();
+          EXPECT_NEAR(total_bitrate_bps, expected_bitrate_bps, 500);
+          EXPECT_NEAR(retransmission_bitrate_bps, expected_bitrate_bps, 500);
+        });
+
+    sender->SendPacket(packet.get(), PacedPacketInfo());
+    time_controller_.AdvanceTime(kPacketInterval);
+  }
+}
+
+TEST_F(RtpSenderEgressTest, DoesNotPutNotRetransmittablePacketsInHistory) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  packet_history_.SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 10);
+
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  packet->set_allow_retransmission(false);
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+  EXPECT_FALSE(packet_history_.GetPacketState(packet->SequenceNumber()));
+}
+
+TEST_F(RtpSenderEgressTest, PutsRetransmittablePacketsInHistory) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  packet_history_.SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 10);
+
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  packet->set_allow_retransmission(true);
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+  EXPECT_TRUE(packet_history_.GetPacketState(packet->SequenceNumber()));
+}
+
+TEST_F(RtpSenderEgressTest, DoesNotPutNonMediaInHistory) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  packet_history_.SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 10);
+
+  // Non-media packets, even when marked as retransmittable, are not put into
+  // the packet history.
+  std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
+  retransmission->set_allow_retransmission(true);
+  retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
+  retransmission->set_retransmitted_sequence_number(
+      retransmission->SequenceNumber());
+  sender->SendPacket(retransmission.get(), PacedPacketInfo());
+  EXPECT_FALSE(
+      packet_history_.GetPacketState(retransmission->SequenceNumber()));
+
+  std::unique_ptr<RtpPacketToSend> fec = BuildRtpPacket();
+  fec->set_allow_retransmission(true);
+  fec->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
+  sender->SendPacket(fec.get(), PacedPacketInfo());
+  EXPECT_FALSE(packet_history_.GetPacketState(fec->SequenceNumber()));
+
+  std::unique_ptr<RtpPacketToSend> padding = BuildRtpPacket();
+  padding->set_allow_retransmission(true);
+  padding->set_packet_type(RtpPacketMediaType::kPadding);
+  sender->SendPacket(padding.get(), PacedPacketInfo());
+  EXPECT_FALSE(packet_history_.GetPacketState(padding->SequenceNumber()));
+}
+
+TEST_F(RtpSenderEgressTest, UpdatesSendStatusOfRetransmittedPackets) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  packet_history_.SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 10);
+
+  // Send a packet, putting it in the history.
+  std::unique_ptr<RtpPacketToSend> media_packet = BuildRtpPacket();
+  media_packet->set_allow_retransmission(true);
+  sender->SendPacket(media_packet.get(), PacedPacketInfo());
+  EXPECT_TRUE(packet_history_.GetPacketState(media_packet->SequenceNumber()));
+
+  // Simulate a retransmission, marking the packet as pending.
+  std::unique_ptr<RtpPacketToSend> retransmission =
+      packet_history_.GetPacketAndMarkAsPending(media_packet->SequenceNumber());
+  retransmission->set_retransmitted_sequence_number(
+      media_packet->SequenceNumber());
+  retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
+  EXPECT_TRUE(packet_history_.GetPacketState(media_packet->SequenceNumber()));
+
+  // Simulate packet leaving pacer, the packet should be marked as non-pending.
+  sender->SendPacket(retransmission.get(), PacedPacketInfo());
+  EXPECT_TRUE(packet_history_.GetPacketState(media_packet->SequenceNumber()));
+}
+
+TEST_F(RtpSenderEgressTest, StreamDataCountersCallbacks) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+
+  const RtpPacketCounter kEmptyCounter;
+  RtpPacketCounter expected_transmitted_counter;
+  RtpPacketCounter expected_retransmission_counter;
+
+  // Send a media packet.
+  std::unique_ptr<RtpPacketToSend> media_packet = BuildRtpPacket();
+  media_packet->SetPayloadSize(6);
+  media_packet->SetSequenceNumber(kStartSequenceNumber);
+  media_packet->set_time_in_send_queue(TimeDelta::Millis(10));
+  expected_transmitted_counter.packets += 1;
+  expected_transmitted_counter.payload_bytes += media_packet->payload_size();
+  expected_transmitted_counter.header_bytes += media_packet->headers_size();
+  expected_transmitted_counter.total_packet_delay += TimeDelta::Millis(10);
+
+  EXPECT_CALL(
+      mock_rtp_stats_callback_,
+      DataCountersUpdated(AllOf(Field(&StreamDataCounters::transmitted,
+                                      expected_transmitted_counter),
+                                Field(&StreamDataCounters::retransmitted,
+                                      expected_retransmission_counter),
+                                Field(&StreamDataCounters::fec, kEmptyCounter)),
+                          kSsrc));
+  sender->SendPacket(media_packet.get(), PacedPacketInfo());
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  // Send a retransmission. Retransmissions are counted into both transmitted
+  // and retransmitted packet statistics.
+  std::unique_ptr<RtpPacketToSend> retransmission_packet = BuildRtpPacket();
+  retransmission_packet->set_packet_type(RtpPacketMediaType::kRetransmission);
+  retransmission_packet->SetSequenceNumber(kStartSequenceNumber);
+  retransmission_packet->set_retransmitted_sequence_number(
+      kStartSequenceNumber);
+  retransmission_packet->set_time_in_send_queue(TimeDelta::Millis(20));
+  media_packet->SetPayloadSize(7);
+  expected_transmitted_counter.packets += 1;
+  expected_transmitted_counter.payload_bytes +=
+      retransmission_packet->payload_size();
+  expected_transmitted_counter.header_bytes +=
+      retransmission_packet->headers_size();
+  expected_transmitted_counter.total_packet_delay += TimeDelta::Millis(20);
+
+  expected_retransmission_counter.packets += 1;
+  expected_retransmission_counter.payload_bytes +=
+      retransmission_packet->payload_size();
+  expected_retransmission_counter.header_bytes +=
+      retransmission_packet->headers_size();
+  expected_retransmission_counter.total_packet_delay += TimeDelta::Millis(20);
+
+  EXPECT_CALL(
+      mock_rtp_stats_callback_,
+      DataCountersUpdated(AllOf(Field(&StreamDataCounters::transmitted,
+                                      expected_transmitted_counter),
+                                Field(&StreamDataCounters::retransmitted,
+                                      expected_retransmission_counter),
+                                Field(&StreamDataCounters::fec, kEmptyCounter)),
+                          kSsrc));
+  sender->SendPacket(retransmission_packet.get(), PacedPacketInfo());
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  // Send a padding packet.
+  std::unique_ptr<RtpPacketToSend> padding_packet = BuildRtpPacket();
+  padding_packet->set_packet_type(RtpPacketMediaType::kPadding);
+  padding_packet->SetPadding(224);
+  padding_packet->SetSequenceNumber(kStartSequenceNumber + 1);
+  padding_packet->set_time_in_send_queue(TimeDelta::Millis(30));
+  expected_transmitted_counter.packets += 1;
+  expected_transmitted_counter.padding_bytes += padding_packet->padding_size();
+  expected_transmitted_counter.header_bytes += padding_packet->headers_size();
+  expected_transmitted_counter.total_packet_delay += TimeDelta::Millis(30);
+
+  EXPECT_CALL(
+      mock_rtp_stats_callback_,
+      DataCountersUpdated(AllOf(Field(&StreamDataCounters::transmitted,
+                                      expected_transmitted_counter),
+                                Field(&StreamDataCounters::retransmitted,
+                                      expected_retransmission_counter),
+                                Field(&StreamDataCounters::fec, kEmptyCounter)),
+                          kSsrc));
+  sender->SendPacket(padding_packet.get(), PacedPacketInfo());
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+}
+
+TEST_F(RtpSenderEgressTest, StreamDataCountersCallbacksFec) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+
+  const RtpPacketCounter kEmptyCounter;
+  RtpPacketCounter expected_transmitted_counter;
+  RtpPacketCounter expected_fec_counter;
+
+  // Send a media packet.
+  std::unique_ptr<RtpPacketToSend> media_packet = BuildRtpPacket();
+  media_packet->SetPayloadSize(6);
+  expected_transmitted_counter.packets += 1;
+  expected_transmitted_counter.payload_bytes += media_packet->payload_size();
+  expected_transmitted_counter.header_bytes += media_packet->headers_size();
+
+  EXPECT_CALL(
+      mock_rtp_stats_callback_,
+      DataCountersUpdated(
+          AllOf(Field(&StreamDataCounters::transmitted,
+                      expected_transmitted_counter),
+                Field(&StreamDataCounters::retransmitted, kEmptyCounter),
+                Field(&StreamDataCounters::fec, expected_fec_counter)),
+          kSsrc));
+  sender->SendPacket(media_packet.get(), PacedPacketInfo());
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  // Send and FEC packet. FEC is counted into both transmitted and FEC packet
+  // statistics.
+  std::unique_ptr<RtpPacketToSend> fec_packet = BuildRtpPacket();
+  fec_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
+  fec_packet->SetPayloadSize(6);
+  expected_transmitted_counter.packets += 1;
+  expected_transmitted_counter.payload_bytes += fec_packet->payload_size();
+  expected_transmitted_counter.header_bytes += fec_packet->headers_size();
+
+  expected_fec_counter.packets += 1;
+  expected_fec_counter.payload_bytes += fec_packet->payload_size();
+  expected_fec_counter.header_bytes += fec_packet->headers_size();
+
+  EXPECT_CALL(
+      mock_rtp_stats_callback_,
+      DataCountersUpdated(
+          AllOf(Field(&StreamDataCounters::transmitted,
+                      expected_transmitted_counter),
+                Field(&StreamDataCounters::retransmitted, kEmptyCounter),
+                Field(&StreamDataCounters::fec, expected_fec_counter)),
+          kSsrc));
+  sender->SendPacket(fec_packet.get(), PacedPacketInfo());
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+}
+
+TEST_F(RtpSenderEgressTest, UpdatesDataCounters) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+
+  const RtpPacketCounter kEmptyCounter;
+
+  // Send a media packet.
+  std::unique_ptr<RtpPacketToSend> media_packet = BuildRtpPacket();
+  media_packet->SetPayloadSize(6);
+  sender->SendPacket(media_packet.get(), PacedPacketInfo());
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  // Send an RTX retransmission packet.
+  std::unique_ptr<RtpPacketToSend> rtx_packet = BuildRtpPacket();
+  rtx_packet->set_packet_type(RtpPacketMediaType::kRetransmission);
+  rtx_packet->SetSsrc(kRtxSsrc);
+  rtx_packet->SetPayloadSize(7);
+  rtx_packet->set_retransmitted_sequence_number(media_packet->SequenceNumber());
+  sender->SendPacket(rtx_packet.get(), PacedPacketInfo());
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  StreamDataCounters rtp_stats;
+  StreamDataCounters rtx_stats;
+  sender->GetDataCounters(&rtp_stats, &rtx_stats);
+
+  EXPECT_EQ(rtp_stats.transmitted.packets, 1u);
+  EXPECT_EQ(rtp_stats.transmitted.payload_bytes, media_packet->payload_size());
+  EXPECT_EQ(rtp_stats.transmitted.padding_bytes, media_packet->padding_size());
+  EXPECT_EQ(rtp_stats.transmitted.header_bytes, media_packet->headers_size());
+  EXPECT_EQ(rtp_stats.retransmitted, kEmptyCounter);
+  EXPECT_EQ(rtp_stats.fec, kEmptyCounter);
+
+  // Retransmissions are counted both into transmitted and retransmitted
+  // packet counts.
+  EXPECT_EQ(rtx_stats.transmitted.packets, 1u);
+  EXPECT_EQ(rtx_stats.transmitted.payload_bytes, rtx_packet->payload_size());
+  EXPECT_EQ(rtx_stats.transmitted.padding_bytes, rtx_packet->padding_size());
+  EXPECT_EQ(rtx_stats.transmitted.header_bytes, rtx_packet->headers_size());
+  EXPECT_EQ(rtx_stats.retransmitted, rtx_stats.transmitted);
+  EXPECT_EQ(rtx_stats.fec, kEmptyCounter);
+}
+
+TEST_F(RtpSenderEgressTest, SendPacketUpdatesExtensions) {
+  header_extensions_.RegisterByUri(kVideoTimingExtensionId,
+                                   VideoTimingExtension::Uri());
+  header_extensions_.RegisterByUri(kAbsoluteSendTimeExtensionId,
+                                   AbsoluteSendTime::Uri());
+  header_extensions_.RegisterByUri(kTransmissionOffsetExtensionId,
+                                   TransmissionOffset::Uri());
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  packet->set_packetization_finish_time(clock_->CurrentTime());
+
+  const int32_t kDiffMs = 10;
+  time_controller_.AdvanceTime(TimeDelta::Millis(kDiffMs));
+
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+
+  RtpPacketReceived received_packet = transport_.last_packet()->packet;
+
+  EXPECT_EQ(received_packet.GetExtension<TransmissionOffset>(), kDiffMs * 90);
+
+  EXPECT_EQ(received_packet.GetExtension<AbsoluteSendTime>(),
+            AbsoluteSendTime::To24Bits(clock_->CurrentTime()));
+
+  VideoSendTiming timing;
+  EXPECT_TRUE(received_packet.GetExtension<VideoTimingExtension>(&timing));
+  EXPECT_EQ(timing.pacer_exit_delta_ms, kDiffMs);
+}
+
+TEST_F(RtpSenderEgressTest, SendPacketSetsPacketOptions) {
+  const uint16_t kPacketId = 42;
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+
+  std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+  packet->SetExtension<TransportSequenceNumber>(kPacketId);
+  EXPECT_CALL(send_packet_observer_, OnSendPacket);
+  sender->SendPacket(packet.get(), PacedPacketInfo());
+
+  PacketOptions packet_options = transport_.last_packet()->options;
+
+  EXPECT_EQ(packet_options.packet_id, kPacketId);
+  EXPECT_TRUE(packet_options.included_in_allocation);
+  EXPECT_TRUE(packet_options.included_in_feedback);
+  EXPECT_FALSE(packet_options.is_retransmit);
+
+  // Send another packet as retransmission, verify options are populated.
+  std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
+  retransmission->SetExtension<TransportSequenceNumber>(kPacketId + 1);
+  retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
+  retransmission->set_retransmitted_sequence_number(packet->SequenceNumber());
+  sender->SendPacket(retransmission.get(), PacedPacketInfo());
+  EXPECT_TRUE(transport_.last_packet()->options.is_retransmit);
+}
+
+TEST_F(RtpSenderEgressTest, SendPacketUpdatesStats) {
+  const size_t kPayloadSize = 1000;
+  StrictMock<MockSendSideDelayObserver> send_side_delay_observer;
+
+  const rtc::ArrayView<const RtpExtensionSize> kNoRtpHeaderExtensionSizes;
+  FlexfecSender flexfec(kFlexfectPayloadType, kFlexFecSsrc, kSsrc, /*mid=*/"",
+                        /*header_extensions=*/{}, kNoRtpHeaderExtensionSizes,
+                        /*rtp_state=*/nullptr, time_controller_.GetClock());
+  RtpRtcpInterface::Configuration config = DefaultConfig();
+  config.fec_generator = &flexfec;
+  config.send_side_delay_observer = &send_side_delay_observer;
+  auto sender = std::make_unique<RtpSenderEgress>(config, &packet_history_);
+
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+
+  const int64_t capture_time_ms = clock_->TimeInMilliseconds();
+
+  std::unique_ptr<RtpPacketToSend> video_packet = BuildRtpPacket();
+  video_packet->set_packet_type(RtpPacketMediaType::kVideo);
+  video_packet->SetPayloadSize(kPayloadSize);
+  video_packet->SetExtension<TransportSequenceNumber>(1);
+
+  std::unique_ptr<RtpPacketToSend> rtx_packet = BuildRtpPacket();
+  rtx_packet->SetSsrc(kRtxSsrc);
+  rtx_packet->set_packet_type(RtpPacketMediaType::kRetransmission);
+  rtx_packet->set_retransmitted_sequence_number(video_packet->SequenceNumber());
+  rtx_packet->SetPayloadSize(kPayloadSize);
+  rtx_packet->SetExtension<TransportSequenceNumber>(2);
+
+  std::unique_ptr<RtpPacketToSend> fec_packet = BuildRtpPacket();
+  fec_packet->SetSsrc(kFlexFecSsrc);
+  fec_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
+  fec_packet->SetPayloadSize(kPayloadSize);
+  fec_packet->SetExtension<TransportSequenceNumber>(3);
+
+  const int64_t kDiffMs = 25;
+  time_controller_.AdvanceTime(TimeDelta::Millis(kDiffMs));
+
+  EXPECT_CALL(send_side_delay_observer,
+              SendSideDelayUpdated(kDiffMs, kDiffMs, kSsrc));
+  EXPECT_CALL(send_side_delay_observer,
+              SendSideDelayUpdated(kDiffMs, kDiffMs, kFlexFecSsrc));
+
+  EXPECT_CALL(send_packet_observer_, OnSendPacket(1, capture_time_ms, kSsrc));
+
+  sender->SendPacket(video_packet.get(), PacedPacketInfo());
+
+  // Send packet observer not called for padding/retransmissions.
+  EXPECT_CALL(send_packet_observer_, OnSendPacket(2, _, _)).Times(0);
+  sender->SendPacket(rtx_packet.get(), PacedPacketInfo());
+
+  EXPECT_CALL(send_packet_observer_,
+              OnSendPacket(3, capture_time_ms, kFlexFecSsrc));
+  sender->SendPacket(fec_packet.get(), PacedPacketInfo());
+
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+  StreamDataCounters rtp_stats;
+  StreamDataCounters rtx_stats;
+  sender->GetDataCounters(&rtp_stats, &rtx_stats);
+  EXPECT_EQ(rtp_stats.transmitted.packets, 2u);
+  EXPECT_EQ(rtp_stats.fec.packets, 1u);
+  EXPECT_EQ(rtx_stats.retransmitted.packets, 1u);
+}
+
+TEST_F(RtpSenderEgressTest, TransportFeedbackObserverWithRetransmission) {
+  const uint16_t kTransportSequenceNumber = 17;
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+  std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
+  retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
+  retransmission->SetExtension<TransportSequenceNumber>(
+      kTransportSequenceNumber);
+  uint16_t retransmitted_seq = retransmission->SequenceNumber() - 2;
+  retransmission->set_retransmitted_sequence_number(retransmitted_seq);
+
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  EXPECT_CALL(
+      feedback_observer_,
+      OnAddPacket(AllOf(
+          Field(&RtpPacketSendInfo::media_ssrc, kSsrc),
+          Field(&RtpPacketSendInfo::rtp_sequence_number, retransmitted_seq),
+          Field(&RtpPacketSendInfo::transport_sequence_number,
+                kTransportSequenceNumber))));
+  sender->SendPacket(retransmission.get(), PacedPacketInfo());
+}
+
+TEST_F(RtpSenderEgressTest, TransportFeedbackObserverWithRtxRetransmission) {
+  const uint16_t kTransportSequenceNumber = 17;
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+
+  std::unique_ptr<RtpPacketToSend> rtx_retransmission = BuildRtpPacket();
+  rtx_retransmission->SetSsrc(kRtxSsrc);
+  rtx_retransmission->SetExtension<TransportSequenceNumber>(
+      kTransportSequenceNumber);
+  rtx_retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
+  uint16_t rtx_retransmitted_seq = rtx_retransmission->SequenceNumber() - 2;
+  rtx_retransmission->set_retransmitted_sequence_number(rtx_retransmitted_seq);
+
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  EXPECT_CALL(
+      feedback_observer_,
+      OnAddPacket(AllOf(
+          Field(&RtpPacketSendInfo::media_ssrc, kSsrc),
+          Field(&RtpPacketSendInfo::rtp_sequence_number, rtx_retransmitted_seq),
+          Field(&RtpPacketSendInfo::transport_sequence_number,
+                kTransportSequenceNumber))));
+  sender->SendPacket(rtx_retransmission.get(), PacedPacketInfo());
+}
+
+TEST_F(RtpSenderEgressTest, TransportFeedbackObserverPadding) {
+  const uint16_t kTransportSequenceNumber = 17;
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+  std::unique_ptr<RtpPacketToSend> padding = BuildRtpPacket();
+  padding->SetPadding(224);
+  padding->set_packet_type(RtpPacketMediaType::kPadding);
+  padding->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  EXPECT_CALL(
+      feedback_observer_,
+      OnAddPacket(AllOf(Field(&RtpPacketSendInfo::media_ssrc, absl::nullopt),
+                        Field(&RtpPacketSendInfo::transport_sequence_number,
+                              kTransportSequenceNumber))));
+  sender->SendPacket(padding.get(), PacedPacketInfo());
+}
+
+TEST_F(RtpSenderEgressTest, TransportFeedbackObserverRtxPadding) {
+  const uint16_t kTransportSequenceNumber = 17;
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+
+  std::unique_ptr<RtpPacketToSend> rtx_padding = BuildRtpPacket();
+  rtx_padding->SetPadding(224);
+  rtx_padding->SetSsrc(kRtxSsrc);
+  rtx_padding->set_packet_type(RtpPacketMediaType::kPadding);
+  rtx_padding->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  EXPECT_CALL(
+      feedback_observer_,
+      OnAddPacket(AllOf(Field(&RtpPacketSendInfo::media_ssrc, absl::nullopt),
+                        Field(&RtpPacketSendInfo::transport_sequence_number,
+                              kTransportSequenceNumber))));
+  sender->SendPacket(rtx_padding.get(), PacedPacketInfo());
+}
+
+TEST_F(RtpSenderEgressTest, TransportFeedbackObserverFec) {
+  const uint16_t kTransportSequenceNumber = 17;
+  header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
+                                   TransportSequenceNumber::Uri());
+
+  std::unique_ptr<RtpPacketToSend> fec_packet = BuildRtpPacket();
+  fec_packet->SetSsrc(kFlexFecSsrc);
+  fec_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
+  fec_packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+
+  const rtc::ArrayView<const RtpExtensionSize> kNoRtpHeaderExtensionSizes;
+  FlexfecSender flexfec(kFlexfectPayloadType, kFlexFecSsrc, kSsrc, /*mid=*/"",
+                        /*header_extensions=*/{}, kNoRtpHeaderExtensionSizes,
+                        /*rtp_state=*/nullptr, time_controller_.GetClock());
+  RtpRtcpInterface::Configuration config = DefaultConfig();
+  config.fec_generator = &flexfec;
+  auto sender = std::make_unique<RtpSenderEgress>(config, &packet_history_);
+  EXPECT_CALL(
+      feedback_observer_,
+      OnAddPacket(AllOf(Field(&RtpPacketSendInfo::media_ssrc, absl::nullopt),
+                        Field(&RtpPacketSendInfo::transport_sequence_number,
+                              kTransportSequenceNumber))));
+  sender->SendPacket(fec_packet.get(), PacedPacketInfo());
+}
+
+TEST_F(RtpSenderEgressTest, SupportsAbortingRetransmissions) {
+  std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
+  packet_history_.SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 10);
+
+  // Create a packet and send it so it is put in the history.
+  std::unique_ptr<RtpPacketToSend> media_packet = BuildRtpPacket();
+  media_packet->set_packet_type(RtpPacketMediaType::kVideo);
+  media_packet->set_allow_retransmission(true);
+  const uint16_t media_sequence_number = media_packet->SequenceNumber();
+  sender->SendPacket(media_packet.get(), PacedPacketInfo());
+
+  // Fetch a retranmission packet from the history, this should mark the
+  // media packets as pending so it is not available to grab again.
+  std::unique_ptr<RtpPacketToSend> retransmission_packet =
+      packet_history_.GetPacketAndMarkAsPending(media_sequence_number);
+  ASSERT_TRUE(retransmission_packet);
+  EXPECT_FALSE(
+      packet_history_.GetPacketAndMarkAsPending(media_sequence_number));
+
+  // Mark retransmission as aborted, fetching packet is possible again.
+  retransmission_packet.reset();
+  uint16_t kAbortedSequenceNumbers[] = {media_sequence_number};
+  sender->OnAbortedRetransmissions(kAbortedSequenceNumbers);
+  EXPECT_TRUE(packet_history_.GetPacketAndMarkAsPending(media_sequence_number));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
new file mode 100644
index 0000000000..432c5e72d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -0,0 +1,1373 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sender.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video/video_timing.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/rtp_rtcp/include/rtp_cvo.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/packet_sequencer.h"
+#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
+#include "modules/rtp_rtcp/source/video_fec_generator.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/rate_limiter.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
+#include "test/scoped_key_value_config.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+
+namespace {
+enum : int {  // The first valid value is 1.
+  kAbsoluteSendTimeExtensionId = 1,
+  kAudioLevelExtensionId,
+  kGenericDescriptorId,
+  kMidExtensionId,
+  kRepairedRidExtensionId,
+  kRidExtensionId,
+  kTransmissionTimeOffsetExtensionId,
+  kTransportSequenceNumberExtensionId,
+  kVideoRotationExtensionId,
+  kVideoTimingExtensionId,
+};
+
+const int kPayload = 100;
+const int kRtxPayload = 98;
+const uint32_t kTimestamp = 10;
+const uint16_t kSeqNum = 33;
+const uint32_t kSsrc = 725242;
+const uint32_t kRtxSsrc = 12345;
+const uint32_t kFlexFecSsrc = 45678;
+const uint64_t kStartTime = 123456789;
+const size_t kMaxPaddingSize = 224u;
+const uint8_t kPayloadData[] = {47, 11, 32, 93, 89};
+const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
+const size_t kMaxPaddingLength = 224;      // Value taken from rtp_sender.cc.
+const uint32_t kTimestampTicksPerMs = 90;  // 90kHz clock.
+constexpr absl::string_view kMid = "mid";
+constexpr absl::string_view kRid = "f";
+constexpr bool kMarkerBit = true;
+
+using ::testing::_;
+using ::testing::AllOf;
+using ::testing::AtLeast;
+using ::testing::Contains;
+using ::testing::Each;
+using ::testing::ElementsAre;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::Gt;
+using ::testing::IsEmpty;
+using ::testing::NiceMock;
+using ::testing::Not;
+using ::testing::Pointee;
+using ::testing::Property;
+using ::testing::Return;
+using ::testing::SizeIs;
+
+class MockRtpPacketPacer : public RtpPacketSender {
+ public:
+  MockRtpPacketPacer() {}
+  virtual ~MockRtpPacketPacer() {}
+
+  MOCK_METHOD(void,
+              EnqueuePackets,
+              (std::vector<std::unique_ptr<RtpPacketToSend>>),
+              (override));
+  MOCK_METHOD(void, RemovePacketsForSsrc, (uint32_t), (override));
+};
+
+}  // namespace
+
+class RtpSenderTest : public ::testing::Test {
+ protected:
+  RtpSenderTest()
+      : time_controller_(Timestamp::Millis(kStartTime)),
+        clock_(time_controller_.GetClock()),
+        retransmission_rate_limiter_(clock_, 1000),
+        flexfec_sender_(0,
+                        kFlexFecSsrc,
+                        kSsrc,
+                        "",
+                        std::vector<RtpExtension>(),
+                        std::vector<RtpExtensionSize>(),
+                        nullptr,
+                        clock_) {}
+
+  void SetUp() override { SetUpRtpSender(true, false, nullptr); }
+
+  void SetUpRtpSender(bool populate_network2,
+                      bool always_send_mid_and_rid,
+                      VideoFecGenerator* fec_generator) {
+    RtpRtcpInterface::Configuration config = GetDefaultConfig();
+    config.fec_generator = fec_generator;
+    config.populate_network2_timestamp = populate_network2;
+    config.always_send_mid_and_rid = always_send_mid_and_rid;
+    CreateSender(config);
+  }
+
+  RtpRtcpInterface::Configuration GetDefaultConfig() {
+    RtpRtcpInterface::Configuration config;
+    config.clock = clock_;
+    config.local_media_ssrc = kSsrc;
+    config.rtx_send_ssrc = kRtxSsrc;
+    config.event_log = &mock_rtc_event_log_;
+    config.retransmission_rate_limiter = &retransmission_rate_limiter_;
+    config.paced_sender = &mock_paced_sender_;
+    config.field_trials = &field_trials_;
+    // Configure rid unconditionally, it has effect only if
+    // corresponding header extension is enabled.
+    config.rid = std::string(kRid);
+    return config;
+  }
+
+  void CreateSender(const RtpRtcpInterface::Configuration& config) {
+    packet_history_ = std::make_unique<RtpPacketHistory>(
+        config.clock, config.enable_rtx_padding_prioritization);
+    sequencer_.emplace(kSsrc, kRtxSsrc,
+                       /*require_marker_before_media_padding=*/!config.audio,
+                       clock_);
+    rtp_sender_ = std::make_unique<RTPSender>(config, packet_history_.get(),
+                                              config.paced_sender);
+    sequencer_->set_media_sequence_number(kSeqNum);
+    rtp_sender_->SetTimestampOffset(0);
+  }
+
+  GlobalSimulatedTimeController time_controller_;
+  Clock* const clock_;
+  NiceMock<MockRtcEventLog> mock_rtc_event_log_;
+  MockRtpPacketPacer mock_paced_sender_;
+  RateLimiter retransmission_rate_limiter_;
+  FlexfecSender flexfec_sender_;
+
+  absl::optional<PacketSequencer> sequencer_;
+  std::unique_ptr<RtpPacketHistory> packet_history_;
+  std::unique_ptr<RTPSender> rtp_sender_;
+
+  const test::ScopedKeyValueConfig field_trials_;
+
+  std::unique_ptr<RtpPacketToSend> BuildRtpPacket(int payload_type,
+                                                  bool marker_bit,
+                                                  uint32_t timestamp,
+                                                  int64_t capture_time_ms) {
+    auto packet = rtp_sender_->AllocatePacket();
+    packet->SetPayloadType(payload_type);
+    packet->set_packet_type(RtpPacketMediaType::kVideo);
+    packet->SetMarker(marker_bit);
+    packet->SetTimestamp(timestamp);
+    packet->set_capture_time(Timestamp::Millis(capture_time_ms));
+    return packet;
+  }
+
+  std::unique_ptr<RtpPacketToSend> SendPacket(int64_t capture_time_ms,
+                                              int payload_length) {
+    uint32_t timestamp = capture_time_ms * 90;
+    auto packet =
+        BuildRtpPacket(kPayload, kMarkerBit, timestamp, capture_time_ms);
+    packet->AllocatePayload(payload_length);
+    packet->set_allow_retransmission(true);
+
+    // Packet should be stored in a send bucket.
+    EXPECT_TRUE(
+        rtp_sender_->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
+    return packet;
+  }
+
+  std::unique_ptr<RtpPacketToSend> SendGenericPacket() {
+    const int64_t kCaptureTimeMs = clock_->TimeInMilliseconds();
+    // Use maximum allowed size to catch corner cases when packet is dropped
+    // because of lack of capacity for the media packet, or for an rtx packet
+    // containing the media packet.
+    return SendPacket(kCaptureTimeMs,
+                      /*payload_length=*/rtp_sender_->MaxRtpPacketSize() -
+                          rtp_sender_->ExpectedPerPacketOverhead());
+  }
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      size_t target_size_bytes) {
+    return rtp_sender_->GeneratePadding(
+        target_size_bytes, /*media_has_been_sent=*/true,
+        sequencer_->CanSendPaddingOnMediaSsrc());
+  }
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> Sequence(
+      std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+    for (auto& packet : packets) {
+      sequencer_->Sequence(*packet);
+    }
+    return packets;
+  }
+
+  size_t GenerateAndSendPadding(size_t target_size_bytes) {
+    size_t generated_bytes = 0;
+    for (auto& packet : GeneratePadding(target_size_bytes)) {
+      generated_bytes += packet->payload_size() + packet->padding_size();
+      rtp_sender_->SendToNetwork(std::move(packet));
+    }
+    return generated_bytes;
+  }
+
+  // The following are helpers for configuring the RTPSender. They must be
+  // called before sending any packets.
+
+  // Enable the retransmission stream with sizable packet storage.
+  void EnableRtx() {
+    // RTX needs to be able to read the source packets from the packet store.
+    // Pick a number of packets to store big enough for any unit test.
+    constexpr uint16_t kNumberOfPacketsToStore = 100;
+    packet_history_->SetStorePacketsStatus(
+        RtpPacketHistory::StorageMode::kStoreAndCull, kNumberOfPacketsToStore);
+    rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
+    rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+  }
+
+  // Enable sending of the MID header extension for both the primary SSRC and
+  // the RTX SSRC.
+  void EnableMidSending(absl::string_view mid) {
+    rtp_sender_->RegisterRtpHeaderExtension(RtpMid::Uri(), kMidExtensionId);
+    rtp_sender_->SetMid(mid);
+  }
+
+  // Enable sending of the RSID header extension for the primary SSRC and the
+  // RRSID header extension for the RTX SSRC.
+  void EnableRidSending() {
+    rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::Uri(),
+                                            kRidExtensionId);
+    rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(),
+                                            kRepairedRidExtensionId);
+  }
+};
+
+TEST_F(RtpSenderTest, AllocatePacketSetCsrc) {
+  // Configure rtp_sender with csrc.
+  std::vector<uint32_t> csrcs;
+  csrcs.push_back(0x23456789);
+  rtp_sender_->SetCsrcs(csrcs);
+
+  auto packet = rtp_sender_->AllocatePacket();
+
+  ASSERT_TRUE(packet);
+  EXPECT_EQ(rtp_sender_->SSRC(), packet->Ssrc());
+  EXPECT_EQ(csrcs, packet->Csrcs());
+}
+
+TEST_F(RtpSenderTest, AllocatePacketReserveExtensions) {
+  // Configure rtp_sender with extensions.
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransmissionOffset::Uri(), kTransmissionTimeOffsetExtensionId));
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(AudioLevel::Uri(),
+                                                      kAudioLevelExtensionId));
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      VideoOrientation::Uri(), kVideoRotationExtensionId));
+
+  auto packet = rtp_sender_->AllocatePacket();
+
+  ASSERT_TRUE(packet);
+  // Preallocate BWE extensions RtpSender set itself.
+  EXPECT_TRUE(packet->HasExtension<TransmissionOffset>());
+  EXPECT_TRUE(packet->HasExtension<AbsoluteSendTime>());
+  EXPECT_TRUE(packet->HasExtension<TransportSequenceNumber>());
+  // Do not allocate media specific extensions.
+  EXPECT_FALSE(packet->HasExtension<AudioLevel>());
+  EXPECT_FALSE(packet->HasExtension<VideoOrientation>());
+}
+
+TEST_F(RtpSenderTest, PaddingAlwaysAllowedOnAudio) {
+  RtpRtcpInterface::Configuration config = GetDefaultConfig();
+  config.audio = true;
+  CreateSender(config);
+
+  std::unique_ptr<RtpPacketToSend> audio_packet = rtp_sender_->AllocatePacket();
+  // Padding on audio stream allowed regardless of marker in the last packet.
+  audio_packet->SetMarker(false);
+  audio_packet->SetPayloadType(kPayload);
+  sequencer_->Sequence(*audio_packet);
+
+  const size_t kPaddingSize = 59;
+
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(AllOf(
+          Pointee(Property(&RtpPacketToSend::packet_type,
+                           RtpPacketMediaType::kPadding)),
+          Pointee(Property(&RtpPacketToSend::padding_size, kPaddingSize))))));
+  EXPECT_EQ(kPaddingSize, GenerateAndSendPadding(kPaddingSize));
+
+  // Requested padding size is too small, will send a larger one.
+  const size_t kMinPaddingSize = 50;
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(
+                  AllOf(Pointee(Property(&RtpPacketToSend::packet_type,
+                                         RtpPacketMediaType::kPadding)),
+                        Pointee(Property(&RtpPacketToSend::padding_size,
+                                         kMinPaddingSize))))));
+  EXPECT_EQ(kMinPaddingSize, GenerateAndSendPadding(kMinPaddingSize - 5));
+}
+
+TEST_F(RtpSenderTest, SendToNetworkForwardsPacketsToPacer) {
+  auto packet = BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, 0);
+  Timestamp now = clock_->CurrentTime();
+
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(AllOf(
+                  Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
+                  Pointee(Property(&RtpPacketToSend::capture_time, now))))));
+  EXPECT_TRUE(
+      rtp_sender_->SendToNetwork(std::make_unique<RtpPacketToSend>(*packet)));
+}
+
+TEST_F(RtpSenderTest, ReSendPacketForwardsPacketsToPacer) {
+  packet_history_->SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 10);
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  auto packet = BuildRtpPacket(kPayload, kMarkerBit, kTimestamp, now_ms);
+  packet->SetSequenceNumber(kSeqNum);
+  packet->set_allow_retransmission(true);
+  packet_history_->PutRtpPacket(std::move(packet), Timestamp::Millis(now_ms));
+
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(AllOf(
+                  Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
+                  Pointee(Property(&RtpPacketToSend::SequenceNumber, kSeqNum)),
+                  Pointee(Property(&RtpPacketToSend::capture_time,
+                                   Timestamp::Millis(now_ms))),
+                  Pointee(Property(&RtpPacketToSend::packet_type,
+                                   RtpPacketMediaType::kRetransmission))))));
+  EXPECT_TRUE(rtp_sender_->ReSendPacket(kSeqNum));
+}
+
+// This test sends 1 regular video packet, then 4 padding packets, and then
+// 1 more regular packet.
+TEST_F(RtpSenderTest, SendPadding) {
+  constexpr int kNumPaddingPackets = 4;
+  EXPECT_CALL(mock_paced_sender_, EnqueuePackets);
+  std::unique_ptr<RtpPacketToSend> media_packet =
+      SendPacket(/*capture_time_ms=*/clock_->TimeInMilliseconds(),
+                 /*payload_size=*/100);
+  sequencer_->Sequence(*media_packet);
+
+  // Wait 50 ms before generating each padding packet.
+  for (int i = 0; i < kNumPaddingPackets; ++i) {
+    time_controller_.AdvanceTime(TimeDelta::Millis(50));
+    const size_t kPaddingTargetBytes = 100;  // Request 100 bytes of padding.
+
+    // Padding should be sent on the media ssrc, with a continous sequence
+    // number range. Size will be forced to full pack size and the timestamp
+    // shall be that of the last media packet.
+    EXPECT_CALL(mock_paced_sender_,
+                EnqueuePackets(ElementsAre(Pointee(AllOf(
+                    Property(&RtpPacketToSend::Ssrc, kSsrc),
+                    Property(&RtpPacketToSend::padding_size, kMaxPaddingLength),
+                    Property(&RtpPacketToSend::SequenceNumber,
+                             media_packet->SequenceNumber() + i + 1),
+                    Property(&RtpPacketToSend::Timestamp,
+                             media_packet->Timestamp()))))));
+    std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
+        Sequence(GeneratePadding(kPaddingTargetBytes));
+    ASSERT_THAT(padding_packets, SizeIs(1));
+    rtp_sender_->SendToNetwork(std::move(padding_packets[0]));
+  }
+
+  // Send a regular video packet again.
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(Pointee(Property(
+          &RtpPacketToSend::Timestamp, Gt(media_packet->Timestamp()))))));
+
+  std::unique_ptr<RtpPacketToSend> next_media_packet =
+      SendPacket(/*capture_time_ms=*/clock_->TimeInMilliseconds(),
+                 /*payload_size=*/100);
+}
+
+TEST_F(RtpSenderTest, NoPaddingAsFirstPacketWithoutBweExtensions) {
+  EXPECT_THAT(rtp_sender_->GeneratePadding(
+                  /*target_size_bytes=*/100,
+                  /*media_has_been_sent=*/false,
+                  /*can_send_padding_on_media_ssrc=*/false),
+              IsEmpty());
+
+  // Don't send padding before media even with RTX.
+  EnableRtx();
+  EXPECT_THAT(rtp_sender_->GeneratePadding(
+                  /*target_size_bytes=*/100,
+                  /*media_has_been_sent=*/false,
+                  /*can_send_padding_on_media_ssrc=*/false),
+              IsEmpty());
+}
+
+TEST_F(RtpSenderTest, RequiresRtxSsrcToEnableRtx) {
+  RtpRtcpInterface::Configuration config = GetDefaultConfig();
+  config.rtx_send_ssrc = absl::nullopt;
+  RTPSender rtp_sender(config, packet_history_.get(), config.paced_sender);
+  rtp_sender.SetRtxPayloadType(kRtxPayload, kPayload);
+
+  rtp_sender.SetRtxStatus(kRtxRetransmitted);
+
+  EXPECT_EQ(rtp_sender.RtxStatus(), kRtxOff);
+}
+
+TEST_F(RtpSenderTest, RequiresRtxPayloadTypesToEnableRtx) {
+  RtpRtcpInterface::Configuration config = GetDefaultConfig();
+  config.rtx_send_ssrc = kRtxSsrc;
+  RTPSender rtp_sender(config, packet_history_.get(), config.paced_sender);
+
+  rtp_sender.SetRtxStatus(kRtxRetransmitted);
+
+  EXPECT_EQ(rtp_sender.RtxStatus(), kRtxOff);
+}
+
+TEST_F(RtpSenderTest, CanEnableRtxWhenRtxSsrcAndPayloadTypeAreConfigured) {
+  RtpRtcpInterface::Configuration config = GetDefaultConfig();
+  config.rtx_send_ssrc = kRtxSsrc;
+  RTPSender rtp_sender(config, packet_history_.get(), config.paced_sender);
+  rtp_sender.SetRtxPayloadType(kRtxPayload, kPayload);
+
+  ASSERT_EQ(rtp_sender.RtxStatus(), kRtxOff);
+  rtp_sender.SetRtxStatus(kRtxRetransmitted);
+
+  EXPECT_EQ(rtp_sender.RtxStatus(), kRtxRetransmitted);
+}
+
+TEST_F(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithTransportCc) {
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
+
+  // Padding can't be sent as first packet on media SSRC since we don't know
+  // what payload type to assign.
+  EXPECT_THAT(rtp_sender_->GeneratePadding(
+                  /*target_size_bytes=*/100,
+                  /*media_has_been_sent=*/false,
+                  /*can_send_padding_on_media_ssrc=*/false),
+              IsEmpty());
+
+  // With transportcc padding can be sent as first packet on the RTX SSRC.
+  EnableRtx();
+  EXPECT_THAT(rtp_sender_->GeneratePadding(
+                  /*target_size_bytes=*/100,
+                  /*media_has_been_sent=*/false,
+                  /*can_send_padding_on_media_ssrc=*/false),
+              Not(IsEmpty()));
+}
+
+TEST_F(RtpSenderTest, AllowPaddingAsFirstPacketOnRtxWithAbsSendTime) {
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
+
+  // Padding can't be sent as first packet on media SSRC since we don't know
+  // what payload type to assign.
+  EXPECT_THAT(rtp_sender_->GeneratePadding(
+                  /*target_size_bytes=*/100,
+                  /*media_has_been_sent=*/false,
+                  /*can_send_padding_on_media_ssrc=*/false),
+              IsEmpty());
+
+  // With abs send time, padding can be sent as first packet on the RTX SSRC.
+  EnableRtx();
+  EXPECT_THAT(rtp_sender_->GeneratePadding(
+                  /*target_size_bytes=*/100,
+                  /*media_has_been_sent=*/false,
+                  /*can_send_padding_on_media_ssrc=*/false),
+              Not(IsEmpty()));
+}
+
+TEST_F(RtpSenderTest, UpdatesTimestampsOnPlainRtxPadding) {
+  EnableRtx();
+  // Timestamps as set based on capture time in RtpSenderTest.
+  const int64_t start_time = clock_->TimeInMilliseconds();
+  const uint32_t start_timestamp = start_time * kTimestampTicksPerMs;
+
+  // Start by sending one media packet.
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(
+          AllOf(Pointee(Property(&RtpPacketToSend::padding_size, 0u)),
+                Pointee(Property(&RtpPacketToSend::Timestamp, start_timestamp)),
+                Pointee(Property(&RtpPacketToSend::capture_time,
+                                 Timestamp::Millis(start_time)))))));
+  std::unique_ptr<RtpPacketToSend> media_packet =
+      SendPacket(start_time, /*payload_size=*/600);
+  sequencer_->Sequence(*media_packet);
+
+  // Advance time before sending padding.
+  const TimeDelta kTimeDiff = TimeDelta::Millis(17);
+  time_controller_.AdvanceTime(kTimeDiff);
+
+  // Timestamps on padding should be offset from the sent media.
+  EXPECT_THAT(
+      Sequence(GeneratePadding(/*target_size_bytes=*/100)),
+      Each(Pointee(AllOf(
+          Property(&RtpPacketToSend::padding_size, kMaxPaddingLength),
+          Property(&RtpPacketToSend::Timestamp,
+                   start_timestamp + (kTimestampTicksPerMs * kTimeDiff.ms())),
+          Property(&RtpPacketToSend::capture_time,
+                   Timestamp::Millis(start_time) + kTimeDiff)))));
+}
+
+TEST_F(RtpSenderTest, KeepsTimestampsOnPayloadPadding) {
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
+  EnableRtx();
+  // Timestamps as set based on capture time in RtpSenderTest.
+  const int64_t start_time = clock_->TimeInMilliseconds();
+  const uint32_t start_timestamp = start_time * kTimestampTicksPerMs;
+  const size_t kPayloadSize = 200;
+  const size_t kRtxHeaderSize = 2;
+
+  // Start by sending one media packet and putting in the packet history.
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(
+          AllOf(Pointee(Property(&RtpPacketToSend::padding_size, 0u)),
+                Pointee(Property(&RtpPacketToSend::Timestamp, start_timestamp)),
+                Pointee(Property(&RtpPacketToSend::capture_time,
+                                 Timestamp::Millis(start_time)))))));
+  std::unique_ptr<RtpPacketToSend> media_packet =
+      SendPacket(start_time, kPayloadSize);
+  packet_history_->PutRtpPacket(std::move(media_packet),
+                                Timestamp::Millis(start_time));
+
+  // Advance time before sending padding.
+  const TimeDelta kTimeDiff = TimeDelta::Millis(17);
+  time_controller_.AdvanceTime(kTimeDiff);
+
+  // Timestamps on payload padding should be set to original.
+  EXPECT_THAT(GeneratePadding(/*target_size_bytes=*/100),
+              Each(AllOf(Pointee(Property(&RtpPacketToSend::padding_size, 0u)),
+                         Pointee(Property(&RtpPacketToSend::payload_size,
+                                          kPayloadSize + kRtxHeaderSize)),
+                         Pointee(Property(&RtpPacketToSend::Timestamp,
+                                          start_timestamp)),
+                         Pointee(Property(&RtpPacketToSend::capture_time,
+                                          Timestamp::Millis(start_time))))));
+}
+
+// Test that the MID header extension is included on sent packets when
+// configured.
+TEST_F(RtpSenderTest, MidIncludedOnSentPackets) {
+  EnableMidSending(kMid);
+
+  // Send a couple packets, expect both packets to have the MID set.
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(Pointee(
+                  Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid)))))
+      .Times(2);
+  SendGenericPacket();
+  SendGenericPacket();
+}
+
+TEST_F(RtpSenderTest, RidIncludedOnSentPackets) {
+  EnableRidSending();
+
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(Pointee(Property(
+                  &RtpPacketToSend::GetExtension<RtpStreamId>, kRid)))));
+  SendGenericPacket();
+}
+
+TEST_F(RtpSenderTest, RidIncludedOnRtxSentPackets) {
+  EnableRtx();
+  EnableRidSending();
+
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(Pointee(AllOf(
+                  Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid),
+                  Property(&RtpPacketToSend::HasExtension<RepairedRtpStreamId>,
+                           false))))))
+      .WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+        sequencer_->Sequence(*packets[0]);
+        packet_history_->PutRtpPacket(std::move(packets[0]),
+                                      clock_->CurrentTime());
+      });
+  SendGenericPacket();
+
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(Pointee(AllOf(
+          Property(&RtpPacketToSend::GetExtension<RepairedRtpStreamId>, kRid),
+          Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
+  rtp_sender_->ReSendPacket(kSeqNum);
+}
+
+TEST_F(RtpSenderTest, MidAndRidNotIncludedOnSentPacketsAfterAck) {
+  EnableMidSending(kMid);
+  EnableRidSending();
+
+  // This first packet should include both MID and RID.
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(Pointee(AllOf(
+          Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
+          Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid))))));
+  auto first_built_packet = SendGenericPacket();
+  rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
+
+  // The second packet should include neither since an ack was received.
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(Pointee(AllOf(
+          Property(&RtpPacketToSend::HasExtension<RtpMid>, false),
+          Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
+  SendGenericPacket();
+}
+
+TEST_F(RtpSenderTest, MidAndRidAlwaysIncludedOnSentPacketsWhenConfigured) {
+  SetUpRtpSender(false, /*always_send_mid_and_rid=*/true, nullptr);
+  EnableMidSending(kMid);
+  EnableRidSending();
+
+  // Send two media packets: one before and one after the ack.
+  // Due to the configuration, both sent packets should contain MID and RID.
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(Pointee(
+          AllOf(Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
+                Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid))))))
+      .Times(2);
+  auto first_built_packet = SendGenericPacket();
+  rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
+  SendGenericPacket();
+}
+
+// Test that the first RTX packet includes both MID and RRID even if the packet
+// being retransmitted did not have MID or RID. The MID and RID are needed on
+// the first packets for a given SSRC, and RTX packets are sent on a separate
+// SSRC.
+TEST_F(RtpSenderTest, MidAndRidIncludedOnFirstRtxPacket) {
+  EnableRtx();
+  EnableMidSending(kMid);
+  EnableRidSending();
+
+  // This first packet will include both MID and RID.
+  EXPECT_CALL(mock_paced_sender_, EnqueuePackets);
+  auto first_built_packet = SendGenericPacket();
+  rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
+
+  // The second packet will include neither since an ack was received, put
+  // it in the packet history for retransmission.
+  EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
+      .WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+        packet_history_->PutRtpPacket(std::move(packets[0]),
+                                      clock_->CurrentTime());
+      });
+  auto second_built_packet = SendGenericPacket();
+
+  // The first RTX packet should include MID and RRID.
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(Pointee(AllOf(
+                  Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
+                  Property(&RtpPacketToSend::GetExtension<RepairedRtpStreamId>,
+                           kRid))))));
+  rtp_sender_->ReSendPacket(second_built_packet->SequenceNumber());
+}
+
+// Test that the RTX packets sent after receving an ACK on the RTX SSRC does
+// not include either MID or RRID even if the packet being retransmitted did
+// had a MID or RID.
+TEST_F(RtpSenderTest, MidAndRidNotIncludedOnRtxPacketsAfterAck) {
+  EnableRtx();
+  EnableMidSending(kMid);
+  EnableRidSending();
+
+  // This first packet will include both MID and RID.
+  auto first_built_packet = SendGenericPacket();
+  sequencer_->Sequence(*first_built_packet);
+  packet_history_->PutRtpPacket(
+      std::make_unique<RtpPacketToSend>(*first_built_packet),
+      /*send_time=*/clock_->CurrentTime());
+  rtp_sender_->OnReceivedAckOnSsrc(first_built_packet->SequenceNumber());
+
+  // The second packet will include neither since an ack was received.
+  auto second_built_packet = SendGenericPacket();
+  sequencer_->Sequence(*second_built_packet);
+  packet_history_->PutRtpPacket(
+      std::make_unique<RtpPacketToSend>(*second_built_packet),
+      /*send_time=*/clock_->CurrentTime());
+
+  // The first RTX packet will include MID and RRID.
+  EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
+      .WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+        rtp_sender_->OnReceivedAckOnRtxSsrc(packets[0]->SequenceNumber());
+        packet_history_->MarkPacketAsSent(
+            *packets[0]->retransmitted_sequence_number());
+      });
+  rtp_sender_->ReSendPacket(second_built_packet->SequenceNumber());
+
+  // The second and third RTX packets should not include MID nor RRID.
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(Pointee(AllOf(
+                  Property(&RtpPacketToSend::HasExtension<RtpMid>, false),
+                  Property(&RtpPacketToSend::HasExtension<RepairedRtpStreamId>,
+                           false))))))
+      .Times(2);
+  rtp_sender_->ReSendPacket(first_built_packet->SequenceNumber());
+  rtp_sender_->ReSendPacket(second_built_packet->SequenceNumber());
+}
+
+TEST_F(RtpSenderTest, MidAndRidAlwaysIncludedOnRtxPacketsWhenConfigured) {
+  SetUpRtpSender(false, /*always_send_mid_and_rid=*/true, nullptr);
+  EnableRtx();
+  EnableMidSending(kMid);
+  EnableRidSending();
+
+  // Send two media packets: one before and one after the ack.
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(Pointee(
+          AllOf(Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
+                Property(&RtpPacketToSend::GetExtension<RtpStreamId>, kRid))))))
+      .Times(2)
+      .WillRepeatedly(
+          [&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+            packet_history_->PutRtpPacket(std::move(packets[0]),
+                                          clock_->CurrentTime());
+          });
+  auto media_packet1 = SendGenericPacket();
+  rtp_sender_->OnReceivedAckOnSsrc(media_packet1->SequenceNumber());
+  auto media_packet2 = SendGenericPacket();
+
+  // Send three RTX packets with different combinations of orders w.r.t. the
+  // media and RTX acks.
+  // Due to the configuration, all sent packets should contain MID
+  // and either RID (media) or RRID (RTX).
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(Pointee(AllOf(
+                  Property(&RtpPacketToSend::GetExtension<RtpMid>, kMid),
+                  Property(&RtpPacketToSend::GetExtension<RepairedRtpStreamId>,
+                           kRid))))))
+      .Times(3)
+      .WillRepeatedly(
+          [&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+            rtp_sender_->OnReceivedAckOnRtxSsrc(packets[0]->SequenceNumber());
+            packet_history_->MarkPacketAsSent(
+                *packets[0]->retransmitted_sequence_number());
+          });
+  rtp_sender_->ReSendPacket(media_packet2->SequenceNumber());
+  rtp_sender_->ReSendPacket(media_packet1->SequenceNumber());
+  rtp_sender_->ReSendPacket(media_packet2->SequenceNumber());
+}
+
+// Test that if the RtpState indicates an ACK has been received on that SSRC
+// then neither the MID nor RID header extensions will be sent.
+TEST_F(RtpSenderTest, MidAndRidNotIncludedOnSentPacketsAfterRtpStateRestored) {
+  EnableMidSending(kMid);
+  EnableRidSending();
+
+  RtpState state = rtp_sender_->GetRtpState();
+  EXPECT_FALSE(state.ssrc_has_acked);
+  state.ssrc_has_acked = true;
+  rtp_sender_->SetRtpState(state);
+
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(Pointee(AllOf(
+          Property(&RtpPacketToSend::HasExtension<RtpMid>, false),
+          Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
+  SendGenericPacket();
+}
+
+// Test that if the RTX RtpState indicates an ACK has been received on that
+// RTX SSRC then neither the MID nor RRID header extensions will be sent on
+// RTX packets.
+TEST_F(RtpSenderTest, MidAndRridNotIncludedOnRtxPacketsAfterRtpStateRestored) {
+  EnableRtx();
+  EnableMidSending(kMid);
+  EnableRidSending();
+
+  RtpState rtx_state = rtp_sender_->GetRtxRtpState();
+  EXPECT_FALSE(rtx_state.ssrc_has_acked);
+  rtx_state.ssrc_has_acked = true;
+  rtp_sender_->SetRtxRtpState(rtx_state);
+
+  EXPECT_CALL(mock_paced_sender_, EnqueuePackets(SizeIs(1)))
+      .WillOnce([&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+        packet_history_->PutRtpPacket(std::move(packets[0]),
+                                      clock_->CurrentTime());
+      });
+  auto built_packet = SendGenericPacket();
+
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(Pointee(AllOf(
+          Property(&RtpPacketToSend::HasExtension<RtpMid>, false),
+          Property(&RtpPacketToSend::HasExtension<RtpStreamId>, false))))));
+  ASSERT_LT(0, rtp_sender_->ReSendPacket(built_packet->SequenceNumber()));
+}
+
+TEST_F(RtpSenderTest, RespectsNackBitrateLimit) {
+  const int32_t kPacketSize = 1400;
+  const int32_t kNumPackets = 30;
+  retransmission_rate_limiter_.SetMaxRate(kPacketSize * kNumPackets * 8);
+  EnableRtx();
+
+  std::vector<uint16_t> sequence_numbers;
+  for (int32_t i = 0; i < kNumPackets; ++i) {
+    std::unique_ptr<RtpPacketToSend> packet =
+        BuildRtpPacket(kPayload, /*marker_bit=*/true, /*timestamp=*/0,
+                       /*capture_time_ms=*/clock_->TimeInMilliseconds());
+    packet->set_allow_retransmission(true);
+    sequencer_->Sequence(*packet);
+    sequence_numbers.push_back(packet->SequenceNumber());
+    packet_history_->PutRtpPacket(std::move(packet),
+                                  /*send_time=*/clock_->CurrentTime());
+    time_controller_.AdvanceTime(TimeDelta::Millis(1));
+  }
+
+  time_controller_.AdvanceTime(TimeDelta::Millis(1000 - kNumPackets));
+
+  // Resending should work - brings the bandwidth up to the limit.
+  // NACK bitrate is capped to the same bitrate as the encoder, since the max
+  // protection overhead is 50% (see MediaOptimization::SetTargetRates).
+  EXPECT_CALL(mock_paced_sender_, EnqueuePackets(ElementsAre(Pointee(Property(
+                                      &RtpPacketToSend::packet_type,
+                                      RtpPacketMediaType::kRetransmission)))))
+      .Times(kNumPackets)
+      .WillRepeatedly(
+          [&](std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+            for (const auto& packet : packets) {
+              packet_history_->MarkPacketAsSent(
+                  *packet->retransmitted_sequence_number());
+            }
+          });
+  rtp_sender_->OnReceivedNack(sequence_numbers, 0);
+
+  // Must be at least 5ms in between retransmission attempts.
+  time_controller_.AdvanceTime(TimeDelta::Millis(5));
+
+  // Resending should not work, bandwidth exceeded.
+  EXPECT_CALL(mock_paced_sender_, EnqueuePackets).Times(0);
+  rtp_sender_->OnReceivedNack(sequence_numbers, 0);
+}
+
+TEST_F(RtpSenderTest, UpdatingCsrcsUpdatedOverhead) {
+  RtpRtcpInterface::Configuration config = GetDefaultConfig();
+  config.rtx_send_ssrc = {};
+  CreateSender(config);
+
+  // Base RTP overhead is 12B.
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
+
+  // Adding two csrcs adds 2*4 bytes to the header.
+  rtp_sender_->SetCsrcs({1, 2});
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 20u);
+}
+
+TEST_F(RtpSenderTest, OnOverheadChanged) {
+  RtpRtcpInterface::Configuration config = GetDefaultConfig();
+  config.rtx_send_ssrc = {};
+  CreateSender(config);
+
+  // Base RTP overhead is 12B.
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
+
+  rtp_sender_->RegisterRtpHeaderExtension(TransmissionOffset::Uri(),
+                                          kTransmissionTimeOffsetExtensionId);
+
+  // TransmissionTimeOffset extension has a size of 3B, but with the addition
+  // of header index and rounding to 4 byte boundary we end up with 20B total.
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 20u);
+}
+
+TEST_F(RtpSenderTest, CountMidOnlyUntilAcked) {
+  RtpRtcpInterface::Configuration config = GetDefaultConfig();
+  config.rtx_send_ssrc = {};
+  CreateSender(config);
+
+  // Base RTP overhead is 12B.
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
+
+  rtp_sender_->RegisterRtpHeaderExtension(RtpMid::Uri(), kMidExtensionId);
+
+  // Counted only if set.
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
+  rtp_sender_->SetMid("foo");
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 36u);
+  rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::Uri(), kRidExtensionId);
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 52u);
+
+  // Ack received, mid/rid no longer sent.
+  rtp_sender_->OnReceivedAckOnSsrc(0);
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
+}
+
+TEST_F(RtpSenderTest, CountMidRidRridUntilAcked) {
+  RtpRtcpInterface::Configuration config = GetDefaultConfig();
+  CreateSender(config);
+
+  // Base RTP overhead is 12B and we use RTX which has an additional 2 bytes
+  // overhead.
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 14u);
+
+  rtp_sender_->RegisterRtpHeaderExtension(RtpMid::Uri(), kMidExtensionId);
+
+  // Counted only if set.
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 14u);
+  rtp_sender_->SetMid("foo");
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 38u);
+
+  rtp_sender_->RegisterRtpHeaderExtension(RtpStreamId::Uri(), kRidExtensionId);
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 54u);
+
+  // mid/rrid may be shared with mid/rid when both are active.
+  rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(),
+                                          kRepairedRidExtensionId);
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 54u);
+
+  // Ack received, mid/rid no longer sent but we still need space for
+  // mid/rrid which can no longer be shared with mid/rid.
+  rtp_sender_->OnReceivedAckOnSsrc(0);
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 54u);
+
+  // Ack received for RTX, no need to send RRID anymore.
+  rtp_sender_->OnReceivedAckOnRtxSsrc(0);
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 14u);
+}
+
+TEST_F(RtpSenderTest, DontCountVolatileExtensionsIntoOverhead) {
+  RtpRtcpInterface::Configuration config = GetDefaultConfig();
+  config.rtx_send_ssrc = {};
+  CreateSender(config);
+
+  // Base RTP overhead is 12B.
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
+
+  rtp_sender_->RegisterRtpHeaderExtension(InbandComfortNoiseExtension::Uri(),
+                                          1);
+  rtp_sender_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
+                                          2);
+  rtp_sender_->RegisterRtpHeaderExtension(VideoOrientation::Uri(), 3);
+  rtp_sender_->RegisterRtpHeaderExtension(PlayoutDelayLimits::Uri(), 4);
+  rtp_sender_->RegisterRtpHeaderExtension(VideoContentTypeExtension::Uri(), 5);
+  rtp_sender_->RegisterRtpHeaderExtension(VideoTimingExtension::Uri(), 6);
+  rtp_sender_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(), 7);
+  rtp_sender_->RegisterRtpHeaderExtension(ColorSpaceExtension::Uri(), 8);
+
+  // Still only 12B counted since can't count on above being sent.
+  EXPECT_EQ(rtp_sender_->ExpectedPerPacketOverhead(), 12u);
+}
+
+TEST_F(RtpSenderTest, SendPacketHandlesRetransmissionHistory) {
+  packet_history_->SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 10);
+
+  // Ignore calls to EnqueuePackets() for this test.
+  EXPECT_CALL(mock_paced_sender_, EnqueuePackets).WillRepeatedly(Return());
+
+  // Build a media packet and put in the packet history.
+  std::unique_ptr<RtpPacketToSend> packet =
+      BuildRtpPacket(kPayload, true, 0, clock_->TimeInMilliseconds());
+  const uint16_t media_sequence_number = packet->SequenceNumber();
+  packet->set_allow_retransmission(true);
+  packet_history_->PutRtpPacket(std::move(packet), clock_->CurrentTime());
+
+  // Simulate successful retransmission request.
+  time_controller_.AdvanceTime(TimeDelta::Millis(30));
+  EXPECT_THAT(rtp_sender_->ReSendPacket(media_sequence_number), Gt(0));
+
+  // Packet already pending, retransmission not allowed.
+  time_controller_.AdvanceTime(TimeDelta::Millis(30));
+  EXPECT_THAT(rtp_sender_->ReSendPacket(media_sequence_number), Eq(0));
+
+  // Simulate packet exiting pacer, mark as not longer pending.
+  packet_history_->MarkPacketAsSent(media_sequence_number);
+
+  // Retransmissions allowed again.
+  time_controller_.AdvanceTime(TimeDelta::Millis(30));
+  EXPECT_THAT(rtp_sender_->ReSendPacket(media_sequence_number), Gt(0));
+}
+
+TEST_F(RtpSenderTest, MarksRetransmittedPackets) {
+  packet_history_->SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 10);
+
+  // Build a media packet and put in the packet history.
+  std::unique_ptr<RtpPacketToSend> packet =
+      BuildRtpPacket(kPayload, true, 0, clock_->TimeInMilliseconds());
+  const uint16_t media_sequence_number = packet->SequenceNumber();
+  packet->set_allow_retransmission(true);
+  packet_history_->PutRtpPacket(std::move(packet), clock_->CurrentTime());
+
+  // Expect a retransmission packet marked with which packet it is a
+  // retransmit of.
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(AllOf(
+          Pointee(Property(&RtpPacketToSend::packet_type,
+                           RtpPacketMediaType::kRetransmission)),
+          Pointee(Property(&RtpPacketToSend::retransmitted_sequence_number,
+                           Eq(media_sequence_number)))))));
+  EXPECT_THAT(rtp_sender_->ReSendPacket(media_sequence_number), Gt(0));
+}
+
+TEST_F(RtpSenderTest, GeneratedPaddingHasBweExtensions) {
+  // Min requested size in order to use RTX payload.
+  const size_t kMinPaddingSize = 50;
+  EnableRtx();
+
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransmissionOffset::Uri(), kTransmissionTimeOffsetExtensionId));
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
+
+  // Put a packet in the history, in order to facilitate payload padding.
+  std::unique_ptr<RtpPacketToSend> packet =
+      BuildRtpPacket(kPayload, true, 0, clock_->TimeInMilliseconds());
+  packet->set_allow_retransmission(true);
+  packet->SetPayloadSize(kMinPaddingSize);
+  packet->set_packet_type(RtpPacketMediaType::kVideo);
+  packet_history_->PutRtpPacket(std::move(packet), clock_->CurrentTime());
+
+  // Generate a plain padding packet, check that extensions are registered.
+  std::vector<std::unique_ptr<RtpPacketToSend>> generated_packets =
+      GeneratePadding(/*target_size_bytes=*/1);
+  ASSERT_THAT(generated_packets, SizeIs(1));
+  auto& plain_padding = generated_packets.front();
+  EXPECT_GT(plain_padding->padding_size(), 0u);
+  EXPECT_TRUE(plain_padding->HasExtension<TransportSequenceNumber>());
+  EXPECT_TRUE(plain_padding->HasExtension<AbsoluteSendTime>());
+  EXPECT_TRUE(plain_padding->HasExtension<TransmissionOffset>());
+  EXPECT_GT(plain_padding->padding_size(), 0u);
+
+  // Generate a payload padding packets, check that extensions are registered.
+  generated_packets = GeneratePadding(kMinPaddingSize);
+  ASSERT_EQ(generated_packets.size(), 1u);
+  auto& payload_padding = generated_packets.front();
+  EXPECT_EQ(payload_padding->padding_size(), 0u);
+  EXPECT_TRUE(payload_padding->HasExtension<TransportSequenceNumber>());
+  EXPECT_TRUE(payload_padding->HasExtension<AbsoluteSendTime>());
+  EXPECT_TRUE(payload_padding->HasExtension<TransmissionOffset>());
+  EXPECT_GT(payload_padding->payload_size(), 0u);
+}
+
+TEST_F(RtpSenderTest, GeneratePaddingResendsOldPacketsWithRtx) {
+  // Min requested size in order to use RTX payload.
+  const size_t kMinPaddingSize = 50;
+
+  rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
+  rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+  packet_history_->SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 1);
+
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
+
+  const size_t kPayloadPacketSize = kMinPaddingSize;
+  std::unique_ptr<RtpPacketToSend> packet =
+      BuildRtpPacket(kPayload, true, 0, clock_->TimeInMilliseconds());
+  packet->set_allow_retransmission(true);
+  packet->SetPayloadSize(kPayloadPacketSize);
+  packet->set_packet_type(RtpPacketMediaType::kVideo);
+  packet_history_->PutRtpPacket(std::move(packet), clock_->CurrentTime());
+
+  // Generated padding has large enough budget that the video packet should be
+  // retransmitted as padding.
+  std::vector<std::unique_ptr<RtpPacketToSend>> generated_packets =
+      GeneratePadding(kMinPaddingSize);
+  ASSERT_EQ(generated_packets.size(), 1u);
+  auto& padding_packet = generated_packets.front();
+  EXPECT_EQ(padding_packet->packet_type(), RtpPacketMediaType::kPadding);
+  EXPECT_EQ(padding_packet->Ssrc(), kRtxSsrc);
+  EXPECT_EQ(padding_packet->payload_size(),
+            kPayloadPacketSize + kRtxHeaderSize);
+
+  // Not enough budged for payload padding, use plain padding instead.
+  const size_t kPaddingBytesRequested = kMinPaddingSize - 1;
+
+  size_t padding_bytes_generated = 0;
+  generated_packets = GeneratePadding(kPaddingBytesRequested);
+  EXPECT_EQ(generated_packets.size(), 1u);
+  for (auto& packet : generated_packets) {
+    EXPECT_EQ(packet->packet_type(), RtpPacketMediaType::kPadding);
+    EXPECT_EQ(packet->Ssrc(), kRtxSsrc);
+    EXPECT_EQ(packet->payload_size(), 0u);
+    EXPECT_GT(packet->padding_size(), 0u);
+    padding_bytes_generated += packet->padding_size();
+  }
+
+  EXPECT_EQ(padding_bytes_generated, kMaxPaddingSize);
+}
+
+TEST_F(RtpSenderTest, LimitsPayloadPaddingSize) {
+  // RTX payload padding is limited to 3x target size.
+  const double kFactor = 3.0;
+  SetUpRtpSender(false, false, nullptr);
+  rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
+  rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
+  packet_history_->SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 1);
+
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
+
+  // Send a dummy video packet so it ends up in the packet history.
+  const size_t kPayloadPacketSize = 1234u;
+  std::unique_ptr<RtpPacketToSend> packet =
+      BuildRtpPacket(kPayload, true, 0, clock_->TimeInMilliseconds());
+  packet->set_allow_retransmission(true);
+  packet->SetPayloadSize(kPayloadPacketSize);
+  packet->set_packet_type(RtpPacketMediaType::kVideo);
+  packet_history_->PutRtpPacket(std::move(packet), clock_->CurrentTime());
+
+  // Smallest target size that will result in the sent packet being returned as
+  // padding.
+  const size_t kMinTargerSizeForPayload =
+      (kPayloadPacketSize + kRtxHeaderSize) / kFactor;
+
+  // Generated padding has large enough budget that the video packet should be
+  // retransmitted as padding.
+  EXPECT_THAT(
+      GeneratePadding(kMinTargerSizeForPayload),
+      AllOf(Not(IsEmpty()),
+            Each(Pointee(Property(&RtpPacketToSend::padding_size, Eq(0u))))));
+
+  // If payload padding is > 2x requested size, plain padding is returned
+  // instead.
+  EXPECT_THAT(
+      GeneratePadding(kMinTargerSizeForPayload - 1),
+      AllOf(Not(IsEmpty()),
+            Each(Pointee(Property(&RtpPacketToSend::padding_size, Gt(0u))))));
+}
+
+TEST_F(RtpSenderTest, GeneratePaddingCreatesPurePaddingWithoutRtx) {
+  packet_history_->SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 1);
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransmissionOffset::Uri(), kTransmissionTimeOffsetExtensionId));
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      AbsoluteSendTime::Uri(), kAbsoluteSendTimeExtensionId));
+  ASSERT_TRUE(rtp_sender_->RegisterRtpHeaderExtension(
+      TransportSequenceNumber::Uri(), kTransportSequenceNumberExtensionId));
+
+  const size_t kPayloadPacketSize = 1234;
+  // Send a dummy video packet so it ends up in the packet history. Since we
+  // are not using RTX, it should never be used as padding.
+  std::unique_ptr<RtpPacketToSend> packet =
+      BuildRtpPacket(kPayload, true, 0, clock_->TimeInMilliseconds());
+  packet->set_allow_retransmission(true);
+  packet->SetPayloadSize(kPayloadPacketSize);
+  packet->set_packet_type(RtpPacketMediaType::kVideo);
+  sequencer_->Sequence(*packet);
+  packet_history_->PutRtpPacket(std::move(packet), clock_->CurrentTime());
+
+  // Payload padding not available without RTX, only generate plain padding on
+  // the media SSRC.
+  // Number of padding packets is the requested padding size divided by max
+  // padding packet size, rounded up. Pure padding packets are always of the
+  // maximum size.
+  const size_t kPaddingBytesRequested = kPayloadPacketSize + kRtxHeaderSize;
+  const size_t kExpectedNumPaddingPackets =
+      (kPaddingBytesRequested + kMaxPaddingSize - 1) / kMaxPaddingSize;
+  size_t padding_bytes_generated = 0;
+  std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
+      GeneratePadding(kPaddingBytesRequested);
+  EXPECT_EQ(padding_packets.size(), kExpectedNumPaddingPackets);
+  for (auto& packet : padding_packets) {
+    EXPECT_EQ(packet->packet_type(), RtpPacketMediaType::kPadding);
+    EXPECT_EQ(packet->Ssrc(), kSsrc);
+    EXPECT_EQ(packet->payload_size(), 0u);
+    EXPECT_GT(packet->padding_size(), 0u);
+    padding_bytes_generated += packet->padding_size();
+    EXPECT_TRUE(packet->HasExtension<TransportSequenceNumber>());
+    EXPECT_TRUE(packet->HasExtension<AbsoluteSendTime>());
+    EXPECT_TRUE(packet->HasExtension<TransmissionOffset>());
+  }
+
+  EXPECT_EQ(padding_bytes_generated,
+            kExpectedNumPaddingPackets * kMaxPaddingSize);
+}
+
+TEST_F(RtpSenderTest, SupportsPadding) {
+  bool kSendingMediaStats[] = {true, false};
+  bool kEnableRedundantPayloads[] = {true, false};
+  absl::string_view kBweExtensionUris[] = {
+      TransportSequenceNumber::Uri(), TransportSequenceNumberV2::Uri(),
+      AbsoluteSendTime::Uri(), TransmissionOffset::Uri()};
+  const int kExtensionsId = 7;
+
+  for (bool sending_media : kSendingMediaStats) {
+    rtp_sender_->SetSendingMediaStatus(sending_media);
+    for (bool redundant_payloads : kEnableRedundantPayloads) {
+      int rtx_mode = kRtxRetransmitted;
+      if (redundant_payloads) {
+        rtx_mode |= kRtxRedundantPayloads;
+      }
+      rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
+      rtp_sender_->SetRtxStatus(rtx_mode);
+
+      for (auto extension_uri : kBweExtensionUris) {
+        EXPECT_FALSE(rtp_sender_->SupportsPadding());
+        rtp_sender_->RegisterRtpHeaderExtension(extension_uri, kExtensionsId);
+        if (!sending_media) {
+          EXPECT_FALSE(rtp_sender_->SupportsPadding());
+        } else {
+          EXPECT_TRUE(rtp_sender_->SupportsPadding());
+          if (redundant_payloads) {
+            EXPECT_TRUE(rtp_sender_->SupportsRtxPayloadPadding());
+          } else {
+            EXPECT_FALSE(rtp_sender_->SupportsRtxPayloadPadding());
+          }
+        }
+        rtp_sender_->DeregisterRtpHeaderExtension(extension_uri);
+        EXPECT_FALSE(rtp_sender_->SupportsPadding());
+      }
+    }
+  }
+}
+
+TEST_F(RtpSenderTest, SetsCaptureTimeOnRtxRetransmissions) {
+  EnableRtx();
+
+  // Put a packet in the packet history, with current time as capture time.
+  const int64_t start_time_ms = clock_->TimeInMilliseconds();
+  std::unique_ptr<RtpPacketToSend> packet =
+      BuildRtpPacket(kPayload, kMarkerBit, start_time_ms,
+                     /*capture_time_ms=*/start_time_ms);
+  packet->set_allow_retransmission(true);
+  sequencer_->Sequence(*packet);
+  packet_history_->PutRtpPacket(std::move(packet),
+                                Timestamp::Millis(start_time_ms));
+
+  // Advance time, request an RTX retransmission. Capture timestamp should be
+  // preserved.
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+
+  EXPECT_CALL(
+      mock_paced_sender_,
+      EnqueuePackets(ElementsAre(Pointee(Property(
+          &RtpPacketToSend::capture_time, Timestamp::Millis(start_time_ms))))));
+  EXPECT_GT(rtp_sender_->ReSendPacket(kSeqNum), 0);
+}
+
+TEST_F(RtpSenderTest, IgnoresNackAfterDisablingMedia) {
+  const TimeDelta kRtt = TimeDelta::Millis(10);
+
+  EnableRtx();
+  packet_history_->SetRtt(kRtt);
+
+  // Put a packet in the history.
+  const int64_t start_time_ms = clock_->TimeInMilliseconds();
+  std::unique_ptr<RtpPacketToSend> packet =
+      BuildRtpPacket(kPayload, kMarkerBit, start_time_ms,
+                     /*capture_time_ms=*/start_time_ms);
+  packet->set_allow_retransmission(true);
+  sequencer_->Sequence(*packet);
+  packet_history_->PutRtpPacket(std::move(packet),
+                                Timestamp::Millis(start_time_ms));
+
+  // Disable media sending and try to retransmit the packet, it should fail.
+  rtp_sender_->SetSendingMediaStatus(false);
+  time_controller_.AdvanceTime(kRtt);
+  EXPECT_LT(rtp_sender_->ReSendPacket(kSeqNum), 0);
+}
+
+TEST_F(RtpSenderTest, DoesntFecProtectRetransmissions) {
+  // Set up retranmission without RTX, so that a plain copy of the old packet is
+  // re-sent instead.
+  const TimeDelta kRtt = TimeDelta::Millis(10);
+  rtp_sender_->SetSendingMediaStatus(true);
+  rtp_sender_->SetRtxStatus(kRtxOff);
+  packet_history_->SetStorePacketsStatus(
+      RtpPacketHistory::StorageMode::kStoreAndCull, 10);
+  packet_history_->SetRtt(kRtt);
+
+  // Put a fec protected packet in the history.
+  const int64_t start_time_ms = clock_->TimeInMilliseconds();
+  std::unique_ptr<RtpPacketToSend> packet =
+      BuildRtpPacket(kPayload, kMarkerBit, start_time_ms,
+                     /*capture_time_ms=*/start_time_ms);
+  packet->set_allow_retransmission(true);
+  packet->set_fec_protect_packet(true);
+  sequencer_->Sequence(*packet);
+  packet_history_->PutRtpPacket(std::move(packet),
+                                Timestamp::Millis(start_time_ms));
+
+  // Re-send packet, the retransmitted packet should not have the FEC protection
+  // flag set.
+  EXPECT_CALL(mock_paced_sender_,
+              EnqueuePackets(ElementsAre(Pointee(
+                  Property(&RtpPacketToSend::fec_protect_packet, false)))));
+
+  time_controller_.AdvanceTime(kRtt);
+  EXPECT_GT(rtp_sender_->ReSendPacket(kSeqNum), 0);
+}
+
+TEST_F(RtpSenderTest, MarksPacketsWithKeyframeStatus) {
+  RTPSenderVideo::Config video_config;
+  video_config.clock = clock_;
+  video_config.rtp_sender = rtp_sender_.get();
+  video_config.field_trials = &field_trials_;
+  RTPSenderVideo rtp_sender_video(video_config);
+
+  const uint8_t kPayloadType = 127;
+  const absl::optional<VideoCodecType> kCodecType =
+      VideoCodecType::kVideoCodecGeneric;
+
+  const uint32_t kCaptureTimeMsToRtpTimestamp = 90;  // 90 kHz clock
+
+  {
+    EXPECT_CALL(mock_paced_sender_,
+                EnqueuePackets(Each(
+                    Pointee(Property(&RtpPacketToSend::is_key_frame, true)))))
+        .Times(AtLeast(1));
+    RTPVideoHeader video_header;
+    video_header.frame_type = VideoFrameType::kVideoFrameKey;
+    int64_t capture_time_ms = clock_->TimeInMilliseconds();
+    EXPECT_TRUE(rtp_sender_video.SendVideo(
+        kPayloadType, kCodecType,
+        capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
+        kPayloadData, video_header, kDefaultExpectedRetransmissionTimeMs));
+
+    time_controller_.AdvanceTime(TimeDelta::Millis(33));
+  }
+
+  {
+    EXPECT_CALL(mock_paced_sender_,
+                EnqueuePackets(Each(
+                    Pointee(Property(&RtpPacketToSend::is_key_frame, false)))))
+        .Times(AtLeast(1));
+    RTPVideoHeader video_header;
+    video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+    int64_t capture_time_ms = clock_->TimeInMilliseconds();
+    EXPECT_TRUE(rtp_sender_video.SendVideo(
+        kPayloadType, kCodecType,
+        capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
+        kPayloadData, video_header, kDefaultExpectedRetransmissionTimeMs));
+
+    time_controller_.AdvanceTime(TimeDelta::Millis(33));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
new file mode 100644
index 0000000000..99a00025c1
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -0,0 +1,929 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/memory/memory.h"
+#include "absl/strings/match.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+namespace {
+constexpr size_t kRedForFecHeaderLength = 1;
+constexpr int64_t kMaxUnretransmittableFrameIntervalMs = 33 * 4;
+constexpr char kIncludeCaptureClockOffset[] =
+    "WebRTC-IncludeCaptureClockOffset";
+
+void BuildRedPayload(const RtpPacketToSend& media_packet,
+                     RtpPacketToSend* red_packet) {
+  uint8_t* red_payload = red_packet->AllocatePayload(
+      kRedForFecHeaderLength + media_packet.payload_size());
+  RTC_DCHECK(red_payload);
+  red_payload[0] = media_packet.PayloadType();
+
+  auto media_payload = media_packet.payload();
+  memcpy(&red_payload[kRedForFecHeaderLength], media_payload.data(),
+         media_payload.size());
+}
+
+bool MinimizeDescriptor(RTPVideoHeader* video_header) {
+  if (auto* vp8 =
+          absl::get_if<RTPVideoHeaderVP8>(&video_header->video_type_header)) {
+    // Set minimum fields the RtpPacketizer is using to create vp8 packets.
+    // nonReference is the only field that doesn't require extra space.
+    bool non_reference = vp8->nonReference;
+    vp8->InitRTPVideoHeaderVP8();
+    vp8->nonReference = non_reference;
+    return true;
+  }
+  // TODO(danilchap): Reduce vp9 codec specific descriptor too.
+  return false;
+}
+
+bool IsBaseLayer(const RTPVideoHeader& video_header) {
+  switch (video_header.codec) {
+    case kVideoCodecVP8: {
+      const auto& vp8 =
+          absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
+      return (vp8.temporalIdx == 0 || vp8.temporalIdx == kNoTemporalIdx);
+    }
+    case kVideoCodecVP9: {
+      const auto& vp9 =
+          absl::get<RTPVideoHeaderVP9>(video_header.video_type_header);
+      return (vp9.temporal_idx == 0 || vp9.temporal_idx == kNoTemporalIdx);
+    }
+    case kVideoCodecH264:
+      // TODO(kron): Implement logic for H264 once WebRTC supports temporal
+      // layers for H264.
+      break;
+    default:
+      break;
+  }
+  return true;
+}
+
+[[maybe_unused]] const char* FrameTypeToString(VideoFrameType frame_type) {
+  switch (frame_type) {
+    case VideoFrameType::kEmptyFrame:
+      return "empty";
+    case VideoFrameType::kVideoFrameKey:
+      return "video_key";
+    case VideoFrameType::kVideoFrameDelta:
+      return "video_delta";
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return "";
+  }
+}
+
+bool IsNoopDelay(const VideoPlayoutDelay& delay) {
+  return delay.min_ms == -1 && delay.max_ms == -1;
+}
+
+absl::optional<VideoPlayoutDelay> LoadVideoPlayoutDelayOverride(
+    const FieldTrialsView* key_value_config) {
+  RTC_DCHECK(key_value_config);
+  FieldTrialOptional<int> playout_delay_min_ms("min_ms", absl::nullopt);
+  FieldTrialOptional<int> playout_delay_max_ms("max_ms", absl::nullopt);
+  ParseFieldTrial({&playout_delay_max_ms, &playout_delay_min_ms},
+                  key_value_config->Lookup("WebRTC-ForceSendPlayoutDelay"));
+  return playout_delay_max_ms && playout_delay_min_ms
+             ? absl::make_optional<VideoPlayoutDelay>(*playout_delay_min_ms,
+                                                      *playout_delay_max_ms)
+             : absl::nullopt;
+}
+
+// Some packets can be skipped and the stream can still be decoded. Those
+// packets are less likely to be retransmitted if they are lost.
+bool PacketWillLikelyBeRequestedForRestransmitionIfLost(
+    const RTPVideoHeader& video_header) {
+  return IsBaseLayer(video_header) &&
+         !(video_header.generic.has_value()
+               ? absl::c_linear_search(
+                     video_header.generic->decode_target_indications,
+                     DecodeTargetIndication::kDiscardable)
+               : false);
+}
+
+}  // namespace
+
+RTPSenderVideo::RTPSenderVideo(const Config& config)
+    : rtp_sender_(config.rtp_sender),
+      clock_(config.clock),
+      retransmission_settings_(
+          config.enable_retransmit_all_layers
+              ? kRetransmitAllLayers
+              : (kRetransmitBaseLayer | kConditionallyRetransmitHigherLayers)),
+      last_rotation_(kVideoRotation_0),
+      transmit_color_space_next_frame_(false),
+      send_allocation_(SendVideoLayersAllocation::kDontSend),
+      current_playout_delay_{-1, -1},
+      playout_delay_pending_(false),
+      forced_playout_delay_(LoadVideoPlayoutDelayOverride(config.field_trials)),
+      red_payload_type_(config.red_payload_type),
+      fec_type_(config.fec_type),
+      fec_overhead_bytes_(config.fec_overhead_bytes),
+      packetization_overhead_bitrate_(1000, RateStatistics::kBpsScale),
+      frame_encryptor_(config.frame_encryptor),
+      require_frame_encryption_(config.require_frame_encryption),
+      generic_descriptor_auth_experiment_(!absl::StartsWith(
+          config.field_trials->Lookup("WebRTC-GenericDescriptorAuth"),
+          "Disabled")),
+      absolute_capture_time_sender_(config.clock),
+      frame_transformer_delegate_(
+          config.frame_transformer
+              ? rtc::make_ref_counted<RTPSenderVideoFrameTransformerDelegate>(
+                    this,
+                    config.frame_transformer,
+                    rtp_sender_->SSRC(),
+                    config.task_queue_factory)
+              : nullptr),
+      include_capture_clock_offset_(!absl::StartsWith(
+          config.field_trials->Lookup(kIncludeCaptureClockOffset),
+          "Disabled")) {
+  if (frame_transformer_delegate_)
+    frame_transformer_delegate_->Init();
+}
+
+RTPSenderVideo::~RTPSenderVideo() {
+  if (frame_transformer_delegate_)
+    frame_transformer_delegate_->Reset();
+}
+
+void RTPSenderVideo::LogAndSendToNetwork(
+    std::vector<std::unique_ptr<RtpPacketToSend>> packets,
+    size_t unpacketized_payload_size) {
+  {
+    MutexLock lock(&stats_mutex_);
+    size_t packetized_payload_size = 0;
+    for (const auto& packet : packets) {
+      if (*packet->packet_type() == RtpPacketMediaType::kVideo) {
+        packetized_payload_size += packet->payload_size();
+      }
+    }
+    // AV1 and H264 packetizers may produce less packetized bytes than
+    // unpacketized.
+    if (packetized_payload_size >= unpacketized_payload_size) {
+      packetization_overhead_bitrate_.Update(
+          packetized_payload_size - unpacketized_payload_size,
+          clock_->TimeInMilliseconds());
+    }
+  }
+
+  rtp_sender_->EnqueuePackets(std::move(packets));
+}
+
+size_t RTPSenderVideo::FecPacketOverhead() const {
+  size_t overhead = fec_overhead_bytes_;
+  if (red_enabled()) {
+    // The RED overhead is due to a small header.
+    overhead += kRedForFecHeaderLength;
+
+    if (fec_type_ == VideoFecGenerator::FecType::kUlpFec) {
+      // For ULPFEC, the overhead is the FEC headers plus RED for FEC header
+      // (see above) plus anything in RTP header beyond the 12 bytes base header
+      // (CSRC list, extensions...)
+      // This reason for the header extensions to be included here is that
+      // from an FEC viewpoint, they are part of the payload to be protected.
+      // (The base RTP header is already protected by the FEC header.)
+      overhead +=
+          rtp_sender_->FecOrPaddingPacketMaxRtpHeaderLength() - kRtpHeaderSize;
+    }
+  }
+  return overhead;
+}
+
+void RTPSenderVideo::SetVideoStructure(
+    const FrameDependencyStructure* video_structure) {
+  if (frame_transformer_delegate_) {
+    frame_transformer_delegate_->SetVideoStructureUnderLock(video_structure);
+    return;
+  }
+  SetVideoStructureInternal(video_structure);
+}
+
+void RTPSenderVideo::SetVideoStructureAfterTransformation(
+    const FrameDependencyStructure* video_structure) {
+  SetVideoStructureInternal(video_structure);
+}
+
+void RTPSenderVideo::SetVideoStructureInternal(
+    const FrameDependencyStructure* video_structure) {
+  RTC_DCHECK_RUNS_SERIALIZED(&send_checker_);
+  if (video_structure == nullptr) {
+    video_structure_ = nullptr;
+    return;
+  }
+  // Simple sanity checks video structure is set up.
+  RTC_DCHECK_GT(video_structure->num_decode_targets, 0);
+  RTC_DCHECK_GT(video_structure->templates.size(), 0);
+
+  int structure_id = 0;
+  if (video_structure_) {
+    if (*video_structure_ == *video_structure) {
+      // Same structure (just a new key frame), no update required.
+      return;
+    }
+    // When setting different video structure make sure structure_id is updated
+    // so that templates from different structures do not collide.
+    static constexpr int kMaxTemplates = 64;
+    structure_id =
+        (video_structure_->structure_id + video_structure_->templates.size()) %
+        kMaxTemplates;
+  }
+
+  video_structure_ =
+      std::make_unique<FrameDependencyStructure>(*video_structure);
+  video_structure_->structure_id = structure_id;
+}
+
+void RTPSenderVideo::SetVideoLayersAllocation(
+    VideoLayersAllocation allocation) {
+  if (frame_transformer_delegate_) {
+    frame_transformer_delegate_->SetVideoLayersAllocationUnderLock(
+        std::move(allocation));
+    return;
+  }
+  SetVideoLayersAllocationInternal(std::move(allocation));
+}
+
+void RTPSenderVideo::SetVideoLayersAllocationAfterTransformation(
+    VideoLayersAllocation allocation) {
+  SetVideoLayersAllocationInternal(std::move(allocation));
+}
+
+void RTPSenderVideo::SetVideoLayersAllocationInternal(
+    VideoLayersAllocation allocation) {
+  RTC_DCHECK_RUNS_SERIALIZED(&send_checker_);
+  if (!allocation_ || allocation.active_spatial_layers.size() !=
+                          allocation_->active_spatial_layers.size()) {
+    send_allocation_ = SendVideoLayersAllocation::kSendWithResolution;
+  } else if (send_allocation_ == SendVideoLayersAllocation::kDontSend) {
+    send_allocation_ = SendVideoLayersAllocation::kSendWithoutResolution;
+  }
+  if (send_allocation_ == SendVideoLayersAllocation::kSendWithoutResolution) {
+    // Check if frame rate changed more than 5fps since the last time the
+    // extension was sent with frame rate and resolution.
+    for (size_t i = 0; i < allocation.active_spatial_layers.size(); ++i) {
+      if (abs(static_cast<int>(
+                  allocation.active_spatial_layers[i].frame_rate_fps) -
+              static_cast<int>(
+                  last_full_sent_allocation_->active_spatial_layers[i]
+                      .frame_rate_fps)) > 5) {
+        send_allocation_ = SendVideoLayersAllocation::kSendWithResolution;
+        break;
+      }
+    }
+  }
+  allocation_ = std::move(allocation);
+}
+
+void RTPSenderVideo::AddRtpHeaderExtensions(const RTPVideoHeader& video_header,
+                                            bool first_packet,
+                                            bool last_packet,
+                                            RtpPacketToSend* packet) const {
+  // Send color space when changed or if the frame is a key frame. Keep
+  // sending color space information until the first base layer frame to
+  // guarantee that the information is retrieved by the receiver.
+  bool set_color_space =
+      video_header.color_space != last_color_space_ ||
+      video_header.frame_type == VideoFrameType::kVideoFrameKey ||
+      transmit_color_space_next_frame_;
+  // Color space requires two-byte header extensions if HDR metadata is
+  // included. Therefore, it's best to add this extension first so that the
+  // other extensions in the same packet are written as two-byte headers at
+  // once.
+  if (last_packet && set_color_space && video_header.color_space)
+    packet->SetExtension<ColorSpaceExtension>(video_header.color_space.value());
+
+  // According to
+  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
+  // ts_126114v120700p.pdf Section 7.4.5:
+  // The MTSI client shall add the payload bytes as defined in this clause
+  // onto the last RTP packet in each group of packets which make up a key
+  // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
+  // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP
+  // packet in each group of packets which make up another type of frame
+  // (e.g. a P-Frame) only if the current value is different from the previous
+  // value sent.
+  // Set rotation when key frame or when changed (to follow standard).
+  // Or when different from 0 (to follow current receiver implementation).
+  bool set_video_rotation =
+      video_header.frame_type == VideoFrameType::kVideoFrameKey ||
+      video_header.rotation != last_rotation_ ||
+      video_header.rotation != kVideoRotation_0;
+  if (last_packet && set_video_rotation)
+    packet->SetExtension<VideoOrientation>(video_header.rotation);
+
+  // Report content type only for key frames.
+  if (last_packet &&
+      video_header.frame_type == VideoFrameType::kVideoFrameKey &&
+      video_header.content_type != VideoContentType::UNSPECIFIED)
+    packet->SetExtension<VideoContentTypeExtension>(video_header.content_type);
+
+  if (last_packet &&
+      video_header.video_timing.flags != VideoSendTiming::kInvalid)
+    packet->SetExtension<VideoTimingExtension>(video_header.video_timing);
+
+  // If transmitted, add to all packets; ack logic depends on this.
+  if (playout_delay_pending_) {
+    packet->SetExtension<PlayoutDelayLimits>(current_playout_delay_);
+  }
+
+  if (first_packet && video_header.absolute_capture_time.has_value()) {
+    packet->SetExtension<AbsoluteCaptureTimeExtension>(
+        *video_header.absolute_capture_time);
+  }
+
+  if (video_header.generic) {
+    bool extension_is_set = false;
+    if (packet->IsRegistered<RtpDependencyDescriptorExtension>() &&
+        video_structure_ != nullptr) {
+      DependencyDescriptor descriptor;
+      descriptor.first_packet_in_frame = first_packet;
+      descriptor.last_packet_in_frame = last_packet;
+      descriptor.frame_number = video_header.generic->frame_id & 0xFFFF;
+      descriptor.frame_dependencies.spatial_id =
+          video_header.generic->spatial_index;
+      descriptor.frame_dependencies.temporal_id =
+          video_header.generic->temporal_index;
+      for (int64_t dep : video_header.generic->dependencies) {
+        descriptor.frame_dependencies.frame_diffs.push_back(
+            video_header.generic->frame_id - dep);
+      }
+      descriptor.frame_dependencies.chain_diffs =
+          video_header.generic->chain_diffs;
+      descriptor.frame_dependencies.decode_target_indications =
+          video_header.generic->decode_target_indications;
+      RTC_DCHECK_EQ(
+          descriptor.frame_dependencies.decode_target_indications.size(),
+          video_structure_->num_decode_targets);
+
+      if (first_packet) {
+        descriptor.active_decode_targets_bitmask =
+            active_decode_targets_tracker_.ActiveDecodeTargetsBitmask();
+      }
+      // VP9 mark all layer frames of the first picture as kVideoFrameKey,
+      // Structure should be attached to the descriptor to lowest spatial layer
+      // when inter layer dependency is used, i.e. L structures; or to all
+      // layers when inter layer dependency is not used, i.e. S structures.
+      // Distinguish these two cases by checking if there are any dependencies.
+      if (video_header.frame_type == VideoFrameType::kVideoFrameKey &&
+          video_header.generic->dependencies.empty() && first_packet) {
+        // To avoid extra structure copy, temporary share ownership of the
+        // video_structure with the dependency descriptor.
+        descriptor.attached_structure =
+            absl::WrapUnique(video_structure_.get());
+      }
+      extension_is_set = packet->SetExtension<RtpDependencyDescriptorExtension>(
+          *video_structure_,
+          active_decode_targets_tracker_.ActiveChainsBitmask(), descriptor);
+
+      // Remove the temporary shared ownership.
+      descriptor.attached_structure.release();
+    }
+
+    // Do not use generic frame descriptor when dependency descriptor is stored.
+    if (packet->IsRegistered<RtpGenericFrameDescriptorExtension00>() &&
+        !extension_is_set) {
+      RtpGenericFrameDescriptor generic_descriptor;
+      generic_descriptor.SetFirstPacketInSubFrame(first_packet);
+      generic_descriptor.SetLastPacketInSubFrame(last_packet);
+
+      if (first_packet) {
+        generic_descriptor.SetFrameId(
+            static_cast<uint16_t>(video_header.generic->frame_id));
+        for (int64_t dep : video_header.generic->dependencies) {
+          generic_descriptor.AddFrameDependencyDiff(
+              video_header.generic->frame_id - dep);
+        }
+
+        uint8_t spatial_bimask = 1 << video_header.generic->spatial_index;
+        generic_descriptor.SetSpatialLayersBitmask(spatial_bimask);
+
+        generic_descriptor.SetTemporalLayer(
+            video_header.generic->temporal_index);
+
+        if (video_header.frame_type == VideoFrameType::kVideoFrameKey) {
+          generic_descriptor.SetResolution(video_header.width,
+                                           video_header.height);
+        }
+      }
+
+      packet->SetExtension<RtpGenericFrameDescriptorExtension00>(
+          generic_descriptor);
+    }
+  }
+
+  if (packet->IsRegistered<RtpVideoLayersAllocationExtension>() &&
+      first_packet &&
+      send_allocation_ != SendVideoLayersAllocation::kDontSend &&
+      (video_header.frame_type == VideoFrameType::kVideoFrameKey ||
+       PacketWillLikelyBeRequestedForRestransmitionIfLost(video_header))) {
+    VideoLayersAllocation allocation = allocation_.value();
+    allocation.resolution_and_frame_rate_is_valid =
+        send_allocation_ == SendVideoLayersAllocation::kSendWithResolution;
+    packet->SetExtension<RtpVideoLayersAllocationExtension>(allocation);
+  }
+
+  if (first_packet && video_header.video_frame_tracking_id) {
+    packet->SetExtension<VideoFrameTrackingIdExtension>(
+        *video_header.video_frame_tracking_id);
+  }
+}
+
+bool RTPSenderVideo::SendVideo(
+    int payload_type,
+    absl::optional<VideoCodecType> codec_type,
+    uint32_t rtp_timestamp,
+    int64_t capture_time_ms,
+    rtc::ArrayView<const uint8_t> payload,
+    RTPVideoHeader video_header,
+    absl::optional<int64_t> expected_retransmission_time_ms) {
+  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
+                          FrameTypeToString(video_header.frame_type));
+  RTC_CHECK_RUNS_SERIALIZED(&send_checker_);
+
+  if (video_header.frame_type == VideoFrameType::kEmptyFrame)
+    return true;
+
+  if (payload.empty())
+    return false;
+  if (!rtp_sender_->SendingMedia()) {
+    return false;
+  }
+
+  int32_t retransmission_settings = retransmission_settings_;
+  if (codec_type == VideoCodecType::kVideoCodecH264) {
+    // Backward compatibility for older receivers without temporal layer logic.
+    retransmission_settings = kRetransmitBaseLayer | kRetransmitHigherLayers;
+  }
+  const uint8_t temporal_id = GetTemporalId(video_header);
+  // TODO(bugs.webrtc.org/10714): retransmission_settings_ should generally be
+  // replaced by expected_retransmission_time_ms.has_value().
+  const bool allow_retransmission =
+      expected_retransmission_time_ms.has_value() &&
+      AllowRetransmission(temporal_id, retransmission_settings,
+                          *expected_retransmission_time_ms);
+
+  MaybeUpdateCurrentPlayoutDelay(video_header);
+  if (video_header.frame_type == VideoFrameType::kVideoFrameKey) {
+    if (!IsNoopDelay(current_playout_delay_)) {
+      // Force playout delay on key-frames, if set.
+      playout_delay_pending_ = true;
+    }
+    if (allocation_) {
+      // Send the bitrate allocation on every key frame.
+      send_allocation_ = SendVideoLayersAllocation::kSendWithResolution;
+    }
+  }
+
+  if (video_structure_ != nullptr && video_header.generic) {
+    active_decode_targets_tracker_.OnFrame(
+        video_structure_->decode_target_protected_by_chain,
+        video_header.generic->active_decode_targets,
+        video_header.frame_type == VideoFrameType::kVideoFrameKey,
+        video_header.generic->frame_id, video_header.generic->chain_diffs);
+  }
+
+  // No FEC protection for upper temporal layers, if used.
+  const bool use_fec = fec_type_.has_value() &&
+                       (temporal_id == 0 || temporal_id == kNoTemporalIdx);
+
+  // Maximum size of packet including rtp headers.
+  // Extra space left in case packet will be resent using fec or rtx.
+  int packet_capacity = rtp_sender_->MaxRtpPacketSize();
+  if (use_fec) {
+    packet_capacity -= FecPacketOverhead();
+  }
+  if (allow_retransmission) {
+    packet_capacity -= rtp_sender_->RtxPacketOverhead();
+  }
+
+  absl::optional<Timestamp> capture_time;
+  if (capture_time_ms > 0) {
+    capture_time = Timestamp::Millis(capture_time_ms);
+  }
+
+  std::unique_ptr<RtpPacketToSend> single_packet =
+      rtp_sender_->AllocatePacket();
+  RTC_DCHECK_LE(packet_capacity, single_packet->capacity());
+  single_packet->SetPayloadType(payload_type);
+  single_packet->SetTimestamp(rtp_timestamp);
+  if (capture_time)
+    single_packet->set_capture_time(*capture_time);
+
+  // Construct the absolute capture time extension if not provided.
+  if (!video_header.absolute_capture_time.has_value() &&
+      capture_time.has_value()) {
+    video_header.absolute_capture_time.emplace();
+    video_header.absolute_capture_time->absolute_capture_timestamp =
+        Int64MsToUQ32x32(
+            clock_->ConvertTimestampToNtpTime(*capture_time).ToMs());
+    if (include_capture_clock_offset_) {
+      video_header.absolute_capture_time->estimated_capture_clock_offset = 0;
+    }
+  }
+
+  // Let `absolute_capture_time_sender_` decide if the extension should be sent.
+  if (video_header.absolute_capture_time.has_value()) {
+    video_header.absolute_capture_time =
+        absolute_capture_time_sender_.OnSendPacket(
+            AbsoluteCaptureTimeSender::GetSource(single_packet->Ssrc(),
+                                                 single_packet->Csrcs()),
+            single_packet->Timestamp(), kVideoPayloadTypeFrequency,
+            video_header.absolute_capture_time->absolute_capture_timestamp,
+            video_header.absolute_capture_time->estimated_capture_clock_offset);
+  }
+
+  auto first_packet = std::make_unique<RtpPacketToSend>(*single_packet);
+  auto middle_packet = std::make_unique<RtpPacketToSend>(*single_packet);
+  auto last_packet = std::make_unique<RtpPacketToSend>(*single_packet);
+  // Simplest way to estimate how much extensions would occupy is to set them.
+  AddRtpHeaderExtensions(video_header,
+                         /*first_packet=*/true, /*last_packet=*/true,
+                         single_packet.get());
+  if (video_structure_ != nullptr &&
+      single_packet->IsRegistered<RtpDependencyDescriptorExtension>() &&
+      !single_packet->HasExtension<RtpDependencyDescriptorExtension>()) {
+    RTC_DCHECK_EQ(video_header.frame_type, VideoFrameType::kVideoFrameKey);
+    // Disable attaching dependency descriptor to delta packets (including
+    // non-first packet of a key frame) when it wasn't attached to a key frame,
+    // as dependency descriptor can't be usable in such case.
+    RTC_LOG(LS_WARNING) << "Disable dependency descriptor because failed to "
+                           "attach it to a key frame.";
+    video_structure_ = nullptr;
+  }
+
+  AddRtpHeaderExtensions(video_header,
+                         /*first_packet=*/true, /*last_packet=*/false,
+                         first_packet.get());
+  AddRtpHeaderExtensions(video_header,
+                         /*first_packet=*/false, /*last_packet=*/false,
+                         middle_packet.get());
+  AddRtpHeaderExtensions(video_header,
+                         /*first_packet=*/false, /*last_packet=*/true,
+                         last_packet.get());
+
+  RTC_DCHECK_GT(packet_capacity, single_packet->headers_size());
+  RTC_DCHECK_GT(packet_capacity, first_packet->headers_size());
+  RTC_DCHECK_GT(packet_capacity, middle_packet->headers_size());
+  RTC_DCHECK_GT(packet_capacity, last_packet->headers_size());
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = packet_capacity - middle_packet->headers_size();
+
+  RTC_DCHECK_GE(single_packet->headers_size(), middle_packet->headers_size());
+  limits.single_packet_reduction_len =
+      single_packet->headers_size() - middle_packet->headers_size();
+
+  RTC_DCHECK_GE(first_packet->headers_size(), middle_packet->headers_size());
+  limits.first_packet_reduction_len =
+      first_packet->headers_size() - middle_packet->headers_size();
+
+  RTC_DCHECK_GE(last_packet->headers_size(), middle_packet->headers_size());
+  limits.last_packet_reduction_len =
+      last_packet->headers_size() - middle_packet->headers_size();
+
+  bool has_generic_descriptor =
+      first_packet->HasExtension<RtpGenericFrameDescriptorExtension00>() ||
+      first_packet->HasExtension<RtpDependencyDescriptorExtension>();
+
+  // Minimization of the vp8 descriptor may erase temporal_id, so use
+  // `temporal_id` rather than reference `video_header` beyond this point.
+  if (has_generic_descriptor) {
+    MinimizeDescriptor(&video_header);
+  }
+
+  // TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
+  rtc::Buffer encrypted_video_payload;
+  if (frame_encryptor_ != nullptr) {
+    const size_t max_ciphertext_size =
+        frame_encryptor_->GetMaxCiphertextByteSize(cricket::MEDIA_TYPE_VIDEO,
+                                                   payload.size());
+    encrypted_video_payload.SetSize(max_ciphertext_size);
+
+    size_t bytes_written = 0;
+
+    // Enable header authentication if the field trial isn't disabled.
+    std::vector<uint8_t> additional_data;
+    if (generic_descriptor_auth_experiment_) {
+      additional_data = RtpDescriptorAuthentication(video_header);
+    }
+
+    if (frame_encryptor_->Encrypt(
+            cricket::MEDIA_TYPE_VIDEO, first_packet->Ssrc(), additional_data,
+            payload, encrypted_video_payload, &bytes_written) != 0) {
+      return false;
+    }
+
+    encrypted_video_payload.SetSize(bytes_written);
+    payload = encrypted_video_payload;
+  } else if (require_frame_encryption_) {
+    RTC_LOG(LS_WARNING)
+        << "No FrameEncryptor is attached to this video sending stream but "
+           "one is required since require_frame_encryptor is set";
+  }
+
+  std::unique_ptr<RtpPacketizer> packetizer =
+      RtpPacketizer::Create(codec_type, payload, limits, video_header);
+
+  const size_t num_packets = packetizer->NumPackets();
+
+  if (num_packets == 0)
+    return false;
+
+  bool first_frame = first_frame_sent_();
+  std::vector<std::unique_ptr<RtpPacketToSend>> rtp_packets;
+  for (size_t i = 0; i < num_packets; ++i) {
+    std::unique_ptr<RtpPacketToSend> packet;
+    int expected_payload_capacity;
+    // Choose right packet template:
+    if (num_packets == 1) {
+      packet = std::move(single_packet);
+      expected_payload_capacity =
+          limits.max_payload_len - limits.single_packet_reduction_len;
+    } else if (i == 0) {
+      packet = std::move(first_packet);
+      expected_payload_capacity =
+          limits.max_payload_len - limits.first_packet_reduction_len;
+    } else if (i == num_packets - 1) {
+      packet = std::move(last_packet);
+      expected_payload_capacity =
+          limits.max_payload_len - limits.last_packet_reduction_len;
+    } else {
+      packet = std::make_unique<RtpPacketToSend>(*middle_packet);
+      expected_payload_capacity = limits.max_payload_len;
+    }
+
+    packet->set_first_packet_of_frame(i == 0);
+
+    if (!packetizer->NextPacket(packet.get()))
+      return false;
+    RTC_DCHECK_LE(packet->payload_size(), expected_payload_capacity);
+
+    packet->set_allow_retransmission(allow_retransmission);
+    packet->set_is_key_frame(video_header.frame_type ==
+                             VideoFrameType::kVideoFrameKey);
+
+    // Put packetization finish timestamp into extension.
+    if (packet->HasExtension<VideoTimingExtension>()) {
+      packet->set_packetization_finish_time(clock_->CurrentTime());
+    }
+
+    packet->set_fec_protect_packet(use_fec);
+
+    if (red_enabled()) {
+      // TODO(sprang): Consider packetizing directly into packets with the RED
+      // header already in place, to avoid this copy.
+      std::unique_ptr<RtpPacketToSend> red_packet(new RtpPacketToSend(*packet));
+      BuildRedPayload(*packet, red_packet.get());
+      red_packet->SetPayloadType(*red_payload_type_);
+      red_packet->set_is_red(true);
+
+      // Append `red_packet` instead of `packet` to output.
+      red_packet->set_packet_type(RtpPacketMediaType::kVideo);
+      red_packet->set_allow_retransmission(packet->allow_retransmission());
+      rtp_packets.emplace_back(std::move(red_packet));
+    } else {
+      packet->set_packet_type(RtpPacketMediaType::kVideo);
+      rtp_packets.emplace_back(std::move(packet));
+    }
+
+    if (first_frame) {
+      if (i == 0) {
+        RTC_LOG(LS_INFO)
+            << "Sent first RTP packet of the first video frame (pre-pacer)";
+      }
+      if (i == num_packets - 1) {
+        RTC_LOG(LS_INFO)
+            << "Sent last RTP packet of the first video frame (pre-pacer)";
+      }
+    }
+  }
+
+  LogAndSendToNetwork(std::move(rtp_packets), payload.size());
+
+  // Update details about the last sent frame.
+  last_rotation_ = video_header.rotation;
+
+  if (video_header.color_space != last_color_space_) {
+    last_color_space_ = video_header.color_space;
+    transmit_color_space_next_frame_ = !IsBaseLayer(video_header);
+  } else {
+    transmit_color_space_next_frame_ =
+        transmit_color_space_next_frame_ ? !IsBaseLayer(video_header) : false;
+  }
+
+  if (video_header.frame_type == VideoFrameType::kVideoFrameKey ||
+      PacketWillLikelyBeRequestedForRestransmitionIfLost(video_header)) {
+    // This frame will likely be delivered, no need to populate playout
+    // delay extensions until it changes again.
+    playout_delay_pending_ = false;
+    if (send_allocation_ == SendVideoLayersAllocation::kSendWithResolution) {
+      last_full_sent_allocation_ = allocation_;
+    }
+    send_allocation_ = SendVideoLayersAllocation::kDontSend;
+  }
+
+  TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp",
+                         rtp_timestamp);
+  return true;
+}
+
+bool RTPSenderVideo::SendEncodedImage(
+    int payload_type,
+    absl::optional<VideoCodecType> codec_type,
+    uint32_t rtp_timestamp,
+    const EncodedImage& encoded_image,
+    RTPVideoHeader video_header,
+    absl::optional<int64_t> expected_retransmission_time_ms) {
+  if (frame_transformer_delegate_) {
+    // The frame will be sent async once transformed.
+    return frame_transformer_delegate_->TransformFrame(
+        payload_type, codec_type, rtp_timestamp, encoded_image, video_header,
+        expected_retransmission_time_ms);
+  }
+  return SendVideo(payload_type, codec_type, rtp_timestamp,
+                   encoded_image.capture_time_ms_, encoded_image, video_header,
+                   expected_retransmission_time_ms);
+}
+
+uint32_t RTPSenderVideo::PacketizationOverheadBps() const {
+  MutexLock lock(&stats_mutex_);
+  return packetization_overhead_bitrate_.Rate(clock_->TimeInMilliseconds())
+      .value_or(0);
+}
+
+bool RTPSenderVideo::AllowRetransmission(
+    uint8_t temporal_id,
+    int32_t retransmission_settings,
+    int64_t expected_retransmission_time_ms) {
+  if (retransmission_settings == kRetransmitOff)
+    return false;
+
+  MutexLock lock(&stats_mutex_);
+  // Media packet storage.
+  if ((retransmission_settings & kConditionallyRetransmitHigherLayers) &&
+      UpdateConditionalRetransmit(temporal_id,
+                                  expected_retransmission_time_ms)) {
+    retransmission_settings |= kRetransmitHigherLayers;
+  }
+
+  if (temporal_id == kNoTemporalIdx)
+    return true;
+
+  if ((retransmission_settings & kRetransmitBaseLayer) && temporal_id == 0)
+    return true;
+
+  if ((retransmission_settings & kRetransmitHigherLayers) && temporal_id > 0)
+    return true;
+
+  return false;
+}
+
+uint8_t RTPSenderVideo::GetTemporalId(const RTPVideoHeader& header) {
+  struct TemporalIdGetter {
+    uint8_t operator()(const RTPVideoHeaderVP8& vp8) { return vp8.temporalIdx; }
+    uint8_t operator()(const RTPVideoHeaderVP9& vp9) {
+      return vp9.temporal_idx;
+    }
+    uint8_t operator()(const RTPVideoHeaderH264&) { return kNoTemporalIdx; }
+    uint8_t operator()(const RTPVideoHeaderLegacyGeneric&) {
+      return kNoTemporalIdx;
+    }
+    uint8_t operator()(const absl::monostate&) { return kNoTemporalIdx; }
+  };
+  return absl::visit(TemporalIdGetter(), header.video_type_header);
+}
+
+bool RTPSenderVideo::UpdateConditionalRetransmit(
+    uint8_t temporal_id,
+    int64_t expected_retransmission_time_ms) {
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  // Update stats for any temporal layer.
+  TemporalLayerStats* current_layer_stats =
+      &frame_stats_by_temporal_layer_[temporal_id];
+  current_layer_stats->frame_rate_fp1000s.Update(1, now_ms);
+  int64_t tl_frame_interval = now_ms - current_layer_stats->last_frame_time_ms;
+  current_layer_stats->last_frame_time_ms = now_ms;
+
+  // Conditional retransmit only applies to upper layers.
+  if (temporal_id != kNoTemporalIdx && temporal_id > 0) {
+    if (tl_frame_interval >= kMaxUnretransmittableFrameIntervalMs) {
+      // Too long since a retransmittable frame in this layer, enable NACK
+      // protection.
+      return true;
+    } else {
+      // Estimate when the next frame of any lower layer will be sent.
+      const int64_t kUndefined = std::numeric_limits<int64_t>::max();
+      int64_t expected_next_frame_time = kUndefined;
+      for (int i = temporal_id - 1; i >= 0; --i) {
+        TemporalLayerStats* stats = &frame_stats_by_temporal_layer_[i];
+        absl::optional<uint32_t> rate = stats->frame_rate_fp1000s.Rate(now_ms);
+        if (rate) {
+          int64_t tl_next = stats->last_frame_time_ms + 1000000 / *rate;
+          if (tl_next - now_ms > -expected_retransmission_time_ms &&
+              tl_next < expected_next_frame_time) {
+            expected_next_frame_time = tl_next;
+          }
+        }
+      }
+
+      if (expected_next_frame_time == kUndefined ||
+          expected_next_frame_time - now_ms > expected_retransmission_time_ms) {
+        // The next frame in a lower layer is expected at a later time (or
+        // unable to tell due to lack of data) than a retransmission is
+        // estimated to be able to arrive, so allow this packet to be nacked.
+        return true;
+      }
+    }
+  }
+
+  return false;
+}
+
+void RTPSenderVideo::MaybeUpdateCurrentPlayoutDelay(
+    const RTPVideoHeader& header) {
+  VideoPlayoutDelay requested_delay =
+      forced_playout_delay_.value_or(header.playout_delay);
+
+  if (IsNoopDelay(requested_delay)) {
+    return;
+  }
+
+  if (requested_delay.min_ms > PlayoutDelayLimits::kMaxMs ||
+      requested_delay.max_ms > PlayoutDelayLimits::kMaxMs) {
+    RTC_DLOG(LS_ERROR)
+        << "Requested playout delay values out of range, ignored";
+    return;
+  }
+  if (requested_delay.max_ms != -1 &&
+      requested_delay.min_ms > requested_delay.max_ms) {
+    RTC_DLOG(LS_ERROR) << "Requested playout delay values out of order";
+    return;
+  }
+
+  if (!playout_delay_pending_) {
+    current_playout_delay_ = requested_delay;
+    playout_delay_pending_ = true;
+    return;
+  }
+
+  if ((requested_delay.min_ms == -1 ||
+       requested_delay.min_ms == current_playout_delay_.min_ms) &&
+      (requested_delay.max_ms == -1 ||
+       requested_delay.max_ms == current_playout_delay_.max_ms)) {
+    // No change, ignore.
+    return;
+  }
+
+  if (requested_delay.min_ms == -1) {
+    RTC_DCHECK_GE(requested_delay.max_ms, 0);
+    requested_delay.min_ms =
+        std::min(current_playout_delay_.min_ms, requested_delay.max_ms);
+  }
+  if (requested_delay.max_ms == -1) {
+    requested_delay.max_ms =
+        std::max(current_playout_delay_.max_ms, requested_delay.min_ms);
+  }
+
+  current_playout_delay_ = requested_delay;
+  playout_delay_pending_ = true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.h
new file mode 100644
index 0000000000..ecff8d42de
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -0,0 +1,255 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_codec_type.h"
+#include "api/video/video_frame_type.h"
+#include "api/video/video_layers_allocation.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
+#include "modules/rtp_rtcp/source/active_decode_targets_helper.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/rtp_sender.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/rtp_rtcp/source/video_fec_generator.h"
+#include "rtc_base/one_time_event.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class FrameEncryptorInterface;
+class RtpPacketizer;
+class RtpPacketToSend;
+
+// kConditionallyRetransmitHigherLayers allows retransmission of video frames
+// in higher layers if either the last frame in that layer was too far back in
+// time, or if we estimate that a new frame will be available in a lower layer
+// in a shorter time than it would take to request and receive a retransmission.
+enum RetransmissionMode : uint8_t {
+  kRetransmitOff = 0x0,
+  kRetransmitBaseLayer = 0x2,
+  kRetransmitHigherLayers = 0x4,
+  kRetransmitAllLayers = 0x6,
+  kConditionallyRetransmitHigherLayers = 0x8
+};
+
+class RTPSenderVideo {
+ public:
+  static constexpr int64_t kTLRateWindowSizeMs = 2500;
+
+  struct Config {
+    Config() = default;
+    Config(const Config&) = delete;
+    Config(Config&&) = default;
+
+    // All members of this struct, with the exception of `field_trials`, are
+    // expected to outlive the RTPSenderVideo object they are passed to.
+    Clock* clock = nullptr;
+    RTPSender* rtp_sender = nullptr;
+    // Some FEC data is duplicated here in preparation of moving FEC to
+    // the egress stage.
+    absl::optional<VideoFecGenerator::FecType> fec_type;
+    size_t fec_overhead_bytes = 0;  // Per packet max FEC overhead.
+    FrameEncryptorInterface* frame_encryptor = nullptr;
+    bool require_frame_encryption = false;
+    bool enable_retransmit_all_layers = false;
+    absl::optional<int> red_payload_type;
+    const FieldTrialsView* field_trials = nullptr;
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer;
+    TaskQueueFactory* task_queue_factory = nullptr;
+  };
+
+  explicit RTPSenderVideo(const Config& config);
+
+  virtual ~RTPSenderVideo();
+
+  // expected_retransmission_time_ms.has_value() -> retransmission allowed.
+  // `capture_time_ms` and `clock::CurrentTime` should be using the same epoch.
+  // Calls to this method are assumed to be externally serialized.
+  bool SendVideo(int payload_type,
+                 absl::optional<VideoCodecType> codec_type,
+                 uint32_t rtp_timestamp,
+                 int64_t capture_time_ms,
+                 rtc::ArrayView<const uint8_t> payload,
+                 RTPVideoHeader video_header,
+                 absl::optional<int64_t> expected_retransmission_time_ms);
+
+  bool SendEncodedImage(
+      int payload_type,
+      absl::optional<VideoCodecType> codec_type,
+      uint32_t rtp_timestamp,
+      const EncodedImage& encoded_image,
+      RTPVideoHeader video_header,
+      absl::optional<int64_t> expected_retransmission_time_ms);
+
+  // Configures video structures produced by encoder to send using the
+  // dependency descriptor rtp header extension. Next call to SendVideo should
+  // have video_header.frame_type == kVideoFrameKey.
+  // All calls to SendVideo after this call must use video_header compatible
+  // with the video_structure.
+  void SetVideoStructure(const FrameDependencyStructure* video_structure);
+  // Should only be used by a RTPSenderVideoFrameTransformerDelegate and exists
+  // to ensure correct syncronization.
+  void SetVideoStructureAfterTransformation(
+      const FrameDependencyStructure* video_structure);
+
+  // Sets current active VideoLayersAllocation. The allocation will be sent
+  // using the rtp video layers allocation extension. The allocation will be
+  // sent in full on every key frame. The allocation will be sent once on a
+  // none discardable delta frame per call to this method and will not contain
+  // resolution and frame rate.
+  void SetVideoLayersAllocation(VideoLayersAllocation allocation);
+  // Should only be used by a RTPSenderVideoFrameTransformerDelegate and exists
+  // to ensure correct syncronization.
+  void SetVideoLayersAllocationAfterTransformation(
+      VideoLayersAllocation allocation);
+
+  // Returns the current packetization overhead rate, in bps. Note that this is
+  // the payload overhead, eg the VP8 payload headers, not the RTP headers
+  // or extension/
+  // TODO(sprang): Consider moving this to RtpSenderEgress so it's in the same
+  // place as the other rate stats.
+  uint32_t PacketizationOverheadBps() const;
+
+ protected:
+  static uint8_t GetTemporalId(const RTPVideoHeader& header);
+  bool AllowRetransmission(uint8_t temporal_id,
+                           int32_t retransmission_settings,
+                           int64_t expected_retransmission_time_ms);
+
+ private:
+  struct TemporalLayerStats {
+    TemporalLayerStats()
+        : frame_rate_fp1000s(kTLRateWindowSizeMs, 1000 * 1000),
+          last_frame_time_ms(0) {}
+    // Frame rate, in frames per 1000 seconds. This essentially turns the fps
+    // value into a fixed point value with three decimals. Improves precision at
+    // low frame rates.
+    RateStatistics frame_rate_fp1000s;
+    int64_t last_frame_time_ms;
+  };
+
+  enum class SendVideoLayersAllocation {
+    kSendWithResolution,
+    kSendWithoutResolution,
+    kDontSend
+  };
+
+  void SetVideoStructureInternal(
+      const FrameDependencyStructure* video_structure);
+  void SetVideoLayersAllocationInternal(VideoLayersAllocation allocation);
+
+  void AddRtpHeaderExtensions(const RTPVideoHeader& video_header,
+                              bool first_packet,
+                              bool last_packet,
+                              RtpPacketToSend* packet) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(send_checker_);
+
+  size_t FecPacketOverhead() const RTC_EXCLUSIVE_LOCKS_REQUIRED(send_checker_);
+
+  void LogAndSendToNetwork(
+      std::vector<std::unique_ptr<RtpPacketToSend>> packets,
+      size_t unpacketized_payload_size);
+
+  bool red_enabled() const { return red_payload_type_.has_value(); }
+
+  bool UpdateConditionalRetransmit(uint8_t temporal_id,
+                                   int64_t expected_retransmission_time_ms)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(stats_mutex_);
+
+  void MaybeUpdateCurrentPlayoutDelay(const RTPVideoHeader& header)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(send_checker_);
+
+  RTPSender* const rtp_sender_;
+  Clock* const clock_;
+
+  const int32_t retransmission_settings_;
+
+  // These members should only be accessed from within SendVideo() to avoid
+  // potential race conditions.
+  rtc::RaceChecker send_checker_;
+  VideoRotation last_rotation_ RTC_GUARDED_BY(send_checker_);
+  absl::optional<ColorSpace> last_color_space_ RTC_GUARDED_BY(send_checker_);
+  bool transmit_color_space_next_frame_ RTC_GUARDED_BY(send_checker_);
+  std::unique_ptr<FrameDependencyStructure> video_structure_
+      RTC_GUARDED_BY(send_checker_);
+  absl::optional<VideoLayersAllocation> allocation_
+      RTC_GUARDED_BY(send_checker_);
+  // Flag indicating if we should send `allocation_`.
+  SendVideoLayersAllocation send_allocation_ RTC_GUARDED_BY(send_checker_);
+  absl::optional<VideoLayersAllocation> last_full_sent_allocation_
+      RTC_GUARDED_BY(send_checker_);
+
+  // Current target playout delay.
+  VideoPlayoutDelay current_playout_delay_ RTC_GUARDED_BY(send_checker_);
+  // Flag indicating if we need to send `current_playout_delay_` in order
+  // to guarantee it gets delivered.
+  bool playout_delay_pending_;
+  // Set by the field trial WebRTC-ForceSendPlayoutDelay to override the playout
+  // delay of outgoing video frames.
+  const absl::optional<VideoPlayoutDelay> forced_playout_delay_;
+
+  // Should never be held when calling out of this class.
+  Mutex mutex_;
+
+  const absl::optional<int> red_payload_type_;
+  absl::optional<VideoFecGenerator::FecType> fec_type_;
+  const size_t fec_overhead_bytes_;  // Per packet max FEC overhead.
+
+  mutable Mutex stats_mutex_;
+  RateStatistics packetization_overhead_bitrate_ RTC_GUARDED_BY(stats_mutex_);
+
+  std::map<int, TemporalLayerStats> frame_stats_by_temporal_layer_
+      RTC_GUARDED_BY(stats_mutex_);
+
+  OneTimeEvent first_frame_sent_;
+
+  // E2EE Custom Video Frame Encryptor (optional)
+  FrameEncryptorInterface* const frame_encryptor_ = nullptr;
+  // If set to true will require all outgoing frames to pass through an
+  // initialized frame_encryptor_ before being sent out of the network.
+  // Otherwise these payloads will be dropped.
+  const bool require_frame_encryption_;
+  // Set to true if the generic descriptor should be authenticated.
+  const bool generic_descriptor_auth_experiment_;
+
+  AbsoluteCaptureTimeSender absolute_capture_time_sender_;
+  // Tracks updates to the active decode targets and decides when active decode
+  // targets bitmask should be attached to the dependency descriptor.
+  ActiveDecodeTargetsHelper active_decode_targets_tracker_;
+
+  const rtc::scoped_refptr<RTPSenderVideoFrameTransformerDelegate>
+      frame_transformer_delegate_;
+
+  const bool include_capture_clock_offset_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
new file mode 100644
index 0000000000..27b6a17a20
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
@@ -0,0 +1,227 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h"
+
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+class TransformableVideoSenderFrame : public TransformableVideoFrameInterface {
+ public:
+  TransformableVideoSenderFrame(
+      const EncodedImage& encoded_image,
+      const RTPVideoHeader& video_header,
+      int payload_type,
+      absl::optional<VideoCodecType> codec_type,
+      uint32_t rtp_timestamp,
+      absl::optional<int64_t> expected_retransmission_time_ms,
+      uint32_t ssrc)
+      : encoded_data_(encoded_image.GetEncodedData()),
+        header_(video_header),
+        metadata_(header_.GetAsMetadata()),
+        frame_type_(encoded_image._frameType),
+        payload_type_(payload_type),
+        codec_type_(codec_type),
+        timestamp_(rtp_timestamp),
+        capture_time_ms_(encoded_image.capture_time_ms_),
+        expected_retransmission_time_ms_(expected_retransmission_time_ms),
+        ssrc_(ssrc) {
+    RTC_DCHECK_GE(payload_type_, 0);
+    RTC_DCHECK_LE(payload_type_, 127);
+  }
+
+  ~TransformableVideoSenderFrame() override = default;
+
+  // Implements TransformableVideoFrameInterface.
+  rtc::ArrayView<const uint8_t> GetData() const override {
+    return *encoded_data_;
+  }
+
+  void SetData(rtc::ArrayView<const uint8_t> data) override {
+    encoded_data_ = EncodedImageBuffer::Create(data.data(), data.size());
+  }
+
+  uint32_t GetTimestamp() const override { return timestamp_; }
+  uint32_t GetSsrc() const override { return ssrc_; }
+
+  bool IsKeyFrame() const override {
+    return frame_type_ == VideoFrameType::kVideoFrameKey;
+  }
+
+  std::vector<uint8_t> GetAdditionalData() const override {
+    return RtpDescriptorAuthentication(header_);
+  }
+
+  const VideoFrameMetadata& GetMetadata() const override { return metadata_; }
+  void SetMetadata(const VideoFrameMetadata& metadata) override {
+    header_.SetFromMetadata(metadata);
+    // We have to keep a local copy because GetMetadata() has to return a
+    // reference.
+    metadata_ = header_.GetAsMetadata();
+  }
+
+  const RTPVideoHeader& GetHeader() const { return header_; }
+  uint8_t GetPayloadType() const override { return payload_type_; }
+  absl::optional<VideoCodecType> GetCodecType() const { return codec_type_; }
+  int64_t GetCaptureTimeMs() const { return capture_time_ms_; }
+
+  const absl::optional<int64_t>& GetExpectedRetransmissionTimeMs() const {
+    return expected_retransmission_time_ms_;
+  }
+
+  Direction GetDirection() const override { return Direction::kSender; }
+
+ private:
+  rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data_;
+  RTPVideoHeader header_;
+  // This is a copy of `header_.GetAsMetadata()`, only needed because the
+  // interface says GetMetadata() must return a const ref rather than a value.
+  // TODO(crbug.com/webrtc/14709): Change the interface and delete this variable
+  // to reduce risk of it getting out-of-sync with `header_.GetAsMetadata()`.
+  VideoFrameMetadata metadata_;
+  const VideoFrameType frame_type_;
+  const uint8_t payload_type_;
+  const absl::optional<VideoCodecType> codec_type_ = absl::nullopt;
+  const uint32_t timestamp_;
+  const int64_t capture_time_ms_;
+  const absl::optional<int64_t> expected_retransmission_time_ms_;
+  const uint32_t ssrc_;
+};
+}  // namespace
+
+RTPSenderVideoFrameTransformerDelegate::RTPSenderVideoFrameTransformerDelegate(
+    RTPSenderVideo* sender,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    uint32_t ssrc,
+    TaskQueueFactory* task_queue_factory)
+    : sender_(sender),
+      frame_transformer_(std::move(frame_transformer)),
+      ssrc_(ssrc),
+      transformation_queue_(task_queue_factory->CreateTaskQueue(
+          "video_frame_transformer",
+          TaskQueueFactory::Priority::NORMAL)) {}
+
+void RTPSenderVideoFrameTransformerDelegate::Init() {
+  frame_transformer_->RegisterTransformedFrameSinkCallback(
+      rtc::scoped_refptr<TransformedFrameCallback>(this), ssrc_);
+}
+
+bool RTPSenderVideoFrameTransformerDelegate::TransformFrame(
+    int payload_type,
+    absl::optional<VideoCodecType> codec_type,
+    uint32_t rtp_timestamp,
+    const EncodedImage& encoded_image,
+    RTPVideoHeader video_header,
+    absl::optional<int64_t> expected_retransmission_time_ms) {
+  frame_transformer_->Transform(std::make_unique<TransformableVideoSenderFrame>(
+      encoded_image, video_header, payload_type, codec_type, rtp_timestamp,
+      expected_retransmission_time_ms, ssrc_));
+  return true;
+}
+
+void RTPSenderVideoFrameTransformerDelegate::OnTransformedFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) {
+  MutexLock lock(&sender_lock_);
+
+  if (!sender_) {
+    return;
+  }
+  rtc::scoped_refptr<RTPSenderVideoFrameTransformerDelegate> delegate(this);
+  transformation_queue_->PostTask(
+      [delegate = std::move(delegate), frame = std::move(frame)]() mutable {
+        RTC_DCHECK_RUN_ON(delegate->transformation_queue_.get());
+        delegate->SendVideo(std::move(frame));
+      });
+}
+
+void RTPSenderVideoFrameTransformerDelegate::SendVideo(
+    std::unique_ptr<TransformableFrameInterface> transformed_frame) const {
+  RTC_DCHECK_RUN_ON(transformation_queue_.get());
+  RTC_CHECK_EQ(transformed_frame->GetDirection(),
+               TransformableFrameInterface::Direction::kSender);
+  MutexLock lock(&sender_lock_);
+  if (!sender_)
+    return;
+  auto* transformed_video_frame =
+      static_cast<TransformableVideoSenderFrame*>(transformed_frame.get());
+  sender_->SendVideo(
+      transformed_video_frame->GetPayloadType(),
+      transformed_video_frame->GetCodecType(),
+      transformed_video_frame->GetTimestamp(),
+      transformed_video_frame->GetCaptureTimeMs(),
+      transformed_video_frame->GetData(), transformed_video_frame->GetHeader(),
+      transformed_video_frame->GetExpectedRetransmissionTimeMs());
+}
+
+void RTPSenderVideoFrameTransformerDelegate::SetVideoStructureUnderLock(
+    const FrameDependencyStructure* video_structure) {
+  MutexLock lock(&sender_lock_);
+  RTC_CHECK(sender_);
+  sender_->SetVideoStructureAfterTransformation(video_structure);
+}
+
+void RTPSenderVideoFrameTransformerDelegate::SetVideoLayersAllocationUnderLock(
+    VideoLayersAllocation allocation) {
+  MutexLock lock(&sender_lock_);
+  RTC_CHECK(sender_);
+  sender_->SetVideoLayersAllocationAfterTransformation(std::move(allocation));
+}
+
+void RTPSenderVideoFrameTransformerDelegate::Reset() {
+  frame_transformer_->UnregisterTransformedFrameSinkCallback(ssrc_);
+  frame_transformer_ = nullptr;
+  {
+    MutexLock lock(&sender_lock_);
+    sender_ = nullptr;
+  }
+}
+
+std::unique_ptr<TransformableVideoFrameInterface> CloneSenderVideoFrame(
+    TransformableVideoFrameInterface* original) {
+  auto encoded_image_buffer = EncodedImageBuffer::Create(
+      original->GetData().data(), original->GetData().size());
+  EncodedImage encoded_image;
+  encoded_image.SetEncodedData(encoded_image_buffer);
+  RTPVideoHeader new_header;
+  absl::optional<VideoCodecType> new_codec_type;
+  // TODO(bugs.webrtc.org/14708): Figure out a way to get the header information
+  // without casting to TransformableVideoSenderFrame.
+  if (original->GetDirection() ==
+      TransformableFrameInterface::Direction::kSender) {
+    // TODO(bugs.webrtc.org/14708): Figure out a way to bulletproof this cast.
+    auto original_as_sender =
+        static_cast<TransformableVideoSenderFrame*>(original);
+    new_header = original_as_sender->GetHeader();
+    new_codec_type = original_as_sender->GetCodecType();
+  } else {
+    // TODO(bugs.webrtc.org/14708): Make this codec dependent
+    new_header.video_type_header.emplace<RTPVideoHeaderVP8>();
+    new_codec_type = kVideoCodecVP8;
+    // TODO(bugs.webrtc.org/14708): Fill in the new_header when it's not
+    // `Direction::kSender`
+  }
+  // TODO(bugs.webrtc.org/14708): Fill in other EncodedImage parameters
+  return std::make_unique<TransformableVideoSenderFrame>(
+      encoded_image, new_header, original->GetPayloadType(), new_codec_type,
+      original->GetTimestamp(),
+      absl::nullopt,  // expected_retransmission_time_ms
+      original->GetSsrc());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
new file mode 100644
index 0000000000..55f7961e2d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_FRAME_TRANSFORMER_DELEGATE_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_FRAME_TRANSFORMER_DELEGATE_H_
+
+#include <memory>
+
+#include "api/frame_transformer_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/video/video_layers_allocation.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class RTPSenderVideo;
+
+// Delegates calls to FrameTransformerInterface to transform frames, and to
+// RTPSenderVideo to send the transformed frames. Ensures thread-safe access to
+// the sender.
+class RTPSenderVideoFrameTransformerDelegate : public TransformedFrameCallback {
+ public:
+  RTPSenderVideoFrameTransformerDelegate(
+      RTPSenderVideo* sender,
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+      uint32_t ssrc,
+      TaskQueueFactory* send_transport_queue);
+
+  void Init();
+
+  // Delegates the call to FrameTransformerInterface::TransformFrame.
+  bool TransformFrame(int payload_type,
+                      absl::optional<VideoCodecType> codec_type,
+                      uint32_t rtp_timestamp,
+                      const EncodedImage& encoded_image,
+                      RTPVideoHeader video_header,
+                      absl::optional<int64_t> expected_retransmission_time_ms);
+
+  // Implements TransformedFrameCallback. Can be called on any thread. Posts
+  // the transformed frame to be sent on the `encoder_queue_`.
+  void OnTransformedFrame(
+      std::unique_ptr<TransformableFrameInterface> frame) override;
+
+  // Delegates the call to RTPSendVideo::SendVideo on the `encoder_queue_`.
+  void SendVideo(std::unique_ptr<TransformableFrameInterface> frame) const
+      RTC_RUN_ON(transformation_queue_);
+
+  // Delegates the call to RTPSendVideo::SetVideoStructureAfterTransformation
+  // under `sender_lock_`.
+  void SetVideoStructureUnderLock(
+      const FrameDependencyStructure* video_structure);
+
+  // Delegates the call to
+  // RTPSendVideo::SetVideoLayersAllocationAfterTransformation under
+  // `sender_lock_`.
+  void SetVideoLayersAllocationUnderLock(VideoLayersAllocation allocation);
+
+  // Unregisters and releases the `frame_transformer_` reference, and resets
+  // `sender_` under lock. Called from RTPSenderVideo destructor to prevent the
+  // `sender_` to dangle.
+  void Reset();
+
+ protected:
+  ~RTPSenderVideoFrameTransformerDelegate() override = default;
+
+ private:
+  void EnsureEncoderQueueCreated();
+
+  mutable Mutex sender_lock_;
+  RTPSenderVideo* sender_ RTC_GUARDED_BY(sender_lock_);
+  rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_;
+  const uint32_t ssrc_;
+  // Used when the encoded frames arrives without a current task queue. This can
+  // happen if a hardware encoder was used.
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> transformation_queue_;
+};
+
+// Method to support cloning a Sender frame from another frame
+std::unique_ptr<TransformableVideoFrameInterface> CloneSenderVideoFrame(
+    TransformableVideoFrameInterface* original);
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_FRAME_TRANSFORMER_DELEGATE_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
new file mode 100644
index 0000000000..d6fbba7bd8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -0,0 +1,1697 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "api/field_trials_registry.h"
+#include "api/frame_transformer_factory.h"
+#include "api/rtp_headers.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/mock_frame_encryptor.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video/video_timing.h"
+#include "modules/rtp_rtcp/include/rtp_cvo.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/rate_limiter.h"
+#include "rtc_base/thread.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_frame_transformer.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+
+namespace {
+
+using ::testing::_;
+using ::testing::ContainerEq;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+using ::testing::NiceMock;
+using ::testing::Not;
+using ::testing::Return;
+using ::testing::ReturnArg;
+using ::testing::SaveArg;
+using ::testing::SizeIs;
+using ::testing::WithArgs;
+
+enum : int {  // The first valid value is 1.
+  kAbsoluteSendTimeExtensionId = 1,
+  kGenericDescriptorId,
+  kDependencyDescriptorId,
+  kTransmissionTimeOffsetExtensionId,
+  kTransportSequenceNumberExtensionId,
+  kVideoRotationExtensionId,
+  kVideoTimingExtensionId,
+  kAbsoluteCaptureTimeExtensionId,
+  kPlayoutDelayExtensionId,
+  kVideoLayersAllocationExtensionId,
+};
+
+constexpr int kPayload = 100;
+constexpr VideoCodecType kType = VideoCodecType::kVideoCodecGeneric;
+constexpr uint32_t kTimestamp = 10;
+constexpr uint16_t kSeqNum = 33;
+constexpr uint32_t kSsrc = 725242;
+constexpr uint32_t kRtxSsrc = 912364;
+constexpr int kMaxPacketLength = 1500;
+constexpr Timestamp kStartTime = Timestamp::Millis(123456789);
+constexpr int64_t kDefaultExpectedRetransmissionTimeMs = 125;
+
+class LoopbackTransportTest : public webrtc::Transport {
+ public:
+  LoopbackTransportTest() {
+    receivers_extensions_.Register<TransmissionOffset>(
+        kTransmissionTimeOffsetExtensionId);
+    receivers_extensions_.Register<AbsoluteSendTime>(
+        kAbsoluteSendTimeExtensionId);
+    receivers_extensions_.Register<TransportSequenceNumber>(
+        kTransportSequenceNumberExtensionId);
+    receivers_extensions_.Register<VideoOrientation>(kVideoRotationExtensionId);
+    receivers_extensions_.Register<VideoTimingExtension>(
+        kVideoTimingExtensionId);
+    receivers_extensions_.Register<RtpGenericFrameDescriptorExtension00>(
+        kGenericDescriptorId);
+    receivers_extensions_.Register<RtpDependencyDescriptorExtension>(
+        kDependencyDescriptorId);
+    receivers_extensions_.Register<AbsoluteCaptureTimeExtension>(
+        kAbsoluteCaptureTimeExtensionId);
+    receivers_extensions_.Register<PlayoutDelayLimits>(
+        kPlayoutDelayExtensionId);
+    receivers_extensions_.Register<RtpVideoLayersAllocationExtension>(
+        kVideoLayersAllocationExtensionId);
+  }
+
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override {
+    sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
+    EXPECT_TRUE(sent_packets_.back().Parse(data, len));
+    return true;
+  }
+  bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
+  const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); }
+  int packets_sent() { return sent_packets_.size(); }
+  const std::vector<RtpPacketReceived>& sent_packets() const {
+    return sent_packets_;
+  }
+
+ private:
+  RtpHeaderExtensionMap receivers_extensions_;
+  std::vector<RtpPacketReceived> sent_packets_;
+};
+
+class TestRtpSenderVideo : public RTPSenderVideo {
+ public:
+  TestRtpSenderVideo(Clock* clock,
+                     RTPSender* rtp_sender,
+                     const FieldTrialsView& field_trials)
+      : RTPSenderVideo([&] {
+          Config config;
+          config.clock = clock;
+          config.rtp_sender = rtp_sender;
+          config.field_trials = &field_trials;
+          return config;
+        }()) {}
+  ~TestRtpSenderVideo() override {}
+
+  bool AllowRetransmission(const RTPVideoHeader& header,
+                           int32_t retransmission_settings,
+                           int64_t expected_retransmission_time_ms) {
+    return RTPSenderVideo::AllowRetransmission(GetTemporalId(header),
+                                               retransmission_settings,
+                                               expected_retransmission_time_ms);
+  }
+};
+
+class FieldTrials : public FieldTrialsRegistry {
+ public:
+  FieldTrials() : include_capture_clock_offset_(false) {}
+
+  void set_include_capture_clock_offset(bool include_capture_clock_offset) {
+    include_capture_clock_offset_ = include_capture_clock_offset;
+  }
+
+ private:
+  std::string GetValue(absl::string_view key) const override {
+    if (key == "WebRTC-IncludeCaptureClockOffset") {
+      return include_capture_clock_offset_ ? "" : "Disabled";
+    }
+    return "";
+  }
+
+  bool include_capture_clock_offset_;
+};
+
+class RtpSenderVideoTest : public ::testing::Test {
+ public:
+  RtpSenderVideoTest()
+      : fake_clock_(kStartTime),
+        retransmission_rate_limiter_(&fake_clock_, 1000),
+        rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
+          RtpRtcpInterface::Configuration config;
+          config.clock = &fake_clock_;
+          config.outgoing_transport = &transport_;
+          config.retransmission_rate_limiter = &retransmission_rate_limiter_;
+          config.field_trials = &field_trials_;
+          config.local_media_ssrc = kSsrc;
+          config.rtx_send_ssrc = kRtxSsrc;
+          config.rid = "rid";
+          return config;
+        }())),
+        rtp_sender_video_(
+            std::make_unique<TestRtpSenderVideo>(&fake_clock_,
+                                                 rtp_module_->RtpSender(),
+                                                 field_trials_)) {
+    rtp_module_->SetSequenceNumber(kSeqNum);
+    rtp_module_->SetStartTimestamp(0);
+  }
+
+  void UsesMinimalVp8DescriptorWhenGenericFrameDescriptorExtensionIsUsed(
+      int version);
+
+ protected:
+  rtc::AutoThread main_thread_;
+  const RtpRtcpInterface::Configuration config_;
+  FieldTrials field_trials_;
+  SimulatedClock fake_clock_;
+  LoopbackTransportTest transport_;
+  RateLimiter retransmission_rate_limiter_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_module_;
+  std::unique_ptr<TestRtpSenderVideo> rtp_sender_video_;
+};
+
+TEST_F(RtpSenderVideoTest, KeyFrameHasCVO) {
+  uint8_t kFrame[kMaxPacketLength];
+  rtp_module_->RegisterRtpHeaderExtension(VideoOrientation::Uri(),
+                                          kVideoRotationExtensionId);
+
+  RTPVideoHeader hdr;
+  hdr.rotation = kVideoRotation_0;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  VideoRotation rotation;
+  EXPECT_TRUE(
+      transport_.last_sent_packet().GetExtension<VideoOrientation>(&rotation));
+  EXPECT_EQ(kVideoRotation_0, rotation);
+}
+
+TEST_F(RtpSenderVideoTest, TimingFrameHasPacketizationTimstampSet) {
+  uint8_t kFrame[kMaxPacketLength];
+  const int64_t kPacketizationTimeMs = 100;
+  const int64_t kEncodeStartDeltaMs = 10;
+  const int64_t kEncodeFinishDeltaMs = 50;
+  rtp_module_->RegisterRtpHeaderExtension(VideoTimingExtension::Uri(),
+                                          kVideoTimingExtensionId);
+
+  const int64_t kCaptureTimestamp = fake_clock_.TimeInMilliseconds();
+
+  RTPVideoHeader hdr;
+  hdr.video_timing.flags = VideoSendTiming::kTriggeredByTimer;
+  hdr.video_timing.encode_start_delta_ms = kEncodeStartDeltaMs;
+  hdr.video_timing.encode_finish_delta_ms = kEncodeFinishDeltaMs;
+
+  fake_clock_.AdvanceTimeMilliseconds(kPacketizationTimeMs);
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, kCaptureTimestamp,
+                               kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  VideoSendTiming timing;
+  EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
+      &timing));
+  EXPECT_EQ(kPacketizationTimeMs, timing.packetization_finish_delta_ms);
+  EXPECT_EQ(kEncodeStartDeltaMs, timing.encode_start_delta_ms);
+  EXPECT_EQ(kEncodeFinishDeltaMs, timing.encode_finish_delta_ms);
+}
+
+TEST_F(RtpSenderVideoTest, DeltaFrameHasCVOWhenChanged) {
+  uint8_t kFrame[kMaxPacketLength];
+  rtp_module_->RegisterRtpHeaderExtension(VideoOrientation::Uri(),
+                                          kVideoRotationExtensionId);
+
+  RTPVideoHeader hdr;
+  hdr.rotation = kVideoRotation_90;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  EXPECT_TRUE(
+      rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                                   kDefaultExpectedRetransmissionTimeMs));
+
+  hdr.rotation = kVideoRotation_0;
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  EXPECT_TRUE(
+      rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp + 1, 0, kFrame,
+                                   hdr, kDefaultExpectedRetransmissionTimeMs));
+
+  VideoRotation rotation;
+  EXPECT_TRUE(
+      transport_.last_sent_packet().GetExtension<VideoOrientation>(&rotation));
+  EXPECT_EQ(kVideoRotation_0, rotation);
+}
+
+TEST_F(RtpSenderVideoTest, DeltaFrameHasCVOWhenNonZero) {
+  uint8_t kFrame[kMaxPacketLength];
+  rtp_module_->RegisterRtpHeaderExtension(VideoOrientation::Uri(),
+                                          kVideoRotationExtensionId);
+
+  RTPVideoHeader hdr;
+  hdr.rotation = kVideoRotation_90;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  EXPECT_TRUE(
+      rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                                   kDefaultExpectedRetransmissionTimeMs));
+
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  EXPECT_TRUE(
+      rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp + 1, 0, kFrame,
+                                   hdr, kDefaultExpectedRetransmissionTimeMs));
+
+  VideoRotation rotation;
+  EXPECT_TRUE(
+      transport_.last_sent_packet().GetExtension<VideoOrientation>(&rotation));
+  EXPECT_EQ(kVideoRotation_90, rotation);
+}
+
+// Make sure rotation is parsed correctly when the Camera (C) and Flip (F) bits
+// are set in the CVO byte.
+TEST_F(RtpSenderVideoTest, SendVideoWithCameraAndFlipCVO) {
+  // Test extracting rotation when Camera (C) and Flip (F) bits are zero.
+  EXPECT_EQ(kVideoRotation_0, ConvertCVOByteToVideoRotation(0));
+  EXPECT_EQ(kVideoRotation_90, ConvertCVOByteToVideoRotation(1));
+  EXPECT_EQ(kVideoRotation_180, ConvertCVOByteToVideoRotation(2));
+  EXPECT_EQ(kVideoRotation_270, ConvertCVOByteToVideoRotation(3));
+  // Test extracting rotation when Camera (C) and Flip (F) bits are set.
+  const int flip_bit = 1 << 2;
+  const int camera_bit = 1 << 3;
+  EXPECT_EQ(kVideoRotation_0,
+            ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 0));
+  EXPECT_EQ(kVideoRotation_90,
+            ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 1));
+  EXPECT_EQ(kVideoRotation_180,
+            ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 2));
+  EXPECT_EQ(kVideoRotation_270,
+            ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 3));
+}
+
+TEST_F(RtpSenderVideoTest, RetransmissionTypesGeneric) {
+  RTPVideoHeader header;
+  header.codec = kVideoCodecGeneric;
+
+  EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitOff, kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitBaseLayer, kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitHigherLayers, kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+      header, kConditionallyRetransmitHigherLayers,
+      kDefaultExpectedRetransmissionTimeMs));
+}
+
+TEST_F(RtpSenderVideoTest, RetransmissionTypesH264) {
+  RTPVideoHeader header;
+  header.video_type_header.emplace<RTPVideoHeaderH264>().packetization_mode =
+      H264PacketizationMode::NonInterleaved;
+  header.codec = kVideoCodecH264;
+
+  EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitOff, kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitBaseLayer, kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitHigherLayers, kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+      header, kConditionallyRetransmitHigherLayers,
+      kDefaultExpectedRetransmissionTimeMs));
+}
+
+TEST_F(RtpSenderVideoTest, RetransmissionTypesVP8BaseLayer) {
+  RTPVideoHeader header;
+  header.codec = kVideoCodecVP8;
+  auto& vp8_header = header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.temporalIdx = 0;
+
+  EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitOff, kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitBaseLayer, kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitHigherLayers, kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitHigherLayers | kRetransmitBaseLayer,
+      kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
+      header, kConditionallyRetransmitHigherLayers,
+      kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+      header, kRetransmitBaseLayer | kConditionallyRetransmitHigherLayers,
+      kDefaultExpectedRetransmissionTimeMs));
+}
+
+TEST_F(RtpSenderVideoTest, RetransmissionTypesVP8HigherLayers) {
+  RTPVideoHeader header;
+  header.codec = kVideoCodecVP8;
+
+  auto& vp8_header = header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  for (int tid = 1; tid <= kMaxTemporalStreams; ++tid) {
+    vp8_header.temporalIdx = tid;
+
+    EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
+        header, kRetransmitOff, kDefaultExpectedRetransmissionTimeMs));
+    EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
+        header, kRetransmitBaseLayer, kDefaultExpectedRetransmissionTimeMs));
+    EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+        header, kRetransmitHigherLayers, kDefaultExpectedRetransmissionTimeMs));
+    EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+        header, kRetransmitHigherLayers | kRetransmitBaseLayer,
+        kDefaultExpectedRetransmissionTimeMs));
+  }
+}
+
+TEST_F(RtpSenderVideoTest, RetransmissionTypesVP9) {
+  RTPVideoHeader header;
+  header.codec = kVideoCodecVP9;
+
+  auto& vp9_header = header.video_type_header.emplace<RTPVideoHeaderVP9>();
+  for (int tid = 1; tid <= kMaxTemporalStreams; ++tid) {
+    vp9_header.temporal_idx = tid;
+
+    EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
+        header, kRetransmitOff, kDefaultExpectedRetransmissionTimeMs));
+    EXPECT_FALSE(rtp_sender_video_->AllowRetransmission(
+        header, kRetransmitBaseLayer, kDefaultExpectedRetransmissionTimeMs));
+    EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+        header, kRetransmitHigherLayers, kDefaultExpectedRetransmissionTimeMs));
+    EXPECT_TRUE(rtp_sender_video_->AllowRetransmission(
+        header, kRetransmitHigherLayers | kRetransmitBaseLayer,
+        kDefaultExpectedRetransmissionTimeMs));
+  }
+}
+
+TEST_F(RtpSenderVideoTest, ConditionalRetransmit) {
+  const int64_t kFrameIntervalMs = 33;
+  const int64_t kRttMs = (kFrameIntervalMs * 3) / 2;
+  const uint8_t kSettings =
+      kRetransmitBaseLayer | kConditionallyRetransmitHigherLayers;
+
+  // Insert VP8 frames for all temporal layers, but stop before the final index.
+  RTPVideoHeader header;
+  header.codec = kVideoCodecVP8;
+
+  // Fill averaging window to prevent rounding errors.
+  constexpr int kNumRepetitions =
+      (RTPSenderVideo::kTLRateWindowSizeMs + (kFrameIntervalMs / 2)) /
+      kFrameIntervalMs;
+  constexpr int kPattern[] = {0, 2, 1, 2};
+  auto& vp8_header = header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  for (size_t i = 0; i < arraysize(kPattern) * kNumRepetitions; ++i) {
+    vp8_header.temporalIdx = kPattern[i % arraysize(kPattern)];
+    rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs);
+    fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+  }
+
+  // Since we're at the start of the pattern, the next expected frame in TL0 is
+  // right now. We will wait at most one expected retransmission time before
+  // acknowledging that it did not arrive, which means this frame and the next
+  // will not be retransmitted.
+  vp8_header.temporalIdx = 1;
+  EXPECT_FALSE(
+      rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+  EXPECT_FALSE(
+      rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+
+  // The TL0 frame did not arrive. So allow retransmission.
+  EXPECT_TRUE(
+      rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+
+  // Insert a frame for TL2. We just had frame in TL1, so the next one there is
+  // in three frames away. TL0 is still too far in the past. So, allow
+  // retransmission.
+  vp8_header.temporalIdx = 2;
+  EXPECT_TRUE(
+      rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+
+  // Another TL2, next in TL1 is two frames away. Allow again.
+  EXPECT_TRUE(
+      rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+
+  // Yet another TL2, next in TL1 is now only one frame away, so don't store
+  // for retransmission.
+  EXPECT_FALSE(
+      rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs));
+}
+
+TEST_F(RtpSenderVideoTest, ConditionalRetransmitLimit) {
+  const int64_t kFrameIntervalMs = 200;
+  const int64_t kRttMs = (kFrameIntervalMs * 3) / 2;
+  const int32_t kSettings =
+      kRetransmitBaseLayer | kConditionallyRetransmitHigherLayers;
+
+  // Insert VP8 frames for all temporal layers, but stop before the final index.
+  RTPVideoHeader header;
+  header.codec = kVideoCodecVP8;
+
+  // Fill averaging window to prevent rounding errors.
+  constexpr int kNumRepetitions =
+      (RTPSenderVideo::kTLRateWindowSizeMs + (kFrameIntervalMs / 2)) /
+      kFrameIntervalMs;
+  constexpr int kPattern[] = {0, 2, 2, 2};
+  auto& vp8_header = header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  for (size_t i = 0; i < arraysize(kPattern) * kNumRepetitions; ++i) {
+    vp8_header.temporalIdx = kPattern[i % arraysize(kPattern)];
+
+    rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs);
+    fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+  }
+
+  // Since we're at the start of the pattern, the next expected frame will be
+  // right now in TL0. Put it in TL1 instead. Regular rules would dictate that
+  // we don't store for retransmission because we expect a frame in a lower
+  // layer, but that last frame in TL1 was a long time ago in absolute terms,
+  // so allow retransmission anyway.
+  vp8_header.temporalIdx = 1;
+  EXPECT_TRUE(
+      rtp_sender_video_->AllowRetransmission(header, kSettings, kRttMs));
+}
+
+TEST_F(RtpSenderVideoTest,
+       ReservesEnoughSpaceForRtxPacketWhenMidAndRsidAreRegistered) {
+  constexpr int kMediaPayloadId = 100;
+  constexpr int kRtxPayloadId = 101;
+  constexpr size_t kMaxPacketSize = 1'000;
+
+  rtp_module_->SetMaxRtpPacketSize(kMaxPacketSize);
+  rtp_module_->RegisterRtpHeaderExtension(RtpMid::Uri(), 1);
+  rtp_module_->RegisterRtpHeaderExtension(RtpStreamId::Uri(), 2);
+  rtp_module_->RegisterRtpHeaderExtension(RepairedRtpStreamId::Uri(), 3);
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteSendTime::Uri(), 4);
+  rtp_module_->SetMid("long_mid");
+  rtp_module_->SetRtxSendPayloadType(kRtxPayloadId, kMediaPayloadId);
+  rtp_module_->SetStorePacketsStatus(/*enable=*/true, 10);
+  rtp_module_->SetRtxSendStatus(kRtxRetransmitted);
+
+  RTPVideoHeader header;
+  header.codec = kVideoCodecVP8;
+  header.frame_type = VideoFrameType::kVideoFrameDelta;
+  auto& vp8_header = header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.temporalIdx = 0;
+
+  uint8_t kPayload[kMaxPacketSize] = {};
+  EXPECT_TRUE(rtp_sender_video_->SendVideo(
+      kMediaPayloadId, /*codec_type=*/kVideoCodecVP8, /*rtp_timestamp=*/0,
+      /*capture_time_ms=*/1'000, kPayload, header,
+      /*expected_retransmission_time_ms=*/absl::nullopt, /*csrcs=*/{}));
+  ASSERT_THAT(transport_.sent_packets(), Not(IsEmpty()));
+  // Ack media ssrc, but not rtx ssrc.
+  rtcp::ReceiverReport rr;
+  rtcp::ReportBlock rb;
+  rb.SetMediaSsrc(kSsrc);
+  rb.SetExtHighestSeqNum(transport_.last_sent_packet().SequenceNumber());
+  rr.AddReportBlock(rb);
+  rtp_module_->IncomingRtcpPacket(rr.Build());
+
+  // Test for various frame size close to `kMaxPacketSize` to catch edge cases
+  // when rtx packet barely fit.
+  for (size_t frame_size = 800; frame_size < kMaxPacketSize; ++frame_size) {
+    SCOPED_TRACE(frame_size);
+    rtc::ArrayView<const uint8_t> payload(kPayload, frame_size);
+
+    EXPECT_TRUE(rtp_sender_video_->SendVideo(
+        kMediaPayloadId, /*codec_type=*/kVideoCodecVP8, /*rtp_timestamp=*/0,
+        /*capture_time_ms=*/1'000, payload, header,
+        /*expected_retransmission_time_ms=*/1'000, /*csrcs=*/{}));
+    const RtpPacketReceived& media_packet = transport_.last_sent_packet();
+    EXPECT_EQ(media_packet.Ssrc(), kSsrc);
+
+    rtcp::Nack nack;
+    nack.SetMediaSsrc(kSsrc);
+    nack.SetPacketIds({media_packet.SequenceNumber()});
+    rtp_module_->IncomingRtcpPacket(nack.Build());
+
+    const RtpPacketReceived& rtx_packet = transport_.last_sent_packet();
+    EXPECT_EQ(rtx_packet.Ssrc(), kRtxSsrc);
+    EXPECT_LE(rtx_packet.size(), kMaxPacketSize);
+  }
+}
+
+TEST_F(RtpSenderVideoTest, SendsDependencyDescriptorWhenVideoStructureIsSet) {
+  const int64_t kFrameId = 100000;
+  uint8_t kFrame[100];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
+  FrameDependencyStructure video_structure;
+  video_structure.num_decode_targets = 2;
+  video_structure.templates = {
+      FrameDependencyTemplate().S(0).T(0).Dtis("SS"),
+      FrameDependencyTemplate().S(1).T(0).Dtis("-S"),
+      FrameDependencyTemplate().S(1).T(1).Dtis("-D"),
+  };
+  rtp_sender_video_->SetVideoStructure(&video_structure);
+
+  // Send key frame.
+  RTPVideoHeader hdr;
+  RTPVideoHeader::GenericDescriptorInfo& generic = hdr.generic.emplace();
+  generic.frame_id = kFrameId;
+  generic.temporal_index = 0;
+  generic.spatial_index = 0;
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch,
+                                       DecodeTargetIndication::kSwitch};
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  ASSERT_EQ(transport_.packets_sent(), 1);
+  DependencyDescriptor descriptor_key;
+  ASSERT_TRUE(transport_.last_sent_packet()
+                  .GetExtension<RtpDependencyDescriptorExtension>(
+                      nullptr, &descriptor_key));
+  ASSERT_TRUE(descriptor_key.attached_structure);
+  EXPECT_EQ(descriptor_key.attached_structure->num_decode_targets, 2);
+  EXPECT_THAT(descriptor_key.attached_structure->templates, SizeIs(3));
+  EXPECT_EQ(descriptor_key.frame_number, kFrameId & 0xFFFF);
+  EXPECT_EQ(descriptor_key.frame_dependencies.spatial_id, 0);
+  EXPECT_EQ(descriptor_key.frame_dependencies.temporal_id, 0);
+  EXPECT_EQ(descriptor_key.frame_dependencies.decode_target_indications,
+            generic.decode_target_indications);
+  EXPECT_THAT(descriptor_key.frame_dependencies.frame_diffs, IsEmpty());
+
+  // Send delta frame.
+  generic.frame_id = kFrameId + 1;
+  generic.temporal_index = 1;
+  generic.spatial_index = 1;
+  generic.dependencies = {kFrameId, kFrameId - 500};
+  generic.decode_target_indications = {DecodeTargetIndication::kNotPresent,
+                                       DecodeTargetIndication::kRequired};
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  EXPECT_EQ(transport_.packets_sent(), 2);
+  DependencyDescriptor descriptor_delta;
+  ASSERT_TRUE(
+      transport_.last_sent_packet()
+          .GetExtension<RtpDependencyDescriptorExtension>(
+              descriptor_key.attached_structure.get(), &descriptor_delta));
+  EXPECT_EQ(descriptor_delta.attached_structure, nullptr);
+  EXPECT_EQ(descriptor_delta.frame_number, (kFrameId + 1) & 0xFFFF);
+  EXPECT_EQ(descriptor_delta.frame_dependencies.spatial_id, 1);
+  EXPECT_EQ(descriptor_delta.frame_dependencies.temporal_id, 1);
+  EXPECT_EQ(descriptor_delta.frame_dependencies.decode_target_indications,
+            generic.decode_target_indications);
+  EXPECT_THAT(descriptor_delta.frame_dependencies.frame_diffs,
+              ElementsAre(1, 501));
+}
+
+TEST_F(RtpSenderVideoTest,
+       SkipsDependencyDescriptorOnDeltaFrameWhenFailedToAttachToKeyFrame) {
+  const int64_t kFrameId = 100000;
+  uint8_t kFrame[100];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
+  rtp_module_->SetExtmapAllowMixed(false);
+  FrameDependencyStructure video_structure;
+  video_structure.num_decode_targets = 2;
+  // Use many templates so that key dependency descriptor would be too large
+  // to fit into 16 bytes (max size of one byte header rtp header extension)
+  video_structure.templates = {
+      FrameDependencyTemplate().S(0).T(0).Dtis("SS"),
+      FrameDependencyTemplate().S(1).T(0).Dtis("-S"),
+      FrameDependencyTemplate().S(1).T(1).Dtis("-D").FrameDiffs({1, 2, 3, 4}),
+      FrameDependencyTemplate().S(1).T(1).Dtis("-D").FrameDiffs({2, 3, 4, 5}),
+      FrameDependencyTemplate().S(1).T(1).Dtis("-D").FrameDiffs({3, 4, 5, 6}),
+      FrameDependencyTemplate().S(1).T(1).Dtis("-D").FrameDiffs({4, 5, 6, 7}),
+  };
+  rtp_sender_video_->SetVideoStructure(&video_structure);
+
+  // Send key frame.
+  RTPVideoHeader hdr;
+  RTPVideoHeader::GenericDescriptorInfo& generic = hdr.generic.emplace();
+  generic.frame_id = kFrameId;
+  generic.temporal_index = 0;
+  generic.spatial_index = 0;
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch,
+                                       DecodeTargetIndication::kSwitch};
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  ASSERT_EQ(transport_.packets_sent(), 1);
+  DependencyDescriptor descriptor_key;
+  ASSERT_FALSE(transport_.last_sent_packet()
+                   .HasExtension<RtpDependencyDescriptorExtension>());
+
+  // Send delta frame.
+  generic.frame_id = kFrameId + 1;
+  generic.temporal_index = 1;
+  generic.spatial_index = 1;
+  generic.dependencies = {kFrameId, kFrameId - 500};
+  generic.decode_target_indications = {DecodeTargetIndication::kNotPresent,
+                                       DecodeTargetIndication::kRequired};
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  EXPECT_EQ(transport_.packets_sent(), 2);
+  EXPECT_FALSE(transport_.last_sent_packet()
+                   .HasExtension<RtpDependencyDescriptorExtension>());
+}
+
+TEST_F(RtpSenderVideoTest, PropagatesChainDiffsIntoDependencyDescriptor) {
+  const int64_t kFrameId = 100000;
+  uint8_t kFrame[100];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
+  FrameDependencyStructure video_structure;
+  video_structure.num_decode_targets = 2;
+  video_structure.num_chains = 1;
+  video_structure.decode_target_protected_by_chain = {0, 0};
+  video_structure.templates = {
+      FrameDependencyTemplate().S(0).T(0).Dtis("SS").ChainDiffs({1}),
+  };
+  rtp_sender_video_->SetVideoStructure(&video_structure);
+
+  RTPVideoHeader hdr;
+  RTPVideoHeader::GenericDescriptorInfo& generic = hdr.generic.emplace();
+  generic.frame_id = kFrameId;
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch,
+                                       DecodeTargetIndication::kSwitch};
+  generic.chain_diffs = {2};
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  ASSERT_EQ(transport_.packets_sent(), 1);
+  DependencyDescriptor descriptor_key;
+  ASSERT_TRUE(transport_.last_sent_packet()
+                  .GetExtension<RtpDependencyDescriptorExtension>(
+                      nullptr, &descriptor_key));
+  EXPECT_THAT(descriptor_key.frame_dependencies.chain_diffs,
+              ContainerEq(generic.chain_diffs));
+}
+
+TEST_F(RtpSenderVideoTest,
+       PropagatesActiveDecodeTargetsIntoDependencyDescriptor) {
+  const int64_t kFrameId = 100000;
+  uint8_t kFrame[100];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
+  FrameDependencyStructure video_structure;
+  video_structure.num_decode_targets = 2;
+  video_structure.num_chains = 1;
+  video_structure.decode_target_protected_by_chain = {0, 0};
+  video_structure.templates = {
+      FrameDependencyTemplate().S(0).T(0).Dtis("SS").ChainDiffs({1}),
+  };
+  rtp_sender_video_->SetVideoStructure(&video_structure);
+
+  RTPVideoHeader hdr;
+  RTPVideoHeader::GenericDescriptorInfo& generic = hdr.generic.emplace();
+  generic.frame_id = kFrameId;
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch,
+                                       DecodeTargetIndication::kSwitch};
+  generic.active_decode_targets = 0b01;
+  generic.chain_diffs = {1};
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  ASSERT_EQ(transport_.packets_sent(), 1);
+  DependencyDescriptor descriptor_key;
+  ASSERT_TRUE(transport_.last_sent_packet()
+                  .GetExtension<RtpDependencyDescriptorExtension>(
+                      nullptr, &descriptor_key));
+  EXPECT_EQ(descriptor_key.active_decode_targets_bitmask, 0b01u);
+}
+
+TEST_F(RtpSenderVideoTest,
+       SetDiffentVideoStructureAvoidsCollisionWithThePreviousStructure) {
+  const int64_t kFrameId = 100000;
+  uint8_t kFrame[100];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
+  FrameDependencyStructure video_structure1;
+  video_structure1.num_decode_targets = 2;
+  video_structure1.templates = {
+      FrameDependencyTemplate().S(0).T(0).Dtis("SS"),
+      FrameDependencyTemplate().S(0).T(1).Dtis("D-"),
+  };
+  FrameDependencyStructure video_structure2;
+  video_structure2.num_decode_targets = 2;
+  video_structure2.templates = {
+      FrameDependencyTemplate().S(0).T(0).Dtis("SS"),
+      FrameDependencyTemplate().S(0).T(1).Dtis("R-"),
+  };
+
+  // Send 1st key frame.
+  RTPVideoHeader hdr;
+  RTPVideoHeader::GenericDescriptorInfo& generic = hdr.generic.emplace();
+  generic.frame_id = kFrameId;
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch,
+                                       DecodeTargetIndication::kSwitch};
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SetVideoStructure(&video_structure1);
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  // Parse 1st extension.
+  ASSERT_EQ(transport_.packets_sent(), 1);
+  DependencyDescriptor descriptor_key1;
+  ASSERT_TRUE(transport_.last_sent_packet()
+                  .GetExtension<RtpDependencyDescriptorExtension>(
+                      nullptr, &descriptor_key1));
+  ASSERT_TRUE(descriptor_key1.attached_structure);
+
+  // Send the delta frame.
+  generic.frame_id = kFrameId + 1;
+  generic.temporal_index = 1;
+  generic.decode_target_indications = {DecodeTargetIndication::kDiscardable,
+                                       DecodeTargetIndication::kNotPresent};
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  ASSERT_EQ(transport_.packets_sent(), 2);
+  RtpPacket delta_packet = transport_.last_sent_packet();
+
+  // Send 2nd key frame.
+  generic.frame_id = kFrameId + 2;
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch,
+                                       DecodeTargetIndication::kSwitch};
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SetVideoStructure(&video_structure2);
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  // Parse the 2nd key frame.
+  ASSERT_EQ(transport_.packets_sent(), 3);
+  DependencyDescriptor descriptor_key2;
+  ASSERT_TRUE(transport_.last_sent_packet()
+                  .GetExtension<RtpDependencyDescriptorExtension>(
+                      nullptr, &descriptor_key2));
+  ASSERT_TRUE(descriptor_key2.attached_structure);
+
+  // Try to parse the 1st delta frame. It should parseble using the structure
+  // from the 1st key frame, but not using the structure from the 2nd key frame.
+  DependencyDescriptor descriptor_delta;
+  EXPECT_TRUE(delta_packet.GetExtension<RtpDependencyDescriptorExtension>(
+      descriptor_key1.attached_structure.get(), &descriptor_delta));
+  EXPECT_FALSE(delta_packet.GetExtension<RtpDependencyDescriptorExtension>(
+      descriptor_key2.attached_structure.get(), &descriptor_delta));
+}
+
+TEST_F(RtpSenderVideoTest,
+       AuthenticateVideoHeaderWhenDependencyDescriptorExtensionIsUsed) {
+  static constexpr size_t kFrameSize = 100;
+  uint8_t kFrame[kFrameSize] = {1, 2, 3, 4};
+
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpDependencyDescriptorExtension::Uri(), kDependencyDescriptorId);
+  auto encryptor = rtc::make_ref_counted<NiceMock<MockFrameEncryptor>>();
+  ON_CALL(*encryptor, GetMaxCiphertextByteSize).WillByDefault(ReturnArg<1>());
+  ON_CALL(*encryptor, Encrypt)
+      .WillByDefault(WithArgs<3, 5>(
+          [](rtc::ArrayView<const uint8_t> frame, size_t* bytes_written) {
+            *bytes_written = frame.size();
+            return 0;
+          }));
+  RTPSenderVideo::Config config;
+  config.clock = &fake_clock_;
+  config.rtp_sender = rtp_module_->RtpSender();
+  config.field_trials = &field_trials_;
+  config.frame_encryptor = encryptor.get();
+  RTPSenderVideo rtp_sender_video(config);
+
+  FrameDependencyStructure video_structure;
+  video_structure.num_decode_targets = 1;
+  video_structure.templates = {FrameDependencyTemplate().Dtis("S")};
+  rtp_sender_video.SetVideoStructure(&video_structure);
+
+  // Send key frame.
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  hdr.generic.emplace().decode_target_indications =
+      video_structure.templates[0].decode_target_indications;
+
+  EXPECT_CALL(*encryptor,
+              Encrypt(_, _, Not(IsEmpty()), ElementsAreArray(kFrame), _, _));
+  rtp_sender_video.SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                             kDefaultExpectedRetransmissionTimeMs);
+  // Double check packet with the dependency descriptor is sent.
+  ASSERT_EQ(transport_.packets_sent(), 1);
+  EXPECT_TRUE(transport_.last_sent_packet()
+                  .HasExtension<RtpDependencyDescriptorExtension>());
+}
+
+TEST_F(RtpSenderVideoTest, PopulateGenericFrameDescriptor) {
+  const int64_t kFrameId = 100000;
+  uint8_t kFrame[100];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpGenericFrameDescriptorExtension00::Uri(), kGenericDescriptorId);
+
+  RTPVideoHeader hdr;
+  RTPVideoHeader::GenericDescriptorInfo& generic = hdr.generic.emplace();
+  generic.frame_id = kFrameId;
+  generic.temporal_index = 3;
+  generic.spatial_index = 2;
+  generic.dependencies.push_back(kFrameId - 1);
+  generic.dependencies.push_back(kFrameId - 500);
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  RtpGenericFrameDescriptor descriptor_wire;
+  EXPECT_EQ(1, transport_.packets_sent());
+  ASSERT_TRUE(transport_.last_sent_packet()
+                  .GetExtension<RtpGenericFrameDescriptorExtension00>(
+                      &descriptor_wire));
+  EXPECT_EQ(static_cast<uint16_t>(generic.frame_id), descriptor_wire.FrameId());
+  EXPECT_EQ(generic.temporal_index, descriptor_wire.TemporalLayer());
+  EXPECT_THAT(descriptor_wire.FrameDependenciesDiffs(), ElementsAre(1, 500));
+  EXPECT_EQ(descriptor_wire.SpatialLayersBitmask(), 0b0000'0100);
+}
+
+void RtpSenderVideoTest::
+    UsesMinimalVp8DescriptorWhenGenericFrameDescriptorExtensionIsUsed(
+        int version) {
+  const int64_t kFrameId = 100000;
+  const size_t kFrameSize = 100;
+  uint8_t kFrame[kFrameSize];
+
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpGenericFrameDescriptorExtension00::Uri(), kGenericDescriptorId);
+
+  RTPVideoHeader hdr;
+  hdr.codec = kVideoCodecVP8;
+  RTPVideoHeaderVP8& vp8 = hdr.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8.pictureId = kFrameId % 0X7FFF;
+  vp8.tl0PicIdx = 13;
+  vp8.temporalIdx = 1;
+  vp8.keyIdx = 2;
+  RTPVideoHeader::GenericDescriptorInfo& generic = hdr.generic.emplace();
+  generic.frame_id = kFrameId;
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, VideoCodecType::kVideoCodecVP8,
+                               kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  ASSERT_EQ(transport_.packets_sent(), 1);
+  // Expect only minimal 1-byte vp8 descriptor was generated.
+  EXPECT_EQ(transport_.last_sent_packet().payload_size(), 1 + kFrameSize);
+}
+
+TEST_F(RtpSenderVideoTest,
+       UsesMinimalVp8DescriptorWhenGenericFrameDescriptorExtensionIsUsed00) {
+  UsesMinimalVp8DescriptorWhenGenericFrameDescriptorExtensionIsUsed(0);
+}
+
+TEST_F(RtpSenderVideoTest,
+       UsesMinimalVp8DescriptorWhenGenericFrameDescriptorExtensionIsUsed01) {
+  UsesMinimalVp8DescriptorWhenGenericFrameDescriptorExtensionIsUsed(1);
+}
+
+TEST_F(RtpSenderVideoTest, VideoLayersAllocationWithResolutionSentOnKeyFrames) {
+  const size_t kFrameSize = 100;
+  uint8_t kFrame[kFrameSize];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpVideoLayersAllocationExtension::Uri(),
+      kVideoLayersAllocationExtensionId);
+
+  VideoLayersAllocation allocation;
+  VideoLayersAllocation::SpatialLayer layer;
+  layer.width = 360;
+  layer.height = 180;
+  layer.target_bitrate_per_temporal_layer.push_back(
+      DataRate::KilobitsPerSec(50));
+  allocation.resolution_and_frame_rate_is_valid = true;
+  allocation.active_spatial_layers.push_back(layer);
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  VideoLayersAllocation sent_allocation;
+  EXPECT_TRUE(
+      transport_.last_sent_packet()
+          .GetExtension<RtpVideoLayersAllocationExtension>(&sent_allocation));
+  EXPECT_THAT(sent_allocation.active_spatial_layers, ElementsAre(layer));
+
+  // Next key frame also have the allocation.
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  EXPECT_TRUE(
+      transport_.last_sent_packet()
+          .GetExtension<RtpVideoLayersAllocationExtension>(&sent_allocation));
+}
+
+TEST_F(RtpSenderVideoTest,
+       VideoLayersAllocationWithoutResolutionSentOnDeltaWhenUpdated) {
+  const size_t kFrameSize = 100;
+  uint8_t kFrame[kFrameSize];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpVideoLayersAllocationExtension::Uri(),
+      kVideoLayersAllocationExtensionId);
+
+  VideoLayersAllocation allocation;
+  VideoLayersAllocation::SpatialLayer layer;
+  layer.width = 360;
+  layer.height = 180;
+  allocation.resolution_and_frame_rate_is_valid = true;
+  layer.target_bitrate_per_temporal_layer.push_back(
+      DataRate::KilobitsPerSec(50));
+  allocation.active_spatial_layers.push_back(layer);
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  EXPECT_TRUE(transport_.last_sent_packet()
+                  .HasExtension<RtpVideoLayersAllocationExtension>());
+
+  // No allocation sent on delta frame unless it has been updated.
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  EXPECT_FALSE(transport_.last_sent_packet()
+                   .HasExtension<RtpVideoLayersAllocationExtension>());
+
+  // Update the allocation.
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  VideoLayersAllocation sent_allocation;
+  EXPECT_TRUE(
+      transport_.last_sent_packet()
+          .GetExtension<RtpVideoLayersAllocationExtension>(&sent_allocation));
+  ASSERT_THAT(sent_allocation.active_spatial_layers, SizeIs(1));
+  EXPECT_FALSE(sent_allocation.resolution_and_frame_rate_is_valid);
+  EXPECT_THAT(sent_allocation.active_spatial_layers[0]
+                  .target_bitrate_per_temporal_layer,
+              SizeIs(1));
+}
+
+TEST_F(RtpSenderVideoTest,
+       VideoLayersAllocationWithResolutionSentOnDeltaWhenSpatialLayerAdded) {
+  const size_t kFrameSize = 100;
+  uint8_t kFrame[kFrameSize];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpVideoLayersAllocationExtension::Uri(),
+      kVideoLayersAllocationExtensionId);
+
+  VideoLayersAllocation allocation;
+  allocation.resolution_and_frame_rate_is_valid = true;
+  VideoLayersAllocation::SpatialLayer layer;
+  layer.width = 360;
+  layer.height = 180;
+  layer.spatial_id = 0;
+  layer.target_bitrate_per_temporal_layer.push_back(
+      DataRate::KilobitsPerSec(50));
+  allocation.active_spatial_layers.push_back(layer);
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  ASSERT_TRUE(transport_.last_sent_packet()
+                  .HasExtension<RtpVideoLayersAllocationExtension>());
+
+  // Update the allocation.
+  layer.width = 640;
+  layer.height = 320;
+  layer.spatial_id = 1;
+  layer.target_bitrate_per_temporal_layer.push_back(
+      DataRate::KilobitsPerSec(100));
+  allocation.active_spatial_layers.push_back(layer);
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  VideoLayersAllocation sent_allocation;
+  EXPECT_TRUE(
+      transport_.last_sent_packet()
+          .GetExtension<RtpVideoLayersAllocationExtension>(&sent_allocation));
+  EXPECT_THAT(sent_allocation.active_spatial_layers, SizeIs(2));
+  EXPECT_TRUE(sent_allocation.resolution_and_frame_rate_is_valid);
+}
+
+TEST_F(RtpSenderVideoTest,
+       VideoLayersAllocationWithResolutionSentOnLargeFrameRateChange) {
+  const size_t kFrameSize = 100;
+  uint8_t kFrame[kFrameSize];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpVideoLayersAllocationExtension::Uri(),
+      kVideoLayersAllocationExtensionId);
+
+  VideoLayersAllocation allocation;
+  allocation.resolution_and_frame_rate_is_valid = true;
+  VideoLayersAllocation::SpatialLayer layer;
+  layer.width = 360;
+  layer.height = 180;
+  layer.spatial_id = 0;
+  layer.frame_rate_fps = 10;
+  layer.target_bitrate_per_temporal_layer.push_back(
+      DataRate::KilobitsPerSec(50));
+  allocation.active_spatial_layers.push_back(layer);
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  ASSERT_TRUE(transport_.last_sent_packet()
+                  .HasExtension<RtpVideoLayersAllocationExtension>());
+
+  // Update frame rate only.
+  allocation.active_spatial_layers[0].frame_rate_fps = 20;
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  VideoLayersAllocation sent_allocation;
+  EXPECT_TRUE(
+      transport_.last_sent_packet()
+          .GetExtension<RtpVideoLayersAllocationExtension>(&sent_allocation));
+  ASSERT_TRUE(sent_allocation.resolution_and_frame_rate_is_valid);
+  EXPECT_EQ(sent_allocation.active_spatial_layers[0].frame_rate_fps, 20);
+}
+
+TEST_F(RtpSenderVideoTest,
+       VideoLayersAllocationWithoutResolutionSentOnSmallFrameRateChange) {
+  const size_t kFrameSize = 100;
+  uint8_t kFrame[kFrameSize];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpVideoLayersAllocationExtension::Uri(),
+      kVideoLayersAllocationExtensionId);
+
+  VideoLayersAllocation allocation;
+  allocation.resolution_and_frame_rate_is_valid = true;
+  VideoLayersAllocation::SpatialLayer layer;
+  layer.width = 360;
+  layer.height = 180;
+  layer.spatial_id = 0;
+  layer.frame_rate_fps = 10;
+  layer.target_bitrate_per_temporal_layer.push_back(
+      DataRate::KilobitsPerSec(50));
+  allocation.active_spatial_layers.push_back(layer);
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  ASSERT_TRUE(transport_.last_sent_packet()
+                  .HasExtension<RtpVideoLayersAllocationExtension>());
+
+  // Update frame rate slightly.
+  allocation.active_spatial_layers[0].frame_rate_fps = 9;
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  VideoLayersAllocation sent_allocation;
+  EXPECT_TRUE(
+      transport_.last_sent_packet()
+          .GetExtension<RtpVideoLayersAllocationExtension>(&sent_allocation));
+  EXPECT_FALSE(sent_allocation.resolution_and_frame_rate_is_valid);
+}
+
+TEST_F(RtpSenderVideoTest, VideoLayersAllocationSentOnDeltaFramesOnlyOnUpdate) {
+  const size_t kFrameSize = 100;
+  uint8_t kFrame[kFrameSize];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpVideoLayersAllocationExtension::Uri(),
+      kVideoLayersAllocationExtensionId);
+
+  VideoLayersAllocation allocation;
+  VideoLayersAllocation::SpatialLayer layer;
+  layer.width = 360;
+  layer.height = 180;
+  layer.target_bitrate_per_temporal_layer.push_back(
+      DataRate::KilobitsPerSec(50));
+  allocation.active_spatial_layers.push_back(layer);
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  VideoLayersAllocation sent_allocation;
+  EXPECT_TRUE(
+      transport_.last_sent_packet()
+          .GetExtension<RtpVideoLayersAllocationExtension>(&sent_allocation));
+  EXPECT_THAT(sent_allocation.active_spatial_layers, SizeIs(1));
+
+  // VideoLayersAllocation not sent on the next delta frame.
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  EXPECT_FALSE(transport_.last_sent_packet()
+                   .HasExtension<RtpVideoLayersAllocationExtension>());
+
+  // Update allocation. VideoLayesAllocation should be sent on the next frame.
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  EXPECT_TRUE(
+      transport_.last_sent_packet()
+          .GetExtension<RtpVideoLayersAllocationExtension>(&sent_allocation));
+}
+
+TEST_F(RtpSenderVideoTest, VideoLayersAllocationNotSentOnHigherTemporalLayers) {
+  const size_t kFrameSize = 100;
+  uint8_t kFrame[kFrameSize];
+  rtp_module_->RegisterRtpHeaderExtension(
+      RtpVideoLayersAllocationExtension::Uri(),
+      kVideoLayersAllocationExtensionId);
+
+  VideoLayersAllocation allocation;
+  allocation.resolution_and_frame_rate_is_valid = true;
+  VideoLayersAllocation::SpatialLayer layer;
+  layer.width = 360;
+  layer.height = 180;
+  layer.target_bitrate_per_temporal_layer.push_back(
+      DataRate::KilobitsPerSec(50));
+  allocation.active_spatial_layers.push_back(layer);
+  rtp_sender_video_->SetVideoLayersAllocation(allocation);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  hdr.codec = VideoCodecType::kVideoCodecVP8;
+  auto& vp8_header = hdr.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.temporalIdx = 1;
+
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  EXPECT_FALSE(transport_.last_sent_packet()
+                   .HasExtension<RtpVideoLayersAllocationExtension>());
+
+  // Send a delta frame on tl0.
+  vp8_header.temporalIdx = 0;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  EXPECT_TRUE(transport_.last_sent_packet()
+                  .HasExtension<RtpVideoLayersAllocationExtension>());
+}
+
+TEST_F(RtpSenderVideoTest, AbsoluteCaptureTime) {
+  constexpr int64_t kAbsoluteCaptureTimestampMs = 12345678;
+  uint8_t kFrame[kMaxPacketLength];
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
+                                          kAbsoluteCaptureTimeExtensionId);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp,
+                               kAbsoluteCaptureTimestampMs, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+
+  // It is expected that one and only one of the packets sent on this video
+  // frame has absolute capture time header extension.
+  for (const RtpPacketReceived& packet : transport_.sent_packets()) {
+    if (absolute_capture_time.has_value()) {
+      EXPECT_FALSE(packet.HasExtension<AbsoluteCaptureTimeExtension>());
+    } else {
+      absolute_capture_time =
+          packet.GetExtension<AbsoluteCaptureTimeExtension>();
+    }
+  }
+
+  // Verify the capture timestamp and that the clock offset is not set.
+  ASSERT_TRUE(absolute_capture_time.has_value());
+  EXPECT_EQ(
+      absolute_capture_time->absolute_capture_timestamp,
+      Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
+          kAbsoluteCaptureTimestampMs)));
+  EXPECT_FALSE(
+      absolute_capture_time->estimated_capture_clock_offset.has_value());
+}
+
+TEST_F(RtpSenderVideoTest,
+       AbsoluteCaptureTimeNotForwardedWhenImageHasNoCaptureTime) {
+  uint8_t kFrame[kMaxPacketLength];
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
+                                          kAbsoluteCaptureTimeExtensionId);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp,
+                               /*capture_time_ms=*/0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  // No absolute capture time should be set as the capture_time_ms was the
+  // default value.
+  for (const RtpPacketReceived& packet : transport_.sent_packets()) {
+    EXPECT_FALSE(packet.HasExtension<AbsoluteCaptureTimeExtension>());
+  }
+}
+
+// Essentially the same test as AbsoluteCaptureTime but with a field trial.
+// After the field trial is experimented, we will remove AbsoluteCaptureTime.
+TEST_F(RtpSenderVideoTest, AbsoluteCaptureTimeWithCaptureClockOffset) {
+  field_trials_.set_include_capture_clock_offset(true);
+  rtp_sender_video_ = std::make_unique<TestRtpSenderVideo>(
+      &fake_clock_, rtp_module_->RtpSender(), field_trials_);
+
+  constexpr int64_t kAbsoluteCaptureTimestampMs = 12345678;
+  uint8_t kFrame[kMaxPacketLength];
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
+                                          kAbsoluteCaptureTimeExtensionId);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp,
+                               kAbsoluteCaptureTimestampMs, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+
+  // It is expected that one and only one of the packets sent on this video
+  // frame has absolute capture time header extension.
+  for (const RtpPacketReceived& packet : transport_.sent_packets()) {
+    if (absolute_capture_time.has_value()) {
+      EXPECT_FALSE(packet.HasExtension<AbsoluteCaptureTimeExtension>());
+    } else {
+      absolute_capture_time =
+          packet.GetExtension<AbsoluteCaptureTimeExtension>();
+    }
+  }
+
+  // Verify the capture timestamp and that the clock offset is set to zero.
+  ASSERT_TRUE(absolute_capture_time.has_value());
+  EXPECT_EQ(
+      absolute_capture_time->absolute_capture_timestamp,
+      Int64MsToUQ32x32(fake_clock_.ConvertTimestampToNtpTimeInMilliseconds(
+          kAbsoluteCaptureTimestampMs)));
+  EXPECT_EQ(absolute_capture_time->estimated_capture_clock_offset, 0);
+}
+
+TEST_F(RtpSenderVideoTest, AbsoluteCaptureTimeWithExtensionProvided) {
+  constexpr AbsoluteCaptureTime kAbsoluteCaptureTime = {
+      123,
+      absl::optional<int64_t>(456),
+  };
+  uint8_t kFrame[kMaxPacketLength];
+  rtp_module_->RegisterRtpHeaderExtension(AbsoluteCaptureTimeExtension::Uri(),
+                                          kAbsoluteCaptureTimeExtensionId);
+
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  hdr.absolute_capture_time = kAbsoluteCaptureTime;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp,
+                               /*capture_time_ms=*/789, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+
+  absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+
+  // It is expected that one and only one of the packets sent on this video
+  // frame has absolute capture time header extension.
+  for (const RtpPacketReceived& packet : transport_.sent_packets()) {
+    if (absolute_capture_time.has_value()) {
+      EXPECT_FALSE(packet.HasExtension<AbsoluteCaptureTimeExtension>());
+    } else {
+      absolute_capture_time =
+          packet.GetExtension<AbsoluteCaptureTimeExtension>();
+    }
+  }
+
+  // Verify the extension.
+  EXPECT_EQ(absolute_capture_time, kAbsoluteCaptureTime);
+}
+
+TEST_F(RtpSenderVideoTest, PopulatesPlayoutDelay) {
+  // Single packet frames.
+  constexpr size_t kPacketSize = 123;
+  uint8_t kFrame[kPacketSize];
+  rtp_module_->RegisterRtpHeaderExtension(PlayoutDelayLimits::Uri(),
+                                          kPlayoutDelayExtensionId);
+  const VideoPlayoutDelay kExpectedDelay = {10, 20};
+
+  // Send initial key-frame without playout delay.
+  RTPVideoHeader hdr;
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  hdr.codec = VideoCodecType::kVideoCodecVP8;
+  auto& vp8_header = hdr.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.temporalIdx = 0;
+
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  EXPECT_FALSE(
+      transport_.last_sent_packet().HasExtension<PlayoutDelayLimits>());
+
+  // Set playout delay on a discardable frame.
+  hdr.playout_delay = kExpectedDelay;
+  hdr.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp8_header.temporalIdx = 1;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  VideoPlayoutDelay received_delay = VideoPlayoutDelay();
+  ASSERT_TRUE(transport_.last_sent_packet().GetExtension<PlayoutDelayLimits>(
+      &received_delay));
+  EXPECT_EQ(received_delay, kExpectedDelay);
+
+  // Set playout delay on a non-discardable frame, the extension should still
+  // be populated since dilvery wasn't guaranteed on the last one.
+  hdr.playout_delay = VideoPlayoutDelay();  // Indicates "no change".
+  vp8_header.temporalIdx = 0;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  ASSERT_TRUE(transport_.last_sent_packet().GetExtension<PlayoutDelayLimits>(
+      &received_delay));
+  EXPECT_EQ(received_delay, kExpectedDelay);
+
+  // The next frame does not need the extensions since it's delivery has
+  // already been guaranteed.
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  EXPECT_FALSE(
+      transport_.last_sent_packet().HasExtension<PlayoutDelayLimits>());
+
+  // Insert key-frame, we need to refresh the state here.
+  hdr.frame_type = VideoFrameType::kVideoFrameKey;
+  rtp_sender_video_->SendVideo(kPayload, kType, kTimestamp, 0, kFrame, hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
+  ASSERT_TRUE(transport_.last_sent_packet().GetExtension<PlayoutDelayLimits>(
+      &received_delay));
+  EXPECT_EQ(received_delay, kExpectedDelay);
+}
+
+TEST_F(RtpSenderVideoTest, SendGenericVideo) {
+  const uint8_t kPayloadType = 127;
+  const VideoCodecType kCodecType = VideoCodecType::kVideoCodecGeneric;
+  const uint8_t kPayload[] = {47, 11, 32, 93, 89};
+
+  // Send keyframe.
+  RTPVideoHeader video_header;
+  video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  ASSERT_TRUE(rtp_sender_video_->SendVideo(kPayloadType, kCodecType, 1234, 4321,
+                                           kPayload, video_header,
+                                           absl::nullopt));
+
+  rtc::ArrayView<const uint8_t> sent_payload =
+      transport_.last_sent_packet().payload();
+  uint8_t generic_header = sent_payload[0];
+  EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
+  EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
+  EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(kPayload));
+
+  // Send delta frame.
+  const uint8_t kDeltaPayload[] = {13, 42, 32, 93, 13};
+  video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  ASSERT_TRUE(rtp_sender_video_->SendVideo(kPayloadType, kCodecType, 1234, 4321,
+                                           kDeltaPayload, video_header,
+                                           absl::nullopt));
+
+  sent_payload = sent_payload = transport_.last_sent_packet().payload();
+  generic_header = sent_payload[0];
+  EXPECT_FALSE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
+  EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
+  EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(kDeltaPayload));
+}
+
+TEST_F(RtpSenderVideoTest, SendRawVideo) {
+  const uint8_t kPayloadType = 111;
+  const uint8_t kPayload[] = {11, 22, 33, 44, 55};
+
+  // Send a frame.
+  RTPVideoHeader video_header;
+  video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  ASSERT_TRUE(rtp_sender_video_->SendVideo(kPayloadType, absl::nullopt, 1234,
+                                           4321, kPayload, video_header,
+                                           absl::nullopt));
+
+  rtc::ArrayView<const uint8_t> sent_payload =
+      transport_.last_sent_packet().payload();
+  EXPECT_THAT(sent_payload, ElementsAreArray(kPayload));
+}
+
+class RtpSenderVideoWithFrameTransformerTest : public ::testing::Test {
+ public:
+  RtpSenderVideoWithFrameTransformerTest()
+      : time_controller_(kStartTime),
+        retransmission_rate_limiter_(time_controller_.GetClock(), 1000),
+        rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
+          RtpRtcpInterface::Configuration config;
+          config.clock = time_controller_.GetClock();
+          config.outgoing_transport = &transport_;
+          config.retransmission_rate_limiter = &retransmission_rate_limiter_;
+          config.field_trials = &field_trials_;
+          config.local_media_ssrc = kSsrc;
+          return config;
+        }())) {
+    rtp_module_->SetSequenceNumber(kSeqNum);
+    rtp_module_->SetStartTimestamp(0);
+  }
+
+  std::unique_ptr<RTPSenderVideo> CreateSenderWithFrameTransformer(
+      rtc::scoped_refptr<FrameTransformerInterface> transformer) {
+    RTPSenderVideo::Config config;
+    config.clock = time_controller_.GetClock();
+    config.rtp_sender = rtp_module_->RtpSender();
+    config.field_trials = &field_trials_;
+    config.frame_transformer = transformer;
+    config.task_queue_factory = time_controller_.GetTaskQueueFactory();
+    return std::make_unique<RTPSenderVideo>(config);
+  }
+
+ protected:
+  GlobalSimulatedTimeController time_controller_;
+  FieldTrialBasedConfig field_trials_;
+  LoopbackTransportTest transport_;
+  RateLimiter retransmission_rate_limiter_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_module_;
+};
+
+std::unique_ptr<EncodedImage> CreateDefaultEncodedImage() {
+  const uint8_t data[] = {1, 2, 3, 4};
+  auto encoded_image = std::make_unique<EncodedImage>();
+  encoded_image->SetEncodedData(
+      webrtc::EncodedImageBuffer::Create(data, sizeof(data)));
+  return encoded_image;
+}
+
+TEST_F(RtpSenderVideoWithFrameTransformerTest,
+       CreateSenderRegistersFrameTransformer) {
+  auto mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  EXPECT_CALL(*mock_frame_transformer,
+              RegisterTransformedFrameSinkCallback(_, kSsrc));
+  std::unique_ptr<RTPSenderVideo> rtp_sender_video =
+      CreateSenderWithFrameTransformer(mock_frame_transformer);
+}
+
+TEST_F(RtpSenderVideoWithFrameTransformerTest,
+       DestroySenderUnregistersFrameTransformer) {
+  auto mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  std::unique_ptr<RTPSenderVideo> rtp_sender_video =
+      CreateSenderWithFrameTransformer(mock_frame_transformer);
+  EXPECT_CALL(*mock_frame_transformer,
+              UnregisterTransformedFrameSinkCallback(kSsrc));
+  rtp_sender_video = nullptr;
+}
+
+TEST_F(RtpSenderVideoWithFrameTransformerTest,
+       SendEncodedImageTransformsFrame) {
+  auto mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  std::unique_ptr<RTPSenderVideo> rtp_sender_video =
+      CreateSenderWithFrameTransformer(mock_frame_transformer);
+  auto encoded_image = CreateDefaultEncodedImage();
+  RTPVideoHeader video_header;
+
+  EXPECT_CALL(*mock_frame_transformer, Transform);
+  rtp_sender_video->SendEncodedImage(kPayload, kType, kTimestamp,
+                                     *encoded_image, video_header,
+                                     kDefaultExpectedRetransmissionTimeMs);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST_F(RtpSenderVideoWithFrameTransformerTest, ValidPayloadTypes) {
+  auto mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  std::unique_ptr<RTPSenderVideo> rtp_sender_video =
+      CreateSenderWithFrameTransformer(mock_frame_transformer);
+  auto encoded_image = CreateDefaultEncodedImage();
+  RTPVideoHeader video_header;
+
+  EXPECT_TRUE(rtp_sender_video->SendEncodedImage(
+      0, kType, kTimestamp, *encoded_image, video_header,
+      kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_TRUE(rtp_sender_video->SendEncodedImage(
+      127, kType, kTimestamp, *encoded_image, video_header,
+      kDefaultExpectedRetransmissionTimeMs));
+  EXPECT_DEATH(rtp_sender_video->SendEncodedImage(
+                   -1, kType, kTimestamp, *encoded_image, video_header,
+                   kDefaultExpectedRetransmissionTimeMs),
+               "");
+  EXPECT_DEATH(rtp_sender_video->SendEncodedImage(
+                   128, kType, kTimestamp, *encoded_image, video_header,
+                   kDefaultExpectedRetransmissionTimeMs),
+               "");
+}
+#endif
+
+TEST_F(RtpSenderVideoWithFrameTransformerTest, OnTransformedFrameSendsVideo) {
+  auto mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameSinkCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  std::unique_ptr<RTPSenderVideo> rtp_sender_video =
+      CreateSenderWithFrameTransformer(mock_frame_transformer);
+  ASSERT_TRUE(callback);
+
+  auto encoded_image = CreateDefaultEncodedImage();
+  RTPVideoHeader video_header;
+  video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  ON_CALL(*mock_frame_transformer, Transform)
+      .WillByDefault(
+          [&callback](std::unique_ptr<TransformableFrameInterface> frame) {
+            callback->OnTransformedFrame(std::move(frame));
+          });
+  auto encoder_queue = time_controller_.GetTaskQueueFactory()->CreateTaskQueue(
+      "encoder_queue", TaskQueueFactory::Priority::NORMAL);
+  encoder_queue->PostTask([&] {
+    rtp_sender_video->SendEncodedImage(kPayload, kType, kTimestamp,
+                                       *encoded_image, video_header,
+                                       kDefaultExpectedRetransmissionTimeMs);
+  });
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+  EXPECT_EQ(transport_.packets_sent(), 1);
+  encoder_queue->PostTask([&] {
+    rtp_sender_video->SendEncodedImage(kPayload, kType, kTimestamp,
+                                       *encoded_image, video_header,
+                                       kDefaultExpectedRetransmissionTimeMs);
+  });
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+  EXPECT_EQ(transport_.packets_sent(), 2);
+}
+
+TEST_F(RtpSenderVideoWithFrameTransformerTest,
+       TransformableFrameMetadataHasCorrectValue) {
+  auto mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  std::unique_ptr<RTPSenderVideo> rtp_sender_video =
+      CreateSenderWithFrameTransformer(mock_frame_transformer);
+  auto encoded_image = CreateDefaultEncodedImage();
+  RTPVideoHeader video_header;
+  video_header.width = 1280u;
+  video_header.height = 720u;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.frame_id = 10;
+  generic.temporal_index = 3;
+  generic.spatial_index = 2;
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch};
+  generic.dependencies = {5};
+
+  // Check that the transformable frame passed to the frame transformer has the
+  // correct metadata.
+  EXPECT_CALL(*mock_frame_transformer, Transform)
+      .WillOnce(
+          [](std::unique_ptr<TransformableFrameInterface> transformable_frame) {
+            auto frame =
+                absl::WrapUnique(static_cast<TransformableVideoFrameInterface*>(
+                    transformable_frame.release()));
+            ASSERT_TRUE(frame);
+            auto metadata = frame->GetMetadata();
+            EXPECT_EQ(metadata.GetWidth(), 1280u);
+            EXPECT_EQ(metadata.GetHeight(), 720u);
+            EXPECT_EQ(metadata.GetFrameId(), 10);
+            EXPECT_EQ(metadata.GetTemporalIndex(), 3);
+            EXPECT_EQ(metadata.GetSpatialIndex(), 2);
+            EXPECT_THAT(metadata.GetFrameDependencies(), ElementsAre(5));
+            EXPECT_THAT(metadata.GetDecodeTargetIndications(),
+                        ElementsAre(DecodeTargetIndication::kSwitch));
+          });
+  rtp_sender_video->SendEncodedImage(kPayload, kType, kTimestamp,
+                                     *encoded_image, video_header,
+                                     kDefaultExpectedRetransmissionTimeMs);
+}
+
+TEST_F(RtpSenderVideoWithFrameTransformerTest,
+       OnTransformedFrameSendsVideoWhenCloned) {
+  auto mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameSinkCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  std::unique_ptr<RTPSenderVideo> rtp_sender_video =
+      CreateSenderWithFrameTransformer(mock_frame_transformer);
+  ASSERT_TRUE(callback);
+
+  auto encoded_image = CreateDefaultEncodedImage();
+  RTPVideoHeader video_header;
+  video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  ON_CALL(*mock_frame_transformer, Transform)
+      .WillByDefault(
+          [&callback](std::unique_ptr<TransformableFrameInterface> frame) {
+            auto clone = CloneVideoFrame(
+                static_cast<TransformableVideoFrameInterface*>(frame.get()));
+            EXPECT_TRUE(clone);
+            callback->OnTransformedFrame(std::move(clone));
+          });
+  auto encoder_queue = time_controller_.GetTaskQueueFactory()->CreateTaskQueue(
+      "encoder_queue", TaskQueueFactory::Priority::NORMAL);
+  encoder_queue->PostTask([&] {
+    rtp_sender_video->SendEncodedImage(kPayload, kType, kTimestamp,
+                                       *encoded_image, video_header,
+                                       kDefaultExpectedRetransmissionTimeMs);
+  });
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+  EXPECT_EQ(transport_.packets_sent(), 1);
+  encoder_queue->PostTask([&] {
+    rtp_sender_video->SendEncodedImage(kPayload, kType, kTimestamp,
+                                       *encoded_image, video_header,
+                                       kDefaultExpectedRetransmissionTimeMs);
+  });
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+  EXPECT_EQ(transport_.packets_sent(), 2);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map.cc
new file mode 100644
index 0000000000..441429d442
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map.cc
@@ -0,0 +1,129 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
+
+#include <algorithm>
+#include <iterator>
+#include <limits>
+
+#include "absl/algorithm/container.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+
+namespace webrtc {
+
+RtpSequenceNumberMap::RtpSequenceNumberMap(size_t max_entries)
+    : max_entries_(max_entries) {
+  RTC_DCHECK_GT(max_entries_, 4);  // See code paring down to `max_entries_`.
+  RTC_DCHECK_LE(max_entries_, 1 << 15);
+}
+
+RtpSequenceNumberMap::~RtpSequenceNumberMap() = default;
+
+void RtpSequenceNumberMap::InsertPacket(uint16_t sequence_number, Info info) {
+  RTC_DCHECK(associations_.size() < 2 ||
+             AheadOf(associations_.back().sequence_number,
+                     associations_.front().sequence_number));
+
+  if (associations_.empty()) {
+    associations_.emplace_back(sequence_number, info);
+    return;
+  }
+
+  if (AheadOrAt(sequence_number, associations_.front().sequence_number) &&
+      AheadOrAt(associations_.back().sequence_number, sequence_number)) {
+    // The sequence number has wrapped around and is within the range
+    // currently held by `associations_` - we should invalidate all entries.
+    RTC_LOG(LS_WARNING) << "Sequence number wrapped-around unexpectedly.";
+    associations_.clear();
+    associations_.emplace_back(sequence_number, info);
+    return;
+  }
+
+  std::deque<Association>::iterator erase_to = associations_.begin();
+
+  RTC_DCHECK_LE(associations_.size(), max_entries_);
+  if (associations_.size() == max_entries_) {
+    // Pare down the container so that inserting some additional elements
+    // would not exceed the maximum size.
+    const size_t new_size = 3 * max_entries_ / 4;
+    erase_to = std::next(erase_to, max_entries_ - new_size);
+  }
+
+  // It is guaranteed that `associations_` can be split into two partitions,
+  // either partition possibly empty, such that:
+  // * In the first partition, all elements are AheadOf the new element.
+  //   This is the partition of the obsolete elements.
+  // * In the second partition, the new element is AheadOf all the elements.
+  //   The elements of this partition may stay.
+  auto cmp = [](const Association& a, uint16_t sequence_number) {
+    return AheadOf(a.sequence_number, sequence_number);
+  };
+  RTC_DCHECK(erase_to != associations_.end());
+  erase_to =
+      std::lower_bound(erase_to, associations_.end(), sequence_number, cmp);
+  associations_.erase(associations_.begin(), erase_to);
+
+  associations_.emplace_back(sequence_number, info);
+
+  RTC_DCHECK(associations_.size() == 1 ||
+             AheadOf(associations_.back().sequence_number,
+                     associations_.front().sequence_number));
+}
+
+void RtpSequenceNumberMap::InsertFrame(uint16_t first_sequence_number,
+                                       size_t packet_count,
+                                       uint32_t timestamp) {
+  RTC_DCHECK_GT(packet_count, 0);
+  RTC_DCHECK_LE(packet_count, std::numeric_limits<size_t>::max());
+
+  for (size_t i = 0; i < packet_count; ++i) {
+    const bool is_first = (i == 0);
+    const bool is_last = (i == packet_count - 1);
+    InsertPacket(static_cast<uint16_t>(first_sequence_number + i),
+                 Info(timestamp, is_first, is_last));
+  }
+}
+
+absl::optional<RtpSequenceNumberMap::Info> RtpSequenceNumberMap::Get(
+    uint16_t sequence_number) const {
+  // To make the binary search easier to understand, we use the fact that
+  // adding a constant offset to all elements, as well as to the searched
+  // element, does not change the relative ordering. This way, we can find
+  // an offset that would make all of the elements strictly ascending according
+  // to normal integer comparison.
+  // Finding such an offset is easy - the offset that would map the oldest
+  // element to 0 would serve this purpose.
+
+  if (associations_.empty()) {
+    return absl::nullopt;
+  }
+
+  const uint16_t offset =
+      static_cast<uint16_t>(0) - associations_.front().sequence_number;
+
+  auto cmp = [offset](const Association& a, uint16_t sequence_number) {
+    return static_cast<uint16_t>(a.sequence_number + offset) <
+           static_cast<uint16_t>(sequence_number + offset);
+  };
+  const auto elem = absl::c_lower_bound(associations_, sequence_number, cmp);
+
+  return elem != associations_.end() && elem->sequence_number == sequence_number
+             ? absl::optional<Info>(elem->info)
+             : absl::nullopt;
+}
+
+size_t RtpSequenceNumberMap::AssociationCountForTesting() const {
+  return associations_.size();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map.h
new file mode 100644
index 0000000000..8a036c25a4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_SEQUENCE_NUMBER_MAP_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_SEQUENCE_NUMBER_MAP_H_
+
+#include <cstddef>
+#include <cstdint>
+#include <deque>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+// Records the association of RTP sequence numbers to timestamps and to whether
+// the packet was first and/or last in the frame.
+//
+// 1. Limits number of entries. Whenever `max_entries` is about to be exceeded,
+//    the size is reduced by approximately 25%.
+// 2. RTP sequence numbers wrap around relatively infrequently.
+//    This class therefore only remembers at most the last 2^15 RTP packets,
+//    so that the newest packet's sequence number is still AheadOf the oldest
+//    packet's sequence number.
+// 3. Media frames are sometimes split into several RTP packets.
+//    In such a case, Insert() is expected to be called once for each packet.
+//    The timestamp is not expected to change between those calls.
+class RtpSequenceNumberMap final {
+ public:
+  struct Info final {
+    Info(uint32_t timestamp, bool is_first, bool is_last)
+        : timestamp(timestamp), is_first(is_first), is_last(is_last) {}
+
+    friend bool operator==(const Info& lhs, const Info& rhs) {
+      return lhs.timestamp == rhs.timestamp && lhs.is_first == rhs.is_first &&
+             lhs.is_last == rhs.is_last;
+    }
+
+    uint32_t timestamp;
+    bool is_first;
+    bool is_last;
+  };
+
+  explicit RtpSequenceNumberMap(size_t max_entries);
+  RtpSequenceNumberMap(const RtpSequenceNumberMap& other) = delete;
+  RtpSequenceNumberMap& operator=(const RtpSequenceNumberMap& other) = delete;
+  ~RtpSequenceNumberMap();
+
+  void InsertPacket(uint16_t sequence_number, Info info);
+  void InsertFrame(uint16_t first_sequence_number,
+                   size_t packet_count,
+                   uint32_t timestamp);
+
+  absl::optional<Info> Get(uint16_t sequence_number) const;
+
+  size_t AssociationCountForTesting() const;
+
+ private:
+  struct Association {
+    explicit Association(uint16_t sequence_number)
+        : Association(sequence_number, Info(0, false, false)) {}
+
+    Association(uint16_t sequence_number, Info info)
+        : sequence_number(sequence_number), info(info) {}
+
+    uint16_t sequence_number;
+    Info info;
+  };
+
+  const size_t max_entries_;
+
+  // The non-transitivity of AheadOf() would be problematic with a map,
+  // so we use a deque instead.
+  std::deque<Association> associations_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SEQUENCE_NUMBER_MAP_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc
new file mode 100644
index 0000000000..78c9e4a251
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sequence_number_map_unittest.cc
@@ -0,0 +1,502 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
+
+#include <algorithm>
+#include <iterator>
+#include <limits>
+#include <memory>
+#include <tuple>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/random.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+using Info = RtpSequenceNumberMap::Info;
+
+constexpr uint16_t kUint16Max = std::numeric_limits<uint16_t>::max();
+constexpr size_t kMaxPossibleMaxEntries = 1 << 15;
+
+// Just a named pair.
+struct Association final {
+  Association(uint16_t sequence_number, Info info)
+      : sequence_number(sequence_number), info(info) {}
+
+  uint16_t sequence_number;
+  Info info;
+};
+
+class RtpSequenceNumberMapTest : public ::testing::Test {
+ protected:
+  RtpSequenceNumberMapTest() : random_(1983) {}
+  ~RtpSequenceNumberMapTest() override = default;
+
+  Association CreateAssociation(uint16_t sequence_number, uint32_t timestamp) {
+    return Association(sequence_number,
+                       {timestamp, random_.Rand<bool>(), random_.Rand<bool>()});
+  }
+
+  void VerifyAssociations(const RtpSequenceNumberMap& uut,
+                          const std::vector<Association>& associations) {
+    return VerifyAssociations(uut, associations.begin(), associations.end());
+  }
+
+  void VerifyAssociations(
+      const RtpSequenceNumberMap& uut,
+      std::vector<Association>::const_iterator associations_begin,
+      std::vector<Association>::const_iterator associations_end) {
+    RTC_DCHECK(associations_begin < associations_end);
+    ASSERT_EQ(static_cast<size_t>(associations_end - associations_begin),
+              uut.AssociationCountForTesting());
+    for (auto association = associations_begin; association != associations_end;
+         ++association) {
+      EXPECT_EQ(uut.Get(association->sequence_number), association->info);
+    }
+  }
+
+  // Allows several variations of the same test; definition next to the tests.
+  void GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted(
+      bool with_wrap_around,
+      bool last_element_kept);
+
+  // Allows several variations of the same test; definition next to the tests.
+  void RepeatedSequenceNumberInvalidatesAll(size_t index_of_repeated);
+
+  // Allows several variations of the same test; definition next to the tests.
+  void MaxEntriesReachedAtSameTimeAsObsoletionOfItem(size_t max_entries,
+                                                     size_t obsoleted_count);
+
+  Random random_;
+};
+
+class RtpSequenceNumberMapTestWithParams
+    : public RtpSequenceNumberMapTest,
+      public ::testing::WithParamInterface<std::tuple<size_t, uint16_t>> {
+ protected:
+  RtpSequenceNumberMapTestWithParams() = default;
+  ~RtpSequenceNumberMapTestWithParams() override = default;
+
+  std::vector<Association> ProduceRandomAssociationSequence(
+      size_t association_count,
+      uint16_t first_sequence_number,
+      bool allow_obsoletion) {
+    std::vector<Association> associations;
+    associations.reserve(association_count);
+
+    if (association_count == 0) {
+      return associations;
+    }
+
+    associations.emplace_back(
+        first_sequence_number,
+        Info(0, random_.Rand<bool>(), random_.Rand<bool>()));
+
+    for (size_t i = 1; i < association_count; ++i) {
+      const uint16_t sequence_number =
+          associations[i - 1].sequence_number + random_.Rand(1, 100);
+      RTC_DCHECK(allow_obsoletion ||
+                 AheadOf(sequence_number, associations[0].sequence_number));
+
+      const uint32_t timestamp =
+          associations[i - 1].info.timestamp + random_.Rand(1, 10000);
+
+      associations.emplace_back(
+          sequence_number,
+          Info(timestamp, random_.Rand<bool>(), random_.Rand<bool>()));
+    }
+
+    return associations;
+  }
+};
+
+INSTANTIATE_TEST_SUITE_P(All,
+                         RtpSequenceNumberMapTestWithParams,
+                         ::testing::Combine(
+                             // Association count.
+                             ::testing::Values(1, 2, 100),
+                             // First sequence number.
+                             ::testing::Values(0,
+                                               100,
+                                               kUint16Max - 100,
+                                               kUint16Max - 1,
+                                               kUint16Max)));
+
+TEST_F(RtpSequenceNumberMapTest, GetBeforeAssociationsRecordedReturnsNullOpt) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+  constexpr uint16_t kArbitrarySequenceNumber = 321;
+  EXPECT_FALSE(uut.Get(kArbitrarySequenceNumber));
+}
+
+// Version #1 - any old unknown sequence number.
+TEST_F(RtpSequenceNumberMapTest, GetUnknownSequenceNumberReturnsNullOpt1) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  constexpr uint16_t kKnownSequenceNumber = 10;
+  constexpr uint32_t kArbitrary = 987;
+  uut.InsertPacket(kKnownSequenceNumber, {kArbitrary, false, false});
+
+  constexpr uint16_t kUnknownSequenceNumber = kKnownSequenceNumber + 1;
+  EXPECT_FALSE(uut.Get(kUnknownSequenceNumber));
+}
+
+// Version #2 - intentionally pick a value in the range of currently held
+// values, so as to trigger lower_bound / upper_bound.
+TEST_F(RtpSequenceNumberMapTest, GetUnknownSequenceNumberReturnsNullOpt2) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  const std::vector<Association> setup = {CreateAssociation(1000, 500),  //
+                                          CreateAssociation(1020, 501)};
+  for (const Association& association : setup) {
+    uut.InsertPacket(association.sequence_number, association.info);
+  }
+
+  EXPECT_FALSE(uut.Get(1001));
+}
+
+TEST_P(RtpSequenceNumberMapTestWithParams,
+       GetKnownSequenceNumberReturnsCorrectValue) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  const size_t association_count = std::get<0>(GetParam());
+  const uint16_t first_sequence_number = std::get<1>(GetParam());
+
+  const std::vector<Association> associations =
+      ProduceRandomAssociationSequence(association_count, first_sequence_number,
+                                       /*allow_obsoletion=*/false);
+
+  for (const Association& association : associations) {
+    uut.InsertPacket(association.sequence_number, association.info);
+  }
+
+  VerifyAssociations(uut, associations);
+}
+
+TEST_F(RtpSequenceNumberMapTest, InsertFrameOnSinglePacketFrame) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  constexpr uint16_t kSequenceNumber = 888;
+  constexpr uint32_t kTimestamp = 98765;
+  uut.InsertFrame(kSequenceNumber, 1, kTimestamp);
+
+  EXPECT_EQ(uut.Get(kSequenceNumber), Info(kTimestamp, true, true));
+}
+
+TEST_F(RtpSequenceNumberMapTest, InsertFrameOnMultiPacketFrameNoWrapAround) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  constexpr uint16_t kFirstSequenceNumber = 0;
+  constexpr uint32_t kTimestamp = 98765;
+  uut.InsertFrame(kFirstSequenceNumber, 3, kTimestamp);
+
+  EXPECT_EQ(uut.Get(kFirstSequenceNumber + 0), Info(kTimestamp, true, false));
+  EXPECT_EQ(uut.Get(kFirstSequenceNumber + 1), Info(kTimestamp, false, false));
+  EXPECT_EQ(uut.Get(kFirstSequenceNumber + 2), Info(kTimestamp, false, true));
+}
+
+TEST_F(RtpSequenceNumberMapTest, InsertFrameOnMultiPacketFrameWithWrapAround) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  constexpr uint16_t kFirstSequenceNumber = kUint16Max;
+  constexpr uint32_t kTimestamp = 98765;
+  uut.InsertFrame(kFirstSequenceNumber, 3, kTimestamp);
+
+// Suppress "truncation of constant value" warning; wrap-around is intended.
+#ifdef _MSC_VER
+#pragma warning(push)
+#pragma warning(disable : 4309)
+#endif
+  EXPECT_EQ(uut.Get(static_cast<uint16_t>(kFirstSequenceNumber + 0u)),
+            Info(kTimestamp, true, false));
+  EXPECT_EQ(uut.Get(static_cast<uint16_t>(kFirstSequenceNumber + 1u)),
+            Info(kTimestamp, false, false));
+  EXPECT_EQ(uut.Get(static_cast<uint16_t>(kFirstSequenceNumber + 2u)),
+            Info(kTimestamp, false, true));
+#ifdef _MSC_VER
+#pragma warning(pop)
+#endif
+}
+
+TEST_F(RtpSequenceNumberMapTest,
+       GetObsoleteSequenceNumberReturnsNullOptSingleValueObsoleted) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  const std::vector<Association> associations = {
+      CreateAssociation(0, 10),       //
+      CreateAssociation(0x8000, 20),  //
+      CreateAssociation(0x8001u, 30)};
+
+  uut.InsertPacket(associations[0].sequence_number, associations[0].info);
+
+  // First association not yet obsolete, and therefore remembered.
+  RTC_DCHECK(AheadOf(associations[1].sequence_number,
+                     associations[0].sequence_number));
+  uut.InsertPacket(associations[1].sequence_number, associations[1].info);
+  VerifyAssociations(uut, {associations[0], associations[1]});
+
+  // Test focus - new entry obsoletes first entry.
+  RTC_DCHECK(!AheadOf(associations[2].sequence_number,
+                      associations[0].sequence_number));
+  uut.InsertPacket(associations[2].sequence_number, associations[2].info);
+  VerifyAssociations(uut, {associations[1], associations[2]});
+}
+
+void RtpSequenceNumberMapTest::
+    GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted(
+        bool with_wrap_around,
+        bool last_element_kept) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  std::vector<Association> associations;
+  if (with_wrap_around) {
+    associations = {CreateAssociation(kUint16Max - 1, 10),  //
+                    CreateAssociation(kUint16Max, 20),      //
+                    CreateAssociation(0, 30),               //
+                    CreateAssociation(1, 40),               //
+                    CreateAssociation(2, 50)};
+  } else {
+    associations = {CreateAssociation(1, 10),  //
+                    CreateAssociation(2, 20),  //
+                    CreateAssociation(3, 30),  //
+                    CreateAssociation(4, 40),  //
+                    CreateAssociation(5, 50)};
+  }
+
+  for (auto association : associations) {
+    uut.InsertPacket(association.sequence_number, association.info);
+  }
+  VerifyAssociations(uut, associations);
+
+  // Define a new association that will obsolete either all previous entries,
+  // or all previous entries except for the last one, depending on the
+  // parameter instantiation of this test.
+  RTC_DCHECK_EQ(
+      static_cast<uint16_t>(
+          associations[associations.size() - 1].sequence_number),
+      static_cast<uint16_t>(
+          associations[associations.size() - 2].sequence_number + 1u));
+  uint16_t new_sequence_number;
+  if (last_element_kept) {
+    new_sequence_number =
+        associations[associations.size() - 1].sequence_number + 0x8000;
+    RTC_DCHECK(AheadOf(new_sequence_number,
+                       associations[associations.size() - 1].sequence_number));
+  } else {
+    new_sequence_number =
+        associations[associations.size() - 1].sequence_number + 0x8001;
+    RTC_DCHECK(!AheadOf(new_sequence_number,
+                        associations[associations.size() - 1].sequence_number));
+  }
+  RTC_DCHECK(!AheadOf(new_sequence_number,
+                      associations[associations.size() - 2].sequence_number));
+
+  // Record the new association.
+  const Association new_association =
+      CreateAssociation(new_sequence_number, 60);
+  uut.InsertPacket(new_association.sequence_number, new_association.info);
+
+  // Make sure all obsoleted elements were removed.
+  const size_t obsoleted_count =
+      associations.size() - (last_element_kept ? 1 : 0);
+  for (size_t i = 0; i < obsoleted_count; ++i) {
+    EXPECT_FALSE(uut.Get(associations[i].sequence_number));
+  }
+
+  // Make sure the expected elements were not removed, and return the
+  // expected value.
+  if (last_element_kept) {
+    EXPECT_EQ(uut.Get(associations.back().sequence_number),
+              associations.back().info);
+  }
+  EXPECT_EQ(uut.Get(new_association.sequence_number), new_association.info);
+}
+
+TEST_F(RtpSequenceNumberMapTest,
+       GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted0) {
+  const bool with_wrap_around = false;
+  const bool last_element_kept = false;
+  GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted(
+      with_wrap_around, last_element_kept);
+}
+
+TEST_F(RtpSequenceNumberMapTest,
+       GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted1) {
+  const bool with_wrap_around = true;
+  const bool last_element_kept = false;
+  GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted(
+      with_wrap_around, last_element_kept);
+}
+
+TEST_F(RtpSequenceNumberMapTest,
+       GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted2) {
+  const bool with_wrap_around = false;
+  const bool last_element_kept = true;
+  GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted(
+      with_wrap_around, last_element_kept);
+}
+
+TEST_F(RtpSequenceNumberMapTest,
+       GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted3) {
+  const bool with_wrap_around = true;
+  const bool last_element_kept = true;
+  GetObsoleteSequenceNumberReturnsNullOptMultipleEntriesObsoleted(
+      with_wrap_around, last_element_kept);
+}
+
+void RtpSequenceNumberMapTest::RepeatedSequenceNumberInvalidatesAll(
+    size_t index_of_repeated) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  const std::vector<Association> setup = {CreateAssociation(100, 500),  //
+                                          CreateAssociation(101, 501),  //
+                                          CreateAssociation(102, 502)};
+  RTC_DCHECK_LT(index_of_repeated, setup.size());
+  for (const Association& association : setup) {
+    uut.InsertPacket(association.sequence_number, association.info);
+  }
+
+  const Association new_association =
+      CreateAssociation(setup[index_of_repeated].sequence_number, 503);
+  uut.InsertPacket(new_association.sequence_number, new_association.info);
+
+  // All entries from setup invalidated.
+  // New entry valid and mapped to new value.
+  for (size_t i = 0; i < setup.size(); ++i) {
+    if (i == index_of_repeated) {
+      EXPECT_EQ(uut.Get(new_association.sequence_number), new_association.info);
+    } else {
+      EXPECT_FALSE(uut.Get(setup[i].sequence_number));
+    }
+  }
+}
+
+TEST_F(RtpSequenceNumberMapTest,
+       RepeatedSequenceNumberInvalidatesAllRepeatFirst) {
+  RepeatedSequenceNumberInvalidatesAll(0);
+}
+
+TEST_F(RtpSequenceNumberMapTest,
+       RepeatedSequenceNumberInvalidatesAllRepeatMiddle) {
+  RepeatedSequenceNumberInvalidatesAll(1);
+}
+
+TEST_F(RtpSequenceNumberMapTest,
+       RepeatedSequenceNumberInvalidatesAllRepeatLast) {
+  RepeatedSequenceNumberInvalidatesAll(2);
+}
+
+TEST_F(RtpSequenceNumberMapTest,
+       SequenceNumberInsideMemorizedRangeInvalidatesAll) {
+  RtpSequenceNumberMap uut(kMaxPossibleMaxEntries);
+
+  const std::vector<Association> setup = {CreateAssociation(1000, 500),  //
+                                          CreateAssociation(1020, 501),  //
+                                          CreateAssociation(1030, 502)};
+  for (const Association& association : setup) {
+    uut.InsertPacket(association.sequence_number, association.info);
+  }
+
+  const Association new_association = CreateAssociation(1010, 503);
+  uut.InsertPacket(new_association.sequence_number, new_association.info);
+
+  // All entries from setup invalidated.
+  // New entry valid and mapped to new value.
+  for (size_t i = 0; i < setup.size(); ++i) {
+    EXPECT_FALSE(uut.Get(setup[i].sequence_number));
+  }
+  EXPECT_EQ(uut.Get(new_association.sequence_number), new_association.info);
+}
+
+TEST_F(RtpSequenceNumberMapTest, MaxEntriesObserved) {
+  constexpr size_t kMaxEntries = 100;
+  RtpSequenceNumberMap uut(kMaxEntries);
+
+  std::vector<Association> associations;
+  associations.reserve(kMaxEntries);
+  uint32_t timestamp = 789;
+  for (size_t i = 0; i < kMaxEntries; ++i) {
+    associations.push_back(CreateAssociation(i, ++timestamp));
+    uut.InsertPacket(associations[i].sequence_number, associations[i].info);
+  }
+  VerifyAssociations(uut, associations);  // Sanity.
+
+  const Association new_association =
+      CreateAssociation(kMaxEntries, ++timestamp);
+  uut.InsertPacket(new_association.sequence_number, new_association.info);
+  associations.push_back(new_association);
+
+  // The +1 is for `new_association`.
+  const size_t kExpectedAssociationCount = 3 * kMaxEntries / 4 + 1;
+  const auto expected_begin =
+      std::prev(associations.end(), kExpectedAssociationCount);
+  VerifyAssociations(uut, expected_begin, associations.end());
+}
+
+void RtpSequenceNumberMapTest::MaxEntriesReachedAtSameTimeAsObsoletionOfItem(
+    size_t max_entries,
+    size_t obsoleted_count) {
+  RtpSequenceNumberMap uut(max_entries);
+
+  std::vector<Association> associations;
+  associations.reserve(max_entries);
+  uint32_t timestamp = 789;
+  for (size_t i = 0; i < max_entries; ++i) {
+    associations.push_back(CreateAssociation(i, ++timestamp));
+    uut.InsertPacket(associations[i].sequence_number, associations[i].info);
+  }
+  VerifyAssociations(uut, associations);  // Sanity.
+
+  const uint16_t new_association_sequence_number =
+      static_cast<uint16_t>(obsoleted_count) + (1 << 15);
+  const Association new_association =
+      CreateAssociation(new_association_sequence_number, ++timestamp);
+  uut.InsertPacket(new_association.sequence_number, new_association.info);
+  associations.push_back(new_association);
+
+  // The +1 is for `new_association`.
+  const size_t kExpectedAssociationCount =
+      std::min(3 * max_entries / 4, max_entries - obsoleted_count) + 1;
+  const auto expected_begin =
+      std::prev(associations.end(), kExpectedAssociationCount);
+  VerifyAssociations(uut, expected_begin, associations.end());
+}
+
+// Version #1 - #(obsoleted entries) < #(entries after paring down below max).
+TEST_F(RtpSequenceNumberMapTest,
+       MaxEntriesReachedAtSameTimeAsObsoletionOfItem1) {
+  constexpr size_t kMaxEntries = 100;
+  constexpr size_t kObsoletionTarget = (kMaxEntries / 4) - 1;
+  MaxEntriesReachedAtSameTimeAsObsoletionOfItem(kMaxEntries, kObsoletionTarget);
+}
+
+// Version #2 - #(obsoleted entries) == #(entries after paring down below max).
+TEST_F(RtpSequenceNumberMapTest,
+       MaxEntriesReachedAtSameTimeAsObsoletionOfItem2) {
+  constexpr size_t kMaxEntries = 100;
+  constexpr size_t kObsoletionTarget = kMaxEntries / 4;
+  MaxEntriesReachedAtSameTimeAsObsoletionOfItem(kMaxEntries, kObsoletionTarget);
+}
+
+// Version #3 - #(obsoleted entries) > #(entries after paring down below max).
+TEST_F(RtpSequenceNumberMapTest,
+       MaxEntriesReachedAtSameTimeAsObsoletionOfItem3) {
+  constexpr size_t kMaxEntries = 100;
+  constexpr size_t kObsoletionTarget = (kMaxEntries / 4) + 1;
+  MaxEntriesReachedAtSameTimeAsObsoletionOfItem(kMaxEntries, kObsoletionTarget);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util.cc
new file mode 100644
index 0000000000..cf1e54254a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_util.h"
+
+#include <cstddef>
+#include <cstdint>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr uint8_t kRtpVersion = 2;
+constexpr size_t kMinRtpPacketLen = 12;
+constexpr size_t kMinRtcpPacketLen = 4;
+
+bool HasCorrectRtpVersion(rtc::ArrayView<const uint8_t> packet) {
+  return packet[0] >> 6 == kRtpVersion;
+}
+
+// For additional details, see http://tools.ietf.org/html/rfc5761#section-4
+bool PayloadTypeIsReservedForRtcp(uint8_t payload_type) {
+  return 64 <= payload_type && payload_type < 96;
+}
+
+}  // namespace
+
+bool IsRtpPacket(rtc::ArrayView<const uint8_t> packet) {
+  return packet.size() >= kMinRtpPacketLen && HasCorrectRtpVersion(packet) &&
+         !PayloadTypeIsReservedForRtcp(packet[1] & 0x7F);
+}
+
+bool IsRtcpPacket(rtc::ArrayView<const uint8_t> packet) {
+  return packet.size() >= kMinRtcpPacketLen && HasCorrectRtpVersion(packet) &&
+         PayloadTypeIsReservedForRtcp(packet[1] & 0x7F);
+}
+
+int ParseRtpPayloadType(rtc::ArrayView<const uint8_t> rtp_packet) {
+  RTC_DCHECK(IsRtpPacket(rtp_packet));
+  return rtp_packet[1] & 0x7F;
+}
+
+uint16_t ParseRtpSequenceNumber(rtc::ArrayView<const uint8_t> rtp_packet) {
+  RTC_DCHECK(IsRtpPacket(rtp_packet));
+  return ByteReader<uint16_t>::ReadBigEndian(rtp_packet.data() + 2);
+}
+
+uint32_t ParseRtpSsrc(rtc::ArrayView<const uint8_t> rtp_packet) {
+  RTC_DCHECK(IsRtpPacket(rtp_packet));
+  return ByteReader<uint32_t>::ReadBigEndian(rtp_packet.data() + 8);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util.h
new file mode 100644
index 0000000000..835cfcd6c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_UTIL_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_UTIL_H_
+
+#include <cstdint>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+bool IsRtcpPacket(rtc::ArrayView<const uint8_t> packet);
+bool IsRtpPacket(rtc::ArrayView<const uint8_t> packet);
+
+// Returns base rtp header fields of the rtp packet.
+// Behaviour is undefined when `!IsRtpPacket(rtp_packet)`.
+int ParseRtpPayloadType(rtc::ArrayView<const uint8_t> rtp_packet);
+uint16_t ParseRtpSequenceNumber(rtc::ArrayView<const uint8_t> rtp_packet);
+uint32_t ParseRtpSsrc(rtc::ArrayView<const uint8_t> rtp_packet);
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_UTIL_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util_unittest.cc
new file mode 100644
index 0000000000..3e23416ff4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util_unittest.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_util.h"
+
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace {
+
+TEST(RtpUtilTest, IsRtpPacket) {
+  constexpr uint8_t kMinimalisticRtpPacket[] = {0x80, 97, 0, 0,  //
+                                                0,    0,  0, 0,  //
+                                                0,    0,  0, 0};
+  EXPECT_TRUE(IsRtpPacket(kMinimalisticRtpPacket));
+
+  constexpr uint8_t kWrongRtpVersion[] = {0xc0, 97, 0, 0,  //
+                                          0,    0,  0, 0,  //
+                                          0,    0,  0, 0};
+  EXPECT_FALSE(IsRtpPacket(kWrongRtpVersion));
+
+  constexpr uint8_t kPacketWithPayloadForRtcp[] = {0x80, 200, 0, 0,  //
+                                                   0,    0,   0, 0,  //
+                                                   0,    0,   0, 0};
+  EXPECT_FALSE(IsRtpPacket(kPacketWithPayloadForRtcp));
+
+  constexpr uint8_t kTooSmallRtpPacket[] = {0x80, 97, 0, 0,  //
+                                            0,    0,  0, 0,  //
+                                            0,    0,  0};
+  EXPECT_FALSE(IsRtpPacket(kTooSmallRtpPacket));
+
+  EXPECT_FALSE(IsRtpPacket({}));
+}
+
+TEST(RtpUtilTest, IsRtcpPacket) {
+  constexpr uint8_t kMinimalisticRtcpPacket[] = {0x80, 202, 0, 0};
+  EXPECT_TRUE(IsRtcpPacket(kMinimalisticRtcpPacket));
+
+  constexpr uint8_t kWrongRtpVersion[] = {0xc0, 202, 0, 0};
+  EXPECT_FALSE(IsRtcpPacket(kWrongRtpVersion));
+
+  constexpr uint8_t kPacketWithPayloadForRtp[] = {0x80, 225, 0, 0};
+  EXPECT_FALSE(IsRtcpPacket(kPacketWithPayloadForRtp));
+
+  constexpr uint8_t kTooSmallRtcpPacket[] = {0x80, 202, 0};
+  EXPECT_FALSE(IsRtcpPacket(kTooSmallRtcpPacket));
+
+  EXPECT_FALSE(IsRtcpPacket({}));
+}
+
+TEST(RtpUtilTest, ParseRtpPayloadType) {
+  constexpr uint8_t kMinimalisticRtpPacket[] = {0x80, 97,   0,    0,  //
+                                                0,    0,    0,    0,  //
+                                                0x12, 0x34, 0x56, 0x78};
+  EXPECT_EQ(ParseRtpPayloadType(kMinimalisticRtpPacket), 97);
+
+  constexpr uint8_t kMinimalisticRtpPacketWithMarker[] = {
+      0x80, 0x80 | 97, 0,    0,  //
+      0,    0,         0,    0,  //
+      0x12, 0x34,      0x56, 0x78};
+  EXPECT_EQ(ParseRtpPayloadType(kMinimalisticRtpPacketWithMarker), 97);
+}
+
+TEST(RtpUtilTest, ParseRtpSequenceNumber) {
+  constexpr uint8_t kMinimalisticRtpPacket[] = {0x80, 97, 0x12, 0x34,  //
+                                                0,    0,  0,    0,     //
+                                                0,    0,  0,    0};
+  EXPECT_EQ(ParseRtpSequenceNumber(kMinimalisticRtpPacket), 0x1234);
+}
+
+TEST(RtpUtilTest, ParseRtpSsrc) {
+  constexpr uint8_t kMinimalisticRtpPacket[] = {0x80, 97,   0,    0,  //
+                                                0,    0,    0,    0,  //
+                                                0x12, 0x34, 0x56, 0x78};
+  EXPECT_EQ(ParseRtpSsrc(kMinimalisticRtpPacket), 0x12345678u);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header.cc
new file mode 100644
index 0000000000..b07a7beec4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header.cc
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+
+namespace webrtc {
+
+RTPVideoHeader::GenericDescriptorInfo::GenericDescriptorInfo() = default;
+RTPVideoHeader::GenericDescriptorInfo::GenericDescriptorInfo(
+    const GenericDescriptorInfo& other) = default;
+RTPVideoHeader::GenericDescriptorInfo::~GenericDescriptorInfo() = default;
+
+// static
+RTPVideoHeader RTPVideoHeader::FromMetadata(
+    const VideoFrameMetadata& metadata) {
+  RTPVideoHeader rtp_video_header;
+  rtp_video_header.SetFromMetadata(metadata);
+  return rtp_video_header;
+}
+
+RTPVideoHeader::RTPVideoHeader() : video_timing() {}
+RTPVideoHeader::RTPVideoHeader(const RTPVideoHeader& other) = default;
+RTPVideoHeader::~RTPVideoHeader() = default;
+
+VideoFrameMetadata RTPVideoHeader::GetAsMetadata() const {
+  VideoFrameMetadata metadata;
+  metadata.SetFrameType(frame_type);
+  metadata.SetWidth(width);
+  metadata.SetHeight(height);
+  metadata.SetRotation(rotation);
+  metadata.SetContentType(content_type);
+  if (generic) {
+    metadata.SetFrameId(generic->frame_id);
+    metadata.SetSpatialIndex(generic->spatial_index);
+    metadata.SetTemporalIndex(generic->temporal_index);
+    metadata.SetFrameDependencies(generic->dependencies);
+    metadata.SetDecodeTargetIndications(generic->decode_target_indications);
+  }
+  metadata.SetIsLastFrameInPicture(is_last_frame_in_picture);
+  metadata.SetSimulcastIdx(simulcastIdx);
+  metadata.SetCodec(codec);
+  switch (codec) {
+    case VideoCodecType::kVideoCodecVP8:
+      metadata.SetRTPVideoHeaderCodecSpecifics(
+          absl::get<RTPVideoHeaderVP8>(video_type_header));
+      break;
+    case VideoCodecType::kVideoCodecVP9:
+      metadata.SetRTPVideoHeaderCodecSpecifics(
+          absl::get<RTPVideoHeaderVP9>(video_type_header));
+      break;
+    case VideoCodecType::kVideoCodecH264:
+      metadata.SetRTPVideoHeaderCodecSpecifics(
+          absl::get<RTPVideoHeaderH264>(video_type_header));
+      break;
+    default:
+      // Codec-specifics are not supported for this codec.
+      break;
+  }
+  return metadata;
+}
+
+void RTPVideoHeader::SetFromMetadata(const VideoFrameMetadata& metadata) {
+  frame_type = metadata.GetFrameType();
+  width = metadata.GetWidth();
+  height = metadata.GetHeight();
+  rotation = metadata.GetRotation();
+  content_type = metadata.GetContentType();
+  if (!metadata.GetFrameId().has_value()) {
+    generic = absl::nullopt;
+  } else {
+    generic.emplace();
+    generic->frame_id = metadata.GetFrameId().value();
+    generic->spatial_index = metadata.GetSpatialIndex();
+    generic->temporal_index = metadata.GetTemporalIndex();
+    generic->dependencies.assign(metadata.GetFrameDependencies().begin(),
+                                 metadata.GetFrameDependencies().end());
+    generic->decode_target_indications.assign(
+        metadata.GetDecodeTargetIndications().begin(),
+        metadata.GetDecodeTargetIndications().end());
+  }
+  is_last_frame_in_picture = metadata.GetIsLastFrameInPicture();
+  simulcastIdx = metadata.GetSimulcastIdx();
+  codec = metadata.GetCodec();
+  switch (codec) {
+    case VideoCodecType::kVideoCodecVP8:
+      video_type_header = absl::get<RTPVideoHeaderVP8>(
+          metadata.GetRTPVideoHeaderCodecSpecifics());
+      break;
+    case VideoCodecType::kVideoCodecVP9:
+      video_type_header = absl::get<RTPVideoHeaderVP9>(
+          metadata.GetRTPVideoHeaderCodecSpecifics());
+      break;
+    case VideoCodecType::kVideoCodecH264:
+      video_type_header = absl::get<RTPVideoHeaderH264>(
+          metadata.GetRTPVideoHeaderCodecSpecifics());
+      break;
+    default:
+      // Codec-specifics are not supported for this codec.
+      break;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header.h
new file mode 100644
index 0000000000..d185c20c83
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header.h
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
+
+#include <bitset>
+#include <cstdint>
+
+#include "absl/container/inlined_vector.h"
+#include "absl/types/optional.h"
+#include "absl/types/variant.h"
+#include "api/rtp_headers.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/color_space.h"
+#include "api/video/video_codec_type.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_frame_metadata.h"
+#include "api/video/video_frame_type.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_timing.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+
+namespace webrtc {
+// Details passed in the rtp payload for legacy generic rtp packetizer.
+// TODO(bugs.webrtc.org/9772): Deprecate in favor of passing generic video
+// details in an rtp header extension.
+struct RTPVideoHeaderLegacyGeneric {
+  uint16_t picture_id;
+};
+
+using RTPVideoTypeHeader = absl::variant<absl::monostate,
+                                         RTPVideoHeaderVP8,
+                                         RTPVideoHeaderVP9,
+                                         RTPVideoHeaderH264,
+                                         RTPVideoHeaderLegacyGeneric>;
+
+struct RTPVideoHeader {
+  struct GenericDescriptorInfo {
+    GenericDescriptorInfo();
+    GenericDescriptorInfo(const GenericDescriptorInfo& other);
+    ~GenericDescriptorInfo();
+
+    int64_t frame_id = 0;
+    int spatial_index = 0;
+    int temporal_index = 0;
+    absl::InlinedVector<DecodeTargetIndication, 10> decode_target_indications;
+    absl::InlinedVector<int64_t, 5> dependencies;
+    absl::InlinedVector<int, 4> chain_diffs;
+    std::bitset<32> active_decode_targets = ~uint32_t{0};
+  };
+
+  static RTPVideoHeader FromMetadata(const VideoFrameMetadata& metadata);
+
+  RTPVideoHeader();
+  RTPVideoHeader(const RTPVideoHeader& other);
+
+  ~RTPVideoHeader();
+
+  // The subset of RTPVideoHeader that is exposed in the Insertable Streams API.
+  VideoFrameMetadata GetAsMetadata() const;
+  void SetFromMetadata(const VideoFrameMetadata& metadata);
+
+  absl::optional<GenericDescriptorInfo> generic;
+
+  VideoFrameType frame_type = VideoFrameType::kEmptyFrame;
+  uint16_t width = 0;
+  uint16_t height = 0;
+  VideoRotation rotation = VideoRotation::kVideoRotation_0;
+  VideoContentType content_type = VideoContentType::UNSPECIFIED;
+  bool is_first_packet_in_frame = false;
+  bool is_last_packet_in_frame = false;
+  bool is_last_frame_in_picture = true;
+  uint8_t simulcastIdx = 0;
+  VideoCodecType codec = VideoCodecType::kVideoCodecGeneric;
+
+  VideoPlayoutDelay playout_delay;
+  VideoSendTiming video_timing;
+  absl::optional<ColorSpace> color_space;
+  // This field is meant for media quality testing purpose only. When enabled it
+  // carries the webrtc::VideoFrame id field from the sender to the receiver.
+  absl::optional<uint16_t> video_frame_tracking_id;
+  RTPVideoTypeHeader video_type_header;
+
+  // When provided, is sent as is as an RTP header extension according to
+  // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time.
+  // Otherwise, it is derived from other relevant information.
+  absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header_unittest.cc
new file mode 100644
index 0000000000..335fa1a8a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_header_unittest.cc
@@ -0,0 +1,351 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+
+#include "api/video/video_frame_metadata.h"
+#include "api/video/video_frame_type.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+
+TEST(RTPVideoHeaderTest, FrameType_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetFrameType(), VideoFrameType::kVideoFrameKey);
+}
+
+TEST(RTPVideoHeaderTest, FrameType_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetFrameType(VideoFrameType::kVideoFrameKey);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_EQ(video_header.frame_type, VideoFrameType::kVideoFrameKey);
+}
+
+TEST(RTPVideoHeaderTest, Width_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  video_header.width = 1280u;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetWidth(), 1280u);
+}
+
+TEST(RTPVideoHeaderTest, Width_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetWidth(1280u);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_EQ(video_header.width, 1280u);
+}
+
+TEST(RTPVideoHeaderTest, Height_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  video_header.height = 720u;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetHeight(), 720u);
+}
+
+TEST(RTPVideoHeaderTest, Height_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetHeight(720u);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_EQ(video_header.height, 720u);
+}
+
+TEST(RTPVideoHeaderTest, Rotation_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  video_header.rotation = VideoRotation::kVideoRotation_90;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetRotation(), VideoRotation::kVideoRotation_90);
+}
+
+TEST(RTPVideoHeaderTest, Rotation_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetRotation(VideoRotation::kVideoRotation_90);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_EQ(video_header.rotation, VideoRotation::kVideoRotation_90);
+}
+
+TEST(RTPVideoHeaderTest, ContentType_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  video_header.content_type = VideoContentType::SCREENSHARE;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetContentType(), VideoContentType::SCREENSHARE);
+}
+
+TEST(RTPVideoHeaderTest, ContentType_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetContentType(VideoContentType::SCREENSHARE);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_EQ(video_header.content_type, VideoContentType::SCREENSHARE);
+}
+
+TEST(RTPVideoHeaderTest, FrameId_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.frame_id = 10;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetFrameId().value(), 10);
+}
+
+TEST(RTPVideoHeaderTest, FrameId_GetAsMetadataWhenGenericIsMissing) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_FALSE(metadata.GetFrameId().has_value());
+}
+
+TEST(RTPVideoHeaderTest, FrameId_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetFrameId(10);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_TRUE(video_header.generic.has_value());
+  EXPECT_EQ(video_header.generic->frame_id, 10);
+}
+
+TEST(RTPVideoHeaderTest, FrameId_FromMetadataWhenFrameIdIsMissing) {
+  VideoFrameMetadata metadata;
+  metadata.SetFrameId(absl::nullopt);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_FALSE(video_header.generic.has_value());
+}
+
+TEST(RTPVideoHeaderTest, SpatialIndex_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.spatial_index = 2;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetSpatialIndex(), 2);
+}
+
+TEST(RTPVideoHeaderTest, SpatialIndex_GetAsMetadataWhenGenericIsMissing) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_EQ(metadata.GetSpatialIndex(), 0);
+}
+
+TEST(RTPVideoHeaderTest, SpatialIndex_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetFrameId(123);  // Must have a frame ID for related properties.
+  metadata.SetSpatialIndex(2);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_TRUE(video_header.generic.has_value());
+  EXPECT_EQ(video_header.generic->spatial_index, 2);
+}
+
+TEST(RTPVideoHeaderTest, TemporalIndex_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.temporal_index = 3;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetTemporalIndex(), 3);
+}
+
+TEST(RTPVideoHeaderTest, TemporalIndex_GetAsMetadataWhenGenericIsMissing) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_EQ(metadata.GetTemporalIndex(), 0);
+}
+
+TEST(RTPVideoHeaderTest, TemporalIndex_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetFrameId(123);  // Must have a frame ID for related properties.
+  metadata.SetTemporalIndex(3);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_TRUE(video_header.generic.has_value());
+  EXPECT_EQ(video_header.generic->temporal_index, 3);
+}
+
+TEST(RTPVideoHeaderTest, FrameDependencies_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.dependencies = {5, 6, 7};
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_THAT(metadata.GetFrameDependencies(), ElementsAre(5, 6, 7));
+}
+
+TEST(RTPVideoHeaderTest, FrameDependency_GetAsMetadataWhenGenericIsMissing) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_THAT(metadata.GetFrameDependencies(), IsEmpty());
+}
+
+TEST(RTPVideoHeaderTest, FrameDependencies_FromMetadata) {
+  VideoFrameMetadata metadata;
+  absl::InlinedVector<int64_t, 5> dependencies = {5, 6, 7};
+  metadata.SetFrameId(123);  // Must have a frame ID for related properties.
+  metadata.SetFrameDependencies(dependencies);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_TRUE(video_header.generic.has_value());
+  EXPECT_THAT(video_header.generic->dependencies, ElementsAre(5, 6, 7));
+}
+
+TEST(RTPVideoHeaderTest, DecodeTargetIndications_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch};
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_THAT(metadata.GetDecodeTargetIndications(),
+              ElementsAre(DecodeTargetIndication::kSwitch));
+}
+
+TEST(RTPVideoHeaderTest,
+     DecodeTargetIndications_GetAsMetadataWhenGenericIsMissing) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_THAT(metadata.GetDecodeTargetIndications(), IsEmpty());
+}
+
+TEST(RTPVideoHeaderTest, DecodeTargetIndications_FromMetadata) {
+  VideoFrameMetadata metadata;
+  absl::InlinedVector<DecodeTargetIndication, 10> decode_target_indications = {
+      DecodeTargetIndication::kSwitch};
+  metadata.SetFrameId(123);  // Must have a frame ID for related properties.
+  metadata.SetDecodeTargetIndications(decode_target_indications);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_TRUE(video_header.generic.has_value());
+  EXPECT_THAT(video_header.generic->decode_target_indications,
+              ElementsAre(DecodeTargetIndication::kSwitch));
+}
+
+TEST(RTPVideoHeaderTest, IsLastFrameInPicture_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  video_header.is_last_frame_in_picture = false;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_FALSE(metadata.GetIsLastFrameInPicture());
+}
+
+TEST(RTPVideoHeaderTest, IsLastFrameInPicture_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetIsLastFrameInPicture(false);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_FALSE(video_header.is_last_frame_in_picture);
+}
+
+TEST(RTPVideoHeaderTest, SimulcastIdx_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  video_header.simulcastIdx = 123;
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetSimulcastIdx(), 123);
+}
+
+TEST(RTPVideoHeaderTest, SimulcastIdx_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetSimulcastIdx(123);
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_EQ(video_header.simulcastIdx, 123);
+}
+
+TEST(RTPVideoHeaderTest, Codec_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  video_header.codec = VideoCodecType::kVideoCodecVP9;
+  video_header.video_type_header = RTPVideoHeaderVP9();
+  VideoFrameMetadata metadata = video_header.GetAsMetadata();
+  EXPECT_EQ(metadata.GetCodec(), VideoCodecType::kVideoCodecVP9);
+}
+
+TEST(RTPVideoHeaderTest, Codec_FromMetadata) {
+  VideoFrameMetadata metadata;
+  metadata.SetCodec(VideoCodecType::kVideoCodecVP9);
+  metadata.SetRTPVideoHeaderCodecSpecifics(RTPVideoHeaderVP9());
+  RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+  EXPECT_EQ(video_header.codec, VideoCodecType::kVideoCodecVP9);
+}
+
+TEST(RTPVideoHeaderTest, RTPVideoHeaderCodecSpecifics_GetAsMetadata) {
+  RTPVideoHeader video_header;
+  {
+    video_header.codec = VideoCodecType::kVideoCodecVP8;
+    RTPVideoHeaderVP8 vp8_specifics;
+    vp8_specifics.InitRTPVideoHeaderVP8();
+    vp8_specifics.pictureId = 42;
+    video_header.video_type_header = vp8_specifics;
+    VideoFrameMetadata metadata = video_header.GetAsMetadata();
+    EXPECT_EQ(
+        absl::get<RTPVideoHeaderVP8>(metadata.GetRTPVideoHeaderCodecSpecifics())
+            .pictureId,
+        vp8_specifics.pictureId);
+  }
+  {
+    video_header.codec = VideoCodecType::kVideoCodecVP9;
+    RTPVideoHeaderVP9 vp9_specifics;
+    vp9_specifics.InitRTPVideoHeaderVP9();
+    vp9_specifics.max_picture_id = 42;
+    video_header.video_type_header = vp9_specifics;
+    VideoFrameMetadata metadata = video_header.GetAsMetadata();
+    EXPECT_EQ(
+        absl::get<RTPVideoHeaderVP9>(metadata.GetRTPVideoHeaderCodecSpecifics())
+            .max_picture_id,
+        vp9_specifics.max_picture_id);
+  }
+  {
+    video_header.codec = VideoCodecType::kVideoCodecH264;
+    RTPVideoHeaderH264 h264_specifics;
+    h264_specifics.nalu_type = 42;
+    video_header.video_type_header = h264_specifics;
+    VideoFrameMetadata metadata = video_header.GetAsMetadata();
+    EXPECT_EQ(absl::get<RTPVideoHeaderH264>(
+                  metadata.GetRTPVideoHeaderCodecSpecifics())
+                  .nalu_type,
+              h264_specifics.nalu_type);
+  }
+}
+
+TEST(RTPVideoHeaderTest, RTPVideoHeaderCodecSpecifics_FromMetadata) {
+  VideoFrameMetadata metadata;
+  {
+    metadata.SetCodec(VideoCodecType::kVideoCodecVP8);
+    RTPVideoHeaderVP8 vp8_specifics;
+    vp8_specifics.InitRTPVideoHeaderVP8();
+    vp8_specifics.pictureId = 42;
+    metadata.SetRTPVideoHeaderCodecSpecifics(vp8_specifics);
+    RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+    EXPECT_EQ(
+        absl::get<RTPVideoHeaderVP8>(video_header.video_type_header).pictureId,
+        42);
+  }
+  {
+    metadata.SetCodec(VideoCodecType::kVideoCodecVP9);
+    RTPVideoHeaderVP9 vp9_specifics;
+    vp9_specifics.InitRTPVideoHeaderVP9();
+    vp9_specifics.max_picture_id = 42;
+    metadata.SetRTPVideoHeaderCodecSpecifics(vp9_specifics);
+    RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+    EXPECT_EQ(absl::get<RTPVideoHeaderVP9>(video_header.video_type_header)
+                  .max_picture_id,
+              42);
+  }
+  {
+    metadata.SetCodec(VideoCodecType::kVideoCodecH264);
+    RTPVideoHeaderH264 h264_specifics;
+    h264_specifics.nalu_type = 42;
+    metadata.SetRTPVideoHeaderCodecSpecifics(h264_specifics);
+    RTPVideoHeader video_header = RTPVideoHeader::FromMetadata(metadata);
+    EXPECT_EQ(
+        absl::get<RTPVideoHeaderH264>(video_header.video_type_header).nalu_type,
+        42);
+  }
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.cc
new file mode 100644
index 0000000000..78fd7c9808
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.cc
@@ -0,0 +1,323 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/algorithm/container.h"
+#include "api/video/video_layers_allocation.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/leb128.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+constexpr RTPExtensionType RtpVideoLayersAllocationExtension::kId;
+
+namespace {
+
+constexpr int kMaxNumRtpStreams = 4;
+
+bool AllocationIsValid(const VideoLayersAllocation& allocation) {
+  // Since all multivalue fields are stored in (rtp_stream_id, spatial_id) order
+  // assume `allocation.active_spatial_layers` is already sorted. It is simpler
+  // to assemble it in the sorted way than to resort during serialization.
+  if (!absl::c_is_sorted(
+          allocation.active_spatial_layers,
+          [](const VideoLayersAllocation::SpatialLayer& lhs,
+             const VideoLayersAllocation::SpatialLayer& rhs) {
+            return std::make_tuple(lhs.rtp_stream_index, lhs.spatial_id) <
+                   std::make_tuple(rhs.rtp_stream_index, rhs.spatial_id);
+          })) {
+    return false;
+  }
+
+  int max_rtp_stream_idx = 0;
+  for (const auto& spatial_layer : allocation.active_spatial_layers) {
+    if (spatial_layer.rtp_stream_index < 0 ||
+        spatial_layer.rtp_stream_index >= 4) {
+      return false;
+    }
+    if (spatial_layer.spatial_id < 0 || spatial_layer.spatial_id >= 4) {
+      return false;
+    }
+    if (spatial_layer.target_bitrate_per_temporal_layer.empty() ||
+        spatial_layer.target_bitrate_per_temporal_layer.size() > 4) {
+      return false;
+    }
+    if (max_rtp_stream_idx < spatial_layer.rtp_stream_index) {
+      max_rtp_stream_idx = spatial_layer.rtp_stream_index;
+    }
+    if (allocation.resolution_and_frame_rate_is_valid) {
+      // TODO(danilchap): Add check width and height are no more than 0x10000
+      // when width and height become larger type and thus would support maximum
+      // resolution.
+      if (spatial_layer.width <= 0) {
+        return false;
+      }
+      if (spatial_layer.height <= 0) {
+        return false;
+      }
+      if (spatial_layer.frame_rate_fps > 255) {
+        return false;
+      }
+    }
+  }
+  if (allocation.rtp_stream_index < 0 ||
+      (!allocation.active_spatial_layers.empty() &&
+       allocation.rtp_stream_index > max_rtp_stream_idx)) {
+    return false;
+  }
+  return true;
+}
+
+struct SpatialLayersBitmasks {
+  int max_rtp_stream_id = 0;
+  uint8_t spatial_layer_bitmask[kMaxNumRtpStreams] = {};
+  bool bitmasks_are_the_same = true;
+};
+
+SpatialLayersBitmasks SpatialLayersBitmasksPerRtpStream(
+    const VideoLayersAllocation& allocation) {
+  RTC_DCHECK(AllocationIsValid(allocation));
+  SpatialLayersBitmasks result;
+  for (const auto& layer : allocation.active_spatial_layers) {
+    result.spatial_layer_bitmask[layer.rtp_stream_index] |=
+        (1u << layer.spatial_id);
+    if (result.max_rtp_stream_id < layer.rtp_stream_index) {
+      result.max_rtp_stream_id = layer.rtp_stream_index;
+    }
+  }
+  for (int i = 1; i <= result.max_rtp_stream_id; ++i) {
+    if (result.spatial_layer_bitmask[i] != result.spatial_layer_bitmask[0]) {
+      result.bitmasks_are_the_same = false;
+      break;
+    }
+  }
+  return result;
+}
+
+}  // namespace
+
+// See /docs/native-code/rtp-rtpext/video-layers-allocation00/README.md
+// for the description of the format.
+
+bool RtpVideoLayersAllocationExtension::Write(
+    rtc::ArrayView<uint8_t> data,
+    const VideoLayersAllocation& allocation) {
+  RTC_DCHECK(AllocationIsValid(allocation));
+  RTC_DCHECK_GE(data.size(), ValueSize(allocation));
+
+  if (allocation.active_spatial_layers.empty()) {
+    data[0] = 0;
+    return true;
+  }
+
+  SpatialLayersBitmasks slb = SpatialLayersBitmasksPerRtpStream(allocation);
+  uint8_t* write_at = data.data();
+  // First half of the header byte.
+  *write_at = (allocation.rtp_stream_index << 6);
+  // number of rtp stream - 1 is the same as the maximum rtp_stream_id.
+  *write_at |= slb.max_rtp_stream_id << 4;
+  if (slb.bitmasks_are_the_same) {
+    // Second half of the header byte.
+    *write_at |= slb.spatial_layer_bitmask[0];
+  } else {
+    // spatial layer bitmasks when they are different for different RTP streams.
+    *++write_at =
+        (slb.spatial_layer_bitmask[0] << 4) | slb.spatial_layer_bitmask[1];
+    if (slb.max_rtp_stream_id >= 2) {
+      *++write_at =
+          (slb.spatial_layer_bitmask[2] << 4) | slb.spatial_layer_bitmask[3];
+    }
+  }
+  ++write_at;
+
+  {  // Number of temporal layers.
+    int bit_offset = 8;
+    *write_at = 0;
+    for (const auto& layer : allocation.active_spatial_layers) {
+      if (bit_offset == 0) {
+        bit_offset = 6;
+        *++write_at = 0;
+      } else {
+        bit_offset -= 2;
+      }
+      *write_at |=
+          ((layer.target_bitrate_per_temporal_layer.size() - 1) << bit_offset);
+    }
+    ++write_at;
+  }
+
+  // Target bitrates.
+  for (const auto& spatial_layer : allocation.active_spatial_layers) {
+    for (const DataRate& bitrate :
+         spatial_layer.target_bitrate_per_temporal_layer) {
+      write_at += WriteLeb128(bitrate.kbps(), write_at);
+    }
+  }
+
+  if (allocation.resolution_and_frame_rate_is_valid) {
+    for (const auto& spatial_layer : allocation.active_spatial_layers) {
+      ByteWriter<uint16_t>::WriteBigEndian(write_at, spatial_layer.width - 1);
+      write_at += 2;
+      ByteWriter<uint16_t>::WriteBigEndian(write_at, spatial_layer.height - 1);
+      write_at += 2;
+      *write_at = spatial_layer.frame_rate_fps;
+      ++write_at;
+    }
+  }
+  RTC_DCHECK_EQ(write_at - data.data(), ValueSize(allocation));
+  return true;
+}
+
+bool RtpVideoLayersAllocationExtension::Parse(
+    rtc::ArrayView<const uint8_t> data,
+    VideoLayersAllocation* allocation) {
+  if (data.empty() || allocation == nullptr) {
+    return false;
+  }
+
+  allocation->active_spatial_layers.clear();
+
+  const uint8_t* read_at = data.data();
+  const uint8_t* const end = data.data() + data.size();
+
+  if (data.size() == 1 && *read_at == 0) {
+    allocation->rtp_stream_index = 0;
+    allocation->resolution_and_frame_rate_is_valid = true;
+    return AllocationIsValid(*allocation);
+  }
+
+  // Header byte.
+  allocation->rtp_stream_index = *read_at >> 6;
+  int num_rtp_streams = 1 + ((*read_at >> 4) & 0b11);
+  uint8_t spatial_layers_bitmasks[kMaxNumRtpStreams];
+  spatial_layers_bitmasks[0] = *read_at & 0b1111;
+
+  if (spatial_layers_bitmasks[0] != 0) {
+    for (int i = 1; i < num_rtp_streams; ++i) {
+      spatial_layers_bitmasks[i] = spatial_layers_bitmasks[0];
+    }
+  } else {
+    // Spatial layer bitmasks when they are different for different RTP streams.
+    if (++read_at == end) {
+      return false;
+    }
+    spatial_layers_bitmasks[0] = *read_at >> 4;
+    spatial_layers_bitmasks[1] = *read_at & 0b1111;
+    if (num_rtp_streams > 2) {
+      if (++read_at == end) {
+        return false;
+      }
+      spatial_layers_bitmasks[2] = *read_at >> 4;
+      spatial_layers_bitmasks[3] = *read_at & 0b1111;
+    }
+  }
+  if (++read_at == end) {
+    return false;
+  }
+
+  // Read number of temporal layers,
+  // Create `allocation->active_spatial_layers` while iterating though it.
+  int bit_offset = 8;
+  for (int stream_idx = 0; stream_idx < num_rtp_streams; ++stream_idx) {
+    for (int sid = 0; sid < VideoLayersAllocation::kMaxSpatialIds; ++sid) {
+      if ((spatial_layers_bitmasks[stream_idx] & (1 << sid)) == 0) {
+        continue;
+      }
+
+      if (bit_offset == 0) {
+        bit_offset = 6;
+        if (++read_at == end) {
+          return false;
+        }
+      } else {
+        bit_offset -= 2;
+      }
+      int num_temporal_layers = 1 + ((*read_at >> bit_offset) & 0b11);
+      allocation->active_spatial_layers.emplace_back();
+      auto& layer = allocation->active_spatial_layers.back();
+      layer.rtp_stream_index = stream_idx;
+      layer.spatial_id = sid;
+      layer.target_bitrate_per_temporal_layer.resize(num_temporal_layers,
+                                                     DataRate::Zero());
+    }
+  }
+  if (++read_at == end) {
+    return false;
+  }
+
+  // Target bitrates.
+  for (auto& layer : allocation->active_spatial_layers) {
+    for (DataRate& rate : layer.target_bitrate_per_temporal_layer) {
+      uint64_t bitrate_kbps = ReadLeb128(read_at, end);
+      // bitrate_kbps might represent larger values than DataRate type,
+      // discard unreasonably large values.
+      if (read_at == nullptr || bitrate_kbps > 1'000'000) {
+        return false;
+      }
+      rate = DataRate::KilobitsPerSec(bitrate_kbps);
+    }
+  }
+
+  if (read_at == end) {
+    allocation->resolution_and_frame_rate_is_valid = false;
+    return AllocationIsValid(*allocation);
+  }
+
+  if (read_at + 5 * allocation->active_spatial_layers.size() != end) {
+    // data is left, but it size is not what can be used for resolutions and
+    // framerates.
+    return false;
+  }
+  allocation->resolution_and_frame_rate_is_valid = true;
+  for (auto& layer : allocation->active_spatial_layers) {
+    layer.width = 1 + ByteReader<uint16_t, 2>::ReadBigEndian(read_at);
+    read_at += 2;
+    layer.height = 1 + ByteReader<uint16_t, 2>::ReadBigEndian(read_at);
+    read_at += 2;
+    layer.frame_rate_fps = *read_at;
+    ++read_at;
+  }
+
+  return AllocationIsValid(*allocation);
+}
+
+size_t RtpVideoLayersAllocationExtension::ValueSize(
+    const VideoLayersAllocation& allocation) {
+  if (allocation.active_spatial_layers.empty()) {
+    return 1;
+  }
+  size_t result = 1;  // header
+  SpatialLayersBitmasks slb = SpatialLayersBitmasksPerRtpStream(allocation);
+  if (!slb.bitmasks_are_the_same) {
+    ++result;
+    if (slb.max_rtp_stream_id >= 2) {
+      ++result;
+    }
+  }
+  // 2 bits per active spatial layer, rounded up to full byte, i.e.
+  // 0.25 byte per active spatial layer.
+  result += (allocation.active_spatial_layers.size() + 3) / 4;
+  for (const auto& spatial_layer : allocation.active_spatial_layers) {
+    for (DataRate value : spatial_layer.target_bitrate_per_temporal_layer) {
+      result += Leb128Size(value.kbps());
+    }
+  }
+  if (allocation.resolution_and_frame_rate_is_valid) {
+    result += 5 * allocation.active_spatial_layers.size();
+  }
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h
new file mode 100644
index 0000000000..3f1603bcd8
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_LAYERS_ALLOCATION_EXTENSION_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_LAYERS_ALLOCATION_EXTENSION_H_
+
+#include "absl/strings/string_view.h"
+#include "api/rtp_parameters.h"
+#include "api/video/video_layers_allocation.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+// TODO(bugs.webrtc.org/12000): Note that this extensions is being developed and
+// the wire format will likely change.
+class RtpVideoLayersAllocationExtension {
+ public:
+  using value_type = VideoLayersAllocation;
+  static constexpr RTPExtensionType kId = kRtpExtensionVideoLayersAllocation;
+  static constexpr absl::string_view Uri() {
+    return RtpExtension::kVideoLayersAllocationUri;
+  }
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    VideoLayersAllocation* allocation);
+  static size_t ValueSize(const VideoLayersAllocation& allocation);
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    const VideoLayersAllocation& allocation);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_LAYERS_ALLOCATION_EXTENSION_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc
new file mode 100644
index 0000000000..db077409ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc
@@ -0,0 +1,287 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
+
+#include "api/video/video_layers_allocation.h"
+#include "rtc_base/bit_buffer.h"
+#include "rtc_base/buffer.h"
+
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace {
+
+TEST(RtpVideoLayersAllocationExtension, WriteEmptyLayersAllocationReturnsTrue) {
+  VideoLayersAllocation written_allocation;
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+}
+
+TEST(RtpVideoLayersAllocationExtension,
+     CanWriteAndParseLayersAllocationWithZeroSpatialLayers) {
+  // We require the resolution_and_frame_rate_is_valid to be set to true in
+  // order to send an "empty" allocation.
+  VideoLayersAllocation written_allocation;
+  written_allocation.resolution_and_frame_rate_is_valid = true;
+  written_allocation.rtp_stream_index = 0;
+
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+
+  VideoLayersAllocation parsed_allocation;
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Parse(buffer, &parsed_allocation));
+  EXPECT_EQ(written_allocation, parsed_allocation);
+}
+
+TEST(RtpVideoLayersAllocationExtension,
+     CanWriteAndParse2SpatialWith2TemporalLayers) {
+  VideoLayersAllocation written_allocation;
+  written_allocation.rtp_stream_index = 1;
+  written_allocation.active_spatial_layers = {
+      {
+          /*rtp_stream_index*/ 0,
+          /*spatial_id*/ 0,
+          /*target_bitrate_per_temporal_layer*/
+          {DataRate::KilobitsPerSec(25), DataRate::KilobitsPerSec(50)},
+          /*width*/ 0,
+          /*height*/ 0,
+          /*frame_rate_fps*/ 0,
+      },
+      {
+          /*rtp_stream_index*/ 1,
+          /*spatial_id*/ 0,
+          /*target_bitrate_per_temporal_layer*/
+          {DataRate::KilobitsPerSec(100), DataRate::KilobitsPerSec(200)},
+          /*width*/ 0,
+          /*height*/ 0,
+          /*frame_rate_fps*/ 0,
+      },
+  };
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+  VideoLayersAllocation parsed_allocation;
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Parse(buffer, &parsed_allocation));
+  EXPECT_EQ(written_allocation, parsed_allocation);
+}
+
+TEST(RtpVideoLayersAllocationExtension,
+     CanWriteAndParseAllocationWithDifferentNumerOfSpatialLayers) {
+  VideoLayersAllocation written_allocation;
+  written_allocation.rtp_stream_index = 1;
+  written_allocation.active_spatial_layers = {
+      {/*rtp_stream_index*/ 0,
+       /*spatial_id*/ 0,
+       /*target_bitrate_per_temporal_layer*/ {DataRate::KilobitsPerSec(50)},
+       /*width*/ 0,
+       /*height*/ 0,
+       /*frame_rate_fps*/ 0},
+      {/*rtp_stream_index*/ 1,
+       /*spatial_id*/ 0,
+       /*target_bitrate_per_temporal_layer*/ {DataRate::KilobitsPerSec(100)},
+       /*width*/ 0,
+       /*height*/ 0,
+       /*frame_rate_fps*/ 0},
+      {/*rtp_stream_index*/ 1,
+       /*spatial_id*/ 1,
+       /*target_bitrate_per_temporal_layer*/ {DataRate::KilobitsPerSec(200)},
+       /*width*/ 0,
+       /*height*/ 0,
+       /*frame_rate_fps*/ 0},
+  };
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+  VideoLayersAllocation parsed_allocation;
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Parse(buffer, &parsed_allocation));
+  EXPECT_EQ(written_allocation, parsed_allocation);
+}
+
+TEST(RtpVideoLayersAllocationExtension,
+     CanWriteAndParseAllocationWithSkippedLowerSpatialLayer) {
+  VideoLayersAllocation written_allocation;
+  written_allocation.rtp_stream_index = 1;
+  written_allocation.active_spatial_layers = {
+      {/*rtp_stream_index*/ 0,
+       /*spatial_id*/ 0,
+       /*target_bitrate_per_temporal_layer*/ {DataRate::KilobitsPerSec(50)},
+       /*width*/ 0,
+       /*height*/ 0,
+       /*frame_rate_fps*/ 0},
+      {/*rtp_stream_index*/ 1,
+       /*spatial_id*/ 1,
+       /*target_bitrate_per_temporal_layer*/ {DataRate::KilobitsPerSec(200)},
+       /*width*/ 0,
+       /*height*/ 0,
+       /*frame_rate_fps*/ 0},
+  };
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+  VideoLayersAllocation parsed_allocation;
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Parse(buffer, &parsed_allocation));
+  EXPECT_EQ(written_allocation, parsed_allocation);
+}
+
+TEST(RtpVideoLayersAllocationExtension,
+     CanWriteAndParseAllocationWithSkippedRtpStreamIds) {
+  VideoLayersAllocation written_allocation;
+  written_allocation.rtp_stream_index = 2;
+  written_allocation.active_spatial_layers = {
+      {/*rtp_stream_index*/ 0,
+       /*spatial_id*/ 0,
+       /*target_bitrate_per_temporal_layer*/ {DataRate::KilobitsPerSec(50)},
+       /*width*/ 0,
+       /*height*/ 0,
+       /*frame_rate_fps*/ 0},
+      {/*rtp_stream_index*/ 2,
+       /*spatial_id*/ 0,
+       /*target_bitrate_per_temporal_layer*/ {DataRate::KilobitsPerSec(200)},
+       /*width*/ 0,
+       /*height*/ 0,
+       /*frame_rate_fps*/ 0},
+  };
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+  VideoLayersAllocation parsed_allocation;
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Parse(buffer, &parsed_allocation));
+  EXPECT_EQ(written_allocation, parsed_allocation);
+}
+
+TEST(RtpVideoLayersAllocationExtension,
+     CanWriteAndParseAllocationWithDifferentNumerOfTemporalLayers) {
+  VideoLayersAllocation written_allocation;
+  written_allocation.rtp_stream_index = 1;
+  written_allocation.active_spatial_layers = {
+      {
+          /*rtp_stream_index*/ 0,
+          /*spatial_id*/ 0,
+          /*target_bitrate_per_temporal_layer*/
+          {DataRate::KilobitsPerSec(25), DataRate::KilobitsPerSec(50)},
+          /*width*/ 0,
+          /*height*/ 0,
+          /*frame_rate_fps*/ 0,
+      },
+      {
+          /*rtp_stream_index*/ 1,
+          /*spatial_id*/ 0,
+          /*target_bitrate_per_temporal_layer*/ {DataRate::KilobitsPerSec(100)},
+          /*width*/ 0,
+          /*height*/ 0,
+          /*frame_rate_fps*/ 0,
+      },
+  };
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+  VideoLayersAllocation parsed_allocation;
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Parse(buffer, &parsed_allocation));
+  EXPECT_EQ(written_allocation, parsed_allocation);
+}
+
+TEST(RtpVideoLayersAllocationExtension,
+     CanWriteAndParseAllocationWithResolution) {
+  VideoLayersAllocation written_allocation;
+  written_allocation.rtp_stream_index = 1;
+  written_allocation.resolution_and_frame_rate_is_valid = true;
+  written_allocation.active_spatial_layers = {
+      {
+          /*rtp_stream_index*/ 0,
+          /*spatial_id*/ 0,
+          /*target_bitrate_per_temporal_layer*/
+          {DataRate::KilobitsPerSec(25), DataRate::KilobitsPerSec(50)},
+          /*width*/ 320,
+          /*height*/ 240,
+          /*frame_rate_fps*/ 8,
+      },
+      {
+          /*rtp_stream_index*/ 1,
+          /*spatial_id*/ 1,
+          /*target_bitrate_per_temporal_layer*/
+          {DataRate::KilobitsPerSec(100), DataRate::KilobitsPerSec(200)},
+          /*width*/ 640,
+          /*height*/ 320,
+          /*frame_rate_fps*/ 30,
+      },
+  };
+
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+  VideoLayersAllocation parsed_allocation;
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Parse(buffer, &parsed_allocation));
+  EXPECT_EQ(written_allocation, parsed_allocation);
+}
+
+TEST(RtpVideoLayersAllocationExtension,
+     WriteEmptyAllocationCanHaveAnyRtpStreamIndex) {
+  VideoLayersAllocation written_allocation;
+  written_allocation.rtp_stream_index = 1;
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  EXPECT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+}
+
+TEST(RtpVideoLayersAllocationExtension, DiscardsOverLargeDataRate) {
+  constexpr uint8_t buffer[] = {0x4b, 0xf6, 0xff, 0xff, 0xff, 0xff, 0xff,
+                                0xff, 0xcb, 0x78, 0xeb, 0x8d, 0xb5, 0x31};
+  VideoLayersAllocation allocation;
+  EXPECT_FALSE(RtpVideoLayersAllocationExtension::Parse(buffer, &allocation));
+}
+
+TEST(RtpVideoLayersAllocationExtension, DiscardsInvalidHeight) {
+  VideoLayersAllocation written_allocation;
+  written_allocation.rtp_stream_index = 0;
+  written_allocation.resolution_and_frame_rate_is_valid = true;
+  written_allocation.active_spatial_layers = {
+      {
+          /*rtp_stream_index*/ 0,
+          /*spatial_id*/ 0,
+          /*target_bitrate_per_temporal_layer*/
+          {DataRate::KilobitsPerSec(25), DataRate::KilobitsPerSec(50)},
+          /*width*/ 320,
+          /*height*/ 240,
+          /*frame_rate_fps*/ 8,
+      },
+  };
+  rtc::Buffer buffer(
+      RtpVideoLayersAllocationExtension::ValueSize(written_allocation));
+  ASSERT_TRUE(
+      RtpVideoLayersAllocationExtension::Write(buffer, written_allocation));
+
+  // Modify the height to be invalid.
+  buffer[buffer.size() - 3] = 0xff;
+  buffer[buffer.size() - 2] = 0xff;
+  VideoLayersAllocation allocation;
+  EXPECT_FALSE(RtpVideoLayersAllocationExtension::Parse(buffer, &allocation));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker.cc
new file mode 100644
index 0000000000..65f21700d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/source_tracker.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+constexpr int64_t SourceTracker::kTimeoutMs;
+
+SourceTracker::SourceTracker(Clock* clock) : clock_(clock) {}
+
+void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos) {
+  if (packet_infos.empty()) {
+    return;
+  }
+
+  TRACE_EVENT0("webrtc", "SourceTracker::OnFrameDelivered");
+
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  MutexLock lock_scope(&lock_);
+
+  for (const RtpPacketInfo& packet_info : packet_infos) {
+    for (uint32_t csrc : packet_info.csrcs()) {
+      SourceKey key(RtpSourceType::CSRC, csrc);
+      SourceEntry& entry = UpdateEntry(key);
+
+      const auto packet_time = packet_info.receive_time().ms();
+      entry.timestamp_ms = packet_time ? packet_time : now_ms;
+      entry.audio_level = packet_info.audio_level();
+      entry.absolute_capture_time = packet_info.absolute_capture_time();
+      entry.local_capture_clock_offset =
+          packet_info.local_capture_clock_offset();
+      entry.rtp_timestamp = packet_info.rtp_timestamp();
+    }
+
+    SourceKey key(RtpSourceType::SSRC, packet_info.ssrc());
+    SourceEntry& entry = UpdateEntry(key);
+
+    entry.timestamp_ms = now_ms;
+    entry.audio_level = packet_info.audio_level();
+    entry.absolute_capture_time = packet_info.absolute_capture_time();
+    entry.local_capture_clock_offset = packet_info.local_capture_clock_offset();
+    entry.rtp_timestamp = packet_info.rtp_timestamp();
+  }
+
+  PruneEntries(now_ms);
+}
+
+std::vector<RtpSource> SourceTracker::GetSources() const {
+  std::vector<RtpSource> sources;
+
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  MutexLock lock_scope(&lock_);
+
+  PruneEntries(now_ms);
+
+  for (const auto& pair : list_) {
+    const SourceKey& key = pair.first;
+    const SourceEntry& entry = pair.second;
+
+    sources.emplace_back(
+        entry.timestamp_ms, key.source, key.source_type, entry.rtp_timestamp,
+        RtpSource::Extensions{
+            .audio_level = entry.audio_level,
+            .absolute_capture_time = entry.absolute_capture_time,
+            .local_capture_clock_offset = entry.local_capture_clock_offset});
+  }
+
+  std::sort(sources.begin(), sources.end(), [](const auto &a, const auto &b){
+    return a.timestamp_ms() > b.timestamp_ms();
+  });
+
+  return sources;
+}
+
+SourceTracker::SourceEntry& SourceTracker::UpdateEntry(const SourceKey& key) {
+  // We intentionally do |find() + emplace()|, instead of checking the return
+  // value of `emplace()`, for performance reasons. It's much more likely for
+  // the key to already exist than for it not to.
+  auto map_it = map_.find(key);
+  if (map_it == map_.end()) {
+    // Insert a new entry at the front of the list.
+    list_.emplace_front(key, SourceEntry());
+    map_.emplace(key, list_.begin());
+  } else if (map_it->second != list_.begin()) {
+    // Move the old entry to the front of the list.
+    list_.splice(list_.begin(), list_, map_it->second);
+  }
+
+  return list_.front().second;
+}
+
+void SourceTracker::PruneEntries(int64_t now_ms) const {
+  int64_t prune_ms = now_ms - kTimeoutMs;
+
+  while (!list_.empty() && list_.back().second.timestamp_ms < prune_ms) {
+    map_.erase(list_.back().first);
+    list_.pop_back();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker.h b/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker.h
new file mode 100644
index 0000000000..f9e8354d44
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker.h
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_SOURCE_TRACKER_H_
+#define MODULES_RTP_RTCP_SOURCE_SOURCE_TRACKER_H_
+
+#include <cstdint>
+#include <list>
+#include <unordered_map>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/rtp_packet_infos.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+//
+// Tracker for `RTCRtpContributingSource` and `RTCRtpSynchronizationSource`:
+//   - https://w3c.github.io/webrtc-pc/#dom-rtcrtpcontributingsource
+//   - https://w3c.github.io/webrtc-pc/#dom-rtcrtpsynchronizationsource
+//
+class SourceTracker {
+ public:
+  // Amount of time before the entry associated with an update is removed. See:
+  // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
+  static constexpr int64_t kTimeoutMs = 10000;  // 10 seconds
+
+  explicit SourceTracker(Clock* clock);
+
+  SourceTracker(const SourceTracker& other) = delete;
+  SourceTracker(SourceTracker&& other) = delete;
+  SourceTracker& operator=(const SourceTracker& other) = delete;
+  SourceTracker& operator=(SourceTracker&& other) = delete;
+
+  // Updates the source entries when a frame is delivered to the
+  // RTCRtpReceiver's MediaStreamTrack.
+  void OnFrameDelivered(const RtpPacketInfos& packet_infos);
+
+  // Returns an `RtpSource` for each unique SSRC and CSRC identifier updated in
+  // the last `kTimeoutMs` milliseconds. Entries appear in reverse chronological
+  // order (i.e. with the most recently updated entries appearing first).
+  std::vector<RtpSource> GetSources() const;
+
+ private:
+  struct SourceKey {
+    SourceKey(RtpSourceType source_type, uint32_t source)
+        : source_type(source_type), source(source) {}
+
+    // Type of `source`.
+    RtpSourceType source_type;
+
+    // CSRC or SSRC identifier of the contributing or synchronization source.
+    uint32_t source;
+  };
+
+  struct SourceKeyComparator {
+    bool operator()(const SourceKey& lhs, const SourceKey& rhs) const {
+      return (lhs.source_type == rhs.source_type) && (lhs.source == rhs.source);
+    }
+  };
+
+  struct SourceKeyHasher {
+    size_t operator()(const SourceKey& value) const {
+      return static_cast<size_t>(value.source_type) +
+             static_cast<size_t>(value.source) * 11076425802534262905ULL;
+    }
+  };
+
+  struct SourceEntry {
+    // Timestamp indicating the most recent time a frame from an RTP packet,
+    // originating from this source, was delivered to the RTCRtpReceiver's
+    // MediaStreamTrack. Its reference clock is the outer class's `clock_`.
+    int64_t timestamp_ms;
+
+    // Audio level from an RFC 6464 or RFC 6465 header extension received with
+    // the most recent packet used to assemble the frame associated with
+    // `timestamp_ms`. May be absent. Only relevant for audio receivers. See the
+    // specs for `RTCRtpContributingSource` for more info.
+    absl::optional<uint8_t> audio_level;
+
+    // Absolute capture time header extension received or interpolated from the
+    // most recent packet used to assemble the frame. For more info see
+    // https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
+    absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+
+    // Clock offset between the local clock and the capturer's clock.
+    // Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset`
+    // which instead represents the clock offset between a remote sender and the
+    // capturer. The following holds:
+    //   Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset
+    absl::optional<TimeDelta> local_capture_clock_offset;
+
+    // RTP timestamp of the most recent packet used to assemble the frame
+    // associated with `timestamp_ms`.
+    uint32_t rtp_timestamp;
+  };
+
+  using SourceList = std::list<std::pair<const SourceKey, SourceEntry>>;
+  using SourceMap = std::unordered_map<SourceKey,
+                                       SourceList::iterator,
+                                       SourceKeyHasher,
+                                       SourceKeyComparator>;
+
+  // Updates an entry by creating it (if it didn't previously exist) and moving
+  // it to the front of the list. Returns a reference to the entry.
+  SourceEntry& UpdateEntry(const SourceKey& key)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+
+  // Removes entries that have timed out. Marked as "const" so that we can do
+  // pruning in getters.
+  void PruneEntries(int64_t now_ms) const RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+
+  Clock* const clock_;
+  mutable Mutex lock_;
+
+  // Entries are stored in reverse chronological order (i.e. with the most
+  // recently updated entries appearing first). Mutability is needed for timeout
+  // pruning in const functions.
+  mutable SourceList list_ RTC_GUARDED_BY(lock_);
+  mutable SourceMap map_ RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_SOURCE_TRACKER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker_unittest.cc
new file mode 100644
index 0000000000..b3e3102d7e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/source_tracker_unittest.cc
@@ -0,0 +1,533 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/source_tracker.h"
+
+#include <algorithm>
+#include <list>
+#include <random>
+#include <set>
+#include <tuple>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/rtp_headers.h"
+#include "api/rtp_packet_info.h"
+#include "api/rtp_packet_infos.h"
+#include "system_wrappers/include/ntp_time.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Combine;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+using ::testing::TestWithParam;
+using ::testing::Values;
+
+constexpr size_t kPacketInfosCountMax = 5;
+
+// Simple "guaranteed to be correct" re-implementation of `SourceTracker` for
+// dual-implementation testing purposes.
+class ExpectedSourceTracker {
+ public:
+  explicit ExpectedSourceTracker(Clock* clock) : clock_(clock) {}
+
+  void OnFrameDelivered(const RtpPacketInfos& packet_infos) {
+    const int64_t now_ms = clock_->TimeInMilliseconds();
+
+    for (const auto& packet_info : packet_infos) {
+      RtpSource::Extensions extensions = {
+          packet_info.audio_level(), packet_info.absolute_capture_time(),
+          packet_info.local_capture_clock_offset()};
+
+      for (const auto& csrc : packet_info.csrcs()) {
+        entries_.emplace_front(now_ms, csrc, RtpSourceType::CSRC,
+                               packet_info.rtp_timestamp(), extensions);
+      }
+
+      entries_.emplace_front(now_ms, packet_info.ssrc(), RtpSourceType::SSRC,
+                             packet_info.rtp_timestamp(), extensions);
+    }
+
+    PruneEntries(now_ms);
+  }
+
+  std::vector<RtpSource> GetSources() const {
+    PruneEntries(clock_->TimeInMilliseconds());
+
+    return std::vector<RtpSource>(entries_.begin(), entries_.end());
+  }
+
+ private:
+  void PruneEntries(int64_t now_ms) const {
+    const int64_t prune_ms = now_ms - 10000;  // 10 seconds
+
+    std::set<std::pair<RtpSourceType, uint32_t>> seen;
+
+    auto it = entries_.begin();
+    auto end = entries_.end();
+    while (it != end) {
+      auto next = it;
+      ++next;
+
+      auto key = std::make_pair(it->source_type(), it->source_id());
+      if (!seen.insert(key).second || it->timestamp_ms() < prune_ms) {
+        entries_.erase(it);
+      }
+
+      it = next;
+    }
+  }
+
+  Clock* const clock_;
+
+  mutable std::list<RtpSource> entries_;
+};
+
+class SourceTrackerRandomTest
+    : public TestWithParam<std::tuple<uint32_t, uint32_t>> {
+ protected:
+  SourceTrackerRandomTest()
+      : ssrcs_count_(std::get<0>(GetParam())),
+        csrcs_count_(std::get<1>(GetParam())),
+        generator_(42) {}
+
+  RtpPacketInfos GeneratePacketInfos() {
+    size_t count = std::uniform_int_distribution<size_t>(
+        1, kPacketInfosCountMax)(generator_);
+
+    RtpPacketInfos::vector_type packet_infos;
+    for (size_t i = 0; i < count; ++i) {
+      packet_infos
+          .emplace_back(GenerateSsrc(), GenerateCsrcs(), GenerateRtpTimestamp(),
+                        GenerateReceiveTime())
+          .set_audio_level(GenerateAudioLevel())
+          .set_absolute_capture_time(GenerateAbsoluteCaptureTime())
+          .set_local_capture_clock_offset(GenerateLocalCaptureClockOffset());
+    }
+
+    return RtpPacketInfos(std::move(packet_infos));
+  }
+
+  int64_t GenerateClockAdvanceTimeMilliseconds() {
+    double roll = std::uniform_real_distribution<double>(0.0, 1.0)(generator_);
+
+    if (roll < 0.05) {
+      return 0;
+    }
+
+    if (roll < 0.08) {
+      return SourceTracker::kTimeoutMs - 1;
+    }
+
+    if (roll < 0.11) {
+      return SourceTracker::kTimeoutMs;
+    }
+
+    if (roll < 0.19) {
+      return std::uniform_int_distribution<int64_t>(
+          SourceTracker::kTimeoutMs,
+          SourceTracker::kTimeoutMs * 1000)(generator_);
+    }
+
+    return std::uniform_int_distribution<int64_t>(
+        1, SourceTracker::kTimeoutMs - 1)(generator_);
+  }
+
+ private:
+  uint32_t GenerateSsrc() {
+    return std::uniform_int_distribution<uint32_t>(1, ssrcs_count_)(generator_);
+  }
+
+  std::vector<uint32_t> GenerateCsrcs() {
+    std::vector<uint32_t> csrcs;
+    for (size_t i = 1; i <= csrcs_count_ && csrcs.size() < kRtpCsrcSize; ++i) {
+      if (std::bernoulli_distribution(0.5)(generator_)) {
+        csrcs.push_back(i);
+      }
+    }
+
+    return csrcs;
+  }
+
+  uint32_t GenerateRtpTimestamp() {
+    return std::uniform_int_distribution<uint32_t>()(generator_);
+  }
+
+  absl::optional<uint8_t> GenerateAudioLevel() {
+    if (std::bernoulli_distribution(0.25)(generator_)) {
+      return absl::nullopt;
+    }
+
+    // Workaround for std::uniform_int_distribution<uint8_t> not being allowed.
+    return static_cast<uint8_t>(
+        std::uniform_int_distribution<uint16_t>()(generator_));
+  }
+
+  absl::optional<AbsoluteCaptureTime> GenerateAbsoluteCaptureTime() {
+    if (std::bernoulli_distribution(0.25)(generator_)) {
+      return absl::nullopt;
+    }
+
+    AbsoluteCaptureTime value;
+
+    value.absolute_capture_timestamp =
+        std::uniform_int_distribution<uint64_t>()(generator_);
+
+    if (std::bernoulli_distribution(0.5)(generator_)) {
+      value.estimated_capture_clock_offset = absl::nullopt;
+    } else {
+      value.estimated_capture_clock_offset =
+          std::uniform_int_distribution<int64_t>()(generator_);
+    }
+
+    return value;
+  }
+
+  absl::optional<TimeDelta> GenerateLocalCaptureClockOffset() {
+    if (std::bernoulli_distribution(0.5)(generator_)) {
+      return absl::nullopt;
+    }
+    return TimeDelta::Millis(
+        UQ32x32ToInt64Ms(std::uniform_int_distribution<int64_t>()(generator_)));
+  }
+
+  Timestamp GenerateReceiveTime() {
+    return Timestamp::Micros(
+        std::uniform_int_distribution<int64_t>()(generator_));
+  }
+
+  const uint32_t ssrcs_count_;
+  const uint32_t csrcs_count_;
+
+  std::mt19937 generator_;
+};
+
+}  // namespace
+
+TEST_P(SourceTrackerRandomTest, RandomOperations) {
+  constexpr size_t kIterationsCount = 200;
+
+  SimulatedClock clock(1000000000000ULL);
+  SourceTracker actual_tracker(&clock);
+  ExpectedSourceTracker expected_tracker(&clock);
+
+  ASSERT_THAT(actual_tracker.GetSources(), IsEmpty());
+  ASSERT_THAT(expected_tracker.GetSources(), IsEmpty());
+
+  for (size_t i = 0; i < kIterationsCount; ++i) {
+    RtpPacketInfos packet_infos = GeneratePacketInfos();
+
+    actual_tracker.OnFrameDelivered(packet_infos);
+    expected_tracker.OnFrameDelivered(packet_infos);
+
+    clock.AdvanceTimeMilliseconds(GenerateClockAdvanceTimeMilliseconds());
+
+    ASSERT_THAT(actual_tracker.GetSources(),
+                ElementsAreArray(expected_tracker.GetSources()));
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(All,
+                         SourceTrackerRandomTest,
+                         Combine(/*ssrcs_count_=*/Values(1, 2, 4),
+                                 /*csrcs_count_=*/Values(0, 1, 3, 7)));
+
+TEST(SourceTrackerTest, StartEmpty) {
+  SimulatedClock clock(1000000000000ULL);
+  SourceTracker tracker(&clock);
+
+  EXPECT_THAT(tracker.GetSources(), IsEmpty());
+}
+
+TEST(SourceTrackerTest, OnFrameDeliveredRecordsSourcesDistinctSsrcs) {
+  constexpr uint32_t kSsrc1 = 10;
+  constexpr uint32_t kSsrc2 = 11;
+  constexpr uint32_t kCsrcs0 = 20;
+  constexpr uint32_t kCsrcs1 = 21;
+  constexpr uint32_t kCsrcs2 = 22;
+  constexpr uint32_t kRtpTimestamp0 = 40;
+  constexpr uint32_t kRtpTimestamp1 = 50;
+  constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
+  constexpr absl::optional<uint8_t> kAudioLevel1 = 20;
+  constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime =
+      AbsoluteCaptureTime{/*absolute_capture_timestamp=*/12,
+                          /*estimated_capture_clock_offset=*/absl::nullopt};
+  constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset = absl::nullopt;
+  constexpr Timestamp kReceiveTime0 = Timestamp::Millis(60);
+  constexpr Timestamp kReceiveTime1 = Timestamp::Millis(70);
+
+  SimulatedClock clock(1000000000000ULL);
+  SourceTracker tracker(&clock);
+
+  tracker.OnFrameDelivered(RtpPacketInfos(
+      {RtpPacketInfo(kSsrc1, {kCsrcs0, kCsrcs1}, kRtpTimestamp0, kReceiveTime0)
+           .set_audio_level(kAudioLevel0)
+           .set_absolute_capture_time(kAbsoluteCaptureTime)
+           .set_local_capture_clock_offset(kLocalCaptureClockOffset),
+       RtpPacketInfo(kSsrc2, {kCsrcs2}, kRtpTimestamp1, kReceiveTime1)
+           .set_audio_level(kAudioLevel1)
+           .set_absolute_capture_time(kAbsoluteCaptureTime)
+           .set_local_capture_clock_offset(kLocalCaptureClockOffset)}));
+
+  int64_t timestamp_ms = clock.TimeInMilliseconds();
+  constexpr RtpSource::Extensions extensions0 = {
+      .audio_level = kAudioLevel0,
+      .absolute_capture_time = kAbsoluteCaptureTime,
+      .local_capture_clock_offset = kLocalCaptureClockOffset};
+  constexpr RtpSource::Extensions extensions1 = {
+      .audio_level = kAudioLevel1,
+      .absolute_capture_time = kAbsoluteCaptureTime,
+      .local_capture_clock_offset = kLocalCaptureClockOffset};
+
+  EXPECT_THAT(tracker.GetSources(),
+              ElementsAre(RtpSource(timestamp_ms, kSsrc2, RtpSourceType::SSRC,
+                                    kRtpTimestamp1, extensions1),
+                          RtpSource(timestamp_ms, kCsrcs2, RtpSourceType::CSRC,
+                                    kRtpTimestamp1, extensions1),
+                          RtpSource(timestamp_ms, kSsrc1, RtpSourceType::SSRC,
+                                    kRtpTimestamp0, extensions0),
+                          RtpSource(timestamp_ms, kCsrcs1, RtpSourceType::CSRC,
+                                    kRtpTimestamp0, extensions0),
+                          RtpSource(timestamp_ms, kCsrcs0, RtpSourceType::CSRC,
+                                    kRtpTimestamp0, extensions0)));
+}
+
+TEST(SourceTrackerTest, OnFrameDeliveredRecordsSourcesSameSsrc) {
+  constexpr uint32_t kSsrc = 10;
+  constexpr uint32_t kCsrcs0 = 20;
+  constexpr uint32_t kCsrcs1 = 21;
+  constexpr uint32_t kCsrcs2 = 22;
+  constexpr uint32_t kRtpTimestamp0 = 40;
+  constexpr uint32_t kRtpTimestamp1 = 45;
+  constexpr uint32_t kRtpTimestamp2 = 50;
+  constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
+  constexpr absl::optional<uint8_t> kAudioLevel1 = 20;
+  constexpr absl::optional<uint8_t> kAudioLevel2 = 10;
+  constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime =
+      AbsoluteCaptureTime{/*absolute_capture_timestamp=*/12,
+                          /*estimated_capture_clock_offset=*/absl::nullopt};
+  constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset = absl::nullopt;
+  constexpr Timestamp kReceiveTime0 = Timestamp::Millis(60);
+  constexpr Timestamp kReceiveTime1 = Timestamp::Millis(70);
+  constexpr Timestamp kReceiveTime2 = Timestamp::Millis(80);
+
+  SimulatedClock clock(1000000000000ULL);
+  SourceTracker tracker(&clock);
+
+  tracker.OnFrameDelivered(RtpPacketInfos({
+      RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp0, kReceiveTime0)
+          .set_audio_level(kAudioLevel0)
+          .set_absolute_capture_time(kAbsoluteCaptureTime)
+          .set_local_capture_clock_offset(kLocalCaptureClockOffset),
+      RtpPacketInfo(kSsrc, {kCsrcs2}, kRtpTimestamp1, kReceiveTime1)
+          .set_audio_level(kAudioLevel1)
+          .set_absolute_capture_time(kAbsoluteCaptureTime)
+          .set_local_capture_clock_offset(kLocalCaptureClockOffset),
+      RtpPacketInfo(kSsrc, {kCsrcs0}, kRtpTimestamp2, kReceiveTime2)
+          .set_audio_level(kAudioLevel2)
+          .set_absolute_capture_time(kAbsoluteCaptureTime)
+          .set_local_capture_clock_offset(kLocalCaptureClockOffset),
+  }));
+
+  int64_t timestamp_ms = clock.TimeInMilliseconds();
+  constexpr RtpSource::Extensions extensions0 = {
+      .audio_level = kAudioLevel0,
+      .absolute_capture_time = kAbsoluteCaptureTime,
+      .local_capture_clock_offset = kLocalCaptureClockOffset};
+  constexpr RtpSource::Extensions extensions1 = {
+      .audio_level = kAudioLevel1,
+      .absolute_capture_time = kAbsoluteCaptureTime,
+      .local_capture_clock_offset = kLocalCaptureClockOffset};
+  constexpr RtpSource::Extensions extensions2 = {
+      .audio_level = kAudioLevel2,
+      .absolute_capture_time = kAbsoluteCaptureTime,
+      .local_capture_clock_offset = kLocalCaptureClockOffset};
+
+  EXPECT_THAT(tracker.GetSources(),
+              ElementsAre(RtpSource(timestamp_ms, kSsrc, RtpSourceType::SSRC,
+                                    kRtpTimestamp2, extensions2),
+                          RtpSource(timestamp_ms, kCsrcs0, RtpSourceType::CSRC,
+                                    kRtpTimestamp2, extensions2),
+                          RtpSource(timestamp_ms, kCsrcs2, RtpSourceType::CSRC,
+                                    kRtpTimestamp1, extensions1),
+                          RtpSource(timestamp_ms, kCsrcs1, RtpSourceType::CSRC,
+                                    kRtpTimestamp0, extensions0)));
+}
+
+TEST(SourceTrackerTest, OnFrameDeliveredUpdatesSources) {
+  constexpr uint32_t kSsrc1 = 10;
+  constexpr uint32_t kSsrc2 = 11;
+  constexpr uint32_t kCsrcs0 = 20;
+  constexpr uint32_t kCsrcs1 = 21;
+  constexpr uint32_t kCsrcs2 = 22;
+  constexpr uint32_t kRtpTimestamp0 = 40;
+  constexpr uint32_t kRtpTimestamp1 = 41;
+  constexpr uint32_t kRtpTimestamp2 = 42;
+  constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
+  constexpr absl::optional<uint8_t> kAudioLevel1 = absl::nullopt;
+  constexpr absl::optional<uint8_t> kAudioLevel2 = 10;
+  constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime0 =
+      AbsoluteCaptureTime{12, 34};
+  constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime1 =
+      AbsoluteCaptureTime{56, 78};
+  constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime2 =
+      AbsoluteCaptureTime{89, 90};
+  constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset0 =
+      TimeDelta::Millis(123);
+  constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset1 =
+      TimeDelta::Millis(456);
+  constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset2 =
+      TimeDelta::Millis(789);
+  constexpr Timestamp kReceiveTime0 = Timestamp::Millis(60);
+  constexpr Timestamp kReceiveTime1 = Timestamp::Millis(61);
+  constexpr Timestamp kReceiveTime2 = Timestamp::Millis(62);
+
+  constexpr RtpSource::Extensions extensions0 = {
+      .audio_level = kAudioLevel0,
+      .absolute_capture_time = kAbsoluteCaptureTime0,
+      .local_capture_clock_offset = kLocalCaptureClockOffset0};
+  constexpr RtpSource::Extensions extensions1 = {
+      .audio_level = kAudioLevel1,
+      .absolute_capture_time = kAbsoluteCaptureTime1,
+      .local_capture_clock_offset = kLocalCaptureClockOffset1};
+  constexpr RtpSource::Extensions extensions2 = {
+      .audio_level = kAudioLevel2,
+      .absolute_capture_time = kAbsoluteCaptureTime2,
+      .local_capture_clock_offset = kLocalCaptureClockOffset2};
+
+  SimulatedClock clock(1000000000000ULL);
+  SourceTracker tracker(&clock);
+
+  tracker.OnFrameDelivered(RtpPacketInfos(
+      {RtpPacketInfo(kSsrc1, {kCsrcs0, kCsrcs1}, kRtpTimestamp0, kReceiveTime0)
+           .set_audio_level(kAudioLevel0)
+           .set_absolute_capture_time(kAbsoluteCaptureTime0)
+           .set_local_capture_clock_offset(kLocalCaptureClockOffset0)}));
+
+  int64_t timestamp_ms_0 = clock.TimeInMilliseconds();
+  EXPECT_THAT(
+      tracker.GetSources(),
+      ElementsAre(RtpSource(timestamp_ms_0, kSsrc1, RtpSourceType::SSRC,
+                            kRtpTimestamp0, extensions0),
+                  RtpSource(timestamp_ms_0, kCsrcs1, RtpSourceType::CSRC,
+                            kRtpTimestamp0, extensions0),
+                  RtpSource(timestamp_ms_0, kCsrcs0, RtpSourceType::CSRC,
+                            kRtpTimestamp0, extensions0)));
+
+  // Deliver packets with updated sources.
+
+  clock.AdvanceTimeMilliseconds(17);
+  tracker.OnFrameDelivered(RtpPacketInfos(
+      {RtpPacketInfo(kSsrc1, {kCsrcs0, kCsrcs2}, kRtpTimestamp1, kReceiveTime1)
+           .set_audio_level(kAudioLevel1)
+           .set_absolute_capture_time(kAbsoluteCaptureTime1)
+           .set_local_capture_clock_offset(kLocalCaptureClockOffset1)}));
+
+  int64_t timestamp_ms_1 = clock.TimeInMilliseconds();
+
+  EXPECT_THAT(
+      tracker.GetSources(),
+      ElementsAre(RtpSource(timestamp_ms_1, kSsrc1, RtpSourceType::SSRC,
+                            kRtpTimestamp1, extensions1),
+                  RtpSource(timestamp_ms_1, kCsrcs2, RtpSourceType::CSRC,
+                            kRtpTimestamp1, extensions1),
+                  RtpSource(timestamp_ms_1, kCsrcs0, RtpSourceType::CSRC,
+                            kRtpTimestamp1, extensions1),
+                  RtpSource(timestamp_ms_0, kCsrcs1, RtpSourceType::CSRC,
+                            kRtpTimestamp0, extensions0)));
+
+  // Deliver more packets with update csrcs and a new ssrc.
+  clock.AdvanceTimeMilliseconds(17);
+
+  tracker.OnFrameDelivered(RtpPacketInfos(
+      {RtpPacketInfo(kSsrc2, {kCsrcs0}, kRtpTimestamp2, kReceiveTime2)
+           .set_audio_level(kAudioLevel2)
+           .set_absolute_capture_time(kAbsoluteCaptureTime2)
+           .set_local_capture_clock_offset(kLocalCaptureClockOffset2)}));
+
+  int64_t timestamp_ms_2 = clock.TimeInMilliseconds();
+
+  EXPECT_THAT(
+      tracker.GetSources(),
+      ElementsAre(RtpSource(timestamp_ms_2, kSsrc2, RtpSourceType::SSRC,
+                            kRtpTimestamp2, extensions2),
+                  RtpSource(timestamp_ms_2, kCsrcs0, RtpSourceType::CSRC,
+                            kRtpTimestamp2, extensions2),
+                  RtpSource(timestamp_ms_1, kSsrc1, RtpSourceType::SSRC,
+                            kRtpTimestamp1, extensions1),
+                  RtpSource(timestamp_ms_1, kCsrcs2, RtpSourceType::CSRC,
+                            kRtpTimestamp1, extensions1),
+                  RtpSource(timestamp_ms_0, kCsrcs1, RtpSourceType::CSRC,
+                            kRtpTimestamp0, extensions0)));
+}
+
+TEST(SourceTrackerTest, TimedOutSourcesAreRemoved) {
+  constexpr uint32_t kSsrc = 10;
+  constexpr uint32_t kCsrcs0 = 20;
+  constexpr uint32_t kCsrcs1 = 21;
+  constexpr uint32_t kCsrcs2 = 22;
+  constexpr uint32_t kRtpTimestamp0 = 40;
+  constexpr uint32_t kRtpTimestamp1 = 41;
+  constexpr absl::optional<uint8_t> kAudioLevel0 = 50;
+  constexpr absl::optional<uint8_t> kAudioLevel1 = absl::nullopt;
+  constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime0 =
+      AbsoluteCaptureTime{12, 34};
+  constexpr absl::optional<AbsoluteCaptureTime> kAbsoluteCaptureTime1 =
+      AbsoluteCaptureTime{56, 78};
+  constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset0 =
+      TimeDelta::Millis(123);
+  constexpr absl::optional<TimeDelta> kLocalCaptureClockOffset1 =
+      TimeDelta::Millis(456);
+  constexpr Timestamp kReceiveTime0 = Timestamp::Millis(60);
+  constexpr Timestamp kReceiveTime1 = Timestamp::Millis(61);
+
+  SimulatedClock clock(1000000000000ULL);
+  SourceTracker tracker(&clock);
+
+  tracker.OnFrameDelivered(RtpPacketInfos(
+      {RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs1}, kRtpTimestamp0, kReceiveTime0)
+           .set_audio_level(kAudioLevel0)
+           .set_absolute_capture_time(kAbsoluteCaptureTime0)
+           .set_local_capture_clock_offset(kLocalCaptureClockOffset0)}));
+
+  clock.AdvanceTimeMilliseconds(17);
+
+  tracker.OnFrameDelivered(RtpPacketInfos(
+      {RtpPacketInfo(kSsrc, {kCsrcs0, kCsrcs2}, kRtpTimestamp1, kReceiveTime1)
+           .set_audio_level(kAudioLevel1)
+           .set_absolute_capture_time(kAbsoluteCaptureTime1)
+           .set_local_capture_clock_offset(kLocalCaptureClockOffset1)}));
+
+  int64_t timestamp_ms_1 = clock.TimeInMilliseconds();
+
+  clock.AdvanceTimeMilliseconds(SourceTracker::kTimeoutMs);
+
+  constexpr RtpSource::Extensions extensions1 = {
+      .audio_level = kAudioLevel1,
+      .absolute_capture_time = kAbsoluteCaptureTime1,
+      .local_capture_clock_offset = kLocalCaptureClockOffset1};
+
+  EXPECT_THAT(
+      tracker.GetSources(),
+      ElementsAre(RtpSource(timestamp_ms_1, kSsrc, RtpSourceType::SSRC,
+                            kRtpTimestamp1, extensions1),
+                  RtpSource(timestamp_ms_1, kCsrcs2, RtpSourceType::CSRC,
+                            kRtpTimestamp1, extensions1),
+                  RtpSource(timestamp_ms_1, kCsrcs0, RtpSourceType::CSRC,
+                            kRtpTimestamp1, extensions1)));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/time_util.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/time_util.cc
new file mode 100644
index 0000000000..44ca07dabe
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/time_util.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/time_util.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/divide_round.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+uint32_t SaturatedToCompactNtp(TimeDelta delta) {
+  constexpr uint32_t kMaxCompactNtp = 0xFFFFFFFF;
+  constexpr int kCompactNtpInSecond = 0x10000;
+  if (delta <= TimeDelta::Zero())
+    return 0;
+  if (delta.us() >=
+      kMaxCompactNtp * rtc::kNumMicrosecsPerSec / kCompactNtpInSecond)
+    return kMaxCompactNtp;
+  // To convert to compact ntp need to divide by 1e6 to get seconds,
+  // then multiply by 0x10000 to get the final result.
+  // To avoid float operations, multiplication and division swapped.
+  return DivideRoundToNearest(delta.us() * kCompactNtpInSecond,
+                              rtc::kNumMicrosecsPerSec);
+}
+
+TimeDelta CompactNtpRttToTimeDelta(uint32_t compact_ntp_interval) {
+  static constexpr TimeDelta kMinRtt = TimeDelta::Millis(1);
+  // Interval to convert expected to be positive, e.g. RTT or delay.
+  // Because interval can be derived from non-monotonic ntp clock,
+  // it might become negative that is indistinguishable from very large values.
+  // Since very large RTT/delay is less likely than non-monotonic ntp clock,
+  // such value is considered negative and converted to minimum value of 1ms.
+  if (compact_ntp_interval > 0x80000000)
+    return kMinRtt;
+  // Convert to 64bit value to avoid multiplication overflow.
+  int64_t value = static_cast<int64_t>(compact_ntp_interval);
+  // To convert to TimeDelta need to divide by 2^16 to get seconds,
+  // then multiply by 1'000'000 to get microseconds. To avoid float operations,
+  // multiplication and division are swapped.
+  int64_t us = DivideRoundToNearest(value * rtc::kNumMicrosecsPerSec, 1 << 16);
+  // Small RTT value is considered too good to be true and increased to 1ms.
+  return std::max(TimeDelta::Micros(us), kMinRtt);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/time_util.h b/third_party/libwebrtc/modules/rtp_rtcp/source/time_util.h
new file mode 100644
index 0000000000..9ff444b12e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/time_util.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
+#define MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
+
+#include <stdint.h>
+
+#include "api/units/time_delta.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+
+// Helper function for compact ntp representation:
+// RFC 3550, Section 4. Time Format.
+// Wallclock time is represented using the timestamp format of
+// the Network Time Protocol (NTP).
+// ...
+// In some fields where a more compact representation is
+// appropriate, only the middle 32 bits are used; that is, the low 16
+// bits of the integer part and the high 16 bits of the fractional part.
+inline uint32_t CompactNtp(NtpTime ntp) {
+  return (ntp.seconds() << 16) | (ntp.fractions() >> 16);
+}
+
+// Converts interval to compact ntp (1/2^16 seconds) resolution.
+// Negative values converted to 0, Overlarge values converted to max uint32_t.
+uint32_t SaturatedToCompactNtp(TimeDelta delta);
+
+// Convert interval to the NTP time resolution (1/2^32 seconds ~= 0.2 ns).
+// For deltas with absolute value larger than 35 minutes result is unspecified.
+inline constexpr int64_t ToNtpUnits(TimeDelta delta) {
+  // For better precision `delta` is taken with best TimeDelta precision (us),
+  // then multiplaction and conversion to seconds are swapped to avoid float
+  // arithmetic.
+  // 2^31 us ~= 35.8 minutes.
+  return (rtc::saturated_cast<int32_t>(delta.us()) * (int64_t{1} << 32)) /
+         1'000'000;
+}
+
+// Converts interval from compact ntp (1/2^16 seconds) resolution to TimeDelta.
+// This interval can be up to ~9.1 hours (2^15 seconds).
+// Values close to 2^16 seconds are considered negative and are converted to
+// minimum value of 1ms.
+TimeDelta CompactNtpRttToTimeDelta(uint32_t compact_ntp_interval);
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_TIME_UTIL_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/time_util_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/time_util_unittest.cc
new file mode 100644
index 0000000000..b3d557fd83
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/time_util_unittest.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/time_util.h"
+
+#include <cstdint>
+#include <limits>
+
+#include "api/units/time_delta.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(TimeUtilTest, CompactNtp) {
+  const uint32_t kNtpSec = 0x12345678;
+  const uint32_t kNtpFrac = 0x23456789;
+  const NtpTime kNtp(kNtpSec, kNtpFrac);
+  const uint32_t kNtpMid = 0x56782345;
+  EXPECT_EQ(kNtpMid, CompactNtp(kNtp));
+}
+
+TEST(TimeUtilTest, CompactNtpRttToTimeDelta) {
+  const NtpTime ntp1(0x12345, 0x23456);
+  const NtpTime ntp2(0x12654, 0x64335);
+  int64_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+
+  EXPECT_NEAR(CompactNtpRttToTimeDelta(ntp_diff).ms(), ms_diff, 1);
+}
+
+TEST(TimeUtilTest, CompactNtpRttToTimeDeltaWithWrap) {
+  const NtpTime ntp1(0x1ffff, 0x23456);
+  const NtpTime ntp2(0x20000, 0x64335);
+  int64_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+
+  // While ntp2 > ntp1, there compact ntp presentation happen to be opposite.
+  // That shouldn't be a problem as long as unsigned arithmetic is used.
+  ASSERT_GT(ntp2.ToMs(), ntp1.ToMs());
+  ASSERT_LT(CompactNtp(ntp2), CompactNtp(ntp1));
+
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+  EXPECT_NEAR(CompactNtpRttToTimeDelta(ntp_diff).ms(), ms_diff, 1);
+}
+
+TEST(TimeUtilTest, CompactNtpRttToTimeDeltaLarge) {
+  const NtpTime ntp1(0x10000, 0x00006);
+  const NtpTime ntp2(0x17fff, 0xffff5);
+  int64_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+  // Ntp difference close to 2^15 seconds should convert correctly too.
+  ASSERT_NEAR(ms_diff, ((1 << 15) - 1) * 1000, 1);
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+  EXPECT_NEAR(CompactNtpRttToTimeDelta(ntp_diff).ms(), ms_diff, 1);
+}
+
+TEST(TimeUtilTest, CompactNtpRttToTimeDeltaNegative) {
+  const NtpTime ntp1(0x20000, 0x23456);
+  const NtpTime ntp2(0x1ffff, 0x64335);
+  int64_t ms_diff = ntp2.ToMs() - ntp1.ToMs();
+  ASSERT_GT(0, ms_diff);
+  // Ntp difference close to 2^16 seconds should be treated as negative.
+  uint32_t ntp_diff = CompactNtp(ntp2) - CompactNtp(ntp1);
+  EXPECT_EQ(CompactNtpRttToTimeDelta(ntp_diff), TimeDelta::Millis(1));
+}
+
+TEST(TimeUtilTest, SaturatedToCompactNtp) {
+  // Converts negative to zero.
+  EXPECT_EQ(SaturatedToCompactNtp(TimeDelta::Micros(-1)), 0u);
+  EXPECT_EQ(SaturatedToCompactNtp(TimeDelta::Zero()), 0u);
+  // Converts values just above and just below max uint32_t.
+  EXPECT_EQ(SaturatedToCompactNtp(TimeDelta::Micros(65536000000)), 0xffffffff);
+  EXPECT_EQ(SaturatedToCompactNtp(TimeDelta::Micros(65535999985)), 0xffffffff);
+  EXPECT_EQ(SaturatedToCompactNtp(TimeDelta::Micros(65535999970)), 0xfffffffe);
+  // Converts half-seconds.
+  EXPECT_EQ(SaturatedToCompactNtp(TimeDelta::Millis(500)), 0x8000u);
+  EXPECT_EQ(SaturatedToCompactNtp(TimeDelta::Seconds(1)), 0x10000u);
+  EXPECT_EQ(SaturatedToCompactNtp(TimeDelta::Millis(1'500)), 0x18000u);
+  // Convert us -> compact_ntp -> TimeDelta. Compact ntp precision is ~15us.
+  EXPECT_NEAR(
+      CompactNtpRttToTimeDelta(SaturatedToCompactNtp(TimeDelta::Micros(1'516)))
+          .us(),
+      1'516, 16);
+  EXPECT_NEAR(
+      CompactNtpRttToTimeDelta(SaturatedToCompactNtp(TimeDelta::Millis(15)))
+          .us(),
+      15'000, 16);
+  EXPECT_NEAR(
+      CompactNtpRttToTimeDelta(SaturatedToCompactNtp(TimeDelta::Micros(5'485)))
+          .us(),
+      5'485, 16);
+  EXPECT_NEAR(
+      CompactNtpRttToTimeDelta(SaturatedToCompactNtp(TimeDelta::Micros(5'515)))
+          .us(),
+      5'515, 16);
+}
+
+TEST(TimeUtilTest, ToNtpUnits) {
+  EXPECT_EQ(ToNtpUnits(TimeDelta::Zero()), 0);
+  EXPECT_EQ(ToNtpUnits(TimeDelta::Seconds(1)), int64_t{1} << 32);
+  EXPECT_EQ(ToNtpUnits(TimeDelta::Seconds(-1)), -(int64_t{1} << 32));
+
+  EXPECT_EQ(ToNtpUnits(TimeDelta::Millis(500)), int64_t{1} << 31);
+  EXPECT_EQ(ToNtpUnits(TimeDelta::Millis(-1'500)), -(int64_t{3} << 31));
+
+  // Smallest TimeDelta that can be converted without precision loss.
+  EXPECT_EQ(ToNtpUnits(TimeDelta::Micros(15'625)), int64_t{1} << 26);
+
+  // 1 us ~= 4'294.97 NTP units. ToNtpUnits makes no rounding promises.
+  EXPECT_GE(ToNtpUnits(TimeDelta::Micros(1)), 4'294);
+  EXPECT_LE(ToNtpUnits(TimeDelta::Micros(1)), 4'295);
+
+  // Test near maximum and minimum supported values.
+  static constexpr int64_t k35MinutesInNtpUnits = int64_t{35 * 60} << 32;
+  EXPECT_EQ(ToNtpUnits(TimeDelta::Seconds(35 * 60)), k35MinutesInNtpUnits);
+  EXPECT_EQ(ToNtpUnits(TimeDelta::Seconds(-35 * 60)), -k35MinutesInNtpUnits);
+
+  // The result for too large or too small values is unspecified, but
+  // shouldn't cause integer overflow or other undefined behavior.
+  ToNtpUnits(TimeDelta::Micros(std::numeric_limits<int64_t>::max() - 1));
+  ToNtpUnits(TimeDelta::Micros(std::numeric_limits<int64_t>::min() + 1));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/tmmbr_help.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/tmmbr_help.cc
new file mode 100644
index 0000000000..569ed4d8e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/tmmbr_help.cc
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/tmmbr_help.h"
+
+#include <stddef.h>
+
+#include <limits>
+
+#include "absl/algorithm/container.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+std::vector<rtcp::TmmbItem> TMMBRHelp::FindBoundingSet(
+    std::vector<rtcp::TmmbItem> candidates) {
+  // Filter out candidates with 0 bitrate.
+  for (auto it = candidates.begin(); it != candidates.end();) {
+    if (!it->bitrate_bps())
+      it = candidates.erase(it);
+    else
+      ++it;
+  }
+
+  if (candidates.size() <= 1)
+    return candidates;
+
+  size_t num_candidates = candidates.size();
+
+  // 1. Sort by increasing packet overhead.
+  absl::c_sort(candidates,
+               [](const rtcp::TmmbItem& lhs, const rtcp::TmmbItem& rhs) {
+                 return lhs.packet_overhead() < rhs.packet_overhead();
+               });
+
+  // 2. For tuples with same overhead, keep the one with the lowest bitrate.
+  for (auto it = candidates.begin(); it != candidates.end();) {
+    RTC_DCHECK(it->bitrate_bps());
+    auto current_min = it;
+    auto next_it = it + 1;
+    // Use fact candidates are sorted by overhead, so candidates with same
+    // overhead are adjusted.
+    while (next_it != candidates.end() &&
+           next_it->packet_overhead() == current_min->packet_overhead()) {
+      if (next_it->bitrate_bps() < current_min->bitrate_bps()) {
+        current_min->set_bitrate_bps(0);
+        current_min = next_it;
+      } else {
+        next_it->set_bitrate_bps(0);
+      }
+      ++next_it;
+      --num_candidates;
+    }
+    it = next_it;
+  }
+
+  // 3. Select and remove tuple with lowest bitrate.
+  // (If more than 1, choose the one with highest overhead).
+  auto min_bitrate_it = candidates.end();
+  for (auto it = candidates.begin(); it != candidates.end(); ++it) {
+    if (it->bitrate_bps()) {
+      min_bitrate_it = it;
+      break;
+    }
+  }
+
+  for (auto it = min_bitrate_it; it != candidates.end(); ++it) {
+    if (it->bitrate_bps() &&
+        it->bitrate_bps() <= min_bitrate_it->bitrate_bps()) {
+      // Get min bitrate.
+      min_bitrate_it = it;
+    }
+  }
+
+  std::vector<rtcp::TmmbItem> bounding_set;
+  bounding_set.reserve(num_candidates);
+  std::vector<float> intersection(num_candidates);
+  std::vector<float> max_packet_rate(num_candidates);
+
+  // First member of selected list.
+  bounding_set.push_back(*min_bitrate_it);
+  intersection[0] = 0;
+  // Calculate its maximum packet rate (where its line crosses x-axis).
+  uint16_t packet_overhead = bounding_set.back().packet_overhead();
+  if (packet_overhead == 0) {
+    // Avoid division by zero.
+    max_packet_rate[0] = std::numeric_limits<float>::max();
+  } else {
+    max_packet_rate[0] =
+        bounding_set.back().bitrate_bps() / static_cast<float>(packet_overhead);
+  }
+  // Remove from candidate list.
+  min_bitrate_it->set_bitrate_bps(0);
+  --num_candidates;
+
+  // 4. Discard from candidate list all tuple with lower overhead
+  // (next tuple must be steeper).
+  for (auto it = candidates.begin(); it != candidates.end(); ++it) {
+    if (it->bitrate_bps() &&
+        it->packet_overhead() < bounding_set.front().packet_overhead()) {
+      it->set_bitrate_bps(0);
+      --num_candidates;
+    }
+  }
+
+  bool get_new_candidate = true;
+  rtcp::TmmbItem cur_candidate;
+  while (num_candidates > 0) {
+    if (get_new_candidate) {
+      // 5. Remove first remaining tuple from candidate list.
+      for (auto it = candidates.begin(); it != candidates.end(); ++it) {
+        if (it->bitrate_bps()) {
+          cur_candidate = *it;
+          it->set_bitrate_bps(0);
+          break;
+        }
+      }
+    }
+
+    // 6. Calculate packet rate and intersection of the current
+    // line with line of last tuple in selected list.
+    RTC_DCHECK_NE(cur_candidate.packet_overhead(),
+                  bounding_set.back().packet_overhead());
+    float packet_rate = static_cast<float>(cur_candidate.bitrate_bps() -
+                                           bounding_set.back().bitrate_bps()) /
+                        (cur_candidate.packet_overhead() -
+                         bounding_set.back().packet_overhead());
+
+    // 7. If the packet rate is equal or lower than intersection of
+    //    last tuple in selected list,
+    //    remove last tuple in selected list & go back to step 6.
+    if (packet_rate <= intersection[bounding_set.size() - 1]) {
+      // Remove last tuple and goto step 6.
+      bounding_set.pop_back();
+      get_new_candidate = false;
+    } else {
+      // 8. If packet rate is lower than maximum packet rate of
+      // last tuple in selected list, add current tuple to selected
+      // list.
+      if (packet_rate < max_packet_rate[bounding_set.size() - 1]) {
+        bounding_set.push_back(cur_candidate);
+        intersection[bounding_set.size() - 1] = packet_rate;
+        uint16_t packet_overhead = bounding_set.back().packet_overhead();
+        RTC_DCHECK_NE(packet_overhead, 0);
+        max_packet_rate[bounding_set.size() - 1] =
+            bounding_set.back().bitrate_bps() /
+            static_cast<float>(packet_overhead);
+      }
+      --num_candidates;
+      get_new_candidate = true;
+    }
+
+    // 9. Go back to step 5 if any tuple remains in candidate list.
+  }
+  RTC_DCHECK(!bounding_set.empty());
+  return bounding_set;
+}
+
+bool TMMBRHelp::IsOwner(const std::vector<rtcp::TmmbItem>& bounding,
+                        uint32_t ssrc) {
+  for (const rtcp::TmmbItem& item : bounding) {
+    if (item.ssrc() == ssrc) {
+      return true;
+    }
+  }
+  return false;
+}
+
+uint64_t TMMBRHelp::CalcMinBitrateBps(
+    const std::vector<rtcp::TmmbItem>& candidates) {
+  RTC_DCHECK(!candidates.empty());
+  uint64_t min_bitrate_bps = std::numeric_limits<uint64_t>::max();
+  for (const rtcp::TmmbItem& item : candidates)
+    if (item.bitrate_bps() < min_bitrate_bps)
+      min_bitrate_bps = item.bitrate_bps();
+  return min_bitrate_bps;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/tmmbr_help.h b/third_party/libwebrtc/modules/rtp_rtcp/source/tmmbr_help.h
new file mode 100644
index 0000000000..8c26b22eb7
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/tmmbr_help.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
+#define MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+
+namespace webrtc {
+
+class TMMBRHelp {
+ public:
+  static std::vector<rtcp::TmmbItem> FindBoundingSet(
+      std::vector<rtcp::TmmbItem> candidates);
+
+  static bool IsOwner(const std::vector<rtcp::TmmbItem>& bounding,
+                      uint32_t ssrc);
+
+  static uint64_t CalcMinBitrateBps(
+      const std::vector<rtcp::TmmbItem>& candidates);
+};
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
new file mode 100644
index 0000000000..20402fc4d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
@@ -0,0 +1,268 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/ulpfec_generator.h"
+
+#include <string.h>
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr size_t kRedForFecHeaderLength = 1;
+
+// This controls the maximum amount of excess overhead (actual - target)
+// allowed in order to trigger EncodeFec(), before `params_.max_fec_frames`
+// is reached. Overhead here is defined as relative to number of media packets.
+constexpr int kMaxExcessOverhead = 50;  // Q8.
+
+// This is the minimum number of media packets required (above some protection
+// level) in order to trigger EncodeFec(), before `params_.max_fec_frames` is
+// reached.
+constexpr size_t kMinMediaPackets = 4;
+
+// Threshold on the received FEC protection level, above which we enforce at
+// least `kMinMediaPackets` packets for the FEC code. Below this
+// threshold `kMinMediaPackets` is set to default value of 1.
+//
+// The range is between 0 and 255, where 255 corresponds to 100% overhead
+// (relative to the number of protected media packets).
+constexpr uint8_t kHighProtectionThreshold = 80;
+
+// This threshold is used to adapt the `kMinMediaPackets` threshold, based
+// on the average number of packets per frame seen so far. When there are few
+// packets per frame (as given by this threshold), at least
+// `kMinMediaPackets` + 1 packets are sent to the FEC code.
+constexpr float kMinMediaPacketsAdaptationThreshold = 2.0f;
+
+// At construction time, we don't know the SSRC that is used for the generated
+// FEC packets, but we still need to give it to the ForwardErrorCorrection ctor
+// to be used in the decoding.
+// TODO(brandtr): Get rid of this awkwardness by splitting
+// ForwardErrorCorrection in two objects -- one encoder and one decoder.
+constexpr uint32_t kUnknownSsrc = 0;
+
+}  // namespace
+
+UlpfecGenerator::Params::Params() = default;
+UlpfecGenerator::Params::Params(FecProtectionParams delta_params,
+                                FecProtectionParams keyframe_params)
+    : delta_params(delta_params), keyframe_params(keyframe_params) {}
+
+UlpfecGenerator::UlpfecGenerator(int red_payload_type,
+                                 int ulpfec_payload_type,
+                                 Clock* clock)
+    : red_payload_type_(red_payload_type),
+      ulpfec_payload_type_(ulpfec_payload_type),
+      clock_(clock),
+      fec_(ForwardErrorCorrection::CreateUlpfec(kUnknownSsrc)),
+      num_protected_frames_(0),
+      min_num_media_packets_(1),
+      media_contains_keyframe_(false),
+      fec_bitrate_(/*max_window_size_ms=*/1000, RateStatistics::kBpsScale) {}
+
+// Used by FlexFecSender, payload types are unused.
+UlpfecGenerator::UlpfecGenerator(std::unique_ptr<ForwardErrorCorrection> fec,
+                                 Clock* clock)
+    : red_payload_type_(0),
+      ulpfec_payload_type_(0),
+      clock_(clock),
+      fec_(std::move(fec)),
+      num_protected_frames_(0),
+      min_num_media_packets_(1),
+      media_contains_keyframe_(false),
+      fec_bitrate_(/*max_window_size_ms=*/1000, RateStatistics::kBpsScale) {}
+
+UlpfecGenerator::~UlpfecGenerator() = default;
+
+void UlpfecGenerator::SetProtectionParameters(
+    const FecProtectionParams& delta_params,
+    const FecProtectionParams& key_params) {
+  RTC_DCHECK_GE(delta_params.fec_rate, 0);
+  RTC_DCHECK_LE(delta_params.fec_rate, 255);
+  RTC_DCHECK_GE(key_params.fec_rate, 0);
+  RTC_DCHECK_LE(key_params.fec_rate, 255);
+  // Store the new params and apply them for the next set of FEC packets being
+  // produced.
+  MutexLock lock(&mutex_);
+  pending_params_.emplace(delta_params, key_params);
+}
+
+void UlpfecGenerator::AddPacketAndGenerateFec(const RtpPacketToSend& packet) {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  RTC_DCHECK(generated_fec_packets_.empty());
+
+  {
+    MutexLock lock(&mutex_);
+    if (pending_params_) {
+      current_params_ = *pending_params_;
+      pending_params_.reset();
+
+      if (CurrentParams().fec_rate > kHighProtectionThreshold) {
+        min_num_media_packets_ = kMinMediaPackets;
+      } else {
+        min_num_media_packets_ = 1;
+      }
+    }
+  }
+
+  if (packet.is_key_frame()) {
+    media_contains_keyframe_ = true;
+  }
+  const bool complete_frame = packet.Marker();
+  if (media_packets_.size() < kUlpfecMaxMediaPackets) {
+    // Our packet masks can only protect up to `kUlpfecMaxMediaPackets` packets.
+    auto fec_packet = std::make_unique<ForwardErrorCorrection::Packet>();
+    fec_packet->data = packet.Buffer();
+    media_packets_.push_back(std::move(fec_packet));
+
+    // Keep a copy of the last RTP packet, so we can copy the RTP header
+    // from it when creating newly generated ULPFEC+RED packets.
+    RTC_DCHECK_GE(packet.headers_size(), kRtpHeaderSize);
+    last_media_packet_ = packet;
+  }
+
+  if (complete_frame) {
+    ++num_protected_frames_;
+  }
+
+  auto params = CurrentParams();
+
+  // Produce FEC over at most `params_.max_fec_frames` frames, or as soon as:
+  // (1) the excess overhead (actual overhead - requested/target overhead) is
+  // less than `kMaxExcessOverhead`, and
+  // (2) at least `min_num_media_packets_` media packets is reached.
+  if (complete_frame &&
+      (num_protected_frames_ >= params.max_fec_frames ||
+       (ExcessOverheadBelowMax() && MinimumMediaPacketsReached()))) {
+    // We are not using Unequal Protection feature of the parity erasure code.
+    constexpr int kNumImportantPackets = 0;
+    constexpr bool kUseUnequalProtection = false;
+    fec_->EncodeFec(media_packets_, params.fec_rate, kNumImportantPackets,
+                    kUseUnequalProtection, params.fec_mask_type,
+                    &generated_fec_packets_);
+    if (generated_fec_packets_.empty()) {
+      ResetState();
+    }
+  }
+}
+
+bool UlpfecGenerator::ExcessOverheadBelowMax() const {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+
+  return ((Overhead() - CurrentParams().fec_rate) < kMaxExcessOverhead);
+}
+
+bool UlpfecGenerator::MinimumMediaPacketsReached() const {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  float average_num_packets_per_frame =
+      static_cast<float>(media_packets_.size()) / num_protected_frames_;
+  int num_media_packets = static_cast<int>(media_packets_.size());
+  if (average_num_packets_per_frame < kMinMediaPacketsAdaptationThreshold) {
+    return num_media_packets >= min_num_media_packets_;
+  } else {
+    // For larger rates (more packets/frame), increase the threshold.
+    // TODO(brandtr): Investigate what impact this adaptation has.
+    return num_media_packets >= min_num_media_packets_ + 1;
+  }
+}
+
+const FecProtectionParams& UlpfecGenerator::CurrentParams() const {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  return media_contains_keyframe_ ? current_params_.keyframe_params
+                                  : current_params_.delta_params;
+}
+
+size_t UlpfecGenerator::MaxPacketOverhead() const {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  return fec_->MaxPacketOverhead();
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> UlpfecGenerator::GetFecPackets() {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  if (generated_fec_packets_.empty()) {
+    return std::vector<std::unique_ptr<RtpPacketToSend>>();
+  }
+
+  // Wrap FEC packet (including FEC headers) in a RED packet. Since the
+  // FEC packets in `generated_fec_packets_` don't have RTP headers, we
+  // reuse the header from the last media packet.
+  RTC_CHECK(last_media_packet_.has_value());
+  last_media_packet_->SetPayloadSize(0);
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets;
+  fec_packets.reserve(generated_fec_packets_.size());
+
+  size_t total_fec_size_bytes = 0;
+  for (const auto* fec_packet : generated_fec_packets_) {
+    std::unique_ptr<RtpPacketToSend> red_packet =
+        std::make_unique<RtpPacketToSend>(*last_media_packet_);
+    red_packet->SetPayloadType(red_payload_type_);
+    red_packet->SetMarker(false);
+    uint8_t* payload_buffer = red_packet->SetPayloadSize(
+        kRedForFecHeaderLength + fec_packet->data.size());
+    // Primary RED header with F bit unset.
+    // See https://tools.ietf.org/html/rfc2198#section-3
+    payload_buffer[0] = ulpfec_payload_type_;  // RED header.
+    memcpy(&payload_buffer[1], fec_packet->data.data(),
+           fec_packet->data.size());
+    total_fec_size_bytes += red_packet->size();
+    red_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
+    red_packet->set_allow_retransmission(false);
+    red_packet->set_is_red(true);
+    red_packet->set_fec_protect_packet(false);
+    fec_packets.push_back(std::move(red_packet));
+  }
+
+  ResetState();
+
+  MutexLock lock(&mutex_);
+  fec_bitrate_.Update(total_fec_size_bytes, clock_->TimeInMilliseconds());
+
+  return fec_packets;
+}
+
+DataRate UlpfecGenerator::CurrentFecRate() const {
+  MutexLock lock(&mutex_);
+  return DataRate::BitsPerSec(
+      fec_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0));
+}
+
+int UlpfecGenerator::Overhead() const {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  RTC_DCHECK(!media_packets_.empty());
+  int num_fec_packets =
+      fec_->NumFecPackets(media_packets_.size(), CurrentParams().fec_rate);
+
+  // Return the overhead in Q8.
+  return (num_fec_packets << 8) / media_packets_.size();
+}
+
+void UlpfecGenerator::ResetState() {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  media_packets_.clear();
+  last_media_packet_.reset();
+  generated_fec_packets_.clear();
+  num_protected_frames_ = 0;
+  media_contains_keyframe_ = false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator.h b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator.h
new file mode 100644
index 0000000000..88a8b459e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator.h
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_ULPFEC_GENERATOR_H_
+#define MODULES_RTP_RTCP_SOURCE_ULPFEC_GENERATOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+#include <memory>
+#include <vector>
+
+#include "modules/include/module_fec_types.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/video_fec_generator.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class FlexfecSender;
+
+class UlpfecGenerator : public VideoFecGenerator {
+  friend class FlexfecSender;
+
+ public:
+  UlpfecGenerator(int red_payload_type, int ulpfec_payload_type, Clock* clock);
+  ~UlpfecGenerator();
+
+  FecType GetFecType() const override {
+    return VideoFecGenerator::FecType::kUlpFec;
+  }
+  absl::optional<uint32_t> FecSsrc() override { return absl::nullopt; }
+
+  void SetProtectionParameters(const FecProtectionParams& delta_params,
+                               const FecProtectionParams& key_params) override;
+
+  // Adds a media packet to the internal buffer. When enough media packets
+  // have been added, the FEC packets are generated and stored internally.
+  // These FEC packets are then obtained by calling GetFecPacketsAsRed().
+  void AddPacketAndGenerateFec(const RtpPacketToSend& packet) override;
+
+  // Returns the overhead, per packet, for FEC (and possibly RED).
+  size_t MaxPacketOverhead() const override;
+
+  std::vector<std::unique_ptr<RtpPacketToSend>> GetFecPackets() override;
+
+  // Current rate of FEC packets generated, including all RTP-level headers.
+  DataRate CurrentFecRate() const override;
+
+  absl::optional<RtpState> GetRtpState() override { return absl::nullopt; }
+
+  // Currently used protection params.
+  const FecProtectionParams& CurrentParams() const;
+
+ private:
+  struct Params {
+    Params();
+    Params(FecProtectionParams delta_params,
+           FecProtectionParams keyframe_params);
+
+    FecProtectionParams delta_params;
+    FecProtectionParams keyframe_params;
+  };
+
+  UlpfecGenerator(std::unique_ptr<ForwardErrorCorrection> fec, Clock* clock);
+
+  // Overhead is defined as relative to the number of media packets, and not
+  // relative to total number of packets. This definition is inherited from the
+  // protection factor produced by video_coding module and how the FEC
+  // generation is implemented.
+  int Overhead() const;
+
+  // Returns true if the excess overhead (actual - target) for the FEC is below
+  // the amount `kMaxExcessOverhead`. This effects the lower protection level
+  // cases and low number of media packets/frame. The target overhead is given
+  // by `params_.fec_rate`, and is only achievable in the limit of large number
+  // of media packets.
+  bool ExcessOverheadBelowMax() const;
+
+  // Returns true if the number of added media packets is at least
+  // `min_num_media_packets_`. This condition tries to capture the effect
+  // that, for the same amount of protection/overhead, longer codes
+  // (e.g. (2k,2m) vs (k,m)) are generally more effective at recovering losses.
+  bool MinimumMediaPacketsReached() const;
+
+  void ResetState();
+
+  const int red_payload_type_;
+  const int ulpfec_payload_type_;
+  Clock* const clock_;
+
+  rtc::RaceChecker race_checker_;
+  const std::unique_ptr<ForwardErrorCorrection> fec_
+      RTC_GUARDED_BY(race_checker_);
+  ForwardErrorCorrection::PacketList media_packets_
+      RTC_GUARDED_BY(race_checker_);
+  absl::optional<RtpPacketToSend> last_media_packet_
+      RTC_GUARDED_BY(race_checker_);
+  std::list<ForwardErrorCorrection::Packet*> generated_fec_packets_
+      RTC_GUARDED_BY(race_checker_);
+  int num_protected_frames_ RTC_GUARDED_BY(race_checker_);
+  int min_num_media_packets_ RTC_GUARDED_BY(race_checker_);
+  Params current_params_ RTC_GUARDED_BY(race_checker_);
+  bool media_contains_keyframe_ RTC_GUARDED_BY(race_checker_);
+
+  mutable Mutex mutex_;
+  absl::optional<Params> pending_params_ RTC_GUARDED_BY(mutex_);
+  RateStatistics fec_bitrate_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ULPFEC_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc
new file mode 100644
index 0000000000..18f5685791
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc
@@ -0,0 +1,273 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/ulpfec_generator.h"
+
+#include <list>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/fec_test_helper.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+using test::fec::AugmentedPacket;
+using test::fec::AugmentedPacketGenerator;
+
+constexpr int kFecPayloadType = 96;
+constexpr int kRedPayloadType = 97;
+constexpr uint32_t kMediaSsrc = 835424;
+}  // namespace
+
+void VerifyHeader(uint16_t seq_num,
+                  uint32_t timestamp,
+                  int red_payload_type,
+                  int fec_payload_type,
+                  bool marker_bit,
+                  const rtc::CopyOnWriteBuffer& data) {
+  // Marker bit not set.
+  EXPECT_EQ(marker_bit ? 0x80 : 0, data[1] & 0x80);
+  EXPECT_EQ(red_payload_type, data[1] & 0x7F);
+  EXPECT_EQ(seq_num, (data[2] << 8) + data[3]);
+  uint32_t parsed_timestamp =
+      (data[4] << 24) + (data[5] << 16) + (data[6] << 8) + data[7];
+  EXPECT_EQ(timestamp, parsed_timestamp);
+  EXPECT_EQ(static_cast<uint8_t>(fec_payload_type), data[kRtpHeaderSize]);
+}
+
+class UlpfecGeneratorTest : public ::testing::Test {
+ protected:
+  UlpfecGeneratorTest()
+      : fake_clock_(1),
+        ulpfec_generator_(kRedPayloadType, kFecPayloadType, &fake_clock_),
+        packet_generator_(kMediaSsrc) {}
+
+  SimulatedClock fake_clock_;
+  UlpfecGenerator ulpfec_generator_;
+  AugmentedPacketGenerator packet_generator_;
+};
+
+// Verifies bug found via fuzzing, where a gap in the packet sequence caused us
+// to move past the end of the current FEC packet mask byte without moving to
+// the next byte. That likely caused us to repeatedly read from the same byte,
+// and if that byte didn't protect packets we would generate empty FEC.
+TEST_F(UlpfecGeneratorTest, NoEmptyFecWithSeqNumGaps) {
+  struct Packet {
+    size_t header_size;
+    size_t payload_size;
+    uint16_t seq_num;
+    bool marker_bit;
+  };
+  std::vector<Packet> protected_packets;
+  protected_packets.push_back({15, 3, 41, 0});
+  protected_packets.push_back({14, 1, 43, 0});
+  protected_packets.push_back({19, 0, 48, 0});
+  protected_packets.push_back({19, 0, 50, 0});
+  protected_packets.push_back({14, 3, 51, 0});
+  protected_packets.push_back({13, 8, 52, 0});
+  protected_packets.push_back({19, 2, 53, 0});
+  protected_packets.push_back({12, 3, 54, 0});
+  protected_packets.push_back({21, 0, 55, 0});
+  protected_packets.push_back({13, 3, 57, 1});
+  FecProtectionParams params = {117, 3, kFecMaskBursty};
+  ulpfec_generator_.SetProtectionParameters(params, params);
+  for (Packet p : protected_packets) {
+    RtpPacketToSend packet(nullptr);
+    packet.SetMarker(p.marker_bit);
+    packet.AllocateExtension(RTPExtensionType::kRtpExtensionMid,
+                             p.header_size - packet.headers_size());
+    packet.SetSequenceNumber(p.seq_num);
+    packet.AllocatePayload(p.payload_size);
+    ulpfec_generator_.AddPacketAndGenerateFec(packet);
+
+    std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+        ulpfec_generator_.GetFecPackets();
+    if (!p.marker_bit) {
+      EXPECT_TRUE(fec_packets.empty());
+    } else {
+      EXPECT_FALSE(fec_packets.empty());
+    }
+  }
+}
+
+TEST_F(UlpfecGeneratorTest, OneFrameFec) {
+  // The number of media packets (`kNumPackets`), number of frames (one for
+  // this test), and the protection factor (|params->fec_rate|) are set to make
+  // sure the conditions for generating FEC are satisfied. This means:
+  // (1) protection factor is high enough so that actual overhead over 1 frame
+  // of packets is within `kMaxExcessOverhead`, and (2) the total number of
+  // media packets for 1 frame is at least `minimum_media_packets_fec_`.
+  constexpr size_t kNumPackets = 4;
+  FecProtectionParams params = {15, 3, kFecMaskRandom};
+  packet_generator_.NewFrame(kNumPackets);
+  // Expecting one FEC packet.
+  ulpfec_generator_.SetProtectionParameters(params, params);
+  uint32_t last_timestamp = 0;
+  for (size_t i = 0; i < kNumPackets; ++i) {
+    std::unique_ptr<AugmentedPacket> packet =
+        packet_generator_.NextPacket(i, 10);
+    RtpPacketToSend rtp_packet(nullptr);
+    EXPECT_TRUE(rtp_packet.Parse(packet->data.data(), packet->data.size()));
+    ulpfec_generator_.AddPacketAndGenerateFec(rtp_packet);
+    last_timestamp = packet->header.timestamp;
+  }
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+      ulpfec_generator_.GetFecPackets();
+  EXPECT_EQ(fec_packets.size(), 1u);
+  uint16_t seq_num = packet_generator_.NextPacketSeqNum();
+  fec_packets[0]->SetSequenceNumber(seq_num);
+  EXPECT_TRUE(ulpfec_generator_.GetFecPackets().empty());
+
+  EXPECT_EQ(fec_packets[0]->headers_size(), kRtpHeaderSize);
+
+  VerifyHeader(seq_num, last_timestamp, kRedPayloadType, kFecPayloadType, false,
+               fec_packets[0]->Buffer());
+}
+
+TEST_F(UlpfecGeneratorTest, TwoFrameFec) {
+  // The number of media packets/frame (`kNumPackets`), the number of frames
+  // (`kNumFrames`), and the protection factor (|params->fec_rate|) are set to
+  // make sure the conditions for generating FEC are satisfied. This means:
+  // (1) protection factor is high enough so that actual overhead over
+  // `kNumFrames` is within `kMaxExcessOverhead`, and (2) the total number of
+  // media packets for `kNumFrames` frames is at least
+  // `minimum_media_packets_fec_`.
+  constexpr size_t kNumPackets = 2;
+  constexpr size_t kNumFrames = 2;
+
+  FecProtectionParams params = {15, 3, kFecMaskRandom};
+  // Expecting one FEC packet.
+  ulpfec_generator_.SetProtectionParameters(params, params);
+  uint32_t last_timestamp = 0;
+  for (size_t i = 0; i < kNumFrames; ++i) {
+    packet_generator_.NewFrame(kNumPackets);
+    for (size_t j = 0; j < kNumPackets; ++j) {
+      std::unique_ptr<AugmentedPacket> packet =
+          packet_generator_.NextPacket(i * kNumPackets + j, 10);
+      RtpPacketToSend rtp_packet(nullptr);
+      EXPECT_TRUE(rtp_packet.Parse(packet->data.data(), packet->data.size()));
+      ulpfec_generator_.AddPacketAndGenerateFec(rtp_packet);
+      last_timestamp = packet->header.timestamp;
+    }
+  }
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+      ulpfec_generator_.GetFecPackets();
+  EXPECT_EQ(fec_packets.size(), 1u);
+  const uint16_t seq_num = packet_generator_.NextPacketSeqNum();
+  fec_packets[0]->SetSequenceNumber(seq_num);
+  VerifyHeader(seq_num, last_timestamp, kRedPayloadType, kFecPayloadType, false,
+               fec_packets[0]->Buffer());
+}
+
+TEST_F(UlpfecGeneratorTest, MixedMediaRtpHeaderLengths) {
+  constexpr size_t kShortRtpHeaderLength = 12;
+  constexpr size_t kLongRtpHeaderLength = 16;
+
+  // Only one frame required to generate FEC.
+  FecProtectionParams params = {127, 1, kFecMaskRandom};
+  ulpfec_generator_.SetProtectionParameters(params, params);
+
+  // Fill up internal buffer with media packets with short RTP header length.
+  packet_generator_.NewFrame(kUlpfecMaxMediaPackets + 1);
+  for (size_t i = 0; i < kUlpfecMaxMediaPackets; ++i) {
+    std::unique_ptr<AugmentedPacket> packet =
+        packet_generator_.NextPacket(i, 10);
+    RtpPacketToSend rtp_packet(nullptr);
+    EXPECT_TRUE(rtp_packet.Parse(packet->data.data(), packet->data.size()));
+    EXPECT_EQ(rtp_packet.headers_size(), kShortRtpHeaderLength);
+    ulpfec_generator_.AddPacketAndGenerateFec(rtp_packet);
+    EXPECT_TRUE(ulpfec_generator_.GetFecPackets().empty());
+  }
+
+  // Kick off FEC generation with media packet with long RTP header length.
+  // Since the internal buffer is full, this packet will not be protected.
+  std::unique_ptr<AugmentedPacket> packet =
+      packet_generator_.NextPacket(kUlpfecMaxMediaPackets, 10);
+  RtpPacketToSend rtp_packet(nullptr);
+  EXPECT_TRUE(rtp_packet.Parse(packet->data.data(), packet->data.size()));
+  EXPECT_TRUE(rtp_packet.SetPayloadSize(0) != nullptr);
+  const uint32_t csrcs[]{1};
+  rtp_packet.SetCsrcs(csrcs);
+
+  EXPECT_EQ(rtp_packet.headers_size(), kLongRtpHeaderLength);
+
+  ulpfec_generator_.AddPacketAndGenerateFec(rtp_packet);
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+      ulpfec_generator_.GetFecPackets();
+  EXPECT_FALSE(fec_packets.empty());
+
+  // Ensure that the RED header is placed correctly, i.e. the correct
+  // RTP header length was used in the RED packet creation.
+  uint16_t seq_num = packet_generator_.NextPacketSeqNum();
+  for (const auto& fec_packet : fec_packets) {
+    fec_packet->SetSequenceNumber(seq_num++);
+    EXPECT_EQ(kFecPayloadType, fec_packet->data()[kShortRtpHeaderLength]);
+  }
+}
+
+TEST_F(UlpfecGeneratorTest, UpdatesProtectionParameters) {
+  const FecProtectionParams kKeyFrameParams = {25, /*max_fec_frames=*/2,
+                                               kFecMaskRandom};
+  const FecProtectionParams kDeltaFrameParams = {25, /*max_fec_frames=*/5,
+                                                 kFecMaskRandom};
+
+  ulpfec_generator_.SetProtectionParameters(kDeltaFrameParams, kKeyFrameParams);
+
+  // No params applied yet.
+  EXPECT_EQ(ulpfec_generator_.CurrentParams().max_fec_frames, 0);
+
+  // Helper function to add a single-packet frame market as either key-frame
+  // or delta-frame.
+  auto add_frame = [&](bool is_keyframe) {
+    packet_generator_.NewFrame(1);
+    std::unique_ptr<AugmentedPacket> packet =
+        packet_generator_.NextPacket(0, 10);
+    RtpPacketToSend rtp_packet(nullptr);
+    EXPECT_TRUE(rtp_packet.Parse(packet->data.data(), packet->data.size()));
+    rtp_packet.set_is_key_frame(is_keyframe);
+    ulpfec_generator_.AddPacketAndGenerateFec(rtp_packet);
+  };
+
+  // Add key-frame, keyframe params should apply, no FEC generated yet.
+  add_frame(true);
+  EXPECT_EQ(ulpfec_generator_.CurrentParams().max_fec_frames, 2);
+  EXPECT_TRUE(ulpfec_generator_.GetFecPackets().empty());
+
+  // Add delta-frame, generated FEC packet. Params will not be updated until
+  // next added packet though.
+  add_frame(false);
+  EXPECT_EQ(ulpfec_generator_.CurrentParams().max_fec_frames, 2);
+  EXPECT_FALSE(ulpfec_generator_.GetFecPackets().empty());
+
+  // Add delta-frame, now params get updated.
+  add_frame(false);
+  EXPECT_EQ(ulpfec_generator_.CurrentParams().max_fec_frames, 5);
+  EXPECT_TRUE(ulpfec_generator_.GetFecPackets().empty());
+
+  // Add yet another delta-frame.
+  add_frame(false);
+  EXPECT_EQ(ulpfec_generator_.CurrentParams().max_fec_frames, 5);
+  EXPECT_TRUE(ulpfec_generator_.GetFecPackets().empty());
+
+  // Add key-frame, params immediately switch to key-frame ones. The two
+  // buffered frames plus the key-frame is protected and fec emitted,
+  // even though the frame count is technically over the keyframe frame count
+  // threshold.
+  add_frame(true);
+  EXPECT_EQ(ulpfec_generator_.CurrentParams().max_fec_frames, 2);
+  EXPECT_FALSE(ulpfec_generator_.GetFecPackets().empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
new file mode 100644
index 0000000000..8378a8f19f
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
+
+#include <string.h>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Maximum number of media packets that can be protected in one batch.
+constexpr size_t kMaxMediaPackets = 48;
+
+// Maximum number of media packets tracked by FEC decoder.
+// Maintain a sufficiently larger tracking window than `kMaxMediaPackets`
+// to account for packet reordering in pacer/ network.
+constexpr size_t kMaxTrackedMediaPackets = 4 * kMaxMediaPackets;
+
+// Maximum number of FEC packets stored inside ForwardErrorCorrection.
+constexpr size_t kMaxFecPackets = kMaxMediaPackets;
+
+// FEC Level 0 header size in bytes.
+constexpr size_t kFecLevel0HeaderSize = 10;
+
+// FEC Level 1 (ULP) header size in bytes (L bit is set).
+constexpr size_t kFecLevel1HeaderSizeLBitSet = 2 + kUlpfecPacketMaskSizeLBitSet;
+
+// FEC Level 1 (ULP) header size in bytes (L bit is cleared).
+constexpr size_t kFecLevel1HeaderSizeLBitClear =
+    2 + kUlpfecPacketMaskSizeLBitClear;
+
+constexpr size_t kPacketMaskOffset = kFecLevel0HeaderSize + 2;
+
+size_t UlpfecHeaderSize(size_t packet_mask_size) {
+  RTC_DCHECK_LE(packet_mask_size, kUlpfecPacketMaskSizeLBitSet);
+  if (packet_mask_size <= kUlpfecPacketMaskSizeLBitClear) {
+    return kFecLevel0HeaderSize + kFecLevel1HeaderSizeLBitClear;
+  } else {
+    return kFecLevel0HeaderSize + kFecLevel1HeaderSizeLBitSet;
+  }
+}
+
+}  // namespace
+
+UlpfecHeaderReader::UlpfecHeaderReader()
+    : FecHeaderReader(kMaxTrackedMediaPackets, kMaxFecPackets) {}
+
+UlpfecHeaderReader::~UlpfecHeaderReader() = default;
+
+bool UlpfecHeaderReader::ReadFecHeader(
+    ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const {
+  uint8_t* data = fec_packet->pkt->data.MutableData();
+  if (fec_packet->pkt->data.size() < kPacketMaskOffset) {
+    return false;  // Truncated packet.
+  }
+  bool l_bit = (data[0] & 0x40) != 0u;
+  size_t packet_mask_size =
+      l_bit ? kUlpfecPacketMaskSizeLBitSet : kUlpfecPacketMaskSizeLBitClear;
+  fec_packet->fec_header_size = UlpfecHeaderSize(packet_mask_size);
+  uint16_t seq_num_base = ByteReader<uint16_t>::ReadBigEndian(&data[2]);
+  fec_packet->protected_ssrc = fec_packet->ssrc;  // Due to RED.
+  fec_packet->seq_num_base = seq_num_base;
+  fec_packet->packet_mask_offset = kPacketMaskOffset;
+  fec_packet->packet_mask_size = packet_mask_size;
+  fec_packet->protection_length =
+      ByteReader<uint16_t>::ReadBigEndian(&data[10]);
+
+  // Store length recovery field in temporary location in header.
+  // This makes the header "compatible" with the corresponding
+  // FlexFEC location of the length recovery field, thus simplifying
+  // the XORing operations.
+  memcpy(&data[2], &data[8], 2);
+
+  return true;
+}
+
+UlpfecHeaderWriter::UlpfecHeaderWriter()
+    : FecHeaderWriter(kMaxMediaPackets,
+                      kMaxFecPackets,
+                      kFecLevel0HeaderSize + kFecLevel1HeaderSizeLBitSet) {}
+
+UlpfecHeaderWriter::~UlpfecHeaderWriter() = default;
+
+// TODO(brandtr): Consider updating this implementation (which actually
+// returns a bound on the sequence number spread), if logic is added to
+// UlpfecHeaderWriter::FinalizeFecHeader to truncate packet masks which end
+// in a string of zeroes. (Similar to how it is done in the FlexFEC case.)
+size_t UlpfecHeaderWriter::MinPacketMaskSize(const uint8_t* packet_mask,
+                                             size_t packet_mask_size) const {
+  return packet_mask_size;
+}
+
+size_t UlpfecHeaderWriter::FecHeaderSize(size_t packet_mask_size) const {
+  return UlpfecHeaderSize(packet_mask_size);
+}
+
+void UlpfecHeaderWriter::FinalizeFecHeader(
+    uint32_t /* media_ssrc */,
+    uint16_t seq_num_base,
+    const uint8_t* packet_mask,
+    size_t packet_mask_size,
+    ForwardErrorCorrection::Packet* fec_packet) const {
+  uint8_t* data = fec_packet->data.MutableData();
+  // Set E bit to zero.
+  data[0] &= 0x7f;
+  // Set L bit based on packet mask size. (Note that the packet mask
+  // can only take on two discrete values.)
+  bool l_bit = (packet_mask_size == kUlpfecPacketMaskSizeLBitSet);
+  if (l_bit) {
+    data[0] |= 0x40;  // Set the L bit.
+  } else {
+    RTC_DCHECK_EQ(packet_mask_size, kUlpfecPacketMaskSizeLBitClear);
+    data[0] &= 0xbf;  // Clear the L bit.
+  }
+  // Copy length recovery field from temporary location.
+  memcpy(&data[8], &data[2], 2);
+  // Write sequence number base.
+  ByteWriter<uint16_t>::WriteBigEndian(&data[2], seq_num_base);
+  // Protection length is set to entire packet. (This is not
+  // required in general.)
+  const size_t fec_header_size = FecHeaderSize(packet_mask_size);
+  ByteWriter<uint16_t>::WriteBigEndian(
+      &data[10], fec_packet->data.size() - fec_header_size);
+  // Copy the packet mask.
+  memcpy(&data[12], packet_mask, packet_mask_size);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
new file mode 100644
index 0000000000..a8bb737dbb
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_ULPFEC_HEADER_READER_WRITER_H_
+#define MODULES_RTP_RTCP_SOURCE_ULPFEC_HEADER_READER_WRITER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+
+namespace webrtc {
+
+// FEC Level 0 Header, 10 bytes.
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |E|L|P|X|  CC   |M| PT recovery |            SN base            |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |                          TS recovery                          |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |        length recovery        |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// FEC Level 1 Header, 4 bytes (L = 0) or 8 bytes (L = 1).
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |       Protection Length       |             mask              |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |              mask cont. (present only when L = 1)             |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+class UlpfecHeaderReader : public FecHeaderReader {
+ public:
+  UlpfecHeaderReader();
+  ~UlpfecHeaderReader() override;
+
+  bool ReadFecHeader(
+      ForwardErrorCorrection::ReceivedFecPacket* fec_packet) const override;
+};
+
+class UlpfecHeaderWriter : public FecHeaderWriter {
+ public:
+  UlpfecHeaderWriter();
+  ~UlpfecHeaderWriter() override;
+
+  size_t MinPacketMaskSize(const uint8_t* packet_mask,
+                           size_t packet_mask_size) const override;
+
+  size_t FecHeaderSize(size_t packet_mask_row_size) const override;
+
+  void FinalizeFecHeader(
+      uint32_t media_ssrc,  // Unused by ULPFEC.
+      uint16_t seq_num_base,
+      const uint8_t* packet_mask,
+      size_t packet_mask_size,
+      ForwardErrorCorrection::Packet* fec_packet) const override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ULPFEC_HEADER_READER_WRITER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer_unittest.cc
new file mode 100644
index 0000000000..a190a548e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer_unittest.cc
@@ -0,0 +1,245 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
+
+#include <string.h>
+
+#include <memory>
+#include <utility>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/random.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using Packet = ForwardErrorCorrection::Packet;
+using ReceivedFecPacket = ForwardErrorCorrection::ReceivedFecPacket;
+
+constexpr uint32_t kMediaSsrc = 1254983;
+constexpr uint16_t kMediaStartSeqNum = 825;
+constexpr size_t kMediaPacketLength = 1234;
+
+constexpr size_t kUlpfecHeaderSizeLBitClear = 14;
+constexpr size_t kUlpfecHeaderSizeLBitSet = 18;
+constexpr size_t kUlpfecPacketMaskOffset = 12;
+
+std::unique_ptr<uint8_t[]> GeneratePacketMask(size_t packet_mask_size,
+                                              uint64_t seed) {
+  Random random(seed);
+  std::unique_ptr<uint8_t[]> packet_mask(new uint8_t[packet_mask_size]);
+  for (size_t i = 0; i < packet_mask_size; ++i) {
+    packet_mask[i] = random.Rand<uint8_t>();
+  }
+  return packet_mask;
+}
+
+std::unique_ptr<Packet> WriteHeader(const uint8_t* packet_mask,
+                                    size_t packet_mask_size) {
+  UlpfecHeaderWriter writer;
+  std::unique_ptr<Packet> written_packet(new Packet());
+  written_packet->data.SetSize(kMediaPacketLength);
+  uint8_t* data = written_packet->data.MutableData();
+  for (size_t i = 0; i < written_packet->data.size(); ++i) {
+    data[i] = i;  // Actual content doesn't matter.
+  }
+  writer.FinalizeFecHeader(kMediaSsrc, kMediaStartSeqNum, packet_mask,
+                           packet_mask_size, written_packet.get());
+  return written_packet;
+}
+
+std::unique_ptr<ReceivedFecPacket> ReadHeader(const Packet& written_packet) {
+  UlpfecHeaderReader reader;
+  std::unique_ptr<ReceivedFecPacket> read_packet(new ReceivedFecPacket());
+  read_packet->ssrc = kMediaSsrc;
+  read_packet->pkt = rtc::scoped_refptr<Packet>(new Packet());
+  read_packet->pkt->data = written_packet.data;
+  EXPECT_TRUE(reader.ReadFecHeader(read_packet.get()));
+  return read_packet;
+}
+
+void VerifyHeaders(size_t expected_fec_header_size,
+                   const uint8_t* expected_packet_mask,
+                   size_t expected_packet_mask_size,
+                   const Packet& written_packet,
+                   const ReceivedFecPacket& read_packet) {
+  EXPECT_EQ(kMediaSsrc, read_packet.ssrc);
+  EXPECT_EQ(expected_fec_header_size, read_packet.fec_header_size);
+  EXPECT_EQ(kMediaSsrc, read_packet.protected_ssrc);
+  EXPECT_EQ(kMediaStartSeqNum, read_packet.seq_num_base);
+  EXPECT_EQ(kUlpfecPacketMaskOffset, read_packet.packet_mask_offset);
+  ASSERT_EQ(expected_packet_mask_size, read_packet.packet_mask_size);
+  EXPECT_EQ(written_packet.data.size() - expected_fec_header_size,
+            read_packet.protection_length);
+  EXPECT_EQ(0, memcmp(expected_packet_mask,
+                      read_packet.pkt->data.MutableData() +
+                          read_packet.packet_mask_offset,
+                      read_packet.packet_mask_size));
+  // Verify that the call to ReadFecHeader did not tamper with the payload.
+  EXPECT_EQ(0, memcmp(written_packet.data.data() + expected_fec_header_size,
+                      read_packet.pkt->data.cdata() + expected_fec_header_size,
+                      written_packet.data.size() - expected_fec_header_size));
+}
+
+}  // namespace
+
+TEST(UlpfecHeaderReaderTest, ReadsSmallHeader) {
+  const uint8_t packet[] = {
+      0x00, 0x12, 0xab, 0xcd,  // L bit clear, "random" payload type and SN base
+      0x12, 0x34, 0x56, 0x78,  // "random" TS recovery
+      0xab, 0xcd, 0x11, 0x22,  // "random" length recovery and protection length
+      0x33, 0x44,              // "random" packet mask
+      0x00, 0x00, 0x00, 0x00   // payload
+  };
+  const size_t packet_length = sizeof(packet);
+  ReceivedFecPacket read_packet;
+  read_packet.pkt = rtc::scoped_refptr<Packet>(new Packet());
+  read_packet.pkt->data.SetData(packet, packet_length);
+
+  UlpfecHeaderReader reader;
+  EXPECT_TRUE(reader.ReadFecHeader(&read_packet));
+
+  EXPECT_EQ(14U, read_packet.fec_header_size);
+  EXPECT_EQ(0xabcdU, read_packet.seq_num_base);
+  EXPECT_EQ(12U, read_packet.packet_mask_offset);
+  EXPECT_EQ(2U, read_packet.packet_mask_size);
+  EXPECT_EQ(0x1122U, read_packet.protection_length);
+}
+
+TEST(UlpfecHeaderReaderTest, ReadsLargeHeader) {
+  const uint8_t packet[] = {
+      0x40, 0x12, 0xab, 0xcd,  // L bit set, "random" payload type and SN base
+      0x12, 0x34, 0x56, 0x78,  // "random" TS recovery
+      0xab, 0xcd, 0x11, 0x22,  // "random" length recovery and protection length
+      0x33, 0x44, 0x55, 0x66,  // "random" packet mask
+      0x77, 0x88,              //
+      0x00, 0x00, 0x00, 0x00   // payload
+  };
+  const size_t packet_length = sizeof(packet);
+  ReceivedFecPacket read_packet;
+  read_packet.pkt = rtc::scoped_refptr<Packet>(new Packet());
+  read_packet.pkt->data.SetData(packet, packet_length);
+
+  UlpfecHeaderReader reader;
+  EXPECT_TRUE(reader.ReadFecHeader(&read_packet));
+
+  EXPECT_EQ(18U, read_packet.fec_header_size);
+  EXPECT_EQ(0xabcdU, read_packet.seq_num_base);
+  EXPECT_EQ(12U, read_packet.packet_mask_offset);
+  EXPECT_EQ(6U, read_packet.packet_mask_size);
+  EXPECT_EQ(0x1122U, read_packet.protection_length);
+}
+
+TEST(UlpfecHeaderWriterTest, FinalizesSmallHeader) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  Packet written_packet;
+  written_packet.data.SetSize(kMediaPacketLength);
+  uint8_t* data = written_packet.data.MutableData();
+  for (size_t i = 0; i < written_packet.data.size(); ++i) {
+    data[i] = i;
+  }
+
+  UlpfecHeaderWriter writer;
+  writer.FinalizeFecHeader(kMediaSsrc, kMediaStartSeqNum, packet_mask.get(),
+                           packet_mask_size, &written_packet);
+
+  const uint8_t* packet = written_packet.data.cdata();
+  EXPECT_EQ(0x00, packet[0] & 0x80);  // E bit.
+  EXPECT_EQ(0x00, packet[0] & 0x40);  // L bit.
+  EXPECT_EQ(kMediaStartSeqNum, ByteReader<uint16_t>::ReadBigEndian(packet + 2));
+  EXPECT_EQ(
+      static_cast<uint16_t>(kMediaPacketLength - kUlpfecHeaderSizeLBitClear),
+      ByteReader<uint16_t>::ReadBigEndian(packet + 10));
+  EXPECT_EQ(0, memcmp(packet + kUlpfecPacketMaskOffset, packet_mask.get(),
+                      packet_mask_size));
+}
+
+TEST(UlpfecHeaderWriterTest, FinalizesLargeHeader) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+  Packet written_packet;
+  written_packet.data.SetSize(kMediaPacketLength);
+  uint8_t* data = written_packet.data.MutableData();
+  for (size_t i = 0; i < written_packet.data.size(); ++i) {
+    data[i] = i;
+  }
+
+  UlpfecHeaderWriter writer;
+  writer.FinalizeFecHeader(kMediaSsrc, kMediaStartSeqNum, packet_mask.get(),
+                           packet_mask_size, &written_packet);
+
+  const uint8_t* packet = written_packet.data.cdata();
+  EXPECT_EQ(0x00, packet[0] & 0x80);  // E bit.
+  EXPECT_EQ(0x40, packet[0] & 0x40);  // L bit.
+  EXPECT_EQ(kMediaStartSeqNum, ByteReader<uint16_t>::ReadBigEndian(packet + 2));
+  EXPECT_EQ(
+      static_cast<uint16_t>(kMediaPacketLength - kUlpfecHeaderSizeLBitSet),
+      ByteReader<uint16_t>::ReadBigEndian(packet + 10));
+  EXPECT_EQ(0, memcmp(packet + kUlpfecPacketMaskOffset, packet_mask.get(),
+                      packet_mask_size));
+}
+
+TEST(UlpfecHeaderWriterTest, CalculateSmallHeaderSize) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+
+  UlpfecHeaderWriter writer;
+  size_t min_packet_mask_size =
+      writer.MinPacketMaskSize(packet_mask.get(), packet_mask_size);
+
+  EXPECT_EQ(kUlpfecPacketMaskSizeLBitClear, min_packet_mask_size);
+  EXPECT_EQ(kUlpfecHeaderSizeLBitClear,
+            writer.FecHeaderSize(min_packet_mask_size));
+}
+
+TEST(UlpfecHeaderWriterTest, CalculateLargeHeaderSize) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+
+  UlpfecHeaderWriter writer;
+  size_t min_packet_mask_size =
+      writer.MinPacketMaskSize(packet_mask.get(), packet_mask_size);
+
+  EXPECT_EQ(kUlpfecPacketMaskSizeLBitSet, min_packet_mask_size);
+  EXPECT_EQ(kUlpfecHeaderSizeLBitSet,
+            writer.FecHeaderSize(min_packet_mask_size));
+}
+
+TEST(UlpfecHeaderReaderWriterTest, WriteAndReadSmallHeader) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitClear;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+  auto read_packet = ReadHeader(*written_packet);
+
+  VerifyHeaders(kUlpfecHeaderSizeLBitClear, packet_mask.get(), packet_mask_size,
+                *written_packet, *read_packet);
+}
+
+TEST(UlpfecHeaderReaderWriterTest, WriteAndReadLargeHeader) {
+  const size_t packet_mask_size = kUlpfecPacketMaskSizeLBitSet;
+  auto packet_mask = GeneratePacketMask(packet_mask_size, 0xabcd);
+
+  auto written_packet = WriteHeader(packet_mask.get(), packet_mask_size);
+  auto read_packet = ReadHeader(*written_packet);
+
+  VerifyHeaders(kUlpfecHeaderSizeLBitSet, packet_mask.get(), packet_mask_size,
+                *written_packet, *read_packet);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver.cc
new file mode 100644
index 0000000000..4452bdbba5
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver.cc
@@ -0,0 +1,245 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/ulpfec_receiver.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+UlpfecReceiver::UlpfecReceiver(uint32_t ssrc,
+                               int ulpfec_payload_type,
+                               RecoveredPacketReceiver* callback,
+                               Clock* clock)
+    : ssrc_(ssrc),
+      ulpfec_payload_type_(ulpfec_payload_type),
+      clock_(clock),
+      recovered_packet_callback_(callback),
+      fec_(ForwardErrorCorrection::CreateUlpfec(ssrc_)) {
+  // TODO(tommi, brandtr): Once considerations for red have been split
+  // away from this implementation, we can require the ulpfec payload type
+  // to always be valid and use uint8 for storage (as is done elsewhere).
+  RTC_DCHECK_GE(ulpfec_payload_type_, -1);
+}
+
+UlpfecReceiver::~UlpfecReceiver() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  if (packet_counter_.first_packet_time != Timestamp::MinusInfinity()) {
+    const Timestamp now = clock_->CurrentTime();
+    TimeDelta elapsed = (now - packet_counter_.first_packet_time);
+    if (elapsed.seconds() >= metrics::kMinRunTimeInSeconds) {
+      if (packet_counter_.num_packets > 0) {
+        RTC_HISTOGRAM_PERCENTAGE(
+            "WebRTC.Video.ReceivedFecPacketsInPercent",
+            static_cast<int>(packet_counter_.num_fec_packets * 100 /
+                             packet_counter_.num_packets));
+      }
+      if (packet_counter_.num_fec_packets > 0) {
+        RTC_HISTOGRAM_PERCENTAGE(
+            "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
+            static_cast<int>(packet_counter_.num_recovered_packets * 100 /
+                             packet_counter_.num_fec_packets));
+      }
+      if (ulpfec_payload_type_ != -1) {
+        RTC_HISTOGRAM_COUNTS_10000(
+            "WebRTC.Video.FecBitrateReceivedInKbps",
+            static_cast<int>(packet_counter_.num_bytes * 8 / elapsed.seconds() /
+                             1000));
+      }
+    }
+  }
+
+  received_packets_.clear();
+  fec_->ResetState(&recovered_packets_);
+}
+
+FecPacketCounter UlpfecReceiver::GetPacketCounter() const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  return packet_counter_;
+}
+
+//     0                   1                    2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3  4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |F|   block PT  |  timestamp offset         |   block length    |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+//
+// RFC 2198          RTP Payload for Redundant Audio Data    September 1997
+//
+//    The bits in the header are specified as follows:
+//
+//    F: 1 bit First bit in header indicates whether another header block
+//        follows.  If 1 further header blocks follow, if 0 this is the
+//        last header block.
+//        If 0 there is only 1 byte RED header
+//
+//    block PT: 7 bits RTP payload type for this block.
+//
+//    timestamp offset:  14 bits Unsigned offset of timestamp of this block
+//        relative to timestamp given in RTP header.  The use of an unsigned
+//        offset implies that redundant data must be sent after the primary
+//        data, and is hence a time to be subtracted from the current
+//        timestamp to determine the timestamp of the data for which this
+//        block is the redundancy.
+//
+//    block length:  10 bits Length in bytes of the corresponding data
+//        block excluding header.
+
+bool UlpfecReceiver::AddReceivedRedPacket(const RtpPacketReceived& rtp_packet) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  // TODO(bugs.webrtc.org/11993): We get here via Call::DeliverRtp, so should be
+  // moved to the network thread.
+
+  if (rtp_packet.Ssrc() != ssrc_) {
+    RTC_LOG(LS_WARNING)
+        << "Received RED packet with different SSRC than expected; dropping.";
+    return false;
+  }
+  if (rtp_packet.size() > IP_PACKET_SIZE) {
+    RTC_LOG(LS_WARNING) << "Received RED packet with length exceeds maximum IP "
+                           "packet size; dropping.";
+    return false;
+  }
+
+  static constexpr uint8_t kRedHeaderLength = 1;
+
+  if (rtp_packet.payload_size() == 0) {
+    RTC_LOG(LS_WARNING) << "Corrupt/truncated FEC packet.";
+    return false;
+  }
+
+  // Remove RED header of incoming packet and store as a virtual RTP packet.
+  auto received_packet =
+      std::make_unique<ForwardErrorCorrection::ReceivedPacket>();
+  received_packet->pkt = new ForwardErrorCorrection::Packet();
+
+  // Get payload type from RED header and sequence number from RTP header.
+  uint8_t payload_type = rtp_packet.payload()[0] & 0x7f;
+  received_packet->is_fec = payload_type == ulpfec_payload_type_;
+  received_packet->is_recovered = rtp_packet.recovered();
+  received_packet->ssrc = rtp_packet.Ssrc();
+  received_packet->seq_num = rtp_packet.SequenceNumber();
+  received_packet->extensions = rtp_packet.extension_manager();
+
+  if (rtp_packet.payload()[0] & 0x80) {
+    // f bit set in RED header, i.e. there are more than one RED header blocks.
+    // WebRTC never generates multiple blocks in a RED packet for FEC.
+    RTC_LOG(LS_WARNING) << "More than 1 block in RED packet is not supported.";
+    return false;
+  }
+
+  ++packet_counter_.num_packets;
+  packet_counter_.num_bytes += rtp_packet.size();
+  if (packet_counter_.first_packet_time == Timestamp::MinusInfinity()) {
+    packet_counter_.first_packet_time = clock_->CurrentTime();
+  }
+
+  if (received_packet->is_fec) {
+    ++packet_counter_.num_fec_packets;
+    // everything behind the RED header
+    received_packet->pkt->data =
+        rtp_packet.Buffer().Slice(rtp_packet.headers_size() + kRedHeaderLength,
+                                  rtp_packet.payload_size() - kRedHeaderLength);
+  } else {
+    received_packet->pkt->data.EnsureCapacity(rtp_packet.size() -
+                                              kRedHeaderLength);
+    // Copy RTP header.
+    received_packet->pkt->data.SetData(rtp_packet.data(),
+                                       rtp_packet.headers_size());
+    // Set payload type.
+    uint8_t& payload_type_byte = received_packet->pkt->data.MutableData()[1];
+    payload_type_byte &= 0x80;          // Reset RED payload type.
+    payload_type_byte += payload_type;  // Set media payload type.
+    // Copy payload and padding data, after the RED header.
+    received_packet->pkt->data.AppendData(
+        rtp_packet.data() + rtp_packet.headers_size() + kRedHeaderLength,
+        rtp_packet.size() - rtp_packet.headers_size() - kRedHeaderLength);
+  }
+
+  if (received_packet->pkt->data.size() > 0) {
+    received_packets_.push_back(std::move(received_packet));
+  }
+  return true;
+}
+
+void UlpfecReceiver::ProcessReceivedFec() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // If we iterate over `received_packets_` and it contains a packet that cause
+  // us to recurse back to this function (for example a RED packet encapsulating
+  // a RED packet), then we will recurse forever. To avoid this we swap
+  // `received_packets_` with an empty vector so that the next recursive call
+  // wont iterate over the same packet again. This also solves the problem of
+  // not modifying the vector we are currently iterating over (packets are added
+  // in AddReceivedRedPacket).
+  std::vector<std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>>
+      received_packets;
+  received_packets.swap(received_packets_);
+  RtpHeaderExtensionMap* last_recovered_extension_map = nullptr;
+
+  for (const auto& received_packet : received_packets) {
+    // Send received media packet to VCM.
+    if (!received_packet->is_fec) {
+      ForwardErrorCorrection::Packet* packet = received_packet->pkt.get();
+
+      RtpPacketReceived rtp_packet(&received_packet->extensions);
+      if (!rtp_packet.Parse(std::move(packet->data))) {
+        RTC_LOG(LS_WARNING) << "Corrupted media packet";
+        continue;
+      }
+      recovered_packet_callback_->OnRecoveredPacket(rtp_packet);
+      // Some header extensions need to be zeroed in `received_packet` since
+      // they are written to the packet after FEC encoding. We try to do it
+      // without a copy of the underlying Copy-On-Write buffer, but if a
+      // reference is held by `recovered_packet_callback_->OnRecoveredPacket` a
+      // copy will still be made in 'rtp_packet.ZeroMutableExtensions()'.
+      rtp_packet.ZeroMutableExtensions();
+      packet->data = rtp_packet.Buffer();
+    }
+    if (!received_packet->is_recovered) {
+      // Do not pass recovered packets to FEC. Recovered packet might have
+      // different set of the RTP header extensions and thus different byte
+      // representation than the original packet, That will corrupt
+      // FEC calculation.
+      fec_->DecodeFec(*received_packet, &recovered_packets_);
+      last_recovered_extension_map = &received_packet->extensions;
+    }
+  }
+
+  // Send any recovered media packets to VCM.
+  for (const auto& recovered_packet : recovered_packets_) {
+    if (recovered_packet->returned) {
+      // Already sent to the VCM and the jitter buffer.
+      continue;
+    }
+    ForwardErrorCorrection::Packet* packet = recovered_packet->pkt.get();
+    ++packet_counter_.num_recovered_packets;
+    // Set this flag first; in case the recovered packet carries a RED
+    // header, OnRecoveredPacket will recurse back here.
+    recovered_packet->returned = true;
+    RtpPacketReceived parsed_packet(last_recovered_extension_map);
+    if (!parsed_packet.Parse(packet->data)) {
+      continue;
+    }
+    parsed_packet.set_recovered(true);
+    recovered_packet_callback_->OnRecoveredPacket(parsed_packet);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver.h b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver.h
new file mode 100644
index 0000000000..3ad1c75e09
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_ULPFEC_RECEIVER_H_
+#define MODULES_RTP_RTCP_SOURCE_ULPFEC_RECEIVER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "modules/rtp_rtcp/include/recovered_packet_receiver.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+struct FecPacketCounter {
+  FecPacketCounter() = default;
+  size_t num_packets = 0;  // Number of received packets.
+  size_t num_bytes = 0;
+  size_t num_fec_packets = 0;  // Number of received FEC packets.
+  size_t num_recovered_packets =
+      0;  // Number of recovered media packets using FEC.
+  // Time when first packet is received.
+  Timestamp first_packet_time = Timestamp::MinusInfinity();
+};
+
+class UlpfecReceiver {
+ public:
+  UlpfecReceiver(uint32_t ssrc,
+                 int ulpfec_payload_type,
+                 RecoveredPacketReceiver* callback,
+                 Clock* clock);
+  ~UlpfecReceiver();
+
+  int ulpfec_payload_type() const { return ulpfec_payload_type_; }
+
+  bool AddReceivedRedPacket(const RtpPacketReceived& rtp_packet);
+
+  void ProcessReceivedFec();
+
+  FecPacketCounter GetPacketCounter() const;
+
+  void SetRtpExtensions(rtc::ArrayView<const RtpExtension> extensions);
+
+ private:
+  const uint32_t ssrc_;
+  const int ulpfec_payload_type_;
+  Clock* const clock_;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+  RecoveredPacketReceiver* const recovered_packet_callback_;
+  const std::unique_ptr<ForwardErrorCorrection> fec_;
+  // TODO(nisse): The AddReceivedRedPacket method adds one or two packets to
+  // this list at a time, after which it is emptied by ProcessReceivedFec. It
+  // will make things simpler to merge AddReceivedRedPacket and
+  // ProcessReceivedFec into a single method, and we can then delete this list.
+  std::vector<std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>>
+      received_packets_ RTC_GUARDED_BY(&sequence_checker_);
+  ForwardErrorCorrection::RecoveredPacketList recovered_packets_
+      RTC_GUARDED_BY(&sequence_checker_);
+  FecPacketCounter packet_counter_ RTC_GUARDED_BY(&sequence_checker_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ULPFEC_RECEIVER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
new file mode 100644
index 0000000000..676e20c795
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
@@ -0,0 +1,543 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/ulpfec_receiver.h"
+
+#include <string.h>
+
+#include <list>
+#include <memory>
+#include <utility>
+
+#include "modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/fec_test_helper.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+using ::testing::_;
+using ::testing::Eq;
+using ::testing::Property;
+
+using test::fec::AugmentedPacket;
+using Packet = ForwardErrorCorrection::Packet;
+using test::fec::UlpfecPacketGenerator;
+
+constexpr int kFecPayloadType = 96;
+constexpr uint32_t kMediaSsrc = 835424;
+
+class NullRecoveredPacketReceiver : public RecoveredPacketReceiver {
+ public:
+  void OnRecoveredPacket(const RtpPacketReceived& packet) override {}
+};
+
+}  // namespace
+
+class UlpfecReceiverTest : public ::testing::Test {
+ protected:
+  UlpfecReceiverTest()
+      : fec_(ForwardErrorCorrection::CreateUlpfec(kMediaSsrc)),
+        receiver_fec_(kMediaSsrc,
+                      kFecPayloadType,
+                      &recovered_packet_receiver_,
+                      Clock::GetRealTimeClock()),
+        packet_generator_(kMediaSsrc) {}
+
+  // Generates `num_fec_packets` FEC packets, given `media_packets`.
+  void EncodeFec(const ForwardErrorCorrection::PacketList& media_packets,
+                 size_t num_fec_packets,
+                 std::list<ForwardErrorCorrection::Packet*>* fec_packets);
+
+  // Generates `num_media_packets` corresponding to a single frame.
+  void PacketizeFrame(size_t num_media_packets,
+                      size_t frame_offset,
+                      std::list<AugmentedPacket*>* augmented_packets,
+                      ForwardErrorCorrection::PacketList* packets);
+
+  // Build a media packet using `packet_generator_` and add it
+  // to the receiver.
+  void BuildAndAddRedMediaPacket(AugmentedPacket* packet,
+                                 bool is_recovered = false);
+
+  // Build a FEC packet using `packet_generator_` and add it
+  // to the receiver.
+  void BuildAndAddRedFecPacket(Packet* packet);
+
+  // Ensure that `recovered_packet_receiver_` will be called correctly
+  // and that the recovered packet will be identical to the lost packet.
+  void VerifyReconstructedMediaPacket(const AugmentedPacket& packet,
+                                      size_t times);
+
+  void InjectGarbagePacketLength(size_t fec_garbage_offset);
+
+  static void SurvivesMaliciousPacket(const uint8_t* data,
+                                      size_t length,
+                                      uint8_t ulpfec_payload_type);
+
+  MockRecoveredPacketReceiver recovered_packet_receiver_;
+  std::unique_ptr<ForwardErrorCorrection> fec_;
+  UlpfecReceiver receiver_fec_;
+  UlpfecPacketGenerator packet_generator_;
+};
+
+void UlpfecReceiverTest::EncodeFec(
+    const ForwardErrorCorrection::PacketList& media_packets,
+    size_t num_fec_packets,
+    std::list<ForwardErrorCorrection::Packet*>* fec_packets) {
+  const uint8_t protection_factor =
+      num_fec_packets * 255 / media_packets.size();
+  // Unequal protection is turned off, and the number of important
+  // packets is thus irrelevant.
+  constexpr int kNumImportantPackets = 0;
+  constexpr bool kUseUnequalProtection = false;
+  constexpr FecMaskType kFecMaskType = kFecMaskBursty;
+  EXPECT_EQ(
+      0, fec_->EncodeFec(media_packets, protection_factor, kNumImportantPackets,
+                         kUseUnequalProtection, kFecMaskType, fec_packets));
+  ASSERT_EQ(num_fec_packets, fec_packets->size());
+}
+
+void UlpfecReceiverTest::PacketizeFrame(
+    size_t num_media_packets,
+    size_t frame_offset,
+    std::list<AugmentedPacket*>* augmented_packets,
+    ForwardErrorCorrection::PacketList* packets) {
+  packet_generator_.NewFrame(num_media_packets);
+  for (size_t i = 0; i < num_media_packets; ++i) {
+    std::unique_ptr<AugmentedPacket> next_packet(
+        packet_generator_.NextPacket(frame_offset + i, kRtpHeaderSize + 10));
+    augmented_packets->push_back(next_packet.get());
+    packets->push_back(std::move(next_packet));
+  }
+}
+
+void UlpfecReceiverTest::BuildAndAddRedMediaPacket(AugmentedPacket* packet,
+                                                   bool is_recovered) {
+  RtpPacketReceived red_packet =
+      packet_generator_.BuildMediaRedPacket(*packet, is_recovered);
+  EXPECT_TRUE(receiver_fec_.AddReceivedRedPacket(red_packet));
+}
+
+void UlpfecReceiverTest::BuildAndAddRedFecPacket(Packet* packet) {
+  RtpPacketReceived red_packet =
+      packet_generator_.BuildUlpfecRedPacket(*packet);
+  EXPECT_TRUE(receiver_fec_.AddReceivedRedPacket(red_packet));
+}
+
+void UlpfecReceiverTest::VerifyReconstructedMediaPacket(
+    const AugmentedPacket& packet,
+    size_t times) {
+  // Verify that the content of the reconstructed packet is equal to the
+  // content of `packet`, and that the same content is received `times` number
+  // of times in a row.
+  EXPECT_CALL(
+      recovered_packet_receiver_,
+      OnRecoveredPacket(Property(&RtpPacketReceived::Buffer, Eq(packet.data))))
+      .Times(times);
+}
+
+void UlpfecReceiverTest::InjectGarbagePacketLength(size_t fec_garbage_offset) {
+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_));
+
+  const size_t kNumFecPackets = 1;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  PacketizeFrame(2, 0, &augmented_media_packets, &media_packets);
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets, kNumFecPackets, &fec_packets);
+  ByteWriter<uint16_t>::WriteBigEndian(
+      fec_packets.front()->data.MutableData() + fec_garbage_offset, 0x4711);
+
+  // Inject first media packet, then first FEC packet, skipping the second media
+  // packet to cause a recovery from the FEC packet.
+  BuildAndAddRedMediaPacket(augmented_media_packets.front());
+  BuildAndAddRedFecPacket(fec_packets.front());
+  receiver_fec_.ProcessReceivedFec();
+
+  FecPacketCounter counter = receiver_fec_.GetPacketCounter();
+  EXPECT_EQ(2U, counter.num_packets);
+  EXPECT_EQ(1U, counter.num_fec_packets);
+  EXPECT_EQ(0U, counter.num_recovered_packets);
+}
+
+void UlpfecReceiverTest::SurvivesMaliciousPacket(const uint8_t* data,
+                                                 size_t length,
+                                                 uint8_t ulpfec_payload_type) {
+  NullRecoveredPacketReceiver null_callback;
+  UlpfecReceiver receiver_fec(kMediaSsrc, ulpfec_payload_type, &null_callback,
+                              Clock::GetRealTimeClock());
+
+  RtpPacketReceived rtp_packet;
+  ASSERT_TRUE(rtp_packet.Parse(data, length));
+  receiver_fec.AddReceivedRedPacket(rtp_packet);
+}
+
+TEST_F(UlpfecReceiverTest, TwoMediaOneFec) {
+  constexpr size_t kNumFecPackets = 1u;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  PacketizeFrame(2, 0, &augmented_media_packets, &media_packets);
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets, kNumFecPackets, &fec_packets);
+
+  FecPacketCounter counter = receiver_fec_.GetPacketCounter();
+  EXPECT_EQ(0u, counter.num_packets);
+  EXPECT_EQ(Timestamp::MinusInfinity(), counter.first_packet_time);
+
+  // Recovery
+  auto it = augmented_media_packets.begin();
+  BuildAndAddRedMediaPacket(*it);
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+  counter = receiver_fec_.GetPacketCounter();
+  EXPECT_EQ(1u, counter.num_packets);
+  EXPECT_EQ(0u, counter.num_fec_packets);
+  EXPECT_EQ(0u, counter.num_recovered_packets);
+  const Timestamp first_packet_time = counter.first_packet_time;
+  EXPECT_NE(Timestamp::MinusInfinity(), first_packet_time);
+
+  // Drop one media packet.
+  auto fec_it = fec_packets.begin();
+  BuildAndAddRedFecPacket(*fec_it);
+  ++it;
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+
+  counter = receiver_fec_.GetPacketCounter();
+  EXPECT_EQ(2u, counter.num_packets);
+  EXPECT_EQ(1u, counter.num_fec_packets);
+  EXPECT_EQ(1u, counter.num_recovered_packets);
+  EXPECT_EQ(first_packet_time, counter.first_packet_time);
+}
+
+TEST_F(UlpfecReceiverTest, TwoMediaOneFecNotUsesRecoveredPackets) {
+  constexpr size_t kNumFecPackets = 1u;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  PacketizeFrame(2, 0, &augmented_media_packets, &media_packets);
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets, kNumFecPackets, &fec_packets);
+
+  FecPacketCounter counter = receiver_fec_.GetPacketCounter();
+  EXPECT_EQ(0u, counter.num_packets);
+  EXPECT_EQ(Timestamp::MinusInfinity(), counter.first_packet_time);
+
+  // Recovery
+  auto it = augmented_media_packets.begin();
+  BuildAndAddRedMediaPacket(*it, /*is_recovered=*/true);
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+  counter = receiver_fec_.GetPacketCounter();
+  EXPECT_EQ(1u, counter.num_packets);
+  EXPECT_EQ(0u, counter.num_fec_packets);
+  EXPECT_EQ(0u, counter.num_recovered_packets);
+  const Timestamp first_packet_time = counter.first_packet_time;
+  EXPECT_NE(Timestamp::MinusInfinity(), first_packet_time);
+
+  // Drop one media packet.
+  auto fec_it = fec_packets.begin();
+  BuildAndAddRedFecPacket(*fec_it);
+  ++it;
+  receiver_fec_.ProcessReceivedFec();
+
+  counter = receiver_fec_.GetPacketCounter();
+  EXPECT_EQ(2u, counter.num_packets);
+  EXPECT_EQ(1u, counter.num_fec_packets);
+  EXPECT_EQ(0u, counter.num_recovered_packets);
+  EXPECT_EQ(first_packet_time, counter.first_packet_time);
+}
+
+TEST_F(UlpfecReceiverTest, InjectGarbageFecHeaderLengthRecovery) {
+  // Byte offset 8 is the 'length recovery' field of the FEC header.
+  InjectGarbagePacketLength(8);
+}
+
+TEST_F(UlpfecReceiverTest, InjectGarbageFecLevelHeaderProtectionLength) {
+  // Byte offset 10 is the 'protection length' field in the first FEC level
+  // header.
+  InjectGarbagePacketLength(10);
+}
+
+TEST_F(UlpfecReceiverTest, TwoMediaTwoFec) {
+  const size_t kNumFecPackets = 2;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  PacketizeFrame(2, 0, &augmented_media_packets, &media_packets);
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets, kNumFecPackets, &fec_packets);
+
+  // Recovery
+  // Drop both media packets.
+  auto it = augmented_media_packets.begin();
+  auto fec_it = fec_packets.begin();
+  BuildAndAddRedFecPacket(*fec_it);
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+  ++fec_it;
+  BuildAndAddRedFecPacket(*fec_it);
+  ++it;
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+}
+
+TEST_F(UlpfecReceiverTest, TwoFramesOneFec) {
+  const size_t kNumFecPackets = 1;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  PacketizeFrame(1, 0, &augmented_media_packets, &media_packets);
+  PacketizeFrame(1, 1, &augmented_media_packets, &media_packets);
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets, kNumFecPackets, &fec_packets);
+
+  // Recovery
+  auto it = augmented_media_packets.begin();
+  BuildAndAddRedMediaPacket(augmented_media_packets.front());
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+  // Drop one media packet.
+  BuildAndAddRedFecPacket(fec_packets.front());
+  ++it;
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+}
+
+TEST_F(UlpfecReceiverTest, OneCompleteOneUnrecoverableFrame) {
+  const size_t kNumFecPackets = 1;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  PacketizeFrame(1, 0, &augmented_media_packets, &media_packets);
+  PacketizeFrame(2, 1, &augmented_media_packets, &media_packets);
+
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets, kNumFecPackets, &fec_packets);
+
+  // Recovery
+  auto it = augmented_media_packets.begin();
+  BuildAndAddRedMediaPacket(*it);  // First frame: one packet.
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+  ++it;
+  BuildAndAddRedMediaPacket(*it);  // First packet of second frame.
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+}
+
+TEST_F(UlpfecReceiverTest, MaxFramesOneFec) {
+  const size_t kNumFecPackets = 1;
+  const size_t kNumMediaPackets = 48;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  for (size_t i = 0; i < kNumMediaPackets; ++i) {
+    PacketizeFrame(1, i, &augmented_media_packets, &media_packets);
+  }
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets, kNumFecPackets, &fec_packets);
+
+  // Recovery
+  auto it = augmented_media_packets.begin();
+  ++it;  // Drop first packet.
+  for (; it != augmented_media_packets.end(); ++it) {
+    BuildAndAddRedMediaPacket(*it);
+    VerifyReconstructedMediaPacket(**it, 1);
+    receiver_fec_.ProcessReceivedFec();
+  }
+  BuildAndAddRedFecPacket(fec_packets.front());
+  it = augmented_media_packets.begin();
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+}
+
+TEST_F(UlpfecReceiverTest, TooManyFrames) {
+  const size_t kNumFecPackets = 1;
+  const size_t kNumMediaPackets = 49;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  for (size_t i = 0; i < kNumMediaPackets; ++i) {
+    PacketizeFrame(1, i, &augmented_media_packets, &media_packets);
+  }
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EXPECT_EQ(-1, fec_->EncodeFec(media_packets,
+                                kNumFecPackets * 255 / kNumMediaPackets, 0,
+                                false, kFecMaskBursty, &fec_packets));
+}
+
+TEST_F(UlpfecReceiverTest, PacketNotDroppedTooEarly) {
+  // 1 frame with 2 media packets and one FEC packet. One media packet missing.
+  // Delay the FEC packet.
+  Packet* delayed_fec = nullptr;
+  const size_t kNumFecPacketsBatch1 = 1;
+  const size_t kNumMediaPacketsBatch1 = 2;
+  std::list<AugmentedPacket*> augmented_media_packets_batch1;
+  ForwardErrorCorrection::PacketList media_packets_batch1;
+  PacketizeFrame(kNumMediaPacketsBatch1, 0, &augmented_media_packets_batch1,
+                 &media_packets_batch1);
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets_batch1, kNumFecPacketsBatch1, &fec_packets);
+
+  BuildAndAddRedMediaPacket(augmented_media_packets_batch1.front());
+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_)).Times(1);
+  receiver_fec_.ProcessReceivedFec();
+  delayed_fec = fec_packets.front();
+
+  // Fill the FEC decoder. No packets should be dropped.
+  const size_t kNumMediaPacketsBatch2 = 191;
+  std::list<AugmentedPacket*> augmented_media_packets_batch2;
+  ForwardErrorCorrection::PacketList media_packets_batch2;
+  for (size_t i = 0; i < kNumMediaPacketsBatch2; ++i) {
+    PacketizeFrame(1, i, &augmented_media_packets_batch2,
+                   &media_packets_batch2);
+  }
+  for (auto it = augmented_media_packets_batch2.begin();
+       it != augmented_media_packets_batch2.end(); ++it) {
+    BuildAndAddRedMediaPacket(*it);
+    EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_)).Times(1);
+    receiver_fec_.ProcessReceivedFec();
+  }
+
+  // Add the delayed FEC packet. One packet should be reconstructed.
+  BuildAndAddRedFecPacket(delayed_fec);
+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_)).Times(1);
+  receiver_fec_.ProcessReceivedFec();
+}
+
+TEST_F(UlpfecReceiverTest, PacketDroppedWhenTooOld) {
+  // 1 frame with 2 media packets and one FEC packet. One media packet missing.
+  // Delay the FEC packet.
+  Packet* delayed_fec = nullptr;
+  const size_t kNumFecPacketsBatch1 = 1;
+  const size_t kNumMediaPacketsBatch1 = 2;
+  std::list<AugmentedPacket*> augmented_media_packets_batch1;
+  ForwardErrorCorrection::PacketList media_packets_batch1;
+  PacketizeFrame(kNumMediaPacketsBatch1, 0, &augmented_media_packets_batch1,
+                 &media_packets_batch1);
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets_batch1, kNumFecPacketsBatch1, &fec_packets);
+
+  BuildAndAddRedMediaPacket(augmented_media_packets_batch1.front());
+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_)).Times(1);
+  receiver_fec_.ProcessReceivedFec();
+  delayed_fec = fec_packets.front();
+
+  // Fill the FEC decoder and force the last packet to be dropped.
+  const size_t kNumMediaPacketsBatch2 = 192;
+  std::list<AugmentedPacket*> augmented_media_packets_batch2;
+  ForwardErrorCorrection::PacketList media_packets_batch2;
+  for (size_t i = 0; i < kNumMediaPacketsBatch2; ++i) {
+    PacketizeFrame(1, i, &augmented_media_packets_batch2,
+                   &media_packets_batch2);
+  }
+  for (auto it = augmented_media_packets_batch2.begin();
+       it != augmented_media_packets_batch2.end(); ++it) {
+    BuildAndAddRedMediaPacket(*it);
+    EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_)).Times(1);
+    receiver_fec_.ProcessReceivedFec();
+  }
+
+  // Add the delayed FEC packet. No packet should be reconstructed since the
+  // first media packet of that frame has been dropped due to being too old.
+  BuildAndAddRedFecPacket(delayed_fec);
+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_)).Times(0);
+  receiver_fec_.ProcessReceivedFec();
+}
+
+TEST_F(UlpfecReceiverTest, OldFecPacketDropped) {
+  // 49 frames with 2 media packets and one FEC packet. All media packets
+  // missing.
+  const size_t kNumMediaPackets = 49 * 2;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  for (size_t i = 0; i < kNumMediaPackets / 2; ++i) {
+    std::list<AugmentedPacket*> frame_augmented_media_packets;
+    ForwardErrorCorrection::PacketList frame_media_packets;
+    std::list<ForwardErrorCorrection::Packet*> fec_packets;
+    PacketizeFrame(2, 0, &frame_augmented_media_packets, &frame_media_packets);
+    EncodeFec(frame_media_packets, 1, &fec_packets);
+    for (auto it = fec_packets.begin(); it != fec_packets.end(); ++it) {
+      // Only FEC packets inserted. No packets recoverable at this time.
+      BuildAndAddRedFecPacket(*it);
+      EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_)).Times(0);
+      receiver_fec_.ProcessReceivedFec();
+    }
+    // Move unique_ptr's to media_packets for lifetime management.
+    media_packets.insert(media_packets.end(),
+                         std::make_move_iterator(frame_media_packets.begin()),
+                         std::make_move_iterator(frame_media_packets.end()));
+    augmented_media_packets.insert(augmented_media_packets.end(),
+                                   frame_augmented_media_packets.begin(),
+                                   frame_augmented_media_packets.end());
+  }
+  // Insert the oldest media packet. The corresponding FEC packet is too old
+  // and should have been dropped. Only the media packet we inserted will be
+  // returned.
+  BuildAndAddRedMediaPacket(augmented_media_packets.front());
+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_)).Times(1);
+  receiver_fec_.ProcessReceivedFec();
+}
+
+TEST_F(UlpfecReceiverTest, TruncatedPacketWithFBitSet) {
+  const uint8_t kTruncatedPacket[] = {0x80, 0x2a, 0x68, 0x71, 0x29, 0xa1, 0x27,
+                                      0x3a, 0x29, 0x12, 0x2a, 0x98, 0xe0, 0x29};
+
+  SurvivesMaliciousPacket(kTruncatedPacket, sizeof(kTruncatedPacket), 100);
+}
+
+TEST_F(UlpfecReceiverTest,
+       TruncatedPacketWithFBitSetEndingAfterFirstRedHeader) {
+  const uint8_t kPacket[] = {
+      0x89, 0x27, 0x3a, 0x83, 0x27, 0x3a, 0x3a, 0xf3, 0x67, 0xbe, 0x2a,
+      0xa9, 0x27, 0x54, 0x3a, 0x3a, 0x2a, 0x67, 0x3a, 0xf3, 0x67, 0xbe,
+      0x2a, 0x27, 0xe6, 0xf6, 0x03, 0x3e, 0x29, 0x27, 0x21, 0x27, 0x2a,
+      0x29, 0x21, 0x4b, 0x29, 0x3a, 0x28, 0x29, 0xbf, 0x29, 0x2a, 0x26,
+      0x29, 0xae, 0x27, 0xa6, 0xf6, 0x00, 0x03, 0x3e};
+  SurvivesMaliciousPacket(kPacket, sizeof(kPacket), 100);
+}
+
+TEST_F(UlpfecReceiverTest, TruncatedPacketWithoutDataPastFirstBlock) {
+  const uint8_t kPacket[] = {
+      0x82, 0x38, 0x92, 0x38, 0x92, 0x38, 0xde, 0x2a, 0x11, 0xc8, 0xa3, 0xc4,
+      0x82, 0x38, 0x2a, 0x21, 0x2a, 0x28, 0x92, 0x38, 0x92, 0x00, 0x00, 0x0a,
+      0x3a, 0xc8, 0xa3, 0x3a, 0x27, 0xc4, 0x2a, 0x21, 0x2a, 0x28};
+  SurvivesMaliciousPacket(kPacket, sizeof(kPacket), 100);
+}
+
+TEST_F(UlpfecReceiverTest, MediaWithPadding) {
+  const size_t kNumFecPackets = 1;
+  std::list<AugmentedPacket*> augmented_media_packets;
+  ForwardErrorCorrection::PacketList media_packets;
+  PacketizeFrame(2, 0, &augmented_media_packets, &media_packets);
+
+  // Append four bytes of padding to the first media packet.
+  const uint8_t kPadding[] = {0, 0, 0, 4};
+  augmented_media_packets.front()->data.AppendData(kPadding);
+  augmented_media_packets.front()->data.MutableData()[0] |= 1 << 5;  // P bit.
+  augmented_media_packets.front()->header.paddingLength = 4;
+
+  std::list<ForwardErrorCorrection::Packet*> fec_packets;
+  EncodeFec(media_packets, kNumFecPackets, &fec_packets);
+
+  auto it = augmented_media_packets.begin();
+  BuildAndAddRedMediaPacket(augmented_media_packets.front());
+
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+
+  BuildAndAddRedFecPacket(fec_packets.front());
+  ++it;
+  VerifyReconstructedMediaPacket(**it, 1);
+  receiver_fec_.ProcessReceivedFec();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_fec_generator.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_fec_generator.h
new file mode 100644
index 0000000000..38e4103cb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_fec_generator.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_FEC_GENERATOR_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_FEC_GENERATOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/units/data_rate.h"
+#include "modules/include/module_fec_types.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+class VideoFecGenerator {
+ public:
+  VideoFecGenerator() = default;
+  virtual ~VideoFecGenerator() = default;
+
+  enum class FecType { kFlexFec, kUlpFec };
+  virtual FecType GetFecType() const = 0;
+  // Returns the SSRC used for FEC packets (i.e. FlexFec SSRC).
+  virtual absl::optional<uint32_t> FecSsrc() = 0;
+  // Returns the overhead, in bytes per packet, for FEC (and possibly RED).
+  virtual size_t MaxPacketOverhead() const = 0;
+  // Current rate of FEC packets generated, including all RTP-level headers.
+  virtual DataRate CurrentFecRate() const = 0;
+  // Set FEC rates, max frames before FEC is sent, and type of FEC masks.
+  virtual void SetProtectionParameters(
+      const FecProtectionParams& delta_params,
+      const FecProtectionParams& key_params) = 0;
+  // Called on new media packet to be protected. The generator may choose
+  // to generate FEC packets at this time, if so they will be stored in an
+  // internal buffer.
+  virtual void AddPacketAndGenerateFec(const RtpPacketToSend& packet) = 0;
+  // Get (and remove) and FEC packets pending in the generator. These packets
+  // will lack sequence numbers, that needs to be set externally.
+  // TODO(bugs.webrtc.org/11340): Actually FlexFec sets seq#, fix that!
+  virtual std::vector<std::unique_ptr<RtpPacketToSend>> GetFecPackets() = 0;
+  // Only called on the VideoSendStream queue, after operation has shut down,
+  // and only populated if there is an RtpState (e.g. FlexFec).
+  virtual absl::optional<RtpState> GetRtpState() = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_FEC_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer.cc
new file mode 100644
index 0000000000..bb0bf09e90
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/video/encoded_image.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<EncodedImageBuffer> VideoRtpDepacketizer::AssembleFrame(
+    rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads) {
+  size_t frame_size = 0;
+  for (rtc::ArrayView<const uint8_t> payload : rtp_payloads) {
+    frame_size += payload.size();
+  }
+
+  rtc::scoped_refptr<EncodedImageBuffer> bitstream =
+      EncodedImageBuffer::Create(frame_size);
+
+  uint8_t* write_at = bitstream->data();
+  for (rtc::ArrayView<const uint8_t> payload : rtp_payloads) {
+    memcpy(write_at, payload.data(), payload.size());
+    write_at += payload.size();
+  }
+  RTC_DCHECK_EQ(write_at - bitstream->data(), bitstream->size());
+  return bitstream;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer.h
new file mode 100644
index 0000000000..2266120799
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/video/encoded_image.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class VideoRtpDepacketizer {
+ public:
+  struct ParsedRtpPayload {
+    RTPVideoHeader video_header;
+    rtc::CopyOnWriteBuffer video_payload;
+  };
+
+  virtual ~VideoRtpDepacketizer() = default;
+  virtual absl::optional<ParsedRtpPayload> Parse(
+      rtc::CopyOnWriteBuffer rtp_payload) = 0;
+  virtual rtc::scoped_refptr<EncodedImageBuffer> AssembleFrame(
+      rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
new file mode 100644
index 0000000000..870f788538
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.cc
@@ -0,0 +1,395 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <utility>
+
+#include "modules/rtp_rtcp/source/leb128.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "rtc_base/byte_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace {
+// AV1 format:
+//
+// RTP payload syntax:
+//     0 1 2 3 4 5 6 7
+//    +-+-+-+-+-+-+-+-+
+//    |Z|Y| W |N|-|-|-| (REQUIRED)
+//    +=+=+=+=+=+=+=+=+ (REPEATED W-1 times, or any times if W = 0)
+//    |1|             |
+//    +-+ OBU fragment|
+//    |1|             | (REQUIRED, leb128 encoded)
+//    +-+    size     |
+//    |0|             |
+//    +-+-+-+-+-+-+-+-+
+//    |  OBU fragment |
+//    |     ...       |
+//    +=+=+=+=+=+=+=+=+
+//    |     ...       |
+//    +=+=+=+=+=+=+=+=+ if W > 0, last fragment MUST NOT have size field
+//    |  OBU fragment |
+//    |     ...       |
+//    +=+=+=+=+=+=+=+=+
+//
+//
+// OBU syntax:
+//     0 1 2 3 4 5 6 7
+//    +-+-+-+-+-+-+-+-+
+//    |0| type  |X|S|-| (REQUIRED)
+//    +-+-+-+-+-+-+-+-+
+// X: | TID |SID|-|-|-| (OPTIONAL)
+//    +-+-+-+-+-+-+-+-+
+//    |1|             |
+//    +-+ OBU payload |
+// S: |1|             | (OPTIONAL, variable length leb128 encoded)
+//    +-+    size     |
+//    |0|             |
+//    +-+-+-+-+-+-+-+-+
+//    |  OBU payload  |
+//    |     ...       |
+class ArrayOfArrayViews {
+ public:
+  class const_iterator;
+  ArrayOfArrayViews() = default;
+  ArrayOfArrayViews(const ArrayOfArrayViews&) = default;
+  ArrayOfArrayViews& operator=(const ArrayOfArrayViews&) = default;
+  ~ArrayOfArrayViews() = default;
+
+  const_iterator begin() const;
+  const_iterator end() const;
+  bool empty() const { return data_.empty(); }
+  size_t size() const { return size_; }
+  void CopyTo(uint8_t* destination, const_iterator first) const;
+
+  void Append(const uint8_t* data, size_t size) {
+    data_.emplace_back(data, size);
+    size_ += size;
+  }
+
+ private:
+  using Storage = absl::InlinedVector<rtc::ArrayView<const uint8_t>, 2>;
+
+  size_t size_ = 0;
+  Storage data_;
+};
+
+class ArrayOfArrayViews::const_iterator {
+ public:
+  const_iterator() = default;
+  const_iterator(const const_iterator&) = default;
+  const_iterator& operator=(const const_iterator&) = default;
+
+  const_iterator& operator++() {
+    if (++inner_ == outer_->size()) {
+      ++outer_;
+      inner_ = 0;
+    }
+    return *this;
+  }
+  uint8_t operator*() const { return (*outer_)[inner_]; }
+
+  friend bool operator==(const const_iterator& lhs, const const_iterator& rhs) {
+    return lhs.outer_ == rhs.outer_ && lhs.inner_ == rhs.inner_;
+  }
+
+ private:
+  friend ArrayOfArrayViews;
+  const_iterator(ArrayOfArrayViews::Storage::const_iterator outer, size_t inner)
+      : outer_(outer), inner_(inner) {}
+
+  Storage::const_iterator outer_;
+  size_t inner_;
+};
+
+ArrayOfArrayViews::const_iterator ArrayOfArrayViews::begin() const {
+  return const_iterator(data_.begin(), 0);
+}
+
+ArrayOfArrayViews::const_iterator ArrayOfArrayViews::end() const {
+  return const_iterator(data_.end(), 0);
+}
+
+void ArrayOfArrayViews::CopyTo(uint8_t* destination,
+                               const_iterator first) const {
+  if (first == end()) {
+    // Empty OBU payload. E.g. Temporal Delimiters are always empty.
+    return;
+  }
+  size_t first_chunk_size = first.outer_->size() - first.inner_;
+  memcpy(destination, first.outer_->data() + first.inner_, first_chunk_size);
+  destination += first_chunk_size;
+  for (auto it = std::next(first.outer_); it != data_.end(); ++it) {
+    memcpy(destination, it->data(), it->size());
+    destination += it->size();
+  }
+}
+
+struct ObuInfo {
+  // Size of the obu_header and obu_size fields in the ouput frame.
+  size_t prefix_size = 0;
+  // obu_header() and obu_size (leb128 encoded payload_size).
+  // obu_header can be up to 2 bytes, obu_size - up to 5.
+  std::array<uint8_t, 7> prefix;
+  // Size of the obu payload in the output frame, i.e. excluding header
+  size_t payload_size = 0;
+  // iterator pointing to the beginning of the obu payload.
+  ArrayOfArrayViews::const_iterator payload_offset;
+  // OBU payloads as written in the rtp packet payloads.
+  ArrayOfArrayViews data;
+};
+// Expect that majority of the frame won't use more than 4 obus.
+// In a simple stream delta frame consist of single Frame OBU, while key frame
+// also has Sequence Header OBU.
+using VectorObuInfo = absl::InlinedVector<ObuInfo, 4>;
+
+constexpr uint8_t kObuSizePresentBit = 0b0'0000'010;
+
+bool ObuHasExtension(uint8_t obu_header) {
+  return obu_header & 0b0'0000'100u;
+}
+
+bool ObuHasSize(uint8_t obu_header) {
+  return obu_header & kObuSizePresentBit;
+}
+
+bool RtpStartsWithFragment(uint8_t aggregation_header) {
+  return aggregation_header & 0b1000'0000u;
+}
+bool RtpEndsWithFragment(uint8_t aggregation_header) {
+  return aggregation_header & 0b0100'0000u;
+}
+int RtpNumObus(uint8_t aggregation_header) {  // 0 for any number of obus.
+  return (aggregation_header & 0b0011'0000u) >> 4;
+}
+int RtpStartsNewCodedVideoSequence(uint8_t aggregation_header) {
+  return aggregation_header & 0b0000'1000u;
+}
+
+// Reorgonizes array of rtp payloads into array of obus:
+// fills ObuInfo::data field.
+// Returns empty vector on error.
+VectorObuInfo ParseObus(
+    rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads) {
+  VectorObuInfo obu_infos;
+  bool expect_continues_obu = false;
+  for (rtc::ArrayView<const uint8_t> rtp_payload : rtp_payloads) {
+    rtc::ByteBufferReader payload(
+        reinterpret_cast<const char*>(rtp_payload.data()), rtp_payload.size());
+    uint8_t aggregation_header;
+    if (!payload.ReadUInt8(&aggregation_header)) {
+      RTC_DLOG(LS_WARNING)
+          << "Failed to find aggregation header in the packet.";
+      return {};
+    }
+    // Z-bit: 1 if the first OBU contained in the packet is a continuation of a
+    // previous OBU.
+    bool continues_obu = RtpStartsWithFragment(aggregation_header);
+    if (continues_obu != expect_continues_obu) {
+      RTC_DLOG(LS_WARNING) << "Unexpected Z-bit " << continues_obu;
+      return {};
+    }
+    int num_expected_obus = RtpNumObus(aggregation_header);
+    if (payload.Length() == 0) {
+      // rtp packet has just the aggregation header. That may be valid only when
+      // there is exactly one fragment in the packet of size 0.
+      if (num_expected_obus != 1) {
+        RTC_DLOG(LS_WARNING)
+            << "Invalid packet with just an aggregation header.";
+        return {};
+      }
+      if (!continues_obu) {
+        // Empty packet just to notify there is a new OBU.
+        obu_infos.emplace_back();
+      }
+      expect_continues_obu = RtpEndsWithFragment(aggregation_header);
+      continue;
+    }
+
+    for (int obu_index = 1; payload.Length() > 0; ++obu_index) {
+      ObuInfo& obu_info = (obu_index == 1 && continues_obu)
+                              ? obu_infos.back()
+                              : obu_infos.emplace_back();
+      uint64_t fragment_size;
+      // When num_expected_obus > 0, last OBU (fragment) is not preceeded by
+      // the size field. See W field in
+      // https://aomediacodec.github.io/av1-rtp-spec/#43-av1-aggregation-header
+      bool has_fragment_size = (obu_index != num_expected_obus);
+      if (has_fragment_size) {
+        if (!payload.ReadUVarint(&fragment_size)) {
+          RTC_DLOG(LS_WARNING) << "Failed to read fragment size for obu #"
+                               << obu_index << "/" << num_expected_obus;
+          return {};
+        }
+        if (fragment_size > payload.Length()) {
+          // Malformed input: written size is larger than remaining buffer.
+          RTC_DLOG(LS_WARNING) << "Malformed fragment size " << fragment_size
+                               << " is larger than remaining size "
+                               << payload.Length() << " while reading obu #"
+                               << obu_index << "/" << num_expected_obus;
+          return {};
+        }
+      } else {
+        fragment_size = payload.Length();
+      }
+      // While it is in-practical to pass empty fragments, it is still possible.
+      if (fragment_size > 0) {
+        obu_info.data.Append(reinterpret_cast<const uint8_t*>(payload.Data()),
+                             fragment_size);
+        payload.Consume(fragment_size);
+      }
+    }
+    // Z flag should be same as Y flag of the next packet.
+    expect_continues_obu = RtpEndsWithFragment(aggregation_header);
+  }
+  if (expect_continues_obu) {
+    RTC_DLOG(LS_WARNING) << "Last packet shouldn't have last obu fragmented.";
+    return {};
+  }
+  return obu_infos;
+}
+
+// Calculates sizes for the Obu, i.e. base on ObuInfo::data field calculates
+// all other fields in the ObuInfo structure.
+// Returns false if obu found to be misformed.
+bool CalculateObuSizes(ObuInfo* obu_info) {
+  if (obu_info->data.empty()) {
+    RTC_DLOG(LS_WARNING) << "Invalid bitstream: empty obu provided.";
+    return false;
+  }
+  auto it = obu_info->data.begin();
+  uint8_t obu_header = *it;
+  obu_info->prefix[0] = obu_header | kObuSizePresentBit;
+  obu_info->prefix_size = 1;
+  ++it;
+  if (ObuHasExtension(obu_header)) {
+    if (it == obu_info->data.end()) {
+      return false;
+    }
+    obu_info->prefix[1] = *it;  // obu_extension_header
+    obu_info->prefix_size = 2;
+    ++it;
+  }
+  // Read, validate, and skip size, if present.
+  if (!ObuHasSize(obu_header)) {
+    obu_info->payload_size = obu_info->data.size() - obu_info->prefix_size;
+  } else {
+    // Read leb128 encoded field obu_size.
+    uint64_t obu_size_bytes = 0;
+    // Number of bytes obu_size field occupy in the bitstream.
+    int size_of_obu_size_bytes = 0;
+    uint8_t leb128_byte;
+    do {
+      if (it == obu_info->data.end() || size_of_obu_size_bytes >= 8) {
+        RTC_DLOG(LS_WARNING)
+            << "Failed to read obu_size. obu_size field is too long: "
+            << size_of_obu_size_bytes << " bytes processed.";
+        return false;
+      }
+      leb128_byte = *it;
+      obu_size_bytes |= uint64_t{leb128_byte & 0x7Fu}
+                        << (size_of_obu_size_bytes * 7);
+      ++size_of_obu_size_bytes;
+      ++it;
+    } while ((leb128_byte & 0x80) != 0);
+
+    obu_info->payload_size =
+        obu_info->data.size() - obu_info->prefix_size - size_of_obu_size_bytes;
+    if (obu_size_bytes != obu_info->payload_size) {
+      // obu_size was present in the bitstream and mismatches calculated size.
+      RTC_DLOG(LS_WARNING) << "Mismatch in obu_size. signaled: "
+                           << obu_size_bytes
+                           << ", actual: " << obu_info->payload_size;
+      return false;
+    }
+  }
+  obu_info->payload_offset = it;
+  obu_info->prefix_size +=
+      WriteLeb128(rtc::dchecked_cast<uint64_t>(obu_info->payload_size),
+                  obu_info->prefix.data() + obu_info->prefix_size);
+  return true;
+}
+
+}  // namespace
+
+rtc::scoped_refptr<EncodedImageBuffer> VideoRtpDepacketizerAv1::AssembleFrame(
+    rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads) {
+  VectorObuInfo obu_infos = ParseObus(rtp_payloads);
+  if (obu_infos.empty()) {
+    return nullptr;
+  }
+
+  size_t frame_size = 0;
+  for (ObuInfo& obu_info : obu_infos) {
+    if (!CalculateObuSizes(&obu_info)) {
+      return nullptr;
+    }
+    frame_size += (obu_info.prefix_size + obu_info.payload_size);
+  }
+
+  rtc::scoped_refptr<EncodedImageBuffer> bitstream =
+      EncodedImageBuffer::Create(frame_size);
+  uint8_t* write_at = bitstream->data();
+  for (const ObuInfo& obu_info : obu_infos) {
+    // Copy the obu_header and obu_size fields.
+    memcpy(write_at, obu_info.prefix.data(), obu_info.prefix_size);
+    write_at += obu_info.prefix_size;
+    // Copy the obu payload.
+    obu_info.data.CopyTo(write_at, obu_info.payload_offset);
+    write_at += obu_info.payload_size;
+  }
+  RTC_CHECK_EQ(write_at - bitstream->data(), bitstream->size());
+  return bitstream;
+}
+
+absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+VideoRtpDepacketizerAv1::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
+  if (rtp_payload.size() == 0) {
+    RTC_DLOG(LS_ERROR) << "Empty rtp payload.";
+    return absl::nullopt;
+  }
+  uint8_t aggregation_header = rtp_payload.cdata()[0];
+  if (RtpStartsNewCodedVideoSequence(aggregation_header) &&
+      RtpStartsWithFragment(aggregation_header)) {
+    // new coded video sequence can't start from an OBU fragment.
+    return absl::nullopt;
+  }
+  absl::optional<ParsedRtpPayload> parsed(absl::in_place);
+
+  // To assemble frame, all of the rtp payload is required, including
+  // aggregation header.
+  parsed->video_payload = std::move(rtp_payload);
+
+  parsed->video_header.codec = VideoCodecType::kVideoCodecAV1;
+  // These are not accurate since frame may consist of several packet aligned
+  // chunks of obus, but should be good enough for most cases. It might produce
+  // frame that do not map to any real frame, but av1 decoder should be able to
+  // handle it since it promise to handle individual obus rather than full
+  // frames.
+  parsed->video_header.is_first_packet_in_frame =
+      !RtpStartsWithFragment(aggregation_header);
+  parsed->video_header.is_last_packet_in_frame =
+      !RtpEndsWithFragment(aggregation_header);
+
+  parsed->video_header.frame_type =
+      RtpStartsNewCodedVideoSequence(aggregation_header)
+          ? VideoFrameType::kVideoFrameKey
+          : VideoFrameType::kVideoFrameDelta;
+  return parsed;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h
new file mode 100644
index 0000000000..ac8c7e6d11
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_AV1_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_AV1_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/video/encoded_image.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class VideoRtpDepacketizerAv1 : public VideoRtpDepacketizer {
+ public:
+  VideoRtpDepacketizerAv1() = default;
+  VideoRtpDepacketizerAv1(const VideoRtpDepacketizerAv1&) = delete;
+  VideoRtpDepacketizerAv1& operator=(const VideoRtpDepacketizerAv1&) = delete;
+  ~VideoRtpDepacketizerAv1() override = default;
+
+  rtc::scoped_refptr<EncodedImageBuffer> AssembleFrame(
+      rtc::ArrayView<const rtc::ArrayView<const uint8_t>> rtp_payloads)
+      override;
+
+  absl::optional<ParsedRtpPayload> Parse(
+      rtc::CopyOnWriteBuffer rtp_payload) override;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_AV1_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc
new file mode 100644
index 0000000000..e9ad1a1b8e
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_av1_unittest.cc
@@ -0,0 +1,392 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+
+// Signals number of the OBU (fragments) in the packet.
+constexpr uint8_t kObuCountOne = 0b00'01'0000;
+
+constexpr uint8_t kObuHeaderSequenceHeader = 0b0'0001'000;
+constexpr uint8_t kObuHeaderFrame = 0b0'0110'000;
+
+constexpr uint8_t kObuHeaderHasSize = 0b0'0000'010;
+
+TEST(VideoRtpDepacketizerAv1Test, ParsePassFullRtpPayloadAsCodecPayload) {
+  const uint8_t packet[] = {(uint8_t{1} << 7) | kObuCountOne, 1, 2, 3, 4};
+  rtc::CopyOnWriteBuffer rtp_payload(packet);
+  VideoRtpDepacketizerAv1 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+  ASSERT_TRUE(parsed);
+  EXPECT_EQ(parsed->video_payload.size(), sizeof(packet));
+  EXPECT_TRUE(parsed->video_payload.cdata() == rtp_payload.cdata());
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     ParseTreatsContinuationFlagAsNotBeginningOfFrame) {
+  const uint8_t packet[] = {
+      (uint8_t{1} << 7) | kObuCountOne,
+      kObuHeaderFrame};  // Value doesn't matter since it is a
+                         // continuation of the OBU from previous packet.
+  VideoRtpDepacketizerAv1 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
+  ASSERT_TRUE(parsed);
+  EXPECT_FALSE(parsed->video_header.is_first_packet_in_frame);
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     ParseTreatsNoContinuationFlagAsBeginningOfFrame) {
+  const uint8_t packet[] = {(uint8_t{0} << 7) | kObuCountOne, kObuHeaderFrame};
+  VideoRtpDepacketizerAv1 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
+  ASSERT_TRUE(parsed);
+  EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
+}
+
+TEST(VideoRtpDepacketizerAv1Test, ParseTreatsWillContinueFlagAsNotEndOfFrame) {
+  const uint8_t packet[] = {(uint8_t{1} << 6) | kObuCountOne, kObuHeaderFrame};
+  rtc::CopyOnWriteBuffer rtp_payload(packet);
+  VideoRtpDepacketizerAv1 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+  ASSERT_TRUE(parsed);
+  EXPECT_FALSE(parsed->video_header.is_last_packet_in_frame);
+}
+
+TEST(VideoRtpDepacketizerAv1Test, ParseTreatsNoWillContinueFlagAsEndOfFrame) {
+  const uint8_t packet[] = {(uint8_t{0} << 6) | kObuCountOne, kObuHeaderFrame};
+  VideoRtpDepacketizerAv1 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
+  ASSERT_TRUE(parsed);
+  EXPECT_TRUE(parsed->video_header.is_last_packet_in_frame);
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     ParseUsesNewCodedVideoSequenceBitAsKeyFrameIndidcator) {
+  const uint8_t packet[] = {(uint8_t{1} << 3) | kObuCountOne,
+                            kObuHeaderSequenceHeader};
+  VideoRtpDepacketizerAv1 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
+  ASSERT_TRUE(parsed);
+  EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
+  EXPECT_TRUE(parsed->video_header.frame_type ==
+              VideoFrameType::kVideoFrameKey);
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     ParseUsesUnsetNewCodedVideoSequenceBitAsDeltaFrameIndidcator) {
+  const uint8_t packet[] = {(uint8_t{0} << 3) | kObuCountOne,
+                            kObuHeaderSequenceHeader};
+  VideoRtpDepacketizerAv1 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtc::CopyOnWriteBuffer(packet));
+  ASSERT_TRUE(parsed);
+  EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
+  EXPECT_TRUE(parsed->video_header.frame_type ==
+              VideoFrameType::kVideoFrameDelta);
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     ParseRejectsPacketWithNewCVSAndContinuationFlagsBothSet) {
+  const uint8_t packet[] = {0b10'00'1000 | kObuCountOne,
+                            kObuHeaderSequenceHeader};
+  VideoRtpDepacketizerAv1 depacketizer;
+  ASSERT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(packet)));
+}
+
+TEST(VideoRtpDepacketizerAv1Test, AssembleFrameSetsOBUPayloadSizeWhenAbsent) {
+  const uint8_t payload1[] = {0b00'01'0000,  // aggregation header
+                              0b0'0110'000,  // /  Frame
+                              20, 30, 40};   // \  OBU
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  rtc::ArrayView<const uint8_t> frame_view(*frame);
+  EXPECT_TRUE(frame_view[0] & kObuHeaderHasSize);
+  EXPECT_EQ(frame_view[1], 3);
+}
+
+TEST(VideoRtpDepacketizerAv1Test, AssembleFrameSetsOBUPayloadSizeWhenPresent) {
+  const uint8_t payload1[] = {0b00'01'0000,  // aggregation header
+                              0b0'0110'010,  // /  Frame OBU header
+                              3,             // obu_size
+                              20,
+                              30,
+                              40};  // \  obu_payload
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  rtc::ArrayView<const uint8_t> frame_view(*frame);
+  EXPECT_TRUE(frame_view[0] & kObuHeaderHasSize);
+  EXPECT_EQ(frame_view[1], 3);
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     AssembleFrameSetsOBUPayloadSizeAfterExtensionWhenAbsent) {
+  const uint8_t payload1[] = {0b00'01'0000,           // aggregation header
+                              0b0'0110'100,           // /  Frame
+                              0b010'01'000,           // | extension_header
+                              20,           30, 40};  // \  OBU
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  rtc::ArrayView<const uint8_t> frame_view(*frame);
+  EXPECT_TRUE(frame_view[0] & kObuHeaderHasSize);
+  EXPECT_EQ(frame_view[2], 3);
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     AssembleFrameSetsOBUPayloadSizeAfterExtensionWhenPresent) {
+  const uint8_t payload1[] = {0b00'01'0000,  // aggregation header
+                              0b0'0110'110,  // /  Frame OBU header
+                              0b010'01'000,  // | extension_header
+                              3,             // | obu_size
+                              20,
+                              30,
+                              40};  // \  obu_payload
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  rtc::ArrayView<const uint8_t> frame_view(*frame);
+  EXPECT_TRUE(frame_view[0] & kObuHeaderHasSize);
+  EXPECT_EQ(frame_view[2], 3);
+}
+
+TEST(VideoRtpDepacketizerAv1Test, AssembleFrameFromOnePacketWithOneObu) {
+  const uint8_t payload1[] = {0b00'01'0000,  // aggregation header
+                              0b0'0110'000,  // /  Frame
+                              20};           // \  OBU
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_THAT(rtc::ArrayView<const uint8_t>(*frame),
+              ElementsAre(0b0'0110'010, 1, 20));
+}
+
+TEST(VideoRtpDepacketizerAv1Test, AssembleFrameFromOnePacketWithTwoObus) {
+  const uint8_t payload1[] = {0b00'10'0000,  // aggregation header
+                              2,             // /  Sequence
+                              0b0'0001'000,  // |  Header
+                              10,            // \  OBU
+                              0b0'0110'000,  // /  Frame
+                              20};           // \  OBU
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_THAT(rtc::ArrayView<const uint8_t>(*frame),
+              ElementsAre(0b0'0001'010, 1, 10,    // Sequence Header OBU
+                          0b0'0110'010, 1, 20));  // Frame OBU
+}
+
+TEST(VideoRtpDepacketizerAv1Test, AssembleFrameFromTwoPacketsWithOneObu) {
+  const uint8_t payload1[] = {0b01'01'0000,  // aggregation header
+                              0b0'0110'000, 20, 30};
+  const uint8_t payload2[] = {0b10'01'0000,  // aggregation header
+                              40};
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_THAT(rtc::ArrayView<const uint8_t>(*frame),
+              ElementsAre(0b0'0110'010, 3, 20, 30, 40));
+}
+
+TEST(VideoRtpDepacketizerAv1Test, AssembleFrameFromTwoPacketsWithTwoObu) {
+  const uint8_t payload1[] = {0b01'10'0000,  // aggregation header
+                              2,             // /  Sequence
+                              0b0'0001'000,  // |  Header
+                              10,            // \  OBU
+                              0b0'0110'000,  //
+                              20,
+                              30};           //
+  const uint8_t payload2[] = {0b10'01'0000,  // aggregation header
+                              40};           //
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_THAT(rtc::ArrayView<const uint8_t>(*frame),
+              ElementsAre(0b0'0001'010, 1, 10,            // SH
+                          0b0'0110'010, 3, 20, 30, 40));  // Frame
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     AssembleFrameFromTwoPacketsWithManyObusSomeWithExtensions) {
+  const uint8_t payload1[] = {0b01'00'0000,  // aggregation header
+                              2,             // /
+                              0b0'0001'000,  // |  Sequence Header
+                              10,            // \  OBU
+                              2,             // /
+                              0b0'0101'000,  // |  Metadata OBU
+                              20,            // \  without extension
+                              4,             // /
+                              0b0'0101'100,  // |  Metadata OBU
+                              0b001'10'000,  // |  with extension
+                              20,            // |
+                              30,            // \  metadata payload
+                              5,             // /
+                              0b0'0110'100,  // |  Frame OBU
+                              0b001'10'000,  // |  with extension
+                              40,            // |
+                              50,            // |
+                              60};           // |
+  const uint8_t payload2[] = {0b10'01'0000,  // aggregation header
+                              70, 80, 90};   // \  tail of the frame OBU
+
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_THAT(rtc::ArrayView<const uint8_t>(*frame),
+              ElementsAre(  // Sequence header OBU
+                  0b0'0001'010, 1, 10,
+                  // Metadata OBU without extension
+                  0b0'0101'010, 1, 20,
+                  // Metadata OBU with extenion
+                  0b0'0101'110, 0b001'10'000, 2, 20, 30,
+                  // Frame OBU with extension
+                  0b0'0110'110, 0b001'10'000, 6, 40, 50, 60, 70, 80, 90));
+}
+
+TEST(VideoRtpDepacketizerAv1Test, AssembleFrameWithOneObuFromManyPackets) {
+  const uint8_t payload1[] = {0b01'01'0000,  // aggregation header
+                              0b0'0110'000, 11, 12};
+  const uint8_t payload2[] = {0b11'01'0000,  // aggregation header
+                              13, 14};
+  const uint8_t payload3[] = {0b11'01'0000,  // aggregation header
+                              15, 16, 17};
+  const uint8_t payload4[] = {0b10'01'0000,  // aggregation header
+                              18};
+
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1, payload2, payload3,
+                                              payload4};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_THAT(rtc::ArrayView<const uint8_t>(*frame),
+              ElementsAre(0b0'0110'010, 8, 11, 12, 13, 14, 15, 16, 17, 18));
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     AssembleFrameFromManyPacketsWithSomeObuBorderAligned) {
+  const uint8_t payload1[] = {0b01'10'0000,  // aggregation header
+                              3,             // size of the 1st fragment
+                              0b0'0011'000,  // Frame header OBU
+                              11,
+                              12,
+                              0b0'0100'000,  // Tile group OBU
+                              21,
+                              22,
+                              23};
+  const uint8_t payload2[] = {0b10'01'0000,  // aggregation header
+                              24, 25, 26, 27};
+  // payload2 ends an OBU, payload3 starts a new one.
+  const uint8_t payload3[] = {0b01'10'0000,  // aggregation header
+                              3,             // size of the 1st fragment
+                              0b0'0111'000,  // Redundant frame header OBU
+                              11,
+                              12,
+                              0b0'0100'000,  // Tile group OBU
+                              31,
+                              32};
+  const uint8_t payload4[] = {0b10'01'0000,  // aggregation header
+                              33, 34, 35, 36};
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1, payload2, payload3,
+                                              payload4};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_THAT(rtc::ArrayView<const uint8_t>(*frame),
+              ElementsAre(0b0'0011'010, 2, 11, 12,  // Frame header
+                          0b0'0100'010, 7, 21, 22, 23, 24, 25, 26, 27,  //
+                          0b0'0111'010, 2, 11, 12,                      //
+                          0b0'0100'010, 6, 31, 32, 33, 34, 35, 36));
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     AssembleFrameFromOnePacketsOneObuPayloadSize127Bytes) {
+  uint8_t payload1[4 + 127];
+  memset(payload1, 0, sizeof(payload1));
+  payload1[0] = 0b00'00'0000;  // aggregation header
+  payload1[1] = 0x80;          // leb128 encoded size of 128 bytes
+  payload1[2] = 0x01;          // in two bytes
+  payload1[3] = 0b0'0110'000;  // obu_header with size and extension bits unset.
+  payload1[4 + 42] = 0x42;
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_EQ(frame->size(), 2 + 127u);
+  rtc::ArrayView<const uint8_t> frame_view(*frame);
+  EXPECT_EQ(frame_view[0], 0b0'0110'010);  // obu_header with size bit set.
+  EXPECT_EQ(frame_view[1], 127);  // obu payload size, 1 byte enough to encode.
+  // Check 'random' byte from the payload is at the same 'random' offset.
+  EXPECT_EQ(frame_view[2 + 42], 0x42);
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     AssembleFrameFromTwoPacketsOneObuPayloadSize128Bytes) {
+  uint8_t payload1[3 + 32];
+  memset(payload1, 0, sizeof(payload1));
+  payload1[0] = 0b01'00'0000;  // aggregation header
+  payload1[1] = 33;            // leb128 encoded size of 33 bytes in one byte
+  payload1[2] = 0b0'0110'000;  // obu_header with size and extension bits unset.
+  payload1[3 + 10] = 0x10;
+  uint8_t payload2[2 + 96];
+  memset(payload2, 0, sizeof(payload2));
+  payload2[0] = 0b10'00'0000;  // aggregation header
+  payload2[1] = 96;            // leb128 encoded size of 96 bytes in one byte
+  payload2[2 + 20] = 0x20;
+
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_EQ(frame->size(), 3 + 128u);
+  rtc::ArrayView<const uint8_t> frame_view(*frame);
+  EXPECT_EQ(frame_view[0], 0b0'0110'010);  // obu_header with size bit set.
+  EXPECT_EQ(frame_view[1], 0x80);          // obu payload size of 128 bytes.
+  EXPECT_EQ(frame_view[2], 0x01);          // encoded in two byes
+  // Check two 'random' byte from the payload is at the same 'random' offset.
+  EXPECT_EQ(frame_view[3 + 10], 0x10);
+  EXPECT_EQ(frame_view[3 + 32 + 20], 0x20);
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     AssembleFrameFromAlmostEmptyPacketStartingAnOBU) {
+  const uint8_t payload1[] = {0b01'01'0000};
+  const uint8_t payload2[] = {0b10'01'0000, 0b0'0110'000, 10, 20, 30};
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_THAT(rtc::ArrayView<const uint8_t>(*frame),
+              ElementsAre(0b0'0110'010, 3, 10, 20, 30));
+}
+
+TEST(VideoRtpDepacketizerAv1Test,
+     AssembleFrameFromAlmostEmptyPacketFinishingAnOBU) {
+  const uint8_t payload1[] = {0b01'01'0000, 0b0'0110'000, 10, 20, 30};
+  const uint8_t payload2[] = {0b10'01'0000};
+  rtc::ArrayView<const uint8_t> payloads[] = {payload1, payload2};
+
+  auto frame = VideoRtpDepacketizerAv1().AssembleFrame(payloads);
+  ASSERT_TRUE(frame);
+  EXPECT_THAT(rtc::ArrayView<const uint8_t>(*frame),
+              ElementsAre(0b0'0110'010, 3, 10, 20, 30));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc
new file mode 100644
index 0000000000..6010771318
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+constexpr uint8_t kKeyFrameBit = 0b0000'0001;
+constexpr uint8_t kFirstPacketBit = 0b0000'0010;
+// If this bit is set, there will be an extended header contained in this
+// packet. This was added later so old clients will not send this.
+constexpr uint8_t kExtendedHeaderBit = 0b0000'0100;
+
+constexpr size_t kGenericHeaderLength = 1;
+constexpr size_t kExtendedHeaderLength = 2;
+}  // namespace
+
+absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+VideoRtpDepacketizerGeneric::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
+  if (rtp_payload.size() == 0) {
+    RTC_LOG(LS_WARNING) << "Empty payload.";
+    return absl::nullopt;
+  }
+  absl::optional<ParsedRtpPayload> parsed(absl::in_place);
+  const uint8_t* payload_data = rtp_payload.cdata();
+
+  uint8_t generic_header = payload_data[0];
+  size_t offset = kGenericHeaderLength;
+
+  parsed->video_header.frame_type = (generic_header & kKeyFrameBit)
+                                        ? VideoFrameType::kVideoFrameKey
+                                        : VideoFrameType::kVideoFrameDelta;
+  parsed->video_header.is_first_packet_in_frame =
+      (generic_header & kFirstPacketBit) != 0;
+  parsed->video_header.codec = kVideoCodecGeneric;
+  parsed->video_header.width = 0;
+  parsed->video_header.height = 0;
+
+  if (generic_header & kExtendedHeaderBit) {
+    if (rtp_payload.size() < offset + kExtendedHeaderLength) {
+      RTC_LOG(LS_WARNING) << "Too short payload for generic header.";
+      return absl::nullopt;
+    }
+    parsed->video_header.video_type_header
+        .emplace<RTPVideoHeaderLegacyGeneric>()
+        .picture_id = ((payload_data[1] & 0x7F) << 8) | payload_data[2];
+    offset += kExtendedHeaderLength;
+  }
+
+  parsed->video_payload =
+      rtp_payload.Slice(offset, rtp_payload.size() - offset);
+  return parsed;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h
new file mode 100644
index 0000000000..27056da481
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_GENERIC_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_GENERIC_H_
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class VideoRtpDepacketizerGeneric : public VideoRtpDepacketizer {
+ public:
+  ~VideoRtpDepacketizerGeneric() override = default;
+
+  absl::optional<ParsedRtpPayload> Parse(
+      rtc::CopyOnWriteBuffer rtp_payload) override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_GENERIC_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic_unittest.cc
new file mode 100644
index 0000000000..860ddab4fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic_unittest.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h"
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::SizeIs;
+
+TEST(VideoRtpDepacketizerGeneric, NonExtendedHeaderNoFrameId) {
+  const size_t kRtpPayloadSize = 10;
+  const uint8_t kPayload[kRtpPayloadSize] = {0x01};
+  rtc::CopyOnWriteBuffer rtp_payload(kPayload);
+
+  VideoRtpDepacketizerGeneric depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+
+  ASSERT_TRUE(parsed);
+  EXPECT_EQ(parsed->video_header.generic, absl::nullopt);
+  EXPECT_THAT(parsed->video_payload, SizeIs(kRtpPayloadSize - 1));
+}
+
+TEST(VideoRtpDepacketizerGeneric, ExtendedHeaderParsesFrameId) {
+  const size_t kRtpPayloadSize = 10;
+  const uint8_t kPayload[kRtpPayloadSize] = {0x05, 0x13, 0x37};
+  rtc::CopyOnWriteBuffer rtp_payload(kPayload);
+
+  VideoRtpDepacketizerGeneric depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+
+  ASSERT_TRUE(parsed);
+  const auto* generic_header = absl::get_if<RTPVideoHeaderLegacyGeneric>(
+      &parsed->video_header.video_type_header);
+  ASSERT_TRUE(generic_header);
+  EXPECT_EQ(generic_header->picture_id, 0x1337);
+  EXPECT_THAT(parsed->video_payload, SizeIs(kRtpPayloadSize - 3));
+}
+
+TEST(VideoRtpDepacketizerGeneric, PassRtpPayloadAsVideoPayload) {
+  const uint8_t kPayload[] = {0x01, 0x25, 0x52};
+  rtc::CopyOnWriteBuffer rtp_payload(kPayload);
+
+  VideoRtpDepacketizerGeneric depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+
+  ASSERT_TRUE(parsed);
+  // Check there was no memcpy involved by verifying return and original buffers
+  // point to the same buffer.
+  EXPECT_EQ(parsed->video_payload.cdata(), rtp_payload.cdata() + 1);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
new file mode 100644
index 0000000000..9978e5f5fc
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc
@@ -0,0 +1,310 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"
+
+#include <cstddef>
+#include <cstdint>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "absl/types/variant.h"
+#include "common_video/h264/h264_common.h"
+#include "common_video/h264/pps_parser.h"
+#include "common_video/h264/sps_parser.h"
+#include "common_video/h264/sps_vui_rewriter.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_format_h264.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kNalHeaderSize = 1;
+constexpr size_t kFuAHeaderSize = 2;
+constexpr size_t kLengthFieldSize = 2;
+constexpr size_t kStapAHeaderSize = kNalHeaderSize + kLengthFieldSize;
+
+// TODO(pbos): Avoid parsing this here as well as inside the jitter buffer.
+bool ParseStapAStartOffsets(const uint8_t* nalu_ptr,
+                            size_t length_remaining,
+                            std::vector<size_t>* offsets) {
+  size_t offset = 0;
+  while (length_remaining > 0) {
+    // Buffer doesn't contain room for additional nalu length.
+    if (length_remaining < sizeof(uint16_t))
+      return false;
+    uint16_t nalu_size = ByteReader<uint16_t>::ReadBigEndian(nalu_ptr);
+    nalu_ptr += sizeof(uint16_t);
+    length_remaining -= sizeof(uint16_t);
+    if (nalu_size > length_remaining)
+      return false;
+    nalu_ptr += nalu_size;
+    length_remaining -= nalu_size;
+
+    offsets->push_back(offset + kStapAHeaderSize);
+    offset += kLengthFieldSize + nalu_size;
+  }
+  return true;
+}
+
+absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
+    rtc::CopyOnWriteBuffer rtp_payload) {
+  const uint8_t* const payload_data = rtp_payload.cdata();
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
+      absl::in_place);
+  bool modified_buffer = false;
+  parsed_payload->video_payload = rtp_payload;
+  parsed_payload->video_header.width = 0;
+  parsed_payload->video_header.height = 0;
+  parsed_payload->video_header.codec = kVideoCodecH264;
+  parsed_payload->video_header.simulcastIdx = 0;
+  parsed_payload->video_header.is_first_packet_in_frame = true;
+  auto& h264_header = parsed_payload->video_header.video_type_header
+                          .emplace<RTPVideoHeaderH264>();
+
+  const uint8_t* nalu_start = payload_data + kNalHeaderSize;
+  const size_t nalu_length = rtp_payload.size() - kNalHeaderSize;
+  uint8_t nal_type = payload_data[0] & kH264TypeMask;
+  std::vector<size_t> nalu_start_offsets;
+  if (nal_type == H264::NaluType::kStapA) {
+    // Skip the StapA header (StapA NAL type + length).
+    if (rtp_payload.size() <= kStapAHeaderSize) {
+      RTC_LOG(LS_ERROR) << "StapA header truncated.";
+      return absl::nullopt;
+    }
+
+    if (!ParseStapAStartOffsets(nalu_start, nalu_length, &nalu_start_offsets)) {
+      RTC_LOG(LS_ERROR) << "StapA packet with incorrect NALU packet lengths.";
+      return absl::nullopt;
+    }
+
+    h264_header.packetization_type = kH264StapA;
+    nal_type = payload_data[kStapAHeaderSize] & kH264TypeMask;
+  } else {
+    h264_header.packetization_type = kH264SingleNalu;
+    nalu_start_offsets.push_back(0);
+  }
+  h264_header.nalu_type = nal_type;
+  parsed_payload->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+
+  nalu_start_offsets.push_back(rtp_payload.size() +
+                               kLengthFieldSize);  // End offset.
+  for (size_t i = 0; i < nalu_start_offsets.size() - 1; ++i) {
+    size_t start_offset = nalu_start_offsets[i];
+    // End offset is actually start offset for next unit, excluding length field
+    // so remove that from this units length.
+    size_t end_offset = nalu_start_offsets[i + 1] - kLengthFieldSize;
+    if (end_offset - start_offset < H264::kNaluTypeSize) {
+      RTC_LOG(LS_ERROR) << "STAP-A packet too short";
+      return absl::nullopt;
+    }
+
+    NaluInfo nalu;
+    nalu.type = payload_data[start_offset] & kH264TypeMask;
+    nalu.sps_id = -1;
+    nalu.pps_id = -1;
+    start_offset += H264::kNaluTypeSize;
+
+    switch (nalu.type) {
+      case H264::NaluType::kSps: {
+        // Check if VUI is present in SPS and if it needs to be modified to
+        // avoid
+        // excessive decoder latency.
+
+        // Copy any previous data first (likely just the first header).
+        rtc::Buffer output_buffer;
+        if (start_offset)
+          output_buffer.AppendData(payload_data, start_offset);
+
+        absl::optional<SpsParser::SpsState> sps;
+
+        SpsVuiRewriter::ParseResult result = SpsVuiRewriter::ParseAndRewriteSps(
+            &payload_data[start_offset], end_offset - start_offset, &sps,
+            nullptr, &output_buffer, SpsVuiRewriter::Direction::kIncoming);
+
+        if (result == SpsVuiRewriter::ParseResult::kVuiRewritten) {
+          if (modified_buffer) {
+            RTC_LOG(LS_WARNING)
+                << "More than one H264 SPS NAL units needing "
+                   "rewriting found within a single STAP-A packet. "
+                   "Keeping the first and rewriting the last.";
+          }
+
+          // Rewrite length field to new SPS size.
+          if (h264_header.packetization_type == kH264StapA) {
+            size_t length_field_offset =
+                start_offset - (H264::kNaluTypeSize + kLengthFieldSize);
+            // Stap-A Length includes payload data and type header.
+            size_t rewritten_size =
+                output_buffer.size() - start_offset + H264::kNaluTypeSize;
+            ByteWriter<uint16_t>::WriteBigEndian(
+                &output_buffer[length_field_offset], rewritten_size);
+          }
+
+          parsed_payload->video_payload.SetData(output_buffer.data(),
+                                                output_buffer.size());
+          // Append rest of packet.
+          parsed_payload->video_payload.AppendData(
+              &payload_data[end_offset],
+              nalu_length + kNalHeaderSize - end_offset);
+
+          modified_buffer = true;
+        }
+
+        if (sps) {
+          parsed_payload->video_header.width = sps->width;
+          parsed_payload->video_header.height = sps->height;
+          nalu.sps_id = sps->id;
+        } else {
+          RTC_LOG(LS_WARNING) << "Failed to parse SPS id from SPS slice.";
+        }
+        parsed_payload->video_header.frame_type =
+            VideoFrameType::kVideoFrameKey;
+        break;
+      }
+      case H264::NaluType::kPps: {
+        uint32_t pps_id;
+        uint32_t sps_id;
+        if (PpsParser::ParsePpsIds(&payload_data[start_offset],
+                                   end_offset - start_offset, &pps_id,
+                                   &sps_id)) {
+          nalu.pps_id = pps_id;
+          nalu.sps_id = sps_id;
+        } else {
+          RTC_LOG(LS_WARNING)
+              << "Failed to parse PPS id and SPS id from PPS slice.";
+        }
+        break;
+      }
+      case H264::NaluType::kIdr:
+        parsed_payload->video_header.frame_type =
+            VideoFrameType::kVideoFrameKey;
+        [[fallthrough]];
+      case H264::NaluType::kSlice: {
+        absl::optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
+            &payload_data[start_offset], end_offset - start_offset);
+        if (pps_id) {
+          nalu.pps_id = *pps_id;
+        } else {
+          RTC_LOG(LS_WARNING) << "Failed to parse PPS id from slice of type: "
+                              << static_cast<int>(nalu.type);
+        }
+        break;
+      }
+      // Slices below don't contain SPS or PPS ids.
+      case H264::NaluType::kAud:
+      case H264::NaluType::kEndOfSequence:
+      case H264::NaluType::kEndOfStream:
+      case H264::NaluType::kFiller:
+      case H264::NaluType::kSei:
+        break;
+      case H264::NaluType::kStapA:
+      case H264::NaluType::kFuA:
+        RTC_LOG(LS_WARNING) << "Unexpected STAP-A or FU-A received.";
+        return absl::nullopt;
+    }
+
+    if (h264_header.nalus_length == kMaxNalusPerPacket) {
+      RTC_LOG(LS_WARNING)
+          << "Received packet containing more than " << kMaxNalusPerPacket
+          << " NAL units. Will not keep track sps and pps ids for all of them.";
+    } else {
+      h264_header.nalus[h264_header.nalus_length++] = nalu;
+    }
+  }
+
+  return parsed_payload;
+}
+
+absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuaNalu(
+    rtc::CopyOnWriteBuffer rtp_payload) {
+  if (rtp_payload.size() < kFuAHeaderSize) {
+    RTC_LOG(LS_ERROR) << "FU-A NAL units truncated.";
+    return absl::nullopt;
+  }
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload(
+      absl::in_place);
+  uint8_t fnri = rtp_payload.cdata()[0] & (kH264FBit | kH264NriMask);
+  uint8_t original_nal_type = rtp_payload.cdata()[1] & kH264TypeMask;
+  bool first_fragment = (rtp_payload.cdata()[1] & kH264SBit) > 0;
+  NaluInfo nalu;
+  nalu.type = original_nal_type;
+  nalu.sps_id = -1;
+  nalu.pps_id = -1;
+  if (first_fragment) {
+    absl::optional<uint32_t> pps_id =
+        PpsParser::ParsePpsIdFromSlice(rtp_payload.cdata() + 2 * kNalHeaderSize,
+                                       rtp_payload.size() - 2 * kNalHeaderSize);
+    if (pps_id) {
+      nalu.pps_id = *pps_id;
+    } else {
+      RTC_LOG(LS_WARNING)
+          << "Failed to parse PPS from first fragment of FU-A NAL "
+             "unit with original type: "
+          << static_cast<int>(nalu.type);
+    }
+    uint8_t original_nal_header = fnri | original_nal_type;
+    rtp_payload =
+        rtp_payload.Slice(kNalHeaderSize, rtp_payload.size() - kNalHeaderSize);
+    rtp_payload.MutableData()[0] = original_nal_header;
+    parsed_payload->video_payload = std::move(rtp_payload);
+  } else {
+    parsed_payload->video_payload =
+        rtp_payload.Slice(kFuAHeaderSize, rtp_payload.size() - kFuAHeaderSize);
+  }
+
+  if (original_nal_type == H264::NaluType::kIdr) {
+    parsed_payload->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  } else {
+    parsed_payload->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  }
+  parsed_payload->video_header.width = 0;
+  parsed_payload->video_header.height = 0;
+  parsed_payload->video_header.codec = kVideoCodecH264;
+  parsed_payload->video_header.simulcastIdx = 0;
+  parsed_payload->video_header.is_first_packet_in_frame = first_fragment;
+  auto& h264_header = parsed_payload->video_header.video_type_header
+                          .emplace<RTPVideoHeaderH264>();
+  h264_header.packetization_type = kH264FuA;
+  h264_header.nalu_type = original_nal_type;
+  if (first_fragment) {
+    h264_header.nalus[h264_header.nalus_length] = nalu;
+    h264_header.nalus_length = 1;
+  }
+  return parsed_payload;
+}
+
+}  // namespace
+
+absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+VideoRtpDepacketizerH264::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
+  if (rtp_payload.size() == 0) {
+    RTC_LOG(LS_ERROR) << "Empty payload.";
+    return absl::nullopt;
+  }
+
+  uint8_t nal_type = rtp_payload.cdata()[0] & kH264TypeMask;
+
+  if (nal_type == H264::NaluType::kFuA) {
+    // Fragmented NAL units (FU-A).
+    return ParseFuaNalu(std::move(rtp_payload));
+  } else {
+    // We handle STAP-A and single NALU's the same way here. The jitter buffer
+    // will depacketize the STAP-A into NAL units later.
+    return ProcessStapAOrSingleNalu(std::move(rtp_payload));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h
new file mode 100644
index 0000000000..cbea860049
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+class VideoRtpDepacketizerH264 : public VideoRtpDepacketizer {
+ public:
+  ~VideoRtpDepacketizerH264() override = default;
+
+  absl::optional<ParsedRtpPayload> Parse(
+      rtc::CopyOnWriteBuffer rtp_payload) override;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_H264_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc
new file mode 100644
index 0000000000..f569c45fd3
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264_unittest.cc
@@ -0,0 +1,425 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "common_video/h264/h264_common.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_format_h264.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Each;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::IsEmpty;
+using ::testing::SizeIs;
+
+enum Nalu {
+  kSlice = 1,
+  kIdr = 5,
+  kSei = 6,
+  kSps = 7,
+  kPps = 8,
+  kStapA = 24,
+  kFuA = 28
+};
+
+constexpr uint8_t kOriginalSps[] = {kSps, 0x00, 0x00, 0x03, 0x03,
+                                    0xF4, 0x05, 0x03, 0xC7, 0xC0};
+constexpr uint8_t kRewrittenSps[] = {kSps, 0x00, 0x00, 0x03, 0x03,
+                                     0xF4, 0x05, 0x03, 0xC7, 0xE0,
+                                     0x1B, 0x41, 0x10, 0x8D, 0x00};
+constexpr uint8_t kIdrOne[] = {kIdr, 0xFF, 0x00, 0x00, 0x04};
+constexpr uint8_t kIdrTwo[] = {kIdr, 0xFF, 0x00, 0x11};
+
+TEST(VideoRtpDepacketizerH264Test, SingleNalu) {
+  uint8_t packet[2] = {0x05, 0xFF};  // F=0, NRI=0, Type=5 (IDR).
+  rtc::CopyOnWriteBuffer rtp_payload(packet);
+
+  VideoRtpDepacketizerH264 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+  ASSERT_TRUE(parsed);
+
+  EXPECT_EQ(parsed->video_payload, rtp_payload);
+  EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameKey);
+  EXPECT_EQ(parsed->video_header.codec, kVideoCodecH264);
+  EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
+  const RTPVideoHeaderH264& h264 =
+      absl::get<RTPVideoHeaderH264>(parsed->video_header.video_type_header);
+  EXPECT_EQ(h264.packetization_type, kH264SingleNalu);
+  EXPECT_EQ(h264.nalu_type, kIdr);
+}
+
+TEST(VideoRtpDepacketizerH264Test, SingleNaluSpsWithResolution) {
+  uint8_t packet[] = {kSps, 0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40, 0x50,
+                      0x05, 0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0,
+                      0x00, 0x00, 0x03, 0x2A, 0xE0, 0xF1, 0x83, 0x25};
+  rtc::CopyOnWriteBuffer rtp_payload(packet);
+
+  VideoRtpDepacketizerH264 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+  ASSERT_TRUE(parsed);
+
+  EXPECT_EQ(parsed->video_payload, rtp_payload);
+  EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameKey);
+  EXPECT_EQ(parsed->video_header.codec, kVideoCodecH264);
+  EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
+  EXPECT_EQ(parsed->video_header.width, 1280u);
+  EXPECT_EQ(parsed->video_header.height, 720u);
+  const auto& h264 =
+      absl::get<RTPVideoHeaderH264>(parsed->video_header.video_type_header);
+  EXPECT_EQ(h264.packetization_type, kH264SingleNalu);
+}
+
+TEST(VideoRtpDepacketizerH264Test, StapAKey) {
+  // clang-format off
+  const NaluInfo kExpectedNalus[] = { {H264::kSps, 0, -1},
+                                      {H264::kPps, 1, 2},
+                                      {H264::kIdr, -1, 0} };
+  uint8_t packet[] = {kStapA,  // F=0, NRI=0, Type=24.
+                      // Length, nal header, payload.
+                      0, 0x18, kExpectedNalus[0].type,
+                        0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40, 0x50, 0x05, 0xBA,
+                        0x10, 0x00, 0x00, 0x03, 0x00, 0xC0, 0x00, 0x00, 0x03,
+                        0x2A, 0xE0, 0xF1, 0x83, 0x25,
+                      0, 0xD, kExpectedNalus[1].type,
+                        0x69, 0xFC, 0x0, 0x0, 0x3, 0x0, 0x7, 0xFF, 0xFF, 0xFF,
+                        0xF6, 0x40,
+                      0, 0xB, kExpectedNalus[2].type,
+                        0x85, 0xB8, 0x0, 0x4, 0x0, 0x0, 0x13, 0x93, 0x12, 0x0};
+  // clang-format on
+  rtc::CopyOnWriteBuffer rtp_payload(packet);
+
+  VideoRtpDepacketizerH264 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+  ASSERT_TRUE(parsed);
+
+  EXPECT_EQ(parsed->video_payload, rtp_payload);
+  EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameKey);
+  EXPECT_EQ(parsed->video_header.codec, kVideoCodecH264);
+  EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
+  const auto& h264 =
+      absl::get<RTPVideoHeaderH264>(parsed->video_header.video_type_header);
+  EXPECT_EQ(h264.packetization_type, kH264StapA);
+  // NALU type for aggregated packets is the type of the first packet only.
+  EXPECT_EQ(h264.nalu_type, kSps);
+  ASSERT_EQ(h264.nalus_length, 3u);
+  for (size_t i = 0; i < h264.nalus_length; ++i) {
+    EXPECT_EQ(h264.nalus[i].type, kExpectedNalus[i].type)
+        << "Failed parsing nalu " << i;
+    EXPECT_EQ(h264.nalus[i].sps_id, kExpectedNalus[i].sps_id)
+        << "Failed parsing nalu " << i;
+    EXPECT_EQ(h264.nalus[i].pps_id, kExpectedNalus[i].pps_id)
+        << "Failed parsing nalu " << i;
+  }
+}
+
+TEST(VideoRtpDepacketizerH264Test, StapANaluSpsWithResolution) {
+  uint8_t packet[] = {kStapA,  // F=0, NRI=0, Type=24.
+                               // Length (2 bytes), nal header, payload.
+                      0x00, 0x19, kSps, 0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40,
+                      0x50, 0x05, 0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0,
+                      0x00, 0x00, 0x03, 0x2A, 0xE0, 0xF1, 0x83, 0x25, 0x80,
+                      0x00, 0x03, kIdr, 0xFF, 0x00, 0x00, 0x04, kIdr, 0xFF,
+                      0x00, 0x11};
+  rtc::CopyOnWriteBuffer rtp_payload(packet);
+
+  VideoRtpDepacketizerH264 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+  ASSERT_TRUE(parsed);
+
+  EXPECT_EQ(parsed->video_payload, rtp_payload);
+  EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameKey);
+  EXPECT_EQ(parsed->video_header.codec, kVideoCodecH264);
+  EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
+  EXPECT_EQ(parsed->video_header.width, 1280u);
+  EXPECT_EQ(parsed->video_header.height, 720u);
+  const auto& h264 =
+      absl::get<RTPVideoHeaderH264>(parsed->video_header.video_type_header);
+  EXPECT_EQ(h264.packetization_type, kH264StapA);
+}
+
+TEST(VideoRtpDepacketizerH264Test, EmptyStapARejected) {
+  uint8_t lone_empty_packet[] = {kStapA, 0x00, 0x00};
+  uint8_t leading_empty_packet[] = {kStapA, 0x00, 0x00, 0x00, 0x04,
+                                    kIdr,   0xFF, 0x00, 0x11};
+  uint8_t middle_empty_packet[] = {kStapA, 0x00, 0x03, kIdr, 0xFF, 0x00, 0x00,
+                                   0x00,   0x00, 0x04, kIdr, 0xFF, 0x00, 0x11};
+  uint8_t trailing_empty_packet[] = {kStapA, 0x00, 0x03, kIdr,
+                                     0xFF,   0x00, 0x00, 0x00};
+
+  VideoRtpDepacketizerH264 depacketizer;
+  EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(lone_empty_packet)));
+  EXPECT_FALSE(
+      depacketizer.Parse(rtc::CopyOnWriteBuffer(leading_empty_packet)));
+  EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(middle_empty_packet)));
+  EXPECT_FALSE(
+      depacketizer.Parse(rtc::CopyOnWriteBuffer(trailing_empty_packet)));
+}
+
+TEST(VideoRtpDepacketizerH264Test, DepacketizeWithRewriting) {
+  rtc::CopyOnWriteBuffer in_buffer;
+  rtc::Buffer out_buffer;
+
+  uint8_t kHeader[2] = {kStapA};
+  in_buffer.AppendData(kHeader, 1);
+  out_buffer.AppendData(kHeader, 1);
+
+  ByteWriter<uint16_t>::WriteBigEndian(kHeader, sizeof(kOriginalSps));
+  in_buffer.AppendData(kHeader, 2);
+  in_buffer.AppendData(kOriginalSps);
+  ByteWriter<uint16_t>::WriteBigEndian(kHeader, sizeof(kRewrittenSps));
+  out_buffer.AppendData(kHeader, 2);
+  out_buffer.AppendData(kRewrittenSps);
+
+  ByteWriter<uint16_t>::WriteBigEndian(kHeader, sizeof(kIdrOne));
+  in_buffer.AppendData(kHeader, 2);
+  in_buffer.AppendData(kIdrOne);
+  out_buffer.AppendData(kHeader, 2);
+  out_buffer.AppendData(kIdrOne);
+
+  ByteWriter<uint16_t>::WriteBigEndian(kHeader, sizeof(kIdrTwo));
+  in_buffer.AppendData(kHeader, 2);
+  in_buffer.AppendData(kIdrTwo);
+  out_buffer.AppendData(kHeader, 2);
+  out_buffer.AppendData(kIdrTwo);
+
+  VideoRtpDepacketizerH264 depacketizer;
+  auto parsed = depacketizer.Parse(in_buffer);
+  ASSERT_TRUE(parsed);
+  EXPECT_THAT(rtc::MakeArrayView(parsed->video_payload.cdata(),
+                                 parsed->video_payload.size()),
+              ElementsAreArray(out_buffer));
+}
+
+TEST(VideoRtpDepacketizerH264Test, DepacketizeWithDoubleRewriting) {
+  rtc::CopyOnWriteBuffer in_buffer;
+  rtc::Buffer out_buffer;
+
+  uint8_t kHeader[2] = {kStapA};
+  in_buffer.AppendData(kHeader, 1);
+  out_buffer.AppendData(kHeader, 1);
+
+  // First SPS will be kept...
+  ByteWriter<uint16_t>::WriteBigEndian(kHeader, sizeof(kOriginalSps));
+  in_buffer.AppendData(kHeader, 2);
+  in_buffer.AppendData(kOriginalSps);
+  out_buffer.AppendData(kHeader, 2);
+  out_buffer.AppendData(kOriginalSps);
+
+  // ...only the second one will be rewritten.
+  ByteWriter<uint16_t>::WriteBigEndian(kHeader, sizeof(kOriginalSps));
+  in_buffer.AppendData(kHeader, 2);
+  in_buffer.AppendData(kOriginalSps);
+  ByteWriter<uint16_t>::WriteBigEndian(kHeader, sizeof(kRewrittenSps));
+  out_buffer.AppendData(kHeader, 2);
+  out_buffer.AppendData(kRewrittenSps);
+
+  ByteWriter<uint16_t>::WriteBigEndian(kHeader, sizeof(kIdrOne));
+  in_buffer.AppendData(kHeader, 2);
+  in_buffer.AppendData(kIdrOne);
+  out_buffer.AppendData(kHeader, 2);
+  out_buffer.AppendData(kIdrOne);
+
+  ByteWriter<uint16_t>::WriteBigEndian(kHeader, sizeof(kIdrTwo));
+  in_buffer.AppendData(kHeader, 2);
+  in_buffer.AppendData(kIdrTwo);
+  out_buffer.AppendData(kHeader, 2);
+  out_buffer.AppendData(kIdrTwo);
+
+  VideoRtpDepacketizerH264 depacketizer;
+  auto parsed = depacketizer.Parse(in_buffer);
+  ASSERT_TRUE(parsed);
+  std::vector<uint8_t> expected_packet_payload(
+      out_buffer.data(), &out_buffer.data()[out_buffer.size()]);
+  EXPECT_THAT(rtc::MakeArrayView(parsed->video_payload.cdata(),
+                                 parsed->video_payload.size()),
+              ElementsAreArray(out_buffer));
+}
+
+TEST(VideoRtpDepacketizerH264Test, StapADelta) {
+  uint8_t packet[16] = {kStapA,  // F=0, NRI=0, Type=24.
+                                 // Length, nal header, payload.
+                        0, 0x02, kSlice, 0xFF, 0, 0x03, kSlice, 0xFF, 0x00, 0,
+                        0x04, kSlice, 0xFF, 0x00, 0x11};
+  rtc::CopyOnWriteBuffer rtp_payload(packet);
+
+  VideoRtpDepacketizerH264 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+  ASSERT_TRUE(parsed);
+
+  EXPECT_EQ(parsed->video_payload.size(), rtp_payload.size());
+  EXPECT_EQ(parsed->video_payload.cdata(), rtp_payload.cdata());
+
+  EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameDelta);
+  EXPECT_EQ(parsed->video_header.codec, kVideoCodecH264);
+  EXPECT_TRUE(parsed->video_header.is_first_packet_in_frame);
+  const RTPVideoHeaderH264& h264 =
+      absl::get<RTPVideoHeaderH264>(parsed->video_header.video_type_header);
+  EXPECT_EQ(h264.packetization_type, kH264StapA);
+  // NALU type for aggregated packets is the type of the first packet only.
+  EXPECT_EQ(h264.nalu_type, kSlice);
+}
+
+TEST(VideoRtpDepacketizerH264Test, FuA) {
+  // clang-format off
+  uint8_t packet1[] = {
+      kFuA,              // F=0, NRI=0, Type=28.
+      kH264SBit | kIdr,  // FU header.
+      0x85, 0xB8, 0x0, 0x4, 0x0, 0x0, 0x13, 0x93, 0x12, 0x0  // Payload.
+  };
+  // clang-format on
+  const uint8_t kExpected1[] = {kIdr, 0x85, 0xB8, 0x0,  0x4, 0x0,
+                                0x0,  0x13, 0x93, 0x12, 0x0};
+
+  uint8_t packet2[] = {
+      kFuA,  // F=0, NRI=0, Type=28.
+      kIdr,  // FU header.
+      0x02   // Payload.
+  };
+  const uint8_t kExpected2[] = {0x02};
+
+  uint8_t packet3[] = {
+      kFuA,              // F=0, NRI=0, Type=28.
+      kH264EBit | kIdr,  // FU header.
+      0x03               // Payload.
+  };
+  const uint8_t kExpected3[] = {0x03};
+
+  VideoRtpDepacketizerH264 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed1 =
+      depacketizer.Parse(rtc::CopyOnWriteBuffer(packet1));
+  ASSERT_TRUE(parsed1);
+  // We expect that the first packet is one byte shorter since the FU-A header
+  // has been replaced by the original nal header.
+  EXPECT_THAT(rtc::MakeArrayView(parsed1->video_payload.cdata(),
+                                 parsed1->video_payload.size()),
+              ElementsAreArray(kExpected1));
+  EXPECT_EQ(parsed1->video_header.frame_type, VideoFrameType::kVideoFrameKey);
+  EXPECT_EQ(parsed1->video_header.codec, kVideoCodecH264);
+  EXPECT_TRUE(parsed1->video_header.is_first_packet_in_frame);
+  {
+    const RTPVideoHeaderH264& h264 =
+        absl::get<RTPVideoHeaderH264>(parsed1->video_header.video_type_header);
+    EXPECT_EQ(h264.packetization_type, kH264FuA);
+    EXPECT_EQ(h264.nalu_type, kIdr);
+    ASSERT_EQ(h264.nalus_length, 1u);
+    EXPECT_EQ(h264.nalus[0].type, static_cast<H264::NaluType>(kIdr));
+    EXPECT_EQ(h264.nalus[0].sps_id, -1);
+    EXPECT_EQ(h264.nalus[0].pps_id, 0);
+  }
+
+  // Following packets will be 2 bytes shorter since they will only be appended
+  // onto the first packet.
+  auto parsed2 = depacketizer.Parse(rtc::CopyOnWriteBuffer(packet2));
+  EXPECT_THAT(rtc::MakeArrayView(parsed2->video_payload.cdata(),
+                                 parsed2->video_payload.size()),
+              ElementsAreArray(kExpected2));
+  EXPECT_FALSE(parsed2->video_header.is_first_packet_in_frame);
+  EXPECT_EQ(parsed2->video_header.codec, kVideoCodecH264);
+  {
+    const RTPVideoHeaderH264& h264 =
+        absl::get<RTPVideoHeaderH264>(parsed2->video_header.video_type_header);
+    EXPECT_EQ(h264.packetization_type, kH264FuA);
+    EXPECT_EQ(h264.nalu_type, kIdr);
+    // NALU info is only expected for the first FU-A packet.
+    EXPECT_EQ(h264.nalus_length, 0u);
+  }
+
+  auto parsed3 = depacketizer.Parse(rtc::CopyOnWriteBuffer(packet3));
+  EXPECT_THAT(rtc::MakeArrayView(parsed3->video_payload.cdata(),
+                                 parsed3->video_payload.size()),
+              ElementsAreArray(kExpected3));
+  EXPECT_FALSE(parsed3->video_header.is_first_packet_in_frame);
+  EXPECT_EQ(parsed3->video_header.codec, kVideoCodecH264);
+  {
+    const RTPVideoHeaderH264& h264 =
+        absl::get<RTPVideoHeaderH264>(parsed3->video_header.video_type_header);
+    EXPECT_EQ(h264.packetization_type, kH264FuA);
+    EXPECT_EQ(h264.nalu_type, kIdr);
+    // NALU info is only expected for the first FU-A packet.
+    ASSERT_EQ(h264.nalus_length, 0u);
+  }
+}
+
+TEST(VideoRtpDepacketizerH264Test, EmptyPayload) {
+  rtc::CopyOnWriteBuffer empty;
+  VideoRtpDepacketizerH264 depacketizer;
+  EXPECT_FALSE(depacketizer.Parse(empty));
+}
+
+TEST(VideoRtpDepacketizerH264Test, TruncatedFuaNalu) {
+  const uint8_t kPayload[] = {0x9c};
+  VideoRtpDepacketizerH264 depacketizer;
+  EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
+}
+
+TEST(VideoRtpDepacketizerH264Test, TruncatedSingleStapANalu) {
+  const uint8_t kPayload[] = {0xd8, 0x27};
+  VideoRtpDepacketizerH264 depacketizer;
+  EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
+}
+
+TEST(VideoRtpDepacketizerH264Test, StapAPacketWithTruncatedNalUnits) {
+  const uint8_t kPayload[] = {0x58, 0xCB, 0xED, 0xDF};
+  VideoRtpDepacketizerH264 depacketizer;
+  EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
+}
+
+TEST(VideoRtpDepacketizerH264Test, TruncationJustAfterSingleStapANalu) {
+  const uint8_t kPayload[] = {0x38, 0x27, 0x27};
+  VideoRtpDepacketizerH264 depacketizer;
+  EXPECT_FALSE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
+}
+
+TEST(VideoRtpDepacketizerH264Test, ShortSpsPacket) {
+  const uint8_t kPayload[] = {0x27, 0x80, 0x00};
+  VideoRtpDepacketizerH264 depacketizer;
+  EXPECT_TRUE(depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload)));
+}
+
+TEST(VideoRtpDepacketizerH264Test, SeiPacket) {
+  const uint8_t kPayload[] = {
+      kSei,                   // F=0, NRI=0, Type=6.
+      0x03, 0x03, 0x03, 0x03  // Payload.
+  };
+  VideoRtpDepacketizerH264 depacketizer;
+  auto parsed = depacketizer.Parse(rtc::CopyOnWriteBuffer(kPayload));
+  ASSERT_TRUE(parsed);
+  const RTPVideoHeaderH264& h264 =
+      absl::get<RTPVideoHeaderH264>(parsed->video_header.video_type_header);
+  EXPECT_EQ(parsed->video_header.frame_type, VideoFrameType::kVideoFrameDelta);
+  EXPECT_EQ(h264.packetization_type, kH264SingleNalu);
+  EXPECT_EQ(h264.nalu_type, kSei);
+  ASSERT_EQ(h264.nalus_length, 1u);
+  EXPECT_EQ(h264.nalus[0].type, static_cast<H264::NaluType>(kSei));
+  EXPECT_EQ(h264.nalus[0].sps_id, -1);
+  EXPECT_EQ(h264.nalus[0].pps_id, -1);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc
new file mode 100644
index 0000000000..81b4e4ab53
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.cc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
+
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+VideoRtpDepacketizerRaw::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
+  absl::optional<ParsedRtpPayload> parsed(absl::in_place);
+  parsed->video_payload = std::move(rtp_payload);
+  return parsed;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h
new file mode 100644
index 0000000000..59c8695352
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_RAW_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_RAW_H_
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class VideoRtpDepacketizerRaw : public VideoRtpDepacketizer {
+ public:
+  ~VideoRtpDepacketizerRaw() override = default;
+
+  absl::optional<ParsedRtpPayload> Parse(
+      rtc::CopyOnWriteBuffer rtp_payload) override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_RAW_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw_unittest.cc
new file mode 100644
index 0000000000..36c826ab84
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_raw_unittest.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
+
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+TEST(VideoRtpDepacketizerRaw, PassRtpPayloadAsVideoPayload) {
+  const uint8_t kPayload[] = {0x05, 0x25, 0x52};
+  rtc::CopyOnWriteBuffer rtp_payload(kPayload);
+
+  VideoRtpDepacketizerRaw depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+
+  ASSERT_TRUE(parsed);
+  EXPECT_EQ(parsed->video_payload.size(), rtp_payload.size());
+  // Check there was no memcpy involved by verifying return and original buffers
+  // point to the same buffer.
+  EXPECT_EQ(parsed->video_payload.cdata(), rtp_payload.cdata());
+}
+
+TEST(VideoRtpDepacketizerRaw, UsesDefaultValuesForVideoHeader) {
+  const uint8_t kPayload[] = {0x05, 0x25, 0x52};
+  rtc::CopyOnWriteBuffer rtp_payload(kPayload);
+
+  VideoRtpDepacketizerRaw depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+
+  ASSERT_TRUE(parsed);
+  EXPECT_FALSE(parsed->video_header.generic);
+  EXPECT_EQ(parsed->video_header.codec, kVideoCodecGeneric);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
new file mode 100644
index 0000000000..d6bd33c24d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.cc
@@ -0,0 +1,201 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+// VP8 payload descriptor
+// https://datatracker.ietf.org/doc/html/rfc7741#section-4.2
+//
+//       0 1 2 3 4 5 6 7
+//      +-+-+-+-+-+-+-+-+
+//      |X|R|N|S|R| PID | (REQUIRED)
+//      +-+-+-+-+-+-+-+-+
+// X:   |I|L|T|K| RSV   | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+// I:   |M| PictureID   | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+//      |   PictureID   |
+//      +-+-+-+-+-+-+-+-+
+// L:   |   TL0PICIDX   | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+// T/K: |TID|Y| KEYIDX  | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+//
+// VP8 payload header. Considered part of the actual payload, sent to decoder.
+// https://datatracker.ietf.org/doc/html/rfc7741#section-4.3
+//
+//       0 1 2 3 4 5 6 7
+//      +-+-+-+-+-+-+-+-+
+//      |Size0|H| VER |P|
+//      +-+-+-+-+-+-+-+-+
+//      :      ...      :
+//      +-+-+-+-+-+-+-+-+
+
+namespace webrtc {
+namespace {
+
+constexpr int kFailedToParse = 0;
+
+int ParseVP8Descriptor(RTPVideoHeaderVP8* vp8,
+                       const uint8_t* data,
+                       size_t data_length) {
+  RTC_DCHECK_GT(data_length, 0);
+  int parsed_bytes = 0;
+  // Parse mandatory first byte of payload descriptor.
+  bool extension = (*data & 0x80) ? true : false;             // X bit
+  vp8->nonReference = (*data & 0x20) ? true : false;          // N bit
+  vp8->beginningOfPartition = (*data & 0x10) ? true : false;  // S bit
+  vp8->partitionId = (*data & 0x07);                          // PID field
+
+  data++;
+  parsed_bytes++;
+  data_length--;
+
+  if (!extension)
+    return parsed_bytes;
+
+  if (data_length == 0)
+    return kFailedToParse;
+  // Optional X field is present.
+  bool has_picture_id = (*data & 0x80) ? true : false;   // I bit
+  bool has_tl0_pic_idx = (*data & 0x40) ? true : false;  // L bit
+  bool has_tid = (*data & 0x20) ? true : false;          // T bit
+  bool has_key_idx = (*data & 0x10) ? true : false;      // K bit
+
+  // Advance data and decrease remaining payload size.
+  data++;
+  parsed_bytes++;
+  data_length--;
+
+  if (has_picture_id) {
+    if (data_length == 0)
+      return kFailedToParse;
+
+    vp8->pictureId = (*data & 0x7F);
+    if (*data & 0x80) {
+      data++;
+      parsed_bytes++;
+      if (--data_length == 0)
+        return kFailedToParse;
+      // PictureId is 15 bits
+      vp8->pictureId = (vp8->pictureId << 8) + *data;
+    }
+    data++;
+    parsed_bytes++;
+    data_length--;
+  }
+
+  if (has_tl0_pic_idx) {
+    if (data_length == 0)
+      return kFailedToParse;
+
+    vp8->tl0PicIdx = *data;
+    data++;
+    parsed_bytes++;
+    data_length--;
+  }
+
+  if (has_tid || has_key_idx) {
+    if (data_length == 0)
+      return kFailedToParse;
+
+    if (has_tid) {
+      vp8->temporalIdx = ((*data >> 6) & 0x03);
+      vp8->layerSync = (*data & 0x20) ? true : false;  // Y bit
+    }
+    if (has_key_idx) {
+      vp8->keyIdx = *data & 0x1F;
+    }
+    data++;
+    parsed_bytes++;
+    data_length--;
+  }
+  return parsed_bytes;
+}
+
+}  // namespace
+
+absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+VideoRtpDepacketizerVp8::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
+  rtc::ArrayView<const uint8_t> payload(rtp_payload.cdata(),
+                                        rtp_payload.size());
+  absl::optional<ParsedRtpPayload> result(absl::in_place);
+  int offset = ParseRtpPayload(payload, &result->video_header);
+  if (offset == kFailedToParse)
+    return absl::nullopt;
+  RTC_DCHECK_LT(offset, rtp_payload.size());
+  result->video_payload =
+      rtp_payload.Slice(offset, rtp_payload.size() - offset);
+  return result;
+}
+
+int VideoRtpDepacketizerVp8::ParseRtpPayload(
+    rtc::ArrayView<const uint8_t> rtp_payload,
+    RTPVideoHeader* video_header) {
+  RTC_DCHECK(video_header);
+  if (rtp_payload.empty()) {
+    RTC_LOG(LS_ERROR) << "Empty rtp payload.";
+    return kFailedToParse;
+  }
+
+  video_header->simulcastIdx = 0;
+  video_header->codec = kVideoCodecVP8;
+  auto& vp8_header =
+      video_header->video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.InitRTPVideoHeaderVP8();
+
+  const int descriptor_size =
+      ParseVP8Descriptor(&vp8_header, rtp_payload.data(), rtp_payload.size());
+  if (descriptor_size == kFailedToParse)
+    return kFailedToParse;
+
+  RTC_DCHECK_LT(vp8_header.partitionId, 8);
+
+  video_header->is_first_packet_in_frame =
+      vp8_header.beginningOfPartition && vp8_header.partitionId == 0;
+
+  int vp8_payload_size = rtp_payload.size() - descriptor_size;
+  if (vp8_payload_size == 0) {
+    RTC_LOG(LS_WARNING) << "Empty vp8 payload.";
+    return kFailedToParse;
+  }
+  const uint8_t* vp8_payload = rtp_payload.data() + descriptor_size;
+
+  // Read P bit from payload header (only at beginning of first partition).
+  if (video_header->is_first_packet_in_frame && (*vp8_payload & 0x01) == 0) {
+    video_header->frame_type = VideoFrameType::kVideoFrameKey;
+
+    if (vp8_payload_size < 10) {
+      // For an I-frame we should always have the uncompressed VP8 header
+      // in the beginning of the partition.
+      return kFailedToParse;
+    }
+    video_header->width = ((vp8_payload[7] << 8) + vp8_payload[6]) & 0x3FFF;
+    video_header->height = ((vp8_payload[9] << 8) + vp8_payload[8]) & 0x3FFF;
+  } else {
+    video_header->frame_type = VideoFrameType::kVideoFrameDelta;
+
+    video_header->width = 0;
+    video_header->height = 0;
+  }
+
+  return descriptor_size;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h
new file mode 100644
index 0000000000..3d7cb3291d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP8_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP8_H_
+
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class VideoRtpDepacketizerVp8 : public VideoRtpDepacketizer {
+ public:
+  VideoRtpDepacketizerVp8() = default;
+  VideoRtpDepacketizerVp8(const VideoRtpDepacketizerVp8&) = delete;
+  VideoRtpDepacketizerVp8& operator=(const VideoRtpDepacketizerVp8&) = delete;
+  ~VideoRtpDepacketizerVp8() override = default;
+
+  // Parses vp8 rtp payload descriptor.
+  // Returns zero on error or vp8 payload header offset on success.
+  static int ParseRtpPayload(rtc::ArrayView<const uint8_t> rtp_payload,
+                             RTPVideoHeader* video_header);
+
+  absl::optional<ParsedRtpPayload> Parse(
+      rtc::CopyOnWriteBuffer rtp_payload) override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP8_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc
new file mode 100644
index 0000000000..77469cf935
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc
@@ -0,0 +1,244 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+// VP8 payload descriptor
+// https://datatracker.ietf.org/doc/html/rfc7741#section-4.2
+//
+//       0 1 2 3 4 5 6 7
+//      +-+-+-+-+-+-+-+-+
+//      |X|R|N|S|R| PID | (REQUIRED)
+//      +-+-+-+-+-+-+-+-+
+// X:   |I|L|T|K| RSV   | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+// I:   |M| PictureID   | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+//      |   PictureID   |
+//      +-+-+-+-+-+-+-+-+
+// L:   |   TL0PICIDX   | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+// T/K: |TID|Y| KEYIDX  | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+
+//
+// VP8 payload header. Considered part of the actual payload, sent to decoder.
+// https://datatracker.ietf.org/doc/html/rfc7741#section-4.3
+//
+//       0 1 2 3 4 5 6 7
+//      +-+-+-+-+-+-+-+-+
+//      |Size0|H| VER |P|
+//      +-+-+-+-+-+-+-+-+
+//      :      ...      :
+//      +-+-+-+-+-+-+-+-+
+
+namespace webrtc {
+namespace {
+
+TEST(VideoRtpDepacketizerVp8Test, BasicHeader) {
+  uint8_t packet[4] = {0};
+  packet[0] = 0b0001'0100;  // S = 1, partition ID = 4.
+  packet[1] = 0x01;         // P frame.
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp8::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 1);
+  EXPECT_EQ(video_header.frame_type, VideoFrameType::kVideoFrameDelta);
+  EXPECT_EQ(video_header.codec, kVideoCodecVP8);
+  const auto& vp8_header =
+      absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
+  EXPECT_FALSE(vp8_header.nonReference);
+  EXPECT_TRUE(vp8_header.beginningOfPartition);
+  EXPECT_EQ(vp8_header.partitionId, 4);
+  EXPECT_EQ(vp8_header.pictureId, kNoPictureId);
+  EXPECT_EQ(vp8_header.tl0PicIdx, kNoTl0PicIdx);
+  EXPECT_EQ(vp8_header.temporalIdx, kNoTemporalIdx);
+  EXPECT_EQ(vp8_header.keyIdx, kNoKeyIdx);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, OneBytePictureID) {
+  const uint8_t kPictureId = 17;
+  uint8_t packet[10] = {0};
+  packet[0] = 0b1010'0000;
+  packet[1] = 0b1000'0000;
+  packet[2] = kPictureId;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp8::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 3);
+  const auto& vp8_header =
+      absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
+  EXPECT_EQ(vp8_header.pictureId, kPictureId);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, TwoBytePictureID) {
+  const uint16_t kPictureId = 0x1234;
+  uint8_t packet[10] = {0};
+  packet[0] = 0b1010'0000;
+  packet[1] = 0b1000'0000;
+  packet[2] = 0x80 | 0x12;
+  packet[3] = 0x34;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp8::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 4);
+  const auto& vp8_header =
+      absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
+  EXPECT_EQ(vp8_header.pictureId, kPictureId);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, Tl0PicIdx) {
+  const uint8_t kTl0PicIdx = 17;
+  uint8_t packet[13] = {0};
+  packet[0] = 0b1000'0000;
+  packet[1] = 0b0100'0000;
+  packet[2] = kTl0PicIdx;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp8::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 3);
+  const auto& vp8_header =
+      absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
+  EXPECT_EQ(vp8_header.tl0PicIdx, kTl0PicIdx);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, TIDAndLayerSync) {
+  uint8_t packet[10] = {0};
+  packet[0] = 0b1000'0000;
+  packet[1] = 0b0010'0000;
+  packet[2] = 0b10'0'00000;  // TID(2) + LayerSync(false)
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp8::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 3);
+  const auto& vp8_header =
+      absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
+  EXPECT_EQ(vp8_header.temporalIdx, 2);
+  EXPECT_FALSE(vp8_header.layerSync);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, KeyIdx) {
+  const uint8_t kKeyIdx = 17;
+  uint8_t packet[10] = {0};
+  packet[0] = 0b1000'0000;
+  packet[1] = 0b0001'0000;
+  packet[2] = kKeyIdx;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp8::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 3);
+  const auto& vp8_header =
+      absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
+  EXPECT_EQ(vp8_header.keyIdx, kKeyIdx);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, MultipleExtensions) {
+  uint8_t packet[10] = {0};
+  packet[0] = 0b1010'0110;  // X and N bit set, partition ID = 6
+  packet[1] = 0b1111'0000;
+  packet[2] = 0x80 | 0x12;  // PictureID, high 7 bits.
+  packet[3] = 0x34;         // PictureID, low 8 bits.
+  packet[4] = 42;           // Tl0PicIdx.
+  packet[5] = 0b01'1'10001;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp8::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 6);
+  const auto& vp8_header =
+      absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
+  EXPECT_TRUE(vp8_header.nonReference);
+  EXPECT_EQ(vp8_header.partitionId, 0b0110);
+  EXPECT_EQ(vp8_header.pictureId, 0x1234);
+  EXPECT_EQ(vp8_header.tl0PicIdx, 42);
+  EXPECT_EQ(vp8_header.temporalIdx, 1);
+  EXPECT_TRUE(vp8_header.layerSync);
+  EXPECT_EQ(vp8_header.keyIdx, 0b10001);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, TooShortHeader) {
+  uint8_t packet[4] = {0};
+  packet[0] = 0b1000'0000;
+  packet[1] = 0b1111'0000;  // All extensions are enabled...
+  packet[2] = 0x80 | 17;    // ... but only 2 bytes PictureID is provided.
+  packet[3] = 17;           // PictureID, low 8 bits.
+
+  RTPVideoHeader unused;
+  EXPECT_EQ(VideoRtpDepacketizerVp8::ParseRtpPayload(packet, &unused), 0);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, WithPacketizer) {
+  uint8_t data[10] = {0};
+  RtpPacketToSend packet(/*extenions=*/nullptr);
+  RTPVideoHeaderVP8 input_header;
+  input_header.nonReference = true;
+  input_header.pictureId = 300;
+  input_header.temporalIdx = 1;
+  input_header.layerSync = false;
+  input_header.tl0PicIdx = kNoTl0PicIdx;  // Disable.
+  input_header.keyIdx = 31;
+  RtpPacketizerVp8 packetizer(data, /*limits=*/{}, input_header);
+  EXPECT_EQ(packetizer.NumPackets(), 1u);
+  ASSERT_TRUE(packetizer.NextPacket(&packet));
+
+  VideoRtpDepacketizerVp8 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(packet.PayloadBuffer());
+  ASSERT_TRUE(parsed);
+
+  EXPECT_EQ(parsed->video_header.codec, kVideoCodecVP8);
+  const auto& vp8_header =
+      absl::get<RTPVideoHeaderVP8>(parsed->video_header.video_type_header);
+  EXPECT_EQ(vp8_header.nonReference, input_header.nonReference);
+  EXPECT_EQ(vp8_header.pictureId, input_header.pictureId);
+  EXPECT_EQ(vp8_header.tl0PicIdx, input_header.tl0PicIdx);
+  EXPECT_EQ(vp8_header.temporalIdx, input_header.temporalIdx);
+  EXPECT_EQ(vp8_header.layerSync, input_header.layerSync);
+  EXPECT_EQ(vp8_header.keyIdx, input_header.keyIdx);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, ReferencesInputCopyOnWriteBuffer) {
+  constexpr size_t kHeaderSize = 5;
+  uint8_t packet[16] = {0};
+  packet[0] = 0b1000'0000;
+  packet[1] = 0b1111'0000;  // with all extensions,
+  packet[2] = 15;           // and one-byte picture id.
+
+  rtc::CopyOnWriteBuffer rtp_payload(packet);
+  VideoRtpDepacketizerVp8 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+  ASSERT_TRUE(parsed);
+
+  EXPECT_EQ(parsed->video_payload.size(), rtp_payload.size() - kHeaderSize);
+  // Compare pointers to check there was no copy on write buffer unsharing.
+  EXPECT_EQ(parsed->video_payload.cdata(), rtp_payload.cdata() + kHeaderSize);
+}
+
+TEST(VideoRtpDepacketizerVp8Test, FailsOnEmptyPayload) {
+  rtc::ArrayView<const uint8_t> empty;
+  RTPVideoHeader video_header;
+  EXPECT_EQ(VideoRtpDepacketizerVp8::ParseRtpPayload(empty, &video_header), 0);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
new file mode 100644
index 0000000000..41f363d221
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc
@@ -0,0 +1,226 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
+
+#include <string.h>
+
+#include "api/video/video_codec_constants.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "rtc_base/bitstream_reader.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+// Picture ID:
+//
+//      +-+-+-+-+-+-+-+-+
+// I:   |M| PICTURE ID  |   M:0 => picture id is 7 bits.
+//      +-+-+-+-+-+-+-+-+   M:1 => picture id is 15 bits.
+// M:   | EXTENDED PID  |
+//      +-+-+-+-+-+-+-+-+
+//
+void ParsePictureId(BitstreamReader& parser, RTPVideoHeaderVP9* vp9) {
+  if (parser.ReadBit()) {  // m_bit
+    vp9->picture_id = parser.ReadBits(15);
+    vp9->max_picture_id = kMaxTwoBytePictureId;
+  } else {
+    vp9->picture_id = parser.ReadBits(7);
+    vp9->max_picture_id = kMaxOneBytePictureId;
+  }
+}
+
+// Layer indices :
+//
+//      +-+-+-+-+-+-+-+-+
+// L:   |  T  |U|  S  |D|
+//      +-+-+-+-+-+-+-+-+
+//      |   TL0PICIDX   |  (non-flexible mode only)
+//      +-+-+-+-+-+-+-+-+
+//
+void ParseLayerInfo(BitstreamReader& parser, RTPVideoHeaderVP9* vp9) {
+  vp9->temporal_idx = parser.ReadBits(3);
+  vp9->temporal_up_switch = parser.Read<bool>();
+  vp9->spatial_idx = parser.ReadBits(3);
+  vp9->inter_layer_predicted = parser.Read<bool>();
+  if (vp9->spatial_idx >= kMaxSpatialLayers) {
+    parser.Invalidate();
+    return;
+  }
+
+  if (!vp9->flexible_mode) {
+    vp9->tl0_pic_idx = parser.Read<uint8_t>();
+  }
+}
+
+// Reference indices:
+//
+//      +-+-+-+-+-+-+-+-+                P=1,F=1: At least one reference index
+// P,F: | P_DIFF      |N|  up to 3 times          has to be specified.
+//      +-+-+-+-+-+-+-+-+                    N=1: An additional P_DIFF follows
+//                                                current P_DIFF.
+//
+void ParseRefIndices(BitstreamReader& parser, RTPVideoHeaderVP9* vp9) {
+  if (vp9->picture_id == kNoPictureId) {
+    parser.Invalidate();
+    return;
+  }
+
+  vp9->num_ref_pics = 0;
+  bool n_bit;
+  do {
+    if (vp9->num_ref_pics == kMaxVp9RefPics) {
+      parser.Invalidate();
+      return;
+    }
+
+    uint8_t p_diff = parser.ReadBits(7);
+    n_bit = parser.Read<bool>();
+
+    vp9->pid_diff[vp9->num_ref_pics] = p_diff;
+    uint32_t scaled_pid = vp9->picture_id;
+    if (p_diff > scaled_pid) {
+      // TODO(asapersson): Max should correspond to the picture id of last wrap.
+      scaled_pid += vp9->max_picture_id + 1;
+    }
+    vp9->ref_picture_id[vp9->num_ref_pics++] = scaled_pid - p_diff;
+  } while (n_bit);
+}
+
+// Scalability structure (SS).
+//
+//      +-+-+-+-+-+-+-+-+
+// V:   | N_S |Y|G|-|-|-|
+//      +-+-+-+-+-+-+-+-+              -|
+// Y:   |     WIDTH     | (OPTIONAL)    .
+//      +               +               .
+//      |               | (OPTIONAL)    .
+//      +-+-+-+-+-+-+-+-+               . N_S + 1 times
+//      |     HEIGHT    | (OPTIONAL)    .
+//      +               +               .
+//      |               | (OPTIONAL)    .
+//      +-+-+-+-+-+-+-+-+              -|
+// G:   |      N_G      | (OPTIONAL)
+//      +-+-+-+-+-+-+-+-+                           -|
+// N_G: |  T  |U| R |-|-| (OPTIONAL)                 .
+//      +-+-+-+-+-+-+-+-+              -|            . N_G times
+//      |    P_DIFF     | (OPTIONAL)    . R times    .
+//      +-+-+-+-+-+-+-+-+              -|           -|
+//
+void ParseSsData(BitstreamReader& parser, RTPVideoHeaderVP9* vp9) {
+  vp9->num_spatial_layers = parser.ReadBits(3) + 1;
+  vp9->spatial_layer_resolution_present = parser.Read<bool>();
+  bool g_bit = parser.Read<bool>();
+  parser.ConsumeBits(3);
+  vp9->gof.num_frames_in_gof = 0;
+
+  if (vp9->spatial_layer_resolution_present) {
+    for (size_t i = 0; i < vp9->num_spatial_layers; ++i) {
+      vp9->width[i] = parser.Read<uint16_t>();
+      vp9->height[i] = parser.Read<uint16_t>();
+    }
+  }
+  if (g_bit) {
+    vp9->gof.num_frames_in_gof = parser.Read<uint8_t>();
+  }
+  for (size_t i = 0; i < vp9->gof.num_frames_in_gof; ++i) {
+    vp9->gof.temporal_idx[i] = parser.ReadBits(3);
+    vp9->gof.temporal_up_switch[i] = parser.Read<bool>();
+    vp9->gof.num_ref_pics[i] = parser.ReadBits(2);
+    parser.ConsumeBits(2);
+
+    for (uint8_t p = 0; p < vp9->gof.num_ref_pics[i]; ++p) {
+      vp9->gof.pid_diff[i][p] = parser.Read<uint8_t>();
+    }
+  }
+}
+}  // namespace
+
+absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
+VideoRtpDepacketizerVp9::Parse(rtc::CopyOnWriteBuffer rtp_payload) {
+  absl::optional<ParsedRtpPayload> result(absl::in_place);
+  int offset = ParseRtpPayload(rtp_payload, &result->video_header);
+  if (offset == 0)
+    return absl::nullopt;
+  RTC_DCHECK_LT(offset, rtp_payload.size());
+  result->video_payload =
+      rtp_payload.Slice(offset, rtp_payload.size() - offset);
+  return result;
+}
+
+int VideoRtpDepacketizerVp9::ParseRtpPayload(
+    rtc::ArrayView<const uint8_t> rtp_payload,
+    RTPVideoHeader* video_header) {
+  RTC_DCHECK(video_header);
+  // Parse mandatory first byte of payload descriptor.
+  BitstreamReader parser(rtp_payload);
+  uint8_t first_byte = parser.Read<uint8_t>();
+  bool i_bit = first_byte & 0b1000'0000;  // PictureId present .
+  bool p_bit = first_byte & 0b0100'0000;  // Inter-picture predicted.
+  bool l_bit = first_byte & 0b0010'0000;  // Layer indices present.
+  bool f_bit = first_byte & 0b0001'0000;  // Flexible mode.
+  bool b_bit = first_byte & 0b0000'1000;  // Begins frame flag.
+  bool e_bit = first_byte & 0b0000'0100;  // Ends frame flag.
+  bool v_bit = first_byte & 0b0000'0010;  // Scalability structure present.
+  bool z_bit = first_byte & 0b0000'0001;  // Not used for inter-layer prediction
+
+  // Parsed payload.
+  video_header->width = 0;
+  video_header->height = 0;
+  video_header->simulcastIdx = 0;
+  video_header->codec = kVideoCodecVP9;
+
+  video_header->frame_type =
+      p_bit ? VideoFrameType::kVideoFrameDelta : VideoFrameType::kVideoFrameKey;
+
+  auto& vp9_header =
+      video_header->video_type_header.emplace<RTPVideoHeaderVP9>();
+  vp9_header.InitRTPVideoHeaderVP9();
+  vp9_header.inter_pic_predicted = p_bit;
+  vp9_header.flexible_mode = f_bit;
+  vp9_header.beginning_of_frame = b_bit;
+  vp9_header.end_of_frame = e_bit;
+  vp9_header.ss_data_available = v_bit;
+  vp9_header.non_ref_for_inter_layer_pred = z_bit;
+
+  // Parse fields that are present.
+  if (i_bit) {
+    ParsePictureId(parser, &vp9_header);
+  }
+  if (l_bit) {
+    ParseLayerInfo(parser, &vp9_header);
+  }
+  if (p_bit && f_bit) {
+    ParseRefIndices(parser, &vp9_header);
+  }
+  if (v_bit) {
+    ParseSsData(parser, &vp9_header);
+    if (vp9_header.spatial_layer_resolution_present) {
+      // TODO(asapersson): Add support for spatial layers.
+      video_header->width = vp9_header.width[0];
+      video_header->height = vp9_header.height[0];
+    }
+  }
+  video_header->is_first_packet_in_frame = b_bit;
+  video_header->is_last_packet_in_frame = e_bit;
+
+  int num_remaining_bits = parser.RemainingBitCount();
+  if (num_remaining_bits <= 0) {
+    // Failed to parse or empty vp9 payload data.
+    return 0;
+  }
+  // vp9 descriptor is byte aligned.
+  RTC_DCHECK_EQ(num_remaining_bits % 8, 0);
+  return rtp_payload.size() - num_remaining_bits / 8;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
new file mode 100644
index 0000000000..4bb358a15f
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP9_H_
+#define MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP9_H_
+
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+class VideoRtpDepacketizerVp9 : public VideoRtpDepacketizer {
+ public:
+  VideoRtpDepacketizerVp9() = default;
+  VideoRtpDepacketizerVp9(const VideoRtpDepacketizerVp9&) = delete;
+  VideoRtpDepacketizerVp9& operator=(const VideoRtpDepacketizerVp9&) = delete;
+  ~VideoRtpDepacketizerVp9() override = default;
+
+  // Parses vp9 rtp payload descriptor.
+  // Returns zero on error or vp9 payload header offset on success.
+  static int ParseRtpPayload(rtc::ArrayView<const uint8_t> rtp_payload,
+                             RTPVideoHeader* video_header);
+
+  absl::optional<ParsedRtpPayload> Parse(
+      rtc::CopyOnWriteBuffer rtp_payload) override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_RTP_DEPACKETIZER_VP9_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc
new file mode 100644
index 0000000000..36af59a779
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9_unittest.cc
@@ -0,0 +1,373 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+void VerifyHeader(const RTPVideoHeaderVP9& expected,
+                  const RTPVideoHeaderVP9& actual) {
+  EXPECT_EQ(expected.inter_layer_predicted, actual.inter_layer_predicted);
+  EXPECT_EQ(expected.inter_pic_predicted, actual.inter_pic_predicted);
+  EXPECT_EQ(expected.flexible_mode, actual.flexible_mode);
+  EXPECT_EQ(expected.beginning_of_frame, actual.beginning_of_frame);
+  EXPECT_EQ(expected.end_of_frame, actual.end_of_frame);
+  EXPECT_EQ(expected.ss_data_available, actual.ss_data_available);
+  EXPECT_EQ(expected.non_ref_for_inter_layer_pred,
+            actual.non_ref_for_inter_layer_pred);
+  EXPECT_EQ(expected.picture_id, actual.picture_id);
+  EXPECT_EQ(expected.max_picture_id, actual.max_picture_id);
+  EXPECT_EQ(expected.temporal_idx, actual.temporal_idx);
+  EXPECT_EQ(expected.spatial_idx, actual.spatial_idx);
+  EXPECT_EQ(expected.gof_idx, actual.gof_idx);
+  EXPECT_EQ(expected.tl0_pic_idx, actual.tl0_pic_idx);
+  EXPECT_EQ(expected.temporal_up_switch, actual.temporal_up_switch);
+
+  EXPECT_EQ(expected.num_ref_pics, actual.num_ref_pics);
+  for (uint8_t i = 0; i < expected.num_ref_pics; ++i) {
+    EXPECT_EQ(expected.pid_diff[i], actual.pid_diff[i]);
+    EXPECT_EQ(expected.ref_picture_id[i], actual.ref_picture_id[i]);
+  }
+  if (expected.ss_data_available) {
+    EXPECT_EQ(expected.spatial_layer_resolution_present,
+              actual.spatial_layer_resolution_present);
+    EXPECT_EQ(expected.num_spatial_layers, actual.num_spatial_layers);
+    if (expected.spatial_layer_resolution_present) {
+      for (size_t i = 0; i < expected.num_spatial_layers; i++) {
+        EXPECT_EQ(expected.width[i], actual.width[i]);
+        EXPECT_EQ(expected.height[i], actual.height[i]);
+      }
+    }
+    EXPECT_EQ(expected.gof.num_frames_in_gof, actual.gof.num_frames_in_gof);
+    for (size_t i = 0; i < expected.gof.num_frames_in_gof; i++) {
+      EXPECT_EQ(expected.gof.temporal_up_switch[i],
+                actual.gof.temporal_up_switch[i]);
+      EXPECT_EQ(expected.gof.temporal_idx[i], actual.gof.temporal_idx[i]);
+      EXPECT_EQ(expected.gof.num_ref_pics[i], actual.gof.num_ref_pics[i]);
+      for (uint8_t j = 0; j < expected.gof.num_ref_pics[i]; j++) {
+        EXPECT_EQ(expected.gof.pid_diff[i][j], actual.gof.pid_diff[i][j]);
+      }
+    }
+  }
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseBasicHeader) {
+  uint8_t packet[4] = {0};
+  packet[0] = 0x0C;  // I:0 P:0 L:0 F:0 B:1 E:1 V:0 Z:0
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 1);
+  RTPVideoHeaderVP9 expected;
+  expected.InitRTPVideoHeaderVP9();
+  expected.beginning_of_frame = true;
+  expected.end_of_frame = true;
+  VerifyHeader(expected,
+               absl::get<RTPVideoHeaderVP9>(video_header.video_type_header));
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseOneBytePictureId) {
+  uint8_t packet[10] = {0};
+  packet[0] = 0x80;  // I:1 P:0 L:0 F:0 B:0 E:0 V:0 Z:0
+  packet[1] = kMaxOneBytePictureId;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 2);
+  RTPVideoHeaderVP9 expected;
+  expected.InitRTPVideoHeaderVP9();
+  expected.picture_id = kMaxOneBytePictureId;
+  expected.max_picture_id = kMaxOneBytePictureId;
+  VerifyHeader(expected,
+               absl::get<RTPVideoHeaderVP9>(video_header.video_type_header));
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseTwoBytePictureId) {
+  uint8_t packet[10] = {0};
+  packet[0] = 0x80;  // I:1 P:0 L:0 F:0 B:0 E:0 V:0 Z:0
+  packet[1] = 0x80 | ((kMaxTwoBytePictureId >> 8) & 0x7F);
+  packet[2] = kMaxTwoBytePictureId & 0xFF;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 3);
+  RTPVideoHeaderVP9 expected;
+  expected.InitRTPVideoHeaderVP9();
+  expected.picture_id = kMaxTwoBytePictureId;
+  expected.max_picture_id = kMaxTwoBytePictureId;
+  VerifyHeader(expected,
+               absl::get<RTPVideoHeaderVP9>(video_header.video_type_header));
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseLayerInfoWithNonFlexibleMode) {
+  const uint8_t kTemporalIdx = 2;
+  const uint8_t kUbit = 1;
+  const uint8_t kSpatialIdx = 1;
+  const uint8_t kDbit = 1;
+  const uint8_t kTl0PicIdx = 17;
+  uint8_t packet[13] = {0};
+  packet[0] = 0x20;  // I:0 P:0 L:1 F:0 B:0 E:0 V:0 Z:0
+  packet[1] = (kTemporalIdx << 5) | (kUbit << 4) | (kSpatialIdx << 1) | kDbit;
+  packet[2] = kTl0PicIdx;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 3);
+  RTPVideoHeaderVP9 expected;
+  expected.InitRTPVideoHeaderVP9();
+  // T:2 U:1 S:1 D:1
+  // TL0PICIDX:17
+  expected.temporal_idx = kTemporalIdx;
+  expected.temporal_up_switch = kUbit ? true : false;
+  expected.spatial_idx = kSpatialIdx;
+  expected.inter_layer_predicted = kDbit ? true : false;
+  expected.tl0_pic_idx = kTl0PicIdx;
+  VerifyHeader(expected,
+               absl::get<RTPVideoHeaderVP9>(video_header.video_type_header));
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseLayerInfoWithFlexibleMode) {
+  const uint8_t kTemporalIdx = 2;
+  const uint8_t kUbit = 1;
+  const uint8_t kSpatialIdx = 0;
+  const uint8_t kDbit = 0;
+  uint8_t packet[13] = {0};
+  packet[0] = 0x38;  // I:0 P:0 L:1 F:1 B:1 E:0 V:0 Z:0
+  packet[1] = (kTemporalIdx << 5) | (kUbit << 4) | (kSpatialIdx << 1) | kDbit;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 2);
+  RTPVideoHeaderVP9 expected;
+  expected.InitRTPVideoHeaderVP9();
+  // I:0 P:0 L:1 F:1 B:1 E:0 V:0 Z:0
+  // L:   T:2 U:1 S:0 D:0
+  expected.beginning_of_frame = true;
+  expected.flexible_mode = true;
+  expected.temporal_idx = kTemporalIdx;
+  expected.temporal_up_switch = kUbit ? true : false;
+  expected.spatial_idx = kSpatialIdx;
+  expected.inter_layer_predicted = kDbit ? true : false;
+  VerifyHeader(expected,
+               absl::get<RTPVideoHeaderVP9>(video_header.video_type_header));
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseRefIdx) {
+  const int16_t kPictureId = 17;
+  const uint8_t kPdiff1 = 17;
+  const uint8_t kPdiff2 = 18;
+  const uint8_t kPdiff3 = 127;
+  uint8_t packet[13] = {0};
+  packet[0] = 0xD8;  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 Z:0
+  packet[1] = 0x80 | ((kPictureId >> 8) & 0x7F);  // Two byte pictureID.
+  packet[2] = kPictureId;
+  packet[3] = (kPdiff1 << 1) | 1;  // P_DIFF N:1
+  packet[4] = (kPdiff2 << 1) | 1;  // P_DIFF N:1
+  packet[5] = (kPdiff3 << 1) | 0;  // P_DIFF N:0
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 6);
+  RTPVideoHeaderVP9 expected;
+  expected.InitRTPVideoHeaderVP9();
+  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 Z:0
+  // I:    PICTURE ID:17
+  // I:
+  // P,F:  P_DIFF:17  N:1 => refPicId = 17 - 17 = 0
+  // P,F:  P_DIFF:18  N:1 => refPicId = (kMaxPictureId + 1) + 17 - 18 = 0x7FFF
+  // P,F:  P_DIFF:127 N:0 => refPicId = (kMaxPictureId + 1) + 17 - 127 = 32658
+  expected.beginning_of_frame = true;
+  expected.inter_pic_predicted = true;
+  expected.flexible_mode = true;
+  expected.picture_id = kPictureId;
+  expected.num_ref_pics = 3;
+  expected.pid_diff[0] = kPdiff1;
+  expected.pid_diff[1] = kPdiff2;
+  expected.pid_diff[2] = kPdiff3;
+  expected.ref_picture_id[0] = 0;
+  expected.ref_picture_id[1] = 0x7FFF;
+  expected.ref_picture_id[2] = 32658;
+  VerifyHeader(expected,
+               absl::get<RTPVideoHeaderVP9>(video_header.video_type_header));
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseRefIdxFailsWithNoPictureId) {
+  const uint8_t kPdiff = 3;
+  uint8_t packet[13] = {0};
+  packet[0] = 0x58;           // I:0 P:1 L:0 F:1 B:1 E:0 V:0 Z:0
+  packet[1] = (kPdiff << 1);  // P,F:  P_DIFF:3 N:0
+
+  RTPVideoHeader video_header;
+  EXPECT_EQ(VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header), 0);
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseRefIdxFailsWithTooManyRefPics) {
+  const uint8_t kPdiff = 3;
+  uint8_t packet[13] = {0};
+  packet[0] = 0xD8;                  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 Z:0
+  packet[1] = kMaxOneBytePictureId;  // I:    PICTURE ID:127
+  packet[2] = (kPdiff << 1) | 1;     // P,F:  P_DIFF:3 N:1
+  packet[3] = (kPdiff << 1) | 1;     // P,F:  P_DIFF:3 N:1
+  packet[4] = (kPdiff << 1) | 1;     // P,F:  P_DIFF:3 N:1
+  packet[5] = (kPdiff << 1) | 0;     // P,F:  P_DIFF:3 N:0
+
+  RTPVideoHeader video_header;
+  EXPECT_EQ(VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header), 0);
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseSsData) {
+  const uint8_t kYbit = 0;
+  const size_t kNs = 2;
+  const size_t kNg = 2;
+  uint8_t packet[23] = {0};
+  packet[0] = 0x0A;  // I:0 P:0 L:0 F:0 B:1 E:0 V:1 Z:0
+  packet[1] = ((kNs - 1) << 5) | (kYbit << 4) | (1 << 3);  // N_S Y G:1 -
+  packet[2] = kNg;                                         // N_G
+  packet[3] = (0 << 5) | (1 << 4) | (0 << 2) | 0;          // T:0 U:1 R:0 -
+  packet[4] = (2 << 5) | (0 << 4) | (1 << 2) | 0;          // T:2 U:0 R:1 -
+  packet[5] = 33;
+
+  RTPVideoHeader video_header;
+  int offset = VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(offset, 6);
+  RTPVideoHeaderVP9 expected;
+  expected.InitRTPVideoHeaderVP9();
+  expected.beginning_of_frame = true;
+  expected.ss_data_available = true;
+  expected.num_spatial_layers = kNs;
+  expected.spatial_layer_resolution_present = kYbit ? true : false;
+  expected.gof.num_frames_in_gof = kNg;
+  expected.gof.temporal_idx[0] = 0;
+  expected.gof.temporal_idx[1] = 2;
+  expected.gof.temporal_up_switch[0] = true;
+  expected.gof.temporal_up_switch[1] = false;
+  expected.gof.num_ref_pics[0] = 0;
+  expected.gof.num_ref_pics[1] = 1;
+  expected.gof.pid_diff[1][0] = 33;
+  VerifyHeader(expected,
+               absl::get<RTPVideoHeaderVP9>(video_header.video_type_header));
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseFirstPacketInKeyFrame) {
+  uint8_t packet[2] = {0};
+  packet[0] = 0x08;  // I:0 P:0 L:0 F:0 B:1 E:0 V:0 Z:0
+
+  RTPVideoHeader video_header;
+  VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(video_header.frame_type, VideoFrameType::kVideoFrameKey);
+  EXPECT_TRUE(video_header.is_first_packet_in_frame);
+  EXPECT_FALSE(video_header.is_last_packet_in_frame);
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseLastPacketInDeltaFrame) {
+  uint8_t packet[2] = {0};
+  packet[0] = 0x44;  // I:0 P:1 L:0 F:0 B:0 E:1 V:0 Z:0
+
+  RTPVideoHeader video_header;
+  VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(video_header.frame_type, VideoFrameType::kVideoFrameDelta);
+  EXPECT_FALSE(video_header.is_first_packet_in_frame);
+  EXPECT_TRUE(video_header.is_last_packet_in_frame);
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseResolution) {
+  const uint16_t kWidth[2] = {640, 1280};
+  const uint16_t kHeight[2] = {360, 720};
+  uint8_t packet[20] = {0};
+  packet[0] = 0x0A;                     // I:0 P:0 L:0 F:0 B:1 E:0 V:1 Z:0
+  packet[1] = (1 << 5) | (1 << 4) | 0;  // N_S:1 Y:1 G:0
+  packet[2] = kWidth[0] >> 8;
+  packet[3] = kWidth[0] & 0xFF;
+  packet[4] = kHeight[0] >> 8;
+  packet[5] = kHeight[0] & 0xFF;
+  packet[6] = kWidth[1] >> 8;
+  packet[7] = kWidth[1] & 0xFF;
+  packet[8] = kHeight[1] >> 8;
+  packet[9] = kHeight[1] & 0xFF;
+
+  RTPVideoHeader video_header;
+  VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  EXPECT_EQ(video_header.width, kWidth[0]);
+  EXPECT_EQ(video_header.height, kHeight[0]);
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseFailsForNoPayloadLength) {
+  rtc::ArrayView<const uint8_t> empty;
+
+  RTPVideoHeader video_header;
+  EXPECT_EQ(VideoRtpDepacketizerVp9::ParseRtpPayload(empty, &video_header), 0);
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseFailsForTooShortBufferToFitPayload) {
+  uint8_t packet[] = {0};
+
+  RTPVideoHeader video_header;
+  EXPECT_EQ(VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header), 0);
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ParseNonRefForInterLayerPred) {
+  RTPVideoHeader video_header;
+  RTPVideoHeaderVP9 expected;
+  expected.InitRTPVideoHeaderVP9();
+  uint8_t packet[2] = {0};
+
+  packet[0] = 0x08;  // I:0 P:0 L:0 F:0 B:1 E:0 V:0 Z:0
+  VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  expected.beginning_of_frame = true;
+  expected.non_ref_for_inter_layer_pred = false;
+  VerifyHeader(expected,
+               absl::get<RTPVideoHeaderVP9>(video_header.video_type_header));
+
+  packet[0] = 0x05;  // I:0 P:0 L:0 F:0 B:0 E:1 V:0 Z:1
+  VideoRtpDepacketizerVp9::ParseRtpPayload(packet, &video_header);
+
+  expected.beginning_of_frame = false;
+  expected.end_of_frame = true;
+  expected.non_ref_for_inter_layer_pred = true;
+  VerifyHeader(expected,
+               absl::get<RTPVideoHeaderVP9>(video_header.video_type_header));
+}
+
+TEST(VideoRtpDepacketizerVp9Test, ReferencesInputCopyOnWriteBuffer) {
+  constexpr size_t kHeaderSize = 1;
+  uint8_t packet[4] = {0};
+  packet[0] = 0x0C;  // I:0 P:0 L:0 F:0 B:1 E:1 V:0 Z:0
+
+  rtc::CopyOnWriteBuffer rtp_payload(packet);
+  VideoRtpDepacketizerVp9 depacketizer;
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed =
+      depacketizer.Parse(rtp_payload);
+  ASSERT_TRUE(parsed);
+
+  EXPECT_EQ(parsed->video_payload.size(), rtp_payload.size() - kHeaderSize);
+  // Compare pointers to check there was no copy on write buffer unsharing.
+  EXPECT_EQ(parsed->video_payload.cdata(), rtp_payload.cdata() + kHeaderSize);
+}
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h b/third_party/libwebrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
new file mode 100644
index 0000000000..0bc2f56917
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
+#define MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
+
+namespace webrtc {
+
+// Maximum number of media packets allowed in this test. The burst mask types
+// are currently defined up to (kMaxMediaPacketsTest, kMaxMediaPacketsTest).
+const int kMaxMediaPacketsTest = 12;
+
+// Maximum number of FEC codes considered in this test.
+const int kNumberCodes = kMaxMediaPacketsTest * (kMaxMediaPacketsTest + 1) / 2;
+
+// For the random mask type: reference level for the maximum average residual
+// loss expected for each code size up to:
+// (kMaxMediaPacketsTest, kMaxMediaPacketsTest).
+const float kMaxResidualLossRandomMask[kNumberCodes] = {
+    0.009463f, 0.022436f, 0.007376f, 0.033895f, 0.012423f, 0.004644f, 0.043438f,
+    0.019937f, 0.008820f, 0.003438f, 0.051282f, 0.025795f, 0.012872f, 0.006458f,
+    0.003195f, 0.057728f, 0.032146f, 0.016708f, 0.009242f, 0.005054f, 0.003078f,
+    0.063050f, 0.037261f, 0.021767f, 0.012447f, 0.007099f, 0.003826f, 0.002504f,
+    0.067476f, 0.042348f, 0.026169f, 0.015695f, 0.009478f, 0.005887f, 0.003568f,
+    0.001689f, 0.071187f, 0.046575f, 0.031697f, 0.019797f, 0.012433f, 0.007027f,
+    0.004845f, 0.002777f, 0.001753f, 0.074326f, 0.050628f, 0.034978f, 0.021955f,
+    0.014821f, 0.009462f, 0.006393f, 0.004181f, 0.003105f, 0.001231f, 0.077008f,
+    0.054226f, 0.038407f, 0.026251f, 0.018634f, 0.011568f, 0.008130f, 0.004957f,
+    0.003334f, 0.002069f, 0.001304f, 0.079318f, 0.057180f, 0.041268f, 0.028842f,
+    0.020033f, 0.014061f, 0.009636f, 0.006411f, 0.004583f, 0.002817f, 0.001770f,
+    0.001258f};
+
+// For the bursty mask type: reference level for the maximum average residual
+// loss expected for each code size up to:
+// (kMaxMediaPacketsTestf, kMaxMediaPacketsTest).
+const float kMaxResidualLossBurstyMask[kNumberCodes] = {
+    0.033236f, 0.053344f, 0.026616f, 0.064129f, 0.036589f, 0.021892f, 0.071055f,
+    0.043890f, 0.028009f, 0.018524f, 0.075968f, 0.049828f, 0.033288f, 0.022791f,
+    0.016088f, 0.079672f, 0.054586f, 0.037872f, 0.026679f, 0.019326f, 0.014293f,
+    0.082582f, 0.058719f, 0.042045f, 0.030504f, 0.022391f, 0.016894f, 0.012946f,
+    0.084935f, 0.062169f, 0.045620f, 0.033713f, 0.025570f, 0.019439f, 0.015121f,
+    0.011920f, 0.086881f, 0.065267f, 0.048721f, 0.037613f, 0.028278f, 0.022152f,
+    0.017314f, 0.013791f, 0.011130f, 0.088516f, 0.067911f, 0.051709f, 0.040819f,
+    0.030777f, 0.024547f, 0.019689f, 0.015877f, 0.012773f, 0.010516f, 0.089909f,
+    0.070332f, 0.054402f, 0.043210f, 0.034096f, 0.026625f, 0.021823f, 0.017648f,
+    0.014649f, 0.011982f, 0.010035f, 0.091109f, 0.072428f, 0.056775f, 0.045418f,
+    0.036679f, 0.028599f, 0.023693f, 0.019966f, 0.016603f, 0.013690f, 0.011359f,
+    0.009657f};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/test/testFec/test_fec.cc b/third_party/libwebrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
new file mode 100644
index 0000000000..5ac8feca21
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
@@ -0,0 +1,474 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * Test application for core FEC algorithm. Calls encoding and decoding
+ * functions in ForwardErrorCorrection directly.
+ */
+
+#include <string.h>
+#include <time.h>
+
+#include <list>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "rtc_base/random.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+// #define VERBOSE_OUTPUT
+
+namespace webrtc {
+namespace fec_private_tables {
+extern const uint8_t** kPacketMaskBurstyTbl[12];
+}
+namespace test {
+using fec_private_tables::kPacketMaskBurstyTbl;
+
+void ReceivePackets(
+    std::vector<std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>>*
+        to_decode_list,
+    std::vector<std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>>*
+        received_packet_list,
+    size_t num_packets_to_decode,
+    float reorder_rate,
+    float duplicate_rate,
+    Random* random) {
+  RTC_DCHECK(to_decode_list->empty());
+  RTC_DCHECK_LE(num_packets_to_decode, received_packet_list->size());
+
+  for (size_t i = 0; i < num_packets_to_decode; i++) {
+    auto it = received_packet_list->begin();
+    // Reorder packets.
+    float random_variable = random->Rand<float>();
+    while (random_variable < reorder_rate) {
+      ++it;
+      if (it == received_packet_list->end()) {
+        --it;
+        break;
+      }
+      random_variable = random->Rand<float>();
+    }
+    to_decode_list->push_back(std::move(*it));
+    received_packet_list->erase(it);
+
+    // Duplicate packets.
+    ForwardErrorCorrection::ReceivedPacket* received_packet =
+        to_decode_list->back().get();
+    random_variable = random->Rand<float>();
+    while (random_variable < duplicate_rate) {
+      std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> duplicate_packet(
+          new ForwardErrorCorrection::ReceivedPacket());
+      *duplicate_packet = *received_packet;
+      duplicate_packet->pkt = new ForwardErrorCorrection::Packet();
+      duplicate_packet->pkt->data = received_packet->pkt->data;
+
+      to_decode_list->push_back(std::move(duplicate_packet));
+      random_variable = random->Rand<float>();
+    }
+  }
+}
+
+void RunTest(bool use_flexfec) {
+  // TODO(marpan): Split this function into subroutines/helper functions.
+  enum { kMaxNumberMediaPackets = 48 };
+  enum { kMaxNumberFecPackets = 48 };
+
+  const uint32_t kNumMaskBytesL0 = 2;
+  const uint32_t kNumMaskBytesL1 = 6;
+
+  // FOR UEP
+  const bool kUseUnequalProtection = true;
+
+  // FEC mask types.
+  const FecMaskType kMaskTypes[] = {kFecMaskRandom, kFecMaskBursty};
+  const int kNumFecMaskTypes = sizeof(kMaskTypes) / sizeof(*kMaskTypes);
+
+  // Maximum number of media packets allowed for the mask type.
+  const uint16_t kMaxMediaPackets[] = {
+      kMaxNumberMediaPackets,
+      sizeof(kPacketMaskBurstyTbl) / sizeof(*kPacketMaskBurstyTbl)};
+
+  ASSERT_EQ(12, kMaxMediaPackets[1]) << "Max media packets for bursty mode not "
+                                        "equal to 12.";
+
+  ForwardErrorCorrection::PacketList media_packet_list;
+  std::list<ForwardErrorCorrection::Packet*> fec_packet_list;
+  std::vector<std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>>
+      to_decode_list;
+  std::vector<std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>>
+      received_packet_list;
+  ForwardErrorCorrection::RecoveredPacketList recovered_packet_list;
+  std::list<uint8_t*> fec_mask_list;
+
+  // Running over only two loss rates to limit execution time.
+  const float loss_rate[] = {0.05f, 0.01f};
+  const uint32_t loss_rate_size = sizeof(loss_rate) / sizeof(*loss_rate);
+  const float reorder_rate = 0.1f;
+  const float duplicate_rate = 0.1f;
+
+  uint8_t media_loss_mask[kMaxNumberMediaPackets];
+  uint8_t fec_loss_mask[kMaxNumberFecPackets];
+  uint8_t fec_packet_masks[kMaxNumberFecPackets][kMaxNumberMediaPackets];
+
+  // Seed the random number generator, storing the seed to file in order to
+  // reproduce past results.
+  const unsigned int random_seed = static_cast<unsigned int>(time(nullptr));
+  Random random(random_seed);
+  std::string filename = webrtc::test::OutputPath() + "randomSeedLog.txt";
+  FILE* random_seed_file = fopen(filename.c_str(), "a");
+  fprintf(random_seed_file, "%u\n", random_seed);
+  fclose(random_seed_file);
+  random_seed_file = nullptr;
+
+  uint16_t seq_num = 0;
+  uint32_t timestamp = random.Rand<uint32_t>();
+  const uint32_t media_ssrc = random.Rand(1u, 0xfffffffe);
+  uint32_t fec_ssrc;
+  uint16_t fec_seq_num_offset;
+  if (use_flexfec) {
+    fec_ssrc = random.Rand(1u, 0xfffffffe);
+    fec_seq_num_offset = random.Rand(0, 1 << 15);
+  } else {
+    fec_ssrc = media_ssrc;
+    fec_seq_num_offset = 0;
+  }
+
+  std::unique_ptr<ForwardErrorCorrection> fec;
+  if (use_flexfec) {
+    fec = ForwardErrorCorrection::CreateFlexfec(fec_ssrc, media_ssrc);
+  } else {
+    RTC_DCHECK_EQ(media_ssrc, fec_ssrc);
+    fec = ForwardErrorCorrection::CreateUlpfec(fec_ssrc);
+  }
+
+  // Loop over the mask types: random and bursty.
+  for (int mask_type_idx = 0; mask_type_idx < kNumFecMaskTypes;
+       ++mask_type_idx) {
+    for (uint32_t loss_rate_idx = 0; loss_rate_idx < loss_rate_size;
+         ++loss_rate_idx) {
+      printf("Loss rate: %.2f, Mask type %d \n", loss_rate[loss_rate_idx],
+             mask_type_idx);
+
+      const uint32_t packet_mask_max = kMaxMediaPackets[mask_type_idx];
+      std::unique_ptr<uint8_t[]> packet_mask(
+          new uint8_t[packet_mask_max * kNumMaskBytesL1]);
+
+      FecMaskType fec_mask_type = kMaskTypes[mask_type_idx];
+
+      for (uint32_t num_media_packets = 1; num_media_packets <= packet_mask_max;
+           num_media_packets++) {
+        internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
+
+        for (uint32_t num_fec_packets = 1;
+             num_fec_packets <= num_media_packets &&
+             num_fec_packets <= packet_mask_max;
+             num_fec_packets++) {
+          // Loop over num_imp_packets: usually <= (0.3*num_media_packets).
+          // For this test we check up to ~ (num_media_packets / 4).
+          uint32_t max_num_imp_packets = num_media_packets / 4 + 1;
+          for (uint32_t num_imp_packets = 0;
+               num_imp_packets <= max_num_imp_packets &&
+               num_imp_packets <= packet_mask_max;
+               num_imp_packets++) {
+            uint8_t protection_factor =
+                static_cast<uint8_t>(num_fec_packets * 255 / num_media_packets);
+
+            const uint32_t mask_bytes_per_fec_packet =
+                (num_media_packets > 16) ? kNumMaskBytesL1 : kNumMaskBytesL0;
+
+            memset(packet_mask.get(), 0,
+                   num_media_packets * mask_bytes_per_fec_packet);
+
+            // Transfer packet masks from bit-mask to byte-mask.
+            internal::GeneratePacketMasks(
+                num_media_packets, num_fec_packets, num_imp_packets,
+                kUseUnequalProtection, &mask_table, packet_mask.get());
+
+#ifdef VERBOSE_OUTPUT
+            printf(
+                "%u media packets, %u FEC packets, %u num_imp_packets, "
+                "loss rate = %.2f \n",
+                num_media_packets, num_fec_packets, num_imp_packets,
+                loss_rate[loss_rate_idx]);
+            printf("Packet mask matrix \n");
+#endif
+
+            for (uint32_t i = 0; i < num_fec_packets; i++) {
+              for (uint32_t j = 0; j < num_media_packets; j++) {
+                const uint8_t byte_mask =
+                    packet_mask[i * mask_bytes_per_fec_packet + j / 8];
+                const uint32_t bit_position = (7 - j % 8);
+                fec_packet_masks[i][j] =
+                    (byte_mask & (1 << bit_position)) >> bit_position;
+#ifdef VERBOSE_OUTPUT
+                printf("%u ", fec_packet_masks[i][j]);
+#endif
+              }
+#ifdef VERBOSE_OUTPUT
+              printf("\n");
+#endif
+            }
+#ifdef VERBOSE_OUTPUT
+            printf("\n");
+#endif
+            // Check for all zero rows or columns: indicates incorrect mask.
+            uint32_t row_limit = num_media_packets;
+            for (uint32_t i = 0; i < num_fec_packets; ++i) {
+              uint32_t row_sum = 0;
+              for (uint32_t j = 0; j < row_limit; ++j) {
+                row_sum += fec_packet_masks[i][j];
+              }
+              ASSERT_NE(0u, row_sum) << "Row is all zero " << i;
+            }
+            for (uint32_t j = 0; j < row_limit; ++j) {
+              uint32_t column_sum = 0;
+              for (uint32_t i = 0; i < num_fec_packets; ++i) {
+                column_sum += fec_packet_masks[i][j];
+              }
+              ASSERT_NE(0u, column_sum) << "Column is all zero " << j;
+            }
+
+            // Construct media packets.
+            // Reset the sequence number here for each FEC code/mask tested
+            // below, to avoid sequence number wrap-around. In actual decoding,
+            // old FEC packets in list are dropped if sequence number wrap
+            // around is detected. This case is currently not handled below.
+            seq_num = 0;
+            for (uint32_t i = 0; i < num_media_packets; ++i) {
+              std::unique_ptr<ForwardErrorCorrection::Packet> media_packet(
+                  new ForwardErrorCorrection::Packet());
+              const uint32_t kMinPacketSize = 12;
+              const uint32_t kMaxPacketSize = static_cast<uint32_t>(
+                  IP_PACKET_SIZE - 12 - 28 - fec->MaxPacketOverhead());
+              size_t packet_length =
+                  random.Rand(kMinPacketSize, kMaxPacketSize);
+              media_packet->data.SetSize(packet_length);
+
+              uint8_t* data = media_packet->data.MutableData();
+              // Generate random values for the first 2 bytes.
+              data[0] = random.Rand<uint8_t>();
+              data[1] = random.Rand<uint8_t>();
+
+              // The first two bits are assumed to be 10 by the
+              // FEC encoder. In fact the FEC decoder will set the
+              // two first bits to 10 regardless of what they
+              // actually were. Set the first two bits to 10
+              // so that a memcmp can be performed for the
+              // whole restored packet.
+              data[0] |= 0x80;
+              data[0] &= 0xbf;
+
+              // FEC is applied to a whole frame.
+              // A frame is signaled by multiple packets without
+              // the marker bit set followed by the last packet of
+              // the frame for which the marker bit is set.
+              // Only push one (fake) frame to the FEC.
+              data[1] &= 0x7f;
+
+              ByteWriter<uint16_t>::WriteBigEndian(&data[2], seq_num);
+              ByteWriter<uint32_t>::WriteBigEndian(&data[4], timestamp);
+              ByteWriter<uint32_t>::WriteBigEndian(&data[8], media_ssrc);
+              // Generate random values for payload
+              for (size_t j = 12; j < packet_length; ++j) {
+                data[j] = random.Rand<uint8_t>();
+              }
+              media_packet_list.push_back(std::move(media_packet));
+              seq_num++;
+            }
+            media_packet_list.back()->data.MutableData()[1] |= 0x80;
+
+            ASSERT_EQ(0, fec->EncodeFec(media_packet_list, protection_factor,
+                                        num_imp_packets, kUseUnequalProtection,
+                                        fec_mask_type, &fec_packet_list))
+                << "EncodeFec() failed";
+
+            ASSERT_EQ(num_fec_packets, fec_packet_list.size())
+                << "We requested " << num_fec_packets
+                << " FEC packets, but "
+                   "EncodeFec() produced "
+                << fec_packet_list.size();
+
+            memset(media_loss_mask, 0, sizeof(media_loss_mask));
+            uint32_t media_packet_idx = 0;
+            for (const auto& media_packet : media_packet_list) {
+              // We want a value between 0 and 1.
+              const float loss_random_variable = random.Rand<float>();
+
+              if (loss_random_variable >= loss_rate[loss_rate_idx]) {
+                media_loss_mask[media_packet_idx] = 1;
+                std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>
+                    received_packet(
+                        new ForwardErrorCorrection::ReceivedPacket());
+                received_packet->pkt = new ForwardErrorCorrection::Packet();
+                received_packet->pkt->data = media_packet->data;
+                received_packet->ssrc = media_ssrc;
+                received_packet->seq_num = ByteReader<uint16_t>::ReadBigEndian(
+                    media_packet->data.data() + 2);
+                received_packet->is_fec = false;
+                received_packet_list.push_back(std::move(received_packet));
+              }
+              media_packet_idx++;
+            }
+
+            memset(fec_loss_mask, 0, sizeof(fec_loss_mask));
+            uint32_t fec_packet_idx = 0;
+            for (auto* fec_packet : fec_packet_list) {
+              const float loss_random_variable = random.Rand<float>();
+              if (loss_random_variable >= loss_rate[loss_rate_idx]) {
+                fec_loss_mask[fec_packet_idx] = 1;
+                std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>
+                    received_packet(
+                        new ForwardErrorCorrection::ReceivedPacket());
+                received_packet->pkt = new ForwardErrorCorrection::Packet();
+                received_packet->pkt->data = fec_packet->data;
+                received_packet->seq_num = fec_seq_num_offset + seq_num;
+                received_packet->is_fec = true;
+                received_packet->ssrc = fec_ssrc;
+                received_packet_list.push_back(std::move(received_packet));
+
+                fec_mask_list.push_back(fec_packet_masks[fec_packet_idx]);
+              }
+              ++fec_packet_idx;
+              ++seq_num;
+            }
+
+#ifdef VERBOSE_OUTPUT
+            printf("Media loss mask:\n");
+            for (uint32_t i = 0; i < num_media_packets; i++) {
+              printf("%u ", media_loss_mask[i]);
+            }
+            printf("\n\n");
+
+            printf("FEC loss mask:\n");
+            for (uint32_t i = 0; i < num_fec_packets; i++) {
+              printf("%u ", fec_loss_mask[i]);
+            }
+            printf("\n\n");
+#endif
+
+            auto fec_mask_it = fec_mask_list.begin();
+            while (fec_mask_it != fec_mask_list.end()) {
+              uint32_t hamming_dist = 0;
+              uint32_t recovery_position = 0;
+              for (uint32_t i = 0; i < num_media_packets; i++) {
+                if (media_loss_mask[i] == 0 && (*fec_mask_it)[i] == 1) {
+                  recovery_position = i;
+                  ++hamming_dist;
+                }
+              }
+              auto item_to_delete = fec_mask_it;
+              ++fec_mask_it;
+
+              if (hamming_dist == 1) {
+                // Recovery possible. Restart search.
+                media_loss_mask[recovery_position] = 1;
+                fec_mask_it = fec_mask_list.begin();
+              } else if (hamming_dist == 0) {
+                // FEC packet cannot provide further recovery.
+                fec_mask_list.erase(item_to_delete);
+              }
+            }
+#ifdef VERBOSE_OUTPUT
+            printf("Recovery mask:\n");
+            for (uint32_t i = 0; i < num_media_packets; ++i) {
+              printf("%u ", media_loss_mask[i]);
+            }
+            printf("\n\n");
+#endif
+            // For error-checking frame completion.
+            bool fec_packet_received = false;
+            while (!received_packet_list.empty()) {
+              size_t num_packets_to_decode = random.Rand(
+                  1u, static_cast<uint32_t>(received_packet_list.size()));
+              ReceivePackets(&to_decode_list, &received_packet_list,
+                             num_packets_to_decode, reorder_rate,
+                             duplicate_rate, &random);
+
+              if (fec_packet_received == false) {
+                for (const auto& received_packet : to_decode_list) {
+                  if (received_packet->is_fec) {
+                    fec_packet_received = true;
+                  }
+                }
+              }
+              for (const auto& received_packet : to_decode_list) {
+                fec->DecodeFec(*received_packet, &recovered_packet_list);
+              }
+              to_decode_list.clear();
+            }
+            media_packet_idx = 0;
+            for (const auto& media_packet : media_packet_list) {
+              if (media_loss_mask[media_packet_idx] == 1) {
+                // Should have recovered this packet.
+                auto recovered_packet_list_it = recovered_packet_list.cbegin();
+
+                ASSERT_FALSE(recovered_packet_list_it ==
+                             recovered_packet_list.end())
+                    << "Insufficient number of recovered packets.";
+                ForwardErrorCorrection::RecoveredPacket* recovered_packet =
+                    recovered_packet_list_it->get();
+
+                ASSERT_EQ(recovered_packet->pkt->data.size(),
+                          media_packet->data.size())
+                    << "Recovered packet length not identical to original "
+                       "media packet";
+                ASSERT_EQ(0, memcmp(recovered_packet->pkt->data.cdata(),
+                                    media_packet->data.cdata(),
+                                    media_packet->data.size()))
+                    << "Recovered packet payload not identical to original "
+                       "media packet";
+                recovered_packet_list.pop_front();
+              }
+              ++media_packet_idx;
+            }
+            fec->ResetState(&recovered_packet_list);
+            ASSERT_TRUE(recovered_packet_list.empty())
+                << "Excessive number of recovered packets.\t size is: "
+                << recovered_packet_list.size();
+            // -- Teardown --
+            media_packet_list.clear();
+
+            // Clear FEC packet list, so we don't pass in a non-empty
+            // list in the next call to DecodeFec().
+            fec_packet_list.clear();
+
+            // Delete received packets we didn't pass to DecodeFec(), due to
+            // early frame completion.
+            received_packet_list.clear();
+
+            while (!fec_mask_list.empty()) {
+              fec_mask_list.pop_front();
+            }
+            timestamp += 90000 / 30;
+          }  // loop over num_imp_packets
+        }    // loop over FecPackets
+      }      // loop over num_media_packets
+    }        // loop over loss rates
+  }          // loop over mask types
+
+  // Have DecodeFec clear the recovered packet list.
+  fec->ResetState(&recovered_packet_list);
+  ASSERT_TRUE(recovered_packet_list.empty())
+      << "Recovered packet list is not empty";
+}
+
+TEST(FecTest, UlpfecTest) {
+  RunTest(false);
+}
+
+TEST(FecTest, FlexfecTest) {
+  RunTest(true);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc b/third_party/libwebrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
new file mode 100644
index 0000000000..25ceee585a
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
@@ -0,0 +1,1060 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * The purpose of this test is to compute metrics to characterize the properties
+ * and efficiency of the packets masks used in the generic XOR FEC code.
+ *
+ * The metrics measure the efficiency (recovery potential or residual loss) of
+ * the FEC code, under various statistical loss models for the packet/symbol
+ * loss events. Various constraints on the behavior of these metrics are
+ * verified, and compared to the reference RS (Reed-Solomon) code. This serves
+ * in some way as a basic check/benchmark for the packet masks.
+ *
+ * By an FEC code, we mean an erasure packet/symbol code, characterized by:
+ * (1) The code size parameters (k,m), where k = number of source/media packets,
+ * and m = number of FEC packets,
+ * (2) The code type: XOR or RS.
+ * In the case of XOR, the residual loss is determined via the set of packet
+ * masks (generator matrix). In the case of RS, the residual loss is determined
+ * directly from the MDS (maximum distance separable) property of RS.
+ *
+ * Currently two classes of packets masks are available (random type and bursty
+ * type), so three codes are considered below: RS, XOR-random, and XOR-bursty.
+ * The bursty class is defined up to k=12, so (k=12,m=12) is largest code size
+ * considered in this test.
+ *
+ * The XOR codes are defined via the RFC 5109 and correspond to the class of
+ * LDGM (low density generator matrix) codes, which is a subset of the LDPC
+ * (low density parity check) codes. Future implementation will consider
+ * extending our XOR codes to include LDPC codes, which explicitly include
+ * protection of FEC packets.
+ *
+ * The type of packet/symbol loss models considered in this test are:
+ * (1) Random loss: Bernoulli process, characterized by the average loss rate.
+ * (2) Bursty loss: Markov chain (Gilbert-Elliot model), characterized by two
+ * parameters: average loss rate and average burst length.
+ */
+
+#include <cmath>
+#include <memory>
+
+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
+#include "modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+// Maximum number of media packets allows for XOR (RFC 5109) code.
+enum { kMaxNumberMediaPackets = 48 };
+
+// Maximum number of media packets allowed for each mask type.
+const uint16_t kMaxMediaPackets[] = {kMaxNumberMediaPackets, 12};
+
+// Maximum gap size for characterizing the consecutiveness of the loss.
+const int kMaxGapSize = 2 * kMaxMediaPacketsTest;
+
+// Number of gap levels written to file/output.
+const int kGapSizeOutput = 5;
+
+// Maximum number of states for characterizing the residual loss distribution.
+const int kNumStatesDistribution = 2 * kMaxMediaPacketsTest * kMaxGapSize + 1;
+
+// The code type.
+enum CodeType {
+  xor_random_code,  // XOR with random mask type.
+  xor_bursty_code,  // XOR with bursty mask type.
+  rs_code           // Reed_solomon.
+};
+
+// The code size parameters.
+struct CodeSizeParams {
+  int num_media_packets;
+  int num_fec_packets;
+  // Protection level: num_fec_packets / (num_media_packets + num_fec_packets).
+  float protection_level;
+  // Number of loss configurations, for a given loss number and gap number.
+  // The gap number refers to the maximum gap/hole of a loss configuration
+  // (used to measure the "consecutiveness" of the loss).
+  int configuration_density[kNumStatesDistribution];
+};
+
+// The type of loss models.
+enum LossModelType { kRandomLossModel, kBurstyLossModel };
+
+struct LossModel {
+  LossModelType loss_type;
+  float average_loss_rate;
+  float average_burst_length;
+};
+
+// Average loss rates.
+const float kAverageLossRate[] = {0.025f, 0.05f, 0.1f, 0.25f};
+
+// Average burst lengths. The case of |kAverageBurstLength = 1.0| refers to
+// the random model. Note that for the random (Bernoulli) model, the average
+// burst length is determined by the average loss rate, i.e.,
+// AverageBurstLength = 1 / (1 - AverageLossRate) for random model.
+const float kAverageBurstLength[] = {1.0f, 2.0f, 4.0f};
+
+// Total number of loss models: For each burst length case, there are
+// a number of models corresponding to the loss rates.
+const int kNumLossModels =
+    (sizeof(kAverageBurstLength) / sizeof(*kAverageBurstLength)) *
+    (sizeof(kAverageLossRate) / sizeof(*kAverageLossRate));
+
+// Thresholds on the average loss rate of the packet loss model, below which
+// certain properties of the codes are expected.
+float loss_rate_upper_threshold = 0.20f;
+float loss_rate_lower_threshold = 0.025f;
+
+// Set of thresholds on the expected average recovery rate, for each code type.
+// These are global thresholds for now; in future version we may condition them
+// on the code length/size and protection level.
+const float kRecoveryRateXorRandom[3] = {0.94f, 0.50f, 0.19f};
+const float kRecoveryRateXorBursty[3] = {0.90f, 0.54f, 0.22f};
+
+// Metrics for a given FEC code; each code is defined by the code type
+// (RS, XOR-random/bursty), and the code size parameters (k,m), where
+// k = num_media_packets, m = num_fec_packets.
+struct MetricsFecCode {
+  // The average and variance of the residual loss, as a function of the
+  // packet/symbol loss model. The average/variance is computed by averaging
+  // over all loss configurations wrt the loss probability given by the
+  // underlying loss model.
+  double average_residual_loss[kNumLossModels];
+  double variance_residual_loss[kNumLossModels];
+  // The residual loss, as a function of the loss number and the gap number of
+  // the loss configurations. The gap number refers to the maximum gap/hole of
+  // a loss configuration (used to measure the "consecutiveness" of the loss).
+  double residual_loss_per_loss_gap[kNumStatesDistribution];
+  // The recovery rate as a function of the loss number.
+  double recovery_rate_per_loss[2 * kMaxMediaPacketsTest + 1];
+};
+
+MetricsFecCode kMetricsXorRandom[kNumberCodes];
+MetricsFecCode kMetricsXorBursty[kNumberCodes];
+MetricsFecCode kMetricsReedSolomon[kNumberCodes];
+
+class FecPacketMaskMetricsTest : public ::testing::Test {
+ protected:
+  FecPacketMaskMetricsTest() {}
+
+  int max_num_codes_;
+  LossModel loss_model_[kNumLossModels];
+  CodeSizeParams code_params_[kNumberCodes];
+
+  uint8_t fec_packet_masks_[kMaxNumberMediaPackets][kMaxNumberMediaPackets];
+  FILE* fp_mask_;
+
+  // Measure of the gap of the loss for configuration given by `state`.
+  // This is to measure degree of consecutiveness for the loss configuration.
+  // Useful if the packets are sent out in order of sequence numbers and there
+  // is little/no re-ordering during transmission.
+  int GapLoss(int tot_num_packets, uint8_t* state) {
+    int max_gap_loss = 0;
+    // Find the first loss.
+    int first_loss = 0;
+    for (int i = 0; i < tot_num_packets; i++) {
+      if (state[i] == 1) {
+        first_loss = i;
+        break;
+      }
+    }
+    int prev_loss = first_loss;
+    for (int i = first_loss + 1; i < tot_num_packets; i++) {
+      if (state[i] == 1) {  // Lost state.
+        int gap_loss = (i - prev_loss) - 1;
+        if (gap_loss > max_gap_loss) {
+          max_gap_loss = gap_loss;
+        }
+        prev_loss = i;
+      }
+    }
+    return max_gap_loss;
+  }
+
+  // Returns the number of recovered media packets for the XOR code, given the
+  // packet mask `fec_packet_masks_`, for the loss state/configuration given by
+  // `state`.
+  int RecoveredMediaPackets(int num_media_packets,
+                            int num_fec_packets,
+                            uint8_t* state) {
+    std::unique_ptr<uint8_t[]> state_tmp(
+        new uint8_t[num_media_packets + num_fec_packets]);
+    memcpy(state_tmp.get(), state, num_media_packets + num_fec_packets);
+    int num_recovered_packets = 0;
+    bool loop_again = true;
+    while (loop_again) {
+      loop_again = false;
+      bool recovered_new_packet = false;
+      // Check if we can recover anything: loop over all possible FEC packets.
+      for (int i = 0; i < num_fec_packets; i++) {
+        if (state_tmp[i + num_media_packets] == 0) {
+          // We have this FEC packet.
+          int num_packets_in_mask = 0;
+          int num_received_packets_in_mask = 0;
+          for (int j = 0; j < num_media_packets; j++) {
+            if (fec_packet_masks_[i][j] == 1) {
+              num_packets_in_mask++;
+              if (state_tmp[j] == 0) {
+                num_received_packets_in_mask++;
+              }
+            }
+          }
+          if ((num_packets_in_mask - 1) == num_received_packets_in_mask) {
+            // We can recover the missing media packet for this FEC packet.
+            num_recovered_packets++;
+            recovered_new_packet = true;
+            int jsel = -1;
+            int check_num_recovered = 0;
+            // Update the state with newly recovered media packet.
+            for (int j = 0; j < num_media_packets; j++) {
+              if (fec_packet_masks_[i][j] == 1 && state_tmp[j] == 1) {
+                // This is the lost media packet we will recover.
+                jsel = j;
+                check_num_recovered++;
+              }
+            }
+            // Check that we can only recover 1 packet.
+            RTC_DCHECK_EQ(check_num_recovered, 1);
+            // Update the state with the newly recovered media packet.
+            state_tmp[jsel] = 0;
+          }
+        }
+      }  // Go to the next FEC packet in the loop.
+      // If we have recovered at least one new packet in this FEC loop,
+      // go through loop again, otherwise we leave loop.
+      if (recovered_new_packet) {
+        loop_again = true;
+      }
+    }
+    return num_recovered_packets;
+  }
+
+  // Compute the probability of occurence of the loss state/configuration,
+  // given by `state`, for all the loss models considered in this test.
+  void ComputeProbabilityWeight(double* prob_weight,
+                                uint8_t* state,
+                                int tot_num_packets) {
+    // Loop over the loss models.
+    for (int k = 0; k < kNumLossModels; k++) {
+      double loss_rate = static_cast<double>(loss_model_[k].average_loss_rate);
+      double burst_length =
+          static_cast<double>(loss_model_[k].average_burst_length);
+      double result = 1.0;
+      if (loss_model_[k].loss_type == kRandomLossModel) {
+        for (int i = 0; i < tot_num_packets; i++) {
+          if (state[i] == 0) {
+            result *= (1.0 - loss_rate);
+          } else {
+            result *= loss_rate;
+          }
+        }
+      } else {  // Gilbert-Elliot model for burst model.
+        RTC_DCHECK_EQ(loss_model_[k].loss_type, kBurstyLossModel);
+        // Transition probabilities: from previous to current state.
+        // Prob. of previous = lost --> current = received.
+        double prob10 = 1.0 / burst_length;
+        // Prob. of previous = lost --> currrent = lost.
+        double prob11 = 1.0 - prob10;
+        // Prob. of previous = received --> current = lost.
+        double prob01 = prob10 * (loss_rate / (1.0 - loss_rate));
+        // Prob. of previous = received --> current = received.
+        double prob00 = 1.0 - prob01;
+
+        // Use stationary probability for first state/packet.
+        if (state[0] == 0) {  // Received
+          result = (1.0 - loss_rate);
+        } else {  // Lost
+          result = loss_rate;
+        }
+
+        // Subsequent states: use transition probabilities.
+        for (int i = 1; i < tot_num_packets; i++) {
+          // Current state is received
+          if (state[i] == 0) {
+            if (state[i - 1] == 0) {
+              result *= prob00;  // Previous received, current received.
+            } else {
+              result *= prob10;  // Previous lost, current received.
+            }
+          } else {  // Current state is lost
+            if (state[i - 1] == 0) {
+              result *= prob01;  // Previous received, current lost.
+            } else {
+              result *= prob11;  // Previous lost, current lost.
+            }
+          }
+        }
+      }
+      prob_weight[k] = result;
+    }
+  }
+
+  void CopyMetrics(MetricsFecCode* metrics_output,
+                   MetricsFecCode metrics_input) {
+    memcpy(metrics_output->average_residual_loss,
+           metrics_input.average_residual_loss,
+           sizeof(double) * kNumLossModels);
+    memcpy(metrics_output->variance_residual_loss,
+           metrics_input.variance_residual_loss,
+           sizeof(double) * kNumLossModels);
+    memcpy(metrics_output->residual_loss_per_loss_gap,
+           metrics_input.residual_loss_per_loss_gap,
+           sizeof(double) * kNumStatesDistribution);
+    memcpy(metrics_output->recovery_rate_per_loss,
+           metrics_input.recovery_rate_per_loss,
+           sizeof(double) * 2 * kMaxMediaPacketsTest);
+  }
+
+  // Compute the residual loss per gap, by summing the
+  // `residual_loss_per_loss_gap` over all loss configurations up to loss number
+  // = `num_fec_packets`.
+  double ComputeResidualLossPerGap(MetricsFecCode metrics,
+                                   int gap_number,
+                                   int num_fec_packets,
+                                   int code_index) {
+    double residual_loss_gap = 0.0;
+    int tot_num_configs = 0;
+    for (int loss = 1; loss <= num_fec_packets; loss++) {
+      int index = gap_number * (2 * kMaxMediaPacketsTest) + loss;
+      residual_loss_gap += metrics.residual_loss_per_loss_gap[index];
+      tot_num_configs += code_params_[code_index].configuration_density[index];
+    }
+    // Normalize, to compare across code sizes.
+    if (tot_num_configs > 0) {
+      residual_loss_gap =
+          residual_loss_gap / static_cast<double>(tot_num_configs);
+    }
+    return residual_loss_gap;
+  }
+
+  // Compute the recovery rate per loss number, by summing the
+  // `residual_loss_per_loss_gap` over all gap configurations.
+  void ComputeRecoveryRatePerLoss(MetricsFecCode* metrics,
+                                  int num_media_packets,
+                                  int num_fec_packets,
+                                  int code_index) {
+    for (int loss = 1; loss <= num_media_packets + num_fec_packets; loss++) {
+      metrics->recovery_rate_per_loss[loss] = 0.0;
+      int tot_num_configs = 0;
+      double arl = 0.0;
+      for (int gap = 0; gap < kMaxGapSize; gap++) {
+        int index = gap * (2 * kMaxMediaPacketsTest) + loss;
+        arl += metrics->residual_loss_per_loss_gap[index];
+        tot_num_configs +=
+            code_params_[code_index].configuration_density[index];
+      }
+      // Normalize, to compare across code sizes.
+      if (tot_num_configs > 0) {
+        arl = arl / static_cast<double>(tot_num_configs);
+      }
+      // Recovery rate for a given loss `loss` is 1 minus the scaled `arl`,
+      // where the scale factor is relative to code size/parameters.
+      double scaled_loss =
+          static_cast<double>(loss * num_media_packets) /
+          static_cast<double>(num_media_packets + num_fec_packets);
+      metrics->recovery_rate_per_loss[loss] = 1.0 - arl / scaled_loss;
+    }
+  }
+
+  void SetMetricsZero(MetricsFecCode* metrics) {
+    memset(metrics->average_residual_loss, 0, sizeof(double) * kNumLossModels);
+    memset(metrics->variance_residual_loss, 0, sizeof(double) * kNumLossModels);
+    memset(metrics->residual_loss_per_loss_gap, 0,
+           sizeof(double) * kNumStatesDistribution);
+    memset(metrics->recovery_rate_per_loss, 0,
+           sizeof(double) * 2 * kMaxMediaPacketsTest + 1);
+  }
+
+  // Compute the metrics for an FEC code, given by the code type `code_type`
+  // (XOR-random/ bursty or RS), and by the code index `code_index`
+  // (which containes the code size parameters/protection length).
+  void ComputeMetricsForCode(CodeType code_type, int code_index) {
+    std::unique_ptr<double[]> prob_weight(new double[kNumLossModels]);
+    memset(prob_weight.get(), 0, sizeof(double) * kNumLossModels);
+    MetricsFecCode metrics_code;
+    SetMetricsZero(&metrics_code);
+
+    int num_media_packets = code_params_[code_index].num_media_packets;
+    int num_fec_packets = code_params_[code_index].num_fec_packets;
+    int tot_num_packets = num_media_packets + num_fec_packets;
+    std::unique_ptr<uint8_t[]> state(new uint8_t[tot_num_packets]);
+    memset(state.get(), 0, tot_num_packets);
+
+    int num_loss_configurations = 1 << tot_num_packets;
+    // Loop over all loss configurations for the symbol sequence of length
+    // `tot_num_packets`. In this version we process up to (k=12, m=12) codes,
+    // and get exact expressions for the residual loss.
+    // TODO(marpan): For larger codes, loop over some random sample of loss
+    // configurations, sampling driven by the underlying statistical loss model
+    // (importance sampling).
+
+    // The symbols/packets are arranged as a sequence of source/media packets
+    // followed by FEC packets. This is the sequence ordering used in the RTP.
+    // A configuration refers to a sequence of received/lost (0/1 bit) states
+    // for the string of packets/symbols. For example, for a (k=4,m=3) code
+    // (4 media packets, 3 FEC packets), with 2 losses (one media and one FEC),
+    // the loss configurations is:
+    // Media1   Media2   Media3   Media4   FEC1   FEC2   FEC3
+    //   0         0        1       0        0      1     0
+    for (int i = 1; i < num_loss_configurations; i++) {
+      // Counter for number of packets lost.
+      int num_packets_lost = 0;
+      // Counters for the number of media packets lost.
+      int num_media_packets_lost = 0;
+
+      // Map configuration number to a loss state.
+      for (int j = 0; j < tot_num_packets; j++) {
+        state[j] = 0;  // Received state.
+        int bit_value = i >> (tot_num_packets - j - 1) & 1;
+        if (bit_value == 1) {
+          state[j] = 1;  // Lost state.
+          num_packets_lost++;
+          if (j < num_media_packets) {
+            num_media_packets_lost++;
+          }
+        }
+      }  // Done with loop over total number of packets.
+      RTC_DCHECK_LE(num_media_packets_lost, num_media_packets);
+      RTC_DCHECK_LE(num_packets_lost, tot_num_packets && num_packets_lost > 0);
+      double residual_loss = 0.0;
+      // Only need to compute residual loss (number of recovered packets) for
+      // configurations that have at least one media packet lost.
+      if (num_media_packets_lost >= 1) {
+        // Compute the number of recovered packets.
+        int num_recovered_packets = 0;
+        if (code_type == xor_random_code || code_type == xor_bursty_code) {
+          num_recovered_packets = RecoveredMediaPackets(
+              num_media_packets, num_fec_packets, state.get());
+        } else {
+          // For the RS code, we can either completely recover all the packets
+          // if the loss is less than or equal to the number of FEC packets,
+          // otherwise we can recover none of the missing packets. This is the
+          // all or nothing (MDS) property of the RS code.
+          if (num_packets_lost <= num_fec_packets) {
+            num_recovered_packets = num_media_packets_lost;
+          }
+        }
+        RTC_DCHECK_LE(num_recovered_packets, num_media_packets);
+        // Compute the residual loss. We only care about recovering media/source
+        // packets, so residual loss is based on lost/recovered media packets.
+        residual_loss =
+            static_cast<double>(num_media_packets_lost - num_recovered_packets);
+        // Compute the probability weights for this configuration.
+        ComputeProbabilityWeight(prob_weight.get(), state.get(),
+                                 tot_num_packets);
+        // Update the average and variance of the residual loss.
+        for (int k = 0; k < kNumLossModels; k++) {
+          metrics_code.average_residual_loss[k] +=
+              residual_loss * prob_weight[k];
+          metrics_code.variance_residual_loss[k] +=
+              residual_loss * residual_loss * prob_weight[k];
+        }
+      }  // Done with processing for num_media_packets_lost >= 1.
+      // Update the distribution statistics.
+      // Compute the gap of the loss (the "consecutiveness" of the loss).
+      int gap_loss = GapLoss(tot_num_packets, state.get());
+      RTC_DCHECK_LT(gap_loss, kMaxGapSize);
+      int index = gap_loss * (2 * kMaxMediaPacketsTest) + num_packets_lost;
+      RTC_DCHECK_LT(index, kNumStatesDistribution);
+      metrics_code.residual_loss_per_loss_gap[index] += residual_loss;
+      if (code_type == xor_random_code) {
+        // The configuration density is only a function of the code length and
+        // only needs to computed for the first `code_type` passed here.
+        code_params_[code_index].configuration_density[index]++;
+      }
+    }  // Done with loop over configurations.
+    // Normalize the average residual loss and compute/normalize the variance.
+    for (int k = 0; k < kNumLossModels; k++) {
+      // Normalize the average residual loss by the total number of packets
+      // `tot_num_packets` (i.e., the code length). For a code with no (zero)
+      // recovery, the average residual loss for that code would be reduced like
+      // ~`average_loss_rate` * `num_media_packets` / `tot_num_packets`. This is
+      // the expected reduction in the average residual loss just from adding
+      // FEC packets to the symbol sequence.
+      metrics_code.average_residual_loss[k] =
+          metrics_code.average_residual_loss[k] /
+          static_cast<double>(tot_num_packets);
+      metrics_code.variance_residual_loss[k] =
+          metrics_code.variance_residual_loss[k] /
+          static_cast<double>(num_media_packets * num_media_packets);
+      metrics_code.variance_residual_loss[k] =
+          metrics_code.variance_residual_loss[k] -
+          (metrics_code.average_residual_loss[k] *
+           metrics_code.average_residual_loss[k]);
+      RTC_DCHECK_GE(metrics_code.variance_residual_loss[k], 0.0);
+      RTC_DCHECK_GT(metrics_code.average_residual_loss[k], 0.0);
+      metrics_code.variance_residual_loss[k] =
+          std::sqrt(metrics_code.variance_residual_loss[k]) /
+          metrics_code.average_residual_loss[k];
+    }
+
+    // Compute marginal distribution as a function of loss parameter.
+    ComputeRecoveryRatePerLoss(&metrics_code, num_media_packets,
+                               num_fec_packets, code_index);
+    if (code_type == rs_code) {
+      CopyMetrics(&kMetricsReedSolomon[code_index], metrics_code);
+    } else if (code_type == xor_random_code) {
+      CopyMetrics(&kMetricsXorRandom[code_index], metrics_code);
+    } else if (code_type == xor_bursty_code) {
+      CopyMetrics(&kMetricsXorBursty[code_index], metrics_code);
+    } else {
+      RTC_DCHECK_NOTREACHED();
+    }
+  }
+
+  void WriteOutMetricsAllFecCodes() {
+    std::string filename = test::OutputPath() + "data_metrics_all_codes";
+    FILE* fp = fopen(filename.c_str(), "wb");
+    // Loop through codes up to `kMaxMediaPacketsTest`.
+    int code_index = 0;
+    for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
+         num_media_packets++) {
+      for (int num_fec_packets = 1; num_fec_packets <= num_media_packets;
+           num_fec_packets++) {
+        fprintf(fp, "FOR CODE: (%d, %d) \n", num_media_packets,
+                num_fec_packets);
+        for (int k = 0; k < kNumLossModels; k++) {
+          float loss_rate = loss_model_[k].average_loss_rate;
+          float burst_length = loss_model_[k].average_burst_length;
+          fprintf(
+              fp,
+              "Loss rate = %.2f, Burst length = %.2f:  %.4f  %.4f  %.4f"
+              " **** %.4f %.4f %.4f \n",
+              loss_rate, burst_length,
+              100 * kMetricsReedSolomon[code_index].average_residual_loss[k],
+              100 * kMetricsXorRandom[code_index].average_residual_loss[k],
+              100 * kMetricsXorBursty[code_index].average_residual_loss[k],
+              kMetricsReedSolomon[code_index].variance_residual_loss[k],
+              kMetricsXorRandom[code_index].variance_residual_loss[k],
+              kMetricsXorBursty[code_index].variance_residual_loss[k]);
+        }
+        for (int gap = 0; gap < kGapSizeOutput; gap++) {
+          double rs_residual_loss =
+              ComputeResidualLossPerGap(kMetricsReedSolomon[code_index], gap,
+                                        num_fec_packets, code_index);
+          double xor_random_residual_loss = ComputeResidualLossPerGap(
+              kMetricsXorRandom[code_index], gap, num_fec_packets, code_index);
+          double xor_bursty_residual_loss = ComputeResidualLossPerGap(
+              kMetricsXorBursty[code_index], gap, num_fec_packets, code_index);
+          fprintf(fp,
+                  "Residual loss as a function of gap "
+                  "%d: %.4f %.4f %.4f \n",
+                  gap, rs_residual_loss, xor_random_residual_loss,
+                  xor_bursty_residual_loss);
+        }
+        fprintf(fp, "Recovery rate as a function of loss number \n");
+        for (int loss = 1; loss <= num_media_packets + num_fec_packets;
+             loss++) {
+          fprintf(fp, "For loss number %d: %.4f %.4f %.4f \n", loss,
+                  kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss],
+                  kMetricsXorRandom[code_index].recovery_rate_per_loss[loss],
+                  kMetricsXorBursty[code_index].recovery_rate_per_loss[loss]);
+        }
+        fprintf(fp, "******************\n");
+        fprintf(fp, "\n");
+        code_index++;
+      }
+    }
+    fclose(fp);
+  }
+
+  void SetLossModels() {
+    int num_loss_rates = sizeof(kAverageLossRate) / sizeof(*kAverageLossRate);
+    int num_burst_lengths =
+        sizeof(kAverageBurstLength) / sizeof(*kAverageBurstLength);
+    int num_loss_models = 0;
+    for (int k = 0; k < num_burst_lengths; k++) {
+      for (int k2 = 0; k2 < num_loss_rates; k2++) {
+        loss_model_[num_loss_models].average_loss_rate = kAverageLossRate[k2];
+        loss_model_[num_loss_models].average_burst_length =
+            kAverageBurstLength[k];
+        // First set of loss models are of random type.
+        if (k == 0) {
+          loss_model_[num_loss_models].loss_type = kRandomLossModel;
+        } else {
+          loss_model_[num_loss_models].loss_type = kBurstyLossModel;
+        }
+        num_loss_models++;
+      }
+    }
+    RTC_DCHECK_EQ(num_loss_models, kNumLossModels);
+  }
+
+  void SetCodeParams() {
+    int code_index = 0;
+    for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
+         num_media_packets++) {
+      for (int num_fec_packets = 1; num_fec_packets <= num_media_packets;
+           num_fec_packets++) {
+        code_params_[code_index].num_media_packets = num_media_packets;
+        code_params_[code_index].num_fec_packets = num_fec_packets;
+        code_params_[code_index].protection_level =
+            static_cast<float>(num_fec_packets) /
+            static_cast<float>(num_media_packets + num_fec_packets);
+        for (int k = 0; k < kNumStatesDistribution; k++) {
+          code_params_[code_index].configuration_density[k] = 0;
+        }
+        code_index++;
+      }
+    }
+    max_num_codes_ = code_index;
+  }
+
+  // Make some basic checks on the packet masks. Return -1 if any of these
+  // checks fail.
+  int RejectInvalidMasks(int num_media_packets, int num_fec_packets) {
+    // Make sure every FEC packet protects something.
+    for (int i = 0; i < num_fec_packets; i++) {
+      int row_degree = 0;
+      for (int j = 0; j < num_media_packets; j++) {
+        if (fec_packet_masks_[i][j] == 1) {
+          row_degree++;
+        }
+      }
+      if (row_degree == 0) {
+        printf(
+            "Invalid mask: FEC packet has empty mask (does not protect "
+            "anything) %d %d %d \n",
+            i, num_media_packets, num_fec_packets);
+        return -1;
+      }
+    }
+    // Mask sure every media packet has some protection.
+    for (int j = 0; j < num_media_packets; j++) {
+      int column_degree = 0;
+      for (int i = 0; i < num_fec_packets; i++) {
+        if (fec_packet_masks_[i][j] == 1) {
+          column_degree++;
+        }
+      }
+      if (column_degree == 0) {
+        printf(
+            "Invalid mask: Media packet has no protection at all %d %d %d "
+            "\n",
+            j, num_media_packets, num_fec_packets);
+        return -1;
+      }
+    }
+    // Make sure we do not have two identical FEC packets.
+    for (int i = 0; i < num_fec_packets; i++) {
+      for (int i2 = i + 1; i2 < num_fec_packets; i2++) {
+        int overlap = 0;
+        for (int j = 0; j < num_media_packets; j++) {
+          if (fec_packet_masks_[i][j] == fec_packet_masks_[i2][j]) {
+            overlap++;
+          }
+        }
+        if (overlap == num_media_packets) {
+          printf("Invalid mask: Two FEC packets are identical %d %d %d %d \n",
+                 i, i2, num_media_packets, num_fec_packets);
+          return -1;
+        }
+      }
+    }
+    // Avoid codes that have two media packets with full protection (all 1s in
+    // their corresponding columns). This would mean that if we lose those
+    // two packets, we can never recover them even if we receive all the other
+    // packets. Exclude the special cases of 1 or 2 FEC packets.
+    if (num_fec_packets > 2) {
+      for (int j = 0; j < num_media_packets; j++) {
+        for (int j2 = j + 1; j2 < num_media_packets; j2++) {
+          int degree = 0;
+          for (int i = 0; i < num_fec_packets; i++) {
+            if (fec_packet_masks_[i][j] == fec_packet_masks_[i][j2] &&
+                fec_packet_masks_[i][j] == 1) {
+              degree++;
+            }
+          }
+          if (degree == num_fec_packets) {
+            printf(
+                "Invalid mask: Two media packets are have full degree "
+                "%d %d %d %d \n",
+                j, j2, num_media_packets, num_fec_packets);
+            return -1;
+          }
+        }
+      }
+    }
+    return 0;
+  }
+
+  void GetPacketMaskConvertToBitMask(uint8_t* packet_mask,
+                                     int num_media_packets,
+                                     int num_fec_packets,
+                                     int mask_bytes_fec_packet,
+                                     CodeType code_type) {
+    for (int i = 0; i < num_fec_packets; i++) {
+      for (int j = 0; j < num_media_packets; j++) {
+        const uint8_t byte_mask =
+            packet_mask[i * mask_bytes_fec_packet + j / 8];
+        const int bit_position = (7 - j % 8);
+        fec_packet_masks_[i][j] =
+            (byte_mask & (1 << bit_position)) >> bit_position;
+        fprintf(fp_mask_, "%d ", fec_packet_masks_[i][j]);
+      }
+      fprintf(fp_mask_, "\n");
+    }
+    fprintf(fp_mask_, "\n");
+  }
+
+  int ProcessXORPacketMasks(CodeType code_type, FecMaskType fec_mask_type) {
+    int code_index = 0;
+    // Maximum number of media packets allowed for the mask type.
+    const int packet_mask_max = kMaxMediaPackets[fec_mask_type];
+    std::unique_ptr<uint8_t[]> packet_mask(
+        new uint8_t[packet_mask_max * kUlpfecMaxPacketMaskSize]);
+    // Loop through codes up to `kMaxMediaPacketsTest`.
+    for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
+         ++num_media_packets) {
+      const int mask_bytes_fec_packet =
+          static_cast<int>(internal::PacketMaskSize(num_media_packets));
+      internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
+      for (int num_fec_packets = 1; num_fec_packets <= num_media_packets;
+           num_fec_packets++) {
+        memset(packet_mask.get(), 0, num_media_packets * mask_bytes_fec_packet);
+        rtc::ArrayView<const uint8_t> mask =
+            mask_table.LookUp(num_media_packets, num_fec_packets);
+        memcpy(packet_mask.get(), &mask[0], mask.size());
+        // Convert to bit mask.
+        GetPacketMaskConvertToBitMask(packet_mask.get(), num_media_packets,
+                                      num_fec_packets, mask_bytes_fec_packet,
+                                      code_type);
+        if (RejectInvalidMasks(num_media_packets, num_fec_packets) < 0) {
+          return -1;
+        }
+        // Compute the metrics for this code/mask.
+        ComputeMetricsForCode(code_type, code_index);
+        code_index++;
+      }
+    }
+    RTC_DCHECK_EQ(code_index, kNumberCodes);
+    return 0;
+  }
+
+  void ProcessRS(CodeType code_type) {
+    int code_index = 0;
+    for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
+         num_media_packets++) {
+      for (int num_fec_packets = 1; num_fec_packets <= num_media_packets;
+           num_fec_packets++) {
+        // Compute the metrics for this code type.
+        ComputeMetricsForCode(code_type, code_index);
+        code_index++;
+      }
+    }
+  }
+
+  // Compute metrics for all code types and sizes.
+  void ComputeMetricsAllCodes() {
+    SetLossModels();
+    SetCodeParams();
+    // Get metrics for XOR code with packet masks of random type.
+    std::string filename = test::OutputPath() + "data_packet_masks";
+    fp_mask_ = fopen(filename.c_str(), "wb");
+    fprintf(fp_mask_, "MASK OF TYPE RANDOM: \n");
+    EXPECT_EQ(ProcessXORPacketMasks(xor_random_code, kFecMaskRandom), 0);
+    // Get metrics for XOR code with packet masks of bursty type.
+    fprintf(fp_mask_, "MASK OF TYPE BURSTY: \n");
+    EXPECT_EQ(ProcessXORPacketMasks(xor_bursty_code, kFecMaskBursty), 0);
+    fclose(fp_mask_);
+    // Get metrics for Reed-Solomon code.
+    ProcessRS(rs_code);
+  }
+};
+
+// Verify that the average residual loss, averaged over loss models
+// appropriate to each mask type, is below some maximum acceptable level. The
+// acceptable levels are read in from a file, and correspond to a current set
+// of packet masks. The levels for each code may be updated over time.
+TEST_F(FecPacketMaskMetricsTest, FecXorMaxResidualLoss) {
+  SetLossModels();
+  SetCodeParams();
+  ComputeMetricsAllCodes();
+  WriteOutMetricsAllFecCodes();
+  int num_loss_rates = sizeof(kAverageLossRate) / sizeof(*kAverageLossRate);
+  int num_burst_lengths =
+      sizeof(kAverageBurstLength) / sizeof(*kAverageBurstLength);
+  for (int code_index = 0; code_index < max_num_codes_; code_index++) {
+    double sum_residual_loss_random_mask_random_loss = 0.0;
+    double sum_residual_loss_bursty_mask_bursty_loss = 0.0;
+    // Compute the sum residual loss across the models, for each mask type.
+    for (int k = 0; k < kNumLossModels; k++) {
+      if (loss_model_[k].loss_type == kRandomLossModel) {
+        sum_residual_loss_random_mask_random_loss +=
+            kMetricsXorRandom[code_index].average_residual_loss[k];
+      } else if (loss_model_[k].loss_type == kBurstyLossModel) {
+        sum_residual_loss_bursty_mask_bursty_loss +=
+            kMetricsXorBursty[code_index].average_residual_loss[k];
+      }
+    }
+    float average_residual_loss_random_mask_random_loss =
+        sum_residual_loss_random_mask_random_loss / num_loss_rates;
+    float average_residual_loss_bursty_mask_bursty_loss =
+        sum_residual_loss_bursty_mask_bursty_loss /
+        (num_loss_rates * (num_burst_lengths - 1));
+    const float ref_random_mask = kMaxResidualLossRandomMask[code_index];
+    const float ref_bursty_mask = kMaxResidualLossBurstyMask[code_index];
+    EXPECT_LE(average_residual_loss_random_mask_random_loss, ref_random_mask);
+    EXPECT_LE(average_residual_loss_bursty_mask_bursty_loss, ref_bursty_mask);
+  }
+}
+
+// Verify the behavior of the XOR codes vs the RS codes.
+// For random loss model with average loss rates <= the code protection level,
+// the RS code (optimal MDS code) is more efficient than XOR codes.
+// However, for larger loss rates (above protection level) and/or bursty
+// loss models, the RS is not always more efficient than XOR (though in most
+// cases it still is).
+TEST_F(FecPacketMaskMetricsTest, FecXorVsRS) {
+  SetLossModels();
+  SetCodeParams();
+  for (int code_index = 0; code_index < max_num_codes_; code_index++) {
+    for (int k = 0; k < kNumLossModels; k++) {
+      float loss_rate = loss_model_[k].average_loss_rate;
+      float protection_level = code_params_[code_index].protection_level;
+      // Under these conditions we expect XOR to not be better than RS.
+      if (loss_model_[k].loss_type == kRandomLossModel &&
+          loss_rate <= protection_level) {
+        EXPECT_GE(kMetricsXorRandom[code_index].average_residual_loss[k],
+                  kMetricsReedSolomon[code_index].average_residual_loss[k]);
+        EXPECT_GE(kMetricsXorBursty[code_index].average_residual_loss[k],
+                  kMetricsReedSolomon[code_index].average_residual_loss[k]);
+      }
+      // TODO(marpan): There are some cases (for high loss rates and/or
+      // burst loss models) where XOR is better than RS. Is there some pattern
+      // we can identify and enforce as a constraint?
+    }
+  }
+}
+
+// Verify the trend (change) in the average residual loss, as a function of
+// loss rate, of the XOR code relative to the RS code.
+// The difference between XOR and RS should not get worse as we increase
+// the average loss rate.
+TEST_F(FecPacketMaskMetricsTest, FecTrendXorVsRsLossRate) {
+  SetLossModels();
+  SetCodeParams();
+  // TODO(marpan): Examine this further to see if the condition can be strictly
+  // satisfied (i.e., scale = 1.0) for all codes with different/better masks.
+  double scale = 0.90;
+  int num_loss_rates = sizeof(kAverageLossRate) / sizeof(*kAverageLossRate);
+  int num_burst_lengths =
+      sizeof(kAverageBurstLength) / sizeof(*kAverageBurstLength);
+  for (int code_index = 0; code_index < max_num_codes_; code_index++) {
+    for (int i = 0; i < num_burst_lengths; i++) {
+      for (int j = 0; j < num_loss_rates - 1; j++) {
+        int k = num_loss_rates * i + j;
+        // For XOR random.
+        if (kMetricsXorRandom[code_index].average_residual_loss[k] >
+            kMetricsReedSolomon[code_index].average_residual_loss[k]) {
+          double diff_rs_xor_random_loss1 =
+              (kMetricsXorRandom[code_index].average_residual_loss[k] -
+               kMetricsReedSolomon[code_index].average_residual_loss[k]) /
+              kMetricsXorRandom[code_index].average_residual_loss[k];
+          double diff_rs_xor_random_loss2 =
+              (kMetricsXorRandom[code_index].average_residual_loss[k + 1] -
+               kMetricsReedSolomon[code_index].average_residual_loss[k + 1]) /
+              kMetricsXorRandom[code_index].average_residual_loss[k + 1];
+          EXPECT_GE(diff_rs_xor_random_loss1, scale * diff_rs_xor_random_loss2);
+        }
+        // TODO(marpan): Investigate the cases for the bursty mask where
+        // this trend is not strictly satisfied.
+      }
+    }
+  }
+}
+
+// Verify the average residual loss behavior via the protection level and
+// the code length. The average residual loss for a given (k1,m1) code
+// should generally be higher than that of another code (k2,m2), which has
+// either of the two conditions satisfied:
+// 1) higher protection & code length at least as large: (k2+m2) >= (k1+m1),
+// 2) equal protection and larger code length: (k2+m2) > (k1+m1).
+// Currently does not hold for some cases of the XOR code with random mask.
+TEST_F(FecPacketMaskMetricsTest, FecBehaviorViaProtectionLevelAndLength) {
+  SetLossModels();
+  SetCodeParams();
+  for (int code_index1 = 0; code_index1 < max_num_codes_; code_index1++) {
+    float protection_level1 = code_params_[code_index1].protection_level;
+    int length1 = code_params_[code_index1].num_media_packets +
+                  code_params_[code_index1].num_fec_packets;
+    for (int code_index2 = 0; code_index2 < max_num_codes_; code_index2++) {
+      float protection_level2 = code_params_[code_index2].protection_level;
+      int length2 = code_params_[code_index2].num_media_packets +
+                    code_params_[code_index2].num_fec_packets;
+      // Codes with higher protection are more efficient, conditioned on the
+      // length of the code (higher protection but shorter length codes are
+      // generally not more efficient). For two codes with equal protection,
+      // the longer code is generally more efficient. For high loss rate
+      // models, this condition may be violated for some codes with equal or
+      // very close protection levels. High loss rate case is excluded below.
+      if ((protection_level2 > protection_level1 && length2 >= length1) ||
+          (protection_level2 == protection_level1 && length2 > length1)) {
+        for (int k = 0; k < kNumLossModels; k++) {
+          float loss_rate = loss_model_[k].average_loss_rate;
+          if (loss_rate < loss_rate_upper_threshold) {
+            EXPECT_LT(
+                kMetricsReedSolomon[code_index2].average_residual_loss[k],
+                kMetricsReedSolomon[code_index1].average_residual_loss[k]);
+            // TODO(marpan): There are some corner cases where this is not
+            // satisfied with the current packet masks. Look into updating
+            // these cases to see if this behavior should/can be satisfied,
+            // with overall lower residual loss for those XOR codes.
+            // EXPECT_LT(
+            //    kMetricsXorBursty[code_index2].average_residual_loss[k],
+            //    kMetricsXorBursty[code_index1].average_residual_loss[k]);
+            // EXPECT_LT(
+            //   kMetricsXorRandom[code_index2].average_residual_loss[k],
+            //   kMetricsXorRandom[code_index1].average_residual_loss[k]);
+          }
+        }
+      }
+    }
+  }
+}
+
+// Verify the beheavior of the variance of the XOR codes.
+// The partial recovery of the XOR versus the all or nothing behavior of the RS
+// code means that the variance of the residual loss for XOR should generally
+// not be worse than RS.
+TEST_F(FecPacketMaskMetricsTest, FecVarianceBehaviorXorVsRs) {
+  SetLossModels();
+  SetCodeParams();
+  // The condition is not strictly satisfied with the current masks,
+  // i.e., for some codes, the variance of XOR may be slightly higher than RS.
+  // TODO(marpan): Examine this further to see if the condition can be strictly
+  // satisfied (i.e., scale = 1.0) for all codes with different/better masks.
+  double scale = 0.95;
+  for (int code_index = 0; code_index < max_num_codes_; code_index++) {
+    for (int k = 0; k < kNumLossModels; k++) {
+      EXPECT_LE(scale * kMetricsXorRandom[code_index].variance_residual_loss[k],
+                kMetricsReedSolomon[code_index].variance_residual_loss[k]);
+      EXPECT_LE(scale * kMetricsXorBursty[code_index].variance_residual_loss[k],
+                kMetricsReedSolomon[code_index].variance_residual_loss[k]);
+    }
+  }
+}
+
+// For the bursty mask type, the residual loss must be strictly zero for all
+// consecutive losses (i.e, gap = 0) with number of losses <= num_fec_packets.
+// This is a design property of the bursty mask type.
+TEST_F(FecPacketMaskMetricsTest, FecXorBurstyPerfectRecoveryConsecutiveLoss) {
+  SetLossModels();
+  SetCodeParams();
+  for (int code_index = 0; code_index < max_num_codes_; code_index++) {
+    int num_fec_packets = code_params_[code_index].num_fec_packets;
+    for (int loss = 1; loss <= num_fec_packets; loss++) {
+      int index = loss;  // `gap` is zero.
+      EXPECT_EQ(kMetricsXorBursty[code_index].residual_loss_per_loss_gap[index],
+                0.0);
+    }
+  }
+}
+
+// The XOR codes with random mask type are generally better than the ones with
+// bursty mask type, for random loss models at low loss rates.
+// The XOR codes with bursty mask types are generally better than the one with
+// random mask type, for bursty loss models and/or high loss rates.
+// TODO(marpan): Enable this test when some of the packet masks are updated.
+// Some isolated cases of the codes don't pass this currently.
+/*
+TEST_F(FecPacketMaskMetricsTest, FecXorRandomVsBursty) {
+  SetLossModels();
+  SetCodeParams();
+  for (int code_index = 0; code_index < max_num_codes_; code_index++) {
+    double sum_residual_loss_random_mask_random_loss = 0.0;
+    double sum_residual_loss_bursty_mask_random_loss = 0.0;
+    double sum_residual_loss_random_mask_bursty_loss = 0.0;
+    double sum_residual_loss_bursty_mask_bursty_loss = 0.0;
+    // Compute the sum residual loss across the models, for each mask type.
+    for (int k = 0; k < kNumLossModels; k++) {
+      float loss_rate = loss_model_[k].average_loss_rate;
+      if (loss_model_[k].loss_type == kRandomLossModel &&
+          loss_rate < loss_rate_upper_threshold) {
+        sum_residual_loss_random_mask_random_loss +=
+            kMetricsXorRandom[code_index].average_residual_loss[k];
+        sum_residual_loss_bursty_mask_random_loss +=
+            kMetricsXorBursty[code_index].average_residual_loss[k];
+      } else if (loss_model_[k].loss_type == kBurstyLossModel &&
+          loss_rate > loss_rate_lower_threshold) {
+        sum_residual_loss_random_mask_bursty_loss +=
+            kMetricsXorRandom[code_index].average_residual_loss[k];
+        sum_residual_loss_bursty_mask_bursty_loss +=
+            kMetricsXorBursty[code_index].average_residual_loss[k];
+      }
+    }
+    EXPECT_LE(sum_residual_loss_random_mask_random_loss,
+              sum_residual_loss_bursty_mask_random_loss);
+    EXPECT_LE(sum_residual_loss_bursty_mask_bursty_loss,
+              sum_residual_loss_random_mask_bursty_loss);
+  }
+}
+*/
+
+// Verify that the average recovery rate for each code is equal or above some
+// threshold, for certain loss number conditions.
+TEST_F(FecPacketMaskMetricsTest, FecRecoveryRateUnderLossConditions) {
+  SetLossModels();
+  SetCodeParams();
+  for (int code_index = 0; code_index < max_num_codes_; code_index++) {
+    int num_media_packets = code_params_[code_index].num_media_packets;
+    int num_fec_packets = code_params_[code_index].num_fec_packets;
+    // Perfect recovery (`recovery_rate_per_loss` == 1) is expected for
+    // `loss_number` = 1, for all codes.
+    int loss_number = 1;
+    EXPECT_EQ(
+        kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss_number],
+        1.0);
+    EXPECT_EQ(kMetricsXorRandom[code_index].recovery_rate_per_loss[loss_number],
+              1.0);
+    EXPECT_EQ(kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
+              1.0);
+    // For `loss_number` = `num_fec_packets` / 2, we expect the following:
+    // Perfect recovery for RS, and recovery for XOR above the threshold.
+    loss_number = num_fec_packets / 2 > 0 ? num_fec_packets / 2 : 1;
+    EXPECT_EQ(
+        kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss_number],
+        1.0);
+    EXPECT_GE(kMetricsXorRandom[code_index].recovery_rate_per_loss[loss_number],
+              kRecoveryRateXorRandom[0]);
+    EXPECT_GE(kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
+              kRecoveryRateXorBursty[0]);
+    // For `loss_number` = `num_fec_packets`, we expect the following:
+    // Perfect recovery for RS, and recovery for XOR above the threshold.
+    loss_number = num_fec_packets;
+    EXPECT_EQ(
+        kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss_number],
+        1.0);
+    EXPECT_GE(kMetricsXorRandom[code_index].recovery_rate_per_loss[loss_number],
+              kRecoveryRateXorRandom[1]);
+    EXPECT_GE(kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
+              kRecoveryRateXorBursty[1]);
+    // For `loss_number` = `num_fec_packets` + 1, we expect the following:
+    // Zero recovery for RS, but non-zero recovery for XOR.
+    if (num_fec_packets > 1 && num_media_packets > 2) {
+      loss_number = num_fec_packets + 1;
+      EXPECT_EQ(
+          kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss_number],
+          0.0);
+      EXPECT_GE(
+          kMetricsXorRandom[code_index].recovery_rate_per_loss[loss_number],
+          kRecoveryRateXorRandom[2]);
+      EXPECT_GE(
+          kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
+          kRecoveryRateXorBursty[2]);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/third_party/fft/BUILD.gn b/third_party/libwebrtc/modules/third_party/fft/BUILD.gn
new file mode 100644
index 0000000000..49dbd6f9b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/fft/BUILD.gn
@@ -0,0 +1,16 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the ../../../LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("fft") {
+  sources = [
+    "fft.c",
+    "fft.h",
+  ]
+}
diff --git a/third_party/libwebrtc/modules/third_party/fft/LICENSE b/third_party/libwebrtc/modules/third_party/fft/LICENSE
new file mode 100644
index 0000000000..c0a78058cf
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/fft/LICENSE
@@ -0,0 +1,25 @@
+/*
+ * Copyright(c)1995,97 Mark Olesen <olesen@me.QueensU.CA>
+ *    Queen's Univ at Kingston (Canada)
+ *
+ * Permission to use, copy, modify, and distribute this software for
+ * any purpose without fee is hereby granted, provided that this
+ * entire notice is included in all copies of any software which is
+ * or includes a copy or modification of this software and in all
+ * copies of the supporting documentation for such software.
+ *
+ * THIS SOFTWARE IS BEING PROVIDED "AS IS", WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTY.  IN PARTICULAR, NEITHER THE AUTHOR NOR QUEEN'S
+ * UNIVERSITY AT KINGSTON MAKES ANY REPRESENTATION OR WARRANTY OF ANY
+ * KIND CONCERNING THE MERCHANTABILITY OF THIS SOFTWARE OR ITS
+ * FITNESS FOR ANY PARTICULAR PURPOSE.
+ *
+ * All of which is to say that you can do what you like with this
+ * source code provided you don't try to sell it as your own and you
+ * include an unaltered copy of this message (including the
+ * copyright).
+ *
+ * It is also implicitly understood that bug fixes and improvements
+ * should make their way back to the general Internet community so
+ * that everyone benefits.
+ */
diff --git a/third_party/libwebrtc/modules/third_party/fft/README.chromium b/third_party/libwebrtc/modules/third_party/fft/README.chromium
new file mode 100644
index 0000000000..94d20d422f
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/fft/README.chromium
@@ -0,0 +1,12 @@
+Name: fft
+Short Name: fft
+URL:
+Version: 0
+Date: 2018-07-26
+License: Custom license
+License File: LICENSE
+Security Critical: yes
+
+Description:
+Multivariate complex Fourier transform, computed in place
+using mixed-radix Fast Fourier Transform algorithm.
diff --git a/third_party/libwebrtc/modules/third_party/fft/fft.c b/third_party/libwebrtc/modules/third_party/fft/fft.c
new file mode 100644
index 0000000000..72604626d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/fft/fft.c
@@ -0,0 +1,942 @@
+/*
+ * Copyright(c)1995,97 Mark Olesen <olesen@me.QueensU.CA>
+ *    Queen's Univ at Kingston (Canada)
+ *
+ * Permission to use, copy, modify, and distribute this software for
+ * any purpose without fee is hereby granted, provided that this
+ * entire notice is included in all copies of any software which is
+ * or includes a copy or modification of this software and in all
+ * copies of the supporting documentation for such software.
+ *
+ * THIS SOFTWARE IS BEING PROVIDED "AS IS", WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTY.  IN PARTICULAR, NEITHER THE AUTHOR NOR QUEEN'S
+ * UNIVERSITY AT KINGSTON MAKES ANY REPRESENTATION OR WARRANTY OF ANY
+ * KIND CONCERNING THE MERCHANTABILITY OF THIS SOFTWARE OR ITS
+ * FITNESS FOR ANY PARTICULAR PURPOSE.
+ *
+ * All of which is to say that you can do what you like with this
+ * source code provided you don't try to sell it as your own and you
+ * include an unaltered copy of this message (including the
+ * copyright).
+ *
+ * It is also implicitly understood that bug fixes and improvements
+ * should make their way back to the general Internet community so
+ * that everyone benefits.
+ *
+ * Changes:
+ *   Trivial type modifications by the WebRTC authors.
+ */
+
+
+/*
+ * File:
+ * WebRtcIsac_Fftn.c
+ *
+ * Public:
+ * WebRtcIsac_Fftn / fftnf ();
+ *
+ * Private:
+ * WebRtcIsac_Fftradix / fftradixf ();
+ *
+ * Descript:
+ * multivariate complex Fourier transform, computed in place
+ * using mixed-radix Fast Fourier Transform algorithm.
+ *
+ * Fortran code by:
+ * RC Singleton, Stanford Research Institute, Sept. 1968
+ *
+ * translated by f2c (version 19950721).
+ *
+ * int WebRtcIsac_Fftn (int ndim, const int dims[], REAL Re[], REAL Im[],
+ *     int iSign, double scaling);
+ *
+ * NDIM = the total number dimensions
+ * DIMS = a vector of array sizes
+ * if NDIM is zero then DIMS must be zero-terminated
+ *
+ * RE and IM hold the real and imaginary components of the data, and return
+ * the resulting real and imaginary Fourier coefficients.  Multidimensional
+ * data *must* be allocated contiguously.  There is no limit on the number
+ * of dimensions.
+ *
+ * ISIGN = the sign of the complex exponential (ie, forward or inverse FFT)
+ * the magnitude of ISIGN (normally 1) is used to determine the
+ * correct indexing increment (see below).
+ *
+ * SCALING = normalizing constant by which the final result is *divided*
+ * if SCALING == -1, normalize by total dimension of the transform
+ * if SCALING <  -1, normalize by the square-root of the total dimension
+ *
+ * example:
+ * tri-variate transform with Re[n1][n2][n3], Im[n1][n2][n3]
+ *
+ * int dims[3] = {n1,n2,n3}
+ * WebRtcIsac_Fftn (3, dims, Re, Im, 1, scaling);
+ *
+ *-----------------------------------------------------------------------*
+ * int WebRtcIsac_Fftradix (REAL Re[], REAL Im[], size_t nTotal, size_t nPass,
+ *   size_t nSpan, int iSign, size_t max_factors,
+ *   size_t max_perm);
+ *
+ * RE, IM - see above documentation
+ *
+ * Although there is no limit on the number of dimensions, WebRtcIsac_Fftradix() must
+ * be called once for each dimension, but the calls may be in any order.
+ *
+ * NTOTAL = the total number of complex data values
+ * NPASS  = the dimension of the current variable
+ * NSPAN/NPASS = the spacing of consecutive data values while indexing the
+ * current variable
+ * ISIGN - see above documentation
+ *
+ * example:
+ * tri-variate transform with Re[n1][n2][n3], Im[n1][n2][n3]
+ *
+ * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n1,       n1, 1, maxf, maxp);
+ * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n2,    n1*n2, 1, maxf, maxp);
+ * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n3, n1*n2*n3, 1, maxf, maxp);
+ *
+ * single-variate transform,
+ *    NTOTAL = N = NSPAN = (number of complex data values),
+ *
+ * WebRtcIsac_Fftradix (Re, Im, n, n, n, 1, maxf, maxp);
+ *
+ * The data can also be stored in a single array with alternating real and
+ * imaginary parts, the magnitude of ISIGN is changed to 2 to give correct
+ * indexing increment, and data [0] and data [1] used to pass the initial
+ * addresses for the sequences of real and imaginary values,
+ *
+ * example:
+ * REAL data [2*NTOTAL];
+ * WebRtcIsac_Fftradix ( &data[0], &data[1], NTOTAL, nPass, nSpan, 2, maxf, maxp);
+ *
+ * for temporary allocation:
+ *
+ * MAX_FACTORS >= the maximum prime factor of NPASS
+ * MAX_PERM >= the number of prime factors of NPASS.  In addition,
+ * if the square-free portion K of NPASS has two or more prime
+ * factors, then MAX_PERM >= (K-1)
+ *
+ * storage in FACTOR for a maximum of 15 prime factors of NPASS. if NPASS
+ * has more than one square-free factor, the product of the square-free
+ * factors must be <= 210 array storage for maximum prime factor of 23 the
+ * following two constants should agree with the array dimensions.
+ *
+ *----------------------------------------------------------------------*/
+
+#include <stdlib.h>
+#include <math.h>
+
+#include "modules/third_party/fft/fft.h"
+
+/* double precision routine */
+static int
+WebRtcIsac_Fftradix (double Re[], double Im[],
+                    size_t nTotal, size_t nPass, size_t nSpan, int isign,
+                    int max_factors, unsigned int max_perm,
+                    FFTstr *fftstate);
+
+
+
+#ifndef M_PI
+# define M_PI 3.14159265358979323846264338327950288
+#endif
+
+#ifndef SIN60
+# define SIN60 0.86602540378443865 /* sin(60 deg) */
+# define COS72 0.30901699437494742 /* cos(72 deg) */
+# define SIN72 0.95105651629515357 /* sin(72 deg) */
+#endif
+
+# define REAL  double
+# define FFTN  WebRtcIsac_Fftn
+# define FFTNS  "fftn"
+# define FFTRADIX WebRtcIsac_Fftradix
+# define FFTRADIXS "fftradix"
+
+
+int  WebRtcIsac_Fftns(unsigned int ndim, const int dims[],
+                     double Re[],
+                     double Im[],
+                     int iSign,
+                     double scaling,
+                     FFTstr *fftstate)
+{
+
+  size_t nSpan, nPass, nTotal;
+  unsigned int i;
+  int ret, max_factors, max_perm;
+
+  /*
+   * tally the number of elements in the data array
+   * and determine the number of dimensions
+   */
+  nTotal = 1;
+  if (ndim && dims [0])
+  {
+    for (i = 0; i < ndim; i++)
+    {
+      if (dims [i] <= 0)
+      {
+        return -1;
+      }
+      nTotal *= dims [i];
+    }
+  }
+  else
+  {
+    ndim = 0;
+    for (i = 0; dims [i]; i++)
+    {
+      if (dims [i] <= 0)
+      {
+        return -1;
+      }
+      nTotal *= dims [i];
+      ndim++;
+    }
+  }
+
+  /* determine maximum number of factors and permuations */
+#if 1
+  /*
+   * follow John Beale's example, just use the largest dimension and don't
+   * worry about excess allocation.  May be someone else will do it?
+   */
+  max_factors = max_perm = 1;
+  for (i = 0; i < ndim; i++)
+  {
+    nSpan = dims [i];
+    if ((int)nSpan > max_factors)
+    {
+      max_factors = (int)nSpan;
+    }
+    if ((int)nSpan > max_perm)
+    {
+      max_perm = (int)nSpan;
+    }
+  }
+#else
+  /* use the constants used in the original Fortran code */
+  max_factors = 23;
+  max_perm = 209;
+#endif
+  /* loop over the dimensions: */
+  nPass = 1;
+  for (i = 0; i < ndim; i++)
+  {
+    nSpan = dims [i];
+    nPass *= nSpan;
+    ret = FFTRADIX (Re, Im, nTotal, nSpan, nPass, iSign,
+                    max_factors, max_perm, fftstate);
+    /* exit, clean-up already done */
+    if (ret)
+      return ret;
+  }
+
+  /* Divide through by the normalizing constant: */
+  if (scaling && scaling != 1.0)
+  {
+    if (iSign < 0) iSign = -iSign;
+    if (scaling < 0.0)
+    {
+      scaling = (double)nTotal;
+      if (scaling < -1.0)
+        scaling = sqrt (scaling);
+    }
+    scaling = 1.0 / scaling; /* multiply is often faster */
+    for (i = 0; i < nTotal; i += iSign)
+    {
+      Re [i] *= scaling;
+      Im [i] *= scaling;
+    }
+  }
+  return 0;
+}
+
+/*
+ * singleton's mixed radix routine
+ *
+ * could move allocation out to WebRtcIsac_Fftn(), but leave it here so that it's
+ * possible to make this a standalone function
+ */
+
+static int   FFTRADIX (REAL Re[],
+                       REAL Im[],
+                       size_t nTotal,
+                       size_t nPass,
+                       size_t nSpan,
+                       int iSign,
+                       int max_factors,
+                       unsigned int max_perm,
+                       FFTstr *fftstate)
+{
+  int ii, mfactor, kspan, ispan, inc;
+  int j, jc, jf, jj, k, k1, k2, k3, k4, kk, kt, nn, ns, nt;
+
+
+  REAL radf;
+  REAL c1, c2, c3, cd, aa, aj, ak, ajm, ajp, akm, akp;
+  REAL s1, s2, s3, sd, bb, bj, bk, bjm, bjp, bkm, bkp;
+
+  REAL *Rtmp = NULL; /* temp space for real part*/
+  REAL *Itmp = NULL; /* temp space for imaginary part */
+  REAL *Cos = NULL; /* Cosine values */
+  REAL *Sin = NULL; /* Sine values */
+
+  REAL s60 = SIN60;  /* sin(60 deg) */
+  REAL c72 = COS72;  /* cos(72 deg) */
+  REAL s72 = SIN72;  /* sin(72 deg) */
+  REAL pi2 = M_PI;  /* use PI first, 2 PI later */
+
+
+  fftstate->SpaceAlloced = 0;
+  fftstate->MaxPermAlloced = 0;
+
+
+  // initialize to avoid warnings
+  k3 = c2 = c3 = s2 = s3 = 0.0;
+
+  if (nPass < 2)
+    return 0;
+
+  /*  allocate storage */
+  if (fftstate->SpaceAlloced < max_factors * sizeof (REAL))
+  {
+#ifdef SUN_BROKEN_REALLOC
+    if (!fftstate->SpaceAlloced) /* first time */
+    {
+      fftstate->SpaceAlloced = max_factors * sizeof (REAL);
+    }
+    else
+    {
+#endif
+      fftstate->SpaceAlloced = max_factors * sizeof (REAL);
+#ifdef SUN_BROKEN_REALLOC
+    }
+#endif
+  }
+  else
+  {
+    /* allow full use of alloc'd space */
+    max_factors = fftstate->SpaceAlloced / sizeof (REAL);
+  }
+  if (fftstate->MaxPermAlloced < max_perm)
+  {
+#ifdef SUN_BROKEN_REALLOC
+    if (!fftstate->MaxPermAlloced) /* first time */
+    else
+#endif
+      fftstate->MaxPermAlloced = max_perm;
+  }
+  else
+  {
+    /* allow full use of alloc'd space */
+    max_perm = fftstate->MaxPermAlloced;
+  }
+
+  /* assign pointers */
+  Rtmp = (REAL *) fftstate->Tmp0;
+  Itmp = (REAL *) fftstate->Tmp1;
+  Cos  = (REAL *) fftstate->Tmp2;
+  Sin  = (REAL *) fftstate->Tmp3;
+
+  /*
+   * Function Body
+   */
+  inc = iSign;
+  if (iSign < 0) {
+    s72 = -s72;
+    s60 = -s60;
+    pi2 = -pi2;
+    inc = -inc;  /* absolute value */
+  }
+
+  /* adjust for strange increments */
+  nt = inc * (int)nTotal;
+  ns = inc * (int)nSpan;
+  kspan = ns;
+
+  nn = nt - inc;
+  jc = ns / (int)nPass;
+  radf = pi2 * (double) jc;
+  pi2 *= 2.0;   /* use 2 PI from here on */
+
+  ii = 0;
+  jf = 0;
+  /*  determine the factors of n */
+  mfactor = 0;
+  k = (int)nPass;
+  while (k % 16 == 0) {
+    mfactor++;
+    fftstate->factor [mfactor - 1] = 4;
+    k /= 16;
+  }
+  j = 3;
+  jj = 9;
+  do {
+    while (k % jj == 0) {
+      mfactor++;
+      fftstate->factor [mfactor - 1] = j;
+      k /= jj;
+    }
+    j += 2;
+    jj = j * j;
+  } while (jj <= k);
+  if (k <= 4) {
+    kt = mfactor;
+    fftstate->factor [mfactor] = k;
+    if (k != 1)
+      mfactor++;
+  } else {
+    if (k - (k / 4 << 2) == 0) {
+      mfactor++;
+      fftstate->factor [mfactor - 1] = 2;
+      k /= 4;
+    }
+    kt = mfactor;
+    j = 2;
+    do {
+      if (k % j == 0) {
+        mfactor++;
+        fftstate->factor [mfactor - 1] = j;
+        k /= j;
+      }
+      j = ((j + 1) / 2 << 1) + 1;
+    } while (j <= k);
+  }
+  if (kt) {
+    j = kt;
+    do {
+      mfactor++;
+      fftstate->factor [mfactor - 1] = fftstate->factor [j - 1];
+      j--;
+    } while (j);
+  }
+
+  /* test that mfactors is in range */
+  if (mfactor > FFT_NFACTOR)
+  {
+    return -1;
+  }
+
+  /* compute fourier transform */
+  for (;;) {
+    sd = radf / (double) kspan;
+    cd = sin(sd);
+    cd = 2.0 * cd * cd;
+    sd = sin(sd + sd);
+    kk = 0;
+    ii++;
+
+    switch (fftstate->factor [ii - 1]) {
+      case 2:
+        /* transform for factor of 2 (including rotation factor) */
+        kspan /= 2;
+        k1 = kspan + 2;
+        do {
+          do {
+            k2 = kk + kspan;
+            ak = Re [k2];
+            bk = Im [k2];
+            Re [k2] = Re [kk] - ak;
+            Im [k2] = Im [kk] - bk;
+            Re [kk] += ak;
+            Im [kk] += bk;
+            kk = k2 + kspan;
+          } while (kk < nn);
+          kk -= nn;
+        } while (kk < jc);
+        if (kk >= kspan)
+          goto Permute_Results_Label;  /* exit infinite loop */
+        do {
+          c1 = 1.0 - cd;
+          s1 = sd;
+          do {
+            do {
+              do {
+                k2 = kk + kspan;
+                ak = Re [kk] - Re [k2];
+                bk = Im [kk] - Im [k2];
+                Re [kk] += Re [k2];
+                Im [kk] += Im [k2];
+                Re [k2] = c1 * ak - s1 * bk;
+                Im [k2] = s1 * ak + c1 * bk;
+                kk = k2 + kspan;
+              } while (kk < (nt-1));
+              k2 = kk - nt;
+              c1 = -c1;
+              kk = k1 - k2;
+            } while (kk > k2);
+            ak = c1 - (cd * c1 + sd * s1);
+            s1 = sd * c1 - cd * s1 + s1;
+            c1 = 2.0 - (ak * ak + s1 * s1);
+            s1 *= c1;
+            c1 *= ak;
+            kk += jc;
+          } while (kk < k2);
+          k1 += inc + inc;
+          kk = (k1 - kspan + 1) / 2 + jc - 1;
+        } while (kk < (jc + jc));
+        break;
+
+      case 4:   /* transform for factor of 4 */
+        ispan = kspan;
+        kspan /= 4;
+
+        do {
+          c1 = 1.0;
+          s1 = 0.0;
+          do {
+            do {
+              k1 = kk + kspan;
+              k2 = k1 + kspan;
+              k3 = k2 + kspan;
+              akp = Re [kk] + Re [k2];
+              akm = Re [kk] - Re [k2];
+              ajp = Re [k1] + Re [k3];
+              ajm = Re [k1] - Re [k3];
+              bkp = Im [kk] + Im [k2];
+              bkm = Im [kk] - Im [k2];
+              bjp = Im [k1] + Im [k3];
+              bjm = Im [k1] - Im [k3];
+              Re [kk] = akp + ajp;
+              Im [kk] = bkp + bjp;
+              ajp = akp - ajp;
+              bjp = bkp - bjp;
+              if (iSign < 0) {
+                akp = akm + bjm;
+                bkp = bkm - ajm;
+                akm -= bjm;
+                bkm += ajm;
+              } else {
+                akp = akm - bjm;
+                bkp = bkm + ajm;
+                akm += bjm;
+                bkm -= ajm;
+              }
+              /* avoid useless multiplies */
+              if (s1 == 0.0) {
+                Re [k1] = akp;
+                Re [k2] = ajp;
+                Re [k3] = akm;
+                Im [k1] = bkp;
+                Im [k2] = bjp;
+                Im [k3] = bkm;
+              } else {
+                Re [k1] = akp * c1 - bkp * s1;
+                Re [k2] = ajp * c2 - bjp * s2;
+                Re [k3] = akm * c3 - bkm * s3;
+                Im [k1] = akp * s1 + bkp * c1;
+                Im [k2] = ajp * s2 + bjp * c2;
+                Im [k3] = akm * s3 + bkm * c3;
+              }
+              kk = k3 + kspan;
+            } while (kk < nt);
+
+            c2 = c1 - (cd * c1 + sd * s1);
+            s1 = sd * c1 - cd * s1 + s1;
+            c1 = 2.0 - (c2 * c2 + s1 * s1);
+            s1 *= c1;
+            c1 *= c2;
+            /* values of c2, c3, s2, s3 that will get used next time */
+            c2 = c1 * c1 - s1 * s1;
+            s2 = 2.0 * c1 * s1;
+            c3 = c2 * c1 - s2 * s1;
+            s3 = c2 * s1 + s2 * c1;
+            kk = kk - nt + jc;
+          } while (kk < kspan);
+          kk = kk - kspan + inc;
+        } while (kk < jc);
+        if (kspan == jc)
+          goto Permute_Results_Label;  /* exit infinite loop */
+        break;
+
+      default:
+        /*  transform for odd factors */
+#ifdef FFT_RADIX4
+        return -1;
+        break;
+#else /* FFT_RADIX4 */
+        k = fftstate->factor [ii - 1];
+        ispan = kspan;
+        kspan /= k;
+
+        switch (k) {
+          case 3: /* transform for factor of 3 (optional code) */
+            do {
+              do {
+                k1 = kk + kspan;
+                k2 = k1 + kspan;
+                ak = Re [kk];
+                bk = Im [kk];
+                aj = Re [k1] + Re [k2];
+                bj = Im [k1] + Im [k2];
+                Re [kk] = ak + aj;
+                Im [kk] = bk + bj;
+                ak -= 0.5 * aj;
+                bk -= 0.5 * bj;
+                aj = (Re [k1] - Re [k2]) * s60;
+                bj = (Im [k1] - Im [k2]) * s60;
+                Re [k1] = ak - bj;
+                Re [k2] = ak + bj;
+                Im [k1] = bk + aj;
+                Im [k2] = bk - aj;
+                kk = k2 + kspan;
+              } while (kk < (nn - 1));
+              kk -= nn;
+            } while (kk < kspan);
+            break;
+
+          case 5: /*  transform for factor of 5 (optional code) */
+            c2 = c72 * c72 - s72 * s72;
+            s2 = 2.0 * c72 * s72;
+            do {
+              do {
+                k1 = kk + kspan;
+                k2 = k1 + kspan;
+                k3 = k2 + kspan;
+                k4 = k3 + kspan;
+                akp = Re [k1] + Re [k4];
+                akm = Re [k1] - Re [k4];
+                bkp = Im [k1] + Im [k4];
+                bkm = Im [k1] - Im [k4];
+                ajp = Re [k2] + Re [k3];
+                ajm = Re [k2] - Re [k3];
+                bjp = Im [k2] + Im [k3];
+                bjm = Im [k2] - Im [k3];
+                aa = Re [kk];
+                bb = Im [kk];
+                Re [kk] = aa + akp + ajp;
+                Im [kk] = bb + bkp + bjp;
+                ak = akp * c72 + ajp * c2 + aa;
+                bk = bkp * c72 + bjp * c2 + bb;
+                aj = akm * s72 + ajm * s2;
+                bj = bkm * s72 + bjm * s2;
+                Re [k1] = ak - bj;
+                Re [k4] = ak + bj;
+                Im [k1] = bk + aj;
+                Im [k4] = bk - aj;
+                ak = akp * c2 + ajp * c72 + aa;
+                bk = bkp * c2 + bjp * c72 + bb;
+                aj = akm * s2 - ajm * s72;
+                bj = bkm * s2 - bjm * s72;
+                Re [k2] = ak - bj;
+                Re [k3] = ak + bj;
+                Im [k2] = bk + aj;
+                Im [k3] = bk - aj;
+                kk = k4 + kspan;
+              } while (kk < (nn-1));
+              kk -= nn;
+            } while (kk < kspan);
+            break;
+
+          default:
+            if (k != jf) {
+              jf = k;
+              s1 = pi2 / (double) k;
+              c1 = cos(s1);
+              s1 = sin(s1);
+              if (jf > max_factors){
+                return -1;
+              }
+              Cos [jf - 1] = 1.0;
+              Sin [jf - 1] = 0.0;
+              j = 1;
+              do {
+                Cos [j - 1] = Cos [k - 1] * c1 + Sin [k - 1] * s1;
+                Sin [j - 1] = Cos [k - 1] * s1 - Sin [k - 1] * c1;
+                k--;
+                Cos [k - 1] = Cos [j - 1];
+                Sin [k - 1] = -Sin [j - 1];
+                j++;
+              } while (j < k);
+            }
+            do {
+              do {
+                k1 = kk;
+                k2 = kk + ispan;
+                ak = aa = Re [kk];
+                bk = bb = Im [kk];
+                j = 1;
+                k1 += kspan;
+                do {
+                  k2 -= kspan;
+                  j++;
+                  Rtmp [j - 1] = Re [k1] + Re [k2];
+                  ak += Rtmp [j - 1];
+                  Itmp [j - 1] = Im [k1] + Im [k2];
+                  bk += Itmp [j - 1];
+                  j++;
+                  Rtmp [j - 1] = Re [k1] - Re [k2];
+                  Itmp [j - 1] = Im [k1] - Im [k2];
+                  k1 += kspan;
+                } while (k1 < k2);
+                Re [kk] = ak;
+                Im [kk] = bk;
+                k1 = kk;
+                k2 = kk + ispan;
+                j = 1;
+                do {
+                  k1 += kspan;
+                  k2 -= kspan;
+                  jj = j;
+                  ak = aa;
+                  bk = bb;
+                  aj = 0.0;
+                  bj = 0.0;
+                  k = 1;
+                  do {
+                    k++;
+                    ak += Rtmp [k - 1] * Cos [jj - 1];
+                    bk += Itmp [k - 1] * Cos [jj - 1];
+                    k++;
+                    aj += Rtmp [k - 1] * Sin [jj - 1];
+                    bj += Itmp [k - 1] * Sin [jj - 1];
+                    jj += j;
+                    if (jj > jf) {
+                      jj -= jf;
+                    }
+                  } while (k < jf);
+                  k = jf - j;
+                  Re [k1] = ak - bj;
+                  Im [k1] = bk + aj;
+                  Re [k2] = ak + bj;
+                  Im [k2] = bk - aj;
+                  j++;
+                } while (j < k);
+                kk += ispan;
+              } while (kk < nn);
+              kk -= nn;
+            } while (kk < kspan);
+            break;
+        }
+
+        /*  multiply by rotation factor (except for factors of 2 and 4) */
+        if (ii == mfactor)
+          goto Permute_Results_Label;  /* exit infinite loop */
+        kk = jc;
+        do {
+          c2 = 1.0 - cd;
+          s1 = sd;
+          do {
+            c1 = c2;
+            s2 = s1;
+            kk += kspan;
+            do {
+              do {
+                ak = Re [kk];
+                Re [kk] = c2 * ak - s2 * Im [kk];
+                Im [kk] = s2 * ak + c2 * Im [kk];
+                kk += ispan;
+              } while (kk < nt);
+              ak = s1 * s2;
+              s2 = s1 * c2 + c1 * s2;
+              c2 = c1 * c2 - ak;
+              kk = kk - nt + kspan;
+            } while (kk < ispan);
+            c2 = c1 - (cd * c1 + sd * s1);
+            s1 += sd * c1 - cd * s1;
+            c1 = 2.0 - (c2 * c2 + s1 * s1);
+            s1 *= c1;
+            c2 *= c1;
+            kk = kk - ispan + jc;
+          } while (kk < kspan);
+          kk = kk - kspan + jc + inc;
+        } while (kk < (jc + jc));
+        break;
+#endif /* FFT_RADIX4 */
+    }
+  }
+
+  /*  permute the results to normal order---done in two stages */
+  /*  permutation for square factors of n */
+Permute_Results_Label:
+  fftstate->Perm [0] = ns;
+  if (kt) {
+    k = kt + kt + 1;
+    if (mfactor < k)
+      k--;
+    j = 1;
+    fftstate->Perm [k] = jc;
+    do {
+      fftstate->Perm [j] = fftstate->Perm [j - 1] / fftstate->factor [j - 1];
+      fftstate->Perm [k - 1] = fftstate->Perm [k] * fftstate->factor [j - 1];
+      j++;
+      k--;
+    } while (j < k);
+    k3 = fftstate->Perm [k];
+    kspan = fftstate->Perm [1];
+    kk = jc;
+    k2 = kspan;
+    j = 1;
+    if (nPass != nTotal) {
+      /*  permutation for multivariate transform */
+   Permute_Multi_Label:
+      do {
+        do {
+          k = kk + jc;
+          do {
+            /* swap Re [kk] <> Re [k2], Im [kk] <> Im [k2] */
+            ak = Re [kk]; Re [kk] = Re [k2]; Re [k2] = ak;
+            bk = Im [kk]; Im [kk] = Im [k2]; Im [k2] = bk;
+            kk += inc;
+            k2 += inc;
+          } while (kk < (k-1));
+          kk += ns - jc;
+          k2 += ns - jc;
+        } while (kk < (nt-1));
+        k2 = k2 - nt + kspan;
+        kk = kk - nt + jc;
+      } while (k2 < (ns-1));
+      do {
+        do {
+          k2 -= fftstate->Perm [j - 1];
+          j++;
+          k2 = fftstate->Perm [j] + k2;
+        } while (k2 > fftstate->Perm [j - 1]);
+        j = 1;
+        do {
+          if (kk < (k2-1))
+            goto Permute_Multi_Label;
+          kk += jc;
+          k2 += kspan;
+        } while (k2 < (ns-1));
+      } while (kk < (ns-1));
+    } else {
+      /*  permutation for single-variate transform (optional code) */
+   Permute_Single_Label:
+      do {
+        /* swap Re [kk] <> Re [k2], Im [kk] <> Im [k2] */
+        ak = Re [kk]; Re [kk] = Re [k2]; Re [k2] = ak;
+        bk = Im [kk]; Im [kk] = Im [k2]; Im [k2] = bk;
+        kk += inc;
+        k2 += kspan;
+      } while (k2 < (ns-1));
+      do {
+        do {
+          k2 -= fftstate->Perm [j - 1];
+          j++;
+          k2 = fftstate->Perm [j] + k2;
+        } while (k2 >= fftstate->Perm [j - 1]);
+        j = 1;
+        do {
+          if (kk < k2)
+            goto Permute_Single_Label;
+          kk += inc;
+          k2 += kspan;
+        } while (k2 < (ns-1));
+      } while (kk < (ns-1));
+    }
+    jc = k3;
+  }
+
+  if ((kt << 1) + 1 >= mfactor)
+    return 0;
+  ispan = fftstate->Perm [kt];
+  /* permutation for square-free factors of n */
+  j = mfactor - kt;
+  fftstate->factor [j] = 1;
+  do {
+    fftstate->factor [j - 1] *= fftstate->factor [j];
+    j--;
+  } while (j != kt);
+  kt++;
+  nn = fftstate->factor [kt - 1] - 1;
+  if (nn > (int) max_perm) {
+    return -1;
+  }
+  j = jj = 0;
+  for (;;) {
+    k = kt + 1;
+    k2 = fftstate->factor [kt - 1];
+    kk = fftstate->factor [k - 1];
+    j++;
+    if (j > nn)
+      break;    /* exit infinite loop */
+    jj += kk;
+    while (jj >= k2) {
+      jj -= k2;
+      k2 = kk;
+      k++;
+      kk = fftstate->factor [k - 1];
+      jj += kk;
+    }
+    fftstate->Perm [j - 1] = jj;
+  }
+  /*  determine the permutation cycles of length greater than 1 */
+  j = 0;
+  for (;;) {
+    do {
+      j++;
+      kk = fftstate->Perm [j - 1];
+    } while (kk < 0);
+    if (kk != j) {
+      do {
+        k = kk;
+        kk = fftstate->Perm [k - 1];
+        fftstate->Perm [k - 1] = -kk;
+      } while (kk != j);
+      k3 = kk;
+    } else {
+      fftstate->Perm [j - 1] = -j;
+      if (j == nn)
+        break;  /* exit infinite loop */
+    }
+  }
+  max_factors *= inc;
+  /*  reorder a and b, following the permutation cycles */
+  for (;;) {
+    j = k3 + 1;
+    nt -= ispan;
+    ii = nt - inc + 1;
+    if (nt < 0)
+      break;   /* exit infinite loop */
+    do {
+      do {
+        j--;
+      } while (fftstate->Perm [j - 1] < 0);
+      jj = jc;
+      do {
+        kspan = jj;
+        if (jj > max_factors) {
+          kspan = max_factors;
+        }
+        jj -= kspan;
+        k = fftstate->Perm [j - 1];
+        kk = jc * k + ii + jj;
+        k1 = kk + kspan - 1;
+        k2 = 0;
+        do {
+          k2++;
+          Rtmp [k2 - 1] = Re [k1];
+          Itmp [k2 - 1] = Im [k1];
+          k1 -= inc;
+        } while (k1 != (kk-1));
+        do {
+          k1 = kk + kspan - 1;
+          k2 = k1 - jc * (k + fftstate->Perm [k - 1]);
+          k = -fftstate->Perm [k - 1];
+          do {
+            Re [k1] = Re [k2];
+            Im [k1] = Im [k2];
+            k1 -= inc;
+            k2 -= inc;
+          } while (k1 != (kk-1));
+          kk = k2 + 1;
+        } while (k != j);
+        k1 = kk + kspan - 1;
+        k2 = 0;
+        do {
+          k2++;
+          Re [k1] = Rtmp [k2 - 1];
+          Im [k1] = Itmp [k2 - 1];
+          k1 -= inc;
+        } while (k1 != (kk-1));
+      } while (jj);
+    } while (j != 1);
+  }
+  return 0;   /* exit point here */
+}
+/* ---------------------- end-of-file (c source) ---------------------- */
+
diff --git a/third_party/libwebrtc/modules/third_party/fft/fft.h b/third_party/libwebrtc/modules/third_party/fft/fft.h
new file mode 100644
index 0000000000..f8f8b6f17b
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/fft/fft.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the ../../../LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*--------------------------------*-C-*---------------------------------*
+ * File:
+ *       fftn.h
+ * ---------------------------------------------------------------------*
+ * Re[]:        real value array
+ * Im[]:        imaginary value array
+ * nTotal:      total number of complex values
+ * nPass:       number of elements involved in this pass of transform
+ * nSpan:       nspan/nPass = number of bytes to increment pointer
+ *              in Re[] and Im[]
+ * isign: exponent: +1 = forward  -1 = reverse
+ * scaling: normalizing constant by which the final result is *divided*
+ * scaling == -1, normalize by total dimension of the transform
+ * scaling <  -1, normalize by the square-root of the total dimension
+ *
+ * ----------------------------------------------------------------------
+ * See the comments in the code for correct usage!
+ */
+
+#ifndef MODULES_THIRD_PARTY_FFT_FFT_H_
+#define MODULES_THIRD_PARTY_FFT_FFT_H_
+
+#define FFT_MAXFFTSIZE 2048
+#define FFT_NFACTOR 11
+
+typedef struct {
+  unsigned int SpaceAlloced;
+  unsigned int MaxPermAlloced;
+  double Tmp0[FFT_MAXFFTSIZE];
+  double Tmp1[FFT_MAXFFTSIZE];
+  double Tmp2[FFT_MAXFFTSIZE];
+  double Tmp3[FFT_MAXFFTSIZE];
+  int Perm[FFT_MAXFFTSIZE];
+  int factor[FFT_NFACTOR];
+
+} FFTstr;
+
+/* double precision routine */
+
+int WebRtcIsac_Fftns(unsigned int ndim,
+                     const int dims[],
+                     double Re[],
+                     double Im[],
+                     int isign,
+                     double scaling,
+                     FFTstr* fftstate);
+
+#endif /* MODULES_THIRD_PARTY_FFT_FFT_H_ */
diff --git a/third_party/libwebrtc/modules/third_party/fft/fft_gn/moz.build b/third_party/libwebrtc/modules/third_party/fft/fft_gn/moz.build
new file mode 100644
index 0000000000..4002282040
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/fft/fft_gn/moz.build
@@ -0,0 +1,217 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/third_party/fft/fft.c"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("fft_gn")
diff --git a/third_party/libwebrtc/modules/third_party/g711/BUILD.gn b/third_party/libwebrtc/modules/third_party/g711/BUILD.gn
new file mode 100644
index 0000000000..f7735a6fe7
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g711/BUILD.gn
@@ -0,0 +1,17 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the ../../../LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("g711_3p") {
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "g711.c",
+    "g711.h",
+  ]
+}
diff --git a/third_party/libwebrtc/modules/third_party/g711/LICENSE b/third_party/libwebrtc/modules/third_party/g711/LICENSE
new file mode 100644
index 0000000000..3cdf910304
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g711/LICENSE
@@ -0,0 +1,14 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g711.h - In line A-law and u-law conversion routines
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2001 Steve Underwood
+ *
+ *  Despite my general liking of the GPL, I place this code in the
+ *  public domain for the benefit of all mankind - even the slimy
+ *  ones who might try to proprietize my work and use it to my
+ *  detriment.
+ */
diff --git a/third_party/libwebrtc/modules/third_party/g711/README.chromium b/third_party/libwebrtc/modules/third_party/g711/README.chromium
new file mode 100644
index 0000000000..1baa2637a9
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g711/README.chromium
@@ -0,0 +1,11 @@
+Name: In line A-law and u-law conversion routines
+Short Name: g711
+URL:
+Version: 0
+Date: 2018-06-25
+License: Custom license
+License File: LICENSE
+Security Critical: yes
+
+Description:
+In line A-law and u-law conversion routines
diff --git a/third_party/libwebrtc/modules/third_party/g711/g711.c b/third_party/libwebrtc/modules/third_party/g711/g711.c
new file mode 100644
index 0000000000..b7ae6bb109
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g711/g711.c
@@ -0,0 +1,72 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g711.c - A-law and u-law transcoding routines
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ *  Despite my general liking of the GPL, I place this code in the
+ *  public domain for the benefit of all mankind - even the slimy
+ *  ones who might try to proprietize my work and use it to my
+ *  detriment.
+ *
+ * $Id: g711.c,v 1.1 2006/06/07 15:46:39 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Removed unused include files
+ * -Changed to use WebRtc types
+ * -Added option to run encoder bitexact with ITU-T reference implementation
+ */
+
+#include "modules/third_party/g711/g711.h"
+
+/* Copied from the CCITT G.711 specification */
+static const uint8_t ulaw_to_alaw_table[256] = {
+   42,  43,  40,  41,  46,  47,  44,  45,  34,  35,  32,  33,  38,  39,  36,
+   37,  58,  59,  56,  57,  62,  63,  60,  61,  50,  51,  48,  49,  54,  55,
+   52,  53,  10,  11,   8,   9,  14,  15,  12,  13,   2,   3,   0,   1,   6,
+    7,   4,  26,  27,  24,  25,  30,  31,  28,  29,  18,  19,  16,  17,  22,
+   23,  20,  21, 106, 104, 105, 110, 111, 108, 109,  98,  99,  96,  97, 102,
+  103, 100, 101, 122, 120, 126, 127, 124, 125, 114, 115, 112, 113, 118, 119,
+  116, 117,  75,  73,  79,  77,  66,  67,  64,  65,  70,  71,  68,  69,  90,
+   91,  88,  89,  94,  95,  92,  93,  82,  82,  83,  83,  80,  80,  81,  81,
+   86,  86,  87,  87,  84,  84,  85,  85, 170, 171, 168, 169, 174, 175, 172,
+  173, 162, 163, 160, 161, 166, 167, 164, 165, 186, 187, 184, 185, 190, 191,
+  188, 189, 178, 179, 176, 177, 182, 183, 180, 181, 138, 139, 136, 137, 142,
+  143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 154, 155, 152, 153, 158,
+  159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149, 234, 232, 233, 238,
+  239, 236, 237, 226, 227, 224, 225, 230, 231, 228, 229, 250, 248, 254, 255,
+  252, 253, 242, 243, 240, 241, 246, 247, 244, 245, 203, 201, 207, 205, 194,
+  195, 192, 193, 198, 199, 196, 197, 218, 219, 216, 217, 222, 223, 220, 221,
+  210, 210, 211, 211, 208, 208, 209, 209, 214, 214, 215, 215, 212, 212, 213,
+  213
+};
+
+/* These transcoding tables are copied from the CCITT G.711 specification. To
+   achieve optimal results, do not change them. */
+static const uint8_t alaw_to_ulaw_table[256] = {
+   42,  43,  40,  41,  46,  47,  44,  45,  34,  35,  32,  33,  38,  39,  36,
+   37,  57,  58,  55,  56,  61,  62,  59,  60,  49,  50,  47,  48,  53,  54,
+   51,  52,  10,  11,   8,   9,  14,  15,  12,  13,   2,   3,   0,   1,   6,
+    7,   4,   5,  26,  27,  24,  25,  30,  31,  28,  29,  18,  19,  16,  17,
+   22,  23,  20,  21,  98,  99,  96,  97, 102, 103, 100, 101,  93,  93,  92,
+   92,  95,  95,  94,  94, 116, 118, 112, 114, 124, 126, 120, 122, 106, 107,
+  104, 105, 110, 111, 108, 109,  72,  73,  70,  71,  76,  77,  74,  75,  64,
+   65,  63,  63,  68,  69,  66,  67,  86,  87,  84,  85,  90,  91,  88,  89,
+   79,  79,  78,  78,  82,  83,  80,  81, 170, 171, 168, 169, 174, 175, 172,
+  173, 162, 163, 160, 161, 166, 167, 164, 165, 185, 186, 183, 184, 189, 190,
+  187, 188, 177, 178, 175, 176, 181, 182, 179, 180, 138, 139, 136, 137, 142,
+  143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 133, 154, 155, 152, 153,
+  158, 159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149, 226, 227, 224,
+  225, 230, 231, 228, 229, 221, 221, 220, 220, 223, 223, 222, 222, 244, 246,
+  240, 242, 252, 254, 248, 250, 234, 235, 232, 233, 238, 239, 236, 237, 200,
+  201, 198, 199, 204, 205, 202, 203, 192, 193, 191, 191, 196, 197, 194, 195,
+  214, 215, 212, 213, 218, 219, 216, 217, 207, 207, 206, 206, 210, 211, 208,
+  209
+};
+
+uint8_t alaw_to_ulaw(uint8_t alaw) { return alaw_to_ulaw_table[alaw]; }
+
+uint8_t ulaw_to_alaw(uint8_t ulaw) { return ulaw_to_alaw_table[ulaw]; }
diff --git a/third_party/libwebrtc/modules/third_party/g711/g711.h b/third_party/libwebrtc/modules/third_party/g711/g711.h
new file mode 100644
index 0000000000..4eef42c0bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g711/g711.h
@@ -0,0 +1,350 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g711.h - In line A-law and u-law conversion routines
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2001 Steve Underwood
+ *
+ *  Despite my general liking of the GPL, I place this code in the
+ *  public domain for the benefit of all mankind - even the slimy
+ *  ones who might try to proprietize my work and use it to my
+ *  detriment.
+ *
+ * $Id: g711.h,v 1.1 2006/06/07 15:46:39 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Changed to use WebRtc types
+ * -Changed __inline__ to __inline
+ * -Two changes to make implementation bitexact with ITU-T reference
+ * implementation
+ */
+
+/*! \page g711_page A-law and mu-law handling
+Lookup tables for A-law and u-law look attractive, until you consider the impact
+on the CPU cache. If it causes a substantial area of your processor cache to get
+hit too often, cache sloshing will severely slow things down. The main reason
+these routines are slow in C, is the lack of direct access to the CPU's "find
+the first 1" instruction. A little in-line assembler fixes that, and the
+conversion routines can be faster than lookup tables, in most real world usage.
+A "find the first 1" instruction is available on most modern CPUs, and is a
+much underused feature.
+
+If an assembly language method of bit searching is not available, these routines
+revert to a method that can be a little slow, so the cache thrashing might not
+seem so bad :(
+
+Feel free to submit patches to add fast "find the first 1" support for your own
+favourite processor.
+
+Look up tables are used for transcoding between A-law and u-law, since it is
+difficult to achieve the precise transcoding procedure laid down in the G.711
+specification by other means.
+*/
+
+#ifndef MODULES_THIRD_PARTY_G711_G711_H_
+#define MODULES_THIRD_PARTY_G711_G711_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include <stdint.h>
+
+#if defined(__i386__)
+/*! \brief Find the bit position of the highest set bit in a word
+    \param bits The word to be searched
+    \return The bit number of the highest set bit, or -1 if the word is zero. */
+static __inline__ int top_bit(unsigned int bits) {
+  int res;
+
+  __asm__ __volatile__(
+      " movl $-1,%%edx;\n"
+      " bsrl %%eax,%%edx;\n"
+      : "=d"(res)
+      : "a"(bits));
+  return res;
+}
+
+/*! \brief Find the bit position of the lowest set bit in a word
+    \param bits The word to be searched
+    \return The bit number of the lowest set bit, or -1 if the word is zero. */
+static __inline__ int bottom_bit(unsigned int bits) {
+  int res;
+
+  __asm__ __volatile__(
+      " movl $-1,%%edx;\n"
+      " bsfl %%eax,%%edx;\n"
+      : "=d"(res)
+      : "a"(bits));
+  return res;
+}
+#elif defined(__x86_64__)
+static __inline__ int top_bit(unsigned int bits) {
+  int res;
+
+  __asm__ __volatile__(
+      " movq $-1,%%rdx;\n"
+      " bsrq %%rax,%%rdx;\n"
+      : "=d"(res)
+      : "a"(bits));
+  return res;
+}
+
+static __inline__ int bottom_bit(unsigned int bits) {
+  int res;
+
+  __asm__ __volatile__(
+      " movq $-1,%%rdx;\n"
+      " bsfq %%rax,%%rdx;\n"
+      : "=d"(res)
+      : "a"(bits));
+  return res;
+}
+#else
+static __inline int top_bit(unsigned int bits) {
+  int i;
+
+  if (bits == 0) {
+    return -1;
+  }
+  i = 0;
+  if (bits & 0xFFFF0000) {
+    bits &= 0xFFFF0000;
+    i += 16;
+  }
+  if (bits & 0xFF00FF00) {
+    bits &= 0xFF00FF00;
+    i += 8;
+  }
+  if (bits & 0xF0F0F0F0) {
+    bits &= 0xF0F0F0F0;
+    i += 4;
+  }
+  if (bits & 0xCCCCCCCC) {
+    bits &= 0xCCCCCCCC;
+    i += 2;
+  }
+  if (bits & 0xAAAAAAAA) {
+    bits &= 0xAAAAAAAA;
+    i += 1;
+  }
+  return i;
+}
+
+static __inline int bottom_bit(unsigned int bits) {
+  int i;
+
+  if (bits == 0) {
+    return -1;
+  }
+  i = 32;
+  if (bits & 0x0000FFFF) {
+    bits &= 0x0000FFFF;
+    i -= 16;
+  }
+  if (bits & 0x00FF00FF) {
+    bits &= 0x00FF00FF;
+    i -= 8;
+  }
+  if (bits & 0x0F0F0F0F) {
+    bits &= 0x0F0F0F0F;
+    i -= 4;
+  }
+  if (bits & 0x33333333) {
+    bits &= 0x33333333;
+    i -= 2;
+  }
+  if (bits & 0x55555555) {
+    bits &= 0x55555555;
+    i -= 1;
+  }
+  return i;
+}
+#endif
+
+/* N.B. It is tempting to use look-up tables for A-law and u-law conversion.
+ *      However, you should consider the cache footprint.
+ *
+ *      A 64K byte table for linear to x-law and a 512 byte table for x-law to
+ *      linear sound like peanuts these days, and shouldn't an array lookup be
+ *      real fast? No! When the cache sloshes as badly as this one will, a tight
+ *      calculation may be better. The messiest part is normally finding the
+ *      segment, but a little inline assembly can fix that on an i386, x86_64
+ * and many other modern processors.
+ */
+
+/*
+ * Mu-law is basically as follows:
+ *
+ *      Biased Linear Input Code        Compressed Code
+ *      ------------------------        ---------------
+ *      00000001wxyza                   000wxyz
+ *      0000001wxyzab                   001wxyz
+ *      000001wxyzabc                   010wxyz
+ *      00001wxyzabcd                   011wxyz
+ *      0001wxyzabcde                   100wxyz
+ *      001wxyzabcdef                   101wxyz
+ *      01wxyzabcdefg                   110wxyz
+ *      1wxyzabcdefgh                   111wxyz
+ *
+ * Each biased linear code has a leading 1 which identifies the segment
+ * number. The value of the segment number is equal to 7 minus the number
+ * of leading 0's. The quantization interval is directly available as the
+ * four bits wxyz.  * The trailing bits (a - h) are ignored.
+ *
+ * Ordinarily the complement of the resulting code word is used for
+ * transmission, and so the code word is complemented before it is returned.
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+
+//#define ULAW_ZEROTRAP                 /* turn on the trap as per the MIL-STD
+//*/
+#define ULAW_BIAS 0x84 /* Bias for linear code. */
+
+/*! \brief Encode a linear sample to u-law
+    \param linear The sample to encode.
+    \return The u-law value.
+*/
+static __inline uint8_t linear_to_ulaw(int linear) {
+  uint8_t u_val;
+  int mask;
+  int seg;
+
+  /* Get the sign and the magnitude of the value. */
+  if (linear < 0) {
+    /* WebRtc, tlegrand: -1 added to get bitexact to reference implementation */
+    linear = ULAW_BIAS - linear - 1;
+    mask = 0x7F;
+  } else {
+    linear = ULAW_BIAS + linear;
+    mask = 0xFF;
+  }
+
+  seg = top_bit(linear | 0xFF) - 7;
+
+  /*
+   * Combine the sign, segment, quantization bits,
+   * and complement the code word.
+   */
+  if (seg >= 8)
+    u_val = (uint8_t)(0x7F ^ mask);
+  else
+    u_val = (uint8_t)(((seg << 4) | ((linear >> (seg + 3)) & 0xF)) ^ mask);
+#ifdef ULAW_ZEROTRAP
+  /* Optional ITU trap */
+  if (u_val == 0)
+    u_val = 0x02;
+#endif
+  return u_val;
+}
+
+/*! \brief Decode an u-law sample to a linear value.
+    \param ulaw The u-law sample to decode.
+    \return The linear value.
+*/
+static __inline int16_t ulaw_to_linear(uint8_t ulaw) {
+  int t;
+
+  /* Complement to obtain normal u-law value. */
+  ulaw = ~ulaw;
+  /*
+   * Extract and bias the quantization bits. Then
+   * shift up by the segment number and subtract out the bias.
+   */
+  t = (((ulaw & 0x0F) << 3) + ULAW_BIAS) << (((int)ulaw & 0x70) >> 4);
+  return (int16_t)((ulaw & 0x80) ? (ULAW_BIAS - t) : (t - ULAW_BIAS));
+}
+
+/*
+ * A-law is basically as follows:
+ *
+ *      Linear Input Code        Compressed Code
+ *      -----------------        ---------------
+ *      0000000wxyza             000wxyz
+ *      0000001wxyza             001wxyz
+ *      000001wxyzab             010wxyz
+ *      00001wxyzabc             011wxyz
+ *      0001wxyzabcd             100wxyz
+ *      001wxyzabcde             101wxyz
+ *      01wxyzabcdef             110wxyz
+ *      1wxyzabcdefg             111wxyz
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+
+#define ALAW_AMI_MASK 0x55
+
+/*! \brief Encode a linear sample to A-law
+    \param linear The sample to encode.
+    \return The A-law value.
+*/
+static __inline uint8_t linear_to_alaw(int linear) {
+  int mask;
+  int seg;
+
+  if (linear >= 0) {
+    /* Sign (bit 7) bit = 1 */
+    mask = ALAW_AMI_MASK | 0x80;
+  } else {
+    /* Sign (bit 7) bit = 0 */
+    mask = ALAW_AMI_MASK;
+    /* WebRtc, tlegrand: Changed from -8 to -1 to get bitexact to reference
+     * implementation */
+    linear = -linear - 1;
+  }
+
+  /* Convert the scaled magnitude to segment number. */
+  seg = top_bit(linear | 0xFF) - 7;
+  if (seg >= 8) {
+    if (linear >= 0) {
+      /* Out of range. Return maximum value. */
+      return (uint8_t)(0x7F ^ mask);
+    }
+    /* We must be just a tiny step below zero */
+    return (uint8_t)(0x00 ^ mask);
+  }
+  /* Combine the sign, segment, and quantization bits. */
+  return (uint8_t)(((seg << 4) | ((linear >> ((seg) ? (seg + 3) : 4)) & 0x0F)) ^
+                   mask);
+}
+
+/*! \brief Decode an A-law sample to a linear value.
+    \param alaw The A-law sample to decode.
+    \return The linear value.
+*/
+static __inline int16_t alaw_to_linear(uint8_t alaw) {
+  int i;
+  int seg;
+
+  alaw ^= ALAW_AMI_MASK;
+  i = ((alaw & 0x0F) << 4);
+  seg = (((int)alaw & 0x70) >> 4);
+  if (seg)
+    i = (i + 0x108) << (seg - 1);
+  else
+    i += 8;
+  return (int16_t)((alaw & 0x80) ? i : -i);
+}
+
+/*! \brief Transcode from A-law to u-law, using the procedure defined in G.711.
+    \param alaw The A-law sample to transcode.
+    \return The best matching u-law value.
+*/
+uint8_t alaw_to_ulaw(uint8_t alaw);
+
+/*! \brief Transcode from u-law to A-law, using the procedure defined in G.711.
+    \param alaw The u-law sample to transcode.
+    \return The best matching A-law value.
+*/
+uint8_t ulaw_to_alaw(uint8_t ulaw);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* MODULES_THIRD_PARTY_G711_G711_H_ */
diff --git a/third_party/libwebrtc/modules/third_party/g711/g711_3p_gn/moz.build b/third_party/libwebrtc/modules/third_party/g711/g711_3p_gn/moz.build
new file mode 100644
index 0000000000..3b2ee5ee3d
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g711/g711_3p_gn/moz.build
@@ -0,0 +1,217 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/third_party/g711/g711.c"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("g711_3p_gn")
diff --git a/third_party/libwebrtc/modules/third_party/g722/BUILD.gn b/third_party/libwebrtc/modules/third_party/g722/BUILD.gn
new file mode 100644
index 0000000000..316cdc7415
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g722/BUILD.gn
@@ -0,0 +1,18 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the ../../../LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("g722_3p") {
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "g722_decode.c",
+    "g722_enc_dec.h",
+    "g722_encode.c",
+  ]
+}
diff --git a/third_party/libwebrtc/modules/third_party/g722/LICENSE b/third_party/libwebrtc/modules/third_party/g722/LICENSE
new file mode 100644
index 0000000000..ea6308f0db
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g722/LICENSE
@@ -0,0 +1,20 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722_decode.c - The ITU G.722 codec, decode part.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ *  Despite my general liking of the GPL, I place my own contributions
+ *  to this code in the public domain for the benefit of all mankind -
+ *  even the slimy ones who might try to proprietize my work and use it
+ *  to my detriment.
+ *
+ * Based in part on a single channel G.722 codec which is:
+ *
+ * Copyright (c) CMU 1993
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ */
diff --git a/third_party/libwebrtc/modules/third_party/g722/README.chromium b/third_party/libwebrtc/modules/third_party/g722/README.chromium
new file mode 100644
index 0000000000..c427ed8cf2
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g722/README.chromium
@@ -0,0 +1,11 @@
+Name: The ITU G.722 codec, encode and decode part.
+Short Name: g722
+URL:
+Version: 0
+Date: 2018-06-25
+License: Custom license
+License File: LICENSE
+Security Critical: yes
+
+Description:
+The ITU G.722 codec, encode and decode part.
diff --git a/third_party/libwebrtc/modules/third_party/g722/g722_3p_gn/moz.build b/third_party/libwebrtc/modules/third_party/g722/g722_3p_gn/moz.build
new file mode 100644
index 0000000000..f40abbac14
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g722/g722_3p_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/third_party/g722/g722_encode.c"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/third_party/g722/g722_decode.c"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("g722_3p_gn")
diff --git a/third_party/libwebrtc/modules/third_party/g722/g722_decode.c b/third_party/libwebrtc/modules/third_party/g722/g722_decode.c
new file mode 100644
index 0000000000..012aeb5028
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g722/g722_decode.c
@@ -0,0 +1,399 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722_decode.c - The ITU G.722 codec, decode part.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ *  Despite my general liking of the GPL, I place my own contributions
+ *  to this code in the public domain for the benefit of all mankind -
+ *  even the slimy ones who might try to proprietize my work and use it
+ *  to my detriment.
+ *
+ * Based in part on a single channel G.722 codec which is:
+ *
+ * Copyright (c) CMU 1993
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722_decode.c,v 1.15 2006/07/07 16:37:49 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Removed usage of inttypes.h and tgmath.h
+ * -Changed to use WebRtc types
+ * -Changed __inline__ to __inline
+ * -Added saturation check on output
+ */
+
+/*! \file */
+
+
+#include <memory.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "modules/third_party/g722/g722_enc_dec.h"
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+static __inline int16_t saturate(int32_t amp)
+{
+    int16_t amp16;
+
+    /* Hopefully this is optimised for the common case - not clipping */
+    amp16 = (int16_t) amp;
+    if (amp == amp16)
+        return amp16;
+    if (amp > WEBRTC_INT16_MAX)
+        return  WEBRTC_INT16_MAX;
+    return  WEBRTC_INT16_MIN;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void block4(G722DecoderState *s, int band, int d);
+
+static void block4(G722DecoderState *s, int band, int d)
+{
+    int wd1;
+    int wd2;
+    int wd3;
+    int i;
+
+    /* Block 4, RECONS */
+    s->band[band].d[0] = d;
+    s->band[band].r[0] = saturate(s->band[band].s + d);
+
+    /* Block 4, PARREC */
+    s->band[band].p[0] = saturate(s->band[band].sz + d);
+
+    /* Block 4, UPPOL2 */
+    for (i = 0;  i < 3;  i++)
+        s->band[band].sg[i] = s->band[band].p[i] >> 15;
+    wd1 = saturate(s->band[band].a[1] * 4);
+
+    wd2 = (s->band[band].sg[0] == s->band[band].sg[1])  ?  -wd1  :  wd1;
+    if (wd2 > 32767)
+        wd2 = 32767;
+    wd3 = (s->band[band].sg[0] == s->band[band].sg[2])  ?  128  :  -128;
+    wd3 += (wd2 >> 7);
+    wd3 += (s->band[band].a[2]*32512) >> 15;
+    if (wd3 > 12288)
+        wd3 = 12288;
+    else if (wd3 < -12288)
+        wd3 = -12288;
+    s->band[band].ap[2] = wd3;
+
+    /* Block 4, UPPOL1 */
+    s->band[band].sg[0] = s->band[band].p[0] >> 15;
+    s->band[band].sg[1] = s->band[band].p[1] >> 15;
+    wd1 = (s->band[band].sg[0] == s->band[band].sg[1])  ?  192  :  -192;
+    wd2 = (s->band[band].a[1]*32640) >> 15;
+
+    s->band[band].ap[1] = saturate(wd1 + wd2);
+    wd3 = saturate(15360 - s->band[band].ap[2]);
+    if (s->band[band].ap[1] > wd3)
+        s->band[band].ap[1] = wd3;
+    else if (s->band[band].ap[1] < -wd3)
+        s->band[band].ap[1] = -wd3;
+
+    /* Block 4, UPZERO */
+    wd1 = (d == 0)  ?  0  :  128;
+    s->band[band].sg[0] = d >> 15;
+    for (i = 1;  i < 7;  i++)
+    {
+        s->band[band].sg[i] = s->band[band].d[i] >> 15;
+        wd2 = (s->band[band].sg[i] == s->band[band].sg[0])  ?  wd1  :  -wd1;
+        wd3 = (s->band[band].b[i]*32640) >> 15;
+        s->band[band].bp[i] = saturate(wd2 + wd3);
+    }
+
+    /* Block 4, DELAYA */
+    for (i = 6;  i > 0;  i--)
+    {
+        s->band[band].d[i] = s->band[band].d[i - 1];
+        s->band[band].b[i] = s->band[band].bp[i];
+    }
+
+    for (i = 2;  i > 0;  i--)
+    {
+        s->band[band].r[i] = s->band[band].r[i - 1];
+        s->band[band].p[i] = s->band[band].p[i - 1];
+        s->band[band].a[i] = s->band[band].ap[i];
+    }
+
+    /* Block 4, FILTEP */
+    wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
+    wd1 = (s->band[band].a[1]*wd1) >> 15;
+    wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
+    wd2 = (s->band[band].a[2]*wd2) >> 15;
+    s->band[band].sp = saturate(wd1 + wd2);
+
+    /* Block 4, FILTEZ */
+    s->band[band].sz = 0;
+    for (i = 6;  i > 0;  i--)
+    {
+        wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
+        s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
+    }
+    s->band[band].sz = saturate(s->band[band].sz);
+
+    /* Block 4, PREDIC */
+    s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
+}
+/*- End of function --------------------------------------------------------*/
+
+G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
+                                          int rate,
+                                          int options) {
+    s = s ? s : malloc(sizeof(*s));
+    memset(s, 0, sizeof(*s));
+    if (rate == 48000)
+        s->bits_per_sample = 6;
+    else if (rate == 56000)
+        s->bits_per_sample = 7;
+    else
+        s->bits_per_sample = 8;
+    if ((options & G722_SAMPLE_RATE_8000))
+        s->eight_k = TRUE;
+    if ((options & G722_PACKED)  &&  s->bits_per_sample != 8)
+        s->packed = TRUE;
+    else
+        s->packed = FALSE;
+    s->band[0].det = 32;
+    s->band[1].det = 8;
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int WebRtc_g722_decode_release(G722DecoderState *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+size_t WebRtc_g722_decode(G722DecoderState *s, int16_t amp[],
+                          const uint8_t g722_data[], size_t len)
+{
+    static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
+    static const int rl42[16] = {0, 7, 6, 5, 4, 3, 2, 1,
+                                 7, 6, 5, 4, 3,  2, 1, 0 };
+    static const int ilb[32] =
+    {
+        2048, 2093, 2139, 2186, 2233, 2282, 2332,
+        2383, 2435, 2489, 2543, 2599, 2656, 2714,
+        2774, 2834, 2896, 2960, 3025, 3091, 3158,
+        3228, 3298, 3371, 3444, 3520, 3597, 3676,
+        3756, 3838, 3922, 4008
+    };
+    static const int wh[3] = {0, -214, 798};
+    static const int rh2[4] = {2, 1, 2, 1};
+    static const int qm2[4] = {-7408, -1616,  7408,   1616};
+    static const int qm4[16] =
+    {
+              0, -20456, -12896,  -8968,
+          -6288,  -4240,  -2584,  -1200,
+          20456,  12896,   8968,   6288,
+           4240,   2584,   1200,      0
+    };
+    static const int qm5[32] =
+    {
+           -280,   -280, -23352, -17560,
+         -14120, -11664,  -9752,  -8184,
+          -6864,  -5712,  -4696,  -3784,
+          -2960,  -2208,  -1520,   -880,
+          23352,  17560,  14120,  11664,
+           9752,   8184,   6864,   5712,
+           4696,   3784,   2960,   2208,
+           1520,    880,    280,   -280
+    };
+    static const int qm6[64] =
+    {
+           -136,   -136,   -136,   -136,
+         -24808, -21904, -19008, -16704,
+         -14984, -13512, -12280, -11192,
+         -10232,  -9360,  -8576,  -7856,
+          -7192,  -6576,  -6000,  -5456,
+          -4944,  -4464,  -4008,  -3576,
+          -3168,  -2776,  -2400,  -2032,
+          -1688,  -1360,  -1040,   -728,
+          24808,  21904,  19008,  16704,
+          14984,  13512,  12280,  11192,
+          10232,   9360,   8576,   7856,
+           7192,   6576,   6000,   5456,
+           4944,   4464,   4008,   3576,
+           3168,   2776,   2400,   2032,
+           1688,   1360,   1040,    728,
+            432,    136,   -432,   -136
+    };
+    static const int qmf_coeffs[12] =
+    {
+           3,  -11,   12,   32, -210,  951, 3876, -805,  362, -156,   53,  -11,
+    };
+
+    int dlowt;
+    int rlow;
+    int ihigh;
+    int dhigh;
+    int rhigh;
+    int xout1;
+    int xout2;
+    int wd1;
+    int wd2;
+    int wd3;
+    int code;
+    size_t outlen;
+    int i;
+    size_t j;
+
+    outlen = 0;
+    rhigh = 0;
+    for (j = 0;  j < len;  )
+    {
+        if (s->packed)
+        {
+            /* Unpack the code bits */
+            if (s->in_bits < s->bits_per_sample)
+            {
+                s->in_buffer |= (g722_data[j++] << s->in_bits);
+                s->in_bits += 8;
+            }
+            code = s->in_buffer & ((1 << s->bits_per_sample) - 1);
+            s->in_buffer >>= s->bits_per_sample;
+            s->in_bits -= s->bits_per_sample;
+        }
+        else
+        {
+            code = g722_data[j++];
+        }
+
+        switch (s->bits_per_sample)
+        {
+        default:
+        case 8:
+            wd1 = code & 0x3F;
+            ihigh = (code >> 6) & 0x03;
+            wd2 = qm6[wd1];
+            wd1 >>= 2;
+            break;
+        case 7:
+            wd1 = code & 0x1F;
+            ihigh = (code >> 5) & 0x03;
+            wd2 = qm5[wd1];
+            wd1 >>= 1;
+            break;
+        case 6:
+            wd1 = code & 0x0F;
+            ihigh = (code >> 4) & 0x03;
+            wd2 = qm4[wd1];
+            break;
+        }
+        /* Block 5L, LOW BAND INVQBL */
+        wd2 = (s->band[0].det*wd2) >> 15;
+        /* Block 5L, RECONS */
+        rlow = s->band[0].s + wd2;
+        /* Block 6L, LIMIT */
+        if (rlow > 16383)
+            rlow = 16383;
+        else if (rlow < -16384)
+            rlow = -16384;
+
+        /* Block 2L, INVQAL */
+        wd2 = qm4[wd1];
+        dlowt = (s->band[0].det*wd2) >> 15;
+
+        /* Block 3L, LOGSCL */
+        wd2 = rl42[wd1];
+        wd1 = (s->band[0].nb*127) >> 7;
+        wd1 += wl[wd2];
+        if (wd1 < 0)
+            wd1 = 0;
+        else if (wd1 > 18432)
+            wd1 = 18432;
+        s->band[0].nb = wd1;
+
+        /* Block 3L, SCALEL */
+        wd1 = (s->band[0].nb >> 6) & 31;
+        wd2 = 8 - (s->band[0].nb >> 11);
+        wd3 = (wd2 < 0)  ?  (ilb[wd1] << -wd2)  :  (ilb[wd1] >> wd2);
+        s->band[0].det = wd3 << 2;
+
+        block4(s, 0, dlowt);
+
+        if (!s->eight_k)
+        {
+            /* Block 2H, INVQAH */
+            wd2 = qm2[ihigh];
+            dhigh = (s->band[1].det*wd2) >> 15;
+            /* Block 5H, RECONS */
+            rhigh = dhigh + s->band[1].s;
+            /* Block 6H, LIMIT */
+            if (rhigh > 16383)
+                rhigh = 16383;
+            else if (rhigh < -16384)
+                rhigh = -16384;
+
+            /* Block 2H, INVQAH */
+            wd2 = rh2[ihigh];
+            wd1 = (s->band[1].nb*127) >> 7;
+            wd1 += wh[wd2];
+            if (wd1 < 0)
+                wd1 = 0;
+            else if (wd1 > 22528)
+                wd1 = 22528;
+            s->band[1].nb = wd1;
+
+            /* Block 3H, SCALEH */
+            wd1 = (s->band[1].nb >> 6) & 31;
+            wd2 = 10 - (s->band[1].nb >> 11);
+            wd3 = (wd2 < 0)  ?  (ilb[wd1] << -wd2)  :  (ilb[wd1] >> wd2);
+            s->band[1].det = wd3 << 2;
+
+            block4(s, 1, dhigh);
+        }
+
+        if (s->itu_test_mode)
+        {
+            amp[outlen++] = (int16_t) (rlow << 1);
+            amp[outlen++] = (int16_t) (rhigh << 1);
+        }
+        else
+        {
+            if (s->eight_k)
+            {
+                amp[outlen++] = (int16_t) (rlow << 1);
+            }
+            else
+            {
+                /* Apply the receive QMF */
+                for (i = 0;  i < 22;  i++)
+                    s->x[i] = s->x[i + 2];
+                s->x[22] = rlow + rhigh;
+                s->x[23] = rlow - rhigh;
+
+                xout1 = 0;
+                xout2 = 0;
+                for (i = 0;  i < 12;  i++)
+                {
+                    xout2 += s->x[2*i]*qmf_coeffs[i];
+                    xout1 += s->x[2*i + 1]*qmf_coeffs[11 - i];
+                }
+                /* We shift by 12 to allow for the QMF filters (DC gain = 4096), less 1
+                   to allow for the 15 bit input to the G.722 algorithm. */
+                /* WebRtc, tlegrand: added saturation */
+                amp[outlen++] = saturate(xout1 >> 11);
+                amp[outlen++] = saturate(xout2 >> 11);
+            }
+        }
+    }
+    return outlen;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/
diff --git a/third_party/libwebrtc/modules/third_party/g722/g722_enc_dec.h b/third_party/libwebrtc/modules/third_party/g722/g722_enc_dec.h
new file mode 100644
index 0000000000..47b0936d82
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g722/g722_enc_dec.h
@@ -0,0 +1,154 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722.h - The ITU G.722 codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ *  Despite my general liking of the GPL, I place my own contributions
+ *  to this code in the public domain for the benefit of all mankind -
+ *  even the slimy ones who might try to proprietize my work and use it
+ *  to my detriment.
+ *
+ * Based on a single channel G.722 codec which is:
+ *
+ *****    Copyright (c) CMU    1993      *****
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722.h,v 1.10 2006/06/16 12:45:53 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Changed to use WebRtc types
+ * -Added new defines for minimum and maximum values of short int
+ */
+
+/*! \file */
+
+#ifndef MODULES_THIRD_PARTY_G722_G722_H_
+#define MODULES_THIRD_PARTY_G722_G722_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*! \page g722_page G.722 encoding and decoding
+\section g722_page_sec_1 What does it do?
+The G.722 module is a bit exact implementation of the ITU G.722 specification
+for all three specified bit rates - 64000bps, 56000bps and 48000bps. It passes
+the ITU tests.
+
+To allow fast and flexible interworking with narrow band telephony, the encoder
+and decoder support an option for the linear audio to be an 8k samples/second
+stream. In this mode the codec is considerably faster, and still fully
+compatible with wideband terminals using G.722.
+
+\section g722_page_sec_2 How does it work?
+???.
+*/
+
+#define WEBRTC_INT16_MAX 32767
+#define WEBRTC_INT16_MIN -32768
+
+enum { G722_SAMPLE_RATE_8000 = 0x0001, G722_PACKED = 0x0002 };
+
+typedef struct {
+  /*! TRUE if the operating in the special ITU test mode, with the band split
+     filters disabled. */
+  int itu_test_mode;
+  /*! TRUE if the G.722 data is packed */
+  int packed;
+  /*! TRUE if encode from 8k samples/second */
+  int eight_k;
+  /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
+  int bits_per_sample;
+
+  /*! Signal history for the QMF */
+  int x[24];
+
+  struct {
+    int s;
+    int sp;
+    int sz;
+    int r[3];
+    int a[3];
+    int ap[3];
+    int p[3];
+    int d[7];
+    int b[7];
+    int bp[7];
+    int sg[7];
+    int nb;
+    int det;
+  } band[2];
+
+  unsigned int in_buffer;
+  int in_bits;
+  unsigned int out_buffer;
+  int out_bits;
+} G722EncoderState;
+
+typedef struct {
+  /*! TRUE if the operating in the special ITU test mode, with the band split
+     filters disabled. */
+  int itu_test_mode;
+  /*! TRUE if the G.722 data is packed */
+  int packed;
+  /*! TRUE if decode to 8k samples/second */
+  int eight_k;
+  /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
+  int bits_per_sample;
+
+  /*! Signal history for the QMF */
+  int x[24];
+
+  struct {
+    int s;
+    int sp;
+    int sz;
+    int r[3];
+    int a[3];
+    int ap[3];
+    int p[3];
+    int d[7];
+    int b[7];
+    int bp[7];
+    int sg[7];
+    int nb;
+    int det;
+  } band[2];
+
+  unsigned int in_buffer;
+  int in_bits;
+  unsigned int out_buffer;
+  int out_bits;
+} G722DecoderState;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
+                                          int rate,
+                                          int options);
+int WebRtc_g722_encode_release(G722EncoderState* s);
+size_t WebRtc_g722_encode(G722EncoderState* s,
+                          uint8_t g722_data[],
+                          const int16_t amp[],
+                          size_t len);
+
+G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
+                                          int rate,
+                                          int options);
+int WebRtc_g722_decode_release(G722DecoderState* s);
+size_t WebRtc_g722_decode(G722DecoderState* s,
+                          int16_t amp[],
+                          const uint8_t g722_data[],
+                          size_t len);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* MODULES_THIRD_PARTY_G722_G722_H_ */
diff --git a/third_party/libwebrtc/modules/third_party/g722/g722_encode.c b/third_party/libwebrtc/modules/third_party/g722/g722_encode.c
new file mode 100644
index 0000000000..fedf9f5961
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/g722/g722_encode.c
@@ -0,0 +1,429 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722_encode.c - The ITU G.722 codec, encode part.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ *  Despite my general liking of the GPL, I place my own contributions
+ *  to this code in the public domain for the benefit of all mankind -
+ *  even the slimy ones who might try to proprietize my work and use it
+ *  to my detriment.
+ *
+ * Based on a single channel 64kbps only G.722 codec which is:
+ *
+ *****    Copyright (c) CMU    1993      *****
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722_encode.c,v 1.14 2006/07/07 16:37:49 steveu Exp $
+ *
+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
+ * -Removed usage of inttypes.h and tgmath.h
+ * -Changed to use WebRtc types
+ * -Added option to run encoder bitexact with ITU-T reference implementation
+ */
+
+/*! \file */
+
+#include <memory.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "modules/third_party/g722/g722_enc_dec.h"
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+static __inline int16_t saturate(int32_t amp)
+{
+    int16_t amp16;
+
+    /* Hopefully this is optimised for the common case - not clipping */
+    amp16 = (int16_t) amp;
+    if (amp == amp16)
+        return amp16;
+    if (amp > WEBRTC_INT16_MAX)
+        return  WEBRTC_INT16_MAX;
+    return  WEBRTC_INT16_MIN;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void block4(G722EncoderState *s, int band, int d)
+{
+    int wd1;
+    int wd2;
+    int wd3;
+    int i;
+
+    /* Block 4, RECONS */
+    s->band[band].d[0] = d;
+    s->band[band].r[0] = saturate(s->band[band].s + d);
+
+    /* Block 4, PARREC */
+    s->band[band].p[0] = saturate(s->band[band].sz + d);
+
+    /* Block 4, UPPOL2 */
+    for (i = 0;  i < 3;  i++)
+        s->band[band].sg[i] = s->band[band].p[i] >> 15;
+    wd1 = saturate(s->band[band].a[1] * 4);
+
+    wd2 = (s->band[band].sg[0] == s->band[band].sg[1])  ?  -wd1  :  wd1;
+    if (wd2 > 32767)
+        wd2 = 32767;
+    wd3 = (wd2 >> 7) + ((s->band[band].sg[0] == s->band[band].sg[2])  ?  128  :  -128);
+    wd3 += (s->band[band].a[2]*32512) >> 15;
+    if (wd3 > 12288)
+        wd3 = 12288;
+    else if (wd3 < -12288)
+        wd3 = -12288;
+    s->band[band].ap[2] = wd3;
+
+    /* Block 4, UPPOL1 */
+    s->band[band].sg[0] = s->band[band].p[0] >> 15;
+    s->band[band].sg[1] = s->band[band].p[1] >> 15;
+    wd1 = (s->band[band].sg[0] == s->band[band].sg[1])  ?  192  :  -192;
+    wd2 = (s->band[band].a[1]*32640) >> 15;
+
+    s->band[band].ap[1] = saturate(wd1 + wd2);
+    wd3 = saturate(15360 - s->band[band].ap[2]);
+    if (s->band[band].ap[1] > wd3)
+        s->band[band].ap[1] = wd3;
+    else if (s->band[band].ap[1] < -wd3)
+        s->band[band].ap[1] = -wd3;
+
+    /* Block 4, UPZERO */
+    wd1 = (d == 0)  ?  0  :  128;
+    s->band[band].sg[0] = d >> 15;
+    for (i = 1;  i < 7;  i++)
+    {
+        s->band[band].sg[i] = s->band[band].d[i] >> 15;
+        wd2 = (s->band[band].sg[i] == s->band[band].sg[0])  ?  wd1  :  -wd1;
+        wd3 = (s->band[band].b[i]*32640) >> 15;
+        s->band[band].bp[i] = saturate(wd2 + wd3);
+    }
+
+    /* Block 4, DELAYA */
+    for (i = 6;  i > 0;  i--)
+    {
+        s->band[band].d[i] = s->band[band].d[i - 1];
+        s->band[band].b[i] = s->band[band].bp[i];
+    }
+
+    for (i = 2;  i > 0;  i--)
+    {
+        s->band[band].r[i] = s->band[band].r[i - 1];
+        s->band[band].p[i] = s->band[band].p[i - 1];
+        s->band[band].a[i] = s->band[band].ap[i];
+    }
+
+    /* Block 4, FILTEP */
+    wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
+    wd1 = (s->band[band].a[1]*wd1) >> 15;
+    wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
+    wd2 = (s->band[band].a[2]*wd2) >> 15;
+    s->band[band].sp = saturate(wd1 + wd2);
+
+    /* Block 4, FILTEZ */
+    s->band[band].sz = 0;
+    for (i = 6;  i > 0;  i--)
+    {
+        wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
+        s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
+    }
+    s->band[band].sz = saturate(s->band[band].sz);
+
+    /* Block 4, PREDIC */
+    s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
+}
+/*- End of function --------------------------------------------------------*/
+
+G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
+                                          int rate,
+                                          int options) {
+    if (s == NULL)
+    {
+        if ((s = (G722EncoderState *) malloc(sizeof(*s))) == NULL)
+            return NULL;
+    }
+    memset(s, 0, sizeof(*s));
+    if (rate == 48000)
+        s->bits_per_sample = 6;
+    else if (rate == 56000)
+        s->bits_per_sample = 7;
+    else
+        s->bits_per_sample = 8;
+    if ((options & G722_SAMPLE_RATE_8000))
+        s->eight_k = TRUE;
+    if ((options & G722_PACKED)  &&  s->bits_per_sample != 8)
+        s->packed = TRUE;
+    else
+        s->packed = FALSE;
+    s->band[0].det = 32;
+    s->band[1].det = 8;
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int WebRtc_g722_encode_release(G722EncoderState *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+/* WebRtc, tlegrand:
+ * Only define the following if bit-exactness with reference implementation
+ * is needed. Will only have any effect if input signal is saturated.
+ */
+//#define RUN_LIKE_REFERENCE_G722
+#ifdef RUN_LIKE_REFERENCE_G722
+int16_t limitValues (int16_t rl)
+{
+
+    int16_t yl;
+
+    yl = (rl > 16383) ? 16383 : ((rl < -16384) ? -16384 : rl);
+
+    return (yl);
+}
+#endif
+
+size_t WebRtc_g722_encode(G722EncoderState *s, uint8_t g722_data[],
+                          const int16_t amp[], size_t len)
+{
+    static const int q6[32] =
+    {
+           0,   35,   72,  110,  150,  190,  233,  276,
+         323,  370,  422,  473,  530,  587,  650,  714,
+         786,  858,  940, 1023, 1121, 1219, 1339, 1458,
+        1612, 1765, 1980, 2195, 2557, 2919,    0,    0
+    };
+    static const int iln[32] =
+    {
+         0, 63, 62, 31, 30, 29, 28, 27,
+        26, 25, 24, 23, 22, 21, 20, 19,
+        18, 17, 16, 15, 14, 13, 12, 11,
+        10,  9,  8,  7,  6,  5,  4,  0
+    };
+    static const int ilp[32] =
+    {
+         0, 61, 60, 59, 58, 57, 56, 55,
+        54, 53, 52, 51, 50, 49, 48, 47,
+        46, 45, 44, 43, 42, 41, 40, 39,
+        38, 37, 36, 35, 34, 33, 32,  0
+    };
+    static const int wl[8] =
+    {
+        -60, -30, 58, 172, 334, 538, 1198, 3042
+    };
+    static const int rl42[16] =
+    {
+        0, 7, 6, 5, 4, 3, 2, 1, 7, 6, 5, 4, 3, 2, 1, 0
+    };
+    static const int ilb[32] =
+    {
+        2048, 2093, 2139, 2186, 2233, 2282, 2332,
+        2383, 2435, 2489, 2543, 2599, 2656, 2714,
+        2774, 2834, 2896, 2960, 3025, 3091, 3158,
+        3228, 3298, 3371, 3444, 3520, 3597, 3676,
+        3756, 3838, 3922, 4008
+    };
+    static const int qm4[16] =
+    {
+             0, -20456, -12896, -8968,
+         -6288,  -4240,  -2584, -1200,
+         20456,  12896,   8968,  6288,
+          4240,   2584,   1200,     0
+    };
+    static const int qm2[4] =
+    {
+        -7408,  -1616,   7408,   1616
+    };
+    static const int qmf_coeffs[12] =
+    {
+           3,  -11,   12,   32, -210,  951, 3876, -805,  362, -156,   53,  -11,
+    };
+    static const int ihn[3] = {0, 1, 0};
+    static const int ihp[3] = {0, 3, 2};
+    static const int wh[3] = {0, -214, 798};
+    static const int rh2[4] = {2, 1, 2, 1};
+
+    int dlow;
+    int dhigh;
+    int el;
+    int wd;
+    int wd1;
+    int ril;
+    int wd2;
+    int il4;
+    int ih2;
+    int wd3;
+    int eh;
+    int mih;
+    int i;
+    size_t j;
+    /* Low and high band PCM from the QMF */
+    int xlow;
+    int xhigh;
+    size_t g722_bytes;
+    /* Even and odd tap accumulators */
+    int sumeven;
+    int sumodd;
+    int ihigh;
+    int ilow;
+    int code;
+
+    g722_bytes = 0;
+    xhigh = 0;
+    for (j = 0;  j < len;  )
+    {
+        if (s->itu_test_mode)
+        {
+            xlow =
+            xhigh = amp[j++] >> 1;
+        }
+        else
+        {
+            if (s->eight_k)
+            {
+                /* We shift by 1 to allow for the 15 bit input to the G.722 algorithm. */
+                xlow = amp[j++] >> 1;
+            }
+            else
+            {
+                /* Apply the transmit QMF */
+                /* Shuffle the buffer down */
+                for (i = 0;  i < 22;  i++)
+                    s->x[i] = s->x[i + 2];
+                s->x[22] = amp[j++];
+                s->x[23] = amp[j++];
+
+                /* Discard every other QMF output */
+                sumeven = 0;
+                sumodd = 0;
+                for (i = 0;  i < 12;  i++)
+                {
+                    sumodd += s->x[2*i]*qmf_coeffs[i];
+                    sumeven += s->x[2*i + 1]*qmf_coeffs[11 - i];
+                }
+                /* We shift by 12 to allow for the QMF filters (DC gain = 4096), plus 1
+                   to allow for us summing two filters, plus 1 to allow for the 15 bit
+                   input to the G.722 algorithm. */
+                xlow = (sumeven + sumodd) >> 14;
+                xhigh = (sumeven - sumodd) >> 14;
+
+#ifdef RUN_LIKE_REFERENCE_G722
+                /* The following lines are only used to verify bit-exactness
+                 * with reference implementation of G.722. Higher precision
+                 * is achieved without limiting the values.
+                 */
+                xlow = limitValues(xlow);
+                xhigh = limitValues(xhigh);
+#endif
+            }
+        }
+        /* Block 1L, SUBTRA */
+        el = saturate(xlow - s->band[0].s);
+
+        /* Block 1L, QUANTL */
+        wd = (el >= 0)  ?  el  :  -(el + 1);
+
+        for (i = 1;  i < 30;  i++)
+        {
+            wd1 = (q6[i]*s->band[0].det) >> 12;
+            if (wd < wd1)
+                break;
+        }
+        ilow = (el < 0)  ?  iln[i]  :  ilp[i];
+
+        /* Block 2L, INVQAL */
+        ril = ilow >> 2;
+        wd2 = qm4[ril];
+        dlow = (s->band[0].det*wd2) >> 15;
+
+        /* Block 3L, LOGSCL */
+        il4 = rl42[ril];
+        wd = (s->band[0].nb*127) >> 7;
+        s->band[0].nb = wd + wl[il4];
+        if (s->band[0].nb < 0)
+            s->band[0].nb = 0;
+        else if (s->band[0].nb > 18432)
+            s->band[0].nb = 18432;
+
+        /* Block 3L, SCALEL */
+        wd1 = (s->band[0].nb >> 6) & 31;
+        wd2 = 8 - (s->band[0].nb >> 11);
+        wd3 = (wd2 < 0)  ?  (ilb[wd1] << -wd2)  :  (ilb[wd1] >> wd2);
+        s->band[0].det = wd3 << 2;
+
+        block4(s, 0, dlow);
+
+        if (s->eight_k)
+        {
+            /* Just leave the high bits as zero */
+            code = (0xC0 | ilow) >> (8 - s->bits_per_sample);
+        }
+        else
+        {
+            /* Block 1H, SUBTRA */
+            eh = saturate(xhigh - s->band[1].s);
+
+            /* Block 1H, QUANTH */
+            wd = (eh >= 0)  ?  eh  :  -(eh + 1);
+            wd1 = (564*s->band[1].det) >> 12;
+            mih = (wd >= wd1)  ?  2  :  1;
+            ihigh = (eh < 0)  ?  ihn[mih]  :  ihp[mih];
+
+            /* Block 2H, INVQAH */
+            wd2 = qm2[ihigh];
+            dhigh = (s->band[1].det*wd2) >> 15;
+
+            /* Block 3H, LOGSCH */
+            ih2 = rh2[ihigh];
+            wd = (s->band[1].nb*127) >> 7;
+            s->band[1].nb = wd + wh[ih2];
+            if (s->band[1].nb < 0)
+                s->band[1].nb = 0;
+            else if (s->band[1].nb > 22528)
+                s->band[1].nb = 22528;
+
+            /* Block 3H, SCALEH */
+            wd1 = (s->band[1].nb >> 6) & 31;
+            wd2 = 10 - (s->band[1].nb >> 11);
+            wd3 = (wd2 < 0)  ?  (ilb[wd1] << -wd2)  :  (ilb[wd1] >> wd2);
+            s->band[1].det = wd3 << 2;
+
+            block4(s, 1, dhigh);
+            code = ((ihigh << 6) | ilow) >> (8 - s->bits_per_sample);
+        }
+
+        if (s->packed)
+        {
+            /* Pack the code bits */
+            s->out_buffer |= (code << s->out_bits);
+            s->out_bits += s->bits_per_sample;
+            if (s->out_bits >= 8)
+            {
+                g722_data[g722_bytes++] = (uint8_t) (s->out_buffer & 0xFF);
+                s->out_bits -= 8;
+                s->out_buffer >>= 8;
+            }
+        }
+        else
+        {
+            g722_data[g722_bytes++] = (uint8_t) code;
+        }
+    }
+    return g722_bytes;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/
diff --git a/third_party/libwebrtc/modules/third_party/portaudio/BUILD.gn b/third_party/libwebrtc/modules/third_party/portaudio/BUILD.gn
new file mode 100644
index 0000000000..c49c544e9d
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/portaudio/BUILD.gn
@@ -0,0 +1,18 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the ../../../LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("mac_portaudio") {
+  visibility = [ "../../audio_device:*" ]
+  sources = [
+    "pa_memorybarrier.h",
+    "pa_ringbuffer.c",
+    "pa_ringbuffer.h",
+  ]
+}
diff --git a/third_party/libwebrtc/modules/third_party/portaudio/LICENSE b/third_party/libwebrtc/modules/third_party/portaudio/LICENSE
new file mode 100644
index 0000000000..6ccaca2910
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/portaudio/LICENSE
@@ -0,0 +1,91 @@
+/*
+ * $Id: pa_memorybarrier.h 1240 2007-07-17 13:05:07Z bjornroche $
+ * Portable Audio I/O Library
+ * Memory barrier utilities
+ *
+ * Author: Bjorn Roche, XO Audio, LLC
+ *
+ * This program uses the PortAudio Portable Audio Library.
+ * For more information see: http://www.portaudio.com
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however,
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
+ * license above.
+ */
+
+/*
+ * $Id: pa_ringbuffer.c 1421 2009-11-18 16:09:05Z bjornroche $
+ * Portable Audio I/O Library
+ * Ring Buffer utility.
+ *
+ * Author: Phil Burk, http://www.softsynth.com
+ * modified for SMP safety on Mac OS X by Bjorn Roche
+ * modified for SMP safety on Linux by Leland Lucius
+ * also, allowed for const where possible
+ * modified for multiple-byte-sized data elements by Sven Fischer
+ *
+ * Note that this is safe only for a single-thread reader and a
+ * single-thread writer.
+ *
+ * This program uses the PortAudio Portable Audio Library.
+ * For more information see: http://www.portaudio.com
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however,
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
+ * license above.
+ */
+
diff --git a/third_party/libwebrtc/modules/third_party/portaudio/README.chromium b/third_party/libwebrtc/modules/third_party/portaudio/README.chromium
new file mode 100644
index 0000000000..722dd94345
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/portaudio/README.chromium
@@ -0,0 +1,14 @@
+Name: Portaudio library for mac
+Short Name: portaudio
+URL: https://github.com/PortAudio/portaudio/tree/master/src/common
+Version: 9d8563100d841300f1689b186d131347ad43a0f6
+Date: 2022-04-12
+License: Custom license
+License File: LICENSE
+Security Critical: yes
+
+Description:
+Part of portaudio library to operate with memory barriers and ring buffer.
+
+Local changes:
+  - Minor formatting to make 'git cl format' happy.
diff --git a/third_party/libwebrtc/modules/third_party/portaudio/pa_memorybarrier.h b/third_party/libwebrtc/modules/third_party/portaudio/pa_memorybarrier.h
new file mode 100644
index 0000000000..f8c1852f4e
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/portaudio/pa_memorybarrier.h
@@ -0,0 +1,144 @@
+/*
+ * $Id: pa_memorybarrier.h 1240 2007-07-17 13:05:07Z bjornroche $
+ * Portable Audio I/O Library
+ * Memory barrier utilities
+ *
+ * Author: Bjorn Roche, XO Audio, LLC
+ *
+ * This program uses the PortAudio Portable Audio Library.
+ * For more information see: http://www.portaudio.com
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however,
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
+ * license above.
+ */
+
+/**
+ @file pa_memorybarrier.h
+ @ingroup common_src
+*/
+
+/****************
+ * Some memory barrier primitives based on the system.
+ * right now only OS X, FreeBSD, and Linux are supported. In addition to
+ *providing memory barriers, these functions should ensure that data cached in
+ *registers is written out to cache where it can be snooped by other CPUs. (ie,
+ *the volatile keyword should not be required)
+ *
+ * the primitives that must be defined are:
+ *
+ * PaUtil_FullMemoryBarrier()
+ * PaUtil_ReadMemoryBarrier()
+ * PaUtil_WriteMemoryBarrier()
+ *
+ ****************/
+
+#ifndef MODULES_THIRD_PARTY_PORTAUDIO_PA_MEMORYBARRIER_H_
+#define MODULES_THIRD_PARTY_PORTAUDIO_PA_MEMORYBARRIER_H_
+
+#if defined(__APPLE__)
+/* Support for the atomic library was added in C11.
+ */
+#if (__STDC_VERSION__ < 201112L) || defined(__STDC_NO_ATOMICS__)
+#include <libkern/OSAtomic.h>
+/* Here are the memory barrier functions. Mac OS X only provides
+   full memory barriers, so the three types of barriers are the same,
+   however, these barriers are superior to compiler-based ones.
+   These were deprecated in MacOS 10.12. */
+#define PaUtil_FullMemoryBarrier() OSMemoryBarrier()
+#define PaUtil_ReadMemoryBarrier() OSMemoryBarrier()
+#define PaUtil_WriteMemoryBarrier() OSMemoryBarrier()
+#else
+#include <stdatomic.h>
+#define PaUtil_FullMemoryBarrier() atomic_thread_fence(memory_order_seq_cst)
+#define PaUtil_ReadMemoryBarrier() atomic_thread_fence(memory_order_acquire)
+#define PaUtil_WriteMemoryBarrier() atomic_thread_fence(memory_order_release)
+#endif
+#elif defined(__GNUC__)
+/* GCC >= 4.1 has built-in intrinsics. We'll use those */
+#if (__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)
+#define PaUtil_FullMemoryBarrier() __sync_synchronize()
+#define PaUtil_ReadMemoryBarrier() __sync_synchronize()
+#define PaUtil_WriteMemoryBarrier() __sync_synchronize()
+/* as a fallback, GCC understands volatile asm and "memory" to mean it
+ * should not reorder memory read/writes */
+/* Note that it is not clear that any compiler actually defines __PPC__,
+ * it can probably removed safely. */
+#elif defined(__ppc__) || defined(__powerpc__) || defined(__PPC__)
+#define PaUtil_FullMemoryBarrier() asm volatile("sync" ::: "memory")
+#define PaUtil_ReadMemoryBarrier() asm volatile("sync" ::: "memory")
+#define PaUtil_WriteMemoryBarrier() asm volatile("sync" ::: "memory")
+#elif defined(__i386__) || defined(__i486__) || defined(__i586__) || \
+    defined(__i686__) || defined(__x86_64__)
+#define PaUtil_FullMemoryBarrier() asm volatile("mfence" ::: "memory")
+#define PaUtil_ReadMemoryBarrier() asm volatile("lfence" ::: "memory")
+#define PaUtil_WriteMemoryBarrier() asm volatile("sfence" ::: "memory")
+#else
+#ifdef ALLOW_SMP_DANGERS
+#warning Memory barriers not defined on this system or system unknown
+#warning For SMP safety, you should fix this.
+#define PaUtil_FullMemoryBarrier()
+#define PaUtil_ReadMemoryBarrier()
+#define PaUtil_WriteMemoryBarrier()
+#else
+#           error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
+#endif
+#endif
+#elif (_MSC_VER >= 1400) && !defined(_WIN32_WCE)
+#include <intrin.h>
+#pragma intrinsic(_ReadWriteBarrier)
+#pragma intrinsic(_ReadBarrier)
+#pragma intrinsic(_WriteBarrier)
+/* note that MSVC intrinsics _ReadWriteBarrier(), _ReadBarrier(),
+ * _WriteBarrier() are just compiler barriers *not* memory barriers */
+#define PaUtil_FullMemoryBarrier() _ReadWriteBarrier()
+#define PaUtil_ReadMemoryBarrier() _ReadBarrier()
+#define PaUtil_WriteMemoryBarrier() _WriteBarrier()
+#elif defined(_WIN32_WCE)
+#define PaUtil_FullMemoryBarrier()
+#define PaUtil_ReadMemoryBarrier()
+#define PaUtil_WriteMemoryBarrier()
+#elif defined(_MSC_VER) || defined(__BORLANDC__)
+#define PaUtil_FullMemoryBarrier() _asm { lock add    [esp], 0}
+#define PaUtil_ReadMemoryBarrier() _asm { lock add    [esp], 0}
+#define PaUtil_WriteMemoryBarrier() _asm { lock add    [esp], 0}
+#else
+#ifdef ALLOW_SMP_DANGERS
+#warning Memory barriers not defined on this system or system unknown
+#warning For SMP safety, you should fix this.
+#define PaUtil_FullMemoryBarrier()
+#define PaUtil_ReadMemoryBarrier()
+#define PaUtil_WriteMemoryBarrier()
+#else
+#       error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
+#endif
+#endif
+
+#endif /* MODULES_THIRD_PARTY_PORTAUDIO_PA_MEMORYBARRIER_H_ */
diff --git a/third_party/libwebrtc/modules/third_party/portaudio/pa_ringbuffer.c b/third_party/libwebrtc/modules/third_party/portaudio/pa_ringbuffer.c
new file mode 100644
index 0000000000..b978d54f19
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/portaudio/pa_ringbuffer.c
@@ -0,0 +1,237 @@
+/*
+ * $Id$
+ * Portable Audio I/O Library
+ * Ring Buffer utility.
+ *
+ * Author: Phil Burk, http://www.softsynth.com
+ * modified for SMP safety on Mac OS X by Bjorn Roche
+ * modified for SMP safety on Linux by Leland Lucius
+ * also, allowed for const where possible
+ * modified for multiple-byte-sized data elements by Sven Fischer
+ *
+ * Note that this is safe only for a single-thread reader and a
+ * single-thread writer.
+ *
+ * This program uses the PortAudio Portable Audio Library.
+ * For more information see: http://www.portaudio.com
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however,
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
+ * license above.
+ */
+
+/**
+ @file
+ @ingroup common_src
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <math.h>
+#include "pa_ringbuffer.h"
+#include <string.h>
+#include "pa_memorybarrier.h"
+
+/***************************************************************************
+ * Initialize FIFO.
+ * elementCount must be power of 2, returns -1 if not.
+ */
+ring_buffer_size_t PaUtil_InitializeRingBuffer( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementSizeBytes, ring_buffer_size_t elementCount, void *dataPtr )
+{
+    if( ((elementCount-1) & elementCount) != 0) return -1; /* Not Power of two. */
+    rbuf->bufferSize = elementCount;
+    rbuf->buffer = (char *)dataPtr;
+    PaUtil_FlushRingBuffer( rbuf );
+    rbuf->bigMask = (elementCount*2)-1;
+    rbuf->smallMask = (elementCount)-1;
+    rbuf->elementSizeBytes = elementSizeBytes;
+    return 0;
+}
+
+/***************************************************************************
+** Return number of elements available for reading. */
+ring_buffer_size_t PaUtil_GetRingBufferReadAvailable( const PaUtilRingBuffer *rbuf )
+{
+    return ( (rbuf->writeIndex - rbuf->readIndex) & rbuf->bigMask );
+}
+/***************************************************************************
+** Return number of elements available for writing. */
+ring_buffer_size_t PaUtil_GetRingBufferWriteAvailable( const PaUtilRingBuffer *rbuf )
+{
+    return ( rbuf->bufferSize - PaUtil_GetRingBufferReadAvailable(rbuf));
+}
+
+/***************************************************************************
+** Clear buffer. Should only be called when buffer is NOT being read or written. */
+void PaUtil_FlushRingBuffer( PaUtilRingBuffer *rbuf )
+{
+    rbuf->writeIndex = rbuf->readIndex = 0;
+}
+
+/***************************************************************************
+** Get address of region(s) to which we can write data.
+** If the region is contiguous, size2 will be zero.
+** If non-contiguous, size2 will be the size of second region.
+** Returns room available to be written or elementCount, whichever is smaller.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferWriteRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
+                                       void **dataPtr1, ring_buffer_size_t *sizePtr1,
+                                       void **dataPtr2, ring_buffer_size_t *sizePtr2 )
+{
+    ring_buffer_size_t   index;
+    ring_buffer_size_t   available = PaUtil_GetRingBufferWriteAvailable( rbuf );
+    if( elementCount > available ) elementCount = available;
+    /* Check to see if write is not contiguous. */
+    index = rbuf->writeIndex & rbuf->smallMask;
+    if( (index + elementCount) > rbuf->bufferSize )
+    {
+        /* Write data in two blocks that wrap the buffer. */
+        ring_buffer_size_t   firstHalf = rbuf->bufferSize - index;
+        *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
+        *sizePtr1 = firstHalf;
+        *dataPtr2 = &rbuf->buffer[0];
+        *sizePtr2 = elementCount - firstHalf;
+    }
+    else
+    {
+        *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
+        *sizePtr1 = elementCount;
+        *dataPtr2 = NULL;
+        *sizePtr2 = 0;
+    }
+
+    if( available )
+        PaUtil_FullMemoryBarrier(); /* (write-after-read) => full barrier */
+
+    return elementCount;
+}
+
+
+/***************************************************************************
+*/
+ring_buffer_size_t PaUtil_AdvanceRingBufferWriteIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount )
+{
+    /* ensure that previous writes are seen before we update the write index
+       (write after write)
+    */
+    PaUtil_WriteMemoryBarrier();
+    return rbuf->writeIndex = (rbuf->writeIndex + elementCount) & rbuf->bigMask;
+}
+
+/***************************************************************************
+** Get address of region(s) from which we can read data.
+** If the region is contiguous, size2 will be zero.
+** If non-contiguous, size2 will be the size of second region.
+** Returns room available to be read or elementCount, whichever is smaller.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferReadRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
+                                void **dataPtr1, ring_buffer_size_t *sizePtr1,
+                                void **dataPtr2, ring_buffer_size_t *sizePtr2 )
+{
+    ring_buffer_size_t   index;
+    ring_buffer_size_t   available = PaUtil_GetRingBufferReadAvailable( rbuf ); /* doesn't use memory barrier */
+    if( elementCount > available ) elementCount = available;
+    /* Check to see if read is not contiguous. */
+    index = rbuf->readIndex & rbuf->smallMask;
+    if( (index + elementCount) > rbuf->bufferSize )
+    {
+        /* Write data in two blocks that wrap the buffer. */
+        ring_buffer_size_t firstHalf = rbuf->bufferSize - index;
+        *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
+        *sizePtr1 = firstHalf;
+        *dataPtr2 = &rbuf->buffer[0];
+        *sizePtr2 = elementCount - firstHalf;
+    }
+    else
+    {
+        *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
+        *sizePtr1 = elementCount;
+        *dataPtr2 = NULL;
+        *sizePtr2 = 0;
+    }
+
+    if( available )
+        PaUtil_ReadMemoryBarrier(); /* (read-after-read) => read barrier */
+
+    return elementCount;
+}
+/***************************************************************************
+*/
+ring_buffer_size_t PaUtil_AdvanceRingBufferReadIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount )
+{
+    /* ensure that previous reads (copies out of the ring buffer) are always completed before updating (writing) the read index.
+       (write-after-read) => full barrier
+    */
+    PaUtil_FullMemoryBarrier();
+    return rbuf->readIndex = (rbuf->readIndex + elementCount) & rbuf->bigMask;
+}
+
+/***************************************************************************
+** Return elements written. */
+ring_buffer_size_t PaUtil_WriteRingBuffer( PaUtilRingBuffer *rbuf, const void *data, ring_buffer_size_t elementCount )
+{
+    ring_buffer_size_t size1, size2, numWritten;
+    void *data1, *data2;
+    numWritten = PaUtil_GetRingBufferWriteRegions( rbuf, elementCount, &data1, &size1, &data2, &size2 );
+    if( size2 > 0 )
+    {
+
+        memcpy( data1, data, size1*rbuf->elementSizeBytes );
+        data = ((char *)data) + size1*rbuf->elementSizeBytes;
+        memcpy( data2, data, size2*rbuf->elementSizeBytes );
+    }
+    else
+    {
+        memcpy( data1, data, size1*rbuf->elementSizeBytes );
+    }
+    PaUtil_AdvanceRingBufferWriteIndex( rbuf, numWritten );
+    return numWritten;
+}
+
+/***************************************************************************
+** Return elements read. */
+ring_buffer_size_t PaUtil_ReadRingBuffer( PaUtilRingBuffer *rbuf, void *data, ring_buffer_size_t elementCount )
+{
+    ring_buffer_size_t size1, size2, numRead;
+    void *data1, *data2;
+    numRead = PaUtil_GetRingBufferReadRegions( rbuf, elementCount, &data1, &size1, &data2, &size2 );
+    if( size2 > 0 )
+    {
+        memcpy( data, data1, size1*rbuf->elementSizeBytes );
+        data = ((char *)data) + size1*rbuf->elementSizeBytes;
+        memcpy( data, data2, size2*rbuf->elementSizeBytes );
+    }
+    else
+    {
+        memcpy( data, data1, size1*rbuf->elementSizeBytes );
+    }
+    PaUtil_AdvanceRingBufferReadIndex( rbuf, numRead );
+    return numRead;
+}
diff --git a/third_party/libwebrtc/modules/third_party/portaudio/pa_ringbuffer.h b/third_party/libwebrtc/modules/third_party/portaudio/pa_ringbuffer.h
new file mode 100644
index 0000000000..aa9d0aa772
--- /dev/null
+++ b/third_party/libwebrtc/modules/third_party/portaudio/pa_ringbuffer.h
@@ -0,0 +1,263 @@
+#ifndef MODULES_THIRD_PARTY_PORTAUDIO_PA_RINGBUFFER_H_
+#define MODULES_THIRD_PARTY_PORTAUDIO_PA_RINGBUFFER_H_
+/*
+ * $Id$
+ * Portable Audio I/O Library
+ * Ring Buffer utility.
+ *
+ * Author: Phil Burk, http://www.softsynth.com
+ * modified for SMP safety on OS X by Bjorn Roche.
+ * also allowed for const where possible.
+ * modified for multiple-byte-sized data elements by Sven Fischer
+ *
+ * Note that this is safe only for a single-thread reader
+ * and a single-thread writer.
+ *
+ * This program is distributed with the PortAudio Portable Audio Library.
+ * For more information see: http://www.portaudio.com
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however,
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also
+ * requested that these non-binding requests be included along with the
+ * license above.
+ */
+
+/** @file
+ @ingroup common_src
+ @brief Single-reader single-writer lock-free ring buffer
+
+ PaUtilRingBuffer is a ring buffer used to transport samples between
+ different execution contexts (threads, OS callbacks, interrupt handlers)
+ without requiring the use of any locks. This only works when there is
+ a single reader and a single writer (ie. one thread or callback writes
+ to the ring buffer, another thread or callback reads from it).
+
+ The PaUtilRingBuffer structure manages a ring buffer containing N
+ elements, where N must be a power of two. An element may be any size
+ (specified in bytes).
+
+ The memory area used to store the buffer elements must be allocated by
+ the client prior to calling PaUtil_InitializeRingBuffer() and must outlive
+ the use of the ring buffer.
+
+ @note The ring buffer functions are not normally exposed in the PortAudio
+ libraries. If you want to call them then you will need to add pa_ringbuffer.c
+ to your application source code.
+*/
+
+#if defined(__APPLE__)
+#include <sys/types.h>
+typedef int32_t ring_buffer_size_t;
+#elif defined(__GNUC__)
+typedef long ring_buffer_size_t;
+#elif (_MSC_VER >= 1400)
+typedef long ring_buffer_size_t;
+#elif defined(_MSC_VER) || defined(__BORLANDC__)
+typedef long ring_buffer_size_t;
+#else
+typedef long ring_buffer_size_t;
+#endif
+
+#ifdef __cplusplus
+extern "C" {
+#endif /* __cplusplus */
+
+typedef struct PaUtilRingBuffer {
+  ring_buffer_size_t bufferSize; /**< Number of elements in FIFO. Power of 2.
+                                    Set by PaUtil_InitRingBuffer. */
+  volatile ring_buffer_size_t
+      writeIndex; /**< Index of next writable element. Set by
+                     PaUtil_AdvanceRingBufferWriteIndex. */
+  volatile ring_buffer_size_t
+      readIndex;                /**< Index of next readable element. Set by
+                                   PaUtil_AdvanceRingBufferReadIndex. */
+  ring_buffer_size_t bigMask;   /**< Used for wrapping indices with extra bit to
+                                   distinguish full/empty. */
+  ring_buffer_size_t smallMask; /**< Used for fitting indices to buffer. */
+  ring_buffer_size_t elementSizeBytes; /**< Number of bytes per element. */
+  char* buffer; /**< Pointer to the buffer containing the actual data. */
+} PaUtilRingBuffer;
+
+/** Initialize Ring Buffer to empty state ready to have elements written to it.
+
+ @param rbuf The ring buffer.
+
+ @param elementSizeBytes The size of a single data element in bytes.
+
+ @param elementCount The number of elements in the buffer (must be a power of
+ 2).
+
+ @param dataPtr A pointer to a previously allocated area where the data
+ will be maintained.  It must be elementCount*elementSizeBytes long.
+
+ @return -1 if elementCount is not a power of 2, otherwise 0.
+*/
+ring_buffer_size_t PaUtil_InitializeRingBuffer(
+    PaUtilRingBuffer* rbuf,
+    ring_buffer_size_t elementSizeBytes,
+    ring_buffer_size_t elementCount,
+    void* dataPtr);
+
+/** Reset buffer to empty. Should only be called when buffer is NOT being read
+ or written.
+
+ @param rbuf The ring buffer.
+*/
+void PaUtil_FlushRingBuffer(PaUtilRingBuffer* rbuf);
+
+/** Retrieve the number of elements available in the ring buffer for writing.
+
+ @param rbuf The ring buffer.
+
+ @return The number of elements available for writing.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferWriteAvailable(
+    const PaUtilRingBuffer* rbuf);
+
+/** Retrieve the number of elements available in the ring buffer for reading.
+
+ @param rbuf The ring buffer.
+
+ @return The number of elements available for reading.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferReadAvailable(
+    const PaUtilRingBuffer* rbuf);
+
+/** Write data to the ring buffer.
+
+ @param rbuf The ring buffer.
+
+ @param data The address of new data to write to the buffer.
+
+ @param elementCount The number of elements to be written.
+
+ @return The number of elements written.
+*/
+ring_buffer_size_t PaUtil_WriteRingBuffer(PaUtilRingBuffer* rbuf,
+                                          const void* data,
+                                          ring_buffer_size_t elementCount);
+
+/** Read data from the ring buffer.
+
+ @param rbuf The ring buffer.
+
+ @param data The address where the data should be stored.
+
+ @param elementCount The number of elements to be read.
+
+ @return The number of elements read.
+*/
+ring_buffer_size_t PaUtil_ReadRingBuffer(PaUtilRingBuffer* rbuf,
+                                         void* data,
+                                         ring_buffer_size_t elementCount);
+
+/** Get address of region(s) to which we can write data.
+
+ @param rbuf The ring buffer.
+
+ @param elementCount The number of elements desired.
+
+ @param dataPtr1 The address where the first (or only) region pointer will be
+ stored.
+
+ @param sizePtr1 The address where the first (or only) region length will be
+ stored.
+
+ @param dataPtr2 The address where the second region pointer will be stored if
+ the first region is too small to satisfy elementCount.
+
+ @param sizePtr2 The address where the second region length will be stored if
+ the first region is too small to satisfy elementCount.
+
+ @return The room available to be written or elementCount, whichever is smaller.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferWriteRegions(
+    PaUtilRingBuffer* rbuf,
+    ring_buffer_size_t elementCount,
+    void** dataPtr1,
+    ring_buffer_size_t* sizePtr1,
+    void** dataPtr2,
+    ring_buffer_size_t* sizePtr2);
+
+/** Advance the write index to the next location to be written.
+
+ @param rbuf The ring buffer.
+
+ @param elementCount The number of elements to advance.
+
+ @return The new position.
+*/
+ring_buffer_size_t PaUtil_AdvanceRingBufferWriteIndex(
+    PaUtilRingBuffer* rbuf,
+    ring_buffer_size_t elementCount);
+
+/** Get address of region(s) from which we can read data.
+
+ @param rbuf The ring buffer.
+
+ @param elementCount The number of elements desired.
+
+ @param dataPtr1 The address where the first (or only) region pointer will be
+ stored.
+
+ @param sizePtr1 The address where the first (or only) region length will be
+ stored.
+
+ @param dataPtr2 The address where the second region pointer will be stored if
+ the first region is too small to satisfy elementCount.
+
+ @param sizePtr2 The address where the second region length will be stored if
+ the first region is too small to satisfy elementCount.
+
+ @return The number of elements available for reading.
+*/
+ring_buffer_size_t PaUtil_GetRingBufferReadRegions(
+    PaUtilRingBuffer* rbuf,
+    ring_buffer_size_t elementCount,
+    void** dataPtr1,
+    ring_buffer_size_t* sizePtr1,
+    void** dataPtr2,
+    ring_buffer_size_t* sizePtr2);
+
+/** Advance the read index to the next location to be read.
+
+ @param rbuf The ring buffer.
+
+ @param elementCount The number of elements to advance.
+
+ @return The new position.
+*/
+ring_buffer_size_t PaUtil_AdvanceRingBufferReadIndex(
+    PaUtilRingBuffer* rbuf,
+    ring_buffer_size_t elementCount);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif /* MODULES_THIRD_PARTY_PORTAUDIO_PA_RINGBUFFER_H_ */
diff --git a/third_party/libwebrtc/modules/utility/BUILD.gn b/third_party/libwebrtc/modules/utility/BUILD.gn
new file mode 100644
index 0000000000..46bca17f02
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/BUILD.gn
@@ -0,0 +1,78 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_source_set("utility") {
+  sources = [
+    "maybe_worker_thread.cc",
+    "maybe_worker_thread.h",
+  ]
+
+  deps = [
+    "../../api:field_trials_view",
+    "../../api:sequence_checker",
+    "../../api/task_queue",
+    "../../api/task_queue:pending_task_safety_flag",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rtc_event",
+    "../../rtc_base:rtc_task_queue",
+  ]
+
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
+    "//third_party/abseil-cpp/absl/strings:strings",
+  ]
+
+  if (is_android) {
+    visibility = [ "*" ]
+    sources += [
+      "include/helpers_android.h",
+      "include/jvm_android.h",
+      "source/helpers_android.cc",
+      "source/jvm_android.cc",
+    ]
+
+    deps += [
+      "../../api:sequence_checker",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:platform_thread",
+      "../../rtc_base/system:arch",
+    ]
+
+    if (build_with_mozilla) {
+      sources -= [ "source/jvm_android.cc" ]
+    }
+  }
+}
+
+if (rtc_include_tests) {
+  rtc_library("utility_unittests") {
+    testonly = true
+
+    sources = [ "maybe_worker_thread_unittests.cc" ]
+    deps = [
+      ":utility",
+      "../../api:sequence_checker",
+      "../../api/task_queue",
+      "../../api/task_queue:default_task_queue_factory",
+      "../../api/task_queue:pending_task_safety_flag",
+      "../../api/units:time_delta",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:threading",
+      "../../test:explicit_key_value_config",
+      "../../test:field_trial",
+      "../../test:test_main",
+      "../../test:test_support",
+      "../../test/time_controller",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/utility/DEPS b/third_party/libwebrtc/modules/utility/DEPS
new file mode 100644
index 0000000000..e9017d6b27
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+  "+common_audio",
+  "+common_video",
+  "+system_wrappers",
+]
diff --git a/third_party/libwebrtc/modules/utility/OWNERS b/third_party/libwebrtc/modules/utility/OWNERS
new file mode 100644
index 0000000000..5904b95df7
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/OWNERS
@@ -0,0 +1 @@
+perkj@webrtc.org
diff --git a/third_party/libwebrtc/modules/utility/include/helpers_android.h b/third_party/libwebrtc/modules/utility/include/helpers_android.h
new file mode 100644
index 0000000000..5e8deb9536
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/include/helpers_android.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+#define MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+
+#include <jni.h>
+
+#include <string>
+
+#include "rtc_base/system/arch.h"
+
+// Abort the process if `jni` has a Java exception pending.
+// TODO(henrika): merge with CHECK_JNI_EXCEPTION() in jni_helpers.h.
+#define CHECK_EXCEPTION(jni)        \
+  RTC_CHECK(!jni->ExceptionCheck()) \
+      << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
+
+#if defined(WEBRTC_ARCH_X86)
+// Dalvik JIT generated code doesn't guarantee 16-byte stack alignment on
+// x86 - use force_align_arg_pointer to realign the stack at the JNI
+// boundary. bugs.webrtc.org/9050
+#define JNI_FUNCTION_ALIGN __attribute__((force_align_arg_pointer))
+#else
+#define JNI_FUNCTION_ALIGN
+#endif
+
+namespace webrtc {
+
+// Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
+JNIEnv* GetEnv(JavaVM* jvm);
+
+// Return a `jlong` that will correctly convert back to `ptr`.  This is needed
+// because the alternative (of silently passing a 32-bit pointer to a vararg
+// function expecting a 64-bit param) picks up garbage in the high 32 bits.
+jlong PointerTojlong(void* ptr);
+
+// JNIEnv-helper methods that wraps the API which uses the JNI interface
+// pointer (JNIEnv*). It allows us to RTC_CHECK success and that no Java
+// exception is thrown while calling the method.
+jmethodID GetMethodID(JNIEnv* jni,
+                      jclass c,
+                      const char* name,
+                      const char* signature);
+
+jmethodID GetStaticMethodID(JNIEnv* jni,
+                            jclass c,
+                            const char* name,
+                            const char* signature);
+
+jclass FindClass(JNIEnv* jni, const char* name);
+
+jobject NewGlobalRef(JNIEnv* jni, jobject o);
+
+void DeleteGlobalRef(JNIEnv* jni, jobject o);
+
+// Attach thread to JVM if necessary and detach at scope end if originally
+// attached.
+class AttachThreadScoped {
+ public:
+  explicit AttachThreadScoped(JavaVM* jvm);
+  ~AttachThreadScoped();
+  JNIEnv* env();
+
+ private:
+  bool attached_;
+  JavaVM* jvm_;
+  JNIEnv* env_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
diff --git a/third_party/libwebrtc/modules/utility/include/jvm_android.h b/third_party/libwebrtc/modules/utility/include/jvm_android.h
new file mode 100644
index 0000000000..bafb8f2c15
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/include/jvm_android.h
@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+#define MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+
+#include <jni.h>
+
+#include <memory>
+#include <string>
+
+#include "api/sequence_checker.h"
+#include "modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+// RAII JavaVM AttachCurrentThread/DetachCurrentThread object.
+//
+// The JNI interface pointer (JNIEnv) is valid only in the current thread.
+// Should another thread need to access the Java VM, it must first call
+// AttachCurrentThread() to attach itself to the VM and obtain a JNI interface
+// pointer. The native thread remains attached to the VM until it calls
+// DetachCurrentThread() to detach.
+class JvmThreadConnector {
+ public:
+  JvmThreadConnector();
+  ~JvmThreadConnector();
+
+ private:
+  SequenceChecker thread_checker_;
+  bool attached_;
+};
+
+// This class is created by the NativeRegistration class and is used to wrap
+// the actual Java object handle (jobject) on which we can call methods from
+// C++ in to Java. See example in JVM for more details.
+// TODO(henrika): extend support for type of function calls.
+class GlobalRef {
+ public:
+  GlobalRef(JNIEnv* jni, jobject object);
+  ~GlobalRef();
+
+  jboolean CallBooleanMethod(jmethodID methodID, ...);
+  jint CallIntMethod(jmethodID methodID, ...);
+  void CallVoidMethod(jmethodID methodID, ...);
+
+ private:
+  JNIEnv* const jni_;
+  const jobject j_object_;
+};
+
+// Wraps the jclass object on which we can call GetMethodId() functions to
+// query method IDs.
+class JavaClass {
+ public:
+  JavaClass(JNIEnv* jni, jclass clazz) : jni_(jni), j_class_(clazz) {}
+  ~JavaClass() {}
+
+  jmethodID GetMethodId(const char* name, const char* signature);
+  jmethodID GetStaticMethodId(const char* name, const char* signature);
+  jobject CallStaticObjectMethod(jmethodID methodID, ...);
+  jint CallStaticIntMethod(jmethodID methodID, ...);
+
+ protected:
+  JNIEnv* const jni_;
+  jclass const j_class_;
+};
+
+// Adds support of the NewObject factory method to the JavaClass class.
+// See example in JVM for more details on how to use it.
+class NativeRegistration : public JavaClass {
+ public:
+  NativeRegistration(JNIEnv* jni, jclass clazz);
+  ~NativeRegistration();
+
+  std::unique_ptr<GlobalRef> NewObject(const char* name,
+                                       const char* signature,
+                                       ...);
+
+ private:
+  JNIEnv* const jni_;
+};
+
+// This class is created by the JVM class and is used to expose methods that
+// needs the JNI interface pointer but its main purpose is to create a
+// NativeRegistration object given name of a Java class and a list of native
+// methods. See example in JVM for more details.
+class JNIEnvironment {
+ public:
+  explicit JNIEnvironment(JNIEnv* jni);
+  ~JNIEnvironment();
+
+  // Registers native methods with the Java class specified by `name`.
+  // Note that the class name must be one of the names in the static
+  // `loaded_classes` array defined in jvm_android.cc.
+  // This method must be called on the construction thread.
+  std::unique_ptr<NativeRegistration> RegisterNatives(
+      const char* name,
+      const JNINativeMethod* methods,
+      int num_methods);
+
+  // Converts from Java string to std::string.
+  // This method must be called on the construction thread.
+  std::string JavaToStdString(const jstring& j_string);
+
+ private:
+  SequenceChecker thread_checker_;
+  JNIEnv* const jni_;
+};
+
+// Main class for working with Java from C++ using JNI in WebRTC.
+//
+// Example usage:
+//
+//   // At initialization (e.g. in JNI_OnLoad), call JVM::Initialize.
+//   JNIEnv* jni = ::base::android::AttachCurrentThread();
+//   JavaVM* jvm = NULL;
+//   jni->GetJavaVM(&jvm);
+//   webrtc::JVM::Initialize(jvm);
+//
+//   // Header (.h) file of example class called User.
+//   std::unique_ptr<JNIEnvironment> env;
+//   std::unique_ptr<NativeRegistration> reg;
+//   std::unique_ptr<GlobalRef> obj;
+//
+//   // Construction (in .cc file) of User class.
+//   User::User() {
+//     // Calling thread must be attached to the JVM.
+//     env = JVM::GetInstance()->environment();
+//     reg = env->RegisterNatives("org/webrtc/WebRtcTest", ,);
+//     obj = reg->NewObject("<init>", ,);
+//   }
+//
+//   // Each User method can now use `reg` and `obj` and call Java functions
+//   // in WebRtcTest.java, e.g. boolean init() {}.
+//   bool User::Foo() {
+//     jmethodID id = reg->GetMethodId("init", "()Z");
+//     return obj->CallBooleanMethod(id);
+//   }
+//
+//   // And finally, e.g. in JNI_OnUnLoad, call JVM::Uninitialize.
+//   JVM::Uninitialize();
+class JVM {
+ public:
+  // Stores global handles to the Java VM interface.
+  // Should be called once on a thread that is attached to the JVM.
+  static void Initialize(JavaVM* jvm);
+  // Like the method above but also passes the context to the ContextUtils
+  // class. This method should be used by pure-C++ Android users that can't call
+  // ContextUtils.initialize directly.
+  static void Initialize(JavaVM* jvm, jobject context);
+  // Clears handles stored in Initialize(). Must be called on same thread as
+  // Initialize().
+  static void Uninitialize();
+  // Gives access to the global Java VM interface pointer, which then can be
+  // used to create a valid JNIEnvironment object or to get a JavaClass object.
+  static JVM* GetInstance();
+
+  // Creates a JNIEnvironment object.
+  // This method returns a NULL pointer if AttachCurrentThread() has not been
+  // called successfully. Use the AttachCurrentThreadIfNeeded class if needed.
+  std::unique_ptr<JNIEnvironment> environment();
+
+  // Returns a JavaClass object given class `name`.
+  // Note that the class name must be one of the names in the static
+  // `loaded_classes` array defined in jvm_android.cc.
+  // This method must be called on the construction thread.
+  JavaClass GetClass(const char* name);
+
+  // TODO(henrika): can we make these private?
+  JavaVM* jvm() const { return jvm_; }
+
+ protected:
+  JVM(JavaVM* jvm);
+  ~JVM();
+
+ private:
+  JNIEnv* jni() const { return GetEnv(jvm_); }
+
+  SequenceChecker thread_checker_;
+  JavaVM* const jvm_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
diff --git a/third_party/libwebrtc/modules/utility/maybe_worker_thread.cc b/third_party/libwebrtc/modules/utility/maybe_worker_thread.cc
new file mode 100644
index 0000000000..abb52d4691
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/maybe_worker_thread.cc
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/utility/maybe_worker_thread.h"
+
+#include <utility>
+
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/task_queue.h"
+
+namespace webrtc {
+
+MaybeWorkerThread::MaybeWorkerThread(const FieldTrialsView& field_trials,
+                                     absl::string_view task_queue_name,
+                                     TaskQueueFactory* factory)
+    : owned_task_queue_(
+          field_trials.IsEnabled("WebRTC-SendPacketsOnWorkerThread")
+              ? nullptr
+              : factory->CreateTaskQueue(task_queue_name,
+                                         rtc::TaskQueue::Priority::NORMAL)),
+      worker_thread_(TaskQueueBase::Current()) {
+  RTC_DCHECK(worker_thread_);
+  RTC_LOG(LS_INFO) << "WebRTC-SendPacketsOnWorkerThread"
+                   << (owned_task_queue_ ? " Disabled" : " Enabled");
+}
+
+MaybeWorkerThread::~MaybeWorkerThread() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  if (owned_task_queue_) {
+    // owned_task_queue_ must be a valid pointer when the task queue is
+    // destroyed since there may be tasks that use this object that run when the
+    // task queue is deleted.
+    owned_task_queue_->Delete();
+    owned_task_queue_.release();
+  }
+}
+
+void MaybeWorkerThread::RunSynchronous(absl::AnyInvocable<void() &&> task) {
+  if (owned_task_queue_) {
+    rtc::Event thread_sync_event;
+    auto closure = [&thread_sync_event, task = std::move(task)]() mutable {
+      std::move(task)();
+      thread_sync_event.Set();
+    };
+    owned_task_queue_->PostTask(std::move(closure));
+    thread_sync_event.Wait(rtc::Event::kForever);
+  } else {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    std::move(task)();
+  }
+}
+
+void MaybeWorkerThread::RunOrPost(absl::AnyInvocable<void() &&> task) {
+  if (owned_task_queue_) {
+    owned_task_queue_->PostTask(std::move(task));
+  } else {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    std::move(task)();
+  }
+}
+
+TaskQueueBase* MaybeWorkerThread::TaskQueueForDelayedTasks() const {
+  RTC_DCHECK(IsCurrent());
+  return owned_task_queue_ ? owned_task_queue_.get() : worker_thread_;
+}
+
+TaskQueueBase* MaybeWorkerThread::TaskQueueForPost() const {
+  return owned_task_queue_ ? owned_task_queue_.get() : worker_thread_;
+}
+
+bool MaybeWorkerThread::IsCurrent() const {
+  if (owned_task_queue_) {
+    return owned_task_queue_->IsCurrent();
+  }
+  return worker_thread_->IsCurrent();
+}
+
+absl::AnyInvocable<void() &&> MaybeWorkerThread::MaybeSafeTask(
+    rtc::scoped_refptr<PendingTaskSafetyFlag> flag,
+    absl::AnyInvocable<void() &&> task) {
+  if (owned_task_queue_) {
+    return task;
+  } else {
+    return SafeTask(std::move(flag), std::move(task));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/utility/maybe_worker_thread.h b/third_party/libwebrtc/modules/utility/maybe_worker_thread.h
new file mode 100644
index 0000000000..a93a173bf6
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/maybe_worker_thread.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_UTILITY_MAYBE_WORKER_THREAD_H_
+#define MODULES_UTILITY_MAYBE_WORKER_THREAD_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Helper class used by experiment to replace usage of the
+// RTP worker task queue owned by RtpTransportControllerSend, and the pacer task
+// queue owned by TaskQueuePacedSender with the one and only worker thread.
+// Tasks will run on the target sequence which is either the worker thread or
+// one of these task queues depending on the field trial
+// "WebRTC-SendPacketsOnWorkerThread".
+// This class is assumed to be created on the worker thread and the worker
+// thread is assumed to outlive an instance of this class.
+//
+// Experiment can be tracked in
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=14502
+//
+// After experiment evaluation, this class should be deleted.
+// Calls to RunOrPost and RunSynchronous should be removed and the task should
+// be invoked immediately.
+// Instead of MaybeSafeTask a SafeTask should be used when posting tasks.
+class RTC_LOCKABLE MaybeWorkerThread {
+ public:
+  MaybeWorkerThread(const FieldTrialsView& field_trials,
+                    absl::string_view task_queue_name,
+                    TaskQueueFactory* factory);
+  ~MaybeWorkerThread();
+
+  // Runs `task` immediately on the worker thread if in experiment, otherwise
+  // post the task on the task queue.
+  void RunOrPost(absl::AnyInvocable<void() &&> task);
+  // Runs `task` immediately on the worker thread if in experiment, otherwise
+  // post the task on the task queue and use an even to wait for completion.
+  void RunSynchronous(absl::AnyInvocable<void() &&> task);
+
+  // Used for posting delayed or repeated tasks on the worker thread or task
+  // queue depending on the field trial. DCHECKs that this method is called on
+  // the target sequence.
+  TaskQueueBase* TaskQueueForDelayedTasks() const;
+
+  // Used when a task has to be posted from one sequence to the target
+  // sequence. A task should only be posted if a sequence hop is needed.
+  TaskQueueBase* TaskQueueForPost() const;
+
+  // Workaround to use a SafeTask only if the target sequence is the worker
+  // thread. This is used when a SafeTask can not be used because the object
+  // that posted the task is not destroyed on the target sequence. Instead, the
+  // caller has to guarantee that this MaybeWorkerThread is destroyed first
+  // since that guarantee that the posted task is deleted or run before the
+  // owning class.
+  absl::AnyInvocable<void() &&> MaybeSafeTask(
+      rtc::scoped_refptr<PendingTaskSafetyFlag> flag,
+      absl::AnyInvocable<void() &&> task);
+
+  // To implement macro RTC_DCHECK_RUN_ON.
+  // Implementation delegate to the actual used sequence.
+  bool IsCurrent() const;
+
+ private:
+  SequenceChecker sequence_checker_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> owned_task_queue_;
+  TaskQueueBase* const worker_thread_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_UTILITY_MAYBE_WORKER_THREAD_H_
diff --git a/third_party/libwebrtc/modules/utility/maybe_worker_thread_unittests.cc b/third_party/libwebrtc/modules/utility/maybe_worker_thread_unittests.cc
new file mode 100644
index 0000000000..f6a981f90a
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/maybe_worker_thread_unittests.cc
@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "modules/utility/maybe_worker_thread.h"
+#include "rtc_base/event.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gtest.h"
+#include "test/time_controller/real_time_controller.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr char kFieldTrialString[] =
+    "WebRTC-SendPacketsOnWorkerThread/Enabled/";
+
+TEST(MaybeWorkerThreadTest, RunOrPostRunOnWorkerThreadInExperiment) {
+  test::ExplicitKeyValueConfig field_trial(kFieldTrialString);
+  RealTimeController controller;
+  MaybeWorkerThread m(field_trial, "test_tq", controller.GetTaskQueueFactory());
+
+  SequenceChecker checker;
+  bool run = false;
+  m.RunOrPost([&] {
+    EXPECT_TRUE(checker.IsCurrent());
+    run = true;
+  });
+  EXPECT_TRUE(run);
+}
+
+TEST(MaybeWorkerThreadTest, RunOrPostPostsOnTqPerDefault) {
+  test::ExplicitKeyValueConfig field_trial("");
+  RealTimeController controller;
+  MaybeWorkerThread m(field_trial, "test_tq", controller.GetTaskQueueFactory());
+
+  SequenceChecker checker;
+  rtc::Event event;
+  m.RunOrPost([&] {
+    EXPECT_FALSE(checker.IsCurrent());
+    event.Set();
+  });
+  EXPECT_TRUE(event.Wait(TimeDelta::Seconds(10)));
+}
+
+TEST(MaybeWorkerThreadTest, RunSynchronousRunOnWorkerThreadInExperiment) {
+  test::ExplicitKeyValueConfig field_trial(kFieldTrialString);
+  RealTimeController controller;
+  MaybeWorkerThread m(field_trial, "test_tq", controller.GetTaskQueueFactory());
+
+  SequenceChecker checker;
+  bool run = false;
+  m.RunSynchronous([&] {
+    EXPECT_TRUE(checker.IsCurrent());
+    run = true;
+  });
+  EXPECT_TRUE(run);
+}
+
+TEST(MaybeWorkerThreadTest, RunSynchronousRunOnTqPerDefault) {
+  test::ExplicitKeyValueConfig field_trial("");
+  RealTimeController controller;
+  MaybeWorkerThread m(field_trial, "test_tq", controller.GetTaskQueueFactory());
+
+  SequenceChecker checker;
+  bool run = false;
+  m.RunSynchronous([&] {
+    EXPECT_FALSE(checker.IsCurrent());
+    run = true;
+  });
+  EXPECT_TRUE(run);
+}
+
+TEST(MaybeWorkerThreadTest, MaybeSafeTaskDoesNotReturnSafeTaskPerDefault) {
+  // We cant really test that the return value from MaybeSafeTask is a SafeTask.
+  // But we can test that the safety flag does not have more references after a
+  // call.
+  test::ExplicitKeyValueConfig field_trial("");
+  RealTimeController controller;
+  MaybeWorkerThread m(field_trial, "test_tq", controller.GetTaskQueueFactory());
+
+  rtc::scoped_refptr<PendingTaskSafetyFlag> flag =
+      PendingTaskSafetyFlag::Create();
+  auto closure = m.MaybeSafeTask(flag, [] {});
+  EXPECT_EQ(flag->Release(), rtc::RefCountReleaseStatus::kDroppedLastRef);
+  flag.release();
+}
+
+TEST(MaybeWorkerThreadTest, MaybeSafeTaskDoesNotReturnSafeTaskInExperiment) {
+  // We cant really test that the return value from MaybeSafeTask is a SafeTask.
+  // But we can test that the safety flag does have one more references after a
+  // call.
+  test::ExplicitKeyValueConfig field_trial(kFieldTrialString);
+  RealTimeController controller;
+  MaybeWorkerThread m(field_trial, "test_tq", controller.GetTaskQueueFactory());
+
+  rtc::scoped_refptr<PendingTaskSafetyFlag> flag =
+      PendingTaskSafetyFlag::Create();
+  auto closure = m.MaybeSafeTask(flag, [] {});
+  EXPECT_EQ(flag->Release(), rtc::RefCountReleaseStatus::kOtherRefsRemained);
+  flag.release();
+}
+
+TEST(MaybeWorkerThreadTest, IsCurrentBehavesCorrectPerDefault) {
+  test::ExplicitKeyValueConfig field_trial("");
+  RealTimeController controller;
+  MaybeWorkerThread m(field_trial, "test_tq", controller.GetTaskQueueFactory());
+
+  EXPECT_FALSE(m.IsCurrent());
+  m.RunSynchronous([&] { EXPECT_TRUE(m.IsCurrent()); });
+}
+
+TEST(MaybeWorkerThreadTest, IsCurrentBehavesCorrectInExperiment) {
+  test::ExplicitKeyValueConfig field_trial(kFieldTrialString);
+  RealTimeController controller;
+  MaybeWorkerThread m(field_trial, "test_tq", controller.GetTaskQueueFactory());
+
+  EXPECT_TRUE(m.IsCurrent());
+  auto tq = controller.GetTaskQueueFactory()->CreateTaskQueue(
+      "tq", TaskQueueFactory::Priority::NORMAL);
+  rtc::Event event;
+  tq->PostTask([&] {
+    EXPECT_FALSE(m.IsCurrent());
+    event.Set();
+  });
+  ASSERT_TRUE(event.Wait(TimeDelta::Seconds(10)));
+}
+
+TEST(MaybeWorkerThreadTest, IsCurrentCanBeCalledInDestructorPerDefault) {
+  test::ExplicitKeyValueConfig field_trial("");
+  RealTimeController controller;
+  {
+    MaybeWorkerThread m(field_trial, "test_tq",
+                        controller.GetTaskQueueFactory());
+    m.RunOrPost([&] { EXPECT_TRUE(m.IsCurrent()); });
+  }
+}
+
+TEST(MaybeWorkerThreadTest, IsCurrentCanBeCalledInDestructorInExperiment) {
+  test::ExplicitKeyValueConfig field_trial(kFieldTrialString);
+  RealTimeController controller;
+  {
+    MaybeWorkerThread m(field_trial, "test_tq",
+                        controller.GetTaskQueueFactory());
+    m.RunOrPost([&] { EXPECT_TRUE(m.IsCurrent()); });
+  }
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/utility/source/helpers_android.cc b/third_party/libwebrtc/modules/utility/source/helpers_android.cc
new file mode 100644
index 0000000000..9cfee8a2af
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/source/helpers_android.cc
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/utility/include/helpers_android.h"
+
+#include <android/log.h>
+#include <pthread.h>
+#include <stddef.h>
+#include <unistd.h>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/platform_thread.h"
+
+#define TAG "HelpersAndroid"
+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
+
+namespace webrtc {
+
+JNIEnv* GetEnv(JavaVM* jvm) {
+  void* env = NULL;
+  jint status = jvm->GetEnv(&env, JNI_VERSION_1_6);
+  RTC_CHECK(((env != NULL) && (status == JNI_OK)) ||
+            ((env == NULL) && (status == JNI_EDETACHED)))
+      << "Unexpected GetEnv return: " << status << ":" << env;
+  return reinterpret_cast<JNIEnv*>(env);
+}
+
+// Return a `jlong` that will correctly convert back to `ptr`.  This is needed
+// because the alternative (of silently passing a 32-bit pointer to a vararg
+// function expecting a 64-bit param) picks up garbage in the high 32 bits.
+jlong PointerTojlong(void* ptr) {
+  static_assert(sizeof(intptr_t) <= sizeof(jlong),
+                "Time to rethink the use of jlongs");
+  // Going through intptr_t to be obvious about the definedness of the
+  // conversion from pointer to integral type.  intptr_t to jlong is a standard
+  // widening by the static_assert above.
+  jlong ret = reinterpret_cast<intptr_t>(ptr);
+  RTC_DCHECK(reinterpret_cast<void*>(ret) == ptr);
+  return ret;
+}
+
+jmethodID GetMethodID(JNIEnv* jni,
+                      jclass c,
+                      const char* name,
+                      const char* signature) {
+  jmethodID m = jni->GetMethodID(c, name, signature);
+  CHECK_EXCEPTION(jni) << "Error during GetMethodID: " << name << ", "
+                       << signature;
+  RTC_CHECK(m) << name << ", " << signature;
+  return m;
+}
+
+jmethodID GetStaticMethodID(JNIEnv* jni,
+                            jclass c,
+                            const char* name,
+                            const char* signature) {
+  jmethodID m = jni->GetStaticMethodID(c, name, signature);
+  CHECK_EXCEPTION(jni) << "Error during GetStaticMethodID: " << name << ", "
+                       << signature;
+  RTC_CHECK(m) << name << ", " << signature;
+  return m;
+}
+
+jclass FindClass(JNIEnv* jni, const char* name) {
+  jclass c = jni->FindClass(name);
+  CHECK_EXCEPTION(jni) << "Error during FindClass: " << name;
+  RTC_CHECK(c) << name;
+  return c;
+}
+
+jobject NewGlobalRef(JNIEnv* jni, jobject o) {
+  jobject ret = jni->NewGlobalRef(o);
+  CHECK_EXCEPTION(jni) << "Error during NewGlobalRef";
+  RTC_CHECK(ret);
+  return ret;
+}
+
+void DeleteGlobalRef(JNIEnv* jni, jobject o) {
+  jni->DeleteGlobalRef(o);
+  CHECK_EXCEPTION(jni) << "Error during DeleteGlobalRef";
+}
+
+AttachThreadScoped::AttachThreadScoped(JavaVM* jvm)
+    : attached_(false), jvm_(jvm), env_(NULL) {
+  env_ = GetEnv(jvm);
+  if (!env_) {
+    // Adding debug log here so we can track down potential leaks and figure
+    // out why we sometimes see "Native thread exiting without having called
+    // DetachCurrentThread" in logcat outputs.
+    ALOGD("Attaching thread to JVM[tid=%d]", rtc::CurrentThreadId());
+    jint res = jvm->AttachCurrentThread(&env_, NULL);
+    attached_ = (res == JNI_OK);
+    RTC_CHECK(attached_) << "AttachCurrentThread failed: " << res;
+  }
+}
+
+AttachThreadScoped::~AttachThreadScoped() {
+  if (attached_) {
+    ALOGD("Detaching thread from JVM[tid=%d]", rtc::CurrentThreadId());
+    jint res = jvm_->DetachCurrentThread();
+    RTC_CHECK(res == JNI_OK) << "DetachCurrentThread failed: " << res;
+    RTC_CHECK(!GetEnv(jvm_));
+  }
+}
+
+JNIEnv* AttachThreadScoped::env() {
+  return env_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/utility/source/jvm_android.cc b/third_party/libwebrtc/modules/utility/source/jvm_android.cc
new file mode 100644
index 0000000000..39ef12f428
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/source/jvm_android.cc
@@ -0,0 +1,292 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/utility/include/jvm_android.h"
+
+#include <android/log.h>
+
+#include <memory>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+
+namespace mozilla {
+namespace jni {
+jclass GetClassRef(JNIEnv* aEnv, const char* aClassName);
+}
+}
+
+#define TAG "JVM"
+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
+
+namespace webrtc {
+
+JVM* g_jvm;
+
+// TODO(henrika): add more clases here if needed.
+struct {
+  const char* name;
+  jclass clazz;
+} loaded_classes[] = {
+    {"org/webrtc/voiceengine/BuildInfo", nullptr},
+    {"org/webrtc/voiceengine/WebRtcAudioManager", nullptr},
+    {"org/webrtc/voiceengine/WebRtcAudioRecord", nullptr},
+    {"org/webrtc/voiceengine/WebRtcAudioTrack", nullptr},
+};
+
+// Android's FindClass() is trickier than usual because the app-specific
+// ClassLoader is not consulted when there is no app-specific frame on the
+// stack.  Consequently, we only look up all classes once in native WebRTC.
+// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
+void LoadClasses(JNIEnv* jni) {
+  RTC_LOG(LS_INFO) << "LoadClasses:";
+  for (auto& c : loaded_classes) {
+    ALOGD("name: %s", c.name);
+    jclass clsRef = mozilla::jni::GetClassRef(jni, c.name);
+    RTC_CHECK(clsRef) << c.name;
+    c.clazz = static_cast<jclass>(jni->NewGlobalRef(clsRef));
+    jni->DeleteLocalRef(clsRef);
+  }
+}
+
+void FreeClassReferences(JNIEnv* jni) {
+  for (auto& c : loaded_classes) {
+    jni->DeleteGlobalRef(c.clazz);
+    c.clazz = nullptr;
+  }
+}
+
+jclass LookUpClass(const char* name) {
+  for (auto& c : loaded_classes) {
+    if (strcmp(c.name, name) == 0)
+      return c.clazz;
+  }
+  RTC_CHECK(false) << "Unable to find class in lookup table";
+  return 0;
+}
+
+// JvmThreadConnector implementation.
+JvmThreadConnector::JvmThreadConnector() : attached_(false) {
+  RTC_LOG(LS_INFO) << "JvmThreadConnector::ctor";
+  JavaVM* jvm = JVM::GetInstance()->jvm();
+  RTC_CHECK(jvm);
+  JNIEnv* jni = GetEnv(jvm);
+  if (!jni) {
+    RTC_LOG(LS_INFO) << "Attaching thread to JVM";
+    JNIEnv* env = nullptr;
+    jint ret = jvm->AttachCurrentThread(&env, nullptr);
+    attached_ = (ret == JNI_OK);
+  }
+}
+
+JvmThreadConnector::~JvmThreadConnector() {
+  RTC_LOG(LS_INFO) << "JvmThreadConnector::dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (attached_) {
+    RTC_LOG(LS_INFO) << "Detaching thread from JVM";
+    jint res = JVM::GetInstance()->jvm()->DetachCurrentThread();
+    RTC_CHECK(res == JNI_OK) << "DetachCurrentThread failed: " << res;
+  }
+}
+
+// GlobalRef implementation.
+GlobalRef::GlobalRef(JNIEnv* jni, jobject object)
+    : jni_(jni), j_object_(NewGlobalRef(jni, object)) {
+  RTC_LOG(LS_INFO) << "GlobalRef::ctor";
+}
+
+GlobalRef::~GlobalRef() {
+  RTC_LOG(LS_INFO) << "GlobalRef::dtor";
+  DeleteGlobalRef(jni_, j_object_);
+}
+
+jboolean GlobalRef::CallBooleanMethod(jmethodID methodID, ...) {
+  va_list args;
+  va_start(args, methodID);
+  jboolean res = jni_->CallBooleanMethodV(j_object_, methodID, args);
+  CHECK_EXCEPTION(jni_) << "Error during CallBooleanMethod";
+  va_end(args);
+  return res;
+}
+
+jint GlobalRef::CallIntMethod(jmethodID methodID, ...) {
+  va_list args;
+  va_start(args, methodID);
+  jint res = jni_->CallIntMethodV(j_object_, methodID, args);
+  CHECK_EXCEPTION(jni_) << "Error during CallIntMethod";
+  va_end(args);
+  return res;
+}
+
+void GlobalRef::CallVoidMethod(jmethodID methodID, ...) {
+  va_list args;
+  va_start(args, methodID);
+  jni_->CallVoidMethodV(j_object_, methodID, args);
+  CHECK_EXCEPTION(jni_) << "Error during CallVoidMethod";
+  va_end(args);
+}
+
+// NativeRegistration implementation.
+NativeRegistration::NativeRegistration(JNIEnv* jni, jclass clazz)
+    : JavaClass(jni, clazz), jni_(jni) {
+  RTC_LOG(LS_INFO) << "NativeRegistration::ctor";
+}
+
+NativeRegistration::~NativeRegistration() {
+  RTC_LOG(LS_INFO) << "NativeRegistration::dtor";
+  jni_->UnregisterNatives(j_class_);
+  CHECK_EXCEPTION(jni_) << "Error during UnregisterNatives";
+}
+
+std::unique_ptr<GlobalRef> NativeRegistration::NewObject(const char* name,
+                                                         const char* signature,
+                                                         ...) {
+  RTC_LOG(LS_INFO) << "NativeRegistration::NewObject";
+  va_list args;
+  va_start(args, signature);
+  jobject obj = jni_->NewObjectV(
+      j_class_, GetMethodID(jni_, j_class_, name, signature), args);
+  CHECK_EXCEPTION(jni_) << "Error during NewObjectV";
+  va_end(args);
+  return std::unique_ptr<GlobalRef>(new GlobalRef(jni_, obj));
+}
+
+// JavaClass implementation.
+jmethodID JavaClass::GetMethodId(const char* name, const char* signature) {
+  return GetMethodID(jni_, j_class_, name, signature);
+}
+
+jmethodID JavaClass::GetStaticMethodId(const char* name,
+                                       const char* signature) {
+  return GetStaticMethodID(jni_, j_class_, name, signature);
+}
+
+jobject JavaClass::CallStaticObjectMethod(jmethodID methodID, ...) {
+  va_list args;
+  va_start(args, methodID);
+  jobject res = jni_->CallStaticObjectMethodV(j_class_, methodID, args);
+  CHECK_EXCEPTION(jni_) << "Error during CallStaticObjectMethod";
+  return res;
+}
+
+jint JavaClass::CallStaticIntMethod(jmethodID methodID, ...) {
+  va_list args;
+  va_start(args, methodID);
+  jint res = jni_->CallStaticIntMethodV(j_class_, methodID, args);
+  CHECK_EXCEPTION(jni_) << "Error during CallStaticIntMethod";
+  return res;
+}
+
+// JNIEnvironment implementation.
+JNIEnvironment::JNIEnvironment(JNIEnv* jni) : jni_(jni) {
+  RTC_LOG(LS_INFO) << "JNIEnvironment::ctor";
+}
+
+JNIEnvironment::~JNIEnvironment() {
+  RTC_LOG(LS_INFO) << "JNIEnvironment::dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+}
+
+std::unique_ptr<NativeRegistration> JNIEnvironment::RegisterNatives(
+    const char* name,
+    const JNINativeMethod* methods,
+    int num_methods) {
+  RTC_LOG(LS_INFO) << "JNIEnvironment::RegisterNatives: " << name;
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  jclass clazz = LookUpClass(name);
+  jni_->RegisterNatives(clazz, methods, num_methods);
+  CHECK_EXCEPTION(jni_) << "Error during RegisterNatives";
+  return std::unique_ptr<NativeRegistration>(
+      new NativeRegistration(jni_, clazz));
+}
+
+std::string JNIEnvironment::JavaToStdString(const jstring& j_string) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  const char* jchars = jni_->GetStringUTFChars(j_string, nullptr);
+  CHECK_EXCEPTION(jni_);
+  const int size = jni_->GetStringUTFLength(j_string);
+  CHECK_EXCEPTION(jni_);
+  std::string ret(jchars, size);
+  jni_->ReleaseStringUTFChars(j_string, jchars);
+  CHECK_EXCEPTION(jni_);
+  return ret;
+}
+
+// static
+void JVM::Initialize(JavaVM* jvm) {
+  if (g_jvm) {
+    return;
+  }
+  g_jvm = new JVM(jvm);
+}
+
+void JVM::Initialize(JavaVM* jvm, jobject context) {
+  Initialize(jvm);
+
+  // Pass in the context to the new ContextUtils class.
+  JNIEnv* jni = g_jvm->jni();
+  jclass context_utils = FindClass(jni, "org/webrtc/ContextUtils");
+  jmethodID initialize_method = jni->GetStaticMethodID(
+      context_utils, "initialize", "(Landroid/content/Context;)V");
+  jni->CallStaticVoidMethod(context_utils, initialize_method, context);
+}
+
+// static
+void JVM::Uninitialize() {
+  RTC_LOG(LS_INFO) << "JVM::Uninitialize";
+  RTC_DCHECK(g_jvm);
+  delete g_jvm;
+  g_jvm = nullptr;
+}
+
+// static
+JVM* JVM::GetInstance() {
+  RTC_DCHECK(g_jvm);
+  return g_jvm;
+}
+
+JVM::JVM(JavaVM* jvm) : jvm_(jvm) {
+  RTC_LOG(LS_INFO) << "JVM::JVM";
+  RTC_CHECK(jni()) << "AttachCurrentThread() must be called on this thread.";
+  LoadClasses(jni());
+}
+
+JVM::~JVM() {
+  RTC_LOG(LS_INFO) << "JVM::~JVM";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  FreeClassReferences(jni());
+}
+
+std::unique_ptr<JNIEnvironment> JVM::environment() {
+  RTC_LOG(LS_INFO) << "JVM::environment";
+  ;
+  // The JNIEnv is used for thread-local storage. For this reason, we cannot
+  // share a JNIEnv between threads. If a piece of code has no other way to get
+  // its JNIEnv, we should share the JavaVM, and use GetEnv to discover the
+  // thread's JNIEnv. (Assuming it has one, if not, use AttachCurrentThread).
+  // See // http://developer.android.com/training/articles/perf-jni.html.
+  JNIEnv* jni = GetEnv(jvm_);
+  if (!jni) {
+    RTC_LOG(LS_ERROR)
+        << "AttachCurrentThread() has not been called on this thread";
+    return std::unique_ptr<JNIEnvironment>();
+  }
+  return std::unique_ptr<JNIEnvironment>(new JNIEnvironment(jni));
+}
+
+JavaClass JVM::GetClass(const char* name) {
+  RTC_LOG(LS_INFO) << "JVM::GetClass: " << name;
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return JavaClass(jni(), LookUpClass(name));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/utility/utility_gn/moz.build b/third_party/libwebrtc/modules/utility/utility_gn/moz.build
new file mode 100644
index 0000000000..b932f5da73
--- /dev/null
+++ b/third_party/libwebrtc/modules/utility/utility_gn/moz.build
@@ -0,0 +1,229 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/utility/maybe_worker_thread.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/utility/source/helpers_android.cc"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("utility_gn")
diff --git a/third_party/libwebrtc/modules/video_capture/BUILD.gn b/third_party/libwebrtc/modules/video_capture/BUILD.gn
new file mode 100644
index 0000000000..4a5bf62433
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/BUILD.gn
@@ -0,0 +1,152 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+# Note this target is missing an implementation for the video capture.
+# Targets must link with either 'video_capture' or
+# 'video_capture_internal_impl' depending on whether they want to
+# use the internal capturer.
+rtc_library("video_capture_module") {
+  visibility = [ "*" ]
+  sources = [
+    "device_info_impl.cc",
+    "device_info_impl.h",
+    "raw_video_sink_interface.h",
+    "video_capture.h",
+    "video_capture_config.h",
+    "video_capture_defines.h",
+    "video_capture_factory.cc",
+    "video_capture_factory.h",
+    "video_capture_impl.cc",
+    "video_capture_impl.h",
+  ]
+
+  deps = [
+    "../../api:scoped_refptr",
+    "../../api/video:video_frame",
+    "../../api/video:video_rtp_headers",
+    "../../common_video",
+    "../../media:rtc_media_base",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:refcount",
+    "../../rtc_base:stringutils",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
+    "//third_party/libyuv",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+if (!build_with_chromium) {
+  rtc_source_set("video_capture_internal_impl") {
+    visibility = [ "*" ]
+    deps = [
+      ":video_capture_module",
+      "../../api:scoped_refptr",
+      "../../api:sequence_checker",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:platform_thread",
+      "../../rtc_base:refcount",
+      "../../rtc_base:stringutils",
+      "../../rtc_base/synchronization:mutex",
+      "../../system_wrappers",
+    ]
+
+    if (is_linux || is_bsd || is_chromeos) {
+      sources = [
+        "linux/device_info_linux.cc",
+        "linux/device_info_v4l2.cc",
+        "linux/device_info_v4l2.h",
+        "linux/video_capture_linux.cc",
+        "linux/video_capture_v4l2.cc",
+        "linux/video_capture_v4l2.h",
+      ]
+      deps += [ "../../media:rtc_media_base" ]
+    }
+    if (is_win) {
+      sources = [
+        "windows/device_info_ds.cc",
+        "windows/device_info_ds.h",
+        "windows/help_functions_ds.cc",
+        "windows/help_functions_ds.h",
+        "windows/sink_filter_ds.cc",
+        "windows/sink_filter_ds.h",
+        "windows/video_capture_ds.cc",
+        "windows/video_capture_ds.h",
+        "windows/video_capture_factory_windows.cc",
+      ]
+
+      libs = [
+        "ole32.lib",
+        "oleaut32.lib",
+        "strmiids.lib",
+        "user32.lib",
+      ]
+    }
+    if (is_fuchsia) {
+      sources = [ "video_capture_factory_null.cc" ]
+    }
+
+    if (!build_with_mozilla && is_android) {
+      include_dirs = [
+        "/config/external/nspr",
+        "/nsprpub/lib/ds",
+        "/nsprpub/pr/include",
+      ]
+
+      sources = [
+        "android/device_info_android.cc",
+        "android/video_capture_android.cc",
+      ]
+    }
+  }
+
+  if (!is_android && rtc_include_tests) {
+    rtc_test("video_capture_tests") {
+      sources = [ "test/video_capture_unittest.cc" ]
+      ldflags = []
+      if (is_linux || is_chromeos || is_mac) {
+        ldflags += [
+          "-lpthread",
+          "-lm",
+        ]
+      }
+      if (is_linux || is_chromeos) {
+        ldflags += [
+          "-lrt",
+          "-lXext",
+          "-lX11",
+        ]
+      }
+
+      deps = [
+        ":video_capture_internal_impl",
+        ":video_capture_module",
+        "../../api:scoped_refptr",
+        "../../api/video:video_frame",
+        "../../api/video:video_rtp_headers",
+        "../../common_video",
+        "../../rtc_base:timeutils",
+        "../../rtc_base/synchronization:mutex",
+        "../../system_wrappers",
+        "../../test:frame_utils",
+        "../../test:test_main",
+        "../../test:test_support",
+        "../../test:video_test_common",
+        "//testing/gtest",
+        "//third_party/abseil-cpp/absl/memory",
+      ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/video_capture/DEPS b/third_party/libwebrtc/modules/video_capture/DEPS
new file mode 100644
index 0000000000..9ad1d576bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/DEPS
@@ -0,0 +1,6 @@
+include_rules = [
+  "+common_video",
+  "+media/base",
+  "+system_wrappers",
+  "+third_party/libyuv",
+]
diff --git a/third_party/libwebrtc/modules/video_capture/OWNERS b/third_party/libwebrtc/modules/video_capture/OWNERS
new file mode 100644
index 0000000000..364d66d36f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/OWNERS
@@ -0,0 +1,4 @@
+ilnik@webrtc.org
+mflodman@webrtc.org
+perkj@webrtc.org
+tkchin@webrtc.org
diff --git a/third_party/libwebrtc/modules/video_capture/device_info_impl.cc b/third_party/libwebrtc/modules/video_capture/device_info_impl.cc
new file mode 100644
index 0000000000..2a6afb3147
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/device_info_impl.cc
@@ -0,0 +1,224 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/device_info_impl.h"
+
+#include <stdlib.h>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "rtc_base/logging.h"
+
+#ifndef abs
+#define abs(a) (a >= 0 ? a : -a)
+#endif
+
+namespace webrtc {
+namespace videocapturemodule {
+
+DeviceInfoImpl::DeviceInfoImpl()
+    : _lastUsedDeviceName(NULL), _lastUsedDeviceNameLength(0) {}
+
+DeviceInfoImpl::~DeviceInfoImpl(void) {
+  MutexLock lock(&_apiLock);
+  free(_lastUsedDeviceName);
+}
+
+int32_t DeviceInfoImpl::NumberOfCapabilities(const char* deviceUniqueIdUTF8) {
+  if (!deviceUniqueIdUTF8)
+    return -1;
+
+  MutexLock lock(&_apiLock);
+
+  // Is it the same device that is asked for again.
+  if (absl::EqualsIgnoreCase(
+          deviceUniqueIdUTF8,
+          absl::string_view(_lastUsedDeviceName, _lastUsedDeviceNameLength))) {
+    return static_cast<int32_t>(_captureCapabilities.size());
+  }
+
+  int32_t ret = CreateCapabilityMap(deviceUniqueIdUTF8);
+  return ret;
+}
+
+int32_t DeviceInfoImpl::GetCapability(const char* deviceUniqueIdUTF8,
+                                      const uint32_t deviceCapabilityNumber,
+                                      VideoCaptureCapability& capability) {
+  RTC_DCHECK(deviceUniqueIdUTF8);
+
+  MutexLock lock(&_apiLock);
+
+  if (!absl::EqualsIgnoreCase(
+          deviceUniqueIdUTF8,
+          absl::string_view(_lastUsedDeviceName, _lastUsedDeviceNameLength))) {
+    if (-1 == CreateCapabilityMap(deviceUniqueIdUTF8)) {
+      return -1;
+    }
+  }
+
+  // Make sure the number is valid
+  if (deviceCapabilityNumber >= (unsigned int)_captureCapabilities.size()) {
+    RTC_LOG(LS_ERROR) << deviceUniqueIdUTF8 << " Invalid deviceCapabilityNumber "
+                      << deviceCapabilityNumber << ">= number of capabilities ("
+                      << _captureCapabilities.size() << ").";
+    return -1;
+  }
+
+  capability = _captureCapabilities[deviceCapabilityNumber];
+  return 0;
+}
+
+int32_t DeviceInfoImpl::GetBestMatchedCapability(
+    const char* deviceUniqueIdUTF8,
+    const VideoCaptureCapability& requested,
+    VideoCaptureCapability& resulting) {
+  if (!deviceUniqueIdUTF8)
+    return -1;
+
+  MutexLock lock(&_apiLock);
+  if (!absl::EqualsIgnoreCase(
+          deviceUniqueIdUTF8,
+          absl::string_view(_lastUsedDeviceName, _lastUsedDeviceNameLength))) {
+    if (-1 == CreateCapabilityMap(deviceUniqueIdUTF8)) {
+      return -1;
+    }
+  }
+
+  int32_t bestformatIndex = -1;
+  int32_t bestWidth = 0;
+  int32_t bestHeight = 0;
+  int32_t bestFrameRate = 0;
+  VideoType bestVideoType = VideoType::kUnknown;
+
+  const int32_t numberOfCapabilies =
+      static_cast<int32_t>(_captureCapabilities.size());
+
+  bool hasNonRGB24Capability = false;
+  for (int32_t tmp = 0; tmp < numberOfCapabilies;
+       ++tmp)  // Loop through all capabilities
+  {
+    VideoCaptureCapability& capability = _captureCapabilities[tmp];
+    if (capability.videoType != VideoType::kRGB24) {
+      hasNonRGB24Capability = true;
+    }
+  }
+
+  for (int32_t tmp = 0; tmp < numberOfCapabilies;
+       ++tmp)  // Loop through all capabilities
+  {
+    VideoCaptureCapability& capability = _captureCapabilities[tmp];
+    if (hasNonRGB24Capability && capability.videoType == VideoType::kRGB24) {
+      continue;
+    }
+
+    const int32_t diffWidth = capability.width - requested.width;
+    const int32_t diffHeight = capability.height - requested.height;
+    const int32_t diffFrameRate = capability.maxFPS - requested.maxFPS;
+
+    const int32_t currentbestDiffWith = bestWidth - requested.width;
+    const int32_t currentbestDiffHeight = bestHeight - requested.height;
+    const int32_t currentbestDiffFrameRate = bestFrameRate - requested.maxFPS;
+
+    if ((diffHeight >= 0 &&
+         diffHeight <= abs(currentbestDiffHeight))  // Height better or equalt
+                                                    // that previouse.
+        || (currentbestDiffHeight < 0 && diffHeight >= currentbestDiffHeight)) {
+      if (diffHeight ==
+          currentbestDiffHeight)  // Found best height. Care about the width)
+      {
+        if ((diffWidth >= 0 &&
+             diffWidth <= abs(currentbestDiffWith))  // Width better or equal
+            || (currentbestDiffWith < 0 && diffWidth >= currentbestDiffWith)) {
+          if (diffWidth == currentbestDiffWith &&
+              diffHeight == currentbestDiffHeight)  // Same size as previously
+          {
+            // Also check the best frame rate if the diff is the same as
+            // previouse
+            if (((diffFrameRate >= 0 &&
+                  diffFrameRate <=
+                      currentbestDiffFrameRate)  // Frame rate to high but
+                                                 // better match than previouse
+                                                 // and we have not selected IUV
+                 || (currentbestDiffFrameRate < 0 &&
+                     diffFrameRate >=
+                         currentbestDiffFrameRate))  // Current frame rate is
+                                                     // lower than requested.
+                                                     // This is better.
+            ) {
+              if ((currentbestDiffFrameRate ==
+                   diffFrameRate)  // Same frame rate as previous  or frame rate
+                                   // allready good enough
+                  || (currentbestDiffFrameRate >= 0)) {
+                if (bestVideoType != requested.videoType &&
+                    requested.videoType != VideoType::kUnknown &&
+                    (capability.videoType == requested.videoType ||
+                     capability.videoType == VideoType::kI420 ||
+                     capability.videoType == VideoType::kYUY2 ||
+                     capability.videoType == VideoType::kYV12 ||
+                     capability.videoType == VideoType::kNV12)) {
+                  bestVideoType = capability.videoType;
+                  bestformatIndex = tmp;
+                }
+                // If width height and frame rate is full filled we can use the
+                // camera for encoding if it is supported.
+                if (capability.height == requested.height &&
+                    capability.width == requested.width &&
+                    capability.maxFPS >= requested.maxFPS) {
+                  bestformatIndex = tmp;
+                }
+              } else  // Better frame rate
+              {
+                bestWidth = capability.width;
+                bestHeight = capability.height;
+                bestFrameRate = capability.maxFPS;
+                bestVideoType = capability.videoType;
+                bestformatIndex = tmp;
+              }
+            }
+          } else  // Better width than previously
+          {
+            bestWidth = capability.width;
+            bestHeight = capability.height;
+            bestFrameRate = capability.maxFPS;
+            bestVideoType = capability.videoType;
+            bestformatIndex = tmp;
+          }
+        }     // else width no good
+      } else  // Better height
+      {
+        bestWidth = capability.width;
+        bestHeight = capability.height;
+        bestFrameRate = capability.maxFPS;
+        bestVideoType = capability.videoType;
+        bestformatIndex = tmp;
+      }
+    }  // else height not good
+  }    // end for
+
+  RTC_LOG(LS_VERBOSE) << "Best camera format: " << bestWidth << "x"
+                      << bestHeight << "@" << bestFrameRate
+                      << "fps, color format: "
+                      << static_cast<int>(bestVideoType);
+
+  // Copy the capability
+  if (bestformatIndex < 0)
+    return -1;
+  resulting = _captureCapabilities[bestformatIndex];
+  return bestformatIndex;
+}
+
+// Default implementation. This should be overridden by Mobile implementations.
+int32_t DeviceInfoImpl::GetOrientation(const char* deviceUniqueIdUTF8,
+                                       VideoRotation& orientation) {
+  orientation = kVideoRotation_0;
+  return -1;
+}
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/device_info_impl.h b/third_party/libwebrtc/modules/video_capture/device_info_impl.h
new file mode 100644
index 0000000000..8acbef6d69
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/device_info_impl.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
+#define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/video/video_rotation.h"
+#include "modules/video_capture/video_capture.h"
+#include "modules/video_capture/video_capture_defines.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+class DeviceInfoImpl : public VideoCaptureModule::DeviceInfo {
+ public:
+  DeviceInfoImpl();
+  ~DeviceInfoImpl(void) override;
+  int32_t NumberOfCapabilities(const char* deviceUniqueIdUTF8) override;
+  int32_t GetCapability(const char* deviceUniqueIdUTF8,
+                        uint32_t deviceCapabilityNumber,
+                        VideoCaptureCapability& capability) override;
+
+  int32_t GetBestMatchedCapability(const char* deviceUniqueIdUTF8,
+                                   const VideoCaptureCapability& requested,
+                                   VideoCaptureCapability& resulting) override;
+  int32_t GetOrientation(const char* deviceUniqueIdUTF8,
+                         VideoRotation& orientation) override;
+
+ protected:
+  /* Initialize this object*/
+
+  virtual int32_t Init() = 0;
+  int32_t Refresh() override { return 0; }
+  /*
+   * Fills the member variable _captureCapabilities with capabilities for the
+   * given device name.
+   */
+  virtual int32_t CreateCapabilityMap(const char* deviceUniqueIdUTF8)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(_apiLock) = 0;
+
+ protected:
+  // Data members
+  typedef std::vector<VideoCaptureCapability> VideoCaptureCapabilities;
+  VideoCaptureCapabilities _captureCapabilities RTC_GUARDED_BY(_apiLock);
+  Mutex _apiLock;
+  char* _lastUsedDeviceName RTC_GUARDED_BY(_apiLock);
+  uint32_t _lastUsedDeviceNameLength RTC_GUARDED_BY(_apiLock);
+};
+}  // namespace videocapturemodule
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
diff --git a/third_party/libwebrtc/modules/video_capture/linux/device_info_linux.cc b/third_party/libwebrtc/modules/video_capture/linux/device_info_linux.cc
new file mode 100644
index 0000000000..4821cbccd5
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/linux/device_info_linux.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <errno.h>
+#include <fcntl.h>
+#include <poll.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
+#include <linux/videodev2.h>
+#endif
+
+#include <vector>
+
+#include "modules/video_capture/linux/device_info_v4l2.h"
+#include "modules/video_capture/video_capture.h"
+#include "modules/video_capture/video_capture_defines.h"
+#include "modules/video_capture/video_capture_impl.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo() {
+  return new videocapturemodule::DeviceInfoV4l2();
+}
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.cc b/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.cc
new file mode 100644
index 0000000000..ccd4b2bd2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.cc
@@ -0,0 +1,517 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/linux/device_info_v4l2.h"
+
+#include <errno.h>
+#include <fcntl.h>
+#include <poll.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
+#include <linux/videodev2.h>
+#endif
+
+#include <vector>
+
+#include "modules/video_capture/video_capture.h"
+#include "modules/video_capture/video_capture_defines.h"
+#include "modules/video_capture/video_capture_impl.h"
+#include "rtc_base/logging.h"
+
+#ifdef WEBRTC_LINUX
+#define EVENT_SIZE  ( sizeof (struct inotify_event) )
+#define BUF_LEN     ( 1024 * ( EVENT_SIZE + 16 ) )
+#endif
+
+// These defines are here to support building on kernel 3.16 which some
+// downstream projects, e.g. Firefox, use.
+// TODO(apehrson): Remove them and their undefs when no longer needed.
+#ifndef V4L2_PIX_FMT_ABGR32
+#define ABGR32_OVERRIDE 1
+#define V4L2_PIX_FMT_ABGR32 v4l2_fourcc('A', 'R', '2', '4')
+#endif
+
+#ifndef V4L2_PIX_FMT_ARGB32
+#define ARGB32_OVERRIDE 1
+#define V4L2_PIX_FMT_ARGB32 v4l2_fourcc('B', 'A', '2', '4')
+#endif
+
+#ifndef V4L2_PIX_FMT_RGBA32
+#define RGBA32_OVERRIDE 1
+#define V4L2_PIX_FMT_RGBA32 v4l2_fourcc('A', 'B', '2', '4')
+#endif
+
+namespace webrtc {
+namespace videocapturemodule {
+#ifdef WEBRTC_LINUX
+void DeviceInfoV4l2::HandleEvent(inotify_event* event, int fd)
+{
+    if (event->mask & IN_CREATE) {
+        if (fd == _fd_v4l) {
+            DeviceChange();
+        } else if ((event->mask & IN_ISDIR) && (fd == _fd_dev)) {
+            if (_wd_v4l < 0) {
+                // Sometimes inotify_add_watch failed if we call it immediately after receiving this event
+                // Adding 5ms delay to let file system settle down
+                usleep(5*1000);
+                _wd_v4l = inotify_add_watch(_fd_v4l, "/dev/v4l/by-path/", IN_CREATE | IN_DELETE | IN_DELETE_SELF);
+                if (_wd_v4l >= 0) {
+                    DeviceChange();
+                }
+            }
+        }
+    } else if (event->mask & IN_DELETE) {
+        if (fd == _fd_v4l) {
+            DeviceChange();
+        }
+    } else if (event->mask & IN_DELETE_SELF) {
+        if (fd == _fd_v4l) {
+            inotify_rm_watch(_fd_v4l, _wd_v4l);
+            _wd_v4l = -1;
+        } else {
+            assert(false);
+        }
+    }
+}
+
+int DeviceInfoV4l2::EventCheck(int fd)
+{
+    struct pollfd fds = {
+      .fd = fd,
+      .events = POLLIN,
+      .revents = 0
+    };
+
+    return poll(&fds, 1, 100);
+}
+
+int DeviceInfoV4l2::HandleEvents(int fd)
+{
+    char buffer[BUF_LEN];
+
+    ssize_t r = read(fd, buffer, BUF_LEN);
+
+    if (r <= 0) {
+        return r;
+    }
+
+    ssize_t buffer_i = 0;
+    inotify_event* pevent;
+    size_t eventSize;
+    int count = 0;
+
+    while (buffer_i < r)
+    {
+        pevent = (inotify_event *) (&buffer[buffer_i]);
+        eventSize = sizeof(inotify_event) + pevent->len;
+        char event[sizeof(inotify_event) + FILENAME_MAX + 1] // null-terminated
+            __attribute__ ((aligned(__alignof__(struct inotify_event))));
+
+        memcpy(event, pevent, eventSize);
+
+        HandleEvent((inotify_event*)(event), fd);
+
+        buffer_i += eventSize;
+        count++;
+    }
+
+    return count;
+}
+
+int DeviceInfoV4l2::ProcessInotifyEvents()
+{
+    while (!_isShutdown) {
+        if (EventCheck(_fd_dev) > 0) {
+            if (HandleEvents(_fd_dev) < 0) {
+                break;
+            }
+        }
+        if (EventCheck(_fd_v4l) > 0) {
+            if (HandleEvents(_fd_v4l) < 0) {
+                break;
+            }
+        }
+    }
+    return 0;
+}
+
+void DeviceInfoV4l2::InotifyProcess()
+{
+    _fd_v4l = inotify_init();
+    _fd_dev = inotify_init();
+    if (_fd_v4l >= 0 && _fd_dev >= 0) {
+        _wd_v4l = inotify_add_watch(_fd_v4l, "/dev/v4l/by-path/", IN_CREATE | IN_DELETE | IN_DELETE_SELF);
+        _wd_dev = inotify_add_watch(_fd_dev, "/dev/", IN_CREATE);
+        ProcessInotifyEvents();
+
+        if (_wd_v4l >= 0) {
+          inotify_rm_watch(_fd_v4l, _wd_v4l);
+        }
+
+        if (_wd_dev >= 0) {
+          inotify_rm_watch(_fd_dev, _wd_dev);
+        }
+
+        close(_fd_v4l);
+        close(_fd_dev);
+    }
+}
+#endif
+
+DeviceInfoV4l2::DeviceInfoV4l2() : DeviceInfoImpl()
+#ifdef WEBRTC_LINUX
+    , _isShutdown(false)
+#endif
+{
+#ifdef WEBRTC_LINUX
+  _inotifyEventThread = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        InotifyProcess();
+      }, "InotifyEventThread");
+#endif
+}
+
+int32_t DeviceInfoV4l2::Init() {
+  return 0;
+}
+
+DeviceInfoV4l2::~DeviceInfoV4l2() {
+#ifdef WEBRTC_LINUX
+    _isShutdown = true;
+
+    if (!_inotifyEventThread.empty()) {
+      _inotifyEventThread.Finalize();
+    }
+#endif
+}
+
+uint32_t DeviceInfoV4l2::NumberOfDevices() {
+  uint32_t count = 0;
+  char device[20];
+  int fd = -1;
+  struct v4l2_capability cap;
+
+  /* detect /dev/video [0-63]VideoCaptureModule entries */
+  for (int n = 0; n < 64; n++) {
+    snprintf(device, sizeof(device), "/dev/video%d", n);
+    if ((fd = open(device, O_RDONLY)) != -1) {
+      // query device capabilities and make sure this is a video capture device
+      if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 || !IsVideoCaptureDevice(&cap)) {
+        close(fd);
+        continue;
+      }
+
+      close(fd);
+      count++;
+    }
+  }
+
+  return count;
+}
+
+int32_t DeviceInfoV4l2::GetDeviceName(uint32_t deviceNumber,
+                                      char* deviceNameUTF8,
+                                      uint32_t deviceNameLength,
+                                      char* deviceUniqueIdUTF8,
+                                      uint32_t deviceUniqueIdUTF8Length,
+                                      char* /*productUniqueIdUTF8*/,
+                                      uint32_t /*productUniqueIdUTF8Length*/,
+                                      pid_t* /*pid*/) {
+  // Travel through /dev/video [0-63]
+  uint32_t count = 0;
+  char device[20];
+  int fd = -1;
+  bool found = false;
+  struct v4l2_capability cap;
+  int device_index;
+  for (device_index = 0; device_index < 64; device_index++) {
+    sprintf(device, "/dev/video%d", device_index);
+    if ((fd = open(device, O_RDONLY)) != -1) {
+      // query device capabilities and make sure this is a video capture device
+      if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 || !IsVideoCaptureDevice(&cap)) {
+        close(fd);
+        continue;
+      }
+      if (count == deviceNumber) {
+        // Found the device
+        found = true;
+        break;
+      } else {
+        close(fd);
+        count++;
+      }
+    }
+  }
+
+  if (!found)
+    return -1;
+
+  // query device capabilities
+  if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0) {
+    RTC_LOG(LS_INFO) << "error in querying the device capability for device "
+                     << device << ". errno = " << errno;
+    close(fd);
+    return -1;
+  }
+
+  close(fd);
+
+  char cameraName[64];
+  memset(deviceNameUTF8, 0, deviceNameLength);
+  memcpy(cameraName, cap.card, sizeof(cap.card));
+
+  if (deviceNameLength > strlen(cameraName)) {
+    memcpy(deviceNameUTF8, cameraName, strlen(cameraName));
+  } else {
+    RTC_LOG(LS_INFO) << "buffer passed is too small";
+    return -1;
+  }
+
+  if (cap.bus_info[0] != 0) {  // may not available in all drivers
+    // copy device id
+    size_t len = strlen(reinterpret_cast<const char*>(cap.bus_info));
+    if (deviceUniqueIdUTF8Length > len) {
+      memset(deviceUniqueIdUTF8, 0, deviceUniqueIdUTF8Length);
+      memcpy(deviceUniqueIdUTF8, cap.bus_info, len);
+    } else {
+      RTC_LOG(LS_INFO) << "buffer passed is too small";
+      return -1;
+    }
+  } else {
+    // if there's no bus info to use for uniqueId, invent one - and it has to be repeatable
+    if (snprintf(deviceUniqueIdUTF8,
+                 deviceUniqueIdUTF8Length, "fake_%u", device_index) >=
+        (int) deviceUniqueIdUTF8Length)
+    {
+      return -1;
+    }
+  }
+  return 0;
+}
+
+int32_t DeviceInfoV4l2::CreateCapabilityMap(const char* deviceUniqueIdUTF8) {
+  int fd;
+  char device[32];
+  bool found = false;
+
+  const int32_t deviceUniqueIdUTF8Length = strlen(deviceUniqueIdUTF8);
+  if (deviceUniqueIdUTF8Length >= kVideoCaptureUniqueNameLength) {
+    RTC_LOG(LS_INFO) << "Device name too long";
+    return -1;
+  }
+  RTC_LOG(LS_INFO) << "CreateCapabilityMap called for device "
+                   << deviceUniqueIdUTF8;
+
+  /* detect /dev/video [0-63] entries */
+  for (int n = 0; n < 64; ++n) {
+    snprintf(device, sizeof(device), "/dev/video%d", n);
+    fd = open(device, O_RDONLY);
+    if (fd == -1)
+      continue;
+
+    // query device capabilities
+    struct v4l2_capability cap;
+    if (ioctl(fd, VIDIOC_QUERYCAP, &cap) == 0) {
+      // skip devices without video capture capability
+      if (!IsVideoCaptureDevice(&cap)) {
+        close(fd);
+        continue;
+      }
+
+      if (cap.bus_info[0] != 0) {
+        if (strncmp(reinterpret_cast<const char*>(cap.bus_info),
+                    deviceUniqueIdUTF8,
+                    strlen(deviceUniqueIdUTF8)) == 0) {  // match with device id
+          found = true;
+          break;  // fd matches with device unique id supplied
+        }
+      } else {  // match for device name
+        if (IsDeviceNameMatches(reinterpret_cast<const char*>(cap.card),
+                                deviceUniqueIdUTF8)) {
+          found = true;
+          break;
+        }
+      }
+    }
+    close(fd);  // close since this is not the matching device
+  }
+
+  if (!found) {
+    RTC_LOG(LS_INFO) << "no matching device found";
+    return -1;
+  }
+
+  // now fd will point to the matching device
+  // reset old capability list.
+  _captureCapabilities.clear();
+
+  int size = FillCapabilities(fd);
+  close(fd);
+
+  // Store the new used device name
+  _lastUsedDeviceNameLength = deviceUniqueIdUTF8Length;
+  _lastUsedDeviceName = reinterpret_cast<char*>(
+      realloc(_lastUsedDeviceName, _lastUsedDeviceNameLength + 1));
+  memcpy(_lastUsedDeviceName, deviceUniqueIdUTF8,
+         _lastUsedDeviceNameLength + 1);
+
+  RTC_LOG(LS_INFO) << "CreateCapabilityMap " << _captureCapabilities.size();
+
+  return size;
+}
+
+int32_t DeviceInfoV4l2::DisplayCaptureSettingsDialogBox(
+    const char* /*deviceUniqueIdUTF8*/,
+    const char* /*dialogTitleUTF8*/,
+    void* /*parentWindow*/,
+    uint32_t /*positionX*/,
+    uint32_t /*positionY*/) {
+  return -1;
+}
+
+bool DeviceInfoV4l2::IsDeviceNameMatches(const char* name,
+                                         const char* deviceUniqueIdUTF8) {
+  if (strncmp(deviceUniqueIdUTF8, name, strlen(name)) == 0)
+    return true;
+  return false;
+}
+
+bool DeviceInfoV4l2::IsVideoCaptureDevice(struct v4l2_capability* cap)
+{
+  if (cap->capabilities & V4L2_CAP_DEVICE_CAPS) {
+    return cap->device_caps & V4L2_CAP_VIDEO_CAPTURE;
+  } else {
+    return cap->capabilities & V4L2_CAP_VIDEO_CAPTURE;
+  }
+}
+
+int32_t DeviceInfoV4l2::FillCapabilities(int fd) {
+  // set image format
+  struct v4l2_format video_fmt;
+  memset(&video_fmt, 0, sizeof(struct v4l2_format));
+
+  video_fmt.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+  video_fmt.fmt.pix.sizeimage = 0;
+
+  unsigned int videoFormats[] = {
+      V4L2_PIX_FMT_MJPEG,  V4L2_PIX_FMT_JPEG,   V4L2_PIX_FMT_YUV420,
+      V4L2_PIX_FMT_YVU420, V4L2_PIX_FMT_YUYV,   V4L2_PIX_FMT_UYVY,
+      V4L2_PIX_FMT_NV12,   V4L2_PIX_FMT_BGR24,  V4L2_PIX_FMT_RGB24,
+      V4L2_PIX_FMT_RGB565, V4L2_PIX_FMT_ABGR32, V4L2_PIX_FMT_ARGB32,
+      V4L2_PIX_FMT_RGBA32, V4L2_PIX_FMT_BGR32,  V4L2_PIX_FMT_RGB32,
+  };
+  constexpr int totalFmts = sizeof(videoFormats) / sizeof(unsigned int);
+
+  int sizes = 13;
+  unsigned int size[][2] = {{128, 96},   {160, 120},  {176, 144},  {320, 240},
+                            {352, 288},  {640, 480},  {704, 576},  {800, 600},
+                            {960, 720},  {1280, 720}, {1024, 768}, {1440, 1080},
+                            {1920, 1080}};
+
+  for (int fmts = 0; fmts < totalFmts; fmts++) {
+    for (int i = 0; i < sizes; i++) {
+      video_fmt.fmt.pix.pixelformat = videoFormats[fmts];
+      video_fmt.fmt.pix.width = size[i][0];
+      video_fmt.fmt.pix.height = size[i][1];
+
+      if (ioctl(fd, VIDIOC_TRY_FMT, &video_fmt) >= 0) {
+        if ((video_fmt.fmt.pix.width == size[i][0]) &&
+            (video_fmt.fmt.pix.height == size[i][1])) {
+          VideoCaptureCapability cap;
+          cap.width = video_fmt.fmt.pix.width;
+          cap.height = video_fmt.fmt.pix.height;
+          if (videoFormats[fmts] == V4L2_PIX_FMT_YUYV) {
+            cap.videoType = VideoType::kYUY2;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_YUV420) {
+            cap.videoType = VideoType::kI420;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_YVU420) {
+            cap.videoType = VideoType::kYV12;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_MJPEG ||
+                     videoFormats[fmts] == V4L2_PIX_FMT_JPEG) {
+            cap.videoType = VideoType::kMJPEG;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_UYVY) {
+            cap.videoType = VideoType::kUYVY;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_NV12) {
+            cap.videoType = VideoType::kNV12;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_BGR24) {
+            // NB that for RGB formats, `VideoType` follows naming conventions
+            // of libyuv[1], where e.g. the format for FOURCC "ARGB" stores
+            // pixels in BGRA order in memory. V4L2[2] on the other hand names
+            // its formats based on the order of the RGB components as stored in
+            // memory. Applies to all RGB formats below.
+            // [1]https://chromium.googlesource.com/libyuv/libyuv/+/refs/heads/main/docs/formats.md#the-argb-fourcc
+            // [2]https://www.kernel.org/doc/html/v6.2/userspace-api/media/v4l/pixfmt-rgb.html#bits-per-component
+            cap.videoType = VideoType::kRGB24;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_RGB24) {
+            cap.videoType = VideoType::kBGR24;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_RGB565) {
+            cap.videoType = VideoType::kRGB565;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_ABGR32) {
+            cap.videoType = VideoType::kARGB;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_ARGB32) {
+            cap.videoType = VideoType::kBGRA;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_BGR32) {
+            cap.videoType = VideoType::kARGB;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_RGB32) {
+            cap.videoType = VideoType::kBGRA;
+          } else if (videoFormats[fmts] == V4L2_PIX_FMT_RGBA32) {
+            cap.videoType = VideoType::kABGR;
+          } else {
+            RTC_DCHECK_NOTREACHED();
+          }
+
+          // get fps of current camera mode
+          // V4l2 does not have a stable method of knowing so we just guess.
+          if (cap.width >= 800 && cap.videoType != VideoType::kMJPEG) {
+            cap.maxFPS = 15;
+          } else {
+            cap.maxFPS = 30;
+          }
+
+          _captureCapabilities.push_back(cap);
+          RTC_LOG(LS_VERBOSE) << "Camera capability, width:" << cap.width
+                              << " height:" << cap.height
+                              << " type:" << static_cast<int32_t>(cap.videoType)
+                              << " fps:" << cap.maxFPS;
+        }
+      }
+    }
+  }
+
+  RTC_LOG(LS_INFO) << "CreateCapabilityMap " << _captureCapabilities.size();
+  return _captureCapabilities.size();
+}
+
+}  // namespace videocapturemodule
+}  // namespace webrtc
+
+#ifdef ABGR32_OVERRIDE
+#undef ABGR32_OVERRIDE
+#undef V4L2_PIX_FMT_ABGR32
+#endif
+
+#ifdef ARGB32_OVERRIDE
+#undef ARGB32_OVERRIDE
+#undef V4L2_PIX_FMT_ARGB32
+#endif
+
+#ifdef RGBA32_OVERRIDE
+#undef RGBA32_OVERRIDE
+#undef V4L2_PIX_FMT_RGBA32
+#endif
diff --git a/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.h b/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.h
new file mode 100644
index 0000000000..0bec3eb765
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_LINUX_DEVICE_INFO_V4L2_H_
+#define MODULES_VIDEO_CAPTURE_LINUX_DEVICE_INFO_V4L2_H_
+
+#include <stdint.h>
+
+#include "modules/video_capture/device_info_impl.h"
+
+#include "rtc_base/platform_thread.h"
+#ifdef WEBRTC_LINUX
+#include <sys/inotify.h>
+#endif
+
+struct v4l2_capability;
+
+namespace webrtc {
+namespace videocapturemodule {
+class DeviceInfoV4l2 : public DeviceInfoImpl {
+ public:
+  DeviceInfoV4l2();
+  ~DeviceInfoV4l2() override;
+  uint32_t NumberOfDevices() override;
+  int32_t GetDeviceName(uint32_t deviceNumber,
+                        char* deviceNameUTF8,
+                        uint32_t deviceNameLength,
+                        char* deviceUniqueIdUTF8,
+                        uint32_t deviceUniqueIdUTF8Length,
+                        char* productUniqueIdUTF8 = 0,
+                        uint32_t productUniqueIdUTF8Length = 0,
+                        pid_t* pid=0) override;
+  /*
+   * Fills the membervariable _captureCapabilities with capabilites for the
+   * given device name.
+   */
+  int32_t CreateCapabilityMap(const char* deviceUniqueIdUTF8) override
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(_apiLock);
+  int32_t DisplayCaptureSettingsDialogBox(const char* /*deviceUniqueIdUTF8*/,
+                                          const char* /*dialogTitleUTF8*/,
+                                          void* /*parentWindow*/,
+                                          uint32_t /*positionX*/,
+                                          uint32_t /*positionY*/) override;
+  int32_t FillCapabilities(int fd) RTC_EXCLUSIVE_LOCKS_REQUIRED(_apiLock);
+  int32_t Init() override;
+
+ private:
+  bool IsDeviceNameMatches(const char* name, const char* deviceUniqueIdUTF8);
+  bool IsVideoCaptureDevice(struct v4l2_capability* cap);
+
+#ifdef WEBRTC_LINUX
+  void HandleEvent(inotify_event* event, int fd);
+  int EventCheck(int fd);
+  int HandleEvents(int fd);
+  int ProcessInotifyEvents();
+  rtc::PlatformThread _inotifyEventThread;
+  void InotifyProcess();
+  int _fd_v4l, _fd_dev, _wd_v4l, _wd_dev; /* accessed on InotifyEventThread thread */
+  std::atomic<bool> _isShutdown;
+#endif
+};
+}  // namespace videocapturemodule
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CAPTURE_LINUX_DEVICE_INFO_V4L2_H_
diff --git a/third_party/libwebrtc/modules/video_capture/linux/video_capture_linux.cc b/third_party/libwebrtc/modules/video_capture/linux/video_capture_linux.cc
new file mode 100644
index 0000000000..f3324a8e68
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/linux/video_capture_linux.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <errno.h>
+#include <fcntl.h>
+#include <poll.h>
+#include <stdio.h>
+#include <string.h>
+#include <sys/ioctl.h>
+#include <sys/mman.h>
+#include <sys/select.h>
+#include <time.h>
+#include <unistd.h>
+// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
+#include <linux/videodev2.h>
+#endif
+
+#include <new>
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "media/base/video_common.h"
+#include "modules/video_capture/linux/video_capture_v4l2.h"
+#include "modules/video_capture/video_capture.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+rtc::scoped_refptr<VideoCaptureModule> VideoCaptureImpl::Create(
+    const char* deviceUniqueId) {
+  auto implementation = rtc::make_ref_counted<VideoCaptureModuleV4L2>();
+
+  if (implementation->Init(deviceUniqueId) != 0)
+    return nullptr;
+
+  return implementation;
+}
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.cc b/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.cc
new file mode 100644
index 0000000000..2935cd027d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.cc
@@ -0,0 +1,489 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/linux/video_capture_v4l2.h"
+
+#include <errno.h>
+#include <fcntl.h>
+#include <poll.h>
+#include <stdio.h>
+#include <string.h>
+#include <sys/ioctl.h>
+#include <sys/mman.h>
+#include <sys/select.h>
+#include <time.h>
+#include <unistd.h>
+// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
+#include <linux/videodev2.h>
+#endif
+
+#include <new>
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "media/base/video_common.h"
+#include "modules/video_capture/video_capture.h"
+#include "rtc_base/logging.h"
+
+// These defines are here to support building on kernel 3.16 which some
+// downstream projects, e.g. Firefox, use.
+// TODO(apehrson): Remove them and their undefs when no longer needed.
+#ifndef V4L2_PIX_FMT_ABGR32
+#define ABGR32_OVERRIDE 1
+#define V4L2_PIX_FMT_ABGR32 v4l2_fourcc('A', 'R', '2', '4')
+#endif
+
+#ifndef V4L2_PIX_FMT_ARGB32
+#define ARGB32_OVERRIDE 1
+#define V4L2_PIX_FMT_ARGB32 v4l2_fourcc('B', 'A', '2', '4')
+#endif
+
+#ifndef V4L2_PIX_FMT_RGBA32
+#define RGBA32_OVERRIDE 1
+#define V4L2_PIX_FMT_RGBA32 v4l2_fourcc('A', 'B', '2', '4')
+#endif
+
+namespace webrtc {
+namespace videocapturemodule {
+VideoCaptureModuleV4L2::VideoCaptureModuleV4L2()
+    : VideoCaptureImpl(),
+      _deviceId(-1),
+      _deviceFd(-1),
+      _buffersAllocatedByDevice(-1),
+      _currentWidth(-1),
+      _currentHeight(-1),
+      _currentFrameRate(-1),
+      _captureStarted(false),
+      _captureVideoType(VideoType::kI420),
+      _pool(NULL) {}
+
+int32_t VideoCaptureModuleV4L2::Init(const char* deviceUniqueIdUTF8) {
+  int len = strlen((const char*)deviceUniqueIdUTF8);
+  _deviceUniqueId = new (std::nothrow) char[len + 1];
+  if (_deviceUniqueId) {
+    memcpy(_deviceUniqueId, deviceUniqueIdUTF8, len + 1);
+  }
+
+  int device_index;
+  if (sscanf(deviceUniqueIdUTF8,"fake_%d", &device_index) == 1)
+  {
+    _deviceId = device_index;
+    return 0;
+  }
+
+  int fd;
+  char device[32];
+  bool found = false;
+
+  /* detect /dev/video [0-63] entries */
+  int n;
+  for (n = 0; n < 64; n++) {
+    snprintf(device, sizeof(device), "/dev/video%d", n);
+    if ((fd = open(device, O_RDONLY)) != -1) {
+      // query device capabilities
+      struct v4l2_capability cap;
+      if (ioctl(fd, VIDIOC_QUERYCAP, &cap) == 0) {
+        if (cap.bus_info[0] != 0) {
+          if (strncmp((const char*)cap.bus_info,
+                      (const char*)deviceUniqueIdUTF8,
+                      strlen((const char*)deviceUniqueIdUTF8)) ==
+              0) {  // match with device id
+            close(fd);
+            found = true;
+            break;  // fd matches with device unique id supplied
+          }
+        }
+      }
+      close(fd);  // close since this is not the matching device
+    }
+  }
+  if (!found) {
+    RTC_LOG(LS_INFO) << "no matching device found";
+    return -1;
+  }
+  _deviceId = n;  // store the device id
+  return 0;
+}
+
+VideoCaptureModuleV4L2::~VideoCaptureModuleV4L2() {
+  StopCapture();
+  if (_deviceFd != -1)
+    close(_deviceFd);
+}
+
+int32_t VideoCaptureModuleV4L2::StartCapture(
+    const VideoCaptureCapability& capability) {
+  if (_captureStarted) {
+    if (capability.width == _currentWidth &&
+        capability.height == _currentHeight &&
+        _captureVideoType == capability.videoType) {
+      return 0;
+    } else {
+      StopCapture();
+    }
+  }
+
+  MutexLock lock(&capture_lock_);
+  // first open /dev/video device
+  char device[20];
+  snprintf(device, sizeof(device), "/dev/video%d", _deviceId);
+
+  if ((_deviceFd = open(device, O_RDWR | O_NONBLOCK, 0)) < 0) {
+    RTC_LOG(LS_INFO) << "error in opening " << device << " errono = " << errno;
+    return -1;
+  }
+
+  // Supported video formats in preferred order.
+  // If the requested resolution is larger than VGA, we prefer MJPEG. Go for
+  // I420 otherwise.
+  unsigned int hdFmts[] = {
+      V4L2_PIX_FMT_MJPEG,  V4L2_PIX_FMT_YUV420, V4L2_PIX_FMT_YVU420,
+      V4L2_PIX_FMT_YUYV,   V4L2_PIX_FMT_UYVY,   V4L2_PIX_FMT_NV12,
+      V4L2_PIX_FMT_ABGR32, V4L2_PIX_FMT_ARGB32, V4L2_PIX_FMT_RGBA32,
+      V4L2_PIX_FMT_BGR32,  V4L2_PIX_FMT_RGB32,  V4L2_PIX_FMT_BGR24,
+      V4L2_PIX_FMT_RGB24,  V4L2_PIX_FMT_RGB565, V4L2_PIX_FMT_JPEG,
+  };
+  unsigned int sdFmts[] = {
+      V4L2_PIX_FMT_YUV420, V4L2_PIX_FMT_YVU420, V4L2_PIX_FMT_YUYV,
+      V4L2_PIX_FMT_UYVY,   V4L2_PIX_FMT_NV12,   V4L2_PIX_FMT_ABGR32,
+      V4L2_PIX_FMT_ARGB32, V4L2_PIX_FMT_RGBA32, V4L2_PIX_FMT_BGR32,
+      V4L2_PIX_FMT_RGB32,  V4L2_PIX_FMT_BGR24,  V4L2_PIX_FMT_RGB24,
+      V4L2_PIX_FMT_RGB565, V4L2_PIX_FMT_MJPEG,  V4L2_PIX_FMT_JPEG,
+  };
+  const bool isHd = capability.width > 640 || capability.height > 480;
+  unsigned int* fmts = isHd ? hdFmts : sdFmts;
+  static_assert(sizeof(hdFmts) == sizeof(sdFmts));
+  constexpr int nFormats = sizeof(hdFmts) / sizeof(unsigned int);
+
+  // Enumerate image formats.
+  struct v4l2_fmtdesc fmt;
+  int fmtsIdx = nFormats;
+  memset(&fmt, 0, sizeof(fmt));
+  fmt.index = 0;
+  fmt.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+  RTC_LOG(LS_INFO) << "Video Capture enumerats supported image formats:";
+  while (ioctl(_deviceFd, VIDIOC_ENUM_FMT, &fmt) == 0) {
+    RTC_LOG(LS_INFO) << "  { pixelformat = "
+                     << cricket::GetFourccName(fmt.pixelformat)
+                     << ", description = '" << fmt.description << "' }";
+    // Match the preferred order.
+    for (int i = 0; i < nFormats; i++) {
+      if (fmt.pixelformat == fmts[i] && i < fmtsIdx)
+        fmtsIdx = i;
+    }
+    // Keep enumerating.
+    fmt.index++;
+  }
+
+  if (fmtsIdx == nFormats) {
+    RTC_LOG(LS_INFO) << "no supporting video formats found";
+    return -1;
+  } else {
+    RTC_LOG(LS_INFO) << "We prefer format "
+                     << cricket::GetFourccName(fmts[fmtsIdx]);
+  }
+
+  struct v4l2_format video_fmt;
+  memset(&video_fmt, 0, sizeof(struct v4l2_format));
+  video_fmt.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+  video_fmt.fmt.pix.sizeimage = 0;
+  video_fmt.fmt.pix.width = capability.width;
+  video_fmt.fmt.pix.height = capability.height;
+  video_fmt.fmt.pix.pixelformat = fmts[fmtsIdx];
+
+  if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_YUYV)
+    _captureVideoType = VideoType::kYUY2;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_YUV420)
+    _captureVideoType = VideoType::kI420;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_YVU420)
+    _captureVideoType = VideoType::kYV12;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_UYVY)
+    _captureVideoType = VideoType::kUYVY;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_NV12)
+    _captureVideoType = VideoType::kNV12;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_BGR24)
+    _captureVideoType = VideoType::kRGB24;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_RGB24)
+    _captureVideoType = VideoType::kBGR24;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_RGB565)
+    _captureVideoType = VideoType::kRGB565;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_ABGR32 ||
+           video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_BGR32)
+    _captureVideoType = VideoType::kARGB;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_ARGB32 ||
+           video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_RGB32)
+    _captureVideoType = VideoType::kBGRA;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_RGBA32)
+    _captureVideoType = VideoType::kABGR;
+  else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_MJPEG ||
+           video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_JPEG)
+    _captureVideoType = VideoType::kMJPEG;
+  else
+    RTC_DCHECK_NOTREACHED();
+
+  // set format and frame size now
+  if (ioctl(_deviceFd, VIDIOC_S_FMT, &video_fmt) < 0) {
+    RTC_LOG(LS_INFO) << "error in VIDIOC_S_FMT, errno = " << errno;
+    return -1;
+  }
+
+  // initialize current width and height
+  _currentWidth = video_fmt.fmt.pix.width;
+  _currentHeight = video_fmt.fmt.pix.height;
+
+  // Trying to set frame rate, before check driver capability.
+  bool driver_framerate_support = true;
+  struct v4l2_streamparm streamparms;
+  memset(&streamparms, 0, sizeof(streamparms));
+  streamparms.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+  if (ioctl(_deviceFd, VIDIOC_G_PARM, &streamparms) < 0) {
+    RTC_LOG(LS_INFO) << "error in VIDIOC_G_PARM errno = " << errno;
+    driver_framerate_support = false;
+    // continue
+  } else {
+    // check the capability flag is set to V4L2_CAP_TIMEPERFRAME.
+    if (streamparms.parm.capture.capability & V4L2_CAP_TIMEPERFRAME) {
+      // driver supports the feature. Set required framerate.
+      memset(&streamparms, 0, sizeof(streamparms));
+      streamparms.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+      streamparms.parm.capture.timeperframe.numerator = 1;
+      streamparms.parm.capture.timeperframe.denominator = capability.maxFPS;
+      if (ioctl(_deviceFd, VIDIOC_S_PARM, &streamparms) < 0) {
+        RTC_LOG(LS_INFO) << "Failed to set the framerate. errno=" << errno;
+        driver_framerate_support = false;
+      } else {
+        _currentFrameRate = capability.maxFPS;
+      }
+    }
+  }
+  // If driver doesn't support framerate control, need to hardcode.
+  // Hardcoding the value based on the frame size.
+  if (!driver_framerate_support) {
+    if (_currentWidth >= 800 && _captureVideoType != VideoType::kMJPEG) {
+      _currentFrameRate = 15;
+    } else {
+      _currentFrameRate = 30;
+    }
+  }
+
+  if (!AllocateVideoBuffers()) {
+    RTC_LOG(LS_INFO) << "failed to allocate video capture buffers";
+    return -1;
+  }
+
+  // start capture thread;
+  if (_captureThread.empty()) {
+    quit_ = false;
+    _captureThread = rtc::PlatformThread::SpawnJoinable(
+        [self = rtc::scoped_refptr(this)] {
+          while (self->CaptureProcess()) {
+          }
+        },
+        "CaptureThread",
+        rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kHigh));
+  }
+
+  // Needed to start UVC camera - from the uvcview application
+  enum v4l2_buf_type type;
+  type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+  if (ioctl(_deviceFd, VIDIOC_STREAMON, &type) == -1) {
+    RTC_LOG(LS_INFO) << "Failed to turn on stream";
+    return -1;
+  }
+
+  _captureStarted = true;
+  return 0;
+}
+
+int32_t VideoCaptureModuleV4L2::StopCapture() {
+  if (!_captureThread.empty()) {
+    {
+      MutexLock lock(&capture_lock_);
+      quit_ = true;
+    }
+    // Make sure the capture thread stops using the mutex.
+    _captureThread.Finalize();
+  }
+
+  MutexLock lock(&capture_lock_);
+  if (_captureStarted) {
+    _captureStarted = false;
+
+    DeAllocateVideoBuffers();
+    close(_deviceFd);
+    _deviceFd = -1;
+  }
+
+  return 0;
+}
+
+// critical section protected by the caller
+
+bool VideoCaptureModuleV4L2::AllocateVideoBuffers() {
+  struct v4l2_requestbuffers rbuffer;
+  memset(&rbuffer, 0, sizeof(v4l2_requestbuffers));
+
+  rbuffer.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+  rbuffer.memory = V4L2_MEMORY_MMAP;
+  rbuffer.count = kNoOfV4L2Bufffers;
+
+  if (ioctl(_deviceFd, VIDIOC_REQBUFS, &rbuffer) < 0) {
+    RTC_LOG(LS_INFO) << "Could not get buffers from device. errno = " << errno;
+    return false;
+  }
+
+  if (rbuffer.count > kNoOfV4L2Bufffers)
+    rbuffer.count = kNoOfV4L2Bufffers;
+
+  _buffersAllocatedByDevice = rbuffer.count;
+
+  // Map the buffers
+  _pool = new Buffer[rbuffer.count];
+
+  for (unsigned int i = 0; i < rbuffer.count; i++) {
+    struct v4l2_buffer buffer;
+    memset(&buffer, 0, sizeof(v4l2_buffer));
+    buffer.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+    buffer.memory = V4L2_MEMORY_MMAP;
+    buffer.index = i;
+
+    if (ioctl(_deviceFd, VIDIOC_QUERYBUF, &buffer) < 0) {
+      return false;
+    }
+
+    _pool[i].start = mmap(NULL, buffer.length, PROT_READ | PROT_WRITE,
+                          MAP_SHARED, _deviceFd, buffer.m.offset);
+
+    if (MAP_FAILED == _pool[i].start) {
+      for (unsigned int j = 0; j < i; j++)
+        munmap(_pool[j].start, _pool[j].length);
+      return false;
+    }
+
+    _pool[i].length = buffer.length;
+
+    if (ioctl(_deviceFd, VIDIOC_QBUF, &buffer) < 0) {
+      return false;
+    }
+  }
+  return true;
+}
+
+bool VideoCaptureModuleV4L2::DeAllocateVideoBuffers() {
+  // unmap buffers
+  for (int i = 0; i < _buffersAllocatedByDevice; i++)
+    munmap(_pool[i].start, _pool[i].length);
+
+  delete[] _pool;
+
+  // turn off stream
+  enum v4l2_buf_type type;
+  type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+  if (ioctl(_deviceFd, VIDIOC_STREAMOFF, &type) < 0) {
+    RTC_LOG(LS_INFO) << "VIDIOC_STREAMOFF error. errno: " << errno;
+  }
+
+  return true;
+}
+
+bool VideoCaptureModuleV4L2::CaptureStarted() {
+  return _captureStarted;
+}
+
+bool VideoCaptureModuleV4L2::CaptureProcess() {
+  int retVal = 0;
+  struct pollfd rSet;
+
+  rSet.fd = _deviceFd;
+  rSet.events = POLLIN;
+  rSet.revents = 0;
+
+  retVal = poll(&rSet, 1, 1000);
+
+  {
+    MutexLock lock(&capture_lock_);
+
+    if (quit_) {
+      return false;
+    }
+
+    if (retVal < 0 && errno != EINTR) {  // continue if interrupted
+      // poll failed
+      return false;
+    } else if (retVal == 0) {
+      // poll timed out
+      return true;
+    } else if (!(rSet.revents & POLLIN)) {
+      // not event on camera handle
+      return true;
+    }
+
+    if (_captureStarted) {
+      struct v4l2_buffer buf;
+      memset(&buf, 0, sizeof(struct v4l2_buffer));
+      buf.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
+      buf.memory = V4L2_MEMORY_MMAP;
+      // dequeue a buffer - repeat until dequeued properly!
+      while (ioctl(_deviceFd, VIDIOC_DQBUF, &buf) < 0) {
+        if (errno != EINTR) {
+          RTC_LOG(LS_INFO) << "could not sync on a buffer on device "
+                           << strerror(errno);
+          return true;
+        }
+      }
+      VideoCaptureCapability frameInfo;
+      frameInfo.width = _currentWidth;
+      frameInfo.height = _currentHeight;
+      frameInfo.videoType = _captureVideoType;
+
+      // convert to to I420 if needed
+      IncomingFrame(reinterpret_cast<uint8_t*>(_pool[buf.index].start),
+                    buf.bytesused, frameInfo);
+      // enqueue the buffer again
+      if (ioctl(_deviceFd, VIDIOC_QBUF, &buf) == -1) {
+        RTC_LOG(LS_INFO) << "Failed to enqueue capture buffer";
+      }
+    }
+  }
+  usleep(0);
+  return true;
+}
+
+int32_t VideoCaptureModuleV4L2::CaptureSettings(
+    VideoCaptureCapability& settings) {
+  settings.width = _currentWidth;
+  settings.height = _currentHeight;
+  settings.maxFPS = _currentFrameRate;
+  settings.videoType = _captureVideoType;
+
+  return 0;
+}
+}  // namespace videocapturemodule
+}  // namespace webrtc
+
+#ifdef ABGR32_OVERRIDE
+#undef ABGR32_OVERRIDE
+#undef V4L2_PIX_FMT_ABGR32
+#endif
+
+#ifdef ARGB32_OVERRIDE
+#undef ARGB32_OVERRIDE
+#undef V4L2_PIX_FMT_ARGB32
+#endif
+
+#ifdef RGBA32_OVERRIDE
+#undef RGBA32_OVERRIDE
+#undef V4L2_PIX_FMT_RGBA32
+#endif
diff --git a/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.h b/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.h
new file mode 100644
index 0000000000..65e89e2daa
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_LINUX_VIDEO_CAPTURE_V4L2_H_
+#define MODULES_VIDEO_CAPTURE_LINUX_VIDEO_CAPTURE_V4L2_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "modules/video_capture/video_capture_defines.h"
+#include "modules/video_capture/video_capture_impl.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+class VideoCaptureModuleV4L2 : public VideoCaptureImpl {
+ public:
+  VideoCaptureModuleV4L2();
+  ~VideoCaptureModuleV4L2() override;
+  int32_t Init(const char* deviceUniqueId);
+  int32_t StartCapture(const VideoCaptureCapability& capability) override;
+  int32_t StopCapture() override;
+  bool CaptureStarted() override;
+  int32_t CaptureSettings(VideoCaptureCapability& settings) override;
+
+ private:
+  enum { kNoOfV4L2Bufffers = 4 };
+
+  static void CaptureThread(void*);
+  bool CaptureProcess();
+  bool AllocateVideoBuffers();
+  bool DeAllocateVideoBuffers();
+
+  rtc::PlatformThread _captureThread;
+  Mutex capture_lock_;
+  bool quit_ RTC_GUARDED_BY(capture_lock_);
+  int32_t _deviceId;
+  int32_t _deviceFd;
+
+  int32_t _buffersAllocatedByDevice;
+  int32_t _currentWidth;
+  int32_t _currentHeight;
+  int32_t _currentFrameRate;
+  bool _captureStarted;
+  VideoType _captureVideoType;
+  struct Buffer {
+    void* start;
+    size_t length;
+  };
+  Buffer* _pool;
+};
+}  // namespace videocapturemodule
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CAPTURE_LINUX_VIDEO_CAPTURE_V4L2_H_
diff --git a/third_party/libwebrtc/modules/video_capture/raw_video_sink_interface.h b/third_party/libwebrtc/modules/video_capture/raw_video_sink_interface.h
new file mode 100644
index 0000000000..094e9e20bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/raw_video_sink_interface.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains interfaces used for creating the VideoCaptureModule
+// and DeviceInfo.
+
+#ifndef MODULES_VIDEO_CAPTURE_RAW_VIDEO_SINK_INTERFACE_H_
+#define MODULES_VIDEO_CAPTURE_RAW_VIDEO_SINK_INTERFACE_H_
+
+#include "modules/video_capture/video_capture_defines.h"
+
+namespace webrtc {
+
+class RawVideoSinkInterface {
+ public:
+  virtual ~RawVideoSinkInterface() = default;
+
+  virtual int32_t OnRawFrame(uint8_t* videoFrame,
+                             size_t videoFrameLength,
+                             const webrtc::VideoCaptureCapability& frameInfo,
+                             VideoRotation rotation,
+                             int64_t captureTime) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CAPTURE_RAW_VIDEO_SINK_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/video_capture/test/video_capture_unittest.cc b/third_party/libwebrtc/modules/video_capture/test/video_capture_unittest.cc
new file mode 100644
index 0000000000..c8af222b57
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/test/video_capture_unittest.cc
@@ -0,0 +1,376 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/video_capture.h"
+
+#include <stdio.h>
+
+#include <map>
+#include <memory>
+#include <sstream>
+
+#include "absl/memory/memory.h"
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "modules/video_capture/video_capture_factory.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/frame_utils.h"
+#include "test/gtest.h"
+
+using webrtc::SleepMs;
+using webrtc::VideoCaptureCapability;
+using webrtc::VideoCaptureFactory;
+using webrtc::VideoCaptureModule;
+
+#define WAIT_(ex, timeout, res)                           \
+  do {                                                    \
+    res = (ex);                                           \
+    int64_t start = rtc::TimeMillis();                    \
+    while (!res && rtc::TimeMillis() < start + timeout) { \
+      SleepMs(5);                                         \
+      res = (ex);                                         \
+    }                                                     \
+  } while (0)
+
+#define EXPECT_TRUE_WAIT(ex, timeout) \
+  do {                                \
+    bool res;                         \
+    WAIT_(ex, timeout, res);          \
+    if (!res)                         \
+      EXPECT_TRUE(ex);                \
+  } while (0)
+
+static const int kTimeOut = 5000;
+#ifdef WEBRTC_MAC
+static const int kTestHeight = 288;
+static const int kTestWidth = 352;
+static const int kTestFramerate = 30;
+#endif
+
+class TestVideoCaptureCallback
+    : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  TestVideoCaptureCallback()
+      : last_render_time_ms_(0),
+        incoming_frames_(0),
+        timing_warnings_(0),
+        rotate_frame_(webrtc::kVideoRotation_0) {}
+
+  ~TestVideoCaptureCallback() override {
+    if (timing_warnings_ > 0)
+      printf("No of timing warnings %d\n", timing_warnings_);
+  }
+
+  void OnFrame(const webrtc::VideoFrame& videoFrame) override {
+    webrtc::MutexLock lock(&capture_lock_);
+    int height = videoFrame.height();
+    int width = videoFrame.width();
+#if defined(WEBRTC_ANDROID) && WEBRTC_ANDROID
+    // Android camera frames may be rotated depending on test device
+    // orientation.
+    EXPECT_TRUE(height == capability_.height || height == capability_.width);
+    EXPECT_TRUE(width == capability_.width || width == capability_.height);
+#else
+    EXPECT_EQ(height, capability_.height);
+    EXPECT_EQ(width, capability_.width);
+    EXPECT_EQ(rotate_frame_, videoFrame.rotation());
+#endif
+    // RenderTimstamp should be the time now.
+    EXPECT_TRUE(videoFrame.render_time_ms() >= rtc::TimeMillis() - 30 &&
+                videoFrame.render_time_ms() <= rtc::TimeMillis());
+
+    if ((videoFrame.render_time_ms() >
+             last_render_time_ms_ + (1000 * 1.1) / capability_.maxFPS &&
+         last_render_time_ms_ > 0) ||
+        (videoFrame.render_time_ms() <
+             last_render_time_ms_ + (1000 * 0.9) / capability_.maxFPS &&
+         last_render_time_ms_ > 0)) {
+      timing_warnings_++;
+    }
+
+    incoming_frames_++;
+    last_render_time_ms_ = videoFrame.render_time_ms();
+    last_frame_ = videoFrame.video_frame_buffer();
+  }
+
+  void SetExpectedCapability(VideoCaptureCapability capability) {
+    webrtc::MutexLock lock(&capture_lock_);
+    capability_ = capability;
+    incoming_frames_ = 0;
+    last_render_time_ms_ = 0;
+  }
+  int incoming_frames() {
+    webrtc::MutexLock lock(&capture_lock_);
+    return incoming_frames_;
+  }
+
+  int timing_warnings() {
+    webrtc::MutexLock lock(&capture_lock_);
+    return timing_warnings_;
+  }
+  VideoCaptureCapability capability() {
+    webrtc::MutexLock lock(&capture_lock_);
+    return capability_;
+  }
+
+  bool CompareLastFrame(const webrtc::VideoFrame& frame) {
+    webrtc::MutexLock lock(&capture_lock_);
+    return webrtc::test::FrameBufsEqual(last_frame_,
+                                        frame.video_frame_buffer());
+  }
+
+  void SetExpectedCaptureRotation(webrtc::VideoRotation rotation) {
+    webrtc::MutexLock lock(&capture_lock_);
+    rotate_frame_ = rotation;
+  }
+
+ private:
+  webrtc::Mutex capture_lock_;
+  VideoCaptureCapability capability_;
+  int64_t last_render_time_ms_;
+  int incoming_frames_;
+  int timing_warnings_;
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> last_frame_;
+  webrtc::VideoRotation rotate_frame_;
+};
+
+class VideoCaptureTest : public ::testing::Test {
+ public:
+  VideoCaptureTest() : number_of_devices_(0) {}
+
+  void SetUp() override {
+    device_info_.reset(VideoCaptureFactory::CreateDeviceInfo());
+    RTC_DCHECK(device_info_.get());
+    number_of_devices_ = device_info_->NumberOfDevices();
+    ASSERT_GT(number_of_devices_, 0u);
+  }
+
+  rtc::scoped_refptr<VideoCaptureModule> OpenVideoCaptureDevice(
+      unsigned int device,
+      rtc::VideoSinkInterface<webrtc::VideoFrame>* callback) {
+    char device_name[256];
+    char unique_name[256];
+
+    EXPECT_EQ(0, device_info_->GetDeviceName(device, device_name, 256,
+                                             unique_name, 256));
+
+    rtc::scoped_refptr<VideoCaptureModule> module(
+        VideoCaptureFactory::Create(unique_name));
+    if (module.get() == NULL)
+      return nullptr;
+
+    EXPECT_FALSE(module->CaptureStarted());
+
+    module->RegisterCaptureDataCallback(callback);
+    return module;
+  }
+
+  void StartCapture(VideoCaptureModule* capture_module,
+                    VideoCaptureCapability capability) {
+    ASSERT_EQ(0, capture_module->StartCapture(capability));
+    EXPECT_TRUE(capture_module->CaptureStarted());
+
+    VideoCaptureCapability resulting_capability;
+    EXPECT_EQ(0, capture_module->CaptureSettings(resulting_capability));
+    EXPECT_EQ(capability.width, resulting_capability.width);
+    EXPECT_EQ(capability.height, resulting_capability.height);
+  }
+
+  std::unique_ptr<VideoCaptureModule::DeviceInfo> device_info_;
+  unsigned int number_of_devices_;
+};
+
+#ifdef WEBRTC_MAC
+// Currently fails on Mac 64-bit, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5406
+#define MAYBE_CreateDelete DISABLED_CreateDelete
+#else
+#define MAYBE_CreateDelete CreateDelete
+#endif
+TEST_F(VideoCaptureTest, MAYBE_CreateDelete) {
+  for (int i = 0; i < 5; ++i) {
+    int64_t start_time = rtc::TimeMillis();
+    TestVideoCaptureCallback capture_observer;
+    rtc::scoped_refptr<VideoCaptureModule> module(
+        OpenVideoCaptureDevice(0, &capture_observer));
+    ASSERT_TRUE(module.get() != NULL);
+
+    VideoCaptureCapability capability;
+#ifndef WEBRTC_MAC
+    device_info_->GetCapability(module->CurrentDeviceName(), 0, capability);
+#else
+    capability.width = kTestWidth;
+    capability.height = kTestHeight;
+    capability.maxFPS = kTestFramerate;
+    capability.videoType = webrtc::VideoType::kUnknown;
+#endif
+    capture_observer.SetExpectedCapability(capability);
+    ASSERT_NO_FATAL_FAILURE(StartCapture(module.get(), capability));
+
+    // Less than 4s to start the camera.
+    EXPECT_LE(rtc::TimeMillis() - start_time, 4000);
+
+    // Make sure 5 frames are captured.
+    EXPECT_TRUE_WAIT(capture_observer.incoming_frames() >= 5, kTimeOut);
+
+    int64_t stop_time = rtc::TimeMillis();
+    EXPECT_EQ(0, module->StopCapture());
+    EXPECT_FALSE(module->CaptureStarted());
+
+    // Less than 3s to stop the camera.
+    EXPECT_LE(rtc::TimeMillis() - stop_time, 3000);
+  }
+}
+
+#ifdef WEBRTC_MAC
+// Currently fails on Mac 64-bit, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5406
+#define MAYBE_Capabilities DISABLED_Capabilities
+#else
+#define MAYBE_Capabilities Capabilities
+#endif
+TEST_F(VideoCaptureTest, MAYBE_Capabilities) {
+  TestVideoCaptureCallback capture_observer;
+
+  rtc::scoped_refptr<VideoCaptureModule> module(
+      OpenVideoCaptureDevice(0, &capture_observer));
+  ASSERT_TRUE(module.get() != NULL);
+
+  int number_of_capabilities =
+      device_info_->NumberOfCapabilities(module->CurrentDeviceName());
+  EXPECT_GT(number_of_capabilities, 0);
+  // Key is <width>x<height>, value is vector of maxFPS values at that
+  // resolution.
+  typedef std::map<std::string, std::vector<int> > FrameRatesByResolution;
+  FrameRatesByResolution frame_rates_by_resolution;
+  for (int i = 0; i < number_of_capabilities; ++i) {
+    VideoCaptureCapability capability;
+    EXPECT_EQ(0, device_info_->GetCapability(module->CurrentDeviceName(), i,
+                                             capability));
+    std::ostringstream resolutionStream;
+    resolutionStream << capability.width << "x" << capability.height;
+    resolutionStream.flush();
+    std::string resolution = resolutionStream.str();
+    frame_rates_by_resolution[resolution].push_back(capability.maxFPS);
+
+    // Since Android presents so many resolution/FPS combinations and the test
+    // runner imposes a timeout, we only actually start the capture and test
+    // that a frame was captured for 2 frame-rates at each resolution.
+    if (frame_rates_by_resolution[resolution].size() > 2)
+      continue;
+
+    capture_observer.SetExpectedCapability(capability);
+    ASSERT_NO_FATAL_FAILURE(StartCapture(module.get(), capability));
+    // Make sure at least one frame is captured.
+    EXPECT_TRUE_WAIT(capture_observer.incoming_frames() >= 1, kTimeOut);
+
+    EXPECT_EQ(0, module->StopCapture());
+  }
+
+#if defined(WEBRTC_ANDROID) && WEBRTC_ANDROID
+  // There's no reason for this to _necessarily_ be true, but in practice all
+  // Android devices this test runs on in fact do support multiple capture
+  // resolutions and multiple frame-rates per captured resolution, so we assert
+  // this fact here as a regression-test against the time that we only noticed a
+  // single frame-rate per resolution (bug 2974).  If this test starts being run
+  // on devices for which this is untrue (e.g. Nexus4) then the following should
+  // probably be wrapped in a base::android::BuildInfo::model()/device() check.
+  EXPECT_GT(frame_rates_by_resolution.size(), 1U);
+  for (FrameRatesByResolution::const_iterator it =
+           frame_rates_by_resolution.begin();
+       it != frame_rates_by_resolution.end(); ++it) {
+    EXPECT_GT(it->second.size(), 1U) << it->first;
+  }
+#endif  // WEBRTC_ANDROID
+}
+
+// NOTE: flaky, crashes sometimes.
+// http://code.google.com/p/webrtc/issues/detail?id=777
+TEST_F(VideoCaptureTest, DISABLED_TestTwoCameras) {
+  if (number_of_devices_ < 2) {
+    printf("There are not two cameras available. Aborting test. \n");
+    return;
+  }
+
+  TestVideoCaptureCallback capture_observer1;
+  rtc::scoped_refptr<VideoCaptureModule> module1(
+      OpenVideoCaptureDevice(0, &capture_observer1));
+  ASSERT_TRUE(module1.get() != NULL);
+  VideoCaptureCapability capability1;
+#ifndef WEBRTC_MAC
+  device_info_->GetCapability(module1->CurrentDeviceName(), 0, capability1);
+#else
+  capability1.width = kTestWidth;
+  capability1.height = kTestHeight;
+  capability1.maxFPS = kTestFramerate;
+  capability1.videoType = webrtc::VideoType::kUnknown;
+#endif
+  capture_observer1.SetExpectedCapability(capability1);
+
+  TestVideoCaptureCallback capture_observer2;
+  rtc::scoped_refptr<VideoCaptureModule> module2(
+      OpenVideoCaptureDevice(1, &capture_observer2));
+  ASSERT_TRUE(module1.get() != NULL);
+
+  VideoCaptureCapability capability2;
+#ifndef WEBRTC_MAC
+  device_info_->GetCapability(module2->CurrentDeviceName(), 0, capability2);
+#else
+  capability2.width = kTestWidth;
+  capability2.height = kTestHeight;
+  capability2.maxFPS = kTestFramerate;
+  capability2.videoType = webrtc::VideoType::kUnknown;
+#endif
+  capture_observer2.SetExpectedCapability(capability2);
+
+  ASSERT_NO_FATAL_FAILURE(StartCapture(module1.get(), capability1));
+  ASSERT_NO_FATAL_FAILURE(StartCapture(module2.get(), capability2));
+  EXPECT_TRUE_WAIT(capture_observer1.incoming_frames() >= 5, kTimeOut);
+  EXPECT_TRUE_WAIT(capture_observer2.incoming_frames() >= 5, kTimeOut);
+  EXPECT_EQ(0, module2->StopCapture());
+  EXPECT_EQ(0, module1->StopCapture());
+}
+
+#ifdef WEBRTC_MAC
+// No VideoCaptureImpl on Mac.
+#define MAYBE_ConcurrentAccess DISABLED_ConcurrentAccess
+#else
+#define MAYBE_ConcurrentAccess ConcurrentAccess
+#endif
+TEST_F(VideoCaptureTest, MAYBE_ConcurrentAccess) {
+  TestVideoCaptureCallback capture_observer1;
+  rtc::scoped_refptr<VideoCaptureModule> module1(
+      OpenVideoCaptureDevice(0, &capture_observer1));
+  ASSERT_TRUE(module1.get() != NULL);
+  VideoCaptureCapability capability;
+  device_info_->GetCapability(module1->CurrentDeviceName(), 0, capability);
+  capture_observer1.SetExpectedCapability(capability);
+
+  TestVideoCaptureCallback capture_observer2;
+  rtc::scoped_refptr<VideoCaptureModule> module2(
+      OpenVideoCaptureDevice(0, &capture_observer2));
+  ASSERT_TRUE(module2.get() != NULL);
+  capture_observer2.SetExpectedCapability(capability);
+
+  // Starting module1 should work.
+  ASSERT_NO_FATAL_FAILURE(StartCapture(module1.get(), capability));
+  EXPECT_TRUE_WAIT(capture_observer1.incoming_frames() >= 5, kTimeOut);
+
+  // When module1 is stopped, starting module2 for the same device should work.
+  EXPECT_EQ(0, module1->StopCapture());
+  ASSERT_NO_FATAL_FAILURE(StartCapture(module2.get(), capability));
+  EXPECT_TRUE_WAIT(capture_observer2.incoming_frames() >= 5, kTimeOut);
+
+  EXPECT_EQ(0, module2->StopCapture());
+}
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture.h b/third_party/libwebrtc/modules/video_capture/video_capture.h
new file mode 100644
index 0000000000..7e181c538e
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture.h
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
+#define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
+
+#include "api/video/video_rotation.h"
+#include "api/video/video_sink_interface.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
+#include "modules/video_capture/raw_video_sink_interface.h"
+#include "modules/video_capture/video_capture_defines.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include <set>
+
+#if defined(ANDROID)
+#include <jni.h>
+#endif
+
+namespace webrtc {
+
+class VideoInputFeedBack
+{
+public:
+    virtual void OnDeviceChange() = 0;
+protected:
+    virtual ~VideoInputFeedBack(){}
+};
+
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+  int32_t SetCaptureAndroidVM(JavaVM* javaVM);
+#endif
+
+class VideoCaptureModule : public rtc::RefCountInterface {
+ public:
+  // Interface for receiving information about available camera devices.
+  class DeviceInfo {
+   public:
+    virtual uint32_t NumberOfDevices() = 0;
+    virtual int32_t Refresh() = 0;
+    virtual void DeviceChange() {
+      MutexLock lock(&_inputCallbacksMutex);
+      for (auto inputCallBack : _inputCallBacks) {
+        inputCallBack->OnDeviceChange();
+      }
+    }
+    virtual void RegisterVideoInputFeedBack(VideoInputFeedBack* callBack) {
+      MutexLock lock(&_inputCallbacksMutex);
+      _inputCallBacks.insert(callBack);
+    }
+
+    virtual void DeRegisterVideoInputFeedBack(VideoInputFeedBack* callBack) {
+      MutexLock lock(&_inputCallbacksMutex);
+      auto it = _inputCallBacks.find(callBack);
+      if (it != _inputCallBacks.end()) {
+        _inputCallBacks.erase(it);
+      }
+    }
+
+    // Returns the available capture devices.
+    // deviceNumber   - Index of capture device.
+    // deviceNameUTF8 - Friendly name of the capture device.
+    // deviceUniqueIdUTF8 - Unique name of the capture device if it exist.
+    //                      Otherwise same as deviceNameUTF8.
+    // productUniqueIdUTF8 - Unique product id if it exist.
+    //                       Null terminated otherwise.
+    virtual int32_t GetDeviceName(uint32_t deviceNumber,
+                                  char* deviceNameUTF8,
+                                  uint32_t deviceNameLength,
+                                  char* deviceUniqueIdUTF8,
+                                  uint32_t deviceUniqueIdUTF8Length,
+                                  char* productUniqueIdUTF8 = 0,
+                                  uint32_t productUniqueIdUTF8Length = 0,
+                                  pid_t* pid = 0) = 0;
+
+    // Returns the number of capabilities this device.
+    virtual int32_t NumberOfCapabilities(const char* deviceUniqueIdUTF8) = 0;
+
+    // Gets the capabilities of the named device.
+    virtual int32_t GetCapability(const char* deviceUniqueIdUTF8,
+                                  uint32_t deviceCapabilityNumber,
+                                  VideoCaptureCapability& capability) = 0;
+
+    // Gets clockwise angle the captured frames should be rotated in order
+    // to be displayed correctly on a normally rotated display.
+    virtual int32_t GetOrientation(const char* deviceUniqueIdUTF8,
+                                   VideoRotation& orientation) = 0;
+
+    // Gets the capability that best matches the requested width, height and
+    // frame rate.
+    // Returns the deviceCapabilityNumber on success.
+    virtual int32_t GetBestMatchedCapability(
+        const char* deviceUniqueIdUTF8,
+        const VideoCaptureCapability& requested,
+        VideoCaptureCapability& resulting) = 0;
+
+    // Display OS /capture device specific settings dialog
+    virtual int32_t DisplayCaptureSettingsDialogBox(
+        const char* deviceUniqueIdUTF8,
+        const char* dialogTitleUTF8,
+        void* parentWindow,
+        uint32_t positionX,
+        uint32_t positionY) = 0;
+
+    virtual ~DeviceInfo() {}
+   private:
+    Mutex _inputCallbacksMutex;
+    std::set<VideoInputFeedBack*> _inputCallBacks RTC_GUARDED_BY(_inputCallbacksMutex);
+  };
+
+  //   Register capture data callback
+  virtual void RegisterCaptureDataCallback(
+      rtc::VideoSinkInterface<VideoFrame>* dataCallback) = 0;
+  virtual void RegisterCaptureDataCallback(
+      RawVideoSinkInterface* dataCallback) = 0;
+
+  //  Remove capture data callback
+  virtual void DeRegisterCaptureDataCallback(
+      rtc::VideoSinkInterface<VideoFrame> *dataCallback) = 0;
+
+  // Start capture device
+  virtual int32_t StartCapture(const VideoCaptureCapability& capability) = 0;
+
+  virtual int32_t StopCaptureIfAllClientsClose() = 0;
+
+  virtual bool FocusOnSelectedSource() { return false; }
+
+  virtual int32_t StopCapture() = 0;
+
+  // Returns the name of the device used by this module.
+  virtual const char* CurrentDeviceName() const = 0;
+
+  // Returns true if the capture device is running
+  virtual bool CaptureStarted() = 0;
+
+  // Gets the current configuration.
+  virtual int32_t CaptureSettings(VideoCaptureCapability& settings) = 0;
+
+  // Set the rotation of the captured frames.
+  // If the rotation is set to the same as returned by
+  // DeviceInfo::GetOrientation the captured frames are
+  // displayed correctly if rendered.
+  virtual int32_t SetCaptureRotation(VideoRotation rotation) = 0;
+
+  // Tells the capture module whether to apply the pending rotation. By default,
+  // the rotation is applied and the generated frame is up right. When set to
+  // false, generated frames will carry the rotation information from
+  // SetCaptureRotation. Return value indicates whether this operation succeeds.
+  virtual bool SetApplyRotation(bool enable) = 0;
+
+  // Return whether the rotation is applied or left pending.
+  virtual bool GetApplyRotation() = 0;
+
+  // Mozilla: TrackingId setter for use in profiler markers.
+  virtual void SetTrackingId(uint32_t aTrackingIdProcId) {}
+
+ protected:
+  ~VideoCaptureModule() override {}
+};
+
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_config.h b/third_party/libwebrtc/modules/video_capture/video_capture_config.h
new file mode 100644
index 0000000000..f285b9eeb1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture_config.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
+#define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
+
+namespace webrtc {
+namespace videocapturemodule {
+enum { kDefaultWidth = 640 };     // Start width
+enum { kDefaultHeight = 480 };    // Start heigt
+enum { kDefaultFrameRate = 30 };  // Start frame rate
+
+enum { kMaxFrameRate = 60 };  // Max allowed frame rate of the start image
+
+enum { kDefaultCaptureDelay = 120 };
+enum {
+  kMaxCaptureDelay = 270
+};  // Max capture delay allowed in the precompiled capture delay values.
+
+enum { kFrameRateCallbackInterval = 1000 };
+enum { kFrameRateCountHistorySize = 90 };
+enum { kFrameRateHistoryWindowMs = 2000 };
+}  // namespace videocapturemodule
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_defines.h b/third_party/libwebrtc/modules/video_capture/video_capture_defines.h
new file mode 100644
index 0000000000..63534600a9
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture_defines.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_
+#define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_
+
+#include "api/video/video_frame.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+
+namespace webrtc {
+
+enum {
+  kVideoCaptureUniqueNameLength = 1024
+};  // Max unique capture device name lenght
+enum { kVideoCaptureDeviceNameLength = 256 };  // Max capture device name lenght
+enum { kVideoCaptureProductIdLength = 128 };   // Max product id length
+
+struct VideoCaptureCapability {
+  int32_t width;
+  int32_t height;
+  int32_t maxFPS;
+  VideoType videoType;
+  bool interlaced;
+
+  VideoCaptureCapability() {
+    width = 0;
+    height = 0;
+    maxFPS = 0;
+    videoType = VideoType::kUnknown;
+    interlaced = false;
+  }
+  bool operator!=(const VideoCaptureCapability& other) const {
+    if (width != other.width)
+      return true;
+    if (height != other.height)
+      return true;
+    if (maxFPS != other.maxFPS)
+      return true;
+    if (videoType != other.videoType)
+      return true;
+    if (interlaced != other.interlaced)
+      return true;
+    return false;
+  }
+  bool operator==(const VideoCaptureCapability& other) const {
+    return !operator!=(other);
+  }
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_DEFINES_H_
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_factory.cc b/third_party/libwebrtc/modules/video_capture/video_capture_factory.cc
new file mode 100644
index 0000000000..e4a46902e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture_factory.cc
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/video_capture_factory.h"
+
+#include "modules/video_capture/video_capture_impl.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<VideoCaptureModule> VideoCaptureFactory::Create(
+    const char* deviceUniqueIdUTF8) {
+  return videocapturemodule::VideoCaptureImpl::Create(deviceUniqueIdUTF8);
+}
+
+VideoCaptureModule::DeviceInfo* VideoCaptureFactory::CreateDeviceInfo() {
+  return videocapturemodule::VideoCaptureImpl::CreateDeviceInfo();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_factory.h b/third_party/libwebrtc/modules/video_capture/video_capture_factory.h
new file mode 100644
index 0000000000..1fe47d9fab
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture_factory.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains interfaces used for creating the VideoCaptureModule
+// and DeviceInfo.
+
+#ifndef MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_FACTORY_H_
+#define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_FACTORY_H_
+
+#include "api/scoped_refptr.h"
+#include "modules/video_capture/video_capture.h"
+#include "modules/video_capture/video_capture_defines.h"
+
+namespace webrtc {
+
+class VideoCaptureFactory {
+ public:
+  // Create a video capture module object
+  // id - unique identifier of this video capture module object.
+  // deviceUniqueIdUTF8 - name of the device.
+  //                      Available names can be found by using GetDeviceName
+  static rtc::scoped_refptr<VideoCaptureModule> Create(
+      const char* deviceUniqueIdUTF8);
+
+  static VideoCaptureModule::DeviceInfo* CreateDeviceInfo();
+
+ private:
+  ~VideoCaptureFactory();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_factory_null.cc b/third_party/libwebrtc/modules/video_capture/video_capture_factory_null.cc
new file mode 100644
index 0000000000..7808d19851
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture_factory_null.cc
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/video_capture_impl.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+
+// static
+VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo() {
+  return nullptr;
+}
+
+rtc::scoped_refptr<VideoCaptureModule> VideoCaptureImpl::Create(
+    const char* device_id) {
+  return nullptr;
+}
+
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_impl.cc b/third_party/libwebrtc/modules/video_capture/video_capture_impl.cc
new file mode 100644
index 0000000000..9d9a1471e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture_impl.cc
@@ -0,0 +1,334 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/video_capture_impl.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "modules/video_capture/video_capture_config.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "third_party/libyuv/include/libyuv.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+
+const char* VideoCaptureImpl::CurrentDeviceName() const {
+  return _deviceUniqueId;
+}
+
+// static
+int32_t VideoCaptureImpl::RotationFromDegrees(int degrees,
+                                              VideoRotation* rotation) {
+  switch (degrees) {
+    case 0:
+      *rotation = kVideoRotation_0;
+      return 0;
+    case 90:
+      *rotation = kVideoRotation_90;
+      return 0;
+    case 180:
+      *rotation = kVideoRotation_180;
+      return 0;
+    case 270:
+      *rotation = kVideoRotation_270;
+      return 0;
+    default:
+      return -1;
+      ;
+  }
+}
+
+// static
+int32_t VideoCaptureImpl::RotationInDegrees(VideoRotation rotation,
+                                            int* degrees) {
+  switch (rotation) {
+    case kVideoRotation_0:
+      *degrees = 0;
+      return 0;
+    case kVideoRotation_90:
+      *degrees = 90;
+      return 0;
+    case kVideoRotation_180:
+      *degrees = 180;
+      return 0;
+    case kVideoRotation_270:
+      *degrees = 270;
+      return 0;
+  }
+  return -1;
+}
+
+VideoCaptureImpl::VideoCaptureImpl()
+    : _deviceUniqueId(NULL),
+      _requestedCapability(),
+      _lastProcessTimeNanos(rtc::TimeNanos()),
+      _lastFrameRateCallbackTimeNanos(rtc::TimeNanos()),
+      _rawDataCallBack(NULL),
+      _lastProcessFrameTimeNanos(rtc::TimeNanos()),
+      _rotateFrame(kVideoRotation_0),
+      apply_rotation_(false) {
+  _requestedCapability.width = kDefaultWidth;
+  _requestedCapability.height = kDefaultHeight;
+  _requestedCapability.maxFPS = 30;
+  _requestedCapability.videoType = VideoType::kI420;
+  memset(_incomingFrameTimesNanos, 0, sizeof(_incomingFrameTimesNanos));
+}
+
+VideoCaptureImpl::~VideoCaptureImpl() {
+  if (_deviceUniqueId)
+    delete[] _deviceUniqueId;
+}
+
+void VideoCaptureImpl::RegisterCaptureDataCallback(
+    rtc::VideoSinkInterface<VideoFrame>* dataCallBack) {
+  MutexLock lock(&api_lock_);
+  RTC_DCHECK(!_rawDataCallBack);
+  _dataCallBacks.insert(dataCallBack);
+}
+
+void VideoCaptureImpl::RegisterCaptureDataCallback(
+    RawVideoSinkInterface* dataCallBack) {
+  MutexLock lock(&api_lock_);
+  RTC_DCHECK(_dataCallBacks.empty());
+  _rawDataCallBack = dataCallBack;
+}
+
+void VideoCaptureImpl::DeRegisterCaptureDataCallback(
+    rtc::VideoSinkInterface<VideoFrame>* dataCallBack) {
+  MutexLock lock(&api_lock_);
+  auto it = _dataCallBacks.find(dataCallBack);
+  if (it != _dataCallBacks.end()) {
+    _dataCallBacks.erase(it);
+  }
+  _rawDataCallBack = NULL;
+}
+
+int32_t VideoCaptureImpl::StopCaptureIfAllClientsClose() {
+  if (_dataCallBacks.empty()) {
+    return StopCapture();
+  } else {
+    return 0;
+  }
+}
+
+int32_t VideoCaptureImpl::DeliverCapturedFrame(VideoFrame& captureFrame) {
+  UpdateFrameCount();  // frame count used for local frame rate callback.
+
+  for (auto dataCallBack : _dataCallBacks) {
+    dataCallBack->OnFrame(captureFrame);
+  }
+
+  return 0;
+}
+
+void VideoCaptureImpl::DeliverRawFrame(uint8_t* videoFrame,
+                                       size_t videoFrameLength,
+                                       const VideoCaptureCapability& frameInfo,
+                                       int64_t captureTime) {
+  UpdateFrameCount();
+  _rawDataCallBack->OnRawFrame(videoFrame, videoFrameLength, frameInfo,
+                               _rotateFrame, captureTime);
+}
+
+int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
+                                        size_t videoFrameLength,
+                                        const VideoCaptureCapability& frameInfo,
+                                        int64_t captureTime /*=0*/) {
+  MutexLock lock(&api_lock_);
+
+  const int32_t width = frameInfo.width;
+  const int32_t height = frameInfo.height;
+
+  TRACE_EVENT1("webrtc", "VC::IncomingFrame", "capture_time", captureTime);
+
+  if (_rawDataCallBack) {
+    DeliverRawFrame(videoFrame, videoFrameLength, frameInfo, captureTime);
+    return 0;
+  }
+
+  // Not encoded, convert to I420.
+  if (frameInfo.videoType != VideoType::kMJPEG) {
+    // Allow buffers larger than expected. On linux gstreamer allocates buffers
+    // page-aligned and v4l2loopback passes us the buffer size verbatim which
+    // for most cases is larger than expected.
+    // See https://github.com/umlaeute/v4l2loopback/issues/190.
+    if (auto size = CalcBufferSize(frameInfo.videoType, width, abs(height));
+        videoFrameLength < size) {
+      RTC_LOG(LS_ERROR) << "Wrong incoming frame length. Expected " << size
+                        << ", Got " << videoFrameLength << ".";
+      return -1;
+    }
+  }
+
+  int target_width = width;
+  int target_height = abs(height);
+
+  // SetApplyRotation doesn't take any lock. Make a local copy here.
+  bool apply_rotation = apply_rotation_;
+
+  if (apply_rotation) {
+    // Rotating resolution when for 90/270 degree rotations.
+    if (_rotateFrame == kVideoRotation_90 ||
+        _rotateFrame == kVideoRotation_270) {
+      target_width = abs(height);
+      target_height = width;
+    }
+  }
+
+  int stride_y = target_width;
+  int stride_uv = (target_width + 1) / 2;
+
+  // Setting absolute height (in case it was negative).
+  // In Windows, the image starts bottom left, instead of top left.
+  // Setting a negative source height, inverts the image (within LibYuv).
+  rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(
+      target_width, target_height, stride_y, stride_uv, stride_uv);
+
+  libyuv::RotationMode rotation_mode = libyuv::kRotate0;
+  if (apply_rotation) {
+    switch (_rotateFrame) {
+      case kVideoRotation_0:
+        rotation_mode = libyuv::kRotate0;
+        break;
+      case kVideoRotation_90:
+        rotation_mode = libyuv::kRotate90;
+        break;
+      case kVideoRotation_180:
+        rotation_mode = libyuv::kRotate180;
+        break;
+      case kVideoRotation_270:
+        rotation_mode = libyuv::kRotate270;
+        break;
+    }
+  }
+
+  int dst_width = buffer->width();
+  int dst_height = buffer->height();
+
+  // LibYuv expects pre-rotation_mode values for dst.
+  // Stride values should correspond to the destination values.
+  if (rotation_mode == libyuv::kRotate90 || rotation_mode == libyuv::kRotate270) {
+    std::swap(dst_width, dst_height);
+  }
+
+  const int conversionResult = libyuv::ConvertToI420(
+      videoFrame, videoFrameLength, buffer.get()->MutableDataY(),
+      buffer.get()->StrideY(), buffer.get()->MutableDataU(),
+      buffer.get()->StrideU(), buffer.get()->MutableDataV(),
+      buffer.get()->StrideV(), 0, 0,  // No Cropping
+      width, height, dst_width, dst_height, rotation_mode,
+      ConvertVideoType(frameInfo.videoType));
+  if (conversionResult != 0) {
+    RTC_LOG(LS_ERROR) << "Failed to convert capture frame from type "
+                      << static_cast<int>(frameInfo.videoType) << "to I420.";
+    return -1;
+  }
+
+  VideoFrame captureFrame =
+      VideoFrame::Builder()
+          .set_video_frame_buffer(buffer)
+          .set_timestamp_rtp(0)
+          .set_timestamp_ms(rtc::TimeMillis())
+          .set_rotation(!apply_rotation ? _rotateFrame : kVideoRotation_0)
+          .build();
+  captureFrame.set_ntp_time_ms(captureTime);
+
+  // This is one ugly hack to let CamerasParent know what rotation
+  // the frame was captured at. Note that this goes against the intended
+  // meaning of rotation of the frame (how to rotate it before rendering).
+  // We do this so CamerasChild can scale to the proper dimensions
+  // later on in the pipe.
+  captureFrame.set_rotation(_rotateFrame);
+
+  DeliverCapturedFrame(captureFrame);
+
+  return 0;
+}
+
+int32_t VideoCaptureImpl::StartCapture(
+    const VideoCaptureCapability& capability) {
+  _requestedCapability = capability;
+  return -1;
+}
+
+int32_t VideoCaptureImpl::StopCapture() {
+  return -1;
+}
+
+bool VideoCaptureImpl::CaptureStarted() {
+  return false;
+}
+
+int32_t VideoCaptureImpl::CaptureSettings(
+    VideoCaptureCapability& /*settings*/) {
+  return -1;
+}
+
+int32_t VideoCaptureImpl::SetCaptureRotation(VideoRotation rotation) {
+  MutexLock lock(&api_lock_);
+  _rotateFrame = rotation;
+  return 0;
+}
+
+bool VideoCaptureImpl::SetApplyRotation(bool enable) {
+  // We can't take any lock here as it'll cause deadlock with IncomingFrame.
+
+  // The effect of this is the last caller wins.
+  apply_rotation_ = enable;
+  return true;
+}
+
+bool VideoCaptureImpl::GetApplyRotation() {
+  return apply_rotation_;
+}
+
+void VideoCaptureImpl::UpdateFrameCount() {
+  if (_incomingFrameTimesNanos[0] / rtc::kNumNanosecsPerMicrosec == 0) {
+    // first no shift
+  } else {
+    // shift
+    for (int i = (kFrameRateCountHistorySize - 2); i >= 0; --i) {
+      _incomingFrameTimesNanos[i + 1] = _incomingFrameTimesNanos[i];
+    }
+  }
+  _incomingFrameTimesNanos[0] = rtc::TimeNanos();
+}
+
+uint32_t VideoCaptureImpl::CalculateFrameRate(int64_t now_ns) {
+  int32_t num = 0;
+  int32_t nrOfFrames = 0;
+  for (num = 1; num < (kFrameRateCountHistorySize - 1); ++num) {
+    if (_incomingFrameTimesNanos[num] <= 0 ||
+        (now_ns - _incomingFrameTimesNanos[num]) /
+                rtc::kNumNanosecsPerMillisec >
+            kFrameRateHistoryWindowMs) {  // don't use data older than 2sec
+      break;
+    } else {
+      nrOfFrames++;
+    }
+  }
+  if (num > 1) {
+    int64_t diff = (now_ns - _incomingFrameTimesNanos[num - 1]) /
+                   rtc::kNumNanosecsPerMillisec;
+    if (diff > 0) {
+      return uint32_t((nrOfFrames * 1000.0f / diff) + 0.5f);
+    }
+  }
+
+  return nrOfFrames;
+}
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_impl.h b/third_party/libwebrtc/modules/video_capture/video_capture_impl.h
new file mode 100644
index 0000000000..b9af5f2441
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture_impl.h
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_IMPL_H_
+#define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_IMPL_H_
+
+/*
+ * video_capture_impl.h
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/scoped_refptr.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_sink_interface.h"
+#include "modules/video_capture/video_capture.h"
+#include "modules/video_capture/video_capture_config.h"
+#include "modules/video_capture/video_capture_defines.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+namespace videocapturemodule {
+// Class definitions
+class VideoCaptureImpl : public VideoCaptureModule {
+ public:
+  /*
+   *   Create a video capture module object
+   *
+   *   id              - unique identifier of this video capture module object
+   *   deviceUniqueIdUTF8 -  name of the device. Available names can be found by
+   * using GetDeviceName
+   */
+  static rtc::scoped_refptr<VideoCaptureModule> Create(
+      const char* deviceUniqueIdUTF8);
+
+  static DeviceInfo* CreateDeviceInfo();
+
+  // Helpers for converting between (integral) degrees and
+  // VideoRotation values.  Return 0 on success.
+  static int32_t RotationFromDegrees(int degrees, VideoRotation* rotation);
+  static int32_t RotationInDegrees(VideoRotation rotation, int* degrees);
+
+  // Call backs
+  void RegisterCaptureDataCallback(
+      rtc::VideoSinkInterface<VideoFrame>* dataCallback) override;
+  virtual void RegisterCaptureDataCallback(
+      RawVideoSinkInterface* dataCallback) override;
+  void DeRegisterCaptureDataCallback(
+      rtc::VideoSinkInterface<VideoFrame>* dataCallback) override;
+
+  int32_t StopCaptureIfAllClientsClose() override;
+  int32_t SetCaptureRotation(VideoRotation rotation) override;
+  bool SetApplyRotation(bool enable) override;
+  bool GetApplyRotation() override;
+
+  const char* CurrentDeviceName() const override;
+
+  // `capture_time` must be specified in NTP time format in milliseconds.
+  int32_t IncomingFrame(uint8_t* videoFrame,
+                        size_t videoFrameLength,
+                        const VideoCaptureCapability& frameInfo,
+                        int64_t captureTime = 0);
+
+  // Platform dependent
+  int32_t StartCapture(const VideoCaptureCapability& capability) override;
+  int32_t StopCapture() override;
+  bool CaptureStarted() override;
+  int32_t CaptureSettings(VideoCaptureCapability& /*settings*/) override;
+
+ protected:
+  VideoCaptureImpl();
+  ~VideoCaptureImpl() override;
+
+  // moved DeliverCapturedFrame to protected for VideoCaptureAndroid (mjf)
+  int32_t DeliverCapturedFrame(VideoFrame& captureFrame);
+
+  char* _deviceUniqueId;  // current Device unique name;
+  Mutex api_lock_;
+  VideoCaptureCapability _requestedCapability;  // Should be set by platform
+                                                // dependent code in
+                                                // StartCapture.
+ private:
+  void UpdateFrameCount();
+  uint32_t CalculateFrameRate(int64_t now_ns);
+  void DeliverRawFrame(uint8_t* videoFrame,
+                       size_t videoFrameLength,
+                       const VideoCaptureCapability& frameInfo,
+                       int64_t captureTime);
+
+  // last time the module process function was called.
+  int64_t _lastProcessTimeNanos;
+  // last time the frame rate callback function was called.
+  int64_t _lastFrameRateCallbackTimeNanos;
+
+  std::set<rtc::VideoSinkInterface<VideoFrame>*> _dataCallBacks;
+  RawVideoSinkInterface* _rawDataCallBack;
+
+  int64_t _lastProcessFrameTimeNanos;
+  // timestamp for local captured frames
+  int64_t _incomingFrameTimesNanos[kFrameRateCountHistorySize];
+  VideoRotation _rotateFrame;  // Set if the frame should be rotated by the
+                               // capture module.
+
+  // Indicate whether rotation should be applied before delivered externally.
+  bool apply_rotation_;
+};
+}  // namespace videocapturemodule
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_IMPL_H_
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_internal_impl_gn/moz.build b/third_party/libwebrtc/modules/video_capture/video_capture_internal_impl_gn/moz.build
new file mode 100644
index 0000000000..cd5557b826
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture_internal_impl_gn/moz.build
@@ -0,0 +1,254 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "GLESv2",
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/video_capture/linux/device_info_linux.cc",
+        "/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.cc",
+        "/third_party/libwebrtc/modules/video_capture/linux/video_capture_linux.cc",
+        "/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.cc"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/video_capture/linux/device_info_linux.cc",
+        "/third_party/libwebrtc/modules/video_capture/linux/device_info_v4l2.cc",
+        "/third_party/libwebrtc/modules/video_capture/linux/video_capture_linux.cc",
+        "/third_party/libwebrtc/modules/video_capture/linux/video_capture_v4l2.cc"
+    ]
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "ole32",
+        "oleaut32",
+        "secur32",
+        "strmiids",
+        "user32",
+        "winmm"
+    ]
+
+    SOURCES += [
+        "/third_party/libwebrtc/modules/video_capture/windows/device_info_ds.cc",
+        "/third_party/libwebrtc/modules/video_capture/windows/help_functions_ds.cc",
+        "/third_party/libwebrtc/modules/video_capture/windows/sink_filter_ds.cc"
+    ]
+
+    UNIFIED_SOURCES += [
+        "/third_party/libwebrtc/modules/video_capture/windows/video_capture_ds.cc",
+        "/third_party/libwebrtc/modules/video_capture/windows/video_capture_factory_windows.cc"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("video_capture_internal_impl_gn")
diff --git a/third_party/libwebrtc/modules/video_capture/video_capture_module_gn/moz.build b/third_party/libwebrtc/modules/video_capture/video_capture_module_gn/moz.build
new file mode 100644
index 0000000000..8aa245b127
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/video_capture_module_gn/moz.build
@@ -0,0 +1,237 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/media/libyuv/",
+    "/media/libyuv/libyuv/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_capture/device_info_impl.cc",
+    "/third_party/libwebrtc/modules/video_capture/video_capture_factory.cc",
+    "/third_party/libwebrtc/modules/video_capture/video_capture_impl.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "GLESv2",
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("video_capture_module_gn")
diff --git a/third_party/libwebrtc/modules/video_capture/windows/device_info_ds.cc b/third_party/libwebrtc/modules/video_capture/windows/device_info_ds.cc
new file mode 100644
index 0000000000..2b01fc6930
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/windows/device_info_ds.cc
@@ -0,0 +1,713 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/windows/device_info_ds.h"
+
+#include <dvdmedia.h>
+
+#include "modules/video_capture/video_capture_config.h"
+#include "modules/video_capture/windows/help_functions_ds.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+
+BOOL isVideoDevice(DEV_BROADCAST_HDR *pHdr)
+{
+  if (pHdr == NULL) {
+    return FALSE;
+  }
+  if (pHdr->dbch_devicetype != DBT_DEVTYP_DEVICEINTERFACE) {
+    return FALSE;
+  }
+  DEV_BROADCAST_DEVICEINTERFACE* pDi = (DEV_BROADCAST_DEVICEINTERFACE*)pHdr;
+  return pDi->dbcc_classguid == KSCATEGORY_VIDEO_CAMERA;
+}
+
+LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
+{
+    DeviceInfoDS* pParent;
+    if (uiMsg == WM_CREATE)
+    {
+        pParent = (DeviceInfoDS*)((LPCREATESTRUCT)lParam)->lpCreateParams;
+        SetWindowLongPtr(hWnd, GWLP_USERDATA, (LONG_PTR)pParent);
+    }
+    else if (uiMsg == WM_DESTROY)
+    {
+        SetWindowLongPtr(hWnd, GWLP_USERDATA, NULL);
+    }
+    else if (uiMsg == WM_DEVICECHANGE)
+    {
+        pParent = (DeviceInfoDS*)GetWindowLongPtr(hWnd, GWLP_USERDATA);
+        if (pParent && isVideoDevice((PDEV_BROADCAST_HDR)lParam))
+        {
+            pParent->DeviceChange();
+        }
+    }
+    return DefWindowProc(hWnd, uiMsg, wParam, lParam);
+}
+
+// static
+DeviceInfoDS* DeviceInfoDS::Create() {
+  DeviceInfoDS* dsInfo = new DeviceInfoDS();
+  if (!dsInfo || dsInfo->Init() != 0) {
+    delete dsInfo;
+    dsInfo = NULL;
+  }
+  return dsInfo;
+}
+
+DeviceInfoDS::DeviceInfoDS()
+    : _dsDevEnum(NULL),
+      _dsMonikerDevEnum(NULL),
+      _CoUninitializeIsRequired(true),
+      _hdevnotify(NULL) {
+  // 1) Initialize the COM library (make Windows load the DLLs).
+  //
+  // CoInitializeEx must be called at least once, and is usually called only
+  // once, for each thread that uses the COM library. Multiple calls to
+  // CoInitializeEx by the same thread are allowed as long as they pass the same
+  // concurrency flag, but subsequent valid calls return S_FALSE. To close the
+  // COM library gracefully on a thread, each successful call to CoInitializeEx,
+  // including any call that returns S_FALSE, must be balanced by a
+  // corresponding call to CoUninitialize.
+  //
+
+  /*Apartment-threading, while allowing for multiple threads of execution,
+   serializes all incoming calls by requiring that calls to methods of objects
+   created by this thread always run on the same thread the apartment/thread
+   that created them. In addition, calls can arrive only at message-queue
+   boundaries (i.e., only during a PeekMessage, SendMessage, DispatchMessage,
+   etc.). Because of this serialization, it is not typically necessary to write
+   concurrency control into the code for the object, other than to avoid calls
+   to PeekMessage and SendMessage during processing that must not be interrupted
+   by other method invocations or calls to other objects in the same
+   apartment/thread.*/
+
+  /// CoInitializeEx(NULL, COINIT_APARTMENTTHREADED ); //|
+  /// COINIT_SPEED_OVER_MEMORY
+  HRESULT hr = CoInitializeEx(
+      NULL, COINIT_MULTITHREADED);  // Use COINIT_MULTITHREADED since Voice
+                                    // Engine uses COINIT_MULTITHREADED
+  if (FAILED(hr)) {
+    // Avoid calling CoUninitialize() since CoInitializeEx() failed.
+    _CoUninitializeIsRequired = FALSE;
+
+    if (hr == RPC_E_CHANGED_MODE) {
+      // Calling thread has already initialized COM to be used in a
+      // single-threaded apartment (STA). We are then prevented from using STA.
+      // Details: hr = 0x80010106 <=> "Cannot change thread mode after it is
+      // set".
+      //
+      RTC_DLOG(LS_INFO) << __FUNCTION__
+                        << ": CoInitializeEx(NULL, COINIT_APARTMENTTHREADED)"
+                           " => RPC_E_CHANGED_MODE, error 0x"
+                        << rtc::ToHex(hr);
+    }
+  }
+
+  _hInstance = reinterpret_cast<HINSTANCE>(GetModuleHandle(NULL));
+  _wndClass = {0};
+  _wndClass.lpfnWndProc = &WndProc;
+  _wndClass.lpszClassName = TEXT("DeviceInfoDS");
+  _wndClass.hInstance = _hInstance;
+
+  if (RegisterClass(&_wndClass)) {
+    _hwnd = CreateWindow(_wndClass.lpszClassName, NULL, 0, CW_USEDEFAULT,
+                         CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT, NULL,
+                         NULL, _hInstance, this);
+
+    DEV_BROADCAST_DEVICEINTERFACE di = { 0 };
+    di.dbcc_size = sizeof(di);
+    di.dbcc_devicetype  = DBT_DEVTYP_DEVICEINTERFACE;
+    di.dbcc_classguid  = KSCATEGORY_VIDEO_CAMERA;
+
+    _hdevnotify = RegisterDeviceNotification(_hwnd, &di,
+                                             DEVICE_NOTIFY_WINDOW_HANDLE);
+  }
+}
+
+DeviceInfoDS::~DeviceInfoDS() {
+  RELEASE_AND_CLEAR(_dsMonikerDevEnum);
+  RELEASE_AND_CLEAR(_dsDevEnum);
+  if (_CoUninitializeIsRequired) {
+    CoUninitialize();
+  }
+  if (_hdevnotify)
+  {
+    UnregisterDeviceNotification(_hdevnotify);
+  }
+  if (_hwnd != NULL) {
+    DestroyWindow(_hwnd);
+  }
+  UnregisterClass(_wndClass.lpszClassName, _hInstance);
+}
+
+int32_t DeviceInfoDS::Init() {
+  HRESULT hr = CoCreateInstance(CLSID_SystemDeviceEnum, NULL, CLSCTX_INPROC,
+                                IID_ICreateDevEnum, (void**)&_dsDevEnum);
+  if (hr != NOERROR) {
+    RTC_LOG(LS_INFO) << "Failed to create CLSID_SystemDeviceEnum, error 0x"
+                     << rtc::ToHex(hr);
+    return -1;
+  }
+  return 0;
+}
+uint32_t DeviceInfoDS::NumberOfDevices() {
+  MutexLock lock(&_apiLock);
+  return GetDeviceInfo(0, 0, 0, 0, 0, 0, 0);
+}
+
+int32_t DeviceInfoDS::GetDeviceName(uint32_t deviceNumber,
+                                    char* deviceNameUTF8,
+                                    uint32_t deviceNameLength,
+                                    char* deviceUniqueIdUTF8,
+                                    uint32_t deviceUniqueIdUTF8Length,
+                                    char* productUniqueIdUTF8,
+                                    uint32_t productUniqueIdUTF8Length,
+                                    pid_t* pid) {
+  MutexLock lock(&_apiLock);
+  const int32_t result = GetDeviceInfo(
+      deviceNumber, deviceNameUTF8, deviceNameLength, deviceUniqueIdUTF8,
+      deviceUniqueIdUTF8Length, productUniqueIdUTF8, productUniqueIdUTF8Length);
+  return result > (int32_t)deviceNumber ? 0 : -1;
+}
+
+int32_t DeviceInfoDS::GetDeviceInfo(uint32_t deviceNumber,
+                                    char* deviceNameUTF8,
+                                    uint32_t deviceNameLength,
+                                    char* deviceUniqueIdUTF8,
+                                    uint32_t deviceUniqueIdUTF8Length,
+                                    char* productUniqueIdUTF8,
+                                    uint32_t productUniqueIdUTF8Length)
+
+{
+  // enumerate all video capture devices
+  RELEASE_AND_CLEAR(_dsMonikerDevEnum);
+  HRESULT hr = _dsDevEnum->CreateClassEnumerator(CLSID_VideoInputDeviceCategory,
+                                                 &_dsMonikerDevEnum, 0);
+  if (hr != NOERROR) {
+    RTC_LOG(LS_INFO) << "Failed to enumerate CLSID_SystemDeviceEnum, error 0x"
+                     << rtc::ToHex(hr) << ". No webcam exist?";
+    return 0;
+  }
+
+  _dsMonikerDevEnum->Reset();
+  ULONG cFetched;
+  IMoniker* pM;
+  int index = 0;
+  while (S_OK == _dsMonikerDevEnum->Next(1, &pM, &cFetched)) {
+    IPropertyBag* pBag;
+    hr = pM->BindToStorage(0, 0, IID_IPropertyBag, (void**)&pBag);
+    if (S_OK == hr) {
+      // Find the description or friendly name.
+      VARIANT varName;
+      VariantInit(&varName);
+      hr = pBag->Read(L"Description", &varName, 0);
+      if (FAILED(hr)) {
+        hr = pBag->Read(L"FriendlyName", &varName, 0);
+      }
+      if (SUCCEEDED(hr)) {
+        // ignore all VFW drivers
+        if ((wcsstr(varName.bstrVal, (L"(VFW)")) == NULL) &&
+            (_wcsnicmp(varName.bstrVal, (L"Google Camera Adapter"), 21) != 0)) {
+          // Found a valid device.
+          if (index == static_cast<int>(deviceNumber)) {
+            int convResult = 0;
+            if (deviceNameLength > 0) {
+              convResult = WideCharToMultiByte(CP_UTF8, 0, varName.bstrVal, -1,
+                                               (char*)deviceNameUTF8,
+                                               deviceNameLength, NULL, NULL);
+              if (convResult == 0) {
+                RTC_LOG(LS_INFO) << "Failed to convert device name to UTF8, "
+                                    "error = "
+                                 << GetLastError();
+                return -1;
+              }
+            }
+            if (deviceUniqueIdUTF8Length > 0) {
+              hr = pBag->Read(L"DevicePath", &varName, 0);
+              if (FAILED(hr)) {
+                strncpy_s((char*)deviceUniqueIdUTF8, deviceUniqueIdUTF8Length,
+                          (char*)deviceNameUTF8, convResult);
+                RTC_LOG(LS_INFO) << "Failed to get "
+                                    "deviceUniqueIdUTF8 using "
+                                    "deviceNameUTF8";
+              } else {
+                convResult = WideCharToMultiByte(
+                    CP_UTF8, 0, varName.bstrVal, -1, (char*)deviceUniqueIdUTF8,
+                    deviceUniqueIdUTF8Length, NULL, NULL);
+                if (convResult == 0) {
+                  RTC_LOG(LS_INFO) << "Failed to convert device "
+                                      "name to UTF8, error = "
+                                   << GetLastError();
+                  return -1;
+                }
+                if (productUniqueIdUTF8 && productUniqueIdUTF8Length > 0) {
+                  GetProductId(deviceUniqueIdUTF8, productUniqueIdUTF8,
+                               productUniqueIdUTF8Length);
+                }
+              }
+            }
+          }
+          ++index;  // increase the number of valid devices
+        }
+      }
+      VariantClear(&varName);
+      pBag->Release();
+      pM->Release();
+    }
+  }
+  if (deviceNameLength) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << " " << deviceNameUTF8;
+  }
+  return index;
+}
+
+IBaseFilter* DeviceInfoDS::GetDeviceFilter(const char* deviceUniqueIdUTF8,
+                                           char* productUniqueIdUTF8,
+                                           uint32_t productUniqueIdUTF8Length) {
+  const int32_t deviceUniqueIdUTF8Length = (int32_t)strlen(
+      (char*)deviceUniqueIdUTF8);  // UTF8 is also NULL terminated
+  if (deviceUniqueIdUTF8Length >= kVideoCaptureUniqueNameLength) {
+    RTC_LOG(LS_INFO) << "Device name too long";
+    return NULL;
+  }
+
+  // enumerate all video capture devices
+  RELEASE_AND_CLEAR(_dsMonikerDevEnum);
+  HRESULT hr = _dsDevEnum->CreateClassEnumerator(CLSID_VideoInputDeviceCategory,
+                                                 &_dsMonikerDevEnum, 0);
+  if (hr != NOERROR) {
+    RTC_LOG(LS_INFO) << "Failed to enumerate CLSID_SystemDeviceEnum, error 0x"
+                     << rtc::ToHex(hr) << ". No webcam exist?";
+    return 0;
+  }
+  _dsMonikerDevEnum->Reset();
+  ULONG cFetched;
+  IMoniker* pM;
+
+  IBaseFilter* captureFilter = NULL;
+  bool deviceFound = false;
+  while (S_OK == _dsMonikerDevEnum->Next(1, &pM, &cFetched) && !deviceFound) {
+    IPropertyBag* pBag;
+    hr = pM->BindToStorage(0, 0, IID_IPropertyBag, (void**)&pBag);
+    if (S_OK == hr) {
+      // Find the description or friendly name.
+      VARIANT varName;
+      VariantInit(&varName);
+      if (deviceUniqueIdUTF8Length > 0) {
+        hr = pBag->Read(L"DevicePath", &varName, 0);
+        if (FAILED(hr)) {
+          hr = pBag->Read(L"Description", &varName, 0);
+          if (FAILED(hr)) {
+            hr = pBag->Read(L"FriendlyName", &varName, 0);
+          }
+        }
+        if (SUCCEEDED(hr)) {
+          char tempDevicePathUTF8[256];
+          tempDevicePathUTF8[0] = 0;
+          WideCharToMultiByte(CP_UTF8, 0, varName.bstrVal, -1,
+                              tempDevicePathUTF8, sizeof(tempDevicePathUTF8),
+                              NULL, NULL);
+          if (strncmp(tempDevicePathUTF8, (const char*)deviceUniqueIdUTF8,
+                      deviceUniqueIdUTF8Length) == 0) {
+            // We have found the requested device
+            deviceFound = true;
+            hr =
+                pM->BindToObject(0, 0, IID_IBaseFilter, (void**)&captureFilter);
+            if
+              FAILED(hr) {
+                RTC_LOG(LS_ERROR) << "Failed to bind to the selected "
+                                     "capture device "
+                                  << hr;
+              }
+
+            if (productUniqueIdUTF8 &&
+                productUniqueIdUTF8Length > 0)  // Get the device name
+            {
+              GetProductId(deviceUniqueIdUTF8, productUniqueIdUTF8,
+                           productUniqueIdUTF8Length);
+            }
+          }
+        }
+      }
+      VariantClear(&varName);
+      pBag->Release();
+    }
+    pM->Release();
+  }
+  return captureFilter;
+}
+
+int32_t DeviceInfoDS::GetWindowsCapability(
+    const int32_t capabilityIndex,
+    VideoCaptureCapabilityWindows& windowsCapability) {
+  MutexLock lock(&_apiLock);
+
+  if (capabilityIndex < 0 || static_cast<size_t>(capabilityIndex) >=
+                                 _captureCapabilitiesWindows.size()) {
+    return -1;
+  }
+
+  windowsCapability = _captureCapabilitiesWindows[capabilityIndex];
+  return 0;
+}
+
+int32_t DeviceInfoDS::CreateCapabilityMap(const char* deviceUniqueIdUTF8)
+
+{
+  // Reset old capability list
+  _captureCapabilities.clear();
+
+  const int32_t deviceUniqueIdUTF8Length =
+      (int32_t)strlen((char*)deviceUniqueIdUTF8);
+  if (deviceUniqueIdUTF8Length >= kVideoCaptureUniqueNameLength) {
+    RTC_LOG(LS_INFO) << "Device name too long";
+    return -1;
+  }
+  RTC_LOG(LS_INFO) << "CreateCapabilityMap called for device "
+                   << deviceUniqueIdUTF8;
+
+  char productId[kVideoCaptureProductIdLength];
+  IBaseFilter* captureDevice = DeviceInfoDS::GetDeviceFilter(
+      deviceUniqueIdUTF8, productId, kVideoCaptureProductIdLength);
+  if (!captureDevice)
+    return -1;
+  IPin* outputCapturePin = GetOutputPin(captureDevice, GUID_NULL);
+  if (!outputCapturePin) {
+    RTC_LOG(LS_INFO) << "Failed to get capture device output pin";
+    RELEASE_AND_CLEAR(captureDevice);
+    return -1;
+  }
+  IAMExtDevice* extDevice = NULL;
+  HRESULT hr =
+      captureDevice->QueryInterface(IID_IAMExtDevice, (void**)&extDevice);
+  if (SUCCEEDED(hr) && extDevice) {
+    RTC_LOG(LS_INFO) << "This is an external device";
+    extDevice->Release();
+  }
+
+  IAMStreamConfig* streamConfig = NULL;
+  hr = outputCapturePin->QueryInterface(IID_IAMStreamConfig,
+                                        (void**)&streamConfig);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO) << "Failed to get IID_IAMStreamConfig interface "
+                        "from capture device";
+    return -1;
+  }
+
+  // this  gets the FPS
+  IAMVideoControl* videoControlConfig = NULL;
+  HRESULT hrVC = captureDevice->QueryInterface(IID_IAMVideoControl,
+                                               (void**)&videoControlConfig);
+  if (FAILED(hrVC)) {
+    RTC_LOG(LS_INFO) << "IID_IAMVideoControl Interface NOT SUPPORTED";
+  }
+
+  AM_MEDIA_TYPE* pmt = NULL;
+  VIDEO_STREAM_CONFIG_CAPS caps;
+  int count, size;
+
+  hr = streamConfig->GetNumberOfCapabilities(&count, &size);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO) << "Failed to GetNumberOfCapabilities";
+    RELEASE_AND_CLEAR(videoControlConfig);
+    RELEASE_AND_CLEAR(streamConfig);
+    RELEASE_AND_CLEAR(outputCapturePin);
+    RELEASE_AND_CLEAR(captureDevice);
+    return -1;
+  }
+
+  // Check if the device support formattype == FORMAT_VideoInfo2 and
+  // FORMAT_VideoInfo. Prefer FORMAT_VideoInfo since some cameras (ZureCam) has
+  // been seen having problem with MJPEG and FORMAT_VideoInfo2 Interlace flag is
+  // only supported in FORMAT_VideoInfo2
+  bool supportFORMAT_VideoInfo2 = false;
+  bool supportFORMAT_VideoInfo = false;
+  bool foundInterlacedFormat = false;
+  GUID preferedVideoFormat = FORMAT_VideoInfo;
+  for (int32_t tmp = 0; tmp < count; ++tmp) {
+    hr = streamConfig->GetStreamCaps(tmp, &pmt, reinterpret_cast<BYTE*>(&caps));
+    if (hr == S_OK) {
+      if (pmt->majortype == MEDIATYPE_Video &&
+          pmt->formattype == FORMAT_VideoInfo2) {
+        RTC_LOG(LS_INFO) << "Device support FORMAT_VideoInfo2";
+        supportFORMAT_VideoInfo2 = true;
+        VIDEOINFOHEADER2* h =
+            reinterpret_cast<VIDEOINFOHEADER2*>(pmt->pbFormat);
+        RTC_DCHECK(h);
+        foundInterlacedFormat |=
+            h->dwInterlaceFlags &
+            (AMINTERLACE_IsInterlaced | AMINTERLACE_DisplayModeBobOnly);
+      }
+      if (pmt->majortype == MEDIATYPE_Video &&
+          pmt->formattype == FORMAT_VideoInfo) {
+        RTC_LOG(LS_INFO) << "Device support FORMAT_VideoInfo2";
+        supportFORMAT_VideoInfo = true;
+      }
+
+      FreeMediaType(pmt);
+      pmt = NULL;
+    }
+  }
+  if (supportFORMAT_VideoInfo2) {
+    if (supportFORMAT_VideoInfo && !foundInterlacedFormat) {
+      preferedVideoFormat = FORMAT_VideoInfo;
+    } else {
+      preferedVideoFormat = FORMAT_VideoInfo2;
+    }
+  }
+
+  for (int32_t tmp = 0; tmp < count; ++tmp) {
+    hr = streamConfig->GetStreamCaps(tmp, &pmt, reinterpret_cast<BYTE*>(&caps));
+    if (hr != S_OK) {
+      RTC_LOG(LS_INFO) << "Failed to GetStreamCaps";
+      RELEASE_AND_CLEAR(videoControlConfig);
+      RELEASE_AND_CLEAR(streamConfig);
+      RELEASE_AND_CLEAR(outputCapturePin);
+      RELEASE_AND_CLEAR(captureDevice);
+      return -1;
+    }
+
+    if (pmt->majortype == MEDIATYPE_Video &&
+        pmt->formattype == preferedVideoFormat) {
+      VideoCaptureCapabilityWindows capability;
+      int64_t avgTimePerFrame = 0;
+
+      if (pmt->formattype == FORMAT_VideoInfo) {
+        VIDEOINFOHEADER* h = reinterpret_cast<VIDEOINFOHEADER*>(pmt->pbFormat);
+        RTC_DCHECK(h);
+        capability.directShowCapabilityIndex = tmp;
+        capability.width = h->bmiHeader.biWidth;
+        capability.height = h->bmiHeader.biHeight;
+        avgTimePerFrame = h->AvgTimePerFrame;
+      }
+      if (pmt->formattype == FORMAT_VideoInfo2) {
+        VIDEOINFOHEADER2* h =
+            reinterpret_cast<VIDEOINFOHEADER2*>(pmt->pbFormat);
+        RTC_DCHECK(h);
+        capability.directShowCapabilityIndex = tmp;
+        capability.width = h->bmiHeader.biWidth;
+        capability.height = h->bmiHeader.biHeight;
+        capability.interlaced =
+            h->dwInterlaceFlags &
+            (AMINTERLACE_IsInterlaced | AMINTERLACE_DisplayModeBobOnly);
+        avgTimePerFrame = h->AvgTimePerFrame;
+      }
+
+      if (hrVC == S_OK) {
+        LONGLONG* frameDurationList = NULL;
+        LONGLONG maxFPS = 0;
+        long listSize = 0;
+        SIZE size;
+        size.cx = capability.width;
+        size.cy = capability.height;
+
+        // GetMaxAvailableFrameRate doesn't return max frame rate always
+        // eg: Logitech Notebook. This may be due to a bug in that API
+        // because GetFrameRateList array is reversed in the above camera. So
+        // a util method written. Can't assume the first value will return
+        // the max fps.
+        hrVC = videoControlConfig->GetFrameRateList(
+            outputCapturePin, tmp, size, &listSize, &frameDurationList);
+
+        if (hrVC == S_OK) {
+          maxFPS = GetMaxOfFrameArray(frameDurationList, listSize);
+        }
+
+        CoTaskMemFree(frameDurationList);
+        frameDurationList = NULL;
+        listSize = 0;
+
+        // On some odd cameras, you may get a 0 for duration. Some others may
+        // not update the out vars. GetMaxOfFrameArray returns the lowest
+        // duration (highest FPS), or 0 if there was no list with elements.
+        if (0 != maxFPS) {
+          capability.maxFPS = static_cast<int>(10000000 / maxFPS);
+          capability.supportFrameRateControl = true;
+        } else  // use existing method
+        {
+          RTC_LOG(LS_INFO) << "GetMaxAvailableFrameRate NOT SUPPORTED";
+          if (avgTimePerFrame > 0)
+            capability.maxFPS = static_cast<int>(10000000 / avgTimePerFrame);
+          else
+            capability.maxFPS = 0;
+        }
+      } else  // use existing method in case IAMVideoControl is not supported
+      {
+        if (avgTimePerFrame > 0)
+          capability.maxFPS = static_cast<int>(10000000 / avgTimePerFrame);
+        else
+          capability.maxFPS = 0;
+      }
+
+      // can't switch MEDIATYPE :~(
+      if (pmt->subtype == MEDIASUBTYPE_I420) {
+        capability.videoType = VideoType::kI420;
+      } else if (pmt->subtype == MEDIASUBTYPE_IYUV) {
+        capability.videoType = VideoType::kIYUV;
+      } else if (pmt->subtype == MEDIASUBTYPE_RGB24) {
+        capability.videoType = VideoType::kRGB24;
+      } else if (pmt->subtype == MEDIASUBTYPE_YUY2) {
+        capability.videoType = VideoType::kYUY2;
+      } else if (pmt->subtype == MEDIASUBTYPE_RGB565) {
+        capability.videoType = VideoType::kRGB565;
+      } else if (pmt->subtype == MEDIASUBTYPE_MJPG) {
+        capability.videoType = VideoType::kMJPEG;
+      } else if (pmt->subtype == MEDIASUBTYPE_dvsl ||
+                 pmt->subtype == MEDIASUBTYPE_dvsd ||
+                 pmt->subtype ==
+                     MEDIASUBTYPE_dvhd)  // If this is an external DV camera
+      {
+        capability.videoType =
+            VideoType::kYUY2;  // MS DV filter seems to create this type
+      } else if (pmt->subtype ==
+                 MEDIASUBTYPE_UYVY)  // Seen used by Declink capture cards
+      {
+        capability.videoType = VideoType::kUYVY;
+      } else if (pmt->subtype ==
+                 MEDIASUBTYPE_HDYC)  // Seen used by Declink capture cards. Uses
+                                     // BT. 709 color. Not entiry correct to use
+                                     // UYVY. http://en.wikipedia.org/wiki/YCbCr
+      {
+        RTC_LOG(LS_INFO) << "Device support HDYC.";
+        capability.videoType = VideoType::kUYVY;
+      } else {
+        WCHAR strGuid[39];
+        StringFromGUID2(pmt->subtype, strGuid, 39);
+        RTC_LOG(LS_WARNING)
+            << "Device support unknown media type " << strGuid << ", width "
+            << capability.width << ", height " << capability.height;
+        continue;
+      }
+
+      _captureCapabilities.push_back(capability);
+      _captureCapabilitiesWindows.push_back(capability);
+      RTC_LOG(LS_INFO) << "Camera capability, width:" << capability.width
+                       << " height:" << capability.height
+                       << " type:" << static_cast<int>(capability.videoType)
+                       << " fps:" << capability.maxFPS;
+    }
+    FreeMediaType(pmt);
+    pmt = NULL;
+  }
+  RELEASE_AND_CLEAR(streamConfig);
+  RELEASE_AND_CLEAR(videoControlConfig);
+  RELEASE_AND_CLEAR(outputCapturePin);
+  RELEASE_AND_CLEAR(captureDevice);  // Release the capture device
+
+  // Store the new used device name
+  _lastUsedDeviceNameLength = deviceUniqueIdUTF8Length;
+  _lastUsedDeviceName =
+      (char*)realloc(_lastUsedDeviceName, _lastUsedDeviceNameLength + 1);
+  memcpy(_lastUsedDeviceName, deviceUniqueIdUTF8,
+         _lastUsedDeviceNameLength + 1);
+  RTC_LOG(LS_INFO) << "CreateCapabilityMap " << _captureCapabilities.size();
+
+  return static_cast<int32_t>(_captureCapabilities.size());
+}
+
+// Constructs a product ID from the Windows DevicePath. on a USB device the
+// devicePath contains product id and vendor id. This seems to work for firewire
+// as well.
+// Example of device path:
+// "\\?\usb#vid_0408&pid_2010&mi_00#7&258e7aaf&0&0000#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
+// "\\?\avc#sony&dv-vcr&camcorder&dv#65b2d50301460008#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
+void DeviceInfoDS::GetProductId(const char* devicePath,
+                                char* productUniqueIdUTF8,
+                                uint32_t productUniqueIdUTF8Length) {
+  *productUniqueIdUTF8 = '\0';
+  char* startPos = strstr((char*)devicePath, "\\\\?\\");
+  if (!startPos) {
+    strncpy_s((char*)productUniqueIdUTF8, productUniqueIdUTF8Length, "", 1);
+    RTC_LOG(LS_INFO) << "Failed to get the product Id";
+    return;
+  }
+  startPos += 4;
+
+  char* pos = strchr(startPos, '&');
+  if (!pos || pos >= (char*)devicePath + strlen((char*)devicePath)) {
+    strncpy_s((char*)productUniqueIdUTF8, productUniqueIdUTF8Length, "", 1);
+    RTC_LOG(LS_INFO) << "Failed to get the product Id";
+    return;
+  }
+  // Find the second occurrence.
+  pos = strchr(pos + 1, '&');
+  uint32_t bytesToCopy = (uint32_t)(pos - startPos);
+  if (pos && (bytesToCopy < productUniqueIdUTF8Length) &&
+      bytesToCopy <= kVideoCaptureProductIdLength) {
+    strncpy_s((char*)productUniqueIdUTF8, productUniqueIdUTF8Length,
+              (char*)startPos, bytesToCopy);
+  } else {
+    strncpy_s((char*)productUniqueIdUTF8, productUniqueIdUTF8Length, "", 1);
+    RTC_LOG(LS_INFO) << "Failed to get the product Id";
+  }
+}
+
+int32_t DeviceInfoDS::DisplayCaptureSettingsDialogBox(
+    const char* deviceUniqueIdUTF8,
+    const char* dialogTitleUTF8,
+    void* parentWindow,
+    uint32_t positionX,
+    uint32_t positionY) {
+  MutexLock lock(&_apiLock);
+  HWND window = (HWND)parentWindow;
+
+  IBaseFilter* filter = GetDeviceFilter(deviceUniqueIdUTF8, NULL, 0);
+  if (!filter)
+    return -1;
+
+  ISpecifyPropertyPages* pPages = NULL;
+  CAUUID uuid;
+  HRESULT hr = S_OK;
+
+  hr = filter->QueryInterface(IID_ISpecifyPropertyPages, (LPVOID*)&pPages);
+  if (!SUCCEEDED(hr)) {
+    filter->Release();
+    return -1;
+  }
+  hr = pPages->GetPages(&uuid);
+  if (!SUCCEEDED(hr)) {
+    filter->Release();
+    return -1;
+  }
+
+  WCHAR tempDialogTitleWide[256];
+  tempDialogTitleWide[0] = 0;
+  int size = 255;
+
+  // UTF-8 to wide char
+  MultiByteToWideChar(CP_UTF8, 0, (char*)dialogTitleUTF8, -1,
+                      tempDialogTitleWide, size);
+
+  // Invoke a dialog box to display.
+
+  hr = OleCreatePropertyFrame(
+      window,               // You must create the parent window.
+      positionX,            // Horizontal position for the dialog box.
+      positionY,            // Vertical position for the dialog box.
+      tempDialogTitleWide,  // String used for the dialog box caption.
+      1,                    // Number of pointers passed in pPlugin.
+      (LPUNKNOWN*)&filter,  // Pointer to the filter.
+      uuid.cElems,          // Number of property pages.
+      uuid.pElems,          // Array of property page CLSIDs.
+      LOCALE_USER_DEFAULT,  // Locale ID for the dialog box.
+      0, NULL);             // Reserved
+  // Release memory.
+  if (uuid.pElems) {
+    CoTaskMemFree(uuid.pElems);
+  }
+  filter->Release();
+  return 0;
+}
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/windows/device_info_ds.h b/third_party/libwebrtc/modules/video_capture/windows/device_info_ds.h
new file mode 100644
index 0000000000..e6dfaed366
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/windows/device_info_ds.h
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_DS_H_
+#define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_DS_H_
+
+#include <dshow.h>
+#include <Ks.h>
+#include <dbt.h>
+
+#include "modules/video_capture/device_info_impl.h"
+#include "modules/video_capture/video_capture.h"
+#include "modules/video_capture/video_capture_impl.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+struct VideoCaptureCapabilityWindows : public VideoCaptureCapability {
+  uint32_t directShowCapabilityIndex;
+  bool supportFrameRateControl;
+  VideoCaptureCapabilityWindows() {
+    directShowCapabilityIndex = 0;
+    supportFrameRateControl = false;
+  }
+};
+
+class DeviceInfoDS : public DeviceInfoImpl {
+ public:
+  // Factory function.
+  static DeviceInfoDS* Create();
+
+  DeviceInfoDS();
+  ~DeviceInfoDS() override;
+
+  int32_t Init() override;
+  uint32_t NumberOfDevices() override;
+
+  /*
+   * Returns the available capture devices.
+   */
+  int32_t GetDeviceName(uint32_t deviceNumber,
+                        char* deviceNameUTF8,
+                        uint32_t deviceNameLength,
+                        char* deviceUniqueIdUTF8,
+                        uint32_t deviceUniqueIdUTF8Length,
+                        char* productUniqueIdUTF8,
+                        uint32_t productUniqueIdUTF8Length,
+                        pid_t* pid) override;
+
+  /*
+   * Display OS /capture device specific settings dialog
+   */
+  int32_t DisplayCaptureSettingsDialogBox(const char* deviceUniqueIdUTF8,
+                                          const char* dialogTitleUTF8,
+                                          void* parentWindow,
+                                          uint32_t positionX,
+                                          uint32_t positionY) override;
+
+  // Windows specific
+
+  /* Gets a capture device filter
+   The user of this API is responsible for releasing the filter when it not
+   needed.
+   */
+  IBaseFilter* GetDeviceFilter(const char* deviceUniqueIdUTF8,
+                               char* productUniqueIdUTF8 = NULL,
+                               uint32_t productUniqueIdUTF8Length = 0);
+
+  int32_t GetWindowsCapability(
+      int32_t capabilityIndex,
+      VideoCaptureCapabilityWindows& windowsCapability);
+
+  static void GetProductId(const char* devicePath,
+                           char* productUniqueIdUTF8,
+                           uint32_t productUniqueIdUTF8Length);
+
+ protected:
+  int32_t GetDeviceInfo(uint32_t deviceNumber,
+                        char* deviceNameUTF8,
+                        uint32_t deviceNameLength,
+                        char* deviceUniqueIdUTF8,
+                        uint32_t deviceUniqueIdUTF8Length,
+                        char* productUniqueIdUTF8,
+                        uint32_t productUniqueIdUTF8Length);
+
+  int32_t CreateCapabilityMap(const char* deviceUniqueIdUTF8) override
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(_apiLock);
+
+ private:
+  ICreateDevEnum* _dsDevEnum;
+  IEnumMoniker* _dsMonikerDevEnum;
+  bool _CoUninitializeIsRequired;
+  std::vector<VideoCaptureCapabilityWindows> _captureCapabilitiesWindows;
+  HWND _hwnd;
+  WNDCLASS _wndClass;
+  HINSTANCE _hInstance;
+  HDEVNOTIFY _hdevnotify;
+};
+}  // namespace videocapturemodule
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_DS_H_
diff --git a/third_party/libwebrtc/modules/video_capture/windows/help_functions_ds.cc b/third_party/libwebrtc/modules/video_capture/windows/help_functions_ds.cc
new file mode 100644
index 0000000000..47fecfe4a1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/windows/help_functions_ds.cc
@@ -0,0 +1,158 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <initguid.h>  // Must come before the help_functions_ds.h include so
+                       // that DEFINE_GUID() entries will be defined in this
+                       // object file.
+
+#include <cguid.h>
+
+#include "modules/video_capture/windows/help_functions_ds.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+// This returns minimum :), which will give max frame rate...
+LONGLONG GetMaxOfFrameArray(LONGLONG* maxFps, long size) {
+  if (!maxFps || size <= 0) {
+    return 0;
+  }
+  LONGLONG maxFPS = maxFps[0];
+  for (int i = 0; i < size; i++) {
+    if (maxFPS > maxFps[i])
+      maxFPS = maxFps[i];
+  }
+  return maxFPS;
+}
+
+IPin* GetInputPin(IBaseFilter* filter) {
+  IPin* pin = NULL;
+  IEnumPins* pPinEnum = NULL;
+  filter->EnumPins(&pPinEnum);
+  if (pPinEnum == NULL) {
+    return NULL;
+  }
+
+  // get first unconnected pin
+  pPinEnum->Reset();  // set to first pin
+
+  while (S_OK == pPinEnum->Next(1, &pin, NULL)) {
+    PIN_DIRECTION pPinDir;
+    pin->QueryDirection(&pPinDir);
+    if (PINDIR_INPUT == pPinDir)  // This is an input pin
+    {
+      IPin* tempPin = NULL;
+      if (S_OK != pin->ConnectedTo(&tempPin))  // The pint is not connected
+      {
+        pPinEnum->Release();
+        return pin;
+      }
+    }
+    pin->Release();
+  }
+  pPinEnum->Release();
+  return NULL;
+}
+
+IPin* GetOutputPin(IBaseFilter* filter, REFGUID Category) {
+  IPin* pin = NULL;
+  IEnumPins* pPinEnum = NULL;
+  filter->EnumPins(&pPinEnum);
+  if (pPinEnum == NULL) {
+    return NULL;
+  }
+  // get first unconnected pin
+  pPinEnum->Reset();  // set to first pin
+  while (S_OK == pPinEnum->Next(1, &pin, NULL)) {
+    PIN_DIRECTION pPinDir;
+    pin->QueryDirection(&pPinDir);
+    if (PINDIR_OUTPUT == pPinDir)  // This is an output pin
+    {
+      if (Category == GUID_NULL || PinMatchesCategory(pin, Category)) {
+        pPinEnum->Release();
+        return pin;
+      }
+    }
+    pin->Release();
+    pin = NULL;
+  }
+  pPinEnum->Release();
+  return NULL;
+}
+
+BOOL PinMatchesCategory(IPin* pPin, REFGUID Category) {
+  BOOL bFound = FALSE;
+  IKsPropertySet* pKs = NULL;
+  HRESULT hr = pPin->QueryInterface(IID_PPV_ARGS(&pKs));
+  if (SUCCEEDED(hr)) {
+    GUID PinCategory;
+    DWORD cbReturned;
+    hr = pKs->Get(AMPROPSETID_Pin, AMPROPERTY_PIN_CATEGORY, NULL, 0,
+                  &PinCategory, sizeof(GUID), &cbReturned);
+    if (SUCCEEDED(hr) && (cbReturned == sizeof(GUID))) {
+      bFound = (PinCategory == Category);
+    }
+    pKs->Release();
+  }
+  return bFound;
+}
+
+void ResetMediaType(AM_MEDIA_TYPE* media_type) {
+  if (!media_type)
+    return;
+  if (media_type->cbFormat != 0) {
+    CoTaskMemFree(media_type->pbFormat);
+    media_type->cbFormat = 0;
+    media_type->pbFormat = nullptr;
+  }
+  if (media_type->pUnk) {
+    media_type->pUnk->Release();
+    media_type->pUnk = nullptr;
+  }
+}
+
+void FreeMediaType(AM_MEDIA_TYPE* media_type) {
+  if (!media_type)
+    return;
+  ResetMediaType(media_type);
+  CoTaskMemFree(media_type);
+}
+
+HRESULT CopyMediaType(AM_MEDIA_TYPE* target, const AM_MEDIA_TYPE* source) {
+  RTC_DCHECK_NE(source, target);
+  *target = *source;
+  if (source->cbFormat != 0) {
+    RTC_DCHECK(source->pbFormat);
+    target->pbFormat =
+        reinterpret_cast<BYTE*>(CoTaskMemAlloc(source->cbFormat));
+    if (target->pbFormat == nullptr) {
+      target->cbFormat = 0;
+      return E_OUTOFMEMORY;
+    } else {
+      CopyMemory(target->pbFormat, source->pbFormat, target->cbFormat);
+    }
+  }
+
+  if (target->pUnk != nullptr)
+    target->pUnk->AddRef();
+
+  return S_OK;
+}
+
+wchar_t* DuplicateWideString(const wchar_t* str) {
+  size_t len = lstrlenW(str);
+  wchar_t* ret =
+      reinterpret_cast<LPWSTR>(CoTaskMemAlloc((len + 1) * sizeof(wchar_t)));
+  lstrcpyW(ret, str);
+  return ret;
+}
+
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/windows/help_functions_ds.h b/third_party/libwebrtc/modules/video_capture/windows/help_functions_ds.h
new file mode 100644
index 0000000000..29479157a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/windows/help_functions_ds.h
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_DS_H_
+#define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_DS_H_
+
+#include <dshow.h>
+
+#include <type_traits>
+#include <utility>
+
+#include "api/scoped_refptr.h"
+#include "rtc_base/ref_counter.h"
+
+DEFINE_GUID(MEDIASUBTYPE_I420,
+            0x30323449,
+            0x0000,
+            0x0010,
+            0x80,
+            0x00,
+            0x00,
+            0xAA,
+            0x00,
+            0x38,
+            0x9B,
+            0x71);
+DEFINE_GUID(MEDIASUBTYPE_HDYC,
+            0x43594448,
+            0x0000,
+            0x0010,
+            0x80,
+            0x00,
+            0x00,
+            0xAA,
+            0x00,
+            0x38,
+            0x9B,
+            0x71);
+
+#define RELEASE_AND_CLEAR(p) \
+  if (p) {                   \
+    (p)->Release();          \
+    (p) = NULL;              \
+  }
+
+namespace webrtc {
+namespace videocapturemodule {
+LONGLONG GetMaxOfFrameArray(LONGLONG* maxFps, long size);
+
+IPin* GetInputPin(IBaseFilter* filter);
+IPin* GetOutputPin(IBaseFilter* filter, REFGUID Category);
+BOOL PinMatchesCategory(IPin* pPin, REFGUID Category);
+void ResetMediaType(AM_MEDIA_TYPE* media_type);
+void FreeMediaType(AM_MEDIA_TYPE* media_type);
+HRESULT CopyMediaType(AM_MEDIA_TYPE* target, const AM_MEDIA_TYPE* source);
+
+// Helper function to make using scoped_refptr with COM interface pointers
+// a little less awkward. rtc::scoped_refptr doesn't support the & operator
+// or a way to receive values via an out ptr.
+// The function is intentionally not called QueryInterface to make things less
+// confusing for the compiler to figure out what the caller wants to do when
+// called from within the context of a class that also implements COM
+// interfaces.
+template <class T>
+HRESULT GetComInterface(IUnknown* object, rtc::scoped_refptr<T>* ptr) {
+  // This helper function is not meant to magically free ptr. If we do that
+  // we add code bloat to most places where it's not needed and make the code
+  // less readable since it's not clear at the call site that the pointer
+  // would get freed even inf QI() fails.
+  RTC_DCHECK(!ptr->get());
+  void* new_ptr = nullptr;
+  HRESULT hr = object->QueryInterface(__uuidof(T), &new_ptr);
+  if (SUCCEEDED(hr))
+    ptr->swap(reinterpret_cast<T**>(&new_ptr));
+  return hr;
+}
+
+// Provides a reference count implementation for COM (IUnknown derived) classes.
+// The implementation uses atomics for managing the ref count.
+template <class T>
+class ComRefCount : public T {
+ public:
+  ComRefCount() {}
+
+  template <class P0>
+  explicit ComRefCount(P0&& p0) : T(std::forward<P0>(p0)) {}
+
+  STDMETHOD_(ULONG, AddRef)() override {
+    ref_count_.IncRef();
+    return 1;
+  }
+
+  STDMETHOD_(ULONG, Release)() override {
+    const auto status = ref_count_.DecRef();
+    if (status == rtc::RefCountReleaseStatus::kDroppedLastRef) {
+      delete this;
+      return 0;
+    }
+    return 1;
+  }
+
+ protected:
+  ~ComRefCount() {}
+
+ private:
+  webrtc::webrtc_impl::RefCounter ref_count_{0};
+};
+
+}  // namespace videocapturemodule
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_DS_H_
diff --git a/third_party/libwebrtc/modules/video_capture/windows/sink_filter_ds.cc b/third_party/libwebrtc/modules/video_capture/windows/sink_filter_ds.cc
new file mode 100644
index 0000000000..0c5acb668d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/windows/sink_filter_ds.cc
@@ -0,0 +1,959 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/windows/sink_filter_ds.h"
+
+#include <dvdmedia.h>  // VIDEOINFOHEADER2
+#include <initguid.h>
+
+#include <algorithm>
+#include <list>
+
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/string_utils.h"
+
+DEFINE_GUID(CLSID_SINKFILTER,
+            0x88cdbbdc,
+            0xa73b,
+            0x4afa,
+            0xac,
+            0xbf,
+            0x15,
+            0xd5,
+            0xe2,
+            0xce,
+            0x12,
+            0xc3);
+
+namespace webrtc {
+namespace videocapturemodule {
+namespace {
+
+// Simple enumeration implementation that enumerates over a single pin :-/
+class EnumPins : public IEnumPins {
+ public:
+  EnumPins(IPin* pin) : pin_(pin) {}
+
+ protected:
+  virtual ~EnumPins() {}
+
+ private:
+  STDMETHOD(QueryInterface)(REFIID riid, void** ppv) override {
+    if (riid == IID_IUnknown || riid == IID_IEnumPins) {
+      *ppv = static_cast<IEnumPins*>(this);
+      AddRef();
+      return S_OK;
+    }
+    return E_NOINTERFACE;
+  }
+
+  STDMETHOD(Clone)(IEnumPins** pins) {
+    RTC_DCHECK_NOTREACHED();
+    return E_NOTIMPL;
+  }
+
+  STDMETHOD(Next)(ULONG count, IPin** pins, ULONG* fetched) {
+    RTC_DCHECK(count > 0);
+    RTC_DCHECK(pins);
+    // fetched may be NULL.
+
+    if (pos_ > 0) {
+      if (fetched)
+        *fetched = 0;
+      return S_FALSE;
+    }
+
+    ++pos_;
+    pins[0] = pin_.get();
+    pins[0]->AddRef();
+    if (fetched)
+      *fetched = 1;
+
+    return count == 1 ? S_OK : S_FALSE;
+  }
+
+  STDMETHOD(Skip)(ULONG count) {
+    RTC_DCHECK_NOTREACHED();
+    return E_NOTIMPL;
+  }
+
+  STDMETHOD(Reset)() {
+    pos_ = 0;
+    return S_OK;
+  }
+
+  rtc::scoped_refptr<IPin> pin_;
+  int pos_ = 0;
+};
+
+bool IsMediaTypePartialMatch(const AM_MEDIA_TYPE& a, const AM_MEDIA_TYPE& b) {
+  if (b.majortype != GUID_NULL && a.majortype != b.majortype)
+    return false;
+
+  if (b.subtype != GUID_NULL && a.subtype != b.subtype)
+    return false;
+
+  if (b.formattype != GUID_NULL) {
+    // if the format block is specified then it must match exactly
+    if (a.formattype != b.formattype)
+      return false;
+
+    if (a.cbFormat != b.cbFormat)
+      return false;
+
+    if (a.cbFormat != 0 && memcmp(a.pbFormat, b.pbFormat, a.cbFormat) != 0)
+      return false;
+  }
+
+  return true;
+}
+
+bool IsMediaTypeFullySpecified(const AM_MEDIA_TYPE& type) {
+  return type.majortype != GUID_NULL && type.formattype != GUID_NULL;
+}
+
+BYTE* AllocMediaTypeFormatBuffer(AM_MEDIA_TYPE* media_type, ULONG length) {
+  RTC_DCHECK(length);
+  if (media_type->cbFormat == length)
+    return media_type->pbFormat;
+
+  BYTE* buffer = static_cast<BYTE*>(CoTaskMemAlloc(length));
+  if (!buffer)
+    return nullptr;
+
+  if (media_type->pbFormat) {
+    RTC_DCHECK(media_type->cbFormat);
+    CoTaskMemFree(media_type->pbFormat);
+    media_type->pbFormat = nullptr;
+  }
+
+  media_type->cbFormat = length;
+  media_type->pbFormat = buffer;
+  return buffer;
+}
+
+void GetSampleProperties(IMediaSample* sample, AM_SAMPLE2_PROPERTIES* props) {
+  rtc::scoped_refptr<IMediaSample2> sample2;
+  if (SUCCEEDED(GetComInterface(sample, &sample2))) {
+    sample2->GetProperties(sizeof(*props), reinterpret_cast<BYTE*>(props));
+    return;
+  }
+
+  //  Get the properties the hard way.
+  props->cbData = sizeof(*props);
+  props->dwTypeSpecificFlags = 0;
+  props->dwStreamId = AM_STREAM_MEDIA;
+  props->dwSampleFlags = 0;
+
+  if (sample->IsDiscontinuity() == S_OK)
+    props->dwSampleFlags |= AM_SAMPLE_DATADISCONTINUITY;
+
+  if (sample->IsPreroll() == S_OK)
+    props->dwSampleFlags |= AM_SAMPLE_PREROLL;
+
+  if (sample->IsSyncPoint() == S_OK)
+    props->dwSampleFlags |= AM_SAMPLE_SPLICEPOINT;
+
+  if (SUCCEEDED(sample->GetTime(&props->tStart, &props->tStop)))
+    props->dwSampleFlags |= AM_SAMPLE_TIMEVALID | AM_SAMPLE_STOPVALID;
+
+  if (sample->GetMediaType(&props->pMediaType) == S_OK)
+    props->dwSampleFlags |= AM_SAMPLE_TYPECHANGED;
+
+  sample->GetPointer(&props->pbBuffer);
+  props->lActual = sample->GetActualDataLength();
+  props->cbBuffer = sample->GetSize();
+}
+
+// Returns true if the media type is supported, false otherwise.
+// For supported types, the `capability` will be populated accordingly.
+bool TranslateMediaTypeToVideoCaptureCapability(
+    const AM_MEDIA_TYPE* media_type,
+    VideoCaptureCapability* capability) {
+  RTC_DCHECK(capability);
+  if (!media_type || media_type->majortype != MEDIATYPE_Video ||
+      !media_type->pbFormat) {
+    return false;
+  }
+
+  const BITMAPINFOHEADER* bih = nullptr;
+  if (media_type->formattype == FORMAT_VideoInfo) {
+    bih = &reinterpret_cast<VIDEOINFOHEADER*>(media_type->pbFormat)->bmiHeader;
+  } else if (media_type->formattype != FORMAT_VideoInfo2) {
+    bih = &reinterpret_cast<VIDEOINFOHEADER2*>(media_type->pbFormat)->bmiHeader;
+  } else {
+    return false;
+  }
+
+  RTC_LOG(LS_INFO) << "TranslateMediaTypeToVideoCaptureCapability width:"
+                   << bih->biWidth << " height:" << bih->biHeight
+                   << " Compression:0x" << rtc::ToHex(bih->biCompression);
+
+  const GUID& sub_type = media_type->subtype;
+  if (sub_type == MEDIASUBTYPE_MJPG &&
+      bih->biCompression == MAKEFOURCC('M', 'J', 'P', 'G')) {
+    capability->videoType = VideoType::kMJPEG;
+  } else if (sub_type == MEDIASUBTYPE_I420 &&
+             bih->biCompression == MAKEFOURCC('I', '4', '2', '0')) {
+    capability->videoType = VideoType::kI420;
+  } else if (sub_type == MEDIASUBTYPE_YUY2 &&
+             bih->biCompression == MAKEFOURCC('Y', 'U', 'Y', '2')) {
+    capability->videoType = VideoType::kYUY2;
+  } else if (sub_type == MEDIASUBTYPE_UYVY &&
+             bih->biCompression == MAKEFOURCC('U', 'Y', 'V', 'Y')) {
+    capability->videoType = VideoType::kUYVY;
+  } else if (sub_type == MEDIASUBTYPE_HDYC) {
+    capability->videoType = VideoType::kUYVY;
+  } else if (sub_type == MEDIASUBTYPE_RGB24 && bih->biCompression == BI_RGB) {
+    capability->videoType = VideoType::kRGB24;
+  } else {
+    return false;
+  }
+
+  // Store the incoming width and height
+  capability->width = bih->biWidth;
+
+  // Store the incoming height,
+  // for RGB24 we assume the frame to be upside down
+  if (sub_type == MEDIASUBTYPE_RGB24 && bih->biHeight > 0) {
+    capability->height = -(bih->biHeight);
+  } else {
+    capability->height = abs(bih->biHeight);
+  }
+
+  return true;
+}
+
+class MediaTypesEnum : public IEnumMediaTypes {
+ public:
+  MediaTypesEnum(const VideoCaptureCapability& capability)
+      : capability_(capability),
+        format_preference_order_(
+            {// Default preferences, sorted by cost-to-convert-to-i420.
+             VideoType::kI420, VideoType::kYUY2, VideoType::kRGB24,
+             VideoType::kUYVY, VideoType::kMJPEG}) {
+    // Use the preferred video type, if supported.
+    auto it = std::find(format_preference_order_.begin(),
+                        format_preference_order_.end(), capability_.videoType);
+    if (it != format_preference_order_.end()) {
+      RTC_LOG(LS_INFO) << "Selected video type: " << *it;
+      // Move it to the front of the list, if it isn't already there.
+      if (it != format_preference_order_.begin()) {
+        format_preference_order_.splice(format_preference_order_.begin(),
+                                        format_preference_order_, it,
+                                        std::next(it));
+      }
+    } else {
+      RTC_LOG(LS_WARNING) << "Unsupported video type: " << *it
+                          << ", using default preference list.";
+    }
+  }
+
+ protected:
+  virtual ~MediaTypesEnum() {}
+
+ private:
+  STDMETHOD(QueryInterface)(REFIID riid, void** ppv) override {
+    if (riid == IID_IUnknown || riid == IID_IEnumMediaTypes) {
+      *ppv = static_cast<IEnumMediaTypes*>(this);
+      AddRef();
+      return S_OK;
+    }
+    return E_NOINTERFACE;
+  }
+
+  // IEnumMediaTypes
+  STDMETHOD(Clone)(IEnumMediaTypes** pins) {
+    RTC_DCHECK_NOTREACHED();
+    return E_NOTIMPL;
+  }
+
+  STDMETHOD(Next)(ULONG count, AM_MEDIA_TYPE** types, ULONG* fetched) {
+    RTC_DCHECK(count > 0);
+    RTC_DCHECK(types);
+    // fetched may be NULL.
+    if (fetched)
+      *fetched = 0;
+
+    for (ULONG i = 0;
+         i < count && pos_ < static_cast<int>(format_preference_order_.size());
+         ++i) {
+      AM_MEDIA_TYPE* media_type = reinterpret_cast<AM_MEDIA_TYPE*>(
+          CoTaskMemAlloc(sizeof(AM_MEDIA_TYPE)));
+      ZeroMemory(media_type, sizeof(*media_type));
+      types[i] = media_type;
+      VIDEOINFOHEADER* vih = reinterpret_cast<VIDEOINFOHEADER*>(
+          AllocMediaTypeFormatBuffer(media_type, sizeof(VIDEOINFOHEADER)));
+      ZeroMemory(vih, sizeof(*vih));
+      vih->bmiHeader.biSize = sizeof(BITMAPINFOHEADER);
+      vih->bmiHeader.biPlanes = 1;
+      vih->bmiHeader.biClrImportant = 0;
+      vih->bmiHeader.biClrUsed = 0;
+      if (capability_.maxFPS != 0)
+        vih->AvgTimePerFrame = 10000000 / capability_.maxFPS;
+
+      SetRectEmpty(&vih->rcSource);  // we want the whole image area rendered.
+      SetRectEmpty(&vih->rcTarget);  // no particular destination rectangle
+
+      media_type->majortype = MEDIATYPE_Video;
+      media_type->formattype = FORMAT_VideoInfo;
+      media_type->bTemporalCompression = FALSE;
+
+      // Set format information.
+      auto format_it = std::next(format_preference_order_.begin(), pos_++);
+      SetMediaInfoFromVideoType(*format_it, &vih->bmiHeader, media_type);
+
+      vih->bmiHeader.biWidth = capability_.width;
+      vih->bmiHeader.biHeight = capability_.height;
+      vih->bmiHeader.biSizeImage = ((vih->bmiHeader.biBitCount / 4) *
+                                    capability_.height * capability_.width) /
+                                   2;
+
+      RTC_DCHECK(vih->bmiHeader.biSizeImage);
+      media_type->lSampleSize = vih->bmiHeader.biSizeImage;
+      media_type->bFixedSizeSamples = true;
+      if (fetched)
+        ++(*fetched);
+    }
+    return pos_ == static_cast<int>(format_preference_order_.size()) ? S_FALSE
+                                                                     : S_OK;
+  }
+
+  static void SetMediaInfoFromVideoType(VideoType video_type,
+                                        BITMAPINFOHEADER* bitmap_header,
+                                        AM_MEDIA_TYPE* media_type) {
+    switch (video_type) {
+      case VideoType::kI420:
+        bitmap_header->biCompression = MAKEFOURCC('I', '4', '2', '0');
+        bitmap_header->biBitCount = 12;  // bit per pixel
+        media_type->subtype = MEDIASUBTYPE_I420;
+        break;
+      case VideoType::kYUY2:
+        bitmap_header->biCompression = MAKEFOURCC('Y', 'U', 'Y', '2');
+        bitmap_header->biBitCount = 16;  // bit per pixel
+        media_type->subtype = MEDIASUBTYPE_YUY2;
+        break;
+      case VideoType::kRGB24:
+        bitmap_header->biCompression = BI_RGB;
+        bitmap_header->biBitCount = 24;  // bit per pixel
+        media_type->subtype = MEDIASUBTYPE_RGB24;
+        break;
+      case VideoType::kUYVY:
+        bitmap_header->biCompression = MAKEFOURCC('U', 'Y', 'V', 'Y');
+        bitmap_header->biBitCount = 16;  // bit per pixel
+        media_type->subtype = MEDIASUBTYPE_UYVY;
+        break;
+      case VideoType::kMJPEG:
+        bitmap_header->biCompression = MAKEFOURCC('M', 'J', 'P', 'G');
+        bitmap_header->biBitCount = 12;  // bit per pixel
+        media_type->subtype = MEDIASUBTYPE_MJPG;
+        break;
+      default:
+        RTC_DCHECK_NOTREACHED();
+    }
+  }
+
+  STDMETHOD(Skip)(ULONG count) {
+    RTC_DCHECK_NOTREACHED();
+    return E_NOTIMPL;
+  }
+
+  STDMETHOD(Reset)() {
+    pos_ = 0;
+    return S_OK;
+  }
+
+  int pos_ = 0;
+  const VideoCaptureCapability capability_;
+  std::list<VideoType> format_preference_order_;
+};
+
+}  // namespace
+
+CaptureInputPin::CaptureInputPin(CaptureSinkFilter* filter) {
+  capture_checker_.Detach();
+  // No reference held to avoid circular references.
+  info_.pFilter = filter;
+  info_.dir = PINDIR_INPUT;
+}
+
+CaptureInputPin::~CaptureInputPin() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  ResetMediaType(&media_type_);
+}
+
+HRESULT CaptureInputPin::SetRequestedCapability(
+    const VideoCaptureCapability& capability) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  RTC_DCHECK(Filter()->IsStopped());
+  requested_capability_ = capability;
+  resulting_capability_ = VideoCaptureCapability();
+  return S_OK;
+}
+
+void CaptureInputPin::OnFilterActivated() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  runtime_error_ = false;
+  flushing_ = false;
+  capture_checker_.Detach();
+  capture_thread_id_ = 0;
+}
+
+void CaptureInputPin::OnFilterDeactivated() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  // Expedite shutdown by raising the flushing flag so no further processing
+  // on the capture thread occurs. When the graph is stopped and all filters
+  // have been told to stop, the media controller (graph) will wait for the
+  // capture thread to stop.
+  flushing_ = true;
+  if (allocator_)
+    allocator_->Decommit();
+}
+
+CaptureSinkFilter* CaptureInputPin::Filter() const {
+  return static_cast<CaptureSinkFilter*>(info_.pFilter);
+}
+
+HRESULT CaptureInputPin::AttemptConnection(IPin* receive_pin,
+                                           const AM_MEDIA_TYPE* media_type) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  RTC_DCHECK(Filter()->IsStopped());
+
+  // Check that the connection is valid  -- need to do this for every
+  // connect attempt since BreakConnect will undo it.
+  HRESULT hr = CheckDirection(receive_pin);
+  if (FAILED(hr))
+    return hr;
+
+  if (!TranslateMediaTypeToVideoCaptureCapability(media_type,
+                                                  &resulting_capability_)) {
+    ClearAllocator(true);
+    return VFW_E_TYPE_NOT_ACCEPTED;
+  }
+
+  // See if the other pin will accept this type.
+  hr = receive_pin->ReceiveConnection(static_cast<IPin*>(this), media_type);
+  if (FAILED(hr)) {
+    receive_pin_ = nullptr;  // Should already be null, but just in case.
+    return hr;
+  }
+
+  // Should have been set as part of the connect process.
+  RTC_DCHECK_EQ(receive_pin_, receive_pin);
+
+  ResetMediaType(&media_type_);
+  CopyMediaType(&media_type_, media_type);
+
+  return S_OK;
+}
+
+std::vector<AM_MEDIA_TYPE*> CaptureInputPin::DetermineCandidateFormats(
+    IPin* receive_pin,
+    const AM_MEDIA_TYPE* media_type) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  RTC_DCHECK(receive_pin);
+  RTC_DCHECK(media_type);
+
+  std::vector<AM_MEDIA_TYPE*> ret;
+
+  for (int i = 0; i < 2; i++) {
+    IEnumMediaTypes* types = nullptr;
+    if (i == 0) {
+      // First time around, try types from receive_pin.
+      receive_pin->EnumMediaTypes(&types);
+    } else {
+      // Then try ours.
+      EnumMediaTypes(&types);
+    }
+
+    if (types) {
+      while (true) {
+        ULONG fetched = 0;
+        AM_MEDIA_TYPE* this_type = nullptr;
+        if (types->Next(1, &this_type, &fetched) != S_OK)
+          break;
+
+        if (IsMediaTypePartialMatch(*this_type, *media_type)) {
+          ret.push_back(this_type);
+        } else {
+          FreeMediaType(this_type);
+        }
+      }
+      types->Release();
+    }
+  }
+
+  return ret;
+}
+
+void CaptureInputPin::ClearAllocator(bool decommit) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  if (!allocator_)
+    return;
+  if (decommit)
+    allocator_->Decommit();
+  allocator_ = nullptr;
+}
+
+HRESULT CaptureInputPin::CheckDirection(IPin* pin) const {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  PIN_DIRECTION pd;
+  pin->QueryDirection(&pd);
+  // Fairly basic check, make sure we don't pair input with input etc.
+  return pd == info_.dir ? VFW_E_INVALID_DIRECTION : S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureInputPin::QueryInterface(REFIID riid,
+                                                                  void** ppv) {
+  (*ppv) = nullptr;
+  if (riid == IID_IUnknown || riid == IID_IMemInputPin) {
+    *ppv = static_cast<IMemInputPin*>(this);
+  } else if (riid == IID_IPin) {
+    *ppv = static_cast<IPin*>(this);
+  }
+
+  if (!(*ppv))
+    return E_NOINTERFACE;
+
+  static_cast<IMemInputPin*>(this)->AddRef();
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::Connect(IPin* receive_pin, const AM_MEDIA_TYPE* media_type) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  if (!media_type || !receive_pin)
+    return E_POINTER;
+
+  if (!Filter()->IsStopped())
+    return VFW_E_NOT_STOPPED;
+
+  if (receive_pin_) {
+    RTC_DCHECK_NOTREACHED();
+    return VFW_E_ALREADY_CONNECTED;
+  }
+
+  if (IsMediaTypeFullySpecified(*media_type))
+    return AttemptConnection(receive_pin, media_type);
+
+  auto types = DetermineCandidateFormats(receive_pin, media_type);
+  bool connected = false;
+  for (auto* type : types) {
+    if (!connected && AttemptConnection(receive_pin, media_type) == S_OK)
+      connected = true;
+
+    FreeMediaType(type);
+  }
+
+  return connected ? S_OK : VFW_E_NO_ACCEPTABLE_TYPES;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::ReceiveConnection(IPin* connector,
+                                   const AM_MEDIA_TYPE* media_type) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  RTC_DCHECK(Filter()->IsStopped());
+
+  if (receive_pin_) {
+    RTC_DCHECK_NOTREACHED();
+    return VFW_E_ALREADY_CONNECTED;
+  }
+
+  HRESULT hr = CheckDirection(connector);
+  if (FAILED(hr))
+    return hr;
+
+  if (!TranslateMediaTypeToVideoCaptureCapability(media_type,
+                                                  &resulting_capability_))
+    return VFW_E_TYPE_NOT_ACCEPTED;
+
+  // Complete the connection
+
+  receive_pin_ = connector;
+  ResetMediaType(&media_type_);
+  CopyMediaType(&media_type_, media_type);
+
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureInputPin::Disconnect() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  if (!Filter()->IsStopped())
+    return VFW_E_NOT_STOPPED;
+
+  if (!receive_pin_)
+    return S_FALSE;
+
+  ClearAllocator(true);
+  receive_pin_ = nullptr;
+
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureInputPin::ConnectedTo(IPin** pin) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+
+  if (!receive_pin_)
+    return VFW_E_NOT_CONNECTED;
+
+  *pin = receive_pin_.get();
+  receive_pin_->AddRef();
+
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::ConnectionMediaType(AM_MEDIA_TYPE* media_type) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+
+  if (!receive_pin_)
+    return VFW_E_NOT_CONNECTED;
+
+  CopyMediaType(media_type, &media_type_);
+
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::QueryPinInfo(PIN_INFO* info) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  *info = info_;
+  if (info_.pFilter)
+    info_.pFilter->AddRef();
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::QueryDirection(PIN_DIRECTION* pin_dir) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  *pin_dir = info_.dir;
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureInputPin::QueryId(LPWSTR* id) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  size_t len = lstrlenW(info_.achName);
+  *id = reinterpret_cast<LPWSTR>(CoTaskMemAlloc((len + 1) * sizeof(wchar_t)));
+  lstrcpyW(*id, info_.achName);
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::QueryAccept(const AM_MEDIA_TYPE* media_type) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  RTC_DCHECK(Filter()->IsStopped());
+  VideoCaptureCapability capability(resulting_capability_);
+  return TranslateMediaTypeToVideoCaptureCapability(media_type, &capability)
+             ? S_FALSE
+             : S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::EnumMediaTypes(IEnumMediaTypes** types) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  *types = new ComRefCount<MediaTypesEnum>(requested_capability_);
+  (*types)->AddRef();
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::QueryInternalConnections(IPin** pins, ULONG* count) {
+  return E_NOTIMPL;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureInputPin::EndOfStream() {
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureInputPin::BeginFlush() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  flushing_ = true;
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureInputPin::EndFlush() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  flushing_ = false;
+  runtime_error_ = false;
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::NewSegment(REFERENCE_TIME start,
+                            REFERENCE_TIME stop,
+                            double rate) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::GetAllocator(IMemAllocator** allocator) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  if (allocator_ == nullptr) {
+    HRESULT hr = CoCreateInstance(CLSID_MemoryAllocator, 0,
+                                  CLSCTX_INPROC_SERVER, IID_IMemAllocator,
+                                  reinterpret_cast<void**>(allocator));
+    if (FAILED(hr))
+      return hr;
+    allocator_.swap(allocator);
+  }
+  *allocator = allocator_.get();
+  allocator_->AddRef();
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::NotifyAllocator(IMemAllocator* allocator, BOOL read_only) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  allocator_.swap(&allocator);
+  if (allocator_)
+    allocator_->AddRef();
+  if (allocator)
+    allocator->Release();
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::GetAllocatorRequirements(ALLOCATOR_PROPERTIES* props) {
+  return E_NOTIMPL;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::Receive(IMediaSample* media_sample) {
+  RTC_DCHECK_RUN_ON(&capture_checker_);
+
+  CaptureSinkFilter* const filter = static_cast<CaptureSinkFilter*>(Filter());
+
+  if (flushing_.load(std::memory_order_relaxed))
+    return S_FALSE;
+
+  if (runtime_error_.load(std::memory_order_relaxed))
+    return VFW_E_RUNTIME_ERROR;
+
+  if (!capture_thread_id_) {
+    // Make sure we set the thread name only once.
+    capture_thread_id_ = GetCurrentThreadId();
+    rtc::SetCurrentThreadName("webrtc_video_capture");
+  }
+
+  AM_SAMPLE2_PROPERTIES sample_props = {};
+  GetSampleProperties(media_sample, &sample_props);
+  // Has the format changed in this sample?
+  if (sample_props.dwSampleFlags & AM_SAMPLE_TYPECHANGED) {
+    // Check the derived class accepts the new format.
+    // This shouldn't fail as the source must call QueryAccept first.
+
+    // Note: This will modify resulting_capability_.
+    // That should be OK as long as resulting_capability_ is only modified
+    // on this thread while it is running (filter is not stopped), and only
+    // modified on the main thread when the filter is stopped (i.e. this thread
+    // is not running).
+    if (!TranslateMediaTypeToVideoCaptureCapability(sample_props.pMediaType,
+                                                    &resulting_capability_)) {
+      // Raise a runtime error if we fail the media type
+      runtime_error_ = true;
+      EndOfStream();
+      Filter()->NotifyEvent(EC_ERRORABORT, VFW_E_TYPE_NOT_ACCEPTED, 0);
+      return VFW_E_INVALIDMEDIATYPE;
+    }
+  }
+
+  filter->ProcessCapturedFrame(sample_props.pbBuffer, sample_props.lActual,
+                               resulting_capability_);
+
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureInputPin::ReceiveMultiple(IMediaSample** samples,
+                                 long count,
+                                 long* processed) {
+  HRESULT hr = S_OK;
+  *processed = 0;
+  while (count-- > 0) {
+    hr = Receive(samples[*processed]);
+    if (hr != S_OK)
+      break;
+    ++(*processed);
+  }
+  return hr;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureInputPin::ReceiveCanBlock() {
+  return S_FALSE;
+}
+
+//  ----------------------------------------------------------------------------
+
+CaptureSinkFilter::CaptureSinkFilter(VideoCaptureImpl* capture_observer)
+    : input_pin_(new ComRefCount<CaptureInputPin>(this)),
+      capture_observer_(capture_observer) {}
+
+CaptureSinkFilter::~CaptureSinkFilter() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+}
+
+HRESULT CaptureSinkFilter::SetRequestedCapability(
+    const VideoCaptureCapability& capability) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  // Called on the same thread as capture is started on.
+  return input_pin_->SetRequestedCapability(capability);
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureSinkFilter::GetState(DWORD msecs, FILTER_STATE* state) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  *state = state_;
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureSinkFilter::SetSyncSource(IReferenceClock* clock) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureSinkFilter::GetSyncSource(IReferenceClock** clock) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  return E_NOTIMPL;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureSinkFilter::Pause() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  state_ = State_Paused;
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureSinkFilter::Run(REFERENCE_TIME tStart) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  if (state_ == State_Stopped)
+    Pause();
+
+  state_ = State_Running;
+  input_pin_->OnFilterActivated();
+
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureSinkFilter::Stop() {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  if (state_ == State_Stopped)
+    return S_OK;
+
+  state_ = State_Stopped;
+  input_pin_->OnFilterDeactivated();
+
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureSinkFilter::EnumPins(IEnumPins** pins) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  *pins = new ComRefCount<class EnumPins>(input_pin_.get());
+  (*pins)->AddRef();
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureSinkFilter::FindPin(LPCWSTR id,
+                                                             IPin** pin) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  // There's no ID assigned to our input pin, so looking it up based on one
+  // is pointless (and in practice, this method isn't being used).
+  return VFW_E_NOT_FOUND;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureSinkFilter::QueryFilterInfo(FILTER_INFO* info) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  *info = info_;
+  if (info->pGraph)
+    info->pGraph->AddRef();
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureSinkFilter::JoinFilterGraph(IFilterGraph* graph, LPCWSTR name) {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  RTC_DCHECK(IsStopped());
+
+  // Note, since a reference to the filter is held by the graph manager,
+  // filters must not hold a reference to the graph. If they would, we'd have
+  // a circular reference. Instead, a pointer to the graph can be held without
+  // reference. See documentation for IBaseFilter::JoinFilterGraph for more.
+  info_.pGraph = graph;  // No AddRef().
+  sink_ = nullptr;
+
+  if (info_.pGraph) {
+    // make sure we don't hold on to the reference we may receive.
+    // Note that this assumes the same object identity, but so be it.
+    rtc::scoped_refptr<IMediaEventSink> sink;
+    GetComInterface(info_.pGraph, &sink);
+    sink_ = sink.get();
+  }
+
+  info_.achName[0] = L'\0';
+  if (name)
+    lstrcpynW(info_.achName, name, arraysize(info_.achName));
+
+  return S_OK;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureSinkFilter::QueryVendorInfo(LPWSTR* vendor_info) {
+  return E_NOTIMPL;
+}
+
+void CaptureSinkFilter::ProcessCapturedFrame(
+    unsigned char* buffer,
+    size_t length,
+    const VideoCaptureCapability& frame_info) {
+  // Called on the capture thread.
+  capture_observer_->IncomingFrame(buffer, length, frame_info);
+}
+
+void CaptureSinkFilter::NotifyEvent(long code,
+                                    LONG_PTR param1,
+                                    LONG_PTR param2) {
+  // Called on the capture thread.
+  if (!sink_)
+    return;
+
+  if (EC_COMPLETE == code)
+    param2 = reinterpret_cast<LONG_PTR>(static_cast<IBaseFilter*>(this));
+  sink_->Notify(code, param1, param2);
+}
+
+bool CaptureSinkFilter::IsStopped() const {
+  RTC_DCHECK_RUN_ON(&main_checker_);
+  return state_ == State_Stopped;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP
+CaptureSinkFilter::QueryInterface(REFIID riid, void** ppv) {
+  if (riid == IID_IUnknown || riid == IID_IPersist || riid == IID_IBaseFilter) {
+    *ppv = static_cast<IBaseFilter*>(this);
+    AddRef();
+    return S_OK;
+  }
+  return E_NOINTERFACE;
+}
+
+COM_DECLSPEC_NOTHROW STDMETHODIMP CaptureSinkFilter::GetClassID(CLSID* clsid) {
+  *clsid = CLSID_SINKFILTER;
+  return S_OK;
+}
+
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/windows/sink_filter_ds.h b/third_party/libwebrtc/modules/video_capture/windows/sink_filter_ds.h
new file mode 100644
index 0000000000..b0fabda3cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/windows/sink_filter_ds.h
@@ -0,0 +1,162 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
+#define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
+
+#include <dshow.h>
+
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "modules/video_capture/video_capture_impl.h"
+#include "modules/video_capture/windows/help_functions_ds.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+// forward declarations
+class CaptureSinkFilter;
+
+// Input pin for camera input
+// Implements IMemInputPin, IPin.
+class CaptureInputPin : public IMemInputPin, public IPin {
+ public:
+  CaptureInputPin(CaptureSinkFilter* filter);
+
+  HRESULT SetRequestedCapability(const VideoCaptureCapability& capability);
+
+  // Notifications from the filter.
+  void OnFilterActivated();
+  void OnFilterDeactivated();
+
+ protected:
+  virtual ~CaptureInputPin();
+
+ private:
+  CaptureSinkFilter* Filter() const;
+
+  HRESULT AttemptConnection(IPin* receive_pin, const AM_MEDIA_TYPE* media_type);
+  std::vector<AM_MEDIA_TYPE*> DetermineCandidateFormats(
+      IPin* receive_pin,
+      const AM_MEDIA_TYPE* media_type);
+  void ClearAllocator(bool decommit);
+  HRESULT CheckDirection(IPin* pin) const;
+
+  // IUnknown
+  STDMETHOD(QueryInterface)(REFIID riid, void** ppv) override;
+
+  // clang-format off
+  // clang isn't sure what to do with the longer STDMETHOD() function
+  // declarations.
+
+  // IPin
+  STDMETHOD(Connect)(IPin* receive_pin,
+                     const AM_MEDIA_TYPE* media_type) override;
+  STDMETHOD(ReceiveConnection)(IPin* connector,
+                               const AM_MEDIA_TYPE* media_type) override;
+  STDMETHOD(Disconnect)() override;
+  STDMETHOD(ConnectedTo)(IPin** pin) override;
+  STDMETHOD(ConnectionMediaType)(AM_MEDIA_TYPE* media_type) override;
+  STDMETHOD(QueryPinInfo)(PIN_INFO* info) override;
+  STDMETHOD(QueryDirection)(PIN_DIRECTION* pin_dir) override;
+  STDMETHOD(QueryId)(LPWSTR* id) override;
+  STDMETHOD(QueryAccept)(const AM_MEDIA_TYPE* media_type) override;
+  STDMETHOD(EnumMediaTypes)(IEnumMediaTypes** types) override;
+  STDMETHOD(QueryInternalConnections)(IPin** pins, ULONG* count) override;
+  STDMETHOD(EndOfStream)() override;
+  STDMETHOD(BeginFlush)() override;
+  STDMETHOD(EndFlush)() override;
+  STDMETHOD(NewSegment)(REFERENCE_TIME start, REFERENCE_TIME stop,
+                        double rate) override;
+
+  // IMemInputPin
+  STDMETHOD(GetAllocator)(IMemAllocator** allocator) override;
+  STDMETHOD(NotifyAllocator)(IMemAllocator* allocator, BOOL read_only) override;
+  STDMETHOD(GetAllocatorRequirements)(ALLOCATOR_PROPERTIES* props) override;
+  STDMETHOD(Receive)(IMediaSample* sample) override;
+  STDMETHOD(ReceiveMultiple)(IMediaSample** samples, long count,
+                             long* processed) override;
+  STDMETHOD(ReceiveCanBlock)() override;
+  // clang-format on
+
+  SequenceChecker main_checker_;
+  SequenceChecker capture_checker_;
+
+  VideoCaptureCapability requested_capability_ RTC_GUARDED_BY(main_checker_);
+  // Accessed on the main thread when Filter()->IsStopped() (capture thread not
+  // running), otherwise accessed on the capture thread.
+  VideoCaptureCapability resulting_capability_;
+  DWORD capture_thread_id_ = 0;
+  rtc::scoped_refptr<IMemAllocator> allocator_ RTC_GUARDED_BY(main_checker_);
+  rtc::scoped_refptr<IPin> receive_pin_ RTC_GUARDED_BY(main_checker_);
+  std::atomic_bool flushing_{false};
+  std::atomic_bool runtime_error_{false};
+  // Holds a referenceless pointer to the owning filter, the name and
+  // direction of the pin. The filter pointer can be considered const.
+  PIN_INFO info_ = {};
+  AM_MEDIA_TYPE media_type_ RTC_GUARDED_BY(main_checker_) = {};
+};
+
+// Implement IBaseFilter (including IPersist and IMediaFilter).
+class CaptureSinkFilter : public IBaseFilter {
+ public:
+  CaptureSinkFilter(VideoCaptureImpl* capture_observer);
+
+  HRESULT SetRequestedCapability(const VideoCaptureCapability& capability);
+
+  // Called on the capture thread.
+  void ProcessCapturedFrame(unsigned char* buffer,
+                            size_t length,
+                            const VideoCaptureCapability& frame_info);
+
+  void NotifyEvent(long code, LONG_PTR param1, LONG_PTR param2);
+  bool IsStopped() const;
+
+  //  IUnknown
+  STDMETHOD(QueryInterface)(REFIID riid, void** ppv) override;
+
+  // IPersist
+  STDMETHOD(GetClassID)(CLSID* clsid) override;
+
+  // IMediaFilter.
+  STDMETHOD(GetState)(DWORD msecs, FILTER_STATE* state) override;
+  STDMETHOD(SetSyncSource)(IReferenceClock* clock) override;
+  STDMETHOD(GetSyncSource)(IReferenceClock** clock) override;
+  STDMETHOD(Pause)() override;
+  STDMETHOD(Run)(REFERENCE_TIME start) override;
+  STDMETHOD(Stop)() override;
+
+  // IBaseFilter
+  STDMETHOD(EnumPins)(IEnumPins** pins) override;
+  STDMETHOD(FindPin)(LPCWSTR id, IPin** pin) override;
+  STDMETHOD(QueryFilterInfo)(FILTER_INFO* info) override;
+  STDMETHOD(JoinFilterGraph)(IFilterGraph* graph, LPCWSTR name) override;
+  STDMETHOD(QueryVendorInfo)(LPWSTR* vendor_info) override;
+
+ protected:
+  virtual ~CaptureSinkFilter();
+
+ private:
+  SequenceChecker main_checker_;
+  const rtc::scoped_refptr<ComRefCount<CaptureInputPin>> input_pin_;
+  VideoCaptureImpl* const capture_observer_;
+  FILTER_INFO info_ RTC_GUARDED_BY(main_checker_) = {};
+  // Set/cleared in JoinFilterGraph. The filter must be stopped (no capture)
+  // at that time, so no lock is required. While the state is not stopped,
+  // the sink will be used from the capture thread.
+  IMediaEventSink* sink_ = nullptr;
+  FILTER_STATE state_ RTC_GUARDED_BY(main_checker_) = State_Stopped;
+};
+}  // namespace videocapturemodule
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
diff --git a/third_party/libwebrtc/modules/video_capture/windows/video_capture_ds.cc b/third_party/libwebrtc/modules/video_capture/windows/video_capture_ds.cc
new file mode 100644
index 0000000000..b13ac074f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/windows/video_capture_ds.cc
@@ -0,0 +1,322 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_capture/windows/video_capture_ds.h"
+
+#include <dvdmedia.h>  // VIDEOINFOHEADER2
+
+#include "modules/video_capture/video_capture_config.h"
+#include "modules/video_capture/windows/help_functions_ds.h"
+#include "modules/video_capture/windows/sink_filter_ds.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+VideoCaptureDS::VideoCaptureDS()
+    : _captureFilter(NULL),
+      _graphBuilder(NULL),
+      _mediaControl(NULL),
+      _inputSendPin(NULL),
+      _outputCapturePin(NULL),
+      _dvFilter(NULL),
+      _inputDvPin(NULL),
+      _outputDvPin(NULL) {}
+
+VideoCaptureDS::~VideoCaptureDS() {
+  if (_mediaControl) {
+    _mediaControl->Stop();
+  }
+  if (_graphBuilder) {
+    if (sink_filter_)
+      _graphBuilder->RemoveFilter(sink_filter_.get());
+    if (_captureFilter)
+      _graphBuilder->RemoveFilter(_captureFilter);
+    if (_dvFilter)
+      _graphBuilder->RemoveFilter(_dvFilter);
+  }
+  RELEASE_AND_CLEAR(_inputSendPin);
+  RELEASE_AND_CLEAR(_outputCapturePin);
+
+  RELEASE_AND_CLEAR(_captureFilter);  // release the capture device
+  RELEASE_AND_CLEAR(_dvFilter);
+
+  RELEASE_AND_CLEAR(_mediaControl);
+
+  RELEASE_AND_CLEAR(_inputDvPin);
+  RELEASE_AND_CLEAR(_outputDvPin);
+
+  RELEASE_AND_CLEAR(_graphBuilder);
+}
+
+int32_t VideoCaptureDS::Init(const char* deviceUniqueIdUTF8) {
+  const int32_t nameLength = (int32_t)strlen((char*)deviceUniqueIdUTF8);
+  if (nameLength >= kVideoCaptureUniqueNameLength)
+    return -1;
+
+  // Store the device name
+  _deviceUniqueId = new (std::nothrow) char[nameLength + 1];
+  memcpy(_deviceUniqueId, deviceUniqueIdUTF8, nameLength + 1);
+
+  if (_dsInfo.Init() != 0)
+    return -1;
+
+  _captureFilter = _dsInfo.GetDeviceFilter(deviceUniqueIdUTF8);
+  if (!_captureFilter) {
+    RTC_LOG(LS_INFO) << "Failed to create capture filter.";
+    return -1;
+  }
+
+  // Get the interface for DirectShow's GraphBuilder
+  HRESULT hr = CoCreateInstance(CLSID_FilterGraph, NULL, CLSCTX_INPROC_SERVER,
+                                IID_IGraphBuilder, (void**)&_graphBuilder);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO) << "Failed to create graph builder.";
+    return -1;
+  }
+
+  hr = _graphBuilder->QueryInterface(IID_IMediaControl, (void**)&_mediaControl);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO) << "Failed to create media control builder.";
+    return -1;
+  }
+  hr = _graphBuilder->AddFilter(_captureFilter, CAPTURE_FILTER_NAME);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO) << "Failed to add the capture device to the graph.";
+    return -1;
+  }
+
+  _outputCapturePin = GetOutputPin(_captureFilter, PIN_CATEGORY_CAPTURE);
+  if (!_outputCapturePin) {
+    RTC_LOG(LS_INFO) << "Failed to get output capture pin";
+    return -1;
+  }
+
+  // Create the sink filte used for receiving Captured frames.
+  sink_filter_ = new ComRefCount<CaptureSinkFilter>(this);
+
+  hr = _graphBuilder->AddFilter(sink_filter_.get(), SINK_FILTER_NAME);
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO) << "Failed to add the send filter to the graph.";
+    return -1;
+  }
+
+  _inputSendPin = GetInputPin(sink_filter_.get());
+  if (!_inputSendPin) {
+    RTC_LOG(LS_INFO) << "Failed to get input send pin";
+    return -1;
+  }
+
+  if (SetCameraOutput(_requestedCapability) != 0) {
+    return -1;
+  }
+  RTC_LOG(LS_INFO) << "Capture device '" << deviceUniqueIdUTF8
+                   << "' initialized.";
+  return 0;
+}
+
+int32_t VideoCaptureDS::StartCapture(const VideoCaptureCapability& capability) {
+  if (capability != _requestedCapability) {
+    DisconnectGraph();
+
+    if (SetCameraOutput(capability) != 0) {
+      return -1;
+    }
+  }
+  HRESULT hr = _mediaControl->Pause();
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO)
+        << "Failed to Pause the Capture device. Is it already occupied? " << hr;
+    return -1;
+  }
+  hr = _mediaControl->Run();
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO) << "Failed to start the Capture device.";
+    return -1;
+  }
+  return 0;
+}
+
+int32_t VideoCaptureDS::StopCapture() {
+  HRESULT hr = _mediaControl->StopWhenReady();
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO) << "Failed to stop the capture graph. " << hr;
+    return -1;
+  }
+  return 0;
+}
+
+bool VideoCaptureDS::CaptureStarted() {
+  OAFilterState state = 0;
+  HRESULT hr = _mediaControl->GetState(1000, &state);
+  if (hr != S_OK && hr != VFW_S_CANT_CUE) {
+    RTC_LOG(LS_INFO) << "Failed to get the CaptureStarted status";
+  }
+  RTC_LOG(LS_INFO) << "CaptureStarted " << state;
+  return state == State_Running;
+}
+
+int32_t VideoCaptureDS::CaptureSettings(VideoCaptureCapability& settings) {
+  settings = _requestedCapability;
+  return 0;
+}
+
+int32_t VideoCaptureDS::SetCameraOutput(
+    const VideoCaptureCapability& requestedCapability) {
+  // Get the best matching capability
+  VideoCaptureCapability capability;
+  int32_t capabilityIndex;
+
+  // Store the new requested size
+  _requestedCapability = requestedCapability;
+  // Match the requested capability with the supported.
+  if ((capabilityIndex = _dsInfo.GetBestMatchedCapability(
+           _deviceUniqueId, _requestedCapability, capability)) < 0) {
+    return -1;
+  }
+  // Reduce the frame rate if possible.
+  if (capability.maxFPS > requestedCapability.maxFPS) {
+    capability.maxFPS = requestedCapability.maxFPS;
+  } else if (capability.maxFPS <= 0) {
+    capability.maxFPS = 30;
+  }
+
+  // Convert it to the windows capability index since they are not nexessary
+  // the same
+  VideoCaptureCapabilityWindows windowsCapability;
+  if (_dsInfo.GetWindowsCapability(capabilityIndex, windowsCapability) != 0) {
+    return -1;
+  }
+
+  IAMStreamConfig* streamConfig = NULL;
+  AM_MEDIA_TYPE* pmt = NULL;
+  VIDEO_STREAM_CONFIG_CAPS caps;
+
+  HRESULT hr = _outputCapturePin->QueryInterface(IID_IAMStreamConfig,
+                                                 (void**)&streamConfig);
+  if (hr) {
+    RTC_LOG(LS_INFO) << "Can't get the Capture format settings.";
+    return -1;
+  }
+
+  // Get the windows capability from the capture device
+  bool isDVCamera = false;
+  hr = streamConfig->GetStreamCaps(windowsCapability.directShowCapabilityIndex,
+                                   &pmt, reinterpret_cast<BYTE*>(&caps));
+  if (hr == S_OK) {
+    if (pmt->formattype == FORMAT_VideoInfo2) {
+      VIDEOINFOHEADER2* h = reinterpret_cast<VIDEOINFOHEADER2*>(pmt->pbFormat);
+      if (capability.maxFPS > 0 && windowsCapability.supportFrameRateControl) {
+        h->AvgTimePerFrame = REFERENCE_TIME(10000000.0 / capability.maxFPS);
+      }
+    } else {
+      VIDEOINFOHEADER* h = reinterpret_cast<VIDEOINFOHEADER*>(pmt->pbFormat);
+      if (capability.maxFPS > 0 && windowsCapability.supportFrameRateControl) {
+        h->AvgTimePerFrame = REFERENCE_TIME(10000000.0 / capability.maxFPS);
+      }
+    }
+
+    // Set the sink filter to request this capability
+    sink_filter_->SetRequestedCapability(capability);
+    // Order the capture device to use this capability
+    hr += streamConfig->SetFormat(pmt);
+
+    // Check if this is a DV camera and we need to add MS DV Filter
+    if (pmt->subtype == MEDIASUBTYPE_dvsl ||
+        pmt->subtype == MEDIASUBTYPE_dvsd ||
+        pmt->subtype == MEDIASUBTYPE_dvhd) {
+      isDVCamera = true;  // This is a DV camera. Use MS DV filter
+    }
+
+    FreeMediaType(pmt);
+    pmt = NULL;
+  }
+  RELEASE_AND_CLEAR(streamConfig);
+
+  if (FAILED(hr)) {
+    RTC_LOG(LS_INFO) << "Failed to set capture device output format";
+    return -1;
+  }
+
+  if (isDVCamera) {
+    hr = ConnectDVCamera();
+  } else {
+    hr = _graphBuilder->ConnectDirect(_outputCapturePin, _inputSendPin, NULL);
+  }
+  if (hr != S_OK) {
+    RTC_LOG(LS_INFO) << "Failed to connect the Capture graph " << hr;
+    return -1;
+  }
+  return 0;
+}
+
+int32_t VideoCaptureDS::DisconnectGraph() {
+  HRESULT hr = _mediaControl->Stop();
+  hr += _graphBuilder->Disconnect(_outputCapturePin);
+  hr += _graphBuilder->Disconnect(_inputSendPin);
+
+  // if the DV camera filter exist
+  if (_dvFilter) {
+    _graphBuilder->Disconnect(_inputDvPin);
+    _graphBuilder->Disconnect(_outputDvPin);
+  }
+  if (hr != S_OK) {
+    RTC_LOG(LS_ERROR)
+        << "Failed to Stop the Capture device for reconfiguration " << hr;
+    return -1;
+  }
+  return 0;
+}
+
+HRESULT VideoCaptureDS::ConnectDVCamera() {
+  HRESULT hr = S_OK;
+
+  if (!_dvFilter) {
+    hr = CoCreateInstance(CLSID_DVVideoCodec, NULL, CLSCTX_INPROC,
+                          IID_IBaseFilter, (void**)&_dvFilter);
+    if (hr != S_OK) {
+      RTC_LOG(LS_INFO) << "Failed to create the dv decoder: " << hr;
+      return hr;
+    }
+    hr = _graphBuilder->AddFilter(_dvFilter, L"VideoDecoderDV");
+    if (hr != S_OK) {
+      RTC_LOG(LS_INFO) << "Failed to add the dv decoder to the graph: " << hr;
+      return hr;
+    }
+    _inputDvPin = GetInputPin(_dvFilter);
+    if (_inputDvPin == NULL) {
+      RTC_LOG(LS_INFO) << "Failed to get input pin from DV decoder";
+      return -1;
+    }
+    _outputDvPin = GetOutputPin(_dvFilter, GUID_NULL);
+    if (_outputDvPin == NULL) {
+      RTC_LOG(LS_INFO) << "Failed to get output pin from DV decoder";
+      return -1;
+    }
+  }
+  hr = _graphBuilder->ConnectDirect(_outputCapturePin, _inputDvPin, NULL);
+  if (hr != S_OK) {
+    RTC_LOG(LS_INFO) << "Failed to connect capture device to the dv devoder: "
+                     << hr;
+    return hr;
+  }
+
+  hr = _graphBuilder->ConnectDirect(_outputDvPin, _inputSendPin, NULL);
+  if (hr != S_OK) {
+    if (hr == HRESULT_FROM_WIN32(ERROR_TOO_MANY_OPEN_FILES)) {
+      RTC_LOG(LS_INFO) << "Failed to connect the capture device, busy";
+    } else {
+      RTC_LOG(LS_INFO) << "Failed to connect capture device to the send graph: "
+                       << hr;
+    }
+  }
+  return hr;
+}
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_capture/windows/video_capture_ds.h b/third_party/libwebrtc/modules/video_capture/windows/video_capture_ds.h
new file mode 100644
index 0000000000..0f01cfaf67
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/windows/video_capture_ds.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_DS_H_
+#define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_DS_H_
+
+#include "api/scoped_refptr.h"
+#include "modules/video_capture/video_capture_impl.h"
+#include "modules/video_capture/windows/device_info_ds.h"
+
+#define CAPTURE_FILTER_NAME L"VideoCaptureFilter"
+#define SINK_FILTER_NAME L"SinkFilter"
+
+namespace webrtc {
+namespace videocapturemodule {
+// Forward declaraion
+class CaptureSinkFilter;
+
+class VideoCaptureDS : public VideoCaptureImpl {
+ public:
+  VideoCaptureDS();
+
+  virtual int32_t Init(const char* deviceUniqueIdUTF8);
+
+  /*************************************************************************
+   *
+   *   Start/Stop
+   *
+   *************************************************************************/
+  int32_t StartCapture(const VideoCaptureCapability& capability) override;
+  int32_t StopCapture() override;
+
+  /**************************************************************************
+   *
+   *   Properties of the set device
+   *
+   **************************************************************************/
+
+  bool CaptureStarted() override;
+  int32_t CaptureSettings(VideoCaptureCapability& settings) override;
+
+ protected:
+  ~VideoCaptureDS() override;
+
+  // Help functions
+
+  int32_t SetCameraOutput(const VideoCaptureCapability& requestedCapability);
+  int32_t DisconnectGraph();
+  HRESULT ConnectDVCamera();
+
+  DeviceInfoDS _dsInfo;
+
+  IBaseFilter* _captureFilter;
+  IGraphBuilder* _graphBuilder;
+  IMediaControl* _mediaControl;
+  rtc::scoped_refptr<CaptureSinkFilter> sink_filter_;
+  IPin* _inputSendPin;
+  IPin* _outputCapturePin;
+
+  // Microsoft DV interface (external DV cameras)
+  IBaseFilter* _dvFilter;
+  IPin* _inputDvPin;
+  IPin* _outputDvPin;
+};
+}  // namespace videocapturemodule
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_DS_H_
diff --git a/third_party/libwebrtc/modules/video_capture/windows/video_capture_factory_windows.cc b/third_party/libwebrtc/modules/video_capture/windows/video_capture_factory_windows.cc
new file mode 100644
index 0000000000..481326c1d2
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_capture/windows/video_capture_factory_windows.cc
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/scoped_refptr.h"
+#include "modules/video_capture/windows/video_capture_ds.h"
+
+namespace webrtc {
+namespace videocapturemodule {
+
+// static
+VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo() {
+  // TODO(tommi): Use the Media Foundation version on Vista and up.
+  return DeviceInfoDS::Create();
+}
+
+rtc::scoped_refptr<VideoCaptureModule> VideoCaptureImpl::Create(
+    const char* device_id) {
+  if (device_id == nullptr)
+    return nullptr;
+
+  // TODO(tommi): Use Media Foundation implementation for Vista and up.
+  auto capture = rtc::make_ref_counted<VideoCaptureDS>();
+  if (capture->Init(device_id) != 0) {
+    return nullptr;
+  }
+
+  return capture;
+}
+
+}  // namespace videocapturemodule
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/BUILD.gn b/third_party/libwebrtc/modules/video_coding/BUILD.gn
new file mode 100644
index 0000000000..e1b5e4ba84
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/BUILD.gn
@@ -0,0 +1,1329 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//third_party/libaom/options.gni")
+import("../../webrtc.gni")
+
+rtc_library("encoded_frame") {
+  visibility = [ "*" ]
+  sources = [
+    "encoded_frame.cc",
+    "encoded_frame.h",
+  ]
+  deps = [
+    ":codec_globals_headers",
+    ":video_codec_interface",
+    "../../api/video:encoded_image",
+    "../../api/video:video_frame",
+    "../../api/video:video_rtp_headers",
+    "../../modules:module_api_public",
+    "../../modules/rtp_rtcp:rtp_video_header",
+    "../../rtc_base:checks",
+    "../../rtc_base/experiments:alr_experiment",
+    "../../rtc_base/experiments:rtt_mult_experiment",
+    "../../rtc_base/system:rtc_export",
+    "../../system_wrappers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/types:optional",
+    "//third_party/abseil-cpp/absl/types:variant",
+  ]
+}
+
+rtc_library("chain_diff_calculator") {
+  sources = [
+    "chain_diff_calculator.cc",
+    "chain_diff_calculator.h",
+  ]
+
+  deps = [
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("frame_dependencies_calculator") {
+  sources = [
+    "frame_dependencies_calculator.cc",
+    "frame_dependencies_calculator.h",
+  ]
+
+  deps = [
+    "../../api:array_view",
+    "../../common_video/generic_frame_descriptor",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("nack_requester") {
+  sources = [
+    "histogram.cc",
+    "histogram.h",
+    "nack_requester.cc",
+    "nack_requester.h",
+  ]
+
+  deps = [
+    "..:module_api",
+    "../../api:field_trials_view",
+    "../../api:sequence_checker",
+    "../../api/task_queue",
+    "../../api/task_queue:pending_task_safety_flag",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/task_utils:repeating_task",
+    "../../system_wrappers",
+  ]
+}
+
+rtc_library("packet_buffer") {
+  sources = [
+    "packet_buffer.cc",
+    "packet_buffer.h",
+  ]
+  deps = [
+    ":codec_globals_headers",
+    "../../api:array_view",
+    "../../api:rtp_packet_info",
+    "../../api/units:timestamp",
+    "../../api/video:encoded_image",
+    "../../api/video:video_frame_type",
+    "../../common_video",
+    "../../rtc_base:checks",
+    "../../rtc_base:copy_on_write_buffer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:mod_ops",
+    "../../rtc_base:rtc_numerics",
+    "../rtp_rtcp:rtp_rtcp_format",
+    "../rtp_rtcp:rtp_video_header",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/types:variant",
+  ]
+}
+
+rtc_library("h264_packet_buffer") {
+  sources = [
+    "h264_packet_buffer.cc",
+    "h264_packet_buffer.h",
+  ]
+  deps = [
+    ":codec_globals_headers",
+    ":packet_buffer",
+    "../../api:array_view",
+    "../../api:rtp_packet_info",
+    "../../api/units:timestamp",
+    "../../api/video:encoded_image",
+    "../../api/video:video_frame_type",
+    "../../common_video",
+    "../../rtc_base:checks",
+    "../../rtc_base:copy_on_write_buffer",
+    "../../rtc_base:logging",
+    "../../rtc_base:rtc_numerics",
+    "../rtp_rtcp:rtp_rtcp_format",
+    "../rtp_rtcp:rtp_video_header",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("frame_helpers") {
+  sources = [
+    "frame_helpers.cc",
+    "frame_helpers.h",
+  ]
+  deps = [
+    "../../api/video:encoded_frame",
+    "../../rtc_base:logging",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/container:inlined_vector" ]
+}
+
+rtc_library("frame_buffer2") {
+  sources = [
+    "frame_buffer2.cc",
+    "frame_buffer2.h",
+  ]
+  deps = [
+    ":frame_helpers",
+    ":video_codec_interface",
+    ":video_coding_utility",
+    "../../api:field_trials_view",
+    "../../api:sequence_checker",
+    "../../api/task_queue",
+    "../../api/units:data_size",
+    "../../api/units:time_delta",
+    "../../api/video:encoded_frame",
+    "../../api/video:encoded_image",
+    "../../api/video:video_rtp_headers",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rtc_event",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/experiments:rtt_mult_experiment",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:no_unique_address",
+    "../../rtc_base/task_utils:repeating_task",
+    "../../system_wrappers",
+    "timing:inter_frame_delay",
+    "timing:jitter_estimator",
+    "timing:timing_module",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/container:inlined_vector" ]
+}
+
+rtc_library("video_coding") {
+  visibility = [ "*" ]
+  sources = [
+    "decoder_database.cc",
+    "decoder_database.h",
+    "fec_controller_default.cc",
+    "fec_controller_default.h",
+    "fec_rate_table.h",
+    "frame_object.cc",
+    "frame_object.h",
+    "generic_decoder.cc",
+    "generic_decoder.h",
+    "h264_sprop_parameter_sets.cc",
+    "h264_sprop_parameter_sets.h",
+    "h264_sps_pps_tracker.cc",
+    "h264_sps_pps_tracker.h",
+    "include/video_codec_initializer.h",
+    "internal_defines.h",
+    "loss_notification_controller.cc",
+    "loss_notification_controller.h",
+    "media_opt_util.cc",
+    "media_opt_util.h",
+    "rtp_frame_id_only_ref_finder.cc",
+    "rtp_frame_id_only_ref_finder.h",
+    "rtp_frame_reference_finder.cc",
+    "rtp_frame_reference_finder.h",
+    "rtp_generic_ref_finder.cc",
+    "rtp_generic_ref_finder.h",
+    "rtp_seq_num_only_ref_finder.cc",
+    "rtp_seq_num_only_ref_finder.h",
+    "rtp_vp8_ref_finder.cc",
+    "rtp_vp8_ref_finder.h",
+    "rtp_vp9_ref_finder.cc",
+    "rtp_vp9_ref_finder.h",
+    "video_codec_initializer.cc",
+    "video_receiver2.cc",
+    "video_receiver2.h",
+  ]
+  if (build_with_mozilla) {
+    sources += [
+      "event_wrapper.cc",
+      "event_wrapper.h",
+    ]
+  }
+
+  deps = [
+    ":codec_globals_headers",
+    ":encoded_frame",
+    ":frame_helpers",
+    ":video_codec_interface",
+    ":video_coding_utility",
+    ":webrtc_vp8_scalability",
+    ":webrtc_vp9_helpers",
+    "..:module_api",
+    "..:module_api_public",
+    "..:module_fec_api",
+    "../../api:array_view",
+    "../../api:fec_controller_api",
+    "../../api:field_trials_view",
+    "../../api:rtp_headers",
+    "../../api:rtp_packet_info",
+    "../../api:scoped_refptr",
+    "../../api:sequence_checker",
+    "../../api/task_queue",
+    "../../api/units:data_rate",
+    "../../api/units:data_size",
+    "../../api/units:frequency",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../api/video:builtin_video_bitrate_allocator_factory",
+    "../../api/video:encoded_frame",
+    "../../api/video:encoded_image",
+    "../../api/video:video_adaptation",
+    "../../api/video:video_adaptation",
+    "../../api/video:video_bitrate_allocation",
+    "../../api/video:video_bitrate_allocator",
+    "../../api/video:video_bitrate_allocator_factory",
+    "../../api/video:video_frame",
+    "../../api/video:video_frame_type",
+    "../../api/video:video_rtp_headers",
+    "../../api/video_codecs:video_codecs_api",
+    "../../common_video",
+    "../../rtc_base:checks",
+    "../../rtc_base:copy_on_write_buffer",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rtc_event",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base:threading",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/experiments:alr_experiment",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/experiments:min_video_bitrate_experiment",
+    "../../rtc_base/experiments:rate_control_settings",
+    "../../rtc_base/experiments:rtt_mult_experiment",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:no_unique_address",
+    "../../rtc_base/task_utils:repeating_task",
+    "../../rtc_base/third_party/base64",
+    "../../system_wrappers",
+    "../../system_wrappers:field_trial",
+    "../../system_wrappers:metrics",
+    "../../video/config:encoder_config",
+    "../rtp_rtcp",
+    "../rtp_rtcp:rtp_rtcp_format",
+    "../rtp_rtcp:rtp_video_header",
+    "codecs/av1:av1_svc_config",
+    "svc:scalability_mode_util",
+    "timing:inter_frame_delay",
+    "timing:jitter_estimator",
+    "timing:rtt_filter",
+    "timing:timing_module",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
+    "//third_party/abseil-cpp/absl/types:optional",
+    "//third_party/abseil-cpp/absl/types:variant",
+  ]
+}
+
+rtc_library("video_codec_interface") {
+  visibility = [ "*" ]
+  sources = [
+    "include/video_codec_interface.cc",
+    "include/video_codec_interface.h",
+    "include/video_coding_defines.h",
+    "include/video_error_codes.h",
+    "video_coding_defines.cc",
+  ]
+  deps = [
+    ":codec_globals_headers",
+    "../../api/video:video_frame",
+    "../../api/video:video_rtp_headers",
+    "../../api/video_codecs:scalability_mode",
+    "../../api/video_codecs:video_codecs_api",
+    "../../common_video",
+    "../../common_video/generic_frame_descriptor",
+    "../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("video_coding_legacy") {
+  visibility = [ ":video_coding_unittests" ]
+  sources = [
+    "decoding_state.cc",
+    "decoding_state.h",
+    "event_wrapper.cc",
+    "event_wrapper.h",
+    "frame_buffer.cc",
+    "frame_buffer.h",
+    "include/video_coding.h",
+    "jitter_buffer.cc",
+    "jitter_buffer.h",
+    "jitter_buffer_common.h",
+    "packet.cc",
+    "packet.h",
+    "receiver.cc",
+    "receiver.h",
+    "session_info.cc",
+    "session_info.h",
+    "video_coding_impl.cc",
+    "video_coding_impl.h",
+    "video_receiver.cc",
+  ]
+  deps = [
+    ":codec_globals_headers",
+    ":encoded_frame",
+    ":video_codec_interface",
+    ":video_coding",
+    "..:module_api",
+    "..:module_api_public",
+    "../../api:field_trials_view",
+    "../../api:rtp_headers",
+    "../../api:rtp_packet_info",
+    "../../api:sequence_checker",
+    "../../api/transport:field_trial_based_config",
+    "../../api/units:timestamp",
+    "../../api/video:encoded_image",
+    "../../api/video:video_frame",
+    "../../api/video:video_frame_type",
+    "../../api/video:video_rtp_headers",
+    "../../api/video_codecs:video_codecs_api",
+    "../../common_video",
+    "../../modules/rtp_rtcp:rtp_video_header",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:one_time_event",
+    "../../rtc_base:rtc_event",
+    "../../rtc_base:safe_conversions",
+    "../../rtc_base/memory:always_valid_pointer",
+    "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers",
+    "../rtp_rtcp:rtp_rtcp_format",
+    "../rtp_rtcp:rtp_video_header",
+    "timing:inter_frame_delay",
+    "timing:jitter_estimator",
+    "timing:timing_module",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/types:optional",
+    "//third_party/abseil-cpp/absl/types:variant",
+  ]
+}
+
+rtc_source_set("codec_globals_headers") {
+  visibility = [ "*" ]
+  sources = [
+    "codecs/h264/include/h264_globals.h",
+    "codecs/interface/common_constants.h",
+    "codecs/vp8/include/vp8_globals.h",
+    "codecs/vp9/include/vp9_globals.h",
+  ]
+
+  deps = [ "../../rtc_base:checks" ]
+}
+
+rtc_library("video_coding_utility") {
+  visibility = [ "*" ]
+  sources = [
+    "utility/bandwidth_quality_scaler.cc",
+    "utility/bandwidth_quality_scaler.h",
+    "utility/decoded_frames_history.cc",
+    "utility/decoded_frames_history.h",
+    "utility/frame_dropper.cc",
+    "utility/frame_dropper.h",
+    "utility/framerate_controller_deprecated.cc",
+    "utility/framerate_controller_deprecated.h",
+    "utility/ivf_defines.h",
+    "utility/ivf_file_reader.cc",
+    "utility/ivf_file_reader.h",
+    "utility/ivf_file_writer.cc",
+    "utility/ivf_file_writer.h",
+    "utility/qp_parser.cc",
+    "utility/qp_parser.h",
+    "utility/quality_scaler.cc",
+    "utility/quality_scaler.h",
+    "utility/simulcast_rate_allocator.cc",
+    "utility/simulcast_rate_allocator.h",
+    "utility/simulcast_utility.cc",
+    "utility/simulcast_utility.h",
+    "utility/vp8_constants.h",
+    "utility/vp8_header_parser.cc",
+    "utility/vp8_header_parser.h",
+    "utility/vp9_constants.h",
+    "utility/vp9_uncompressed_header_parser.cc",
+    "utility/vp9_uncompressed_header_parser.h",
+  ]
+
+  deps = [
+    ":video_codec_interface",
+    "../../api:array_view",
+    "../../api:field_trials_view",
+    "../../api:scoped_refptr",
+    "../../api:sequence_checker",
+    "../../api/units:time_delta",
+    "../../api/video:encoded_frame",
+    "../../api/video:encoded_image",
+    "../../api/video:video_adaptation",
+    "../../api/video:video_bitrate_allocation",
+    "../../api/video:video_bitrate_allocator",
+    "../../api/video:video_codec_constants",
+    "../../api/video:video_frame",
+    "../../api/video_codecs:video_codecs_api",
+    "../../common_video",
+    "../../modules/rtp_rtcp",
+    "../../rtc_base:bitstream_reader",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:rate_statistics",
+    "../../rtc_base:refcount",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:stringutils",
+    "../../rtc_base:timeutils",
+    "../../rtc_base:weak_ptr",
+    "../../rtc_base/experiments:bandwidth_quality_scaler_settings",
+    "../../rtc_base/experiments:encoder_info_settings",
+    "../../rtc_base/experiments:quality_scaler_settings",
+    "../../rtc_base/experiments:quality_scaling_experiment",
+    "../../rtc_base/experiments:rate_control_settings",
+    "../../rtc_base/experiments:stable_target_rate_experiment",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:arch",
+    "../../rtc_base/system:file_wrapper",
+    "../../rtc_base/system:no_unique_address",
+    "../../rtc_base/task_utils:repeating_task",
+    "../../system_wrappers:field_trial",
+    "../rtp_rtcp:rtp_rtcp_format",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/numeric:bits",
+    "//third_party/abseil-cpp/absl/strings:strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("webrtc_h264") {
+  visibility = [ "*" ]
+  sources = [
+    "codecs/h264/h264.cc",
+    "codecs/h264/h264_color_space.cc",
+    "codecs/h264/h264_color_space.h",
+    "codecs/h264/h264_decoder_impl.cc",
+    "codecs/h264/h264_decoder_impl.h",
+    "codecs/h264/h264_encoder_impl.cc",
+    "codecs/h264/h264_encoder_impl.h",
+    "codecs/h264/include/h264.h",
+  ]
+
+  defines = []
+  deps = [
+    ":video_codec_interface",
+    ":video_coding_utility",
+    "../../api/transport/rtp:dependency_descriptor",
+    "../../api/video:video_codec_constants",
+    "../../api/video:video_frame",
+    "../../api/video:video_frame_i010",
+    "../../api/video:video_rtp_headers",
+    "../../api/video_codecs:scalability_mode",
+    "../../api/video_codecs:video_codecs_api",
+    "../../common_video",
+    "../../media:codec",
+    "../../media:media_constants",
+    "../../media:rtc_media_base",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/system:rtc_export",
+    "../../system_wrappers:field_trial",
+    "../../system_wrappers:metrics",
+    "svc:scalability_structures",
+    "svc:scalable_video_controller",
+    "//third_party/libyuv",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+
+  if (rtc_use_h264) {
+    deps += [
+      "//third_party/ffmpeg",
+      "//third_party/openh264:encoder",
+    ]
+    if (!build_with_mozilla) {
+      deps += [ "../../media:rtc_media_base" ]
+    }
+  }
+}
+
+rtc_library("webrtc_multiplex") {
+  sources = [
+    "codecs/multiplex/augmented_video_frame_buffer.cc",
+    "codecs/multiplex/include/augmented_video_frame_buffer.h",
+    "codecs/multiplex/include/multiplex_decoder_adapter.h",
+    "codecs/multiplex/include/multiplex_encoder_adapter.h",
+    "codecs/multiplex/multiplex_decoder_adapter.cc",
+    "codecs/multiplex/multiplex_encoded_image_packer.cc",
+    "codecs/multiplex/multiplex_encoded_image_packer.h",
+    "codecs/multiplex/multiplex_encoder_adapter.cc",
+  ]
+
+  deps = [
+    ":video_codec_interface",
+    ":video_coding_utility",
+    "../../api:fec_controller_api",
+    "../../api:scoped_refptr",
+    "../../api/video:encoded_image",
+    "../../api/video:video_frame",
+    "../../api/video:video_rtp_headers",
+    "../../api/video_codecs:video_codecs_api",
+    "../../common_video",
+    "../../media:rtc_media_base",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base/synchronization:mutex",
+    "../rtp_rtcp:rtp_rtcp_format",
+  ]
+}
+
+# This target defines a bare-bones interface towards libvpx, used by the
+# VP8 and VP9 wrappers below.
+rtc_library("webrtc_libvpx_interface") {
+  visibility = [ "*" ]
+  sources = [
+    "codecs/interface/libvpx_interface.cc",
+    "codecs/interface/libvpx_interface.h",
+  ]
+  deps = [ "../../rtc_base:checks" ]
+  if (rtc_build_libvpx) {
+    deps += [ rtc_libvpx_dir ]
+  }
+}
+
+rtc_library("mock_libvpx_interface") {
+  testonly = true
+  sources = [ "codecs/interface/mock_libvpx_interface.h" ]
+  deps = [
+    ":webrtc_libvpx_interface",
+    "../../test:test_support",
+  ]
+}
+
+# This target includes the internal SW codec.
+rtc_library("webrtc_vp8") {
+  visibility = [ "*" ]
+  poisonous = [ "software_video_codecs" ]
+  sources = [
+    "codecs/vp8/include/vp8.h",
+    "codecs/vp8/libvpx_vp8_decoder.cc",
+    "codecs/vp8/libvpx_vp8_decoder.h",
+    "codecs/vp8/libvpx_vp8_encoder.cc",
+    "codecs/vp8/libvpx_vp8_encoder.h",
+  ]
+
+  deps = [
+    ":codec_globals_headers",
+    ":video_codec_interface",
+    ":video_coding_utility",
+    ":webrtc_libvpx_interface",
+    ":webrtc_vp8_scalability",
+    ":webrtc_vp8_temporal_layers",
+    "../../api:fec_controller_api",
+    "../../api:scoped_refptr",
+    "../../api/video:encoded_image",
+    "../../api/video:video_frame",
+    "../../api/video:video_rtp_headers",
+    "../../api/video_codecs:scalability_mode",
+    "../../api/video_codecs:video_codecs_api",
+    "../../api/video_codecs:vp8_temporal_layers_factory",
+    "../../common_video",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/experiments:cpu_speed_experiment",
+    "../../rtc_base/experiments:encoder_info_settings",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/experiments:rate_control_settings",
+    "../../system_wrappers:field_trial",
+    "../../system_wrappers:metrics",
+    "svc:scalability_mode_util",
+    "//third_party/libyuv",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings:strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  if (rtc_build_libvpx) {
+    deps += [ rtc_libvpx_dir ]
+  }
+}
+
+rtc_source_set("webrtc_vp8_scalability") {
+  sources = [
+    "codecs/vp8/vp8_scalability.cc",
+    "codecs/vp8/vp8_scalability.h",
+  ]
+  deps = [ "../../api/video_codecs:scalability_mode" ]
+}
+
+rtc_library("webrtc_vp8_temporal_layers") {
+  visibility = [ "*" ]
+  sources = [
+    "codecs/vp8/default_temporal_layers.cc",
+    "codecs/vp8/default_temporal_layers.h",
+    "codecs/vp8/include/temporal_layers_checker.h",
+    "codecs/vp8/screenshare_layers.cc",
+    "codecs/vp8/screenshare_layers.h",
+    "codecs/vp8/temporal_layers.h",
+    "codecs/vp8/temporal_layers_checker.cc",
+  ]
+
+  deps = [
+    ":codec_globals_headers",
+    ":video_codec_interface",
+    ":video_coding_utility",
+    "../../api:fec_controller_api",
+    "../../api/video_codecs:video_codecs_api",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rate_statistics",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:timeutils",
+    "../../system_wrappers:field_trial",
+    "../../system_wrappers:metrics",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+# This target includes VP9 files that may be used for any VP9 codec, internal SW or external HW.
+rtc_library("webrtc_vp9_helpers") {
+  sources = [
+    "codecs/vp9/svc_config.cc",
+    "codecs/vp9/svc_config.h",
+  ]
+
+  deps = [
+    ":codec_globals_headers",
+    ":video_codec_interface",
+    "../../api/video:video_bitrate_allocation",
+    "../../api/video:video_bitrate_allocator",
+    "../../api/video:video_codec_constants",
+    "../../api/video_codecs:video_codecs_api",
+    "../../common_video",
+    "../../media:rtc_media_base",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base/experiments:stable_target_rate_experiment",
+    "svc:scalability_mode_util",
+    "svc:scalability_structures",
+    "svc:scalable_video_controller",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/container:inlined_vector" ]
+}
+
+rtc_library("webrtc_vp9") {
+  visibility = [ "*" ]
+  poisonous = [ "software_video_codecs" ]
+  sources = [
+    "codecs/vp9/include/vp9.h",
+    "codecs/vp9/libvpx_vp9_decoder.cc",
+    "codecs/vp9/libvpx_vp9_decoder.h",
+    "codecs/vp9/libvpx_vp9_encoder.cc",
+    "codecs/vp9/libvpx_vp9_encoder.h",
+    "codecs/vp9/vp9.cc",
+    "codecs/vp9/vp9_frame_buffer_pool.cc",
+    "codecs/vp9/vp9_frame_buffer_pool.h",
+  ]
+
+  deps = [
+    ":video_codec_interface",
+    ":video_coding_utility",
+    ":webrtc_libvpx_interface",
+    ":webrtc_vp9_helpers",
+    "../../api:fec_controller_api",
+    "../../api:field_trials_view",
+    "../../api:refcountedbase",
+    "../../api:scoped_refptr",
+    "../../api/transport:field_trial_based_config",
+    "../../api/video:video_frame",
+    "../../api/video:video_frame_i010",
+    "../../api/video:video_rtp_headers",
+    "../../api/video_codecs:scalability_mode",
+    "../../api/video_codecs:video_codecs_api",
+    "../../common_video",
+    "../../media:codec",
+    "../../media:rtc_media_base",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:logging",
+    "../../rtc_base:stringutils",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/containers:flat_map",
+    "../../rtc_base/experiments:encoder_info_settings",
+    "../../rtc_base/experiments:field_trial_parser",
+    "../../rtc_base/experiments:rate_control_settings",
+    "../../rtc_base/synchronization:mutex",
+    "../../system_wrappers:field_trial",
+    "../rtp_rtcp:rtp_rtcp_format",
+    "svc:scalability_mode_util",
+    "svc:scalability_structures",
+    "svc:scalable_video_controller",
+    "svc:svc_rate_allocator",
+    "//third_party/libyuv",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings:strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  if (rtc_build_libvpx) {
+    deps += [ rtc_libvpx_dir ]
+  }
+}
+
+if (rtc_include_tests) {
+  if (is_android) {
+    rtc_library("android_codec_factory_helper") {
+      sources = [
+        "codecs/test/android_codec_factory_helper.cc",
+        "codecs/test/android_codec_factory_helper.h",
+      ]
+
+      deps = [
+        "../../api/video_codecs:video_codecs_api",
+        "../../modules/utility:utility",
+        "../../rtc_base:checks",
+        "../../rtc_base:ignore_wundef",
+        "../../sdk/android:internal_jni",
+        "../../sdk/android:native_api_base",
+        "../../sdk/android:native_api_codecs",
+        "../../sdk/android:native_api_jni",
+      ]
+    }
+  }
+
+  if (is_ios || is_mac) {
+    rtc_library("objc_codec_factory_helper") {
+      sources = [
+        "codecs/test/objc_codec_factory_helper.h",
+        "codecs/test/objc_codec_factory_helper.mm",
+      ]
+
+      deps = [
+        "../../api/video:video_frame",
+        "../../api/video:video_rtp_headers",
+        "../../api/video_codecs:video_codecs_api",
+        "../../media:rtc_audio_video",
+        "../../media:rtc_media_base",
+        "../../sdk:native_api",
+        "../../sdk:peerconnectionfactory_base_objc",
+        "../../sdk:videocodec_objc",
+        "../../sdk:videosource_objc",
+        "../../sdk:videotoolbox_objc",
+      ]
+    }
+  }
+
+  rtc_library("encoded_video_frame_producer") {
+    testonly = true
+    sources = [
+      "codecs/test/encoded_video_frame_producer.cc",
+      "codecs/test/encoded_video_frame_producer.h",
+    ]
+    deps = [
+      ":video_codec_interface",
+      "../../api:create_frame_generator",
+      "../../api:frame_generator_api",
+      "../../api/transport/rtp:dependency_descriptor",
+      "../../api/video:encoded_image",
+      "../../api/video:video_frame",
+      "../../api/video:video_frame_type",
+      "../../api/video_codecs:video_codecs_api",
+      "../../rtc_base:checks",
+    ]
+  }
+
+  rtc_library("simulcast_test_fixture_impl") {
+    testonly = true
+    sources = [
+      "utility/simulcast_test_fixture_impl.cc",
+      "utility/simulcast_test_fixture_impl.h",
+    ]
+
+    deps = [
+      ":video_codec_interface",
+      ":video_coding",
+      ":video_coding_utility",
+      "../../api:mock_video_decoder",
+      "../../api:mock_video_encoder",
+      "../../api:simulcast_test_fixture_api",
+      "../../api/video:encoded_image",
+      "../../api/video:video_frame",
+      "../../api/video:video_rtp_headers",
+      "../../api/video_codecs:video_codecs_api",
+      "../../common_video",
+      "../../rtc_base:checks",
+      "../../test:test_support",
+    ]
+  }
+
+  rtc_library("video_codecs_test_framework") {
+    testonly = true
+    sources = [
+      "codecs/test/video_codec_analyzer.cc",
+      "codecs/test/video_codec_analyzer.h",
+      "codecs/test/video_codec_unittest.cc",
+      "codecs/test/video_codec_unittest.h",
+      "codecs/test/videoprocessor.cc",
+      "codecs/test/videoprocessor.h",
+    ]
+
+    deps = [
+      ":codec_globals_headers",
+      ":video_codec_interface",
+      ":video_coding",
+      ":video_coding_utility",
+      ":videocodec_test_stats_impl",
+      ":webrtc_vp9_helpers",
+      "..:module_api",
+      "../../api:create_frame_generator",
+      "../../api:frame_generator_api",
+      "../../api:scoped_refptr",
+      "../../api:sequence_checker",
+      "../../api:video_codec_tester_api",
+      "../../api:videocodec_test_fixture_api",
+      "../../api/task_queue",
+      "../../api/task_queue:default_task_queue_factory",
+      "../../api/video:builtin_video_bitrate_allocator_factory",
+      "../../api/video:encoded_image",
+      "../../api/video:resolution",
+      "../../api/video:video_bitrate_allocation",
+      "../../api/video:video_bitrate_allocator",
+      "../../api/video:video_bitrate_allocator_factory",
+      "../../api/video:video_codec_constants",
+      "../../api/video:video_frame",
+      "../../api/video:video_rtp_headers",
+      "../../api/video_codecs:video_codecs_api",
+      "../../common_video",
+      "../../rtc_base:buffer",
+      "../../rtc_base:checks",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:task_queue_for_test",
+      "../../rtc_base:timeutils",
+      "../../rtc_base/synchronization:mutex",
+      "../../rtc_base/system:no_unique_address",
+      "../../test:test_support",
+      "../../test:video_test_common",
+      "../../test:video_test_support",
+      "../rtp_rtcp:rtp_rtcp_format",
+      "//third_party/libyuv",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
+  video_coding_modules_tests_resources = []
+  if (is_android) {
+    video_coding_modules_tests_resources += [
+      "../../resources/foreman_128x96.yuv",
+      "../../resources/foreman_160x120.yuv",
+      "../../resources/foreman_176x144.yuv",
+      "../../resources/foreman_240x136.yuv",
+      "../../resources/foreman_320x240.yuv",
+      "../../resources/foreman_480x272.yuv",
+    ]
+  }
+  if (!is_android) {
+    video_coding_modules_tests_resources += [
+      "../../resources/ConferenceMotion_1280_720_50.yuv",
+      "../../resources/FourPeople_1280x720_30.yuv",
+    ]
+  }
+
+  num_video_coding_modules_tests_resources = 0
+  foreach(i, video_coding_modules_tests_resources) {
+    num_video_coding_modules_tests_resources += 1
+  }
+
+  if (num_video_coding_modules_tests_resources > 0) {
+    if (is_ios || is_mac) {
+      bundle_data("video_coding_modules_tests_resources_bundle_data") {
+        testonly = true
+        sources = video_coding_modules_tests_resources
+        outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+      }
+    }
+  }
+
+  rtc_library("videocodec_test_impl") {
+    testonly = true
+    sources = [
+      "codecs/test/video_codec_tester_impl.cc",
+      "codecs/test/video_codec_tester_impl.h",
+      "codecs/test/videocodec_test_fixture_impl.cc",
+      "codecs/test/videocodec_test_fixture_impl.h",
+    ]
+    deps = [
+      ":codec_globals_headers",
+      ":video_codec_interface",
+      ":video_codecs_test_framework",
+      ":video_coding_utility",
+      ":videocodec_test_stats_impl",
+      ":webrtc_vp9_helpers",
+      "../../api:array_view",
+      "../../api:video_codec_tester_api",
+      "../../api:videocodec_test_fixture_api",
+      "../../api/task_queue:default_task_queue_factory",
+      "../../api/task_queue:task_queue",
+      "../../api/test/metrics:global_metrics_logger_and_exporter",
+      "../../api/test/metrics:metric",
+      "../../api/test/video:function_video_factory",
+      "../../api/transport:field_trial_based_config",
+      "../../api/units:frequency",
+      "../../api/units:time_delta",
+      "../../api/units:timestamp",
+      "../../api/video:encoded_image",
+      "../../api/video:video_bitrate_allocation",
+      "../../api/video:video_frame",
+      "../../api/video_codecs:video_codecs_api",
+      "../../api/video_codecs:video_decoder_factory_template",
+      "../../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
+      "../../api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
+      "../../api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
+      "../../api/video_codecs:video_decoder_factory_template_open_h264_adapter",
+      "../../api/video_codecs:video_encoder_factory_template",
+      "../../api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
+      "../../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
+      "../../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
+      "../../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
+      "../../call:video_stream_api",
+      "../../common_video",
+      "../../media:media_constants",
+      "../../media:rtc_audio_video",
+      "../../media:rtc_media_base",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:rtc_base_tests_utils",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:stringutils",
+      "../../rtc_base:task_queue_for_test",
+      "../../rtc_base:timeutils",
+      "../../system_wrappers",
+      "../../test:fileutils",
+      "../../test:test_support",
+      "../../test:video_test_common",
+      "../../test:video_test_support",
+      "../../video/config:encoder_config",
+      "../../video/config:streams_config",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings:strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+
+  rtc_library("videocodec_test_stats_impl") {
+    testonly = true
+    sources = [
+      "codecs/test/videocodec_test_stats_impl.cc",
+      "codecs/test/videocodec_test_stats_impl.h",
+    ]
+    deps = [
+      "../../api:videocodec_test_stats_api",
+      "../../api/numerics",
+      "../../rtc_base:checks",
+      "../../rtc_base:rtc_numerics",
+      "../../rtc_base:stringutils",
+      "../../test:test_common",
+      "../rtp_rtcp:rtp_rtcp_format",
+    ]
+  }
+
+  rtc_library("video_coding_modules_tests") {
+    testonly = true
+    defines = []
+
+    sources = [
+      "codecs/h264/test/h264_impl_unittest.cc",
+      "codecs/multiplex/test/multiplex_adapter_unittest.cc",
+      "codecs/test/video_codec_test.cc",
+      "codecs/test/video_encoder_decoder_instantiation_tests.cc",
+      "codecs/test/videocodec_test_av1.cc",
+      "codecs/test/videocodec_test_libvpx.cc",
+      "codecs/vp8/test/vp8_impl_unittest.cc",
+    ]
+
+    if (rtc_libvpx_build_vp9) {
+      sources += [ "codecs/vp9/test/vp9_impl_unittest.cc" ]
+    }
+
+    if (rtc_use_h264) {
+      sources += [ "codecs/test/videocodec_test_openh264.cc" ]
+    }
+
+    deps = [
+      ":encoded_video_frame_producer",
+      ":mock_libvpx_interface",
+      ":video_codec_interface",
+      ":video_codecs_test_framework",
+      ":video_coding_utility",
+      ":videocodec_test_impl",
+      ":webrtc_h264",
+      ":webrtc_libvpx_interface",
+      ":webrtc_multiplex",
+      ":webrtc_vp8",
+      ":webrtc_vp9",
+      ":webrtc_vp9_helpers",
+      "../../api:create_frame_generator",
+      "../../api:create_video_codec_tester_api",
+      "../../api:create_videocodec_test_fixture_api",
+      "../../api:frame_generator_api",
+      "../../api:mock_video_codec_factory",
+      "../../api:mock_video_decoder",
+      "../../api:mock_video_encoder",
+      "../../api:scoped_refptr",
+      "../../api:video_codec_tester_api",
+      "../../api:videocodec_test_fixture_api",
+      "../../api:videocodec_test_stats_api",
+      "../../api/test/video:function_video_factory",
+      "../../api/units:data_rate",
+      "../../api/units:frequency",
+      "../../api/video:encoded_image",
+      "../../api/video:resolution",
+      "../../api/video:video_frame",
+      "../../api/video:video_rtp_headers",
+      "../../api/video_codecs:builtin_video_decoder_factory",
+      "../../api/video_codecs:builtin_video_encoder_factory",
+      "../../api/video_codecs:rtc_software_fallback_wrappers",
+      "../../api/video_codecs:scalability_mode",
+      "../../api/video_codecs:video_codecs_api",
+      "../../common_video",
+      "../../common_video/test:utilities",
+      "../../media:codec",
+      "../../media:media_constants",
+      "../../media:rtc_internal_video_codecs",
+      "../../media:rtc_media_base",
+      "../../media:rtc_simulcast_encoder_adapter",
+      "../../rtc_base:refcount",
+      "../../rtc_base:stringutils",
+      "../../rtc_base:timeutils",
+      "../../test:explicit_key_value_config",
+      "../../test:field_trial",
+      "../../test:fileutils",
+      "../../test:test_support",
+      "../../test:video_test_common",
+      "../../test:video_test_support",
+      "../rtp_rtcp:rtp_rtcp_format",
+      "codecs/av1:dav1d_decoder",
+      "svc:scalability_mode_util",
+      "//third_party/libyuv",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/functional:any_invocable",
+      "//third_party/abseil-cpp/absl/memory",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+
+    data = video_coding_modules_tests_resources
+
+    if (is_android) {
+      sources += [ "codecs/test/videocodec_test_mediacodec.cc" ]
+
+      deps += [
+        ":android_codec_factory_helper",
+        "../../rtc_base:stringutils",
+      ]
+    }
+
+    if (is_ios || is_mac) {
+      sources += [ "codecs/test/videocodec_test_videotoolbox.cc" ]
+
+      deps += [ ":objc_codec_factory_helper" ]
+
+      if (num_video_coding_modules_tests_resources > 0) {
+        deps += [ ":video_coding_modules_tests_resources_bundle_data" ]
+      }
+    }
+
+    if (rtc_build_libvpx) {
+      deps += [ rtc_libvpx_dir ]
+    }
+  }
+
+  rtc_library("video_coding_unittests") {
+    testonly = true
+
+    sources = [
+      "chain_diff_calculator_unittest.cc",
+      "codecs/test/video_codec_analyzer_unittest.cc",
+      "codecs/test/video_codec_tester_impl_unittest.cc",
+      "codecs/test/videocodec_test_fixture_config_unittest.cc",
+      "codecs/test/videocodec_test_stats_impl_unittest.cc",
+      "codecs/test/videoprocessor_unittest.cc",
+      "codecs/vp8/default_temporal_layers_unittest.cc",
+      "codecs/vp8/libvpx_vp8_simulcast_test.cc",
+      "codecs/vp8/screenshare_layers_unittest.cc",
+      "codecs/vp9/svc_config_unittest.cc",
+      "decoder_database_unittest.cc",
+      "decoding_state_unittest.cc",
+      "fec_controller_unittest.cc",
+      "frame_buffer2_unittest.cc",
+      "frame_dependencies_calculator_unittest.cc",
+      "frame_helpers_unittest.cc",
+      "generic_decoder_unittest.cc",
+      "h264_packet_buffer_unittest.cc",
+      "h264_sprop_parameter_sets_unittest.cc",
+      "h264_sps_pps_tracker_unittest.cc",
+      "histogram_unittest.cc",
+      "jitter_buffer_unittest.cc",
+      "loss_notification_controller_unittest.cc",
+      "nack_requester_unittest.cc",
+      "packet_buffer_unittest.cc",
+      "receiver_unittest.cc",
+      "rtp_frame_reference_finder_unittest.cc",
+      "rtp_vp8_ref_finder_unittest.cc",
+      "rtp_vp9_ref_finder_unittest.cc",
+      "session_info_unittest.cc",
+      "test/stream_generator.cc",
+      "test/stream_generator.h",
+      "utility/bandwidth_quality_scaler_unittest.cc",
+      "utility/decoded_frames_history_unittest.cc",
+      "utility/frame_dropper_unittest.cc",
+      "utility/framerate_controller_deprecated_unittest.cc",
+      "utility/ivf_file_reader_unittest.cc",
+      "utility/ivf_file_writer_unittest.cc",
+      "utility/qp_parser_unittest.cc",
+      "utility/quality_scaler_unittest.cc",
+      "utility/simulcast_rate_allocator_unittest.cc",
+      "utility/vp9_uncompressed_header_parser_unittest.cc",
+      "video_codec_initializer_unittest.cc",
+      "video_receiver2_unittest.cc",
+      "video_receiver_unittest.cc",
+    ]
+    if (rtc_use_h264) {
+      sources += [
+        "codecs/h264/h264_encoder_impl_unittest.cc",
+        "codecs/h264/h264_simulcast_unittest.cc",
+      ]
+    }
+
+    deps = [
+      ":chain_diff_calculator",
+      ":codec_globals_headers",
+      ":encoded_frame",
+      ":frame_buffer2",
+      ":frame_dependencies_calculator",
+      ":frame_helpers",
+      ":h264_packet_buffer",
+      ":nack_requester",
+      ":packet_buffer",
+      ":simulcast_test_fixture_impl",
+      ":video_codec_interface",
+      ":video_codecs_test_framework",
+      ":video_coding",
+      ":video_coding_legacy",
+      ":video_coding_utility",
+      ":videocodec_test_impl",
+      ":videocodec_test_stats_impl",
+      ":webrtc_h264",
+      ":webrtc_vp8",
+      ":webrtc_vp8_temporal_layers",
+      ":webrtc_vp9",
+      ":webrtc_vp9_helpers",
+      "..:module_fec_api",
+      "../../api:array_view",
+      "../../api:create_simulcast_test_fixture_api",
+      "../../api:fec_controller_api",
+      "../../api:mock_fec_controller_override",
+      "../../api:mock_video_decoder",
+      "../../api:mock_video_encoder",
+      "../../api:rtp_packet_info",
+      "../../api:scoped_refptr",
+      "../../api:simulcast_test_fixture_api",
+      "../../api:video_codec_tester_api",
+      "../../api:videocodec_test_fixture_api",
+      "../../api/task_queue",
+      "../../api/task_queue:default_task_queue_factory",
+      "../../api/task_queue/test:mock_task_queue_base",
+      "../../api/test/video:function_video_factory",
+      "../../api/units:data_size",
+      "../../api/units:frequency",
+      "../../api/units:time_delta",
+      "../../api/units:timestamp",
+      "../../api/video:builtin_video_bitrate_allocator_factory",
+      "../../api/video:encoded_frame",
+      "../../api/video:encoded_image",
+      "../../api/video:render_resolution",
+      "../../api/video:video_adaptation",
+      "../../api/video:video_bitrate_allocation",
+      "../../api/video:video_bitrate_allocator",
+      "../../api/video:video_bitrate_allocator_factory",
+      "../../api/video:video_frame",
+      "../../api/video:video_frame_type",
+      "../../api/video:video_rtp_headers",
+      "../../api/video_codecs:video_codecs_api",
+      "../../api/video_codecs:vp8_temporal_layers_factory",
+      "../../common_video",
+      "../../common_video/generic_frame_descriptor",
+      "../../common_video/test:utilities",
+      "../../media:media_constants",
+      "../../media:rtc_media_base",
+      "../../rtc_base:checks",
+      "../../rtc_base:gunit_helpers",
+      "../../rtc_base:histogram_percentile_counter",
+      "../../rtc_base:platform_thread",
+      "../../rtc_base:random",
+      "../../rtc_base:refcount",
+      "../../rtc_base:rtc_base_tests_utils",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:rtc_numerics",
+      "../../rtc_base:stringutils",
+      "../../rtc_base:task_queue_for_test",
+      "../../rtc_base:timeutils",
+      "../../rtc_base/experiments:encoder_info_settings",
+      "../../rtc_base/synchronization:mutex",
+      "../../rtc_base/system:unused",
+      "../../system_wrappers",
+      "../../system_wrappers:field_trial",
+      "../../system_wrappers:metrics",
+      "../../test:fake_encoded_frame",
+      "../../test:fake_video_codecs",
+      "../../test:field_trial",
+      "../../test:fileutils",
+      "../../test:run_loop",
+      "../../test:scoped_key_value_config",
+      "../../test:test_support",
+      "../../test:video_test_common",
+      "../../test:video_test_support",
+      "../../test/time_controller:time_controller",
+      "../../third_party/libyuv:libyuv",
+      "../rtp_rtcp:rtp_rtcp_format",
+      "../rtp_rtcp:rtp_video_header",
+      "codecs/av1:video_coding_codecs_av1_tests",
+      "svc:scalability_structure_tests",
+      "svc:svc_rate_allocator_tests",
+      "timing:jitter_estimator",
+      "timing:timing_module",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/memory",
+      "//third_party/abseil-cpp/absl/types:optional",
+      "//third_party/abseil-cpp/absl/types:variant",
+    ]
+    if (rtc_build_libvpx) {
+      deps += [ rtc_libvpx_dir ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/video_coding/DEPS b/third_party/libwebrtc/modules/video_coding/DEPS
new file mode 100644
index 0000000000..d62707c2f9
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/DEPS
@@ -0,0 +1,25 @@
+include_rules = [
+  "+vpx",
+  "+call",
+  "+common_video",
+  "+sdk",
+  "+system_wrappers",
+  "+rtc_tools",
+  "+third_party/libyuv",
+  "+rtc_base/system/rtc_export.h",
+  "+video/config",
+]
+
+specific_include_rules = {
+  "android_codec_factory_helper\.cc": [
+    "+base/android",
+  ],
+  "multiplex_encoder_adapter\.cc": [
+    "+media/base",
+  ],
+  ".*test.*\.cc": [
+     "+media/base",
+     "+media/engine",
+     "+video/config",
+  ],
+}
diff --git a/third_party/libwebrtc/modules/video_coding/OWNERS b/third_party/libwebrtc/modules/video_coding/OWNERS
new file mode 100644
index 0000000000..2e4d968c98
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/OWNERS
@@ -0,0 +1,7 @@
+asapersson@webrtc.org
+brandtr@webrtc.org
+ilnik@webrtc.org
+marpan@webrtc.org
+philipel@webrtc.org
+sprang@webrtc.org
+stefan@webrtc.org
diff --git a/third_party/libwebrtc/modules/video_coding/chain_diff_calculator.cc b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator.cc
new file mode 100644
index 0000000000..5f852717b5
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/chain_diff_calculator.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <algorithm>
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "absl/types/optional.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+void ChainDiffCalculator::Reset(const std::vector<bool>& chains) {
+  last_frame_in_chain_.resize(chains.size());
+  for (size_t i = 0; i < chains.size(); ++i) {
+    if (chains[i]) {
+      last_frame_in_chain_[i] = absl::nullopt;
+    }
+  }
+}
+
+absl::InlinedVector<int, 4> ChainDiffCalculator::ChainDiffs(
+    int64_t frame_id) const {
+  absl::InlinedVector<int, 4> result;
+  result.reserve(last_frame_in_chain_.size());
+  for (const auto& frame_id_in_chain : last_frame_in_chain_) {
+    result.push_back(frame_id_in_chain ? (frame_id - *frame_id_in_chain) : 0);
+  }
+  return result;
+}
+
+absl::InlinedVector<int, 4> ChainDiffCalculator::From(
+    int64_t frame_id,
+    const std::vector<bool>& chains) {
+  auto result = ChainDiffs(frame_id);
+  if (chains.size() != last_frame_in_chain_.size()) {
+    RTC_LOG(LS_ERROR) << "Insconsistent chain configuration for frame#"
+                      << frame_id << ": expected "
+                      << last_frame_in_chain_.size() << " chains, found "
+                      << chains.size();
+  }
+  size_t num_chains = std::min(last_frame_in_chain_.size(), chains.size());
+  for (size_t i = 0; i < num_chains; ++i) {
+    if (chains[i]) {
+      last_frame_in_chain_[i] = frame_id;
+    }
+  }
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/chain_diff_calculator.h b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator.h
new file mode 100644
index 0000000000..bca7340c6f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CHAIN_DIFF_CALCULATOR_H_
+#define MODULES_VIDEO_CODING_CHAIN_DIFF_CALCULATOR_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+// This class is thread compatible.
+class ChainDiffCalculator {
+ public:
+  ChainDiffCalculator() = default;
+  ChainDiffCalculator(const ChainDiffCalculator&) = default;
+  ChainDiffCalculator& operator=(const ChainDiffCalculator&) = default;
+
+  // Restarts chains, i.e. for position where chains[i] == true next chain_diff
+  // will be 0. Saves chains.size() as number of chains in the stream.
+  void Reset(const std::vector<bool>& chains);
+
+  // Returns chain diffs based on flags if frame is part of the chain.
+  absl::InlinedVector<int, 4> From(int64_t frame_id,
+                                   const std::vector<bool>& chains);
+
+ private:
+  absl::InlinedVector<int, 4> ChainDiffs(int64_t frame_id) const;
+
+  absl::InlinedVector<absl::optional<int64_t>, 4> last_frame_in_chain_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CHAIN_DIFF_CALCULATOR_H_
diff --git a/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_gn/moz.build
new file mode 100644
index 0000000000..9020f11067
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/chain_diff_calculator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("chain_diff_calculator_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_unittest.cc b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_unittest.cc
new file mode 100644
index 0000000000..efd09bd888
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/chain_diff_calculator_unittest.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/chain_diff_calculator.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::SizeIs;
+
+TEST(ChainDiffCalculatorTest, SingleChain) {
+  // Simulate  a stream with 2 temporal layer where chain
+  // protects temporal layer 0.
+  ChainDiffCalculator calculator;
+  // Key frame.
+  calculator.Reset({true});
+  EXPECT_THAT(calculator.From(1, {true}), ElementsAre(0));
+  // T1 delta frame.
+  EXPECT_THAT(calculator.From(2, {false}), ElementsAre(1));
+  // T0 delta frame.
+  EXPECT_THAT(calculator.From(3, {true}), ElementsAre(2));
+}
+
+TEST(ChainDiffCalculatorTest, TwoChainsFullSvc) {
+  // Simulate a full svc stream with 2 spatial and 2 temporal layers.
+  // chains are protecting temporal layers 0.
+  ChainDiffCalculator calculator;
+  // S0 Key frame.
+  calculator.Reset({true, true});
+  EXPECT_THAT(calculator.From(1, {true, true}), ElementsAre(0, 0));
+  // S1 Key frame.
+  EXPECT_THAT(calculator.From(2, {false, true}), ElementsAre(1, 1));
+  // S0T1 delta frame.
+  EXPECT_THAT(calculator.From(3, {false, false}), ElementsAre(2, 1));
+  // S1T1 delta frame.
+  EXPECT_THAT(calculator.From(4, {false, false}), ElementsAre(3, 2));
+  // S0T0 delta frame.
+  EXPECT_THAT(calculator.From(5, {true, true}), ElementsAre(4, 3));
+  // S1T0 delta frame.
+  EXPECT_THAT(calculator.From(6, {false, true}), ElementsAre(1, 1));
+}
+
+TEST(ChainDiffCalculatorTest, TwoChainsKSvc) {
+  // Simulate a k-svc stream with 2 spatial and 2 temporal layers.
+  // chains are protecting temporal layers 0.
+  ChainDiffCalculator calculator;
+  // S0 Key frame.
+  calculator.Reset({true, true});
+  EXPECT_THAT(calculator.From(1, {true, true}), ElementsAre(0, 0));
+  // S1 Key frame.
+  EXPECT_THAT(calculator.From(2, {false, true}), ElementsAre(1, 1));
+  // S0T1 delta frame.
+  EXPECT_THAT(calculator.From(3, {false, false}), ElementsAre(2, 1));
+  // S1T1 delta frame.
+  EXPECT_THAT(calculator.From(4, {false, false}), ElementsAre(3, 2));
+  // S0T0 delta frame.
+  EXPECT_THAT(calculator.From(5, {true, false}), ElementsAre(4, 3));
+  // S1T0 delta frame.
+  EXPECT_THAT(calculator.From(6, {false, true}), ElementsAre(1, 4));
+}
+
+TEST(ChainDiffCalculatorTest, TwoChainsSimulcast) {
+  // Simulate a k-svc stream with 2 spatial and 2 temporal layers.
+  // chains are protecting temporal layers 0.
+  ChainDiffCalculator calculator;
+  // S0 Key frame.
+  calculator.Reset({true, false});
+  EXPECT_THAT(calculator.From(1, {true, false}), ElementsAre(0, 0));
+  // S1 Key frame.
+  calculator.Reset({false, true});
+  EXPECT_THAT(calculator.From(2, {false, true}), ElementsAre(1, 0));
+  // S0T1 delta frame.
+  EXPECT_THAT(calculator.From(3, {false, false}), ElementsAre(2, 1));
+  // S1T1 delta frame.
+  EXPECT_THAT(calculator.From(4, {false, false}), ElementsAre(3, 2));
+  // S0T0 delta frame.
+  EXPECT_THAT(calculator.From(5, {true, false}), ElementsAre(4, 3));
+  // S1T0 delta frame.
+  EXPECT_THAT(calculator.From(6, {false, true}), ElementsAre(1, 4));
+}
+
+TEST(ChainDiffCalculatorTest, ResilentToAbsentChainConfig) {
+  ChainDiffCalculator calculator;
+  // Key frame.
+  calculator.Reset({true, false});
+  EXPECT_THAT(calculator.From(1, {true, false}), ElementsAre(0, 0));
+  // Forgot to set chains. should still return 2 chain_diffs.
+  EXPECT_THAT(calculator.From(2, {}), ElementsAre(1, 0));
+  // chain diffs for next frame(s) are undefined, but still there should be
+  // correct number of them.
+  EXPECT_THAT(calculator.From(3, {true, false}), SizeIs(2));
+  EXPECT_THAT(calculator.From(4, {false, true}), SizeIs(2));
+  // Since previous two frames updated all the chains, can expect what
+  // chain_diffs would be.
+  EXPECT_THAT(calculator.From(5, {false, false}), ElementsAre(2, 1));
+}
+
+TEST(ChainDiffCalculatorTest, ResilentToTooMainChains) {
+  ChainDiffCalculator calculator;
+  // Key frame.
+  calculator.Reset({true, false});
+  EXPECT_THAT(calculator.From(1, {true, false}), ElementsAre(0, 0));
+  // Set wrong number of chains. Expect number of chain_diffs is not changed.
+  EXPECT_THAT(calculator.From(2, {true, true, true}), ElementsAre(1, 0));
+  // chain diffs for next frame(s) are undefined, but still there should be
+  // correct number of them.
+  EXPECT_THAT(calculator.From(3, {true, false}), SizeIs(2));
+  EXPECT_THAT(calculator.From(4, {false, true}), SizeIs(2));
+  // Since previous two frames updated all the chains, can expect what
+  // chain_diffs would be.
+  EXPECT_THAT(calculator.From(5, {false, false}), ElementsAre(2, 1));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codec_globals_headers_gn/moz.build b/third_party/libwebrtc/modules/video_coding/codec_globals_headers_gn/moz.build
new file mode 100644
index 0000000000..7c87e340af
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codec_globals_headers_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("codec_globals_headers_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/BUILD.gn b/third_party/libwebrtc/modules/video_coding/codecs/av1/BUILD.gn
new file mode 100644
index 0000000000..610f958ad1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/BUILD.gn
@@ -0,0 +1,110 @@
+# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//third_party/libaom/options.gni")
+import("../../../../webrtc.gni")
+
+rtc_library("av1_svc_config") {
+  sources = [
+    "av1_svc_config.cc",
+    "av1_svc_config.h",
+  ]
+  deps = [
+    "../../../../api/video_codecs:video_codecs_api",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:logging",
+    "../../../../rtc_base:stringutils",
+    "../../svc:scalability_mode_util",
+    "../../svc:scalability_structures",
+    "../../svc:scalable_video_controller",
+  ]
+
+  absl_deps = [ "//third_party/abseil-cpp/absl/container:inlined_vector" ]
+}
+
+rtc_library("dav1d_decoder") {
+  visibility = [ "*" ]
+  poisonous = [ "software_video_codecs" ]
+  public = [ "dav1d_decoder.h" ]
+  sources = [ "dav1d_decoder.cc" ]
+
+  deps = [
+    "../..:video_codec_interface",
+    "../../../../api:scoped_refptr",
+    "../../../../api/video:encoded_image",
+    "../../../../api/video:video_frame",
+    "../../../../api/video_codecs:video_codecs_api",
+    "../../../../common_video",
+    "../../../../rtc_base:logging",
+    "//third_party/dav1d",
+    "//third_party/libyuv",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("libaom_av1_encoder") {
+  visibility = [ "*" ]
+  poisonous = [ "software_video_codecs" ]
+  public = [ "libaom_av1_encoder.h" ]
+  sources = [ "libaom_av1_encoder.cc" ]
+  deps = [
+    "../..:video_codec_interface",
+    "../../../../api:scoped_refptr",
+    "../../../../api/video:encoded_image",
+    "../../../../api/video:video_frame",
+    "../../../../api/video_codecs:scalability_mode",
+    "../../../../api/video_codecs:video_codecs_api",
+    "../../../../common_video",
+    "../../../../rtc_base:checks",
+    "../../../../rtc_base:logging",
+    "../../../../rtc_base:rtc_numerics",
+    "../../svc:scalability_structures",
+    "../../svc:scalable_video_controller",
+    "//third_party/libaom",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings:strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("video_coding_codecs_av1_tests") {
+    testonly = true
+
+    sources = [ "av1_svc_config_unittest.cc" ]
+    deps = [
+      ":av1_svc_config",
+      "../../../../api/video_codecs:video_codecs_api",
+      "../../../../test:test_support",
+    ]
+
+    if (enable_libaom) {
+      sources += [
+        "libaom_av1_encoder_unittest.cc",
+        "libaom_av1_unittest.cc",
+      ]
+      deps += [
+        ":dav1d_decoder",
+        ":libaom_av1_encoder",
+        "../..:encoded_video_frame_producer",
+        "../..:video_codec_interface",
+        "../../../../api:mock_video_encoder",
+        "../../../../api/units:data_size",
+        "../../../../api/units:time_delta",
+        "../../../../api/video:video_frame",
+        "../../svc:scalability_mode_util",
+        "../../svc:scalability_structures",
+        "../../svc:scalable_video_controller",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/DEPS b/third_party/libwebrtc/modules/video_coding/codecs/av1/DEPS
new file mode 100644
index 0000000000..bfb1c733d4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+  "+third_party/libaom",
+  "+third_party/dav1d",
+]
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config.cc b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config.cc
new file mode 100644
index 0000000000..43dcf96ab7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/av1/av1_svc_config.h"
+
+#include <algorithm>
+#include <cmath>
+#include <memory>
+
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace {
+absl::optional<ScalabilityMode> BuildScalabilityMode(int num_temporal_layers,
+                                                     int num_spatial_layers) {
+  char name[20];
+  rtc::SimpleStringBuilder ss(name);
+  ss << "L" << num_spatial_layers << "T" << num_temporal_layers;
+  if (num_spatial_layers > 1) {
+    ss << "_KEY";
+  }
+
+  return ScalabilityModeFromString(name);
+}
+}  // namespace
+
+absl::InlinedVector<ScalabilityMode, kScalabilityModeCount>
+LibaomAv1EncoderSupportedScalabilityModes() {
+  absl::InlinedVector<ScalabilityMode, kScalabilityModeCount> scalability_modes;
+  for (ScalabilityMode scalability_mode : kAllScalabilityModes) {
+    if (ScalabilityStructureConfig(scalability_mode) != absl::nullopt) {
+      scalability_modes.push_back(scalability_mode);
+    }
+  }
+  return scalability_modes;
+}
+
+bool LibaomAv1EncoderSupportsScalabilityMode(ScalabilityMode scalability_mode) {
+  // For libaom AV1, the scalability mode is supported if we can create the
+  // scalability structure.
+  return ScalabilityStructureConfig(scalability_mode) != absl::nullopt;
+}
+
+bool SetAv1SvcConfig(VideoCodec& video_codec,
+                     int num_temporal_layers,
+                     int num_spatial_layers) {
+  RTC_DCHECK_EQ(video_codec.codecType, kVideoCodecAV1);
+
+  absl::optional<ScalabilityMode> scalability_mode =
+      video_codec.GetScalabilityMode();
+  if (!scalability_mode.has_value()) {
+    scalability_mode =
+        BuildScalabilityMode(num_temporal_layers, num_spatial_layers);
+    if (!scalability_mode) {
+      RTC_LOG(LS_WARNING) << "Scalability mode is not set, using 'L1T1'.";
+      scalability_mode = ScalabilityMode::kL1T1;
+    }
+  }
+
+  std::unique_ptr<ScalableVideoController> structure =
+      CreateScalabilityStructure(*scalability_mode);
+  if (structure == nullptr) {
+    RTC_LOG(LS_WARNING) << "Failed to create structure "
+                        << static_cast<int>(*scalability_mode);
+    return false;
+  }
+
+  video_codec.SetScalabilityMode(*scalability_mode);
+
+  ScalableVideoController::StreamLayersConfig info = structure->StreamConfig();
+  for (int sl_idx = 0; sl_idx < info.num_spatial_layers; ++sl_idx) {
+    SpatialLayer& spatial_layer = video_codec.spatialLayers[sl_idx];
+    spatial_layer.width = video_codec.width * info.scaling_factor_num[sl_idx] /
+                          info.scaling_factor_den[sl_idx];
+    spatial_layer.height = video_codec.height *
+                           info.scaling_factor_num[sl_idx] /
+                           info.scaling_factor_den[sl_idx];
+    spatial_layer.maxFramerate = video_codec.maxFramerate;
+    spatial_layer.numberOfTemporalLayers = info.num_temporal_layers;
+    spatial_layer.active = true;
+  }
+
+  if (info.num_spatial_layers == 1) {
+    SpatialLayer& spatial_layer = video_codec.spatialLayers[0];
+    spatial_layer.minBitrate = video_codec.minBitrate;
+    spatial_layer.maxBitrate = video_codec.maxBitrate;
+    spatial_layer.targetBitrate =
+        (video_codec.minBitrate + video_codec.maxBitrate) / 2;
+    return true;
+  }
+
+  for (int sl_idx = 0; sl_idx < info.num_spatial_layers; ++sl_idx) {
+    SpatialLayer& spatial_layer = video_codec.spatialLayers[sl_idx];
+    // minBitrate and maxBitrate formulas are copied from vp9 settings and
+    // are not yet tuned for av1.
+    const int num_pixels = spatial_layer.width * spatial_layer.height;
+    int min_bitrate_kbps = (600.0 * std::sqrt(num_pixels) - 95'000.0) / 1000.0;
+    spatial_layer.minBitrate = std::max(min_bitrate_kbps, 20);
+    spatial_layer.maxBitrate = 50 + static_cast<int>(1.6 * num_pixels / 1000.0);
+    spatial_layer.targetBitrate =
+        (spatial_layer.minBitrate + spatial_layer.maxBitrate) / 2;
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config.h b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config.h
new file mode 100644
index 0000000000..05b886b9f4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config.h
@@ -0,0 +1,32 @@
+/* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_AV1_AV1_SVC_CONFIG_H_
+#define MODULES_VIDEO_CODING_CODECS_AV1_AV1_SVC_CONFIG_H_
+
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "api/video_codecs/video_codec.h"
+
+namespace webrtc {
+
+absl::InlinedVector<ScalabilityMode, kScalabilityModeCount>
+LibaomAv1EncoderSupportedScalabilityModes();
+
+bool LibaomAv1EncoderSupportsScalabilityMode(ScalabilityMode scalability_mode);
+
+// Fills `video_codec.spatialLayers` using other members.
+bool SetAv1SvcConfig(VideoCodec& video_codec,
+                     int num_temporal_layers,
+                     int num_spatial_layers);
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_AV1_AV1_SVC_CONFIG_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_gn/moz.build b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_gn/moz.build
new file mode 100644
index 0000000000..cc297017e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("av1_svc_config_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_unittest.cc
new file mode 100644
index 0000000000..9f1da9865c
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/av1_svc_config_unittest.cc
@@ -0,0 +1,171 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/av1/av1_svc_config.h"
+
+#include "api/video_codecs/video_codec.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+constexpr int kDontCare = 0;
+
+TEST(Av1SvcConfigTest, TreatsEmptyAsL1T1) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/kDontCare,
+                              /*num_spatial_layers=*/kDontCare));
+
+  EXPECT_TRUE(video_codec.spatialLayers[0].active);
+  EXPECT_EQ(video_codec.spatialLayers[0].numberOfTemporalLayers, 1);
+  EXPECT_FALSE(video_codec.spatialLayers[1].active);
+}
+
+TEST(Av1SvcConfigTest, ScalabilityModeFromNumberOfTemporalLayers) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/3,
+                              /*num_spatial_layers=*/1));
+  EXPECT_EQ(video_codec.spatialLayers[0].numberOfTemporalLayers, 3);
+}
+
+TEST(Av1SvcConfigTest, ScalabilityModeFromNumberOfSpatialLayers) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/3,
+                              /*num_spatial_layers=*/2));
+  EXPECT_EQ(video_codec.spatialLayers[0].numberOfTemporalLayers, 3);
+  EXPECT_TRUE(video_codec.spatialLayers[0].active);
+  EXPECT_TRUE(video_codec.spatialLayers[1].active);
+  EXPECT_FALSE(video_codec.spatialLayers[2].active);
+}
+
+TEST(Av1SvcConfigTest, SetsActiveSpatialLayersFromScalabilityMode) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+  video_codec.SetScalabilityMode(ScalabilityMode::kL2T1);
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/kDontCare,
+                              /*num_spatial_layers=*/kDontCare));
+
+  EXPECT_TRUE(video_codec.spatialLayers[0].active);
+  EXPECT_TRUE(video_codec.spatialLayers[1].active);
+  EXPECT_FALSE(video_codec.spatialLayers[2].active);
+}
+
+TEST(Av1SvcConfigTest, ConfiguresDobuleResolutionRatioFromScalabilityMode) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+  video_codec.SetScalabilityMode(ScalabilityMode::kL2T1);
+  video_codec.width = 1200;
+  video_codec.height = 800;
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/kDontCare,
+                              /*num_spatial_layers=*/kDontCare));
+
+  EXPECT_EQ(video_codec.spatialLayers[0].width, 600);
+  EXPECT_EQ(video_codec.spatialLayers[0].height, 400);
+  EXPECT_EQ(video_codec.spatialLayers[1].width, 1200);
+  EXPECT_EQ(video_codec.spatialLayers[1].height, 800);
+}
+
+TEST(Av1SvcConfigTest, ConfiguresSmallResolutionRatioFromScalabilityMode) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+  // h mode uses 1.5:1 ratio
+  video_codec.SetScalabilityMode(ScalabilityMode::kL2T1h);
+  video_codec.width = 1500;
+  video_codec.height = 900;
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/kDontCare,
+                              /*num_spatial_layers=*/kDontCare));
+
+  EXPECT_EQ(video_codec.spatialLayers[0].width, 1000);
+  EXPECT_EQ(video_codec.spatialLayers[0].height, 600);
+  EXPECT_EQ(video_codec.spatialLayers[1].width, 1500);
+  EXPECT_EQ(video_codec.spatialLayers[1].height, 900);
+}
+
+TEST(Av1SvcConfigTest, CopiesFramrate) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+  // h mode uses 1.5:1 ratio
+  video_codec.SetScalabilityMode(ScalabilityMode::kL2T1);
+  video_codec.maxFramerate = 27;
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/kDontCare,
+                              /*num_spatial_layers=*/kDontCare));
+
+  EXPECT_EQ(video_codec.spatialLayers[0].maxFramerate, 27);
+  EXPECT_EQ(video_codec.spatialLayers[1].maxFramerate, 27);
+}
+
+TEST(Av1SvcConfigTest, SetsNumberOfTemporalLayers) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+  video_codec.SetScalabilityMode(ScalabilityMode::kL1T3);
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/kDontCare,
+                              /*num_spatial_layers=*/kDontCare));
+
+  EXPECT_EQ(video_codec.spatialLayers[0].numberOfTemporalLayers, 3);
+}
+
+TEST(Av1SvcConfigTest, CopiesMinMaxBitrateForSingleSpatialLayer) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+  video_codec.SetScalabilityMode(ScalabilityMode::kL1T3);
+  video_codec.minBitrate = 100;
+  video_codec.maxBitrate = 500;
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/kDontCare,
+                              /*num_spatial_layers=*/kDontCare));
+
+  EXPECT_EQ(video_codec.spatialLayers[0].minBitrate, 100u);
+  EXPECT_EQ(video_codec.spatialLayers[0].maxBitrate, 500u);
+  EXPECT_LE(video_codec.spatialLayers[0].minBitrate,
+            video_codec.spatialLayers[0].targetBitrate);
+  EXPECT_LE(video_codec.spatialLayers[0].targetBitrate,
+            video_codec.spatialLayers[0].maxBitrate);
+}
+
+TEST(Av1SvcConfigTest, SetsBitratesForMultipleSpatialLayers) {
+  VideoCodec video_codec;
+  video_codec.codecType = kVideoCodecAV1;
+  video_codec.SetScalabilityMode(ScalabilityMode::kL3T3);
+
+  EXPECT_TRUE(SetAv1SvcConfig(video_codec, /*num_temporal_layers=*/kDontCare,
+                              /*num_spatial_layers=*/kDontCare));
+
+  EXPECT_GT(video_codec.spatialLayers[0].minBitrate, 0u);
+  EXPECT_LE(video_codec.spatialLayers[0].minBitrate,
+            video_codec.spatialLayers[0].targetBitrate);
+  EXPECT_LE(video_codec.spatialLayers[0].targetBitrate,
+            video_codec.spatialLayers[0].maxBitrate);
+
+  EXPECT_GT(video_codec.spatialLayers[1].minBitrate, 0u);
+  EXPECT_LE(video_codec.spatialLayers[1].minBitrate,
+            video_codec.spatialLayers[1].targetBitrate);
+  EXPECT_LE(video_codec.spatialLayers[1].targetBitrate,
+            video_codec.spatialLayers[1].maxBitrate);
+
+  EXPECT_GT(video_codec.spatialLayers[2].minBitrate, 0u);
+  EXPECT_LE(video_codec.spatialLayers[2].minBitrate,
+            video_codec.spatialLayers[2].targetBitrate);
+  EXPECT_LE(video_codec.spatialLayers[2].targetBitrate,
+            video_codec.spatialLayers[2].maxBitrate);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.cc b/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.cc
new file mode 100644
index 0000000000..a2cd6d868c
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.cc
@@ -0,0 +1,205 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/av1/dav1d_decoder.h"
+
+#include <algorithm>
+
+#include "api/scoped_refptr.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_frame_buffer.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/logging.h"
+#include "third_party/dav1d/libdav1d/include/dav1d/dav1d.h"
+#include "third_party/libyuv/include/libyuv/convert.h"
+#include "third_party/libyuv/include/libyuv/planar_functions.h"
+
+namespace webrtc {
+namespace {
+
+class Dav1dDecoder : public VideoDecoder {
+ public:
+  Dav1dDecoder();
+  Dav1dDecoder(const Dav1dDecoder&) = delete;
+  Dav1dDecoder& operator=(const Dav1dDecoder&) = delete;
+
+  ~Dav1dDecoder() override;
+
+  bool Configure(const Settings& settings) override;
+  int32_t Decode(const EncodedImage& encoded_image,
+                 bool missing_frames,
+                 int64_t render_time_ms) override;
+  int32_t RegisterDecodeCompleteCallback(
+      DecodedImageCallback* callback) override;
+  int32_t Release() override;
+  DecoderInfo GetDecoderInfo() const override;
+  const char* ImplementationName() const override;
+
+ private:
+  Dav1dContext* context_ = nullptr;
+  DecodedImageCallback* decode_complete_callback_ = nullptr;
+};
+
+class ScopedDav1dData {
+ public:
+  ~ScopedDav1dData() { dav1d_data_unref(&data_); }
+
+  Dav1dData& Data() { return data_; }
+
+ private:
+  Dav1dData data_ = {};
+};
+
+class ScopedDav1dPicture
+    : public rtc::RefCountedNonVirtual<ScopedDav1dPicture> {
+ public:
+  ~ScopedDav1dPicture() { dav1d_picture_unref(&picture_); }
+
+  Dav1dPicture& Picture() { return picture_; }
+  using rtc::RefCountedNonVirtual<ScopedDav1dPicture>::HasOneRef;
+
+ private:
+  Dav1dPicture picture_ = {};
+};
+
+constexpr char kDav1dName[] = "dav1d";
+
+// Calling `dav1d_data_wrap` requires a `free_callback` to be registered.
+void NullFreeCallback(const uint8_t* buffer, void* opaque) {}
+
+Dav1dDecoder::Dav1dDecoder() = default;
+
+Dav1dDecoder::~Dav1dDecoder() {
+  Release();
+}
+
+bool Dav1dDecoder::Configure(const Settings& settings) {
+  Dav1dSettings s;
+  dav1d_default_settings(&s);
+
+  s.n_threads = std::max(2, settings.number_of_cores());
+  s.max_frame_delay = 1;   // For low latency decoding.
+  s.all_layers = 0;        // Don't output a frame for every spatial layer.
+  s.operating_point = 31;  // Decode all operating points.
+
+  return dav1d_open(&context_, &s) == 0;
+}
+
+int32_t Dav1dDecoder::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* decode_complete_callback) {
+  decode_complete_callback_ = decode_complete_callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t Dav1dDecoder::Release() {
+  dav1d_close(&context_);
+  if (context_ != nullptr) {
+    return WEBRTC_VIDEO_CODEC_MEMORY;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+VideoDecoder::DecoderInfo Dav1dDecoder::GetDecoderInfo() const {
+  DecoderInfo info;
+  info.implementation_name = kDav1dName;
+  info.is_hardware_accelerated = false;
+  return info;
+}
+
+const char* Dav1dDecoder::ImplementationName() const {
+  return kDav1dName;
+}
+
+int32_t Dav1dDecoder::Decode(const EncodedImage& encoded_image,
+                             bool /*missing_frames*/,
+                             int64_t /*render_time_ms*/) {
+  if (!context_ || decode_complete_callback_ == nullptr) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  ScopedDav1dData scoped_dav1d_data;
+  Dav1dData& dav1d_data = scoped_dav1d_data.Data();
+  dav1d_data_wrap(&dav1d_data, encoded_image.data(), encoded_image.size(),
+                  /*free_callback=*/&NullFreeCallback,
+                  /*user_data=*/nullptr);
+
+  if (int decode_res = dav1d_send_data(context_, &dav1d_data)) {
+    RTC_LOG(LS_WARNING)
+        << "Dav1dDecoder::Decode decoding failed with error code "
+        << decode_res;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  rtc::scoped_refptr<ScopedDav1dPicture> scoped_dav1d_picture(
+      new ScopedDav1dPicture{});
+  Dav1dPicture& dav1d_picture = scoped_dav1d_picture->Picture();
+  if (int get_picture_res = dav1d_get_picture(context_, &dav1d_picture)) {
+    RTC_LOG(LS_WARNING)
+        << "Dav1dDecoder::Decode getting picture failed with error code "
+        << get_picture_res;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  if (dav1d_picture.p.bpc != 8) {
+    // Only accept 8 bit depth.
+    RTC_LOG(LS_ERROR) << "Dav1dDecoder::Decode unhandled bit depth: "
+                      << dav1d_picture.p.bpc;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  rtc::scoped_refptr<VideoFrameBuffer> wrapped_buffer;
+  if (dav1d_picture.p.layout == DAV1D_PIXEL_LAYOUT_I420) {
+    wrapped_buffer = WrapI420Buffer(
+        dav1d_picture.p.w, dav1d_picture.p.h,
+        static_cast<uint8_t*>(dav1d_picture.data[0]), dav1d_picture.stride[0],
+        static_cast<uint8_t*>(dav1d_picture.data[1]), dav1d_picture.stride[1],
+        static_cast<uint8_t*>(dav1d_picture.data[2]), dav1d_picture.stride[1],
+        // To keep |scoped_dav1d_picture.Picture()| alive
+        [scoped_dav1d_picture] {});
+  } else if (dav1d_picture.p.layout == DAV1D_PIXEL_LAYOUT_I444) {
+    wrapped_buffer = WrapI444Buffer(
+        dav1d_picture.p.w, dav1d_picture.p.h,
+        static_cast<uint8_t*>(dav1d_picture.data[0]), dav1d_picture.stride[0],
+        static_cast<uint8_t*>(dav1d_picture.data[1]), dav1d_picture.stride[1],
+        static_cast<uint8_t*>(dav1d_picture.data[2]), dav1d_picture.stride[1],
+        // To keep |scoped_dav1d_picture.Picture()| alive
+        [scoped_dav1d_picture] {});
+  } else {
+    // Only accept I420 or I444 pixel format.
+    RTC_LOG(LS_ERROR) << "Dav1dDecoder::Decode unhandled pixel layout: "
+                      << dav1d_picture.p.layout;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  if (!wrapped_buffer.get()) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  VideoFrame decoded_frame = VideoFrame::Builder()
+                                 .set_video_frame_buffer(wrapped_buffer)
+                                 .set_timestamp_rtp(encoded_image.Timestamp())
+                                 .set_ntp_time_ms(encoded_image.ntp_time_ms_)
+                                 .set_color_space(encoded_image.ColorSpace())
+                                 .build();
+
+  decode_complete_callback_->Decoded(decoded_frame, absl::nullopt,
+                                     absl::nullopt);
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+}  // namespace
+
+std::unique_ptr<VideoDecoder> CreateDav1dDecoder() {
+  return std::make_unique<Dav1dDecoder>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.h b/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.h
new file mode 100644
index 0000000000..c9396d1e03
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_CODECS_AV1_DAV1D_DECODER_H_
+#define MODULES_VIDEO_CODING_CODECS_AV1_DAV1D_DECODER_H_
+
+#include <memory>
+
+#include "api/video_codecs/video_decoder.h"
+
+namespace webrtc {
+
+std::unique_ptr<VideoDecoder> CreateDav1dDecoder();
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_AV1_DAV1D_DECODER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_encoder.cc b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_encoder.cc
new file mode 100644
index 0000000000..2713171616
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_encoder.cc
@@ -0,0 +1,825 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/codecs/av1/libaom_av1_encoder.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/base/macros.h"
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "modules/video_coding/svc/scalable_video_controller_no_layering.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "third_party/libaom/source/libaom/aom/aom_codec.h"
+#include "third_party/libaom/source/libaom/aom/aom_encoder.h"
+#include "third_party/libaom/source/libaom/aom/aomcx.h"
+
+#define SET_ENCODER_PARAM_OR_RETURN_ERROR(param_id, param_value) \
+  do {                                                           \
+    if (!SetEncoderControlParameters(param_id, param_value)) {   \
+      return WEBRTC_VIDEO_CODEC_ERROR;                           \
+    }                                                            \
+  } while (0)
+
+namespace webrtc {
+namespace {
+
+// Encoder configuration parameters
+constexpr int kQpMin = 10;
+constexpr int kUsageProfile = AOM_USAGE_REALTIME;
+constexpr int kMinQindex = 145;  // Min qindex threshold for QP scaling.
+constexpr int kMaxQindex = 205;  // Max qindex threshold for QP scaling.
+constexpr int kBitDepth = 8;
+constexpr int kLagInFrames = 0;  // No look ahead.
+constexpr int kRtpTicksPerSecond = 90000;
+constexpr float kMinimumFrameRate = 1.0;
+
+aom_superblock_size_t GetSuperblockSize(int width, int height, int threads) {
+  int resolution = width * height;
+  if (threads >= 4 && resolution >= 960 * 540 && resolution < 1920 * 1080)
+    return AOM_SUPERBLOCK_SIZE_64X64;
+  else
+    return AOM_SUPERBLOCK_SIZE_DYNAMIC;
+}
+
+class LibaomAv1Encoder final : public VideoEncoder {
+ public:
+  explicit LibaomAv1Encoder(
+      const absl::optional<LibaomAv1EncoderAuxConfig>& aux_config);
+  ~LibaomAv1Encoder();
+
+  int InitEncode(const VideoCodec* codec_settings,
+                 const Settings& settings) override;
+
+  int32_t RegisterEncodeCompleteCallback(
+      EncodedImageCallback* encoded_image_callback) override;
+
+  int32_t Release() override;
+
+  int32_t Encode(const VideoFrame& frame,
+                 const std::vector<VideoFrameType>* frame_types) override;
+
+  void SetRates(const RateControlParameters& parameters) override;
+
+  EncoderInfo GetEncoderInfo() const override;
+
+ private:
+  template <typename P>
+  bool SetEncoderControlParameters(int param_id, P param_value);
+
+  // Get value to be used for encoder cpu_speed setting
+  int GetCpuSpeed(int width, int height);
+
+  // Determine number of encoder threads to use.
+  int NumberOfThreads(int width, int height, int number_of_cores);
+
+  bool SvcEnabled() const { return svc_params_.has_value(); }
+  // Fills svc_params_ memeber value. Returns false on error.
+  bool SetSvcParams(ScalableVideoController::StreamLayersConfig svc_config);
+  // Configures the encoder with layer for the next frame.
+  void SetSvcLayerId(
+      const ScalableVideoController::LayerFrameConfig& layer_frame);
+  // Configures the encoder which buffers next frame updates and can reference.
+  void SetSvcRefFrameConfig(
+      const ScalableVideoController::LayerFrameConfig& layer_frame);
+  // If pixel format doesn't match, then reallocate.
+  void MaybeRewrapImgWithFormat(const aom_img_fmt_t fmt);
+
+  std::unique_ptr<ScalableVideoController> svc_controller_;
+  absl::optional<ScalabilityMode> scalability_mode_;
+  bool inited_;
+  bool rates_configured_;
+  absl::optional<aom_svc_params_t> svc_params_;
+  VideoCodec encoder_settings_;
+  absl::optional<LibaomAv1EncoderAuxConfig> aux_config_;
+  aom_image_t* frame_for_encode_;
+  aom_codec_ctx_t ctx_;
+  aom_codec_enc_cfg_t cfg_;
+  EncodedImageCallback* encoded_image_callback_;
+  SeqNumUnwrapper<uint32_t> rtp_timestamp_unwrapper_;
+};
+
+int32_t VerifyCodecSettings(const VideoCodec& codec_settings) {
+  if (codec_settings.width < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (codec_settings.height < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  // maxBitrate == 0 represents an unspecified maxBitRate.
+  if (codec_settings.maxBitrate > 0 &&
+      codec_settings.minBitrate > codec_settings.maxBitrate) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (codec_settings.maxBitrate > 0 &&
+      codec_settings.startBitrate > codec_settings.maxBitrate) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (codec_settings.startBitrate < codec_settings.minBitrate) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (codec_settings.maxFramerate < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+LibaomAv1Encoder::LibaomAv1Encoder(
+    const absl::optional<LibaomAv1EncoderAuxConfig>& aux_config)
+    : inited_(false),
+      rates_configured_(false),
+      aux_config_(aux_config),
+      frame_for_encode_(nullptr),
+      encoded_image_callback_(nullptr) {}
+
+LibaomAv1Encoder::~LibaomAv1Encoder() {
+  Release();
+}
+
+int LibaomAv1Encoder::InitEncode(const VideoCodec* codec_settings,
+                                 const Settings& settings) {
+  if (codec_settings == nullptr) {
+    RTC_LOG(LS_WARNING) << "No codec settings provided to "
+                           "LibaomAv1Encoder.";
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (settings.number_of_cores < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (inited_) {
+    RTC_LOG(LS_WARNING) << "Initing LibaomAv1Encoder without first releasing.";
+    Release();
+  }
+  encoder_settings_ = *codec_settings;
+
+  // Sanity checks for encoder configuration.
+  const int32_t result = VerifyCodecSettings(encoder_settings_);
+  if (result < 0) {
+    RTC_LOG(LS_WARNING) << "Incorrect codec settings provided to "
+                           "LibaomAv1Encoder.";
+    return result;
+  }
+  if (encoder_settings_.numberOfSimulcastStreams > 1) {
+    RTC_LOG(LS_WARNING) << "Simulcast is not implemented by LibaomAv1Encoder.";
+    return result;
+  }
+  scalability_mode_ = encoder_settings_.GetScalabilityMode();
+  if (!scalability_mode_.has_value()) {
+    RTC_LOG(LS_WARNING) << "Scalability mode is not set, using 'L1T1'.";
+    scalability_mode_ = ScalabilityMode::kL1T1;
+  }
+  svc_controller_ = CreateScalabilityStructure(*scalability_mode_);
+  if (svc_controller_ == nullptr) {
+    RTC_LOG(LS_WARNING) << "Failed to set scalability mode "
+                        << static_cast<int>(*scalability_mode_);
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  if (!SetSvcParams(svc_controller_->StreamConfig())) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  // Initialize encoder configuration structure with default values
+  aom_codec_err_t ret =
+      aom_codec_enc_config_default(aom_codec_av1_cx(), &cfg_, kUsageProfile);
+  if (ret != AOM_CODEC_OK) {
+    RTC_LOG(LS_WARNING) << "LibaomAv1Encoder::EncodeInit returned " << ret
+                        << " on aom_codec_enc_config_default.";
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  // Overwrite default config with input encoder settings & RTC-relevant values.
+  cfg_.g_w = encoder_settings_.width;
+  cfg_.g_h = encoder_settings_.height;
+  cfg_.g_threads =
+      NumberOfThreads(cfg_.g_w, cfg_.g_h, settings.number_of_cores);
+  cfg_.g_timebase.num = 1;
+  cfg_.g_timebase.den = kRtpTicksPerSecond;
+  cfg_.rc_target_bitrate = encoder_settings_.maxBitrate;  // kilobits/sec.
+  cfg_.g_input_bit_depth = kBitDepth;
+  cfg_.kf_mode = AOM_KF_DISABLED;
+  cfg_.rc_min_quantizer = kQpMin;
+  cfg_.rc_max_quantizer = encoder_settings_.qpMax;
+  cfg_.rc_undershoot_pct = 50;
+  cfg_.rc_overshoot_pct = 50;
+  cfg_.rc_buf_initial_sz = 600;
+  cfg_.rc_buf_optimal_sz = 600;
+  cfg_.rc_buf_sz = 1000;
+  cfg_.g_usage = kUsageProfile;
+  cfg_.g_error_resilient = 0;
+  // Low-latency settings.
+  cfg_.rc_end_usage = AOM_CBR;          // Constant Bit Rate (CBR) mode
+  cfg_.g_pass = AOM_RC_ONE_PASS;        // One-pass rate control
+  cfg_.g_lag_in_frames = kLagInFrames;  // No look ahead when lag equals 0.
+
+  if (frame_for_encode_ != nullptr) {
+    aom_img_free(frame_for_encode_);
+    frame_for_encode_ = nullptr;
+  }
+
+  // Flag options: AOM_CODEC_USE_PSNR and AOM_CODEC_USE_HIGHBITDEPTH
+  aom_codec_flags_t flags = 0;
+
+  // Initialize an encoder instance.
+  ret = aom_codec_enc_init(&ctx_, aom_codec_av1_cx(), &cfg_, flags);
+  if (ret != AOM_CODEC_OK) {
+    RTC_LOG(LS_WARNING) << "LibaomAv1Encoder::EncodeInit returned " << ret
+                        << " on aom_codec_enc_init.";
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  inited_ = true;
+
+  // Set control parameters
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AOME_SET_CPUUSED,
+                                    GetCpuSpeed(cfg_.g_w, cfg_.g_h));
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_CDEF, 1);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_TPL_MODEL, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_DELTAQ_MODE, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_ORDER_HINT, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_AQ_MODE, 3);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AOME_SET_MAX_INTRA_BITRATE_PCT, 300);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_COEFF_COST_UPD_FREQ, 3);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_MODE_COST_UPD_FREQ, 3);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_MV_COST_UPD_FREQ, 3);
+
+  if (codec_settings->mode == VideoCodecMode::kScreensharing) {
+    SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_TUNE_CONTENT,
+                                      AOM_CONTENT_SCREEN);
+    SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_PALETTE, 1);
+  } else {
+    SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_PALETTE, 0);
+  }
+
+  if (cfg_.g_threads == 4 && cfg_.g_w == 640 &&
+      (cfg_.g_h == 360 || cfg_.g_h == 480)) {
+    SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_TILE_ROWS,
+                                      static_cast<int>(log2(cfg_.g_threads)));
+  } else {
+    SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_TILE_COLUMNS,
+                                      static_cast<int>(log2(cfg_.g_threads)));
+  }
+
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ROW_MT, 1);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_OBMC, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_NOISE_SENSITIVITY, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_WARPED_MOTION, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_GLOBAL_MOTION, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_REF_FRAME_MVS, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(
+      AV1E_SET_SUPERBLOCK_SIZE,
+      GetSuperblockSize(cfg_.g_w, cfg_.g_h, cfg_.g_threads));
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_CFL_INTRA, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_SMOOTH_INTRA, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_ANGLE_DELTA, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_FILTER_INTRA, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_INTRA_DEFAULT_TX_ONLY, 1);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_DISABLE_TRELLIS_QUANT, 1);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_DIST_WTD_COMP, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_DIFF_WTD_COMP, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_DUAL_FILTER, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_INTERINTRA_COMP, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_INTERINTRA_WEDGE, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_INTRA_EDGE_FILTER, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_INTRABC, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_MASKED_COMP, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_PAETH_INTRA, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_QM, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_RECT_PARTITIONS, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_RESTORATION, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_SMOOTH_INTERINTRA, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ENABLE_TX64, 0);
+  SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_MAX_REFERENCE_FRAMES, 3);
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+template <typename P>
+bool LibaomAv1Encoder::SetEncoderControlParameters(int param_id,
+                                                   P param_value) {
+  aom_codec_err_t error_code = aom_codec_control(&ctx_, param_id, param_value);
+  if (error_code != AOM_CODEC_OK) {
+    RTC_LOG(LS_WARNING)
+        << "LibaomAv1Encoder::SetEncoderControlParameters returned "
+        << error_code << " on id:  " << param_id << ".";
+  }
+  return error_code == AOM_CODEC_OK;
+}
+
+// Only positive speeds, range for real-time coding currently is: 6 - 8.
+// Lower means slower/better quality, higher means fastest/lower quality.
+int LibaomAv1Encoder::GetCpuSpeed(int width, int height) {
+  if (aux_config_) {
+    if (auto it = aux_config_->max_pixel_count_to_cpu_speed.lower_bound(width *
+                                                                        height);
+        it != aux_config_->max_pixel_count_to_cpu_speed.end()) {
+      return it->second;
+    }
+
+    return 10;
+  } else {
+    // For smaller resolutions, use lower speed setting (get some coding gain at
+    // the cost of increased encoding complexity).
+    switch (encoder_settings_.GetVideoEncoderComplexity()) {
+      case VideoCodecComplexity::kComplexityHigh:
+        if (width * height <= 320 * 180)
+          return 8;
+        else if (width * height <= 640 * 360)
+          return 9;
+        else
+          return 10;
+      case VideoCodecComplexity::kComplexityHigher:
+        if (width * height <= 320 * 180)
+          return 7;
+        else if (width * height <= 640 * 360)
+          return 8;
+        else if (width * height <= 1280 * 720)
+          return 9;
+        else
+          return 10;
+      case VideoCodecComplexity::kComplexityMax:
+        if (width * height <= 320 * 180)
+          return 6;
+        else if (width * height <= 640 * 360)
+          return 7;
+        else if (width * height <= 1280 * 720)
+          return 8;
+        else
+          return 9;
+      default:
+        return 10;
+    }
+  }
+}
+
+int LibaomAv1Encoder::NumberOfThreads(int width,
+                                      int height,
+                                      int number_of_cores) {
+  // Keep the number of encoder threads equal to the possible number of
+  // column/row tiles, which is (1, 2, 4, 8). See comments below for
+  // AV1E_SET_TILE_COLUMNS/ROWS.
+  if (width * height >= 640 * 360 && number_of_cores > 4) {
+    return 4;
+  } else if (width * height >= 320 * 180 && number_of_cores > 2) {
+    return 2;
+  } else {
+// Use 2 threads for low res on ARM.
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
+    defined(WEBRTC_ANDROID)
+    if (width * height >= 320 * 180 && number_of_cores > 2) {
+      return 2;
+    }
+#endif
+    // 1 thread less than VGA.
+    return 1;
+  }
+}
+
+bool LibaomAv1Encoder::SetSvcParams(
+    ScalableVideoController::StreamLayersConfig svc_config) {
+  bool svc_enabled =
+      svc_config.num_spatial_layers > 1 || svc_config.num_temporal_layers > 1;
+  if (!svc_enabled) {
+    svc_params_ = absl::nullopt;
+    return true;
+  }
+  if (svc_config.num_spatial_layers < 1 || svc_config.num_spatial_layers > 4) {
+    RTC_LOG(LS_WARNING) << "Av1 supports up to 4 spatial layers. "
+                        << svc_config.num_spatial_layers << " configured.";
+    return false;
+  }
+  if (svc_config.num_temporal_layers < 1 ||
+      svc_config.num_temporal_layers > 8) {
+    RTC_LOG(LS_WARNING) << "Av1 supports up to 8 temporal layers. "
+                        << svc_config.num_temporal_layers << " configured.";
+    return false;
+  }
+  aom_svc_params_t& svc_params = svc_params_.emplace();
+  svc_params.number_spatial_layers = svc_config.num_spatial_layers;
+  svc_params.number_temporal_layers = svc_config.num_temporal_layers;
+
+  int num_layers =
+      svc_config.num_spatial_layers * svc_config.num_temporal_layers;
+  for (int i = 0; i < num_layers; ++i) {
+    svc_params.min_quantizers[i] = kQpMin;
+    svc_params.max_quantizers[i] = encoder_settings_.qpMax;
+  }
+
+  // Assume each temporal layer doubles framerate.
+  for (int tid = 0; tid < svc_config.num_temporal_layers; ++tid) {
+    svc_params.framerate_factor[tid] =
+        1 << (svc_config.num_temporal_layers - tid - 1);
+  }
+
+  for (int sid = 0; sid < svc_config.num_spatial_layers; ++sid) {
+    svc_params.scaling_factor_num[sid] = svc_config.scaling_factor_num[sid];
+    svc_params.scaling_factor_den[sid] = svc_config.scaling_factor_den[sid];
+  }
+
+  return true;
+}
+
+void LibaomAv1Encoder::SetSvcLayerId(
+    const ScalableVideoController::LayerFrameConfig& layer_frame) {
+  aom_svc_layer_id_t layer_id = {};
+  layer_id.spatial_layer_id = layer_frame.SpatialId();
+  layer_id.temporal_layer_id = layer_frame.TemporalId();
+  SetEncoderControlParameters(AV1E_SET_SVC_LAYER_ID, &layer_id);
+}
+
+void LibaomAv1Encoder::SetSvcRefFrameConfig(
+    const ScalableVideoController::LayerFrameConfig& layer_frame) {
+  // Buffer name to use for each layer_frame.buffers position. In particular
+  // when there are 2 buffers are referenced, prefer name them last and golden,
+  // because av1 bitstream format has dedicated fields for these two names.
+  // See last_frame_idx and golden_frame_idx in the av1 spec
+  // https://aomediacodec.github.io/av1-spec/av1-spec.pdf
+  static constexpr int kPreferedSlotName[] = {0,  // Last
+                                              3,  // Golden
+                                              1, 2, 4, 5, 6};
+  static constexpr int kAv1NumBuffers = 8;
+
+  aom_svc_ref_frame_config_t ref_frame_config = {};
+  RTC_CHECK_LE(layer_frame.Buffers().size(), ABSL_ARRAYSIZE(kPreferedSlotName));
+  for (size_t i = 0; i < layer_frame.Buffers().size(); ++i) {
+    const CodecBufferUsage& buffer = layer_frame.Buffers()[i];
+    int slot_name = kPreferedSlotName[i];
+    RTC_CHECK_GE(buffer.id, 0);
+    RTC_CHECK_LT(buffer.id, kAv1NumBuffers);
+    ref_frame_config.ref_idx[slot_name] = buffer.id;
+    if (buffer.referenced) {
+      ref_frame_config.reference[slot_name] = 1;
+    }
+    if (buffer.updated) {
+      ref_frame_config.refresh[buffer.id] = 1;
+    }
+  }
+
+  SetEncoderControlParameters(AV1E_SET_SVC_REF_FRAME_CONFIG, &ref_frame_config);
+}
+
+int32_t LibaomAv1Encoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* encoded_image_callback) {
+  encoded_image_callback_ = encoded_image_callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t LibaomAv1Encoder::Release() {
+  if (frame_for_encode_ != nullptr) {
+    aom_img_free(frame_for_encode_);
+    frame_for_encode_ = nullptr;
+  }
+  if (inited_) {
+    if (aom_codec_destroy(&ctx_)) {
+      return WEBRTC_VIDEO_CODEC_MEMORY;
+    }
+    inited_ = false;
+  }
+  rates_configured_ = false;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void LibaomAv1Encoder::MaybeRewrapImgWithFormat(const aom_img_fmt_t fmt) {
+  if (!frame_for_encode_) {
+    frame_for_encode_ =
+        aom_img_wrap(nullptr, fmt, cfg_.g_w, cfg_.g_h, 1, nullptr);
+
+  } else if (frame_for_encode_->fmt != fmt) {
+    RTC_LOG(LS_INFO) << "Switching AV1 encoder pixel format to "
+                     << (fmt == AOM_IMG_FMT_NV12 ? "NV12" : "I420");
+    aom_img_free(frame_for_encode_);
+    frame_for_encode_ =
+        aom_img_wrap(nullptr, fmt, cfg_.g_w, cfg_.g_h, 1, nullptr);
+  }
+  // else no-op since the image is already in the right format.
+}
+
+int32_t LibaomAv1Encoder::Encode(
+    const VideoFrame& frame,
+    const std::vector<VideoFrameType>* frame_types) {
+  if (!inited_ || encoded_image_callback_ == nullptr || !rates_configured_) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  bool keyframe_required =
+      frame_types != nullptr &&
+      absl::c_linear_search(*frame_types, VideoFrameType::kVideoFrameKey);
+
+  std::vector<ScalableVideoController::LayerFrameConfig> layer_frames =
+      svc_controller_->NextFrameConfig(keyframe_required);
+
+  if (layer_frames.empty()) {
+    RTC_LOG(LS_ERROR) << "SVCController returned no configuration for a frame.";
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  rtc::scoped_refptr<VideoFrameBuffer> buffer = frame.video_frame_buffer();
+  absl::InlinedVector<VideoFrameBuffer::Type, kMaxPreferredPixelFormats>
+      supported_formats = {VideoFrameBuffer::Type::kI420,
+                           VideoFrameBuffer::Type::kNV12};
+  rtc::scoped_refptr<VideoFrameBuffer> mapped_buffer;
+  if (buffer->type() != VideoFrameBuffer::Type::kNative) {
+    // `buffer` is already mapped.
+    mapped_buffer = buffer;
+  } else {
+    // Attempt to map to one of the supported formats.
+    mapped_buffer = buffer->GetMappedFrameBuffer(supported_formats);
+  }
+
+  // Convert input frame to I420, if needed.
+  if (!mapped_buffer ||
+      (absl::c_find(supported_formats, mapped_buffer->type()) ==
+           supported_formats.end() &&
+       mapped_buffer->type() != VideoFrameBuffer::Type::kI420A)) {
+    rtc::scoped_refptr<I420BufferInterface> converted_buffer(buffer->ToI420());
+    if (!converted_buffer) {
+      RTC_LOG(LS_ERROR) << "Failed to convert "
+                        << VideoFrameBufferTypeToString(
+                               frame.video_frame_buffer()->type())
+                        << " image to I420. Can't encode frame.";
+      return WEBRTC_VIDEO_CODEC_ENCODER_FAILURE;
+    }
+    RTC_CHECK(converted_buffer->type() == VideoFrameBuffer::Type::kI420 ||
+              converted_buffer->type() == VideoFrameBuffer::Type::kI420A);
+
+    mapped_buffer = converted_buffer;
+  }
+
+  switch (mapped_buffer->type()) {
+    case VideoFrameBuffer::Type::kI420:
+    case VideoFrameBuffer::Type::kI420A: {
+      // Set frame_for_encode_ data pointers and strides.
+      MaybeRewrapImgWithFormat(AOM_IMG_FMT_I420);
+      auto i420_buffer = mapped_buffer->GetI420();
+      RTC_DCHECK(i420_buffer);
+      frame_for_encode_->planes[AOM_PLANE_Y] =
+          const_cast<unsigned char*>(i420_buffer->DataY());
+      frame_for_encode_->planes[AOM_PLANE_U] =
+          const_cast<unsigned char*>(i420_buffer->DataU());
+      frame_for_encode_->planes[AOM_PLANE_V] =
+          const_cast<unsigned char*>(i420_buffer->DataV());
+      frame_for_encode_->stride[AOM_PLANE_Y] = i420_buffer->StrideY();
+      frame_for_encode_->stride[AOM_PLANE_U] = i420_buffer->StrideU();
+      frame_for_encode_->stride[AOM_PLANE_V] = i420_buffer->StrideV();
+      break;
+    }
+    case VideoFrameBuffer::Type::kNV12: {
+      MaybeRewrapImgWithFormat(AOM_IMG_FMT_NV12);
+      const NV12BufferInterface* nv12_buffer = mapped_buffer->GetNV12();
+      RTC_DCHECK(nv12_buffer);
+      frame_for_encode_->planes[AOM_PLANE_Y] =
+          const_cast<unsigned char*>(nv12_buffer->DataY());
+      frame_for_encode_->planes[AOM_PLANE_U] =
+          const_cast<unsigned char*>(nv12_buffer->DataUV());
+      frame_for_encode_->planes[AOM_PLANE_V] = nullptr;
+      frame_for_encode_->stride[AOM_PLANE_Y] = nv12_buffer->StrideY();
+      frame_for_encode_->stride[AOM_PLANE_U] = nv12_buffer->StrideUV();
+      frame_for_encode_->stride[AOM_PLANE_V] = 0;
+      break;
+    }
+    default:
+      return WEBRTC_VIDEO_CODEC_ENCODER_FAILURE;
+  }
+
+  const uint32_t duration =
+      kRtpTicksPerSecond / static_cast<float>(encoder_settings_.maxFramerate);
+
+  const size_t num_spatial_layers =
+      svc_params_ ? svc_params_->number_spatial_layers : 1;
+  auto next_layer_frame = layer_frames.begin();
+  for (size_t i = 0; i < num_spatial_layers; ++i) {
+    // The libaom AV1 encoder requires that `aom_codec_encode` is called for
+    // every spatial layer, even if the configured bitrate for that layer is
+    // zero. For zero bitrate spatial layers no frames will be produced.
+    absl::optional<ScalableVideoController::LayerFrameConfig>
+        non_encoded_layer_frame;
+    ScalableVideoController::LayerFrameConfig* layer_frame;
+    if (next_layer_frame != layer_frames.end() &&
+        next_layer_frame->SpatialId() == static_cast<int>(i)) {
+      layer_frame = &*next_layer_frame;
+      ++next_layer_frame;
+    } else {
+      // For layers that are not encoded only the spatial id matters.
+      non_encoded_layer_frame.emplace().S(i);
+      layer_frame = &*non_encoded_layer_frame;
+    }
+    const bool end_of_picture = (next_layer_frame == layer_frames.end());
+
+    aom_enc_frame_flags_t flags =
+        layer_frame->IsKeyframe() ? AOM_EFLAG_FORCE_KF : 0;
+
+    if (SvcEnabled()) {
+      SetSvcLayerId(*layer_frame);
+      SetSvcRefFrameConfig(*layer_frame);
+
+      SET_ENCODER_PARAM_OR_RETURN_ERROR(AV1E_SET_ERROR_RESILIENT_MODE,
+                                        layer_frame->TemporalId() > 0 ? 1 : 0);
+    }
+
+    // Encode a frame. The presentation timestamp `pts` should never wrap, hence
+    // the unwrapping.
+    aom_codec_err_t ret = aom_codec_encode(
+        &ctx_, frame_for_encode_,
+        rtp_timestamp_unwrapper_.Unwrap(frame.timestamp()), duration, flags);
+    if (ret != AOM_CODEC_OK) {
+      RTC_LOG(LS_WARNING) << "LibaomAv1Encoder::Encode returned " << ret
+                          << " on aom_codec_encode.";
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
+    if (non_encoded_layer_frame) {
+      continue;
+    }
+
+    // Get encoded image data.
+    EncodedImage encoded_image;
+    aom_codec_iter_t iter = nullptr;
+    int data_pkt_count = 0;
+    while (const aom_codec_cx_pkt_t* pkt =
+               aom_codec_get_cx_data(&ctx_, &iter)) {
+      if (pkt->kind == AOM_CODEC_CX_FRAME_PKT && pkt->data.frame.sz > 0) {
+        if (data_pkt_count > 0) {
+          RTC_LOG(LS_WARNING) << "LibaomAv1Encoder::Encoder returned more than "
+                                 "one data packet for an input video frame.";
+          Release();
+        }
+        encoded_image.SetEncodedData(EncodedImageBuffer::Create(
+            /*data=*/static_cast<const uint8_t*>(pkt->data.frame.buf),
+            /*size=*/pkt->data.frame.sz));
+
+        if ((pkt->data.frame.flags & AOM_EFLAG_FORCE_KF) != 0) {
+          layer_frame->Keyframe();
+        }
+
+        encoded_image._frameType = layer_frame->IsKeyframe()
+                                       ? VideoFrameType::kVideoFrameKey
+                                       : VideoFrameType::kVideoFrameDelta;
+        encoded_image.SetTimestamp(frame.timestamp());
+        encoded_image.capture_time_ms_ = frame.render_time_ms();
+        encoded_image.rotation_ = frame.rotation();
+        encoded_image.content_type_ = VideoContentType::UNSPECIFIED;
+        // If encoded image width/height info are added to aom_codec_cx_pkt_t,
+        // use those values in lieu of the values in frame.
+        if (svc_params_) {
+          int n = svc_params_->scaling_factor_num[layer_frame->SpatialId()];
+          int d = svc_params_->scaling_factor_den[layer_frame->SpatialId()];
+          encoded_image._encodedWidth = cfg_.g_w * n / d;
+          encoded_image._encodedHeight = cfg_.g_h * n / d;
+          encoded_image.SetSpatialIndex(layer_frame->SpatialId());
+          encoded_image.SetTemporalIndex(layer_frame->TemporalId());
+        } else {
+          encoded_image._encodedWidth = cfg_.g_w;
+          encoded_image._encodedHeight = cfg_.g_h;
+        }
+        encoded_image.timing_.flags = VideoSendTiming::kInvalid;
+
+        int qp = -1;
+        SET_ENCODER_PARAM_OR_RETURN_ERROR(AOME_GET_LAST_QUANTIZER, &qp);
+        encoded_image.qp_ = qp;
+
+        encoded_image.SetColorSpace(frame.color_space());
+        ++data_pkt_count;
+      }
+    }
+
+    // Deliver encoded image data.
+    if (encoded_image.size() > 0) {
+      CodecSpecificInfo codec_specific_info;
+      codec_specific_info.codecType = kVideoCodecAV1;
+      codec_specific_info.end_of_picture = end_of_picture;
+      codec_specific_info.scalability_mode = scalability_mode_;
+      bool is_keyframe = layer_frame->IsKeyframe();
+      codec_specific_info.generic_frame_info =
+          svc_controller_->OnEncodeDone(*layer_frame);
+      if (is_keyframe && codec_specific_info.generic_frame_info) {
+        codec_specific_info.template_structure =
+            svc_controller_->DependencyStructure();
+        auto& resolutions = codec_specific_info.template_structure->resolutions;
+        if (SvcEnabled()) {
+          resolutions.resize(svc_params_->number_spatial_layers);
+          for (int sid = 0; sid < svc_params_->number_spatial_layers; ++sid) {
+            int n = svc_params_->scaling_factor_num[sid];
+            int d = svc_params_->scaling_factor_den[sid];
+            resolutions[sid] =
+                RenderResolution(cfg_.g_w * n / d, cfg_.g_h * n / d);
+          }
+        } else {
+          resolutions = {RenderResolution(cfg_.g_w, cfg_.g_h)};
+        }
+      }
+      encoded_image_callback_->OnEncodedImage(encoded_image,
+                                              &codec_specific_info);
+    }
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void LibaomAv1Encoder::SetRates(const RateControlParameters& parameters) {
+  if (!inited_) {
+    RTC_LOG(LS_WARNING) << "SetRates() while encoder is not initialized";
+    return;
+  }
+  if (parameters.framerate_fps < kMinimumFrameRate) {
+    RTC_LOG(LS_WARNING) << "Unsupported framerate (must be >= "
+                        << kMinimumFrameRate
+                        << " ): " << parameters.framerate_fps;
+    return;
+  }
+  if (parameters.bitrate.get_sum_bps() == 0) {
+    RTC_LOG(LS_WARNING) << "Attempt to set target bit rate to zero";
+    return;
+  }
+
+  // The bitrates caluclated internally in libaom when `AV1E_SET_SVC_PARAMS` is
+  // called depends on the currently configured `rc_target_bitrate`. If the
+  // total target bitrate is not updated first a division by zero could happen.
+  svc_controller_->OnRatesUpdated(parameters.bitrate);
+  cfg_.rc_target_bitrate = parameters.bitrate.get_sum_kbps();
+  aom_codec_err_t error_code = aom_codec_enc_config_set(&ctx_, &cfg_);
+  if (error_code != AOM_CODEC_OK) {
+    RTC_LOG(LS_WARNING) << "Error configuring encoder, error code: "
+                        << error_code;
+  }
+
+  if (SvcEnabled()) {
+    for (int sid = 0; sid < svc_params_->number_spatial_layers; ++sid) {
+      // libaom bitrate for spatial id S and temporal id T means bitrate
+      // of frames with spatial_id=S and temporal_id<=T
+      // while `parameters.bitrate` provdies bitrate of frames with
+      // spatial_id=S and temporal_id=T
+      int accumulated_bitrate_bps = 0;
+      for (int tid = 0; tid < svc_params_->number_temporal_layers; ++tid) {
+        int layer_index = sid * svc_params_->number_temporal_layers + tid;
+        accumulated_bitrate_bps += parameters.bitrate.GetBitrate(sid, tid);
+        // `svc_params.layer_target_bitrate` expects bitrate in kbps.
+        svc_params_->layer_target_bitrate[layer_index] =
+            accumulated_bitrate_bps / 1000;
+      }
+    }
+    SetEncoderControlParameters(AV1E_SET_SVC_PARAMS, &*svc_params_);
+  }
+
+  rates_configured_ = true;
+
+  // Set frame rate to closest integer value.
+  encoder_settings_.maxFramerate =
+      static_cast<uint32_t>(parameters.framerate_fps + 0.5);
+}
+
+VideoEncoder::EncoderInfo LibaomAv1Encoder::GetEncoderInfo() const {
+  EncoderInfo info;
+  info.supports_native_handle = false;
+  info.implementation_name = "libaom";
+  info.has_trusted_rate_controller = true;
+  info.is_hardware_accelerated = false;
+  info.scaling_settings = VideoEncoder::ScalingSettings(kMinQindex, kMaxQindex);
+  info.preferred_pixel_formats = {VideoFrameBuffer::Type::kI420,
+                                  VideoFrameBuffer::Type::kNV12};
+  if (SvcEnabled()) {
+    for (int sid = 0; sid < svc_params_->number_spatial_layers; ++sid) {
+      info.fps_allocation[sid].resize(svc_params_->number_temporal_layers);
+      for (int tid = 0; tid < svc_params_->number_temporal_layers; ++tid) {
+        info.fps_allocation[sid][tid] =
+            encoder_settings_.maxFramerate / svc_params_->framerate_factor[tid];
+      }
+    }
+  }
+  return info;
+}
+
+}  // namespace
+
+std::unique_ptr<VideoEncoder> CreateLibaomAv1Encoder() {
+  return std::make_unique<LibaomAv1Encoder>(absl::nullopt);
+}
+
+std::unique_ptr<VideoEncoder> CreateLibaomAv1Encoder(
+    const LibaomAv1EncoderAuxConfig& aux_config) {
+  return std::make_unique<LibaomAv1Encoder>(aux_config);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_encoder.h b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_encoder.h
new file mode 100644
index 0000000000..2fd1d5a754
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_encoder.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_CODECS_AV1_LIBAOM_AV1_ENCODER_H_
+#define MODULES_VIDEO_CODING_CODECS_AV1_LIBAOM_AV1_ENCODER_H_
+
+#include <map>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/video_codecs/video_encoder.h"
+
+namespace webrtc {
+struct LibaomAv1EncoderAuxConfig {
+  // A map of max pixel count --> cpu speed.
+  std::map<int, int> max_pixel_count_to_cpu_speed;
+};
+
+std::unique_ptr<VideoEncoder> CreateLibaomAv1Encoder();
+std::unique_ptr<VideoEncoder> CreateLibaomAv1Encoder(
+    const LibaomAv1EncoderAuxConfig& aux_config);
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_AV1_LIBAOM_AV1_ENCODER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc
new file mode 100644
index 0000000000..d194cef35b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_encoder_unittest.cc
@@ -0,0 +1,264 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/av1/libaom_av1_encoder.h"
+
+#include <limits>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/codecs/test/encoded_video_frame_producer.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::IsEmpty;
+using ::testing::SizeIs;
+
+VideoCodec DefaultCodecSettings() {
+  VideoCodec codec_settings;
+  codec_settings.width = 320;
+  codec_settings.height = 180;
+  codec_settings.maxFramerate = 30;
+  codec_settings.maxBitrate = 1000;
+  codec_settings.qpMax = 63;
+  return codec_settings;
+}
+
+VideoEncoder::Settings DefaultEncoderSettings() {
+  return VideoEncoder::Settings(
+      VideoEncoder::Capabilities(/*loss_notification=*/false),
+      /*number_of_cores=*/1, /*max_payload_size=*/1200);
+}
+
+TEST(LibaomAv1EncoderTest, CanCreate) {
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  EXPECT_TRUE(encoder);
+}
+
+TEST(LibaomAv1EncoderTest, InitAndRelease) {
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  ASSERT_TRUE(encoder);
+  VideoCodec codec_settings = DefaultCodecSettings();
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+  EXPECT_EQ(encoder->Release(), WEBRTC_VIDEO_CODEC_OK);
+}
+
+TEST(LibaomAv1EncoderTest, NoBitrateOnTopLayerRefecltedInActiveDecodeTargets) {
+  // Configure encoder with 2 temporal layers.
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.SetScalabilityMode(ScalabilityMode::kL1T2);
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  VideoEncoder::RateControlParameters rate_parameters;
+  rate_parameters.framerate_fps = 30;
+  rate_parameters.bitrate.SetBitrate(0, /*temporal_index=*/0, 300'000);
+  rate_parameters.bitrate.SetBitrate(0, /*temporal_index=*/1, 0);
+  encoder->SetRates(rate_parameters);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> encoded_frames =
+      EncodedVideoFrameProducer(*encoder).SetNumInputFrames(1).Encode();
+  ASSERT_THAT(encoded_frames, SizeIs(1));
+  ASSERT_NE(encoded_frames[0].codec_specific_info.generic_frame_info,
+            absl::nullopt);
+  // Assuming L1T2 structure uses 1st decode target for T0 and 2nd decode target
+  // for T0+T1 frames, expect only 1st decode target is active.
+  EXPECT_EQ(encoded_frames[0]
+                .codec_specific_info.generic_frame_info->active_decode_targets,
+            0b01);
+}
+
+TEST(LibaomAv1EncoderTest,
+     SpatialScalabilityInTemporalUnitReportedAsDeltaFrame) {
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.SetScalabilityMode(ScalabilityMode::kL2T1);
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  VideoEncoder::RateControlParameters rate_parameters;
+  rate_parameters.framerate_fps = 30;
+  rate_parameters.bitrate.SetBitrate(/*spatial_index=*/0, 0, 300'000);
+  rate_parameters.bitrate.SetBitrate(/*spatial_index=*/1, 0, 300'000);
+  encoder->SetRates(rate_parameters);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> encoded_frames =
+      EncodedVideoFrameProducer(*encoder).SetNumInputFrames(1).Encode();
+  ASSERT_THAT(encoded_frames, SizeIs(2));
+  EXPECT_THAT(encoded_frames[0].encoded_image._frameType,
+              Eq(VideoFrameType::kVideoFrameKey));
+  EXPECT_THAT(encoded_frames[1].encoded_image._frameType,
+              Eq(VideoFrameType::kVideoFrameDelta));
+}
+
+TEST(LibaomAv1EncoderTest, NoBitrateOnTopSpatialLayerProduceDeltaFrames) {
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.SetScalabilityMode(ScalabilityMode::kL2T1);
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  VideoEncoder::RateControlParameters rate_parameters;
+  rate_parameters.framerate_fps = 30;
+  rate_parameters.bitrate.SetBitrate(/*spatial_index=*/0, 0, 300'000);
+  rate_parameters.bitrate.SetBitrate(/*spatial_index=*/1, 0, 0);
+  encoder->SetRates(rate_parameters);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> encoded_frames =
+      EncodedVideoFrameProducer(*encoder).SetNumInputFrames(2).Encode();
+  ASSERT_THAT(encoded_frames, SizeIs(2));
+  EXPECT_THAT(encoded_frames[0].encoded_image._frameType,
+              Eq(VideoFrameType::kVideoFrameKey));
+  EXPECT_THAT(encoded_frames[1].encoded_image._frameType,
+              Eq(VideoFrameType::kVideoFrameDelta));
+}
+
+TEST(LibaomAv1EncoderTest, SetsEndOfPictureForLastFrameInTemporalUnit) {
+  VideoBitrateAllocation allocation;
+  allocation.SetBitrate(0, 0, 30000);
+  allocation.SetBitrate(1, 0, 40000);
+  allocation.SetBitrate(2, 0, 30000);
+
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  // Configure encoder with 3 spatial layers.
+  codec_settings.SetScalabilityMode(ScalabilityMode::kL3T1);
+  codec_settings.maxBitrate = allocation.get_sum_kbps();
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      allocation, codec_settings.maxFramerate));
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> encoded_frames =
+      EncodedVideoFrameProducer(*encoder).SetNumInputFrames(2).Encode();
+  ASSERT_THAT(encoded_frames, SizeIs(6));
+  EXPECT_FALSE(encoded_frames[0].codec_specific_info.end_of_picture);
+  EXPECT_FALSE(encoded_frames[1].codec_specific_info.end_of_picture);
+  EXPECT_TRUE(encoded_frames[2].codec_specific_info.end_of_picture);
+  EXPECT_FALSE(encoded_frames[3].codec_specific_info.end_of_picture);
+  EXPECT_FALSE(encoded_frames[4].codec_specific_info.end_of_picture);
+  EXPECT_TRUE(encoded_frames[5].codec_specific_info.end_of_picture);
+}
+
+TEST(LibaomAv1EncoderTest, CheckOddDimensionsWithSpatialLayers) {
+  VideoBitrateAllocation allocation;
+  allocation.SetBitrate(0, 0, 30000);
+  allocation.SetBitrate(1, 0, 40000);
+  allocation.SetBitrate(2, 0, 30000);
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  // Configure encoder with 3 spatial layers.
+  codec_settings.SetScalabilityMode(ScalabilityMode::kL3T1);
+  // Odd width and height values should not make encoder crash.
+  codec_settings.width = 623;
+  codec_settings.height = 405;
+  codec_settings.maxBitrate = allocation.get_sum_kbps();
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      allocation, codec_settings.maxFramerate));
+  EncodedVideoFrameProducer evfp(*encoder);
+  evfp.SetResolution(RenderResolution{623, 405});
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> encoded_frames =
+      evfp.SetNumInputFrames(2).Encode();
+  ASSERT_THAT(encoded_frames, SizeIs(6));
+}
+
+TEST(LibaomAv1EncoderTest, EncoderInfoProvidesFpsAllocation) {
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.SetScalabilityMode(ScalabilityMode::kL3T3);
+  codec_settings.maxFramerate = 60;
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  const auto& encoder_info = encoder->GetEncoderInfo();
+  EXPECT_THAT(encoder_info.fps_allocation[0], ElementsAre(15, 30, 60));
+  EXPECT_THAT(encoder_info.fps_allocation[1], ElementsAre(15, 30, 60));
+  EXPECT_THAT(encoder_info.fps_allocation[2], ElementsAre(15, 30, 60));
+  EXPECT_THAT(encoder_info.fps_allocation[3], IsEmpty());
+}
+
+TEST(LibaomAv1EncoderTest, PopulatesEncodedFrameSize) {
+  VideoBitrateAllocation allocation;
+  allocation.SetBitrate(0, 0, 30000);
+  allocation.SetBitrate(1, 0, 40000);
+  allocation.SetBitrate(2, 0, 30000);
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.maxBitrate = allocation.get_sum_kbps();
+  ASSERT_GT(codec_settings.width, 4);
+  // Configure encoder with 3 spatial layers.
+  codec_settings.SetScalabilityMode(ScalabilityMode::kL3T1);
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      allocation, codec_settings.maxFramerate));
+  using Frame = EncodedVideoFrameProducer::EncodedFrame;
+  std::vector<Frame> encoded_frames =
+      EncodedVideoFrameProducer(*encoder).SetNumInputFrames(1).Encode();
+  EXPECT_THAT(
+      encoded_frames,
+      ElementsAre(
+          Field(&Frame::encoded_image,
+                AllOf(Field(&EncodedImage::_encodedWidth,
+                            codec_settings.width / 4),
+                      Field(&EncodedImage::_encodedHeight,
+                            codec_settings.height / 4))),
+          Field(&Frame::encoded_image,
+                AllOf(Field(&EncodedImage::_encodedWidth,
+                            codec_settings.width / 2),
+                      Field(&EncodedImage::_encodedHeight,
+                            codec_settings.height / 2))),
+          Field(&Frame::encoded_image,
+                AllOf(Field(&EncodedImage::_encodedWidth, codec_settings.width),
+                      Field(&EncodedImage::_encodedHeight,
+                            codec_settings.height)))));
+}
+
+TEST(LibaomAv1EncoderTest, RtpTimestampWrap) {
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.SetScalabilityMode(ScalabilityMode::kL1T1);
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  VideoEncoder::RateControlParameters rate_parameters;
+  rate_parameters.framerate_fps = 30;
+  rate_parameters.bitrate.SetBitrate(/*spatial_index=*/0, 0, 300'000);
+  encoder->SetRates(rate_parameters);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> encoded_frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(2)
+          .SetRtpTimestamp(std::numeric_limits<uint32_t>::max())
+          .Encode();
+  ASSERT_THAT(encoded_frames, SizeIs(2));
+  EXPECT_THAT(encoded_frames[0].encoded_image._frameType,
+              Eq(VideoFrameType::kVideoFrameKey));
+  EXPECT_THAT(encoded_frames[1].encoded_image._frameType,
+              Eq(VideoFrameType::kVideoFrameDelta));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
new file mode 100644
index 0000000000..86e317f94b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
@@ -0,0 +1,368 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+#include <memory>
+#include <ostream>
+#include <tuple>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/codecs/av1/dav1d_decoder.h"
+#include "modules/video_coding/codecs/av1/libaom_av1_encoder.h"
+#include "modules/video_coding/codecs/test/encoded_video_frame_producer.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "modules/video_coding/svc/scalable_video_controller_no_layering.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ContainerEq;
+using ::testing::Each;
+using ::testing::ElementsAreArray;
+using ::testing::Ge;
+using ::testing::IsEmpty;
+using ::testing::Not;
+using ::testing::NotNull;
+using ::testing::Optional;
+using ::testing::Pointwise;
+using ::testing::SizeIs;
+using ::testing::Truly;
+using ::testing::Values;
+
+// Use small resolution for this test to make it faster.
+constexpr int kWidth = 320;
+constexpr int kHeight = 180;
+constexpr int kFramerate = 30;
+
+VideoCodec DefaultCodecSettings() {
+  VideoCodec codec_settings;
+  codec_settings.SetScalabilityMode(ScalabilityMode::kL1T1);
+  codec_settings.width = kWidth;
+  codec_settings.height = kHeight;
+  codec_settings.maxFramerate = kFramerate;
+  codec_settings.maxBitrate = 1000;
+  codec_settings.qpMax = 63;
+  return codec_settings;
+}
+VideoEncoder::Settings DefaultEncoderSettings() {
+  return VideoEncoder::Settings(
+      VideoEncoder::Capabilities(/*loss_notification=*/false),
+      /*number_of_cores=*/1, /*max_payload_size=*/1200);
+}
+
+class TestAv1Decoder {
+ public:
+  explicit TestAv1Decoder(int decoder_id)
+      : decoder_id_(decoder_id), decoder_(CreateDav1dDecoder()) {
+    if (decoder_ == nullptr) {
+      ADD_FAILURE() << "Failed to create a decoder#" << decoder_id_;
+      return;
+    }
+    EXPECT_TRUE(decoder_->Configure({}));
+    EXPECT_EQ(decoder_->RegisterDecodeCompleteCallback(&callback_),
+              WEBRTC_VIDEO_CODEC_OK);
+  }
+  // This class requires pointer stability and thus not copyable nor movable.
+  TestAv1Decoder(const TestAv1Decoder&) = delete;
+  TestAv1Decoder& operator=(const TestAv1Decoder&) = delete;
+
+  void Decode(int64_t frame_id, const EncodedImage& image) {
+    ASSERT_THAT(decoder_, NotNull());
+    int32_t error = decoder_->Decode(image, /*missing_frames=*/false,
+                                     /*render_time_ms=*/image.capture_time_ms_);
+    if (error != WEBRTC_VIDEO_CODEC_OK) {
+      ADD_FAILURE() << "Failed to decode frame id " << frame_id
+                    << " with error code " << error << " by decoder#"
+                    << decoder_id_;
+      return;
+    }
+    decoded_ids_.push_back(frame_id);
+  }
+
+  const std::vector<int64_t>& decoded_frame_ids() const { return decoded_ids_; }
+  size_t num_output_frames() const { return callback_.num_called(); }
+
+ private:
+  // Decoder callback that only counts how many times it was called.
+  // While it is tempting to replace it with a simple mock, that one requires
+  // to set expectation on number of calls in advance. Tests below unsure about
+  // expected number of calls until after calls are done.
+  class DecoderCallback : public DecodedImageCallback {
+   public:
+    size_t num_called() const { return num_called_; }
+
+   private:
+    int32_t Decoded(VideoFrame& /*decoded_image*/) override {
+      ++num_called_;
+      return 0;
+    }
+    void Decoded(VideoFrame& /*decoded_image*/,
+                 absl::optional<int32_t> /*decode_time_ms*/,
+                 absl::optional<uint8_t> /*qp*/) override {
+      ++num_called_;
+    }
+
+    int num_called_ = 0;
+  };
+
+  const int decoder_id_;
+  std::vector<int64_t> decoded_ids_;
+  DecoderCallback callback_;
+  const std::unique_ptr<VideoDecoder> decoder_;
+};
+
+TEST(LibaomAv1Test, EncodeDecode) {
+  TestAv1Decoder decoder(0);
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  VideoBitrateAllocation allocation;
+  allocation.SetBitrate(0, 0, 300000);
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      allocation, codec_settings.maxFramerate));
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> encoded_frames =
+      EncodedVideoFrameProducer(*encoder).SetNumInputFrames(4).Encode();
+  for (size_t frame_id = 0; frame_id < encoded_frames.size(); ++frame_id) {
+    decoder.Decode(static_cast<int64_t>(frame_id),
+                   encoded_frames[frame_id].encoded_image);
+  }
+
+  // Check encoder produced some frames for decoder to decode.
+  ASSERT_THAT(encoded_frames, Not(IsEmpty()));
+  // Check decoder found all of them valid.
+  EXPECT_THAT(decoder.decoded_frame_ids(), SizeIs(encoded_frames.size()));
+  // Check each of them produced an output frame.
+  EXPECT_EQ(decoder.num_output_frames(), decoder.decoded_frame_ids().size());
+}
+
+struct LayerId {
+  friend bool operator==(const LayerId& lhs, const LayerId& rhs) {
+    return std::tie(lhs.spatial_id, lhs.temporal_id) ==
+           std::tie(rhs.spatial_id, rhs.temporal_id);
+  }
+  friend bool operator<(const LayerId& lhs, const LayerId& rhs) {
+    return std::tie(lhs.spatial_id, lhs.temporal_id) <
+           std::tie(rhs.spatial_id, rhs.temporal_id);
+  }
+  friend std::ostream& operator<<(std::ostream& s, const LayerId& layer) {
+    return s << "S" << layer.spatial_id << "T" << layer.temporal_id;
+  }
+
+  int spatial_id = 0;
+  int temporal_id = 0;
+};
+
+struct SvcTestParam {
+  ScalabilityMode GetScalabilityMode() const {
+    absl::optional<ScalabilityMode> scalability_mode =
+        ScalabilityModeFromString(name);
+    RTC_CHECK(scalability_mode.has_value());
+    return *scalability_mode;
+  }
+
+  std::string name;
+  int num_frames_to_generate;
+  std::map<LayerId, DataRate> configured_bitrates;
+};
+
+class LibaomAv1SvcTest : public ::testing::TestWithParam<SvcTestParam> {};
+
+TEST_P(LibaomAv1SvcTest, EncodeAndDecodeAllDecodeTargets) {
+  const SvcTestParam param = GetParam();
+  std::unique_ptr<ScalableVideoController> svc_controller =
+      CreateScalabilityStructure(param.GetScalabilityMode());
+  ASSERT_TRUE(svc_controller);
+  VideoBitrateAllocation allocation;
+  if (param.configured_bitrates.empty()) {
+    ScalableVideoController::StreamLayersConfig config =
+        svc_controller->StreamConfig();
+    for (int sid = 0; sid < config.num_spatial_layers; ++sid) {
+      for (int tid = 0; tid < config.num_temporal_layers; ++tid) {
+        allocation.SetBitrate(sid, tid, 100'000);
+      }
+    }
+  } else {
+    for (const auto& kv : param.configured_bitrates) {
+      allocation.SetBitrate(kv.first.spatial_id, kv.first.temporal_id,
+                            kv.second.bps());
+    }
+  }
+
+  size_t num_decode_targets =
+      svc_controller->DependencyStructure().num_decode_targets;
+
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.SetScalabilityMode(GetParam().GetScalabilityMode());
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      allocation, codec_settings.maxFramerate));
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> encoded_frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(GetParam().num_frames_to_generate)
+          .SetResolution({kWidth, kHeight})
+          .Encode();
+
+  ASSERT_THAT(
+      encoded_frames,
+      Each(Truly([&](const EncodedVideoFrameProducer::EncodedFrame& frame) {
+        return frame.codec_specific_info.generic_frame_info &&
+               frame.codec_specific_info.generic_frame_info
+                       ->decode_target_indications.size() == num_decode_targets;
+      })));
+
+  for (size_t dt = 0; dt < num_decode_targets; ++dt) {
+    TestAv1Decoder decoder(dt);
+    std::vector<int64_t> requested_ids;
+    for (int64_t frame_id = 0;
+         frame_id < static_cast<int64_t>(encoded_frames.size()); ++frame_id) {
+      const EncodedVideoFrameProducer::EncodedFrame& frame =
+          encoded_frames[frame_id];
+      if (frame.codec_specific_info.generic_frame_info
+              ->decode_target_indications[dt] !=
+          DecodeTargetIndication::kNotPresent) {
+        requested_ids.push_back(frame_id);
+        decoder.Decode(frame_id, frame.encoded_image);
+      }
+      EXPECT_THAT(frame.codec_specific_info.scalability_mode,
+                  Optional(param.GetScalabilityMode()));
+    }
+
+    ASSERT_THAT(requested_ids, SizeIs(Ge(2u)));
+    // Check decoder found all of them valid.
+    EXPECT_THAT(decoder.decoded_frame_ids(), ContainerEq(requested_ids))
+        << "Decoder#" << dt;
+    // Check each of them produced an output frame.
+    EXPECT_EQ(decoder.num_output_frames(), decoder.decoded_frame_ids().size())
+        << "Decoder#" << dt;
+  }
+}
+
+MATCHER(SameLayerIdAndBitrateIsNear, "") {
+  // First check if layer id is the same.
+  return std::get<0>(arg).first == std::get<1>(arg).first &&
+         // check measured bitrate is not much lower than requested.
+         std::get<0>(arg).second >= std::get<1>(arg).second * 0.8 &&
+         // check measured bitrate is not much larger than requested.
+         std::get<0>(arg).second <= std::get<1>(arg).second * 1.1;
+}
+
+TEST_P(LibaomAv1SvcTest, SetRatesMatchMeasuredBitrate) {
+  const SvcTestParam param = GetParam();
+  if (param.configured_bitrates.empty()) {
+    // Rates are not configured for this particular structure, skip the test.
+    return;
+  }
+  constexpr TimeDelta kDuration = TimeDelta::Seconds(5);
+
+  VideoBitrateAllocation allocation;
+  for (const auto& kv : param.configured_bitrates) {
+    allocation.SetBitrate(kv.first.spatial_id, kv.first.temporal_id,
+                          kv.second.bps());
+  }
+
+  std::unique_ptr<VideoEncoder> encoder = CreateLibaomAv1Encoder();
+  ASSERT_TRUE(encoder);
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.SetScalabilityMode(param.GetScalabilityMode());
+  codec_settings.maxBitrate = allocation.get_sum_kbps();
+  codec_settings.maxFramerate = 30;
+  ASSERT_EQ(encoder->InitEncode(&codec_settings, DefaultEncoderSettings()),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      allocation, codec_settings.maxFramerate));
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> encoded_frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(codec_settings.maxFramerate * kDuration.seconds())
+          .SetResolution({codec_settings.width, codec_settings.height})
+          .SetFramerateFps(codec_settings.maxFramerate)
+          .Encode();
+
+  // Calculate size of each layer.
+  std::map<LayerId, DataSize> layer_size;
+  for (const auto& frame : encoded_frames) {
+    ASSERT_TRUE(frame.codec_specific_info.generic_frame_info);
+    const auto& layer = *frame.codec_specific_info.generic_frame_info;
+    LayerId layer_id = {layer.spatial_id, layer.temporal_id};
+    // This is almost same as
+    // layer_size[layer_id] += DataSize::Bytes(frame.encoded_image.size());
+    // but avoids calling deleted default constructor for DataSize.
+    layer_size.emplace(layer_id, DataSize::Zero()).first->second +=
+        DataSize::Bytes(frame.encoded_image.size());
+  }
+  // Convert size of the layer into bitrate of that layer.
+  std::vector<std::pair<LayerId, DataRate>> measured_bitrates;
+  for (const auto& kv : layer_size) {
+    measured_bitrates.emplace_back(kv.first, kv.second / kDuration);
+  }
+  EXPECT_THAT(measured_bitrates, Pointwise(SameLayerIdAndBitrateIsNear(),
+                                           param.configured_bitrates));
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    Svc,
+    LibaomAv1SvcTest,
+    Values(SvcTestParam{"L1T1", /*num_frames_to_generate=*/4},
+           SvcTestParam{"L1T2",
+                        /*num_frames_to_generate=*/4,
+                        /*configured_bitrates=*/
+                        {{{0, 0}, DataRate::KilobitsPerSec(60)},
+                         {{0, 1}, DataRate::KilobitsPerSec(40)}}},
+           SvcTestParam{"L1T3", /*num_frames_to_generate=*/8},
+           SvcTestParam{"L2T1",
+                        /*num_frames_to_generate=*/3,
+                        /*configured_bitrates=*/
+                        {{{0, 0}, DataRate::KilobitsPerSec(30)},
+                         {{1, 0}, DataRate::KilobitsPerSec(70)}}},
+           SvcTestParam{"L2T1h",
+                        /*num_frames_to_generate=*/3,
+                        /*configured_bitrates=*/
+                        {{{0, 0}, DataRate::KilobitsPerSec(30)},
+                         {{1, 0}, DataRate::KilobitsPerSec(70)}}},
+           SvcTestParam{"L2T1_KEY", /*num_frames_to_generate=*/3},
+           SvcTestParam{"L3T1", /*num_frames_to_generate=*/3},
+           SvcTestParam{"L3T3", /*num_frames_to_generate=*/8},
+           SvcTestParam{"S2T1", /*num_frames_to_generate=*/3},
+           SvcTestParam{"S3T3", /*num_frames_to_generate=*/8},
+           SvcTestParam{"L2T2", /*num_frames_to_generate=*/4},
+           SvcTestParam{"L2T2_KEY", /*num_frames_to_generate=*/4},
+           SvcTestParam{"L2T2_KEY_SHIFT",
+                        /*num_frames_to_generate=*/4,
+                        /*configured_bitrates=*/
+                        {{{0, 0}, DataRate::KilobitsPerSec(70)},
+                         {{0, 1}, DataRate::KilobitsPerSec(30)},
+                         {{1, 0}, DataRate::KilobitsPerSec(110)},
+                         {{1, 1}, DataRate::KilobitsPerSec(80)}}}),
+    [](const testing::TestParamInfo<SvcTestParam>& info) {
+      return info.param.name;
+    });
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/DEPS b/third_party/libwebrtc/modules/video_coding/codecs/h264/DEPS
new file mode 100644
index 0000000000..4e110917d8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+  "+third_party/ffmpeg",
+  "+third_party/openh264",
+  "+media/base",
+]
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/OWNERS b/third_party/libwebrtc/modules/video_coding/codecs/h264/OWNERS
new file mode 100644
index 0000000000..4b06c4e32b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/OWNERS
@@ -0,0 +1,2 @@
+sprang@webrtc.org
+ssilkin@webrtc.org
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264.cc b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264.cc
new file mode 100644
index 0000000000..23580d7a4a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264.cc
@@ -0,0 +1,166 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "modules/video_coding/codecs/h264/include/h264.h"
+
+#include <memory>
+#include <string>
+
+#include "absl/container/inlined_vector.h"
+#include "absl/types/optional.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "media/base/media_constants.h"
+#include "rtc_base/trace_event.h"
+
+#if defined(WEBRTC_USE_H264)
+#include "modules/video_coding/codecs/h264/h264_decoder_impl.h"
+#include "modules/video_coding/codecs/h264/h264_encoder_impl.h"
+#endif
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+#if defined(WEBRTC_USE_H264)
+bool g_rtc_use_h264 = true;
+#endif
+
+// If H.264 OpenH264/FFmpeg codec is supported.
+bool IsH264CodecSupported() {
+#if defined(WEBRTC_USE_H264)
+  return g_rtc_use_h264;
+#else
+  return false;
+#endif
+}
+
+constexpr ScalabilityMode kSupportedScalabilityModes[] = {
+    ScalabilityMode::kL1T1, ScalabilityMode::kL1T2, ScalabilityMode::kL1T3};
+
+}  // namespace
+
+SdpVideoFormat CreateH264Format(H264Profile profile,
+                                H264Level level,
+                                const std::string& packetization_mode,
+                                bool add_scalability_modes) {
+  const absl::optional<std::string> profile_string =
+      H264ProfileLevelIdToString(H264ProfileLevelId(profile, level));
+  RTC_CHECK(profile_string);
+  absl::InlinedVector<ScalabilityMode, kScalabilityModeCount> scalability_modes;
+  if (add_scalability_modes) {
+    for (const auto scalability_mode : kSupportedScalabilityModes) {
+      scalability_modes.push_back(scalability_mode);
+    }
+  }
+  return SdpVideoFormat(
+      cricket::kH264CodecName,
+      {{cricket::kH264FmtpProfileLevelId, *profile_string},
+       {cricket::kH264FmtpLevelAsymmetryAllowed, "1"},
+       {cricket::kH264FmtpPacketizationMode, packetization_mode}},
+      scalability_modes);
+}
+
+void DisableRtcUseH264() {
+#if defined(WEBRTC_USE_H264)
+  g_rtc_use_h264 = false;
+#endif
+}
+
+std::vector<SdpVideoFormat> SupportedH264Codecs(bool add_scalability_modes) {
+  TRACE_EVENT0("webrtc", __func__);
+  if (!IsH264CodecSupported())
+    return std::vector<SdpVideoFormat>();
+  // We only support encoding Constrained Baseline Profile (CBP), but the
+  // decoder supports more profiles. We can list all profiles here that are
+  // supported by the decoder and that are also supersets of CBP, i.e. the
+  // decoder for that profile is required to be able to decode CBP. This means
+  // we can encode and send CBP even though we negotiated a potentially
+  // higher profile. See the H264 spec for more information.
+  //
+  // We support both packetization modes 0 (mandatory) and 1 (optional,
+  // preferred).
+  return {CreateH264Format(H264Profile::kProfileBaseline, H264Level::kLevel3_1,
+                           "1", add_scalability_modes),
+          CreateH264Format(H264Profile::kProfileBaseline, H264Level::kLevel3_1,
+                           "0", add_scalability_modes),
+          CreateH264Format(H264Profile::kProfileConstrainedBaseline,
+                           H264Level::kLevel3_1, "1", add_scalability_modes),
+          CreateH264Format(H264Profile::kProfileConstrainedBaseline,
+                           H264Level::kLevel3_1, "0", add_scalability_modes),
+          CreateH264Format(H264Profile::kProfileMain, H264Level::kLevel3_1, "1",
+                           add_scalability_modes),
+          CreateH264Format(H264Profile::kProfileMain, H264Level::kLevel3_1, "0",
+                           add_scalability_modes)};
+}
+
+std::vector<SdpVideoFormat> SupportedH264DecoderCodecs() {
+  TRACE_EVENT0("webrtc", __func__);
+  if (!IsH264CodecSupported())
+    return std::vector<SdpVideoFormat>();
+
+  std::vector<SdpVideoFormat> supportedCodecs = SupportedH264Codecs();
+
+  // OpenH264 doesn't yet support High Predictive 4:4:4 encoding but it does
+  // support decoding.
+  supportedCodecs.push_back(CreateH264Format(
+      H264Profile::kProfilePredictiveHigh444, H264Level::kLevel3_1, "1"));
+  supportedCodecs.push_back(CreateH264Format(
+      H264Profile::kProfilePredictiveHigh444, H264Level::kLevel3_1, "0"));
+
+  return supportedCodecs;
+}
+
+std::unique_ptr<H264Encoder> H264Encoder::Create(
+    const cricket::VideoCodec& codec) {
+  RTC_DCHECK(H264Encoder::IsSupported());
+#if defined(WEBRTC_USE_H264)
+  RTC_CHECK(g_rtc_use_h264);
+  RTC_LOG(LS_INFO) << "Creating H264EncoderImpl.";
+  return std::make_unique<H264EncoderImpl>(codec);
+#else
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+#endif
+}
+
+bool H264Encoder::IsSupported() {
+  return IsH264CodecSupported();
+}
+
+bool H264Encoder::SupportsScalabilityMode(ScalabilityMode scalability_mode) {
+  for (const auto& entry : kSupportedScalabilityModes) {
+    if (entry == scalability_mode) {
+      return true;
+    }
+  }
+  return false;
+}
+
+std::unique_ptr<H264Decoder> H264Decoder::Create() {
+  RTC_DCHECK(H264Decoder::IsSupported());
+#if defined(WEBRTC_USE_H264)
+  RTC_CHECK(g_rtc_use_h264);
+  RTC_LOG(LS_INFO) << "Creating H264DecoderImpl.";
+  return std::make_unique<H264DecoderImpl>();
+#else
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+#endif
+}
+
+bool H264Decoder::IsSupported() {
+  return IsH264CodecSupported();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_color_space.cc b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_color_space.cc
new file mode 100644
index 0000000000..59921263e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_color_space.cc
@@ -0,0 +1,178 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Everything declared/defined in this header is only required when WebRTC is
+// build with H264 support, please do not move anything out of the
+// #ifdef unless needed and tested.
+#ifdef WEBRTC_USE_H264
+
+#include "modules/video_coding/codecs/h264/h264_color_space.h"
+
+namespace webrtc {
+
+ColorSpace ExtractH264ColorSpace(AVCodecContext* codec) {
+  ColorSpace::PrimaryID primaries = ColorSpace::PrimaryID::kUnspecified;
+  switch (codec->color_primaries) {
+    case AVCOL_PRI_BT709:
+      primaries = ColorSpace::PrimaryID::kBT709;
+      break;
+    case AVCOL_PRI_BT470M:
+      primaries = ColorSpace::PrimaryID::kBT470M;
+      break;
+    case AVCOL_PRI_BT470BG:
+      primaries = ColorSpace::PrimaryID::kBT470BG;
+      break;
+    case AVCOL_PRI_SMPTE170M:
+      primaries = ColorSpace::PrimaryID::kSMPTE170M;
+      break;
+    case AVCOL_PRI_SMPTE240M:
+      primaries = ColorSpace::PrimaryID::kSMPTE240M;
+      break;
+    case AVCOL_PRI_FILM:
+      primaries = ColorSpace::PrimaryID::kFILM;
+      break;
+    case AVCOL_PRI_BT2020:
+      primaries = ColorSpace::PrimaryID::kBT2020;
+      break;
+    case AVCOL_PRI_SMPTE428:
+      primaries = ColorSpace::PrimaryID::kSMPTEST428;
+      break;
+    case AVCOL_PRI_SMPTE431:
+      primaries = ColorSpace::PrimaryID::kSMPTEST431;
+      break;
+    case AVCOL_PRI_SMPTE432:
+      primaries = ColorSpace::PrimaryID::kSMPTEST432;
+      break;
+    case AVCOL_PRI_JEDEC_P22:
+      primaries = ColorSpace::PrimaryID::kJEDECP22;
+      break;
+    case AVCOL_PRI_RESERVED0:
+    case AVCOL_PRI_UNSPECIFIED:
+    case AVCOL_PRI_RESERVED:
+    default:
+      break;
+  }
+
+  ColorSpace::TransferID transfer = ColorSpace::TransferID::kUnspecified;
+  switch (codec->color_trc) {
+    case AVCOL_TRC_BT709:
+      transfer = ColorSpace::TransferID::kBT709;
+      break;
+    case AVCOL_TRC_GAMMA22:
+      transfer = ColorSpace::TransferID::kGAMMA22;
+      break;
+    case AVCOL_TRC_GAMMA28:
+      transfer = ColorSpace::TransferID::kGAMMA28;
+      break;
+    case AVCOL_TRC_SMPTE170M:
+      transfer = ColorSpace::TransferID::kSMPTE170M;
+      break;
+    case AVCOL_TRC_SMPTE240M:
+      transfer = ColorSpace::TransferID::kSMPTE240M;
+      break;
+    case AVCOL_TRC_LINEAR:
+      transfer = ColorSpace::TransferID::kLINEAR;
+      break;
+    case AVCOL_TRC_LOG:
+      transfer = ColorSpace::TransferID::kLOG;
+      break;
+    case AVCOL_TRC_LOG_SQRT:
+      transfer = ColorSpace::TransferID::kLOG_SQRT;
+      break;
+    case AVCOL_TRC_IEC61966_2_4:
+      transfer = ColorSpace::TransferID::kIEC61966_2_4;
+      break;
+    case AVCOL_TRC_BT1361_ECG:
+      transfer = ColorSpace::TransferID::kBT1361_ECG;
+      break;
+    case AVCOL_TRC_IEC61966_2_1:
+      transfer = ColorSpace::TransferID::kIEC61966_2_1;
+      break;
+    case AVCOL_TRC_BT2020_10:
+      transfer = ColorSpace::TransferID::kBT2020_10;
+      break;
+    case AVCOL_TRC_BT2020_12:
+      transfer = ColorSpace::TransferID::kBT2020_12;
+      break;
+    case AVCOL_TRC_SMPTE2084:
+      transfer = ColorSpace::TransferID::kSMPTEST2084;
+      break;
+    case AVCOL_TRC_SMPTE428:
+      transfer = ColorSpace::TransferID::kSMPTEST428;
+      break;
+    case AVCOL_TRC_ARIB_STD_B67:
+      transfer = ColorSpace::TransferID::kARIB_STD_B67;
+      break;
+    case AVCOL_TRC_RESERVED0:
+    case AVCOL_TRC_UNSPECIFIED:
+    case AVCOL_TRC_RESERVED:
+    default:
+      break;
+  }
+
+  ColorSpace::MatrixID matrix = ColorSpace::MatrixID::kUnspecified;
+  switch (codec->colorspace) {
+    case AVCOL_SPC_RGB:
+      matrix = ColorSpace::MatrixID::kRGB;
+      break;
+    case AVCOL_SPC_BT709:
+      matrix = ColorSpace::MatrixID::kBT709;
+      break;
+    case AVCOL_SPC_FCC:
+      matrix = ColorSpace::MatrixID::kFCC;
+      break;
+    case AVCOL_SPC_BT470BG:
+      matrix = ColorSpace::MatrixID::kBT470BG;
+      break;
+    case AVCOL_SPC_SMPTE170M:
+      matrix = ColorSpace::MatrixID::kSMPTE170M;
+      break;
+    case AVCOL_SPC_SMPTE240M:
+      matrix = ColorSpace::MatrixID::kSMPTE240M;
+      break;
+    case AVCOL_SPC_YCGCO:
+      matrix = ColorSpace::MatrixID::kYCOCG;
+      break;
+    case AVCOL_SPC_BT2020_NCL:
+      matrix = ColorSpace::MatrixID::kBT2020_NCL;
+      break;
+    case AVCOL_SPC_BT2020_CL:
+      matrix = ColorSpace::MatrixID::kBT2020_CL;
+      break;
+    case AVCOL_SPC_SMPTE2085:
+      matrix = ColorSpace::MatrixID::kSMPTE2085;
+      break;
+    case AVCOL_SPC_CHROMA_DERIVED_NCL:
+    case AVCOL_SPC_CHROMA_DERIVED_CL:
+    case AVCOL_SPC_ICTCP:
+    case AVCOL_SPC_UNSPECIFIED:
+    case AVCOL_SPC_RESERVED:
+    default:
+      break;
+  }
+
+  ColorSpace::RangeID range = ColorSpace::RangeID::kInvalid;
+  switch (codec->color_range) {
+    case AVCOL_RANGE_MPEG:
+      range = ColorSpace::RangeID::kLimited;
+      break;
+    case AVCOL_RANGE_JPEG:
+      range = ColorSpace::RangeID::kFull;
+      break;
+    case AVCOL_RANGE_UNSPECIFIED:
+    default:
+      break;
+  }
+  return ColorSpace(primaries, transfer, matrix, range);
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_USE_H264
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_color_space.h b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_color_space.h
new file mode 100644
index 0000000000..392ccaf563
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_color_space.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_H264_H264_COLOR_SPACE_H_
+#define MODULES_VIDEO_CODING_CODECS_H264_H264_COLOR_SPACE_H_
+
+// Everything declared in this header is only required when WebRTC is
+// build with H264 support, please do not move anything out of the
+// #ifdef unless needed and tested.
+#ifdef WEBRTC_USE_H264
+
+#if defined(WEBRTC_WIN) && !defined(__clang__)
+#error "See: bugs.webrtc.org/9213#c13."
+#endif
+
+#include "api/video/color_space.h"
+
+extern "C" {
+#include "third_party/ffmpeg/libavcodec/avcodec.h"
+}  // extern "C"
+
+namespace webrtc {
+
+// Helper class for extracting color space information from H264 stream.
+ColorSpace ExtractH264ColorSpace(AVCodecContext* codec);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_USE_H264
+
+#endif  // MODULES_VIDEO_CODING_CODECS_H264_H264_COLOR_SPACE_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
new file mode 100644
index 0000000000..f67718cb23
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
@@ -0,0 +1,657 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+// Everything declared/defined in this header is only required when WebRTC is
+// build with H264 support, please do not move anything out of the
+// #ifdef unless needed and tested.
+#ifdef WEBRTC_USE_H264
+
+#include "modules/video_coding/codecs/h264/h264_decoder_impl.h"
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+
+extern "C" {
+#include "third_party/ffmpeg/libavcodec/avcodec.h"
+#include "third_party/ffmpeg/libavformat/avformat.h"
+#include "third_party/ffmpeg/libavutil/imgutils.h"
+}  // extern "C"
+
+#include "api/video/color_space.h"
+#include "api/video/i010_buffer.h"
+#include "api/video/i420_buffer.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "modules/video_coding/codecs/h264/h264_color_space.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr std::array<AVPixelFormat, 9> kPixelFormatsSupported = {
+    AV_PIX_FMT_YUV420P,     AV_PIX_FMT_YUV422P,     AV_PIX_FMT_YUV444P,
+    AV_PIX_FMT_YUVJ420P,    AV_PIX_FMT_YUVJ422P,    AV_PIX_FMT_YUVJ444P,
+    AV_PIX_FMT_YUV420P10LE, AV_PIX_FMT_YUV422P10LE, AV_PIX_FMT_YUV444P10LE};
+const size_t kYPlaneIndex = 0;
+const size_t kUPlaneIndex = 1;
+const size_t kVPlaneIndex = 2;
+
+// Used by histograms. Values of entries should not be changed.
+enum H264DecoderImplEvent {
+  kH264DecoderEventInit = 0,
+  kH264DecoderEventError = 1,
+  kH264DecoderEventMax = 16,
+};
+
+struct ScopedPtrAVFreePacket {
+  void operator()(AVPacket* packet) { av_packet_free(&packet); }
+};
+typedef std::unique_ptr<AVPacket, ScopedPtrAVFreePacket> ScopedAVPacket;
+
+ScopedAVPacket MakeScopedAVPacket() {
+  ScopedAVPacket packet(av_packet_alloc());
+  return packet;
+}
+
+}  // namespace
+
+int H264DecoderImpl::AVGetBuffer2(AVCodecContext* context,
+                                  AVFrame* av_frame,
+                                  int flags) {
+  // Set in `Configure`.
+  H264DecoderImpl* decoder = static_cast<H264DecoderImpl*>(context->opaque);
+  // DCHECK values set in `Configure`.
+  RTC_DCHECK(decoder);
+  // Necessary capability to be allowed to provide our own buffers.
+  RTC_DCHECK(context->codec->capabilities | AV_CODEC_CAP_DR1);
+
+  auto pixelFormatSupported = std::find_if(
+      kPixelFormatsSupported.begin(), kPixelFormatsSupported.end(),
+      [context](AVPixelFormat format) { return context->pix_fmt == format; });
+
+  RTC_CHECK(pixelFormatSupported != kPixelFormatsSupported.end());
+
+  // `av_frame->width` and `av_frame->height` are set by FFmpeg. These are the
+  // actual image's dimensions and may be different from `context->width` and
+  // `context->coded_width` due to reordering.
+  int width = av_frame->width;
+  int height = av_frame->height;
+  // See `lowres`, if used the decoder scales the image by 1/2^(lowres). This
+  // has implications on which resolutions are valid, but we don't use it.
+  RTC_CHECK_EQ(context->lowres, 0);
+  // Adjust the `width` and `height` to values acceptable by the decoder.
+  // Without this, FFmpeg may overflow the buffer. If modified, `width` and/or
+  // `height` are larger than the actual image and the image has to be cropped
+  // (top-left corner) after decoding to avoid visible borders to the right and
+  // bottom of the actual image.
+  avcodec_align_dimensions(context, &width, &height);
+
+  RTC_CHECK_GE(width, 0);
+  RTC_CHECK_GE(height, 0);
+  int ret = av_image_check_size(static_cast<unsigned int>(width),
+                                static_cast<unsigned int>(height), 0, nullptr);
+  if (ret < 0) {
+    RTC_LOG(LS_ERROR) << "Invalid picture size " << width << "x" << height;
+    decoder->ReportError();
+    return ret;
+  }
+
+  // The video frame is stored in `frame_buffer`. `av_frame` is FFmpeg's version
+  // of a video frame and will be set up to reference `frame_buffer`'s data.
+
+  // FFmpeg expects the initial allocation to be zero-initialized according to
+  // http://crbug.com/390941. Our pool is set up to zero-initialize new buffers.
+  // TODO(https://crbug.com/390941): Delete that feature from the video pool,
+  // instead add an explicit call to InitializeData here.
+  rtc::scoped_refptr<PlanarYuvBuffer> frame_buffer;
+  rtc::scoped_refptr<I444Buffer> i444_buffer;
+  rtc::scoped_refptr<I420Buffer> i420_buffer;
+  rtc::scoped_refptr<I422Buffer> i422_buffer;
+  rtc::scoped_refptr<I010Buffer> i010_buffer;
+  rtc::scoped_refptr<I210Buffer> i210_buffer;
+  rtc::scoped_refptr<I410Buffer> i410_buffer;
+  int bytes_per_pixel = 1;
+  switch (context->pix_fmt) {
+    case AV_PIX_FMT_YUV420P:
+    case AV_PIX_FMT_YUVJ420P:
+      i420_buffer =
+          decoder->ffmpeg_buffer_pool_.CreateI420Buffer(width, height);
+      // Set `av_frame` members as required by FFmpeg.
+      av_frame->data[kYPlaneIndex] = i420_buffer->MutableDataY();
+      av_frame->linesize[kYPlaneIndex] = i420_buffer->StrideY();
+      av_frame->data[kUPlaneIndex] = i420_buffer->MutableDataU();
+      av_frame->linesize[kUPlaneIndex] = i420_buffer->StrideU();
+      av_frame->data[kVPlaneIndex] = i420_buffer->MutableDataV();
+      av_frame->linesize[kVPlaneIndex] = i420_buffer->StrideV();
+      RTC_DCHECK_EQ(av_frame->extended_data, av_frame->data);
+      frame_buffer = i420_buffer;
+      break;
+    case AV_PIX_FMT_YUV444P:
+    case AV_PIX_FMT_YUVJ444P:
+      i444_buffer =
+          decoder->ffmpeg_buffer_pool_.CreateI444Buffer(width, height);
+      // Set `av_frame` members as required by FFmpeg.
+      av_frame->data[kYPlaneIndex] = i444_buffer->MutableDataY();
+      av_frame->linesize[kYPlaneIndex] = i444_buffer->StrideY();
+      av_frame->data[kUPlaneIndex] = i444_buffer->MutableDataU();
+      av_frame->linesize[kUPlaneIndex] = i444_buffer->StrideU();
+      av_frame->data[kVPlaneIndex] = i444_buffer->MutableDataV();
+      av_frame->linesize[kVPlaneIndex] = i444_buffer->StrideV();
+      frame_buffer = i444_buffer;
+      break;
+    case AV_PIX_FMT_YUV422P:
+    case AV_PIX_FMT_YUVJ422P:
+      i422_buffer =
+          decoder->ffmpeg_buffer_pool_.CreateI422Buffer(width, height);
+      // Set `av_frame` members as required by FFmpeg.
+      av_frame->data[kYPlaneIndex] = i422_buffer->MutableDataY();
+      av_frame->linesize[kYPlaneIndex] = i422_buffer->StrideY();
+      av_frame->data[kUPlaneIndex] = i422_buffer->MutableDataU();
+      av_frame->linesize[kUPlaneIndex] = i422_buffer->StrideU();
+      av_frame->data[kVPlaneIndex] = i422_buffer->MutableDataV();
+      av_frame->linesize[kVPlaneIndex] = i422_buffer->StrideV();
+      frame_buffer = i422_buffer;
+      break;
+    case AV_PIX_FMT_YUV420P10LE:
+      i010_buffer =
+          decoder->ffmpeg_buffer_pool_.CreateI010Buffer(width, height);
+      // Set `av_frame` members as required by FFmpeg.
+      av_frame->data[kYPlaneIndex] =
+          reinterpret_cast<uint8_t*>(i010_buffer->MutableDataY());
+      av_frame->linesize[kYPlaneIndex] = i010_buffer->StrideY() * 2;
+      av_frame->data[kUPlaneIndex] =
+          reinterpret_cast<uint8_t*>(i010_buffer->MutableDataU());
+      av_frame->linesize[kUPlaneIndex] = i010_buffer->StrideU() * 2;
+      av_frame->data[kVPlaneIndex] =
+          reinterpret_cast<uint8_t*>(i010_buffer->MutableDataV());
+      av_frame->linesize[kVPlaneIndex] = i010_buffer->StrideV() * 2;
+      frame_buffer = i010_buffer;
+      bytes_per_pixel = 2;
+      break;
+    case AV_PIX_FMT_YUV422P10LE:
+      i210_buffer =
+          decoder->ffmpeg_buffer_pool_.CreateI210Buffer(width, height);
+      // Set `av_frame` members as required by FFmpeg.
+      av_frame->data[kYPlaneIndex] =
+          reinterpret_cast<uint8_t*>(i210_buffer->MutableDataY());
+      av_frame->linesize[kYPlaneIndex] = i210_buffer->StrideY() * 2;
+      av_frame->data[kUPlaneIndex] =
+          reinterpret_cast<uint8_t*>(i210_buffer->MutableDataU());
+      av_frame->linesize[kUPlaneIndex] = i210_buffer->StrideU() * 2;
+      av_frame->data[kVPlaneIndex] =
+          reinterpret_cast<uint8_t*>(i210_buffer->MutableDataV());
+      av_frame->linesize[kVPlaneIndex] = i210_buffer->StrideV() * 2;
+      frame_buffer = i210_buffer;
+      bytes_per_pixel = 2;
+      break;
+    case AV_PIX_FMT_YUV444P10LE:
+      i410_buffer =
+          decoder->ffmpeg_buffer_pool_.CreateI410Buffer(width, height);
+      // Set `av_frame` members as required by FFmpeg.
+      av_frame->data[kYPlaneIndex] =
+          reinterpret_cast<uint8_t*>(i410_buffer->MutableDataY());
+      av_frame->linesize[kYPlaneIndex] = i410_buffer->StrideY() * 2;
+      av_frame->data[kUPlaneIndex] =
+          reinterpret_cast<uint8_t*>(i410_buffer->MutableDataU());
+      av_frame->linesize[kUPlaneIndex] = i410_buffer->StrideU() * 2;
+      av_frame->data[kVPlaneIndex] =
+          reinterpret_cast<uint8_t*>(i410_buffer->MutableDataV());
+      av_frame->linesize[kVPlaneIndex] = i410_buffer->StrideV() * 2;
+      frame_buffer = i410_buffer;
+      bytes_per_pixel = 2;
+      break;
+    default:
+      RTC_LOG(LS_ERROR) << "Unsupported buffer type " << context->pix_fmt
+                        << ". Check supported supported pixel formats!";
+      decoder->ReportError();
+      return -1;
+  }
+
+  int y_size = width * height * bytes_per_pixel;
+  int uv_size = frame_buffer->ChromaWidth() * frame_buffer->ChromaHeight() *
+                bytes_per_pixel;
+  // DCHECK that we have a continuous buffer as is required.
+  RTC_DCHECK_EQ(av_frame->data[kUPlaneIndex],
+                av_frame->data[kYPlaneIndex] + y_size);
+  RTC_DCHECK_EQ(av_frame->data[kVPlaneIndex],
+                av_frame->data[kUPlaneIndex] + uv_size);
+  int total_size = y_size + 2 * uv_size;
+
+  av_frame->format = context->pix_fmt;
+  av_frame->reordered_opaque = context->reordered_opaque;
+
+  // Create a VideoFrame object, to keep a reference to the buffer.
+  // TODO(nisse): The VideoFrame's timestamp and rotation info is not used.
+  // Refactor to do not use a VideoFrame object at all.
+  av_frame->buf[0] = av_buffer_create(
+      av_frame->data[kYPlaneIndex], total_size, AVFreeBuffer2,
+      static_cast<void*>(
+          std::make_unique<VideoFrame>(VideoFrame::Builder()
+                                           .set_video_frame_buffer(frame_buffer)
+                                           .set_rotation(kVideoRotation_0)
+                                           .set_timestamp_us(0)
+                                           .build())
+              .release()),
+      0);
+  RTC_CHECK(av_frame->buf[0]);
+  return 0;
+}
+
+void H264DecoderImpl::AVFreeBuffer2(void* opaque, uint8_t* data) {
+  // The buffer pool recycles the buffer used by `video_frame` when there are no
+  // more references to it. `video_frame` is a thin buffer holder and is not
+  // recycled.
+  VideoFrame* video_frame = static_cast<VideoFrame*>(opaque);
+  delete video_frame;
+}
+
+H264DecoderImpl::H264DecoderImpl()
+    : ffmpeg_buffer_pool_(true),
+      decoded_image_callback_(nullptr),
+      has_reported_init_(false),
+      has_reported_error_(false) {}
+
+H264DecoderImpl::~H264DecoderImpl() {
+  Release();
+}
+
+bool H264DecoderImpl::Configure(const Settings& settings) {
+  ReportInit();
+  if (settings.codec_type() != kVideoCodecH264) {
+    ReportError();
+    return false;
+  }
+
+  // Release necessary in case of re-initializing.
+  int32_t ret = Release();
+  if (ret != WEBRTC_VIDEO_CODEC_OK) {
+    ReportError();
+    return false;
+  }
+  RTC_DCHECK(!av_context_);
+
+  // Initialize AVCodecContext.
+  av_context_.reset(avcodec_alloc_context3(nullptr));
+
+  av_context_->codec_type = AVMEDIA_TYPE_VIDEO;
+  av_context_->codec_id = AV_CODEC_ID_H264;
+  const RenderResolution& resolution = settings.max_render_resolution();
+  if (resolution.Valid()) {
+    av_context_->coded_width = resolution.Width();
+    av_context_->coded_height = resolution.Height();
+  }
+  av_context_->extradata = nullptr;
+  av_context_->extradata_size = 0;
+
+  // If this is ever increased, look at `av_context_->thread_safe_callbacks` and
+  // make it possible to disable the thread checker in the frame buffer pool.
+  av_context_->thread_count = 1;
+  av_context_->thread_type = FF_THREAD_SLICE;
+
+  // Function used by FFmpeg to get buffers to store decoded frames in.
+  av_context_->get_buffer2 = AVGetBuffer2;
+  // `get_buffer2` is called with the context, there `opaque` can be used to get
+  // a pointer `this`.
+  av_context_->opaque = this;
+
+  const AVCodec* codec = avcodec_find_decoder(av_context_->codec_id);
+  if (!codec) {
+    // This is an indication that FFmpeg has not been initialized or it has not
+    // been compiled/initialized with the correct set of codecs.
+    RTC_LOG(LS_ERROR) << "FFmpeg H.264 decoder not found.";
+    Release();
+    ReportError();
+    return false;
+  }
+  int res = avcodec_open2(av_context_.get(), codec, nullptr);
+  if (res < 0) {
+    RTC_LOG(LS_ERROR) << "avcodec_open2 error: " << res;
+    Release();
+    ReportError();
+    return false;
+  }
+
+  av_frame_.reset(av_frame_alloc());
+
+  if (absl::optional<int> buffer_pool_size = settings.buffer_pool_size()) {
+    if (!ffmpeg_buffer_pool_.Resize(*buffer_pool_size)) {
+      return false;
+    }
+  }
+  return true;
+}
+
+int32_t H264DecoderImpl::Release() {
+  av_context_.reset();
+  av_frame_.reset();
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264DecoderImpl::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  decoded_image_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
+                                bool /*missing_frames*/,
+                                int64_t /*render_time_ms*/) {
+  if (!IsInitialized()) {
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (!decoded_image_callback_) {
+    RTC_LOG(LS_WARNING)
+        << "Configure() has been called, but a callback function "
+           "has not been set with RegisterDecodeCompleteCallback()";
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (!input_image.data() || !input_image.size()) {
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  ScopedAVPacket packet = MakeScopedAVPacket();
+  if (!packet) {
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  // packet.data has a non-const type, but isn't modified by
+  // avcodec_send_packet.
+  packet->data = const_cast<uint8_t*>(input_image.data());
+  if (input_image.size() >
+      static_cast<size_t>(std::numeric_limits<int>::max())) {
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  packet->size = static_cast<int>(input_image.size());
+  int64_t frame_timestamp_us = input_image.ntp_time_ms_ * 1000;  // ms -> μs
+  av_context_->reordered_opaque = frame_timestamp_us;
+
+  int result = avcodec_send_packet(av_context_.get(), packet.get());
+
+  if (result < 0) {
+    RTC_LOG(LS_ERROR) << "avcodec_send_packet error: " << result;
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  result = avcodec_receive_frame(av_context_.get(), av_frame_.get());
+  if (result < 0) {
+    RTC_LOG(LS_ERROR) << "avcodec_receive_frame error: " << result;
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  // We don't expect reordering. Decoded frame timestamp should match
+  // the input one.
+  RTC_DCHECK_EQ(av_frame_->reordered_opaque, frame_timestamp_us);
+
+  // TODO(sakal): Maybe it is possible to get QP directly from FFmpeg.
+  h264_bitstream_parser_.ParseBitstream(input_image);
+  absl::optional<int> qp = h264_bitstream_parser_.GetLastSliceQp();
+
+  // Obtain the `video_frame` containing the decoded image.
+  VideoFrame* input_frame =
+      static_cast<VideoFrame*>(av_buffer_get_opaque(av_frame_->buf[0]));
+  RTC_DCHECK(input_frame);
+  rtc::scoped_refptr<VideoFrameBuffer> frame_buffer =
+      input_frame->video_frame_buffer();
+
+  // Instantiate Planar YUV buffer according to video frame buffer type
+  const webrtc::PlanarYuvBuffer* planar_yuv_buffer = nullptr;
+  const webrtc::PlanarYuv8Buffer* planar_yuv8_buffer = nullptr;
+  const webrtc::PlanarYuv16BBuffer* planar_yuv16_buffer = nullptr;
+  VideoFrameBuffer::Type video_frame_buffer_type = frame_buffer->type();
+  switch (video_frame_buffer_type) {
+    case VideoFrameBuffer::Type::kI420:
+      planar_yuv_buffer = frame_buffer->GetI420();
+      planar_yuv8_buffer =
+          reinterpret_cast<const webrtc::PlanarYuv8Buffer*>(planar_yuv_buffer);
+      break;
+    case VideoFrameBuffer::Type::kI444:
+      planar_yuv_buffer = frame_buffer->GetI444();
+      planar_yuv8_buffer =
+          reinterpret_cast<const webrtc::PlanarYuv8Buffer*>(planar_yuv_buffer);
+      break;
+    case VideoFrameBuffer::Type::kI422:
+      planar_yuv_buffer = frame_buffer->GetI422();
+      planar_yuv8_buffer =
+          reinterpret_cast<const webrtc::PlanarYuv8Buffer*>(planar_yuv_buffer);
+      break;
+    case VideoFrameBuffer::Type::kI010:
+      planar_yuv_buffer = frame_buffer->GetI010();
+      planar_yuv16_buffer = reinterpret_cast<const webrtc::PlanarYuv16BBuffer*>(
+          planar_yuv_buffer);
+      break;
+    case VideoFrameBuffer::Type::kI210:
+      planar_yuv_buffer = frame_buffer->GetI210();
+      planar_yuv16_buffer = reinterpret_cast<const webrtc::PlanarYuv16BBuffer*>(
+          planar_yuv_buffer);
+      break;
+    case VideoFrameBuffer::Type::kI410:
+      planar_yuv_buffer = frame_buffer->GetI410();
+      planar_yuv16_buffer = reinterpret_cast<const webrtc::PlanarYuv16BBuffer*>(
+          planar_yuv_buffer);
+      break;
+    default:
+      // If this code is changed to allow other video frame buffer type,
+      // make sure that the code below which wraps I420/I422/I444 buffer and
+      // code which converts to NV12 is changed
+      // to work with new video frame buffer type
+
+      RTC_LOG(LS_ERROR) << "frame_buffer type: "
+                        << static_cast<int32_t>(video_frame_buffer_type)
+                        << " is not supported!";
+      ReportError();
+      return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  // When needed, FFmpeg applies cropping by moving plane pointers and adjusting
+  // frame width/height. Ensure that cropped buffers lie within the allocated
+  // memory.
+  RTC_DCHECK_LE(av_frame_->width, planar_yuv_buffer->width());
+  RTC_DCHECK_LE(av_frame_->height, planar_yuv_buffer->height());
+  switch (video_frame_buffer_type) {
+    case VideoFrameBuffer::Type::kI420:
+    case VideoFrameBuffer::Type::kI444:
+    case VideoFrameBuffer::Type::kI422: {
+      RTC_DCHECK_GE(av_frame_->data[kYPlaneIndex], planar_yuv8_buffer->DataY());
+      RTC_DCHECK_LE(
+          av_frame_->data[kYPlaneIndex] +
+              av_frame_->linesize[kYPlaneIndex] * av_frame_->height,
+          planar_yuv8_buffer->DataY() +
+              planar_yuv8_buffer->StrideY() * planar_yuv8_buffer->height());
+      RTC_DCHECK_GE(av_frame_->data[kUPlaneIndex], planar_yuv8_buffer->DataU());
+      RTC_DCHECK_LE(
+          av_frame_->data[kUPlaneIndex] +
+              av_frame_->linesize[kUPlaneIndex] *
+                  planar_yuv8_buffer->ChromaHeight(),
+          planar_yuv8_buffer->DataU() + planar_yuv8_buffer->StrideU() *
+                                            planar_yuv8_buffer->ChromaHeight());
+      RTC_DCHECK_GE(av_frame_->data[kVPlaneIndex], planar_yuv8_buffer->DataV());
+      RTC_DCHECK_LE(
+          av_frame_->data[kVPlaneIndex] +
+              av_frame_->linesize[kVPlaneIndex] *
+                  planar_yuv8_buffer->ChromaHeight(),
+          planar_yuv8_buffer->DataV() + planar_yuv8_buffer->StrideV() *
+                                            planar_yuv8_buffer->ChromaHeight());
+      break;
+    }
+    case VideoFrameBuffer::Type::kI010:
+    case VideoFrameBuffer::Type::kI210:
+    case VideoFrameBuffer::Type::kI410: {
+      RTC_DCHECK_GE(
+          av_frame_->data[kYPlaneIndex],
+          reinterpret_cast<const uint8_t*>(planar_yuv16_buffer->DataY()));
+      RTC_DCHECK_LE(
+          av_frame_->data[kYPlaneIndex] +
+              av_frame_->linesize[kYPlaneIndex] * av_frame_->height,
+          reinterpret_cast<const uint8_t*>(planar_yuv16_buffer->DataY()) +
+              planar_yuv16_buffer->StrideY() * 2 *
+                  planar_yuv16_buffer->height());
+      RTC_DCHECK_GE(
+          av_frame_->data[kUPlaneIndex],
+          reinterpret_cast<const uint8_t*>(planar_yuv16_buffer->DataU()));
+      RTC_DCHECK_LE(
+          av_frame_->data[kUPlaneIndex] +
+              av_frame_->linesize[kUPlaneIndex] *
+                  planar_yuv16_buffer->ChromaHeight(),
+          reinterpret_cast<const uint8_t*>(planar_yuv16_buffer->DataU()) +
+              planar_yuv16_buffer->StrideU() * 2 *
+                  planar_yuv16_buffer->ChromaHeight());
+      RTC_DCHECK_GE(
+          av_frame_->data[kVPlaneIndex],
+          reinterpret_cast<const uint8_t*>(planar_yuv16_buffer->DataV()));
+      RTC_DCHECK_LE(
+          av_frame_->data[kVPlaneIndex] +
+              av_frame_->linesize[kVPlaneIndex] *
+                  planar_yuv16_buffer->ChromaHeight(),
+          reinterpret_cast<const uint8_t*>(planar_yuv16_buffer->DataV()) +
+              planar_yuv16_buffer->StrideV() * 2 *
+                  planar_yuv16_buffer->ChromaHeight());
+      break;
+    }
+    default:
+      RTC_LOG(LS_ERROR) << "frame_buffer type: "
+                        << static_cast<int32_t>(video_frame_buffer_type)
+                        << " is not supported!";
+      ReportError();
+      return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> cropped_buffer;
+  switch (video_frame_buffer_type) {
+    case VideoFrameBuffer::Type::kI420:
+      cropped_buffer = WrapI420Buffer(
+          av_frame_->width, av_frame_->height, av_frame_->data[kYPlaneIndex],
+          av_frame_->linesize[kYPlaneIndex], av_frame_->data[kUPlaneIndex],
+          av_frame_->linesize[kUPlaneIndex], av_frame_->data[kVPlaneIndex],
+          av_frame_->linesize[kVPlaneIndex],
+          // To keep reference alive.
+          [frame_buffer] {});
+      break;
+    case VideoFrameBuffer::Type::kI444:
+      cropped_buffer = WrapI444Buffer(
+          av_frame_->width, av_frame_->height, av_frame_->data[kYPlaneIndex],
+          av_frame_->linesize[kYPlaneIndex], av_frame_->data[kUPlaneIndex],
+          av_frame_->linesize[kUPlaneIndex], av_frame_->data[kVPlaneIndex],
+          av_frame_->linesize[kVPlaneIndex],
+          // To keep reference alive.
+          [frame_buffer] {});
+      break;
+    case VideoFrameBuffer::Type::kI422:
+      cropped_buffer = WrapI422Buffer(
+          av_frame_->width, av_frame_->height, av_frame_->data[kYPlaneIndex],
+          av_frame_->linesize[kYPlaneIndex], av_frame_->data[kUPlaneIndex],
+          av_frame_->linesize[kUPlaneIndex], av_frame_->data[kVPlaneIndex],
+          av_frame_->linesize[kVPlaneIndex],
+          // To keep reference alive.
+          [frame_buffer] {});
+      break;
+    case VideoFrameBuffer::Type::kI010:
+      cropped_buffer = WrapI010Buffer(
+          av_frame_->width, av_frame_->height,
+          reinterpret_cast<const uint16_t*>(av_frame_->data[kYPlaneIndex]),
+          av_frame_->linesize[kYPlaneIndex] / 2,
+          reinterpret_cast<const uint16_t*>(av_frame_->data[kUPlaneIndex]),
+          av_frame_->linesize[kUPlaneIndex] / 2,
+          reinterpret_cast<const uint16_t*>(av_frame_->data[kVPlaneIndex]),
+          av_frame_->linesize[kVPlaneIndex] / 2,
+          // To keep reference alive.
+          [frame_buffer] {});
+      break;
+    case VideoFrameBuffer::Type::kI210:
+      cropped_buffer = WrapI210Buffer(
+          av_frame_->width, av_frame_->height,
+          reinterpret_cast<const uint16_t*>(av_frame_->data[kYPlaneIndex]),
+          av_frame_->linesize[kYPlaneIndex] / 2,
+          reinterpret_cast<const uint16_t*>(av_frame_->data[kUPlaneIndex]),
+          av_frame_->linesize[kUPlaneIndex] / 2,
+          reinterpret_cast<const uint16_t*>(av_frame_->data[kVPlaneIndex]),
+          av_frame_->linesize[kVPlaneIndex] / 2,
+          // To keep reference alive.
+          [frame_buffer] {});
+      break;
+    case VideoFrameBuffer::Type::kI410:
+      cropped_buffer = WrapI410Buffer(
+          av_frame_->width, av_frame_->height,
+          reinterpret_cast<const uint16_t*>(av_frame_->data[kYPlaneIndex]),
+          av_frame_->linesize[kYPlaneIndex] / 2,
+          reinterpret_cast<const uint16_t*>(av_frame_->data[kUPlaneIndex]),
+          av_frame_->linesize[kUPlaneIndex] / 2,
+          reinterpret_cast<const uint16_t*>(av_frame_->data[kVPlaneIndex]),
+          av_frame_->linesize[kVPlaneIndex] / 2,
+          // To keep reference alive.
+          [frame_buffer] {});
+      break;
+    default:
+      RTC_LOG(LS_ERROR) << "frame_buffer type: "
+                        << static_cast<int32_t>(video_frame_buffer_type)
+                        << " is not supported!";
+      ReportError();
+      return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  // Pass on color space from input frame if explicitly specified.
+  const ColorSpace& color_space =
+      input_image.ColorSpace() ? *input_image.ColorSpace()
+                               : ExtractH264ColorSpace(av_context_.get());
+
+  VideoFrame decoded_frame = VideoFrame::Builder()
+                                 .set_video_frame_buffer(cropped_buffer)
+                                 .set_timestamp_rtp(input_image.Timestamp())
+                                 .set_color_space(color_space)
+                                 .build();
+
+  // Return decoded frame.
+  // TODO(nisse): Timestamp and rotation are all zero here. Change decoder
+  // interface to pass a VideoFrameBuffer instead of a VideoFrame?
+  decoded_image_callback_->Decoded(decoded_frame, absl::nullopt, qp);
+
+  // Stop referencing it, possibly freeing `input_frame`.
+  av_frame_unref(av_frame_.get());
+  input_frame = nullptr;
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+const char* H264DecoderImpl::ImplementationName() const {
+  return "FFmpeg";
+}
+
+bool H264DecoderImpl::IsInitialized() const {
+  return av_context_ != nullptr;
+}
+
+void H264DecoderImpl::ReportInit() {
+  if (has_reported_init_)
+    return;
+  RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264DecoderImpl.Event",
+                            kH264DecoderEventInit, kH264DecoderEventMax);
+  has_reported_init_ = true;
+}
+
+void H264DecoderImpl::ReportError() {
+  if (has_reported_error_)
+    return;
+  RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264DecoderImpl.Event",
+                            kH264DecoderEventError, kH264DecoderEventMax);
+  has_reported_error_ = true;
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_USE_H264
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h
new file mode 100644
index 0000000000..97d091cf4b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
+#define MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
+
+// Everything declared in this header is only required when WebRTC is
+// build with H264 support, please do not move anything out of the
+// #ifdef unless needed and tested.
+#ifdef WEBRTC_USE_H264
+
+#if defined(WEBRTC_WIN) && !defined(__clang__)
+#error "See: bugs.webrtc.org/9213#c13."
+#endif
+
+#include <memory>
+
+#include "modules/video_coding/codecs/h264/include/h264.h"
+
+// CAVEAT: According to ffmpeg docs for avcodec_send_packet, ffmpeg requires a
+// few extra padding bytes after the end of input. And in addition, docs for
+// AV_INPUT_BUFFER_PADDING_SIZE says "If the first 23 bits of the additional
+// bytes are not 0, then damaged MPEG bitstreams could cause overread and
+// segfault."
+//
+// WebRTC doesn't ensure any such padding, and REQUIRES ffmpeg to be compiled
+// with CONFIG_SAFE_BITSTREAM_READER, which is intended to eliminate
+// out-of-bounds reads. ffmpeg docs doesn't say explicitly what effects this
+// flag has on the h.264 decoder or avcodec_send_packet, though, so this is in
+// some way depending on undocumented behavior. If any problems turn up, we may
+// have to add an extra copy operation, to enforce padding before buffers are
+// passed to ffmpeg.
+
+extern "C" {
+#include "third_party/ffmpeg/libavcodec/avcodec.h"
+}  // extern "C"
+
+#include "common_video/h264/h264_bitstream_parser.h"
+#include "common_video/include/video_frame_buffer_pool.h"
+
+namespace webrtc {
+
+struct AVCodecContextDeleter {
+  void operator()(AVCodecContext* ptr) const { avcodec_free_context(&ptr); }
+};
+struct AVFrameDeleter {
+  void operator()(AVFrame* ptr) const { av_frame_free(&ptr); }
+};
+
+class H264DecoderImpl : public H264Decoder {
+ public:
+  H264DecoderImpl();
+  ~H264DecoderImpl() override;
+
+  bool Configure(const Settings& settings) override;
+  int32_t Release() override;
+
+  int32_t RegisterDecodeCompleteCallback(
+      DecodedImageCallback* callback) override;
+
+  // `missing_frames`, `fragmentation` and `render_time_ms` are ignored.
+  int32_t Decode(const EncodedImage& input_image,
+                 bool /*missing_frames*/,
+                 int64_t render_time_ms = -1) override;
+
+  const char* ImplementationName() const override;
+
+ private:
+  // Called by FFmpeg when it needs a frame buffer to store decoded frames in.
+  // The `VideoFrame` returned by FFmpeg at `Decode` originate from here. Their
+  // buffers are reference counted and freed by FFmpeg using `AVFreeBuffer2`.
+  static int AVGetBuffer2(AVCodecContext* context,
+                          AVFrame* av_frame,
+                          int flags);
+  // Called by FFmpeg when it is done with a video frame, see `AVGetBuffer2`.
+  static void AVFreeBuffer2(void* opaque, uint8_t* data);
+
+  bool IsInitialized() const;
+
+  // Reports statistics with histograms.
+  void ReportInit();
+  void ReportError();
+
+  // Used by ffmpeg via `AVGetBuffer2()` to allocate I420 images.
+  VideoFrameBufferPool ffmpeg_buffer_pool_;
+  std::unique_ptr<AVCodecContext, AVCodecContextDeleter> av_context_;
+  std::unique_ptr<AVFrame, AVFrameDeleter> av_frame_;
+
+  DecodedImageCallback* decoded_image_callback_;
+
+  bool has_reported_init_;
+  bool has_reported_error_;
+
+  webrtc::H264BitstreamParser h264_bitstream_parser_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_USE_H264
+
+#endif  // MODULES_VIDEO_CODING_CODECS_H264_H264_DECODER_IMPL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
new file mode 100644
index 0000000000..b8055ac85f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
@@ -0,0 +1,713 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+// Everything declared/defined in this header is only required when WebRTC is
+// build with H264 support, please do not move anything out of the
+// #ifdef unless needed and tested.
+#ifdef WEBRTC_USE_H264
+
+#include "modules/video_coding/codecs/h264/h264_encoder_impl.h"
+
+#include <algorithm>
+#include <limits>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
+#include "modules/video_coding/utility/simulcast_utility.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+#include "third_party/libyuv/include/libyuv/convert.h"
+#include "third_party/libyuv/include/libyuv/scale.h"
+#include "third_party/openh264/src/codec/api/wels/codec_api.h"
+#include "third_party/openh264/src/codec/api/wels/codec_app_def.h"
+#include "third_party/openh264/src/codec/api/wels/codec_def.h"
+#include "third_party/openh264/src/codec/api/wels/codec_ver.h"
+
+namespace webrtc {
+
+namespace {
+
+const bool kOpenH264EncoderDetailedLogging = false;
+
+// QP scaling thresholds.
+static const int kLowH264QpThreshold = 24;
+static const int kHighH264QpThreshold = 37;
+
+// Used by histograms. Values of entries should not be changed.
+enum H264EncoderImplEvent {
+  kH264EncoderEventInit = 0,
+  kH264EncoderEventError = 1,
+  kH264EncoderEventMax = 16,
+};
+
+int NumberOfThreads(int width, int height, int number_of_cores) {
+  // TODO(hbos): In Chromium, multiple threads do not work with sandbox on Mac,
+  // see crbug.com/583348. Until further investigated, only use one thread.
+  //  if (width * height >= 1920 * 1080 && number_of_cores > 8) {
+  //    return 8;  // 8 threads for 1080p on high perf machines.
+  //  } else if (width * height > 1280 * 960 && number_of_cores >= 6) {
+  //    return 3;  // 3 threads for 1080p.
+  //  } else if (width * height > 640 * 480 && number_of_cores >= 3) {
+  //    return 2;  // 2 threads for qHD/HD.
+  //  } else {
+  //    return 1;  // 1 thread for VGA or less.
+  //  }
+  // TODO(sprang): Also check sSliceArgument.uiSliceNum om GetEncoderPrams(),
+  //               before enabling multithreading here.
+  return 1;
+}
+
+VideoFrameType ConvertToVideoFrameType(EVideoFrameType type) {
+  switch (type) {
+    case videoFrameTypeIDR:
+      return VideoFrameType::kVideoFrameKey;
+    case videoFrameTypeSkip:
+    case videoFrameTypeI:
+    case videoFrameTypeP:
+    case videoFrameTypeIPMixed:
+      return VideoFrameType::kVideoFrameDelta;
+    case videoFrameTypeInvalid:
+      break;
+  }
+  RTC_DCHECK_NOTREACHED() << "Unexpected/invalid frame type: " << type;
+  return VideoFrameType::kEmptyFrame;
+}
+
+absl::optional<ScalabilityMode> ScalabilityModeFromTemporalLayers(
+    int num_temporal_layers) {
+  switch (num_temporal_layers) {
+    case 0:
+      break;
+    case 1:
+      return ScalabilityMode::kL1T1;
+    case 2:
+      return ScalabilityMode::kL1T2;
+    case 3:
+      return ScalabilityMode::kL1T3;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  return absl::nullopt;
+}
+
+}  // namespace
+
+// Helper method used by H264EncoderImpl::Encode.
+// Copies the encoded bytes from `info` to `encoded_image`. The
+// `encoded_image->_buffer` may be deleted and reallocated if a bigger buffer is
+// required.
+//
+// After OpenH264 encoding, the encoded bytes are stored in `info` spread out
+// over a number of layers and "NAL units". Each NAL unit is a fragment starting
+// with the four-byte start code {0,0,0,1}. All of this data (including the
+// start codes) is copied to the `encoded_image->_buffer`.
+static void RtpFragmentize(EncodedImage* encoded_image, SFrameBSInfo* info) {
+  // Calculate minimum buffer size required to hold encoded data.
+  size_t required_capacity = 0;
+  size_t fragments_count = 0;
+  for (int layer = 0; layer < info->iLayerNum; ++layer) {
+    const SLayerBSInfo& layerInfo = info->sLayerInfo[layer];
+    for (int nal = 0; nal < layerInfo.iNalCount; ++nal, ++fragments_count) {
+      RTC_CHECK_GE(layerInfo.pNalLengthInByte[nal], 0);
+      // Ensure `required_capacity` will not overflow.
+      RTC_CHECK_LE(layerInfo.pNalLengthInByte[nal],
+                   std::numeric_limits<size_t>::max() - required_capacity);
+      required_capacity += layerInfo.pNalLengthInByte[nal];
+    }
+  }
+  auto buffer = EncodedImageBuffer::Create(required_capacity);
+  encoded_image->SetEncodedData(buffer);
+
+  // Iterate layers and NAL units, note each NAL unit as a fragment and copy
+  // the data to `encoded_image->_buffer`.
+  const uint8_t start_code[4] = {0, 0, 0, 1};
+  size_t frag = 0;
+  encoded_image->set_size(0);
+  for (int layer = 0; layer < info->iLayerNum; ++layer) {
+    const SLayerBSInfo& layerInfo = info->sLayerInfo[layer];
+    // Iterate NAL units making up this layer, noting fragments.
+    size_t layer_len = 0;
+    for (int nal = 0; nal < layerInfo.iNalCount; ++nal, ++frag) {
+      // Because the sum of all layer lengths, `required_capacity`, fits in a
+      // `size_t`, we know that any indices in-between will not overflow.
+      RTC_DCHECK_GE(layerInfo.pNalLengthInByte[nal], 4);
+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 0], start_code[0]);
+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 1], start_code[1]);
+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 2], start_code[2]);
+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 3], start_code[3]);
+      layer_len += layerInfo.pNalLengthInByte[nal];
+    }
+    // Copy the entire layer's data (including start codes).
+    memcpy(buffer->data() + encoded_image->size(), layerInfo.pBsBuf, layer_len);
+    encoded_image->set_size(encoded_image->size() + layer_len);
+  }
+}
+
+H264EncoderImpl::H264EncoderImpl(const cricket::VideoCodec& codec)
+    : packetization_mode_(H264PacketizationMode::SingleNalUnit),
+      max_payload_size_(0),
+      number_of_cores_(0),
+      encoded_image_callback_(nullptr),
+      has_reported_init_(false),
+      has_reported_error_(false) {
+  RTC_CHECK(absl::EqualsIgnoreCase(codec.name, cricket::kH264CodecName));
+  std::string packetization_mode_string;
+  if (codec.GetParam(cricket::kH264FmtpPacketizationMode,
+                     &packetization_mode_string) &&
+      packetization_mode_string == "1") {
+    packetization_mode_ = H264PacketizationMode::NonInterleaved;
+  }
+  downscaled_buffers_.reserve(kMaxSimulcastStreams - 1);
+  encoded_images_.reserve(kMaxSimulcastStreams);
+  encoders_.reserve(kMaxSimulcastStreams);
+  configurations_.reserve(kMaxSimulcastStreams);
+  tl0sync_limit_.reserve(kMaxSimulcastStreams);
+  svc_controllers_.reserve(kMaxSimulcastStreams);
+}
+
+H264EncoderImpl::~H264EncoderImpl() {
+  Release();
+}
+
+int32_t H264EncoderImpl::InitEncode(const VideoCodec* inst,
+                                    const VideoEncoder::Settings& settings) {
+  ReportInit();
+  if (!inst || inst->codecType != kVideoCodecH264) {
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (inst->maxFramerate == 0) {
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (inst->width < 1 || inst->height < 1) {
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  int32_t release_ret = Release();
+  if (release_ret != WEBRTC_VIDEO_CODEC_OK) {
+    ReportError();
+    return release_ret;
+  }
+
+  int number_of_streams = SimulcastUtility::NumberOfSimulcastStreams(*inst);
+  bool doing_simulcast = (number_of_streams > 1);
+
+  if (doing_simulcast &&
+      !SimulcastUtility::ValidSimulcastParameters(*inst, number_of_streams)) {
+    return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
+  }
+  downscaled_buffers_.resize(number_of_streams - 1);
+  encoded_images_.resize(number_of_streams);
+  encoders_.resize(number_of_streams);
+  pictures_.resize(number_of_streams);
+  svc_controllers_.resize(number_of_streams);
+  scalability_modes_.resize(number_of_streams);
+  configurations_.resize(number_of_streams);
+  tl0sync_limit_.resize(number_of_streams);
+
+  number_of_cores_ = settings.number_of_cores;
+  max_payload_size_ = settings.max_payload_size;
+  codec_ = *inst;
+
+  // Code expects simulcastStream resolutions to be correct, make sure they are
+  // filled even when there are no simulcast layers.
+  if (codec_.numberOfSimulcastStreams == 0) {
+    codec_.simulcastStream[0].width = codec_.width;
+    codec_.simulcastStream[0].height = codec_.height;
+  }
+
+  for (int i = 0, idx = number_of_streams - 1; i < number_of_streams;
+       ++i, --idx) {
+    ISVCEncoder* openh264_encoder;
+    // Create encoder.
+    if (WelsCreateSVCEncoder(&openh264_encoder) != 0) {
+      // Failed to create encoder.
+      RTC_LOG(LS_ERROR) << "Failed to create OpenH264 encoder";
+      RTC_DCHECK(!openh264_encoder);
+      Release();
+      ReportError();
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    RTC_DCHECK(openh264_encoder);
+    if (kOpenH264EncoderDetailedLogging) {
+      int trace_level = WELS_LOG_DETAIL;
+      openh264_encoder->SetOption(ENCODER_OPTION_TRACE_LEVEL, &trace_level);
+    }
+    // else WELS_LOG_DEFAULT is used by default.
+
+    // Store h264 encoder.
+    encoders_[i] = openh264_encoder;
+
+    // Set internal settings from codec_settings
+    configurations_[i].simulcast_idx = idx;
+    configurations_[i].sending = false;
+    configurations_[i].width = codec_.simulcastStream[idx].width;
+    configurations_[i].height = codec_.simulcastStream[idx].height;
+    configurations_[i].max_frame_rate = static_cast<float>(codec_.maxFramerate);
+    configurations_[i].frame_dropping_on = codec_.GetFrameDropEnabled();
+    configurations_[i].key_frame_interval = codec_.H264()->keyFrameInterval;
+    configurations_[i].num_temporal_layers =
+        std::max(codec_.H264()->numberOfTemporalLayers,
+                 codec_.simulcastStream[idx].numberOfTemporalLayers);
+
+    // Create downscaled image buffers.
+    if (i > 0) {
+      downscaled_buffers_[i - 1] = I420Buffer::Create(
+          configurations_[i].width, configurations_[i].height,
+          configurations_[i].width, configurations_[i].width / 2,
+          configurations_[i].width / 2);
+    }
+
+    // Codec_settings uses kbits/second; encoder uses bits/second.
+    configurations_[i].max_bps = codec_.maxBitrate * 1000;
+    configurations_[i].target_bps = codec_.startBitrate * 1000;
+
+    // Create encoder parameters based on the layer configuration.
+    SEncParamExt encoder_params = CreateEncoderParams(i);
+
+    // Initialize.
+    if (openh264_encoder->InitializeExt(&encoder_params) != 0) {
+      RTC_LOG(LS_ERROR) << "Failed to initialize OpenH264 encoder";
+      Release();
+      ReportError();
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    // TODO(pbos): Base init params on these values before submitting.
+    int video_format = EVideoFormatType::videoFormatI420;
+    openh264_encoder->SetOption(ENCODER_OPTION_DATAFORMAT, &video_format);
+
+    // Initialize encoded image. Default buffer size: size of unencoded data.
+
+    const size_t new_capacity =
+        CalcBufferSize(VideoType::kI420, codec_.simulcastStream[idx].width,
+                       codec_.simulcastStream[idx].height);
+    encoded_images_[i].SetEncodedData(EncodedImageBuffer::Create(new_capacity));
+    encoded_images_[i]._encodedWidth = codec_.simulcastStream[idx].width;
+    encoded_images_[i]._encodedHeight = codec_.simulcastStream[idx].height;
+    encoded_images_[i].set_size(0);
+
+    tl0sync_limit_[i] = configurations_[i].num_temporal_layers;
+    scalability_modes_[i] = ScalabilityModeFromTemporalLayers(
+        configurations_[i].num_temporal_layers);
+    if (scalability_modes_[i].has_value()) {
+      svc_controllers_[i] = CreateScalabilityStructure(*scalability_modes_[i]);
+      if (svc_controllers_[i] == nullptr) {
+        RTC_LOG(LS_ERROR) << "Failed to create scalability structure";
+        Release();
+        ReportError();
+        return WEBRTC_VIDEO_CODEC_ERROR;
+      }
+    }
+  }
+
+  SimulcastRateAllocator init_allocator(codec_);
+  VideoBitrateAllocation allocation =
+      init_allocator.Allocate(VideoBitrateAllocationParameters(
+          DataRate::KilobitsPerSec(codec_.startBitrate), codec_.maxFramerate));
+  SetRates(RateControlParameters(allocation, codec_.maxFramerate));
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264EncoderImpl::Release() {
+  while (!encoders_.empty()) {
+    ISVCEncoder* openh264_encoder = encoders_.back();
+    if (openh264_encoder) {
+      RTC_CHECK_EQ(0, openh264_encoder->Uninitialize());
+      WelsDestroySVCEncoder(openh264_encoder);
+    }
+    encoders_.pop_back();
+  }
+  downscaled_buffers_.clear();
+  configurations_.clear();
+  encoded_images_.clear();
+  pictures_.clear();
+  tl0sync_limit_.clear();
+  svc_controllers_.clear();
+  scalability_modes_.clear();
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t H264EncoderImpl::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  encoded_image_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void H264EncoderImpl::SetRates(const RateControlParameters& parameters) {
+  if (encoders_.empty()) {
+    RTC_LOG(LS_WARNING) << "SetRates() while uninitialized.";
+    return;
+  }
+
+  if (parameters.framerate_fps < 1.0) {
+    RTC_LOG(LS_WARNING) << "Invalid frame rate: " << parameters.framerate_fps;
+    return;
+  }
+
+  if (parameters.bitrate.get_sum_bps() == 0) {
+    // Encoder paused, turn off all encoding.
+    for (size_t i = 0; i < configurations_.size(); ++i) {
+      configurations_[i].SetStreamState(false);
+    }
+    return;
+  }
+
+  codec_.maxFramerate = static_cast<uint32_t>(parameters.framerate_fps);
+
+  size_t stream_idx = encoders_.size() - 1;
+  for (size_t i = 0; i < encoders_.size(); ++i, --stream_idx) {
+    // Update layer config.
+    configurations_[i].target_bps =
+        parameters.bitrate.GetSpatialLayerSum(stream_idx);
+    configurations_[i].max_frame_rate = parameters.framerate_fps;
+
+    if (configurations_[i].target_bps) {
+      configurations_[i].SetStreamState(true);
+
+      // Update h264 encoder.
+      SBitrateInfo target_bitrate;
+      memset(&target_bitrate, 0, sizeof(SBitrateInfo));
+      target_bitrate.iLayer = SPATIAL_LAYER_ALL,
+      target_bitrate.iBitrate = configurations_[i].target_bps;
+      encoders_[i]->SetOption(ENCODER_OPTION_BITRATE, &target_bitrate);
+      encoders_[i]->SetOption(ENCODER_OPTION_FRAME_RATE,
+                              &configurations_[i].max_frame_rate);
+    } else {
+      configurations_[i].SetStreamState(false);
+    }
+  }
+}
+
+int32_t H264EncoderImpl::Encode(
+    const VideoFrame& input_frame,
+    const std::vector<VideoFrameType>* frame_types) {
+  if (encoders_.empty()) {
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (!encoded_image_callback_) {
+    RTC_LOG(LS_WARNING)
+        << "InitEncode() has been called, but a callback function "
+           "has not been set with RegisterEncodeCompleteCallback()";
+    ReportError();
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  rtc::scoped_refptr<I420BufferInterface> frame_buffer =
+      input_frame.video_frame_buffer()->ToI420();
+  if (!frame_buffer) {
+    RTC_LOG(LS_ERROR) << "Failed to convert "
+                      << VideoFrameBufferTypeToString(
+                             input_frame.video_frame_buffer()->type())
+                      << " image to I420. Can't encode frame.";
+    return WEBRTC_VIDEO_CODEC_ENCODER_FAILURE;
+  }
+  RTC_CHECK(frame_buffer->type() == VideoFrameBuffer::Type::kI420 ||
+            frame_buffer->type() == VideoFrameBuffer::Type::kI420A);
+
+  bool is_keyframe_needed = false;
+  for (size_t i = 0; i < configurations_.size(); ++i) {
+    if (configurations_[i].key_frame_request && configurations_[i].sending) {
+      // This is legacy behavior, generating a keyframe on all layers
+      // when generating one for a layer that became active for the first time
+      // or after being disabled.
+      is_keyframe_needed = true;
+      break;
+    }
+  }
+
+  RTC_DCHECK_EQ(configurations_[0].width, frame_buffer->width());
+  RTC_DCHECK_EQ(configurations_[0].height, frame_buffer->height());
+
+  // Encode image for each layer.
+  for (size_t i = 0; i < encoders_.size(); ++i) {
+    // EncodeFrame input.
+    pictures_[i] = {0};
+    pictures_[i].iPicWidth = configurations_[i].width;
+    pictures_[i].iPicHeight = configurations_[i].height;
+    pictures_[i].iColorFormat = EVideoFormatType::videoFormatI420;
+    pictures_[i].uiTimeStamp = input_frame.ntp_time_ms();
+    // Downscale images on second and ongoing layers.
+    if (i == 0) {
+      pictures_[i].iStride[0] = frame_buffer->StrideY();
+      pictures_[i].iStride[1] = frame_buffer->StrideU();
+      pictures_[i].iStride[2] = frame_buffer->StrideV();
+      pictures_[i].pData[0] = const_cast<uint8_t*>(frame_buffer->DataY());
+      pictures_[i].pData[1] = const_cast<uint8_t*>(frame_buffer->DataU());
+      pictures_[i].pData[2] = const_cast<uint8_t*>(frame_buffer->DataV());
+    } else {
+      pictures_[i].iStride[0] = downscaled_buffers_[i - 1]->StrideY();
+      pictures_[i].iStride[1] = downscaled_buffers_[i - 1]->StrideU();
+      pictures_[i].iStride[2] = downscaled_buffers_[i - 1]->StrideV();
+      pictures_[i].pData[0] =
+          const_cast<uint8_t*>(downscaled_buffers_[i - 1]->DataY());
+      pictures_[i].pData[1] =
+          const_cast<uint8_t*>(downscaled_buffers_[i - 1]->DataU());
+      pictures_[i].pData[2] =
+          const_cast<uint8_t*>(downscaled_buffers_[i - 1]->DataV());
+      // Scale the image down a number of times by downsampling factor.
+      libyuv::I420Scale(pictures_[i - 1].pData[0], pictures_[i - 1].iStride[0],
+                        pictures_[i - 1].pData[1], pictures_[i - 1].iStride[1],
+                        pictures_[i - 1].pData[2], pictures_[i - 1].iStride[2],
+                        configurations_[i - 1].width,
+                        configurations_[i - 1].height, pictures_[i].pData[0],
+                        pictures_[i].iStride[0], pictures_[i].pData[1],
+                        pictures_[i].iStride[1], pictures_[i].pData[2],
+                        pictures_[i].iStride[2], configurations_[i].width,
+                        configurations_[i].height, libyuv::kFilterBox);
+    }
+
+    if (!configurations_[i].sending) {
+      continue;
+    }
+    if (frame_types != nullptr && i < frame_types->size()) {
+      // Skip frame?
+      if ((*frame_types)[i] == VideoFrameType::kEmptyFrame) {
+        continue;
+      }
+    }
+    // Send a key frame either when this layer is configured to require one
+    // or we have explicitly been asked to.
+    const size_t simulcast_idx =
+        static_cast<size_t>(configurations_[i].simulcast_idx);
+    bool send_key_frame =
+        is_keyframe_needed ||
+        (frame_types && simulcast_idx < frame_types->size() &&
+         (*frame_types)[simulcast_idx] == VideoFrameType::kVideoFrameKey);
+    if (send_key_frame) {
+      // API doc says ForceIntraFrame(false) does nothing, but calling this
+      // function forces a key frame regardless of the `bIDR` argument's value.
+      // (If every frame is a key frame we get lag/delays.)
+      encoders_[i]->ForceIntraFrame(true);
+      configurations_[i].key_frame_request = false;
+    }
+    // EncodeFrame output.
+    SFrameBSInfo info;
+    memset(&info, 0, sizeof(SFrameBSInfo));
+
+    std::vector<ScalableVideoController::LayerFrameConfig> layer_frames;
+    if (svc_controllers_[i]) {
+      layer_frames = svc_controllers_[i]->NextFrameConfig(send_key_frame);
+      RTC_CHECK_EQ(layer_frames.size(), 1);
+    }
+
+    // Encode!
+    int enc_ret = encoders_[i]->EncodeFrame(&pictures_[i], &info);
+    if (enc_ret != 0) {
+      RTC_LOG(LS_ERROR)
+          << "OpenH264 frame encoding failed, EncodeFrame returned " << enc_ret
+          << ".";
+      ReportError();
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
+    encoded_images_[i]._encodedWidth = configurations_[i].width;
+    encoded_images_[i]._encodedHeight = configurations_[i].height;
+    encoded_images_[i].SetTimestamp(input_frame.timestamp());
+    encoded_images_[i].SetColorSpace(input_frame.color_space());
+    encoded_images_[i]._frameType = ConvertToVideoFrameType(info.eFrameType);
+    encoded_images_[i].SetSpatialIndex(configurations_[i].simulcast_idx);
+
+    // Split encoded image up into fragments. This also updates
+    // `encoded_image_`.
+    RtpFragmentize(&encoded_images_[i], &info);
+
+    // Encoder can skip frames to save bandwidth in which case
+    // `encoded_images_[i]._length` == 0.
+    if (encoded_images_[i].size() > 0) {
+      // Parse QP.
+      h264_bitstream_parser_.ParseBitstream(encoded_images_[i]);
+      encoded_images_[i].qp_ =
+          h264_bitstream_parser_.GetLastSliceQp().value_or(-1);
+
+      // Deliver encoded image.
+      CodecSpecificInfo codec_specific;
+      codec_specific.codecType = kVideoCodecH264;
+      codec_specific.codecSpecific.H264.packetization_mode =
+          packetization_mode_;
+      codec_specific.codecSpecific.H264.temporal_idx = kNoTemporalIdx;
+      codec_specific.codecSpecific.H264.idr_frame =
+          info.eFrameType == videoFrameTypeIDR;
+      codec_specific.codecSpecific.H264.base_layer_sync = false;
+      if (configurations_[i].num_temporal_layers > 1) {
+        const uint8_t tid = info.sLayerInfo[0].uiTemporalId;
+        codec_specific.codecSpecific.H264.temporal_idx = tid;
+        codec_specific.codecSpecific.H264.base_layer_sync =
+            tid > 0 && tid < tl0sync_limit_[i];
+        if (svc_controllers_[i]) {
+          if (layer_frames[0].TemporalId() != tid) {
+            RTC_LOG(LS_WARNING)
+                << "Encoder produced a frame for layer S" << (i + 1) << "T"
+                << tid + 1 << " that wasn't requested.";
+            continue;
+          }
+          encoded_images_[i].SetTemporalIndex(tid);
+        }
+        if (codec_specific.codecSpecific.H264.base_layer_sync) {
+          tl0sync_limit_[i] = tid;
+        }
+        if (tid == 0) {
+          tl0sync_limit_[i] = configurations_[i].num_temporal_layers;
+        }
+      }
+      if (svc_controllers_[i]) {
+        codec_specific.generic_frame_info =
+            svc_controllers_[i]->OnEncodeDone(layer_frames[0]);
+        if (send_key_frame && codec_specific.generic_frame_info.has_value()) {
+          codec_specific.template_structure =
+              svc_controllers_[i]->DependencyStructure();
+        }
+        codec_specific.scalability_mode = scalability_modes_[i];
+      }
+      encoded_image_callback_->OnEncodedImage(encoded_images_[i],
+                                              &codec_specific);
+    }
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+// Initialization parameters.
+// There are two ways to initialize. There is SEncParamBase (cleared with
+// memset(&p, 0, sizeof(SEncParamBase)) used in Initialize, and SEncParamExt
+// which is a superset of SEncParamBase (cleared with GetDefaultParams) used
+// in InitializeExt.
+SEncParamExt H264EncoderImpl::CreateEncoderParams(size_t i) const {
+  SEncParamExt encoder_params;
+  encoders_[i]->GetDefaultParams(&encoder_params);
+  if (codec_.mode == VideoCodecMode::kRealtimeVideo) {
+    encoder_params.iUsageType = CAMERA_VIDEO_REAL_TIME;
+  } else if (codec_.mode == VideoCodecMode::kScreensharing) {
+    encoder_params.iUsageType = SCREEN_CONTENT_REAL_TIME;
+  } else {
+    RTC_DCHECK_NOTREACHED();
+  }
+  encoder_params.iPicWidth = configurations_[i].width;
+  encoder_params.iPicHeight = configurations_[i].height;
+  encoder_params.iTargetBitrate = configurations_[i].target_bps;
+  // Keep unspecified. WebRTC's max codec bitrate is not the same setting
+  // as OpenH264's iMaxBitrate. More details in https://crbug.com/webrtc/11543
+  encoder_params.iMaxBitrate = UNSPECIFIED_BIT_RATE;
+  // Rate Control mode
+  encoder_params.iRCMode = RC_BITRATE_MODE;
+  encoder_params.fMaxFrameRate = configurations_[i].max_frame_rate;
+
+  // The following parameters are extension parameters (they're in SEncParamExt,
+  // not in SEncParamBase).
+  encoder_params.bEnableFrameSkip = configurations_[i].frame_dropping_on;
+  // `uiIntraPeriod`    - multiple of GOP size
+  // `keyFrameInterval` - number of frames
+  encoder_params.uiIntraPeriod = configurations_[i].key_frame_interval;
+  // Reuse SPS id if possible. This helps to avoid reset of chromium HW decoder
+  // on each key-frame.
+  // Note that WebRTC resets encoder on resolution change which makes all
+  // EParameterSetStrategy modes except INCREASING_ID (default) essentially
+  // equivalent to CONSTANT_ID.
+  encoder_params.eSpsPpsIdStrategy = SPS_LISTING;
+  encoder_params.uiMaxNalSize = 0;
+  // Threading model: use auto.
+  //  0: auto (dynamic imp. internal encoder)
+  //  1: single thread (default value)
+  // >1: number of threads
+  encoder_params.iMultipleThreadIdc = NumberOfThreads(
+      encoder_params.iPicWidth, encoder_params.iPicHeight, number_of_cores_);
+  // The base spatial layer 0 is the only one we use.
+  encoder_params.sSpatialLayers[0].iVideoWidth = encoder_params.iPicWidth;
+  encoder_params.sSpatialLayers[0].iVideoHeight = encoder_params.iPicHeight;
+  encoder_params.sSpatialLayers[0].fFrameRate = encoder_params.fMaxFrameRate;
+  encoder_params.sSpatialLayers[0].iSpatialBitrate =
+      encoder_params.iTargetBitrate;
+  encoder_params.sSpatialLayers[0].iMaxSpatialBitrate =
+      encoder_params.iMaxBitrate;
+  encoder_params.iTemporalLayerNum = configurations_[i].num_temporal_layers;
+  if (encoder_params.iTemporalLayerNum > 1) {
+    // iNumRefFrame specifies total number of reference buffers to allocate.
+    // For N temporal layers we need at least (N - 1) buffers to store last
+    // encoded frames of all reference temporal layers.
+    // Note that there is no API in OpenH264 encoder to specify exact set of
+    // references to be used to prediction of a given frame. Encoder can
+    // theoretically use all available reference buffers.
+    encoder_params.iNumRefFrame = encoder_params.iTemporalLayerNum - 1;
+  }
+  RTC_LOG(LS_INFO) << "OpenH264 version is " << OPENH264_MAJOR << "."
+                   << OPENH264_MINOR;
+  switch (packetization_mode_) {
+    case H264PacketizationMode::SingleNalUnit:
+      // Limit the size of the packets produced.
+      encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceNum = 1;
+      encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceMode =
+          SM_SIZELIMITED_SLICE;
+      encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceSizeConstraint =
+          static_cast<unsigned int>(max_payload_size_);
+      RTC_LOG(LS_INFO) << "Encoder is configured with NALU constraint: "
+                       << max_payload_size_ << " bytes";
+      break;
+    case H264PacketizationMode::NonInterleaved:
+      // When uiSliceMode = SM_FIXEDSLCNUM_SLICE, uiSliceNum = 0 means auto
+      // design it with cpu core number.
+      // TODO(sprang): Set to 0 when we understand why the rate controller borks
+      //               when uiSliceNum > 1.
+      encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceNum = 1;
+      encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceMode =
+          SM_FIXEDSLCNUM_SLICE;
+      break;
+  }
+  return encoder_params;
+}
+
+void H264EncoderImpl::ReportInit() {
+  if (has_reported_init_)
+    return;
+  RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264EncoderImpl.Event",
+                            kH264EncoderEventInit, kH264EncoderEventMax);
+  has_reported_init_ = true;
+}
+
+void H264EncoderImpl::ReportError() {
+  if (has_reported_error_)
+    return;
+  RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264EncoderImpl.Event",
+                            kH264EncoderEventError, kH264EncoderEventMax);
+  has_reported_error_ = true;
+}
+
+VideoEncoder::EncoderInfo H264EncoderImpl::GetEncoderInfo() const {
+  EncoderInfo info;
+  info.supports_native_handle = false;
+  info.implementation_name = "OpenH264";
+  info.scaling_settings =
+      VideoEncoder::ScalingSettings(kLowH264QpThreshold, kHighH264QpThreshold);
+  info.is_hardware_accelerated = false;
+  info.supports_simulcast = true;
+  info.preferred_pixel_formats = {VideoFrameBuffer::Type::kI420};
+  return info;
+}
+
+void H264EncoderImpl::LayerConfig::SetStreamState(bool send_stream) {
+  if (send_stream && !sending) {
+    // Need a key frame if we have not sent this stream before.
+    key_frame_request = true;
+  }
+  sending = send_stream;
+}
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_USE_H264
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
new file mode 100644
index 0000000000..f02521f0dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
@@ -0,0 +1,125 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
+#define MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
+
+// Everything declared in this header is only required when WebRTC is
+// build with H264 support, please do not move anything out of the
+// #ifdef unless needed and tested.
+#ifdef WEBRTC_USE_H264
+
+#if defined(WEBRTC_WIN) && !defined(__clang__)
+#error "See: bugs.webrtc.org/9213#c13."
+#endif
+
+#include <memory>
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "api/video_codecs/video_encoder.h"
+#include "common_video/h264/h264_bitstream_parser.h"
+#include "modules/video_coding/codecs/h264/include/h264.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "modules/video_coding/utility/quality_scaler.h"
+#include "third_party/openh264/src/codec/api/wels/codec_app_def.h"
+
+class ISVCEncoder;
+
+namespace webrtc {
+
+class H264EncoderImpl : public H264Encoder {
+ public:
+  struct LayerConfig {
+    int simulcast_idx = 0;
+    int width = -1;
+    int height = -1;
+    bool sending = true;
+    bool key_frame_request = false;
+    float max_frame_rate = 0;
+    uint32_t target_bps = 0;
+    uint32_t max_bps = 0;
+    bool frame_dropping_on = false;
+    int key_frame_interval = 0;
+    int num_temporal_layers = 1;
+
+    void SetStreamState(bool send_stream);
+  };
+
+ public:
+  explicit H264EncoderImpl(const cricket::VideoCodec& codec);
+  ~H264EncoderImpl() override;
+
+  // `settings.max_payload_size` is ignored.
+  // The following members of `codec_settings` are used. The rest are ignored.
+  // - codecType (must be kVideoCodecH264)
+  // - targetBitrate
+  // - maxFramerate
+  // - width
+  // - height
+  int32_t InitEncode(const VideoCodec* codec_settings,
+                     const VideoEncoder::Settings& settings) override;
+  int32_t Release() override;
+
+  int32_t RegisterEncodeCompleteCallback(
+      EncodedImageCallback* callback) override;
+  void SetRates(const RateControlParameters& parameters) override;
+
+  // The result of encoding - an EncodedImage and CodecSpecificInfo - are
+  // passed to the encode complete callback.
+  int32_t Encode(const VideoFrame& frame,
+                 const std::vector<VideoFrameType>* frame_types) override;
+
+  EncoderInfo GetEncoderInfo() const override;
+
+  // Exposed for testing.
+  H264PacketizationMode PacketizationModeForTesting() const {
+    return packetization_mode_;
+  }
+
+ private:
+  SEncParamExt CreateEncoderParams(size_t i) const;
+
+  webrtc::H264BitstreamParser h264_bitstream_parser_;
+  // Reports statistics with histograms.
+  void ReportInit();
+  void ReportError();
+
+  std::vector<ISVCEncoder*> encoders_;
+  std::vector<SSourcePicture> pictures_;
+  std::vector<rtc::scoped_refptr<I420Buffer>> downscaled_buffers_;
+  std::vector<LayerConfig> configurations_;
+  std::vector<EncodedImage> encoded_images_;
+  std::vector<std::unique_ptr<ScalableVideoController>> svc_controllers_;
+  absl::InlinedVector<absl::optional<ScalabilityMode>, kMaxSimulcastStreams>
+      scalability_modes_;
+
+  VideoCodec codec_;
+  H264PacketizationMode packetization_mode_;
+  size_t max_payload_size_;
+  int32_t number_of_cores_;
+  EncodedImageCallback* encoded_image_callback_;
+
+  bool has_reported_init_;
+  bool has_reported_error_;
+
+  std::vector<uint8_t> tl0sync_limit_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_USE_H264
+
+#endif  // MODULES_VIDEO_CODING_CODECS_H264_H264_ENCODER_IMPL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
new file mode 100644
index 0000000000..52d26955ab
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "modules/video_coding/codecs/h264/h264_encoder_impl.h"
+
+#include "api/video_codecs/video_encoder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kMaxPayloadSize = 1024;
+const int kNumCores = 1;
+
+const VideoEncoder::Capabilities kCapabilities(false);
+const VideoEncoder::Settings kSettings(kCapabilities,
+                                       kNumCores,
+                                       kMaxPayloadSize);
+
+void SetDefaultSettings(VideoCodec* codec_settings) {
+  codec_settings->codecType = kVideoCodecH264;
+  codec_settings->maxFramerate = 60;
+  codec_settings->width = 640;
+  codec_settings->height = 480;
+  // If frame dropping is false, we get a warning that bitrate can't
+  // be controlled for RC_QUALITY_MODE; RC_BITRATE_MODE and RC_TIMESTAMP_MODE
+  codec_settings->SetFrameDropEnabled(true);
+  codec_settings->startBitrate = 2000;
+  codec_settings->maxBitrate = 4000;
+}
+
+TEST(H264EncoderImplTest, CanInitializeWithDefaultParameters) {
+  H264EncoderImpl encoder(cricket::VideoCodec("H264"));
+  VideoCodec codec_settings;
+  SetDefaultSettings(&codec_settings);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings, kSettings));
+  EXPECT_EQ(H264PacketizationMode::NonInterleaved,
+            encoder.PacketizationModeForTesting());
+}
+
+TEST(H264EncoderImplTest, CanInitializeWithNonInterleavedModeExplicitly) {
+  cricket::VideoCodec codec("H264");
+  codec.SetParam(cricket::kH264FmtpPacketizationMode, "1");
+  H264EncoderImpl encoder(codec);
+  VideoCodec codec_settings;
+  SetDefaultSettings(&codec_settings);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings, kSettings));
+  EXPECT_EQ(H264PacketizationMode::NonInterleaved,
+            encoder.PacketizationModeForTesting());
+}
+
+TEST(H264EncoderImplTest, CanInitializeWithSingleNalUnitModeExplicitly) {
+  cricket::VideoCodec codec("H264");
+  codec.SetParam(cricket::kH264FmtpPacketizationMode, "0");
+  H264EncoderImpl encoder(codec);
+  VideoCodec codec_settings;
+  SetDefaultSettings(&codec_settings);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings, kSettings));
+  EXPECT_EQ(H264PacketizationMode::SingleNalUnit,
+            encoder.PacketizationModeForTesting());
+}
+
+TEST(H264EncoderImplTest, CanInitializeWithRemovedParameter) {
+  cricket::VideoCodec codec("H264");
+  codec.RemoveParam(cricket::kH264FmtpPacketizationMode);
+  H264EncoderImpl encoder(codec);
+  VideoCodec codec_settings;
+  SetDefaultSettings(&codec_settings);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings, kSettings));
+  EXPECT_EQ(H264PacketizationMode::SingleNalUnit,
+            encoder.PacketizationModeForTesting());
+}
+
+}  // anonymous namespace
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc
new file mode 100644
index 0000000000..2acb629a76
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "api/test/create_simulcast_test_fixture.h"
+#include "api/test/simulcast_test_fixture.h"
+#include "api/test/video/function_video_decoder_factory.h"
+#include "api/test/video/function_video_encoder_factory.h"
+#include "modules/video_coding/codecs/h264/include/h264.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+std::unique_ptr<SimulcastTestFixture> CreateSpecificSimulcastTestFixture() {
+  std::unique_ptr<VideoEncoderFactory> encoder_factory =
+      std::make_unique<FunctionVideoEncoderFactory>(
+          []() { return H264Encoder::Create(cricket::VideoCodec("H264")); });
+  std::unique_ptr<VideoDecoderFactory> decoder_factory =
+      std::make_unique<FunctionVideoDecoderFactory>(
+          []() { return H264Decoder::Create(); });
+  return CreateSimulcastTestFixture(std::move(encoder_factory),
+                                    std::move(decoder_factory),
+                                    SdpVideoFormat("H264"));
+}
+}  // namespace
+
+TEST(TestH264Simulcast, TestKeyFrameRequestsOnAllStreams) {
+  GTEST_SKIP() << "Not applicable to H264.";
+}
+
+TEST(TestH264Simulcast, TestKeyFrameRequestsOnSpecificStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestKeyFrameRequestsOnSpecificStreams();
+}
+
+TEST(TestH264Simulcast, TestPaddingAllStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingAllStreams();
+}
+
+TEST(TestH264Simulcast, TestPaddingTwoStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingTwoStreams();
+}
+
+TEST(TestH264Simulcast, TestPaddingTwoStreamsOneMaxedOut) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingTwoStreamsOneMaxedOut();
+}
+
+TEST(TestH264Simulcast, TestPaddingOneStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingOneStream();
+}
+
+TEST(TestH264Simulcast, TestPaddingOneStreamTwoMaxedOut) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingOneStreamTwoMaxedOut();
+}
+
+TEST(TestH264Simulcast, TestSendAllStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSendAllStreams();
+}
+
+TEST(TestH264Simulcast, TestDisablingStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestDisablingStreams();
+}
+
+TEST(TestH264Simulcast, TestActiveStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestActiveStreams();
+}
+
+TEST(TestH264Simulcast, TestSwitchingToOneStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSwitchingToOneStream();
+}
+
+TEST(TestH264Simulcast, TestSwitchingToOneOddStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSwitchingToOneOddStream();
+}
+
+TEST(TestH264Simulcast, TestStrideEncodeDecode) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestStrideEncodeDecode();
+}
+
+TEST(TestH264Simulcast, TestSpatioTemporalLayers333PatternEncoder) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSpatioTemporalLayers333PatternEncoder();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/include/h264.h b/third_party/libwebrtc/modules/video_coding/codecs/h264/include/h264.h
new file mode 100644
index 0000000000..2635b53842
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/include/h264.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_H_
+#define MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/video_codecs/h264_profile_level_id.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "media/base/codec.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+struct SdpVideoFormat;
+
+// Creates an H264 SdpVideoFormat entry with specified paramters.
+RTC_EXPORT SdpVideoFormat
+CreateH264Format(H264Profile profile,
+                 H264Level level,
+                 const std::string& packetization_mode,
+                 bool add_scalability_modes = false);
+
+// Set to disable the H.264 encoder/decoder implementations that are provided if
+// `rtc_use_h264` build flag is true (if false, this function does nothing).
+// This function should only be called before or during WebRTC initialization
+// and is not thread-safe.
+RTC_EXPORT void DisableRtcUseH264();
+
+// Returns a vector with all supported internal H264 encode profiles that we can
+// negotiate in SDP, in order of preference.
+std::vector<SdpVideoFormat> SupportedH264Codecs(
+    bool add_scalability_modes = false);
+
+// Returns a vector with all supported internal H264 decode profiles that we can
+// negotiate in SDP, in order of preference. This will be available for receive
+// only connections.
+std::vector<SdpVideoFormat> SupportedH264DecoderCodecs();
+
+class RTC_EXPORT H264Encoder : public VideoEncoder {
+ public:
+  static std::unique_ptr<H264Encoder> Create(const cricket::VideoCodec& codec);
+  // If H.264 is supported (any implementation).
+  static bool IsSupported();
+  static bool SupportsScalabilityMode(ScalabilityMode scalability_mode);
+
+  ~H264Encoder() override {}
+};
+
+class RTC_EXPORT H264Decoder : public VideoDecoder {
+ public:
+  static std::unique_ptr<H264Decoder> Create();
+  static bool IsSupported();
+
+  ~H264Decoder() override {}
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/include/h264_globals.h b/third_party/libwebrtc/modules/video_coding/codecs/h264/include/h264_globals.h
new file mode 100644
index 0000000000..b61dc8c507
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/include/h264_globals.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains codec dependent definitions that are needed in
+// order to compile the WebRTC codebase, even if this codec is not used.
+
+#ifndef MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
+#define MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
+
+#include <string>
+
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// The packetization types that we support: single, aggregated, and fragmented.
+enum H264PacketizationTypes {
+  kH264SingleNalu,  // This packet contains a single NAL unit.
+  kH264StapA,       // This packet contains STAP-A (single time
+                    // aggregation) packets. If this packet has an
+                    // associated NAL unit type, it'll be for the
+                    // first such aggregated packet.
+  kH264FuA,         // This packet contains a FU-A (fragmentation
+                    // unit) packet, meaning it is a part of a frame
+                    // that was too large to fit into a single packet.
+};
+
+// Packetization modes are defined in RFC 6184 section 6
+// Due to the structure containing this being initialized with zeroes
+// in some places, and mode 1 being default, mode 1 needs to have the value
+// zero. https://crbug.com/webrtc/6803
+enum class H264PacketizationMode {
+  NonInterleaved = 0,  // Mode 1 - STAP-A, FU-A is allowed
+  SingleNalUnit        // Mode 0 - only single NALU allowed
+};
+
+// This function is declared inline because it is not clear which
+// .cc file it should belong to.
+// TODO(hta): Refactor. https://bugs.webrtc.org/6842
+// TODO(jonasolsson): Use absl::string_view instead when that's available.
+inline std::string ToString(H264PacketizationMode mode) {
+  if (mode == H264PacketizationMode::NonInterleaved) {
+    return "NonInterleaved";
+  } else if (mode == H264PacketizationMode::SingleNalUnit) {
+    return "SingleNalUnit";
+  }
+  RTC_DCHECK_NOTREACHED();
+  return "";
+}
+
+struct NaluInfo {
+  uint8_t type;
+  int sps_id;
+  int pps_id;
+};
+
+const size_t kMaxNalusPerPacket = 10;
+
+struct RTPVideoHeaderH264 {
+  // The NAL unit type. If this is a header for a
+  // fragmented packet, it's the NAL unit type of
+  // the original data. If this is the header for an
+  // aggregated packet, it's the NAL unit type of
+  // the first NAL unit in the packet.
+  uint8_t nalu_type;
+  // The packetization type of this buffer - single, aggregated or fragmented.
+  H264PacketizationTypes packetization_type;
+  NaluInfo nalus[kMaxNalusPerPacket];
+  size_t nalus_length;
+  // The packetization mode of this transport. Packetization mode
+  // determines which packetization types are allowed when packetizing.
+  H264PacketizationMode packetization_mode;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
new file mode 100644
index 0000000000..595e627bcc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/video/color_space.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "media/base/codec.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/h264/include/h264.h"
+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "test/gtest.h"
+#include "test/video_codec_settings.h"
+
+namespace webrtc {
+
+class TestH264Impl : public VideoCodecUnitTest {
+ protected:
+  std::unique_ptr<VideoEncoder> CreateEncoder() override {
+    return H264Encoder::Create(cricket::VideoCodec(cricket::kH264CodecName));
+  }
+
+  std::unique_ptr<VideoDecoder> CreateDecoder() override {
+    return H264Decoder::Create();
+  }
+
+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
+    webrtc::test::CodecSettings(kVideoCodecH264, codec_settings);
+  }
+};
+
+#ifdef WEBRTC_USE_H264
+#define MAYBE_EncodeDecode EncodeDecode
+#define MAYBE_DecodedQpEqualsEncodedQp DecodedQpEqualsEncodedQp
+#else
+#define MAYBE_EncodeDecode DISABLED_EncodeDecode
+#define MAYBE_DecodedQpEqualsEncodedQp DISABLED_DecodedQpEqualsEncodedQp
+#endif
+
+TEST_F(TestH264Impl, MAYBE_EncodeDecode) {
+  VideoFrame input_frame = NextInputFrame();
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(input_frame, nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  // First frame should be a key frame.
+  encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, 0));
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+  EXPECT_GT(I420PSNR(&input_frame, decoded_frame.get()), 36);
+
+  const ColorSpace color_space = *decoded_frame->color_space();
+  EXPECT_EQ(ColorSpace::PrimaryID::kUnspecified, color_space.primaries());
+  EXPECT_EQ(ColorSpace::TransferID::kUnspecified, color_space.transfer());
+  EXPECT_EQ(ColorSpace::MatrixID::kUnspecified, color_space.matrix());
+  EXPECT_EQ(ColorSpace::RangeID::kInvalid, color_space.range());
+  EXPECT_EQ(ColorSpace::ChromaSiting::kUnspecified,
+            color_space.chroma_siting_horizontal());
+  EXPECT_EQ(ColorSpace::ChromaSiting::kUnspecified,
+            color_space.chroma_siting_vertical());
+}
+
+TEST_F(TestH264Impl, MAYBE_DecodedQpEqualsEncodedQp) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  // First frame should be a key frame.
+  encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, 0));
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+  ASSERT_TRUE(decoded_qp);
+  EXPECT_EQ(encoded_frame.qp_, *decoded_qp);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/interface/common_constants.h b/third_party/libwebrtc/modules/video_coding/codecs/interface/common_constants.h
new file mode 100644
index 0000000000..a8fc6290b9
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/interface/common_constants.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains constants that are used by multiple global
+// codec definitions (modules/video_coding/codecs/*/include/*_globals.h)
+
+#ifndef MODULES_VIDEO_CODING_CODECS_INTERFACE_COMMON_CONSTANTS_H_
+#define MODULES_VIDEO_CODING_CODECS_INTERFACE_COMMON_CONSTANTS_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+const int16_t kNoPictureId = -1;
+const int16_t kNoTl0PicIdx = -1;
+const uint8_t kNoTemporalIdx = 0xFF;
+const int kNoKeyIdx = -1;
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_INTERFACE_COMMON_CONSTANTS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/interface/libvpx_interface.cc b/third_party/libwebrtc/modules/video_coding/codecs/interface/libvpx_interface.cc
new file mode 100644
index 0000000000..4f33bef2ba
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/interface/libvpx_interface.cc
@@ -0,0 +1,373 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/interface/libvpx_interface.h"
+
+#include <memory>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+class LibvpxFacade : public LibvpxInterface {
+ public:
+  LibvpxFacade() = default;
+  ~LibvpxFacade() override = default;
+
+  vpx_image_t* img_alloc(vpx_image_t* img,
+                         vpx_img_fmt_t fmt,
+                         unsigned int d_w,
+                         unsigned int d_h,
+                         unsigned int align) const override {
+    return ::vpx_img_alloc(img, fmt, d_w, d_h, align);
+  }
+
+  vpx_image_t* img_wrap(vpx_image_t* img,
+                        vpx_img_fmt_t fmt,
+                        unsigned int d_w,
+                        unsigned int d_h,
+                        unsigned int stride_align,
+                        unsigned char* img_data) const override {
+    return ::vpx_img_wrap(img, fmt, d_w, d_h, stride_align, img_data);
+  }
+
+  void img_free(vpx_image_t* img) const override { ::vpx_img_free(img); }
+
+  vpx_codec_err_t codec_enc_config_set(
+      vpx_codec_ctx_t* ctx,
+      const vpx_codec_enc_cfg_t* cfg) const override {
+    return ::vpx_codec_enc_config_set(ctx, cfg);
+  }
+
+  vpx_codec_err_t codec_enc_config_default(vpx_codec_iface_t* iface,
+                                           vpx_codec_enc_cfg_t* cfg,
+                                           unsigned int usage) const override {
+    return ::vpx_codec_enc_config_default(iface, cfg, usage);
+  }
+
+  vpx_codec_err_t codec_enc_init(vpx_codec_ctx_t* ctx,
+                                 vpx_codec_iface_t* iface,
+                                 const vpx_codec_enc_cfg_t* cfg,
+                                 vpx_codec_flags_t flags) const override {
+    return ::vpx_codec_enc_init(ctx, iface, cfg, flags);
+  }
+
+  vpx_codec_err_t codec_enc_init_multi(vpx_codec_ctx_t* ctx,
+                                       vpx_codec_iface_t* iface,
+                                       vpx_codec_enc_cfg_t* cfg,
+                                       int num_enc,
+                                       vpx_codec_flags_t flags,
+                                       vpx_rational_t* dsf) const override {
+    return ::vpx_codec_enc_init_multi(ctx, iface, cfg, num_enc, flags, dsf);
+  }
+
+  vpx_codec_err_t codec_destroy(vpx_codec_ctx_t* ctx) const override {
+    return ::vpx_codec_destroy(ctx);
+  }
+
+  // For types related to these parameters, see section
+  // "VP8 encoder control function parameter type" in vpx/vp8cx.h.
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                uint32_t param) const override {
+    // We need an explicit call for each type since vpx_codec_control is a
+    // macro that gets expanded into another call based on the parameter name.
+    switch (ctrl_id) {
+      case VP8E_SET_ENABLEAUTOALTREF:
+        return vpx_codec_control(ctx, VP8E_SET_ENABLEAUTOALTREF, param);
+      case VP8E_SET_NOISE_SENSITIVITY:
+        return vpx_codec_control(ctx, VP8E_SET_NOISE_SENSITIVITY, param);
+      case VP8E_SET_SHARPNESS:
+        return vpx_codec_control(ctx, VP8E_SET_SHARPNESS, param);
+      case VP8E_SET_STATIC_THRESHOLD:
+        return vpx_codec_control(ctx, VP8E_SET_STATIC_THRESHOLD, param);
+      case VP8E_SET_ARNR_MAXFRAMES:
+        return vpx_codec_control(ctx, VP8E_SET_ARNR_MAXFRAMES, param);
+      case VP8E_SET_ARNR_STRENGTH:
+        return vpx_codec_control(ctx, VP8E_SET_ARNR_STRENGTH, param);
+      case VP8E_SET_CQ_LEVEL:
+        return vpx_codec_control(ctx, VP8E_SET_CQ_LEVEL, param);
+      case VP8E_SET_MAX_INTRA_BITRATE_PCT:
+        return vpx_codec_control(ctx, VP8E_SET_MAX_INTRA_BITRATE_PCT, param);
+      case VP9E_SET_MAX_INTER_BITRATE_PCT:
+        return vpx_codec_control(ctx, VP9E_SET_MAX_INTER_BITRATE_PCT, param);
+      case VP8E_SET_GF_CBR_BOOST_PCT:
+        return vpx_codec_control(ctx, VP8E_SET_GF_CBR_BOOST_PCT, param);
+      case VP8E_SET_SCREEN_CONTENT_MODE:
+        return vpx_codec_control(ctx, VP8E_SET_SCREEN_CONTENT_MODE, param);
+      case VP9E_SET_GF_CBR_BOOST_PCT:
+        return vpx_codec_control(ctx, VP9E_SET_GF_CBR_BOOST_PCT, param);
+      case VP9E_SET_LOSSLESS:
+        return vpx_codec_control(ctx, VP9E_SET_LOSSLESS, param);
+      case VP9E_SET_FRAME_PARALLEL_DECODING:
+        return vpx_codec_control(ctx, VP9E_SET_FRAME_PARALLEL_DECODING, param);
+      case VP9E_SET_AQ_MODE:
+        return vpx_codec_control(ctx, VP9E_SET_AQ_MODE, param);
+      case VP9E_SET_FRAME_PERIODIC_BOOST:
+        return vpx_codec_control(ctx, VP9E_SET_FRAME_PERIODIC_BOOST, param);
+      case VP9E_SET_NOISE_SENSITIVITY:
+        return vpx_codec_control(ctx, VP9E_SET_NOISE_SENSITIVITY, param);
+      case VP9E_SET_MIN_GF_INTERVAL:
+        return vpx_codec_control(ctx, VP9E_SET_MIN_GF_INTERVAL, param);
+      case VP9E_SET_MAX_GF_INTERVAL:
+        return vpx_codec_control(ctx, VP9E_SET_MAX_GF_INTERVAL, param);
+      case VP9E_SET_TARGET_LEVEL:
+        return vpx_codec_control(ctx, VP9E_SET_TARGET_LEVEL, param);
+      case VP9E_SET_ROW_MT:
+        return vpx_codec_control(ctx, VP9E_SET_ROW_MT, param);
+      case VP9E_ENABLE_MOTION_VECTOR_UNIT_TEST:
+        return vpx_codec_control(ctx, VP9E_ENABLE_MOTION_VECTOR_UNIT_TEST,
+                                 param);
+      case VP9E_SET_SVC_INTER_LAYER_PRED:
+        return vpx_codec_control(ctx, VP9E_SET_SVC_INTER_LAYER_PRED, param);
+      case VP9E_SET_SVC_GF_TEMPORAL_REF:
+        return vpx_codec_control(ctx, VP9E_SET_SVC_GF_TEMPORAL_REF, param);
+      case VP9E_SET_POSTENCODE_DROP:
+        return vpx_codec_control(ctx, VP9E_SET_POSTENCODE_DROP, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                int param) const override {
+    switch (ctrl_id) {
+      case VP8E_SET_FRAME_FLAGS:
+        return vpx_codec_control(ctx, VP8E_SET_FRAME_FLAGS, param);
+      case VP8E_SET_TEMPORAL_LAYER_ID:
+        return vpx_codec_control(ctx, VP8E_SET_TEMPORAL_LAYER_ID, param);
+      case VP9E_SET_SVC:
+        return vpx_codec_control(ctx, VP9E_SET_SVC, param);
+      case VP8E_SET_CPUUSED:
+        return vpx_codec_control(ctx, VP8E_SET_CPUUSED, param);
+      case VP8E_SET_TOKEN_PARTITIONS:
+        return vpx_codec_control(ctx, VP8E_SET_TOKEN_PARTITIONS, param);
+      case VP8E_SET_TUNING:
+        return vpx_codec_control(ctx, VP8E_SET_TUNING, param);
+      case VP9E_SET_TILE_COLUMNS:
+        return vpx_codec_control(ctx, VP9E_SET_TILE_COLUMNS, param);
+      case VP9E_SET_TILE_ROWS:
+        return vpx_codec_control(ctx, VP9E_SET_TILE_ROWS, param);
+      case VP9E_SET_TPL:
+        return vpx_codec_control(ctx, VP9E_SET_TPL, param);
+      case VP9E_SET_ALT_REF_AQ:
+        return vpx_codec_control(ctx, VP9E_SET_ALT_REF_AQ, param);
+      case VP9E_SET_TUNE_CONTENT:
+        return vpx_codec_control(ctx, VP9E_SET_TUNE_CONTENT, param);
+      case VP9E_SET_COLOR_SPACE:
+        return vpx_codec_control(ctx, VP9E_SET_COLOR_SPACE, param);
+      case VP9E_SET_COLOR_RANGE:
+        return vpx_codec_control(ctx, VP9E_SET_COLOR_RANGE, param);
+      case VP9E_SET_DELTA_Q_UV:
+        return vpx_codec_control(ctx, VP9E_SET_DELTA_Q_UV, param);
+      case VP9E_SET_DISABLE_OVERSHOOT_MAXQ_CBR:
+        return vpx_codec_control(ctx, VP9E_SET_DISABLE_OVERSHOOT_MAXQ_CBR,
+                                 param);
+      case VP9E_SET_DISABLE_LOOPFILTER:
+        return vpx_codec_control(ctx, VP9E_SET_DISABLE_LOOPFILTER, param);
+
+      default:
+        if (param >= 0) {
+          // Might be intended for uint32_t but int literal used, try fallback.
+          return codec_control(ctx, ctrl_id, static_cast<uint32_t>(param));
+        }
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                int* param) const override {
+    switch (ctrl_id) {
+      case VP8E_GET_LAST_QUANTIZER:
+        return vpx_codec_control(ctx, VP8E_GET_LAST_QUANTIZER, param);
+      case VP8E_GET_LAST_QUANTIZER_64:
+        return vpx_codec_control(ctx, VP8E_GET_LAST_QUANTIZER_64, param);
+      case VP9E_SET_RENDER_SIZE:
+        return vpx_codec_control(ctx, VP9E_SET_RENDER_SIZE, param);
+      case VP9E_GET_LEVEL:
+        return vpx_codec_control(ctx, VP9E_GET_LEVEL, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_roi_map* param) const override {
+    switch (ctrl_id) {
+      case VP8E_SET_ROI_MAP:
+        return vpx_codec_control(ctx, VP8E_SET_ROI_MAP, param);
+      case VP9E_SET_ROI_MAP:
+        return vpx_codec_control(ctx, VP9E_SET_ROI_MAP, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_active_map* param) const override {
+    switch (ctrl_id) {
+      case VP8E_SET_ACTIVEMAP:
+        return vpx_codec_control(ctx, VP8E_SET_ACTIVEMAP, param);
+      case VP9E_GET_ACTIVEMAP:
+        return vpx_codec_control(ctx, VP8E_SET_ACTIVEMAP, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_scaling_mode* param) const override {
+    switch (ctrl_id) {
+      case VP8E_SET_SCALEMODE:
+        return vpx_codec_control(ctx, VP8E_SET_SCALEMODE, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_svc_extra_cfg_t* param) const override {
+    switch (ctrl_id) {
+      case VP9E_SET_SVC_PARAMETERS:
+        return vpx_codec_control_(ctx, VP9E_SET_SVC_PARAMETERS, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_svc_frame_drop_t* param) const override {
+    switch (ctrl_id) {
+      case VP9E_SET_SVC_FRAME_DROP_LAYER:
+        return vpx_codec_control_(ctx, VP9E_SET_SVC_FRAME_DROP_LAYER, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                void* param) const override {
+    switch (ctrl_id) {
+      case VP9E_SET_SVC_PARAMETERS:
+        return vpx_codec_control_(ctx, VP9E_SET_SVC_PARAMETERS, param);
+      case VP9E_REGISTER_CX_CALLBACK:
+        return vpx_codec_control_(ctx, VP9E_REGISTER_CX_CALLBACK, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_svc_layer_id_t* param) const override {
+    switch (ctrl_id) {
+      case VP9E_SET_SVC_LAYER_ID:
+        return vpx_codec_control_(ctx, VP9E_SET_SVC_LAYER_ID, param);
+      case VP9E_GET_SVC_LAYER_ID:
+        return vpx_codec_control_(ctx, VP9E_GET_SVC_LAYER_ID, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(
+      vpx_codec_ctx_t* ctx,
+      vp8e_enc_control_id ctrl_id,
+      vpx_svc_ref_frame_config_t* param) const override {
+    switch (ctrl_id) {
+      case VP9E_SET_SVC_REF_FRAME_CONFIG:
+        return vpx_codec_control_(ctx, VP9E_SET_SVC_REF_FRAME_CONFIG, param);
+      case VP9E_GET_SVC_REF_FRAME_CONFIG:
+        return vpx_codec_control_(ctx, VP9E_GET_SVC_REF_FRAME_CONFIG, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(
+      vpx_codec_ctx_t* ctx,
+      vp8e_enc_control_id ctrl_id,
+      vpx_svc_spatial_layer_sync_t* param) const override {
+    switch (ctrl_id) {
+      case VP9E_SET_SVC_SPATIAL_LAYER_SYNC:
+        return vpx_codec_control_(ctx, VP9E_SET_SVC_SPATIAL_LAYER_SYNC, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_rc_funcs_t* param) const override {
+    switch (ctrl_id) {
+      case VP9E_SET_EXTERNAL_RATE_CONTROL:
+        return vpx_codec_control_(ctx, VP9E_SET_EXTERNAL_RATE_CONTROL, param);
+      default:
+        RTC_DCHECK_NOTREACHED() << "Unsupported libvpx ctrl_id: " << ctrl_id;
+    }
+    return VPX_CODEC_ERROR;
+  }
+
+  vpx_codec_err_t codec_encode(vpx_codec_ctx_t* ctx,
+                               const vpx_image_t* img,
+                               vpx_codec_pts_t pts,
+                               uint64_t duration,
+                               vpx_enc_frame_flags_t flags,
+                               uint64_t deadline) const override {
+    return ::vpx_codec_encode(ctx, img, pts, duration, flags, deadline);
+  }
+
+  const vpx_codec_cx_pkt_t* codec_get_cx_data(
+      vpx_codec_ctx_t* ctx,
+      vpx_codec_iter_t* iter) const override {
+    return ::vpx_codec_get_cx_data(ctx, iter);
+  }
+
+  const char* codec_error_detail(vpx_codec_ctx_t* ctx) const override {
+    return ::vpx_codec_error_detail(ctx);
+  }
+
+  const char* codec_error(vpx_codec_ctx_t* ctx) const override {
+    return ::vpx_codec_error(ctx);
+  }
+
+  const char* codec_err_to_string(vpx_codec_err_t err) const override {
+    return ::vpx_codec_err_to_string(err);
+  }
+};
+
+}  // namespace
+
+std::unique_ptr<LibvpxInterface> LibvpxInterface::Create() {
+  return std::make_unique<LibvpxFacade>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/interface/libvpx_interface.h b/third_party/libwebrtc/modules/video_coding/codecs/interface/libvpx_interface.h
new file mode 100644
index 0000000000..3dea24dd6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/interface/libvpx_interface.h
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_INTERFACE_LIBVPX_INTERFACE_H_
+#define MODULES_VIDEO_CODING_CODECS_INTERFACE_LIBVPX_INTERFACE_H_
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "vpx/vp8cx.h"
+#include "vpx/vpx_codec.h"
+#include "vpx/vpx_encoder.h"
+#include "vpx/vpx_image.h"
+
+namespace webrtc {
+
+// This interface is a proxy to the static libvpx functions, so that they
+// can be mocked for testing. Currently supports VP8 encoder functions.
+// TODO(sprang): Extend this to VP8 decoder and VP9 encoder/decoder too.
+class LibvpxInterface {
+ public:
+  LibvpxInterface() = default;
+  virtual ~LibvpxInterface() = default;
+
+  virtual vpx_image_t* img_alloc(vpx_image_t* img,
+                                 vpx_img_fmt_t fmt,
+                                 unsigned int d_w,
+                                 unsigned int d_h,
+                                 unsigned int align) const = 0;
+  virtual vpx_image_t* img_wrap(vpx_image_t* img,
+                                vpx_img_fmt_t fmt,
+                                unsigned int d_w,
+                                unsigned int d_h,
+                                unsigned int stride_align,
+                                unsigned char* img_data) const = 0;
+  virtual void img_free(vpx_image_t* img) const = 0;
+
+  virtual vpx_codec_err_t codec_enc_config_set(
+      vpx_codec_ctx_t* ctx,
+      const vpx_codec_enc_cfg_t* cfg) const = 0;
+  virtual vpx_codec_err_t codec_enc_config_default(
+      vpx_codec_iface_t* iface,
+      vpx_codec_enc_cfg_t* cfg,
+      unsigned int usage) const = 0;
+
+  virtual vpx_codec_err_t codec_enc_init(vpx_codec_ctx_t* ctx,
+                                         vpx_codec_iface_t* iface,
+                                         const vpx_codec_enc_cfg_t* cfg,
+                                         vpx_codec_flags_t flags) const = 0;
+  virtual vpx_codec_err_t codec_enc_init_multi(vpx_codec_ctx_t* ctx,
+                                               vpx_codec_iface_t* iface,
+                                               vpx_codec_enc_cfg_t* cfg,
+                                               int num_enc,
+                                               vpx_codec_flags_t flags,
+                                               vpx_rational_t* dsf) const = 0;
+  virtual vpx_codec_err_t codec_destroy(vpx_codec_ctx_t* ctx) const = 0;
+
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        uint32_t param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        int param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        int* param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        vpx_roi_map* param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        vpx_active_map* param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        vpx_scaling_mode* param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        vpx_svc_extra_cfg_t* param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        vpx_svc_frame_drop_t* param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        void* param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        vpx_svc_layer_id_t* param) const = 0;
+  virtual vpx_codec_err_t codec_control(
+      vpx_codec_ctx_t* ctx,
+      vp8e_enc_control_id ctrl_id,
+      vpx_svc_ref_frame_config_t* param) const = 0;
+  virtual vpx_codec_err_t codec_control(
+      vpx_codec_ctx_t* ctx,
+      vp8e_enc_control_id ctrl_id,
+      vpx_svc_spatial_layer_sync_t* param) const = 0;
+  virtual vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                        vp8e_enc_control_id ctrl_id,
+                                        vpx_rc_funcs_t* param) const = 0;
+  virtual vpx_codec_err_t codec_encode(vpx_codec_ctx_t* ctx,
+                                       const vpx_image_t* img,
+                                       vpx_codec_pts_t pts,
+                                       uint64_t duration,
+                                       vpx_enc_frame_flags_t flags,
+                                       uint64_t deadline) const = 0;
+
+  virtual const vpx_codec_cx_pkt_t* codec_get_cx_data(
+      vpx_codec_ctx_t* ctx,
+      vpx_codec_iter_t* iter) const = 0;
+
+  virtual const char* codec_error_detail(vpx_codec_ctx_t* ctx) const = 0;
+  virtual const char* codec_error(vpx_codec_ctx_t* ctx) const = 0;
+  virtual const char* codec_err_to_string(vpx_codec_err_t err) const = 0;
+
+  // Returns interface wrapping the actual libvpx functions.
+  static std::unique_ptr<LibvpxInterface> Create();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_INTERFACE_LIBVPX_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/interface/mock_libvpx_interface.h b/third_party/libwebrtc/modules/video_coding/codecs/interface/mock_libvpx_interface.h
new file mode 100644
index 0000000000..6dfe733dd0
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/interface/mock_libvpx_interface.h
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_LIBVPX_INTERFACE_H_
+#define MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_LIBVPX_INTERFACE_H_
+
+#include "modules/video_coding/codecs/interface/libvpx_interface.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class MockLibvpxInterface : public LibvpxInterface {
+ public:
+  MOCK_METHOD(
+      vpx_image_t*,
+      img_alloc,
+      (vpx_image_t*, vpx_img_fmt_t, unsigned int, unsigned int, unsigned int),
+      (const, override));
+  MOCK_METHOD(vpx_image_t*,
+              img_wrap,
+              (vpx_image_t*,
+               vpx_img_fmt_t,
+               unsigned int,
+               unsigned int,
+               unsigned int,
+               unsigned char*),
+              (const, override));
+  MOCK_METHOD(void, img_free, (vpx_image_t * img), (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_enc_config_set,
+              (vpx_codec_ctx_t*, const vpx_codec_enc_cfg_t*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_enc_config_default,
+              (vpx_codec_iface_t*, vpx_codec_enc_cfg_t*, unsigned int),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_enc_init,
+              (vpx_codec_ctx_t*,
+               vpx_codec_iface_t*,
+               const vpx_codec_enc_cfg_t*,
+               vpx_codec_flags_t),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_enc_init_multi,
+              (vpx_codec_ctx_t*,
+               vpx_codec_iface_t*,
+               vpx_codec_enc_cfg_t*,
+               int,
+               vpx_codec_flags_t,
+               vpx_rational_t*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_destroy,
+              (vpx_codec_ctx_t*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, uint32_t),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, int),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, int*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, vpx_roi_map*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, vpx_active_map*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, vpx_scaling_mode*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, vpx_svc_extra_cfg_t*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, vpx_svc_frame_drop_t*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, void*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, vpx_svc_layer_id_t*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*,
+               vp8e_enc_control_id,
+               vpx_svc_ref_frame_config_t*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*,
+               vp8e_enc_control_id,
+               vpx_svc_spatial_layer_sync_t*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_control,
+              (vpx_codec_ctx_t*, vp8e_enc_control_id, vpx_rc_funcs_t*),
+              (const, override));
+  MOCK_METHOD(vpx_codec_err_t,
+              codec_encode,
+              (vpx_codec_ctx_t*,
+               const vpx_image_t*,
+               vpx_codec_pts_t,
+               uint64_t,
+               vpx_enc_frame_flags_t,
+               uint64_t),
+              (const, override));
+  MOCK_METHOD(const vpx_codec_cx_pkt_t*,
+              codec_get_cx_data,
+              (vpx_codec_ctx_t*, vpx_codec_iter_t*),
+              (const, override));
+  MOCK_METHOD(const char*,
+              codec_error_detail,
+              (vpx_codec_ctx_t*),
+              (const, override));
+  MOCK_METHOD(const char*, codec_error, (vpx_codec_ctx_t*), (const, override));
+  MOCK_METHOD(const char*,
+              codec_err_to_string,
+              (vpx_codec_err_t),
+              (const, override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_LIBVPX_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/multiplex/augmented_video_frame_buffer.cc b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/augmented_video_frame_buffer.cc
new file mode 100644
index 0000000000..8740884f5b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/augmented_video_frame_buffer.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h"
+
+#include <stdint.h>
+
+#include <utility>
+
+#include "api/video/video_frame_buffer.h"
+
+namespace webrtc {
+
+AugmentedVideoFrameBuffer::AugmentedVideoFrameBuffer(
+    const rtc::scoped_refptr<VideoFrameBuffer>& video_frame_buffer,
+    std::unique_ptr<uint8_t[]> augmenting_data,
+    uint16_t augmenting_data_size)
+    : augmenting_data_size_(augmenting_data_size),
+      augmenting_data_(std::move(augmenting_data)),
+      video_frame_buffer_(video_frame_buffer) {}
+
+rtc::scoped_refptr<VideoFrameBuffer>
+AugmentedVideoFrameBuffer::GetVideoFrameBuffer() const {
+  return video_frame_buffer_;
+}
+
+uint8_t* AugmentedVideoFrameBuffer::GetAugmentingData() const {
+  return augmenting_data_.get();
+}
+
+uint16_t AugmentedVideoFrameBuffer::GetAugmentingDataSize() const {
+  return augmenting_data_size_;
+}
+
+VideoFrameBuffer::Type AugmentedVideoFrameBuffer::type() const {
+  return video_frame_buffer_->type();
+}
+
+int AugmentedVideoFrameBuffer::width() const {
+  return video_frame_buffer_->width();
+}
+
+int AugmentedVideoFrameBuffer::height() const {
+  return video_frame_buffer_->height();
+}
+
+rtc::scoped_refptr<I420BufferInterface> AugmentedVideoFrameBuffer::ToI420() {
+  return video_frame_buffer_->ToI420();
+}
+
+const I420BufferInterface* AugmentedVideoFrameBuffer::GetI420() const {
+  // TODO(https://crbug.com/webrtc/12021): When AugmentedVideoFrameBuffer is
+  // updated to implement the buffer interfaces of relevant
+  // VideoFrameBuffer::Types, stop overriding GetI420() as a workaround to
+  // AugmentedVideoFrameBuffer not being the type that is returned by type().
+  return video_frame_buffer_->GetI420();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h
new file mode 100644
index 0000000000..d711cd07da
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_AUGMENTED_VIDEO_FRAME_BUFFER_H_
+#define MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_AUGMENTED_VIDEO_FRAME_BUFFER_H_
+
+#include <cstdint>
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "api/video/video_frame_buffer.h"
+
+namespace webrtc {
+class AugmentedVideoFrameBuffer : public VideoFrameBuffer {
+ public:
+  AugmentedVideoFrameBuffer(
+      const rtc::scoped_refptr<VideoFrameBuffer>& video_frame_buffer,
+      std::unique_ptr<uint8_t[]> augmenting_data,
+      uint16_t augmenting_data_size);
+
+  // Retrieves the underlying VideoFrameBuffer without the augmented data
+  rtc::scoped_refptr<VideoFrameBuffer> GetVideoFrameBuffer() const;
+
+  // Gets a pointer to the augmenting data and moves ownership to the caller
+  uint8_t* GetAugmentingData() const;
+
+  // Get the size of the augmenting data
+  uint16_t GetAugmentingDataSize() const;
+
+  // Returns the type of the underlying VideoFrameBuffer
+  Type type() const final;
+
+  // Returns the width of the underlying VideoFrameBuffer
+  int width() const final;
+
+  // Returns the height of the underlying VideoFrameBuffer
+  int height() const final;
+
+  // Get the I140 Buffer from the underlying frame buffer
+  rtc::scoped_refptr<I420BufferInterface> ToI420() final;
+  // Returns GetI420() of the underlying VideoFrameBuffer.
+  // TODO(hbos): AugmentedVideoFrameBuffer should not return a type (such as
+  // kI420) without also implementing that type's interface (i.e.
+  // I420BufferInterface). Either implement all possible Type's interfaces or
+  // return kNative.
+  const I420BufferInterface* GetI420() const final;
+
+ private:
+  uint16_t augmenting_data_size_;
+  std::unique_ptr<uint8_t[]> augmenting_data_;
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_AUGMENTED_VIDEO_FRAME_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h
new file mode 100644
index 0000000000..e73f7d0e9f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_DECODER_ADAPTER_H_
+#define MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_DECODER_ADAPTER_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
+
+namespace webrtc {
+
+class MultiplexDecoderAdapter : public VideoDecoder {
+ public:
+  // `factory` is not owned and expected to outlive this class.
+  MultiplexDecoderAdapter(VideoDecoderFactory* factory,
+                          const SdpVideoFormat& associated_format,
+                          bool supports_augmenting_data = false);
+  virtual ~MultiplexDecoderAdapter();
+
+  // Implements VideoDecoder
+  bool Configure(const Settings& settings) override;
+  int32_t Decode(const EncodedImage& input_image,
+                 bool missing_frames,
+                 int64_t render_time_ms) override;
+  int32_t RegisterDecodeCompleteCallback(
+      DecodedImageCallback* callback) override;
+  int32_t Release() override;
+
+  void Decoded(AlphaCodecStream stream_idx,
+               VideoFrame* decoded_image,
+               absl::optional<int32_t> decode_time_ms,
+               absl::optional<uint8_t> qp);
+
+ private:
+  // Wrapper class that redirects Decoded() calls.
+  class AdapterDecodedImageCallback;
+
+  // Holds the decoded image output of a frame.
+  struct DecodedImageData;
+
+  // Holds the augmenting data of an image
+  struct AugmentingData;
+
+  void MergeAlphaImages(VideoFrame* decoded_image,
+                        const absl::optional<int32_t>& decode_time_ms,
+                        const absl::optional<uint8_t>& qp,
+                        VideoFrame* multiplex_decoded_image,
+                        const absl::optional<int32_t>& multiplex_decode_time_ms,
+                        const absl::optional<uint8_t>& multiplex_qp,
+                        std::unique_ptr<uint8_t[]> augmenting_data,
+                        uint16_t augmenting_data_length);
+
+  VideoDecoderFactory* const factory_;
+  const SdpVideoFormat associated_format_;
+  std::vector<std::unique_ptr<VideoDecoder>> decoders_;
+  std::vector<std::unique_ptr<AdapterDecodedImageCallback>> adapter_callbacks_;
+  DecodedImageCallback* decoded_complete_callback_;
+
+  // Holds YUV or AXX decode output of a frame that is identified by timestamp.
+  std::map<uint32_t /* timestamp */, DecodedImageData> decoded_data_;
+  std::map<uint32_t /* timestamp */, AugmentingData> decoded_augmenting_data_;
+  const bool supports_augmenting_data_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_DECODER_ADAPTER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h
new file mode 100644
index 0000000000..2e5aad8a5b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_ENCODER_ADAPTER_H_
+#define MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_ENCODER_ADAPTER_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "api/fec_controller_override.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+enum AlphaCodecStream {
+  kYUVStream = 0,
+  kAXXStream = 1,
+  kAlphaCodecStreams = 2,
+};
+
+class MultiplexEncoderAdapter : public VideoEncoder {
+ public:
+  // `factory` is not owned and expected to outlive this class.
+  MultiplexEncoderAdapter(VideoEncoderFactory* factory,
+                          const SdpVideoFormat& associated_format,
+                          bool supports_augmenting_data = false);
+  virtual ~MultiplexEncoderAdapter();
+
+  // Implements VideoEncoder
+  void SetFecControllerOverride(
+      FecControllerOverride* fec_controller_override) override;
+  int InitEncode(const VideoCodec* inst,
+                 const VideoEncoder::Settings& settings) override;
+  int Encode(const VideoFrame& input_image,
+             const std::vector<VideoFrameType>* frame_types) override;
+  int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
+  void SetRates(const RateControlParameters& parameters) override;
+  void OnPacketLossRateUpdate(float packet_loss_rate) override;
+  void OnRttUpdate(int64_t rtt_ms) override;
+  void OnLossNotification(const LossNotification& loss_notification) override;
+  int Release() override;
+  EncoderInfo GetEncoderInfo() const override;
+
+  EncodedImageCallback::Result OnEncodedImage(
+      AlphaCodecStream stream_idx,
+      const EncodedImage& encodedImage,
+      const CodecSpecificInfo* codecSpecificInfo);
+
+ private:
+  // Wrapper class that redirects OnEncodedImage() calls.
+  class AdapterEncodedImageCallback;
+
+  VideoEncoderFactory* const factory_;
+  const SdpVideoFormat associated_format_;
+  std::vector<std::unique_ptr<VideoEncoder>> encoders_;
+  std::vector<std::unique_ptr<AdapterEncodedImageCallback>> adapter_callbacks_;
+  EncodedImageCallback* encoded_complete_callback_;
+
+  std::map<uint32_t /* timestamp */, MultiplexImage> stashed_images_
+      RTC_GUARDED_BY(mutex_);
+
+  uint16_t picture_index_ = 0;
+  std::vector<uint8_t> multiplex_dummy_planes_;
+
+  int key_frame_interval_;
+  EncodedImage combined_image_;
+
+  Mutex mutex_;
+
+  const bool supports_augmented_data_;
+  int augmenting_data_size_ = 0;
+
+  EncoderInfo encoder_info_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_ENCODER_ADAPTER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
new file mode 100644
index 0000000000..0ad3d3883a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
@@ -0,0 +1,266 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h"
+
+#include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame_buffer.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h"
+#include "modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+class MultiplexDecoderAdapter::AdapterDecodedImageCallback
+    : public webrtc::DecodedImageCallback {
+ public:
+  AdapterDecodedImageCallback(webrtc::MultiplexDecoderAdapter* adapter,
+                              AlphaCodecStream stream_idx)
+      : adapter_(adapter), stream_idx_(stream_idx) {}
+
+  void Decoded(VideoFrame& decoded_image,
+               absl::optional<int32_t> decode_time_ms,
+               absl::optional<uint8_t> qp) override {
+    if (!adapter_)
+      return;
+    adapter_->Decoded(stream_idx_, &decoded_image, decode_time_ms, qp);
+  }
+  int32_t Decoded(VideoFrame& decoded_image) override {
+    RTC_DCHECK_NOTREACHED();
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+  int32_t Decoded(VideoFrame& decoded_image, int64_t decode_time_ms) override {
+    RTC_DCHECK_NOTREACHED();
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+ private:
+  MultiplexDecoderAdapter* adapter_;
+  const AlphaCodecStream stream_idx_;
+};
+
+struct MultiplexDecoderAdapter::DecodedImageData {
+  explicit DecodedImageData(AlphaCodecStream stream_idx)
+      : stream_idx_(stream_idx),
+        decoded_image_(
+            VideoFrame::Builder()
+                .set_video_frame_buffer(
+                    I420Buffer::Create(1 /* width */, 1 /* height */))
+                .set_timestamp_rtp(0)
+                .set_timestamp_us(0)
+                .set_rotation(kVideoRotation_0)
+                .build()) {
+    RTC_DCHECK_EQ(kAXXStream, stream_idx);
+  }
+  DecodedImageData(AlphaCodecStream stream_idx,
+                   const VideoFrame& decoded_image,
+                   const absl::optional<int32_t>& decode_time_ms,
+                   const absl::optional<uint8_t>& qp)
+      : stream_idx_(stream_idx),
+        decoded_image_(decoded_image),
+        decode_time_ms_(decode_time_ms),
+        qp_(qp) {}
+
+  DecodedImageData() = delete;
+  DecodedImageData(const DecodedImageData&) = delete;
+  DecodedImageData& operator=(const DecodedImageData&) = delete;
+
+  const AlphaCodecStream stream_idx_;
+  VideoFrame decoded_image_;
+  const absl::optional<int32_t> decode_time_ms_;
+  const absl::optional<uint8_t> qp_;
+};
+
+struct MultiplexDecoderAdapter::AugmentingData {
+  AugmentingData(std::unique_ptr<uint8_t[]> augmenting_data, uint16_t data_size)
+      : data_(std::move(augmenting_data)), size_(data_size) {}
+  AugmentingData() = delete;
+  AugmentingData(const AugmentingData&) = delete;
+  AugmentingData& operator=(const AugmentingData&) = delete;
+
+  std::unique_ptr<uint8_t[]> data_;
+  const uint16_t size_;
+};
+
+MultiplexDecoderAdapter::MultiplexDecoderAdapter(
+    VideoDecoderFactory* factory,
+    const SdpVideoFormat& associated_format,
+    bool supports_augmenting_data)
+    : factory_(factory),
+      associated_format_(associated_format),
+      supports_augmenting_data_(supports_augmenting_data) {}
+
+MultiplexDecoderAdapter::~MultiplexDecoderAdapter() {
+  Release();
+}
+
+bool MultiplexDecoderAdapter::Configure(const Settings& settings) {
+  RTC_DCHECK_EQ(settings.codec_type(), kVideoCodecMultiplex);
+  Settings associated_settings = settings;
+  associated_settings.set_codec_type(
+      PayloadStringToCodecType(associated_format_.name));
+  for (size_t i = 0; i < kAlphaCodecStreams; ++i) {
+    std::unique_ptr<VideoDecoder> decoder =
+        factory_->CreateVideoDecoder(associated_format_);
+    if (!decoder->Configure(associated_settings)) {
+      return false;
+    }
+    adapter_callbacks_.emplace_back(
+        new MultiplexDecoderAdapter::AdapterDecodedImageCallback(
+            this, static_cast<AlphaCodecStream>(i)));
+    decoder->RegisterDecodeCompleteCallback(adapter_callbacks_.back().get());
+    decoders_.emplace_back(std::move(decoder));
+  }
+  return true;
+}
+
+int32_t MultiplexDecoderAdapter::Decode(const EncodedImage& input_image,
+                                        bool missing_frames,
+                                        int64_t render_time_ms) {
+  MultiplexImage image = MultiplexEncodedImagePacker::Unpack(input_image);
+
+  if (supports_augmenting_data_) {
+    RTC_DCHECK(decoded_augmenting_data_.find(input_image.Timestamp()) ==
+               decoded_augmenting_data_.end());
+    decoded_augmenting_data_.emplace(
+        std::piecewise_construct,
+        std::forward_as_tuple(input_image.Timestamp()),
+        std::forward_as_tuple(std::move(image.augmenting_data),
+                              image.augmenting_data_size));
+  }
+
+  if (image.component_count == 1) {
+    RTC_DCHECK(decoded_data_.find(input_image.Timestamp()) ==
+               decoded_data_.end());
+    decoded_data_.emplace(std::piecewise_construct,
+                          std::forward_as_tuple(input_image.Timestamp()),
+                          std::forward_as_tuple(kAXXStream));
+  }
+  int32_t rv = 0;
+  for (size_t i = 0; i < image.image_components.size(); i++) {
+    rv = decoders_[image.image_components[i].component_index]->Decode(
+        image.image_components[i].encoded_image, missing_frames,
+        render_time_ms);
+    if (rv != WEBRTC_VIDEO_CODEC_OK)
+      return rv;
+  }
+  return rv;
+}
+
+int32_t MultiplexDecoderAdapter::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  decoded_complete_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t MultiplexDecoderAdapter::Release() {
+  for (auto& decoder : decoders_) {
+    const int32_t rv = decoder->Release();
+    if (rv)
+      return rv;
+  }
+  decoders_.clear();
+  adapter_callbacks_.clear();
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void MultiplexDecoderAdapter::Decoded(AlphaCodecStream stream_idx,
+                                      VideoFrame* decoded_image,
+                                      absl::optional<int32_t> decode_time_ms,
+                                      absl::optional<uint8_t> qp) {
+  const auto& other_decoded_data_it =
+      decoded_data_.find(decoded_image->timestamp());
+  const auto& augmenting_data_it =
+      decoded_augmenting_data_.find(decoded_image->timestamp());
+  const bool has_augmenting_data =
+      augmenting_data_it != decoded_augmenting_data_.end();
+  if (other_decoded_data_it != decoded_data_.end()) {
+    uint16_t augmenting_data_size =
+        has_augmenting_data ? augmenting_data_it->second.size_ : 0;
+    std::unique_ptr<uint8_t[]> augmenting_data =
+        has_augmenting_data ? std::move(augmenting_data_it->second.data_)
+                            : nullptr;
+    auto& other_image_data = other_decoded_data_it->second;
+    if (stream_idx == kYUVStream) {
+      RTC_DCHECK_EQ(kAXXStream, other_image_data.stream_idx_);
+      MergeAlphaImages(decoded_image, decode_time_ms, qp,
+                       &other_image_data.decoded_image_,
+                       other_image_data.decode_time_ms_, other_image_data.qp_,
+                       std::move(augmenting_data), augmenting_data_size);
+    } else {
+      RTC_DCHECK_EQ(kYUVStream, other_image_data.stream_idx_);
+      RTC_DCHECK_EQ(kAXXStream, stream_idx);
+      MergeAlphaImages(&other_image_data.decoded_image_,
+                       other_image_data.decode_time_ms_, other_image_data.qp_,
+                       decoded_image, decode_time_ms, qp,
+                       std::move(augmenting_data), augmenting_data_size);
+    }
+    decoded_data_.erase(decoded_data_.begin(), other_decoded_data_it);
+    if (has_augmenting_data) {
+      decoded_augmenting_data_.erase(decoded_augmenting_data_.begin(),
+                                     augmenting_data_it);
+    }
+    return;
+  }
+  RTC_DCHECK(decoded_data_.find(decoded_image->timestamp()) ==
+             decoded_data_.end());
+  decoded_data_.emplace(
+      std::piecewise_construct,
+      std::forward_as_tuple(decoded_image->timestamp()),
+      std::forward_as_tuple(stream_idx, *decoded_image, decode_time_ms, qp));
+}
+
+void MultiplexDecoderAdapter::MergeAlphaImages(
+    VideoFrame* decoded_image,
+    const absl::optional<int32_t>& decode_time_ms,
+    const absl::optional<uint8_t>& qp,
+    VideoFrame* alpha_decoded_image,
+    const absl::optional<int32_t>& alpha_decode_time_ms,
+    const absl::optional<uint8_t>& alpha_qp,
+    std::unique_ptr<uint8_t[]> augmenting_data,
+    uint16_t augmenting_data_length) {
+  rtc::scoped_refptr<VideoFrameBuffer> merged_buffer;
+  if (!alpha_decoded_image->timestamp()) {
+    merged_buffer = decoded_image->video_frame_buffer();
+  } else {
+    rtc::scoped_refptr<webrtc::I420BufferInterface> yuv_buffer =
+        decoded_image->video_frame_buffer()->ToI420();
+    rtc::scoped_refptr<webrtc::I420BufferInterface> alpha_buffer =
+        alpha_decoded_image->video_frame_buffer()->ToI420();
+    RTC_DCHECK_EQ(yuv_buffer->width(), alpha_buffer->width());
+    RTC_DCHECK_EQ(yuv_buffer->height(), alpha_buffer->height());
+    merged_buffer = WrapI420ABuffer(
+        yuv_buffer->width(), yuv_buffer->height(), yuv_buffer->DataY(),
+        yuv_buffer->StrideY(), yuv_buffer->DataU(), yuv_buffer->StrideU(),
+        yuv_buffer->DataV(), yuv_buffer->StrideV(), alpha_buffer->DataY(),
+        alpha_buffer->StrideY(),
+        // To keep references alive.
+        [yuv_buffer, alpha_buffer] {});
+  }
+  if (supports_augmenting_data_) {
+    merged_buffer = rtc::make_ref_counted<AugmentedVideoFrameBuffer>(
+        merged_buffer, std::move(augmenting_data), augmenting_data_length);
+  }
+
+  VideoFrame merged_image = VideoFrame::Builder()
+                                .set_video_frame_buffer(merged_buffer)
+                                .set_timestamp_rtp(decoded_image->timestamp())
+                                .set_timestamp_us(0)
+                                .set_rotation(decoded_image->rotation())
+                                .set_id(decoded_image->id())
+                                .set_packet_infos(decoded_image->packet_infos())
+                                .build();
+  decoded_complete_callback_->Decoded(merged_image, decode_time_ms, qp);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc
new file mode 100644
index 0000000000..0f05d1a89c
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc
@@ -0,0 +1,277 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h"
+
+#include <cstring>
+#include <utility>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+int PackHeader(uint8_t* buffer, MultiplexImageHeader header) {
+  int offset = 0;
+  ByteWriter<uint8_t>::WriteBigEndian(buffer + offset, header.component_count);
+  offset += sizeof(uint8_t);
+
+  ByteWriter<uint16_t>::WriteBigEndian(buffer + offset, header.image_index);
+  offset += sizeof(uint16_t);
+
+  ByteWriter<uint16_t>::WriteBigEndian(buffer + offset,
+                                       header.augmenting_data_size);
+  offset += sizeof(uint16_t);
+
+  ByteWriter<uint32_t>::WriteBigEndian(buffer + offset,
+                                       header.augmenting_data_offset);
+  offset += sizeof(uint32_t);
+
+  ByteWriter<uint32_t>::WriteBigEndian(buffer + offset,
+                                       header.first_component_header_offset);
+  offset += sizeof(uint32_t);
+
+  RTC_DCHECK_EQ(offset, kMultiplexImageHeaderSize);
+  return offset;
+}
+
+MultiplexImageHeader UnpackHeader(const uint8_t* buffer) {
+  MultiplexImageHeader header;
+  int offset = 0;
+  header.component_count = ByteReader<uint8_t>::ReadBigEndian(buffer + offset);
+  offset += sizeof(uint8_t);
+
+  header.image_index = ByteReader<uint16_t>::ReadBigEndian(buffer + offset);
+  offset += sizeof(uint16_t);
+
+  header.augmenting_data_size =
+      ByteReader<uint16_t>::ReadBigEndian(buffer + offset);
+  offset += sizeof(uint16_t);
+
+  header.augmenting_data_offset =
+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
+  offset += sizeof(uint32_t);
+
+  header.first_component_header_offset =
+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
+  offset += sizeof(uint32_t);
+
+  RTC_DCHECK_EQ(offset, kMultiplexImageHeaderSize);
+  return header;
+}
+
+int PackFrameHeader(uint8_t* buffer,
+                    MultiplexImageComponentHeader frame_header) {
+  int offset = 0;
+  ByteWriter<uint32_t>::WriteBigEndian(
+      buffer + offset, frame_header.next_component_header_offset);
+  offset += sizeof(uint32_t);
+
+  ByteWriter<uint8_t>::WriteBigEndian(buffer + offset,
+                                      frame_header.component_index);
+  offset += sizeof(uint8_t);
+
+  ByteWriter<uint32_t>::WriteBigEndian(buffer + offset,
+                                       frame_header.bitstream_offset);
+  offset += sizeof(uint32_t);
+
+  ByteWriter<uint32_t>::WriteBigEndian(buffer + offset,
+                                       frame_header.bitstream_length);
+  offset += sizeof(uint32_t);
+
+  ByteWriter<uint8_t>::WriteBigEndian(buffer + offset, frame_header.codec_type);
+  offset += sizeof(uint8_t);
+
+  ByteWriter<uint8_t>::WriteBigEndian(
+      buffer + offset, static_cast<uint8_t>(frame_header.frame_type));
+  offset += sizeof(uint8_t);
+
+  RTC_DCHECK_EQ(offset, kMultiplexImageComponentHeaderSize);
+  return offset;
+}
+
+MultiplexImageComponentHeader UnpackFrameHeader(const uint8_t* buffer) {
+  MultiplexImageComponentHeader frame_header;
+  int offset = 0;
+
+  frame_header.next_component_header_offset =
+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
+  offset += sizeof(uint32_t);
+
+  frame_header.component_index =
+      ByteReader<uint8_t>::ReadBigEndian(buffer + offset);
+  offset += sizeof(uint8_t);
+
+  frame_header.bitstream_offset =
+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
+  offset += sizeof(uint32_t);
+
+  frame_header.bitstream_length =
+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
+  offset += sizeof(uint32_t);
+
+  // This makes the wire format depend on the numeric values of the
+  // VideoCodecType and VideoFrameType enum constants.
+  frame_header.codec_type = static_cast<VideoCodecType>(
+      ByteReader<uint8_t>::ReadBigEndian(buffer + offset));
+  offset += sizeof(uint8_t);
+
+  frame_header.frame_type = static_cast<VideoFrameType>(
+      ByteReader<uint8_t>::ReadBigEndian(buffer + offset));
+  offset += sizeof(uint8_t);
+
+  RTC_DCHECK_EQ(offset, kMultiplexImageComponentHeaderSize);
+  return frame_header;
+}
+
+void PackBitstream(uint8_t* buffer, MultiplexImageComponent image) {
+  memcpy(buffer, image.encoded_image.data(), image.encoded_image.size());
+}
+
+MultiplexImage::MultiplexImage(uint16_t picture_index,
+                               uint8_t frame_count,
+                               std::unique_ptr<uint8_t[]> augmenting_data,
+                               uint16_t augmenting_data_size)
+    : image_index(picture_index),
+      component_count(frame_count),
+      augmenting_data_size(augmenting_data_size),
+      augmenting_data(std::move(augmenting_data)) {}
+
+EncodedImage MultiplexEncodedImagePacker::PackAndRelease(
+    const MultiplexImage& multiplex_image) {
+  MultiplexImageHeader header;
+  std::vector<MultiplexImageComponentHeader> frame_headers;
+
+  header.component_count = multiplex_image.component_count;
+  header.image_index = multiplex_image.image_index;
+  int header_offset = kMultiplexImageHeaderSize;
+  header.first_component_header_offset = header_offset;
+  header.augmenting_data_offset =
+      header_offset +
+      kMultiplexImageComponentHeaderSize * header.component_count;
+  header.augmenting_data_size = multiplex_image.augmenting_data_size;
+  int bitstream_offset =
+      header.augmenting_data_offset + header.augmenting_data_size;
+
+  const std::vector<MultiplexImageComponent>& images =
+      multiplex_image.image_components;
+  EncodedImage combined_image = images[0].encoded_image;
+  for (size_t i = 0; i < images.size(); i++) {
+    MultiplexImageComponentHeader frame_header;
+    header_offset += kMultiplexImageComponentHeaderSize;
+    frame_header.next_component_header_offset =
+        (i == images.size() - 1) ? 0 : header_offset;
+    frame_header.component_index = images[i].component_index;
+
+    frame_header.bitstream_offset = bitstream_offset;
+    frame_header.bitstream_length =
+        static_cast<uint32_t>(images[i].encoded_image.size());
+    bitstream_offset += frame_header.bitstream_length;
+
+    frame_header.codec_type = images[i].codec_type;
+    frame_header.frame_type = images[i].encoded_image._frameType;
+
+    // As long as one component is delta frame, we have to mark the combined
+    // frame as delta frame, because it is necessary for all components to be
+    // key frame so as to decode the whole image without previous frame data.
+    // Thus only when all components are key frames, we can mark the combined
+    // frame as key frame.
+    if (frame_header.frame_type == VideoFrameType::kVideoFrameDelta) {
+      combined_image._frameType = VideoFrameType::kVideoFrameDelta;
+    }
+
+    frame_headers.push_back(frame_header);
+  }
+
+  auto buffer = EncodedImageBuffer::Create(bitstream_offset);
+  combined_image.SetEncodedData(buffer);
+
+  // header
+  header_offset = PackHeader(buffer->data(), header);
+  RTC_DCHECK_EQ(header.first_component_header_offset,
+                kMultiplexImageHeaderSize);
+
+  // Frame Header
+  for (size_t i = 0; i < images.size(); i++) {
+    int relative_offset =
+        PackFrameHeader(buffer->data() + header_offset, frame_headers[i]);
+    RTC_DCHECK_EQ(relative_offset, kMultiplexImageComponentHeaderSize);
+
+    header_offset = frame_headers[i].next_component_header_offset;
+    RTC_DCHECK_EQ(header_offset,
+                  (i == images.size() - 1)
+                      ? 0
+                      : (kMultiplexImageHeaderSize +
+                         kMultiplexImageComponentHeaderSize * (i + 1)));
+  }
+
+  // Augmenting Data
+  if (multiplex_image.augmenting_data_size != 0) {
+    memcpy(buffer->data() + header.augmenting_data_offset,
+           multiplex_image.augmenting_data.get(),
+           multiplex_image.augmenting_data_size);
+  }
+
+  // Bitstreams
+  for (size_t i = 0; i < images.size(); i++) {
+    PackBitstream(buffer->data() + frame_headers[i].bitstream_offset,
+                  images[i]);
+  }
+
+  return combined_image;
+}
+
+MultiplexImage MultiplexEncodedImagePacker::Unpack(
+    const EncodedImage& combined_image) {
+  const MultiplexImageHeader& header = UnpackHeader(combined_image.data());
+
+  std::vector<MultiplexImageComponentHeader> frame_headers;
+  int header_offset = header.first_component_header_offset;
+
+  while (header_offset > 0) {
+    frame_headers.push_back(
+        UnpackFrameHeader(combined_image.data() + header_offset));
+    header_offset = frame_headers.back().next_component_header_offset;
+  }
+
+  RTC_DCHECK_LE(frame_headers.size(), header.component_count);
+  std::unique_ptr<uint8_t[]> augmenting_data = nullptr;
+  if (header.augmenting_data_size != 0) {
+    augmenting_data =
+        std::unique_ptr<uint8_t[]>(new uint8_t[header.augmenting_data_size]);
+    memcpy(augmenting_data.get(),
+           combined_image.data() + header.augmenting_data_offset,
+           header.augmenting_data_size);
+  }
+
+  MultiplexImage multiplex_image(header.image_index, header.component_count,
+                                 std::move(augmenting_data),
+                                 header.augmenting_data_size);
+
+  for (size_t i = 0; i < frame_headers.size(); i++) {
+    MultiplexImageComponent image_component;
+    image_component.component_index = frame_headers[i].component_index;
+    image_component.codec_type = frame_headers[i].codec_type;
+
+    EncodedImage encoded_image = combined_image;
+    encoded_image.SetTimestamp(combined_image.Timestamp());
+    encoded_image._frameType = frame_headers[i].frame_type;
+    encoded_image.SetEncodedData(EncodedImageBuffer::Create(
+        combined_image.data() + frame_headers[i].bitstream_offset,
+        frame_headers[i].bitstream_length));
+
+    image_component.encoded_image = encoded_image;
+
+    multiplex_image.image_components.push_back(image_component);
+  }
+
+  return multiplex_image;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h
new file mode 100644
index 0000000000..299a0159d5
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_MULTIPLEX_MULTIPLEX_ENCODED_IMAGE_PACKER_H_
+#define MODULES_VIDEO_CODING_CODECS_MULTIPLEX_MULTIPLEX_ENCODED_IMAGE_PACKER_H_
+
+#include <cstdint>
+#include <memory>
+#include <vector>
+
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_codec.h"
+
+namespace webrtc {
+
+// Struct describing the whole bundle of multiple frames of an image.
+// This struct is expected to be the set in the beginning of a picture's
+// bitstream data.
+struct MultiplexImageHeader {
+  // The number of frame components making up the complete picture data.
+  // For example, `frame_count` = 2 for the case of YUV frame with Alpha frame.
+  uint8_t component_count;
+
+  // The increasing image ID given by the encoder. For different components
+  // of a single picture, they have the same `picture_index`.
+  uint16_t image_index;
+
+  // The location of the first MultiplexImageComponentHeader in the bitstream,
+  // in terms of byte from the beginning of the bitstream.
+  uint32_t first_component_header_offset;
+
+  // The location of the augmenting data in the bitstream, in terms of bytes
+  // from the beginning of the bitstream
+  uint32_t augmenting_data_offset;
+
+  // The size of the augmenting data in the bitstream it terms of byte
+  uint16_t augmenting_data_size;
+};
+const int kMultiplexImageHeaderSize =
+    sizeof(uint8_t) + 2 * sizeof(uint16_t) + 2 * sizeof(uint32_t);
+
+// Struct describing the individual image component's content.
+struct MultiplexImageComponentHeader {
+  // The location of the next MultiplexImageComponentHeader in the bitstream,
+  // in terms of the byte from the beginning of the bitstream;
+  uint32_t next_component_header_offset;
+
+  // Identifies which component this frame represent, i.e. YUV frame vs Alpha
+  // frame.
+  uint8_t component_index;
+
+  // The location of the real encoded image data of the frame in the bitstream,
+  // in terms of byte from the beginning of the bitstream.
+  uint32_t bitstream_offset;
+
+  // Indicates the number of bytes of the encoded image data.
+  uint32_t bitstream_length;
+
+  // Indicated the underlying VideoCodecType of the frame, i.e. VP9 or VP8 etc.
+  VideoCodecType codec_type;
+
+  // Indicated the underlying frame is a key frame or delta frame.
+  VideoFrameType frame_type;
+};
+const int kMultiplexImageComponentHeaderSize =
+    sizeof(uint32_t) + sizeof(uint8_t) + sizeof(uint32_t) + sizeof(uint32_t) +
+    sizeof(uint8_t) + sizeof(uint8_t);
+
+// Struct holding the encoded image for one component.
+struct MultiplexImageComponent {
+  // Indicated the underlying VideoCodecType of the frame, i.e. VP9 or VP8 etc.
+  VideoCodecType codec_type;
+
+  // Identifies which component this frame represent, i.e. YUV frame vs Alpha
+  // frame.
+  uint8_t component_index;
+
+  // Stores the actual frame data of the encoded image.
+  EncodedImage encoded_image;
+};
+
+// Struct holding the whole frame bundle of components of an image.
+struct MultiplexImage {
+  uint16_t image_index;
+  uint8_t component_count;
+  uint16_t augmenting_data_size;
+  std::unique_ptr<uint8_t[]> augmenting_data;
+  std::vector<MultiplexImageComponent> image_components;
+
+  MultiplexImage(uint16_t picture_index,
+                 uint8_t component_count,
+                 std::unique_ptr<uint8_t[]> augmenting_data,
+                 uint16_t augmenting_data_size);
+};
+
+// A utility class providing conversion between two representations of a
+// multiplex image frame:
+// 1. Packed version is just one encoded image, we pack all necessary metadata
+//    in the bitstream as headers.
+// 2. Unpacked version is essentially a list of encoded images, one for one
+//    component.
+class MultiplexEncodedImagePacker {
+ public:
+  // Note: It is caller responsibility to release the buffer of the result.
+  static EncodedImage PackAndRelease(const MultiplexImage& image);
+
+  // Note: The image components just share the memory with `combined_image`.
+  static MultiplexImage Unpack(const EncodedImage& combined_image);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_MULTIPLEX_MULTIPLEX_ENCODED_IMAGE_PACKER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc
new file mode 100644
index 0000000000..80744e2d8c
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc
@@ -0,0 +1,353 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
+
+#include <cstring>
+
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_encoder.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "media/base/video_common.h"
+#include "modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+// Callback wrapper that helps distinguish returned results from `encoders_`
+// instances.
+class MultiplexEncoderAdapter::AdapterEncodedImageCallback
+    : public webrtc::EncodedImageCallback {
+ public:
+  AdapterEncodedImageCallback(webrtc::MultiplexEncoderAdapter* adapter,
+                              AlphaCodecStream stream_idx)
+      : adapter_(adapter), stream_idx_(stream_idx) {}
+
+  EncodedImageCallback::Result OnEncodedImage(
+      const EncodedImage& encoded_image,
+      const CodecSpecificInfo* codec_specific_info) override {
+    if (!adapter_)
+      return Result(Result::OK);
+    return adapter_->OnEncodedImage(stream_idx_, encoded_image,
+                                    codec_specific_info);
+  }
+
+ private:
+  MultiplexEncoderAdapter* adapter_;
+  const AlphaCodecStream stream_idx_;
+};
+
+MultiplexEncoderAdapter::MultiplexEncoderAdapter(
+    VideoEncoderFactory* factory,
+    const SdpVideoFormat& associated_format,
+    bool supports_augmented_data)
+    : factory_(factory),
+      associated_format_(associated_format),
+      encoded_complete_callback_(nullptr),
+      key_frame_interval_(0),
+      supports_augmented_data_(supports_augmented_data) {}
+
+MultiplexEncoderAdapter::~MultiplexEncoderAdapter() {
+  Release();
+}
+
+void MultiplexEncoderAdapter::SetFecControllerOverride(
+    FecControllerOverride* fec_controller_override) {
+  // Ignored.
+}
+
+int MultiplexEncoderAdapter::InitEncode(
+    const VideoCodec* inst,
+    const VideoEncoder::Settings& settings) {
+  const size_t buffer_size =
+      CalcBufferSize(VideoType::kI420, inst->width, inst->height);
+  multiplex_dummy_planes_.resize(buffer_size);
+  // It is more expensive to encode 0x00, so use 0x80 instead.
+  std::fill(multiplex_dummy_planes_.begin(), multiplex_dummy_planes_.end(),
+            0x80);
+
+  RTC_DCHECK_EQ(kVideoCodecMultiplex, inst->codecType);
+  VideoCodec video_codec = *inst;
+  video_codec.codecType = PayloadStringToCodecType(associated_format_.name);
+
+  // Take over the key frame interval at adapter level, because we have to
+  // sync the key frames for both sub-encoders.
+  switch (video_codec.codecType) {
+    case kVideoCodecVP8:
+      key_frame_interval_ = video_codec.VP8()->keyFrameInterval;
+      video_codec.VP8()->keyFrameInterval = 0;
+      break;
+    case kVideoCodecVP9:
+      key_frame_interval_ = video_codec.VP9()->keyFrameInterval;
+      video_codec.VP9()->keyFrameInterval = 0;
+      break;
+    case kVideoCodecH264:
+      key_frame_interval_ = video_codec.H264()->keyFrameInterval;
+      video_codec.H264()->keyFrameInterval = 0;
+      break;
+    default:
+      break;
+  }
+
+  encoder_info_ = EncoderInfo();
+  encoder_info_.implementation_name = "MultiplexEncoderAdapter (";
+  encoder_info_.requested_resolution_alignment = 1;
+  encoder_info_.apply_alignment_to_all_simulcast_layers = false;
+  // This needs to be false so that we can do the split in Encode().
+  encoder_info_.supports_native_handle = false;
+
+  for (size_t i = 0; i < kAlphaCodecStreams; ++i) {
+    std::unique_ptr<VideoEncoder> encoder =
+        factory_->CreateVideoEncoder(associated_format_);
+    const int rv = encoder->InitEncode(&video_codec, settings);
+    if (rv) {
+      RTC_LOG(LS_ERROR) << "Failed to create multiplex codec index " << i;
+      return rv;
+    }
+    adapter_callbacks_.emplace_back(new AdapterEncodedImageCallback(
+        this, static_cast<AlphaCodecStream>(i)));
+    encoder->RegisterEncodeCompleteCallback(adapter_callbacks_.back().get());
+
+    const EncoderInfo& encoder_impl_info = encoder->GetEncoderInfo();
+    encoder_info_.implementation_name += encoder_impl_info.implementation_name;
+    if (i != kAlphaCodecStreams - 1) {
+      encoder_info_.implementation_name += ", ";
+    }
+    // Uses hardware support if any of the encoders uses it.
+    // For example, if we are having issues with down-scaling due to
+    // pipelining delay in HW encoders we need higher encoder usage
+    // thresholds in CPU adaptation.
+    if (i == 0) {
+      encoder_info_.is_hardware_accelerated =
+          encoder_impl_info.is_hardware_accelerated;
+    } else {
+      encoder_info_.is_hardware_accelerated |=
+          encoder_impl_info.is_hardware_accelerated;
+    }
+
+    encoder_info_.requested_resolution_alignment = cricket::LeastCommonMultiple(
+        encoder_info_.requested_resolution_alignment,
+        encoder_impl_info.requested_resolution_alignment);
+
+    if (encoder_impl_info.apply_alignment_to_all_simulcast_layers) {
+      encoder_info_.apply_alignment_to_all_simulcast_layers = true;
+    }
+
+    encoders_.emplace_back(std::move(encoder));
+  }
+  encoder_info_.implementation_name += ")";
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int MultiplexEncoderAdapter::Encode(
+    const VideoFrame& input_image,
+    const std::vector<VideoFrameType>* frame_types) {
+  if (!encoded_complete_callback_) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  // The input image is forwarded as-is, unless it is a native buffer and
+  // `supports_augmented_data_` is true in which case we need to map it in order
+  // to access the underlying AugmentedVideoFrameBuffer.
+  VideoFrame forwarded_image = input_image;
+  if (supports_augmented_data_ &&
+      forwarded_image.video_frame_buffer()->type() ==
+          VideoFrameBuffer::Type::kNative) {
+    auto info = GetEncoderInfo();
+    rtc::scoped_refptr<VideoFrameBuffer> mapped_buffer =
+        forwarded_image.video_frame_buffer()->GetMappedFrameBuffer(
+            info.preferred_pixel_formats);
+    if (!mapped_buffer) {
+      // Unable to map the buffer.
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    forwarded_image.set_video_frame_buffer(std::move(mapped_buffer));
+  }
+
+  std::vector<VideoFrameType> adjusted_frame_types;
+  if (key_frame_interval_ > 0 && picture_index_ % key_frame_interval_ == 0) {
+    adjusted_frame_types.push_back(VideoFrameType::kVideoFrameKey);
+  } else {
+    adjusted_frame_types.push_back(VideoFrameType::kVideoFrameDelta);
+  }
+  const bool has_alpha = forwarded_image.video_frame_buffer()->type() ==
+                         VideoFrameBuffer::Type::kI420A;
+  std::unique_ptr<uint8_t[]> augmenting_data = nullptr;
+  uint16_t augmenting_data_length = 0;
+  AugmentedVideoFrameBuffer* augmented_video_frame_buffer = nullptr;
+  if (supports_augmented_data_) {
+    augmented_video_frame_buffer = static_cast<AugmentedVideoFrameBuffer*>(
+        forwarded_image.video_frame_buffer().get());
+    augmenting_data_length =
+        augmented_video_frame_buffer->GetAugmentingDataSize();
+    augmenting_data =
+        std::unique_ptr<uint8_t[]>(new uint8_t[augmenting_data_length]);
+    memcpy(augmenting_data.get(),
+           augmented_video_frame_buffer->GetAugmentingData(),
+           augmenting_data_length);
+    augmenting_data_size_ = augmenting_data_length;
+  }
+
+  {
+    MutexLock lock(&mutex_);
+    stashed_images_.emplace(
+        std::piecewise_construct,
+        std::forward_as_tuple(forwarded_image.timestamp()),
+        std::forward_as_tuple(
+            picture_index_, has_alpha ? kAlphaCodecStreams : 1,
+            std::move(augmenting_data), augmenting_data_length));
+  }
+
+  ++picture_index_;
+
+  // Encode YUV
+  int rv =
+      encoders_[kYUVStream]->Encode(forwarded_image, &adjusted_frame_types);
+
+  // If we do not receive an alpha frame, we send a single frame for this
+  // `picture_index_`. The receiver will receive `frame_count` as 1 which
+  // specifies this case.
+  if (rv || !has_alpha)
+    return rv;
+
+  // Encode AXX
+  rtc::scoped_refptr<VideoFrameBuffer> frame_buffer =
+      supports_augmented_data_
+          ? augmented_video_frame_buffer->GetVideoFrameBuffer()
+          : forwarded_image.video_frame_buffer();
+  const I420ABufferInterface* yuva_buffer = frame_buffer->GetI420A();
+  rtc::scoped_refptr<I420BufferInterface> alpha_buffer =
+      WrapI420Buffer(forwarded_image.width(), forwarded_image.height(),
+                     yuva_buffer->DataA(), yuva_buffer->StrideA(),
+                     multiplex_dummy_planes_.data(), yuva_buffer->StrideU(),
+                     multiplex_dummy_planes_.data(), yuva_buffer->StrideV(),
+                     // To keep reference alive.
+                     [frame_buffer] {});
+  VideoFrame alpha_image =
+      VideoFrame::Builder()
+          .set_video_frame_buffer(alpha_buffer)
+          .set_timestamp_rtp(forwarded_image.timestamp())
+          .set_timestamp_ms(forwarded_image.render_time_ms())
+          .set_rotation(forwarded_image.rotation())
+          .set_id(forwarded_image.id())
+          .set_packet_infos(forwarded_image.packet_infos())
+          .build();
+  rv = encoders_[kAXXStream]->Encode(alpha_image, &adjusted_frame_types);
+  return rv;
+}
+
+int MultiplexEncoderAdapter::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  encoded_complete_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void MultiplexEncoderAdapter::SetRates(
+    const RateControlParameters& parameters) {
+  VideoBitrateAllocation bitrate_allocation(parameters.bitrate);
+  bitrate_allocation.SetBitrate(
+      0, 0, parameters.bitrate.GetBitrate(0, 0) - augmenting_data_size_);
+  for (auto& encoder : encoders_) {
+    // TODO(emircan): `framerate` is used to calculate duration in encoder
+    // instances. We report the total frame rate to keep real time for now.
+    // Remove this after refactoring duration logic.
+    encoder->SetRates(RateControlParameters(
+        bitrate_allocation,
+        static_cast<uint32_t>(encoders_.size() * parameters.framerate_fps),
+        parameters.bandwidth_allocation -
+            DataRate::BitsPerSec(augmenting_data_size_)));
+  }
+}
+
+void MultiplexEncoderAdapter::OnPacketLossRateUpdate(float packet_loss_rate) {
+  for (auto& encoder : encoders_) {
+    encoder->OnPacketLossRateUpdate(packet_loss_rate);
+  }
+}
+
+void MultiplexEncoderAdapter::OnRttUpdate(int64_t rtt_ms) {
+  for (auto& encoder : encoders_) {
+    encoder->OnRttUpdate(rtt_ms);
+  }
+}
+
+void MultiplexEncoderAdapter::OnLossNotification(
+    const LossNotification& loss_notification) {
+  for (auto& encoder : encoders_) {
+    encoder->OnLossNotification(loss_notification);
+  }
+}
+
+int MultiplexEncoderAdapter::Release() {
+  for (auto& encoder : encoders_) {
+    const int rv = encoder->Release();
+    if (rv)
+      return rv;
+  }
+  encoders_.clear();
+  adapter_callbacks_.clear();
+  MutexLock lock(&mutex_);
+  stashed_images_.clear();
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+VideoEncoder::EncoderInfo MultiplexEncoderAdapter::GetEncoderInfo() const {
+  return encoder_info_;
+}
+
+EncodedImageCallback::Result MultiplexEncoderAdapter::OnEncodedImage(
+    AlphaCodecStream stream_idx,
+    const EncodedImage& encodedImage,
+    const CodecSpecificInfo* codecSpecificInfo) {
+  // Save the image
+  MultiplexImageComponent image_component;
+  image_component.component_index = stream_idx;
+  image_component.codec_type =
+      PayloadStringToCodecType(associated_format_.name);
+  image_component.encoded_image = encodedImage;
+
+  MutexLock lock(&mutex_);
+  const auto& stashed_image_itr =
+      stashed_images_.find(encodedImage.Timestamp());
+  const auto& stashed_image_next_itr = std::next(stashed_image_itr, 1);
+  RTC_DCHECK(stashed_image_itr != stashed_images_.end());
+  MultiplexImage& stashed_image = stashed_image_itr->second;
+  const uint8_t frame_count = stashed_image.component_count;
+
+  stashed_image.image_components.push_back(image_component);
+
+  if (stashed_image.image_components.size() == frame_count) {
+    // Complete case
+    for (auto iter = stashed_images_.begin();
+         iter != stashed_images_.end() && iter != stashed_image_next_itr;
+         iter++) {
+      // No image at all, skip.
+      if (iter->second.image_components.size() == 0)
+        continue;
+
+      // We have to send out those stashed frames, otherwise the delta frame
+      // dependency chain is broken.
+      combined_image_ =
+          MultiplexEncodedImagePacker::PackAndRelease(iter->second);
+
+      CodecSpecificInfo codec_info = *codecSpecificInfo;
+      codec_info.codecType = kVideoCodecMultiplex;
+      encoded_complete_callback_->OnEncodedImage(combined_image_, &codec_info);
+    }
+
+    stashed_images_.erase(stashed_images_.begin(), stashed_image_next_itr);
+  }
+  return EncodedImageCallback::Result(EncodedImageCallback::Result::OK);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc
new file mode 100644
index 0000000000..be0f5deb52
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc
@@ -0,0 +1,319 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/test/mock_video_decoder_factory.h"
+#include "api/test/mock_video_encoder_factory.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h"
+#include "modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h"
+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
+#include "modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h"
+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/video_codec_settings.h"
+
+using ::testing::_;
+using ::testing::Return;
+
+namespace webrtc {
+
+constexpr const char* kMultiplexAssociatedCodecName = cricket::kVp9CodecName;
+const VideoCodecType kMultiplexAssociatedCodecType =
+    PayloadStringToCodecType(kMultiplexAssociatedCodecName);
+
+class TestMultiplexAdapter : public VideoCodecUnitTest,
+                             public ::testing::WithParamInterface<
+                                 bool /* supports_augmenting_data */> {
+ public:
+  TestMultiplexAdapter()
+      : decoder_factory_(new webrtc::MockVideoDecoderFactory),
+        encoder_factory_(new webrtc::MockVideoEncoderFactory),
+        supports_augmenting_data_(GetParam()) {}
+
+ protected:
+  std::unique_ptr<VideoDecoder> CreateDecoder() override {
+    return std::make_unique<MultiplexDecoderAdapter>(
+        decoder_factory_.get(), SdpVideoFormat(kMultiplexAssociatedCodecName),
+        supports_augmenting_data_);
+  }
+
+  std::unique_ptr<VideoEncoder> CreateEncoder() override {
+    return std::make_unique<MultiplexEncoderAdapter>(
+        encoder_factory_.get(), SdpVideoFormat(kMultiplexAssociatedCodecName),
+        supports_augmenting_data_);
+  }
+
+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
+    webrtc::test::CodecSettings(kMultiplexAssociatedCodecType, codec_settings);
+    codec_settings->VP9()->numberOfTemporalLayers = 1;
+    codec_settings->VP9()->numberOfSpatialLayers = 1;
+    codec_settings->codecType = webrtc::kVideoCodecMultiplex;
+  }
+
+  std::unique_ptr<VideoFrame> CreateDataAugmentedInputFrame(
+      VideoFrame* video_frame) {
+    rtc::scoped_refptr<VideoFrameBuffer> video_buffer =
+        video_frame->video_frame_buffer();
+    std::unique_ptr<uint8_t[]> data =
+        std::unique_ptr<uint8_t[]>(new uint8_t[16]);
+    for (int i = 0; i < 16; i++) {
+      data[i] = i;
+    }
+    auto augmented_video_frame_buffer =
+        rtc::make_ref_counted<AugmentedVideoFrameBuffer>(video_buffer,
+                                                         std::move(data), 16);
+    return std::make_unique<VideoFrame>(
+        VideoFrame::Builder()
+            .set_video_frame_buffer(augmented_video_frame_buffer)
+            .set_timestamp_rtp(video_frame->timestamp())
+            .set_timestamp_ms(video_frame->render_time_ms())
+            .set_rotation(video_frame->rotation())
+            .set_id(video_frame->id())
+            .build());
+  }
+
+  std::unique_ptr<VideoFrame> CreateI420AInputFrame() {
+    VideoFrame input_frame = NextInputFrame();
+    rtc::scoped_refptr<webrtc::I420BufferInterface> yuv_buffer =
+        input_frame.video_frame_buffer()->ToI420();
+    rtc::scoped_refptr<I420ABufferInterface> yuva_buffer = WrapI420ABuffer(
+        yuv_buffer->width(), yuv_buffer->height(), yuv_buffer->DataY(),
+        yuv_buffer->StrideY(), yuv_buffer->DataU(), yuv_buffer->StrideU(),
+        yuv_buffer->DataV(), yuv_buffer->StrideV(), yuv_buffer->DataY(),
+        yuv_buffer->StrideY(),
+        // To keep reference alive.
+        [yuv_buffer] {});
+    return std::make_unique<VideoFrame>(VideoFrame::Builder()
+                                            .set_video_frame_buffer(yuva_buffer)
+                                            .set_timestamp_rtp(123)
+                                            .set_timestamp_ms(345)
+                                            .set_rotation(kVideoRotation_0)
+                                            .build());
+  }
+
+  std::unique_ptr<VideoFrame> CreateInputFrame(bool contains_alpha) {
+    std::unique_ptr<VideoFrame> video_frame;
+    if (contains_alpha) {
+      video_frame = CreateI420AInputFrame();
+    } else {
+      VideoFrame next_frame = NextInputFrame();
+      video_frame = std::make_unique<VideoFrame>(
+          VideoFrame::Builder()
+              .set_video_frame_buffer(next_frame.video_frame_buffer())
+              .set_timestamp_rtp(next_frame.timestamp())
+              .set_timestamp_ms(next_frame.render_time_ms())
+              .set_rotation(next_frame.rotation())
+              .set_id(next_frame.id())
+              .build());
+    }
+    if (supports_augmenting_data_) {
+      video_frame = CreateDataAugmentedInputFrame(video_frame.get());
+    }
+
+    return video_frame;
+  }
+
+  void CheckData(rtc::scoped_refptr<VideoFrameBuffer> video_frame_buffer) {
+    if (!supports_augmenting_data_) {
+      return;
+    }
+    AugmentedVideoFrameBuffer* augmented_buffer =
+        static_cast<AugmentedVideoFrameBuffer*>(video_frame_buffer.get());
+    EXPECT_EQ(augmented_buffer->GetAugmentingDataSize(), 16);
+    uint8_t* data = augmented_buffer->GetAugmentingData();
+    for (int i = 0; i < 16; i++) {
+      EXPECT_EQ(data[i], i);
+    }
+  }
+
+  std::unique_ptr<VideoFrame> ExtractAXXFrame(const VideoFrame& video_frame) {
+    rtc::scoped_refptr<VideoFrameBuffer> video_frame_buffer =
+        video_frame.video_frame_buffer();
+    if (supports_augmenting_data_) {
+      AugmentedVideoFrameBuffer* augmentedBuffer =
+          static_cast<AugmentedVideoFrameBuffer*>(video_frame_buffer.get());
+      video_frame_buffer = augmentedBuffer->GetVideoFrameBuffer();
+    }
+    const I420ABufferInterface* yuva_buffer = video_frame_buffer->GetI420A();
+    rtc::scoped_refptr<I420BufferInterface> axx_buffer = WrapI420Buffer(
+        yuva_buffer->width(), yuva_buffer->height(), yuva_buffer->DataA(),
+        yuva_buffer->StrideA(), yuva_buffer->DataU(), yuva_buffer->StrideU(),
+        yuva_buffer->DataV(), yuva_buffer->StrideV(), [video_frame_buffer] {});
+    return std::make_unique<VideoFrame>(VideoFrame::Builder()
+                                            .set_video_frame_buffer(axx_buffer)
+                                            .set_timestamp_rtp(123)
+                                            .set_timestamp_ms(345)
+                                            .set_rotation(kVideoRotation_0)
+                                            .build());
+  }
+
+ private:
+  void SetUp() override {
+    EXPECT_CALL(*decoder_factory_, Die);
+    // The decoders/encoders will be owned by the caller of
+    // CreateVideoDecoder()/CreateVideoEncoder().
+    EXPECT_CALL(*decoder_factory_, CreateVideoDecoder)
+        .Times(2)
+        .WillRepeatedly([] { return VP9Decoder::Create(); });
+
+    EXPECT_CALL(*encoder_factory_, Die);
+    EXPECT_CALL(*encoder_factory_, CreateVideoEncoder)
+        .Times(2)
+        .WillRepeatedly([] { return VP9Encoder::Create(); });
+
+    VideoCodecUnitTest::SetUp();
+  }
+
+  const std::unique_ptr<webrtc::MockVideoDecoderFactory> decoder_factory_;
+  const std::unique_ptr<webrtc::MockVideoEncoderFactory> encoder_factory_;
+  const bool supports_augmenting_data_;
+};
+
+// TODO(emircan): Currently VideoCodecUnitTest tests do a complete setup
+// step that goes beyond constructing `decoder_`. Simplify these tests to do
+// less.
+TEST_P(TestMultiplexAdapter, ConstructAndDestructDecoder) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
+}
+
+TEST_P(TestMultiplexAdapter, ConstructAndDestructEncoder) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+}
+
+TEST_P(TestMultiplexAdapter, EncodeDecodeI420Frame) {
+  std::unique_ptr<VideoFrame> input_frame = CreateInputFrame(false);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(*input_frame, nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  EXPECT_EQ(kVideoCodecMultiplex, codec_specific_info.codecType);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, -1));
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+  EXPECT_GT(I420PSNR(input_frame.get(), decoded_frame.get()), 36);
+  CheckData(decoded_frame->video_frame_buffer());
+}
+
+TEST_P(TestMultiplexAdapter, EncodeDecodeI420AFrame) {
+  std::unique_ptr<VideoFrame> yuva_frame = CreateInputFrame(true);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(*yuva_frame, nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  EXPECT_EQ(kVideoCodecMultiplex, codec_specific_info.codecType);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, 0));
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+  EXPECT_GT(I420PSNR(yuva_frame.get(), decoded_frame.get()), 36);
+
+  // Find PSNR for AXX bits.
+  std::unique_ptr<VideoFrame> input_axx_frame = ExtractAXXFrame(*yuva_frame);
+  std::unique_ptr<VideoFrame> output_axx_frame =
+      ExtractAXXFrame(*decoded_frame);
+  EXPECT_GT(I420PSNR(input_axx_frame.get(), output_axx_frame.get()), 47);
+
+  CheckData(decoded_frame->video_frame_buffer());
+}
+
+TEST_P(TestMultiplexAdapter, CheckSingleFrameEncodedBitstream) {
+  std::unique_ptr<VideoFrame> input_frame = CreateInputFrame(false);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(*input_frame, nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  EXPECT_EQ(kVideoCodecMultiplex, codec_specific_info.codecType);
+  EXPECT_FALSE(encoded_frame.SpatialIndex());
+
+  const MultiplexImage& unpacked_frame =
+      MultiplexEncodedImagePacker::Unpack(encoded_frame);
+  EXPECT_EQ(0, unpacked_frame.image_index);
+  EXPECT_EQ(1, unpacked_frame.component_count);
+  const MultiplexImageComponent& component = unpacked_frame.image_components[0];
+  EXPECT_EQ(0, component.component_index);
+  EXPECT_NE(nullptr, component.encoded_image.data());
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, component.encoded_image._frameType);
+}
+
+TEST_P(TestMultiplexAdapter, CheckDoubleFramesEncodedBitstream) {
+  std::unique_ptr<VideoFrame> yuva_frame = CreateInputFrame(true);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(*yuva_frame, nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  EXPECT_EQ(kVideoCodecMultiplex, codec_specific_info.codecType);
+  EXPECT_FALSE(encoded_frame.SpatialIndex());
+
+  const MultiplexImage& unpacked_frame =
+      MultiplexEncodedImagePacker::Unpack(encoded_frame);
+  EXPECT_EQ(0, unpacked_frame.image_index);
+  EXPECT_EQ(2, unpacked_frame.component_count);
+  EXPECT_EQ(unpacked_frame.image_components.size(),
+            unpacked_frame.component_count);
+  for (int i = 0; i < unpacked_frame.component_count; ++i) {
+    const MultiplexImageComponent& component =
+        unpacked_frame.image_components[i];
+    EXPECT_EQ(i, component.component_index);
+    EXPECT_NE(nullptr, component.encoded_image.data());
+    EXPECT_EQ(VideoFrameType::kVideoFrameKey,
+              component.encoded_image._frameType);
+  }
+}
+
+TEST_P(TestMultiplexAdapter, ImageIndexIncreases) {
+  std::unique_ptr<VideoFrame> yuva_frame = CreateInputFrame(true);
+  const size_t expected_num_encoded_frames = 3;
+  for (size_t i = 0; i < expected_num_encoded_frames; ++i) {
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(*yuva_frame, nullptr));
+    EncodedImage encoded_frame;
+    CodecSpecificInfo codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+    const MultiplexImage& unpacked_frame =
+        MultiplexEncodedImagePacker::Unpack(encoded_frame);
+    EXPECT_EQ(i, unpacked_frame.image_index);
+    EXPECT_EQ(
+        i ? VideoFrameType::kVideoFrameDelta : VideoFrameType::kVideoFrameKey,
+        encoded_frame._frameType);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(TestMultiplexAdapter,
+                         TestMultiplexAdapter,
+                         ::testing::Bool());
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/android_codec_factory_helper.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/android_codec_factory_helper.cc
new file mode 100644
index 0000000000..d1be684cbb
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/android_codec_factory_helper.cc
@@ -0,0 +1,78 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/android_codec_factory_helper.h"
+
+#include <jni.h>
+#include <pthread.h>
+#include <stddef.h>
+
+#include <memory>
+
+#include "modules/utility/include/jvm_android.h"
+#include "rtc_base/checks.h"
+#include "sdk/android/native_api/codecs/wrapper.h"
+#include "sdk/android/native_api/jni/class_loader.h"
+#include "sdk/android/native_api/jni/jvm.h"
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
+#include "sdk/android/src/jni/jvm.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
+
+void EnsureInitializedOnce() {
+  RTC_CHECK(::webrtc::jni::GetJVM() != nullptr);
+
+  JNIEnv* jni = ::webrtc::jni::AttachCurrentThreadIfNeeded();
+  JavaVM* jvm = NULL;
+  RTC_CHECK_EQ(0, jni->GetJavaVM(&jvm));
+
+  // Initialize the Java environment (currently only used by the audio manager).
+  webrtc::JVM::Initialize(jvm);
+}
+
+}  // namespace
+
+void InitializeAndroidObjects() {
+  RTC_CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
+}
+
+std::unique_ptr<VideoEncoderFactory> CreateAndroidEncoderFactory() {
+  JNIEnv* env = AttachCurrentThreadIfNeeded();
+  ScopedJavaLocalRef<jclass> factory_class =
+      GetClass(env, "org/webrtc/HardwareVideoEncoderFactory");
+  jmethodID factory_constructor = env->GetMethodID(
+      factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;ZZ)V");
+  ScopedJavaLocalRef<jobject> factory_object(
+      env, env->NewObject(factory_class.obj(), factory_constructor,
+                          nullptr /* shared_context */,
+                          false /* enable_intel_vp8_encoder */,
+                          true /* enable_h264_high_profile */));
+  return JavaToNativeVideoEncoderFactory(env, factory_object.obj());
+}
+
+std::unique_ptr<VideoDecoderFactory> CreateAndroidDecoderFactory() {
+  JNIEnv* env = AttachCurrentThreadIfNeeded();
+  ScopedJavaLocalRef<jclass> factory_class =
+      GetClass(env, "org/webrtc/HardwareVideoDecoderFactory");
+  jmethodID factory_constructor = env->GetMethodID(
+      factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;)V");
+  ScopedJavaLocalRef<jobject> factory_object(
+      env, env->NewObject(factory_class.obj(), factory_constructor,
+                          nullptr /* shared_context */));
+  return JavaToNativeVideoDecoderFactory(env, factory_object.obj());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/android_codec_factory_helper.h b/third_party/libwebrtc/modules/video_coding/codecs/test/android_codec_factory_helper.h
new file mode 100644
index 0000000000..ad9cf35162
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/android_codec_factory_helper.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_CODEC_FACTORY_HELPER_H_
+#define MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_CODEC_FACTORY_HELPER_H_
+
+#include <memory>
+
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+
+namespace webrtc {
+namespace test {
+
+void InitializeAndroidObjects();
+
+std::unique_ptr<VideoEncoderFactory> CreateAndroidEncoderFactory();
+std::unique_ptr<VideoDecoderFactory> CreateAndroidDecoderFactory();
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_CODEC_FACTORY_HELPER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/batch/empty-runtime-deps b/third_party/libwebrtc/modules/video_coding/codecs/test/batch/empty-runtime-deps
new file mode 100644
index 0000000000..6702195ca9
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/batch/empty-runtime-deps
@@ -0,0 +1 @@
+does-not-exist
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/batch/run-instantiation-tests.sh b/third_party/libwebrtc/modules/video_coding/codecs/test/batch/run-instantiation-tests.sh
new file mode 100755
index 0000000000..28083b1808
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/batch/run-instantiation-tests.sh
@@ -0,0 +1,56 @@
+#!/bin/bash
+
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+if [ $# -ne 1 ]; then
+    echo "Usage: run-instantiation-tests.sh ADB-DEVICE-ID"
+    exit 1
+fi
+
+# Paths: update these based on your git checkout and gn output folder names.
+WEBRTC_DIR=$HOME/src/webrtc/src
+BUILD_DIR=$WEBRTC_DIR/out/Android_Release
+
+# Other settings.
+ADB=`which adb`
+SERIAL=$1
+TIMEOUT=7200
+
+# Ensure we are using the latest version.
+ninja -C $BUILD_DIR modules_tests
+
+# Transfer the required files by trying to run a test that doesn't exist.
+echo "===> Transferring required resources to device $1."
+$WEBRTC_DIR/build/android/test_runner.py gtest \
+  --output-directory $BUILD_DIR \
+  --suite modules_tests \
+  --gtest_filter "DoesNotExist" \
+  --shard-timeout $TIMEOUT \
+  --runtime-deps-path $BUILD_DIR/gen.runtime/modules/modules_tests__test_runner_script.runtime_deps \
+  --adb-path $ADB \
+  --device $SERIAL \
+  --verbose
+
+# Run all tests as separate test invocations.
+mkdir $SERIAL
+pushd $SERIAL
+$WEBRTC_DIR/build/android/test_runner.py gtest \
+  --output-directory $BUILD_DIR \
+  --suite modules_tests \
+  --gtest_filter "*InstantiationTest*" \
+  --gtest_also_run_disabled_tests \
+  --shard-timeout $TIMEOUT \
+  --runtime-deps-path ../empty-runtime-deps \
+  --test-launcher-retry-limit 0 \
+  --adb-path $ADB \
+  --device $SERIAL \
+  --verbose \
+  --num-retries 0 \
+  2>&1 | tee -a instantiation-tests.log
+popd
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/batch/run-videoprocessor-tests.sh b/third_party/libwebrtc/modules/video_coding/codecs/test/batch/run-videoprocessor-tests.sh
new file mode 100755
index 0000000000..25c971ba61
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/batch/run-videoprocessor-tests.sh
@@ -0,0 +1,70 @@
+#!/bin/bash
+
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+if [ $# -ne 1 ]; then
+    echo "Usage: run.sh ADB-DEVICE-ID"
+    exit 1
+fi
+
+# Paths: update these based on your git checkout and gn output folder names.
+WEBRTC_DIR=$HOME/src/webrtc/src
+BUILD_DIR=$WEBRTC_DIR/out/Android_Release
+
+# Clips: update these to encode/decode other content.
+CLIPS=('Foreman')
+RESOLUTIONS=('128x96' '160x120' '176x144' '320x240' '352x288')
+FRAMERATES=(30)
+
+# Other settings.
+ADB=`which adb`
+SERIAL=$1
+TIMEOUT=7200
+
+# Ensure we are using the latest version.
+ninja -C $BUILD_DIR modules_tests
+
+# Transfer the required files by trying to run a test that doesn't exist.
+echo "===> Transferring required resources to device $1."
+$WEBRTC_DIR/build/android/test_runner.py gtest \
+  --output-directory $BUILD_DIR \
+  --suite modules_tests \
+  --gtest_filter "DoesNotExist" \
+  --shard-timeout $TIMEOUT \
+  --runtime-deps-path $BUILD_DIR/gen.runtime/modules/modules_tests__test_runner_script.runtime_deps \
+  --adb-path $ADB \
+  --device $SERIAL \
+  --verbose
+
+# Run all tests as separate test invocations.
+mkdir $SERIAL
+pushd $SERIAL
+for clip in "${CLIPS[@]}"; do
+  for resolution in "${RESOLUTIONS[@]}"; do
+    for framerate in "${FRAMERATES[@]}"; do
+      test_name="${clip}_${resolution}_${framerate}"
+      log_name="${test_name}.log"
+
+      echo "===> Running ${test_name} on device $1."
+
+      $WEBRTC_DIR/build/android/test_runner.py gtest \
+        --output-directory $BUILD_DIR \
+        --suite modules_tests \
+        --gtest_filter "CodecSettings/*${test_name}*" \
+        --shard-timeout $TIMEOUT \
+        --runtime-deps-path ../empty-runtime-deps \
+        --test-launcher-retry-limit 0 \
+        --adb-path $ADB \
+        --device $SERIAL \
+        --verbose \
+        2>&1 | tee -a ${log_name}
+    done
+  done
+done
+popd
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/encoded_video_frame_producer.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/encoded_video_frame_producer.cc
new file mode 100644
index 0000000000..899826eee4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/encoded_video_frame_producer.cc
@@ -0,0 +1,77 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/encoded_video_frame_producer.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/test/create_frame_generator.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_type.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+class EncoderCallback : public EncodedImageCallback {
+ public:
+  explicit EncoderCallback(
+      std::vector<EncodedVideoFrameProducer::EncodedFrame>& output_frames)
+      : output_frames_(output_frames) {}
+
+ private:
+  Result OnEncodedImage(const EncodedImage& encoded_image,
+                        const CodecSpecificInfo* codec_specific_info) override {
+    output_frames_.push_back({encoded_image, *codec_specific_info});
+    return Result(Result::Error::OK);
+  }
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame>& output_frames_;
+};
+
+}  // namespace
+
+std::vector<EncodedVideoFrameProducer::EncodedFrame>
+EncodedVideoFrameProducer::Encode() {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_buffer_generator =
+      test::CreateSquareFrameGenerator(
+          resolution_.Width(), resolution_.Height(),
+          test::FrameGeneratorInterface::OutputType::kI420, absl::nullopt);
+
+  std::vector<EncodedFrame> encoded_frames;
+  EncoderCallback encoder_callback(encoded_frames);
+  RTC_CHECK_EQ(encoder_.RegisterEncodeCompleteCallback(&encoder_callback),
+               WEBRTC_VIDEO_CODEC_OK);
+
+  uint32_t rtp_tick = 90000 / framerate_fps_;
+  for (int i = 0; i < num_input_frames_; ++i) {
+    VideoFrame frame =
+        VideoFrame::Builder()
+            .set_video_frame_buffer(frame_buffer_generator->NextFrame().buffer)
+            .set_timestamp_rtp(rtp_timestamp_)
+            .build();
+    rtp_timestamp_ += rtp_tick;
+    RTC_CHECK_EQ(encoder_.Encode(frame, &next_frame_type_),
+                 WEBRTC_VIDEO_CODEC_OK);
+    next_frame_type_[0] = VideoFrameType::kVideoFrameDelta;
+  }
+
+  RTC_CHECK_EQ(encoder_.RegisterEncodeCompleteCallback(nullptr),
+               WEBRTC_VIDEO_CODEC_OK);
+  return encoded_frames;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/encoded_video_frame_producer.h b/third_party/libwebrtc/modules/video_coding/codecs/test/encoded_video_frame_producer.h
new file mode 100644
index 0000000000..04f4a64950
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/encoded_video_frame_producer.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_ENCODED_VIDEO_FRAME_PRODUCER_H_
+#define MODULES_VIDEO_CODING_CODECS_TEST_ENCODED_VIDEO_FRAME_PRODUCER_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+
+// Wrapper around VideoEncoder::Encode for convenient input (generates frames)
+// and output (returns encoded frames instead of passing them to callback)
+class EncodedVideoFrameProducer {
+ public:
+  struct EncodedFrame {
+    EncodedImage encoded_image;
+    CodecSpecificInfo codec_specific_info;
+  };
+
+  // `encoder` should be initialized, but shouldn't have `EncoderCallback` set.
+  explicit EncodedVideoFrameProducer(VideoEncoder& encoder)
+      : encoder_(encoder) {}
+  EncodedVideoFrameProducer(const EncodedVideoFrameProducer&) = delete;
+  EncodedVideoFrameProducer& operator=(const EncodedVideoFrameProducer&) =
+      delete;
+
+  // Number of the input frames to pass to the encoder.
+  EncodedVideoFrameProducer& SetNumInputFrames(int value);
+  // Encode next frame as key frame.
+  EncodedVideoFrameProducer& ForceKeyFrame();
+  // Resolution of the input frames.
+  EncodedVideoFrameProducer& SetResolution(RenderResolution value);
+
+  EncodedVideoFrameProducer& SetFramerateFps(int value);
+
+  EncodedVideoFrameProducer& SetRtpTimestamp(uint32_t value);
+
+  // Generates input video frames and encodes them with `encoder` provided in
+  // the constructor. Returns frame passed to the `OnEncodedImage` by wraping
+  // `EncodedImageCallback` underneath.
+  std::vector<EncodedFrame> Encode();
+
+ private:
+  VideoEncoder& encoder_;
+
+  uint32_t rtp_timestamp_ = 1000;
+  int num_input_frames_ = 1;
+  int framerate_fps_ = 30;
+  RenderResolution resolution_ = {320, 180};
+  std::vector<VideoFrameType> next_frame_type_ = {
+      VideoFrameType::kVideoFrameKey};
+};
+
+inline EncodedVideoFrameProducer& EncodedVideoFrameProducer::SetNumInputFrames(
+    int value) {
+  RTC_DCHECK_GT(value, 0);
+  num_input_frames_ = value;
+  return *this;
+}
+
+inline EncodedVideoFrameProducer& EncodedVideoFrameProducer::ForceKeyFrame() {
+  next_frame_type_ = {VideoFrameType::kVideoFrameKey};
+  return *this;
+}
+
+inline EncodedVideoFrameProducer& EncodedVideoFrameProducer::SetResolution(
+    RenderResolution value) {
+  resolution_ = value;
+  return *this;
+}
+
+inline EncodedVideoFrameProducer& EncodedVideoFrameProducer::SetFramerateFps(
+    int value) {
+  RTC_DCHECK_GT(value, 0);
+  framerate_fps_ = value;
+  return *this;
+}
+
+inline EncodedVideoFrameProducer& EncodedVideoFrameProducer::SetRtpTimestamp(
+    uint32_t value) {
+  rtp_timestamp_ = value;
+  return *this;
+}
+
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_ENCODED_VIDEO_FRAME_PRODUCER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.h b/third_party/libwebrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.h
new file mode 100644
index 0000000000..475d0fdd08
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_OBJC_CODEC_FACTORY_HELPER_H_
+#define MODULES_VIDEO_CODING_CODECS_TEST_OBJC_CODEC_FACTORY_HELPER_H_
+
+#include <memory>
+
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+
+namespace webrtc {
+namespace test {
+
+std::unique_ptr<VideoEncoderFactory> CreateObjCEncoderFactory();
+std::unique_ptr<VideoDecoderFactory> CreateObjCDecoderFactory();
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_OBJC_CODEC_FACTORY_HELPER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.mm b/third_party/libwebrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.mm
new file mode 100644
index 0000000000..ed82376251
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.mm
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/objc_codec_factory_helper.h"
+
+#import "sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h"
+#import "sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h"
+#include "sdk/objc/native/api/video_decoder_factory.h"
+#include "sdk/objc/native/api/video_encoder_factory.h"
+
+namespace webrtc {
+namespace test {
+
+std::unique_ptr<VideoEncoderFactory> CreateObjCEncoderFactory() {
+  return ObjCToNativeVideoEncoderFactory([[RTC_OBJC_TYPE(RTCVideoEncoderFactoryH264) alloc] init]);
+}
+
+std::unique_ptr<VideoDecoderFactory> CreateObjCDecoderFactory() {
+  return ObjCToNativeVideoDecoderFactory([[RTC_OBJC_TYPE(RTCVideoDecoderFactoryH264) alloc] init]);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/plot_webrtc_test_logs.py b/third_party/libwebrtc/modules/video_coding/codecs/test/plot_webrtc_test_logs.py
new file mode 100755
index 0000000000..29e2d6f65a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/plot_webrtc_test_logs.py
@@ -0,0 +1,438 @@
+#  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+#  Use of this source code is governed by a BSD-style license
+#  that can be found in the LICENSE file in the root of the source
+#  tree. An additional intellectual property rights grant can be found
+#  in the file PATENTS.  All contributing project authors may
+#  be found in the AUTHORS file in the root of the source tree.
+"""Plots statistics from WebRTC integration test logs.
+
+Usage: $ python plot_webrtc_test_logs.py filename.txt
+"""
+
+import numpy
+import sys
+import re
+
+import matplotlib.pyplot as plt
+
+# Log events.
+EVENT_START = 'RUN      ] CodecSettings/VideoCodecTestParameterized.'
+EVENT_END = 'OK ] CodecSettings/VideoCodecTestParameterized.'
+
+# Metrics to plot, tuple: (name to parse in file, label to use when plotting).
+WIDTH = ('width', 'width')
+HEIGHT = ('height', 'height')
+FILENAME = ('filename', 'clip')
+CODEC_TYPE = ('codec_type', 'Codec')
+ENCODER_IMPLEMENTATION_NAME = ('enc_impl_name', 'enc name')
+DECODER_IMPLEMENTATION_NAME = ('dec_impl_name', 'dec name')
+CODEC_IMPLEMENTATION_NAME = ('codec_impl_name', 'codec name')
+CORES = ('num_cores', 'CPU cores used')
+DENOISING = ('denoising', 'denoising')
+RESILIENCE = ('resilience', 'resilience')
+ERROR_CONCEALMENT = ('error_concealment', 'error concealment')
+CPU_USAGE = ('cpu_usage_percent', 'CPU usage (%)')
+BITRATE = ('target_bitrate_kbps', 'target bitrate (kbps)')
+FRAMERATE = ('input_framerate_fps', 'fps')
+QP = ('avg_qp', 'QP avg')
+PSNR = ('avg_psnr', 'PSNR (dB)')
+SSIM = ('avg_ssim', 'SSIM')
+ENC_BITRATE = ('bitrate_kbps', 'encoded bitrate (kbps)')
+NUM_FRAMES = ('num_input_frames', 'num frames')
+NUM_DROPPED_FRAMES = ('num_dropped_frames', 'num dropped frames')
+TIME_TO_TARGET = ('time_to_reach_target_bitrate_sec',
+                  'time to reach target rate (sec)')
+ENCODE_SPEED_FPS = ('enc_speed_fps', 'encode speed (fps)')
+DECODE_SPEED_FPS = ('dec_speed_fps', 'decode speed (fps)')
+AVG_KEY_FRAME_SIZE = ('avg_key_frame_size_bytes', 'avg key frame size (bytes)')
+AVG_DELTA_FRAME_SIZE = ('avg_delta_frame_size_bytes',
+                        'avg delta frame size (bytes)')
+
+# Settings.
+SETTINGS = [
+    WIDTH,
+    HEIGHT,
+    FILENAME,
+    NUM_FRAMES,
+]
+
+# Settings, options for x-axis.
+X_SETTINGS = [
+    CORES,
+    FRAMERATE,
+    DENOISING,
+    RESILIENCE,
+    ERROR_CONCEALMENT,
+    BITRATE,  # TODO(asapersson): Needs to be last.
+]
+
+# Settings, options for subplots.
+SUBPLOT_SETTINGS = [
+    CODEC_TYPE,
+    ENCODER_IMPLEMENTATION_NAME,
+    DECODER_IMPLEMENTATION_NAME,
+    CODEC_IMPLEMENTATION_NAME,
+] + X_SETTINGS
+
+# Results.
+RESULTS = [
+    PSNR,
+    SSIM,
+    ENC_BITRATE,
+    NUM_DROPPED_FRAMES,
+    TIME_TO_TARGET,
+    ENCODE_SPEED_FPS,
+    DECODE_SPEED_FPS,
+    QP,
+    CPU_USAGE,
+    AVG_KEY_FRAME_SIZE,
+    AVG_DELTA_FRAME_SIZE,
+]
+
+METRICS_TO_PARSE = SETTINGS + SUBPLOT_SETTINGS + RESULTS
+
+Y_METRICS = [res[1] for res in RESULTS]
+
+# Parameters for plotting.
+FIG_SIZE_SCALE_FACTOR_X = 1.6
+FIG_SIZE_SCALE_FACTOR_Y = 1.8
+GRID_COLOR = [0.45, 0.45, 0.45]
+
+
+def ParseSetting(filename, setting):
+    """Parses setting from file.
+
+  Args:
+    filename: The name of the file.
+    setting: Name of setting to parse (e.g. width).
+
+  Returns:
+    A list holding parsed settings, e.g. ['width: 128.0', 'width: 160.0'] """
+
+    settings = []
+
+    settings_file = open(filename)
+    while True:
+        line = settings_file.readline()
+        if not line:
+            break
+        if re.search(r'%s' % EVENT_START, line):
+            # Parse event.
+            parsed = {}
+            while True:
+                line = settings_file.readline()
+                if not line:
+                    break
+                if re.search(r'%s' % EVENT_END, line):
+                    # Add parsed setting to list.
+                    if setting in parsed:
+                        s = setting + ': ' + str(parsed[setting])
+                        if s not in settings:
+                            settings.append(s)
+                    break
+
+                TryFindMetric(parsed, line)
+
+    settings_file.close()
+    return settings
+
+
+def ParseMetrics(filename, setting1, setting2):
+    """Parses metrics from file.
+
+  Args:
+    filename: The name of the file.
+    setting1: First setting for sorting metrics (e.g. width).
+    setting2: Second setting for sorting metrics (e.g. CPU cores used).
+
+  Returns:
+    A dictionary holding parsed metrics.
+
+  For example:
+    metrics[key1][key2][measurement]
+
+  metrics = {
+  "width: 352": {
+    "CPU cores used: 1.0": {
+      "encode time (us)": [0.718005, 0.806925, 0.909726, 0.931835, 0.953642],
+      "PSNR (dB)": [25.546029, 29.465518, 34.723535, 36.428493, 38.686551],
+      "bitrate (kbps)": [50, 100, 300, 500, 1000]
+    },
+    "CPU cores used: 2.0": {
+      "encode time (us)": [0.718005, 0.806925, 0.909726, 0.931835, 0.953642],
+      "PSNR (dB)": [25.546029, 29.465518, 34.723535, 36.428493, 38.686551],
+      "bitrate (kbps)": [50, 100, 300, 500, 1000]
+    },
+  },
+  "width: 176": {
+    "CPU cores used: 1.0": {
+      "encode time (us)": [0.857897, 0.91608, 0.959173, 0.971116, 0.980961],
+      "PSNR (dB)": [30.243646, 33.375592, 37.574387, 39.42184, 41.437897],
+      "bitrate (kbps)": [50, 100, 300, 500, 1000]
+    },
+  }
+  } """
+
+    metrics = {}
+
+    # Parse events.
+    settings_file = open(filename)
+    while True:
+        line = settings_file.readline()
+        if not line:
+            break
+        if re.search(r'%s' % EVENT_START, line):
+            # Parse event.
+            parsed = {}
+            while True:
+                line = settings_file.readline()
+                if not line:
+                    break
+                if re.search(r'%s' % EVENT_END, line):
+                    # Add parsed values to metrics.
+                    key1 = setting1 + ': ' + str(parsed[setting1])
+                    key2 = setting2 + ': ' + str(parsed[setting2])
+                    if key1 not in metrics:
+                        metrics[key1] = {}
+                    if key2 not in metrics[key1]:
+                        metrics[key1][key2] = {}
+
+                    for label in parsed:
+                        if label not in metrics[key1][key2]:
+                            metrics[key1][key2][label] = []
+                        metrics[key1][key2][label].append(parsed[label])
+
+                    break
+
+                TryFindMetric(parsed, line)
+
+    settings_file.close()
+    return metrics
+
+
+def TryFindMetric(parsed, line):
+    for metric in METRICS_TO_PARSE:
+        name = metric[0]
+        label = metric[1]
+        if re.search(r'%s' % name, line):
+            found, value = GetMetric(name, line)
+            if found:
+                parsed[label] = value
+            return
+
+
+def GetMetric(name, string):
+    # Float (e.g. bitrate = 98.8253).
+    pattern = r'%s\s*[:=]\s*([+-]?\d+\.*\d*)' % name
+    m = re.search(r'%s' % pattern, string)
+    if m is not None:
+        return StringToFloat(m.group(1))
+
+    # Alphanumeric characters (e.g. codec type : VP8).
+    pattern = r'%s\s*[:=]\s*(\w+)' % name
+    m = re.search(r'%s' % pattern, string)
+    if m is not None:
+        return True, m.group(1)
+
+    return False, -1
+
+
+def StringToFloat(value):
+    try:
+        value = float(value)
+    except ValueError:
+        print "Not a float, skipped %s" % value
+        return False, -1
+
+    return True, value
+
+
+def Plot(y_metric, x_metric, metrics):
+    """Plots y_metric vs x_metric per key in metrics.
+
+  For example:
+    y_metric = 'PSNR (dB)'
+    x_metric = 'bitrate (kbps)'
+    metrics = {
+      "CPU cores used: 1.0": {
+        "PSNR (dB)": [25.546029, 29.465518, 34.723535, 36.428493, 38.686551],
+        "bitrate (kbps)": [50, 100, 300, 500, 1000]
+      },
+      "CPU cores used: 2.0": {
+        "PSNR (dB)": [25.546029, 29.465518, 34.723535, 36.428493, 38.686551],
+        "bitrate (kbps)": [50, 100, 300, 500, 1000]
+      },
+    }
+    """
+    for key in sorted(metrics):
+        data = metrics[key]
+        if y_metric not in data:
+            print "Failed to find metric: %s" % y_metric
+            continue
+
+        y = numpy.array(data[y_metric])
+        x = numpy.array(data[x_metric])
+        if len(y) != len(x):
+            print "Length mismatch for %s, %s" % (y, x)
+            continue
+
+        label = y_metric + ' - ' + str(key)
+
+        plt.plot(x,
+                 y,
+                 label=label,
+                 linewidth=1.5,
+                 marker='o',
+                 markersize=5,
+                 markeredgewidth=0.0)
+
+
+def PlotFigure(settings, y_metrics, x_metric, metrics, title):
+    """Plots metrics in y_metrics list. One figure is plotted and each entry
+  in the list is plotted in a subplot (and sorted per settings).
+
+  For example:
+    settings = ['width: 128.0', 'width: 160.0']. Sort subplot per setting.
+    y_metrics = ['PSNR (dB)', 'PSNR (dB)']. Metric to plot per subplot.
+    x_metric = 'bitrate (kbps)'
+
+  """
+
+    plt.figure()
+    plt.suptitle(title, fontsize='large', fontweight='bold')
+    settings.sort()
+    rows = len(settings)
+    cols = 1
+    pos = 1
+    while pos <= rows:
+        plt.rc('grid', color=GRID_COLOR)
+        ax = plt.subplot(rows, cols, pos)
+        plt.grid()
+        plt.setp(ax.get_xticklabels(), visible=(pos == rows), fontsize='large')
+        plt.setp(ax.get_yticklabels(), fontsize='large')
+        setting = settings[pos - 1]
+        Plot(y_metrics[pos - 1], x_metric, metrics[setting])
+        if setting.startswith(WIDTH[1]):
+            plt.title(setting, fontsize='medium')
+        plt.legend(fontsize='large', loc='best')
+        pos += 1
+
+    plt.xlabel(x_metric, fontsize='large')
+    plt.subplots_adjust(left=0.06,
+                        right=0.98,
+                        bottom=0.05,
+                        top=0.94,
+                        hspace=0.08)
+
+
+def GetTitle(filename, setting):
+    title = ''
+    if setting != CODEC_IMPLEMENTATION_NAME[1] and setting != CODEC_TYPE[1]:
+        codec_types = ParseSetting(filename, CODEC_TYPE[1])
+        for i in range(0, len(codec_types)):
+            title += codec_types[i] + ', '
+
+    if setting != CORES[1]:
+        cores = ParseSetting(filename, CORES[1])
+        for i in range(0, len(cores)):
+            title += cores[i].split('.')[0] + ', '
+
+    if setting != FRAMERATE[1]:
+        framerate = ParseSetting(filename, FRAMERATE[1])
+        for i in range(0, len(framerate)):
+            title += framerate[i].split('.')[0] + ', '
+
+    if (setting != CODEC_IMPLEMENTATION_NAME[1]
+            and setting != ENCODER_IMPLEMENTATION_NAME[1]):
+        enc_names = ParseSetting(filename, ENCODER_IMPLEMENTATION_NAME[1])
+        for i in range(0, len(enc_names)):
+            title += enc_names[i] + ', '
+
+    if (setting != CODEC_IMPLEMENTATION_NAME[1]
+            and setting != DECODER_IMPLEMENTATION_NAME[1]):
+        dec_names = ParseSetting(filename, DECODER_IMPLEMENTATION_NAME[1])
+        for i in range(0, len(dec_names)):
+            title += dec_names[i] + ', '
+
+    filenames = ParseSetting(filename, FILENAME[1])
+    title += filenames[0].split('_')[0]
+
+    num_frames = ParseSetting(filename, NUM_FRAMES[1])
+    for i in range(0, len(num_frames)):
+        title += ' (' + num_frames[i].split('.')[0] + ')'
+
+    return title
+
+
+def ToString(input_list):
+    return ToStringWithoutMetric(input_list, ('', ''))
+
+
+def ToStringWithoutMetric(input_list, metric):
+    i = 1
+    output_str = ""
+    for m in input_list:
+        if m != metric:
+            output_str = output_str + ("%s. %s\n" % (i, m[1]))
+            i += 1
+    return output_str
+
+
+def GetIdx(text_list):
+    return int(raw_input(text_list)) - 1
+
+
+def main():
+    filename = sys.argv[1]
+
+    # Setup.
+    idx_metric = GetIdx("Choose metric:\n0. All\n%s" % ToString(RESULTS))
+    if idx_metric == -1:
+        # Plot all metrics. One subplot for each metric.
+        # Per subplot: metric vs bitrate (per resolution).
+        cores = ParseSetting(filename, CORES[1])
+        setting1 = CORES[1]
+        setting2 = WIDTH[1]
+        sub_keys = [cores[0]] * len(Y_METRICS)
+        y_metrics = Y_METRICS
+        x_metric = BITRATE[1]
+    else:
+        resolutions = ParseSetting(filename, WIDTH[1])
+        idx = GetIdx("Select metric for x-axis:\n%s" % ToString(X_SETTINGS))
+        if X_SETTINGS[idx] == BITRATE:
+            idx = GetIdx("Plot per:\n%s" %
+                         ToStringWithoutMetric(SUBPLOT_SETTINGS, BITRATE))
+            idx_setting = METRICS_TO_PARSE.index(SUBPLOT_SETTINGS[idx])
+            # Plot one metric. One subplot for each resolution.
+            # Per subplot: metric vs bitrate (per setting).
+            setting1 = WIDTH[1]
+            setting2 = METRICS_TO_PARSE[idx_setting][1]
+            sub_keys = resolutions
+            y_metrics = [RESULTS[idx_metric][1]] * len(sub_keys)
+            x_metric = BITRATE[1]
+        else:
+            # Plot one metric. One subplot for each resolution.
+            # Per subplot: metric vs setting (per bitrate).
+            setting1 = WIDTH[1]
+            setting2 = BITRATE[1]
+            sub_keys = resolutions
+            y_metrics = [RESULTS[idx_metric][1]] * len(sub_keys)
+            x_metric = X_SETTINGS[idx][1]
+
+    metrics = ParseMetrics(filename, setting1, setting2)
+
+    # Stretch fig size.
+    figsize = plt.rcParams["figure.figsize"]
+    figsize[0] *= FIG_SIZE_SCALE_FACTOR_X
+    figsize[1] *= FIG_SIZE_SCALE_FACTOR_Y
+    plt.rcParams["figure.figsize"] = figsize
+
+    PlotFigure(sub_keys, y_metrics, x_metric, metrics,
+               GetTitle(filename, setting2))
+
+    plt.show()
+
+
+if __name__ == '__main__':
+    main()
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.cc
new file mode 100644
index 0000000000..50af417bcf
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.cc
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/video_codec_analyzer.h"
+
+#include <memory>
+
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/test/video_codec_tester.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/time_utils.h"
+#include "third_party/libyuv/include/libyuv/compare.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+struct Psnr {
+  double y;
+  double u;
+  double v;
+  double yuv;
+};
+
+Psnr CalcPsnr(const I420BufferInterface& ref_buffer,
+              const I420BufferInterface& dec_buffer) {
+  RTC_CHECK_EQ(ref_buffer.width(), dec_buffer.width());
+  RTC_CHECK_EQ(ref_buffer.height(), dec_buffer.height());
+
+  uint64_t sse_y = libyuv::ComputeSumSquareErrorPlane(
+      dec_buffer.DataY(), dec_buffer.StrideY(), ref_buffer.DataY(),
+      ref_buffer.StrideY(), dec_buffer.width(), dec_buffer.height());
+
+  uint64_t sse_u = libyuv::ComputeSumSquareErrorPlane(
+      dec_buffer.DataU(), dec_buffer.StrideU(), ref_buffer.DataU(),
+      ref_buffer.StrideU(), dec_buffer.width() / 2, dec_buffer.height() / 2);
+
+  uint64_t sse_v = libyuv::ComputeSumSquareErrorPlane(
+      dec_buffer.DataV(), dec_buffer.StrideV(), ref_buffer.DataV(),
+      ref_buffer.StrideV(), dec_buffer.width() / 2, dec_buffer.height() / 2);
+
+  int num_y_samples = dec_buffer.width() * dec_buffer.height();
+  Psnr psnr;
+  psnr.y = libyuv::SumSquareErrorToPsnr(sse_y, num_y_samples);
+  psnr.u = libyuv::SumSquareErrorToPsnr(sse_u, num_y_samples / 4);
+  psnr.v = libyuv::SumSquareErrorToPsnr(sse_v, num_y_samples / 4);
+  psnr.yuv = libyuv::SumSquareErrorToPsnr(sse_y + sse_u + sse_v,
+                                          num_y_samples + num_y_samples / 2);
+  return psnr;
+}
+
+}  // namespace
+
+VideoCodecAnalyzer::VideoCodecAnalyzer(
+    rtc::TaskQueue& task_queue,
+    ReferenceVideoSource* reference_video_source)
+    : task_queue_(task_queue), reference_video_source_(reference_video_source) {
+  sequence_checker_.Detach();
+}
+
+void VideoCodecAnalyzer::StartEncode(const VideoFrame& input_frame) {
+  int64_t encode_started_ns = rtc::TimeNanos();
+  task_queue_.PostTask(
+      [this, timestamp_rtp = input_frame.timestamp(), encode_started_ns]() {
+        RTC_DCHECK_RUN_ON(&sequence_checker_);
+        VideoCodecTestStats::FrameStatistics* fs =
+            stats_.GetOrAddFrame(timestamp_rtp, /*spatial_idx=*/0);
+        fs->encode_start_ns = encode_started_ns;
+      });
+}
+
+void VideoCodecAnalyzer::FinishEncode(const EncodedImage& frame) {
+  int64_t encode_finished_ns = rtc::TimeNanos();
+
+  task_queue_.PostTask([this, timestamp_rtp = frame.Timestamp(),
+                        spatial_idx = frame.SpatialIndex().value_or(0),
+                        temporal_idx = frame.TemporalIndex().value_or(0),
+                        frame_type = frame._frameType, qp = frame.qp_,
+                        frame_size_bytes = frame.size(), encode_finished_ns]() {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    VideoCodecTestStats::FrameStatistics* fs =
+        stats_.GetOrAddFrame(timestamp_rtp, spatial_idx);
+    VideoCodecTestStats::FrameStatistics* fs_base =
+        stats_.GetOrAddFrame(timestamp_rtp, 0);
+
+    fs->encode_start_ns = fs_base->encode_start_ns;
+    fs->spatial_idx = spatial_idx;
+    fs->temporal_idx = temporal_idx;
+    fs->frame_type = frame_type;
+    fs->qp = qp;
+
+    fs->encode_time_us = (encode_finished_ns - fs->encode_start_ns) /
+                         rtc::kNumNanosecsPerMicrosec;
+    fs->length_bytes = frame_size_bytes;
+
+    fs->encoding_successful = true;
+  });
+}
+
+void VideoCodecAnalyzer::StartDecode(const EncodedImage& frame) {
+  int64_t decode_start_ns = rtc::TimeNanos();
+  task_queue_.PostTask([this, timestamp_rtp = frame.Timestamp(),
+                        spatial_idx = frame.SpatialIndex().value_or(0),
+                        frame_size_bytes = frame.size(), decode_start_ns]() {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    VideoCodecTestStats::FrameStatistics* fs =
+        stats_.GetOrAddFrame(timestamp_rtp, spatial_idx);
+    if (fs->length_bytes == 0) {
+      // In encode-decode test the frame size is set in EncodeFinished. In
+      // decode-only test set it here.
+      fs->length_bytes = frame_size_bytes;
+    }
+    fs->decode_start_ns = decode_start_ns;
+  });
+}
+
+void VideoCodecAnalyzer::FinishDecode(const VideoFrame& frame,
+                                      int spatial_idx) {
+  int64_t decode_finished_ns = rtc::TimeNanos();
+  task_queue_.PostTask([this, timestamp_rtp = frame.timestamp(), spatial_idx,
+                        width = frame.width(), height = frame.height(),
+                        decode_finished_ns]() {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    VideoCodecTestStats::FrameStatistics* fs =
+        stats_.GetFrameWithTimestamp(timestamp_rtp, spatial_idx);
+    fs->decode_time_us = (decode_finished_ns - fs->decode_start_ns) /
+                         rtc::kNumNanosecsPerMicrosec;
+    fs->decoded_width = width;
+    fs->decoded_height = height;
+    fs->decoding_successful = true;
+  });
+
+  if (reference_video_source_ != nullptr) {
+    // Copy hardware-backed frame into main memory to release output buffers
+    // which number may be limited in hardware decoders.
+    rtc::scoped_refptr<I420BufferInterface> decoded_buffer =
+        frame.video_frame_buffer()->ToI420();
+
+    task_queue_.PostTask([this, decoded_buffer,
+                          timestamp_rtp = frame.timestamp(), spatial_idx]() {
+      RTC_DCHECK_RUN_ON(&sequence_checker_);
+      VideoFrame ref_frame = reference_video_source_->GetFrame(
+          timestamp_rtp, {.width = decoded_buffer->width(),
+                          .height = decoded_buffer->height()});
+      rtc::scoped_refptr<I420BufferInterface> ref_buffer =
+          ref_frame.video_frame_buffer()->ToI420();
+
+      Psnr psnr = CalcPsnr(*decoded_buffer, *ref_buffer);
+      VideoCodecTestStats::FrameStatistics* fs =
+          this->stats_.GetFrameWithTimestamp(timestamp_rtp, spatial_idx);
+      fs->psnr_y = static_cast<float>(psnr.y);
+      fs->psnr_u = static_cast<float>(psnr.u);
+      fs->psnr_v = static_cast<float>(psnr.v);
+      fs->psnr = static_cast<float>(psnr.yuv);
+
+      fs->quality_analysis_successful = true;
+    });
+  }
+}
+
+std::unique_ptr<VideoCodecTestStats> VideoCodecAnalyzer::GetStats() {
+  std::unique_ptr<VideoCodecTestStats> stats;
+  rtc::Event ready;
+  task_queue_.PostTask([this, &stats, &ready]() mutable {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    stats.reset(new VideoCodecTestStatsImpl(stats_));
+    ready.Set();
+  });
+  ready.Wait(rtc::Event::kForever);
+  return stats;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.h b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.h
new file mode 100644
index 0000000000..63a864e810
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_ANALYZER_H_
+#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_ANALYZER_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/video/encoded_image.h"
+#include "api/video/resolution.h"
+#include "api/video/video_frame.h"
+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/task_queue_for_test.h"
+
+namespace webrtc {
+namespace test {
+
+// Analyzer measures and collects metrics necessary for evaluation of video
+// codec quality and performance. This class is thread-safe.
+class VideoCodecAnalyzer {
+ public:
+  // An interface that provides reference frames for spatial quality analysis.
+  class ReferenceVideoSource {
+   public:
+    virtual ~ReferenceVideoSource() = default;
+
+    virtual VideoFrame GetFrame(uint32_t timestamp_rtp,
+                                Resolution resolution) = 0;
+  };
+
+  VideoCodecAnalyzer(rtc::TaskQueue& task_queue,
+                     ReferenceVideoSource* reference_video_source = nullptr);
+
+  void StartEncode(const VideoFrame& frame);
+
+  void FinishEncode(const EncodedImage& frame);
+
+  void StartDecode(const EncodedImage& frame);
+
+  void FinishDecode(const VideoFrame& frame, int spatial_idx);
+
+  std::unique_ptr<VideoCodecTestStats> GetStats();
+
+ protected:
+  rtc::TaskQueue& task_queue_;
+  ReferenceVideoSource* const reference_video_source_;
+  VideoCodecTestStatsImpl stats_ RTC_GUARDED_BY(sequence_checker_);
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_ANALYZER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer_unittest.cc
new file mode 100644
index 0000000000..3f9de6dac2
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_analyzer_unittest.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/video_codec_analyzer.h"
+
+#include "absl/types/optional.h"
+#include "api/video/i420_buffer.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "third_party/libyuv/include/libyuv/planar_functions.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+using ::testing::Return;
+using ::testing::Values;
+
+const size_t kTimestamp = 3000;
+const size_t kSpatialIdx = 2;
+
+class MockReferenceVideoSource
+    : public VideoCodecAnalyzer::ReferenceVideoSource {
+ public:
+  MOCK_METHOD(VideoFrame, GetFrame, (uint32_t, Resolution), (override));
+};
+
+VideoFrame CreateVideoFrame(uint32_t timestamp_rtp,
+                            uint8_t y = 0,
+                            uint8_t u = 0,
+                            uint8_t v = 0) {
+  rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(2, 2));
+
+  libyuv::I420Rect(buffer->MutableDataY(), buffer->StrideY(),
+                   buffer->MutableDataU(), buffer->StrideU(),
+                   buffer->MutableDataV(), buffer->StrideV(), 0, 0,
+                   buffer->width(), buffer->height(), y, u, v);
+
+  return VideoFrame::Builder()
+      .set_video_frame_buffer(buffer)
+      .set_timestamp_rtp(timestamp_rtp)
+      .build();
+}
+
+EncodedImage CreateEncodedImage(uint32_t timestamp_rtp, int spatial_idx = 0) {
+  EncodedImage encoded_image;
+  encoded_image.SetTimestamp(timestamp_rtp);
+  encoded_image.SetSpatialIndex(spatial_idx);
+  return encoded_image;
+}
+}  // namespace
+
+TEST(VideoCodecAnalyzerTest, EncodeStartedCreatesFrameStats) {
+  TaskQueueForTest task_queue;
+  VideoCodecAnalyzer analyzer(task_queue);
+  analyzer.StartEncode(CreateVideoFrame(kTimestamp));
+
+  auto fs = analyzer.GetStats()->GetFrameStatistics();
+  EXPECT_EQ(1u, fs.size());
+  EXPECT_EQ(fs[0].rtp_timestamp, kTimestamp);
+}
+
+TEST(VideoCodecAnalyzerTest, EncodeFinishedUpdatesFrameStats) {
+  TaskQueueForTest task_queue;
+  VideoCodecAnalyzer analyzer(task_queue);
+  analyzer.StartEncode(CreateVideoFrame(kTimestamp));
+
+  EncodedImage encoded_frame = CreateEncodedImage(kTimestamp, kSpatialIdx);
+  analyzer.FinishEncode(encoded_frame);
+
+  auto fs = analyzer.GetStats()->GetFrameStatistics();
+  EXPECT_EQ(2u, fs.size());
+  EXPECT_TRUE(fs[1].encoding_successful);
+}
+
+TEST(VideoCodecAnalyzerTest, DecodeStartedNoFrameStatsCreatesFrameStats) {
+  TaskQueueForTest task_queue;
+  VideoCodecAnalyzer analyzer(task_queue);
+  analyzer.StartDecode(CreateEncodedImage(kTimestamp, kSpatialIdx));
+
+  auto fs = analyzer.GetStats()->GetFrameStatistics();
+  EXPECT_EQ(1u, fs.size());
+  EXPECT_EQ(fs[0].rtp_timestamp, kTimestamp);
+}
+
+TEST(VideoCodecAnalyzerTest, DecodeStartedFrameStatsExistsReusesFrameStats) {
+  TaskQueueForTest task_queue;
+  VideoCodecAnalyzer analyzer(task_queue);
+  analyzer.StartEncode(CreateVideoFrame(kTimestamp));
+  analyzer.StartDecode(CreateEncodedImage(kTimestamp, /*spatial_idx=*/0));
+
+  auto fs = analyzer.GetStats()->GetFrameStatistics();
+  EXPECT_EQ(1u, fs.size());
+}
+
+TEST(VideoCodecAnalyzerTest, DecodeFinishedUpdatesFrameStats) {
+  TaskQueueForTest task_queue;
+  VideoCodecAnalyzer analyzer(task_queue);
+  analyzer.StartDecode(CreateEncodedImage(kTimestamp, kSpatialIdx));
+  VideoFrame decoded_frame = CreateVideoFrame(kTimestamp);
+  analyzer.FinishDecode(decoded_frame, kSpatialIdx);
+
+  auto fs = analyzer.GetStats()->GetFrameStatistics();
+  EXPECT_EQ(1u, fs.size());
+
+  EXPECT_TRUE(fs[0].decoding_successful);
+  EXPECT_EQ(static_cast<int>(fs[0].decoded_width), decoded_frame.width());
+  EXPECT_EQ(static_cast<int>(fs[0].decoded_height), decoded_frame.height());
+}
+
+TEST(VideoCodecAnalyzerTest, DecodeFinishedComputesPsnr) {
+  TaskQueueForTest task_queue;
+  MockReferenceVideoSource reference_video_source;
+  VideoCodecAnalyzer analyzer(task_queue, &reference_video_source);
+  analyzer.StartDecode(CreateEncodedImage(kTimestamp, kSpatialIdx));
+
+  EXPECT_CALL(reference_video_source, GetFrame)
+      .WillOnce(Return(CreateVideoFrame(kTimestamp, /*y=*/0,
+                                        /*u=*/0, /*v=*/0)));
+
+  analyzer.FinishDecode(
+      CreateVideoFrame(kTimestamp, /*value_y=*/1, /*value_u=*/2, /*value_v=*/3),
+      kSpatialIdx);
+
+  auto fs = analyzer.GetStats()->GetFrameStatistics();
+  EXPECT_EQ(1u, fs.size());
+
+  EXPECT_NEAR(fs[0].psnr_y, 48, 1);
+  EXPECT_NEAR(fs[0].psnr_u, 42, 1);
+  EXPECT_NEAR(fs[0].psnr_v, 38, 1);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc
new file mode 100644
index 0000000000..bd4c8e07f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc
@@ -0,0 +1,456 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/video_codecs/video_codec.h"
+
+#include <cstddef>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "api/test/create_video_codec_tester.h"
+#include "api/test/videocodec_test_stats.h"
+#include "api/units/data_rate.h"
+#include "api/units/frequency.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/resolution.h"
+#include "api/video_codecs/builtin_video_decoder_factory.h"
+#include "api/video_codecs/builtin_video_encoder_factory.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "media/base/media_constants.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+using ::testing::Combine;
+using ::testing::Values;
+using Layer = std::pair<int, int>;
+
+struct VideoInfo {
+  std::string name;
+  Resolution resolution;
+};
+
+struct CodecInfo {
+  std::string type;
+  std::string encoder;
+  std::string decoder;
+};
+
+struct EncodingSettings {
+  ScalabilityMode scalability_mode;
+  // Spatial layer resolution.
+  std::map<int, Resolution> resolution;
+  // Top temporal layer frame rate.
+  Frequency framerate;
+  // Bitrate of spatial and temporal layers.
+  std::map<Layer, DataRate> bitrate;
+};
+
+struct EncodingTestSettings {
+  std::string name;
+  int num_frames = 1;
+  std::map<int, EncodingSettings> frame_settings;
+};
+
+struct DecodingTestSettings {
+  std::string name;
+};
+
+struct QualityExpectations {
+  double min_apsnr_y;
+};
+
+struct EncodeDecodeTestParams {
+  CodecInfo codec;
+  VideoInfo video;
+  VideoCodecTester::EncoderSettings encoder_settings;
+  VideoCodecTester::DecoderSettings decoder_settings;
+  EncodingTestSettings encoding_settings;
+  DecodingTestSettings decoding_settings;
+  QualityExpectations quality_expectations;
+};
+
+const EncodingSettings kQvga64Kbps30Fps = {
+    .scalability_mode = ScalabilityMode::kL1T1,
+    .resolution = {{0, {.width = 320, .height = 180}}},
+    .framerate = Frequency::Hertz(30),
+    .bitrate = {{Layer(0, 0), DataRate::KilobitsPerSec(64)}}};
+
+const EncodingTestSettings kConstantRateQvga64Kbps30Fps = {
+    .name = "ConstantRateQvga64Kbps30Fps",
+    .num_frames = 300,
+    .frame_settings = {{/*frame_num=*/0, kQvga64Kbps30Fps}}};
+
+const QualityExpectations kLowQuality = {.min_apsnr_y = 30};
+
+const VideoInfo kFourPeople_1280x720_30 = {
+    .name = "FourPeople_1280x720_30",
+    .resolution = {.width = 1280, .height = 720}};
+
+const CodecInfo kLibvpxVp8 = {.type = "VP8",
+                              .encoder = "libvpx",
+                              .decoder = "libvpx"};
+
+const CodecInfo kLibvpxVp9 = {.type = "VP9",
+                              .encoder = "libvpx",
+                              .decoder = "libvpx"};
+
+const CodecInfo kOpenH264 = {.type = "H264",
+                             .encoder = "openh264",
+                             .decoder = "ffmpeg"};
+
+class TestRawVideoSource : public VideoCodecTester::RawVideoSource {
+ public:
+  static constexpr Frequency k90kHz = Frequency::Hertz(90000);
+
+  TestRawVideoSource(std::unique_ptr<FrameReader> frame_reader,
+                     const EncodingTestSettings& test_settings)
+      : frame_reader_(std::move(frame_reader)),
+        test_settings_(test_settings),
+        frame_num_(0),
+        timestamp_rtp_(0) {
+    // Ensure settings for the first frame are provided.
+    RTC_CHECK_GT(test_settings_.frame_settings.size(), 0u);
+    RTC_CHECK_EQ(test_settings_.frame_settings.begin()->first, 0);
+  }
+
+  // Pulls next frame. Frame RTP timestamp is set accordingly to
+  // `EncodingSettings::framerate`.
+  absl::optional<VideoFrame> PullFrame() override {
+    if (frame_num_ >= test_settings_.num_frames) {
+      // End of stream.
+      return absl::nullopt;
+    }
+
+    EncodingSettings frame_settings =
+        std::prev(test_settings_.frame_settings.upper_bound(frame_num_))
+            ->second;
+
+    int pulled_frame;
+    auto buffer = frame_reader_->PullFrame(
+        &pulled_frame, frame_settings.resolution.rbegin()->second,
+        {.num = 30, .den = static_cast<int>(frame_settings.framerate.hertz())});
+    RTC_CHECK(buffer) << "Cannot pull frame " << frame_num_;
+
+    auto frame = VideoFrame::Builder()
+                     .set_video_frame_buffer(buffer)
+                     .set_timestamp_rtp(timestamp_rtp_)
+                     .build();
+
+    pulled_frames_[timestamp_rtp_] = pulled_frame;
+    timestamp_rtp_ += k90kHz / frame_settings.framerate;
+    ++frame_num_;
+
+    return frame;
+  }
+
+  // Reads frame specified by `timestamp_rtp`, scales it to `resolution` and
+  // returns. Frame with the given `timestamp_rtp` is expected to be pulled
+  // before.
+  VideoFrame GetFrame(uint32_t timestamp_rtp, Resolution resolution) override {
+    RTC_CHECK(pulled_frames_.find(timestamp_rtp) != pulled_frames_.end())
+        << "Frame with RTP timestamp " << timestamp_rtp
+        << " was not pulled before";
+    auto buffer =
+        frame_reader_->ReadFrame(pulled_frames_[timestamp_rtp], resolution);
+    return VideoFrame::Builder()
+        .set_video_frame_buffer(buffer)
+        .set_timestamp_rtp(timestamp_rtp)
+        .build();
+  }
+
+ protected:
+  std::unique_ptr<FrameReader> frame_reader_;
+  const EncodingTestSettings& test_settings_;
+  int frame_num_;
+  uint32_t timestamp_rtp_;
+  std::map<uint32_t, int> pulled_frames_;
+};
+
+class TestEncoder : public VideoCodecTester::Encoder,
+                    public EncodedImageCallback {
+ public:
+  TestEncoder(std::unique_ptr<VideoEncoder> encoder,
+              const CodecInfo& codec_info,
+              const std::map<int, EncodingSettings>& frame_settings)
+      : encoder_(std::move(encoder)),
+        codec_info_(codec_info),
+        frame_settings_(frame_settings),
+        frame_num_(0) {
+    // Ensure settings for the first frame is provided.
+    RTC_CHECK_GT(frame_settings_.size(), 0u);
+    RTC_CHECK_EQ(frame_settings_.begin()->first, 0);
+
+    encoder_->RegisterEncodeCompleteCallback(this);
+  }
+
+  void Encode(const VideoFrame& frame, EncodeCallback callback) override {
+    callbacks_[frame.timestamp()] = std::move(callback);
+
+    if (auto fs = frame_settings_.find(frame_num_);
+        fs != frame_settings_.end()) {
+      if (fs == frame_settings_.begin() ||
+          ConfigChanged(fs->second, std::prev(fs)->second)) {
+        Configure(fs->second);
+      }
+      if (fs == frame_settings_.begin() ||
+          RateChanged(fs->second, std::prev(fs)->second)) {
+        SetRates(fs->second);
+      }
+    }
+
+    int result = encoder_->Encode(frame, nullptr);
+    RTC_CHECK_EQ(result, WEBRTC_VIDEO_CODEC_OK);
+    ++frame_num_;
+  }
+
+ protected:
+  Result OnEncodedImage(const EncodedImage& encoded_image,
+                        const CodecSpecificInfo* codec_specific_info) override {
+    auto cb = callbacks_.find(encoded_image.Timestamp());
+    RTC_CHECK(cb != callbacks_.end());
+    cb->second(encoded_image);
+
+    callbacks_.erase(callbacks_.begin(), cb);
+    return Result(Result::Error::OK);
+  }
+
+  void Configure(const EncodingSettings& es) {
+    VideoCodec vc;
+    const Resolution& resolution = es.resolution.rbegin()->second;
+    vc.width = resolution.width;
+    vc.height = resolution.height;
+    const DataRate& bitrate = es.bitrate.rbegin()->second;
+    vc.startBitrate = bitrate.kbps();
+    vc.maxBitrate = bitrate.kbps();
+    vc.minBitrate = 0;
+    vc.maxFramerate = static_cast<uint32_t>(es.framerate.hertz());
+    vc.active = true;
+    vc.qpMax = 0;
+    vc.numberOfSimulcastStreams = 0;
+    vc.mode = webrtc::VideoCodecMode::kRealtimeVideo;
+    vc.SetFrameDropEnabled(true);
+
+    vc.codecType = PayloadStringToCodecType(codec_info_.type);
+    if (vc.codecType == kVideoCodecVP8) {
+      *(vc.VP8()) = VideoEncoder::GetDefaultVp8Settings();
+    } else if (vc.codecType == kVideoCodecVP9) {
+      *(vc.VP9()) = VideoEncoder::GetDefaultVp9Settings();
+    } else if (vc.codecType == kVideoCodecH264) {
+      *(vc.H264()) = VideoEncoder::GetDefaultH264Settings();
+    }
+
+    VideoEncoder::Settings ves(
+        VideoEncoder::Capabilities(/*loss_notification=*/false),
+        /*number_of_cores=*/1,
+        /*max_payload_size=*/1440);
+
+    int result = encoder_->InitEncode(&vc, ves);
+    RTC_CHECK_EQ(result, WEBRTC_VIDEO_CODEC_OK);
+  }
+
+  void SetRates(const EncodingSettings& es) {
+    VideoEncoder::RateControlParameters rc;
+    int num_spatial_layers =
+        ScalabilityModeToNumSpatialLayers(es.scalability_mode);
+    int num_temporal_layers =
+        ScalabilityModeToNumSpatialLayers(es.scalability_mode);
+    for (int sidx = 0; sidx < num_spatial_layers; ++sidx) {
+      for (int tidx = 0; tidx < num_temporal_layers; ++tidx) {
+        RTC_CHECK(es.bitrate.find(Layer(sidx, tidx)) != es.bitrate.end())
+            << "Bitrate for layer S=" << sidx << " T=" << tidx << " is not set";
+        rc.bitrate.SetBitrate(sidx, tidx,
+                              es.bitrate.at(Layer(sidx, tidx)).bps());
+      }
+    }
+
+    rc.framerate_fps = es.framerate.millihertz() / 1000.0;
+    encoder_->SetRates(rc);
+  }
+
+  bool ConfigChanged(const EncodingSettings& es,
+                     const EncodingSettings& prev_es) const {
+    return es.scalability_mode != prev_es.scalability_mode ||
+           es.resolution != prev_es.resolution;
+  }
+
+  bool RateChanged(const EncodingSettings& es,
+                   const EncodingSettings& prev_es) const {
+    return es.bitrate != prev_es.bitrate || es.framerate != prev_es.framerate;
+  }
+
+  std::unique_ptr<VideoEncoder> encoder_;
+  const CodecInfo& codec_info_;
+  const std::map<int, EncodingSettings>& frame_settings_;
+  int frame_num_;
+  std::map<uint32_t, EncodeCallback> callbacks_;
+};
+
+class TestDecoder : public VideoCodecTester::Decoder,
+                    public DecodedImageCallback {
+ public:
+  TestDecoder(std::unique_ptr<VideoDecoder> decoder,
+              const CodecInfo& codec_info)
+      : decoder_(std::move(decoder)), codec_info_(codec_info), frame_num_(0) {
+    decoder_->RegisterDecodeCompleteCallback(this);
+  }
+  void Decode(const EncodedImage& frame, DecodeCallback callback) override {
+    callbacks_[frame.Timestamp()] = std::move(callback);
+
+    if (frame_num_ == 0) {
+      Configure();
+    }
+
+    decoder_->Decode(frame, /*missing_frames=*/false,
+                     /*render_time_ms=*/0);
+    ++frame_num_;
+  }
+
+  void Configure() {
+    VideoDecoder::Settings ds;
+    ds.set_codec_type(PayloadStringToCodecType(codec_info_.type));
+    ds.set_number_of_cores(1);
+
+    bool result = decoder_->Configure(ds);
+    RTC_CHECK(result);
+  }
+
+ protected:
+  int Decoded(VideoFrame& decoded_frame) override {
+    auto cb = callbacks_.find(decoded_frame.timestamp());
+    RTC_CHECK(cb != callbacks_.end());
+    cb->second(decoded_frame);
+
+    callbacks_.erase(callbacks_.begin(), cb);
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  std::unique_ptr<VideoDecoder> decoder_;
+  const CodecInfo& codec_info_;
+  int frame_num_;
+  std::map<uint32_t, DecodeCallback> callbacks_;
+};
+
+std::unique_ptr<VideoCodecTester::Encoder> CreateEncoder(
+    const CodecInfo& codec_info,
+    const std::map<int, EncodingSettings>& frame_settings) {
+  auto factory = CreateBuiltinVideoEncoderFactory();
+  auto encoder = factory->CreateVideoEncoder(SdpVideoFormat(codec_info.type));
+  return std::make_unique<TestEncoder>(std::move(encoder), codec_info,
+                                       frame_settings);
+}
+
+std::unique_ptr<VideoCodecTester::Decoder> CreateDecoder(
+    const CodecInfo& codec_info) {
+  auto factory = CreateBuiltinVideoDecoderFactory();
+  auto decoder = factory->CreateVideoDecoder(SdpVideoFormat(codec_info.type));
+  return std::make_unique<TestDecoder>(std::move(decoder), codec_info);
+}
+
+}  // namespace
+
+class EncodeDecodeTest
+    : public ::testing::TestWithParam<EncodeDecodeTestParams> {
+ public:
+  EncodeDecodeTest() : test_params_(GetParam()) {}
+
+  void SetUp() override {
+    std::unique_ptr<FrameReader> frame_reader =
+        CreateYuvFrameReader(ResourcePath(test_params_.video.name, "yuv"),
+                             test_params_.video.resolution,
+                             YuvFrameReaderImpl::RepeatMode::kPingPong);
+    video_source_ = std::make_unique<TestRawVideoSource>(
+        std::move(frame_reader), test_params_.encoding_settings);
+
+    encoder_ = CreateEncoder(test_params_.codec,
+                             test_params_.encoding_settings.frame_settings);
+    decoder_ = CreateDecoder(test_params_.codec);
+
+    tester_ = CreateVideoCodecTester();
+  }
+
+  static std::string TestParametersToStr(
+      const ::testing::TestParamInfo<EncodeDecodeTest::ParamType>& info) {
+    return std::string(info.param.encoding_settings.name +
+                       info.param.codec.type + info.param.codec.encoder +
+                       info.param.codec.decoder);
+  }
+
+ protected:
+  EncodeDecodeTestParams test_params_;
+  std::unique_ptr<TestRawVideoSource> video_source_;
+  std::unique_ptr<VideoCodecTester::Encoder> encoder_;
+  std::unique_ptr<VideoCodecTester::Decoder> decoder_;
+  std::unique_ptr<VideoCodecTester> tester_;
+};
+
+TEST_P(EncodeDecodeTest, DISABLED_TestEncodeDecode) {
+  std::unique_ptr<VideoCodecTestStats> stats = tester_->RunEncodeDecodeTest(
+      std::move(video_source_), std::move(encoder_), std::move(decoder_),
+      test_params_.encoder_settings, test_params_.decoder_settings);
+
+  const auto& frame_settings = test_params_.encoding_settings.frame_settings;
+  for (auto fs = frame_settings.begin(); fs != frame_settings.end(); ++fs) {
+    int first_frame = fs->first;
+    int last_frame = std::next(fs) != frame_settings.end()
+                         ? std::next(fs)->first - 1
+                         : test_params_.encoding_settings.num_frames - 1;
+
+    const EncodingSettings& encoding_settings = fs->second;
+    auto metrics = stats->CalcVideoStatistic(
+        first_frame, last_frame, encoding_settings.bitrate.rbegin()->second,
+        encoding_settings.framerate);
+
+    EXPECT_GE(metrics.avg_psnr_y,
+              test_params_.quality_expectations.min_apsnr_y);
+  }
+}
+
+std::list<EncodeDecodeTestParams> ConstantRateTestParameters() {
+  std::list<EncodeDecodeTestParams> test_params;
+  std::vector<CodecInfo> codecs = {kLibvpxVp8};
+  std::vector<VideoInfo> videos = {kFourPeople_1280x720_30};
+  std::vector<std::pair<EncodingTestSettings, QualityExpectations>>
+      encoding_settings = {{kConstantRateQvga64Kbps30Fps, kLowQuality}};
+  for (const CodecInfo& codec : codecs) {
+    for (const VideoInfo& video : videos) {
+      for (const auto& es : encoding_settings) {
+        EncodeDecodeTestParams p;
+        p.codec = codec;
+        p.video = video;
+        p.encoding_settings = es.first;
+        p.quality_expectations = es.second;
+        test_params.push_back(p);
+      }
+    }
+  }
+  return test_params;
+}
+
+INSTANTIATE_TEST_SUITE_P(ConstantRate,
+                         EncodeDecodeTest,
+                         ::testing::ValuesIn(ConstantRateTestParameters()),
+                         EncodeDecodeTest::TestParametersToStr);
+}  // namespace test
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.cc
new file mode 100644
index 0000000000..3000c1adee
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.cc
@@ -0,0 +1,325 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/video_codec_tester_impl.h"
+
+#include <map>
+#include <memory>
+#include <utility>
+
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "modules/video_coding/codecs/test/video_codec_analyzer.h"
+#include "rtc_base/event.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/sleep.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+using RawVideoSource = VideoCodecTester::RawVideoSource;
+using CodedVideoSource = VideoCodecTester::CodedVideoSource;
+using Decoder = VideoCodecTester::Decoder;
+using Encoder = VideoCodecTester::Encoder;
+using EncoderSettings = VideoCodecTester::EncoderSettings;
+using DecoderSettings = VideoCodecTester::DecoderSettings;
+using PacingSettings = VideoCodecTester::PacingSettings;
+using PacingMode = PacingSettings::PacingMode;
+
+constexpr Frequency k90kHz = Frequency::Hertz(90000);
+
+// A thread-safe wrapper for video source to be shared with the quality analyzer
+// that reads reference frames from a separate thread.
+class SyncRawVideoSource : public VideoCodecAnalyzer::ReferenceVideoSource {
+ public:
+  explicit SyncRawVideoSource(std::unique_ptr<RawVideoSource> video_source)
+      : video_source_(std::move(video_source)) {}
+
+  absl::optional<VideoFrame> PullFrame() {
+    MutexLock lock(&mutex_);
+    return video_source_->PullFrame();
+  }
+
+  VideoFrame GetFrame(uint32_t timestamp_rtp, Resolution resolution) override {
+    MutexLock lock(&mutex_);
+    return video_source_->GetFrame(timestamp_rtp, resolution);
+  }
+
+ protected:
+  std::unique_ptr<RawVideoSource> video_source_ RTC_GUARDED_BY(mutex_);
+  Mutex mutex_;
+};
+
+// Pacer calculates delay necessary to keep frame encode or decode call spaced
+// from the previous calls by the pacing time. `Delay` is expected to be called
+// as close as possible to posting frame encode or decode task. This class is
+// not thread safe.
+class Pacer {
+ public:
+  explicit Pacer(PacingSettings settings)
+      : settings_(settings), delay_(TimeDelta::Zero()) {}
+  TimeDelta Delay(Timestamp beat) {
+    if (settings_.mode == PacingMode::kNoPacing) {
+      return TimeDelta::Zero();
+    }
+
+    Timestamp now = Timestamp::Micros(rtc::TimeMicros());
+    if (prev_time_.has_value()) {
+      delay_ += PacingTime(beat);
+      delay_ -= (now - *prev_time_);
+      if (delay_.ns() < 0) {
+        delay_ = TimeDelta::Zero();
+      }
+    }
+
+    prev_beat_ = beat;
+    prev_time_ = now;
+    return delay_;
+  }
+
+ private:
+  TimeDelta PacingTime(Timestamp beat) {
+    if (settings_.mode == PacingMode::kRealTime) {
+      return beat - *prev_beat_;
+    }
+    RTC_CHECK_EQ(PacingMode::kConstantRate, settings_.mode);
+    return 1 / settings_.constant_rate;
+  }
+
+  PacingSettings settings_;
+  absl::optional<Timestamp> prev_beat_;
+  absl::optional<Timestamp> prev_time_;
+  TimeDelta delay_;
+};
+
+// Task queue that keeps the number of queued tasks below a certain limit. If
+// the limit is reached, posting of a next task is blocked until execution of a
+// previously posted task starts. This class is not thread-safe.
+class LimitedTaskQueue {
+ public:
+  // The codec tester reads frames from video source in the main thread.
+  // Encoding and decoding are done in separate threads. If encoding or
+  // decoding is slow, the reading may go far ahead and may buffer too many
+  // frames in memory. To prevent this we limit the encoding/decoding queue
+  // size. When the queue is full, the main thread and, hence, reading frames
+  // from video source is blocked until a previously posted encoding/decoding
+  // task starts.
+  static constexpr int kMaxTaskQueueSize = 3;
+
+  explicit LimitedTaskQueue(rtc::TaskQueue& task_queue)
+      : task_queue_(task_queue), queue_size_(0) {}
+
+  void PostDelayedTask(absl::AnyInvocable<void() &&> task, TimeDelta delay) {
+    ++queue_size_;
+    task_queue_.PostDelayedTask(
+        [this, task = std::move(task)]() mutable {
+          std::move(task)();
+          --queue_size_;
+          task_executed_.Set();
+        },
+        delay);
+
+    task_executed_.Reset();
+    if (queue_size_ > kMaxTaskQueueSize) {
+      task_executed_.Wait(rtc::Event::kForever);
+    }
+    RTC_CHECK(queue_size_ <= kMaxTaskQueueSize);
+  }
+
+  void WaitForPreviouslyPostedTasks() {
+    while (queue_size_ > 0) {
+      task_executed_.Wait(rtc::Event::kForever);
+      task_executed_.Reset();
+    }
+  }
+
+  rtc::TaskQueue& task_queue_;
+  std::atomic_int queue_size_;
+  rtc::Event task_executed_;
+};
+
+class TesterDecoder {
+ public:
+  TesterDecoder(std::unique_ptr<Decoder> decoder,
+                VideoCodecAnalyzer* analyzer,
+                const DecoderSettings& settings,
+                rtc::TaskQueue& task_queue)
+      : decoder_(std::move(decoder)),
+        analyzer_(analyzer),
+        settings_(settings),
+        pacer_(settings.pacing),
+        task_queue_(task_queue) {
+    RTC_CHECK(analyzer_) << "Analyzer must be provided";
+  }
+
+  void Decode(const EncodedImage& frame) {
+    Timestamp timestamp = Timestamp::Micros((frame.Timestamp() / k90kHz).us());
+
+    task_queue_.PostDelayedTask(
+        [this, frame] {
+          analyzer_->StartDecode(frame);
+          decoder_->Decode(frame, [this](const VideoFrame& decoded_frame) {
+            this->analyzer_->FinishDecode(decoded_frame, /*spatial_idx=*/0);
+          });
+        },
+        pacer_.Delay(timestamp));
+  }
+
+  void Flush() { task_queue_.WaitForPreviouslyPostedTasks(); }
+
+ protected:
+  std::unique_ptr<Decoder> decoder_;
+  VideoCodecAnalyzer* const analyzer_;
+  const DecoderSettings& settings_;
+  Pacer pacer_;
+  LimitedTaskQueue task_queue_;
+};
+
+class TesterEncoder {
+ public:
+  TesterEncoder(std::unique_ptr<Encoder> encoder,
+                TesterDecoder* decoder,
+                VideoCodecAnalyzer* analyzer,
+                const EncoderSettings& settings,
+                rtc::TaskQueue& task_queue)
+      : encoder_(std::move(encoder)),
+        decoder_(decoder),
+        analyzer_(analyzer),
+        settings_(settings),
+        pacer_(settings.pacing),
+        task_queue_(task_queue) {
+    RTC_CHECK(analyzer_) << "Analyzer must be provided";
+  }
+
+  void Encode(const VideoFrame& frame) {
+    Timestamp timestamp = Timestamp::Micros((frame.timestamp() / k90kHz).us());
+
+    task_queue_.PostDelayedTask(
+        [this, frame] {
+          analyzer_->StartEncode(frame);
+          encoder_->Encode(frame, [this](const EncodedImage& encoded_frame) {
+            this->analyzer_->FinishEncode(encoded_frame);
+            if (decoder_ != nullptr) {
+              this->decoder_->Decode(encoded_frame);
+            }
+          });
+        },
+        pacer_.Delay(timestamp));
+  }
+
+  void Flush() { task_queue_.WaitForPreviouslyPostedTasks(); }
+
+ protected:
+  std::unique_ptr<Encoder> encoder_;
+  TesterDecoder* const decoder_;
+  VideoCodecAnalyzer* const analyzer_;
+  const EncoderSettings& settings_;
+  Pacer pacer_;
+  LimitedTaskQueue task_queue_;
+};
+
+}  // namespace
+
+VideoCodecTesterImpl::VideoCodecTesterImpl()
+    : VideoCodecTesterImpl(/*task_queue_factory=*/nullptr) {}
+
+VideoCodecTesterImpl::VideoCodecTesterImpl(TaskQueueFactory* task_queue_factory)
+    : task_queue_factory_(task_queue_factory) {
+  if (task_queue_factory_ == nullptr) {
+    owned_task_queue_factory_ = CreateDefaultTaskQueueFactory();
+    task_queue_factory_ = owned_task_queue_factory_.get();
+  }
+}
+
+std::unique_ptr<VideoCodecTestStats> VideoCodecTesterImpl::RunDecodeTest(
+    std::unique_ptr<CodedVideoSource> video_source,
+    std::unique_ptr<Decoder> decoder,
+    const DecoderSettings& decoder_settings) {
+  rtc::TaskQueue analyser_task_queue(task_queue_factory_->CreateTaskQueue(
+      "Analyzer", TaskQueueFactory::Priority::NORMAL));
+  rtc::TaskQueue decoder_task_queue(task_queue_factory_->CreateTaskQueue(
+      "Decoder", TaskQueueFactory::Priority::NORMAL));
+
+  VideoCodecAnalyzer perf_analyzer(analyser_task_queue);
+  TesterDecoder tester_decoder(std::move(decoder), &perf_analyzer,
+                               decoder_settings, decoder_task_queue);
+
+  while (auto frame = video_source->PullFrame()) {
+    tester_decoder.Decode(*frame);
+  }
+
+  tester_decoder.Flush();
+
+  return perf_analyzer.GetStats();
+}
+
+std::unique_ptr<VideoCodecTestStats> VideoCodecTesterImpl::RunEncodeTest(
+    std::unique_ptr<RawVideoSource> video_source,
+    std::unique_ptr<Encoder> encoder,
+    const EncoderSettings& encoder_settings) {
+  rtc::TaskQueue analyser_task_queue(task_queue_factory_->CreateTaskQueue(
+      "Analyzer", TaskQueueFactory::Priority::NORMAL));
+  rtc::TaskQueue encoder_task_queue(task_queue_factory_->CreateTaskQueue(
+      "Encoder", TaskQueueFactory::Priority::NORMAL));
+
+  SyncRawVideoSource sync_source(std::move(video_source));
+  VideoCodecAnalyzer perf_analyzer(analyser_task_queue);
+  TesterEncoder tester_encoder(std::move(encoder), /*decoder=*/nullptr,
+                               &perf_analyzer, encoder_settings,
+                               encoder_task_queue);
+
+  while (auto frame = sync_source.PullFrame()) {
+    tester_encoder.Encode(*frame);
+  }
+
+  tester_encoder.Flush();
+
+  return perf_analyzer.GetStats();
+}
+
+std::unique_ptr<VideoCodecTestStats> VideoCodecTesterImpl::RunEncodeDecodeTest(
+    std::unique_ptr<RawVideoSource> video_source,
+    std::unique_ptr<Encoder> encoder,
+    std::unique_ptr<Decoder> decoder,
+    const EncoderSettings& encoder_settings,
+    const DecoderSettings& decoder_settings) {
+  rtc::TaskQueue analyser_task_queue(task_queue_factory_->CreateTaskQueue(
+      "Analyzer", TaskQueueFactory::Priority::NORMAL));
+  rtc::TaskQueue decoder_task_queue(task_queue_factory_->CreateTaskQueue(
+      "Decoder", TaskQueueFactory::Priority::NORMAL));
+  rtc::TaskQueue encoder_task_queue(task_queue_factory_->CreateTaskQueue(
+      "Encoder", TaskQueueFactory::Priority::NORMAL));
+
+  SyncRawVideoSource sync_source(std::move(video_source));
+  VideoCodecAnalyzer perf_analyzer(analyser_task_queue, &sync_source);
+  TesterDecoder tester_decoder(std::move(decoder), &perf_analyzer,
+                               decoder_settings, decoder_task_queue);
+  TesterEncoder tester_encoder(std::move(encoder), &tester_decoder,
+                               &perf_analyzer, encoder_settings,
+                               encoder_task_queue);
+
+  while (auto frame = sync_source.PullFrame()) {
+    tester_encoder.Encode(*frame);
+  }
+
+  tester_encoder.Flush();
+  tester_decoder.Flush();
+
+  return perf_analyzer.GetStats();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.h b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.h
new file mode 100644
index 0000000000..b64adeb882
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_TESTER_IMPL_H_
+#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_TESTER_IMPL_H_
+
+#include <memory>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/video_codec_tester.h"
+
+namespace webrtc {
+namespace test {
+
+// A stateless implementation of `VideoCodecTester`. This class is thread safe.
+class VideoCodecTesterImpl : public VideoCodecTester {
+ public:
+  VideoCodecTesterImpl();
+  explicit VideoCodecTesterImpl(TaskQueueFactory* task_queue_factory);
+
+  std::unique_ptr<VideoCodecTestStats> RunDecodeTest(
+      std::unique_ptr<CodedVideoSource> video_source,
+      std::unique_ptr<Decoder> decoder,
+      const DecoderSettings& decoder_settings) override;
+
+  std::unique_ptr<VideoCodecTestStats> RunEncodeTest(
+      std::unique_ptr<RawVideoSource> video_source,
+      std::unique_ptr<Encoder> encoder,
+      const EncoderSettings& encoder_settings) override;
+
+  std::unique_ptr<VideoCodecTestStats> RunEncodeDecodeTest(
+      std::unique_ptr<RawVideoSource> video_source,
+      std::unique_ptr<Encoder> encoder,
+      std::unique_ptr<Decoder> decoder,
+      const EncoderSettings& encoder_settings,
+      const DecoderSettings& decoder_settings) override;
+
+ protected:
+  std::unique_ptr<TaskQueueFactory> owned_task_queue_factory_;
+  TaskQueueFactory* task_queue_factory_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_TESTER_IMPL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl_unittest.cc
new file mode 100644
index 0000000000..29fb006fb5
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_tester_impl_unittest.cc
@@ -0,0 +1,259 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/video_codec_tester_impl.h"
+
+#include <memory>
+#include <tuple>
+#include <utility>
+#include <vector>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "api/task_queue/test/mock_task_queue_base.h"
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/time_utils.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+using ::testing::_;
+using ::testing::Invoke;
+using ::testing::InvokeWithoutArgs;
+using ::testing::Return;
+
+using Decoder = VideoCodecTester::Decoder;
+using Encoder = VideoCodecTester::Encoder;
+using CodedVideoSource = VideoCodecTester::CodedVideoSource;
+using RawVideoSource = VideoCodecTester::RawVideoSource;
+using DecoderSettings = VideoCodecTester::DecoderSettings;
+using EncoderSettings = VideoCodecTester::EncoderSettings;
+using PacingSettings = VideoCodecTester::PacingSettings;
+using PacingMode = PacingSettings::PacingMode;
+
+constexpr Frequency k90kHz = Frequency::Hertz(90000);
+
+VideoFrame CreateVideoFrame(uint32_t timestamp_rtp) {
+  rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(2, 2));
+  return VideoFrame::Builder()
+      .set_video_frame_buffer(buffer)
+      .set_timestamp_rtp(timestamp_rtp)
+      .build();
+}
+
+EncodedImage CreateEncodedImage(uint32_t timestamp_rtp) {
+  EncodedImage encoded_image;
+  encoded_image.SetTimestamp(timestamp_rtp);
+  return encoded_image;
+}
+
+class MockRawVideoSource : public RawVideoSource {
+ public:
+  MOCK_METHOD(absl::optional<VideoFrame>, PullFrame, (), (override));
+  MOCK_METHOD(VideoFrame,
+              GetFrame,
+              (uint32_t timestamp_rtp, Resolution),
+              (override));
+};
+
+class MockCodedVideoSource : public CodedVideoSource {
+ public:
+  MOCK_METHOD(absl::optional<EncodedImage>, PullFrame, (), (override));
+};
+
+class MockDecoder : public Decoder {
+ public:
+  MOCK_METHOD(void,
+              Decode,
+              (const EncodedImage& frame, DecodeCallback callback),
+              (override));
+};
+
+class MockEncoder : public Encoder {
+ public:
+  MOCK_METHOD(void,
+              Encode,
+              (const VideoFrame& frame, EncodeCallback callback),
+              (override));
+};
+
+class MockTaskQueueFactory : public TaskQueueFactory {
+ public:
+  explicit MockTaskQueueFactory(TaskQueueBase& task_queue)
+      : task_queue_(task_queue) {}
+
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
+      absl::string_view name,
+      Priority priority) const override {
+    return std::unique_ptr<TaskQueueBase, TaskQueueDeleter>(&task_queue_);
+  }
+
+ protected:
+  TaskQueueBase& task_queue_;
+};
+}  // namespace
+
+class VideoCodecTesterImplPacingTest
+    : public ::testing::TestWithParam<std::tuple<PacingSettings,
+                                                 std::vector<int>,
+                                                 std::vector<int>,
+                                                 std::vector<int>>> {
+ public:
+  VideoCodecTesterImplPacingTest()
+      : pacing_settings_(std::get<0>(GetParam())),
+        frame_timestamp_ms_(std::get<1>(GetParam())),
+        frame_capture_delay_ms_(std::get<2>(GetParam())),
+        expected_frame_start_ms_(std::get<3>(GetParam())),
+        num_frames_(frame_timestamp_ms_.size()),
+        task_queue_factory_(task_queue_) {}
+
+  void SetUp() override {
+    ON_CALL(task_queue_, PostTask)
+        .WillByDefault(Invoke(
+            [](absl::AnyInvocable<void() &&> task) { std::move(task)(); }));
+
+    ON_CALL(task_queue_, PostDelayedTask)
+        .WillByDefault(
+            Invoke([&](absl::AnyInvocable<void() &&> task, TimeDelta delay) {
+              clock_.AdvanceTime(delay);
+              std::move(task)();
+            }));
+  }
+
+ protected:
+  PacingSettings pacing_settings_;
+  std::vector<int> frame_timestamp_ms_;
+  std::vector<int> frame_capture_delay_ms_;
+  std::vector<int> expected_frame_start_ms_;
+  size_t num_frames_;
+
+  rtc::ScopedFakeClock clock_;
+  MockTaskQueueBase task_queue_;
+  MockTaskQueueFactory task_queue_factory_;
+};
+
+TEST_P(VideoCodecTesterImplPacingTest, PaceEncode) {
+  auto video_source = std::make_unique<MockRawVideoSource>();
+
+  size_t frame_num = 0;
+  EXPECT_CALL(*video_source, PullFrame).WillRepeatedly(Invoke([&]() mutable {
+    if (frame_num >= num_frames_) {
+      return absl::optional<VideoFrame>();
+    }
+    clock_.AdvanceTime(TimeDelta::Millis(frame_capture_delay_ms_[frame_num]));
+
+    uint32_t timestamp_rtp = frame_timestamp_ms_[frame_num] * k90kHz.hertz() /
+                             rtc::kNumMillisecsPerSec;
+    ++frame_num;
+    return absl::optional<VideoFrame>(CreateVideoFrame(timestamp_rtp));
+  }));
+
+  auto encoder = std::make_unique<MockEncoder>();
+  EncoderSettings encoder_settings;
+  encoder_settings.pacing = pacing_settings_;
+
+  VideoCodecTesterImpl tester(&task_queue_factory_);
+  auto fs = tester
+                .RunEncodeTest(std::move(video_source), std::move(encoder),
+                               encoder_settings)
+                ->GetFrameStatistics();
+  ASSERT_EQ(fs.size(), num_frames_);
+
+  for (size_t i = 0; i < fs.size(); ++i) {
+    int encode_start_ms = (fs[i].encode_start_ns - fs[0].encode_start_ns) /
+                          rtc::kNumNanosecsPerMillisec;
+    EXPECT_NEAR(encode_start_ms, expected_frame_start_ms_[i], 10);
+  }
+}
+
+TEST_P(VideoCodecTesterImplPacingTest, PaceDecode) {
+  auto video_source = std::make_unique<MockCodedVideoSource>();
+
+  size_t frame_num = 0;
+  EXPECT_CALL(*video_source, PullFrame).WillRepeatedly(Invoke([&]() mutable {
+    if (frame_num >= num_frames_) {
+      return absl::optional<EncodedImage>();
+    }
+    clock_.AdvanceTime(TimeDelta::Millis(frame_capture_delay_ms_[frame_num]));
+
+    uint32_t timestamp_rtp = frame_timestamp_ms_[frame_num] * k90kHz.hertz() /
+                             rtc::kNumMillisecsPerSec;
+    ++frame_num;
+    return absl::optional<EncodedImage>(CreateEncodedImage(timestamp_rtp));
+  }));
+
+  auto decoder = std::make_unique<MockDecoder>();
+  DecoderSettings decoder_settings;
+  decoder_settings.pacing = pacing_settings_;
+
+  VideoCodecTesterImpl tester(&task_queue_factory_);
+  auto fs = tester
+                .RunDecodeTest(std::move(video_source), std::move(decoder),
+                               decoder_settings)
+                ->GetFrameStatistics();
+  ASSERT_EQ(fs.size(), num_frames_);
+
+  for (size_t i = 0; i < fs.size(); ++i) {
+    int decode_start_ms = (fs[i].decode_start_ns - fs[0].decode_start_ns) /
+                          rtc::kNumNanosecsPerMillisec;
+    EXPECT_NEAR(decode_start_ms, expected_frame_start_ms_[i], 10);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    All,
+    VideoCodecTesterImplPacingTest,
+    ::testing::ValuesIn(
+        {std::make_tuple(PacingSettings({.mode = PacingMode::kNoPacing}),
+                         /*frame_timestamp_ms=*/std::vector<int>{0, 100},
+                         /*frame_capture_delay_ms=*/std::vector<int>{0, 0},
+                         /*expected_frame_start_ms=*/std::vector<int>{0, 0}),
+         // Pace with rate equal to the source frame rate. Frames are captured
+         // instantly. Verify that frames are paced with the source frame rate.
+         std::make_tuple(PacingSettings({.mode = PacingMode::kRealTime}),
+                         /*frame_timestamp_ms=*/std::vector<int>{0, 100},
+                         /*frame_capture_delay_ms=*/std::vector<int>{0, 0},
+                         /*expected_frame_start_ms=*/std::vector<int>{0, 100}),
+         // Pace with rate equal to the source frame rate. Frame capture is
+         // delayed by more than pacing time. Verify that no extra delay is
+         // added.
+         std::make_tuple(PacingSettings({.mode = PacingMode::kRealTime}),
+                         /*frame_timestamp_ms=*/std::vector<int>{0, 100},
+                         /*frame_capture_delay_ms=*/std::vector<int>{0, 200},
+                         /*expected_frame_start_ms=*/std::vector<int>{0, 200}),
+         // Pace with constant rate less then source frame rate. Frames are
+         // captured instantly. Verify that frames are paced with the requested
+         // constant rate.
+         std::make_tuple(
+             PacingSettings({.mode = PacingMode::kConstantRate,
+                             .constant_rate = Frequency::Hertz(20)}),
+             /*frame_timestamp_ms=*/std::vector<int>{0, 100},
+             /*frame_capture_delay_ms=*/std::vector<int>{0, 0},
+             /*expected_frame_start_ms=*/std::vector<int>{0, 50}),
+         // Pace with constant rate less then source frame rate. Frame capture
+         // is delayed by more than the pacing time. Verify that no extra delay
+         // is added.
+         std::make_tuple(
+             PacingSettings({.mode = PacingMode::kConstantRate,
+                             .constant_rate = Frequency::Hertz(20)}),
+             /*frame_timestamp_ms=*/std::vector<int>{0, 100},
+             /*frame_capture_delay_ms=*/std::vector<int>{0, 200},
+             /*expected_frame_start_ms=*/std::vector<int>{0, 200})}));
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_unittest.cc
new file mode 100644
index 0000000000..a4a8b253fc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_unittest.cc
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
+
+#include <utility>
+
+#include "api/test/create_frame_generator.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "test/video_codec_settings.h"
+
+static constexpr webrtc::TimeDelta kEncodeTimeout =
+    webrtc::TimeDelta::Millis(100);
+static constexpr webrtc::TimeDelta kDecodeTimeout =
+    webrtc::TimeDelta::Millis(25);
+// Set bitrate to get higher quality.
+static const int kStartBitrate = 300;
+static const int kMaxBitrate = 4000;
+static const int kWidth = 176;        // Width of the input image.
+static const int kHeight = 144;       // Height of the input image.
+static const int kMaxFramerate = 30;  // Arbitrary value.
+
+namespace webrtc {
+namespace {
+const VideoEncoder::Capabilities kCapabilities(false);
+}
+
+EncodedImageCallback::Result
+VideoCodecUnitTest::FakeEncodeCompleteCallback::OnEncodedImage(
+    const EncodedImage& frame,
+    const CodecSpecificInfo* codec_specific_info) {
+  MutexLock lock(&test_->encoded_frame_section_);
+  test_->encoded_frames_.push_back(frame);
+  RTC_DCHECK(codec_specific_info);
+  test_->codec_specific_infos_.push_back(*codec_specific_info);
+  if (!test_->wait_for_encoded_frames_threshold_) {
+    test_->encoded_frame_event_.Set();
+    return Result(Result::OK);
+  }
+
+  if (test_->encoded_frames_.size() ==
+      test_->wait_for_encoded_frames_threshold_) {
+    test_->wait_for_encoded_frames_threshold_ = 1;
+    test_->encoded_frame_event_.Set();
+  }
+  return Result(Result::OK);
+}
+
+void VideoCodecUnitTest::FakeDecodeCompleteCallback::Decoded(
+    VideoFrame& frame,
+    absl::optional<int32_t> decode_time_ms,
+    absl::optional<uint8_t> qp) {
+  MutexLock lock(&test_->decoded_frame_section_);
+  test_->decoded_frame_.emplace(frame);
+  test_->decoded_qp_ = qp;
+  test_->decoded_frame_event_.Set();
+}
+
+void VideoCodecUnitTest::SetUp() {
+  webrtc::test::CodecSettings(kVideoCodecVP8, &codec_settings_);
+  codec_settings_.startBitrate = kStartBitrate;
+  codec_settings_.maxBitrate = kMaxBitrate;
+  codec_settings_.maxFramerate = kMaxFramerate;
+  codec_settings_.width = kWidth;
+  codec_settings_.height = kHeight;
+
+  ModifyCodecSettings(&codec_settings_);
+
+  input_frame_generator_ = test::CreateSquareFrameGenerator(
+      codec_settings_.width, codec_settings_.height,
+      test::FrameGeneratorInterface::OutputType::kI420, absl::optional<int>());
+
+  encoder_ = CreateEncoder();
+  decoder_ = CreateDecoder();
+  encoder_->RegisterEncodeCompleteCallback(&encode_complete_callback_);
+  decoder_->RegisterDecodeCompleteCallback(&decode_complete_callback_);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(
+                &codec_settings_,
+                VideoEncoder::Settings(kCapabilities, 1 /* number of cores */,
+                                       0 /* max payload size (unused) */)));
+
+  VideoDecoder::Settings decoder_settings;
+  decoder_settings.set_codec_type(codec_settings_.codecType);
+  decoder_settings.set_max_render_resolution(
+      {codec_settings_.width, codec_settings_.height});
+  EXPECT_TRUE(decoder_->Configure(decoder_settings));
+}
+
+void VideoCodecUnitTest::ModifyCodecSettings(VideoCodec* codec_settings) {}
+
+VideoFrame VideoCodecUnitTest::NextInputFrame() {
+  test::FrameGeneratorInterface::VideoFrameData frame_data =
+      input_frame_generator_->NextFrame();
+  VideoFrame input_frame = VideoFrame::Builder()
+                               .set_video_frame_buffer(frame_data.buffer)
+                               .set_update_rect(frame_data.update_rect)
+                               .build();
+
+  const uint32_t timestamp =
+      last_input_frame_timestamp_ +
+      kVideoPayloadTypeFrequency / codec_settings_.maxFramerate;
+  input_frame.set_timestamp(timestamp);
+
+  last_input_frame_timestamp_ = timestamp;
+  return input_frame;
+}
+
+bool VideoCodecUnitTest::WaitForEncodedFrame(
+    EncodedImage* frame,
+    CodecSpecificInfo* codec_specific_info) {
+  std::vector<EncodedImage> frames;
+  std::vector<CodecSpecificInfo> codec_specific_infos;
+  if (!WaitForEncodedFrames(&frames, &codec_specific_infos))
+    return false;
+  EXPECT_EQ(frames.size(), static_cast<size_t>(1));
+  EXPECT_EQ(frames.size(), codec_specific_infos.size());
+  *frame = frames[0];
+  *codec_specific_info = codec_specific_infos[0];
+  return true;
+}
+
+void VideoCodecUnitTest::SetWaitForEncodedFramesThreshold(size_t num_frames) {
+  MutexLock lock(&encoded_frame_section_);
+  wait_for_encoded_frames_threshold_ = num_frames;
+}
+
+bool VideoCodecUnitTest::WaitForEncodedFrames(
+    std::vector<EncodedImage>* frames,
+    std::vector<CodecSpecificInfo>* codec_specific_info) {
+  EXPECT_TRUE(encoded_frame_event_.Wait(kEncodeTimeout))
+      << "Timed out while waiting for encoded frame.";
+  // This becomes unsafe if there are multiple threads waiting for frames.
+  MutexLock lock(&encoded_frame_section_);
+  EXPECT_FALSE(encoded_frames_.empty());
+  EXPECT_FALSE(codec_specific_infos_.empty());
+  EXPECT_EQ(encoded_frames_.size(), codec_specific_infos_.size());
+  if (!encoded_frames_.empty()) {
+    *frames = encoded_frames_;
+    encoded_frames_.clear();
+    RTC_DCHECK(!codec_specific_infos_.empty());
+    *codec_specific_info = codec_specific_infos_;
+    codec_specific_infos_.clear();
+    return true;
+  } else {
+    return false;
+  }
+}
+
+bool VideoCodecUnitTest::WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
+                                             absl::optional<uint8_t>* qp) {
+  bool ret = decoded_frame_event_.Wait(kDecodeTimeout);
+  EXPECT_TRUE(ret) << "Timed out while waiting for a decoded frame.";
+  // This becomes unsafe if there are multiple threads waiting for frames.
+  MutexLock lock(&decoded_frame_section_);
+  EXPECT_TRUE(decoded_frame_);
+  if (decoded_frame_) {
+    frame->reset(new VideoFrame(std::move(*decoded_frame_)));
+    *qp = decoded_qp_;
+    decoded_frame_.reset();
+    return true;
+  } else {
+    return false;
+  }
+}
+
+size_t VideoCodecUnitTest::GetNumEncodedFrames() {
+  MutexLock lock(&encoded_frame_section_);
+  return encoded_frames_.size();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_unittest.h b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_unittest.h
new file mode 100644
index 0000000000..7d05882b63
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_unittest.h
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_UNITTEST_H_
+#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_UNITTEST_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/test/frame_generator_interface.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/utility/vp8_header_parser.h"
+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+#include "rtc_base/event.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class VideoCodecUnitTest : public ::testing::Test {
+ public:
+  VideoCodecUnitTest()
+      : encode_complete_callback_(this),
+        decode_complete_callback_(this),
+        wait_for_encoded_frames_threshold_(1),
+        last_input_frame_timestamp_(0) {}
+
+ protected:
+  class FakeEncodeCompleteCallback : public webrtc::EncodedImageCallback {
+   public:
+    explicit FakeEncodeCompleteCallback(VideoCodecUnitTest* test)
+        : test_(test) {}
+
+    Result OnEncodedImage(const EncodedImage& frame,
+                          const CodecSpecificInfo* codec_specific_info);
+
+   private:
+    VideoCodecUnitTest* const test_;
+  };
+
+  class FakeDecodeCompleteCallback : public webrtc::DecodedImageCallback {
+   public:
+    explicit FakeDecodeCompleteCallback(VideoCodecUnitTest* test)
+        : test_(test) {}
+
+    int32_t Decoded(VideoFrame& frame) override {
+      RTC_DCHECK_NOTREACHED();
+      return -1;
+    }
+    int32_t Decoded(VideoFrame& frame, int64_t decode_time_ms) override {
+      RTC_DCHECK_NOTREACHED();
+      return -1;
+    }
+    void Decoded(VideoFrame& frame,
+                 absl::optional<int32_t> decode_time_ms,
+                 absl::optional<uint8_t> qp) override;
+
+   private:
+    VideoCodecUnitTest* const test_;
+  };
+
+  virtual std::unique_ptr<VideoEncoder> CreateEncoder() = 0;
+  virtual std::unique_ptr<VideoDecoder> CreateDecoder() = 0;
+
+  void SetUp() override;
+
+  virtual void ModifyCodecSettings(VideoCodec* codec_settings);
+
+  VideoFrame NextInputFrame();
+
+  // Helper method for waiting a single encoded frame.
+  bool WaitForEncodedFrame(EncodedImage* frame,
+                           CodecSpecificInfo* codec_specific_info);
+
+  // Helper methods for waiting for multiple encoded frames. Caller must
+  // define how many frames are to be waited for via `num_frames` before calling
+  // Encode(). Then, they can expect to retrive them via WaitForEncodedFrames().
+  void SetWaitForEncodedFramesThreshold(size_t num_frames);
+  bool WaitForEncodedFrames(
+      std::vector<EncodedImage>* frames,
+      std::vector<CodecSpecificInfo>* codec_specific_info);
+
+  // Helper method for waiting a single decoded frame.
+  bool WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
+                           absl::optional<uint8_t>* qp);
+
+  size_t GetNumEncodedFrames();
+
+  VideoCodec codec_settings_;
+
+  std::unique_ptr<VideoEncoder> encoder_;
+  std::unique_ptr<VideoDecoder> decoder_;
+  std::unique_ptr<test::FrameGeneratorInterface> input_frame_generator_;
+
+ private:
+  FakeEncodeCompleteCallback encode_complete_callback_;
+  FakeDecodeCompleteCallback decode_complete_callback_;
+
+  rtc::Event encoded_frame_event_;
+  Mutex encoded_frame_section_;
+  size_t wait_for_encoded_frames_threshold_;
+  std::vector<EncodedImage> encoded_frames_
+      RTC_GUARDED_BY(encoded_frame_section_);
+  std::vector<CodecSpecificInfo> codec_specific_infos_
+      RTC_GUARDED_BY(encoded_frame_section_);
+
+  rtc::Event decoded_frame_event_;
+  Mutex decoded_frame_section_;
+  absl::optional<VideoFrame> decoded_frame_
+      RTC_GUARDED_BY(decoded_frame_section_);
+  absl::optional<uint8_t> decoded_qp_ RTC_GUARDED_BY(decoded_frame_section_);
+
+  uint32_t last_input_frame_timestamp_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_UNITTEST_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_encoder_decoder_instantiation_tests.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_encoder_decoder_instantiation_tests.cc
new file mode 100644
index 0000000000..41f2304748
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_encoder_decoder_instantiation_tests.cc
@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#if defined(WEBRTC_ANDROID)
+#include "modules/video_coding/codecs/test/android_codec_factory_helper.h"
+#elif defined(WEBRTC_IOS)
+#include "modules/video_coding/codecs/test/objc_codec_factory_helper.h"
+#endif
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/video_codec_settings.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+using ::testing::NotNull;
+
+const VideoEncoder::Capabilities kCapabilities(false);
+
+int32_t InitEncoder(VideoCodecType codec_type, VideoEncoder* encoder) {
+  VideoCodec codec;
+  CodecSettings(codec_type, &codec);
+  codec.width = 640;
+  codec.height = 480;
+  codec.maxFramerate = 30;
+  RTC_CHECK(encoder);
+  return encoder->InitEncode(
+      &codec, VideoEncoder::Settings(kCapabilities, 1 /* number_of_cores */,
+                                     1200 /* max_payload_size */));
+}
+
+VideoDecoder::Settings DecoderSettings(VideoCodecType codec_type) {
+  VideoDecoder::Settings settings;
+  settings.set_max_render_resolution({640, 480});
+  settings.set_codec_type(codec_type);
+  return settings;
+}
+
+}  // namespace
+
+class VideoEncoderDecoderInstantiationTest
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<::testing::tuple<int, int>> {
+ protected:
+  VideoEncoderDecoderInstantiationTest()
+      : vp8_format_("VP8"),
+        vp9_format_("VP9"),
+        h264cbp_format_("H264"),
+        num_encoders_(::testing::get<0>(GetParam())),
+        num_decoders_(::testing::get<1>(GetParam())) {
+#if defined(WEBRTC_ANDROID)
+    InitializeAndroidObjects();
+    encoder_factory_ = CreateAndroidEncoderFactory();
+    decoder_factory_ = CreateAndroidDecoderFactory();
+#elif defined(WEBRTC_IOS)
+    encoder_factory_ = CreateObjCEncoderFactory();
+    decoder_factory_ = CreateObjCDecoderFactory();
+#else
+    RTC_DCHECK_NOTREACHED() << "Only support Android and iOS.";
+#endif
+  }
+
+  ~VideoEncoderDecoderInstantiationTest() {
+    for (auto& encoder : encoders_) {
+      encoder->Release();
+    }
+    for (auto& decoder : decoders_) {
+      decoder->Release();
+    }
+  }
+
+  const SdpVideoFormat vp8_format_;
+  const SdpVideoFormat vp9_format_;
+  const SdpVideoFormat h264cbp_format_;
+  std::unique_ptr<VideoEncoderFactory> encoder_factory_;
+  std::unique_ptr<VideoDecoderFactory> decoder_factory_;
+
+  const int num_encoders_;
+  const int num_decoders_;
+  std::vector<std::unique_ptr<VideoEncoder>> encoders_;
+  std::vector<std::unique_ptr<VideoDecoder>> decoders_;
+};
+
+INSTANTIATE_TEST_SUITE_P(MultipleEncoders,
+                         VideoEncoderDecoderInstantiationTest,
+                         ::testing::Combine(::testing::Range(1, 4),
+                                            ::testing::Range(1, 2)));
+
+INSTANTIATE_TEST_SUITE_P(MultipleDecoders,
+                         VideoEncoderDecoderInstantiationTest,
+                         ::testing::Combine(::testing::Range(1, 2),
+                                            ::testing::Range(1, 9)));
+
+INSTANTIATE_TEST_SUITE_P(MultipleEncodersDecoders,
+                         VideoEncoderDecoderInstantiationTest,
+                         ::testing::Combine(::testing::Range(1, 4),
+                                            ::testing::Range(1, 9)));
+
+// TODO(brandtr): Check that the factories actually support the codecs before
+// trying to instantiate. Currently, we will just crash with a Java exception
+// if the factory does not support the codec.
+TEST_P(VideoEncoderDecoderInstantiationTest, DISABLED_InstantiateVp8Codecs) {
+  for (int i = 0; i < num_encoders_; ++i) {
+    std::unique_ptr<VideoEncoder> encoder =
+        encoder_factory_->CreateVideoEncoder(vp8_format_);
+    EXPECT_EQ(0, InitEncoder(kVideoCodecVP8, encoder.get()));
+    encoders_.emplace_back(std::move(encoder));
+  }
+
+  for (int i = 0; i < num_decoders_; ++i) {
+    std::unique_ptr<VideoDecoder> decoder =
+        decoder_factory_->CreateVideoDecoder(vp8_format_);
+    ASSERT_THAT(decoder, NotNull());
+    EXPECT_TRUE(decoder->Configure(DecoderSettings(kVideoCodecVP8)));
+    decoders_.emplace_back(std::move(decoder));
+  }
+}
+
+TEST_P(VideoEncoderDecoderInstantiationTest,
+       DISABLED_InstantiateH264CBPCodecs) {
+  for (int i = 0; i < num_encoders_; ++i) {
+    std::unique_ptr<VideoEncoder> encoder =
+        encoder_factory_->CreateVideoEncoder(h264cbp_format_);
+    EXPECT_EQ(0, InitEncoder(kVideoCodecH264, encoder.get()));
+    encoders_.emplace_back(std::move(encoder));
+  }
+
+  for (int i = 0; i < num_decoders_; ++i) {
+    std::unique_ptr<VideoDecoder> decoder =
+        decoder_factory_->CreateVideoDecoder(h264cbp_format_);
+    ASSERT_THAT(decoder, NotNull());
+    EXPECT_TRUE(decoder->Configure(DecoderSettings(kVideoCodecH264)));
+    decoders_.push_back(std::move(decoder));
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_av1.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_av1.cc
new file mode 100644
index 0000000000..9189f5abe5
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_av1.cc
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <vector>
+
+#include "api/test/create_videocodec_test_fixture.h"
+#include "api/test/video/function_video_encoder_factory.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "media/base/media_constants.h"
+#include "media/engine/internal_decoder_factory.h"
+#include "media/engine/internal_encoder_factory.h"
+#include "media/engine/simulcast_encoder_adapter.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+// Test clips settings.
+constexpr int kCifWidth = 352;
+constexpr int kCifHeight = 288;
+constexpr int kNumFramesLong = 300;
+
+VideoCodecTestFixture::Config CreateConfig(std::string filename) {
+  VideoCodecTestFixture::Config config;
+  config.filename = filename;
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = kNumFramesLong;
+  config.use_single_core = true;
+  return config;
+}
+
+TEST(VideoCodecTestAv1, HighBitrate) {
+  auto config = CreateConfig("foreman_cif");
+  config.SetCodecSettings(cricket::kAv1CodecName, 1, 1, 1, false, true, true,
+                          kCifWidth, kCifHeight);
+  config.codec_settings.SetScalabilityMode(ScalabilityMode::kL1T1);
+  config.num_frames = kNumFramesLong;
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {12, 1, 0, 1, 0.3, 0.1, 0, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{37, 34, 0.94, 0.91}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestAv1, VeryLowBitrate) {
+  auto config = CreateConfig("foreman_cif");
+  config.SetCodecSettings(cricket::kAv1CodecName, 1, 1, 1, false, true, true,
+                          kCifWidth, kCifHeight);
+  config.codec_settings.SetScalabilityMode(ScalabilityMode::kL1T1);
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{50, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {15, 8, 75, 2, 2, 2, 2, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{28, 24.8, 0.70, 0.55}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+#if !defined(WEBRTC_ANDROID)
+constexpr int kHdWidth = 1280;
+constexpr int kHdHeight = 720;
+TEST(VideoCodecTestAv1, Hd) {
+  auto config = CreateConfig("ConferenceMotion_1280_720_50");
+  config.SetCodecSettings(cricket::kAv1CodecName, 1, 1, 1, false, true, true,
+                          kHdWidth, kHdHeight);
+  config.codec_settings.SetScalabilityMode(ScalabilityMode::kL1T1);
+  config.num_frames = kNumFramesLong;
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{1000, 50, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {13, 3, 0, 1, 0.3, 0.1, 0, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {
+      {35.9, 31.5, 0.925, 0.865}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+#endif
+
+}  // namespace
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc
new file mode 100644
index 0000000000..126aa93ee8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+
+#include "api/test/videocodec_test_fixture.h"
+#include "api/video_codecs/video_codec.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/video_codec_settings.h"
+
+using ::testing::ElementsAre;
+
+namespace webrtc {
+namespace test {
+
+using Config = VideoCodecTestFixture::Config;
+
+namespace {
+const size_t kNumTemporalLayers = 2;
+}  // namespace
+
+TEST(Config, NumberOfCoresWithUseSingleCore) {
+  Config config;
+  config.use_single_core = true;
+  EXPECT_EQ(1u, config.NumberOfCores());
+}
+
+TEST(Config, NumberOfCoresWithoutUseSingleCore) {
+  Config config;
+  config.use_single_core = false;
+  EXPECT_GE(config.NumberOfCores(), 1u);
+}
+
+TEST(Config, NumberOfTemporalLayersIsOne) {
+  Config config;
+  webrtc::test::CodecSettings(kVideoCodecH264, &config.codec_settings);
+  EXPECT_EQ(1u, config.NumberOfTemporalLayers());
+}
+
+TEST(Config, NumberOfTemporalLayers_Vp8) {
+  Config config;
+  webrtc::test::CodecSettings(kVideoCodecVP8, &config.codec_settings);
+  config.codec_settings.VP8()->numberOfTemporalLayers = kNumTemporalLayers;
+  EXPECT_EQ(kNumTemporalLayers, config.NumberOfTemporalLayers());
+}
+
+TEST(Config, NumberOfTemporalLayers_Vp9) {
+  Config config;
+  webrtc::test::CodecSettings(kVideoCodecVP9, &config.codec_settings);
+  config.codec_settings.VP9()->numberOfTemporalLayers = kNumTemporalLayers;
+  EXPECT_EQ(kNumTemporalLayers, config.NumberOfTemporalLayers());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
new file mode 100644
index 0000000000..e56e8a92af
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
@@ -0,0 +1,860 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/videocodec_test_fixture_impl.h"
+
+#include <stdint.h>
+#include <stdio.h>
+
+#include <algorithm>
+#include <cmath>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/str_replace.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video_codecs/h264_profile_level_id.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_decoder_factory_template.h"
+#include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h"
+#include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h"
+#include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h"
+#include "api/video_codecs/video_decoder_factory_template_open_h264_adapter.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "api/video_codecs/video_encoder_factory_template.h"
+#include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h"
+#include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h"
+#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
+#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
+#include "common_video/h264/h264_common.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "modules/video_coding/codecs/vp9/svc_config.h"
+#include "modules/video_coding/utility/ivf_file_writer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/cpu_time.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/cpu_info.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_writer.h"
+#include "test/video_codec_settings.h"
+#include "video/config/simulcast.h"
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using VideoStatistics = VideoCodecTestStats::VideoStatistics;
+
+const int kBaseKeyFrameInterval = 3000;
+const double kBitratePriority = 1.0;
+const int kDefaultMaxFramerateFps = 30;
+const int kMaxQp = 56;
+
+void ConfigureSimulcast(VideoCodec* codec_settings) {
+  FieldTrialBasedConfig trials;
+  const std::vector<webrtc::VideoStream> streams = cricket::GetSimulcastConfig(
+      /*min_layer=*/1, codec_settings->numberOfSimulcastStreams,
+      codec_settings->width, codec_settings->height, kBitratePriority, kMaxQp,
+      /* is_screenshare = */ false, true, trials);
+
+  for (size_t i = 0; i < streams.size(); ++i) {
+    SimulcastStream* ss = &codec_settings->simulcastStream[i];
+    ss->width = static_cast<uint16_t>(streams[i].width);
+    ss->height = static_cast<uint16_t>(streams[i].height);
+    ss->numberOfTemporalLayers =
+        static_cast<unsigned char>(*streams[i].num_temporal_layers);
+    ss->maxBitrate = streams[i].max_bitrate_bps / 1000;
+    ss->targetBitrate = streams[i].target_bitrate_bps / 1000;
+    ss->minBitrate = streams[i].min_bitrate_bps / 1000;
+    ss->qpMax = streams[i].max_qp;
+    ss->active = true;
+  }
+}
+
+void ConfigureSvc(VideoCodec* codec_settings) {
+  RTC_CHECK_EQ(kVideoCodecVP9, codec_settings->codecType);
+
+  const std::vector<SpatialLayer> layers = GetSvcConfig(
+      codec_settings->width, codec_settings->height, kDefaultMaxFramerateFps,
+      /*first_active_layer=*/0, codec_settings->VP9()->numberOfSpatialLayers,
+      codec_settings->VP9()->numberOfTemporalLayers,
+      /* is_screen_sharing = */ false);
+  ASSERT_EQ(codec_settings->VP9()->numberOfSpatialLayers, layers.size())
+      << "GetSvcConfig returned fewer spatial layers than configured.";
+
+  for (size_t i = 0; i < layers.size(); ++i) {
+    codec_settings->spatialLayers[i] = layers[i];
+  }
+}
+
+std::string CodecSpecificToString(const VideoCodec& codec) {
+  char buf[1024];
+  rtc::SimpleStringBuilder ss(buf);
+  switch (codec.codecType) {
+    case kVideoCodecVP8:
+      ss << "\nnum_temporal_layers: "
+         << static_cast<int>(codec.VP8().numberOfTemporalLayers);
+      ss << "\ndenoising: " << codec.VP8().denoisingOn;
+      ss << "\nautomatic_resize: " << codec.VP8().automaticResizeOn;
+      ss << "\nkey_frame_interval: " << codec.VP8().keyFrameInterval;
+      break;
+    case kVideoCodecVP9:
+      ss << "\nnum_temporal_layers: "
+         << static_cast<int>(codec.VP9().numberOfTemporalLayers);
+      ss << "\nnum_spatial_layers: "
+         << static_cast<int>(codec.VP9().numberOfSpatialLayers);
+      ss << "\ndenoising: " << codec.VP9().denoisingOn;
+      ss << "\nkey_frame_interval: " << codec.VP9().keyFrameInterval;
+      ss << "\nadaptive_qp_mode: " << codec.VP9().adaptiveQpMode;
+      ss << "\nautomatic_resize: " << codec.VP9().automaticResizeOn;
+      ss << "\nflexible_mode: " << codec.VP9().flexibleMode;
+      break;
+    case kVideoCodecH264:
+      ss << "\nkey_frame_interval: " << codec.H264().keyFrameInterval;
+      ss << "\nnum_temporal_layers: "
+         << static_cast<int>(codec.H264().numberOfTemporalLayers);
+      break;
+    default:
+      break;
+  }
+  return ss.str();
+}
+
+bool RunEncodeInRealTime(const VideoCodecTestFixtureImpl::Config& config) {
+  if (config.measure_cpu || config.encode_in_real_time) {
+    return true;
+  }
+  return false;
+}
+
+std::string FilenameWithParams(
+    const VideoCodecTestFixtureImpl::Config& config) {
+  return config.filename + "_" + config.CodecName() + "_" +
+         std::to_string(config.codec_settings.startBitrate);
+}
+
+SdpVideoFormat CreateSdpVideoFormat(
+    const VideoCodecTestFixtureImpl::Config& config) {
+  if (config.codec_settings.codecType == kVideoCodecH264) {
+    const char* packetization_mode =
+        config.h264_codec_settings.packetization_mode ==
+                H264PacketizationMode::NonInterleaved
+            ? "1"
+            : "0";
+    SdpVideoFormat::Parameters codec_params = {
+        {cricket::kH264FmtpProfileLevelId,
+         *H264ProfileLevelIdToString(H264ProfileLevelId(
+             config.h264_codec_settings.profile, H264Level::kLevel3_1))},
+        {cricket::kH264FmtpPacketizationMode, packetization_mode},
+        {cricket::kH264FmtpLevelAsymmetryAllowed, "1"}};
+
+    return SdpVideoFormat(config.codec_name, codec_params);
+  } else if (config.codec_settings.codecType == kVideoCodecVP9) {
+    return SdpVideoFormat(config.codec_name, {{"profile-id", "0"}});
+  }
+
+  return SdpVideoFormat(config.codec_name);
+}
+
+}  // namespace
+
+VideoCodecTestFixtureImpl::Config::Config() = default;
+
+void VideoCodecTestFixtureImpl::Config::SetCodecSettings(
+    std::string codec_name,
+    size_t num_simulcast_streams,
+    size_t num_spatial_layers,
+    size_t num_temporal_layers,
+    bool denoising_on,
+    bool frame_dropper_on,
+    bool spatial_resize_on,
+    size_t width,
+    size_t height) {
+  this->codec_name = codec_name;
+  VideoCodecType codec_type = PayloadStringToCodecType(codec_name);
+  webrtc::test::CodecSettings(codec_type, &codec_settings);
+
+  // TODO(brandtr): Move the setting of `width` and `height` to the tests, and
+  // DCHECK that they are set before initializing the codec instead.
+  codec_settings.width = static_cast<uint16_t>(width);
+  codec_settings.height = static_cast<uint16_t>(height);
+
+  RTC_CHECK(num_simulcast_streams >= 1 &&
+            num_simulcast_streams <= kMaxSimulcastStreams);
+  RTC_CHECK(num_spatial_layers >= 1 && num_spatial_layers <= kMaxSpatialLayers);
+  RTC_CHECK(num_temporal_layers >= 1 &&
+            num_temporal_layers <= kMaxTemporalStreams);
+
+  // Simulcast is only available with VP8.
+  RTC_CHECK(num_simulcast_streams < 2 || codec_type == kVideoCodecVP8);
+
+  // Spatial scalability is only available with VP9.
+  RTC_CHECK(num_spatial_layers < 2 || codec_type == kVideoCodecVP9);
+
+  // Some base code requires numberOfSimulcastStreams to be set to zero
+  // when simulcast is not used.
+  codec_settings.numberOfSimulcastStreams =
+      num_simulcast_streams <= 1 ? 0
+                                 : static_cast<uint8_t>(num_simulcast_streams);
+
+  codec_settings.SetFrameDropEnabled(frame_dropper_on);
+  switch (codec_settings.codecType) {
+    case kVideoCodecVP8:
+      codec_settings.VP8()->numberOfTemporalLayers =
+          static_cast<uint8_t>(num_temporal_layers);
+      codec_settings.VP8()->denoisingOn = denoising_on;
+      codec_settings.VP8()->automaticResizeOn = spatial_resize_on;
+      codec_settings.VP8()->keyFrameInterval = kBaseKeyFrameInterval;
+      break;
+    case kVideoCodecVP9:
+      codec_settings.VP9()->numberOfTemporalLayers =
+          static_cast<uint8_t>(num_temporal_layers);
+      codec_settings.VP9()->denoisingOn = denoising_on;
+      codec_settings.VP9()->keyFrameInterval = kBaseKeyFrameInterval;
+      codec_settings.VP9()->automaticResizeOn = spatial_resize_on;
+      codec_settings.VP9()->numberOfSpatialLayers =
+          static_cast<uint8_t>(num_spatial_layers);
+      break;
+    case kVideoCodecAV1:
+      codec_settings.qpMax = 63;
+      break;
+    case kVideoCodecH264:
+      codec_settings.H264()->keyFrameInterval = kBaseKeyFrameInterval;
+      codec_settings.H264()->numberOfTemporalLayers =
+          static_cast<uint8_t>(num_temporal_layers);
+      break;
+    default:
+      break;
+  }
+
+  if (codec_settings.numberOfSimulcastStreams > 1) {
+    ConfigureSimulcast(&codec_settings);
+  } else if (codec_settings.codecType == kVideoCodecVP9 &&
+             codec_settings.VP9()->numberOfSpatialLayers > 1) {
+    ConfigureSvc(&codec_settings);
+  }
+}
+
+size_t VideoCodecTestFixtureImpl::Config::NumberOfCores() const {
+  return use_single_core ? 1 : CpuInfo::DetectNumberOfCores();
+}
+
+size_t VideoCodecTestFixtureImpl::Config::NumberOfTemporalLayers() const {
+  if (codec_settings.codecType == kVideoCodecVP8) {
+    return codec_settings.VP8().numberOfTemporalLayers;
+  } else if (codec_settings.codecType == kVideoCodecVP9) {
+    return codec_settings.VP9().numberOfTemporalLayers;
+  } else if (codec_settings.codecType == kVideoCodecH264) {
+    return codec_settings.H264().numberOfTemporalLayers;
+  } else {
+    return 1;
+  }
+}
+
+size_t VideoCodecTestFixtureImpl::Config::NumberOfSpatialLayers() const {
+  if (codec_settings.codecType == kVideoCodecVP9) {
+    return codec_settings.VP9().numberOfSpatialLayers;
+  } else {
+    return 1;
+  }
+}
+
+size_t VideoCodecTestFixtureImpl::Config::NumberOfSimulcastStreams() const {
+  return codec_settings.numberOfSimulcastStreams;
+}
+
+std::string VideoCodecTestFixtureImpl::Config::ToString() const {
+  std::string codec_type = CodecTypeToPayloadString(codec_settings.codecType);
+  rtc::StringBuilder ss;
+  ss << "test_name: " << test_name;
+  ss << "\nfilename: " << filename;
+  ss << "\nnum_frames: " << num_frames;
+  ss << "\nmax_payload_size_bytes: " << max_payload_size_bytes;
+  ss << "\ndecode: " << decode;
+  ss << "\nuse_single_core: " << use_single_core;
+  ss << "\nmeasure_cpu: " << measure_cpu;
+  ss << "\nnum_cores: " << NumberOfCores();
+  ss << "\ncodec_type: " << codec_type;
+  ss << "\n\n--> codec_settings";
+  ss << "\nwidth: " << codec_settings.width;
+  ss << "\nheight: " << codec_settings.height;
+  ss << "\nmax_framerate_fps: " << codec_settings.maxFramerate;
+  ss << "\nstart_bitrate_kbps: " << codec_settings.startBitrate;
+  ss << "\nmax_bitrate_kbps: " << codec_settings.maxBitrate;
+  ss << "\nmin_bitrate_kbps: " << codec_settings.minBitrate;
+  ss << "\nmax_qp: " << codec_settings.qpMax;
+  ss << "\nnum_simulcast_streams: "
+     << static_cast<int>(codec_settings.numberOfSimulcastStreams);
+  ss << "\n\n--> codec_settings." << codec_type;
+  ss << "complexity: "
+     << static_cast<int>(codec_settings.GetVideoEncoderComplexity());
+  ss << "\nframe_dropping: " << codec_settings.GetFrameDropEnabled();
+  ss << "\n" << CodecSpecificToString(codec_settings);
+  if (codec_settings.numberOfSimulcastStreams > 1) {
+    for (int i = 0; i < codec_settings.numberOfSimulcastStreams; ++i) {
+      ss << "\n\n--> codec_settings.simulcastStream[" << i << "]";
+      const SimulcastStream& simulcast_stream =
+          codec_settings.simulcastStream[i];
+      ss << "\nwidth: " << simulcast_stream.width;
+      ss << "\nheight: " << simulcast_stream.height;
+      ss << "\nnum_temporal_layers: "
+         << static_cast<int>(simulcast_stream.numberOfTemporalLayers);
+      ss << "\nmin_bitrate_kbps: " << simulcast_stream.minBitrate;
+      ss << "\ntarget_bitrate_kbps: " << simulcast_stream.targetBitrate;
+      ss << "\nmax_bitrate_kbps: " << simulcast_stream.maxBitrate;
+      ss << "\nmax_qp: " << simulcast_stream.qpMax;
+      ss << "\nactive: " << simulcast_stream.active;
+    }
+  }
+  ss << "\n";
+  return ss.Release();
+}
+
+std::string VideoCodecTestFixtureImpl::Config::CodecName() const {
+  std::string name = codec_name;
+  if (name.empty()) {
+    name = CodecTypeToPayloadString(codec_settings.codecType);
+  }
+  if (codec_settings.codecType == kVideoCodecH264) {
+    if (h264_codec_settings.profile == H264Profile::kProfileConstrainedHigh) {
+      return name + "-CHP";
+    } else {
+      RTC_DCHECK_EQ(h264_codec_settings.profile,
+                    H264Profile::kProfileConstrainedBaseline);
+      return name + "-CBP";
+    }
+  }
+  return name;
+}
+
+// TODO(kthelgason): Move this out of the test fixture impl and
+// make available as a shared utility class.
+void VideoCodecTestFixtureImpl::H264KeyframeChecker::CheckEncodedFrame(
+    webrtc::VideoCodecType codec,
+    const EncodedImage& encoded_frame) const {
+  EXPECT_EQ(kVideoCodecH264, codec);
+  bool contains_sps = false;
+  bool contains_pps = false;
+  bool contains_idr = false;
+  const std::vector<webrtc::H264::NaluIndex> nalu_indices =
+      webrtc::H264::FindNaluIndices(encoded_frame.data(), encoded_frame.size());
+  for (const webrtc::H264::NaluIndex& index : nalu_indices) {
+    webrtc::H264::NaluType nalu_type = webrtc::H264::ParseNaluType(
+        encoded_frame.data()[index.payload_start_offset]);
+    if (nalu_type == webrtc::H264::NaluType::kSps) {
+      contains_sps = true;
+    } else if (nalu_type == webrtc::H264::NaluType::kPps) {
+      contains_pps = true;
+    } else if (nalu_type == webrtc::H264::NaluType::kIdr) {
+      contains_idr = true;
+    }
+  }
+  if (encoded_frame._frameType == VideoFrameType::kVideoFrameKey) {
+    EXPECT_TRUE(contains_sps) << "Keyframe should contain SPS.";
+    EXPECT_TRUE(contains_pps) << "Keyframe should contain PPS.";
+    EXPECT_TRUE(contains_idr) << "Keyframe should contain IDR.";
+  } else if (encoded_frame._frameType == VideoFrameType::kVideoFrameDelta) {
+    EXPECT_FALSE(contains_sps) << "Delta frame should not contain SPS.";
+    EXPECT_FALSE(contains_pps) << "Delta frame should not contain PPS.";
+    EXPECT_FALSE(contains_idr) << "Delta frame should not contain IDR.";
+  } else {
+    RTC_DCHECK_NOTREACHED();
+  }
+}
+
+class VideoCodecTestFixtureImpl::CpuProcessTime final {
+ public:
+  explicit CpuProcessTime(const Config& config) : config_(config) {}
+  ~CpuProcessTime() {}
+
+  void Start() {
+    if (config_.measure_cpu) {
+      cpu_time_ -= rtc::GetProcessCpuTimeNanos();
+      wallclock_time_ -= rtc::SystemTimeNanos();
+    }
+  }
+  void Stop() {
+    if (config_.measure_cpu) {
+      cpu_time_ += rtc::GetProcessCpuTimeNanos();
+      wallclock_time_ += rtc::SystemTimeNanos();
+    }
+  }
+  void Print() const {
+    if (config_.measure_cpu) {
+      RTC_LOG(LS_INFO) << "cpu_usage_percent: "
+                       << GetUsagePercent() / config_.NumberOfCores();
+    }
+  }
+
+ private:
+  double GetUsagePercent() const {
+    return static_cast<double>(cpu_time_) / wallclock_time_ * 100.0;
+  }
+
+  const Config config_;
+  int64_t cpu_time_ = 0;
+  int64_t wallclock_time_ = 0;
+};
+
+VideoCodecTestFixtureImpl::VideoCodecTestFixtureImpl(Config config)
+    : encoder_factory_(std::make_unique<webrtc::VideoEncoderFactoryTemplate<
+                           webrtc::LibvpxVp8EncoderTemplateAdapter,
+                           webrtc::LibvpxVp9EncoderTemplateAdapter,
+                           webrtc::OpenH264EncoderTemplateAdapter,
+                           webrtc::LibaomAv1EncoderTemplateAdapter>>()),
+      decoder_factory_(std::make_unique<webrtc::VideoDecoderFactoryTemplate<
+                           webrtc::LibvpxVp8DecoderTemplateAdapter,
+                           webrtc::LibvpxVp9DecoderTemplateAdapter,
+                           webrtc::OpenH264DecoderTemplateAdapter,
+                           webrtc::Dav1dDecoderTemplateAdapter>>()),
+      config_(config) {}
+
+VideoCodecTestFixtureImpl::VideoCodecTestFixtureImpl(
+    Config config,
+    std::unique_ptr<VideoDecoderFactory> decoder_factory,
+    std::unique_ptr<VideoEncoderFactory> encoder_factory)
+    : encoder_factory_(std::move(encoder_factory)),
+      decoder_factory_(std::move(decoder_factory)),
+      config_(config) {}
+
+VideoCodecTestFixtureImpl::~VideoCodecTestFixtureImpl() = default;
+
+// Processes all frames in the clip and verifies the result.
+void VideoCodecTestFixtureImpl::RunTest(
+    const std::vector<RateProfile>& rate_profiles,
+    const std::vector<RateControlThresholds>* rc_thresholds,
+    const std::vector<QualityThresholds>* quality_thresholds,
+    const BitstreamThresholds* bs_thresholds) {
+  RTC_DCHECK(!rate_profiles.empty());
+
+  // To emulate operation on a production VideoStreamEncoder, we call the
+  // codecs on a task queue.
+  TaskQueueForTest task_queue("VidProc TQ");
+
+  bool is_setup_succeeded = SetUpAndInitObjects(
+      &task_queue, rate_profiles[0].target_kbps, rate_profiles[0].input_fps);
+  EXPECT_TRUE(is_setup_succeeded);
+  if (!is_setup_succeeded) {
+    ReleaseAndCloseObjects(&task_queue);
+    return;
+  }
+
+  PrintSettings(&task_queue);
+  ProcessAllFrames(&task_queue, rate_profiles);
+  ReleaseAndCloseObjects(&task_queue);
+
+  AnalyzeAllFrames(rate_profiles, rc_thresholds, quality_thresholds,
+                   bs_thresholds);
+}
+
+void VideoCodecTestFixtureImpl::ProcessAllFrames(
+    TaskQueueForTest* task_queue,
+    const std::vector<RateProfile>& rate_profiles) {
+  // Set initial rates.
+  auto rate_profile = rate_profiles.begin();
+  task_queue->PostTask([this, rate_profile] {
+    processor_->SetRates(rate_profile->target_kbps, rate_profile->input_fps);
+  });
+
+  cpu_process_time_->Start();
+
+  for (size_t frame_num = 0; frame_num < config_.num_frames; ++frame_num) {
+    auto next_rate_profile = std::next(rate_profile);
+    if (next_rate_profile != rate_profiles.end() &&
+        frame_num == next_rate_profile->frame_num) {
+      rate_profile = next_rate_profile;
+      task_queue->PostTask([this, rate_profile] {
+        processor_->SetRates(rate_profile->target_kbps,
+                             rate_profile->input_fps);
+      });
+    }
+
+    task_queue->PostTask([this] { processor_->ProcessFrame(); });
+
+    if (RunEncodeInRealTime(config_)) {
+      // Roughly pace the frames.
+      const int frame_duration_ms =
+          std::ceil(rtc::kNumMillisecsPerSec / rate_profile->input_fps);
+      SleepMs(frame_duration_ms);
+    }
+  }
+
+  task_queue->PostTask([this] { processor_->Finalize(); });
+
+  // Wait until we know that the last frame has been sent for encode.
+  task_queue->SendTask([] {});
+
+  // Give the VideoProcessor pipeline some time to process the last frame,
+  // and then release the codecs.
+  SleepMs(1 * rtc::kNumMillisecsPerSec);
+  cpu_process_time_->Stop();
+}
+
+void VideoCodecTestFixtureImpl::AnalyzeAllFrames(
+    const std::vector<RateProfile>& rate_profiles,
+    const std::vector<RateControlThresholds>* rc_thresholds,
+    const std::vector<QualityThresholds>* quality_thresholds,
+    const BitstreamThresholds* bs_thresholds) {
+
+  for (size_t rate_profile_idx = 0; rate_profile_idx < rate_profiles.size();
+       ++rate_profile_idx) {
+    const size_t first_frame_num = rate_profiles[rate_profile_idx].frame_num;
+    const size_t last_frame_num =
+        rate_profile_idx + 1 < rate_profiles.size()
+            ? rate_profiles[rate_profile_idx + 1].frame_num - 1
+            : config_.num_frames - 1;
+    RTC_CHECK(last_frame_num >= first_frame_num);
+
+    VideoStatistics send_stat = stats_.SliceAndCalcAggregatedVideoStatistic(
+        first_frame_num, last_frame_num);
+    RTC_LOG(LS_INFO) << "==> Send stats";
+    RTC_LOG(LS_INFO) << send_stat.ToString("send_") << "\n";
+
+    std::vector<VideoStatistics> layer_stats =
+        stats_.SliceAndCalcLayerVideoStatistic(first_frame_num, last_frame_num);
+    RTC_LOG(LS_INFO) << "==> Receive stats";
+    for (const auto& layer_stat : layer_stats) {
+      RTC_LOG(LS_INFO) << layer_stat.ToString("recv_") << "\n";
+
+      // For perf dashboard.
+      char modifier_buf[256];
+      rtc::SimpleStringBuilder modifier(modifier_buf);
+      modifier << "_r" << rate_profile_idx << "_sl" << layer_stat.spatial_idx;
+
+      auto PrintResultHelper = [&modifier, this](
+                                   absl::string_view measurement, double value,
+                                   Unit unit,
+                                   absl::string_view non_standard_unit_suffix,
+                                   ImprovementDirection improvement_direction) {
+        rtc::StringBuilder metric_name(measurement);
+        metric_name << modifier.str() << non_standard_unit_suffix;
+        GetGlobalMetricsLogger()->LogSingleValueMetric(
+            metric_name.str(), config_.test_name, value, unit,
+            improvement_direction);
+      };
+
+      if (layer_stat.temporal_idx == config_.NumberOfTemporalLayers() - 1) {
+        PrintResultHelper("enc_speed", layer_stat.enc_speed_fps,
+                          Unit::kUnitless, /*non_standard_unit_suffix=*/"_fps",
+                          ImprovementDirection::kBiggerIsBetter);
+        PrintResultHelper("avg_key_frame_size",
+                          layer_stat.avg_key_frame_size_bytes, Unit::kBytes,
+                          /*non_standard_unit_suffix=*/"",
+                          ImprovementDirection::kNeitherIsBetter);
+        PrintResultHelper("num_key_frames", layer_stat.num_key_frames,
+                          Unit::kCount,
+                          /*non_standard_unit_suffix=*/"",
+                          ImprovementDirection::kNeitherIsBetter);
+        printf("\n");
+      }
+
+      modifier << "tl" << layer_stat.temporal_idx;
+      PrintResultHelper("dec_speed", layer_stat.dec_speed_fps, Unit::kUnitless,
+                        /*non_standard_unit_suffix=*/"_fps",
+                        ImprovementDirection::kBiggerIsBetter);
+      PrintResultHelper("avg_delta_frame_size",
+                        layer_stat.avg_delta_frame_size_bytes, Unit::kBytes,
+                        /*non_standard_unit_suffix=*/"",
+                        ImprovementDirection::kNeitherIsBetter);
+      PrintResultHelper("bitrate", layer_stat.bitrate_kbps,
+                        Unit::kKilobitsPerSecond,
+                        /*non_standard_unit_suffix=*/"",
+                        ImprovementDirection::kNeitherIsBetter);
+      PrintResultHelper("framerate", layer_stat.framerate_fps, Unit::kUnitless,
+                        /*non_standard_unit_suffix=*/"_fps",
+                        ImprovementDirection::kNeitherIsBetter);
+      PrintResultHelper("avg_psnr_y", layer_stat.avg_psnr_y, Unit::kUnitless,
+                        /*non_standard_unit_suffix=*/"_dB",
+                        ImprovementDirection::kBiggerIsBetter);
+      PrintResultHelper("avg_psnr_u", layer_stat.avg_psnr_u, Unit::kUnitless,
+                        /*non_standard_unit_suffix=*/"_dB",
+                        ImprovementDirection::kBiggerIsBetter);
+      PrintResultHelper("avg_psnr_v", layer_stat.avg_psnr_v, Unit::kUnitless,
+                        /*non_standard_unit_suffix=*/"_dB",
+                        ImprovementDirection::kBiggerIsBetter);
+      PrintResultHelper("min_psnr_yuv", layer_stat.min_psnr, Unit::kUnitless,
+                        /*non_standard_unit_suffix=*/"_dB",
+                        ImprovementDirection::kBiggerIsBetter);
+      PrintResultHelper("avg_qp", layer_stat.avg_qp, Unit::kUnitless,
+                        /*non_standard_unit_suffix=*/"",
+                        ImprovementDirection::kSmallerIsBetter);
+      printf("\n");
+      if (layer_stat.temporal_idx == config_.NumberOfTemporalLayers() - 1) {
+        printf("\n");
+      }
+    }
+
+    const RateControlThresholds* rc_threshold =
+        rc_thresholds ? &(*rc_thresholds)[rate_profile_idx] : nullptr;
+    const QualityThresholds* quality_threshold =
+        quality_thresholds ? &(*quality_thresholds)[rate_profile_idx] : nullptr;
+
+    VerifyVideoStatistic(send_stat, rc_threshold, quality_threshold,
+                         bs_thresholds,
+                         rate_profiles[rate_profile_idx].target_kbps,
+                         rate_profiles[rate_profile_idx].input_fps);
+  }
+
+  if (config_.print_frame_level_stats) {
+    RTC_LOG(LS_INFO) << "==> Frame stats";
+    std::vector<VideoCodecTestStats::FrameStatistics> frame_stats =
+        stats_.GetFrameStatistics();
+    for (const auto& frame_stat : frame_stats) {
+      RTC_LOG(LS_INFO) << frame_stat.ToString();
+    }
+  }
+
+  cpu_process_time_->Print();
+}
+
+void VideoCodecTestFixtureImpl::VerifyVideoStatistic(
+    const VideoStatistics& video_stat,
+    const RateControlThresholds* rc_thresholds,
+    const QualityThresholds* quality_thresholds,
+    const BitstreamThresholds* bs_thresholds,
+    size_t target_bitrate_kbps,
+    double input_framerate_fps) {
+  if (rc_thresholds) {
+    const float bitrate_mismatch_percent =
+        100 * std::fabs(1.0f * video_stat.bitrate_kbps - target_bitrate_kbps) /
+        target_bitrate_kbps;
+    const float framerate_mismatch_percent =
+        100 * std::fabs(video_stat.framerate_fps - input_framerate_fps) /
+        input_framerate_fps;
+    EXPECT_LE(bitrate_mismatch_percent,
+              rc_thresholds->max_avg_bitrate_mismatch_percent);
+    EXPECT_LE(video_stat.time_to_reach_target_bitrate_sec,
+              rc_thresholds->max_time_to_reach_target_bitrate_sec);
+    EXPECT_LE(framerate_mismatch_percent,
+              rc_thresholds->max_avg_framerate_mismatch_percent);
+    EXPECT_LE(video_stat.avg_delay_sec,
+              rc_thresholds->max_avg_buffer_level_sec);
+    EXPECT_LE(video_stat.max_key_frame_delay_sec,
+              rc_thresholds->max_max_key_frame_delay_sec);
+    EXPECT_LE(video_stat.max_delta_frame_delay_sec,
+              rc_thresholds->max_max_delta_frame_delay_sec);
+    EXPECT_LE(video_stat.num_spatial_resizes,
+              rc_thresholds->max_num_spatial_resizes);
+    EXPECT_LE(video_stat.num_key_frames, rc_thresholds->max_num_key_frames);
+  }
+
+  if (quality_thresholds) {
+    EXPECT_GT(video_stat.avg_psnr, quality_thresholds->min_avg_psnr);
+    EXPECT_GT(video_stat.min_psnr, quality_thresholds->min_min_psnr);
+
+    // SSIM calculation is not optimized and thus it is disabled in real-time
+    // mode.
+    if (!config_.encode_in_real_time) {
+      EXPECT_GT(video_stat.avg_ssim, quality_thresholds->min_avg_ssim);
+      EXPECT_GT(video_stat.min_ssim, quality_thresholds->min_min_ssim);
+    }
+  }
+
+  if (bs_thresholds) {
+    EXPECT_LE(video_stat.max_nalu_size_bytes,
+              bs_thresholds->max_max_nalu_size_bytes);
+  }
+}
+
+bool VideoCodecTestFixtureImpl::CreateEncoderAndDecoder() {
+  SdpVideoFormat encoder_format(CreateSdpVideoFormat(config_));
+  SdpVideoFormat decoder_format = encoder_format;
+
+  // Override encoder and decoder formats with explicitly provided ones.
+  if (config_.encoder_format) {
+    RTC_DCHECK_EQ(config_.encoder_format->name, config_.codec_name);
+    encoder_format = *config_.encoder_format;
+  }
+
+  if (config_.decoder_format) {
+    RTC_DCHECK_EQ(config_.decoder_format->name, config_.codec_name);
+    decoder_format = *config_.decoder_format;
+  }
+
+  encoder_ = encoder_factory_->CreateVideoEncoder(encoder_format);
+  EXPECT_TRUE(encoder_) << "Encoder not successfully created.";
+  if (encoder_ == nullptr) {
+    return false;
+  }
+
+  const size_t num_simulcast_or_spatial_layers = std::max(
+      config_.NumberOfSimulcastStreams(), config_.NumberOfSpatialLayers());
+  for (size_t i = 0; i < num_simulcast_or_spatial_layers; ++i) {
+    std::unique_ptr<VideoDecoder> decoder =
+        decoder_factory_->CreateVideoDecoder(decoder_format);
+    EXPECT_TRUE(decoder) << "Decoder not successfully created.";
+    if (decoder == nullptr) {
+      return false;
+    }
+    decoders_.push_back(std::move(decoder));
+  }
+
+  return true;
+}
+
+void VideoCodecTestFixtureImpl::DestroyEncoderAndDecoder() {
+  decoders_.clear();
+  encoder_.reset();
+}
+
+VideoCodecTestStats& VideoCodecTestFixtureImpl::GetStats() {
+  return stats_;
+}
+
+bool VideoCodecTestFixtureImpl::SetUpAndInitObjects(
+    TaskQueueForTest* task_queue,
+    size_t initial_bitrate_kbps,
+    double initial_framerate_fps) {
+  config_.codec_settings.minBitrate = 0;
+  config_.codec_settings.startBitrate = static_cast<int>(initial_bitrate_kbps);
+  config_.codec_settings.maxFramerate = std::ceil(initial_framerate_fps);
+
+  int clip_width = config_.clip_width.value_or(config_.codec_settings.width);
+  int clip_height = config_.clip_height.value_or(config_.codec_settings.height);
+
+  // Create file objects for quality analysis.
+  source_frame_reader_ = CreateYuvFrameReader(
+      config_.filepath,
+      Resolution({.width = clip_width, .height = clip_height}),
+      YuvFrameReaderImpl::RepeatMode::kPingPong);
+
+  RTC_DCHECK(encoded_frame_writers_.empty());
+  RTC_DCHECK(decoded_frame_writers_.empty());
+
+  stats_.Clear();
+
+  cpu_process_time_.reset(new CpuProcessTime(config_));
+
+  bool is_codec_created = false;
+  task_queue->SendTask([this, &is_codec_created]() {
+    is_codec_created = CreateEncoderAndDecoder();
+  });
+
+  if (!is_codec_created) {
+    return false;
+  }
+
+  if (config_.visualization_params.save_encoded_ivf ||
+      config_.visualization_params.save_decoded_y4m) {
+    std::string encoder_name = GetCodecName(task_queue, /*is_encoder=*/true);
+    encoder_name = absl::StrReplaceAll(encoder_name, {{":", ""}, {" ", "-"}});
+
+    const size_t num_simulcast_or_spatial_layers = std::max(
+        config_.NumberOfSimulcastStreams(), config_.NumberOfSpatialLayers());
+    const size_t num_temporal_layers = config_.NumberOfTemporalLayers();
+    for (size_t simulcast_svc_idx = 0;
+         simulcast_svc_idx < num_simulcast_or_spatial_layers;
+         ++simulcast_svc_idx) {
+      const std::string output_filename_base =
+          JoinFilename(config_.output_path,
+                       FilenameWithParams(config_) + "_" + encoder_name +
+                           "_sl" + std::to_string(simulcast_svc_idx));
+
+      if (config_.visualization_params.save_encoded_ivf) {
+        for (size_t temporal_idx = 0; temporal_idx < num_temporal_layers;
+             ++temporal_idx) {
+          const std::string output_file_path = output_filename_base + "tl" +
+                                               std::to_string(temporal_idx) +
+                                               ".ivf";
+          FileWrapper ivf_file = FileWrapper::OpenWriteOnly(output_file_path);
+
+          const VideoProcessor::LayerKey layer_key(simulcast_svc_idx,
+                                                   temporal_idx);
+          encoded_frame_writers_[layer_key] =
+              IvfFileWriter::Wrap(std::move(ivf_file), /*byte_limit=*/0);
+        }
+      }
+
+      if (config_.visualization_params.save_decoded_y4m) {
+        FrameWriter* decoded_frame_writer = new Y4mFrameWriterImpl(
+            output_filename_base + ".y4m", config_.codec_settings.width,
+            config_.codec_settings.height, config_.codec_settings.maxFramerate);
+        EXPECT_TRUE(decoded_frame_writer->Init());
+        decoded_frame_writers_.push_back(
+            std::unique_ptr<FrameWriter>(decoded_frame_writer));
+      }
+    }
+  }
+
+  task_queue->SendTask(
+      [this]() {
+        processor_ = std::make_unique<VideoProcessor>(
+            encoder_.get(), &decoders_, source_frame_reader_.get(), config_,
+            &stats_, &encoded_frame_writers_,
+            decoded_frame_writers_.empty() ? nullptr : &decoded_frame_writers_);
+      });
+  return true;
+}
+
+void VideoCodecTestFixtureImpl::ReleaseAndCloseObjects(
+    TaskQueueForTest* task_queue) {
+  task_queue->SendTask([this]() {
+    processor_.reset();
+    // The VideoProcessor must be destroyed before the codecs.
+    DestroyEncoderAndDecoder();
+  });
+
+  source_frame_reader_.reset();
+
+  // Close visualization files.
+  for (auto& encoded_frame_writer : encoded_frame_writers_) {
+    EXPECT_TRUE(encoded_frame_writer.second->Close());
+  }
+  encoded_frame_writers_.clear();
+  for (auto& decoded_frame_writer : decoded_frame_writers_) {
+    decoded_frame_writer->Close();
+  }
+  decoded_frame_writers_.clear();
+}
+
+std::string VideoCodecTestFixtureImpl::GetCodecName(
+    TaskQueueForTest* task_queue,
+    bool is_encoder) const {
+  std::string codec_name;
+  task_queue->SendTask([this, is_encoder, &codec_name] {
+    if (is_encoder) {
+      codec_name = encoder_->GetEncoderInfo().implementation_name;
+    } else {
+      codec_name = decoders_.at(0)->ImplementationName();
+    }
+  });
+  return codec_name;
+}
+
+void VideoCodecTestFixtureImpl::PrintSettings(
+    TaskQueueForTest* task_queue) const {
+  RTC_LOG(LS_INFO) << "==> Config";
+  RTC_LOG(LS_INFO) << config_.ToString();
+
+  RTC_LOG(LS_INFO) << "==> Codec names";
+  RTC_LOG(LS_INFO) << "enc_impl_name: "
+                   << GetCodecName(task_queue, /*is_encoder=*/true);
+  RTC_LOG(LS_INFO) << "dec_impl_name: "
+                   << GetCodecName(task_queue, /*is_encoder=*/false);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.h b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.h
new file mode 100644
index 0000000000..005b7c0a8e
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.h
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_FIXTURE_IMPL_H_
+#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_FIXTURE_IMPL_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/test/videocodec_test_fixture.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "common_video/h264/h264_common.h"
+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
+#include "modules/video_coding/codecs/test/videoprocessor.h"
+#include "modules/video_coding/utility/ivf_file_writer.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/testsupport/frame_reader.h"
+#include "test/testsupport/frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+// Integration test for video processor. It does rate control and frame quality
+// analysis using frame statistics collected by video processor and logs the
+// results. If thresholds are specified it checks that corresponding metrics
+// are in desirable range.
+class VideoCodecTestFixtureImpl : public VideoCodecTestFixture {
+  // Verifies that all H.264 keyframes contain SPS/PPS/IDR NALUs.
+ public:
+  class H264KeyframeChecker : public EncodedFrameChecker {
+   public:
+    void CheckEncodedFrame(webrtc::VideoCodecType codec,
+                           const EncodedImage& encoded_frame) const override;
+  };
+
+  explicit VideoCodecTestFixtureImpl(Config config);
+  VideoCodecTestFixtureImpl(
+      Config config,
+      std::unique_ptr<VideoDecoderFactory> decoder_factory,
+      std::unique_ptr<VideoEncoderFactory> encoder_factory);
+  ~VideoCodecTestFixtureImpl() override;
+
+  void RunTest(const std::vector<RateProfile>& rate_profiles,
+               const std::vector<RateControlThresholds>* rc_thresholds,
+               const std::vector<QualityThresholds>* quality_thresholds,
+               const BitstreamThresholds* bs_thresholds) override;
+
+  VideoCodecTestStats& GetStats() override;
+
+ private:
+  class CpuProcessTime;
+
+  bool CreateEncoderAndDecoder();
+  void DestroyEncoderAndDecoder();
+  bool SetUpAndInitObjects(TaskQueueForTest* task_queue,
+                           size_t initial_bitrate_kbps,
+                           double initial_framerate_fps);
+  void ReleaseAndCloseObjects(TaskQueueForTest* task_queue);
+
+  void ProcessAllFrames(TaskQueueForTest* task_queue,
+                        const std::vector<RateProfile>& rate_profiles);
+  void AnalyzeAllFrames(
+      const std::vector<RateProfile>& rate_profiles,
+      const std::vector<RateControlThresholds>* rc_thresholds,
+      const std::vector<QualityThresholds>* quality_thresholds,
+      const BitstreamThresholds* bs_thresholds);
+
+  void VerifyVideoStatistic(
+      const VideoCodecTestStats::VideoStatistics& video_stat,
+      const RateControlThresholds* rc_thresholds,
+      const QualityThresholds* quality_thresholds,
+      const BitstreamThresholds* bs_thresholds,
+      size_t target_bitrate_kbps,
+      double input_framerate_fps);
+
+  std::string GetCodecName(TaskQueueForTest* task_queue, bool is_encoder) const;
+  void PrintSettings(TaskQueueForTest* task_queue) const;
+
+  // Codecs.
+  const std::unique_ptr<VideoEncoderFactory> encoder_factory_;
+  std::unique_ptr<VideoEncoder> encoder_;
+  const std::unique_ptr<VideoDecoderFactory> decoder_factory_;
+  VideoProcessor::VideoDecoderList decoders_;
+
+  // Helper objects.
+  Config config_;
+  VideoCodecTestStatsImpl stats_;
+  std::unique_ptr<FrameReader> source_frame_reader_;
+  VideoProcessor::IvfFileWriterMap encoded_frame_writers_;
+  VideoProcessor::FrameWriterList decoded_frame_writers_;
+  std::unique_ptr<VideoProcessor> processor_;
+  std::unique_ptr<CpuProcessTime> cpu_process_time_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_FIXTURE_IMPL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
new file mode 100644
index 0000000000..062375bd60
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
@@ -0,0 +1,465 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <vector>
+
+#include "api/test/create_videocodec_test_fixture.h"
+#include "api/test/video/function_video_encoder_factory.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "media/base/media_constants.h"
+#include "media/engine/internal_decoder_factory.h"
+#include "media/engine/internal_encoder_factory.h"
+#include "media/engine/simulcast_encoder_adapter.h"
+#include "modules/video_coding/utility/vp8_header_parser.h"
+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+
+using VideoStatistics = VideoCodecTestStats::VideoStatistics;
+
+namespace {
+// Codec settings.
+const int kCifWidth = 352;
+const int kCifHeight = 288;
+const int kNumFramesShort = 100;
+const int kNumFramesLong = 300;
+const size_t kBitrateRdPerfKbps[] = {100,  200,  300,  400,  500,  600,
+                                     700,  800,  1000, 1250, 1400, 1600,
+                                     1800, 2000, 2200, 2500};
+const size_t kNumFirstFramesToSkipAtRdPerfAnalysis = 60;
+
+class QpFrameChecker : public VideoCodecTestFixture::EncodedFrameChecker {
+ public:
+  void CheckEncodedFrame(webrtc::VideoCodecType codec,
+                         const EncodedImage& encoded_frame) const override {
+    int qp;
+    if (codec == kVideoCodecVP8) {
+      EXPECT_TRUE(vp8::GetQp(encoded_frame.data(), encoded_frame.size(), &qp));
+    } else if (codec == kVideoCodecVP9) {
+      EXPECT_TRUE(vp9::GetQp(encoded_frame.data(), encoded_frame.size(), &qp));
+    } else {
+      RTC_DCHECK_NOTREACHED();
+    }
+    EXPECT_EQ(encoded_frame.qp_, qp) << "Encoder QP != parsed bitstream QP.";
+  }
+};
+
+VideoCodecTestFixture::Config CreateConfig() {
+  VideoCodecTestFixture::Config config;
+  config.filename = "foreman_cif";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = kNumFramesLong;
+  config.use_single_core = true;
+  return config;
+}
+
+void PrintRdPerf(std::map<size_t, std::vector<VideoStatistics>> rd_stats) {
+  printf("--> Summary\n");
+  printf("%11s %5s %6s %11s %12s %11s %13s %13s %5s %7s %7s %7s %13s %13s\n",
+         "uplink_kbps", "width", "height", "spatial_idx", "temporal_idx",
+         "target_kbps", "downlink_kbps", "framerate_fps", "psnr", "psnr_y",
+         "psnr_u", "psnr_v", "enc_speed_fps", "dec_speed_fps");
+  for (const auto& rd_stat : rd_stats) {
+    const size_t bitrate_kbps = rd_stat.first;
+    for (const auto& layer_stat : rd_stat.second) {
+      printf(
+          "%11zu %5zu %6zu %11zu %12zu %11zu %13zu %13.2f %5.2f %7.2f %7.2f "
+          "%7.2f"
+          "%13.2f %13.2f\n",
+          bitrate_kbps, layer_stat.width, layer_stat.height,
+          layer_stat.spatial_idx, layer_stat.temporal_idx,
+          layer_stat.target_bitrate_kbps, layer_stat.bitrate_kbps,
+          layer_stat.framerate_fps, layer_stat.avg_psnr, layer_stat.avg_psnr_y,
+          layer_stat.avg_psnr_u, layer_stat.avg_psnr_v,
+          layer_stat.enc_speed_fps, layer_stat.dec_speed_fps);
+    }
+  }
+}
+}  // namespace
+
+#if defined(RTC_ENABLE_VP9)
+TEST(VideoCodecTestLibvpx, HighBitrateVP9) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, false, true, false,
+                          kCifWidth, kCifHeight);
+  config.num_frames = kNumFramesShort;
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 0, 1, 0.3, 0.1, 0, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{37, 36, 0.94, 0.92}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestLibvpx, ChangeBitrateVP9) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, false, true, false,
+                          kCifWidth, kCifHeight);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {
+      {200, 30, 0},  // target_kbps, input_fps, frame_num
+      {700, 30, 100},
+      {500, 30, 200}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 2, 0, 1, 0.5, 0.1, 0, 1},
+      {15, 3, 0, 1, 0.5, 0.1, 0, 0},
+      {11, 2, 0, 1, 0.5, 0.1, 0, 0}};
+
+  std::vector<QualityThresholds> quality_thresholds = {
+      {34, 33, 0.90, 0.88}, {38, 35, 0.95, 0.91}, {35, 34, 0.93, 0.90}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestLibvpx, ChangeFramerateVP9) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, false, true, false,
+                          kCifWidth, kCifHeight);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {
+      {100, 24, 0},  // target_kbps, input_fps, frame_num
+      {100, 15, 100},
+      {100, 10, 200}};
+
+  // Framerate mismatch should be lower for lower framerate.
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {10, 2, 40, 1, 0.5, 0.2, 0, 1},
+      {8, 2, 5, 1, 0.5, 0.2, 0, 0},
+      {5, 2, 0, 1, 0.5, 0.3, 0, 0}};
+
+  // Quality should be higher for lower framerates for the same content.
+  std::vector<QualityThresholds> quality_thresholds = {
+      {33, 32, 0.88, 0.86}, {33.5, 32, 0.90, 0.86}, {33.5, 31.5, 0.90, 0.85}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestLibvpx, DenoiserOnVP9) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, true, true, false,
+                          kCifWidth, kCifHeight);
+  config.num_frames = kNumFramesShort;
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 0, 1, 0.3, 0.1, 0, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{37.5, 36, 0.94, 0.93}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestLibvpx, VeryLowBitrateVP9) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, false, true, true,
+                          kCifWidth, kCifHeight);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{50, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {15, 3, 75, 1, 0.5, 0.4, 2, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{28, 25, 0.80, 0.65}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+// TODO(marpan): Add temporal layer test for VP9, once changes are in
+// vp9 wrapper for this.
+
+#endif  // defined(RTC_ENABLE_VP9)
+
+TEST(VideoCodecTestLibvpx, HighBitrateVP8) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, true, true, false,
+                          kCifWidth, kCifHeight);
+  config.num_frames = kNumFramesShort;
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 0, 1, 0.2, 0.1, 0, 1}};
+
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+  std::vector<QualityThresholds> quality_thresholds = {{35, 33, 0.91, 0.89}};
+#else
+  std::vector<QualityThresholds> quality_thresholds = {{37, 35, 0.93, 0.91}};
+#endif
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestLibvpx, MAYBE_ChangeBitrateVP8) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, true, true, false,
+                          kCifWidth, kCifHeight);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {
+      {200, 30, 0},  // target_kbps, input_fps, frame_num
+      {800, 30, 100},
+      {500, 30, 200}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 0, 1, 0.2, 0.1, 0, 1},
+      {15.5, 1, 0, 1, 0.2, 0.1, 0, 0},
+      {15, 1, 0, 1, 0.2, 0.1, 0, 0}};
+
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+  std::vector<QualityThresholds> quality_thresholds = {
+      {31.8, 31, 0.86, 0.85}, {36, 34.8, 0.92, 0.90}, {33.5, 32, 0.90, 0.88}};
+#else
+  std::vector<QualityThresholds> quality_thresholds = {
+      {33, 32, 0.89, 0.88}, {38, 36, 0.94, 0.93}, {35, 34, 0.92, 0.91}};
+#endif
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestLibvpx, MAYBE_ChangeFramerateVP8) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, true, true, false,
+                          kCifWidth, kCifHeight);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {
+      {80, 24, 0},  // target_kbps, input_fps, frame_index_rate_update
+      {80, 15, 100},
+      {80, 10, 200}};
+
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {10, 2.42, 60, 1, 0.3, 0.3, 0, 1},
+      {10, 2, 30, 1, 0.3, 0.3, 0, 0},
+      {10, 2, 10, 1, 0.3, 0.2, 0, 0}};
+#else
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {10, 2, 20, 1, 0.3, 0.15, 0, 1},
+      {5, 2, 5, 1, 0.3, 0.15, 0, 0},
+      {4, 2, 1, 1, 0.3, 0.2, 0, 0}};
+#endif
+
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+  std::vector<QualityThresholds> quality_thresholds = {
+      {31, 30, 0.85, 0.84}, {31.4, 30.5, 0.86, 0.84}, {30.5, 29, 0.83, 0.78}};
+#else
+  std::vector<QualityThresholds> quality_thresholds = {
+      {31, 30, 0.87, 0.85}, {32, 31, 0.88, 0.85}, {32, 30, 0.87, 0.82}};
+#endif
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_TemporalLayersVP8 DISABLED_TemporalLayersVP8
+#else
+#define MAYBE_TemporalLayersVP8 TemporalLayersVP8
+#endif
+TEST(VideoCodecTestLibvpx, MAYBE_TemporalLayersVP8) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 3, true, true, false,
+                          kCifWidth, kCifHeight);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{200, 30, 0}, {400, 30, 150}};
+
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {10, 1, 2.1, 1, 0.2, 0.1, 0, 1}, {12, 2, 3, 1, 0.2, 0.1, 0, 1}};
+#else
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 0, 1, 0.2, 0.1, 0, 1}, {10, 2, 0, 1, 0.2, 0.1, 0, 1}};
+#endif
+// Min SSIM drops because of high motion scene with complex backgound (trees).
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+  std::vector<QualityThresholds> quality_thresholds = {{31, 30, 0.85, 0.83},
+                                                       {31, 28, 0.85, 0.75}};
+#else
+  std::vector<QualityThresholds> quality_thresholds = {{32, 30, 0.88, 0.85},
+                                                       {33, 30, 0.89, 0.83}};
+#endif
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_MultiresVP8 DISABLED_MultiresVP8
+#else
+#define MAYBE_MultiresVP8 MultiresVP8
+#endif
+TEST(VideoCodecTestLibvpx, MAYBE_MultiresVP8) {
+  auto config = CreateConfig();
+  config.filename = "ConferenceMotion_1280_720_50";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = 100;
+  config.SetCodecSettings(cricket::kVp8CodecName, 3, 1, 3, true, true, false,
+                          1280, 720);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{1500, 30, 0}};
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {4.1, 1.04, 7, 0.18, 0.14, 0.08, 0, 1}};
+#else
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 5, 1, 0.3, 0.1, 0, 1}};
+#endif
+  std::vector<QualityThresholds> quality_thresholds = {{34, 32, 0.90, 0.88}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_SimulcastVP8 DISABLED_SimulcastVP8
+#else
+#define MAYBE_SimulcastVP8 SimulcastVP8
+#endif
+TEST(VideoCodecTestLibvpx, MAYBE_SimulcastVP8) {
+  auto config = CreateConfig();
+  config.filename = "ConferenceMotion_1280_720_50";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = 100;
+  config.SetCodecSettings(cricket::kVp8CodecName, 3, 1, 3, true, true, false,
+                          1280, 720);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+
+  InternalEncoderFactory internal_encoder_factory;
+  std::unique_ptr<VideoEncoderFactory> adapted_encoder_factory =
+      std::make_unique<FunctionVideoEncoderFactory>([&]() {
+        return std::make_unique<SimulcastEncoderAdapter>(
+            &internal_encoder_factory, SdpVideoFormat(cricket::kVp8CodecName));
+      });
+  std::unique_ptr<InternalDecoderFactory> internal_decoder_factory(
+      new InternalDecoderFactory());
+
+  auto fixture =
+      CreateVideoCodecTestFixture(config, std::move(internal_decoder_factory),
+                                  std::move(adapted_encoder_factory));
+
+  std::vector<RateProfile> rate_profiles = {{1500, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {20, 5, 90, 1, 0.5, 0.3, 0, 1}};
+  std::vector<QualityThresholds> quality_thresholds = {{34, 32, 0.90, 0.88}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_SvcVP9 DISABLED_SvcVP9
+#else
+#define MAYBE_SvcVP9 SvcVP9
+#endif
+TEST(VideoCodecTestLibvpx, MAYBE_SvcVP9) {
+  auto config = CreateConfig();
+  config.filename = "ConferenceMotion_1280_720_50";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = 100;
+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 3, 3, true, true, false,
+                          1280, 720);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{1500, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 5, 1, 0.3, 0.1, 0, 1}};
+  std::vector<QualityThresholds> quality_thresholds = {{36, 34, 0.93, 0.90}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestLibvpx, DISABLED_MultiresVP8RdPerf) {
+  auto config = CreateConfig();
+  config.filename = "FourPeople_1280x720_30";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = 300;
+  config.print_frame_level_stats = true;
+  config.SetCodecSettings(cricket::kVp8CodecName, 3, 1, 3, true, true, false,
+                          1280, 720);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::map<size_t, std::vector<VideoStatistics>> rd_stats;
+  for (size_t bitrate_kbps : kBitrateRdPerfKbps) {
+    std::vector<RateProfile> rate_profiles = {{bitrate_kbps, 30, 0}};
+
+    fixture->RunTest(rate_profiles, nullptr, nullptr, nullptr);
+
+    rd_stats[bitrate_kbps] =
+        fixture->GetStats().SliceAndCalcLayerVideoStatistic(
+            kNumFirstFramesToSkipAtRdPerfAnalysis, config.num_frames - 1);
+  }
+
+  PrintRdPerf(rd_stats);
+}
+
+TEST(VideoCodecTestLibvpx, DISABLED_SvcVP9RdPerf) {
+  auto config = CreateConfig();
+  config.filename = "FourPeople_1280x720_30";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = 300;
+  config.print_frame_level_stats = true;
+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 3, 3, true, true, false,
+                          1280, 720);
+  const auto frame_checker = std::make_unique<QpFrameChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::map<size_t, std::vector<VideoStatistics>> rd_stats;
+  for (size_t bitrate_kbps : kBitrateRdPerfKbps) {
+    std::vector<RateProfile> rate_profiles = {{bitrate_kbps, 30, 0}};
+
+    fixture->RunTest(rate_profiles, nullptr, nullptr, nullptr);
+
+    rd_stats[bitrate_kbps] =
+        fixture->GetStats().SliceAndCalcLayerVideoStatistic(
+            kNumFirstFramesToSkipAtRdPerfAnalysis, config.num_frames - 1);
+  }
+
+  PrintRdPerf(rd_stats);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc
new file mode 100644
index 0000000000..fce21544b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc
@@ -0,0 +1,267 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <string>
+#include <tuple>
+#include <vector>
+
+#include "api/test/create_videocodec_test_fixture.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/test/android_codec_factory_helper.h"
+#include "modules/video_coding/codecs/test/videocodec_test_fixture_impl.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+const int kForemanNumFrames = 300;
+const int kForemanFramerateFps = 30;
+
+struct RateProfileData {
+  std::string name;
+  std::vector<webrtc::test::RateProfile> rate_profile;
+};
+
+const size_t kConstRateIntervalSec = 10;
+
+const RateProfileData kBitRateHighLowHigh = {
+    /*name=*/"BitRateHighLowHigh",
+    /*rate_profile=*/{
+        {/*target_kbps=*/3000, /*input_fps=*/30, /*frame_num=*/0},
+        {/*target_kbps=*/1500, /*input_fps=*/30, /*frame_num=*/300},
+        {/*target_kbps=*/750, /*input_fps=*/30, /*frame_num=*/600},
+        {/*target_kbps=*/1500, /*input_fps=*/30, /*frame_num=*/900},
+        {/*target_kbps=*/3000, /*input_fps=*/30, /*frame_num=*/1200}}};
+
+const RateProfileData kBitRateLowHighLow = {
+    /*name=*/"BitRateLowHighLow",
+    /*rate_profile=*/{
+        {/*target_kbps=*/750, /*input_fps=*/30, /*frame_num=*/0},
+        {/*target_kbps=*/1500, /*input_fps=*/30, /*frame_num=*/300},
+        {/*target_kbps=*/3000, /*input_fps=*/30, /*frame_num=*/600},
+        {/*target_kbps=*/1500, /*input_fps=*/30, /*frame_num=*/900},
+        {/*target_kbps=*/750, /*input_fps=*/30, /*frame_num=*/1200}}};
+
+const RateProfileData kFrameRateHighLowHigh = {
+    /*name=*/"FrameRateHighLowHigh",
+    /*rate_profile=*/{
+        {/*target_kbps=*/2000, /*input_fps=*/30, /*frame_num=*/0},
+        {/*target_kbps=*/2000, /*input_fps=*/15, /*frame_num=*/300},
+        {/*target_kbps=*/2000, /*input_fps=*/7.5, /*frame_num=*/450},
+        {/*target_kbps=*/2000, /*input_fps=*/15, /*frame_num=*/525},
+        {/*target_kbps=*/2000, /*input_fps=*/30, /*frame_num=*/675}}};
+
+const RateProfileData kFrameRateLowHighLow = {
+    /*name=*/"FrameRateLowHighLow",
+    /*rate_profile=*/{
+        {/*target_kbps=*/2000, /*input_fps=*/7.5, /*frame_num=*/0},
+        {/*target_kbps=*/2000, /*input_fps=*/15, /*frame_num=*/75},
+        {/*target_kbps=*/2000, /*input_fps=*/30, /*frame_num=*/225},
+        {/*target_kbps=*/2000, /*input_fps=*/15, /*frame_num=*/525},
+        {/*target_kbps=*/2000, /*input_fps=*/7.5, /*frame_num=*/775}}};
+
+VideoCodecTestFixture::Config CreateConfig() {
+  VideoCodecTestFixture::Config config;
+  config.filename = "foreman_cif";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = kForemanNumFrames;
+  // In order to not overwhelm the OpenMAX buffers in the Android MediaCodec.
+  config.encode_in_real_time = true;
+  return config;
+}
+
+std::unique_ptr<VideoCodecTestFixture> CreateTestFixtureWithConfig(
+    VideoCodecTestFixture::Config config) {
+  InitializeAndroidObjects();  // Idempotent.
+  auto encoder_factory = CreateAndroidEncoderFactory();
+  auto decoder_factory = CreateAndroidDecoderFactory();
+  return CreateVideoCodecTestFixture(config, std::move(decoder_factory),
+                                     std::move(encoder_factory));
+}
+}  // namespace
+
+TEST(VideoCodecTestMediaCodec, ForemanCif500kbpsVp8) {
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, false, false, false,
+                          352, 288);
+  auto fixture = CreateTestFixtureWithConfig(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, kForemanFramerateFps, 0}};
+
+  // The thresholds below may have to be tweaked to let even poor MediaCodec
+  // implementations pass. If this test fails on the bots, disable it and
+  // ping brandtr@.
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {10, 1, 1, 0.1, 0.2, 0.1, 0, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{36, 31, 0.92, 0.86}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestMediaCodec, ForemanCif500kbpsH264CBP) {
+  auto config = CreateConfig();
+  const auto frame_checker =
+      std::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
+  config.encoded_frame_checker = frame_checker.get();
+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, false, false,
+                          352, 288);
+  auto fixture = CreateTestFixtureWithConfig(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, kForemanFramerateFps, 0}};
+
+  // The thresholds below may have to be tweaked to let even poor MediaCodec
+  // implementations pass. If this test fails on the bots, disable it and
+  // ping brandtr@.
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {10, 1, 1, 0.1, 0.2, 0.1, 0, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{36, 31, 0.92, 0.86}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+// TODO(brandtr): Enable this test when we have trybots/buildbots with
+// HW encoders that support CHP.
+TEST(VideoCodecTestMediaCodec, DISABLED_ForemanCif500kbpsH264CHP) {
+  auto config = CreateConfig();
+  const auto frame_checker =
+      std::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
+
+  config.h264_codec_settings.profile = H264Profile::kProfileConstrainedHigh;
+  config.encoded_frame_checker = frame_checker.get();
+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, false, false,
+                          352, 288);
+  auto fixture = CreateTestFixtureWithConfig(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, kForemanFramerateFps, 0}};
+
+  // The thresholds below may have to be tweaked to let even poor MediaCodec
+  // implementations pass. If this test fails on the bots, disable it and
+  // ping brandtr@.
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{37, 35, 0.93, 0.91}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+TEST(VideoCodecTestMediaCodec, ForemanMixedRes100kbpsVp8H264) {
+  auto config = CreateConfig();
+  const int kNumFrames = 30;
+  const std::vector<std::string> codecs = {cricket::kVp8CodecName,
+                                           cricket::kH264CodecName};
+  const std::vector<std::tuple<int, int>> resolutions = {
+      {128, 96}, {176, 144}, {320, 240}, {480, 272}};
+  const std::vector<RateProfile> rate_profiles = {
+      {100, kForemanFramerateFps, 0}};
+  const std::vector<QualityThresholds> quality_thresholds = {
+      {29, 26, 0.8, 0.75}};
+
+  for (const auto& codec : codecs) {
+    for (const auto& resolution : resolutions) {
+      const int width = std::get<0>(resolution);
+      const int height = std::get<1>(resolution);
+      config.filename = std::string("foreman_") + std::to_string(width) + "x" +
+                        std::to_string(height);
+      config.filepath = ResourcePath(config.filename, "yuv");
+      config.num_frames = kNumFrames;
+      config.SetCodecSettings(codec, 1, 1, 1, false, false, false, width,
+                              height);
+
+      auto fixture = CreateTestFixtureWithConfig(config);
+      fixture->RunTest(rate_profiles, nullptr /* rc_thresholds */,
+                       &quality_thresholds, nullptr /* bs_thresholds */);
+    }
+  }
+}
+
+class VideoCodecTestMediaCodecRateAdaptation
+    : public ::testing::TestWithParam<
+          std::tuple<RateProfileData, std::string>> {
+ public:
+  static std::string ParamInfoToStr(
+      const ::testing::TestParamInfo<
+          VideoCodecTestMediaCodecRateAdaptation::ParamType>& info) {
+    char buf[512];
+    rtc::SimpleStringBuilder ss(buf);
+    ss << std::get<0>(info.param).name << "_" << std::get<1>(info.param);
+    return ss.str();
+  }
+};
+
+TEST_P(VideoCodecTestMediaCodecRateAdaptation, DISABLED_RateAdaptation) {
+  const std::vector<webrtc::test::RateProfile> rate_profile =
+      std::get<0>(GetParam()).rate_profile;
+  const std::string codec_name = std::get<1>(GetParam());
+
+  VideoCodecTestFixture::Config config;
+  config.filename = "FourPeople_1280x720_30";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = rate_profile.back().frame_num +
+                      static_cast<size_t>(kConstRateIntervalSec *
+                                          rate_profile.back().input_fps);
+  config.encode_in_real_time = true;
+  config.SetCodecSettings(codec_name, 1, 1, 1, false, false, false, 1280, 720);
+
+  auto fixture = CreateTestFixtureWithConfig(config);
+  fixture->RunTest(rate_profile, nullptr, nullptr, nullptr);
+
+  for (size_t i = 0; i < rate_profile.size(); ++i) {
+    const size_t num_frames =
+        static_cast<size_t>(rate_profile[i].input_fps * kConstRateIntervalSec);
+
+    auto stats = fixture->GetStats().SliceAndCalcLayerVideoStatistic(
+        rate_profile[i].frame_num, rate_profile[i].frame_num + num_frames - 1);
+    ASSERT_EQ(stats.size(), 1u);
+
+    // Bitrate mismatch is <= 10%.
+    EXPECT_LE(stats[0].avg_bitrate_mismatch_pct, 10);
+    EXPECT_GE(stats[0].avg_bitrate_mismatch_pct, -10);
+
+    // Avg frame transmission delay and processing latency is <=100..250ms
+    // depending on frame rate.
+    const double expected_delay_sec =
+        std::min(std::max(1 / rate_profile[i].input_fps, 0.1), 0.25);
+    EXPECT_LE(stats[0].avg_delay_sec, expected_delay_sec);
+    EXPECT_LE(stats[0].avg_encode_latency_sec, expected_delay_sec);
+    EXPECT_LE(stats[0].avg_decode_latency_sec, expected_delay_sec);
+
+    // Frame drops are not expected.
+    EXPECT_EQ(stats[0].num_encoded_frames, num_frames);
+    EXPECT_EQ(stats[0].num_decoded_frames, num_frames);
+
+    // Periodic keyframes are not expected.
+    EXPECT_EQ(stats[0].num_key_frames, i == 0 ? 1u : 0);
+
+    // Ensure codec delivers a reasonable spatial quality.
+    EXPECT_GE(stats[0].avg_psnr_y, 35);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    RateAdaptation,
+    VideoCodecTestMediaCodecRateAdaptation,
+    ::testing::Combine(::testing::Values(kBitRateLowHighLow,
+                                         kBitRateHighLowHigh,
+                                         kFrameRateLowHighLow,
+                                         kFrameRateHighLowHigh),
+                       ::testing::Values(cricket::kVp8CodecName,
+                                         cricket::kVp9CodecName,
+                                         cricket::kH264CodecName)),
+    VideoCodecTestMediaCodecRateAdaptation::ParamInfoToStr);
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc
new file mode 100644
index 0000000000..6513074bad
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <vector>
+
+#include "api/test/create_videocodec_test_fixture.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/test/videocodec_test_fixture_impl.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+// Codec settings.
+const int kCifWidth = 352;
+const int kCifHeight = 288;
+const int kNumFrames = 100;
+
+VideoCodecTestFixture::Config CreateConfig() {
+  VideoCodecTestFixture::Config config;
+  config.filename = "foreman_cif";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = kNumFrames;
+  // Only allow encoder/decoder to use single core, for predictability.
+  config.use_single_core = true;
+  return config;
+}
+}  // namespace
+
+TEST(VideoCodecTestOpenH264, ConstantHighBitrate) {
+  auto frame_checker =
+      std::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, true, false,
+                          kCifWidth, kCifHeight);
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{37, 35, 0.93, 0.91}};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
+}
+
+// H264: Enable SingleNalUnit packetization mode. Encoder should split
+// large frames into multiple slices and limit length of NAL units.
+TEST(VideoCodecTestOpenH264, SingleNalUnit) {
+  auto frame_checker =
+      std::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
+  auto config = CreateConfig();
+  config.h264_codec_settings.packetization_mode =
+      H264PacketizationMode::SingleNalUnit;
+  config.max_payload_size_bytes = 500;
+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, true, false,
+                          kCifWidth, kCifHeight);
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateVideoCodecTestFixture(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, 30, 0}};
+
+  std::vector<RateControlThresholds> rc_thresholds = {
+      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{37, 35, 0.93, 0.91}};
+
+  BitstreamThresholds bs_thresholds = {config.max_payload_size_bytes};
+
+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds,
+                   &bs_thresholds);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc
new file mode 100644
index 0000000000..390348b97a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc
@@ -0,0 +1,441 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
+
+#include <algorithm>
+#include <cmath>
+#include <iterator>
+#include <limits>
+#include <numeric>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/running_statistics.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace test {
+
+using FrameStatistics = VideoCodecTestStats::FrameStatistics;
+using VideoStatistics = VideoCodecTestStats::VideoStatistics;
+
+namespace {
+const int kMaxBitrateMismatchPercent = 20;
+}
+
+VideoCodecTestStatsImpl::VideoCodecTestStatsImpl() = default;
+VideoCodecTestStatsImpl::~VideoCodecTestStatsImpl() = default;
+
+void VideoCodecTestStatsImpl::AddFrame(const FrameStatistics& frame_stat) {
+  const size_t timestamp = frame_stat.rtp_timestamp;
+  const size_t layer_idx = frame_stat.spatial_idx;
+  RTC_DCHECK(rtp_timestamp_to_frame_num_[layer_idx].find(timestamp) ==
+             rtp_timestamp_to_frame_num_[layer_idx].end());
+  rtp_timestamp_to_frame_num_[layer_idx][timestamp] = frame_stat.frame_number;
+  layer_stats_[layer_idx].push_back(frame_stat);
+}
+
+FrameStatistics* VideoCodecTestStatsImpl::GetFrame(size_t frame_num,
+                                                   size_t layer_idx) {
+  RTC_CHECK_LT(frame_num, layer_stats_[layer_idx].size());
+  return &layer_stats_[layer_idx][frame_num];
+}
+
+FrameStatistics* VideoCodecTestStatsImpl::GetFrameWithTimestamp(
+    size_t timestamp,
+    size_t layer_idx) {
+  RTC_DCHECK(rtp_timestamp_to_frame_num_[layer_idx].find(timestamp) !=
+             rtp_timestamp_to_frame_num_[layer_idx].end());
+
+  return GetFrame(rtp_timestamp_to_frame_num_[layer_idx][timestamp], layer_idx);
+}
+
+FrameStatistics* VideoCodecTestStatsImpl::GetOrAddFrame(size_t timestamp_rtp,
+                                                        size_t spatial_idx) {
+  if (rtp_timestamp_to_frame_num_[spatial_idx].count(timestamp_rtp) > 0) {
+    return GetFrameWithTimestamp(timestamp_rtp, spatial_idx);
+  }
+
+  size_t frame_num = layer_stats_[spatial_idx].size();
+  AddFrame(FrameStatistics(frame_num, timestamp_rtp, spatial_idx));
+
+  return GetFrameWithTimestamp(timestamp_rtp, spatial_idx);
+}
+
+std::vector<FrameStatistics> VideoCodecTestStatsImpl::GetFrameStatistics()
+    const {
+  size_t capacity = 0;
+  for (const auto& layer_stat : layer_stats_) {
+    capacity += layer_stat.second.size();
+  }
+
+  std::vector<FrameStatistics> frame_statistics;
+  frame_statistics.reserve(capacity);
+  for (const auto& layer_stat : layer_stats_) {
+    std::copy(layer_stat.second.cbegin(), layer_stat.second.cend(),
+              std::back_inserter(frame_statistics));
+  }
+
+  return frame_statistics;
+}
+
+std::vector<VideoStatistics>
+VideoCodecTestStatsImpl::SliceAndCalcLayerVideoStatistic(
+    size_t first_frame_num,
+    size_t last_frame_num) {
+  std::vector<VideoStatistics> layer_stats;
+
+  size_t num_spatial_layers = 0;
+  size_t num_temporal_layers = 0;
+  GetNumberOfEncodedLayers(first_frame_num, last_frame_num, &num_spatial_layers,
+                           &num_temporal_layers);
+  RTC_CHECK_GT(num_spatial_layers, 0);
+  RTC_CHECK_GT(num_temporal_layers, 0);
+
+  for (size_t spatial_idx = 0; spatial_idx < num_spatial_layers;
+       ++spatial_idx) {
+    for (size_t temporal_idx = 0; temporal_idx < num_temporal_layers;
+         ++temporal_idx) {
+      VideoStatistics layer_stat = SliceAndCalcVideoStatistic(
+          first_frame_num, last_frame_num, spatial_idx, temporal_idx, false,
+          /*target_bitrate=*/absl::nullopt, /*target_framerate=*/absl::nullopt);
+      layer_stats.push_back(layer_stat);
+    }
+  }
+
+  return layer_stats;
+}
+
+VideoStatistics VideoCodecTestStatsImpl::SliceAndCalcAggregatedVideoStatistic(
+    size_t first_frame_num,
+    size_t last_frame_num) {
+  size_t num_spatial_layers = 0;
+  size_t num_temporal_layers = 0;
+  GetNumberOfEncodedLayers(first_frame_num, last_frame_num, &num_spatial_layers,
+                           &num_temporal_layers);
+  RTC_CHECK_GT(num_spatial_layers, 0);
+  RTC_CHECK_GT(num_temporal_layers, 0);
+
+  return SliceAndCalcVideoStatistic(
+      first_frame_num, last_frame_num, num_spatial_layers - 1,
+      num_temporal_layers - 1, true, /*target_bitrate=*/absl::nullopt,
+      /*target_framerate=*/absl::nullopt);
+}
+
+VideoStatistics VideoCodecTestStatsImpl::CalcVideoStatistic(
+    size_t first_frame_num,
+    size_t last_frame_num,
+    DataRate target_bitrate,
+    Frequency target_framerate) {
+  size_t num_spatial_layers = 0;
+  size_t num_temporal_layers = 0;
+  GetNumberOfEncodedLayers(first_frame_num, last_frame_num, &num_spatial_layers,
+                           &num_temporal_layers);
+  return SliceAndCalcVideoStatistic(
+      first_frame_num, last_frame_num, num_spatial_layers - 1,
+      num_temporal_layers - 1, true, target_bitrate, target_framerate);
+}
+
+size_t VideoCodecTestStatsImpl::Size(size_t spatial_idx) {
+  return layer_stats_[spatial_idx].size();
+}
+
+void VideoCodecTestStatsImpl::Clear() {
+  layer_stats_.clear();
+  rtp_timestamp_to_frame_num_.clear();
+}
+
+FrameStatistics VideoCodecTestStatsImpl::AggregateFrameStatistic(
+    size_t frame_num,
+    size_t spatial_idx,
+    bool aggregate_independent_layers) {
+  FrameStatistics frame_stat = *GetFrame(frame_num, spatial_idx);
+  bool inter_layer_predicted = frame_stat.inter_layer_predicted;
+  while (spatial_idx-- > 0) {
+    if (aggregate_independent_layers || inter_layer_predicted) {
+      FrameStatistics* base_frame_stat = GetFrame(frame_num, spatial_idx);
+      frame_stat.length_bytes += base_frame_stat->length_bytes;
+      frame_stat.target_bitrate_kbps += base_frame_stat->target_bitrate_kbps;
+
+      inter_layer_predicted = base_frame_stat->inter_layer_predicted;
+    }
+  }
+
+  return frame_stat;
+}
+
+size_t VideoCodecTestStatsImpl::CalcLayerTargetBitrateKbps(
+    size_t first_frame_num,
+    size_t last_frame_num,
+    size_t spatial_idx,
+    size_t temporal_idx,
+    bool aggregate_independent_layers) {
+  size_t target_bitrate_kbps = 0;
+
+  // We don't know if superframe includes all required spatial layers because
+  // of possible frame drops. Run through all frames in specified range, find
+  // and return maximum target bitrate. Assume that target bitrate in frame
+  // statistic is specified per temporal layer.
+  for (size_t frame_num = first_frame_num; frame_num <= last_frame_num;
+       ++frame_num) {
+    FrameStatistics superframe = AggregateFrameStatistic(
+        frame_num, spatial_idx, aggregate_independent_layers);
+
+    if (superframe.temporal_idx <= temporal_idx) {
+      target_bitrate_kbps =
+          std::max(target_bitrate_kbps, superframe.target_bitrate_kbps);
+    }
+  }
+
+  RTC_DCHECK_GT(target_bitrate_kbps, 0);
+  return target_bitrate_kbps;
+}
+
+VideoStatistics VideoCodecTestStatsImpl::SliceAndCalcVideoStatistic(
+    size_t first_frame_num,
+    size_t last_frame_num,
+    size_t spatial_idx,
+    size_t temporal_idx,
+    bool aggregate_independent_layers,
+    absl::optional<DataRate> target_bitrate,
+    absl::optional<Frequency> target_framerate) {
+  VideoStatistics video_stat;
+
+  float buffer_level_bits = 0.0f;
+  webrtc_impl::RunningStatistics<float> buffer_level_sec;
+
+  webrtc_impl::RunningStatistics<size_t> key_frame_size_bytes;
+  webrtc_impl::RunningStatistics<size_t> delta_frame_size_bytes;
+
+  webrtc_impl::RunningStatistics<size_t> frame_encoding_time_us;
+  webrtc_impl::RunningStatistics<size_t> frame_decoding_time_us;
+
+  webrtc_impl::RunningStatistics<float> psnr_y;
+  webrtc_impl::RunningStatistics<float> psnr_u;
+  webrtc_impl::RunningStatistics<float> psnr_v;
+  webrtc_impl::RunningStatistics<float> psnr;
+  webrtc_impl::RunningStatistics<float> ssim;
+  webrtc_impl::RunningStatistics<int> qp;
+
+  size_t rtp_timestamp_first_frame = 0;
+  size_t rtp_timestamp_prev_frame = 0;
+
+  FrameStatistics last_successfully_decoded_frame(0, 0, 0);
+
+  const size_t target_bitrate_kbps =
+      target_bitrate.has_value()
+          ? target_bitrate->kbps()
+          : CalcLayerTargetBitrateKbps(first_frame_num, last_frame_num,
+                                       spatial_idx, temporal_idx,
+                                       aggregate_independent_layers);
+  const size_t target_bitrate_bps = 1000 * target_bitrate_kbps;
+  RTC_CHECK_GT(target_bitrate_kbps, 0);  // We divide by `target_bitrate_kbps`.
+
+  for (size_t frame_num = first_frame_num; frame_num <= last_frame_num;
+       ++frame_num) {
+    FrameStatistics frame_stat = AggregateFrameStatistic(
+        frame_num, spatial_idx, aggregate_independent_layers);
+
+    float time_since_first_frame_sec =
+        1.0f * (frame_stat.rtp_timestamp - rtp_timestamp_first_frame) /
+        kVideoPayloadTypeFrequency;
+    float time_since_prev_frame_sec =
+        1.0f * (frame_stat.rtp_timestamp - rtp_timestamp_prev_frame) /
+        kVideoPayloadTypeFrequency;
+
+    if (frame_stat.temporal_idx > temporal_idx) {
+      continue;
+    }
+
+    buffer_level_bits -= time_since_prev_frame_sec * 1000 * target_bitrate_kbps;
+    buffer_level_bits = std::max(0.0f, buffer_level_bits);
+    buffer_level_bits += 8.0 * frame_stat.length_bytes;
+    buffer_level_sec.AddSample(buffer_level_bits /
+                               (1000 * target_bitrate_kbps));
+
+    video_stat.length_bytes += frame_stat.length_bytes;
+
+    if (frame_stat.encoding_successful) {
+      ++video_stat.num_encoded_frames;
+
+      if (frame_stat.frame_type == VideoFrameType::kVideoFrameKey) {
+        key_frame_size_bytes.AddSample(frame_stat.length_bytes);
+        ++video_stat.num_key_frames;
+      } else {
+        delta_frame_size_bytes.AddSample(frame_stat.length_bytes);
+      }
+
+      frame_encoding_time_us.AddSample(frame_stat.encode_time_us);
+      qp.AddSample(frame_stat.qp);
+
+      video_stat.max_nalu_size_bytes = std::max(video_stat.max_nalu_size_bytes,
+                                                frame_stat.max_nalu_size_bytes);
+    }
+
+    if (frame_stat.decoding_successful) {
+      ++video_stat.num_decoded_frames;
+
+      video_stat.width = std::max(video_stat.width, frame_stat.decoded_width);
+      video_stat.height =
+          std::max(video_stat.height, frame_stat.decoded_height);
+
+      if (video_stat.num_decoded_frames > 1) {
+        if (last_successfully_decoded_frame.decoded_width !=
+                frame_stat.decoded_width ||
+            last_successfully_decoded_frame.decoded_height !=
+                frame_stat.decoded_height) {
+          ++video_stat.num_spatial_resizes;
+        }
+      }
+
+      frame_decoding_time_us.AddSample(frame_stat.decode_time_us);
+      last_successfully_decoded_frame = frame_stat;
+    }
+
+    if (frame_stat.quality_analysis_successful) {
+      psnr_y.AddSample(frame_stat.psnr_y);
+      psnr_u.AddSample(frame_stat.psnr_u);
+      psnr_v.AddSample(frame_stat.psnr_v);
+      psnr.AddSample(frame_stat.psnr);
+      ssim.AddSample(frame_stat.ssim);
+    }
+
+    if (video_stat.num_input_frames > 0) {
+      if (video_stat.time_to_reach_target_bitrate_sec == 0.0f) {
+        RTC_CHECK_GT(time_since_first_frame_sec, 0);
+        const float curr_kbps =
+            8.0 * video_stat.length_bytes / 1000 / time_since_first_frame_sec;
+        const float bitrate_mismatch_percent =
+            100 * std::fabs(curr_kbps - target_bitrate_kbps) /
+            target_bitrate_kbps;
+        if (bitrate_mismatch_percent < kMaxBitrateMismatchPercent) {
+          video_stat.time_to_reach_target_bitrate_sec =
+              time_since_first_frame_sec;
+        }
+      }
+    }
+
+    rtp_timestamp_prev_frame = frame_stat.rtp_timestamp;
+    if (video_stat.num_input_frames == 0) {
+      rtp_timestamp_first_frame = frame_stat.rtp_timestamp;
+    }
+
+    ++video_stat.num_input_frames;
+  }
+
+  const size_t num_frames = last_frame_num - first_frame_num + 1;
+  const size_t timestamp_delta =
+      GetFrame(first_frame_num + 1, spatial_idx)->rtp_timestamp -
+      GetFrame(first_frame_num, spatial_idx)->rtp_timestamp;
+  RTC_CHECK_GT(timestamp_delta, 0);
+  const float input_framerate_fps =
+      target_framerate.has_value()
+          ? target_framerate->millihertz() / 1000.0
+          : 1.0 * kVideoPayloadTypeFrequency / timestamp_delta;
+  RTC_CHECK_GT(input_framerate_fps, 0);
+  const float duration_sec = num_frames / input_framerate_fps;
+
+  video_stat.target_bitrate_kbps = target_bitrate_kbps;
+  video_stat.input_framerate_fps = input_framerate_fps;
+
+  video_stat.spatial_idx = spatial_idx;
+  video_stat.temporal_idx = temporal_idx;
+
+  RTC_CHECK_GT(duration_sec, 0);
+  const float bitrate_bps = 8 * video_stat.length_bytes / duration_sec;
+  video_stat.bitrate_kbps = static_cast<size_t>((bitrate_bps + 500) / 1000);
+  video_stat.framerate_fps = video_stat.num_encoded_frames / duration_sec;
+
+  // http://bugs.webrtc.org/10400: On Windows, we only get millisecond
+  // granularity in the frame encode/decode timing measurements.
+  // So we need to softly avoid a div-by-zero here.
+  const float mean_encode_time_us =
+      frame_encoding_time_us.GetMean().value_or(0);
+  video_stat.enc_speed_fps = mean_encode_time_us > 0.0f
+                                 ? 1000000.0f / mean_encode_time_us
+                                 : std::numeric_limits<float>::max();
+  const float mean_decode_time_us =
+      frame_decoding_time_us.GetMean().value_or(0);
+  video_stat.dec_speed_fps = mean_decode_time_us > 0.0f
+                                 ? 1000000.0f / mean_decode_time_us
+                                 : std::numeric_limits<float>::max();
+
+  video_stat.avg_encode_latency_sec =
+      frame_encoding_time_us.GetMean().value_or(0) / 1000000.0f;
+  video_stat.max_encode_latency_sec =
+      frame_encoding_time_us.GetMax().value_or(0) / 1000000.0f;
+
+  video_stat.avg_decode_latency_sec =
+      frame_decoding_time_us.GetMean().value_or(0) / 1000000.0f;
+  video_stat.max_decode_latency_sec =
+      frame_decoding_time_us.GetMax().value_or(0) / 1000000.0f;
+
+  auto MaxDelaySec = [target_bitrate_kbps](
+                         const webrtc_impl::RunningStatistics<size_t>& stats) {
+    return 8 * stats.GetMax().value_or(0) / 1000 / target_bitrate_kbps;
+  };
+
+  video_stat.avg_delay_sec = buffer_level_sec.GetMean().value_or(0);
+  video_stat.max_key_frame_delay_sec = MaxDelaySec(key_frame_size_bytes);
+  video_stat.max_delta_frame_delay_sec = MaxDelaySec(delta_frame_size_bytes);
+
+  video_stat.avg_bitrate_mismatch_pct =
+      100 * (bitrate_bps - target_bitrate_bps) / target_bitrate_bps;
+  video_stat.avg_framerate_mismatch_pct =
+      100 * (video_stat.framerate_fps - input_framerate_fps) /
+      input_framerate_fps;
+
+  video_stat.avg_key_frame_size_bytes =
+      key_frame_size_bytes.GetMean().value_or(0);
+  video_stat.avg_delta_frame_size_bytes =
+      delta_frame_size_bytes.GetMean().value_or(0);
+  video_stat.avg_qp = qp.GetMean().value_or(0);
+
+  video_stat.avg_psnr_y = psnr_y.GetMean().value_or(0);
+  video_stat.avg_psnr_u = psnr_u.GetMean().value_or(0);
+  video_stat.avg_psnr_v = psnr_v.GetMean().value_or(0);
+  video_stat.avg_psnr = psnr.GetMean().value_or(0);
+  video_stat.min_psnr =
+      psnr.GetMin().value_or(std::numeric_limits<float>::max());
+  video_stat.avg_ssim = ssim.GetMean().value_or(0);
+  video_stat.min_ssim =
+      ssim.GetMin().value_or(std::numeric_limits<float>::max());
+
+  return video_stat;
+}
+
+void VideoCodecTestStatsImpl::GetNumberOfEncodedLayers(
+    size_t first_frame_num,
+    size_t last_frame_num,
+    size_t* num_encoded_spatial_layers,
+    size_t* num_encoded_temporal_layers) {
+  *num_encoded_spatial_layers = 0;
+  *num_encoded_temporal_layers = 0;
+
+  const size_t num_spatial_layers = layer_stats_.size();
+
+  for (size_t frame_num = first_frame_num; frame_num <= last_frame_num;
+       ++frame_num) {
+    for (size_t spatial_idx = 0; spatial_idx < num_spatial_layers;
+         ++spatial_idx) {
+      FrameStatistics* frame_stat = GetFrame(frame_num, spatial_idx);
+      if (frame_stat->encoding_successful) {
+        *num_encoded_spatial_layers =
+            std::max(*num_encoded_spatial_layers, frame_stat->spatial_idx + 1);
+        *num_encoded_temporal_layers = std::max(*num_encoded_temporal_layers,
+                                                frame_stat->temporal_idx + 1);
+      }
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.h b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.h
new file mode 100644
index 0000000000..1a7980aa0a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_STATS_IMPL_H_
+#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_STATS_IMPL_H_
+
+#include <stddef.h>
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "api/test/videocodec_test_stats.h"  // NOLINT(build/include)
+
+namespace webrtc {
+namespace test {
+
+// Statistics for a sequence of processed frames. This class is not thread safe.
+class VideoCodecTestStatsImpl : public VideoCodecTestStats {
+ public:
+  VideoCodecTestStatsImpl();
+  ~VideoCodecTestStatsImpl() override;
+
+  // Creates a FrameStatistics for the next frame to be processed.
+  void AddFrame(const FrameStatistics& frame_stat);
+
+  // Returns the FrameStatistics corresponding to `frame_number` or `timestamp`.
+  FrameStatistics* GetFrame(size_t frame_number, size_t spatial_idx);
+  FrameStatistics* GetFrameWithTimestamp(size_t timestamp, size_t spatial_idx);
+
+  // Creates FrameStatisticts if it doesn't exists and/or returns
+  // created/existing FrameStatisticts.
+  FrameStatistics* GetOrAddFrame(size_t timestamp_rtp, size_t spatial_idx);
+
+  // Implements VideoCodecTestStats.
+  std::vector<FrameStatistics> GetFrameStatistics() const override;
+  std::vector<VideoStatistics> SliceAndCalcLayerVideoStatistic(
+      size_t first_frame_num,
+      size_t last_frame_num) override;
+
+  VideoStatistics SliceAndCalcAggregatedVideoStatistic(size_t first_frame_num,
+                                                       size_t last_frame_num);
+
+  VideoStatistics CalcVideoStatistic(size_t first_frame,
+                                     size_t last_frame,
+                                     DataRate target_bitrate,
+                                     Frequency target_framerate) override;
+
+  size_t Size(size_t spatial_idx);
+
+  void Clear();
+
+ private:
+  VideoCodecTestStats::FrameStatistics AggregateFrameStatistic(
+      size_t frame_num,
+      size_t spatial_idx,
+      bool aggregate_independent_layers);
+
+  size_t CalcLayerTargetBitrateKbps(size_t first_frame_num,
+                                    size_t last_frame_num,
+                                    size_t spatial_idx,
+                                    size_t temporal_idx,
+                                    bool aggregate_independent_layers);
+
+  VideoCodecTestStats::VideoStatistics SliceAndCalcVideoStatistic(
+      size_t first_frame_num,
+      size_t last_frame_num,
+      size_t spatial_idx,
+      size_t temporal_idx,
+      bool aggregate_independent_layers,
+      absl::optional<DataRate> target_bitrate,
+      absl::optional<Frequency> target_framerate);
+
+  void GetNumberOfEncodedLayers(size_t first_frame_num,
+                                size_t last_frame_num,
+                                size_t* num_encoded_spatial_layers,
+                                size_t* num_encoded_temporal_layers);
+
+  // layer_idx -> stats.
+  std::map<size_t, std::vector<FrameStatistics>> layer_stats_;
+  // layer_idx -> rtp_timestamp -> frame_num.
+  std::map<size_t, std::map<size_t, size_t>> rtp_timestamp_to_frame_num_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_STATS_IMPL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl_unittest.cc
new file mode 100644
index 0000000000..89e7d2e1c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
+
+#include <vector>
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+using FrameStatistics = VideoCodecTestStatsImpl::FrameStatistics;
+
+namespace {
+
+const size_t kTimestamp = 12345;
+
+using ::testing::AllOf;
+using ::testing::Contains;
+using ::testing::Field;
+
+}  // namespace
+
+TEST(StatsTest, AddAndGetFrame) {
+  VideoCodecTestStatsImpl stats;
+  stats.AddFrame(FrameStatistics(0, kTimestamp, 0));
+  FrameStatistics* frame_stat = stats.GetFrame(0u, 0);
+  EXPECT_EQ(0u, frame_stat->frame_number);
+  EXPECT_EQ(kTimestamp, frame_stat->rtp_timestamp);
+}
+
+TEST(StatsTest, GetOrAddFrame_noFrame_createsNewFrameStat) {
+  VideoCodecTestStatsImpl stats;
+  stats.GetOrAddFrame(kTimestamp, 0);
+  FrameStatistics* frame_stat = stats.GetFrameWithTimestamp(kTimestamp, 0);
+  EXPECT_EQ(kTimestamp, frame_stat->rtp_timestamp);
+}
+
+TEST(StatsTest, GetOrAddFrame_frameExists_returnsExistingFrameStat) {
+  VideoCodecTestStatsImpl stats;
+  stats.AddFrame(FrameStatistics(0, kTimestamp, 0));
+  FrameStatistics* frame_stat1 = stats.GetFrameWithTimestamp(kTimestamp, 0);
+  FrameStatistics* frame_stat2 = stats.GetOrAddFrame(kTimestamp, 0);
+  EXPECT_EQ(frame_stat1, frame_stat2);
+}
+
+TEST(StatsTest, AddAndGetFrames) {
+  VideoCodecTestStatsImpl stats;
+  const size_t kNumFrames = 1000;
+  for (size_t i = 0; i < kNumFrames; ++i) {
+    stats.AddFrame(FrameStatistics(i, kTimestamp + i, 0));
+    FrameStatistics* frame_stat = stats.GetFrame(i, 0);
+    EXPECT_EQ(i, frame_stat->frame_number);
+    EXPECT_EQ(kTimestamp + i, frame_stat->rtp_timestamp);
+  }
+  EXPECT_EQ(kNumFrames, stats.Size(0));
+  // Get frame.
+  size_t i = 22;
+  FrameStatistics* frame_stat = stats.GetFrameWithTimestamp(kTimestamp + i, 0);
+  EXPECT_EQ(i, frame_stat->frame_number);
+  EXPECT_EQ(kTimestamp + i, frame_stat->rtp_timestamp);
+}
+
+TEST(StatsTest, AddFrameLayering) {
+  VideoCodecTestStatsImpl stats;
+  for (size_t i = 0; i < 3; ++i) {
+    stats.AddFrame(FrameStatistics(0, kTimestamp + i, i));
+    FrameStatistics* frame_stat = stats.GetFrame(0u, i);
+    EXPECT_EQ(0u, frame_stat->frame_number);
+    EXPECT_EQ(kTimestamp, frame_stat->rtp_timestamp - i);
+    EXPECT_EQ(1u, stats.Size(i));
+  }
+}
+
+TEST(StatsTest, GetFrameStatistics) {
+  VideoCodecTestStatsImpl stats;
+
+  stats.AddFrame(FrameStatistics(0, kTimestamp, 0));
+  stats.AddFrame(FrameStatistics(0, kTimestamp, 1));
+  stats.AddFrame(FrameStatistics(1, kTimestamp + 3000, 0));
+  stats.AddFrame(FrameStatistics(1, kTimestamp + 3000, 1));
+
+  const std::vector<FrameStatistics> frame_stats = stats.GetFrameStatistics();
+
+  auto field_matcher = [](size_t frame_number, size_t spatial_idx) {
+    return AllOf(Field(&FrameStatistics::frame_number, frame_number),
+                 Field(&FrameStatistics::spatial_idx, spatial_idx));
+  };
+  EXPECT_THAT(frame_stats, Contains(field_matcher(0, 0)));
+  EXPECT_THAT(frame_stats, Contains(field_matcher(0, 1)));
+  EXPECT_THAT(frame_stats, Contains(field_matcher(1, 0)));
+  EXPECT_THAT(frame_stats, Contains(field_matcher(1, 1)));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc
new file mode 100644
index 0000000000..6df974362f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <vector>
+
+#include "api/test/create_videocodec_test_fixture.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/test/objc_codec_factory_helper.h"
+#include "modules/video_coding/codecs/test/videocodec_test_fixture_impl.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+const int kForemanNumFrames = 300;
+
+VideoCodecTestFixture::Config CreateConfig() {
+  VideoCodecTestFixture::Config config;
+  config.filename = "foreman_cif";
+  config.filepath = ResourcePath(config.filename, "yuv");
+  config.num_frames = kForemanNumFrames;
+  return config;
+}
+
+std::unique_ptr<VideoCodecTestFixture> CreateTestFixtureWithConfig(
+    VideoCodecTestFixture::Config config) {
+  auto decoder_factory = CreateObjCDecoderFactory();
+  auto encoder_factory = CreateObjCEncoderFactory();
+  return CreateVideoCodecTestFixture(config, std::move(decoder_factory),
+                                     std::move(encoder_factory));
+}
+}  // namespace
+
+// TODO(webrtc:9099): Disabled until the issue is fixed.
+// HW codecs don't work on simulators. Only run these tests on device.
+// #if TARGET_OS_IPHONE && !TARGET_IPHONE_SIMULATOR
+// #define MAYBE_TEST TEST
+// #else
+#define MAYBE_TEST(s, name) TEST(s, DISABLED_##name)
+// #endif
+
+// TODO(kthelgason): Use RC Thresholds when the internal bitrateAdjuster is no
+// longer in use.
+MAYBE_TEST(VideoCodecTestVideoToolbox, ForemanCif500kbpsH264CBP) {
+  const auto frame_checker =
+      std::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
+  auto config = CreateConfig();
+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, false, false,
+                          352, 288);
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateTestFixtureWithConfig(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, 30, 0}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{33, 29, 0.9, 0.82}};
+
+  fixture->RunTest(rate_profiles, nullptr, &quality_thresholds, nullptr);
+}
+
+MAYBE_TEST(VideoCodecTestVideoToolbox, ForemanCif500kbpsH264CHP) {
+  const auto frame_checker =
+      std::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
+  auto config = CreateConfig();
+  config.h264_codec_settings.profile = H264Profile::kProfileConstrainedHigh;
+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, false, false,
+                          352, 288);
+  config.encoded_frame_checker = frame_checker.get();
+  auto fixture = CreateTestFixtureWithConfig(config);
+
+  std::vector<RateProfile> rate_profiles = {{500, 30, 0}};
+
+  std::vector<QualityThresholds> quality_thresholds = {{33, 30, 0.91, 0.83}};
+
+  fixture->RunTest(rate_profiles, nullptr, &quality_thresholds, nullptr);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videoprocessor.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videoprocessor.cc
new file mode 100644
index 0000000000..13266c40df
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videoprocessor.cc
@@ -0,0 +1,722 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/videoprocessor.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstddef>
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "api/scoped_refptr.h"
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_bitrate_allocator_factory.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "common_video/h264/h264_common.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/time_utils.h"
+#include "test/gtest.h"
+#include "third_party/libyuv/include/libyuv/compare.h"
+#include "third_party/libyuv/include/libyuv/scale.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+const int kMsToRtpTimestamp = kVideoPayloadTypeFrequency / 1000;
+const int kMaxBufferedInputFrames = 20;
+
+const VideoEncoder::Capabilities kCapabilities(false);
+
+size_t GetMaxNaluSizeBytes(const EncodedImage& encoded_frame,
+                           const VideoCodecTestFixture::Config& config) {
+  if (config.codec_settings.codecType != kVideoCodecH264)
+    return 0;
+
+  std::vector<webrtc::H264::NaluIndex> nalu_indices =
+      webrtc::H264::FindNaluIndices(encoded_frame.data(), encoded_frame.size());
+
+  RTC_CHECK(!nalu_indices.empty());
+
+  size_t max_size = 0;
+  for (const webrtc::H264::NaluIndex& index : nalu_indices)
+    max_size = std::max(max_size, index.payload_size);
+
+  return max_size;
+}
+
+size_t GetTemporalLayerIndex(const CodecSpecificInfo& codec_specific) {
+  size_t temporal_idx = 0;
+  if (codec_specific.codecType == kVideoCodecVP8) {
+    temporal_idx = codec_specific.codecSpecific.VP8.temporalIdx;
+  } else if (codec_specific.codecType == kVideoCodecVP9) {
+    temporal_idx = codec_specific.codecSpecific.VP9.temporal_idx;
+  }
+  if (temporal_idx == kNoTemporalIdx) {
+    temporal_idx = 0;
+  }
+  return temporal_idx;
+}
+
+int GetElapsedTimeMicroseconds(int64_t start_ns, int64_t stop_ns) {
+  int64_t diff_us = (stop_ns - start_ns) / rtc::kNumNanosecsPerMicrosec;
+  RTC_DCHECK_GE(diff_us, std::numeric_limits<int>::min());
+  RTC_DCHECK_LE(diff_us, std::numeric_limits<int>::max());
+  return static_cast<int>(diff_us);
+}
+
+void CalculateFrameQuality(const I420BufferInterface& ref_buffer,
+                           const I420BufferInterface& dec_buffer,
+                           VideoCodecTestStats::FrameStatistics* frame_stat,
+                           bool calc_ssim) {
+  if (ref_buffer.width() != dec_buffer.width() ||
+      ref_buffer.height() != dec_buffer.height()) {
+    RTC_CHECK_GE(ref_buffer.width(), dec_buffer.width());
+    RTC_CHECK_GE(ref_buffer.height(), dec_buffer.height());
+    // Downscale reference frame.
+    rtc::scoped_refptr<I420Buffer> scaled_buffer =
+        I420Buffer::Create(dec_buffer.width(), dec_buffer.height());
+    I420Scale(ref_buffer.DataY(), ref_buffer.StrideY(), ref_buffer.DataU(),
+              ref_buffer.StrideU(), ref_buffer.DataV(), ref_buffer.StrideV(),
+              ref_buffer.width(), ref_buffer.height(),
+              scaled_buffer->MutableDataY(), scaled_buffer->StrideY(),
+              scaled_buffer->MutableDataU(), scaled_buffer->StrideU(),
+              scaled_buffer->MutableDataV(), scaled_buffer->StrideV(),
+              scaled_buffer->width(), scaled_buffer->height(),
+              libyuv::kFilterBox);
+
+    CalculateFrameQuality(*scaled_buffer, dec_buffer, frame_stat, calc_ssim);
+  } else {
+    const uint64_t sse_y = libyuv::ComputeSumSquareErrorPlane(
+        dec_buffer.DataY(), dec_buffer.StrideY(), ref_buffer.DataY(),
+        ref_buffer.StrideY(), dec_buffer.width(), dec_buffer.height());
+
+    const uint64_t sse_u = libyuv::ComputeSumSquareErrorPlane(
+        dec_buffer.DataU(), dec_buffer.StrideU(), ref_buffer.DataU(),
+        ref_buffer.StrideU(), dec_buffer.width() / 2, dec_buffer.height() / 2);
+
+    const uint64_t sse_v = libyuv::ComputeSumSquareErrorPlane(
+        dec_buffer.DataV(), dec_buffer.StrideV(), ref_buffer.DataV(),
+        ref_buffer.StrideV(), dec_buffer.width() / 2, dec_buffer.height() / 2);
+
+    const size_t num_y_samples = dec_buffer.width() * dec_buffer.height();
+    const size_t num_u_samples =
+        dec_buffer.width() / 2 * dec_buffer.height() / 2;
+
+    frame_stat->psnr_y = libyuv::SumSquareErrorToPsnr(sse_y, num_y_samples);
+    frame_stat->psnr_u = libyuv::SumSquareErrorToPsnr(sse_u, num_u_samples);
+    frame_stat->psnr_v = libyuv::SumSquareErrorToPsnr(sse_v, num_u_samples);
+    frame_stat->psnr = libyuv::SumSquareErrorToPsnr(
+        sse_y + sse_u + sse_v, num_y_samples + 2 * num_u_samples);
+
+    if (calc_ssim) {
+      frame_stat->ssim = I420SSIM(ref_buffer, dec_buffer);
+    }
+  }
+}
+
+}  // namespace
+
+VideoProcessor::VideoProcessor(webrtc::VideoEncoder* encoder,
+                               VideoDecoderList* decoders,
+                               FrameReader* input_frame_reader,
+                               const VideoCodecTestFixture::Config& config,
+                               VideoCodecTestStatsImpl* stats,
+                               IvfFileWriterMap* encoded_frame_writers,
+                               FrameWriterList* decoded_frame_writers)
+    : config_(config),
+      num_simulcast_or_spatial_layers_(
+          std::max(config_.NumberOfSimulcastStreams(),
+                   config_.NumberOfSpatialLayers())),
+      analyze_frame_quality_(!config_.measure_cpu),
+      stats_(stats),
+      encoder_(encoder),
+      decoders_(decoders),
+      bitrate_allocator_(
+          CreateBuiltinVideoBitrateAllocatorFactory()
+              ->CreateVideoBitrateAllocator(config_.codec_settings)),
+      encode_callback_(this),
+      input_frame_reader_(input_frame_reader),
+      merged_encoded_frames_(num_simulcast_or_spatial_layers_),
+      encoded_frame_writers_(encoded_frame_writers),
+      decoded_frame_writers_(decoded_frame_writers),
+      last_inputed_frame_num_(0),
+      last_inputed_timestamp_(0),
+      first_encoded_frame_(num_simulcast_or_spatial_layers_, true),
+      last_encoded_frame_num_(num_simulcast_or_spatial_layers_),
+      first_decoded_frame_(num_simulcast_or_spatial_layers_, true),
+      last_decoded_frame_num_(num_simulcast_or_spatial_layers_),
+      last_decoded_frame_buffer_(num_simulcast_or_spatial_layers_),
+      post_encode_time_ns_(0),
+      is_finalized_(false) {
+  // Sanity checks.
+  RTC_CHECK(TaskQueueBase::Current())
+      << "VideoProcessor must be run on a task queue.";
+  RTC_CHECK(stats_);
+  RTC_CHECK(encoder_);
+  RTC_CHECK(decoders_);
+  RTC_CHECK_EQ(decoders_->size(), num_simulcast_or_spatial_layers_);
+  RTC_CHECK(input_frame_reader_);
+  RTC_CHECK(encoded_frame_writers_);
+  RTC_CHECK(!decoded_frame_writers ||
+            decoded_frame_writers->size() == num_simulcast_or_spatial_layers_);
+
+  // Setup required callbacks for the encoder and decoder and initialize them.
+  RTC_CHECK_EQ(encoder_->RegisterEncodeCompleteCallback(&encode_callback_),
+               WEBRTC_VIDEO_CODEC_OK);
+
+  // Initialize codecs so that they are ready to receive frames.
+  RTC_CHECK_EQ(encoder_->InitEncode(
+                   &config_.codec_settings,
+                   VideoEncoder::Settings(
+                       kCapabilities, static_cast<int>(config_.NumberOfCores()),
+                       config_.max_payload_size_bytes)),
+               WEBRTC_VIDEO_CODEC_OK);
+
+  for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
+    decode_callback_.push_back(
+        std::make_unique<VideoProcessorDecodeCompleteCallback>(this, i));
+    VideoDecoder::Settings decoder_settings;
+    decoder_settings.set_max_render_resolution(
+        {config_.codec_settings.width, config_.codec_settings.height});
+    decoder_settings.set_codec_type(config_.codec_settings.codecType);
+    decoder_settings.set_number_of_cores(config_.NumberOfCores());
+    RTC_CHECK(decoders_->at(i)->Configure(decoder_settings));
+    RTC_CHECK_EQ(decoders_->at(i)->RegisterDecodeCompleteCallback(
+                     decode_callback_.at(i).get()),
+                 WEBRTC_VIDEO_CODEC_OK);
+  }
+}
+
+VideoProcessor::~VideoProcessor() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  if (!is_finalized_) {
+    Finalize();
+  }
+
+  // Explicitly reset codecs, in case they don't do that themselves when they
+  // go out of scope.
+  RTC_CHECK_EQ(encoder_->Release(), WEBRTC_VIDEO_CODEC_OK);
+  encoder_->RegisterEncodeCompleteCallback(nullptr);
+  for (auto& decoder : *decoders_) {
+    RTC_CHECK_EQ(decoder->Release(), WEBRTC_VIDEO_CODEC_OK);
+    decoder->RegisterDecodeCompleteCallback(nullptr);
+  }
+
+  // Sanity check.
+  RTC_CHECK_LE(input_frames_.size(), kMaxBufferedInputFrames);
+}
+
+void VideoProcessor::ProcessFrame() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  RTC_DCHECK(!is_finalized_);
+
+  RTC_DCHECK_GT(target_rates_.size(), 0u);
+  RTC_DCHECK_EQ(target_rates_.begin()->first, 0u);
+  RateProfile target_rate =
+      std::prev(target_rates_.upper_bound(last_inputed_frame_num_))->second;
+
+  const size_t frame_number = last_inputed_frame_num_++;
+
+  // Get input frame and store for future quality calculation.
+  Resolution resolution = Resolution({.width = config_.codec_settings.width,
+                                      .height = config_.codec_settings.height});
+  FrameReader::Ratio framerate_scale = FrameReader::Ratio(
+      {.num = config_.clip_fps.value_or(config_.codec_settings.maxFramerate),
+       .den = static_cast<int>(config_.codec_settings.maxFramerate)});
+  rtc::scoped_refptr<I420BufferInterface> buffer =
+      input_frame_reader_->PullFrame(
+          /*frame_num*/ nullptr, resolution, framerate_scale);
+
+  RTC_CHECK(buffer) << "Tried to read too many frames from the file.";
+  const size_t timestamp =
+      last_inputed_timestamp_ +
+      static_cast<size_t>(kVideoPayloadTypeFrequency / target_rate.input_fps);
+  VideoFrame input_frame =
+      VideoFrame::Builder()
+          .set_video_frame_buffer(buffer)
+          .set_timestamp_rtp(static_cast<uint32_t>(timestamp))
+          .set_timestamp_ms(static_cast<int64_t>(timestamp / kMsToRtpTimestamp))
+          .set_rotation(webrtc::kVideoRotation_0)
+          .build();
+  // Store input frame as a reference for quality calculations.
+  if (config_.decode && !config_.measure_cpu) {
+    if (input_frames_.size() == kMaxBufferedInputFrames) {
+      input_frames_.erase(input_frames_.begin());
+    }
+
+    if (config_.reference_width != -1 && config_.reference_height != -1 &&
+        (input_frame.width() != config_.reference_width ||
+         input_frame.height() != config_.reference_height)) {
+      rtc::scoped_refptr<I420Buffer> scaled_buffer = I420Buffer::Create(
+          config_.codec_settings.width, config_.codec_settings.height);
+      scaled_buffer->ScaleFrom(*input_frame.video_frame_buffer()->ToI420());
+
+      VideoFrame scaled_reference_frame = input_frame;
+      scaled_reference_frame.set_video_frame_buffer(scaled_buffer);
+      input_frames_.emplace(frame_number, scaled_reference_frame);
+
+      if (config_.reference_width == config_.codec_settings.width &&
+          config_.reference_height == config_.codec_settings.height) {
+        // Both encoding and comparison uses the same down-scale factor, reuse
+        // it for encoder below.
+        input_frame = scaled_reference_frame;
+      }
+    } else {
+      input_frames_.emplace(frame_number, input_frame);
+    }
+  }
+  last_inputed_timestamp_ = timestamp;
+
+  post_encode_time_ns_ = 0;
+
+  // Create frame statistics object for all simulcast/spatial layers.
+  for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
+    FrameStatistics frame_stat(frame_number, timestamp, i);
+    stats_->AddFrame(frame_stat);
+  }
+
+  // For the highest measurement accuracy of the encode time, the start/stop
+  // time recordings should wrap the Encode call as tightly as possible.
+  const int64_t encode_start_ns = rtc::TimeNanos();
+  for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
+    FrameStatistics* frame_stat = stats_->GetFrame(frame_number, i);
+    frame_stat->encode_start_ns = encode_start_ns;
+  }
+
+  if (input_frame.width() != config_.codec_settings.width ||
+      input_frame.height() != config_.codec_settings.height) {
+    rtc::scoped_refptr<I420Buffer> scaled_buffer = I420Buffer::Create(
+        config_.codec_settings.width, config_.codec_settings.height);
+    scaled_buffer->ScaleFrom(*input_frame.video_frame_buffer()->ToI420());
+    input_frame.set_video_frame_buffer(scaled_buffer);
+  }
+
+  // Encode.
+  const std::vector<VideoFrameType> frame_types =
+      (frame_number == 0)
+          ? std::vector<VideoFrameType>(num_simulcast_or_spatial_layers_,
+                                        VideoFrameType::kVideoFrameKey)
+          : std::vector<VideoFrameType>(num_simulcast_or_spatial_layers_,
+                                        VideoFrameType::kVideoFrameDelta);
+  const int encode_return_code = encoder_->Encode(input_frame, &frame_types);
+  for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
+    FrameStatistics* frame_stat = stats_->GetFrame(frame_number, i);
+    frame_stat->encode_return_code = encode_return_code;
+  }
+}
+
+void VideoProcessor::SetRates(size_t bitrate_kbps, double framerate_fps) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  RTC_DCHECK(!is_finalized_);
+
+  target_rates_[last_inputed_frame_num_] =
+      RateProfile({.target_kbps = bitrate_kbps, .input_fps = framerate_fps});
+
+  auto bitrate_allocation =
+      bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
+          static_cast<uint32_t>(bitrate_kbps * 1000), framerate_fps));
+  encoder_->SetRates(
+      VideoEncoder::RateControlParameters(bitrate_allocation, framerate_fps));
+}
+
+int32_t VideoProcessor::VideoProcessorDecodeCompleteCallback::Decoded(
+    VideoFrame& image) {
+  // Post the callback to the right task queue, if needed.
+  if (!task_queue_->IsCurrent()) {
+    // There might be a limited amount of output buffers, make a copy to make
+    // sure we don't block the decoder.
+    VideoFrame copy = VideoFrame::Builder()
+                          .set_video_frame_buffer(I420Buffer::Copy(
+                              *image.video_frame_buffer()->ToI420()))
+                          .set_rotation(image.rotation())
+                          .set_timestamp_us(image.timestamp_us())
+                          .set_id(image.id())
+                          .build();
+    copy.set_timestamp(image.timestamp());
+
+    task_queue_->PostTask([this, copy]() {
+      video_processor_->FrameDecoded(copy, simulcast_svc_idx_);
+    });
+    return 0;
+  }
+  video_processor_->FrameDecoded(image, simulcast_svc_idx_);
+  return 0;
+}
+
+void VideoProcessor::FrameEncoded(
+    const webrtc::EncodedImage& encoded_image,
+    const webrtc::CodecSpecificInfo& codec_specific) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // For the highest measurement accuracy of the encode time, the start/stop
+  // time recordings should wrap the Encode call as tightly as possible.
+  const int64_t encode_stop_ns = rtc::TimeNanos();
+
+  const VideoCodecType codec_type = codec_specific.codecType;
+  if (config_.encoded_frame_checker) {
+    config_.encoded_frame_checker->CheckEncodedFrame(codec_type, encoded_image);
+  }
+
+  // Layer metadata.
+  size_t spatial_idx = encoded_image.SpatialIndex().value_or(0);
+  size_t temporal_idx = GetTemporalLayerIndex(codec_specific);
+
+  FrameStatistics* frame_stat =
+      stats_->GetFrameWithTimestamp(encoded_image.Timestamp(), spatial_idx);
+  const size_t frame_number = frame_stat->frame_number;
+
+  // Ensure that the encode order is monotonically increasing, within this
+  // simulcast/spatial layer.
+  RTC_CHECK(first_encoded_frame_[spatial_idx] ||
+            last_encoded_frame_num_[spatial_idx] < frame_number);
+
+  // Ensure SVC spatial layers are delivered in ascending order.
+  const size_t num_spatial_layers = config_.NumberOfSpatialLayers();
+  if (!first_encoded_frame_[spatial_idx] && num_spatial_layers > 1) {
+    for (size_t i = 0; i < spatial_idx; ++i) {
+      RTC_CHECK_LE(last_encoded_frame_num_[i], frame_number);
+    }
+    for (size_t i = spatial_idx + 1; i < num_simulcast_or_spatial_layers_;
+         ++i) {
+      RTC_CHECK_GT(frame_number, last_encoded_frame_num_[i]);
+    }
+  }
+  first_encoded_frame_[spatial_idx] = false;
+  last_encoded_frame_num_[spatial_idx] = frame_number;
+
+  RateProfile target_rate =
+      std::prev(target_rates_.upper_bound(frame_number))->second;
+  auto bitrate_allocation =
+      bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
+          static_cast<uint32_t>(target_rate.target_kbps * 1000),
+          target_rate.input_fps));
+
+  // Update frame statistics.
+  frame_stat->encoding_successful = true;
+  frame_stat->encode_time_us = GetElapsedTimeMicroseconds(
+      frame_stat->encode_start_ns, encode_stop_ns - post_encode_time_ns_);
+  frame_stat->target_bitrate_kbps =
+      bitrate_allocation.GetTemporalLayerSum(spatial_idx, temporal_idx) / 1000;
+  frame_stat->target_framerate_fps = target_rate.input_fps;
+  frame_stat->length_bytes = encoded_image.size();
+  frame_stat->frame_type = encoded_image._frameType;
+  frame_stat->temporal_idx = temporal_idx;
+  frame_stat->max_nalu_size_bytes = GetMaxNaluSizeBytes(encoded_image, config_);
+  frame_stat->qp = encoded_image.qp_;
+
+  if (codec_type == kVideoCodecVP9) {
+    const CodecSpecificInfoVP9& vp9_info = codec_specific.codecSpecific.VP9;
+    frame_stat->inter_layer_predicted = vp9_info.inter_layer_predicted;
+    frame_stat->non_ref_for_inter_layer_pred =
+        vp9_info.non_ref_for_inter_layer_pred;
+  } else {
+    frame_stat->inter_layer_predicted = false;
+    frame_stat->non_ref_for_inter_layer_pred = true;
+  }
+
+  const webrtc::EncodedImage* encoded_image_for_decode = &encoded_image;
+  if (config_.decode || !encoded_frame_writers_->empty()) {
+    if (num_spatial_layers > 1) {
+      encoded_image_for_decode = BuildAndStoreSuperframe(
+          encoded_image, codec_type, frame_number, spatial_idx,
+          frame_stat->inter_layer_predicted);
+    }
+  }
+
+  if (config_.decode) {
+    DecodeFrame(*encoded_image_for_decode, spatial_idx);
+
+    if (codec_specific.end_of_picture && num_spatial_layers > 1) {
+      // If inter-layer prediction is enabled and upper layer was dropped then
+      // base layer should be passed to upper layer decoder. Otherwise decoder
+      // won't be able to decode next superframe.
+      const EncodedImage* base_image = nullptr;
+      const FrameStatistics* base_stat = nullptr;
+      for (size_t i = 0; i < num_spatial_layers; ++i) {
+        const bool layer_dropped = (first_decoded_frame_[i] ||
+                                    last_decoded_frame_num_[i] < frame_number);
+
+        // Ensure current layer was decoded.
+        RTC_CHECK(layer_dropped == false || i != spatial_idx);
+
+        if (!layer_dropped) {
+          base_image = &merged_encoded_frames_[i];
+          base_stat =
+              stats_->GetFrameWithTimestamp(encoded_image.Timestamp(), i);
+        } else if (base_image && !base_stat->non_ref_for_inter_layer_pred) {
+          DecodeFrame(*base_image, i);
+        }
+      }
+    }
+  } else {
+    frame_stat->decode_return_code = WEBRTC_VIDEO_CODEC_NO_OUTPUT;
+  }
+
+  // Since frames in higher TLs typically depend on frames in lower TLs,
+  // write out frames in lower TLs to bitstream dumps of higher TLs.
+  for (size_t write_temporal_idx = temporal_idx;
+       write_temporal_idx < config_.NumberOfTemporalLayers();
+       ++write_temporal_idx) {
+    const VideoProcessor::LayerKey layer_key(spatial_idx, write_temporal_idx);
+    auto it = encoded_frame_writers_->find(layer_key);
+    if (it != encoded_frame_writers_->cend()) {
+      RTC_CHECK(it->second->WriteFrame(*encoded_image_for_decode,
+                                       config_.codec_settings.codecType));
+    }
+  }
+
+  if (!config_.encode_in_real_time) {
+    // To get pure encode time for next layers, measure time spent in encode
+    // callback and subtract it from encode time of next layers.
+    post_encode_time_ns_ += rtc::TimeNanos() - encode_stop_ns;
+  }
+}
+
+void VideoProcessor::CalcFrameQuality(const I420BufferInterface& decoded_frame,
+                                      FrameStatistics* frame_stat) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  const auto reference_frame = input_frames_.find(frame_stat->frame_number);
+  RTC_CHECK(reference_frame != input_frames_.cend())
+      << "The codecs are either buffering too much, dropping too much, or "
+         "being too slow relative to the input frame rate.";
+
+  // SSIM calculation is not optimized. Skip it in real-time mode.
+  const bool calc_ssim = !config_.encode_in_real_time;
+  CalculateFrameQuality(*reference_frame->second.video_frame_buffer()->ToI420(),
+                        decoded_frame, frame_stat, calc_ssim);
+
+  frame_stat->quality_analysis_successful = true;
+}
+
+void VideoProcessor::WriteDecodedFrame(const I420BufferInterface& decoded_frame,
+                                       FrameWriter& frame_writer) {
+  int input_video_width = config_.codec_settings.width;
+  int input_video_height = config_.codec_settings.height;
+
+  rtc::scoped_refptr<I420Buffer> scaled_buffer;
+  const I420BufferInterface* scaled_frame;
+
+  if (decoded_frame.width() == input_video_width &&
+      decoded_frame.height() == input_video_height) {
+    scaled_frame = &decoded_frame;
+  } else {
+    EXPECT_DOUBLE_EQ(
+        static_cast<double>(input_video_width) / input_video_height,
+        static_cast<double>(decoded_frame.width()) / decoded_frame.height());
+
+    scaled_buffer = I420Buffer::Create(input_video_width, input_video_height);
+    scaled_buffer->ScaleFrom(decoded_frame);
+
+    scaled_frame = scaled_buffer.get();
+  }
+
+  // Ensure there is no padding.
+  RTC_CHECK_EQ(scaled_frame->StrideY(), input_video_width);
+  RTC_CHECK_EQ(scaled_frame->StrideU(), input_video_width / 2);
+  RTC_CHECK_EQ(scaled_frame->StrideV(), input_video_width / 2);
+
+  RTC_CHECK_EQ(3 * input_video_width * input_video_height / 2,
+               frame_writer.FrameLength());
+
+  RTC_CHECK(frame_writer.WriteFrame(scaled_frame->DataY()));
+}
+
+void VideoProcessor::FrameDecoded(const VideoFrame& decoded_frame,
+                                  size_t spatial_idx) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // For the highest measurement accuracy of the decode time, the start/stop
+  // time recordings should wrap the Decode call as tightly as possible.
+  const int64_t decode_stop_ns = rtc::TimeNanos();
+
+  FrameStatistics* frame_stat =
+      stats_->GetFrameWithTimestamp(decoded_frame.timestamp(), spatial_idx);
+  const size_t frame_number = frame_stat->frame_number;
+
+  if (!first_decoded_frame_[spatial_idx]) {
+    for (size_t dropped_frame_number = last_decoded_frame_num_[spatial_idx] + 1;
+         dropped_frame_number < frame_number; ++dropped_frame_number) {
+      FrameStatistics* dropped_frame_stat =
+          stats_->GetFrame(dropped_frame_number, spatial_idx);
+
+      if (analyze_frame_quality_ && config_.analyze_quality_of_dropped_frames) {
+        // Calculate frame quality comparing input frame with last decoded one.
+        CalcFrameQuality(*last_decoded_frame_buffer_[spatial_idx],
+                         dropped_frame_stat);
+      }
+
+      if (decoded_frame_writers_ != nullptr) {
+        // Fill drops with last decoded frame to make them look like freeze at
+        // playback and to keep decoded layers in sync.
+        WriteDecodedFrame(*last_decoded_frame_buffer_[spatial_idx],
+                          *decoded_frame_writers_->at(spatial_idx));
+      }
+    }
+  }
+
+  // Ensure that the decode order is monotonically increasing, within this
+  // simulcast/spatial layer.
+  RTC_CHECK(first_decoded_frame_[spatial_idx] ||
+            last_decoded_frame_num_[spatial_idx] < frame_number);
+  first_decoded_frame_[spatial_idx] = false;
+  last_decoded_frame_num_[spatial_idx] = frame_number;
+
+  // Update frame statistics.
+  frame_stat->decoding_successful = true;
+  frame_stat->decode_time_us =
+      GetElapsedTimeMicroseconds(frame_stat->decode_start_ns, decode_stop_ns);
+  frame_stat->decoded_width = decoded_frame.width();
+  frame_stat->decoded_height = decoded_frame.height();
+
+  // Skip quality metrics calculation to not affect CPU usage.
+  if (analyze_frame_quality_ || decoded_frame_writers_) {
+    // Save last decoded frame to handle possible future drops.
+    rtc::scoped_refptr<I420BufferInterface> i420buffer =
+        decoded_frame.video_frame_buffer()->ToI420();
+
+    // Copy decoded frame to a buffer without padding/stride such that we can
+    // dump Y, U and V planes into a file in one shot.
+    last_decoded_frame_buffer_[spatial_idx] = I420Buffer::Copy(
+        i420buffer->width(), i420buffer->height(), i420buffer->DataY(),
+        i420buffer->StrideY(), i420buffer->DataU(), i420buffer->StrideU(),
+        i420buffer->DataV(), i420buffer->StrideV());
+  }
+
+  if (analyze_frame_quality_) {
+    CalcFrameQuality(*decoded_frame.video_frame_buffer()->ToI420(), frame_stat);
+  }
+
+  if (decoded_frame_writers_ != nullptr) {
+    WriteDecodedFrame(*last_decoded_frame_buffer_[spatial_idx],
+                      *decoded_frame_writers_->at(spatial_idx));
+  }
+
+  // Erase all buffered input frames that we have moved past for all
+  // simulcast/spatial layers. Never buffer more than
+  // `kMaxBufferedInputFrames` frames, to protect against long runs of
+  // consecutive frame drops for a particular layer.
+  const auto min_last_decoded_frame_num = std::min_element(
+      last_decoded_frame_num_.cbegin(), last_decoded_frame_num_.cend());
+  const size_t min_buffered_frame_num =
+      std::max(0, static_cast<int>(frame_number) - kMaxBufferedInputFrames + 1);
+  RTC_CHECK(min_last_decoded_frame_num != last_decoded_frame_num_.cend());
+  const auto input_frames_erase_before = input_frames_.lower_bound(
+      std::max(*min_last_decoded_frame_num, min_buffered_frame_num));
+  input_frames_.erase(input_frames_.cbegin(), input_frames_erase_before);
+}
+
+void VideoProcessor::DecodeFrame(const EncodedImage& encoded_image,
+                                 size_t spatial_idx) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  FrameStatistics* frame_stat =
+      stats_->GetFrameWithTimestamp(encoded_image.Timestamp(), spatial_idx);
+
+  frame_stat->decode_start_ns = rtc::TimeNanos();
+  frame_stat->decode_return_code =
+      decoders_->at(spatial_idx)->Decode(encoded_image, false, 0);
+}
+
+const webrtc::EncodedImage* VideoProcessor::BuildAndStoreSuperframe(
+    const EncodedImage& encoded_image,
+    const VideoCodecType codec,
+    size_t frame_number,
+    size_t spatial_idx,
+    bool inter_layer_predicted) {
+  // Should only be called for SVC.
+  RTC_CHECK_GT(config_.NumberOfSpatialLayers(), 1);
+
+  EncodedImage base_image;
+  RTC_CHECK_EQ(base_image.size(), 0);
+
+  // Each SVC layer is decoded with dedicated decoder. Find the nearest
+  // non-dropped base frame and merge it and current frame into superframe.
+  if (inter_layer_predicted) {
+    for (int base_idx = static_cast<int>(spatial_idx) - 1; base_idx >= 0;
+         --base_idx) {
+      EncodedImage lower_layer = merged_encoded_frames_.at(base_idx);
+      if (lower_layer.Timestamp() == encoded_image.Timestamp()) {
+        base_image = lower_layer;
+        break;
+      }
+    }
+  }
+  const size_t payload_size_bytes = base_image.size() + encoded_image.size();
+
+  auto buffer = EncodedImageBuffer::Create(payload_size_bytes);
+  if (base_image.size()) {
+    RTC_CHECK(base_image.data());
+    memcpy(buffer->data(), base_image.data(), base_image.size());
+  }
+  memcpy(buffer->data() + base_image.size(), encoded_image.data(),
+         encoded_image.size());
+
+  EncodedImage copied_image = encoded_image;
+  copied_image.SetEncodedData(buffer);
+  if (base_image.size())
+    copied_image._frameType = base_image._frameType;
+
+  // Replace previous EncodedImage for this spatial layer.
+  merged_encoded_frames_.at(spatial_idx) = std::move(copied_image);
+
+  return &merged_encoded_frames_.at(spatial_idx);
+}
+
+void VideoProcessor::Finalize() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  RTC_DCHECK(!is_finalized_);
+  is_finalized_ = true;
+
+  if (!(analyze_frame_quality_ && config_.analyze_quality_of_dropped_frames) &&
+      decoded_frame_writers_ == nullptr) {
+    return;
+  }
+
+  for (size_t spatial_idx = 0; spatial_idx < num_simulcast_or_spatial_layers_;
+       ++spatial_idx) {
+    if (first_decoded_frame_[spatial_idx]) {
+      continue;  // No decoded frames on this spatial layer.
+    }
+
+    for (size_t dropped_frame_number = last_decoded_frame_num_[spatial_idx] + 1;
+         dropped_frame_number < last_inputed_frame_num_;
+         ++dropped_frame_number) {
+      FrameStatistics* frame_stat =
+          stats_->GetFrame(dropped_frame_number, spatial_idx);
+
+      RTC_DCHECK(!frame_stat->decoding_successful);
+
+      if (analyze_frame_quality_ && config_.analyze_quality_of_dropped_frames) {
+        CalcFrameQuality(*last_decoded_frame_buffer_[spatial_idx], frame_stat);
+      }
+
+      if (decoded_frame_writers_ != nullptr) {
+        WriteDecodedFrame(*last_decoded_frame_buffer_[spatial_idx],
+                          *decoded_frame_writers_->at(spatial_idx));
+      }
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videoprocessor.h b/third_party/libwebrtc/modules/video_coding/codecs/test/videoprocessor.h
new file mode 100644
index 0000000000..0a5fdf8622
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videoprocessor.h
@@ -0,0 +1,263 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
+#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/videocodec_test_fixture.h"
+#include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/resolution.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_bitrate_allocator.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/include/module_common_types.h"
+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/utility/ivf_file_writer.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/thread_annotations.h"
+#include "test/testsupport/frame_reader.h"
+#include "test/testsupport/frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+// Handles encoding/decoding of video using the VideoEncoder/VideoDecoder
+// interfaces. This is done in a sequential manner in order to be able to
+// measure times properly.
+// The class processes a frame at the time for the configured input file.
+// It maintains state of where in the source input file the processing is at.
+class VideoProcessor {
+ public:
+  using VideoDecoderList = std::vector<std::unique_ptr<VideoDecoder>>;
+  using LayerKey = std::pair<int /* spatial_idx */, int /* temporal_idx */>;
+  using IvfFileWriterMap = std::map<LayerKey, std::unique_ptr<IvfFileWriter>>;
+  // TODO(brandtr): Consider changing FrameWriterList to be a FrameWriterMap,
+  // to be able to save different TLs separately.
+  using FrameWriterList = std::vector<std::unique_ptr<FrameWriter>>;
+  using FrameStatistics = VideoCodecTestStats::FrameStatistics;
+
+  VideoProcessor(webrtc::VideoEncoder* encoder,
+                 VideoDecoderList* decoders,
+                 FrameReader* input_frame_reader,
+                 const VideoCodecTestFixture::Config& config,
+                 VideoCodecTestStatsImpl* stats,
+                 IvfFileWriterMap* encoded_frame_writers,
+                 FrameWriterList* decoded_frame_writers);
+  ~VideoProcessor();
+
+  VideoProcessor(const VideoProcessor&) = delete;
+  VideoProcessor& operator=(const VideoProcessor&) = delete;
+
+  // Reads a frame and sends it to the encoder. When the encode callback
+  // is received, the encoded frame is buffered. After encoding is finished
+  // buffered frame is sent to decoder. Quality evaluation is done in
+  // the decode callback.
+  void ProcessFrame();
+
+  // Updates the encoder with target rates. Must be called at least once.
+  void SetRates(size_t bitrate_kbps, double framerate_fps);
+
+  // Signals processor to finalize frame processing and handle possible tail
+  // drops. If not called expelicitly, this will be called in dtor. It is
+  // unexpected to get ProcessFrame() or SetRates() calls after Finalize().
+  void Finalize();
+
+ private:
+  class VideoProcessorEncodeCompleteCallback
+      : public webrtc::EncodedImageCallback {
+   public:
+    explicit VideoProcessorEncodeCompleteCallback(
+        VideoProcessor* video_processor)
+        : video_processor_(video_processor),
+          task_queue_(TaskQueueBase::Current()) {
+      RTC_DCHECK(video_processor_);
+      RTC_DCHECK(task_queue_);
+    }
+
+    Result OnEncodedImage(
+        const webrtc::EncodedImage& encoded_image,
+        const webrtc::CodecSpecificInfo* codec_specific_info) override {
+      RTC_CHECK(codec_specific_info);
+
+      // Post the callback to the right task queue, if needed.
+      if (!task_queue_->IsCurrent()) {
+        VideoProcessor* video_processor = video_processor_;
+        task_queue_->PostTask([video_processor, encoded_image,
+                               codec_specific_info = *codec_specific_info] {
+          video_processor->FrameEncoded(encoded_image, codec_specific_info);
+        });
+        return Result(Result::OK, 0);
+      }
+
+      video_processor_->FrameEncoded(encoded_image, *codec_specific_info);
+      return Result(Result::OK, 0);
+    }
+
+   private:
+    VideoProcessor* const video_processor_;
+    TaskQueueBase* const task_queue_;
+  };
+
+  class VideoProcessorDecodeCompleteCallback
+      : public webrtc::DecodedImageCallback {
+   public:
+    explicit VideoProcessorDecodeCompleteCallback(
+        VideoProcessor* video_processor,
+        size_t simulcast_svc_idx)
+        : video_processor_(video_processor),
+          simulcast_svc_idx_(simulcast_svc_idx),
+          task_queue_(TaskQueueBase::Current()) {
+      RTC_DCHECK(video_processor_);
+      RTC_DCHECK(task_queue_);
+    }
+
+    int32_t Decoded(webrtc::VideoFrame& image) override;
+
+    int32_t Decoded(webrtc::VideoFrame& image,
+                    int64_t decode_time_ms) override {
+      return Decoded(image);
+    }
+
+    void Decoded(webrtc::VideoFrame& image,
+                 absl::optional<int32_t> decode_time_ms,
+                 absl::optional<uint8_t> qp) override {
+      Decoded(image);
+    }
+
+   private:
+    VideoProcessor* const video_processor_;
+    const size_t simulcast_svc_idx_;
+    TaskQueueBase* const task_queue_;
+  };
+
+  // Invoked by the callback adapter when a frame has completed encoding.
+  void FrameEncoded(const webrtc::EncodedImage& encoded_image,
+                    const webrtc::CodecSpecificInfo& codec_specific);
+
+  // Invoked by the callback adapter when a frame has completed decoding.
+  void FrameDecoded(const webrtc::VideoFrame& image, size_t simulcast_svc_idx);
+
+  void DecodeFrame(const EncodedImage& encoded_image, size_t simulcast_svc_idx);
+
+  // In order to supply the SVC decoders with super frames containing all
+  // lower layer frames, we merge and store the layer frames in this method.
+  const webrtc::EncodedImage* BuildAndStoreSuperframe(
+      const EncodedImage& encoded_image,
+      VideoCodecType codec,
+      size_t frame_number,
+      size_t simulcast_svc_idx,
+      bool inter_layer_predicted) RTC_RUN_ON(sequence_checker_);
+
+  void CalcFrameQuality(const I420BufferInterface& decoded_frame,
+                        FrameStatistics* frame_stat);
+
+  void WriteDecodedFrame(const I420BufferInterface& decoded_frame,
+                         FrameWriter& frame_writer);
+
+  void HandleTailDrops();
+
+  // Test config.
+  const VideoCodecTestFixture::Config config_;
+  const size_t num_simulcast_or_spatial_layers_;
+  const bool analyze_frame_quality_;
+
+  // Frame statistics.
+  VideoCodecTestStatsImpl* const stats_;
+
+  // Codecs.
+  webrtc::VideoEncoder* const encoder_;
+  VideoDecoderList* const decoders_;
+  const std::unique_ptr<VideoBitrateAllocator> bitrate_allocator_;
+
+  // Target bitrate and framerate per frame.
+  std::map<size_t, RateProfile> target_rates_ RTC_GUARDED_BY(sequence_checker_);
+
+  // Adapters for the codec callbacks.
+  VideoProcessorEncodeCompleteCallback encode_callback_;
+  // Assign separate callback object to each decoder. This allows us to identify
+  // decoded layer in frame decode callback.
+  // simulcast_svc_idx -> decode callback.
+  std::vector<std::unique_ptr<VideoProcessorDecodeCompleteCallback>>
+      decode_callback_;
+
+  // Each call to ProcessFrame() will read one frame from `input_frame_reader_`.
+  FrameReader* const input_frame_reader_;
+
+  // Input frames are used as reference for frame quality evaluations.
+  // Async codecs might queue frames. To handle that we keep input frame
+  // and release it after corresponding coded frame is decoded and quality
+  // measurement is done.
+  // frame_number -> frame.
+  std::map<size_t, VideoFrame> input_frames_ RTC_GUARDED_BY(sequence_checker_);
+
+  // Encoder delivers coded frame layer-by-layer. We store coded frames and
+  // then, after all layers are encoded, decode them. Such separation of
+  // frame processing on superframe level simplifies encoding/decoding time
+  // measurement.
+  // simulcast_svc_idx -> merged SVC encoded frame.
+  std::vector<EncodedImage> merged_encoded_frames_
+      RTC_GUARDED_BY(sequence_checker_);
+
+  // These (optional) file writers are used to persistently store the encoded
+  // and decoded bitstreams. Each frame writer is enabled by being non-null.
+  IvfFileWriterMap* const encoded_frame_writers_;
+  FrameWriterList* const decoded_frame_writers_;
+
+  // Metadata for inputed/encoded/decoded frames. Used for frame identification,
+  // frame drop detection, etc. We assume that encoded/decoded frames are
+  // ordered within each simulcast/spatial layer, but we do not make any
+  // assumptions of frame ordering between layers.
+  size_t last_inputed_frame_num_ RTC_GUARDED_BY(sequence_checker_);
+  size_t last_inputed_timestamp_ RTC_GUARDED_BY(sequence_checker_);
+  // simulcast_svc_idx -> encode status.
+  std::vector<bool> first_encoded_frame_ RTC_GUARDED_BY(sequence_checker_);
+  // simulcast_svc_idx -> frame_number.
+  std::vector<size_t> last_encoded_frame_num_ RTC_GUARDED_BY(sequence_checker_);
+  // simulcast_svc_idx -> decode status.
+  std::vector<bool> first_decoded_frame_ RTC_GUARDED_BY(sequence_checker_);
+  // simulcast_svc_idx -> frame_number.
+  std::vector<size_t> last_decoded_frame_num_ RTC_GUARDED_BY(sequence_checker_);
+  // simulcast_svc_idx -> buffer.
+  std::vector<rtc::scoped_refptr<I420Buffer>> last_decoded_frame_buffer_
+      RTC_GUARDED_BY(sequence_checker_);
+
+  // Time spent in frame encode callback. It is accumulated for layers and
+  // reset when frame encode starts. When next layer is encoded post-encode time
+  // is substracted from measured encode time. Thus we get pure encode time.
+  int64_t post_encode_time_ns_ RTC_GUARDED_BY(sequence_checker_);
+
+  // Indicates whether Finalize() was called or not.
+  bool is_finalized_ RTC_GUARDED_BY(sequence_checker_);
+
+  // This class must be operated on a TaskQueue.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
new file mode 100644
index 0000000000..f1774af5df
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
@@ -0,0 +1,197 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/test/videoprocessor.h"
+
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "api/test/mock_video_decoder.h"
+#include "api/test/mock_video_encoder.h"
+#include "api/test/videocodec_test_fixture.h"
+#include "api/video/i420_buffer.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/mock/mock_frame_reader.h"
+
+using ::testing::_;
+using ::testing::AllOf;
+using ::testing::Field;
+using ::testing::Property;
+using ::testing::ResultOf;
+using ::testing::Return;
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+const int kWidth = 352;
+const int kHeight = 288;
+
+}  // namespace
+
+class VideoProcessorTest : public ::testing::Test {
+ protected:
+  VideoProcessorTest() : q_("VP queue") {
+    config_.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, false, false,
+                             false, kWidth, kHeight);
+
+    decoder_mock_ = new MockVideoDecoder();
+    decoders_.push_back(std::unique_ptr<VideoDecoder>(decoder_mock_));
+
+    ExpectInit();
+    q_.SendTask(
+        [this] {
+          video_processor_ = std::make_unique<VideoProcessor>(
+              &encoder_mock_, &decoders_, &frame_reader_mock_, config_, &stats_,
+              &encoded_frame_writers_, /*decoded_frame_writers=*/nullptr);
+        });
+  }
+
+  ~VideoProcessorTest() {
+    q_.SendTask([this] { video_processor_.reset(); });
+  }
+
+  void ExpectInit() {
+    EXPECT_CALL(encoder_mock_, InitEncode(_, _));
+    EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback);
+    EXPECT_CALL(*decoder_mock_, Configure);
+    EXPECT_CALL(*decoder_mock_, RegisterDecodeCompleteCallback);
+  }
+
+  void ExpectRelease() {
+    EXPECT_CALL(encoder_mock_, Release()).Times(1);
+    EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback(_)).Times(1);
+    EXPECT_CALL(*decoder_mock_, Release()).Times(1);
+    EXPECT_CALL(*decoder_mock_, RegisterDecodeCompleteCallback(_)).Times(1);
+  }
+
+  TaskQueueForTest q_;
+
+  VideoCodecTestFixture::Config config_;
+
+  MockVideoEncoder encoder_mock_;
+  MockVideoDecoder* decoder_mock_;
+  std::vector<std::unique_ptr<VideoDecoder>> decoders_;
+  MockFrameReader frame_reader_mock_;
+  VideoCodecTestStatsImpl stats_;
+  VideoProcessor::IvfFileWriterMap encoded_frame_writers_;
+  std::unique_ptr<VideoProcessor> video_processor_;
+};
+
+TEST_F(VideoProcessorTest, InitRelease) {
+  ExpectRelease();
+}
+
+TEST_F(VideoProcessorTest, ProcessFrames_FixedFramerate) {
+  const int kBitrateKbps = 456;
+  const int kFramerateFps = 31;
+  EXPECT_CALL(
+      encoder_mock_,
+      SetRates(Field(&VideoEncoder::RateControlParameters::framerate_fps,
+                     static_cast<double>(kFramerateFps))))
+      .Times(1);
+  q_.SendTask([=] { video_processor_->SetRates(kBitrateKbps, kFramerateFps); });
+
+  EXPECT_CALL(frame_reader_mock_, PullFrame(_, _, _))
+      .WillRepeatedly(Return(I420Buffer::Create(kWidth, kHeight)));
+  EXPECT_CALL(
+      encoder_mock_,
+      Encode(Property(&VideoFrame::timestamp, 1 * 90000 / kFramerateFps), _))
+      .Times(1);
+  q_.SendTask([this] { video_processor_->ProcessFrame(); });
+
+  EXPECT_CALL(
+      encoder_mock_,
+      Encode(Property(&VideoFrame::timestamp, 2 * 90000 / kFramerateFps), _))
+      .Times(1);
+  q_.SendTask([this] { video_processor_->ProcessFrame(); });
+
+  ExpectRelease();
+}
+
+TEST_F(VideoProcessorTest, ProcessFrames_VariableFramerate) {
+  const int kBitrateKbps = 456;
+  const int kStartFramerateFps = 27;
+  const int kStartTimestamp = 90000 / kStartFramerateFps;
+  EXPECT_CALL(
+      encoder_mock_,
+      SetRates(Field(&VideoEncoder::RateControlParameters::framerate_fps,
+                     static_cast<double>(kStartFramerateFps))))
+      .Times(1);
+  q_.SendTask(
+      [=] { video_processor_->SetRates(kBitrateKbps, kStartFramerateFps); });
+
+  EXPECT_CALL(frame_reader_mock_, PullFrame(_, _, _))
+      .WillRepeatedly(Return(I420Buffer::Create(kWidth, kHeight)));
+  EXPECT_CALL(encoder_mock_,
+              Encode(Property(&VideoFrame::timestamp, kStartTimestamp), _))
+      .Times(1);
+  q_.SendTask([this] { video_processor_->ProcessFrame(); });
+
+  const int kNewFramerateFps = 13;
+  EXPECT_CALL(
+      encoder_mock_,
+      SetRates(Field(&VideoEncoder::RateControlParameters::framerate_fps,
+                     static_cast<double>(kNewFramerateFps))))
+      .Times(1);
+  q_.SendTask(
+      [=] { video_processor_->SetRates(kBitrateKbps, kNewFramerateFps); });
+
+  EXPECT_CALL(encoder_mock_,
+              Encode(Property(&VideoFrame::timestamp,
+                              kStartTimestamp + 90000 / kNewFramerateFps),
+                     _))
+      .Times(1);
+  q_.SendTask([this] { video_processor_->ProcessFrame(); });
+
+  ExpectRelease();
+}
+
+TEST_F(VideoProcessorTest, SetRates) {
+  const uint32_t kBitrateKbps = 123;
+  const int kFramerateFps = 17;
+
+  EXPECT_CALL(
+      encoder_mock_,
+      SetRates(AllOf(ResultOf(
+                         [](const VideoEncoder::RateControlParameters& params) {
+                           return params.bitrate.get_sum_kbps();
+                         },
+                         kBitrateKbps),
+                     Field(&VideoEncoder::RateControlParameters::framerate_fps,
+                           static_cast<double>(kFramerateFps)))))
+      .Times(1);
+  q_.SendTask([=] { video_processor_->SetRates(kBitrateKbps, kFramerateFps); });
+
+  const uint32_t kNewBitrateKbps = 456;
+  const int kNewFramerateFps = 34;
+  EXPECT_CALL(
+      encoder_mock_,
+      SetRates(AllOf(ResultOf(
+                         [](const VideoEncoder::RateControlParameters& params) {
+                           return params.bitrate.get_sum_kbps();
+                         },
+                         kNewBitrateKbps),
+                     Field(&VideoEncoder::RateControlParameters::framerate_fps,
+                           static_cast<double>(kNewFramerateFps)))))
+      .Times(1);
+  q_.SendTask(
+      [=] { video_processor_->SetRates(kNewBitrateKbps, kNewFramerateFps); });
+
+  ExpectRelease();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
new file mode 100644
index 0000000000..94860da1b6
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
@@ -0,0 +1,884 @@
+/* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
+
+#include <stdlib.h>
+
+#include <algorithm>
+#include <array>
+#include <memory>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+DefaultTemporalLayers::PendingFrame::PendingFrame() = default;
+DefaultTemporalLayers::PendingFrame::PendingFrame(
+    uint32_t timestamp,
+    bool expired,
+    uint8_t updated_buffers_mask,
+    const DependencyInfo& dependency_info)
+    : timestamp(timestamp),
+      expired(expired),
+      updated_buffer_mask(updated_buffers_mask),
+      dependency_info(dependency_info) {}
+
+namespace {
+using BufferFlags = Vp8FrameConfig::BufferFlags;
+using FreezeEntropy = Vp8FrameConfig::FreezeEntropy;
+using Vp8BufferReference = Vp8FrameConfig::Vp8BufferReference;
+
+constexpr BufferFlags kNone = BufferFlags::kNone;
+constexpr BufferFlags kReference = BufferFlags::kReference;
+constexpr BufferFlags kUpdate = BufferFlags::kUpdate;
+constexpr BufferFlags kReferenceAndUpdate = BufferFlags::kReferenceAndUpdate;
+constexpr FreezeEntropy kFreezeEntropy = FreezeEntropy::kFreezeEntropy;
+
+static constexpr uint8_t kUninitializedPatternIndex =
+    std::numeric_limits<uint8_t>::max();
+static constexpr std::array<Vp8BufferReference, 3> kAllBuffers = {
+    {Vp8BufferReference::kLast, Vp8BufferReference::kGolden,
+     Vp8BufferReference::kAltref}};
+
+std::vector<unsigned int> GetTemporalIds(size_t num_layers) {
+  switch (num_layers) {
+    case 1:
+      // Temporal layer structure (single layer):
+      // 0 0 0 0 ...
+      return {0};
+    case 2:
+      // Temporal layer structure:
+      //   1   1 ...
+      // 0   0   ...
+      return {0, 1};
+    case 3:
+      // Temporal layer structure:
+      //   2   2   2   2 ...
+      //     1       1   ...
+      // 0       0       ...
+      return {0, 2, 1, 2};
+    case 4:
+      // Temporal layer structure:
+      //   3   3   3   3   3   3   3   3 ...
+      //     2       2       2       2   ...
+      //         1               1       ...
+      // 0               0               ...
+      return {0, 3, 2, 3, 1, 3, 2, 3};
+    default:
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return {0};
+}
+
+uint8_t GetUpdatedBuffers(const Vp8FrameConfig& config) {
+  uint8_t flags = 0;
+  if (config.last_buffer_flags & BufferFlags::kUpdate) {
+    flags |= static_cast<uint8_t>(Vp8BufferReference::kLast);
+  }
+  if (config.golden_buffer_flags & BufferFlags::kUpdate) {
+    flags |= static_cast<uint8_t>(Vp8BufferReference::kGolden);
+  }
+  if (config.arf_buffer_flags & BufferFlags::kUpdate) {
+    flags |= static_cast<uint8_t>(Vp8BufferReference::kAltref);
+  }
+  return flags;
+}
+
+size_t BufferToIndex(Vp8BufferReference buffer) {
+  switch (buffer) {
+    case Vp8FrameConfig::Vp8BufferReference::kLast:
+      return 0;
+    case Vp8FrameConfig::Vp8BufferReference::kGolden:
+      return 1;
+    case Vp8FrameConfig::Vp8BufferReference::kAltref:
+      return 2;
+    case Vp8FrameConfig::Vp8BufferReference::kNone:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
+}  // namespace
+
+constexpr size_t DefaultTemporalLayers::kNumReferenceBuffers;
+
+std::vector<DefaultTemporalLayers::DependencyInfo>
+DefaultTemporalLayers::GetDependencyInfo(size_t num_layers) {
+  // For indexing in the patterns described below (which temporal layers they
+  // belong to), see the diagram above.
+  // Layer sync is done similarly for all patterns (except single stream) and
+  // happens every 8 frames:
+  // TL1 layer syncs by periodically by only referencing TL0 ('last'), but still
+  // updating 'golden', so it can be used as a reference by future TL1 frames.
+  // TL2 layer syncs just before TL1 by only depending on TL0 (and not depending
+  // on TL1's buffer before TL1 has layer synced).
+  // TODO(pbos): Consider cyclically updating 'arf' (and 'golden' for 1TL) for
+  // the base layer in 1-3TL instead of 'last' periodically on long intervals,
+  // so that if scene changes occur (user walks between rooms or rotates webcam)
+  // the 'arf' (or 'golden' respectively) is not stuck on a no-longer relevant
+  // keyframe.
+
+  switch (num_layers) {
+    case 1:
+      // Always reference and update the same buffer.
+      return {{"S", {kReferenceAndUpdate, kNone, kNone}}};
+    case 2:
+      // All layers can reference but not update the 'alt' buffer, this means
+      // that the 'alt' buffer reference is effectively the last keyframe.
+      // TL0 also references and updates the 'last' buffer.
+      // TL1 also references 'last' and references and updates 'golden'.
+      if (!field_trial::IsDisabled("WebRTC-UseShortVP8TL2Pattern")) {
+        // Shortened 4-frame pattern:
+        //   1---1   1---1 ...
+        //  /   /   /   /
+        // 0---0---0---0 ...
+        return {{"SS", {kReferenceAndUpdate, kNone, kNone}},
+                {"-S", {kReference, kUpdate, kNone}},
+                {"SR", {kReferenceAndUpdate, kNone, kNone}},
+                {"-D", {kReference, kReference, kNone, kFreezeEntropy}}};
+      } else {
+        // "Default" 8-frame pattern:
+        //   1---1---1---1   1---1---1---1 ...
+        //  /   /   /   /   /   /   /   /
+        // 0---0---0---0---0---0---0---0 ...
+        return {{"SS", {kReferenceAndUpdate, kNone, kNone}},
+                {"-S", {kReference, kUpdate, kNone}},
+                {"SR", {kReferenceAndUpdate, kNone, kNone}},
+                {"-R", {kReference, kReferenceAndUpdate, kNone}},
+                {"SR", {kReferenceAndUpdate, kNone, kNone}},
+                {"-R", {kReference, kReferenceAndUpdate, kNone}},
+                {"SR", {kReferenceAndUpdate, kNone, kNone}},
+                {"-D", {kReference, kReference, kNone, kFreezeEntropy}}};
+      }
+    case 3:
+      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
+        // This field trial is intended to check if it is worth using a shorter
+        // temporal pattern, trading some coding efficiency for less risk of
+        // dropped frames.
+        // The coding efficiency will decrease somewhat since the higher layer
+        // state is more volatile, but it will be offset slightly by updating
+        // the altref buffer with TL2 frames, instead of just referencing lower
+        // layers.
+        // If a frame is dropped in a higher layer, the jitter
+        // buffer on the receive side won't be able to decode any higher layer
+        // frame until the next sync frame. So we expect a noticeable decrease
+        // in frame drops on links with high packet loss.
+
+        // TL0 references and updates the 'last' buffer.
+        // TL1  references 'last' and references and updates 'golden'.
+        // TL2 references both 'last' & 'golden' and references and updates
+        // 'arf'.
+        //     2-------2       2-------2       2
+        //    /     __/       /     __/       /
+        //   /   __1         /   __1         /
+        //  /___/           /___/           /
+        // 0---------------0---------------0-----
+        // 0   1   2   3   4   5   6   7   8   9 ...
+        return {{"SSS", {kReferenceAndUpdate, kNone, kNone}},
+                {"--S", {kReference, kNone, kUpdate}},
+                {"-DR", {kReference, kUpdate, kNone}},
+                {"--D", {kReference, kReference, kReference, kFreezeEntropy}}};
+      } else {
+        // All layers can reference but not update the 'alt' buffer, this means
+        // that the 'alt' buffer reference is effectively the last keyframe.
+        // TL0 also references and updates the 'last' buffer.
+        // TL1 also references 'last' and references and updates 'golden'.
+        // TL2 references both 'last' and 'golden' but updates no buffer.
+        //     2     __2  _____2     __2       2
+        //    /     /____/    /     /         /
+        //   /     1---------/-----1         /
+        //  /_____/         /_____/         /
+        // 0---------------0---------------0-----
+        // 0   1   2   3   4   5   6   7   8   9 ...
+        return {{"SSS", {kReferenceAndUpdate, kNone, kNone}},
+                {"--D", {kReference, kNone, kNone, kFreezeEntropy}},
+                {"-SS", {kReference, kUpdate, kNone}},
+                {"--D", {kReference, kReference, kNone, kFreezeEntropy}},
+                {"SRR", {kReferenceAndUpdate, kNone, kNone}},
+                {"--D", {kReference, kReference, kNone, kFreezeEntropy}},
+                {"-DS", {kReference, kReferenceAndUpdate, kNone}},
+                {"--D", {kReference, kReference, kNone, kFreezeEntropy}}};
+      }
+    case 4:
+      // TL0 references and updates only the 'last' buffer.
+      // TL1 references 'last' and updates and references 'golden'.
+      // TL2 references 'last' and 'golden', and references and updates 'arf'.
+      // TL3 references all buffers but update none of them.
+      // TODO(philipel): Set decode target information for this structure.
+      return {{"----", {kReferenceAndUpdate, kNone, kNone}},
+              {"----", {kReference, kNone, kNone, kFreezeEntropy}},
+              {"----", {kReference, kNone, kUpdate}},
+              {"----", {kReference, kNone, kReference, kFreezeEntropy}},
+              {"----", {kReference, kUpdate, kNone}},
+              {"----", {kReference, kReference, kReference, kFreezeEntropy}},
+              {"----", {kReference, kReference, kReferenceAndUpdate}},
+              {"----", {kReference, kReference, kReference, kFreezeEntropy}},
+              {"----", {kReferenceAndUpdate, kNone, kNone}},
+              {"----", {kReference, kReference, kReference, kFreezeEntropy}},
+              {"----", {kReference, kReference, kReferenceAndUpdate}},
+              {"----", {kReference, kReference, kReference, kFreezeEntropy}},
+              {"----", {kReference, kReferenceAndUpdate, kNone}},
+              {"----", {kReference, kReference, kReference, kFreezeEntropy}},
+              {"----", {kReference, kReference, kReferenceAndUpdate}},
+              {"----", {kReference, kReference, kReference, kFreezeEntropy}}};
+    default:
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return {{"", {kNone, kNone, kNone}}};
+}
+
+std::bitset<DefaultTemporalLayers::kNumReferenceBuffers>
+DefaultTemporalLayers::DetermineStaticBuffers(
+    const std::vector<DependencyInfo>& temporal_pattern) {
+  std::bitset<kNumReferenceBuffers> buffers;
+  buffers.set();
+  for (const DependencyInfo& info : temporal_pattern) {
+    uint8_t updated_buffers = GetUpdatedBuffers(info.frame_config);
+
+    for (Vp8BufferReference buffer : kAllBuffers) {
+      if (static_cast<uint8_t>(buffer) & updated_buffers) {
+        buffers.reset(BufferToIndex(buffer));
+      }
+    }
+  }
+  return buffers;
+}
+
+DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers)
+    : num_layers_(std::max(1, number_of_temporal_layers)),
+      temporal_ids_(GetTemporalIds(num_layers_)),
+      temporal_pattern_(GetDependencyInfo(num_layers_)),
+      is_static_buffer_(DetermineStaticBuffers(temporal_pattern_)),
+      pattern_idx_(kUninitializedPatternIndex),
+      new_bitrates_bps_(std::vector<uint32_t>(num_layers_, 0u)) {
+  RTC_CHECK_GE(kMaxTemporalStreams, number_of_temporal_layers);
+  RTC_CHECK_GE(number_of_temporal_layers, 0);
+  RTC_CHECK_LE(number_of_temporal_layers, 4);
+  // pattern_idx_ wraps around temporal_pattern_.size, this is incorrect if
+  // temporal_ids_ are ever longer. If this is no longer correct it needs to
+  // wrap at max(temporal_ids_.size(), temporal_pattern_.size()).
+  RTC_DCHECK_LE(temporal_ids_.size(), temporal_pattern_.size());
+
+  RTC_DCHECK(
+      checker_ = TemporalLayersChecker::CreateTemporalLayersChecker(
+          Vp8TemporalLayersType::kFixedPattern, number_of_temporal_layers));
+
+  // Always need to start with a keyframe, so pre-populate all frame counters.
+  frames_since_buffer_refresh_.fill(0);
+}
+
+DefaultTemporalLayers::~DefaultTemporalLayers() = default;
+
+void DefaultTemporalLayers::SetQpLimits(size_t stream_index,
+                                        int min_qp,
+                                        int max_qp) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+  // Ignore.
+}
+
+size_t DefaultTemporalLayers::StreamCount() const {
+  return 1;
+}
+
+bool DefaultTemporalLayers::SupportsEncoderFrameDropping(
+    size_t stream_index) const {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+  // This class allows the encoder drop frames as it sees fit.
+  return true;
+}
+
+void DefaultTemporalLayers::OnRatesUpdated(
+    size_t stream_index,
+    const std::vector<uint32_t>& bitrates_bps,
+    int framerate_fps) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+  RTC_DCHECK_GT(bitrates_bps.size(), 0);
+  RTC_DCHECK_LE(bitrates_bps.size(), num_layers_);
+  // `bitrates_bps` uses individual rate per layer, but Vp8EncoderConfig wants
+  // the accumulated rate, so sum them up.
+  new_bitrates_bps_ = bitrates_bps;
+  new_bitrates_bps_->resize(num_layers_);
+  for (size_t i = 1; i < num_layers_; ++i) {
+    (*new_bitrates_bps_)[i] += (*new_bitrates_bps_)[i - 1];
+  }
+}
+
+Vp8EncoderConfig DefaultTemporalLayers::UpdateConfiguration(
+    size_t stream_index) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+
+  Vp8EncoderConfig config;
+
+  if (!new_bitrates_bps_) {
+    return config;
+  }
+
+  config.temporal_layer_config.emplace();
+  Vp8EncoderConfig::TemporalLayerConfig& ts_config =
+      config.temporal_layer_config.value();
+
+  for (size_t i = 0; i < num_layers_; ++i) {
+    ts_config.ts_target_bitrate[i] = (*new_bitrates_bps_)[i] / 1000;
+    // ..., 4, 2, 1
+    ts_config.ts_rate_decimator[i] = 1 << (num_layers_ - i - 1);
+  }
+
+  ts_config.ts_number_layers = num_layers_;
+  ts_config.ts_periodicity = temporal_ids_.size();
+  std::copy(temporal_ids_.begin(), temporal_ids_.end(),
+            ts_config.ts_layer_id.begin());
+
+  new_bitrates_bps_.reset();
+
+  return config;
+}
+
+bool DefaultTemporalLayers::IsSyncFrame(const Vp8FrameConfig& config) const {
+  // Since we always assign TL0 to 'last' in these patterns, we can infer layer
+  // sync by checking if temporal id > 0 and we only reference TL0 or buffers
+  // containing the last key-frame.
+  if (config.packetizer_temporal_idx == 0) {
+    // TL0 frames are per definition not sync frames.
+    return false;
+  }
+
+  if ((config.last_buffer_flags & BufferFlags::kReference) == 0) {
+    // Sync frames must reference TL0.
+    return false;
+  }
+
+  if ((config.golden_buffer_flags & BufferFlags::kReference) &&
+      !is_static_buffer_[BufferToIndex(Vp8BufferReference::kGolden)]) {
+    // Referencing a golden frame that contains a non-(base layer|key frame).
+    return false;
+  }
+  if ((config.arf_buffer_flags & BufferFlags::kReference) &&
+      !is_static_buffer_[BufferToIndex(Vp8BufferReference::kAltref)]) {
+    // Referencing an altref frame that contains a non-(base layer|key frame).
+    return false;
+  }
+
+  return true;
+}
+
+Vp8FrameConfig DefaultTemporalLayers::NextFrameConfig(size_t stream_index,
+                                                      uint32_t timestamp) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+  RTC_DCHECK_GT(num_layers_, 0);
+  RTC_DCHECK_GT(temporal_pattern_.size(), 0);
+
+  RTC_DCHECK_GT(kUninitializedPatternIndex, temporal_pattern_.size());
+  const bool first_frame = (pattern_idx_ == kUninitializedPatternIndex);
+
+  pattern_idx_ = (pattern_idx_ + 1) % temporal_pattern_.size();
+  DependencyInfo dependency_info = temporal_pattern_[pattern_idx_];
+  Vp8FrameConfig& tl_config = dependency_info.frame_config;
+  tl_config.encoder_layer_id = tl_config.packetizer_temporal_idx =
+      temporal_ids_[pattern_idx_ % temporal_ids_.size()];
+
+  if (pattern_idx_ == 0) {
+    // Start of new pattern iteration, set up clear state by invalidating any
+    // pending frames, so that we don't make an invalid reference to a buffer
+    // containing data from a previous iteration.
+    for (auto& frame : pending_frames_) {
+      frame.expired = true;
+    }
+  }
+
+  if (first_frame) {
+    tl_config = Vp8FrameConfig::GetIntraFrameConfig();
+  } else {
+    // Last is always ok to reference as it contains the base layer. For other
+    // buffers though, we need to check if the buffer has actually been
+    // refreshed this cycle of the temporal pattern. If the encoder dropped
+    // a frame, it might not have.
+    ValidateReferences(&tl_config.golden_buffer_flags,
+                       Vp8BufferReference::kGolden);
+    ValidateReferences(&tl_config.arf_buffer_flags,
+                       Vp8BufferReference::kAltref);
+    // Update search order to let the encoder know which buffers contains the
+    // most recent data.
+    UpdateSearchOrder(&tl_config);
+    // Figure out if this a sync frame (non-base-layer frame with only
+    // base-layer references).
+    tl_config.layer_sync = IsSyncFrame(tl_config);
+
+    // Increment frame age, this needs to be in sync with `pattern_idx_`,
+    // so must update it here. Resetting age to 0 must be done when encoding is
+    // complete though, and so in the case of pipelining encoder it might lag.
+    // To prevent this data spill over into the next iteration,
+    // the `pedning_frames_` map is reset in loops. If delay is constant,
+    // the relative age should still be OK for the search order.
+    for (size_t& n : frames_since_buffer_refresh_) {
+      ++n;
+    }
+  }
+
+  // Add frame to set of pending frames, awaiting completion.
+  pending_frames_.emplace_back(timestamp, false, GetUpdatedBuffers(tl_config),
+                               dependency_info);
+
+  // Checker does not yet support encoder frame dropping, so validate flags
+  // here before they can be dropped.
+  // TODO(sprang): Update checker to support dropping.
+  RTC_DCHECK(checker_->CheckTemporalConfig(first_frame, tl_config));
+
+  return tl_config;
+}
+
+void DefaultTemporalLayers::ValidateReferences(BufferFlags* flags,
+                                               Vp8BufferReference ref) const {
+  // Check if the buffer specified by `ref` is actually referenced, and if so
+  // if it also a dynamically updating one (buffers always just containing
+  // keyframes are always safe to reference).
+  if ((*flags & BufferFlags::kReference) &&
+      !is_static_buffer_[BufferToIndex(ref)]) {
+    if (NumFramesSinceBufferRefresh(ref) >= pattern_idx_) {
+      // No valid buffer state, or buffer contains frame that is older than the
+      // current pattern. This reference is not valid, so remove it.
+      *flags = static_cast<BufferFlags>(*flags & ~BufferFlags::kReference);
+    }
+  }
+}
+
+void DefaultTemporalLayers::UpdateSearchOrder(Vp8FrameConfig* config) {
+  // Figure out which of the buffers we can reference, and order them so that
+  // the most recently refreshed is first. Otherwise prioritize last first,
+  // golden second, and altref third.
+  using BufferRefAge = std::pair<Vp8BufferReference, size_t>;
+  std::vector<BufferRefAge> eligible_buffers;
+  if (config->last_buffer_flags & BufferFlags::kReference) {
+    eligible_buffers.emplace_back(
+        Vp8BufferReference::kLast,
+        NumFramesSinceBufferRefresh(Vp8BufferReference::kLast));
+  }
+  if (config->golden_buffer_flags & BufferFlags::kReference) {
+    eligible_buffers.emplace_back(
+        Vp8BufferReference::kGolden,
+        NumFramesSinceBufferRefresh(Vp8BufferReference::kGolden));
+  }
+  if (config->arf_buffer_flags & BufferFlags::kReference) {
+    eligible_buffers.emplace_back(
+        Vp8BufferReference::kAltref,
+        NumFramesSinceBufferRefresh(Vp8BufferReference::kAltref));
+  }
+
+  std::sort(eligible_buffers.begin(), eligible_buffers.end(),
+            [](const BufferRefAge& lhs, const BufferRefAge& rhs) {
+              if (lhs.second != rhs.second) {
+                // Lower count has highest precedence.
+                return lhs.second < rhs.second;
+              }
+              return lhs.first < rhs.first;
+            });
+
+  // Populate the search order fields where possible.
+  if (!eligible_buffers.empty()) {
+    config->first_reference = eligible_buffers.front().first;
+    if (eligible_buffers.size() > 1)
+      config->second_reference = eligible_buffers[1].first;
+  }
+}
+
+size_t DefaultTemporalLayers::NumFramesSinceBufferRefresh(
+    Vp8FrameConfig::Vp8BufferReference ref) const {
+  return frames_since_buffer_refresh_[BufferToIndex(ref)];
+}
+
+void DefaultTemporalLayers::ResetNumFramesSinceBufferRefresh(
+    Vp8FrameConfig::Vp8BufferReference ref) {
+  frames_since_buffer_refresh_[BufferToIndex(ref)] = 0;
+}
+
+void DefaultTemporalLayers::CullPendingFramesBefore(uint32_t timestamp) {
+  while (!pending_frames_.empty() &&
+         pending_frames_.front().timestamp != timestamp) {
+    pending_frames_.pop_front();
+  }
+}
+
+void DefaultTemporalLayers::OnEncodeDone(size_t stream_index,
+                                         uint32_t rtp_timestamp,
+                                         size_t size_bytes,
+                                         bool is_keyframe,
+                                         int qp,
+                                         CodecSpecificInfo* info) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+  RTC_DCHECK_GT(num_layers_, 0);
+
+  if (size_bytes == 0) {
+    RTC_LOG(LS_WARNING) << "Empty frame; treating as dropped.";
+    OnFrameDropped(stream_index, rtp_timestamp);
+    return;
+  }
+
+  CullPendingFramesBefore(rtp_timestamp);
+  RTC_CHECK(!pending_frames_.empty());
+  PendingFrame& frame = pending_frames_.front();
+  RTC_DCHECK_EQ(frame.timestamp, rtp_timestamp);
+  const Vp8FrameConfig& frame_config = frame.dependency_info.frame_config;
+  if (is_keyframe) {
+    // Signal key-frame so checker resets state.
+    RTC_DCHECK(checker_->CheckTemporalConfig(true, frame_config));
+  }
+
+  CodecSpecificInfoVP8& vp8_info = info->codecSpecific.VP8;
+  if (num_layers_ == 1) {
+    vp8_info.temporalIdx = kNoTemporalIdx;
+    vp8_info.layerSync = false;
+  } else {
+    if (is_keyframe) {
+      // Restart the temporal pattern on keyframes.
+      pattern_idx_ = 0;
+      vp8_info.temporalIdx = 0;
+      vp8_info.layerSync = true;  // Keyframes are always sync frames.
+
+      for (Vp8BufferReference buffer : kAllBuffers) {
+        if (is_static_buffer_[BufferToIndex(buffer)]) {
+          // Update frame count of all kf-only buffers, regardless of state of
+          // `pending_frames_`.
+          ResetNumFramesSinceBufferRefresh(buffer);
+        } else {
+          // Key-frames update all buffers, this should be reflected when
+          // updating state in FrameEncoded().
+          frame.updated_buffer_mask |= static_cast<uint8_t>(buffer);
+        }
+      }
+    } else {
+      // Delta frame, update codec specifics with temporal id and sync flag.
+      vp8_info.temporalIdx = frame_config.packetizer_temporal_idx;
+      vp8_info.layerSync = frame_config.layer_sync;
+    }
+  }
+
+  vp8_info.useExplicitDependencies = true;
+  RTC_DCHECK_EQ(vp8_info.referencedBuffersCount, 0u);
+  RTC_DCHECK_EQ(vp8_info.updatedBuffersCount, 0u);
+
+  GenericFrameInfo& generic_frame_info = info->generic_frame_info.emplace();
+
+  for (int i = 0; i < static_cast<int>(Vp8FrameConfig::Buffer::kCount); ++i) {
+    bool references = false;
+    bool updates = is_keyframe;
+
+    if (!is_keyframe &&
+        frame_config.References(static_cast<Vp8FrameConfig::Buffer>(i))) {
+      RTC_DCHECK_LT(vp8_info.referencedBuffersCount,
+                    arraysize(CodecSpecificInfoVP8::referencedBuffers));
+      references = true;
+      vp8_info.referencedBuffers[vp8_info.referencedBuffersCount++] = i;
+    }
+
+    if (is_keyframe ||
+        frame_config.Updates(static_cast<Vp8FrameConfig::Buffer>(i))) {
+      RTC_DCHECK_LT(vp8_info.updatedBuffersCount,
+                    arraysize(CodecSpecificInfoVP8::updatedBuffers));
+      updates = true;
+      vp8_info.updatedBuffers[vp8_info.updatedBuffersCount++] = i;
+    }
+
+    if (references || updates) {
+      generic_frame_info.encoder_buffers.emplace_back(i, references, updates);
+    }
+  }
+
+  // The templates are always present on keyframes, and then refered to by
+  // subsequent frames.
+  if (is_keyframe) {
+    info->template_structure = GetTemplateStructure(num_layers_);
+    generic_frame_info.decode_target_indications =
+        temporal_pattern_.front().decode_target_indications;
+    generic_frame_info.temporal_id = 0;
+  } else {
+    generic_frame_info.decode_target_indications =
+        frame.dependency_info.decode_target_indications;
+    generic_frame_info.temporal_id = frame_config.packetizer_temporal_idx;
+  }
+
+  if (!frame.expired) {
+    for (Vp8BufferReference buffer : kAllBuffers) {
+      if (frame.updated_buffer_mask & static_cast<uint8_t>(buffer)) {
+        ResetNumFramesSinceBufferRefresh(buffer);
+      }
+    }
+  }
+
+  pending_frames_.pop_front();
+}
+
+void DefaultTemporalLayers::OnFrameDropped(size_t stream_index,
+                                           uint32_t rtp_timestamp) {
+  CullPendingFramesBefore(rtp_timestamp);
+  RTC_CHECK(!pending_frames_.empty());
+  RTC_DCHECK_EQ(pending_frames_.front().timestamp, rtp_timestamp);
+  pending_frames_.pop_front();
+}
+
+void DefaultTemporalLayers::OnPacketLossRateUpdate(float packet_loss_rate) {}
+
+void DefaultTemporalLayers::OnRttUpdate(int64_t rtt_ms) {}
+
+void DefaultTemporalLayers::OnLossNotification(
+    const VideoEncoder::LossNotification& loss_notification) {}
+
+FrameDependencyStructure DefaultTemporalLayers::GetTemplateStructure(
+    int num_layers) const {
+  RTC_CHECK_LT(num_layers, 5);
+  RTC_CHECK_GT(num_layers, 0);
+
+  FrameDependencyStructure template_structure;
+  template_structure.num_decode_targets = num_layers;
+
+  switch (num_layers) {
+    case 1: {
+      template_structure.templates.resize(2);
+      template_structure.templates[0].T(0).Dtis("S");
+      template_structure.templates[1].T(0).Dtis("S").FrameDiffs({1});
+      return template_structure;
+    }
+    case 2: {
+      template_structure.templates.resize(5);
+      template_structure.templates[0].T(0).Dtis("SS");
+      template_structure.templates[1].T(0).Dtis("SS").FrameDiffs({2});
+      template_structure.templates[2].T(0).Dtis("SR").FrameDiffs({2});
+      template_structure.templates[3].T(1).Dtis("-S").FrameDiffs({1});
+      template_structure.templates[4].T(1).Dtis("-D").FrameDiffs({2, 1});
+      return template_structure;
+    }
+    case 3: {
+      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
+        template_structure.templates.resize(5);
+        template_structure.templates[0].T(0).Dtis("SSS");
+        template_structure.templates[1].T(0).Dtis("SSS").FrameDiffs({4});
+        template_structure.templates[2].T(1).Dtis("-DR").FrameDiffs({2});
+        template_structure.templates[3].T(2).Dtis("--S").FrameDiffs({1});
+        template_structure.templates[4].T(2).Dtis("--D").FrameDiffs({2, 1});
+      } else {
+        template_structure.templates.resize(7);
+        template_structure.templates[0].T(0).Dtis("SSS");
+        template_structure.templates[1].T(0).Dtis("SSS").FrameDiffs({4});
+        template_structure.templates[2].T(0).Dtis("SRR").FrameDiffs({4});
+        template_structure.templates[3].T(1).Dtis("-SS").FrameDiffs({2});
+        template_structure.templates[4].T(1).Dtis("-DS").FrameDiffs({4, 2});
+        template_structure.templates[5].T(2).Dtis("--D").FrameDiffs({1});
+        template_structure.templates[6].T(2).Dtis("--D").FrameDiffs({3, 1});
+      }
+      return template_structure;
+    }
+    case 4: {
+      template_structure.templates.resize(8);
+      template_structure.templates[0].T(0).Dtis("SSSS");
+      template_structure.templates[1].T(0).Dtis("SSSS").FrameDiffs({8});
+      template_structure.templates[2].T(1).Dtis("-SRR").FrameDiffs({4});
+      template_structure.templates[3].T(1).Dtis("-SRR").FrameDiffs({4, 8});
+      template_structure.templates[4].T(2).Dtis("--SR").FrameDiffs({2});
+      template_structure.templates[5].T(2).Dtis("--SR").FrameDiffs({2, 4});
+      template_structure.templates[6].T(3).Dtis("---D").FrameDiffs({1});
+      template_structure.templates[7].T(3).Dtis("---D").FrameDiffs({1, 3});
+      return template_structure;
+    }
+    default:
+      RTC_DCHECK_NOTREACHED();
+      // To make the compiler happy!
+      return template_structure;
+  }
+}
+
+// Returns list of temporal dependencies for each frame in the temporal pattern.
+// Values are lists of indecies in the pattern.
+std::vector<std::set<uint8_t>> GetTemporalDependencies(
+    int num_temporal_layers) {
+  switch (num_temporal_layers) {
+    case 1:
+      return {{0}};
+    case 2:
+      if (!field_trial::IsDisabled("WebRTC-UseShortVP8TL2Pattern")) {
+        return {{2}, {0}, {0}, {1, 2}};
+      } else {
+        return {{6}, {0}, {0}, {1, 2}, {2}, {3, 4}, {4}, {5, 6}};
+      }
+    case 3:
+      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
+        return {{0}, {0}, {0}, {0, 1, 2}};
+      } else {
+        return {{4}, {0}, {0}, {0, 2}, {0}, {2, 4}, {2, 4}, {4, 6}};
+      }
+    case 4:
+      return {{8},    {0},         {0},         {0, 2},
+              {0},    {0, 2, 4},   {0, 2, 4},   {0, 4, 6},
+              {0},    {4, 6, 8},   {4, 6, 8},   {4, 8, 10},
+              {4, 8}, {8, 10, 12}, {8, 10, 12}, {8, 12, 14}};
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return {};
+  }
+}
+
+DefaultTemporalLayersChecker::DefaultTemporalLayersChecker(
+    int num_temporal_layers)
+    : TemporalLayersChecker(num_temporal_layers),
+      num_layers_(std::max(1, num_temporal_layers)),
+      temporal_ids_(GetTemporalIds(num_layers_)),
+      temporal_dependencies_(GetTemporalDependencies(num_layers_)),
+      pattern_idx_(255) {
+  int i = 0;
+  while (temporal_ids_.size() < temporal_dependencies_.size()) {
+    temporal_ids_.push_back(temporal_ids_[i++]);
+  }
+}
+
+DefaultTemporalLayersChecker::~DefaultTemporalLayersChecker() = default;
+
+bool DefaultTemporalLayersChecker::CheckTemporalConfig(
+    bool frame_is_keyframe,
+    const Vp8FrameConfig& frame_config) {
+  if (!TemporalLayersChecker::CheckTemporalConfig(frame_is_keyframe,
+                                                  frame_config)) {
+    return false;
+  }
+  if (frame_config.drop_frame) {
+    return true;
+  }
+
+  if (frame_is_keyframe) {
+    pattern_idx_ = 0;
+    last_ = BufferState();
+    golden_ = BufferState();
+    arf_ = BufferState();
+    return true;
+  }
+
+  ++pattern_idx_;
+  if (pattern_idx_ == temporal_ids_.size()) {
+    // All non key-frame buffers should be updated each pattern cycle.
+    if (!last_.is_keyframe && !last_.is_updated_this_cycle) {
+      RTC_LOG(LS_ERROR) << "Last buffer was not updated during pattern cycle.";
+      return false;
+    }
+    if (!arf_.is_keyframe && !arf_.is_updated_this_cycle) {
+      RTC_LOG(LS_ERROR) << "Arf buffer was not updated during pattern cycle.";
+      return false;
+    }
+    if (!golden_.is_keyframe && !golden_.is_updated_this_cycle) {
+      RTC_LOG(LS_ERROR)
+          << "Golden buffer was not updated during pattern cycle.";
+      return false;
+    }
+    last_.is_updated_this_cycle = false;
+    arf_.is_updated_this_cycle = false;
+    golden_.is_updated_this_cycle = false;
+    pattern_idx_ = 0;
+  }
+  uint8_t expected_tl_idx = temporal_ids_[pattern_idx_];
+  if (frame_config.packetizer_temporal_idx != expected_tl_idx) {
+    RTC_LOG(LS_ERROR) << "Frame has an incorrect temporal index. Expected: "
+                      << static_cast<int>(expected_tl_idx) << " Actual: "
+                      << static_cast<int>(frame_config.packetizer_temporal_idx);
+    return false;
+  }
+
+  bool need_sync = temporal_ids_[pattern_idx_] > 0 &&
+                   temporal_ids_[pattern_idx_] != kNoTemporalIdx;
+  std::vector<int> dependencies;
+
+  if (frame_config.last_buffer_flags & BufferFlags::kReference) {
+    uint8_t referenced_layer = temporal_ids_[last_.pattern_idx];
+    if (referenced_layer > 0) {
+      need_sync = false;
+    }
+    if (!last_.is_keyframe) {
+      dependencies.push_back(last_.pattern_idx);
+    }
+  } else if (frame_config.first_reference == Vp8BufferReference::kLast ||
+             frame_config.second_reference == Vp8BufferReference::kLast) {
+    RTC_LOG(LS_ERROR)
+        << "Last buffer not referenced, but present in search order.";
+    return false;
+  }
+
+  if (frame_config.arf_buffer_flags & BufferFlags::kReference) {
+    uint8_t referenced_layer = temporal_ids_[arf_.pattern_idx];
+    if (referenced_layer > 0) {
+      need_sync = false;
+    }
+    if (!arf_.is_keyframe) {
+      dependencies.push_back(arf_.pattern_idx);
+    }
+  } else if (frame_config.first_reference == Vp8BufferReference::kAltref ||
+             frame_config.second_reference == Vp8BufferReference::kAltref) {
+    RTC_LOG(LS_ERROR)
+        << "Altret buffer not referenced, but present in search order.";
+    return false;
+  }
+
+  if (frame_config.golden_buffer_flags & BufferFlags::kReference) {
+    uint8_t referenced_layer = temporal_ids_[golden_.pattern_idx];
+    if (referenced_layer > 0) {
+      need_sync = false;
+    }
+    if (!golden_.is_keyframe) {
+      dependencies.push_back(golden_.pattern_idx);
+    }
+  } else if (frame_config.first_reference == Vp8BufferReference::kGolden ||
+             frame_config.second_reference == Vp8BufferReference::kGolden) {
+    RTC_LOG(LS_ERROR)
+        << "Golden buffer not referenced, but present in search order.";
+    return false;
+  }
+
+  if (need_sync != frame_config.layer_sync) {
+    RTC_LOG(LS_ERROR) << "Sync bit is set incorrectly on a frame. Expected: "
+                      << need_sync << " Actual: " << frame_config.layer_sync;
+    return false;
+  }
+
+  if (!frame_is_keyframe) {
+    size_t i;
+    for (i = 0; i < dependencies.size(); ++i) {
+      if (temporal_dependencies_[pattern_idx_].find(dependencies[i]) ==
+          temporal_dependencies_[pattern_idx_].end()) {
+        RTC_LOG(LS_ERROR)
+            << "Illegal temporal dependency out of defined pattern "
+               "from position "
+            << static_cast<int>(pattern_idx_) << " to position "
+            << static_cast<int>(dependencies[i]);
+        return false;
+      }
+    }
+  }
+
+  if (frame_config.last_buffer_flags & BufferFlags::kUpdate) {
+    last_.is_updated_this_cycle = true;
+    last_.pattern_idx = pattern_idx_;
+    last_.is_keyframe = false;
+  }
+  if (frame_config.arf_buffer_flags & BufferFlags::kUpdate) {
+    arf_.is_updated_this_cycle = true;
+    arf_.pattern_idx = pattern_idx_;
+    arf_.is_keyframe = false;
+  }
+  if (frame_config.golden_buffer_flags & BufferFlags::kUpdate) {
+    golden_.is_updated_this_cycle = true;
+    golden_.pattern_idx = pattern_idx_;
+    golden_.is_keyframe = false;
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h b/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
new file mode 100644
index 0000000000..bc6574c54c
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
@@ -0,0 +1,168 @@
+/* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+/*
+ * This file defines classes for doing temporal layers with VP8.
+ */
+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
+#define MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <bitset>
+#include <deque>
+#include <limits>
+#include <memory>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/video_codecs/vp8_frame_config.h"
+#include "api/video_codecs/vp8_temporal_layers.h"
+#include "modules/video_coding/codecs/vp8/include/temporal_layers_checker.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+
+class DefaultTemporalLayers final : public Vp8FrameBufferController {
+ public:
+  explicit DefaultTemporalLayers(int number_of_temporal_layers);
+  ~DefaultTemporalLayers() override;
+
+  void SetQpLimits(size_t stream_index, int min_qp, int max_qp) override;
+
+  size_t StreamCount() const override;
+
+  bool SupportsEncoderFrameDropping(size_t stream_index) const override;
+
+  // Returns the recommended VP8 encode flags needed. May refresh the decoder
+  // and/or update the reference buffers.
+  Vp8FrameConfig NextFrameConfig(size_t stream_index,
+                                 uint32_t timestamp) override;
+
+  // New target bitrate, per temporal layer.
+  void OnRatesUpdated(size_t stream_index,
+                      const std::vector<uint32_t>& bitrates_bps,
+                      int framerate_fps) override;
+
+  Vp8EncoderConfig UpdateConfiguration(size_t stream_index) override;
+
+  // Callbacks methods on frame completion. OnEncodeDone() or OnFrameDropped()
+  // should be called once for each NextFrameConfig() call (using the RTP
+  // timestamp as ID), and the calls MUST be in the same order.
+  void OnEncodeDone(size_t stream_index,
+                    uint32_t rtp_timestamp,
+                    size_t size_bytes,
+                    bool is_keyframe,
+                    int qp,
+                    CodecSpecificInfo* info) override;
+  void OnFrameDropped(size_t stream_index, uint32_t rtp_timestamp) override;
+
+  void OnPacketLossRateUpdate(float packet_loss_rate) override;
+
+  void OnRttUpdate(int64_t rtt_ms) override;
+
+  void OnLossNotification(
+      const VideoEncoder::LossNotification& loss_notification) override;
+
+ private:
+  static constexpr size_t kNumReferenceBuffers = 3;  // Last, golden, altref.
+  struct DependencyInfo {
+    DependencyInfo() = default;
+    DependencyInfo(absl::string_view indication_symbols,
+                   Vp8FrameConfig frame_config)
+        : decode_target_indications(
+              webrtc_impl::StringToDecodeTargetIndications(indication_symbols)),
+          frame_config(frame_config) {}
+
+    absl::InlinedVector<DecodeTargetIndication, 10> decode_target_indications;
+    Vp8FrameConfig frame_config;
+  };
+  struct PendingFrame {
+    PendingFrame();
+    PendingFrame(uint32_t timestamp,
+                 bool expired,
+                 uint8_t updated_buffers_mask,
+                 const DependencyInfo& dependency_info);
+    uint32_t timestamp = 0;
+    // Flag indicating if this frame has expired, ie it belongs to a previous
+    // iteration of the temporal pattern.
+    bool expired = false;
+    // Bitmask of Vp8BufferReference flags, indicating which buffers this frame
+    // updates.
+    uint8_t updated_buffer_mask = 0;
+    // The frame config returned by NextFrameConfig() for this frame.
+    DependencyInfo dependency_info;
+  };
+
+  static std::vector<DependencyInfo> GetDependencyInfo(size_t num_layers);
+  static std::bitset<kNumReferenceBuffers> DetermineStaticBuffers(
+      const std::vector<DependencyInfo>& temporal_pattern);
+  bool IsSyncFrame(const Vp8FrameConfig& config) const;
+  void ValidateReferences(Vp8FrameConfig::BufferFlags* flags,
+                          Vp8FrameConfig::Vp8BufferReference ref) const;
+  void UpdateSearchOrder(Vp8FrameConfig* config);
+  size_t NumFramesSinceBufferRefresh(
+      Vp8FrameConfig::Vp8BufferReference ref) const;
+  void ResetNumFramesSinceBufferRefresh(Vp8FrameConfig::Vp8BufferReference ref);
+  void CullPendingFramesBefore(uint32_t timestamp);
+
+  const size_t num_layers_;
+  const std::vector<unsigned int> temporal_ids_;
+  const std::vector<DependencyInfo> temporal_pattern_;
+  // Per reference buffer flag indicating if it is static, meaning it is only
+  // updated by key-frames.
+  const std::bitset<kNumReferenceBuffers> is_static_buffer_;
+  FrameDependencyStructure GetTemplateStructure(int num_layers) const;
+
+  uint8_t pattern_idx_;
+  // Updated cumulative bitrates, per temporal layer.
+  absl::optional<std::vector<uint32_t>> new_bitrates_bps_;
+
+  // Status for each pending frame, in
+  std::deque<PendingFrame> pending_frames_;
+
+  // One counter per reference buffer, indicating number of frames since last
+  // refresh. For non-base-layer frames (ie golden, altref buffers), this is
+  // reset when the pattern loops.
+  std::array<size_t, kNumReferenceBuffers> frames_since_buffer_refresh_;
+
+  // Optional utility used to verify reference validity.
+  std::unique_ptr<TemporalLayersChecker> checker_;
+};
+
+class DefaultTemporalLayersChecker : public TemporalLayersChecker {
+ public:
+  explicit DefaultTemporalLayersChecker(int number_of_temporal_layers);
+  ~DefaultTemporalLayersChecker() override;
+
+  bool CheckTemporalConfig(bool frame_is_keyframe,
+                           const Vp8FrameConfig& frame_config) override;
+
+ private:
+  struct BufferState {
+    BufferState()
+        : is_updated_this_cycle(false), is_keyframe(true), pattern_idx(0) {}
+
+    bool is_updated_this_cycle;
+    bool is_keyframe;
+    uint8_t pattern_idx;
+  };
+  const size_t num_layers_;
+  std::vector<unsigned int> temporal_ids_;
+  const std::vector<std::set<uint8_t>> temporal_dependencies_;
+  BufferState last_;
+  BufferState arf_;
+  BufferState golden_;
+  uint8_t pattern_idx_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
new file mode 100644
index 0000000000..ae027a9d8a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
@@ -0,0 +1,781 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
+
+#include <cstdint>
+#include <memory>
+
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/vp8_frame_config.h"
+#include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "vpx/vp8cx.h"
+
+// TODO(bugs.webrtc.org/10582): Test the behavior of UpdateConfiguration().
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using ::testing::Each;
+
+enum {
+  kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
+                        VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF,
+  kTemporalUpdateGoldenWithoutDependency =
+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF |
+      VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateGolden =
+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateAltrefWithoutDependency =
+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF |
+      VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
+                        VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
+  kTemporalUpdateNoneNoRefAltRef =
+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
+      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
+  kTemporalUpdateNoneNoRefGolden =
+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
+      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
+  kTemporalUpdateNoneNoRefGoldenAltRef =
+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_REF_ARF |
+      VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
+  kTemporalUpdateGoldenWithoutDependencyRefAltRef =
+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
+  kTemporalUpdateLastRefAltRef =
+      VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_REF_GF,
+  kTemporalUpdateLastAndGoldenRefAltRef =
+      VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_REF_GF,
+};
+
+using BufferFlags = Vp8FrameConfig::BufferFlags;
+using Vp8BufferReference = Vp8FrameConfig::Vp8BufferReference;
+
+constexpr uint8_t kNone = static_cast<uint8_t>(Vp8BufferReference::kNone);
+constexpr uint8_t kLast = static_cast<uint8_t>(Vp8BufferReference::kLast);
+constexpr uint8_t kGolden = static_cast<uint8_t>(Vp8BufferReference::kGolden);
+constexpr uint8_t kAltref = static_cast<uint8_t>(Vp8BufferReference::kAltref);
+constexpr uint8_t kAll = kLast | kGolden | kAltref;
+
+constexpr int ToVp8CodecFlags(uint8_t referenced_buffers,
+                              uint8_t updated_buffers,
+                              bool update_entropy) {
+  return (((referenced_buffers & kLast) == 0) ? VP8_EFLAG_NO_REF_LAST : 0) |
+         (((referenced_buffers & kGolden) == 0) ? VP8_EFLAG_NO_REF_GF : 0) |
+         (((referenced_buffers & kAltref) == 0) ? VP8_EFLAG_NO_REF_ARF : 0) |
+         (((updated_buffers & kLast) == 0) ? VP8_EFLAG_NO_UPD_LAST : 0) |
+         (((updated_buffers & kGolden) == 0) ? VP8_EFLAG_NO_UPD_GF : 0) |
+         (((updated_buffers & kAltref) == 0) ? VP8_EFLAG_NO_UPD_ARF : 0) |
+         (update_entropy ? 0 : VP8_EFLAG_NO_UPD_ENTROPY);
+}
+
+constexpr int kKeyFrameFlags = ToVp8CodecFlags(kNone, kAll, true);
+
+std::vector<uint32_t> GetTemporalLayerRates(int target_bitrate_kbps,
+                                            int framerate_fps,
+                                            int num_temporal_layers) {
+  VideoCodec codec;
+  codec.codecType = VideoCodecType::kVideoCodecVP8;
+  codec.numberOfSimulcastStreams = 1;
+  codec.maxBitrate = target_bitrate_kbps;
+  codec.maxFramerate = framerate_fps;
+  codec.simulcastStream[0].targetBitrate = target_bitrate_kbps;
+  codec.simulcastStream[0].maxBitrate = target_bitrate_kbps;
+  codec.simulcastStream[0].numberOfTemporalLayers = num_temporal_layers;
+  codec.simulcastStream[0].active = true;
+  SimulcastRateAllocator allocator(codec);
+  return allocator
+      .Allocate(
+          VideoBitrateAllocationParameters(target_bitrate_kbps, framerate_fps))
+      .GetTemporalLayerAllocation(0);
+}
+
+constexpr int kDefaultBitrateBps = 500;
+constexpr int kDefaultFramerate = 30;
+constexpr int kDefaultBytesPerFrame =
+    (kDefaultBitrateBps / 8) / kDefaultFramerate;
+constexpr int kDefaultQp = 2;
+}  // namespace
+
+class TemporalLayersTest : public ::testing::Test {
+ public:
+  ~TemporalLayersTest() override = default;
+
+  CodecSpecificInfo* IgnoredCodecSpecificInfo() {
+    codec_specific_info_ = std::make_unique<CodecSpecificInfo>();
+    return codec_specific_info_.get();
+  }
+
+ private:
+  std::unique_ptr<CodecSpecificInfo> codec_specific_info_;
+};
+
+TEST_F(TemporalLayersTest, 2Layers) {
+  constexpr int kNumLayers = 2;
+  DefaultTemporalLayers tl(kNumLayers);
+  DefaultTemporalLayersChecker checker(kNumLayers);
+  tl.OnRatesUpdated(0,
+                    GetTemporalLayerRates(kDefaultBytesPerFrame,
+                                          kDefaultFramerate, kNumLayers),
+                    kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  constexpr size_t kPatternSize = 4;
+  constexpr size_t kRepetitions = 4;
+
+  const int expected_flags[kPatternSize] = {
+      ToVp8CodecFlags(kLast, kLast, true),
+      ToVp8CodecFlags(kLast, kGolden, true),
+      ToVp8CodecFlags(kLast, kLast, true),
+      ToVp8CodecFlags(kLast | kGolden, kNone, false),
+  };
+  const int expected_temporal_idx[kPatternSize] = {0, 1, 0, 1};
+  const bool expected_layer_sync[kPatternSize] = {false, true, false, false};
+
+  uint32_t timestamp = 0;
+  for (size_t i = 0; i < kPatternSize * kRepetitions; ++i) {
+    const size_t ind = i % kPatternSize;
+    const bool is_keyframe = (i == 0);
+    CodecSpecificInfo info;
+    Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+    EXPECT_EQ(is_keyframe ? kKeyFrameFlags : expected_flags[ind],
+              LibvpxVp8Encoder::EncodeFlags(tl_config))
+        << i;
+    tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, is_keyframe,
+                    kDefaultQp, &info);
+    EXPECT_TRUE(checker.CheckTemporalConfig(is_keyframe, tl_config));
+    EXPECT_EQ(expected_temporal_idx[ind], info.codecSpecific.VP8.temporalIdx);
+    EXPECT_EQ(expected_temporal_idx[ind], tl_config.packetizer_temporal_idx);
+    EXPECT_EQ(expected_temporal_idx[ind], tl_config.encoder_layer_id);
+    EXPECT_EQ(is_keyframe || expected_layer_sync[ind],
+              info.codecSpecific.VP8.layerSync);
+    EXPECT_EQ(expected_layer_sync[ind], tl_config.layer_sync);
+    timestamp += 3000;
+  }
+}
+
+TEST_F(TemporalLayersTest, 3Layers) {
+  constexpr int kNumLayers = 3;
+  DefaultTemporalLayers tl(kNumLayers);
+  DefaultTemporalLayersChecker checker(kNumLayers);
+  tl.OnRatesUpdated(0,
+                    GetTemporalLayerRates(kDefaultBytesPerFrame,
+                                          kDefaultFramerate, kNumLayers),
+                    kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  int expected_flags[16] = {
+      kTemporalUpdateLast,
+      kTemporalUpdateNoneNoRefGoldenAltRef,
+      kTemporalUpdateGoldenWithoutDependency,
+      kTemporalUpdateNoneNoRefAltRef,
+      kTemporalUpdateLast,
+      kTemporalUpdateNoneNoRefAltRef,
+      kTemporalUpdateGolden,
+      kTemporalUpdateNoneNoRefAltRef,
+      kTemporalUpdateLast,
+      kTemporalUpdateNoneNoRefGoldenAltRef,
+      kTemporalUpdateGoldenWithoutDependency,
+      kTemporalUpdateNoneNoRefAltRef,
+      kTemporalUpdateLast,
+      kTemporalUpdateNoneNoRefAltRef,
+      kTemporalUpdateGolden,
+      kTemporalUpdateNoneNoRefAltRef,
+  };
+  int expected_temporal_idx[16] = {0, 2, 1, 2, 0, 2, 1, 2,
+                                   0, 2, 1, 2, 0, 2, 1, 2};
+
+  bool expected_layer_sync[16] = {false, true,  true,  false, false, false,
+                                  false, false, false, true,  true,  false,
+                                  false, false, false, false};
+
+  unsigned int timestamp = 0;
+  for (int i = 0; i < 16; ++i) {
+    const bool is_keyframe = (i == 0);
+    CodecSpecificInfo info;
+    Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+    EXPECT_EQ(is_keyframe ? kKeyFrameFlags : expected_flags[i],
+              LibvpxVp8Encoder::EncodeFlags(tl_config))
+        << i;
+    tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, is_keyframe,
+                    kDefaultQp, &info);
+    EXPECT_TRUE(checker.CheckTemporalConfig(is_keyframe, tl_config));
+    EXPECT_EQ(expected_temporal_idx[i], info.codecSpecific.VP8.temporalIdx);
+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
+    EXPECT_EQ(is_keyframe || expected_layer_sync[i],
+              info.codecSpecific.VP8.layerSync);
+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
+    timestamp += 3000;
+  }
+}
+
+TEST_F(TemporalLayersTest, Alternative3Layers) {
+  constexpr int kNumLayers = 3;
+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
+  DefaultTemporalLayers tl(kNumLayers);
+  DefaultTemporalLayersChecker checker(kNumLayers);
+  tl.OnRatesUpdated(0,
+                    GetTemporalLayerRates(kDefaultBytesPerFrame,
+                                          kDefaultFramerate, kNumLayers),
+                    kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  int expected_flags[8] = {kTemporalUpdateLast,
+                           kTemporalUpdateAltrefWithoutDependency,
+                           kTemporalUpdateGoldenWithoutDependency,
+                           kTemporalUpdateNone,
+                           kTemporalUpdateLast,
+                           kTemporalUpdateAltrefWithoutDependency,
+                           kTemporalUpdateGoldenWithoutDependency,
+                           kTemporalUpdateNone};
+  int expected_temporal_idx[8] = {0, 2, 1, 2, 0, 2, 1, 2};
+
+  bool expected_layer_sync[8] = {false, true, true, false,
+                                 false, true, true, false};
+
+  unsigned int timestamp = 0;
+  for (int i = 0; i < 8; ++i) {
+    const bool is_keyframe = (i == 0);
+    CodecSpecificInfo info;
+    Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+    EXPECT_EQ(is_keyframe ? kKeyFrameFlags : expected_flags[i],
+              LibvpxVp8Encoder::EncodeFlags(tl_config))
+        << i;
+    tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, is_keyframe,
+                    kDefaultQp, &info);
+    EXPECT_TRUE(checker.CheckTemporalConfig(is_keyframe, tl_config));
+    EXPECT_EQ(expected_temporal_idx[i], info.codecSpecific.VP8.temporalIdx);
+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
+    EXPECT_EQ(is_keyframe || expected_layer_sync[i],
+              info.codecSpecific.VP8.layerSync);
+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
+    timestamp += 3000;
+  }
+}
+
+TEST_F(TemporalLayersTest, SearchOrder) {
+  constexpr int kNumLayers = 3;
+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
+  DefaultTemporalLayers tl(kNumLayers);
+  DefaultTemporalLayersChecker checker(kNumLayers);
+  tl.OnRatesUpdated(0,
+                    GetTemporalLayerRates(kDefaultBytesPerFrame,
+                                          kDefaultFramerate, kNumLayers),
+                    kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  // Use a repeating pattern of tl 0, 2, 1, 2.
+  // Tl 0, 1, 2 update last, golden, altref respectively.
+
+  // Start with a key-frame. tl_config flags can be ignored.
+  uint32_t timestamp = 0;
+  Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // TL2 frame. First one only references TL0. Updates altref.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kLast);
+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kNone);
+
+  // TL1 frame. Can only reference TL0. Updated golden.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kLast);
+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kNone);
+
+  // TL2 frame. Can reference all three buffers. Golden was the last to be
+  // updated, the next to last was altref.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kGolden);
+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kAltref);
+}
+
+TEST_F(TemporalLayersTest, SearchOrderWithDrop) {
+  constexpr int kNumLayers = 3;
+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
+  DefaultTemporalLayers tl(kNumLayers);
+  DefaultTemporalLayersChecker checker(kNumLayers);
+  tl.OnRatesUpdated(0,
+                    GetTemporalLayerRates(kDefaultBytesPerFrame,
+                                          kDefaultFramerate, kNumLayers),
+                    kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  // Use a repeating pattern of tl 0, 2, 1, 2.
+  // Tl 0, 1, 2 update last, golden, altref respectively.
+
+  // Start with a key-frame. tl_config flags can be ignored.
+  uint32_t timestamp = 0;
+  Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // TL2 frame. First one only references TL0. Updates altref.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kLast);
+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kNone);
+
+  // Dropped TL1 frame. Can only reference TL0. Should have updated golden.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, 0, false, 0, nullptr);
+
+  // TL2 frame. Can normally reference all three buffers, but golden has not
+  // been populated this cycle. Altref was last to be updated, before that last.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kAltref);
+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kLast);
+}
+
+TEST_F(TemporalLayersTest, DoesNotReferenceDroppedFrames) {
+  constexpr int kNumLayers = 3;
+  // Use a repeating pattern of tl 0, 2, 1, 2.
+  // Tl 0, 1, 2 update last, golden, altref respectively.
+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
+  DefaultTemporalLayers tl(kNumLayers);
+  DefaultTemporalLayersChecker checker(kNumLayers);
+  tl.OnRatesUpdated(0,
+                    GetTemporalLayerRates(kDefaultBytesPerFrame,
+                                          kDefaultFramerate, kNumLayers),
+                    kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  // Start with a keyframe.
+  uint32_t timestamp = 0;
+  Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // Dropped TL2 frame.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, 0, false, 0, nullptr);
+
+  // Dropped TL1 frame.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, 0, false, 0, nullptr);
+
+  // TL2 frame. Can reference all three buffers, valid since golden and altref
+  // both contain the last keyframe.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
+  EXPECT_TRUE(tl_config.golden_buffer_flags & BufferFlags::kReference);
+  EXPECT_TRUE(tl_config.arf_buffer_flags & BufferFlags::kReference);
+
+  // Restart of cycle!
+
+  // TL0 base layer frame, updating and referencing last.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // TL2 frame, updating altref.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // TL1 frame, updating golden.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // TL2 frame. Can still reference all buffer since they have been update this
+  // cycle.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
+  EXPECT_TRUE(tl_config.golden_buffer_flags & BufferFlags::kReference);
+  EXPECT_TRUE(tl_config.arf_buffer_flags & BufferFlags::kReference);
+
+  // Restart of cycle!
+
+  // TL0 base layer frame, updating and referencing last.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // Dropped TL2 frame.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, 0, false, 0, nullptr);
+
+  // Dropped TL1 frame.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, 0, false, 0, nullptr);
+
+  // TL2 frame. This time golden and altref contain data from the previous cycle
+  // and cannot be referenced.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
+  EXPECT_FALSE(tl_config.golden_buffer_flags & BufferFlags::kReference);
+  EXPECT_FALSE(tl_config.arf_buffer_flags & BufferFlags::kReference);
+}
+
+TEST_F(TemporalLayersTest, DoesNotReferenceUnlessGuaranteedToExist) {
+  constexpr int kNumLayers = 3;
+  // Use a repeating pattern of tl 0, 2, 1, 2.
+  // Tl 0, 1 updates last, golden respectively. Altref is always last keyframe.
+  DefaultTemporalLayers tl(kNumLayers);
+  DefaultTemporalLayersChecker checker(kNumLayers);
+  tl.OnRatesUpdated(0,
+                    GetTemporalLayerRates(kDefaultBytesPerFrame,
+                                          kDefaultFramerate, kNumLayers),
+                    kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  // Start with a keyframe.
+  uint32_t timestamp = 0;
+  Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // Do a full cycle of the pattern.
+  for (int i = 0; i < 7; ++i) {
+    tl_config = tl.NextFrameConfig(0, ++timestamp);
+    tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                    IgnoredCodecSpecificInfo());
+  }
+
+  // TL0 base layer frame, starting the cycle over.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // TL2 frame.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // Encoder has a hiccup and builds a queue, so frame encoding is delayed.
+  // TL1 frame, updating golden.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+
+  // TL2 frame, that should be referencing golden, but we can't be certain it's
+  // not going to be dropped, so that is not allowed.
+  tl_config = tl.NextFrameConfig(0, timestamp + 1);
+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
+  EXPECT_FALSE(tl_config.golden_buffer_flags & BufferFlags::kReference);
+  EXPECT_FALSE(tl_config.arf_buffer_flags & BufferFlags::kReference);
+
+  // TL0 base layer frame.
+  tl_config = tl.NextFrameConfig(0, timestamp + 2);
+
+  // The previous four enqueued frames finally get encoded, and the updated
+  // buffers are now OK to reference.
+  // Enqueued TL1 frame ready.
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  // Enqueued TL2 frame.
+  tl.OnEncodeDone(0, ++timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  // Enqueued TL0 frame.
+  tl.OnEncodeDone(0, ++timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // TL2 frame, all buffers are now in a known good state, OK to reference.
+  tl_config = tl.NextFrameConfig(0, ++timestamp + 1);
+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
+  EXPECT_TRUE(tl_config.golden_buffer_flags & BufferFlags::kReference);
+  EXPECT_FALSE(tl_config.arf_buffer_flags & BufferFlags::kReference);
+}
+
+TEST_F(TemporalLayersTest, DoesNotReferenceUnlessGuaranteedToExistLongDelay) {
+  constexpr int kNumLayers = 3;
+  // Use a repeating pattern of tl 0, 2, 1, 2.
+  // Tl 0, 1 updates last, golden, altref respectively.
+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
+  DefaultTemporalLayers tl(kNumLayers);
+  DefaultTemporalLayersChecker checker(kNumLayers);
+  tl.OnRatesUpdated(0,
+                    GetTemporalLayerRates(kDefaultBytesPerFrame,
+                                          kDefaultFramerate, kNumLayers),
+                    kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  // Start with a keyframe.
+  uint32_t timestamp = 0;
+  Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // Do a full cycle of the pattern.
+  for (int i = 0; i < 3; ++i) {
+    tl_config = tl.NextFrameConfig(0, ++timestamp);
+    tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                    IgnoredCodecSpecificInfo());
+  }
+
+  // TL0 base layer frame, starting the cycle over.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // TL2 frame.
+  tl_config = tl.NextFrameConfig(0, ++timestamp);
+  tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // Encoder has a hiccup and builds a queue, so frame encoding is delayed.
+  // Encoded, but delayed frames in TL 1, 2.
+  tl_config = tl.NextFrameConfig(0, timestamp + 1);
+  tl_config = tl.NextFrameConfig(0, timestamp + 2);
+
+  // Restart of the pattern!
+
+  // Encoded, but delayed frames in TL 2, 1.
+  tl_config = tl.NextFrameConfig(0, timestamp + 3);
+  tl_config = tl.NextFrameConfig(0, timestamp + 4);
+
+  // TL1 frame from last cycle is ready.
+  tl.OnEncodeDone(0, timestamp + 1, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+  // TL2 frame from last cycle is ready.
+  tl.OnEncodeDone(0, timestamp + 2, kDefaultBytesPerFrame, false, kDefaultQp,
+                  IgnoredCodecSpecificInfo());
+
+  // TL2 frame, that should be referencing all buffers, but altref and golden
+  // haven not been updated this cycle. (Don't be fooled by the late frames from
+  // the last cycle!)
+  tl_config = tl.NextFrameConfig(0, timestamp + 5);
+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
+  EXPECT_FALSE(tl_config.golden_buffer_flags & BufferFlags::kReference);
+  EXPECT_FALSE(tl_config.arf_buffer_flags & BufferFlags::kReference);
+}
+
+TEST_F(TemporalLayersTest, KeyFrame) {
+  constexpr int kNumLayers = 3;
+  DefaultTemporalLayers tl(kNumLayers);
+  DefaultTemporalLayersChecker checker(kNumLayers);
+  tl.OnRatesUpdated(0,
+                    GetTemporalLayerRates(kDefaultBytesPerFrame,
+                                          kDefaultFramerate, kNumLayers),
+                    kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  int expected_flags[8] = {
+      kTemporalUpdateLastRefAltRef,
+      kTemporalUpdateNoneNoRefGoldenAltRef,
+      kTemporalUpdateGoldenWithoutDependency,
+      kTemporalUpdateNoneNoRefAltRef,
+      kTemporalUpdateLast,
+      kTemporalUpdateNoneNoRefAltRef,
+      kTemporalUpdateGolden,
+      kTemporalUpdateNone,
+  };
+  int expected_temporal_idx[8] = {0, 2, 1, 2, 0, 2, 1, 2};
+  bool expected_layer_sync[8] = {true,  true,  true,  false,
+                                 false, false, false, false};
+
+  uint32_t timestamp = 0;
+  for (int i = 0; i < 7; ++i) {
+    // Temporal pattern starts from 0 after key frame. Let the first `i` - 1
+    // frames be delta frames, and the `i`th one key frame.
+    for (int j = 1; j <= i; ++j) {
+      // Since last frame was always a keyframe and thus index 0 in the pattern,
+      // this loop starts at index 1.
+      Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+      EXPECT_EQ(expected_flags[j], LibvpxVp8Encoder::EncodeFlags(tl_config))
+          << j;
+      tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
+                      IgnoredCodecSpecificInfo());
+      EXPECT_TRUE(checker.CheckTemporalConfig(false, tl_config));
+      EXPECT_EQ(expected_temporal_idx[j], tl_config.packetizer_temporal_idx);
+      EXPECT_EQ(expected_temporal_idx[j], tl_config.encoder_layer_id);
+      EXPECT_EQ(expected_layer_sync[j], tl_config.layer_sync);
+      timestamp += 3000;
+    }
+
+    CodecSpecificInfo info;
+    Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp);
+    tl.OnEncodeDone(0, timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
+                    &info);
+    EXPECT_TRUE(info.codecSpecific.VP8.layerSync)
+        << "Key frame should be marked layer sync.";
+    EXPECT_EQ(0, info.codecSpecific.VP8.temporalIdx)
+        << "Key frame should always be packetized as layer 0";
+    EXPECT_EQ(0, info.generic_frame_info->temporal_id)
+        << "Key frame should always be packetized as layer 0";
+    EXPECT_THAT(info.generic_frame_info->decode_target_indications,
+                Each(DecodeTargetIndication::kSwitch))
+        << "Key frame is universal switch";
+    EXPECT_TRUE(checker.CheckTemporalConfig(true, tl_config));
+  }
+}
+
+TEST_F(TemporalLayersTest, SetsTlCountOnFirstConfigUpdate) {
+  // Create an instance and fetch config update without setting any rate.
+  constexpr int kNumLayers = 2;
+  DefaultTemporalLayers tl(kNumLayers);
+  Vp8EncoderConfig config = tl.UpdateConfiguration(0);
+
+  // Config should indicate correct number of temporal layers, but zero bitrate.
+  ASSERT_TRUE(config.temporal_layer_config.has_value());
+  EXPECT_EQ(config.temporal_layer_config->ts_number_layers,
+            uint32_t{kNumLayers});
+  std::array<uint32_t, Vp8EncoderConfig::TemporalLayerConfig::kMaxLayers>
+      kZeroRate = {};
+  EXPECT_EQ(config.temporal_layer_config->ts_target_bitrate, kZeroRate);
+
+  // On second call, no new update.
+  config = tl.UpdateConfiguration(0);
+  EXPECT_FALSE(config.temporal_layer_config.has_value());
+}
+
+class TemporalLayersReferenceTest : public TemporalLayersTest,
+                                    public ::testing::WithParamInterface<int> {
+ public:
+  TemporalLayersReferenceTest()
+      : timestamp_(1),
+        last_sync_timestamp_(timestamp_),
+        tl0_reference_(nullptr) {}
+  virtual ~TemporalLayersReferenceTest() {}
+
+ protected:
+  static const int kMaxPatternLength = 32;
+
+  struct BufferState {
+    BufferState() : BufferState(-1, 0, false) {}
+    BufferState(int temporal_idx, uint32_t timestamp, bool sync)
+        : temporal_idx(temporal_idx), timestamp(timestamp), sync(sync) {}
+    int temporal_idx;
+    uint32_t timestamp;
+    bool sync;
+  };
+
+  bool UpdateSyncRefState(const BufferFlags& flags, BufferState* buffer_state) {
+    if (flags & BufferFlags::kReference) {
+      if (buffer_state->temporal_idx == -1)
+        return true;  // References key-frame.
+      if (buffer_state->temporal_idx == 0) {
+        // No more than one reference to TL0 frame.
+        EXPECT_EQ(nullptr, tl0_reference_);
+        tl0_reference_ = buffer_state;
+        return true;
+      }
+      return false;  // References higher layer.
+    }
+    return true;  // No reference, does not affect sync frame status.
+  }
+
+  void ValidateReference(const BufferFlags& flags,
+                         const BufferState& buffer_state,
+                         int temporal_layer) {
+    if (flags & BufferFlags::kReference) {
+      if (temporal_layer > 0 && buffer_state.timestamp > 0) {
+        // Check that high layer reference does not go past last sync frame.
+        EXPECT_GE(buffer_state.timestamp, last_sync_timestamp_);
+      }
+      // No reference to buffer in higher layer.
+      EXPECT_LE(buffer_state.temporal_idx, temporal_layer);
+    }
+  }
+
+  uint32_t timestamp_ = 1;
+  uint32_t last_sync_timestamp_ = timestamp_;
+  BufferState* tl0_reference_;
+
+  BufferState last_state;
+  BufferState golden_state;
+  BufferState altref_state;
+};
+
+INSTANTIATE_TEST_SUITE_P(DefaultTemporalLayersTest,
+                         TemporalLayersReferenceTest,
+                         ::testing::Range(1, kMaxTemporalStreams + 1));
+
+TEST_P(TemporalLayersReferenceTest, ValidFrameConfigs) {
+  const int num_layers = GetParam();
+  DefaultTemporalLayers tl(num_layers);
+  tl.OnRatesUpdated(
+      0, GetTemporalLayerRates(kDefaultBytesPerFrame, kDefaultFramerate, 1),
+      kDefaultFramerate);
+  tl.UpdateConfiguration(0);
+
+  // Run through the pattern and store the frame dependencies, plus keep track
+  // of the buffer state; which buffers references which temporal layers (if
+  // (any). If a given buffer is never updated, it is legal to reference it
+  // even for sync frames. In order to be general, don't assume TL0 always
+  // updates `last`.
+  std::vector<Vp8FrameConfig> tl_configs(kMaxPatternLength);
+  for (int i = 0; i < kMaxPatternLength; ++i) {
+    Vp8FrameConfig tl_config = tl.NextFrameConfig(0, timestamp_);
+    tl.OnEncodeDone(0, timestamp_, kDefaultBytesPerFrame, i == 0, kDefaultQp,
+                    IgnoredCodecSpecificInfo());
+    ++timestamp_;
+    EXPECT_FALSE(tl_config.drop_frame);
+    tl_configs.push_back(tl_config);
+    int temporal_idx = tl_config.encoder_layer_id;
+    // For the default layers, always keep encoder and rtp layers in sync.
+    EXPECT_EQ(tl_config.packetizer_temporal_idx, temporal_idx);
+
+    // Determine if this frame is in a higher layer but references only TL0
+    // or untouched buffers, if so verify it is marked as a layer sync.
+    bool is_sync_frame = true;
+    tl0_reference_ = nullptr;
+    if (temporal_idx <= 0) {
+      is_sync_frame = false;  // TL0 by definition not a sync frame.
+    } else if (!UpdateSyncRefState(tl_config.last_buffer_flags, &last_state)) {
+      is_sync_frame = false;
+    } else if (!UpdateSyncRefState(tl_config.golden_buffer_flags,
+                                   &golden_state)) {
+      is_sync_frame = false;
+    } else if (!UpdateSyncRefState(tl_config.arf_buffer_flags, &altref_state)) {
+      is_sync_frame = false;
+    }
+    if (is_sync_frame) {
+      // Cache timestamp for last found sync frame, so that we can verify no
+      // references back past this frame.
+      ASSERT_TRUE(tl0_reference_);
+      last_sync_timestamp_ = tl0_reference_->timestamp;
+    }
+    EXPECT_EQ(tl_config.layer_sync, is_sync_frame);
+
+    // Validate no reference from lower to high temporal layer, or backwards
+    // past last reference frame.
+    ValidateReference(tl_config.last_buffer_flags, last_state, temporal_idx);
+    ValidateReference(tl_config.golden_buffer_flags, golden_state,
+                      temporal_idx);
+    ValidateReference(tl_config.arf_buffer_flags, altref_state, temporal_idx);
+
+    // Update the current layer state.
+    BufferState state = {temporal_idx, timestamp_, is_sync_frame};
+    if (tl_config.last_buffer_flags & BufferFlags::kUpdate)
+      last_state = state;
+    if (tl_config.golden_buffer_flags & BufferFlags::kUpdate)
+      golden_state = state;
+    if (tl_config.arf_buffer_flags & BufferFlags::kUpdate)
+      altref_state = state;
+  }
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/include/temporal_layers_checker.h b/third_party/libwebrtc/modules/video_coding/codecs/vp8/include/temporal_layers_checker.h
new file mode 100644
index 0000000000..3d1671a676
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/include/temporal_layers_checker.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_TEMPORAL_LAYERS_CHECKER_H_
+#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_TEMPORAL_LAYERS_CHECKER_H_
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/video_codecs/vp8_frame_config.h"
+#include "api/video_codecs/vp8_temporal_layers.h"
+
+namespace webrtc {
+
+// Interface for a class that verifies correctness of temporal layer
+// configurations (dependencies, sync flag, etc).
+// Intended to be used in tests as well as with real apps in debug mode.
+class TemporalLayersChecker {
+ public:
+  explicit TemporalLayersChecker(int num_temporal_layers);
+  virtual ~TemporalLayersChecker() {}
+
+  virtual bool CheckTemporalConfig(bool frame_is_keyframe,
+                                   const Vp8FrameConfig& frame_config);
+
+  static std::unique_ptr<TemporalLayersChecker> CreateTemporalLayersChecker(
+      Vp8TemporalLayersType type,
+      int num_temporal_layers);
+
+ private:
+  struct BufferState {
+    BufferState() : is_keyframe(true), temporal_layer(0), sequence_number(0) {}
+    bool is_keyframe;
+    uint8_t temporal_layer;
+    uint32_t sequence_number;
+  };
+  bool CheckAndUpdateBufferState(BufferState* state,
+                                 bool* need_sync,
+                                 bool frame_is_keyframe,
+                                 uint8_t temporal_layer,
+                                 Vp8FrameConfig::BufferFlags flags,
+                                 uint32_t sequence_number,
+                                 uint32_t* lowest_sequence_referenced);
+  BufferState last_;
+  BufferState arf_;
+  BufferState golden_;
+  int num_temporal_layers_;
+  uint32_t sequence_number_;
+  uint32_t last_sync_sequence_number_;
+  uint32_t last_tl0_sequence_number_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_TEMPORAL_LAYERS_CHECKER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/include/vp8.h b/third_party/libwebrtc/modules/video_coding/codecs/vp8/include/vp8.h
new file mode 100644
index 0000000000..2fc647874f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/include/vp8.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
+#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/vp8_frame_buffer_controller.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+
+// TODO(brandtr): Move these interfaces to the api/ folder.
+class VP8Encoder {
+ public:
+  struct Settings {
+    // Allows for overriding the Vp8FrameBufferController used by the encoder.
+    // If unset, a default Vp8FrameBufferController will be instantiated
+    // internally.
+    std::unique_ptr<Vp8FrameBufferControllerFactory>
+        frame_buffer_controller_factory = nullptr;
+
+    // Allows for overriding the resolution/bitrate limits exposed through
+    // VideoEncoder::GetEncoderInfo(). No override is done if empty.
+    std::vector<VideoEncoder::ResolutionBitrateLimits>
+        resolution_bitrate_limits = {};
+  };
+
+  static std::unique_ptr<VideoEncoder> Create();
+  static std::unique_ptr<VideoEncoder> Create(Settings settings);
+};
+
+class VP8Decoder {
+ public:
+  static std::unique_ptr<VideoDecoder> Create();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h b/third_party/libwebrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
new file mode 100644
index 0000000000..1fab5f45a6
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains codec dependent definitions that are needed in
+// order to compile the WebRTC codebase, even if this codec is not used.
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
+#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
+
+#include "modules/video_coding/codecs/interface/common_constants.h"
+
+namespace webrtc {
+
+struct RTPVideoHeaderVP8 {
+  void InitRTPVideoHeaderVP8() {
+    nonReference = false;
+    pictureId = kNoPictureId;
+    tl0PicIdx = kNoTl0PicIdx;
+    temporalIdx = kNoTemporalIdx;
+    layerSync = false;
+    keyIdx = kNoKeyIdx;
+    partitionId = 0;
+    beginningOfPartition = false;
+  }
+
+  bool nonReference;          // Frame is discardable.
+  int16_t pictureId;          // Picture ID index, 15 bits;
+                              // kNoPictureId if PictureID does not exist.
+  int16_t tl0PicIdx;          // TL0PIC_IDX, 8 bits;
+                              // kNoTl0PicIdx means no value provided.
+  uint8_t temporalIdx;        // Temporal layer index, or kNoTemporalIdx.
+  bool layerSync;             // This frame is a layer sync frame.
+                              // Disabled if temporalIdx == kNoTemporalIdx.
+  int keyIdx;                 // 5 bits; kNoKeyIdx means not used.
+  int partitionId;            // VP8 partition ID
+  bool beginningOfPartition;  // True if this packet is the first
+                              // in a VP8 partition. Otherwise false
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
new file mode 100644
index 0000000000..3fe86f2f85
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
@@ -0,0 +1,384 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include <algorithm>
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/exp_filter.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+#include "third_party/libyuv/include/libyuv/convert.h"
+#include "vpx/vp8.h"
+#include "vpx/vp8dx.h"
+#include "vpx/vpx_decoder.h"
+
+namespace webrtc {
+namespace {
+constexpr int kVp8ErrorPropagationTh = 30;
+// vpx_decoder.h documentation indicates decode deadline is time in us, with
+// "Set to zero for unlimited.", but actual implementation requires this to be
+// a mode with 0 meaning allow delay and 1 not allowing it.
+constexpr long kDecodeDeadlineRealtime = 1;  // NOLINT
+
+const char kVp8PostProcArmFieldTrial[] = "WebRTC-VP8-Postproc-Config-Arm";
+const char kVp8PostProcFieldTrial[] = "WebRTC-VP8-Postproc-Config";
+
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
+    defined(WEBRTC_ANDROID)  || defined(WEBRTC_ARCH_MIPS)
+constexpr bool kIsArm = true;
+#else
+constexpr bool kIsArm = false;
+#endif
+
+absl::optional<LibvpxVp8Decoder::DeblockParams> DefaultDeblockParams() {
+  return LibvpxVp8Decoder::DeblockParams(/*max_level=*/8,
+                                         /*degrade_qp=*/60,
+                                         /*min_qp=*/30);
+}
+
+absl::optional<LibvpxVp8Decoder::DeblockParams>
+GetPostProcParamsFromFieldTrialGroup() {
+  std::string group = webrtc::field_trial::FindFullName(
+      kIsArm ? kVp8PostProcArmFieldTrial : kVp8PostProcFieldTrial);
+  if (group.empty()) {
+    return DefaultDeblockParams();
+  }
+
+  LibvpxVp8Decoder::DeblockParams params;
+  if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &params.max_level,
+             &params.min_qp, &params.degrade_qp) != 3) {
+    return DefaultDeblockParams();
+  }
+
+  if (params.max_level < 0 || params.max_level > 16) {
+    return DefaultDeblockParams();
+  }
+
+  if (params.min_qp < 0 || params.degrade_qp <= params.min_qp) {
+    return DefaultDeblockParams();
+  }
+
+  return params;
+}
+
+}  // namespace
+
+std::unique_ptr<VideoDecoder> VP8Decoder::Create() {
+  return std::make_unique<LibvpxVp8Decoder>();
+}
+
+class LibvpxVp8Decoder::QpSmoother {
+ public:
+  QpSmoother() : last_sample_ms_(rtc::TimeMillis()), smoother_(kAlpha) {}
+
+  int GetAvg() const {
+    float value = smoother_.filtered();
+    return (value == rtc::ExpFilter::kValueUndefined) ? 0
+                                                      : static_cast<int>(value);
+  }
+
+  void Add(float sample) {
+    int64_t now_ms = rtc::TimeMillis();
+    smoother_.Apply(static_cast<float>(now_ms - last_sample_ms_), sample);
+    last_sample_ms_ = now_ms;
+  }
+
+  void Reset() { smoother_.Reset(kAlpha); }
+
+ private:
+  const float kAlpha = 0.95f;
+  int64_t last_sample_ms_;
+  rtc::ExpFilter smoother_;
+};
+
+LibvpxVp8Decoder::LibvpxVp8Decoder()
+    : use_postproc_(
+          kIsArm ? webrtc::field_trial::IsEnabled(kVp8PostProcArmFieldTrial)
+                 : true),
+      buffer_pool_(false, 300 /* max_number_of_buffers*/),
+      decode_complete_callback_(NULL),
+      inited_(false),
+      decoder_(NULL),
+      propagation_cnt_(-1),
+      last_frame_width_(0),
+      last_frame_height_(0),
+      key_frame_required_(true),
+      deblock_params_(use_postproc_ ? GetPostProcParamsFromFieldTrialGroup()
+                                    : absl::nullopt),
+      qp_smoother_(use_postproc_ ? new QpSmoother() : nullptr) {}
+
+LibvpxVp8Decoder::~LibvpxVp8Decoder() {
+  inited_ = true;  // in order to do the actual release
+  Release();
+}
+
+bool LibvpxVp8Decoder::Configure(const Settings& settings) {
+  if (Release() < 0) {
+    return false;
+  }
+  if (decoder_ == NULL) {
+    decoder_ = new vpx_codec_ctx_t;
+    memset(decoder_, 0, sizeof(*decoder_));
+  }
+  vpx_codec_dec_cfg_t cfg;
+  // Setting number of threads to a constant value (1)
+  cfg.threads = 1;
+  cfg.h = cfg.w = 0;  // set after decode
+
+  vpx_codec_flags_t flags = use_postproc_ ? VPX_CODEC_USE_POSTPROC : 0;
+
+  if (vpx_codec_dec_init(decoder_, vpx_codec_vp8_dx(), &cfg, flags)) {
+    delete decoder_;
+    decoder_ = nullptr;
+    return false;
+  }
+
+  propagation_cnt_ = -1;
+  inited_ = true;
+
+  // Always start with a complete key frame.
+  key_frame_required_ = true;
+  if (absl::optional<int> buffer_pool_size = settings.buffer_pool_size()) {
+    if (!buffer_pool_.Resize(*buffer_pool_size)) {
+      return false;
+    }
+  }
+  return true;
+}
+
+int LibvpxVp8Decoder::Decode(const EncodedImage& input_image,
+                             bool missing_frames,
+                             int64_t /*render_time_ms*/) {
+  if (!inited_) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (decode_complete_callback_ == NULL) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (input_image.data() == NULL && input_image.size() > 0) {
+    // Reset to avoid requesting key frames too often.
+    if (propagation_cnt_ > 0)
+      propagation_cnt_ = 0;
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  // Post process configurations.
+  if (use_postproc_) {
+    vp8_postproc_cfg_t ppcfg;
+    // MFQE enabled to reduce key frame popping.
+    ppcfg.post_proc_flag = VP8_MFQE;
+
+    if (kIsArm) {
+      RTC_DCHECK(deblock_params_.has_value());
+    }
+    if (deblock_params_.has_value()) {
+      // For low resolutions, use stronger deblocking filter.
+      int last_width_x_height = last_frame_width_ * last_frame_height_;
+      if (last_width_x_height > 0 && last_width_x_height <= 320 * 240) {
+        // Enable the deblock and demacroblocker based on qp thresholds.
+        RTC_DCHECK(qp_smoother_);
+        int qp = qp_smoother_->GetAvg();
+        if (qp > deblock_params_->min_qp) {
+          int level = deblock_params_->max_level;
+          if (qp < deblock_params_->degrade_qp) {
+            // Use lower level.
+            level = deblock_params_->max_level *
+                    (qp - deblock_params_->min_qp) /
+                    (deblock_params_->degrade_qp - deblock_params_->min_qp);
+          }
+          // Deblocking level only affects VP8_DEMACROBLOCK.
+          ppcfg.deblocking_level = std::max(level, 1);
+          ppcfg.post_proc_flag |= VP8_DEBLOCK | VP8_DEMACROBLOCK;
+        }
+      }
+    } else {
+      // Non-arm with no explicit deblock params set.
+      ppcfg.post_proc_flag |= VP8_DEBLOCK;
+      // For VGA resolutions and lower, enable the demacroblocker postproc.
+      if (last_frame_width_ * last_frame_height_ <= 640 * 360) {
+        ppcfg.post_proc_flag |= VP8_DEMACROBLOCK;
+      }
+      // Strength of deblocking filter. Valid range:[0,16]
+      ppcfg.deblocking_level = 3;
+    }
+
+    vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
+  }
+
+  // Always start with a complete key frame.
+  if (key_frame_required_) {
+    if (input_image._frameType != VideoFrameType::kVideoFrameKey)
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    key_frame_required_ = false;
+  }
+  // Restrict error propagation using key frame requests.
+  // Reset on a key frame refresh.
+  if (input_image._frameType == VideoFrameType::kVideoFrameKey) {
+    propagation_cnt_ = -1;
+    // Start count on first loss.
+  } else if (missing_frames && propagation_cnt_ == -1) {
+    propagation_cnt_ = 0;
+  }
+  if (propagation_cnt_ >= 0) {
+    propagation_cnt_++;
+  }
+
+  vpx_codec_iter_t iter = NULL;
+  vpx_image_t* img;
+  int ret;
+
+  // Check for missing frames.
+  if (missing_frames) {
+    // Call decoder with zero data length to signal missing frames.
+    if (vpx_codec_decode(decoder_, NULL, 0, 0, kDecodeDeadlineRealtime)) {
+      // Reset to avoid requesting key frames too often.
+      if (propagation_cnt_ > 0)
+        propagation_cnt_ = 0;
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    img = vpx_codec_get_frame(decoder_, &iter);
+    iter = NULL;
+  }
+
+  const uint8_t* buffer = input_image.data();
+  if (input_image.size() == 0) {
+    buffer = NULL;  // Triggers full frame concealment.
+  }
+  if (vpx_codec_decode(decoder_, buffer, input_image.size(), 0,
+                       kDecodeDeadlineRealtime)) {
+    // Reset to avoid requesting key frames too often.
+    if (propagation_cnt_ > 0) {
+      propagation_cnt_ = 0;
+    }
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  img = vpx_codec_get_frame(decoder_, &iter);
+  int qp;
+  vpx_codec_err_t vpx_ret =
+      vpx_codec_control(decoder_, VPXD_GET_LAST_QUANTIZER, &qp);
+  RTC_DCHECK_EQ(vpx_ret, VPX_CODEC_OK);
+  ret = ReturnFrame(img, input_image.Timestamp(), qp, input_image.ColorSpace());
+  if (ret != 0) {
+    // Reset to avoid requesting key frames too often.
+    if (ret < 0 && propagation_cnt_ > 0)
+      propagation_cnt_ = 0;
+    return ret;
+  }
+  // Check Vs. threshold
+  if (propagation_cnt_ > kVp8ErrorPropagationTh) {
+    // Reset to avoid requesting key frames too often.
+    propagation_cnt_ = 0;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int LibvpxVp8Decoder::ReturnFrame(
+    const vpx_image_t* img,
+    uint32_t timestamp,
+    int qp,
+    const webrtc::ColorSpace* explicit_color_space) {
+  if (img == NULL) {
+    // Decoder OK and NULL image => No show frame
+    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
+  }
+  if (qp_smoother_) {
+    if (last_frame_width_ != static_cast<int>(img->d_w) ||
+        last_frame_height_ != static_cast<int>(img->d_h)) {
+      qp_smoother_->Reset();
+    }
+    qp_smoother_->Add(qp);
+  }
+  last_frame_width_ = img->d_w;
+  last_frame_height_ = img->d_h;
+  // Allocate memory for decoded image.
+  rtc::scoped_refptr<VideoFrameBuffer> buffer;
+
+  rtc::scoped_refptr<I420Buffer> i420_buffer =
+      buffer_pool_.CreateI420Buffer(img->d_w, img->d_h);
+  buffer = i420_buffer;
+  if (i420_buffer.get()) {
+    libyuv::I420Copy(img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
+                     img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
+                     img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
+                     i420_buffer->MutableDataY(), i420_buffer->StrideY(),
+                     i420_buffer->MutableDataU(), i420_buffer->StrideU(),
+                     i420_buffer->MutableDataV(), i420_buffer->StrideV(),
+                     img->d_w, img->d_h);
+  }
+
+  if (!buffer.get()) {
+    // Pool has too many pending frames.
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Video.LibvpxVp8Decoder.TooManyPendingFrames",
+                          1);
+    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
+  }
+
+  VideoFrame decoded_image = VideoFrame::Builder()
+                                 .set_video_frame_buffer(buffer)
+                                 .set_timestamp_rtp(timestamp)
+                                 .set_color_space(explicit_color_space)
+                                 .build();
+  decode_complete_callback_->Decoded(decoded_image, absl::nullopt, qp);
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int LibvpxVp8Decoder::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  decode_complete_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int LibvpxVp8Decoder::Release() {
+  int ret_val = WEBRTC_VIDEO_CODEC_OK;
+
+  if (decoder_ != NULL) {
+    if (inited_) {
+      if (vpx_codec_destroy(decoder_)) {
+        ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
+      }
+    }
+    delete decoder_;
+    decoder_ = NULL;
+  }
+  buffer_pool_.Release();
+  inited_ = false;
+  return ret_val;
+}
+
+VideoDecoder::DecoderInfo LibvpxVp8Decoder::GetDecoderInfo() const {
+  DecoderInfo info;
+  info.implementation_name = "libvpx";
+  info.is_hardware_accelerated = false;
+  return info;
+}
+
+const char* LibvpxVp8Decoder::ImplementationName() const {
+  return "libvpx";
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
new file mode 100644
index 0000000000..f9acd70bad
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_DECODER_H_
+#define MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_DECODER_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_decoder.h"
+#include "common_video/include/video_frame_buffer_pool.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "vpx/vp8dx.h"
+#include "vpx/vpx_decoder.h"
+
+namespace webrtc {
+
+class LibvpxVp8Decoder : public VideoDecoder {
+ public:
+  LibvpxVp8Decoder();
+  ~LibvpxVp8Decoder() override;
+
+  bool Configure(const Settings& settings) override;
+  int Decode(const EncodedImage& input_image,
+             bool missing_frames,
+             int64_t /*render_time_ms*/) override;
+
+  int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
+  int Release() override;
+
+  DecoderInfo GetDecoderInfo() const override;
+  const char* ImplementationName() const override;
+
+  struct DeblockParams {
+    DeblockParams() : max_level(6), degrade_qp(1), min_qp(0) {}
+    DeblockParams(int max_level, int degrade_qp, int min_qp)
+        : max_level(max_level), degrade_qp(degrade_qp), min_qp(min_qp) {}
+    int max_level;   // Deblocking strength: [0, 16].
+    int degrade_qp;  // If QP value is below, start lowering `max_level`.
+    int min_qp;      // If QP value is below, turn off deblocking.
+  };
+
+ private:
+  class QpSmoother;
+  int ReturnFrame(const vpx_image_t* img,
+                  uint32_t timeStamp,
+                  int qp,
+                  const webrtc::ColorSpace* explicit_color_space);
+  const bool use_postproc_;
+
+  VideoFrameBufferPool buffer_pool_;
+  DecodedImageCallback* decode_complete_callback_;
+  bool inited_;
+  vpx_codec_ctx_t* decoder_;
+  int propagation_cnt_;
+  int last_frame_width_;
+  int last_frame_height_;
+  bool key_frame_required_;
+  const absl::optional<DeblockParams> deblock_params_;
+  const std::unique_ptr<QpSmoother> qp_smoother_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_DECODER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
new file mode 100644
index 0000000000..cc84605ce7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
@@ -0,0 +1,1438 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <iterator>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "api/scoped_refptr.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_timing.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "api/video_codecs/vp8_temporal_layers.h"
+#include "api/video_codecs/vp8_temporal_layers_factory.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/codecs/vp8/vp8_scalability.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
+#include "modules/video_coding/utility/simulcast_utility.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/field_trial_units.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/field_trial.h"
+#include "third_party/libyuv/include/libyuv/scale.h"
+#include "vpx/vp8cx.h"
+
+namespace webrtc {
+namespace {
+#if defined(WEBRTC_IOS)
+constexpr char kVP8IosMaxNumberOfThreadFieldTrial[] =
+    "WebRTC-VP8IosMaxNumberOfThread";
+constexpr char kVP8IosMaxNumberOfThreadFieldTrialParameter[] = "max_thread";
+#endif
+
+constexpr char kVp8ForcePartitionResilience[] =
+    "WebRTC-VP8-ForcePartitionResilience";
+
+// QP is obtained from VP8-bitstream for HW, so the QP corresponds to the
+// bitstream range of [0, 127] and not the user-level range of [0,63].
+constexpr int kLowVp8QpThreshold = 29;
+constexpr int kHighVp8QpThreshold = 95;
+
+constexpr int kTokenPartitions = VP8_ONE_TOKENPARTITION;
+constexpr uint32_t kVp832ByteAlign = 32u;
+
+constexpr int kRtpTicksPerSecond = 90000;
+constexpr int kRtpTicksPerMs = kRtpTicksPerSecond / 1000;
+
+// VP8 denoiser states.
+enum denoiserState : uint32_t {
+  kDenoiserOff,
+  kDenoiserOnYOnly,
+  kDenoiserOnYUV,
+  kDenoiserOnYUVAggressive,
+  // Adaptive mode defaults to kDenoiserOnYUV on key frame, but may switch
+  // to kDenoiserOnYUVAggressive based on a computed noise metric.
+  kDenoiserOnAdaptive
+};
+
+// Greatest common divisior
+int GCD(int a, int b) {
+  int c = a % b;
+  while (c != 0) {
+    a = b;
+    b = c;
+    c = a % b;
+  }
+  return b;
+}
+
+static_assert(Vp8EncoderConfig::TemporalLayerConfig::kMaxPeriodicity ==
+                  VPX_TS_MAX_PERIODICITY,
+              "Vp8EncoderConfig::kMaxPeriodicity must be kept in sync with the "
+              "constant in libvpx.");
+static_assert(Vp8EncoderConfig::TemporalLayerConfig::kMaxLayers ==
+                  VPX_TS_MAX_LAYERS,
+              "Vp8EncoderConfig::kMaxLayers must be kept in sync with the "
+              "constant in libvpx.");
+
+// Allow a newer value to override a current value only if the new value
+// is set.
+template <typename T>
+bool MaybeSetNewValue(const absl::optional<T>& new_value,
+                      absl::optional<T>* base_value) {
+  if (new_value.has_value() && new_value != *base_value) {
+    *base_value = new_value;
+    return true;
+  } else {
+    return false;
+  }
+}
+
+// Adds configuration from `new_config` to `base_config`. Both configs consist
+// of optionals, and only optionals which are set in `new_config` can have
+// an effect. (That is, set values in `base_config` cannot be unset.)
+// Returns `true` iff any changes were made to `base_config`.
+bool MaybeExtendVp8EncoderConfig(const Vp8EncoderConfig& new_config,
+                                 Vp8EncoderConfig* base_config) {
+  bool changes_made = false;
+  changes_made |= MaybeSetNewValue(new_config.temporal_layer_config,
+                                   &base_config->temporal_layer_config);
+  changes_made |= MaybeSetNewValue(new_config.rc_target_bitrate,
+                                   &base_config->rc_target_bitrate);
+  changes_made |= MaybeSetNewValue(new_config.rc_max_quantizer,
+                                   &base_config->rc_max_quantizer);
+  changes_made |= MaybeSetNewValue(new_config.g_error_resilient,
+                                   &base_config->g_error_resilient);
+  return changes_made;
+}
+
+void ApplyVp8EncoderConfigToVpxConfig(const Vp8EncoderConfig& encoder_config,
+                                      vpx_codec_enc_cfg_t* vpx_config) {
+  if (encoder_config.temporal_layer_config.has_value()) {
+    const Vp8EncoderConfig::TemporalLayerConfig& ts_config =
+        encoder_config.temporal_layer_config.value();
+    vpx_config->ts_number_layers = ts_config.ts_number_layers;
+    std::copy(ts_config.ts_target_bitrate.begin(),
+              ts_config.ts_target_bitrate.end(),
+              std::begin(vpx_config->ts_target_bitrate));
+    std::copy(ts_config.ts_rate_decimator.begin(),
+              ts_config.ts_rate_decimator.end(),
+              std::begin(vpx_config->ts_rate_decimator));
+    vpx_config->ts_periodicity = ts_config.ts_periodicity;
+    std::copy(ts_config.ts_layer_id.begin(), ts_config.ts_layer_id.end(),
+              std::begin(vpx_config->ts_layer_id));
+  } else {
+    vpx_config->ts_number_layers = 1;
+    vpx_config->ts_rate_decimator[0] = 1;
+    vpx_config->ts_periodicity = 1;
+    vpx_config->ts_layer_id[0] = 0;
+  }
+
+  if (encoder_config.rc_target_bitrate.has_value()) {
+    vpx_config->rc_target_bitrate = encoder_config.rc_target_bitrate.value();
+  }
+
+  if (encoder_config.rc_max_quantizer.has_value()) {
+    vpx_config->rc_max_quantizer = encoder_config.rc_max_quantizer.value();
+  }
+
+  if (encoder_config.g_error_resilient.has_value()) {
+    vpx_config->g_error_resilient = encoder_config.g_error_resilient.value();
+  }
+}
+
+bool IsCompatibleVideoFrameBufferType(VideoFrameBuffer::Type left,
+                                      VideoFrameBuffer::Type right) {
+  if (left == VideoFrameBuffer::Type::kI420 ||
+      left == VideoFrameBuffer::Type::kI420A) {
+    // LibvpxVp8Encoder does not care about the alpha channel, I420A and I420
+    // are considered compatible.
+    return right == VideoFrameBuffer::Type::kI420 ||
+           right == VideoFrameBuffer::Type::kI420A;
+  }
+  return left == right;
+}
+
+void SetRawImagePlanes(vpx_image_t* raw_image, VideoFrameBuffer* buffer) {
+  switch (buffer->type()) {
+    case VideoFrameBuffer::Type::kI420:
+    case VideoFrameBuffer::Type::kI420A: {
+      const I420BufferInterface* i420_buffer = buffer->GetI420();
+      RTC_DCHECK(i420_buffer);
+      raw_image->planes[VPX_PLANE_Y] =
+          const_cast<uint8_t*>(i420_buffer->DataY());
+      raw_image->planes[VPX_PLANE_U] =
+          const_cast<uint8_t*>(i420_buffer->DataU());
+      raw_image->planes[VPX_PLANE_V] =
+          const_cast<uint8_t*>(i420_buffer->DataV());
+      raw_image->stride[VPX_PLANE_Y] = i420_buffer->StrideY();
+      raw_image->stride[VPX_PLANE_U] = i420_buffer->StrideU();
+      raw_image->stride[VPX_PLANE_V] = i420_buffer->StrideV();
+      break;
+    }
+    case VideoFrameBuffer::Type::kNV12: {
+      const NV12BufferInterface* nv12_buffer = buffer->GetNV12();
+      RTC_DCHECK(nv12_buffer);
+      raw_image->planes[VPX_PLANE_Y] =
+          const_cast<uint8_t*>(nv12_buffer->DataY());
+      raw_image->planes[VPX_PLANE_U] =
+          const_cast<uint8_t*>(nv12_buffer->DataUV());
+      raw_image->planes[VPX_PLANE_V] = raw_image->planes[VPX_PLANE_U] + 1;
+      raw_image->stride[VPX_PLANE_Y] = nv12_buffer->StrideY();
+      raw_image->stride[VPX_PLANE_U] = nv12_buffer->StrideUV();
+      raw_image->stride[VPX_PLANE_V] = nv12_buffer->StrideUV();
+      break;
+    }
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+}
+
+}  // namespace
+
+std::unique_ptr<VideoEncoder> VP8Encoder::Create() {
+  return std::make_unique<LibvpxVp8Encoder>(LibvpxInterface::Create(),
+                                            VP8Encoder::Settings());
+}
+
+std::unique_ptr<VideoEncoder> VP8Encoder::Create(
+    VP8Encoder::Settings settings) {
+  return std::make_unique<LibvpxVp8Encoder>(LibvpxInterface::Create(),
+                                            std::move(settings));
+}
+
+vpx_enc_frame_flags_t LibvpxVp8Encoder::EncodeFlags(
+    const Vp8FrameConfig& references) {
+  RTC_DCHECK(!references.drop_frame);
+
+  vpx_enc_frame_flags_t flags = 0;
+
+  if ((references.last_buffer_flags &
+       Vp8FrameConfig::BufferFlags::kReference) == 0)
+    flags |= VP8_EFLAG_NO_REF_LAST;
+  if ((references.last_buffer_flags & Vp8FrameConfig::BufferFlags::kUpdate) ==
+      0)
+    flags |= VP8_EFLAG_NO_UPD_LAST;
+  if ((references.golden_buffer_flags &
+       Vp8FrameConfig::BufferFlags::kReference) == 0)
+    flags |= VP8_EFLAG_NO_REF_GF;
+  if ((references.golden_buffer_flags & Vp8FrameConfig::BufferFlags::kUpdate) ==
+      0)
+    flags |= VP8_EFLAG_NO_UPD_GF;
+  if ((references.arf_buffer_flags & Vp8FrameConfig::BufferFlags::kReference) ==
+      0)
+    flags |= VP8_EFLAG_NO_REF_ARF;
+  if ((references.arf_buffer_flags & Vp8FrameConfig::BufferFlags::kUpdate) == 0)
+    flags |= VP8_EFLAG_NO_UPD_ARF;
+  if (references.freeze_entropy)
+    flags |= VP8_EFLAG_NO_UPD_ENTROPY;
+
+  return flags;
+}
+
+LibvpxVp8Encoder::LibvpxVp8Encoder(std::unique_ptr<LibvpxInterface> interface,
+                                   VP8Encoder::Settings settings)
+    : libvpx_(std::move(interface)),
+      rate_control_settings_(RateControlSettings::ParseFromFieldTrials()),
+      frame_buffer_controller_factory_(
+          std::move(settings.frame_buffer_controller_factory)),
+      resolution_bitrate_limits_(std::move(settings.resolution_bitrate_limits)),
+      key_frame_request_(kMaxSimulcastStreams, false),
+      variable_framerate_experiment_(ParseVariableFramerateConfig(
+          "WebRTC-VP8VariableFramerateScreenshare")),
+      framerate_controller_(variable_framerate_experiment_.framerate_limit) {
+  // TODO(eladalon/ilnik): These reservations might be wasting memory.
+  // InitEncode() is resizing to the actual size, which might be smaller.
+  raw_images_.reserve(kMaxSimulcastStreams);
+  encoded_images_.reserve(kMaxSimulcastStreams);
+  send_stream_.reserve(kMaxSimulcastStreams);
+  cpu_speed_.assign(kMaxSimulcastStreams, cpu_speed_default_);
+  encoders_.reserve(kMaxSimulcastStreams);
+  vpx_configs_.reserve(kMaxSimulcastStreams);
+  config_overrides_.reserve(kMaxSimulcastStreams);
+  downsampling_factors_.reserve(kMaxSimulcastStreams);
+}
+
+LibvpxVp8Encoder::~LibvpxVp8Encoder() {
+  Release();
+}
+
+int LibvpxVp8Encoder::Release() {
+  int ret_val = WEBRTC_VIDEO_CODEC_OK;
+
+  encoded_images_.clear();
+
+  if (inited_) {
+    for (auto it = encoders_.rbegin(); it != encoders_.rend(); ++it) {
+      if (libvpx_->codec_destroy(&*it)) {
+        ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
+      }
+    }
+  }
+  encoders_.clear();
+
+  vpx_configs_.clear();
+  config_overrides_.clear();
+  send_stream_.clear();
+  cpu_speed_.clear();
+
+  for (auto it = raw_images_.rbegin(); it != raw_images_.rend(); ++it) {
+    libvpx_->img_free(&*it);
+  }
+  raw_images_.clear();
+
+  frame_buffer_controller_.reset();
+  inited_ = false;
+  return ret_val;
+}
+
+void LibvpxVp8Encoder::SetRates(const RateControlParameters& parameters) {
+  if (!inited_) {
+    RTC_LOG(LS_WARNING) << "SetRates() while not initialize";
+    return;
+  }
+
+  if (encoders_[0].err) {
+    RTC_LOG(LS_WARNING) << "Encoder in error state.";
+    return;
+  }
+
+  if (parameters.framerate_fps < 1.0) {
+    RTC_LOG(LS_WARNING) << "Unsupported framerate (must be >= 1.0): "
+                        << parameters.framerate_fps;
+    return;
+  }
+
+  if (parameters.bitrate.get_sum_bps() == 0) {
+    // Encoder paused, turn off all encoding.
+    const int num_streams = static_cast<size_t>(encoders_.size());
+    for (int i = 0; i < num_streams; ++i)
+      SetStreamState(false, i);
+    return;
+  }
+
+  codec_.maxFramerate = static_cast<uint32_t>(parameters.framerate_fps + 0.5);
+
+  if (encoders_.size() > 1) {
+    // If we have more than 1 stream, reduce the qp_max for the low resolution
+    // stream if frame rate is not too low. The trade-off with lower qp_max is
+    // possibly more dropped frames, so we only do this if the frame rate is
+    // above some threshold (base temporal layer is down to 1/4 for 3 layers).
+    // We may want to condition this on bitrate later.
+    if (rate_control_settings_.Vp8BoostBaseLayerQuality() &&
+        parameters.framerate_fps > 20.0) {
+      vpx_configs_[encoders_.size() - 1].rc_max_quantizer = 45;
+    } else {
+      // Go back to default value set in InitEncode.
+      vpx_configs_[encoders_.size() - 1].rc_max_quantizer = qp_max_;
+    }
+  }
+
+  for (size_t i = 0; i < encoders_.size(); ++i) {
+    const size_t stream_idx = encoders_.size() - 1 - i;
+
+    unsigned int target_bitrate_kbps =
+        parameters.bitrate.GetSpatialLayerSum(stream_idx) / 1000;
+
+    bool send_stream = target_bitrate_kbps > 0;
+    if (send_stream || encoders_.size() > 1)
+      SetStreamState(send_stream, stream_idx);
+
+    vpx_configs_[i].rc_target_bitrate = target_bitrate_kbps;
+    if (send_stream) {
+      frame_buffer_controller_->OnRatesUpdated(
+          stream_idx, parameters.bitrate.GetTemporalLayerAllocation(stream_idx),
+          static_cast<int>(parameters.framerate_fps + 0.5));
+    }
+
+    UpdateVpxConfiguration(stream_idx);
+
+    vpx_codec_err_t err =
+        libvpx_->codec_enc_config_set(&encoders_[i], &vpx_configs_[i]);
+    if (err != VPX_CODEC_OK) {
+      RTC_LOG(LS_WARNING) << "Error configuring codec, error code: " << err
+                          << ", details: "
+                          << libvpx_->codec_error_detail(&encoders_[i]);
+    }
+  }
+}
+
+void LibvpxVp8Encoder::OnPacketLossRateUpdate(float packet_loss_rate) {
+  // TODO(bugs.webrtc.org/10431): Replace condition by DCHECK.
+  if (frame_buffer_controller_) {
+    frame_buffer_controller_->OnPacketLossRateUpdate(packet_loss_rate);
+  }
+}
+
+void LibvpxVp8Encoder::OnRttUpdate(int64_t rtt_ms) {
+  // TODO(bugs.webrtc.org/10431): Replace condition by DCHECK.
+  if (frame_buffer_controller_) {
+    frame_buffer_controller_->OnRttUpdate(rtt_ms);
+  }
+}
+
+void LibvpxVp8Encoder::OnLossNotification(
+    const LossNotification& loss_notification) {
+  if (frame_buffer_controller_) {
+    frame_buffer_controller_->OnLossNotification(loss_notification);
+  }
+}
+
+void LibvpxVp8Encoder::SetStreamState(bool send_stream, int stream_idx) {
+  if (send_stream && !send_stream_[stream_idx]) {
+    // Need a key frame if we have not sent this stream before.
+    key_frame_request_[stream_idx] = true;
+  }
+  send_stream_[stream_idx] = send_stream;
+}
+
+void LibvpxVp8Encoder::SetFecControllerOverride(
+    FecControllerOverride* fec_controller_override) {
+  // TODO(bugs.webrtc.org/10769): Update downstream and remove ability to
+  // pass nullptr.
+  // RTC_DCHECK(fec_controller_override);
+  RTC_DCHECK(!fec_controller_override_);
+  fec_controller_override_ = fec_controller_override;
+}
+
+// TODO(eladalon): s/inst/codec_settings/g.
+int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst,
+                                 const VideoEncoder::Settings& settings) {
+  if (inst == NULL) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (inst->maxFramerate < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  // allow zero to represent an unspecified maxBitRate
+  if (inst->maxBitrate > 0 && inst->startBitrate > inst->maxBitrate) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (inst->width < 1 || inst->height < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (settings.number_of_cores < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  if (absl::optional<ScalabilityMode> scalability_mode =
+          inst->GetScalabilityMode();
+      scalability_mode.has_value() &&
+      !VP8SupportsScalabilityMode(*scalability_mode)) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  num_active_streams_ = 0;
+  for (int i = 0; i < inst->numberOfSimulcastStreams; ++i) {
+    if (inst->simulcastStream[i].active) {
+      ++num_active_streams_;
+    }
+  }
+  if (inst->numberOfSimulcastStreams == 0 && inst->active) {
+    num_active_streams_ = 1;
+  }
+
+  if (inst->VP8().automaticResizeOn && num_active_streams_ > 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  // Use the previous pixel format to avoid extra image allocations.
+  vpx_img_fmt_t pixel_format =
+      raw_images_.empty() ? VPX_IMG_FMT_I420 : raw_images_[0].fmt;
+
+  int retVal = Release();
+  if (retVal < 0) {
+    return retVal;
+  }
+
+  int number_of_streams = SimulcastUtility::NumberOfSimulcastStreams(*inst);
+  if (number_of_streams > 1 &&
+      !SimulcastUtility::ValidSimulcastParameters(*inst, number_of_streams)) {
+    return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
+  }
+
+  RTC_DCHECK(!frame_buffer_controller_);
+  if (frame_buffer_controller_factory_) {
+    frame_buffer_controller_ = frame_buffer_controller_factory_->Create(
+        *inst, settings, fec_controller_override_);
+  } else {
+    Vp8TemporalLayersFactory factory;
+    frame_buffer_controller_ =
+        factory.Create(*inst, settings, fec_controller_override_);
+  }
+  RTC_DCHECK(frame_buffer_controller_);
+
+  number_of_cores_ = settings.number_of_cores;
+  timestamp_ = 0;
+  codec_ = *inst;
+
+  // Code expects simulcastStream resolutions to be correct, make sure they are
+  // filled even when there are no simulcast layers.
+  if (codec_.numberOfSimulcastStreams == 0) {
+    codec_.simulcastStream[0].width = codec_.width;
+    codec_.simulcastStream[0].height = codec_.height;
+  }
+
+  encoded_images_.resize(number_of_streams);
+  encoders_.resize(number_of_streams);
+  vpx_configs_.resize(number_of_streams);
+  config_overrides_.resize(number_of_streams);
+  downsampling_factors_.resize(number_of_streams);
+  raw_images_.resize(number_of_streams);
+  send_stream_.resize(number_of_streams);
+  send_stream_[0] = true;  // For non-simulcast case.
+  cpu_speed_.resize(number_of_streams);
+  std::fill(key_frame_request_.begin(), key_frame_request_.end(), false);
+
+  int idx = number_of_streams - 1;
+  for (int i = 0; i < (number_of_streams - 1); ++i, --idx) {
+    int gcd = GCD(inst->simulcastStream[idx].width,
+                  inst->simulcastStream[idx - 1].width);
+    downsampling_factors_[i].num = inst->simulcastStream[idx].width / gcd;
+    downsampling_factors_[i].den = inst->simulcastStream[idx - 1].width / gcd;
+    send_stream_[i] = false;
+  }
+  if (number_of_streams > 1) {
+    send_stream_[number_of_streams - 1] = false;
+    downsampling_factors_[number_of_streams - 1].num = 1;
+    downsampling_factors_[number_of_streams - 1].den = 1;
+  }
+
+  // populate encoder configuration with default values
+  if (libvpx_->codec_enc_config_default(vpx_codec_vp8_cx(), &vpx_configs_[0],
+                                        0)) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  // setting the time base of the codec
+  vpx_configs_[0].g_timebase.num = 1;
+  vpx_configs_[0].g_timebase.den = kRtpTicksPerSecond;
+  vpx_configs_[0].g_lag_in_frames = 0;  // 0- no frame lagging
+
+  // Set the error resilience mode for temporal layers (but not simulcast).
+  vpx_configs_[0].g_error_resilient =
+      (SimulcastUtility::NumberOfTemporalLayers(*inst, 0) > 1)
+          ? VPX_ERROR_RESILIENT_DEFAULT
+          : 0;
+
+  // Override the error resilience mode if this is not simulcast, but we are
+  // using temporal layers.
+  if (field_trial::IsEnabled(kVp8ForcePartitionResilience) &&
+      (number_of_streams == 1) &&
+      (SimulcastUtility::NumberOfTemporalLayers(*inst, 0) > 1)) {
+    RTC_LOG(LS_INFO) << "Overriding g_error_resilient from "
+                     << vpx_configs_[0].g_error_resilient << " to "
+                     << VPX_ERROR_RESILIENT_PARTITIONS;
+    vpx_configs_[0].g_error_resilient = VPX_ERROR_RESILIENT_PARTITIONS;
+  }
+
+  // rate control settings
+  vpx_configs_[0].rc_dropframe_thresh = FrameDropThreshold(0);
+  vpx_configs_[0].rc_end_usage = VPX_CBR;
+  vpx_configs_[0].g_pass = VPX_RC_ONE_PASS;
+  // Handle resizing outside of libvpx.
+  vpx_configs_[0].rc_resize_allowed = 0;
+  vpx_configs_[0].rc_min_quantizer =
+      codec_.mode == VideoCodecMode::kScreensharing ? 12 : 2;
+  if (inst->qpMax >= vpx_configs_[0].rc_min_quantizer) {
+    qp_max_ = inst->qpMax;
+  }
+  if (rate_control_settings_.LibvpxVp8QpMax()) {
+    qp_max_ = std::max(rate_control_settings_.LibvpxVp8QpMax().value(),
+                       static_cast<int>(vpx_configs_[0].rc_min_quantizer));
+  }
+  vpx_configs_[0].rc_max_quantizer = qp_max_;
+  vpx_configs_[0].rc_undershoot_pct = 100;
+  vpx_configs_[0].rc_overshoot_pct = 15;
+  vpx_configs_[0].rc_buf_initial_sz = 500;
+  vpx_configs_[0].rc_buf_optimal_sz = 600;
+  vpx_configs_[0].rc_buf_sz = 1000;
+
+  // Set the maximum target size of any key-frame.
+  rc_max_intra_target_ = MaxIntraTarget(vpx_configs_[0].rc_buf_optimal_sz);
+
+  if (inst->VP8().keyFrameInterval > 0) {
+    vpx_configs_[0].kf_mode = VPX_KF_AUTO;
+    vpx_configs_[0].kf_max_dist = inst->VP8().keyFrameInterval;
+  } else {
+    vpx_configs_[0].kf_mode = VPX_KF_DISABLED;
+  }
+
+  // Allow the user to set the complexity for the base stream.
+  switch (inst->GetVideoEncoderComplexity()) {
+    case VideoCodecComplexity::kComplexityHigh:
+      cpu_speed_[0] = -5;
+      break;
+    case VideoCodecComplexity::kComplexityHigher:
+      cpu_speed_[0] = -4;
+      break;
+    case VideoCodecComplexity::kComplexityMax:
+      cpu_speed_[0] = -3;
+      break;
+    default:
+      cpu_speed_[0] = -6;
+      break;
+  }
+  cpu_speed_default_ = cpu_speed_[0];
+  // Set encoding complexity (cpu_speed) based on resolution and/or platform.
+  cpu_speed_[0] = GetCpuSpeed(inst->width, inst->height);
+  for (int i = 1; i < number_of_streams; ++i) {
+    cpu_speed_[i] =
+        GetCpuSpeed(inst->simulcastStream[number_of_streams - 1 - i].width,
+                    inst->simulcastStream[number_of_streams - 1 - i].height);
+  }
+  vpx_configs_[0].g_w = inst->width;
+  vpx_configs_[0].g_h = inst->height;
+
+  // Determine number of threads based on the image size and #cores.
+  // TODO(fbarchard): Consider number of Simulcast layers.
+  vpx_configs_[0].g_threads = NumberOfThreads(
+      vpx_configs_[0].g_w, vpx_configs_[0].g_h, settings.number_of_cores);
+
+  // Creating a wrapper to the image - setting image data to NULL.
+  // Actual pointer will be set in encode. Setting align to 1, as it
+  // is meaningless (no memory allocation is done here).
+  libvpx_->img_wrap(&raw_images_[0], pixel_format, inst->width, inst->height, 1,
+                    NULL);
+
+  // Note the order we use is different from webm, we have lowest resolution
+  // at position 0 and they have highest resolution at position 0.
+  const size_t stream_idx_cfg_0 = encoders_.size() - 1;
+  SimulcastRateAllocator init_allocator(codec_);
+  VideoBitrateAllocation allocation =
+      init_allocator.Allocate(VideoBitrateAllocationParameters(
+          inst->startBitrate * 1000, inst->maxFramerate));
+  std::vector<uint32_t> stream_bitrates;
+  for (int i = 0; i == 0 || i < inst->numberOfSimulcastStreams; ++i) {
+    uint32_t bitrate = allocation.GetSpatialLayerSum(i) / 1000;
+    stream_bitrates.push_back(bitrate);
+  }
+
+  vpx_configs_[0].rc_target_bitrate = stream_bitrates[stream_idx_cfg_0];
+  if (stream_bitrates[stream_idx_cfg_0] > 0) {
+    uint32_t maxFramerate =
+        inst->simulcastStream[stream_idx_cfg_0].maxFramerate;
+    if (!maxFramerate) {
+      maxFramerate = inst->maxFramerate;
+    }
+
+    frame_buffer_controller_->OnRatesUpdated(
+        stream_idx_cfg_0,
+        allocation.GetTemporalLayerAllocation(stream_idx_cfg_0), maxFramerate);
+  }
+  frame_buffer_controller_->SetQpLimits(stream_idx_cfg_0,
+                                        vpx_configs_[0].rc_min_quantizer,
+                                        vpx_configs_[0].rc_max_quantizer);
+  UpdateVpxConfiguration(stream_idx_cfg_0);
+  vpx_configs_[0].rc_dropframe_thresh = FrameDropThreshold(stream_idx_cfg_0);
+
+  for (size_t i = 1; i < encoders_.size(); ++i) {
+    const size_t stream_idx = encoders_.size() - 1 - i;
+    memcpy(&vpx_configs_[i], &vpx_configs_[0], sizeof(vpx_configs_[0]));
+
+    vpx_configs_[i].g_w = inst->simulcastStream[stream_idx].width;
+    vpx_configs_[i].g_h = inst->simulcastStream[stream_idx].height;
+
+    // Use 1 thread for lower resolutions.
+    vpx_configs_[i].g_threads = 1;
+
+    vpx_configs_[i].rc_dropframe_thresh = FrameDropThreshold(stream_idx);
+
+    // Setting alignment to 32 - as that ensures at least 16 for all
+    // planes (32 for Y, 16 for U,V). Libvpx sets the requested stride for
+    // the y plane, but only half of it to the u and v planes.
+    libvpx_->img_alloc(
+        &raw_images_[i], pixel_format, inst->simulcastStream[stream_idx].width,
+        inst->simulcastStream[stream_idx].height, kVp832ByteAlign);
+    SetStreamState(stream_bitrates[stream_idx] > 0, stream_idx);
+    vpx_configs_[i].rc_target_bitrate = stream_bitrates[stream_idx];
+    if (stream_bitrates[stream_idx] > 0) {
+      uint32_t maxFramerate = inst->simulcastStream[stream_idx].maxFramerate;
+      if (!maxFramerate) {
+        maxFramerate = inst->maxFramerate;
+      }
+      frame_buffer_controller_->OnRatesUpdated(
+          stream_idx, allocation.GetTemporalLayerAllocation(stream_idx),
+          maxFramerate);
+    }
+    frame_buffer_controller_->SetQpLimits(stream_idx,
+                                          vpx_configs_[i].rc_min_quantizer,
+                                          vpx_configs_[i].rc_max_quantizer);
+    UpdateVpxConfiguration(stream_idx);
+  }
+
+  return InitAndSetControlSettings();
+}
+
+int LibvpxVp8Encoder::GetCpuSpeed(int width, int height) {
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
+    defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
+  // On mobile platform, use a lower speed setting for lower resolutions for
+  // CPUs with 4 or more cores.
+  RTC_DCHECK_GT(number_of_cores_, 0);
+  if (experimental_cpu_speed_config_arm_
+          .GetValue(width * height, number_of_cores_)
+          .has_value()) {
+    return experimental_cpu_speed_config_arm_
+        .GetValue(width * height, number_of_cores_)
+        .value();
+  }
+
+  if (number_of_cores_ <= 3)
+    return -12;
+
+  if (width * height <= 352 * 288)
+    return -8;
+  else if (width * height <= 640 * 480)
+    return -10;
+  else
+    return -12;
+#else
+  // For non-ARM, increase encoding complexity (i.e., use lower speed setting)
+  // if resolution is below CIF. Otherwise, keep the default/user setting
+  // (`cpu_speed_default_`) set on InitEncode via VP8().complexity.
+  if (width * height < 352 * 288)
+    return (cpu_speed_default_ < -4) ? -4 : cpu_speed_default_;
+  else
+    return cpu_speed_default_;
+#endif
+}
+
+int LibvpxVp8Encoder::NumberOfThreads(int width, int height, int cpus) {
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
+  if (width * height >= 320 * 180) {
+    if (cpus >= 4) {
+      // 3 threads for CPUs with 4 and more cores since most of times only 4
+      // cores will be active.
+      return 3;
+    } else if (cpus == 3 || cpus == 2) {
+      return 2;
+    } else {
+      return 1;
+    }
+  }
+  return 1;
+#else
+#if defined(WEBRTC_IOS)
+  std::string trial_string =
+      field_trial::FindFullName(kVP8IosMaxNumberOfThreadFieldTrial);
+  FieldTrialParameter<int> max_thread_number(
+      kVP8IosMaxNumberOfThreadFieldTrialParameter, 0);
+  ParseFieldTrial({&max_thread_number}, trial_string);
+  if (max_thread_number.Get() > 0) {
+    if (width * height < 320 * 180) {
+      return 1;  // Use single thread for small screens
+    }
+    // thread number must be less than or equal to the number of CPUs.
+    return std::min(cpus, max_thread_number.Get());
+  }
+#endif  // defined(WEBRTC_IOS)
+  if (width * height >= 1920 * 1080 && cpus > 8) {
+    return 8;  // 8 threads for 1080p on high perf machines.
+  } else if (width * height > 1280 * 960 && cpus >= 6) {
+    // 3 threads for 1080p.
+    return 3;
+  } else if (width * height > 640 * 480 && cpus >= 3) {
+    // Default 2 threads for qHD/HD, but allow 3 if core count is high enough,
+    // as this will allow more margin for high-core/low clock machines or if
+    // not built with highest optimization.
+    if (cpus >= 6) {
+      return 3;
+    }
+    return 2;
+  } else {
+    // 1 thread for VGA or less.
+    return 1;
+  }
+#endif
+}
+
+int LibvpxVp8Encoder::InitAndSetControlSettings() {
+  vpx_codec_flags_t flags = 0;
+  flags |= VPX_CODEC_USE_OUTPUT_PARTITION;
+
+  if (encoders_.size() > 1) {
+    int error = libvpx_->codec_enc_init_multi(
+        &encoders_[0], vpx_codec_vp8_cx(), &vpx_configs_[0], encoders_.size(),
+        flags, &downsampling_factors_[0]);
+    if (error) {
+      return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+  } else {
+    if (libvpx_->codec_enc_init(&encoders_[0], vpx_codec_vp8_cx(),
+                                &vpx_configs_[0], flags)) {
+      return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+  }
+  // Enable denoising for the highest resolution stream, and for
+  // the second highest resolution if we are doing more than 2
+  // spatial layers/streams.
+  // TODO(holmer): Investigate possibility of adding a libvpx API
+  // for getting the denoised frame from the encoder and using that
+  // when encoding lower resolution streams. Would it work with the
+  // multi-res encoding feature?
+  denoiserState denoiser_state = kDenoiserOnYOnly;
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
+    defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
+  denoiser_state = kDenoiserOnYOnly;
+#else
+  denoiser_state = kDenoiserOnAdaptive;
+#endif
+  libvpx_->codec_control(
+      &encoders_[0], VP8E_SET_NOISE_SENSITIVITY,
+      codec_.VP8()->denoisingOn ? denoiser_state : kDenoiserOff);
+  if (encoders_.size() > 2) {
+    libvpx_->codec_control(
+        &encoders_[1], VP8E_SET_NOISE_SENSITIVITY,
+        codec_.VP8()->denoisingOn ? denoiser_state : kDenoiserOff);
+  }
+  for (size_t i = 0; i < encoders_.size(); ++i) {
+    // Allow more screen content to be detected as static.
+    libvpx_->codec_control(
+        &(encoders_[i]), VP8E_SET_STATIC_THRESHOLD,
+        codec_.mode == VideoCodecMode::kScreensharing ? 100u : 1u);
+    libvpx_->codec_control(&(encoders_[i]), VP8E_SET_CPUUSED, cpu_speed_[i]);
+    libvpx_->codec_control(
+        &(encoders_[i]), VP8E_SET_TOKEN_PARTITIONS,
+        static_cast<vp8e_token_partitions>(kTokenPartitions));
+    libvpx_->codec_control(&(encoders_[i]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
+                           rc_max_intra_target_);
+    // VP8E_SET_SCREEN_CONTENT_MODE 2 = screen content with more aggressive
+    // rate control (drop frames on large target bitrate overshoot)
+    libvpx_->codec_control(
+        &(encoders_[i]), VP8E_SET_SCREEN_CONTENT_MODE,
+        codec_.mode == VideoCodecMode::kScreensharing ? 2u : 0u);
+  }
+  inited_ = true;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+uint32_t LibvpxVp8Encoder::MaxIntraTarget(uint32_t optimalBuffersize) {
+  // Set max to the optimal buffer level (normalized by target BR),
+  // and scaled by a scalePar.
+  // Max target size = scalePar * optimalBufferSize * targetBR[Kbps].
+  // This values is presented in percentage of perFrameBw:
+  // perFrameBw = targetBR[Kbps] * 1000 / frameRate.
+  // The target in % is as follows:
+
+  float scalePar = 0.5;
+  uint32_t targetPct = optimalBuffersize * scalePar * codec_.maxFramerate / 10;
+
+  // Don't go below 3 times the per frame bandwidth.
+  const uint32_t minIntraTh = 300;
+  return (targetPct < minIntraTh) ? minIntraTh : targetPct;
+}
+
+uint32_t LibvpxVp8Encoder::FrameDropThreshold(size_t spatial_idx) const {
+  if (!codec_.GetFrameDropEnabled()) {
+    return 0;
+  }
+
+  // If temporal layers are used, they get to override the frame dropping
+  // setting, as eg. ScreenshareLayers does not work as intended with frame
+  // dropping on and DefaultTemporalLayers will have performance issues with
+  // frame dropping off.
+  RTC_DCHECK(frame_buffer_controller_);
+  RTC_DCHECK_LT(spatial_idx, frame_buffer_controller_->StreamCount());
+  return frame_buffer_controller_->SupportsEncoderFrameDropping(spatial_idx)
+             ? 30
+             : 0;
+}
+
+size_t LibvpxVp8Encoder::SteadyStateSize(int sid, int tid) {
+  const int encoder_id = encoders_.size() - 1 - sid;
+  size_t bitrate_bps;
+  float fps;
+  if ((SimulcastUtility::IsConferenceModeScreenshare(codec_) && sid == 0) ||
+      vpx_configs_[encoder_id].ts_number_layers <= 1) {
+    // In conference screenshare there's no defined per temporal layer bitrate
+    // and framerate.
+    bitrate_bps = vpx_configs_[encoder_id].rc_target_bitrate * 1000;
+    fps = codec_.maxFramerate;
+  } else {
+    bitrate_bps = vpx_configs_[encoder_id].ts_target_bitrate[tid] * 1000;
+    fps = codec_.maxFramerate /
+          fmax(vpx_configs_[encoder_id].ts_rate_decimator[tid], 1.0);
+    if (tid > 0) {
+      // Layer bitrate and fps are counted as a partial sums.
+      bitrate_bps -= vpx_configs_[encoder_id].ts_target_bitrate[tid - 1] * 1000;
+      fps = codec_.maxFramerate /
+            fmax(vpx_configs_[encoder_id].ts_rate_decimator[tid - 1], 1.0);
+    }
+  }
+
+  if (fps < 1e-9)
+    return 0;
+  return static_cast<size_t>(
+      bitrate_bps / (8 * fps) *
+          (100 -
+           variable_framerate_experiment_.steady_state_undershoot_percentage) /
+          100 +
+      0.5);
+}
+
+bool LibvpxVp8Encoder::UpdateVpxConfiguration(size_t stream_index) {
+  RTC_DCHECK(frame_buffer_controller_);
+
+  const size_t config_index = vpx_configs_.size() - 1 - stream_index;
+
+  RTC_DCHECK_LT(config_index, config_overrides_.size());
+  Vp8EncoderConfig* config = &config_overrides_[config_index];
+
+  const Vp8EncoderConfig new_config =
+      frame_buffer_controller_->UpdateConfiguration(stream_index);
+
+  if (new_config.reset_previous_configuration_overrides) {
+    *config = new_config;
+    return true;
+  }
+
+  const bool changes_made = MaybeExtendVp8EncoderConfig(new_config, config);
+
+  // Note that overrides must be applied even if they haven't changed.
+  RTC_DCHECK_LT(config_index, vpx_configs_.size());
+  vpx_codec_enc_cfg_t* vpx_config = &vpx_configs_[config_index];
+  ApplyVp8EncoderConfigToVpxConfig(*config, vpx_config);
+
+  return changes_made;
+}
+
+int LibvpxVp8Encoder::Encode(const VideoFrame& frame,
+                             const std::vector<VideoFrameType>* frame_types) {
+  RTC_DCHECK_EQ(frame.width(), codec_.width);
+  RTC_DCHECK_EQ(frame.height(), codec_.height);
+
+  if (!inited_)
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  if (encoded_complete_callback_ == NULL)
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+
+  bool key_frame_requested = false;
+  for (size_t i = 0; i < key_frame_request_.size() && i < send_stream_.size();
+       ++i) {
+    if (key_frame_request_[i] && send_stream_[i]) {
+      key_frame_requested = true;
+      break;
+    }
+  }
+  if (!key_frame_requested && frame_types) {
+    for (size_t i = 0; i < frame_types->size() && i < send_stream_.size();
+         ++i) {
+      if ((*frame_types)[i] == VideoFrameType::kVideoFrameKey &&
+          send_stream_[i]) {
+        key_frame_requested = true;
+        break;
+      }
+    }
+  }
+
+  if (frame.update_rect().IsEmpty() && num_steady_state_frames_ >= 3 &&
+      !key_frame_requested) {
+    if (variable_framerate_experiment_.enabled &&
+        framerate_controller_.DropFrame(frame.timestamp() / kRtpTicksPerMs)) {
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+    framerate_controller_.AddFrame(frame.timestamp() / kRtpTicksPerMs);
+  }
+
+  bool send_key_frame = key_frame_requested;
+  bool drop_frame = false;
+  bool retransmission_allowed = true;
+  Vp8FrameConfig tl_configs[kMaxSimulcastStreams];
+  for (size_t i = 0; i < encoders_.size(); ++i) {
+    tl_configs[i] =
+        frame_buffer_controller_->NextFrameConfig(i, frame.timestamp());
+    send_key_frame |= tl_configs[i].IntraFrame();
+    drop_frame |= tl_configs[i].drop_frame;
+    RTC_DCHECK(i == 0 ||
+               retransmission_allowed == tl_configs[i].retransmission_allowed);
+    retransmission_allowed = tl_configs[i].retransmission_allowed;
+  }
+
+  if (drop_frame && !send_key_frame) {
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  vpx_enc_frame_flags_t flags[kMaxSimulcastStreams];
+  for (size_t i = 0; i < encoders_.size(); ++i) {
+    flags[i] = send_key_frame ? VPX_EFLAG_FORCE_KF : EncodeFlags(tl_configs[i]);
+  }
+
+  // Scale and map buffers and set `raw_images_` to hold pointers to the result.
+  // Because `raw_images_` are set to hold pointers to the prepared buffers, we
+  // need to keep these buffers alive through reference counting until after
+  // encoding is complete.
+  std::vector<rtc::scoped_refptr<VideoFrameBuffer>> prepared_buffers =
+      PrepareBuffers(frame.video_frame_buffer());
+  if (prepared_buffers.empty()) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  struct CleanUpOnExit {
+    explicit CleanUpOnExit(
+        vpx_image_t* raw_image,
+        std::vector<rtc::scoped_refptr<VideoFrameBuffer>> prepared_buffers)
+        : raw_image_(raw_image),
+          prepared_buffers_(std::move(prepared_buffers)) {}
+    ~CleanUpOnExit() {
+      raw_image_->planes[VPX_PLANE_Y] = nullptr;
+      raw_image_->planes[VPX_PLANE_U] = nullptr;
+      raw_image_->planes[VPX_PLANE_V] = nullptr;
+    }
+    vpx_image_t* raw_image_;
+    std::vector<rtc::scoped_refptr<VideoFrameBuffer>> prepared_buffers_;
+  } clean_up_on_exit(&raw_images_[0], std::move(prepared_buffers));
+
+  if (send_key_frame) {
+    // Adapt the size of the key frame when in screenshare with 1 temporal
+    // layer.
+    if (encoders_.size() == 1 &&
+        codec_.mode == VideoCodecMode::kScreensharing &&
+        codec_.VP8()->numberOfTemporalLayers <= 1) {
+      const uint32_t forceKeyFrameIntraTh = 100;
+      libvpx_->codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
+                             forceKeyFrameIntraTh);
+    }
+
+    std::fill(key_frame_request_.begin(), key_frame_request_.end(), false);
+  }
+
+  // Set the encoder frame flags and temporal layer_id for each spatial stream.
+  // Note that streams are defined starting from lowest resolution at
+  // position 0 to highest resolution at position |encoders_.size() - 1|,
+  // whereas `encoder_` is from highest to lowest resolution.
+  for (size_t i = 0; i < encoders_.size(); ++i) {
+    const size_t stream_idx = encoders_.size() - 1 - i;
+
+    if (UpdateVpxConfiguration(stream_idx)) {
+      if (libvpx_->codec_enc_config_set(&encoders_[i], &vpx_configs_[i]))
+        return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
+    libvpx_->codec_control(&encoders_[i], VP8E_SET_FRAME_FLAGS,
+                           static_cast<int>(flags[stream_idx]));
+    libvpx_->codec_control(&encoders_[i], VP8E_SET_TEMPORAL_LAYER_ID,
+                           tl_configs[i].encoder_layer_id);
+  }
+  // TODO(holmer): Ideally the duration should be the timestamp diff of this
+  // frame and the next frame to be encoded, which we don't have. Instead we
+  // would like to use the duration of the previous frame. Unfortunately the
+  // rate control seems to be off with that setup. Using the average input
+  // frame rate to calculate an average duration for now.
+  RTC_DCHECK_GT(codec_.maxFramerate, 0);
+  uint32_t duration = kRtpTicksPerSecond / codec_.maxFramerate;
+
+  int error = WEBRTC_VIDEO_CODEC_OK;
+  int num_tries = 0;
+  // If the first try returns WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT
+  // the frame must be reencoded with the same parameters again because
+  // target bitrate is exceeded and encoder state has been reset.
+  while (num_tries == 0 ||
+         (num_tries == 1 &&
+          error == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT)) {
+    ++num_tries;
+    // Note we must pass 0 for `flags` field in encode call below since they are
+    // set above in `libvpx_interface_->vpx_codec_control_` function for each
+    // encoder/spatial layer.
+    error = libvpx_->codec_encode(&encoders_[0], &raw_images_[0], timestamp_,
+                                  duration, 0, VPX_DL_REALTIME);
+    // Reset specific intra frame thresholds, following the key frame.
+    if (send_key_frame) {
+      libvpx_->codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
+                             rc_max_intra_target_);
+    }
+    if (error)
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    // Examines frame timestamps only.
+    error = GetEncodedPartitions(frame, retransmission_allowed);
+  }
+  // TODO(sprang): Shouldn't we use the frame timestamp instead?
+  timestamp_ += duration;
+  return error;
+}
+
+void LibvpxVp8Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
+                                             const vpx_codec_cx_pkt_t& pkt,
+                                             int stream_idx,
+                                             int encoder_idx,
+                                             uint32_t timestamp) {
+  RTC_DCHECK(codec_specific);
+  codec_specific->codecType = kVideoCodecVP8;
+  codec_specific->codecSpecific.VP8.keyIdx =
+      kNoKeyIdx;  // TODO(hlundin) populate this
+  codec_specific->codecSpecific.VP8.nonReference =
+      (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) != 0;
+
+  int qp = 0;
+  vpx_codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER_64, &qp);
+  bool is_keyframe = (pkt.data.frame.flags & VPX_FRAME_IS_KEY) != 0;
+  frame_buffer_controller_->OnEncodeDone(stream_idx, timestamp,
+                                         encoded_images_[encoder_idx].size(),
+                                         is_keyframe, qp, codec_specific);
+  if (is_keyframe && codec_specific->template_structure != absl::nullopt) {
+    // Number of resolutions must match number of spatial layers, VP8 structures
+    // expected to use single spatial layer. Templates must be ordered by
+    // spatial_id, so assumption there is exactly one spatial layer is same as
+    // assumption last template uses spatial_id = 0.
+    // This check catches potential scenario where template_structure is shared
+    // across multiple vp8 streams and they are distinguished using spatial_id.
+    // Assigning single resolution doesn't support such scenario, i.e. assumes
+    // vp8 simulcast is sent using multiple ssrcs.
+    RTC_DCHECK(!codec_specific->template_structure->templates.empty());
+    RTC_DCHECK_EQ(
+        codec_specific->template_structure->templates.back().spatial_id, 0);
+    codec_specific->template_structure->resolutions = {
+        RenderResolution(pkt.data.frame.width[0], pkt.data.frame.height[0])};
+  }
+  switch (vpx_configs_[encoder_idx].ts_number_layers) {
+    case 1:
+      codec_specific->scalability_mode = ScalabilityMode::kL1T1;
+      break;
+    case 2:
+      codec_specific->scalability_mode = ScalabilityMode::kL1T2;
+      break;
+    case 3:
+      codec_specific->scalability_mode = ScalabilityMode::kL1T3;
+      break;
+  }
+}
+
+int LibvpxVp8Encoder::GetEncodedPartitions(const VideoFrame& input_image,
+                                           bool retransmission_allowed) {
+  int stream_idx = static_cast<int>(encoders_.size()) - 1;
+  int result = WEBRTC_VIDEO_CODEC_OK;
+  for (size_t encoder_idx = 0; encoder_idx < encoders_.size();
+       ++encoder_idx, --stream_idx) {
+    vpx_codec_iter_t iter = NULL;
+    encoded_images_[encoder_idx].set_size(0);
+    encoded_images_[encoder_idx]._frameType = VideoFrameType::kVideoFrameDelta;
+    CodecSpecificInfo codec_specific;
+    const vpx_codec_cx_pkt_t* pkt = NULL;
+
+    size_t encoded_size = 0;
+    while ((pkt = libvpx_->codec_get_cx_data(&encoders_[encoder_idx], &iter)) !=
+           NULL) {
+      if (pkt->kind == VPX_CODEC_CX_FRAME_PKT) {
+        encoded_size += pkt->data.frame.sz;
+      }
+    }
+
+    auto buffer = EncodedImageBuffer::Create(encoded_size);
+
+    iter = NULL;
+    size_t encoded_pos = 0;
+    while ((pkt = libvpx_->codec_get_cx_data(&encoders_[encoder_idx], &iter)) !=
+           NULL) {
+      switch (pkt->kind) {
+        case VPX_CODEC_CX_FRAME_PKT: {
+          RTC_CHECK_LE(encoded_pos + pkt->data.frame.sz, buffer->size());
+          memcpy(&buffer->data()[encoded_pos], pkt->data.frame.buf,
+                 pkt->data.frame.sz);
+          encoded_pos += pkt->data.frame.sz;
+          break;
+        }
+        default:
+          break;
+      }
+      // End of frame
+      if ((pkt->data.frame.flags & VPX_FRAME_IS_FRAGMENT) == 0) {
+        // check if encoded frame is a key frame
+        if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
+          encoded_images_[encoder_idx]._frameType =
+              VideoFrameType::kVideoFrameKey;
+        }
+        encoded_images_[encoder_idx].SetEncodedData(buffer);
+        encoded_images_[encoder_idx].set_size(encoded_pos);
+        encoded_images_[encoder_idx].SetSpatialIndex(stream_idx);
+        PopulateCodecSpecific(&codec_specific, *pkt, stream_idx, encoder_idx,
+                              input_image.timestamp());
+        if (codec_specific.codecSpecific.VP8.temporalIdx != kNoTemporalIdx) {
+          encoded_images_[encoder_idx].SetTemporalIndex(
+              codec_specific.codecSpecific.VP8.temporalIdx);
+        }
+        break;
+      }
+    }
+    encoded_images_[encoder_idx].SetTimestamp(input_image.timestamp());
+    encoded_images_[encoder_idx].SetColorSpace(input_image.color_space());
+    encoded_images_[encoder_idx].SetRetransmissionAllowed(
+        retransmission_allowed);
+
+    if (send_stream_[stream_idx]) {
+      if (encoded_images_[encoder_idx].size() > 0) {
+        TRACE_COUNTER_ID1("webrtc", "EncodedFrameSize", encoder_idx,
+                          encoded_images_[encoder_idx].size());
+        encoded_images_[encoder_idx]._encodedHeight =
+            codec_.simulcastStream[stream_idx].height;
+        encoded_images_[encoder_idx]._encodedWidth =
+            codec_.simulcastStream[stream_idx].width;
+        int qp_128 = -1;
+        libvpx_->codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER,
+                               &qp_128);
+        encoded_images_[encoder_idx].qp_ = qp_128;
+        encoded_complete_callback_->OnEncodedImage(encoded_images_[encoder_idx],
+                                                   &codec_specific);
+        const size_t steady_state_size = SteadyStateSize(
+            stream_idx, codec_specific.codecSpecific.VP8.temporalIdx);
+        if (qp_128 > variable_framerate_experiment_.steady_state_qp ||
+            encoded_images_[encoder_idx].size() > steady_state_size) {
+          num_steady_state_frames_ = 0;
+        } else {
+          ++num_steady_state_frames_;
+        }
+      } else if (!frame_buffer_controller_->SupportsEncoderFrameDropping(
+                     stream_idx)) {
+        result = WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT;
+        if (encoded_images_[encoder_idx].size() == 0) {
+          // Dropped frame that will be re-encoded.
+          frame_buffer_controller_->OnFrameDropped(stream_idx,
+                                                   input_image.timestamp());
+        }
+      }
+    }
+  }
+  return result;
+}
+
+VideoEncoder::EncoderInfo LibvpxVp8Encoder::GetEncoderInfo() const {
+  EncoderInfo info;
+  info.supports_native_handle = false;
+  info.implementation_name = "libvpx";
+  info.has_trusted_rate_controller =
+      rate_control_settings_.LibvpxVp8TrustedRateController();
+  info.is_hardware_accelerated = false;
+  info.supports_simulcast = true;
+  if (!resolution_bitrate_limits_.empty()) {
+    info.resolution_bitrate_limits = resolution_bitrate_limits_;
+  }
+  if (encoder_info_override_.requested_resolution_alignment()) {
+    info.requested_resolution_alignment =
+        *encoder_info_override_.requested_resolution_alignment();
+    info.apply_alignment_to_all_simulcast_layers =
+        encoder_info_override_.apply_alignment_to_all_simulcast_layers();
+  }
+  if (!encoder_info_override_.resolution_bitrate_limits().empty()) {
+    info.resolution_bitrate_limits =
+        encoder_info_override_.resolution_bitrate_limits();
+  }
+
+  const bool enable_scaling =
+      num_active_streams_ == 1 &&
+      (vpx_configs_.empty() || vpx_configs_[0].rc_dropframe_thresh > 0) &&
+      codec_.VP8().automaticResizeOn;
+
+  info.scaling_settings = enable_scaling
+                              ? VideoEncoder::ScalingSettings(
+                                    kLowVp8QpThreshold, kHighVp8QpThreshold)
+                              : VideoEncoder::ScalingSettings::kOff;
+  if (rate_control_settings_.LibvpxVp8MinPixels()) {
+    info.scaling_settings.min_pixels_per_frame =
+        rate_control_settings_.LibvpxVp8MinPixels().value();
+  }
+  info.preferred_pixel_formats = {VideoFrameBuffer::Type::kI420,
+                                  VideoFrameBuffer::Type::kNV12};
+
+  if (inited_) {
+    // `encoder_idx` is libvpx index where 0 is highest resolution.
+    // `si` is simulcast index, where 0 is lowest resolution.
+    for (size_t si = 0, encoder_idx = encoders_.size() - 1;
+         si < encoders_.size(); ++si, --encoder_idx) {
+      info.fps_allocation[si].clear();
+      if ((codec_.numberOfSimulcastStreams > si &&
+           !codec_.simulcastStream[si].active) ||
+          (si == 0 && SimulcastUtility::IsConferenceModeScreenshare(codec_))) {
+        // No defined frame rate fractions if not active or if using
+        // ScreenshareLayers, leave vector empty and continue;
+        continue;
+      }
+      if (vpx_configs_[encoder_idx].ts_number_layers <= 1) {
+        info.fps_allocation[si].push_back(EncoderInfo::kMaxFramerateFraction);
+      } else {
+        for (size_t ti = 0; ti < vpx_configs_[encoder_idx].ts_number_layers;
+             ++ti) {
+          RTC_DCHECK_GT(vpx_configs_[encoder_idx].ts_rate_decimator[ti], 0);
+          info.fps_allocation[si].push_back(rtc::saturated_cast<uint8_t>(
+              EncoderInfo::kMaxFramerateFraction /
+                  vpx_configs_[encoder_idx].ts_rate_decimator[ti] +
+              0.5));
+        }
+      }
+    }
+  }
+
+  return info;
+}
+
+int LibvpxVp8Encoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  encoded_complete_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void LibvpxVp8Encoder::MaybeUpdatePixelFormat(vpx_img_fmt fmt) {
+  RTC_DCHECK(!raw_images_.empty());
+  if (raw_images_[0].fmt == fmt) {
+    RTC_DCHECK(std::all_of(
+        std::next(raw_images_.begin()), raw_images_.end(),
+        [fmt](const vpx_image_t& raw_img) { return raw_img.fmt == fmt; }))
+        << "Not all raw images had the right format!";
+    return;
+  }
+  RTC_LOG(LS_INFO) << "Updating vp8 encoder pixel format to "
+                   << (fmt == VPX_IMG_FMT_NV12 ? "NV12" : "I420");
+  for (size_t i = 0; i < raw_images_.size(); ++i) {
+    vpx_image_t& img = raw_images_[i];
+    auto d_w = img.d_w;
+    auto d_h = img.d_h;
+    libvpx_->img_free(&img);
+    // First image is wrapping the input frame, the rest are allocated.
+    if (i == 0) {
+      libvpx_->img_wrap(&img, fmt, d_w, d_h, 1, NULL);
+    } else {
+      libvpx_->img_alloc(&img, fmt, d_w, d_h, kVp832ByteAlign);
+    }
+  }
+}
+
+std::vector<rtc::scoped_refptr<VideoFrameBuffer>>
+LibvpxVp8Encoder::PrepareBuffers(rtc::scoped_refptr<VideoFrameBuffer> buffer) {
+  RTC_DCHECK_EQ(buffer->width(), raw_images_[0].d_w);
+  RTC_DCHECK_EQ(buffer->height(), raw_images_[0].d_h);
+  absl::InlinedVector<VideoFrameBuffer::Type, kMaxPreferredPixelFormats>
+      supported_formats = {VideoFrameBuffer::Type::kI420,
+                           VideoFrameBuffer::Type::kNV12};
+
+  rtc::scoped_refptr<VideoFrameBuffer> mapped_buffer;
+  if (buffer->type() != VideoFrameBuffer::Type::kNative) {
+    // `buffer` is already mapped.
+    mapped_buffer = buffer;
+  } else {
+    // Attempt to map to one of the supported formats.
+    mapped_buffer = buffer->GetMappedFrameBuffer(supported_formats);
+  }
+  if (!mapped_buffer ||
+      (absl::c_find(supported_formats, mapped_buffer->type()) ==
+           supported_formats.end() &&
+       mapped_buffer->type() != VideoFrameBuffer::Type::kI420A)) {
+    // Unknown pixel format or unable to map, convert to I420 and prepare that
+    // buffer instead to ensure Scale() is safe to use.
+    auto converted_buffer = buffer->ToI420();
+    if (!converted_buffer) {
+      RTC_LOG(LS_ERROR) << "Failed to convert "
+                        << VideoFrameBufferTypeToString(buffer->type())
+                        << " image to I420. Can't encode frame.";
+      return {};
+    }
+    RTC_CHECK(converted_buffer->type() == VideoFrameBuffer::Type::kI420 ||
+              converted_buffer->type() == VideoFrameBuffer::Type::kI420A);
+
+    // Because `buffer` had to be converted, use `converted_buffer` instead...
+    buffer = mapped_buffer = converted_buffer;
+  }
+
+  // Maybe update pixel format.
+  absl::InlinedVector<VideoFrameBuffer::Type, kMaxPreferredPixelFormats>
+      mapped_type = {mapped_buffer->type()};
+  switch (mapped_buffer->type()) {
+    case VideoFrameBuffer::Type::kI420:
+    case VideoFrameBuffer::Type::kI420A:
+      MaybeUpdatePixelFormat(VPX_IMG_FMT_I420);
+      break;
+    case VideoFrameBuffer::Type::kNV12:
+      MaybeUpdatePixelFormat(VPX_IMG_FMT_NV12);
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+
+  // Prepare `raw_images_` from `mapped_buffer` and, if simulcast, scaled
+  // versions of `buffer`.
+  std::vector<rtc::scoped_refptr<VideoFrameBuffer>> prepared_buffers;
+  SetRawImagePlanes(&raw_images_[0], mapped_buffer.get());
+  prepared_buffers.push_back(mapped_buffer);
+  for (size_t i = 1; i < encoders_.size(); ++i) {
+    // Native buffers should implement optimized scaling and is the preferred
+    // buffer to scale. But if the buffer isn't native, it should be cheaper to
+    // scale from the previously prepared buffer which is smaller than `buffer`.
+    VideoFrameBuffer* buffer_to_scale =
+        buffer->type() == VideoFrameBuffer::Type::kNative
+            ? buffer.get()
+            : prepared_buffers.back().get();
+
+    auto scaled_buffer =
+        buffer_to_scale->Scale(raw_images_[i].d_w, raw_images_[i].d_h);
+    if (scaled_buffer->type() == VideoFrameBuffer::Type::kNative) {
+      auto mapped_scaled_buffer =
+          scaled_buffer->GetMappedFrameBuffer(mapped_type);
+      RTC_DCHECK(mapped_scaled_buffer) << "Unable to map the scaled buffer.";
+      if (!mapped_scaled_buffer) {
+        RTC_LOG(LS_ERROR) << "Failed to map scaled "
+                          << VideoFrameBufferTypeToString(scaled_buffer->type())
+                          << " image to "
+                          << VideoFrameBufferTypeToString(mapped_buffer->type())
+                          << ". Can't encode frame.";
+        return {};
+      }
+      scaled_buffer = mapped_scaled_buffer;
+    }
+    if (!IsCompatibleVideoFrameBufferType(scaled_buffer->type(),
+                                          mapped_buffer->type())) {
+      RTC_LOG(LS_ERROR) << "When scaling "
+                        << VideoFrameBufferTypeToString(buffer_to_scale->type())
+                        << ", the image was unexpectedly converted to "
+                        << VideoFrameBufferTypeToString(scaled_buffer->type())
+                        << " instead of "
+                        << VideoFrameBufferTypeToString(mapped_buffer->type())
+                        << ". Can't encode frame.";
+      RTC_DCHECK_NOTREACHED()
+          << "Scaled buffer type "
+          << VideoFrameBufferTypeToString(scaled_buffer->type())
+          << " is not compatible with mapped buffer type "
+          << VideoFrameBufferTypeToString(mapped_buffer->type());
+      return {};
+    }
+    SetRawImagePlanes(&raw_images_[i], scaled_buffer.get());
+    prepared_buffers.push_back(scaled_buffer);
+  }
+  return prepared_buffers;
+}
+
+// static
+LibvpxVp8Encoder::VariableFramerateExperiment
+LibvpxVp8Encoder::ParseVariableFramerateConfig(std::string group_name) {
+  FieldTrialFlag disabled = FieldTrialFlag("Disabled");
+  FieldTrialParameter<double> framerate_limit("min_fps", 5.0);
+  FieldTrialParameter<int> qp("min_qp", 15);
+  FieldTrialParameter<int> undershoot_percentage("undershoot", 30);
+  ParseFieldTrial({&disabled, &framerate_limit, &qp, &undershoot_percentage},
+                  field_trial::FindFullName(group_name));
+  VariableFramerateExperiment config;
+  config.enabled = !disabled.Get();
+  config.framerate_limit = framerate_limit.Get();
+  config.steady_state_qp = qp.Get();
+  config.steady_state_undershoot_percentage = undershoot_percentage.Get();
+
+  return config;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
new file mode 100644
index 0000000000..74477eac7e
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
@@ -0,0 +1,159 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_ENCODER_H_
+#define MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_ENCODER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/fec_controller_override.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/vp8_frame_buffer_controller.h"
+#include "api/video_codecs/vp8_frame_config.h"
+#include "modules/video_coding/codecs/interface/libvpx_interface.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/utility/framerate_controller_deprecated.h"
+#include "modules/video_coding/utility/vp8_constants.h"
+#include "rtc_base/experiments/cpu_speed_experiment.h"
+#include "rtc_base/experiments/encoder_info_settings.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+#include "vpx/vp8cx.h"
+#include "vpx/vpx_encoder.h"
+
+namespace webrtc {
+
+class LibvpxVp8Encoder : public VideoEncoder {
+ public:
+  LibvpxVp8Encoder(std::unique_ptr<LibvpxInterface> interface,
+                   VP8Encoder::Settings settings);
+  ~LibvpxVp8Encoder() override;
+
+  int Release() override;
+
+  void SetFecControllerOverride(
+      FecControllerOverride* fec_controller_override) override;
+
+  int InitEncode(const VideoCodec* codec_settings,
+                 const VideoEncoder::Settings& settings) override;
+
+  int Encode(const VideoFrame& input_image,
+             const std::vector<VideoFrameType>* frame_types) override;
+
+  int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
+
+  void SetRates(const RateControlParameters& parameters) override;
+
+  void OnPacketLossRateUpdate(float packet_loss_rate) override;
+
+  void OnRttUpdate(int64_t rtt_ms) override;
+
+  void OnLossNotification(const LossNotification& loss_notification) override;
+
+  EncoderInfo GetEncoderInfo() const override;
+
+  static vpx_enc_frame_flags_t EncodeFlags(const Vp8FrameConfig& references);
+
+ private:
+  // Get the cpu_speed setting for encoder based on resolution and/or platform.
+  int GetCpuSpeed(int width, int height);
+
+  // Determine number of encoder threads to use.
+  int NumberOfThreads(int width, int height, int number_of_cores);
+
+  // Call encoder initialize function and set control settings.
+  int InitAndSetControlSettings();
+
+  void PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
+                             const vpx_codec_cx_pkt& pkt,
+                             int stream_idx,
+                             int encoder_idx,
+                             uint32_t timestamp);
+
+  int GetEncodedPartitions(const VideoFrame& input_image,
+                           bool retransmission_allowed);
+
+  // Set the stream state for stream `stream_idx`.
+  void SetStreamState(bool send_stream, int stream_idx);
+
+  uint32_t MaxIntraTarget(uint32_t optimal_buffer_size);
+
+  uint32_t FrameDropThreshold(size_t spatial_idx) const;
+
+  size_t SteadyStateSize(int sid, int tid);
+
+  bool UpdateVpxConfiguration(size_t stream_index);
+
+  void MaybeUpdatePixelFormat(vpx_img_fmt fmt);
+  // Prepares `raw_image_` to reference image data of `buffer`, or of mapped or
+  // scaled versions of `buffer`. Returns a list of buffers that got referenced
+  // as a result, allowing the caller to keep references to them until after
+  // encoding has finished. On failure to convert the buffer, an empty list is
+  // returned.
+  std::vector<rtc::scoped_refptr<VideoFrameBuffer>> PrepareBuffers(
+      rtc::scoped_refptr<VideoFrameBuffer> buffer);
+
+  const std::unique_ptr<LibvpxInterface> libvpx_;
+
+  const CpuSpeedExperiment experimental_cpu_speed_config_arm_;
+  const RateControlSettings rate_control_settings_;
+
+  EncodedImageCallback* encoded_complete_callback_ = nullptr;
+  VideoCodec codec_;
+  bool inited_ = false;
+  int64_t timestamp_ = 0;
+  int qp_max_ = 56;
+  int cpu_speed_default_ = -6;
+  int number_of_cores_ = 0;
+  uint32_t rc_max_intra_target_ = 0;
+  int num_active_streams_ = 0;
+  const std::unique_ptr<Vp8FrameBufferControllerFactory>
+      frame_buffer_controller_factory_;
+  std::unique_ptr<Vp8FrameBufferController> frame_buffer_controller_;
+  const std::vector<VideoEncoder::ResolutionBitrateLimits>
+      resolution_bitrate_limits_;
+  std::vector<bool> key_frame_request_;
+  std::vector<bool> send_stream_;
+  std::vector<int> cpu_speed_;
+  std::vector<vpx_image_t> raw_images_;
+  std::vector<EncodedImage> encoded_images_;
+  std::vector<vpx_codec_ctx_t> encoders_;
+  std::vector<vpx_codec_enc_cfg_t> vpx_configs_;
+  std::vector<Vp8EncoderConfig> config_overrides_;
+  std::vector<vpx_rational_t> downsampling_factors_;
+
+  // Variable frame-rate screencast related fields and methods.
+  const struct VariableFramerateExperiment {
+    bool enabled = false;
+    // Framerate is limited to this value in steady state.
+    float framerate_limit = 5.0;
+    // This qp or below is considered a steady state.
+    int steady_state_qp = kVp8SteadyStateQpThreshold;
+    // Frames of at least this percentage below ideal for configured bitrate are
+    // considered in a steady state.
+    int steady_state_undershoot_percentage = 30;
+  } variable_framerate_experiment_;
+  static VariableFramerateExperiment ParseVariableFramerateConfig(
+      std::string group_name);
+  FramerateControllerDeprecated framerate_controller_;
+  int num_steady_state_frames_ = 0;
+
+  FecControllerOverride* fec_controller_override_ = nullptr;
+
+  const LibvpxVp8EncoderInfoSettings encoder_info_override_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_ENCODER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc
new file mode 100644
index 0000000000..4ca3de20d5
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "api/test/create_simulcast_test_fixture.h"
+#include "api/test/simulcast_test_fixture.h"
+#include "api/test/video/function_video_decoder_factory.h"
+#include "api/test/video/function_video_encoder_factory.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+std::unique_ptr<SimulcastTestFixture> CreateSpecificSimulcastTestFixture() {
+  std::unique_ptr<VideoEncoderFactory> encoder_factory =
+      std::make_unique<FunctionVideoEncoderFactory>(
+          []() { return VP8Encoder::Create(); });
+  std::unique_ptr<VideoDecoderFactory> decoder_factory =
+      std::make_unique<FunctionVideoDecoderFactory>(
+          []() { return VP8Decoder::Create(); });
+  return CreateSimulcastTestFixture(std::move(encoder_factory),
+                                    std::move(decoder_factory),
+                                    SdpVideoFormat("VP8"));
+}
+}  // namespace
+
+TEST(LibvpxVp8SimulcastTest, TestKeyFrameRequestsOnAllStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestKeyFrameRequestsOnAllStreams();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestKeyFrameRequestsOnSpecificStreams) {
+  GTEST_SKIP() << "Not applicable to VP8.";
+}
+
+TEST(LibvpxVp8SimulcastTest, TestPaddingAllStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingAllStreams();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestPaddingTwoStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingTwoStreams();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestPaddingTwoStreamsOneMaxedOut) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingTwoStreamsOneMaxedOut();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestPaddingOneStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingOneStream();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestPaddingOneStreamTwoMaxedOut) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingOneStreamTwoMaxedOut();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestSendAllStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSendAllStreams();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestDisablingStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestDisablingStreams();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestActiveStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestActiveStreams();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestSwitchingToOneStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSwitchingToOneStream();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestSwitchingToOneOddStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSwitchingToOneOddStream();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestSwitchingToOneSmallStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSwitchingToOneSmallStream();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestSpatioTemporalLayers333PatternEncoder) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSpatioTemporalLayers333PatternEncoder();
+}
+
+TEST(LibvpxVp8SimulcastTest, TestStrideEncodeDecode) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestStrideEncodeDecode();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
new file mode 100644
index 0000000000..71db0b22c2
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
@@ -0,0 +1,624 @@
+/* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
+
+#include <stdlib.h>
+
+#include <algorithm>
+#include <memory>
+
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+using BufferFlags = Vp8FrameConfig::BufferFlags;
+
+constexpr BufferFlags kNone = Vp8FrameConfig::BufferFlags::kNone;
+constexpr BufferFlags kReference = Vp8FrameConfig::BufferFlags::kReference;
+constexpr BufferFlags kUpdate = Vp8FrameConfig::BufferFlags::kUpdate;
+constexpr BufferFlags kReferenceAndUpdate =
+    Vp8FrameConfig::BufferFlags::kReferenceAndUpdate;
+
+constexpr int kOneSecond90Khz = 90000;
+constexpr int kMinTimeBetweenSyncs = kOneSecond90Khz * 2;
+constexpr int kMaxTimeBetweenSyncs = kOneSecond90Khz * 4;
+constexpr int kQpDeltaThresholdForSync = 8;
+constexpr int kMinBitrateKbpsForQpBoost = 500;
+constexpr auto kSwitch = DecodeTargetIndication::kSwitch;
+}  // namespace
+
+const double ScreenshareLayers::kMaxTL0FpsReduction = 2.5;
+const double ScreenshareLayers::kAcceptableTargetOvershoot = 2.0;
+
+constexpr int ScreenshareLayers::kMaxNumTemporalLayers;
+
+// Always emit a frame with certain interval, even if bitrate targets have
+// been exceeded. This prevents needless keyframe requests.
+const int ScreenshareLayers::kMaxFrameIntervalMs = 2750;
+
+ScreenshareLayers::ScreenshareLayers(int num_temporal_layers)
+    : number_of_temporal_layers_(
+          std::min(kMaxNumTemporalLayers, num_temporal_layers)),
+      active_layer_(-1),
+      last_timestamp_(-1),
+      last_sync_timestamp_(-1),
+      last_emitted_tl0_timestamp_(-1),
+      last_frame_time_ms_(-1),
+      max_debt_bytes_(0),
+      encode_framerate_(1000.0f, 1000.0f),  // 1 second window, second scale.
+      bitrate_updated_(false),
+      checker_(TemporalLayersChecker::CreateTemporalLayersChecker(
+          Vp8TemporalLayersType::kBitrateDynamic,
+          num_temporal_layers)) {
+  RTC_CHECK_GT(number_of_temporal_layers_, 0);
+  RTC_CHECK_LE(number_of_temporal_layers_, kMaxNumTemporalLayers);
+}
+
+ScreenshareLayers::~ScreenshareLayers() {
+  UpdateHistograms();
+}
+
+void ScreenshareLayers::SetQpLimits(size_t stream_index,
+                                    int min_qp,
+                                    int max_qp) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+  // 0 < min_qp <= max_qp
+  RTC_DCHECK_LT(0, min_qp);
+  RTC_DCHECK_LE(min_qp, max_qp);
+
+  RTC_DCHECK_EQ(min_qp_.has_value(), max_qp_.has_value());
+  if (!min_qp_.has_value()) {
+    min_qp_ = min_qp;
+    max_qp_ = max_qp;
+  } else {
+    RTC_DCHECK_EQ(min_qp, min_qp_.value());
+    RTC_DCHECK_EQ(max_qp, max_qp_.value());
+  }
+}
+
+size_t ScreenshareLayers::StreamCount() const {
+  return 1;
+}
+
+bool ScreenshareLayers::SupportsEncoderFrameDropping(
+    size_t stream_index) const {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+  // Frame dropping is handled internally by this class.
+  return false;
+}
+
+Vp8FrameConfig ScreenshareLayers::NextFrameConfig(size_t stream_index,
+                                                  uint32_t timestamp) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+
+  auto it = pending_frame_configs_.find(timestamp);
+  if (it != pending_frame_configs_.end()) {
+    // Drop and re-encode, reuse the previous config.
+    return it->second.frame_config;
+  }
+
+  if (number_of_temporal_layers_ <= 1) {
+    // No flags needed for 1 layer screenshare.
+    // TODO(pbos): Consider updating only last, and not all buffers.
+    DependencyInfo dependency_info{
+        "S", {kReferenceAndUpdate, kReferenceAndUpdate, kReferenceAndUpdate}};
+    pending_frame_configs_[timestamp] = dependency_info;
+    return dependency_info.frame_config;
+  }
+
+  const int64_t now_ms = rtc::TimeMillis();
+
+  int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
+  int64_t ts_diff;
+  if (last_timestamp_ == -1) {
+    ts_diff = kOneSecond90Khz / capture_framerate_.value_or(*target_framerate_);
+  } else {
+    ts_diff = unwrapped_timestamp - last_timestamp_;
+  }
+
+  if (target_framerate_) {
+    // If input frame rate exceeds target frame rate, either over a one second
+    // averaging window, or if frame interval is below 90% of desired value,
+    // drop frame.
+    if (encode_framerate_.Rate(now_ms).value_or(0) > *target_framerate_)
+      return Vp8FrameConfig(kNone, kNone, kNone);
+
+    // Primarily check if frame interval is too short using frame timestamps,
+    // as if they are correct they won't be affected by queuing in webrtc.
+    const int64_t expected_frame_interval_90khz =
+        kOneSecond90Khz / *target_framerate_;
+    if (last_timestamp_ != -1 && ts_diff > 0) {
+      if (ts_diff < 85 * expected_frame_interval_90khz / 100) {
+        return Vp8FrameConfig(kNone, kNone, kNone);
+      }
+    } else {
+      // Timestamps looks off, use realtime clock here instead.
+      const int64_t expected_frame_interval_ms = 1000 / *target_framerate_;
+      if (last_frame_time_ms_ != -1 &&
+          now_ms - last_frame_time_ms_ <
+              (85 * expected_frame_interval_ms) / 100) {
+        return Vp8FrameConfig(kNone, kNone, kNone);
+      }
+    }
+  }
+
+  if (stats_.first_frame_time_ms_ == -1)
+    stats_.first_frame_time_ms_ = now_ms;
+
+  // Make sure both frame droppers leak out bits.
+  layers_[0].UpdateDebt(ts_diff / 90);
+  layers_[1].UpdateDebt(ts_diff / 90);
+  last_timestamp_ = timestamp;
+  last_frame_time_ms_ = now_ms;
+
+  TemporalLayerState layer_state = TemporalLayerState::kDrop;
+
+  if (active_layer_ == -1 ||
+      layers_[active_layer_].state != TemporalLayer::State::kDropped) {
+    if (last_emitted_tl0_timestamp_ != -1 &&
+        (unwrapped_timestamp - last_emitted_tl0_timestamp_) / 90 >
+            kMaxFrameIntervalMs) {
+      // Too long time has passed since the last frame was emitted, cancel
+      // enough debt to allow a single frame.
+      layers_[0].debt_bytes_ = max_debt_bytes_ - 1;
+    }
+    if (layers_[0].debt_bytes_ > max_debt_bytes_) {
+      // Must drop TL0, encode TL1 instead.
+      if (layers_[1].debt_bytes_ > max_debt_bytes_) {
+        // Must drop both TL0 and TL1.
+        active_layer_ = -1;
+      } else {
+        active_layer_ = 1;
+      }
+    } else {
+      active_layer_ = 0;
+    }
+  }
+
+  switch (active_layer_) {
+    case 0:
+      layer_state = TemporalLayerState::kTl0;
+      last_emitted_tl0_timestamp_ = unwrapped_timestamp;
+      break;
+    case 1:
+      if (layers_[1].state != TemporalLayer::State::kDropped) {
+        if (TimeToSync(unwrapped_timestamp) ||
+            layers_[1].state == TemporalLayer::State::kKeyFrame) {
+          last_sync_timestamp_ = unwrapped_timestamp;
+          layer_state = TemporalLayerState::kTl1Sync;
+        } else {
+          layer_state = TemporalLayerState::kTl1;
+        }
+      } else {
+        layer_state = last_sync_timestamp_ == unwrapped_timestamp
+                          ? TemporalLayerState::kTl1Sync
+                          : TemporalLayerState::kTl1;
+      }
+      break;
+    case -1:
+      layer_state = TemporalLayerState::kDrop;
+      ++stats_.num_dropped_frames_;
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+
+  DependencyInfo dependency_info;
+  // TODO(pbos): Consider referencing but not updating the 'alt' buffer for all
+  // layers.
+  switch (layer_state) {
+    case TemporalLayerState::kDrop:
+      dependency_info = {"", {kNone, kNone, kNone}};
+      break;
+    case TemporalLayerState::kTl0:
+      // TL0 only references and updates 'last'.
+      dependency_info = {"SS", {kReferenceAndUpdate, kNone, kNone}};
+      dependency_info.frame_config.packetizer_temporal_idx = 0;
+      break;
+    case TemporalLayerState::kTl1:
+      // TL1 references both 'last' and 'golden' but only updates 'golden'.
+      dependency_info = {"-R", {kReference, kReferenceAndUpdate, kNone}};
+      dependency_info.frame_config.packetizer_temporal_idx = 1;
+      break;
+    case TemporalLayerState::kTl1Sync:
+      // Predict from only TL0 to allow participants to switch to the high
+      // bitrate stream. Updates 'golden' so that TL1 can continue to refer to
+      // and update 'golden' from this point on.
+      dependency_info = {"-S", {kReference, kUpdate, kNone}};
+      dependency_info.frame_config.packetizer_temporal_idx = 1;
+      dependency_info.frame_config.layer_sync = true;
+      break;
+  }
+
+  pending_frame_configs_[timestamp] = dependency_info;
+  return dependency_info.frame_config;
+}
+
+void ScreenshareLayers::OnRatesUpdated(
+    size_t stream_index,
+    const std::vector<uint32_t>& bitrates_bps,
+    int framerate_fps) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+  RTC_DCHECK_GT(framerate_fps, 0);
+  RTC_DCHECK_GE(bitrates_bps.size(), 1);
+  RTC_DCHECK_LE(bitrates_bps.size(), 2);
+
+  // `bitrates_bps` uses individual rates per layer, but we want to use the
+  // accumulated rate here.
+  uint32_t tl0_kbps = bitrates_bps[0] / 1000;
+  uint32_t tl1_kbps = tl0_kbps;
+  if (bitrates_bps.size() > 1) {
+    tl1_kbps += bitrates_bps[1] / 1000;
+  }
+
+  if (!target_framerate_) {
+    // First OnRatesUpdated() is called during construction, with the
+    // configured targets as parameters.
+    target_framerate_ = framerate_fps;
+    capture_framerate_ = target_framerate_;
+    bitrate_updated_ = true;
+  } else {
+    if ((capture_framerate_ &&
+         framerate_fps != static_cast<int>(*capture_framerate_)) ||
+        (tl0_kbps != layers_[0].target_rate_kbps_) ||
+        (tl1_kbps != layers_[1].target_rate_kbps_)) {
+      bitrate_updated_ = true;
+    }
+
+    if (framerate_fps < 0) {
+      capture_framerate_.reset();
+    } else {
+      capture_framerate_ = framerate_fps;
+    }
+  }
+
+  layers_[0].target_rate_kbps_ = tl0_kbps;
+  layers_[1].target_rate_kbps_ = tl1_kbps;
+}
+
+void ScreenshareLayers::OnEncodeDone(size_t stream_index,
+                                     uint32_t rtp_timestamp,
+                                     size_t size_bytes,
+                                     bool is_keyframe,
+                                     int qp,
+                                     CodecSpecificInfo* info) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+
+  if (size_bytes == 0) {
+    RTC_LOG(LS_WARNING) << "Empty frame; treating as dropped.";
+    OnFrameDropped(stream_index, rtp_timestamp);
+    return;
+  }
+
+  absl::optional<DependencyInfo> dependency_info;
+  auto it = pending_frame_configs_.find(rtp_timestamp);
+  if (it != pending_frame_configs_.end()) {
+    dependency_info = it->second;
+    pending_frame_configs_.erase(it);
+
+    if (checker_) {
+      RTC_DCHECK(checker_->CheckTemporalConfig(is_keyframe,
+                                               dependency_info->frame_config));
+    }
+  }
+
+  CodecSpecificInfoVP8& vp8_info = info->codecSpecific.VP8;
+  GenericFrameInfo& generic_frame_info = info->generic_frame_info.emplace();
+
+  if (number_of_temporal_layers_ == 1) {
+    vp8_info.temporalIdx = kNoTemporalIdx;
+    vp8_info.layerSync = false;
+    generic_frame_info.temporal_id = 0;
+    generic_frame_info.decode_target_indications = {kSwitch};
+    generic_frame_info.encoder_buffers.emplace_back(
+        0, /*referenced=*/!is_keyframe, /*updated=*/true);
+  } else {
+    int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(rtp_timestamp);
+    if (dependency_info) {
+      vp8_info.temporalIdx =
+          dependency_info->frame_config.packetizer_temporal_idx;
+      vp8_info.layerSync = dependency_info->frame_config.layer_sync;
+      generic_frame_info.temporal_id = vp8_info.temporalIdx;
+      generic_frame_info.decode_target_indications =
+          dependency_info->decode_target_indications;
+    } else {
+      RTC_DCHECK(is_keyframe);
+    }
+
+    if (is_keyframe) {
+      vp8_info.temporalIdx = 0;
+      last_sync_timestamp_ = unwrapped_timestamp;
+      vp8_info.layerSync = true;
+      layers_[0].state = TemporalLayer::State::kKeyFrame;
+      layers_[1].state = TemporalLayer::State::kKeyFrame;
+      active_layer_ = 1;
+      info->template_structure =
+          GetTemplateStructure(number_of_temporal_layers_);
+      generic_frame_info.temporal_id = vp8_info.temporalIdx;
+      generic_frame_info.decode_target_indications = {kSwitch, kSwitch};
+    } else if (active_layer_ >= 0 && layers_[active_layer_].state ==
+                                         TemporalLayer::State::kKeyFrame) {
+      layers_[active_layer_].state = TemporalLayer::State::kNormal;
+    }
+
+    vp8_info.useExplicitDependencies = true;
+    RTC_DCHECK_EQ(vp8_info.referencedBuffersCount, 0u);
+    RTC_DCHECK_EQ(vp8_info.updatedBuffersCount, 0u);
+
+    // Note that `frame_config` is not derefernced if `is_keyframe`,
+    // meaning it's never dereferenced if the optional may be unset.
+    for (int i = 0; i < static_cast<int>(Vp8FrameConfig::Buffer::kCount); ++i) {
+      bool references = false;
+      bool updates = is_keyframe;
+      if (!is_keyframe && dependency_info->frame_config.References(
+                              static_cast<Vp8FrameConfig::Buffer>(i))) {
+        RTC_DCHECK_LT(vp8_info.referencedBuffersCount,
+                      arraysize(CodecSpecificInfoVP8::referencedBuffers));
+        references = true;
+        vp8_info.referencedBuffers[vp8_info.referencedBuffersCount++] = i;
+      }
+
+      if (is_keyframe || dependency_info->frame_config.Updates(
+                             static_cast<Vp8FrameConfig::Buffer>(i))) {
+        RTC_DCHECK_LT(vp8_info.updatedBuffersCount,
+                      arraysize(CodecSpecificInfoVP8::updatedBuffers));
+        updates = true;
+        vp8_info.updatedBuffers[vp8_info.updatedBuffersCount++] = i;
+      }
+
+      if (references || updates)
+        generic_frame_info.encoder_buffers.emplace_back(i, references, updates);
+    }
+  }
+
+  encode_framerate_.Update(1, rtc::TimeMillis());
+
+  if (number_of_temporal_layers_ == 1)
+    return;
+
+  RTC_DCHECK_NE(-1, active_layer_);
+  if (layers_[active_layer_].state == TemporalLayer::State::kDropped) {
+    layers_[active_layer_].state = TemporalLayer::State::kQualityBoost;
+  }
+
+  if (qp != -1)
+    layers_[active_layer_].last_qp = qp;
+
+  if (active_layer_ == 0) {
+    layers_[0].debt_bytes_ += size_bytes;
+    layers_[1].debt_bytes_ += size_bytes;
+    ++stats_.num_tl0_frames_;
+    stats_.tl0_target_bitrate_sum_ += layers_[0].target_rate_kbps_;
+    stats_.tl0_qp_sum_ += qp;
+  } else if (active_layer_ == 1) {
+    layers_[1].debt_bytes_ += size_bytes;
+    ++stats_.num_tl1_frames_;
+    stats_.tl1_target_bitrate_sum_ += layers_[1].target_rate_kbps_;
+    stats_.tl1_qp_sum_ += qp;
+  }
+}
+
+void ScreenshareLayers::OnFrameDropped(size_t stream_index,
+                                       uint32_t rtp_timestamp) {
+  layers_[active_layer_].state = TemporalLayer::State::kDropped;
+  ++stats_.num_overshoots_;
+}
+
+void ScreenshareLayers::OnPacketLossRateUpdate(float packet_loss_rate) {}
+
+void ScreenshareLayers::OnRttUpdate(int64_t rtt_ms) {}
+
+void ScreenshareLayers::OnLossNotification(
+    const VideoEncoder::LossNotification& loss_notification) {}
+
+FrameDependencyStructure ScreenshareLayers::GetTemplateStructure(
+    int num_layers) const {
+  RTC_CHECK_LT(num_layers, 3);
+  RTC_CHECK_GT(num_layers, 0);
+
+  FrameDependencyStructure template_structure;
+  template_structure.num_decode_targets = num_layers;
+
+  switch (num_layers) {
+    case 1: {
+      template_structure.templates.resize(2);
+      template_structure.templates[0].T(0).Dtis("S");
+      template_structure.templates[1].T(0).Dtis("S").FrameDiffs({1});
+      return template_structure;
+    }
+    case 2: {
+      template_structure.templates.resize(3);
+      template_structure.templates[0].T(0).Dtis("SS");
+      template_structure.templates[1].T(0).Dtis("SS").FrameDiffs({1});
+      template_structure.templates[2].T(1).Dtis("-S").FrameDiffs({1});
+      return template_structure;
+    }
+    default:
+      RTC_DCHECK_NOTREACHED();
+      // To make the compiler happy!
+      return template_structure;
+  }
+}
+
+bool ScreenshareLayers::TimeToSync(int64_t timestamp) const {
+  RTC_DCHECK_EQ(1, active_layer_);
+  RTC_DCHECK_NE(-1, layers_[0].last_qp);
+  if (layers_[1].last_qp == -1) {
+    // First frame in TL1 should only depend on TL0 since there are no
+    // previous frames in TL1.
+    return true;
+  }
+
+  RTC_DCHECK_NE(-1, last_sync_timestamp_);
+  int64_t timestamp_diff = timestamp - last_sync_timestamp_;
+  if (timestamp_diff > kMaxTimeBetweenSyncs) {
+    // After a certain time, force a sync frame.
+    return true;
+  } else if (timestamp_diff < kMinTimeBetweenSyncs) {
+    // If too soon from previous sync frame, don't issue a new one.
+    return false;
+  }
+  // Issue a sync frame if difference in quality between TL0 and TL1 isn't too
+  // large.
+  if (layers_[0].last_qp - layers_[1].last_qp < kQpDeltaThresholdForSync)
+    return true;
+  return false;
+}
+
+uint32_t ScreenshareLayers::GetCodecTargetBitrateKbps() const {
+  uint32_t target_bitrate_kbps = layers_[0].target_rate_kbps_;
+
+  if (number_of_temporal_layers_ > 1) {
+    // Calculate a codec target bitrate. This may be higher than TL0, gaining
+    // quality at the expense of frame rate at TL0. Constraints:
+    // - TL0 frame rate no less than framerate / kMaxTL0FpsReduction.
+    // - Target rate * kAcceptableTargetOvershoot should not exceed TL1 rate.
+    target_bitrate_kbps =
+        std::min(layers_[0].target_rate_kbps_ * kMaxTL0FpsReduction,
+                 layers_[1].target_rate_kbps_ / kAcceptableTargetOvershoot);
+  }
+
+  return std::max(layers_[0].target_rate_kbps_, target_bitrate_kbps);
+}
+
+Vp8EncoderConfig ScreenshareLayers::UpdateConfiguration(size_t stream_index) {
+  RTC_DCHECK_LT(stream_index, StreamCount());
+  RTC_DCHECK(min_qp_.has_value());
+  RTC_DCHECK(max_qp_.has_value());
+
+  const uint32_t target_bitrate_kbps = GetCodecTargetBitrateKbps();
+
+  // TODO(sprang): We _really_ need to make an overhaul of this class. :(
+  // If we're dropping frames in order to meet a target framerate, adjust the
+  // bitrate assigned to the encoder so the total average bitrate is correct.
+  float encoder_config_bitrate_kbps = target_bitrate_kbps;
+  if (target_framerate_ && capture_framerate_ &&
+      *target_framerate_ < *capture_framerate_) {
+    encoder_config_bitrate_kbps *=
+        static_cast<float>(*capture_framerate_) / *target_framerate_;
+  }
+
+  if (bitrate_updated_ ||
+      encoder_config_.rc_target_bitrate !=
+          absl::make_optional(encoder_config_bitrate_kbps)) {
+    encoder_config_.rc_target_bitrate = encoder_config_bitrate_kbps;
+
+    // Don't reconfigure qp limits during quality boost frames.
+    if (active_layer_ == -1 ||
+        layers_[active_layer_].state != TemporalLayer::State::kQualityBoost) {
+      const int min_qp = min_qp_.value();
+      const int max_qp = max_qp_.value();
+
+      // After a dropped frame, a frame with max qp will be encoded and the
+      // quality will then ramp up from there. To boost the speed of recovery,
+      // encode the next frame with lower max qp, if there is sufficient
+      // bandwidth to do so without causing excessive delay.
+      // TL0 is the most important to improve since the errors in this layer
+      // will propagate to TL1.
+      // Currently, reduce max qp by 20% for TL0 and 15% for TL1.
+      if (layers_[1].target_rate_kbps_ >= kMinBitrateKbpsForQpBoost) {
+        layers_[0].enhanced_max_qp = min_qp + (((max_qp - min_qp) * 80) / 100);
+        layers_[1].enhanced_max_qp = min_qp + (((max_qp - min_qp) * 85) / 100);
+      } else {
+        layers_[0].enhanced_max_qp = -1;
+        layers_[1].enhanced_max_qp = -1;
+      }
+    }
+
+    if (capture_framerate_) {
+      int avg_frame_size =
+          (target_bitrate_kbps * 1000) / (8 * *capture_framerate_);
+      // Allow max debt to be the size of a single optimal frame.
+      // TODO(sprang): Determine if this needs to be adjusted by some factor.
+      // (Lower values may cause more frame drops, higher may lead to queuing
+      // delays.)
+      max_debt_bytes_ = avg_frame_size;
+    }
+
+    bitrate_updated_ = false;
+  }
+
+  // Don't try to update boosts state if not active yet.
+  if (active_layer_ == -1)
+    return encoder_config_;
+
+  if (number_of_temporal_layers_ <= 1)
+    return encoder_config_;
+
+  // If layer is in the quality boost state (following a dropped frame), update
+  // the configuration with the adjusted (lower) qp and set the state back to
+  // normal.
+  unsigned int adjusted_max_qp = max_qp_.value();  // Set the normal max qp.
+  if (layers_[active_layer_].state == TemporalLayer::State::kQualityBoost) {
+    if (layers_[active_layer_].enhanced_max_qp != -1) {
+      // Bitrate is high enough for quality boost, update max qp.
+      adjusted_max_qp = layers_[active_layer_].enhanced_max_qp;
+    }
+    // Regardless of qp, reset the boost state for the next frame.
+    layers_[active_layer_].state = TemporalLayer::State::kNormal;
+  }
+  encoder_config_.rc_max_quantizer = adjusted_max_qp;
+
+  return encoder_config_;
+}
+
+void ScreenshareLayers::TemporalLayer::UpdateDebt(int64_t delta_ms) {
+  uint32_t debt_reduction_bytes = target_rate_kbps_ * delta_ms / 8;
+  if (debt_reduction_bytes >= debt_bytes_) {
+    debt_bytes_ = 0;
+  } else {
+    debt_bytes_ -= debt_reduction_bytes;
+  }
+}
+
+void ScreenshareLayers::UpdateHistograms() {
+  if (stats_.first_frame_time_ms_ == -1)
+    return;
+  int64_t duration_sec =
+      (rtc::TimeMillis() - stats_.first_frame_time_ms_ + 500) / 1000;
+  if (duration_sec >= metrics::kMinRunTimeInSeconds) {
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.Screenshare.Layer0.FrameRate",
+        (stats_.num_tl0_frames_ + (duration_sec / 2)) / duration_sec);
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.Screenshare.Layer1.FrameRate",
+        (stats_.num_tl1_frames_ + (duration_sec / 2)) / duration_sec);
+    int total_frames = stats_.num_tl0_frames_ + stats_.num_tl1_frames_;
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.Screenshare.FramesPerDrop",
+        (stats_.num_dropped_frames_ == 0
+             ? 0
+             : total_frames / stats_.num_dropped_frames_));
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.Screenshare.FramesPerOvershoot",
+        (stats_.num_overshoots_ == 0 ? 0
+                                     : total_frames / stats_.num_overshoots_));
+    if (stats_.num_tl0_frames_ > 0) {
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.Screenshare.Layer0.Qp",
+                                 stats_.tl0_qp_sum_ / stats_.num_tl0_frames_);
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.Screenshare.Layer0.TargetBitrate",
+          stats_.tl0_target_bitrate_sum_ / stats_.num_tl0_frames_);
+    }
+    if (stats_.num_tl1_frames_ > 0) {
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.Screenshare.Layer1.Qp",
+                                 stats_.tl1_qp_sum_ / stats_.num_tl1_frames_);
+      RTC_HISTOGRAM_COUNTS_10000(
+          "WebRTC.Video.Screenshare.Layer1.TargetBitrate",
+          stats_.tl1_target_bitrate_sum_ / stats_.num_tl1_frames_);
+    }
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers.h b/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
new file mode 100644
index 0000000000..47d6b401f4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
@@ -0,0 +1,164 @@
+/* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
+#define MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
+
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/video_codecs/vp8_frame_config.h"
+#include "api/video_codecs/vp8_temporal_layers.h"
+#include "modules/video_coding/codecs/vp8/include/temporal_layers_checker.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/utility/frame_dropper.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/rate_statistics.h"
+
+namespace webrtc {
+
+struct CodecSpecificInfoVP8;
+class Clock;
+
+class ScreenshareLayers final : public Vp8FrameBufferController {
+ public:
+  static const double kMaxTL0FpsReduction;
+  static const double kAcceptableTargetOvershoot;
+  static const int kMaxFrameIntervalMs;
+
+  explicit ScreenshareLayers(int num_temporal_layers);
+  ~ScreenshareLayers() override;
+
+  void SetQpLimits(size_t stream_index, int min_qp, int max_qp) override;
+
+  size_t StreamCount() const override;
+
+  bool SupportsEncoderFrameDropping(size_t stream_index) const override;
+
+  // Returns the recommended VP8 encode flags needed. May refresh the decoder
+  // and/or update the reference buffers.
+  Vp8FrameConfig NextFrameConfig(size_t stream_index,
+                                 uint32_t rtp_timestamp) override;
+
+  // New target bitrate, per temporal layer.
+  void OnRatesUpdated(size_t stream_index,
+                      const std::vector<uint32_t>& bitrates_bps,
+                      int framerate_fps) override;
+
+  Vp8EncoderConfig UpdateConfiguration(size_t stream_index) override;
+
+  void OnEncodeDone(size_t stream_index,
+                    uint32_t rtp_timestamp,
+                    size_t size_bytes,
+                    bool is_keyframe,
+                    int qp,
+                    CodecSpecificInfo* info) override;
+
+  void OnFrameDropped(size_t stream_index, uint32_t rtp_timestamp) override;
+
+  void OnPacketLossRateUpdate(float packet_loss_rate) override;
+
+  void OnRttUpdate(int64_t rtt_ms) override;
+
+  void OnLossNotification(
+      const VideoEncoder::LossNotification& loss_notification) override;
+
+ private:
+  enum class TemporalLayerState : int { kDrop, kTl0, kTl1, kTl1Sync };
+
+  struct DependencyInfo {
+    DependencyInfo() = default;
+    DependencyInfo(absl::string_view indication_symbols,
+                   Vp8FrameConfig frame_config)
+        : decode_target_indications(
+              webrtc_impl::StringToDecodeTargetIndications(indication_symbols)),
+          frame_config(frame_config) {}
+
+    absl::InlinedVector<DecodeTargetIndication, 10> decode_target_indications;
+    Vp8FrameConfig frame_config;
+  };
+
+  bool TimeToSync(int64_t timestamp) const;
+  uint32_t GetCodecTargetBitrateKbps() const;
+
+  const int number_of_temporal_layers_;
+
+  // TODO(eladalon/sprang): These should be made into const-int set in the ctor.
+  absl::optional<int> min_qp_;
+  absl::optional<int> max_qp_;
+
+  int active_layer_;
+  int64_t last_timestamp_;
+  int64_t last_sync_timestamp_;
+  int64_t last_emitted_tl0_timestamp_;
+  int64_t last_frame_time_ms_;
+  RtpTimestampUnwrapper time_wrap_handler_;
+  uint32_t max_debt_bytes_;
+
+  std::map<uint32_t, DependencyInfo> pending_frame_configs_;
+
+  // Configured max framerate.
+  absl::optional<uint32_t> target_framerate_;
+  // Incoming framerate from capturer.
+  absl::optional<uint32_t> capture_framerate_;
+
+  // Tracks what framerate we actually encode, and drops frames on overshoot.
+  RateStatistics encode_framerate_;
+  bool bitrate_updated_;
+
+  static constexpr int kMaxNumTemporalLayers = 2;
+  struct TemporalLayer {
+    TemporalLayer()
+        : state(State::kNormal),
+          enhanced_max_qp(-1),
+          last_qp(-1),
+          debt_bytes_(0),
+          target_rate_kbps_(0) {}
+
+    enum class State {
+      kNormal,
+      kDropped,
+      kReencoded,
+      kQualityBoost,
+      kKeyFrame
+    } state;
+
+    int enhanced_max_qp;
+    int last_qp;
+    uint32_t debt_bytes_;
+    uint32_t target_rate_kbps_;
+
+    void UpdateDebt(int64_t delta_ms);
+  } layers_[kMaxNumTemporalLayers];
+
+  void UpdateHistograms();
+  FrameDependencyStructure GetTemplateStructure(int num_layers) const;
+
+  // Data for histogram statistics.
+  struct Stats {
+    int64_t first_frame_time_ms_ = -1;
+    int64_t num_tl0_frames_ = 0;
+    int64_t num_tl1_frames_ = 0;
+    int64_t num_dropped_frames_ = 0;
+    int64_t num_overshoots_ = 0;
+    int64_t tl0_qp_sum_ = 0;
+    int64_t tl1_qp_sum_ = 0;
+    int64_t tl0_target_bitrate_sum_ = 0;
+    int64_t tl1_target_bitrate_sum_ = 0;
+  } stats_;
+
+  Vp8EncoderConfig encoder_config_;
+
+  // Optional utility used to verify reference validity.
+  std::unique_ptr<TemporalLayersChecker> checker_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
new file mode 100644
index 0000000000..e5b3bd4fdf
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
@@ -0,0 +1,788 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include <cstdint>
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/vp8_frame_config.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fake_clock.h"
+#include "system_wrappers/include/metrics.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "vpx/vp8cx.h"
+
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::NiceMock;
+
+namespace webrtc {
+namespace {
+// 5 frames per second at 90 kHz.
+const uint32_t kTimestampDelta5Fps = 90000 / 5;
+const int kDefaultQp = 54;
+const int kDefaultTl0BitrateKbps = 200;
+const int kDefaultTl1BitrateKbps = 2000;
+const int kFrameRate = 5;
+const int kSyncPeriodSeconds = 2;
+const int kMaxSyncPeriodSeconds = 4;
+
+// Expected flags for corresponding temporal layers.
+const int kTl0Flags = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
+                      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF;
+const int kTl1Flags =
+    VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST;
+const int kTl1SyncFlags = VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF |
+                          VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST;
+const std::vector<uint32_t> kDefault2TlBitratesBps = {
+    kDefaultTl0BitrateKbps * 1000,
+    (kDefaultTl1BitrateKbps - kDefaultTl0BitrateKbps) * 1000};
+
+}  // namespace
+
+class ScreenshareLayerTest : public ::testing::Test {
+ protected:
+  ScreenshareLayerTest()
+      : min_qp_(2),
+        max_qp_(kDefaultQp),
+        frame_size_(-1),
+        timestamp_(90),
+        config_updated_(false) {}
+  virtual ~ScreenshareLayerTest() {}
+
+  void SetUp() override {
+    layers_.reset(new ScreenshareLayers(2));
+    cfg_ = ConfigureBitrates();
+  }
+
+  int EncodeFrame(bool base_sync, CodecSpecificInfo* info = nullptr) {
+    CodecSpecificInfo ignored_info;
+    if (!info) {
+      info = &ignored_info;
+    }
+
+    int flags = ConfigureFrame(base_sync);
+    if (flags != -1)
+      layers_->OnEncodeDone(0, timestamp_, frame_size_, base_sync, kDefaultQp,
+                            info);
+    return flags;
+  }
+
+  int ConfigureFrame(bool key_frame) {
+    tl_config_ = NextFrameConfig(0, timestamp_);
+    EXPECT_EQ(0, tl_config_.encoder_layer_id)
+        << "ScreenshareLayers always encodes using the bitrate allocator for "
+           "layer 0, but may reference different buffers and packetize "
+           "differently.";
+    if (tl_config_.drop_frame) {
+      return -1;
+    }
+    const uint32_t prev_rc_target_bitrate = cfg_.rc_target_bitrate.value_or(-1);
+    const uint32_t prev_rc_max_quantizer = cfg_.rc_max_quantizer.value_or(-1);
+
+    cfg_ = layers_->UpdateConfiguration(0);
+
+    config_updated_ =
+        cfg_.temporal_layer_config.has_value() ||
+        (cfg_.rc_target_bitrate.has_value() &&
+         cfg_.rc_target_bitrate.value() != prev_rc_target_bitrate) ||
+        (cfg_.rc_max_quantizer.has_value() &&
+         cfg_.rc_max_quantizer.value() != prev_rc_max_quantizer) ||
+        cfg_.g_error_resilient.has_value();
+
+    int flags = LibvpxVp8Encoder::EncodeFlags(tl_config_);
+    EXPECT_NE(-1, frame_size_);
+    return flags;
+  }
+
+  Vp8FrameConfig NextFrameConfig(size_t stream_index, uint32_t timestamp) {
+    int64_t timestamp_ms = timestamp / 90;
+    clock_.AdvanceTime(TimeDelta::Millis(timestamp_ms - rtc::TimeMillis()));
+    return layers_->NextFrameConfig(stream_index, timestamp);
+  }
+
+  int FrameSizeForBitrate(int bitrate_kbps) {
+    return ((bitrate_kbps * 1000) / 8) / kFrameRate;
+  }
+
+  Vp8EncoderConfig ConfigureBitrates() {
+    layers_->SetQpLimits(0, min_qp_, max_qp_);
+    layers_->OnRatesUpdated(0, kDefault2TlBitratesBps, kFrameRate);
+    const Vp8EncoderConfig vp8_cfg = layers_->UpdateConfiguration(0);
+    EXPECT_TRUE(vp8_cfg.rc_target_bitrate.has_value());
+    frame_size_ = FrameSizeForBitrate(vp8_cfg.rc_target_bitrate.value());
+    return vp8_cfg;
+  }
+
+  void WithQpLimits(int min_qp, int max_qp) {
+    min_qp_ = min_qp;
+    max_qp_ = max_qp;
+  }
+
+  // Runs a few initial frames and makes sure we have seen frames on both
+  // temporal layers, including sync and non-sync frames.
+  bool RunGracePeriod() {
+    bool got_tl0 = false;
+    bool got_tl1 = false;
+    bool got_tl1_sync = false;
+    for (int i = 0; i < 10; ++i) {
+      CodecSpecificInfo info;
+      EXPECT_NE(-1, EncodeFrame(false, &info));
+      timestamp_ += kTimestampDelta5Fps;
+      if (info.codecSpecific.VP8.temporalIdx == 0) {
+        got_tl0 = true;
+      } else if (info.codecSpecific.VP8.layerSync) {
+        got_tl1_sync = true;
+      } else {
+        got_tl1 = true;
+      }
+      if (got_tl0 && got_tl1 && got_tl1_sync)
+        return true;
+    }
+    return false;
+  }
+
+  // Adds frames until we get one in the specified temporal layer. The last
+  // FrameEncoded() call will be omitted and needs to be done by the caller.
+  // Returns the flags for the last frame.
+  int SkipUntilTl(int layer) {
+    return SkipUntilTlAndSync(layer, absl::nullopt);
+  }
+
+  // Same as SkipUntilTl, but also waits until the sync bit condition is met.
+  int SkipUntilTlAndSync(int layer, absl::optional<bool> sync) {
+    int flags = 0;
+    const int kMaxFramesToSkip =
+        1 + (sync.value_or(false) ? kMaxSyncPeriodSeconds : 1) * kFrameRate;
+    for (int i = 0; i < kMaxFramesToSkip; ++i) {
+      flags = ConfigureFrame(false);
+      if (tl_config_.packetizer_temporal_idx != layer ||
+          (sync && *sync != tl_config_.layer_sync)) {
+        if (flags != -1) {
+          // If flags do not request a frame drop, report some default values
+          // for frame size etc.
+          CodecSpecificInfo info;
+          layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp,
+                                &info);
+        }
+        timestamp_ += kTimestampDelta5Fps;
+      } else {
+        // Found frame from sought after layer.
+        return flags;
+      }
+    }
+    ADD_FAILURE() << "Did not get a frame of TL" << layer << " in time.";
+    return -1;
+  }
+
+  int min_qp_;
+  uint32_t max_qp_;
+  int frame_size_;
+  rtc::ScopedFakeClock clock_;
+  std::unique_ptr<ScreenshareLayers> layers_;
+
+  uint32_t timestamp_;
+  Vp8FrameConfig tl_config_;
+  Vp8EncoderConfig cfg_;
+  bool config_updated_;
+
+  CodecSpecificInfo* IgnoredCodecSpecificInfo() {
+    ignored_codec_specific_info_ = std::make_unique<CodecSpecificInfo>();
+    return ignored_codec_specific_info_.get();
+  }
+
+ private:
+  std::unique_ptr<CodecSpecificInfo> ignored_codec_specific_info_;
+};
+
+TEST_F(ScreenshareLayerTest, 1Layer) {
+  layers_.reset(new ScreenshareLayers(1));
+  ConfigureBitrates();
+  // One layer screenshare should not use the frame dropper as all frames will
+  // belong to the base layer.
+  const int kSingleLayerFlags = 0;
+  auto info = std::make_unique<CodecSpecificInfo>();
+  int flags = EncodeFrame(/*base_sync=*/false, info.get());
+  timestamp_ += kTimestampDelta5Fps;
+  EXPECT_EQ(static_cast<uint8_t>(kNoTemporalIdx),
+            info->codecSpecific.VP8.temporalIdx);
+  EXPECT_FALSE(info->codecSpecific.VP8.layerSync);
+  EXPECT_EQ(info->generic_frame_info->temporal_id, 0);
+
+  info = std::make_unique<CodecSpecificInfo>();
+  flags = EncodeFrame(/*base_sync=*/false, info.get());
+  EXPECT_EQ(kSingleLayerFlags, flags);
+  EXPECT_EQ(static_cast<uint8_t>(kNoTemporalIdx),
+            info->codecSpecific.VP8.temporalIdx);
+  EXPECT_FALSE(info->codecSpecific.VP8.layerSync);
+  EXPECT_EQ(info->generic_frame_info->temporal_id, 0);
+}
+
+TEST_F(ScreenshareLayerTest, 2LayersPeriodicSync) {
+  std::vector<int> sync_times;
+  const int kNumFrames = kSyncPeriodSeconds * kFrameRate * 2 - 1;
+  for (int i = 0; i < kNumFrames; ++i) {
+    CodecSpecificInfo info;
+    EncodeFrame(false, &info);
+    timestamp_ += kTimestampDelta5Fps;
+    if (info.codecSpecific.VP8.temporalIdx == 1 &&
+        info.codecSpecific.VP8.layerSync) {
+      sync_times.push_back(timestamp_);
+    }
+  }
+
+  ASSERT_EQ(2u, sync_times.size());
+  EXPECT_GE(sync_times[1] - sync_times[0], 90000 * kSyncPeriodSeconds);
+}
+
+TEST_F(ScreenshareLayerTest, 2LayersSyncAfterTimeout) {
+  std::vector<int> sync_times;
+  const int kNumFrames = kMaxSyncPeriodSeconds * kFrameRate * 2 - 1;
+  for (int i = 0; i < kNumFrames; ++i) {
+    CodecSpecificInfo info;
+
+    tl_config_ = NextFrameConfig(0, timestamp_);
+    cfg_ = layers_->UpdateConfiguration(0);
+
+    // Simulate TL1 being at least 8 qp steps better.
+    if (tl_config_.packetizer_temporal_idx == 0) {
+      layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp,
+                            &info);
+    } else {
+      layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp - 8,
+                            &info);
+    }
+
+    if (info.codecSpecific.VP8.temporalIdx == 1 &&
+        info.codecSpecific.VP8.layerSync)
+      sync_times.push_back(timestamp_);
+
+    timestamp_ += kTimestampDelta5Fps;
+  }
+
+  ASSERT_EQ(2u, sync_times.size());
+  EXPECT_GE(sync_times[1] - sync_times[0], 90000 * kMaxSyncPeriodSeconds);
+}
+
+TEST_F(ScreenshareLayerTest, 2LayersSyncAfterSimilarQP) {
+  std::vector<int> sync_times;
+
+  const int kNumFrames = (kSyncPeriodSeconds +
+                          ((kMaxSyncPeriodSeconds - kSyncPeriodSeconds) / 2)) *
+                         kFrameRate;
+  for (int i = 0; i < kNumFrames; ++i) {
+    CodecSpecificInfo info;
+
+    ConfigureFrame(false);
+
+    // Simulate TL1 being at least 8 qp steps better.
+    if (tl_config_.packetizer_temporal_idx == 0) {
+      layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp,
+                            &info);
+    } else {
+      layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp - 8,
+                            &info);
+    }
+
+    if (info.codecSpecific.VP8.temporalIdx == 1 &&
+        info.codecSpecific.VP8.layerSync)
+      sync_times.push_back(timestamp_);
+
+    timestamp_ += kTimestampDelta5Fps;
+  }
+
+  ASSERT_EQ(1u, sync_times.size());
+
+  bool bumped_tl0_quality = false;
+  for (int i = 0; i < 3; ++i) {
+    CodecSpecificInfo info;
+
+    int flags = ConfigureFrame(false);
+    layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp - 8,
+                          &info);
+    if (info.codecSpecific.VP8.temporalIdx == 0) {
+      // Bump TL0 to same quality as TL1.
+      bumped_tl0_quality = true;
+    } else {
+      if (bumped_tl0_quality) {
+        EXPECT_TRUE(info.codecSpecific.VP8.layerSync);
+        EXPECT_EQ(kTl1SyncFlags, flags);
+        return;
+      }
+    }
+    timestamp_ += kTimestampDelta5Fps;
+  }
+  ADD_FAILURE() << "No TL1 frame arrived within time limit.";
+}
+
+TEST_F(ScreenshareLayerTest, 2LayersToggling) {
+  EXPECT_TRUE(RunGracePeriod());
+
+  // Insert 50 frames. 2/5 should be TL0.
+  int tl0_frames = 0;
+  int tl1_frames = 0;
+  for (int i = 0; i < 50; ++i) {
+    CodecSpecificInfo info;
+    EncodeFrame(/*base_sync=*/false, &info);
+    EXPECT_EQ(info.codecSpecific.VP8.temporalIdx,
+              info.generic_frame_info->temporal_id);
+    timestamp_ += kTimestampDelta5Fps;
+    switch (info.codecSpecific.VP8.temporalIdx) {
+      case 0:
+        ++tl0_frames;
+        break;
+      case 1:
+        ++tl1_frames;
+        break;
+      default:
+        abort();
+    }
+  }
+  EXPECT_EQ(20, tl0_frames);
+  EXPECT_EQ(30, tl1_frames);
+}
+
+TEST_F(ScreenshareLayerTest, AllFitsLayer0) {
+  frame_size_ = FrameSizeForBitrate(kDefaultTl0BitrateKbps);
+
+  // Insert 50 frames, small enough that all fits in TL0.
+  for (int i = 0; i < 50; ++i) {
+    CodecSpecificInfo info;
+    int flags = EncodeFrame(false, &info);
+    timestamp_ += kTimestampDelta5Fps;
+    EXPECT_EQ(kTl0Flags, flags);
+    EXPECT_EQ(0, info.codecSpecific.VP8.temporalIdx);
+  }
+}
+
+TEST_F(ScreenshareLayerTest, TooHighBitrate) {
+  frame_size_ = 2 * FrameSizeForBitrate(kDefaultTl1BitrateKbps);
+
+  // Insert 100 frames. Half should be dropped.
+  int tl0_frames = 0;
+  int tl1_frames = 0;
+  int dropped_frames = 0;
+  for (int i = 0; i < 100; ++i) {
+    CodecSpecificInfo info;
+    int flags = EncodeFrame(false, &info);
+    timestamp_ += kTimestampDelta5Fps;
+    if (flags == -1) {
+      ++dropped_frames;
+    } else {
+      switch (info.codecSpecific.VP8.temporalIdx) {
+        case 0:
+          ++tl0_frames;
+          break;
+        case 1:
+          ++tl1_frames;
+          break;
+        default:
+          ADD_FAILURE() << "Unexpected temporal id";
+      }
+    }
+  }
+
+  EXPECT_NEAR(50, tl0_frames + tl1_frames, 1);
+  EXPECT_NEAR(50, dropped_frames, 1);
+}
+
+TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL0) {
+  const int kTl0_kbps = 100;
+  const int kTl1_kbps = 1000;
+  const std::vector<uint32_t> layer_rates = {kTl0_kbps * 1000,
+                                             (kTl1_kbps - kTl0_kbps) * 1000};
+  layers_->OnRatesUpdated(0, layer_rates, kFrameRate);
+  cfg_ = layers_->UpdateConfiguration(0);
+
+  EXPECT_EQ(static_cast<unsigned int>(
+                ScreenshareLayers::kMaxTL0FpsReduction * kTl0_kbps + 0.5),
+            cfg_.rc_target_bitrate);
+}
+
+TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL1) {
+  const int kTl0_kbps = 100;
+  const int kTl1_kbps = 450;
+  const std::vector<uint32_t> layer_rates = {kTl0_kbps * 1000,
+                                             (kTl1_kbps - kTl0_kbps) * 1000};
+  layers_->OnRatesUpdated(0, layer_rates, kFrameRate);
+  cfg_ = layers_->UpdateConfiguration(0);
+
+  EXPECT_EQ(static_cast<unsigned int>(
+                kTl1_kbps / ScreenshareLayers::kAcceptableTargetOvershoot),
+            cfg_.rc_target_bitrate);
+}
+
+TEST_F(ScreenshareLayerTest, TargetBitrateBelowTL0) {
+  const int kTl0_kbps = 100;
+  const std::vector<uint32_t> layer_rates = {kTl0_kbps * 1000};
+  layers_->OnRatesUpdated(0, layer_rates, kFrameRate);
+  cfg_ = layers_->UpdateConfiguration(0);
+
+  EXPECT_EQ(static_cast<uint32_t>(kTl0_kbps), cfg_.rc_target_bitrate);
+}
+
+TEST_F(ScreenshareLayerTest, EncoderDrop) {
+  EXPECT_TRUE(RunGracePeriod());
+  SkipUntilTl(0);
+
+  // Size 0 indicates dropped frame.
+  layers_->OnEncodeDone(0, timestamp_, 0, false, 0, IgnoredCodecSpecificInfo());
+
+  // Re-encode frame (so don't advance timestamp).
+  int flags = EncodeFrame(false);
+  timestamp_ += kTimestampDelta5Fps;
+  EXPECT_FALSE(config_updated_);
+  EXPECT_EQ(kTl0Flags, flags);
+
+  // Next frame should have boosted quality...
+  SkipUntilTl(0);
+  EXPECT_TRUE(config_updated_);
+  EXPECT_LT(cfg_.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
+  layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp,
+                        IgnoredCodecSpecificInfo());
+  timestamp_ += kTimestampDelta5Fps;
+
+  // ...then back to standard setup.
+  SkipUntilTl(0);
+  layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp,
+                        IgnoredCodecSpecificInfo());
+  timestamp_ += kTimestampDelta5Fps;
+  EXPECT_EQ(cfg_.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
+
+  // Next drop in TL1.
+  SkipUntilTl(1);
+  layers_->OnEncodeDone(0, timestamp_, 0, false, 0, IgnoredCodecSpecificInfo());
+
+  // Re-encode frame (so don't advance timestamp).
+  flags = EncodeFrame(false);
+  timestamp_ += kTimestampDelta5Fps;
+  EXPECT_FALSE(config_updated_);
+  EXPECT_EQ(kTl1Flags, flags);
+
+  // Next frame should have boosted QP.
+  SkipUntilTl(1);
+  EXPECT_TRUE(config_updated_);
+  EXPECT_LT(cfg_.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
+  layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp,
+                        IgnoredCodecSpecificInfo());
+  timestamp_ += kTimestampDelta5Fps;
+
+  // ...and back to normal.
+  SkipUntilTl(1);
+  EXPECT_EQ(cfg_.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
+  layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp,
+                        IgnoredCodecSpecificInfo());
+  timestamp_ += kTimestampDelta5Fps;
+}
+
+TEST_F(ScreenshareLayerTest, RespectsMaxIntervalBetweenFrames) {
+  const int kLowBitrateKbps = 50;
+  const int kLargeFrameSizeBytes = 100000;
+  const uint32_t kStartTimestamp = 1234;
+
+  const std::vector<uint32_t> layer_rates = {kLowBitrateKbps * 1000};
+  layers_->OnRatesUpdated(0, layer_rates, kFrameRate);
+  cfg_ = layers_->UpdateConfiguration(0);
+
+  EXPECT_EQ(kTl0Flags,
+            LibvpxVp8Encoder::EncodeFlags(NextFrameConfig(0, kStartTimestamp)));
+  layers_->OnEncodeDone(0, kStartTimestamp, kLargeFrameSizeBytes, false,
+                        kDefaultQp, IgnoredCodecSpecificInfo());
+
+  const uint32_t kTwoSecondsLater =
+      kStartTimestamp + (ScreenshareLayers::kMaxFrameIntervalMs * 90);
+
+  // Sanity check, repayment time should exceed kMaxFrameIntervalMs.
+  ASSERT_GT(kStartTimestamp + 90 * (kLargeFrameSizeBytes * 8) / kLowBitrateKbps,
+            kStartTimestamp + (ScreenshareLayers::kMaxFrameIntervalMs * 90));
+
+  // Expect drop one frame interval before the two second timeout. If we try
+  // any later, the frame will be dropped anyway by the frame rate throttling
+  // logic.
+  EXPECT_TRUE(
+      NextFrameConfig(0, kTwoSecondsLater - kTimestampDelta5Fps).drop_frame);
+
+  // More than two seconds has passed since last frame, one should be emitted
+  // even if bitrate target is then exceeded.
+  EXPECT_EQ(kTl0Flags, LibvpxVp8Encoder::EncodeFlags(
+                           NextFrameConfig(0, kTwoSecondsLater + 90)));
+}
+
+TEST_F(ScreenshareLayerTest, UpdatesHistograms) {
+  metrics::Reset();
+  bool trigger_drop = false;
+  bool dropped_frame = false;
+  bool overshoot = false;
+  const int kTl0Qp = 35;
+  const int kTl1Qp = 30;
+  for (int64_t timestamp = 0;
+       timestamp < kTimestampDelta5Fps * 5 * metrics::kMinRunTimeInSeconds;
+       timestamp += kTimestampDelta5Fps) {
+    tl_config_ = NextFrameConfig(0, timestamp);
+    if (tl_config_.drop_frame) {
+      dropped_frame = true;
+      continue;
+    }
+    int flags = LibvpxVp8Encoder::EncodeFlags(tl_config_);
+    if (flags != -1)
+      cfg_ = layers_->UpdateConfiguration(0);
+
+    if (timestamp >= kTimestampDelta5Fps * 5 && !overshoot && flags != -1) {
+      // Simulate one overshoot.
+      layers_->OnEncodeDone(0, timestamp, 0, false, 0, nullptr);
+      overshoot = true;
+    }
+
+    if (flags == kTl0Flags) {
+      if (timestamp >= kTimestampDelta5Fps * 20 && !trigger_drop) {
+        // Simulate a too large frame, to cause frame drop.
+        layers_->OnEncodeDone(0, timestamp, frame_size_ * 10, false, kTl0Qp,
+                              IgnoredCodecSpecificInfo());
+        trigger_drop = true;
+      } else {
+        layers_->OnEncodeDone(0, timestamp, frame_size_, false, kTl0Qp,
+                              IgnoredCodecSpecificInfo());
+      }
+    } else if (flags == kTl1Flags || flags == kTl1SyncFlags) {
+      layers_->OnEncodeDone(0, timestamp, frame_size_, false, kTl1Qp,
+                            IgnoredCodecSpecificInfo());
+    } else if (flags == -1) {
+      dropped_frame = true;
+    } else {
+      RTC_DCHECK_NOTREACHED() << "Unexpected flags";
+    }
+    clock_.AdvanceTime(TimeDelta::Millis(1000 / 5));
+  }
+
+  EXPECT_TRUE(overshoot);
+  EXPECT_TRUE(dropped_frame);
+
+  layers_.reset();  // Histograms are reported on destruction.
+
+  EXPECT_METRIC_EQ(
+      1, metrics::NumSamples("WebRTC.Video.Screenshare.Layer0.FrameRate"));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumSamples("WebRTC.Video.Screenshare.Layer1.FrameRate"));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumSamples("WebRTC.Video.Screenshare.FramesPerDrop"));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumSamples("WebRTC.Video.Screenshare.FramesPerOvershoot"));
+  EXPECT_METRIC_EQ(1,
+                   metrics::NumSamples("WebRTC.Video.Screenshare.Layer0.Qp"));
+  EXPECT_METRIC_EQ(1,
+                   metrics::NumSamples("WebRTC.Video.Screenshare.Layer1.Qp"));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumSamples("WebRTC.Video.Screenshare.Layer0.TargetBitrate"));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumSamples("WebRTC.Video.Screenshare.Layer1.TargetBitrate"));
+
+  EXPECT_METRIC_GT(
+      metrics::MinSample("WebRTC.Video.Screenshare.Layer0.FrameRate"), 1);
+  EXPECT_METRIC_GT(
+      metrics::MinSample("WebRTC.Video.Screenshare.Layer1.FrameRate"), 1);
+  EXPECT_METRIC_GT(metrics::MinSample("WebRTC.Video.Screenshare.FramesPerDrop"),
+                   1);
+  EXPECT_METRIC_GT(
+      metrics::MinSample("WebRTC.Video.Screenshare.FramesPerOvershoot"), 1);
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.Video.Screenshare.Layer0.Qp", kTl0Qp));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.Video.Screenshare.Layer1.Qp", kTl1Qp));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.Video.Screenshare.Layer0.TargetBitrate",
+                            kDefaultTl0BitrateKbps));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.Video.Screenshare.Layer1.TargetBitrate",
+                            kDefaultTl1BitrateKbps));
+}
+
+TEST_F(ScreenshareLayerTest, RespectsConfiguredFramerate) {
+  int64_t kTestSpanMs = 2000;
+  int64_t kFrameIntervalsMs = 1000 / kFrameRate;
+
+  uint32_t timestamp = 1234;
+  int num_input_frames = 0;
+  int num_discarded_frames = 0;
+
+  // Send at regular rate - no drops expected.
+  for (int64_t i = 0; i < kTestSpanMs; i += kFrameIntervalsMs) {
+    if (NextFrameConfig(0, timestamp).drop_frame) {
+      ++num_discarded_frames;
+    } else {
+      size_t frame_size_bytes = kDefaultTl0BitrateKbps * kFrameIntervalsMs / 8;
+      layers_->OnEncodeDone(0, timestamp, frame_size_bytes, false, kDefaultQp,
+                            IgnoredCodecSpecificInfo());
+    }
+    timestamp += kFrameIntervalsMs * 90;
+    clock_.AdvanceTime(TimeDelta::Millis(kFrameIntervalsMs));
+
+    ++num_input_frames;
+  }
+  EXPECT_EQ(0, num_discarded_frames);
+
+  // Send at twice the configured rate - drop every other frame.
+  num_input_frames = 0;
+  num_discarded_frames = 0;
+  for (int64_t i = 0; i < kTestSpanMs; i += kFrameIntervalsMs / 2) {
+    if (NextFrameConfig(0, timestamp).drop_frame) {
+      ++num_discarded_frames;
+    } else {
+      size_t frame_size_bytes = kDefaultTl0BitrateKbps * kFrameIntervalsMs / 8;
+      layers_->OnEncodeDone(0, timestamp, frame_size_bytes, false, kDefaultQp,
+                            IgnoredCodecSpecificInfo());
+    }
+    timestamp += kFrameIntervalsMs * 90 / 2;
+    clock_.AdvanceTime(TimeDelta::Millis(kFrameIntervalsMs));
+    ++num_input_frames;
+  }
+
+  // Allow for some rounding errors in the measurements.
+  EXPECT_NEAR(num_discarded_frames, num_input_frames / 2, 2);
+}
+
+TEST_F(ScreenshareLayerTest, 2LayersSyncAtOvershootDrop) {
+  // Run grace period so we have existing frames in both TL0 and Tl1.
+  EXPECT_TRUE(RunGracePeriod());
+
+  // Move ahead until we have a sync frame in TL1.
+  EXPECT_EQ(kTl1SyncFlags, SkipUntilTlAndSync(1, true));
+  ASSERT_TRUE(tl_config_.layer_sync);
+
+  // Simulate overshoot of this frame.
+  layers_->OnEncodeDone(0, timestamp_, 0, false, 0, nullptr);
+
+  cfg_ = layers_->UpdateConfiguration(0);
+  EXPECT_EQ(kTl1SyncFlags, LibvpxVp8Encoder::EncodeFlags(tl_config_));
+
+  CodecSpecificInfo new_info;
+  layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp,
+                        &new_info);
+  EXPECT_TRUE(new_info.codecSpecific.VP8.layerSync);
+}
+
+TEST_F(ScreenshareLayerTest, DropOnTooShortFrameInterval) {
+  // Run grace period so we have existing frames in both TL0 and Tl1.
+  EXPECT_TRUE(RunGracePeriod());
+
+  // Add a large gap, so there's plenty of room in the rate tracker.
+  timestamp_ += kTimestampDelta5Fps * 3;
+  EXPECT_FALSE(NextFrameConfig(0, timestamp_).drop_frame);
+  layers_->OnEncodeDone(0, timestamp_, frame_size_, false, kDefaultQp,
+                        IgnoredCodecSpecificInfo());
+
+  // Frame interval below 90% if desired time is not allowed, try inserting
+  // frame just before this limit.
+  const int64_t kMinFrameInterval = (kTimestampDelta5Fps * 85) / 100;
+  timestamp_ += kMinFrameInterval - 90;
+  EXPECT_TRUE(NextFrameConfig(0, timestamp_).drop_frame);
+
+  // Try again at the limit, now it should pass.
+  timestamp_ += 90;
+  EXPECT_FALSE(NextFrameConfig(0, timestamp_).drop_frame);
+}
+
+TEST_F(ScreenshareLayerTest, AdjustsBitrateWhenDroppingFrames) {
+  const uint32_t kTimestampDelta10Fps = kTimestampDelta5Fps / 2;
+  const int kNumFrames = 30;
+  ASSERT_TRUE(cfg_.rc_target_bitrate.has_value());
+  const uint32_t default_bitrate = cfg_.rc_target_bitrate.value();
+  layers_->OnRatesUpdated(0, kDefault2TlBitratesBps, 10);
+
+  int num_dropped_frames = 0;
+  for (int i = 0; i < kNumFrames; ++i) {
+    if (EncodeFrame(false) == -1)
+      ++num_dropped_frames;
+    timestamp_ += kTimestampDelta10Fps;
+  }
+  cfg_ = layers_->UpdateConfiguration(0);
+
+  EXPECT_EQ(num_dropped_frames, kNumFrames / 2);
+  EXPECT_EQ(cfg_.rc_target_bitrate, default_bitrate * 2);
+}
+
+TEST_F(ScreenshareLayerTest, UpdatesConfigurationAfterRateChange) {
+  // Set inital rate again, no need to update configuration.
+  layers_->OnRatesUpdated(0, kDefault2TlBitratesBps, kFrameRate);
+  cfg_ = layers_->UpdateConfiguration(0);
+
+  // Rate changed, now update config.
+  std::vector<uint32_t> bitrates = kDefault2TlBitratesBps;
+  bitrates[1] -= 100000;
+  layers_->OnRatesUpdated(0, bitrates, 5);
+  cfg_ = layers_->UpdateConfiguration(0);
+
+  // Changed rate, but then set changed rate again before trying to update
+  // configuration, update should still apply.
+  bitrates[1] -= 100000;
+  layers_->OnRatesUpdated(0, bitrates, 5);
+  layers_->OnRatesUpdated(0, bitrates, 5);
+  cfg_ = layers_->UpdateConfiguration(0);
+}
+
+TEST_F(ScreenshareLayerTest, MaxQpRestoredAfterDoubleDrop) {
+  // Run grace period so we have existing frames in both TL0 and Tl1.
+  EXPECT_TRUE(RunGracePeriod());
+
+  // Move ahead until we have a sync frame in TL1.
+  EXPECT_EQ(kTl1SyncFlags, SkipUntilTlAndSync(1, true));
+  ASSERT_TRUE(tl_config_.layer_sync);
+
+  // Simulate overshoot of this frame.
+  layers_->OnEncodeDone(0, timestamp_, 0, false, -1, nullptr);
+
+  // Simulate re-encoded frame.
+  layers_->OnEncodeDone(0, timestamp_, 1, false, max_qp_,
+                        IgnoredCodecSpecificInfo());
+
+  // Next frame, expect boosted quality.
+  // Slightly alter bitrate between each frame.
+  std::vector<uint32_t> kDefault2TlBitratesBpsAlt = kDefault2TlBitratesBps;
+  kDefault2TlBitratesBpsAlt[1] += 4000;
+  layers_->OnRatesUpdated(0, kDefault2TlBitratesBpsAlt, kFrameRate);
+  EXPECT_EQ(kTl1Flags, SkipUntilTlAndSync(1, false));
+  EXPECT_TRUE(config_updated_);
+  EXPECT_LT(cfg_.rc_max_quantizer, max_qp_);
+  ASSERT_TRUE(cfg_.rc_max_quantizer.has_value());
+  const uint32_t adjusted_qp = cfg_.rc_max_quantizer.value();
+
+  // Simulate overshoot of this frame.
+  layers_->OnEncodeDone(0, timestamp_, 0, false, -1, nullptr);
+
+  // Simulate re-encoded frame.
+  layers_->OnEncodeDone(0, timestamp_, frame_size_, false, max_qp_,
+                        IgnoredCodecSpecificInfo());
+
+  // A third frame, expect boosted quality.
+  layers_->OnRatesUpdated(0, kDefault2TlBitratesBps, kFrameRate);
+  EXPECT_EQ(kTl1Flags, SkipUntilTlAndSync(1, false));
+  EXPECT_TRUE(config_updated_);
+  EXPECT_LT(cfg_.rc_max_quantizer, max_qp_);
+  EXPECT_EQ(adjusted_qp, cfg_.rc_max_quantizer);
+
+  // Frame encoded.
+  layers_->OnEncodeDone(0, timestamp_, frame_size_, false, max_qp_,
+                        IgnoredCodecSpecificInfo());
+
+  // A fourth frame, max qp should be restored.
+  layers_->OnRatesUpdated(0, kDefault2TlBitratesBpsAlt, kFrameRate);
+  EXPECT_EQ(kTl1Flags, SkipUntilTlAndSync(1, false));
+  EXPECT_EQ(cfg_.rc_max_quantizer, max_qp_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers.h b/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers.h
new file mode 100644
index 0000000000..9576fb27be
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
+#define MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
+
+// TODO(webrtc:9012) Remove this file when downstream projects have updated.
+#include "api/video_codecs/vp8_temporal_layers.h"
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
new file mode 100644
index 0000000000..5aebd2c526
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
@@ -0,0 +1,146 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp8/include/temporal_layers_checker.h"
+
+#include <memory>
+
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+std::unique_ptr<TemporalLayersChecker>
+TemporalLayersChecker::CreateTemporalLayersChecker(Vp8TemporalLayersType type,
+                                                   int num_temporal_layers) {
+  switch (type) {
+    case Vp8TemporalLayersType::kFixedPattern:
+      return std::make_unique<DefaultTemporalLayersChecker>(
+          num_temporal_layers);
+    case Vp8TemporalLayersType::kBitrateDynamic:
+      // Conference mode temporal layering for screen content in base stream.
+      return std::make_unique<TemporalLayersChecker>(num_temporal_layers);
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+TemporalLayersChecker::TemporalLayersChecker(int num_temporal_layers)
+    : num_temporal_layers_(num_temporal_layers),
+      sequence_number_(0),
+      last_sync_sequence_number_(0),
+      last_tl0_sequence_number_(0) {}
+
+bool TemporalLayersChecker::CheckAndUpdateBufferState(
+    BufferState* state,
+    bool* need_sync,
+    bool frame_is_keyframe,
+    uint8_t temporal_layer,
+    Vp8FrameConfig::BufferFlags flags,
+    uint32_t sequence_number,
+    uint32_t* lowest_sequence_referenced) {
+  if (flags & Vp8FrameConfig::BufferFlags::kReference) {
+    if (state->temporal_layer > 0 && !state->is_keyframe) {
+      *need_sync = false;
+    }
+    if (!state->is_keyframe && !frame_is_keyframe &&
+        state->sequence_number < *lowest_sequence_referenced) {
+      *lowest_sequence_referenced = state->sequence_number;
+    }
+    if (!frame_is_keyframe && !state->is_keyframe &&
+        state->temporal_layer > temporal_layer) {
+      RTC_LOG(LS_ERROR) << "Frame is referencing higher temporal layer.";
+      return false;
+    }
+  }
+  if ((flags & Vp8FrameConfig::BufferFlags::kUpdate)) {
+    state->temporal_layer = temporal_layer;
+    state->sequence_number = sequence_number;
+    state->is_keyframe = frame_is_keyframe;
+  }
+  if (frame_is_keyframe)
+    state->is_keyframe = true;
+  return true;
+}
+
+bool TemporalLayersChecker::CheckTemporalConfig(
+    bool frame_is_keyframe,
+    const Vp8FrameConfig& frame_config) {
+  if (frame_config.drop_frame ||
+      frame_config.packetizer_temporal_idx == kNoTemporalIdx) {
+    return true;
+  }
+  ++sequence_number_;
+  if (frame_config.packetizer_temporal_idx >= num_temporal_layers_ ||
+      (frame_config.packetizer_temporal_idx == kNoTemporalIdx &&
+       num_temporal_layers_ > 1)) {
+    RTC_LOG(LS_ERROR) << "Incorrect temporal layer set for frame: "
+                      << frame_config.packetizer_temporal_idx
+                      << " num_temporal_layers: " << num_temporal_layers_;
+    return false;
+  }
+
+  uint32_t lowest_sequence_referenced = sequence_number_;
+  bool need_sync = frame_config.packetizer_temporal_idx > 0 &&
+                   frame_config.packetizer_temporal_idx != kNoTemporalIdx;
+
+  if (!CheckAndUpdateBufferState(
+          &last_, &need_sync, frame_is_keyframe,
+          frame_config.packetizer_temporal_idx, frame_config.last_buffer_flags,
+          sequence_number_, &lowest_sequence_referenced)) {
+    RTC_LOG(LS_ERROR) << "Error in the Last buffer";
+    return false;
+  }
+  if (!CheckAndUpdateBufferState(&golden_, &need_sync, frame_is_keyframe,
+                                 frame_config.packetizer_temporal_idx,
+                                 frame_config.golden_buffer_flags,
+                                 sequence_number_,
+                                 &lowest_sequence_referenced)) {
+    RTC_LOG(LS_ERROR) << "Error in the Golden buffer";
+    return false;
+  }
+  if (!CheckAndUpdateBufferState(
+          &arf_, &need_sync, frame_is_keyframe,
+          frame_config.packetizer_temporal_idx, frame_config.arf_buffer_flags,
+          sequence_number_, &lowest_sequence_referenced)) {
+    RTC_LOG(LS_ERROR) << "Error in the Arf buffer";
+    return false;
+  }
+
+  if (lowest_sequence_referenced < last_sync_sequence_number_ &&
+      !frame_is_keyframe) {
+    RTC_LOG(LS_ERROR) << "Reference past the last sync frame. Referenced "
+                      << lowest_sequence_referenced << ", but sync was at "
+                      << last_sync_sequence_number_;
+    return false;
+  }
+
+  if (frame_config.packetizer_temporal_idx == 0) {
+    last_tl0_sequence_number_ = sequence_number_;
+  }
+
+  if (frame_is_keyframe) {
+    last_sync_sequence_number_ = sequence_number_;
+  }
+
+  if (need_sync) {
+    last_sync_sequence_number_ = last_tl0_sequence_number_;
+  }
+
+  // Ignore sync flag on key-frames as it really doesn't matter.
+  if (need_sync != frame_config.layer_sync && !frame_is_keyframe) {
+    RTC_LOG(LS_ERROR) << "Sync bit is set incorrectly on a frame. Expected: "
+                      << need_sync << " Actual: " << frame_config.layer_sync;
+    return false;
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
new file mode 100644
index 0000000000..c5a8b659c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -0,0 +1,913 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "api/test/create_frame_generator.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/test/mock_video_decoder.h"
+#include "api/test/mock_video_encoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/vp8_temporal_layers.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "common_video/test/utilities.h"
+#include "modules/video_coding/codecs/interface/mock_libvpx_interface.h"
+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
+#include "modules/video_coding/utility/vp8_header_parser.h"
+#include "rtc_base/time_utils.h"
+#include "test/field_trial.h"
+#include "test/mappable_native_buffer.h"
+#include "test/video_codec_settings.h"
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::AllOf;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::Field;
+using ::testing::Invoke;
+using ::testing::NiceMock;
+using ::testing::Return;
+using EncoderInfo = webrtc::VideoEncoder::EncoderInfo;
+using FramerateFractions =
+    absl::InlinedVector<uint8_t, webrtc::kMaxTemporalStreams>;
+
+namespace {
+constexpr uint32_t kLegacyScreenshareTl0BitrateKbps = 200;
+constexpr uint32_t kLegacyScreenshareTl1BitrateKbps = 1000;
+constexpr uint32_t kInitialTimestampRtp = 123;
+constexpr int64_t kTestNtpTimeMs = 456;
+constexpr int64_t kInitialTimestampMs = 789;
+constexpr int kNumCores = 1;
+constexpr size_t kMaxPayloadSize = 1440;
+constexpr int kWidth = 172;
+constexpr int kHeight = 144;
+constexpr float kFramerateFps = 30;
+
+const VideoEncoder::Capabilities kCapabilities(false);
+const VideoEncoder::Settings kSettings(kCapabilities,
+                                       kNumCores,
+                                       kMaxPayloadSize);
+}  // namespace
+
+class TestVp8Impl : public VideoCodecUnitTest {
+ protected:
+  std::unique_ptr<VideoEncoder> CreateEncoder() override {
+    return VP8Encoder::Create();
+  }
+
+  std::unique_ptr<VideoDecoder> CreateDecoder() override {
+    return VP8Decoder::Create();
+  }
+
+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
+    webrtc::test::CodecSettings(kVideoCodecVP8, codec_settings);
+    codec_settings->width = kWidth;
+    codec_settings->height = kHeight;
+    codec_settings->SetVideoEncoderComplexity(
+        VideoCodecComplexity::kComplexityNormal);
+  }
+
+  void EncodeAndWaitForFrame(const VideoFrame& input_frame,
+                             EncodedImage* encoded_frame,
+                             CodecSpecificInfo* codec_specific_info,
+                             bool keyframe = false) {
+    std::vector<VideoFrameType> frame_types;
+    if (keyframe) {
+      frame_types.emplace_back(VideoFrameType::kVideoFrameKey);
+    } else {
+      frame_types.emplace_back(VideoFrameType::kVideoFrameDelta);
+    }
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(input_frame, &frame_types));
+    ASSERT_TRUE(WaitForEncodedFrame(encoded_frame, codec_specific_info));
+    VerifyQpParser(*encoded_frame);
+    EXPECT_EQ(kVideoCodecVP8, codec_specific_info->codecType);
+    EXPECT_EQ(0, encoded_frame->SpatialIndex());
+  }
+
+  void EncodeAndExpectFrameWith(const VideoFrame& input_frame,
+                                uint8_t temporal_idx,
+                                bool keyframe = false) {
+    EncodedImage encoded_frame;
+    CodecSpecificInfo codec_specific_info;
+    EncodeAndWaitForFrame(input_frame, &encoded_frame, &codec_specific_info,
+                          keyframe);
+    EXPECT_EQ(temporal_idx, codec_specific_info.codecSpecific.VP8.temporalIdx);
+  }
+
+  void VerifyQpParser(const EncodedImage& encoded_frame) const {
+    int qp;
+    EXPECT_GT(encoded_frame.size(), 0u);
+    ASSERT_TRUE(vp8::GetQp(encoded_frame.data(), encoded_frame.size(), &qp));
+    EXPECT_EQ(encoded_frame.qp_, qp) << "Encoder QP != parsed bitstream QP.";
+  }
+};
+
+TEST_F(TestVp8Impl, ErrorResilienceDisabledForNoTemporalLayers) {
+  codec_settings_.simulcastStream[0].numberOfTemporalLayers = 1;
+
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           VP8Encoder::Settings());
+  EXPECT_CALL(*vpx,
+              codec_enc_init(
+                  _, _, Field(&vpx_codec_enc_cfg_t::g_error_resilient, 0), _));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings_, kSettings));
+}
+
+TEST_F(TestVp8Impl, DefaultErrorResilienceEnabledForTemporalLayers) {
+  codec_settings_.simulcastStream[0].numberOfTemporalLayers = 2;
+  codec_settings_.VP8()->numberOfTemporalLayers = 2;
+
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           VP8Encoder::Settings());
+  EXPECT_CALL(*vpx,
+              codec_enc_init(_, _,
+                             Field(&vpx_codec_enc_cfg_t::g_error_resilient,
+                                   VPX_ERROR_RESILIENT_DEFAULT),
+                             _));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings_, kSettings));
+}
+
+TEST_F(TestVp8Impl,
+       PartitionErrorResilienceEnabledForTemporalLayersWithFieldTrial) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VP8-ForcePartitionResilience/Enabled/");
+  codec_settings_.simulcastStream[0].numberOfTemporalLayers = 2;
+  codec_settings_.VP8()->numberOfTemporalLayers = 2;
+
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           VP8Encoder::Settings());
+  EXPECT_CALL(*vpx,
+              codec_enc_init(_, _,
+                             Field(&vpx_codec_enc_cfg_t::g_error_resilient,
+                                   VPX_ERROR_RESILIENT_PARTITIONS),
+                             _));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings_, kSettings));
+}
+
+TEST_F(TestVp8Impl, SetRates) {
+  codec_settings_.SetFrameDropEnabled(true);
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           VP8Encoder::Settings());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings_,
+                               VideoEncoder::Settings(kCapabilities, 1, 1000)));
+
+  const uint32_t kBitrateBps = 300000;
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(0, 0, kBitrateBps);
+  EXPECT_CALL(
+      *vpx,
+      codec_enc_config_set(
+          _, AllOf(Field(&vpx_codec_enc_cfg_t::rc_target_bitrate,
+                         kBitrateBps / 1000),
+                   Field(&vpx_codec_enc_cfg_t::rc_undershoot_pct, 100u),
+                   Field(&vpx_codec_enc_cfg_t::rc_overshoot_pct, 15u),
+                   Field(&vpx_codec_enc_cfg_t::rc_buf_sz, 1000u),
+                   Field(&vpx_codec_enc_cfg_t::rc_buf_optimal_sz, 600u),
+                   Field(&vpx_codec_enc_cfg_t::rc_dropframe_thresh, 30u))))
+      .WillOnce(Return(VPX_CODEC_OK));
+  encoder.SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, static_cast<double>(codec_settings_.maxFramerate)));
+}
+
+TEST_F(TestVp8Impl, EncodeFrameAndRelease) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_UNINITIALIZED,
+            encoder_->Encode(NextInputFrame(), nullptr));
+}
+
+TEST_F(TestVp8Impl, EncodeNv12FrameSimulcast) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  input_frame_generator_ = test::CreateSquareFrameGenerator(
+      kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kNV12,
+      absl::nullopt);
+  EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_UNINITIALIZED,
+            encoder_->Encode(NextInputFrame(), nullptr));
+}
+
+TEST_F(TestVp8Impl, EncodeI420FrameAfterNv12Frame) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  input_frame_generator_ = test::CreateSquareFrameGenerator(
+      kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kNV12,
+      absl::nullopt);
+  EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info);
+  input_frame_generator_ = test::CreateSquareFrameGenerator(
+      kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kI420,
+      absl::nullopt);
+  EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_UNINITIALIZED,
+            encoder_->Encode(NextInputFrame(), nullptr));
+}
+
+TEST_F(TestVp8Impl, Configure) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
+  EXPECT_TRUE(decoder_->Configure({}));
+}
+
+TEST_F(TestVp8Impl, OnEncodedImageReportsInfo) {
+  VideoFrame input_frame = NextInputFrame();
+  input_frame.set_timestamp(kInitialTimestampRtp);
+  input_frame.set_timestamp_us(kInitialTimestampMs *
+                               rtc::kNumMicrosecsPerMillisec);
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  EncodeAndWaitForFrame(input_frame, &encoded_frame, &codec_specific_info);
+
+  EXPECT_EQ(kInitialTimestampRtp, encoded_frame.Timestamp());
+  EXPECT_EQ(kWidth, static_cast<int>(encoded_frame._encodedWidth));
+  EXPECT_EQ(kHeight, static_cast<int>(encoded_frame._encodedHeight));
+}
+
+TEST_F(TestVp8Impl,
+       EncoderFillsResolutionInCodecAgnosticSectionOfCodecSpecificInfo) {
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info);
+
+  ASSERT_TRUE(codec_specific_info.template_structure);
+  EXPECT_THAT(codec_specific_info.template_structure->resolutions,
+              ElementsAre(RenderResolution(kWidth, kHeight)));
+}
+
+TEST_F(TestVp8Impl, DecodedQpEqualsEncodedQp) {
+  VideoFrame input_frame = NextInputFrame();
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  EncodeAndWaitForFrame(input_frame, &encoded_frame, &codec_specific_info);
+
+  // First frame should be a key frame.
+  encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, -1));
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+  ASSERT_TRUE(decoded_qp);
+  EXPECT_GT(I420PSNR(&input_frame, decoded_frame.get()), 36);
+  EXPECT_EQ(encoded_frame.qp_, *decoded_qp);
+}
+
+TEST_F(TestVp8Impl, ChecksSimulcastSettings) {
+  codec_settings_.numberOfSimulcastStreams = 2;
+  // Resolutions are not in ascending order, temporal layers do not match.
+  codec_settings_.simulcastStream[0] = {.width = kWidth,
+                                        .height = kHeight,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 2,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[1] = {.width = kWidth / 2,
+                                        .height = kHeight / 2,
+                                        .maxFramerate = 30,
+                                        .numberOfTemporalLayers = 3,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+  codec_settings_.numberOfSimulcastStreams = 3;
+  // Resolutions are not in ascending order.
+  codec_settings_.simulcastStream[0] = {.width = kWidth / 2,
+                                        .height = kHeight / 2,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[1] = {.width = kWidth / 2 - 1,
+                                        .height = kHeight / 2 - 1,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[2] = {.width = kWidth,
+                                        .height = kHeight,
+                                        .maxFramerate = 30,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+  // Resolutions are not in ascending order.
+  codec_settings_.simulcastStream[0] = {.width = kWidth,
+                                        .height = kHeight,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[1] = {.width = kWidth,
+                                        .height = kHeight,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[2] = {.width = kWidth - 1,
+                                        .height = kHeight - 1,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+  // Temporal layers do not match.
+  codec_settings_.simulcastStream[0] = {.width = kWidth / 4,
+                                        .height = kHeight / 4,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[1] = {.width = kWidth / 2,
+                                        .height = kHeight / 2,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 2,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[2] = {.width = kWidth,
+                                        .height = kHeight,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 3,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+  // Resolutions do not match codec config.
+  codec_settings_.simulcastStream[0] = {.width = kWidth / 4 + 1,
+                                        .height = kHeight / 4 + 1,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[1] = {.width = kWidth / 2 + 2,
+                                        .height = kHeight / 2 + 2,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[2] = {.width = kWidth + 4,
+                                        .height = kHeight + 4,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+  // Everything fine: scaling by 2, top resolution matches video, temporal
+  // settings are the same for all layers.
+  codec_settings_.simulcastStream[0] = {.width = kWidth / 4,
+                                        .height = kHeight / 4,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[1] = {.width = kWidth / 2,
+                                        .height = kHeight / 2,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[2] = {.width = kWidth,
+                                        .height = kHeight,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+  // Everything fine: custom scaling, top resolution matches video, temporal
+  // settings are the same for all layers.
+  codec_settings_.simulcastStream[0] = {.width = kWidth / 4,
+                                        .height = kHeight / 4,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[1] = {.width = kWidth,
+                                        .height = kHeight,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  codec_settings_.simulcastStream[2] = {.width = kWidth,
+                                        .height = kHeight,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80};
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+}
+
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_AlignedStrideEncodeDecode DISABLED_AlignedStrideEncodeDecode
+#else
+#define MAYBE_AlignedStrideEncodeDecode AlignedStrideEncodeDecode
+#endif
+TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
+  VideoFrame input_frame = NextInputFrame();
+  input_frame.set_timestamp(kInitialTimestampRtp);
+  input_frame.set_timestamp_us(kInitialTimestampMs *
+                               rtc::kNumMicrosecsPerMillisec);
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  EncodeAndWaitForFrame(input_frame, &encoded_frame, &codec_specific_info);
+
+  // First frame should be a key frame.
+  encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
+  encoded_frame.ntp_time_ms_ = kTestNtpTimeMs;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, -1));
+
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+  // Compute PSNR on all planes (faster than SSIM).
+  EXPECT_GT(I420PSNR(&input_frame, decoded_frame.get()), 36);
+  EXPECT_EQ(kInitialTimestampRtp, decoded_frame->timestamp());
+}
+
+TEST_F(TestVp8Impl, EncoderWith2TemporalLayers) {
+  codec_settings_.VP8()->numberOfTemporalLayers = 2;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // Temporal layer 0.
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info);
+
+  EXPECT_EQ(0, codec_specific_info.codecSpecific.VP8.temporalIdx);
+  // Temporal layer 1.
+  EncodeAndExpectFrameWith(NextInputFrame(), 1);
+  // Temporal layer 0.
+  EncodeAndExpectFrameWith(NextInputFrame(), 0);
+  // Temporal layer 1.
+  EncodeAndExpectFrameWith(NextInputFrame(), 1);
+}
+
+TEST_F(TestVp8Impl, ScalingDisabledIfAutomaticResizeOff) {
+  codec_settings_.VP8()->automaticResizeOn = false;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoEncoder::ScalingSettings settings =
+      encoder_->GetEncoderInfo().scaling_settings;
+  EXPECT_FALSE(settings.thresholds.has_value());
+}
+
+TEST_F(TestVp8Impl, ScalingEnabledIfAutomaticResizeOn) {
+  codec_settings_.SetFrameDropEnabled(true);
+  codec_settings_.VP8()->automaticResizeOn = true;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoEncoder::ScalingSettings settings =
+      encoder_->GetEncoderInfo().scaling_settings;
+  EXPECT_TRUE(settings.thresholds.has_value());
+  EXPECT_EQ(kDefaultMinPixelsPerFrame, settings.min_pixels_per_frame);
+}
+
+TEST_F(TestVp8Impl, DontDropKeyframes) {
+  // Set very high resolution to trigger overuse more easily.
+  const int kScreenWidth = 1920;
+  const int kScreenHeight = 1080;
+
+  codec_settings_.width = kScreenWidth;
+  codec_settings_.height = kScreenHeight;
+
+  // Screensharing has the internal frame dropper off, and instead per frame
+  // asks ScreenshareLayers to decide if it should be dropped or not.
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.mode = VideoCodecMode::kScreensharing;
+  // ScreenshareLayers triggers on 2 temporal layers and 1000kbps max bitrate.
+  codec_settings_.VP8()->numberOfTemporalLayers = 2;
+  codec_settings_.maxBitrate = 1000;
+
+  // Reset the frame generator with large number of squares, leading to lots of
+  // details and high probability of overshoot.
+  input_frame_generator_ = test::CreateSquareFrameGenerator(
+      codec_settings_.width, codec_settings_.height,
+      test::FrameGeneratorInterface::OutputType::kI420,
+      /* num_squares = */ absl::optional<int>(300));
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoBitrateAllocation bitrate_allocation;
+  // Bitrate only enough for TL0.
+  bitrate_allocation.SetBitrate(0, 0, 200000);
+  encoder_->SetRates(
+      VideoEncoder::RateControlParameters(bitrate_allocation, 5.0));
+
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  EncodeAndWaitForFrame(NextInputFrame(), &encoded_frame, &codec_specific_info,
+                        true);
+  EncodeAndExpectFrameWith(NextInputFrame(), 0, true);
+  EncodeAndExpectFrameWith(NextInputFrame(), 0, true);
+  EncodeAndExpectFrameWith(NextInputFrame(), 0, true);
+}
+
+TEST_F(TestVp8Impl, KeepsTimestampOnReencode) {
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           VP8Encoder::Settings());
+
+  // Settings needed to trigger ScreenshareLayers usage, which is required for
+  // overshoot-drop-reencode logic.
+  codec_settings_.maxBitrate = 1000;
+  codec_settings_.mode = VideoCodecMode::kScreensharing;
+  codec_settings_.VP8()->numberOfTemporalLayers = 2;
+  codec_settings_.legacy_conference_mode = true;
+
+  EXPECT_CALL(*vpx, img_wrap(_, _, _, _, _, _))
+      .WillOnce(Invoke([](vpx_image_t* img, vpx_img_fmt_t fmt, unsigned int d_w,
+                          unsigned int d_h, unsigned int stride_align,
+                          unsigned char* img_data) {
+        img->fmt = fmt;
+        img->d_w = d_w;
+        img->d_h = d_h;
+        img->img_data = img_data;
+        return img;
+      }));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder.InitEncode(&codec_settings_,
+                               VideoEncoder::Settings(kCapabilities, 1, 1000)));
+  MockEncodedImageCallback callback;
+  encoder.RegisterEncodeCompleteCallback(&callback);
+
+  // Simulate overshoot drop, re-encode: encode function will be called twice
+  // with the same parameters. codec_get_cx_data() will by default return no
+  // image data and be interpreted as drop.
+  EXPECT_CALL(*vpx, codec_encode(_, _, /* pts = */ 0, _, _, _))
+      .Times(2)
+      .WillRepeatedly(Return(vpx_codec_err_t::VPX_CODEC_OK));
+
+  auto delta_frame =
+      std::vector<VideoFrameType>{VideoFrameType::kVideoFrameDelta};
+  encoder.Encode(NextInputFrame(), &delta_frame);
+}
+
+TEST(LibvpxVp8EncoderTest, GetEncoderInfoReturnsStaticInformation) {
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           VP8Encoder::Settings());
+
+  const auto info = encoder.GetEncoderInfo();
+
+  EXPECT_FALSE(info.supports_native_handle);
+  EXPECT_FALSE(info.is_hardware_accelerated);
+  EXPECT_TRUE(info.supports_simulcast);
+  EXPECT_EQ(info.implementation_name, "libvpx");
+  EXPECT_EQ(info.requested_resolution_alignment, 1u);
+  EXPECT_THAT(info.preferred_pixel_formats,
+              testing::UnorderedElementsAre(VideoFrameBuffer::Type::kNV12,
+                                            VideoFrameBuffer::Type::kI420));
+}
+
+TEST(LibvpxVp8EncoderTest, RequestedResolutionAlignmentFromFieldTrial) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VP8-GetEncoderInfoOverride/"
+      "requested_resolution_alignment:10/");
+
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           VP8Encoder::Settings());
+
+  EXPECT_EQ(encoder.GetEncoderInfo().requested_resolution_alignment, 10u);
+  EXPECT_FALSE(
+      encoder.GetEncoderInfo().apply_alignment_to_all_simulcast_layers);
+  EXPECT_TRUE(encoder.GetEncoderInfo().resolution_bitrate_limits.empty());
+}
+
+TEST(LibvpxVp8EncoderTest, ResolutionBitrateLimitsFromFieldTrial) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VP8-GetEncoderInfoOverride/"
+      "frame_size_pixels:123|456|789,"
+      "min_start_bitrate_bps:11000|22000|33000,"
+      "min_bitrate_bps:44000|55000|66000,"
+      "max_bitrate_bps:77000|88000|99000/");
+
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           VP8Encoder::Settings());
+
+  EXPECT_THAT(
+      encoder.GetEncoderInfo().resolution_bitrate_limits,
+      ::testing::ElementsAre(
+          VideoEncoder::ResolutionBitrateLimits{123, 11000, 44000, 77000},
+          VideoEncoder::ResolutionBitrateLimits{456, 22000, 55000, 88000},
+          VideoEncoder::ResolutionBitrateLimits{789, 33000, 66000, 99000}));
+}
+
+TEST(LibvpxVp8EncoderTest,
+     GetEncoderInfoReturnsEmptyResolutionBitrateLimitsByDefault) {
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           VP8Encoder::Settings());
+
+  const auto info = encoder.GetEncoderInfo();
+
+  EXPECT_TRUE(info.resolution_bitrate_limits.empty());
+}
+
+TEST(LibvpxVp8EncoderTest,
+     GetEncoderInfoReturnsResolutionBitrateLimitsAsConfigured) {
+  std::vector<VideoEncoder::ResolutionBitrateLimits> resolution_bitrate_limits =
+      {VideoEncoder::ResolutionBitrateLimits(/*frame_size_pixels=*/640 * 360,
+                                             /*min_start_bitrate_bps=*/300,
+                                             /*min_bitrate_bps=*/100,
+                                             /*max_bitrate_bps=*/1000),
+       VideoEncoder::ResolutionBitrateLimits(320 * 180, 100, 30, 500)};
+  VP8Encoder::Settings settings;
+  settings.resolution_bitrate_limits = resolution_bitrate_limits;
+
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)),
+                           std::move(settings));
+
+  const auto info = encoder.GetEncoderInfo();
+
+  EXPECT_EQ(info.resolution_bitrate_limits, resolution_bitrate_limits);
+}
+
+TEST_F(TestVp8Impl, GetEncoderInfoFpsAllocationNoLayers) {
+  FramerateFractions expected_fps_allocation[kMaxSpatialLayers] = {
+      FramerateFractions(1, EncoderInfo::kMaxFramerateFraction)};
+
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(expected_fps_allocation));
+}
+
+TEST_F(TestVp8Impl, GetEncoderInfoFpsAllocationTwoTemporalLayers) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+  codec_settings_.numberOfSimulcastStreams = 1;
+  codec_settings_.simulcastStream[0].active = true;
+  codec_settings_.simulcastStream[0].targetBitrate = 100;
+  codec_settings_.simulcastStream[0].maxBitrate = 100;
+  codec_settings_.simulcastStream[0].numberOfTemporalLayers = 2;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  FramerateFractions expected_fps_allocation[kMaxSpatialLayers];
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction / 2);
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction);
+
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(expected_fps_allocation));
+}
+
+TEST_F(TestVp8Impl, GetEncoderInfoFpsAllocationThreeTemporalLayers) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+  codec_settings_.numberOfSimulcastStreams = 1;
+  codec_settings_.simulcastStream[0].active = true;
+  codec_settings_.simulcastStream[0].targetBitrate = 100;
+  codec_settings_.simulcastStream[0].maxBitrate = 100;
+  codec_settings_.simulcastStream[0].numberOfTemporalLayers = 3;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  FramerateFractions expected_fps_allocation[kMaxSpatialLayers];
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction / 4);
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction / 2);
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction);
+
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(expected_fps_allocation));
+}
+
+TEST_F(TestVp8Impl, GetEncoderInfoFpsAllocationScreenshareLayers) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+  codec_settings_.numberOfSimulcastStreams = 1;
+  codec_settings_.mode = VideoCodecMode::kScreensharing;
+  codec_settings_.simulcastStream[0].active = true;
+  codec_settings_.simulcastStream[0].minBitrate = 30;
+  codec_settings_.simulcastStream[0].targetBitrate =
+      kLegacyScreenshareTl0BitrateKbps;
+  codec_settings_.simulcastStream[0].maxBitrate =
+      kLegacyScreenshareTl1BitrateKbps;
+  codec_settings_.simulcastStream[0].numberOfTemporalLayers = 2;
+  codec_settings_.legacy_conference_mode = true;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // Expect empty vector, since this mode doesn't have a fixed framerate.
+  FramerateFractions expected_fps_allocation[kMaxSpatialLayers];
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(expected_fps_allocation));
+}
+
+TEST_F(TestVp8Impl, GetEncoderInfoFpsAllocationSimulcastVideo) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+
+  // Set up three simulcast streams with three temporal layers each.
+  codec_settings_.numberOfSimulcastStreams = 3;
+  for (int i = 0; i < codec_settings_.numberOfSimulcastStreams; ++i) {
+    codec_settings_.simulcastStream[i].active = true;
+    codec_settings_.simulcastStream[i].minBitrate = 30;
+    codec_settings_.simulcastStream[i].targetBitrate = 30;
+    codec_settings_.simulcastStream[i].maxBitrate = 30;
+    codec_settings_.simulcastStream[i].numberOfTemporalLayers = 3;
+    codec_settings_.simulcastStream[i].width =
+        codec_settings_.width >>
+        (codec_settings_.numberOfSimulcastStreams - i - 1);
+    codec_settings_.simulcastStream[i].height =
+        codec_settings_.height >>
+        (codec_settings_.numberOfSimulcastStreams - i - 1);
+  }
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  FramerateFractions expected_fps_allocation[kMaxSpatialLayers];
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction / 4);
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction / 2);
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction);
+  expected_fps_allocation[1] = expected_fps_allocation[0];
+  expected_fps_allocation[2] = expected_fps_allocation[0];
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(expected_fps_allocation));
+
+  // Release encoder and re-init without temporal layers.
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+
+  // Sanity check fps allocation when not inited.
+  FramerateFractions default_fps_fraction[kMaxSpatialLayers];
+  default_fps_fraction[0].push_back(EncoderInfo::kMaxFramerateFraction);
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(default_fps_fraction));
+
+  for (int i = 0; i < codec_settings_.numberOfSimulcastStreams; ++i) {
+    codec_settings_.simulcastStream[i].numberOfTemporalLayers = 1;
+  }
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  for (size_t i = 0; i < 3; ++i) {
+    expected_fps_allocation[i].clear();
+    expected_fps_allocation[i].push_back(EncoderInfo::kMaxFramerateFraction);
+  }
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(expected_fps_allocation));
+}
+
+class TestVp8ImplForPixelFormat
+    : public TestVp8Impl,
+      public ::testing::WithParamInterface<VideoFrameBuffer::Type> {
+ public:
+  TestVp8ImplForPixelFormat() : TestVp8Impl(), mappable_type_(GetParam()) {}
+
+ protected:
+  VideoFrameBuffer::Type mappable_type_;
+};
+
+TEST_P(TestVp8ImplForPixelFormat, EncodeNativeFrameSimulcast) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+
+  // Configure simulcast.
+  codec_settings_.numberOfSimulcastStreams = 3;
+  codec_settings_.simulcastStream[0] = {.width = kWidth / 4,
+                                        .height = kHeight / 4,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80,
+                                        .active = true};
+  codec_settings_.simulcastStream[1] = {.width = kWidth / 2,
+                                        .height = kHeight / 2,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80,
+                                        .active = true};
+  codec_settings_.simulcastStream[2] = {.width = kWidth,
+                                        .height = kHeight,
+                                        .maxFramerate = kFramerateFps,
+                                        .numberOfTemporalLayers = 1,
+                                        .maxBitrate = 4000,
+                                        .targetBitrate = 3000,
+                                        .minBitrate = 2000,
+                                        .qpMax = 80,
+                                        .active = true};
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // Create a zero-conversion NV12 frame (calling ToI420 on it crashes).
+  VideoFrame input_frame =
+      test::CreateMappableNativeFrame(1, mappable_type_, kWidth, kHeight);
+
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  EncodeAndWaitForFrame(input_frame, &encoded_frame, &codec_specific_info);
+
+  // After encoding, we expect one mapping per simulcast layer.
+  rtc::scoped_refptr<test::MappableNativeBuffer> mappable_buffer =
+      test::GetMappableNativeBufferFromVideoFrame(input_frame);
+  std::vector<rtc::scoped_refptr<VideoFrameBuffer>> mapped_buffers =
+      mappable_buffer->GetMappedFramedBuffers();
+  ASSERT_EQ(mapped_buffers.size(), 3u);
+  EXPECT_EQ(mapped_buffers[0]->type(), mappable_type_);
+  EXPECT_EQ(mapped_buffers[0]->width(), kWidth);
+  EXPECT_EQ(mapped_buffers[0]->height(), kHeight);
+  EXPECT_EQ(mapped_buffers[1]->type(), mappable_type_);
+  EXPECT_EQ(mapped_buffers[1]->width(), kWidth / 2);
+  EXPECT_EQ(mapped_buffers[1]->height(), kHeight / 2);
+  EXPECT_EQ(mapped_buffers[2]->type(), mappable_type_);
+  EXPECT_EQ(mapped_buffers[2]->width(), kWidth / 4);
+  EXPECT_EQ(mapped_buffers[2]->height(), kHeight / 4);
+  EXPECT_FALSE(mappable_buffer->DidConvertToI420());
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
+}
+
+INSTANTIATE_TEST_SUITE_P(All,
+                         TestVp8ImplForPixelFormat,
+                         ::testing::Values(VideoFrameBuffer::Type::kI420,
+                                           VideoFrameBuffer::Type::kNV12));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/vp8_scalability.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp8/vp8_scalability.cc
new file mode 100644
index 0000000000..9c7495ddf7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/vp8_scalability.cc
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp8/vp8_scalability.h"
+
+namespace webrtc {
+
+bool VP8SupportsScalabilityMode(ScalabilityMode scalability_mode) {
+  for (const auto& entry : kVP8SupportedScalabilityModes) {
+    if (entry == scalability_mode) {
+      return true;
+    }
+  }
+  return false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp8/vp8_scalability.h b/third_party/libwebrtc/modules/video_coding/codecs/vp8/vp8_scalability.h
new file mode 100644
index 0000000000..923f159118
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp8/vp8_scalability.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_VP8_SCALABILITY_H_
+#define MODULES_VIDEO_CODING_CODECS_VP8_VP8_SCALABILITY_H_
+
+#include "api/video_codecs/scalability_mode.h"
+
+namespace webrtc {
+
+inline constexpr ScalabilityMode kVP8SupportedScalabilityModes[] = {
+    ScalabilityMode::kL1T1, ScalabilityMode::kL1T2, ScalabilityMode::kL1T3};
+bool VP8SupportsScalabilityMode(ScalabilityMode scalability_mode);
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_VP8_SCALABILITY_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/DEPS b/third_party/libwebrtc/modules/video_coding/codecs/vp9/DEPS
new file mode 100644
index 0000000000..cc5cd70142
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/DEPS
@@ -0,0 +1,3 @@
+include_rules = [
+  "+media/base",
+]
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/include/vp9.h b/third_party/libwebrtc/modules/video_coding/codecs/vp9/include/vp9.h
new file mode 100644
index 0000000000..79d403ded3
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/include/vp9.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_H_
+#define MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/scalability_mode.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "media/base/codec.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+
+// Returns a vector with all supported internal VP9 profiles that we can
+// negotiate in SDP, in order of preference.
+std::vector<SdpVideoFormat> SupportedVP9Codecs(
+    bool add_scalability_modes = false);
+
+// Returns a vector with all supported internal VP9 decode profiles in order of
+// preference. These will be availble for receive-only connections.
+std::vector<SdpVideoFormat> SupportedVP9DecoderCodecs();
+
+class VP9Encoder : public VideoEncoder {
+ public:
+  // Deprecated. Returns default implementation using VP9 Profile 0.
+  // TODO(emircan): Remove once this is no longer used.
+  static std::unique_ptr<VP9Encoder> Create();
+  // Parses VP9 Profile from `codec` and returns the appropriate implementation.
+  static std::unique_ptr<VP9Encoder> Create(const cricket::VideoCodec& codec);
+  static bool SupportsScalabilityMode(ScalabilityMode scalability_mode);
+
+  ~VP9Encoder() override {}
+};
+
+class VP9Decoder : public VideoDecoder {
+ public:
+  static std::unique_ptr<VP9Decoder> Create();
+
+  ~VP9Decoder() override {}
+};
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h b/third_party/libwebrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h
new file mode 100644
index 0000000000..f67215ec77
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file contains codec dependent definitions that are needed in
+// order to compile the WebRTC codebase, even if this codec is not used.
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_GLOBALS_H_
+#define MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_GLOBALS_H_
+
+#include <stdint.h>
+
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+const int16_t kMaxOneBytePictureId = 0x7F;    // 7 bits
+const int16_t kMaxTwoBytePictureId = 0x7FFF;  // 15 bits
+const uint8_t kNoSpatialIdx = 0xFF;
+const uint8_t kNoGofIdx = 0xFF;
+const uint8_t kNumVp9Buffers = 8;
+const size_t kMaxVp9RefPics = 3;
+const size_t kMaxVp9FramesInGof = 0xFF;  // 8 bits
+const size_t kMaxVp9NumberOfSpatialLayers = 8;
+
+const size_t kMinVp9SpatialLayerLongSideLength = 240;
+const size_t kMinVp9SpatialLayerShortSideLength = 135;
+
+enum TemporalStructureMode {
+  kTemporalStructureMode1,  // 1 temporal layer structure - i.e., IPPP...
+  kTemporalStructureMode2,  // 2 temporal layers 01...
+  kTemporalStructureMode3,  // 3 temporal layers 0212...
+};
+
+struct GofInfoVP9 {
+  void SetGofInfoVP9(TemporalStructureMode tm) {
+    switch (tm) {
+      case kTemporalStructureMode1:
+        num_frames_in_gof = 1;
+        temporal_idx[0] = 0;
+        temporal_up_switch[0] = true;
+        num_ref_pics[0] = 1;
+        pid_diff[0][0] = 1;
+        break;
+      case kTemporalStructureMode2:
+        num_frames_in_gof = 2;
+        temporal_idx[0] = 0;
+        temporal_up_switch[0] = true;
+        num_ref_pics[0] = 1;
+        pid_diff[0][0] = 2;
+
+        temporal_idx[1] = 1;
+        temporal_up_switch[1] = true;
+        num_ref_pics[1] = 1;
+        pid_diff[1][0] = 1;
+        break;
+      case kTemporalStructureMode3:
+        num_frames_in_gof = 4;
+        temporal_idx[0] = 0;
+        temporal_up_switch[0] = true;
+        num_ref_pics[0] = 1;
+        pid_diff[0][0] = 4;
+
+        temporal_idx[1] = 2;
+        temporal_up_switch[1] = true;
+        num_ref_pics[1] = 1;
+        pid_diff[1][0] = 1;
+
+        temporal_idx[2] = 1;
+        temporal_up_switch[2] = true;
+        num_ref_pics[2] = 1;
+        pid_diff[2][0] = 2;
+
+        temporal_idx[3] = 2;
+        temporal_up_switch[3] = true;
+        num_ref_pics[3] = 1;
+        pid_diff[3][0] = 1;
+        break;
+      default:
+        RTC_DCHECK_NOTREACHED();
+    }
+  }
+
+  void CopyGofInfoVP9(const GofInfoVP9& src) {
+    num_frames_in_gof = src.num_frames_in_gof;
+    for (size_t i = 0; i < num_frames_in_gof; ++i) {
+      temporal_idx[i] = src.temporal_idx[i];
+      temporal_up_switch[i] = src.temporal_up_switch[i];
+      num_ref_pics[i] = src.num_ref_pics[i];
+      for (uint8_t r = 0; r < num_ref_pics[i]; ++r) {
+        pid_diff[i][r] = src.pid_diff[i][r];
+      }
+    }
+  }
+
+  size_t num_frames_in_gof;
+  uint8_t temporal_idx[kMaxVp9FramesInGof];
+  bool temporal_up_switch[kMaxVp9FramesInGof];
+  uint8_t num_ref_pics[kMaxVp9FramesInGof];
+  uint8_t pid_diff[kMaxVp9FramesInGof][kMaxVp9RefPics];
+  uint16_t pid_start;
+};
+
+struct RTPVideoHeaderVP9 {
+  void InitRTPVideoHeaderVP9() {
+    inter_pic_predicted = false;
+    flexible_mode = false;
+    beginning_of_frame = false;
+    end_of_frame = false;
+    ss_data_available = false;
+    non_ref_for_inter_layer_pred = false;
+    picture_id = kNoPictureId;
+    max_picture_id = kMaxTwoBytePictureId;
+    tl0_pic_idx = kNoTl0PicIdx;
+    temporal_idx = kNoTemporalIdx;
+    spatial_idx = kNoSpatialIdx;
+    temporal_up_switch = false;
+    inter_layer_predicted = false;
+    gof_idx = kNoGofIdx;
+    num_ref_pics = 0;
+    num_spatial_layers = 1;
+    first_active_layer = 0;
+    end_of_picture = true;
+  }
+
+  bool inter_pic_predicted;  // This layer frame is dependent on previously
+                             // coded frame(s).
+  bool flexible_mode;        // This frame is in flexible mode.
+  bool beginning_of_frame;   // True if this packet is the first in a VP9 layer
+                             // frame.
+  bool end_of_frame;  // True if this packet is the last in a VP9 layer frame.
+  bool ss_data_available;  // True if SS data is available in this payload
+                           // descriptor.
+  bool non_ref_for_inter_layer_pred;  // True for frame which is not used as
+                                      // reference for inter-layer prediction.
+  int16_t picture_id;                 // PictureID index, 15 bits;
+                       // kNoPictureId if PictureID does not exist.
+  int16_t max_picture_id;   // Maximum picture ID index; either 0x7F or 0x7FFF;
+  int16_t tl0_pic_idx;      // TL0PIC_IDX, 8 bits;
+                            // kNoTl0PicIdx means no value provided.
+  uint8_t temporal_idx;     // Temporal layer index, or kNoTemporalIdx.
+  uint8_t spatial_idx;      // Spatial layer index, or kNoSpatialIdx.
+  bool temporal_up_switch;  // True if upswitch to higher frame rate is possible
+                            // meaning subsequent higher temporal layer pictures
+                            // will not depend on any picture before the current
+                            // picture (in coding order) with temporal layer ID
+                            // greater than `temporal_idx` of this frame.
+  bool inter_layer_predicted;  // Frame is dependent on directly lower spatial
+                               // layer frame.
+
+  uint8_t gof_idx;  // Index to predefined temporal frame info in SS data.
+
+  uint8_t num_ref_pics;  // Number of reference pictures used by this layer
+                         // frame.
+  uint8_t pid_diff[kMaxVp9RefPics];  // P_DIFF signaled to derive the PictureID
+                                     // of the reference pictures.
+  int16_t ref_picture_id[kMaxVp9RefPics];  // PictureID of reference pictures.
+
+  // SS data.
+  size_t num_spatial_layers;  // Always populated.
+  size_t first_active_layer;  // Not sent on wire, used to adjust ss data.
+  bool spatial_layer_resolution_present;
+  uint16_t width[kMaxVp9NumberOfSpatialLayers];
+  uint16_t height[kMaxVp9NumberOfSpatialLayers];
+  GofInfoVP9 gof;
+
+  bool end_of_picture;  // This frame is the last frame in picture.
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_GLOBALS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc
new file mode 100644
index 0000000000..a981f259cf
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc
@@ -0,0 +1,403 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifdef RTC_ENABLE_VP9
+
+#include "modules/video_coding/codecs/vp9/libvpx_vp9_decoder.h"
+
+#include <algorithm>
+
+#include "absl/strings/match.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/video/color_space.h"
+#include "api/video/i010_buffer.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "third_party/libyuv/include/libyuv/convert.h"
+#include "vpx/vp8dx.h"
+#include "vpx/vpx_decoder.h"
+
+namespace webrtc {
+namespace {
+
+// Helper class for extracting VP9 colorspace.
+ColorSpace ExtractVP9ColorSpace(vpx_color_space_t space_t,
+                                vpx_color_range_t range_t,
+                                unsigned int bit_depth) {
+  ColorSpace::PrimaryID primaries = ColorSpace::PrimaryID::kUnspecified;
+  ColorSpace::TransferID transfer = ColorSpace::TransferID::kUnspecified;
+  ColorSpace::MatrixID matrix = ColorSpace::MatrixID::kUnspecified;
+  switch (space_t) {
+    case VPX_CS_BT_601:
+    case VPX_CS_SMPTE_170:
+      primaries = ColorSpace::PrimaryID::kSMPTE170M;
+      transfer = ColorSpace::TransferID::kSMPTE170M;
+      matrix = ColorSpace::MatrixID::kSMPTE170M;
+      break;
+    case VPX_CS_SMPTE_240:
+      primaries = ColorSpace::PrimaryID::kSMPTE240M;
+      transfer = ColorSpace::TransferID::kSMPTE240M;
+      matrix = ColorSpace::MatrixID::kSMPTE240M;
+      break;
+    case VPX_CS_BT_709:
+      primaries = ColorSpace::PrimaryID::kBT709;
+      transfer = ColorSpace::TransferID::kBT709;
+      matrix = ColorSpace::MatrixID::kBT709;
+      break;
+    case VPX_CS_BT_2020:
+      primaries = ColorSpace::PrimaryID::kBT2020;
+      switch (bit_depth) {
+        case 8:
+          transfer = ColorSpace::TransferID::kBT709;
+          break;
+        case 10:
+          transfer = ColorSpace::TransferID::kBT2020_10;
+          break;
+        default:
+          RTC_DCHECK_NOTREACHED();
+          break;
+      }
+      matrix = ColorSpace::MatrixID::kBT2020_NCL;
+      break;
+    case VPX_CS_SRGB:
+      primaries = ColorSpace::PrimaryID::kBT709;
+      transfer = ColorSpace::TransferID::kIEC61966_2_1;
+      matrix = ColorSpace::MatrixID::kBT709;
+      break;
+    default:
+      break;
+  }
+
+  ColorSpace::RangeID range = ColorSpace::RangeID::kInvalid;
+  switch (range_t) {
+    case VPX_CR_STUDIO_RANGE:
+      range = ColorSpace::RangeID::kLimited;
+      break;
+    case VPX_CR_FULL_RANGE:
+      range = ColorSpace::RangeID::kFull;
+      break;
+    default:
+      break;
+  }
+  return ColorSpace(primaries, transfer, matrix, range);
+}
+
+}  // namespace
+
+LibvpxVp9Decoder::LibvpxVp9Decoder()
+    : decode_complete_callback_(nullptr),
+      inited_(false),
+      decoder_(nullptr),
+      key_frame_required_(true) {}
+
+LibvpxVp9Decoder::~LibvpxVp9Decoder() {
+  inited_ = true;  // in order to do the actual release
+  Release();
+  int num_buffers_in_use = libvpx_buffer_pool_.GetNumBuffersInUse();
+  if (num_buffers_in_use > 0) {
+    // The frame buffers are reference counted and frames are exposed after
+    // decoding. There may be valid usage cases where previous frames are still
+    // referenced after ~LibvpxVp9Decoder that is not a leak.
+    RTC_LOG(LS_INFO) << num_buffers_in_use
+                     << " Vp9FrameBuffers are still "
+                        "referenced during ~LibvpxVp9Decoder.";
+  }
+}
+
+bool LibvpxVp9Decoder::Configure(const Settings& settings) {
+  if (Release() < 0) {
+    return false;
+  }
+
+  if (decoder_ == nullptr) {
+    decoder_ = new vpx_codec_ctx_t;
+    memset(decoder_, 0, sizeof(*decoder_));
+  }
+  vpx_codec_dec_cfg_t cfg;
+  memset(&cfg, 0, sizeof(cfg));
+
+#ifdef FUZZING_BUILD_MODE_UNSAFE_FOR_PRODUCTION
+  // We focus on webrtc fuzzing here, not libvpx itself. Use single thread for
+  // fuzzing, because:
+  //  - libvpx's VP9 single thread decoder is more fuzzer friendly. It detects
+  //    errors earlier than the multi-threads version.
+  //  - Make peak CPU usage under control (not depending on input)
+  cfg.threads = 1;
+#else
+  const RenderResolution& resolution = settings.max_render_resolution();
+  if (!resolution.Valid()) {
+    // Postpone configuring number of threads until resolution is known.
+    cfg.threads = 1;
+  } else {
+    // We want to use multithreading when decoding high resolution videos. But
+    // not too many in order to avoid overhead when many stream are decoded
+    // concurrently.
+    // Set 2 thread as target for 1280x720 pixel count, and then scale up
+    // linearly from there - but cap at physical core count.
+    // For common resolutions this results in:
+    // 1 for 360p
+    // 2 for 720p
+    // 4 for 1080p
+    // 8 for 1440p
+    // 18 for 4K
+    int num_threads = std::max(
+        1, 2 * resolution.Width() * resolution.Height() / (1280 * 720));
+    cfg.threads = std::min(settings.number_of_cores(), num_threads);
+  }
+#endif
+
+  current_settings_ = settings;
+
+  vpx_codec_flags_t flags = 0;
+  if (vpx_codec_dec_init(decoder_, vpx_codec_vp9_dx(), &cfg, flags)) {
+    return false;
+  }
+
+  if (!libvpx_buffer_pool_.InitializeVpxUsePool(decoder_)) {
+    return false;
+  }
+
+  inited_ = true;
+  // Always start with a complete key frame.
+  key_frame_required_ = true;
+  if (absl::optional<int> buffer_pool_size = settings.buffer_pool_size()) {
+    if (!libvpx_buffer_pool_.Resize(*buffer_pool_size)) {
+      return false;
+    }
+  }
+
+  vpx_codec_err_t status =
+      vpx_codec_control(decoder_, VP9D_SET_LOOP_FILTER_OPT, 1);
+  if (status != VPX_CODEC_OK) {
+    RTC_LOG(LS_ERROR) << "Failed to enable VP9D_SET_LOOP_FILTER_OPT. "
+                      << vpx_codec_error(decoder_);
+    return false;
+  }
+
+  return true;
+}
+
+int LibvpxVp9Decoder::Decode(const EncodedImage& input_image,
+                             bool missing_frames,
+                             int64_t /*render_time_ms*/) {
+  if (!inited_) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (decode_complete_callback_ == nullptr) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  if (input_image._frameType == VideoFrameType::kVideoFrameKey) {
+    absl::optional<Vp9UncompressedHeader> frame_info =
+        ParseUncompressedVp9Header(
+            rtc::MakeArrayView(input_image.data(), input_image.size()));
+    if (frame_info) {
+      RenderResolution frame_resolution(frame_info->frame_width,
+                                        frame_info->frame_height);
+      if (frame_resolution != current_settings_.max_render_resolution()) {
+        // Resolution has changed, tear down and re-init a new decoder in
+        // order to get correct sizing.
+        Release();
+        current_settings_.set_max_render_resolution(frame_resolution);
+        if (!Configure(current_settings_)) {
+          RTC_LOG(LS_WARNING) << "Failed to re-init decoder.";
+          return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+        }
+      }
+    } else {
+      RTC_LOG(LS_WARNING) << "Failed to parse VP9 header from key-frame.";
+    }
+  }
+
+  // Always start with a complete key frame.
+  if (key_frame_required_) {
+    if (input_image._frameType != VideoFrameType::kVideoFrameKey)
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    key_frame_required_ = false;
+  }
+  vpx_codec_iter_t iter = nullptr;
+  vpx_image_t* img;
+  const uint8_t* buffer = input_image.data();
+  if (input_image.size() == 0) {
+    buffer = nullptr;  // Triggers full frame concealment.
+  }
+  // During decode libvpx may get and release buffers from
+  // `libvpx_buffer_pool_`. In practice libvpx keeps a few (~3-4) buffers alive
+  // at a time.
+  if (vpx_codec_decode(decoder_, buffer,
+                       static_cast<unsigned int>(input_image.size()), 0,
+                       VPX_DL_REALTIME)) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  // `img->fb_priv` contains the image data, a reference counted Vp9FrameBuffer.
+  // It may be released by libvpx during future vpx_codec_decode or
+  // vpx_codec_destroy calls.
+  img = vpx_codec_get_frame(decoder_, &iter);
+  int qp;
+  vpx_codec_err_t vpx_ret =
+      vpx_codec_control(decoder_, VPXD_GET_LAST_QUANTIZER, &qp);
+  RTC_DCHECK_EQ(vpx_ret, VPX_CODEC_OK);
+  int ret =
+      ReturnFrame(img, input_image.Timestamp(), qp, input_image.ColorSpace());
+  if (ret != 0) {
+    return ret;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int LibvpxVp9Decoder::ReturnFrame(
+    const vpx_image_t* img,
+    uint32_t timestamp,
+    int qp,
+    const webrtc::ColorSpace* explicit_color_space) {
+  if (img == nullptr) {
+    // Decoder OK and nullptr image => No show frame.
+    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
+  }
+
+  // This buffer contains all of `img`'s image data, a reference counted
+  // Vp9FrameBuffer. (libvpx is done with the buffers after a few
+  // vpx_codec_decode calls or vpx_codec_destroy).
+  rtc::scoped_refptr<Vp9FrameBufferPool::Vp9FrameBuffer> img_buffer(
+      static_cast<Vp9FrameBufferPool::Vp9FrameBuffer*>(img->fb_priv));
+
+  // The buffer can be used directly by the VideoFrame (without copy) by
+  // using a Wrapped*Buffer.
+  rtc::scoped_refptr<VideoFrameBuffer> img_wrapped_buffer;
+  switch (img->fmt) {
+    case VPX_IMG_FMT_I420:
+      img_wrapped_buffer = WrapI420Buffer(
+          img->d_w, img->d_h, img->planes[VPX_PLANE_Y],
+          img->stride[VPX_PLANE_Y], img->planes[VPX_PLANE_U],
+          img->stride[VPX_PLANE_U], img->planes[VPX_PLANE_V],
+          img->stride[VPX_PLANE_V],
+          // WrappedI420Buffer's mechanism for allowing the release of its
+          // frame buffer is through a callback function. This is where we
+          // should release `img_buffer`.
+          [img_buffer] {});
+      break;
+    case VPX_IMG_FMT_I422:
+      img_wrapped_buffer = WrapI422Buffer(
+          img->d_w, img->d_h, img->planes[VPX_PLANE_Y],
+          img->stride[VPX_PLANE_Y], img->planes[VPX_PLANE_U],
+          img->stride[VPX_PLANE_U], img->planes[VPX_PLANE_V],
+          img->stride[VPX_PLANE_V],
+          // WrappedI444Buffer's mechanism for allowing the release of its
+          // frame buffer is through a callback function. This is where we
+          // should release `img_buffer`.
+          [img_buffer] {});
+      break;
+    case VPX_IMG_FMT_I444:
+      img_wrapped_buffer = WrapI444Buffer(
+          img->d_w, img->d_h, img->planes[VPX_PLANE_Y],
+          img->stride[VPX_PLANE_Y], img->planes[VPX_PLANE_U],
+          img->stride[VPX_PLANE_U], img->planes[VPX_PLANE_V],
+          img->stride[VPX_PLANE_V],
+          // WrappedI444Buffer's mechanism for allowing the release of its
+          // frame buffer is through a callback function. This is where we
+          // should release `img_buffer`.
+          [img_buffer] {});
+      break;
+    case VPX_IMG_FMT_I42016:
+      img_wrapped_buffer = WrapI010Buffer(
+          img->d_w, img->d_h,
+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_Y]),
+          img->stride[VPX_PLANE_Y] / 2,
+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_U]),
+          img->stride[VPX_PLANE_U] / 2,
+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_V]),
+          img->stride[VPX_PLANE_V] / 2, [img_buffer] {});
+      break;
+    case VPX_IMG_FMT_I42216:
+      img_wrapped_buffer = WrapI210Buffer(
+          img->d_w, img->d_h,
+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_Y]),
+          img->stride[VPX_PLANE_Y] / 2,
+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_U]),
+          img->stride[VPX_PLANE_U] / 2,
+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_V]),
+          img->stride[VPX_PLANE_V] / 2, [img_buffer] {});
+      break;
+    case VPX_IMG_FMT_I44416:
+      img_wrapped_buffer = WrapI410Buffer(
+          img->d_w, img->d_h,
+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_Y]),
+          img->stride[VPX_PLANE_Y] / 2,
+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_U]),
+          img->stride[VPX_PLANE_U] / 2,
+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_V]),
+          img->stride[VPX_PLANE_V] / 2, [img_buffer] {});
+      break;
+    default:
+      RTC_LOG(LS_ERROR) << "Unsupported pixel format produced by the decoder: "
+                        << static_cast<int>(img->fmt);
+      return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
+  }
+
+  auto builder = VideoFrame::Builder()
+                     .set_video_frame_buffer(img_wrapped_buffer)
+                     .set_timestamp_rtp(timestamp);
+  if (explicit_color_space) {
+    builder.set_color_space(*explicit_color_space);
+  } else {
+    builder.set_color_space(
+        ExtractVP9ColorSpace(img->cs, img->range, img->bit_depth));
+  }
+  VideoFrame decoded_image = builder.build();
+
+  decode_complete_callback_->Decoded(decoded_image, absl::nullopt, qp);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int LibvpxVp9Decoder::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  decode_complete_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int LibvpxVp9Decoder::Release() {
+  int ret_val = WEBRTC_VIDEO_CODEC_OK;
+
+  if (decoder_ != nullptr) {
+    if (inited_) {
+      // When a codec is destroyed libvpx will release any buffers of
+      // `libvpx_buffer_pool_` it is currently using.
+      if (vpx_codec_destroy(decoder_)) {
+        ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
+      }
+    }
+    delete decoder_;
+    decoder_ = nullptr;
+  }
+  // Releases buffers from the pool. Any buffers not in use are deleted. Buffers
+  // still referenced externally are deleted once fully released, not returning
+  // to the pool.
+  libvpx_buffer_pool_.ClearPool();
+  inited_ = false;
+  return ret_val;
+}
+
+VideoDecoder::DecoderInfo LibvpxVp9Decoder::GetDecoderInfo() const {
+  DecoderInfo info;
+  info.implementation_name = "libvpx";
+  info.is_hardware_accelerated = false;
+  return info;
+}
+
+const char* LibvpxVp9Decoder::ImplementationName() const {
+  return "libvpx";
+}
+
+}  // namespace webrtc
+
+#endif  // RTC_ENABLE_VP9
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.h b/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.h
new file mode 100644
index 0000000000..65fc553b82
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP9_LIBVPX_VP9_DECODER_H_
+#define MODULES_VIDEO_CODING_CODECS_VP9_LIBVPX_VP9_DECODER_H_
+
+#ifdef RTC_ENABLE_VP9
+
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+#include "modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h"
+#include "vpx/vp8cx.h"
+
+namespace webrtc {
+
+class LibvpxVp9Decoder : public VP9Decoder {
+ public:
+  LibvpxVp9Decoder();
+  virtual ~LibvpxVp9Decoder();
+
+  bool Configure(const Settings& settings) override;
+
+  int Decode(const EncodedImage& input_image,
+             bool missing_frames,
+             int64_t /*render_time_ms*/) override;
+
+  int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
+
+  int Release() override;
+
+  DecoderInfo GetDecoderInfo() const override;
+  const char* ImplementationName() const override;
+
+ private:
+  int ReturnFrame(const vpx_image_t* img,
+                  uint32_t timestamp,
+                  int qp,
+                  const webrtc::ColorSpace* explicit_color_space);
+
+  // Memory pool used to share buffers between libvpx and webrtc.
+  Vp9FrameBufferPool libvpx_buffer_pool_;
+  DecodedImageCallback* decode_complete_callback_;
+  bool inited_;
+  vpx_codec_ctx_t* decoder_;
+  bool key_frame_required_;
+  Settings current_settings_;
+};
+}  // namespace webrtc
+
+#endif  // RTC_ENABLE_VP9
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP9_LIBVPX_VP9_DECODER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
new file mode 100644
index 0000000000..5877373b76
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
@@ -0,0 +1,2194 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include <memory>
+#ifdef RTC_ENABLE_VP9
+
+#include <algorithm>
+#include <limits>
+#include <tuple>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/memory/memory.h"
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/video/color_space.h"
+#include "api/video/i010_buffer.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h"
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "modules/video_coding/svc/scalable_video_controller_no_layering.h"
+#include "modules/video_coding/svc/svc_rate_allocator.h"
+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_list.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "third_party/libyuv/include/libyuv/convert.h"
+#include "vpx/vp8cx.h"
+#include "vpx/vpx_encoder.h"
+
+namespace webrtc {
+
+namespace {
+// Maps from gof_idx to encoder internal reference frame buffer index. These
+// maps work for 1,2 and 3 temporal layers with GOF length of 1,2 and 4 frames.
+uint8_t kRefBufIdx[4] = {0, 0, 0, 1};
+uint8_t kUpdBufIdx[4] = {0, 0, 1, 0};
+
+// Maximum allowed PID difference for differnet per-layer frame-rate case.
+const int kMaxAllowedPidDiff = 30;
+
+// TODO(ilink): Tune these thresholds further.
+// Selected using ConverenceMotion_1280_720_50.yuv clip.
+// No toggling observed on any link capacity from 100-2000kbps.
+// HD was reached consistently when link capacity was 1500kbps.
+// Set resolutions are a bit more conservative than svc_config.cc sets, e.g.
+// for 300kbps resolution converged to 270p instead of 360p.
+constexpr int kLowVp9QpThreshold = 149;
+constexpr int kHighVp9QpThreshold = 205;
+
+std::pair<size_t, size_t> GetActiveLayers(
+    const VideoBitrateAllocation& allocation) {
+  for (size_t sl_idx = 0; sl_idx < kMaxSpatialLayers; ++sl_idx) {
+    if (allocation.GetSpatialLayerSum(sl_idx) > 0) {
+      size_t last_layer = sl_idx + 1;
+      while (last_layer < kMaxSpatialLayers &&
+             allocation.GetSpatialLayerSum(last_layer) > 0) {
+        ++last_layer;
+      }
+      return std::make_pair(sl_idx, last_layer);
+    }
+  }
+  return {0, 0};
+}
+
+using Vp9ScalabilityStructure =
+    std::tuple<std::unique_ptr<ScalableVideoController>, ScalabilityMode>;
+absl::optional<Vp9ScalabilityStructure> CreateVp9ScalabilityStructure(
+    const VideoCodec& codec) {
+  int num_spatial_layers = codec.VP9().numberOfSpatialLayers;
+  int num_temporal_layers =
+      std::max(1, int{codec.VP9().numberOfTemporalLayers});
+  if (num_spatial_layers == 1 && num_temporal_layers == 1) {
+    return absl::make_optional<Vp9ScalabilityStructure>(
+        std::make_unique<ScalableVideoControllerNoLayering>(),
+        ScalabilityMode::kL1T1);
+  }
+
+  char name[20];
+  rtc::SimpleStringBuilder ss(name);
+  if (codec.mode == VideoCodecMode::kScreensharing) {
+    // TODO(bugs.webrtc.org/11999): Compose names of the structures when they
+    // are implemented.
+    return absl::nullopt;
+  } else if (codec.VP9().interLayerPred == InterLayerPredMode::kOn ||
+             num_spatial_layers == 1) {
+    ss << "L" << num_spatial_layers << "T" << num_temporal_layers;
+  } else if (codec.VP9().interLayerPred == InterLayerPredMode::kOnKeyPic) {
+    ss << "L" << num_spatial_layers << "T" << num_temporal_layers << "_KEY";
+  } else {
+    RTC_DCHECK_EQ(codec.VP9().interLayerPred, InterLayerPredMode::kOff);
+    ss << "S" << num_spatial_layers << "T" << num_temporal_layers;
+  }
+
+  // Check spatial ratio.
+  if (num_spatial_layers > 1 && codec.spatialLayers[0].targetBitrate > 0) {
+    if (codec.width != codec.spatialLayers[num_spatial_layers - 1].width ||
+        codec.height != codec.spatialLayers[num_spatial_layers - 1].height) {
+      RTC_LOG(LS_WARNING)
+          << "Top layer resolution expected to match overall resolution";
+      return absl::nullopt;
+    }
+    // Check if the ratio is one of the supported.
+    int numerator;
+    int denominator;
+    if (codec.spatialLayers[1].width == 2 * codec.spatialLayers[0].width) {
+      numerator = 1;
+      denominator = 2;
+      // no suffix for 1:2 ratio.
+    } else if (2 * codec.spatialLayers[1].width ==
+               3 * codec.spatialLayers[0].width) {
+      numerator = 2;
+      denominator = 3;
+      ss << "h";
+    } else {
+      RTC_LOG(LS_WARNING) << "Unsupported scalability ratio "
+                          << codec.spatialLayers[0].width << ":"
+                          << codec.spatialLayers[1].width;
+      return absl::nullopt;
+    }
+    // Validate ratio is consistent for all spatial layer transitions.
+    for (int sid = 1; sid < num_spatial_layers; ++sid) {
+      if (codec.spatialLayers[sid].width * numerator !=
+              codec.spatialLayers[sid - 1].width * denominator ||
+          codec.spatialLayers[sid].height * numerator !=
+              codec.spatialLayers[sid - 1].height * denominator) {
+        RTC_LOG(LS_WARNING) << "Inconsistent scalability ratio " << numerator
+                            << ":" << denominator;
+        return absl::nullopt;
+      }
+    }
+  }
+
+  absl::optional<ScalabilityMode> scalability_mode =
+      ScalabilityModeFromString(name);
+  if (!scalability_mode.has_value()) {
+    RTC_LOG(LS_WARNING) << "Invalid scalability mode " << name;
+    return absl::nullopt;
+  }
+  auto scalability_structure_controller =
+      CreateScalabilityStructure(*scalability_mode);
+  if (scalability_structure_controller == nullptr) {
+    RTC_LOG(LS_WARNING) << "Unsupported scalability structure " << name;
+  } else {
+    RTC_LOG(LS_INFO) << "Created scalability structure " << name;
+  }
+  return absl::make_optional<Vp9ScalabilityStructure>(
+      std::move(scalability_structure_controller), *scalability_mode);
+}
+
+vpx_svc_ref_frame_config_t Vp9References(
+    rtc::ArrayView<const ScalableVideoController::LayerFrameConfig> layers) {
+  vpx_svc_ref_frame_config_t ref_config = {};
+  for (const ScalableVideoController::LayerFrameConfig& layer_frame : layers) {
+    const auto& buffers = layer_frame.Buffers();
+    RTC_DCHECK_LE(buffers.size(), 3);
+    int sid = layer_frame.SpatialId();
+    if (!buffers.empty()) {
+      ref_config.lst_fb_idx[sid] = buffers[0].id;
+      ref_config.reference_last[sid] = buffers[0].referenced;
+      if (buffers[0].updated) {
+        ref_config.update_buffer_slot[sid] |= (1 << buffers[0].id);
+      }
+    }
+    if (buffers.size() > 1) {
+      ref_config.gld_fb_idx[sid] = buffers[1].id;
+      ref_config.reference_golden[sid] = buffers[1].referenced;
+      if (buffers[1].updated) {
+        ref_config.update_buffer_slot[sid] |= (1 << buffers[1].id);
+      }
+    }
+    if (buffers.size() > 2) {
+      ref_config.alt_fb_idx[sid] = buffers[2].id;
+      ref_config.reference_alt_ref[sid] = buffers[2].referenced;
+      if (buffers[2].updated) {
+        ref_config.update_buffer_slot[sid] |= (1 << buffers[2].id);
+      }
+    }
+  }
+  // TODO(bugs.webrtc.org/11999): Fill ref_config.duration
+  return ref_config;
+}
+
+bool AllowDenoising() {
+  // Do not enable the denoiser on ARM since optimization is pending.
+  // Denoiser is on by default on other platforms.
+#if !defined(WEBRTC_ARCH_ARM) && !defined(WEBRTC_ARCH_ARM64) && \
+    !defined(ANDROID)
+  return true;
+#else
+  return false;
+#endif
+}
+
+}  // namespace
+
+void LibvpxVp9Encoder::EncoderOutputCodedPacketCallback(vpx_codec_cx_pkt* pkt,
+                                                        void* user_data) {
+  LibvpxVp9Encoder* enc = static_cast<LibvpxVp9Encoder*>(user_data);
+  enc->GetEncodedLayerFrame(pkt);
+}
+
+LibvpxVp9Encoder::LibvpxVp9Encoder(const cricket::VideoCodec& codec,
+                                   std::unique_ptr<LibvpxInterface> interface,
+                                   const FieldTrialsView& trials)
+    : libvpx_(std::move(interface)),
+      encoded_image_(),
+      encoded_complete_callback_(nullptr),
+      profile_(
+          ParseSdpForVP9Profile(codec.params).value_or(VP9Profile::kProfile0)),
+      inited_(false),
+      timestamp_(0),
+      rc_max_intra_target_(0),
+      encoder_(nullptr),
+      config_(nullptr),
+      raw_(nullptr),
+      input_image_(nullptr),
+      force_key_frame_(true),
+      pics_since_key_(0),
+      num_temporal_layers_(0),
+      num_spatial_layers_(0),
+      num_active_spatial_layers_(0),
+      first_active_layer_(0),
+      layer_deactivation_requires_key_frame_(absl::StartsWith(
+          trials.Lookup("WebRTC-Vp9IssueKeyFrameOnLayerDeactivation"),
+          "Enabled")),
+      is_svc_(false),
+      inter_layer_pred_(InterLayerPredMode::kOn),
+      external_ref_control_(false),  // Set in InitEncode because of tests.
+      trusted_rate_controller_(
+          RateControlSettings::ParseFromKeyValueConfig(&trials)
+              .LibvpxVp9TrustedRateController()),
+      layer_buffering_(false),
+      full_superframe_drop_(true),
+      first_frame_in_picture_(true),
+      ss_info_needed_(false),
+      force_all_active_layers_(false),
+      num_cores_(0),
+      is_flexible_mode_(false),
+      variable_framerate_experiment_(ParseVariableFramerateConfig(trials)),
+      variable_framerate_controller_(
+          variable_framerate_experiment_.framerate_limit),
+      quality_scaler_experiment_(ParseQualityScalerConfig(trials)),
+      external_ref_ctrl_(
+          !absl::StartsWith(trials.Lookup("WebRTC-Vp9ExternalRefCtrl"),
+                            "Disabled")),
+      performance_flags_(ParsePerformanceFlagsFromTrials(trials)),
+      num_steady_state_frames_(0),
+      config_changed_(true) {
+  codec_ = {};
+  memset(&svc_params_, 0, sizeof(vpx_svc_extra_cfg_t));
+}
+
+LibvpxVp9Encoder::~LibvpxVp9Encoder() {
+  Release();
+}
+
+void LibvpxVp9Encoder::SetFecControllerOverride(FecControllerOverride*) {
+  // Ignored.
+}
+
+int LibvpxVp9Encoder::Release() {
+  int ret_val = WEBRTC_VIDEO_CODEC_OK;
+
+  if (encoder_ != nullptr) {
+    if (inited_) {
+      if (libvpx_->codec_destroy(encoder_)) {
+        ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
+      }
+    }
+    delete encoder_;
+    encoder_ = nullptr;
+  }
+  if (config_ != nullptr) {
+    delete config_;
+    config_ = nullptr;
+  }
+  if (raw_ != nullptr) {
+    libvpx_->img_free(raw_);
+    raw_ = nullptr;
+  }
+  inited_ = false;
+  return ret_val;
+}
+
+bool LibvpxVp9Encoder::ExplicitlyConfiguredSpatialLayers() const {
+  // We check target_bitrate_bps of the 0th layer to see if the spatial layers
+  // (i.e. bitrates) were explicitly configured.
+  return codec_.spatialLayers[0].targetBitrate > 0;
+}
+
+bool LibvpxVp9Encoder::SetSvcRates(
+    const VideoBitrateAllocation& bitrate_allocation) {
+  std::pair<size_t, size_t> current_layers =
+      GetActiveLayers(current_bitrate_allocation_);
+  std::pair<size_t, size_t> new_layers = GetActiveLayers(bitrate_allocation);
+
+  const bool layer_activation_requires_key_frame =
+      inter_layer_pred_ == InterLayerPredMode::kOff ||
+      inter_layer_pred_ == InterLayerPredMode::kOnKeyPic;
+  const bool lower_layers_enabled = new_layers.first < current_layers.first;
+  const bool higher_layers_enabled = new_layers.second > current_layers.second;
+  const bool disabled_layers = new_layers.first > current_layers.first ||
+                               new_layers.second < current_layers.second;
+
+  if (lower_layers_enabled ||
+      (higher_layers_enabled && layer_activation_requires_key_frame) ||
+      (disabled_layers && layer_deactivation_requires_key_frame_)) {
+    force_key_frame_ = true;
+  }
+
+  if (current_layers != new_layers) {
+    ss_info_needed_ = true;
+  }
+
+  config_->rc_target_bitrate = bitrate_allocation.get_sum_kbps();
+
+  if (ExplicitlyConfiguredSpatialLayers()) {
+    for (size_t sl_idx = 0; sl_idx < num_spatial_layers_; ++sl_idx) {
+      const bool was_layer_active = (config_->ss_target_bitrate[sl_idx] > 0);
+      config_->ss_target_bitrate[sl_idx] =
+          bitrate_allocation.GetSpatialLayerSum(sl_idx) / 1000;
+
+      for (size_t tl_idx = 0; tl_idx < num_temporal_layers_; ++tl_idx) {
+        config_->layer_target_bitrate[sl_idx * num_temporal_layers_ + tl_idx] =
+            bitrate_allocation.GetTemporalLayerSum(sl_idx, tl_idx) / 1000;
+      }
+
+      if (!was_layer_active) {
+        // Reset frame rate controller if layer is resumed after pause.
+        framerate_controller_[sl_idx].Reset();
+      }
+
+      framerate_controller_[sl_idx].SetTargetRate(
+          codec_.spatialLayers[sl_idx].maxFramerate);
+    }
+  } else {
+    float rate_ratio[VPX_MAX_LAYERS] = {0};
+    float total = 0;
+    for (int i = 0; i < num_spatial_layers_; ++i) {
+      if (svc_params_.scaling_factor_num[i] <= 0 ||
+          svc_params_.scaling_factor_den[i] <= 0) {
+        RTC_LOG(LS_ERROR) << "Scaling factors not specified!";
+        return false;
+      }
+      rate_ratio[i] = static_cast<float>(svc_params_.scaling_factor_num[i]) /
+                      svc_params_.scaling_factor_den[i];
+      total += rate_ratio[i];
+    }
+
+    for (int i = 0; i < num_spatial_layers_; ++i) {
+      RTC_CHECK_GT(total, 0);
+      config_->ss_target_bitrate[i] = static_cast<unsigned int>(
+          config_->rc_target_bitrate * rate_ratio[i] / total);
+      if (num_temporal_layers_ == 1) {
+        config_->layer_target_bitrate[i] = config_->ss_target_bitrate[i];
+      } else if (num_temporal_layers_ == 2) {
+        config_->layer_target_bitrate[i * num_temporal_layers_] =
+            config_->ss_target_bitrate[i] * 2 / 3;
+        config_->layer_target_bitrate[i * num_temporal_layers_ + 1] =
+            config_->ss_target_bitrate[i];
+      } else if (num_temporal_layers_ == 3) {
+        config_->layer_target_bitrate[i * num_temporal_layers_] =
+            config_->ss_target_bitrate[i] / 2;
+        config_->layer_target_bitrate[i * num_temporal_layers_ + 1] =
+            config_->layer_target_bitrate[i * num_temporal_layers_] +
+            (config_->ss_target_bitrate[i] / 4);
+        config_->layer_target_bitrate[i * num_temporal_layers_ + 2] =
+            config_->ss_target_bitrate[i];
+      } else {
+        RTC_LOG(LS_ERROR) << "Unsupported number of temporal layers: "
+                          << num_temporal_layers_;
+        return false;
+      }
+
+      framerate_controller_[i].SetTargetRate(codec_.maxFramerate);
+    }
+  }
+
+  num_active_spatial_layers_ = 0;
+  first_active_layer_ = 0;
+  bool seen_active_layer = false;
+  bool expect_no_more_active_layers = false;
+  for (int i = 0; i < num_spatial_layers_; ++i) {
+    if (config_->ss_target_bitrate[i] > 0) {
+      RTC_DCHECK(!expect_no_more_active_layers) << "Only middle layer is "
+                                                   "deactivated.";
+      if (!seen_active_layer) {
+        first_active_layer_ = i;
+      }
+      num_active_spatial_layers_ = i + 1;
+      seen_active_layer = true;
+    } else {
+      expect_no_more_active_layers = seen_active_layer;
+    }
+  }
+
+  if (seen_active_layer && performance_flags_.use_per_layer_speed) {
+    bool denoiser_on =
+        AllowDenoising() && codec_.VP9()->denoisingOn &&
+        performance_flags_by_spatial_index_[num_active_spatial_layers_ - 1]
+            .allow_denoising;
+    libvpx_->codec_control(encoder_, VP9E_SET_NOISE_SENSITIVITY,
+                           denoiser_on ? 1 : 0);
+  }
+
+  if (higher_layers_enabled && !force_key_frame_) {
+    // Prohibit drop of all layers for the next frame, so newly enabled
+    // layer would have a valid spatial reference.
+    for (size_t i = 0; i < num_spatial_layers_; ++i) {
+      svc_drop_frame_.framedrop_thresh[i] = 0;
+    }
+    force_all_active_layers_ = true;
+  }
+
+  if (svc_controller_) {
+    for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+      // Bitrates in `layer_target_bitrate` are accumulated for each temporal
+      // layer but in `VideoBitrateAllocation` they should be separated.
+      int previous_bitrate_kbps = 0;
+      for (int tid = 0; tid < num_temporal_layers_; ++tid) {
+        int accumulated_bitrate_kbps =
+            config_->layer_target_bitrate[sid * num_temporal_layers_ + tid];
+        int single_layer_bitrate_kbps =
+            accumulated_bitrate_kbps - previous_bitrate_kbps;
+        RTC_DCHECK_GE(single_layer_bitrate_kbps, 0);
+        current_bitrate_allocation_.SetBitrate(
+            sid, tid, single_layer_bitrate_kbps * 1'000);
+        previous_bitrate_kbps = accumulated_bitrate_kbps;
+      }
+    }
+    svc_controller_->OnRatesUpdated(current_bitrate_allocation_);
+  } else {
+    current_bitrate_allocation_ = bitrate_allocation;
+  }
+  config_changed_ = true;
+  return true;
+}
+
+void LibvpxVp9Encoder::DisableSpatialLayer(int sid) {
+  RTC_DCHECK_LT(sid, num_spatial_layers_);
+  if (config_->ss_target_bitrate[sid] == 0) {
+    return;
+  }
+  config_->ss_target_bitrate[sid] = 0;
+  for (int tid = 0; tid < num_temporal_layers_; ++tid) {
+    config_->layer_target_bitrate[sid * num_temporal_layers_ + tid] = 0;
+  }
+  config_changed_ = true;
+}
+
+void LibvpxVp9Encoder::EnableSpatialLayer(int sid) {
+  RTC_DCHECK_LT(sid, num_spatial_layers_);
+  if (config_->ss_target_bitrate[sid] > 0) {
+    return;
+  }
+  for (int tid = 0; tid < num_temporal_layers_; ++tid) {
+    config_->layer_target_bitrate[sid * num_temporal_layers_ + tid] =
+        current_bitrate_allocation_.GetTemporalLayerSum(sid, tid) / 1000;
+  }
+  config_->ss_target_bitrate[sid] =
+      current_bitrate_allocation_.GetSpatialLayerSum(sid) / 1000;
+  RTC_DCHECK_GT(config_->ss_target_bitrate[sid], 0);
+  config_changed_ = true;
+}
+
+void LibvpxVp9Encoder::SetActiveSpatialLayers() {
+  // Svc controller may decide to skip a frame at certain spatial layer even
+  // when bitrate for it is non-zero, however libvpx uses configured bitrate as
+  // a signal which layers should be produced.
+  RTC_DCHECK(svc_controller_);
+  RTC_DCHECK(!layer_frames_.empty());
+  RTC_DCHECK(absl::c_is_sorted(
+      layer_frames_, [](const ScalableVideoController::LayerFrameConfig& lhs,
+                        const ScalableVideoController::LayerFrameConfig& rhs) {
+        return lhs.SpatialId() < rhs.SpatialId();
+      }));
+
+  auto frame_it = layer_frames_.begin();
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    if (frame_it != layer_frames_.end() && frame_it->SpatialId() == sid) {
+      EnableSpatialLayer(sid);
+      ++frame_it;
+    } else {
+      DisableSpatialLayer(sid);
+    }
+  }
+}
+
+void LibvpxVp9Encoder::SetRates(const RateControlParameters& parameters) {
+  if (!inited_) {
+    RTC_LOG(LS_WARNING) << "SetRates() called while uninitialized.";
+    return;
+  }
+  if (encoder_->err) {
+    RTC_LOG(LS_WARNING) << "Encoder in error state: " << encoder_->err;
+    return;
+  }
+  if (parameters.framerate_fps < 1.0) {
+    RTC_LOG(LS_WARNING) << "Unsupported framerate: "
+                        << parameters.framerate_fps;
+    return;
+  }
+
+  codec_.maxFramerate = static_cast<uint32_t>(parameters.framerate_fps + 0.5);
+
+  bool res = SetSvcRates(parameters.bitrate);
+  RTC_DCHECK(res) << "Failed to set new bitrate allocation";
+  config_changed_ = true;
+}
+
+// TODO(eladalon): s/inst/codec_settings/g.
+int LibvpxVp9Encoder::InitEncode(const VideoCodec* inst,
+                                 const Settings& settings) {
+  if (inst == nullptr) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (inst->maxFramerate < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  // Allow zero to represent an unspecified maxBitRate
+  if (inst->maxBitrate > 0 && inst->startBitrate > inst->maxBitrate) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (inst->width < 1 || inst->height < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (settings.number_of_cores < 1) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  if (inst->VP9().numberOfTemporalLayers > 3) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  // libvpx probably does not support more than 3 spatial layers.
+  if (inst->VP9().numberOfSpatialLayers > 3) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  absl::optional<vpx_img_fmt_t> previous_img_fmt =
+      raw_ ? absl::make_optional<vpx_img_fmt_t>(raw_->fmt) : absl::nullopt;
+
+  int ret_val = Release();
+  if (ret_val < 0) {
+    return ret_val;
+  }
+  if (encoder_ == nullptr) {
+    encoder_ = new vpx_codec_ctx_t;
+    memset(encoder_, 0, sizeof(*encoder_));
+  }
+  if (config_ == nullptr) {
+    config_ = new vpx_codec_enc_cfg_t;
+    memset(config_, 0, sizeof(*config_));
+  }
+  timestamp_ = 0;
+  if (&codec_ != inst) {
+    codec_ = *inst;
+  }
+  memset(&svc_params_, 0, sizeof(vpx_svc_extra_cfg_t));
+
+  force_key_frame_ = true;
+  pics_since_key_ = 0;
+  num_cores_ = settings.number_of_cores;
+
+  scalability_mode_ = inst->GetScalabilityMode();
+  if (scalability_mode_.has_value()) {
+    // Use settings from `ScalabilityMode` identifier.
+    RTC_LOG(LS_INFO) << "Create scalability structure "
+                     << ScalabilityModeToString(*scalability_mode_);
+    svc_controller_ = CreateScalabilityStructure(*scalability_mode_);
+    if (!svc_controller_) {
+      RTC_LOG(LS_WARNING) << "Failed to create scalability structure.";
+      return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+    ScalableVideoController::StreamLayersConfig info =
+        svc_controller_->StreamConfig();
+    num_spatial_layers_ = info.num_spatial_layers;
+    num_temporal_layers_ = info.num_temporal_layers;
+    inter_layer_pred_ = ScalabilityModeToInterLayerPredMode(*scalability_mode_);
+  } else {
+    num_spatial_layers_ = inst->VP9().numberOfSpatialLayers;
+    RTC_DCHECK_GT(num_spatial_layers_, 0);
+    num_temporal_layers_ = inst->VP9().numberOfTemporalLayers;
+    if (num_temporal_layers_ == 0) {
+      num_temporal_layers_ = 1;
+    }
+    inter_layer_pred_ = inst->VP9().interLayerPred;
+    auto vp9_scalability = CreateVp9ScalabilityStructure(*inst);
+    if (vp9_scalability.has_value()) {
+      std::tie(svc_controller_, scalability_mode_) =
+          std::move(vp9_scalability.value());
+    } else {
+      svc_controller_ = nullptr;
+      scalability_mode_ = absl::nullopt;
+    }
+  }
+
+  framerate_controller_ = std::vector<FramerateControllerDeprecated>(
+      num_spatial_layers_, FramerateControllerDeprecated(codec_.maxFramerate));
+
+  is_svc_ = (num_spatial_layers_ > 1 || num_temporal_layers_ > 1);
+
+  // Populate encoder configuration with default values.
+  if (libvpx_->codec_enc_config_default(vpx_codec_vp9_cx(), config_, 0)) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  vpx_img_fmt img_fmt = VPX_IMG_FMT_NONE;
+  unsigned int bits_for_storage = 8;
+  switch (profile_) {
+    case VP9Profile::kProfile0:
+      img_fmt = previous_img_fmt.value_or(VPX_IMG_FMT_I420);
+      bits_for_storage = 8;
+      config_->g_bit_depth = VPX_BITS_8;
+      config_->g_profile = 0;
+      config_->g_input_bit_depth = 8;
+      break;
+    case VP9Profile::kProfile1:
+      // Encoding of profile 1 is not implemented. It would require extended
+      // support for I444, I422, and I440 buffers.
+      RTC_DCHECK_NOTREACHED();
+      break;
+    case VP9Profile::kProfile2:
+      img_fmt = VPX_IMG_FMT_I42016;
+      bits_for_storage = 16;
+      config_->g_bit_depth = VPX_BITS_10;
+      config_->g_profile = 2;
+      config_->g_input_bit_depth = 10;
+      break;
+    case VP9Profile::kProfile3:
+      // Encoding of profile 3 is not implemented.
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+
+  // Creating a wrapper to the image - setting image data to nullptr. Actual
+  // pointer will be set in encode. Setting align to 1, as it is meaningless
+  // (actual memory is not allocated).
+  raw_ = libvpx_->img_wrap(nullptr, img_fmt, codec_.width, codec_.height, 1,
+                           nullptr);
+  raw_->bit_depth = bits_for_storage;
+
+  config_->g_w = codec_.width;
+  config_->g_h = codec_.height;
+  config_->rc_target_bitrate = inst->startBitrate;  // in kbit/s
+  config_->g_error_resilient = is_svc_ ? VPX_ERROR_RESILIENT_DEFAULT : 0;
+  // Setting the time base of the codec.
+  config_->g_timebase.num = 1;
+  config_->g_timebase.den = 90000;
+  config_->g_lag_in_frames = 0;  // 0- no frame lagging
+  config_->g_threads = 1;
+  // Rate control settings.
+  config_->rc_dropframe_thresh = inst->GetFrameDropEnabled() ? 30 : 0;
+  config_->rc_end_usage = VPX_CBR;
+  config_->g_pass = VPX_RC_ONE_PASS;
+  config_->rc_min_quantizer =
+      codec_.mode == VideoCodecMode::kScreensharing ? 8 : 2;
+  config_->rc_max_quantizer = 52;
+  config_->rc_undershoot_pct = 50;
+  config_->rc_overshoot_pct = 50;
+  config_->rc_buf_initial_sz = 500;
+  config_->rc_buf_optimal_sz = 600;
+  config_->rc_buf_sz = 1000;
+  // Set the maximum target size of any key-frame.
+  rc_max_intra_target_ = MaxIntraTarget(config_->rc_buf_optimal_sz);
+  // Key-frame interval is enforced manually by this wrapper.
+  config_->kf_mode = VPX_KF_DISABLED;
+  // TODO(webm:1592): work-around for libvpx issue, as it can still
+  // put some key-frames at will even in VPX_KF_DISABLED kf_mode.
+  config_->kf_max_dist = inst->VP9().keyFrameInterval;
+  config_->kf_min_dist = config_->kf_max_dist;
+  if (quality_scaler_experiment_.enabled) {
+    // In that experiment webrtc wide quality scaler is used instead of libvpx
+    // internal scaler.
+    config_->rc_resize_allowed = 0;
+  } else {
+    config_->rc_resize_allowed = inst->VP9().automaticResizeOn ? 1 : 0;
+  }
+  // Determine number of threads based on the image size and #cores.
+  config_->g_threads =
+      NumberOfThreads(config_->g_w, config_->g_h, settings.number_of_cores);
+
+  is_flexible_mode_ = inst->VP9().flexibleMode;
+
+  if (num_spatial_layers_ > 1 &&
+      codec_.mode == VideoCodecMode::kScreensharing && !is_flexible_mode_) {
+    RTC_LOG(LS_ERROR) << "Flexible mode is required for screenshare with "
+                         "several spatial layers";
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  // External reference control is required for different frame rate on spatial
+  // layers because libvpx generates rtp incompatible references in this case.
+  external_ref_control_ = external_ref_ctrl_ ||
+                          (num_spatial_layers_ > 1 &&
+                           codec_.mode == VideoCodecMode::kScreensharing) ||
+                          inter_layer_pred_ == InterLayerPredMode::kOn;
+
+  if (num_temporal_layers_ == 1) {
+    gof_.SetGofInfoVP9(kTemporalStructureMode1);
+    config_->temporal_layering_mode = VP9E_TEMPORAL_LAYERING_MODE_NOLAYERING;
+    config_->ts_number_layers = 1;
+    config_->ts_rate_decimator[0] = 1;
+    config_->ts_periodicity = 1;
+    config_->ts_layer_id[0] = 0;
+  } else if (num_temporal_layers_ == 2) {
+    gof_.SetGofInfoVP9(kTemporalStructureMode2);
+    config_->temporal_layering_mode = VP9E_TEMPORAL_LAYERING_MODE_0101;
+    config_->ts_number_layers = 2;
+    config_->ts_rate_decimator[0] = 2;
+    config_->ts_rate_decimator[1] = 1;
+    config_->ts_periodicity = 2;
+    config_->ts_layer_id[0] = 0;
+    config_->ts_layer_id[1] = 1;
+  } else if (num_temporal_layers_ == 3) {
+    gof_.SetGofInfoVP9(kTemporalStructureMode3);
+    config_->temporal_layering_mode = VP9E_TEMPORAL_LAYERING_MODE_0212;
+    config_->ts_number_layers = 3;
+    config_->ts_rate_decimator[0] = 4;
+    config_->ts_rate_decimator[1] = 2;
+    config_->ts_rate_decimator[2] = 1;
+    config_->ts_periodicity = 4;
+    config_->ts_layer_id[0] = 0;
+    config_->ts_layer_id[1] = 2;
+    config_->ts_layer_id[2] = 1;
+    config_->ts_layer_id[3] = 2;
+  } else {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  if (external_ref_control_) {
+    config_->temporal_layering_mode = VP9E_TEMPORAL_LAYERING_MODE_BYPASS;
+    if (num_temporal_layers_ > 1 && num_spatial_layers_ > 1 &&
+        codec_.mode == VideoCodecMode::kScreensharing) {
+      // External reference control for several temporal layers with different
+      // frame rates on spatial layers is not implemented yet.
+      return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+    }
+  }
+  ref_buf_ = {};
+
+  return InitAndSetControlSettings(inst);
+}
+
+int LibvpxVp9Encoder::NumberOfThreads(int width,
+                                      int height,
+                                      int number_of_cores) {
+  // Keep the number of encoder threads equal to the possible number of column
+  // tiles, which is (1, 2, 4, 8). See comments below for VP9E_SET_TILE_COLUMNS.
+  if (width * height >= 1280 * 720 && number_of_cores > 4) {
+    return 4;
+  } else if (width * height >= 640 * 360 && number_of_cores > 2) {
+    return 2;
+  } else {
+// Use 2 threads for low res on ARM.
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
+    defined(WEBRTC_ANDROID)
+    if (width * height >= 320 * 180 && number_of_cores > 2) {
+      return 2;
+    }
+#endif
+    // 1 thread less than VGA.
+    return 1;
+  }
+}
+
+int LibvpxVp9Encoder::InitAndSetControlSettings(const VideoCodec* inst) {
+  // Set QP-min/max per spatial and temporal layer.
+  int tot_num_layers = num_spatial_layers_ * num_temporal_layers_;
+  for (int i = 0; i < tot_num_layers; ++i) {
+    svc_params_.max_quantizers[i] = config_->rc_max_quantizer;
+    svc_params_.min_quantizers[i] = config_->rc_min_quantizer;
+  }
+  config_->ss_number_layers = num_spatial_layers_;
+  if (svc_controller_) {
+    auto stream_config = svc_controller_->StreamConfig();
+    for (int i = 0; i < stream_config.num_spatial_layers; ++i) {
+      svc_params_.scaling_factor_num[i] = stream_config.scaling_factor_num[i];
+      svc_params_.scaling_factor_den[i] = stream_config.scaling_factor_den[i];
+    }
+  } else if (ExplicitlyConfiguredSpatialLayers()) {
+    for (int i = 0; i < num_spatial_layers_; ++i) {
+      const auto& layer = codec_.spatialLayers[i];
+      RTC_CHECK_GT(layer.width, 0);
+      const int scale_factor = codec_.width / layer.width;
+      RTC_DCHECK_GT(scale_factor, 0);
+
+      // Ensure scaler factor is integer.
+      if (scale_factor * layer.width != codec_.width) {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+      }
+
+      // Ensure scale factor is the same in both dimensions.
+      if (scale_factor * layer.height != codec_.height) {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+      }
+
+      // Ensure scale factor is power of two.
+      const bool is_pow_of_two = (scale_factor & (scale_factor - 1)) == 0;
+      if (!is_pow_of_two) {
+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+      }
+
+      svc_params_.scaling_factor_num[i] = 1;
+      svc_params_.scaling_factor_den[i] = scale_factor;
+
+      RTC_DCHECK_GT(codec_.spatialLayers[i].maxFramerate, 0);
+      RTC_DCHECK_LE(codec_.spatialLayers[i].maxFramerate, codec_.maxFramerate);
+      if (i > 0) {
+        // Frame rate of high spatial layer is supposed to be equal or higher
+        // than frame rate of low spatial layer.
+        RTC_DCHECK_GE(codec_.spatialLayers[i].maxFramerate,
+                      codec_.spatialLayers[i - 1].maxFramerate);
+      }
+    }
+  } else {
+    int scaling_factor_num = 256;
+    for (int i = num_spatial_layers_ - 1; i >= 0; --i) {
+      // 1:2 scaling in each dimension.
+      svc_params_.scaling_factor_num[i] = scaling_factor_num;
+      svc_params_.scaling_factor_den[i] = 256;
+    }
+  }
+
+  UpdatePerformanceFlags();
+  RTC_DCHECK_EQ(performance_flags_by_spatial_index_.size(),
+                static_cast<size_t>(num_spatial_layers_));
+
+  SvcRateAllocator init_allocator(codec_);
+  current_bitrate_allocation_ =
+      init_allocator.Allocate(VideoBitrateAllocationParameters(
+          inst->startBitrate * 1000, inst->maxFramerate));
+  if (!SetSvcRates(current_bitrate_allocation_)) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  const vpx_codec_err_t rv = libvpx_->codec_enc_init(
+      encoder_, vpx_codec_vp9_cx(), config_,
+      config_->g_bit_depth == VPX_BITS_8 ? 0 : VPX_CODEC_USE_HIGHBITDEPTH);
+  if (rv != VPX_CODEC_OK) {
+    RTC_LOG(LS_ERROR) << "Init error: " << libvpx_->codec_err_to_string(rv);
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  if (performance_flags_.use_per_layer_speed) {
+    for (int si = 0; si < num_spatial_layers_; ++si) {
+      svc_params_.speed_per_layer[si] =
+          performance_flags_by_spatial_index_[si].base_layer_speed;
+      svc_params_.loopfilter_ctrl[si] =
+          performance_flags_by_spatial_index_[si].deblock_mode;
+    }
+    bool denoiser_on =
+        AllowDenoising() && inst->VP9().denoisingOn &&
+        performance_flags_by_spatial_index_[num_spatial_layers_ - 1]
+            .allow_denoising;
+    libvpx_->codec_control(encoder_, VP9E_SET_NOISE_SENSITIVITY,
+                           denoiser_on ? 1 : 0);
+  }
+
+  libvpx_->codec_control(encoder_, VP8E_SET_MAX_INTRA_BITRATE_PCT,
+                         rc_max_intra_target_);
+  libvpx_->codec_control(encoder_, VP9E_SET_AQ_MODE,
+                         inst->VP9().adaptiveQpMode ? 3 : 0);
+
+  libvpx_->codec_control(encoder_, VP9E_SET_FRAME_PARALLEL_DECODING, 0);
+  libvpx_->codec_control(encoder_, VP9E_SET_SVC_GF_TEMPORAL_REF, 0);
+
+  if (is_svc_) {
+    libvpx_->codec_control(encoder_, VP9E_SET_SVC, 1);
+    libvpx_->codec_control(encoder_, VP9E_SET_SVC_PARAMETERS, &svc_params_);
+  }
+  if (!is_svc_ || !performance_flags_.use_per_layer_speed) {
+    libvpx_->codec_control(
+        encoder_, VP8E_SET_CPUUSED,
+        performance_flags_by_spatial_index_.rbegin()->base_layer_speed);
+  }
+
+  if (num_spatial_layers_ > 1) {
+    switch (inter_layer_pred_) {
+      case InterLayerPredMode::kOn:
+        libvpx_->codec_control(encoder_, VP9E_SET_SVC_INTER_LAYER_PRED, 0);
+        break;
+      case InterLayerPredMode::kOff:
+        libvpx_->codec_control(encoder_, VP9E_SET_SVC_INTER_LAYER_PRED, 1);
+        break;
+      case InterLayerPredMode::kOnKeyPic:
+        libvpx_->codec_control(encoder_, VP9E_SET_SVC_INTER_LAYER_PRED, 2);
+        break;
+      default:
+        RTC_DCHECK_NOTREACHED();
+    }
+
+    memset(&svc_drop_frame_, 0, sizeof(svc_drop_frame_));
+    const bool reverse_constrained_drop_mode =
+        inter_layer_pred_ == InterLayerPredMode::kOn &&
+        codec_.mode == VideoCodecMode::kScreensharing &&
+        num_spatial_layers_ > 1;
+    if (reverse_constrained_drop_mode) {
+      // Screenshare dropping mode: drop a layer only together with all lower
+      // layers. This ensures that drops on lower layers won't reduce frame-rate
+      // for higher layers and reference structure is RTP-compatible.
+      svc_drop_frame_.framedrop_mode = CONSTRAINED_FROM_ABOVE_DROP;
+      svc_drop_frame_.max_consec_drop = 5;
+      for (size_t i = 0; i < num_spatial_layers_; ++i) {
+        svc_drop_frame_.framedrop_thresh[i] = config_->rc_dropframe_thresh;
+      }
+      // No buffering is needed because the highest layer is always present in
+      // all frames in CONSTRAINED_FROM_ABOVE drop mode.
+      layer_buffering_ = false;
+    } else {
+      // Configure encoder to drop entire superframe whenever it needs to drop
+      // a layer. This mode is preferred over per-layer dropping which causes
+      // quality flickering and is not compatible with RTP non-flexible mode.
+      svc_drop_frame_.framedrop_mode =
+          full_superframe_drop_ ? FULL_SUPERFRAME_DROP : CONSTRAINED_LAYER_DROP;
+      // Buffering is needed only for constrained layer drop, as it's not clear
+      // which frame is the last.
+      layer_buffering_ = !full_superframe_drop_;
+      svc_drop_frame_.max_consec_drop = std::numeric_limits<int>::max();
+      for (size_t i = 0; i < num_spatial_layers_; ++i) {
+        svc_drop_frame_.framedrop_thresh[i] = config_->rc_dropframe_thresh;
+      }
+    }
+    libvpx_->codec_control(encoder_, VP9E_SET_SVC_FRAME_DROP_LAYER,
+                           &svc_drop_frame_);
+  }
+
+  // Register callback for getting each spatial layer.
+  vpx_codec_priv_output_cx_pkt_cb_pair_t cbp = {
+      LibvpxVp9Encoder::EncoderOutputCodedPacketCallback,
+      reinterpret_cast<void*>(this)};
+  libvpx_->codec_control(encoder_, VP9E_REGISTER_CX_CALLBACK,
+                         reinterpret_cast<void*>(&cbp));
+
+  // Control function to set the number of column tiles in encoding a frame, in
+  // log2 unit: e.g., 0 = 1 tile column, 1 = 2 tile columns, 2 = 4 tile columns.
+  // The number tile columns will be capped by the encoder based on image size
+  // (minimum width of tile column is 256 pixels, maximum is 4096).
+  libvpx_->codec_control(encoder_, VP9E_SET_TILE_COLUMNS,
+                         static_cast<int>((config_->g_threads >> 1)));
+
+  // Turn on row-based multithreading.
+  libvpx_->codec_control(encoder_, VP9E_SET_ROW_MT, 1);
+
+  if (AllowDenoising() && !performance_flags_.use_per_layer_speed) {
+    libvpx_->codec_control(encoder_, VP9E_SET_NOISE_SENSITIVITY,
+                           inst->VP9().denoisingOn ? 1 : 0);
+  }
+
+  if (codec_.mode == VideoCodecMode::kScreensharing) {
+    // Adjust internal parameters to screen content.
+    libvpx_->codec_control(encoder_, VP9E_SET_TUNE_CONTENT, 1);
+  }
+  // Enable encoder skip of static/low content blocks.
+  libvpx_->codec_control(encoder_, VP8E_SET_STATIC_THRESHOLD, 1);
+  inited_ = true;
+  config_changed_ = true;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+uint32_t LibvpxVp9Encoder::MaxIntraTarget(uint32_t optimal_buffer_size) {
+  // Set max to the optimal buffer level (normalized by target BR),
+  // and scaled by a scale_par.
+  // Max target size = scale_par * optimal_buffer_size * targetBR[Kbps].
+  // This value is presented in percentage of perFrameBw:
+  // perFrameBw = targetBR[Kbps] * 1000 / framerate.
+  // The target in % is as follows:
+  float scale_par = 0.5;
+  uint32_t target_pct =
+      optimal_buffer_size * scale_par * codec_.maxFramerate / 10;
+  // Don't go below 3 times the per frame bandwidth.
+  const uint32_t min_intra_size = 300;
+  return (target_pct < min_intra_size) ? min_intra_size : target_pct;
+}
+
+int LibvpxVp9Encoder::Encode(const VideoFrame& input_image,
+                             const std::vector<VideoFrameType>* frame_types) {
+  if (!inited_) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (encoded_complete_callback_ == nullptr) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  if (num_active_spatial_layers_ == 0) {
+    // All spatial layers are disabled, return without encoding anything.
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  // We only support one stream at the moment.
+  if (frame_types && !frame_types->empty()) {
+    if ((*frame_types)[0] == VideoFrameType::kVideoFrameKey) {
+      force_key_frame_ = true;
+    }
+  }
+
+  if (pics_since_key_ + 1 ==
+      static_cast<size_t>(codec_.VP9()->keyFrameInterval)) {
+    force_key_frame_ = true;
+  }
+
+  if (svc_controller_) {
+    layer_frames_ = svc_controller_->NextFrameConfig(force_key_frame_);
+    if (layer_frames_.empty()) {
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    if (layer_frames_.front().IsKeyframe()) {
+      force_key_frame_ = true;
+    }
+  }
+
+  vpx_svc_layer_id_t layer_id = {0};
+  if (!force_key_frame_) {
+    const size_t gof_idx = (pics_since_key_ + 1) % gof_.num_frames_in_gof;
+    layer_id.temporal_layer_id = gof_.temporal_idx[gof_idx];
+
+    if (codec_.mode == VideoCodecMode::kScreensharing) {
+      const uint32_t frame_timestamp_ms =
+          1000 * input_image.timestamp() / kVideoPayloadTypeFrequency;
+
+      // To ensure that several rate-limiters with different limits don't
+      // interfere, they must be queried in order of increasing limit.
+
+      bool use_steady_state_limiter =
+          variable_framerate_experiment_.enabled &&
+          input_image.update_rect().IsEmpty() &&
+          num_steady_state_frames_ >=
+              variable_framerate_experiment_.frames_before_steady_state;
+
+      // Need to check all frame limiters, even if lower layers are disabled,
+      // because variable frame-rate limiter should be checked after the first
+      // layer. It's easier to overwrite active layers after, then check all
+      // cases.
+      for (uint8_t sl_idx = 0; sl_idx < num_active_spatial_layers_; ++sl_idx) {
+        const float layer_fps =
+            framerate_controller_[layer_id.spatial_layer_id].GetTargetRate();
+        // Use steady state rate-limiter at the correct place.
+        if (use_steady_state_limiter &&
+            layer_fps > variable_framerate_experiment_.framerate_limit - 1e-9) {
+          if (variable_framerate_controller_.DropFrame(frame_timestamp_ms)) {
+            layer_id.spatial_layer_id = num_active_spatial_layers_;
+          }
+          // Break always: if rate limiter triggered frame drop, no need to
+          // continue; otherwise, the rate is less than the next limiters.
+          break;
+        }
+        if (framerate_controller_[sl_idx].DropFrame(frame_timestamp_ms)) {
+          ++layer_id.spatial_layer_id;
+        } else {
+          break;
+        }
+      }
+
+      if (use_steady_state_limiter &&
+          layer_id.spatial_layer_id < num_active_spatial_layers_) {
+        variable_framerate_controller_.AddFrame(frame_timestamp_ms);
+      }
+    }
+
+    if (force_all_active_layers_) {
+      layer_id.spatial_layer_id = first_active_layer_;
+      force_all_active_layers_ = false;
+    }
+
+    RTC_DCHECK_LE(layer_id.spatial_layer_id, num_active_spatial_layers_);
+    if (layer_id.spatial_layer_id >= num_active_spatial_layers_) {
+      // Drop entire picture.
+      return WEBRTC_VIDEO_CODEC_OK;
+    }
+  }
+
+  // Need to set temporal layer id on ALL layers, even disabled ones.
+  // Otherwise libvpx might produce frames on a disabled layer:
+  // http://crbug.com/1051476
+  for (int sl_idx = 0; sl_idx < num_spatial_layers_; ++sl_idx) {
+    layer_id.temporal_layer_id_per_spatial[sl_idx] = layer_id.temporal_layer_id;
+  }
+
+  if (layer_id.spatial_layer_id < first_active_layer_) {
+    layer_id.spatial_layer_id = first_active_layer_;
+  }
+
+  if (svc_controller_) {
+    layer_id.spatial_layer_id = layer_frames_.front().SpatialId();
+    layer_id.temporal_layer_id = layer_frames_.front().TemporalId();
+    for (const auto& layer : layer_frames_) {
+      layer_id.temporal_layer_id_per_spatial[layer.SpatialId()] =
+          layer.TemporalId();
+    }
+    SetActiveSpatialLayers();
+  }
+
+  if (is_svc_ && performance_flags_.use_per_layer_speed) {
+    // Update speed settings that might depend on temporal index.
+    bool speed_updated = false;
+    for (int sl_idx = 0; sl_idx < num_spatial_layers_; ++sl_idx) {
+      const int target_speed =
+          layer_id.temporal_layer_id_per_spatial[sl_idx] == 0
+              ? performance_flags_by_spatial_index_[sl_idx].base_layer_speed
+              : performance_flags_by_spatial_index_[sl_idx].high_layer_speed;
+      if (svc_params_.speed_per_layer[sl_idx] != target_speed) {
+        svc_params_.speed_per_layer[sl_idx] = target_speed;
+        speed_updated = true;
+      }
+    }
+    if (speed_updated) {
+      libvpx_->codec_control(encoder_, VP9E_SET_SVC_PARAMETERS, &svc_params_);
+    }
+  }
+
+  libvpx_->codec_control(encoder_, VP9E_SET_SVC_LAYER_ID, &layer_id);
+
+  if (num_spatial_layers_ > 1) {
+    // Update frame dropping settings as they may change on per-frame basis.
+    libvpx_->codec_control(encoder_, VP9E_SET_SVC_FRAME_DROP_LAYER,
+                           &svc_drop_frame_);
+  }
+
+  if (config_changed_) {
+    if (libvpx_->codec_enc_config_set(encoder_, config_)) {
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+
+    if (!performance_flags_.use_per_layer_speed) {
+      // Not setting individual speeds per layer, find the highest active
+      // resolution instead and base the speed on that.
+      for (int i = num_spatial_layers_ - 1; i >= 0; --i) {
+        if (config_->ss_target_bitrate[i] > 0) {
+          int width = (svc_params_.scaling_factor_num[i] * config_->g_w) /
+                      svc_params_.scaling_factor_den[i];
+          int height = (svc_params_.scaling_factor_num[i] * config_->g_h) /
+                       svc_params_.scaling_factor_den[i];
+          int speed =
+              std::prev(performance_flags_.settings_by_resolution.lower_bound(
+                            width * height))
+                  ->second.base_layer_speed;
+          libvpx_->codec_control(encoder_, VP8E_SET_CPUUSED, speed);
+          break;
+        }
+      }
+    }
+    config_changed_ = false;
+  }
+
+  if (input_image.width() != codec_.width ||
+      input_image.height() != codec_.height) {
+    int ret = UpdateCodecFrameSize(input_image);
+    if (ret < 0) {
+      return ret;
+    }
+  }
+
+  RTC_DCHECK_EQ(input_image.width(), raw_->d_w);
+  RTC_DCHECK_EQ(input_image.height(), raw_->d_h);
+
+  // Set input image for use in the callback.
+  // This was necessary since you need some information from input_image.
+  // You can save only the necessary information (such as timestamp) instead of
+  // doing this.
+  input_image_ = &input_image;
+
+  // In case we need to map the buffer, `mapped_buffer` is used to keep it alive
+  // through reference counting until after encoding has finished.
+  rtc::scoped_refptr<const VideoFrameBuffer> mapped_buffer;
+  const I010BufferInterface* i010_buffer;
+  rtc::scoped_refptr<const I010BufferInterface> i010_copy;
+  switch (profile_) {
+    case VP9Profile::kProfile0: {
+      mapped_buffer =
+          PrepareBufferForProfile0(input_image.video_frame_buffer());
+      if (!mapped_buffer) {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+      }
+      break;
+    }
+    case VP9Profile::kProfile1: {
+      RTC_DCHECK_NOTREACHED();
+      break;
+    }
+    case VP9Profile::kProfile2: {
+      // We can inject kI010 frames directly for encode. All other formats
+      // should be converted to it.
+      switch (input_image.video_frame_buffer()->type()) {
+        case VideoFrameBuffer::Type::kI010: {
+          i010_buffer = input_image.video_frame_buffer()->GetI010();
+          break;
+        }
+        default: {
+          auto i420_buffer = input_image.video_frame_buffer()->ToI420();
+          if (!i420_buffer) {
+            RTC_LOG(LS_ERROR) << "Failed to convert "
+                              << VideoFrameBufferTypeToString(
+                                     input_image.video_frame_buffer()->type())
+                              << " image to I420. Can't encode frame.";
+            return WEBRTC_VIDEO_CODEC_ERROR;
+          }
+          i010_copy = I010Buffer::Copy(*i420_buffer);
+          i010_buffer = i010_copy.get();
+        }
+      }
+      raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(
+          reinterpret_cast<const uint8_t*>(i010_buffer->DataY()));
+      raw_->planes[VPX_PLANE_U] = const_cast<uint8_t*>(
+          reinterpret_cast<const uint8_t*>(i010_buffer->DataU()));
+      raw_->planes[VPX_PLANE_V] = const_cast<uint8_t*>(
+          reinterpret_cast<const uint8_t*>(i010_buffer->DataV()));
+      raw_->stride[VPX_PLANE_Y] = i010_buffer->StrideY() * 2;
+      raw_->stride[VPX_PLANE_U] = i010_buffer->StrideU() * 2;
+      raw_->stride[VPX_PLANE_V] = i010_buffer->StrideV() * 2;
+      break;
+    }
+    case VP9Profile::kProfile3: {
+      RTC_DCHECK_NOTREACHED();
+      break;
+    }
+  }
+
+  vpx_enc_frame_flags_t flags = 0;
+  if (force_key_frame_) {
+    flags = VPX_EFLAG_FORCE_KF;
+  }
+
+  if (svc_controller_) {
+    vpx_svc_ref_frame_config_t ref_config = Vp9References(layer_frames_);
+    libvpx_->codec_control(encoder_, VP9E_SET_SVC_REF_FRAME_CONFIG,
+                           &ref_config);
+  } else if (external_ref_control_) {
+    vpx_svc_ref_frame_config_t ref_config =
+        SetReferences(force_key_frame_, layer_id.spatial_layer_id);
+
+    if (VideoCodecMode::kScreensharing == codec_.mode) {
+      for (uint8_t sl_idx = 0; sl_idx < num_active_spatial_layers_; ++sl_idx) {
+        ref_config.duration[sl_idx] = static_cast<int64_t>(
+            90000 / (std::min(static_cast<float>(codec_.maxFramerate),
+                              framerate_controller_[sl_idx].GetTargetRate())));
+      }
+    }
+
+    libvpx_->codec_control(encoder_, VP9E_SET_SVC_REF_FRAME_CONFIG,
+                           &ref_config);
+  }
+
+  first_frame_in_picture_ = true;
+
+  // TODO(ssilkin): Frame duration should be specified per spatial layer
+  // since their frame rate can be different. For now calculate frame duration
+  // based on target frame rate of the highest spatial layer, which frame rate
+  // is supposed to be equal or higher than frame rate of low spatial layers.
+  // Also, timestamp should represent actual time passed since previous frame
+  // (not 'expected' time). Then rate controller can drain buffer more
+  // accurately.
+  RTC_DCHECK_GE(framerate_controller_.size(), num_active_spatial_layers_);
+  float target_framerate_fps =
+      (codec_.mode == VideoCodecMode::kScreensharing)
+          ? std::min(static_cast<float>(codec_.maxFramerate),
+                     framerate_controller_[num_active_spatial_layers_ - 1]
+                         .GetTargetRate())
+          : codec_.maxFramerate;
+  uint32_t duration = static_cast<uint32_t>(90000 / target_framerate_fps);
+  const vpx_codec_err_t rv = libvpx_->codec_encode(
+      encoder_, raw_, timestamp_, duration, flags, VPX_DL_REALTIME);
+  if (rv != VPX_CODEC_OK) {
+    RTC_LOG(LS_ERROR) << "Encoding error: " << libvpx_->codec_err_to_string(rv)
+                      << "\n"
+                         "Details: "
+                      << libvpx_->codec_error(encoder_) << "\n"
+                      << libvpx_->codec_error_detail(encoder_);
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  timestamp_ += duration;
+
+  if (layer_buffering_) {
+    const bool end_of_picture = true;
+    DeliverBufferedFrame(end_of_picture);
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int LibvpxVp9Encoder::UpdateCodecFrameSize(
+    const VideoFrame& input_image) {
+  RTC_LOG(LS_INFO) << "Reconfiging VP from " <<
+          codec_.width << "x" << codec_.height << " to " <<
+          input_image.width() << "x" << input_image.height();
+  // Preserve latest bitrate/framerate setting
+  // TODO: Mozilla - see below, we need to save more state here.
+  //uint32_t old_bitrate_kbit = config_->rc_target_bitrate;
+  //uint32_t old_framerate = codec_.maxFramerate;
+
+  codec_.width = input_image.width();
+  codec_.height = input_image.height();
+
+  vpx_img_free(raw_);
+  raw_ = vpx_img_wrap(NULL, VPX_IMG_FMT_I420, codec_.width, codec_.height,
+                      1, NULL);
+  // Update encoder context for new frame size.
+  config_->g_w = codec_.width;
+  config_->g_h = codec_.height;
+
+  // Determine number of threads based on the image size and #cores.
+  config_->g_threads = NumberOfThreads(codec_.width, codec_.height,
+                                       num_cores_);
+
+  // NOTE: We would like to do this the same way vp8 does it
+  // (with vpx_codec_enc_config_set()), but that causes asserts
+  // in AQ 3 (cyclic); and in AQ 0 it works, but on a resize to smaller
+  // than 1/2 x 1/2 original it asserts in convolve().  Given these
+  // bugs in trying to do it the "right" way, we basically re-do
+  // the initialization.
+  vpx_codec_destroy(encoder_); // clean up old state
+  int result = InitAndSetControlSettings(&codec_);
+  if (result == WEBRTC_VIDEO_CODEC_OK) {
+    // TODO: Mozilla rates have become much more complicated, we need to store
+    // more state or find another way of doing this.
+    //return SetRates(old_bitrate_kbit, old_framerate);
+    RTC_CHECK(false);
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  return result;
+}
+
+bool LibvpxVp9Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
+                                             absl::optional<int>* spatial_idx,
+                                             absl::optional<int>* temporal_idx,
+                                             const vpx_codec_cx_pkt& pkt) {
+  RTC_CHECK(codec_specific != nullptr);
+  codec_specific->codecType = kVideoCodecVP9;
+  CodecSpecificInfoVP9* vp9_info = &(codec_specific->codecSpecific.VP9);
+
+  vp9_info->first_frame_in_picture = first_frame_in_picture_;
+  vp9_info->flexible_mode = is_flexible_mode_;
+
+  if (pkt.data.frame.flags & VPX_FRAME_IS_KEY) {
+    pics_since_key_ = 0;
+  } else if (first_frame_in_picture_) {
+    ++pics_since_key_;
+  }
+
+  vpx_svc_layer_id_t layer_id = {0};
+  libvpx_->codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
+
+  // Can't have keyframe with non-zero temporal layer.
+  RTC_DCHECK(pics_since_key_ != 0 || layer_id.temporal_layer_id == 0);
+
+  RTC_CHECK_GT(num_temporal_layers_, 0);
+  RTC_CHECK_GT(num_active_spatial_layers_, 0);
+  if (num_temporal_layers_ == 1) {
+    RTC_CHECK_EQ(layer_id.temporal_layer_id, 0);
+    vp9_info->temporal_idx = kNoTemporalIdx;
+    *temporal_idx = absl::nullopt;
+  } else {
+    vp9_info->temporal_idx = layer_id.temporal_layer_id;
+    *temporal_idx = layer_id.temporal_layer_id;
+  }
+  if (num_active_spatial_layers_ == 1) {
+    RTC_CHECK_EQ(layer_id.spatial_layer_id, 0);
+    *spatial_idx = absl::nullopt;
+  } else {
+    *spatial_idx = layer_id.spatial_layer_id;
+  }
+
+  const bool is_key_pic = (pics_since_key_ == 0);
+  const bool is_inter_layer_pred_allowed =
+      (inter_layer_pred_ == InterLayerPredMode::kOn ||
+       (inter_layer_pred_ == InterLayerPredMode::kOnKeyPic && is_key_pic));
+
+  // Always set inter_layer_predicted to true on high layer frame if inter-layer
+  // prediction (ILP) is allowed even if encoder didn't actually use it.
+  // Setting inter_layer_predicted to false would allow receiver to decode high
+  // layer frame without decoding low layer frame. If that would happen (e.g.
+  // if low layer frame is lost) then receiver won't be able to decode next high
+  // layer frame which uses ILP.
+  vp9_info->inter_layer_predicted =
+      first_frame_in_picture_ ? false : is_inter_layer_pred_allowed;
+
+  // Mark all low spatial layer frames as references (not just frames of
+  // active low spatial layers) if inter-layer prediction is enabled since
+  // these frames are indirect references of high spatial layer, which can
+  // later be enabled without key frame.
+  vp9_info->non_ref_for_inter_layer_pred =
+      !is_inter_layer_pred_allowed ||
+      layer_id.spatial_layer_id + 1 == num_spatial_layers_;
+
+  // Always populate this, so that the packetizer can properly set the marker
+  // bit.
+  vp9_info->num_spatial_layers = num_active_spatial_layers_;
+  vp9_info->first_active_layer = first_active_layer_;
+
+  vp9_info->num_ref_pics = 0;
+  FillReferenceIndices(pkt, pics_since_key_, vp9_info->inter_layer_predicted,
+                       vp9_info);
+  if (vp9_info->flexible_mode) {
+    vp9_info->gof_idx = kNoGofIdx;
+    if (!svc_controller_) {
+      if (num_temporal_layers_ == 1) {
+        vp9_info->temporal_up_switch = true;
+      } else {
+        // In flexible mode with > 1 temporal layer but no SVC controller we
+        // can't techincally determine if a frame is an upswitch point, use
+        // gof-based data as proxy for now.
+        // TODO(sprang): Remove once SVC controller is the only choice.
+        vp9_info->gof_idx =
+            static_cast<uint8_t>(pics_since_key_ % gof_.num_frames_in_gof);
+        vp9_info->temporal_up_switch =
+            gof_.temporal_up_switch[vp9_info->gof_idx];
+      }
+    }
+  } else {
+    vp9_info->gof_idx =
+        static_cast<uint8_t>(pics_since_key_ % gof_.num_frames_in_gof);
+    vp9_info->temporal_up_switch = gof_.temporal_up_switch[vp9_info->gof_idx];
+    RTC_DCHECK(vp9_info->num_ref_pics == gof_.num_ref_pics[vp9_info->gof_idx] ||
+               vp9_info->num_ref_pics == 0);
+  }
+
+  vp9_info->inter_pic_predicted = (!is_key_pic && vp9_info->num_ref_pics > 0);
+
+  // Write SS on key frame of independently coded spatial layers and on base
+  // temporal/spatial layer frame if number of layers changed without issuing
+  // of key picture (inter-layer prediction is enabled).
+  const bool is_key_frame = is_key_pic && !vp9_info->inter_layer_predicted;
+  if (is_key_frame || (ss_info_needed_ && layer_id.temporal_layer_id == 0 &&
+                       layer_id.spatial_layer_id == first_active_layer_)) {
+    vp9_info->ss_data_available = true;
+    vp9_info->spatial_layer_resolution_present = true;
+    // Signal disabled layers.
+    for (size_t i = 0; i < first_active_layer_; ++i) {
+      vp9_info->width[i] = 0;
+      vp9_info->height[i] = 0;
+    }
+    for (size_t i = first_active_layer_; i < num_active_spatial_layers_; ++i) {
+      vp9_info->width[i] = codec_.width * svc_params_.scaling_factor_num[i] /
+                           svc_params_.scaling_factor_den[i];
+      vp9_info->height[i] = codec_.height * svc_params_.scaling_factor_num[i] /
+                            svc_params_.scaling_factor_den[i];
+    }
+    if (vp9_info->flexible_mode) {
+      vp9_info->gof.num_frames_in_gof = 0;
+    } else {
+      vp9_info->gof.CopyGofInfoVP9(gof_);
+    }
+
+    ss_info_needed_ = false;
+  } else {
+    vp9_info->ss_data_available = false;
+  }
+
+  first_frame_in_picture_ = false;
+
+  // Populate codec-agnostic section in the codec specific structure.
+  if (svc_controller_) {
+    auto it = absl::c_find_if(
+        layer_frames_,
+        [&](const ScalableVideoController::LayerFrameConfig& config) {
+          return config.SpatialId() == layer_id.spatial_layer_id;
+        });
+    if (it == layer_frames_.end()) {
+      RTC_LOG(LS_ERROR) << "Encoder produced a frame for layer S"
+                        << layer_id.spatial_layer_id << "T"
+                        << layer_id.temporal_layer_id
+                        << " that wasn't requested.";
+      return false;
+    }
+    codec_specific->generic_frame_info = svc_controller_->OnEncodeDone(*it);
+    if (is_key_frame) {
+      codec_specific->template_structure =
+          svc_controller_->DependencyStructure();
+      auto& resolutions = codec_specific->template_structure->resolutions;
+      resolutions.resize(num_spatial_layers_);
+      for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+        resolutions[sid] = RenderResolution(
+            /*width=*/codec_.width * svc_params_.scaling_factor_num[sid] /
+                svc_params_.scaling_factor_den[sid],
+            /*height=*/codec_.height * svc_params_.scaling_factor_num[sid] /
+                svc_params_.scaling_factor_den[sid]);
+      }
+    }
+    if (is_flexible_mode_) {
+      // Populate data for legacy temporal-upswitch state.
+      // We can switch up to a higher temporal layer only if all temporal layers
+      // higher than this (within the current spatial layer) are switch points.
+      vp9_info->temporal_up_switch = true;
+      for (int i = layer_id.temporal_layer_id + 1; i < num_temporal_layers_;
+           ++i) {
+        // Assumes decode targets are always ordered first by spatial then by
+        // temporal id.
+        size_t dti_index =
+            (layer_id.spatial_layer_id * num_temporal_layers_) + i;
+        vp9_info->temporal_up_switch &=
+            (codec_specific->generic_frame_info
+                 ->decode_target_indications[dti_index] ==
+             DecodeTargetIndication::kSwitch);
+      }
+    }
+  }
+  codec_specific->scalability_mode = scalability_mode_;
+  return true;
+}
+
+void LibvpxVp9Encoder::FillReferenceIndices(const vpx_codec_cx_pkt& pkt,
+                                            const size_t pic_num,
+                                            const bool inter_layer_predicted,
+                                            CodecSpecificInfoVP9* vp9_info) {
+  vpx_svc_layer_id_t layer_id = {0};
+  libvpx_->codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
+
+  const bool is_key_frame =
+      (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ? true : false;
+
+  std::vector<RefFrameBuffer> ref_buf_list;
+
+  if (is_svc_) {
+    vpx_svc_ref_frame_config_t enc_layer_conf = {{0}};
+    libvpx_->codec_control(encoder_, VP9E_GET_SVC_REF_FRAME_CONFIG,
+                           &enc_layer_conf);
+    char ref_buf_flags[] = "00000000";
+    // There should be one character per buffer + 1 termination '\0'.
+    static_assert(sizeof(ref_buf_flags) == kNumVp9Buffers + 1);
+
+    if (enc_layer_conf.reference_last[layer_id.spatial_layer_id]) {
+      const size_t fb_idx =
+          enc_layer_conf.lst_fb_idx[layer_id.spatial_layer_id];
+      RTC_DCHECK_LT(fb_idx, ref_buf_.size());
+      if (std::find(ref_buf_list.begin(), ref_buf_list.end(),
+                    ref_buf_[fb_idx]) == ref_buf_list.end()) {
+        ref_buf_list.push_back(ref_buf_[fb_idx]);
+        ref_buf_flags[fb_idx] = '1';
+      }
+    }
+
+    if (enc_layer_conf.reference_alt_ref[layer_id.spatial_layer_id]) {
+      const size_t fb_idx =
+          enc_layer_conf.alt_fb_idx[layer_id.spatial_layer_id];
+      RTC_DCHECK_LT(fb_idx, ref_buf_.size());
+      if (std::find(ref_buf_list.begin(), ref_buf_list.end(),
+                    ref_buf_[fb_idx]) == ref_buf_list.end()) {
+        ref_buf_list.push_back(ref_buf_[fb_idx]);
+        ref_buf_flags[fb_idx] = '1';
+      }
+    }
+
+    if (enc_layer_conf.reference_golden[layer_id.spatial_layer_id]) {
+      const size_t fb_idx =
+          enc_layer_conf.gld_fb_idx[layer_id.spatial_layer_id];
+      RTC_DCHECK_LT(fb_idx, ref_buf_.size());
+      if (std::find(ref_buf_list.begin(), ref_buf_list.end(),
+                    ref_buf_[fb_idx]) == ref_buf_list.end()) {
+        ref_buf_list.push_back(ref_buf_[fb_idx]);
+        ref_buf_flags[fb_idx] = '1';
+      }
+    }
+
+    RTC_LOG(LS_VERBOSE) << "Frame " << pic_num << " sl "
+                        << layer_id.spatial_layer_id << " tl "
+                        << layer_id.temporal_layer_id << " refered buffers "
+                        << ref_buf_flags;
+
+  } else if (!is_key_frame) {
+    RTC_DCHECK_EQ(num_spatial_layers_, 1);
+    RTC_DCHECK_EQ(num_temporal_layers_, 1);
+    // In non-SVC mode encoder doesn't provide reference list. Assume each frame
+    // refers previous one, which is stored in buffer 0.
+    ref_buf_list.push_back(ref_buf_[0]);
+  }
+
+  std::vector<size_t> ref_pid_list;
+
+  vp9_info->num_ref_pics = 0;
+  for (const RefFrameBuffer& ref_buf : ref_buf_list) {
+    RTC_DCHECK_LE(ref_buf.pic_num, pic_num);
+    if (ref_buf.pic_num < pic_num) {
+      if (inter_layer_pred_ != InterLayerPredMode::kOn) {
+        // RTP spec limits temporal prediction to the same spatial layer.
+        // It is safe to ignore this requirement if inter-layer prediction is
+        // enabled for all frames when all base frames are relayed to receiver.
+        RTC_DCHECK_EQ(ref_buf.spatial_layer_id, layer_id.spatial_layer_id);
+      } else {
+        RTC_DCHECK_LE(ref_buf.spatial_layer_id, layer_id.spatial_layer_id);
+      }
+      RTC_DCHECK_LE(ref_buf.temporal_layer_id, layer_id.temporal_layer_id);
+
+      // Encoder may reference several spatial layers on the same previous
+      // frame in case if some spatial layers are skipped on the current frame.
+      // We shouldn't put duplicate references as it may break some old
+      // clients and isn't RTP compatible.
+      if (std::find(ref_pid_list.begin(), ref_pid_list.end(),
+                    ref_buf.pic_num) != ref_pid_list.end()) {
+        continue;
+      }
+      ref_pid_list.push_back(ref_buf.pic_num);
+
+      const size_t p_diff = pic_num - ref_buf.pic_num;
+      RTC_DCHECK_LE(p_diff, 127UL);
+
+      vp9_info->p_diff[vp9_info->num_ref_pics] = static_cast<uint8_t>(p_diff);
+      ++vp9_info->num_ref_pics;
+    } else {
+      RTC_DCHECK(inter_layer_predicted);
+      // RTP spec only allows to use previous spatial layer for inter-layer
+      // prediction.
+      RTC_DCHECK_EQ(ref_buf.spatial_layer_id + 1, layer_id.spatial_layer_id);
+    }
+  }
+}
+
+void LibvpxVp9Encoder::UpdateReferenceBuffers(const vpx_codec_cx_pkt& pkt,
+                                              const size_t pic_num) {
+  vpx_svc_layer_id_t layer_id = {0};
+  libvpx_->codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
+
+  RefFrameBuffer frame_buf = {.pic_num = pic_num,
+                              .spatial_layer_id = layer_id.spatial_layer_id,
+                              .temporal_layer_id = layer_id.temporal_layer_id};
+
+  if (is_svc_) {
+    vpx_svc_ref_frame_config_t enc_layer_conf = {{0}};
+    libvpx_->codec_control(encoder_, VP9E_GET_SVC_REF_FRAME_CONFIG,
+                           &enc_layer_conf);
+    const int update_buffer_slot =
+        enc_layer_conf.update_buffer_slot[layer_id.spatial_layer_id];
+
+    for (size_t i = 0; i < ref_buf_.size(); ++i) {
+      if (update_buffer_slot & (1 << i)) {
+        ref_buf_[i] = frame_buf;
+      }
+    }
+
+    RTC_LOG(LS_VERBOSE) << "Frame " << pic_num << " sl "
+                        << layer_id.spatial_layer_id << " tl "
+                        << layer_id.temporal_layer_id << " updated buffers "
+                        << (update_buffer_slot & (1 << 0) ? 1 : 0)
+                        << (update_buffer_slot & (1 << 1) ? 1 : 0)
+                        << (update_buffer_slot & (1 << 2) ? 1 : 0)
+                        << (update_buffer_slot & (1 << 3) ? 1 : 0)
+                        << (update_buffer_slot & (1 << 4) ? 1 : 0)
+                        << (update_buffer_slot & (1 << 5) ? 1 : 0)
+                        << (update_buffer_slot & (1 << 6) ? 1 : 0)
+                        << (update_buffer_slot & (1 << 7) ? 1 : 0);
+  } else {
+    RTC_DCHECK_EQ(num_spatial_layers_, 1);
+    RTC_DCHECK_EQ(num_temporal_layers_, 1);
+    // In non-svc mode encoder doesn't provide reference list. Assume each frame
+    // is reference and stored in buffer 0.
+    ref_buf_[0] = frame_buf;
+  }
+}
+
+vpx_svc_ref_frame_config_t LibvpxVp9Encoder::SetReferences(
+    bool is_key_pic,
+    int first_active_spatial_layer_id) {
+  // kRefBufIdx, kUpdBufIdx need to be updated to support longer GOFs.
+  RTC_DCHECK_LE(gof_.num_frames_in_gof, 4);
+
+  vpx_svc_ref_frame_config_t ref_config;
+  memset(&ref_config, 0, sizeof(ref_config));
+
+  const size_t num_temporal_refs = std::max(1, num_temporal_layers_ - 1);
+  const bool is_inter_layer_pred_allowed =
+      inter_layer_pred_ == InterLayerPredMode::kOn ||
+      (inter_layer_pred_ == InterLayerPredMode::kOnKeyPic && is_key_pic);
+  absl::optional<int> last_updated_buf_idx;
+
+  // Put temporal reference to LAST and spatial reference to GOLDEN. Update
+  // frame buffer (i.e. store encoded frame) if current frame is a temporal
+  // reference (i.e. it belongs to a low temporal layer) or it is a spatial
+  // reference. In later case, always store spatial reference in the last
+  // reference frame buffer.
+  // For the case of 3 temporal and 3 spatial layers we need 6 frame buffers
+  // for temporal references plus 1 buffer for spatial reference. 7 buffers
+  // in total.
+
+  for (int sl_idx = first_active_spatial_layer_id;
+       sl_idx < num_active_spatial_layers_; ++sl_idx) {
+    const size_t curr_pic_num = is_key_pic ? 0 : pics_since_key_ + 1;
+    const size_t gof_idx = curr_pic_num % gof_.num_frames_in_gof;
+
+    if (!is_key_pic) {
+      // Set up temporal reference.
+      const int buf_idx = sl_idx * num_temporal_refs + kRefBufIdx[gof_idx];
+
+      // Last reference frame buffer is reserved for spatial reference. It is
+      // not supposed to be used for temporal prediction.
+      RTC_DCHECK_LT(buf_idx, kNumVp9Buffers - 1);
+
+      const int pid_diff = curr_pic_num - ref_buf_[buf_idx].pic_num;
+      // Incorrect spatial layer may be in the buffer due to a key-frame.
+      const bool same_spatial_layer =
+          ref_buf_[buf_idx].spatial_layer_id == sl_idx;
+      bool correct_pid = false;
+      if (is_flexible_mode_) {
+        correct_pid = pid_diff > 0 && pid_diff < kMaxAllowedPidDiff;
+      } else {
+        // Below code assumes single temporal referecence.
+        RTC_DCHECK_EQ(gof_.num_ref_pics[gof_idx], 1);
+        correct_pid = pid_diff == gof_.pid_diff[gof_idx][0];
+      }
+
+      if (same_spatial_layer && correct_pid) {
+        ref_config.lst_fb_idx[sl_idx] = buf_idx;
+        ref_config.reference_last[sl_idx] = 1;
+      } else {
+        // This reference doesn't match with one specified by GOF. This can
+        // only happen if spatial layer is enabled dynamically without key
+        // frame. Spatial prediction is supposed to be enabled in this case.
+        RTC_DCHECK(is_inter_layer_pred_allowed &&
+                   sl_idx > first_active_spatial_layer_id);
+      }
+    }
+
+    if (is_inter_layer_pred_allowed && sl_idx > first_active_spatial_layer_id) {
+      // Set up spatial reference.
+      RTC_DCHECK(last_updated_buf_idx);
+      ref_config.gld_fb_idx[sl_idx] = *last_updated_buf_idx;
+      ref_config.reference_golden[sl_idx] = 1;
+    } else {
+      RTC_DCHECK(ref_config.reference_last[sl_idx] != 0 ||
+                 sl_idx == first_active_spatial_layer_id ||
+                 inter_layer_pred_ == InterLayerPredMode::kOff);
+    }
+
+    last_updated_buf_idx.reset();
+
+    if (gof_.temporal_idx[gof_idx] < num_temporal_layers_ - 1 ||
+        num_temporal_layers_ == 1) {
+      last_updated_buf_idx = sl_idx * num_temporal_refs + kUpdBufIdx[gof_idx];
+
+      // Ensure last frame buffer is not used for temporal prediction (it is
+      // reserved for spatial reference).
+      RTC_DCHECK_LT(*last_updated_buf_idx, kNumVp9Buffers - 1);
+    } else if (is_inter_layer_pred_allowed) {
+      last_updated_buf_idx = kNumVp9Buffers - 1;
+    }
+
+    if (last_updated_buf_idx) {
+      ref_config.update_buffer_slot[sl_idx] = 1 << *last_updated_buf_idx;
+    }
+  }
+
+  return ref_config;
+}
+
+void LibvpxVp9Encoder::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
+  RTC_DCHECK_EQ(pkt->kind, VPX_CODEC_CX_FRAME_PKT);
+
+  if (pkt->data.frame.sz == 0) {
+    // Ignore dropped frame.
+    return;
+  }
+
+  vpx_svc_layer_id_t layer_id = {0};
+  libvpx_->codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
+
+  if (layer_buffering_) {
+    // Deliver buffered low spatial layer frame.
+    const bool end_of_picture = false;
+    DeliverBufferedFrame(end_of_picture);
+  }
+
+  encoded_image_.SetEncodedData(EncodedImageBuffer::Create(
+      static_cast<const uint8_t*>(pkt->data.frame.buf), pkt->data.frame.sz));
+
+  codec_specific_ = {};
+  absl::optional<int> spatial_index;
+  absl::optional<int> temporal_index;
+  if (!PopulateCodecSpecific(&codec_specific_, &spatial_index, &temporal_index,
+                             *pkt)) {
+    // Drop the frame.
+    encoded_image_.set_size(0);
+    return;
+  }
+  encoded_image_.SetSpatialIndex(spatial_index);
+  encoded_image_.SetTemporalIndex(temporal_index);
+
+  const bool is_key_frame =
+      ((pkt->data.frame.flags & VPX_FRAME_IS_KEY) ? true : false) &&
+      !codec_specific_.codecSpecific.VP9.inter_layer_predicted;
+
+  // Ensure encoder issued key frame on request.
+  RTC_DCHECK(is_key_frame || !force_key_frame_);
+
+  // Check if encoded frame is a key frame.
+  encoded_image_._frameType = VideoFrameType::kVideoFrameDelta;
+  if (is_key_frame) {
+    encoded_image_._frameType = VideoFrameType::kVideoFrameKey;
+    force_key_frame_ = false;
+  }
+
+  UpdateReferenceBuffers(*pkt, pics_since_key_);
+
+  TRACE_COUNTER1("webrtc", "EncodedFrameSize", encoded_image_.size());
+  encoded_image_.SetTimestamp(input_image_->timestamp());
+  encoded_image_.SetColorSpace(input_image_->color_space());
+  encoded_image_._encodedHeight =
+      pkt->data.frame.height[layer_id.spatial_layer_id];
+  encoded_image_._encodedWidth =
+      pkt->data.frame.width[layer_id.spatial_layer_id];
+  int qp = -1;
+  libvpx_->codec_control(encoder_, VP8E_GET_LAST_QUANTIZER, &qp);
+  encoded_image_.qp_ = qp;
+
+  if (!layer_buffering_) {
+    const bool end_of_picture = encoded_image_.SpatialIndex().value_or(0) + 1 ==
+                                num_active_spatial_layers_;
+    DeliverBufferedFrame(end_of_picture);
+  }
+}
+
+void LibvpxVp9Encoder::DeliverBufferedFrame(bool end_of_picture) {
+  if (encoded_image_.size() > 0) {
+    if (num_spatial_layers_ > 1) {
+      // Restore frame dropping settings, as dropping may be temporary forbidden
+      // due to dynamically enabled layers.
+      for (size_t i = 0; i < num_spatial_layers_; ++i) {
+        svc_drop_frame_.framedrop_thresh[i] = config_->rc_dropframe_thresh;
+      }
+    }
+
+    codec_specific_.end_of_picture = end_of_picture;
+
+    encoded_complete_callback_->OnEncodedImage(encoded_image_,
+                                               &codec_specific_);
+
+    if (codec_.mode == VideoCodecMode::kScreensharing) {
+      const uint8_t spatial_idx = encoded_image_.SpatialIndex().value_or(0);
+      const uint32_t frame_timestamp_ms =
+          1000 * encoded_image_.Timestamp() / kVideoPayloadTypeFrequency;
+      framerate_controller_[spatial_idx].AddFrame(frame_timestamp_ms);
+
+      const size_t steady_state_size = SteadyStateSize(
+          spatial_idx, codec_specific_.codecSpecific.VP9.temporal_idx);
+
+      // Only frames on spatial layers, which may be limited in a steady state
+      // are considered for steady state detection.
+      if (framerate_controller_[spatial_idx].GetTargetRate() >
+          variable_framerate_experiment_.framerate_limit + 1e-9) {
+        if (encoded_image_.qp_ <=
+                variable_framerate_experiment_.steady_state_qp &&
+            encoded_image_.size() <= steady_state_size) {
+          ++num_steady_state_frames_;
+        } else {
+          num_steady_state_frames_ = 0;
+        }
+      }
+    }
+    encoded_image_.set_size(0);
+  }
+}
+
+int LibvpxVp9Encoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  encoded_complete_callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+VideoEncoder::EncoderInfo LibvpxVp9Encoder::GetEncoderInfo() const {
+  EncoderInfo info;
+  info.supports_native_handle = false;
+  info.implementation_name = "libvpx";
+  if (quality_scaler_experiment_.enabled && inited_ &&
+      codec_.VP9().automaticResizeOn) {
+    info.scaling_settings = VideoEncoder::ScalingSettings(
+        quality_scaler_experiment_.low_qp, quality_scaler_experiment_.high_qp);
+  } else {
+    info.scaling_settings = VideoEncoder::ScalingSettings::kOff;
+  }
+  info.has_trusted_rate_controller = trusted_rate_controller_;
+  info.is_hardware_accelerated = false;
+  if (inited_) {
+    // Find the max configured fps of any active spatial layer.
+    float max_fps = 0.0;
+    for (size_t si = 0; si < num_spatial_layers_; ++si) {
+      if (codec_.spatialLayers[si].active &&
+          codec_.spatialLayers[si].maxFramerate > max_fps) {
+        max_fps = codec_.spatialLayers[si].maxFramerate;
+      }
+    }
+
+    for (size_t si = 0; si < num_spatial_layers_; ++si) {
+      info.fps_allocation[si].clear();
+      if (!codec_.spatialLayers[si].active) {
+        continue;
+      }
+
+      // This spatial layer may already use a fraction of the total frame rate.
+      const float sl_fps_fraction =
+          codec_.spatialLayers[si].maxFramerate / max_fps;
+      for (size_t ti = 0; ti < num_temporal_layers_; ++ti) {
+        const uint32_t decimator =
+            num_temporal_layers_ <= 1 ? 1 : config_->ts_rate_decimator[ti];
+        RTC_DCHECK_GT(decimator, 0);
+        info.fps_allocation[si].push_back(
+            rtc::saturated_cast<uint8_t>(EncoderInfo::kMaxFramerateFraction *
+                                         (sl_fps_fraction / decimator)));
+      }
+    }
+    if (profile_ == VP9Profile::kProfile0) {
+      info.preferred_pixel_formats = {VideoFrameBuffer::Type::kI420,
+                                      VideoFrameBuffer::Type::kNV12};
+    }
+  }
+  if (!encoder_info_override_.resolution_bitrate_limits().empty()) {
+    info.resolution_bitrate_limits =
+        encoder_info_override_.resolution_bitrate_limits();
+  }
+  return info;
+}
+
+size_t LibvpxVp9Encoder::SteadyStateSize(int sid, int tid) {
+  const size_t bitrate_bps = current_bitrate_allocation_.GetBitrate(
+      sid, tid == kNoTemporalIdx ? 0 : tid);
+  const float fps = (codec_.mode == VideoCodecMode::kScreensharing)
+                        ? std::min(static_cast<float>(codec_.maxFramerate),
+                                   framerate_controller_[sid].GetTargetRate())
+                        : codec_.maxFramerate;
+  return static_cast<size_t>(
+      bitrate_bps / (8 * fps) *
+          (100 -
+           variable_framerate_experiment_.steady_state_undershoot_percentage) /
+          100 +
+      0.5);
+}
+
+// static
+LibvpxVp9Encoder::VariableFramerateExperiment
+LibvpxVp9Encoder::ParseVariableFramerateConfig(const FieldTrialsView& trials) {
+  FieldTrialFlag enabled = FieldTrialFlag("Enabled");
+  FieldTrialParameter<double> framerate_limit("min_fps", 5.0);
+  FieldTrialParameter<int> qp("min_qp", 32);
+  FieldTrialParameter<int> undershoot_percentage("undershoot", 30);
+  FieldTrialParameter<int> frames_before_steady_state(
+      "frames_before_steady_state", 5);
+  ParseFieldTrial({&enabled, &framerate_limit, &qp, &undershoot_percentage,
+                   &frames_before_steady_state},
+                  trials.Lookup("WebRTC-VP9VariableFramerateScreenshare"));
+  VariableFramerateExperiment config;
+  config.enabled = enabled.Get();
+  config.framerate_limit = framerate_limit.Get();
+  config.steady_state_qp = qp.Get();
+  config.steady_state_undershoot_percentage = undershoot_percentage.Get();
+  config.frames_before_steady_state = frames_before_steady_state.Get();
+
+  return config;
+}
+
+// static
+LibvpxVp9Encoder::QualityScalerExperiment
+LibvpxVp9Encoder::ParseQualityScalerConfig(const FieldTrialsView& trials) {
+  FieldTrialFlag disabled = FieldTrialFlag("Disabled");
+  FieldTrialParameter<int> low_qp("low_qp", kLowVp9QpThreshold);
+  FieldTrialParameter<int> high_qp("hihg_qp", kHighVp9QpThreshold);
+  ParseFieldTrial({&disabled, &low_qp, &high_qp},
+                  trials.Lookup("WebRTC-VP9QualityScaler"));
+  QualityScalerExperiment config;
+  config.enabled = !disabled.Get();
+  RTC_LOG(LS_INFO) << "Webrtc quality scaler for vp9 is "
+                   << (config.enabled ? "enabled." : "disabled");
+  config.low_qp = low_qp.Get();
+  config.high_qp = high_qp.Get();
+
+  return config;
+}
+
+void LibvpxVp9Encoder::UpdatePerformanceFlags() {
+  flat_map<int, PerformanceFlags::ParameterSet> params_by_resolution;
+  if (codec_.GetVideoEncoderComplexity() ==
+      VideoCodecComplexity::kComplexityLow) {
+    // For low tier devices, always use speed 9. Only disable upper
+    // layer deblocking below QCIF.
+    params_by_resolution[0] = {.base_layer_speed = 9,
+                               .high_layer_speed = 9,
+                               .deblock_mode = 1,
+                               .allow_denoising = true};
+    params_by_resolution[352 * 288] = {.base_layer_speed = 9,
+                                       .high_layer_speed = 9,
+                                       .deblock_mode = 0,
+                                       .allow_denoising = true};
+  } else {
+    params_by_resolution = performance_flags_.settings_by_resolution;
+  }
+
+  const auto find_speed = [&](int min_pixel_count) {
+    RTC_DCHECK(!params_by_resolution.empty());
+    auto it = params_by_resolution.upper_bound(min_pixel_count);
+    return std::prev(it)->second;
+  };
+  performance_flags_by_spatial_index_.clear();
+
+  if (is_svc_) {
+    for (int si = 0; si < num_spatial_layers_; ++si) {
+      performance_flags_by_spatial_index_.push_back(find_speed(
+          codec_.spatialLayers[si].width * codec_.spatialLayers[si].height));
+    }
+  } else {
+    performance_flags_by_spatial_index_.push_back(
+        find_speed(codec_.width * codec_.height));
+  }
+}
+
+// static
+LibvpxVp9Encoder::PerformanceFlags
+LibvpxVp9Encoder::ParsePerformanceFlagsFromTrials(
+    const FieldTrialsView& trials) {
+  struct Params : public PerformanceFlags::ParameterSet {
+    int min_pixel_count = 0;
+  };
+
+  FieldTrialStructList<Params> trials_list(
+      {FieldTrialStructMember("min_pixel_count",
+                              [](Params* p) { return &p->min_pixel_count; }),
+       FieldTrialStructMember("high_layer_speed",
+                              [](Params* p) { return &p->high_layer_speed; }),
+       FieldTrialStructMember("base_layer_speed",
+                              [](Params* p) { return &p->base_layer_speed; }),
+       FieldTrialStructMember("deblock_mode",
+                              [](Params* p) { return &p->deblock_mode; }),
+       FieldTrialStructMember("denoiser",
+                              [](Params* p) { return &p->allow_denoising; })},
+      {});
+
+  FieldTrialFlag per_layer_speed("use_per_layer_speed");
+
+  ParseFieldTrial({&trials_list, &per_layer_speed},
+                  trials.Lookup("WebRTC-VP9-PerformanceFlags"));
+
+  PerformanceFlags flags;
+  flags.use_per_layer_speed = per_layer_speed.Get();
+
+  constexpr int kMinSpeed = 1;
+  constexpr int kMaxSpeed = 9;
+  for (auto& f : trials_list.Get()) {
+    if (f.base_layer_speed < kMinSpeed || f.base_layer_speed > kMaxSpeed ||
+        f.high_layer_speed < kMinSpeed || f.high_layer_speed > kMaxSpeed ||
+        f.deblock_mode < 0 || f.deblock_mode > 2) {
+      RTC_LOG(LS_WARNING) << "Ignoring invalid performance flags: "
+                          << "min_pixel_count = " << f.min_pixel_count
+                          << ", high_layer_speed = " << f.high_layer_speed
+                          << ", base_layer_speed = " << f.base_layer_speed
+                          << ", deblock_mode = " << f.deblock_mode;
+      continue;
+    }
+    flags.settings_by_resolution[f.min_pixel_count] = f;
+  }
+
+  if (flags.settings_by_resolution.empty()) {
+    return GetDefaultPerformanceFlags();
+  }
+
+  return flags;
+}
+
+// static
+LibvpxVp9Encoder::PerformanceFlags
+LibvpxVp9Encoder::GetDefaultPerformanceFlags() {
+  PerformanceFlags flags;
+  flags.use_per_layer_speed = true;
+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || defined(ANDROID)
+  // Speed 8 on all layers for all resolutions.
+  flags.settings_by_resolution[0] = {.base_layer_speed = 8,
+                                     .high_layer_speed = 8,
+                                     .deblock_mode = 0,
+                                     .allow_denoising = true};
+#else
+
+  // For smaller resolutions, use lower speed setting for the temporal base
+  // layer (get some coding gain at the cost of increased encoding complexity).
+  // Set encoder Speed 5 for TL0, encoder Speed 8 for upper temporal layers, and
+  // disable deblocking for upper-most temporal layers.
+  flags.settings_by_resolution[0] = {.base_layer_speed = 5,
+                                     .high_layer_speed = 8,
+                                     .deblock_mode = 1,
+                                     .allow_denoising = true};
+
+  // Use speed 7 for QCIF and above.
+  // Set encoder Speed 7 for TL0, encoder Speed 8 for upper temporal layers, and
+  // enable deblocking for all temporal layers.
+  flags.settings_by_resolution[352 * 288] = {.base_layer_speed = 7,
+                                             .high_layer_speed = 8,
+                                             .deblock_mode = 0,
+                                             .allow_denoising = true};
+
+  // For very high resolution (1080p and up), turn the speed all the way up
+  // since this is very CPU intensive. Also disable denoising to save CPU, at
+  // these resolutions denoising appear less effective and hopefully you also
+  // have a less noisy video source at this point.
+  flags.settings_by_resolution[1920 * 1080] = {.base_layer_speed = 9,
+                                               .high_layer_speed = 9,
+                                               .deblock_mode = 0,
+                                               .allow_denoising = false};
+
+#endif
+  return flags;
+}
+
+void LibvpxVp9Encoder::MaybeRewrapRawWithFormat(const vpx_img_fmt fmt) {
+  if (!raw_) {
+    raw_ = libvpx_->img_wrap(nullptr, fmt, codec_.width, codec_.height, 1,
+                             nullptr);
+  } else if (raw_->fmt != fmt) {
+    RTC_LOG(LS_INFO) << "Switching VP9 encoder pixel format to "
+                     << (fmt == VPX_IMG_FMT_NV12 ? "NV12" : "I420");
+    libvpx_->img_free(raw_);
+    raw_ = libvpx_->img_wrap(nullptr, fmt, codec_.width, codec_.height, 1,
+                             nullptr);
+  }
+  // else no-op since the image is already in the right format.
+}
+
+rtc::scoped_refptr<VideoFrameBuffer> LibvpxVp9Encoder::PrepareBufferForProfile0(
+    rtc::scoped_refptr<VideoFrameBuffer> buffer) {
+  absl::InlinedVector<VideoFrameBuffer::Type, kMaxPreferredPixelFormats>
+      supported_formats = {VideoFrameBuffer::Type::kI420,
+                           VideoFrameBuffer::Type::kNV12};
+
+  rtc::scoped_refptr<VideoFrameBuffer> mapped_buffer;
+  if (buffer->type() != VideoFrameBuffer::Type::kNative) {
+    // `buffer` is already mapped.
+    mapped_buffer = buffer;
+  } else {
+    // Attempt to map to one of the supported formats.
+    mapped_buffer = buffer->GetMappedFrameBuffer(supported_formats);
+  }
+  if (!mapped_buffer ||
+      (absl::c_find(supported_formats, mapped_buffer->type()) ==
+           supported_formats.end() &&
+       mapped_buffer->type() != VideoFrameBuffer::Type::kI420A)) {
+    // Unknown pixel format or unable to map, convert to I420 and prepare that
+    // buffer instead to ensure Scale() is safe to use.
+    auto converted_buffer = buffer->ToI420();
+    if (!converted_buffer) {
+      RTC_LOG(LS_ERROR) << "Failed to convert "
+                        << VideoFrameBufferTypeToString(buffer->type())
+                        << " image to I420. Can't encode frame.";
+      return {};
+    }
+    RTC_CHECK(converted_buffer->type() == VideoFrameBuffer::Type::kI420 ||
+              converted_buffer->type() == VideoFrameBuffer::Type::kI420A);
+
+    // Because `buffer` had to be converted, use `converted_buffer` instead.
+    buffer = mapped_buffer = converted_buffer;
+  }
+
+  // Prepare `raw_` from `mapped_buffer`.
+  switch (mapped_buffer->type()) {
+    case VideoFrameBuffer::Type::kI420:
+    case VideoFrameBuffer::Type::kI420A: {
+      MaybeRewrapRawWithFormat(VPX_IMG_FMT_I420);
+      const I420BufferInterface* i420_buffer = mapped_buffer->GetI420();
+      RTC_DCHECK(i420_buffer);
+      raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(i420_buffer->DataY());
+      raw_->planes[VPX_PLANE_U] = const_cast<uint8_t*>(i420_buffer->DataU());
+      raw_->planes[VPX_PLANE_V] = const_cast<uint8_t*>(i420_buffer->DataV());
+      raw_->stride[VPX_PLANE_Y] = i420_buffer->StrideY();
+      raw_->stride[VPX_PLANE_U] = i420_buffer->StrideU();
+      raw_->stride[VPX_PLANE_V] = i420_buffer->StrideV();
+      break;
+    }
+    case VideoFrameBuffer::Type::kNV12: {
+      MaybeRewrapRawWithFormat(VPX_IMG_FMT_NV12);
+      const NV12BufferInterface* nv12_buffer = mapped_buffer->GetNV12();
+      RTC_DCHECK(nv12_buffer);
+      raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(nv12_buffer->DataY());
+      raw_->planes[VPX_PLANE_U] = const_cast<uint8_t*>(nv12_buffer->DataUV());
+      raw_->planes[VPX_PLANE_V] = raw_->planes[VPX_PLANE_U] + 1;
+      raw_->stride[VPX_PLANE_Y] = nv12_buffer->StrideY();
+      raw_->stride[VPX_PLANE_U] = nv12_buffer->StrideUV();
+      raw_->stride[VPX_PLANE_V] = nv12_buffer->StrideUV();
+      break;
+    }
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  return mapped_buffer;
+}
+
+}  // namespace webrtc
+
+#endif  // RTC_ENABLE_VP9
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h b/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h
new file mode 100644
index 0000000000..bb871f8498
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h
@@ -0,0 +1,251 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP9_LIBVPX_VP9_ENCODER_H_
+#define MODULES_VIDEO_CODING_CODECS_VP9_LIBVPX_VP9_ENCODER_H_
+
+#ifdef RTC_ENABLE_VP9
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "api/fec_controller_override.h"
+#include "api/field_trials_view.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/vp9_profile.h"
+#include "common_video/include/video_frame_buffer_pool.h"
+#include "modules/video_coding/codecs/interface/libvpx_interface.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+#include "modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "modules/video_coding/utility/framerate_controller_deprecated.h"
+#include "rtc_base/containers/flat_map.h"
+#include "rtc_base/experiments/encoder_info_settings.h"
+#include "vpx/vp8cx.h"
+
+namespace webrtc {
+
+class LibvpxVp9Encoder : public VP9Encoder {
+ public:
+  LibvpxVp9Encoder(const cricket::VideoCodec& codec,
+                   std::unique_ptr<LibvpxInterface> interface,
+                   const FieldTrialsView& trials);
+
+  ~LibvpxVp9Encoder() override;
+
+  void SetFecControllerOverride(
+      FecControllerOverride* fec_controller_override) override;
+
+  int Release() override;
+
+  int InitEncode(const VideoCodec* codec_settings,
+                 const Settings& settings) override;
+
+  int Encode(const VideoFrame& input_image,
+             const std::vector<VideoFrameType>* frame_types) override;
+
+  int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
+
+  void SetRates(const RateControlParameters& parameters) override;
+
+  EncoderInfo GetEncoderInfo() const override;
+
+ private:
+  // Determine number of encoder threads to use.
+  int NumberOfThreads(int width, int height, int number_of_cores);
+
+  // Call encoder initialize function and set control settings.
+  int InitAndSetControlSettings(const VideoCodec* inst);
+
+  // Update frame size for codec.
+  int UpdateCodecFrameSize(const VideoFrame& input_image);
+
+  bool PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
+                             absl::optional<int>* spatial_idx,
+                             absl::optional<int>* temporal_idx,
+                             const vpx_codec_cx_pkt& pkt);
+  void FillReferenceIndices(const vpx_codec_cx_pkt& pkt,
+                            size_t pic_num,
+                            bool inter_layer_predicted,
+                            CodecSpecificInfoVP9* vp9_info);
+  void UpdateReferenceBuffers(const vpx_codec_cx_pkt& pkt, size_t pic_num);
+  vpx_svc_ref_frame_config_t SetReferences(bool is_key_pic,
+                                           int first_active_spatial_layer_id);
+
+  bool ExplicitlyConfiguredSpatialLayers() const;
+  bool SetSvcRates(const VideoBitrateAllocation& bitrate_allocation);
+
+  // Configures which spatial layers libvpx should encode according to
+  // configuration provided by svc_controller_.
+  void EnableSpatialLayer(int sid);
+  void DisableSpatialLayer(int sid);
+  void SetActiveSpatialLayers();
+
+  void GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt);
+
+  // Callback function for outputting packets per spatial layer.
+  static void EncoderOutputCodedPacketCallback(vpx_codec_cx_pkt* pkt,
+                                               void* user_data);
+
+  void DeliverBufferedFrame(bool end_of_picture);
+
+  bool DropFrame(uint8_t spatial_idx, uint32_t rtp_timestamp);
+
+  // Determine maximum target for Intra frames
+  //
+  // Input:
+  //    - optimal_buffer_size : Optimal buffer size
+  // Return Value             : Max target size for Intra frames represented as
+  //                            percentage of the per frame bandwidth
+  uint32_t MaxIntraTarget(uint32_t optimal_buffer_size);
+
+  size_t SteadyStateSize(int sid, int tid);
+
+  void MaybeRewrapRawWithFormat(vpx_img_fmt fmt);
+  // Prepares `raw_` to reference image data of `buffer`, or of mapped or scaled
+  // versions of `buffer`. Returns the buffer that got referenced as a result,
+  // allowing the caller to keep a reference to it until after encoding has
+  // finished. On failure to convert the buffer, null is returned.
+  rtc::scoped_refptr<VideoFrameBuffer> PrepareBufferForProfile0(
+      rtc::scoped_refptr<VideoFrameBuffer> buffer);
+
+  const std::unique_ptr<LibvpxInterface> libvpx_;
+  EncodedImage encoded_image_;
+  CodecSpecificInfo codec_specific_;
+  EncodedImageCallback* encoded_complete_callback_;
+  VideoCodec codec_;
+  const VP9Profile profile_;
+  bool inited_;
+  int64_t timestamp_;
+  uint32_t rc_max_intra_target_;
+  vpx_codec_ctx_t* encoder_;
+  vpx_codec_enc_cfg_t* config_;
+  vpx_image_t* raw_;
+  vpx_svc_extra_cfg_t svc_params_;
+  const VideoFrame* input_image_;
+  GofInfoVP9 gof_;  // Contains each frame's temporal information for
+                    // non-flexible mode.
+  bool force_key_frame_;
+  size_t pics_since_key_;
+  uint8_t num_temporal_layers_;
+  uint8_t num_spatial_layers_;         // Number of configured SLs
+  uint8_t num_active_spatial_layers_;  // Number of actively encoded SLs
+  uint8_t first_active_layer_;
+  bool layer_deactivation_requires_key_frame_;
+  bool is_svc_;
+  InterLayerPredMode inter_layer_pred_;
+  bool external_ref_control_;
+  const bool trusted_rate_controller_;
+  bool layer_buffering_;
+  const bool full_superframe_drop_;
+  vpx_svc_frame_drop_t svc_drop_frame_;
+  bool first_frame_in_picture_;
+  VideoBitrateAllocation current_bitrate_allocation_;
+  bool ss_info_needed_;
+  bool force_all_active_layers_;
+  uint8_t num_cores_;
+
+  std::unique_ptr<ScalableVideoController> svc_controller_;
+  absl::optional<ScalabilityMode> scalability_mode_;
+  std::vector<FramerateControllerDeprecated> framerate_controller_;
+
+  // Used for flexible mode.
+  bool is_flexible_mode_;
+  struct RefFrameBuffer {
+    bool operator==(const RefFrameBuffer& o) {
+      return pic_num == o.pic_num && spatial_layer_id == o.spatial_layer_id &&
+             temporal_layer_id == o.temporal_layer_id;
+    }
+
+    size_t pic_num = 0;
+    int spatial_layer_id = 0;
+    int temporal_layer_id = 0;
+  };
+  std::array<RefFrameBuffer, kNumVp9Buffers> ref_buf_;
+  std::vector<ScalableVideoController::LayerFrameConfig> layer_frames_;
+
+  // Variable frame-rate related fields and methods.
+  const struct VariableFramerateExperiment {
+    bool enabled;
+    // Framerate is limited to this value in steady state.
+    float framerate_limit;
+    // This qp or below is considered a steady state.
+    int steady_state_qp;
+    // Frames of at least this percentage below ideal for configured bitrate are
+    // considered in a steady state.
+    int steady_state_undershoot_percentage;
+    // Number of consecutive frames with good QP and size required to detect
+    // the steady state.
+    int frames_before_steady_state;
+  } variable_framerate_experiment_;
+  static VariableFramerateExperiment ParseVariableFramerateConfig(
+      const FieldTrialsView& trials);
+  FramerateControllerDeprecated variable_framerate_controller_;
+
+  const struct QualityScalerExperiment {
+    int low_qp;
+    int high_qp;
+    bool enabled;
+  } quality_scaler_experiment_;
+  static QualityScalerExperiment ParseQualityScalerConfig(
+      const FieldTrialsView& trials);
+  const bool external_ref_ctrl_;
+
+  // Flags that can affect speed vs quality tradeoff, and are configureable per
+  // resolution ranges.
+  struct PerformanceFlags {
+    // If false, a lookup will be made in `settings_by_resolution` base on the
+    // highest currently active resolution, and the overall speed then set to
+    // to the `base_layer_speed` matching that entry.
+    // If true, each active resolution will have it's speed and deblock_mode set
+    // based on it resolution, and the high layer speed configured for non
+    // base temporal layer frames.
+    bool use_per_layer_speed = false;
+
+    struct ParameterSet {
+      int base_layer_speed = -1;  // Speed setting for TL0.
+      int high_layer_speed = -1;  // Speed setting for TL1-TL3.
+      //  0 = deblock all temporal layers (TL)
+      //  1 = disable deblock for top-most TL
+      //  2 = disable deblock for all TLs
+      int deblock_mode = 0;
+      bool allow_denoising = true;
+    };
+    // Map from min pixel count to settings for that resolution and above.
+    // E.g. if you want some settings A if below wvga (640x360) and some other
+    // setting B at wvga and above, you'd use map {{0, A}, {230400, B}}.
+    flat_map<int, ParameterSet> settings_by_resolution;
+  };
+  // Performance flags, ordered by `min_pixel_count`.
+  const PerformanceFlags performance_flags_;
+  // Caching of of `speed_configs_`, where index i maps to the resolution as
+  // specified in `codec_.spatialLayer[i]`.
+  std::vector<PerformanceFlags::ParameterSet>
+      performance_flags_by_spatial_index_;
+  void UpdatePerformanceFlags();
+  static PerformanceFlags ParsePerformanceFlagsFromTrials(
+      const FieldTrialsView& trials);
+  static PerformanceFlags GetDefaultPerformanceFlags();
+
+  int num_steady_state_frames_;
+  // Only set config when this flag is set.
+  bool config_changed_;
+
+  const LibvpxVp9EncoderInfoSettings encoder_info_override_;
+};
+
+}  // namespace webrtc
+
+#endif  // RTC_ENABLE_VP9
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP9_LIBVPX_VP9_ENCODER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config.cc
new file mode 100644
index 0000000000..3a32a43622
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config.cc
@@ -0,0 +1,240 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp9/svc_config.h"
+
+#include <algorithm>
+#include <cmath>
+#include <memory>
+#include <vector>
+
+#include "media/base/video_common.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+const size_t kMinVp9SvcBitrateKbps = 30;
+
+const size_t kMaxNumLayersForScreenSharing = 3;
+const float kMaxScreenSharingLayerFramerateFps[] = {5.0, 10.0, 30.0};
+const size_t kMinScreenSharingLayerBitrateKbps[] = {30, 200, 500};
+const size_t kTargetScreenSharingLayerBitrateKbps[] = {150, 350, 950};
+const size_t kMaxScreenSharingLayerBitrateKbps[] = {250, 500, 950};
+
+// Gets limited number of layers for given resolution.
+size_t GetLimitedNumSpatialLayers(size_t width, size_t height) {
+  const bool is_landscape = width >= height;
+  const size_t min_width = is_landscape ? kMinVp9SpatialLayerLongSideLength
+                                        : kMinVp9SpatialLayerShortSideLength;
+  const size_t min_height = is_landscape ? kMinVp9SpatialLayerShortSideLength
+                                         : kMinVp9SpatialLayerLongSideLength;
+  const size_t num_layers_fit_horz = static_cast<size_t>(
+      std::floor(1 + std::max(0.0f, std::log2(1.0f * width / min_width))));
+  const size_t num_layers_fit_vert = static_cast<size_t>(
+      std::floor(1 + std::max(0.0f, std::log2(1.0f * height / min_height))));
+  return std::min(num_layers_fit_horz, num_layers_fit_vert);
+}
+}  // namespace
+
+std::vector<SpatialLayer> ConfigureSvcScreenSharing(size_t input_width,
+                                                    size_t input_height,
+                                                    float max_framerate_fps,
+                                                    size_t num_spatial_layers) {
+  num_spatial_layers =
+      std::min(num_spatial_layers, kMaxNumLayersForScreenSharing);
+  std::vector<SpatialLayer> spatial_layers;
+
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    SpatialLayer spatial_layer = {0};
+    spatial_layer.width = input_width;
+    spatial_layer.height = input_height;
+    spatial_layer.maxFramerate =
+        std::min(kMaxScreenSharingLayerFramerateFps[sl_idx], max_framerate_fps);
+    spatial_layer.numberOfTemporalLayers = 1;
+    spatial_layer.minBitrate =
+        static_cast<int>(kMinScreenSharingLayerBitrateKbps[sl_idx]);
+    spatial_layer.maxBitrate =
+        static_cast<int>(kMaxScreenSharingLayerBitrateKbps[sl_idx]);
+    spatial_layer.targetBitrate =
+        static_cast<int>(kTargetScreenSharingLayerBitrateKbps[sl_idx]);
+    spatial_layer.active = true;
+    spatial_layers.push_back(spatial_layer);
+  }
+
+  return spatial_layers;
+}
+
+std::vector<SpatialLayer> ConfigureSvcNormalVideo(
+    size_t input_width,
+    size_t input_height,
+    float max_framerate_fps,
+    size_t first_active_layer,
+    size_t num_spatial_layers,
+    size_t num_temporal_layers,
+    absl::optional<ScalableVideoController::StreamLayersConfig> config) {
+  RTC_DCHECK_LT(first_active_layer, num_spatial_layers);
+
+  // Limit number of layers for given resolution.
+  size_t limited_num_spatial_layers =
+      GetLimitedNumSpatialLayers(input_width, input_height);
+  if (limited_num_spatial_layers < num_spatial_layers) {
+    RTC_LOG(LS_WARNING) << "Reducing number of spatial layers from "
+                        << num_spatial_layers << " to "
+                        << limited_num_spatial_layers
+                        << " due to low input resolution.";
+    num_spatial_layers = limited_num_spatial_layers;
+  }
+
+  // First active layer must be configured.
+  num_spatial_layers = std::max(num_spatial_layers, first_active_layer + 1);
+
+  // Ensure top layer is even enough.
+  int required_divisiblity = 1 << (num_spatial_layers - first_active_layer - 1);
+  if (config) {
+    required_divisiblity = 1;
+    for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+      required_divisiblity = cricket::LeastCommonMultiple(
+          required_divisiblity, config->scaling_factor_den[sl_idx]);
+    }
+  }
+  input_width = input_width - input_width % required_divisiblity;
+  input_height = input_height - input_height % required_divisiblity;
+
+  std::vector<SpatialLayer> spatial_layers;
+  for (size_t sl_idx = first_active_layer; sl_idx < num_spatial_layers;
+       ++sl_idx) {
+    SpatialLayer spatial_layer = {0};
+    spatial_layer.width = input_width >> (num_spatial_layers - sl_idx - 1);
+    spatial_layer.height = input_height >> (num_spatial_layers - sl_idx - 1);
+    spatial_layer.maxFramerate = max_framerate_fps;
+    spatial_layer.numberOfTemporalLayers = num_temporal_layers;
+    spatial_layer.active = true;
+
+    if (config) {
+      spatial_layer.width = input_width * config->scaling_factor_num[sl_idx] /
+                            config->scaling_factor_den[sl_idx];
+      spatial_layer.height = input_height * config->scaling_factor_num[sl_idx] /
+                             config->scaling_factor_den[sl_idx];
+    }
+
+    // minBitrate and maxBitrate formulas were derived from
+    // subjective-quality data to determing bit rates below which video
+    // quality is unacceptable and above which additional bits do not provide
+    // benefit. The formulas express rate in units of kbps.
+
+    // TODO(ssilkin): Add to the comment PSNR/SSIM we get at encoding certain
+    // video to min/max bitrate specified by those formulas.
+    const size_t num_pixels = spatial_layer.width * spatial_layer.height;
+    int min_bitrate =
+        static_cast<int>((600. * std::sqrt(num_pixels) - 95000.) / 1000.);
+    min_bitrate = std::max(min_bitrate, 0);
+    spatial_layer.minBitrate =
+        std::max(static_cast<size_t>(min_bitrate), kMinVp9SvcBitrateKbps);
+    spatial_layer.maxBitrate =
+        static_cast<int>((1.6 * num_pixels + 50 * 1000) / 1000);
+    spatial_layer.targetBitrate =
+        (spatial_layer.minBitrate + spatial_layer.maxBitrate) / 2;
+    spatial_layers.push_back(spatial_layer);
+  }
+
+  // A workaround for situation when single HD layer is left with minBitrate
+  // about 500kbps. This would mean that there will always be at least 500kbps
+  // allocated to video regardless of how low is the actual BWE.
+  // Also, boost maxBitrate for the first layer to account for lost ability to
+  // predict from previous layers.
+  if (first_active_layer > 0) {
+    spatial_layers[0].minBitrate = kMinVp9SvcBitrateKbps;
+    // TODO(ilnik): tune this value or come up with a different formula to
+    // ensure that all singlecast configurations look good and not too much
+    // bitrate is added.
+    spatial_layers[0].maxBitrate *= 1.1;
+  }
+
+  return spatial_layers;
+}
+
+// Uses scalability mode to configure spatial layers.
+std::vector<SpatialLayer> GetVp9SvcConfig(VideoCodec& codec) {
+  RTC_DCHECK_EQ(codec.codecType, kVideoCodecVP9);
+
+  absl::optional<ScalabilityMode> scalability_mode = codec.GetScalabilityMode();
+  RTC_DCHECK(scalability_mode.has_value());
+
+  // Limit number of spatial layers for given resolution.
+  int limited_num_spatial_layers =
+      GetLimitedNumSpatialLayers(codec.width, codec.height);
+  if (limited_num_spatial_layers <
+      ScalabilityModeToNumSpatialLayers(*scalability_mode)) {
+    ScalabilityMode limited_scalability_mode =
+        LimitNumSpatialLayers(*scalability_mode, limited_num_spatial_layers);
+    RTC_LOG(LS_WARNING)
+        << "Reducing number of spatial layers due to low input resolution: "
+        << ScalabilityModeToString(*scalability_mode) << " to "
+        << ScalabilityModeToString(limited_scalability_mode);
+    scalability_mode = limited_scalability_mode;
+    codec.SetScalabilityMode(limited_scalability_mode);
+  }
+
+  absl::optional<ScalableVideoController::StreamLayersConfig> info =
+      ScalabilityStructureConfig(*scalability_mode);
+  if (!info.has_value()) {
+    RTC_LOG(LS_WARNING) << "Failed to create structure "
+                        << ScalabilityModeToString(*scalability_mode);
+    return {};
+  }
+
+  // TODO(bugs.webrtc.org/11607): Add support for screensharing.
+  std::vector<SpatialLayer> spatial_layers =
+      GetSvcConfig(codec.width, codec.height, codec.maxFramerate,
+                   /*first_active_layer=*/0, info->num_spatial_layers,
+                   info->num_temporal_layers, /*is_screen_sharing=*/false,
+                   codec.GetScalabilityMode() ? info : absl::nullopt);
+  RTC_DCHECK(!spatial_layers.empty());
+
+  // Use codec bitrate limits if spatial layering is not requested.
+  if (info->num_spatial_layers == 1) {
+    spatial_layers.back().minBitrate = codec.minBitrate;
+    spatial_layers.back().targetBitrate = codec.maxBitrate;
+    spatial_layers.back().maxBitrate = codec.maxBitrate;
+  }
+
+  return spatial_layers;
+}
+
+std::vector<SpatialLayer> GetSvcConfig(
+    size_t input_width,
+    size_t input_height,
+    float max_framerate_fps,
+    size_t first_active_layer,
+    size_t num_spatial_layers,
+    size_t num_temporal_layers,
+    bool is_screen_sharing,
+    absl::optional<ScalableVideoController::StreamLayersConfig> config) {
+  RTC_DCHECK_GT(input_width, 0);
+  RTC_DCHECK_GT(input_height, 0);
+  RTC_DCHECK_GT(num_spatial_layers, 0);
+  RTC_DCHECK_GT(num_temporal_layers, 0);
+
+  if (is_screen_sharing) {
+    return ConfigureSvcScreenSharing(input_width, input_height,
+                                     max_framerate_fps, num_spatial_layers);
+  } else {
+    return ConfigureSvcNormalVideo(input_width, input_height, max_framerate_fps,
+                                   first_active_layer, num_spatial_layers,
+                                   num_temporal_layers, config);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config.h b/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config.h
new file mode 100644
index 0000000000..adeaf0f161
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config.h
@@ -0,0 +1,39 @@
+/* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP9_SVC_CONFIG_H_
+#define MODULES_VIDEO_CODING_CODECS_VP9_SVC_CONFIG_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "api/video_codecs/spatial_layer.h"
+#include "api/video_codecs/video_codec.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+
+namespace webrtc {
+
+// Uses scalability mode to configure spatial layers.
+std::vector<SpatialLayer> GetVp9SvcConfig(VideoCodec& video_codec);
+
+std::vector<SpatialLayer> GetSvcConfig(
+    size_t input_width,
+    size_t input_height,
+    float max_framerate_fps,
+    size_t first_active_layer,
+    size_t num_spatial_layers,
+    size_t num_temporal_layers,
+    bool is_screen_sharing,
+    absl::optional<ScalableVideoController::StreamLayersConfig> config =
+        absl::nullopt);
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP9_SVC_CONFIG_H_
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc
new file mode 100644
index 0000000000..762fd39287
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc
@@ -0,0 +1,285 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp9/svc_config.h"
+
+#include <cstddef>
+#include <vector>
+
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::ElementsAre;
+using ::testing::Field;
+
+namespace webrtc {
+TEST(SvcConfig, NumSpatialLayers) {
+  const size_t max_num_spatial_layers = 6;
+  const size_t first_active_layer = 0;
+  const size_t num_spatial_layers = 2;
+
+  std::vector<SpatialLayer> spatial_layers = GetSvcConfig(
+      kMinVp9SpatialLayerLongSideLength << (num_spatial_layers - 1),
+      kMinVp9SpatialLayerShortSideLength << (num_spatial_layers - 1), 30,
+      first_active_layer, max_num_spatial_layers, 1, false);
+
+  EXPECT_EQ(spatial_layers.size(), num_spatial_layers);
+}
+
+TEST(SvcConfig, NumSpatialLayersPortrait) {
+  const size_t max_num_spatial_layers = 6;
+  const size_t first_active_layer = 0;
+  const size_t num_spatial_layers = 2;
+
+  std::vector<SpatialLayer> spatial_layers = GetSvcConfig(
+      kMinVp9SpatialLayerShortSideLength << (num_spatial_layers - 1),
+      kMinVp9SpatialLayerLongSideLength << (num_spatial_layers - 1), 30,
+      first_active_layer, max_num_spatial_layers, 1, false);
+
+  EXPECT_EQ(spatial_layers.size(), num_spatial_layers);
+}
+
+TEST(SvcConfig, NumSpatialLayersWithScalabilityMode) {
+  VideoCodec codec;
+  codec.codecType = kVideoCodecVP9;
+  codec.width = 960;
+  codec.height = 540;
+  codec.SetScalabilityMode(ScalabilityMode::kL3T3_KEY);
+
+  std::vector<SpatialLayer> spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers, ElementsAre(Field(&SpatialLayer::height, 135),
+                                          Field(&SpatialLayer::height, 270),
+                                          Field(&SpatialLayer::height, 540)));
+  EXPECT_THAT(spatial_layers,
+              ElementsAre(Field(&SpatialLayer::numberOfTemporalLayers, 3),
+                          Field(&SpatialLayer::numberOfTemporalLayers, 3),
+                          Field(&SpatialLayer::numberOfTemporalLayers, 3)));
+  EXPECT_EQ(codec.GetScalabilityMode(), ScalabilityMode::kL3T3_KEY);
+}
+
+TEST(SvcConfig, NumSpatialLayersLimitedWithScalabilityMode) {
+  VideoCodec codec;
+  codec.codecType = kVideoCodecVP9;
+  codec.width = 480;
+  codec.height = 270;
+  codec.SetScalabilityMode(ScalabilityMode::kL3T3_KEY);
+
+  // Scalability mode updated.
+  std::vector<SpatialLayer> spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers, ElementsAre(Field(&SpatialLayer::height, 135),
+                                          Field(&SpatialLayer::height, 270)));
+  EXPECT_THAT(spatial_layers,
+              ElementsAre(Field(&SpatialLayer::numberOfTemporalLayers, 3),
+                          Field(&SpatialLayer::numberOfTemporalLayers, 3)));
+  EXPECT_EQ(codec.GetScalabilityMode(), ScalabilityMode::kL2T3_KEY);
+}
+
+TEST(SvcConfig, NumSpatialLayersLimitedWithScalabilityModePortrait) {
+  VideoCodec codec;
+  codec.codecType = kVideoCodecVP9;
+  codec.width = 270;
+  codec.height = 480;
+  codec.SetScalabilityMode(ScalabilityMode::kL3T1);
+
+  // Scalability mode updated.
+  std::vector<SpatialLayer> spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers, ElementsAre(Field(&SpatialLayer::width, 135),
+                                          Field(&SpatialLayer::width, 270)));
+  EXPECT_THAT(spatial_layers,
+              ElementsAre(Field(&SpatialLayer::numberOfTemporalLayers, 1),
+                          Field(&SpatialLayer::numberOfTemporalLayers, 1)));
+  EXPECT_EQ(codec.GetScalabilityMode(), ScalabilityMode::kL2T1);
+}
+
+TEST(SvcConfig, NumSpatialLayersWithScalabilityModeResolutionRatio1_5) {
+  VideoCodec codec;
+  codec.codecType = kVideoCodecVP9;
+  codec.width = 270;
+  codec.height = 480;
+  codec.SetScalabilityMode(ScalabilityMode::kL2T1h);  // 1.5:1
+
+  std::vector<SpatialLayer> spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers, ElementsAre(Field(&SpatialLayer::width, 180),
+                                          Field(&SpatialLayer::width, 270)));
+  EXPECT_THAT(spatial_layers,
+              ElementsAre(Field(&SpatialLayer::numberOfTemporalLayers, 1),
+                          Field(&SpatialLayer::numberOfTemporalLayers, 1)));
+  EXPECT_EQ(codec.GetScalabilityMode(), ScalabilityMode::kL2T1h);
+}
+
+TEST(SvcConfig, NumSpatialLayersLimitedWithScalabilityModeResolutionRatio1_5) {
+  VideoCodec codec;
+  codec.codecType = kVideoCodecVP9;
+  codec.width = 320;
+  codec.height = 180;
+  codec.SetScalabilityMode(ScalabilityMode::kL3T1h);  // 1.5:1
+
+  // Scalability mode updated.
+  std::vector<SpatialLayer> spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers, ElementsAre(Field(&SpatialLayer::width, 320)));
+  EXPECT_THAT(spatial_layers,
+              ElementsAre(Field(&SpatialLayer::numberOfTemporalLayers, 1)));
+  EXPECT_EQ(codec.GetScalabilityMode(), ScalabilityMode::kL1T1);
+}
+
+TEST(SvcConfig, AlwaysSendsAtLeastOneLayer) {
+  const size_t max_num_spatial_layers = 6;
+  const size_t first_active_layer = 5;
+
+  std::vector<SpatialLayer> spatial_layers = GetSvcConfig(
+      kMinVp9SpatialLayerLongSideLength, kMinVp9SpatialLayerShortSideLength, 30,
+      first_active_layer, max_num_spatial_layers, 1, false);
+  EXPECT_EQ(spatial_layers.size(), 1u);
+  EXPECT_EQ(spatial_layers.back().width, kMinVp9SpatialLayerLongSideLength);
+}
+
+TEST(SvcConfig, AlwaysSendsAtLeastOneLayerPortrait) {
+  const size_t max_num_spatial_layers = 6;
+  const size_t first_active_layer = 5;
+
+  std::vector<SpatialLayer> spatial_layers = GetSvcConfig(
+      kMinVp9SpatialLayerShortSideLength, kMinVp9SpatialLayerLongSideLength, 30,
+      first_active_layer, max_num_spatial_layers, 1, false);
+  EXPECT_EQ(spatial_layers.size(), 1u);
+  EXPECT_EQ(spatial_layers.back().width, kMinVp9SpatialLayerShortSideLength);
+}
+
+TEST(SvcConfig, EnforcesMinimalRequiredParity) {
+  const size_t max_num_spatial_layers = 3;
+  const size_t kOddSize = 1023;
+
+  std::vector<SpatialLayer> spatial_layers =
+      GetSvcConfig(kOddSize, kOddSize, 30,
+                   /*first_active_layer=*/1, max_num_spatial_layers, 1, false);
+  // Since there are 2 layers total (1, 2), divisiblity by 2 is required.
+  EXPECT_EQ(spatial_layers.back().width, kOddSize - 1);
+  EXPECT_EQ(spatial_layers.back().width, kOddSize - 1);
+
+  spatial_layers =
+      GetSvcConfig(kOddSize, kOddSize, 30,
+                   /*first_active_layer=*/0, max_num_spatial_layers, 1, false);
+  // Since there are 3 layers total (0, 1, 2), divisiblity by 4 is required.
+  EXPECT_EQ(spatial_layers.back().width, kOddSize - 3);
+  EXPECT_EQ(spatial_layers.back().width, kOddSize - 3);
+
+  spatial_layers =
+      GetSvcConfig(kOddSize, kOddSize, 30,
+                   /*first_active_layer=*/2, max_num_spatial_layers, 1, false);
+  // Since there is only 1 layer active (2), divisiblity by 1 is required.
+  EXPECT_EQ(spatial_layers.back().width, kOddSize);
+  EXPECT_EQ(spatial_layers.back().width, kOddSize);
+}
+
+TEST(SvcConfig, EnforcesMinimalRequiredParityWithScalabilityMode) {
+  VideoCodec codec;
+  codec.codecType = kVideoCodecVP9;
+  codec.width = 1023;
+  codec.height = 1023;
+  codec.SetScalabilityMode(ScalabilityMode::kL3T1);
+
+  std::vector<SpatialLayer> spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers,  // Divisiblity by 4 required.
+              ElementsAre(Field(&SpatialLayer::width, 255),
+                          Field(&SpatialLayer::width, 510),
+                          Field(&SpatialLayer::width, 1020)));
+
+  codec.SetScalabilityMode(ScalabilityMode::kL2T1);
+  spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers,  // Divisiblity by 2 required.
+              ElementsAre(Field(&SpatialLayer::width, 511),
+                          Field(&SpatialLayer::width, 1022)));
+
+  codec.SetScalabilityMode(ScalabilityMode::kL1T1);
+  spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers,  // Divisiblity by 1 required.
+              ElementsAre(Field(&SpatialLayer::width, 1023)));
+}
+
+TEST(SvcConfig, EnforcesMinimalRequiredParityWithScalabilityModeResRatio1_5) {
+  VideoCodec codec;
+  codec.codecType = kVideoCodecVP9;
+  codec.width = 1280;
+  codec.height = 1280;
+  codec.SetScalabilityMode(ScalabilityMode::kL2T1h);  // 1.5:1
+
+  std::vector<SpatialLayer> spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers,  // Divisiblity by 3 required.
+              ElementsAre(Field(&SpatialLayer::width, 852),
+                          Field(&SpatialLayer::width, 1278)));
+}
+
+TEST(SvcConfig, SkipsInactiveLayers) {
+  const size_t num_spatial_layers = 4;
+  const size_t first_active_layer = 2;
+
+  std::vector<SpatialLayer> spatial_layers = GetSvcConfig(
+      kMinVp9SpatialLayerLongSideLength << (num_spatial_layers - 1),
+      kMinVp9SpatialLayerShortSideLength << (num_spatial_layers - 1), 30,
+      first_active_layer, num_spatial_layers, 1, false);
+  EXPECT_EQ(spatial_layers.size(), 2u);
+  EXPECT_EQ(spatial_layers.back().width,
+            kMinVp9SpatialLayerLongSideLength << (num_spatial_layers - 1));
+}
+
+TEST(SvcConfig, BitrateThresholds) {
+  const size_t first_active_layer = 0;
+  const size_t num_spatial_layers = 3;
+  std::vector<SpatialLayer> spatial_layers = GetSvcConfig(
+      kMinVp9SpatialLayerLongSideLength << (num_spatial_layers - 1),
+      kMinVp9SpatialLayerShortSideLength << (num_spatial_layers - 1), 30,
+      first_active_layer, num_spatial_layers, 1, false);
+
+  EXPECT_EQ(spatial_layers.size(), num_spatial_layers);
+
+  for (const SpatialLayer& layer : spatial_layers) {
+    EXPECT_LE(layer.minBitrate, layer.maxBitrate);
+    EXPECT_LE(layer.minBitrate, layer.targetBitrate);
+    EXPECT_LE(layer.targetBitrate, layer.maxBitrate);
+  }
+}
+
+TEST(SvcConfig, BitrateThresholdsWithScalabilityMode) {
+  VideoCodec codec;
+  codec.codecType = kVideoCodecVP9;
+  codec.width = 960;
+  codec.height = 540;
+  codec.SetScalabilityMode(ScalabilityMode::kS3T3);
+
+  std::vector<SpatialLayer> spatial_layers = GetVp9SvcConfig(codec);
+  EXPECT_THAT(spatial_layers, ElementsAre(Field(&SpatialLayer::height, 135),
+                                          Field(&SpatialLayer::height, 270),
+                                          Field(&SpatialLayer::height, 540)));
+
+  for (const SpatialLayer& layer : spatial_layers) {
+    EXPECT_LE(layer.minBitrate, layer.maxBitrate);
+    EXPECT_LE(layer.minBitrate, layer.targetBitrate);
+    EXPECT_LE(layer.targetBitrate, layer.maxBitrate);
+  }
+}
+
+TEST(SvcConfig, ScreenSharing) {
+  std::vector<SpatialLayer> spatial_layers =
+      GetSvcConfig(1920, 1080, 30, 1, 3, 3, true);
+
+  EXPECT_EQ(spatial_layers.size(), 3UL);
+
+  for (size_t i = 0; i < 3; ++i) {
+    const SpatialLayer& layer = spatial_layers[i];
+    EXPECT_EQ(layer.width, 1920);
+    EXPECT_EQ(layer.height, 1080);
+    EXPECT_EQ(layer.maxFramerate, (i < 1) ? 5 : (i < 2 ? 10 : 30));
+    EXPECT_EQ(layer.numberOfTemporalLayers, 1);
+    EXPECT_LE(layer.minBitrate, layer.maxBitrate);
+    EXPECT_LE(layer.minBitrate, layer.targetBitrate);
+    EXPECT_LE(layer.targetBitrate, layer.maxBitrate);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
new file mode 100644
index 0000000000..b6293a342e
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
@@ -0,0 +1,2446 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "absl/memory/memory.h"
+#include "api/test/create_frame_generator.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/test/mock_video_encoder.h"
+#include "api/video/color_space.h"
+#include "api/video/i420_buffer.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/vp9_profile.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/video_coding/codecs/interface/libvpx_interface.h"
+#include "modules/video_coding/codecs/interface/mock_libvpx_interface.h"
+#include "modules/video_coding/codecs/test/encoded_video_frame_producer.h"
+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+#include "modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h"
+#include "modules/video_coding/codecs/vp9/svc_config.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/explicit_key_value_config.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mappable_native_buffer.h"
+#include "test/video_codec_settings.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::A;
+using ::testing::AllOf;
+using ::testing::An;
+using ::testing::AnyNumber;
+using ::testing::ByRef;
+using ::testing::DoAll;
+using ::testing::Each;
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::Field;
+using ::testing::IsEmpty;
+using ::testing::Mock;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::SafeMatcherCast;
+using ::testing::SaveArgPointee;
+using ::testing::SetArgPointee;
+using ::testing::SizeIs;
+using ::testing::TypedEq;
+using ::testing::UnorderedElementsAreArray;
+using ::testing::WithArg;
+using EncoderInfo = webrtc::VideoEncoder::EncoderInfo;
+using FramerateFractions =
+    absl::InlinedVector<uint8_t, webrtc::kMaxTemporalStreams>;
+
+constexpr size_t kWidth = 1280;
+constexpr size_t kHeight = 720;
+
+const VideoEncoder::Capabilities kCapabilities(false);
+const VideoEncoder::Settings kSettings(kCapabilities,
+                                       /*number_of_cores=*/1,
+                                       /*max_payload_size=*/0);
+
+VideoCodec DefaultCodecSettings() {
+  VideoCodec codec_settings;
+  webrtc::test::CodecSettings(kVideoCodecVP9, &codec_settings);
+  codec_settings.width = kWidth;
+  codec_settings.height = kHeight;
+  codec_settings.VP9()->numberOfTemporalLayers = 1;
+  codec_settings.VP9()->numberOfSpatialLayers = 1;
+  return codec_settings;
+}
+
+void ConfigureSvc(VideoCodec& codec_settings,
+                  int num_spatial_layers,
+                  int num_temporal_layers = 1) {
+  codec_settings.VP9()->numberOfSpatialLayers = num_spatial_layers;
+  codec_settings.VP9()->numberOfTemporalLayers = num_temporal_layers;
+  codec_settings.SetFrameDropEnabled(false);
+
+  std::vector<SpatialLayer> layers = GetSvcConfig(
+      codec_settings.width, codec_settings.height, codec_settings.maxFramerate,
+      /*first_active_layer=*/0, num_spatial_layers, num_temporal_layers, false);
+  for (size_t i = 0; i < layers.size(); ++i) {
+    codec_settings.spatialLayers[i] = layers[i];
+  }
+}
+
+}  // namespace
+
+class TestVp9Impl : public VideoCodecUnitTest {
+ protected:
+  std::unique_ptr<VideoEncoder> CreateEncoder() override {
+    return VP9Encoder::Create();
+  }
+
+  std::unique_ptr<VideoDecoder> CreateDecoder() override {
+    return VP9Decoder::Create();
+  }
+
+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
+    webrtc::test::CodecSettings(kVideoCodecVP9, codec_settings);
+    codec_settings->width = kWidth;
+    codec_settings->height = kHeight;
+    codec_settings->VP9()->numberOfTemporalLayers = 1;
+    codec_settings->VP9()->numberOfSpatialLayers = 1;
+  }
+};
+
+class TestVp9ImplForPixelFormat
+    : public TestVp9Impl,
+      public ::testing::WithParamInterface<
+          test::FrameGeneratorInterface::OutputType> {
+ protected:
+  void SetUp() override {
+    input_frame_generator_ = test::CreateSquareFrameGenerator(
+        kWidth, kHeight, GetParam(), absl::optional<int>());
+    TestVp9Impl::SetUp();
+  }
+};
+
+// Disabled on ios as flake, see https://crbug.com/webrtc/7057
+#if defined(WEBRTC_IOS)
+TEST_P(TestVp9ImplForPixelFormat, DISABLED_EncodeDecode) {
+#else
+TEST_P(TestVp9ImplForPixelFormat, EncodeDecode) {
+#endif
+  VideoFrame input_frame = NextInputFrame();
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(input_frame, nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  // First frame should be a key frame.
+  encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, 0));
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+  EXPECT_GT(I420PSNR(&input_frame, decoded_frame.get()), 36);
+
+  const ColorSpace color_space = *decoded_frame->color_space();
+  EXPECT_EQ(ColorSpace::PrimaryID::kUnspecified, color_space.primaries());
+  EXPECT_EQ(ColorSpace::TransferID::kUnspecified, color_space.transfer());
+  EXPECT_EQ(ColorSpace::MatrixID::kUnspecified, color_space.matrix());
+  EXPECT_EQ(ColorSpace::RangeID::kLimited, color_space.range());
+  EXPECT_EQ(ColorSpace::ChromaSiting::kUnspecified,
+            color_space.chroma_siting_horizontal());
+  EXPECT_EQ(ColorSpace::ChromaSiting::kUnspecified,
+            color_space.chroma_siting_vertical());
+}
+
+TEST_P(TestVp9ImplForPixelFormat, EncodeNativeBuffer) {
+  VideoFrame input_frame = NextInputFrame();
+  // Replace the input frame with a fake native buffer of the same size and
+  // underlying pixel format. Do not allow ToI420() for non-I420 buffers,
+  // ensuring zero-conversion.
+  input_frame = test::CreateMappableNativeFrame(
+      input_frame.ntp_time_ms(), input_frame.video_frame_buffer()->type(),
+      input_frame.width(), input_frame.height());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(input_frame, nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+
+  // After encoding, we would expect a single mapping to have happened.
+  rtc::scoped_refptr<test::MappableNativeBuffer> mappable_buffer =
+      test::GetMappableNativeBufferFromVideoFrame(input_frame);
+  std::vector<rtc::scoped_refptr<VideoFrameBuffer>> mapped_buffers =
+      mappable_buffer->GetMappedFramedBuffers();
+  ASSERT_EQ(mapped_buffers.size(), 1u);
+  EXPECT_EQ(mapped_buffers[0]->type(), mappable_buffer->mappable_type());
+  EXPECT_EQ(mapped_buffers[0]->width(), input_frame.width());
+  EXPECT_EQ(mapped_buffers[0]->height(), input_frame.height());
+  EXPECT_FALSE(mappable_buffer->DidConvertToI420());
+}
+
+TEST_P(TestVp9ImplForPixelFormat, DecodedColorSpaceFromBitstream) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+
+  // Encoded frame without explicit color space information.
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, 0));
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+  // Color space present from encoded bitstream.
+  ASSERT_TRUE(decoded_frame->color_space());
+  // No HDR metadata present.
+  EXPECT_FALSE(decoded_frame->color_space()->hdr_metadata());
+}
+
+TEST_P(TestVp9ImplForPixelFormat, DecodedQpEqualsEncodedQp) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  // First frame should be a key frame.
+  encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, 0));
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+  ASSERT_TRUE(decoded_qp);
+  EXPECT_EQ(encoded_frame.qp_, *decoded_qp);
+}
+
+TEST_F(TestVp9Impl, SwitchInputPixelFormatsWithoutReconfigure) {
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+
+  // Change the input frame type from I420 to NV12, encoding should still work.
+  input_frame_generator_ = test::CreateSquareFrameGenerator(
+      kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kNV12,
+      absl::optional<int>());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+
+  // Flipping back to I420, encoding should still work.
+  input_frame_generator_ = test::CreateSquareFrameGenerator(
+      kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kI420,
+      absl::optional<int>());
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+}
+
+TEST(Vp9ImplTest, ParserQpEqualsEncodedQp) {
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  encoder->InitEncode(&codec_settings, kSettings);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(1)
+          .SetResolution({kWidth, kHeight})
+          .Encode();
+  ASSERT_THAT(frames, SizeIs(1));
+  const auto& encoded_frame = frames.front().encoded_image;
+  int qp = 0;
+  ASSERT_TRUE(vp9::GetQp(encoded_frame.data(), encoded_frame.size(), &qp));
+  EXPECT_EQ(encoded_frame.qp_, qp);
+}
+
+TEST(Vp9ImplTest, EncodeAttachesTemplateStructureWithSvcController) {
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, kSettings),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(2)
+          .SetResolution({kWidth, kHeight})
+          .Encode();
+
+  ASSERT_THAT(frames, SizeIs(2));
+  EXPECT_TRUE(frames[0].codec_specific_info.template_structure);
+  EXPECT_TRUE(frames[0].codec_specific_info.generic_frame_info);
+
+  EXPECT_FALSE(frames[1].codec_specific_info.template_structure);
+  EXPECT_TRUE(frames[1].codec_specific_info.generic_frame_info);
+}
+
+TEST(Vp9ImplTest, EncoderWith2TemporalLayers) {
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.VP9()->numberOfTemporalLayers = 2;
+  // Tl0PidIdx is only used in non-flexible mode.
+  codec_settings.VP9()->flexibleMode = false;
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, kSettings),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(4)
+          .SetResolution({kWidth, kHeight})
+          .Encode();
+
+  ASSERT_THAT(frames, SizeIs(4));
+  EXPECT_EQ(frames[0].codec_specific_info.codecSpecific.VP9.temporal_idx, 0);
+  EXPECT_EQ(frames[1].codec_specific_info.codecSpecific.VP9.temporal_idx, 1);
+  EXPECT_EQ(frames[2].codec_specific_info.codecSpecific.VP9.temporal_idx, 0);
+  EXPECT_EQ(frames[3].codec_specific_info.codecSpecific.VP9.temporal_idx, 1);
+}
+
+TEST(Vp9ImplTest, EncodeTemporalLayersWithSvcController) {
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.VP9()->numberOfTemporalLayers = 2;
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, kSettings),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(4)
+          .SetResolution({kWidth, kHeight})
+          .Encode();
+
+  ASSERT_THAT(frames, SizeIs(4));
+  EXPECT_EQ(frames[0].codec_specific_info.codecSpecific.VP9.temporal_idx, 0);
+  EXPECT_EQ(frames[1].codec_specific_info.codecSpecific.VP9.temporal_idx, 1);
+  EXPECT_EQ(frames[2].codec_specific_info.codecSpecific.VP9.temporal_idx, 0);
+  EXPECT_EQ(frames[3].codec_specific_info.codecSpecific.VP9.temporal_idx, 1);
+  // Verify codec agnostic part
+  ASSERT_TRUE(frames[0].codec_specific_info.generic_frame_info);
+  ASSERT_TRUE(frames[1].codec_specific_info.generic_frame_info);
+  ASSERT_TRUE(frames[2].codec_specific_info.generic_frame_info);
+  ASSERT_TRUE(frames[3].codec_specific_info.generic_frame_info);
+  EXPECT_EQ(frames[0].codec_specific_info.generic_frame_info->temporal_id, 0);
+  EXPECT_EQ(frames[1].codec_specific_info.generic_frame_info->temporal_id, 1);
+  EXPECT_EQ(frames[2].codec_specific_info.generic_frame_info->temporal_id, 0);
+  EXPECT_EQ(frames[3].codec_specific_info.generic_frame_info->temporal_id, 1);
+}
+
+TEST(Vp9ImplTest, EncoderWith2SpatialLayers) {
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.VP9()->numberOfSpatialLayers = 2;
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, kSettings),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(1)
+          .SetResolution({kWidth, kHeight})
+          .Encode();
+
+  ASSERT_THAT(frames, SizeIs(2));
+  EXPECT_EQ(frames[0].encoded_image.SpatialIndex(), 0);
+  EXPECT_EQ(frames[1].encoded_image.SpatialIndex(), 1);
+}
+
+TEST(Vp9ImplTest, EncodeSpatialLayersWithSvcController) {
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.VP9()->numberOfSpatialLayers = 2;
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, kSettings),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(2)
+          .SetResolution({kWidth, kHeight})
+          .Encode();
+
+  ASSERT_THAT(frames, SizeIs(4));
+  EXPECT_EQ(frames[0].encoded_image.SpatialIndex(), 0);
+  EXPECT_EQ(frames[1].encoded_image.SpatialIndex(), 1);
+  EXPECT_EQ(frames[2].encoded_image.SpatialIndex(), 0);
+  EXPECT_EQ(frames[3].encoded_image.SpatialIndex(), 1);
+  // Verify codec agnostic part
+  ASSERT_TRUE(frames[0].codec_specific_info.generic_frame_info);
+  ASSERT_TRUE(frames[1].codec_specific_info.generic_frame_info);
+  ASSERT_TRUE(frames[2].codec_specific_info.generic_frame_info);
+  ASSERT_TRUE(frames[3].codec_specific_info.generic_frame_info);
+  EXPECT_EQ(frames[0].codec_specific_info.generic_frame_info->spatial_id, 0);
+  EXPECT_EQ(frames[1].codec_specific_info.generic_frame_info->spatial_id, 1);
+  EXPECT_EQ(frames[2].codec_specific_info.generic_frame_info->spatial_id, 0);
+  EXPECT_EQ(frames[3].codec_specific_info.generic_frame_info->spatial_id, 1);
+}
+
+TEST_F(TestVp9Impl, EncoderExplicitLayering) {
+  // Override default settings.
+  codec_settings_.VP9()->numberOfTemporalLayers = 1;
+  codec_settings_.VP9()->numberOfSpatialLayers = 2;
+
+  codec_settings_.width = 960;
+  codec_settings_.height = 540;
+  codec_settings_.spatialLayers[0].minBitrate = 200;
+  codec_settings_.spatialLayers[0].maxBitrate = 500;
+  codec_settings_.spatialLayers[0].targetBitrate =
+      (codec_settings_.spatialLayers[0].minBitrate +
+       codec_settings_.spatialLayers[0].maxBitrate) /
+      2;
+  codec_settings_.spatialLayers[0].active = true;
+
+  codec_settings_.spatialLayers[1].minBitrate = 400;
+  codec_settings_.spatialLayers[1].maxBitrate = 1500;
+  codec_settings_.spatialLayers[1].targetBitrate =
+      (codec_settings_.spatialLayers[1].minBitrate +
+       codec_settings_.spatialLayers[1].maxBitrate) /
+      2;
+  codec_settings_.spatialLayers[1].active = true;
+
+  codec_settings_.spatialLayers[0].width = codec_settings_.width / 2;
+  codec_settings_.spatialLayers[0].height = codec_settings_.height / 2;
+  codec_settings_.spatialLayers[0].maxFramerate = codec_settings_.maxFramerate;
+  codec_settings_.spatialLayers[1].width = codec_settings_.width;
+  codec_settings_.spatialLayers[1].height = codec_settings_.height;
+  codec_settings_.spatialLayers[1].maxFramerate = codec_settings_.maxFramerate;
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // Ensure it fails if scaling factors in horz/vert dimentions are different.
+  codec_settings_.spatialLayers[0].width = codec_settings_.width;
+  codec_settings_.spatialLayers[0].height = codec_settings_.height / 2;
+  codec_settings_.spatialLayers[1].width = codec_settings_.width;
+  codec_settings_.spatialLayers[1].height = codec_settings_.height;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_PARAMETER,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // Ensure it fails if scaling factor is not power of two.
+  codec_settings_.spatialLayers[0].width = codec_settings_.width / 3;
+  codec_settings_.spatialLayers[0].height = codec_settings_.height / 3;
+  codec_settings_.spatialLayers[1].width = codec_settings_.width;
+  codec_settings_.spatialLayers[1].height = codec_settings_.height;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_PARAMETER,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+}
+
+TEST_F(TestVp9Impl, EnableDisableSpatialLayers) {
+  // Configure encoder to produce N spatial layers. Encode frames of layer 0
+  // then enable layer 1 and encode more frames and so on until layer N-1.
+  // Then disable layers one by one in the same way.
+  // Note: bit rate allocation is high to avoid frame dropping due to rate
+  // control, the encoder should always produce a frame. A dropped
+  // frame indicates a problem and the test will fail.
+  const size_t num_spatial_layers = 3;
+  const size_t num_frames_to_encode = 5;
+
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+  codec_settings_.SetFrameDropEnabled(true);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    // Allocate high bit rate to avoid frame dropping due to rate control.
+    bitrate_allocation.SetBitrate(
+        sl_idx, 0,
+        codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000 * 2);
+    encoder_->SetRates(VideoEncoder::RateControlParameters(
+        bitrate_allocation, codec_settings_.maxFramerate));
+
+    for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+      SetWaitForEncodedFramesThreshold(sl_idx + 1);
+      EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+                encoder_->Encode(NextInputFrame(), nullptr));
+      std::vector<EncodedImage> encoded_frame;
+      std::vector<CodecSpecificInfo> codec_specific_info;
+      ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+      EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+                frame_num == 0);
+    }
+  }
+
+  for (size_t i = 0; i < num_spatial_layers - 1; ++i) {
+    const size_t sl_idx = num_spatial_layers - i - 1;
+    bitrate_allocation.SetBitrate(sl_idx, 0, 0);
+    encoder_->SetRates(VideoEncoder::RateControlParameters(
+        bitrate_allocation, codec_settings_.maxFramerate));
+
+    for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+      SetWaitForEncodedFramesThreshold(sl_idx);
+      EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+                encoder_->Encode(NextInputFrame(), nullptr));
+      std::vector<EncodedImage> encoded_frame;
+      std::vector<CodecSpecificInfo> codec_specific_info;
+      ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+      EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+                frame_num == 0);
+    }
+  }
+}
+
+TEST(Vp9ImplTest, EnableDisableSpatialLayersWithSvcController) {
+  const int num_spatial_layers = 3;
+  // Configure encoder to produce 3 spatial layers. Encode frames of layer 0
+  // then enable layer 1 and encode more frames and so on.
+  // Then disable layers one by one in the same way.
+  // Note: bit rate allocation is high to avoid frame dropping due to rate
+  // control, the encoder should always produce a frame. A dropped
+  // frame indicates a problem and the test will fail.
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  ConfigureSvc(codec_settings, num_spatial_layers);
+  codec_settings.SetFrameDropEnabled(true);
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, kSettings),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  EncodedVideoFrameProducer producer(*encoder);
+  producer.SetResolution({kWidth, kHeight});
+
+  // Encode a key frame to validate all other frames are delta frames.
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> frames =
+      producer.SetNumInputFrames(1).Encode();
+  ASSERT_THAT(frames, Not(IsEmpty()));
+  EXPECT_TRUE(frames[0].codec_specific_info.template_structure);
+
+  const size_t num_frames_to_encode = 5;
+
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    // Allocate high bit rate to avoid frame dropping due to rate control.
+    bitrate_allocation.SetBitrate(
+        sl_idx, 0,
+        codec_settings.spatialLayers[sl_idx].targetBitrate * 1000 * 2);
+    encoder->SetRates(VideoEncoder::RateControlParameters(
+        bitrate_allocation, codec_settings.maxFramerate));
+
+    frames = producer.SetNumInputFrames(num_frames_to_encode).Encode();
+    // With (sl_idx+1) spatial layers expect (sl_idx+1) frames per input frame.
+    ASSERT_THAT(frames, SizeIs(num_frames_to_encode * (sl_idx + 1)));
+    for (size_t i = 0; i < frames.size(); ++i) {
+      EXPECT_TRUE(frames[i].codec_specific_info.generic_frame_info);
+      EXPECT_FALSE(frames[i].codec_specific_info.template_structure);
+    }
+  }
+
+  for (int sl_idx = num_spatial_layers - 1; sl_idx > 0; --sl_idx) {
+    bitrate_allocation.SetBitrate(sl_idx, 0, 0);
+    encoder->SetRates(VideoEncoder::RateControlParameters(
+        bitrate_allocation, codec_settings.maxFramerate));
+
+    frames = producer.SetNumInputFrames(num_frames_to_encode).Encode();
+    // With `sl_idx` spatial layer disabled, there are `sl_idx` spatial layers
+    // left.
+    ASSERT_THAT(frames, SizeIs(num_frames_to_encode * sl_idx));
+    for (size_t i = 0; i < frames.size(); ++i) {
+      EXPECT_TRUE(frames[i].codec_specific_info.generic_frame_info);
+      EXPECT_FALSE(frames[i].codec_specific_info.template_structure);
+    }
+  }
+}
+
+MATCHER_P2(GenericLayerIs, spatial_id, temporal_id, "") {
+  if (arg.codec_specific_info.generic_frame_info == absl::nullopt) {
+    *result_listener << " miss generic_frame_info";
+    return false;
+  }
+  const auto& layer = *arg.codec_specific_info.generic_frame_info;
+  if (layer.spatial_id != spatial_id || layer.temporal_id != temporal_id) {
+    *result_listener << " frame from layer (" << layer.spatial_id << ", "
+                     << layer.temporal_id << ")";
+    return false;
+  }
+  return true;
+}
+
+TEST(Vp9ImplTest, SpatialUpswitchNotAtGOFBoundary) {
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  ConfigureSvc(codec_settings, /*num_spatial_layers=*/3,
+               /*num_temporal_layers=*/3);
+  codec_settings.SetFrameDropEnabled(true);
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, kSettings),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  EncodedVideoFrameProducer producer(*encoder);
+  producer.SetResolution({kWidth, kHeight});
+
+  // Disable all but spatial_layer = 0;
+  VideoBitrateAllocation bitrate_allocation;
+  int layer_bitrate_bps = codec_settings.spatialLayers[0].targetBitrate * 1000;
+  bitrate_allocation.SetBitrate(0, 0, layer_bitrate_bps);
+  bitrate_allocation.SetBitrate(0, 1, layer_bitrate_bps);
+  bitrate_allocation.SetBitrate(0, 2, layer_bitrate_bps);
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings.maxFramerate));
+  EXPECT_THAT(producer.SetNumInputFrames(3).Encode(),
+              ElementsAre(GenericLayerIs(0, 0), GenericLayerIs(0, 2),
+                          GenericLayerIs(0, 1)));
+
+  // Upswitch to spatial_layer = 1
+  layer_bitrate_bps = codec_settings.spatialLayers[1].targetBitrate * 1000;
+  bitrate_allocation.SetBitrate(1, 0, layer_bitrate_bps);
+  bitrate_allocation.SetBitrate(1, 1, layer_bitrate_bps);
+  bitrate_allocation.SetBitrate(1, 2, layer_bitrate_bps);
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings.maxFramerate));
+  // Expect upswitch doesn't happen immediately since there is no S1 frame that
+  // S1T2 frame can reference.
+  EXPECT_THAT(producer.SetNumInputFrames(1).Encode(),
+              ElementsAre(GenericLayerIs(0, 2)));
+  // Expect spatial upswitch happens now, at T0 frame.
+  EXPECT_THAT(producer.SetNumInputFrames(1).Encode(),
+              ElementsAre(GenericLayerIs(0, 0), GenericLayerIs(1, 0)));
+}
+// TODO(bugs.webrtc.org/13442) Enable once a forward fix has landed in WebRTC.
+TEST_F(TestVp9Impl, DISABLED_DisableEnableBaseLayerTriggersKeyFrame) {
+  // Configure encoder to produce N spatial layers. Encode frames for all
+  // layers. Then disable all but the last layer. Then reenable all back again.
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Vp9ExternalRefCtrl/Enabled/");
+  const size_t num_spatial_layers = 3;
+  const size_t num_temporal_layers = 3;
+  // Must not be multiple of temporal period to exercise all code paths.
+  const size_t num_frames_to_encode = 5;
+
+  ConfigureSvc(codec_settings_, num_spatial_layers, num_temporal_layers);
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.VP9()->flexibleMode = false;
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOnKeyPic;
+  codec_settings_.mode = VideoCodecMode::kRealtimeVideo;
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+      // Allocate high bit rate to avoid frame dropping due to rate control.
+      bitrate_allocation.SetBitrate(
+          sl_idx, tl_idx,
+          codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000 * 2);
+    }
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(num_spatial_layers);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frame;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+              frame_num == 0);
+  }
+
+  // Disable all but top layer.
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers - 1; ++sl_idx) {
+    for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+      bitrate_allocation.SetBitrate(sl_idx, tl_idx, 0);
+    }
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  bool seen_ss_data = false;
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(1);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frame;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    // SS available immediatly after switching on base temporal layer.
+    if (seen_ss_data) {
+      EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+                false);
+    } else {
+      EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+                codec_specific_info[0].codecSpecific.VP9.temporal_idx == 0);
+      seen_ss_data |=
+          codec_specific_info[0].codecSpecific.VP9.ss_data_available;
+    }
+    // No key-frames generated for disabling layers.
+    EXPECT_EQ(encoded_frame[0]._frameType, VideoFrameType::kVideoFrameDelta);
+    EXPECT_EQ(encoded_frame[0].SpatialIndex().value_or(-1), 2);
+  }
+  EXPECT_TRUE(seen_ss_data);
+
+  // Force key-frame.
+  std::vector<VideoFrameType> frame_types = {VideoFrameType::kVideoFrameKey};
+  SetWaitForEncodedFramesThreshold(1);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->Encode(NextInputFrame(), &frame_types));
+  std::vector<EncodedImage> encoded_frame;
+  std::vector<CodecSpecificInfo> codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+  // Key-frame should be produced.
+  EXPECT_EQ(encoded_frame[0]._frameType, VideoFrameType::kVideoFrameKey);
+  EXPECT_EQ(encoded_frame[0].SpatialIndex().value_or(-1), 2);
+
+  // Encode some more frames.
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(1);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frame;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    EXPECT_EQ(encoded_frame[0]._frameType, VideoFrameType::kVideoFrameDelta);
+    EXPECT_EQ(encoded_frame[0].SpatialIndex().value_or(-1), 2);
+  }
+
+  // Enable the second layer back.
+  // Allocate high bit rate to avoid frame dropping due to rate control.
+  for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+    bitrate_allocation.SetBitrate(
+        1, tl_idx, codec_settings_.spatialLayers[0].targetBitrate * 1000 * 2);
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(2);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frame;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    ASSERT_EQ(encoded_frame.size(), 2u);
+    // SS available immediatly after switching on.
+    EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+              frame_num == 0);
+    // Keyframe should be generated when enabling lower layers.
+    const VideoFrameType expected_type = frame_num == 0
+                                             ? VideoFrameType::kVideoFrameKey
+                                             : VideoFrameType::kVideoFrameDelta;
+    EXPECT_EQ(encoded_frame[0]._frameType, expected_type);
+    EXPECT_EQ(encoded_frame[0].SpatialIndex().value_or(-1), 1);
+    EXPECT_EQ(encoded_frame[1].SpatialIndex().value_or(-1), 2);
+  }
+
+  // Enable the first layer back.
+  // Allocate high bit rate to avoid frame dropping due to rate control.
+  for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+    bitrate_allocation.SetBitrate(
+        0, tl_idx, codec_settings_.spatialLayers[1].targetBitrate * 1000 * 2);
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(num_spatial_layers);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frame;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    ASSERT_EQ(encoded_frame.size(), 3u);
+    // SS available immediatly after switching on.
+    EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+              frame_num == 0);
+    // Keyframe should be generated when enabling lower layers.
+    const VideoFrameType expected_type = frame_num == 0
+                                             ? VideoFrameType::kVideoFrameKey
+                                             : VideoFrameType::kVideoFrameDelta;
+    EXPECT_EQ(encoded_frame[0]._frameType, expected_type);
+  }
+}
+// TODO(bugs.webrtc.org/13442) Enable once a forward fix has landed in WebRTC.
+TEST(Vp9ImplTest,
+     DISABLED_DisableEnableBaseLayerWithSvcControllerTriggersKeyFrame) {
+  // Configure encoder to produce N spatial layers. Encode frames for all
+  // layers. Then disable all but the last layer. Then reenable all back again.
+  const size_t num_spatial_layers = 3;
+  const size_t num_temporal_layers = 3;
+  // Must not be multiple of temporal period to exercise all code paths.
+  const size_t num_frames_to_encode = 5;
+
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  ConfigureSvc(codec_settings, num_spatial_layers, num_temporal_layers);
+  codec_settings.SetFrameDropEnabled(false);
+  codec_settings.VP9()->flexibleMode = false;
+  codec_settings.VP9()->interLayerPred = InterLayerPredMode::kOnKeyPic;
+  codec_settings.mode = VideoCodecMode::kRealtimeVideo;
+
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, kSettings),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+      // Allocate high bit rate to avoid frame dropping due to rate control.
+      bitrate_allocation.SetBitrate(
+          sl_idx, tl_idx,
+          codec_settings.spatialLayers[sl_idx].targetBitrate * 1000 * 2);
+    }
+  }
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings.maxFramerate));
+
+  EncodedVideoFrameProducer producer(*encoder);
+  producer.SetResolution({kWidth, kHeight});
+
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> frames =
+      producer.SetNumInputFrames(num_frames_to_encode).Encode();
+  ASSERT_THAT(frames, SizeIs(num_frames_to_encode * num_spatial_layers));
+
+  // Disable all but top spatial layer.
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers - 1; ++sl_idx) {
+    for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+      bitrate_allocation.SetBitrate(sl_idx, tl_idx, 0);
+    }
+  }
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings.maxFramerate));
+
+  frames = producer.SetNumInputFrames(num_frames_to_encode).Encode();
+  EXPECT_THAT(frames, SizeIs(num_frames_to_encode));
+  for (const auto& frame : frames) {
+    // Expect no key-frames generated.
+    EXPECT_FALSE(frame.codec_specific_info.template_structure);
+    ASSERT_TRUE(frame.codec_specific_info.generic_frame_info);
+    EXPECT_EQ(frame.codec_specific_info.generic_frame_info->spatial_id, 2);
+  }
+
+  frames = producer.ForceKeyFrame().SetNumInputFrames(1).Encode();
+  ASSERT_THAT(frames, SizeIs(1));
+  // Key-frame should be produced.
+  EXPECT_EQ(frames[0].encoded_image._frameType, VideoFrameType::kVideoFrameKey);
+  ASSERT_TRUE(frames[0].codec_specific_info.template_structure);
+  ASSERT_TRUE(frames[0].codec_specific_info.generic_frame_info);
+  EXPECT_EQ(frames[0].codec_specific_info.generic_frame_info->spatial_id, 2);
+
+  frames = producer.SetNumInputFrames(num_frames_to_encode).Encode();
+  ASSERT_THAT(frames, SizeIs(num_frames_to_encode));
+  for (const auto& frame : frames) {
+    EXPECT_EQ(frame.encoded_image._frameType, VideoFrameType::kVideoFrameDelta);
+    EXPECT_FALSE(frame.codec_specific_info.template_structure);
+    ASSERT_TRUE(frame.codec_specific_info.generic_frame_info);
+    EXPECT_EQ(frame.codec_specific_info.generic_frame_info->spatial_id, 2);
+  }
+
+  // Enable the second layer back.
+  // Allocate high bit rate to avoid frame dropping due to rate control.
+  for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+    bitrate_allocation.SetBitrate(
+        1, tl_idx, codec_settings.spatialLayers[0].targetBitrate * 1000 * 2);
+  }
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings.maxFramerate));
+
+  frames = producer.SetNumInputFrames(num_frames_to_encode).Encode();
+  ASSERT_THAT(frames, SizeIs(num_frames_to_encode * 2));
+  EXPECT_EQ(frames[0].encoded_image._frameType, VideoFrameType::kVideoFrameKey);
+  EXPECT_TRUE(frames[0].codec_specific_info.template_structure);
+  ASSERT_TRUE(frames[0].codec_specific_info.generic_frame_info);
+  EXPECT_EQ(frames[0].codec_specific_info.generic_frame_info->spatial_id, 1);
+  for (size_t i = 1; i < frames.size(); ++i) {
+    EXPECT_EQ(frames[i].encoded_image._frameType,
+              VideoFrameType::kVideoFrameDelta);
+    EXPECT_FALSE(frames[i].codec_specific_info.template_structure);
+    ASSERT_TRUE(frames[i].codec_specific_info.generic_frame_info);
+    EXPECT_EQ(frames[i].codec_specific_info.generic_frame_info->spatial_id,
+              1 + static_cast<int>(i % 2));
+  }
+
+  // Enable the first layer back.
+  // Allocate high bit rate to avoid frame dropping due to rate control.
+  for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+    bitrate_allocation.SetBitrate(
+        0, tl_idx, codec_settings.spatialLayers[1].targetBitrate * 1000 * 2);
+  }
+  encoder->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings.maxFramerate));
+
+  frames = producer.SetNumInputFrames(num_frames_to_encode).Encode();
+  ASSERT_THAT(frames, SizeIs(num_frames_to_encode * 3));
+  EXPECT_TRUE(frames[0].codec_specific_info.template_structure);
+  ASSERT_TRUE(frames[0].codec_specific_info.generic_frame_info);
+  EXPECT_EQ(frames[0].codec_specific_info.generic_frame_info->spatial_id, 0);
+  for (size_t i = 1; i < frames.size(); ++i) {
+    EXPECT_FALSE(frames[i].codec_specific_info.template_structure);
+    ASSERT_TRUE(frames[i].codec_specific_info.generic_frame_info);
+    EXPECT_EQ(frames[i].codec_specific_info.generic_frame_info->spatial_id,
+              static_cast<int>(i % 3));
+  }
+}
+
+TEST_F(TestVp9Impl, DisableEnableBaseLayerTriggersKeyFrameForScreenshare) {
+  // Configure encoder to produce N spatial layers. Encode frames for all
+  // layers. Then disable all but the last layer. Then reenable all back again.
+  const size_t num_spatial_layers = 3;
+  const size_t num_frames_to_encode = 5;
+
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.mode = VideoCodecMode::kScreensharing;
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOn;
+  codec_settings_.VP9()->flexibleMode = true;
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    // Allocate high bit rate to avoid frame dropping due to rate control.
+    bitrate_allocation.SetBitrate(
+        sl_idx, 0,
+        codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000 * 2);
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(num_spatial_layers);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frame;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+              frame_num == 0);
+  }
+
+  // Disable all but top layer.
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers - 1; ++sl_idx) {
+    bitrate_allocation.SetBitrate(sl_idx, 0, 0);
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(1);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frame;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    // SS available immediatly after switching off.
+    EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+              frame_num == 0);
+    // No key-frames generated for disabling layers.
+    EXPECT_EQ(encoded_frame[0]._frameType, VideoFrameType::kVideoFrameDelta);
+    EXPECT_EQ(encoded_frame[0].SpatialIndex().value_or(-1), 2);
+  }
+
+  // Force key-frame.
+  std::vector<VideoFrameType> frame_types = {VideoFrameType::kVideoFrameKey};
+  SetWaitForEncodedFramesThreshold(1);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->Encode(NextInputFrame(), &frame_types));
+  std::vector<EncodedImage> encoded_frame;
+  std::vector<CodecSpecificInfo> codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+  // Key-frame should be produced.
+  EXPECT_EQ(encoded_frame[0]._frameType, VideoFrameType::kVideoFrameKey);
+
+  // Enable the second layer back.
+  // Allocate high bit rate to avoid frame dropping due to rate control.
+  bitrate_allocation.SetBitrate(
+      1, 0, codec_settings_.spatialLayers[0].targetBitrate * 1000 * 2);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(2);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frame;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    ASSERT_EQ(encoded_frame.size(), 2u);
+    // SS available immediatly after switching on.
+    EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+              frame_num == 0);
+    // Keyframe should be generated when enabling lower layers.
+    const VideoFrameType expected_type = frame_num == 0
+                                             ? VideoFrameType::kVideoFrameKey
+                                             : VideoFrameType::kVideoFrameDelta;
+    EXPECT_EQ(encoded_frame[0]._frameType, expected_type);
+    EXPECT_EQ(encoded_frame[0].SpatialIndex().value_or(-1), 1);
+    EXPECT_EQ(encoded_frame[1].SpatialIndex().value_or(-1), 2);
+  }
+
+  // Enable the first layer back.
+  // Allocate high bit rate to avoid frame dropping due to rate control.
+  bitrate_allocation.SetBitrate(
+      0, 0, codec_settings_.spatialLayers[1].targetBitrate * 1000 * 2);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(num_spatial_layers);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frame;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    ASSERT_EQ(encoded_frame.size(), 3u);
+    // SS available immediatly after switching on.
+    EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.ss_data_available,
+              frame_num == 0);
+    // Keyframe should be generated when enabling lower layers.
+    const VideoFrameType expected_type = frame_num == 0
+                                             ? VideoFrameType::kVideoFrameKey
+                                             : VideoFrameType::kVideoFrameDelta;
+    EXPECT_EQ(encoded_frame[0]._frameType, expected_type);
+  }
+}
+
+TEST_F(TestVp9Impl, EndOfPicture) {
+  const size_t num_spatial_layers = 2;
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // Encode both base and upper layers. Check that end-of-superframe flag is
+  // set on upper layer frame but not on base layer frame.
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(
+      0, 0, codec_settings_.spatialLayers[0].targetBitrate * 1000);
+  bitrate_allocation.SetBitrate(
+      1, 0, codec_settings_.spatialLayers[1].targetBitrate * 1000);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+  SetWaitForEncodedFramesThreshold(2);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+
+  std::vector<EncodedImage> frames;
+  std::vector<CodecSpecificInfo> codec_specific;
+  ASSERT_TRUE(WaitForEncodedFrames(&frames, &codec_specific));
+  EXPECT_FALSE(codec_specific[0].end_of_picture);
+  EXPECT_TRUE(codec_specific[1].end_of_picture);
+
+  // Encode only base layer. Check that end-of-superframe flag is
+  // set on base layer frame.
+  bitrate_allocation.SetBitrate(1, 0, 0);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  SetWaitForEncodedFramesThreshold(1);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+
+  ASSERT_TRUE(WaitForEncodedFrames(&frames, &codec_specific));
+  EXPECT_FALSE(frames[0].SpatialIndex());
+  EXPECT_TRUE(codec_specific[0].end_of_picture);
+}
+
+TEST_F(TestVp9Impl, InterLayerPred) {
+  const size_t num_spatial_layers = 2;
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+  codec_settings_.SetFrameDropEnabled(false);
+
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t i = 0; i < num_spatial_layers; ++i) {
+    bitrate_allocation.SetBitrate(
+        i, 0, codec_settings_.spatialLayers[i].targetBitrate * 1000);
+  }
+
+  const std::vector<InterLayerPredMode> inter_layer_pred_modes = {
+      InterLayerPredMode::kOff, InterLayerPredMode::kOn,
+      InterLayerPredMode::kOnKeyPic};
+
+  for (const InterLayerPredMode inter_layer_pred : inter_layer_pred_modes) {
+    codec_settings_.VP9()->interLayerPred = inter_layer_pred;
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->InitEncode(&codec_settings_, kSettings));
+
+    encoder_->SetRates(VideoEncoder::RateControlParameters(
+        bitrate_allocation, codec_settings_.maxFramerate));
+
+    SetWaitForEncodedFramesThreshold(2);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+
+    std::vector<EncodedImage> frames;
+    std::vector<CodecSpecificInfo> codec_specific;
+    ASSERT_TRUE(WaitForEncodedFrames(&frames, &codec_specific));
+
+    // Key frame.
+    ASSERT_EQ(frames[0].SpatialIndex(), 0);
+    ASSERT_FALSE(codec_specific[0].codecSpecific.VP9.inter_pic_predicted);
+    EXPECT_FALSE(codec_specific[0].codecSpecific.VP9.inter_layer_predicted);
+    EXPECT_EQ(codec_specific[0].codecSpecific.VP9.non_ref_for_inter_layer_pred,
+              inter_layer_pred == InterLayerPredMode::kOff);
+    EXPECT_TRUE(codec_specific[0].codecSpecific.VP9.ss_data_available);
+
+    ASSERT_EQ(frames[1].SpatialIndex(), 1);
+    ASSERT_FALSE(codec_specific[1].codecSpecific.VP9.inter_pic_predicted);
+    EXPECT_EQ(codec_specific[1].codecSpecific.VP9.inter_layer_predicted,
+              inter_layer_pred == InterLayerPredMode::kOn ||
+                  inter_layer_pred == InterLayerPredMode::kOnKeyPic);
+    EXPECT_EQ(codec_specific[1].codecSpecific.VP9.ss_data_available,
+              inter_layer_pred == InterLayerPredMode::kOff);
+    EXPECT_TRUE(
+        codec_specific[1].codecSpecific.VP9.non_ref_for_inter_layer_pred);
+
+    // Delta frame.
+    SetWaitForEncodedFramesThreshold(2);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    ASSERT_TRUE(WaitForEncodedFrames(&frames, &codec_specific));
+
+    ASSERT_EQ(frames[0].SpatialIndex(), 0);
+    ASSERT_TRUE(codec_specific[0].codecSpecific.VP9.inter_pic_predicted);
+    EXPECT_FALSE(codec_specific[0].codecSpecific.VP9.inter_layer_predicted);
+    EXPECT_EQ(codec_specific[0].codecSpecific.VP9.non_ref_for_inter_layer_pred,
+              inter_layer_pred != InterLayerPredMode::kOn);
+    EXPECT_FALSE(codec_specific[0].codecSpecific.VP9.ss_data_available);
+
+    ASSERT_EQ(frames[1].SpatialIndex(), 1);
+    ASSERT_TRUE(codec_specific[1].codecSpecific.VP9.inter_pic_predicted);
+    EXPECT_EQ(codec_specific[1].codecSpecific.VP9.inter_layer_predicted,
+              inter_layer_pred == InterLayerPredMode::kOn);
+    EXPECT_TRUE(
+        codec_specific[1].codecSpecific.VP9.non_ref_for_inter_layer_pred);
+    EXPECT_FALSE(codec_specific[1].codecSpecific.VP9.ss_data_available);
+  }
+}
+
+TEST_F(TestVp9Impl,
+       EnablingUpperLayerTriggersKeyFrameIfInterLayerPredIsDisabled) {
+  const size_t num_spatial_layers = 3;
+  const size_t num_frames_to_encode = 2;
+
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+  codec_settings_.SetFrameDropEnabled(false);
+
+  const std::vector<InterLayerPredMode> inter_layer_pred_modes = {
+      InterLayerPredMode::kOff, InterLayerPredMode::kOn,
+      InterLayerPredMode::kOnKeyPic};
+
+  for (const InterLayerPredMode inter_layer_pred : inter_layer_pred_modes) {
+    codec_settings_.VP9()->interLayerPred = inter_layer_pred;
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->InitEncode(&codec_settings_, kSettings));
+
+    VideoBitrateAllocation bitrate_allocation;
+    for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+      bitrate_allocation.SetBitrate(
+          sl_idx, 0,
+          codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000);
+      encoder_->SetRates(VideoEncoder::RateControlParameters(
+          bitrate_allocation, codec_settings_.maxFramerate));
+
+      for (size_t frame_num = 0; frame_num < num_frames_to_encode;
+           ++frame_num) {
+        SetWaitForEncodedFramesThreshold(sl_idx + 1);
+        EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+                  encoder_->Encode(NextInputFrame(), nullptr));
+        std::vector<EncodedImage> encoded_frame;
+        std::vector<CodecSpecificInfo> codec_specific_info;
+        ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+
+        const bool is_first_upper_layer_frame = (sl_idx > 0 && frame_num == 0);
+        if (is_first_upper_layer_frame) {
+          if (inter_layer_pred == InterLayerPredMode::kOn) {
+            EXPECT_EQ(encoded_frame[0]._frameType,
+                      VideoFrameType::kVideoFrameDelta);
+          } else {
+            EXPECT_EQ(encoded_frame[0]._frameType,
+                      VideoFrameType::kVideoFrameKey);
+          }
+        } else if (sl_idx == 0 && frame_num == 0) {
+          EXPECT_EQ(encoded_frame[0]._frameType,
+                    VideoFrameType::kVideoFrameKey);
+        } else {
+          for (size_t i = 0; i <= sl_idx; ++i) {
+            EXPECT_EQ(encoded_frame[i]._frameType,
+                      VideoFrameType::kVideoFrameDelta);
+          }
+        }
+      }
+    }
+  }
+}
+
+TEST_F(TestVp9Impl,
+       EnablingUpperLayerUnsetsInterPicPredictedInInterlayerPredModeOn) {
+  const size_t num_spatial_layers = 3;
+  const size_t num_frames_to_encode = 2;
+
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.VP9()->flexibleMode = false;
+
+  const std::vector<InterLayerPredMode> inter_layer_pred_modes = {
+      InterLayerPredMode::kOff, InterLayerPredMode::kOn,
+      InterLayerPredMode::kOnKeyPic};
+
+  for (const InterLayerPredMode inter_layer_pred : inter_layer_pred_modes) {
+    codec_settings_.VP9()->interLayerPred = inter_layer_pred;
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->InitEncode(&codec_settings_, kSettings));
+
+    VideoBitrateAllocation bitrate_allocation;
+    for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+      bitrate_allocation.SetBitrate(
+          sl_idx, 0,
+          codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000);
+      encoder_->SetRates(VideoEncoder::RateControlParameters(
+          bitrate_allocation, codec_settings_.maxFramerate));
+
+      for (size_t frame_num = 0; frame_num < num_frames_to_encode;
+           ++frame_num) {
+        SetWaitForEncodedFramesThreshold(sl_idx + 1);
+        EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+                  encoder_->Encode(NextInputFrame(), nullptr));
+        std::vector<EncodedImage> encoded_frame;
+        std::vector<CodecSpecificInfo> codec_specific_info;
+        ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+
+        ASSERT_EQ(codec_specific_info.size(), sl_idx + 1);
+
+        for (size_t i = 0; i <= sl_idx; ++i) {
+          const bool is_keyframe =
+              encoded_frame[0]._frameType == VideoFrameType::kVideoFrameKey;
+          const bool is_first_upper_layer_frame =
+              (i == sl_idx && frame_num == 0);
+          // Interframe references are there, unless it's a keyframe,
+          // or it's a first activated frame in a upper layer
+          const bool expect_no_references =
+              is_keyframe || (is_first_upper_layer_frame &&
+                              inter_layer_pred == InterLayerPredMode::kOn);
+          EXPECT_EQ(
+              codec_specific_info[i].codecSpecific.VP9.inter_pic_predicted,
+              !expect_no_references);
+        }
+      }
+    }
+  }
+}
+
+TEST_F(TestVp9Impl, EnablingDisablingUpperLayerInTheSameGof) {
+  const size_t num_spatial_layers = 2;
+  const size_t num_temporal_layers = 2;
+
+  ConfigureSvc(codec_settings_, num_spatial_layers, num_temporal_layers);
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.VP9()->flexibleMode = false;
+
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOn;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoBitrateAllocation bitrate_allocation;
+
+  // Enable both spatial and both temporal layers.
+  bitrate_allocation.SetBitrate(
+      0, 0, codec_settings_.spatialLayers[0].targetBitrate * 1000 / 2);
+  bitrate_allocation.SetBitrate(
+      0, 1, codec_settings_.spatialLayers[0].targetBitrate * 1000 / 2);
+  bitrate_allocation.SetBitrate(
+      1, 0, codec_settings_.spatialLayers[1].targetBitrate * 1000 / 2);
+  bitrate_allocation.SetBitrate(
+      1, 1, codec_settings_.spatialLayers[1].targetBitrate * 1000 / 2);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  std::vector<EncodedImage> encoded_frame;
+  std::vector<CodecSpecificInfo> codec_specific_info;
+
+  // Encode 3 frames.
+  for (int i = 0; i < 3; ++i) {
+    SetWaitForEncodedFramesThreshold(2);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    ASSERT_EQ(codec_specific_info.size(), 2u);
+  }
+
+  // Disable SL1 layer.
+  bitrate_allocation.SetBitrate(1, 0, 0);
+  bitrate_allocation.SetBitrate(1, 1, 0);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  // Encode 1 frame.
+  SetWaitForEncodedFramesThreshold(1);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+  ASSERT_EQ(codec_specific_info.size(), 1u);
+  EXPECT_EQ(encoded_frame[0]._frameType, VideoFrameType::kVideoFrameDelta);
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.temporal_idx, 1);
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.inter_pic_predicted, true);
+
+  // Enable SL1 layer.
+  bitrate_allocation.SetBitrate(
+      1, 0, codec_settings_.spatialLayers[1].targetBitrate * 1000 / 2);
+  bitrate_allocation.SetBitrate(
+      1, 1, codec_settings_.spatialLayers[1].targetBitrate * 1000 / 2);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  // Encode 1 frame.
+  SetWaitForEncodedFramesThreshold(2);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+  ASSERT_EQ(codec_specific_info.size(), 2u);
+  EXPECT_EQ(encoded_frame[0]._frameType, VideoFrameType::kVideoFrameDelta);
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.temporal_idx, 0);
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.inter_pic_predicted, true);
+  EXPECT_EQ(codec_specific_info[1].codecSpecific.VP9.inter_pic_predicted, true);
+}
+
+TEST_F(TestVp9Impl, EnablingDisablingUpperLayerAccrossGof) {
+  const size_t num_spatial_layers = 2;
+  const size_t num_temporal_layers = 2;
+
+  ConfigureSvc(codec_settings_, num_spatial_layers, num_temporal_layers);
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.VP9()->flexibleMode = false;
+
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOn;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoBitrateAllocation bitrate_allocation;
+
+  // Enable both spatial and both temporal layers.
+  bitrate_allocation.SetBitrate(
+      0, 0, codec_settings_.spatialLayers[0].targetBitrate * 1000 / 2);
+  bitrate_allocation.SetBitrate(
+      0, 1, codec_settings_.spatialLayers[0].targetBitrate * 1000 / 2);
+  bitrate_allocation.SetBitrate(
+      1, 0, codec_settings_.spatialLayers[1].targetBitrate * 1000 / 2);
+  bitrate_allocation.SetBitrate(
+      1, 1, codec_settings_.spatialLayers[1].targetBitrate * 1000 / 2);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  std::vector<EncodedImage> encoded_frame;
+  std::vector<CodecSpecificInfo> codec_specific_info;
+
+  // Encode 3 frames.
+  for (int i = 0; i < 3; ++i) {
+    SetWaitForEncodedFramesThreshold(2);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    ASSERT_EQ(codec_specific_info.size(), 2u);
+  }
+
+  // Disable SL1 layer.
+  bitrate_allocation.SetBitrate(1, 0, 0);
+  bitrate_allocation.SetBitrate(1, 1, 0);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  // Encode 11 frames. More than Gof length 2, and odd to end at TL1 frame.
+  for (int i = 0; i < 11; ++i) {
+    SetWaitForEncodedFramesThreshold(1);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+    ASSERT_EQ(codec_specific_info.size(), 1u);
+    EXPECT_EQ(encoded_frame[0]._frameType, VideoFrameType::kVideoFrameDelta);
+    EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.temporal_idx, 1 - i % 2);
+    EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.inter_pic_predicted,
+              true);
+  }
+
+  // Enable SL1 layer.
+  bitrate_allocation.SetBitrate(
+      1, 0, codec_settings_.spatialLayers[1].targetBitrate * 1000 / 2);
+  bitrate_allocation.SetBitrate(
+      1, 1, codec_settings_.spatialLayers[1].targetBitrate * 1000 / 2);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  // Encode 1 frame.
+  SetWaitForEncodedFramesThreshold(2);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
+  ASSERT_EQ(codec_specific_info.size(), 2u);
+  EXPECT_EQ(encoded_frame[0]._frameType, VideoFrameType::kVideoFrameDelta);
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.temporal_idx, 0);
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.inter_pic_predicted, true);
+  EXPECT_EQ(codec_specific_info[1].codecSpecific.VP9.inter_pic_predicted,
+            false);
+}
+
+TEST_F(TestVp9Impl, EnablingNewLayerInScreenshareForcesAllLayersWithSS) {
+  const size_t num_spatial_layers = 3;
+  // Chosen by hand, the 2nd frame is dropped with configured per-layer max
+  // framerate.
+  const size_t num_frames_to_encode_before_drop = 1;
+
+  codec_settings_.maxFramerate = 30;
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+  codec_settings_.spatialLayers[0].maxFramerate = 5.0;
+  // use 30 for the SL 1 instead of 10, so even if SL 0 frame is dropped due to
+  // framerate capping we would still get back at least a middle layer. It
+  // simplifies the test.
+  codec_settings_.spatialLayers[1].maxFramerate = 30.0;
+  codec_settings_.spatialLayers[2].maxFramerate = 30.0;
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.mode = VideoCodecMode::kScreensharing;
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOn;
+  codec_settings_.VP9()->flexibleMode = true;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // Enable all but the last layer.
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers - 1; ++sl_idx) {
+    bitrate_allocation.SetBitrate(
+        sl_idx, 0, codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000);
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  // Encode enough frames to force drop due to framerate capping.
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode_before_drop;
+       ++frame_num) {
+    SetWaitForEncodedFramesThreshold(num_spatial_layers - 1);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frames;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+  }
+
+  // Enable the last layer.
+  bitrate_allocation.SetBitrate(
+      num_spatial_layers - 1, 0,
+      codec_settings_.spatialLayers[num_spatial_layers - 1].targetBitrate *
+          1000);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  // All layers are encoded, even though frame dropping should happen.
+  SetWaitForEncodedFramesThreshold(num_spatial_layers);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  // Now all 3 layers should be encoded.
+  std::vector<EncodedImage> encoded_frames;
+  std::vector<CodecSpecificInfo> codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+  EXPECT_EQ(encoded_frames.size(), 3u);
+  // Scalability structure has to be triggered.
+  EXPECT_TRUE(codec_specific_info[0].codecSpecific.VP9.ss_data_available);
+}
+
+TEST_F(TestVp9Impl, ScreenshareFrameDropping) {
+  const int num_spatial_layers = 3;
+  const int num_frames_to_detect_drops = 2;
+
+  codec_settings_.maxFramerate = 30;
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+  // use 30 for the SL0 and SL1 because it simplifies the test.
+  codec_settings_.spatialLayers[0].maxFramerate = 30.0;
+  codec_settings_.spatialLayers[1].maxFramerate = 30.0;
+  codec_settings_.spatialLayers[2].maxFramerate = 30.0;
+  codec_settings_.SetFrameDropEnabled(true);
+  codec_settings_.mode = VideoCodecMode::kScreensharing;
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOn;
+  codec_settings_.VP9()->flexibleMode = true;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // Enable all but the last layer.
+  VideoBitrateAllocation bitrate_allocation;
+  // Very low bitrate for the lowest spatial layer to ensure rate-control drops.
+  bitrate_allocation.SetBitrate(0, 0, 1000);
+  bitrate_allocation.SetBitrate(
+      1, 0, codec_settings_.spatialLayers[1].targetBitrate * 1000);
+  // Disable highest layer.
+  bitrate_allocation.SetBitrate(2, 0, 0);
+
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  bool frame_dropped = false;
+  // Encode enough frames to force drop due to rate-control.
+  for (size_t frame_num = 0; frame_num < num_frames_to_detect_drops;
+       ++frame_num) {
+    SetWaitForEncodedFramesThreshold(1);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frames;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+    EXPECT_LE(encoded_frames.size(), 2u);
+    EXPECT_GE(encoded_frames.size(), 1u);
+    if (encoded_frames.size() == 1) {
+      frame_dropped = true;
+      // Dropped frame is on the SL0.
+      EXPECT_EQ(encoded_frames[0].SpatialIndex(), 1);
+    }
+  }
+  EXPECT_TRUE(frame_dropped);
+
+  // Enable the last layer.
+  bitrate_allocation.SetBitrate(
+      2, 0, codec_settings_.spatialLayers[2].targetBitrate * 1000);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+  SetWaitForEncodedFramesThreshold(1);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  std::vector<EncodedImage> encoded_frames;
+  std::vector<CodecSpecificInfo> codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+  // No drop allowed.
+  EXPECT_EQ(encoded_frames.size(), 3u);
+
+  // Verify that frame-dropping is re-enabled back.
+  frame_dropped = false;
+  // Encode enough frames to force drop due to rate-control.
+  for (size_t frame_num = 0; frame_num < num_frames_to_detect_drops;
+       ++frame_num) {
+    SetWaitForEncodedFramesThreshold(1);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frames;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+    EXPECT_LE(encoded_frames.size(), 3u);
+    EXPECT_GE(encoded_frames.size(), 2u);
+    if (encoded_frames.size() == 2) {
+      frame_dropped = true;
+      // Dropped frame is on the SL0.
+      EXPECT_EQ(encoded_frames[0].SpatialIndex(), 1);
+      EXPECT_EQ(encoded_frames[1].SpatialIndex(), 2);
+    }
+  }
+  EXPECT_TRUE(frame_dropped);
+}
+
+TEST_F(TestVp9Impl, RemovingLayerIsNotDelayedInScreenshareAndAddsSsInfo) {
+  const size_t num_spatial_layers = 3;
+  // Chosen by hand, the 2nd frame is dropped with configured per-layer max
+  // framerate.
+  const size_t num_frames_to_encode_before_drop = 1;
+  // Chosen by hand, exactly 5 frames are dropped for input fps=30 and max
+  // framerate = 5.
+  const size_t num_dropped_frames = 5;
+
+  codec_settings_.maxFramerate = 30;
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+  codec_settings_.spatialLayers[0].maxFramerate = 5.0;
+  // use 30 for the SL 1 instead of 5, so even if SL 0 frame is dropped due to
+  // framerate capping we would still get back at least a middle layer. It
+  // simplifies the test.
+  codec_settings_.spatialLayers[1].maxFramerate = 30.0;
+  codec_settings_.spatialLayers[2].maxFramerate = 30.0;
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.mode = VideoCodecMode::kScreensharing;
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOn;
+  codec_settings_.VP9()->flexibleMode = true;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // All layers are enabled from the start.
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    bitrate_allocation.SetBitrate(
+        sl_idx, 0, codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000);
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  // Encode enough frames to force drop due to framerate capping.
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode_before_drop;
+       ++frame_num) {
+    SetWaitForEncodedFramesThreshold(num_spatial_layers);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    std::vector<EncodedImage> encoded_frames;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+  }
+
+  // Now the first layer should not have frames in it.
+  for (size_t frame_num = 0; frame_num < num_dropped_frames - 2; ++frame_num) {
+    SetWaitForEncodedFramesThreshold(2);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    // First layer is dropped due to frame rate cap. The last layer should not
+    // be enabled yet.
+    std::vector<EncodedImage> encoded_frames;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+    // First layer is skipped.
+    EXPECT_EQ(encoded_frames[0].SpatialIndex().value_or(-1), 1);
+  }
+
+  // Disable the last layer.
+  bitrate_allocation.SetBitrate(num_spatial_layers - 1, 0, 0);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  // Still expected to drop first layer. Last layer has to be disable also.
+  for (size_t frame_num = num_dropped_frames - 2;
+       frame_num < num_dropped_frames; ++frame_num) {
+    // Expect back one frame.
+    SetWaitForEncodedFramesThreshold(1);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    // First layer is dropped due to frame rate cap. The last layer should not
+    // be enabled yet.
+    std::vector<EncodedImage> encoded_frames;
+    std::vector<CodecSpecificInfo> codec_specific_info;
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+    // First layer is skipped.
+    EXPECT_EQ(encoded_frames[0].SpatialIndex().value_or(-1), 1);
+    // No SS data on non-base spatial layer.
+    EXPECT_FALSE(codec_specific_info[0].codecSpecific.VP9.ss_data_available);
+  }
+
+  SetWaitForEncodedFramesThreshold(2);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  std::vector<EncodedImage> encoded_frames;
+  std::vector<CodecSpecificInfo> codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+  // First layer is not skipped now.
+  EXPECT_EQ(encoded_frames[0].SpatialIndex().value_or(-1), 0);
+  // SS data should be present.
+  EXPECT_TRUE(codec_specific_info[0].codecSpecific.VP9.ss_data_available);
+}
+
+TEST_F(TestVp9Impl, DisableNewLayerInVideoDelaysSsInfoTillTL0) {
+  const size_t num_spatial_layers = 3;
+  const size_t num_temporal_layers = 2;
+  // Chosen by hand, the 2nd frame is dropped with configured per-layer max
+  // framerate.
+  ConfigureSvc(codec_settings_, num_spatial_layers, num_temporal_layers);
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.mode = VideoCodecMode::kRealtimeVideo;
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOnKeyPic;
+  codec_settings_.VP9()->flexibleMode = false;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // Enable all the layers.
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+      bitrate_allocation.SetBitrate(
+          sl_idx, tl_idx,
+          codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000 /
+              num_temporal_layers);
+    }
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  std::vector<EncodedImage> encoded_frames;
+  std::vector<CodecSpecificInfo> codec_specific_info;
+
+  // Encode one TL0 frame
+  SetWaitForEncodedFramesThreshold(num_spatial_layers);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.temporal_idx, 0u);
+
+  // Disable the last layer.
+  for (size_t tl_idx = 0; tl_idx < num_temporal_layers; ++tl_idx) {
+    bitrate_allocation.SetBitrate(num_spatial_layers - 1, tl_idx, 0);
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  // Next is TL1 frame. The last layer is disabled immediately, but SS structure
+  // is not provided here.
+  SetWaitForEncodedFramesThreshold(num_spatial_layers - 1);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.temporal_idx, 1u);
+  EXPECT_FALSE(codec_specific_info[0].codecSpecific.VP9.ss_data_available);
+
+  // Next is TL0 frame, which should have delayed SS structure.
+  SetWaitForEncodedFramesThreshold(num_spatial_layers - 1);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_info));
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.temporal_idx, 0u);
+  EXPECT_TRUE(codec_specific_info[0].codecSpecific.VP9.ss_data_available);
+  EXPECT_TRUE(codec_specific_info[0]
+                  .codecSpecific.VP9.spatial_layer_resolution_present);
+  EXPECT_EQ(codec_specific_info[0].codecSpecific.VP9.num_spatial_layers,
+            num_spatial_layers - 1);
+}
+
+TEST_F(TestVp9Impl,
+       LowLayerMarkedAsRefIfHighLayerNotEncodedAndInterLayerPredIsEnabled) {
+  ConfigureSvc(codec_settings_, 3);
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOn;
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoBitrateAllocation bitrate_allocation;
+  bitrate_allocation.SetBitrate(
+      0, 0, codec_settings_.spatialLayers[0].targetBitrate * 1000);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_info));
+  EXPECT_TRUE(codec_info.codecSpecific.VP9.ss_data_available);
+  EXPECT_FALSE(codec_info.codecSpecific.VP9.non_ref_for_inter_layer_pred);
+}
+
+TEST_F(TestVp9Impl, ScalabilityStructureIsAvailableInFlexibleMode) {
+  codec_settings_.VP9()->flexibleMode = true;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  EXPECT_TRUE(codec_specific_info.codecSpecific.VP9.ss_data_available);
+}
+
+TEST_F(TestVp9Impl, Profile0PreferredPixelFormats) {
+  EXPECT_THAT(encoder_->GetEncoderInfo().preferred_pixel_formats,
+              testing::UnorderedElementsAre(VideoFrameBuffer::Type::kNV12,
+                                            VideoFrameBuffer::Type::kI420));
+}
+
+TEST_F(TestVp9Impl, EncoderInfoWithoutResolutionBitrateLimits) {
+  EXPECT_TRUE(encoder_->GetEncoderInfo().resolution_bitrate_limits.empty());
+}
+
+TEST_F(TestVp9Impl, EncoderInfoWithBitrateLimitsFromFieldTrial) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VP9-GetEncoderInfoOverride/"
+      "frame_size_pixels:123|456|789,"
+      "min_start_bitrate_bps:11000|22000|33000,"
+      "min_bitrate_bps:44000|55000|66000,"
+      "max_bitrate_bps:77000|88000|99000/");
+  SetUp();
+
+  EXPECT_THAT(
+      encoder_->GetEncoderInfo().resolution_bitrate_limits,
+      ::testing::ElementsAre(
+          VideoEncoder::ResolutionBitrateLimits{123, 11000, 44000, 77000},
+          VideoEncoder::ResolutionBitrateLimits{456, 22000, 55000, 88000},
+          VideoEncoder::ResolutionBitrateLimits{789, 33000, 66000, 99000}));
+}
+
+TEST_F(TestVp9Impl, EncoderInfoFpsAllocation) {
+  const uint8_t kNumSpatialLayers = 3;
+  const uint8_t kNumTemporalLayers = 3;
+
+  codec_settings_.maxFramerate = 30;
+  codec_settings_.VP9()->numberOfSpatialLayers = kNumSpatialLayers;
+  codec_settings_.VP9()->numberOfTemporalLayers = kNumTemporalLayers;
+
+  for (uint8_t sl_idx = 0; sl_idx < kNumSpatialLayers; ++sl_idx) {
+    codec_settings_.spatialLayers[sl_idx].width = codec_settings_.width;
+    codec_settings_.spatialLayers[sl_idx].height = codec_settings_.height;
+    codec_settings_.spatialLayers[sl_idx].minBitrate =
+        codec_settings_.startBitrate;
+    codec_settings_.spatialLayers[sl_idx].maxBitrate =
+        codec_settings_.startBitrate;
+    codec_settings_.spatialLayers[sl_idx].targetBitrate =
+        codec_settings_.startBitrate;
+    codec_settings_.spatialLayers[sl_idx].active = true;
+    codec_settings_.spatialLayers[sl_idx].maxFramerate =
+        codec_settings_.maxFramerate;
+  }
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  FramerateFractions expected_fps_allocation[kMaxSpatialLayers];
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction / 4);
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction / 2);
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction);
+  expected_fps_allocation[1] = expected_fps_allocation[0];
+  expected_fps_allocation[2] = expected_fps_allocation[0];
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ElementsAreArray(expected_fps_allocation));
+}
+
+TEST_F(TestVp9Impl, EncoderInfoFpsAllocationFlexibleMode) {
+  const uint8_t kNumSpatialLayers = 3;
+
+  codec_settings_.maxFramerate = 30;
+  codec_settings_.VP9()->numberOfSpatialLayers = kNumSpatialLayers;
+  codec_settings_.VP9()->numberOfTemporalLayers = 1;
+  codec_settings_.VP9()->flexibleMode = true;
+
+  VideoEncoder::RateControlParameters rate_params;
+  for (uint8_t sl_idx = 0; sl_idx < kNumSpatialLayers; ++sl_idx) {
+    codec_settings_.spatialLayers[sl_idx].width = codec_settings_.width;
+    codec_settings_.spatialLayers[sl_idx].height = codec_settings_.height;
+    codec_settings_.spatialLayers[sl_idx].minBitrate =
+        codec_settings_.startBitrate;
+    codec_settings_.spatialLayers[sl_idx].maxBitrate =
+        codec_settings_.startBitrate;
+    codec_settings_.spatialLayers[sl_idx].targetBitrate =
+        codec_settings_.startBitrate;
+    codec_settings_.spatialLayers[sl_idx].active = true;
+    // Force different frame rates for different layers, to verify that total
+    // fraction is correct.
+    codec_settings_.spatialLayers[sl_idx].maxFramerate =
+        codec_settings_.maxFramerate / (kNumSpatialLayers - sl_idx);
+    rate_params.bitrate.SetBitrate(sl_idx, 0,
+                                   codec_settings_.startBitrate * 1000);
+  }
+  rate_params.bandwidth_allocation =
+      DataRate::BitsPerSec(rate_params.bitrate.get_sum_bps());
+  rate_params.framerate_fps = codec_settings_.maxFramerate;
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  // No temporal layers allowed when spatial layers have different fps targets.
+  FramerateFractions expected_fps_allocation[kMaxSpatialLayers];
+  expected_fps_allocation[0].push_back(EncoderInfo::kMaxFramerateFraction / 3);
+  expected_fps_allocation[1].push_back(EncoderInfo::kMaxFramerateFraction / 2);
+  expected_fps_allocation[2].push_back(EncoderInfo::kMaxFramerateFraction);
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(expected_fps_allocation));
+
+  // SetRates with current fps does not alter outcome.
+  encoder_->SetRates(rate_params);
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(expected_fps_allocation));
+
+  // Higher fps than the codec wants, should still not affect outcome.
+  rate_params.framerate_fps *= 2;
+  encoder_->SetRates(rate_params);
+  EXPECT_THAT(encoder_->GetEncoderInfo().fps_allocation,
+              ::testing::ElementsAreArray(expected_fps_allocation));
+}
+
+class Vp9ImplWithLayeringTest
+    : public ::testing::TestWithParam<std::tuple<int, int, bool>> {
+ protected:
+  Vp9ImplWithLayeringTest()
+      : num_spatial_layers_(std::get<0>(GetParam())),
+        num_temporal_layers_(std::get<1>(GetParam())),
+        override_field_trials_(std::get<2>(GetParam())
+                                   ? "WebRTC-Vp9ExternalRefCtrl/Enabled/"
+                                   : "") {}
+
+  const uint8_t num_spatial_layers_;
+  const uint8_t num_temporal_layers_;
+  const test::ScopedFieldTrials override_field_trials_;
+};
+
+TEST_P(Vp9ImplWithLayeringTest, FlexibleMode) {
+  // In flexible mode encoder wrapper obtains actual list of references from
+  // encoder and writes it into RTP payload descriptor. Check that reference
+  // list in payload descriptor matches the predefined one, which is used
+  // in non-flexible mode.
+  std::unique_ptr<VideoEncoder> encoder = VP9Encoder::Create();
+  VideoCodec codec_settings = DefaultCodecSettings();
+  codec_settings.VP9()->flexibleMode = true;
+  codec_settings.SetFrameDropEnabled(false);
+  codec_settings.VP9()->numberOfSpatialLayers = num_spatial_layers_;
+  codec_settings.VP9()->numberOfTemporalLayers = num_temporal_layers_;
+  EXPECT_EQ(encoder->InitEncode(&codec_settings, kSettings),
+            WEBRTC_VIDEO_CODEC_OK);
+
+  GofInfoVP9 gof;
+  if (num_temporal_layers_ == 1) {
+    gof.SetGofInfoVP9(kTemporalStructureMode1);
+  } else if (num_temporal_layers_ == 2) {
+    gof.SetGofInfoVP9(kTemporalStructureMode2);
+  } else if (num_temporal_layers_ == 3) {
+    gof.SetGofInfoVP9(kTemporalStructureMode3);
+  }
+
+  // Encode at least (num_frames_in_gof + 1) frames to verify references
+  // of non-key frame with gof_idx = 0.
+  int num_input_frames = gof.num_frames_in_gof + 1;
+  std::vector<EncodedVideoFrameProducer::EncodedFrame> frames =
+      EncodedVideoFrameProducer(*encoder)
+          .SetNumInputFrames(num_input_frames)
+          .SetResolution({kWidth, kHeight})
+          .Encode();
+  ASSERT_THAT(frames, SizeIs(num_input_frames * num_spatial_layers_));
+
+  for (size_t i = 0; i < frames.size(); ++i) {
+    const EncodedVideoFrameProducer::EncodedFrame& frame = frames[i];
+    const size_t picture_idx = i / num_spatial_layers_;
+    const size_t gof_idx = picture_idx % gof.num_frames_in_gof;
+
+    const CodecSpecificInfoVP9& vp9 =
+        frame.codec_specific_info.codecSpecific.VP9;
+    EXPECT_EQ(frame.encoded_image.SpatialIndex(),
+              num_spatial_layers_ == 1
+                  ? absl::nullopt
+                  : absl::optional<int>(i % num_spatial_layers_))
+        << "Frame " << i;
+    EXPECT_EQ(vp9.temporal_idx, num_temporal_layers_ == 1
+                                    ? kNoTemporalIdx
+                                    : gof.temporal_idx[gof_idx])
+        << "Frame " << i;
+    EXPECT_EQ(vp9.temporal_up_switch, gof.temporal_up_switch[gof_idx])
+        << "Frame " << i;
+    if (picture_idx == 0) {
+      EXPECT_EQ(vp9.num_ref_pics, 0) << "Frame " << i;
+    } else {
+      EXPECT_THAT(rtc::MakeArrayView(vp9.p_diff, vp9.num_ref_pics),
+                  UnorderedElementsAreArray(gof.pid_diff[gof_idx],
+                                            gof.num_ref_pics[gof_idx]))
+          << "Frame " << i;
+    }
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(All,
+                         Vp9ImplWithLayeringTest,
+                         ::testing::Combine(::testing::Values(1, 2, 3),
+                                            ::testing::Values(1, 2, 3),
+                                            ::testing::Bool()));
+
+class TestVp9ImplFrameDropping : public TestVp9Impl {
+ protected:
+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
+    webrtc::test::CodecSettings(kVideoCodecVP9, codec_settings);
+    // We need to encode quite a lot of frames in this test. Use low resolution
+    // to reduce execution time.
+    codec_settings->width = 64;
+    codec_settings->height = 64;
+    codec_settings->mode = VideoCodecMode::kScreensharing;
+  }
+};
+
+TEST_F(TestVp9ImplFrameDropping, PreEncodeFrameDropping) {
+  const size_t num_frames_to_encode = 100;
+  const float input_framerate_fps = 30.0;
+  const float video_duration_secs = num_frames_to_encode / input_framerate_fps;
+  const float expected_framerate_fps = 5.0f;
+  const float max_abs_framerate_error_fps = expected_framerate_fps * 0.1f;
+
+  codec_settings_.maxFramerate = static_cast<uint32_t>(expected_framerate_fps);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoFrame input_frame = NextInputFrame();
+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(input_frame, nullptr));
+    const size_t timestamp = input_frame.timestamp() +
+                             kVideoPayloadTypeFrequency / input_framerate_fps;
+    input_frame.set_timestamp(static_cast<uint32_t>(timestamp));
+  }
+
+  const size_t num_encoded_frames = GetNumEncodedFrames();
+  const float encoded_framerate_fps = num_encoded_frames / video_duration_secs;
+  EXPECT_NEAR(encoded_framerate_fps, expected_framerate_fps,
+              max_abs_framerate_error_fps);
+}
+
+TEST_F(TestVp9ImplFrameDropping, DifferentFrameratePerSpatialLayer) {
+  // Assign different frame rate to spatial layers and check that result frame
+  // rate is close to the assigned one.
+  const uint8_t num_spatial_layers = 3;
+  const float input_framerate_fps = 30.0;
+  const size_t video_duration_secs = 3;
+  const size_t num_input_frames = video_duration_secs * input_framerate_fps;
+
+  codec_settings_.VP9()->numberOfSpatialLayers = num_spatial_layers;
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.VP9()->flexibleMode = true;
+
+  VideoBitrateAllocation bitrate_allocation;
+  for (uint8_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    // Frame rate increases from low to high layer.
+    const uint32_t framerate_fps = 10 * (sl_idx + 1);
+
+    codec_settings_.spatialLayers[sl_idx].width = codec_settings_.width;
+    codec_settings_.spatialLayers[sl_idx].height = codec_settings_.height;
+    codec_settings_.spatialLayers[sl_idx].maxFramerate = framerate_fps;
+    codec_settings_.spatialLayers[sl_idx].minBitrate =
+        codec_settings_.startBitrate;
+    codec_settings_.spatialLayers[sl_idx].maxBitrate =
+        codec_settings_.startBitrate;
+    codec_settings_.spatialLayers[sl_idx].targetBitrate =
+        codec_settings_.startBitrate;
+    codec_settings_.spatialLayers[sl_idx].active = true;
+
+    bitrate_allocation.SetBitrate(
+        sl_idx, 0, codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000);
+  }
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  VideoFrame input_frame = NextInputFrame();
+  for (size_t frame_num = 0; frame_num < num_input_frames; ++frame_num) {
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(input_frame, nullptr));
+    const size_t timestamp = input_frame.timestamp() +
+                             kVideoPayloadTypeFrequency / input_framerate_fps;
+    input_frame.set_timestamp(static_cast<uint32_t>(timestamp));
+  }
+
+  std::vector<EncodedImage> encoded_frames;
+  std::vector<CodecSpecificInfo> codec_infos;
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_infos));
+
+  std::vector<size_t> num_encoded_frames(num_spatial_layers, 0);
+  for (EncodedImage& encoded_frame : encoded_frames) {
+    ++num_encoded_frames[encoded_frame.SpatialIndex().value_or(0)];
+  }
+
+  for (uint8_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    const float layer_target_framerate_fps =
+        codec_settings_.spatialLayers[sl_idx].maxFramerate;
+    const float layer_output_framerate_fps =
+        static_cast<float>(num_encoded_frames[sl_idx]) / video_duration_secs;
+    const float max_framerate_error_fps = layer_target_framerate_fps * 0.1f;
+    EXPECT_NEAR(layer_output_framerate_fps, layer_target_framerate_fps,
+                max_framerate_error_fps);
+  }
+}
+
+class TestVp9ImplProfile2 : public TestVp9Impl {
+ protected:
+  void SetUp() override {
+    // Profile 2 might not be available on some platforms until
+    // https://bugs.chromium.org/p/webm/issues/detail?id=1544 is solved.
+    bool profile_2_is_supported = false;
+    for (const auto& codec : SupportedVP9Codecs()) {
+      if (ParseSdpForVP9Profile(codec.parameters)
+              .value_or(VP9Profile::kProfile0) == VP9Profile::kProfile2) {
+        profile_2_is_supported = true;
+      }
+    }
+    if (!profile_2_is_supported)
+      return;
+
+    TestVp9Impl::SetUp();
+    input_frame_generator_ = test::CreateSquareFrameGenerator(
+        codec_settings_.width, codec_settings_.height,
+        test::FrameGeneratorInterface::OutputType::kI010,
+        absl::optional<int>());
+  }
+
+  std::unique_ptr<VideoEncoder> CreateEncoder() override {
+    cricket::VideoCodec profile2_codec;
+    profile2_codec.SetParam(kVP9FmtpProfileId,
+                            VP9ProfileToString(VP9Profile::kProfile2));
+    return VP9Encoder::Create(profile2_codec);
+  }
+
+  std::unique_ptr<VideoDecoder> CreateDecoder() override {
+    return VP9Decoder::Create();
+  }
+};
+
+TEST_F(TestVp9ImplProfile2, EncodeDecode) {
+  if (!encoder_)
+    return;
+
+  VideoFrame input_frame = NextInputFrame();
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(input_frame, nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+  // First frame should be a key frame.
+  encoded_frame._frameType = VideoFrameType::kVideoFrameKey;
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frame, false, 0));
+  std::unique_ptr<VideoFrame> decoded_frame;
+  absl::optional<uint8_t> decoded_qp;
+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
+  ASSERT_TRUE(decoded_frame);
+
+  // TODO(emircan): Add PSNR for different color depths.
+  EXPECT_GT(I420PSNR(*input_frame.video_frame_buffer()->ToI420(),
+                     *decoded_frame->video_frame_buffer()->ToI420()),
+            31);
+}
+
+TEST_F(TestVp9Impl, EncodeWithDynamicRate) {
+  // Configured dynamic rate field trial and re-create the encoder.
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VideoRateControl/vp9_dynamic_rate:true/");
+  SetUp();
+
+  // Set 300kbps target with 100% headroom.
+  VideoEncoder::RateControlParameters params;
+  params.bandwidth_allocation = DataRate::BitsPerSec(300000);
+  params.bitrate.SetBitrate(0, 0, params.bandwidth_allocation.bps());
+  params.framerate_fps = 30.0;
+
+  encoder_->SetRates(params);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  EncodedImage encoded_frame;
+  CodecSpecificInfo codec_specific_info;
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+
+  // Set no headroom and encode again.
+  params.bandwidth_allocation = DataRate::Zero();
+  encoder_->SetRates(params);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
+}
+
+TEST_F(TestVp9Impl, ReenablingUpperLayerAfterKFWithInterlayerPredIsEnabled) {
+  const size_t num_spatial_layers = 2;
+  const int num_frames_to_encode = 10;
+  codec_settings_.VP9()->flexibleMode = true;
+  codec_settings_.SetFrameDropEnabled(false);
+  codec_settings_.VP9()->numberOfSpatialLayers = num_spatial_layers;
+  codec_settings_.VP9()->numberOfTemporalLayers = 1;
+  codec_settings_.VP9()->interLayerPred = InterLayerPredMode::kOn;
+  // Force low frame-rate, so all layers are present for all frames.
+  codec_settings_.maxFramerate = 5;
+
+  ConfigureSvc(codec_settings_, num_spatial_layers);
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->InitEncode(&codec_settings_, kSettings));
+
+  VideoBitrateAllocation bitrate_allocation;
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    bitrate_allocation.SetBitrate(
+        sl_idx, 0, codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000);
+  }
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  std::vector<EncodedImage> encoded_frames;
+  std::vector<CodecSpecificInfo> codec_specific;
+
+  for (int i = 0; i < num_frames_to_encode; ++i) {
+    SetWaitForEncodedFramesThreshold(num_spatial_layers);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific));
+    EXPECT_EQ(encoded_frames.size(), num_spatial_layers);
+  }
+
+  // Disable the last layer.
+  bitrate_allocation.SetBitrate(num_spatial_layers - 1, 0, 0);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  for (int i = 0; i < num_frames_to_encode; ++i) {
+    SetWaitForEncodedFramesThreshold(num_spatial_layers - 1);
+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              encoder_->Encode(NextInputFrame(), nullptr));
+    ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific));
+    EXPECT_EQ(encoded_frames.size(), num_spatial_layers - 1);
+  }
+
+  std::vector<VideoFrameType> frame_types = {VideoFrameType::kVideoFrameKey};
+
+  // Force a key-frame with the last layer still disabled.
+  SetWaitForEncodedFramesThreshold(num_spatial_layers - 1);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
+            encoder_->Encode(NextInputFrame(), &frame_types));
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific));
+  EXPECT_EQ(encoded_frames.size(), num_spatial_layers - 1);
+  ASSERT_EQ(encoded_frames[0]._frameType, VideoFrameType::kVideoFrameKey);
+
+  // Re-enable the last layer.
+  bitrate_allocation.SetBitrate(
+      num_spatial_layers - 1, 0,
+      codec_settings_.spatialLayers[num_spatial_layers - 1].targetBitrate *
+          1000);
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      bitrate_allocation, codec_settings_.maxFramerate));
+
+  SetWaitForEncodedFramesThreshold(num_spatial_layers);
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Encode(NextInputFrame(), nullptr));
+  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific));
+  EXPECT_EQ(encoded_frames.size(), num_spatial_layers);
+  EXPECT_EQ(encoded_frames[0]._frameType, VideoFrameType::kVideoFrameDelta);
+}
+
+TEST_F(TestVp9Impl, HandlesEmptyDecoderConfigure) {
+  std::unique_ptr<VideoDecoder> decoder = CreateDecoder();
+  // Check that default settings are ok for decoder.
+  EXPECT_TRUE(decoder->Configure({}));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder->Release());
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    TestVp9ImplForPixelFormat,
+    TestVp9ImplForPixelFormat,
+    ::testing::Values(test::FrameGeneratorInterface::OutputType::kI420,
+                      test::FrameGeneratorInterface::OutputType::kNV12),
+    [](const auto& info) {
+      return test::FrameGeneratorInterface::OutputTypeToString(info.param);
+    });
+
+// Helper function to populate an vpx_image_t instance with dimensions and
+// potential image data.
+std::function<vpx_image_t*(vpx_image_t*,
+                           vpx_img_fmt_t,
+                           unsigned int,
+                           unsigned int,
+                           unsigned int,
+                           unsigned char* img_data)>
+GetWrapImageFunction(vpx_image_t* img) {
+  return [img](vpx_image_t* /*img*/, vpx_img_fmt_t fmt, unsigned int d_w,
+               unsigned int d_h, unsigned int /*stride_align*/,
+               unsigned char* img_data) {
+    img->fmt = fmt;
+    img->d_w = d_w;
+    img->d_h = d_h;
+    img->img_data = img_data;
+    return img;
+  };
+}
+
+TEST(Vp9SpeedSettingsTrialsTest, NoSvcUsesGlobalSpeedFromTl0InLayerConfig) {
+  // TL0 speed 8 at >= 480x270, 5 if below that.
+  test::ExplicitKeyValueConfig trials(
+      "WebRTC-VP9-PerformanceFlags/"
+      "use_per_layer_speed,"
+      "min_pixel_count:0|129600,"
+      "base_layer_speed:4|8,"
+      "high_layer_speed:5|9,"
+      "deblock_mode:1|0/");
+
+  // Keep a raw pointer for EXPECT calls and the like. Ownership is otherwise
+  // passed on to LibvpxVp9Encoder.
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp9Encoder encoder(cricket::VideoCodec(),
+                           absl::WrapUnique<LibvpxInterface>(vpx), trials);
+
+  VideoCodec settings = DefaultCodecSettings();
+  settings.width = 480;
+  settings.height = 270;
+  vpx_image_t img;
+
+  ON_CALL(*vpx, img_wrap).WillByDefault(GetWrapImageFunction(&img));
+  ON_CALL(*vpx, codec_enc_config_default)
+      .WillByDefault(DoAll(WithArg<1>([](vpx_codec_enc_cfg_t* cfg) {
+                             memset(cfg, 0, sizeof(vpx_codec_enc_cfg_t));
+                           }),
+                           Return(VPX_CODEC_OK)));
+  EXPECT_CALL(*vpx, codec_control(_, _, An<int>())).Times(AnyNumber());
+
+  EXPECT_CALL(*vpx, codec_control(_, VP9E_SET_SVC_PARAMETERS,
+                                  A<vpx_svc_extra_cfg_t*>()))
+      .Times(0);
+
+  EXPECT_CALL(*vpx, codec_control(_, VP8E_SET_CPUUSED, TypedEq<int>(8)));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder.InitEncode(&settings, kSettings));
+
+  encoder.Release();
+  settings.width = 352;
+  settings.height = 216;
+
+  EXPECT_CALL(*vpx, codec_control(_, VP8E_SET_CPUUSED, TypedEq<int>(4)));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder.InitEncode(&settings, kSettings));
+}
+
+TEST(Vp9SpeedSettingsTrialsTest,
+     NoPerLayerFlagUsesGlobalSpeedFromTopLayerInConfig) {
+  // TL0 speed 8 at >= 480x270, 5 if below that.
+  test::ExplicitKeyValueConfig trials(
+      "WebRTC-VP9-PerformanceFlags/"
+      "min_pixel_count:0|129600,"
+      "base_layer_speed:4|8,"
+      "high_layer_speed:5|9,"
+      "deblock_mode:1|0/");
+
+  // Keep a raw pointer for EXPECT calls and the like. Ownership is otherwise
+  // passed on to LibvpxVp9Encoder.
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp9Encoder encoder(cricket::VideoCodec(),
+                           absl::WrapUnique<LibvpxInterface>(vpx), trials);
+
+  VideoCodec settings = DefaultCodecSettings();
+  settings.width = 480;
+  settings.height = 270;
+  ConfigureSvc(settings, 2, 3);
+  vpx_image_t img;
+
+  ON_CALL(*vpx, img_wrap).WillByDefault(GetWrapImageFunction(&img));
+  ON_CALL(*vpx, codec_enc_config_default)
+      .WillByDefault(DoAll(WithArg<1>([](vpx_codec_enc_cfg_t* cfg) {
+                             memset(cfg, 0, sizeof(vpx_codec_enc_cfg_t));
+                           }),
+                           Return(VPX_CODEC_OK)));
+  EXPECT_CALL(*vpx, codec_control(_, _, An<int>())).Times(AnyNumber());
+
+  // Speed settings not populated when 'use_per_layer_speed' flag is absent.
+  EXPECT_CALL(*vpx,
+              codec_control(
+                  _, VP9E_SET_SVC_PARAMETERS,
+                  SafeMatcherCast<vpx_svc_extra_cfg_t*>(AllOf(
+                      Field(&vpx_svc_extra_cfg_t::speed_per_layer, Each(0)),
+                      Field(&vpx_svc_extra_cfg_t::loopfilter_ctrl, Each(0))))))
+      .Times(2);
+
+  EXPECT_CALL(*vpx, codec_control(_, VP8E_SET_CPUUSED, TypedEq<int>(8)));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder.InitEncode(&settings, kSettings));
+
+  encoder.Release();
+  settings.width = 476;
+  settings.height = 268;
+  settings.spatialLayers[0].width = settings.width / 2;
+  settings.spatialLayers[0].height = settings.height / 2;
+  settings.spatialLayers[1].width = settings.width;
+  settings.spatialLayers[1].height = settings.height;
+
+  EXPECT_CALL(*vpx, codec_control(_, VP8E_SET_CPUUSED, TypedEq<int>(4)));
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder.InitEncode(&settings, kSettings));
+}
+
+TEST(Vp9SpeedSettingsTrialsTest, DefaultPerLayerFlagsWithSvc) {
+  // Per-temporal and spatial layer speed settings:
+  // SL0:   TL0 = speed 5, TL1/TL2 = speed 8.
+  // SL1/2: TL0 = speed 7, TL1/TL2 = speed 8.
+  // Deblocking-mode per spatial layer:
+  // SL0: mode 1, SL1/2: mode 0.
+  test::ExplicitKeyValueConfig trials(
+      "WebRTC-VP9-PerformanceFlags/"
+      "use_per_layer_speed,"
+      "min_pixel_count:0|129600,"
+      "base_layer_speed:5|7,"
+      "high_layer_speed:8|8,"
+      "deblock_mode:1|0/");
+
+  // Keep a raw pointer for EXPECT calls and the like. Ownership is otherwise
+  // passed on to LibvpxVp9Encoder.
+  auto* const vpx = new NiceMock<MockLibvpxInterface>();
+  LibvpxVp9Encoder encoder(cricket::VideoCodec(),
+                           absl::WrapUnique<LibvpxInterface>(vpx), trials);
+
+  VideoCodec settings = DefaultCodecSettings();
+  constexpr int kNumSpatialLayers = 3;
+  constexpr int kNumTemporalLayers = 3;
+  ConfigureSvc(settings, kNumSpatialLayers, kNumTemporalLayers);
+  VideoBitrateAllocation bitrate_allocation;
+  for (int si = 0; si < kNumSpatialLayers; ++si) {
+    for (int ti = 0; ti < kNumTemporalLayers; ++ti) {
+      uint32_t bitrate_bps =
+          settings.spatialLayers[si].targetBitrate * 1'000 / kNumTemporalLayers;
+      bitrate_allocation.SetBitrate(si, ti, bitrate_bps);
+    }
+  }
+  vpx_image_t img;
+
+  // Speed settings per spatial layer, for TL0.
+  const int kBaseTlSpeed[VPX_MAX_LAYERS] = {5, 7, 7};
+  // Speed settings per spatial layer, for TL1, TL2.
+  const int kHighTlSpeed[VPX_MAX_LAYERS] = {8, 8, 8};
+  // Loopfilter settings are handled within libvpx, so this array is valid for
+  // both TL0 and higher.
+  const int kLoopFilter[VPX_MAX_LAYERS] = {1, 0, 0};
+
+  ON_CALL(*vpx, img_wrap).WillByDefault(GetWrapImageFunction(&img));
+  ON_CALL(*vpx, codec_enc_init)
+      .WillByDefault(WithArg<0>([](vpx_codec_ctx_t* ctx) {
+        memset(ctx, 0, sizeof(*ctx));
+        return VPX_CODEC_OK;
+      }));
+  ON_CALL(*vpx, codec_enc_config_default)
+      .WillByDefault(DoAll(WithArg<1>([](vpx_codec_enc_cfg_t* cfg) {
+                             memset(cfg, 0, sizeof(vpx_codec_enc_cfg_t));
+                           }),
+                           Return(VPX_CODEC_OK)));
+  EXPECT_CALL(
+      *vpx, codec_control(_, VP9E_SET_SVC_PARAMETERS,
+                          SafeMatcherCast<vpx_svc_extra_cfg_t*>(
+                              AllOf(Field(&vpx_svc_extra_cfg_t::speed_per_layer,
+                                          ElementsAreArray(kBaseTlSpeed)),
+                                    Field(&vpx_svc_extra_cfg_t::loopfilter_ctrl,
+                                          ElementsAreArray(kLoopFilter))))));
+
+  // Capture the callback into the vp9 wrapper.
+  vpx_codec_priv_output_cx_pkt_cb_pair_t callback_pointer = {};
+  EXPECT_CALL(*vpx, codec_control(_, VP9E_REGISTER_CX_CALLBACK, A<void*>()))
+      .WillOnce(WithArg<2>([&](void* cbp) {
+        callback_pointer =
+            *reinterpret_cast<vpx_codec_priv_output_cx_pkt_cb_pair_t*>(cbp);
+        return VPX_CODEC_OK;
+      }));
+
+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder.InitEncode(&settings, kSettings));
+
+  encoder.SetRates(VideoEncoder::RateControlParameters(bitrate_allocation,
+                                                       settings.maxFramerate));
+
+  MockEncodedImageCallback callback;
+  encoder.RegisterEncodeCompleteCallback(&callback);
+  auto frame_generator = test::CreateSquareFrameGenerator(
+      kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kI420, 10);
+  Mock::VerifyAndClearExpectations(vpx);
+
+  uint8_t data[1] = {0};
+  vpx_codec_cx_pkt encoded_data = {};
+  encoded_data.data.frame.buf = &data;
+  encoded_data.data.frame.sz = 1;
+
+  const auto kImageOk =
+      EncodedImageCallback::Result(EncodedImageCallback::Result::OK);
+
+  int spatial_id = 0;
+  int temporal_id = 0;
+  EXPECT_CALL(*vpx,
+              codec_control(_, VP9E_SET_SVC_LAYER_ID, A<vpx_svc_layer_id_t*>()))
+      .Times(AnyNumber());
+  EXPECT_CALL(*vpx,
+              codec_control(_, VP9E_GET_SVC_LAYER_ID, A<vpx_svc_layer_id_t*>()))
+      .WillRepeatedly(WithArg<2>([&](vpx_svc_layer_id_t* layer_id) {
+        layer_id->spatial_layer_id = spatial_id;
+        layer_id->temporal_layer_id = temporal_id;
+        return VPX_CODEC_OK;
+      }));
+  vpx_svc_ref_frame_config_t stored_refs = {};
+  ON_CALL(*vpx, codec_control(_, VP9E_SET_SVC_REF_FRAME_CONFIG,
+                              A<vpx_svc_ref_frame_config_t*>()))
+      .WillByDefault(
+          DoAll(SaveArgPointee<2>(&stored_refs), Return(VPX_CODEC_OK)));
+  ON_CALL(*vpx, codec_control(_, VP9E_GET_SVC_REF_FRAME_CONFIG,
+                              A<vpx_svc_ref_frame_config_t*>()))
+      .WillByDefault(
+          DoAll(SetArgPointee<2>(ByRef(stored_refs)), Return(VPX_CODEC_OK)));
+
+  // First frame is keyframe.
+  encoded_data.data.frame.flags = VPX_FRAME_IS_KEY;
+
+  // Default 3-layer temporal pattern: 0-2-1-2, then repeat and do two more.
+  for (int ti : {0, 2, 1, 2, 0, 2}) {
+    EXPECT_CALL(*vpx, codec_encode).WillOnce(Return(VPX_CODEC_OK));
+    // No update expected if flags haven't changed, and they change we we move
+    // between base temporal layer and non-base temporal layer.
+    if ((ti > 0) != (temporal_id > 0)) {
+      EXPECT_CALL(*vpx, codec_control(
+                            _, VP9E_SET_SVC_PARAMETERS,
+                            SafeMatcherCast<vpx_svc_extra_cfg_t*>(AllOf(
+                                Field(&vpx_svc_extra_cfg_t::speed_per_layer,
+                                      ElementsAreArray(ti == 0 ? kBaseTlSpeed
+                                                               : kHighTlSpeed)),
+                                Field(&vpx_svc_extra_cfg_t::loopfilter_ctrl,
+                                      ElementsAreArray(kLoopFilter))))));
+    } else {
+      EXPECT_CALL(*vpx, codec_control(_, VP9E_SET_SVC_PARAMETERS,
+                                      A<vpx_svc_extra_cfg_t*>()))
+          .Times(0);
+    }
+
+    VideoFrame frame =
+        VideoFrame::Builder()
+            .set_video_frame_buffer(frame_generator->NextFrame().buffer)
+            .build();
+    encoder.Encode(frame, nullptr);
+
+    temporal_id = ti;
+    for (int si = 0; si < kNumSpatialLayers; ++si) {
+      spatial_id = si;
+
+      EXPECT_CALL(callback, OnEncodedImage).WillOnce(Return(kImageOk));
+      callback_pointer.output_cx_pkt(&encoded_data, callback_pointer.user_priv);
+    }
+
+    encoded_data.data.frame.flags = 0;  // Following frames are delta frames.
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9.cc
new file mode 100644
index 0000000000..222e57b6ba
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+
+#include <memory>
+
+#include "absl/container/inlined_vector.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/vp9_profile.h"
+#include "modules/video_coding/codecs/vp9/libvpx_vp9_decoder.h"
+#include "modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h"
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "rtc_base/checks.h"
+#include "vpx/vp8cx.h"
+#include "vpx/vp8dx.h"
+#include "vpx/vpx_codec.h"
+
+namespace webrtc {
+
+std::vector<SdpVideoFormat> SupportedVP9Codecs(bool add_scalability_modes) {
+#ifdef RTC_ENABLE_VP9
+  // Profile 2 might not be available on some platforms until
+  // https://bugs.chromium.org/p/webm/issues/detail?id=1544 is solved.
+  static bool vpx_supports_high_bit_depth =
+      (vpx_codec_get_caps(vpx_codec_vp9_cx()) & VPX_CODEC_CAP_HIGHBITDEPTH) !=
+          0 &&
+      (vpx_codec_get_caps(vpx_codec_vp9_dx()) & VPX_CODEC_CAP_HIGHBITDEPTH) !=
+          0;
+
+  absl::InlinedVector<ScalabilityMode, kScalabilityModeCount> scalability_modes;
+  if (add_scalability_modes) {
+    for (const auto scalability_mode : kAllScalabilityModes) {
+      if (ScalabilityStructureConfig(scalability_mode).has_value()) {
+        scalability_modes.push_back(scalability_mode);
+      }
+    }
+  }
+  std::vector<SdpVideoFormat> supported_formats{SdpVideoFormat(
+      cricket::kVp9CodecName,
+      {{kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile0)}},
+      scalability_modes)};
+  if (vpx_supports_high_bit_depth) {
+    supported_formats.push_back(SdpVideoFormat(
+        cricket::kVp9CodecName,
+        {{kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile2)}},
+        scalability_modes));
+  }
+
+  return supported_formats;
+#else
+  return std::vector<SdpVideoFormat>();
+#endif
+}
+
+std::vector<SdpVideoFormat> SupportedVP9DecoderCodecs() {
+#ifdef RTC_ENABLE_VP9
+  std::vector<SdpVideoFormat> supported_formats = SupportedVP9Codecs();
+  // The WebRTC internal decoder supports VP9 profile 1 and 3. However, there's
+  // currently no way of sending VP9 profile 1 or 3 using the internal encoder.
+  // It would require extended support for I444, I422, and I440 buffers.
+  supported_formats.push_back(SdpVideoFormat(
+      cricket::kVp9CodecName,
+      {{kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile1)}}));
+  supported_formats.push_back(SdpVideoFormat(
+      cricket::kVp9CodecName,
+      {{kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile3)}}));
+  return supported_formats;
+#else
+  return std::vector<SdpVideoFormat>();
+#endif
+}
+
+std::unique_ptr<VP9Encoder> VP9Encoder::Create() {
+#ifdef RTC_ENABLE_VP9
+  return std::make_unique<LibvpxVp9Encoder>(cricket::VideoCodec(),
+                                            LibvpxInterface::Create(),
+                                            FieldTrialBasedConfig());
+#else
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+#endif
+}
+
+std::unique_ptr<VP9Encoder> VP9Encoder::Create(
+    const cricket::VideoCodec& codec) {
+#ifdef RTC_ENABLE_VP9
+  return std::make_unique<LibvpxVp9Encoder>(codec, LibvpxInterface::Create(),
+                                            FieldTrialBasedConfig());
+#else
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+#endif
+}
+
+bool VP9Encoder::SupportsScalabilityMode(ScalabilityMode scalability_mode) {
+  return ScalabilityStructureConfig(scalability_mode).has_value();
+}
+
+std::unique_ptr<VP9Decoder> VP9Decoder::Create() {
+#ifdef RTC_ENABLE_VP9
+  return std::make_unique<LibvpxVp9Decoder>();
+#else
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+#endif
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc b/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
new file mode 100644
index 0000000000..181550ce91
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifdef RTC_ENABLE_VP9
+
+#include "modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "vpx/vpx_codec.h"
+#include "vpx/vpx_decoder.h"
+#include "vpx/vpx_frame_buffer.h"
+
+namespace webrtc {
+
+uint8_t* Vp9FrameBufferPool::Vp9FrameBuffer::GetData() {
+  return data_.data<uint8_t>();
+}
+
+size_t Vp9FrameBufferPool::Vp9FrameBuffer::GetDataSize() const {
+  return data_.size();
+}
+
+void Vp9FrameBufferPool::Vp9FrameBuffer::SetSize(size_t size) {
+  data_.SetSize(size);
+}
+
+bool Vp9FrameBufferPool::InitializeVpxUsePool(
+    vpx_codec_ctx* vpx_codec_context) {
+  RTC_DCHECK(vpx_codec_context);
+  // Tell libvpx to use this pool.
+  if (vpx_codec_set_frame_buffer_functions(
+          // In which context to use these callback functions.
+          vpx_codec_context,
+          // Called by libvpx when it needs another frame buffer.
+          &Vp9FrameBufferPool::VpxGetFrameBuffer,
+          // Called by libvpx when it no longer uses a frame buffer.
+          &Vp9FrameBufferPool::VpxReleaseFrameBuffer,
+          // `this` will be passed as `user_priv` to VpxGetFrameBuffer.
+          this)) {
+    // Failed to configure libvpx to use Vp9FrameBufferPool.
+    return false;
+  }
+  return true;
+}
+
+rtc::scoped_refptr<Vp9FrameBufferPool::Vp9FrameBuffer>
+Vp9FrameBufferPool::GetFrameBuffer(size_t min_size) {
+  RTC_DCHECK_GT(min_size, 0);
+  rtc::scoped_refptr<Vp9FrameBuffer> available_buffer = nullptr;
+  {
+    MutexLock lock(&buffers_lock_);
+    // Do we have a buffer we can recycle?
+    for (const auto& buffer : allocated_buffers_) {
+      if (buffer->HasOneRef()) {
+        available_buffer = buffer;
+        break;
+      }
+    }
+    // Otherwise create one.
+    if (available_buffer == nullptr) {
+      available_buffer = new Vp9FrameBuffer();
+      allocated_buffers_.push_back(available_buffer);
+      if (allocated_buffers_.size() > max_num_buffers_) {
+        RTC_LOG(LS_WARNING)
+            << allocated_buffers_.size()
+            << " Vp9FrameBuffers have been "
+               "allocated by a Vp9FrameBufferPool (exceeding what is "
+               "considered reasonable, "
+            << max_num_buffers_ << ").";
+
+        // TODO(phoglund): this limit is being hit in tests since Oct 5 2016.
+        // See https://bugs.chromium.org/p/webrtc/issues/detail?id=6484.
+        // RTC_DCHECK_NOTREACHED();
+      }
+    }
+  }
+
+  available_buffer->SetSize(min_size);
+  return available_buffer;
+}
+
+int Vp9FrameBufferPool::GetNumBuffersInUse() const {
+  int num_buffers_in_use = 0;
+  MutexLock lock(&buffers_lock_);
+  for (const auto& buffer : allocated_buffers_) {
+    if (!buffer->HasOneRef())
+      ++num_buffers_in_use;
+  }
+  return num_buffers_in_use;
+}
+
+bool Vp9FrameBufferPool::Resize(size_t max_number_of_buffers) {
+  MutexLock lock(&buffers_lock_);
+  size_t used_buffers_count = 0;
+  for (const auto& buffer : allocated_buffers_) {
+    // If the buffer is in use, the ref count will be >= 2, one from the list we
+    // are looping over and one from the application. If the ref count is 1,
+    // then the list we are looping over holds the only reference and it's safe
+    // to reuse.
+    if (!buffer->HasOneRef()) {
+      used_buffers_count++;
+    }
+  }
+  if (used_buffers_count > max_number_of_buffers) {
+    return false;
+  }
+  max_num_buffers_ = max_number_of_buffers;
+
+  size_t buffers_to_purge = allocated_buffers_.size() - max_num_buffers_;
+  auto iter = allocated_buffers_.begin();
+  while (iter != allocated_buffers_.end() && buffers_to_purge > 0) {
+    if ((*iter)->HasOneRef()) {
+      iter = allocated_buffers_.erase(iter);
+      buffers_to_purge--;
+    } else {
+      ++iter;
+    }
+  }
+  return true;
+}
+
+void Vp9FrameBufferPool::ClearPool() {
+  MutexLock lock(&buffers_lock_);
+  allocated_buffers_.clear();
+}
+
+// static
+int32_t Vp9FrameBufferPool::VpxGetFrameBuffer(void* user_priv,
+                                              size_t min_size,
+                                              vpx_codec_frame_buffer* fb) {
+  RTC_DCHECK(user_priv);
+  RTC_DCHECK(fb);
+
+#ifdef FUZZING_BUILD_MODE_UNSAFE_FOR_PRODUCTION
+  // Limit size of 8k YUV highdef frame
+  size_t size_limit = 7680 * 4320 * 3 / 2 * 2;
+  if (min_size > size_limit)
+    return -1;
+#endif
+
+  Vp9FrameBufferPool* pool = static_cast<Vp9FrameBufferPool*>(user_priv);
+
+  rtc::scoped_refptr<Vp9FrameBuffer> buffer = pool->GetFrameBuffer(min_size);
+  fb->data = buffer->GetData();
+  fb->size = buffer->GetDataSize();
+  // Store Vp9FrameBuffer* in `priv` for use in VpxReleaseFrameBuffer.
+  // This also makes vpx_codec_get_frame return images with their `fb_priv` set
+  // to `buffer` which is important for external reference counting.
+  // Release from refptr so that the buffer's `ref_count_` remains 1 when
+  // `buffer` goes out of scope.
+  fb->priv = static_cast<void*>(buffer.release());
+  return 0;
+}
+
+// static
+int32_t Vp9FrameBufferPool::VpxReleaseFrameBuffer(void* user_priv,
+                                                  vpx_codec_frame_buffer* fb) {
+  RTC_DCHECK(user_priv);
+  RTC_DCHECK(fb);
+  Vp9FrameBuffer* buffer = static_cast<Vp9FrameBuffer*>(fb->priv);
+  if (buffer != nullptr) {
+    buffer->Release();
+    // When libvpx fails to decode and you continue to try to decode (and fail)
+    // libvpx can for some reason try to release the same buffer multiple times.
+    // Setting `priv` to null protects against trying to Release multiple times.
+    fb->priv = nullptr;
+  }
+  return 0;
+}
+
+}  // namespace webrtc
+
+#endif  // RTC_ENABLE_VP9
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h b/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h
new file mode 100644
index 0000000000..f46f1b7ea2
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h
@@ -0,0 +1,134 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef MODULES_VIDEO_CODING_CODECS_VP9_VP9_FRAME_BUFFER_POOL_H_
+#define MODULES_VIDEO_CODING_CODECS_VP9_VP9_FRAME_BUFFER_POOL_H_
+
+#ifdef RTC_ENABLE_VP9
+
+#include <vector>
+
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/synchronization/mutex.h"
+
+struct vpx_codec_ctx;
+struct vpx_codec_frame_buffer;
+
+namespace webrtc {
+
+// If more buffers than this are allocated we print warnings and crash if in
+// debug mode. VP9 is defined to have 8 reference buffers, of which 3 can be
+// referenced by any frame, see
+// https://tools.ietf.org/html/draft-grange-vp9-bitstream-00#section-2.2.2.
+// Assuming VP9 holds on to at most 8 buffers, any more buffers than that
+// would have to be by application code. Decoded frames should not be
+// referenced for longer than necessary. If we allow ~60 additional buffers
+// then the application has ~1 second to e.g. render each frame of a 60 fps
+// video.
+constexpr size_t kDefaultMaxNumBuffers = 68;
+
+// This memory pool is used to serve buffers to libvpx for decoding purposes in
+// VP9, which is set up in InitializeVPXUsePool. After the initialization any
+// time libvpx wants to decode a frame it will use buffers provided and released
+// through VpxGetFrameBuffer and VpxReleaseFrameBuffer.
+// The benefit of owning the pool that libvpx relies on for decoding is that the
+// decoded frames returned by libvpx (from vpx_codec_get_frame) use parts of our
+// buffers for the decoded image data. By retaining ownership of this buffer
+// using scoped_refptr, the image buffer can be reused by VideoFrames and no
+// frame copy has to occur during decoding and frame delivery.
+//
+// Pseudo example usage case:
+//    Vp9FrameBufferPool pool;
+//    pool.InitializeVpxUsePool(decoder_ctx);
+//    ...
+//
+//    // During decoding, libvpx will get and release buffers from the pool.
+//    vpx_codec_decode(decoder_ctx, ...);
+//
+//    vpx_image_t* img = vpx_codec_get_frame(decoder_ctx, &iter);
+//    // Important to use scoped_refptr to protect it against being recycled by
+//    // the pool.
+//    scoped_refptr<Vp9FrameBuffer> img_buffer = (Vp9FrameBuffer*)img->fb_priv;
+//    ...
+//
+//    // Destroying the codec will make libvpx release any buffers it was using.
+//    vpx_codec_destroy(decoder_ctx);
+class Vp9FrameBufferPool {
+ public:
+  class Vp9FrameBuffer final
+      : public rtc::RefCountedNonVirtual<Vp9FrameBuffer> {
+   public:
+    uint8_t* GetData();
+    size_t GetDataSize() const;
+    void SetSize(size_t size);
+
+    using rtc::RefCountedNonVirtual<Vp9FrameBuffer>::HasOneRef;
+
+   private:
+    // Data as an easily resizable buffer.
+    rtc::Buffer data_;
+  };
+
+  // Configures libvpx to, in the specified context, use this memory pool for
+  // buffers used to decompress frames. This is only supported for VP9.
+  bool InitializeVpxUsePool(vpx_codec_ctx* vpx_codec_context);
+
+  // Gets a frame buffer of at least `min_size`, recycling an available one or
+  // creating a new one. When no longer referenced from the outside the buffer
+  // becomes recyclable.
+  rtc::scoped_refptr<Vp9FrameBuffer> GetFrameBuffer(size_t min_size);
+  // Gets the number of buffers currently in use (not ready to be recycled).
+  int GetNumBuffersInUse() const;
+  // Changes the max amount of buffers in the pool to the new value.
+  // Returns true if change was successful and false if the amount of already
+  // allocated buffers is bigger than new value.
+  bool Resize(size_t max_number_of_buffers);
+  // Releases allocated buffers, deleting available buffers. Buffers in use are
+  // not deleted until they are no longer referenced.
+  void ClearPool();
+
+  // InitializeVpxUsePool configures libvpx to call this function when it needs
+  // a new frame buffer. Parameters:
+  // `user_priv` Private data passed to libvpx, InitializeVpxUsePool sets it up
+  //             to be a pointer to the pool.
+  // `min_size`  Minimum size needed by libvpx (to decompress a frame).
+  // `fb`        Pointer to the libvpx frame buffer object, this is updated to
+  //             use the pool's buffer.
+  // Returns 0 on success. Returns < 0 on failure.
+  static int32_t VpxGetFrameBuffer(void* user_priv,
+                                   size_t min_size,
+                                   vpx_codec_frame_buffer* fb);
+
+  // InitializeVpxUsePool configures libvpx to call this function when it has
+  // finished using one of the pool's frame buffer. Parameters:
+  // `user_priv` Private data passed to libvpx, InitializeVpxUsePool sets it up
+  //             to be a pointer to the pool.
+  // `fb`        Pointer to the libvpx frame buffer object, its `priv` will be
+  //             a pointer to one of the pool's Vp9FrameBuffer.
+  static int32_t VpxReleaseFrameBuffer(void* user_priv,
+                                       vpx_codec_frame_buffer* fb);
+
+ private:
+  // Protects `allocated_buffers_`.
+  mutable Mutex buffers_lock_;
+  // All buffers, in use or ready to be recycled.
+  std::vector<rtc::scoped_refptr<Vp9FrameBuffer>> allocated_buffers_
+      RTC_GUARDED_BY(buffers_lock_);
+  size_t max_num_buffers_ = kDefaultMaxNumBuffers;
+};
+
+}  // namespace webrtc
+
+#endif  // RTC_ENABLE_VP9
+
+#endif  // MODULES_VIDEO_CODING_CODECS_VP9_VP9_FRAME_BUFFER_POOL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/decoder_database.cc b/third_party/libwebrtc/modules/video_coding/decoder_database.cc
new file mode 100644
index 0000000000..3410edc624
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/decoder_database.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/decoder_database.h"
+
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+VCMDecoderDatabase::VCMDecoderDatabase() {
+  decoder_sequence_checker_.Detach();
+}
+
+void VCMDecoderDatabase::DeregisterExternalDecoder(uint8_t payload_type) {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  auto it = decoders_.find(payload_type);
+  if (it == decoders_.end()) {
+    return;
+  }
+
+  // We can't use payload_type to check if the decoder is currently in use,
+  // because payload type may be out of date (e.g. before we decode the first
+  // frame after RegisterReceiveCodec).
+  if (current_decoder_ && current_decoder_->IsSameDecoder(it->second.get())) {
+    // Release it if it was registered and in use.
+    current_decoder_ = absl::nullopt;
+  }
+  decoders_.erase(it);
+}
+
+// Add the external decoder object to the list of external decoders.
+// Won't be registered as a receive codec until RegisterReceiveCodec is called.
+void VCMDecoderDatabase::RegisterExternalDecoder(
+    uint8_t payload_type,
+    std::unique_ptr<VideoDecoder> external_decoder) {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  // If payload value already exists, erase old and insert new.
+  DeregisterExternalDecoder(payload_type);
+  if (external_decoder) {
+    decoders_.emplace(
+        std::make_pair(payload_type, std::move(external_decoder)));
+  }
+}
+
+bool VCMDecoderDatabase::IsExternalDecoderRegistered(
+    uint8_t payload_type) const {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  return decoders_.find(payload_type) != decoders_.end();
+}
+
+void VCMDecoderDatabase::RegisterReceiveCodec(
+    uint8_t payload_type,
+    const VideoDecoder::Settings& settings) {
+  // If payload value already exists, erase old and insert new.
+  if (payload_type == current_payload_type_) {
+    current_payload_type_ = absl::nullopt;
+  }
+  decoder_settings_[payload_type] = settings;
+}
+
+bool VCMDecoderDatabase::DeregisterReceiveCodec(uint8_t payload_type) {
+  if (decoder_settings_.erase(payload_type) == 0) {
+    return false;
+  }
+  if (payload_type == current_payload_type_) {
+    // This codec is currently in use.
+    current_payload_type_ = absl::nullopt;
+  }
+  return true;
+}
+
+void VCMDecoderDatabase::DeregisterReceiveCodecs() {
+  current_payload_type_ = absl::nullopt;
+  decoder_settings_.clear();
+}
+
+VCMGenericDecoder* VCMDecoderDatabase::GetDecoder(
+    const VCMEncodedFrame& frame,
+    VCMDecodedFrameCallback* decoded_frame_callback) {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  RTC_DCHECK(decoded_frame_callback->UserReceiveCallback());
+  uint8_t payload_type = frame.PayloadType();
+  if (payload_type == current_payload_type_ || payload_type == 0) {
+    return current_decoder_.has_value() ? &*current_decoder_ : nullptr;
+  }
+  // If decoder exists - delete.
+  if (current_decoder_.has_value()) {
+    current_decoder_ = absl::nullopt;
+    current_payload_type_ = absl::nullopt;
+  }
+
+  CreateAndInitDecoder(frame);
+  if (current_decoder_ == absl::nullopt) {
+    return nullptr;
+  }
+
+  VCMReceiveCallback* callback = decoded_frame_callback->UserReceiveCallback();
+  callback->OnIncomingPayloadType(payload_type);
+  if (current_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback) <
+      0) {
+    current_decoder_ = absl::nullopt;
+    return nullptr;
+  }
+
+  current_payload_type_ = payload_type;
+  return &*current_decoder_;
+}
+
+void VCMDecoderDatabase::CreateAndInitDecoder(const VCMEncodedFrame& frame) {
+  uint8_t payload_type = frame.PayloadType();
+  RTC_DLOG(LS_INFO) << "Initializing decoder with payload type '"
+                    << int{payload_type} << "'.";
+  auto decoder_item = decoder_settings_.find(payload_type);
+  if (decoder_item == decoder_settings_.end()) {
+    RTC_LOG(LS_ERROR) << "Can't find a decoder associated with payload type: "
+                      << int{payload_type};
+    return;
+  }
+  auto external_dec_item = decoders_.find(payload_type);
+  if (external_dec_item == decoders_.end()) {
+    RTC_LOG(LS_ERROR) << "No decoder of this type exists.";
+    return;
+  }
+  current_decoder_.emplace(external_dec_item->second.get());
+
+  // Copy over input resolutions to prevent codec reinitialization due to
+  // the first frame being of a different resolution than the database values.
+  // This is best effort, since there's no guarantee that width/height have been
+  // parsed yet (and may be zero).
+  RenderResolution frame_resolution(frame.EncodedImage()._encodedWidth,
+                                    frame.EncodedImage()._encodedHeight);
+  if (frame_resolution.Valid()) {
+    decoder_item->second.set_max_render_resolution(frame_resolution);
+  }
+  if (!current_decoder_->Configure(decoder_item->second)) {
+    current_decoder_ = absl::nullopt;
+    RTC_LOG(LS_ERROR) << "Failed to initialize decoder.";
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/decoder_database.h b/third_party/libwebrtc/modules/video_coding/decoder_database.h
new file mode 100644
index 0000000000..98f4335621
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/decoder_database.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_DECODER_DATABASE_H_
+#define MODULES_VIDEO_CODING_DECODER_DATABASE_H_
+
+#include <stdint.h>
+
+#include <map>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/encoded_frame.h"
+#include "modules/video_coding/generic_decoder.h"
+
+namespace webrtc {
+
+class VCMDecoderDatabase {
+ public:
+  VCMDecoderDatabase();
+  VCMDecoderDatabase(const VCMDecoderDatabase&) = delete;
+  VCMDecoderDatabase& operator=(const VCMDecoderDatabase&) = delete;
+  ~VCMDecoderDatabase() = default;
+
+  // Returns a pointer to the previously registered decoder or nullptr if none
+  // was registered for the `payload_type`.
+  void DeregisterExternalDecoder(uint8_t payload_type);
+  void RegisterExternalDecoder(uint8_t payload_type,
+                               std::unique_ptr<VideoDecoder> external_decoder);
+  bool IsExternalDecoderRegistered(uint8_t payload_type) const;
+
+  void RegisterReceiveCodec(uint8_t payload_type,
+                            const VideoDecoder::Settings& settings);
+  bool DeregisterReceiveCodec(uint8_t payload_type);
+  void DeregisterReceiveCodecs();
+
+  // Returns a decoder specified by frame.PayloadType. The decoded frame
+  // callback of the decoder is set to `decoded_frame_callback`. If no such
+  // decoder already exists an instance will be created and initialized.
+  // nullptr is returned if no decoder with the specified payload type was found
+  // and the function failed to create one.
+  VCMGenericDecoder* GetDecoder(
+      const VCMEncodedFrame& frame,
+      VCMDecodedFrameCallback* decoded_frame_callback);
+
+ private:
+  void CreateAndInitDecoder(const VCMEncodedFrame& frame)
+      RTC_RUN_ON(decoder_sequence_checker_);
+
+  SequenceChecker decoder_sequence_checker_;
+
+  absl::optional<uint8_t> current_payload_type_;
+  absl::optional<VCMGenericDecoder> current_decoder_
+      RTC_GUARDED_BY(decoder_sequence_checker_);
+  // Initialization paramaters for decoders keyed by payload type.
+  std::map<uint8_t, VideoDecoder::Settings> decoder_settings_;
+  // Decoders keyed by payload type.
+  std::map<uint8_t, std::unique_ptr<VideoDecoder>> decoders_
+      RTC_GUARDED_BY(decoder_sequence_checker_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_DECODER_DATABASE_H_
diff --git a/third_party/libwebrtc/modules/video_coding/decoder_database_unittest.cc b/third_party/libwebrtc/modules/video_coding/decoder_database_unittest.cc
new file mode 100644
index 0000000000..2e9c91b1c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/decoder_database_unittest.cc
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/decoder_database.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/test/mock_video_decoder.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::NiceMock;
+
+// Test registering and unregistering an external decoder instance.
+TEST(VCMDecoderDatabaseTest, RegisterExternalDecoder) {
+  VCMDecoderDatabase db;
+  constexpr int kPayloadType = 1;
+  ASSERT_FALSE(db.IsExternalDecoderRegistered(kPayloadType));
+
+  auto decoder = std::make_unique<NiceMock<MockVideoDecoder>>();
+  bool decoder_deleted = false;
+  EXPECT_CALL(*decoder, Destruct).WillOnce([&decoder_deleted] {
+    decoder_deleted = true;
+  });
+
+  db.RegisterExternalDecoder(kPayloadType, std::move(decoder));
+  EXPECT_TRUE(db.IsExternalDecoderRegistered(kPayloadType));
+  db.DeregisterExternalDecoder(kPayloadType);
+  EXPECT_TRUE(decoder_deleted);
+  EXPECT_FALSE(db.IsExternalDecoderRegistered(kPayloadType));
+}
+
+TEST(VCMDecoderDatabaseTest, RegisterReceiveCodec) {
+  VCMDecoderDatabase db;
+  constexpr int kPayloadType = 1;
+  ASSERT_FALSE(db.DeregisterReceiveCodec(kPayloadType));
+
+  VideoDecoder::Settings settings;
+  settings.set_codec_type(kVideoCodecVP8);
+  settings.set_max_render_resolution({10, 10});
+  settings.set_number_of_cores(4);
+  db.RegisterReceiveCodec(kPayloadType, settings);
+
+  EXPECT_TRUE(db.DeregisterReceiveCodec(kPayloadType));
+}
+
+TEST(VCMDecoderDatabaseTest, DeregisterReceiveCodecs) {
+  VCMDecoderDatabase db;
+  constexpr int kPayloadType1 = 1;
+  constexpr int kPayloadType2 = 2;
+  ASSERT_FALSE(db.DeregisterReceiveCodec(kPayloadType1));
+  ASSERT_FALSE(db.DeregisterReceiveCodec(kPayloadType2));
+
+  VideoDecoder::Settings settings1;
+  settings1.set_codec_type(kVideoCodecVP8);
+  settings1.set_max_render_resolution({10, 10});
+  settings1.set_number_of_cores(4);
+
+  VideoDecoder::Settings settings2 = settings1;
+  settings2.set_codec_type(kVideoCodecVP9);
+
+  db.RegisterReceiveCodec(kPayloadType1, settings1);
+  db.RegisterReceiveCodec(kPayloadType2, settings2);
+
+  db.DeregisterReceiveCodecs();
+
+  // All receive codecs must have been removed.
+  EXPECT_FALSE(db.DeregisterReceiveCodec(kPayloadType1));
+  EXPECT_FALSE(db.DeregisterReceiveCodec(kPayloadType2));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/decoding_state.cc b/third_party/libwebrtc/modules/video_coding/decoding_state.cc
new file mode 100644
index 0000000000..5e405cbd05
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/decoding_state.cc
@@ -0,0 +1,368 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/decoding_state.h"
+
+#include "common_video/h264/h264_common.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/video_coding/frame_buffer.h"
+#include "modules/video_coding/jitter_buffer_common.h"
+#include "modules/video_coding/packet.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+VCMDecodingState::VCMDecodingState()
+    : sequence_num_(0),
+      time_stamp_(0),
+      picture_id_(kNoPictureId),
+      temporal_id_(kNoTemporalIdx),
+      tl0_pic_id_(kNoTl0PicIdx),
+      full_sync_(true),
+      in_initial_state_(true) {
+  memset(frame_decoded_, 0, sizeof(frame_decoded_));
+}
+
+VCMDecodingState::~VCMDecodingState() {}
+
+void VCMDecodingState::Reset() {
+  // TODO(mikhal): Verify - not always would want to reset the sync
+  sequence_num_ = 0;
+  time_stamp_ = 0;
+  picture_id_ = kNoPictureId;
+  temporal_id_ = kNoTemporalIdx;
+  tl0_pic_id_ = kNoTl0PicIdx;
+  full_sync_ = true;
+  in_initial_state_ = true;
+  memset(frame_decoded_, 0, sizeof(frame_decoded_));
+  received_sps_.clear();
+  received_pps_.clear();
+}
+
+uint32_t VCMDecodingState::time_stamp() const {
+  return time_stamp_;
+}
+
+uint16_t VCMDecodingState::sequence_num() const {
+  return sequence_num_;
+}
+
+bool VCMDecodingState::IsOldFrame(const VCMFrameBuffer* frame) const {
+  RTC_DCHECK(frame);
+  if (in_initial_state_)
+    return false;
+  return !IsNewerTimestamp(frame->Timestamp(), time_stamp_);
+}
+
+bool VCMDecodingState::IsOldPacket(const VCMPacket* packet) const {
+  RTC_DCHECK(packet);
+  if (in_initial_state_)
+    return false;
+  return !IsNewerTimestamp(packet->timestamp, time_stamp_);
+}
+
+void VCMDecodingState::SetState(const VCMFrameBuffer* frame) {
+  RTC_DCHECK(frame);
+  RTC_CHECK_GE(frame->GetHighSeqNum(), 0);
+  if (!UsingFlexibleMode(frame))
+    UpdateSyncState(frame);
+  sequence_num_ = static_cast<uint16_t>(frame->GetHighSeqNum());
+  time_stamp_ = frame->Timestamp();
+  picture_id_ = frame->PictureId();
+  temporal_id_ = frame->TemporalId();
+  tl0_pic_id_ = frame->Tl0PicId();
+
+  for (const NaluInfo& nalu : frame->GetNaluInfos()) {
+    if (nalu.type == H264::NaluType::kPps) {
+      if (nalu.pps_id < 0) {
+        RTC_LOG(LS_WARNING) << "Received pps without pps id.";
+      } else if (nalu.sps_id < 0) {
+        RTC_LOG(LS_WARNING) << "Received pps without sps id.";
+      } else {
+        received_pps_[nalu.pps_id] = nalu.sps_id;
+      }
+    } else if (nalu.type == H264::NaluType::kSps) {
+      if (nalu.sps_id < 0) {
+        RTC_LOG(LS_WARNING) << "Received sps without sps id.";
+      } else {
+        received_sps_.insert(nalu.sps_id);
+      }
+    }
+  }
+
+  if (UsingFlexibleMode(frame)) {
+    uint16_t frame_index = picture_id_ % kFrameDecodedLength;
+    if (in_initial_state_) {
+      frame_decoded_cleared_to_ = frame_index;
+    } else if (frame->FrameType() == VideoFrameType::kVideoFrameKey) {
+      memset(frame_decoded_, 0, sizeof(frame_decoded_));
+      frame_decoded_cleared_to_ = frame_index;
+    } else {
+      if (AheadOfFramesDecodedClearedTo(frame_index)) {
+        while (frame_decoded_cleared_to_ != frame_index) {
+          frame_decoded_cleared_to_ =
+              (frame_decoded_cleared_to_ + 1) % kFrameDecodedLength;
+          frame_decoded_[frame_decoded_cleared_to_] = false;
+        }
+      }
+    }
+    frame_decoded_[frame_index] = true;
+  }
+
+  in_initial_state_ = false;
+}
+
+void VCMDecodingState::CopyFrom(const VCMDecodingState& state) {
+  sequence_num_ = state.sequence_num_;
+  time_stamp_ = state.time_stamp_;
+  picture_id_ = state.picture_id_;
+  temporal_id_ = state.temporal_id_;
+  tl0_pic_id_ = state.tl0_pic_id_;
+  full_sync_ = state.full_sync_;
+  in_initial_state_ = state.in_initial_state_;
+  frame_decoded_cleared_to_ = state.frame_decoded_cleared_to_;
+  memcpy(frame_decoded_, state.frame_decoded_, sizeof(frame_decoded_));
+  received_sps_ = state.received_sps_;
+  received_pps_ = state.received_pps_;
+}
+
+bool VCMDecodingState::UpdateEmptyFrame(const VCMFrameBuffer* frame) {
+  bool empty_packet = frame->GetHighSeqNum() == frame->GetLowSeqNum();
+  if (in_initial_state_ && empty_packet) {
+    // Drop empty packets as long as we are in the initial state.
+    return true;
+  }
+  if ((empty_packet && ContinuousSeqNum(frame->GetHighSeqNum())) ||
+      ContinuousFrame(frame)) {
+    // Continuous empty packets or continuous frames can be dropped if we
+    // advance the sequence number.
+    sequence_num_ = frame->GetHighSeqNum();
+    time_stamp_ = frame->Timestamp();
+    return true;
+  }
+  return false;
+}
+
+void VCMDecodingState::UpdateOldPacket(const VCMPacket* packet) {
+  RTC_DCHECK(packet);
+  if (packet->timestamp == time_stamp_) {
+    // Late packet belonging to the last decoded frame - make sure we update the
+    // last decoded sequence number.
+    sequence_num_ = LatestSequenceNumber(packet->seqNum, sequence_num_);
+  }
+}
+
+void VCMDecodingState::SetSeqNum(uint16_t new_seq_num) {
+  sequence_num_ = new_seq_num;
+}
+
+bool VCMDecodingState::in_initial_state() const {
+  return in_initial_state_;
+}
+
+bool VCMDecodingState::full_sync() const {
+  return full_sync_;
+}
+
+void VCMDecodingState::UpdateSyncState(const VCMFrameBuffer* frame) {
+  if (in_initial_state_)
+    return;
+  if (frame->TemporalId() == kNoTemporalIdx ||
+      frame->Tl0PicId() == kNoTl0PicIdx) {
+    full_sync_ = true;
+  } else if (frame->FrameType() == VideoFrameType::kVideoFrameKey ||
+             frame->LayerSync()) {
+    full_sync_ = true;
+  } else if (full_sync_) {
+    // Verify that we are still in sync.
+    // Sync will be broken if continuity is true for layers but not for the
+    // other methods (PictureId and SeqNum).
+    if (UsingPictureId(frame)) {
+      // First check for a valid tl0PicId.
+      if (frame->Tl0PicId() - tl0_pic_id_ > 1) {
+        full_sync_ = false;
+      } else {
+        full_sync_ = ContinuousPictureId(frame->PictureId());
+      }
+    } else {
+      full_sync_ =
+          ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()));
+    }
+  }
+}
+
+bool VCMDecodingState::ContinuousFrame(const VCMFrameBuffer* frame) const {
+  // Check continuity based on the following hierarchy:
+  // - Temporal layers (stop here if out of sync).
+  // - Picture Id when available.
+  // - Sequence numbers.
+  // Return true when in initial state.
+  // Note that when a method is not applicable it will return false.
+  RTC_DCHECK(frame);
+  // A key frame is always considered continuous as it doesn't refer to any
+  // frames and therefore won't introduce any errors even if prior frames are
+  // missing.
+  if (frame->FrameType() == VideoFrameType::kVideoFrameKey &&
+      HaveSpsAndPps(frame->GetNaluInfos())) {
+    return true;
+  }
+  // When in the initial state we always require a key frame to start decoding.
+  if (in_initial_state_)
+    return false;
+  if (ContinuousLayer(frame->TemporalId(), frame->Tl0PicId()))
+    return true;
+  // tl0picId is either not used, or should remain unchanged.
+  if (frame->Tl0PicId() != tl0_pic_id_)
+    return false;
+  // Base layers are not continuous or temporal layers are inactive.
+  // In the presence of temporal layers, check for Picture ID/sequence number
+  // continuity if sync can be restored by this frame.
+  if (!full_sync_ && !frame->LayerSync())
+    return false;
+  if (UsingPictureId(frame)) {
+    if (UsingFlexibleMode(frame)) {
+      return ContinuousFrameRefs(frame);
+    } else {
+      return ContinuousPictureId(frame->PictureId());
+    }
+  } else {
+    return ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum())) &&
+           HaveSpsAndPps(frame->GetNaluInfos());
+  }
+}
+
+bool VCMDecodingState::ContinuousPictureId(int picture_id) const {
+  int next_picture_id = picture_id_ + 1;
+  if (picture_id < picture_id_) {
+    // Wrap
+    if (picture_id_ >= 0x80) {
+      // 15 bits used for picture id
+      return ((next_picture_id & 0x7FFF) == picture_id);
+    } else {
+      // 7 bits used for picture id
+      return ((next_picture_id & 0x7F) == picture_id);
+    }
+  }
+  // No wrap
+  return (next_picture_id == picture_id);
+}
+
+bool VCMDecodingState::ContinuousSeqNum(uint16_t seq_num) const {
+  return seq_num == static_cast<uint16_t>(sequence_num_ + 1);
+}
+
+bool VCMDecodingState::ContinuousLayer(int temporal_id, int tl0_pic_id) const {
+  // First, check if applicable.
+  if (temporal_id == kNoTemporalIdx || tl0_pic_id == kNoTl0PicIdx)
+    return false;
+  // If this is the first frame to use temporal layers, make sure we start
+  // from base.
+  else if (tl0_pic_id_ == kNoTl0PicIdx && temporal_id_ == kNoTemporalIdx &&
+           temporal_id == 0)
+    return true;
+
+  // Current implementation: Look for base layer continuity.
+  if (temporal_id != 0)
+    return false;
+  return (static_cast<uint8_t>(tl0_pic_id_ + 1) == tl0_pic_id);
+}
+
+bool VCMDecodingState::ContinuousFrameRefs(const VCMFrameBuffer* frame) const {
+  uint8_t num_refs = frame->CodecSpecific()->codecSpecific.VP9.num_ref_pics;
+  for (uint8_t r = 0; r < num_refs; ++r) {
+    uint16_t frame_ref = frame->PictureId() -
+                         frame->CodecSpecific()->codecSpecific.VP9.p_diff[r];
+    uint16_t frame_index = frame_ref % kFrameDecodedLength;
+    if (AheadOfFramesDecodedClearedTo(frame_index) ||
+        !frame_decoded_[frame_index]) {
+      return false;
+    }
+  }
+  return true;
+}
+
+bool VCMDecodingState::UsingPictureId(const VCMFrameBuffer* frame) const {
+  return (frame->PictureId() != kNoPictureId && picture_id_ != kNoPictureId);
+}
+
+bool VCMDecodingState::UsingFlexibleMode(const VCMFrameBuffer* frame) const {
+  bool is_flexible_mode =
+      frame->CodecSpecific()->codecType == kVideoCodecVP9 &&
+      frame->CodecSpecific()->codecSpecific.VP9.flexible_mode;
+  if (is_flexible_mode && frame->PictureId() == kNoPictureId) {
+    RTC_LOG(LS_WARNING) << "Frame is marked as using flexible mode but no"
+                           "picture id is set.";
+    return false;
+  }
+  return is_flexible_mode;
+}
+
+// TODO(philipel): change how check work, this check practially
+// limits the max p_diff to 64.
+bool VCMDecodingState::AheadOfFramesDecodedClearedTo(uint16_t index) const {
+  // No way of knowing for sure if we are actually ahead of
+  // frame_decoded_cleared_to_. We just make the assumption
+  // that we are not trying to reference back to a very old
+  // index, but instead are referencing a newer index.
+  uint16_t diff =
+      index > frame_decoded_cleared_to_
+          ? kFrameDecodedLength - (index - frame_decoded_cleared_to_)
+          : frame_decoded_cleared_to_ - index;
+  return diff > kFrameDecodedLength / 2;
+}
+
+bool VCMDecodingState::HaveSpsAndPps(const std::vector<NaluInfo>& nalus) const {
+  std::set<int> new_sps;
+  std::map<int, int> new_pps;
+  for (const NaluInfo& nalu : nalus) {
+    // Check if this nalu actually contains sps/pps information or dependencies.
+    if (nalu.sps_id == -1 && nalu.pps_id == -1)
+      continue;
+    switch (nalu.type) {
+      case H264::NaluType::kPps:
+        if (nalu.pps_id < 0) {
+          RTC_LOG(LS_WARNING) << "Received pps without pps id.";
+        } else if (nalu.sps_id < 0) {
+          RTC_LOG(LS_WARNING) << "Received pps without sps id.";
+        } else {
+          new_pps[nalu.pps_id] = nalu.sps_id;
+        }
+        break;
+      case H264::NaluType::kSps:
+        if (nalu.sps_id < 0) {
+          RTC_LOG(LS_WARNING) << "Received sps without sps id.";
+        } else {
+          new_sps.insert(nalu.sps_id);
+        }
+        break;
+      default: {
+        int needed_sps = -1;
+        auto pps_it = new_pps.find(nalu.pps_id);
+        if (pps_it != new_pps.end()) {
+          needed_sps = pps_it->second;
+        } else {
+          auto pps_it2 = received_pps_.find(nalu.pps_id);
+          if (pps_it2 == received_pps_.end()) {
+            return false;
+          }
+          needed_sps = pps_it2->second;
+        }
+        if (new_sps.find(needed_sps) == new_sps.end() &&
+            received_sps_.find(needed_sps) == received_sps_.end()) {
+          return false;
+        }
+        break;
+      }
+    }
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/decoding_state.h b/third_party/libwebrtc/modules/video_coding/decoding_state.h
new file mode 100644
index 0000000000..ec972949d8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/decoding_state.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_DECODING_STATE_H_
+#define MODULES_VIDEO_CODING_DECODING_STATE_H_
+
+#include <cstdint>
+#include <map>
+#include <set>
+#include <vector>
+
+namespace webrtc {
+
+// Forward declarations
+struct NaluInfo;
+class VCMFrameBuffer;
+class VCMPacket;
+
+class VCMDecodingState {
+ public:
+  // The max number of bits used to reference back
+  // to a previous frame when using flexible mode.
+  static const uint16_t kNumRefBits = 7;
+  static const uint16_t kFrameDecodedLength = 1 << kNumRefBits;
+
+  VCMDecodingState();
+  ~VCMDecodingState();
+  // Check for old frame
+  bool IsOldFrame(const VCMFrameBuffer* frame) const;
+  // Check for old packet
+  bool IsOldPacket(const VCMPacket* packet) const;
+  // Check for frame continuity based on current decoded state. Use best method
+  // possible, i.e. temporal info, picture ID or sequence number.
+  bool ContinuousFrame(const VCMFrameBuffer* frame) const;
+  void SetState(const VCMFrameBuffer* frame);
+  void CopyFrom(const VCMDecodingState& state);
+  bool UpdateEmptyFrame(const VCMFrameBuffer* frame);
+  // Update the sequence number if the timestamp matches current state and the
+  // sequence number is higher than the current one. This accounts for packets
+  // arriving late.
+  void UpdateOldPacket(const VCMPacket* packet);
+  void SetSeqNum(uint16_t new_seq_num);
+  void Reset();
+  uint32_t time_stamp() const;
+  uint16_t sequence_num() const;
+  // Return true if at initial state.
+  bool in_initial_state() const;
+  // Return true when sync is on - decode all layers.
+  bool full_sync() const;
+
+ private:
+  void UpdateSyncState(const VCMFrameBuffer* frame);
+  // Designated continuity functions
+  bool ContinuousPictureId(int picture_id) const;
+  bool ContinuousSeqNum(uint16_t seq_num) const;
+  bool ContinuousLayer(int temporal_id, int tl0_pic_id) const;
+  bool ContinuousFrameRefs(const VCMFrameBuffer* frame) const;
+  bool UsingPictureId(const VCMFrameBuffer* frame) const;
+  bool UsingFlexibleMode(const VCMFrameBuffer* frame) const;
+  bool AheadOfFramesDecodedClearedTo(uint16_t index) const;
+  bool HaveSpsAndPps(const std::vector<NaluInfo>& nalus) const;
+
+  // Keep state of last decoded frame.
+  // TODO(mikhal/stefan): create designated classes to handle these types.
+  uint16_t sequence_num_;
+  uint32_t time_stamp_;
+  int picture_id_;
+  int temporal_id_;
+  int tl0_pic_id_;
+  bool full_sync_;  // Sync flag when temporal layers are used.
+  bool in_initial_state_;
+
+  // Used to check references in flexible mode.
+  bool frame_decoded_[kFrameDecodedLength];
+  uint16_t frame_decoded_cleared_to_;
+  std::set<int> received_sps_;
+  std::map<int, int> received_pps_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_DECODING_STATE_H_
diff --git a/third_party/libwebrtc/modules/video_coding/decoding_state_unittest.cc b/third_party/libwebrtc/modules/video_coding/decoding_state_unittest.cc
new file mode 100644
index 0000000000..bef7f81c62
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/decoding_state_unittest.cc
@@ -0,0 +1,713 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/decoding_state.h"
+
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "modules/video_coding/frame_buffer.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/packet.h"
+#include "modules/video_coding/session_info.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(TestDecodingState, Sanity) {
+  VCMDecodingState dec_state;
+  dec_state.Reset();
+  EXPECT_TRUE(dec_state.in_initial_state());
+  EXPECT_TRUE(dec_state.full_sync());
+}
+
+TEST(TestDecodingState, FrameContinuity) {
+  VCMDecodingState dec_state;
+  // Check that makes decision based on correct method.
+  VCMFrameBuffer frame;
+  VCMFrameBuffer frame_key;
+  VCMPacket packet;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.video_header.codec = kVideoCodecVP8;
+  auto& vp8_header =
+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.pictureId = 0x007F;
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  // Always start with a key frame.
+  dec_state.Reset();
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  EXPECT_LE(0, frame_key.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame_key));
+  dec_state.SetState(&frame);
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  // Use pictureId
+  packet.video_header.is_first_packet_in_frame = false;
+  vp8_header.pictureId = 0x0002;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+  frame.Reset();
+  vp8_header.pictureId = 0;
+  packet.seqNum = 10;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+
+  // Use sequence numbers.
+  vp8_header.pictureId = kNoPictureId;
+  frame.Reset();
+  packet.seqNum = dec_state.sequence_num() - 1u;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+  frame.Reset();
+  packet.seqNum = dec_state.sequence_num() + 1u;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  // Insert another packet to this frame
+  packet.seqNum++;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  // Verify wrap.
+  EXPECT_LE(dec_state.sequence_num(), 0xffff);
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Insert packet with temporal info.
+  dec_state.Reset();
+  frame.Reset();
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 0;
+  packet.seqNum = 1;
+  packet.timestamp = 1;
+  EXPECT_TRUE(dec_state.full_sync());
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  EXPECT_TRUE(dec_state.full_sync());
+  frame.Reset();
+  // 1 layer up - still good.
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 1;
+  vp8_header.pictureId = 1;
+  packet.seqNum = 2;
+  packet.timestamp = 2;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+  EXPECT_TRUE(dec_state.full_sync());
+  frame.Reset();
+  // Lost non-base layer packet => should update sync parameter.
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 3;
+  vp8_header.pictureId = 3;
+  packet.seqNum = 4;
+  packet.timestamp = 4;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+  // Now insert the next non-base layer (belonging to a next tl0PicId).
+  frame.Reset();
+  vp8_header.tl0PicIdx = 1;
+  vp8_header.temporalIdx = 2;
+  vp8_header.pictureId = 4;
+  packet.seqNum = 5;
+  packet.timestamp = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  // Checking continuity and not updating the state - this should not trigger
+  // an update of sync state.
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+  EXPECT_TRUE(dec_state.full_sync());
+  // Next base layer (dropped interim non-base layers) - should update sync.
+  frame.Reset();
+  vp8_header.tl0PicIdx = 1;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 5;
+  packet.seqNum = 6;
+  packet.timestamp = 6;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+  EXPECT_FALSE(dec_state.full_sync());
+
+  // Check wrap for temporal layers.
+  frame.Reset();
+  vp8_header.tl0PicIdx = 0x00FF;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 6;
+  packet.seqNum = 7;
+  packet.timestamp = 7;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  EXPECT_FALSE(dec_state.full_sync());
+  frame.Reset();
+  vp8_header.tl0PicIdx = 0x0000;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 7;
+  packet.seqNum = 8;
+  packet.timestamp = 8;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  // The current frame is not continuous
+  dec_state.SetState(&frame);
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+}
+
+TEST(TestDecodingState, UpdateOldPacket) {
+  VCMDecodingState dec_state;
+  // Update only if zero size and newer than previous.
+  // Should only update if the timeStamp match.
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.timestamp = 1;
+  packet.seqNum = 1;
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  EXPECT_EQ(dec_state.sequence_num(), 1);
+  // Insert an empty packet that does not belong to the same frame.
+  // => Sequence num should be the same.
+  packet.timestamp = 2;
+  dec_state.UpdateOldPacket(&packet);
+  EXPECT_EQ(dec_state.sequence_num(), 1);
+  // Now insert empty packet belonging to the same frame.
+  packet.timestamp = 1;
+  packet.seqNum = 2;
+  packet.video_header.frame_type = VideoFrameType::kEmptyFrame;
+  packet.sizeBytes = 0;
+  dec_state.UpdateOldPacket(&packet);
+  EXPECT_EQ(dec_state.sequence_num(), 2);
+  // Now insert delta packet belonging to the same frame.
+  packet.timestamp = 1;
+  packet.seqNum = 3;
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.sizeBytes = 1400;
+  dec_state.UpdateOldPacket(&packet);
+  EXPECT_EQ(dec_state.sequence_num(), 3);
+  // Insert a packet belonging to an older timestamp - should not update the
+  // sequence number.
+  packet.timestamp = 0;
+  packet.seqNum = 4;
+  packet.video_header.frame_type = VideoFrameType::kEmptyFrame;
+  packet.sizeBytes = 0;
+  dec_state.UpdateOldPacket(&packet);
+  EXPECT_EQ(dec_state.sequence_num(), 3);
+}
+
+TEST(TestDecodingState, MultiLayerBehavior) {
+  // Identify sync/non-sync when more than one layer.
+  VCMDecodingState dec_state;
+  // Identify packets belonging to old frames/packets.
+  // Set state for current frames.
+  // tl0PicIdx 0, temporal id 0.
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.video_header.codec = kVideoCodecVP8;
+  packet.timestamp = 0;
+  packet.seqNum = 0;
+  auto& vp8_header =
+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 0;
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  // tl0PicIdx 0, temporal id 1.
+  frame.Reset();
+  packet.timestamp = 1;
+  packet.seqNum = 1;
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 1;
+  vp8_header.pictureId = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+  EXPECT_TRUE(dec_state.full_sync());
+  // Lost tl0PicIdx 0, temporal id 2.
+  // Insert tl0PicIdx 0, temporal id 3.
+  frame.Reset();
+  packet.timestamp = 3;
+  packet.seqNum = 3;
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 3;
+  vp8_header.pictureId = 3;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+  EXPECT_FALSE(dec_state.full_sync());
+  // Insert next base layer
+  frame.Reset();
+  packet.timestamp = 4;
+  packet.seqNum = 4;
+  vp8_header.tl0PicIdx = 1;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 4;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+  EXPECT_FALSE(dec_state.full_sync());
+  // Insert key frame - should update sync value.
+  // A key frame is always a base layer.
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.timestamp = 5;
+  packet.seqNum = 5;
+  vp8_header.tl0PicIdx = 2;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+  EXPECT_TRUE(dec_state.full_sync());
+  // After sync, a continuous PictureId is required
+  // (continuous base layer is not enough )
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.timestamp = 6;
+  packet.seqNum = 6;
+  vp8_header.tl0PicIdx = 3;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 6;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  EXPECT_TRUE(dec_state.full_sync());
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.timestamp = 8;
+  packet.seqNum = 8;
+  vp8_header.tl0PicIdx = 4;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 8;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+  EXPECT_TRUE(dec_state.full_sync());
+  dec_state.SetState(&frame);
+  EXPECT_FALSE(dec_state.full_sync());
+
+  // Insert a non-ref frame - should update sync value.
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.timestamp = 9;
+  packet.seqNum = 9;
+  vp8_header.tl0PicIdx = 4;
+  vp8_header.temporalIdx = 2;
+  vp8_header.pictureId = 9;
+  vp8_header.layerSync = true;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  EXPECT_TRUE(dec_state.full_sync());
+
+  // The following test will verify the sync flag behavior after a loss.
+  // Create the following pattern:
+  // Update base layer, lose packet 1 (sync flag on, layer 2), insert packet 3
+  // (sync flag on, layer 2) check continuity and sync flag after inserting
+  // packet 2 (sync flag on, layer 1).
+  // Base layer.
+  frame.Reset();
+  dec_state.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.markerBit = 1;
+  packet.timestamp = 0;
+  packet.seqNum = 0;
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 0;
+  vp8_header.layerSync = false;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  EXPECT_TRUE(dec_state.full_sync());
+  // Layer 2 - 2 packets (insert one, lose one).
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.markerBit = 0;
+  packet.timestamp = 1;
+  packet.seqNum = 1;
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 2;
+  vp8_header.pictureId = 1;
+  vp8_header.layerSync = true;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  // Layer 1
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.markerBit = 1;
+  packet.timestamp = 2;
+  packet.seqNum = 3;
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 1;
+  vp8_header.pictureId = 2;
+  vp8_header.layerSync = true;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+  EXPECT_TRUE(dec_state.full_sync());
+}
+
+TEST(TestDecodingState, DiscontinuousPicIdContinuousSeqNum) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet.video_header.codec = kVideoCodecVP8;
+  packet.timestamp = 0;
+  packet.seqNum = 0;
+  auto& vp8_header =
+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 0;
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  EXPECT_TRUE(dec_state.full_sync());
+
+  // Continuous sequence number but discontinuous picture id. This implies a
+  // a loss and we have to fall back to only decoding the base layer.
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.timestamp += 3000;
+  ++packet.seqNum;
+  vp8_header.temporalIdx = 1;
+  vp8_header.pictureId = 2;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+  EXPECT_FALSE(dec_state.full_sync());
+}
+
+TEST(TestDecodingState, OldInput) {
+  VCMDecodingState dec_state;
+  // Identify packets belonging to old frames/packets.
+  // Set state for current frames.
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.timestamp = 10;
+  packet.seqNum = 1;
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  packet.timestamp = 9;
+  EXPECT_TRUE(dec_state.IsOldPacket(&packet));
+  // Check for old frame
+  frame.Reset();
+  frame.InsertPacket(packet, 0, frame_data);
+  EXPECT_TRUE(dec_state.IsOldFrame(&frame));
+}
+
+TEST(TestDecodingState, PictureIdRepeat) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet.video_header.codec = kVideoCodecVP8;
+  packet.timestamp = 0;
+  packet.seqNum = 0;
+  auto& vp8_header =
+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.tl0PicIdx = 0;
+  vp8_header.temporalIdx = 0;
+  vp8_header.pictureId = 0;
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  // tl0PicIdx 0, temporal id 1.
+  frame.Reset();
+  ++packet.timestamp;
+  ++packet.seqNum;
+  vp8_header.temporalIdx++;
+  vp8_header.pictureId++;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  frame.Reset();
+  // Testing only gap in tl0PicIdx when tl0PicIdx in continuous.
+  vp8_header.tl0PicIdx += 3;
+  vp8_header.temporalIdx++;
+  vp8_header.tl0PicIdx = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+}
+
+TEST(TestDecodingState, FrameContinuityFlexibleModeKeyFrame) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  uint8_t data[] = "I need a data pointer for this test!";
+  packet.sizeBytes = sizeof(data);
+  packet.dataPtr = data;
+  packet.video_header.codec = kVideoCodecVP9;
+
+  auto& vp9_hdr =
+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
+  vp9_hdr.picture_id = 10;
+  vp9_hdr.flexible_mode = true;
+
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+
+  // Key frame as first frame
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Key frame again
+  vp9_hdr.picture_id = 11;
+  frame.Reset();
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 11, continuous
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_hdr.picture_id = 12;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+}
+
+TEST(TestDecodingState, FrameContinuityFlexibleModeOutOfOrderFrames) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  uint8_t data[] = "I need a data pointer for this test!";
+  packet.sizeBytes = sizeof(data);
+  packet.dataPtr = data;
+  packet.video_header.codec = kVideoCodecVP9;
+
+  auto& vp9_hdr =
+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
+  vp9_hdr.picture_id = 10;
+  vp9_hdr.flexible_mode = true;
+
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+
+  // Key frame as first frame
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 10, continuous
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_hdr.picture_id = 15;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Out of order, last id 15, this id 12, ref to 10, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 12;
+  vp9_hdr.pid_diff[0] = 2;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref 10, 12, 15, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 20;
+  vp9_hdr.num_ref_pics = 3;
+  vp9_hdr.pid_diff[0] = 10;
+  vp9_hdr.pid_diff[1] = 8;
+  vp9_hdr.pid_diff[2] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+}
+
+TEST(TestDecodingState, FrameContinuityFlexibleModeGeneral) {
+  VCMDecodingState dec_state;
+  VCMFrameBuffer frame;
+  VCMPacket packet;
+  packet.video_header.is_first_packet_in_frame = true;
+  packet.timestamp = 1;
+  packet.seqNum = 0xffff;
+  uint8_t data[] = "I need a data pointer for this test!";
+  packet.sizeBytes = sizeof(data);
+  packet.dataPtr = data;
+  packet.video_header.codec = kVideoCodecVP9;
+
+  auto& vp9_hdr =
+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
+  vp9_hdr.picture_id = 10;
+  vp9_hdr.flexible_mode = true;
+
+  FrameData frame_data;
+  frame_data.rtt_ms = 0;
+  frame_data.rolling_average_packets_per_frame = -1;
+
+  // Key frame as first frame
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+
+  // Delta frame as first frame
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Key frame then delta frame
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  dec_state.SetState(&frame);
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.picture_id = 15;
+  vp9_hdr.pid_diff[0] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 11, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 16;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Ref to 15, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 16;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to 11 and 15, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 20;
+  vp9_hdr.num_ref_pics = 2;
+  vp9_hdr.pid_diff[0] = 9;
+  vp9_hdr.pid_diff[1] = 5;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Ref to 10, 15 and 16, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 22;
+  vp9_hdr.num_ref_pics = 3;
+  vp9_hdr.pid_diff[0] = 12;
+  vp9_hdr.pid_diff[1] = 7;
+  vp9_hdr.pid_diff[2] = 6;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Key Frame, continuous
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  vp9_hdr.picture_id = VCMDecodingState::kFrameDecodedLength - 2;
+  vp9_hdr.num_ref_pics = 0;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame at last index, ref to KF, continuous
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_hdr.picture_id = VCMDecodingState::kFrameDecodedLength - 1;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame after wrapping buffer length, ref to last index, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 0;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame after wrapping start frame, ref to 0, continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 20;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 20;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Frame after wrapping start frame, ref to 10, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 23;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 13;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+
+  // Key frame, continuous
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  vp9_hdr.picture_id = 25;
+  vp9_hdr.num_ref_pics = 0;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to KF, continuous
+  frame.Reset();
+  packet.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_hdr.picture_id = 26;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 1;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
+  dec_state.SetState(&frame);
+
+  // Ref to frame previous to KF, not continuous
+  frame.Reset();
+  vp9_hdr.picture_id = 30;
+  vp9_hdr.num_ref_pics = 1;
+  vp9_hdr.pid_diff[0] = 30;
+  EXPECT_LE(0, frame.InsertPacket(packet, 0, frame_data));
+  EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/encoded_frame.cc b/third_party/libwebrtc/modules/video_coding/encoded_frame.cc
new file mode 100644
index 0000000000..637a20cfc9
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/encoded_frame.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/encoded_frame.h"
+
+#include <string.h>
+
+#include "absl/types/variant.h"
+#include "api/video/video_timing.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+
+namespace webrtc {
+
+VCMEncodedFrame::VCMEncodedFrame()
+    : webrtc::EncodedImage(),
+      _renderTimeMs(-1),
+      _payloadType(0),
+      _missingFrame(false),
+      _codec(kVideoCodecGeneric) {
+  _codecSpecificInfo.codecType = kVideoCodecGeneric;
+}
+
+VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame&) = default;
+
+VCMEncodedFrame::~VCMEncodedFrame() {
+  Reset();
+}
+
+void VCMEncodedFrame::Reset() {
+  SetTimestamp(0);
+  SetSpatialIndex(absl::nullopt);
+  _renderTimeMs = -1;
+  _payloadType = 0;
+  _frameType = VideoFrameType::kVideoFrameDelta;
+  _encodedWidth = 0;
+  _encodedHeight = 0;
+  _missingFrame = false;
+  set_size(0);
+  _codecSpecificInfo.codecType = kVideoCodecGeneric;
+  _codec = kVideoCodecGeneric;
+  rotation_ = kVideoRotation_0;
+  content_type_ = VideoContentType::UNSPECIFIED;
+  timing_.flags = VideoSendTiming::kInvalid;
+}
+
+void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header) {
+  if (header) {
+    switch (header->codec) {
+      case kVideoCodecVP8: {
+        const auto& vp8_header =
+            absl::get<RTPVideoHeaderVP8>(header->video_type_header);
+        if (_codecSpecificInfo.codecType != kVideoCodecVP8) {
+          // This is the first packet for this frame.
+          _codecSpecificInfo.codecSpecific.VP8.temporalIdx = 0;
+          _codecSpecificInfo.codecSpecific.VP8.layerSync = false;
+          _codecSpecificInfo.codecSpecific.VP8.keyIdx = -1;
+          _codecSpecificInfo.codecType = kVideoCodecVP8;
+        }
+        _codecSpecificInfo.codecSpecific.VP8.nonReference =
+            vp8_header.nonReference;
+        if (vp8_header.temporalIdx != kNoTemporalIdx) {
+          _codecSpecificInfo.codecSpecific.VP8.temporalIdx =
+              vp8_header.temporalIdx;
+          _codecSpecificInfo.codecSpecific.VP8.layerSync = vp8_header.layerSync;
+        }
+        if (vp8_header.keyIdx != kNoKeyIdx) {
+          _codecSpecificInfo.codecSpecific.VP8.keyIdx = vp8_header.keyIdx;
+        }
+        break;
+      }
+      case kVideoCodecVP9: {
+        const auto& vp9_header =
+            absl::get<RTPVideoHeaderVP9>(header->video_type_header);
+        if (_codecSpecificInfo.codecType != kVideoCodecVP9) {
+          // This is the first packet for this frame.
+          _codecSpecificInfo.codecSpecific.VP9.temporal_idx = 0;
+          _codecSpecificInfo.codecSpecific.VP9.gof_idx = 0;
+          _codecSpecificInfo.codecSpecific.VP9.inter_layer_predicted = false;
+          _codecSpecificInfo.codecType = kVideoCodecVP9;
+        }
+        _codecSpecificInfo.codecSpecific.VP9.inter_pic_predicted =
+            vp9_header.inter_pic_predicted;
+        _codecSpecificInfo.codecSpecific.VP9.flexible_mode =
+            vp9_header.flexible_mode;
+        _codecSpecificInfo.codecSpecific.VP9.num_ref_pics =
+            vp9_header.num_ref_pics;
+        for (uint8_t r = 0; r < vp9_header.num_ref_pics; ++r) {
+          _codecSpecificInfo.codecSpecific.VP9.p_diff[r] =
+              vp9_header.pid_diff[r];
+        }
+        _codecSpecificInfo.codecSpecific.VP9.ss_data_available =
+            vp9_header.ss_data_available;
+        if (vp9_header.temporal_idx != kNoTemporalIdx) {
+          _codecSpecificInfo.codecSpecific.VP9.temporal_idx =
+              vp9_header.temporal_idx;
+          _codecSpecificInfo.codecSpecific.VP9.temporal_up_switch =
+              vp9_header.temporal_up_switch;
+        }
+        if (vp9_header.spatial_idx != kNoSpatialIdx) {
+          _codecSpecificInfo.codecSpecific.VP9.inter_layer_predicted =
+              vp9_header.inter_layer_predicted;
+          SetSpatialIndex(vp9_header.spatial_idx);
+        }
+        if (vp9_header.gof_idx != kNoGofIdx) {
+          _codecSpecificInfo.codecSpecific.VP9.gof_idx = vp9_header.gof_idx;
+        }
+        if (vp9_header.ss_data_available) {
+          _codecSpecificInfo.codecSpecific.VP9.num_spatial_layers =
+              vp9_header.num_spatial_layers;
+          _codecSpecificInfo.codecSpecific.VP9
+              .spatial_layer_resolution_present =
+              vp9_header.spatial_layer_resolution_present;
+          if (vp9_header.spatial_layer_resolution_present) {
+            for (size_t i = 0; i < vp9_header.num_spatial_layers; ++i) {
+              _codecSpecificInfo.codecSpecific.VP9.width[i] =
+                  vp9_header.width[i];
+              _codecSpecificInfo.codecSpecific.VP9.height[i] =
+                  vp9_header.height[i];
+            }
+          }
+          _codecSpecificInfo.codecSpecific.VP9.gof.CopyGofInfoVP9(
+              vp9_header.gof);
+        }
+        break;
+      }
+      case kVideoCodecH264: {
+        _codecSpecificInfo.codecType = kVideoCodecH264;
+        break;
+      }
+      case kVideoCodecAV1: {
+        _codecSpecificInfo.codecType = kVideoCodecAV1;
+        break;
+      }
+      default: {
+        _codecSpecificInfo.codecType = kVideoCodecGeneric;
+        break;
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/encoded_frame.h b/third_party/libwebrtc/modules/video_coding/encoded_frame.h
new file mode 100644
index 0000000000..9cc769277d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/encoded_frame.h
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+#define MODULES_VIDEO_CODING_ENCODED_FRAME_H_
+
+#include <vector>
+
+#include "api/video/encoded_image.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class RTC_EXPORT VCMEncodedFrame : public EncodedImage {
+ public:
+  VCMEncodedFrame();
+  VCMEncodedFrame(const VCMEncodedFrame&);
+
+  ~VCMEncodedFrame();
+  /**
+   *   Set render time in milliseconds
+   */
+  void SetRenderTime(const int64_t renderTimeMs) {
+    _renderTimeMs = renderTimeMs;
+  }
+
+  VideoPlayoutDelay PlayoutDelay() const { return playout_delay_; }
+
+  void SetPlayoutDelay(VideoPlayoutDelay playout_delay) {
+    playout_delay_ = playout_delay;
+  }
+
+  /**
+   *   Get the encoded image
+   */
+  const webrtc::EncodedImage& EncodedImage() const {
+    return static_cast<const webrtc::EncodedImage&>(*this);
+  }
+
+  using EncodedImage::ColorSpace;
+  using EncodedImage::data;
+  using EncodedImage::GetEncodedData;
+  using EncodedImage::NtpTimeMs;
+  using EncodedImage::PacketInfos;
+  using EncodedImage::set_size;
+  using EncodedImage::SetColorSpace;
+  using EncodedImage::SetEncodedData;
+  using EncodedImage::SetPacketInfos;
+  using EncodedImage::SetSpatialIndex;
+  using EncodedImage::SetSpatialLayerFrameSize;
+  using EncodedImage::SetTimestamp;
+  using EncodedImage::size;
+  using EncodedImage::SpatialIndex;
+  using EncodedImage::SpatialLayerFrameSize;
+  using EncodedImage::Timestamp;
+
+  /**
+   *   Get render time in milliseconds
+   */
+  int64_t RenderTimeMs() const { return _renderTimeMs; }
+  /**
+   *   Get frame type
+   */
+  webrtc::VideoFrameType FrameType() const { return _frameType; }
+  /**
+   *   Set frame type
+   */
+  void SetFrameType(webrtc::VideoFrameType frame_type) {
+    _frameType = frame_type;
+  }
+  /**
+   *   Get frame rotation
+   */
+  VideoRotation rotation() const { return rotation_; }
+  /**
+   *  Get video content type
+   */
+  VideoContentType contentType() const { return content_type_; }
+  /**
+   * Get video timing
+   */
+  EncodedImage::Timing video_timing() const { return timing_; }
+  EncodedImage::Timing* video_timing_mutable() { return &timing_; }
+  /**
+   *   True if there's a frame missing before this frame
+   */
+  bool MissingFrame() const { return _missingFrame; }
+  /**
+   *   Payload type of the encoded payload
+   */
+  uint8_t PayloadType() const { return _payloadType; }
+  /**
+   *   Get codec specific info.
+   *   The returned pointer is only valid as long as the VCMEncodedFrame
+   *   is valid. Also, VCMEncodedFrame owns the pointer and will delete
+   *   the object.
+   */
+  const CodecSpecificInfo* CodecSpecific() const { return &_codecSpecificInfo; }
+  void SetCodecSpecific(const CodecSpecificInfo* codec_specific) {
+    _codecSpecificInfo = *codec_specific;
+  }
+
+ protected:
+  void Reset();
+
+  void CopyCodecSpecific(const RTPVideoHeader* header);
+
+  int64_t _renderTimeMs;
+  uint8_t _payloadType;
+  bool _missingFrame;
+  CodecSpecificInfo _codecSpecificInfo;
+  webrtc::VideoCodecType _codec;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_ENCODED_FRAME_H_
diff --git a/third_party/libwebrtc/modules/video_coding/encoded_frame_gn/moz.build b/third_party/libwebrtc/modules/video_coding/encoded_frame_gn/moz.build
new file mode 100644
index 0000000000..159b77d10c
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/encoded_frame_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/encoded_frame.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("encoded_frame_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/event_wrapper.cc b/third_party/libwebrtc/modules/video_coding/event_wrapper.cc
new file mode 100644
index 0000000000..748c92f637
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/event_wrapper.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/event_wrapper.h"
+
+#include "rtc_base/event.h"
+
+namespace webrtc {
+
+class EventWrapperImpl : public EventWrapper {
+ public:
+  ~EventWrapperImpl() override {}
+
+  bool Set() override {
+    event_.Set();
+    return true;
+  }
+
+  // TODO(bugs.webrtc.org/14366): Migrate to TimeDelta.
+  EventTypeWrapper Wait(int max_time_ms) override {
+    return event_.Wait(TimeDelta::Millis(max_time_ms)) ? kEventSignaled
+                                                       : kEventTimeout;
+  }
+
+ private:
+  rtc::Event event_;
+};
+
+// static
+EventWrapper* EventWrapper::Create() {
+  return new EventWrapperImpl();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/event_wrapper.h b/third_party/libwebrtc/modules/video_coding/event_wrapper.h
new file mode 100644
index 0000000000..c5e5661282
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/event_wrapper.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_EVENT_WRAPPER_H_
+#define MODULES_VIDEO_CODING_EVENT_WRAPPER_H_
+
+namespace webrtc {
+enum EventTypeWrapper { kEventSignaled = 1, kEventTimeout = 2 };
+
+class EventWrapper {
+ public:
+  // Factory method. Constructor disabled.
+  static EventWrapper* Create();
+
+  virtual ~EventWrapper() {}
+
+  // Releases threads who are calling Wait() and has started waiting. Please
+  // note that a thread calling Wait() will not start waiting immediately.
+  // assumptions to the contrary is a very common source of issues in
+  // multithreaded programming.
+  // Set is sticky in the sense that it will release at least one thread
+  // either immediately or some time in the future.
+  virtual bool Set() = 0;
+
+  // Puts the calling thread into a wait state. The thread may be released
+  // by a Set() call depending on if other threads are waiting and if so on
+  // timing. The thread that was released will reset the event before leaving
+  // preventing more threads from being released. If multiple threads
+  // are waiting for the same Set(), only one (random) thread is guaranteed to
+  // be released. It is possible that multiple (random) threads are released
+  // Depending on timing.
+  //
+  // `max_time_ms` is the maximum time to wait in milliseconds.
+  // TODO(bugs.webrtc.org/14366): Migrate to TimeDelta.
+  virtual EventTypeWrapper Wait(int max_time_ms) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_EVENT_WRAPPER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/fec_controller_default.cc b/third_party/libwebrtc/modules/video_coding/fec_controller_default.cc
new file mode 100644
index 0000000000..f204b01c7c
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/fec_controller_default.cc
@@ -0,0 +1,211 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/fec_controller_default.h"  // NOLINT
+
+#include <stdlib.h>
+
+#include <algorithm>
+#include <string>
+
+#include "modules/include/module_fec_types.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+const float kProtectionOverheadRateThreshold = 0.5;
+
+FecControllerDefault::FecControllerDefault(
+    Clock* clock,
+    VCMProtectionCallback* protection_callback)
+    : clock_(clock),
+      protection_callback_(protection_callback),
+      loss_prot_logic_(new media_optimization::VCMLossProtectionLogic(
+          clock_->TimeInMilliseconds())),
+      max_payload_size_(1460),
+      overhead_threshold_(GetProtectionOverheadRateThreshold()) {}
+
+FecControllerDefault::FecControllerDefault(Clock* clock)
+    : clock_(clock),
+      loss_prot_logic_(new media_optimization::VCMLossProtectionLogic(
+          clock_->TimeInMilliseconds())),
+      max_payload_size_(1460),
+      overhead_threshold_(GetProtectionOverheadRateThreshold()) {}
+
+FecControllerDefault::~FecControllerDefault(void) {
+  loss_prot_logic_->Release();
+}
+
+void FecControllerDefault::SetProtectionCallback(
+    VCMProtectionCallback* protection_callback) {
+  protection_callback_ = protection_callback;
+}
+
+void FecControllerDefault::SetEncodingData(size_t width,
+                                           size_t height,
+                                           size_t num_temporal_layers,
+                                           size_t max_payload_size) {
+  MutexLock lock(&mutex_);
+  loss_prot_logic_->UpdateFrameSize(width, height);
+  loss_prot_logic_->UpdateNumLayers(num_temporal_layers);
+  max_payload_size_ = max_payload_size;
+}
+
+float FecControllerDefault::GetProtectionOverheadRateThreshold() {
+  float overhead_threshold =
+      strtof(webrtc::field_trial::FindFullName(
+                 "WebRTC-ProtectionOverheadRateThreshold")
+                 .c_str(),
+             nullptr);
+  if (overhead_threshold > 0 && overhead_threshold <= 1) {
+    RTC_LOG(LS_INFO) << "ProtectionOverheadRateThreshold is set to "
+                     << overhead_threshold;
+    return overhead_threshold;
+  } else if (overhead_threshold < 0 || overhead_threshold > 1) {
+    RTC_LOG(LS_WARNING)
+        << "ProtectionOverheadRateThreshold field trial is set to "
+           "an invalid value, expecting a value between (0, 1].";
+  }
+  // WebRTC-ProtectionOverheadRateThreshold field trial string is not found, use
+  // the default value.
+  return kProtectionOverheadRateThreshold;
+}
+
+uint32_t FecControllerDefault::UpdateFecRates(
+    uint32_t estimated_bitrate_bps,
+    int actual_framerate_fps,
+    uint8_t fraction_lost,
+    std::vector<bool> loss_mask_vector,
+    int64_t round_trip_time_ms) {
+  float target_bitrate_kbps =
+      static_cast<float>(estimated_bitrate_bps) / 1000.0f;
+  // Sanity check.
+  if (actual_framerate_fps < 1.0) {
+    actual_framerate_fps = 1.0;
+  }
+  FecProtectionParams delta_fec_params;
+  FecProtectionParams key_fec_params;
+  {
+    MutexLock lock(&mutex_);
+    loss_prot_logic_->UpdateBitRate(target_bitrate_kbps);
+    loss_prot_logic_->UpdateRtt(round_trip_time_ms);
+    // Update frame rate for the loss protection logic class: frame rate should
+    // be the actual/sent rate.
+    loss_prot_logic_->UpdateFrameRate(actual_framerate_fps);
+    // Returns the filtered packet loss, used for the protection setting.
+    // The filtered loss may be the received loss (no filter), or some
+    // filtered value (average or max window filter).
+    // Use max window filter for now.
+    media_optimization::FilterPacketLossMode filter_mode =
+        media_optimization::kMaxFilter;
+    uint8_t packet_loss_enc = loss_prot_logic_->FilteredLoss(
+        clock_->TimeInMilliseconds(), filter_mode, fraction_lost);
+    // For now use the filtered loss for computing the robustness settings.
+    loss_prot_logic_->UpdateFilteredLossPr(packet_loss_enc);
+    if (loss_prot_logic_->SelectedType() == media_optimization::kNone) {
+      return estimated_bitrate_bps;
+    }
+    // Update method will compute the robustness settings for the given
+    // protection method and the overhead cost
+    // the protection method is set by the user via SetVideoProtection.
+    loss_prot_logic_->UpdateMethod();
+    // Get the bit cost of protection method, based on the amount of
+    // overhead data actually transmitted (including headers) the last
+    // second.
+    // Get the FEC code rate for Key frames (set to 0 when NA).
+    key_fec_params.fec_rate =
+        loss_prot_logic_->SelectedMethod()->RequiredProtectionFactorK();
+    // Get the FEC code rate for Delta frames (set to 0 when NA).
+    delta_fec_params.fec_rate =
+        loss_prot_logic_->SelectedMethod()->RequiredProtectionFactorD();
+    // The RTP module currently requires the same `max_fec_frames` for both
+    // key and delta frames.
+    delta_fec_params.max_fec_frames =
+        loss_prot_logic_->SelectedMethod()->MaxFramesFec();
+    key_fec_params.max_fec_frames =
+        loss_prot_logic_->SelectedMethod()->MaxFramesFec();
+  }
+  // Set the FEC packet mask type. `kFecMaskBursty` is more effective for
+  // consecutive losses and little/no packet re-ordering. As we currently
+  // do not have feedback data on the degree of correlated losses and packet
+  // re-ordering, we keep default setting to `kFecMaskRandom` for now.
+  delta_fec_params.fec_mask_type = kFecMaskRandom;
+  key_fec_params.fec_mask_type = kFecMaskRandom;
+  // Update protection callback with protection settings.
+  uint32_t sent_video_rate_bps = 0;
+  uint32_t sent_nack_rate_bps = 0;
+  uint32_t sent_fec_rate_bps = 0;
+  // Rate cost of the protection methods.
+  float protection_overhead_rate = 0.0f;
+  // TODO(Marco): Pass FEC protection values per layer.
+  protection_callback_->ProtectionRequest(
+      &delta_fec_params, &key_fec_params, &sent_video_rate_bps,
+      &sent_nack_rate_bps, &sent_fec_rate_bps);
+  uint32_t sent_total_rate_bps =
+      sent_video_rate_bps + sent_nack_rate_bps + sent_fec_rate_bps;
+  // Estimate the overhead costs of the next second as staying the same
+  // wrt the source bitrate.
+  if (sent_total_rate_bps > 0) {
+    protection_overhead_rate =
+        static_cast<float>(sent_nack_rate_bps + sent_fec_rate_bps) /
+        sent_total_rate_bps;
+  }
+  // Cap the overhead estimate to a threshold, default is 50%.
+  protection_overhead_rate =
+      std::min(protection_overhead_rate, overhead_threshold_);
+  // Source coding rate: total rate - protection overhead.
+  return estimated_bitrate_bps * (1.0 - protection_overhead_rate);
+}
+
+void FecControllerDefault::SetProtectionMethod(bool enable_fec,
+                                               bool enable_nack) {
+  media_optimization::VCMProtectionMethodEnum method(media_optimization::kNone);
+  if (enable_fec && enable_nack) {
+    method = media_optimization::kNackFec;
+  } else if (enable_nack) {
+    method = media_optimization::kNack;
+  } else if (enable_fec) {
+    method = media_optimization::kFec;
+  }
+  MutexLock lock(&mutex_);
+  loss_prot_logic_->SetMethod(method);
+}
+
+void FecControllerDefault::UpdateWithEncodedData(
+    const size_t encoded_image_length,
+    const VideoFrameType encoded_image_frametype) {
+  const size_t encoded_length = encoded_image_length;
+  MutexLock lock(&mutex_);
+  if (encoded_length > 0) {
+    const bool delta_frame =
+        encoded_image_frametype != VideoFrameType::kVideoFrameKey;
+    if (max_payload_size_ > 0 && encoded_length > 0) {
+      const float min_packets_per_frame =
+          encoded_length / static_cast<float>(max_payload_size_);
+      if (delta_frame) {
+        loss_prot_logic_->UpdatePacketsPerFrame(min_packets_per_frame,
+                                                clock_->TimeInMilliseconds());
+      } else {
+        loss_prot_logic_->UpdatePacketsPerFrameKey(
+            min_packets_per_frame, clock_->TimeInMilliseconds());
+      }
+    }
+    if (!delta_frame && encoded_length > 0) {
+      loss_prot_logic_->UpdateKeyFrameSize(static_cast<float>(encoded_length));
+    }
+  }
+}
+
+bool FecControllerDefault::UseLossVectorMask() {
+  return false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/fec_controller_default.h b/third_party/libwebrtc/modules/video_coding/fec_controller_default.h
new file mode 100644
index 0000000000..a97dea011b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/fec_controller_default.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_FEC_CONTROLLER_DEFAULT_H_
+#define MODULES_VIDEO_CODING_FEC_CONTROLLER_DEFAULT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/fec_controller.h"
+#include "modules/video_coding/media_opt_util.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class FecControllerDefault : public FecController {
+ public:
+  FecControllerDefault(Clock* clock,
+                       VCMProtectionCallback* protection_callback);
+  explicit FecControllerDefault(Clock* clock);
+  ~FecControllerDefault() override;
+
+  FecControllerDefault(const FecControllerDefault&) = delete;
+  FecControllerDefault& operator=(const FecControllerDefault&) = delete;
+
+  void SetProtectionCallback(
+      VCMProtectionCallback* protection_callback) override;
+  void SetProtectionMethod(bool enable_fec, bool enable_nack) override;
+  void SetEncodingData(size_t width,
+                       size_t height,
+                       size_t num_temporal_layers,
+                       size_t max_payload_size) override;
+  uint32_t UpdateFecRates(uint32_t estimated_bitrate_bps,
+                          int actual_framerate_fps,
+                          uint8_t fraction_lost,
+                          std::vector<bool> loss_mask_vector,
+                          int64_t round_trip_time_ms) override;
+  void UpdateWithEncodedData(size_t encoded_image_length,
+                             VideoFrameType encoded_image_frametype) override;
+  bool UseLossVectorMask() override;
+  float GetProtectionOverheadRateThreshold();
+
+ private:
+  enum { kBitrateAverageWinMs = 1000 };
+  Clock* const clock_;
+  VCMProtectionCallback* protection_callback_;
+  Mutex mutex_;
+  std::unique_ptr<media_optimization::VCMLossProtectionLogic> loss_prot_logic_
+      RTC_GUARDED_BY(mutex_);
+  size_t max_payload_size_ RTC_GUARDED_BY(mutex_);
+
+  const float overhead_threshold_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CODING_FEC_CONTROLLER_DEFAULT_H_
diff --git a/third_party/libwebrtc/modules/video_coding/fec_controller_unittest.cc b/third_party/libwebrtc/modules/video_coding/fec_controller_unittest.cc
new file mode 100644
index 0000000000..fda3d309a4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/fec_controller_unittest.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/fec_controller.h"
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "modules/include/module_fec_types.h"
+#include "modules/video_coding/fec_controller_default.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+static const int kCodecBitrateBps = 100000;
+
+class ProtectionBitrateCalculatorTest : public ::testing::Test {
+ protected:
+  enum {
+    kSampleRate = 90000  // RTP timestamps per second.
+  };
+
+  class ProtectionCallback : public VCMProtectionCallback {
+   public:
+    int ProtectionRequest(const FecProtectionParams* delta_params,
+                          const FecProtectionParams* key_params,
+                          uint32_t* sent_video_rate_bps,
+                          uint32_t* sent_nack_rate_bps,
+                          uint32_t* sent_fec_rate_bps) override {
+      *sent_video_rate_bps = kCodecBitrateBps;
+      *sent_nack_rate_bps = nack_rate_bps_;
+      *sent_fec_rate_bps = fec_rate_bps_;
+      return 0;
+    }
+
+    uint32_t fec_rate_bps_ = 0;
+    uint32_t nack_rate_bps_ = 0;
+  };
+
+  // Note: simulated clock starts at 1 seconds, since parts of webrtc use 0 as
+  // a special case (e.g. frame rate in media optimization).
+  ProtectionBitrateCalculatorTest()
+      : clock_(1000), fec_controller_(&clock_, &protection_callback_) {}
+
+  SimulatedClock clock_;
+  ProtectionCallback protection_callback_;
+  FecControllerDefault fec_controller_;
+};
+
+TEST_F(ProtectionBitrateCalculatorTest, ProtectsUsingFecBitrate) {
+  static const uint32_t kMaxBitrateBps = 130000;
+
+  fec_controller_.SetProtectionMethod(true /*enable_fec*/,
+                                      false /* enable_nack */);
+  fec_controller_.SetEncodingData(640, 480, 1, 1000);
+
+  // Using 10% of codec bitrate for FEC.
+  protection_callback_.fec_rate_bps_ = kCodecBitrateBps / 10;
+  uint32_t target_bitrate = fec_controller_.UpdateFecRates(
+      kMaxBitrateBps, 30, 0, std::vector<bool>(1, false), 0);
+
+  EXPECT_GT(target_bitrate, 0u);
+  EXPECT_GT(kMaxBitrateBps, target_bitrate);
+
+  // Using as much for codec bitrate as fec rate, new target rate should share
+  // both equally, but only be half of max (since that ceiling should be hit).
+  protection_callback_.fec_rate_bps_ = kCodecBitrateBps;
+  target_bitrate = fec_controller_.UpdateFecRates(
+      kMaxBitrateBps, 30, 128, std::vector<bool>(1, false), 100);
+  EXPECT_EQ(kMaxBitrateBps / 2, target_bitrate);
+}
+
+TEST_F(ProtectionBitrateCalculatorTest, ProtectsUsingNackBitrate) {
+  static const uint32_t kMaxBitrateBps = 130000;
+
+  fec_controller_.SetProtectionMethod(false /*enable_fec*/,
+                                      true /* enable_nack */);
+  fec_controller_.SetEncodingData(640, 480, 1, 1000);
+
+  uint32_t target_bitrate = fec_controller_.UpdateFecRates(
+      kMaxBitrateBps, 30, 0, std::vector<bool>(1, false), 0);
+
+  EXPECT_EQ(kMaxBitrateBps, target_bitrate);
+
+  // Using as much for codec bitrate as nack rate, new target rate should share
+  // both equally, but only be half of max (since that ceiling should be hit).
+  protection_callback_.nack_rate_bps_ = kMaxBitrateBps;
+  target_bitrate = fec_controller_.UpdateFecRates(
+      kMaxBitrateBps, 30, 128, std::vector<bool>(1, false), 100);
+  EXPECT_EQ(kMaxBitrateBps / 2, target_bitrate);
+}
+
+TEST_F(ProtectionBitrateCalculatorTest, NoProtection) {
+  static const uint32_t kMaxBitrateBps = 130000;
+
+  fec_controller_.SetProtectionMethod(false /*enable_fec*/,
+                                      false /* enable_nack */);
+  fec_controller_.SetEncodingData(640, 480, 1, 1000);
+
+  uint32_t target_bitrate = fec_controller_.UpdateFecRates(
+      kMaxBitrateBps, 30, 128, std::vector<bool>(1, false), 100);
+  EXPECT_EQ(kMaxBitrateBps, target_bitrate);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/fec_rate_table.h b/third_party/libwebrtc/modules/video_coding/fec_rate_table.h
new file mode 100644
index 0000000000..91ec0ce159
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/fec_rate_table.h
@@ -0,0 +1,461 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_FEC_RATE_TABLE_H_
+#define MODULES_VIDEO_CODING_FEC_RATE_TABLE_H_
+
+// This is a private header for media_opt_util.cc.
+// It should not be included by other files.
+
+namespace webrtc {
+
+// Table for Protection factor (code rate) of delta frames, for the XOR FEC.
+// Input is the packet loss and an effective rate (bits/frame).
+// Output is array kFecRateTable[k], where k = rate_i*129 + loss_j;
+// loss_j = 0,1,..128, and rate_i varies over some range.
+// TODO(brandtr): Consider replacing this big static table with a closed-form
+// expression instead.
+static const int kFecRateTableSize = 6450;
+static const unsigned char kFecRateTable[kFecRateTableSize] = {
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,
+    0,   0,   0,   11,  11,  11,  11,  11,  11,  11,  11,  11,  11,  11,  11,
+    11,  11,  11,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,
+    39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,
+    39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,  39,
+    39,  39,  39,  39,  39,  39,  51,  51,  51,  51,  51,  51,  51,  51,  51,
+    51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,  51,
+    51,  51,  51,  51,  51,  51,  51,  51,  51,  0,   0,   0,   0,   0,   0,
+    0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   8,   8,   8,
+    8,   8,   8,   8,   8,   8,   8,   8,   8,   8,   8,   8,   30,  30,  30,
+    30,  30,  30,  30,  30,  30,  30,  30,  30,  30,  30,  30,  56,  56,  56,
+    56,  56,  56,  56,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,
+    65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,
+    65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,  65,
+    87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,  87,
+    87,  87,  87,  87,  87,  87,  87,  87,  87,  78,  78,  78,  78,  78,  78,
+    78,  78,  78,  0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,
+    0,   0,   0,   6,   6,   6,   23,  23,  23,  23,  23,  23,  23,  23,  23,
+    23,  23,  23,  23,  23,  23,  44,  44,  44,  44,  44,  44,  50,  50,  50,
+    50,  50,  50,  50,  50,  50,  68,  68,  68,  68,  68,  68,  68,  85,  85,
+    85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,
+    85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,  85,
+    85,  85,  85,  85,  85,  85,  85,  85,  85,  105, 105, 105, 105, 105, 105,
+    105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105, 105,
+    105, 105, 105, 88,  88,  88,  88,  88,  88,  88,  88,  88,  0,   0,   0,
+    0,   0,   0,   0,   0,   0,   5,   5,   5,   5,   5,   5,   19,  19,  19,
+    36,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,  41,
+    55,  55,  55,  55,  55,  55,  69,  69,  69,  69,  69,  69,  69,  69,  69,
+    75,  75,  80,  80,  80,  80,  80,  97,  97,  97,  97,  97,  97,  97,  97,
+    97,  97,  102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102,
+    102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102, 102,
+    102, 102, 102, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116,
+    116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 116, 100, 100, 100,
+    100, 100, 100, 100, 100, 100, 0,   0,   0,   0,   0,   0,   0,   0,   4,
+    16,  16,  16,  16,  16,  16,  30,  35,  35,  47,  58,  58,  58,  58,  58,
+    58,  58,  58,  58,  58,  58,  58,  58,  58,  63,  63,  63,  63,  63,  63,
+    77,  77,  77,  77,  77,  77,  77,  82,  82,  82,  82,  94,  94,  94,  94,
+    94,  105, 105, 105, 105, 110, 110, 110, 110, 110, 110, 122, 122, 122, 122,
+    122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122,
+    122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 122, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 115, 115, 115, 115, 115, 115, 115, 115, 115,
+    0,   0,   0,   0,   0,   0,   0,   4,   14,  27,  27,  27,  27,  27,  31,
+    41,  52,  52,  56,  69,  69,  69,  69,  69,  69,  69,  69,  69,  69,  69,
+    69,  69,  69,  69,  69,  69,  69,  69,  69,  79,  79,  79,  79,  83,  83,
+    83,  94,  94,  94,  94,  106, 106, 106, 106, 106, 115, 115, 115, 115, 125,
+    125, 125, 125, 125, 125, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   0,   0,   0,   3,   3,
+    3,   17,  28,  38,  38,  38,  38,  38,  47,  51,  63,  63,  63,  72,  72,
+    72,  72,  72,  72,  72,  76,  76,  76,  76,  80,  80,  80,  80,  80,  80,
+    80,  80,  80,  84,  84,  84,  84,  93,  93,  93,  105, 105, 105, 105, 114,
+    114, 114, 114, 114, 124, 124, 124, 124, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
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+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   19,  21,
+    38,  40,  58,  58,  60,  75,  77,  77,  77,  81,  81,  107, 109, 109, 109,
+    114, 116, 116, 116, 116, 116, 116, 116, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   18,  25,  37,  44,  56,  56,  63,  75,
+    75,  75,  75,  88,  88,  111, 111, 111, 111, 112, 112, 112, 112, 112, 112,
+    112, 114, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   18,  30,  36,  48,  55,  55,  67,  73,  73,  73,  73,  97,  97,  110,
+    110, 110, 110, 110, 110, 110, 110, 110, 110, 110, 116, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   18,  34,  36,  52,  55,
+    55,  70,  72,  73,  73,  73,  102, 104, 108, 108, 108, 108, 109, 109, 109,
+    109, 109, 109, 109, 119, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   17,  35,  35,  52,  59,  59,  70,  70,  76,  76,  76,
+    99,  105, 105, 105, 105, 105, 111, 111, 111, 111, 111, 111, 111, 121, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   17,  34,
+    36,  51,  61,  62,  70,  70,  80,  80,  80,  93,  103, 103, 103, 103, 103,
+    112, 112, 112, 112, 112, 116, 118, 124, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   16,  33,  39,  50,  59,  65,  72,  72,
+    82,  82,  82,  91,  100, 100, 100, 100, 100, 109, 109, 109, 109, 109, 121,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   16,  32,  43,  48,  54,  66,  75,  75,  81,  83,  83,  92,  97,  97,
+    97,  99,  99,  105, 105, 105, 105, 105, 123, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   15,  31,  46,  47,  49,
+    69,  77,  77,  81,  85,  85,  93,  95,  95,  95,  100, 100, 102, 102, 102,
+    102, 102, 120, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   15,  30,  46,  48,  48,  70,  75,  79,  82,  87,  87,
+    92,  94,  94,  94,  103, 103, 103, 103, 103, 104, 104, 115, 120, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   15,  30,
+    45,  50,  50,  68,  70,  80,  85,  89,  89,  90,  95,  95,  95,  104, 104,
+    104, 104, 104, 109, 109, 112, 114, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   14,  29,  44,  54,  54,  64,  64,  83,
+    87,  88,  88,  88,  98,  98,  98,  103, 103, 103, 103, 103, 113, 113, 113,
+    113, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    0,   14,  29,  43,  56,  56,  61,  61,  84,  85,  88,  88,  88,  100, 100,
+    100, 102, 102, 102, 102, 102, 113, 116, 116, 116, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   14,  28,  42,  57,  57,
+    62,  62,  80,  80,  91,  91,  91,  100, 100, 100, 100, 100, 100, 100, 100,
+    109, 119, 119, 119, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 0,   14,  28,  42,  56,  56,  65,  66,  76,  76,  92,  92,
+    92,  97,  97,  97,  101, 101, 101, 101, 101, 106, 121, 121, 121, 126, 126,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 0,   13,  27,
+    41,  55,  55,  67,  72,  74,  74,  90,  90,  90,  91,  91,  91,  105, 105,
+    105, 105, 105, 107, 122, 122, 122, 123, 123, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 0,   13,  27,  40,  54,  54,  67,  76,  76,
+    76,  85,  85,  85,  85,  85,  85,  112, 112, 112, 112, 112, 112, 121, 121,
+    121, 121, 121, 126, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+    127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_FEC_RATE_TABLE_H_
diff --git a/third_party/libwebrtc/modules/video_coding/frame_buffer.cc b/third_party/libwebrtc/modules/video_coding/frame_buffer.cc
new file mode 100644
index 0000000000..787da1e5a9
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_buffer.cc
@@ -0,0 +1,265 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/frame_buffer.h"
+
+#include <string.h>
+
+#include "api/video/encoded_image.h"
+#include "api/video/video_timing.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/packet.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+VCMFrameBuffer::VCMFrameBuffer()
+    : _state(kStateEmpty), _nackCount(0), _latestPacketTimeMs(-1) {}
+
+VCMFrameBuffer::~VCMFrameBuffer() {}
+
+webrtc::VideoFrameType VCMFrameBuffer::FrameType() const {
+  return _sessionInfo.FrameType();
+}
+
+int32_t VCMFrameBuffer::GetLowSeqNum() const {
+  return _sessionInfo.LowSequenceNumber();
+}
+
+int32_t VCMFrameBuffer::GetHighSeqNum() const {
+  return _sessionInfo.HighSequenceNumber();
+}
+
+int VCMFrameBuffer::PictureId() const {
+  return _sessionInfo.PictureId();
+}
+
+int VCMFrameBuffer::TemporalId() const {
+  return _sessionInfo.TemporalId();
+}
+
+bool VCMFrameBuffer::LayerSync() const {
+  return _sessionInfo.LayerSync();
+}
+
+int VCMFrameBuffer::Tl0PicId() const {
+  return _sessionInfo.Tl0PicId();
+}
+
+std::vector<NaluInfo> VCMFrameBuffer::GetNaluInfos() const {
+  return _sessionInfo.GetNaluInfos();
+}
+
+void VCMFrameBuffer::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::SetGofInfo");
+  _sessionInfo.SetGofInfo(gof_info, idx);
+  // TODO(asapersson): Consider adding hdr->VP9.ref_picture_id for testing.
+  _codecSpecificInfo.codecSpecific.VP9.temporal_idx =
+      gof_info.temporal_idx[idx];
+  _codecSpecificInfo.codecSpecific.VP9.temporal_up_switch =
+      gof_info.temporal_up_switch[idx];
+}
+
+// Insert packet
+VCMFrameBufferEnum VCMFrameBuffer::InsertPacket(const VCMPacket& packet,
+                                                int64_t timeInMs,
+                                                const FrameData& frame_data) {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::InsertPacket");
+  RTC_DCHECK(!(NULL == packet.dataPtr && packet.sizeBytes > 0));
+  if (packet.dataPtr != NULL) {
+    _payloadType = packet.payloadType;
+  }
+
+  if (kStateEmpty == _state) {
+    // First packet (empty and/or media) inserted into this frame.
+    // store some info and set some initial values.
+    SetTimestamp(packet.timestamp);
+    // We only take the ntp timestamp of the first packet of a frame.
+    ntp_time_ms_ = packet.ntp_time_ms_;
+    _codec = packet.codec();
+    if (packet.video_header.frame_type != VideoFrameType::kEmptyFrame) {
+      // first media packet
+      SetState(kStateIncomplete);
+    }
+  }
+
+  size_t oldSize = encoded_image_buffer_ ? encoded_image_buffer_->size() : 0;
+  uint32_t requiredSizeBytes =
+      size() + packet.sizeBytes +
+      (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
+  if (requiredSizeBytes > oldSize) {
+    const uint8_t* prevBuffer = data();
+    const uint32_t increments =
+        requiredSizeBytes / kBufferIncStepSizeBytes +
+        (requiredSizeBytes % kBufferIncStepSizeBytes > 0);
+    const uint32_t newSize = oldSize + increments * kBufferIncStepSizeBytes;
+    if (newSize > kMaxJBFrameSizeBytes) {
+      RTC_LOG(LS_ERROR) << "Failed to insert packet due to frame being too "
+                           "big.";
+      return kSizeError;
+    }
+    if (data() == nullptr) {
+      encoded_image_buffer_ = EncodedImageBuffer::Create(newSize);
+      SetEncodedData(encoded_image_buffer_);
+      set_size(0);
+    } else {
+      RTC_CHECK(encoded_image_buffer_ != nullptr);
+      RTC_DCHECK_EQ(encoded_image_buffer_->data(), data());
+      encoded_image_buffer_->Realloc(newSize);
+    }
+    _sessionInfo.UpdateDataPointers(prevBuffer, data());
+  }
+
+  if (packet.width() > 0 && packet.height() > 0) {
+    _encodedWidth = packet.width();
+    _encodedHeight = packet.height();
+  }
+
+  // Don't copy payload specific data for empty packets (e.g padding packets).
+  if (packet.sizeBytes > 0)
+    CopyCodecSpecific(&packet.video_header);
+
+  int retVal = _sessionInfo.InsertPacket(
+      packet, encoded_image_buffer_ ? encoded_image_buffer_->data() : nullptr,
+      frame_data);
+  if (retVal == -1) {
+    return kSizeError;
+  } else if (retVal == -2) {
+    return kDuplicatePacket;
+  } else if (retVal == -3) {
+    return kOutOfBoundsPacket;
+  }
+  // update size
+  set_size(size() + static_cast<uint32_t>(retVal));
+
+  _latestPacketTimeMs = timeInMs;
+
+  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
+  // ts_126114v120700p.pdf Section 7.4.5.
+  // The MTSI client shall add the payload bytes as defined in this clause
+  // onto the last RTP packet in each group of packets which make up a key
+  // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
+  // (HEVC)).
+  if (packet.markerBit) {
+    rotation_ = packet.video_header.rotation;
+    content_type_ = packet.video_header.content_type;
+    if (packet.video_header.video_timing.flags != VideoSendTiming::kInvalid) {
+      timing_.encode_start_ms =
+          ntp_time_ms_ + packet.video_header.video_timing.encode_start_delta_ms;
+      timing_.encode_finish_ms =
+          ntp_time_ms_ +
+          packet.video_header.video_timing.encode_finish_delta_ms;
+      timing_.packetization_finish_ms =
+          ntp_time_ms_ +
+          packet.video_header.video_timing.packetization_finish_delta_ms;
+      timing_.pacer_exit_ms =
+          ntp_time_ms_ + packet.video_header.video_timing.pacer_exit_delta_ms;
+      timing_.network_timestamp_ms =
+          ntp_time_ms_ +
+          packet.video_header.video_timing.network_timestamp_delta_ms;
+      timing_.network2_timestamp_ms =
+          ntp_time_ms_ +
+          packet.video_header.video_timing.network2_timestamp_delta_ms;
+    }
+    timing_.flags = packet.video_header.video_timing.flags;
+  }
+
+  if (packet.is_first_packet_in_frame()) {
+    playout_delay_ = packet.video_header.playout_delay;
+  }
+
+  if (_sessionInfo.complete()) {
+    SetState(kStateComplete);
+    return kCompleteSession;
+  }
+  return kIncomplete;
+}
+
+int64_t VCMFrameBuffer::LatestPacketTimeMs() const {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::LatestPacketTimeMs");
+  return _latestPacketTimeMs;
+}
+
+void VCMFrameBuffer::IncrementNackCount() {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::IncrementNackCount");
+  _nackCount++;
+}
+
+int16_t VCMFrameBuffer::GetNackCount() const {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::GetNackCount");
+  return _nackCount;
+}
+
+bool VCMFrameBuffer::HaveFirstPacket() const {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::HaveFirstPacket");
+  return _sessionInfo.HaveFirstPacket();
+}
+
+int VCMFrameBuffer::NumPackets() const {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::NumPackets");
+  return _sessionInfo.NumPackets();
+}
+
+void VCMFrameBuffer::Reset() {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::Reset");
+  set_size(0);
+  _sessionInfo.Reset();
+  _payloadType = 0;
+  _nackCount = 0;
+  _latestPacketTimeMs = -1;
+  _state = kStateEmpty;
+  VCMEncodedFrame::Reset();
+}
+
+// Set state of frame
+void VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state) {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::SetState");
+  if (_state == state) {
+    return;
+  }
+  switch (state) {
+    case kStateIncomplete:
+      // we can go to this state from state kStateEmpty
+      RTC_DCHECK_EQ(_state, kStateEmpty);
+
+      // Do nothing, we received a packet
+      break;
+
+    case kStateComplete:
+      RTC_DCHECK(_state == kStateEmpty || _state == kStateIncomplete);
+
+      break;
+
+    case kStateEmpty:
+      // Should only be set to empty through Reset().
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+  _state = state;
+}
+
+// Get current state of frame
+VCMFrameBufferStateEnum VCMFrameBuffer::GetState() const {
+  return _state;
+}
+
+void VCMFrameBuffer::PrepareForDecode(bool continuous) {
+  TRACE_EVENT0("webrtc", "VCMFrameBuffer::PrepareForDecode");
+  size_t bytes_removed = _sessionInfo.MakeDecodable();
+  set_size(size() - bytes_removed);
+  // Transfer frame information to EncodedFrame and create any codec
+  // specific information.
+  _frameType = _sessionInfo.FrameType();
+  _missingFrame = !continuous;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/frame_buffer.h b/third_party/libwebrtc/modules/video_coding/frame_buffer.h
new file mode 100644
index 0000000000..76df28e588
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_buffer.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+#define MODULES_VIDEO_CODING_FRAME_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "modules/video_coding/encoded_frame.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/jitter_buffer_common.h"
+#include "modules/video_coding/packet.h"
+#include "modules/video_coding/session_info.h"
+
+namespace webrtc {
+
+class VCMFrameBuffer : public VCMEncodedFrame {
+ public:
+  VCMFrameBuffer();
+  virtual ~VCMFrameBuffer();
+
+  virtual void Reset();
+
+  VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
+                                  int64_t timeInMs,
+                                  const FrameData& frame_data);
+
+  // State
+  // Get current state of frame
+  VCMFrameBufferStateEnum GetState() const;
+  void PrepareForDecode(bool continuous);
+
+  bool IsSessionComplete() const;
+  bool HaveFirstPacket() const;
+  int NumPackets() const;
+
+  // Sequence numbers
+  // Get lowest packet sequence number in frame
+  int32_t GetLowSeqNum() const;
+  // Get highest packet sequence number in frame
+  int32_t GetHighSeqNum() const;
+
+  int PictureId() const;
+  int TemporalId() const;
+  bool LayerSync() const;
+  int Tl0PicId() const;
+
+  std::vector<NaluInfo> GetNaluInfos() const;
+
+  void SetGofInfo(const GofInfoVP9& gof_info, size_t idx);
+
+  // Increments a counter to keep track of the number of packets of this frame
+  // which were NACKed before they arrived.
+  void IncrementNackCount();
+  // Returns the number of packets of this frame which were NACKed before they
+  // arrived.
+  int16_t GetNackCount() const;
+
+  int64_t LatestPacketTimeMs() const;
+
+  webrtc::VideoFrameType FrameType() const;
+
+ private:
+  void SetState(VCMFrameBufferStateEnum state);  // Set state of frame
+
+  VCMFrameBufferStateEnum _state;  // Current state of the frame
+  // Set with SetEncodedData, but keep pointer to the concrete class here, to
+  // enable reallocation and mutation.
+  rtc::scoped_refptr<EncodedImageBuffer> encoded_image_buffer_;
+  VCMSessionInfo _sessionInfo;
+  uint16_t _nackCount;
+  int64_t _latestPacketTimeMs;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_FRAME_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/frame_buffer2.cc b/third_party/libwebrtc/modules/video_coding/frame_buffer2.cc
new file mode 100644
index 0000000000..a70b143a29
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_buffer2.cc
@@ -0,0 +1,625 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/frame_buffer2.h"
+
+#include <algorithm>
+#include <cstdlib>
+#include <iterator>
+#include <memory>
+#include <queue>
+#include <utility>
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_timing.h"
+#include "modules/video_coding/frame_helpers.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "modules/video_coding/timing/jitter_estimator.h"
+#include "modules/video_coding/timing/timing.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/rtt_mult_experiment.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace video_coding {
+
+namespace {
+// Max number of frames the buffer will hold.
+constexpr size_t kMaxFramesBuffered = 800;
+
+// Default value for the maximum decode queue size that is used when the
+// low-latency renderer is used.
+constexpr size_t kZeroPlayoutDelayDefaultMaxDecodeQueueSize = 8;
+
+// Max number of decoded frame info that will be saved.
+constexpr int kMaxFramesHistory = 1 << 13;
+
+// The time it's allowed for a frame to be late to its rendering prediction and
+// still be rendered.
+constexpr int kMaxAllowedFrameDelayMs = 5;
+
+constexpr int64_t kLogNonDecodedIntervalMs = 5000;
+}  // namespace
+
+FrameBuffer::FrameBuffer(Clock* clock,
+                         VCMTiming* timing,
+                         const FieldTrialsView& field_trials)
+    : decoded_frames_history_(kMaxFramesHistory),
+      clock_(clock),
+      callback_queue_(nullptr),
+      jitter_estimator_(clock, field_trials),
+      timing_(timing),
+      stopped_(false),
+      protection_mode_(kProtectionNack),
+      last_log_non_decoded_ms_(-kLogNonDecodedIntervalMs),
+      rtt_mult_settings_(RttMultExperiment::GetRttMultValue()),
+      zero_playout_delay_max_decode_queue_size_(
+          "max_decode_queue_size",
+          kZeroPlayoutDelayDefaultMaxDecodeQueueSize) {
+  ParseFieldTrial({&zero_playout_delay_max_decode_queue_size_},
+                  field_trials.Lookup("WebRTC-ZeroPlayoutDelay"));
+  callback_checker_.Detach();
+}
+
+FrameBuffer::~FrameBuffer() {
+  RTC_DCHECK_RUN_ON(&construction_checker_);
+}
+
+void FrameBuffer::NextFrame(int64_t max_wait_time_ms,
+                            bool keyframe_required,
+                            TaskQueueBase* callback_queue,
+                            NextFrameCallback handler) {
+  RTC_DCHECK_RUN_ON(&callback_checker_);
+  RTC_DCHECK(callback_queue->IsCurrent());
+  TRACE_EVENT0("webrtc", "FrameBuffer::NextFrame");
+  int64_t latest_return_time_ms =
+      clock_->TimeInMilliseconds() + max_wait_time_ms;
+
+  MutexLock lock(&mutex_);
+  if (stopped_) {
+    return;
+  }
+  latest_return_time_ms_ = latest_return_time_ms;
+  keyframe_required_ = keyframe_required;
+  frame_handler_ = handler;
+  callback_queue_ = callback_queue;
+  StartWaitForNextFrameOnQueue();
+}
+
+void FrameBuffer::StartWaitForNextFrameOnQueue() {
+  RTC_DCHECK(callback_queue_);
+  RTC_DCHECK(!callback_task_.Running());
+  int64_t wait_ms = FindNextFrame(clock_->CurrentTime());
+  callback_task_ = RepeatingTaskHandle::DelayedStart(
+      callback_queue_, TimeDelta::Millis(wait_ms),
+      [this] {
+        RTC_DCHECK_RUN_ON(&callback_checker_);
+        // If this task has not been cancelled, we did not get any new frames
+        // while waiting. Continue with frame delivery.
+        std::unique_ptr<EncodedFrame> frame;
+        NextFrameCallback frame_handler;
+        {
+          MutexLock lock(&mutex_);
+          if (!frames_to_decode_.empty()) {
+            // We have frames, deliver!
+            frame = GetNextFrame();
+            timing_->SetLastDecodeScheduledTimestamp(clock_->CurrentTime());
+          } else if (clock_->TimeInMilliseconds() < latest_return_time_ms_) {
+            // If there's no frames to decode and there is still time left, it
+            // means that the frame buffer was cleared between creation and
+            // execution of this task. Continue waiting for the remaining time.
+            int64_t wait_ms = FindNextFrame(clock_->CurrentTime());
+            return TimeDelta::Millis(wait_ms);
+          }
+          frame_handler = std::move(frame_handler_);
+          CancelCallback();
+        }
+        // Deliver frame, if any. Otherwise signal timeout.
+        frame_handler(std::move(frame));
+        return TimeDelta::Zero();  // Ignored.
+      },
+      TaskQueueBase::DelayPrecision::kHigh);
+}
+
+int64_t FrameBuffer::FindNextFrame(Timestamp now) {
+  int64_t wait_ms = latest_return_time_ms_ - now.ms();
+  frames_to_decode_.clear();
+
+  // `last_continuous_frame_` may be empty below, but nullopt is smaller
+  // than everything else and loop will immediately terminate as expected.
+  for (auto frame_it = frames_.begin();
+       frame_it != frames_.end() && frame_it->first <= last_continuous_frame_;
+       ++frame_it) {
+    if (!frame_it->second.continuous ||
+        frame_it->second.num_missing_decodable > 0) {
+      continue;
+    }
+
+    EncodedFrame* frame = frame_it->second.frame.get();
+
+    if (keyframe_required_ && !frame->is_keyframe())
+      continue;
+
+    auto last_decoded_frame_timestamp =
+        decoded_frames_history_.GetLastDecodedFrameTimestamp();
+
+    // TODO(https://bugs.webrtc.org/9974): consider removing this check
+    // as it may make a stream undecodable after a very long delay between
+    // frames.
+    if (last_decoded_frame_timestamp &&
+        AheadOf(*last_decoded_frame_timestamp, frame->Timestamp())) {
+      continue;
+    }
+
+    // Gather all remaining frames for the same superframe.
+    std::vector<FrameMap::iterator> current_superframe;
+    current_superframe.push_back(frame_it);
+    bool last_layer_completed = frame_it->second.frame->is_last_spatial_layer;
+    FrameMap::iterator next_frame_it = frame_it;
+    while (!last_layer_completed) {
+      ++next_frame_it;
+
+      if (next_frame_it == frames_.end() || !next_frame_it->second.frame) {
+        break;
+      }
+
+      if (next_frame_it->second.frame->Timestamp() != frame->Timestamp() ||
+          !next_frame_it->second.continuous) {
+        break;
+      }
+
+      if (next_frame_it->second.num_missing_decodable > 0) {
+        bool has_inter_layer_dependency = false;
+        for (size_t i = 0; i < EncodedFrame::kMaxFrameReferences &&
+                           i < next_frame_it->second.frame->num_references;
+             ++i) {
+          if (next_frame_it->second.frame->references[i] >= frame_it->first) {
+            has_inter_layer_dependency = true;
+            break;
+          }
+        }
+
+        // If the frame has an undecoded dependency that is not within the same
+        // temporal unit then this frame is not yet ready to be decoded. If it
+        // is within the same temporal unit then the not yet decoded dependency
+        // is just a lower spatial frame, which is ok.
+        if (!has_inter_layer_dependency ||
+            next_frame_it->second.num_missing_decodable > 1) {
+          break;
+        }
+      }
+
+      current_superframe.push_back(next_frame_it);
+      last_layer_completed = next_frame_it->second.frame->is_last_spatial_layer;
+    }
+    // Check if the current superframe is complete.
+    // TODO(bugs.webrtc.org/10064): consider returning all available to
+    // decode frames even if the superframe is not complete yet.
+    if (!last_layer_completed) {
+      continue;
+    }
+
+    frames_to_decode_ = std::move(current_superframe);
+
+    absl::optional<Timestamp> render_time = frame->RenderTimestamp();
+    if (!render_time) {
+      render_time = timing_->RenderTime(frame->Timestamp(), now);
+      frame->SetRenderTime(render_time->ms());
+    }
+    bool too_many_frames_queued =
+        frames_.size() > zero_playout_delay_max_decode_queue_size_ ? true
+                                                                   : false;
+    wait_ms =
+        timing_->MaxWaitingTime(*render_time, now, too_many_frames_queued).ms();
+
+    // This will cause the frame buffer to prefer high framerate rather
+    // than high resolution in the case of the decoder not decoding fast
+    // enough and the stream has multiple spatial and temporal layers.
+    // For multiple temporal layers it may cause non-base layer frames to be
+    // skipped if they are late.
+    if (wait_ms < -kMaxAllowedFrameDelayMs)
+      continue;
+
+    break;
+  }
+  wait_ms = std::min<int64_t>(wait_ms, latest_return_time_ms_ - now.ms());
+  wait_ms = std::max<int64_t>(wait_ms, 0);
+  return wait_ms;
+}
+
+std::unique_ptr<EncodedFrame> FrameBuffer::GetNextFrame() {
+  RTC_DCHECK_RUN_ON(&callback_checker_);
+  Timestamp now = clock_->CurrentTime();
+  // TODO(ilnik): remove `frames_out` use frames_to_decode_ directly.
+  std::vector<std::unique_ptr<EncodedFrame>> frames_out;
+
+  RTC_DCHECK(!frames_to_decode_.empty());
+  bool superframe_delayed_by_retransmission = false;
+  DataSize superframe_size = DataSize::Zero();
+  const EncodedFrame& first_frame = *frames_to_decode_[0]->second.frame;
+  absl::optional<Timestamp> render_time = first_frame.RenderTimestamp();
+  int64_t receive_time_ms = first_frame.ReceivedTime();
+  // Gracefully handle bad RTP timestamps and render time issues.
+  if (!render_time || FrameHasBadRenderTiming(*render_time, now) ||
+      TargetVideoDelayIsTooLarge(timing_->TargetVideoDelay())) {
+    RTC_LOG(LS_WARNING) << "Resetting jitter estimator and timing module due "
+                           "to bad render timing for rtp_timestamp="
+                        << first_frame.Timestamp();
+    jitter_estimator_.Reset();
+    timing_->Reset();
+    render_time = timing_->RenderTime(first_frame.Timestamp(), now);
+  }
+
+  for (FrameMap::iterator& frame_it : frames_to_decode_) {
+    RTC_DCHECK(frame_it != frames_.end());
+    std::unique_ptr<EncodedFrame> frame = std::move(frame_it->second.frame);
+
+    frame->SetRenderTime(render_time->ms());
+
+    superframe_delayed_by_retransmission |= frame->delayed_by_retransmission();
+    receive_time_ms = std::max(receive_time_ms, frame->ReceivedTime());
+    superframe_size += DataSize::Bytes(frame->size());
+
+    PropagateDecodability(frame_it->second);
+    decoded_frames_history_.InsertDecoded(frame_it->first, frame->Timestamp());
+
+    frames_.erase(frames_.begin(), ++frame_it);
+
+    frames_out.emplace_back(std::move(frame));
+  }
+
+  if (!superframe_delayed_by_retransmission) {
+    auto frame_delay = inter_frame_delay_.CalculateDelay(
+        first_frame.Timestamp(), Timestamp::Millis(receive_time_ms));
+
+    if (frame_delay) {
+      jitter_estimator_.UpdateEstimate(*frame_delay, superframe_size);
+    }
+
+    float rtt_mult = protection_mode_ == kProtectionNackFEC ? 0.0 : 1.0;
+    absl::optional<TimeDelta> rtt_mult_add_cap_ms = absl::nullopt;
+    if (rtt_mult_settings_.has_value()) {
+      rtt_mult = rtt_mult_settings_->rtt_mult_setting;
+      rtt_mult_add_cap_ms =
+          TimeDelta::Millis(rtt_mult_settings_->rtt_mult_add_cap_ms);
+    }
+    timing_->SetJitterDelay(
+        jitter_estimator_.GetJitterEstimate(rtt_mult, rtt_mult_add_cap_ms));
+    timing_->UpdateCurrentDelay(*render_time, now);
+  } else {
+    if (RttMultExperiment::RttMultEnabled())
+      jitter_estimator_.FrameNacked();
+  }
+
+  if (frames_out.size() == 1) {
+    return std::move(frames_out[0]);
+  } else {
+    return CombineAndDeleteFrames(std::move(frames_out));
+  }
+}
+
+void FrameBuffer::SetProtectionMode(VCMVideoProtection mode) {
+  TRACE_EVENT0("webrtc", "FrameBuffer::SetProtectionMode");
+  MutexLock lock(&mutex_);
+  protection_mode_ = mode;
+}
+
+void FrameBuffer::Stop() {
+  TRACE_EVENT0("webrtc", "FrameBuffer::Stop");
+  MutexLock lock(&mutex_);
+  if (stopped_)
+    return;
+  stopped_ = true;
+
+  CancelCallback();
+}
+
+void FrameBuffer::Clear() {
+  MutexLock lock(&mutex_);
+  ClearFramesAndHistory();
+}
+
+int FrameBuffer::Size() {
+  MutexLock lock(&mutex_);
+  return frames_.size();
+}
+
+void FrameBuffer::UpdateRtt(int64_t rtt_ms) {
+  MutexLock lock(&mutex_);
+  jitter_estimator_.UpdateRtt(TimeDelta::Millis(rtt_ms));
+}
+
+bool FrameBuffer::ValidReferences(const EncodedFrame& frame) const {
+  for (size_t i = 0; i < frame.num_references; ++i) {
+    if (frame.references[i] >= frame.Id())
+      return false;
+
+    for (size_t j = i + 1; j < frame.num_references; ++j) {
+      if (frame.references[i] == frame.references[j])
+        return false;
+    }
+  }
+
+  return true;
+}
+
+void FrameBuffer::CancelCallback() {
+  // Called from the callback queue or from within Stop().
+  frame_handler_ = {};
+  callback_task_.Stop();
+  callback_queue_ = nullptr;
+  callback_checker_.Detach();
+}
+
+int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
+  TRACE_EVENT0("webrtc", "FrameBuffer::InsertFrame");
+  RTC_DCHECK(frame);
+
+  MutexLock lock(&mutex_);
+
+  const auto& pis = frame->PacketInfos();
+  int64_t last_continuous_frame_id = last_continuous_frame_.value_or(-1);
+
+  if (!ValidReferences(*frame)) {
+    TRACE_EVENT2("webrtc",
+                 "FrameBuffer::InsertFrame Frame dropped (Invalid references)",
+                 "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+                 frame->Id());
+    RTC_LOG(LS_WARNING) << "Frame " << frame->Id()
+                        << " has invalid frame references, dropping frame.";
+    return last_continuous_frame_id;
+  }
+
+  if (frames_.size() >= kMaxFramesBuffered) {
+    if (frame->is_keyframe()) {
+      TRACE_EVENT2("webrtc",
+                   "FrameBuffer::InsertFrame Frames dropped (KF + Full buffer)",
+                   "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+                   frame->Id());
+      RTC_LOG(LS_WARNING) << "Inserting keyframe " << frame->Id()
+                          << " but buffer is full, clearing"
+                             " buffer and inserting the frame.";
+      ClearFramesAndHistory();
+    } else {
+      TRACE_EVENT2("webrtc",
+                   "FrameBuffer::InsertFrame Frame dropped (Full buffer)",
+                   "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+                   frame->Id());
+      RTC_LOG(LS_WARNING) << "Frame " << frame->Id()
+                          << " could not be inserted due to the frame "
+                             "buffer being full, dropping frame.";
+      return last_continuous_frame_id;
+    }
+  }
+
+  auto last_decoded_frame = decoded_frames_history_.GetLastDecodedFrameId();
+  auto last_decoded_frame_timestamp =
+      decoded_frames_history_.GetLastDecodedFrameTimestamp();
+  if (last_decoded_frame && frame->Id() <= *last_decoded_frame) {
+    if (AheadOf(frame->Timestamp(), *last_decoded_frame_timestamp) &&
+        frame->is_keyframe()) {
+      // If this frame has a newer timestamp but an earlier frame id then we
+      // assume there has been a jump in the frame id due to some encoder
+      // reconfiguration or some other reason. Even though this is not according
+      // to spec we can still continue to decode from this frame if it is a
+      // keyframe.
+      TRACE_EVENT2("webrtc",
+                   "FrameBuffer::InsertFrame Frames dropped (OOO + PicId jump)",
+                   "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+                   frame->Id());
+      RTC_LOG(LS_WARNING)
+          << "A jump in frame id was detected, clearing buffer.";
+      ClearFramesAndHistory();
+      last_continuous_frame_id = -1;
+    } else {
+      TRACE_EVENT2("webrtc",
+                   "FrameBuffer::InsertFrame Frame dropped (Out of order)",
+                   "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+                   frame->Id());
+      RTC_LOG(LS_WARNING) << "Frame " << frame->Id() << " inserted after frame "
+                          << *last_decoded_frame
+                          << " was handed off for decoding, dropping frame.";
+      return last_continuous_frame_id;
+    }
+  }
+
+  // Test if inserting this frame would cause the order of the frames to become
+  // ambiguous (covering more than half the interval of 2^16). This can happen
+  // when the frame id make large jumps mid stream.
+  if (!frames_.empty() && frame->Id() < frames_.begin()->first &&
+      frames_.rbegin()->first < frame->Id()) {
+    TRACE_EVENT2("webrtc",
+                 "FrameBuffer::InsertFrame Frames dropped (PicId big-jump)",
+                 "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+                 frame->Id());
+    RTC_LOG(LS_WARNING) << "A jump in frame id was detected, clearing buffer.";
+    ClearFramesAndHistory();
+    last_continuous_frame_id = -1;
+  }
+
+  auto info = frames_.emplace(frame->Id(), FrameInfo()).first;
+
+  if (info->second.frame) {
+    return last_continuous_frame_id;
+  }
+
+  if (!UpdateFrameInfoWithIncomingFrame(*frame, info))
+    return last_continuous_frame_id;
+
+  // If ReceiveTime is negative then it is not a valid timestamp.
+  if (!frame->delayed_by_retransmission() && frame->ReceivedTime() >= 0)
+    timing_->IncomingTimestamp(frame->Timestamp(),
+                               Timestamp::Millis(frame->ReceivedTime()));
+
+  // It can happen that a frame will be reported as fully received even if a
+  // lower spatial layer frame is missing.
+  info->second.frame = std::move(frame);
+
+  if (info->second.num_missing_continuous == 0) {
+    info->second.continuous = true;
+    PropagateContinuity(info);
+    last_continuous_frame_id = *last_continuous_frame_;
+
+    // Since we now have new continuous frames there might be a better frame
+    // to return from NextFrame.
+    if (callback_queue_) {
+      callback_queue_->PostTask([this] {
+        MutexLock lock(&mutex_);
+        if (!callback_task_.Running())
+          return;
+        RTC_CHECK(frame_handler_);
+        callback_task_.Stop();
+        StartWaitForNextFrameOnQueue();
+      });
+    }
+  }
+
+  return last_continuous_frame_id;
+}
+
+void FrameBuffer::PropagateContinuity(FrameMap::iterator start) {
+  TRACE_EVENT0("webrtc", "FrameBuffer::PropagateContinuity");
+  RTC_DCHECK(start->second.continuous);
+
+  std::queue<FrameMap::iterator> continuous_frames;
+  continuous_frames.push(start);
+
+  // A simple BFS to traverse continuous frames.
+  while (!continuous_frames.empty()) {
+    auto frame = continuous_frames.front();
+    continuous_frames.pop();
+
+    if (!last_continuous_frame_ || *last_continuous_frame_ < frame->first) {
+      last_continuous_frame_ = frame->first;
+    }
+
+    // Loop through all dependent frames, and if that frame no longer has
+    // any unfulfilled dependencies then that frame is continuous as well.
+    for (size_t d = 0; d < frame->second.dependent_frames.size(); ++d) {
+      auto frame_ref = frames_.find(frame->second.dependent_frames[d]);
+      RTC_DCHECK(frame_ref != frames_.end());
+
+      // TODO(philipel): Look into why we've seen this happen.
+      if (frame_ref != frames_.end()) {
+        --frame_ref->second.num_missing_continuous;
+        if (frame_ref->second.num_missing_continuous == 0) {
+          frame_ref->second.continuous = true;
+          continuous_frames.push(frame_ref);
+        }
+      }
+    }
+  }
+}
+
+void FrameBuffer::PropagateDecodability(const FrameInfo& info) {
+  TRACE_EVENT0("webrtc", "FrameBuffer::PropagateDecodability");
+  for (size_t d = 0; d < info.dependent_frames.size(); ++d) {
+    auto ref_info = frames_.find(info.dependent_frames[d]);
+    RTC_DCHECK(ref_info != frames_.end());
+    // TODO(philipel): Look into why we've seen this happen.
+    if (ref_info != frames_.end()) {
+      RTC_DCHECK_GT(ref_info->second.num_missing_decodable, 0U);
+      --ref_info->second.num_missing_decodable;
+    }
+  }
+}
+
+bool FrameBuffer::UpdateFrameInfoWithIncomingFrame(const EncodedFrame& frame,
+                                                   FrameMap::iterator info) {
+  TRACE_EVENT0("webrtc", "FrameBuffer::UpdateFrameInfoWithIncomingFrame");
+  auto last_decoded_frame = decoded_frames_history_.GetLastDecodedFrameId();
+  RTC_DCHECK(!last_decoded_frame || *last_decoded_frame < info->first);
+
+  // In this function we determine how many missing dependencies this `frame`
+  // has to become continuous/decodable. If a frame that this `frame` depend
+  // on has already been decoded then we can ignore that dependency since it has
+  // already been fulfilled.
+  //
+  // For all other frames we will register a backwards reference to this `frame`
+  // so that `num_missing_continuous` and `num_missing_decodable` can be
+  // decremented as frames become continuous/are decoded.
+  struct Dependency {
+    int64_t frame_id;
+    bool continuous;
+  };
+  std::vector<Dependency> not_yet_fulfilled_dependencies;
+
+  // Find all dependencies that have not yet been fulfilled.
+  for (size_t i = 0; i < frame.num_references; ++i) {
+    // Does `frame` depend on a frame earlier than the last decoded one?
+    if (last_decoded_frame && frame.references[i] <= *last_decoded_frame) {
+      // Was that frame decoded? If not, this `frame` will never become
+      // decodable.
+      if (!decoded_frames_history_.WasDecoded(frame.references[i])) {
+        int64_t now_ms = clock_->TimeInMilliseconds();
+        if (last_log_non_decoded_ms_ + kLogNonDecodedIntervalMs < now_ms) {
+          RTC_LOG(LS_WARNING)
+              << "Frame " << frame.Id()
+              << " depends on a non-decoded frame more previous than the last "
+                 "decoded frame, dropping frame.";
+          last_log_non_decoded_ms_ = now_ms;
+        }
+        return false;
+      }
+    } else {
+      auto ref_info = frames_.find(frame.references[i]);
+      bool ref_continuous =
+          ref_info != frames_.end() && ref_info->second.continuous;
+      not_yet_fulfilled_dependencies.push_back(
+          {frame.references[i], ref_continuous});
+    }
+  }
+
+  info->second.num_missing_continuous = not_yet_fulfilled_dependencies.size();
+  info->second.num_missing_decodable = not_yet_fulfilled_dependencies.size();
+
+  for (const Dependency& dep : not_yet_fulfilled_dependencies) {
+    if (dep.continuous)
+      --info->second.num_missing_continuous;
+
+    frames_[dep.frame_id].dependent_frames.push_back(frame.Id());
+  }
+
+  return true;
+}
+
+void FrameBuffer::ClearFramesAndHistory() {
+  TRACE_EVENT0("webrtc", "FrameBuffer::ClearFramesAndHistory");
+  frames_.clear();
+  last_continuous_frame_.reset();
+  frames_to_decode_.clear();
+  decoded_frames_history_.Clear();
+}
+
+// TODO(philipel): Avoid the concatenation of frames here, by replacing
+// NextFrame and GetNextFrame with methods returning multiple frames.
+std::unique_ptr<EncodedFrame> FrameBuffer::CombineAndDeleteFrames(
+    std::vector<std::unique_ptr<EncodedFrame>> frames) const {
+  RTC_DCHECK(!frames.empty());
+  absl::InlinedVector<std::unique_ptr<EncodedFrame>, 4> inlined;
+  for (auto& frame : frames) {
+    inlined.push_back(std::move(frame));
+  }
+  return webrtc::CombineAndDeleteFrames(std::move(inlined));
+}
+
+FrameBuffer::FrameInfo::FrameInfo() = default;
+FrameBuffer::FrameInfo::FrameInfo(FrameInfo&&) = default;
+FrameBuffer::FrameInfo::~FrameInfo() = default;
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/frame_buffer2.h b/third_party/libwebrtc/modules/video_coding/frame_buffer2.h
new file mode 100644
index 0000000000..1383c40ae3
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_buffer2.h
@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_FRAME_BUFFER2_H_
+#define MODULES_VIDEO_CODING_FRAME_BUFFER2_H_
+
+#include <array>
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/video/encoded_frame.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "modules/video_coding/timing/inter_frame_delay.h"
+#include "modules/video_coding/timing/jitter_estimator.h"
+#include "modules/video_coding/utility/decoded_frames_history.h"
+#include "rtc_base/event.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/rtt_mult_experiment.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class Clock;
+class VCMReceiveStatisticsCallback;
+class JitterEstimator;
+class VCMTiming;
+
+namespace video_coding {
+
+class FrameBuffer {
+ public:
+  FrameBuffer(Clock* clock,
+              VCMTiming* timing,
+              const FieldTrialsView& field_trials);
+
+  FrameBuffer() = delete;
+  FrameBuffer(const FrameBuffer&) = delete;
+  FrameBuffer& operator=(const FrameBuffer&) = delete;
+
+  virtual ~FrameBuffer();
+
+  // Insert a frame into the frame buffer. Returns the picture id
+  // of the last continuous frame or -1 if there is no continuous frame.
+  int64_t InsertFrame(std::unique_ptr<EncodedFrame> frame);
+
+  using NextFrameCallback = std::function<void(std::unique_ptr<EncodedFrame>)>;
+  // Get the next frame for decoding. `handler` is invoked with the next frame
+  // or with nullptr if no frame is ready for decoding after `max_wait_time_ms`.
+  void NextFrame(int64_t max_wait_time_ms,
+                 bool keyframe_required,
+                 TaskQueueBase* callback_queue,
+                 NextFrameCallback handler);
+
+  // Tells the FrameBuffer which protection mode that is in use. Affects
+  // the frame timing.
+  // TODO(philipel): Remove this when new timing calculations has been
+  //                 implemented.
+  void SetProtectionMode(VCMVideoProtection mode);
+
+  // Stop the frame buffer, causing any sleeping thread in NextFrame to
+  // return immediately.
+  void Stop();
+
+  // Updates the RTT for jitter buffer estimation.
+  void UpdateRtt(int64_t rtt_ms);
+
+  // Clears the FrameBuffer, removing all the buffered frames.
+  void Clear();
+
+  int Size();
+
+ private:
+  struct FrameInfo {
+    FrameInfo();
+    FrameInfo(FrameInfo&&);
+    ~FrameInfo();
+
+    // Which other frames that have direct unfulfilled dependencies
+    // on this frame.
+    absl::InlinedVector<int64_t, 8> dependent_frames;
+
+    // A frame is continiuous if it has all its referenced/indirectly
+    // referenced frames.
+    //
+    // How many unfulfilled frames this frame have until it becomes continuous.
+    size_t num_missing_continuous = 0;
+
+    // A frame is decodable if all its referenced frames have been decoded.
+    //
+    // How many unfulfilled frames this frame have until it becomes decodable.
+    size_t num_missing_decodable = 0;
+
+    // If this frame is continuous or not.
+    bool continuous = false;
+
+    // The actual EncodedFrame.
+    std::unique_ptr<EncodedFrame> frame;
+  };
+
+  using FrameMap = std::map<int64_t, FrameInfo>;
+
+  // Check that the references of `frame` are valid.
+  bool ValidReferences(const EncodedFrame& frame) const;
+
+  int64_t FindNextFrame(Timestamp now) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  std::unique_ptr<EncodedFrame> GetNextFrame()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  void StartWaitForNextFrameOnQueue() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  void CancelCallback() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Update all directly dependent and indirectly dependent frames and mark
+  // them as continuous if all their references has been fulfilled.
+  void PropagateContinuity(FrameMap::iterator start)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Marks the frame as decoded and updates all directly dependent frames.
+  void PropagateDecodability(const FrameInfo& info)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Update the corresponding FrameInfo of `frame` and all FrameInfos that
+  // `frame` references.
+  // Return false if `frame` will never be decodable, true otherwise.
+  bool UpdateFrameInfoWithIncomingFrame(const EncodedFrame& frame,
+                                        FrameMap::iterator info)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  void ClearFramesAndHistory() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // The cleaner solution would be to have the NextFrame function return a
+  // vector of frames, but until the decoding pipeline can support decoding
+  // multiple frames at the same time we combine all frames to one frame and
+  // return it. See bugs.webrtc.org/10064
+  std::unique_ptr<EncodedFrame> CombineAndDeleteFrames(
+      std::vector<std::unique_ptr<EncodedFrame>> frames) const;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker construction_checker_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker callback_checker_;
+
+  // Stores only undecoded frames.
+  FrameMap frames_ RTC_GUARDED_BY(mutex_);
+  DecodedFramesHistory decoded_frames_history_ RTC_GUARDED_BY(mutex_);
+
+  Mutex mutex_;
+  Clock* const clock_;
+
+  TaskQueueBase* callback_queue_ RTC_GUARDED_BY(mutex_);
+  RepeatingTaskHandle callback_task_ RTC_GUARDED_BY(mutex_);
+  NextFrameCallback frame_handler_ RTC_GUARDED_BY(mutex_);
+  int64_t latest_return_time_ms_ RTC_GUARDED_BY(mutex_);
+  bool keyframe_required_ RTC_GUARDED_BY(mutex_);
+
+  JitterEstimator jitter_estimator_ RTC_GUARDED_BY(mutex_);
+  VCMTiming* const timing_ RTC_GUARDED_BY(mutex_);
+  InterFrameDelay inter_frame_delay_ RTC_GUARDED_BY(mutex_);
+  absl::optional<int64_t> last_continuous_frame_ RTC_GUARDED_BY(mutex_);
+  std::vector<FrameMap::iterator> frames_to_decode_ RTC_GUARDED_BY(mutex_);
+  bool stopped_ RTC_GUARDED_BY(mutex_);
+  VCMVideoProtection protection_mode_ RTC_GUARDED_BY(mutex_);
+  int64_t last_log_non_decoded_ms_ RTC_GUARDED_BY(mutex_);
+
+  // rtt_mult experiment settings.
+  const absl::optional<RttMultExperiment::Settings> rtt_mult_settings_;
+
+  // Maximum number of frames in the decode queue to allow pacing. If the
+  // queue grows beyond the max limit, pacing will be disabled and frames will
+  // be pushed to the decoder as soon as possible. This only has an effect
+  // when the low-latency rendering path is active, which is indicated by
+  // the frame's render time == 0.
+  FieldTrialParameter<unsigned> zero_playout_delay_max_decode_queue_size_;
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_FRAME_BUFFER2_H_
diff --git a/third_party/libwebrtc/modules/video_coding/frame_buffer2_unittest.cc b/third_party/libwebrtc/modules/video_coding/frame_buffer2_unittest.cc
new file mode 100644
index 0000000000..0fabd9b496
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_buffer2_unittest.cc
@@ -0,0 +1,665 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/frame_buffer2.h"
+
+#include <algorithm>
+#include <cstring>
+#include <limits>
+#include <memory>
+#include <vector>
+
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/timing/jitter_estimator.h"
+#include "modules/video_coding/timing/timing.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/random.h"
+#include "system_wrappers/include/clock.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+using ::testing::_;
+using ::testing::IsEmpty;
+using ::testing::Return;
+using ::testing::SizeIs;
+
+namespace webrtc {
+namespace video_coding {
+
+class VCMTimingFake : public VCMTiming {
+ public:
+  explicit VCMTimingFake(Clock* clock, const FieldTrialsView& field_trials)
+      : VCMTiming(clock, field_trials) {}
+
+  Timestamp RenderTime(uint32_t frame_timestamp, Timestamp now) const override {
+    if (last_render_time_.IsMinusInfinity()) {
+      last_render_time_ = now + kDelay;
+      last_timestamp_ = frame_timestamp;
+    }
+
+    auto diff = MinDiff(frame_timestamp, last_timestamp_);
+    auto timeDiff = TimeDelta::Millis(diff / 90);
+    if (AheadOf(frame_timestamp, last_timestamp_))
+      last_render_time_ += timeDiff;
+    else
+      last_render_time_ -= timeDiff;
+
+    last_timestamp_ = frame_timestamp;
+    return last_render_time_;
+  }
+
+  TimeDelta MaxWaitingTime(Timestamp render_time,
+                           Timestamp now,
+                           bool too_many_frames_queued) const override {
+    return render_time - now - kDecodeTime;
+  }
+
+  TimeDelta GetCurrentJitter() {
+    return VCMTiming::GetTimings().jitter_buffer_delay;
+  }
+
+ private:
+  static constexpr TimeDelta kDelay = TimeDelta::Millis(50);
+  const TimeDelta kDecodeTime = kDelay / 2;
+  mutable uint32_t last_timestamp_ = 0;
+  mutable Timestamp last_render_time_ = Timestamp::MinusInfinity();
+};
+
+class FrameObjectFake : public EncodedFrame {
+ public:
+  int64_t ReceivedTime() const override { return 0; }
+
+  int64_t RenderTime() const override { return _renderTimeMs; }
+
+  bool delayed_by_retransmission() const override {
+    return delayed_by_retransmission_;
+  }
+  void set_delayed_by_retransmission(bool delayed) {
+    delayed_by_retransmission_ = delayed;
+  }
+
+ private:
+  bool delayed_by_retransmission_ = false;
+};
+
+class VCMReceiveStatisticsCallbackMock : public VCMReceiveStatisticsCallback {
+ public:
+  MOCK_METHOD(void,
+              OnCompleteFrame,
+              (bool is_keyframe,
+               size_t size_bytes,
+               VideoContentType content_type),
+              (override));
+  MOCK_METHOD(void, OnDroppedFrames, (uint32_t frames_dropped), (override));
+  MOCK_METHOD(void,
+              OnFrameBufferTimingsUpdated,
+              (int max_decode,
+               int current_delay,
+               int target_delay,
+               int jitter_buffer,
+               int min_playout_delay,
+               int render_delay),
+              (override));
+  MOCK_METHOD(void,
+              OnTimingFrameInfoUpdated,
+              (const TimingFrameInfo& info),
+              (override));
+};
+
+class TestFrameBuffer2 : public ::testing::Test {
+ protected:
+  static constexpr int kMaxReferences = 5;
+  static constexpr int kFps1 = 1000;
+  static constexpr int kFps10 = kFps1 / 10;
+  static constexpr int kFps20 = kFps1 / 20;
+  static constexpr size_t kFrameSize = 10;
+
+  TestFrameBuffer2()
+      : time_controller_(Timestamp::Seconds(0)),
+        time_task_queue_(
+            time_controller_.GetTaskQueueFactory()->CreateTaskQueue(
+                "extract queue",
+                TaskQueueFactory::Priority::NORMAL)),
+        timing_(time_controller_.GetClock(), field_trials_),
+        buffer_(new FrameBuffer(time_controller_.GetClock(),
+                                &timing_,
+                                field_trials_)),
+        rand_(0x34678213) {}
+
+  template <typename... T>
+  std::unique_ptr<FrameObjectFake> CreateFrame(uint16_t picture_id,
+                                               uint8_t spatial_layer,
+                                               int64_t ts_ms,
+                                               bool last_spatial_layer,
+                                               size_t frame_size_bytes,
+                                               T... refs) {
+    static_assert(sizeof...(refs) <= kMaxReferences,
+                  "To many references specified for EncodedFrame.");
+    std::array<uint16_t, sizeof...(refs)> references = {
+        {rtc::checked_cast<uint16_t>(refs)...}};
+
+    auto frame = std::make_unique<FrameObjectFake>();
+    frame->SetId(picture_id);
+    frame->SetSpatialIndex(spatial_layer);
+    frame->SetTimestamp(ts_ms * 90);
+    frame->num_references = references.size();
+    frame->is_last_spatial_layer = last_spatial_layer;
+    // Add some data to buffer.
+    frame->SetEncodedData(EncodedImageBuffer::Create(frame_size_bytes));
+    for (size_t r = 0; r < references.size(); ++r)
+      frame->references[r] = references[r];
+    return frame;
+  }
+
+  template <typename... T>
+  int InsertFrame(uint16_t picture_id,
+                  uint8_t spatial_layer,
+                  int64_t ts_ms,
+                  bool last_spatial_layer,
+                  size_t frame_size_bytes,
+                  T... refs) {
+    return buffer_->InsertFrame(CreateFrame(picture_id, spatial_layer, ts_ms,
+                                            last_spatial_layer,
+                                            frame_size_bytes, refs...));
+  }
+
+  int InsertNackedFrame(uint16_t picture_id, int64_t ts_ms) {
+    std::unique_ptr<FrameObjectFake> frame =
+        CreateFrame(picture_id, 0, ts_ms, true, kFrameSize);
+    frame->set_delayed_by_retransmission(true);
+    return buffer_->InsertFrame(std::move(frame));
+  }
+
+  void ExtractFrame(int64_t max_wait_time = 0, bool keyframe_required = false) {
+    time_task_queue_->PostTask([this, max_wait_time, keyframe_required]() {
+      buffer_->NextFrame(max_wait_time, keyframe_required,
+                         time_task_queue_.get(),
+                         [this](std::unique_ptr<EncodedFrame> frame) {
+                           frames_.emplace_back(std::move(frame));
+                         });
+    });
+    if (max_wait_time == 0) {
+      time_controller_.AdvanceTime(TimeDelta::Zero());
+    }
+  }
+
+  void CheckFrame(size_t index, int picture_id, int spatial_layer) {
+    ASSERT_LT(index, frames_.size());
+    ASSERT_TRUE(frames_[index]);
+    ASSERT_EQ(picture_id, frames_[index]->Id());
+    ASSERT_EQ(spatial_layer, frames_[index]->SpatialIndex().value_or(0));
+  }
+
+  void CheckFrameSize(size_t index, size_t size) {
+    ASSERT_LT(index, frames_.size());
+    ASSERT_TRUE(frames_[index]);
+    ASSERT_EQ(frames_[index]->size(), size);
+  }
+
+  void CheckNoFrame(size_t index) {
+    ASSERT_LT(index, frames_.size());
+    ASSERT_FALSE(frames_[index]);
+  }
+
+  uint32_t Rand() { return rand_.Rand<uint32_t>(); }
+
+  test::ScopedKeyValueConfig field_trials_;
+  webrtc::GlobalSimulatedTimeController time_controller_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> time_task_queue_;
+  VCMTimingFake timing_;
+  std::unique_ptr<FrameBuffer> buffer_;
+  std::vector<std::unique_ptr<EncodedFrame>> frames_;
+  Random rand_;
+};
+
+// From https://en.cppreference.com/w/cpp/language/static: "If ... a constexpr
+// static data member (since C++11) is odr-used, a definition at namespace scope
+// is still required... This definition is deprecated for constexpr data members
+// since C++17."
+// kFrameSize is odr-used since it is passed by reference to EXPECT_EQ().
+#if __cplusplus < 201703L
+constexpr size_t TestFrameBuffer2::kFrameSize;
+#endif
+
+TEST_F(TestFrameBuffer2, WaitForFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  ExtractFrame(50);
+  InsertFrame(pid, 0, ts, true, kFrameSize);
+  time_controller_.AdvanceTime(TimeDelta::Millis(50));
+  CheckFrame(0, pid, 0);
+}
+
+TEST_F(TestFrameBuffer2, ClearWhileWaitingForFrame) {
+  const uint16_t pid = Rand();
+
+  // Insert a frame and wait for it for max 100ms.
+  InsertFrame(pid, 0, 25, true, kFrameSize);
+  ExtractFrame(100);
+  // After 10ms, clear the buffer.
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  buffer_->Clear();
+  // Confirm that the frame was not sent for rendering.
+  time_controller_.AdvanceTime(TimeDelta::Millis(15));
+  EXPECT_THAT(frames_, IsEmpty());
+
+  // We are still waiting for a frame, since 100ms has not passed. Insert a new
+  // frame. This new frame should be the one that is returned as the old frame
+  // was cleared.
+  const uint16_t new_pid = pid + 1;
+  InsertFrame(new_pid, 0, 50, true, kFrameSize);
+  time_controller_.AdvanceTime(TimeDelta::Millis(25));
+  ASSERT_THAT(frames_, SizeIs(1));
+  CheckFrame(0, new_pid, 0);
+}
+
+TEST_F(TestFrameBuffer2, OneSuperFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, false, kFrameSize);
+  InsertFrame(pid + 1, 1, ts, true, kFrameSize);
+  ExtractFrame();
+
+  CheckFrame(0, pid, 1);
+}
+
+TEST_F(TestFrameBuffer2, ZeroPlayoutDelay) {
+  test::ScopedKeyValueConfig field_trials;
+  VCMTiming timing(time_controller_.GetClock(), field_trials);
+  buffer_ = std::make_unique<FrameBuffer>(time_controller_.GetClock(), &timing,
+                                          field_trials);
+  const VideoPlayoutDelay kPlayoutDelayMs = {0, 0};
+  std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
+  test_frame->SetId(0);
+  test_frame->SetPlayoutDelay(kPlayoutDelayMs);
+  buffer_->InsertFrame(std::move(test_frame));
+  ExtractFrame(0, false);
+  CheckFrame(0, 0, 0);
+  EXPECT_EQ(0, frames_[0]->RenderTimeMs());
+}
+
+// Flaky test, see bugs.webrtc.org/7068.
+TEST_F(TestFrameBuffer2, DISABLED_OneUnorderedSuperFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  ExtractFrame(50);
+  InsertFrame(pid, 1, ts, true, kFrameSize);
+  InsertFrame(pid, 0, ts, false, kFrameSize);
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+
+  CheckFrame(0, pid, 0);
+  CheckFrame(1, pid, 1);
+}
+
+TEST_F(TestFrameBuffer2, DISABLED_OneLayerStreamReordered) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, false, true, kFrameSize);
+  ExtractFrame();
+  CheckFrame(0, pid, 0);
+  for (int i = 1; i < 10; i += 2) {
+    ExtractFrame(50);
+    InsertFrame(pid + i + 1, 0, ts + (i + 1) * kFps10, true, kFrameSize,
+                pid + i);
+    time_controller_.AdvanceTime(TimeDelta::Millis(kFps10));
+    InsertFrame(pid + i, 0, ts + i * kFps10, true, kFrameSize, pid + i - 1);
+    time_controller_.AdvanceTime(TimeDelta::Millis(kFps10));
+    ExtractFrame();
+    CheckFrame(i, pid + i, 0);
+    CheckFrame(i + 1, pid + i + 1, 0);
+  }
+}
+
+TEST_F(TestFrameBuffer2, ExtractFromEmptyBuffer) {
+  ExtractFrame();
+  CheckNoFrame(0);
+}
+
+TEST_F(TestFrameBuffer2, MissingFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, true, kFrameSize);
+  InsertFrame(pid + 2, 0, ts, true, kFrameSize, pid);
+  InsertFrame(pid + 3, 0, ts, true, kFrameSize, pid + 1, pid + 2);
+  ExtractFrame();
+  ExtractFrame();
+  ExtractFrame();
+
+  CheckFrame(0, pid, 0);
+  CheckFrame(1, pid + 2, 0);
+  CheckNoFrame(2);
+}
+
+TEST_F(TestFrameBuffer2, OneLayerStream) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, true, kFrameSize);
+  ExtractFrame();
+  CheckFrame(0, pid, 0);
+  for (int i = 1; i < 10; ++i) {
+    InsertFrame(pid + i, 0, ts + i * kFps10, true, kFrameSize, pid + i - 1);
+    ExtractFrame();
+    time_controller_.AdvanceTime(TimeDelta::Millis(kFps10));
+    CheckFrame(i, pid + i, 0);
+  }
+}
+
+TEST_F(TestFrameBuffer2, DropTemporalLayerSlowDecoder) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, true, kFrameSize);
+  InsertFrame(pid + 1, 0, ts + kFps20, true, kFrameSize, pid);
+  for (int i = 2; i < 10; i += 2) {
+    uint32_t ts_tl0 = ts + i / 2 * kFps10;
+    InsertFrame(pid + i, 0, ts_tl0, true, kFrameSize, pid + i - 2);
+    InsertFrame(pid + i + 1, 0, ts_tl0 + kFps20, true, kFrameSize, pid + i,
+                pid + i - 1);
+  }
+
+  for (int i = 0; i < 10; ++i) {
+    ExtractFrame();
+    time_controller_.AdvanceTime(TimeDelta::Millis(70));
+  }
+
+  CheckFrame(0, pid, 0);
+  CheckFrame(1, pid + 1, 0);
+  CheckFrame(2, pid + 2, 0);
+  CheckFrame(3, pid + 4, 0);
+  CheckFrame(4, pid + 6, 0);
+  CheckFrame(5, pid + 8, 0);
+  CheckNoFrame(6);
+  CheckNoFrame(7);
+  CheckNoFrame(8);
+  CheckNoFrame(9);
+}
+
+TEST_F(TestFrameBuffer2, DropFramesIfSystemIsStalled) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, true, kFrameSize);
+  InsertFrame(pid + 1, 0, ts + 1 * kFps10, true, kFrameSize, pid);
+  InsertFrame(pid + 2, 0, ts + 2 * kFps10, true, kFrameSize, pid + 1);
+  InsertFrame(pid + 3, 0, ts + 3 * kFps10, true, kFrameSize);
+
+  ExtractFrame();
+  // Jump forward in time, simulating the system being stalled for some reason.
+  time_controller_.AdvanceTime(TimeDelta::Millis(3) * kFps10);
+  // Extract one more frame, expect second and third frame to be dropped.
+  ExtractFrame();
+
+  CheckFrame(0, pid + 0, 0);
+  CheckFrame(1, pid + 3, 0);
+}
+
+TEST_F(TestFrameBuffer2, DroppedFramesCountedOnClear) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, true, kFrameSize);
+  for (int i = 1; i < 5; ++i) {
+    InsertFrame(pid + i, 0, ts + i * kFps10, true, kFrameSize, pid + i - 1);
+  }
+
+  // All frames should be dropped when Clear is called.
+  buffer_->Clear();
+}
+
+TEST_F(TestFrameBuffer2, InsertLateFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, true, kFrameSize);
+  ExtractFrame();
+  InsertFrame(pid + 2, 0, ts, true, kFrameSize);
+  ExtractFrame();
+  InsertFrame(pid + 1, 0, ts, true, kFrameSize, pid);
+  ExtractFrame();
+
+  CheckFrame(0, pid, 0);
+  CheckFrame(1, pid + 2, 0);
+  CheckNoFrame(2);
+}
+
+TEST_F(TestFrameBuffer2, ProtectionModeNackFEC) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+  constexpr int64_t kRttMs = 200;
+  buffer_->UpdateRtt(kRttMs);
+
+  // Jitter estimate unaffected by RTT in this protection mode.
+  buffer_->SetProtectionMode(kProtectionNackFEC);
+  InsertNackedFrame(pid, ts);
+  InsertNackedFrame(pid + 1, ts + 100);
+  InsertNackedFrame(pid + 2, ts + 200);
+  InsertFrame(pid + 3, 0, ts + 300, true, kFrameSize);
+  ExtractFrame();
+  ExtractFrame();
+  ExtractFrame();
+  ExtractFrame();
+  ASSERT_EQ(4u, frames_.size());
+  EXPECT_LT(timing_.GetCurrentJitter().ms(), kRttMs);
+}
+
+TEST_F(TestFrameBuffer2, NoContinuousFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  EXPECT_EQ(-1, InsertFrame(pid + 1, 0, ts, true, kFrameSize, pid));
+}
+
+TEST_F(TestFrameBuffer2, LastContinuousFrameSingleLayer) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  EXPECT_EQ(pid, InsertFrame(pid, 0, ts, true, kFrameSize));
+  EXPECT_EQ(pid, InsertFrame(pid + 2, 0, ts, true, kFrameSize, pid + 1));
+  EXPECT_EQ(pid + 2, InsertFrame(pid + 1, 0, ts, true, kFrameSize, pid));
+  EXPECT_EQ(pid + 2, InsertFrame(pid + 4, 0, ts, true, kFrameSize, pid + 3));
+  EXPECT_EQ(pid + 5, InsertFrame(pid + 5, 0, ts, true, kFrameSize));
+}
+
+TEST_F(TestFrameBuffer2, LastContinuousFrameTwoLayers) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  EXPECT_EQ(pid, InsertFrame(pid, 0, ts, false, kFrameSize));
+  EXPECT_EQ(pid + 1, InsertFrame(pid + 1, 1, ts, true, kFrameSize));
+  EXPECT_EQ(pid + 1,
+            InsertFrame(pid + 3, 1, ts, true, kFrameSize, pid + 1, pid + 2));
+  EXPECT_EQ(pid + 1, InsertFrame(pid + 4, 0, ts, false, kFrameSize, pid + 2));
+  EXPECT_EQ(pid + 1,
+            InsertFrame(pid + 5, 1, ts, true, kFrameSize, pid + 3, pid + 4));
+  EXPECT_EQ(pid + 1, InsertFrame(pid + 6, 0, ts, false, kFrameSize, pid + 4));
+  EXPECT_EQ(pid + 6, InsertFrame(pid + 2, 0, ts, false, kFrameSize, pid));
+  EXPECT_EQ(pid + 7,
+            InsertFrame(pid + 7, 1, ts, true, kFrameSize, pid + 5, pid + 6));
+}
+
+TEST_F(TestFrameBuffer2, PictureIdJumpBack) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  EXPECT_EQ(pid, InsertFrame(pid, 0, ts, true, kFrameSize));
+  EXPECT_EQ(pid + 1, InsertFrame(pid + 1, 0, ts + 1, true, kFrameSize, pid));
+  ExtractFrame();
+  CheckFrame(0, pid, 0);
+
+  // Jump back in pid but increase ts.
+  EXPECT_EQ(pid - 1, InsertFrame(pid - 1, 0, ts + 2, true, kFrameSize));
+  ExtractFrame();
+  ExtractFrame();
+  CheckFrame(1, pid - 1, 0);
+  CheckNoFrame(2);
+}
+
+TEST_F(TestFrameBuffer2, ForwardJumps) {
+  EXPECT_EQ(5453, InsertFrame(5453, 0, 1, true, kFrameSize));
+  ExtractFrame();
+  EXPECT_EQ(5454, InsertFrame(5454, 0, 1, true, kFrameSize, 5453));
+  ExtractFrame();
+  EXPECT_EQ(15670, InsertFrame(15670, 0, 1, true, kFrameSize));
+  ExtractFrame();
+  EXPECT_EQ(29804, InsertFrame(29804, 0, 1, true, kFrameSize));
+  ExtractFrame();
+  EXPECT_EQ(29805, InsertFrame(29805, 0, 1, true, kFrameSize, 29804));
+  ExtractFrame();
+  EXPECT_EQ(29806, InsertFrame(29806, 0, 1, true, kFrameSize, 29805));
+  ExtractFrame();
+  EXPECT_EQ(33819, InsertFrame(33819, 0, 1, true, kFrameSize));
+  ExtractFrame();
+  EXPECT_EQ(41248, InsertFrame(41248, 0, 1, true, kFrameSize));
+  ExtractFrame();
+}
+
+TEST_F(TestFrameBuffer2, DuplicateFrames) {
+  EXPECT_EQ(22256, InsertFrame(22256, 0, 1, true, kFrameSize));
+  ExtractFrame();
+  EXPECT_EQ(22256, InsertFrame(22256, 0, 1, true, kFrameSize));
+}
+
+// TODO(philipel): implement more unittests related to invalid references.
+TEST_F(TestFrameBuffer2, InvalidReferences) {
+  EXPECT_EQ(-1, InsertFrame(0, 0, 1000, true, kFrameSize, 2));
+  EXPECT_EQ(1, InsertFrame(1, 0, 2000, true, kFrameSize));
+  ExtractFrame();
+  EXPECT_EQ(2, InsertFrame(2, 0, 3000, true, kFrameSize, 1));
+}
+
+TEST_F(TestFrameBuffer2, KeyframeRequired) {
+  EXPECT_EQ(1, InsertFrame(1, 0, 1000, true, kFrameSize));
+  EXPECT_EQ(2, InsertFrame(2, 0, 2000, true, kFrameSize, 1));
+  EXPECT_EQ(3, InsertFrame(3, 0, 3000, true, kFrameSize));
+  ExtractFrame();
+  ExtractFrame(0, true);
+  ExtractFrame();
+
+  CheckFrame(0, 1, 0);
+  CheckFrame(1, 3, 0);
+  CheckNoFrame(2);
+}
+
+TEST_F(TestFrameBuffer2, KeyframeClearsFullBuffer) {
+  const int kMaxBufferSize = 600;
+
+  for (int i = 1; i <= kMaxBufferSize; ++i)
+    EXPECT_EQ(-1, InsertFrame(i, 0, i * 1000, true, kFrameSize, i - 1));
+  ExtractFrame();
+  CheckNoFrame(0);
+
+  EXPECT_EQ(kMaxBufferSize + 1,
+            InsertFrame(kMaxBufferSize + 1, 0, (kMaxBufferSize + 1) * 1000,
+                        true, kFrameSize));
+  ExtractFrame();
+  CheckFrame(1, kMaxBufferSize + 1, 0);
+}
+
+TEST_F(TestFrameBuffer2, DontUpdateOnUndecodableFrame) {
+  InsertFrame(1, 0, 0, true, kFrameSize);
+  ExtractFrame(0, true);
+  InsertFrame(3, 0, 0, true, kFrameSize, 2, 0);
+  InsertFrame(3, 0, 0, true, kFrameSize, 0);
+  InsertFrame(2, 0, 0, true, kFrameSize);
+  ExtractFrame(0, true);
+  ExtractFrame(0, true);
+}
+
+TEST_F(TestFrameBuffer2, DontDecodeOlderTimestamp) {
+  InsertFrame(2, 0, 1, true, kFrameSize);
+  InsertFrame(1, 0, 2, true,
+              kFrameSize);  // Older picture id but newer timestamp.
+  ExtractFrame(0);
+  ExtractFrame(0);
+  CheckFrame(0, 1, 0);
+  CheckNoFrame(1);
+
+  InsertFrame(3, 0, 4, true, kFrameSize);
+  InsertFrame(4, 0, 3, true,
+              kFrameSize);  // Newer picture id but older timestamp.
+  ExtractFrame(0);
+  ExtractFrame(0);
+  CheckFrame(2, 3, 0);
+  CheckNoFrame(3);
+}
+
+TEST_F(TestFrameBuffer2, CombineFramesToSuperframe) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, false, kFrameSize);
+  InsertFrame(pid + 1, 1, ts, true, 2 * kFrameSize, pid);
+  ExtractFrame(0);
+  ExtractFrame(0);
+  CheckFrame(0, pid, 1);
+  CheckNoFrame(1);
+  // Two frames should be combined and returned together.
+  CheckFrameSize(0, 3 * kFrameSize);
+
+  EXPECT_EQ(frames_[0]->SpatialIndex(), 1);
+  EXPECT_EQ(frames_[0]->SpatialLayerFrameSize(0), kFrameSize);
+  EXPECT_EQ(frames_[0]->SpatialLayerFrameSize(1), 2 * kFrameSize);
+}
+
+TEST_F(TestFrameBuffer2, HigherSpatialLayerNonDecodable) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, false, kFrameSize);
+  InsertFrame(pid + 1, 1, ts, true, kFrameSize, pid);
+
+  ExtractFrame(0);
+  CheckFrame(0, pid, 1);
+
+  InsertFrame(pid + 3, 1, ts + kFps20, true, kFrameSize, pid);
+  InsertFrame(pid + 4, 0, ts + kFps10, false, kFrameSize, pid);
+  InsertFrame(pid + 5, 1, ts + kFps10, true, kFrameSize, pid + 3, pid + 4);
+
+  time_controller_.AdvanceTime(TimeDelta::Millis(1000));
+  // Frame pid+3 is decodable but too late.
+  // In superframe pid+4 is decodable, but frame pid+5 is not.
+  // Incorrect implementation might skip pid+2 frame and output undecodable
+  // pid+5 instead.
+  ExtractFrame();
+  ExtractFrame();
+  CheckFrame(1, pid + 3, 1);
+  CheckFrame(2, pid + 4, 1);
+}
+
+TEST_F(TestFrameBuffer2, StopWhileWaitingForFrame) {
+  uint16_t pid = Rand();
+  uint32_t ts = Rand();
+
+  InsertFrame(pid, 0, ts, true, kFrameSize);
+  ExtractFrame(10);
+  buffer_->Stop();
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  EXPECT_THAT(frames_, IsEmpty());
+
+  // A new frame request should exit immediately and return no new frame.
+  ExtractFrame(0);
+  EXPECT_THAT(frames_, IsEmpty());
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator.cc b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator.cc
new file mode 100644
index 0000000000..7ca59f779a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator.cc
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/frame_dependencies_calculator.h"
+
+#include <stdint.h>
+
+#include <iterator>
+#include <set>
+
+#include "absl/algorithm/container.h"
+#include "absl/container/inlined_vector.h"
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+absl::InlinedVector<int64_t, 5> FrameDependenciesCalculator::FromBuffersUsage(
+    int64_t frame_id,
+    rtc::ArrayView<const CodecBufferUsage> buffers_usage) {
+  absl::InlinedVector<int64_t, 5> dependencies;
+  RTC_DCHECK_GT(buffers_usage.size(), 0);
+  for (const CodecBufferUsage& buffer_usage : buffers_usage) {
+    RTC_CHECK_GE(buffer_usage.id, 0);
+    if (buffers_.size() <= static_cast<size_t>(buffer_usage.id)) {
+      buffers_.resize(buffer_usage.id + 1);
+    }
+  }
+  std::set<int64_t> direct_depenendencies;
+  std::set<int64_t> indirect_depenendencies;
+
+  for (const CodecBufferUsage& buffer_usage : buffers_usage) {
+    if (!buffer_usage.referenced) {
+      continue;
+    }
+    const BufferUsage& buffer = buffers_[buffer_usage.id];
+    if (buffer.frame_id == absl::nullopt) {
+      RTC_LOG(LS_ERROR) << "Odd configuration: frame " << frame_id
+                        << " references buffer #" << buffer_usage.id
+                        << " that was never updated.";
+      continue;
+    }
+    direct_depenendencies.insert(*buffer.frame_id);
+    indirect_depenendencies.insert(buffer.dependencies.begin(),
+                                   buffer.dependencies.end());
+  }
+  // Reduce references: if frame #3 depends on frame #2 and #1, and frame #2
+  // depends on frame #1, then frame #3 needs to depend just on frame #2.
+  // Though this set diff removes only 1 level of indirection, it seems
+  // enough for all currently used structures.
+  absl::c_set_difference(direct_depenendencies, indirect_depenendencies,
+                         std::back_inserter(dependencies));
+
+  // Update buffers.
+  for (const CodecBufferUsage& buffer_usage : buffers_usage) {
+    if (!buffer_usage.updated) {
+      continue;
+    }
+    BufferUsage& buffer = buffers_[buffer_usage.id];
+    buffer.frame_id = frame_id;
+    buffer.dependencies.assign(direct_depenendencies.begin(),
+                               direct_depenendencies.end());
+  }
+
+  return dependencies;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator.h b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator.h
new file mode 100644
index 0000000000..2c4a8502e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_FRAME_DEPENDENCIES_CALCULATOR_H_
+#define MODULES_VIDEO_CODING_FRAME_DEPENDENCIES_CALCULATOR_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+
+namespace webrtc {
+
+// This class is thread compatible.
+class FrameDependenciesCalculator {
+ public:
+  FrameDependenciesCalculator() = default;
+  FrameDependenciesCalculator(const FrameDependenciesCalculator&) = default;
+  FrameDependenciesCalculator& operator=(const FrameDependenciesCalculator&) =
+      default;
+
+  // Calculates frame dependencies based on previous encoder buffer usage.
+  absl::InlinedVector<int64_t, 5> FromBuffersUsage(
+      int64_t frame_id,
+      rtc::ArrayView<const CodecBufferUsage> buffers_usage);
+
+ private:
+  struct BufferUsage {
+    absl::optional<int64_t> frame_id;
+    absl::InlinedVector<int64_t, 4> dependencies;
+  };
+
+  absl::InlinedVector<BufferUsage, 4> buffers_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_FRAME_DEPENDENCIES_CALCULATOR_H_
diff --git a/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_gn/moz.build
new file mode 100644
index 0000000000..c518044c94
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("frame_dependencies_calculator_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_unittest.cc b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_unittest.cc
new file mode 100644
index 0000000000..a09650401a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_dependencies_calculator_unittest.cc
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/frame_dependencies_calculator.h"
+
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+using ::testing::UnorderedElementsAre;
+
+constexpr CodecBufferUsage ReferenceAndUpdate(int id) {
+  return CodecBufferUsage(id, /*referenced=*/true, /*updated=*/true);
+}
+constexpr CodecBufferUsage Reference(int id) {
+  return CodecBufferUsage(id, /*referenced=*/true, /*updated=*/false);
+}
+constexpr CodecBufferUsage Update(int id) {
+  return CodecBufferUsage(id, /*referenced=*/false, /*updated=*/true);
+}
+
+TEST(FrameDependenciesCalculatorTest, SingleLayer) {
+  CodecBufferUsage pattern[] = {ReferenceAndUpdate(0)};
+  FrameDependenciesCalculator calculator;
+
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/1, pattern), IsEmpty());
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/3, pattern),
+              ElementsAre(1));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/6, pattern),
+              ElementsAre(3));
+}
+
+TEST(FrameDependenciesCalculatorTest, TwoTemporalLayers) {
+  // Shortened 4-frame pattern:
+  // T1:  2---4   6---8 ...
+  //      /   /   /   /
+  // T0: 1---3---5---7 ...
+  CodecBufferUsage pattern0[] = {ReferenceAndUpdate(0)};
+  CodecBufferUsage pattern1[] = {Reference(0), Update(1)};
+  CodecBufferUsage pattern2[] = {ReferenceAndUpdate(0)};
+  CodecBufferUsage pattern3[] = {Reference(0), Reference(1)};
+  FrameDependenciesCalculator calculator;
+
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/1, pattern0), IsEmpty());
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/2, pattern1),
+              ElementsAre(1));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/3, pattern2),
+              ElementsAre(1));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/4, pattern3),
+              UnorderedElementsAre(2, 3));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/5, pattern0),
+              ElementsAre(3));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/6, pattern1),
+              ElementsAre(5));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/7, pattern2),
+              ElementsAre(5));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/8, pattern3),
+              UnorderedElementsAre(6, 7));
+}
+
+TEST(FrameDependenciesCalculatorTest, ThreeTemporalLayers4FramePattern) {
+  // T2:   2---4   6---8 ...
+  //      /   /   /   /
+  // T1:  |  3    |  7   ...
+  //      /_/     /_/
+  // T0: 1-------5-----  ...
+  CodecBufferUsage pattern0[] = {ReferenceAndUpdate(0)};
+  CodecBufferUsage pattern1[] = {Reference(0), Update(2)};
+  CodecBufferUsage pattern2[] = {Reference(0), Update(1)};
+  CodecBufferUsage pattern3[] = {Reference(0), Reference(1), Reference(2)};
+  FrameDependenciesCalculator calculator;
+
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/1, pattern0), IsEmpty());
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/2, pattern1),
+              ElementsAre(1));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/3, pattern2),
+              ElementsAre(1));
+  // Note that frame#4 references buffer#0 that is updated by frame#1,
+  // yet there is no direct dependency from frame#4 to frame#1.
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/4, pattern3),
+              UnorderedElementsAre(2, 3));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/5, pattern0),
+              ElementsAre(1));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/6, pattern1),
+              ElementsAre(5));
+}
+
+TEST(FrameDependenciesCalculatorTest, SimulcastWith2Layers) {
+  // S1: 2---4---6-  ...
+  //
+  // S0: 1---3---5-  ...
+  CodecBufferUsage pattern0[] = {ReferenceAndUpdate(0)};
+  CodecBufferUsage pattern1[] = {ReferenceAndUpdate(1)};
+  FrameDependenciesCalculator calculator;
+
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/1, pattern0), IsEmpty());
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/2, pattern1), IsEmpty());
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/3, pattern0),
+              ElementsAre(1));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/4, pattern1),
+              ElementsAre(2));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/5, pattern0),
+              ElementsAre(3));
+  EXPECT_THAT(calculator.FromBuffersUsage(/*frame_id=*/6, pattern1),
+              ElementsAre(4));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/frame_helpers.cc b/third_party/libwebrtc/modules/video_coding/frame_helpers.cc
new file mode 100644
index 0000000000..e25eac8a18
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_helpers.cc
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/frame_helpers.h"
+
+#include <utility>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+constexpr TimeDelta kMaxVideoDelay = TimeDelta::Millis(10000);
+}
+
+bool FrameHasBadRenderTiming(Timestamp render_time, Timestamp now) {
+  // Zero render time means render immediately.
+  if (render_time.IsZero()) {
+    return false;
+  }
+  if (render_time < Timestamp::Zero()) {
+    return true;
+  }
+  TimeDelta frame_delay = render_time - now;
+  if (frame_delay.Abs() > kMaxVideoDelay) {
+    RTC_LOG(LS_WARNING) << "Frame has bad render timing because it is out of "
+                           "the delay bounds (frame_delay_ms="
+                        << frame_delay.ms()
+                        << ", kMaxVideoDelay_ms=" << kMaxVideoDelay.ms() << ")";
+    return true;
+  }
+  return false;
+}
+
+bool TargetVideoDelayIsTooLarge(TimeDelta target_video_delay) {
+  if (target_video_delay > kMaxVideoDelay) {
+    RTC_LOG(LS_WARNING)
+        << "Target video delay is too large. (target_video_delay_ms="
+        << target_video_delay.ms()
+        << ", kMaxVideoDelay_ms=" << kMaxVideoDelay.ms() << ")";
+    return true;
+  }
+  return false;
+}
+
+std::unique_ptr<EncodedFrame> CombineAndDeleteFrames(
+    absl::InlinedVector<std::unique_ptr<EncodedFrame>, 4> frames) {
+  RTC_DCHECK(!frames.empty());
+
+  if (frames.size() == 1) {
+    return std::move(frames[0]);
+  }
+
+  size_t total_length = 0;
+  for (const auto& frame : frames) {
+    total_length += frame->size();
+  }
+  const EncodedFrame& last_frame = *frames.back();
+  std::unique_ptr<EncodedFrame> first_frame = std::move(frames[0]);
+  auto encoded_image_buffer = EncodedImageBuffer::Create(total_length);
+  uint8_t* buffer = encoded_image_buffer->data();
+  first_frame->SetSpatialLayerFrameSize(first_frame->SpatialIndex().value_or(0),
+                                        first_frame->size());
+  memcpy(buffer, first_frame->data(), first_frame->size());
+  buffer += first_frame->size();
+
+  // Spatial index of combined frame is set equal to spatial index of its top
+  // spatial layer.
+  first_frame->SetSpatialIndex(last_frame.SpatialIndex().value_or(0));
+
+  first_frame->video_timing_mutable()->network2_timestamp_ms =
+      last_frame.video_timing().network2_timestamp_ms;
+  first_frame->video_timing_mutable()->receive_finish_ms =
+      last_frame.video_timing().receive_finish_ms;
+
+  // Append all remaining frames to the first one.
+  for (size_t i = 1; i < frames.size(); ++i) {
+    // Let |next_frame| fall out of scope so it is deleted after copying.
+    std::unique_ptr<EncodedFrame> next_frame = std::move(frames[i]);
+    first_frame->SetSpatialLayerFrameSize(
+        next_frame->SpatialIndex().value_or(0), next_frame->size());
+    memcpy(buffer, next_frame->data(), next_frame->size());
+    buffer += next_frame->size();
+  }
+  first_frame->SetEncodedData(encoded_image_buffer);
+  return first_frame;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/frame_helpers.h b/third_party/libwebrtc/modules/video_coding/frame_helpers.h
new file mode 100644
index 0000000000..56ee593678
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_helpers.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_FRAME_HELPERS_H_
+#define MODULES_VIDEO_CODING_FRAME_HELPERS_H_
+
+#include <memory>
+
+#include "absl/container/inlined_vector.h"
+#include "api/video/encoded_frame.h"
+
+namespace webrtc {
+
+bool FrameHasBadRenderTiming(Timestamp render_time, Timestamp now);
+
+bool TargetVideoDelayIsTooLarge(TimeDelta target_video_delay);
+
+std::unique_ptr<EncodedFrame> CombineAndDeleteFrames(
+    absl::InlinedVector<std::unique_ptr<EncodedFrame>, 4> frames);
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_FRAME_HELPERS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/frame_helpers_gn/moz.build b/third_party/libwebrtc/modules/video_coding/frame_helpers_gn/moz.build
new file mode 100644
index 0000000000..d6342c48f7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_helpers_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/frame_helpers.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("frame_helpers_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/frame_helpers_unittest.cc b/third_party/libwebrtc/modules/video_coding/frame_helpers_unittest.cc
new file mode 100644
index 0000000000..1f73689c0a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_helpers_unittest.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/frame_helpers.h"
+
+#include "api/units/timestamp.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+TEST(FrameHasBadRenderTimingTest, LargePositiveFrameDelayIsBad) {
+  Timestamp render_time = Timestamp::Seconds(12);
+  Timestamp now = Timestamp::Seconds(0);
+
+  EXPECT_TRUE(FrameHasBadRenderTiming(render_time, now));
+}
+
+TEST(FrameHasBadRenderTimingTest, LargeNegativeFrameDelayIsBad) {
+  Timestamp render_time = Timestamp::Seconds(12);
+  Timestamp now = Timestamp::Seconds(24);
+
+  EXPECT_TRUE(FrameHasBadRenderTiming(render_time, now));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/frame_object.cc b/third_party/libwebrtc/modules/video_coding/frame_object.cc
new file mode 100644
index 0000000000..d226dcd013
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_object.cc
@@ -0,0 +1,131 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/frame_object.h"
+
+#include <string.h>
+
+#include <utility>
+
+#include "api/video/encoded_image.h"
+#include "api/video/video_timing.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+RtpFrameObject::RtpFrameObject(
+    uint16_t first_seq_num,
+    uint16_t last_seq_num,
+    bool markerBit,
+    int times_nacked,
+    int64_t first_packet_received_time,
+    int64_t last_packet_received_time,
+    uint32_t rtp_timestamp,
+    int64_t ntp_time_ms,
+    const VideoSendTiming& timing,
+    uint8_t payload_type,
+    VideoCodecType codec,
+    VideoRotation rotation,
+    VideoContentType content_type,
+    const RTPVideoHeader& video_header,
+    const absl::optional<webrtc::ColorSpace>& color_space,
+    RtpPacketInfos packet_infos,
+    rtc::scoped_refptr<EncodedImageBuffer> image_buffer)
+    : image_buffer_(image_buffer),
+      first_seq_num_(first_seq_num),
+      last_seq_num_(last_seq_num),
+      last_packet_received_time_(last_packet_received_time),
+      times_nacked_(times_nacked) {
+  rtp_video_header_ = video_header;
+
+  // EncodedFrame members
+  codec_type_ = codec;
+
+  // TODO(philipel): Remove when encoded image is replaced by EncodedFrame.
+  // VCMEncodedFrame members
+  CopyCodecSpecific(&rtp_video_header_);
+  _payloadType = payload_type;
+  SetTimestamp(rtp_timestamp);
+  ntp_time_ms_ = ntp_time_ms;
+  _frameType = rtp_video_header_.frame_type;
+
+  // Setting frame's playout delays to the same values
+  // as of the first packet's.
+  SetPlayoutDelay(rtp_video_header_.playout_delay);
+
+  SetEncodedData(image_buffer_);
+  _encodedWidth = rtp_video_header_.width;
+  _encodedHeight = rtp_video_header_.height;
+
+  // EncodedFrame members
+  SetPacketInfos(std::move(packet_infos));
+
+  rotation_ = rotation;
+  SetColorSpace(color_space);
+  SetVideoFrameTrackingId(rtp_video_header_.video_frame_tracking_id);
+  content_type_ = content_type;
+  if (timing.flags != VideoSendTiming::kInvalid) {
+    // ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
+    // as this will be dealt with at the time of reporting.
+    timing_.encode_start_ms = ntp_time_ms_ + timing.encode_start_delta_ms;
+    timing_.encode_finish_ms = ntp_time_ms_ + timing.encode_finish_delta_ms;
+    timing_.packetization_finish_ms =
+        ntp_time_ms_ + timing.packetization_finish_delta_ms;
+    timing_.pacer_exit_ms = ntp_time_ms_ + timing.pacer_exit_delta_ms;
+    timing_.network_timestamp_ms =
+        ntp_time_ms_ + timing.network_timestamp_delta_ms;
+    timing_.network2_timestamp_ms =
+        ntp_time_ms_ + timing.network2_timestamp_delta_ms;
+  }
+  timing_.receive_start_ms = first_packet_received_time;
+  timing_.receive_finish_ms = last_packet_received_time;
+  timing_.flags = timing.flags;
+  is_last_spatial_layer = markerBit;
+}
+
+RtpFrameObject::~RtpFrameObject() {
+}
+
+uint16_t RtpFrameObject::first_seq_num() const {
+  return first_seq_num_;
+}
+
+uint16_t RtpFrameObject::last_seq_num() const {
+  return last_seq_num_;
+}
+
+int RtpFrameObject::times_nacked() const {
+  return times_nacked_;
+}
+
+VideoFrameType RtpFrameObject::frame_type() const {
+  return rtp_video_header_.frame_type;
+}
+
+VideoCodecType RtpFrameObject::codec_type() const {
+  return codec_type_;
+}
+
+int64_t RtpFrameObject::ReceivedTime() const {
+  return last_packet_received_time_;
+}
+
+int64_t RtpFrameObject::RenderTime() const {
+  return _renderTimeMs;
+}
+
+bool RtpFrameObject::delayed_by_retransmission() const {
+  return times_nacked() > 0;
+}
+
+const RTPVideoHeader& RtpFrameObject::GetRtpVideoHeader() const {
+  return rtp_video_header_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/frame_object.h b/third_party/libwebrtc/modules/video_coding/frame_object.h
new file mode 100644
index 0000000000..c6f069f241
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/frame_object.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_FRAME_OBJECT_H_
+#define MODULES_VIDEO_CODING_FRAME_OBJECT_H_
+
+#include "absl/types/optional.h"
+#include "api/video/encoded_frame.h"
+
+namespace webrtc {
+
+class RtpFrameObject : public EncodedFrame {
+ public:
+  RtpFrameObject(uint16_t first_seq_num,
+                 uint16_t last_seq_num,
+                 bool markerBit,
+                 int times_nacked,
+                 int64_t first_packet_received_time,
+                 int64_t last_packet_received_time,
+                 uint32_t rtp_timestamp,
+                 int64_t ntp_time_ms,
+                 const VideoSendTiming& timing,
+                 uint8_t payload_type,
+                 VideoCodecType codec,
+                 VideoRotation rotation,
+                 VideoContentType content_type,
+                 const RTPVideoHeader& video_header,
+                 const absl::optional<webrtc::ColorSpace>& color_space,
+                 RtpPacketInfos packet_infos,
+                 rtc::scoped_refptr<EncodedImageBuffer> image_buffer);
+
+  ~RtpFrameObject() override;
+  uint16_t first_seq_num() const;
+  uint16_t last_seq_num() const;
+  int times_nacked() const;
+  VideoFrameType frame_type() const;
+  VideoCodecType codec_type() const;
+  int64_t ReceivedTime() const override;
+  int64_t RenderTime() const override;
+  bool delayed_by_retransmission() const override;
+  const RTPVideoHeader& GetRtpVideoHeader() const;
+
+  uint8_t* mutable_data() { return image_buffer_->data(); }
+
+ private:
+  // Reference for mutable access.
+  rtc::scoped_refptr<EncodedImageBuffer> image_buffer_;
+  RTPVideoHeader rtp_video_header_;
+  VideoCodecType codec_type_;
+  uint16_t first_seq_num_;
+  uint16_t last_seq_num_;
+  int64_t last_packet_received_time_;
+
+  // Equal to times nacked of the packet with the highet times nacked
+  // belonging to this frame.
+  int times_nacked_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_FRAME_OBJECT_H_
diff --git a/third_party/libwebrtc/modules/video_coding/g3doc/index.md b/third_party/libwebrtc/modules/video_coding/g3doc/index.md
new file mode 100644
index 0000000000..fdf39982fa
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/g3doc/index.md
@@ -0,0 +1,177 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'brandtr' reviewed: '2021-04-15'} *-->
+
+# Video coding in WebRTC
+
+## Introduction to layered video coding
+
+[Video coding][video-coding-wiki] is the process of encoding a stream of
+uncompressed video frames into a compressed bitstream, whose bitrate is lower
+than that of the original stream.
+
+### Block-based hybrid video coding
+
+All video codecs in WebRTC are based on the block-based hybrid video coding
+paradigm, which entails prediction of the original video frame using either
+[information from previously encoded frames][motion-compensation-wiki] or
+information from previously encoded portions of the current frame, subtraction
+of the prediction from the original video, and
+[transform][transform-coding-wiki] and [quantization][quantization-wiki] of the
+resulting difference. The output of the quantization process, quantized
+transform coefficients, is losslessly [entropy coded][entropy-coding-wiki] along
+with other encoder parameters (e.g., those related to the prediction process)
+and then a reconstruction is constructed by inverse quantizing and inverse
+transforming the quantized transform coefficients and adding the result to the
+prediction. Finally, in-loop filtering is applied and the resulting
+reconstruction is stored as a reference frame to be used to develop predictions
+for future frames.
+
+### Frame types
+
+When an encoded frame depends on previously encoded frames (i.e., it has one or
+more inter-frame dependencies), the prior frames must be available at the
+receiver before the current frame can be decoded. In order for a receiver to
+start decoding an encoded bitstream, a frame which has no prior dependencies is
+required. Such a frame is called a "key frame". For real-time-communications
+encoding, key frames typically compress less efficiently than "delta frames"
+(i.e., frames whose predictions are derived from previously encoded frames).
+
+### Single-layer coding
+
+In 1:1 calls, the encoded bitstream has a single recipient. Using end-to-end
+bandwidth estimation, the target bitrate can thus be well tailored for the
+intended recipient. The number of key frames can be kept to a minimum and the
+compressability of the stream can be maximized. One way of achiving this is by
+using "single-layer coding", where each delta frame only depends on the frame
+that was most recently encoded.
+
+### Scalable video coding
+
+In multiway conferences, on the other hand, the encoded bitstream has multiple
+recipients each of whom may have different downlink bandwidths. In order to
+tailor the encoded bitstreams to a heterogeneous network of receivers,
+[scalable video coding][svc-wiki] can be used. The idea is to introduce
+structure into the dependency graph of the encoded bitstream, such that _layers_ of
+the full stream can be decoded using only available lower layers. This structure
+allows for a [selective forwarding unit][sfu-webrtc-glossary] to discard upper
+layers of the of the bitstream in order to achieve the intended downlink
+bandwidth.
+
+There are multiple types of scalability:
+
+* _Temporal scalability_ are layers whose framerate (and bitrate) is lower than that of the upper layer(s)
+* _Spatial scalability_ are layers whose resolution (and bitrate) is lower than that of the upper layer(s)
+* _Quality scalability_ are layers whose bitrate is lower than that of the upper layer(s)
+
+WebRTC supports temporal scalability for `VP8`, `VP9` and `AV1`, and spatial
+scalability for `VP9` and `AV1`.
+
+### Simulcast
+
+Simulcast is another approach for multiway conferencing, where multiple
+_independent_ bitstreams are produced by the encoder.
+
+In cases where multiple encodings of the same source are required (e.g., uplink
+transmission in a multiway call), spatial scalability with inter-layer
+prediction generally offers superior coding efficiency compared with simulcast.
+When a single encoding is required (e.g., downlink transmission in any call),
+simulcast generally provides better coding efficiency for the upper spatial
+layers. The `K-SVC` concept, where spatial inter-layer dependencies are only
+used to encode key frames, for which inter-layer prediction is typically
+significantly more effective than it is for delta frames, can be seen as a
+compromise between full spatial scalability and simulcast.
+
+## Overview of implementation in `modules/video_coding`
+
+Given the general introduction to video coding above, we now describe some
+specifics of the [`modules/video_coding`][modules-video-coding] folder in WebRTC.
+
+### Built-in software codecs in [`modules/video_coding/codecs`][modules-video-coding-codecs]
+
+This folder contains WebRTC-specific classes that wrap software codec
+implementations for different video coding standards:
+
+* [libaom][libaom-src] for [AV1][av1-spec]
+* [libvpx][libvpx-src] for [VP8][vp8-spec] and [VP9][vp9-spec]
+* [OpenH264][openh264-src] for [H.264 constrained baseline profile][h264-spec]
+
+Users of the library can also inject their own codecs, using the
+[VideoEncoderFactory][video-encoder-factory-interface] and
+[VideoDecoderFactory][video-decoder-factory-interface] interfaces. This is how
+platform-supported codecs, such as hardware backed codecs, are implemented.
+
+### Video codec test framework in [`modules/video_coding/codecs/test`][modules-video-coding-codecs-test]
+
+This folder contains a test framework that can be used to evaluate video quality
+performance of different video codec implementations.
+
+### SVC helper classes in [`modules/video_coding/svc`][modules-video-coding-svc]
+
+*   [`ScalabilityStructure*`][scalabilitystructure] - different
+    [standardized scalability structures][scalability-structure-spec]
+*   [`ScalableVideoController`][scalablevideocontroller] - provides instructions to the video encoder how
+    to create a scalable stream
+*   [`SvcRateAllocator`][svcrateallocator] - bitrate allocation to different spatial and temporal
+    layers
+
+### Utility classes in [`modules/video_coding/utility`][modules-video-coding-utility]
+
+*   [`FrameDropper`][framedropper] - drops incoming frames when encoder systematically
+    overshoots its target bitrate
+*   [`FramerateController`][frameratecontroller] - drops incoming frames to achieve a target framerate
+*   [`QpParser`][qpparser] - parses the quantization parameter from a bitstream
+*   [`QualityScaler`][qualityscaler] - signals when an encoder generates encoded frames whose
+    quantization parameter is outside the window of acceptable values
+*   [`SimulcastRateAllocator`][simulcastrateallocator] - bitrate allocation to simulcast layers
+
+### General helper classes in [`modules/video_coding`][modules-video-coding]
+
+*   [`FecControllerDefault`][feccontrollerdefault] - provides a default implementation for rate
+    allocation to [forward error correction][fec-wiki]
+*   [`VideoCodecInitializer`][videocodecinitializer] - converts between different encoder configuration
+    structs
+
+### Receiver buffer classes in [`modules/video_coding`][modules-video-coding]
+
+*   [`PacketBuffer`][packetbuffer] - (re-)combines RTP packets into frames
+*   [`RtpFrameReferenceFinder`][rtpframereferencefinder] - determines dependencies between frames based on information in the RTP header, payload header and RTP extensions
+*   [`FrameBuffer`][framebuffer] - order frames based on their dependencies to be fed to the decoder
+
+[video-coding-wiki]: https://en.wikipedia.org/wiki/Video_coding_format
+[motion-compensation-wiki]: https://en.wikipedia.org/wiki/Motion_compensation
+[transform-coding-wiki]: https://en.wikipedia.org/wiki/Transform_coding
+[motion-vector-wiki]: https://en.wikipedia.org/wiki/Motion_vector
+[mpeg-wiki]: https://en.wikipedia.org/wiki/Moving_Picture_Experts_Group
+[svc-wiki]: https://en.wikipedia.org/wiki/Scalable_Video_Coding
+[sfu-webrtc-glossary]: https://webrtcglossary.com/sfu/
+[libvpx-src]: https://chromium.googlesource.com/webm/libvpx/
+[libaom-src]: https://aomedia.googlesource.com/aom/
+[openh264-src]: https://github.com/cisco/openh264
+[vp8-spec]: https://tools.ietf.org/html/rfc6386
+[vp9-spec]: https://storage.googleapis.com/downloads.webmproject.org/docs/vp9/vp9-bitstream-specification-v0.6-20160331-draft.pdf
+[av1-spec]: https://aomediacodec.github.io/av1-spec/
+[h264-spec]: https://www.itu.int/rec/T-REC-H.264-201906-I/en
+[video-encoder-factory-interface]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/video_codecs/video_encoder_factory.h;l=27;drc=afadfb24a5e608da6ae102b20b0add53a083dcf3
+[video-decoder-factory-interface]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/video_codecs/video_decoder_factory.h;l=27;drc=49c293f03d8f593aa3aca282577fcb14daa63207
+[scalability-structure-spec]: https://w3c.github.io/webrtc-svc/#scalabilitymodes*
+[fec-wiki]: https://en.wikipedia.org/wiki/Error_correction_code#Forward_error_correction
+[entropy-coding-wiki]: https://en.wikipedia.org/wiki/Entropy_encoding
+[modules-video-coding]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/
+[modules-video-coding-codecs]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/codecs/
+[modules-video-coding-codecs-test]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/codecs/test/
+[modules-video-coding-svc]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/svc/
+[modules-video-coding-utility]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/utility/
+[scalabilitystructure]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/svc/create_scalability_structure.h?q=CreateScalabilityStructure
+[scalablevideocontroller]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/svc/scalable_video_controller.h?q=ScalableVideoController
+[svcrateallocator]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/svc/svc_rate_allocator.h?q=SvcRateAllocator
+[framedropper]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/utility/frame_dropper.h?q=FrameDropper
+[frameratecontroller]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/utility/framerate_controller.h?q=FramerateController
+[qpparser]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/utility/qp_parser.h?q=QpParser
+[qualityscaler]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/utility/quality_scaler.h?q=QualityScaler
+[simulcastrateallocator]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/utility/simulcast_rate_allocator.h?q=SimulcastRateAllocator
+[feccontrollerdefault]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/fec_controller_default.h?q=FecControllerDefault
+[videocodecinitializer]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/include/video_codec_initializer.h?q=VideoCodecInitializer
+[packetbuffer]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/packet_buffer.h?q=PacketBuffer
+[rtpframereferencefinder]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/rtp_frame_reference_finder.h?q=RtpFrameReferenceFinder
+[framebuffer]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/frame_buffer2.h?q=FrameBuffer
+[quantization-wiki]: https://en.wikipedia.org/wiki/Quantization_(signal_processing)
diff --git a/third_party/libwebrtc/modules/video_coding/generic_decoder.cc b/third_party/libwebrtc/modules/video_coding/generic_decoder.cc
new file mode 100644
index 0000000000..54467d1477
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/generic_decoder.cc
@@ -0,0 +1,325 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/generic_decoder.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <cmath>
+#include <iterator>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/types/optional.h"
+#include "api/video/video_timing.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace {
+
+// Changed from 10 to 30 in Mozilla Bug 989944: Increase decode
+// timestamp map to handle delayed decode on 8x10.  The map is
+// now a deque (as of libwebrtc upstream commit 1c51ec4d74).
+constexpr size_t kDecoderFrameMemoryLength = 30;
+
+}
+
+VCMDecodedFrameCallback::VCMDecodedFrameCallback(
+    VCMTiming* timing,
+    Clock* clock,
+    const FieldTrialsView& field_trials)
+    : _clock(clock), _timing(timing) {
+  ntp_offset_ =
+      _clock->CurrentNtpInMilliseconds() - _clock->TimeInMilliseconds();
+}
+
+VCMDecodedFrameCallback::~VCMDecodedFrameCallback() {}
+
+void VCMDecodedFrameCallback::SetUserReceiveCallback(
+    VCMReceiveCallback* receiveCallback) {
+  RTC_DCHECK(construction_thread_.IsCurrent());
+  RTC_DCHECK((!_receiveCallback && receiveCallback) ||
+             (_receiveCallback && !receiveCallback));
+  _receiveCallback = receiveCallback;
+}
+
+VCMReceiveCallback* VCMDecodedFrameCallback::UserReceiveCallback() {
+  // Called on the decode thread via VCMCodecDataBase::GetDecoder.
+  // The callback must always have been set before this happens.
+  RTC_DCHECK(_receiveCallback);
+  return _receiveCallback;
+}
+
+int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage) {
+  // This function may be called on the decode TaskQueue, but may also be called
+  // on an OS provided queue such as on iOS (see e.g. b/153465112).
+  return Decoded(decodedImage, -1);
+}
+
+int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
+                                         int64_t decode_time_ms) {
+  Decoded(decodedImage,
+          decode_time_ms >= 0 ? absl::optional<int32_t>(decode_time_ms)
+                              : absl::nullopt,
+          absl::nullopt);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+std::pair<absl::optional<FrameInfo>, size_t>
+VCMDecodedFrameCallback::FindFrameInfo(uint32_t rtp_timestamp) {
+  absl::optional<FrameInfo> frame_info;
+
+  auto it = absl::c_find_if(frame_infos_, [rtp_timestamp](const auto& entry) {
+    return entry.rtp_timestamp == rtp_timestamp ||
+           IsNewerTimestamp(entry.rtp_timestamp, rtp_timestamp);
+  });
+  size_t dropped_frames = std::distance(frame_infos_.begin(), it);
+
+  if (it != frame_infos_.end() && it->rtp_timestamp == rtp_timestamp) {
+    // Frame was found and should also be removed from the queue.
+    frame_info = std::move(*it);
+    ++it;
+  }
+
+  frame_infos_.erase(frame_infos_.begin(), it);
+  return std::make_pair(std::move(frame_info), dropped_frames);
+}
+
+void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
+                                      absl::optional<int32_t> decode_time_ms,
+                                      absl::optional<uint8_t> qp) {
+  RTC_DCHECK(_receiveCallback) << "Callback must not be null at this point";
+  TRACE_EVENT_INSTANT1("webrtc", "VCMDecodedFrameCallback::Decoded",
+                       "timestamp", decodedImage.timestamp());
+  // TODO(holmer): We should improve this so that we can handle multiple
+  // callbacks from one call to Decode().
+  absl::optional<FrameInfo> frame_info;
+  int timestamp_map_size = 0;
+  int dropped_frames = 0;
+  {
+    MutexLock lock(&lock_);
+    std::tie(frame_info, dropped_frames) =
+        FindFrameInfo(decodedImage.timestamp());
+    timestamp_map_size = frame_infos_.size();
+  }
+  if (dropped_frames > 0) {
+    _receiveCallback->OnDroppedFrames(dropped_frames);
+  }
+
+  if (!frame_info) {
+    RTC_LOG(LS_WARNING) << "Too many frames backed up in the decoder, dropping "
+                           "frame with timestamp "
+                        << decodedImage.timestamp();
+    return;
+  }
+
+  decodedImage.set_ntp_time_ms(frame_info->ntp_time_ms);
+  decodedImage.set_packet_infos(frame_info->packet_infos);
+  decodedImage.set_rotation(frame_info->rotation);
+  VideoFrame::RenderParameters render_parameters = _timing->RenderParameters();
+  if (render_parameters.max_composition_delay_in_frames) {
+    // Subtract frames that are in flight.
+    render_parameters.max_composition_delay_in_frames =
+        std::max(0, *render_parameters.max_composition_delay_in_frames -
+                        timestamp_map_size);
+  }
+  decodedImage.set_render_parameters(render_parameters);
+
+  RTC_DCHECK(frame_info->decode_start);
+  const Timestamp now = _clock->CurrentTime();
+  const TimeDelta decode_time = decode_time_ms
+                                    ? TimeDelta::Millis(*decode_time_ms)
+                                    : now - *frame_info->decode_start;
+  _timing->StopDecodeTimer(decode_time, now);
+  decodedImage.set_processing_time(
+      {*frame_info->decode_start, *frame_info->decode_start + decode_time});
+
+  // Report timing information.
+  TimingFrameInfo timing_frame_info;
+  if (frame_info->timing.flags != VideoSendTiming::kInvalid) {
+    int64_t capture_time_ms = decodedImage.ntp_time_ms() - ntp_offset_;
+    // Convert remote timestamps to local time from ntp timestamps.
+    frame_info->timing.encode_start_ms -= ntp_offset_;
+    frame_info->timing.encode_finish_ms -= ntp_offset_;
+    frame_info->timing.packetization_finish_ms -= ntp_offset_;
+    frame_info->timing.pacer_exit_ms -= ntp_offset_;
+    frame_info->timing.network_timestamp_ms -= ntp_offset_;
+    frame_info->timing.network2_timestamp_ms -= ntp_offset_;
+
+    int64_t sender_delta_ms = 0;
+    if (decodedImage.ntp_time_ms() < 0) {
+      // Sender clock is not estimated yet. Make sure that sender times are all
+      // negative to indicate that. Yet they still should be relatively correct.
+      sender_delta_ms =
+          std::max({capture_time_ms, frame_info->timing.encode_start_ms,
+                    frame_info->timing.encode_finish_ms,
+                    frame_info->timing.packetization_finish_ms,
+                    frame_info->timing.pacer_exit_ms,
+                    frame_info->timing.network_timestamp_ms,
+                    frame_info->timing.network2_timestamp_ms}) +
+          1;
+    }
+
+    timing_frame_info.capture_time_ms = capture_time_ms - sender_delta_ms;
+    timing_frame_info.encode_start_ms =
+        frame_info->timing.encode_start_ms - sender_delta_ms;
+    timing_frame_info.encode_finish_ms =
+        frame_info->timing.encode_finish_ms - sender_delta_ms;
+    timing_frame_info.packetization_finish_ms =
+        frame_info->timing.packetization_finish_ms - sender_delta_ms;
+    timing_frame_info.pacer_exit_ms =
+        frame_info->timing.pacer_exit_ms - sender_delta_ms;
+    timing_frame_info.network_timestamp_ms =
+        frame_info->timing.network_timestamp_ms - sender_delta_ms;
+    timing_frame_info.network2_timestamp_ms =
+        frame_info->timing.network2_timestamp_ms - sender_delta_ms;
+  }
+
+  timing_frame_info.flags = frame_info->timing.flags;
+  timing_frame_info.decode_start_ms = frame_info->decode_start->ms();
+  timing_frame_info.decode_finish_ms = now.ms();
+  timing_frame_info.render_time_ms =
+      frame_info->render_time ? frame_info->render_time->ms() : -1;
+  timing_frame_info.rtp_timestamp = decodedImage.timestamp();
+  timing_frame_info.receive_start_ms = frame_info->timing.receive_start_ms;
+  timing_frame_info.receive_finish_ms = frame_info->timing.receive_finish_ms;
+  _timing->SetTimingFrameInfo(timing_frame_info);
+
+  decodedImage.set_timestamp_us(
+      frame_info->render_time ? frame_info->render_time->us() : -1);
+  _receiveCallback->FrameToRender(decodedImage, qp, decode_time,
+                                  frame_info->content_type);
+}
+
+void VCMDecodedFrameCallback::OnDecoderInfoChanged(
+    const VideoDecoder::DecoderInfo& decoder_info) {
+  _receiveCallback->OnDecoderInfoChanged(decoder_info);
+}
+
+void VCMDecodedFrameCallback::Map(FrameInfo frameInfo) {
+  int dropped_frames = 0;
+  {
+    MutexLock lock(&lock_);
+    int initial_size = frame_infos_.size();
+    if (initial_size == kDecoderFrameMemoryLength) {
+      frame_infos_.pop_front();
+      dropped_frames = 1;
+    }
+    frame_infos_.push_back(std::move(frameInfo));
+    // If no frame is dropped, the new size should be `initial_size` + 1
+  }
+  if (dropped_frames > 0) {
+    _receiveCallback->OnDroppedFrames(dropped_frames);
+  }
+}
+
+void VCMDecodedFrameCallback::ClearTimestampMap() {
+  int dropped_frames = 0;
+  {
+    MutexLock lock(&lock_);
+    dropped_frames = frame_infos_.size();
+    frame_infos_.clear();
+  }
+  if (dropped_frames > 0) {
+    _receiveCallback->OnDroppedFrames(dropped_frames);
+  }
+}
+
+VCMGenericDecoder::VCMGenericDecoder(VideoDecoder* decoder)
+    : _callback(NULL),
+      decoder_(decoder),
+      _last_keyframe_content_type(VideoContentType::UNSPECIFIED) {
+  RTC_DCHECK(decoder_);
+}
+
+VCMGenericDecoder::~VCMGenericDecoder() {
+  decoder_->Release();
+}
+
+bool VCMGenericDecoder::Configure(const VideoDecoder::Settings& settings) {
+  TRACE_EVENT0("webrtc", "VCMGenericDecoder::Configure");
+
+  bool ok = decoder_->Configure(settings);
+  decoder_info_ = decoder_->GetDecoderInfo();
+  RTC_LOG(LS_INFO) << "Decoder implementation: " << decoder_info_.ToString();
+  if (_callback) {
+    _callback->OnDecoderInfoChanged(decoder_info_);
+  }
+  return ok;
+}
+
+int32_t VCMGenericDecoder::Decode(const VCMEncodedFrame& frame, Timestamp now) {
+  TRACE_EVENT1("webrtc", "VCMGenericDecoder::Decode", "timestamp",
+               frame.Timestamp());
+  FrameInfo frame_info;
+  frame_info.rtp_timestamp = frame.Timestamp();
+  frame_info.decode_start = now;
+  frame_info.render_time =
+      frame.RenderTimeMs() >= 0
+          ? absl::make_optional(Timestamp::Millis(frame.RenderTimeMs()))
+          : absl::nullopt;
+  frame_info.rotation = frame.rotation();
+  frame_info.timing = frame.video_timing();
+  frame_info.ntp_time_ms = frame.EncodedImage().ntp_time_ms_;
+  frame_info.packet_infos = frame.PacketInfos();
+
+  // Set correctly only for key frames. Thus, use latest key frame
+  // content type. If the corresponding key frame was lost, decode will fail
+  // and content type will be ignored.
+  if (frame.FrameType() == VideoFrameType::kVideoFrameKey) {
+    frame_info.content_type = frame.contentType();
+    _last_keyframe_content_type = frame.contentType();
+  } else {
+    frame_info.content_type = _last_keyframe_content_type;
+  }
+  _callback->Map(std::move(frame_info));
+
+  int32_t ret = decoder_->Decode(frame.EncodedImage(), frame.MissingFrame(),
+                                 frame.RenderTimeMs());
+  VideoDecoder::DecoderInfo decoder_info = decoder_->GetDecoderInfo();
+  if (decoder_info != decoder_info_) {
+    RTC_LOG(LS_INFO) << "Changed decoder implementation to: "
+                     << decoder_info.ToString();
+    decoder_info_ = decoder_info;
+    if (decoder_info.implementation_name.empty()) {
+      decoder_info.implementation_name = "unknown";
+    }
+    _callback->OnDecoderInfoChanged(std::move(decoder_info));
+  }
+  if (ret < WEBRTC_VIDEO_CODEC_OK) {
+    RTC_LOG(LS_WARNING) << "Failed to decode frame with timestamp "
+                        << frame.Timestamp() << ", error code: " << ret;
+    _callback->ClearTimestampMap();
+  } else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT) {
+    // No output.
+    _callback->ClearTimestampMap();
+  }
+  return ret;
+}
+
+int32_t VCMGenericDecoder::RegisterDecodeCompleteCallback(
+    VCMDecodedFrameCallback* callback) {
+  _callback = callback;
+  int32_t ret = decoder_->RegisterDecodeCompleteCallback(callback);
+  if (callback && !decoder_info_.implementation_name.empty()) {
+    callback->OnDecoderInfoChanged(decoder_info_);
+  }
+  return ret;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/generic_decoder.h b/third_party/libwebrtc/modules/video_coding/generic_decoder.h
new file mode 100644
index 0000000000..7dc6d34c01
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/generic_decoder.h
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+#define MODULES_VIDEO_CODING_GENERIC_DECODER_H_
+
+#include <cstdint>
+#include <deque>
+#include <string>
+#include <utility>
+
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/encoded_frame.h"
+#include "modules/video_coding/timing/timing.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class VCMReceiveCallback;
+
+struct FrameInfo {
+  FrameInfo() = default;
+  FrameInfo(const FrameInfo&) = delete;
+  FrameInfo& operator=(const FrameInfo&) = delete;
+  FrameInfo(FrameInfo&&) = default;
+  FrameInfo& operator=(FrameInfo&&) = default;
+
+  uint32_t rtp_timestamp;
+  // This is likely not optional, but some inputs seem to sometimes be negative.
+  // TODO(bugs.webrtc.org/13756): See if this can be replaced with Timestamp
+  // once all inputs to this field use Timestamp instead of an integer.
+  absl::optional<Timestamp> render_time;
+  absl::optional<Timestamp> decode_start;
+  VideoRotation rotation;
+  VideoContentType content_type;
+  EncodedImage::Timing timing;
+  int64_t ntp_time_ms;
+  RtpPacketInfos packet_infos;
+  // ColorSpace is not stored here, as it might be modified by decoders.
+};
+
+class VCMDecodedFrameCallback : public DecodedImageCallback {
+ public:
+  VCMDecodedFrameCallback(VCMTiming* timing,
+                          Clock* clock,
+                          const FieldTrialsView& field_trials);
+  ~VCMDecodedFrameCallback() override;
+  void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
+  VCMReceiveCallback* UserReceiveCallback();
+
+  int32_t Decoded(VideoFrame& decodedImage) override;
+  int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) override;
+  void Decoded(VideoFrame& decodedImage,
+               absl::optional<int32_t> decode_time_ms,
+               absl::optional<uint8_t> qp) override;
+
+  void OnDecoderInfoChanged(const VideoDecoder::DecoderInfo& decoder_info);
+
+  void Map(FrameInfo frameInfo);
+  void ClearTimestampMap();
+
+ private:
+  std::pair<absl::optional<FrameInfo>, size_t> FindFrameInfo(
+      uint32_t rtp_timestamp) RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+
+  SequenceChecker construction_thread_;
+  Clock* const _clock;
+  // This callback must be set before the decoder thread starts running
+  // and must only be unset when external threads (e.g decoder thread)
+  // have been stopped. Due to that, the variable should regarded as const
+  // while there are more than one threads involved, it must be set
+  // from the same thread, and therfore a lock is not required to access it.
+  VCMReceiveCallback* _receiveCallback = nullptr;
+  VCMTiming* _timing;
+  Mutex lock_;
+  std::deque<FrameInfo> frame_infos_ RTC_GUARDED_BY(lock_);
+  int64_t ntp_offset_;
+};
+
+class VCMGenericDecoder {
+ public:
+  explicit VCMGenericDecoder(VideoDecoder* decoder);
+  ~VCMGenericDecoder();
+
+  /**
+   * Initialize the decoder with the information from the `settings`
+   */
+  bool Configure(const VideoDecoder::Settings& settings);
+
+  /**
+   * Decode to a raw I420 frame,
+   *
+   * inputVideoBuffer reference to encoded video frame
+   */
+  int32_t Decode(const VCMEncodedFrame& inputFrame, Timestamp now);
+
+  /**
+   * Set decode callback. Deregistering while decoding is illegal.
+   */
+  int32_t RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
+
+  bool IsSameDecoder(VideoDecoder* decoder) const {
+    return decoder_ == decoder;
+  }
+
+ private:
+  VCMDecodedFrameCallback* _callback = nullptr;
+  VideoDecoder* const decoder_;
+  VideoContentType _last_keyframe_content_type;
+  VideoDecoder::DecoderInfo decoder_info_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_GENERIC_DECODER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/generic_decoder_unittest.cc b/third_party/libwebrtc/modules/video_coding/generic_decoder_unittest.cc
new file mode 100644
index 0000000000..68bc307e65
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/generic_decoder_unittest.cc
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/generic_decoder.h"
+
+#include <cstdint>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/rtp_packet_infos.h"
+#include "api/video_codecs/video_decoder.h"
+#include "common_video/test/utilities.h"
+#include "modules/video_coding/timing/timing.h"
+#include "system_wrappers/include/clock.h"
+#include "test/fake_decoder.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class ReceiveCallback : public VCMReceiveCallback {
+ public:
+  int32_t FrameToRender(VideoFrame& frame,
+                        absl::optional<uint8_t> qp,
+                        TimeDelta decode_time,
+                        VideoContentType content_type) override {
+    frames_.push_back(frame);
+    return 0;
+  }
+
+  absl::optional<VideoFrame> PopLastFrame() {
+    if (frames_.empty())
+      return absl::nullopt;
+    auto ret = frames_.front();
+    frames_.pop_back();
+    return ret;
+  }
+
+  rtc::ArrayView<const VideoFrame> GetAllFrames() const { return frames_; }
+
+  void OnDroppedFrames(uint32_t frames_dropped) {
+    frames_dropped_ += frames_dropped;
+  }
+
+  uint32_t frames_dropped() const { return frames_dropped_; }
+
+ private:
+  std::vector<VideoFrame> frames_;
+  uint32_t frames_dropped_ = 0;
+};
+
+class GenericDecoderTest : public ::testing::Test {
+ protected:
+  GenericDecoderTest()
+      : time_controller_(Timestamp::Zero()),
+        clock_(time_controller_.GetClock()),
+        timing_(time_controller_.GetClock(), field_trials_),
+        decoder_(time_controller_.GetTaskQueueFactory()),
+        vcm_callback_(&timing_, time_controller_.GetClock(), field_trials_),
+        generic_decoder_(&decoder_) {}
+
+  void SetUp() override {
+    generic_decoder_.RegisterDecodeCompleteCallback(&vcm_callback_);
+    vcm_callback_.SetUserReceiveCallback(&user_callback_);
+    VideoDecoder::Settings settings;
+    settings.set_codec_type(kVideoCodecVP8);
+    settings.set_max_render_resolution({10, 10});
+    settings.set_number_of_cores(4);
+    generic_decoder_.Configure(settings);
+  }
+
+  GlobalSimulatedTimeController time_controller_;
+  Clock* const clock_;
+  test::ScopedKeyValueConfig field_trials_;
+  VCMTiming timing_;
+  webrtc::test::FakeDecoder decoder_;
+  VCMDecodedFrameCallback vcm_callback_;
+  VCMGenericDecoder generic_decoder_;
+  ReceiveCallback user_callback_;
+};
+
+TEST_F(GenericDecoderTest, PassesPacketInfos) {
+  RtpPacketInfos packet_infos = CreatePacketInfos(3);
+  VCMEncodedFrame encoded_frame;
+  encoded_frame.SetPacketInfos(packet_infos);
+  generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_EQ(decoded_frame->packet_infos().size(), 3U);
+}
+
+TEST_F(GenericDecoderTest, FrameDroppedIfTooManyFramesInFlight) {
+  constexpr int kMaxFramesInFlight = 10;
+  decoder_.SetDelayedDecoding(10);
+  for (int i = 0; i < kMaxFramesInFlight + 1; ++i) {
+    VCMEncodedFrame encoded_frame;
+    encoded_frame.SetTimestamp(90000 * i);
+    generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
+  }
+
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+
+  auto frames = user_callback_.GetAllFrames();
+  ASSERT_EQ(10U, frames.size());
+  // Expect that the first frame was dropped since all decodes released at the
+  // same time and the oldest frame info is the first one dropped.
+  EXPECT_EQ(frames[0].timestamp(), 90000u);
+  EXPECT_EQ(1u, user_callback_.frames_dropped());
+}
+
+TEST_F(GenericDecoderTest, PassesPacketInfosForDelayedDecoders) {
+  RtpPacketInfos packet_infos = CreatePacketInfos(3);
+  decoder_.SetDelayedDecoding(100);
+
+  {
+    // Ensure the original frame is destroyed before the decoding is completed.
+    VCMEncodedFrame encoded_frame;
+    encoded_frame.SetPacketInfos(packet_infos);
+    generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
+  }
+
+  time_controller_.AdvanceTime(TimeDelta::Millis(200));
+  absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_EQ(decoded_frame->packet_infos().size(), 3U);
+}
+
+TEST_F(GenericDecoderTest, MaxCompositionDelayNotSetByDefault) {
+  VCMEncodedFrame encoded_frame;
+  generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_THAT(
+      decoded_frame->render_parameters().max_composition_delay_in_frames,
+      testing::Eq(absl::nullopt));
+}
+
+TEST_F(GenericDecoderTest, MaxCompositionDelayActivatedByPlayoutDelay) {
+  VCMEncodedFrame encoded_frame;
+  // VideoReceiveStream2 would set MaxCompositionDelayInFrames if playout delay
+  // is specified as X,Y, where X=0, Y>0.
+  constexpr int kMaxCompositionDelayInFrames = 3;  // ~50 ms at 60 fps.
+  timing_.SetMaxCompositionDelayInFrames(
+      absl::make_optional(kMaxCompositionDelayInFrames));
+  generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_THAT(
+      decoded_frame->render_parameters().max_composition_delay_in_frames,
+      testing::Optional(kMaxCompositionDelayInFrames));
+}
+
+TEST_F(GenericDecoderTest, IsLowLatencyStreamFalseByDefault) {
+  VCMEncodedFrame encoded_frame;
+  generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_FALSE(decoded_frame->render_parameters().use_low_latency_rendering);
+}
+
+TEST_F(GenericDecoderTest, IsLowLatencyStreamActivatedByPlayoutDelay) {
+  VCMEncodedFrame encoded_frame;
+  const VideoPlayoutDelay kPlayoutDelay = {0, 50};
+  timing_.set_min_playout_delay(TimeDelta::Millis(kPlayoutDelay.min_ms));
+  timing_.set_max_playout_delay(TimeDelta::Millis(kPlayoutDelay.max_ms));
+  generic_decoder_.Decode(encoded_frame, clock_->CurrentTime());
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  absl::optional<VideoFrame> decoded_frame = user_callback_.PopLastFrame();
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_TRUE(decoded_frame->render_parameters().use_low_latency_rendering);
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/h264_packet_buffer.cc b/third_party/libwebrtc/modules/video_coding/h264_packet_buffer.cc
new file mode 100644
index 0000000000..6096665bda
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/h264_packet_buffer.cc
@@ -0,0 +1,287 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/h264_packet_buffer.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/rtp_packet_info.h"
+#include "api/video/video_frame_type.h"
+#include "common_video/h264/h264_common.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+
+namespace webrtc {
+namespace {
+int64_t EuclideanMod(int64_t n, int64_t div) {
+  RTC_DCHECK_GT(div, 0);
+  return (n %= div) < 0 ? n + div : n;
+}
+
+rtc::ArrayView<const NaluInfo> GetNaluInfos(
+    const RTPVideoHeaderH264& h264_header) {
+  if (h264_header.nalus_length > kMaxNalusPerPacket) {
+    return {};
+  }
+
+  return rtc::MakeArrayView(h264_header.nalus, h264_header.nalus_length);
+}
+
+bool IsFirstPacketOfFragment(const RTPVideoHeaderH264& h264_header) {
+  return h264_header.nalus_length > 0;
+}
+
+bool BeginningOfIdr(const H264PacketBuffer::Packet& packet) {
+  const auto& h264_header =
+      absl::get<RTPVideoHeaderH264>(packet.video_header.video_type_header);
+  const bool contains_idr_nalu =
+      absl::c_any_of(GetNaluInfos(h264_header), [](const auto& nalu_info) {
+        return nalu_info.type == H264::NaluType::kIdr;
+      });
+  switch (h264_header.packetization_type) {
+    case kH264StapA:
+    case kH264SingleNalu: {
+      return contains_idr_nalu;
+    }
+    case kH264FuA: {
+      return contains_idr_nalu && IsFirstPacketOfFragment(h264_header);
+    }
+  }
+}
+
+bool HasSps(const H264PacketBuffer::Packet& packet) {
+  auto& h264_header =
+      absl::get<RTPVideoHeaderH264>(packet.video_header.video_type_header);
+  return absl::c_any_of(GetNaluInfos(h264_header), [](const auto& nalu_info) {
+    return nalu_info.type == H264::NaluType::kSps;
+  });
+}
+
+// TODO(bugs.webrtc.org/13157): Update the H264 depacketizer so we don't have to
+//                              fiddle with the payload at this point.
+rtc::CopyOnWriteBuffer FixVideoPayload(rtc::ArrayView<const uint8_t> payload,
+                                       const RTPVideoHeader& video_header) {
+  constexpr uint8_t kStartCode[] = {0, 0, 0, 1};
+
+  const auto& h264_header =
+      absl::get<RTPVideoHeaderH264>(video_header.video_type_header);
+
+  rtc::CopyOnWriteBuffer result;
+  switch (h264_header.packetization_type) {
+    case kH264StapA: {
+      const uint8_t* payload_end = payload.data() + payload.size();
+      const uint8_t* nalu_ptr = payload.data() + 1;
+      while (nalu_ptr < payload_end - 1) {
+        // The first two bytes describe the length of the segment, where a
+        // segment is the nalu type plus nalu payload.
+        uint16_t segment_length = nalu_ptr[0] << 8 | nalu_ptr[1];
+        nalu_ptr += 2;
+
+        if (nalu_ptr + segment_length <= payload_end) {
+          result.AppendData(kStartCode);
+          result.AppendData(nalu_ptr, segment_length);
+        }
+        nalu_ptr += segment_length;
+      }
+      return result;
+    }
+
+    case kH264FuA: {
+      if (IsFirstPacketOfFragment(h264_header)) {
+        result.AppendData(kStartCode);
+      }
+      result.AppendData(payload);
+      return result;
+    }
+
+    case kH264SingleNalu: {
+      result.AppendData(kStartCode);
+      result.AppendData(payload);
+      return result;
+    }
+  }
+
+  RTC_DCHECK_NOTREACHED();
+  return result;
+}
+
+}  // namespace
+
+H264PacketBuffer::H264PacketBuffer(bool idr_only_keyframes_allowed)
+    : idr_only_keyframes_allowed_(idr_only_keyframes_allowed) {}
+
+H264PacketBuffer::InsertResult H264PacketBuffer::InsertPacket(
+    std::unique_ptr<Packet> packet) {
+  RTC_DCHECK(packet->video_header.codec == kVideoCodecH264);
+
+  InsertResult result;
+  if (!absl::holds_alternative<RTPVideoHeaderH264>(
+          packet->video_header.video_type_header)) {
+    return result;
+  }
+
+  int64_t unwrapped_seq_num = seq_num_unwrapper_.Unwrap(packet->seq_num);
+  auto& packet_slot = GetPacket(unwrapped_seq_num);
+  if (packet_slot != nullptr &&
+      AheadOrAt(packet_slot->timestamp, packet->timestamp)) {
+    // The incoming `packet` is old or a duplicate.
+    return result;
+  } else {
+    packet_slot = std::move(packet);
+  }
+
+  result.packets = FindFrames(unwrapped_seq_num);
+  return result;
+}
+
+std::unique_ptr<H264PacketBuffer::Packet>& H264PacketBuffer::GetPacket(
+    int64_t unwrapped_seq_num) {
+  return buffer_[EuclideanMod(unwrapped_seq_num, kBufferSize)];
+}
+
+bool H264PacketBuffer::BeginningOfStream(
+    const H264PacketBuffer::Packet& packet) const {
+  return HasSps(packet) ||
+         (idr_only_keyframes_allowed_ && BeginningOfIdr(packet));
+}
+
+std::vector<std::unique_ptr<H264PacketBuffer::Packet>>
+H264PacketBuffer::FindFrames(int64_t unwrapped_seq_num) {
+  std::vector<std::unique_ptr<Packet>> found_frames;
+
+  Packet* packet = GetPacket(unwrapped_seq_num).get();
+  RTC_CHECK(packet != nullptr);
+
+  // Check if the packet is continuous or the beginning of a new coded video
+  // sequence.
+  if (unwrapped_seq_num - 1 != last_continuous_unwrapped_seq_num_) {
+    if (unwrapped_seq_num <= last_continuous_unwrapped_seq_num_ ||
+        !BeginningOfStream(*packet)) {
+      return found_frames;
+    }
+
+    last_continuous_unwrapped_seq_num_ = unwrapped_seq_num;
+  }
+
+  for (int64_t seq_num = unwrapped_seq_num;
+       seq_num < unwrapped_seq_num + kBufferSize;) {
+    RTC_DCHECK_GE(seq_num, *last_continuous_unwrapped_seq_num_);
+
+    // Packets that were never assembled into a completed frame will stay in
+    // the 'buffer_'. Check that the `packet` sequence number match the expected
+    // unwrapped sequence number.
+    if (static_cast<uint16_t>(seq_num) != packet->seq_num) {
+      return found_frames;
+    }
+
+    last_continuous_unwrapped_seq_num_ = seq_num;
+    // Last packet of the frame, try to assemble the frame.
+    if (packet->marker_bit) {
+      uint32_t rtp_timestamp = packet->timestamp;
+
+      // Iterate backwards to find where the frame starts.
+      for (int64_t seq_num_start = seq_num;
+           seq_num_start > seq_num - kBufferSize; --seq_num_start) {
+        auto& prev_packet = GetPacket(seq_num_start - 1);
+
+        if (prev_packet == nullptr || prev_packet->timestamp != rtp_timestamp) {
+          if (MaybeAssembleFrame(seq_num_start, seq_num, found_frames)) {
+            // Frame was assembled, continue to look for more frames.
+            break;
+          } else {
+            // Frame was not assembled, no subsequent frame will be continuous.
+            return found_frames;
+          }
+        }
+      }
+    }
+
+    seq_num++;
+    packet = GetPacket(seq_num).get();
+    if (packet == nullptr) {
+      return found_frames;
+    }
+  }
+
+  return found_frames;
+}
+
+bool H264PacketBuffer::MaybeAssembleFrame(
+    int64_t start_seq_num_unwrapped,
+    int64_t end_sequence_number_unwrapped,
+    std::vector<std::unique_ptr<Packet>>& frames) {
+  bool has_sps = false;
+  bool has_pps = false;
+  bool has_idr = false;
+
+  int width = -1;
+  int height = -1;
+
+  for (int64_t seq_num = start_seq_num_unwrapped;
+       seq_num <= end_sequence_number_unwrapped; ++seq_num) {
+    const auto& packet = GetPacket(seq_num);
+    const auto& h264_header =
+        absl::get<RTPVideoHeaderH264>(packet->video_header.video_type_header);
+    for (const auto& nalu : GetNaluInfos(h264_header)) {
+      has_idr |= nalu.type == H264::NaluType::kIdr;
+      has_sps |= nalu.type == H264::NaluType::kSps;
+      has_pps |= nalu.type == H264::NaluType::kPps;
+    }
+
+    width = std::max<int>(packet->video_header.width, width);
+    height = std::max<int>(packet->video_header.height, height);
+  }
+
+  if (has_idr) {
+    if (!idr_only_keyframes_allowed_ && (!has_sps || !has_pps)) {
+      return false;
+    }
+  }
+
+  for (int64_t seq_num = start_seq_num_unwrapped;
+       seq_num <= end_sequence_number_unwrapped; ++seq_num) {
+    auto& packet = GetPacket(seq_num);
+
+    packet->video_header.is_first_packet_in_frame =
+        (seq_num == start_seq_num_unwrapped);
+    packet->video_header.is_last_packet_in_frame =
+        (seq_num == end_sequence_number_unwrapped);
+
+    if (packet->video_header.is_first_packet_in_frame) {
+      if (width > 0 && height > 0) {
+        packet->video_header.width = width;
+        packet->video_header.height = height;
+      }
+
+      packet->video_header.frame_type = has_idr
+                                            ? VideoFrameType::kVideoFrameKey
+                                            : VideoFrameType::kVideoFrameDelta;
+    }
+
+    packet->video_payload =
+        FixVideoPayload(packet->video_payload, packet->video_header);
+
+    frames.push_back(std::move(packet));
+  }
+
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/h264_packet_buffer.h b/third_party/libwebrtc/modules/video_coding/h264_packet_buffer.h
new file mode 100644
index 0000000000..a72c240e82
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/h264_packet_buffer.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_H264_PACKET_BUFFER_H_
+#define MODULES_VIDEO_CODING_H264_PACKET_BUFFER_H_
+
+#include <array>
+#include <memory>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "modules/video_coding/packet_buffer.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+class H264PacketBuffer {
+ public:
+  // The H264PacketBuffer does the same job as the PacketBuffer but for H264
+  // only. To make it fit in with surronding code the PacketBuffer input/output
+  // classes are used.
+  using Packet = video_coding::PacketBuffer::Packet;
+  using InsertResult = video_coding::PacketBuffer::InsertResult;
+
+  explicit H264PacketBuffer(bool idr_only_keyframes_allowed);
+
+  ABSL_MUST_USE_RESULT InsertResult
+  InsertPacket(std::unique_ptr<Packet> packet);
+
+ private:
+  static constexpr int kBufferSize = 2048;
+
+  std::unique_ptr<Packet>& GetPacket(int64_t unwrapped_seq_num);
+  bool BeginningOfStream(const Packet& packet) const;
+  std::vector<std::unique_ptr<Packet>> FindFrames(int64_t unwrapped_seq_num);
+  bool MaybeAssembleFrame(int64_t start_seq_num_unwrapped,
+                          int64_t end_sequence_number_unwrapped,
+                          std::vector<std::unique_ptr<Packet>>& packets);
+
+  const bool idr_only_keyframes_allowed_;
+  std::array<std::unique_ptr<Packet>, kBufferSize> buffer_;
+  absl::optional<int64_t> last_continuous_unwrapped_seq_num_;
+  SeqNumUnwrapper<uint16_t> seq_num_unwrapper_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_H264_PACKET_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/h264_packet_buffer_unittest.cc b/third_party/libwebrtc/modules/video_coding/h264_packet_buffer_unittest.cc
new file mode 100644
index 0000000000..4f2331da28
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/h264_packet_buffer_unittest.cc
@@ -0,0 +1,778 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/h264_packet_buffer.h"
+
+#include <cstring>
+#include <limits>
+#include <ostream>
+#include <string>
+#include <utility>
+
+#include "api/array_view.h"
+#include "api/video/render_resolution.h"
+#include "common_video/h264/h264_common.h"
+#include "rtc_base/system/unused.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::IsEmpty;
+using ::testing::SizeIs;
+
+using H264::NaluType::kAud;
+using H264::NaluType::kFuA;
+using H264::NaluType::kIdr;
+using H264::NaluType::kPps;
+using H264::NaluType::kSlice;
+using H264::NaluType::kSps;
+using H264::NaluType::kStapA;
+
+constexpr int kBufferSize = 2048;
+
+std::vector<uint8_t> StartCode() {
+  return {0, 0, 0, 1};
+}
+
+NaluInfo MakeNaluInfo(uint8_t type) {
+  NaluInfo res;
+  res.type = type;
+  res.sps_id = -1;
+  res.pps_id = -1;
+  return res;
+}
+
+class Packet {
+ public:
+  explicit Packet(H264PacketizationTypes type);
+
+  Packet& Idr(std::vector<uint8_t> payload = {9, 9, 9});
+  Packet& Slice(std::vector<uint8_t> payload = {9, 9, 9});
+  Packet& Sps(std::vector<uint8_t> payload = {9, 9, 9});
+  Packet& SpsWithResolution(RenderResolution resolution,
+                            std::vector<uint8_t> payload = {9, 9, 9});
+  Packet& Pps(std::vector<uint8_t> payload = {9, 9, 9});
+  Packet& Aud();
+  Packet& Marker();
+  Packet& AsFirstFragment();
+  Packet& Time(uint32_t rtp_timestamp);
+  Packet& SeqNum(uint16_t rtp_seq_num);
+
+  std::unique_ptr<H264PacketBuffer::Packet> Build();
+
+ private:
+  rtc::CopyOnWriteBuffer BuildFuaPayload() const;
+  rtc::CopyOnWriteBuffer BuildSingleNaluPayload() const;
+  rtc::CopyOnWriteBuffer BuildStapAPayload() const;
+
+  RTPVideoHeaderH264& H264Header() {
+    return absl::get<RTPVideoHeaderH264>(video_header_.video_type_header);
+  }
+  const RTPVideoHeaderH264& H264Header() const {
+    return absl::get<RTPVideoHeaderH264>(video_header_.video_type_header);
+  }
+
+  H264PacketizationTypes type_;
+  RTPVideoHeader video_header_;
+  bool first_fragment_ = false;
+  bool marker_bit_ = false;
+  uint32_t rtp_timestamp_ = 0;
+  uint16_t rtp_seq_num_ = 0;
+  std::vector<std::vector<uint8_t>> nalu_payloads_;
+};
+
+Packet::Packet(H264PacketizationTypes type) : type_(type) {
+  video_header_.video_type_header.emplace<RTPVideoHeaderH264>();
+}
+
+Packet& Packet::Idr(std::vector<uint8_t> payload) {
+  auto& h264_header = H264Header();
+  h264_header.nalus[h264_header.nalus_length++] = MakeNaluInfo(kIdr);
+  nalu_payloads_.push_back(std::move(payload));
+  return *this;
+}
+
+Packet& Packet::Slice(std::vector<uint8_t> payload) {
+  auto& h264_header = H264Header();
+  h264_header.nalus[h264_header.nalus_length++] = MakeNaluInfo(kSlice);
+  nalu_payloads_.push_back(std::move(payload));
+  return *this;
+}
+
+Packet& Packet::Sps(std::vector<uint8_t> payload) {
+  auto& h264_header = H264Header();
+  h264_header.nalus[h264_header.nalus_length++] = MakeNaluInfo(kSps);
+  nalu_payloads_.push_back(std::move(payload));
+  return *this;
+}
+
+Packet& Packet::SpsWithResolution(RenderResolution resolution,
+                                  std::vector<uint8_t> payload) {
+  auto& h264_header = H264Header();
+  h264_header.nalus[h264_header.nalus_length++] = MakeNaluInfo(kSps);
+  video_header_.width = resolution.Width();
+  video_header_.height = resolution.Height();
+  nalu_payloads_.push_back(std::move(payload));
+  return *this;
+}
+
+Packet& Packet::Pps(std::vector<uint8_t> payload) {
+  auto& h264_header = H264Header();
+  h264_header.nalus[h264_header.nalus_length++] = MakeNaluInfo(kPps);
+  nalu_payloads_.push_back(std::move(payload));
+  return *this;
+}
+
+Packet& Packet::Aud() {
+  auto& h264_header = H264Header();
+  h264_header.nalus[h264_header.nalus_length++] = MakeNaluInfo(kAud);
+  nalu_payloads_.push_back({});
+  return *this;
+}
+
+Packet& Packet::Marker() {
+  marker_bit_ = true;
+  return *this;
+}
+
+Packet& Packet::AsFirstFragment() {
+  first_fragment_ = true;
+  return *this;
+}
+
+Packet& Packet::Time(uint32_t rtp_timestamp) {
+  rtp_timestamp_ = rtp_timestamp;
+  return *this;
+}
+
+Packet& Packet::SeqNum(uint16_t rtp_seq_num) {
+  rtp_seq_num_ = rtp_seq_num;
+  return *this;
+}
+
+std::unique_ptr<H264PacketBuffer::Packet> Packet::Build() {
+  auto res = std::make_unique<H264PacketBuffer::Packet>();
+
+  auto& h264_header = H264Header();
+  switch (type_) {
+    case kH264FuA: {
+      RTC_CHECK_EQ(h264_header.nalus_length, 1);
+      res->video_payload = BuildFuaPayload();
+      break;
+    }
+    case kH264SingleNalu: {
+      RTC_CHECK_EQ(h264_header.nalus_length, 1);
+      res->video_payload = BuildSingleNaluPayload();
+      break;
+    }
+    case kH264StapA: {
+      RTC_CHECK_GT(h264_header.nalus_length, 1);
+      RTC_CHECK_LE(h264_header.nalus_length, kMaxNalusPerPacket);
+      res->video_payload = BuildStapAPayload();
+      break;
+    }
+  }
+
+  if (type_ == kH264FuA && !first_fragment_) {
+    h264_header.nalus_length = 0;
+  }
+
+  h264_header.packetization_type = type_;
+  res->marker_bit = marker_bit_;
+  res->video_header = video_header_;
+  res->timestamp = rtp_timestamp_;
+  res->seq_num = rtp_seq_num_;
+  res->video_header.codec = kVideoCodecH264;
+
+  return res;
+}
+
+rtc::CopyOnWriteBuffer Packet::BuildFuaPayload() const {
+  return rtc::CopyOnWriteBuffer(nalu_payloads_[0]);
+}
+
+rtc::CopyOnWriteBuffer Packet::BuildSingleNaluPayload() const {
+  rtc::CopyOnWriteBuffer res;
+  auto& h264_header = H264Header();
+  res.AppendData(&h264_header.nalus[0].type, 1);
+  res.AppendData(nalu_payloads_[0]);
+  return res;
+}
+
+rtc::CopyOnWriteBuffer Packet::BuildStapAPayload() const {
+  rtc::CopyOnWriteBuffer res;
+
+  const uint8_t indicator = H264::NaluType::kStapA;
+  res.AppendData(&indicator, 1);
+
+  auto& h264_header = H264Header();
+  for (size_t i = 0; i < h264_header.nalus_length; ++i) {
+    // The two first bytes indicates the nalu segment size.
+    uint8_t length_as_array[2] = {
+        0, static_cast<uint8_t>(nalu_payloads_[i].size() + 1)};
+    res.AppendData(length_as_array);
+
+    res.AppendData(&h264_header.nalus[i].type, 1);
+    res.AppendData(nalu_payloads_[i]);
+  }
+  return res;
+}
+
+rtc::ArrayView<const uint8_t> PacketPayload(
+    const std::unique_ptr<H264PacketBuffer::Packet>& packet) {
+  return packet->video_payload;
+}
+
+std::vector<uint8_t> FlatVector(
+    const std::vector<std::vector<uint8_t>>& elems) {
+  std::vector<uint8_t> res;
+  for (const auto& elem : elems) {
+    res.insert(res.end(), elem.begin(), elem.end());
+  }
+  return res;
+}
+
+TEST(H264PacketBufferTest, IdrIsKeyframe) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/true);
+
+  EXPECT_THAT(
+      packet_buffer.InsertPacket(Packet(kH264SingleNalu).Idr().Marker().Build())
+          .packets,
+      SizeIs(1));
+}
+
+TEST(H264PacketBufferTest, IdrIsNotKeyframe) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  EXPECT_THAT(
+      packet_buffer.InsertPacket(Packet(kH264SingleNalu).Idr().Marker().Build())
+          .packets,
+      IsEmpty());
+}
+
+TEST(H264PacketBufferTest, IdrIsKeyframeFuaRequiresFirstFragmet) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/true);
+
+  // Not marked as the first fragment
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(Packet(kH264FuA).Idr().SeqNum(0).Time(0).Build())
+          .packets,
+      IsEmpty());
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(
+                      Packet(kH264FuA).Idr().SeqNum(1).Time(0).Marker().Build())
+                  .packets,
+              IsEmpty());
+
+  // Marked as first fragment
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264FuA)
+                                    .Idr()
+                                    .SeqNum(2)
+                                    .Time(1)
+                                    .AsFirstFragment()
+                                    .Build())
+                  .packets,
+              IsEmpty());
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(
+                      Packet(kH264FuA).Idr().SeqNum(3).Time(1).Marker().Build())
+                  .packets,
+              SizeIs(2));
+}
+
+TEST(H264PacketBufferTest, SpsPpsIdrIsKeyframeSingleNalus) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Sps().SeqNum(0).Time(0).Build()));
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Pps().SeqNum(1).Time(0).Build()));
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(
+              Packet(kH264SingleNalu).Idr().SeqNum(2).Time(0).Marker().Build())
+          .packets,
+      SizeIs(3));
+}
+
+TEST(H264PacketBufferTest, PpsIdrIsNotKeyframeSingleNalus) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Pps().SeqNum(0).Time(0).Build()));
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(
+              Packet(kH264SingleNalu).Idr().SeqNum(1).Time(0).Marker().Build())
+          .packets,
+      IsEmpty());
+}
+
+TEST(H264PacketBufferTest, SpsIdrIsNotKeyframeSingleNalus) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Sps().SeqNum(0).Time(0).Build()));
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(
+              Packet(kH264SingleNalu).Idr().SeqNum(1).Time(0).Marker().Build())
+          .packets,
+      IsEmpty());
+}
+
+TEST(H264PacketBufferTest, SpsPpsIdrIsKeyframeStapA) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264StapA)
+                                    .Sps()
+                                    .Pps()
+                                    .Idr()
+                                    .SeqNum(0)
+                                    .Time(0)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              SizeIs(1));
+}
+
+TEST(H264PacketBufferTest, PpsIdrIsNotKeyframeStapA) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(
+              Packet(kH264StapA).Pps().Idr().SeqNum(0).Time(0).Marker().Build())
+          .packets,
+      IsEmpty());
+}
+
+TEST(H264PacketBufferTest, SpsIdrIsNotKeyframeStapA) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(
+              Packet(kH264StapA).Sps().Idr().SeqNum(2).Time(2).Marker().Build())
+          .packets,
+      IsEmpty());
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264StapA)
+                                    .Sps()
+                                    .Pps()
+                                    .Idr()
+                                    .SeqNum(3)
+                                    .Time(3)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              SizeIs(1));
+}
+
+TEST(H264PacketBufferTest, InsertingSpsPpsLastCompletesKeyframe) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Idr().SeqNum(2).Time(1).Marker().Build()));
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(
+                      Packet(kH264StapA).Sps().Pps().SeqNum(1).Time(1).Build())
+                  .packets,
+              SizeIs(2));
+}
+
+TEST(H264PacketBufferTest, InsertingMidFuaCompletesFrame) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264StapA)
+                                    .Sps()
+                                    .Pps()
+                                    .Idr()
+                                    .SeqNum(0)
+                                    .Time(0)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              SizeIs(1));
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264FuA).Slice().SeqNum(1).Time(1).AsFirstFragment().Build()));
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264FuA).Slice().SeqNum(3).Time(1).Marker().Build()));
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(Packet(kH264FuA).Slice().SeqNum(2).Time(1).Build())
+          .packets,
+      SizeIs(3));
+}
+
+TEST(H264PacketBufferTest, SeqNumJumpDoesNotCompleteFrame) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264StapA)
+                                    .Sps()
+                                    .Pps()
+                                    .Idr()
+                                    .SeqNum(0)
+                                    .Time(0)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              SizeIs(1));
+
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(Packet(kH264FuA).Slice().SeqNum(1).Time(1).Build())
+          .packets,
+      IsEmpty());
+
+  // Add `kBufferSize` to make the index of the sequence number wrap and end up
+  // where the packet with sequence number 2 would have ended up.
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264FuA)
+                                    .Slice()
+                                    .SeqNum(2 + kBufferSize)
+                                    .Time(3)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              IsEmpty());
+}
+
+TEST(H264PacketBufferTest, OldFramesAreNotCompletedAfterBufferWrap) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264SingleNalu)
+                                    .Slice()
+                                    .SeqNum(1)
+                                    .Time(1)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              IsEmpty());
+
+  // New keyframe, preceedes packet with sequence number 1 in the buffer.
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264StapA)
+                                    .Sps()
+                                    .Pps()
+                                    .Idr()
+                                    .SeqNum(kBufferSize)
+                                    .Time(kBufferSize)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              SizeIs(1));
+}
+
+TEST(H264PacketBufferTest, OldPacketsDontBlockNewPackets) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264StapA)
+                                    .Sps()
+                                    .Pps()
+                                    .Idr()
+                                    .SeqNum(kBufferSize)
+                                    .Time(kBufferSize)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              SizeIs(1));
+
+  RTC_UNUSED(packet_buffer.InsertPacket(Packet(kH264FuA)
+                                            .Slice()
+                                            .SeqNum(kBufferSize + 1)
+                                            .Time(kBufferSize + 1)
+                                            .AsFirstFragment()
+                                            .Build()));
+
+  RTC_UNUSED(packet_buffer.InsertPacket(Packet(kH264FuA)
+                                            .Slice()
+                                            .SeqNum(kBufferSize + 3)
+                                            .Time(kBufferSize + 1)
+                                            .Marker()
+                                            .Build()));
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(Packet(kH264FuA).Slice().SeqNum(2).Time(2).Build())
+          .packets,
+      IsEmpty());
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264FuA)
+                                    .Slice()
+                                    .SeqNum(kBufferSize + 2)
+                                    .Time(kBufferSize + 1)
+                                    .Build())
+                  .packets,
+              SizeIs(3));
+}
+
+TEST(H264PacketBufferTest, OldPacketDoesntCompleteFrame) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264StapA)
+                                    .Sps()
+                                    .Pps()
+                                    .Idr()
+                                    .SeqNum(kBufferSize)
+                                    .Time(kBufferSize)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              SizeIs(1));
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264FuA)
+                                    .Slice()
+                                    .SeqNum(kBufferSize + 3)
+                                    .Time(kBufferSize + 1)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              IsEmpty());
+
+  EXPECT_THAT(
+      packet_buffer
+          .InsertPacket(
+              Packet(kH264FuA).Slice().SeqNum(2).Time(2).Marker().Build())
+          .packets,
+      IsEmpty());
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264FuA)
+                                    .Slice()
+                                    .SeqNum(kBufferSize + 1)
+                                    .Time(kBufferSize + 1)
+                                    .AsFirstFragment()
+                                    .Build())
+                  .packets,
+              IsEmpty());
+}
+
+TEST(H264PacketBufferTest, FrameBoundariesAreSet) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  auto key = packet_buffer.InsertPacket(
+      Packet(kH264StapA).Sps().Pps().Idr().SeqNum(1).Time(1).Marker().Build());
+
+  ASSERT_THAT(key.packets, SizeIs(1));
+  EXPECT_TRUE(key.packets[0]->video_header.is_first_packet_in_frame);
+  EXPECT_TRUE(key.packets[0]->video_header.is_last_packet_in_frame);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264FuA).Slice().SeqNum(2).Time(2).Build()));
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264FuA).Slice().SeqNum(3).Time(2).Build()));
+  auto delta = packet_buffer.InsertPacket(
+      Packet(kH264FuA).Slice().SeqNum(4).Time(2).Marker().Build());
+
+  ASSERT_THAT(delta.packets, SizeIs(3));
+  EXPECT_TRUE(delta.packets[0]->video_header.is_first_packet_in_frame);
+  EXPECT_FALSE(delta.packets[0]->video_header.is_last_packet_in_frame);
+
+  EXPECT_FALSE(delta.packets[1]->video_header.is_first_packet_in_frame);
+  EXPECT_FALSE(delta.packets[1]->video_header.is_last_packet_in_frame);
+
+  EXPECT_FALSE(delta.packets[2]->video_header.is_first_packet_in_frame);
+  EXPECT_TRUE(delta.packets[2]->video_header.is_last_packet_in_frame);
+}
+
+TEST(H264PacketBufferTest, ResolutionSetOnFirstPacket) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Aud().SeqNum(1).Time(1).Build()));
+  auto res = packet_buffer.InsertPacket(Packet(kH264StapA)
+                                            .SpsWithResolution({320, 240})
+                                            .Pps()
+                                            .Idr()
+                                            .SeqNum(2)
+                                            .Time(1)
+                                            .Marker()
+                                            .Build());
+
+  ASSERT_THAT(res.packets, SizeIs(2));
+  EXPECT_THAT(res.packets[0]->video_header.width, Eq(320));
+  EXPECT_THAT(res.packets[0]->video_header.height, Eq(240));
+}
+
+TEST(H264PacketBufferTest, KeyframeAndDeltaFrameSetOnFirstPacket) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Aud().SeqNum(1).Time(1).Build()));
+  auto key = packet_buffer.InsertPacket(
+      Packet(kH264StapA).Sps().Pps().Idr().SeqNum(2).Time(1).Marker().Build());
+
+  auto delta = packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Slice().SeqNum(3).Time(2).Marker().Build());
+
+  ASSERT_THAT(key.packets, SizeIs(2));
+  EXPECT_THAT(key.packets[0]->video_header.frame_type,
+              Eq(VideoFrameType::kVideoFrameKey));
+  ASSERT_THAT(delta.packets, SizeIs(1));
+  EXPECT_THAT(delta.packets[0]->video_header.frame_type,
+              Eq(VideoFrameType::kVideoFrameDelta));
+}
+
+TEST(H264PacketBufferTest, RtpSeqNumWrap) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264StapA).Sps().Pps().SeqNum(0xffff).Time(0).Build()));
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264FuA).Idr().SeqNum(0).Time(0).Build()));
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(
+                      Packet(kH264FuA).Idr().SeqNum(1).Time(0).Marker().Build())
+                  .packets,
+              SizeIs(3));
+}
+
+TEST(H264PacketBufferTest, StapAFixedBitstream) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  auto packets = packet_buffer
+                     .InsertPacket(Packet(kH264StapA)
+                                       .Sps({1, 2, 3})
+                                       .Pps({4, 5, 6})
+                                       .Idr({7, 8, 9})
+                                       .SeqNum(0)
+                                       .Time(0)
+                                       .Marker()
+                                       .Build())
+                     .packets;
+
+  ASSERT_THAT(packets, SizeIs(1));
+  EXPECT_THAT(PacketPayload(packets[0]),
+              ElementsAreArray(FlatVector({StartCode(),
+                                           {kSps, 1, 2, 3},
+                                           StartCode(),
+                                           {kPps, 4, 5, 6},
+                                           StartCode(),
+                                           {kIdr, 7, 8, 9}})));
+}
+
+TEST(H264PacketBufferTest, SingleNaluFixedBitstream) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Sps({1, 2, 3}).SeqNum(0).Time(0).Build()));
+  RTC_UNUSED(packet_buffer.InsertPacket(
+      Packet(kH264SingleNalu).Pps({4, 5, 6}).SeqNum(1).Time(0).Build()));
+  auto packets = packet_buffer
+                     .InsertPacket(Packet(kH264SingleNalu)
+                                       .Idr({7, 8, 9})
+                                       .SeqNum(2)
+                                       .Time(0)
+                                       .Marker()
+                                       .Build())
+                     .packets;
+
+  ASSERT_THAT(packets, SizeIs(3));
+  EXPECT_THAT(PacketPayload(packets[0]),
+              ElementsAreArray(FlatVector({StartCode(), {kSps, 1, 2, 3}})));
+  EXPECT_THAT(PacketPayload(packets[1]),
+              ElementsAreArray(FlatVector({StartCode(), {kPps, 4, 5, 6}})));
+  EXPECT_THAT(PacketPayload(packets[2]),
+              ElementsAreArray(FlatVector({StartCode(), {kIdr, 7, 8, 9}})));
+}
+
+TEST(H264PacketBufferTest, StapaAndFuaFixedBitstream) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  RTC_UNUSED(packet_buffer.InsertPacket(Packet(kH264StapA)
+                                            .Sps({1, 2, 3})
+                                            .Pps({4, 5, 6})
+                                            .SeqNum(0)
+                                            .Time(0)
+                                            .Build()));
+  RTC_UNUSED(packet_buffer.InsertPacket(Packet(kH264FuA)
+                                            .Idr({8, 8, 8})
+                                            .SeqNum(1)
+                                            .Time(0)
+                                            .AsFirstFragment()
+                                            .Build()));
+  auto packets = packet_buffer
+                     .InsertPacket(Packet(kH264FuA)
+                                       .Idr({9, 9, 9})
+                                       .SeqNum(2)
+                                       .Time(0)
+                                       .Marker()
+                                       .Build())
+                     .packets;
+
+  ASSERT_THAT(packets, SizeIs(3));
+  EXPECT_THAT(
+      PacketPayload(packets[0]),
+      ElementsAreArray(FlatVector(
+          {StartCode(), {kSps, 1, 2, 3}, StartCode(), {kPps, 4, 5, 6}})));
+  EXPECT_THAT(PacketPayload(packets[1]),
+              ElementsAreArray(FlatVector({StartCode(), {8, 8, 8}})));
+  // Third is a continuation of second, so only the payload is expected.
+  EXPECT_THAT(PacketPayload(packets[2]),
+              ElementsAreArray(FlatVector({{9, 9, 9}})));
+}
+
+TEST(H264PacketBufferTest, FullPacketBufferDoesNotBlockKeyframe) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  for (int i = 0; i < kBufferSize; ++i) {
+    EXPECT_THAT(
+        packet_buffer
+            .InsertPacket(
+                Packet(kH264SingleNalu).Slice().SeqNum(i).Time(0).Build())
+            .packets,
+        IsEmpty());
+  }
+
+  EXPECT_THAT(packet_buffer
+                  .InsertPacket(Packet(kH264StapA)
+                                    .Sps()
+                                    .Pps()
+                                    .Idr()
+                                    .SeqNum(kBufferSize)
+                                    .Time(1)
+                                    .Marker()
+                                    .Build())
+                  .packets,
+              SizeIs(1));
+}
+
+TEST(H264PacketBufferTest, TooManyNalusInPacket) {
+  H264PacketBuffer packet_buffer(/*allow_idr_only_keyframes=*/false);
+
+  std::unique_ptr<H264PacketBuffer::Packet> packet(
+      Packet(kH264StapA).Sps().Pps().Idr().SeqNum(1).Time(1).Marker().Build());
+  auto& h264_header =
+      absl::get<RTPVideoHeaderH264>(packet->video_header.video_type_header);
+  h264_header.nalus_length = kMaxNalusPerPacket + 1;
+
+  EXPECT_THAT(packet_buffer.InsertPacket(std::move(packet)).packets, IsEmpty());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets.cc b/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets.cc
new file mode 100644
index 0000000000..a64f8885da
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/h264_sprop_parameter_sets.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "rtc_base/logging.h"
+#include "rtc_base/third_party/base64/base64.h"
+
+namespace {
+
+bool DecodeAndConvert(const std::string& base64, std::vector<uint8_t>* binary) {
+  return rtc::Base64::DecodeFromArray(base64.data(), base64.size(),
+                                      rtc::Base64::DO_STRICT, binary, nullptr);
+}
+}  // namespace
+
+namespace webrtc {
+
+bool H264SpropParameterSets::DecodeSprop(const std::string& sprop) {
+  size_t separator_pos = sprop.find(',');
+  RTC_LOG(LS_INFO) << "Parsing sprop \"" << sprop << "\"";
+  if ((separator_pos <= 0) || (separator_pos >= sprop.length() - 1)) {
+    RTC_LOG(LS_WARNING) << "Invalid seperator position " << separator_pos
+                        << " *" << sprop << "*";
+    return false;
+  }
+  std::string sps_str = sprop.substr(0, separator_pos);
+  std::string pps_str = sprop.substr(separator_pos + 1, std::string::npos);
+  if (!DecodeAndConvert(sps_str, &sps_)) {
+    RTC_LOG(LS_WARNING) << "Failed to decode sprop/sps *" << sprop << "*";
+    return false;
+  }
+  if (!DecodeAndConvert(pps_str, &pps_)) {
+    RTC_LOG(LS_WARNING) << "Failed to decode sprop/pps *" << sprop << "*";
+    return false;
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets.h b/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets.h
new file mode 100644
index 0000000000..8a32f31cc0
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_H264_SPROP_PARAMETER_SETS_H_
+#define MODULES_VIDEO_CODING_H264_SPROP_PARAMETER_SETS_H_
+
+#include <cstdint>
+#include <string>
+#include <vector>
+
+namespace webrtc {
+
+class H264SpropParameterSets {
+ public:
+  H264SpropParameterSets() {}
+
+  H264SpropParameterSets(const H264SpropParameterSets&) = delete;
+  H264SpropParameterSets& operator=(const H264SpropParameterSets&) = delete;
+
+  bool DecodeSprop(const std::string& sprop);
+  const std::vector<uint8_t>& sps_nalu() { return sps_; }
+  const std::vector<uint8_t>& pps_nalu() { return pps_; }
+
+ private:
+  std::vector<uint8_t> sps_;
+  std::vector<uint8_t> pps_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_H264_SPROP_PARAMETER_SETS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets_unittest.cc b/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets_unittest.cc
new file mode 100644
index 0000000000..ae263131a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets_unittest.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/h264_sprop_parameter_sets.h"
+
+#include <vector>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class H264SpropParameterSetsTest : public ::testing::Test {
+ public:
+  H264SpropParameterSets h264_sprop;
+};
+
+TEST_F(H264SpropParameterSetsTest, Base64DecodeSprop) {
+  // Example sprop string from https://tools.ietf.org/html/rfc3984 .
+  EXPECT_TRUE(h264_sprop.DecodeSprop("Z0IACpZTBYmI,aMljiA=="));
+  static const std::vector<uint8_t> raw_sps{0x67, 0x42, 0x00, 0x0A, 0x96,
+                                            0x53, 0x05, 0x89, 0x88};
+  static const std::vector<uint8_t> raw_pps{0x68, 0xC9, 0x63, 0x88};
+  EXPECT_EQ(raw_sps, h264_sprop.sps_nalu());
+  EXPECT_EQ(raw_pps, h264_sprop.pps_nalu());
+}
+
+TEST_F(H264SpropParameterSetsTest, InvalidData) {
+  EXPECT_FALSE(h264_sprop.DecodeSprop(","));
+  EXPECT_FALSE(h264_sprop.DecodeSprop(""));
+  EXPECT_FALSE(h264_sprop.DecodeSprop(",iA=="));
+  EXPECT_FALSE(h264_sprop.DecodeSprop("iA==,"));
+  EXPECT_TRUE(h264_sprop.DecodeSprop("iA==,iA=="));
+  EXPECT_FALSE(h264_sprop.DecodeSprop("--,--"));
+  EXPECT_FALSE(h264_sprop.DecodeSprop(",,"));
+  EXPECT_FALSE(h264_sprop.DecodeSprop("iA=="));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker.cc b/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker.cc
new file mode 100644
index 0000000000..0741a261e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker.cc
@@ -0,0 +1,271 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/h264_sps_pps_tracker.h"
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/types/variant.h"
+#include "common_video/h264/h264_common.h"
+#include "common_video/h264/pps_parser.h"
+#include "common_video/h264/sps_parser.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace video_coding {
+
+namespace {
+const uint8_t start_code_h264[] = {0, 0, 0, 1};
+}  // namespace
+
+H264SpsPpsTracker::H264SpsPpsTracker() = default;
+H264SpsPpsTracker::~H264SpsPpsTracker() = default;
+
+H264SpsPpsTracker::PpsInfo::PpsInfo() = default;
+H264SpsPpsTracker::PpsInfo::PpsInfo(PpsInfo&& rhs) = default;
+H264SpsPpsTracker::PpsInfo& H264SpsPpsTracker::PpsInfo::operator=(
+    PpsInfo&& rhs) = default;
+H264SpsPpsTracker::PpsInfo::~PpsInfo() = default;
+
+H264SpsPpsTracker::SpsInfo::SpsInfo() = default;
+H264SpsPpsTracker::SpsInfo::SpsInfo(SpsInfo&& rhs) = default;
+H264SpsPpsTracker::SpsInfo& H264SpsPpsTracker::SpsInfo::operator=(
+    SpsInfo&& rhs) = default;
+H264SpsPpsTracker::SpsInfo::~SpsInfo() = default;
+
+H264SpsPpsTracker::FixedBitstream H264SpsPpsTracker::CopyAndFixBitstream(
+    rtc::ArrayView<const uint8_t> bitstream,
+    RTPVideoHeader* video_header) {
+  RTC_DCHECK(video_header);
+  RTC_DCHECK(video_header->codec == kVideoCodecH264);
+  RTC_DCHECK_GT(bitstream.size(), 0);
+
+  auto& h264_header =
+      absl::get<RTPVideoHeaderH264>(video_header->video_type_header);
+
+  bool append_sps_pps = false;
+  auto sps = sps_data_.end();
+  auto pps = pps_data_.end();
+
+  for (size_t i = 0; i < h264_header.nalus_length; ++i) {
+    const NaluInfo& nalu = h264_header.nalus[i];
+    switch (nalu.type) {
+      case H264::NaluType::kSps: {
+        SpsInfo& sps_info = sps_data_[nalu.sps_id];
+        sps_info.width = video_header->width;
+        sps_info.height = video_header->height;
+        break;
+      }
+      case H264::NaluType::kPps: {
+        pps_data_[nalu.pps_id].sps_id = nalu.sps_id;
+        break;
+      }
+      case H264::NaluType::kIdr: {
+        // If this is the first packet of an IDR, make sure we have the required
+        // SPS/PPS and also calculate how much extra space we need in the buffer
+        // to prepend the SPS/PPS to the bitstream with start codes.
+        if (video_header->is_first_packet_in_frame) {
+          if (nalu.pps_id == -1) {
+            RTC_LOG(LS_WARNING) << "No PPS id in IDR nalu.";
+            return {kRequestKeyframe};
+          }
+
+          pps = pps_data_.find(nalu.pps_id);
+          if (pps == pps_data_.end()) {
+            RTC_LOG(LS_WARNING)
+                << "No PPS with id << " << nalu.pps_id << " received";
+            return {kRequestKeyframe};
+          }
+
+          sps = sps_data_.find(pps->second.sps_id);
+          if (sps == sps_data_.end()) {
+            RTC_LOG(LS_WARNING)
+                << "No SPS with id << " << pps->second.sps_id << " received";
+            return {kRequestKeyframe};
+          }
+
+          // Since the first packet of every keyframe should have its width and
+          // height set we set it here in the case of it being supplied out of
+          // band.
+          video_header->width = sps->second.width;
+          video_header->height = sps->second.height;
+
+          // If the SPS/PPS was supplied out of band then we will have saved
+          // the actual bitstream in `data`.
+          if (sps->second.data && pps->second.data) {
+            RTC_DCHECK_GT(sps->second.size, 0);
+            RTC_DCHECK_GT(pps->second.size, 0);
+            append_sps_pps = true;
+          }
+        }
+        break;
+      }
+      default:
+        break;
+    }
+  }
+
+  RTC_CHECK(!append_sps_pps ||
+            (sps != sps_data_.end() && pps != pps_data_.end()));
+
+  // Calculate how much space we need for the rest of the bitstream.
+  size_t required_size = 0;
+
+  if (append_sps_pps) {
+    required_size += sps->second.size + sizeof(start_code_h264);
+    required_size += pps->second.size + sizeof(start_code_h264);
+  }
+
+  if (h264_header.packetization_type == kH264StapA) {
+    const uint8_t* nalu_ptr = bitstream.data() + 1;
+    while (nalu_ptr < bitstream.data() + bitstream.size() - 1) {
+      RTC_DCHECK(video_header->is_first_packet_in_frame);
+      required_size += sizeof(start_code_h264);
+
+      // The first two bytes describe the length of a segment.
+      uint16_t segment_length = nalu_ptr[0] << 8 | nalu_ptr[1];
+      nalu_ptr += 2;
+
+      required_size += segment_length;
+      nalu_ptr += segment_length;
+    }
+  } else {
+    if (h264_header.nalus_length > 0) {
+      required_size += sizeof(start_code_h264);
+    }
+    required_size += bitstream.size();
+  }
+
+  // Then we copy to the new buffer.
+  H264SpsPpsTracker::FixedBitstream fixed;
+  fixed.bitstream.EnsureCapacity(required_size);
+
+  if (append_sps_pps) {
+    // Insert SPS.
+    fixed.bitstream.AppendData(start_code_h264);
+    fixed.bitstream.AppendData(sps->second.data.get(), sps->second.size);
+
+    // Insert PPS.
+    fixed.bitstream.AppendData(start_code_h264);
+    fixed.bitstream.AppendData(pps->second.data.get(), pps->second.size);
+
+    // Update codec header to reflect the newly added SPS and PPS.
+    NaluInfo sps_info;
+    sps_info.type = H264::NaluType::kSps;
+    sps_info.sps_id = sps->first;
+    sps_info.pps_id = -1;
+    NaluInfo pps_info;
+    pps_info.type = H264::NaluType::kPps;
+    pps_info.sps_id = sps->first;
+    pps_info.pps_id = pps->first;
+    if (h264_header.nalus_length + 2 <= kMaxNalusPerPacket) {
+      h264_header.nalus[h264_header.nalus_length++] = sps_info;
+      h264_header.nalus[h264_header.nalus_length++] = pps_info;
+    } else {
+      RTC_LOG(LS_WARNING) << "Not enough space in H.264 codec header to insert "
+                             "SPS/PPS provided out-of-band.";
+    }
+  }
+
+  // Copy the rest of the bitstream and insert start codes.
+  if (h264_header.packetization_type == kH264StapA) {
+    const uint8_t* nalu_ptr = bitstream.data() + 1;
+    while (nalu_ptr < bitstream.data() + bitstream.size() - 1) {
+      fixed.bitstream.AppendData(start_code_h264);
+
+      // The first two bytes describe the length of a segment.
+      uint16_t segment_length = nalu_ptr[0] << 8 | nalu_ptr[1];
+      nalu_ptr += 2;
+
+      size_t copy_end = nalu_ptr - bitstream.data() + segment_length;
+      if (copy_end > bitstream.size()) {
+        return {kDrop};
+      }
+
+      fixed.bitstream.AppendData(nalu_ptr, segment_length);
+      nalu_ptr += segment_length;
+    }
+  } else {
+    if (h264_header.nalus_length > 0) {
+      fixed.bitstream.AppendData(start_code_h264);
+    }
+    fixed.bitstream.AppendData(bitstream.data(), bitstream.size());
+  }
+
+  fixed.action = kInsert;
+  return fixed;
+}
+
+void H264SpsPpsTracker::InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
+                                          const std::vector<uint8_t>& pps) {
+  constexpr size_t kNaluHeaderOffset = 1;
+  if (sps.size() < kNaluHeaderOffset) {
+    RTC_LOG(LS_WARNING) << "SPS size  " << sps.size() << " is smaller than "
+                        << kNaluHeaderOffset;
+    return;
+  }
+  if ((sps[0] & 0x1f) != H264::NaluType::kSps) {
+    RTC_LOG(LS_WARNING) << "SPS Nalu header missing";
+    return;
+  }
+  if (pps.size() < kNaluHeaderOffset) {
+    RTC_LOG(LS_WARNING) << "PPS size  " << pps.size() << " is smaller than "
+                        << kNaluHeaderOffset;
+    return;
+  }
+  if ((pps[0] & 0x1f) != H264::NaluType::kPps) {
+    RTC_LOG(LS_WARNING) << "SPS Nalu header missing";
+    return;
+  }
+  absl::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
+      sps.data() + kNaluHeaderOffset, sps.size() - kNaluHeaderOffset);
+  absl::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
+      pps.data() + kNaluHeaderOffset, pps.size() - kNaluHeaderOffset);
+
+  if (!parsed_sps) {
+    RTC_LOG(LS_WARNING) << "Failed to parse SPS.";
+  }
+
+  if (!parsed_pps) {
+    RTC_LOG(LS_WARNING) << "Failed to parse PPS.";
+  }
+
+  if (!parsed_pps || !parsed_sps) {
+    return;
+  }
+
+  SpsInfo sps_info;
+  sps_info.size = sps.size();
+  sps_info.width = parsed_sps->width;
+  sps_info.height = parsed_sps->height;
+  uint8_t* sps_data = new uint8_t[sps_info.size];
+  memcpy(sps_data, sps.data(), sps_info.size);
+  sps_info.data.reset(sps_data);
+  sps_data_[parsed_sps->id] = std::move(sps_info);
+
+  PpsInfo pps_info;
+  pps_info.size = pps.size();
+  pps_info.sps_id = parsed_pps->sps_id;
+  uint8_t* pps_data = new uint8_t[pps_info.size];
+  memcpy(pps_data, pps.data(), pps_info.size);
+  pps_info.data.reset(pps_data);
+  pps_data_[parsed_pps->id] = std::move(pps_info);
+
+  RTC_LOG(LS_INFO) << "Inserted SPS id " << parsed_sps->id << " and PPS id "
+                   << parsed_pps->id << " (referencing SPS "
+                   << parsed_pps->sps_id << ")";
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker.h b/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker.h
new file mode 100644
index 0000000000..600e2ee397
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_H264_SPS_PPS_TRACKER_H_
+#define MODULES_VIDEO_CODING_H264_SPS_PPS_TRACKER_H_
+
+#include <cstddef>
+#include <cstdint>
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class H264SpsPpsTracker {
+ public:
+  enum PacketAction { kInsert, kDrop, kRequestKeyframe };
+  struct FixedBitstream {
+    PacketAction action;
+    rtc::CopyOnWriteBuffer bitstream;
+  };
+
+  H264SpsPpsTracker();
+  ~H264SpsPpsTracker();
+
+  // Returns fixed bitstream and modifies `video_header`.
+  FixedBitstream CopyAndFixBitstream(rtc::ArrayView<const uint8_t> bitstream,
+                                     RTPVideoHeader* video_header);
+
+  void InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
+                         const std::vector<uint8_t>& pps);
+
+ private:
+  struct PpsInfo {
+    PpsInfo();
+    PpsInfo(PpsInfo&& rhs);
+    PpsInfo& operator=(PpsInfo&& rhs);
+    ~PpsInfo();
+
+    int sps_id = -1;
+    size_t size = 0;
+    std::unique_ptr<uint8_t[]> data;
+  };
+
+  struct SpsInfo {
+    SpsInfo();
+    SpsInfo(SpsInfo&& rhs);
+    SpsInfo& operator=(SpsInfo&& rhs);
+    ~SpsInfo();
+
+    size_t size = 0;
+    int width = -1;
+    int height = -1;
+    std::unique_ptr<uint8_t[]> data;
+  };
+
+  std::map<uint32_t, PpsInfo> pps_data_;
+  std::map<uint32_t, SpsInfo> sps_data_;
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_H264_SPS_PPS_TRACKER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker_unittest.cc b/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker_unittest.cc
new file mode 100644
index 0000000000..04abb75e4e
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker_unittest.cc
@@ -0,0 +1,368 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/h264_sps_pps_tracker.h"
+
+#include <string.h>
+
+#include <vector>
+
+#include "absl/types/variant.h"
+#include "common_video/h264/h264_common.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "modules/video_coding/packet.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace video_coding {
+namespace {
+
+using ::testing::ElementsAreArray;
+
+const uint8_t start_code[] = {0, 0, 0, 1};
+
+rtc::ArrayView<const uint8_t> Bitstream(
+    const H264SpsPpsTracker::FixedBitstream& fixed) {
+  return fixed.bitstream;
+}
+
+void ExpectSpsPpsIdr(const RTPVideoHeaderH264& codec_header,
+                     uint8_t sps_id,
+                     uint8_t pps_id) {
+  bool contains_sps = false;
+  bool contains_pps = false;
+  bool contains_idr = false;
+  for (const auto& nalu : codec_header.nalus) {
+    if (nalu.type == H264::NaluType::kSps) {
+      EXPECT_EQ(sps_id, nalu.sps_id);
+      contains_sps = true;
+    } else if (nalu.type == H264::NaluType::kPps) {
+      EXPECT_EQ(sps_id, nalu.sps_id);
+      EXPECT_EQ(pps_id, nalu.pps_id);
+      contains_pps = true;
+    } else if (nalu.type == H264::NaluType::kIdr) {
+      EXPECT_EQ(pps_id, nalu.pps_id);
+      contains_idr = true;
+    }
+  }
+  EXPECT_TRUE(contains_sps);
+  EXPECT_TRUE(contains_pps);
+  EXPECT_TRUE(contains_idr);
+}
+
+class H264VideoHeader : public RTPVideoHeader {
+ public:
+  H264VideoHeader() {
+    codec = kVideoCodecH264;
+    is_first_packet_in_frame = false;
+    auto& h264_header = video_type_header.emplace<RTPVideoHeaderH264>();
+    h264_header.nalus_length = 0;
+    h264_header.packetization_type = kH264SingleNalu;
+  }
+
+  RTPVideoHeaderH264& h264() {
+    return absl::get<RTPVideoHeaderH264>(video_type_header);
+  }
+};
+
+}  // namespace
+
+class TestH264SpsPpsTracker : public ::testing::Test {
+ public:
+  void AddSps(H264VideoHeader* header,
+              uint8_t sps_id,
+              std::vector<uint8_t>* data) {
+    NaluInfo info;
+    info.type = H264::NaluType::kSps;
+    info.sps_id = sps_id;
+    info.pps_id = -1;
+    data->push_back(H264::NaluType::kSps);
+    data->push_back(sps_id);  // The sps data, just a single byte.
+
+    header->h264().nalus[header->h264().nalus_length++] = info;
+  }
+
+  void AddPps(H264VideoHeader* header,
+              uint8_t sps_id,
+              uint8_t pps_id,
+              std::vector<uint8_t>* data) {
+    NaluInfo info;
+    info.type = H264::NaluType::kPps;
+    info.sps_id = sps_id;
+    info.pps_id = pps_id;
+    data->push_back(H264::NaluType::kPps);
+    data->push_back(pps_id);  // The pps data, just a single byte.
+
+    header->h264().nalus[header->h264().nalus_length++] = info;
+  }
+
+  void AddIdr(H264VideoHeader* header, int pps_id) {
+    NaluInfo info;
+    info.type = H264::NaluType::kIdr;
+    info.sps_id = -1;
+    info.pps_id = pps_id;
+
+    header->h264().nalus[header->h264().nalus_length++] = info;
+  }
+
+ protected:
+  H264SpsPpsTracker tracker_;
+};
+
+TEST_F(TestH264SpsPpsTracker, NoNalus) {
+  uint8_t data[] = {1, 2, 3};
+  H264VideoHeader header;
+  header.h264().packetization_type = kH264FuA;
+
+  H264SpsPpsTracker::FixedBitstream fixed =
+      tracker_.CopyAndFixBitstream(data, &header);
+
+  EXPECT_EQ(fixed.action, H264SpsPpsTracker::kInsert);
+  EXPECT_THAT(Bitstream(fixed), ElementsAreArray(data));
+}
+
+TEST_F(TestH264SpsPpsTracker, FuAFirstPacket) {
+  uint8_t data[] = {1, 2, 3};
+  H264VideoHeader header;
+  header.h264().packetization_type = kH264FuA;
+  header.h264().nalus_length = 1;
+  header.is_first_packet_in_frame = true;
+
+  H264SpsPpsTracker::FixedBitstream fixed =
+      tracker_.CopyAndFixBitstream(data, &header);
+
+  EXPECT_EQ(fixed.action, H264SpsPpsTracker::kInsert);
+  std::vector<uint8_t> expected;
+  expected.insert(expected.end(), start_code, start_code + sizeof(start_code));
+  expected.insert(expected.end(), {1, 2, 3});
+  EXPECT_THAT(Bitstream(fixed), ElementsAreArray(expected));
+}
+
+TEST_F(TestH264SpsPpsTracker, StapAIncorrectSegmentLength) {
+  uint8_t data[] = {0, 0, 2, 0};
+  H264VideoHeader header;
+  header.h264().packetization_type = kH264StapA;
+  header.is_first_packet_in_frame = true;
+
+  EXPECT_EQ(tracker_.CopyAndFixBitstream(data, &header).action,
+            H264SpsPpsTracker::kDrop);
+}
+
+TEST_F(TestH264SpsPpsTracker, SingleNaluInsertStartCode) {
+  uint8_t data[] = {1, 2, 3};
+  H264VideoHeader header;
+  header.h264().nalus_length = 1;
+
+  H264SpsPpsTracker::FixedBitstream fixed =
+      tracker_.CopyAndFixBitstream(data, &header);
+
+  EXPECT_EQ(fixed.action, H264SpsPpsTracker::kInsert);
+  std::vector<uint8_t> expected;
+  expected.insert(expected.end(), start_code, start_code + sizeof(start_code));
+  expected.insert(expected.end(), {1, 2, 3});
+  EXPECT_THAT(Bitstream(fixed), ElementsAreArray(expected));
+}
+
+TEST_F(TestH264SpsPpsTracker, NoStartCodeInsertedForSubsequentFuAPacket) {
+  std::vector<uint8_t> data = {1, 2, 3};
+  H264VideoHeader header;
+  header.h264().packetization_type = kH264FuA;
+  // Since no NALU begin in this packet the nalus_length is zero.
+  header.h264().nalus_length = 0;
+
+  H264SpsPpsTracker::FixedBitstream fixed =
+      tracker_.CopyAndFixBitstream(data, &header);
+
+  EXPECT_EQ(fixed.action, H264SpsPpsTracker::kInsert);
+  EXPECT_THAT(Bitstream(fixed), ElementsAreArray(data));
+}
+
+TEST_F(TestH264SpsPpsTracker, IdrFirstPacketNoSpsPpsInserted) {
+  std::vector<uint8_t> data = {1, 2, 3};
+  H264VideoHeader header;
+  header.is_first_packet_in_frame = true;
+  AddIdr(&header, 0);
+
+  EXPECT_EQ(tracker_.CopyAndFixBitstream(data, &header).action,
+            H264SpsPpsTracker::kRequestKeyframe);
+}
+
+TEST_F(TestH264SpsPpsTracker, IdrFirstPacketNoPpsInserted) {
+  std::vector<uint8_t> data = {1, 2, 3};
+  H264VideoHeader header;
+  header.is_first_packet_in_frame = true;
+  AddSps(&header, 0, &data);
+  AddIdr(&header, 0);
+
+  EXPECT_EQ(tracker_.CopyAndFixBitstream(data, &header).action,
+            H264SpsPpsTracker::kRequestKeyframe);
+}
+
+TEST_F(TestH264SpsPpsTracker, IdrFirstPacketNoSpsInserted) {
+  std::vector<uint8_t> data = {1, 2, 3};
+  H264VideoHeader header;
+  header.is_first_packet_in_frame = true;
+  AddPps(&header, 0, 0, &data);
+  AddIdr(&header, 0);
+
+  EXPECT_EQ(tracker_.CopyAndFixBitstream(data, &header).action,
+            H264SpsPpsTracker::kRequestKeyframe);
+}
+
+TEST_F(TestH264SpsPpsTracker, SpsPpsPacketThenIdrFirstPacket) {
+  std::vector<uint8_t> data;
+  H264VideoHeader sps_pps_header;
+  // Insert SPS/PPS
+  AddSps(&sps_pps_header, 0, &data);
+  AddPps(&sps_pps_header, 0, 1, &data);
+
+  EXPECT_EQ(tracker_.CopyAndFixBitstream(data, &sps_pps_header).action,
+            H264SpsPpsTracker::kInsert);
+
+  // Insert first packet of the IDR
+  H264VideoHeader idr_header;
+  idr_header.is_first_packet_in_frame = true;
+  AddIdr(&idr_header, 1);
+  data = {1, 2, 3};
+
+  H264SpsPpsTracker::FixedBitstream fixed =
+      tracker_.CopyAndFixBitstream(data, &idr_header);
+  EXPECT_EQ(fixed.action, H264SpsPpsTracker::kInsert);
+
+  std::vector<uint8_t> expected;
+  expected.insert(expected.end(), start_code, start_code + sizeof(start_code));
+  expected.insert(expected.end(), {1, 2, 3});
+  EXPECT_THAT(Bitstream(fixed), ElementsAreArray(expected));
+}
+
+TEST_F(TestH264SpsPpsTracker, SpsPpsIdrInStapA) {
+  std::vector<uint8_t> data;
+  H264VideoHeader header;
+  header.h264().packetization_type = kH264StapA;
+  header.is_first_packet_in_frame = true;  // Always true for StapA
+
+  data.insert(data.end(), {0});     // First byte is ignored
+  data.insert(data.end(), {0, 2});  // Length of segment
+  AddSps(&header, 13, &data);
+  data.insert(data.end(), {0, 2});  // Length of segment
+  AddPps(&header, 13, 27, &data);
+  data.insert(data.end(), {0, 5});  // Length of segment
+  AddIdr(&header, 27);
+  data.insert(data.end(), {1, 2, 3, 2, 1});
+
+  H264SpsPpsTracker::FixedBitstream fixed =
+      tracker_.CopyAndFixBitstream(data, &header);
+
+  EXPECT_THAT(fixed.action, H264SpsPpsTracker::kInsert);
+
+  std::vector<uint8_t> expected;
+  expected.insert(expected.end(), start_code, start_code + sizeof(start_code));
+  expected.insert(expected.end(), {H264::NaluType::kSps, 13});
+  expected.insert(expected.end(), start_code, start_code + sizeof(start_code));
+  expected.insert(expected.end(), {H264::NaluType::kPps, 27});
+  expected.insert(expected.end(), start_code, start_code + sizeof(start_code));
+  expected.insert(expected.end(), {1, 2, 3, 2, 1});
+  EXPECT_THAT(Bitstream(fixed), ElementsAreArray(expected));
+}
+
+TEST_F(TestH264SpsPpsTracker, SpsPpsOutOfBand) {
+  constexpr uint8_t kData[] = {1, 2, 3};
+
+  // Generated by "ffmpeg -r 30 -f avfoundation -i "default" out.h264" on macos.
+  // width: 320, height: 240
+  const std::vector<uint8_t> sps(
+      {0x67, 0x7a, 0x00, 0x0d, 0xbc, 0xd9, 0x41, 0x41, 0xfa, 0x10, 0x00, 0x00,
+       0x03, 0x00, 0x10, 0x00, 0x00, 0x03, 0x03, 0xc0, 0xf1, 0x42, 0x99, 0x60});
+  const std::vector<uint8_t> pps({0x68, 0xeb, 0xe3, 0xcb, 0x22, 0xc0});
+  tracker_.InsertSpsPpsNalus(sps, pps);
+
+  // Insert first packet of the IDR.
+  H264VideoHeader idr_header;
+  idr_header.is_first_packet_in_frame = true;
+  AddIdr(&idr_header, 0);
+  EXPECT_EQ(idr_header.h264().nalus_length, 1u);
+
+  H264SpsPpsTracker::FixedBitstream fixed =
+      tracker_.CopyAndFixBitstream(kData, &idr_header);
+
+  EXPECT_EQ(idr_header.h264().nalus_length, 3u);
+  EXPECT_EQ(idr_header.width, 320u);
+  EXPECT_EQ(idr_header.height, 240u);
+  ExpectSpsPpsIdr(idr_header.h264(), 0, 0);
+}
+
+TEST_F(TestH264SpsPpsTracker, SpsPpsOutOfBandWrongNaluHeader) {
+  constexpr uint8_t kData[] = {1, 2, 3};
+
+  // Generated by "ffmpeg -r 30 -f avfoundation -i "default" out.h264" on macos.
+  // Nalu headers manupilated afterwards.
+  const std::vector<uint8_t> sps(
+      {0xff, 0x7a, 0x00, 0x0d, 0xbc, 0xd9, 0x41, 0x41, 0xfa, 0x10, 0x00, 0x00,
+       0x03, 0x00, 0x10, 0x00, 0x00, 0x03, 0x03, 0xc0, 0xf1, 0x42, 0x99, 0x60});
+  const std::vector<uint8_t> pps({0xff, 0xeb, 0xe3, 0xcb, 0x22, 0xc0});
+  tracker_.InsertSpsPpsNalus(sps, pps);
+
+  // Insert first packet of the IDR.
+  H264VideoHeader idr_header;
+  idr_header.is_first_packet_in_frame = true;
+  AddIdr(&idr_header, 0);
+
+  EXPECT_EQ(tracker_.CopyAndFixBitstream(kData, &idr_header).action,
+            H264SpsPpsTracker::kRequestKeyframe);
+}
+
+TEST_F(TestH264SpsPpsTracker, SpsPpsOutOfBandIncompleteNalu) {
+  constexpr uint8_t kData[] = {1, 2, 3};
+
+  // Generated by "ffmpeg -r 30 -f avfoundation -i "default" out.h264" on macos.
+  // Nalus damaged afterwards.
+  const std::vector<uint8_t> sps({0x67, 0x7a, 0x00, 0x0d, 0xbc, 0xd9});
+  const std::vector<uint8_t> pps({0x68, 0xeb, 0xe3, 0xcb, 0x22, 0xc0});
+  tracker_.InsertSpsPpsNalus(sps, pps);
+
+  // Insert first packet of the IDR.
+  H264VideoHeader idr_header;
+  idr_header.is_first_packet_in_frame = true;
+  AddIdr(&idr_header, 0);
+
+  EXPECT_EQ(tracker_.CopyAndFixBitstream(kData, &idr_header).action,
+            H264SpsPpsTracker::kRequestKeyframe);
+}
+
+TEST_F(TestH264SpsPpsTracker, SaveRestoreWidthHeight) {
+  std::vector<uint8_t> data;
+
+  // Insert an SPS/PPS packet with width/height and make sure
+  // that information is set on the first IDR packet.
+  H264VideoHeader sps_pps_header;
+  AddSps(&sps_pps_header, 0, &data);
+  AddPps(&sps_pps_header, 0, 1, &data);
+  sps_pps_header.width = 320;
+  sps_pps_header.height = 240;
+
+  EXPECT_EQ(tracker_.CopyAndFixBitstream(data, &sps_pps_header).action,
+            H264SpsPpsTracker::kInsert);
+
+  H264VideoHeader idr_header;
+  idr_header.is_first_packet_in_frame = true;
+  AddIdr(&idr_header, 1);
+  data.insert(data.end(), {1, 2, 3});
+
+  EXPECT_EQ(tracker_.CopyAndFixBitstream(data, &idr_header).action,
+            H264SpsPpsTracker::kInsert);
+
+  EXPECT_EQ(idr_header.width, 320);
+  EXPECT_EQ(idr_header.height, 240);
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/histogram.cc b/third_party/libwebrtc/modules/video_coding/histogram.cc
new file mode 100644
index 0000000000..4e90b19eec
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/histogram.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/histogram.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace video_coding {
+Histogram::Histogram(size_t num_buckets, size_t max_num_values) {
+  RTC_DCHECK_GT(num_buckets, 0);
+  RTC_DCHECK_GT(max_num_values, 0);
+  buckets_.resize(num_buckets);
+  values_.reserve(max_num_values);
+  index_ = 0;
+}
+
+void Histogram::Add(size_t value) {
+  value = std::min<size_t>(value, buckets_.size() - 1);
+  if (index_ < values_.size()) {
+    --buckets_[values_[index_]];
+    RTC_DCHECK_LT(values_[index_], buckets_.size());
+    values_[index_] = value;
+  } else {
+    values_.emplace_back(value);
+  }
+
+  ++buckets_[value];
+  index_ = (index_ + 1) % values_.capacity();
+}
+
+size_t Histogram::InverseCdf(float probability) const {
+  RTC_DCHECK_GE(probability, 0.f);
+  RTC_DCHECK_LE(probability, 1.f);
+  RTC_DCHECK_GT(values_.size(), 0ul);
+
+  size_t bucket = 0;
+  float accumulated_probability = 0;
+  while (accumulated_probability < probability && bucket < buckets_.size()) {
+    accumulated_probability +=
+        static_cast<float>(buckets_[bucket]) / values_.size();
+    ++bucket;
+  }
+  return bucket;
+}
+
+size_t Histogram::NumValues() const {
+  return values_.size();
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/histogram.h b/third_party/libwebrtc/modules/video_coding/histogram.h
new file mode 100644
index 0000000000..aa8d44d80f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/histogram.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_HISTOGRAM_H_
+#define MODULES_VIDEO_CODING_HISTOGRAM_H_
+
+#include <cstddef>
+#include <vector>
+
+namespace webrtc {
+namespace video_coding {
+class Histogram {
+ public:
+  // A discrete histogram where every bucket with range [0, num_buckets).
+  // Values greater or equal to num_buckets will be placed in the last bucket.
+  Histogram(size_t num_buckets, size_t max_num_values);
+
+  // Add a value to the histogram. If there already is max_num_values in the
+  // histogram then the oldest value will be replaced with the new value.
+  void Add(size_t value);
+
+  // Calculates how many buckets have to be summed in order to accumulate at
+  // least the given probability.
+  size_t InverseCdf(float probability) const;
+
+  // How many values that make up this histogram.
+  size_t NumValues() const;
+
+ private:
+  // A circular buffer that holds the values that make up the histogram.
+  std::vector<size_t> values_;
+  std::vector<size_t> buckets_;
+  size_t index_;
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_HISTOGRAM_H_
diff --git a/third_party/libwebrtc/modules/video_coding/histogram_unittest.cc b/third_party/libwebrtc/modules/video_coding/histogram_unittest.cc
new file mode 100644
index 0000000000..3690a39398
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/histogram_unittest.cc
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/histogram.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class TestHistogram : public ::testing::Test {
+ protected:
+  TestHistogram() : histogram_(5, 10) {}
+  Histogram histogram_;
+};
+
+TEST_F(TestHistogram, NumValues) {
+  EXPECT_EQ(0ul, histogram_.NumValues());
+  histogram_.Add(0);
+  EXPECT_EQ(1ul, histogram_.NumValues());
+}
+
+TEST_F(TestHistogram, InverseCdf) {
+  histogram_.Add(0);
+  histogram_.Add(1);
+  histogram_.Add(2);
+  histogram_.Add(3);
+  histogram_.Add(4);
+  EXPECT_EQ(5ul, histogram_.NumValues());
+  EXPECT_EQ(1ul, histogram_.InverseCdf(0.2f));
+  EXPECT_EQ(2ul, histogram_.InverseCdf(0.2000001f));
+  EXPECT_EQ(4ul, histogram_.InverseCdf(0.8f));
+
+  histogram_.Add(0);
+  EXPECT_EQ(6ul, histogram_.NumValues());
+  EXPECT_EQ(1ul, histogram_.InverseCdf(0.2f));
+  EXPECT_EQ(1ul, histogram_.InverseCdf(0.2000001f));
+}
+
+TEST_F(TestHistogram, ReplaceOldValues) {
+  histogram_.Add(0);
+  histogram_.Add(0);
+  histogram_.Add(0);
+  histogram_.Add(0);
+  histogram_.Add(0);
+  histogram_.Add(1);
+  histogram_.Add(1);
+  histogram_.Add(1);
+  histogram_.Add(1);
+  histogram_.Add(1);
+  EXPECT_EQ(10ul, histogram_.NumValues());
+  EXPECT_EQ(1ul, histogram_.InverseCdf(0.5f));
+  EXPECT_EQ(2ul, histogram_.InverseCdf(0.5000001f));
+
+  histogram_.Add(4);
+  histogram_.Add(4);
+  histogram_.Add(4);
+  histogram_.Add(4);
+  EXPECT_EQ(10ul, histogram_.NumValues());
+  EXPECT_EQ(1ul, histogram_.InverseCdf(0.1f));
+  EXPECT_EQ(2ul, histogram_.InverseCdf(0.5f));
+
+  histogram_.Add(20);
+  EXPECT_EQ(10ul, histogram_.NumValues());
+  EXPECT_EQ(2ul, histogram_.InverseCdf(0.5f));
+  EXPECT_EQ(5ul, histogram_.InverseCdf(0.5000001f));
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/include/video_codec_initializer.h b/third_party/libwebrtc/modules/video_coding/include/video_codec_initializer.h
new file mode 100644
index 0000000000..270c4dbcd1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/include/video_codec_initializer.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INITIALIZER_H_
+#define MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INITIALIZER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+
+class VideoBitrateAllocator;
+class VideoCodec;
+
+class VideoCodecInitializer {
+ public:
+  // Takes a VideoEncoderConfig and the VideoStream configuration and
+  // translates them into the old school VideoCodec type.
+  // It also creates a VideoBitrateAllocator instance, suitable for the codec
+  // type used. For instance, VP8 will create an allocator than can handle
+  // simulcast and temporal layering.
+  // GetBitrateAllocator is called implicitly from here, no need to call again.
+  static bool SetupCodec(const VideoEncoderConfig& config,
+                         const std::vector<VideoStream>& streams,
+                         VideoCodec* codec);
+
+ private:
+  static VideoCodec VideoEncoderConfigToVideoCodec(
+      const VideoEncoderConfig& config,
+      const std::vector<VideoStream>& streams);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INITIALIZER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.cc b/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.cc
new file mode 100644
index 0000000000..bd033b6c57
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+CodecSpecificInfo::CodecSpecificInfo() : codecType(kVideoCodecGeneric) {
+  memset(&codecSpecific, 0, sizeof(codecSpecific));
+}
+
+CodecSpecificInfo::CodecSpecificInfo(const CodecSpecificInfo&) = default;
+CodecSpecificInfo::~CodecSpecificInfo() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.h b/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.h
new file mode 100644
index 0000000000..46ae0d29e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.h
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
+#define MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
+
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Note: If any pointers are added to this struct, it must be fitted
+// with a copy-constructor. See below.
+// Hack alert - the code assumes that thisstruct is memset when constructed.
+struct CodecSpecificInfoVP8 {
+  bool nonReference;
+  uint8_t temporalIdx;
+  bool layerSync;
+  int8_t keyIdx;  // Negative value to skip keyIdx.
+
+  // Used to generate the list of dependency frames.
+  // `referencedBuffers` and `updatedBuffers` contain buffer IDs.
+  // Note that the buffer IDs here have a one-to-one mapping with the actual
+  // codec buffers, but the exact mapping (i.e. whether 0 refers to Last,
+  // to Golden or to Arf) is not pre-determined.
+  // More references may be specified than are strictly necessary, but not less.
+  // TODO(bugs.webrtc.org/10242): Remove `useExplicitDependencies` once all
+  // encoder-wrappers are updated.
+  bool useExplicitDependencies;
+  static constexpr size_t kBuffersCount = 3;
+  size_t referencedBuffers[kBuffersCount];
+  size_t referencedBuffersCount;
+  size_t updatedBuffers[kBuffersCount];
+  size_t updatedBuffersCount;
+};
+static_assert(std::is_pod<CodecSpecificInfoVP8>::value, "");
+
+// Hack alert - the code assumes that thisstruct is memset when constructed.
+struct CodecSpecificInfoVP9 {
+  bool first_frame_in_picture;  // First frame, increment picture_id.
+  bool inter_pic_predicted;     // This layer frame is dependent on previously
+                                // coded frame(s).
+  bool flexible_mode;
+  bool ss_data_available;
+  bool non_ref_for_inter_layer_pred;
+
+  uint8_t temporal_idx;
+  bool temporal_up_switch;
+  bool inter_layer_predicted;  // Frame is dependent on directly lower spatial
+                               // layer frame.
+  uint8_t gof_idx;
+
+  // SS data.
+  size_t num_spatial_layers;  // Always populated.
+  size_t first_active_layer;
+  bool spatial_layer_resolution_present;
+  uint16_t width[kMaxVp9NumberOfSpatialLayers];
+  uint16_t height[kMaxVp9NumberOfSpatialLayers];
+  GofInfoVP9 gof;
+
+  // Frame reference data.
+  uint8_t num_ref_pics;
+  uint8_t p_diff[kMaxVp9RefPics];
+
+  ABSL_DEPRECATED("") bool end_of_picture;
+};
+static_assert(std::is_pod<CodecSpecificInfoVP9>::value, "");
+
+// Hack alert - the code assumes that thisstruct is memset when constructed.
+struct CodecSpecificInfoH264 {
+  H264PacketizationMode packetization_mode;
+  uint8_t temporal_idx;
+  bool base_layer_sync;
+  bool idr_frame;
+};
+static_assert(std::is_pod<CodecSpecificInfoH264>::value, "");
+
+union CodecSpecificInfoUnion {
+  CodecSpecificInfoVP8 VP8;
+  CodecSpecificInfoVP9 VP9;
+  CodecSpecificInfoH264 H264;
+};
+static_assert(std::is_pod<CodecSpecificInfoUnion>::value, "");
+
+// Note: if any pointers are added to this struct or its sub-structs, it
+// must be fitted with a copy-constructor. This is because it is copied
+// in the copy-constructor of VCMEncodedFrame.
+struct RTC_EXPORT CodecSpecificInfo {
+  CodecSpecificInfo();
+  CodecSpecificInfo(const CodecSpecificInfo&);
+  ~CodecSpecificInfo();
+
+  VideoCodecType codecType;
+  CodecSpecificInfoUnion codecSpecific;
+  bool end_of_picture = true;
+  absl::optional<GenericFrameInfo> generic_frame_info;
+  absl::optional<FrameDependencyStructure> template_structure;
+  absl::optional<ScalabilityMode> scalability_mode;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODEC_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/video_coding/include/video_coding.h b/third_party/libwebrtc/modules/video_coding/include/video_coding.h
new file mode 100644
index 0000000000..ee9326d9fc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/include/video_coding.h
@@ -0,0 +1,150 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_
+#define MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_
+
+#include "api/field_trials_view.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+
+namespace webrtc {
+
+class Clock;
+class EncodedImageCallback;
+class VideoDecoder;
+class VideoEncoder;
+struct CodecSpecificInfo;
+
+class VideoCodingModule {
+ public:
+  // DEPRECATED.
+  static VideoCodingModule* Create(
+      Clock* clock,
+      const FieldTrialsView* field_trials = nullptr);
+
+  virtual ~VideoCodingModule() = default;
+
+  /*
+   *   Receiver
+   */
+
+  // Register possible receive codecs, can be called multiple times for
+  // different codecs.
+  // The module will automatically switch between registered codecs depending on
+  // the
+  // payload type of incoming frames. The actual decoder will be created when
+  // needed.
+  //
+  // Input:
+  //      - payload_type      : RTP payload type
+  //      - settings          : Settings for the decoder to be registered.
+  //
+  virtual void RegisterReceiveCodec(uint8_t payload_type,
+                                    const VideoDecoder::Settings& settings) = 0;
+
+  // Register an external decoder object.
+  //
+  // Input:
+  //      - externalDecoder : Decoder object to be used for decoding frames.
+  //      - payloadType     : The payload type which this decoder is bound to.
+  virtual void RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                                       uint8_t payloadType) = 0;
+
+  // Register a receive callback. Will be called whenever there is a new frame
+  // ready
+  // for rendering.
+  //
+  // Input:
+  //      - receiveCallback        : The callback object to be used by the
+  //      module when a
+  //                                 frame is ready for rendering.
+  //                                 De-register with a NULL pointer.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterReceiveCallback(
+      VCMReceiveCallback* receiveCallback) = 0;
+
+  // Register a frame type request callback. This callback will be called when
+  // the
+  // module needs to request specific frame types from the send side.
+  //
+  // Input:
+  //      - frameTypeCallback      : The callback object to be used by the
+  //      module when
+  //                                 requesting a specific type of frame from
+  //                                 the send side.
+  //                                 De-register with a NULL pointer.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t RegisterFrameTypeCallback(
+      VCMFrameTypeCallback* frameTypeCallback) = 0;
+
+  // Registers a callback which is called whenever the receive side of the VCM
+  // encounters holes in the packet sequence and needs packets to be
+  // retransmitted.
+  //
+  // Input:
+  //              - callback      : The callback to be registered in the VCM.
+  //
+  // Return value     : VCM_OK,     on success.
+  //                    <0,         on error.
+  virtual int32_t RegisterPacketRequestCallback(
+      VCMPacketRequestCallback* callback) = 0;
+
+  // Waits for the next frame in the jitter buffer to become complete
+  // (waits no longer than maxWaitTimeMs), then passes it to the decoder for
+  // decoding.
+  // Should be called as often as possible to get the most out of the decoder.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t Decode(uint16_t maxWaitTimeMs = 200) = 0;
+
+  // Insert a parsed packet into the receiver side of the module. Will be placed
+  // in the
+  // jitter buffer waiting for the frame to become complete. Returns as soon as
+  // the packet
+  // has been placed in the jitter buffer.
+  //
+  // Input:
+  //      - incomingPayload      : Payload of the packet.
+  //      - payloadLength        : Length of the payload.
+  //      - rtp_header           : The parsed RTP header.
+  //      - video_header         : The relevant extensions and payload header.
+  //
+  // Return value      : VCM_OK, on success.
+  //                     < 0,    on error.
+  virtual int32_t IncomingPacket(const uint8_t* incomingPayload,
+                                 size_t payloadLength,
+                                 const RTPHeader& rtp_header,
+                                 const RTPVideoHeader& video_header) = 0;
+
+  // Sets the maximum number of sequence numbers that we are allowed to NACK
+  // and the oldest sequence number that we will consider to NACK. If a
+  // sequence number older than `max_packet_age_to_nack` is missing
+  // a key frame will be requested. A key frame will also be requested if the
+  // time of incomplete or non-continuous frames in the jitter buffer is above
+  // `max_incomplete_time_ms`.
+  virtual void SetNackSettings(size_t max_nack_list_size,
+                               int max_packet_age_to_nack,
+                               int max_incomplete_time_ms) = 0;
+
+  // Runs delayed tasks. Expected to be called periodically.
+  virtual void Process() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_H_
diff --git a/third_party/libwebrtc/modules/video_coding/include/video_coding_defines.h b/third_party/libwebrtc/modules/video_coding/include/video_coding_defines.h
new file mode 100644
index 0000000000..bf98d5e668
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/include/video_coding_defines.h
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
+#define MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_timing.h"
+#include "api/video_codecs/video_decoder.h"
+
+namespace webrtc {
+
+// Error codes
+#define VCM_FRAME_NOT_READY 3
+#define VCM_MISSING_CALLBACK 1
+#define VCM_OK 0
+#define VCM_GENERAL_ERROR -1
+#define VCM_PARAMETER_ERROR -4
+#define VCM_NO_CODEC_REGISTERED -8
+#define VCM_JITTER_BUFFER_ERROR -9
+
+enum {
+  // Timing frames settings. Timing frames are sent every
+  // `kDefaultTimingFramesDelayMs`, or if the frame is at least
+  // `kDefaultOutliserFrameSizePercent` in size of average frame.
+  kDefaultTimingFramesDelayMs = 200,
+  kDefaultOutlierFrameSizePercent = 500,
+  // Maximum number of frames for what we store encode start timing information.
+  kMaxEncodeStartTimeListSize = 150,
+};
+
+enum VCMVideoProtection {
+  kProtectionNack,
+  kProtectionNackFEC,
+};
+
+// Callback class used for passing decoded frames which are ready to be
+// rendered.
+class VCMReceiveCallback {
+ public:
+  virtual int32_t FrameToRender(VideoFrame& videoFrame,  // NOLINT
+                                absl::optional<uint8_t> qp,
+                                TimeDelta decode_time,
+                                VideoContentType content_type) = 0;
+
+  virtual void OnDroppedFrames(uint32_t frames_dropped);
+
+  // Called when the current receive codec changes.
+  virtual void OnIncomingPayloadType(int payload_type);
+  virtual void OnDecoderInfoChanged(
+      const VideoDecoder::DecoderInfo& decoder_info);
+
+ protected:
+  virtual ~VCMReceiveCallback() {}
+};
+
+// Callback class used for informing the user of the incoming bit rate and frame
+// rate.
+class VCMReceiveStatisticsCallback {
+ public:
+  virtual void OnCompleteFrame(bool is_keyframe,
+                               size_t size_bytes,
+                               VideoContentType content_type) = 0;
+
+  virtual void OnDroppedFrames(uint32_t frames_dropped) = 0;
+
+  virtual void OnDiscardedPackets(uint32_t packets_discarded) = 0;
+
+  virtual void OnFrameBufferTimingsUpdated(int max_decode_ms,
+                                           int current_delay_ms,
+                                           int target_delay_ms,
+                                           int jitter_buffer_ms,
+                                           int min_playout_delay_ms,
+                                           int render_delay_ms) = 0;
+
+  virtual void OnTimingFrameInfoUpdated(const TimingFrameInfo& info) = 0;
+
+ protected:
+  virtual ~VCMReceiveStatisticsCallback() {}
+};
+
+// Callback class used for telling the user about what frame type needed to
+// continue decoding.
+// Typically a key frame when the stream has been corrupted in some way.
+class VCMFrameTypeCallback {
+ public:
+  virtual int32_t RequestKeyFrame() = 0;
+
+ protected:
+  virtual ~VCMFrameTypeCallback() {}
+};
+
+// Callback class used for telling the user about which packet sequence numbers
+// are currently
+// missing and need to be resent.
+// TODO(philipel): Deprecate VCMPacketRequestCallback
+//                 and use NackSender instead.
+class VCMPacketRequestCallback {
+ public:
+  virtual int32_t ResendPackets(const uint16_t* sequenceNumbers,
+                                uint16_t length) = 0;
+
+ protected:
+  virtual ~VCMPacketRequestCallback() {}
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
diff --git a/third_party/libwebrtc/modules/video_coding/include/video_error_codes.h b/third_party/libwebrtc/modules/video_coding/include/video_error_codes.h
new file mode 100644
index 0000000000..17146ce205
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/include/video_error_codes.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_INCLUDE_VIDEO_ERROR_CODES_H_
+#define MODULES_VIDEO_CODING_INCLUDE_VIDEO_ERROR_CODES_H_
+
+// NOTE: in sync with video_coding_module_defines.h
+
+// Define return values
+
+#define WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT 5
+#define WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME 4
+#define WEBRTC_VIDEO_CODEC_NO_OUTPUT 1
+#define WEBRTC_VIDEO_CODEC_OK 0
+#define WEBRTC_VIDEO_CODEC_ERROR -1
+#define WEBRTC_VIDEO_CODEC_MEMORY -3
+#define WEBRTC_VIDEO_CODEC_ERR_PARAMETER -4
+#define WEBRTC_VIDEO_CODEC_TIMEOUT -6
+#define WEBRTC_VIDEO_CODEC_UNINITIALIZED -7
+#define WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE -13
+#define WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED -15
+#define WEBRTC_VIDEO_CODEC_ENCODER_FAILURE -16
+
+#endif  // MODULES_VIDEO_CODING_INCLUDE_VIDEO_ERROR_CODES_H_
diff --git a/third_party/libwebrtc/modules/video_coding/internal_defines.h b/third_party/libwebrtc/modules/video_coding/internal_defines.h
new file mode 100644
index 0000000000..f753f200e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/internal_defines.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
+#define MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
+
+namespace webrtc {
+
+#define VCM_MAX(a, b) (((a) > (b)) ? (a) : (b))
+#define VCM_MIN(a, b) (((a) < (b)) ? (a) : (b))
+
+#define VCM_FLUSH_INDICATOR 4
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
diff --git a/third_party/libwebrtc/modules/video_coding/jitter_buffer.cc b/third_party/libwebrtc/modules/video_coding/jitter_buffer.cc
new file mode 100644
index 0000000000..39553c9f3f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/jitter_buffer.cc
@@ -0,0 +1,892 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/jitter_buffer.h"
+
+#include <algorithm>
+#include <limits>
+#include <utility>
+
+#include "api/units/timestamp.h"
+#include "modules/video_coding/frame_buffer.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/internal_defines.h"
+#include "modules/video_coding/jitter_buffer_common.h"
+#include "modules/video_coding/packet.h"
+#include "modules/video_coding/timing/inter_frame_delay.h"
+#include "modules/video_coding/timing/jitter_estimator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+// Use this rtt if no value has been reported.
+static const int64_t kDefaultRtt = 200;
+
+typedef std::pair<uint32_t, VCMFrameBuffer*> FrameListPair;
+
+bool IsKeyFrame(FrameListPair pair) {
+  return pair.second->FrameType() == VideoFrameType::kVideoFrameKey;
+}
+
+bool HasNonEmptyState(FrameListPair pair) {
+  return pair.second->GetState() != kStateEmpty;
+}
+
+void FrameList::InsertFrame(VCMFrameBuffer* frame) {
+  insert(rbegin().base(), FrameListPair(frame->Timestamp(), frame));
+}
+
+VCMFrameBuffer* FrameList::PopFrame(uint32_t timestamp) {
+  FrameList::iterator it = find(timestamp);
+  if (it == end())
+    return NULL;
+  VCMFrameBuffer* frame = it->second;
+  erase(it);
+  return frame;
+}
+
+VCMFrameBuffer* FrameList::Front() const {
+  return begin()->second;
+}
+
+VCMFrameBuffer* FrameList::Back() const {
+  return rbegin()->second;
+}
+
+int FrameList::RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
+                                          UnorderedFrameList* free_frames) {
+  int drop_count = 0;
+  FrameList::iterator it = begin();
+  while (!empty()) {
+    // Throw at least one frame.
+    it->second->Reset();
+    free_frames->push_back(it->second);
+    erase(it++);
+    ++drop_count;
+    if (it != end() &&
+        it->second->FrameType() == VideoFrameType::kVideoFrameKey) {
+      *key_frame_it = it;
+      return drop_count;
+    }
+  }
+  *key_frame_it = end();
+  return drop_count;
+}
+
+void FrameList::CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
+                                        UnorderedFrameList* free_frames) {
+  while (!empty()) {
+    VCMFrameBuffer* oldest_frame = Front();
+    bool remove_frame = false;
+    if (oldest_frame->GetState() == kStateEmpty && size() > 1) {
+      // This frame is empty, try to update the last decoded state and drop it
+      // if successful.
+      remove_frame = decoding_state->UpdateEmptyFrame(oldest_frame);
+    } else {
+      remove_frame = decoding_state->IsOldFrame(oldest_frame);
+    }
+    if (!remove_frame) {
+      break;
+    }
+    free_frames->push_back(oldest_frame);
+    erase(begin());
+  }
+}
+
+void FrameList::Reset(UnorderedFrameList* free_frames) {
+  while (!empty()) {
+    begin()->second->Reset();
+    free_frames->push_back(begin()->second);
+    erase(begin());
+  }
+}
+
+VCMJitterBuffer::VCMJitterBuffer(Clock* clock,
+                                 std::unique_ptr<EventWrapper> event,
+                                 const FieldTrialsView& field_trials)
+    : clock_(clock),
+      running_(false),
+      frame_event_(std::move(event)),
+      max_number_of_frames_(kStartNumberOfFrames),
+      free_frames_(),
+      decodable_frames_(),
+      incomplete_frames_(),
+      last_decoded_state_(),
+      first_packet_since_reset_(true),
+      num_consecutive_old_packets_(0),
+      num_packets_(0),
+      num_duplicated_packets_(0),
+      jitter_estimate_(clock, field_trials),
+      missing_sequence_numbers_(SequenceNumberLessThan()),
+      latest_received_sequence_number_(0),
+      max_nack_list_size_(0),
+      max_packet_age_to_nack_(0),
+      max_incomplete_time_ms_(0),
+      average_packets_per_frame_(0.0f),
+      frame_counter_(0) {
+  for (int i = 0; i < kStartNumberOfFrames; i++)
+    free_frames_.push_back(new VCMFrameBuffer());
+}
+
+VCMJitterBuffer::~VCMJitterBuffer() {
+  Stop();
+  for (UnorderedFrameList::iterator it = free_frames_.begin();
+       it != free_frames_.end(); ++it) {
+    delete *it;
+  }
+  for (FrameList::iterator it = incomplete_frames_.begin();
+       it != incomplete_frames_.end(); ++it) {
+    delete it->second;
+  }
+  for (FrameList::iterator it = decodable_frames_.begin();
+       it != decodable_frames_.end(); ++it) {
+    delete it->second;
+  }
+}
+
+void VCMJitterBuffer::Start() {
+  MutexLock lock(&mutex_);
+  running_ = true;
+
+  num_consecutive_old_packets_ = 0;
+  num_packets_ = 0;
+  num_duplicated_packets_ = 0;
+
+  // Start in a non-signaled state.
+  waiting_for_completion_.frame_size = 0;
+  waiting_for_completion_.timestamp = 0;
+  waiting_for_completion_.latest_packet_time = -1;
+  first_packet_since_reset_ = true;
+  last_decoded_state_.Reset();
+
+  decodable_frames_.Reset(&free_frames_);
+  incomplete_frames_.Reset(&free_frames_);
+}
+
+void VCMJitterBuffer::Stop() {
+  MutexLock lock(&mutex_);
+  running_ = false;
+  last_decoded_state_.Reset();
+
+  // Make sure we wake up any threads waiting on these events.
+  frame_event_->Set();
+}
+
+bool VCMJitterBuffer::Running() const {
+  MutexLock lock(&mutex_);
+  return running_;
+}
+
+void VCMJitterBuffer::Flush() {
+  MutexLock lock(&mutex_);
+  decodable_frames_.Reset(&free_frames_);
+  incomplete_frames_.Reset(&free_frames_);
+  last_decoded_state_.Reset();  // TODO(mikhal): sync reset.
+  num_consecutive_old_packets_ = 0;
+  // Also reset the jitter and delay estimates
+  jitter_estimate_.Reset();
+  inter_frame_delay_.Reset();
+  waiting_for_completion_.frame_size = 0;
+  waiting_for_completion_.timestamp = 0;
+  waiting_for_completion_.latest_packet_time = -1;
+  first_packet_since_reset_ = true;
+  missing_sequence_numbers_.clear();
+}
+
+int VCMJitterBuffer::num_packets() const {
+  MutexLock lock(&mutex_);
+  return num_packets_;
+}
+
+int VCMJitterBuffer::num_duplicated_packets() const {
+  MutexLock lock(&mutex_);
+  return num_duplicated_packets_;
+}
+
+// Returns immediately or a `max_wait_time_ms` ms event hang waiting for a
+// complete frame, `max_wait_time_ms` decided by caller.
+VCMEncodedFrame* VCMJitterBuffer::NextCompleteFrame(uint32_t max_wait_time_ms) {
+  MutexLock lock(&mutex_);
+  if (!running_) {
+    return nullptr;
+  }
+  CleanUpOldOrEmptyFrames();
+
+  if (decodable_frames_.empty() ||
+      decodable_frames_.Front()->GetState() != kStateComplete) {
+    const int64_t end_wait_time_ms =
+        clock_->TimeInMilliseconds() + max_wait_time_ms;
+    int64_t wait_time_ms = max_wait_time_ms;
+    while (wait_time_ms > 0) {
+      mutex_.Unlock();
+      const EventTypeWrapper ret =
+          frame_event_->Wait(static_cast<uint32_t>(wait_time_ms));
+      mutex_.Lock();
+      if (ret == kEventSignaled) {
+        // Are we shutting down the jitter buffer?
+        if (!running_) {
+          return nullptr;
+        }
+        // Finding oldest frame ready for decoder.
+        CleanUpOldOrEmptyFrames();
+        if (decodable_frames_.empty() ||
+            decodable_frames_.Front()->GetState() != kStateComplete) {
+          wait_time_ms = end_wait_time_ms - clock_->TimeInMilliseconds();
+        } else {
+          break;
+        }
+      } else {
+        break;
+      }
+    }
+  }
+  if (decodable_frames_.empty() ||
+      decodable_frames_.Front()->GetState() != kStateComplete) {
+    return nullptr;
+  }
+  return decodable_frames_.Front();
+}
+
+VCMEncodedFrame* VCMJitterBuffer::ExtractAndSetDecode(uint32_t timestamp) {
+  MutexLock lock(&mutex_);
+  if (!running_) {
+    return NULL;
+  }
+  // Extract the frame with the desired timestamp.
+  VCMFrameBuffer* frame = decodable_frames_.PopFrame(timestamp);
+  bool continuous = true;
+  if (!frame) {
+    frame = incomplete_frames_.PopFrame(timestamp);
+    if (frame)
+      continuous = last_decoded_state_.ContinuousFrame(frame);
+    else
+      return NULL;
+  }
+  // Frame pulled out from jitter buffer, update the jitter estimate.
+  const bool retransmitted = (frame->GetNackCount() > 0);
+  if (retransmitted) {
+    jitter_estimate_.FrameNacked();
+  } else if (frame->size() > 0) {
+    // Ignore retransmitted and empty frames.
+    if (waiting_for_completion_.latest_packet_time >= 0) {
+      UpdateJitterEstimate(waiting_for_completion_, true);
+    }
+    if (frame->GetState() == kStateComplete) {
+      UpdateJitterEstimate(*frame, false);
+    } else {
+      // Wait for this one to get complete.
+      waiting_for_completion_.frame_size = frame->size();
+      waiting_for_completion_.latest_packet_time = frame->LatestPacketTimeMs();
+      waiting_for_completion_.timestamp = frame->Timestamp();
+    }
+  }
+
+  // The state must be changed to decoding before cleaning up zero sized
+  // frames to avoid empty frames being cleaned up and then given to the
+  // decoder. Propagates the missing_frame bit.
+  frame->PrepareForDecode(continuous);
+
+  // We have a frame - update the last decoded state and nack list.
+  last_decoded_state_.SetState(frame);
+  DropPacketsFromNackList(last_decoded_state_.sequence_num());
+
+  UpdateAveragePacketsPerFrame(frame->NumPackets());
+
+  return frame;
+}
+
+// Release frame when done with decoding. Should never be used to release
+// frames from within the jitter buffer.
+void VCMJitterBuffer::ReleaseFrame(VCMEncodedFrame* frame) {
+  RTC_CHECK(frame != nullptr);
+  MutexLock lock(&mutex_);
+  VCMFrameBuffer* frame_buffer = static_cast<VCMFrameBuffer*>(frame);
+  RecycleFrameBuffer(frame_buffer);
+}
+
+// Gets frame to use for this timestamp. If no match, get empty frame.
+VCMFrameBufferEnum VCMJitterBuffer::GetFrame(const VCMPacket& packet,
+                                             VCMFrameBuffer** frame,
+                                             FrameList** frame_list) {
+  *frame = incomplete_frames_.PopFrame(packet.timestamp);
+  if (*frame != NULL) {
+    *frame_list = &incomplete_frames_;
+    return kNoError;
+  }
+  *frame = decodable_frames_.PopFrame(packet.timestamp);
+  if (*frame != NULL) {
+    *frame_list = &decodable_frames_;
+    return kNoError;
+  }
+
+  *frame_list = NULL;
+  // No match, return empty frame.
+  *frame = GetEmptyFrame();
+  if (*frame == NULL) {
+    // No free frame! Try to reclaim some...
+    RTC_LOG(LS_WARNING) << "Unable to get empty frame; Recycling.";
+    bool found_key_frame = RecycleFramesUntilKeyFrame();
+    *frame = GetEmptyFrame();
+    RTC_CHECK(*frame);
+    if (!found_key_frame) {
+      RecycleFrameBuffer(*frame);
+      return kFlushIndicator;
+    }
+  }
+  (*frame)->Reset();
+  return kNoError;
+}
+
+int64_t VCMJitterBuffer::LastPacketTime(const VCMEncodedFrame* frame,
+                                        bool* retransmitted) const {
+  RTC_DCHECK(retransmitted);
+  MutexLock lock(&mutex_);
+  const VCMFrameBuffer* frame_buffer =
+      static_cast<const VCMFrameBuffer*>(frame);
+  *retransmitted = (frame_buffer->GetNackCount() > 0);
+  return frame_buffer->LatestPacketTimeMs();
+}
+
+VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
+                                                 bool* retransmitted) {
+  MutexLock lock(&mutex_);
+
+  ++num_packets_;
+  // Does this packet belong to an old frame?
+  if (last_decoded_state_.IsOldPacket(&packet)) {
+    // Account only for media packets.
+    if (packet.sizeBytes > 0) {
+      num_consecutive_old_packets_++;
+    }
+    // Update last decoded sequence number if the packet arrived late and
+    // belongs to a frame with a timestamp equal to the last decoded
+    // timestamp.
+    last_decoded_state_.UpdateOldPacket(&packet);
+    DropPacketsFromNackList(last_decoded_state_.sequence_num());
+
+    // Also see if this old packet made more incomplete frames continuous.
+    FindAndInsertContinuousFramesWithState(last_decoded_state_);
+
+    if (num_consecutive_old_packets_ > kMaxConsecutiveOldPackets) {
+      RTC_LOG(LS_WARNING)
+          << num_consecutive_old_packets_
+          << " consecutive old packets received. Flushing the jitter buffer.";
+      Flush();
+      return kFlushIndicator;
+    }
+    return kOldPacket;
+  }
+
+  num_consecutive_old_packets_ = 0;
+
+  VCMFrameBuffer* frame;
+  FrameList* frame_list;
+  const VCMFrameBufferEnum error = GetFrame(packet, &frame, &frame_list);
+  if (error != kNoError)
+    return error;
+
+  Timestamp now = clock_->CurrentTime();
+  // We are keeping track of the first and latest seq numbers, and
+  // the number of wraps to be able to calculate how many packets we expect.
+  if (first_packet_since_reset_) {
+    // Now it's time to start estimating jitter
+    // reset the delay estimate.
+    inter_frame_delay_.Reset();
+  }
+
+  // Empty packets may bias the jitter estimate (lacking size component),
+  // therefore don't let empty packet trigger the following updates:
+  if (packet.video_header.frame_type != VideoFrameType::kEmptyFrame) {
+    if (waiting_for_completion_.timestamp == packet.timestamp) {
+      // This can get bad if we have a lot of duplicate packets,
+      // we will then count some packet multiple times.
+      waiting_for_completion_.frame_size += packet.sizeBytes;
+      waiting_for_completion_.latest_packet_time = now.ms();
+    } else if (waiting_for_completion_.latest_packet_time >= 0 &&
+               waiting_for_completion_.latest_packet_time + 2000 <= now.ms()) {
+      // A packet should never be more than two seconds late
+      UpdateJitterEstimate(waiting_for_completion_, true);
+      waiting_for_completion_.latest_packet_time = -1;
+      waiting_for_completion_.frame_size = 0;
+      waiting_for_completion_.timestamp = 0;
+    }
+  }
+
+  VCMFrameBufferStateEnum previous_state = frame->GetState();
+  // Insert packet.
+  FrameData frame_data;
+  frame_data.rtt_ms = kDefaultRtt;
+  frame_data.rolling_average_packets_per_frame = average_packets_per_frame_;
+  VCMFrameBufferEnum buffer_state =
+      frame->InsertPacket(packet, now.ms(), frame_data);
+
+  if (buffer_state > 0) {
+    if (first_packet_since_reset_) {
+      latest_received_sequence_number_ = packet.seqNum;
+      first_packet_since_reset_ = false;
+    } else {
+      if (IsPacketRetransmitted(packet)) {
+        frame->IncrementNackCount();
+      }
+      if (!UpdateNackList(packet.seqNum) &&
+          packet.video_header.frame_type != VideoFrameType::kVideoFrameKey) {
+        buffer_state = kFlushIndicator;
+      }
+
+      latest_received_sequence_number_ =
+          LatestSequenceNumber(latest_received_sequence_number_, packet.seqNum);
+    }
+  }
+
+  // Is the frame already in the decodable list?
+  bool continuous = IsContinuous(*frame);
+  switch (buffer_state) {
+    case kGeneralError:
+    case kTimeStampError:
+    case kSizeError: {
+      RecycleFrameBuffer(frame);
+      break;
+    }
+    case kCompleteSession: {
+      if (previous_state != kStateComplete) {
+        if (continuous) {
+          // Signal that we have a complete session.
+          frame_event_->Set();
+        }
+      }
+
+      *retransmitted = (frame->GetNackCount() > 0);
+      if (continuous) {
+        decodable_frames_.InsertFrame(frame);
+        FindAndInsertContinuousFrames(*frame);
+      } else {
+        incomplete_frames_.InsertFrame(frame);
+      }
+      break;
+    }
+    case kIncomplete: {
+      if (frame->GetState() == kStateEmpty &&
+          last_decoded_state_.UpdateEmptyFrame(frame)) {
+        RecycleFrameBuffer(frame);
+        return kNoError;
+      } else {
+        incomplete_frames_.InsertFrame(frame);
+      }
+      break;
+    }
+    case kNoError:
+    case kOutOfBoundsPacket:
+    case kDuplicatePacket: {
+      // Put back the frame where it came from.
+      if (frame_list != NULL) {
+        frame_list->InsertFrame(frame);
+      } else {
+        RecycleFrameBuffer(frame);
+      }
+      ++num_duplicated_packets_;
+      break;
+    }
+    case kFlushIndicator:
+      RecycleFrameBuffer(frame);
+      return kFlushIndicator;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  return buffer_state;
+}
+
+bool VCMJitterBuffer::IsContinuousInState(
+    const VCMFrameBuffer& frame,
+    const VCMDecodingState& decoding_state) const {
+  // Is this frame complete and continuous?
+  return (frame.GetState() == kStateComplete) &&
+         decoding_state.ContinuousFrame(&frame);
+}
+
+bool VCMJitterBuffer::IsContinuous(const VCMFrameBuffer& frame) const {
+  if (IsContinuousInState(frame, last_decoded_state_)) {
+    return true;
+  }
+  VCMDecodingState decoding_state;
+  decoding_state.CopyFrom(last_decoded_state_);
+  for (FrameList::const_iterator it = decodable_frames_.begin();
+       it != decodable_frames_.end(); ++it) {
+    VCMFrameBuffer* decodable_frame = it->second;
+    if (IsNewerTimestamp(decodable_frame->Timestamp(), frame.Timestamp())) {
+      break;
+    }
+    decoding_state.SetState(decodable_frame);
+    if (IsContinuousInState(frame, decoding_state)) {
+      return true;
+    }
+  }
+  return false;
+}
+
+void VCMJitterBuffer::FindAndInsertContinuousFrames(
+    const VCMFrameBuffer& new_frame) {
+  VCMDecodingState decoding_state;
+  decoding_state.CopyFrom(last_decoded_state_);
+  decoding_state.SetState(&new_frame);
+  FindAndInsertContinuousFramesWithState(decoding_state);
+}
+
+void VCMJitterBuffer::FindAndInsertContinuousFramesWithState(
+    const VCMDecodingState& original_decoded_state) {
+  // Copy original_decoded_state so we can move the state forward with each
+  // decodable frame we find.
+  VCMDecodingState decoding_state;
+  decoding_state.CopyFrom(original_decoded_state);
+
+  // When temporal layers are available, we search for a complete or decodable
+  // frame until we hit one of the following:
+  // 1. Continuous base or sync layer.
+  // 2. The end of the list was reached.
+  for (FrameList::iterator it = incomplete_frames_.begin();
+       it != incomplete_frames_.end();) {
+    VCMFrameBuffer* frame = it->second;
+    if (IsNewerTimestamp(original_decoded_state.time_stamp(),
+                         frame->Timestamp())) {
+      ++it;
+      continue;
+    }
+    if (IsContinuousInState(*frame, decoding_state)) {
+      decodable_frames_.InsertFrame(frame);
+      incomplete_frames_.erase(it++);
+      decoding_state.SetState(frame);
+    } else if (frame->TemporalId() <= 0) {
+      break;
+    } else {
+      ++it;
+    }
+  }
+}
+
+uint32_t VCMJitterBuffer::EstimatedJitterMs() {
+  MutexLock lock(&mutex_);
+  const double rtt_mult = 1.0f;
+  return jitter_estimate_.GetJitterEstimate(rtt_mult, absl::nullopt).ms();
+}
+
+void VCMJitterBuffer::SetNackSettings(size_t max_nack_list_size,
+                                      int max_packet_age_to_nack,
+                                      int max_incomplete_time_ms) {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK_GE(max_packet_age_to_nack, 0);
+  RTC_DCHECK_GE(max_incomplete_time_ms_, 0);
+  max_nack_list_size_ = max_nack_list_size;
+  max_packet_age_to_nack_ = max_packet_age_to_nack;
+  max_incomplete_time_ms_ = max_incomplete_time_ms;
+}
+
+int VCMJitterBuffer::NonContinuousOrIncompleteDuration() {
+  if (incomplete_frames_.empty()) {
+    return 0;
+  }
+  uint32_t start_timestamp = incomplete_frames_.Front()->Timestamp();
+  if (!decodable_frames_.empty()) {
+    start_timestamp = decodable_frames_.Back()->Timestamp();
+  }
+  return incomplete_frames_.Back()->Timestamp() - start_timestamp;
+}
+
+uint16_t VCMJitterBuffer::EstimatedLowSequenceNumber(
+    const VCMFrameBuffer& frame) const {
+  RTC_DCHECK_GE(frame.GetLowSeqNum(), 0);
+  if (frame.HaveFirstPacket())
+    return frame.GetLowSeqNum();
+
+  // This estimate is not accurate if more than one packet with lower sequence
+  // number is lost.
+  return frame.GetLowSeqNum() - 1;
+}
+
+std::vector<uint16_t> VCMJitterBuffer::GetNackList(bool* request_key_frame) {
+  MutexLock lock(&mutex_);
+  *request_key_frame = false;
+  if (last_decoded_state_.in_initial_state()) {
+    VCMFrameBuffer* next_frame = NextFrame();
+    const bool first_frame_is_key =
+        next_frame &&
+        next_frame->FrameType() == VideoFrameType::kVideoFrameKey &&
+        next_frame->HaveFirstPacket();
+    if (!first_frame_is_key) {
+      bool have_non_empty_frame =
+          decodable_frames_.end() != find_if(decodable_frames_.begin(),
+                                             decodable_frames_.end(),
+                                             HasNonEmptyState);
+      if (!have_non_empty_frame) {
+        have_non_empty_frame =
+            incomplete_frames_.end() != find_if(incomplete_frames_.begin(),
+                                                incomplete_frames_.end(),
+                                                HasNonEmptyState);
+      }
+      bool found_key_frame = RecycleFramesUntilKeyFrame();
+      if (!found_key_frame) {
+        *request_key_frame = have_non_empty_frame;
+        return std::vector<uint16_t>();
+      }
+    }
+  }
+  if (TooLargeNackList()) {
+    *request_key_frame = !HandleTooLargeNackList();
+  }
+  if (max_incomplete_time_ms_ > 0) {
+    int non_continuous_incomplete_duration =
+        NonContinuousOrIncompleteDuration();
+    if (non_continuous_incomplete_duration > 90 * max_incomplete_time_ms_) {
+      RTC_LOG_F(LS_WARNING) << "Too long non-decodable duration: "
+                            << non_continuous_incomplete_duration << " > "
+                            << 90 * max_incomplete_time_ms_;
+      FrameList::reverse_iterator rit = find_if(
+          incomplete_frames_.rbegin(), incomplete_frames_.rend(), IsKeyFrame);
+      if (rit == incomplete_frames_.rend()) {
+        // Request a key frame if we don't have one already.
+        *request_key_frame = true;
+        return std::vector<uint16_t>();
+      } else {
+        // Skip to the last key frame. If it's incomplete we will start
+        // NACKing it.
+        // Note that the estimated low sequence number is correct for VP8
+        // streams because only the first packet of a key frame is marked.
+        last_decoded_state_.Reset();
+        DropPacketsFromNackList(EstimatedLowSequenceNumber(*rit->second));
+      }
+    }
+  }
+  std::vector<uint16_t> nack_list(missing_sequence_numbers_.begin(),
+                                  missing_sequence_numbers_.end());
+  return nack_list;
+}
+
+VCMFrameBuffer* VCMJitterBuffer::NextFrame() const {
+  if (!decodable_frames_.empty())
+    return decodable_frames_.Front();
+  if (!incomplete_frames_.empty())
+    return incomplete_frames_.Front();
+  return NULL;
+}
+
+bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) {
+  // Make sure we don't add packets which are already too old to be decoded.
+  if (!last_decoded_state_.in_initial_state()) {
+    latest_received_sequence_number_ = LatestSequenceNumber(
+        latest_received_sequence_number_, last_decoded_state_.sequence_num());
+  }
+  if (IsNewerSequenceNumber(sequence_number,
+                            latest_received_sequence_number_)) {
+    // Push any missing sequence numbers to the NACK list.
+    for (uint16_t i = latest_received_sequence_number_ + 1;
+         IsNewerSequenceNumber(sequence_number, i); ++i) {
+      missing_sequence_numbers_.insert(missing_sequence_numbers_.end(), i);
+    }
+    if (TooLargeNackList() && !HandleTooLargeNackList()) {
+      RTC_LOG(LS_WARNING) << "Requesting key frame due to too large NACK list.";
+      return false;
+    }
+    if (MissingTooOldPacket(sequence_number) &&
+        !HandleTooOldPackets(sequence_number)) {
+      RTC_LOG(LS_WARNING)
+          << "Requesting key frame due to missing too old packets";
+      return false;
+    }
+  } else {
+    missing_sequence_numbers_.erase(sequence_number);
+  }
+  return true;
+}
+
+bool VCMJitterBuffer::TooLargeNackList() const {
+  return missing_sequence_numbers_.size() > max_nack_list_size_;
+}
+
+bool VCMJitterBuffer::HandleTooLargeNackList() {
+  // Recycle frames until the NACK list is small enough. It is likely cheaper to
+  // request a key frame than to retransmit this many missing packets.
+  RTC_LOG_F(LS_WARNING) << "NACK list has grown too large: "
+                        << missing_sequence_numbers_.size() << " > "
+                        << max_nack_list_size_;
+  bool key_frame_found = false;
+  while (TooLargeNackList()) {
+    key_frame_found = RecycleFramesUntilKeyFrame();
+  }
+  return key_frame_found;
+}
+
+bool VCMJitterBuffer::MissingTooOldPacket(
+    uint16_t latest_sequence_number) const {
+  if (missing_sequence_numbers_.empty()) {
+    return false;
+  }
+  const uint16_t age_of_oldest_missing_packet =
+      latest_sequence_number - *missing_sequence_numbers_.begin();
+  // Recycle frames if the NACK list contains too old sequence numbers as
+  // the packets may have already been dropped by the sender.
+  return age_of_oldest_missing_packet > max_packet_age_to_nack_;
+}
+
+bool VCMJitterBuffer::HandleTooOldPackets(uint16_t latest_sequence_number) {
+  bool key_frame_found = false;
+  const uint16_t age_of_oldest_missing_packet =
+      latest_sequence_number - *missing_sequence_numbers_.begin();
+  RTC_LOG_F(LS_WARNING) << "NACK list contains too old sequence numbers: "
+                        << age_of_oldest_missing_packet << " > "
+                        << max_packet_age_to_nack_;
+  while (MissingTooOldPacket(latest_sequence_number)) {
+    key_frame_found = RecycleFramesUntilKeyFrame();
+  }
+  return key_frame_found;
+}
+
+void VCMJitterBuffer::DropPacketsFromNackList(
+    uint16_t last_decoded_sequence_number) {
+  // Erase all sequence numbers from the NACK list which we won't need any
+  // longer.
+  missing_sequence_numbers_.erase(
+      missing_sequence_numbers_.begin(),
+      missing_sequence_numbers_.upper_bound(last_decoded_sequence_number));
+}
+
+VCMFrameBuffer* VCMJitterBuffer::GetEmptyFrame() {
+  if (free_frames_.empty()) {
+    if (!TryToIncreaseJitterBufferSize()) {
+      return NULL;
+    }
+  }
+  VCMFrameBuffer* frame = free_frames_.front();
+  free_frames_.pop_front();
+  return frame;
+}
+
+bool VCMJitterBuffer::TryToIncreaseJitterBufferSize() {
+  if (max_number_of_frames_ >= kMaxNumberOfFrames)
+    return false;
+  free_frames_.push_back(new VCMFrameBuffer());
+  ++max_number_of_frames_;
+  return true;
+}
+
+// Recycle oldest frames up to a key frame, used if jitter buffer is completely
+// full.
+bool VCMJitterBuffer::RecycleFramesUntilKeyFrame() {
+  // First release incomplete frames, and only release decodable frames if there
+  // are no incomplete ones.
+  FrameList::iterator key_frame_it;
+  bool key_frame_found = false;
+  int dropped_frames = 0;
+  dropped_frames += incomplete_frames_.RecycleFramesUntilKeyFrame(
+      &key_frame_it, &free_frames_);
+  key_frame_found = key_frame_it != incomplete_frames_.end();
+  if (dropped_frames == 0) {
+    dropped_frames += decodable_frames_.RecycleFramesUntilKeyFrame(
+        &key_frame_it, &free_frames_);
+    key_frame_found = key_frame_it != decodable_frames_.end();
+  }
+  if (key_frame_found) {
+    RTC_LOG(LS_INFO) << "Found key frame while dropping frames.";
+    // Reset last decoded state to make sure the next frame decoded is a key
+    // frame, and start NACKing from here.
+    last_decoded_state_.Reset();
+    DropPacketsFromNackList(EstimatedLowSequenceNumber(*key_frame_it->second));
+  } else if (decodable_frames_.empty()) {
+    // All frames dropped. Reset the decoding state and clear missing sequence
+    // numbers as we're starting fresh.
+    last_decoded_state_.Reset();
+    missing_sequence_numbers_.clear();
+  }
+  return key_frame_found;
+}
+
+void VCMJitterBuffer::UpdateAveragePacketsPerFrame(int current_number_packets) {
+  if (frame_counter_ > kFastConvergeThreshold) {
+    average_packets_per_frame_ =
+        average_packets_per_frame_ * (1 - kNormalConvergeMultiplier) +
+        current_number_packets * kNormalConvergeMultiplier;
+  } else if (frame_counter_ > 0) {
+    average_packets_per_frame_ =
+        average_packets_per_frame_ * (1 - kFastConvergeMultiplier) +
+        current_number_packets * kFastConvergeMultiplier;
+    frame_counter_++;
+  } else {
+    average_packets_per_frame_ = current_number_packets;
+    frame_counter_++;
+  }
+}
+
+// Must be called under the critical section `mutex_`.
+void VCMJitterBuffer::CleanUpOldOrEmptyFrames() {
+  decodable_frames_.CleanUpOldOrEmptyFrames(&last_decoded_state_,
+                                            &free_frames_);
+  incomplete_frames_.CleanUpOldOrEmptyFrames(&last_decoded_state_,
+                                             &free_frames_);
+  if (!last_decoded_state_.in_initial_state()) {
+    DropPacketsFromNackList(last_decoded_state_.sequence_num());
+  }
+}
+
+// Must be called from within `mutex_`.
+bool VCMJitterBuffer::IsPacketRetransmitted(const VCMPacket& packet) const {
+  return missing_sequence_numbers_.find(packet.seqNum) !=
+         missing_sequence_numbers_.end();
+}
+
+// Must be called under the critical section `mutex_`. Should never be
+// called with retransmitted frames, they must be filtered out before this
+// function is called.
+void VCMJitterBuffer::UpdateJitterEstimate(const VCMJitterSample& sample,
+                                           bool incomplete_frame) {
+  if (sample.latest_packet_time == -1) {
+    return;
+  }
+  UpdateJitterEstimate(sample.latest_packet_time, sample.timestamp,
+                       sample.frame_size, incomplete_frame);
+}
+
+// Must be called under the critical section mutex_. Should never be
+// called with retransmitted frames, they must be filtered out before this
+// function is called.
+void VCMJitterBuffer::UpdateJitterEstimate(const VCMFrameBuffer& frame,
+                                           bool incomplete_frame) {
+  if (frame.LatestPacketTimeMs() == -1) {
+    return;
+  }
+  // No retransmitted frames should be a part of the jitter
+  // estimate.
+  UpdateJitterEstimate(frame.LatestPacketTimeMs(), frame.Timestamp(),
+                       frame.size(), incomplete_frame);
+}
+
+// Must be called under the critical section `mutex_`. Should never be
+// called with retransmitted frames, they must be filtered out before this
+// function is called.
+void VCMJitterBuffer::UpdateJitterEstimate(int64_t latest_packet_time_ms,
+                                           uint32_t timestamp,
+                                           unsigned int frame_size,
+                                           bool /*incomplete_frame*/) {
+  if (latest_packet_time_ms == -1) {
+    return;
+  }
+  auto frame_delay = inter_frame_delay_.CalculateDelay(
+      timestamp, Timestamp::Millis(latest_packet_time_ms));
+
+  bool not_reordered = frame_delay.has_value();
+  // Filter out frames which have been reordered in time by the network
+  if (not_reordered) {
+    // Update the jitter estimate with the new samples
+    jitter_estimate_.UpdateEstimate(*frame_delay, DataSize::Bytes(frame_size));
+  }
+}
+
+void VCMJitterBuffer::RecycleFrameBuffer(VCMFrameBuffer* frame) {
+  frame->Reset();
+  free_frames_.push_back(frame);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/jitter_buffer.h b/third_party/libwebrtc/modules/video_coding/jitter_buffer.h
new file mode 100644
index 0000000000..7ca8953428
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/jitter_buffer.h
@@ -0,0 +1,275 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_JITTER_BUFFER_H_
+#define MODULES_VIDEO_CODING_JITTER_BUFFER_H_
+
+#include <list>
+#include <map>
+#include <memory>
+#include <set>
+#include <vector>
+
+#include "api/field_trials_view.h"
+#include "modules/include/module_common_types.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/video_coding/decoding_state.h"
+#include "modules/video_coding/event_wrapper.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "modules/video_coding/jitter_buffer_common.h"
+#include "modules/video_coding/timing/inter_frame_delay.h"
+#include "modules/video_coding/timing/jitter_estimator.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// forward declarations
+class Clock;
+class VCMFrameBuffer;
+class VCMPacket;
+class VCMEncodedFrame;
+
+typedef std::list<VCMFrameBuffer*> UnorderedFrameList;
+
+struct VCMJitterSample {
+  VCMJitterSample() : timestamp(0), frame_size(0), latest_packet_time(-1) {}
+  uint32_t timestamp;
+  uint32_t frame_size;
+  int64_t latest_packet_time;
+};
+
+class TimestampLessThan {
+ public:
+  bool operator()(uint32_t timestamp1, uint32_t timestamp2) const {
+    return IsNewerTimestamp(timestamp2, timestamp1);
+  }
+};
+
+class FrameList
+    : public std::map<uint32_t, VCMFrameBuffer*, TimestampLessThan> {
+ public:
+  void InsertFrame(VCMFrameBuffer* frame);
+  VCMFrameBuffer* PopFrame(uint32_t timestamp);
+  VCMFrameBuffer* Front() const;
+  VCMFrameBuffer* Back() const;
+  int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
+                                 UnorderedFrameList* free_frames);
+  void CleanUpOldOrEmptyFrames(VCMDecodingState* decoding_state,
+                               UnorderedFrameList* free_frames);
+  void Reset(UnorderedFrameList* free_frames);
+};
+
+class VCMJitterBuffer {
+ public:
+  VCMJitterBuffer(Clock* clock,
+                  std::unique_ptr<EventWrapper> event,
+                  const FieldTrialsView& field_trials);
+
+  ~VCMJitterBuffer();
+
+  VCMJitterBuffer(const VCMJitterBuffer&) = delete;
+  VCMJitterBuffer& operator=(const VCMJitterBuffer&) = delete;
+
+  // Initializes and starts jitter buffer.
+  void Start();
+
+  // Signals all internal events and stops the jitter buffer.
+  void Stop();
+
+  // Returns true if the jitter buffer is running.
+  bool Running() const;
+
+  // Empty the jitter buffer of all its data.
+  void Flush();
+
+  // Gets number of packets received.
+  int num_packets() const;
+
+  // Gets number of duplicated packets received.
+  int num_duplicated_packets() const;
+
+  // Wait `max_wait_time_ms` for a complete frame to arrive.
+  // If found, a pointer to the frame is returned. Returns nullptr otherwise.
+  VCMEncodedFrame* NextCompleteFrame(uint32_t max_wait_time_ms);
+
+  // Extract frame corresponding to input timestamp.
+  // Frame will be set to a decoding state.
+  VCMEncodedFrame* ExtractAndSetDecode(uint32_t timestamp);
+
+  // Releases a frame returned from the jitter buffer, should be called when
+  // done with decoding.
+  void ReleaseFrame(VCMEncodedFrame* frame);
+
+  // Returns the time in ms when the latest packet was inserted into the frame.
+  // Retransmitted is set to true if any of the packets belonging to the frame
+  // has been retransmitted.
+  int64_t LastPacketTime(const VCMEncodedFrame* frame,
+                         bool* retransmitted) const;
+
+  // Inserts a packet into a frame returned from GetFrame().
+  // If the return value is <= 0, `frame` is invalidated and the pointer must
+  // be dropped after this function returns.
+  VCMFrameBufferEnum InsertPacket(const VCMPacket& packet, bool* retransmitted);
+
+  // Returns the estimated jitter in milliseconds.
+  uint32_t EstimatedJitterMs();
+
+  void SetNackSettings(size_t max_nack_list_size,
+                       int max_packet_age_to_nack,
+                       int max_incomplete_time_ms);
+
+  // Returns a list of the sequence numbers currently missing.
+  std::vector<uint16_t> GetNackList(bool* request_key_frame);
+
+ private:
+  class SequenceNumberLessThan {
+   public:
+    bool operator()(const uint16_t& sequence_number1,
+                    const uint16_t& sequence_number2) const {
+      return IsNewerSequenceNumber(sequence_number2, sequence_number1);
+    }
+  };
+  typedef std::set<uint16_t, SequenceNumberLessThan> SequenceNumberSet;
+
+  // Gets the frame assigned to the timestamp of the packet. May recycle
+  // existing frames if no free frames are available. Returns an error code if
+  // failing, or kNoError on success. `frame_list` contains which list the
+  // packet was in, or NULL if it was not in a FrameList (a new frame).
+  VCMFrameBufferEnum GetFrame(const VCMPacket& packet,
+                              VCMFrameBuffer** frame,
+                              FrameList** frame_list)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Returns true if `frame` is continuous in `decoding_state`, not taking
+  // decodable frames into account.
+  bool IsContinuousInState(const VCMFrameBuffer& frame,
+                           const VCMDecodingState& decoding_state) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Returns true if `frame` is continuous in the `last_decoded_state_`, taking
+  // all decodable frames into account.
+  bool IsContinuous(const VCMFrameBuffer& frame) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Looks for frames in `incomplete_frames_` which are continuous in the
+  // provided `decoded_state`. Starts the search from the timestamp of
+  // `decoded_state`.
+  void FindAndInsertContinuousFramesWithState(
+      const VCMDecodingState& decoded_state)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Looks for frames in `incomplete_frames_` which are continuous in
+  // `last_decoded_state_` taking all decodable frames into account. Starts
+  // the search from `new_frame`.
+  void FindAndInsertContinuousFrames(const VCMFrameBuffer& new_frame)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  VCMFrameBuffer* NextFrame() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Returns true if the NACK list was updated to cover sequence numbers up to
+  // `sequence_number`. If false a key frame is needed to get into a state where
+  // we can continue decoding.
+  bool UpdateNackList(uint16_t sequence_number)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  bool TooLargeNackList() const;
+  // Returns true if the NACK list was reduced without problem. If false a key
+  // frame is needed to get into a state where we can continue decoding.
+  bool HandleTooLargeNackList() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  bool MissingTooOldPacket(uint16_t latest_sequence_number) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Returns true if the too old packets was successfully removed from the NACK
+  // list. If false, a key frame is needed to get into a state where we can
+  // continue decoding.
+  bool HandleTooOldPackets(uint16_t latest_sequence_number)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Drops all packets in the NACK list up until `last_decoded_sequence_number`.
+  void DropPacketsFromNackList(uint16_t last_decoded_sequence_number);
+
+  // Gets an empty frame, creating a new frame if necessary (i.e. increases
+  // jitter buffer size).
+  VCMFrameBuffer* GetEmptyFrame() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Attempts to increase the size of the jitter buffer. Returns true on
+  // success, false otherwise.
+  bool TryToIncreaseJitterBufferSize() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Recycles oldest frames until a key frame is found. Used if jitter buffer is
+  // completely full. Returns true if a key frame was found.
+  bool RecycleFramesUntilKeyFrame() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Update rolling average of packets per frame.
+  void UpdateAveragePacketsPerFrame(int current_number_packets_);
+
+  // Cleans the frame list in the JB from old/empty frames.
+  // Should only be called prior to actual use.
+  void CleanUpOldOrEmptyFrames() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Returns true if `packet` is likely to have been retransmitted.
+  bool IsPacketRetransmitted(const VCMPacket& packet) const;
+
+  // The following three functions update the jitter estimate with the
+  // payload size, receive time and RTP timestamp of a frame.
+  void UpdateJitterEstimate(const VCMJitterSample& sample,
+                            bool incomplete_frame);
+  void UpdateJitterEstimate(const VCMFrameBuffer& frame, bool incomplete_frame);
+  void UpdateJitterEstimate(int64_t latest_packet_time_ms,
+                            uint32_t timestamp,
+                            unsigned int frame_size,
+                            bool incomplete_frame);
+
+  int NonContinuousOrIncompleteDuration() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  uint16_t EstimatedLowSequenceNumber(const VCMFrameBuffer& frame) const;
+
+  // Reset frame buffer and return it to free_frames_.
+  void RecycleFrameBuffer(VCMFrameBuffer* frame)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock* clock_;
+  // If we are running (have started) or not.
+  bool running_;
+  mutable Mutex mutex_;
+  // Event to signal when we have a frame ready for decoder.
+  std::unique_ptr<EventWrapper> frame_event_;
+  // Number of allocated frames.
+  int max_number_of_frames_;
+  UnorderedFrameList free_frames_ RTC_GUARDED_BY(mutex_);
+  FrameList decodable_frames_ RTC_GUARDED_BY(mutex_);
+  FrameList incomplete_frames_ RTC_GUARDED_BY(mutex_);
+  VCMDecodingState last_decoded_state_ RTC_GUARDED_BY(mutex_);
+  bool first_packet_since_reset_;
+
+  // Number of packets in a row that have been too old.
+  int num_consecutive_old_packets_;
+  // Number of packets received.
+  int num_packets_ RTC_GUARDED_BY(mutex_);
+  // Number of duplicated packets received.
+  int num_duplicated_packets_ RTC_GUARDED_BY(mutex_);
+
+  // Jitter estimation.
+  // Filter for estimating jitter.
+  JitterEstimator jitter_estimate_;
+  // Calculates network delays used for jitter calculations.
+  InterFrameDelay inter_frame_delay_;
+  VCMJitterSample waiting_for_completion_;
+
+  // Holds the internal NACK list (the missing sequence numbers).
+  SequenceNumberSet missing_sequence_numbers_;
+  uint16_t latest_received_sequence_number_;
+  size_t max_nack_list_size_;
+  int max_packet_age_to_nack_;  // Measured in sequence numbers.
+  int max_incomplete_time_ms_;
+
+  // Estimated rolling average of packets per frame
+  float average_packets_per_frame_;
+  // average_packets_per_frame converges fast if we have fewer than this many
+  // frames.
+  int frame_counter_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_JITTER_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/jitter_buffer_common.h b/third_party/libwebrtc/modules/video_coding/jitter_buffer_common.h
new file mode 100644
index 0000000000..6ccfe39199
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/jitter_buffer_common.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
+#define MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
+
+namespace webrtc {
+
+// Used to estimate rolling average of packets per frame.
+static const float kFastConvergeMultiplier = 0.4f;
+static const float kNormalConvergeMultiplier = 0.2f;
+
+enum { kMaxNumberOfFrames = 300 };
+enum { kStartNumberOfFrames = 6 };
+enum { kMaxVideoDelayMs = 10000 };
+enum { kPacketsPerFrameMultiplier = 5 };
+enum { kFastConvergeThreshold = 5 };
+
+enum VCMJitterBufferEnum {
+  kMaxConsecutiveOldFrames = 60,
+  kMaxConsecutiveOldPackets = 300,
+  // TODO(sprang): Reduce this limit once codecs don't sometimes wildly
+  // overshoot bitrate target.
+  kMaxPacketsInSession = 1400,      // Allows ~2MB frames.
+  kBufferIncStepSizeBytes = 30000,  // >20 packets.
+  kMaxJBFrameSizeBytes = 4000000    // sanity don't go above 4Mbyte.
+};
+
+enum VCMFrameBufferEnum {
+  kOutOfBoundsPacket = -7,
+  kNotInitialized = -6,
+  kOldPacket = -5,
+  kGeneralError = -4,
+  kFlushIndicator = -3,  // Indicator that a flush has occurred.
+  kTimeStampError = -2,
+  kSizeError = -1,
+  kNoError = 0,
+  kIncomplete = 1,       // Frame incomplete.
+  kCompleteSession = 3,  // at least one layer in the frame complete.
+  kDuplicatePacket = 5   // We're receiving a duplicate packet.
+};
+
+enum VCMFrameBufferStateEnum {
+  kStateEmpty,       // frame popped by the RTP receiver
+  kStateIncomplete,  // frame that have one or more packet(s) stored
+  kStateComplete,    // frame that have all packets
+};
+
+enum { kH264StartCodeLengthBytes = 4 };
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
diff --git a/third_party/libwebrtc/modules/video_coding/jitter_buffer_unittest.cc b/third_party/libwebrtc/modules/video_coding/jitter_buffer_unittest.cc
new file mode 100644
index 0000000000..cc791fe110
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/jitter_buffer_unittest.cc
@@ -0,0 +1,1848 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/jitter_buffer.h"
+
+#include <list>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "common_video/h264/h264_common.h"
+#include "modules/video_coding/frame_buffer.h"
+#include "modules/video_coding/media_opt_util.h"
+#include "modules/video_coding/packet.h"
+#include "modules/video_coding/test/stream_generator.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/metrics.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+
+class TestBasicJitterBuffer : public ::testing::Test {
+ protected:
+  TestBasicJitterBuffer() {}
+  void SetUp() override {
+    clock_.reset(new SimulatedClock(0));
+    jitter_buffer_.reset(new VCMJitterBuffer(
+        clock_.get(), absl::WrapUnique(EventWrapper::Create()), field_trials_));
+    jitter_buffer_->Start();
+    seq_num_ = 1234;
+    timestamp_ = 0;
+    size_ = 1400;
+    // Data vector -  0, 0, 0x80, 3, 4, 5, 6, 7, 8, 9, 0, 0, 0x80, 3....
+    data_[0] = 0;
+    data_[1] = 0;
+    data_[2] = 0x80;
+    int count = 3;
+    for (unsigned int i = 3; i < sizeof(data_) - 3; ++i) {
+      data_[i] = count;
+      count++;
+      if (count == 10) {
+        data_[i + 1] = 0;
+        data_[i + 2] = 0;
+        data_[i + 3] = 0x80;
+        count = 3;
+        i += 3;
+      }
+    }
+    RTPHeader rtp_header;
+    RTPVideoHeader video_header;
+    rtp_header.sequenceNumber = seq_num_;
+    rtp_header.timestamp = timestamp_;
+    rtp_header.markerBit = true;
+    video_header.codec = kVideoCodecGeneric;
+    video_header.is_first_packet_in_frame = true;
+    video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+    packet_.reset(new VCMPacket(data_, size_, rtp_header, video_header,
+                                /*ntp_time_ms=*/0, clock_->CurrentTime()));
+  }
+
+  VCMEncodedFrame* DecodeCompleteFrame() {
+    VCMEncodedFrame* found_frame = jitter_buffer_->NextCompleteFrame(10);
+    if (!found_frame)
+      return nullptr;
+    return jitter_buffer_->ExtractAndSetDecode(found_frame->Timestamp());
+  }
+
+  void CheckOutFrame(VCMEncodedFrame* frame_out,
+                     unsigned int size,
+                     bool startCode) {
+    ASSERT_TRUE(frame_out);
+
+    const uint8_t* outData = frame_out->data();
+    unsigned int i = 0;
+
+    if (startCode) {
+      EXPECT_EQ(0, outData[0]);
+      EXPECT_EQ(0, outData[1]);
+      EXPECT_EQ(0, outData[2]);
+      EXPECT_EQ(1, outData[3]);
+      i += 4;
+    }
+
+    EXPECT_EQ(size, frame_out->size());
+    int count = 3;
+    for (; i < size; i++) {
+      if (outData[i] == 0 && outData[i + 1] == 0 && outData[i + 2] == 0x80) {
+        i += 2;
+      } else if (startCode && outData[i] == 0 && outData[i + 1] == 0) {
+        EXPECT_EQ(0, outData[0]);
+        EXPECT_EQ(0, outData[1]);
+        EXPECT_EQ(0, outData[2]);
+        EXPECT_EQ(1, outData[3]);
+        i += 3;
+      } else {
+        EXPECT_EQ(count, outData[i]);
+        count++;
+        if (count == 10) {
+          count = 3;
+        }
+      }
+    }
+  }
+
+  uint16_t seq_num_;
+  uint32_t timestamp_;
+  int size_;
+  uint8_t data_[1500];
+  test::ScopedKeyValueConfig field_trials_;
+  std::unique_ptr<VCMPacket> packet_;
+  std::unique_ptr<SimulatedClock> clock_;
+  std::unique_ptr<VCMJitterBuffer> jitter_buffer_;
+};
+
+class TestRunningJitterBuffer : public ::testing::Test {
+ protected:
+  enum { kDataBufferSize = 10 };
+
+  virtual void SetUp() {
+    clock_.reset(new SimulatedClock(0));
+    max_nack_list_size_ = 150;
+    oldest_packet_to_nack_ = 250;
+    jitter_buffer_ = new VCMJitterBuffer(
+        clock_.get(), absl::WrapUnique(EventWrapper::Create()), field_trials_);
+    stream_generator_ = new StreamGenerator(0, clock_->TimeInMilliseconds());
+    jitter_buffer_->Start();
+    jitter_buffer_->SetNackSettings(max_nack_list_size_, oldest_packet_to_nack_,
+                                    0);
+    memset(data_buffer_, 0, kDataBufferSize);
+  }
+
+  virtual void TearDown() {
+    jitter_buffer_->Stop();
+    delete stream_generator_;
+    delete jitter_buffer_;
+  }
+
+  VCMFrameBufferEnum InsertPacketAndPop(int index) {
+    VCMPacket packet;
+    packet.dataPtr = data_buffer_;
+    bool packet_available = stream_generator_->PopPacket(&packet, index);
+    EXPECT_TRUE(packet_available);
+    if (!packet_available)
+      return kGeneralError;  // Return here to avoid crashes below.
+    bool retransmitted = false;
+    return jitter_buffer_->InsertPacket(packet, &retransmitted);
+  }
+
+  VCMFrameBufferEnum InsertPacket(int index) {
+    VCMPacket packet;
+    packet.dataPtr = data_buffer_;
+    bool packet_available = stream_generator_->GetPacket(&packet, index);
+    EXPECT_TRUE(packet_available);
+    if (!packet_available)
+      return kGeneralError;  // Return here to avoid crashes below.
+    bool retransmitted = false;
+    return jitter_buffer_->InsertPacket(packet, &retransmitted);
+  }
+
+  VCMFrameBufferEnum InsertFrame(VideoFrameType frame_type) {
+    stream_generator_->GenerateFrame(
+        frame_type, (frame_type != VideoFrameType::kEmptyFrame) ? 1 : 0,
+        (frame_type == VideoFrameType::kEmptyFrame) ? 1 : 0,
+        clock_->TimeInMilliseconds());
+    VCMFrameBufferEnum ret = InsertPacketAndPop(0);
+    clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+    return ret;
+  }
+
+  VCMFrameBufferEnum InsertFrames(int num_frames, VideoFrameType frame_type) {
+    VCMFrameBufferEnum ret_for_all = kNoError;
+    for (int i = 0; i < num_frames; ++i) {
+      VCMFrameBufferEnum ret = InsertFrame(frame_type);
+      if (ret < kNoError) {
+        ret_for_all = ret;
+      } else if (ret_for_all >= kNoError) {
+        ret_for_all = ret;
+      }
+    }
+    return ret_for_all;
+  }
+
+  void DropFrame(int num_packets) {
+    stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta,
+                                     num_packets, 0,
+                                     clock_->TimeInMilliseconds());
+    for (int i = 0; i < num_packets; ++i)
+      stream_generator_->DropLastPacket();
+    clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+  }
+
+  bool DecodeCompleteFrame() {
+    VCMEncodedFrame* found_frame = jitter_buffer_->NextCompleteFrame(0);
+    if (!found_frame)
+      return false;
+
+    VCMEncodedFrame* frame =
+        jitter_buffer_->ExtractAndSetDecode(found_frame->Timestamp());
+    bool ret = (frame != NULL);
+    jitter_buffer_->ReleaseFrame(frame);
+    return ret;
+  }
+
+  test::ScopedKeyValueConfig field_trials_;
+  VCMJitterBuffer* jitter_buffer_;
+  StreamGenerator* stream_generator_;
+  std::unique_ptr<SimulatedClock> clock_;
+  size_t max_nack_list_size_;
+  int oldest_packet_to_nack_;
+  uint8_t data_buffer_[kDataBufferSize];
+};
+
+class TestJitterBufferNack : public TestRunningJitterBuffer {
+ protected:
+  TestJitterBufferNack() {}
+  virtual void SetUp() { TestRunningJitterBuffer::SetUp(); }
+
+  virtual void TearDown() { TestRunningJitterBuffer::TearDown(); }
+};
+
+TEST_F(TestBasicJitterBuffer, StopRunning) {
+  jitter_buffer_->Stop();
+  EXPECT_TRUE(NULL == DecodeCompleteFrame());
+  jitter_buffer_->Start();
+
+  // No packets inserted.
+  EXPECT_TRUE(NULL == DecodeCompleteFrame());
+}
+
+TEST_F(TestBasicJitterBuffer, SinglePacketFrame) {
+  // Always start with a complete key frame when not allowing errors.
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->timestamp += 123 * 90;
+
+  // Insert the packet to the jitter buffer and get a frame.
+  bool retransmitted = false;
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  CheckOutFrame(frame_out, size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, DualPacketFrame) {
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  // Should not be complete.
+  EXPECT_TRUE(frame_out == NULL);
+
+  ++seq_num_;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+  CheckOutFrame(frame_out, 2 * size_, false);
+
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, 100PacketKeyFrame) {
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+
+  // Frame should not be complete.
+  EXPECT_TRUE(frame_out == NULL);
+
+  // Insert 98 frames.
+  int loop = 0;
+  do {
+    seq_num_++;
+    packet_->video_header.is_first_packet_in_frame = false;
+    packet_->markerBit = false;
+    packet_->seqNum = seq_num_;
+
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+    loop++;
+  } while (loop < 98);
+
+  // Insert last packet.
+  ++seq_num_;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+
+  CheckOutFrame(frame_out, 100 * size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, 100PacketDeltaFrame) {
+  // Always start with a complete key frame.
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_FALSE(frame_out == NULL);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  ++seq_num_;
+  packet_->seqNum = seq_num_;
+  packet_->markerBit = false;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet_->timestamp += 33 * 90;
+
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+
+  // Frame should not be complete.
+  EXPECT_TRUE(frame_out == NULL);
+
+  packet_->video_header.is_first_packet_in_frame = false;
+  // Insert 98 frames.
+  int loop = 0;
+  do {
+    ++seq_num_;
+    packet_->seqNum = seq_num_;
+
+    // Insert a packet into a frame.
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+    loop++;
+  } while (loop < 98);
+
+  // Insert the last packet.
+  ++seq_num_;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+
+  CheckOutFrame(frame_out, 100 * size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, PacketReorderingReverseOrder) {
+  // Insert the "first" packet last.
+  seq_num_ += 100;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+
+  EXPECT_TRUE(frame_out == NULL);
+
+  // Insert 98 packets.
+  int loop = 0;
+  do {
+    seq_num_--;
+    packet_->video_header.is_first_packet_in_frame = false;
+    packet_->markerBit = false;
+    packet_->seqNum = seq_num_;
+
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+    loop++;
+  } while (loop < 98);
+
+  // Insert the last packet.
+  seq_num_--;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+
+  CheckOutFrame(frame_out, 100 * size_, false);
+
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, FrameReordering2Frames2PacketsEach) {
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+
+  EXPECT_TRUE(frame_out == NULL);
+
+  seq_num_++;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  // check that we fail to get frame since seqnum is not continuous
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out == NULL);
+
+  seq_num_ -= 3;
+  timestamp_ -= 33 * 90;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+
+  // It should not be complete.
+  EXPECT_TRUE(frame_out == NULL);
+
+  seq_num_++;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+  CheckOutFrame(frame_out, 2 * size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  frame_out = DecodeCompleteFrame();
+  CheckOutFrame(frame_out, 2 * size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, TestReorderingWithPadding) {
+  jitter_buffer_->SetNackSettings(kMaxNumberOfFrames, kMaxNumberOfFrames, 0);
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+
+  // Send in an initial good packet/frame (Frame A) to start things off.
+  bool retransmitted = false;
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out != NULL);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  // Now send in a complete delta frame (Frame C), but with a sequence number
+  // gap. No pic index either, so no temporal scalability cheating :)
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  // Leave a gap of 2 sequence numbers and two frames.
+  packet_->seqNum = seq_num_ + 3;
+  packet_->timestamp = timestamp_ + (66 * 90);
+  // Still isFirst = marker = true.
+  // Session should be complete (frame is complete), but there's nothing to
+  // decode yet.
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out == NULL);
+
+  // Now send in a complete delta frame (Frame B) that is continuous from A, but
+  // doesn't fill the full gap to C. The rest of the gap is going to be padding.
+  packet_->seqNum = seq_num_ + 1;
+  packet_->timestamp = timestamp_ + (33 * 90);
+  // Still isFirst = marker = true.
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out != NULL);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  // But Frame C isn't continuous yet.
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out == NULL);
+
+  // Add in the padding. These are empty packets (data length is 0) with no
+  // marker bit and matching the timestamp of Frame B.
+  RTPHeader rtp_header;
+  RTPVideoHeader video_header;
+  rtp_header.sequenceNumber = seq_num_ + 2;
+  rtp_header.timestamp = timestamp_ + (33 * 90);
+  rtp_header.markerBit = false;
+  video_header.codec = kVideoCodecGeneric;
+  video_header.frame_type = VideoFrameType::kEmptyFrame;
+  VCMPacket empty_packet(data_, 0, rtp_header, video_header,
+                         /*ntp_time_ms=*/0, clock_->CurrentTime());
+  EXPECT_EQ(kOldPacket,
+            jitter_buffer_->InsertPacket(empty_packet, &retransmitted));
+  empty_packet.seqNum += 1;
+  EXPECT_EQ(kOldPacket,
+            jitter_buffer_->InsertPacket(empty_packet, &retransmitted));
+
+  // But now Frame C should be ready!
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out != NULL);
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, DuplicatePackets) {
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+  EXPECT_EQ(0, jitter_buffer_->num_packets());
+  EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
+
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+
+  EXPECT_TRUE(frame_out == NULL);
+  EXPECT_EQ(1, jitter_buffer_->num_packets());
+  EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
+
+  // Insert a packet into a frame.
+  EXPECT_EQ(kDuplicatePacket,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  EXPECT_EQ(2, jitter_buffer_->num_packets());
+  EXPECT_EQ(1, jitter_buffer_->num_duplicated_packets());
+
+  seq_num_++;
+  packet_->seqNum = seq_num_;
+  packet_->markerBit = true;
+  packet_->video_header.is_first_packet_in_frame = false;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+  ASSERT_TRUE(frame_out != NULL);
+  CheckOutFrame(frame_out, 2 * size_, false);
+
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  EXPECT_EQ(3, jitter_buffer_->num_packets());
+  EXPECT_EQ(1, jitter_buffer_->num_duplicated_packets());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, DuplicatePreviousDeltaFramePacket) {
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+  EXPECT_EQ(0, jitter_buffer_->num_packets());
+  EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
+
+  bool retransmitted = false;
+  // Insert first complete frame.
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  ASSERT_TRUE(frame_out != NULL);
+  CheckOutFrame(frame_out, size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  // Insert 3 delta frames.
+  for (uint16_t i = 1; i <= 3; ++i) {
+    packet_->seqNum = seq_num_ + i;
+    packet_->timestamp = timestamp_ + (i * 33) * 90;
+    packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+    EXPECT_EQ(kCompleteSession,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+    EXPECT_EQ(i + 1, jitter_buffer_->num_packets());
+    EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
+  }
+
+  // Retransmit second delta frame.
+  packet_->seqNum = seq_num_ + 2;
+  packet_->timestamp = timestamp_ + 66 * 90;
+
+  EXPECT_EQ(kDuplicatePacket,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  EXPECT_EQ(5, jitter_buffer_->num_packets());
+  EXPECT_EQ(1, jitter_buffer_->num_duplicated_packets());
+
+  // Should be able to decode 3 delta frames, key frame already decoded.
+  for (size_t i = 0; i < 3; ++i) {
+    frame_out = DecodeCompleteFrame();
+    ASSERT_TRUE(frame_out != NULL);
+    CheckOutFrame(frame_out, size_, false);
+    EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out->FrameType());
+    jitter_buffer_->ReleaseFrame(frame_out);
+  }
+}
+
+TEST_F(TestBasicJitterBuffer, TestSkipForwardVp9) {
+  // Verify that JB skips forward to next base layer frame.
+  //  -------------------------------------------------
+  // | 65485 | 65486 | 65487 | 65488 | 65489 | ...
+  // | pid:5 | pid:6 | pid:7 | pid:8 | pid:9 | ...
+  // | tid:0 | tid:2 | tid:1 | tid:2 | tid:0 | ...
+  // |  ss   |   x   |   x   |   x   |       |
+  //  -------------------------------------------------
+  // |<----------tl0idx:200--------->|<---tl0idx:201---
+
+  jitter_buffer_->SetNackSettings(kMaxNumberOfFrames, kMaxNumberOfFrames, 0);
+  auto& vp9_header =
+      packet_->video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
+
+  bool re = false;
+  packet_->video_header.codec = kVideoCodecVP9;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  vp9_header.flexible_mode = false;
+  vp9_header.spatial_idx = 0;
+  vp9_header.beginning_of_frame = true;
+  vp9_header.end_of_frame = true;
+  vp9_header.temporal_up_switch = false;
+
+  packet_->seqNum = 65485;
+  packet_->timestamp = 1000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  vp9_header.picture_id = 5;
+  vp9_header.tl0_pic_idx = 200;
+  vp9_header.temporal_idx = 0;
+  vp9_header.ss_data_available = true;
+  vp9_header.gof.SetGofInfoVP9(
+      kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  // Insert next temporal layer 0.
+  packet_->seqNum = 65489;
+  packet_->timestamp = 13000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_header.picture_id = 9;
+  vp9_header.tl0_pic_idx = 201;
+  vp9_header.temporal_idx = 0;
+  vp9_header.ss_data_available = false;
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(1000U, frame_out->Timestamp());
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(13000U, frame_out->Timestamp());
+  EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_3TlLayers) {
+  // Verify that frames are updated with SS data when SS packet is reordered.
+  //  --------------------------------
+  // | 65486 | 65487 | 65485 |...
+  // | pid:6 | pid:7 | pid:5 |...
+  // | tid:2 | tid:1 | tid:0 |...
+  // |       |       |  ss   |
+  //  --------------------------------
+  // |<--------tl0idx:200--------->|
+
+  auto& vp9_header =
+      packet_->video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
+
+  bool re = false;
+  packet_->video_header.codec = kVideoCodecVP9;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  vp9_header.flexible_mode = false;
+  vp9_header.spatial_idx = 0;
+  vp9_header.beginning_of_frame = true;
+  vp9_header.end_of_frame = true;
+  vp9_header.tl0_pic_idx = 200;
+
+  packet_->seqNum = 65486;
+  packet_->timestamp = 6000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_header.picture_id = 6;
+  vp9_header.temporal_idx = 2;
+  vp9_header.temporal_up_switch = true;
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  packet_->seqNum = 65487;
+  packet_->timestamp = 9000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_header.picture_id = 7;
+  vp9_header.temporal_idx = 1;
+  vp9_header.temporal_up_switch = true;
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  // Insert first frame with SS data.
+  packet_->seqNum = 65485;
+  packet_->timestamp = 3000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.width = 352;
+  packet_->video_header.height = 288;
+  vp9_header.picture_id = 5;
+  vp9_header.temporal_idx = 0;
+  vp9_header.temporal_up_switch = false;
+  vp9_header.ss_data_available = true;
+  vp9_header.gof.SetGofInfoVP9(
+      kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(3000U, frame_out->Timestamp());
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  EXPECT_EQ(0, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_FALSE(
+      frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(6000U, frame_out->Timestamp());
+  EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out->FrameType());
+  EXPECT_EQ(2, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_TRUE(frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(9000U, frame_out->Timestamp());
+  EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out->FrameType());
+  EXPECT_EQ(1, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_TRUE(frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
+  // Verify that frames are updated with SS data when SS packet is reordered.
+  //  -----------------------------------------
+  // | 65486  | 65487  | 65485  | 65484  |...
+  // | pid:6  | pid:6  | pid:5  | pid:5  |...
+  // | tid:1  | tid:1  | tid:0  | tid:0  |...
+  // | sid:0  | sid:1  | sid:1  | sid:0  |...
+  // | t:6000 | t:6000 | t:3000 | t:3000 |
+  // |        |        |        |  ss    |
+  //  -----------------------------------------
+  // |<-----------tl0idx:200------------>|
+
+  auto& vp9_header =
+      packet_->video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
+
+  bool re = false;
+  packet_->video_header.codec = kVideoCodecVP9;
+  vp9_header.flexible_mode = false;
+  vp9_header.beginning_of_frame = true;
+  vp9_header.end_of_frame = true;
+  vp9_header.tl0_pic_idx = 200;
+
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = 65486;
+  packet_->timestamp = 6000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_header.spatial_idx = 0;
+  vp9_header.picture_id = 6;
+  vp9_header.temporal_idx = 1;
+  vp9_header.temporal_up_switch = true;
+  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = 65487;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  vp9_header.spatial_idx = 1;
+  vp9_header.picture_id = 6;
+  vp9_header.temporal_idx = 1;
+  vp9_header.temporal_up_switch = true;
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = 65485;
+  packet_->timestamp = 3000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  vp9_header.spatial_idx = 1;
+  vp9_header.picture_id = 5;
+  vp9_header.temporal_idx = 0;
+  vp9_header.temporal_up_switch = false;
+  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  // Insert first frame with SS data.
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = 65484;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.width = 352;
+  packet_->video_header.height = 288;
+  vp9_header.spatial_idx = 0;
+  vp9_header.picture_id = 5;
+  vp9_header.temporal_idx = 0;
+  vp9_header.temporal_up_switch = false;
+  vp9_header.ss_data_available = true;
+  vp9_header.gof.SetGofInfoVP9(
+      kTemporalStructureMode2);  // kTemporalStructureMode3: 0-1-0-1..
+  EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(3000U, frame_out->Timestamp());
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  EXPECT_EQ(0, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_FALSE(
+      frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(6000U, frame_out->Timestamp());
+  EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out->FrameType());
+  EXPECT_EQ(1, frame_out->CodecSpecific()->codecSpecific.VP9.temporal_idx);
+  EXPECT_TRUE(frame_out->CodecSpecific()->codecSpecific.VP9.temporal_up_switch);
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, H264InsertStartCode) {
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+  packet_->insertStartCode = true;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+
+  // Frame should not be complete.
+  EXPECT_TRUE(frame_out == NULL);
+
+  seq_num_++;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+  CheckOutFrame(frame_out, size_ * 2 + 4 * 2, true);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, SpsAndPpsHandling) {
+  auto& h264_header =
+      packet_->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+  packet_->timestamp = timestamp_;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->video_header.codec = kVideoCodecH264;
+  h264_header.nalu_type = H264::NaluType::kIdr;
+  h264_header.nalus[0].type = H264::NaluType::kIdr;
+  h264_header.nalus[0].sps_id = -1;
+  h264_header.nalus[0].pps_id = 0;
+  h264_header.nalus_length = 1;
+  bool retransmitted = false;
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  // Not decodable since sps and pps are missing.
+  EXPECT_EQ(nullptr, DecodeCompleteFrame());
+
+  timestamp_ += 3000;
+  packet_->timestamp = timestamp_;
+  ++seq_num_;
+  packet_->seqNum = seq_num_;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->video_header.codec = kVideoCodecH264;
+  h264_header.nalu_type = H264::NaluType::kStapA;
+  h264_header.nalus[0].type = H264::NaluType::kSps;
+  h264_header.nalus[0].sps_id = 0;
+  h264_header.nalus[0].pps_id = -1;
+  h264_header.nalus[1].type = H264::NaluType::kPps;
+  h264_header.nalus[1].sps_id = 0;
+  h264_header.nalus[1].pps_id = 0;
+  h264_header.nalus_length = 2;
+  // Not complete since the marker bit hasn't been received.
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  ++seq_num_;
+  packet_->seqNum = seq_num_;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->video_header.codec = kVideoCodecH264;
+  h264_header.nalu_type = H264::NaluType::kIdr;
+  h264_header.nalus[0].type = H264::NaluType::kIdr;
+  h264_header.nalus[0].sps_id = -1;
+  h264_header.nalus[0].pps_id = 0;
+  h264_header.nalus_length = 1;
+  // Complete and decodable since the pps and sps are received in the first
+  // packet of this frame.
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  ASSERT_NE(nullptr, frame_out);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  timestamp_ += 3000;
+  packet_->timestamp = timestamp_;
+  ++seq_num_;
+  packet_->seqNum = seq_num_;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->video_header.codec = kVideoCodecH264;
+  h264_header.nalu_type = H264::NaluType::kSlice;
+  h264_header.nalus[0].type = H264::NaluType::kSlice;
+  h264_header.nalus[0].sps_id = -1;
+  h264_header.nalus[0].pps_id = 0;
+  h264_header.nalus_length = 1;
+  // Complete and decodable since sps, pps and key frame has been received.
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  frame_out = DecodeCompleteFrame();
+  ASSERT_NE(nullptr, frame_out);
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, DeltaFrame100PacketsWithSeqNumWrap) {
+  seq_num_ = 0xfff0;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+
+  EXPECT_TRUE(frame_out == NULL);
+
+  int loop = 0;
+  do {
+    seq_num_++;
+    packet_->video_header.is_first_packet_in_frame = false;
+    packet_->markerBit = false;
+    packet_->seqNum = seq_num_;
+
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+    frame_out = DecodeCompleteFrame();
+
+    EXPECT_TRUE(frame_out == NULL);
+
+    loop++;
+  } while (loop < 98);
+
+  seq_num_++;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+
+  CheckOutFrame(frame_out, 100 * size_, false);
+
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, PacketReorderingReverseWithNegSeqNumWrap) {
+  // Insert "first" packet last seqnum.
+  seq_num_ = 10;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+
+  // Should not be complete.
+  EXPECT_TRUE(frame_out == NULL);
+
+  // Insert 98 frames.
+  int loop = 0;
+  do {
+    seq_num_--;
+    packet_->video_header.is_first_packet_in_frame = false;
+    packet_->markerBit = false;
+    packet_->seqNum = seq_num_;
+
+    EXPECT_EQ(kIncomplete,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+    frame_out = DecodeCompleteFrame();
+
+    EXPECT_TRUE(frame_out == NULL);
+
+    loop++;
+  } while (loop < 98);
+
+  // Insert last packet.
+  seq_num_--;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+  CheckOutFrame(frame_out, 100 * size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, TestInsertOldFrame) {
+  //   -------      -------
+  //  |   2   |    |   1   |
+  //   -------      -------
+  //  t = 3000     t = 2000
+  seq_num_ = 2;
+  timestamp_ = 3000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->timestamp = timestamp_;
+  packet_->seqNum = seq_num_;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(3000u, frame_out->Timestamp());
+  CheckOutFrame(frame_out, size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  seq_num_--;
+  timestamp_ = 2000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+}
+
+TEST_F(TestBasicJitterBuffer, TestInsertOldFrameWithSeqNumWrap) {
+  //   -------      -------
+  //  |   2   |    |   1   |
+  //   -------      -------
+  //  t = 3000     t = 0xffffff00
+
+  seq_num_ = 2;
+  timestamp_ = 3000;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(timestamp_, frame_out->Timestamp());
+
+  CheckOutFrame(frame_out, size_, false);
+
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  seq_num_--;
+  timestamp_ = 0xffffff00;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  // This timestamp is old.
+  EXPECT_EQ(kOldPacket, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+}
+
+TEST_F(TestBasicJitterBuffer, TimestampWrap) {
+  //  ---------------     ---------------
+  // |   1   |   2   |   |   3   |   4   |
+  //  ---------------     ---------------
+  //  t = 0xffffff00        t = 33*90
+
+  timestamp_ = 0xffffff00;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out == NULL);
+
+  seq_num_++;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+  CheckOutFrame(frame_out, 2 * size_, false);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  seq_num_++;
+  timestamp_ += 33 * 90;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = false;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out == NULL);
+
+  seq_num_++;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  frame_out = DecodeCompleteFrame();
+  CheckOutFrame(frame_out, 2 * size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, 2FrameWithTimestampWrap) {
+  //   -------          -------
+  //  |   1   |        |   2   |
+  //   -------          -------
+  // t = 0xffffff00    t = 2700
+
+  timestamp_ = 0xffffff00;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->timestamp = timestamp_;
+
+  bool retransmitted = false;
+  // Insert first frame (session will be complete).
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  // Insert next frame.
+  seq_num_++;
+  timestamp_ = 2700;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(0xffffff00, frame_out->Timestamp());
+  CheckOutFrame(frame_out, size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  VCMEncodedFrame* frame_out2 = DecodeCompleteFrame();
+  EXPECT_EQ(2700u, frame_out2->Timestamp());
+  CheckOutFrame(frame_out2, size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out2->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out2);
+}
+
+TEST_F(TestBasicJitterBuffer, Insert2FramesReOrderedWithTimestampWrap) {
+  //   -------          -------
+  //  |   2   |        |   1   |
+  //   -------          -------
+  //  t = 2700        t = 0xffffff00
+
+  seq_num_ = 2;
+  timestamp_ = 2700;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  bool retransmitted = false;
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  // Insert second frame
+  seq_num_--;
+  timestamp_ = 0xffffff00;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(0xffffff00, frame_out->Timestamp());
+  CheckOutFrame(frame_out, size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  VCMEncodedFrame* frame_out2 = DecodeCompleteFrame();
+  EXPECT_EQ(2700u, frame_out2->Timestamp());
+  CheckOutFrame(frame_out2, size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameDelta, frame_out2->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out2);
+}
+
+TEST_F(TestBasicJitterBuffer, DeltaFrameWithMoreThanMaxNumberOfPackets) {
+  int loop = 0;
+  bool firstPacket = true;
+  bool retransmitted = false;
+  // Insert kMaxPacketsInJitterBuffer into frame.
+  do {
+    seq_num_++;
+    packet_->video_header.is_first_packet_in_frame = false;
+    packet_->markerBit = false;
+    packet_->seqNum = seq_num_;
+
+    if (firstPacket) {
+      EXPECT_EQ(kIncomplete,
+                jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+      firstPacket = false;
+    } else {
+      EXPECT_EQ(kIncomplete,
+                jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+    }
+
+    loop++;
+  } while (loop < kMaxPacketsInSession);
+
+  // Max number of packets inserted.
+  // Insert one more packet.
+  seq_num_++;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+
+  // Insert the packet -> frame recycled.
+  EXPECT_EQ(kSizeError, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  EXPECT_TRUE(NULL == DecodeCompleteFrame());
+}
+
+TEST_F(TestBasicJitterBuffer, ExceedNumOfFrameWithSeqNumWrap) {
+  // TEST fill JB with more than max number of frame (50 delta frames +
+  // 51 key frames) with wrap in seq_num_
+  //
+  //  --------------------------------------------------------------
+  // | 65485 | 65486 | 65487 | .... | 65535 | 0 | 1 | 2 | .....| 50 |
+  //  --------------------------------------------------------------
+  // |<-----------delta frames------------->|<------key frames----->|
+
+  // Make sure the jitter doesn't request a keyframe after too much non-
+  // decodable frames.
+  jitter_buffer_->SetNackSettings(kMaxNumberOfFrames, kMaxNumberOfFrames, 0);
+
+  int loop = 0;
+  seq_num_ = 65485;
+  uint32_t first_key_frame_timestamp = 0;
+  bool retransmitted = false;
+  // Insert MAX_NUMBER_OF_FRAMES frames.
+  do {
+    timestamp_ += 33 * 90;
+    seq_num_++;
+    packet_->video_header.is_first_packet_in_frame = true;
+    packet_->markerBit = true;
+    packet_->seqNum = seq_num_;
+    packet_->timestamp = timestamp_;
+
+    if (loop == 50) {
+      first_key_frame_timestamp = packet_->timestamp;
+      packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+    }
+
+    // Insert frame.
+    EXPECT_EQ(kCompleteSession,
+              jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+    loop++;
+  } while (loop < kMaxNumberOfFrames);
+
+  // Max number of frames inserted.
+
+  // Insert one more frame.
+  timestamp_ += 33 * 90;
+  seq_num_++;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  packet_->seqNum = seq_num_;
+  packet_->timestamp = timestamp_;
+
+  // Now, no free frame - frames will be recycled until first key frame.
+  EXPECT_EQ(kFlushIndicator,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_EQ(first_key_frame_timestamp, frame_out->Timestamp());
+  CheckOutFrame(frame_out, size_, false);
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, frame_out->FrameType());
+  jitter_buffer_->ReleaseFrame(frame_out);
+}
+
+TEST_F(TestBasicJitterBuffer, EmptyLastFrame) {
+  seq_num_ = 3;
+  // Insert one empty packet per frame, should never return the last timestamp
+  // inserted. Only return empty frames in the presence of subsequent frames.
+  int maxSize = 1000;
+  bool retransmitted = false;
+  for (int i = 0; i < maxSize + 10; i++) {
+    timestamp_ += 33 * 90;
+    seq_num_++;
+    packet_->video_header.is_first_packet_in_frame = false;
+    packet_->markerBit = false;
+    packet_->seqNum = seq_num_;
+    packet_->timestamp = timestamp_;
+    packet_->video_header.frame_type = VideoFrameType::kEmptyFrame;
+
+    EXPECT_EQ(kNoError, jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  }
+}
+
+TEST_F(TestBasicJitterBuffer, NextFrameWhenIncomplete) {
+  // Test that a we cannot get incomplete frames from the JB if we haven't
+  // received the marker bit, unless we have received a packet from a later
+  // timestamp.
+  // Start with a complete key frame - insert and decode.
+  jitter_buffer_->SetNackSettings(kMaxNumberOfFrames, kMaxNumberOfFrames, 0);
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet_->video_header.is_first_packet_in_frame = true;
+  packet_->markerBit = true;
+  bool retransmitted = false;
+
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+  VCMEncodedFrame* frame_out = DecodeCompleteFrame();
+  EXPECT_TRUE(frame_out != NULL);
+  jitter_buffer_->ReleaseFrame(frame_out);
+
+  packet_->seqNum += 2;
+  packet_->timestamp += 33 * 90;
+  packet_->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  packet_->video_header.is_first_packet_in_frame = false;
+  packet_->markerBit = false;
+
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+
+  packet_->seqNum += 2;
+  packet_->timestamp += 33 * 90;
+  packet_->video_header.is_first_packet_in_frame = true;
+
+  EXPECT_EQ(kIncomplete,
+            jitter_buffer_->InsertPacket(*packet_, &retransmitted));
+}
+
+TEST_F(TestRunningJitterBuffer, Full) {
+  // Make sure the jitter doesn't request a keyframe after too much non-
+  // decodable frames.
+  jitter_buffer_->SetNackSettings(kMaxNumberOfFrames, kMaxNumberOfFrames, 0);
+  // Insert a key frame and decode it.
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey), kNoError);
+  EXPECT_TRUE(DecodeCompleteFrame());
+  DropFrame(1);
+  // Fill the jitter buffer.
+  EXPECT_GE(InsertFrames(kMaxNumberOfFrames, VideoFrameType::kVideoFrameDelta),
+            kNoError);
+  // Make sure we can't decode these frames.
+  EXPECT_FALSE(DecodeCompleteFrame());
+  // This frame will make the jitter buffer recycle frames until a key frame.
+  // Since none is found it will have to wait until the next key frame before
+  // decoding.
+  EXPECT_EQ(kFlushIndicator, InsertFrame(VideoFrameType::kVideoFrameDelta));
+  EXPECT_FALSE(DecodeCompleteFrame());
+}
+
+TEST_F(TestRunningJitterBuffer, EmptyPackets) {
+  // Make sure a frame can get complete even though empty packets are missing.
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameKey, 3, 3,
+                                   clock_->TimeInMilliseconds());
+  bool request_key_frame = false;
+  // Insert empty packet.
+  EXPECT_EQ(kNoError, InsertPacketAndPop(4));
+  EXPECT_FALSE(request_key_frame);
+  // Insert 3 media packets.
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  EXPECT_FALSE(request_key_frame);
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  EXPECT_FALSE(request_key_frame);
+  EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
+  EXPECT_FALSE(request_key_frame);
+  // Insert empty packet.
+  EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
+  EXPECT_FALSE(request_key_frame);
+}
+
+TEST_F(TestRunningJitterBuffer, SkipToKeyFrame) {
+  // Insert delta frames.
+  EXPECT_GE(InsertFrames(5, VideoFrameType::kVideoFrameDelta), kNoError);
+  // Can't decode without a key frame.
+  EXPECT_FALSE(DecodeCompleteFrame());
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  // Skip to the next key frame.
+  EXPECT_TRUE(DecodeCompleteFrame());
+}
+
+TEST_F(TestRunningJitterBuffer, DontSkipToKeyFrameIfDecodable) {
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  EXPECT_TRUE(DecodeCompleteFrame());
+  const int kNumDeltaFrames = 5;
+  EXPECT_GE(InsertFrames(kNumDeltaFrames, VideoFrameType::kVideoFrameDelta),
+            kNoError);
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  for (int i = 0; i < kNumDeltaFrames + 1; ++i) {
+    EXPECT_TRUE(DecodeCompleteFrame());
+  }
+}
+
+TEST_F(TestRunningJitterBuffer, KeyDeltaKeyDelta) {
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  EXPECT_TRUE(DecodeCompleteFrame());
+  const int kNumDeltaFrames = 5;
+  EXPECT_GE(InsertFrames(kNumDeltaFrames, VideoFrameType::kVideoFrameDelta),
+            kNoError);
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  EXPECT_GE(InsertFrames(kNumDeltaFrames, VideoFrameType::kVideoFrameDelta),
+            kNoError);
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  for (int i = 0; i < 2 * (kNumDeltaFrames + 1); ++i) {
+    EXPECT_TRUE(DecodeCompleteFrame());
+  }
+}
+
+TEST_F(TestRunningJitterBuffer, TwoPacketsNonContinuous) {
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  EXPECT_TRUE(DecodeCompleteFrame());
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta, 1, 0,
+                                   clock_->TimeInMilliseconds());
+  clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta, 2, 0,
+                                   clock_->TimeInMilliseconds());
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(1));
+  EXPECT_EQ(kCompleteSession, InsertPacketAndPop(1));
+  EXPECT_FALSE(DecodeCompleteFrame());
+  EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
+  EXPECT_TRUE(DecodeCompleteFrame());
+  EXPECT_TRUE(DecodeCompleteFrame());
+}
+
+TEST_F(TestJitterBufferNack, EmptyPackets) {
+  // Make sure empty packets doesn't clog the jitter buffer.
+  EXPECT_GE(InsertFrames(kMaxNumberOfFrames, VideoFrameType::kEmptyFrame),
+            kNoError);
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  EXPECT_TRUE(DecodeCompleteFrame());
+}
+
+TEST_F(TestJitterBufferNack, NackTooOldPackets) {
+  // Insert a key frame and decode it.
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey), kNoError);
+  EXPECT_TRUE(DecodeCompleteFrame());
+
+  // Drop one frame and insert `kNackHistoryLength` to trigger NACKing a too
+  // old packet.
+  DropFrame(1);
+  // Insert a frame which should trigger a recycle until the next key frame.
+  EXPECT_EQ(kFlushIndicator, InsertFrames(oldest_packet_to_nack_ + 1,
+                                          VideoFrameType::kVideoFrameDelta));
+  EXPECT_FALSE(DecodeCompleteFrame());
+
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list =
+      jitter_buffer_->GetNackList(&request_key_frame);
+  // No key frame will be requested since the jitter buffer is empty.
+  EXPECT_FALSE(request_key_frame);
+  EXPECT_EQ(0u, nack_list.size());
+
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta), kNoError);
+  // Waiting for a key frame.
+  EXPECT_FALSE(DecodeCompleteFrame());
+
+  // The next complete continuous frame isn't a key frame, but we're waiting
+  // for one.
+  EXPECT_FALSE(DecodeCompleteFrame());
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey), kNoError);
+  // Skipping ahead to the key frame.
+  EXPECT_TRUE(DecodeCompleteFrame());
+}
+
+TEST_F(TestJitterBufferNack, NackLargeJitterBuffer) {
+  // Insert a key frame and decode it.
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey), kNoError);
+  EXPECT_TRUE(DecodeCompleteFrame());
+
+  // Insert a frame which should trigger a recycle until the next key frame.
+  EXPECT_GE(
+      InsertFrames(oldest_packet_to_nack_, VideoFrameType::kVideoFrameDelta),
+      kNoError);
+
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list =
+      jitter_buffer_->GetNackList(&request_key_frame);
+  // Verify that the jitter buffer does not request a key frame.
+  EXPECT_FALSE(request_key_frame);
+  // Verify that no packets are NACKed.
+  EXPECT_EQ(0u, nack_list.size());
+  // Verify that we can decode the next frame.
+  EXPECT_TRUE(DecodeCompleteFrame());
+}
+
+TEST_F(TestJitterBufferNack, NackListFull) {
+  // Insert a key frame and decode it.
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey), kNoError);
+  EXPECT_TRUE(DecodeCompleteFrame());
+
+  // Generate and drop `kNackHistoryLength` packets to fill the NACK list.
+  DropFrame(max_nack_list_size_ + 1);
+  // Insert a frame which should trigger a recycle until the next key frame.
+  EXPECT_EQ(kFlushIndicator, InsertFrame(VideoFrameType::kVideoFrameDelta));
+  EXPECT_FALSE(DecodeCompleteFrame());
+
+  bool request_key_frame = false;
+  jitter_buffer_->GetNackList(&request_key_frame);
+  // The jitter buffer is empty, so we won't request key frames until we get a
+  // packet.
+  EXPECT_FALSE(request_key_frame);
+
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta), kNoError);
+  // Now we have a packet in the jitter buffer, a key frame will be requested
+  // since it's not a key frame.
+  jitter_buffer_->GetNackList(&request_key_frame);
+  // The jitter buffer is empty, so we won't request key frames until we get a
+  // packet.
+  EXPECT_TRUE(request_key_frame);
+  // The next complete continuous frame isn't a key frame, but we're waiting
+  // for one.
+  EXPECT_FALSE(DecodeCompleteFrame());
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey), kNoError);
+  // Skipping ahead to the key frame.
+  EXPECT_TRUE(DecodeCompleteFrame());
+}
+
+TEST_F(TestJitterBufferNack, NoNackListReturnedBeforeFirstDecode) {
+  DropFrame(10);
+  // Insert a frame and try to generate a NACK list. Shouldn't get one.
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta), kNoError);
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list =
+      jitter_buffer_->GetNackList(&request_key_frame);
+  // No list generated, and a key frame request is signaled.
+  EXPECT_EQ(0u, nack_list.size());
+  EXPECT_TRUE(request_key_frame);
+}
+
+TEST_F(TestJitterBufferNack, NackListBuiltBeforeFirstDecode) {
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta, 2, 0,
+                                   clock_->TimeInMilliseconds());
+  stream_generator_->NextPacket(NULL);  // Drop packet.
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  EXPECT_TRUE(DecodeCompleteFrame());
+  bool extended = false;
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(1u, nack_list.size());
+}
+
+TEST_F(TestJitterBufferNack, VerifyRetransmittedFlag) {
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameKey, 3, 0,
+                                   clock_->TimeInMilliseconds());
+  VCMPacket packet;
+  stream_generator_->PopPacket(&packet, 0);
+  bool retransmitted = false;
+  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(packet, &retransmitted));
+  EXPECT_FALSE(retransmitted);
+  // Drop second packet.
+  stream_generator_->PopPacket(&packet, 1);
+  EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(packet, &retransmitted));
+  EXPECT_FALSE(retransmitted);
+  EXPECT_FALSE(DecodeCompleteFrame());
+  bool extended = false;
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  uint16_t seq_num;
+  EXPECT_EQ(1u, nack_list.size());
+  seq_num = nack_list[0];
+  stream_generator_->PopPacket(&packet, 0);
+  EXPECT_EQ(packet.seqNum, seq_num);
+  EXPECT_EQ(kCompleteSession,
+            jitter_buffer_->InsertPacket(packet, &retransmitted));
+  EXPECT_TRUE(retransmitted);
+  EXPECT_TRUE(DecodeCompleteFrame());
+}
+
+TEST_F(TestJitterBufferNack, UseNackToRecoverFirstKeyFrame) {
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameKey, 3, 0,
+                                   clock_->TimeInMilliseconds());
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  // Drop second packet.
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(1));
+  EXPECT_FALSE(DecodeCompleteFrame());
+  bool extended = false;
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  uint16_t seq_num;
+  ASSERT_EQ(1u, nack_list.size());
+  seq_num = nack_list[0];
+  VCMPacket packet;
+  stream_generator_->GetPacket(&packet, 0);
+  EXPECT_EQ(packet.seqNum, seq_num);
+}
+
+TEST_F(TestJitterBufferNack, UseNackToRecoverFirstKeyFrameSecondInQueue) {
+  VCMPacket packet;
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
+  // First frame is delta.
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta, 3, 0,
+                                   clock_->TimeInMilliseconds());
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  // Drop second packet in frame.
+  ASSERT_TRUE(stream_generator_->PopPacket(&packet, 0));
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  // Second frame is key.
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameKey, 3, 0,
+                                   clock_->TimeInMilliseconds() + 10);
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  // Drop second packet in frame.
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(1));
+  EXPECT_FALSE(DecodeCompleteFrame());
+  bool extended = false;
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  uint16_t seq_num;
+  ASSERT_EQ(1u, nack_list.size());
+  seq_num = nack_list[0];
+  stream_generator_->GetPacket(&packet, 0);
+  EXPECT_EQ(packet.seqNum, seq_num);
+}
+
+TEST_F(TestJitterBufferNack, NormalOperation) {
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey), kNoError);
+  EXPECT_TRUE(DecodeCompleteFrame());
+
+  //  ----------------------------------------------------------------
+  // | 1 | 2 | .. | 8 | 9 | x | 11 | 12 | .. | 19 | x | 21 | .. | 100 |
+  //  ----------------------------------------------------------------
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameKey, 100, 0,
+                                   clock_->TimeInMilliseconds());
+  clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  // Verify that the frame is incomplete.
+  EXPECT_FALSE(DecodeCompleteFrame());
+  while (stream_generator_->PacketsRemaining() > 1) {
+    if (stream_generator_->NextSequenceNumber() % 10 != 0) {
+      EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+    } else {
+      stream_generator_->NextPacket(NULL);  // Drop packet
+    }
+  }
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  EXPECT_EQ(0, stream_generator_->PacketsRemaining());
+  EXPECT_FALSE(DecodeCompleteFrame());
+  bool request_key_frame = false;
+
+  // Verify the NACK list.
+  std::vector<uint16_t> nack_list =
+      jitter_buffer_->GetNackList(&request_key_frame);
+  const size_t kExpectedNackSize = 9;
+  ASSERT_EQ(kExpectedNackSize, nack_list.size());
+  for (size_t i = 0; i < nack_list.size(); ++i)
+    EXPECT_EQ((1 + i) * 10, nack_list[i]);
+}
+
+TEST_F(TestJitterBufferNack, NormalOperationWrap) {
+  bool request_key_frame = false;
+  //  -------   ------------------------------------------------------------
+  // | 65532 | | 65533 | 65534 | 65535 | x | 1 | .. | 9 | x | 11 |.....| 96 |
+  //  -------   ------------------------------------------------------------
+  stream_generator_->Init(65532, clock_->TimeInMilliseconds());
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  EXPECT_FALSE(request_key_frame);
+  EXPECT_TRUE(DecodeCompleteFrame());
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta, 100, 0,
+                                   clock_->TimeInMilliseconds());
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  while (stream_generator_->PacketsRemaining() > 1) {
+    if (stream_generator_->NextSequenceNumber() % 10 != 0) {
+      EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+      EXPECT_FALSE(request_key_frame);
+    } else {
+      stream_generator_->NextPacket(NULL);  // Drop packet
+    }
+  }
+  EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
+  EXPECT_FALSE(request_key_frame);
+  EXPECT_EQ(0, stream_generator_->PacketsRemaining());
+  EXPECT_FALSE(DecodeCompleteFrame());
+  EXPECT_FALSE(DecodeCompleteFrame());
+  bool extended = false;
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  // Verify the NACK list.
+  const size_t kExpectedNackSize = 10;
+  ASSERT_EQ(kExpectedNackSize, nack_list.size());
+  for (size_t i = 0; i < nack_list.size(); ++i)
+    EXPECT_EQ(i * 10, nack_list[i]);
+}
+
+TEST_F(TestJitterBufferNack, NormalOperationWrap2) {
+  bool request_key_frame = false;
+  //  -----------------------------------
+  // | 65532 | 65533 | 65534 | x | 0 | 1 |
+  //  -----------------------------------
+  stream_generator_->Init(65532, clock_->TimeInMilliseconds());
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  EXPECT_FALSE(request_key_frame);
+  EXPECT_TRUE(DecodeCompleteFrame());
+  stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta, 1, 0,
+                                   clock_->TimeInMilliseconds());
+  clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+  for (int i = 0; i < 5; ++i) {
+    if (stream_generator_->NextSequenceNumber() != 65535) {
+      EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
+      EXPECT_FALSE(request_key_frame);
+    } else {
+      stream_generator_->NextPacket(NULL);  // Drop packet
+    }
+    stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameDelta, 1, 0,
+                                     clock_->TimeInMilliseconds());
+    clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+  }
+  EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
+  EXPECT_FALSE(request_key_frame);
+  bool extended = false;
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  // Verify the NACK list.
+  ASSERT_EQ(1u, nack_list.size());
+  EXPECT_EQ(65535, nack_list[0]);
+}
+
+TEST_F(TestJitterBufferNack, ResetByFutureKeyFrameDoesntError) {
+  stream_generator_->Init(0, clock_->TimeInMilliseconds());
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  EXPECT_TRUE(DecodeCompleteFrame());
+  bool extended = false;
+  std::vector<uint16_t> nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(0u, nack_list.size());
+
+  // Far-into-the-future video frame, could be caused by resetting the encoder
+  // or otherwise restarting. This should not fail when error when the packet is
+  // a keyframe, even if all of the nack list needs to be flushed.
+  stream_generator_->Init(10000, clock_->TimeInMilliseconds());
+  clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+  InsertFrame(VideoFrameType::kVideoFrameKey);
+  EXPECT_TRUE(DecodeCompleteFrame());
+  nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(0u, nack_list.size());
+
+  // Stream should be decodable from this point.
+  clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+  InsertFrame(VideoFrameType::kVideoFrameDelta);
+  EXPECT_TRUE(DecodeCompleteFrame());
+  nack_list = jitter_buffer_->GetNackList(&extended);
+  EXPECT_EQ(0u, nack_list.size());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/loss_notification_controller.cc b/third_party/libwebrtc/modules/video_coding/loss_notification_controller.cc
new file mode 100644
index 0000000000..3377ab5a76
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/loss_notification_controller.cc
@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/loss_notification_controller.h"
+
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+
+namespace webrtc {
+namespace {
+// Keep a container's size no higher than `max_allowed_size`, by paring its size
+// down to `target_size` whenever it has more than `max_allowed_size` elements.
+template <typename Container>
+void PareDown(Container* container,
+              size_t max_allowed_size,
+              size_t target_size) {
+  if (container->size() > max_allowed_size) {
+    const size_t entries_to_delete = container->size() - target_size;
+    auto erase_to = container->begin();
+    std::advance(erase_to, entries_to_delete);
+    container->erase(container->begin(), erase_to);
+    RTC_DCHECK_EQ(container->size(), target_size);
+  }
+}
+}  // namespace
+
+LossNotificationController::LossNotificationController(
+    KeyFrameRequestSender* key_frame_request_sender,
+    LossNotificationSender* loss_notification_sender)
+    : key_frame_request_sender_(key_frame_request_sender),
+      loss_notification_sender_(loss_notification_sender),
+      current_frame_potentially_decodable_(true) {
+  RTC_DCHECK(key_frame_request_sender_);
+  RTC_DCHECK(loss_notification_sender_);
+}
+
+LossNotificationController::~LossNotificationController() = default;
+
+void LossNotificationController::OnReceivedPacket(
+    uint16_t rtp_seq_num,
+    const LossNotificationController::FrameDetails* frame) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // Ignore repeated or reordered packets.
+  // TODO(bugs.webrtc.org/10336): Handle packet reordering.
+  if (last_received_seq_num_ &&
+      !AheadOf(rtp_seq_num, *last_received_seq_num_)) {
+    return;
+  }
+
+  DiscardOldInformation();  // Prevent memory overconsumption.
+
+  const bool seq_num_gap =
+      last_received_seq_num_ &&
+      rtp_seq_num != static_cast<uint16_t>(*last_received_seq_num_ + 1u);
+
+  last_received_seq_num_ = rtp_seq_num;
+
+  // `frame` is not nullptr iff the packet is the first packet in the frame.
+  if (frame != nullptr) {
+    // Ignore repeated or reordered frames.
+    // TODO(bugs.webrtc.org/10336): Handle frame reordering.
+    if (last_received_frame_id_.has_value() &&
+        frame->frame_id <= last_received_frame_id_.value()) {
+      RTC_LOG(LS_WARNING) << "Repeated or reordered frame ID ("
+                          << frame->frame_id << ").";
+      return;
+    }
+
+    last_received_frame_id_ = frame->frame_id;
+
+    if (frame->is_keyframe) {
+      // Subsequent frames may not rely on frames before the key frame.
+      // Note that upon receiving a key frame, we do not issue a loss
+      // notification on RTP sequence number gap, unless that gap spanned
+      // the key frame itself. This is because any loss which occurred before
+      // the key frame is no longer relevant.
+      decodable_frame_ids_.clear();
+      current_frame_potentially_decodable_ = true;
+    } else {
+      const bool all_dependencies_decodable =
+          AllDependenciesDecodable(frame->frame_dependencies);
+      current_frame_potentially_decodable_ = all_dependencies_decodable;
+      if (seq_num_gap || !current_frame_potentially_decodable_) {
+        HandleLoss(rtp_seq_num, current_frame_potentially_decodable_);
+      }
+    }
+  } else if (seq_num_gap || !current_frame_potentially_decodable_) {
+    current_frame_potentially_decodable_ = false;
+    // We allow sending multiple loss notifications for a single frame
+    // even if only one of its packets is lost. We do this because the bigger
+    // the frame, the more likely it is to be non-discardable, and therefore
+    // the more robust we wish to be to loss of the feedback messages.
+    HandleLoss(rtp_seq_num, false);
+  }
+}
+
+void LossNotificationController::OnAssembledFrame(
+    uint16_t first_seq_num,
+    int64_t frame_id,
+    bool discardable,
+    rtc::ArrayView<const int64_t> frame_dependencies) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  DiscardOldInformation();  // Prevent memory overconsumption.
+
+  if (discardable) {
+    return;
+  }
+
+  if (!AllDependenciesDecodable(frame_dependencies)) {
+    return;
+  }
+
+  last_decodable_non_discardable_.emplace(first_seq_num);
+  const auto it = decodable_frame_ids_.insert(frame_id);
+  RTC_DCHECK(it.second);
+}
+
+void LossNotificationController::DiscardOldInformation() {
+  constexpr size_t kExpectedKeyFrameIntervalFrames = 3000;
+  constexpr size_t kMaxSize = 2 * kExpectedKeyFrameIntervalFrames;
+  constexpr size_t kTargetSize = kExpectedKeyFrameIntervalFrames;
+  PareDown(&decodable_frame_ids_, kMaxSize, kTargetSize);
+}
+
+bool LossNotificationController::AllDependenciesDecodable(
+    rtc::ArrayView<const int64_t> frame_dependencies) const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // Due to packet reordering, frame buffering and asynchronous decoders, it is
+  // infeasible to make reliable conclusions on the decodability of a frame
+  // immediately when it arrives. We use the following assumptions:
+  // * Intra frames are decodable.
+  // * Inter frames are decodable if all of their references were decodable.
+  // One possibility that is ignored, is that the packet may be corrupt.
+  for (int64_t ref_frame_id : frame_dependencies) {
+    const auto ref_frame_it = decodable_frame_ids_.find(ref_frame_id);
+    if (ref_frame_it == decodable_frame_ids_.end()) {
+      // Reference frame not decodable.
+      return false;
+    }
+  }
+
+  return true;
+}
+
+void LossNotificationController::HandleLoss(uint16_t last_received_seq_num,
+                                            bool decodability_flag) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  if (last_decodable_non_discardable_) {
+    RTC_DCHECK(AheadOf(last_received_seq_num,
+                       last_decodable_non_discardable_->first_seq_num));
+    loss_notification_sender_->SendLossNotification(
+        last_decodable_non_discardable_->first_seq_num, last_received_seq_num,
+        decodability_flag, /*buffering_allowed=*/true);
+  } else {
+    key_frame_request_sender_->RequestKeyFrame();
+  }
+}
+}  //  namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/loss_notification_controller.h b/third_party/libwebrtc/modules/video_coding/loss_notification_controller.h
new file mode 100644
index 0000000000..ecba41267b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/loss_notification_controller.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_LOSS_NOTIFICATION_CONTROLLER_H_
+#define MODULES_VIDEO_CODING_LOSS_NOTIFICATION_CONTROLLER_H_
+
+#include <stdint.h>
+
+#include <set>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/sequence_checker.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace webrtc {
+
+class LossNotificationController {
+ public:
+  struct FrameDetails {
+    bool is_keyframe;
+    int64_t frame_id;
+    rtc::ArrayView<const int64_t> frame_dependencies;
+  };
+
+  LossNotificationController(KeyFrameRequestSender* key_frame_request_sender,
+                             LossNotificationSender* loss_notification_sender);
+  ~LossNotificationController();
+
+  // An RTP packet was received from the network.
+  // `frame` is non-null iff the packet is the first packet in the frame.
+  void OnReceivedPacket(uint16_t rtp_seq_num, const FrameDetails* frame);
+
+  // A frame was assembled from packets previously received.
+  // (Should be called even if the frame was composed of a single packet.)
+  void OnAssembledFrame(uint16_t first_seq_num,
+                        int64_t frame_id,
+                        bool discardable,
+                        rtc::ArrayView<const int64_t> frame_dependencies);
+
+ private:
+  void DiscardOldInformation();
+
+  bool AllDependenciesDecodable(
+      rtc::ArrayView<const int64_t> frame_dependencies) const;
+
+  // When the loss of a packet or the non-decodability of a frame is detected,
+  // produces a key frame request or a loss notification.
+  // 1. `last_received_seq_num` is the last received sequence number.
+  // 2. `decodability_flag` refers to the frame associated with the last packet.
+  //    It is set to `true` if and only if all of that frame's dependencies are
+  //    known to be decodable, and the frame itself is not yet known to be
+  //    unassemblable (i.e. no earlier parts of it were lost).
+  //    Clarifications:
+  //    a. In a multi-packet frame, the first packet reveals the frame's
+  //       dependencies, but it is not yet known whether all parts of the
+  //       current frame will be received.
+  //    b. In a multi-packet frame, if the first packet is missed, the
+  //       dependencies are unknown, but it is known that the frame itself
+  //       is unassemblable.
+  void HandleLoss(uint16_t last_received_seq_num, bool decodability_flag);
+
+  KeyFrameRequestSender* const key_frame_request_sender_
+      RTC_GUARDED_BY(sequence_checker_);
+
+  LossNotificationSender* const loss_notification_sender_
+      RTC_GUARDED_BY(sequence_checker_);
+
+  // Tracked to avoid processing repeated frames (buggy/malicious remote).
+  absl::optional<int64_t> last_received_frame_id_
+      RTC_GUARDED_BY(sequence_checker_);
+
+  // Tracked to avoid processing repeated packets.
+  absl::optional<uint16_t> last_received_seq_num_
+      RTC_GUARDED_BY(sequence_checker_);
+
+  // Tracked in order to correctly report the potential-decodability of
+  // multi-packet frames.
+  bool current_frame_potentially_decodable_ RTC_GUARDED_BY(sequence_checker_);
+
+  // Loss notifications contain the sequence number of the first packet of
+  // the last decodable-and-non-discardable frame. Since this is a bit of
+  // a mouthful, last_decodable_non_discardable_.first_seq_num is used,
+  // which hopefully is a bit easier for human beings to parse
+  // than `first_seq_num_of_last_decodable_non_discardable_`.
+  struct FrameInfo {
+    explicit FrameInfo(uint16_t first_seq_num) : first_seq_num(first_seq_num) {}
+    uint16_t first_seq_num;
+  };
+  absl::optional<FrameInfo> last_decodable_non_discardable_
+      RTC_GUARDED_BY(sequence_checker_);
+
+  // Track which frames are decodable. Later frames are also decodable if
+  // all of their dependencies can be found in this container.
+  // (Naturally, later frames must also be assemblable to be decodable.)
+  std::set<int64_t> decodable_frame_ids_ RTC_GUARDED_BY(sequence_checker_);
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_LOSS_NOTIFICATION_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/loss_notification_controller_unittest.cc b/third_party/libwebrtc/modules/video_coding/loss_notification_controller_unittest.cc
new file mode 100644
index 0000000000..9c4e715b4f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/loss_notification_controller_unittest.cc
@@ -0,0 +1,607 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/loss_notification_controller.h"
+
+#include <stdint.h>
+
+#include <limits>
+#include <string>
+#include <tuple>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+// The information about an RTP packet that is relevant in these tests.
+struct Packet {
+  uint16_t seq_num;
+  bool first_in_frame;
+  bool is_keyframe;
+  int64_t frame_id;
+  std::vector<int64_t> frame_dependencies;
+};
+
+Packet CreatePacket(
+    bool first_in_frame,
+    bool last_in_frame,
+    uint16_t seq_num,
+    uint16_t frame_id,
+    bool is_key_frame,
+    std::vector<int64_t> ref_frame_ids = std::vector<int64_t>()) {
+  Packet packet;
+  packet.seq_num = seq_num;
+  packet.first_in_frame = first_in_frame;
+  if (first_in_frame) {
+    packet.is_keyframe = is_key_frame;
+    packet.frame_id = frame_id;
+    RTC_DCHECK(!is_key_frame || ref_frame_ids.empty());
+    packet.frame_dependencies = std::move(ref_frame_ids);
+  }
+  return packet;
+}
+
+class PacketStreamCreator final {
+ public:
+  PacketStreamCreator() : seq_num_(0), frame_id_(0), next_is_key_frame_(true) {}
+
+  Packet NextPacket() {
+    std::vector<int64_t> ref_frame_ids;
+    if (!next_is_key_frame_) {
+      ref_frame_ids.push_back(frame_id_ - 1);
+    }
+
+    Packet packet = CreatePacket(true, true, seq_num_++, frame_id_++,
+                                 next_is_key_frame_, ref_frame_ids);
+
+    next_is_key_frame_ = false;
+
+    return packet;
+  }
+
+ private:
+  uint16_t seq_num_;
+  int64_t frame_id_;
+  bool next_is_key_frame_;
+};
+}  // namespace
+
+// Most of the logic for the tests is here. Subclasses allow parameterizing
+// the test, or adding some more specific logic.
+class LossNotificationControllerBaseTest : public ::testing::Test,
+                                           public KeyFrameRequestSender,
+                                           public LossNotificationSender {
+ protected:
+  LossNotificationControllerBaseTest()
+      : uut_(this, this), key_frame_requested_(false) {}
+
+  ~LossNotificationControllerBaseTest() override {
+    EXPECT_FALSE(LastKeyFrameRequest());
+    EXPECT_FALSE(LastLossNotification());
+  }
+
+  // KeyFrameRequestSender implementation.
+  void RequestKeyFrame() override {
+    EXPECT_FALSE(LastKeyFrameRequest());
+    EXPECT_FALSE(LastLossNotification());
+    key_frame_requested_ = true;
+  }
+
+  // LossNotificationSender implementation.
+  void SendLossNotification(uint16_t last_decoded_seq_num,
+                            uint16_t last_received_seq_num,
+                            bool decodability_flag,
+                            bool buffering_allowed) override {
+    EXPECT_TRUE(buffering_allowed);  // (Flag useful elsewhere.)
+    EXPECT_FALSE(LastKeyFrameRequest());
+    EXPECT_FALSE(LastLossNotification());
+    last_loss_notification_.emplace(last_decoded_seq_num, last_received_seq_num,
+                                    decodability_flag);
+  }
+
+  void OnReceivedPacket(const Packet& packet) {
+    EXPECT_FALSE(LastKeyFrameRequest());
+    EXPECT_FALSE(LastLossNotification());
+
+    if (packet.first_in_frame) {
+      previous_first_packet_in_frame_ = packet;
+      LossNotificationController::FrameDetails frame;
+      frame.is_keyframe = packet.is_keyframe;
+      frame.frame_id = packet.frame_id;
+      frame.frame_dependencies = packet.frame_dependencies;
+      uut_.OnReceivedPacket(packet.seq_num, &frame);
+    } else {
+      uut_.OnReceivedPacket(packet.seq_num, nullptr);
+    }
+  }
+
+  void OnAssembledFrame(uint16_t first_seq_num,
+                        int64_t frame_id,
+                        bool discardable) {
+    EXPECT_FALSE(LastKeyFrameRequest());
+    EXPECT_FALSE(LastLossNotification());
+
+    ASSERT_TRUE(previous_first_packet_in_frame_);
+    uut_.OnAssembledFrame(first_seq_num, frame_id, discardable,
+                          previous_first_packet_in_frame_->frame_dependencies);
+  }
+
+  void ExpectKeyFrameRequest() {
+    EXPECT_EQ(LastLossNotification(), absl::nullopt);
+    EXPECT_TRUE(LastKeyFrameRequest());
+  }
+
+  void ExpectLossNotification(uint16_t last_decoded_seq_num,
+                              uint16_t last_received_seq_num,
+                              bool decodability_flag) {
+    EXPECT_FALSE(LastKeyFrameRequest());
+    const auto last_ln = LastLossNotification();
+    ASSERT_TRUE(last_ln);
+    const LossNotification expected_ln(
+        last_decoded_seq_num, last_received_seq_num, decodability_flag);
+    EXPECT_EQ(expected_ln, *last_ln)
+        << "Expected loss notification (" << expected_ln.ToString()
+        << ") != received loss notification (" << last_ln->ToString() + ")";
+  }
+
+  struct LossNotification {
+    LossNotification(uint16_t last_decoded_seq_num,
+                     uint16_t last_received_seq_num,
+                     bool decodability_flag)
+        : last_decoded_seq_num(last_decoded_seq_num),
+          last_received_seq_num(last_received_seq_num),
+          decodability_flag(decodability_flag) {}
+
+    LossNotification& operator=(const LossNotification& other) = default;
+
+    bool operator==(const LossNotification& other) const {
+      return last_decoded_seq_num == other.last_decoded_seq_num &&
+             last_received_seq_num == other.last_received_seq_num &&
+             decodability_flag == other.decodability_flag;
+    }
+
+    std::string ToString() const {
+      return std::to_string(last_decoded_seq_num) + ", " +
+             std::to_string(last_received_seq_num) + ", " +
+             std::to_string(decodability_flag);
+    }
+
+    uint16_t last_decoded_seq_num;
+    uint16_t last_received_seq_num;
+    bool decodability_flag;
+  };
+
+  bool LastKeyFrameRequest() {
+    const bool result = key_frame_requested_;
+    key_frame_requested_ = false;
+    return result;
+  }
+
+  absl::optional<LossNotification> LastLossNotification() {
+    const absl::optional<LossNotification> result = last_loss_notification_;
+    last_loss_notification_ = absl::nullopt;
+    return result;
+  }
+
+  LossNotificationController uut_;  // Unit under test.
+
+  bool key_frame_requested_;
+
+  absl::optional<LossNotification> last_loss_notification_;
+
+  // First packet of last frame. (Note that if a test skips the first packet
+  // of a subsequent frame, OnAssembledFrame is not called, and so this is
+  // note read. Therefore, it's not a problem if it is not cleared when
+  // the frame changes.)
+  absl::optional<Packet> previous_first_packet_in_frame_;
+};
+
+class LossNotificationControllerTest
+    : public LossNotificationControllerBaseTest,
+      public ::testing::WithParamInterface<std::tuple<bool, bool, bool>> {
+ protected:
+  // Arbitrary parameterized values, to be used by the tests whenever they
+  // wish to either check some combinations, or wish to demonstrate that
+  // a particular arbitrary value is unimportant.
+  template <size_t N>
+  bool Bool() const {
+    return std::get<N>(GetParam());
+  }
+};
+
+INSTANTIATE_TEST_SUITE_P(_,
+                         LossNotificationControllerTest,
+                         ::testing::Combine(::testing::Bool(),
+                                            ::testing::Bool(),
+                                            ::testing::Bool()));
+
+// If the first frame, which is a key frame, is lost, then a new key frame
+// is requested.
+TEST_P(LossNotificationControllerTest,
+       PacketLossBeforeFirstFrameAssembledTriggersKeyFrameRequest) {
+  OnReceivedPacket(CreatePacket(true, false, 100, 0, true));
+  OnReceivedPacket(CreatePacket(Bool<0>(), Bool<1>(), 103, 1, false, {0}));
+  ExpectKeyFrameRequest();
+}
+
+// If packet loss occurs (but not of the first packet), then a loss notification
+// is issued.
+TEST_P(LossNotificationControllerTest,
+       PacketLossAfterFirstFrameAssembledTriggersLossNotification) {
+  OnReceivedPacket(CreatePacket(true, true, 100, 0, true));
+  OnAssembledFrame(100, 0, false);
+  const bool first = Bool<0>();
+  const bool last = Bool<1>();
+  OnReceivedPacket(CreatePacket(first, last, 103, 1, false, {0}));
+  const bool expected_decodability_flag = first;
+  ExpectLossNotification(100, 103, expected_decodability_flag);
+}
+
+// No key frame or loss notifications issued due to an innocuous wrap-around
+// of the sequence number.
+TEST_P(LossNotificationControllerTest, SeqNumWrapAround) {
+  uint16_t seq_num = std::numeric_limits<uint16_t>::max();
+  OnReceivedPacket(CreatePacket(true, true, seq_num, 0, true));
+  OnAssembledFrame(seq_num, 0, false);
+  const bool first = Bool<0>();
+  const bool last = Bool<1>();
+  OnReceivedPacket(CreatePacket(first, last, ++seq_num, 1, false, {0}));
+}
+
+TEST_F(LossNotificationControllerTest,
+       KeyFrameAfterPacketLossProducesNoLossNotifications) {
+  OnReceivedPacket(CreatePacket(true, true, 100, 1, true));
+  OnAssembledFrame(100, 1, false);
+  OnReceivedPacket(CreatePacket(true, true, 108, 8, true));
+}
+
+TEST_P(LossNotificationControllerTest, LostReferenceProducesLossNotification) {
+  OnReceivedPacket(CreatePacket(true, true, 100, 0, true));
+  OnAssembledFrame(100, 0, false);
+  uint16_t last_decodable_non_discardable_seq_num = 100;
+
+  // RTP gap produces loss notification - not the focus of this test.
+  const bool first = Bool<0>();
+  const bool last = Bool<1>();
+  const bool discardable = Bool<2>();
+  const bool decodable = first;  // Depends on assemblability.
+  OnReceivedPacket(CreatePacket(first, last, 107, 3, false, {0}));
+  ExpectLossNotification(100, 107, decodable);
+  OnAssembledFrame(107, 3, discardable);
+  if (!discardable) {
+    last_decodable_non_discardable_seq_num = 107;
+  }
+
+  // Test focus - a loss notification is produced because of the missing
+  // dependency (frame ID 2), despite the RTP sequence number being the
+  // next expected one.
+  OnReceivedPacket(CreatePacket(true, true, 108, 4, false, {2, 0}));
+  ExpectLossNotification(last_decodable_non_discardable_seq_num, 108, false);
+}
+
+// The difference between this test and the previous one, is that in this test,
+// although the reference frame was received, it was not decodable.
+TEST_P(LossNotificationControllerTest,
+       UndecodableReferenceProducesLossNotification) {
+  OnReceivedPacket(CreatePacket(true, true, 100, 0, true));
+  OnAssembledFrame(100, 0, false);
+  uint16_t last_decodable_non_discardable_seq_num = 100;
+
+  // RTP gap produces loss notification - not the focus of this test.
+  // Also, not decodable; this is important for later in the test.
+  OnReceivedPacket(CreatePacket(true, true, 107, 3, false, {2}));
+  ExpectLossNotification(100, 107, false);
+  const bool discardable = Bool<0>();
+  OnAssembledFrame(107, 3, discardable);
+
+  // Test focus - a loss notification is produced because of the undecodable
+  // dependency (frame ID 3, which depended on the missing frame ID 2).
+  OnReceivedPacket(CreatePacket(true, true, 108, 4, false, {3, 0}));
+  ExpectLossNotification(last_decodable_non_discardable_seq_num, 108, false);
+}
+
+TEST_P(LossNotificationControllerTest, RobustnessAgainstHighInitialRefFrameId) {
+  constexpr uint16_t max_uint16_t = std::numeric_limits<uint16_t>::max();
+  OnReceivedPacket(CreatePacket(true, true, 100, 0, true));
+  OnAssembledFrame(100, 0, false);
+  OnReceivedPacket(CreatePacket(true, true, 101, 1, false, {max_uint16_t}));
+  ExpectLossNotification(100, 101, false);
+  OnAssembledFrame(101, max_uint16_t, Bool<0>());
+}
+
+TEST_P(LossNotificationControllerTest, RepeatedPacketsAreIgnored) {
+  PacketStreamCreator packet_stream;
+
+  const auto key_frame_packet = packet_stream.NextPacket();
+  OnReceivedPacket(key_frame_packet);
+  OnAssembledFrame(key_frame_packet.seq_num, key_frame_packet.frame_id, false);
+
+  const bool gap = Bool<0>();
+
+  if (gap) {
+    // Lose one packet.
+    packet_stream.NextPacket();
+  }
+
+  auto repeated_packet = packet_stream.NextPacket();
+  OnReceivedPacket(repeated_packet);
+  if (gap) {
+    // Loss notification issued because of the gap. This is not the focus of
+    // the test.
+    ExpectLossNotification(key_frame_packet.seq_num, repeated_packet.seq_num,
+                           false);
+  }
+  OnReceivedPacket(repeated_packet);
+}
+
+TEST_F(LossNotificationControllerTest,
+       RecognizesDependencyAcrossIntraFrameThatIsNotAKeyframe) {
+  int last_seq_num = 1;
+  auto receive = [&](bool is_key_frame, int64_t frame_id,
+                     std::vector<int64_t> ref_frame_ids) {
+    ++last_seq_num;
+    OnReceivedPacket(CreatePacket(
+        /*first_in_frame=*/true, /*last_in_frame=*/true, last_seq_num, frame_id,
+        is_key_frame, std::move(ref_frame_ids)));
+    OnAssembledFrame(last_seq_num, frame_id, /*discardable=*/false);
+  };
+  //  11 -- 13
+  //   |     |
+  //  10    12
+  receive(/*is_key_frame=*/true, /*frame_id=*/10, /*ref_frame_ids=*/{});
+  receive(/*is_key_frame=*/false, /*frame_id=*/11, /*ref_frame_ids=*/{10});
+  receive(/*is_key_frame=*/false, /*frame_id=*/12, /*ref_frame_ids=*/{});
+  receive(/*is_key_frame=*/false, /*frame_id=*/13, /*ref_frame_ids=*/{11, 12});
+  EXPECT_FALSE(LastLossNotification());
+}
+
+class LossNotificationControllerTestDecodabilityFlag
+    : public LossNotificationControllerBaseTest {
+ protected:
+  LossNotificationControllerTestDecodabilityFlag()
+      : key_frame_seq_num_(100),
+        key_frame_frame_id_(0),
+        never_received_frame_id_(key_frame_frame_id_ + 1),
+        seq_num_(0),
+        frame_id_(0) {}
+
+  void ReceiveKeyFrame() {
+    RTC_DCHECK_NE(key_frame_frame_id_, never_received_frame_id_);
+    OnReceivedPacket(CreatePacket(true, true, key_frame_seq_num_,
+                                  key_frame_frame_id_, true));
+    OnAssembledFrame(key_frame_seq_num_, key_frame_frame_id_, false);
+    seq_num_ = key_frame_seq_num_;
+    frame_id_ = key_frame_frame_id_;
+  }
+
+  void ReceivePacket(bool first_packet_in_frame,
+                     bool last_packet_in_frame,
+                     const std::vector<int64_t>& ref_frame_ids) {
+    if (first_packet_in_frame) {
+      frame_id_ += 1;
+    }
+    RTC_DCHECK_NE(frame_id_, never_received_frame_id_);
+    constexpr bool is_key_frame = false;
+    OnReceivedPacket(CreatePacket(first_packet_in_frame, last_packet_in_frame,
+                                  ++seq_num_, frame_id_, is_key_frame,
+                                  ref_frame_ids));
+  }
+
+  void CreateGap() {
+    seq_num_ += 50;
+    frame_id_ += 10;
+  }
+
+  const uint16_t key_frame_seq_num_;
+  const uint16_t key_frame_frame_id_;
+
+  // The tests intentionally never receive this, and can therefore always
+  // use this as an unsatisfied dependency.
+  const int64_t never_received_frame_id_ = 123;
+
+  uint16_t seq_num_;
+  int64_t frame_id_;
+};
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       SinglePacketFrameWithDecodableDependencies) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  const std::vector<int64_t> ref_frame_ids = {key_frame_frame_id_};
+  ReceivePacket(true, true, ref_frame_ids);
+
+  const bool expected_decodability_flag = true;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag);
+}
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       SinglePacketFrameWithUndecodableDependencies) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  const std::vector<int64_t> ref_frame_ids = {never_received_frame_id_};
+  ReceivePacket(true, true, ref_frame_ids);
+
+  const bool expected_decodability_flag = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag);
+}
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       FirstPacketOfMultiPacketFrameWithDecodableDependencies) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  const std::vector<int64_t> ref_frame_ids = {key_frame_frame_id_};
+  ReceivePacket(true, false, ref_frame_ids);
+
+  const bool expected_decodability_flag = true;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag);
+}
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       FirstPacketOfMultiPacketFrameWithUndecodableDependencies) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  const std::vector<int64_t> ref_frame_ids = {never_received_frame_id_};
+  ReceivePacket(true, false, ref_frame_ids);
+
+  const bool expected_decodability_flag = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag);
+}
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       MiddlePacketOfMultiPacketFrameWithDecodableDependenciesIfFirstMissed) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  const std::vector<int64_t> ref_frame_ids = {key_frame_frame_id_};
+  ReceivePacket(false, false, ref_frame_ids);
+
+  const bool expected_decodability_flag = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag);
+}
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       MiddlePacketOfMultiPacketFrameWithUndecodableDependenciesIfFirstMissed) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  const std::vector<int64_t> ref_frame_ids = {never_received_frame_id_};
+  ReceivePacket(false, false, ref_frame_ids);
+
+  const bool expected_decodability_flag = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag);
+}
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       MiddlePacketOfMultiPacketFrameWithDecodableDependenciesIfFirstReceived) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  // First packet in multi-packet frame. A loss notification is produced
+  // because of the gap in RTP sequence numbers.
+  const std::vector<int64_t> ref_frame_ids = {key_frame_frame_id_};
+  ReceivePacket(true, false, ref_frame_ids);
+  const bool expected_decodability_flag_first = true;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag_first);
+
+  // Middle packet in multi-packet frame. No additional gap and the frame is
+  // still potentially decodable, so no additional loss indication.
+  ReceivePacket(false, false, ref_frame_ids);
+  EXPECT_FALSE(LastKeyFrameRequest());
+  EXPECT_FALSE(LastLossNotification());
+}
+
+TEST_F(
+    LossNotificationControllerTestDecodabilityFlag,
+    MiddlePacketOfMultiPacketFrameWithUndecodableDependenciesIfFirstReceived) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  // First packet in multi-packet frame. A loss notification is produced
+  // because of the gap in RTP sequence numbers. The frame is also recognized
+  // as having non-decodable dependencies.
+  const std::vector<int64_t> ref_frame_ids = {never_received_frame_id_};
+  ReceivePacket(true, false, ref_frame_ids);
+  const bool expected_decodability_flag_first = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag_first);
+
+  // Middle packet in multi-packet frame. No additional gap, but the frame is
+  // known to be non-decodable, so we keep issuing loss indications.
+  ReceivePacket(false, false, ref_frame_ids);
+  const bool expected_decodability_flag_middle = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag_middle);
+}
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       LastPacketOfMultiPacketFrameWithDecodableDependenciesIfAllPrevMissed) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  const std::vector<int64_t> ref_frame_ids = {key_frame_frame_id_};
+  ReceivePacket(false, true, ref_frame_ids);
+
+  const bool expected_decodability_flag = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag);
+}
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       LastPacketOfMultiPacketFrameWithUndecodableDependenciesIfAllPrevMissed) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  const std::vector<int64_t> ref_frame_ids = {never_received_frame_id_};
+  ReceivePacket(false, true, ref_frame_ids);
+
+  const bool expected_decodability_flag = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag);
+}
+
+TEST_F(LossNotificationControllerTestDecodabilityFlag,
+       LastPacketOfMultiPacketFrameWithDecodableDependenciesIfAllPrevReceived) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  // First packet in multi-packet frame. A loss notification is produced
+  // because of the gap in RTP sequence numbers.
+  const std::vector<int64_t> ref_frame_ids = {key_frame_frame_id_};
+  ReceivePacket(true, false, ref_frame_ids);
+  const bool expected_decodability_flag_first = true;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag_first);
+
+  // Last packet in multi-packet frame. No additional gap and the frame is
+  // still potentially decodable, so no additional loss indication.
+  ReceivePacket(false, true, ref_frame_ids);
+  EXPECT_FALSE(LastKeyFrameRequest());
+  EXPECT_FALSE(LastLossNotification());
+}
+
+TEST_F(
+    LossNotificationControllerTestDecodabilityFlag,
+    LastPacketOfMultiPacketFrameWithUndecodableDependenciesIfAllPrevReceived) {
+  ReceiveKeyFrame();
+  CreateGap();
+
+  // First packet in multi-packet frame. A loss notification is produced
+  // because of the gap in RTP sequence numbers. The frame is also recognized
+  // as having non-decodable dependencies.
+  const std::vector<int64_t> ref_frame_ids = {never_received_frame_id_};
+  ReceivePacket(true, false, ref_frame_ids);
+  const bool expected_decodability_flag_first = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag_first);
+
+  // Last packet in multi-packet frame. No additional gap, but the frame is
+  // known to be non-decodable, so we keep issuing loss indications.
+  ReceivePacket(false, true, ref_frame_ids);
+  const bool expected_decodability_flag_last = false;
+  ExpectLossNotification(key_frame_seq_num_, seq_num_,
+                         expected_decodability_flag_last);
+}
+
+}  //  namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/media_opt_util.cc b/third_party/libwebrtc/modules/video_coding/media_opt_util.cc
new file mode 100644
index 0000000000..7580c95fc7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/media_opt_util.cc
@@ -0,0 +1,704 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/media_opt_util.h"
+
+#include <math.h>
+
+#include <algorithm>
+
+#include "modules/video_coding/fec_rate_table.h"
+#include "modules/video_coding/internal_defines.h"
+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+// Max value of loss rates in off-line model
+static const int kPacketLossMax = 129;
+
+namespace media_optimization {
+
+VCMProtectionParameters::VCMProtectionParameters()
+    : rtt(0),
+      lossPr(0.0f),
+      bitRate(0.0f),
+      packetsPerFrame(0.0f),
+      packetsPerFrameKey(0.0f),
+      frameRate(0.0f),
+      keyFrameSize(0.0f),
+      fecRateDelta(0),
+      fecRateKey(0),
+      codecWidth(0),
+      codecHeight(0),
+      numLayers(1) {}
+
+VCMProtectionMethod::VCMProtectionMethod()
+    : _effectivePacketLoss(0),
+      _protectionFactorK(0),
+      _protectionFactorD(0),
+      _scaleProtKey(2.0f),
+      _maxPayloadSize(1460),
+      _corrFecCost(1.0),
+      _type(kNone) {}
+
+VCMProtectionMethod::~VCMProtectionMethod() {}
+
+enum VCMProtectionMethodEnum VCMProtectionMethod::Type() const {
+  return _type;
+}
+
+uint8_t VCMProtectionMethod::RequiredPacketLossER() {
+  return _effectivePacketLoss;
+}
+
+uint8_t VCMProtectionMethod::RequiredProtectionFactorK() {
+  return _protectionFactorK;
+}
+
+uint8_t VCMProtectionMethod::RequiredProtectionFactorD() {
+  return _protectionFactorD;
+}
+
+bool VCMProtectionMethod::RequiredUepProtectionK() {
+  return _useUepProtectionK;
+}
+
+bool VCMProtectionMethod::RequiredUepProtectionD() {
+  return _useUepProtectionD;
+}
+
+int VCMProtectionMethod::MaxFramesFec() const {
+  return 1;
+}
+
+VCMNackFecMethod::VCMNackFecMethod(int64_t lowRttNackThresholdMs,
+                                   int64_t highRttNackThresholdMs)
+    : VCMFecMethod(),
+      _lowRttNackMs(lowRttNackThresholdMs),
+      _highRttNackMs(highRttNackThresholdMs),
+      _maxFramesFec(1) {
+  RTC_DCHECK(lowRttNackThresholdMs >= -1 && highRttNackThresholdMs >= -1);
+  RTC_DCHECK(highRttNackThresholdMs == -1 ||
+             lowRttNackThresholdMs <= highRttNackThresholdMs);
+  RTC_DCHECK(lowRttNackThresholdMs > -1 || highRttNackThresholdMs == -1);
+  _type = kNackFec;
+}
+
+VCMNackFecMethod::~VCMNackFecMethod() {
+  //
+}
+bool VCMNackFecMethod::ProtectionFactor(
+    const VCMProtectionParameters* parameters) {
+  // Hybrid Nack FEC has three operational modes:
+  // 1. Low RTT (below kLowRttNackMs) - Nack only: Set FEC rate
+  //    (_protectionFactorD) to zero. -1 means no FEC.
+  // 2. High RTT (above _highRttNackMs) - FEC Only: Keep FEC factors.
+  //    -1 means always allow NACK.
+  // 3. Medium RTT values - Hybrid mode: We will only nack the
+  //    residual following the decoding of the FEC (refer to JB logic). FEC
+  //    delta protection factor will be adjusted based on the RTT.
+
+  // Otherwise: we count on FEC; if the RTT is below a threshold, then we
+  // nack the residual, based on a decision made in the JB.
+
+  // Compute the protection factors
+  VCMFecMethod::ProtectionFactor(parameters);
+  if (_lowRttNackMs == -1 || parameters->rtt < _lowRttNackMs) {
+    _protectionFactorD = 0;
+    VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
+
+    // When in Hybrid mode (RTT range), adjust FEC rates based on the
+    // RTT (NACK effectiveness) - adjustment factor is in the range [0,1].
+  } else if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs) {
+    // TODO(mikhal): Disabling adjustment temporarily.
+    // uint16_t rttIndex = (uint16_t) parameters->rtt;
+    float adjustRtt = 1.0f;  // (float)VCMNackFecTable[rttIndex] / 100.0f;
+
+    // Adjust FEC with NACK on (for delta frame only)
+    // table depends on RTT relative to rttMax (NACK Threshold)
+    _protectionFactorD = rtc::saturated_cast<uint8_t>(
+        adjustRtt * rtc::saturated_cast<float>(_protectionFactorD));
+    // update FEC rates after applying adjustment
+    VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
+  }
+
+  return true;
+}
+
+int VCMNackFecMethod::ComputeMaxFramesFec(
+    const VCMProtectionParameters* parameters) {
+  if (parameters->numLayers > 2) {
+    // For more than 2 temporal layers we will only have FEC on the base layer,
+    // and the base layers will be pretty far apart. Therefore we force one
+    // frame FEC.
+    return 1;
+  }
+  // We set the max number of frames to base the FEC on so that on average
+  // we will have complete frames in one RTT. Note that this is an upper
+  // bound, and that the actual number of frames used for FEC is decided by the
+  // RTP module based on the actual number of packets and the protection factor.
+  float base_layer_framerate =
+      parameters->frameRate /
+      rtc::saturated_cast<float>(1 << (parameters->numLayers - 1));
+  int max_frames_fec = std::max(
+      rtc::saturated_cast<int>(
+          2.0f * base_layer_framerate * parameters->rtt / 1000.0f + 0.5f),
+      1);
+  // `kUpperLimitFramesFec` is the upper limit on how many frames we
+  // allow any FEC to be based on.
+  if (max_frames_fec > kUpperLimitFramesFec) {
+    max_frames_fec = kUpperLimitFramesFec;
+  }
+  return max_frames_fec;
+}
+
+int VCMNackFecMethod::MaxFramesFec() const {
+  return _maxFramesFec;
+}
+
+bool VCMNackFecMethod::BitRateTooLowForFec(
+    const VCMProtectionParameters* parameters) {
+  // Bitrate below which we turn off FEC, regardless of reported packet loss.
+  // The condition should depend on resolution and content. For now, use
+  // threshold on bytes per frame, with some effect for the frame size.
+  // The condition for turning off FEC is also based on other factors,
+  // such as `_numLayers`, `_maxFramesFec`, and `_rtt`.
+  int estimate_bytes_per_frame = 1000 * BitsPerFrame(parameters) / 8;
+  int max_bytes_per_frame = kMaxBytesPerFrameForFec;
+  int num_pixels = parameters->codecWidth * parameters->codecHeight;
+  if (num_pixels <= 352 * 288) {
+    max_bytes_per_frame = kMaxBytesPerFrameForFecLow;
+  } else if (num_pixels > 640 * 480) {
+    max_bytes_per_frame = kMaxBytesPerFrameForFecHigh;
+  }
+  // TODO(marpan): add condition based on maximum frames used for FEC,
+  // and expand condition based on frame size.
+  // Max round trip time threshold in ms.
+  const int64_t kMaxRttTurnOffFec = 200;
+  if (estimate_bytes_per_frame < max_bytes_per_frame &&
+      parameters->numLayers < 3 && parameters->rtt < kMaxRttTurnOffFec) {
+    return true;
+  }
+  return false;
+}
+
+bool VCMNackFecMethod::EffectivePacketLoss(
+    const VCMProtectionParameters* parameters) {
+  // Set the effective packet loss for encoder (based on FEC code).
+  // Compute the effective packet loss and residual packet loss due to FEC.
+  VCMFecMethod::EffectivePacketLoss(parameters);
+  return true;
+}
+
+bool VCMNackFecMethod::UpdateParameters(
+    const VCMProtectionParameters* parameters) {
+  ProtectionFactor(parameters);
+  EffectivePacketLoss(parameters);
+  _maxFramesFec = ComputeMaxFramesFec(parameters);
+  if (BitRateTooLowForFec(parameters)) {
+    _protectionFactorK = 0;
+    _protectionFactorD = 0;
+  }
+
+  // Protection/fec rates obtained above are defined relative to total number
+  // of packets (total rate: source + fec) FEC in RTP module assumes
+  // protection factor is defined relative to source number of packets so we
+  // should convert the factor to reduce mismatch between mediaOpt's rate and
+  // the actual one
+  _protectionFactorK = VCMFecMethod::ConvertFECRate(_protectionFactorK);
+  _protectionFactorD = VCMFecMethod::ConvertFECRate(_protectionFactorD);
+
+  return true;
+}
+
+VCMNackMethod::VCMNackMethod() : VCMProtectionMethod() {
+  _type = kNack;
+}
+
+VCMNackMethod::~VCMNackMethod() {
+  //
+}
+
+bool VCMNackMethod::EffectivePacketLoss(
+    const VCMProtectionParameters* parameter) {
+  // Effective Packet Loss, NA in current version.
+  _effectivePacketLoss = 0;
+  return true;
+}
+
+bool VCMNackMethod::UpdateParameters(
+    const VCMProtectionParameters* parameters) {
+  // Compute the effective packet loss
+  EffectivePacketLoss(parameters);
+
+  // nackCost  = (bitRate - nackCost) * (lossPr)
+  return true;
+}
+
+VCMFecMethod::VCMFecMethod()
+    : VCMProtectionMethod(),
+      rate_control_settings_(RateControlSettings::ParseFromFieldTrials()) {
+  _type = kFec;
+}
+
+VCMFecMethod::~VCMFecMethod() = default;
+
+uint8_t VCMFecMethod::BoostCodeRateKey(uint8_t packetFrameDelta,
+                                       uint8_t packetFrameKey) const {
+  uint8_t boostRateKey = 2;
+  // Default: ratio scales the FEC protection up for I frames
+  uint8_t ratio = 1;
+
+  if (packetFrameDelta > 0) {
+    ratio = (int8_t)(packetFrameKey / packetFrameDelta);
+  }
+  ratio = VCM_MAX(boostRateKey, ratio);
+
+  return ratio;
+}
+
+uint8_t VCMFecMethod::ConvertFECRate(uint8_t codeRateRTP) const {
+  return rtc::saturated_cast<uint8_t>(
+      VCM_MIN(255, (0.5 + 255.0 * codeRateRTP /
+                              rtc::saturated_cast<float>(255 - codeRateRTP))));
+}
+
+// Update FEC with protectionFactorD
+void VCMFecMethod::UpdateProtectionFactorD(uint8_t protectionFactorD) {
+  _protectionFactorD = protectionFactorD;
+}
+
+// Update FEC with protectionFactorK
+void VCMFecMethod::UpdateProtectionFactorK(uint8_t protectionFactorK) {
+  _protectionFactorK = protectionFactorK;
+}
+
+bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
+  // FEC PROTECTION SETTINGS: varies with packet loss and bitrate
+
+  // No protection if (filtered) packetLoss is 0
+  uint8_t packetLoss = rtc::saturated_cast<uint8_t>(255 * parameters->lossPr);
+  if (packetLoss == 0) {
+    _protectionFactorK = 0;
+    _protectionFactorD = 0;
+    return true;
+  }
+
+  // Parameters for FEC setting:
+  // first partition size, thresholds, table pars, spatial resoln fac.
+
+  // First partition protection: ~ 20%
+  uint8_t firstPartitionProt = rtc::saturated_cast<uint8_t>(255 * 0.20);
+
+  // Minimum protection level needed to generate one FEC packet for one
+  // source packet/frame (in RTP sender)
+  uint8_t minProtLevelFec = 85;
+
+  // Threshold on packetLoss and bitRrate/frameRate (=average #packets),
+  // above which we allocate protection to cover at least first partition.
+  uint8_t lossThr = 0;
+  uint8_t packetNumThr = 1;
+
+  // Parameters for range of rate index of table.
+  const uint8_t ratePar1 = 5;
+  const uint8_t ratePar2 = 49;
+
+  // Spatial resolution size, relative to a reference size.
+  float spatialSizeToRef = rtc::saturated_cast<float>(parameters->codecWidth *
+                                                      parameters->codecHeight) /
+                           (rtc::saturated_cast<float>(704 * 576));
+  // resolnFac: This parameter will generally increase/decrease the FEC rate
+  // (for fixed bitRate and packetLoss) based on system size.
+  // Use a smaller exponent (< 1) to control/soften system size effect.
+  const float resolnFac = 1.0 / powf(spatialSizeToRef, 0.3f);
+
+  const int bitRatePerFrame = BitsPerFrame(parameters);
+
+  // Average number of packets per frame (source and fec):
+  const uint8_t avgTotPackets = rtc::saturated_cast<uint8_t>(
+      1.5f + rtc::saturated_cast<float>(bitRatePerFrame) * 1000.0f /
+                 rtc::saturated_cast<float>(8.0 * _maxPayloadSize));
+
+  // FEC rate parameters: for P and I frame
+  uint8_t codeRateDelta = 0;
+  uint8_t codeRateKey = 0;
+
+  // Get index for table: the FEC protection depends on an effective rate.
+  // The range on the rate index corresponds to rates (bps)
+  // from ~200k to ~8000k, for 30fps
+  const uint16_t effRateFecTable =
+      rtc::saturated_cast<uint16_t>(resolnFac * bitRatePerFrame);
+  uint8_t rateIndexTable = rtc::saturated_cast<uint8_t>(
+      VCM_MAX(VCM_MIN((effRateFecTable - ratePar1) / ratePar1, ratePar2), 0));
+
+  // Restrict packet loss range to 50:
+  // current tables defined only up to 50%
+  if (packetLoss >= kPacketLossMax) {
+    packetLoss = kPacketLossMax - 1;
+  }
+  uint16_t indexTable = rateIndexTable * kPacketLossMax + packetLoss;
+
+  // Check on table index
+  RTC_DCHECK_LT(indexTable, kFecRateTableSize);
+
+  // Protection factor for P frame
+  codeRateDelta = kFecRateTable[indexTable];
+
+  if (packetLoss > lossThr && avgTotPackets > packetNumThr) {
+    // Set a minimum based on first partition size.
+    if (codeRateDelta < firstPartitionProt) {
+      codeRateDelta = firstPartitionProt;
+    }
+  }
+
+  // Check limit on amount of protection for P frame; 50% is max.
+  if (codeRateDelta >= kPacketLossMax) {
+    codeRateDelta = kPacketLossMax - 1;
+  }
+
+  // For Key frame:
+  // Effectively at a higher rate, so we scale/boost the rate
+  // The boost factor may depend on several factors: ratio of packet
+  // number of I to P frames, how much protection placed on P frames, etc.
+  const uint8_t packetFrameDelta =
+      rtc::saturated_cast<uint8_t>(0.5 + parameters->packetsPerFrame);
+  const uint8_t packetFrameKey =
+      rtc::saturated_cast<uint8_t>(0.5 + parameters->packetsPerFrameKey);
+  const uint8_t boostKey = BoostCodeRateKey(packetFrameDelta, packetFrameKey);
+
+  rateIndexTable = rtc::saturated_cast<uint8_t>(VCM_MAX(
+      VCM_MIN(1 + (boostKey * effRateFecTable - ratePar1) / ratePar1, ratePar2),
+      0));
+  uint16_t indexTableKey = rateIndexTable * kPacketLossMax + packetLoss;
+
+  indexTableKey = VCM_MIN(indexTableKey, kFecRateTableSize);
+
+  // Check on table index
+  RTC_DCHECK_LT(indexTableKey, kFecRateTableSize);
+
+  // Protection factor for I frame
+  codeRateKey = kFecRateTable[indexTableKey];
+
+  // Boosting for Key frame.
+  int boostKeyProt = _scaleProtKey * codeRateDelta;
+  if (boostKeyProt >= kPacketLossMax) {
+    boostKeyProt = kPacketLossMax - 1;
+  }
+
+  // Make sure I frame protection is at least larger than P frame protection,
+  // and at least as high as filtered packet loss.
+  codeRateKey = rtc::saturated_cast<uint8_t>(
+      VCM_MAX(packetLoss, VCM_MAX(boostKeyProt, codeRateKey)));
+
+  // Check limit on amount of protection for I frame: 50% is max.
+  if (codeRateKey >= kPacketLossMax) {
+    codeRateKey = kPacketLossMax - 1;
+  }
+
+  _protectionFactorK = codeRateKey;
+  _protectionFactorD = codeRateDelta;
+
+  // Generally there is a rate mis-match between the FEC cost estimated
+  // in mediaOpt and the actual FEC cost sent out in RTP module.
+  // This is more significant at low rates (small # of source packets), where
+  // the granularity of the FEC decreases. In this case, non-zero protection
+  // in mediaOpt may generate 0 FEC packets in RTP sender (since actual #FEC
+  // is based on rounding off protectionFactor on actual source packet number).
+  // The correction factor (_corrFecCost) attempts to corrects this, at least
+  // for cases of low rates (small #packets) and low protection levels.
+
+  float numPacketsFl =
+      1.0f + (rtc::saturated_cast<float>(bitRatePerFrame) * 1000.0 /
+                  rtc::saturated_cast<float>(8.0 * _maxPayloadSize) +
+              0.5);
+
+  const float estNumFecGen =
+      0.5f +
+      rtc::saturated_cast<float>(_protectionFactorD * numPacketsFl / 255.0f);
+
+  // We reduce cost factor (which will reduce overhead for FEC and
+  // hybrid method) and not the protectionFactor.
+  _corrFecCost = 1.0f;
+  if (estNumFecGen < 1.1f && _protectionFactorD < minProtLevelFec) {
+    _corrFecCost = 0.5f;
+  }
+  if (estNumFecGen < 0.9f && _protectionFactorD < minProtLevelFec) {
+    _corrFecCost = 0.0f;
+  }
+
+  // DONE WITH FEC PROTECTION SETTINGS
+  return true;
+}
+
+int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) {
+  // When temporal layers are available FEC will only be applied on the base
+  // layer.
+  const float bitRateRatio =
+      webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(
+          parameters->numLayers, 0,
+          rate_control_settings_.Vp8BaseHeavyTl3RateAllocation());
+  float frameRateRatio = powf(1 / 2.0, parameters->numLayers - 1);
+  float bitRate = parameters->bitRate * bitRateRatio;
+  float frameRate = parameters->frameRate * frameRateRatio;
+
+  // TODO(mikhal): Update factor following testing.
+  float adjustmentFactor = 1;
+
+  if (frameRate < 1.0f)
+    frameRate = 1.0f;
+  // Average bits per frame (units of kbits)
+  return rtc::saturated_cast<int>(adjustmentFactor * bitRate / frameRate);
+}
+
+bool VCMFecMethod::EffectivePacketLoss(
+    const VCMProtectionParameters* parameters) {
+  // Effective packet loss to encoder is based on RPL (residual packet loss)
+  // this is a soft setting based on degree of FEC protection
+  // RPL = received/input packet loss - average_FEC_recovery
+  // note: received/input packet loss may be filtered based on FilteredLoss
+
+  // Effective Packet Loss, NA in current version.
+  _effectivePacketLoss = 0;
+
+  return true;
+}
+
+bool VCMFecMethod::UpdateParameters(const VCMProtectionParameters* parameters) {
+  // Compute the protection factor
+  ProtectionFactor(parameters);
+
+  // Compute the effective packet loss
+  EffectivePacketLoss(parameters);
+
+  // Protection/fec rates obtained above is defined relative to total number
+  // of packets (total rate: source+fec) FEC in RTP module assumes protection
+  // factor is defined relative to source number of packets so we should
+  // convert the factor to reduce mismatch between mediaOpt suggested rate and
+  // the actual rate
+  _protectionFactorK = ConvertFECRate(_protectionFactorK);
+  _protectionFactorD = ConvertFECRate(_protectionFactorD);
+
+  return true;
+}
+VCMLossProtectionLogic::VCMLossProtectionLogic(int64_t nowMs)
+    : _currentParameters(),
+      _rtt(0),
+      _lossPr(0.0f),
+      _bitRate(0.0f),
+      _frameRate(0.0f),
+      _keyFrameSize(0.0f),
+      _fecRateKey(0),
+      _fecRateDelta(0),
+      _lastPrUpdateT(0),
+      _lossPr255(0.9999f),
+      _lossPrHistory(),
+      _shortMaxLossPr255(0),
+      _packetsPerFrame(0.9999f),
+      _packetsPerFrameKey(0.9999f),
+      _codecWidth(704),
+      _codecHeight(576),
+      _numLayers(1) {
+  Reset(nowMs);
+}
+
+VCMLossProtectionLogic::~VCMLossProtectionLogic() {
+  Release();
+}
+
+void VCMLossProtectionLogic::SetMethod(
+    enum VCMProtectionMethodEnum newMethodType) {
+  if (_selectedMethod && _selectedMethod->Type() == newMethodType)
+    return;
+
+  switch (newMethodType) {
+    case kNack:
+      _selectedMethod.reset(new VCMNackMethod());
+      break;
+    case kFec:
+      _selectedMethod.reset(new VCMFecMethod());
+      break;
+    case kNackFec:
+      _selectedMethod.reset(new VCMNackFecMethod(kLowRttNackMs, -1));
+      break;
+    case kNone:
+      _selectedMethod.reset();
+      break;
+  }
+  UpdateMethod();
+}
+
+void VCMLossProtectionLogic::UpdateRtt(int64_t rtt) {
+  _rtt = rtt;
+}
+
+void VCMLossProtectionLogic::UpdateMaxLossHistory(uint8_t lossPr255,
+                                                  int64_t now) {
+  if (_lossPrHistory[0].timeMs >= 0 &&
+      now - _lossPrHistory[0].timeMs < kLossPrShortFilterWinMs) {
+    if (lossPr255 > _shortMaxLossPr255) {
+      _shortMaxLossPr255 = lossPr255;
+    }
+  } else {
+    // Only add a new value to the history once a second
+    if (_lossPrHistory[0].timeMs == -1) {
+      // First, no shift
+      _shortMaxLossPr255 = lossPr255;
+    } else {
+      // Shift
+      for (int32_t i = (kLossPrHistorySize - 2); i >= 0; i--) {
+        _lossPrHistory[i + 1].lossPr255 = _lossPrHistory[i].lossPr255;
+        _lossPrHistory[i + 1].timeMs = _lossPrHistory[i].timeMs;
+      }
+    }
+    if (_shortMaxLossPr255 == 0) {
+      _shortMaxLossPr255 = lossPr255;
+    }
+
+    _lossPrHistory[0].lossPr255 = _shortMaxLossPr255;
+    _lossPrHistory[0].timeMs = now;
+    _shortMaxLossPr255 = 0;
+  }
+}
+
+uint8_t VCMLossProtectionLogic::MaxFilteredLossPr(int64_t nowMs) const {
+  uint8_t maxFound = _shortMaxLossPr255;
+  if (_lossPrHistory[0].timeMs == -1) {
+    return maxFound;
+  }
+  for (int32_t i = 0; i < kLossPrHistorySize; i++) {
+    if (_lossPrHistory[i].timeMs == -1) {
+      break;
+    }
+    if (nowMs - _lossPrHistory[i].timeMs >
+        kLossPrHistorySize * kLossPrShortFilterWinMs) {
+      // This sample (and all samples after this) is too old
+      break;
+    }
+    if (_lossPrHistory[i].lossPr255 > maxFound) {
+      // This sample is the largest one this far into the history
+      maxFound = _lossPrHistory[i].lossPr255;
+    }
+  }
+  return maxFound;
+}
+
+uint8_t VCMLossProtectionLogic::FilteredLoss(int64_t nowMs,
+                                             FilterPacketLossMode filter_mode,
+                                             uint8_t lossPr255) {
+  // Update the max window filter.
+  UpdateMaxLossHistory(lossPr255, nowMs);
+
+  // Update the recursive average filter.
+  _lossPr255.Apply(rtc::saturated_cast<float>(nowMs - _lastPrUpdateT),
+                   rtc::saturated_cast<float>(lossPr255));
+  _lastPrUpdateT = nowMs;
+
+  // Filtered loss: default is received loss (no filtering).
+  uint8_t filtered_loss = lossPr255;
+
+  switch (filter_mode) {
+    case kNoFilter:
+      break;
+    case kAvgFilter:
+      filtered_loss = rtc::saturated_cast<uint8_t>(_lossPr255.filtered() + 0.5);
+      break;
+    case kMaxFilter:
+      filtered_loss = MaxFilteredLossPr(nowMs);
+      break;
+  }
+
+  return filtered_loss;
+}
+
+void VCMLossProtectionLogic::UpdateFilteredLossPr(uint8_t packetLossEnc) {
+  _lossPr = rtc::saturated_cast<float>(packetLossEnc) / 255.0;
+}
+
+void VCMLossProtectionLogic::UpdateBitRate(float bitRate) {
+  _bitRate = bitRate;
+}
+
+void VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets,
+                                                   int64_t nowMs) {
+  _packetsPerFrame.Apply(
+      rtc::saturated_cast<float>(nowMs - _lastPacketPerFrameUpdateT), nPackets);
+  _lastPacketPerFrameUpdateT = nowMs;
+}
+
+void VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets,
+                                                      int64_t nowMs) {
+  _packetsPerFrameKey.Apply(
+      rtc::saturated_cast<float>(nowMs - _lastPacketPerFrameUpdateTKey),
+      nPackets);
+  _lastPacketPerFrameUpdateTKey = nowMs;
+}
+
+void VCMLossProtectionLogic::UpdateKeyFrameSize(float keyFrameSize) {
+  _keyFrameSize = keyFrameSize;
+}
+
+void VCMLossProtectionLogic::UpdateFrameSize(size_t width, size_t height) {
+  _codecWidth = width;
+  _codecHeight = height;
+}
+
+void VCMLossProtectionLogic::UpdateNumLayers(int numLayers) {
+  _numLayers = (numLayers == 0) ? 1 : numLayers;
+}
+
+bool VCMLossProtectionLogic::UpdateMethod() {
+  if (!_selectedMethod)
+    return false;
+  _currentParameters.rtt = _rtt;
+  _currentParameters.lossPr = _lossPr;
+  _currentParameters.bitRate = _bitRate;
+  _currentParameters.frameRate = _frameRate;  // rename actual frame rate?
+  _currentParameters.keyFrameSize = _keyFrameSize;
+  _currentParameters.fecRateDelta = _fecRateDelta;
+  _currentParameters.fecRateKey = _fecRateKey;
+  _currentParameters.packetsPerFrame = _packetsPerFrame.filtered();
+  _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.filtered();
+  _currentParameters.codecWidth = _codecWidth;
+  _currentParameters.codecHeight = _codecHeight;
+  _currentParameters.numLayers = _numLayers;
+  return _selectedMethod->UpdateParameters(&_currentParameters);
+}
+
+VCMProtectionMethod* VCMLossProtectionLogic::SelectedMethod() const {
+  return _selectedMethod.get();
+}
+
+VCMProtectionMethodEnum VCMLossProtectionLogic::SelectedType() const {
+  return _selectedMethod ? _selectedMethod->Type() : kNone;
+}
+
+void VCMLossProtectionLogic::Reset(int64_t nowMs) {
+  _lastPrUpdateT = nowMs;
+  _lastPacketPerFrameUpdateT = nowMs;
+  _lastPacketPerFrameUpdateTKey = nowMs;
+  _lossPr255.Reset(0.9999f);
+  _packetsPerFrame.Reset(0.9999f);
+  _fecRateDelta = _fecRateKey = 0;
+  for (int32_t i = 0; i < kLossPrHistorySize; i++) {
+    _lossPrHistory[i].lossPr255 = 0;
+    _lossPrHistory[i].timeMs = -1;
+  }
+  _shortMaxLossPr255 = 0;
+  Release();
+}
+
+void VCMLossProtectionLogic::Release() {
+  _selectedMethod.reset();
+}
+
+}  // namespace media_optimization
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/media_opt_util.h b/third_party/libwebrtc/modules/video_coding/media_opt_util.h
new file mode 100644
index 0000000000..a74d1af6cb
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/media_opt_util.h
@@ -0,0 +1,350 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+#define MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
+
+#include <math.h>
+#include <stdlib.h>
+
+#include <memory>
+
+#include "modules/video_coding/internal_defines.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+#include "rtc_base/numerics/exp_filter.h"
+
+namespace webrtc {
+namespace media_optimization {
+
+// Number of time periods used for (max) window filter for packet loss
+// TODO(marpan): set reasonable window size for filtered packet loss,
+// adjustment should be based on logged/real data of loss stats/correlation.
+constexpr int kLossPrHistorySize = 10;
+
+// 1000 ms, total filter length is (kLossPrHistorySize * 1000) ms
+constexpr int kLossPrShortFilterWinMs = 1000;
+
+// The type of filter used on the received packet loss reports.
+enum FilterPacketLossMode {
+  kNoFilter,   // No filtering on received loss.
+  kAvgFilter,  // Recursive average filter.
+  kMaxFilter   // Max-window filter, over the time interval of:
+               // (kLossPrHistorySize * kLossPrShortFilterWinMs) ms.
+};
+
+// Thresholds for hybrid NACK/FEC
+// common to media optimization and the jitter buffer.
+constexpr int64_t kLowRttNackMs = 20;
+
+// If the RTT is higher than this an extra RTT wont be added to to the jitter
+// buffer delay.
+constexpr int kMaxRttDelayThreshold = 500;
+
+struct VCMProtectionParameters {
+  VCMProtectionParameters();
+
+  int64_t rtt;
+  float lossPr;
+  float bitRate;
+  float packetsPerFrame;
+  float packetsPerFrameKey;
+  float frameRate;
+  float keyFrameSize;
+  uint8_t fecRateDelta;
+  uint8_t fecRateKey;
+  uint16_t codecWidth;
+  uint16_t codecHeight;
+  int numLayers;
+};
+
+/******************************/
+/* VCMProtectionMethod class  */
+/******************************/
+
+enum VCMProtectionMethodEnum { kNack, kFec, kNackFec, kNone };
+
+class VCMLossProbabilitySample {
+ public:
+  VCMLossProbabilitySample() : lossPr255(0), timeMs(-1) {}
+
+  uint8_t lossPr255;
+  int64_t timeMs;
+};
+
+class VCMProtectionMethod {
+ public:
+  VCMProtectionMethod();
+  virtual ~VCMProtectionMethod();
+
+  // Updates the efficiency of the method using the parameters provided
+  //
+  // Input:
+  //         - parameters         : Parameters used to calculate efficiency
+  //
+  // Return value                 : True if this method is recommended in
+  //                                the given conditions.
+  virtual bool UpdateParameters(const VCMProtectionParameters* parameters) = 0;
+
+  // Returns the protection type
+  //
+  // Return value                 : The protection type
+  VCMProtectionMethodEnum Type() const;
+
+  // Returns the effective packet loss for ER, required by this protection
+  // method
+  //
+  // Return value                 : Required effective packet loss
+  virtual uint8_t RequiredPacketLossER();
+
+  // Extracts the FEC protection factor for Key frame, required by this
+  // protection method
+  //
+  // Return value                 : Required protectionFactor for Key frame
+  virtual uint8_t RequiredProtectionFactorK();
+
+  // Extracts the FEC protection factor for Delta frame, required by this
+  // protection method
+  //
+  // Return value                 : Required protectionFactor for delta frame
+  virtual uint8_t RequiredProtectionFactorD();
+
+  // Extracts whether the FEC Unequal protection (UEP) is used for Key frame.
+  //
+  // Return value                 : Required Unequal protection on/off state.
+  virtual bool RequiredUepProtectionK();
+
+  // Extracts whether the the FEC Unequal protection (UEP) is used for Delta
+  // frame.
+  //
+  // Return value                 : Required Unequal protection on/off state.
+  virtual bool RequiredUepProtectionD();
+
+  virtual int MaxFramesFec() const;
+
+ protected:
+  uint8_t _effectivePacketLoss;
+  uint8_t _protectionFactorK;
+  uint8_t _protectionFactorD;
+  // Estimation of residual loss after the FEC
+  float _scaleProtKey;
+  int32_t _maxPayloadSize;
+
+  bool _useUepProtectionK;
+  bool _useUepProtectionD;
+  float _corrFecCost;
+  VCMProtectionMethodEnum _type;
+};
+
+class VCMNackMethod : public VCMProtectionMethod {
+ public:
+  VCMNackMethod();
+  ~VCMNackMethod() override;
+  bool UpdateParameters(const VCMProtectionParameters* parameters) override;
+  // Get the effective packet loss
+  bool EffectivePacketLoss(const VCMProtectionParameters* parameter);
+};
+
+class VCMFecMethod : public VCMProtectionMethod {
+ public:
+  VCMFecMethod();
+  ~VCMFecMethod() override;
+  bool UpdateParameters(const VCMProtectionParameters* parameters) override;
+  // Get the effective packet loss for ER
+  bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+  // Get the FEC protection factors
+  bool ProtectionFactor(const VCMProtectionParameters* parameters);
+  // Get the boost for key frame protection
+  uint8_t BoostCodeRateKey(uint8_t packetFrameDelta,
+                           uint8_t packetFrameKey) const;
+  // Convert the rates: defined relative to total# packets or source# packets
+  uint8_t ConvertFECRate(uint8_t codeRate) const;
+  // Get the average effective recovery from FEC: for random loss model
+  float AvgRecoveryFEC(const VCMProtectionParameters* parameters) const;
+  // Update FEC with protectionFactorD
+  void UpdateProtectionFactorD(uint8_t protectionFactorD);
+  // Update FEC with protectionFactorK
+  void UpdateProtectionFactorK(uint8_t protectionFactorK);
+  // Compute the bits per frame. Account for temporal layers when applicable.
+  int BitsPerFrame(const VCMProtectionParameters* parameters);
+
+ protected:
+  static constexpr int kUpperLimitFramesFec = 6;
+  // Thresholds values for the bytes/frame and round trip time, below which we
+  // may turn off FEC, depending on `_numLayers` and `_maxFramesFec`.
+  // Max bytes/frame for VGA, corresponds to ~140k at 25fps.
+  static constexpr int kMaxBytesPerFrameForFec = 700;
+  // Max bytes/frame for CIF and lower: corresponds to ~80k at 25fps.
+  static constexpr int kMaxBytesPerFrameForFecLow = 400;
+  // Max bytes/frame for frame size larger than VGA, ~200k at 25fps.
+  static constexpr int kMaxBytesPerFrameForFecHigh = 1000;
+
+  const RateControlSettings rate_control_settings_;
+};
+
+class VCMNackFecMethod : public VCMFecMethod {
+ public:
+  VCMNackFecMethod(int64_t lowRttNackThresholdMs,
+                   int64_t highRttNackThresholdMs);
+  ~VCMNackFecMethod() override;
+  bool UpdateParameters(const VCMProtectionParameters* parameters) override;
+  // Get the effective packet loss for ER
+  bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
+  // Get the protection factors
+  bool ProtectionFactor(const VCMProtectionParameters* parameters);
+  // Get the max number of frames the FEC is allowed to be based on.
+  int MaxFramesFec() const override;
+  // Turn off the FEC based on low bitrate and other factors.
+  bool BitRateTooLowForFec(const VCMProtectionParameters* parameters);
+
+ private:
+  int ComputeMaxFramesFec(const VCMProtectionParameters* parameters);
+
+  int64_t _lowRttNackMs;
+  int64_t _highRttNackMs;
+  int _maxFramesFec;
+};
+
+class VCMLossProtectionLogic {
+ public:
+  explicit VCMLossProtectionLogic(int64_t nowMs);
+  ~VCMLossProtectionLogic();
+
+  // Set the protection method to be used
+  //
+  // Input:
+  //        - newMethodType    : New requested protection method type. If one
+  //                           is already set, it will be deleted and replaced
+  void SetMethod(VCMProtectionMethodEnum newMethodType);
+
+  // Update the round-trip time
+  //
+  // Input:
+  //          - rtt           : Round-trip time in seconds.
+  void UpdateRtt(int64_t rtt);
+
+  // Update the filtered packet loss.
+  //
+  // Input:
+  //          - packetLossEnc :  The reported packet loss filtered
+  //                             (max window or average)
+  void UpdateFilteredLossPr(uint8_t packetLossEnc);
+
+  // Update the current target bit rate.
+  //
+  // Input:
+  //          - bitRate          : The current target bit rate in kbits/s
+  void UpdateBitRate(float bitRate);
+
+  // Update the number of packets per frame estimate, for delta frames
+  //
+  // Input:
+  //          - nPackets         : Number of packets in the latest sent frame.
+  void UpdatePacketsPerFrame(float nPackets, int64_t nowMs);
+
+  // Update the number of packets per frame estimate, for key frames
+  //
+  // Input:
+  //          - nPackets         : umber of packets in the latest sent frame.
+  void UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs);
+
+  // Update the keyFrameSize estimate
+  //
+  // Input:
+  //          - keyFrameSize     : The size of the latest sent key frame.
+  void UpdateKeyFrameSize(float keyFrameSize);
+
+  // Update the frame rate
+  //
+  // Input:
+  //          - frameRate        : The current target frame rate.
+  void UpdateFrameRate(float frameRate) { _frameRate = frameRate; }
+
+  // Update the frame size
+  //
+  // Input:
+  //          - width        : The codec frame width.
+  //          - height       : The codec frame height.
+  void UpdateFrameSize(size_t width, size_t height);
+
+  // Update the number of active layers
+  //
+  // Input:
+  //          - numLayers    : Number of layers used.
+  void UpdateNumLayers(int numLayers);
+
+  // The amount of packet loss to cover for with FEC.
+  //
+  // Input:
+  //          - fecRateKey      : Packet loss to cover for with FEC when
+  //                              sending key frames.
+  //          - fecRateDelta    : Packet loss to cover for with FEC when
+  //                              sending delta frames.
+  void UpdateFECRates(uint8_t fecRateKey, uint8_t fecRateDelta) {
+    _fecRateKey = fecRateKey;
+    _fecRateDelta = fecRateDelta;
+  }
+
+  // Update the protection methods with the current VCMProtectionParameters
+  // and set the requested protection settings.
+  // Return value     : Returns true on update
+  bool UpdateMethod();
+
+  // Returns the method currently selected.
+  //
+  // Return value                 : The protection method currently selected.
+  VCMProtectionMethod* SelectedMethod() const;
+
+  // Return the protection type of the currently selected method
+  VCMProtectionMethodEnum SelectedType() const;
+
+  // Updates the filtered loss for the average and max window packet loss,
+  // and returns the filtered loss probability in the interval [0, 255].
+  // The returned filtered loss value depends on the parameter `filter_mode`.
+  // The input parameter `lossPr255` is the received packet loss.
+
+  // Return value                 : The filtered loss probability
+  uint8_t FilteredLoss(int64_t nowMs,
+                       FilterPacketLossMode filter_mode,
+                       uint8_t lossPr255);
+
+  void Reset(int64_t nowMs);
+
+  void Release();
+
+ private:
+  // Sets the available loss protection methods.
+  void UpdateMaxLossHistory(uint8_t lossPr255, int64_t now);
+  uint8_t MaxFilteredLossPr(int64_t nowMs) const;
+  std::unique_ptr<VCMProtectionMethod> _selectedMethod;
+  VCMProtectionParameters _currentParameters;
+  int64_t _rtt;
+  float _lossPr;
+  float _bitRate;
+  float _frameRate;
+  float _keyFrameSize;
+  uint8_t _fecRateKey;
+  uint8_t _fecRateDelta;
+  int64_t _lastPrUpdateT;
+  int64_t _lastPacketPerFrameUpdateT;
+  int64_t _lastPacketPerFrameUpdateTKey;
+  rtc::ExpFilter _lossPr255;
+  VCMLossProbabilitySample _lossPrHistory[kLossPrHistorySize];
+  uint8_t _shortMaxLossPr255;
+  rtc::ExpFilter _packetsPerFrame;
+  rtc::ExpFilter _packetsPerFrameKey;
+  size_t _codecWidth;
+  size_t _codecHeight;
+  int _numLayers;
+};
+
+}  // namespace media_optimization
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/nack_requester.cc b/third_party/libwebrtc/modules/video_coding/nack_requester.cc
new file mode 100644
index 0000000000..4e74032d01
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/nack_requester.cc
@@ -0,0 +1,340 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/nack_requester.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kMaxPacketAge = 10'000;
+constexpr int kMaxNackPackets = 1000;
+constexpr TimeDelta kDefaultRtt = TimeDelta::Millis(100);
+constexpr int kMaxNackRetries = 10;
+constexpr int kMaxReorderedPackets = 128;
+constexpr int kNumReorderingBuckets = 10;
+constexpr TimeDelta kDefaultSendNackDelay = TimeDelta::Zero();
+
+TimeDelta GetSendNackDelay(const FieldTrialsView& field_trials) {
+  int64_t delay_ms = strtol(
+      field_trials.Lookup("WebRTC-SendNackDelayMs").c_str(), nullptr, 10);
+  if (delay_ms > 0 && delay_ms <= 20) {
+    RTC_LOG(LS_INFO) << "SendNackDelay is set to " << delay_ms;
+    return TimeDelta::Millis(delay_ms);
+  }
+  return kDefaultSendNackDelay;
+}
+}  // namespace
+
+constexpr TimeDelta NackPeriodicProcessor::kUpdateInterval;
+
+NackPeriodicProcessor::NackPeriodicProcessor(TimeDelta update_interval)
+    : update_interval_(update_interval) {}
+
+NackPeriodicProcessor::~NackPeriodicProcessor() {}
+
+void NackPeriodicProcessor::RegisterNackModule(NackRequesterBase* module) {
+  RTC_DCHECK_RUN_ON(&sequence_);
+  modules_.push_back(module);
+  if (modules_.size() != 1)
+    return;
+  repeating_task_ = RepeatingTaskHandle::DelayedStart(
+      TaskQueueBase::Current(), update_interval_, [this] {
+        RTC_DCHECK_RUN_ON(&sequence_);
+        ProcessNackModules();
+        return update_interval_;
+      });
+}
+
+void NackPeriodicProcessor::UnregisterNackModule(NackRequesterBase* module) {
+  RTC_DCHECK_RUN_ON(&sequence_);
+  auto it = std::find(modules_.begin(), modules_.end(), module);
+  RTC_DCHECK(it != modules_.end());
+  modules_.erase(it);
+  if (modules_.empty())
+    repeating_task_.Stop();
+}
+
+void NackPeriodicProcessor::ProcessNackModules() {
+  RTC_DCHECK_RUN_ON(&sequence_);
+  for (NackRequesterBase* module : modules_)
+    module->ProcessNacks();
+}
+
+ScopedNackPeriodicProcessorRegistration::
+    ScopedNackPeriodicProcessorRegistration(NackRequesterBase* module,
+                                            NackPeriodicProcessor* processor)
+    : module_(module), processor_(processor) {
+  processor_->RegisterNackModule(module_);
+}
+
+ScopedNackPeriodicProcessorRegistration::
+    ~ScopedNackPeriodicProcessorRegistration() {
+  processor_->UnregisterNackModule(module_);
+}
+
+NackRequester::NackInfo::NackInfo()
+    : seq_num(0),
+      send_at_seq_num(0),
+      created_at_time(Timestamp::MinusInfinity()),
+      sent_at_time(Timestamp::MinusInfinity()),
+      retries(0) {}
+
+NackRequester::NackInfo::NackInfo(uint16_t seq_num,
+                                  uint16_t send_at_seq_num,
+                                  Timestamp created_at_time)
+    : seq_num(seq_num),
+      send_at_seq_num(send_at_seq_num),
+      created_at_time(created_at_time),
+      sent_at_time(Timestamp::MinusInfinity()),
+      retries(0) {}
+
+NackRequester::NackRequester(TaskQueueBase* current_queue,
+                             NackPeriodicProcessor* periodic_processor,
+                             Clock* clock,
+                             NackSender* nack_sender,
+                             KeyFrameRequestSender* keyframe_request_sender,
+                             const FieldTrialsView& field_trials)
+    : worker_thread_(current_queue),
+      clock_(clock),
+      nack_sender_(nack_sender),
+      keyframe_request_sender_(keyframe_request_sender),
+      reordering_histogram_(kNumReorderingBuckets, kMaxReorderedPackets),
+      initialized_(false),
+      rtt_(kDefaultRtt),
+      newest_seq_num_(0),
+      send_nack_delay_(GetSendNackDelay(field_trials)),
+      processor_registration_(this, periodic_processor) {
+  RTC_DCHECK(clock_);
+  RTC_DCHECK(nack_sender_);
+  RTC_DCHECK(keyframe_request_sender_);
+  RTC_DCHECK(worker_thread_);
+  RTC_DCHECK(worker_thread_->IsCurrent());
+}
+
+NackRequester::~NackRequester() {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+}
+
+void NackRequester::ProcessNacks() {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  std::vector<uint16_t> nack_batch = GetNackBatch(kTimeOnly);
+  if (!nack_batch.empty()) {
+    // This batch of NACKs is triggered externally; there is no external
+    // initiator who can batch them with other feedback messages.
+    nack_sender_->SendNack(nack_batch, /*buffering_allowed=*/false);
+  }
+}
+
+int NackRequester::OnReceivedPacket(uint16_t seq_num, bool is_keyframe) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  return OnReceivedPacket(seq_num, is_keyframe, false);
+}
+
+int NackRequester::OnReceivedPacket(uint16_t seq_num,
+                                    bool is_keyframe,
+                                    bool is_recovered) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  // TODO(philipel): When the packet includes information whether it is
+  //                 retransmitted or not, use that value instead. For
+  //                 now set it to true, which will cause the reordering
+  //                 statistics to never be updated.
+  bool is_retransmitted = true;
+
+  if (!initialized_) {
+    newest_seq_num_ = seq_num;
+    if (is_keyframe)
+      keyframe_list_.insert(seq_num);
+    initialized_ = true;
+    return 0;
+  }
+
+  // Since the `newest_seq_num_` is a packet we have actually received we know
+  // that packet has never been Nacked.
+  if (seq_num == newest_seq_num_)
+    return 0;
+
+  if (AheadOf(newest_seq_num_, seq_num)) {
+    // An out of order packet has been received.
+    auto nack_list_it = nack_list_.find(seq_num);
+    int nacks_sent_for_packet = 0;
+    if (nack_list_it != nack_list_.end()) {
+      nacks_sent_for_packet = nack_list_it->second.retries;
+      nack_list_.erase(nack_list_it);
+    }
+    if (!is_retransmitted)
+      UpdateReorderingStatistics(seq_num);
+    return nacks_sent_for_packet;
+  }
+
+  // Keep track of new keyframes.
+  if (is_keyframe)
+    keyframe_list_.insert(seq_num);
+
+  // And remove old ones so we don't accumulate keyframes.
+  auto it = keyframe_list_.lower_bound(seq_num - kMaxPacketAge);
+  if (it != keyframe_list_.begin())
+    keyframe_list_.erase(keyframe_list_.begin(), it);
+
+  if (is_recovered) {
+    recovered_list_.insert(seq_num);
+
+    // Remove old ones so we don't accumulate recovered packets.
+    auto it = recovered_list_.lower_bound(seq_num - kMaxPacketAge);
+    if (it != recovered_list_.begin())
+      recovered_list_.erase(recovered_list_.begin(), it);
+
+    // Do not send nack for packets recovered by FEC or RTX.
+    return 0;
+  }
+
+  AddPacketsToNack(newest_seq_num_ + 1, seq_num);
+  newest_seq_num_ = seq_num;
+
+  // Are there any nacks that are waiting for this seq_num.
+  std::vector<uint16_t> nack_batch = GetNackBatch(kSeqNumOnly);
+  if (!nack_batch.empty()) {
+    // This batch of NACKs is triggered externally; the initiator can
+    // batch them with other feedback messages.
+    nack_sender_->SendNack(nack_batch, /*buffering_allowed=*/true);
+  }
+
+  return 0;
+}
+
+void NackRequester::ClearUpTo(uint16_t seq_num) {
+  // Called via RtpVideoStreamReceiver2::FrameContinuous on the network thread.
+  worker_thread_->PostTask(SafeTask(task_safety_.flag(), [seq_num, this]() {
+    RTC_DCHECK_RUN_ON(worker_thread_);
+    nack_list_.erase(nack_list_.begin(), nack_list_.lower_bound(seq_num));
+    keyframe_list_.erase(keyframe_list_.begin(),
+                         keyframe_list_.lower_bound(seq_num));
+    recovered_list_.erase(recovered_list_.begin(),
+                          recovered_list_.lower_bound(seq_num));
+  }));
+}
+
+void NackRequester::UpdateRtt(int64_t rtt_ms) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  rtt_ = TimeDelta::Millis(rtt_ms);
+}
+
+bool NackRequester::RemovePacketsUntilKeyFrame() {
+  // Called on worker_thread_.
+  while (!keyframe_list_.empty()) {
+    auto it = nack_list_.lower_bound(*keyframe_list_.begin());
+
+    if (it != nack_list_.begin()) {
+      // We have found a keyframe that actually is newer than at least one
+      // packet in the nack list.
+      nack_list_.erase(nack_list_.begin(), it);
+      return true;
+    }
+
+    // If this keyframe is so old it does not remove any packets from the list,
+    // remove it from the list of keyframes and try the next keyframe.
+    keyframe_list_.erase(keyframe_list_.begin());
+  }
+  return false;
+}
+
+void NackRequester::AddPacketsToNack(uint16_t seq_num_start,
+                                     uint16_t seq_num_end) {
+  // Called on worker_thread_.
+  // Remove old packets.
+  auto it = nack_list_.lower_bound(seq_num_end - kMaxPacketAge);
+  nack_list_.erase(nack_list_.begin(), it);
+
+  // If the nack list is too large, remove packets from the nack list until
+  // the latest first packet of a keyframe. If the list is still too large,
+  // clear it and request a keyframe.
+  uint16_t num_new_nacks = ForwardDiff(seq_num_start, seq_num_end);
+  if (nack_list_.size() + num_new_nacks > kMaxNackPackets) {
+    while (RemovePacketsUntilKeyFrame() &&
+           nack_list_.size() + num_new_nacks > kMaxNackPackets) {
+    }
+
+    if (nack_list_.size() + num_new_nacks > kMaxNackPackets) {
+      nack_list_.clear();
+      RTC_LOG(LS_WARNING) << "NACK list full, clearing NACK"
+                             " list and requesting keyframe.";
+      keyframe_request_sender_->RequestKeyFrame();
+      return;
+    }
+  }
+
+  for (uint16_t seq_num = seq_num_start; seq_num != seq_num_end; ++seq_num) {
+    // Do not send nack for packets that are already recovered by FEC or RTX
+    if (recovered_list_.find(seq_num) != recovered_list_.end())
+      continue;
+    NackInfo nack_info(seq_num, seq_num + WaitNumberOfPackets(0.5),
+                       clock_->CurrentTime());
+    RTC_DCHECK(nack_list_.find(seq_num) == nack_list_.end());
+    nack_list_[seq_num] = nack_info;
+  }
+}
+
+std::vector<uint16_t> NackRequester::GetNackBatch(NackFilterOptions options) {
+  // Called on worker_thread_.
+
+  bool consider_seq_num = options != kTimeOnly;
+  bool consider_timestamp = options != kSeqNumOnly;
+  Timestamp now = clock_->CurrentTime();
+  std::vector<uint16_t> nack_batch;
+  auto it = nack_list_.begin();
+  while (it != nack_list_.end()) {
+    bool delay_timed_out = now - it->second.created_at_time >= send_nack_delay_;
+    bool nack_on_rtt_passed = now - it->second.sent_at_time >= rtt_;
+    bool nack_on_seq_num_passed =
+        it->second.sent_at_time.IsInfinite() &&
+        AheadOrAt(newest_seq_num_, it->second.send_at_seq_num);
+    if (delay_timed_out && ((consider_seq_num && nack_on_seq_num_passed) ||
+                            (consider_timestamp && nack_on_rtt_passed))) {
+      nack_batch.emplace_back(it->second.seq_num);
+      ++it->second.retries;
+      it->second.sent_at_time = now;
+      if (it->second.retries >= kMaxNackRetries) {
+        RTC_LOG(LS_WARNING) << "Sequence number " << it->second.seq_num
+                            << " removed from NACK list due to max retries.";
+        it = nack_list_.erase(it);
+      } else {
+        ++it;
+      }
+      continue;
+    }
+    ++it;
+  }
+  return nack_batch;
+}
+
+void NackRequester::UpdateReorderingStatistics(uint16_t seq_num) {
+  // Running on worker_thread_.
+  RTC_DCHECK(AheadOf(newest_seq_num_, seq_num));
+  uint16_t diff = ReverseDiff(newest_seq_num_, seq_num);
+  reordering_histogram_.Add(diff);
+}
+
+int NackRequester::WaitNumberOfPackets(float probability) const {
+  // Called on worker_thread_;
+  if (reordering_histogram_.NumValues() == 0)
+    return 0;
+  return reordering_histogram_.InverseCdf(probability);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/nack_requester.h b/third_party/libwebrtc/modules/video_coding/nack_requester.h
new file mode 100644
index 0000000000..c860787dcf
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/nack_requester.h
@@ -0,0 +1,157 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_NACK_REQUESTER_H_
+#define MODULES_VIDEO_CODING_NACK_REQUESTER_H_
+
+#include <stdint.h>
+
+#include <map>
+#include <set>
+#include <vector>
+
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/include/module_common_types.h"
+#include "modules/video_coding/histogram.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class NackRequesterBase {
+ public:
+  virtual ~NackRequesterBase() = default;
+  virtual void ProcessNacks() = 0;
+};
+
+class NackPeriodicProcessor {
+ public:
+  static constexpr TimeDelta kUpdateInterval = TimeDelta::Millis(20);
+  explicit NackPeriodicProcessor(TimeDelta update_interval = kUpdateInterval);
+  ~NackPeriodicProcessor();
+  void RegisterNackModule(NackRequesterBase* module);
+  void UnregisterNackModule(NackRequesterBase* module);
+
+ private:
+  void ProcessNackModules() RTC_RUN_ON(sequence_);
+
+  const TimeDelta update_interval_;
+  RepeatingTaskHandle repeating_task_ RTC_GUARDED_BY(sequence_);
+  std::vector<NackRequesterBase*> modules_ RTC_GUARDED_BY(sequence_);
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_;
+};
+
+class ScopedNackPeriodicProcessorRegistration {
+ public:
+  ScopedNackPeriodicProcessorRegistration(NackRequesterBase* module,
+                                          NackPeriodicProcessor* processor);
+  ~ScopedNackPeriodicProcessorRegistration();
+
+ private:
+  NackRequesterBase* const module_;
+  NackPeriodicProcessor* const processor_;
+};
+
+class NackRequester final : public NackRequesterBase {
+ public:
+  NackRequester(TaskQueueBase* current_queue,
+                NackPeriodicProcessor* periodic_processor,
+                Clock* clock,
+                NackSender* nack_sender,
+                KeyFrameRequestSender* keyframe_request_sender,
+                const FieldTrialsView& field_trials);
+  ~NackRequester();
+
+  void ProcessNacks() override;
+
+  int OnReceivedPacket(uint16_t seq_num, bool is_keyframe);
+  int OnReceivedPacket(uint16_t seq_num, bool is_keyframe, bool is_recovered);
+
+  void ClearUpTo(uint16_t seq_num);
+  void UpdateRtt(int64_t rtt_ms);
+
+ private:
+  // Which fields to consider when deciding which packet to nack in
+  // GetNackBatch.
+  enum NackFilterOptions { kSeqNumOnly, kTimeOnly, kSeqNumAndTime };
+
+  // This class holds the sequence number of the packet that is in the nack list
+  // as well as the meta data about when it should be nacked and how many times
+  // we have tried to nack this packet.
+  struct NackInfo {
+    NackInfo();
+    NackInfo(uint16_t seq_num,
+             uint16_t send_at_seq_num,
+             Timestamp created_at_time);
+
+    uint16_t seq_num;
+    uint16_t send_at_seq_num;
+    Timestamp created_at_time;
+    Timestamp sent_at_time;
+    int retries;
+  };
+
+  void AddPacketsToNack(uint16_t seq_num_start, uint16_t seq_num_end)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_thread_);
+
+  // Removes packets from the nack list until the next keyframe. Returns true
+  // if packets were removed.
+  bool RemovePacketsUntilKeyFrame()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_thread_);
+  std::vector<uint16_t> GetNackBatch(NackFilterOptions options)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_thread_);
+
+  // Update the reordering distribution.
+  void UpdateReorderingStatistics(uint16_t seq_num)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_thread_);
+
+  // Returns how many packets we have to wait in order to receive the packet
+  // with probability `probabilty` or higher.
+  int WaitNumberOfPackets(float probability) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_thread_);
+
+  TaskQueueBase* const worker_thread_;
+  Clock* const clock_;
+  NackSender* const nack_sender_;
+  KeyFrameRequestSender* const keyframe_request_sender_;
+
+  // TODO(philipel): Some of the variables below are consistently used on a
+  // known thread (e.g. see `initialized_`). Those probably do not need
+  // synchronized access.
+  std::map<uint16_t, NackInfo, DescendingSeqNumComp<uint16_t>> nack_list_
+      RTC_GUARDED_BY(worker_thread_);
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> keyframe_list_
+      RTC_GUARDED_BY(worker_thread_);
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> recovered_list_
+      RTC_GUARDED_BY(worker_thread_);
+  video_coding::Histogram reordering_histogram_ RTC_GUARDED_BY(worker_thread_);
+  bool initialized_ RTC_GUARDED_BY(worker_thread_);
+  TimeDelta rtt_ RTC_GUARDED_BY(worker_thread_);
+  uint16_t newest_seq_num_ RTC_GUARDED_BY(worker_thread_);
+
+  // Adds a delay before send nack on packet received.
+  const TimeDelta send_nack_delay_;
+
+  ScopedNackPeriodicProcessorRegistration processor_registration_;
+
+  // Used to signal destruction to potentially pending tasks.
+  ScopedTaskSafety task_safety_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_NACK_REQUESTER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/nack_requester_gn/moz.build b/third_party/libwebrtc/modules/video_coding/nack_requester_gn/moz.build
new file mode 100644
index 0000000000..33e763a48d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/nack_requester_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/histogram.cc",
+    "/third_party/libwebrtc/modules/video_coding/nack_requester.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("nack_requester_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/nack_requester_unittest.cc b/third_party/libwebrtc/modules/video_coding/nack_requester_unittest.cc
new file mode 100644
index 0000000000..6f11cb6e91
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/nack_requester_unittest.cc
@@ -0,0 +1,402 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/nack_requester.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstring>
+#include <memory>
+
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/run_loop.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+// TODO(bugs.webrtc.org/11594): Use the use the GlobalSimulatedTimeController
+// instead of RunLoop. At the moment we mix use of the Clock and the underlying
+// implementation of RunLoop, which is realtime.
+class TestNackRequester : public ::testing::Test,
+                          public NackSender,
+                          public KeyFrameRequestSender {
+ protected:
+  TestNackRequester()
+      : clock_(new SimulatedClock(0)), keyframes_requested_(0) {}
+
+  void SetUp() override {}
+
+  void SendNack(const std::vector<uint16_t>& sequence_numbers,
+                bool buffering_allowed) override {
+    sent_nacks_.insert(sent_nacks_.end(), sequence_numbers.begin(),
+                       sequence_numbers.end());
+    if (waiting_for_send_nack_) {
+      waiting_for_send_nack_ = false;
+      loop_.Quit();
+    }
+  }
+
+  void RequestKeyFrame() override { ++keyframes_requested_; }
+
+  void Flush() {
+    // nack_module.Process();
+    loop_.Flush();
+  }
+
+  bool WaitForSendNack() {
+    if (timed_out_) {
+      RTC_DCHECK_NOTREACHED();
+      return false;
+    }
+
+    RTC_DCHECK(!waiting_for_send_nack_);
+
+    waiting_for_send_nack_ = true;
+    loop_.task_queue()->PostDelayedTask(
+        [this]() {
+          timed_out_ = true;
+          loop_.Quit();
+        },
+        TimeDelta::Seconds(1));
+
+    loop_.Run();
+
+    if (timed_out_)
+      return false;
+
+    RTC_DCHECK(!waiting_for_send_nack_);
+    return true;
+  }
+
+  NackRequester& CreateNackModule(
+      TimeDelta interval = NackPeriodicProcessor::kUpdateInterval) {
+    RTC_DCHECK(!nack_module_.get());
+    nack_periodic_processor_ =
+        std::make_unique<NackPeriodicProcessor>(interval);
+    test::ScopedKeyValueConfig empty_field_trials_;
+    nack_module_ = std::make_unique<NackRequester>(
+        TaskQueueBase::Current(), nack_periodic_processor_.get(), clock_.get(),
+        this, this, empty_field_trials_);
+    nack_module_->UpdateRtt(kDefaultRttMs);
+    return *nack_module_.get();
+  }
+
+  static constexpr int64_t kDefaultRttMs = 20;
+  rtc::AutoThread main_thread_;
+  test::RunLoop loop_;
+  std::unique_ptr<SimulatedClock> clock_;
+  std::unique_ptr<NackPeriodicProcessor> nack_periodic_processor_;
+  std::unique_ptr<NackRequester> nack_module_;
+  std::vector<uint16_t> sent_nacks_;
+  int keyframes_requested_;
+  bool waiting_for_send_nack_ = false;
+  bool timed_out_ = false;
+};
+
+TEST_F(TestNackRequester, NackOnePacket) {
+  NackRequester& nack_module = CreateNackModule();
+  nack_module.OnReceivedPacket(1, false, false);
+  nack_module.OnReceivedPacket(3, false, false);
+  ASSERT_EQ(1u, sent_nacks_.size());
+  EXPECT_EQ(2, sent_nacks_[0]);
+}
+
+TEST_F(TestNackRequester, WrappingSeqNum) {
+  NackRequester& nack_module = CreateNackModule();
+  nack_module.OnReceivedPacket(0xfffe, false, false);
+  nack_module.OnReceivedPacket(1, false, false);
+  ASSERT_EQ(2u, sent_nacks_.size());
+  EXPECT_EQ(0xffff, sent_nacks_[0]);
+  EXPECT_EQ(0, sent_nacks_[1]);
+}
+
+TEST_F(TestNackRequester, WrappingSeqNumClearToKeyframe) {
+  NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(10));
+  nack_module.OnReceivedPacket(0xfffe, false, false);
+  nack_module.OnReceivedPacket(1, false, false);
+  ASSERT_EQ(2u, sent_nacks_.size());
+  EXPECT_EQ(0xffff, sent_nacks_[0]);
+  EXPECT_EQ(0, sent_nacks_[1]);
+
+  sent_nacks_.clear();
+  nack_module.OnReceivedPacket(2, true, false);
+  ASSERT_EQ(0u, sent_nacks_.size());
+
+  nack_module.OnReceivedPacket(501, true, false);
+  ASSERT_EQ(498u, sent_nacks_.size());
+  for (int seq_num = 3; seq_num < 501; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 3]);
+
+  sent_nacks_.clear();
+  nack_module.OnReceivedPacket(1001, false, false);
+  EXPECT_EQ(499u, sent_nacks_.size());
+  for (int seq_num = 502; seq_num < 1001; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 502]);
+
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  ASSERT_TRUE(WaitForSendNack());
+  ASSERT_EQ(999u, sent_nacks_.size());
+  EXPECT_EQ(0xffff, sent_nacks_[0]);
+  EXPECT_EQ(0, sent_nacks_[1]);
+  for (int seq_num = 3; seq_num < 501; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 1]);
+  for (int seq_num = 502; seq_num < 1001; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 2]);
+
+  // Adding packet 1004 will cause the nack list to reach it's max limit.
+  // It will then clear all nacks up to the next keyframe (seq num 2),
+  // thus removing 0xffff and 0 from the nack list.
+  sent_nacks_.clear();
+  nack_module.OnReceivedPacket(1004, false, false);
+  ASSERT_EQ(2u, sent_nacks_.size());
+  EXPECT_EQ(1002, sent_nacks_[0]);
+  EXPECT_EQ(1003, sent_nacks_[1]);
+
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  ASSERT_TRUE(WaitForSendNack());
+  ASSERT_EQ(999u, sent_nacks_.size());
+  for (int seq_num = 3; seq_num < 501; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 3]);
+  for (int seq_num = 502; seq_num < 1001; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 4]);
+
+  // Adding packet 1007 will cause the nack module to overflow again, thus
+  // clearing everything up to 501 which is the next keyframe.
+  nack_module.OnReceivedPacket(1007, false, false);
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  ASSERT_TRUE(WaitForSendNack());
+  ASSERT_EQ(503u, sent_nacks_.size());
+  for (int seq_num = 502; seq_num < 1001; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 502]);
+  EXPECT_EQ(1005, sent_nacks_[501]);
+  EXPECT_EQ(1006, sent_nacks_[502]);
+}
+
+TEST_F(TestNackRequester, ResendNack) {
+  NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(1));
+  nack_module.OnReceivedPacket(1, false, false);
+  nack_module.OnReceivedPacket(3, false, false);
+  size_t expected_nacks_sent = 1;
+  ASSERT_EQ(expected_nacks_sent, sent_nacks_.size());
+  EXPECT_EQ(2, sent_nacks_[0]);
+
+  nack_module.UpdateRtt(1);
+  clock_->AdvanceTimeMilliseconds(1);
+  WaitForSendNack();  // Fast retransmit allowed.
+  EXPECT_EQ(++expected_nacks_sent, sent_nacks_.size());
+
+  // Each try has to wait rtt by default.
+  for (int i = 2; i < 10; ++i) {
+    // Change RTT, above the 40ms max for exponential backoff.
+    TimeDelta rtt = TimeDelta::Millis(160);  // + (i * 10 - 40)
+    nack_module.UpdateRtt(rtt.ms());
+
+    // RTT gets capped at 160ms in backoff calculations.
+    TimeDelta expected_backoff_delay =
+        (i - 1) * std::min(rtt, TimeDelta::Millis(160));
+
+    // Move to one millisecond before next allowed NACK.
+    clock_->AdvanceTimeMilliseconds(expected_backoff_delay.ms() - 1);
+    Flush();
+    EXPECT_EQ(expected_nacks_sent, sent_nacks_.size());
+
+    // Move to one millisecond after next allowed NACK.
+    // After rather than on to avoid rounding errors.
+    clock_->AdvanceTimeMilliseconds(2);
+    WaitForSendNack();  // Now allowed.
+    EXPECT_EQ(++expected_nacks_sent, sent_nacks_.size());
+  }
+
+  // Giving up after 10 tries.
+  clock_->AdvanceTimeMilliseconds(3000);
+  Flush();
+  EXPECT_EQ(expected_nacks_sent, sent_nacks_.size());
+}
+
+TEST_F(TestNackRequester, ResendPacketMaxRetries) {
+  NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(1));
+  nack_module.OnReceivedPacket(1, false, false);
+  nack_module.OnReceivedPacket(3, false, false);
+  ASSERT_EQ(1u, sent_nacks_.size());
+  EXPECT_EQ(2, sent_nacks_[0]);
+
+  int backoff_factor = 1;
+  for (size_t retries = 1; retries < 10; ++retries) {
+    // Exponential backoff, so that we don't reject NACK because of time.
+    clock_->AdvanceTimeMilliseconds(backoff_factor * kDefaultRttMs);
+    backoff_factor *= 2;
+    WaitForSendNack();
+    EXPECT_EQ(retries + 1, sent_nacks_.size());
+  }
+
+  clock_->AdvanceTimeMilliseconds(backoff_factor * kDefaultRttMs);
+  Flush();
+  EXPECT_EQ(10u, sent_nacks_.size());
+}
+
+TEST_F(TestNackRequester, TooLargeNackList) {
+  NackRequester& nack_module = CreateNackModule();
+  nack_module.OnReceivedPacket(0, false, false);
+  nack_module.OnReceivedPacket(1001, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(0, keyframes_requested_);
+  nack_module.OnReceivedPacket(1003, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(1, keyframes_requested_);
+  nack_module.OnReceivedPacket(1004, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(1, keyframes_requested_);
+}
+
+TEST_F(TestNackRequester, TooLargeNackListWithKeyFrame) {
+  NackRequester& nack_module = CreateNackModule();
+  nack_module.OnReceivedPacket(0, false, false);
+  nack_module.OnReceivedPacket(1, true, false);
+  nack_module.OnReceivedPacket(1001, false, false);
+  EXPECT_EQ(999u, sent_nacks_.size());
+  EXPECT_EQ(0, keyframes_requested_);
+  nack_module.OnReceivedPacket(1003, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(0, keyframes_requested_);
+  nack_module.OnReceivedPacket(1005, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(1, keyframes_requested_);
+}
+
+TEST_F(TestNackRequester, ClearUpTo) {
+  NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(1));
+  nack_module.OnReceivedPacket(0, false, false);
+  nack_module.OnReceivedPacket(100, false, false);
+  EXPECT_EQ(99u, sent_nacks_.size());
+
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module.ClearUpTo(50);
+  WaitForSendNack();
+  ASSERT_EQ(50u, sent_nacks_.size());
+  EXPECT_EQ(50, sent_nacks_[0]);
+}
+
+TEST_F(TestNackRequester, ClearUpToWrap) {
+  NackRequester& nack_module = CreateNackModule();
+  nack_module.OnReceivedPacket(0xfff0, false, false);
+  nack_module.OnReceivedPacket(0xf, false, false);
+  EXPECT_EQ(30u, sent_nacks_.size());
+
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module.ClearUpTo(0);
+  WaitForSendNack();
+  ASSERT_EQ(15u, sent_nacks_.size());
+  EXPECT_EQ(0, sent_nacks_[0]);
+}
+
+TEST_F(TestNackRequester, PacketNackCount) {
+  NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(1));
+  EXPECT_EQ(0, nack_module.OnReceivedPacket(0, false, false));
+  EXPECT_EQ(0, nack_module.OnReceivedPacket(2, false, false));
+  EXPECT_EQ(1, nack_module.OnReceivedPacket(1, false, false));
+
+  sent_nacks_.clear();
+  nack_module.UpdateRtt(100);
+  EXPECT_EQ(0, nack_module.OnReceivedPacket(5, false, false));
+  clock_->AdvanceTimeMilliseconds(100);
+  WaitForSendNack();
+  EXPECT_EQ(4u, sent_nacks_.size());
+
+  clock_->AdvanceTimeMilliseconds(125);
+  WaitForSendNack();
+
+  EXPECT_EQ(6u, sent_nacks_.size());
+
+  EXPECT_EQ(3, nack_module.OnReceivedPacket(3, false, false));
+  EXPECT_EQ(3, nack_module.OnReceivedPacket(4, false, false));
+  EXPECT_EQ(0, nack_module.OnReceivedPacket(4, false, false));
+}
+
+TEST_F(TestNackRequester, NackListFullAndNoOverlapWithKeyframes) {
+  NackRequester& nack_module = CreateNackModule();
+  const int kMaxNackPackets = 1000;
+  const unsigned int kFirstGap = kMaxNackPackets - 20;
+  const unsigned int kSecondGap = 200;
+  uint16_t seq_num = 0;
+  nack_module.OnReceivedPacket(seq_num++, true, false);
+  seq_num += kFirstGap;
+  nack_module.OnReceivedPacket(seq_num++, true, false);
+  EXPECT_EQ(kFirstGap, sent_nacks_.size());
+  sent_nacks_.clear();
+  seq_num += kSecondGap;
+  nack_module.OnReceivedPacket(seq_num, true, false);
+  EXPECT_EQ(kSecondGap, sent_nacks_.size());
+}
+
+TEST_F(TestNackRequester, HandleFecRecoveredPacket) {
+  NackRequester& nack_module = CreateNackModule();
+  nack_module.OnReceivedPacket(1, false, false);
+  nack_module.OnReceivedPacket(4, false, true);
+  EXPECT_EQ(0u, sent_nacks_.size());
+  nack_module.OnReceivedPacket(5, false, false);
+  EXPECT_EQ(2u, sent_nacks_.size());
+}
+
+TEST_F(TestNackRequester, SendNackWithoutDelay) {
+  NackRequester& nack_module = CreateNackModule();
+  nack_module.OnReceivedPacket(0, false, false);
+  nack_module.OnReceivedPacket(100, false, false);
+  EXPECT_EQ(99u, sent_nacks_.size());
+}
+
+class TestNackRequesterWithFieldTrial : public ::testing::Test,
+                                        public NackSender,
+                                        public KeyFrameRequestSender {
+ protected:
+  TestNackRequesterWithFieldTrial()
+      : nack_delay_field_trial_("WebRTC-SendNackDelayMs/10/"),
+        clock_(new SimulatedClock(0)),
+        nack_module_(TaskQueueBase::Current(),
+                     &nack_periodic_processor_,
+                     clock_.get(),
+                     this,
+                     this,
+                     nack_delay_field_trial_),
+        keyframes_requested_(0) {}
+
+  void SendNack(const std::vector<uint16_t>& sequence_numbers,
+                bool buffering_allowed) override {
+    sent_nacks_.insert(sent_nacks_.end(), sequence_numbers.begin(),
+                       sequence_numbers.end());
+  }
+
+  void RequestKeyFrame() override { ++keyframes_requested_; }
+
+  test::ScopedKeyValueConfig nack_delay_field_trial_;
+  rtc::AutoThread main_thread_;
+  std::unique_ptr<SimulatedClock> clock_;
+  NackPeriodicProcessor nack_periodic_processor_;
+  NackRequester nack_module_;
+  std::vector<uint16_t> sent_nacks_;
+  int keyframes_requested_;
+};
+
+TEST_F(TestNackRequesterWithFieldTrial, SendNackWithDelay) {
+  nack_module_.OnReceivedPacket(0, false, false);
+  nack_module_.OnReceivedPacket(100, false, false);
+  EXPECT_EQ(0u, sent_nacks_.size());
+  clock_->AdvanceTimeMilliseconds(10);
+  nack_module_.OnReceivedPacket(106, false, false);
+  EXPECT_EQ(99u, sent_nacks_.size());
+  clock_->AdvanceTimeMilliseconds(10);
+  nack_module_.OnReceivedPacket(109, false, false);
+  EXPECT_EQ(104u, sent_nacks_.size());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/packet.cc b/third_party/libwebrtc/modules/video_coding/packet.cc
new file mode 100644
index 0000000000..f1bac4a305
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/packet.cc
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/packet.h"
+
+#include "api/rtp_headers.h"
+
+namespace webrtc {
+
+VCMPacket::VCMPacket()
+    : payloadType(0),
+      timestamp(0),
+      ntp_time_ms_(0),
+      seqNum(0),
+      dataPtr(NULL),
+      sizeBytes(0),
+      markerBit(false),
+      timesNacked(-1),
+      completeNALU(kNaluUnset),
+      insertStartCode(false),
+      video_header() {
+  video_header.playout_delay = {-1, -1};
+}
+
+VCMPacket::VCMPacket(const uint8_t* ptr,
+                     size_t size,
+                     const RTPHeader& rtp_header,
+                     const RTPVideoHeader& videoHeader,
+                     int64_t ntp_time_ms,
+                     Timestamp receive_time)
+    : payloadType(rtp_header.payloadType),
+      timestamp(rtp_header.timestamp),
+      ntp_time_ms_(ntp_time_ms),
+      seqNum(rtp_header.sequenceNumber),
+      dataPtr(ptr),
+      sizeBytes(size),
+      markerBit(rtp_header.markerBit),
+      timesNacked(-1),
+      completeNALU(kNaluIncomplete),
+      insertStartCode(videoHeader.codec == kVideoCodecH264 &&
+                      videoHeader.is_first_packet_in_frame),
+      video_header(videoHeader),
+      packet_info(rtp_header, receive_time) {
+  if (is_first_packet_in_frame() && markerBit) {
+    completeNALU = kNaluComplete;
+  } else if (is_first_packet_in_frame()) {
+    completeNALU = kNaluStart;
+  } else if (markerBit) {
+    completeNALU = kNaluEnd;
+  } else {
+    completeNALU = kNaluIncomplete;
+  }
+
+  // Playout decisions are made entirely based on first packet in a frame.
+  if (!is_first_packet_in_frame()) {
+    video_header.playout_delay = {-1, -1};
+  }
+}
+
+VCMPacket::~VCMPacket() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/packet.h b/third_party/libwebrtc/modules/video_coding/packet.h
new file mode 100644
index 0000000000..9aa2d5ce08
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/packet.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_PACKET_H_
+#define MODULES_VIDEO_CODING_PACKET_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/rtp_headers.h"
+#include "api/rtp_packet_info.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_frame_type.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+
+namespace webrtc {
+
+// Used to indicate if a received packet contain a complete NALU (or equivalent)
+enum VCMNaluCompleteness {
+  kNaluUnset = 0,     // Packet has not been filled.
+  kNaluComplete = 1,  // Packet can be decoded as is.
+  kNaluStart,         // Packet contain beginning of NALU
+  kNaluIncomplete,    // Packet is not beginning or end of NALU
+  kNaluEnd,           // Packet is the end of a NALU
+};
+
+class VCMPacket {
+ public:
+  VCMPacket();
+
+  VCMPacket(const uint8_t* ptr,
+            size_t size,
+            const RTPHeader& rtp_header,
+            const RTPVideoHeader& video_header,
+            int64_t ntp_time_ms,
+            Timestamp receive_time);
+
+  ~VCMPacket();
+
+  VideoCodecType codec() const { return video_header.codec; }
+  int width() const { return video_header.width; }
+  int height() const { return video_header.height; }
+
+  bool is_first_packet_in_frame() const {
+    return video_header.is_first_packet_in_frame;
+  }
+  bool is_last_packet_in_frame() const {
+    return video_header.is_last_packet_in_frame;
+  }
+
+  uint8_t payloadType;
+  uint32_t timestamp;
+  // NTP time of the capture time in local timebase in milliseconds.
+  int64_t ntp_time_ms_;
+  uint16_t seqNum;
+  const uint8_t* dataPtr;
+  size_t sizeBytes;
+  bool markerBit;
+  int timesNacked;
+
+  VCMNaluCompleteness completeNALU;  // Default is kNaluIncomplete.
+  bool insertStartCode;  // True if a start code should be inserted before this
+                         // packet.
+  RTPVideoHeader video_header;
+  absl::optional<RtpGenericFrameDescriptor> generic_descriptor;
+
+  RtpPacketInfo packet_info;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CODING_PACKET_H_
diff --git a/third_party/libwebrtc/modules/video_coding/packet_buffer.cc b/third_party/libwebrtc/modules/video_coding/packet_buffer.cc
new file mode 100644
index 0000000000..04f02fce97
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/packet_buffer.cc
@@ -0,0 +1,422 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/packet_buffer.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <limits>
+#include <utility>
+#include <vector>
+
+#include "absl/types/variant.h"
+#include "api/array_view.h"
+#include "api/rtp_packet_info.h"
+#include "api/video/video_frame_type.h"
+#include "common_video/h264/h264_common.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/mod_ops.h"
+
+namespace webrtc {
+namespace video_coding {
+
+PacketBuffer::Packet::Packet(const RtpPacketReceived& rtp_packet,
+                             const RTPVideoHeader& video_header)
+    : marker_bit(rtp_packet.Marker()),
+      payload_type(rtp_packet.PayloadType()),
+      seq_num(rtp_packet.SequenceNumber()),
+      timestamp(rtp_packet.Timestamp()),
+      times_nacked(-1),
+      video_header(video_header) {}
+
+PacketBuffer::PacketBuffer(size_t start_buffer_size, size_t max_buffer_size)
+    : max_size_(max_buffer_size),
+      first_seq_num_(0),
+      first_packet_received_(false),
+      is_cleared_to_first_seq_num_(false),
+      buffer_(start_buffer_size),
+      sps_pps_idr_is_h264_keyframe_(false) {
+  RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
+  // Buffer size must always be a power of 2.
+  RTC_DCHECK((start_buffer_size & (start_buffer_size - 1)) == 0);
+  RTC_DCHECK((max_buffer_size & (max_buffer_size - 1)) == 0);
+}
+
+PacketBuffer::~PacketBuffer() {
+  Clear();
+}
+
+PacketBuffer::InsertResult PacketBuffer::InsertPacket(
+    std::unique_ptr<PacketBuffer::Packet> packet) {
+  PacketBuffer::InsertResult result;
+
+  uint16_t seq_num = packet->seq_num;
+  size_t index = seq_num % buffer_.size();
+
+  if (!first_packet_received_) {
+    first_seq_num_ = seq_num;
+    first_packet_received_ = true;
+  } else if (AheadOf(first_seq_num_, seq_num)) {
+    // If we have explicitly cleared past this packet then it's old,
+    // don't insert it, just silently ignore it.
+    if (is_cleared_to_first_seq_num_) {
+      return result;
+    }
+
+    first_seq_num_ = seq_num;
+  }
+
+  if (buffer_[index] != nullptr) {
+    // Duplicate packet, just delete the payload.
+    if (buffer_[index]->seq_num == packet->seq_num) {
+      return result;
+    }
+
+    // The packet buffer is full, try to expand the buffer.
+    while (ExpandBufferSize() && buffer_[seq_num % buffer_.size()] != nullptr) {
+    }
+    index = seq_num % buffer_.size();
+
+    // Packet buffer is still full since we were unable to expand the buffer.
+    if (buffer_[index] != nullptr) {
+      // Clear the buffer, delete payload, and return false to signal that a
+      // new keyframe is needed.
+      RTC_LOG(LS_WARNING) << "Clear PacketBuffer and request key frame.";
+      ClearInternal();
+      result.buffer_cleared = true;
+      return result;
+    }
+  }
+
+  packet->continuous = false;
+  buffer_[index] = std::move(packet);
+
+  UpdateMissingPackets(seq_num);
+
+  received_padding_.erase(
+      received_padding_.begin(),
+      received_padding_.lower_bound(seq_num - (buffer_.size() / 4)));
+
+  result.packets = FindFrames(seq_num);
+  return result;
+}
+
+uint32_t PacketBuffer::ClearTo(uint16_t seq_num) {
+  // We have already cleared past this sequence number, no need to do anything.
+  if (is_cleared_to_first_seq_num_ &&
+      AheadOf<uint16_t>(first_seq_num_, seq_num)) {
+    return 0;
+  }
+
+  // If the packet buffer was cleared between a frame was created and returned.
+  if (!first_packet_received_)
+    return 0;
+
+  // Avoid iterating over the buffer more than once by capping the number of
+  // iterations to the `size_` of the buffer.
+  ++seq_num;
+  uint32_t num_cleared_packets = 0;
+  size_t diff = ForwardDiff<uint16_t>(first_seq_num_, seq_num);
+  size_t iterations = std::min(diff, buffer_.size());
+  for (size_t i = 0; i < iterations; ++i) {
+    auto& stored = buffer_[first_seq_num_ % buffer_.size()];
+    if (stored != nullptr && AheadOf<uint16_t>(seq_num, stored->seq_num)) {
+      ++num_cleared_packets;
+      stored = nullptr;
+    }
+    ++first_seq_num_;
+  }
+
+  // If `diff` is larger than `iterations` it means that we don't increment
+  // `first_seq_num_` until we reach `seq_num`, so we set it here.
+  first_seq_num_ = seq_num;
+
+  is_cleared_to_first_seq_num_ = true;
+  missing_packets_.erase(missing_packets_.begin(),
+                         missing_packets_.lower_bound(seq_num));
+
+  received_padding_.erase(received_padding_.begin(),
+                          received_padding_.lower_bound(seq_num));
+
+  return num_cleared_packets;
+}
+
+void PacketBuffer::Clear() {
+  ClearInternal();
+}
+
+PacketBuffer::InsertResult PacketBuffer::InsertPadding(uint16_t seq_num) {
+  PacketBuffer::InsertResult result;
+  UpdateMissingPackets(seq_num);
+  received_padding_.insert(seq_num);
+  result.packets = FindFrames(static_cast<uint16_t>(seq_num + 1));
+  return result;
+}
+
+void PacketBuffer::ForceSpsPpsIdrIsH264Keyframe() {
+  sps_pps_idr_is_h264_keyframe_ = true;
+}
+
+void PacketBuffer::ResetSpsPpsIdrIsH264Keyframe() {
+  sps_pps_idr_is_h264_keyframe_ = false;
+}
+
+void PacketBuffer::ClearInternal() {
+  for (auto& entry : buffer_) {
+    entry = nullptr;
+  }
+
+  first_packet_received_ = false;
+  is_cleared_to_first_seq_num_ = false;
+  newest_inserted_seq_num_.reset();
+  missing_packets_.clear();
+  received_padding_.clear();
+}
+
+bool PacketBuffer::ExpandBufferSize() {
+  if (buffer_.size() == max_size_) {
+    RTC_LOG(LS_WARNING) << "PacketBuffer is already at max size (" << max_size_
+                        << "), failed to increase size.";
+    return false;
+  }
+
+  size_t new_size = std::min(max_size_, 2 * buffer_.size());
+  std::vector<std::unique_ptr<Packet>> new_buffer(new_size);
+  for (std::unique_ptr<Packet>& entry : buffer_) {
+    if (entry != nullptr) {
+      new_buffer[entry->seq_num % new_size] = std::move(entry);
+    }
+  }
+  buffer_ = std::move(new_buffer);
+  RTC_LOG(LS_INFO) << "PacketBuffer size expanded to " << new_size;
+  return true;
+}
+
+bool PacketBuffer::PotentialNewFrame(uint16_t seq_num) const {
+  size_t index = seq_num % buffer_.size();
+  int prev_index = index > 0 ? index - 1 : buffer_.size() - 1;
+  const auto& entry = buffer_[index];
+  const auto& prev_entry = buffer_[prev_index];
+
+  if (entry == nullptr)
+    return false;
+  if (entry->seq_num != seq_num)
+    return false;
+  if (entry->is_first_packet_in_frame())
+    return true;
+  if (prev_entry == nullptr)
+    return false;
+  if (prev_entry->seq_num != static_cast<uint16_t>(entry->seq_num - 1))
+    return false;
+  if (prev_entry->timestamp != entry->timestamp)
+    return false;
+  if (prev_entry->continuous)
+    return true;
+
+  return false;
+}
+
+std::vector<std::unique_ptr<PacketBuffer::Packet>> PacketBuffer::FindFrames(
+    uint16_t seq_num) {
+  std::vector<std::unique_ptr<PacketBuffer::Packet>> found_frames;
+  auto start = seq_num;
+
+  for (size_t i = 0; i < buffer_.size(); ++i) {
+    if (received_padding_.find(seq_num) != received_padding_.end()) {
+      seq_num += 1;
+      continue;
+    }
+
+    if (!PotentialNewFrame(seq_num)) {
+      break;
+    }
+
+    size_t index = seq_num % buffer_.size();
+    buffer_[index]->continuous = true;
+
+    // If all packets of the frame is continuous, find the first packet of the
+    // frame and add all packets of the frame to the returned packets.
+    if (buffer_[index]->is_last_packet_in_frame()) {
+      uint16_t start_seq_num = seq_num;
+
+      // Find the start index by searching backward until the packet with
+      // the `frame_begin` flag is set.
+      int start_index = index;
+      size_t tested_packets = 0;
+      int64_t frame_timestamp = buffer_[start_index]->timestamp;
+
+      // Identify H.264 keyframes by means of SPS, PPS, and IDR.
+      bool is_h264 = buffer_[start_index]->codec() == kVideoCodecH264;
+      bool has_h264_sps = false;
+      bool has_h264_pps = false;
+      bool has_h264_idr = false;
+      bool is_h264_keyframe = false;
+      int idr_width = -1;
+      int idr_height = -1;
+      bool full_frame_found = false;
+      while (true) {
+        ++tested_packets;
+
+        if (!is_h264) {
+          if (buffer_[start_index] == nullptr ||
+              buffer_[start_index]->is_first_packet_in_frame()) {
+            full_frame_found = buffer_[start_index] != nullptr;
+            break;
+          }
+        }
+
+        if (is_h264) {
+          const auto* h264_header = absl::get_if<RTPVideoHeaderH264>(
+              &buffer_[start_index]->video_header.video_type_header);
+          if (!h264_header || h264_header->nalus_length >= kMaxNalusPerPacket)
+            return found_frames;
+
+          for (size_t j = 0; j < h264_header->nalus_length; ++j) {
+            if (h264_header->nalus[j].type == H264::NaluType::kSps) {
+              has_h264_sps = true;
+            } else if (h264_header->nalus[j].type == H264::NaluType::kPps) {
+              has_h264_pps = true;
+            } else if (h264_header->nalus[j].type == H264::NaluType::kIdr) {
+              has_h264_idr = true;
+            }
+          }
+          if ((sps_pps_idr_is_h264_keyframe_ && has_h264_idr && has_h264_sps &&
+               has_h264_pps) ||
+              (!sps_pps_idr_is_h264_keyframe_ && has_h264_idr)) {
+            is_h264_keyframe = true;
+            // Store the resolution of key frame which is the packet with
+            // smallest index and valid resolution; typically its IDR or SPS
+            // packet; there may be packet preceeding this packet, IDR's
+            // resolution will be applied to them.
+            if (buffer_[start_index]->width() > 0 &&
+                buffer_[start_index]->height() > 0) {
+              idr_width = buffer_[start_index]->width();
+              idr_height = buffer_[start_index]->height();
+            }
+          }
+        }
+
+        if (tested_packets == buffer_.size())
+          break;
+
+        start_index = start_index > 0 ? start_index - 1 : buffer_.size() - 1;
+
+        // In the case of H264 we don't have a frame_begin bit (yes,
+        // `frame_begin` might be set to true but that is a lie). So instead
+        // we traverese backwards as long as we have a previous packet and
+        // the timestamp of that packet is the same as this one. This may cause
+        // the PacketBuffer to hand out incomplete frames.
+        // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7106
+        if (is_h264 && (buffer_[start_index] == nullptr ||
+                        buffer_[start_index]->timestamp != frame_timestamp)) {
+          break;
+        }
+
+        --start_seq_num;
+      }
+
+      if (is_h264) {
+        // Warn if this is an unsafe frame.
+        if (has_h264_idr && (!has_h264_sps || !has_h264_pps)) {
+          RTC_LOG(LS_WARNING)
+              << "Received H.264-IDR frame "
+                 "(SPS: "
+              << has_h264_sps << ", PPS: " << has_h264_pps << "). Treating as "
+              << (sps_pps_idr_is_h264_keyframe_ ? "delta" : "key")
+              << " frame since WebRTC-SpsPpsIdrIsH264Keyframe is "
+              << (sps_pps_idr_is_h264_keyframe_ ? "enabled." : "disabled");
+        }
+
+        // Now that we have decided whether to treat this frame as a key frame
+        // or delta frame in the frame buffer, we update the field that
+        // determines if the RtpFrameObject is a key frame or delta frame.
+        const size_t first_packet_index = start_seq_num % buffer_.size();
+        if (is_h264_keyframe) {
+          buffer_[first_packet_index]->video_header.frame_type =
+              VideoFrameType::kVideoFrameKey;
+          if (idr_width > 0 && idr_height > 0) {
+            // IDR frame was finalized and we have the correct resolution for
+            // IDR; update first packet to have same resolution as IDR.
+            buffer_[first_packet_index]->video_header.width = idr_width;
+            buffer_[first_packet_index]->video_header.height = idr_height;
+          }
+        } else {
+          buffer_[first_packet_index]->video_header.frame_type =
+              VideoFrameType::kVideoFrameDelta;
+        }
+
+        // If this is not a keyframe, make sure there are no gaps in the packet
+        // sequence numbers up until this point.
+        if (!is_h264_keyframe && missing_packets_.upper_bound(start_seq_num) !=
+                                     missing_packets_.begin()) {
+          return found_frames;
+        }
+      }
+
+      if (is_h264 || full_frame_found) {
+        const uint16_t end_seq_num = seq_num + 1;
+        // Use uint16_t type to handle sequence number wrap around case.
+        uint16_t num_packets = end_seq_num - start_seq_num;
+        found_frames.reserve(found_frames.size() + num_packets);
+        for (uint16_t i = start_seq_num; i != end_seq_num; ++i) {
+          std::unique_ptr<Packet>& packet = buffer_[i % buffer_.size()];
+          RTC_DCHECK(packet);
+          RTC_DCHECK_EQ(i, packet->seq_num);
+          // Ensure frame boundary flags are properly set.
+          packet->video_header.is_first_packet_in_frame = (i == start_seq_num);
+          packet->video_header.is_last_packet_in_frame = (i == seq_num);
+          found_frames.push_back(std::move(packet));
+        }
+
+        missing_packets_.erase(missing_packets_.begin(),
+                               missing_packets_.upper_bound(seq_num));
+        received_padding_.erase(received_padding_.lower_bound(start),
+                                received_padding_.upper_bound(seq_num));
+      }
+    }
+    ++seq_num;
+  }
+  return found_frames;
+}
+
+void PacketBuffer::UpdateMissingPackets(uint16_t seq_num) {
+  if (!newest_inserted_seq_num_)
+    newest_inserted_seq_num_ = seq_num;
+
+  const int kMaxPaddingAge = 1000;
+  if (AheadOf(seq_num, *newest_inserted_seq_num_)) {
+    uint16_t old_seq_num = seq_num - kMaxPaddingAge;
+    auto erase_to = missing_packets_.lower_bound(old_seq_num);
+    missing_packets_.erase(missing_packets_.begin(), erase_to);
+
+    // Guard against inserting a large amount of missing packets if there is a
+    // jump in the sequence number.
+    if (AheadOf(old_seq_num, *newest_inserted_seq_num_))
+      *newest_inserted_seq_num_ = old_seq_num;
+
+    ++*newest_inserted_seq_num_;
+    while (AheadOf(seq_num, *newest_inserted_seq_num_)) {
+      missing_packets_.insert(*newest_inserted_seq_num_);
+      ++*newest_inserted_seq_num_;
+    }
+  } else {
+    missing_packets_.erase(seq_num);
+  }
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/packet_buffer.h b/third_party/libwebrtc/modules/video_coding/packet_buffer.h
new file mode 100644
index 0000000000..47b2ffe199
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/packet_buffer.h
@@ -0,0 +1,134 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_PACKET_BUFFER_H_
+#define MODULES_VIDEO_CODING_PACKET_BUFFER_H_
+
+#include <memory>
+#include <queue>
+#include <set>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "api/rtp_packet_info.h"
+#include "api/units/timestamp.h"
+#include "api/video/encoded_image.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class PacketBuffer {
+ public:
+  struct Packet {
+    Packet() = default;
+    Packet(const RtpPacketReceived& rtp_packet,
+           const RTPVideoHeader& video_header);
+    Packet(const Packet&) = delete;
+    Packet(Packet&&) = delete;
+    Packet& operator=(const Packet&) = delete;
+    Packet& operator=(Packet&&) = delete;
+    ~Packet() = default;
+
+    VideoCodecType codec() const { return video_header.codec; }
+    int width() const { return video_header.width; }
+    int height() const { return video_header.height; }
+
+    bool is_first_packet_in_frame() const {
+      return video_header.is_first_packet_in_frame;
+    }
+    bool is_last_packet_in_frame() const {
+      return video_header.is_last_packet_in_frame;
+    }
+
+    // If all its previous packets have been inserted into the packet buffer.
+    // Set and used internally by the PacketBuffer.
+    bool continuous = false;
+    bool marker_bit = false;
+    uint8_t payload_type = 0;
+    uint16_t seq_num = 0;
+    uint32_t timestamp = 0;
+    int times_nacked = -1;
+
+    rtc::CopyOnWriteBuffer video_payload;
+    RTPVideoHeader video_header;
+  };
+  struct InsertResult {
+    std::vector<std::unique_ptr<Packet>> packets;
+    // Indicates if the packet buffer was cleared, which means that a key
+    // frame request should be sent.
+    bool buffer_cleared = false;
+  };
+
+  // Both `start_buffer_size` and `max_buffer_size` must be a power of 2.
+  PacketBuffer(size_t start_buffer_size, size_t max_buffer_size);
+  ~PacketBuffer();
+
+  ABSL_MUST_USE_RESULT InsertResult
+  InsertPacket(std::unique_ptr<Packet> packet);
+  ABSL_MUST_USE_RESULT InsertResult InsertPadding(uint16_t seq_num);
+
+  // Clear all packets older than |seq_num|. Returns the number of packets
+  // cleared.
+  uint32_t ClearTo(uint16_t seq_num);
+  void Clear();
+
+  void ForceSpsPpsIdrIsH264Keyframe();
+  void ResetSpsPpsIdrIsH264Keyframe();
+
+ private:
+  void ClearInternal();
+
+  // Tries to expand the buffer.
+  bool ExpandBufferSize();
+
+  // Test if all previous packets has arrived for the given sequence number.
+  bool PotentialNewFrame(uint16_t seq_num) const;
+
+  // Test if all packets of a frame has arrived, and if so, returns packets to
+  // create frames.
+  std::vector<std::unique_ptr<Packet>> FindFrames(uint16_t seq_num);
+
+  void UpdateMissingPackets(uint16_t seq_num);
+
+  // buffer_.size() and max_size_ must always be a power of two.
+  const size_t max_size_;
+
+  // The fist sequence number currently in the buffer.
+  uint16_t first_seq_num_;
+
+  // If the packet buffer has received its first packet.
+  bool first_packet_received_;
+
+  // If the buffer is cleared to `first_seq_num_`.
+  bool is_cleared_to_first_seq_num_;
+
+  // Buffer that holds the the inserted packets and information needed to
+  // determine continuity between them.
+  std::vector<std::unique_ptr<Packet>> buffer_;
+
+  absl::optional<uint16_t> newest_inserted_seq_num_;
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_;
+
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> received_padding_;
+
+  // Indicates if we should require SPS, PPS, and IDR for a particular
+  // RTP timestamp to treat the corresponding frame as a keyframe.
+  bool sps_pps_idr_is_h264_keyframe_;
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_PACKET_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/packet_buffer_gn/moz.build b/third_party/libwebrtc/modules/video_coding/packet_buffer_gn/moz.build
new file mode 100644
index 0000000000..9561cb88e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/packet_buffer_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/packet_buffer.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("packet_buffer_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/packet_buffer_unittest.cc b/third_party/libwebrtc/modules/video_coding/packet_buffer_unittest.cc
new file mode 100644
index 0000000000..b147977ab6
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/packet_buffer_unittest.cc
@@ -0,0 +1,828 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/packet_buffer.h"
+
+#include <cstring>
+#include <limits>
+#include <ostream>
+#include <string>
+#include <utility>
+
+#include "api/array_view.h"
+#include "common_video/h264/h264_common.h"
+#include "modules/video_coding/frame_object.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/random.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace video_coding {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+using ::testing::Matches;
+using ::testing::Pointee;
+using ::testing::SizeIs;
+
+constexpr int kStartSize = 16;
+constexpr int kMaxSize = 64;
+
+void IgnoreResult(PacketBuffer::InsertResult /*result*/) {}
+
+// Validates frame boundaries are valid and returns first sequence_number for
+// each frame.
+std::vector<uint16_t> StartSeqNums(
+    rtc::ArrayView<const std::unique_ptr<PacketBuffer::Packet>> packets) {
+  std::vector<uint16_t> result;
+  bool frame_boundary = true;
+  for (const auto& packet : packets) {
+    EXPECT_EQ(frame_boundary, packet->is_first_packet_in_frame());
+    if (packet->is_first_packet_in_frame()) {
+      result.push_back(packet->seq_num);
+    }
+    frame_boundary = packet->is_last_packet_in_frame();
+  }
+  EXPECT_TRUE(frame_boundary);
+  return result;
+}
+
+MATCHER_P(StartSeqNumsAre, seq_num, "") {
+  return Matches(ElementsAre(seq_num))(StartSeqNums(arg.packets));
+}
+
+MATCHER_P2(StartSeqNumsAre, seq_num1, seq_num2, "") {
+  return Matches(ElementsAre(seq_num1, seq_num2))(StartSeqNums(arg.packets));
+}
+
+MATCHER(KeyFrame, "") {
+  return arg->is_first_packet_in_frame() &&
+         arg->video_header.frame_type == VideoFrameType::kVideoFrameKey;
+}
+
+MATCHER(DeltaFrame, "") {
+  return arg->is_first_packet_in_frame() &&
+         arg->video_header.frame_type == VideoFrameType::kVideoFrameDelta;
+}
+
+struct PacketBufferInsertResult : public PacketBuffer::InsertResult {
+  explicit PacketBufferInsertResult(PacketBuffer::InsertResult result)
+      : InsertResult(std::move(result)) {}
+};
+
+void PrintTo(const PacketBufferInsertResult& result, std::ostream* os) {
+  *os << "frames: { ";
+  for (const auto& packet : result.packets) {
+    if (packet->is_first_packet_in_frame() &&
+        packet->is_last_packet_in_frame()) {
+      *os << "{sn: " << packet->seq_num << " }";
+    } else if (packet->is_first_packet_in_frame()) {
+      *os << "{sn: [" << packet->seq_num << "-";
+    } else if (packet->is_last_packet_in_frame()) {
+      *os << packet->seq_num << "] }, ";
+    }
+  }
+  *os << " }";
+  if (result.buffer_cleared) {
+    *os << ", buffer_cleared";
+  }
+}
+
+class PacketBufferTest : public ::testing::Test {
+ protected:
+  PacketBufferTest() : rand_(0x7732213), packet_buffer_(kStartSize, kMaxSize) {}
+
+  uint16_t Rand() { return rand_.Rand<uint16_t>(); }
+
+  enum IsKeyFrame { kKeyFrame, kDeltaFrame };
+  enum IsFirst { kFirst, kNotFirst };
+  enum IsLast { kLast, kNotLast };
+
+  PacketBufferInsertResult Insert(uint16_t seq_num,  // packet sequence number
+                                  IsKeyFrame keyframe,  // is keyframe
+                                  IsFirst first,  // is first packet of frame
+                                  IsLast last,    // is last packet of frame
+                                  rtc::ArrayView<const uint8_t> data = {},
+                                  uint32_t timestamp = 123u) {  // rtp timestamp
+    auto packet = std::make_unique<PacketBuffer::Packet>();
+    packet->video_header.codec = kVideoCodecGeneric;
+    packet->timestamp = timestamp;
+    packet->seq_num = seq_num;
+    packet->video_header.frame_type = keyframe == kKeyFrame
+                                          ? VideoFrameType::kVideoFrameKey
+                                          : VideoFrameType::kVideoFrameDelta;
+    packet->video_header.is_first_packet_in_frame = first == kFirst;
+    packet->video_header.is_last_packet_in_frame = last == kLast;
+    packet->video_payload.SetData(data.data(), data.size());
+
+    return PacketBufferInsertResult(
+        packet_buffer_.InsertPacket(std::move(packet)));
+  }
+
+  Random rand_;
+  PacketBuffer packet_buffer_;
+};
+
+TEST_F(PacketBufferTest, InsertOnePacket) {
+  const uint16_t seq_num = Rand();
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kLast).packets, SizeIs(1));
+}
+
+TEST_F(PacketBufferTest, InsertMultiplePackets) {
+  const uint16_t seq_num = Rand();
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kLast).packets, SizeIs(1));
+  EXPECT_THAT(Insert(seq_num + 1, kKeyFrame, kFirst, kLast).packets, SizeIs(1));
+  EXPECT_THAT(Insert(seq_num + 2, kKeyFrame, kFirst, kLast).packets, SizeIs(1));
+  EXPECT_THAT(Insert(seq_num + 3, kKeyFrame, kFirst, kLast).packets, SizeIs(1));
+}
+
+TEST_F(PacketBufferTest, InsertDuplicatePacket) {
+  const uint16_t seq_num = Rand();
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kNotLast).packets, IsEmpty());
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kNotLast).packets, IsEmpty());
+  EXPECT_THAT(Insert(seq_num + 1, kKeyFrame, kNotFirst, kLast).packets,
+              SizeIs(2));
+}
+
+TEST_F(PacketBufferTest, SeqNumWrapOneFrame) {
+  Insert(0xFFFF, kKeyFrame, kFirst, kNotLast);
+  EXPECT_THAT(Insert(0x0, kKeyFrame, kNotFirst, kLast),
+              StartSeqNumsAre(0xFFFF));
+}
+
+TEST_F(PacketBufferTest, SeqNumWrapTwoFrames) {
+  EXPECT_THAT(Insert(0xFFFF, kKeyFrame, kFirst, kLast),
+              StartSeqNumsAre(0xFFFF));
+  EXPECT_THAT(Insert(0x0, kKeyFrame, kFirst, kLast), StartSeqNumsAre(0x0));
+}
+
+TEST_F(PacketBufferTest, InsertOldPackets) {
+  EXPECT_THAT(Insert(100, kKeyFrame, kFirst, kNotLast).packets, IsEmpty());
+  EXPECT_THAT(Insert(102, kDeltaFrame, kFirst, kLast).packets, SizeIs(1));
+  EXPECT_THAT(Insert(101, kKeyFrame, kNotFirst, kLast).packets, SizeIs(2));
+
+  EXPECT_THAT(Insert(100, kKeyFrame, kFirst, kNotLast).packets, IsEmpty());
+  EXPECT_THAT(Insert(100, kKeyFrame, kFirst, kNotLast).packets, IsEmpty());
+  EXPECT_THAT(Insert(102, kDeltaFrame, kFirst, kLast).packets, SizeIs(1));
+
+  packet_buffer_.ClearTo(102);
+  EXPECT_THAT(Insert(102, kDeltaFrame, kFirst, kLast).packets, IsEmpty());
+  EXPECT_THAT(Insert(103, kDeltaFrame, kFirst, kLast).packets, SizeIs(1));
+}
+
+TEST_F(PacketBufferTest, FrameSize) {
+  const uint16_t seq_num = Rand();
+  uint8_t data1[5] = {};
+  uint8_t data2[5] = {};
+  uint8_t data3[5] = {};
+  uint8_t data4[5] = {};
+
+  Insert(seq_num, kKeyFrame, kFirst, kNotLast, data1);
+  Insert(seq_num + 1, kKeyFrame, kNotFirst, kNotLast, data2);
+  Insert(seq_num + 2, kKeyFrame, kNotFirst, kNotLast, data3);
+  auto packets =
+      Insert(seq_num + 3, kKeyFrame, kNotFirst, kLast, data4).packets;
+  // Expect one frame of 4 packets.
+  EXPECT_THAT(StartSeqNums(packets), ElementsAre(seq_num));
+  EXPECT_THAT(packets, SizeIs(4));
+}
+
+TEST_F(PacketBufferTest, ExpandBuffer) {
+  const uint16_t seq_num = Rand();
+
+  Insert(seq_num, kKeyFrame, kFirst, kNotLast);
+  for (int i = 1; i < kStartSize; ++i)
+    EXPECT_FALSE(
+        Insert(seq_num + i, kKeyFrame, kNotFirst, kNotLast).buffer_cleared);
+
+  // Already inserted kStartSize number of packets, inserting the last packet
+  // should increase the buffer size and also result in an assembled frame.
+  EXPECT_FALSE(
+      Insert(seq_num + kStartSize, kKeyFrame, kNotFirst, kLast).buffer_cleared);
+}
+
+TEST_F(PacketBufferTest, SingleFrameExpandsBuffer) {
+  const uint16_t seq_num = Rand();
+
+  Insert(seq_num, kKeyFrame, kFirst, kNotLast);
+  for (int i = 1; i < kStartSize; ++i)
+    Insert(seq_num + i, kKeyFrame, kNotFirst, kNotLast);
+  EXPECT_THAT(Insert(seq_num + kStartSize, kKeyFrame, kNotFirst, kLast),
+              StartSeqNumsAre(seq_num));
+}
+
+TEST_F(PacketBufferTest, ExpandBufferOverflow) {
+  const uint16_t seq_num = Rand();
+
+  EXPECT_FALSE(Insert(seq_num, kKeyFrame, kFirst, kNotLast).buffer_cleared);
+  for (int i = 1; i < kMaxSize; ++i)
+    EXPECT_FALSE(
+        Insert(seq_num + i, kKeyFrame, kNotFirst, kNotLast).buffer_cleared);
+
+  // Already inserted kMaxSize number of packets, inserting the last packet
+  // should overflow the buffer and result in false being returned.
+  EXPECT_TRUE(
+      Insert(seq_num + kMaxSize, kKeyFrame, kNotFirst, kLast).buffer_cleared);
+}
+
+TEST_F(PacketBufferTest, OnePacketOneFrame) {
+  const uint16_t seq_num = Rand();
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num));
+}
+
+TEST_F(PacketBufferTest, TwoPacketsTwoFrames) {
+  const uint16_t seq_num = Rand();
+
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num));
+  EXPECT_THAT(Insert(seq_num + 1, kKeyFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num + 1));
+}
+
+TEST_F(PacketBufferTest, TwoPacketsOneFrames) {
+  const uint16_t seq_num = Rand();
+
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kNotLast).packets, IsEmpty());
+  EXPECT_THAT(Insert(seq_num + 1, kKeyFrame, kNotFirst, kLast),
+              StartSeqNumsAre(seq_num));
+}
+
+TEST_F(PacketBufferTest, ThreePacketReorderingOneFrame) {
+  const uint16_t seq_num = Rand();
+
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kNotLast).packets, IsEmpty());
+  EXPECT_THAT(Insert(seq_num + 2, kKeyFrame, kNotFirst, kLast).packets,
+              IsEmpty());
+  EXPECT_THAT(Insert(seq_num + 1, kKeyFrame, kNotFirst, kNotLast),
+              StartSeqNumsAre(seq_num));
+}
+
+TEST_F(PacketBufferTest, Frames) {
+  const uint16_t seq_num = Rand();
+
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num));
+  EXPECT_THAT(Insert(seq_num + 1, kDeltaFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num + 1));
+  EXPECT_THAT(Insert(seq_num + 2, kDeltaFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num + 2));
+  EXPECT_THAT(Insert(seq_num + 3, kDeltaFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num + 3));
+}
+
+TEST_F(PacketBufferTest, ClearSinglePacket) {
+  const uint16_t seq_num = Rand();
+
+  for (int i = 0; i < kMaxSize; ++i)
+    Insert(seq_num + i, kDeltaFrame, kFirst, kLast);
+
+  packet_buffer_.ClearTo(seq_num);
+  EXPECT_FALSE(
+      Insert(seq_num + kMaxSize, kDeltaFrame, kFirst, kLast).buffer_cleared);
+}
+
+TEST_F(PacketBufferTest, ClearPacketBeforeFullyReceivedFrame) {
+  Insert(0, kKeyFrame, kFirst, kNotLast);
+  Insert(1, kKeyFrame, kNotFirst, kNotLast);
+  packet_buffer_.ClearTo(0);
+  EXPECT_THAT(Insert(2, kKeyFrame, kNotFirst, kLast).packets, IsEmpty());
+}
+
+TEST_F(PacketBufferTest, ClearFullBuffer) {
+  for (int i = 0; i < kMaxSize; ++i)
+    Insert(i, kDeltaFrame, kFirst, kLast);
+
+  packet_buffer_.ClearTo(kMaxSize - 1);
+
+  for (int i = kMaxSize; i < 2 * kMaxSize; ++i)
+    EXPECT_FALSE(Insert(i, kDeltaFrame, kFirst, kLast).buffer_cleared);
+}
+
+TEST_F(PacketBufferTest, DontClearNewerPacket) {
+  EXPECT_THAT(Insert(0, kKeyFrame, kFirst, kLast), StartSeqNumsAre(0));
+  packet_buffer_.ClearTo(0);
+  EXPECT_THAT(Insert(2 * kStartSize, kKeyFrame, kFirst, kLast),
+              StartSeqNumsAre(2 * kStartSize));
+  EXPECT_THAT(Insert(3 * kStartSize + 1, kKeyFrame, kFirst, kNotLast).packets,
+              IsEmpty());
+  packet_buffer_.ClearTo(2 * kStartSize);
+  EXPECT_THAT(Insert(3 * kStartSize + 2, kKeyFrame, kNotFirst, kLast),
+              StartSeqNumsAre(3 * kStartSize + 1));
+}
+
+TEST_F(PacketBufferTest, OneIncompleteFrame) {
+  const uint16_t seq_num = Rand();
+
+  EXPECT_THAT(Insert(seq_num, kDeltaFrame, kFirst, kNotLast).packets,
+              IsEmpty());
+  EXPECT_THAT(Insert(seq_num + 1, kDeltaFrame, kNotFirst, kLast),
+              StartSeqNumsAre(seq_num));
+  EXPECT_THAT(Insert(seq_num - 1, kDeltaFrame, kNotFirst, kLast).packets,
+              IsEmpty());
+}
+
+TEST_F(PacketBufferTest, TwoIncompleteFramesFullBuffer) {
+  const uint16_t seq_num = Rand();
+
+  for (int i = 1; i < kMaxSize - 1; ++i)
+    Insert(seq_num + i, kDeltaFrame, kNotFirst, kNotLast);
+  EXPECT_THAT(Insert(seq_num, kDeltaFrame, kFirst, kNotLast).packets,
+              IsEmpty());
+  EXPECT_THAT(Insert(seq_num - 1, kDeltaFrame, kNotFirst, kLast).packets,
+              IsEmpty());
+}
+
+TEST_F(PacketBufferTest, FramesReordered) {
+  const uint16_t seq_num = Rand();
+
+  EXPECT_THAT(Insert(seq_num + 1, kDeltaFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num + 1));
+  EXPECT_THAT(Insert(seq_num, kKeyFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num));
+  EXPECT_THAT(Insert(seq_num + 3, kDeltaFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num + 3));
+  EXPECT_THAT(Insert(seq_num + 2, kDeltaFrame, kFirst, kLast),
+              StartSeqNumsAre(seq_num + 2));
+}
+
+TEST_F(PacketBufferTest, InsertPacketAfterSequenceNumberWrapAround) {
+  uint16_t kFirstSeqNum = 0;
+  uint32_t kTimestampDelta = 100;
+  uint32_t timestamp = 10000;
+  uint16_t seq_num = kFirstSeqNum;
+
+  // Loop until seq_num wraps around.
+  SeqNumUnwrapper<uint16_t> unwrapper;
+  while (unwrapper.Unwrap(seq_num) < std::numeric_limits<uint16_t>::max()) {
+    Insert(seq_num++, kKeyFrame, kFirst, kNotLast, {}, timestamp);
+    for (int i = 0; i < 5; ++i) {
+      Insert(seq_num++, kKeyFrame, kNotFirst, kNotLast, {}, timestamp);
+    }
+    Insert(seq_num++, kKeyFrame, kNotFirst, kLast, {}, timestamp);
+    timestamp += kTimestampDelta;
+  }
+
+  // Receive frame with overlapping sequence numbers.
+  Insert(seq_num++, kKeyFrame, kFirst, kNotLast, {}, timestamp);
+  for (int i = 0; i < 5; ++i) {
+    Insert(seq_num++, kKeyFrame, kNotFirst, kNotLast, {}, timestamp);
+  }
+  auto packets =
+      Insert(seq_num++, kKeyFrame, kNotFirst, kLast, {}, timestamp).packets;
+  // One frame of 7 packets.
+  EXPECT_THAT(StartSeqNums(packets), SizeIs(1));
+  EXPECT_THAT(packets, SizeIs(7));
+}
+
+// If `sps_pps_idr_is_keyframe` is true, we require keyframes to contain
+// SPS/PPS/IDR and the keyframes we create as part of the test do contain
+// SPS/PPS/IDR. If `sps_pps_idr_is_keyframe` is false, we only require and
+// create keyframes containing only IDR.
+class PacketBufferH264Test : public PacketBufferTest {
+ protected:
+  explicit PacketBufferH264Test(bool sps_pps_idr_is_keyframe)
+      : PacketBufferTest(), sps_pps_idr_is_keyframe_(sps_pps_idr_is_keyframe) {
+    if (sps_pps_idr_is_keyframe) {
+      packet_buffer_.ForceSpsPpsIdrIsH264Keyframe();
+    }
+  }
+
+  PacketBufferInsertResult InsertH264(
+      uint16_t seq_num,     // packet sequence number
+      IsKeyFrame keyframe,  // is keyframe
+      IsFirst first,        // is first packet of frame
+      IsLast last,          // is last packet of frame
+      uint32_t timestamp,   // rtp timestamp
+      rtc::ArrayView<const uint8_t> data = {},
+      uint32_t width = 0,     // width of frame (SPS/IDR)
+      uint32_t height = 0) {  // height of frame (SPS/IDR)
+    auto packet = std::make_unique<PacketBuffer::Packet>();
+    packet->video_header.codec = kVideoCodecH264;
+    auto& h264_header =
+        packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+    packet->seq_num = seq_num;
+    packet->timestamp = timestamp;
+    if (keyframe == kKeyFrame) {
+      if (sps_pps_idr_is_keyframe_) {
+        h264_header.nalus[0].type = H264::NaluType::kSps;
+        h264_header.nalus[1].type = H264::NaluType::kPps;
+        h264_header.nalus[2].type = H264::NaluType::kIdr;
+        h264_header.nalus_length = 3;
+      } else {
+        h264_header.nalus[0].type = H264::NaluType::kIdr;
+        h264_header.nalus_length = 1;
+      }
+    }
+    packet->video_header.width = width;
+    packet->video_header.height = height;
+    packet->video_header.is_first_packet_in_frame = first == kFirst;
+    packet->video_header.is_last_packet_in_frame = last == kLast;
+    packet->video_payload.SetData(data.data(), data.size());
+
+    return PacketBufferInsertResult(
+        packet_buffer_.InsertPacket(std::move(packet)));
+  }
+
+  PacketBufferInsertResult InsertH264KeyFrameWithAud(
+      uint16_t seq_num,     // packet sequence number
+      IsKeyFrame keyframe,  // is keyframe
+      IsFirst first,        // is first packet of frame
+      IsLast last,          // is last packet of frame
+      uint32_t timestamp,   // rtp timestamp
+      rtc::ArrayView<const uint8_t> data = {},
+      uint32_t width = 0,     // width of frame (SPS/IDR)
+      uint32_t height = 0) {  // height of frame (SPS/IDR)
+    auto packet = std::make_unique<PacketBuffer::Packet>();
+    packet->video_header.codec = kVideoCodecH264;
+    auto& h264_header =
+        packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+    packet->seq_num = seq_num;
+    packet->timestamp = timestamp;
+
+    // this should be the start of frame.
+    RTC_CHECK(first == kFirst);
+
+    // Insert a AUD NALU / packet without width/height.
+    h264_header.nalus[0].type = H264::NaluType::kAud;
+    h264_header.nalus_length = 1;
+    packet->video_header.is_first_packet_in_frame = true;
+    packet->video_header.is_last_packet_in_frame = false;
+    IgnoreResult(packet_buffer_.InsertPacket(std::move(packet)));
+    // insert IDR
+    return InsertH264(seq_num + 1, keyframe, kNotFirst, last, timestamp, data,
+                      width, height);
+  }
+
+  const bool sps_pps_idr_is_keyframe_;
+};
+
+// This fixture is used to test the general behaviour of the packet buffer
+// in both configurations.
+class PacketBufferH264ParameterizedTest
+    : public ::testing::WithParamInterface<bool>,
+      public PacketBufferH264Test {
+ protected:
+  PacketBufferH264ParameterizedTest() : PacketBufferH264Test(GetParam()) {}
+};
+
+INSTANTIATE_TEST_SUITE_P(SpsPpsIdrIsKeyframe,
+                         PacketBufferH264ParameterizedTest,
+                         ::testing::Bool());
+
+TEST_P(PacketBufferH264ParameterizedTest, DontRemoveMissingPacketOnClearTo) {
+  InsertH264(0, kKeyFrame, kFirst, kLast, 0);
+  InsertH264(2, kDeltaFrame, kFirst, kNotLast, 2);
+  packet_buffer_.ClearTo(0);
+  // Expect no frame because of missing of packet #1
+  EXPECT_THAT(InsertH264(3, kDeltaFrame, kNotFirst, kLast, 2).packets,
+              IsEmpty());
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, GetBitstreamOneFrameFullBuffer) {
+  uint8_t data_arr[kStartSize][1];
+  uint8_t expected[kStartSize];
+
+  for (uint8_t i = 0; i < kStartSize; ++i) {
+    data_arr[i][0] = i;
+    expected[i] = i;
+  }
+
+  InsertH264(0, kKeyFrame, kFirst, kNotLast, 1, data_arr[0]);
+  for (uint8_t i = 1; i < kStartSize - 1; ++i) {
+    InsertH264(i, kKeyFrame, kNotFirst, kNotLast, 1, data_arr[i]);
+  }
+
+  auto packets = InsertH264(kStartSize - 1, kKeyFrame, kNotFirst, kLast, 1,
+                            data_arr[kStartSize - 1])
+                     .packets;
+  ASSERT_THAT(StartSeqNums(packets), ElementsAre(0));
+  EXPECT_THAT(packets, SizeIs(kStartSize));
+  for (size_t i = 0; i < packets.size(); ++i) {
+    EXPECT_THAT(packets[i]->video_payload, SizeIs(1)) << "Packet #" << i;
+  }
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, GetBitstreamBufferPadding) {
+  uint16_t seq_num = Rand();
+  rtc::CopyOnWriteBuffer data = "some plain old data";
+
+  auto packet = std::make_unique<PacketBuffer::Packet>();
+  auto& h264_header =
+      packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+  h264_header.nalus_length = 1;
+  h264_header.nalus[0].type = H264::NaluType::kIdr;
+  h264_header.packetization_type = kH264SingleNalu;
+  packet->seq_num = seq_num;
+  packet->video_header.codec = kVideoCodecH264;
+  packet->video_payload = data;
+  packet->video_header.is_first_packet_in_frame = true;
+  packet->video_header.is_last_packet_in_frame = true;
+  auto frames = packet_buffer_.InsertPacket(std::move(packet)).packets;
+
+  ASSERT_THAT(frames, SizeIs(1));
+  EXPECT_EQ(frames[0]->seq_num, seq_num);
+  EXPECT_EQ(frames[0]->video_payload, data);
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, FrameResolution) {
+  uint16_t seq_num = 100;
+  uint8_t data[] = "some plain old data";
+  uint32_t width = 640;
+  uint32_t height = 360;
+  uint32_t timestamp = 1000;
+
+  auto packets = InsertH264(seq_num, kKeyFrame, kFirst, kLast, timestamp, data,
+                            width, height)
+                     .packets;
+
+  ASSERT_THAT(packets, SizeIs(1));
+  EXPECT_EQ(packets[0]->video_header.width, width);
+  EXPECT_EQ(packets[0]->video_header.height, height);
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, FrameResolutionNaluBeforeSPS) {
+  uint16_t seq_num = 100;
+  uint8_t data[] = "some plain old data";
+  uint32_t width = 640;
+  uint32_t height = 360;
+  uint32_t timestamp = 1000;
+
+  auto packets = InsertH264KeyFrameWithAud(seq_num, kKeyFrame, kFirst, kLast,
+                                           timestamp, data, width, height)
+                     .packets;
+
+  ASSERT_THAT(StartSeqNums(packets), ElementsAre(seq_num));
+  EXPECT_EQ(packets[0]->video_header.width, width);
+  EXPECT_EQ(packets[0]->video_header.height, height);
+}
+
+TEST_F(PacketBufferTest, FreeSlotsOnFrameCreation) {
+  const uint16_t seq_num = Rand();
+
+  Insert(seq_num, kKeyFrame, kFirst, kNotLast);
+  Insert(seq_num + 1, kDeltaFrame, kNotFirst, kNotLast);
+  EXPECT_THAT(Insert(seq_num + 2, kDeltaFrame, kNotFirst, kLast),
+              StartSeqNumsAre(seq_num));
+
+  // Insert frame that fills the whole buffer.
+  Insert(seq_num + 3, kKeyFrame, kFirst, kNotLast);
+  for (int i = 0; i < kMaxSize - 2; ++i)
+    Insert(seq_num + i + 4, kDeltaFrame, kNotFirst, kNotLast);
+  EXPECT_THAT(Insert(seq_num + kMaxSize + 2, kKeyFrame, kNotFirst, kLast),
+              StartSeqNumsAre(seq_num + 3));
+}
+
+TEST_F(PacketBufferTest, Clear) {
+  const uint16_t seq_num = Rand();
+
+  Insert(seq_num, kKeyFrame, kFirst, kNotLast);
+  Insert(seq_num + 1, kDeltaFrame, kNotFirst, kNotLast);
+  EXPECT_THAT(Insert(seq_num + 2, kDeltaFrame, kNotFirst, kLast),
+              StartSeqNumsAre(seq_num));
+
+  packet_buffer_.Clear();
+
+  Insert(seq_num + kStartSize, kKeyFrame, kFirst, kNotLast);
+  Insert(seq_num + kStartSize + 1, kDeltaFrame, kNotFirst, kNotLast);
+  EXPECT_THAT(Insert(seq_num + kStartSize + 2, kDeltaFrame, kNotFirst, kLast),
+              StartSeqNumsAre(seq_num + kStartSize));
+}
+
+TEST_F(PacketBufferTest, FramesAfterClear) {
+  Insert(9025, kDeltaFrame, kFirst, kLast);
+  Insert(9024, kKeyFrame, kFirst, kLast);
+  packet_buffer_.ClearTo(9025);
+  EXPECT_THAT(Insert(9057, kDeltaFrame, kFirst, kLast).packets, SizeIs(1));
+  EXPECT_THAT(Insert(9026, kDeltaFrame, kFirst, kLast).packets, SizeIs(1));
+}
+
+TEST_F(PacketBufferTest, SameFrameDifferentTimestamps) {
+  Insert(0, kKeyFrame, kFirst, kNotLast, {}, 1000);
+  EXPECT_THAT(Insert(1, kKeyFrame, kNotFirst, kLast, {}, 1001).packets,
+              IsEmpty());
+}
+
+TEST_F(PacketBufferTest, ContinuousSeqNumDoubleMarkerBit) {
+  Insert(2, kKeyFrame, kNotFirst, kNotLast);
+  Insert(1, kKeyFrame, kFirst, kLast);
+  EXPECT_THAT(Insert(3, kKeyFrame, kNotFirst, kLast).packets, IsEmpty());
+}
+
+TEST_F(PacketBufferTest, IncomingCodecChange) {
+  auto packet = std::make_unique<PacketBuffer::Packet>();
+  packet->video_header.is_first_packet_in_frame = true;
+  packet->video_header.is_last_packet_in_frame = true;
+  packet->video_header.codec = kVideoCodecVP8;
+  packet->video_header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  packet->timestamp = 1;
+  packet->seq_num = 1;
+  packet->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  EXPECT_THAT(packet_buffer_.InsertPacket(std::move(packet)).packets,
+              SizeIs(1));
+
+  packet = std::make_unique<PacketBuffer::Packet>();
+  packet->video_header.is_first_packet_in_frame = true;
+  packet->video_header.is_last_packet_in_frame = true;
+  packet->video_header.codec = kVideoCodecH264;
+  auto& h264_header =
+      packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+  h264_header.nalus_length = 1;
+  packet->timestamp = 3;
+  packet->seq_num = 3;
+  packet->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  EXPECT_THAT(packet_buffer_.InsertPacket(std::move(packet)).packets,
+              IsEmpty());
+
+  packet = std::make_unique<PacketBuffer::Packet>();
+  packet->video_header.is_first_packet_in_frame = true;
+  packet->video_header.is_last_packet_in_frame = true;
+  packet->video_header.codec = kVideoCodecVP8;
+  packet->video_header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  packet->timestamp = 2;
+  packet->seq_num = 2;
+  packet->video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  EXPECT_THAT(packet_buffer_.InsertPacket(std::move(packet)).packets,
+              SizeIs(2));
+}
+
+TEST_F(PacketBufferTest, TooManyNalusInPacket) {
+  auto packet = std::make_unique<PacketBuffer::Packet>();
+  packet->video_header.codec = kVideoCodecH264;
+  packet->timestamp = 1;
+  packet->seq_num = 1;
+  packet->video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  packet->video_header.is_first_packet_in_frame = true;
+  packet->video_header.is_last_packet_in_frame = true;
+  auto& h264_header =
+      packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+  h264_header.nalus_length = kMaxNalusPerPacket;
+  EXPECT_THAT(packet_buffer_.InsertPacket(std::move(packet)).packets,
+              IsEmpty());
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, OneFrameFillBuffer) {
+  InsertH264(0, kKeyFrame, kFirst, kNotLast, 1000);
+  for (int i = 1; i < kStartSize - 1; ++i)
+    InsertH264(i, kKeyFrame, kNotFirst, kNotLast, 1000);
+  EXPECT_THAT(InsertH264(kStartSize - 1, kKeyFrame, kNotFirst, kLast, 1000),
+              StartSeqNumsAre(0));
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, CreateFramesAfterFilledBuffer) {
+  EXPECT_THAT(InsertH264(kStartSize - 2, kKeyFrame, kFirst, kLast, 0).packets,
+              SizeIs(1));
+
+  InsertH264(kStartSize, kDeltaFrame, kFirst, kNotLast, 2000);
+  for (int i = 1; i < kStartSize; ++i)
+    InsertH264(kStartSize + i, kDeltaFrame, kNotFirst, kNotLast, 2000);
+  EXPECT_THAT(
+      InsertH264(kStartSize + kStartSize, kDeltaFrame, kNotFirst, kLast, 2000)
+          .packets,
+      IsEmpty());
+
+  EXPECT_THAT(InsertH264(kStartSize - 1, kKeyFrame, kFirst, kLast, 1000),
+              StartSeqNumsAre(kStartSize - 1, kStartSize));
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, OneFrameMaxSeqNum) {
+  InsertH264(65534, kKeyFrame, kFirst, kNotLast, 1000);
+  EXPECT_THAT(InsertH264(65535, kKeyFrame, kNotFirst, kLast, 1000),
+              StartSeqNumsAre(65534));
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, ClearMissingPacketsOnKeyframe) {
+  InsertH264(0, kKeyFrame, kFirst, kLast, 1000);
+  InsertH264(2, kKeyFrame, kFirst, kLast, 3000);
+  InsertH264(3, kDeltaFrame, kFirst, kNotLast, 4000);
+  InsertH264(4, kDeltaFrame, kNotFirst, kLast, 4000);
+
+  EXPECT_THAT(InsertH264(kStartSize + 1, kKeyFrame, kFirst, kLast, 18000),
+              StartSeqNumsAre(kStartSize + 1));
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, FindFramesOnPadding) {
+  EXPECT_THAT(InsertH264(0, kKeyFrame, kFirst, kLast, 1000),
+              StartSeqNumsAre(0));
+  EXPECT_THAT(InsertH264(2, kDeltaFrame, kFirst, kLast, 1000).packets,
+              IsEmpty());
+
+  EXPECT_THAT(packet_buffer_.InsertPadding(1), StartSeqNumsAre(2));
+}
+
+TEST_P(PacketBufferH264ParameterizedTest, FindFramesOnReorderedPadding) {
+  EXPECT_THAT(InsertH264(0, kKeyFrame, kFirst, kLast, 1001),
+              StartSeqNumsAre(0));
+  EXPECT_THAT(InsertH264(1, kDeltaFrame, kFirst, kNotLast, 1002).packets,
+              IsEmpty());
+  EXPECT_THAT(packet_buffer_.InsertPadding(3).packets, IsEmpty());
+  EXPECT_THAT(InsertH264(4, kDeltaFrame, kFirst, kLast, 1003).packets,
+              IsEmpty());
+  EXPECT_THAT(InsertH264(2, kDeltaFrame, kNotFirst, kLast, 1002),
+              StartSeqNumsAre(1, 4));
+}
+
+class PacketBufferH264XIsKeyframeTest : public PacketBufferH264Test {
+ protected:
+  const uint16_t kSeqNum = 5;
+
+  explicit PacketBufferH264XIsKeyframeTest(bool sps_pps_idr_is_keyframe)
+      : PacketBufferH264Test(sps_pps_idr_is_keyframe) {}
+
+  std::unique_ptr<PacketBuffer::Packet> CreatePacket() {
+    auto packet = std::make_unique<PacketBuffer::Packet>();
+    packet->video_header.codec = kVideoCodecH264;
+    packet->seq_num = kSeqNum;
+
+    packet->video_header.is_first_packet_in_frame = true;
+    packet->video_header.is_last_packet_in_frame = true;
+    return packet;
+  }
+};
+
+class PacketBufferH264IdrIsKeyframeTest
+    : public PacketBufferH264XIsKeyframeTest {
+ protected:
+  PacketBufferH264IdrIsKeyframeTest()
+      : PacketBufferH264XIsKeyframeTest(false) {}
+};
+
+TEST_F(PacketBufferH264IdrIsKeyframeTest, IdrIsKeyframe) {
+  auto packet = CreatePacket();
+  auto& h264_header =
+      packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+  h264_header.nalus[0].type = H264::NaluType::kIdr;
+  h264_header.nalus_length = 1;
+  EXPECT_THAT(packet_buffer_.InsertPacket(std::move(packet)).packets,
+              ElementsAre(KeyFrame()));
+}
+
+TEST_F(PacketBufferH264IdrIsKeyframeTest, SpsPpsIdrIsKeyframe) {
+  auto packet = CreatePacket();
+  auto& h264_header =
+      packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+  h264_header.nalus[0].type = H264::NaluType::kSps;
+  h264_header.nalus[1].type = H264::NaluType::kPps;
+  h264_header.nalus[2].type = H264::NaluType::kIdr;
+  h264_header.nalus_length = 3;
+
+  EXPECT_THAT(packet_buffer_.InsertPacket(std::move(packet)).packets,
+              ElementsAre(KeyFrame()));
+}
+
+class PacketBufferH264SpsPpsIdrIsKeyframeTest
+    : public PacketBufferH264XIsKeyframeTest {
+ protected:
+  PacketBufferH264SpsPpsIdrIsKeyframeTest()
+      : PacketBufferH264XIsKeyframeTest(true) {}
+};
+
+TEST_F(PacketBufferH264SpsPpsIdrIsKeyframeTest, IdrIsNotKeyframe) {
+  auto packet = CreatePacket();
+  auto& h264_header =
+      packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+  h264_header.nalus[0].type = H264::NaluType::kIdr;
+  h264_header.nalus_length = 1;
+
+  EXPECT_THAT(packet_buffer_.InsertPacket(std::move(packet)).packets,
+              ElementsAre(DeltaFrame()));
+}
+
+TEST_F(PacketBufferH264SpsPpsIdrIsKeyframeTest, SpsPpsIsNotKeyframe) {
+  auto packet = CreatePacket();
+  auto& h264_header =
+      packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+  h264_header.nalus[0].type = H264::NaluType::kSps;
+  h264_header.nalus[1].type = H264::NaluType::kPps;
+  h264_header.nalus_length = 2;
+
+  EXPECT_THAT(packet_buffer_.InsertPacket(std::move(packet)).packets,
+              ElementsAre(DeltaFrame()));
+}
+
+TEST_F(PacketBufferH264SpsPpsIdrIsKeyframeTest, SpsPpsIdrIsKeyframe) {
+  auto packet = CreatePacket();
+  auto& h264_header =
+      packet->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+  h264_header.nalus[0].type = H264::NaluType::kSps;
+  h264_header.nalus[1].type = H264::NaluType::kPps;
+  h264_header.nalus[2].type = H264::NaluType::kIdr;
+  h264_header.nalus_length = 3;
+
+  EXPECT_THAT(packet_buffer_.InsertPacket(std::move(packet)).packets,
+              ElementsAre(KeyFrame()));
+}
+
+}  // namespace
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/receiver.cc b/third_party/libwebrtc/modules/video_coding/receiver.cc
new file mode 100644
index 0000000000..3f954ec9bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/receiver.cc
@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/receiver.h"
+
+
+#include <cstdint>
+#include <cstdlib>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "api/video/encoded_image.h"
+#include "modules/video_coding/encoded_frame.h"
+#include "modules/video_coding/internal_defines.h"
+#include "modules/video_coding/jitter_buffer_common.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+enum { kMaxReceiverDelayMs = 10000 };
+
+VCMReceiver::VCMReceiver(VCMTiming* timing,
+                         Clock* clock,
+                         const FieldTrialsView& field_trials)
+    : VCMReceiver::VCMReceiver(timing,
+                               clock,
+                               absl::WrapUnique(EventWrapper::Create()),
+                               absl::WrapUnique(EventWrapper::Create()),
+                               field_trials) {}
+
+VCMReceiver::VCMReceiver(VCMTiming* timing,
+                         Clock* clock,
+                         std::unique_ptr<EventWrapper> receiver_event,
+                         std::unique_ptr<EventWrapper> jitter_buffer_event,
+                         const FieldTrialsView& field_trials)
+    : clock_(clock),
+      jitter_buffer_(clock_, std::move(jitter_buffer_event), field_trials),
+      timing_(timing),
+      render_wait_event_(std::move(receiver_event)),
+      max_video_delay_ms_(kMaxVideoDelayMs) {
+  jitter_buffer_.Start();
+}
+
+VCMReceiver::~VCMReceiver() {
+  render_wait_event_->Set();
+}
+
+int32_t VCMReceiver::InsertPacket(const VCMPacket& packet) {
+  // Insert the packet into the jitter buffer. The packet can either be empty or
+  // contain media at this point.
+  bool retransmitted = false;
+  const VCMFrameBufferEnum ret =
+      jitter_buffer_.InsertPacket(packet, &retransmitted);
+  if (ret == kOldPacket) {
+    return VCM_OK;
+  } else if (ret == kFlushIndicator) {
+    return VCM_FLUSH_INDICATOR;
+  } else if (ret < 0) {
+    return VCM_JITTER_BUFFER_ERROR;
+  }
+  if (ret == kCompleteSession && !retransmitted) {
+    // We don't want to include timestamps which have suffered from
+    // retransmission here, since we compensate with extra retransmission
+    // delay within the jitter estimate.
+    timing_->IncomingTimestamp(packet.timestamp, clock_->CurrentTime());
+  }
+  return VCM_OK;
+}
+
+VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
+                                               bool prefer_late_decoding) {
+  const int64_t start_time_ms = clock_->TimeInMilliseconds();
+  uint32_t frame_timestamp = 0;
+  int min_playout_delay_ms = -1;
+  int max_playout_delay_ms = -1;
+  int64_t render_time_ms = 0;
+  // Exhaust wait time to get a complete frame for decoding.
+  VCMEncodedFrame* found_frame =
+      jitter_buffer_.NextCompleteFrame(max_wait_time_ms);
+
+  if (found_frame) {
+    frame_timestamp = found_frame->Timestamp();
+    min_playout_delay_ms = found_frame->EncodedImage().playout_delay_.min_ms;
+    max_playout_delay_ms = found_frame->EncodedImage().playout_delay_.max_ms;
+  } else {
+    return nullptr;
+  }
+
+  if (min_playout_delay_ms >= 0)
+    timing_->set_min_playout_delay(TimeDelta::Millis(min_playout_delay_ms));
+
+  if (max_playout_delay_ms >= 0)
+    timing_->set_max_playout_delay(TimeDelta::Millis(max_playout_delay_ms));
+
+  // We have a frame - Set timing and render timestamp.
+  timing_->SetJitterDelay(
+      TimeDelta::Millis(jitter_buffer_.EstimatedJitterMs()));
+  const Timestamp now = clock_->CurrentTime();
+  const int64_t now_ms = now.ms();
+  timing_->UpdateCurrentDelay(frame_timestamp);
+  render_time_ms = timing_->RenderTime(frame_timestamp, now).ms();
+  // Check render timing.
+  bool timing_error = false;
+  // Assume that render timing errors are due to changes in the video stream.
+  if (render_time_ms < 0) {
+    timing_error = true;
+  } else if (std::abs(render_time_ms - now_ms) > max_video_delay_ms_) {
+    int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
+    RTC_LOG(LS_WARNING)
+        << "A frame about to be decoded is out of the configured "
+           "delay bounds ("
+        << frame_delay << " > " << max_video_delay_ms_
+        << "). Resetting the video jitter buffer.";
+    timing_error = true;
+  } else if (static_cast<int>(timing_->TargetVideoDelay().ms()) >
+             max_video_delay_ms_) {
+    RTC_LOG(LS_WARNING) << "The video target delay has grown larger than "
+                        << max_video_delay_ms_
+                        << " ms. Resetting jitter buffer.";
+    timing_error = true;
+  }
+
+  if (timing_error) {
+    // Timing error => reset timing and flush the jitter buffer.
+    jitter_buffer_.Flush();
+    timing_->Reset();
+    return NULL;
+  }
+
+  if (prefer_late_decoding) {
+    // Decode frame as close as possible to the render timestamp.
+    const int32_t available_wait_time =
+        max_wait_time_ms -
+        static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
+    uint16_t new_max_wait_time =
+        static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
+    uint32_t wait_time_ms = rtc::saturated_cast<uint32_t>(
+        timing_
+            ->MaxWaitingTime(Timestamp::Millis(render_time_ms),
+                             clock_->CurrentTime(),
+                             /*too_many_frames_queued=*/false)
+            .ms());
+    if (new_max_wait_time < wait_time_ms) {
+      // We're not allowed to wait until the frame is supposed to be rendered,
+      // waiting as long as we're allowed to avoid busy looping, and then return
+      // NULL. Next call to this function might return the frame.
+      render_wait_event_->Wait(new_max_wait_time);
+      return NULL;
+    }
+    // Wait until it's time to render.
+    render_wait_event_->Wait(wait_time_ms);
+  }
+
+  // Extract the frame from the jitter buffer and set the render time.
+  VCMEncodedFrame* frame = jitter_buffer_.ExtractAndSetDecode(frame_timestamp);
+  if (frame == NULL) {
+    return NULL;
+  }
+  frame->SetRenderTime(render_time_ms);
+  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", frame->Timestamp(), "SetRenderTS",
+                          "render_time", frame->RenderTimeMs());
+  return frame;
+}
+
+void VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame) {
+  jitter_buffer_.ReleaseFrame(frame);
+}
+
+void VCMReceiver::SetNackSettings(size_t max_nack_list_size,
+                                  int max_packet_age_to_nack,
+                                  int max_incomplete_time_ms) {
+  jitter_buffer_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
+                                 max_incomplete_time_ms);
+}
+
+std::vector<uint16_t> VCMReceiver::NackList(bool* request_key_frame) {
+  return jitter_buffer_.GetNackList(request_key_frame);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/receiver.h b/third_party/libwebrtc/modules/video_coding/receiver.h
new file mode 100644
index 0000000000..069f8c55c7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/receiver.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_RECEIVER_H_
+#define MODULES_VIDEO_CODING_RECEIVER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/field_trials_view.h"
+#include "modules/video_coding/event_wrapper.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "modules/video_coding/jitter_buffer.h"
+#include "modules/video_coding/packet.h"
+#include "modules/video_coding/timing/timing.h"
+
+namespace webrtc {
+
+class Clock;
+class VCMEncodedFrame;
+
+class VCMReceiver {
+ public:
+  VCMReceiver(VCMTiming* timing,
+              Clock* clock,
+              const FieldTrialsView& field_trials);
+
+  // Using this constructor, you can specify a different event implemetation for
+  // the jitter buffer. Useful for unit tests when you want to simulate incoming
+  // packets, in which case the jitter buffer's wait event is different from
+  // that of VCMReceiver itself.
+  VCMReceiver(VCMTiming* timing,
+              Clock* clock,
+              std::unique_ptr<EventWrapper> receiver_event,
+              std::unique_ptr<EventWrapper> jitter_buffer_event,
+              const FieldTrialsView& field_trials);
+
+  ~VCMReceiver();
+
+  int32_t InsertPacket(const VCMPacket& packet);
+  VCMEncodedFrame* FrameForDecoding(uint16_t max_wait_time_ms,
+                                    bool prefer_late_decoding);
+  void ReleaseFrame(VCMEncodedFrame* frame);
+
+  // NACK.
+  void SetNackSettings(size_t max_nack_list_size,
+                       int max_packet_age_to_nack,
+                       int max_incomplete_time_ms);
+  std::vector<uint16_t> NackList(bool* request_key_frame);
+
+ private:
+  Clock* const clock_;
+  VCMJitterBuffer jitter_buffer_;
+  VCMTiming* timing_;
+  std::unique_ptr<EventWrapper> render_wait_event_;
+  int max_video_delay_ms_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_RECEIVER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/receiver_unittest.cc b/third_party/libwebrtc/modules/video_coding/receiver_unittest.cc
new file mode 100644
index 0000000000..2beb97e972
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/receiver_unittest.cc
@@ -0,0 +1,493 @@
+/*  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/receiver.h"
+
+#include <string.h>
+
+#include <cstdint>
+#include <memory>
+#include <queue>
+#include <vector>
+
+#include "modules/video_coding/encoded_frame.h"
+#include "modules/video_coding/jitter_buffer_common.h"
+#include "modules/video_coding/packet.h"
+#include "modules/video_coding/test/stream_generator.h"
+#include "modules/video_coding/timing/timing.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+
+class TestVCMReceiver : public ::testing::Test {
+ protected:
+  TestVCMReceiver()
+      : clock_(0),
+        timing_(&clock_, field_trials_),
+        receiver_(&timing_, &clock_, field_trials_),
+        stream_generator_(0, clock_.TimeInMilliseconds()) {}
+
+  int32_t InsertPacket(int index) {
+    VCMPacket packet;
+    bool packet_available = stream_generator_.GetPacket(&packet, index);
+    EXPECT_TRUE(packet_available);
+    if (!packet_available)
+      return kGeneralError;  // Return here to avoid crashes below.
+    return receiver_.InsertPacket(packet);
+  }
+
+  int32_t InsertPacketAndPop(int index) {
+    VCMPacket packet;
+    bool packet_available = stream_generator_.PopPacket(&packet, index);
+    EXPECT_TRUE(packet_available);
+    if (!packet_available)
+      return kGeneralError;  // Return here to avoid crashes below.
+    return receiver_.InsertPacket(packet);
+  }
+
+  int32_t InsertFrame(VideoFrameType frame_type, bool complete) {
+    int num_of_packets = complete ? 1 : 2;
+    stream_generator_.GenerateFrame(
+        frame_type,
+        (frame_type != VideoFrameType::kEmptyFrame) ? num_of_packets : 0,
+        (frame_type == VideoFrameType::kEmptyFrame) ? 1 : 0,
+        clock_.TimeInMilliseconds());
+    int32_t ret = InsertPacketAndPop(0);
+    if (!complete) {
+      // Drop the second packet.
+      VCMPacket packet;
+      stream_generator_.PopPacket(&packet, 0);
+    }
+    clock_.AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
+    return ret;
+  }
+
+  bool DecodeNextFrame() {
+    VCMEncodedFrame* frame = receiver_.FrameForDecoding(0, false);
+    if (!frame)
+      return false;
+    receiver_.ReleaseFrame(frame);
+    return true;
+  }
+
+  test::ScopedKeyValueConfig field_trials_;
+  SimulatedClock clock_;
+  VCMTiming timing_;
+  VCMReceiver receiver_;
+  StreamGenerator stream_generator_;
+};
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_Empty) {
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey, true), kNoError);
+  // Advance time until it's time to decode the key frame.
+  clock_.AdvanceTimeMilliseconds(kMinDelayMs);
+  EXPECT_TRUE(DecodeNextFrame());
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_FALSE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_NoKeyFrame) {
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  const int kNumFrames = kDefaultFrameRate * kMaxNonDecodableDuration / 1000;
+  for (int i = 0; i < kNumFrames; ++i) {
+    EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta, true), kNoError);
+  }
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_TRUE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_OneIncomplete) {
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  timing_.set_min_playout_delay(TimeDelta::Millis(kMinDelayMs));
+  int64_t key_frame_inserted = clock_.TimeInMilliseconds();
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey, true), kNoError);
+  // Insert an incomplete frame.
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta, false), kNoError);
+  // Insert enough frames to have too long non-decodable sequence.
+  for (int i = 0; i < kMaxNonDecodableDurationFrames; ++i) {
+    EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta, true), kNoError);
+  }
+  // Advance time until it's time to decode the key frame.
+  clock_.AdvanceTimeMilliseconds(kMinDelayMs - clock_.TimeInMilliseconds() -
+                                 key_frame_inserted);
+  EXPECT_TRUE(DecodeNextFrame());
+  // Make sure we get a key frame request.
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_TRUE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger) {
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  timing_.set_min_playout_delay(TimeDelta::Millis(kMinDelayMs));
+  int64_t key_frame_inserted = clock_.TimeInMilliseconds();
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey, true), kNoError);
+  // Insert an incomplete frame.
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta, false), kNoError);
+  // Insert all but one frame to not trigger a key frame request due to
+  // too long duration of non-decodable frames.
+  for (int i = 0; i < kMaxNonDecodableDurationFrames - 1; ++i) {
+    EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta, true), kNoError);
+  }
+  // Advance time until it's time to decode the key frame.
+  clock_.AdvanceTimeMilliseconds(kMinDelayMs - clock_.TimeInMilliseconds() -
+                                 key_frame_inserted);
+  EXPECT_TRUE(DecodeNextFrame());
+  // Make sure we don't get a key frame request since we haven't generated
+  // enough frames.
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_FALSE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger2) {
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  timing_.set_min_playout_delay(TimeDelta::Millis(kMinDelayMs));
+  int64_t key_frame_inserted = clock_.TimeInMilliseconds();
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey, true), kNoError);
+  // Insert enough frames to have too long non-decodable sequence, except that
+  // we don't have any losses.
+  for (int i = 0; i < kMaxNonDecodableDurationFrames; ++i) {
+    EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta, true), kNoError);
+  }
+  // Insert an incomplete frame.
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta, false), kNoError);
+  // Advance time until it's time to decode the key frame.
+  clock_.AdvanceTimeMilliseconds(kMinDelayMs - clock_.TimeInMilliseconds() -
+                                 key_frame_inserted);
+  EXPECT_TRUE(DecodeNextFrame());
+  // Make sure we don't get a key frame request since the non-decodable duration
+  // is only one frame.
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_FALSE(request_key_frame);
+}
+
+TEST_F(TestVCMReceiver, NonDecodableDuration_KeyFrameAfterIncompleteFrames) {
+  const size_t kMaxNackListSize = 1000;
+  const int kMaxPacketAgeToNack = 1000;
+  const int kMaxNonDecodableDuration = 500;
+  const int kMaxNonDecodableDurationFrames =
+      (kDefaultFrameRate * kMaxNonDecodableDuration + 500) / 1000;
+  const int kMinDelayMs = 500;
+  receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
+                            kMaxNonDecodableDuration);
+  timing_.set_min_playout_delay(TimeDelta::Millis(kMinDelayMs));
+  int64_t key_frame_inserted = clock_.TimeInMilliseconds();
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey, true), kNoError);
+  // Insert an incomplete frame.
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta, false), kNoError);
+  // Insert enough frames to have too long non-decodable sequence.
+  for (int i = 0; i < kMaxNonDecodableDurationFrames; ++i) {
+    EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameDelta, true), kNoError);
+  }
+  EXPECT_GE(InsertFrame(VideoFrameType::kVideoFrameKey, true), kNoError);
+  // Advance time until it's time to decode the key frame.
+  clock_.AdvanceTimeMilliseconds(kMinDelayMs - clock_.TimeInMilliseconds() -
+                                 key_frame_inserted);
+  EXPECT_TRUE(DecodeNextFrame());
+  // Make sure we don't get a key frame request since we have a key frame
+  // in the list.
+  bool request_key_frame = false;
+  std::vector<uint16_t> nack_list = receiver_.NackList(&request_key_frame);
+  EXPECT_FALSE(request_key_frame);
+}
+
+// A simulated clock, when time elapses, will insert frames into the jitter
+// buffer, based on initial settings.
+class SimulatedClockWithFrames : public SimulatedClock {
+ public:
+  SimulatedClockWithFrames(StreamGenerator* stream_generator,
+                           VCMReceiver* receiver)
+      : SimulatedClock(0),
+        stream_generator_(stream_generator),
+        receiver_(receiver) {}
+  virtual ~SimulatedClockWithFrames() {}
+
+  // If `stop_on_frame` is true and next frame arrives between now and
+  // now+`milliseconds`, the clock will be advanced to the arrival time of next
+  // frame.
+  // Otherwise, the clock will be advanced by `milliseconds`.
+  //
+  // For both cases, a frame will be inserted into the jitter buffer at the
+  // instant when the clock time is timestamps_.front().arrive_time.
+  //
+  // Return true if some frame arrives between now and now+`milliseconds`.
+  bool AdvanceTimeMilliseconds(int64_t milliseconds, bool stop_on_frame) {
+    return AdvanceTimeMicroseconds(milliseconds * 1000, stop_on_frame);
+  }
+
+  bool AdvanceTimeMicroseconds(int64_t microseconds, bool stop_on_frame) {
+    int64_t start_time = TimeInMicroseconds();
+    int64_t end_time = start_time + microseconds;
+    bool frame_injected = false;
+    while (!timestamps_.empty() &&
+           timestamps_.front().arrive_time <= end_time) {
+      RTC_DCHECK_GE(timestamps_.front().arrive_time, start_time);
+
+      SimulatedClock::AdvanceTimeMicroseconds(timestamps_.front().arrive_time -
+                                              TimeInMicroseconds());
+      GenerateAndInsertFrame((timestamps_.front().render_time + 500) / 1000);
+      timestamps_.pop();
+      frame_injected = true;
+
+      if (stop_on_frame)
+        return frame_injected;
+    }
+
+    if (TimeInMicroseconds() < end_time) {
+      SimulatedClock::AdvanceTimeMicroseconds(end_time - TimeInMicroseconds());
+    }
+    return frame_injected;
+  }
+
+  // Input timestamps are in unit Milliseconds.
+  // And `arrive_timestamps` must be positive and in increasing order.
+  // `arrive_timestamps` determine when we are going to insert frames into the
+  // jitter buffer.
+  // `render_timestamps` are the timestamps on the frame.
+  void SetFrames(const int64_t* arrive_timestamps,
+                 const int64_t* render_timestamps,
+                 size_t size) {
+    int64_t previous_arrive_timestamp = 0;
+    for (size_t i = 0; i < size; i++) {
+      RTC_CHECK_GE(arrive_timestamps[i], previous_arrive_timestamp);
+      timestamps_.push(TimestampPair(arrive_timestamps[i] * 1000,
+                                     render_timestamps[i] * 1000));
+      previous_arrive_timestamp = arrive_timestamps[i];
+    }
+  }
+
+ private:
+  struct TimestampPair {
+    TimestampPair(int64_t arrive_timestamp, int64_t render_timestamp)
+        : arrive_time(arrive_timestamp), render_time(render_timestamp) {}
+
+    int64_t arrive_time;
+    int64_t render_time;
+  };
+
+  void GenerateAndInsertFrame(int64_t render_timestamp_ms) {
+    VCMPacket packet;
+    stream_generator_->GenerateFrame(VideoFrameType::kVideoFrameKey,
+                                     1,  // media packets
+                                     0,  // empty packets
+                                     render_timestamp_ms);
+
+    bool packet_available = stream_generator_->PopPacket(&packet, 0);
+    EXPECT_TRUE(packet_available);
+    if (!packet_available)
+      return;  // Return here to avoid crashes below.
+    receiver_->InsertPacket(packet);
+  }
+
+  std::queue<TimestampPair> timestamps_;
+  StreamGenerator* stream_generator_;
+  VCMReceiver* receiver_;
+};
+
+// Use a SimulatedClockWithFrames
+// Wait call will do either of these:
+// 1. If `stop_on_frame` is true, the clock will be turned to the exact instant
+// that the first frame comes and the frame will be inserted into the jitter
+// buffer, or the clock will be turned to now + `max_time` if no frame comes in
+// the window.
+// 2. If `stop_on_frame` is false, the clock will be turn to now + `max_time`,
+// and all the frames arriving between now and now + `max_time` will be
+// inserted into the jitter buffer.
+//
+// This is used to simulate the JitterBuffer getting packets from internet as
+// time elapses.
+
+class FrameInjectEvent : public EventWrapper {
+ public:
+  FrameInjectEvent(SimulatedClockWithFrames* clock, bool stop_on_frame)
+      : clock_(clock), stop_on_frame_(stop_on_frame) {}
+
+  bool Set() override { return true; }
+
+  EventTypeWrapper Wait(int max_time_ms) override {
+    if (clock_->AdvanceTimeMilliseconds(max_time_ms, stop_on_frame_) &&
+        stop_on_frame_) {
+      return EventTypeWrapper::kEventSignaled;
+    } else {
+      return EventTypeWrapper::kEventTimeout;
+    }
+  }
+
+ private:
+  SimulatedClockWithFrames* clock_;
+  bool stop_on_frame_;
+};
+
+class VCMReceiverTimingTest : public ::testing::Test {
+ protected:
+  VCMReceiverTimingTest()
+      : clock_(&stream_generator_, &receiver_),
+        stream_generator_(0, clock_.TimeInMilliseconds()),
+        timing_(&clock_, field_trials_),
+        receiver_(
+            &timing_,
+            &clock_,
+            std::unique_ptr<EventWrapper>(new FrameInjectEvent(&clock_, false)),
+            std::unique_ptr<EventWrapper>(new FrameInjectEvent(&clock_, true)),
+            field_trials_) {}
+
+  virtual void SetUp() {}
+
+  test::ScopedKeyValueConfig field_trials_;
+  SimulatedClockWithFrames clock_;
+  StreamGenerator stream_generator_;
+  VCMTiming timing_;
+  VCMReceiver receiver_;
+};
+
+// Test whether VCMReceiver::FrameForDecoding handles parameter
+// `max_wait_time_ms` correctly:
+// 1. The function execution should never take more than `max_wait_time_ms`.
+// 2. If the function exit before now + `max_wait_time_ms`, a frame must be
+//    returned.
+TEST_F(VCMReceiverTimingTest, FrameForDecoding) {
+  const size_t kNumFrames = 100;
+  const int kFramePeriod = 40;
+  int64_t arrive_timestamps[kNumFrames];
+  int64_t render_timestamps[kNumFrames];
+
+  // Construct test samples.
+  // render_timestamps are the timestamps stored in the Frame;
+  // arrive_timestamps controls when the Frame packet got received.
+  for (size_t i = 0; i < kNumFrames; i++) {
+    // Preset frame rate to 25Hz.
+    // But we add a reasonable deviation to arrive_timestamps to mimic Internet
+    // fluctuation.
+    arrive_timestamps[i] =
+        (i + 1) * kFramePeriod + (i % 10) * ((i % 2) ? 1 : -1);
+    render_timestamps[i] = (i + 1) * kFramePeriod;
+  }
+
+  clock_.SetFrames(arrive_timestamps, render_timestamps, kNumFrames);
+
+  // Record how many frames we finally get out of the receiver.
+  size_t num_frames_return = 0;
+
+  const int64_t kMaxWaitTime = 30;
+
+  // Ideally, we should get all frames that we input in InitializeFrames.
+  // In the case that FrameForDecoding kills frames by error, we rely on the
+  // build bot to kill the test.
+  while (num_frames_return < kNumFrames) {
+    int64_t start_time = clock_.TimeInMilliseconds();
+    VCMEncodedFrame* frame = receiver_.FrameForDecoding(kMaxWaitTime, false);
+    int64_t end_time = clock_.TimeInMilliseconds();
+
+    // In any case the FrameForDecoding should not wait longer than
+    // max_wait_time.
+    // In the case that we did not get a frame, it should have been waiting for
+    // exactly max_wait_time. (By the testing samples we constructed above, we
+    // are sure there is no timing error, so the only case it returns with NULL
+    // is that it runs out of time.)
+    if (frame) {
+      receiver_.ReleaseFrame(frame);
+      ++num_frames_return;
+      EXPECT_GE(kMaxWaitTime, end_time - start_time);
+    } else {
+      EXPECT_EQ(kMaxWaitTime, end_time - start_time);
+    }
+  }
+}
+
+// Test whether VCMReceiver::FrameForDecoding handles parameter
+// `prefer_late_decoding` and `max_wait_time_ms` correctly:
+// 1. The function execution should never take more than `max_wait_time_ms`.
+// 2. If the function exit before now + `max_wait_time_ms`, a frame must be
+//    returned and the end time must be equal to the render timestamp - delay
+//    for decoding and rendering.
+TEST_F(VCMReceiverTimingTest, FrameForDecodingPreferLateDecoding) {
+  const size_t kNumFrames = 100;
+  const int kFramePeriod = 40;
+
+  int64_t arrive_timestamps[kNumFrames];
+  int64_t render_timestamps[kNumFrames];
+
+  auto timings = timing_.GetTimings();
+  TimeDelta render_delay = timings.render_delay;
+  TimeDelta max_decode = timings.max_decode_duration;
+
+  // Construct test samples.
+  // render_timestamps are the timestamps stored in the Frame;
+  // arrive_timestamps controls when the Frame packet got received.
+  for (size_t i = 0; i < kNumFrames; i++) {
+    // Preset frame rate to 25Hz.
+    // But we add a reasonable deviation to arrive_timestamps to mimic Internet
+    // fluctuation.
+    arrive_timestamps[i] =
+        (i + 1) * kFramePeriod + (i % 10) * ((i % 2) ? 1 : -1);
+    render_timestamps[i] = (i + 1) * kFramePeriod;
+  }
+
+  clock_.SetFrames(arrive_timestamps, render_timestamps, kNumFrames);
+
+  // Record how many frames we finally get out of the receiver.
+  size_t num_frames_return = 0;
+  const int64_t kMaxWaitTime = 30;
+  bool prefer_late_decoding = true;
+  while (num_frames_return < kNumFrames) {
+    int64_t start_time = clock_.TimeInMilliseconds();
+
+    VCMEncodedFrame* frame =
+        receiver_.FrameForDecoding(kMaxWaitTime, prefer_late_decoding);
+    int64_t end_time = clock_.TimeInMilliseconds();
+    if (frame) {
+      EXPECT_EQ(frame->RenderTimeMs() - max_decode.ms() - render_delay.ms(),
+                end_time);
+      receiver_.ReleaseFrame(frame);
+      ++num_frames_return;
+    } else {
+      EXPECT_EQ(kMaxWaitTime, end_time - start_time);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_frame_id_only_ref_finder.cc b/third_party/libwebrtc/modules/video_coding/rtp_frame_id_only_ref_finder.cc
new file mode 100644
index 0000000000..9f3d5bb296
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_frame_id_only_ref_finder.cc
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/rtp_frame_id_only_ref_finder.h"
+
+#include <utility>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+RtpFrameReferenceFinder::ReturnVector RtpFrameIdOnlyRefFinder::ManageFrame(
+    std::unique_ptr<RtpFrameObject> frame,
+    int frame_id) {
+  frame->SetSpatialIndex(0);
+  frame->SetId(unwrapper_.Unwrap(frame_id & (kFrameIdLength - 1)));
+  frame->num_references =
+      frame->frame_type() == VideoFrameType::kVideoFrameKey ? 0 : 1;
+  frame->references[0] = frame->Id() - 1;
+
+  RtpFrameReferenceFinder::ReturnVector res;
+  res.push_back(std::move(frame));
+  return res;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_frame_id_only_ref_finder.h b/third_party/libwebrtc/modules/video_coding/rtp_frame_id_only_ref_finder.h
new file mode 100644
index 0000000000..5c0bdc8b26
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_frame_id_only_ref_finder.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_RTP_FRAME_ID_ONLY_REF_FINDER_H_
+#define MODULES_VIDEO_CODING_RTP_FRAME_ID_ONLY_REF_FINDER_H_
+
+#include <memory>
+
+#include "absl/container/inlined_vector.h"
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+class RtpFrameIdOnlyRefFinder {
+ public:
+  RtpFrameIdOnlyRefFinder() = default;
+
+  RtpFrameReferenceFinder::ReturnVector ManageFrame(
+      std::unique_ptr<RtpFrameObject> frame,
+      int frame_id);
+
+ private:
+  static constexpr int kFrameIdLength = 1 << 15;
+  SeqNumUnwrapper<uint16_t, kFrameIdLength> unwrapper_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_RTP_FRAME_ID_ONLY_REF_FINDER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder.cc b/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder.cc
new file mode 100644
index 0000000000..a44b76bf15
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder.cc
@@ -0,0 +1,189 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+
+#include <utility>
+
+#include "absl/types/variant.h"
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/rtp_frame_id_only_ref_finder.h"
+#include "modules/video_coding/rtp_generic_ref_finder.h"
+#include "modules/video_coding/rtp_seq_num_only_ref_finder.h"
+#include "modules/video_coding/rtp_vp8_ref_finder.h"
+#include "modules/video_coding/rtp_vp9_ref_finder.h"
+
+namespace webrtc {
+namespace internal {
+class RtpFrameReferenceFinderImpl {
+ public:
+  RtpFrameReferenceFinderImpl() = default;
+
+  RtpFrameReferenceFinder::ReturnVector ManageFrame(
+      std::unique_ptr<RtpFrameObject> frame);
+  RtpFrameReferenceFinder::ReturnVector PaddingReceived(uint16_t seq_num);
+  void ClearTo(uint16_t seq_num);
+
+ private:
+  using RefFinder = absl::variant<absl::monostate,
+                                  RtpGenericFrameRefFinder,
+                                  RtpFrameIdOnlyRefFinder,
+                                  RtpSeqNumOnlyRefFinder,
+                                  RtpVp8RefFinder,
+                                  RtpVp9RefFinder>;
+
+  template <typename T>
+  T& GetRefFinderAs();
+  RefFinder ref_finder_;
+};
+
+RtpFrameReferenceFinder::ReturnVector RtpFrameReferenceFinderImpl::ManageFrame(
+    std::unique_ptr<RtpFrameObject> frame) {
+  const RTPVideoHeader& video_header = frame->GetRtpVideoHeader();
+
+  if (video_header.generic.has_value()) {
+    return GetRefFinderAs<RtpGenericFrameRefFinder>().ManageFrame(
+        std::move(frame), *video_header.generic);
+  }
+
+  switch (frame->codec_type()) {
+    case kVideoCodecVP8: {
+      const RTPVideoHeaderVP8& vp8_header =
+          absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
+
+      if (vp8_header.temporalIdx == kNoTemporalIdx ||
+          vp8_header.tl0PicIdx == kNoTl0PicIdx) {
+        if (vp8_header.pictureId == kNoPictureId) {
+          return GetRefFinderAs<RtpSeqNumOnlyRefFinder>().ManageFrame(
+              std::move(frame));
+        }
+
+        return GetRefFinderAs<RtpFrameIdOnlyRefFinder>().ManageFrame(
+            std::move(frame), vp8_header.pictureId);
+      }
+
+      return GetRefFinderAs<RtpVp8RefFinder>().ManageFrame(std::move(frame));
+    }
+    case kVideoCodecVP9: {
+      const RTPVideoHeaderVP9& vp9_header =
+          absl::get<RTPVideoHeaderVP9>(video_header.video_type_header);
+
+      if (vp9_header.temporal_idx == kNoTemporalIdx) {
+        if (vp9_header.picture_id == kNoPictureId) {
+          return GetRefFinderAs<RtpSeqNumOnlyRefFinder>().ManageFrame(
+              std::move(frame));
+        }
+
+        return GetRefFinderAs<RtpFrameIdOnlyRefFinder>().ManageFrame(
+            std::move(frame), vp9_header.picture_id);
+      }
+
+      return GetRefFinderAs<RtpVp9RefFinder>().ManageFrame(std::move(frame));
+    }
+    case kVideoCodecGeneric: {
+      if (auto* generic_header = absl::get_if<RTPVideoHeaderLegacyGeneric>(
+              &video_header.video_type_header)) {
+        return GetRefFinderAs<RtpFrameIdOnlyRefFinder>().ManageFrame(
+            std::move(frame), generic_header->picture_id);
+      }
+
+      return GetRefFinderAs<RtpSeqNumOnlyRefFinder>().ManageFrame(
+          std::move(frame));
+    }
+    default: {
+      return GetRefFinderAs<RtpSeqNumOnlyRefFinder>().ManageFrame(
+          std::move(frame));
+    }
+  }
+}
+
+RtpFrameReferenceFinder::ReturnVector
+RtpFrameReferenceFinderImpl::PaddingReceived(uint16_t seq_num) {
+  if (auto* ref_finder = absl::get_if<RtpSeqNumOnlyRefFinder>(&ref_finder_)) {
+    return ref_finder->PaddingReceived(seq_num);
+  }
+  return {};
+}
+
+void RtpFrameReferenceFinderImpl::ClearTo(uint16_t seq_num) {
+  struct ClearToVisitor {
+    void operator()(absl::monostate& ref_finder) {}
+    void operator()(RtpGenericFrameRefFinder& ref_finder) {}
+    void operator()(RtpFrameIdOnlyRefFinder& ref_finder) {}
+    void operator()(RtpSeqNumOnlyRefFinder& ref_finder) {
+      ref_finder.ClearTo(seq_num);
+    }
+    void operator()(RtpVp8RefFinder& ref_finder) {
+      ref_finder.ClearTo(seq_num);
+    }
+    void operator()(RtpVp9RefFinder& ref_finder) {
+      ref_finder.ClearTo(seq_num);
+    }
+    uint16_t seq_num;
+  };
+
+  absl::visit(ClearToVisitor{seq_num}, ref_finder_);
+}
+
+template <typename T>
+T& RtpFrameReferenceFinderImpl::GetRefFinderAs() {
+  if (auto* ref_finder = absl::get_if<T>(&ref_finder_)) {
+    return *ref_finder;
+  }
+  return ref_finder_.emplace<T>();
+}
+
+}  // namespace internal
+
+RtpFrameReferenceFinder::RtpFrameReferenceFinder()
+    : RtpFrameReferenceFinder(0) {}
+
+RtpFrameReferenceFinder::RtpFrameReferenceFinder(
+    int64_t picture_id_offset)
+    : picture_id_offset_(picture_id_offset),
+      impl_(std::make_unique<internal::RtpFrameReferenceFinderImpl>()) {}
+
+RtpFrameReferenceFinder::~RtpFrameReferenceFinder() = default;
+
+RtpFrameReferenceFinder::ReturnVector RtpFrameReferenceFinder::ManageFrame(
+    std::unique_ptr<RtpFrameObject> frame) {
+  // If we have cleared past this frame, drop it.
+  if (cleared_to_seq_num_ != -1 &&
+      AheadOf<uint16_t>(cleared_to_seq_num_, frame->first_seq_num())) {
+    return {};
+  }
+
+  auto frames = impl_->ManageFrame(std::move(frame));
+  AddPictureIdOffset(frames);
+  return frames;
+}
+
+RtpFrameReferenceFinder::ReturnVector RtpFrameReferenceFinder::PaddingReceived(
+    uint16_t seq_num) {
+  auto frames = impl_->PaddingReceived(seq_num);
+  AddPictureIdOffset(frames);
+  return frames;
+}
+
+void RtpFrameReferenceFinder::ClearTo(uint16_t seq_num) {
+  cleared_to_seq_num_ = seq_num;
+  impl_->ClearTo(seq_num);
+}
+
+void RtpFrameReferenceFinder::AddPictureIdOffset(ReturnVector& frames) {
+  for (auto& frame : frames) {
+    frame->SetId(frame->Id() + picture_id_offset_);
+    for (size_t i = 0; i < frame->num_references; ++i) {
+      frame->references[i] += picture_id_offset_;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder.h b/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder.h
new file mode 100644
index 0000000000..9ce63cd8a4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_RTP_FRAME_REFERENCE_FINDER_H_
+#define MODULES_VIDEO_CODING_RTP_FRAME_REFERENCE_FINDER_H_
+
+#include <memory>
+
+#include "modules/video_coding/frame_object.h"
+
+namespace webrtc {
+namespace internal {
+class RtpFrameReferenceFinderImpl;
+}  // namespace internal
+
+class RtpFrameReferenceFinder {
+ public:
+  using ReturnVector = absl::InlinedVector<std::unique_ptr<RtpFrameObject>, 3>;
+
+  RtpFrameReferenceFinder();
+  explicit RtpFrameReferenceFinder(int64_t picture_id_offset);
+  ~RtpFrameReferenceFinder();
+
+  // The RtpFrameReferenceFinder will hold onto the frame until:
+  //  - the required information to determine its references has been received,
+  //    in which case it (and possibly other) frames are returned, or
+  //  - There are too many stashed frames (determined by `kMaxStashedFrames`),
+  //    in which case it gets dropped, or
+  //  - It gets cleared by ClearTo, in which case its dropped.
+  //  - The frame is old, in which case it also gets dropped.
+  ReturnVector ManageFrame(std::unique_ptr<RtpFrameObject> frame);
+
+  // Notifies that padding has been received, which the reference finder
+  // might need to calculate the references of a frame.
+  ReturnVector PaddingReceived(uint16_t seq_num);
+
+  // Clear all stashed frames that include packets older than `seq_num`.
+  void ClearTo(uint16_t seq_num);
+
+ private:
+  void AddPictureIdOffset(ReturnVector& frames);
+
+  // How far frames have been cleared out of the buffer by RTP sequence number.
+  // A frame will be cleared if it contains a packet with a sequence number
+  // older than `cleared_to_seq_num_`.
+  int cleared_to_seq_num_ = -1;
+  const int64_t picture_id_offset_;
+  std::unique_ptr<internal::RtpFrameReferenceFinderImpl> impl_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_RTP_FRAME_REFERENCE_FINDER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder_unittest.cc b/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder_unittest.cc
new file mode 100644
index 0000000000..c58f1a987d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder_unittest.cc
@@ -0,0 +1,322 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstring>
+#include <limits>
+#include <map>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/packet_buffer.h"
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+#include "rtc_base/random.h"
+#include "rtc_base/ref_count.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+std::unique_ptr<RtpFrameObject> CreateFrame(
+    uint16_t seq_num_start,
+    uint16_t seq_num_end,
+    bool keyframe,
+    VideoCodecType codec,
+    const RTPVideoTypeHeader& video_type_header) {
+  RTPVideoHeader video_header;
+  video_header.frame_type = keyframe ? VideoFrameType::kVideoFrameKey
+                                     : VideoFrameType::kVideoFrameDelta;
+  video_header.video_type_header = video_type_header;
+
+  // clang-format off
+  return std::make_unique<RtpFrameObject>(
+      seq_num_start,
+      seq_num_end,
+      /*markerBit=*/true,
+      /*times_nacked=*/0,
+      /*first_packet_received_time=*/0,
+      /*last_packet_received_time=*/0,
+      /*rtp_timestamp=*/0,
+      /*ntp_time_ms=*/0,
+      VideoSendTiming(),
+      /*payload_type=*/0,
+      codec,
+      kVideoRotation_0,
+      VideoContentType::UNSPECIFIED,
+      video_header,
+      /*color_space=*/absl::nullopt,
+      RtpPacketInfos(),
+      EncodedImageBuffer::Create(/*size=*/0));
+  // clang-format on
+}
+}  // namespace
+
+class TestRtpFrameReferenceFinder : public ::testing::Test {
+ protected:
+  TestRtpFrameReferenceFinder()
+      : rand_(0x8739211),
+        reference_finder_(std::make_unique<RtpFrameReferenceFinder>()),
+        frames_from_callback_(FrameComp()) {}
+
+  uint16_t Rand() { return rand_.Rand<uint16_t>(); }
+
+  void OnCompleteFrames(RtpFrameReferenceFinder::ReturnVector frames) {
+    for (auto& frame : frames) {
+      int64_t pid = frame->Id();
+      uint16_t sidx = *frame->SpatialIndex();
+      auto frame_it = frames_from_callback_.find(std::make_pair(pid, sidx));
+      if (frame_it != frames_from_callback_.end()) {
+        ADD_FAILURE() << "Already received frame with (pid:sidx): (" << pid
+                      << ":" << sidx << ")";
+        return;
+      }
+
+      frames_from_callback_.insert(
+          std::make_pair(std::make_pair(pid, sidx), std::move(frame)));
+    }
+  }
+
+  void InsertGeneric(uint16_t seq_num_start,
+                     uint16_t seq_num_end,
+                     bool keyframe) {
+    std::unique_ptr<RtpFrameObject> frame =
+        CreateFrame(seq_num_start, seq_num_end, keyframe, kVideoCodecGeneric,
+                    RTPVideoTypeHeader());
+
+    OnCompleteFrames(reference_finder_->ManageFrame(std::move(frame)));
+  }
+
+  void InsertH264(uint16_t seq_num_start, uint16_t seq_num_end, bool keyframe) {
+    std::unique_ptr<RtpFrameObject> frame =
+        CreateFrame(seq_num_start, seq_num_end, keyframe, kVideoCodecH264,
+                    RTPVideoTypeHeader());
+    OnCompleteFrames(reference_finder_->ManageFrame(std::move(frame)));
+  }
+
+  void InsertPadding(uint16_t seq_num) {
+    OnCompleteFrames(reference_finder_->PaddingReceived(seq_num));
+  }
+
+  // Check if a frame with picture id `pid` and spatial index `sidx` has been
+  // delivered from the packet buffer, and if so, if it has the references
+  // specified by `refs`.
+  template <typename... T>
+  void CheckReferences(int64_t picture_id_offset,
+                       uint16_t sidx,
+                       T... refs) const {
+    int64_t pid = picture_id_offset;
+    auto frame_it = frames_from_callback_.find(std::make_pair(pid, sidx));
+    if (frame_it == frames_from_callback_.end()) {
+      ADD_FAILURE() << "Could not find frame with (pid:sidx): (" << pid << ":"
+                    << sidx << ")";
+      return;
+    }
+
+    std::set<int64_t> actual_refs;
+    for (uint8_t r = 0; r < frame_it->second->num_references; ++r)
+      actual_refs.insert(frame_it->second->references[r]);
+
+    std::set<int64_t> expected_refs;
+    RefsToSet(&expected_refs, refs...);
+
+    ASSERT_EQ(expected_refs, actual_refs);
+  }
+
+  template <typename... T>
+  void CheckReferencesGeneric(int64_t pid, T... refs) const {
+    CheckReferences(pid, 0, refs...);
+  }
+
+  template <typename... T>
+  void CheckReferencesH264(int64_t pid, T... refs) const {
+    CheckReferences(pid, 0, refs...);
+  }
+
+  template <typename... T>
+  void RefsToSet(std::set<int64_t>* m, int64_t ref, T... refs) const {
+    m->insert(ref);
+    RefsToSet(m, refs...);
+  }
+
+  void RefsToSet(std::set<int64_t>* m) const {}
+
+  Random rand_;
+  std::unique_ptr<RtpFrameReferenceFinder> reference_finder_;
+  struct FrameComp {
+    bool operator()(const std::pair<int64_t, uint8_t> f1,
+                    const std::pair<int64_t, uint8_t> f2) const {
+      if (f1.first == f2.first)
+        return f1.second < f2.second;
+      return f1.first < f2.first;
+    }
+  };
+  std::
+      map<std::pair<int64_t, uint8_t>, std::unique_ptr<EncodedFrame>, FrameComp>
+          frames_from_callback_;
+};
+
+TEST_F(TestRtpFrameReferenceFinder, PaddingPackets) {
+  uint16_t sn = Rand();
+
+  InsertGeneric(sn, sn, true);
+  InsertGeneric(sn + 2, sn + 2, false);
+  EXPECT_EQ(1UL, frames_from_callback_.size());
+  InsertPadding(sn + 1);
+  EXPECT_EQ(2UL, frames_from_callback_.size());
+}
+
+TEST_F(TestRtpFrameReferenceFinder, PaddingPacketsReordered) {
+  uint16_t sn = Rand();
+
+  InsertGeneric(sn, sn, true);
+  InsertPadding(sn + 1);
+  InsertPadding(sn + 4);
+  InsertGeneric(sn + 2, sn + 3, false);
+
+  EXPECT_EQ(2UL, frames_from_callback_.size());
+  CheckReferencesGeneric(sn);
+  CheckReferencesGeneric(sn + 3, sn + 0);
+}
+
+TEST_F(TestRtpFrameReferenceFinder, PaddingPacketsReorderedMultipleKeyframes) {
+  uint16_t sn = Rand();
+
+  InsertGeneric(sn, sn, true);
+  InsertPadding(sn + 1);
+  InsertPadding(sn + 4);
+  InsertGeneric(sn + 2, sn + 3, false);
+  InsertGeneric(sn + 5, sn + 5, true);
+  InsertPadding(sn + 6);
+  InsertPadding(sn + 9);
+  InsertGeneric(sn + 7, sn + 8, false);
+
+  EXPECT_EQ(4UL, frames_from_callback_.size());
+}
+
+TEST_F(TestRtpFrameReferenceFinder, AdvanceSavedKeyframe) {
+  uint16_t sn = Rand();
+
+  InsertGeneric(sn, sn, true);
+  InsertGeneric(sn + 1, sn + 1, true);
+  InsertGeneric(sn + 2, sn + 10000, false);
+  InsertGeneric(sn + 10001, sn + 20000, false);
+  InsertGeneric(sn + 20001, sn + 30000, false);
+  InsertGeneric(sn + 30001, sn + 40000, false);
+
+  EXPECT_EQ(6UL, frames_from_callback_.size());
+}
+
+TEST_F(TestRtpFrameReferenceFinder, ClearTo) {
+  uint16_t sn = Rand();
+
+  InsertGeneric(sn, sn + 1, true);
+  InsertGeneric(sn + 4, sn + 5, false);  // stashed
+  EXPECT_EQ(1UL, frames_from_callback_.size());
+
+  InsertGeneric(sn + 6, sn + 7, true);  // keyframe
+  EXPECT_EQ(2UL, frames_from_callback_.size());
+  reference_finder_->ClearTo(sn + 7);
+
+  InsertGeneric(sn + 8, sn + 9, false);  // first frame after keyframe.
+  EXPECT_EQ(3UL, frames_from_callback_.size());
+
+  InsertGeneric(sn + 2, sn + 3, false);  // late, cleared past this frame.
+  EXPECT_EQ(3UL, frames_from_callback_.size());
+}
+
+TEST_F(TestRtpFrameReferenceFinder, H264KeyFrameReferences) {
+  uint16_t sn = Rand();
+  InsertH264(sn, sn, true);
+
+  ASSERT_EQ(1UL, frames_from_callback_.size());
+  CheckReferencesH264(sn);
+}
+
+TEST_F(TestRtpFrameReferenceFinder, H264SequenceNumberWrap) {
+  uint16_t sn = 0xFFFF;
+
+  InsertH264(sn - 1, sn - 1, true);
+  InsertH264(sn, sn, false);
+  InsertH264(sn + 1, sn + 1, false);
+  InsertH264(sn + 2, sn + 2, false);
+
+  ASSERT_EQ(4UL, frames_from_callback_.size());
+  CheckReferencesH264(sn - 1);
+  CheckReferencesH264(sn, sn - 1);
+  CheckReferencesH264(sn + 1, sn);
+  CheckReferencesH264(sn + 2, sn + 1);
+}
+
+TEST_F(TestRtpFrameReferenceFinder, H264Frames) {
+  uint16_t sn = Rand();
+
+  InsertH264(sn, sn, true);
+  InsertH264(sn + 1, sn + 1, false);
+  InsertH264(sn + 2, sn + 2, false);
+  InsertH264(sn + 3, sn + 3, false);
+
+  ASSERT_EQ(4UL, frames_from_callback_.size());
+  CheckReferencesH264(sn);
+  CheckReferencesH264(sn + 1, sn);
+  CheckReferencesH264(sn + 2, sn + 1);
+  CheckReferencesH264(sn + 3, sn + 2);
+}
+
+TEST_F(TestRtpFrameReferenceFinder, H264Reordering) {
+  uint16_t sn = Rand();
+
+  InsertH264(sn, sn, true);
+  InsertH264(sn + 1, sn + 1, false);
+  InsertH264(sn + 3, sn + 3, false);
+  InsertH264(sn + 2, sn + 2, false);
+  InsertH264(sn + 5, sn + 5, false);
+  InsertH264(sn + 6, sn + 6, false);
+  InsertH264(sn + 4, sn + 4, false);
+
+  ASSERT_EQ(7UL, frames_from_callback_.size());
+  CheckReferencesH264(sn);
+  CheckReferencesH264(sn + 1, sn);
+  CheckReferencesH264(sn + 2, sn + 1);
+  CheckReferencesH264(sn + 3, sn + 2);
+  CheckReferencesH264(sn + 4, sn + 3);
+  CheckReferencesH264(sn + 5, sn + 4);
+  CheckReferencesH264(sn + 6, sn + 5);
+}
+
+TEST_F(TestRtpFrameReferenceFinder, H264SequenceNumberWrapMulti) {
+  uint16_t sn = 0xFFFF;
+
+  InsertH264(sn - 3, sn - 2, true);
+  InsertH264(sn - 1, sn + 1, false);
+  InsertH264(sn + 2, sn + 3, false);
+  InsertH264(sn + 4, sn + 7, false);
+
+  ASSERT_EQ(4UL, frames_from_callback_.size());
+  CheckReferencesH264(sn - 2);
+  CheckReferencesH264(sn + 1, sn - 2);
+  CheckReferencesH264(sn + 3, sn + 1);
+  CheckReferencesH264(sn + 7, sn + 3);
+}
+
+TEST_F(TestRtpFrameReferenceFinder, Av1FrameNoDependencyDescriptor) {
+  uint16_t sn = 0xFFFF;
+  std::unique_ptr<RtpFrameObject> frame =
+      CreateFrame(/*seq_num_start=*/sn, /*seq_num_end=*/sn, /*keyframe=*/true,
+                  kVideoCodecAV1, RTPVideoTypeHeader());
+
+  OnCompleteFrames(reference_finder_->ManageFrame(std::move(frame)));
+
+  ASSERT_EQ(1UL, frames_from_callback_.size());
+  CheckReferencesGeneric(sn);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_generic_ref_finder.cc b/third_party/libwebrtc/modules/video_coding/rtp_generic_ref_finder.cc
new file mode 100644
index 0000000000..fd5b8afda1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_generic_ref_finder.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/rtp_generic_ref_finder.h"
+
+#include <utility>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+RtpFrameReferenceFinder::ReturnVector RtpGenericFrameRefFinder::ManageFrame(
+    std::unique_ptr<RtpFrameObject> frame,
+    const RTPVideoHeader::GenericDescriptorInfo& descriptor) {
+  // Frame IDs are unwrapped in the RtpVideoStreamReceiver, no need to unwrap
+  // them here.
+  frame->SetId(descriptor.frame_id);
+  frame->SetSpatialIndex(descriptor.spatial_index);
+  if (descriptor.temporal_index != kNoTemporalIdx)
+    frame->SetTemporalIndex(descriptor.temporal_index);
+
+  RtpFrameReferenceFinder::ReturnVector res;
+  if (EncodedFrame::kMaxFrameReferences < descriptor.dependencies.size()) {
+    RTC_LOG(LS_WARNING) << "Too many dependencies in generic descriptor.";
+    return res;
+  }
+
+  frame->num_references = descriptor.dependencies.size();
+  for (size_t i = 0; i < descriptor.dependencies.size(); ++i) {
+    frame->references[i] = descriptor.dependencies[i];
+  }
+
+  res.push_back(std::move(frame));
+  return res;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_generic_ref_finder.h b/third_party/libwebrtc/modules/video_coding/rtp_generic_ref_finder.h
new file mode 100644
index 0000000000..87d7b59406
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_generic_ref_finder.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_RTP_GENERIC_REF_FINDER_H_
+#define MODULES_VIDEO_CODING_RTP_GENERIC_REF_FINDER_H_
+
+#include <memory>
+
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+
+namespace webrtc {
+
+class RtpGenericFrameRefFinder {
+ public:
+  RtpGenericFrameRefFinder() = default;
+
+  RtpFrameReferenceFinder::ReturnVector ManageFrame(
+      std::unique_ptr<RtpFrameObject> frame,
+      const RTPVideoHeader::GenericDescriptorInfo& descriptor);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_RTP_GENERIC_REF_FINDER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_seq_num_only_ref_finder.cc b/third_party/libwebrtc/modules/video_coding/rtp_seq_num_only_ref_finder.cc
new file mode 100644
index 0000000000..59b027e2ce
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_seq_num_only_ref_finder.cc
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/rtp_seq_num_only_ref_finder.h"
+
+#include <utility>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+RtpFrameReferenceFinder::ReturnVector RtpSeqNumOnlyRefFinder::ManageFrame(
+    std::unique_ptr<RtpFrameObject> frame) {
+  FrameDecision decision = ManageFrameInternal(frame.get());
+
+  RtpFrameReferenceFinder::ReturnVector res;
+  switch (decision) {
+    case kStash:
+      if (stashed_frames_.size() > kMaxStashedFrames)
+        stashed_frames_.pop_back();
+      stashed_frames_.push_front(std::move(frame));
+      return res;
+    case kHandOff:
+      res.push_back(std::move(frame));
+      RetryStashedFrames(res);
+      return res;
+    case kDrop:
+      return res;
+  }
+
+  return res;
+}
+
+RtpSeqNumOnlyRefFinder::FrameDecision
+RtpSeqNumOnlyRefFinder::ManageFrameInternal(RtpFrameObject* frame) {
+  if (frame->frame_type() == VideoFrameType::kVideoFrameKey) {
+    last_seq_num_gop_.insert(std::make_pair(
+        frame->last_seq_num(),
+        std::make_pair(frame->last_seq_num(), frame->last_seq_num())));
+  }
+
+  // We have received a frame but not yet a keyframe, stash this frame.
+  if (last_seq_num_gop_.empty())
+    return kStash;
+
+  // Clean up info for old keyframes but make sure to keep info
+  // for the last keyframe.
+  auto clean_to = last_seq_num_gop_.lower_bound(frame->last_seq_num() - 100);
+  for (auto it = last_seq_num_gop_.begin();
+       it != clean_to && last_seq_num_gop_.size() > 1;) {
+    it = last_seq_num_gop_.erase(it);
+  }
+
+  // Find the last sequence number of the last frame for the keyframe
+  // that this frame indirectly references.
+  auto seq_num_it = last_seq_num_gop_.upper_bound(frame->last_seq_num());
+  if (seq_num_it == last_seq_num_gop_.begin()) {
+    RTC_LOG(LS_WARNING) << "Generic frame with packet range ["
+                        << frame->first_seq_num() << ", "
+                        << frame->last_seq_num()
+                        << "] has no GoP, dropping frame.";
+    return kDrop;
+  }
+  seq_num_it--;
+
+  // Make sure the packet sequence numbers are continuous, otherwise stash
+  // this frame.
+  uint16_t last_picture_id_gop = seq_num_it->second.first;
+  uint16_t last_picture_id_with_padding_gop = seq_num_it->second.second;
+  if (frame->frame_type() == VideoFrameType::kVideoFrameDelta) {
+    uint16_t prev_seq_num = frame->first_seq_num() - 1;
+
+    if (prev_seq_num != last_picture_id_with_padding_gop)
+      return kStash;
+  }
+
+  RTC_DCHECK(AheadOrAt(frame->last_seq_num(), seq_num_it->first));
+
+  // Since keyframes can cause reordering we can't simply assign the
+  // picture id according to some incrementing counter.
+  frame->SetId(frame->last_seq_num());
+  frame->num_references =
+      frame->frame_type() == VideoFrameType::kVideoFrameDelta;
+  frame->references[0] = rtp_seq_num_unwrapper_.Unwrap(last_picture_id_gop);
+  if (AheadOf<uint16_t>(frame->Id(), last_picture_id_gop)) {
+    seq_num_it->second.first = frame->Id();
+    seq_num_it->second.second = frame->Id();
+  }
+
+  UpdateLastPictureIdWithPadding(frame->Id());
+  frame->SetSpatialIndex(0);
+  frame->SetId(rtp_seq_num_unwrapper_.Unwrap(frame->Id()));
+  return kHandOff;
+}
+
+void RtpSeqNumOnlyRefFinder::RetryStashedFrames(
+    RtpFrameReferenceFinder::ReturnVector& res) {
+  bool complete_frame = false;
+  do {
+    complete_frame = false;
+    for (auto frame_it = stashed_frames_.begin();
+         frame_it != stashed_frames_.end();) {
+      FrameDecision decision = ManageFrameInternal(frame_it->get());
+
+      switch (decision) {
+        case kStash:
+          ++frame_it;
+          break;
+        case kHandOff:
+          complete_frame = true;
+          res.push_back(std::move(*frame_it));
+          [[fallthrough]];
+        case kDrop:
+          frame_it = stashed_frames_.erase(frame_it);
+      }
+    }
+  } while (complete_frame);
+}
+
+void RtpSeqNumOnlyRefFinder::UpdateLastPictureIdWithPadding(uint16_t seq_num) {
+  auto gop_seq_num_it = last_seq_num_gop_.upper_bound(seq_num);
+
+  // If this padding packet "belongs" to a group of pictures that we don't track
+  // anymore, do nothing.
+  if (gop_seq_num_it == last_seq_num_gop_.begin())
+    return;
+  --gop_seq_num_it;
+
+  // Calculate the next contiuous sequence number and search for it in
+  // the padding packets we have stashed.
+  uint16_t next_seq_num_with_padding = gop_seq_num_it->second.second + 1;
+  auto padding_seq_num_it =
+      stashed_padding_.lower_bound(next_seq_num_with_padding);
+
+  // While there still are padding packets and those padding packets are
+  // continuous, then advance the "last-picture-id-with-padding" and remove
+  // the stashed padding packet.
+  while (padding_seq_num_it != stashed_padding_.end() &&
+         *padding_seq_num_it == next_seq_num_with_padding) {
+    gop_seq_num_it->second.second = next_seq_num_with_padding;
+    ++next_seq_num_with_padding;
+    padding_seq_num_it = stashed_padding_.erase(padding_seq_num_it);
+  }
+
+  // In the case where the stream has been continuous without any new keyframes
+  // for a while there is a risk that new frames will appear to be older than
+  // the keyframe they belong to due to wrapping sequence number. In order
+  // to prevent this we advance the picture id of the keyframe every so often.
+  if (ForwardDiff(gop_seq_num_it->first, seq_num) > 10000) {
+    auto save = gop_seq_num_it->second;
+    last_seq_num_gop_.clear();
+    last_seq_num_gop_[seq_num] = save;
+  }
+}
+
+RtpFrameReferenceFinder::ReturnVector RtpSeqNumOnlyRefFinder::PaddingReceived(
+    uint16_t seq_num) {
+  auto clean_padding_to =
+      stashed_padding_.lower_bound(seq_num - kMaxPaddingAge);
+  stashed_padding_.erase(stashed_padding_.begin(), clean_padding_to);
+  stashed_padding_.insert(seq_num);
+  UpdateLastPictureIdWithPadding(seq_num);
+  RtpFrameReferenceFinder::ReturnVector res;
+  RetryStashedFrames(res);
+  return res;
+}
+
+void RtpSeqNumOnlyRefFinder::ClearTo(uint16_t seq_num) {
+  auto it = stashed_frames_.begin();
+  while (it != stashed_frames_.end()) {
+    if (AheadOf<uint16_t>(seq_num, (*it)->first_seq_num())) {
+      it = stashed_frames_.erase(it);
+    } else {
+      ++it;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_seq_num_only_ref_finder.h b/third_party/libwebrtc/modules/video_coding/rtp_seq_num_only_ref_finder.h
new file mode 100644
index 0000000000..c05655b579
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_seq_num_only_ref_finder.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_RTP_SEQ_NUM_ONLY_REF_FINDER_H_
+#define MODULES_VIDEO_CODING_RTP_SEQ_NUM_ONLY_REF_FINDER_H_
+
+#include <deque>
+#include <map>
+#include <memory>
+#include <set>
+#include <utility>
+
+#include "absl/container/inlined_vector.h"
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+class RtpSeqNumOnlyRefFinder {
+ public:
+  RtpSeqNumOnlyRefFinder() = default;
+
+  RtpFrameReferenceFinder::ReturnVector ManageFrame(
+      std::unique_ptr<RtpFrameObject> frame);
+  RtpFrameReferenceFinder::ReturnVector PaddingReceived(uint16_t seq_num);
+  void ClearTo(uint16_t seq_num);
+
+ private:
+  static constexpr int kMaxStashedFrames = 100;
+  static constexpr int kMaxPaddingAge = 100;
+
+  enum FrameDecision { kStash, kHandOff, kDrop };
+
+  FrameDecision ManageFrameInternal(RtpFrameObject* frame);
+  void RetryStashedFrames(RtpFrameReferenceFinder::ReturnVector& res);
+  void UpdateLastPictureIdWithPadding(uint16_t seq_num);
+
+  // For every group of pictures, hold two sequence numbers. The first being
+  // the sequence number of the last packet of the last completed frame, and
+  // the second being the sequence number of the last packet of the last
+  // completed frame advanced by any potential continuous packets of padding.
+  std::map<uint16_t,
+           std::pair<uint16_t, uint16_t>,
+           DescendingSeqNumComp<uint16_t>>
+      last_seq_num_gop_;
+
+  // Padding packets that have been received but that are not yet continuous
+  // with any group of pictures.
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> stashed_padding_;
+
+  // Frames that have been fully received but didn't have all the information
+  // needed to determine their references.
+  std::deque<std::unique_ptr<RtpFrameObject>> stashed_frames_;
+
+  // Unwrapper used to unwrap generic RTP streams. In a generic stream we derive
+  // a picture id from the packet sequence number.
+  SeqNumUnwrapper<uint16_t> rtp_seq_num_unwrapper_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_RTP_SEQ_NUM_ONLY_REF_FINDER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder.cc b/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder.cc
new file mode 100644
index 0000000000..185756ce51
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder.cc
@@ -0,0 +1,254 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/rtp_vp8_ref_finder.h"
+
+#include <utility>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+RtpFrameReferenceFinder::ReturnVector RtpVp8RefFinder::ManageFrame(
+    std::unique_ptr<RtpFrameObject> frame) {
+  const RTPVideoHeaderVP8& codec_header = absl::get<RTPVideoHeaderVP8>(
+      frame->GetRtpVideoHeader().video_type_header);
+
+  if (codec_header.temporalIdx != kNoTemporalIdx)
+    frame->SetTemporalIndex(codec_header.temporalIdx);
+
+  int64_t unwrapped_tl0 = tl0_unwrapper_.Unwrap(codec_header.tl0PicIdx & 0xFF);
+  FrameDecision decision =
+      ManageFrameInternal(frame.get(), codec_header, unwrapped_tl0);
+
+  RtpFrameReferenceFinder::ReturnVector res;
+  switch (decision) {
+    case kStash:
+      if (stashed_frames_.size() > kMaxStashedFrames) {
+        stashed_frames_.pop_back();
+      }
+      stashed_frames_.push_front(
+          {.unwrapped_tl0 = unwrapped_tl0, .frame = std::move(frame)});
+      return res;
+    case kHandOff:
+      res.push_back(std::move(frame));
+      RetryStashedFrames(res);
+      return res;
+    case kDrop:
+      return res;
+  }
+
+  return res;
+}
+
+RtpVp8RefFinder::FrameDecision RtpVp8RefFinder::ManageFrameInternal(
+    RtpFrameObject* frame,
+    const RTPVideoHeaderVP8& codec_header,
+    int64_t unwrapped_tl0) {
+  // Protect against corrupted packets with arbitrary large temporal idx.
+  if (codec_header.temporalIdx >= kMaxTemporalLayers)
+    return kDrop;
+
+  frame->SetSpatialIndex(0);
+  frame->SetId(codec_header.pictureId & 0x7FFF);
+
+  if (last_picture_id_ == -1)
+    last_picture_id_ = frame->Id();
+
+  // Clean up info about not yet received frames that are too old.
+  uint16_t old_picture_id =
+      Subtract<kFrameIdLength>(frame->Id(), kMaxNotYetReceivedFrames);
+  auto clean_frames_to = not_yet_received_frames_.lower_bound(old_picture_id);
+  not_yet_received_frames_.erase(not_yet_received_frames_.begin(),
+                                 clean_frames_to);
+  // Avoid re-adding picture ids that were just erased.
+  if (AheadOf<uint16_t, kFrameIdLength>(old_picture_id, last_picture_id_)) {
+    last_picture_id_ = old_picture_id;
+  }
+  // Find if there has been a gap in fully received frames and save the picture
+  // id of those frames in `not_yet_received_frames_`.
+  if (AheadOf<uint16_t, kFrameIdLength>(frame->Id(), last_picture_id_)) {
+    do {
+      last_picture_id_ = Add<kFrameIdLength>(last_picture_id_, 1);
+      not_yet_received_frames_.insert(last_picture_id_);
+    } while (last_picture_id_ != frame->Id());
+  }
+
+  // Clean up info for base layers that are too old.
+  int64_t old_tl0_pic_idx = unwrapped_tl0 - kMaxLayerInfo;
+  auto clean_layer_info_to = layer_info_.lower_bound(old_tl0_pic_idx);
+  layer_info_.erase(layer_info_.begin(), clean_layer_info_to);
+
+  if (frame->frame_type() == VideoFrameType::kVideoFrameKey) {
+    if (codec_header.temporalIdx != 0) {
+      return kDrop;
+    }
+    frame->num_references = 0;
+    layer_info_[unwrapped_tl0].fill(-1);
+    UpdateLayerInfoVp8(frame, unwrapped_tl0, codec_header.temporalIdx);
+    return kHandOff;
+  }
+
+  auto layer_info_it = layer_info_.find(
+      codec_header.temporalIdx == 0 ? unwrapped_tl0 - 1 : unwrapped_tl0);
+
+  // If we don't have the base layer frame yet, stash this frame.
+  if (layer_info_it == layer_info_.end())
+    return kStash;
+
+  // A non keyframe base layer frame has been received, copy the layer info
+  // from the previous base layer frame and set a reference to the previous
+  // base layer frame.
+  if (codec_header.temporalIdx == 0) {
+    layer_info_it =
+        layer_info_.emplace(unwrapped_tl0, layer_info_it->second).first;
+    frame->num_references = 1;
+    int64_t last_pid_on_layer = layer_info_it->second[0];
+
+    // Is this an old frame that has already been used to update the state? If
+    // so, drop it.
+    if (AheadOrAt<uint16_t, kFrameIdLength>(last_pid_on_layer, frame->Id())) {
+      return kDrop;
+    }
+
+    frame->references[0] = last_pid_on_layer;
+    UpdateLayerInfoVp8(frame, unwrapped_tl0, codec_header.temporalIdx);
+    return kHandOff;
+  }
+
+  // Layer sync frame, this frame only references its base layer frame.
+  if (codec_header.layerSync) {
+    frame->num_references = 1;
+    int64_t last_pid_on_layer = layer_info_it->second[codec_header.temporalIdx];
+
+    // Is this an old frame that has already been used to update the state? If
+    // so, drop it.
+    if (last_pid_on_layer != -1 &&
+        AheadOrAt<uint16_t, kFrameIdLength>(last_pid_on_layer, frame->Id())) {
+      return kDrop;
+    }
+
+    frame->references[0] = layer_info_it->second[0];
+    UpdateLayerInfoVp8(frame, unwrapped_tl0, codec_header.temporalIdx);
+    return kHandOff;
+  }
+
+  // Find all references for this frame.
+  frame->num_references = 0;
+  for (uint8_t layer = 0; layer <= codec_header.temporalIdx; ++layer) {
+    // If we have not yet received a previous frame on this temporal layer,
+    // stash this frame.
+    if (layer_info_it->second[layer] == -1)
+      return kStash;
+
+    // If the last frame on this layer is ahead of this frame it means that
+    // a layer sync frame has been received after this frame for the same
+    // base layer frame, drop this frame.
+    if (AheadOf<uint16_t, kFrameIdLength>(layer_info_it->second[layer],
+                                          frame->Id())) {
+      return kDrop;
+    }
+
+    // If we have not yet received a frame between this frame and the referenced
+    // frame then we have to wait for that frame to be completed first.
+    auto not_received_frame_it =
+        not_yet_received_frames_.upper_bound(layer_info_it->second[layer]);
+    if (not_received_frame_it != not_yet_received_frames_.end() &&
+        AheadOf<uint16_t, kFrameIdLength>(frame->Id(),
+                                          *not_received_frame_it)) {
+      return kStash;
+    }
+
+    if (!(AheadOf<uint16_t, kFrameIdLength>(frame->Id(),
+                                            layer_info_it->second[layer]))) {
+      RTC_LOG(LS_WARNING) << "Frame with picture id " << frame->Id()
+                          << " and packet range [" << frame->first_seq_num()
+                          << ", " << frame->last_seq_num()
+                          << "] already received, "
+                             " dropping frame.";
+      return kDrop;
+    }
+
+    ++frame->num_references;
+    frame->references[layer] = layer_info_it->second[layer];
+  }
+
+  UpdateLayerInfoVp8(frame, unwrapped_tl0, codec_header.temporalIdx);
+  return kHandOff;
+}
+
+void RtpVp8RefFinder::UpdateLayerInfoVp8(RtpFrameObject* frame,
+                                         int64_t unwrapped_tl0,
+                                         uint8_t temporal_idx) {
+  auto layer_info_it = layer_info_.find(unwrapped_tl0);
+
+  // Update this layer info and newer.
+  while (layer_info_it != layer_info_.end()) {
+    if (layer_info_it->second[temporal_idx] != -1 &&
+        AheadOf<uint16_t, kFrameIdLength>(layer_info_it->second[temporal_idx],
+                                          frame->Id())) {
+      // The frame was not newer, then no subsequent layer info have to be
+      // update.
+      break;
+    }
+
+    layer_info_it->second[temporal_idx] = frame->Id();
+    ++unwrapped_tl0;
+    layer_info_it = layer_info_.find(unwrapped_tl0);
+  }
+  not_yet_received_frames_.erase(frame->Id());
+
+  UnwrapPictureIds(frame);
+}
+
+void RtpVp8RefFinder::RetryStashedFrames(
+    RtpFrameReferenceFinder::ReturnVector& res) {
+  bool complete_frame = false;
+  do {
+    complete_frame = false;
+    for (auto it = stashed_frames_.begin(); it != stashed_frames_.end();) {
+      const RTPVideoHeaderVP8& codec_header = absl::get<RTPVideoHeaderVP8>(
+          it->frame->GetRtpVideoHeader().video_type_header);
+      FrameDecision decision =
+          ManageFrameInternal(it->frame.get(), codec_header, it->unwrapped_tl0);
+
+      switch (decision) {
+        case kStash:
+          ++it;
+          break;
+        case kHandOff:
+          complete_frame = true;
+          res.push_back(std::move(it->frame));
+          [[fallthrough]];
+        case kDrop:
+          it = stashed_frames_.erase(it);
+      }
+    }
+  } while (complete_frame);
+}
+
+void RtpVp8RefFinder::UnwrapPictureIds(RtpFrameObject* frame) {
+  for (size_t i = 0; i < frame->num_references; ++i)
+    frame->references[i] = unwrapper_.Unwrap(frame->references[i]);
+  frame->SetId(unwrapper_.Unwrap(frame->Id()));
+}
+
+void RtpVp8RefFinder::ClearTo(uint16_t seq_num) {
+  auto it = stashed_frames_.begin();
+  while (it != stashed_frames_.end()) {
+    if (AheadOf<uint16_t>(seq_num, it->frame->first_seq_num())) {
+      it = stashed_frames_.erase(it);
+    } else {
+      ++it;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder.h b/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder.h
new file mode 100644
index 0000000000..26df658a3b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_RTP_VP8_REF_FINDER_H_
+#define MODULES_VIDEO_CODING_RTP_VP8_REF_FINDER_H_
+
+#include <deque>
+#include <map>
+#include <memory>
+#include <set>
+
+#include "absl/container/inlined_vector.h"
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+class RtpVp8RefFinder {
+ public:
+  RtpVp8RefFinder() = default;
+
+  RtpFrameReferenceFinder::ReturnVector ManageFrame(
+      std::unique_ptr<RtpFrameObject> frame);
+  void ClearTo(uint16_t seq_num);
+
+ private:
+  static constexpr int kFrameIdLength = 1 << 15;
+  static constexpr int kMaxLayerInfo = 50;
+  static constexpr int kMaxNotYetReceivedFrames = 100;
+  static constexpr int kMaxStashedFrames = 100;
+  static constexpr int kMaxTemporalLayers = 5;
+
+  struct UnwrappedTl0Frame {
+    int64_t unwrapped_tl0;
+    std::unique_ptr<RtpFrameObject> frame;
+  };
+
+  enum FrameDecision { kStash, kHandOff, kDrop };
+
+  FrameDecision ManageFrameInternal(RtpFrameObject* frame,
+                                    const RTPVideoHeaderVP8& codec_header,
+                                    int64_t unwrapped_tl0);
+  void RetryStashedFrames(RtpFrameReferenceFinder::ReturnVector& res);
+  void UpdateLayerInfoVp8(RtpFrameObject* frame,
+                          int64_t unwrapped_tl0,
+                          uint8_t temporal_idx);
+  void UnwrapPictureIds(RtpFrameObject* frame);
+
+  // Save the last picture id in order to detect when there is a gap in frames
+  // that have not yet been fully received.
+  int last_picture_id_ = -1;
+
+  // Frames earlier than the last received frame that have not yet been
+  // fully received.
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t, kFrameIdLength>>
+      not_yet_received_frames_;
+
+  // Frames that have been fully received but didn't have all the information
+  // needed to determine their references.
+  std::deque<UnwrappedTl0Frame> stashed_frames_;
+
+  // Holds the information about the last completed frame for a given temporal
+  // layer given an unwrapped Tl0 picture index.
+  std::map<int64_t, std::array<int64_t, kMaxTemporalLayers>> layer_info_;
+
+  // Unwrapper used to unwrap VP8/VP9 streams which have their picture id
+  // specified.
+  SeqNumUnwrapper<uint16_t, kFrameIdLength> unwrapper_;
+
+  SeqNumUnwrapper<uint8_t> tl0_unwrapper_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_RTP_VP8_REF_FINDER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder_unittest.cc b/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder_unittest.cc
new file mode 100644
index 0000000000..7dc6cd5521
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder_unittest.cc
@@ -0,0 +1,370 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/rtp_vp8_ref_finder.h"
+
+#include <utility>
+#include <vector>
+
+#include "modules/video_coding/frame_object.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::Contains;
+using ::testing::Eq;
+using ::testing::Matcher;
+using ::testing::Matches;
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAreArray;
+
+namespace webrtc {
+namespace {
+
+MATCHER_P2(HasIdAndRefs, id, refs, "") {
+  return Matches(Eq(id))(arg->Id()) &&
+         Matches(UnorderedElementsAreArray(refs))(
+             rtc::ArrayView<int64_t>(arg->references, arg->num_references));
+}
+
+Matcher<const std::vector<std::unique_ptr<EncodedFrame>>&>
+HasFrameWithIdAndRefs(int64_t frame_id, const std::vector<int64_t>& refs) {
+  return Contains(HasIdAndRefs(frame_id, refs));
+}
+
+class Frame {
+ public:
+  Frame& AsKeyFrame(bool is_keyframe = true) {
+    is_keyframe_ = is_keyframe;
+    return *this;
+  }
+
+  Frame& Pid(int pid) {
+    picture_id_ = pid;
+    return *this;
+  }
+
+  Frame& Tid(int tid) {
+    temporal_id_ = tid;
+    return *this;
+  }
+
+  Frame& Tl0(int tl0) {
+    tl0_idx_ = tl0;
+    return *this;
+  }
+
+  Frame& AsSync(bool is_sync = true) {
+    sync = is_sync;
+    return *this;
+  }
+
+  operator std::unique_ptr<RtpFrameObject>() {
+    RTPVideoHeaderVP8 vp8_header{};
+    vp8_header.pictureId = *picture_id_;
+    vp8_header.temporalIdx = *temporal_id_;
+    vp8_header.tl0PicIdx = *tl0_idx_;
+    vp8_header.layerSync = sync;
+
+    RTPVideoHeader video_header;
+    video_header.frame_type = is_keyframe_ ? VideoFrameType::kVideoFrameKey
+                                           : VideoFrameType::kVideoFrameDelta;
+    video_header.video_type_header = vp8_header;
+    // clang-format off
+    return std::make_unique<RtpFrameObject>(
+        /*seq_num_start=*/0,
+        /*seq_num_end=*/0,
+        /*markerBit=*/true,
+        /*times_nacked=*/0,
+        /*first_packet_received_time=*/0,
+        /*last_packet_received_time=*/0,
+        /*rtp_timestamp=*/0,
+        /*ntp_time_ms=*/0,
+        VideoSendTiming(),
+        /*payload_type=*/0,
+        kVideoCodecVP8,
+        kVideoRotation_0,
+        VideoContentType::UNSPECIFIED,
+        video_header,
+        /*color_space=*/absl::nullopt,
+        RtpPacketInfos(),
+        EncodedImageBuffer::Create(/*size=*/0));
+    // clang-format on
+  }
+
+ private:
+  bool is_keyframe_ = false;
+  absl::optional<int> picture_id_;
+  absl::optional<int> temporal_id_;
+  absl::optional<int> tl0_idx_;
+  bool sync = false;
+};
+
+}  // namespace
+
+class RtpVp8RefFinderTest : public ::testing::Test {
+ protected:
+  RtpVp8RefFinderTest() : ref_finder_(std::make_unique<RtpVp8RefFinder>()) {}
+
+  void Insert(std::unique_ptr<RtpFrameObject> frame) {
+    for (auto& f : ref_finder_->ManageFrame(std::move(frame))) {
+      frames_.push_back(std::move(f));
+    }
+  }
+
+  std::unique_ptr<RtpVp8RefFinder> ref_finder_;
+  std::vector<std::unique_ptr<EncodedFrame>> frames_;
+};
+
+TEST_F(RtpVp8RefFinderTest, Vp8RepeatedFrame_0) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(1).AsKeyFrame());
+  Insert(Frame().Pid(1).Tid(0).Tl0(2));
+  Insert(Frame().Pid(1).Tid(0).Tl0(2));
+
+  EXPECT_THAT(frames_, SizeIs(2));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8RepeatedFrameLayerSync_01) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(1).AsKeyFrame());
+  Insert(Frame().Pid(1).Tid(1).Tl0(1).AsSync());
+  Insert(Frame().Pid(1).Tid(1).Tl0(1).AsSync());
+
+  EXPECT_THAT(frames_, SizeIs(2));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8RepeatedFrame_01) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(1).AsKeyFrame());
+  Insert(Frame().Pid(1).Tid(0).Tl0(2).AsSync());
+  Insert(Frame().Pid(2).Tid(0).Tl0(3));
+  Insert(Frame().Pid(3).Tid(0).Tl0(4));
+  Insert(Frame().Pid(3).Tid(0).Tl0(4));
+
+  EXPECT_THAT(frames_, SizeIs(4));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(2, {1}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(3, {2}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8TemporalLayers_0) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(1).AsKeyFrame());
+  Insert(Frame().Pid(1).Tid(0).Tl0(2));
+
+  EXPECT_THAT(frames_, SizeIs(2));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8DuplicateTl1Frames) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(0).AsKeyFrame());
+  Insert(Frame().Pid(1).Tid(1).Tl0(0).AsSync());
+  Insert(Frame().Pid(2).Tid(0).Tl0(1));
+  Insert(Frame().Pid(3).Tid(1).Tl0(1));
+  Insert(Frame().Pid(3).Tid(1).Tl0(1));
+  Insert(Frame().Pid(4).Tid(0).Tl0(2));
+  Insert(Frame().Pid(5).Tid(1).Tl0(2));
+
+  EXPECT_THAT(frames_, SizeIs(6));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(2, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(3, {1, 2}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(4, {2}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {3, 4}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8TemporalLayersReordering_0) {
+  Insert(Frame().Pid(1).Tid(0).Tl0(2));
+  Insert(Frame().Pid(0).Tid(0).Tl0(1).AsKeyFrame());
+  Insert(Frame().Pid(3).Tid(0).Tl0(4));
+  Insert(Frame().Pid(2).Tid(0).Tl0(3));
+  Insert(Frame().Pid(5).Tid(0).Tl0(6));
+  Insert(Frame().Pid(6).Tid(0).Tl0(7));
+  Insert(Frame().Pid(4).Tid(0).Tl0(5));
+
+  EXPECT_THAT(frames_, SizeIs(7));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(2, {1}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(3, {2}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(4, {3}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {4}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(6, {5}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8TemporalLayers_01) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(255).AsKeyFrame());
+  Insert(Frame().Pid(1).Tid(1).Tl0(255).AsSync());
+  Insert(Frame().Pid(2).Tid(0).Tl0(0));
+  Insert(Frame().Pid(3).Tid(1).Tl0(0));
+
+  EXPECT_THAT(frames_, SizeIs(4));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(2, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(3, {1, 2}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8TemporalLayersReordering_01) {
+  Insert(Frame().Pid(1).Tid(1).Tl0(255).AsSync());
+  Insert(Frame().Pid(0).Tid(0).Tl0(255).AsKeyFrame());
+  Insert(Frame().Pid(3).Tid(1).Tl0(0));
+  Insert(Frame().Pid(5).Tid(1).Tl0(1));
+  Insert(Frame().Pid(2).Tid(0).Tl0(0));
+  Insert(Frame().Pid(4).Tid(0).Tl0(1));
+  Insert(Frame().Pid(6).Tid(0).Tl0(2));
+  Insert(Frame().Pid(7).Tid(1).Tl0(2));
+
+  EXPECT_THAT(frames_, SizeIs(8));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(2, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(3, {1, 2}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(4, {2}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {3, 4}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(6, {4}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(7, {5, 6}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8TemporalLayers_0212) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(55).AsKeyFrame());
+  Insert(Frame().Pid(1).Tid(2).Tl0(55).AsSync());
+  Insert(Frame().Pid(2).Tid(1).Tl0(55).AsSync());
+  Insert(Frame().Pid(3).Tid(2).Tl0(55));
+  Insert(Frame().Pid(4).Tid(0).Tl0(56));
+  Insert(Frame().Pid(5).Tid(2).Tl0(56));
+  Insert(Frame().Pid(6).Tid(1).Tl0(56));
+  Insert(Frame().Pid(7).Tid(2).Tl0(56));
+  Insert(Frame().Pid(8).Tid(0).Tl0(57));
+  Insert(Frame().Pid(9).Tid(2).Tl0(57).AsSync());
+  Insert(Frame().Pid(10).Tid(1).Tl0(57).AsSync());
+  Insert(Frame().Pid(11).Tid(2).Tl0(57));
+
+  EXPECT_THAT(frames_, SizeIs(12));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(2, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(3, {0, 1, 2}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(4, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {2, 3, 4}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(6, {2, 4}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(7, {4, 5, 6}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(8, {4}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(9, {8}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {8}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(11, {8, 9, 10}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8TemporalLayersMissingFrame_0212) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(55).AsKeyFrame());
+  Insert(Frame().Pid(2).Tid(1).Tl0(55).AsSync());
+  Insert(Frame().Pid(3).Tid(2).Tl0(55));
+
+  EXPECT_THAT(frames_, SizeIs(2));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(2, {0}));
+}
+
+// Test with 3 temporal layers in a 0212 pattern.
+TEST_F(RtpVp8RefFinderTest, Vp8TemporalLayersReordering_0212) {
+  Insert(Frame().Pid(127).Tid(2).Tl0(55).AsSync());
+  Insert(Frame().Pid(126).Tid(0).Tl0(55).AsKeyFrame());
+  Insert(Frame().Pid(128).Tid(1).Tl0(55).AsSync());
+  Insert(Frame().Pid(130).Tid(0).Tl0(56));
+  Insert(Frame().Pid(131).Tid(2).Tl0(56));
+  Insert(Frame().Pid(129).Tid(2).Tl0(55));
+  Insert(Frame().Pid(133).Tid(2).Tl0(56));
+  Insert(Frame().Pid(135).Tid(2).Tl0(57).AsSync());
+  Insert(Frame().Pid(132).Tid(1).Tl0(56));
+  Insert(Frame().Pid(134).Tid(0).Tl0(57));
+  Insert(Frame().Pid(137).Tid(2).Tl0(57));
+  Insert(Frame().Pid(136).Tid(1).Tl0(57).AsSync());
+
+  EXPECT_THAT(frames_, SizeIs(12));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(126, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(127, {126}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(128, {126}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(129, {126, 127, 128}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(130, {126}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(131, {128, 129, 130}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(132, {128, 130}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(133, {130, 131, 132}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(134, {130}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(135, {134}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(136, {134}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(137, {134, 135, 136}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8LayerSync) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(0).AsKeyFrame());
+  Insert(Frame().Pid(1).Tid(1).Tl0(0).AsSync());
+  Insert(Frame().Pid(2).Tid(0).Tl0(1));
+  Insert(Frame().Pid(4).Tid(0).Tl0(2));
+  Insert(Frame().Pid(5).Tid(1).Tl0(2).AsSync());
+  Insert(Frame().Pid(6).Tid(0).Tl0(3));
+  Insert(Frame().Pid(7).Tid(1).Tl0(3));
+
+  EXPECT_THAT(frames_, SizeIs(7));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(2, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(4, {2}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {4}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(6, {4}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(7, {5, 6}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8Tl1SyncFrameAfterTl1Frame) {
+  Insert(Frame().Pid(1).Tid(0).Tl0(247).AsKeyFrame().AsSync());
+  Insert(Frame().Pid(3).Tid(0).Tl0(248));
+  Insert(Frame().Pid(4).Tid(1).Tl0(248));
+  Insert(Frame().Pid(5).Tid(1).Tl0(248).AsSync());
+
+  EXPECT_THAT(frames_, SizeIs(3));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(3, {1}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {3}));
+}
+
+TEST_F(RtpVp8RefFinderTest, Vp8DetectMissingFrame_0212) {
+  Insert(Frame().Pid(1).Tid(0).Tl0(1).AsKeyFrame());
+  Insert(Frame().Pid(2).Tid(2).Tl0(1).AsSync());
+  Insert(Frame().Pid(3).Tid(1).Tl0(1).AsSync());
+  Insert(Frame().Pid(4).Tid(2).Tl0(1));
+  Insert(Frame().Pid(6).Tid(2).Tl0(2));
+  Insert(Frame().Pid(7).Tid(1).Tl0(2));
+  Insert(Frame().Pid(8).Tid(2).Tl0(2));
+  Insert(Frame().Pid(5).Tid(0).Tl0(2));
+
+  EXPECT_THAT(frames_, SizeIs(8));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(2, {1}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(3, {1}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(4, {1, 2, 3}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {1}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(6, {3, 4, 5}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(7, {3, 5}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(8, {5, 6, 7}));
+}
+
+TEST_F(RtpVp8RefFinderTest, StashedFramesDoNotWrapTl0Backwards) {
+  Insert(Frame().Pid(0).Tid(0).Tl0(0));
+  EXPECT_THAT(frames_, SizeIs(0));
+
+  Insert(Frame().Pid(128).Tid(0).Tl0(128).AsKeyFrame());
+  EXPECT_THAT(frames_, SizeIs(1));
+  Insert(Frame().Pid(129).Tid(0).Tl0(129));
+  EXPECT_THAT(frames_, SizeIs(2));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder.cc b/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder.cc
new file mode 100644
index 0000000000..175ed3464b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder.cc
@@ -0,0 +1,367 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/rtp_vp9_ref_finder.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+RtpFrameReferenceFinder::ReturnVector RtpVp9RefFinder::ManageFrame(
+    std::unique_ptr<RtpFrameObject> frame) {
+  const RTPVideoHeaderVP9& codec_header = absl::get<RTPVideoHeaderVP9>(
+      frame->GetRtpVideoHeader().video_type_header);
+
+  if (codec_header.temporal_idx != kNoTemporalIdx)
+    frame->SetTemporalIndex(codec_header.temporal_idx);
+  frame->SetSpatialIndex(codec_header.spatial_idx);
+  frame->SetId(codec_header.picture_id & (kFrameIdLength - 1));
+
+  FrameDecision decision;
+  if (codec_header.temporal_idx >= kMaxTemporalLayers ||
+      codec_header.spatial_idx >= kMaxSpatialLayers) {
+    decision = kDrop;
+  } else if (codec_header.flexible_mode) {
+    decision = ManageFrameFlexible(frame.get(), codec_header);
+  } else {
+    if (codec_header.tl0_pic_idx == kNoTl0PicIdx) {
+      RTC_LOG(LS_WARNING) << "TL0PICIDX is expected to be present in "
+                             "non-flexible mode.";
+      decision = kDrop;
+    } else {
+      int64_t unwrapped_tl0 =
+          tl0_unwrapper_.Unwrap(codec_header.tl0_pic_idx & 0xFF);
+      decision = ManageFrameGof(frame.get(), codec_header, unwrapped_tl0);
+
+      if (decision == kStash) {
+        if (stashed_frames_.size() > kMaxStashedFrames) {
+          stashed_frames_.pop_back();
+        }
+
+        stashed_frames_.push_front(
+            {.unwrapped_tl0 = unwrapped_tl0, .frame = std::move(frame)});
+      }
+    }
+  }
+
+  RtpFrameReferenceFinder::ReturnVector res;
+  switch (decision) {
+    case kStash:
+      return res;
+    case kHandOff:
+      res.push_back(std::move(frame));
+      RetryStashedFrames(res);
+      return res;
+    case kDrop:
+      return res;
+  }
+
+  return res;
+}
+
+RtpVp9RefFinder::FrameDecision RtpVp9RefFinder::ManageFrameFlexible(
+    RtpFrameObject* frame,
+    const RTPVideoHeaderVP9& codec_header) {
+  if (codec_header.num_ref_pics > EncodedFrame::kMaxFrameReferences) {
+    return kDrop;
+  }
+
+  frame->num_references = codec_header.num_ref_pics;
+  for (size_t i = 0; i < frame->num_references; ++i) {
+    frame->references[i] =
+        Subtract<kFrameIdLength>(frame->Id(), codec_header.pid_diff[i]);
+  }
+
+  FlattenFrameIdAndRefs(frame, codec_header.inter_layer_predicted);
+  return kHandOff;
+}
+
+RtpVp9RefFinder::FrameDecision RtpVp9RefFinder::ManageFrameGof(
+    RtpFrameObject* frame,
+    const RTPVideoHeaderVP9& codec_header,
+    int64_t unwrapped_tl0) {
+  GofInfo* info;
+  if (codec_header.ss_data_available) {
+    if (codec_header.temporal_idx != 0) {
+      RTC_LOG(LS_WARNING) << "Received scalability structure on a non base "
+                             "layer frame. Scalability structure ignored.";
+    } else {
+      if (codec_header.gof.num_frames_in_gof > kMaxVp9FramesInGof) {
+        return kDrop;
+      }
+
+      for (size_t i = 0; i < codec_header.gof.num_frames_in_gof; ++i) {
+        if (codec_header.gof.num_ref_pics[i] > kMaxVp9RefPics) {
+          return kDrop;
+        }
+      }
+
+      GofInfoVP9 gof = codec_header.gof;
+      if (gof.num_frames_in_gof == 0) {
+        RTC_LOG(LS_WARNING) << "Number of frames in GOF is zero. Assume "
+                               "that stream has only one temporal layer.";
+        gof.SetGofInfoVP9(kTemporalStructureMode1);
+      }
+
+      current_ss_idx_ = Add<kMaxGofSaved>(current_ss_idx_, 1);
+      scalability_structures_[current_ss_idx_] = gof;
+      scalability_structures_[current_ss_idx_].pid_start = frame->Id();
+      gof_info_.emplace(
+          unwrapped_tl0,
+          GofInfo(&scalability_structures_[current_ss_idx_], frame->Id()));
+    }
+
+    const auto gof_info_it = gof_info_.find(unwrapped_tl0);
+    if (gof_info_it == gof_info_.end())
+      return kStash;
+
+    info = &gof_info_it->second;
+
+    if (frame->frame_type() == VideoFrameType::kVideoFrameKey) {
+      frame->num_references = 0;
+      FrameReceivedVp9(frame->Id(), info);
+      FlattenFrameIdAndRefs(frame, codec_header.inter_layer_predicted);
+      return kHandOff;
+    }
+  } else if (frame->frame_type() == VideoFrameType::kVideoFrameKey) {
+    if (frame->SpatialIndex() == 0) {
+      RTC_LOG(LS_WARNING) << "Received keyframe without scalability structure";
+      return kDrop;
+    }
+    const auto gof_info_it = gof_info_.find(unwrapped_tl0);
+    if (gof_info_it == gof_info_.end())
+      return kStash;
+
+    info = &gof_info_it->second;
+
+    frame->num_references = 0;
+    FrameReceivedVp9(frame->Id(), info);
+    FlattenFrameIdAndRefs(frame, codec_header.inter_layer_predicted);
+    return kHandOff;
+  } else {
+    auto gof_info_it = gof_info_.find(
+        (codec_header.temporal_idx == 0) ? unwrapped_tl0 - 1 : unwrapped_tl0);
+
+    // Gof info for this frame is not available yet, stash this frame.
+    if (gof_info_it == gof_info_.end())
+      return kStash;
+
+    if (codec_header.temporal_idx == 0) {
+      gof_info_it = gof_info_
+                        .emplace(unwrapped_tl0,
+                                 GofInfo(gof_info_it->second.gof, frame->Id()))
+                        .first;
+    }
+
+    info = &gof_info_it->second;
+  }
+
+  // Clean up info for base layers that are too old.
+  int64_t old_tl0_pic_idx = unwrapped_tl0 - kMaxGofSaved;
+  auto clean_gof_info_to = gof_info_.lower_bound(old_tl0_pic_idx);
+  gof_info_.erase(gof_info_.begin(), clean_gof_info_to);
+
+  FrameReceivedVp9(frame->Id(), info);
+
+  // Make sure we don't miss any frame that could potentially have the
+  // up switch flag set.
+  if (MissingRequiredFrameVp9(frame->Id(), *info))
+    return kStash;
+
+  if (codec_header.temporal_up_switch)
+    up_switch_.emplace(frame->Id(), codec_header.temporal_idx);
+
+  // Clean out old info about up switch frames.
+  uint16_t old_picture_id = Subtract<kFrameIdLength>(frame->Id(), 50);
+  auto up_switch_erase_to = up_switch_.lower_bound(old_picture_id);
+  up_switch_.erase(up_switch_.begin(), up_switch_erase_to);
+
+  size_t diff =
+      ForwardDiff<uint16_t, kFrameIdLength>(info->gof->pid_start, frame->Id());
+  size_t gof_idx = diff % info->gof->num_frames_in_gof;
+
+  if (info->gof->num_ref_pics[gof_idx] > EncodedFrame::kMaxFrameReferences) {
+    return kDrop;
+  }
+  // Populate references according to the scalability structure.
+  frame->num_references = info->gof->num_ref_pics[gof_idx];
+  for (size_t i = 0; i < frame->num_references; ++i) {
+    frame->references[i] =
+        Subtract<kFrameIdLength>(frame->Id(), info->gof->pid_diff[gof_idx][i]);
+
+    // If this is a reference to a frame earlier than the last up switch point,
+    // then ignore this reference.
+    if (UpSwitchInIntervalVp9(frame->Id(), codec_header.temporal_idx,
+                              frame->references[i])) {
+      --frame->num_references;
+    }
+  }
+
+  // Override GOF references.
+  if (!codec_header.inter_pic_predicted) {
+    frame->num_references = 0;
+  }
+
+  FlattenFrameIdAndRefs(frame, codec_header.inter_layer_predicted);
+  return kHandOff;
+}
+
+bool RtpVp9RefFinder::MissingRequiredFrameVp9(uint16_t picture_id,
+                                              const GofInfo& info) {
+  size_t diff =
+      ForwardDiff<uint16_t, kFrameIdLength>(info.gof->pid_start, picture_id);
+  size_t gof_idx = diff % info.gof->num_frames_in_gof;
+  size_t temporal_idx = info.gof->temporal_idx[gof_idx];
+
+  if (temporal_idx >= kMaxTemporalLayers) {
+    RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers
+                        << " temporal "
+                           "layers are supported.";
+    return true;
+  }
+
+  // For every reference this frame has, check if there is a frame missing in
+  // the interval (`ref_pid`, `picture_id`) in any of the lower temporal
+  // layers. If so, we are missing a required frame.
+  uint8_t num_references = info.gof->num_ref_pics[gof_idx];
+  for (size_t i = 0; i < num_references; ++i) {
+    uint16_t ref_pid =
+        Subtract<kFrameIdLength>(picture_id, info.gof->pid_diff[gof_idx][i]);
+    for (size_t l = 0; l < temporal_idx; ++l) {
+      auto missing_frame_it = missing_frames_for_layer_[l].lower_bound(ref_pid);
+      if (missing_frame_it != missing_frames_for_layer_[l].end() &&
+          AheadOf<uint16_t, kFrameIdLength>(picture_id, *missing_frame_it)) {
+        return true;
+      }
+    }
+  }
+  return false;
+}
+
+void RtpVp9RefFinder::FrameReceivedVp9(uint16_t picture_id, GofInfo* info) {
+  int last_picture_id = info->last_picture_id;
+  size_t gof_size = std::min(info->gof->num_frames_in_gof, kMaxVp9FramesInGof);
+
+  // If there is a gap, find which temporal layer the missing frames
+  // belong to and add the frame as missing for that temporal layer.
+  // Otherwise, remove this frame from the set of missing frames.
+  if (AheadOf<uint16_t, kFrameIdLength>(picture_id, last_picture_id)) {
+    size_t diff = ForwardDiff<uint16_t, kFrameIdLength>(info->gof->pid_start,
+                                                        last_picture_id);
+    size_t gof_idx = diff % gof_size;
+
+    last_picture_id = Add<kFrameIdLength>(last_picture_id, 1);
+    while (last_picture_id != picture_id) {
+      gof_idx = (gof_idx + 1) % gof_size;
+      RTC_CHECK(gof_idx < kMaxVp9FramesInGof);
+
+      size_t temporal_idx = info->gof->temporal_idx[gof_idx];
+      if (temporal_idx >= kMaxTemporalLayers) {
+        RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers
+                            << " temporal "
+                               "layers are supported.";
+        return;
+      }
+
+      missing_frames_for_layer_[temporal_idx].insert(last_picture_id);
+      last_picture_id = Add<kFrameIdLength>(last_picture_id, 1);
+    }
+
+    info->last_picture_id = last_picture_id;
+  } else {
+    size_t diff =
+        ForwardDiff<uint16_t, kFrameIdLength>(info->gof->pid_start, picture_id);
+    size_t gof_idx = diff % gof_size;
+    RTC_CHECK(gof_idx < kMaxVp9FramesInGof);
+
+    size_t temporal_idx = info->gof->temporal_idx[gof_idx];
+    if (temporal_idx >= kMaxTemporalLayers) {
+      RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers
+                          << " temporal "
+                             "layers are supported.";
+      return;
+    }
+
+    missing_frames_for_layer_[temporal_idx].erase(picture_id);
+  }
+}
+
+bool RtpVp9RefFinder::UpSwitchInIntervalVp9(uint16_t picture_id,
+                                            uint8_t temporal_idx,
+                                            uint16_t pid_ref) {
+  for (auto up_switch_it = up_switch_.upper_bound(pid_ref);
+       up_switch_it != up_switch_.end() &&
+       AheadOf<uint16_t, kFrameIdLength>(picture_id, up_switch_it->first);
+       ++up_switch_it) {
+    if (up_switch_it->second < temporal_idx)
+      return true;
+  }
+
+  return false;
+}
+
+void RtpVp9RefFinder::RetryStashedFrames(
+    RtpFrameReferenceFinder::ReturnVector& res) {
+  bool complete_frame = false;
+  do {
+    complete_frame = false;
+    for (auto it = stashed_frames_.begin(); it != stashed_frames_.end();) {
+      const RTPVideoHeaderVP9& codec_header = absl::get<RTPVideoHeaderVP9>(
+          it->frame->GetRtpVideoHeader().video_type_header);
+      RTC_DCHECK(!codec_header.flexible_mode);
+      FrameDecision decision =
+          ManageFrameGof(it->frame.get(), codec_header, it->unwrapped_tl0);
+
+      switch (decision) {
+        case kStash:
+          ++it;
+          break;
+        case kHandOff:
+          complete_frame = true;
+          res.push_back(std::move(it->frame));
+          [[fallthrough]];
+        case kDrop:
+          it = stashed_frames_.erase(it);
+      }
+    }
+  } while (complete_frame);
+}
+
+void RtpVp9RefFinder::FlattenFrameIdAndRefs(RtpFrameObject* frame,
+                                            bool inter_layer_predicted) {
+  for (size_t i = 0; i < frame->num_references; ++i) {
+    frame->references[i] =
+        unwrapper_.Unwrap(frame->references[i]) * kMaxSpatialLayers +
+        *frame->SpatialIndex();
+  }
+  frame->SetId(unwrapper_.Unwrap(frame->Id()) * kMaxSpatialLayers +
+               *frame->SpatialIndex());
+
+  if (inter_layer_predicted &&
+      frame->num_references + 1 <= EncodedFrame::kMaxFrameReferences) {
+    frame->references[frame->num_references] = frame->Id() - 1;
+    ++frame->num_references;
+  }
+}
+
+void RtpVp9RefFinder::ClearTo(uint16_t seq_num) {
+  auto it = stashed_frames_.begin();
+  while (it != stashed_frames_.end()) {
+    if (AheadOf<uint16_t>(seq_num, it->frame->first_seq_num())) {
+      it = stashed_frames_.erase(it);
+    } else {
+      ++it;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder.h b/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder.h
new file mode 100644
index 0000000000..ea5e319fc8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder.h
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_RTP_VP9_REF_FINDER_H_
+#define MODULES_VIDEO_CODING_RTP_VP9_REF_FINDER_H_
+
+#include <deque>
+#include <map>
+#include <memory>
+#include <set>
+
+#include "absl/container/inlined_vector.h"
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+class RtpVp9RefFinder {
+ public:
+  RtpVp9RefFinder() = default;
+
+  RtpFrameReferenceFinder::ReturnVector ManageFrame(
+      std::unique_ptr<RtpFrameObject> frame);
+  void ClearTo(uint16_t seq_num);
+
+ private:
+  static constexpr int kFrameIdLength = 1 << 15;
+  static constexpr int kMaxGofSaved = 50;
+  static constexpr int kMaxLayerInfo = 50;
+  static constexpr int kMaxNotYetReceivedFrames = 100;
+  static constexpr int kMaxStashedFrames = 100;
+  static constexpr int kMaxTemporalLayers = 5;
+
+  enum FrameDecision { kStash, kHandOff, kDrop };
+
+  struct GofInfo {
+    GofInfo(GofInfoVP9* gof, uint16_t last_picture_id)
+        : gof(gof), last_picture_id(last_picture_id) {}
+    GofInfoVP9* gof;
+    uint16_t last_picture_id;
+  };
+
+  struct UnwrappedTl0Frame {
+    int64_t unwrapped_tl0;
+    std::unique_ptr<RtpFrameObject> frame;
+  };
+
+  FrameDecision ManageFrameFlexible(RtpFrameObject* frame,
+                                    const RTPVideoHeaderVP9& vp9_header);
+  FrameDecision ManageFrameGof(RtpFrameObject* frame,
+                               const RTPVideoHeaderVP9& vp9_header,
+                               int64_t unwrapped_tl0);
+  void RetryStashedFrames(RtpFrameReferenceFinder::ReturnVector& res);
+
+  bool MissingRequiredFrameVp9(uint16_t picture_id, const GofInfo& info);
+
+  void FrameReceivedVp9(uint16_t picture_id, GofInfo* info);
+  bool UpSwitchInIntervalVp9(uint16_t picture_id,
+                             uint8_t temporal_idx,
+                             uint16_t pid_ref);
+
+  void FlattenFrameIdAndRefs(RtpFrameObject* frame, bool inter_layer_predicted);
+
+  // Frames that have been fully received but didn't have all the information
+  // needed to determine their references.
+  std::deque<UnwrappedTl0Frame> stashed_frames_;
+
+  // Where the current scalability structure is in the
+  // `scalability_structures_` array.
+  uint8_t current_ss_idx_ = 0;
+
+  // Holds received scalability structures.
+  std::array<GofInfoVP9, kMaxGofSaved> scalability_structures_;
+
+  // Holds the the Gof information for a given unwrapped TL0 picture index.
+  std::map<int64_t, GofInfo> gof_info_;
+
+  // Keep track of which picture id and which temporal layer that had the
+  // up switch flag set.
+  std::map<uint16_t, uint8_t, DescendingSeqNumComp<uint16_t, kFrameIdLength>>
+      up_switch_;
+
+  // For every temporal layer, keep a set of which frames that are missing.
+  std::array<std::set<uint16_t, DescendingSeqNumComp<uint16_t, kFrameIdLength>>,
+             kMaxTemporalLayers>
+      missing_frames_for_layer_;
+
+  // Unwrapper used to unwrap VP8/VP9 streams which have their picture id
+  // specified.
+  SeqNumUnwrapper<uint16_t, kFrameIdLength> unwrapper_;
+
+  SeqNumUnwrapper<uint8_t> tl0_unwrapper_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_RTP_VP9_REF_FINDER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder_unittest.cc b/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder_unittest.cc
new file mode 100644
index 0000000000..51fae50902
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder_unittest.cc
@@ -0,0 +1,637 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <utility>
+#include <vector>
+
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/rtp_vp9_ref_finder.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::Contains;
+using ::testing::Matcher;
+using ::testing::MatcherInterface;
+using ::testing::Matches;
+using ::testing::MatchResultListener;
+using ::testing::Pointee;
+using ::testing::Property;
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAreArray;
+
+namespace webrtc {
+
+namespace {
+class Frame {
+ public:
+  Frame& SeqNum(uint16_t start, uint16_t end) {
+    seq_num_start = start;
+    seq_num_end = end;
+    return *this;
+  }
+
+  Frame& AsKeyFrame(bool is_keyframe = true) {
+    keyframe = is_keyframe;
+    return *this;
+  }
+
+  Frame& Pid(int pid) {
+    picture_id = pid;
+    return *this;
+  }
+
+  Frame& SidAndTid(int sid, int tid) {
+    spatial_id = sid;
+    temporal_id = tid;
+    return *this;
+  }
+
+  Frame& Tl0(int tl0) {
+    tl0_idx = tl0;
+    return *this;
+  }
+
+  Frame& AsUpswitch(bool is_up = true) {
+    up_switch = is_up;
+    return *this;
+  }
+
+  Frame& AsInterLayer(bool is_inter_layer = true) {
+    inter_layer = is_inter_layer;
+    return *this;
+  }
+
+  Frame& NotAsInterPic(bool is_inter_pic = false) {
+    inter_pic = is_inter_pic;
+    return *this;
+  }
+
+  Frame& Gof(GofInfoVP9* ss) {
+    scalability_structure = ss;
+    return *this;
+  }
+
+  Frame& FlexRefs(const std::vector<uint8_t>& refs) {
+    flex_refs = refs;
+    return *this;
+  }
+
+  operator std::unique_ptr<RtpFrameObject>() {
+    RTPVideoHeaderVP9 vp9_header{};
+    vp9_header.picture_id = *picture_id;
+    vp9_header.temporal_idx = *temporal_id;
+    vp9_header.spatial_idx = *spatial_id;
+    if (tl0_idx.has_value()) {
+      RTC_DCHECK(flex_refs.empty());
+      vp9_header.flexible_mode = false;
+      vp9_header.tl0_pic_idx = *tl0_idx;
+    } else {
+      vp9_header.flexible_mode = true;
+      vp9_header.num_ref_pics = flex_refs.size();
+      for (size_t i = 0; i < flex_refs.size(); ++i) {
+        vp9_header.pid_diff[i] = flex_refs.at(i);
+      }
+    }
+    vp9_header.temporal_up_switch = up_switch;
+    vp9_header.inter_layer_predicted = inter_layer;
+    vp9_header.inter_pic_predicted = inter_pic && !keyframe;
+    if (scalability_structure != nullptr) {
+      vp9_header.ss_data_available = true;
+      vp9_header.gof = *scalability_structure;
+    }
+
+    RTPVideoHeader video_header;
+    video_header.frame_type = keyframe ? VideoFrameType::kVideoFrameKey
+                                       : VideoFrameType::kVideoFrameDelta;
+    video_header.video_type_header = vp9_header;
+    // clang-format off
+    return std::make_unique<RtpFrameObject>(
+        seq_num_start,
+        seq_num_end,
+        /*markerBit=*/true,
+        /*times_nacked=*/0,
+        /*first_packet_received_time=*/0,
+        /*last_packet_received_time=*/0,
+        /*rtp_timestamp=*/0,
+        /*ntp_time_ms=*/0,
+        VideoSendTiming(),
+        /*payload_type=*/0,
+        kVideoCodecVP9,
+        kVideoRotation_0,
+        VideoContentType::UNSPECIFIED,
+        video_header,
+        /*color_space=*/absl::nullopt,
+        RtpPacketInfos(),
+        EncodedImageBuffer::Create(/*size=*/0));
+    // clang-format on
+  }
+
+ private:
+  uint16_t seq_num_start = 0;
+  uint16_t seq_num_end = 0;
+  bool keyframe = false;
+  absl::optional<int> picture_id;
+  absl::optional<int> spatial_id;
+  absl::optional<int> temporal_id;
+  absl::optional<int> tl0_idx;
+  bool up_switch = false;
+  bool inter_layer = false;
+  bool inter_pic = true;
+  GofInfoVP9* scalability_structure = nullptr;
+  std::vector<uint8_t> flex_refs;
+};
+
+using FrameVector = std::vector<std::unique_ptr<EncodedFrame>>;
+
+// Would have been nice to use the MATCHER_P3 macro instead, but when used it
+// fails to infer the type of the vector if not explicitly given in the
+class HasFrameMatcher : public MatcherInterface<const FrameVector&> {
+ public:
+  explicit HasFrameMatcher(int64_t frame_id,
+                           const std::vector<int64_t>& expected_refs)
+      : frame_id_(frame_id),
+        expected_refs_(expected_refs) {}
+
+  bool MatchAndExplain(const FrameVector& frames,
+                       MatchResultListener* result_listener) const override {
+    auto it = std::find_if(frames.begin(), frames.end(),
+                           [this](const std::unique_ptr<EncodedFrame>& f) {
+                             return f->Id() == frame_id_;
+                           });
+    if (it == frames.end()) {
+      if (result_listener->IsInterested()) {
+        *result_listener << "No frame with frame_id:" << frame_id_;
+      }
+      return false;
+    }
+
+    rtc::ArrayView<int64_t> actual_refs((*it)->references,
+                                        (*it)->num_references);
+    if (!Matches(UnorderedElementsAreArray(expected_refs_))(actual_refs)) {
+      if (result_listener->IsInterested()) {
+        *result_listener << "Frame with frame_id:" << frame_id_ << " and "
+                         << actual_refs.size() << " references { ";
+        for (auto r : actual_refs) {
+          *result_listener << r << " ";
+        }
+        *result_listener << "}";
+      }
+      return false;
+    }
+
+    return true;
+  }
+
+  void DescribeTo(std::ostream* os) const override {
+    *os << "frame with frame_id:" << frame_id_ << " and "
+        << expected_refs_.size() << " references { ";
+    for (auto r : expected_refs_) {
+      *os << r << " ";
+    }
+    *os << "}";
+  }
+
+ private:
+  const int64_t frame_id_;
+  const std::vector<int64_t> expected_refs_;
+};
+
+}  // namespace
+
+class RtpVp9RefFinderTest : public ::testing::Test {
+ protected:
+  RtpVp9RefFinderTest() : ref_finder_(std::make_unique<RtpVp9RefFinder>()) {}
+
+  void Insert(std::unique_ptr<RtpFrameObject> frame) {
+    for (auto& f : ref_finder_->ManageFrame(std::move(frame))) {
+      frames_.push_back(std::move(f));
+    }
+  }
+
+  std::unique_ptr<RtpVp9RefFinder> ref_finder_;
+  FrameVector frames_;
+};
+
+Matcher<const FrameVector&> HasFrameWithIdAndRefs(int64_t frame_id,
+                                                  std::vector<int64_t> refs) {
+  return MakeMatcher(new HasFrameMatcher(frame_id, refs));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofInsertOneFrame) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode1);
+
+  Insert(Frame().Pid(1).SidAndTid(0, 0).Tl0(0).AsKeyFrame().Gof(&ss));
+
+  EXPECT_EQ(frames_.size(), 1UL);
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofTemporalLayers_0) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode1);  // Only 1 spatial layer.
+
+  Insert(Frame().Pid(1).SidAndTid(0, 0).Tl0(0).AsKeyFrame().Gof(&ss));
+  Insert(Frame().Pid(2).SidAndTid(0, 0).Tl0(1));
+
+  EXPECT_EQ(frames_.size(), 2UL);
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {5}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofSpatialLayers_2) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode1);  // Only 1 spatial layer.
+
+  Insert(Frame().Pid(1).SidAndTid(0, 0).Tl0(0).AsKeyFrame().Gof(&ss));
+  Insert(Frame().Pid(2).SidAndTid(0, 0).Tl0(1));
+  Insert(Frame().Pid(2).SidAndTid(1, 0).Tl0(1).NotAsInterPic());
+  Insert(Frame().Pid(3).SidAndTid(0, 0).Tl0(2));
+  Insert(Frame().Pid(3).SidAndTid(1, 0).Tl0(2));
+
+  EXPECT_EQ(frames_.size(), 5UL);
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {5}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(11, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(15, {10}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(16, {11}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofTemporalLayersReordered_0) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode1);  // Only 1 spatial layer.
+
+  Insert(Frame().Pid(2).SidAndTid(0, 0).Tl0(1));
+  Insert(Frame().Pid(2).SidAndTid(1, 0).Tl0(1).NotAsInterPic());
+  Insert(Frame().Pid(1).SidAndTid(0, 0).Tl0(0).AsKeyFrame().Gof(&ss));
+  Insert(Frame().Pid(3).SidAndTid(0, 0).Tl0(2));
+  Insert(Frame().Pid(3).SidAndTid(1, 0).Tl0(2));
+  Insert(Frame().Pid(4).SidAndTid(0, 0).Tl0(3));
+  Insert(Frame().Pid(5).SidAndTid(1, 0).Tl0(4));
+  Insert(Frame().Pid(4).SidAndTid(1, 0).Tl0(3));
+  Insert(Frame().Pid(5).SidAndTid(0, 0).Tl0(4));
+
+  EXPECT_EQ(frames_.size(), 9UL);
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {5}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(11, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(15, {10}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(16, {11}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(20, {15}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(21, {16}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(25, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(26, {21}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofSkipFramesTemporalLayers_01) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode2);  // 0101 pattern
+
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(1).SidAndTid(0, 1).Tl0(0));
+  // Skip GOF with tl0 1
+  Insert(Frame().Pid(4).SidAndTid(0, 0).Tl0(2).AsKeyFrame().Gof(&ss));
+  Insert(Frame().Pid(5).SidAndTid(0, 1).Tl0(2));
+  // Skip GOF with tl0 3
+  // Skip GOF with tl0 4
+  Insert(Frame().Pid(10).SidAndTid(0, 0).Tl0(5).Gof(&ss));
+  Insert(Frame().Pid(11).SidAndTid(0, 1).Tl0(5));
+
+  ASSERT_EQ(6UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(20, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(25, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(50, {40}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(55, {50}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofSkipFramesTemporalLayers_0212) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode3);  // 02120212 pattern
+
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(1).SidAndTid(0, 2).Tl0(0));
+  Insert(Frame().Pid(2).SidAndTid(0, 1).Tl0(0));
+  Insert(Frame().Pid(3).SidAndTid(0, 2).Tl0(0));
+
+  ASSERT_EQ(4UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(15, {10}));
+
+  // Skip frames with tl0 = 1
+
+  Insert(Frame().Pid(8).SidAndTid(0, 0).Tl0(2).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(9).SidAndTid(0, 2).Tl0(2));
+  Insert(Frame().Pid(10).SidAndTid(0, 1).Tl0(2));
+  Insert(Frame().Pid(11).SidAndTid(0, 2).Tl0(2));
+
+  ASSERT_EQ(8UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(40, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(45, {40}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(50, {40}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(55, {50}));
+
+  // Now insert frames with tl0 = 1
+  Insert(Frame().Pid(4).SidAndTid(0, 0).Tl0(1).AsKeyFrame().Gof(&ss));
+  Insert(Frame().Pid(7).SidAndTid(0, 2).Tl0(1));
+
+  ASSERT_EQ(9UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(20, {}));
+
+  Insert(Frame().Pid(5).SidAndTid(0, 2).Tl0(1));
+  Insert(Frame().Pid(6).SidAndTid(0, 1).Tl0(1));
+
+  ASSERT_EQ(12UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(25, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(30, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(35, {30}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofTemporalLayers_01) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode2);  // 0101 pattern
+
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(1).SidAndTid(0, 1).Tl0(0));
+  Insert(Frame().Pid(2).SidAndTid(0, 0).Tl0(1));
+  Insert(Frame().Pid(3).SidAndTid(0, 1).Tl0(1));
+
+  ASSERT_EQ(4UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(15, {10}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofTemporalLayersReordered_01) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode2);  // 01 pattern
+
+  Insert(Frame().Pid(1).SidAndTid(0, 1).Tl0(0));
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(2).SidAndTid(0, 0).Tl0(1));
+  Insert(Frame().Pid(4).SidAndTid(0, 0).Tl0(2));
+  Insert(Frame().Pid(3).SidAndTid(0, 1).Tl0(1));
+  Insert(Frame().Pid(5).SidAndTid(0, 1).Tl0(2));
+  Insert(Frame().Pid(7).SidAndTid(0, 1).Tl0(3));
+  Insert(Frame().Pid(6).SidAndTid(0, 0).Tl0(3));
+  Insert(Frame().Pid(8).SidAndTid(0, 0).Tl0(4));
+  Insert(Frame().Pid(9).SidAndTid(0, 1).Tl0(4));
+
+  ASSERT_EQ(10UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(15, {10}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(20, {10}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(25, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(30, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(35, {30}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(40, {30}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(45, {40}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofTemporalLayers_0212) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode3);  // 0212 pattern
+
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(1).SidAndTid(0, 2).Tl0(0));
+  Insert(Frame().Pid(2).SidAndTid(0, 1).Tl0(0));
+  Insert(Frame().Pid(3).SidAndTid(0, 2).Tl0(0));
+  Insert(Frame().Pid(4).SidAndTid(0, 0).Tl0(1));
+  Insert(Frame().Pid(5).SidAndTid(0, 2).Tl0(1));
+  Insert(Frame().Pid(6).SidAndTid(0, 1).Tl0(1));
+  Insert(Frame().Pid(7).SidAndTid(0, 2).Tl0(1));
+
+  ASSERT_EQ(8UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(15, {10}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(20, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(25, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(30, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(35, {30}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofTemporalLayersReordered_0212) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode3);  // 0212 pattern
+
+  Insert(Frame().Pid(2).SidAndTid(0, 1).Tl0(0));
+  Insert(Frame().Pid(1).SidAndTid(0, 2).Tl0(0));
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(3).SidAndTid(0, 2).Tl0(0));
+  Insert(Frame().Pid(6).SidAndTid(0, 1).Tl0(1));
+  Insert(Frame().Pid(5).SidAndTid(0, 2).Tl0(1));
+  Insert(Frame().Pid(4).SidAndTid(0, 0).Tl0(1));
+  Insert(Frame().Pid(9).SidAndTid(0, 2).Tl0(2));
+  Insert(Frame().Pid(7).SidAndTid(0, 2).Tl0(1));
+  Insert(Frame().Pid(8).SidAndTid(0, 0).Tl0(2));
+  Insert(Frame().Pid(11).SidAndTid(0, 2).Tl0(2));
+  Insert(Frame().Pid(10).SidAndTid(0, 1).Tl0(2));
+
+  ASSERT_EQ(12UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(15, {10}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(20, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(25, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(30, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(35, {30}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(40, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(45, {40}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(50, {40}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(55, {50}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofTemporalLayersReordered_01_0212) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode2);  // 01 pattern
+
+  Insert(Frame().Pid(1).SidAndTid(0, 1).Tl0(0));
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(3).SidAndTid(0, 1).Tl0(1));
+  Insert(Frame().Pid(6).SidAndTid(0, 1).Tl0(2));
+  ss.SetGofInfoVP9(kTemporalStructureMode3);  // 0212 pattern
+  Insert(Frame().Pid(4).SidAndTid(0, 0).Tl0(2).Gof(&ss));
+  Insert(Frame().Pid(2).SidAndTid(0, 0).Tl0(1));
+  Insert(Frame().Pid(5).SidAndTid(0, 2).Tl0(2));
+  Insert(Frame().Pid(8).SidAndTid(0, 0).Tl0(3));
+  Insert(Frame().Pid(10).SidAndTid(0, 1).Tl0(3));
+  Insert(Frame().Pid(7).SidAndTid(0, 2).Tl0(2));
+  Insert(Frame().Pid(11).SidAndTid(0, 2).Tl0(3));
+  Insert(Frame().Pid(9).SidAndTid(0, 2).Tl0(3));
+
+  ASSERT_EQ(12UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(15, {10}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(20, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(25, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(30, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(35, {30}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(40, {20}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(45, {40}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(50, {40}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(55, {50}));
+}
+
+TEST_F(RtpVp9RefFinderTest, FlexibleModeOneFrame) {
+  Insert(Frame().Pid(0).SidAndTid(0, 0).AsKeyFrame());
+
+  ASSERT_EQ(1UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+}
+
+TEST_F(RtpVp9RefFinderTest, FlexibleModeTwoSpatialLayers) {
+  Insert(Frame().Pid(0).SidAndTid(0, 0).AsKeyFrame());
+  Insert(Frame().Pid(0).SidAndTid(1, 0).AsKeyFrame().AsInterLayer());
+  Insert(Frame().Pid(1).SidAndTid(1, 0).FlexRefs({1}));
+  Insert(Frame().Pid(2).SidAndTid(0, 0).FlexRefs({2}));
+  Insert(Frame().Pid(2).SidAndTid(1, 0).FlexRefs({1}));
+  Insert(Frame().Pid(3).SidAndTid(1, 0).FlexRefs({1}));
+  Insert(Frame().Pid(4).SidAndTid(0, 0).FlexRefs({2}));
+  Insert(Frame().Pid(4).SidAndTid(1, 0).FlexRefs({1}));
+
+  ASSERT_EQ(8UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(6, {1}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(11, {6}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(16, {11}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(20, {10}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(21, {16}));
+}
+
+TEST_F(RtpVp9RefFinderTest, FlexibleModeTwoSpatialLayersReordered) {
+  Insert(Frame().Pid(0).SidAndTid(1, 0).AsKeyFrame().AsInterLayer());
+  Insert(Frame().Pid(1).SidAndTid(1, 0).FlexRefs({1}));
+  Insert(Frame().Pid(0).SidAndTid(0, 0).AsKeyFrame());
+  Insert(Frame().Pid(2).SidAndTid(1, 0).FlexRefs({1}));
+  Insert(Frame().Pid(3).SidAndTid(1, 0).FlexRefs({1}));
+  Insert(Frame().Pid(2).SidAndTid(0, 0).FlexRefs({2}));
+  Insert(Frame().Pid(4).SidAndTid(1, 0).FlexRefs({1}));
+  Insert(Frame().Pid(4).SidAndTid(0, 0).FlexRefs({2}));
+
+  ASSERT_EQ(8UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(1, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(6, {1}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(10, {0}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(11, {6}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(16, {11}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(20, {10}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(21, {16}));
+}
+
+TEST_F(RtpVp9RefFinderTest, WrappingFlexReference) {
+  Insert(Frame().Pid(0).SidAndTid(0, 0).FlexRefs({1}));
+
+  ASSERT_EQ(1UL, frames_.size());
+  const EncodedFrame& frame = *frames_[0];
+
+  ASSERT_EQ(frame.Id() - frame.references[0], 5);
+}
+
+TEST_F(RtpVp9RefFinderTest, GofPidJump) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode3);
+
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(1000).SidAndTid(0, 0).Tl0(1));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofTl0Jump) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode3);
+
+  Insert(Frame()
+             .Pid(0)
+             .SidAndTid(0, 0)
+             .Tl0(125)
+             .AsUpswitch()
+             .AsKeyFrame()
+             .NotAsInterPic()
+             .Gof(&ss));
+  Insert(Frame().Pid(1).SidAndTid(0, 0).Tl0(0).Gof(&ss));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofTidTooHigh) {
+  const int kMaxTemporalLayers = 5;
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode2);
+  ss.temporal_idx[1] = kMaxTemporalLayers;
+
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(1).SidAndTid(0, 0).Tl0(1));
+
+  ASSERT_EQ(1UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+}
+
+TEST_F(RtpVp9RefFinderTest, GofZeroFrames) {
+  GofInfoVP9 ss;
+  ss.num_frames_in_gof = 0;
+
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0).AsKeyFrame().NotAsInterPic().Gof(
+      &ss));
+  Insert(Frame().Pid(1).SidAndTid(0, 0).Tl0(1));
+
+  ASSERT_EQ(2UL, frames_.size());
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(0, {}));
+  EXPECT_THAT(frames_, HasFrameWithIdAndRefs(5, {0}));
+}
+
+TEST_F(RtpVp9RefFinderTest, SpatialIndex) {
+  Insert(Frame().Pid(0).SidAndTid(0, 0).AsKeyFrame());
+  Insert(Frame().Pid(0).SidAndTid(1, 0).AsKeyFrame());
+  Insert(Frame().Pid(0).SidAndTid(2, 0).AsKeyFrame());
+
+  ASSERT_EQ(3UL, frames_.size());
+  EXPECT_THAT(frames_,
+              Contains(Pointee(Property(&EncodedFrame::SpatialIndex, 0))));
+  EXPECT_THAT(frames_,
+              Contains(Pointee(Property(&EncodedFrame::SpatialIndex, 1))));
+  EXPECT_THAT(frames_,
+              Contains(Pointee(Property(&EncodedFrame::SpatialIndex, 2))));
+}
+
+TEST_F(RtpVp9RefFinderTest, StashedFramesDoNotWrapTl0Backwards) {
+  GofInfoVP9 ss;
+  ss.SetGofInfoVP9(kTemporalStructureMode1);
+
+  Insert(Frame().Pid(0).SidAndTid(0, 0).Tl0(0));
+  EXPECT_THAT(frames_, SizeIs(0));
+
+  Insert(Frame().Pid(128).SidAndTid(0, 0).Tl0(128).AsKeyFrame().Gof(&ss));
+  EXPECT_THAT(frames_, SizeIs(1));
+  Insert(Frame().Pid(129).SidAndTid(0, 0).Tl0(129));
+  EXPECT_THAT(frames_, SizeIs(2));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/session_info.cc b/third_party/libwebrtc/modules/video_coding/session_info.cc
new file mode 100644
index 0000000000..e31b8b1d25
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/session_info.cc
@@ -0,0 +1,540 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/session_info.h"
+
+#include <string.h>
+
+#include <vector>
+
+#include "absl/types/variant.h"
+#include "modules/include/module_common_types.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
+#include "modules/video_coding/jitter_buffer_common.h"
+#include "modules/video_coding/packet.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+uint16_t BufferToUWord16(const uint8_t* dataBuffer) {
+  return (dataBuffer[0] << 8) | dataBuffer[1];
+}
+
+}  // namespace
+
+VCMSessionInfo::VCMSessionInfo()
+    : complete_(false),
+      frame_type_(VideoFrameType::kVideoFrameDelta),
+      packets_(),
+      empty_seq_num_low_(-1),
+      empty_seq_num_high_(-1),
+      first_packet_seq_num_(-1),
+      last_packet_seq_num_(-1) {}
+
+VCMSessionInfo::~VCMSessionInfo() {}
+
+void VCMSessionInfo::UpdateDataPointers(const uint8_t* old_base_ptr,
+                                        const uint8_t* new_base_ptr) {
+  for (PacketIterator it = packets_.begin(); it != packets_.end(); ++it)
+    if ((*it).dataPtr != NULL) {
+      RTC_DCHECK(old_base_ptr != NULL && new_base_ptr != NULL);
+      (*it).dataPtr = new_base_ptr + ((*it).dataPtr - old_base_ptr);
+    }
+}
+
+int VCMSessionInfo::LowSequenceNumber() const {
+  if (packets_.empty())
+    return empty_seq_num_low_;
+  return packets_.front().seqNum;
+}
+
+int VCMSessionInfo::HighSequenceNumber() const {
+  if (packets_.empty())
+    return empty_seq_num_high_;
+  if (empty_seq_num_high_ == -1)
+    return packets_.back().seqNum;
+  return LatestSequenceNumber(packets_.back().seqNum, empty_seq_num_high_);
+}
+
+int VCMSessionInfo::PictureId() const {
+  if (packets_.empty())
+    return kNoPictureId;
+  if (packets_.front().video_header.codec == kVideoCodecVP8) {
+    return absl::get<RTPVideoHeaderVP8>(
+               packets_.front().video_header.video_type_header)
+        .pictureId;
+  } else if (packets_.front().video_header.codec == kVideoCodecVP9) {
+    return absl::get<RTPVideoHeaderVP9>(
+               packets_.front().video_header.video_type_header)
+        .picture_id;
+  } else {
+    return kNoPictureId;
+  }
+}
+
+int VCMSessionInfo::TemporalId() const {
+  if (packets_.empty())
+    return kNoTemporalIdx;
+  if (packets_.front().video_header.codec == kVideoCodecVP8) {
+    return absl::get<RTPVideoHeaderVP8>(
+               packets_.front().video_header.video_type_header)
+        .temporalIdx;
+  } else if (packets_.front().video_header.codec == kVideoCodecVP9) {
+    return absl::get<RTPVideoHeaderVP9>(
+               packets_.front().video_header.video_type_header)
+        .temporal_idx;
+  } else {
+    return kNoTemporalIdx;
+  }
+}
+
+bool VCMSessionInfo::LayerSync() const {
+  if (packets_.empty())
+    return false;
+  if (packets_.front().video_header.codec == kVideoCodecVP8) {
+    return absl::get<RTPVideoHeaderVP8>(
+               packets_.front().video_header.video_type_header)
+        .layerSync;
+  } else if (packets_.front().video_header.codec == kVideoCodecVP9) {
+    return absl::get<RTPVideoHeaderVP9>(
+               packets_.front().video_header.video_type_header)
+        .temporal_up_switch;
+  } else {
+    return false;
+  }
+}
+
+int VCMSessionInfo::Tl0PicId() const {
+  if (packets_.empty())
+    return kNoTl0PicIdx;
+  if (packets_.front().video_header.codec == kVideoCodecVP8) {
+    return absl::get<RTPVideoHeaderVP8>(
+               packets_.front().video_header.video_type_header)
+        .tl0PicIdx;
+  } else if (packets_.front().video_header.codec == kVideoCodecVP9) {
+    return absl::get<RTPVideoHeaderVP9>(
+               packets_.front().video_header.video_type_header)
+        .tl0_pic_idx;
+  } else {
+    return kNoTl0PicIdx;
+  }
+}
+
+std::vector<NaluInfo> VCMSessionInfo::GetNaluInfos() const {
+  if (packets_.empty() ||
+      packets_.front().video_header.codec != kVideoCodecH264)
+    return std::vector<NaluInfo>();
+  std::vector<NaluInfo> nalu_infos;
+  for (const VCMPacket& packet : packets_) {
+    const auto& h264 =
+        absl::get<RTPVideoHeaderH264>(packet.video_header.video_type_header);
+    for (size_t i = 0; i < h264.nalus_length; ++i) {
+      nalu_infos.push_back(h264.nalus[i]);
+    }
+  }
+  return nalu_infos;
+}
+
+void VCMSessionInfo::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
+  if (packets_.empty())
+    return;
+
+  auto* vp9_header = absl::get_if<RTPVideoHeaderVP9>(
+      &packets_.front().video_header.video_type_header);
+  if (!vp9_header || vp9_header->flexible_mode)
+    return;
+
+  vp9_header->temporal_idx = gof_info.temporal_idx[idx];
+  vp9_header->temporal_up_switch = gof_info.temporal_up_switch[idx];
+  vp9_header->num_ref_pics = gof_info.num_ref_pics[idx];
+  for (uint8_t i = 0; i < gof_info.num_ref_pics[idx]; ++i) {
+    vp9_header->pid_diff[i] = gof_info.pid_diff[idx][i];
+  }
+}
+
+void VCMSessionInfo::Reset() {
+  complete_ = false;
+  frame_type_ = VideoFrameType::kVideoFrameDelta;
+  packets_.clear();
+  empty_seq_num_low_ = -1;
+  empty_seq_num_high_ = -1;
+  first_packet_seq_num_ = -1;
+  last_packet_seq_num_ = -1;
+}
+
+size_t VCMSessionInfo::SessionLength() const {
+  size_t length = 0;
+  for (PacketIteratorConst it = packets_.begin(); it != packets_.end(); ++it)
+    length += (*it).sizeBytes;
+  return length;
+}
+
+int VCMSessionInfo::NumPackets() const {
+  return packets_.size();
+}
+
+size_t VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
+                                    PacketIterator packet_it) {
+  VCMPacket& packet = *packet_it;
+  PacketIterator it;
+
+  // Calculate the offset into the frame buffer for this packet.
+  size_t offset = 0;
+  for (it = packets_.begin(); it != packet_it; ++it)
+    offset += (*it).sizeBytes;
+
+  // Set the data pointer to pointing to the start of this packet in the
+  // frame buffer.
+  const uint8_t* packet_buffer = packet.dataPtr;
+  packet.dataPtr = frame_buffer + offset;
+
+  // We handle H.264 STAP-A packets in a special way as we need to remove the
+  // two length bytes between each NAL unit, and potentially add start codes.
+  // TODO(pbos): Remove H264 parsing from this step and use a fragmentation
+  // header supplied by the H264 depacketizer.
+  const size_t kH264NALHeaderLengthInBytes = 1;
+  const size_t kLengthFieldLength = 2;
+  const auto* h264 =
+      absl::get_if<RTPVideoHeaderH264>(&packet.video_header.video_type_header);
+  if (h264 && h264->packetization_type == kH264StapA) {
+    size_t required_length = 0;
+    const uint8_t* nalu_ptr = packet_buffer + kH264NALHeaderLengthInBytes;
+    // Must check that incoming data length doesn't extend past end of buffer.
+    // We allow for 100 bytes of expansion due to startcodes being longer than
+    // length fields.
+    while (nalu_ptr + kLengthFieldLength <= packet_buffer + packet.sizeBytes) {
+      size_t length = BufferToUWord16(nalu_ptr);
+      if (nalu_ptr + kLengthFieldLength + length <= packet_buffer + packet.sizeBytes) {
+        required_length +=
+          length + (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
+        nalu_ptr += kLengthFieldLength + length;
+      } else {
+        // Something is very wrong!
+        RTC_LOG(LS_ERROR) << "Failed to insert packet due to corrupt H264 STAP-A";
+        return 0;
+      }
+    }
+    ShiftSubsequentPackets(packet_it, required_length);
+    nalu_ptr = packet_buffer + kH264NALHeaderLengthInBytes;
+    uint8_t* frame_buffer_ptr = frame_buffer + offset;
+    // we already know we won't go past end-of-buffer
+    while (nalu_ptr + kLengthFieldLength <= packet_buffer + packet.sizeBytes) {
+      size_t length = BufferToUWord16(nalu_ptr);
+      nalu_ptr += kLengthFieldLength;
+      frame_buffer_ptr += Insert(nalu_ptr, length, packet.insertStartCode,
+                                 const_cast<uint8_t*>(frame_buffer_ptr));
+      nalu_ptr += length;
+    }
+    packet.sizeBytes = required_length;
+    return packet.sizeBytes;
+  }
+  ShiftSubsequentPackets(
+      packet_it, packet.sizeBytes +
+                     (packet.insertStartCode ? kH264StartCodeLengthBytes : 0));
+
+  packet.sizeBytes =
+      Insert(packet_buffer, packet.sizeBytes, packet.insertStartCode,
+             const_cast<uint8_t*>(packet.dataPtr));
+  return packet.sizeBytes;
+}
+
+size_t VCMSessionInfo::Insert(const uint8_t* buffer,
+                              size_t length,
+                              bool insert_start_code,
+                              uint8_t* frame_buffer) {
+  if (!buffer || !frame_buffer) {
+    return 0;
+  }
+  if (insert_start_code) {
+    const unsigned char startCode[] = {0, 0, 0, 1};
+    memcpy(frame_buffer, startCode, kH264StartCodeLengthBytes);
+  }
+  memcpy(frame_buffer + (insert_start_code ? kH264StartCodeLengthBytes : 0),
+         buffer, length);
+  length += (insert_start_code ? kH264StartCodeLengthBytes : 0);
+
+  return length;
+}
+
+void VCMSessionInfo::ShiftSubsequentPackets(PacketIterator it,
+                                            int steps_to_shift) {
+  ++it;
+  if (it == packets_.end())
+    return;
+  uint8_t* first_packet_ptr = const_cast<uint8_t*>((*it).dataPtr);
+  int shift_length = 0;
+  // Calculate the total move length and move the data pointers in advance.
+  for (; it != packets_.end(); ++it) {
+    shift_length += (*it).sizeBytes;
+    if ((*it).dataPtr != NULL)
+      (*it).dataPtr += steps_to_shift;
+  }
+  memmove(first_packet_ptr + steps_to_shift, first_packet_ptr, shift_length);
+}
+
+void VCMSessionInfo::UpdateCompleteSession() {
+  if (HaveFirstPacket() && HaveLastPacket()) {
+    // Do we have all the packets in this session?
+    bool complete_session = true;
+    PacketIterator it = packets_.begin();
+    PacketIterator prev_it = it;
+    ++it;
+    for (; it != packets_.end(); ++it) {
+      if (!InSequence(it, prev_it)) {
+        complete_session = false;
+        break;
+      }
+      prev_it = it;
+    }
+    complete_ = complete_session;
+  }
+}
+
+bool VCMSessionInfo::complete() const {
+  return complete_;
+}
+
+// Find the end of the NAL unit which the packet pointed to by `packet_it`
+// belongs to. Returns an iterator to the last packet of the frame if the end
+// of the NAL unit wasn't found.
+VCMSessionInfo::PacketIterator VCMSessionInfo::FindNaluEnd(
+    PacketIterator packet_it) const {
+  if ((*packet_it).completeNALU == kNaluEnd ||
+      (*packet_it).completeNALU == kNaluComplete) {
+    return packet_it;
+  }
+  // Find the end of the NAL unit.
+  for (; packet_it != packets_.end(); ++packet_it) {
+    if (((*packet_it).completeNALU == kNaluComplete &&
+         (*packet_it).sizeBytes > 0) ||
+        // Found next NALU.
+        (*packet_it).completeNALU == kNaluStart)
+      return --packet_it;
+    if ((*packet_it).completeNALU == kNaluEnd)
+      return packet_it;
+  }
+  // The end wasn't found.
+  return --packet_it;
+}
+
+size_t VCMSessionInfo::DeletePacketData(PacketIterator start,
+                                        PacketIterator end) {
+  size_t bytes_to_delete = 0;  // The number of bytes to delete.
+  PacketIterator packet_after_end = end;
+  ++packet_after_end;
+
+  // Get the number of bytes to delete.
+  // Clear the size of these packets.
+  for (PacketIterator it = start; it != packet_after_end; ++it) {
+    bytes_to_delete += (*it).sizeBytes;
+    (*it).sizeBytes = 0;
+    (*it).dataPtr = NULL;
+  }
+  if (bytes_to_delete > 0)
+    ShiftSubsequentPackets(end, -static_cast<int>(bytes_to_delete));
+  return bytes_to_delete;
+}
+
+VCMSessionInfo::PacketIterator VCMSessionInfo::FindNextPartitionBeginning(
+    PacketIterator it) const {
+  while (it != packets_.end()) {
+    if (absl::get<RTPVideoHeaderVP8>((*it).video_header.video_type_header)
+            .beginningOfPartition) {
+      return it;
+    }
+    ++it;
+  }
+  return it;
+}
+
+VCMSessionInfo::PacketIterator VCMSessionInfo::FindPartitionEnd(
+    PacketIterator it) const {
+  RTC_DCHECK_EQ((*it).codec(), kVideoCodecVP8);
+  PacketIterator prev_it = it;
+  const int partition_id =
+      absl::get<RTPVideoHeaderVP8>((*it).video_header.video_type_header)
+          .partitionId;
+  while (it != packets_.end()) {
+    bool beginning =
+        absl::get<RTPVideoHeaderVP8>((*it).video_header.video_type_header)
+            .beginningOfPartition;
+    int current_partition_id =
+        absl::get<RTPVideoHeaderVP8>((*it).video_header.video_type_header)
+            .partitionId;
+    bool packet_loss_found = (!beginning && !InSequence(it, prev_it));
+    if (packet_loss_found ||
+        (beginning && current_partition_id != partition_id)) {
+      // Missing packet, the previous packet was the last in sequence.
+      return prev_it;
+    }
+    prev_it = it;
+    ++it;
+  }
+  return prev_it;
+}
+
+bool VCMSessionInfo::InSequence(const PacketIterator& packet_it,
+                                const PacketIterator& prev_packet_it) {
+  // If the two iterators are pointing to the same packet they are considered
+  // to be in sequence.
+  return (packet_it == prev_packet_it ||
+          (static_cast<uint16_t>((*prev_packet_it).seqNum + 1) ==
+           (*packet_it).seqNum));
+}
+
+size_t VCMSessionInfo::MakeDecodable() {
+  size_t return_length = 0;
+  if (packets_.empty()) {
+    return 0;
+  }
+  PacketIterator it = packets_.begin();
+  // Make sure we remove the first NAL unit if it's not decodable.
+  if ((*it).completeNALU == kNaluIncomplete || (*it).completeNALU == kNaluEnd) {
+    PacketIterator nalu_end = FindNaluEnd(it);
+    return_length += DeletePacketData(it, nalu_end);
+    it = nalu_end;
+  }
+  PacketIterator prev_it = it;
+  // Take care of the rest of the NAL units.
+  for (; it != packets_.end(); ++it) {
+    bool start_of_nalu = ((*it).completeNALU == kNaluStart ||
+                          (*it).completeNALU == kNaluComplete);
+    if (!start_of_nalu && !InSequence(it, prev_it)) {
+      // Found a sequence number gap due to packet loss.
+      PacketIterator nalu_end = FindNaluEnd(it);
+      return_length += DeletePacketData(it, nalu_end);
+      it = nalu_end;
+    }
+    prev_it = it;
+  }
+  return return_length;
+}
+
+bool VCMSessionInfo::HaveFirstPacket() const {
+  return !packets_.empty() && (first_packet_seq_num_ != -1);
+}
+
+bool VCMSessionInfo::HaveLastPacket() const {
+  return !packets_.empty() && (last_packet_seq_num_ != -1);
+}
+
+int VCMSessionInfo::InsertPacket(const VCMPacket& packet,
+                                 uint8_t* frame_buffer,
+                                 const FrameData& frame_data) {
+  if (packet.video_header.frame_type == VideoFrameType::kEmptyFrame) {
+    // Update sequence number of an empty packet.
+    // Only media packets are inserted into the packet list.
+    InformOfEmptyPacket(packet.seqNum);
+    return 0;
+  }
+
+  if (packets_.size() == kMaxPacketsInSession) {
+    RTC_LOG(LS_ERROR) << "Max number of packets per frame has been reached.";
+    return -1;
+  }
+
+  // Find the position of this packet in the packet list in sequence number
+  // order and insert it. Loop over the list in reverse order.
+  ReversePacketIterator rit = packets_.rbegin();
+  for (; rit != packets_.rend(); ++rit)
+    if (LatestSequenceNumber(packet.seqNum, (*rit).seqNum) == packet.seqNum)
+      break;
+
+  // Check for duplicate packets.
+  if (rit != packets_.rend() && (*rit).seqNum == packet.seqNum &&
+      (*rit).sizeBytes > 0)
+    return -2;
+
+  if (packet.codec() == kVideoCodecH264) {
+    // H.264 can have leading or trailing non-VCL (Video Coding Layer)
+    // NALUs, such as SPS/PPS/SEI and others.  Also, the RTP marker bit is
+    // not reliable for the last packet of a frame (RFC 6184 5.1 - "Decoders
+    // [] MUST NOT rely on this property"), so allow out-of-order packets to
+    // update the first and last seq# range.  Also mark as a key frame if
+    // any packet is of that type.
+    if (frame_type_ != VideoFrameType::kVideoFrameKey) {
+      frame_type_ = packet.video_header.frame_type;
+    }
+    if (packet.is_first_packet_in_frame() &&
+        (first_packet_seq_num_ == -1 ||
+         IsNewerSequenceNumber(first_packet_seq_num_, packet.seqNum))) {
+      first_packet_seq_num_ = packet.seqNum;
+    }
+    // Note: the code does *not* currently handle the Marker bit being totally
+    // absent from a frame.  It does not, however, depend on it being on the last
+    // packet of the 'frame'/'session'.
+    if (packet.markerBit &&
+        (last_packet_seq_num_ == -1 ||
+         IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_))) {
+      last_packet_seq_num_ = packet.seqNum;
+    }
+  } else {
+    // Only insert media packets between first and last packets (when
+    // available).
+    // Placing check here, as to properly account for duplicate packets.
+    // Check if this is first packet (only valid for some codecs)
+    // Should only be set for one packet per session.
+    if (packet.is_first_packet_in_frame() && first_packet_seq_num_ == -1) {
+      // The first packet in a frame signals the frame type.
+      frame_type_ = packet.video_header.frame_type;
+      // Store the sequence number for the first packet.
+      first_packet_seq_num_ = static_cast<int>(packet.seqNum);
+    } else if (first_packet_seq_num_ != -1 &&
+               IsNewerSequenceNumber(first_packet_seq_num_, packet.seqNum)) {
+      RTC_LOG(LS_WARNING)
+          << "Received packet with a sequence number which is out "
+             "of frame boundaries";
+      return -3;
+    } else if (frame_type_ == VideoFrameType::kEmptyFrame &&
+               packet.video_header.frame_type != VideoFrameType::kEmptyFrame) {
+      // Update the frame type with the type of the first media packet.
+      // TODO(mikhal): Can this trigger?
+      frame_type_ = packet.video_header.frame_type;
+    }
+
+    // Track the marker bit, should only be set for one packet per session.
+    if (packet.markerBit && last_packet_seq_num_ == -1) {
+      last_packet_seq_num_ = static_cast<int>(packet.seqNum);
+    } else if (last_packet_seq_num_ != -1 &&
+               IsNewerSequenceNumber(packet.seqNum, last_packet_seq_num_)) {
+      RTC_LOG(LS_WARNING)
+          << "Received packet with a sequence number which is out "
+             "of frame boundaries";
+      return -3;
+    }
+  }
+
+  // The insert operation invalidates the iterator `rit`.
+  PacketIterator packet_list_it = packets_.insert(rit.base(), packet);
+
+  size_t returnLength = InsertBuffer(frame_buffer, packet_list_it);
+  UpdateCompleteSession();
+  return static_cast<int>(returnLength);
+}
+
+void VCMSessionInfo::InformOfEmptyPacket(uint16_t seq_num) {
+  // Empty packets may be FEC or filler packets. They are sequential and
+  // follow the data packets, therefore, we should only keep track of the high
+  // and low sequence numbers and may assume that the packets in between are
+  // empty packets belonging to the same frame (timestamp).
+  if (empty_seq_num_high_ == -1)
+    empty_seq_num_high_ = seq_num;
+  else
+    empty_seq_num_high_ = LatestSequenceNumber(seq_num, empty_seq_num_high_);
+  if (empty_seq_num_low_ == -1 ||
+      IsNewerSequenceNumber(empty_seq_num_low_, seq_num))
+    empty_seq_num_low_ = seq_num;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/session_info.h b/third_party/libwebrtc/modules/video_coding/session_info.h
new file mode 100644
index 0000000000..6079dbbb72
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/session_info.h
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_SESSION_INFO_H_
+#define MODULES_VIDEO_CODING_SESSION_INFO_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+#include <vector>
+
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/packet.h"
+
+namespace webrtc {
+// Used to pass data from jitter buffer to session info.
+// This data is then used in determining whether a frame is decodable.
+struct FrameData {
+  int64_t rtt_ms;
+  float rolling_average_packets_per_frame;
+};
+
+class VCMSessionInfo {
+ public:
+  VCMSessionInfo();
+  ~VCMSessionInfo();
+
+  void UpdateDataPointers(const uint8_t* old_base_ptr,
+                          const uint8_t* new_base_ptr);
+  void Reset();
+  int InsertPacket(const VCMPacket& packet,
+                   uint8_t* frame_buffer,
+                   const FrameData& frame_data);
+  bool complete() const;
+
+  // Makes the frame decodable. I.e., only contain decodable NALUs. All
+  // non-decodable NALUs will be deleted and packets will be moved to in
+  // memory to remove any empty space.
+  // Returns the number of bytes deleted from the session.
+  size_t MakeDecodable();
+
+  size_t SessionLength() const;
+  int NumPackets() const;
+  bool HaveFirstPacket() const;
+  bool HaveLastPacket() const;
+  webrtc::VideoFrameType FrameType() const { return frame_type_; }
+  int LowSequenceNumber() const;
+
+  // Returns highest sequence number, media or empty.
+  int HighSequenceNumber() const;
+  int PictureId() const;
+  int TemporalId() const;
+  bool LayerSync() const;
+  int Tl0PicId() const;
+
+  std::vector<NaluInfo> GetNaluInfos() const;
+
+  void SetGofInfo(const GofInfoVP9& gof_info, size_t idx);
+
+ private:
+  enum { kMaxVP8Partitions = 9 };
+
+  typedef std::list<VCMPacket> PacketList;
+  typedef PacketList::iterator PacketIterator;
+  typedef PacketList::const_iterator PacketIteratorConst;
+  typedef PacketList::reverse_iterator ReversePacketIterator;
+
+  void InformOfEmptyPacket(uint16_t seq_num);
+
+  // Finds the packet of the beginning of the next VP8 partition. If
+  // none is found the returned iterator points to `packets_.end()`.
+  // `it` is expected to point to the last packet of the previous partition,
+  // or to the first packet of the frame. `packets_skipped` is incremented
+  // for each packet found which doesn't have the beginning bit set.
+  PacketIterator FindNextPartitionBeginning(PacketIterator it) const;
+
+  // Returns an iterator pointing to the last packet of the partition pointed to
+  // by `it`.
+  PacketIterator FindPartitionEnd(PacketIterator it) const;
+  static bool InSequence(const PacketIterator& it,
+                         const PacketIterator& prev_it);
+  size_t InsertBuffer(uint8_t* frame_buffer, PacketIterator packetIterator);
+  size_t Insert(const uint8_t* buffer,
+                size_t length,
+                bool insert_start_code,
+                uint8_t* frame_buffer);
+  void ShiftSubsequentPackets(PacketIterator it, int steps_to_shift);
+  PacketIterator FindNaluEnd(PacketIterator packet_iter) const;
+  // Deletes the data of all packets between `start` and `end`, inclusively.
+  // Note that this function doesn't delete the actual packets.
+  size_t DeletePacketData(PacketIterator start, PacketIterator end);
+  void UpdateCompleteSession();
+
+  bool complete_;
+  webrtc::VideoFrameType frame_type_;
+  // Packets in this frame.
+  PacketList packets_;
+  int empty_seq_num_low_;
+  int empty_seq_num_high_;
+
+  // The following two variables correspond to the first and last media packets
+  // in a session defined by the first packet flag and the marker bit.
+  // They are not necessarily equal to the front and back packets, as packets
+  // may enter out of order.
+  // TODO(mikhal): Refactor the list to use a map.
+  int first_packet_seq_num_;
+  int last_packet_seq_num_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SESSION_INFO_H_
diff --git a/third_party/libwebrtc/modules/video_coding/session_info_unittest.cc b/third_party/libwebrtc/modules/video_coding/session_info_unittest.cc
new file mode 100644
index 0000000000..867997701d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/session_info_unittest.cc
@@ -0,0 +1,469 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/session_info.h"
+
+#include <string.h>
+
+#include "modules/video_coding/packet.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class TestSessionInfo : public ::testing::Test {
+ protected:
+  virtual void SetUp() {
+    memset(packet_buffer_, 0, sizeof(packet_buffer_));
+    memset(frame_buffer_, 0, sizeof(frame_buffer_));
+    session_.Reset();
+    packet_.video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+    packet_.sizeBytes = packet_buffer_size();
+    packet_.dataPtr = packet_buffer_;
+    packet_.seqNum = 0;
+    packet_.timestamp = 0;
+    frame_data.rtt_ms = 0;
+    frame_data.rolling_average_packets_per_frame = -1;
+  }
+
+  void FillPacket(uint8_t start_value) {
+    for (size_t i = 0; i < packet_buffer_size(); ++i)
+      packet_buffer_[i] = start_value + i;
+  }
+
+  void VerifyPacket(uint8_t* start_ptr, uint8_t start_value) {
+    for (size_t j = 0; j < packet_buffer_size(); ++j) {
+      ASSERT_EQ(start_value + j, start_ptr[j]);
+    }
+  }
+
+  size_t packet_buffer_size() const {
+    return sizeof(packet_buffer_) / sizeof(packet_buffer_[0]);
+  }
+  size_t frame_buffer_size() const {
+    return sizeof(frame_buffer_) / sizeof(frame_buffer_[0]);
+  }
+
+  enum { kPacketBufferSize = 10 };
+
+  uint8_t packet_buffer_[kPacketBufferSize];
+  uint8_t frame_buffer_[10 * kPacketBufferSize];
+
+  VCMSessionInfo session_;
+  VCMPacket packet_;
+  FrameData frame_data;
+};
+
+class TestNalUnits : public TestSessionInfo {
+ protected:
+  virtual void SetUp() {
+    TestSessionInfo::SetUp();
+    packet_.video_header.codec = kVideoCodecVP8;
+  }
+
+  bool VerifyNalu(int offset, int packets_expected, int start_value) {
+    EXPECT_GE(session_.SessionLength(),
+              packets_expected * packet_buffer_size());
+    for (int i = 0; i < packets_expected; ++i) {
+      int packet_index = (offset + i) * packet_buffer_size();
+      VerifyPacket(frame_buffer_ + packet_index, start_value + i);
+    }
+    return true;
+  }
+};
+
+class TestNackList : public TestSessionInfo {
+ protected:
+  static const size_t kMaxSeqNumListLength = 30;
+
+  virtual void SetUp() {
+    TestSessionInfo::SetUp();
+    seq_num_list_length_ = 0;
+    memset(seq_num_list_, 0, sizeof(seq_num_list_));
+  }
+
+  void BuildSeqNumList(uint16_t low, uint16_t high) {
+    size_t i = 0;
+    while (low != high + 1) {
+      EXPECT_LT(i, kMaxSeqNumListLength);
+      if (i >= kMaxSeqNumListLength) {
+        seq_num_list_length_ = kMaxSeqNumListLength;
+        return;
+      }
+      seq_num_list_[i] = low;
+      low++;
+      i++;
+    }
+    seq_num_list_length_ = i;
+  }
+
+  void VerifyAll(int value) {
+    for (int i = 0; i < seq_num_list_length_; ++i)
+      EXPECT_EQ(seq_num_list_[i], value);
+  }
+
+  int seq_num_list_[kMaxSeqNumListLength];
+  int seq_num_list_length_;
+};
+
+TEST_F(TestSessionInfo, TestSimpleAPIs) {
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.seqNum = 0xFFFE;
+  packet_.sizeBytes = packet_buffer_size();
+  packet_.video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  FillPacket(0);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  EXPECT_FALSE(session_.HaveLastPacket());
+  EXPECT_EQ(VideoFrameType::kVideoFrameKey, session_.FrameType());
+
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = true;
+  packet_.seqNum += 1;
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  EXPECT_TRUE(session_.HaveLastPacket());
+  EXPECT_EQ(packet_.seqNum, session_.HighSequenceNumber());
+  EXPECT_EQ(0xFFFE, session_.LowSequenceNumber());
+
+  // Insert empty packet which will be the new high sequence number.
+  // To make things more difficult we will make sure to have a wrap here.
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = true;
+  packet_.seqNum = 2;
+  packet_.sizeBytes = 0;
+  packet_.video_header.frame_type = VideoFrameType::kEmptyFrame;
+  EXPECT_EQ(0, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+  EXPECT_EQ(packet_.seqNum, session_.HighSequenceNumber());
+}
+
+TEST_F(TestSessionInfo, NormalOperation) {
+  packet_.seqNum = 0xFFFF;
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.markerBit = false;
+  FillPacket(0);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.video_header.is_first_packet_in_frame = false;
+  for (int i = 1; i < 9; ++i) {
+    packet_.seqNum += 1;
+    FillPacket(i);
+    ASSERT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                        packet_, frame_buffer_, frame_data)));
+  }
+
+  packet_.seqNum += 1;
+  packet_.markerBit = true;
+  FillPacket(9);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  EXPECT_EQ(10 * packet_buffer_size(), session_.SessionLength());
+  for (int i = 0; i < 10; ++i) {
+    SCOPED_TRACE("Calling VerifyPacket");
+    VerifyPacket(frame_buffer_ + i * packet_buffer_size(), i);
+  }
+}
+
+TEST_F(TestSessionInfo, OutOfBoundsPackets1PacketFrame) {
+  packet_.seqNum = 0x0001;
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.markerBit = true;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.seqNum = 0x0004;
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.markerBit = true;
+  FillPacket(1);
+  EXPECT_EQ(-3, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+  packet_.seqNum = 0x0000;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(-3, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+}
+
+TEST_F(TestSessionInfo, SetMarkerBitOnce) {
+  packet_.seqNum = 0x0005;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = true;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  ++packet_.seqNum;
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.markerBit = true;
+  FillPacket(1);
+  EXPECT_EQ(-3, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+}
+
+TEST_F(TestSessionInfo, OutOfBoundsPacketsBase) {
+  // Allow packets in the range 5-6.
+  packet_.seqNum = 0x0005;
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  // Insert an older packet with a first packet set.
+  packet_.seqNum = 0x0004;
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.markerBit = true;
+  FillPacket(1);
+  EXPECT_EQ(-3, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+  packet_.seqNum = 0x0006;
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.markerBit = true;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  packet_.seqNum = 0x0008;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = true;
+  FillPacket(1);
+  EXPECT_EQ(-3, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+}
+
+TEST_F(TestSessionInfo, OutOfBoundsPacketsWrap) {
+  packet_.seqNum = 0xFFFE;
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.seqNum = 0x0004;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = true;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  packet_.seqNum = 0x0002;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = false;
+  FillPacket(1);
+  ASSERT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  packet_.seqNum = 0xFFF0;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(-3, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+  packet_.seqNum = 0x0006;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(-3, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+}
+
+TEST_F(TestSessionInfo, OutOfBoundsOutOfOrder) {
+  // Insert out of bound regular packets, and then the first and last packet.
+  // Verify that correct bounds are maintained.
+  packet_.seqNum = 0x0003;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  // Insert an older packet with a first packet set.
+  packet_.seqNum = 0x0005;
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  packet_.seqNum = 0x0004;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(-3, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+  packet_.seqNum = 0x0010;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+  packet_.seqNum = 0x0008;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = true;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.seqNum = 0x0009;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(-3, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+}
+
+TEST_F(TestNalUnits, OnlyReceivedEmptyPacket) {
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluComplete;
+  packet_.video_header.frame_type = VideoFrameType::kEmptyFrame;
+  packet_.sizeBytes = 0;
+  packet_.seqNum = 0;
+  packet_.markerBit = false;
+  EXPECT_EQ(0, session_.InsertPacket(packet_, frame_buffer_, frame_data));
+
+  EXPECT_EQ(0U, session_.MakeDecodable());
+  EXPECT_EQ(0U, session_.SessionLength());
+}
+
+TEST_F(TestNalUnits, OneIsolatedNaluLoss) {
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.completeNALU = kNaluComplete;
+  packet_.seqNum = 0;
+  packet_.markerBit = false;
+  FillPacket(0);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluComplete;
+  packet_.seqNum += 2;
+  packet_.markerBit = true;
+  FillPacket(2);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  EXPECT_EQ(0U, session_.MakeDecodable());
+  EXPECT_EQ(2 * packet_buffer_size(), session_.SessionLength());
+  SCOPED_TRACE("Calling VerifyNalu");
+  EXPECT_TRUE(VerifyNalu(0, 1, 0));
+  SCOPED_TRACE("Calling VerifyNalu");
+  EXPECT_TRUE(VerifyNalu(1, 1, 2));
+}
+
+TEST_F(TestNalUnits, LossInMiddleOfNalu) {
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.completeNALU = kNaluComplete;
+  packet_.seqNum = 0;
+  packet_.markerBit = false;
+  FillPacket(0);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluEnd;
+  packet_.seqNum += 2;
+  packet_.markerBit = true;
+  FillPacket(2);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  EXPECT_EQ(packet_buffer_size(), session_.MakeDecodable());
+  EXPECT_EQ(packet_buffer_size(), session_.SessionLength());
+  SCOPED_TRACE("Calling VerifyNalu");
+  EXPECT_TRUE(VerifyNalu(0, 1, 0));
+}
+
+TEST_F(TestNalUnits, StartAndEndOfLastNalUnitLost) {
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.completeNALU = kNaluComplete;
+  packet_.seqNum = 0;
+  packet_.markerBit = false;
+  FillPacket(0);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluIncomplete;
+  packet_.seqNum += 2;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  EXPECT_EQ(packet_buffer_size(), session_.MakeDecodable());
+  EXPECT_EQ(packet_buffer_size(), session_.SessionLength());
+  SCOPED_TRACE("Calling VerifyNalu");
+  EXPECT_TRUE(VerifyNalu(0, 1, 0));
+}
+
+TEST_F(TestNalUnits, ReorderWrapNoLoss) {
+  packet_.seqNum = 0xFFFF;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluIncomplete;
+  packet_.seqNum += 1;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.video_header.is_first_packet_in_frame = true;
+  packet_.completeNALU = kNaluComplete;
+  packet_.seqNum -= 1;
+  packet_.markerBit = false;
+  FillPacket(0);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluEnd;
+  packet_.seqNum += 2;
+  packet_.markerBit = true;
+  FillPacket(2);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  EXPECT_EQ(0U, session_.MakeDecodable());
+  EXPECT_EQ(3 * packet_buffer_size(), session_.SessionLength());
+  SCOPED_TRACE("Calling VerifyNalu");
+  EXPECT_TRUE(VerifyNalu(0, 1, 0));
+}
+
+TEST_F(TestNalUnits, WrapLosses) {
+  packet_.seqNum = 0xFFFF;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluIncomplete;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluEnd;
+  packet_.seqNum += 2;
+  packet_.markerBit = true;
+  FillPacket(2);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  EXPECT_EQ(2 * packet_buffer_size(), session_.MakeDecodable());
+  EXPECT_EQ(0U, session_.SessionLength());
+}
+
+TEST_F(TestNalUnits, ReorderWrapLosses) {
+  packet_.seqNum = 0xFFFF;
+
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluEnd;
+  packet_.seqNum += 2;
+  packet_.markerBit = true;
+  FillPacket(2);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  packet_.seqNum -= 2;
+  packet_.video_header.is_first_packet_in_frame = false;
+  packet_.completeNALU = kNaluIncomplete;
+  packet_.markerBit = false;
+  FillPacket(1);
+  EXPECT_EQ(packet_buffer_size(), static_cast<size_t>(session_.InsertPacket(
+                                      packet_, frame_buffer_, frame_data)));
+
+  EXPECT_EQ(2 * packet_buffer_size(), session_.MakeDecodable());
+  EXPECT_EQ(0U, session_.SessionLength());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/BUILD.gn b/third_party/libwebrtc/modules/video_coding/svc/BUILD.gn
new file mode 100644
index 0000000000..b8ce91d99a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/BUILD.gn
@@ -0,0 +1,135 @@
+# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("scalability_mode_util") {
+  sources = [
+    "scalability_mode_util.cc",
+    "scalability_mode_util.h",
+  ]
+  deps = [
+    "../../../api/video_codecs:scalability_mode",
+    "../../../api/video_codecs:video_codecs_api",
+    "../../../rtc_base:checks",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_source_set("scalable_video_controller") {
+  sources = [
+    "scalable_video_controller.h",
+    "scalable_video_controller_no_layering.cc",
+    "scalable_video_controller_no_layering.h",
+  ]
+  deps = [
+    "../../../api/transport/rtp:dependency_descriptor",
+    "../../../api/video:video_bitrate_allocation",
+    "../../../common_video/generic_frame_descriptor",
+    "../../../rtc_base:checks",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_source_set("scalability_structures") {
+  sources = [
+    "create_scalability_structure.cc",
+    "create_scalability_structure.h",
+    "scalability_structure_full_svc.cc",
+    "scalability_structure_full_svc.h",
+    "scalability_structure_key_svc.cc",
+    "scalability_structure_key_svc.h",
+    "scalability_structure_l2t2_key_shift.cc",
+    "scalability_structure_l2t2_key_shift.h",
+    "scalability_structure_simulcast.cc",
+    "scalability_structure_simulcast.h",
+  ]
+  deps = [
+    ":scalable_video_controller",
+    "../../../api/transport/rtp:dependency_descriptor",
+    "../../../api/video:video_bitrate_allocation",
+    "../../../api/video_codecs:scalability_mode",
+    "../../../common_video/generic_frame_descriptor",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_source_set("svc_rate_allocator") {
+  sources = [
+    "svc_rate_allocator.cc",
+    "svc_rate_allocator.h",
+  ]
+  deps = [
+    ":scalability_structures",
+    "../../../api/video:video_bitrate_allocation",
+    "../../../api/video:video_bitrate_allocator",
+    "../../../api/video:video_codec_constants",
+    "../../../api/video_codecs:video_codecs_api",
+    "../../../rtc_base:checks",
+    "../../../rtc_base/experiments:stable_target_rate_experiment",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/container:inlined_vector" ]
+}
+
+if (rtc_include_tests) {
+  rtc_source_set("scalability_structure_tests") {
+    testonly = true
+    sources = [
+      "scalability_mode_util_unittest.cc",
+      "scalability_structure_full_svc_unittest.cc",
+      "scalability_structure_key_svc_unittest.cc",
+      "scalability_structure_l2t2_key_shift_unittest.cc",
+      "scalability_structure_test_helpers.cc",
+      "scalability_structure_test_helpers.h",
+      "scalability_structure_unittest.cc",
+    ]
+    deps = [
+      ":scalability_mode_util",
+      ":scalability_structures",
+      ":scalable_video_controller",
+      "..:chain_diff_calculator",
+      "..:frame_dependencies_calculator",
+      "../../../api:array_view",
+      "../../../api/transport/rtp:dependency_descriptor",
+      "../../../api/video:video_bitrate_allocation",
+      "../../../api/video:video_frame_type",
+      "../../../api/video_codecs:scalability_mode",
+      "../../../common_video/generic_frame_descriptor",
+      "../../../rtc_base:stringutils",
+      "../../../test:test_support",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+
+  rtc_source_set("svc_rate_allocator_tests") {
+    testonly = true
+    sources = [ "svc_rate_allocator_unittest.cc" ]
+    deps = [
+      ":svc_rate_allocator",
+      "..:webrtc_vp9_helpers",
+      "../../../rtc_base:checks",
+      "../../../test:field_trial",
+      "../../../test:test_support",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/video_coding/svc/create_scalability_structure.cc b/third_party/libwebrtc/modules/video_coding/svc/create_scalability_structure.cc
new file mode 100644
index 0000000000..fbcd27b139
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/create_scalability_structure.cc
@@ -0,0 +1,295 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/create_scalability_structure.h"
+
+#include <memory>
+
+#include "api/video_codecs/scalability_mode.h"
+#include "modules/video_coding/svc/scalability_structure_full_svc.h"
+#include "modules/video_coding/svc/scalability_structure_key_svc.h"
+#include "modules/video_coding/svc/scalability_structure_l2t2_key_shift.h"
+#include "modules/video_coding/svc/scalability_structure_simulcast.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "modules/video_coding/svc/scalable_video_controller_no_layering.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+struct NamedStructureFactory {
+  ScalabilityMode name;
+  // Use function pointer to make NamedStructureFactory trivally destructable.
+  std::unique_ptr<ScalableVideoController> (*factory)();
+  ScalableVideoController::StreamLayersConfig config;
+};
+
+// Wrap std::make_unique function to have correct return type.
+template <typename T>
+std::unique_ptr<ScalableVideoController> Create() {
+  return std::make_unique<T>();
+}
+
+template <typename T>
+std::unique_ptr<ScalableVideoController> CreateH() {
+  // 1.5:1 scaling, see https://w3c.github.io/webrtc-svc/#scalabilitymodes*
+  typename T::ScalingFactor factor;
+  factor.num = 2;
+  factor.den = 3;
+  return std::make_unique<T>(factor);
+}
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL1T1 = {
+    /*num_spatial_layers=*/1, /*num_temporal_layers=*/1,
+    /*uses_reference_scaling=*/false};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL1T2 = {
+    /*num_spatial_layers=*/1, /*num_temporal_layers=*/2,
+    /*uses_reference_scaling=*/false};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL1T3 = {
+    /*num_spatial_layers=*/1, /*num_temporal_layers=*/3,
+    /*uses_reference_scaling=*/false};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL2T1 = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/1,
+    /*uses_reference_scaling=*/true,
+    {1, 1},
+    {2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL2T1h = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/1,
+    /*uses_reference_scaling=*/true,
+    {2, 1},
+    {3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL2T2 = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/2,
+    /*uses_reference_scaling=*/true,
+    {1, 1},
+    {2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL2T2h = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/2,
+    /*uses_reference_scaling=*/true,
+    {2, 1},
+    {3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL2T3 = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/3,
+    /*uses_reference_scaling=*/true,
+    {1, 1},
+    {2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL2T3h = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/3,
+    /*uses_reference_scaling=*/true,
+    {2, 1},
+    {3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL3T1 = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/1,
+    /*uses_reference_scaling=*/true,
+    {1, 1, 1},
+    {4, 2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL3T1h = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/1,
+    /*uses_reference_scaling=*/true,
+    {4, 2, 1},
+    {9, 3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL3T2 = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/2,
+    /*uses_reference_scaling=*/true,
+    {1, 1, 1},
+    {4, 2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL3T2h = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/2,
+    /*uses_reference_scaling=*/true,
+    {4, 2, 1},
+    {9, 3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL3T3 = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/3,
+    /*uses_reference_scaling=*/true,
+    {1, 1, 1},
+    {4, 2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigL3T3h = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/3,
+    /*uses_reference_scaling=*/true,
+    {4, 2, 1},
+    {9, 3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS2T1 = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/1,
+    /*uses_reference_scaling=*/false,
+    {1, 1},
+    {2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS2T1h = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/1,
+    /*uses_reference_scaling=*/false,
+    {2, 1},
+    {3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS2T2 = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/2,
+    /*uses_reference_scaling=*/false,
+    {1, 1},
+    {2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS2T2h = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/2,
+    /*uses_reference_scaling=*/false,
+    {2, 1},
+    {3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS2T3 = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/3,
+    /*uses_reference_scaling=*/false,
+    {1, 1},
+    {2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS2T3h = {
+    /*num_spatial_layers=*/2,
+    /*num_temporal_layers=*/3,
+    /*uses_reference_scaling=*/false,
+    {2, 1},
+    {3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS3T1 = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/1,
+    /*uses_reference_scaling=*/false,
+    {1, 1, 1},
+    {4, 2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS3T1h = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/1,
+    /*uses_reference_scaling=*/false,
+    {4, 2, 1},
+    {9, 3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS3T2 = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/2,
+    /*uses_reference_scaling=*/false,
+    {1, 1, 1},
+    {4, 2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS3T2h = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/2,
+    /*uses_reference_scaling=*/false,
+    {4, 2, 1},
+    {9, 3, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS3T3 = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/3,
+    /*uses_reference_scaling=*/false,
+    {1, 1, 1},
+    {4, 2, 1}};
+
+constexpr ScalableVideoController::StreamLayersConfig kConfigS3T3h = {
+    /*num_spatial_layers=*/3,
+    /*num_temporal_layers=*/3,
+    /*uses_reference_scaling=*/false,
+    {4, 2, 1},
+    {9, 3, 1}};
+
+constexpr NamedStructureFactory kFactories[] = {
+    {ScalabilityMode::kL1T1, Create<ScalableVideoControllerNoLayering>,
+     kConfigL1T1},
+    {ScalabilityMode::kL1T2, Create<ScalabilityStructureL1T2>, kConfigL1T2},
+    {ScalabilityMode::kL1T3, Create<ScalabilityStructureL1T3>, kConfigL1T3},
+    {ScalabilityMode::kL2T1, Create<ScalabilityStructureL2T1>, kConfigL2T1},
+    {ScalabilityMode::kL2T1h, CreateH<ScalabilityStructureL2T1>, kConfigL2T1h},
+    {ScalabilityMode::kL2T1_KEY, Create<ScalabilityStructureL2T1Key>,
+     kConfigL2T1},
+    {ScalabilityMode::kL2T2, Create<ScalabilityStructureL2T2>, kConfigL2T2},
+    {ScalabilityMode::kL2T2h, CreateH<ScalabilityStructureL2T2>, kConfigL2T2h},
+    {ScalabilityMode::kL2T2_KEY, Create<ScalabilityStructureL2T2Key>,
+     kConfigL2T2},
+    {ScalabilityMode::kL2T2_KEY_SHIFT, Create<ScalabilityStructureL2T2KeyShift>,
+     kConfigL2T2},
+    {ScalabilityMode::kL2T3, Create<ScalabilityStructureL2T3>, kConfigL2T3},
+    {ScalabilityMode::kL2T3h, CreateH<ScalabilityStructureL2T3>, kConfigL2T3h},
+    {ScalabilityMode::kL2T3_KEY, Create<ScalabilityStructureL2T3Key>,
+     kConfigL2T3},
+    {ScalabilityMode::kL3T1, Create<ScalabilityStructureL3T1>, kConfigL3T1},
+    {ScalabilityMode::kL3T1h, CreateH<ScalabilityStructureL3T1>, kConfigL3T1h},
+    {ScalabilityMode::kL3T1_KEY, Create<ScalabilityStructureL3T1Key>,
+     kConfigL3T1},
+    {ScalabilityMode::kL3T2, Create<ScalabilityStructureL3T2>, kConfigL3T2},
+    {ScalabilityMode::kL3T2h, CreateH<ScalabilityStructureL3T2>, kConfigL3T2h},
+    {ScalabilityMode::kL3T2_KEY, Create<ScalabilityStructureL3T2Key>,
+     kConfigL3T2},
+    {ScalabilityMode::kL3T3, Create<ScalabilityStructureL3T3>, kConfigL3T3},
+    {ScalabilityMode::kL3T3h, CreateH<ScalabilityStructureL3T3>, kConfigL3T3h},
+    {ScalabilityMode::kL3T3_KEY, Create<ScalabilityStructureL3T3Key>,
+     kConfigL3T3},
+    {ScalabilityMode::kS2T1, Create<ScalabilityStructureS2T1>, kConfigS2T1},
+    {ScalabilityMode::kS2T1h, CreateH<ScalabilityStructureS2T1>, kConfigS2T1h},
+    {ScalabilityMode::kS2T2, Create<ScalabilityStructureS2T2>, kConfigS2T2},
+    {ScalabilityMode::kS2T2h, CreateH<ScalabilityStructureS2T2>, kConfigS2T2h},
+    {ScalabilityMode::kS2T3, Create<ScalabilityStructureS2T3>, kConfigS2T3},
+    {ScalabilityMode::kS2T3h, CreateH<ScalabilityStructureS2T3>, kConfigS2T3h},
+    {ScalabilityMode::kS3T1, Create<ScalabilityStructureS3T1>, kConfigS3T1},
+    {ScalabilityMode::kS3T1h, CreateH<ScalabilityStructureS3T1>, kConfigS3T1h},
+    {ScalabilityMode::kS3T2, Create<ScalabilityStructureS3T2>, kConfigS3T2},
+    {ScalabilityMode::kS3T2h, CreateH<ScalabilityStructureS3T2>, kConfigS3T2h},
+    {ScalabilityMode::kS3T3, Create<ScalabilityStructureS3T3>, kConfigS3T3},
+    {ScalabilityMode::kS3T3h, CreateH<ScalabilityStructureS3T3>, kConfigS3T3h},
+};
+
+}  // namespace
+
+std::unique_ptr<ScalableVideoController> CreateScalabilityStructure(
+    ScalabilityMode name) {
+  for (const auto& entry : kFactories) {
+    if (entry.name == name) {
+      return entry.factory();
+    }
+  }
+  return nullptr;
+}
+
+absl::optional<ScalableVideoController::StreamLayersConfig>
+ScalabilityStructureConfig(ScalabilityMode name) {
+  for (const auto& entry : kFactories) {
+    if (entry.name == name) {
+      return entry.config;
+    }
+  }
+  return absl::nullopt;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/create_scalability_structure.h b/third_party/libwebrtc/modules/video_coding/svc/create_scalability_structure.h
new file mode 100644
index 0000000000..3b67443693
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/create_scalability_structure.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_SVC_CREATE_SCALABILITY_STRUCTURE_H_
+#define MODULES_VIDEO_CODING_SVC_CREATE_SCALABILITY_STRUCTURE_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+
+namespace webrtc {
+
+// Creates a structure by name according to
+// https://w3c.github.io/webrtc-svc/#scalabilitymodes*
+// Returns nullptr for unknown name.
+std::unique_ptr<ScalableVideoController> CreateScalabilityStructure(
+    ScalabilityMode name);
+
+// Returns description of the scalability structure identified by 'name',
+// Return nullopt for unknown name.
+absl::optional<ScalableVideoController::StreamLayersConfig>
+ScalabilityStructureConfig(ScalabilityMode name);
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_CREATE_SCALABILITY_STRUCTURE_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util.cc
new file mode 100644
index 0000000000..35d66df203
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util.cc
@@ -0,0 +1,390 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/svc/scalability_mode_util.h"
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+absl::optional<ScalabilityMode> ScalabilityModeFromString(
+    absl::string_view mode_string) {
+  if (mode_string == "L1T1")
+    return ScalabilityMode::kL1T1;
+  if (mode_string == "L1T2")
+    return ScalabilityMode::kL1T2;
+  if (mode_string == "L1T3")
+    return ScalabilityMode::kL1T3;
+
+  if (mode_string == "L2T1")
+    return ScalabilityMode::kL2T1;
+  if (mode_string == "L2T1h")
+    return ScalabilityMode::kL2T1h;
+  if (mode_string == "L2T1_KEY")
+    return ScalabilityMode::kL2T1_KEY;
+
+  if (mode_string == "L2T2")
+    return ScalabilityMode::kL2T2;
+  if (mode_string == "L2T2h")
+    return ScalabilityMode::kL2T2h;
+  if (mode_string == "L2T2_KEY")
+    return ScalabilityMode::kL2T2_KEY;
+  if (mode_string == "L2T2_KEY_SHIFT")
+    return ScalabilityMode::kL2T2_KEY_SHIFT;
+  if (mode_string == "L2T3")
+    return ScalabilityMode::kL2T3;
+  if (mode_string == "L2T3h")
+    return ScalabilityMode::kL2T3h;
+  if (mode_string == "L2T3_KEY")
+    return ScalabilityMode::kL2T3_KEY;
+
+  if (mode_string == "L3T1")
+    return ScalabilityMode::kL3T1;
+  if (mode_string == "L3T1h")
+    return ScalabilityMode::kL3T1h;
+  if (mode_string == "L3T1_KEY")
+    return ScalabilityMode::kL3T1_KEY;
+
+  if (mode_string == "L3T2")
+    return ScalabilityMode::kL3T2;
+  if (mode_string == "L3T2h")
+    return ScalabilityMode::kL3T2h;
+  if (mode_string == "L3T2_KEY")
+    return ScalabilityMode::kL3T2_KEY;
+
+  if (mode_string == "L3T3")
+    return ScalabilityMode::kL3T3;
+  if (mode_string == "L3T3h")
+    return ScalabilityMode::kL3T3h;
+  if (mode_string == "L3T3_KEY")
+    return ScalabilityMode::kL3T3_KEY;
+
+  if (mode_string == "S2T1")
+    return ScalabilityMode::kS2T1;
+  if (mode_string == "S2T1h")
+    return ScalabilityMode::kS2T1h;
+  if (mode_string == "S2T2")
+    return ScalabilityMode::kS2T2;
+  if (mode_string == "S2T2h")
+    return ScalabilityMode::kS2T2h;
+  if (mode_string == "S2T3")
+    return ScalabilityMode::kS2T3;
+  if (mode_string == "S2T3h")
+    return ScalabilityMode::kS2T3h;
+  if (mode_string == "S3T1")
+    return ScalabilityMode::kS3T1;
+  if (mode_string == "S3T1h")
+    return ScalabilityMode::kS3T1h;
+  if (mode_string == "S3T2")
+    return ScalabilityMode::kS3T2;
+  if (mode_string == "S3T2h")
+    return ScalabilityMode::kS3T2h;
+  if (mode_string == "S3T3")
+    return ScalabilityMode::kS3T3;
+  if (mode_string == "S3T3h")
+    return ScalabilityMode::kS3T3h;
+
+  return absl::nullopt;
+}
+
+InterLayerPredMode ScalabilityModeToInterLayerPredMode(
+    ScalabilityMode scalability_mode) {
+  switch (scalability_mode) {
+    case ScalabilityMode::kL1T1:
+    case ScalabilityMode::kL1T2:
+    case ScalabilityMode::kL1T3:
+    case ScalabilityMode::kL2T1:
+    case ScalabilityMode::kL2T1h:
+      return InterLayerPredMode::kOn;
+    case ScalabilityMode::kL2T1_KEY:
+      return InterLayerPredMode::kOnKeyPic;
+    case ScalabilityMode::kL2T2:
+    case ScalabilityMode::kL2T2h:
+      return InterLayerPredMode::kOn;
+    case ScalabilityMode::kL2T2_KEY:
+    case ScalabilityMode::kL2T2_KEY_SHIFT:
+      return InterLayerPredMode::kOnKeyPic;
+    case ScalabilityMode::kL2T3:
+    case ScalabilityMode::kL2T3h:
+      return InterLayerPredMode::kOn;
+    case ScalabilityMode::kL2T3_KEY:
+      return InterLayerPredMode::kOnKeyPic;
+    case ScalabilityMode::kL3T1:
+    case ScalabilityMode::kL3T1h:
+      return InterLayerPredMode::kOn;
+    case ScalabilityMode::kL3T1_KEY:
+      return InterLayerPredMode::kOnKeyPic;
+    case ScalabilityMode::kL3T2:
+    case ScalabilityMode::kL3T2h:
+      return InterLayerPredMode::kOn;
+    case ScalabilityMode::kL3T2_KEY:
+      return InterLayerPredMode::kOnKeyPic;
+    case ScalabilityMode::kL3T3:
+    case ScalabilityMode::kL3T3h:
+      return InterLayerPredMode::kOn;
+    case ScalabilityMode::kL3T3_KEY:
+      return InterLayerPredMode::kOnKeyPic;
+    case ScalabilityMode::kS2T1:
+    case ScalabilityMode::kS2T1h:
+    case ScalabilityMode::kS2T2:
+    case ScalabilityMode::kS2T2h:
+    case ScalabilityMode::kS2T3:
+    case ScalabilityMode::kS2T3h:
+    case ScalabilityMode::kS3T1:
+    case ScalabilityMode::kS3T1h:
+    case ScalabilityMode::kS3T2:
+    case ScalabilityMode::kS3T2h:
+    case ScalabilityMode::kS3T3:
+    case ScalabilityMode::kS3T3h:
+      return InterLayerPredMode::kOff;
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+int ScalabilityModeToNumSpatialLayers(ScalabilityMode scalability_mode) {
+  switch (scalability_mode) {
+    case ScalabilityMode::kL1T1:
+    case ScalabilityMode::kL1T2:
+    case ScalabilityMode::kL1T3:
+      return 1;
+    case ScalabilityMode::kL2T1:
+    case ScalabilityMode::kL2T1h:
+    case ScalabilityMode::kL2T1_KEY:
+    case ScalabilityMode::kL2T2:
+    case ScalabilityMode::kL2T2h:
+    case ScalabilityMode::kL2T2_KEY:
+    case ScalabilityMode::kL2T2_KEY_SHIFT:
+    case ScalabilityMode::kL2T3:
+    case ScalabilityMode::kL2T3h:
+    case ScalabilityMode::kL2T3_KEY:
+      return 2;
+    case ScalabilityMode::kL3T1:
+    case ScalabilityMode::kL3T1h:
+    case ScalabilityMode::kL3T1_KEY:
+    case ScalabilityMode::kL3T2:
+    case ScalabilityMode::kL3T2h:
+    case ScalabilityMode::kL3T2_KEY:
+    case ScalabilityMode::kL3T3:
+    case ScalabilityMode::kL3T3h:
+    case ScalabilityMode::kL3T3_KEY:
+      return 3;
+    case ScalabilityMode::kS2T1:
+    case ScalabilityMode::kS2T1h:
+    case ScalabilityMode::kS2T2:
+    case ScalabilityMode::kS2T2h:
+    case ScalabilityMode::kS2T3:
+    case ScalabilityMode::kS2T3h:
+      return 2;
+    case ScalabilityMode::kS3T1:
+    case ScalabilityMode::kS3T1h:
+    case ScalabilityMode::kS3T2:
+    case ScalabilityMode::kS3T2h:
+    case ScalabilityMode::kS3T3:
+    case ScalabilityMode::kS3T3h:
+      return 3;
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+int ScalabilityModeToNumTemporalLayers(ScalabilityMode scalability_mode) {
+  switch (scalability_mode) {
+    case ScalabilityMode::kL1T1:
+      return 1;
+    case ScalabilityMode::kL1T2:
+      return 2;
+    case ScalabilityMode::kL1T3:
+      return 3;
+    case ScalabilityMode::kL2T1:
+    case ScalabilityMode::kL2T1h:
+    case ScalabilityMode::kL2T1_KEY:
+      return 1;
+    case ScalabilityMode::kL2T2:
+    case ScalabilityMode::kL2T2h:
+    case ScalabilityMode::kL2T2_KEY:
+    case ScalabilityMode::kL2T2_KEY_SHIFT:
+      return 2;
+    case ScalabilityMode::kL2T3:
+    case ScalabilityMode::kL2T3h:
+    case ScalabilityMode::kL2T3_KEY:
+      return 3;
+    case ScalabilityMode::kL3T1:
+    case ScalabilityMode::kL3T1h:
+    case ScalabilityMode::kL3T1_KEY:
+      return 1;
+    case ScalabilityMode::kL3T2:
+    case ScalabilityMode::kL3T2h:
+    case ScalabilityMode::kL3T2_KEY:
+      return 2;
+    case ScalabilityMode::kL3T3:
+    case ScalabilityMode::kL3T3h:
+    case ScalabilityMode::kL3T3_KEY:
+      return 3;
+    case ScalabilityMode::kS2T1:
+    case ScalabilityMode::kS2T1h:
+    case ScalabilityMode::kS3T1:
+    case ScalabilityMode::kS3T1h:
+      return 1;
+    case ScalabilityMode::kS2T2:
+    case ScalabilityMode::kS2T2h:
+    case ScalabilityMode::kS3T2:
+    case ScalabilityMode::kS3T2h:
+      return 2;
+    case ScalabilityMode::kS2T3:
+    case ScalabilityMode::kS2T3h:
+    case ScalabilityMode::kS3T3:
+    case ScalabilityMode::kS3T3h:
+      return 3;
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+absl::optional<ScalabilityModeResolutionRatio> ScalabilityModeToResolutionRatio(
+    ScalabilityMode scalability_mode) {
+  switch (scalability_mode) {
+    case ScalabilityMode::kL1T1:
+    case ScalabilityMode::kL1T2:
+    case ScalabilityMode::kL1T3:
+      return absl::nullopt;
+    case ScalabilityMode::kL2T1:
+    case ScalabilityMode::kL2T1_KEY:
+    case ScalabilityMode::kL2T2:
+    case ScalabilityMode::kL2T2_KEY:
+    case ScalabilityMode::kL2T2_KEY_SHIFT:
+    case ScalabilityMode::kL2T3:
+    case ScalabilityMode::kL2T3_KEY:
+    case ScalabilityMode::kL3T1:
+    case ScalabilityMode::kL3T1_KEY:
+    case ScalabilityMode::kL3T2:
+    case ScalabilityMode::kL3T2_KEY:
+    case ScalabilityMode::kL3T3:
+    case ScalabilityMode::kL3T3_KEY:
+    case ScalabilityMode::kS2T1:
+    case ScalabilityMode::kS2T2:
+    case ScalabilityMode::kS2T3:
+    case ScalabilityMode::kS3T1:
+    case ScalabilityMode::kS3T2:
+    case ScalabilityMode::kS3T3:
+      return ScalabilityModeResolutionRatio::kTwoToOne;
+    case ScalabilityMode::kL2T1h:
+    case ScalabilityMode::kL2T2h:
+    case ScalabilityMode::kL2T3h:
+    case ScalabilityMode::kL3T1h:
+    case ScalabilityMode::kL3T2h:
+    case ScalabilityMode::kL3T3h:
+    case ScalabilityMode::kS2T1h:
+    case ScalabilityMode::kS2T2h:
+    case ScalabilityMode::kS2T3h:
+    case ScalabilityMode::kS3T1h:
+    case ScalabilityMode::kS3T2h:
+    case ScalabilityMode::kS3T3h:
+      return ScalabilityModeResolutionRatio::kThreeToTwo;
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+ScalabilityMode LimitNumSpatialLayers(ScalabilityMode scalability_mode,
+                                      int max_spatial_layers) {
+  int num_spatial_layers = ScalabilityModeToNumSpatialLayers(scalability_mode);
+  if (max_spatial_layers >= num_spatial_layers) {
+    return scalability_mode;
+  }
+
+  switch (scalability_mode) {
+    case ScalabilityMode::kL1T1:
+      return ScalabilityMode::kL1T1;
+    case ScalabilityMode::kL1T2:
+      return ScalabilityMode::kL1T2;
+    case ScalabilityMode::kL1T3:
+      return ScalabilityMode::kL1T3;
+    case ScalabilityMode::kL2T1:
+      return ScalabilityMode::kL1T1;
+    case ScalabilityMode::kL2T1h:
+      return ScalabilityMode::kL1T1;
+    case ScalabilityMode::kL2T1_KEY:
+      return ScalabilityMode::kL1T1;
+    case ScalabilityMode::kL2T2:
+      return ScalabilityMode::kL1T2;
+    case ScalabilityMode::kL2T2h:
+      return ScalabilityMode::kL1T2;
+    case ScalabilityMode::kL2T2_KEY:
+      return ScalabilityMode::kL1T2;
+    case ScalabilityMode::kL2T2_KEY_SHIFT:
+      return ScalabilityMode::kL1T2;
+    case ScalabilityMode::kL2T3:
+      return ScalabilityMode::kL1T3;
+    case ScalabilityMode::kL2T3h:
+      return ScalabilityMode::kL1T3;
+    case ScalabilityMode::kL2T3_KEY:
+      return ScalabilityMode::kL1T3;
+    case ScalabilityMode::kL3T1:
+      return max_spatial_layers == 2 ? ScalabilityMode::kL2T1
+                                     : ScalabilityMode::kL1T1;
+    case ScalabilityMode::kL3T1h:
+      return max_spatial_layers == 2 ? ScalabilityMode::kL2T1h
+                                     : ScalabilityMode::kL1T1;
+    case ScalabilityMode::kL3T1_KEY:
+      return max_spatial_layers == 2 ? ScalabilityMode::kL2T1_KEY
+                                     : ScalabilityMode::kL1T1;
+    case ScalabilityMode::kL3T2:
+      return max_spatial_layers == 2 ? ScalabilityMode::kL2T2
+                                     : ScalabilityMode::kL1T2;
+    case ScalabilityMode::kL3T2h:
+      return max_spatial_layers == 2 ? ScalabilityMode::kL2T2h
+                                     : ScalabilityMode::kL1T2;
+    case ScalabilityMode::kL3T2_KEY:
+      return max_spatial_layers == 2 ? ScalabilityMode::kL2T2_KEY
+                                     : ScalabilityMode::kL1T2;
+    case ScalabilityMode::kL3T3:
+      return max_spatial_layers == 2 ? ScalabilityMode::kL2T3
+                                     : ScalabilityMode::kL1T3;
+    case ScalabilityMode::kL3T3h:
+      return max_spatial_layers == 2 ? ScalabilityMode::kL2T3h
+                                     : ScalabilityMode::kL1T3;
+    case ScalabilityMode::kL3T3_KEY:
+      return max_spatial_layers == 2 ? ScalabilityMode::kL2T3_KEY
+                                     : ScalabilityMode::kL1T3;
+    case ScalabilityMode::kS2T1:
+      return ScalabilityMode::kL1T1;
+    case ScalabilityMode::kS2T1h:
+      return ScalabilityMode::kL1T1;
+    case ScalabilityMode::kS2T2:
+      return ScalabilityMode::kL1T2;
+    case ScalabilityMode::kS2T2h:
+      return ScalabilityMode::kL1T2;
+    case ScalabilityMode::kS2T3:
+      return ScalabilityMode::kL1T3;
+    case ScalabilityMode::kS2T3h:
+      return ScalabilityMode::kL1T3;
+    case ScalabilityMode::kS3T1:
+      return max_spatial_layers == 2 ? ScalabilityMode::kS2T1
+                                     : ScalabilityMode::kL1T1;
+    case ScalabilityMode::kS3T1h:
+      return max_spatial_layers == 2 ? ScalabilityMode::kS2T1h
+                                     : ScalabilityMode::kL1T1;
+    case ScalabilityMode::kS3T2:
+      return max_spatial_layers == 2 ? ScalabilityMode::kS2T2
+                                     : ScalabilityMode::kL1T2;
+    case ScalabilityMode::kS3T2h:
+      return max_spatial_layers == 2 ? ScalabilityMode::kS2T2h
+                                     : ScalabilityMode::kL1T2;
+    case ScalabilityMode::kS3T3:
+      return max_spatial_layers == 2 ? ScalabilityMode::kS2T3
+                                     : ScalabilityMode::kL1T3;
+    case ScalabilityMode::kS3T3h:
+      return max_spatial_layers == 2 ? ScalabilityMode::kS2T3h
+                                     : ScalabilityMode::kL1T3;
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util.h b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util.h
new file mode 100644
index 0000000000..9c8193e037
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_SVC_SCALABILITY_MODE_UTIL_H_
+#define MODULES_VIDEO_CODING_SVC_SCALABILITY_MODE_UTIL_H_
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "api/video_codecs/video_codec.h"
+
+namespace webrtc {
+
+enum class ScalabilityModeResolutionRatio {
+  kTwoToOne,    // The resolution ratio between spatial layers is 2:1.
+  kThreeToTwo,  // The resolution ratio between spatial layers is 1.5:1.
+};
+
+static constexpr char kDefaultScalabilityModeStr[] = "L1T2";
+
+absl::optional<ScalabilityMode> ScalabilityModeFromString(
+    absl::string_view scalability_mode_string);
+
+InterLayerPredMode ScalabilityModeToInterLayerPredMode(
+    ScalabilityMode scalability_mode);
+
+int ScalabilityModeToNumSpatialLayers(ScalabilityMode scalability_mode);
+
+int ScalabilityModeToNumTemporalLayers(ScalabilityMode scalability_mode);
+
+absl::optional<ScalabilityModeResolutionRatio> ScalabilityModeToResolutionRatio(
+    ScalabilityMode scalability_mode);
+
+ScalabilityMode LimitNumSpatialLayers(ScalabilityMode scalability_mode,
+                                      int max_spatial_layers);
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_SCALABILITY_MODE_UTIL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_gn/moz.build b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_gn/moz.build
new file mode 100644
index 0000000000..f786b11616
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("scalability_mode_util_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_unittest.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_unittest.cc
new file mode 100644
index 0000000000..448494ffcc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_mode_util_unittest.cc
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/svc/scalability_mode_util.h"
+
+#include <string>
+#include <tuple>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+TEST(ScalabilityModeUtil, ConvertsL1T2) {
+  EXPECT_EQ(ScalabilityModeFromString("L1T2"), ScalabilityMode::kL1T2);
+  EXPECT_EQ(ScalabilityModeToString(ScalabilityMode::kL1T2), "L1T2");
+}
+
+TEST(ScalabilityModeUtil, RejectsUnknownString) {
+  EXPECT_EQ(ScalabilityModeFromString(""), absl::nullopt);
+  EXPECT_EQ(ScalabilityModeFromString("not-a-mode"), absl::nullopt);
+}
+
+// Check roundtrip conversion of all enum values.
+TEST(ScalabilityModeUtil, ConvertsAllToAndFromString) {
+  const ScalabilityMode kLastEnum = ScalabilityMode::kS3T3h;
+  for (int numerical_enum = 0; numerical_enum <= static_cast<int>(kLastEnum);
+       numerical_enum++) {
+    ScalabilityMode scalability_mode =
+        static_cast<ScalabilityMode>(numerical_enum);
+    absl::string_view scalability_mode_string =
+        ScalabilityModeToString(scalability_mode);
+    EXPECT_FALSE(scalability_mode_string.empty());
+    EXPECT_EQ(ScalabilityModeFromString(scalability_mode_string),
+              scalability_mode);
+  }
+}
+
+struct TestParams {
+  std::string scalability_mode;
+  std::vector<std::tuple<std::vector<int>, std::string>>
+      limited_scalability_mode;
+};
+
+class NumSpatialLayersTest : public ::testing::TestWithParam<TestParams> {};
+
+INSTANTIATE_TEST_SUITE_P(
+    MaxLayers,
+    NumSpatialLayersTest,
+    ::testing::ValuesIn<TestParams>(
+        {{"L1T1", {{{0, 1}, "L1T1"}, {{2}, "L1T1"}, {{3}, "L1T1"}}},
+         {"L1T2", {{{0, 1}, "L1T2"}, {{2}, "L1T2"}, {{3}, "L1T2"}}},
+         {"L1T3", {{{0, 1}, "L1T3"}, {{2}, "L1T3"}, {{3}, "L1T3"}}},
+         {"L2T1", {{{0, 1}, "L1T1"}, {{2}, "L2T1"}, {{3}, "L2T1"}}},
+         {"L2T1h", {{{0, 1}, "L1T1"}, {{2}, "L2T1h"}, {{3}, "L2T1h"}}},
+         {"L2T1_KEY", {{{0, 1}, "L1T1"}, {{2}, "L2T1_KEY"}, {{3}, "L2T1_KEY"}}},
+         {"L2T2", {{{0, 1}, "L1T2"}, {{2}, "L2T2"}, {{3}, "L2T2"}}},
+         {"L2T2h", {{{0, 1}, "L1T2"}, {{2}, "L2T2h"}, {{3}, "L2T2h"}}},
+         {"L2T2_KEY", {{{0, 1}, "L1T2"}, {{2}, "L2T2_KEY"}, {{3}, "L2T2_KEY"}}},
+         {"L2T2_KEY_SHIFT",
+          {{{0, 1}, "L1T2"}, {{2}, "L2T2_KEY_SHIFT"}, {{3}, "L2T2_KEY_SHIFT"}}},
+         {"L2T3", {{{0, 1}, "L1T3"}, {{2}, "L2T3"}, {{3}, "L2T3"}}},
+         {"L2T3h", {{{0, 1}, "L1T3"}, {{2}, "L2T3h"}, {{3}, "L2T3h"}}},
+         {"L2T3_KEY", {{{0, 1}, "L1T3"}, {{2}, "L2T3_KEY"}, {{3}, "L2T3_KEY"}}},
+         {"L3T1", {{{0, 1}, "L1T1"}, {{2}, "L2T1"}, {{3}, "L3T1"}}},
+         {"L3T1h", {{{0, 1}, "L1T1"}, {{2}, "L2T1h"}, {{3}, "L3T1h"}}},
+         {"L3T1_KEY", {{{0, 1}, "L1T1"}, {{2}, "L2T1_KEY"}, {{3}, "L3T1_KEY"}}},
+         {"L3T2", {{{0, 1}, "L1T2"}, {{2}, "L2T2"}, {{3}, "L3T2"}}},
+         {"L3T2h", {{{0, 1}, "L1T2"}, {{2}, "L2T2h"}, {{3}, "L3T2h"}}},
+         {"L3T2_KEY", {{{0, 1}, "L1T2"}, {{2}, "L2T2_KEY"}, {{3}, "L3T2_KEY"}}},
+         {"L3T3", {{{0, 1}, "L1T3"}, {{2}, "L2T3"}, {{3}, "L3T3"}}},
+         {"L3T3h", {{{0, 1}, "L1T3"}, {{2}, "L2T3h"}, {{3}, "L3T3h"}}},
+         {"L3T3_KEY", {{{0, 1}, "L1T3"}, {{2}, "L2T3_KEY"}, {{3}, "L3T3_KEY"}}},
+         {"S2T1", {{{0, 1}, "L1T1"}, {{2}, "S2T1"}, {{3}, "S2T1"}}},
+         {"S2T1h", {{{0, 1}, "L1T1"}, {{2}, "S2T1h"}, {{3}, "S2T1h"}}},
+         {"S2T2", {{{0, 1}, "L1T2"}, {{2}, "S2T2"}, {{3}, "S2T2"}}},
+         {"S2T2h", {{{0, 1}, "L1T2"}, {{2}, "S2T2h"}, {{3}, "S2T2h"}}},
+         {"S2T3", {{{0, 1}, "L1T3"}, {{2}, "S2T3"}, {{3}, "S2T3"}}},
+         {"S2T3h", {{{0, 1}, "L1T3"}, {{2}, "S2T3h"}, {{3}, "S2T3h"}}},
+         {"S3T1", {{{0, 1}, "L1T1"}, {{2}, "S2T1"}, {{3}, "S3T1"}}},
+         {"S3T1h", {{{0, 1}, "L1T1"}, {{2}, "S2T1h"}, {{3}, "S3T1h"}}},
+         {"S3T2", {{{0, 1}, "L1T2"}, {{2}, "S2T2"}, {{3}, "S3T2"}}},
+         {"S3T2h", {{{0, 1}, "L1T2"}, {{2}, "S2T2h"}, {{3}, "S3T2h"}}},
+         {"S3T3", {{{0, 1}, "L1T3"}, {{2}, "S2T3"}, {{3}, "S3T3"}}},
+         {"S3T3h", {{{0, 1}, "L1T3"}, {{2}, "S2T3h"}, {{3}, "S3T3h"}}}}),
+    [](const ::testing::TestParamInfo<TestParams>& info) {
+      return info.param.scalability_mode;
+    });
+
+TEST_P(NumSpatialLayersTest, LimitsSpatialLayers) {
+  const ScalabilityMode mode =
+      *ScalabilityModeFromString(GetParam().scalability_mode);
+  for (const auto& param : GetParam().limited_scalability_mode) {
+    const std::vector<int> max_num_spatial_layers =
+        std::get<std::vector<int>>(param);
+    const ScalabilityMode expected_mode =
+        *ScalabilityModeFromString(std::get<std::string>(param));
+    for (const auto& max_layers : max_num_spatial_layers) {
+      EXPECT_EQ(expected_mode, LimitNumSpatialLayers(mode, max_layers));
+    }
+  }
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc.cc
new file mode 100644
index 0000000000..a262317597
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc.cc
@@ -0,0 +1,444 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/scalability_structure_full_svc.h"
+
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+constexpr int ScalabilityStructureFullSvc::kMaxNumSpatialLayers;
+constexpr int ScalabilityStructureFullSvc::kMaxNumTemporalLayers;
+constexpr absl::string_view ScalabilityStructureFullSvc::kFramePatternNames[];
+
+ScalabilityStructureFullSvc::ScalabilityStructureFullSvc(
+    int num_spatial_layers,
+    int num_temporal_layers,
+    ScalingFactor resolution_factor)
+    : num_spatial_layers_(num_spatial_layers),
+      num_temporal_layers_(num_temporal_layers),
+      resolution_factor_(resolution_factor),
+      active_decode_targets_(
+          (uint32_t{1} << (num_spatial_layers * num_temporal_layers)) - 1) {
+  RTC_DCHECK_LE(num_spatial_layers, kMaxNumSpatialLayers);
+  RTC_DCHECK_LE(num_temporal_layers, kMaxNumTemporalLayers);
+}
+
+ScalabilityStructureFullSvc::~ScalabilityStructureFullSvc() = default;
+
+ScalabilityStructureFullSvc::StreamLayersConfig
+ScalabilityStructureFullSvc::StreamConfig() const {
+  StreamLayersConfig result;
+  result.num_spatial_layers = num_spatial_layers_;
+  result.num_temporal_layers = num_temporal_layers_;
+  result.scaling_factor_num[num_spatial_layers_ - 1] = 1;
+  result.scaling_factor_den[num_spatial_layers_ - 1] = 1;
+  for (int sid = num_spatial_layers_ - 1; sid > 0; --sid) {
+    result.scaling_factor_num[sid - 1] =
+        resolution_factor_.num * result.scaling_factor_num[sid];
+    result.scaling_factor_den[sid - 1] =
+        resolution_factor_.den * result.scaling_factor_den[sid];
+  }
+  result.uses_reference_scaling = num_spatial_layers_ > 1;
+  return result;
+}
+
+bool ScalabilityStructureFullSvc::TemporalLayerIsActive(int tid) const {
+  if (tid >= num_temporal_layers_) {
+    return false;
+  }
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    if (DecodeTargetIsActive(sid, tid)) {
+      return true;
+    }
+  }
+  return false;
+}
+
+DecodeTargetIndication ScalabilityStructureFullSvc::Dti(
+    int sid,
+    int tid,
+    const LayerFrameConfig& config) {
+  if (sid < config.SpatialId() || tid < config.TemporalId()) {
+    return DecodeTargetIndication::kNotPresent;
+  }
+  if (sid == config.SpatialId()) {
+    if (tid == 0) {
+      RTC_DCHECK_EQ(config.TemporalId(), 0);
+      return DecodeTargetIndication::kSwitch;
+    }
+    if (tid == config.TemporalId()) {
+      return DecodeTargetIndication::kDiscardable;
+    }
+    if (tid > config.TemporalId()) {
+      RTC_DCHECK_GT(tid, config.TemporalId());
+      return DecodeTargetIndication::kSwitch;
+    }
+  }
+  RTC_DCHECK_GT(sid, config.SpatialId());
+  RTC_DCHECK_GE(tid, config.TemporalId());
+  if (config.IsKeyframe() || config.Id() == kKey) {
+    return DecodeTargetIndication::kSwitch;
+  }
+  return DecodeTargetIndication::kRequired;
+}
+
+ScalabilityStructureFullSvc::FramePattern
+ScalabilityStructureFullSvc::NextPattern() const {
+  switch (last_pattern_) {
+    case kNone:
+      return kKey;
+    case kDeltaT2B:
+      return kDeltaT0;
+    case kDeltaT2A:
+      if (TemporalLayerIsActive(1)) {
+        return kDeltaT1;
+      }
+      return kDeltaT0;
+    case kDeltaT1:
+      if (TemporalLayerIsActive(2)) {
+        return kDeltaT2B;
+      }
+      return kDeltaT0;
+    case kKey:
+    case kDeltaT0:
+      if (TemporalLayerIsActive(2)) {
+        return kDeltaT2A;
+      }
+      if (TemporalLayerIsActive(1)) {
+        return kDeltaT1;
+      }
+      return kDeltaT0;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return kNone;
+}
+
+std::vector<ScalableVideoController::LayerFrameConfig>
+ScalabilityStructureFullSvc::NextFrameConfig(bool restart) {
+  std::vector<LayerFrameConfig> configs;
+  if (active_decode_targets_.none()) {
+    last_pattern_ = kNone;
+    return configs;
+  }
+  configs.reserve(num_spatial_layers_);
+
+  if (last_pattern_ == kNone || restart) {
+    can_reference_t0_frame_for_spatial_id_.reset();
+    last_pattern_ = kNone;
+  }
+  FramePattern current_pattern = NextPattern();
+
+  absl::optional<int> spatial_dependency_buffer_id;
+  switch (current_pattern) {
+    case kDeltaT0:
+    case kKey:
+      // Disallow temporal references cross T0 on higher temporal layers.
+      can_reference_t1_frame_for_spatial_id_.reset();
+      for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+        if (!DecodeTargetIsActive(sid, /*tid=*/0)) {
+          // Next frame from the spatial layer `sid` shouldn't depend on
+          // potentially old previous frame from the spatial layer `sid`.
+          can_reference_t0_frame_for_spatial_id_.reset(sid);
+          continue;
+        }
+        configs.emplace_back();
+        ScalableVideoController::LayerFrameConfig& config = configs.back();
+        config.Id(current_pattern).S(sid).T(0);
+
+        if (spatial_dependency_buffer_id) {
+          config.Reference(*spatial_dependency_buffer_id);
+        } else if (current_pattern == kKey) {
+          config.Keyframe();
+        }
+
+        if (can_reference_t0_frame_for_spatial_id_[sid]) {
+          config.ReferenceAndUpdate(BufferIndex(sid, /*tid=*/0));
+        } else {
+          // TODO(bugs.webrtc.org/11999): Propagate chain restart on delta frame
+          // to ChainDiffCalculator
+          config.Update(BufferIndex(sid, /*tid=*/0));
+        }
+
+        spatial_dependency_buffer_id = BufferIndex(sid, /*tid=*/0);
+      }
+      break;
+    case kDeltaT1:
+      for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+        if (!DecodeTargetIsActive(sid, /*tid=*/1) ||
+            !can_reference_t0_frame_for_spatial_id_[sid]) {
+          continue;
+        }
+        configs.emplace_back();
+        ScalableVideoController::LayerFrameConfig& config = configs.back();
+        config.Id(current_pattern).S(sid).T(1);
+        // Temporal reference.
+        config.Reference(BufferIndex(sid, /*tid=*/0));
+        // Spatial reference unless this is the lowest active spatial layer.
+        if (spatial_dependency_buffer_id) {
+          config.Reference(*spatial_dependency_buffer_id);
+        }
+        // No frame reference top layer frame, so no need save it into a buffer.
+        if (num_temporal_layers_ > 2 || sid < num_spatial_layers_ - 1) {
+          config.Update(BufferIndex(sid, /*tid=*/1));
+        }
+        spatial_dependency_buffer_id = BufferIndex(sid, /*tid=*/1);
+      }
+      break;
+    case kDeltaT2A:
+    case kDeltaT2B:
+      for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+        if (!DecodeTargetIsActive(sid, /*tid=*/2) ||
+            !can_reference_t0_frame_for_spatial_id_[sid]) {
+          continue;
+        }
+        configs.emplace_back();
+        ScalableVideoController::LayerFrameConfig& config = configs.back();
+        config.Id(current_pattern).S(sid).T(2);
+        // Temporal reference.
+        if (current_pattern == kDeltaT2B &&
+            can_reference_t1_frame_for_spatial_id_[sid]) {
+          config.Reference(BufferIndex(sid, /*tid=*/1));
+        } else {
+          config.Reference(BufferIndex(sid, /*tid=*/0));
+        }
+        // Spatial reference unless this is the lowest active spatial layer.
+        if (spatial_dependency_buffer_id) {
+          config.Reference(*spatial_dependency_buffer_id);
+        }
+        // No frame reference top layer frame, so no need save it into a buffer.
+        if (sid < num_spatial_layers_ - 1) {
+          config.Update(BufferIndex(sid, /*tid=*/2));
+        }
+        spatial_dependency_buffer_id = BufferIndex(sid, /*tid=*/2);
+      }
+      break;
+    case kNone:
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+
+  if (configs.empty() && !restart) {
+    RTC_LOG(LS_WARNING) << "Failed to generate configuration for L"
+                        << num_spatial_layers_ << "T" << num_temporal_layers_
+                        << " with active decode targets "
+                        << active_decode_targets_.to_string('-').substr(
+                               active_decode_targets_.size() -
+                               num_spatial_layers_ * num_temporal_layers_)
+                        << " and transition from "
+                        << kFramePatternNames[last_pattern_] << " to "
+                        << kFramePatternNames[current_pattern]
+                        << ". Resetting.";
+    return NextFrameConfig(/*restart=*/true);
+  }
+
+  return configs;
+}
+
+GenericFrameInfo ScalabilityStructureFullSvc::OnEncodeDone(
+    const LayerFrameConfig& config) {
+  // When encoder drops all frames for a temporal unit, it is better to reuse
+  // old temporal pattern rather than switch to next one, thus switch to next
+  // pattern defered here from the `NextFrameConfig`.
+  // In particular creating VP9 references rely on this behavior.
+  last_pattern_ = static_cast<FramePattern>(config.Id());
+  if (config.TemporalId() == 0) {
+    can_reference_t0_frame_for_spatial_id_.set(config.SpatialId());
+  }
+  if (config.TemporalId() == 1) {
+    can_reference_t1_frame_for_spatial_id_.set(config.SpatialId());
+  }
+
+  GenericFrameInfo frame_info;
+  frame_info.spatial_id = config.SpatialId();
+  frame_info.temporal_id = config.TemporalId();
+  frame_info.encoder_buffers = config.Buffers();
+  frame_info.decode_target_indications.reserve(num_spatial_layers_ *
+                                               num_temporal_layers_);
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    for (int tid = 0; tid < num_temporal_layers_; ++tid) {
+      frame_info.decode_target_indications.push_back(Dti(sid, tid, config));
+    }
+  }
+  if (config.TemporalId() == 0) {
+    frame_info.part_of_chain.resize(num_spatial_layers_);
+    for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+      frame_info.part_of_chain[sid] = config.SpatialId() <= sid;
+    }
+  } else {
+    frame_info.part_of_chain.assign(num_spatial_layers_, false);
+  }
+  frame_info.active_decode_targets = active_decode_targets_;
+  return frame_info;
+}
+
+void ScalabilityStructureFullSvc::OnRatesUpdated(
+    const VideoBitrateAllocation& bitrates) {
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    // Enable/disable spatial layers independetely.
+    bool active = true;
+    for (int tid = 0; tid < num_temporal_layers_; ++tid) {
+      // To enable temporal layer, require bitrates for lower temporal layers.
+      active = active && bitrates.GetBitrate(sid, tid) > 0;
+      SetDecodeTargetIsActive(sid, tid, active);
+    }
+  }
+}
+
+FrameDependencyStructure ScalabilityStructureL1T2::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 2;
+  structure.num_chains = 1;
+  structure.decode_target_protected_by_chain = {0, 0};
+  structure.templates.resize(3);
+  structure.templates[0].T(0).Dtis("SS").ChainDiffs({0});
+  structure.templates[1].T(0).Dtis("SS").ChainDiffs({2}).FrameDiffs({2});
+  structure.templates[2].T(1).Dtis("-D").ChainDiffs({1}).FrameDiffs({1});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureL1T3::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 3;
+  structure.num_chains = 1;
+  structure.decode_target_protected_by_chain = {0, 0, 0};
+  structure.templates.resize(5);
+  structure.templates[0].T(0).Dtis("SSS").ChainDiffs({0});
+  structure.templates[1].T(0).Dtis("SSS").ChainDiffs({4}).FrameDiffs({4});
+  structure.templates[2].T(1).Dtis("-DS").ChainDiffs({2}).FrameDiffs({2});
+  structure.templates[3].T(2).Dtis("--D").ChainDiffs({1}).FrameDiffs({1});
+  structure.templates[4].T(2).Dtis("--D").ChainDiffs({3}).FrameDiffs({1});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureL2T1::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 2;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 1};
+  structure.templates.resize(4);
+  structure.templates[0].S(0).Dtis("SR").ChainDiffs({2, 1}).FrameDiffs({2});
+  structure.templates[1].S(0).Dtis("SS").ChainDiffs({0, 0});
+  structure.templates[2].S(1).Dtis("-S").ChainDiffs({1, 1}).FrameDiffs({2, 1});
+  structure.templates[3].S(1).Dtis("-S").ChainDiffs({1, 1}).FrameDiffs({1});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureL2T2::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 4;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 0, 1, 1};
+  structure.templates.resize(6);
+  auto& templates = structure.templates;
+  templates[0].S(0).T(0).Dtis("SSSS").ChainDiffs({0, 0});
+  templates[1].S(0).T(0).Dtis("SSRR").ChainDiffs({4, 3}).FrameDiffs({4});
+  templates[2].S(0).T(1).Dtis("-D-R").ChainDiffs({2, 1}).FrameDiffs({2});
+  templates[3].S(1).T(0).Dtis("--SS").ChainDiffs({1, 1}).FrameDiffs({1});
+  templates[4].S(1).T(0).Dtis("--SS").ChainDiffs({1, 1}).FrameDiffs({4, 1});
+  templates[5].S(1).T(1).Dtis("---D").ChainDiffs({3, 2}).FrameDiffs({2, 1});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureL2T3::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 6;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 0, 0, 1, 1, 1};
+  auto& t = structure.templates;
+  t.resize(10);
+  t[1].S(0).T(0).Dtis("SSSSSS").ChainDiffs({0, 0});
+  t[6].S(1).T(0).Dtis("---SSS").ChainDiffs({1, 1}).FrameDiffs({1});
+  t[3].S(0).T(2).Dtis("--D--R").ChainDiffs({2, 1}).FrameDiffs({2});
+  t[8].S(1).T(2).Dtis("-----D").ChainDiffs({3, 2}).FrameDiffs({2, 1});
+  t[2].S(0).T(1).Dtis("-DS-RR").ChainDiffs({4, 3}).FrameDiffs({4});
+  t[7].S(1).T(1).Dtis("----DS").ChainDiffs({5, 4}).FrameDiffs({4, 1});
+  t[4].S(0).T(2).Dtis("--D--R").ChainDiffs({6, 5}).FrameDiffs({2});
+  t[9].S(1).T(2).Dtis("-----D").ChainDiffs({7, 6}).FrameDiffs({2, 1});
+  t[0].S(0).T(0).Dtis("SSSRRR").ChainDiffs({8, 7}).FrameDiffs({8});
+  t[5].S(1).T(0).Dtis("---SSS").ChainDiffs({1, 1}).FrameDiffs({8, 1});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureL3T1::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 3;
+  structure.num_chains = 3;
+  structure.decode_target_protected_by_chain = {0, 1, 2};
+  auto& templates = structure.templates;
+  templates.resize(6);
+  templates[0].S(0).Dtis("SRR").ChainDiffs({3, 2, 1}).FrameDiffs({3});
+  templates[1].S(0).Dtis("SSS").ChainDiffs({0, 0, 0});
+  templates[2].S(1).Dtis("-SR").ChainDiffs({1, 1, 1}).FrameDiffs({3, 1});
+  templates[3].S(1).Dtis("-SS").ChainDiffs({1, 1, 1}).FrameDiffs({1});
+  templates[4].S(2).Dtis("--S").ChainDiffs({2, 1, 1}).FrameDiffs({3, 1});
+  templates[5].S(2).Dtis("--S").ChainDiffs({2, 1, 1}).FrameDiffs({1});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureL3T2::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 6;
+  structure.num_chains = 3;
+  structure.decode_target_protected_by_chain = {0, 0, 1, 1, 2, 2};
+  auto& t = structure.templates;
+  t.resize(9);
+  // Templates are shown in the order frames following them appear in the
+  // stream, but in `structure.templates` array templates are sorted by
+  // (`spatial_id`, `temporal_id`) since that is a dependency descriptor
+  // requirement.
+  t[1].S(0).T(0).Dtis("SSSSSS").ChainDiffs({0, 0, 0});
+  t[4].S(1).T(0).Dtis("--SSSS").ChainDiffs({1, 1, 1}).FrameDiffs({1});
+  t[7].S(2).T(0).Dtis("----SS").ChainDiffs({2, 1, 1}).FrameDiffs({1});
+  t[2].S(0).T(1).Dtis("-D-R-R").ChainDiffs({3, 2, 1}).FrameDiffs({3});
+  t[5].S(1).T(1).Dtis("---D-R").ChainDiffs({4, 3, 2}).FrameDiffs({3, 1});
+  t[8].S(2).T(1).Dtis("-----D").ChainDiffs({5, 4, 3}).FrameDiffs({3, 1});
+  t[0].S(0).T(0).Dtis("SSRRRR").ChainDiffs({6, 5, 4}).FrameDiffs({6});
+  t[3].S(1).T(0).Dtis("--SSRR").ChainDiffs({1, 1, 1}).FrameDiffs({6, 1});
+  t[6].S(2).T(0).Dtis("----SS").ChainDiffs({2, 1, 1}).FrameDiffs({6, 1});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureL3T3::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 9;
+  structure.num_chains = 3;
+  structure.decode_target_protected_by_chain = {0, 0, 0, 1, 1, 1, 2, 2, 2};
+  auto& t = structure.templates;
+  t.resize(15);
+  // Templates are shown in the order frames following them appear in the
+  // stream, but in `structure.templates` array templates are sorted by
+  // (`spatial_id`, `temporal_id`) since that is a dependency descriptor
+  // requirement. Indexes are written in hex for nicer alignment.
+  t[0x1].S(0).T(0).Dtis("SSSSSSSSS").ChainDiffs({0, 0, 0});
+  t[0x6].S(1).T(0).Dtis("---SSSSSS").ChainDiffs({1, 1, 1}).FrameDiffs({1});
+  t[0xB].S(2).T(0).Dtis("------SSS").ChainDiffs({2, 1, 1}).FrameDiffs({1});
+  t[0x3].S(0).T(2).Dtis("--D--R--R").ChainDiffs({3, 2, 1}).FrameDiffs({3});
+  t[0x8].S(1).T(2).Dtis("-----D--R").ChainDiffs({4, 3, 2}).FrameDiffs({3, 1});
+  t[0xD].S(2).T(2).Dtis("--------D").ChainDiffs({5, 4, 3}).FrameDiffs({3, 1});
+  t[0x2].S(0).T(1).Dtis("-DS-RR-RR").ChainDiffs({6, 5, 4}).FrameDiffs({6});
+  t[0x7].S(1).T(1).Dtis("----DS-RR").ChainDiffs({7, 6, 5}).FrameDiffs({6, 1});
+  t[0xC].S(2).T(1).Dtis("-------DS").ChainDiffs({8, 7, 6}).FrameDiffs({6, 1});
+  t[0x4].S(0).T(2).Dtis("--D--R--R").ChainDiffs({9, 8, 7}).FrameDiffs({3});
+  t[0x9].S(1).T(2).Dtis("-----D--R").ChainDiffs({10, 9, 8}).FrameDiffs({3, 1});
+  t[0xE].S(2).T(2).Dtis("--------D").ChainDiffs({11, 10, 9}).FrameDiffs({3, 1});
+  t[0x0].S(0).T(0).Dtis("SSSRRRRRR").ChainDiffs({12, 11, 10}).FrameDiffs({12});
+  t[0x5].S(1).T(0).Dtis("---SSSRRR").ChainDiffs({1, 1, 1}).FrameDiffs({12, 1});
+  t[0xA].S(2).T(0).Dtis("------SSS").ChainDiffs({2, 1, 1}).FrameDiffs({12, 1});
+  return structure;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc.h b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc.h
new file mode 100644
index 0000000000..a4ede69342
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc.h
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_FULL_SVC_H_
+#define MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_FULL_SVC_H_
+
+#include <bitset>
+#include <vector>
+
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+
+namespace webrtc {
+
+class ScalabilityStructureFullSvc : public ScalableVideoController {
+ public:
+  struct ScalingFactor {
+    int num = 1;
+    int den = 2;
+  };
+  ScalabilityStructureFullSvc(int num_spatial_layers,
+                              int num_temporal_layers,
+                              ScalingFactor resolution_factor);
+  ~ScalabilityStructureFullSvc() override;
+
+  StreamLayersConfig StreamConfig() const override;
+
+  std::vector<LayerFrameConfig> NextFrameConfig(bool restart) override;
+  GenericFrameInfo OnEncodeDone(const LayerFrameConfig& config) override;
+  void OnRatesUpdated(const VideoBitrateAllocation& bitrates) override;
+
+ private:
+  enum FramePattern {
+    kNone,
+    kKey,
+    kDeltaT2A,
+    kDeltaT1,
+    kDeltaT2B,
+    kDeltaT0,
+  };
+  static constexpr absl::string_view kFramePatternNames[] = {
+      "None", "Key", "DeltaT2A", "DeltaT1", "DeltaT2B", "DeltaT0"};
+  static constexpr int kMaxNumSpatialLayers = 3;
+  static constexpr int kMaxNumTemporalLayers = 3;
+
+  // Index of the buffer to store last frame for layer (`sid`, `tid`)
+  int BufferIndex(int sid, int tid) const {
+    return tid * num_spatial_layers_ + sid;
+  }
+  bool DecodeTargetIsActive(int sid, int tid) const {
+    return active_decode_targets_[sid * num_temporal_layers_ + tid];
+  }
+  void SetDecodeTargetIsActive(int sid, int tid, bool value) {
+    active_decode_targets_.set(sid * num_temporal_layers_ + tid, value);
+  }
+  FramePattern NextPattern() const;
+  bool TemporalLayerIsActive(int tid) const;
+  static DecodeTargetIndication Dti(int sid,
+                                    int tid,
+                                    const LayerFrameConfig& frame);
+
+  const int num_spatial_layers_;
+  const int num_temporal_layers_;
+  const ScalingFactor resolution_factor_;
+
+  FramePattern last_pattern_ = kNone;
+  std::bitset<kMaxNumSpatialLayers> can_reference_t0_frame_for_spatial_id_ = 0;
+  std::bitset<kMaxNumSpatialLayers> can_reference_t1_frame_for_spatial_id_ = 0;
+  std::bitset<32> active_decode_targets_;
+};
+
+// T1       0   0
+//         /   /   / ...
+// T0     0---0---0--
+// Time-> 0 1 2 3 4
+class ScalabilityStructureL1T2 : public ScalabilityStructureFullSvc {
+ public:
+  explicit ScalabilityStructureL1T2(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureFullSvc(1, 2, resolution_factor) {}
+  ~ScalabilityStructureL1T2() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+// T2       0   0   0   0
+//          |  /    |  /
+// T1       / 0     / 0  ...
+//         |_/     |_/
+// T0     0-------0------
+// Time-> 0 1 2 3 4 5 6 7
+class ScalabilityStructureL1T3 : public ScalabilityStructureFullSvc {
+ public:
+  explicit ScalabilityStructureL1T3(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureFullSvc(1, 3, resolution_factor) {}
+  ~ScalabilityStructureL1T3() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+// S1  0--0--0-
+//     |  |  | ...
+// S0  0--0--0-
+class ScalabilityStructureL2T1 : public ScalabilityStructureFullSvc {
+ public:
+  explicit ScalabilityStructureL2T1(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureFullSvc(2, 1, resolution_factor) {}
+  ~ScalabilityStructureL2T1() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+// S1T1     0   0
+//         /|  /|  /
+// S1T0   0-+-0-+-0
+//        | | | | | ...
+// S0T1   | 0 | 0 |
+//        |/  |/  |/
+// S0T0   0---0---0--
+// Time-> 0 1 2 3 4
+class ScalabilityStructureL2T2 : public ScalabilityStructureFullSvc {
+ public:
+  explicit ScalabilityStructureL2T2(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureFullSvc(2, 2, resolution_factor) {}
+  ~ScalabilityStructureL2T2() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+// S1T2      4    ,8
+// S1T1    / |  6' |
+// S1T0   2--+-'+--+-...
+//        |  |  |  |
+// S0T2   |  3  | ,7
+// S0T1   | /   5'
+// S0T0   1----'-----...
+// Time-> 0  1  2  3
+class ScalabilityStructureL2T3 : public ScalabilityStructureFullSvc {
+ public:
+  explicit ScalabilityStructureL2T3(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureFullSvc(2, 3, resolution_factor) {}
+  ~ScalabilityStructureL2T3() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+// S2     0-0-0-
+//        | | |
+// S1     0-0-0-...
+//        | | |
+// S0     0-0-0-
+// Time-> 0 1 2
+class ScalabilityStructureL3T1 : public ScalabilityStructureFullSvc {
+ public:
+  explicit ScalabilityStructureL3T1(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureFullSvc(3, 1, resolution_factor) {}
+  ~ScalabilityStructureL3T1() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+// https://www.w3.org/TR/webrtc-svc/#L3T2*
+class ScalabilityStructureL3T2 : public ScalabilityStructureFullSvc {
+ public:
+  explicit ScalabilityStructureL3T2(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureFullSvc(3, 2, resolution_factor) {}
+  ~ScalabilityStructureL3T2() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+// https://www.w3.org/TR/webrtc-svc/#L3T3*
+class ScalabilityStructureL3T3 : public ScalabilityStructureFullSvc {
+ public:
+  explicit ScalabilityStructureL3T3(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureFullSvc(3, 3, resolution_factor) {}
+  ~ScalabilityStructureL3T3() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_FULL_SVC_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc_unittest.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc_unittest.cc
new file mode 100644
index 0000000000..1c0a8be8f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc_unittest.cc
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/scalability_structure_full_svc.h"
+
+#include <vector>
+
+#include "modules/video_coding/svc/scalability_structure_test_helpers.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::IsEmpty;
+using ::testing::SizeIs;
+
+TEST(ScalabilityStructureL3T3Test, SkipT0FrameByEncoderKeepsReferencesValid) {
+  std::vector<GenericFrameInfo> frames;
+  ScalabilityStructureL3T3 structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  // Only S0T0 decode target is enabled.
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/1, /*s1=*/0));
+  // Encoder generates S0T0 key frame.
+  wrapper.GenerateFrames(/*num_temporal_units=*/1, frames);
+  EXPECT_THAT(frames, SizeIs(1));
+  // Spatial layers 1 is enabled.
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/1, /*s1=*/1));
+  // Encoder tries to generate S0T0 and S1T0 delta frames but they are dropped.
+  structure.NextFrameConfig(/*restart=*/false);
+  // Encoder successfully generates S0T0 and S1T0 delta frames.
+  wrapper.GenerateFrames(/*num_temporal_units=*/1, frames);
+  EXPECT_THAT(frames, SizeIs(3));
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+TEST(ScalabilityStructureL3T3Test, SkipS1T1FrameKeepsStructureValid) {
+  ScalabilityStructureL3T3 structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/3, /*s1=*/3));
+  auto frames = wrapper.GenerateFrames(/*num_temporal_units=*/1);
+  EXPECT_THAT(frames, SizeIs(2));
+  EXPECT_EQ(frames[0].temporal_id, 0);
+
+  frames = wrapper.GenerateFrames(/*num_temporal_units=*/1);
+  EXPECT_THAT(frames, SizeIs(2));
+  EXPECT_EQ(frames[0].temporal_id, 2);
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/3, /*s1=*/0));
+  frames = wrapper.GenerateFrames(/*num_temporal_units=*/1);
+  EXPECT_THAT(frames, SizeIs(1));
+  EXPECT_EQ(frames[0].temporal_id, 1);
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/3, /*s1=*/3));
+  // Rely on checks inside GenerateFrames frame references are valid.
+  frames = wrapper.GenerateFrames(/*num_temporal_units=*/1);
+  EXPECT_THAT(frames, SizeIs(2));
+  EXPECT_EQ(frames[0].temporal_id, 2);
+}
+
+TEST(ScalabilityStructureL3T3Test, SkipT1FrameByEncoderKeepsReferencesValid) {
+  std::vector<GenericFrameInfo> frames;
+  ScalabilityStructureL3T3 structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  // 1st 2 temporal units (T0 and T2)
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  // Simulate T1 frame dropped by the encoder,
+  // i.e. retrieve config, but skip calling OnEncodeDone.
+  structure.NextFrameConfig(/*restart=*/false);
+  // one more temporal units (T2)
+  wrapper.GenerateFrames(/*num_temporal_units=*/1, frames);
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+TEST(ScalabilityStructureL3T3Test,
+     SkippingFrameReusePreviousFrameConfiguration) {
+  std::vector<GenericFrameInfo> frames;
+  ScalabilityStructureL3T3 structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  // 1st 2 temporal units (T0 and T2)
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  ASSERT_THAT(frames, SizeIs(6));
+  ASSERT_EQ(frames[0].temporal_id, 0);
+  ASSERT_EQ(frames[3].temporal_id, 2);
+
+  // Simulate a frame dropped by the encoder,
+  // i.e. retrieve config, but skip calling OnEncodeDone.
+  structure.NextFrameConfig(/*restart=*/false);
+  // two more temporal unit, expect temporal pattern continues
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  ASSERT_THAT(frames, SizeIs(12));
+  // Expect temporal pattern continues as if there were no dropped frames.
+  EXPECT_EQ(frames[6].temporal_id, 1);
+  EXPECT_EQ(frames[9].temporal_id, 2);
+}
+
+TEST(ScalabilityStructureL3T3Test, SwitchSpatialLayerBeforeT1Frame) {
+  ScalabilityStructureL3T3 structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/0));
+  EXPECT_THAT(wrapper.GenerateFrames(1), SizeIs(1));
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/0, /*s1=*/2));
+  auto frames = wrapper.GenerateFrames(1);
+  ASSERT_THAT(frames, SizeIs(1));
+  EXPECT_THAT(frames[0].frame_diffs, IsEmpty());
+  EXPECT_EQ(frames[0].temporal_id, 0);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc.cc
new file mode 100644
index 0000000000..0e6fecfae9
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc.cc
@@ -0,0 +1,427 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/scalability_structure_key_svc.h"
+
+#include <bitset>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+constexpr int ScalabilityStructureKeySvc::kMaxNumSpatialLayers;
+constexpr int ScalabilityStructureKeySvc::kMaxNumTemporalLayers;
+
+ScalabilityStructureKeySvc::ScalabilityStructureKeySvc(int num_spatial_layers,
+                                                       int num_temporal_layers)
+    : num_spatial_layers_(num_spatial_layers),
+      num_temporal_layers_(num_temporal_layers),
+      active_decode_targets_(
+          (uint32_t{1} << (num_spatial_layers * num_temporal_layers)) - 1) {
+  // There is no point to use this structure without spatial scalability.
+  RTC_DCHECK_GT(num_spatial_layers, 1);
+  RTC_DCHECK_LE(num_spatial_layers, kMaxNumSpatialLayers);
+  RTC_DCHECK_LE(num_temporal_layers, kMaxNumTemporalLayers);
+}
+
+ScalabilityStructureKeySvc::~ScalabilityStructureKeySvc() = default;
+
+ScalableVideoController::StreamLayersConfig
+ScalabilityStructureKeySvc::StreamConfig() const {
+  StreamLayersConfig result;
+  result.num_spatial_layers = num_spatial_layers_;
+  result.num_temporal_layers = num_temporal_layers_;
+  result.scaling_factor_num[num_spatial_layers_ - 1] = 1;
+  result.scaling_factor_den[num_spatial_layers_ - 1] = 1;
+  for (int sid = num_spatial_layers_ - 1; sid > 0; --sid) {
+    result.scaling_factor_num[sid - 1] = 1;
+    result.scaling_factor_den[sid - 1] = 2 * result.scaling_factor_den[sid];
+  }
+  result.uses_reference_scaling = true;
+  return result;
+}
+
+bool ScalabilityStructureKeySvc::TemporalLayerIsActive(int tid) const {
+  if (tid >= num_temporal_layers_) {
+    return false;
+  }
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    if (DecodeTargetIsActive(sid, tid)) {
+      return true;
+    }
+  }
+  return false;
+}
+
+DecodeTargetIndication ScalabilityStructureKeySvc::Dti(
+    int sid,
+    int tid,
+    const LayerFrameConfig& config) {
+  if (config.IsKeyframe() || config.Id() == kKey) {
+    RTC_DCHECK_EQ(config.TemporalId(), 0);
+    return sid < config.SpatialId() ? DecodeTargetIndication::kNotPresent
+                                    : DecodeTargetIndication::kSwitch;
+  }
+
+  if (sid != config.SpatialId() || tid < config.TemporalId()) {
+    return DecodeTargetIndication::kNotPresent;
+  }
+  if (tid == config.TemporalId() && tid > 0) {
+    return DecodeTargetIndication::kDiscardable;
+  }
+  return DecodeTargetIndication::kSwitch;
+}
+
+std::vector<ScalableVideoController::LayerFrameConfig>
+ScalabilityStructureKeySvc::KeyframeConfig() {
+  std::vector<LayerFrameConfig> configs;
+  configs.reserve(num_spatial_layers_);
+  absl::optional<int> spatial_dependency_buffer_id;
+  spatial_id_is_enabled_.reset();
+  // Disallow temporal references cross T0 on higher temporal layers.
+  can_reference_t1_frame_for_spatial_id_.reset();
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    if (!DecodeTargetIsActive(sid, /*tid=*/0)) {
+      continue;
+    }
+    configs.emplace_back();
+    ScalableVideoController::LayerFrameConfig& config = configs.back();
+    config.Id(kKey).S(sid).T(0);
+
+    if (spatial_dependency_buffer_id) {
+      config.Reference(*spatial_dependency_buffer_id);
+    } else {
+      config.Keyframe();
+    }
+    config.Update(BufferIndex(sid, /*tid=*/0));
+
+    spatial_id_is_enabled_.set(sid);
+    spatial_dependency_buffer_id = BufferIndex(sid, /*tid=*/0);
+  }
+  return configs;
+}
+
+std::vector<ScalableVideoController::LayerFrameConfig>
+ScalabilityStructureKeySvc::T0Config() {
+  std::vector<LayerFrameConfig> configs;
+  configs.reserve(num_spatial_layers_);
+  // Disallow temporal references cross T0 on higher temporal layers.
+  can_reference_t1_frame_for_spatial_id_.reset();
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    if (!DecodeTargetIsActive(sid, /*tid=*/0)) {
+      spatial_id_is_enabled_.reset(sid);
+      continue;
+    }
+    configs.emplace_back();
+    configs.back().Id(kDeltaT0).S(sid).T(0).ReferenceAndUpdate(
+        BufferIndex(sid, /*tid=*/0));
+  }
+  return configs;
+}
+
+std::vector<ScalableVideoController::LayerFrameConfig>
+ScalabilityStructureKeySvc::T1Config() {
+  std::vector<LayerFrameConfig> configs;
+  configs.reserve(num_spatial_layers_);
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    if (!DecodeTargetIsActive(sid, /*tid=*/1)) {
+      continue;
+    }
+    configs.emplace_back();
+    ScalableVideoController::LayerFrameConfig& config = configs.back();
+    config.Id(kDeltaT1).S(sid).T(1).Reference(BufferIndex(sid, /*tid=*/0));
+    if (num_temporal_layers_ > 2) {
+      config.Update(BufferIndex(sid, /*tid=*/1));
+    }
+  }
+  return configs;
+}
+
+std::vector<ScalableVideoController::LayerFrameConfig>
+ScalabilityStructureKeySvc::T2Config(FramePattern pattern) {
+  std::vector<LayerFrameConfig> configs;
+  configs.reserve(num_spatial_layers_);
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    if (!DecodeTargetIsActive(sid, /*tid=*/2)) {
+      continue;
+    }
+    configs.emplace_back();
+    ScalableVideoController::LayerFrameConfig& config = configs.back();
+    config.Id(pattern).S(sid).T(2);
+    if (can_reference_t1_frame_for_spatial_id_[sid]) {
+      config.Reference(BufferIndex(sid, /*tid=*/1));
+    } else {
+      config.Reference(BufferIndex(sid, /*tid=*/0));
+    }
+  }
+  return configs;
+}
+
+ScalabilityStructureKeySvc::FramePattern
+ScalabilityStructureKeySvc::NextPattern(FramePattern last_pattern) const {
+  switch (last_pattern) {
+    case kNone:
+      return kKey;
+    case kDeltaT2B:
+      return kDeltaT0;
+    case kDeltaT2A:
+      if (TemporalLayerIsActive(1)) {
+        return kDeltaT1;
+      }
+      return kDeltaT0;
+    case kDeltaT1:
+      if (TemporalLayerIsActive(2)) {
+        return kDeltaT2B;
+      }
+      return kDeltaT0;
+    case kDeltaT0:
+    case kKey:
+      if (TemporalLayerIsActive(2)) {
+        return kDeltaT2A;
+      }
+      if (TemporalLayerIsActive(1)) {
+        return kDeltaT1;
+      }
+      return kDeltaT0;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return kNone;
+}
+
+std::vector<ScalableVideoController::LayerFrameConfig>
+ScalabilityStructureKeySvc::NextFrameConfig(bool restart) {
+  if (active_decode_targets_.none()) {
+    last_pattern_ = kNone;
+    return {};
+  }
+
+  if (restart) {
+    last_pattern_ = kNone;
+  }
+
+  FramePattern current_pattern = NextPattern(last_pattern_);
+  switch (current_pattern) {
+    case kKey:
+      return KeyframeConfig();
+    case kDeltaT0:
+      return T0Config();
+    case kDeltaT1:
+      return T1Config();
+    case kDeltaT2A:
+    case kDeltaT2B:
+      return T2Config(current_pattern);
+    case kNone:
+      break;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return {};
+}
+
+GenericFrameInfo ScalabilityStructureKeySvc::OnEncodeDone(
+    const LayerFrameConfig& config) {
+  // When encoder drops all frames for a temporal unit, it is better to reuse
+  // old temporal pattern rather than switch to next one, thus switch to next
+  // pattern defered here from the `NextFrameConfig`.
+  // In particular creating VP9 references rely on this behavior.
+  last_pattern_ = static_cast<FramePattern>(config.Id());
+  if (config.TemporalId() == 1) {
+    can_reference_t1_frame_for_spatial_id_.set(config.SpatialId());
+  }
+
+  GenericFrameInfo frame_info;
+  frame_info.spatial_id = config.SpatialId();
+  frame_info.temporal_id = config.TemporalId();
+  frame_info.encoder_buffers = config.Buffers();
+  frame_info.decode_target_indications.reserve(num_spatial_layers_ *
+                                               num_temporal_layers_);
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    for (int tid = 0; tid < num_temporal_layers_; ++tid) {
+      frame_info.decode_target_indications.push_back(Dti(sid, tid, config));
+    }
+  }
+  frame_info.part_of_chain.assign(num_spatial_layers_, false);
+  if (config.IsKeyframe() || config.Id() == kKey) {
+    RTC_DCHECK_EQ(config.TemporalId(), 0);
+    for (int sid = config.SpatialId(); sid < num_spatial_layers_; ++sid) {
+      frame_info.part_of_chain[sid] = true;
+    }
+  } else if (config.TemporalId() == 0) {
+    frame_info.part_of_chain[config.SpatialId()] = true;
+  }
+  frame_info.active_decode_targets = active_decode_targets_;
+  return frame_info;
+}
+
+void ScalabilityStructureKeySvc::OnRatesUpdated(
+    const VideoBitrateAllocation& bitrates) {
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    // Enable/disable spatial layers independetely.
+    bool active = bitrates.GetBitrate(sid, /*tid=*/0) > 0;
+    SetDecodeTargetIsActive(sid, /*tid=*/0, active);
+    if (!spatial_id_is_enabled_[sid] && active) {
+      // Key frame is required to reenable any spatial layer.
+      last_pattern_ = kNone;
+    }
+
+    for (int tid = 1; tid < num_temporal_layers_; ++tid) {
+      // To enable temporal layer, require bitrates for lower temporal layers.
+      active = active && bitrates.GetBitrate(sid, tid) > 0;
+      SetDecodeTargetIsActive(sid, tid, active);
+    }
+  }
+}
+
+ScalabilityStructureL2T1Key::~ScalabilityStructureL2T1Key() = default;
+
+FrameDependencyStructure ScalabilityStructureL2T1Key::DependencyStructure()
+    const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 2;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 1};
+  structure.templates.resize(4);
+  structure.templates[0].S(0).Dtis("S-").ChainDiffs({2, 1}).FrameDiffs({2});
+  structure.templates[1].S(0).Dtis("SS").ChainDiffs({0, 0});
+  structure.templates[2].S(1).Dtis("-S").ChainDiffs({1, 2}).FrameDiffs({2});
+  structure.templates[3].S(1).Dtis("-S").ChainDiffs({1, 1}).FrameDiffs({1});
+  return structure;
+}
+
+ScalabilityStructureL2T2Key::~ScalabilityStructureL2T2Key() = default;
+
+FrameDependencyStructure ScalabilityStructureL2T2Key::DependencyStructure()
+    const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 4;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 0, 1, 1};
+  structure.templates.resize(6);
+  auto& templates = structure.templates;
+  templates[0].S(0).T(0).Dtis("SSSS").ChainDiffs({0, 0});
+  templates[1].S(0).T(0).Dtis("SS--").ChainDiffs({4, 3}).FrameDiffs({4});
+  templates[2].S(0).T(1).Dtis("-D--").ChainDiffs({2, 1}).FrameDiffs({2});
+  templates[3].S(1).T(0).Dtis("--SS").ChainDiffs({1, 1}).FrameDiffs({1});
+  templates[4].S(1).T(0).Dtis("--SS").ChainDiffs({1, 4}).FrameDiffs({4});
+  templates[5].S(1).T(1).Dtis("---D").ChainDiffs({3, 2}).FrameDiffs({2});
+  return structure;
+}
+
+ScalabilityStructureL2T3Key::~ScalabilityStructureL2T3Key() = default;
+
+FrameDependencyStructure ScalabilityStructureL2T3Key::DependencyStructure()
+    const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 6;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 0, 0, 1, 1, 1};
+  auto& templates = structure.templates;
+  templates.resize(10);
+  templates[0].S(0).T(0).Dtis("SSSSSS").ChainDiffs({0, 0});
+  templates[1].S(0).T(0).Dtis("SSS---").ChainDiffs({8, 7}).FrameDiffs({8});
+  templates[2].S(0).T(1).Dtis("-DS---").ChainDiffs({4, 3}).FrameDiffs({4});
+  templates[3].S(0).T(2).Dtis("--D---").ChainDiffs({2, 1}).FrameDiffs({2});
+  templates[4].S(0).T(2).Dtis("--D---").ChainDiffs({6, 5}).FrameDiffs({2});
+  templates[5].S(1).T(0).Dtis("---SSS").ChainDiffs({1, 1}).FrameDiffs({1});
+  templates[6].S(1).T(0).Dtis("---SSS").ChainDiffs({1, 8}).FrameDiffs({8});
+  templates[7].S(1).T(1).Dtis("----DS").ChainDiffs({5, 4}).FrameDiffs({4});
+  templates[8].S(1).T(2).Dtis("-----D").ChainDiffs({3, 2}).FrameDiffs({2});
+  templates[9].S(1).T(2).Dtis("-----D").ChainDiffs({7, 6}).FrameDiffs({2});
+  return structure;
+}
+
+ScalabilityStructureL3T1Key::~ScalabilityStructureL3T1Key() = default;
+
+FrameDependencyStructure ScalabilityStructureL3T1Key::DependencyStructure()
+    const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 3;
+  structure.num_chains = 3;
+  structure.decode_target_protected_by_chain = {0, 1, 2};
+  auto& t = structure.templates;
+  t.resize(6);
+  // Templates are shown in the order frames following them appear in the
+  // stream, but in `structure.templates` array templates are sorted by
+  // (`spatial_id`, `temporal_id`) since that is a dependency descriptor
+  // requirement.
+  t[1].S(0).Dtis("SSS").ChainDiffs({0, 0, 0});
+  t[3].S(1).Dtis("-SS").ChainDiffs({1, 1, 1}).FrameDiffs({1});
+  t[5].S(2).Dtis("--S").ChainDiffs({2, 1, 1}).FrameDiffs({1});
+  t[0].S(0).Dtis("S--").ChainDiffs({3, 2, 1}).FrameDiffs({3});
+  t[2].S(1).Dtis("-S-").ChainDiffs({1, 3, 2}).FrameDiffs({3});
+  t[4].S(2).Dtis("--S").ChainDiffs({2, 1, 3}).FrameDiffs({3});
+  return structure;
+}
+
+ScalabilityStructureL3T2Key::~ScalabilityStructureL3T2Key() = default;
+
+FrameDependencyStructure ScalabilityStructureL3T2Key::DependencyStructure()
+    const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 6;
+  structure.num_chains = 3;
+  structure.decode_target_protected_by_chain = {0, 0, 1, 1, 2, 2};
+  auto& t = structure.templates;
+  t.resize(9);
+  // Templates are shown in the order frames following them appear in the
+  // stream, but in `structure.templates` array templates are sorted by
+  // (`spatial_id`, `temporal_id`) since that is a dependency descriptor
+  // requirement.
+  t[1].S(0).T(0).Dtis("SSSSSS").ChainDiffs({0, 0, 0});
+  t[4].S(1).T(0).Dtis("--SSSS").ChainDiffs({1, 1, 1}).FrameDiffs({1});
+  t[7].S(2).T(0).Dtis("----SS").ChainDiffs({2, 1, 1}).FrameDiffs({1});
+  t[2].S(0).T(1).Dtis("-D----").ChainDiffs({3, 2, 1}).FrameDiffs({3});
+  t[5].S(1).T(1).Dtis("---D--").ChainDiffs({4, 3, 2}).FrameDiffs({3});
+  t[8].S(2).T(1).Dtis("-----D").ChainDiffs({5, 4, 3}).FrameDiffs({3});
+  t[0].S(0).T(0).Dtis("SS----").ChainDiffs({6, 5, 4}).FrameDiffs({6});
+  t[3].S(1).T(0).Dtis("--SS--").ChainDiffs({1, 6, 5}).FrameDiffs({6});
+  t[6].S(2).T(0).Dtis("----SS").ChainDiffs({2, 1, 6}).FrameDiffs({6});
+  return structure;
+}
+
+ScalabilityStructureL3T3Key::~ScalabilityStructureL3T3Key() = default;
+
+FrameDependencyStructure ScalabilityStructureL3T3Key::DependencyStructure()
+    const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 9;
+  structure.num_chains = 3;
+  structure.decode_target_protected_by_chain = {0, 0, 0, 1, 1, 1, 2, 2, 2};
+  auto& t = structure.templates;
+  t.resize(15);
+  // Templates are shown in the order frames following them appear in the
+  // stream, but in `structure.templates` array templates are sorted by
+  // (`spatial_id`, `temporal_id`) since that is a dependency descriptor
+  // requirement. Indexes are written in hex for nicer alignment.
+  t[0x0].S(0).T(0).Dtis("SSSSSSSSS").ChainDiffs({0, 0, 0});
+  t[0x5].S(1).T(0).Dtis("---SSSSSS").ChainDiffs({1, 1, 1}).FrameDiffs({1});
+  t[0xA].S(2).T(0).Dtis("------SSS").ChainDiffs({2, 1, 1}).FrameDiffs({1});
+  t[0x3].S(0).T(2).Dtis("--D------").ChainDiffs({3, 2, 1}).FrameDiffs({3});
+  t[0x8].S(1).T(2).Dtis("-----D---").ChainDiffs({4, 3, 2}).FrameDiffs({3});
+  t[0xD].S(2).T(2).Dtis("--------D").ChainDiffs({5, 4, 3}).FrameDiffs({3});
+  t[0x2].S(0).T(1).Dtis("-DS------").ChainDiffs({6, 5, 4}).FrameDiffs({6});
+  t[0x7].S(1).T(1).Dtis("----DS---").ChainDiffs({7, 6, 5}).FrameDiffs({6});
+  t[0xC].S(2).T(1).Dtis("-------DS").ChainDiffs({8, 7, 6}).FrameDiffs({6});
+  t[0x4].S(0).T(2).Dtis("--D------").ChainDiffs({9, 8, 7}).FrameDiffs({3});
+  t[0x9].S(1).T(2).Dtis("-----D---").ChainDiffs({10, 9, 8}).FrameDiffs({3});
+  t[0xE].S(2).T(2).Dtis("--------D").ChainDiffs({11, 10, 9}).FrameDiffs({3});
+  t[0x1].S(0).T(0).Dtis("SSS------").ChainDiffs({12, 11, 10}).FrameDiffs({12});
+  t[0x6].S(1).T(0).Dtis("---SSS---").ChainDiffs({1, 12, 11}).FrameDiffs({12});
+  t[0xB].S(2).T(0).Dtis("------SSS").ChainDiffs({2, 1, 12}).FrameDiffs({12});
+  return structure;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc.h b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc.h
new file mode 100644
index 0000000000..54760da431
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc.h
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_KEY_SVC_H_
+#define MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_KEY_SVC_H_
+
+#include <bitset>
+#include <vector>
+
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+
+namespace webrtc {
+
+class ScalabilityStructureKeySvc : public ScalableVideoController {
+ public:
+  ScalabilityStructureKeySvc(int num_spatial_layers, int num_temporal_layers);
+  ~ScalabilityStructureKeySvc() override;
+
+  StreamLayersConfig StreamConfig() const override;
+
+  std::vector<LayerFrameConfig> NextFrameConfig(bool restart) override;
+  GenericFrameInfo OnEncodeDone(const LayerFrameConfig& config) override;
+  void OnRatesUpdated(const VideoBitrateAllocation& bitrates) override;
+
+ private:
+  enum FramePattern : int {
+    kNone,
+    kKey,
+    kDeltaT0,
+    kDeltaT2A,
+    kDeltaT1,
+    kDeltaT2B,
+  };
+  static constexpr int kMaxNumSpatialLayers = 3;
+  static constexpr int kMaxNumTemporalLayers = 3;
+
+  // Index of the buffer to store last frame for layer (`sid`, `tid`)
+  int BufferIndex(int sid, int tid) const {
+    return tid * num_spatial_layers_ + sid;
+  }
+  bool DecodeTargetIsActive(int sid, int tid) const {
+    return active_decode_targets_[sid * num_temporal_layers_ + tid];
+  }
+  void SetDecodeTargetIsActive(int sid, int tid, bool value) {
+    active_decode_targets_.set(sid * num_temporal_layers_ + tid, value);
+  }
+  bool TemporalLayerIsActive(int tid) const;
+  static DecodeTargetIndication Dti(int sid,
+                                    int tid,
+                                    const LayerFrameConfig& config);
+
+  std::vector<LayerFrameConfig> KeyframeConfig();
+  std::vector<LayerFrameConfig> T0Config();
+  std::vector<LayerFrameConfig> T1Config();
+  std::vector<LayerFrameConfig> T2Config(FramePattern pattern);
+
+  FramePattern NextPattern(FramePattern last_pattern) const;
+
+  const int num_spatial_layers_;
+  const int num_temporal_layers_;
+
+  FramePattern last_pattern_ = kNone;
+  std::bitset<kMaxNumSpatialLayers> spatial_id_is_enabled_;
+  std::bitset<kMaxNumSpatialLayers> can_reference_t1_frame_for_spatial_id_;
+  std::bitset<32> active_decode_targets_;
+};
+
+// S1  0--0--0-
+//     |       ...
+// S0  0--0--0-
+class ScalabilityStructureL2T1Key : public ScalabilityStructureKeySvc {
+ public:
+  ScalabilityStructureL2T1Key() : ScalabilityStructureKeySvc(2, 1) {}
+  ~ScalabilityStructureL2T1Key() override;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+// S1T1     0   0
+//         /   /   /
+// S1T0   0---0---0
+//        |         ...
+// S0T1   | 0   0
+//        |/   /   /
+// S0T0   0---0---0
+// Time-> 0 1 2 3 4
+class ScalabilityStructureL2T2Key : public ScalabilityStructureKeySvc {
+ public:
+  ScalabilityStructureL2T2Key() : ScalabilityStructureKeySvc(2, 2) {}
+  ~ScalabilityStructureL2T2Key() override;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+class ScalabilityStructureL2T3Key : public ScalabilityStructureKeySvc {
+ public:
+  ScalabilityStructureL2T3Key() : ScalabilityStructureKeySvc(2, 3) {}
+  ~ScalabilityStructureL2T3Key() override;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+class ScalabilityStructureL3T1Key : public ScalabilityStructureKeySvc {
+ public:
+  ScalabilityStructureL3T1Key() : ScalabilityStructureKeySvc(3, 1) {}
+  ~ScalabilityStructureL3T1Key() override;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+class ScalabilityStructureL3T2Key : public ScalabilityStructureKeySvc {
+ public:
+  ScalabilityStructureL3T2Key() : ScalabilityStructureKeySvc(3, 2) {}
+  ~ScalabilityStructureL3T2Key() override;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+class ScalabilityStructureL3T3Key : public ScalabilityStructureKeySvc {
+ public:
+  ScalabilityStructureL3T3Key() : ScalabilityStructureKeySvc(3, 3) {}
+  ~ScalabilityStructureL3T3Key() override;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_KEY_SVC_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc
new file mode 100644
index 0000000000..5f923bb487
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc_unittest.cc
@@ -0,0 +1,245 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/scalability_structure_key_svc.h"
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/svc/scalability_structure_test_helpers.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+using ::testing::SizeIs;
+
+TEST(ScalabilityStructureL3T3KeyTest,
+     SkipingT1FrameOnOneSpatialLayerKeepsStructureValid) {
+  ScalabilityStructureL3T3Key structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/3, /*s1=*/3));
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  EXPECT_THAT(frames, SizeIs(4));
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/3, /*s1=*/1));
+  wrapper.GenerateFrames(/*num_temporal_units=*/1, frames);
+  EXPECT_THAT(frames, SizeIs(5));
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/3, /*s1=*/3));
+  wrapper.GenerateFrames(/*num_temporal_units=*/1, frames);
+  ASSERT_THAT(frames, SizeIs(7));
+
+  EXPECT_EQ(frames[0].temporal_id, 0);
+  EXPECT_EQ(frames[1].temporal_id, 0);
+  EXPECT_EQ(frames[2].temporal_id, 2);
+  EXPECT_EQ(frames[3].temporal_id, 2);
+  EXPECT_EQ(frames[4].temporal_id, 1);
+  EXPECT_EQ(frames[5].temporal_id, 2);
+  EXPECT_EQ(frames[6].temporal_id, 2);
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+TEST(ScalabilityStructureL3T3KeyTest,
+     SkipT1FrameByEncoderKeepsReferencesValid) {
+  std::vector<GenericFrameInfo> frames;
+  ScalabilityStructureL3T3Key structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  // 1st 2 temporal units (T0 and T2)
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  // Simulate T1 frame dropped by the encoder,
+  // i.e. retrieve config, but skip calling OnEncodeDone.
+  structure.NextFrameConfig(/*restart=*/false);
+  // one more temporal unit.
+  wrapper.GenerateFrames(/*num_temporal_units=*/1, frames);
+
+  EXPECT_THAT(frames, SizeIs(9));
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+TEST(ScalabilityStructureL3T3KeyTest,
+     SkippingFrameReusePreviousFrameConfiguration) {
+  std::vector<GenericFrameInfo> frames;
+  ScalabilityStructureL3T3Key structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  // 1st 2 temporal units (T0 and T2)
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  ASSERT_THAT(frames, SizeIs(6));
+  ASSERT_EQ(frames[0].temporal_id, 0);
+  ASSERT_EQ(frames[3].temporal_id, 2);
+
+  // Simulate a frame dropped by the encoder,
+  // i.e. retrieve config, but skip calling OnEncodeDone.
+  structure.NextFrameConfig(/*restart=*/false);
+  // two more temporal unit, expect temporal pattern continues
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  ASSERT_THAT(frames, SizeIs(12));
+  // Expect temporal pattern continues as if there were no dropped frames.
+  EXPECT_EQ(frames[6].temporal_id, 1);
+  EXPECT_EQ(frames[9].temporal_id, 2);
+}
+
+TEST(ScalabilityStructureL3T3KeyTest, SkippingKeyFrameTriggersNewKeyFrame) {
+  std::vector<GenericFrameInfo> frames;
+  ScalabilityStructureL3T3Key structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  // Ask for a key frame config, but do not return any frames
+  structure.NextFrameConfig(/*restart=*/false);
+
+  // Ask for more frames, expect they start with a key frame.
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  ASSERT_THAT(frames, SizeIs(6));
+  ASSERT_EQ(frames[0].temporal_id, 0);
+  ASSERT_EQ(frames[3].temporal_id, 2);
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+TEST(ScalabilityStructureL3T3KeyTest,
+     SkippingT2FrameAndDisablingT2LayerProduceT1AsNextFrame) {
+  std::vector<GenericFrameInfo> frames;
+  ScalabilityStructureL3T3Key structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  wrapper.GenerateFrames(/*num_temporal_units=*/1, frames);
+  // Ask for next (T2) frame config, but do not return any frames
+  auto config = structure.NextFrameConfig(/*restart=*/false);
+  ASSERT_THAT(config, Not(IsEmpty()));
+  ASSERT_EQ(config.front().TemporalId(), 2);
+
+  // Disable T2 layer,
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/2, /*s2=*/2));
+  // Expect instead of reusing unused config, T1 config is generated.
+  config = structure.NextFrameConfig(/*restart=*/false);
+  ASSERT_THAT(config, Not(IsEmpty()));
+  EXPECT_EQ(config.front().TemporalId(), 1);
+}
+
+TEST(ScalabilityStructureL3T3KeyTest, EnableT2LayerWhileProducingT1Frame) {
+  std::vector<GenericFrameInfo> frames;
+  ScalabilityStructureL3T3Key structure;
+  ScalabilityStructureWrapper wrapper(structure);
+
+  // Disable T2 layer,
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/2, /*s2=*/2));
+
+  // Generate the key frame.
+  wrapper.GenerateFrames(/*num_temporal_units=*/1, frames);
+  ASSERT_THAT(frames, SizeIs(3));
+  EXPECT_EQ(frames[0].temporal_id, 0);
+
+  // Ask for next (T1) frame config, but do not return any frames yet.
+  auto config = structure.NextFrameConfig(/*restart=*/false);
+  ASSERT_THAT(config, Not(IsEmpty()));
+  ASSERT_EQ(config.front().TemporalId(), 1);
+
+  // Reenable T2 layer.
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/3, /*s1=*/3, /*s2=*/3));
+
+  // Finish encoding previously requested config.
+  for (auto layer_config : config) {
+    GenericFrameInfo info = structure.OnEncodeDone(layer_config);
+    EXPECT_EQ(info.temporal_id, 1);
+    frames.push_back(info);
+  }
+  ASSERT_THAT(frames, SizeIs(6));
+
+  // Generate more frames, expect T2 pattern resumes.
+  wrapper.GenerateFrames(/*num_temporal_units=*/4, frames);
+  ASSERT_THAT(frames, SizeIs(18));
+  EXPECT_EQ(frames[6].temporal_id, 2);
+  EXPECT_EQ(frames[9].temporal_id, 0);
+  EXPECT_EQ(frames[12].temporal_id, 2);
+  EXPECT_EQ(frames[15].temporal_id, 1);
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+TEST(ScalabilityStructureL3T3KeyTest,
+     ReenablingSpatialLayerBeforeMissedT0FrameDoesntTriggerAKeyFrame) {
+  ScalabilityStructureL3T3Key structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/2));
+  wrapper.GenerateFrames(1, frames);
+  EXPECT_THAT(frames, SizeIs(2));
+  // Drop a spatial layer.
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/0));
+  wrapper.GenerateFrames(1, frames);
+  EXPECT_THAT(frames, SizeIs(3));
+  // Reenable a spatial layer before T0 frame is encoded.
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/2));
+  wrapper.GenerateFrames(1, frames);
+  EXPECT_THAT(frames, SizeIs(5));
+
+  EXPECT_EQ(frames[0].temporal_id, 0);
+  EXPECT_EQ(frames[1].temporal_id, 0);
+  EXPECT_EQ(frames[2].temporal_id, 1);
+  EXPECT_EQ(frames[3].temporal_id, 0);
+  EXPECT_EQ(frames[4].temporal_id, 0);
+  EXPECT_THAT(frames[3].frame_diffs, SizeIs(1));
+  EXPECT_THAT(frames[4].frame_diffs, SizeIs(1));
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+TEST(ScalabilityStructureL3T3KeyTest, ReenablingSpatialLayerTriggersKeyFrame) {
+  ScalabilityStructureL3T3Key structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  // Start with all spatial layers enabled.
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/2, /*s2=*/2));
+  wrapper.GenerateFrames(3, frames);
+  EXPECT_THAT(frames, SizeIs(9));
+  // Drop a spatial layer. Two remaining spatial layers should just continue.
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/0, /*s2=*/2));
+  wrapper.GenerateFrames(2, frames);
+  EXPECT_THAT(frames, SizeIs(13));
+  // Reenable spatial layer, expect a full restart.
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/2, /*s2=*/2));
+  wrapper.GenerateFrames(1, frames);
+  ASSERT_THAT(frames, SizeIs(16));
+
+  // First 3 temporal units with all spatial layers enabled.
+  EXPECT_EQ(frames[0].temporal_id, 0);
+  EXPECT_EQ(frames[3].temporal_id, 1);
+  EXPECT_EQ(frames[6].temporal_id, 0);
+  // 2 temporal units with spatial layer 1 disabled.
+  EXPECT_EQ(frames[9].spatial_id, 0);
+  EXPECT_EQ(frames[9].temporal_id, 1);
+  EXPECT_EQ(frames[10].spatial_id, 2);
+  EXPECT_EQ(frames[10].temporal_id, 1);
+  // T0 frames were encoded while spatial layer 1 is disabled.
+  EXPECT_EQ(frames[11].spatial_id, 0);
+  EXPECT_EQ(frames[11].temporal_id, 0);
+  EXPECT_EQ(frames[12].spatial_id, 2);
+  EXPECT_EQ(frames[12].temporal_id, 0);
+  // Key frame to reenable spatial layer 1.
+  EXPECT_THAT(frames[13].frame_diffs, IsEmpty());
+  EXPECT_THAT(frames[14].frame_diffs, ElementsAre(1));
+  EXPECT_THAT(frames[15].frame_diffs, ElementsAre(1));
+  EXPECT_EQ(frames[13].temporal_id, 0);
+  EXPECT_EQ(frames[14].temporal_id, 0);
+  EXPECT_EQ(frames[15].temporal_id, 0);
+  auto all_frames = rtc::MakeArrayView(frames.data(), frames.size());
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(all_frames.subview(0, 13)));
+  // Frames starting from the frame#13 should not reference any earlier frames.
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(all_frames.subview(13)));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.cc
new file mode 100644
index 0000000000..4d15942d3e
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.cc
@@ -0,0 +1,177 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/scalability_structure_l2t2_key_shift.h"
+
+#include <utility>
+#include <vector>
+
+#include "absl/base/macros.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+DecodeTargetIndication
+Dti(int sid, int tid, const ScalableVideoController::LayerFrameConfig& config) {
+  if (config.IsKeyframe()) {
+    RTC_DCHECK_EQ(config.TemporalId(), 0);
+    return sid < config.SpatialId() ? DecodeTargetIndication::kNotPresent
+                                    : DecodeTargetIndication::kSwitch;
+  }
+
+  if (sid != config.SpatialId() || tid < config.TemporalId()) {
+    return DecodeTargetIndication::kNotPresent;
+  }
+  if (tid == config.TemporalId() && tid > 0) {
+    return DecodeTargetIndication::kDiscardable;
+  }
+  return DecodeTargetIndication::kSwitch;
+}
+
+}  // namespace
+
+constexpr int ScalabilityStructureL2T2KeyShift::kNumSpatialLayers;
+constexpr int ScalabilityStructureL2T2KeyShift::kNumTemporalLayers;
+
+ScalabilityStructureL2T2KeyShift::~ScalabilityStructureL2T2KeyShift() = default;
+
+ScalableVideoController::StreamLayersConfig
+ScalabilityStructureL2T2KeyShift::StreamConfig() const {
+  StreamLayersConfig result;
+  result.num_spatial_layers = 2;
+  result.num_temporal_layers = 2;
+  result.scaling_factor_num[0] = 1;
+  result.scaling_factor_den[0] = 2;
+  result.uses_reference_scaling = true;
+  return result;
+}
+
+FrameDependencyStructure ScalabilityStructureL2T2KeyShift::DependencyStructure()
+    const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 4;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 0, 1, 1};
+  structure.templates.resize(7);
+  auto& templates = structure.templates;
+  templates[0].S(0).T(0).Dtis("SSSS").ChainDiffs({0, 0});
+  templates[1].S(0).T(0).Dtis("SS--").ChainDiffs({2, 1}).FrameDiffs({2});
+  templates[2].S(0).T(0).Dtis("SS--").ChainDiffs({4, 1}).FrameDiffs({4});
+  templates[3].S(0).T(1).Dtis("-D--").ChainDiffs({2, 3}).FrameDiffs({2});
+  templates[4].S(1).T(0).Dtis("--SS").ChainDiffs({1, 1}).FrameDiffs({1});
+  templates[5].S(1).T(0).Dtis("--SS").ChainDiffs({3, 4}).FrameDiffs({4});
+  templates[6].S(1).T(1).Dtis("---D").ChainDiffs({1, 2}).FrameDiffs({2});
+  return structure;
+}
+
+std::vector<ScalableVideoController::LayerFrameConfig>
+ScalabilityStructureL2T2KeyShift::NextFrameConfig(bool restart) {
+  std::vector<LayerFrameConfig> configs;
+  configs.reserve(2);
+  if (restart) {
+    next_pattern_ = kKey;
+  }
+
+  // Buffer0 keeps latest S0T0 frame,
+  // Buffer1 keeps latest S1T0 frame.
+  switch (next_pattern_) {
+    case kKey:
+      if (DecodeTargetIsActive(/*sid=*/0, /*tid=*/0)) {
+        configs.emplace_back();
+        configs.back().S(0).T(0).Update(0).Keyframe();
+      }
+      if (DecodeTargetIsActive(/*sid=*/1, /*tid=*/0)) {
+        configs.emplace_back();
+        configs.back().S(1).T(0).Update(1);
+        if (DecodeTargetIsActive(/*sid=*/0, /*tid=*/0)) {
+          configs.back().Reference(0);
+        } else {
+          configs.back().Keyframe();
+        }
+      }
+      next_pattern_ = kDelta0;
+      break;
+    case kDelta0:
+      if (DecodeTargetIsActive(/*sid=*/0, /*tid=*/0)) {
+        configs.emplace_back();
+        configs.back().S(0).T(0).ReferenceAndUpdate(0);
+      }
+      if (DecodeTargetIsActive(/*sid=*/1, /*tid=*/1)) {
+        configs.emplace_back();
+        configs.back().S(1).T(1).Reference(1);
+      }
+      if (configs.empty() && DecodeTargetIsActive(/*sid=*/1, /*tid=*/0)) {
+        configs.emplace_back();
+        configs.back().S(1).T(0).ReferenceAndUpdate(1);
+      }
+      next_pattern_ = kDelta1;
+      break;
+    case kDelta1:
+      if (DecodeTargetIsActive(/*sid=*/0, /*tid=*/1)) {
+        configs.emplace_back();
+        configs.back().S(0).T(1).Reference(0);
+      }
+      if (DecodeTargetIsActive(/*sid=*/1, /*tid=*/0)) {
+        configs.emplace_back();
+        configs.back().S(1).T(0).ReferenceAndUpdate(1);
+      }
+      if (configs.empty() && DecodeTargetIsActive(/*sid=*/0, /*tid=*/0)) {
+        configs.emplace_back();
+        configs.back().S(0).T(0).ReferenceAndUpdate(0);
+      }
+      next_pattern_ = kDelta0;
+      break;
+  }
+
+  RTC_DCHECK(!configs.empty() || active_decode_targets_.none());
+  return configs;
+}
+
+GenericFrameInfo ScalabilityStructureL2T2KeyShift::OnEncodeDone(
+    const LayerFrameConfig& config) {
+  GenericFrameInfo frame_info;
+  frame_info.spatial_id = config.SpatialId();
+  frame_info.temporal_id = config.TemporalId();
+  frame_info.encoder_buffers = config.Buffers();
+  for (int sid = 0; sid < kNumSpatialLayers; ++sid) {
+    for (int tid = 0; tid < kNumTemporalLayers; ++tid) {
+      frame_info.decode_target_indications.push_back(Dti(sid, tid, config));
+    }
+  }
+  if (config.IsKeyframe()) {
+    frame_info.part_of_chain = {true, true};
+  } else if (config.TemporalId() == 0) {
+    frame_info.part_of_chain = {config.SpatialId() == 0,
+                                config.SpatialId() == 1};
+  } else {
+    frame_info.part_of_chain = {false, false};
+  }
+  return frame_info;
+}
+
+void ScalabilityStructureL2T2KeyShift::OnRatesUpdated(
+    const VideoBitrateAllocation& bitrates) {
+  for (int sid = 0; sid < kNumSpatialLayers; ++sid) {
+    // Enable/disable spatial layers independetely.
+    bool active = bitrates.GetBitrate(sid, /*tid=*/0) > 0;
+    if (!DecodeTargetIsActive(sid, /*tid=*/0) && active) {
+      // Key frame is required to reenable any spatial layer.
+      next_pattern_ = kKey;
+    }
+
+    SetDecodeTargetIsActive(sid, /*tid=*/0, active);
+    SetDecodeTargetIsActive(sid, /*tid=*/1,
+                            active && bitrates.GetBitrate(sid, /*tid=*/1) > 0);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.h b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.h
new file mode 100644
index 0000000000..26d1afcb29
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_L2T2_KEY_SHIFT_H_
+#define MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_L2T2_KEY_SHIFT_H_
+
+#include <vector>
+
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+
+namespace webrtc {
+
+// S1T1     0   0
+//         /   /   /
+// S1T0   0---0---0
+//        |        ...
+// S0T1   |   0   0
+//        |  /   /
+// S0T0   0-0---0--
+// Time-> 0 1 2 3 4
+class ScalabilityStructureL2T2KeyShift : public ScalableVideoController {
+ public:
+  ~ScalabilityStructureL2T2KeyShift() override;
+
+  StreamLayersConfig StreamConfig() const override;
+  FrameDependencyStructure DependencyStructure() const override;
+
+  std::vector<LayerFrameConfig> NextFrameConfig(bool restart) override;
+  GenericFrameInfo OnEncodeDone(const LayerFrameConfig& config) override;
+  void OnRatesUpdated(const VideoBitrateAllocation& bitrates) override;
+
+ private:
+  enum FramePattern {
+    kKey,
+    kDelta0,
+    kDelta1,
+  };
+
+  static constexpr int kNumSpatialLayers = 2;
+  static constexpr int kNumTemporalLayers = 2;
+
+  bool DecodeTargetIsActive(int sid, int tid) const {
+    return active_decode_targets_[sid * kNumTemporalLayers + tid];
+  }
+  void SetDecodeTargetIsActive(int sid, int tid, bool value) {
+    active_decode_targets_.set(sid * kNumTemporalLayers + tid, value);
+  }
+
+  FramePattern next_pattern_ = kKey;
+  std::bitset<32> active_decode_targets_ = 0b1111;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_L2T2_KEY_SHIFT_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift_unittest.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift_unittest.cc
new file mode 100644
index 0000000000..40fecf1812
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift_unittest.cc
@@ -0,0 +1,358 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/scalability_structure_l2t2_key_shift.h"
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/svc/scalability_structure_test_helpers.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+using ::testing::SizeIs;
+
+// S1T1     3   7
+//         /   /
+// S1T0   1---5---9
+//        |
+// S0T1   |   4   8
+//        |  /   /
+// S0T0   0-2---6
+// Time-> 0 1 2 3 4
+TEST(ScalabilityStructureL2T2KeyShiftTest, DecodeTargetsAreEnabledByDefault) {
+  ScalabilityStructureL2T2KeyShift structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+  wrapper.GenerateFrames(/*num_temporal_units=*/5, frames);
+  ASSERT_THAT(frames, SizeIs(10));
+
+  EXPECT_EQ(frames[0].spatial_id, 0);
+  EXPECT_EQ(frames[1].spatial_id, 1);
+  EXPECT_EQ(frames[2].spatial_id, 0);
+  EXPECT_EQ(frames[3].spatial_id, 1);
+  EXPECT_EQ(frames[4].spatial_id, 0);
+  EXPECT_EQ(frames[5].spatial_id, 1);
+  EXPECT_EQ(frames[6].spatial_id, 0);
+  EXPECT_EQ(frames[7].spatial_id, 1);
+  EXPECT_EQ(frames[8].spatial_id, 0);
+  EXPECT_EQ(frames[9].spatial_id, 1);
+
+  // spatial_id = 0 has the temporal shift.
+  EXPECT_EQ(frames[0].temporal_id, 0);
+  EXPECT_EQ(frames[2].temporal_id, 0);
+  EXPECT_EQ(frames[4].temporal_id, 1);
+  EXPECT_EQ(frames[6].temporal_id, 0);
+  EXPECT_EQ(frames[8].temporal_id, 1);
+
+  // spatial_id = 1 hasn't temporal shift.
+  EXPECT_EQ(frames[1].temporal_id, 0);
+  EXPECT_EQ(frames[3].temporal_id, 1);
+  EXPECT_EQ(frames[5].temporal_id, 0);
+  EXPECT_EQ(frames[7].temporal_id, 1);
+  EXPECT_EQ(frames[9].temporal_id, 0);
+
+  // Key frame diff.
+  EXPECT_THAT(frames[0].frame_diffs, IsEmpty());
+  EXPECT_THAT(frames[1].frame_diffs, ElementsAre(1));
+  // S0T0 frame diffs
+  EXPECT_THAT(frames[2].frame_diffs, ElementsAre(2));
+  EXPECT_THAT(frames[6].frame_diffs, ElementsAre(4));
+  // S1T0 frame diffs
+  EXPECT_THAT(frames[5].frame_diffs, ElementsAre(4));
+  EXPECT_THAT(frames[9].frame_diffs, ElementsAre(4));
+  // T1 frames refer T0 frame of same spatial layer which is 2 frame ids away.
+  EXPECT_THAT(frames[3].frame_diffs, ElementsAre(2));
+  EXPECT_THAT(frames[4].frame_diffs, ElementsAre(2));
+  EXPECT_THAT(frames[7].frame_diffs, ElementsAre(2));
+  EXPECT_THAT(frames[8].frame_diffs, ElementsAre(2));
+}
+
+// S1T0   1---4---7
+//        |
+// S0T1   |   3   6
+//        |  /   /
+// S0T0   0-2---5--
+// Time-> 0 1 2 3 4
+TEST(ScalabilityStructureL2T2KeyShiftTest, DisableS1T1Layer) {
+  ScalabilityStructureL2T2KeyShift structure;
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/1));
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+  wrapper.GenerateFrames(/*num_temporal_units=*/5, frames);
+  ASSERT_THAT(frames, SizeIs(8));
+
+  EXPECT_EQ(frames[0].spatial_id, 0);
+  EXPECT_EQ(frames[1].spatial_id, 1);
+  EXPECT_EQ(frames[2].spatial_id, 0);
+  EXPECT_EQ(frames[3].spatial_id, 0);
+  EXPECT_EQ(frames[4].spatial_id, 1);
+  EXPECT_EQ(frames[5].spatial_id, 0);
+  EXPECT_EQ(frames[6].spatial_id, 0);
+  EXPECT_EQ(frames[7].spatial_id, 1);
+
+  // spatial_id = 0 has the temporal shift.
+  EXPECT_EQ(frames[0].temporal_id, 0);
+  EXPECT_EQ(frames[2].temporal_id, 0);
+  EXPECT_EQ(frames[3].temporal_id, 1);
+  EXPECT_EQ(frames[5].temporal_id, 0);
+  EXPECT_EQ(frames[6].temporal_id, 1);
+
+  // spatial_id = 1 has single temporal layer.
+  EXPECT_EQ(frames[1].temporal_id, 0);
+  EXPECT_EQ(frames[4].temporal_id, 0);
+  EXPECT_EQ(frames[5].temporal_id, 0);
+}
+
+// S1T1     3  |
+//         /   |
+// S1T0   1---5+--7
+//        |    |
+// S0T1   |   4|
+//        |  / |
+// S0T0   0-2--+6---8
+// Time-> 0 1 2 3 4 5
+TEST(ScalabilityStructureL2T2KeyShiftTest, DisableT1LayersAfterFewFrames) {
+  ScalabilityStructureL2T2KeyShift structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  wrapper.GenerateFrames(/*num_temporal_units=*/3, frames);
+  EXPECT_THAT(frames, SizeIs(6));
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/1, /*s1=*/1));
+  wrapper.GenerateFrames(/*num_temporal_units=*/3, frames);
+  ASSERT_THAT(frames, SizeIs(9));
+
+  // Skip validation before T1 was disabled as that is covered by the test
+  // where no layers are disabled.
+  EXPECT_EQ(frames[6].spatial_id, 0);
+  EXPECT_EQ(frames[7].spatial_id, 1);
+  EXPECT_EQ(frames[8].spatial_id, 0);
+
+  EXPECT_EQ(frames[6].temporal_id, 0);
+  EXPECT_EQ(frames[7].temporal_id, 0);
+  EXPECT_EQ(frames[8].temporal_id, 0);
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+// S1T1     1   3
+//         /   /
+// S1T0   0---2
+// Time-> 0 1 2 3 4 5
+TEST(ScalabilityStructureL2T2KeyShiftTest, DisableS0FromTheStart) {
+  ScalabilityStructureL2T2KeyShift structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/0, /*s1=*/2));
+  wrapper.GenerateFrames(/*num_temporal_units=*/4, frames);
+  EXPECT_THAT(frames, SizeIs(4));
+
+  EXPECT_EQ(frames[0].spatial_id, 1);
+  EXPECT_EQ(frames[1].spatial_id, 1);
+  EXPECT_EQ(frames[2].spatial_id, 1);
+  EXPECT_EQ(frames[3].spatial_id, 1);
+
+  EXPECT_EQ(frames[0].temporal_id, 0);
+  EXPECT_EQ(frames[1].temporal_id, 1);
+  EXPECT_EQ(frames[2].temporal_id, 0);
+  EXPECT_EQ(frames[3].temporal_id, 1);
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+// S1T1     3  |6   8
+//         /   /   /
+// S1T0   1---5+--7
+//        |    |
+// S0T1   |   4|
+//        |  / |
+// S0T0   0-2  |
+// Time-> 0 1 2 3 4 5
+TEST(ScalabilityStructureL2T2KeyShiftTest, DisableS0AfterFewFrames) {
+  ScalabilityStructureL2T2KeyShift structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  wrapper.GenerateFrames(/*num_temporal_units=*/3, frames);
+  EXPECT_THAT(frames, SizeIs(6));
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/0, /*s1=*/2));
+  wrapper.GenerateFrames(/*num_temporal_units=*/3, frames);
+  ASSERT_THAT(frames, SizeIs(9));
+
+  // Expect frame[6] is delta frame.
+  EXPECT_THAT(frames[6].frame_diffs, ElementsAre(1));
+  // Skip validation before S0 was disabled as that should be covered by
+  // test where no layers are disabled.
+  EXPECT_EQ(frames[6].spatial_id, 1);
+  EXPECT_EQ(frames[7].spatial_id, 1);
+  EXPECT_EQ(frames[8].spatial_id, 1);
+
+  EXPECT_EQ(frames[6].temporal_id, 1);
+  EXPECT_EQ(frames[7].temporal_id, 0);
+  EXPECT_EQ(frames[8].temporal_id, 1);
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+// S1T1     3| |  8
+//         / | | /
+// S1T0   1  | |6
+//        |  | ||
+// S0T1   |  |4||
+//        |  / ||
+// S0T0   0-2| |5-7
+// Time-> 0 1 2 3 4 5
+TEST(ScalabilityStructureL2T2KeyShiftTest, ReenableS1TriggersKeyFrame) {
+  ScalabilityStructureL2T2KeyShift structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  EXPECT_THAT(frames, SizeIs(4));
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/0));
+  wrapper.GenerateFrames(/*num_temporal_units=*/1, frames);
+  EXPECT_THAT(frames, SizeIs(5));
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/2, /*s1=*/2));
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  ASSERT_THAT(frames, SizeIs(9));
+
+  EXPECT_THAT(frames[4].spatial_id, 0);
+  EXPECT_THAT(frames[4].temporal_id, 1);
+
+  // Expect frame[5] to be a key frame.
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(
+      rtc::MakeArrayView(frames.data() + 5, 4)));
+
+  EXPECT_THAT(frames[5].spatial_id, 0);
+  EXPECT_THAT(frames[6].spatial_id, 1);
+  EXPECT_THAT(frames[7].spatial_id, 0);
+  EXPECT_THAT(frames[8].spatial_id, 1);
+
+  // S0 should do temporal shift after the key frame.
+  EXPECT_THAT(frames[5].temporal_id, 0);
+  EXPECT_THAT(frames[7].temporal_id, 0);
+
+  // No temporal shift for the top spatial layer.
+  EXPECT_THAT(frames[6].temporal_id, 0);
+  EXPECT_THAT(frames[8].temporal_id, 1);
+}
+
+TEST(ScalabilityStructureL2T2KeyShiftTest, EnableOnlyS0T0FromTheStart) {
+  ScalabilityStructureL2T2KeyShift structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/1, /*s1=*/0));
+  wrapper.GenerateFrames(/*num_temporal_units=*/3, frames);
+  ASSERT_THAT(frames, SizeIs(3));
+
+  EXPECT_EQ(frames[0].spatial_id, 0);
+  EXPECT_EQ(frames[1].spatial_id, 0);
+  EXPECT_EQ(frames[2].spatial_id, 0);
+
+  EXPECT_EQ(frames[0].temporal_id, 0);
+  EXPECT_EQ(frames[1].temporal_id, 0);
+  EXPECT_EQ(frames[2].temporal_id, 0);
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+// S1T1     3|
+//         / |
+// S1T0   1  |
+//        |  |
+// S0T1   |  |
+//        |  |
+// S0T0   0-2+4-5-6
+// Time-> 0 1 2 3 4
+TEST(ScalabilityStructureL2T2KeyShiftTest, EnableOnlyS0T0AfterFewFrames) {
+  ScalabilityStructureL2T2KeyShift structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  EXPECT_THAT(frames, SizeIs(4));
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/1, /*s1=*/0));
+  wrapper.GenerateFrames(/*num_temporal_units=*/3, frames);
+  ASSERT_THAT(frames, SizeIs(7));
+
+  EXPECT_EQ(frames[4].spatial_id, 0);
+  EXPECT_EQ(frames[5].spatial_id, 0);
+  EXPECT_EQ(frames[6].spatial_id, 0);
+
+  EXPECT_EQ(frames[4].temporal_id, 0);
+  EXPECT_EQ(frames[5].temporal_id, 0);
+  EXPECT_EQ(frames[6].temporal_id, 0);
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+TEST(ScalabilityStructureL2T2KeyShiftTest, EnableOnlyS1T0FromTheStart) {
+  ScalabilityStructureL2T2KeyShift structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/0, /*s1=*/1));
+  wrapper.GenerateFrames(/*num_temporal_units=*/3, frames);
+  ASSERT_THAT(frames, SizeIs(3));
+
+  EXPECT_EQ(frames[0].spatial_id, 1);
+  EXPECT_EQ(frames[1].spatial_id, 1);
+  EXPECT_EQ(frames[2].spatial_id, 1);
+
+  EXPECT_EQ(frames[0].temporal_id, 0);
+  EXPECT_EQ(frames[1].temporal_id, 0);
+  EXPECT_EQ(frames[2].temporal_id, 0);
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+// S1T1     3|
+//         / |
+// S1T0   1--+4-5-6
+//        |  |
+// S0T1   |  |
+//        |  |
+// S0T0   0-2|
+// Time-> 0 1 2 3 4
+TEST(ScalabilityStructureL2T2KeyShiftTest, EnableOnlyS1T0AfterFewFrames) {
+  ScalabilityStructureL2T2KeyShift structure;
+  ScalabilityStructureWrapper wrapper(structure);
+  std::vector<GenericFrameInfo> frames;
+
+  wrapper.GenerateFrames(/*num_temporal_units=*/2, frames);
+  EXPECT_THAT(frames, SizeIs(4));
+  structure.OnRatesUpdated(EnableTemporalLayers(/*s0=*/0, /*s1=*/1));
+  wrapper.GenerateFrames(/*num_temporal_units=*/3, frames);
+  ASSERT_THAT(frames, SizeIs(7));
+
+  EXPECT_EQ(frames[4].spatial_id, 1);
+  EXPECT_EQ(frames[5].spatial_id, 1);
+  EXPECT_EQ(frames[6].spatial_id, 1);
+
+  EXPECT_EQ(frames[4].temporal_id, 0);
+  EXPECT_EQ(frames[5].temporal_id, 0);
+  EXPECT_EQ(frames[6].temporal_id, 0);
+
+  EXPECT_TRUE(wrapper.FrameReferencesAreValid(frames));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_simulcast.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_simulcast.cc
new file mode 100644
index 0000000000..54e27fda5c
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_simulcast.cc
@@ -0,0 +1,353 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/scalability_structure_simulcast.h"
+
+#include <utility>
+#include <vector>
+
+#include "absl/base/macros.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+DecodeTargetIndication
+Dti(int sid, int tid, const ScalableVideoController::LayerFrameConfig& config) {
+  if (sid != config.SpatialId() || tid < config.TemporalId()) {
+    return DecodeTargetIndication::kNotPresent;
+  }
+  if (tid == 0) {
+    RTC_DCHECK_EQ(config.TemporalId(), 0);
+    return DecodeTargetIndication::kSwitch;
+  }
+  if (tid == config.TemporalId()) {
+    return DecodeTargetIndication::kDiscardable;
+  }
+  RTC_DCHECK_GT(tid, config.TemporalId());
+  return DecodeTargetIndication::kSwitch;
+}
+
+}  // namespace
+
+constexpr int ScalabilityStructureSimulcast::kMaxNumSpatialLayers;
+constexpr int ScalabilityStructureSimulcast::kMaxNumTemporalLayers;
+
+ScalabilityStructureSimulcast::ScalabilityStructureSimulcast(
+    int num_spatial_layers,
+    int num_temporal_layers,
+    ScalingFactor resolution_factor)
+    : num_spatial_layers_(num_spatial_layers),
+      num_temporal_layers_(num_temporal_layers),
+      resolution_factor_(resolution_factor),
+      active_decode_targets_(
+          (uint32_t{1} << (num_spatial_layers * num_temporal_layers)) - 1) {
+  RTC_DCHECK_LE(num_spatial_layers, kMaxNumSpatialLayers);
+  RTC_DCHECK_LE(num_temporal_layers, kMaxNumTemporalLayers);
+}
+
+ScalabilityStructureSimulcast::~ScalabilityStructureSimulcast() = default;
+
+ScalableVideoController::StreamLayersConfig
+ScalabilityStructureSimulcast::StreamConfig() const {
+  StreamLayersConfig result;
+  result.num_spatial_layers = num_spatial_layers_;
+  result.num_temporal_layers = num_temporal_layers_;
+  result.scaling_factor_num[num_spatial_layers_ - 1] = 1;
+  result.scaling_factor_den[num_spatial_layers_ - 1] = 1;
+  for (int sid = num_spatial_layers_ - 1; sid > 0; --sid) {
+    result.scaling_factor_num[sid - 1] =
+        resolution_factor_.num * result.scaling_factor_num[sid];
+    result.scaling_factor_den[sid - 1] =
+        resolution_factor_.den * result.scaling_factor_den[sid];
+  }
+  result.uses_reference_scaling = false;
+  return result;
+}
+
+bool ScalabilityStructureSimulcast::TemporalLayerIsActive(int tid) const {
+  if (tid >= num_temporal_layers_) {
+    return false;
+  }
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    if (DecodeTargetIsActive(sid, tid)) {
+      return true;
+    }
+  }
+  return false;
+}
+
+ScalabilityStructureSimulcast::FramePattern
+ScalabilityStructureSimulcast::NextPattern() const {
+  switch (last_pattern_) {
+    case kNone:
+    case kDeltaT2B:
+      return kDeltaT0;
+    case kDeltaT2A:
+      if (TemporalLayerIsActive(1)) {
+        return kDeltaT1;
+      }
+      return kDeltaT0;
+    case kDeltaT1:
+      if (TemporalLayerIsActive(2)) {
+        return kDeltaT2B;
+      }
+      return kDeltaT0;
+    case kDeltaT0:
+      if (TemporalLayerIsActive(2)) {
+        return kDeltaT2A;
+      }
+      if (TemporalLayerIsActive(1)) {
+        return kDeltaT1;
+      }
+      return kDeltaT0;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return kDeltaT0;
+}
+
+std::vector<ScalableVideoController::LayerFrameConfig>
+ScalabilityStructureSimulcast::NextFrameConfig(bool restart) {
+  std::vector<LayerFrameConfig> configs;
+  if (active_decode_targets_.none()) {
+    last_pattern_ = kNone;
+    return configs;
+  }
+  configs.reserve(num_spatial_layers_);
+
+  if (last_pattern_ == kNone || restart) {
+    can_reference_t0_frame_for_spatial_id_.reset();
+    last_pattern_ = kNone;
+  }
+  FramePattern current_pattern = NextPattern();
+
+  switch (current_pattern) {
+    case kDeltaT0:
+      // Disallow temporal references cross T0 on higher temporal layers.
+      can_reference_t1_frame_for_spatial_id_.reset();
+      for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+        if (!DecodeTargetIsActive(sid, /*tid=*/0)) {
+          // Next frame from the spatial layer `sid` shouldn't depend on
+          // potentially old previous frame from the spatial layer `sid`.
+          can_reference_t0_frame_for_spatial_id_.reset(sid);
+          continue;
+        }
+        configs.emplace_back();
+        ScalableVideoController::LayerFrameConfig& config = configs.back();
+        config.Id(current_pattern).S(sid).T(0);
+
+        if (can_reference_t0_frame_for_spatial_id_[sid]) {
+          config.ReferenceAndUpdate(BufferIndex(sid, /*tid=*/0));
+        } else {
+          config.Keyframe().Update(BufferIndex(sid, /*tid=*/0));
+        }
+        can_reference_t0_frame_for_spatial_id_.set(sid);
+      }
+      break;
+    case kDeltaT1:
+      for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+        if (!DecodeTargetIsActive(sid, /*tid=*/1) ||
+            !can_reference_t0_frame_for_spatial_id_[sid]) {
+          continue;
+        }
+        configs.emplace_back();
+        ScalableVideoController::LayerFrameConfig& config = configs.back();
+        config.Id(current_pattern)
+            .S(sid)
+            .T(1)
+            .Reference(BufferIndex(sid, /*tid=*/0));
+        // Save frame only if there is a higher temporal layer that may need it.
+        if (num_temporal_layers_ > 2) {
+          config.Update(BufferIndex(sid, /*tid=*/1));
+        }
+      }
+      break;
+    case kDeltaT2A:
+    case kDeltaT2B:
+      for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+        if (!DecodeTargetIsActive(sid, /*tid=*/2) ||
+            !can_reference_t0_frame_for_spatial_id_[sid]) {
+          continue;
+        }
+        configs.emplace_back();
+        ScalableVideoController::LayerFrameConfig& config = configs.back();
+        config.Id(current_pattern).S(sid).T(2);
+        if (can_reference_t1_frame_for_spatial_id_[sid]) {
+          config.Reference(BufferIndex(sid, /*tid=*/1));
+        } else {
+          config.Reference(BufferIndex(sid, /*tid=*/0));
+        }
+      }
+      break;
+    case kNone:
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+
+  return configs;
+}
+
+GenericFrameInfo ScalabilityStructureSimulcast::OnEncodeDone(
+    const LayerFrameConfig& config) {
+  last_pattern_ = static_cast<FramePattern>(config.Id());
+  if (config.TemporalId() == 1) {
+    can_reference_t1_frame_for_spatial_id_.set(config.SpatialId());
+  }
+  GenericFrameInfo frame_info;
+  frame_info.spatial_id = config.SpatialId();
+  frame_info.temporal_id = config.TemporalId();
+  frame_info.encoder_buffers = config.Buffers();
+  frame_info.decode_target_indications.reserve(num_spatial_layers_ *
+                                               num_temporal_layers_);
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    for (int tid = 0; tid < num_temporal_layers_; ++tid) {
+      frame_info.decode_target_indications.push_back(Dti(sid, tid, config));
+    }
+  }
+  frame_info.part_of_chain.assign(num_spatial_layers_, false);
+  if (config.TemporalId() == 0) {
+    frame_info.part_of_chain[config.SpatialId()] = true;
+  }
+  frame_info.active_decode_targets = active_decode_targets_;
+  return frame_info;
+}
+
+void ScalabilityStructureSimulcast::OnRatesUpdated(
+    const VideoBitrateAllocation& bitrates) {
+  for (int sid = 0; sid < num_spatial_layers_; ++sid) {
+    // Enable/disable spatial layers independetely.
+    bool active = true;
+    for (int tid = 0; tid < num_temporal_layers_; ++tid) {
+      // To enable temporal layer, require bitrates for lower temporal layers.
+      active = active && bitrates.GetBitrate(sid, tid) > 0;
+      SetDecodeTargetIsActive(sid, tid, active);
+    }
+  }
+}
+
+FrameDependencyStructure ScalabilityStructureS2T1::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 2;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 1};
+  structure.templates.resize(4);
+  structure.templates[0].S(0).Dtis("S-").ChainDiffs({2, 1}).FrameDiffs({2});
+  structure.templates[1].S(0).Dtis("S-").ChainDiffs({0, 0});
+  structure.templates[2].S(1).Dtis("-S").ChainDiffs({1, 2}).FrameDiffs({2});
+  structure.templates[3].S(1).Dtis("-S").ChainDiffs({1, 0});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureS2T2::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 4;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 0, 1, 1};
+  auto& t = structure.templates;
+  t.resize(6);
+  t[1].S(0).T(0).Dtis("SS--").ChainDiffs({0, 0});
+  t[4].S(1).T(0).Dtis("--SS").ChainDiffs({1, 0});
+  t[2].S(0).T(1).Dtis("-D--").ChainDiffs({2, 1}).FrameDiffs({2});
+  t[5].S(1).T(1).Dtis("---D").ChainDiffs({3, 2}).FrameDiffs({2});
+  t[0].S(0).T(0).Dtis("SS--").ChainDiffs({4, 3}).FrameDiffs({4});
+  t[3].S(1).T(0).Dtis("--SS").ChainDiffs({1, 4}).FrameDiffs({4});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureS2T3::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 6;
+  structure.num_chains = 2;
+  structure.decode_target_protected_by_chain = {0, 0, 0, 1, 1, 1};
+  auto& t = structure.templates;
+  t.resize(10);
+  t[1].S(0).T(0).Dtis("SSS---").ChainDiffs({0, 0});
+  t[6].S(1).T(0).Dtis("---SSS").ChainDiffs({1, 0});
+  t[3].S(0).T(2).Dtis("--D---").ChainDiffs({2, 1}).FrameDiffs({2});
+  t[8].S(1).T(2).Dtis("-----D").ChainDiffs({3, 2}).FrameDiffs({2});
+  t[2].S(0).T(1).Dtis("-DS---").ChainDiffs({4, 3}).FrameDiffs({4});
+  t[7].S(1).T(1).Dtis("----DS").ChainDiffs({5, 4}).FrameDiffs({4});
+  t[4].S(0).T(2).Dtis("--D---").ChainDiffs({6, 5}).FrameDiffs({2});
+  t[9].S(1).T(2).Dtis("-----D").ChainDiffs({7, 6}).FrameDiffs({2});
+  t[0].S(0).T(0).Dtis("SSS---").ChainDiffs({8, 7}).FrameDiffs({8});
+  t[5].S(1).T(0).Dtis("---SSS").ChainDiffs({1, 8}).FrameDiffs({8});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureS3T1::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 3;
+  structure.num_chains = 3;
+  structure.decode_target_protected_by_chain = {0, 1, 2};
+  auto& t = structure.templates;
+  t.resize(6);
+  t[1].S(0).T(0).Dtis("S--").ChainDiffs({0, 0, 0});
+  t[3].S(1).T(0).Dtis("-S-").ChainDiffs({1, 0, 0});
+  t[5].S(2).T(0).Dtis("--S").ChainDiffs({2, 1, 0});
+  t[0].S(0).T(0).Dtis("S--").ChainDiffs({3, 2, 1}).FrameDiffs({3});
+  t[2].S(1).T(0).Dtis("-S-").ChainDiffs({1, 3, 2}).FrameDiffs({3});
+  t[4].S(2).T(0).Dtis("--S").ChainDiffs({2, 1, 3}).FrameDiffs({3});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureS3T2::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 6;
+  structure.num_chains = 3;
+  structure.decode_target_protected_by_chain = {0, 0, 1, 1, 2, 2};
+  auto& t = structure.templates;
+  t.resize(9);
+  // Templates are shown in the order frames following them appear in the
+  // stream, but in `structure.templates` array templates are sorted by
+  // (`spatial_id`, `temporal_id`) since that is a dependency descriptor
+  // requirement.
+  t[1].S(0).T(0).Dtis("SS----").ChainDiffs({0, 0, 0});
+  t[4].S(1).T(0).Dtis("--SS--").ChainDiffs({1, 0, 0});
+  t[7].S(2).T(0).Dtis("----SS").ChainDiffs({2, 1, 0});
+  t[2].S(0).T(1).Dtis("-D----").ChainDiffs({3, 2, 1}).FrameDiffs({3});
+  t[5].S(1).T(1).Dtis("---D--").ChainDiffs({4, 3, 2}).FrameDiffs({3});
+  t[8].S(2).T(1).Dtis("-----D").ChainDiffs({5, 4, 3}).FrameDiffs({3});
+  t[0].S(0).T(0).Dtis("SS----").ChainDiffs({6, 5, 4}).FrameDiffs({6});
+  t[3].S(1).T(0).Dtis("--SS--").ChainDiffs({1, 6, 5}).FrameDiffs({6});
+  t[6].S(2).T(0).Dtis("----SS").ChainDiffs({2, 1, 6}).FrameDiffs({6});
+  return structure;
+}
+
+FrameDependencyStructure ScalabilityStructureS3T3::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 9;
+  structure.num_chains = 3;
+  structure.decode_target_protected_by_chain = {0, 0, 0, 1, 1, 1, 2, 2, 2};
+  auto& t = structure.templates;
+  t.resize(15);
+  // Templates are shown in the order frames following them appear in the
+  // stream, but in `structure.templates` array templates are sorted by
+  // (`spatial_id`, `temporal_id`) since that is a dependency descriptor
+  // requirement. Indexes are written in hex for nicer alignment.
+  t[0x1].S(0).T(0).Dtis("SSS------").ChainDiffs({0, 0, 0});
+  t[0x6].S(1).T(0).Dtis("---SSS---").ChainDiffs({1, 0, 0});
+  t[0xB].S(2).T(0).Dtis("------SSS").ChainDiffs({2, 1, 0});
+  t[0x3].S(0).T(2).Dtis("--D------").ChainDiffs({3, 2, 1}).FrameDiffs({3});
+  t[0x8].S(1).T(2).Dtis("-----D---").ChainDiffs({4, 3, 2}).FrameDiffs({3});
+  t[0xD].S(2).T(2).Dtis("--------D").ChainDiffs({5, 4, 3}).FrameDiffs({3});
+  t[0x2].S(0).T(1).Dtis("-DS------").ChainDiffs({6, 5, 4}).FrameDiffs({6});
+  t[0x7].S(1).T(1).Dtis("----DS---").ChainDiffs({7, 6, 5}).FrameDiffs({6});
+  t[0xC].S(2).T(1).Dtis("-------DS").ChainDiffs({8, 7, 6}).FrameDiffs({6});
+  t[0x4].S(0).T(2).Dtis("--D------").ChainDiffs({9, 8, 7}).FrameDiffs({3});
+  t[0x9].S(1).T(2).Dtis("-----D---").ChainDiffs({10, 9, 8}).FrameDiffs({3});
+  t[0xE].S(2).T(2).Dtis("--------D").ChainDiffs({11, 10, 9}).FrameDiffs({3});
+  t[0x0].S(0).T(0).Dtis("SSS------").ChainDiffs({12, 11, 10}).FrameDiffs({12});
+  t[0x5].S(1).T(0).Dtis("---SSS---").ChainDiffs({1, 12, 11}).FrameDiffs({12});
+  t[0xA].S(2).T(0).Dtis("------SSS").ChainDiffs({2, 1, 12}).FrameDiffs({12});
+  return structure;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_simulcast.h b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_simulcast.h
new file mode 100644
index 0000000000..99be9f0d58
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_simulcast.h
@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_SIMULCAST_H_
+#define MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_SIMULCAST_H_
+
+#include <vector>
+
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+
+namespace webrtc {
+
+// Scalability structure with multiple independent spatial layers each with the
+// same temporal layering.
+class ScalabilityStructureSimulcast : public ScalableVideoController {
+ public:
+  struct ScalingFactor {
+    int num = 1;
+    int den = 2;
+  };
+  ScalabilityStructureSimulcast(int num_spatial_layers,
+                                int num_temporal_layers,
+                                ScalingFactor resolution_factor);
+  ~ScalabilityStructureSimulcast() override;
+
+  StreamLayersConfig StreamConfig() const override;
+  std::vector<LayerFrameConfig> NextFrameConfig(bool restart) override;
+  GenericFrameInfo OnEncodeDone(const LayerFrameConfig& config) override;
+  void OnRatesUpdated(const VideoBitrateAllocation& bitrates) override;
+
+ private:
+  enum FramePattern {
+    kNone,
+    kDeltaT2A,
+    kDeltaT1,
+    kDeltaT2B,
+    kDeltaT0,
+  };
+  static constexpr int kMaxNumSpatialLayers = 3;
+  static constexpr int kMaxNumTemporalLayers = 3;
+
+  // Index of the buffer to store last frame for layer (`sid`, `tid`)
+  int BufferIndex(int sid, int tid) const {
+    return tid * num_spatial_layers_ + sid;
+  }
+  bool DecodeTargetIsActive(int sid, int tid) const {
+    return active_decode_targets_[sid * num_temporal_layers_ + tid];
+  }
+  void SetDecodeTargetIsActive(int sid, int tid, bool value) {
+    active_decode_targets_.set(sid * num_temporal_layers_ + tid, value);
+  }
+  FramePattern NextPattern() const;
+  bool TemporalLayerIsActive(int tid) const;
+
+  const int num_spatial_layers_;
+  const int num_temporal_layers_;
+  const ScalingFactor resolution_factor_;
+
+  FramePattern last_pattern_ = kNone;
+  std::bitset<kMaxNumSpatialLayers> can_reference_t0_frame_for_spatial_id_ = 0;
+  std::bitset<kMaxNumSpatialLayers> can_reference_t1_frame_for_spatial_id_ = 0;
+  std::bitset<32> active_decode_targets_;
+};
+
+// S1  0--0--0-
+//             ...
+// S0  0--0--0-
+class ScalabilityStructureS2T1 : public ScalabilityStructureSimulcast {
+ public:
+  explicit ScalabilityStructureS2T1(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureSimulcast(2, 1, resolution_factor) {}
+  ~ScalabilityStructureS2T1() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+class ScalabilityStructureS2T2 : public ScalabilityStructureSimulcast {
+ public:
+  explicit ScalabilityStructureS2T2(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureSimulcast(2, 2, resolution_factor) {}
+  ~ScalabilityStructureS2T2() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+// S1T2       3   7
+//            |  /
+// S1T1       / 5
+//           |_/
+// S1T0     1-------9...
+//
+// S0T2       2   6
+//            |  /
+// S0T1       / 4
+//           |_/
+// S0T0     0-------8...
+// Time->   0 1 2 3 4
+class ScalabilityStructureS2T3 : public ScalabilityStructureSimulcast {
+ public:
+  explicit ScalabilityStructureS2T3(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureSimulcast(2, 3, resolution_factor) {}
+  ~ScalabilityStructureS2T3() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+class ScalabilityStructureS3T1 : public ScalabilityStructureSimulcast {
+ public:
+  explicit ScalabilityStructureS3T1(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureSimulcast(3, 1, resolution_factor) {}
+  ~ScalabilityStructureS3T1() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+class ScalabilityStructureS3T2 : public ScalabilityStructureSimulcast {
+ public:
+  explicit ScalabilityStructureS3T2(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureSimulcast(3, 2, resolution_factor) {}
+  ~ScalabilityStructureS3T2() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+class ScalabilityStructureS3T3 : public ScalabilityStructureSimulcast {
+ public:
+  explicit ScalabilityStructureS3T3(ScalingFactor resolution_factor = {})
+      : ScalabilityStructureSimulcast(3, 3, resolution_factor) {}
+  ~ScalabilityStructureS3T3() override = default;
+
+  FrameDependencyStructure DependencyStructure() const override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_SIMULCAST_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_test_helpers.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_test_helpers.cc
new file mode 100644
index 0000000000..aeb4d88f1a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_test_helpers.cc
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/scalability_structure_test_helpers.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "modules/video_coding/chain_diff_calculator.h"
+#include "modules/video_coding/frame_dependencies_calculator.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+VideoBitrateAllocation EnableTemporalLayers(int s0, int s1, int s2) {
+  VideoBitrateAllocation bitrate;
+  for (int tid = 0; tid < s0; ++tid) {
+    bitrate.SetBitrate(0, tid, 1'000'000);
+  }
+  for (int tid = 0; tid < s1; ++tid) {
+    bitrate.SetBitrate(1, tid, 1'000'000);
+  }
+  for (int tid = 0; tid < s2; ++tid) {
+    bitrate.SetBitrate(2, tid, 1'000'000);
+  }
+  return bitrate;
+}
+
+void ScalabilityStructureWrapper::GenerateFrames(
+    int num_temporal_units,
+    std::vector<GenericFrameInfo>& frames) {
+  for (int i = 0; i < num_temporal_units; ++i) {
+    for (auto& layer_frame :
+         structure_controller_.NextFrameConfig(/*restart=*/false)) {
+      int64_t frame_id = ++frame_id_;
+      bool is_keyframe = layer_frame.IsKeyframe();
+
+      GenericFrameInfo frame_info =
+          structure_controller_.OnEncodeDone(layer_frame);
+      if (is_keyframe) {
+        chain_diff_calculator_.Reset(frame_info.part_of_chain);
+      }
+      frame_info.chain_diffs =
+          chain_diff_calculator_.From(frame_id, frame_info.part_of_chain);
+      for (int64_t base_frame_id : frame_deps_calculator_.FromBuffersUsage(
+               frame_id, frame_info.encoder_buffers)) {
+        frame_info.frame_diffs.push_back(frame_id - base_frame_id);
+      }
+
+      frames.push_back(std::move(frame_info));
+    }
+  }
+}
+
+bool ScalabilityStructureWrapper::FrameReferencesAreValid(
+    rtc::ArrayView<const GenericFrameInfo> frames) const {
+  bool valid = true;
+  // VP9 and AV1 supports up to 8 buffers. Expect no more buffers are not used.
+  std::bitset<8> buffer_contains_frame;
+  for (size_t i = 0; i < frames.size(); ++i) {
+    const GenericFrameInfo& frame = frames[i];
+    for (const CodecBufferUsage& buffer_usage : frame.encoder_buffers) {
+      if (buffer_usage.id < 0 || buffer_usage.id >= 8) {
+        ADD_FAILURE() << "Invalid buffer id " << buffer_usage.id
+                      << " for frame#" << i
+                      << ". Up to 8 buffers are supported.";
+        valid = false;
+        continue;
+      }
+      if (buffer_usage.referenced && !buffer_contains_frame[buffer_usage.id]) {
+        ADD_FAILURE() << "buffer " << buffer_usage.id << " for frame#" << i
+                      << " was reference before updated.";
+        valid = false;
+      }
+      if (buffer_usage.updated) {
+        buffer_contains_frame.set(buffer_usage.id);
+      }
+    }
+    for (int fdiff : frame.frame_diffs) {
+      if (fdiff <= 0 || static_cast<size_t>(fdiff) > i) {
+        ADD_FAILURE() << "Invalid frame diff " << fdiff << " for frame#" << i;
+        valid = false;
+      }
+    }
+  }
+  return valid;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_test_helpers.h b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_test_helpers.h
new file mode 100644
index 0000000000..d183be4766
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_test_helpers.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_TEST_HELPERS_H_
+#define MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_TEST_HELPERS_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/chain_diff_calculator.h"
+#include "modules/video_coding/frame_dependencies_calculator.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+
+namespace webrtc {
+
+// Creates bitrate allocation with non-zero bitrate for given number of temporal
+// layers for each spatial layer.
+VideoBitrateAllocation EnableTemporalLayers(int s0, int s1 = 0, int s2 = 0);
+
+class ScalabilityStructureWrapper {
+ public:
+  explicit ScalabilityStructureWrapper(ScalableVideoController& structure)
+      : structure_controller_(structure) {}
+
+  std::vector<GenericFrameInfo> GenerateFrames(int num_temporal_units) {
+    std::vector<GenericFrameInfo> frames;
+    GenerateFrames(num_temporal_units, frames);
+    return frames;
+  }
+  void GenerateFrames(int num_temporal_units,
+                      std::vector<GenericFrameInfo>& frames);
+
+  // Returns false and ADD_FAILUREs for frames with invalid references.
+  // In particular validates no frame frame reference to frame before frames[0].
+  // In error messages frames are indexed starting with 0.
+  bool FrameReferencesAreValid(
+      rtc::ArrayView<const GenericFrameInfo> frames) const;
+
+ private:
+  ScalableVideoController& structure_controller_;
+  FrameDependenciesCalculator frame_deps_calculator_;
+  ChainDiffCalculator chain_diff_calculator_;
+  int64_t frame_id_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_SCALABILITY_STRUCTURE_TEST_HELPERS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_unittest.cc b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_unittest.cc
new file mode 100644
index 0000000000..2d517c5825
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_unittest.cc
@@ -0,0 +1,395 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <ostream>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "modules/video_coding/svc/scalability_structure_test_helpers.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::AllOf;
+using ::testing::Contains;
+using ::testing::Each;
+using ::testing::ElementsAreArray;
+using ::testing::Field;
+using ::testing::Ge;
+using ::testing::IsEmpty;
+using ::testing::Le;
+using ::testing::Lt;
+using ::testing::Not;
+using ::testing::NotNull;
+using ::testing::SizeIs;
+using ::testing::TestWithParam;
+using ::testing::Values;
+
+std::string FrameDependencyTemplateToString(const FrameDependencyTemplate& t) {
+  rtc::StringBuilder sb;
+  sb << "S" << t.spatial_id << "T" << t.temporal_id;
+  sb << ": dtis = ";
+  for (const auto dtis : t.decode_target_indications) {
+    switch (dtis) {
+      case DecodeTargetIndication::kNotPresent:
+        sb << "-";
+        break;
+      case DecodeTargetIndication::kDiscardable:
+        sb << "D";
+        break;
+      case DecodeTargetIndication::kSwitch:
+        sb << "S";
+        break;
+      case DecodeTargetIndication::kRequired:
+        sb << "R";
+        break;
+      default:
+        sb << "?";
+        break;
+    }
+  }
+  sb << ", frame diffs = { ";
+  for (int d : t.frame_diffs) {
+    sb << d << ", ";
+  }
+  sb << "}, chain diffs = { ";
+  for (int d : t.chain_diffs) {
+    sb << d << ", ";
+  }
+  sb << "}";
+  return sb.Release();
+}
+
+struct SvcTestParam {
+  friend std::ostream& operator<<(std::ostream& os, const SvcTestParam& param) {
+    return os << param.name;
+  }
+
+  ScalabilityMode GetScalabilityMode() const {
+    absl::optional<ScalabilityMode> scalability_mode =
+        ScalabilityModeFromString(name);
+    RTC_CHECK(scalability_mode.has_value());
+    return *scalability_mode;
+  }
+
+  std::string name;
+  int num_temporal_units;
+};
+
+class ScalabilityStructureTest : public TestWithParam<SvcTestParam> {};
+
+TEST_P(ScalabilityStructureTest,
+       StaticConfigMatchesConfigReturnedByController) {
+  std::unique_ptr<ScalableVideoController> controller =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode());
+  absl::optional<ScalableVideoController::StreamLayersConfig> static_config =
+      ScalabilityStructureConfig(GetParam().GetScalabilityMode());
+  ASSERT_THAT(controller, NotNull());
+  ASSERT_NE(static_config, absl::nullopt);
+  ScalableVideoController::StreamLayersConfig config =
+      controller->StreamConfig();
+  EXPECT_EQ(config.num_spatial_layers, static_config->num_spatial_layers);
+  EXPECT_EQ(config.num_temporal_layers, static_config->num_temporal_layers);
+  EXPECT_THAT(
+      rtc::MakeArrayView(config.scaling_factor_num, config.num_spatial_layers),
+      ElementsAreArray(static_config->scaling_factor_num,
+                       static_config->num_spatial_layers));
+  EXPECT_THAT(
+      rtc::MakeArrayView(config.scaling_factor_den, config.num_spatial_layers),
+      ElementsAreArray(static_config->scaling_factor_den,
+                       static_config->num_spatial_layers));
+}
+
+TEST_P(ScalabilityStructureTest,
+       NumberOfDecodeTargetsAndChainsAreInRangeAndConsistent) {
+  FrameDependencyStructure structure =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode())
+          ->DependencyStructure();
+  EXPECT_GT(structure.num_decode_targets, 0);
+  EXPECT_LE(structure.num_decode_targets,
+            DependencyDescriptor::kMaxDecodeTargets);
+  EXPECT_GE(structure.num_chains, 0);
+  EXPECT_LE(structure.num_chains, structure.num_decode_targets);
+  if (structure.num_chains == 0) {
+    EXPECT_THAT(structure.decode_target_protected_by_chain, IsEmpty());
+  } else {
+    EXPECT_THAT(structure.decode_target_protected_by_chain,
+                AllOf(SizeIs(structure.num_decode_targets), Each(Ge(0)),
+                      Each(Lt(structure.num_chains))));
+  }
+  EXPECT_THAT(structure.templates,
+              SizeIs(Lt(size_t{DependencyDescriptor::kMaxTemplates})));
+}
+
+TEST_P(ScalabilityStructureTest, TemplatesAreSortedByLayerId) {
+  FrameDependencyStructure structure =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode())
+          ->DependencyStructure();
+  ASSERT_THAT(structure.templates, Not(IsEmpty()));
+  const auto& first_templates = structure.templates.front();
+  EXPECT_EQ(first_templates.spatial_id, 0);
+  EXPECT_EQ(first_templates.temporal_id, 0);
+  for (size_t i = 1; i < structure.templates.size(); ++i) {
+    const auto& prev_template = structure.templates[i - 1];
+    const auto& next_template = structure.templates[i];
+    if (next_template.spatial_id == prev_template.spatial_id &&
+        next_template.temporal_id == prev_template.temporal_id) {
+      // Same layer, next_layer_idc == 0
+    } else if (next_template.spatial_id == prev_template.spatial_id &&
+               next_template.temporal_id == prev_template.temporal_id + 1) {
+      // Next temporal layer, next_layer_idc == 1
+    } else if (next_template.spatial_id == prev_template.spatial_id + 1 &&
+               next_template.temporal_id == 0) {
+      // Next spatial layer, next_layer_idc == 2
+    } else {
+      // everything else is invalid.
+      ADD_FAILURE() << "Invalid templates order. Template #" << i
+                    << " with layer (" << next_template.spatial_id << ","
+                    << next_template.temporal_id
+                    << ") follows template with layer ("
+                    << prev_template.spatial_id << ","
+                    << prev_template.temporal_id << ").";
+    }
+  }
+}
+
+TEST_P(ScalabilityStructureTest, TemplatesMatchNumberOfDecodeTargetsAndChains) {
+  FrameDependencyStructure structure =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode())
+          ->DependencyStructure();
+  EXPECT_THAT(
+      structure.templates,
+      Each(AllOf(Field(&FrameDependencyTemplate::decode_target_indications,
+                       SizeIs(structure.num_decode_targets)),
+                 Field(&FrameDependencyTemplate::chain_diffs,
+                       SizeIs(structure.num_chains)))));
+}
+
+TEST_P(ScalabilityStructureTest, FrameInfoMatchesFrameDependencyStructure) {
+  std::unique_ptr<ScalableVideoController> svc_controller =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode());
+  FrameDependencyStructure structure = svc_controller->DependencyStructure();
+  std::vector<GenericFrameInfo> frame_infos =
+      ScalabilityStructureWrapper(*svc_controller)
+          .GenerateFrames(GetParam().num_temporal_units);
+  for (size_t frame_id = 0; frame_id < frame_infos.size(); ++frame_id) {
+    const auto& frame = frame_infos[frame_id];
+    EXPECT_GE(frame.spatial_id, 0) << " for frame " << frame_id;
+    EXPECT_GE(frame.temporal_id, 0) << " for frame " << frame_id;
+    EXPECT_THAT(frame.decode_target_indications,
+                SizeIs(structure.num_decode_targets))
+        << " for frame " << frame_id;
+    EXPECT_THAT(frame.part_of_chain, SizeIs(structure.num_chains))
+        << " for frame " << frame_id;
+  }
+}
+
+TEST_P(ScalabilityStructureTest, ThereIsAPerfectTemplateForEachFrame) {
+  std::unique_ptr<ScalableVideoController> svc_controller =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode());
+  FrameDependencyStructure structure = svc_controller->DependencyStructure();
+  std::vector<GenericFrameInfo> frame_infos =
+      ScalabilityStructureWrapper(*svc_controller)
+          .GenerateFrames(GetParam().num_temporal_units);
+  for (size_t frame_id = 0; frame_id < frame_infos.size(); ++frame_id) {
+    EXPECT_THAT(structure.templates, Contains(frame_infos[frame_id]))
+        << " for frame " << frame_id << ", Expected "
+        << FrameDependencyTemplateToString(frame_infos[frame_id]);
+  }
+}
+
+TEST_P(ScalabilityStructureTest, FrameDependsOnSameOrLowerLayer) {
+  std::unique_ptr<ScalableVideoController> svc_controller =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode());
+  std::vector<GenericFrameInfo> frame_infos =
+      ScalabilityStructureWrapper(*svc_controller)
+          .GenerateFrames(GetParam().num_temporal_units);
+  int64_t num_frames = frame_infos.size();
+
+  for (int64_t frame_id = 0; frame_id < num_frames; ++frame_id) {
+    const auto& frame = frame_infos[frame_id];
+    for (int frame_diff : frame.frame_diffs) {
+      int64_t base_frame_id = frame_id - frame_diff;
+      const auto& base_frame = frame_infos[base_frame_id];
+      EXPECT_GE(frame.spatial_id, base_frame.spatial_id)
+          << "Frame " << frame_id << " depends on frame " << base_frame_id;
+      EXPECT_GE(frame.temporal_id, base_frame.temporal_id)
+          << "Frame " << frame_id << " depends on frame " << base_frame_id;
+    }
+  }
+}
+
+TEST_P(ScalabilityStructureTest, NoFrameDependsOnDiscardableOrNotPresent) {
+  std::unique_ptr<ScalableVideoController> svc_controller =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode());
+  std::vector<GenericFrameInfo> frame_infos =
+      ScalabilityStructureWrapper(*svc_controller)
+          .GenerateFrames(GetParam().num_temporal_units);
+  int64_t num_frames = frame_infos.size();
+  FrameDependencyStructure structure = svc_controller->DependencyStructure();
+
+  for (int dt = 0; dt < structure.num_decode_targets; ++dt) {
+    for (int64_t frame_id = 0; frame_id < num_frames; ++frame_id) {
+      const auto& frame = frame_infos[frame_id];
+      if (frame.decode_target_indications[dt] ==
+          DecodeTargetIndication::kNotPresent) {
+        continue;
+      }
+      for (int frame_diff : frame.frame_diffs) {
+        int64_t base_frame_id = frame_id - frame_diff;
+        const auto& base_frame = frame_infos[base_frame_id];
+        EXPECT_NE(base_frame.decode_target_indications[dt],
+                  DecodeTargetIndication::kNotPresent)
+            << "Frame " << frame_id << " depends on frame " << base_frame_id
+            << " that is not part of decode target#" << dt;
+        EXPECT_NE(base_frame.decode_target_indications[dt],
+                  DecodeTargetIndication::kDiscardable)
+            << "Frame " << frame_id << " depends on frame " << base_frame_id
+            << " that is discardable for decode target#" << dt;
+      }
+    }
+  }
+}
+
+TEST_P(ScalabilityStructureTest, NoFrameDependsThroughSwitchIndication) {
+  std::unique_ptr<ScalableVideoController> svc_controller =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode());
+  FrameDependencyStructure structure = svc_controller->DependencyStructure();
+  std::vector<GenericFrameInfo> frame_infos =
+      ScalabilityStructureWrapper(*svc_controller)
+          .GenerateFrames(GetParam().num_temporal_units);
+  int64_t num_frames = frame_infos.size();
+  std::vector<std::set<int64_t>> full_deps(num_frames);
+
+  // For each frame calculate set of all frames it depends on, both directly and
+  // indirectly.
+  for (int64_t frame_id = 0; frame_id < num_frames; ++frame_id) {
+    std::set<int64_t> all_base_frames;
+    for (int frame_diff : frame_infos[frame_id].frame_diffs) {
+      int64_t base_frame_id = frame_id - frame_diff;
+      all_base_frames.insert(base_frame_id);
+      const auto& indirect = full_deps[base_frame_id];
+      all_base_frames.insert(indirect.begin(), indirect.end());
+    }
+    full_deps[frame_id] = std::move(all_base_frames);
+  }
+
+  // Now check the switch indication: frames after the switch indication mustn't
+  // depend on any addition frames before the switch indications.
+  for (int dt = 0; dt < structure.num_decode_targets; ++dt) {
+    for (int64_t switch_frame_id = 0; switch_frame_id < num_frames;
+         ++switch_frame_id) {
+      if (frame_infos[switch_frame_id].decode_target_indications[dt] !=
+          DecodeTargetIndication::kSwitch) {
+        continue;
+      }
+      for (int64_t later_frame_id = switch_frame_id + 1;
+           later_frame_id < num_frames; ++later_frame_id) {
+        if (frame_infos[later_frame_id].decode_target_indications[dt] ==
+            DecodeTargetIndication::kNotPresent) {
+          continue;
+        }
+        for (int frame_diff : frame_infos[later_frame_id].frame_diffs) {
+          int64_t early_frame_id = later_frame_id - frame_diff;
+          if (early_frame_id < switch_frame_id) {
+            EXPECT_THAT(full_deps[switch_frame_id], Contains(early_frame_id))
+                << "For decode target #" << dt << " frame " << later_frame_id
+                << " depends on the frame " << early_frame_id
+                << " that switch indication frame " << switch_frame_id
+                << " doesn't directly on indirectly depend on.";
+          }
+        }
+      }
+    }
+  }
+}
+
+TEST_P(ScalabilityStructureTest, ProduceNoFrameForDisabledLayers) {
+  std::unique_ptr<ScalableVideoController> svc_controller =
+      CreateScalabilityStructure(GetParam().GetScalabilityMode());
+  ScalableVideoController::StreamLayersConfig structure =
+      svc_controller->StreamConfig();
+
+  VideoBitrateAllocation all_bitrates;
+  for (int sid = 0; sid < structure.num_spatial_layers; ++sid) {
+    for (int tid = 0; tid < structure.num_temporal_layers; ++tid) {
+      all_bitrates.SetBitrate(sid, tid, 100'000);
+    }
+  }
+
+  svc_controller->OnRatesUpdated(all_bitrates);
+  ScalabilityStructureWrapper wrapper(*svc_controller);
+  std::vector<GenericFrameInfo> frames =
+      wrapper.GenerateFrames(GetParam().num_temporal_units);
+
+  for (int sid = 0; sid < structure.num_spatial_layers; ++sid) {
+    for (int tid = 0; tid < structure.num_temporal_layers; ++tid) {
+      // When all layers were enabled, expect there was a frame for each layer.
+      EXPECT_THAT(frames,
+                  Contains(AllOf(Field(&GenericFrameInfo::spatial_id, sid),
+                                 Field(&GenericFrameInfo::temporal_id, tid))))
+          << "For layer (" << sid << "," << tid << ")";
+      // Restore bitrates for all layers before disabling single layer.
+      VideoBitrateAllocation bitrates = all_bitrates;
+      bitrates.SetBitrate(sid, tid, 0);
+      svc_controller->OnRatesUpdated(bitrates);
+      // With layer (sid, tid) disabled, expect no frames are produced for it.
+      EXPECT_THAT(
+          wrapper.GenerateFrames(GetParam().num_temporal_units),
+          Not(Contains(AllOf(Field(&GenericFrameInfo::spatial_id, sid),
+                             Field(&GenericFrameInfo::temporal_id, tid)))))
+          << "For layer (" << sid << "," << tid << ")";
+    }
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    Svc,
+    ScalabilityStructureTest,
+    Values(SvcTestParam{"L1T1", /*num_temporal_units=*/3},
+           SvcTestParam{"L1T2", /*num_temporal_units=*/4},
+           SvcTestParam{"L1T3", /*num_temporal_units=*/8},
+           SvcTestParam{"L2T1", /*num_temporal_units=*/3},
+           SvcTestParam{"L2T1_KEY", /*num_temporal_units=*/3},
+           SvcTestParam{"L3T1", /*num_temporal_units=*/3},
+           SvcTestParam{"L3T1_KEY", /*num_temporal_units=*/3},
+           SvcTestParam{"L3T3", /*num_temporal_units=*/8},
+           SvcTestParam{"S2T1", /*num_temporal_units=*/3},
+           SvcTestParam{"S2T2", /*num_temporal_units=*/4},
+           SvcTestParam{"S2T3", /*num_temporal_units=*/8},
+           SvcTestParam{"S3T1", /*num_temporal_units=*/3},
+           SvcTestParam{"S3T2", /*num_temporal_units=*/4},
+           SvcTestParam{"S3T3", /*num_temporal_units=*/8},
+           SvcTestParam{"L2T2", /*num_temporal_units=*/4},
+           SvcTestParam{"L2T2_KEY", /*num_temporal_units=*/4},
+           SvcTestParam{"L2T2_KEY_SHIFT", /*num_temporal_units=*/4},
+           SvcTestParam{"L2T3", /*num_temporal_units=*/8},
+           SvcTestParam{"L2T3_KEY", /*num_temporal_units=*/8},
+           SvcTestParam{"L3T2", /*num_temporal_units=*/4},
+           SvcTestParam{"L3T2_KEY", /*num_temporal_units=*/4},
+           SvcTestParam{"L3T3_KEY", /*num_temporal_units=*/8}),
+    [](const testing::TestParamInfo<SvcTestParam>& info) {
+      return info.param.name;
+    });
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalability_structures_gn/moz.build b/third_party/libwebrtc/modules/video_coding/svc/scalability_structures_gn/moz.build
new file mode 100644
index 0000000000..a3e45b1cd7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalability_structures_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc.cc",
+    "/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_simulcast.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/svc/create_scalability_structure.cc",
+    "/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_full_svc.cc",
+    "/third_party/libwebrtc/modules/video_coding/svc/scalability_structure_l2t2_key_shift.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("scalability_structures_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller.h b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller.h
new file mode 100644
index 0000000000..c7362657ec
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller.h
@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_SVC_SCALABLE_VIDEO_CONTROLLER_H_
+#define MODULES_VIDEO_CODING_SVC_SCALABLE_VIDEO_CONTROLLER_H_
+
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+
+namespace webrtc {
+
+// Controls how video should be encoded to be scalable. Outputs results as
+// buffer usage configuration for encoder and enough details to communicate the
+// scalability structure via dependency descriptor rtp header extension.
+class ScalableVideoController {
+ public:
+  struct StreamLayersConfig {
+    int num_spatial_layers = 1;
+    int num_temporal_layers = 1;
+    // Indicates if frames can reference frames of a different resolution.
+    bool uses_reference_scaling = true;
+    // Spatial layers scaling. Frames with spatial_id = i expected to be encoded
+    // with original_resolution * scaling_factor_num[i] / scaling_factor_den[i].
+    int scaling_factor_num[DependencyDescriptor::kMaxSpatialIds] = {1, 1, 1, 1};
+    int scaling_factor_den[DependencyDescriptor::kMaxSpatialIds] = {1, 1, 1, 1};
+  };
+  class LayerFrameConfig {
+   public:
+    // Builders/setters.
+    LayerFrameConfig& Id(int value);
+    LayerFrameConfig& Keyframe();
+    LayerFrameConfig& S(int value);
+    LayerFrameConfig& T(int value);
+    LayerFrameConfig& Reference(int buffer_id);
+    LayerFrameConfig& Update(int buffer_id);
+    LayerFrameConfig& ReferenceAndUpdate(int buffer_id);
+
+    // Getters.
+    int Id() const { return id_; }
+    bool IsKeyframe() const { return is_keyframe_; }
+    int SpatialId() const { return spatial_id_; }
+    int TemporalId() const { return temporal_id_; }
+    const absl::InlinedVector<CodecBufferUsage, kMaxEncoderBuffers>& Buffers()
+        const {
+      return buffers_;
+    }
+
+   private:
+    // Id to match configuration returned by NextFrameConfig with
+    // (possibly modified) configuration passed back via OnEncoderDone.
+    // The meaning of the id is an implementation detail of
+    // the ScalableVideoController.
+    int id_ = 0;
+
+    // Indication frame should be encoded as a key frame. In particular when
+    // `is_keyframe=true` property `CodecBufferUsage::referenced` should be
+    // ignored and treated as false.
+    bool is_keyframe_ = false;
+
+    int spatial_id_ = 0;
+    int temporal_id_ = 0;
+    // Describes how encoder which buffers encoder allowed to reference and
+    // which buffers encoder should update.
+    absl::InlinedVector<CodecBufferUsage, kMaxEncoderBuffers> buffers_;
+  };
+
+  virtual ~ScalableVideoController() = default;
+
+  // Returns video structure description for encoder to configure itself.
+  virtual StreamLayersConfig StreamConfig() const = 0;
+
+  // Returns video structure description in format compatible with
+  // dependency descriptor rtp header extension.
+  virtual FrameDependencyStructure DependencyStructure() const = 0;
+
+  // Notifies Controller with updated bitrates per layer. In particular notifies
+  // when certain layers should be disabled.
+  // Controller shouldn't produce LayerFrameConfig for disabled layers.
+  virtual void OnRatesUpdated(const VideoBitrateAllocation& bitrates) = 0;
+
+  // When `restart` is true, first `LayerFrameConfig` should have `is_keyframe`
+  // set to true.
+  // Returned vector shouldn't be empty.
+  virtual std::vector<LayerFrameConfig> NextFrameConfig(bool restart) = 0;
+
+  // Returns configuration to pass to EncoderCallback.
+  virtual GenericFrameInfo OnEncodeDone(const LayerFrameConfig& config) = 0;
+};
+
+// Below are implementation details.
+inline ScalableVideoController::LayerFrameConfig&
+ScalableVideoController::LayerFrameConfig::Id(int value) {
+  id_ = value;
+  return *this;
+}
+inline ScalableVideoController::LayerFrameConfig&
+ScalableVideoController::LayerFrameConfig::Keyframe() {
+  is_keyframe_ = true;
+  return *this;
+}
+inline ScalableVideoController::LayerFrameConfig&
+ScalableVideoController::LayerFrameConfig::S(int value) {
+  spatial_id_ = value;
+  return *this;
+}
+inline ScalableVideoController::LayerFrameConfig&
+ScalableVideoController::LayerFrameConfig::T(int value) {
+  temporal_id_ = value;
+  return *this;
+}
+inline ScalableVideoController::LayerFrameConfig&
+ScalableVideoController::LayerFrameConfig::Reference(int buffer_id) {
+  buffers_.emplace_back(buffer_id, /*referenced=*/true, /*updated=*/false);
+  return *this;
+}
+inline ScalableVideoController::LayerFrameConfig&
+ScalableVideoController::LayerFrameConfig::Update(int buffer_id) {
+  buffers_.emplace_back(buffer_id, /*referenced=*/false, /*updated=*/true);
+  return *this;
+}
+inline ScalableVideoController::LayerFrameConfig&
+ScalableVideoController::LayerFrameConfig::ReferenceAndUpdate(int buffer_id) {
+  buffers_.emplace_back(buffer_id, /*referenced=*/true, /*updated=*/true);
+  return *this;
+}
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_SCALABLE_VIDEO_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_gn/moz.build b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_gn/moz.build
new file mode 100644
index 0000000000..42b8307362
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_no_layering.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("scalable_video_controller_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_no_layering.cc b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_no_layering.cc
new file mode 100644
index 0000000000..a9d530dd9d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_no_layering.cc
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/svc/scalable_video_controller_no_layering.h"
+
+#include <utility>
+#include <vector>
+
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+ScalableVideoControllerNoLayering::~ScalableVideoControllerNoLayering() =
+    default;
+
+ScalableVideoController::StreamLayersConfig
+ScalableVideoControllerNoLayering::StreamConfig() const {
+  StreamLayersConfig result;
+  result.num_spatial_layers = 1;
+  result.num_temporal_layers = 1;
+  result.uses_reference_scaling = false;
+  return result;
+}
+
+FrameDependencyStructure
+ScalableVideoControllerNoLayering::DependencyStructure() const {
+  FrameDependencyStructure structure;
+  structure.num_decode_targets = 1;
+  structure.num_chains = 1;
+  structure.decode_target_protected_by_chain = {0};
+
+  FrameDependencyTemplate key_frame;
+  key_frame.decode_target_indications = {DecodeTargetIndication::kSwitch};
+  key_frame.chain_diffs = {0};
+  structure.templates.push_back(key_frame);
+
+  FrameDependencyTemplate delta_frame;
+  delta_frame.decode_target_indications = {DecodeTargetIndication::kSwitch};
+  delta_frame.chain_diffs = {1};
+  delta_frame.frame_diffs = {1};
+  structure.templates.push_back(delta_frame);
+
+  return structure;
+}
+
+std::vector<ScalableVideoController::LayerFrameConfig>
+ScalableVideoControllerNoLayering::NextFrameConfig(bool restart) {
+  if (!enabled_) {
+    return {};
+  }
+  std::vector<LayerFrameConfig> result(1);
+  if (restart || start_) {
+    result[0].Id(0).Keyframe().Update(0);
+  } else {
+    result[0].Id(0).ReferenceAndUpdate(0);
+  }
+  start_ = false;
+  return result;
+}
+
+GenericFrameInfo ScalableVideoControllerNoLayering::OnEncodeDone(
+    const LayerFrameConfig& config) {
+  RTC_DCHECK_EQ(config.Id(), 0);
+  GenericFrameInfo frame_info;
+  frame_info.encoder_buffers = config.Buffers();
+  if (config.IsKeyframe()) {
+    for (auto& buffer : frame_info.encoder_buffers) {
+      buffer.referenced = false;
+    }
+  }
+  frame_info.decode_target_indications = {DecodeTargetIndication::kSwitch};
+  frame_info.part_of_chain = {true};
+  return frame_info;
+}
+
+void ScalableVideoControllerNoLayering::OnRatesUpdated(
+    const VideoBitrateAllocation& bitrates) {
+  enabled_ = bitrates.GetBitrate(0, 0) > 0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_no_layering.h b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_no_layering.h
new file mode 100644
index 0000000000..6d66b61c8b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/scalable_video_controller_no_layering.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_SVC_SCALABLE_VIDEO_CONTROLLER_NO_LAYERING_H_
+#define MODULES_VIDEO_CODING_SVC_SCALABLE_VIDEO_CONTROLLER_NO_LAYERING_H_
+
+#include <vector>
+
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+
+namespace webrtc {
+
+class ScalableVideoControllerNoLayering : public ScalableVideoController {
+ public:
+  ~ScalableVideoControllerNoLayering() override;
+
+  StreamLayersConfig StreamConfig() const override;
+  FrameDependencyStructure DependencyStructure() const override;
+
+  std::vector<LayerFrameConfig> NextFrameConfig(bool restart) override;
+  GenericFrameInfo OnEncodeDone(const LayerFrameConfig& config) override;
+  void OnRatesUpdated(const VideoBitrateAllocation& bitrates) override;
+
+ private:
+  bool start_ = true;
+  bool enabled_ = true;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_SCALABLE_VIDEO_CONTROLLER_NO_LAYERING_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator.cc b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator.cc
new file mode 100644
index 0000000000..b6ae0d7430
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator.cc
@@ -0,0 +1,452 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/svc/svc_rate_allocator.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstddef>
+#include <numeric>
+#include <vector>
+
+#include "absl/container/inlined_vector.h"
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr float kSpatialLayeringRateScalingFactor = 0.55f;
+constexpr float kTemporalLayeringRateScalingFactor = 0.55f;
+
+struct ActiveSpatialLayers {
+  size_t first = 0;
+  size_t num = 0;
+};
+
+ActiveSpatialLayers GetActiveSpatialLayers(const VideoCodec& codec,
+                                           size_t num_spatial_layers) {
+  ActiveSpatialLayers active;
+  for (active.first = 0; active.first < num_spatial_layers; ++active.first) {
+    if (codec.spatialLayers[active.first].active) {
+      break;
+    }
+  }
+
+  size_t last_active_layer = active.first;
+  for (; last_active_layer < num_spatial_layers; ++last_active_layer) {
+    if (!codec.spatialLayers[last_active_layer].active) {
+      break;
+    }
+  }
+  active.num = last_active_layer - active.first;
+
+  return active;
+}
+
+std::vector<DataRate> AdjustAndVerify(
+    const VideoCodec& codec,
+    size_t first_active_layer,
+    const std::vector<DataRate>& spatial_layer_rates) {
+  std::vector<DataRate> adjusted_spatial_layer_rates;
+  // Keep track of rate that couldn't be applied to the previous layer due to
+  // max bitrate constraint, try to pass it forward to the next one.
+  DataRate excess_rate = DataRate::Zero();
+  for (size_t sl_idx = 0; sl_idx < spatial_layer_rates.size(); ++sl_idx) {
+    DataRate min_rate = DataRate::KilobitsPerSec(
+        codec.spatialLayers[first_active_layer + sl_idx].minBitrate);
+    DataRate max_rate = DataRate::KilobitsPerSec(
+        codec.spatialLayers[first_active_layer + sl_idx].maxBitrate);
+
+    DataRate layer_rate = spatial_layer_rates[sl_idx] + excess_rate;
+    if (layer_rate < min_rate) {
+      // Not enough rate to reach min bitrate for desired number of layers,
+      // abort allocation.
+      if (spatial_layer_rates.size() == 1) {
+        return spatial_layer_rates;
+      }
+      return adjusted_spatial_layer_rates;
+    }
+
+    if (layer_rate <= max_rate) {
+      excess_rate = DataRate::Zero();
+      adjusted_spatial_layer_rates.push_back(layer_rate);
+    } else {
+      excess_rate = layer_rate - max_rate;
+      adjusted_spatial_layer_rates.push_back(max_rate);
+    }
+  }
+
+  return adjusted_spatial_layer_rates;
+}
+
+static std::vector<DataRate> SplitBitrate(size_t num_layers,
+                                          DataRate total_bitrate,
+                                          float rate_scaling_factor) {
+  std::vector<DataRate> bitrates;
+
+  double denominator = 0.0;
+  for (size_t layer_idx = 0; layer_idx < num_layers; ++layer_idx) {
+    denominator += std::pow(rate_scaling_factor, layer_idx);
+  }
+
+  double numerator = std::pow(rate_scaling_factor, num_layers - 1);
+  for (size_t layer_idx = 0; layer_idx < num_layers; ++layer_idx) {
+    bitrates.push_back(numerator * total_bitrate / denominator);
+    numerator /= rate_scaling_factor;
+  }
+
+  const DataRate sum =
+      std::accumulate(bitrates.begin(), bitrates.end(), DataRate::Zero());
+
+  // Keep the sum of split bitrates equal to the total bitrate by adding or
+  // subtracting bits, which were lost due to rounding, to the latest layer.
+  if (total_bitrate > sum) {
+    bitrates.back() += total_bitrate - sum;
+  } else if (total_bitrate < sum) {
+    bitrates.back() -= sum - total_bitrate;
+  }
+
+  return bitrates;
+}
+
+// Returns the minimum bitrate needed for `num_active_layers` spatial layers to
+// become active using the configuration specified by `codec`.
+DataRate FindLayerTogglingThreshold(const VideoCodec& codec,
+                                    size_t first_active_layer,
+                                    size_t num_active_layers) {
+  if (num_active_layers == 1) {
+    return DataRate::KilobitsPerSec(codec.spatialLayers[0].minBitrate);
+  }
+
+  if (codec.mode == VideoCodecMode::kRealtimeVideo) {
+    DataRate lower_bound = DataRate::Zero();
+    DataRate upper_bound = DataRate::Zero();
+    if (num_active_layers > 1) {
+      for (size_t i = 0; i < num_active_layers - 1; ++i) {
+        lower_bound += DataRate::KilobitsPerSec(
+            codec.spatialLayers[first_active_layer + i].minBitrate);
+        upper_bound += DataRate::KilobitsPerSec(
+            codec.spatialLayers[first_active_layer + i].maxBitrate);
+      }
+    }
+    upper_bound += DataRate::KilobitsPerSec(
+        codec.spatialLayers[first_active_layer + num_active_layers - 1]
+            .minBitrate);
+
+    // Do a binary search until upper and lower bound is the highest bitrate for
+    // `num_active_layers` - 1 layers and lowest bitrate for `num_active_layers`
+    // layers respectively.
+    while (upper_bound - lower_bound > DataRate::BitsPerSec(1)) {
+      DataRate try_rate = (lower_bound + upper_bound) / 2;
+      if (AdjustAndVerify(codec, first_active_layer,
+                          SplitBitrate(num_active_layers, try_rate,
+                                       kSpatialLayeringRateScalingFactor))
+              .size() == num_active_layers) {
+        upper_bound = try_rate;
+      } else {
+        lower_bound = try_rate;
+      }
+    }
+    return upper_bound;
+  } else {
+    DataRate toggling_rate = DataRate::Zero();
+    for (size_t i = 0; i < num_active_layers - 1; ++i) {
+      toggling_rate += DataRate::KilobitsPerSec(
+          codec.spatialLayers[first_active_layer + i].targetBitrate);
+    }
+    toggling_rate += DataRate::KilobitsPerSec(
+        codec.spatialLayers[first_active_layer + num_active_layers - 1]
+            .minBitrate);
+    return toggling_rate;
+  }
+}
+
+}  // namespace
+
+SvcRateAllocator::NumLayers SvcRateAllocator::GetNumLayers(
+    const VideoCodec& codec) {
+  NumLayers layers;
+  if (absl::optional<ScalabilityMode> scalability_mode =
+          codec.GetScalabilityMode();
+      scalability_mode.has_value()) {
+    if (auto structure = CreateScalabilityStructure(*scalability_mode)) {
+      ScalableVideoController::StreamLayersConfig config =
+          structure->StreamConfig();
+      layers.spatial = config.num_spatial_layers;
+      layers.temporal = config.num_temporal_layers;
+      return layers;
+    }
+  }
+  if (codec.codecType == kVideoCodecVP9) {
+    layers.spatial = codec.VP9().numberOfSpatialLayers;
+    layers.temporal = codec.VP9().numberOfTemporalLayers;
+    return layers;
+  }
+  layers.spatial = 1;
+  layers.temporal = 1;
+  return layers;
+}
+
+SvcRateAllocator::SvcRateAllocator(const VideoCodec& codec)
+    : codec_(codec),
+      num_layers_(GetNumLayers(codec)),
+      experiment_settings_(StableTargetRateExperiment::ParseFromFieldTrials()),
+      cumulative_layer_start_bitrates_(GetLayerStartBitrates(codec)),
+      last_active_layer_count_(0) {
+  RTC_DCHECK_GT(num_layers_.spatial, 0);
+  RTC_DCHECK_LE(num_layers_.spatial, kMaxSpatialLayers);
+  RTC_DCHECK_GT(num_layers_.temporal, 0);
+  RTC_DCHECK_LE(num_layers_.temporal, 3);
+  for (size_t layer_idx = 0; layer_idx < num_layers_.spatial; ++layer_idx) {
+    // Verify min <= target <= max.
+    if (codec.spatialLayers[layer_idx].active) {
+      RTC_DCHECK_GT(codec.spatialLayers[layer_idx].maxBitrate, 0);
+      RTC_DCHECK_GE(codec.spatialLayers[layer_idx].maxBitrate,
+                    codec.spatialLayers[layer_idx].minBitrate);
+      RTC_DCHECK_GE(codec.spatialLayers[layer_idx].targetBitrate,
+                    codec.spatialLayers[layer_idx].minBitrate);
+      RTC_DCHECK_GE(codec.spatialLayers[layer_idx].maxBitrate,
+                    codec.spatialLayers[layer_idx].targetBitrate);
+    }
+  }
+}
+
+VideoBitrateAllocation SvcRateAllocator::Allocate(
+    VideoBitrateAllocationParameters parameters) {
+  DataRate total_bitrate = parameters.total_bitrate;
+  if (codec_.maxBitrate != 0) {
+    total_bitrate =
+        std::min(total_bitrate, DataRate::KilobitsPerSec(codec_.maxBitrate));
+  }
+
+  if (codec_.spatialLayers[0].targetBitrate == 0) {
+    // Delegate rate distribution to encoder wrapper if bitrate thresholds
+    // are not set.
+    VideoBitrateAllocation bitrate_allocation;
+    bitrate_allocation.SetBitrate(0, 0, total_bitrate.bps());
+    return bitrate_allocation;
+  }
+
+  const ActiveSpatialLayers active_layers =
+      GetActiveSpatialLayers(codec_, num_layers_.spatial);
+  size_t num_spatial_layers = active_layers.num;
+
+  if (num_spatial_layers == 0) {
+    return VideoBitrateAllocation();  // All layers are deactivated.
+  }
+
+  // Figure out how many spatial layers should be active.
+  if (experiment_settings_.IsEnabled() &&
+      parameters.stable_bitrate > DataRate::Zero()) {
+    double hysteresis_factor;
+    if (codec_.mode == VideoCodecMode::kScreensharing) {
+      hysteresis_factor = experiment_settings_.GetScreenshareHysteresisFactor();
+    } else {
+      hysteresis_factor = experiment_settings_.GetVideoHysteresisFactor();
+    }
+
+    DataRate stable_rate =
+        std::min(parameters.total_bitrate, parameters.stable_bitrate);
+    // First check if bitrate has grown large enough to enable new layers.
+    size_t num_enabled_with_hysteresis =
+        FindNumEnabledLayers(stable_rate / hysteresis_factor);
+    if (num_enabled_with_hysteresis >= last_active_layer_count_) {
+      num_spatial_layers = num_enabled_with_hysteresis;
+    } else {
+      // We could not enable new layers, check if any should be disabled.
+      num_spatial_layers =
+          std::min(last_active_layer_count_, FindNumEnabledLayers(stable_rate));
+    }
+  } else {
+    num_spatial_layers = FindNumEnabledLayers(parameters.total_bitrate);
+  }
+  last_active_layer_count_ = num_spatial_layers;
+
+  VideoBitrateAllocation allocation;
+  if (codec_.mode == VideoCodecMode::kRealtimeVideo) {
+    allocation = GetAllocationNormalVideo(total_bitrate, active_layers.first,
+                                          num_spatial_layers);
+  } else {
+    allocation = GetAllocationScreenSharing(total_bitrate, active_layers.first,
+                                            num_spatial_layers);
+  }
+  allocation.set_bw_limited(num_spatial_layers < active_layers.num);
+  return allocation;
+}
+
+VideoBitrateAllocation SvcRateAllocator::GetAllocationNormalVideo(
+    DataRate total_bitrate,
+    size_t first_active_layer,
+    size_t num_spatial_layers) const {
+  std::vector<DataRate> spatial_layer_rates;
+  if (num_spatial_layers == 0) {
+    // Not enough rate for even the base layer. Force allocation at the total
+    // bitrate anyway.
+    num_spatial_layers = 1;
+    spatial_layer_rates.push_back(total_bitrate);
+  } else {
+    spatial_layer_rates =
+        AdjustAndVerify(codec_, first_active_layer,
+                        SplitBitrate(num_spatial_layers, total_bitrate,
+                                     kSpatialLayeringRateScalingFactor));
+    RTC_DCHECK_EQ(spatial_layer_rates.size(), num_spatial_layers);
+  }
+
+  VideoBitrateAllocation bitrate_allocation;
+
+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
+    std::vector<DataRate> temporal_layer_rates =
+        SplitBitrate(num_layers_.temporal, spatial_layer_rates[sl_idx],
+                     kTemporalLayeringRateScalingFactor);
+
+    // Distribute rate across temporal layers. Allocate more bits to lower
+    // layers since they are used for prediction of higher layers and their
+    // references are far apart.
+    if (num_layers_.temporal == 1) {
+      bitrate_allocation.SetBitrate(sl_idx + first_active_layer, 0,
+                                    temporal_layer_rates[0].bps());
+    } else if (num_layers_.temporal == 2) {
+      bitrate_allocation.SetBitrate(sl_idx + first_active_layer, 0,
+                                    temporal_layer_rates[1].bps());
+      bitrate_allocation.SetBitrate(sl_idx + first_active_layer, 1,
+                                    temporal_layer_rates[0].bps());
+    } else {
+      RTC_CHECK_EQ(num_layers_.temporal, 3);
+      // In case of three temporal layers the high layer has two frames and the
+      // middle layer has one frame within GOP (in between two consecutive low
+      // layer frames). Thus high layer requires more bits (comparing pure
+      // bitrate of layer, excluding bitrate of base layers) to keep quality on
+      // par with lower layers.
+      bitrate_allocation.SetBitrate(sl_idx + first_active_layer, 0,
+                                    temporal_layer_rates[2].bps());
+      bitrate_allocation.SetBitrate(sl_idx + first_active_layer, 1,
+                                    temporal_layer_rates[0].bps());
+      bitrate_allocation.SetBitrate(sl_idx + first_active_layer, 2,
+                                    temporal_layer_rates[1].bps());
+    }
+  }
+
+  return bitrate_allocation;
+}
+
+// Bit-rate is allocated in such a way, that the highest enabled layer will have
+// between min and max bitrate, and all others will have exactly target
+// bit-rate allocated.
+VideoBitrateAllocation SvcRateAllocator::GetAllocationScreenSharing(
+    DataRate total_bitrate,
+    size_t first_active_layer,
+    size_t num_spatial_layers) const {
+  VideoBitrateAllocation bitrate_allocation;
+
+  if (num_spatial_layers == 0 ||
+      total_bitrate <
+          DataRate::KilobitsPerSec(
+              codec_.spatialLayers[first_active_layer].minBitrate)) {
+    // Always enable at least one layer.
+    bitrate_allocation.SetBitrate(first_active_layer, 0, total_bitrate.bps());
+    return bitrate_allocation;
+  }
+
+  DataRate allocated_rate = DataRate::Zero();
+  DataRate top_layer_rate = DataRate::Zero();
+  size_t sl_idx;
+  for (sl_idx = first_active_layer;
+       sl_idx < first_active_layer + num_spatial_layers; ++sl_idx) {
+    const DataRate min_rate =
+        DataRate::KilobitsPerSec(codec_.spatialLayers[sl_idx].minBitrate);
+    const DataRate target_rate =
+        DataRate::KilobitsPerSec(codec_.spatialLayers[sl_idx].targetBitrate);
+
+    if (allocated_rate + min_rate > total_bitrate) {
+      // Use stable rate to determine if layer should be enabled.
+      break;
+    }
+
+    top_layer_rate = std::min(target_rate, total_bitrate - allocated_rate);
+    bitrate_allocation.SetBitrate(sl_idx, 0, top_layer_rate.bps());
+    allocated_rate += top_layer_rate;
+  }
+
+  if (sl_idx > 0 && total_bitrate - allocated_rate > DataRate::Zero()) {
+    // Add leftover to the last allocated layer.
+    top_layer_rate = std::min(
+        top_layer_rate + (total_bitrate - allocated_rate),
+        DataRate::KilobitsPerSec(codec_.spatialLayers[sl_idx - 1].maxBitrate));
+    bitrate_allocation.SetBitrate(sl_idx - 1, 0, top_layer_rate.bps());
+  }
+
+  return bitrate_allocation;
+}
+
+size_t SvcRateAllocator::FindNumEnabledLayers(DataRate target_rate) const {
+  if (cumulative_layer_start_bitrates_.empty()) {
+    return 0;
+  }
+
+  size_t num_enabled_layers = 0;
+  for (DataRate start_rate : cumulative_layer_start_bitrates_) {
+    // First layer is always enabled.
+    if (num_enabled_layers == 0 || start_rate <= target_rate) {
+      ++num_enabled_layers;
+    } else {
+      break;
+    }
+  }
+
+  return num_enabled_layers;
+}
+
+DataRate SvcRateAllocator::GetMaxBitrate(const VideoCodec& codec) {
+  const NumLayers num_layers = GetNumLayers(codec);
+  const ActiveSpatialLayers active_layers =
+      GetActiveSpatialLayers(codec, num_layers.spatial);
+
+  DataRate max_bitrate = DataRate::Zero();
+  for (size_t sl_idx = 0; sl_idx < active_layers.num; ++sl_idx) {
+    max_bitrate += DataRate::KilobitsPerSec(
+        codec.spatialLayers[active_layers.first + sl_idx].maxBitrate);
+  }
+
+  if (codec.maxBitrate != 0) {
+    max_bitrate =
+        std::min(max_bitrate, DataRate::KilobitsPerSec(codec.maxBitrate));
+  }
+
+  return max_bitrate;
+}
+
+DataRate SvcRateAllocator::GetPaddingBitrate(const VideoCodec& codec) {
+  auto start_bitrate = GetLayerStartBitrates(codec);
+  if (start_bitrate.empty()) {
+    return DataRate::Zero();  // All layers are deactivated.
+  }
+
+  return start_bitrate.back();
+}
+
+absl::InlinedVector<DataRate, kMaxSpatialLayers>
+SvcRateAllocator::GetLayerStartBitrates(const VideoCodec& codec) {
+  absl::InlinedVector<DataRate, kMaxSpatialLayers> start_bitrates;
+  const NumLayers num_layers = GetNumLayers(codec);
+  const ActiveSpatialLayers active_layers =
+      GetActiveSpatialLayers(codec, num_layers.spatial);
+  DataRate last_rate = DataRate::Zero();
+  for (size_t i = 1; i <= active_layers.num; ++i) {
+    DataRate layer_toggling_rate =
+        FindLayerTogglingThreshold(codec, active_layers.first, i);
+    start_bitrates.push_back(layer_toggling_rate);
+    RTC_DCHECK_LE(last_rate, layer_toggling_rate);
+    last_rate = layer_toggling_rate;
+  }
+  return start_bitrates;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator.h b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator.h
new file mode 100644
index 0000000000..bd75fca284
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_SVC_SVC_RATE_ALLOCATOR_H_
+#define MODULES_VIDEO_CODING_SVC_SVC_RATE_ALLOCATOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/container/inlined_vector.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_bitrate_allocator.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video_codecs/video_codec.h"
+#include "rtc_base/experiments/stable_target_rate_experiment.h"
+
+namespace webrtc {
+
+class SvcRateAllocator : public VideoBitrateAllocator {
+ public:
+  explicit SvcRateAllocator(const VideoCodec& codec);
+
+  VideoBitrateAllocation Allocate(
+      VideoBitrateAllocationParameters parameters) override;
+
+  static DataRate GetMaxBitrate(const VideoCodec& codec);
+  static DataRate GetPaddingBitrate(const VideoCodec& codec);
+  static absl::InlinedVector<DataRate, kMaxSpatialLayers> GetLayerStartBitrates(
+      const VideoCodec& codec);
+
+ private:
+  struct NumLayers {
+    size_t spatial = 1;
+    size_t temporal = 1;
+  };
+
+  static NumLayers GetNumLayers(const VideoCodec& codec);
+  VideoBitrateAllocation GetAllocationNormalVideo(
+      DataRate total_bitrate,
+      size_t first_active_layer,
+      size_t num_spatial_layers) const;
+
+  VideoBitrateAllocation GetAllocationScreenSharing(
+      DataRate total_bitrate,
+      size_t first_active_layer,
+      size_t num_spatial_layers) const;
+
+  // Returns the number of layers that are active and have enough bitrate to
+  // actually be enabled.
+  size_t FindNumEnabledLayers(DataRate target_rate) const;
+
+  const VideoCodec codec_;
+  const NumLayers num_layers_;
+  const StableTargetRateExperiment experiment_settings_;
+  const absl::InlinedVector<DataRate, kMaxSpatialLayers>
+      cumulative_layer_start_bitrates_;
+  size_t last_active_layer_count_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_SVC_SVC_RATE_ALLOCATOR_H_
diff --git a/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_gn/moz.build
new file mode 100644
index 0000000000..5d7d952a69
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("svc_rate_allocator_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_unittest.cc b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_unittest.cc
new file mode 100644
index 0000000000..b3a365d722
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/svc/svc_rate_allocator_unittest.cc
@@ -0,0 +1,584 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/svc/svc_rate_allocator.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "modules/video_coding/codecs/vp9/svc_config.h"
+#include "rtc_base/checks.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+static VideoCodec Configure(size_t width,
+                            size_t height,
+                            size_t num_spatial_layers,
+                            size_t num_temporal_layers,
+                            bool is_screen_sharing) {
+  VideoCodec codec;
+  codec.width = width;
+  codec.height = height;
+  codec.codecType = kVideoCodecVP9;
+  codec.mode = is_screen_sharing ? VideoCodecMode::kScreensharing
+                                 : VideoCodecMode::kRealtimeVideo;
+
+  std::vector<SpatialLayer> spatial_layers =
+      GetSvcConfig(width, height, 30, /*first_active_layer=*/0,
+                   num_spatial_layers, num_temporal_layers, is_screen_sharing);
+  RTC_CHECK_LE(spatial_layers.size(), kMaxSpatialLayers);
+
+  codec.VP9()->numberOfSpatialLayers =
+      std::min<unsigned char>(num_spatial_layers, spatial_layers.size());
+  codec.VP9()->numberOfTemporalLayers = std::min<unsigned char>(
+      num_temporal_layers, spatial_layers.back().numberOfTemporalLayers);
+
+  for (size_t sl_idx = 0; sl_idx < spatial_layers.size(); ++sl_idx) {
+    codec.spatialLayers[sl_idx] = spatial_layers[sl_idx];
+  }
+
+  return codec;
+}
+
+}  // namespace
+
+TEST(SvcRateAllocatorTest, SingleLayerFor320x180Input) {
+  VideoCodec codec = Configure(320, 180, 3, 3, false);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  VideoBitrateAllocation allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(1000 * 1000, 30));
+
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(1), 0u);
+}
+
+TEST(SvcRateAllocatorTest, TwoLayersFor640x360Input) {
+  VideoCodec codec = Configure(640, 360, 3, 3, false);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  VideoBitrateAllocation allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(1000 * 1000, 30));
+
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0u);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(2), 0u);
+}
+
+TEST(SvcRateAllocatorTest, ThreeLayersFor1280x720Input) {
+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  VideoBitrateAllocation allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(1000 * 1000, 30));
+
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0u);
+  EXPECT_GT(allocation.GetSpatialLayerSum(2), 0u);
+}
+
+TEST(SvcRateAllocatorTest,
+     BaseLayerNonZeroBitrateEvenIfTotalIfLessThanMinimum) {
+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  const SpatialLayer* layers = codec.spatialLayers;
+
+  VideoBitrateAllocation allocation = allocator.Allocate(
+      VideoBitrateAllocationParameters(layers[0].minBitrate * 1000 / 2, 30));
+
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
+  EXPECT_LT(allocation.GetSpatialLayerSum(0), layers[0].minBitrate * 1000);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(1), 0u);
+}
+
+TEST(SvcRateAllocatorTest, Disable640x360Layer) {
+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  const SpatialLayer* layers = codec.spatialLayers;
+
+  size_t min_bitrate_for_640x360_layer_kbps =
+      layers[0].minBitrate + layers[1].minBitrate;
+
+  VideoBitrateAllocation allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(
+          min_bitrate_for_640x360_layer_kbps * 1000 - 1, 30));
+
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(1), 0u);
+}
+
+TEST(SvcRateAllocatorTest, Disable1280x720Layer) {
+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  const SpatialLayer* layers = codec.spatialLayers;
+
+  size_t min_bitrate_for_1280x720_layer_kbps =
+      layers[0].minBitrate + layers[1].minBitrate + layers[2].minBitrate;
+
+  VideoBitrateAllocation allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(
+          min_bitrate_for_1280x720_layer_kbps * 1000 - 1, 30));
+
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0u);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(2), 0u);
+}
+
+TEST(SvcRateAllocatorTest, BitrateIsCapped) {
+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  const SpatialLayer* layers = codec.spatialLayers;
+
+  const uint32_t link_mbps = 100;
+  VideoBitrateAllocation allocation = allocator.Allocate(
+      VideoBitrateAllocationParameters(link_mbps * 1000000, 30));
+
+  EXPECT_EQ(allocation.get_sum_kbps(),
+            layers[0].maxBitrate + layers[1].maxBitrate + layers[2].maxBitrate);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(0) / 1000, layers[0].maxBitrate);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(1) / 1000, layers[1].maxBitrate);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(2) / 1000, layers[2].maxBitrate);
+}
+
+TEST(SvcRateAllocatorTest, MinBitrateToGetQualityLayer) {
+  VideoCodec codec = Configure(1280, 720, 3, 1, true);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  const SpatialLayer* layers = codec.spatialLayers;
+
+  EXPECT_LE(codec.VP9()->numberOfSpatialLayers, 3U);
+
+  VideoBitrateAllocation allocation = allocator.Allocate(
+      VideoBitrateAllocationParameters(layers[0].minBitrate * 1000, 30));
+  EXPECT_EQ(allocation.GetSpatialLayerSum(0) / 1000, layers[0].minBitrate);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(1), 0UL);
+
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      (layers[0].targetBitrate + layers[1].minBitrate) * 1000, 30));
+  EXPECT_EQ(allocation.GetSpatialLayerSum(0) / 1000, layers[0].targetBitrate);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(1) / 1000, layers[1].minBitrate);
+}
+
+TEST(SvcRateAllocatorTest, DeactivateHigherLayers) {
+  for (int deactivated_idx = 2; deactivated_idx >= 0; --deactivated_idx) {
+    VideoCodec codec = Configure(1280, 720, 3, 1, false);
+    EXPECT_LE(codec.VP9()->numberOfSpatialLayers, 3U);
+
+    for (int i = deactivated_idx; i < 3; ++i)
+      codec.spatialLayers[i].active = false;
+
+    SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+    VideoBitrateAllocation allocation = allocator.Allocate(
+        VideoBitrateAllocationParameters(10 * 1000 * 1000, 30));
+
+    // Ensure layers spatial_idx < deactivated_idx are activated.
+    for (int spatial_idx = 0; spatial_idx < deactivated_idx; ++spatial_idx) {
+      EXPECT_GT(allocation.GetSpatialLayerSum(spatial_idx), 0UL);
+    }
+
+    // Ensure layers spatial_idx >= deactivated_idx are deactivated.
+    for (int spatial_idx = deactivated_idx; spatial_idx < 3; ++spatial_idx) {
+      EXPECT_EQ(allocation.GetSpatialLayerSum(spatial_idx), 0UL);
+    }
+  }
+}
+
+TEST(SvcRateAllocatorTest, DeactivateLowerLayers) {
+  for (int deactivated_idx = 0; deactivated_idx < 3; ++deactivated_idx) {
+    VideoCodec codec = Configure(1280, 720, 3, 1, false);
+    EXPECT_LE(codec.VP9()->numberOfSpatialLayers, 3U);
+
+    for (int i = deactivated_idx; i >= 0; --i)
+      codec.spatialLayers[i].active = false;
+
+    SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+    VideoBitrateAllocation allocation = allocator.Allocate(
+        VideoBitrateAllocationParameters(10 * 1000 * 1000, 30));
+
+    // Ensure layers spatial_idx <= deactivated_idx are deactivated.
+    for (int spatial_idx = 0; spatial_idx <= deactivated_idx; ++spatial_idx) {
+      EXPECT_EQ(allocation.GetSpatialLayerSum(spatial_idx), 0UL);
+    }
+
+    // Ensure layers spatial_idx > deactivated_idx are activated.
+    for (int spatial_idx = deactivated_idx + 1; spatial_idx < 3;
+         ++spatial_idx) {
+      EXPECT_GT(allocation.GetSpatialLayerSum(spatial_idx), 0UL);
+    }
+  }
+}
+
+TEST(SvcRateAllocatorTest, SignalsBwLimited) {
+  VideoCodec codec = Configure(1280, 720, 3, 1, false);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  // Rough estimate calculated by hand.
+  uint32_t min_to_enable_all = 900000;
+
+  EXPECT_TRUE(
+      allocator
+          .Allocate(VideoBitrateAllocationParameters(min_to_enable_all / 2, 30))
+          .is_bw_limited());
+
+  EXPECT_FALSE(
+      allocator
+          .Allocate(VideoBitrateAllocationParameters(min_to_enable_all, 30))
+          .is_bw_limited());
+}
+
+TEST(SvcRateAllocatorTest, NoPaddingIfAllLayersAreDeactivated) {
+  VideoCodec codec = Configure(1280, 720, 3, 1, false);
+  EXPECT_EQ(codec.VP9()->numberOfSpatialLayers, 3U);
+  // Deactivation of base layer deactivates all layers.
+  codec.spatialLayers[0].active = false;
+  codec.spatialLayers[1].active = false;
+  codec.spatialLayers[2].active = false;
+  DataRate padding_rate = SvcRateAllocator::GetPaddingBitrate(codec);
+  EXPECT_EQ(padding_rate, DataRate::Zero());
+}
+
+TEST(SvcRateAllocatorTest, FindLayerTogglingThreshold) {
+  // Let's unit test a utility method of the unit test...
+
+  // Predetermined constants indicating the min bitrate needed for two and three
+  // layers to be enabled respectively, using the config from Configure() with
+  // 1280x720 resolution and three spatial layers.
+  const DataRate kTwoLayerMinRate = DataRate::BitsPerSec(299150);
+  const DataRate kThreeLayerMinRate = DataRate::BitsPerSec(891052);
+
+  VideoCodec codec = Configure(1280, 720, 3, 1, false);
+  absl::InlinedVector<DataRate, kMaxSpatialLayers> layer_start_bitrates =
+      SvcRateAllocator::GetLayerStartBitrates(codec);
+  ASSERT_EQ(layer_start_bitrates.size(), 3u);
+  EXPECT_EQ(layer_start_bitrates[1], kTwoLayerMinRate);
+  EXPECT_EQ(layer_start_bitrates[2], kThreeLayerMinRate);
+}
+
+TEST(SvcRateAllocatorTest, SupportsAv1) {
+  VideoCodec codec;
+  codec.width = 640;
+  codec.height = 360;
+  codec.codecType = kVideoCodecAV1;
+  codec.SetScalabilityMode(ScalabilityMode::kL3T3);
+  codec.spatialLayers[0].active = true;
+  codec.spatialLayers[0].minBitrate = 30;
+  codec.spatialLayers[0].targetBitrate = 51;
+  codec.spatialLayers[0].maxBitrate = 73;
+  codec.spatialLayers[1].active = true;
+  codec.spatialLayers[1].minBitrate = 49;
+  codec.spatialLayers[1].targetBitrate = 64;
+  codec.spatialLayers[1].maxBitrate = 97;
+  codec.spatialLayers[2].active = true;
+  codec.spatialLayers[2].minBitrate = 193;
+  codec.spatialLayers[2].targetBitrate = 305;
+  codec.spatialLayers[2].maxBitrate = 418;
+
+  SvcRateAllocator allocator(codec);
+
+  VideoBitrateAllocation allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(1'000'000, 30));
+
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0u);
+  EXPECT_GT(allocation.GetSpatialLayerSum(2), 0u);
+}
+
+TEST(SvcRateAllocatorTest, SupportsAv1WithSkippedLayer) {
+  VideoCodec codec;
+  codec.width = 640;
+  codec.height = 360;
+  codec.codecType = kVideoCodecAV1;
+  codec.SetScalabilityMode(ScalabilityMode::kL3T3);
+  codec.spatialLayers[0].active = false;
+  codec.spatialLayers[0].minBitrate = 30;
+  codec.spatialLayers[0].targetBitrate = 51;
+  codec.spatialLayers[0].maxBitrate = 73;
+  codec.spatialLayers[1].active = true;
+  codec.spatialLayers[1].minBitrate = 49;
+  codec.spatialLayers[1].targetBitrate = 64;
+  codec.spatialLayers[1].maxBitrate = 97;
+  codec.spatialLayers[2].active = true;
+  codec.spatialLayers[2].minBitrate = 193;
+  codec.spatialLayers[2].targetBitrate = 305;
+  codec.spatialLayers[2].maxBitrate = 418;
+
+  SvcRateAllocator allocator(codec);
+
+  VideoBitrateAllocation allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(1'000'000, 30));
+
+  EXPECT_EQ(allocation.GetSpatialLayerSum(0), 0u);
+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0u);
+  EXPECT_GT(allocation.GetSpatialLayerSum(2), 0u);
+}
+
+TEST(SvcRateAllocatorTest, UsesScalabilityModeToGetNumberOfLayers) {
+  VideoCodec codec;
+  codec.width = 640;
+  codec.height = 360;
+  codec.codecType = kVideoCodecAV1;
+  codec.SetScalabilityMode(ScalabilityMode::kL2T2);
+  codec.spatialLayers[0].active = true;
+  codec.spatialLayers[0].minBitrate = 30;
+  codec.spatialLayers[0].targetBitrate = 51;
+  codec.spatialLayers[0].maxBitrate = 73;
+  codec.spatialLayers[1].active = true;
+  codec.spatialLayers[1].minBitrate = 49;
+  codec.spatialLayers[1].targetBitrate = 64;
+  codec.spatialLayers[1].maxBitrate = 97;
+  codec.spatialLayers[2].active = true;
+  codec.spatialLayers[2].minBitrate = 193;
+  codec.spatialLayers[2].targetBitrate = 305;
+  codec.spatialLayers[2].maxBitrate = 418;
+
+  SvcRateAllocator allocator(codec);
+  VideoBitrateAllocation allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(1'000'000, 30));
+
+  // Expect bitrates for 2 temporal layers.
+  EXPECT_TRUE(allocation.HasBitrate(1, /*temporal_index=*/0));
+  EXPECT_TRUE(allocation.HasBitrate(1, /*temporal_index=*/1));
+  EXPECT_FALSE(allocation.HasBitrate(1, /*temporal_index=*/2));
+
+  // expect codec.spatialLayers[2].active is ignored because scability mode uses
+  // just 2 spatial layers.
+  EXPECT_EQ(allocation.GetSpatialLayerSum(2), 0u);
+}
+
+class SvcRateAllocatorTestParametrizedContentType
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<bool> {
+ public:
+  SvcRateAllocatorTestParametrizedContentType()
+      : is_screen_sharing_(GetParam()) {}
+
+  const bool is_screen_sharing_;
+};
+
+TEST_P(SvcRateAllocatorTestParametrizedContentType, MaxBitrate) {
+  VideoCodec codec = Configure(1280, 720, 3, 1, is_screen_sharing_);
+  EXPECT_EQ(SvcRateAllocator::GetMaxBitrate(codec),
+            DataRate::KilobitsPerSec(codec.spatialLayers[0].maxBitrate +
+                                     codec.spatialLayers[1].maxBitrate +
+                                     codec.spatialLayers[2].maxBitrate));
+
+  // Deactivate middle layer. This causes deactivation of top layer as well.
+  codec.spatialLayers[1].active = false;
+  EXPECT_EQ(SvcRateAllocator::GetMaxBitrate(codec),
+            DataRate::KilobitsPerSec(codec.spatialLayers[0].maxBitrate));
+}
+
+TEST_P(SvcRateAllocatorTestParametrizedContentType, PaddingBitrate) {
+  VideoCodec codec = Configure(1280, 720, 3, 1, is_screen_sharing_);
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  DataRate padding_bitrate = SvcRateAllocator::GetPaddingBitrate(codec);
+
+  VideoBitrateAllocation allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(padding_bitrate, 30));
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0UL);
+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0UL);
+  EXPECT_GT(allocation.GetSpatialLayerSum(2), 0UL);
+
+  // Allocate 90% of padding bitrate. Top layer should be disabled.
+  allocation = allocator.Allocate(
+      VideoBitrateAllocationParameters(9 * padding_bitrate / 10, 30));
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0UL);
+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0UL);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(2), 0UL);
+
+  // Deactivate top layer.
+  codec.spatialLayers[2].active = false;
+
+  padding_bitrate = SvcRateAllocator::GetPaddingBitrate(codec);
+  allocation =
+      allocator.Allocate(VideoBitrateAllocationParameters(padding_bitrate, 30));
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0UL);
+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0UL);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(2), 0UL);
+
+  allocation = allocator.Allocate(
+      VideoBitrateAllocationParameters(9 * padding_bitrate / 10, 30));
+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0UL);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(1), 0UL);
+  EXPECT_EQ(allocation.GetSpatialLayerSum(2), 0UL);
+
+  // Deactivate all layers.
+  codec.spatialLayers[0].active = false;
+  codec.spatialLayers[1].active = false;
+  codec.spatialLayers[2].active = false;
+
+  padding_bitrate = SvcRateAllocator::GetPaddingBitrate(codec);
+  // No padding expected.
+  EXPECT_EQ(DataRate::Zero(), padding_bitrate);
+}
+
+TEST_P(SvcRateAllocatorTestParametrizedContentType, StableBitrate) {
+  ScopedFieldTrials field_trial(
+      "WebRTC-StableTargetRate/enabled:true,video_hysteresis_factor:1.0,"
+      "screenshare_hysteresis_factor:1.0/");
+
+  const VideoCodec codec = Configure(1280, 720, 3, 1, is_screen_sharing_);
+  const auto start_rates = SvcRateAllocator::GetLayerStartBitrates(codec);
+  const DataRate min_rate_two_layers = start_rates[1];
+  const DataRate min_rate_three_layers = start_rates[2];
+
+  const DataRate max_rate_one_layer =
+      DataRate::KilobitsPerSec(codec.spatialLayers[0].maxBitrate);
+  const DataRate max_rate_two_layers =
+      is_screen_sharing_
+          ? DataRate::KilobitsPerSec(codec.spatialLayers[0].targetBitrate +
+                                     codec.spatialLayers[1].maxBitrate)
+          : DataRate::KilobitsPerSec(codec.spatialLayers[0].maxBitrate +
+                                     codec.spatialLayers[1].maxBitrate);
+
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+
+  // Two layers, stable and target equal.
+  auto allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/min_rate_two_layers,
+      /*stable_bitrate=*/min_rate_two_layers, /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_EQ(allocation.get_sum_bps(), min_rate_two_layers.bps());
+
+  // Two layers, stable bitrate too low for two layers.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/min_rate_two_layers,
+      /*stable_bitrate=*/min_rate_two_layers - DataRate::BitsPerSec(1),
+      /*fps=*/30.0));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_EQ(DataRate::BitsPerSec(allocation.get_sum_bps()),
+            std::min(min_rate_two_layers - DataRate::BitsPerSec(1),
+                     max_rate_one_layer));
+
+  // Three layers, stable and target equal.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/min_rate_three_layers,
+      /*stable_bitrate=*/min_rate_three_layers, /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(2));
+  EXPECT_EQ(allocation.get_sum_bps(), min_rate_three_layers.bps());
+
+  // Three layers, stable bitrate too low for three layers.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/min_rate_three_layers,
+      /*stable_bitrate=*/min_rate_three_layers - DataRate::BitsPerSec(1),
+      /*fps=*/30.0));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(2));
+  EXPECT_EQ(DataRate::BitsPerSec(allocation.get_sum_bps()),
+            std::min(min_rate_three_layers - DataRate::BitsPerSec(1),
+                     max_rate_two_layers));
+}
+
+TEST_P(SvcRateAllocatorTestParametrizedContentType,
+       StableBitrateWithHysteresis) {
+  const VideoCodec codec = Configure(1280, 720, 3, 1, is_screen_sharing_);
+  const auto start_rates = SvcRateAllocator::GetLayerStartBitrates(codec);
+  const DataRate min_rate_single_layer = start_rates[0];
+  const DataRate min_rate_two_layers = start_rates[1];
+  const DataRate min_rate_three_layers = start_rates[2];
+
+  ScopedFieldTrials field_trial(
+      "WebRTC-StableTargetRate/enabled:true,video_hysteresis_factor:1.1,"
+      "screenshare_hysteresis_factor:1.1/");
+  SvcRateAllocator allocator = SvcRateAllocator(codec);
+  // Always use max bitrate as target, verify only stable is used for layer
+  // count selection.
+  const DataRate max_bitrate = allocator.GetMaxBitrate(codec);
+
+  // Start with a single layer.
+  auto allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/max_bitrate,
+      /*stable_bitrate=*/min_rate_single_layer, /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(0));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(2));
+
+  // Min bitrate not enough to enable second layer due to 10% hysteresis.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/max_bitrate,
+      /*stable_bitrate=*/min_rate_two_layers, /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(0));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(2));
+
+  // Add hysteresis, second layer should turn on.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/max_bitrate,
+      /*stable_bitrate=*/min_rate_two_layers * 1.1, /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(2));
+
+  // Remove hysteresis, second layer should stay on.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/max_bitrate,
+      /*stable_bitrate=*/min_rate_two_layers, /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(2));
+
+  // Going below min for two layers, second layer should turn off again.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/max_bitrate,
+      /*stable_bitrate=*/min_rate_two_layers - DataRate::BitsPerSec(1),
+      /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(0));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(2));
+
+  // Min bitrate not enough to enable third layer due to 10% hysteresis.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/max_bitrate,
+      /*stable_bitrate=*/min_rate_three_layers, /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(2));
+
+  // Add hysteresis, third layer should turn on.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/max_bitrate,
+      /*stable_bitrate=*/min_rate_three_layers * 1.1, /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(2));
+
+  // Remove hysteresis, third layer should stay on.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/max_bitrate,
+      /*stable_bitrate=*/min_rate_three_layers, /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(2));
+
+  // Going below min for three layers, third layer should turn off again.
+  allocation = allocator.Allocate(VideoBitrateAllocationParameters(
+      /*total_bitrate=*/max_bitrate,
+      /*stable_bitrate=*/min_rate_three_layers - DataRate::BitsPerSec(1),
+      /*fps=*/30.0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(0));
+  EXPECT_TRUE(allocation.IsSpatialLayerUsed(1));
+  EXPECT_FALSE(allocation.IsSpatialLayerUsed(2));
+}
+
+INSTANTIATE_TEST_SUITE_P(_,
+                         SvcRateAllocatorTestParametrizedContentType,
+                         ::testing::Bool());
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/test/stream_generator.cc b/third_party/libwebrtc/modules/video_coding/test/stream_generator.cc
new file mode 100644
index 0000000000..98a0cf1cdc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/test/stream_generator.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/test/stream_generator.h"
+
+#include <string.h>
+
+#include <list>
+
+#include "modules/video_coding/packet.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+StreamGenerator::StreamGenerator(uint16_t start_seq_num, int64_t current_time)
+    : packets_(), sequence_number_(start_seq_num), start_time_(current_time) {}
+
+void StreamGenerator::Init(uint16_t start_seq_num, int64_t current_time) {
+  packets_.clear();
+  sequence_number_ = start_seq_num;
+  start_time_ = current_time;
+  memset(packet_buffer_, 0, sizeof(packet_buffer_));
+}
+
+void StreamGenerator::GenerateFrame(VideoFrameType type,
+                                    int num_media_packets,
+                                    int num_empty_packets,
+                                    int64_t time_ms) {
+  uint32_t timestamp = 90 * (time_ms - start_time_);
+  for (int i = 0; i < num_media_packets; ++i) {
+    const int packet_size =
+        (kFrameSize + num_media_packets / 2) / num_media_packets;
+    bool marker_bit = (i == num_media_packets - 1);
+    packets_.push_back(GeneratePacket(sequence_number_, timestamp, packet_size,
+                                      (i == 0), marker_bit, type));
+    ++sequence_number_;
+  }
+  for (int i = 0; i < num_empty_packets; ++i) {
+    packets_.push_back(GeneratePacket(sequence_number_, timestamp, 0, false,
+                                      false, VideoFrameType::kEmptyFrame));
+    ++sequence_number_;
+  }
+}
+
+VCMPacket StreamGenerator::GeneratePacket(uint16_t sequence_number,
+                                          uint32_t timestamp,
+                                          unsigned int size,
+                                          bool first_packet,
+                                          bool marker_bit,
+                                          VideoFrameType type) {
+  EXPECT_LT(size, kMaxPacketSize);
+  VCMPacket packet;
+  packet.seqNum = sequence_number;
+  packet.timestamp = timestamp;
+  packet.video_header.frame_type = type;
+  packet.video_header.is_first_packet_in_frame = first_packet;
+  packet.markerBit = marker_bit;
+  packet.sizeBytes = size;
+  packet.dataPtr = packet_buffer_;
+  if (packet.is_first_packet_in_frame())
+    packet.completeNALU = kNaluStart;
+  else if (packet.markerBit)
+    packet.completeNALU = kNaluEnd;
+  else
+    packet.completeNALU = kNaluIncomplete;
+  return packet;
+}
+
+bool StreamGenerator::PopPacket(VCMPacket* packet, int index) {
+  std::list<VCMPacket>::iterator it = GetPacketIterator(index);
+  if (it == packets_.end())
+    return false;
+  if (packet)
+    *packet = (*it);
+  packets_.erase(it);
+  return true;
+}
+
+bool StreamGenerator::GetPacket(VCMPacket* packet, int index) {
+  std::list<VCMPacket>::iterator it = GetPacketIterator(index);
+  if (it == packets_.end())
+    return false;
+  if (packet)
+    *packet = (*it);
+  return true;
+}
+
+bool StreamGenerator::NextPacket(VCMPacket* packet) {
+  if (packets_.empty())
+    return false;
+  if (packet != NULL)
+    *packet = packets_.front();
+  packets_.pop_front();
+  return true;
+}
+
+void StreamGenerator::DropLastPacket() {
+  packets_.pop_back();
+}
+
+uint16_t StreamGenerator::NextSequenceNumber() const {
+  if (packets_.empty())
+    return sequence_number_;
+  return packets_.front().seqNum;
+}
+
+int StreamGenerator::PacketsRemaining() const {
+  return packets_.size();
+}
+
+std::list<VCMPacket>::iterator StreamGenerator::GetPacketIterator(int index) {
+  std::list<VCMPacket>::iterator it = packets_.begin();
+  for (int i = 0; i < index; ++i) {
+    ++it;
+    if (it == packets_.end())
+      break;
+  }
+  return it;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/test/stream_generator.h b/third_party/libwebrtc/modules/video_coding/test/stream_generator.h
new file mode 100644
index 0000000000..ddb23ebb76
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/test/stream_generator.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
+#define MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
+
+#include <stdint.h>
+
+#include <list>
+
+#include "modules/video_coding/packet.h"
+
+namespace webrtc {
+
+const unsigned int kDefaultBitrateKbps = 1000;
+const unsigned int kDefaultFrameRate = 25;
+const unsigned int kMaxPacketSize = 1500;
+const unsigned int kFrameSize =
+    (kDefaultBitrateKbps + kDefaultFrameRate * 4) / (kDefaultFrameRate * 8);
+const int kDefaultFramePeriodMs = 1000 / kDefaultFrameRate;
+
+class StreamGenerator {
+ public:
+  StreamGenerator(uint16_t start_seq_num, int64_t current_time);
+
+  StreamGenerator(const StreamGenerator&) = delete;
+  StreamGenerator& operator=(const StreamGenerator&) = delete;
+
+  void Init(uint16_t start_seq_num, int64_t current_time);
+
+  // `time_ms` denotes the timestamp you want to put on the frame, and the unit
+  // is millisecond. GenerateFrame will translate `time_ms` into a 90kHz
+  // timestamp and put it on the frame.
+  void GenerateFrame(VideoFrameType type,
+                     int num_media_packets,
+                     int num_empty_packets,
+                     int64_t time_ms);
+
+  bool PopPacket(VCMPacket* packet, int index);
+  void DropLastPacket();
+
+  bool GetPacket(VCMPacket* packet, int index);
+
+  bool NextPacket(VCMPacket* packet);
+
+  uint16_t NextSequenceNumber() const;
+
+  int PacketsRemaining() const;
+
+ private:
+  VCMPacket GeneratePacket(uint16_t sequence_number,
+                           uint32_t timestamp,
+                           unsigned int size,
+                           bool first_packet,
+                           bool marker_bit,
+                           VideoFrameType type);
+
+  std::list<VCMPacket>::iterator GetPacketIterator(int index);
+
+  std::list<VCMPacket> packets_;
+  uint16_t sequence_number_;
+  int64_t start_time_;
+  uint8_t packet_buffer_[kMaxPacketSize];
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_TEST_STREAM_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/video_coding/timing/BUILD.gn b/third_party/libwebrtc/modules/video_coding/timing/BUILD.gn
new file mode 100644
index 0000000000..38348e6967
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/BUILD.gn
@@ -0,0 +1,153 @@
+# Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("codec_timer") {
+  sources = [
+    "codec_timer.cc",
+    "codec_timer.h",
+  ]
+  deps = [ "../../../rtc_base:rtc_numerics" ]
+}
+
+rtc_library("inter_frame_delay") {
+  sources = [
+    "inter_frame_delay.cc",
+    "inter_frame_delay.h",
+  ]
+  deps = [
+    "../..:module_api_public",
+    "../../../api/units:frequency",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:rtc_numerics",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("frame_delay_variation_kalman_filter") {
+  sources = [
+    "frame_delay_variation_kalman_filter.cc",
+    "frame_delay_variation_kalman_filter.h",
+  ]
+  deps = [
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+  ]
+  visibility = [
+    ":jitter_estimator",
+    ":timing_unittests",
+  ]
+}
+
+rtc_library("jitter_estimator") {
+  sources = [
+    "jitter_estimator.cc",
+    "jitter_estimator.h",
+  ]
+  deps = [
+    ":frame_delay_variation_kalman_filter",
+    ":rtt_filter",
+    "../../../api:field_trials_view",
+    "../../../api/units:data_size",
+    "../../../api/units:frequency",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:rolling_accumulator",
+    "../../../rtc_base:rtc_numerics",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../system_wrappers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("rtt_filter") {
+  sources = [
+    "rtt_filter.cc",
+    "rtt_filter.h",
+  ]
+  deps = [ "../../../api/units:time_delta" ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
+  ]
+}
+
+rtc_library("timestamp_extrapolator") {
+  sources = [
+    "timestamp_extrapolator.cc",
+    "timestamp_extrapolator.h",
+  ]
+  deps = [
+    "../../../api/units:timestamp",
+    "../../../modules:module_api_public",
+    "../../../rtc_base:rtc_numerics",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("timing_module") {
+  sources = [
+    "timing.cc",
+    "timing.h",
+  ]
+  deps = [
+    ":codec_timer",
+    ":timestamp_extrapolator",
+    "../../../api:field_trials_view",
+    "../../../api/units:time_delta",
+    "../../../api/video:video_frame",
+    "../../../api/video:video_rtp_headers",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:macromagic",
+    "../../../rtc_base:rtc_numerics",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../rtc_base/synchronization:mutex",
+    "../../../system_wrappers",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("timing_unittests") {
+  testonly = true
+  sources = [
+    "frame_delay_variation_kalman_filter_unittest.cc",
+    "inter_frame_delay_unittest.cc",
+    "jitter_estimator_unittest.cc",
+    "rtt_filter_unittest.cc",
+    "timestamp_extrapolator_unittest.cc",
+    "timing_unittest.cc",
+  ]
+  deps = [
+    ":frame_delay_variation_kalman_filter",
+    ":inter_frame_delay",
+    ":jitter_estimator",
+    ":rtt_filter",
+    ":timestamp_extrapolator",
+    ":timing_module",
+    "../../../api:array_view",
+    "../../../api:field_trials",
+    "../../../api/units:data_size",
+    "../../../api/units:frequency",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:histogram_percentile_counter",
+    "../../../rtc_base:timeutils",
+    "../../../system_wrappers:system_wrappers",
+    "../../../test:scoped_key_value_config",
+    "../../../test:test_support",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
diff --git a/third_party/libwebrtc/modules/video_coding/timing/codec_timer.cc b/third_party/libwebrtc/modules/video_coding/timing/codec_timer.cc
new file mode 100644
index 0000000000..f57d42d40a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/codec_timer.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/codec_timer.h"
+
+#include <cstdint>
+
+namespace webrtc {
+
+namespace {
+
+// The first kIgnoredSampleCount samples will be ignored.
+const int kIgnoredSampleCount = 5;
+// Return the `kPercentile` value in RequiredDecodeTimeMs().
+const float kPercentile = 0.95f;
+// The window size in ms.
+const int64_t kTimeLimitMs = 10000;
+
+}  // anonymous namespace
+
+CodecTimer::CodecTimer() : ignored_sample_count_(0), filter_(kPercentile) {}
+CodecTimer::~CodecTimer() = default;
+
+void CodecTimer::AddTiming(int64_t decode_time_ms, int64_t now_ms) {
+  // Ignore the first `kIgnoredSampleCount` samples.
+  if (ignored_sample_count_ < kIgnoredSampleCount) {
+    ++ignored_sample_count_;
+    return;
+  }
+
+  // Insert new decode time value.
+  filter_.Insert(decode_time_ms);
+  history_.emplace(decode_time_ms, now_ms);
+
+  // Pop old decode time values.
+  while (!history_.empty() &&
+         now_ms - history_.front().sample_time_ms > kTimeLimitMs) {
+    filter_.Erase(history_.front().decode_time_ms);
+    history_.pop();
+  }
+}
+
+// Get the 95th percentile observed decode time within a time window.
+int64_t CodecTimer::RequiredDecodeTimeMs() const {
+  return filter_.GetPercentileValue();
+}
+
+CodecTimer::Sample::Sample(int64_t decode_time_ms, int64_t sample_time_ms)
+    : decode_time_ms(decode_time_ms), sample_time_ms(sample_time_ms) {}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/codec_timer.h b/third_party/libwebrtc/modules/video_coding/timing/codec_timer.h
new file mode 100644
index 0000000000..9f12d82e98
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/codec_timer.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_TIMING_CODEC_TIMER_H_
+#define MODULES_VIDEO_CODING_TIMING_CODEC_TIMER_H_
+
+#include <queue>
+
+#include "rtc_base/numerics/percentile_filter.h"
+
+namespace webrtc {
+
+class CodecTimer {
+ public:
+  CodecTimer();
+  ~CodecTimer();
+
+  // Add a new decode time to the filter.
+  void AddTiming(int64_t new_decode_time_ms, int64_t now_ms);
+
+  // Get the required decode time in ms. It is the 95th percentile observed
+  // decode time within a time window.
+  int64_t RequiredDecodeTimeMs() const;
+
+ private:
+  struct Sample {
+    Sample(int64_t decode_time_ms, int64_t sample_time_ms);
+    int64_t decode_time_ms;
+    int64_t sample_time_ms;
+  };
+
+  // The number of samples ignored so far.
+  int ignored_sample_count_;
+  // Queue with history of latest decode time values.
+  std::queue<Sample> history_;
+  // `filter_` contains the same values as `history_`, but in a data structure
+  // that allows efficient retrieval of the percentile value.
+  PercentileFilter<int64_t> filter_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_TIMING_CODEC_TIMER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/timing/codec_timer_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/codec_timer_gn/moz.build
new file mode 100644
index 0000000000..59498feb91
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/codec_timer_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/timing/codec_timer.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("codec_timer_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter.cc b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter.cc
new file mode 100644
index 0000000000..ec6aa3445a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/frame_delay_variation_kalman_filter.h"
+
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+namespace {
+// TODO(brandtr): The value below corresponds to 8 Gbps. Is that reasonable?
+constexpr double kMaxBandwidth = 0.000001;  // Unit: [1 / bytes per ms].
+}  // namespace
+
+FrameDelayVariationKalmanFilter::FrameDelayVariationKalmanFilter() {
+  // TODO(brandtr): Is there a factor 1000 missing here?
+  estimate_[0] = 1 / (512e3 / 8);  // Unit: [1 / bytes per ms]
+  estimate_[1] = 0;                // Unit: [ms]
+
+  // Initial estimate covariance.
+  estimate_cov_[0][0] = 1e-4;  // Unit: [(1 / bytes per ms)^2]
+  estimate_cov_[1][1] = 1e2;   // Unit: [ms^2]
+  estimate_cov_[0][1] = estimate_cov_[1][0] = 0;
+
+  // Process noise covariance.
+  process_noise_cov_diag_[0] = 2.5e-10;  // Unit: [(1 / bytes per ms)^2]
+  process_noise_cov_diag_[1] = 1e-10;    // Unit: [ms^2]
+}
+
+void FrameDelayVariationKalmanFilter::PredictAndUpdate(
+    double frame_delay_variation_ms,
+    double frame_size_variation_bytes,
+    double max_frame_size_bytes,
+    double var_noise) {
+  // Sanity checks.
+  if (max_frame_size_bytes < 1) {
+    return;
+  }
+  if (var_noise <= 0.0) {
+    return;
+  }
+
+  // This member function follows the data flow in
+  // https://en.wikipedia.org/wiki/Kalman_filter#Details.
+
+  // 1) Estimate prediction: `x = F*x`.
+  // For this model, there is no need to explicitly predict the estimate, since
+  // the state transition matrix is the identity.
+
+  // 2) Estimate covariance prediction: `P = F*P*F' + Q`.
+  // Again, since the state transition matrix is the identity, this update
+  // is performed by simply adding the process noise covariance.
+  estimate_cov_[0][0] += process_noise_cov_diag_[0];
+  estimate_cov_[1][1] += process_noise_cov_diag_[1];
+
+  // 3) Innovation: `y = z - H*x`.
+  // This is the part of the measurement that cannot be explained by the current
+  // estimate.
+  double innovation =
+      frame_delay_variation_ms -
+      GetFrameDelayVariationEstimateTotal(frame_size_variation_bytes);
+
+  // 4) Innovation variance: `s = H*P*H' + r`.
+  double estim_cov_times_obs[2];
+  estim_cov_times_obs[0] =
+      estimate_cov_[0][0] * frame_size_variation_bytes + estimate_cov_[0][1];
+  estim_cov_times_obs[1] =
+      estimate_cov_[1][0] * frame_size_variation_bytes + estimate_cov_[1][1];
+  double observation_noise_stddev =
+      (300.0 * exp(-fabs(frame_size_variation_bytes) /
+                   (1e0 * max_frame_size_bytes)) +
+       1) *
+      sqrt(var_noise);
+  if (observation_noise_stddev < 1.0) {
+    observation_noise_stddev = 1.0;
+  }
+  // TODO(brandtr): Shouldn't we add observation_noise_stddev^2 here? Otherwise,
+  // the dimensional analysis fails.
+  double innovation_var = frame_size_variation_bytes * estim_cov_times_obs[0] +
+                          estim_cov_times_obs[1] + observation_noise_stddev;
+  if ((innovation_var < 1e-9 && innovation_var >= 0) ||
+      (innovation_var > -1e-9 && innovation_var <= 0)) {
+    RTC_DCHECK_NOTREACHED();
+    return;
+  }
+
+  // 5) Optimal Kalman gain: `K = P*H'/s`.
+  // How much to trust the model vs. how much to trust the measurement.
+  double kalman_gain[2];
+  kalman_gain[0] = estim_cov_times_obs[0] / innovation_var;
+  kalman_gain[1] = estim_cov_times_obs[1] / innovation_var;
+
+  // 6) Estimate update: `x = x + K*y`.
+  // Optimally weight the new information in the innovation and add it to the
+  // old estimate.
+  estimate_[0] += kalman_gain[0] * innovation;
+  estimate_[1] += kalman_gain[1] * innovation;
+
+  // (This clamping is not part of the linear Kalman filter.)
+  if (estimate_[0] < kMaxBandwidth) {
+    estimate_[0] = kMaxBandwidth;
+  }
+
+  // 7) Estimate covariance update: `P = (I - K*H)*P`
+  double t00 = estimate_cov_[0][0];
+  double t01 = estimate_cov_[0][1];
+  estimate_cov_[0][0] =
+      (1 - kalman_gain[0] * frame_size_variation_bytes) * t00 -
+      kalman_gain[0] * estimate_cov_[1][0];
+  estimate_cov_[0][1] =
+      (1 - kalman_gain[0] * frame_size_variation_bytes) * t01 -
+      kalman_gain[0] * estimate_cov_[1][1];
+  estimate_cov_[1][0] = estimate_cov_[1][0] * (1 - kalman_gain[1]) -
+                        kalman_gain[1] * frame_size_variation_bytes * t00;
+  estimate_cov_[1][1] = estimate_cov_[1][1] * (1 - kalman_gain[1]) -
+                        kalman_gain[1] * frame_size_variation_bytes * t01;
+
+  // Covariance matrix, must be positive semi-definite.
+  RTC_DCHECK(estimate_cov_[0][0] + estimate_cov_[1][1] >= 0 &&
+             estimate_cov_[0][0] * estimate_cov_[1][1] -
+                     estimate_cov_[0][1] * estimate_cov_[1][0] >=
+                 0 &&
+             estimate_cov_[0][0] >= 0);
+}
+
+double FrameDelayVariationKalmanFilter::GetFrameDelayVariationEstimateSizeBased(
+    double frame_size_variation_bytes) const {
+  // Unit: [1 / bytes per millisecond] * [bytes] = [milliseconds].
+  return estimate_[0] * frame_size_variation_bytes;
+}
+
+double FrameDelayVariationKalmanFilter::GetFrameDelayVariationEstimateTotal(
+    double frame_size_variation_bytes) const {
+  double frame_transmission_delay_ms =
+      GetFrameDelayVariationEstimateSizeBased(frame_size_variation_bytes);
+  double link_queuing_delay_ms = estimate_[1];
+  return frame_transmission_delay_ms + link_queuing_delay_ms;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter.h b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter.h
new file mode 100644
index 0000000000..a65ceefa10
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter.h
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_TIMING_FRAME_DELAY_VARIATION_KALMAN_FILTER_H_
+#define MODULES_VIDEO_CODING_TIMING_FRAME_DELAY_VARIATION_KALMAN_FILTER_H_
+
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+// This class uses a linear Kalman filter (see
+// https://en.wikipedia.org/wiki/Kalman_filter) to estimate the frame delay
+// variation (i.e., the difference in transmission time between a frame and the
+// prior frame) for a frame, given its size variation in bytes (i.e., the
+// difference in size between a frame and the prior frame). The idea is that,
+// given a fixed link bandwidth, a larger frame (in bytes) would take
+// proportionally longer to arrive than a correspondingly smaller frame. Using
+// the variations of frame delay and frame size, the underlying bandwidth and
+// queuing delay variation of the network link can be estimated.
+//
+// The filter takes as input the frame delay variation, the difference between
+// the actual inter-frame arrival time and the expected inter-frame arrival time
+// (based on RTP timestamp), and frame size variation, the inter-frame size
+// delta for a single frame. The frame delay variation is seen as the
+// measurement and the frame size variation is used in the observation model.
+// The hidden state of the filter is the link bandwidth and queuing delay
+// buildup. The estimated state can be used to get the expected frame delay
+// variation for a frame, given its frame size variation. This information can
+// then be used to estimate the frame delay variation coming from network
+// jitter.
+//
+// Mathematical details:
+//  * The state (`x` in Wikipedia notation) is a 2x1 vector comprising the
+//    reciprocal of link bandwidth [1 / bytes per ms] and the
+//    link queuing delay buildup [ms].
+//  * The state transition matrix (`F`) is the 2x2 identity matrix, meaning that
+//    link bandwidth and link queuing delay buildup are modeled as independent.
+//  * The measurement (`z`) is the (scalar) frame delay variation [ms].
+//  * The observation matrix (`H`) is a 1x2 vector set as
+//    `{frame_size_variation [bytes], 1.0}`.
+//  * The state estimate covariance (`P`) is a symmetric 2x2 matrix.
+//  * The process noise covariance (`Q`) is a constant 2x2 diagonal matrix
+//    [(1 / bytes per ms)^2, ms^2].
+//  * The observation noise covariance (`r`) is a scalar [ms^2] that is
+//    determined externally to this class.
+class FrameDelayVariationKalmanFilter {
+ public:
+  FrameDelayVariationKalmanFilter();
+  ~FrameDelayVariationKalmanFilter() = default;
+
+  // Predicts and updates the filter, given a new pair of frame delay variation
+  // and frame size variation.
+  //
+  // Inputs:
+  // `frame_delay_variation_ms`:
+  //    Frame delay variation as calculated by the `InterFrameDelay` estimator.
+  //
+  // `frame_size_variation_bytes`:
+  //    Frame size variation, i.e., the current frame size minus the previous
+  //    frame size (in bytes). Note that this quantity may be negative.
+  //
+  // `max_frame_size_bytes`:
+  //    Filtered largest frame size received since the last reset.
+  //
+  // `var_noise`:
+  //    Variance of the estimated random jitter.
+  //
+  // TODO(bugs.webrtc.org/14381): For now use doubles as input parameters as
+  // units defined in api/units have insufficient underlying precision for
+  // jitter estimation.
+  void PredictAndUpdate(double frame_delay_variation_ms,
+                        double frame_size_variation_bytes,
+                        double max_frame_size_bytes,
+                        double var_noise);
+
+  // Given a frame size variation, returns the estimated frame delay variation
+  // explained by the link bandwidth alone.
+  double GetFrameDelayVariationEstimateSizeBased(
+      double frame_size_variation_bytes) const;
+
+  // Given a frame size variation, returns the estimated frame delay variation
+  // explained by both link bandwidth and link queuing delay buildup.
+  double GetFrameDelayVariationEstimateTotal(
+      double frame_size_variation_bytes) const;
+
+ private:
+  // State estimate (bandwidth [1 / bytes per ms], queue buildup [ms]).
+  double estimate_[2];
+  double estimate_cov_[2][2];  // Estimate covariance.
+
+  // Process noise covariance. This is a diagonal matrix, so we only store the
+  // diagonal entries.
+  double process_noise_cov_diag_[2];
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_TIMING_FRAME_DELAY_VARIATION_KALMAN_FILTER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_gn/moz.build
new file mode 100644
index 0000000000..ecdcfc93e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("frame_delay_variation_kalman_filter_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_unittest.cc b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_unittest.cc
new file mode 100644
index 0000000000..6103f3a1bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/frame_delay_variation_kalman_filter_unittest.cc
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/frame_delay_variation_kalman_filter.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+// This test verifies that the initial filter state (link bandwidth, link
+// propagation delay) is such that a frame of size zero would take no time to
+// propagate.
+TEST(FrameDelayVariationKalmanFilterTest,
+     InitializedFilterWithZeroSizeFrameTakesNoTimeToPropagate) {
+  FrameDelayVariationKalmanFilter filter;
+
+  // A zero-sized frame...
+  double frame_size_variation_bytes = 0.0;
+
+  // ...should take no time to propagate due to it's size...
+  EXPECT_EQ(filter.GetFrameDelayVariationEstimateSizeBased(
+                frame_size_variation_bytes),
+            0.0);
+
+  // ...and no time due to the initial link propagation delay being zero.
+  EXPECT_EQ(
+      filter.GetFrameDelayVariationEstimateTotal(frame_size_variation_bytes),
+      0.0);
+}
+
+// TODO(brandtr): Look into if there is a factor 1000 missing here? It seems
+// unreasonable to have an initial link bandwidth of 512 _mega_bits per second?
+TEST(FrameDelayVariationKalmanFilterTest,
+     InitializedFilterWithSmallSizeFrameTakesFixedTimeToPropagate) {
+  FrameDelayVariationKalmanFilter filter;
+
+  // A 1000-byte frame...
+  double frame_size_variation_bytes = 1000.0;
+  // ...should take around `1000.0 / (512e3 / 8.0) = 0.015625 ms` to transmit.
+  double expected_frame_delay_variation_estimate_ms = 1000.0 / (512e3 / 8.0);
+
+  EXPECT_EQ(filter.GetFrameDelayVariationEstimateSizeBased(
+                frame_size_variation_bytes),
+            expected_frame_delay_variation_estimate_ms);
+  EXPECT_EQ(
+      filter.GetFrameDelayVariationEstimateTotal(frame_size_variation_bytes),
+      expected_frame_delay_variation_estimate_ms);
+}
+
+TEST(FrameDelayVariationKalmanFilterTest,
+     NegativeNoiseVarianceDoesNotUpdateFilter) {
+  FrameDelayVariationKalmanFilter filter;
+
+  // Negative variance...
+  double var_noise = -0.1;
+  filter.PredictAndUpdate(/*frame_delay_variation_ms=*/3,
+                          /*frame_size_variation_bytes=*/200.0,
+                          /*max_frame_size_bytes=*/2000, var_noise);
+
+  // ...does _not_ update the filter.
+  EXPECT_EQ(filter.GetFrameDelayVariationEstimateTotal(
+                /*frame_size_variation_bytes=*/0.0),
+            0.0);
+
+  // Positive variance...
+  var_noise = 0.1;
+  filter.PredictAndUpdate(/*frame_delay_variation_ms=*/3,
+                          /*frame_size_variation_bytes=*/200.0,
+                          /*max_frame_size_bytes=*/2000, var_noise);
+
+  // ...does update the filter.
+  EXPECT_GT(filter.GetFrameDelayVariationEstimateTotal(
+                /*frame_size_variation_bytes=*/0.0),
+            0.0);
+}
+
+TEST(FrameDelayVariationKalmanFilterTest,
+     VerifyConvergenceWithAlternatingDeviations) {
+  FrameDelayVariationKalmanFilter filter;
+
+  // One frame every 33 ms.
+  int framerate_fps = 30;
+  // Let's assume approximately 10% delay variation.
+  double frame_delay_variation_ms = 3;
+  // With a bitrate of 512 kbps, each frame will be around 2000 bytes.
+  double max_frame_size_bytes = 2000;
+  // And again, let's assume 10% size deviation.
+  double frame_size_variation_bytes = 200;
+  double var_noise = 0.1;
+  int test_duration_s = 60;
+
+  for (int i = 0; i < test_duration_s * framerate_fps; ++i) {
+    // For simplicity, assume alternating variations.
+    double sign = (i % 2 == 0) ? 1.0 : -1.0;
+    filter.PredictAndUpdate(sign * frame_delay_variation_ms,
+                            sign * frame_size_variation_bytes,
+                            max_frame_size_bytes, var_noise);
+  }
+
+  // Verify that the filter has converged within a margin of 0.1 ms.
+  EXPECT_NEAR(
+      filter.GetFrameDelayVariationEstimateTotal(frame_size_variation_bytes),
+      frame_delay_variation_ms, 0.1);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.cc b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.cc
new file mode 100644
index 0000000000..bed9f875ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/inter_frame_delay.h"
+
+#include "absl/types/optional.h"
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "modules/include/module_common_types_public.h"
+
+namespace webrtc {
+
+namespace {
+constexpr Frequency k90kHz = Frequency::KiloHertz(90);
+}
+
+InterFrameDelay::InterFrameDelay() {
+  Reset();
+}
+
+// Resets the delay estimate.
+void InterFrameDelay::Reset() {
+  prev_wall_clock_ = absl::nullopt;
+  prev_rtp_timestamp_unwrapped_ = 0;
+}
+
+// Calculates the delay of a frame with the given timestamp.
+// This method is called when the frame is complete.
+absl::optional<TimeDelta> InterFrameDelay::CalculateDelay(
+    uint32_t rtp_timestamp,
+    Timestamp now) {
+  int64_t rtp_timestamp_unwrapped = unwrapper_.Unwrap(rtp_timestamp);
+  if (!prev_wall_clock_) {
+    // First set of data, initialization, wait for next frame.
+    prev_wall_clock_ = now;
+    prev_rtp_timestamp_unwrapped_ = rtp_timestamp_unwrapped;
+    return TimeDelta::Zero();
+  }
+
+  // Account for reordering in jitter variance estimate in the future?
+  // Note that this also captures incomplete frames which are grabbed for
+  // decoding after a later frame has been complete, i.e. real packet losses.
+  uint32_t cropped_last = static_cast<uint32_t>(prev_rtp_timestamp_unwrapped_);
+  if (rtp_timestamp_unwrapped < prev_rtp_timestamp_unwrapped_ ||
+      !IsNewerTimestamp(rtp_timestamp, cropped_last)) {
+    return absl::nullopt;
+  }
+
+  // Compute the compensated timestamp difference.
+  int64_t d_rtp_ticks = rtp_timestamp_unwrapped - prev_rtp_timestamp_unwrapped_;
+  TimeDelta dts = d_rtp_ticks / k90kHz;
+  TimeDelta dt = now - *prev_wall_clock_;
+
+  // frameDelay is the difference of dT and dTS -- i.e. the difference of the
+  // wall clock time difference and the timestamp difference between two
+  // following frames.
+  TimeDelta delay = dt - dts;
+
+  prev_rtp_timestamp_unwrapped_ = rtp_timestamp_unwrapped;
+  prev_wall_clock_ = now;
+  return delay;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.h b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.h
new file mode 100644
index 0000000000..03b5f78cc1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_TIMING_INTER_FRAME_DELAY_H_
+#define MODULES_VIDEO_CODING_TIMING_INTER_FRAME_DELAY_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+class InterFrameDelay {
+ public:
+  InterFrameDelay();
+
+  // Resets the estimate. Zeros are given as parameters.
+  void Reset();
+
+  // Calculates the delay of a frame with the given timestamp.
+  // This method is called when the frame is complete.
+  absl::optional<TimeDelta> CalculateDelay(uint32_t rtp_timestamp,
+                                           Timestamp now);
+
+ private:
+  // The previous rtp timestamp passed to the delay estimate
+  int64_t prev_rtp_timestamp_unwrapped_;
+  RtpTimestampUnwrapper unwrapper_;
+
+  // The previous wall clock timestamp used by the delay estimate
+  absl::optional<Timestamp> prev_wall_clock_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_TIMING_INTER_FRAME_DELAY_H_
diff --git a/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_gn/moz.build
new file mode 100644
index 0000000000..765d42aade
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("inter_frame_delay_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_unittest.cc b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_unittest.cc
new file mode 100644
index 0000000000..183b378ced
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay_unittest.cc
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/inter_frame_delay.h"
+
+#include <limits>
+
+#include "absl/types/optional.h"
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// Test is for frames at 30fps. At 30fps, RTP timestamps will increase by
+// 90000 / 30 = 3000 ticks per frame.
+constexpr Frequency k30Fps = Frequency::Hertz(30);
+constexpr TimeDelta kFrameDelay = 1 / k30Fps;
+constexpr uint32_t kRtpTicksPerFrame = Frequency::KiloHertz(90) / k30Fps;
+constexpr Timestamp kStartTime = Timestamp::Millis(1337);
+
+}  // namespace
+
+using ::testing::Eq;
+using ::testing::Optional;
+
+TEST(InterFrameDelayTest, OldRtpTimestamp) {
+  InterFrameDelay inter_frame_delay;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(180000, kStartTime),
+              Optional(TimeDelta::Zero()));
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(90000, kStartTime),
+              Eq(absl::nullopt));
+}
+
+TEST(InterFrameDelayTest, NegativeWrapAroundIsSameAsOldRtpTimestamp) {
+  InterFrameDelay inter_frame_delay;
+  uint32_t rtp = 1500;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, kStartTime),
+              Optional(TimeDelta::Zero()));
+  // RTP has wrapped around backwards.
+  rtp -= 3000;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, kStartTime),
+              Eq(absl::nullopt));
+}
+
+TEST(InterFrameDelayTest, CorrectDelayForFrames) {
+  InterFrameDelay inter_frame_delay;
+  // Use a fake clock to simplify time keeping.
+  SimulatedClock clock(kStartTime);
+
+  // First frame is always delay 0.
+  uint32_t rtp = 90000;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Zero()));
+
+  // Perfectly timed frame has 0 delay.
+  clock.AdvanceTime(kFrameDelay);
+  rtp += kRtpTicksPerFrame;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Zero()));
+
+  // Slightly early frame will have a negative delay.
+  clock.AdvanceTime(kFrameDelay - TimeDelta::Millis(3));
+  rtp += kRtpTicksPerFrame;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(-TimeDelta::Millis(3)));
+
+  // Slightly late frame will have positive delay.
+  clock.AdvanceTime(kFrameDelay + TimeDelta::Micros(5125));
+  rtp += kRtpTicksPerFrame;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Micros(5125)));
+
+  // Simulate faster frame RTP at the same clock delay. The frame arrives late,
+  // since the RTP timestamp is faster than the delay, and thus is positive.
+  clock.AdvanceTime(kFrameDelay);
+  rtp += kRtpTicksPerFrame / 2;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(kFrameDelay / 2.0));
+
+  // Simulate slower frame RTP at the same clock delay. The frame is early,
+  // since the RTP timestamp advanced more than the delay, and thus is negative.
+  clock.AdvanceTime(kFrameDelay);
+  rtp += 1.5 * kRtpTicksPerFrame;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(-kFrameDelay / 2.0));
+}
+
+TEST(InterFrameDelayTest, PositiveWrapAround) {
+  InterFrameDelay inter_frame_delay;
+  // Use a fake clock to simplify time keeping.
+  SimulatedClock clock(kStartTime);
+
+  // First frame is behind the max RTP by 1500.
+  uint32_t rtp = std::numeric_limits<uint32_t>::max() - 1500;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Zero()));
+
+  // Rtp wraps around, now 1499.
+  rtp += kRtpTicksPerFrame;
+
+  // Frame delay should be as normal, in this case simulated as 1ms late.
+  clock.AdvanceTime(kFrameDelay + TimeDelta::Millis(1));
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Millis(1)));
+}
+
+TEST(InterFrameDelayTest, MultipleWrapArounds) {
+  // Simulate a long pauses which cause wrap arounds multiple times.
+  constexpr Frequency k90Khz = Frequency::KiloHertz(90);
+  constexpr uint32_t kHalfRtp = std::numeric_limits<uint32_t>::max() / 2;
+  constexpr TimeDelta kWrapAroundDelay = kHalfRtp / k90Khz;
+
+  InterFrameDelay inter_frame_delay;
+  // Use a fake clock to simplify time keeping.
+  SimulatedClock clock(kStartTime);
+  uint32_t rtp = 0;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Zero()));
+
+  rtp += kHalfRtp;
+  clock.AdvanceTime(kWrapAroundDelay);
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Zero()));
+  // 1st wrap around.
+  rtp += kHalfRtp + 1;
+  clock.AdvanceTime(kWrapAroundDelay + TimeDelta::Millis(1));
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Millis(1) - (1 / k90Khz)));
+
+  rtp += kHalfRtp;
+  clock.AdvanceTime(kWrapAroundDelay);
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Zero()));
+  // 2nd wrap arounds.
+  rtp += kHalfRtp + 1;
+  clock.AdvanceTime(kWrapAroundDelay - TimeDelta::Millis(1));
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(-TimeDelta::Millis(1) - (1 / k90Khz)));
+
+  // Ensure short delay (large RTP delay) between wrap-arounds has correct
+  // jitter.
+  rtp += kHalfRtp;
+  clock.AdvanceTime(TimeDelta::Millis(10));
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(-(kWrapAroundDelay - TimeDelta::Millis(10))));
+  // 3nd wrap arounds, this time with large RTP delay.
+  rtp += kHalfRtp + 1;
+  clock.AdvanceTime(TimeDelta::Millis(10));
+  EXPECT_THAT(
+      inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+      Optional(-(kWrapAroundDelay - TimeDelta::Millis(10) + (1 / k90Khz))));
+}
+
+TEST(InterFrameDelayTest, NegativeWrapAroundAfterPositiveWrapAround) {
+  InterFrameDelay inter_frame_delay;
+  // Use a fake clock to simplify time keeping.
+  SimulatedClock clock(kStartTime);
+  uint32_t rtp = std::numeric_limits<uint32_t>::max() - 1500;
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Zero()));
+
+  // Rtp wraps around, now 1499.
+  rtp += kRtpTicksPerFrame;
+  // Frame delay should be as normal, in this case simulated as 1ms late.
+  clock.AdvanceTime(kFrameDelay);
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Optional(TimeDelta::Zero()));
+
+  // Wrap back.
+  rtp -= kRtpTicksPerFrame;
+  // Frame delay should be as normal, in this case simulated as 1ms late.
+  clock.AdvanceTime(kFrameDelay);
+  EXPECT_THAT(inter_frame_delay.CalculateDelay(rtp, clock.CurrentTime()),
+              Eq(absl::nullopt));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator.cc b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator.cc
new file mode 100644
index 0000000000..62757787a1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator.cc
@@ -0,0 +1,476 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/jitter_estimator.h"
+
+#include <math.h>
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_size.h"
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/video_coding/timing/rtt_filter.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+
+// Number of frames to wait for before post processing estimate. Also used in
+// the frame rate estimator ramp-up.
+constexpr size_t kFrameProcessingStartupCount = 30;
+
+// Number of frames to wait for before enabling the frame size filters.
+constexpr size_t kFramesUntilSizeFiltering = 5;
+
+// Initial value for frame size filters.
+constexpr double kInitialAvgAndMaxFrameSizeBytes = 500.0;
+
+// Time constant for average frame size filter.
+constexpr double kPhi = 0.97;
+// Time constant for max frame size filter.
+constexpr double kPsi = 0.9999;
+// Default constants for percentile frame size filter.
+constexpr double kDefaultMaxFrameSizePercentile = 0.95;
+constexpr int kDefaultFrameSizeWindow = 30 * 10;
+
+// Outlier rejection constants.
+constexpr double kNumStdDevDelayClamp = 3.5;
+constexpr double kNumStdDevDelayOutlier = 15.0;
+constexpr double kNumStdDevSizeOutlier = 3.0;
+constexpr double kCongestionRejectionFactor = -0.25;
+
+// Rampup constant for deviation noise filters.
+constexpr size_t kAlphaCountMax = 400;
+
+// Noise threshold constants.
+// ~Less than 1% chance (look up in normal distribution table)...
+constexpr double kNoiseStdDevs = 2.33;
+// ...of getting 30 ms freezes
+constexpr double kNoiseStdDevOffset = 30.0;
+
+// Jitter estimate clamping limits.
+constexpr TimeDelta kMinJitterEstimate = TimeDelta::Millis(1);
+constexpr TimeDelta kMaxJitterEstimate = TimeDelta::Seconds(10);
+
+// A constant describing the delay from the jitter buffer to the delay on the
+// receiving side which is not accounted for by the jitter buffer nor the
+// decoding delay estimate.
+constexpr TimeDelta OPERATING_SYSTEM_JITTER = TimeDelta::Millis(10);
+
+// Time constant for reseting the NACK count.
+constexpr TimeDelta kNackCountTimeout = TimeDelta::Seconds(60);
+
+// RTT mult activation.
+constexpr size_t kNackLimit = 3;
+
+// Frame rate estimate clamping limit.
+constexpr Frequency kMaxFramerateEstimate = Frequency::Hertz(200);
+
+}  // namespace
+
+constexpr char JitterEstimator::Config::kFieldTrialsKey[];
+
+JitterEstimator::Config JitterEstimator::Config::ParseAndValidate(
+    absl::string_view field_trial) {
+  Config config;
+  config.Parser()->Parse(field_trial);
+
+  // The `MovingPercentileFilter` RTC_CHECKs on the validity of the
+  // percentile and window length, so we'd better validate the field trial
+  // provided values here.
+  if (config.max_frame_size_percentile) {
+    double original = *config.max_frame_size_percentile;
+    config.max_frame_size_percentile = std::min(std::max(0.0, original), 1.0);
+    if (config.max_frame_size_percentile != original) {
+      RTC_LOG(LS_ERROR) << "Skipping invalid max_frame_size_percentile="
+                        << original;
+    }
+  }
+  if (config.frame_size_window && config.frame_size_window < 1) {
+    RTC_LOG(LS_ERROR) << "Skipping invalid frame_size_window="
+                      << *config.frame_size_window;
+    config.frame_size_window = 1;
+  }
+
+  // General sanity checks.
+  if (config.num_stddev_delay_clamp && config.num_stddev_delay_clamp < 0.0) {
+    RTC_LOG(LS_ERROR) << "Skipping invalid num_stddev_delay_clamp="
+                      << *config.num_stddev_delay_clamp;
+    config.num_stddev_delay_clamp = 0.0;
+  }
+  if (config.num_stddev_delay_outlier &&
+      config.num_stddev_delay_outlier < 0.0) {
+    RTC_LOG(LS_ERROR) << "Skipping invalid num_stddev_delay_outlier="
+                      << *config.num_stddev_delay_outlier;
+    config.num_stddev_delay_outlier = 0.0;
+  }
+  if (config.num_stddev_size_outlier && config.num_stddev_size_outlier < 0.0) {
+    RTC_LOG(LS_ERROR) << "Skipping invalid num_stddev_size_outlier="
+                      << *config.num_stddev_size_outlier;
+    config.num_stddev_size_outlier = 0.0;
+  }
+
+  return config;
+}
+
+JitterEstimator::JitterEstimator(Clock* clock,
+                                 const FieldTrialsView& field_trials)
+    : config_(Config::ParseAndValidate(
+          field_trials.Lookup(Config::kFieldTrialsKey))),
+      avg_frame_size_median_bytes_(static_cast<size_t>(
+          config_.frame_size_window.value_or(kDefaultFrameSizeWindow))),
+      max_frame_size_bytes_percentile_(
+          config_.max_frame_size_percentile.value_or(
+              kDefaultMaxFrameSizePercentile),
+          static_cast<size_t>(
+              config_.frame_size_window.value_or(kDefaultFrameSizeWindow))),
+      fps_counter_(30),  // TODO(sprang): Use an estimator with limit based
+                         // on time, rather than number of samples.
+      clock_(clock) {
+  Reset();
+}
+
+JitterEstimator::~JitterEstimator() = default;
+
+// Resets the JitterEstimate.
+void JitterEstimator::Reset() {
+  avg_frame_size_bytes_ = kInitialAvgAndMaxFrameSizeBytes;
+  max_frame_size_bytes_ = kInitialAvgAndMaxFrameSizeBytes;
+  var_frame_size_bytes2_ = 100;
+  avg_frame_size_median_bytes_.Reset();
+  max_frame_size_bytes_percentile_.Reset();
+  last_update_time_ = absl::nullopt;
+  prev_estimate_ = absl::nullopt;
+  prev_frame_size_ = absl::nullopt;
+  avg_noise_ms_ = 0.0;
+  var_noise_ms2_ = 4.0;
+  alpha_count_ = 1;
+  filter_jitter_estimate_ = TimeDelta::Zero();
+  latest_nack_ = Timestamp::Zero();
+  nack_count_ = 0;
+  startup_frame_size_sum_bytes_ = 0;
+  startup_frame_size_count_ = 0;
+  startup_count_ = 0;
+  rtt_filter_.Reset();
+  fps_counter_.Reset();
+
+  kalman_filter_ = FrameDelayVariationKalmanFilter();
+}
+
+// Updates the estimates with the new measurements.
+void JitterEstimator::UpdateEstimate(TimeDelta frame_delay,
+                                     DataSize frame_size) {
+  if (frame_size.IsZero()) {
+    return;
+  }
+  // Can't use DataSize since this can be negative.
+  double delta_frame_bytes =
+      frame_size.bytes() - prev_frame_size_.value_or(DataSize::Zero()).bytes();
+  if (startup_frame_size_count_ < kFramesUntilSizeFiltering) {
+    startup_frame_size_sum_bytes_ += frame_size.bytes();
+    startup_frame_size_count_++;
+  } else if (startup_frame_size_count_ == kFramesUntilSizeFiltering) {
+    // Give the frame size filter.
+    avg_frame_size_bytes_ = startup_frame_size_sum_bytes_ /
+                            static_cast<double>(startup_frame_size_count_);
+    startup_frame_size_count_++;
+  }
+
+  double avg_frame_size_bytes =
+      kPhi * avg_frame_size_bytes_ + (1 - kPhi) * frame_size.bytes();
+  double deviation_size_bytes = 2 * sqrt(var_frame_size_bytes2_);
+  if (frame_size.bytes() < avg_frame_size_bytes_ + deviation_size_bytes) {
+    // Only update the average frame size if this sample wasn't a key frame.
+    avg_frame_size_bytes_ = avg_frame_size_bytes;
+  }
+
+  double delta_bytes = frame_size.bytes() - avg_frame_size_bytes;
+  var_frame_size_bytes2_ = std::max(
+      kPhi * var_frame_size_bytes2_ + (1 - kPhi) * (delta_bytes * delta_bytes),
+      1.0);
+
+  // Update non-linear IIR estimate of max frame size.
+  max_frame_size_bytes_ =
+      std::max<double>(kPsi * max_frame_size_bytes_, frame_size.bytes());
+
+  // Maybe update percentile estimates of frame sizes.
+  if (config_.avg_frame_size_median) {
+    avg_frame_size_median_bytes_.Insert(frame_size.bytes());
+  }
+  if (config_.MaxFrameSizePercentileEnabled()) {
+    max_frame_size_bytes_percentile_.Insert(frame_size.bytes());
+  }
+
+  if (!prev_frame_size_) {
+    prev_frame_size_ = frame_size;
+    return;
+  }
+  prev_frame_size_ = frame_size;
+
+  // Cap frame_delay based on the current time deviation noise.
+  double num_stddev_delay_clamp =
+      config_.num_stddev_delay_clamp.value_or(kNumStdDevDelayClamp);
+  TimeDelta max_time_deviation =
+      TimeDelta::Millis(num_stddev_delay_clamp * sqrt(var_noise_ms2_) + 0.5);
+  frame_delay.Clamp(-max_time_deviation, max_time_deviation);
+
+  double delay_deviation_ms =
+      frame_delay.ms() -
+      kalman_filter_.GetFrameDelayVariationEstimateTotal(delta_frame_bytes);
+
+  // Outlier rejection: these conditions depend on filtered versions of the
+  // delay and frame size _means_, respectively, together with a configurable
+  // number of standard deviations. If a sample is large with respect to the
+  // corresponding mean and dispersion (defined by the number of
+  // standard deviations and the sample standard deviation), it is deemed an
+  // outlier. This "empirical rule" is further described in
+  // https://en.wikipedia.org/wiki/68-95-99.7_rule. Note that neither of the
+  // estimated means are true sample means, which implies that they are possibly
+  // not normally distributed. Hence, this rejection method is just a heuristic.
+  double num_stddev_delay_outlier =
+      config_.num_stddev_delay_outlier.value_or(kNumStdDevDelayOutlier);
+  // Delay outlier rejection is two-sided.
+  bool abs_delay_is_not_outlier =
+      fabs(delay_deviation_ms) <
+      num_stddev_delay_outlier * sqrt(var_noise_ms2_);
+  // The reasoning above means, in particular, that we should use the sample
+  // mean-style `avg_frame_size_bytes_` estimate, as opposed to the
+  // median-filtered version, even if configured to use latter for the
+  // calculation in `CalculateEstimate()`.
+  // Size outlier rejection is one-sided.
+  double num_stddev_size_outlier =
+      config_.num_stddev_size_outlier.value_or(kNumStdDevSizeOutlier);
+  bool size_is_positive_outlier =
+      frame_size.bytes() >
+      avg_frame_size_bytes_ +
+          num_stddev_size_outlier * sqrt(var_frame_size_bytes2_);
+
+  // Only update the Kalman filter if the sample is not considered an extreme
+  // outlier. Even if it is an extreme outlier from a delay point of view, if
+  // the frame size also is large the deviation is probably due to an incorrect
+  // line slope.
+  if (abs_delay_is_not_outlier || size_is_positive_outlier) {
+    // Prevent updating with frames which have been congested by a large frame,
+    // and therefore arrives almost at the same time as that frame.
+    // This can occur when we receive a large frame (key frame) which has been
+    // delayed. The next frame is of normal size (delta frame), and thus deltaFS
+    // will be << 0. This removes all frame samples which arrives after a key
+    // frame.
+    double congestion_rejection_factor =
+        config_.congestion_rejection_factor.value_or(
+            kCongestionRejectionFactor);
+    double filtered_max_frame_size_bytes =
+        config_.MaxFrameSizePercentileEnabled()
+            ? max_frame_size_bytes_percentile_.GetFilteredValue()
+            : max_frame_size_bytes_;
+    bool is_not_congested =
+        delta_frame_bytes >
+        congestion_rejection_factor * filtered_max_frame_size_bytes;
+
+    if (is_not_congested || config_.estimate_noise_when_congested) {
+      // Update the variance of the deviation from the line given by the Kalman
+      // filter.
+      EstimateRandomJitter(delay_deviation_ms);
+    }
+    if (is_not_congested) {
+      // Neither a delay outlier nor a congested frame, so we can safely update
+      // the Kalman filter with the sample.
+      kalman_filter_.PredictAndUpdate(frame_delay.ms(), delta_frame_bytes,
+                                      filtered_max_frame_size_bytes,
+                                      var_noise_ms2_);
+    }
+  } else {
+    // Delay outliers affect the noise estimate through a value equal to the
+    // outlier rejection threshold.
+    double num_stddev = (delay_deviation_ms >= 0) ? num_stddev_delay_outlier
+                                                  : -num_stddev_delay_outlier;
+    EstimateRandomJitter(num_stddev * sqrt(var_noise_ms2_));
+  }
+  // Post process the total estimated jitter
+  if (startup_count_ >= kFrameProcessingStartupCount) {
+    PostProcessEstimate();
+  } else {
+    startup_count_++;
+  }
+}
+
+// Updates the nack/packet ratio.
+void JitterEstimator::FrameNacked() {
+  if (nack_count_ < kNackLimit) {
+    nack_count_++;
+  }
+  latest_nack_ = clock_->CurrentTime();
+}
+
+void JitterEstimator::UpdateRtt(TimeDelta rtt) {
+  rtt_filter_.Update(rtt);
+}
+
+JitterEstimator::Config JitterEstimator::GetConfigForTest() const {
+  return config_;
+}
+
+// Estimates the random jitter by calculating the variance of the sample
+// distance from the line given by the Kalman filter.
+void JitterEstimator::EstimateRandomJitter(double d_dT) {
+  Timestamp now = clock_->CurrentTime();
+  if (last_update_time_.has_value()) {
+    fps_counter_.AddSample((now - *last_update_time_).us());
+  }
+  last_update_time_ = now;
+
+  if (alpha_count_ == 0) {
+    RTC_DCHECK_NOTREACHED();
+    return;
+  }
+  double alpha =
+      static_cast<double>(alpha_count_ - 1) / static_cast<double>(alpha_count_);
+  alpha_count_++;
+  if (alpha_count_ > kAlphaCountMax)
+    alpha_count_ = kAlphaCountMax;
+
+  // In order to avoid a low frame rate stream to react slower to changes,
+  // scale the alpha weight relative a 30 fps stream.
+  Frequency fps = GetFrameRate();
+  if (fps > Frequency::Zero()) {
+    constexpr Frequency k30Fps = Frequency::Hertz(30);
+    double rate_scale = k30Fps / fps;
+    // At startup, there can be a lot of noise in the fps estimate.
+    // Interpolate rate_scale linearly, from 1.0 at sample #1, to 30.0 / fps
+    // at sample #kFrameProcessingStartupCount.
+    if (alpha_count_ < kFrameProcessingStartupCount) {
+      rate_scale = (alpha_count_ * rate_scale +
+                    (kFrameProcessingStartupCount - alpha_count_)) /
+                   kFrameProcessingStartupCount;
+    }
+    alpha = pow(alpha, rate_scale);
+  }
+
+  double avg_noise_ms = alpha * avg_noise_ms_ + (1 - alpha) * d_dT;
+  double var_noise_ms2 = alpha * var_noise_ms2_ + (1 - alpha) *
+                                                      (d_dT - avg_noise_ms_) *
+                                                      (d_dT - avg_noise_ms_);
+  avg_noise_ms_ = avg_noise_ms;
+  var_noise_ms2_ = var_noise_ms2;
+  if (var_noise_ms2_ < 1.0) {
+    // The variance should never be zero, since we might get stuck and consider
+    // all samples as outliers.
+    var_noise_ms2_ = 1.0;
+  }
+}
+
+double JitterEstimator::NoiseThreshold() const {
+  double noise_threshold_ms =
+      kNoiseStdDevs * sqrt(var_noise_ms2_) - kNoiseStdDevOffset;
+  if (noise_threshold_ms < 1.0) {
+    noise_threshold_ms = 1.0;
+  }
+  return noise_threshold_ms;
+}
+
+// Calculates the current jitter estimate from the filtered estimates.
+TimeDelta JitterEstimator::CalculateEstimate() {
+  // Using median- and percentile-filtered versions of the frame sizes may be
+  // more robust than using sample mean-style estimates.
+  double filtered_avg_frame_size_bytes =
+      config_.avg_frame_size_median
+          ? avg_frame_size_median_bytes_.GetFilteredValue()
+          : avg_frame_size_bytes_;
+  double filtered_max_frame_size_bytes =
+      config_.MaxFrameSizePercentileEnabled()
+          ? max_frame_size_bytes_percentile_.GetFilteredValue()
+          : max_frame_size_bytes_;
+  double worst_case_frame_size_deviation_bytes =
+      filtered_max_frame_size_bytes - filtered_avg_frame_size_bytes;
+  double ret_ms = kalman_filter_.GetFrameDelayVariationEstimateSizeBased(
+                      worst_case_frame_size_deviation_bytes) +
+                  NoiseThreshold();
+  TimeDelta ret = TimeDelta::Millis(ret_ms);
+
+  // A very low estimate (or negative) is neglected.
+  if (ret < kMinJitterEstimate) {
+    ret = prev_estimate_.value_or(kMinJitterEstimate);
+    // Sanity check to make sure that no other method has set `prev_estimate_`
+    // to a value lower than `kMinJitterEstimate`.
+    RTC_DCHECK_GE(ret, kMinJitterEstimate);
+  } else if (ret > kMaxJitterEstimate) {  // Sanity
+    ret = kMaxJitterEstimate;
+  }
+  prev_estimate_ = ret;
+  return ret;
+}
+
+void JitterEstimator::PostProcessEstimate() {
+  filter_jitter_estimate_ = CalculateEstimate();
+}
+
+// Returns the current filtered estimate if available,
+// otherwise tries to calculate an estimate.
+TimeDelta JitterEstimator::GetJitterEstimate(
+    double rtt_multiplier,
+    absl::optional<TimeDelta> rtt_mult_add_cap) {
+  TimeDelta jitter = CalculateEstimate() + OPERATING_SYSTEM_JITTER;
+  Timestamp now = clock_->CurrentTime();
+
+  if (now - latest_nack_ > kNackCountTimeout)
+    nack_count_ = 0;
+
+  if (filter_jitter_estimate_ > jitter)
+    jitter = filter_jitter_estimate_;
+  if (nack_count_ >= kNackLimit) {
+    if (rtt_mult_add_cap.has_value()) {
+      jitter += std::min(rtt_filter_.Rtt() * rtt_multiplier,
+                         rtt_mult_add_cap.value());
+    } else {
+      jitter += rtt_filter_.Rtt() * rtt_multiplier;
+    }
+  }
+
+  static const Frequency kJitterScaleLowThreshold = Frequency::Hertz(5);
+  static const Frequency kJitterScaleHighThreshold = Frequency::Hertz(10);
+  Frequency fps = GetFrameRate();
+  // Ignore jitter for very low fps streams.
+  if (fps < kJitterScaleLowThreshold) {
+    if (fps.IsZero()) {
+      return std::max(TimeDelta::Zero(), jitter);
+    }
+    return TimeDelta::Zero();
+  }
+
+  // Semi-low frame rate; scale by factor linearly interpolated from 0.0 at
+  // kJitterScaleLowThreshold to 1.0 at kJitterScaleHighThreshold.
+  if (fps < kJitterScaleHighThreshold) {
+    jitter = (1.0 / (kJitterScaleHighThreshold - kJitterScaleLowThreshold)) *
+             (fps - kJitterScaleLowThreshold) * jitter;
+  }
+
+  return std::max(TimeDelta::Zero(), jitter);
+}
+
+Frequency JitterEstimator::GetFrameRate() const {
+  TimeDelta mean_frame_period = TimeDelta::Micros(fps_counter_.ComputeMean());
+  if (mean_frame_period <= TimeDelta::Zero())
+    return Frequency::Zero();
+
+  Frequency fps = 1 / mean_frame_period;
+  // Sanity check.
+  RTC_DCHECK_GE(fps, Frequency::Zero());
+  return std::min(fps, kMaxFramerateEstimate);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator.h b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator.h
new file mode 100644
index 0000000000..a89a4bf1fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator.h
@@ -0,0 +1,218 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_TIMING_JITTER_ESTIMATOR_H_
+#define MODULES_VIDEO_CODING_TIMING_JITTER_ESTIMATOR_H_
+
+#include <algorithm>
+#include <memory>
+#include <queue>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_size.h"
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/video_coding/timing/frame_delay_variation_kalman_filter.h"
+#include "modules/video_coding/timing/rtt_filter.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/numerics/moving_percentile_filter.h"
+#include "rtc_base/rolling_accumulator.h"
+
+namespace webrtc {
+
+class Clock;
+
+class JitterEstimator {
+ public:
+  // Configuration struct for statically overriding some constants and
+  // behaviour, configurable through field trials.
+  struct Config {
+    static constexpr char kFieldTrialsKey[] = "WebRTC-JitterEstimatorConfig";
+
+    // Parses a field trial string and validates the values.
+    static Config ParseAndValidate(absl::string_view field_trial);
+
+    std::unique_ptr<StructParametersParser> Parser() {
+      // clang-format off
+      return StructParametersParser::Create(
+          "avg_frame_size_median", &avg_frame_size_median,
+          "max_frame_size_percentile", &max_frame_size_percentile,
+          "frame_size_window", &frame_size_window,
+          "num_stddev_delay_clamp", &num_stddev_delay_clamp,
+          "num_stddev_delay_outlier", &num_stddev_delay_outlier,
+          "num_stddev_size_outlier", &num_stddev_size_outlier,
+          "congestion_rejection_factor", &congestion_rejection_factor,
+          "estimate_noise_when_congested", &estimate_noise_when_congested);
+      // clang-format on
+    }
+
+    bool MaxFrameSizePercentileEnabled() const {
+      return max_frame_size_percentile.has_value();
+    }
+
+    // If true, the "avg" frame size is calculated as the median over a window
+    // of recent frame sizes.
+    bool avg_frame_size_median = false;
+
+    // If set, the "max" frame size is calculated as this percentile over a
+    // window of recent frame sizes.
+    absl::optional<double> max_frame_size_percentile = absl::nullopt;
+
+    // The length of the percentile filters' window, in number of frames.
+    absl::optional<int> frame_size_window = absl::nullopt;
+
+    // The incoming frame delay variation samples are clamped to be at most
+    // this number of standard deviations away from zero.
+    //
+    // Increasing this value clamps fewer samples.
+    absl::optional<double> num_stddev_delay_clamp = absl::nullopt;
+
+    // A (relative) frame delay variation sample is an outlier if its absolute
+    // deviation from the Kalman filter model falls outside this number of
+    // sample standard deviations.
+    //
+    // Increasing this value rejects fewer samples.
+    absl::optional<double> num_stddev_delay_outlier = absl::nullopt;
+
+    // An (absolute) frame size sample is an outlier if its positive deviation
+    // from the estimated average frame size falls outside this number of sample
+    // standard deviations.
+    //
+    // Increasing this value rejects fewer samples.
+    absl::optional<double> num_stddev_size_outlier = absl::nullopt;
+
+    // A (relative) frame size variation sample is deemed "congested", and is
+    // thus rejected, if its value is less than this factor times the estimated
+    // max frame size.
+    //
+    // Decreasing this value rejects fewer samples.
+    absl::optional<double> congestion_rejection_factor = absl::nullopt;
+
+    // If true, the noise estimate will be updated for congestion rejected
+    // frames. This is currently enabled by default, but that may not be optimal
+    // since congested frames typically are not spread around the line with
+    // Gaussian noise. (This is the whole reason for the congestion rejection!)
+    bool estimate_noise_when_congested = true;
+  };
+
+  JitterEstimator(Clock* clock, const FieldTrialsView& field_trials);
+  JitterEstimator(const JitterEstimator&) = delete;
+  JitterEstimator& operator=(const JitterEstimator&) = delete;
+  ~JitterEstimator();
+
+  // Resets the estimate to the initial state.
+  void Reset();
+
+  // Updates the jitter estimate with the new data.
+  //
+  // Input:
+  //          - frame_delay      : Delay-delta calculated by UTILDelayEstimate.
+  //          - frame_size       : Frame size of the current frame.
+  void UpdateEstimate(TimeDelta frame_delay, DataSize frame_size);
+
+  // Returns the current jitter estimate and adds an RTT dependent term in cases
+  // of retransmission.
+  //  Input:
+  //          - rtt_multiplier   : RTT param multiplier (when applicable).
+  //          - rtt_mult_add_cap : Multiplier cap from the RTTMultExperiment.
+  //
+  // Return value              : Jitter estimate.
+  TimeDelta GetJitterEstimate(double rtt_multiplier,
+                              absl::optional<TimeDelta> rtt_mult_add_cap);
+
+  // Updates the nack counter.
+  void FrameNacked();
+
+  // Updates the RTT filter.
+  //
+  // Input:
+  //          - rtt          : Round trip time.
+  void UpdateRtt(TimeDelta rtt);
+
+  // Returns the configuration. Only to be used by unit tests.
+  Config GetConfigForTest() const;
+
+ private:
+  // Updates the random jitter estimate, i.e. the variance of the time
+  // deviations from the line given by the Kalman filter.
+  //
+  // Input:
+  //          - d_dT              : The deviation from the kalman estimate.
+  void EstimateRandomJitter(double d_dT);
+
+  double NoiseThreshold() const;
+
+  // Calculates the current jitter estimate.
+  //
+  // Return value                 : The current jitter estimate.
+  TimeDelta CalculateEstimate();
+
+  // Post process the calculated estimate.
+  void PostProcessEstimate();
+
+  // Returns the estimated incoming frame rate.
+  Frequency GetFrameRate() const;
+
+  // Configuration that may override some internals.
+  const Config config_;
+
+  // Filters the {frame_delay_delta, frame_size_delta} measurements through
+  // a linear Kalman filter.
+  FrameDelayVariationKalmanFilter kalman_filter_;
+
+  // TODO(bugs.webrtc.org/14381): Update `avg_frame_size_bytes_` to DataSize
+  // when api/units have sufficient precision.
+  double avg_frame_size_bytes_;  // Average frame size
+  double var_frame_size_bytes2_;  // Frame size variance. Unit is bytes^2.
+  // Largest frame size received (descending with a factor kPsi).
+  // Used by default.
+  // TODO(bugs.webrtc.org/14381): Update `max_frame_size_bytes_` to DataSize
+  // when api/units have sufficient precision.
+  double max_frame_size_bytes_;
+  // Percentile frame sized received (over a window). Only used if configured.
+  MovingMedianFilter<int64_t> avg_frame_size_median_bytes_;
+  MovingPercentileFilter<int64_t> max_frame_size_bytes_percentile_;
+  // TODO(bugs.webrtc.org/14381): Update `startup_frame_size_sum_bytes_` to
+  // DataSize when api/units have sufficient precision.
+  double startup_frame_size_sum_bytes_;
+  size_t startup_frame_size_count_;
+
+  absl::optional<Timestamp> last_update_time_;
+  // The previously returned jitter estimate
+  absl::optional<TimeDelta> prev_estimate_;
+  // Frame size of the previous frame
+  absl::optional<DataSize> prev_frame_size_;
+  // Average of the random jitter. Unit is milliseconds.
+  double avg_noise_ms_;
+  // Variance of the time-deviation from the line. Unit is milliseconds^2.
+  double var_noise_ms2_;
+  size_t alpha_count_;
+  // The filtered sum of jitter estimates
+  TimeDelta filter_jitter_estimate_ = TimeDelta::Zero();
+
+  size_t startup_count_;
+  // Time when the latest nack was seen
+  Timestamp latest_nack_ = Timestamp::Zero();
+  // Keeps track of the number of nacks received, but never goes above
+  // kNackLimit.
+  size_t nack_count_;
+  RttFilter rtt_filter_;
+
+  // Tracks frame rates in microseconds.
+  rtc::RollingAccumulator<uint64_t> fps_counter_;
+  Clock* clock_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_TIMING_JITTER_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_gn/moz.build
new file mode 100644
index 0000000000..a0d6154d78
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("jitter_estimator_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_unittest.cc b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_unittest.cc
new file mode 100644
index 0000000000..8e0c01587f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/jitter_estimator_unittest.cc
@@ -0,0 +1,305 @@
+/*  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/jitter_estimator.h"
+
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/field_trials.h"
+#include "api/units/data_size.h"
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/numerics/histogram_percentile_counter.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+// Generates some simple test data in the form of a sawtooth wave.
+class ValueGenerator {
+ public:
+  explicit ValueGenerator(int32_t amplitude)
+      : amplitude_(amplitude), counter_(0) {}
+
+  virtual ~ValueGenerator() = default;
+
+  TimeDelta Delay() const {
+    return TimeDelta::Millis((counter_ % 11) - 5) * amplitude_;
+  }
+
+  DataSize FrameSize() const {
+    return DataSize::Bytes(1000 + Delay().ms() / 5);
+  }
+
+  void Advance() { ++counter_; }
+
+ private:
+  const int32_t amplitude_;
+  int64_t counter_;
+};
+
+class JitterEstimatorTest : public ::testing::Test {
+ protected:
+  explicit JitterEstimatorTest(const std::string& field_trials)
+      : fake_clock_(0),
+        field_trials_(FieldTrials::CreateNoGlobal(field_trials)),
+        estimator_(&fake_clock_, *field_trials_) {}
+  JitterEstimatorTest() : JitterEstimatorTest("") {}
+  virtual ~JitterEstimatorTest() {}
+
+  void Run(int duration_s, int framerate_fps, ValueGenerator& gen) {
+    TimeDelta tick = 1 / Frequency::Hertz(framerate_fps);
+    for (int i = 0; i < duration_s * framerate_fps; ++i) {
+      estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
+      fake_clock_.AdvanceTime(tick);
+      gen.Advance();
+    }
+  }
+
+  SimulatedClock fake_clock_;
+  std::unique_ptr<FieldTrials> field_trials_;
+  JitterEstimator estimator_;
+};
+
+TEST_F(JitterEstimatorTest, SteadyStateConvergence) {
+  ValueGenerator gen(10);
+  Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
+  EXPECT_EQ(estimator_.GetJitterEstimate(0, absl::nullopt).ms(), 54);
+}
+
+TEST_F(JitterEstimatorTest,
+       SizeOutlierIsNotRejectedAndIncreasesJitterEstimate) {
+  ValueGenerator gen(10);
+
+  // Steady state.
+  Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
+  TimeDelta steady_state_jitter =
+      estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // A single outlier frame size...
+  estimator_.UpdateEstimate(gen.Delay(), 10 * gen.FrameSize());
+  TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // ...changes the estimate.
+  EXPECT_GT(outlier_jitter.ms(), 1.25 * steady_state_jitter.ms());
+}
+
+TEST_F(JitterEstimatorTest, LowFramerateDisablesJitterEstimator) {
+  ValueGenerator gen(10);
+  // At 5 fps, we disable jitter delay altogether.
+  TimeDelta time_delta = 1 / Frequency::Hertz(5);
+  for (int i = 0; i < 60; ++i) {
+    estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
+    fake_clock_.AdvanceTime(time_delta);
+    if (i > 2)
+      EXPECT_EQ(estimator_.GetJitterEstimate(0, absl::nullopt),
+                TimeDelta::Zero());
+    gen.Advance();
+  }
+}
+
+TEST_F(JitterEstimatorTest, RttMultAddCap) {
+  std::vector<std::pair<TimeDelta, rtc::HistogramPercentileCounter>>
+      jitter_by_rtt_mult_cap;
+  jitter_by_rtt_mult_cap.emplace_back(
+      /*rtt_mult_add_cap=*/TimeDelta::Millis(10), /*long_tail_boundary=*/1000);
+  jitter_by_rtt_mult_cap.emplace_back(
+      /*rtt_mult_add_cap=*/TimeDelta::Millis(200), /*long_tail_boundary=*/1000);
+
+  for (auto& [rtt_mult_add_cap, jitter] : jitter_by_rtt_mult_cap) {
+    estimator_.Reset();
+
+    ValueGenerator gen(50);
+    TimeDelta time_delta = 1 / Frequency::Hertz(30);
+    constexpr TimeDelta kRtt = TimeDelta::Millis(250);
+    for (int i = 0; i < 100; ++i) {
+      estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
+      fake_clock_.AdvanceTime(time_delta);
+      estimator_.FrameNacked();
+      estimator_.UpdateRtt(kRtt);
+      jitter.Add(
+          estimator_.GetJitterEstimate(/*rtt_mult=*/1.0, rtt_mult_add_cap)
+              .ms());
+      gen.Advance();
+    }
+  }
+
+  // 200ms cap should result in at least 25% higher max compared to 10ms.
+  EXPECT_GT(*jitter_by_rtt_mult_cap[1].second.GetPercentile(1.0),
+            *jitter_by_rtt_mult_cap[0].second.GetPercentile(1.0) * 1.25);
+}
+
+// By default, the `JitterEstimator` is not robust against single large frames.
+TEST_F(JitterEstimatorTest, Single2xFrameSizeImpactsJitterEstimate) {
+  ValueGenerator gen(10);
+
+  // Steady state.
+  Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
+  TimeDelta steady_state_jitter =
+      estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // A single outlier frame size...
+  estimator_.UpdateEstimate(gen.Delay(), 2 * gen.FrameSize());
+  TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // ...impacts the estimate.
+  EXPECT_GT(outlier_jitter.ms(), steady_state_jitter.ms());
+}
+
+// Under the default config, congested frames are used when calculating the
+// noise variance, meaning that they will impact the final jitter estimate.
+TEST_F(JitterEstimatorTest, CongestedFrameImpactsJitterEstimate) {
+  ValueGenerator gen(10);
+
+  // Steady state.
+  Run(/*duration_s=*/10, /*framerate_fps=*/30, gen);
+  TimeDelta steady_state_jitter =
+      estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // Congested frame...
+  estimator_.UpdateEstimate(-10 * gen.Delay(), 0.1 * gen.FrameSize());
+  TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // ...impacts the estimate.
+  EXPECT_GT(outlier_jitter.ms(), steady_state_jitter.ms());
+}
+
+TEST_F(JitterEstimatorTest, EmptyFieldTrialsParsesToUnsetConfig) {
+  JitterEstimator::Config config = estimator_.GetConfigForTest();
+  EXPECT_FALSE(config.avg_frame_size_median);
+  EXPECT_FALSE(config.max_frame_size_percentile.has_value());
+  EXPECT_FALSE(config.frame_size_window.has_value());
+  EXPECT_FALSE(config.num_stddev_delay_clamp.has_value());
+  EXPECT_FALSE(config.num_stddev_delay_outlier.has_value());
+  EXPECT_FALSE(config.num_stddev_size_outlier.has_value());
+  EXPECT_FALSE(config.congestion_rejection_factor.has_value());
+  EXPECT_TRUE(config.estimate_noise_when_congested);
+}
+
+class FieldTrialsOverriddenJitterEstimatorTest : public JitterEstimatorTest {
+ protected:
+  FieldTrialsOverriddenJitterEstimatorTest()
+      : JitterEstimatorTest(
+            "WebRTC-JitterEstimatorConfig/"
+            "avg_frame_size_median:true,"
+            "max_frame_size_percentile:0.9,"
+            "frame_size_window:30,"
+            "num_stddev_delay_clamp:1.1,"
+            "num_stddev_delay_outlier:2,"
+            "num_stddev_size_outlier:3.1,"
+            "congestion_rejection_factor:-1.55,"
+            "estimate_noise_when_congested:false/") {}
+  ~FieldTrialsOverriddenJitterEstimatorTest() {}
+};
+
+TEST_F(FieldTrialsOverriddenJitterEstimatorTest, FieldTrialsParsesCorrectly) {
+  JitterEstimator::Config config = estimator_.GetConfigForTest();
+  EXPECT_TRUE(config.avg_frame_size_median);
+  EXPECT_EQ(*config.max_frame_size_percentile, 0.9);
+  EXPECT_EQ(*config.frame_size_window, 30);
+  EXPECT_EQ(*config.num_stddev_delay_clamp, 1.1);
+  EXPECT_EQ(*config.num_stddev_delay_outlier, 2.0);
+  EXPECT_EQ(*config.num_stddev_size_outlier, 3.1);
+  EXPECT_EQ(*config.congestion_rejection_factor, -1.55);
+  EXPECT_FALSE(config.estimate_noise_when_congested);
+}
+
+TEST_F(FieldTrialsOverriddenJitterEstimatorTest,
+       DelayOutlierIsRejectedAndMaintainsJitterEstimate) {
+  ValueGenerator gen(10);
+
+  // Steady state.
+  Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
+  TimeDelta steady_state_jitter =
+      estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // A single outlier frame size...
+  estimator_.UpdateEstimate(10 * gen.Delay(), gen.FrameSize());
+  TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // ...does not change the estimate.
+  EXPECT_EQ(outlier_jitter.ms(), steady_state_jitter.ms());
+}
+
+// The field trial is configured to be robust against the `(1 - 0.9) = 10%`
+// largest frames over a window of length `30`.
+TEST_F(FieldTrialsOverriddenJitterEstimatorTest,
+       Four2xFrameSizesImpactJitterEstimate) {
+  ValueGenerator gen(10);
+
+  // Steady state.
+  Run(/*duration_s=*/60, /*framerate_fps=*/30, gen);
+  TimeDelta steady_state_jitter =
+      estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // Three outlier frames do not impact the jitter estimate.
+  for (int i = 0; i < 3; ++i) {
+    estimator_.UpdateEstimate(gen.Delay(), 2 * gen.FrameSize());
+  }
+  TimeDelta outlier_jitter_3x = estimator_.GetJitterEstimate(0, absl::nullopt);
+  EXPECT_EQ(outlier_jitter_3x.ms(), steady_state_jitter.ms());
+
+  // Four outlier frames do impact the jitter estimate.
+  estimator_.UpdateEstimate(gen.Delay(), 2 * gen.FrameSize());
+  TimeDelta outlier_jitter_4x = estimator_.GetJitterEstimate(0, absl::nullopt);
+  EXPECT_GT(outlier_jitter_4x.ms(), outlier_jitter_3x.ms());
+}
+
+// When so configured, congested frames are NOT used when calculating the
+// noise variance, meaning that they will NOT impact the final jitter estimate.
+TEST_F(FieldTrialsOverriddenJitterEstimatorTest,
+       CongestedFrameDoesNotImpactJitterEstimate) {
+  ValueGenerator gen(10);
+
+  // Steady state.
+  Run(/*duration_s=*/10, /*framerate_fps=*/30, gen);
+  TimeDelta steady_state_jitter =
+      estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // Congested frame...
+  estimator_.UpdateEstimate(-10 * gen.Delay(), 0.1 * gen.FrameSize());
+  TimeDelta outlier_jitter = estimator_.GetJitterEstimate(0, absl::nullopt);
+
+  // ...does not impact the estimate.
+  EXPECT_EQ(outlier_jitter.ms(), steady_state_jitter.ms());
+}
+
+class MisconfiguredFieldTrialsJitterEstimatorTest : public JitterEstimatorTest {
+ protected:
+  MisconfiguredFieldTrialsJitterEstimatorTest()
+      : JitterEstimatorTest(
+            "WebRTC-JitterEstimatorConfig/"
+            "max_frame_size_percentile:-0.9,"
+            "frame_size_window:-1,"
+            "num_stddev_delay_clamp:-1.9,"
+            "num_stddev_delay_outlier:-2,"
+            "num_stddev_size_outlier:-23.1/") {}
+  ~MisconfiguredFieldTrialsJitterEstimatorTest() {}
+};
+
+TEST_F(MisconfiguredFieldTrialsJitterEstimatorTest, FieldTrialsAreValidated) {
+  JitterEstimator::Config config = estimator_.GetConfigForTest();
+  EXPECT_EQ(*config.max_frame_size_percentile, 0.0);
+  EXPECT_EQ(*config.frame_size_window, 1);
+  EXPECT_EQ(*config.num_stddev_delay_clamp, 0.0);
+  EXPECT_EQ(*config.num_stddev_delay_outlier, 0.0);
+  EXPECT_EQ(*config.num_stddev_size_outlier, 0.0);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/rtt_filter.cc b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter.cc
new file mode 100644
index 0000000000..6962224d61
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter.cc
@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/rtt_filter.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <algorithm>
+
+#include "absl/algorithm/container.h"
+#include "absl/container/inlined_vector.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr TimeDelta kMaxRtt = TimeDelta::Seconds(3);
+constexpr uint32_t kFilterFactorMax = 35;
+constexpr double kJumpStddev = 2.5;
+constexpr double kDriftStdDev = 3.5;
+
+}  // namespace
+
+RttFilter::RttFilter()
+    : avg_rtt_(TimeDelta::Zero()),
+      var_rtt_(0),
+      max_rtt_(TimeDelta::Zero()),
+      jump_buf_(kMaxDriftJumpCount, TimeDelta::Zero()),
+      drift_buf_(kMaxDriftJumpCount, TimeDelta::Zero()) {
+  Reset();
+}
+
+void RttFilter::Reset() {
+  got_non_zero_update_ = false;
+  avg_rtt_ = TimeDelta::Zero();
+  var_rtt_ = 0;
+  max_rtt_ = TimeDelta::Zero();
+  filt_fact_count_ = 1;
+  absl::c_fill(jump_buf_, TimeDelta::Zero());
+  absl::c_fill(drift_buf_, TimeDelta::Zero());
+}
+
+void RttFilter::Update(TimeDelta rtt) {
+  if (!got_non_zero_update_) {
+    if (rtt.IsZero()) {
+      return;
+    }
+    got_non_zero_update_ = true;
+  }
+
+  // Sanity check
+  if (rtt > kMaxRtt) {
+    rtt = kMaxRtt;
+  }
+
+  double filt_factor = 0;
+  if (filt_fact_count_ > 1) {
+    filt_factor = static_cast<double>(filt_fact_count_ - 1) / filt_fact_count_;
+  }
+  filt_fact_count_++;
+  if (filt_fact_count_ > kFilterFactorMax) {
+    // This prevents filt_factor from going above
+    // (_filt_fact_max - 1) / filt_fact_max_,
+    // e.g., filt_fact_max_ = 50 => filt_factor = 49/50 = 0.98
+    filt_fact_count_ = kFilterFactorMax;
+  }
+  TimeDelta old_avg = avg_rtt_;
+  int64_t old_var = var_rtt_;
+  avg_rtt_ = filt_factor * avg_rtt_ + (1 - filt_factor) * rtt;
+  int64_t delta_ms = (rtt - avg_rtt_).ms();
+  var_rtt_ = filt_factor * var_rtt_ + (1 - filt_factor) * (delta_ms * delta_ms);
+  max_rtt_ = std::max(rtt, max_rtt_);
+  if (!JumpDetection(rtt) || !DriftDetection(rtt)) {
+    // In some cases we don't want to update the statistics
+    avg_rtt_ = old_avg;
+    var_rtt_ = old_var;
+  }
+}
+
+bool RttFilter::JumpDetection(TimeDelta rtt) {
+  TimeDelta diff_from_avg = avg_rtt_ - rtt;
+  // Unit of var_rtt_ is ms^2.
+  TimeDelta jump_threshold = TimeDelta::Millis(kJumpStddev * sqrt(var_rtt_));
+  if (diff_from_avg.Abs() > jump_threshold) {
+    bool positive_diff = diff_from_avg >= TimeDelta::Zero();
+    if (!jump_buf_.empty() && positive_diff != last_jump_positive_) {
+      // Since the signs differ the samples currently
+      // in the buffer is useless as they represent a
+      // jump in a different direction.
+      jump_buf_.clear();
+    }
+    if (jump_buf_.size() < kMaxDriftJumpCount) {
+      // Update the buffer used for the short time statistics.
+      // The sign of the diff is used for updating the counter since
+      // we want to use the same buffer for keeping track of when
+      // the RTT jumps down and up.
+      jump_buf_.push_back(rtt);
+      last_jump_positive_ = positive_diff;
+    }
+    if (jump_buf_.size() >= kMaxDriftJumpCount) {
+      // Detected an RTT jump
+      ShortRttFilter(jump_buf_);
+      filt_fact_count_ = kMaxDriftJumpCount + 1;
+      jump_buf_.clear();
+    } else {
+      return false;
+    }
+  } else {
+    jump_buf_.clear();
+  }
+  return true;
+}
+
+bool RttFilter::DriftDetection(TimeDelta rtt) {
+  // Unit of sqrt of var_rtt_ is ms.
+  TimeDelta drift_threshold = TimeDelta::Millis(kDriftStdDev * sqrt(var_rtt_));
+  if (max_rtt_ - avg_rtt_ > drift_threshold) {
+    if (drift_buf_.size() < kMaxDriftJumpCount) {
+      // Update the buffer used for the short time statistics.
+      drift_buf_.push_back(rtt);
+    }
+    if (drift_buf_.size() >= kMaxDriftJumpCount) {
+      // Detected an RTT drift
+      ShortRttFilter(drift_buf_);
+      filt_fact_count_ = kMaxDriftJumpCount + 1;
+      drift_buf_.clear();
+    }
+  } else {
+    drift_buf_.clear();
+  }
+  return true;
+}
+
+void RttFilter::ShortRttFilter(const BufferList& buf) {
+  RTC_DCHECK_EQ(buf.size(), kMaxDriftJumpCount);
+  max_rtt_ = TimeDelta::Zero();
+  avg_rtt_ = TimeDelta::Zero();
+  for (const TimeDelta& rtt : buf) {
+    if (rtt > max_rtt_) {
+      max_rtt_ = rtt;
+    }
+    avg_rtt_ += rtt;
+  }
+  avg_rtt_ = avg_rtt_ / static_cast<double>(buf.size());
+}
+
+TimeDelta RttFilter::Rtt() const {
+  return max_rtt_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/rtt_filter.h b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter.h
new file mode 100644
index 0000000000..b8700b23ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_TIMING_RTT_FILTER_H_
+#define MODULES_VIDEO_CODING_TIMING_RTT_FILTER_H_
+
+#include <stdint.h>
+
+#include "absl/container/inlined_vector.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+class RttFilter {
+ public:
+  RttFilter();
+  RttFilter(const RttFilter&) = delete;
+  RttFilter& operator=(const RttFilter&) = delete;
+
+  // Resets the filter.
+  void Reset();
+  // Updates the filter with a new sample.
+  void Update(TimeDelta rtt);
+  // A getter function for the current RTT level.
+  TimeDelta Rtt() const;
+
+ private:
+  // The size of the drift and jump memory buffers
+  // and thus also the detection threshold for these
+  // detectors in number of samples.
+  static constexpr int kMaxDriftJumpCount = 5;
+  using BufferList = absl::InlinedVector<TimeDelta, kMaxDriftJumpCount>;
+
+  // Detects RTT jumps by comparing the difference between
+  // samples and average to the standard deviation.
+  // Returns true if the long time statistics should be updated
+  // and false otherwise
+  bool JumpDetection(TimeDelta rtt);
+
+  // Detects RTT drifts by comparing the difference between
+  // max and average to the standard deviation.
+  // Returns true if the long time statistics should be updated
+  // and false otherwise
+  bool DriftDetection(TimeDelta rtt);
+
+  // Computes the short time average and maximum of the vector buf.
+  void ShortRttFilter(const BufferList& buf);
+
+  bool got_non_zero_update_;
+  TimeDelta avg_rtt_;
+  // Variance units are TimeDelta^2. Store as ms^2.
+  int64_t var_rtt_;
+  TimeDelta max_rtt_;
+  uint32_t filt_fact_count_;
+  bool last_jump_positive_ = false;
+  BufferList jump_buf_;
+  BufferList drift_buf_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_TIMING_RTT_FILTER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_gn/moz.build
new file mode 100644
index 0000000000..488a49d9c2
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/timing/rtt_filter.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rtt_filter_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_unittest.cc b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_unittest.cc
new file mode 100644
index 0000000000..05502e6f5b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/rtt_filter_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/rtt_filter.h"
+
+#include "api/units/time_delta.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(RttFilterTest, RttIsCapped) {
+  RttFilter rtt_filter;
+  rtt_filter.Update(TimeDelta::Seconds(500));
+
+  EXPECT_EQ(rtt_filter.Rtt(), TimeDelta::Seconds(3));
+}
+
+// If the difference between samples is more than away 2.5 stddev from the mean
+// then this is considered a jump. After more than 5 data points at the new
+// level, the RTT is reset to the new level.
+TEST(RttFilterTest, PositiveJumpDetection) {
+  RttFilter rtt_filter;
+
+  rtt_filter.Update(TimeDelta::Millis(200));
+  rtt_filter.Update(TimeDelta::Millis(200));
+  rtt_filter.Update(TimeDelta::Millis(200));
+
+  // Trigger 5 jumps.
+  rtt_filter.Update(TimeDelta::Millis(1400));
+  rtt_filter.Update(TimeDelta::Millis(1500));
+  rtt_filter.Update(TimeDelta::Millis(1600));
+  rtt_filter.Update(TimeDelta::Millis(1600));
+
+  EXPECT_EQ(rtt_filter.Rtt(), TimeDelta::Millis(1600));
+
+  rtt_filter.Update(TimeDelta::Millis(1600));
+  EXPECT_EQ(rtt_filter.Rtt(), TimeDelta::Millis(1600));
+}
+
+TEST(RttFilterTest, NegativeJumpDetection) {
+  RttFilter rtt_filter;
+
+  for (int i = 0; i < 10; ++i)
+    rtt_filter.Update(TimeDelta::Millis(1500));
+
+  // Trigger 5 negative data points that jump rtt down.
+  rtt_filter.Update(TimeDelta::Millis(200));
+  rtt_filter.Update(TimeDelta::Millis(200));
+  rtt_filter.Update(TimeDelta::Millis(200));
+  rtt_filter.Update(TimeDelta::Millis(200));
+  // Before 5 data points at the new level, max RTT is still 1500.
+  EXPECT_EQ(rtt_filter.Rtt(), TimeDelta::Millis(1500));
+
+  rtt_filter.Update(TimeDelta::Millis(300));
+  EXPECT_EQ(rtt_filter.Rtt(), TimeDelta::Millis(300));
+}
+
+TEST(RttFilterTest, JumpsResetByDirectionShift) {
+  RttFilter rtt_filter;
+  for (int i = 0; i < 10; ++i)
+    rtt_filter.Update(TimeDelta::Millis(1500));
+
+  // Trigger 4 negative jumps, then a positive one. This resets the jump
+  // detection.
+  rtt_filter.Update(TimeDelta::Millis(200));
+  rtt_filter.Update(TimeDelta::Millis(200));
+  rtt_filter.Update(TimeDelta::Millis(200));
+  rtt_filter.Update(TimeDelta::Millis(200));
+  rtt_filter.Update(TimeDelta::Millis(2000));
+  EXPECT_EQ(rtt_filter.Rtt(), TimeDelta::Millis(2000));
+
+  rtt_filter.Update(TimeDelta::Millis(300));
+  EXPECT_EQ(rtt_filter.Rtt(), TimeDelta::Millis(2000));
+}
+
+// If the difference between the max and average is more than 3.5 stddevs away
+// then a drift is detected, and a short filter is applied to find a new max
+// rtt.
+TEST(RttFilterTest, DriftDetection) {
+  RttFilter rtt_filter;
+
+  // Descend RTT by 30ms and settle at 700ms RTT. A drift is detected after rtt
+  // of 700ms is reported around 50 times for these targets.
+  constexpr TimeDelta kStartRtt = TimeDelta::Millis(1000);
+  constexpr TimeDelta kDriftTarget = TimeDelta::Millis(700);
+  constexpr TimeDelta kDelta = TimeDelta::Millis(30);
+  for (TimeDelta rtt = kStartRtt; rtt >= kDriftTarget; rtt -= kDelta)
+    rtt_filter.Update(rtt);
+
+  EXPECT_EQ(rtt_filter.Rtt(), kStartRtt);
+
+  for (int i = 0; i < 50; ++i)
+    rtt_filter.Update(kDriftTarget);
+  EXPECT_EQ(rtt_filter.Rtt(), kDriftTarget);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator.cc b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator.cc
new file mode 100644
index 0000000000..dc62ac674a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator.cc
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/timestamp_extrapolator.h"
+
+#include <algorithm>
+
+#include "absl/types/optional.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr double kLambda = 1;
+constexpr uint32_t kStartUpFilterDelayInPackets = 2;
+constexpr double kAlarmThreshold = 60e3;
+// in timestamp ticks, i.e. 15 ms
+constexpr double kAccDrift = 6600;
+constexpr double kAccMaxError = 7000;
+constexpr double kP11 = 1e10;
+
+}  // namespace
+
+TimestampExtrapolator::TimestampExtrapolator(Timestamp start)
+    : start_(Timestamp::Zero()),
+      prev_(Timestamp::Zero()),
+      packet_count_(0),
+      detector_accumulator_pos_(0),
+      detector_accumulator_neg_(0) {
+  Reset(start);
+}
+
+void TimestampExtrapolator::Reset(Timestamp start) {
+  start_ = start;
+  prev_ = start_;
+  first_unwrapped_timestamp_ = absl::nullopt;
+  w_[0] = 90.0;
+  w_[1] = 0;
+  p_[0][0] = 1;
+  p_[1][1] = kP11;
+  p_[0][1] = p_[1][0] = 0;
+  unwrapper_ = RtpTimestampUnwrapper();
+  packet_count_ = 0;
+  detector_accumulator_pos_ = 0;
+  detector_accumulator_neg_ = 0;
+}
+
+void TimestampExtrapolator::Update(Timestamp now, uint32_t ts90khz) {
+  if (now - prev_ > TimeDelta::Seconds(10)) {
+    // Ten seconds without a complete frame.
+    // Reset the extrapolator
+    Reset(now);
+  } else {
+    prev_ = now;
+  }
+
+  // Remove offset to prevent badly scaled matrices
+  const TimeDelta offset = now - start_;
+  double t_ms = offset.ms();
+
+  int64_t unwrapped_ts90khz = unwrapper_.Unwrap(ts90khz);
+
+  if (!first_unwrapped_timestamp_) {
+    // Make an initial guess of the offset,
+    // should be almost correct since t_ms - start
+    // should about zero at this time.
+    w_[1] = -w_[0] * t_ms;
+    first_unwrapped_timestamp_ = unwrapped_ts90khz;
+  }
+
+  double residual =
+      (static_cast<double>(unwrapped_ts90khz) - *first_unwrapped_timestamp_) -
+      t_ms * w_[0] - w_[1];
+  if (DelayChangeDetection(residual) &&
+      packet_count_ >= kStartUpFilterDelayInPackets) {
+    // A sudden change of average network delay has been detected.
+    // Force the filter to adjust its offset parameter by changing
+    // the offset uncertainty. Don't do this during startup.
+    p_[1][1] = kP11;
+  }
+
+  if (prev_unwrapped_timestamp_ &&
+      unwrapped_ts90khz < prev_unwrapped_timestamp_) {
+    // Drop reordered frames.
+    return;
+  }
+
+  // T = [t(k) 1]';
+  // that = T'*w;
+  // K = P*T/(lambda + T'*P*T);
+  double K[2];
+  K[0] = p_[0][0] * t_ms + p_[0][1];
+  K[1] = p_[1][0] * t_ms + p_[1][1];
+  double TPT = kLambda + t_ms * K[0] + K[1];
+  K[0] /= TPT;
+  K[1] /= TPT;
+  // w = w + K*(ts(k) - that);
+  w_[0] = w_[0] + K[0] * residual;
+  w_[1] = w_[1] + K[1] * residual;
+  // P = 1/lambda*(P - K*T'*P);
+  double p00 =
+      1 / kLambda * (p_[0][0] - (K[0] * t_ms * p_[0][0] + K[0] * p_[1][0]));
+  double p01 =
+      1 / kLambda * (p_[0][1] - (K[0] * t_ms * p_[0][1] + K[0] * p_[1][1]));
+  p_[1][0] =
+      1 / kLambda * (p_[1][0] - (K[1] * t_ms * p_[0][0] + K[1] * p_[1][0]));
+  p_[1][1] =
+      1 / kLambda * (p_[1][1] - (K[1] * t_ms * p_[0][1] + K[1] * p_[1][1]));
+  p_[0][0] = p00;
+  p_[0][1] = p01;
+  prev_unwrapped_timestamp_ = unwrapped_ts90khz;
+  if (packet_count_ < kStartUpFilterDelayInPackets) {
+    packet_count_++;
+  }
+}
+
+absl::optional<Timestamp> TimestampExtrapolator::ExtrapolateLocalTime(
+    uint32_t timestamp90khz) const {
+  int64_t unwrapped_ts90khz = unwrapper_.PeekUnwrap(timestamp90khz);
+  RTC_DCHECK_GE(unwrapped_ts90khz, 0);
+
+  if (!first_unwrapped_timestamp_) {
+    return absl::nullopt;
+  } else if (packet_count_ < kStartUpFilterDelayInPackets) {
+    constexpr double kRtpTicksPerMs = 90;
+    TimeDelta diff = TimeDelta::Millis(
+        (unwrapped_ts90khz - *prev_unwrapped_timestamp_) / kRtpTicksPerMs);
+    if (diff.ms() < 0) {
+      RTC_DCHECK_GE(prev_.ms(), -diff.ms());
+    }
+    return prev_ + diff;
+  } else if (w_[0] < 1e-3) {
+    return start_;
+  } else {
+    double timestampDiff = unwrapped_ts90khz - *first_unwrapped_timestamp_;
+    auto diff_ms = static_cast<int64_t>((timestampDiff - w_[1]) / w_[0] + 0.5);
+    if (diff_ms < 0) {
+      RTC_DCHECK_GE(start_.ms(), -diff_ms);
+    }
+    return start_ + TimeDelta::Millis(diff_ms);
+  }
+}
+
+bool TimestampExtrapolator::DelayChangeDetection(double error) {
+  // CUSUM detection of sudden delay changes
+  error = (error > 0) ? std::min(error, kAccMaxError)
+                      : std::max(error, -kAccMaxError);
+  detector_accumulator_pos_ =
+      std::max(detector_accumulator_pos_ + error - kAccDrift, double{0});
+  detector_accumulator_neg_ =
+      std::min(detector_accumulator_neg_ + error + kAccDrift, double{0});
+  if (detector_accumulator_pos_ > kAlarmThreshold ||
+      detector_accumulator_neg_ < -kAlarmThreshold) {
+    // Alarm
+    detector_accumulator_pos_ = detector_accumulator_neg_ = 0;
+    return true;
+  }
+  return false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator.h b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator.h
new file mode 100644
index 0000000000..6a9763943e
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_TIMING_TIMESTAMP_EXTRAPOLATOR_H_
+#define MODULES_VIDEO_CODING_TIMING_TIMESTAMP_EXTRAPOLATOR_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+// Not thread safe.
+class TimestampExtrapolator {
+ public:
+  explicit TimestampExtrapolator(Timestamp start);
+  void Update(Timestamp now, uint32_t ts90khz);
+  absl::optional<Timestamp> ExtrapolateLocalTime(uint32_t timestamp90khz) const;
+  void Reset(Timestamp start);
+
+ private:
+  void CheckForWrapArounds(uint32_t ts90khz);
+  bool DelayChangeDetection(double error);
+
+  double w_[2];
+  double p_[2][2];
+  Timestamp start_;
+  Timestamp prev_;
+  absl::optional<int64_t> first_unwrapped_timestamp_;
+  RtpTimestampUnwrapper unwrapper_;
+  absl::optional<int64_t> prev_unwrapped_timestamp_;
+  uint32_t packet_count_;
+  double detector_accumulator_pos_;
+  double detector_accumulator_neg_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_TIMING_TIMESTAMP_EXTRAPOLATOR_H_
diff --git a/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_gn/moz.build
new file mode 100644
index 0000000000..4cd085dd09
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("timestamp_extrapolator_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_unittest.cc b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_unittest.cc
new file mode 100644
index 0000000000..0b5fd74a8e
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/timestamp_extrapolator_unittest.cc
@@ -0,0 +1,221 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/timestamp_extrapolator.h"
+
+#include <stdint.h>
+
+#include <limits>
+
+#include "absl/types/optional.h"
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+using ::testing::Eq;
+using ::testing::Optional;
+
+namespace {
+
+constexpr Frequency kRtpHz = Frequency::KiloHertz(90);
+constexpr Frequency k25Fps = Frequency::Hertz(25);
+constexpr TimeDelta k25FpsDelay = 1 / k25Fps;
+
+}  // namespace
+
+TEST(TimestampExtrapolatorTest, ExtrapolationOccursAfter2Packets) {
+  SimulatedClock clock(Timestamp::Millis(1337));
+  TimestampExtrapolator ts_extrapolator(clock.CurrentTime());
+
+  // No packets so no timestamp.
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(90000), Eq(absl::nullopt));
+
+  uint32_t rtp = 90000;
+  clock.AdvanceTime(k25FpsDelay);
+  // First result is a bit confusing since it is based off the "start" time,
+  // which is arbitrary.
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime()));
+
+  rtp += kRtpHz / k25Fps;
+  clock.AdvanceTime(k25FpsDelay);
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime()));
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp + 90000),
+              Optional(clock.CurrentTime() + TimeDelta::Seconds(1)));
+}
+
+TEST(TimestampExtrapolatorTest, ResetsAfter10SecondPause) {
+  SimulatedClock clock(Timestamp::Millis(1337));
+  TimestampExtrapolator ts_extrapolator(clock.CurrentTime());
+
+  uint32_t rtp = 90000;
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime()));
+
+  rtp += kRtpHz / k25Fps;
+  clock.AdvanceTime(k25FpsDelay);
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime()));
+
+  rtp += 10 * kRtpHz.hertz();
+  clock.AdvanceTime(TimeDelta::Seconds(10) + TimeDelta::Micros(1));
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime()));
+}
+
+TEST(TimestampExtrapolatorTest, TimestampExtrapolatesMultipleRtpWrapArounds) {
+  SimulatedClock clock(Timestamp::Millis(1337));
+  TimestampExtrapolator ts_extrapolator(clock.CurrentTime());
+
+  uint32_t rtp = std::numeric_limits<uint32_t>::max();
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime()));
+
+  // One overflow. Static cast to avoid undefined behaviour with +=.
+  rtp += static_cast<uint32_t>(kRtpHz / k25Fps);
+  clock.AdvanceTime(k25FpsDelay);
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime()));
+
+  // Assert that extrapolation works across the boundary as expected.
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp + 90000),
+              Optional(clock.CurrentTime() + TimeDelta::Seconds(1)));
+  // This is not quite 1s since the math always rounds up.
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp - 90000),
+              Optional(clock.CurrentTime() - TimeDelta::Millis(999)));
+
+  // In order to avoid a wrap arounds reset, add a packet every 10s until we
+  // overflow twice.
+  constexpr TimeDelta kRtpOverflowDelay =
+      std::numeric_limits<uint32_t>::max() / kRtpHz;
+  const Timestamp overflow_time = clock.CurrentTime() + kRtpOverflowDelay * 2;
+
+  while (clock.CurrentTime() < overflow_time) {
+    clock.AdvanceTime(TimeDelta::Seconds(10));
+    // Static-cast before += to avoid undefined behaviour of overflow.
+    rtp += static_cast<uint32_t>(kRtpHz * TimeDelta::Seconds(10));
+    ts_extrapolator.Update(clock.CurrentTime(), rtp);
+    EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+                Optional(clock.CurrentTime()));
+  }
+}
+
+TEST(TimestampExtrapolatorTest, NegativeRtpTimestampWrapAround) {
+  SimulatedClock clock(Timestamp::Millis(1337));
+  TimestampExtrapolator ts_extrapolator(clock.CurrentTime());
+  uint32_t rtp = 0;
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime()));
+  // Go backwards!
+  rtp -= kRtpHz.hertz();
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime() - TimeDelta::Seconds(1)));
+}
+
+TEST(TimestampExtrapolatorTest, Slow90KHzClock) {
+  // This simulates a slow camera, which produces frames at 24Hz instead of
+  // 25Hz. The extrapolator should be able to resolve this with enough data.
+  SimulatedClock clock(Timestamp::Millis(1337));
+  TimestampExtrapolator ts_extrapolator(clock.CurrentTime());
+
+  constexpr TimeDelta k24FpsDelay = 1 / Frequency::Hertz(24);
+  uint32_t rtp = 90000;
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+
+  // Slow camera will increment RTP at 25 FPS rate even though its producing at
+  // 24 FPS. After 25 frames the extrapolator should settle at this rate.
+  for (int i = 0; i < 25; ++i) {
+    rtp += kRtpHz / k25Fps;
+    clock.AdvanceTime(k24FpsDelay);
+    ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  }
+
+  // The camera would normally produce 25 frames in 90K ticks, but is slow
+  // so takes 1s + k24FpsDelay for 90K ticks.
+  constexpr Frequency kSlowRtpHz = 90000 / (25 * k24FpsDelay);
+  // The extrapolator will be predicting that time at millisecond precision.
+  auto ts = ts_extrapolator.ExtrapolateLocalTime(rtp + kSlowRtpHz.hertz());
+  ASSERT_TRUE(ts.has_value());
+  EXPECT_EQ(ts->ms(), clock.TimeInMilliseconds() + 1000);
+}
+
+TEST(TimestampExtrapolatorTest, Fast90KHzClock) {
+  // This simulates a fast camera, which produces frames at 26Hz instead of
+  // 25Hz. The extrapolator should be able to resolve this with enough data.
+  SimulatedClock clock(Timestamp::Millis(1337));
+  TimestampExtrapolator ts_extrapolator(clock.CurrentTime());
+
+  constexpr TimeDelta k26FpsDelay = 1 / Frequency::Hertz(26);
+  uint32_t rtp = 90000;
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+
+  // Fast camera will increment RTP at 25 FPS rate even though its producing at
+  // 26 FPS. After 25 frames the extrapolator should settle at this rate.
+  for (int i = 0; i < 25; ++i) {
+    rtp += kRtpHz / k25Fps;
+    clock.AdvanceTime(k26FpsDelay);
+    ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  }
+
+  // The camera would normally produce 25 frames in 90K ticks, but is slow
+  // so takes 1s + k24FpsDelay for 90K ticks.
+  constexpr Frequency kSlowRtpHz = 90000 / (25 * k26FpsDelay);
+  // The extrapolator will be predicting that time at millisecond precision.
+  auto ts = ts_extrapolator.ExtrapolateLocalTime(rtp + kSlowRtpHz.hertz());
+  ASSERT_TRUE(ts.has_value());
+  EXPECT_EQ(ts->ms(), clock.TimeInMilliseconds() + 1000);
+}
+
+TEST(TimestampExtrapolatorTest, TimestampJump) {
+  // This simulates a jump in RTP timestamp, which could occur if a camera was
+  // swapped for example.
+  SimulatedClock clock(Timestamp::Millis(1337));
+  TimestampExtrapolator ts_extrapolator(clock.CurrentTime());
+
+  uint32_t rtp = 90000;
+  clock.AdvanceTime(k25FpsDelay);
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  rtp += kRtpHz / k25Fps;
+  clock.AdvanceTime(k25FpsDelay);
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  rtp += kRtpHz / k25Fps;
+  clock.AdvanceTime(k25FpsDelay);
+  ts_extrapolator.Update(clock.CurrentTime(), rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp),
+              Optional(clock.CurrentTime()));
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(rtp + 90000),
+              Optional(clock.CurrentTime() + TimeDelta::Seconds(1)));
+
+  // Jump RTP.
+  uint32_t new_rtp = 1337 * 90000;
+  clock.AdvanceTime(k25FpsDelay);
+  ts_extrapolator.Update(clock.CurrentTime(), new_rtp);
+  new_rtp += kRtpHz / k25Fps;
+  clock.AdvanceTime(k25FpsDelay);
+  ts_extrapolator.Update(clock.CurrentTime(), new_rtp);
+  EXPECT_THAT(ts_extrapolator.ExtrapolateLocalTime(new_rtp),
+              Optional(clock.CurrentTime()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/timing.cc b/third_party/libwebrtc/modules/video_coding/timing/timing.cc
new file mode 100644
index 0000000000..0b61d5a35e
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/timing.cc
@@ -0,0 +1,297 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/timing.h"
+
+#include <algorithm>
+
+#include "api/units/time_delta.h"
+#include "modules/video_coding/timing/timestamp_extrapolator.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+
+// Default pacing that is used for the low-latency renderer path.
+constexpr TimeDelta kZeroPlayoutDelayDefaultMinPacing = TimeDelta::Millis(8);
+constexpr TimeDelta kLowLatencyStreamMaxPlayoutDelayThreshold =
+    TimeDelta::Millis(500);
+
+void CheckDelaysValid(TimeDelta min_delay, TimeDelta max_delay) {
+  if (min_delay > max_delay) {
+    RTC_LOG(LS_ERROR)
+        << "Playout delays set incorrectly: min playout delay (" << min_delay
+        << ") > max playout delay (" << max_delay
+        << "). This is undefined behaviour. Application writers should "
+           "ensure that the min delay is always less than or equals max "
+           "delay. If trying to use the playout delay header extensions "
+           "described in "
+           "https://webrtc.googlesource.com/src/+/refs/heads/main/docs/"
+           "native-code/rtp-hdrext/playout-delay/, be careful that a playout "
+           "delay hint or A/V sync settings may have caused this conflict.";
+  }
+}
+
+}  // namespace
+
+VCMTiming::VCMTiming(Clock* clock, const FieldTrialsView& field_trials)
+    : clock_(clock),
+      ts_extrapolator_(
+          std::make_unique<TimestampExtrapolator>(clock_->CurrentTime())),
+      codec_timer_(std::make_unique<CodecTimer>()),
+      render_delay_(kDefaultRenderDelay),
+      min_playout_delay_(TimeDelta::Zero()),
+      max_playout_delay_(TimeDelta::Seconds(10)),
+      jitter_delay_(TimeDelta::Zero()),
+      current_delay_(TimeDelta::Zero()),
+      prev_frame_timestamp_(0),
+      num_decoded_frames_(0),
+      zero_playout_delay_min_pacing_("min_pacing",
+                                     kZeroPlayoutDelayDefaultMinPacing),
+      last_decode_scheduled_(Timestamp::Zero()) {
+  ParseFieldTrial({&zero_playout_delay_min_pacing_},
+                  field_trials.Lookup("WebRTC-ZeroPlayoutDelay"));
+}
+
+void VCMTiming::Reset() {
+  MutexLock lock(&mutex_);
+  ts_extrapolator_->Reset(clock_->CurrentTime());
+  codec_timer_ = std::make_unique<CodecTimer>();
+  render_delay_ = kDefaultRenderDelay;
+  min_playout_delay_ = TimeDelta::Zero();
+  jitter_delay_ = TimeDelta::Zero();
+  current_delay_ = TimeDelta::Zero();
+  prev_frame_timestamp_ = 0;
+}
+
+void VCMTiming::set_render_delay(TimeDelta render_delay) {
+  MutexLock lock(&mutex_);
+  render_delay_ = render_delay;
+}
+
+TimeDelta VCMTiming::min_playout_delay() const {
+  MutexLock lock(&mutex_);
+  return min_playout_delay_;
+}
+
+void VCMTiming::set_min_playout_delay(TimeDelta min_playout_delay) {
+  MutexLock lock(&mutex_);
+  if (min_playout_delay_ != min_playout_delay) {
+    CheckDelaysValid(min_playout_delay, max_playout_delay_);
+    min_playout_delay_ = min_playout_delay;
+  }
+}
+
+void VCMTiming::set_max_playout_delay(TimeDelta max_playout_delay) {
+  MutexLock lock(&mutex_);
+  if (max_playout_delay_ != max_playout_delay) {
+    CheckDelaysValid(min_playout_delay_, max_playout_delay);
+    max_playout_delay_ = max_playout_delay;
+  }
+}
+
+void VCMTiming::SetJitterDelay(TimeDelta jitter_delay) {
+  MutexLock lock(&mutex_);
+  if (jitter_delay != jitter_delay_) {
+    jitter_delay_ = jitter_delay;
+    // When in initial state, set current delay to minimum delay.
+    if (current_delay_.IsZero()) {
+      current_delay_ = jitter_delay_;
+    }
+  }
+}
+
+void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
+  MutexLock lock(&mutex_);
+  TimeDelta target_delay = TargetDelayInternal();
+
+  if (current_delay_.IsZero()) {
+    // Not initialized, set current delay to target.
+    current_delay_ = target_delay;
+  } else if (target_delay != current_delay_) {
+    TimeDelta delay_diff = target_delay - current_delay_;
+    // Never change the delay with more than 100 ms every second. If we're
+    // changing the delay in too large steps we will get noticeable freezes. By
+    // limiting the change we can increase the delay in smaller steps, which
+    // will be experienced as the video is played in slow motion. When lowering
+    // the delay the video will be played at a faster pace.
+    TimeDelta max_change = TimeDelta::Zero();
+    if (frame_timestamp < 0x0000ffff && prev_frame_timestamp_ > 0xffff0000) {
+      // wrap
+      max_change =
+          TimeDelta::Millis(kDelayMaxChangeMsPerS *
+                            (frame_timestamp + (static_cast<int64_t>(1) << 32) -
+                             prev_frame_timestamp_) /
+                            90000);
+    } else {
+      max_change =
+          TimeDelta::Millis(kDelayMaxChangeMsPerS *
+                            (frame_timestamp - prev_frame_timestamp_) / 90000);
+    }
+
+    if (max_change <= TimeDelta::Zero()) {
+      // Any changes less than 1 ms are truncated and will be postponed.
+      // Negative change will be due to reordering and should be ignored.
+      return;
+    }
+    delay_diff = std::max(delay_diff, -max_change);
+    delay_diff = std::min(delay_diff, max_change);
+
+    current_delay_ = current_delay_ + delay_diff;
+  }
+  prev_frame_timestamp_ = frame_timestamp;
+}
+
+void VCMTiming::UpdateCurrentDelay(Timestamp render_time,
+                                   Timestamp actual_decode_time) {
+  MutexLock lock(&mutex_);
+  TimeDelta target_delay = TargetDelayInternal();
+  TimeDelta delayed =
+      (actual_decode_time - render_time) + RequiredDecodeTime() + render_delay_;
+
+  // Only consider `delayed` as negative by more than a few microseconds.
+  if (delayed.ms() < 0) {
+    return;
+  }
+  if (current_delay_ + delayed <= target_delay) {
+    current_delay_ += delayed;
+  } else {
+    current_delay_ = target_delay;
+  }
+}
+
+void VCMTiming::StopDecodeTimer(TimeDelta decode_time, Timestamp now) {
+  MutexLock lock(&mutex_);
+  codec_timer_->AddTiming(decode_time.ms(), now.ms());
+  RTC_DCHECK_GE(decode_time, TimeDelta::Zero());
+  ++num_decoded_frames_;
+}
+
+void VCMTiming::IncomingTimestamp(uint32_t rtp_timestamp, Timestamp now) {
+  MutexLock lock(&mutex_);
+  ts_extrapolator_->Update(now, rtp_timestamp);
+}
+
+Timestamp VCMTiming::RenderTime(uint32_t frame_timestamp, Timestamp now) const {
+  MutexLock lock(&mutex_);
+  return RenderTimeInternal(frame_timestamp, now);
+}
+
+void VCMTiming::SetLastDecodeScheduledTimestamp(
+    Timestamp last_decode_scheduled) {
+  MutexLock lock(&mutex_);
+  last_decode_scheduled_ = last_decode_scheduled;
+}
+
+Timestamp VCMTiming::RenderTimeInternal(uint32_t frame_timestamp,
+                                        Timestamp now) const {
+  if (UseLowLatencyRendering()) {
+    // Render as soon as possible or with low-latency renderer algorithm.
+    return Timestamp::Zero();
+  }
+  // Note that TimestampExtrapolator::ExtrapolateLocalTime is not a const
+  // method; it mutates the object's wraparound state.
+  Timestamp estimated_complete_time =
+      ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp).value_or(now);
+
+  // Make sure the actual delay stays in the range of `min_playout_delay_`
+  // and `max_playout_delay_`.
+  TimeDelta actual_delay =
+      current_delay_.Clamped(min_playout_delay_, max_playout_delay_);
+  return estimated_complete_time + actual_delay;
+}
+
+TimeDelta VCMTiming::RequiredDecodeTime() const {
+  const int decode_time_ms = codec_timer_->RequiredDecodeTimeMs();
+  RTC_DCHECK_GE(decode_time_ms, 0);
+  return TimeDelta::Millis(decode_time_ms);
+}
+
+TimeDelta VCMTiming::MaxWaitingTime(Timestamp render_time,
+                                    Timestamp now,
+                                    bool too_many_frames_queued) const {
+  MutexLock lock(&mutex_);
+
+  if (render_time.IsZero() && zero_playout_delay_min_pacing_->us() > 0 &&
+      min_playout_delay_.IsZero() && max_playout_delay_ > TimeDelta::Zero()) {
+    // `render_time` == 0 indicates that the frame should be decoded and
+    // rendered as soon as possible. However, the decoder can be choked if too
+    // many frames are sent at once. Therefore, limit the interframe delay to
+    // |zero_playout_delay_min_pacing_| unless too many frames are queued in
+    // which case the frames are sent to the decoder at once.
+    if (too_many_frames_queued) {
+      return TimeDelta::Zero();
+    }
+    Timestamp earliest_next_decode_start_time =
+        last_decode_scheduled_ + zero_playout_delay_min_pacing_;
+    TimeDelta max_wait_time = now >= earliest_next_decode_start_time
+                                  ? TimeDelta::Zero()
+                                  : earliest_next_decode_start_time - now;
+    return max_wait_time;
+  }
+  return render_time - now - RequiredDecodeTime() - render_delay_;
+}
+
+TimeDelta VCMTiming::TargetVideoDelay() const {
+  MutexLock lock(&mutex_);
+  return TargetDelayInternal();
+}
+
+TimeDelta VCMTiming::TargetDelayInternal() const {
+  return std::max(min_playout_delay_,
+                  jitter_delay_ + RequiredDecodeTime() + render_delay_);
+}
+
+VideoFrame::RenderParameters VCMTiming::RenderParameters() const {
+  MutexLock lock(&mutex_);
+  return {.use_low_latency_rendering = UseLowLatencyRendering(),
+          .max_composition_delay_in_frames = max_composition_delay_in_frames_};
+}
+
+bool VCMTiming::UseLowLatencyRendering() const {
+  // min_playout_delay_==0,
+  // max_playout_delay_<=kLowLatencyStreamMaxPlayoutDelayThreshold indicates
+  // that the low-latency path should be used, which means that frames should be
+  // decoded and rendered as soon as possible.
+  return min_playout_delay_.IsZero() &&
+         max_playout_delay_ <= kLowLatencyStreamMaxPlayoutDelayThreshold;
+}
+
+VCMTiming::VideoDelayTimings VCMTiming::GetTimings() const {
+  MutexLock lock(&mutex_);
+  return VideoDelayTimings{.max_decode_duration = RequiredDecodeTime(),
+                           .current_delay = current_delay_,
+                           .target_delay = TargetDelayInternal(),
+                           .jitter_buffer_delay = jitter_delay_,
+                           .min_playout_delay = min_playout_delay_,
+                           .max_playout_delay = max_playout_delay_,
+                           .render_delay = render_delay_,
+                           .num_decoded_frames = num_decoded_frames_};
+}
+
+void VCMTiming::SetTimingFrameInfo(const TimingFrameInfo& info) {
+  MutexLock lock(&mutex_);
+  timing_frame_info_.emplace(info);
+}
+
+absl::optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
+  MutexLock lock(&mutex_);
+  return timing_frame_info_;
+}
+
+void VCMTiming::SetMaxCompositionDelayInFrames(
+    absl::optional<int> max_composition_delay_in_frames) {
+  MutexLock lock(&mutex_);
+  max_composition_delay_in_frames_ = max_composition_delay_in_frames;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/timing/timing.h b/third_party/libwebrtc/modules/video_coding/timing/timing.h
new file mode 100644
index 0000000000..727527f009
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/timing.h
@@ -0,0 +1,160 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_TIMING_TIMING_H_
+#define MODULES_VIDEO_CODING_TIMING_TIMING_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/time_delta.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_timing.h"
+#include "modules/video_coding/timing/codec_timer.h"
+#include "modules/video_coding/timing/timestamp_extrapolator.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class VCMTiming {
+ public:
+  static constexpr auto kDefaultRenderDelay = TimeDelta::Millis(10);
+  static constexpr auto kDelayMaxChangeMsPerS = 100;
+
+  VCMTiming(Clock* clock, const FieldTrialsView& field_trials);
+  virtual ~VCMTiming() = default;
+
+  // Resets the timing to the initial state.
+  void Reset();
+
+  // Set the amount of time needed to render an image. Defaults to 10 ms.
+  void set_render_delay(TimeDelta render_delay);
+
+  // Set the minimum time the video must be delayed on the receiver to
+  // get the desired jitter buffer level.
+  void SetJitterDelay(TimeDelta required_delay);
+
+  // Set/get the minimum playout delay from capture to render.
+  TimeDelta min_playout_delay() const;
+  void set_min_playout_delay(TimeDelta min_playout_delay);
+
+  // Set/get the maximum playout delay from capture to render in ms.
+  void set_max_playout_delay(TimeDelta max_playout_delay);
+
+  // Increases or decreases the current delay to get closer to the target delay.
+  // Calculates how long it has been since the previous call to this function,
+  // and increases/decreases the delay in proportion to the time difference.
+  void UpdateCurrentDelay(uint32_t frame_timestamp);
+
+  // Increases or decreases the current delay to get closer to the target delay.
+  // Given the actual decode time in ms and the render time in ms for a frame,
+  // this function calculates how late the frame is and increases the delay
+  // accordingly.
+  void UpdateCurrentDelay(Timestamp render_time, Timestamp actual_decode_time);
+
+  // Stops the decoder timer, should be called when the decoder returns a frame
+  // or when the decoded frame callback is called.
+  void StopDecodeTimer(TimeDelta decode_time, Timestamp now);
+
+  // Used to report that a frame is passed to decoding. Updates the timestamp
+  // filter which is used to map between timestamps and receiver system time.
+  virtual void IncomingTimestamp(uint32_t rtp_timestamp,
+                                 Timestamp last_packet_time);
+
+  // Returns the receiver system time when the frame with timestamp
+  // `frame_timestamp` should be rendered, assuming that the system time
+  // currently is `now`.
+  virtual Timestamp RenderTime(uint32_t frame_timestamp, Timestamp now) const;
+
+  // Returns the maximum time in ms that we can wait for a frame to become
+  // complete before we must pass it to the decoder. render_time==0 indicates
+  // that the frames should be processed as quickly as possible, with possibly
+  // only a small delay added to make sure that the decoder is not overloaded.
+  // In this case, the parameter too_many_frames_queued is used to signal that
+  // the decode queue is full and that the frame should be decoded as soon as
+  // possible.
+  virtual TimeDelta MaxWaitingTime(Timestamp render_time,
+                                   Timestamp now,
+                                   bool too_many_frames_queued) const;
+
+  // Returns the current target delay which is required delay + decode time +
+  // render delay.
+  TimeDelta TargetVideoDelay() const;
+
+  // Return current timing information. Returns true if the first frame has been
+  // decoded, false otherwise.
+  struct VideoDelayTimings {
+    TimeDelta max_decode_duration;
+    TimeDelta current_delay;
+    TimeDelta target_delay;
+    TimeDelta jitter_buffer_delay;
+    TimeDelta min_playout_delay;
+    TimeDelta max_playout_delay;
+    TimeDelta render_delay;
+    size_t num_decoded_frames;
+  };
+  VideoDelayTimings GetTimings() const;
+
+  void SetTimingFrameInfo(const TimingFrameInfo& info);
+  absl::optional<TimingFrameInfo> GetTimingFrameInfo();
+
+  void SetMaxCompositionDelayInFrames(
+      absl::optional<int> max_composition_delay_in_frames);
+
+  VideoFrame::RenderParameters RenderParameters() const;
+
+  // Updates the last time a frame was scheduled for decoding.
+  void SetLastDecodeScheduledTimestamp(Timestamp last_decode_scheduled);
+
+ protected:
+  TimeDelta RequiredDecodeTime() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  Timestamp RenderTimeInternal(uint32_t frame_timestamp, Timestamp now) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  TimeDelta TargetDelayInternal() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  bool UseLowLatencyRendering() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+ private:
+  mutable Mutex mutex_;
+  Clock* const clock_;
+  const std::unique_ptr<TimestampExtrapolator> ts_extrapolator_
+      RTC_PT_GUARDED_BY(mutex_);
+  std::unique_ptr<CodecTimer> codec_timer_ RTC_GUARDED_BY(mutex_)
+      RTC_PT_GUARDED_BY(mutex_);
+  TimeDelta render_delay_ RTC_GUARDED_BY(mutex_);
+  // Best-effort playout delay range for frames from capture to render.
+  // The receiver tries to keep the delay between `min_playout_delay_ms_`
+  // and `max_playout_delay_ms_` taking the network jitter into account.
+  // A special case is where min_playout_delay_ms_ = max_playout_delay_ms_ = 0,
+  // in which case the receiver tries to play the frames as they arrive.
+  TimeDelta min_playout_delay_ RTC_GUARDED_BY(mutex_);
+  TimeDelta max_playout_delay_ RTC_GUARDED_BY(mutex_);
+  TimeDelta jitter_delay_ RTC_GUARDED_BY(mutex_);
+  TimeDelta current_delay_ RTC_GUARDED_BY(mutex_);
+  uint32_t prev_frame_timestamp_ RTC_GUARDED_BY(mutex_);
+  absl::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(mutex_);
+  size_t num_decoded_frames_ RTC_GUARDED_BY(mutex_);
+  absl::optional<int> max_composition_delay_in_frames_ RTC_GUARDED_BY(mutex_);
+  // Set by the field trial WebRTC-ZeroPlayoutDelay. The parameter min_pacing
+  // determines the minimum delay between frames scheduled for decoding that is
+  // used when min playout delay=0 and max playout delay>=0.
+  FieldTrialParameter<TimeDelta> zero_playout_delay_min_pacing_
+      RTC_GUARDED_BY(mutex_);
+  // Timestamp at which the last frame was scheduled to be sent to the decoder.
+  // Used only when the RTP header extension playout delay is set to min=0 ms
+  // which is indicated by a render time set to 0.
+  Timestamp last_decode_scheduled_ RTC_GUARDED_BY(mutex_);
+};
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_TIMING_TIMING_H_
diff --git a/third_party/libwebrtc/modules/video_coding/timing/timing_module_gn/moz.build b/third_party/libwebrtc/modules/video_coding/timing/timing_module_gn/moz.build
new file mode 100644
index 0000000000..8ab43fb748
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/timing_module_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/timing/timing.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("timing_module_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/timing/timing_unittest.cc b/third_party/libwebrtc/modules/video_coding/timing/timing_unittest.cc
new file mode 100644
index 0000000000..8633c0de39
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/timing_unittest.cc
@@ -0,0 +1,339 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/timing/timing.h"
+
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+namespace {
+
+constexpr Frequency k25Fps = Frequency::Hertz(25);
+constexpr Frequency k90kHz = Frequency::KiloHertz(90);
+
+}  // namespace
+
+TEST(ReceiverTimingTest, JitterDelay) {
+  test::ScopedKeyValueConfig field_trials;
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.Reset();
+
+  uint32_t timestamp = 0;
+  timing.UpdateCurrentDelay(timestamp);
+
+  timing.Reset();
+
+  timing.IncomingTimestamp(timestamp, clock.CurrentTime());
+  TimeDelta jitter_delay = TimeDelta::Millis(20);
+  timing.SetJitterDelay(jitter_delay);
+  timing.UpdateCurrentDelay(timestamp);
+  timing.set_render_delay(TimeDelta::Zero());
+  auto wait_time = timing.MaxWaitingTime(
+      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
+      /*too_many_frames_queued=*/false);
+  // First update initializes the render time. Since we have no decode delay
+  // we get wait_time = renderTime - now - renderDelay = jitter.
+  EXPECT_EQ(jitter_delay, wait_time);
+
+  jitter_delay += TimeDelta::Millis(VCMTiming::kDelayMaxChangeMsPerS + 10);
+  timestamp += 90000;
+  clock.AdvanceTimeMilliseconds(1000);
+  timing.SetJitterDelay(jitter_delay);
+  timing.UpdateCurrentDelay(timestamp);
+  wait_time = timing.MaxWaitingTime(
+      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
+      /*too_many_frames_queued=*/false);
+  // Since we gradually increase the delay we only get 100 ms every second.
+  EXPECT_EQ(jitter_delay - TimeDelta::Millis(10), wait_time);
+
+  timestamp += 90000;
+  clock.AdvanceTimeMilliseconds(1000);
+  timing.UpdateCurrentDelay(timestamp);
+  wait_time = timing.MaxWaitingTime(
+      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
+      /*too_many_frames_queued=*/false);
+  EXPECT_EQ(jitter_delay, wait_time);
+
+  // Insert frames without jitter, verify that this gives the exact wait time.
+  const int kNumFrames = 300;
+  for (int i = 0; i < kNumFrames; i++) {
+    clock.AdvanceTime(1 / k25Fps);
+    timestamp += k90kHz / k25Fps;
+    timing.IncomingTimestamp(timestamp, clock.CurrentTime());
+  }
+  timing.UpdateCurrentDelay(timestamp);
+  wait_time = timing.MaxWaitingTime(
+      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
+      /*too_many_frames_queued=*/false);
+  EXPECT_EQ(jitter_delay, wait_time);
+
+  // Add decode time estimates for 1 second.
+  const TimeDelta kDecodeTime = TimeDelta::Millis(10);
+  for (int i = 0; i < k25Fps.hertz(); i++) {
+    clock.AdvanceTime(kDecodeTime);
+    timing.StopDecodeTimer(kDecodeTime, clock.CurrentTime());
+    timestamp += k90kHz / k25Fps;
+    clock.AdvanceTime(1 / k25Fps - kDecodeTime);
+    timing.IncomingTimestamp(timestamp, clock.CurrentTime());
+  }
+  timing.UpdateCurrentDelay(timestamp);
+  wait_time = timing.MaxWaitingTime(
+      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
+      /*too_many_frames_queued=*/false);
+  EXPECT_EQ(jitter_delay, wait_time);
+
+  const TimeDelta kMinTotalDelay = TimeDelta::Millis(200);
+  timing.set_min_playout_delay(kMinTotalDelay);
+  clock.AdvanceTimeMilliseconds(5000);
+  timestamp += 5 * 90000;
+  timing.UpdateCurrentDelay(timestamp);
+  const TimeDelta kRenderDelay = TimeDelta::Millis(10);
+  timing.set_render_delay(kRenderDelay);
+  wait_time = timing.MaxWaitingTime(
+      timing.RenderTime(timestamp, clock.CurrentTime()), clock.CurrentTime(),
+      /*too_many_frames_queued=*/false);
+  // We should at least have kMinTotalDelayMs - decodeTime (10) - renderTime
+  // (10) to wait.
+  EXPECT_EQ(kMinTotalDelay - kDecodeTime - kRenderDelay, wait_time);
+  // The total video delay should be equal to the min total delay.
+  EXPECT_EQ(kMinTotalDelay, timing.TargetVideoDelay());
+
+  // Reset playout delay.
+  timing.set_min_playout_delay(TimeDelta::Zero());
+  clock.AdvanceTimeMilliseconds(5000);
+  timestamp += 5 * 90000;
+  timing.UpdateCurrentDelay(timestamp);
+}
+
+TEST(ReceiverTimingTest, TimestampWrapAround) {
+  constexpr auto kStartTime = Timestamp::Millis(1337);
+  test::ScopedKeyValueConfig field_trials;
+  SimulatedClock clock(kStartTime);
+  VCMTiming timing(&clock, field_trials);
+
+  // Provoke a wrap-around. The fifth frame will have wrapped at 25 fps.
+  constexpr uint32_t kRtpTicksPerFrame = k90kHz / k25Fps;
+  uint32_t timestamp = 0xFFFFFFFFu - 3 * kRtpTicksPerFrame;
+  for (int i = 0; i < 5; ++i) {
+    timing.IncomingTimestamp(timestamp, clock.CurrentTime());
+    clock.AdvanceTime(1 / k25Fps);
+    timestamp += kRtpTicksPerFrame;
+    EXPECT_EQ(kStartTime + 3 / k25Fps,
+              timing.RenderTime(0xFFFFFFFFu, clock.CurrentTime()));
+    // One ms later in 90 kHz.
+    EXPECT_EQ(kStartTime + 3 / k25Fps + TimeDelta::Millis(1),
+              timing.RenderTime(89u, clock.CurrentTime()));
+  }
+}
+
+TEST(ReceiverTimingTest, UseLowLatencyRenderer) {
+  test::ScopedKeyValueConfig field_trials;
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.Reset();
+  // Default is false.
+  EXPECT_FALSE(timing.RenderParameters().use_low_latency_rendering);
+  // False if min playout delay > 0.
+  timing.set_min_playout_delay(TimeDelta::Millis(10));
+  timing.set_max_playout_delay(TimeDelta::Millis(20));
+  EXPECT_FALSE(timing.RenderParameters().use_low_latency_rendering);
+  // True if min==0, max > 0.
+  timing.set_min_playout_delay(TimeDelta::Zero());
+  EXPECT_TRUE(timing.RenderParameters().use_low_latency_rendering);
+  // True if min==max==0.
+  timing.set_max_playout_delay(TimeDelta::Zero());
+  EXPECT_TRUE(timing.RenderParameters().use_low_latency_rendering);
+  // True also for max playout delay==500 ms.
+  timing.set_max_playout_delay(TimeDelta::Millis(500));
+  EXPECT_TRUE(timing.RenderParameters().use_low_latency_rendering);
+  // False if max playout delay > 500 ms.
+  timing.set_max_playout_delay(TimeDelta::Millis(501));
+  EXPECT_FALSE(timing.RenderParameters().use_low_latency_rendering);
+}
+
+TEST(ReceiverTimingTest, MaxWaitingTimeIsZeroForZeroRenderTime) {
+  // This is the default path when the RTP playout delay header extension is set
+  // to min==0 and max==0.
+  constexpr int64_t kStartTimeUs = 3.15e13;  // About one year in us.
+  constexpr TimeDelta kTimeDelta = 1 / Frequency::Hertz(60);
+  constexpr Timestamp kZeroRenderTime = Timestamp::Zero();
+  SimulatedClock clock(kStartTimeUs);
+  test::ScopedKeyValueConfig field_trials;
+  VCMTiming timing(&clock, field_trials);
+  timing.Reset();
+  timing.set_max_playout_delay(TimeDelta::Zero());
+  for (int i = 0; i < 10; ++i) {
+    clock.AdvanceTime(kTimeDelta);
+    Timestamp now = clock.CurrentTime();
+    EXPECT_LT(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                    /*too_many_frames_queued=*/false),
+              TimeDelta::Zero());
+  }
+  // Another frame submitted at the same time also returns a negative max
+  // waiting time.
+  Timestamp now = clock.CurrentTime();
+  EXPECT_LT(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                  /*too_many_frames_queued=*/false),
+            TimeDelta::Zero());
+  // MaxWaitingTime should be less than zero even if there's a burst of frames.
+  EXPECT_LT(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                  /*too_many_frames_queued=*/false),
+            TimeDelta::Zero());
+  EXPECT_LT(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                  /*too_many_frames_queued=*/false),
+            TimeDelta::Zero());
+  EXPECT_LT(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                  /*too_many_frames_queued=*/false),
+            TimeDelta::Zero());
+}
+
+TEST(ReceiverTimingTest, MaxWaitingTimeZeroDelayPacingExperiment) {
+  // The minimum pacing is enabled by a field trial and active if the RTP
+  // playout delay header extension is set to min==0.
+  constexpr TimeDelta kMinPacing = TimeDelta::Millis(3);
+  test::ScopedKeyValueConfig field_trials(
+      "WebRTC-ZeroPlayoutDelay/min_pacing:3ms/");
+  constexpr int64_t kStartTimeUs = 3.15e13;  // About one year in us.
+  constexpr TimeDelta kTimeDelta = 1 / Frequency::Hertz(60);
+  constexpr auto kZeroRenderTime = Timestamp::Zero();
+  SimulatedClock clock(kStartTimeUs);
+  VCMTiming timing(&clock, field_trials);
+  timing.Reset();
+  // MaxWaitingTime() returns zero for evenly spaced video frames.
+  for (int i = 0; i < 10; ++i) {
+    clock.AdvanceTime(kTimeDelta);
+    Timestamp now = clock.CurrentTime();
+    EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                    /*too_many_frames_queued=*/false),
+              TimeDelta::Zero());
+    timing.SetLastDecodeScheduledTimestamp(now);
+  }
+  // Another frame submitted at the same time is paced according to the field
+  // trial setting.
+  auto now = clock.CurrentTime();
+  EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                  /*too_many_frames_queued=*/false),
+            kMinPacing);
+  // If there's a burst of frames, the wait time is calculated based on next
+  // decode time.
+  EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                  /*too_many_frames_queued=*/false),
+            kMinPacing);
+  EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                  /*too_many_frames_queued=*/false),
+            kMinPacing);
+  // Allow a few ms to pass, this should be subtracted from the MaxWaitingTime.
+  constexpr TimeDelta kTwoMs = TimeDelta::Millis(2);
+  clock.AdvanceTime(kTwoMs);
+  now = clock.CurrentTime();
+  EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                  /*too_many_frames_queued=*/false),
+            kMinPacing - kTwoMs);
+  // A frame is decoded at the current time, the wait time should be restored to
+  // pacing delay.
+  timing.SetLastDecodeScheduledTimestamp(now);
+  EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                  /*too_many_frames_queued=*/false),
+            kMinPacing);
+}
+
+TEST(ReceiverTimingTest, DefaultMaxWaitingTimeUnaffectedByPacingExperiment) {
+  // The minimum pacing is enabled by a field trial but should not have any
+  // effect if render_time_ms is greater than 0;
+  test::ScopedKeyValueConfig field_trials(
+      "WebRTC-ZeroPlayoutDelay/min_pacing:3ms/");
+  constexpr int64_t kStartTimeUs = 3.15e13;  // About one year in us.
+  const TimeDelta kTimeDelta = TimeDelta::Millis(1000.0 / 60.0);
+  SimulatedClock clock(kStartTimeUs);
+  VCMTiming timing(&clock, field_trials);
+  timing.Reset();
+  clock.AdvanceTime(kTimeDelta);
+  auto now = clock.CurrentTime();
+  Timestamp render_time = now + TimeDelta::Millis(30);
+  // Estimate the internal processing delay from the first frame.
+  TimeDelta estimated_processing_delay =
+      (render_time - now) -
+      timing.MaxWaitingTime(render_time, now,
+                            /*too_many_frames_queued=*/false);
+  EXPECT_GT(estimated_processing_delay, TimeDelta::Zero());
+
+  // Any other frame submitted at the same time should be scheduled according to
+  // its render time.
+  for (int i = 0; i < 5; ++i) {
+    render_time += kTimeDelta;
+    EXPECT_EQ(timing.MaxWaitingTime(render_time, now,
+                                    /*too_many_frames_queued=*/false),
+              render_time - now - estimated_processing_delay);
+  }
+}
+
+TEST(ReceiverTimingTest, MaxWaitingTimeReturnsZeroIfTooManyFramesQueuedIsTrue) {
+  // The minimum pacing is enabled by a field trial and active if the RTP
+  // playout delay header extension is set to min==0.
+  constexpr TimeDelta kMinPacing = TimeDelta::Millis(3);
+  test::ScopedKeyValueConfig field_trials(
+      "WebRTC-ZeroPlayoutDelay/min_pacing:3ms/");
+  constexpr int64_t kStartTimeUs = 3.15e13;  // About one year in us.
+  const TimeDelta kTimeDelta = TimeDelta::Millis(1000.0 / 60.0);
+  constexpr auto kZeroRenderTime = Timestamp::Zero();
+  SimulatedClock clock(kStartTimeUs);
+  VCMTiming timing(&clock, field_trials);
+  timing.Reset();
+  // MaxWaitingTime() returns zero for evenly spaced video frames.
+  for (int i = 0; i < 10; ++i) {
+    clock.AdvanceTime(kTimeDelta);
+    auto now = clock.CurrentTime();
+    EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now,
+                                    /*too_many_frames_queued=*/false),
+              TimeDelta::Zero());
+    timing.SetLastDecodeScheduledTimestamp(now);
+  }
+  // Another frame submitted at the same time is paced according to the field
+  // trial setting.
+  auto now_ms = clock.CurrentTime();
+  EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now_ms,
+                                  /*too_many_frames_queued=*/false),
+            kMinPacing);
+  // MaxWaitingTime returns 0 even if there's a burst of frames if
+  // too_many_frames_queued is set to true.
+  EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now_ms,
+                                  /*too_many_frames_queued=*/true),
+            TimeDelta::Zero());
+  EXPECT_EQ(timing.MaxWaitingTime(kZeroRenderTime, now_ms,
+                                  /*too_many_frames_queued=*/true),
+            TimeDelta::Zero());
+}
+
+TEST(ReceiverTimingTest, UpdateCurrentDelayCapsWhenOffByMicroseconds) {
+  test::ScopedKeyValueConfig field_trials;
+  SimulatedClock clock(0);
+  VCMTiming timing(&clock, field_trials);
+  timing.Reset();
+
+  // Set larger initial current delay.
+  timing.set_min_playout_delay(TimeDelta::Millis(200));
+  timing.UpdateCurrentDelay(Timestamp::Millis(900), Timestamp::Millis(1000));
+
+  // Add a few microseconds to ensure that the delta of decode time is 0 after
+  // rounding, and should reset to the target delay.
+  timing.set_min_playout_delay(TimeDelta::Millis(50));
+  Timestamp decode_time = Timestamp::Millis(1337);
+  Timestamp render_time =
+      decode_time + TimeDelta::Millis(10) + TimeDelta::Micros(37);
+  timing.UpdateCurrentDelay(render_time, decode_time);
+  EXPECT_EQ(timing.GetTimings().current_delay, timing.TargetVideoDelay());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler.cc b/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler.cc
new file mode 100644
index 0000000000..13502a142b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/bandwidth_quality_scaler.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/video/video_adaptation_reason.h"
+#include "api/video_codecs/video_encoder.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/bandwidth_quality_scaler_settings.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/exp_filter.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/weak_ptr.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDefaultMaxWindowSizeMs = 5000;
+constexpr float kHigherMaxBitrateTolerationFactor = 0.95;
+constexpr float kLowerMinBitrateTolerationFactor = 0.8;
+constexpr int kDefaultBitrateStateUpdateIntervalSeconds = 5;
+}  // namespace
+
+BandwidthQualityScaler::BandwidthQualityScaler(
+    BandwidthQualityScalerUsageHandlerInterface* handler)
+    : kBitrateStateUpdateInterval(TimeDelta::Seconds(
+          BandwidthQualityScalerSettings::ParseFromFieldTrials()
+              .BitrateStateUpdateInterval()
+              .value_or(kDefaultBitrateStateUpdateIntervalSeconds))),
+      handler_(handler),
+      encoded_bitrate_(kDefaultMaxWindowSizeMs, RateStatistics::kBpsScale),
+      weak_ptr_factory_(this) {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  RTC_DCHECK(handler_ != nullptr);
+
+  StartCheckForBitrate();
+}
+
+BandwidthQualityScaler::~BandwidthQualityScaler() {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+}
+
+void BandwidthQualityScaler::StartCheckForBitrate() {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  TaskQueueBase::Current()->PostDelayedTask(
+      [this_weak_ptr = weak_ptr_factory_.GetWeakPtr(), this] {
+        if (!this_weak_ptr) {
+          // The caller BandwidthQualityScaler has been deleted.
+          return;
+        }
+        RTC_DCHECK_RUN_ON(&task_checker_);
+        switch (CheckBitrate()) {
+          case BandwidthQualityScaler::CheckBitrateResult::kHighBitRate: {
+            handler_->OnReportUsageBandwidthHigh();
+            last_frame_size_pixels_.reset();
+            break;
+          }
+          case BandwidthQualityScaler::CheckBitrateResult::kLowBitRate: {
+            handler_->OnReportUsageBandwidthLow();
+            last_frame_size_pixels_.reset();
+            break;
+          }
+          case BandwidthQualityScaler::CheckBitrateResult::kNormalBitrate: {
+            break;
+          }
+          case BandwidthQualityScaler::CheckBitrateResult::
+              kInsufficientSamples: {
+            break;
+          }
+        }
+        StartCheckForBitrate();
+      },
+      kBitrateStateUpdateInterval);
+}
+
+void BandwidthQualityScaler::ReportEncodeInfo(int frame_size_bytes,
+                                              int64_t time_sent_in_ms,
+                                              uint32_t encoded_width,
+                                              uint32_t encoded_height) {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  last_time_sent_in_ms_ = time_sent_in_ms;
+  last_frame_size_pixels_ = encoded_width * encoded_height;
+  encoded_bitrate_.Update(frame_size_bytes, time_sent_in_ms);
+}
+
+void BandwidthQualityScaler::SetResolutionBitrateLimits(
+    const std::vector<VideoEncoder::ResolutionBitrateLimits>&
+        resolution_bitrate_limits) {
+  if (resolution_bitrate_limits.empty()) {
+    resolution_bitrate_limits_ = EncoderInfoSettings::
+        GetDefaultSinglecastBitrateLimitsWhenQpIsUntrusted();
+  } else {
+    resolution_bitrate_limits_ = resolution_bitrate_limits;
+  }
+}
+
+BandwidthQualityScaler::CheckBitrateResult
+BandwidthQualityScaler::CheckBitrate() {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  if (!last_frame_size_pixels_.has_value() ||
+      !last_time_sent_in_ms_.has_value()) {
+    return BandwidthQualityScaler::CheckBitrateResult::kInsufficientSamples;
+  }
+
+  absl::optional<int64_t> current_bitrate_bps =
+      encoded_bitrate_.Rate(last_time_sent_in_ms_.value());
+  if (!current_bitrate_bps.has_value()) {
+    // We can't get a valid bitrate due to not enough data points.
+    return BandwidthQualityScaler::CheckBitrateResult::kInsufficientSamples;
+  }
+  absl::optional<VideoEncoder::ResolutionBitrateLimits> suitable_bitrate_limit =
+      EncoderInfoSettings::
+          GetSinglecastBitrateLimitForResolutionWhenQpIsUntrusted(
+              last_frame_size_pixels_, resolution_bitrate_limits_);
+
+  if (!suitable_bitrate_limit.has_value()) {
+    return BandwidthQualityScaler::CheckBitrateResult::kInsufficientSamples;
+  }
+
+  // Multiply by toleration factor to solve the frequent adaptation due to
+  // critical value.
+  if (current_bitrate_bps > suitable_bitrate_limit->max_bitrate_bps *
+                                kHigherMaxBitrateTolerationFactor) {
+    return BandwidthQualityScaler::CheckBitrateResult::kLowBitRate;
+  } else if (current_bitrate_bps <
+             suitable_bitrate_limit->min_start_bitrate_bps *
+                 kLowerMinBitrateTolerationFactor) {
+    return BandwidthQualityScaler::CheckBitrateResult::kHighBitRate;
+  }
+  return BandwidthQualityScaler::CheckBitrateResult::kNormalBitrate;
+}
+
+BandwidthQualityScalerUsageHandlerInterface::
+    ~BandwidthQualityScalerUsageHandlerInterface() {}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler.h b/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler.h
new file mode 100644
index 0000000000..7cd1de0dd2
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_BANDWIDTH_QUALITY_SCALER_H_
+#define MODULES_VIDEO_CODING_UTILITY_BANDWIDTH_QUALITY_SCALER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/video_codecs/video_encoder.h"
+#include "rtc_base/experiments/encoder_info_settings.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/exp_filter.h"
+#include "rtc_base/rate_statistics.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/weak_ptr.h"
+
+namespace webrtc {
+
+class BandwidthQualityScalerUsageHandlerInterface {
+ public:
+  virtual ~BandwidthQualityScalerUsageHandlerInterface();
+
+  virtual void OnReportUsageBandwidthHigh() = 0;
+  virtual void OnReportUsageBandwidthLow() = 0;
+};
+
+// BandwidthQualityScaler runs asynchronously and monitors bandwidth values of
+// encoded frames. It holds a reference to a
+// BandwidthQualityScalerUsageHandlerInterface implementation to signal an
+// overuse or underuse of bandwidth (which indicate a desire to scale the video
+// stream down or up).
+class BandwidthQualityScaler {
+ public:
+  explicit BandwidthQualityScaler(
+      BandwidthQualityScalerUsageHandlerInterface* handler);
+  virtual ~BandwidthQualityScaler();
+
+  void ReportEncodeInfo(int frame_size_bytes,
+                        int64_t time_sent_in_ms,
+                        uint32_t encoded_width,
+                        uint32_t encoded_height);
+
+  // We prioritise to using the |resolution_bitrate_limits| provided by the
+  // current decoder. If not provided, we will use the default data by
+  // GetDefaultResolutionBitrateLimits().
+  void SetResolutionBitrateLimits(
+      const std::vector<VideoEncoder::ResolutionBitrateLimits>&
+          resolution_bitrate_limits);
+
+  const TimeDelta kBitrateStateUpdateInterval;
+
+ private:
+  enum class CheckBitrateResult {
+    kInsufficientSamples,
+    kNormalBitrate,
+    kHighBitRate,
+    kLowBitRate,
+  };
+
+  // We will periodically check encode bitrate, this function will make
+  // resolution up or down decisions and report the decision to the adapter.
+  void StartCheckForBitrate();
+  CheckBitrateResult CheckBitrate();
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker task_checker_;
+  BandwidthQualityScalerUsageHandlerInterface* const handler_
+      RTC_GUARDED_BY(&task_checker_);
+
+  absl::optional<int64_t> last_time_sent_in_ms_ RTC_GUARDED_BY(&task_checker_);
+  RateStatistics encoded_bitrate_ RTC_GUARDED_BY(&task_checker_);
+  absl::optional<int> last_frame_size_pixels_ RTC_GUARDED_BY(&task_checker_);
+  rtc::WeakPtrFactory<BandwidthQualityScaler> weak_ptr_factory_;
+
+  std::vector<VideoEncoder::ResolutionBitrateLimits> resolution_bitrate_limits_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CODING_UTILITY_BANDWIDTH_QUALITY_SCALER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc
new file mode 100644
index 0000000000..d28052e28d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc
@@ -0,0 +1,278 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/bandwidth_quality_scaler.h"
+
+#include <memory>
+#include <string>
+
+#include "api/units/time_delta.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/experiments/encoder_info_settings.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/time_utils.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kFramerateFps = 30;
+constexpr TimeDelta kDefaultBitrateStateUpdateInterval = TimeDelta::Seconds(5);
+constexpr TimeDelta kDefaultEncodeTime = TimeDelta::Seconds(1) / kFramerateFps;
+
+}  // namespace
+
+class FakeBandwidthQualityScalerHandler
+    : public BandwidthQualityScalerUsageHandlerInterface {
+ public:
+  ~FakeBandwidthQualityScalerHandler() override = default;
+  void OnReportUsageBandwidthHigh() override {
+    adapt_down_event_count_++;
+    event_.Set();
+  }
+
+  void OnReportUsageBandwidthLow() override {
+    adapt_up_event_count_++;
+    event_.Set();
+  }
+
+  rtc::Event event_;
+  int adapt_up_event_count_ = 0;
+  int adapt_down_event_count_ = 0;
+};
+
+class BandwidthQualityScalerUnderTest : public BandwidthQualityScaler {
+ public:
+  explicit BandwidthQualityScalerUnderTest(
+      BandwidthQualityScalerUsageHandlerInterface* handler)
+      : BandwidthQualityScaler(handler) {}
+
+  int GetBitrateStateUpdateIntervalMs() {
+    return this->kBitrateStateUpdateInterval.ms() + 200;
+  }
+};
+
+class BandwidthQualityScalerTest
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<std::string> {
+ protected:
+  enum ScaleDirection {
+    kKeepScaleNormalBandwidth,
+    kKeepScaleAboveMaxBandwidth,
+    kKeepScaleUnderMinBandwidth,
+  };
+
+  enum FrameType {
+    kKeyFrame,
+    kNormalFrame,
+    kNormalFrame_Overuse,
+    kNormalFrame_Underuse,
+  };
+  struct FrameConfig {
+    FrameConfig(int frame_num,
+                FrameType frame_type,
+                int actual_width,
+                int actual_height)
+        : frame_num(frame_num),
+          frame_type(frame_type),
+          actual_width(actual_width),
+          actual_height(actual_height) {}
+
+    int frame_num;
+    FrameType frame_type;
+    int actual_width;
+    int actual_height;
+  };
+
+  BandwidthQualityScalerTest()
+      : scoped_field_trial_(GetParam()),
+        task_queue_("BandwidthQualityScalerTestQueue"),
+        handler_(std::make_unique<FakeBandwidthQualityScalerHandler>()) {
+    task_queue_.SendTask(
+        [this] {
+          bandwidth_quality_scaler_ =
+              std::unique_ptr<BandwidthQualityScalerUnderTest>(
+                  new BandwidthQualityScalerUnderTest(handler_.get()));
+          bandwidth_quality_scaler_->SetResolutionBitrateLimits(
+              EncoderInfoSettings::
+                  GetDefaultSinglecastBitrateLimitsWhenQpIsUntrusted());
+          // Only for testing. Set first_timestamp_ in RateStatistics to 0.
+          bandwidth_quality_scaler_->ReportEncodeInfo(0, 0, 0, 0);
+        });
+  }
+
+  ~BandwidthQualityScalerTest() {
+    task_queue_.SendTask([this] { bandwidth_quality_scaler_ = nullptr; });
+  }
+
+  int GetFrameSizeBytes(
+      const FrameConfig& config,
+      const VideoEncoder::ResolutionBitrateLimits& bitrate_limits) {
+    int scale = 8 * kFramerateFps;
+    switch (config.frame_type) {
+      case FrameType::kKeyFrame: {
+        // 4 is experimental value. Based on the test, the number of bytes of
+        // the key frame is about four times of the normal frame
+        return bitrate_limits.max_bitrate_bps * 4 / scale;
+      }
+      case FrameType::kNormalFrame_Overuse: {
+        return bitrate_limits.max_bitrate_bps * 3 / 2 / scale;
+      }
+      case FrameType::kNormalFrame_Underuse: {
+        return bitrate_limits.min_start_bitrate_bps * 3 / 4 / scale;
+      }
+      case FrameType::kNormalFrame: {
+        return (bitrate_limits.max_bitrate_bps +
+                bitrate_limits.min_start_bitrate_bps) /
+               2 / scale;
+      }
+    }
+    return -1;
+  }
+
+  absl::optional<VideoEncoder::ResolutionBitrateLimits>
+  GetDefaultSuitableBitrateLimit(int frame_size_pixels) {
+    return EncoderInfoSettings::
+        GetSinglecastBitrateLimitForResolutionWhenQpIsUntrusted(
+            frame_size_pixels,
+            EncoderInfoSettings::
+                GetDefaultSinglecastBitrateLimitsWhenQpIsUntrusted());
+  }
+
+  void TriggerBandwidthQualityScalerTest(
+      const std::vector<FrameConfig>& frame_configs) {
+    task_queue_.SendTask(
+        [frame_configs, this] {
+          RTC_CHECK(!frame_configs.empty());
+
+          int total_frame_nums = 0;
+          for (const FrameConfig& frame_config : frame_configs) {
+            total_frame_nums += frame_config.frame_num;
+          }
+
+          EXPECT_EQ(
+              kFramerateFps * kDefaultBitrateStateUpdateInterval.seconds(),
+              total_frame_nums);
+
+          uint32_t time_send_to_scaler_ms_ = rtc::TimeMillis();
+          for (size_t i = 0; i < frame_configs.size(); ++i) {
+            const FrameConfig& config = frame_configs[i];
+            absl::optional<VideoEncoder::ResolutionBitrateLimits>
+                suitable_bitrate = GetDefaultSuitableBitrateLimit(
+                    config.actual_width * config.actual_height);
+            EXPECT_TRUE(suitable_bitrate);
+            for (int j = 0; j <= config.frame_num; ++j) {
+              time_send_to_scaler_ms_ += kDefaultEncodeTime.ms();
+              int frame_size_bytes =
+                  GetFrameSizeBytes(config, suitable_bitrate.value());
+              RTC_CHECK(frame_size_bytes > 0);
+              bandwidth_quality_scaler_->ReportEncodeInfo(
+                  frame_size_bytes, time_send_to_scaler_ms_,
+                  config.actual_width, config.actual_height);
+            }
+          }
+        });
+  }
+
+  test::ScopedFieldTrials scoped_field_trial_;
+  TaskQueueForTest task_queue_;
+  std::unique_ptr<BandwidthQualityScalerUnderTest> bandwidth_quality_scaler_;
+  std::unique_ptr<FakeBandwidthQualityScalerHandler> handler_;
+};
+
+INSTANTIATE_TEST_SUITE_P(
+    FieldTrials,
+    BandwidthQualityScalerTest,
+    ::testing::Values("WebRTC-Video-BandwidthQualityScalerSettings/"
+                      "bitrate_state_update_interval_s_:1/",
+                      "WebRTC-Video-BandwidthQualityScalerSettings/"
+                      "bitrate_state_update_interval_s_:2/"));
+
+TEST_P(BandwidthQualityScalerTest, AllNormalFrame_640x360) {
+  const std::vector<FrameConfig> frame_configs{
+      FrameConfig(150, FrameType::kNormalFrame, 640, 360)};
+  TriggerBandwidthQualityScalerTest(frame_configs);
+
+  // When resolution is 640*360, experimental working bitrate range is
+  // [500000,800000] bps. Encoded bitrate is 654253, so it falls in the range
+  // without any operation(up/down).
+  EXPECT_FALSE(handler_->event_.Wait(TimeDelta::Millis(
+      bandwidth_quality_scaler_->GetBitrateStateUpdateIntervalMs())));
+  EXPECT_EQ(0, handler_->adapt_down_event_count_);
+  EXPECT_EQ(0, handler_->adapt_up_event_count_);
+}
+
+TEST_P(BandwidthQualityScalerTest, AllNoramlFrame_AboveMaxBandwidth_640x360) {
+  const std::vector<FrameConfig> frame_configs{
+      FrameConfig(150, FrameType::kNormalFrame_Overuse, 640, 360)};
+  TriggerBandwidthQualityScalerTest(frame_configs);
+
+  // When resolution is 640*360, experimental working bitrate range is
+  // [500000,800000] bps. Encoded bitrate is 1208000 > 800000 * 0.95, so it
+  // triggers adapt_up_event_count_.
+  EXPECT_TRUE(handler_->event_.Wait(TimeDelta::Millis(
+      bandwidth_quality_scaler_->GetBitrateStateUpdateIntervalMs())));
+  EXPECT_EQ(0, handler_->adapt_down_event_count_);
+  EXPECT_EQ(1, handler_->adapt_up_event_count_);
+}
+
+TEST_P(BandwidthQualityScalerTest, AllNormalFrame_Underuse_640x360) {
+  const std::vector<FrameConfig> frame_configs{
+      FrameConfig(150, FrameType::kNormalFrame_Underuse, 640, 360)};
+  TriggerBandwidthQualityScalerTest(frame_configs);
+
+  // When resolution is 640*360, experimental working bitrate range is
+  // [500000,800000] bps. Encoded bitrate is 377379 < 500000 * 0.8, so it
+  // triggers adapt_down_event_count_.
+  EXPECT_TRUE(handler_->event_.Wait(TimeDelta::Millis(
+      bandwidth_quality_scaler_->GetBitrateStateUpdateIntervalMs())));
+  EXPECT_EQ(1, handler_->adapt_down_event_count_);
+  EXPECT_EQ(0, handler_->adapt_up_event_count_);
+}
+
+TEST_P(BandwidthQualityScalerTest, FixedFrameTypeTest1_640x360) {
+  const std::vector<FrameConfig> frame_configs{
+      FrameConfig(5, FrameType::kNormalFrame_Underuse, 640, 360),
+      FrameConfig(110, FrameType::kNormalFrame, 640, 360),
+      FrameConfig(20, FrameType::kNormalFrame_Overuse, 640, 360),
+      FrameConfig(15, FrameType::kKeyFrame, 640, 360),
+  };
+  TriggerBandwidthQualityScalerTest(frame_configs);
+
+  // When resolution is 640*360, experimental working bitrate range is
+  // [500000,800000] bps. Encoded bitrate is 1059462 > 800000 * 0.95, so it
+  // triggers adapt_up_event_count_.
+  EXPECT_TRUE(handler_->event_.Wait(TimeDelta::Millis(
+      bandwidth_quality_scaler_->GetBitrateStateUpdateIntervalMs())));
+  EXPECT_EQ(0, handler_->adapt_down_event_count_);
+  EXPECT_EQ(1, handler_->adapt_up_event_count_);
+}
+
+TEST_P(BandwidthQualityScalerTest, FixedFrameTypeTest2_640x360) {
+  const std::vector<FrameConfig> frame_configs{
+      FrameConfig(10, FrameType::kNormalFrame_Underuse, 640, 360),
+      FrameConfig(50, FrameType::kNormalFrame, 640, 360),
+      FrameConfig(5, FrameType::kKeyFrame, 640, 360),
+      FrameConfig(85, FrameType::kNormalFrame_Overuse, 640, 360),
+  };
+  TriggerBandwidthQualityScalerTest(frame_configs);
+
+  // When resolution is 640*360, experimental working bitrate range is
+  // [500000,800000] bps. Encoded bitrate is 1059462 > 800000 * 0.95, so it
+  // triggers adapt_up_event_count_.
+  EXPECT_TRUE(handler_->event_.Wait(TimeDelta::Millis(
+      bandwidth_quality_scaler_->GetBitrateStateUpdateIntervalMs())));
+  EXPECT_EQ(0, handler_->adapt_down_event_count_);
+  EXPECT_EQ(1, handler_->adapt_up_event_count_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history.cc b/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history.cc
new file mode 100644
index 0000000000..1138aa8448
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history.cc
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/decoded_frames_history.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace video_coding {
+
+DecodedFramesHistory::DecodedFramesHistory(size_t window_size)
+    : buffer_(window_size) {}
+
+DecodedFramesHistory::~DecodedFramesHistory() = default;
+
+void DecodedFramesHistory::InsertDecoded(int64_t frame_id, uint32_t timestamp) {
+  last_decoded_frame_ = frame_id;
+  last_decoded_frame_timestamp_ = timestamp;
+  int new_index = FrameIdToIndex(frame_id);
+
+  RTC_DCHECK(last_frame_id_ < frame_id);
+
+  // Clears expired values from the cyclic buffer_.
+  if (last_frame_id_) {
+    int64_t id_jump = frame_id - *last_frame_id_;
+    int last_index = FrameIdToIndex(*last_frame_id_);
+
+    if (id_jump >= static_cast<int64_t>(buffer_.size())) {
+      std::fill(buffer_.begin(), buffer_.end(), false);
+    } else if (new_index > last_index) {
+      std::fill(buffer_.begin() + last_index + 1, buffer_.begin() + new_index,
+                false);
+    } else {
+      std::fill(buffer_.begin() + last_index + 1, buffer_.end(), false);
+      std::fill(buffer_.begin(), buffer_.begin() + new_index, false);
+    }
+  }
+
+  buffer_[new_index] = true;
+  last_frame_id_ = frame_id;
+}
+
+bool DecodedFramesHistory::WasDecoded(int64_t frame_id) const {
+  if (!last_frame_id_)
+    return false;
+
+  // Reference to the picture_id out of the stored should happen.
+  if (frame_id <= *last_frame_id_ - static_cast<int64_t>(buffer_.size())) {
+    RTC_LOG(LS_WARNING) << "Referencing a frame out of the window. "
+                           "Assuming it was undecoded to avoid artifacts.";
+    return false;
+  }
+
+  if (frame_id > last_frame_id_)
+    return false;
+
+  return buffer_[FrameIdToIndex(frame_id)];
+}
+
+void DecodedFramesHistory::Clear() {
+  last_decoded_frame_timestamp_.reset();
+  last_decoded_frame_.reset();
+  std::fill(buffer_.begin(), buffer_.end(), false);
+  last_frame_id_.reset();
+}
+
+absl::optional<int64_t> DecodedFramesHistory::GetLastDecodedFrameId() const {
+  return last_decoded_frame_;
+}
+
+absl::optional<uint32_t> DecodedFramesHistory::GetLastDecodedFrameTimestamp()
+    const {
+  return last_decoded_frame_timestamp_;
+}
+
+int DecodedFramesHistory::FrameIdToIndex(int64_t frame_id) const {
+  int m = frame_id % buffer_.size();
+  return m >= 0 ? m : m + buffer_.size();
+}
+
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history.h b/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history.h
new file mode 100644
index 0000000000..9b8bf65821
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_DECODED_FRAMES_HISTORY_H_
+#define MODULES_VIDEO_CODING_UTILITY_DECODED_FRAMES_HISTORY_H_
+
+#include <stdint.h>
+
+#include <bitset>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/video/encoded_frame.h"
+
+namespace webrtc {
+namespace video_coding {
+
+class DecodedFramesHistory {
+ public:
+  // window_size - how much frames back to the past are actually remembered.
+  explicit DecodedFramesHistory(size_t window_size);
+  ~DecodedFramesHistory();
+  // Called for each decoded frame. Assumes frame id's are non-decreasing.
+  void InsertDecoded(int64_t frame_id, uint32_t timestamp);
+  // Query if the following (frame_id, spatial_id) pair was inserted before.
+  // Should be at most less by window_size-1 than the last inserted frame id.
+  bool WasDecoded(int64_t frame_id) const;
+
+  void Clear();
+
+  absl::optional<int64_t> GetLastDecodedFrameId() const;
+  absl::optional<uint32_t> GetLastDecodedFrameTimestamp() const;
+
+ private:
+  int FrameIdToIndex(int64_t frame_id) const;
+
+  std::vector<bool> buffer_;
+  absl::optional<int64_t> last_frame_id_;
+  absl::optional<int64_t> last_decoded_frame_;
+  absl::optional<uint32_t> last_decoded_frame_timestamp_;
+};
+
+}  // namespace video_coding
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CODING_UTILITY_DECODED_FRAMES_HISTORY_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history_unittest.cc
new file mode 100644
index 0000000000..ac09a42053
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history_unittest.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/decoded_frames_history.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace video_coding {
+namespace {
+
+constexpr int kHistorySize = 1 << 13;
+
+TEST(DecodedFramesHistory, RequestOnEmptyHistory) {
+  DecodedFramesHistory history(kHistorySize);
+  EXPECT_EQ(history.WasDecoded(1234), false);
+}
+
+TEST(DecodedFramesHistory, FindsLastDecodedFrame) {
+  DecodedFramesHistory history(kHistorySize);
+  history.InsertDecoded(1234, 0);
+  EXPECT_EQ(history.WasDecoded(1234), true);
+}
+
+TEST(DecodedFramesHistory, FindsPreviousFrame) {
+  DecodedFramesHistory history(kHistorySize);
+  history.InsertDecoded(1234, 0);
+  history.InsertDecoded(1235, 0);
+  EXPECT_EQ(history.WasDecoded(1234), true);
+}
+
+TEST(DecodedFramesHistory, ReportsMissingFrame) {
+  DecodedFramesHistory history(kHistorySize);
+  history.InsertDecoded(1234, 0);
+  history.InsertDecoded(1236, 0);
+  EXPECT_EQ(history.WasDecoded(1235), false);
+}
+
+TEST(DecodedFramesHistory, ClearsHistory) {
+  DecodedFramesHistory history(kHistorySize);
+  history.InsertDecoded(1234, 0);
+  history.Clear();
+  EXPECT_EQ(history.WasDecoded(1234), false);
+  EXPECT_EQ(history.GetLastDecodedFrameId(), absl::nullopt);
+  EXPECT_EQ(history.GetLastDecodedFrameTimestamp(), absl::nullopt);
+}
+
+TEST(DecodedFramesHistory, HandlesBigJumpInPictureId) {
+  DecodedFramesHistory history(kHistorySize);
+  history.InsertDecoded(1234, 0);
+  history.InsertDecoded(1235, 0);
+  history.InsertDecoded(1236, 0);
+  history.InsertDecoded(1236 + kHistorySize / 2, 0);
+  EXPECT_EQ(history.WasDecoded(1234), true);
+  EXPECT_EQ(history.WasDecoded(1237), false);
+}
+
+TEST(DecodedFramesHistory, ForgetsTooOldHistory) {
+  DecodedFramesHistory history(kHistorySize);
+  history.InsertDecoded(1234, 0);
+  history.InsertDecoded(1235, 0);
+  history.InsertDecoded(1236, 0);
+  history.InsertDecoded(1236 + kHistorySize * 2, 0);
+  EXPECT_EQ(history.WasDecoded(1234), false);
+  EXPECT_EQ(history.WasDecoded(1237), false);
+}
+
+TEST(DecodedFramesHistory, ReturnsLastDecodedFrameId) {
+  DecodedFramesHistory history(kHistorySize);
+  EXPECT_EQ(history.GetLastDecodedFrameId(), absl::nullopt);
+  history.InsertDecoded(1234, 0);
+  EXPECT_EQ(history.GetLastDecodedFrameId(), 1234);
+  history.InsertDecoded(1235, 0);
+  EXPECT_EQ(history.GetLastDecodedFrameId(), 1235);
+}
+
+TEST(DecodedFramesHistory, ReturnsLastDecodedFrameTimestamp) {
+  DecodedFramesHistory history(kHistorySize);
+  EXPECT_EQ(history.GetLastDecodedFrameTimestamp(), absl::nullopt);
+  history.InsertDecoded(1234, 12345);
+  EXPECT_EQ(history.GetLastDecodedFrameTimestamp(), 12345u);
+  history.InsertDecoded(1235, 12366);
+  EXPECT_EQ(history.GetLastDecodedFrameTimestamp(), 12366u);
+}
+
+TEST(DecodedFramesHistory, NegativePictureIds) {
+  DecodedFramesHistory history(kHistorySize);
+  history.InsertDecoded(-1234, 12345);
+  history.InsertDecoded(-1233, 12366);
+  EXPECT_EQ(*history.GetLastDecodedFrameId(), -1233);
+
+  history.InsertDecoded(-1, 12377);
+  history.InsertDecoded(0, 12388);
+  EXPECT_EQ(*history.GetLastDecodedFrameId(), 0);
+
+  history.InsertDecoded(1, 12399);
+  EXPECT_EQ(*history.GetLastDecodedFrameId(), 1);
+
+  EXPECT_EQ(history.WasDecoded(-1234), true);
+  EXPECT_EQ(history.WasDecoded(-1), true);
+  EXPECT_EQ(history.WasDecoded(0), true);
+  EXPECT_EQ(history.WasDecoded(1), true);
+}
+
+}  // namespace
+}  // namespace video_coding
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/frame_dropper.cc b/third_party/libwebrtc/modules/video_coding/utility/frame_dropper.cc
new file mode 100644
index 0000000000..8ea8a8e268
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/frame_dropper.cc
@@ -0,0 +1,268 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/frame_dropper.h"
+
+#include <algorithm>
+
+namespace webrtc {
+
+namespace {
+
+const float kDefaultFrameSizeAlpha = 0.9f;
+const float kDefaultKeyFrameRatioAlpha = 0.99f;
+// 1 key frame every 10th second in 30 fps.
+const float kDefaultKeyFrameRatioValue = 1 / 300.0f;
+
+const float kDefaultDropRatioAlpha = 0.9f;
+const float kDefaultDropRatioValue = 0.96f;
+// Maximum duration over which frames are continuously dropped.
+const float kDefaultMaxDropDurationSecs = 4.0f;
+
+// Default target bitrate.
+// TODO(isheriff): Should this be higher to avoid dropping too many packets when
+// the bandwidth is unknown at the start ?
+const float kDefaultTargetBitrateKbps = 300.0f;
+const float kDefaultIncomingFrameRate = 30;
+const float kLeakyBucketSizeSeconds = 0.5f;
+
+// A delta frame that is bigger than `kLargeDeltaFactor` times the average
+// delta frame is a large frame that is spread out for accumulation.
+const int kLargeDeltaFactor = 3;
+
+// Cap on the frame size accumulator to prevent excessive drops.
+const float kAccumulatorCapBufferSizeSecs = 3.0f;
+}  // namespace
+
+FrameDropper::FrameDropper()
+    : key_frame_ratio_(kDefaultKeyFrameRatioAlpha),
+      delta_frame_size_avg_kbits_(kDefaultFrameSizeAlpha),
+      drop_ratio_(kDefaultDropRatioAlpha, kDefaultDropRatioValue),
+      enabled_(true),
+      max_drop_duration_secs_(kDefaultMaxDropDurationSecs) {
+  Reset();
+}
+
+FrameDropper::~FrameDropper() = default;
+
+void FrameDropper::Reset() {
+  key_frame_ratio_.Reset(kDefaultKeyFrameRatioAlpha);
+  key_frame_ratio_.Apply(1.0f, kDefaultKeyFrameRatioValue);
+  delta_frame_size_avg_kbits_.Reset(kDefaultFrameSizeAlpha);
+
+  accumulator_ = 0.0f;
+  accumulator_max_ = kDefaultTargetBitrateKbps / 2;
+  target_bitrate_ = kDefaultTargetBitrateKbps;
+  incoming_frame_rate_ = kDefaultIncomingFrameRate;
+
+  large_frame_accumulation_count_ = 0;
+  large_frame_accumulation_chunk_size_ = 0;
+  large_frame_accumulation_spread_ = 0.5 * kDefaultIncomingFrameRate;
+
+  drop_next_ = false;
+  drop_ratio_.Reset(0.9f);
+  drop_ratio_.Apply(0.0f, 0.0f);
+  drop_count_ = 0;
+  was_below_max_ = true;
+}
+
+void FrameDropper::Enable(bool enable) {
+  enabled_ = enable;
+}
+
+void FrameDropper::Fill(size_t framesize_bytes, bool delta_frame) {
+  if (!enabled_) {
+    return;
+  }
+  float framesize_kbits = 8.0f * static_cast<float>(framesize_bytes) / 1000.0f;
+  if (!delta_frame) {
+    key_frame_ratio_.Apply(1.0, 1.0);
+    // Do not spread if we are already doing it (or we risk dropping bits that
+    // need accumulation). Given we compute the key frame ratio and spread
+    // based on that, this should not normally happen.
+    if (large_frame_accumulation_count_ == 0) {
+      if (key_frame_ratio_.filtered() > 1e-5 &&
+          1 / key_frame_ratio_.filtered() < large_frame_accumulation_spread_) {
+        large_frame_accumulation_count_ =
+            static_cast<int32_t>(1 / key_frame_ratio_.filtered() + 0.5);
+      } else {
+        large_frame_accumulation_count_ =
+            static_cast<int32_t>(large_frame_accumulation_spread_ + 0.5);
+      }
+      large_frame_accumulation_chunk_size_ =
+          framesize_kbits / large_frame_accumulation_count_;
+      framesize_kbits = 0;
+    }
+  } else {
+    // Identify if it is an unusually large delta frame and spread accumulation
+    // if that is the case.
+    if (delta_frame_size_avg_kbits_.filtered() != -1 &&
+        (framesize_kbits >
+         kLargeDeltaFactor * delta_frame_size_avg_kbits_.filtered()) &&
+        large_frame_accumulation_count_ == 0) {
+      large_frame_accumulation_count_ =
+          static_cast<int32_t>(large_frame_accumulation_spread_ + 0.5);
+      large_frame_accumulation_chunk_size_ =
+          framesize_kbits / large_frame_accumulation_count_;
+      framesize_kbits = 0;
+    } else {
+      delta_frame_size_avg_kbits_.Apply(1, framesize_kbits);
+    }
+    key_frame_ratio_.Apply(1.0, 0.0);
+  }
+  // Change the level of the accumulator (bucket)
+  accumulator_ += framesize_kbits;
+  CapAccumulator();
+}
+
+void FrameDropper::Leak(uint32_t input_framerate) {
+  if (!enabled_) {
+    return;
+  }
+  if (input_framerate < 1) {
+    return;
+  }
+  if (target_bitrate_ < 0.0f) {
+    return;
+  }
+  // Add lower bound for large frame accumulation spread.
+  large_frame_accumulation_spread_ = std::max(0.5 * input_framerate, 5.0);
+  // Expected bits per frame based on current input frame rate.
+  float expected_bits_per_frame = target_bitrate_ / input_framerate;
+  if (large_frame_accumulation_count_ > 0) {
+    expected_bits_per_frame -= large_frame_accumulation_chunk_size_;
+    --large_frame_accumulation_count_;
+  }
+  accumulator_ -= expected_bits_per_frame;
+  if (accumulator_ < 0.0f) {
+    accumulator_ = 0.0f;
+  }
+  UpdateRatio();
+}
+
+void FrameDropper::UpdateRatio() {
+  if (accumulator_ > 1.3f * accumulator_max_) {
+    // Too far above accumulator max, react faster.
+    drop_ratio_.UpdateBase(0.8f);
+  } else {
+    // Go back to normal reaction.
+    drop_ratio_.UpdateBase(0.9f);
+  }
+  if (accumulator_ > accumulator_max_) {
+    // We are above accumulator max, and should ideally drop a frame. Increase
+    // the drop_ratio_ and drop the frame later.
+    if (was_below_max_) {
+      drop_next_ = true;
+    }
+    drop_ratio_.Apply(1.0f, 1.0f);
+    drop_ratio_.UpdateBase(0.9f);
+  } else {
+    drop_ratio_.Apply(1.0f, 0.0f);
+  }
+  was_below_max_ = accumulator_ < accumulator_max_;
+}
+
+// This function signals when to drop frames to the caller. It makes use of the
+// drop_ratio_ to smooth out the drops over time.
+bool FrameDropper::DropFrame() {
+  if (!enabled_) {
+    return false;
+  }
+  if (drop_next_) {
+    drop_next_ = false;
+    drop_count_ = 0;
+  }
+
+  if (drop_ratio_.filtered() >= 0.5f) {  // Drops per keep
+    // Limit is the number of frames we should drop between each kept frame
+    // to keep our drop ratio. limit is positive in this case.
+    float denom = 1.0f - drop_ratio_.filtered();
+    if (denom < 1e-5) {
+      denom = 1e-5f;
+    }
+    int32_t limit = static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
+    // Put a bound on the max amount of dropped frames between each kept
+    // frame, in terms of frame rate and window size (secs).
+    int max_limit =
+        static_cast<int>(incoming_frame_rate_ * max_drop_duration_secs_);
+    if (limit > max_limit) {
+      limit = max_limit;
+    }
+    if (drop_count_ < 0) {
+      // Reset the drop_count_ since it was negative and should be positive.
+      drop_count_ = -drop_count_;
+    }
+    if (drop_count_ < limit) {
+      // As long we are below the limit we should drop frames.
+      drop_count_++;
+      return true;
+    } else {
+      // Only when we reset drop_count_ a frame should be kept.
+      drop_count_ = 0;
+      return false;
+    }
+  } else if (drop_ratio_.filtered() > 0.0f &&
+             drop_ratio_.filtered() < 0.5f) {  // Keeps per drop
+    // Limit is the number of frames we should keep between each drop
+    // in order to keep the drop ratio. limit is negative in this case,
+    // and the drop_count_ is also negative.
+    float denom = drop_ratio_.filtered();
+    if (denom < 1e-5) {
+      denom = 1e-5f;
+    }
+    int32_t limit = -static_cast<int32_t>(1.0f / denom - 1.0f + 0.5f);
+    if (drop_count_ > 0) {
+      // Reset the drop_count_ since we have a positive
+      // drop_count_, and it should be negative.
+      drop_count_ = -drop_count_;
+    }
+    if (drop_count_ > limit) {
+      if (drop_count_ == 0) {
+        // Drop frames when we reset drop_count_.
+        drop_count_--;
+        return true;
+      } else {
+        // Keep frames as long as we haven't reached limit.
+        drop_count_--;
+        return false;
+      }
+    } else {
+      drop_count_ = 0;
+      return false;
+    }
+  }
+  drop_count_ = 0;
+  return false;
+}
+
+void FrameDropper::SetRates(float bitrate, float incoming_frame_rate) {
+  // Bit rate of -1 means infinite bandwidth.
+  accumulator_max_ = bitrate * kLeakyBucketSizeSeconds;
+  if (target_bitrate_ > 0.0f && bitrate < target_bitrate_ &&
+      accumulator_ > accumulator_max_) {
+    // Rescale the accumulator level if the accumulator max decreases
+    accumulator_ = bitrate / target_bitrate_ * accumulator_;
+  }
+  target_bitrate_ = bitrate;
+  CapAccumulator();
+  incoming_frame_rate_ = incoming_frame_rate;
+}
+
+// Put a cap on the accumulator, i.e., don't let it grow beyond some level.
+// This is a temporary fix for screencasting where very large frames from
+// encoder will cause very slow response (too many frame drops).
+// TODO(isheriff): Remove this now that large delta frames are also spread out ?
+void FrameDropper::CapAccumulator() {
+  float max_accumulator = target_bitrate_ * kAccumulatorCapBufferSizeSecs;
+  if (accumulator_ > max_accumulator) {
+    accumulator_ = max_accumulator;
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/frame_dropper.h b/third_party/libwebrtc/modules/video_coding/utility/frame_dropper.h
new file mode 100644
index 0000000000..b45b7fe27f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/frame_dropper.h
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
+#define MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "rtc_base/numerics/exp_filter.h"
+
+namespace webrtc {
+
+// The Frame Dropper implements a variant of the leaky bucket algorithm
+// for keeping track of when to drop frames to avoid bit rate
+// over use when the encoder can't keep its bit rate.
+class FrameDropper {
+ public:
+  FrameDropper();
+  ~FrameDropper();
+
+  // Resets the FrameDropper to its initial state.
+  void Reset();
+
+  void Enable(bool enable);
+
+  // Answers the question if it's time to drop a frame if we want to reach a
+  // given frame rate. Must be called for every frame.
+  //
+  // Return value     : True if we should drop the current frame.
+  bool DropFrame();
+
+  // Updates the FrameDropper with the size of the latest encoded frame.
+  // The FrameDropper calculates a new drop ratio (can be seen as the
+  // probability to drop a frame) and updates its internal statistics.
+  //
+  // Input:
+  //          - framesize_bytes    : The size of the latest frame returned
+  //                                 from the encoder.
+  //          - delta_frame        : True if the encoder returned a delta frame.
+  void Fill(size_t framesize_bytes, bool delta_frame);
+
+  void Leak(uint32_t input_framerate);
+
+  // Sets the target bit rate and the frame rate produced by the camera.
+  //
+  // Input:
+  //          - bitrate       : The target bit rate.
+  void SetRates(float bitrate, float incoming_frame_rate);
+
+ private:
+  void UpdateRatio();
+  void CapAccumulator();
+
+  rtc::ExpFilter key_frame_ratio_;
+  rtc::ExpFilter delta_frame_size_avg_kbits_;
+
+  // Key frames and large delta frames are not immediately accumulated in the
+  // bucket since they can immediately overflow the bucket leading to large
+  // drops on the following packets that may be much smaller. Instead these
+  // large frames are accumulated over several frames when the bucket leaks.
+
+  // `large_frame_accumulation_spread_` represents the number of frames over
+  // which a large frame is accumulated.
+  float large_frame_accumulation_spread_;
+  // `large_frame_accumulation_count_` represents the number of frames left
+  // to finish accumulating a large frame.
+  int large_frame_accumulation_count_;
+  // `large_frame_accumulation_chunk_size_` represents the size of a single
+  // chunk for large frame accumulation.
+  float large_frame_accumulation_chunk_size_;
+
+  float accumulator_;
+  float accumulator_max_;
+  float target_bitrate_;
+  bool drop_next_;
+  rtc::ExpFilter drop_ratio_;
+  int drop_count_;
+  float incoming_frame_rate_;
+  bool was_below_max_;
+  bool enabled_;
+  const float max_drop_duration_secs_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/frame_dropper_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/frame_dropper_unittest.cc
new file mode 100644
index 0000000000..066103a788
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/frame_dropper_unittest.cc
@@ -0,0 +1,160 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/frame_dropper.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+const float kTargetBitRateKbps = 300;
+const float kIncomingFrameRate = 30;
+const size_t kFrameSizeBytes = 1250;
+
+const size_t kLargeFrameSizeBytes = 25000;
+
+const bool kIncludeKeyFrame = true;
+const bool kDoNotIncludeKeyFrame = false;
+
+}  // namespace
+
+class FrameDropperTest : public ::testing::Test {
+ protected:
+  void SetUp() override {
+    frame_dropper_.SetRates(kTargetBitRateKbps, kIncomingFrameRate);
+  }
+
+  void OverflowLeakyBucket() {
+    // Overflow bucket in frame dropper.
+    for (int i = 0; i < kIncomingFrameRate; ++i) {
+      frame_dropper_.Fill(kFrameSizeBytes, true);
+    }
+    frame_dropper_.Leak(kIncomingFrameRate);
+  }
+
+  void ValidateNoDropsAtTargetBitrate(int large_frame_size_bytes,
+                                      int large_frame_rate,
+                                      bool is_large_frame_delta) {
+    // Smaller frame size is computed to meet `kTargetBitRateKbps`.
+    int small_frame_size_bytes =
+        kFrameSizeBytes -
+        (large_frame_size_bytes * large_frame_rate) / kIncomingFrameRate;
+
+    for (int i = 1; i <= 5 * large_frame_rate; ++i) {
+      // Large frame. First frame is always a key frame.
+      frame_dropper_.Fill(large_frame_size_bytes,
+                          (i == 1) ? false : is_large_frame_delta);
+      frame_dropper_.Leak(kIncomingFrameRate);
+      EXPECT_FALSE(frame_dropper_.DropFrame());
+
+      // Smaller frames.
+      for (int j = 1; j < kIncomingFrameRate / large_frame_rate; ++j) {
+        frame_dropper_.Fill(small_frame_size_bytes, true);
+        frame_dropper_.Leak(kIncomingFrameRate);
+        EXPECT_FALSE(frame_dropper_.DropFrame());
+      }
+    }
+  }
+
+  void ValidateThroughputMatchesTargetBitrate(int bitrate_kbps,
+                                              bool include_keyframe) {
+    int delta_frame_size;
+    int total_bytes = 0;
+
+    if (include_keyframe) {
+      delta_frame_size = ((1000.0 / 8 * bitrate_kbps) - kLargeFrameSizeBytes) /
+                         (kIncomingFrameRate - 1);
+    } else {
+      delta_frame_size = bitrate_kbps * 1000.0 / (8 * kIncomingFrameRate);
+    }
+    const int kNumIterations = 1000;
+    for (int i = 1; i <= kNumIterations; ++i) {
+      int j = 0;
+      if (include_keyframe) {
+        if (!frame_dropper_.DropFrame()) {
+          frame_dropper_.Fill(kLargeFrameSizeBytes, false);
+          total_bytes += kLargeFrameSizeBytes;
+        }
+        frame_dropper_.Leak(kIncomingFrameRate);
+        j++;
+      }
+      for (; j < kIncomingFrameRate; ++j) {
+        if (!frame_dropper_.DropFrame()) {
+          frame_dropper_.Fill(delta_frame_size, true);
+          total_bytes += delta_frame_size;
+        }
+        frame_dropper_.Leak(kIncomingFrameRate);
+      }
+    }
+    float throughput_kbps = total_bytes * 8.0 / (1000 * kNumIterations);
+    float deviation_from_target =
+        (throughput_kbps - kTargetBitRateKbps) * 100.0 / kTargetBitRateKbps;
+    if (deviation_from_target < 0) {
+      deviation_from_target = -deviation_from_target;
+    }
+
+    // Variation is < 0.1%
+    EXPECT_LE(deviation_from_target, 0.1);
+  }
+
+  FrameDropper frame_dropper_;
+};
+
+TEST_F(FrameDropperTest, NoDropsWhenDisabled) {
+  frame_dropper_.Enable(false);
+  OverflowLeakyBucket();
+  EXPECT_FALSE(frame_dropper_.DropFrame());
+}
+
+TEST_F(FrameDropperTest, DropsByDefaultWhenBucketOverflows) {
+  OverflowLeakyBucket();
+  EXPECT_TRUE(frame_dropper_.DropFrame());
+}
+
+TEST_F(FrameDropperTest, NoDropsWhenFillRateMatchesLeakRate) {
+  for (int i = 0; i < 5 * kIncomingFrameRate; ++i) {
+    frame_dropper_.Fill(kFrameSizeBytes, true);
+    frame_dropper_.Leak(kIncomingFrameRate);
+    EXPECT_FALSE(frame_dropper_.DropFrame());
+  }
+}
+
+TEST_F(FrameDropperTest, LargeKeyFrames) {
+  ValidateNoDropsAtTargetBitrate(kLargeFrameSizeBytes, 1, false);
+  frame_dropper_.Reset();
+  ValidateNoDropsAtTargetBitrate(kLargeFrameSizeBytes / 2, 2, false);
+  frame_dropper_.Reset();
+  ValidateNoDropsAtTargetBitrate(kLargeFrameSizeBytes / 4, 4, false);
+  frame_dropper_.Reset();
+  ValidateNoDropsAtTargetBitrate(kLargeFrameSizeBytes / 8, 8, false);
+}
+
+TEST_F(FrameDropperTest, LargeDeltaFrames) {
+  ValidateNoDropsAtTargetBitrate(kLargeFrameSizeBytes, 1, true);
+  frame_dropper_.Reset();
+  ValidateNoDropsAtTargetBitrate(kLargeFrameSizeBytes / 2, 2, true);
+  frame_dropper_.Reset();
+  ValidateNoDropsAtTargetBitrate(kLargeFrameSizeBytes / 4, 4, true);
+  frame_dropper_.Reset();
+  ValidateNoDropsAtTargetBitrate(kLargeFrameSizeBytes / 8, 8, true);
+}
+
+TEST_F(FrameDropperTest, TrafficVolumeAboveAvailableBandwidth) {
+  ValidateThroughputMatchesTargetBitrate(700, kIncludeKeyFrame);
+  ValidateThroughputMatchesTargetBitrate(700, kDoNotIncludeKeyFrame);
+  ValidateThroughputMatchesTargetBitrate(600, kIncludeKeyFrame);
+  ValidateThroughputMatchesTargetBitrate(600, kDoNotIncludeKeyFrame);
+  ValidateThroughputMatchesTargetBitrate(500, kIncludeKeyFrame);
+  ValidateThroughputMatchesTargetBitrate(500, kDoNotIncludeKeyFrame);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated.cc b/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated.cc
new file mode 100644
index 0000000000..5978adc3c4
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated.cc
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/framerate_controller_deprecated.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+
+namespace webrtc {
+
+FramerateControllerDeprecated::FramerateControllerDeprecated(
+    float target_framerate_fps)
+    : min_frame_interval_ms_(0), framerate_estimator_(1000.0, 1000.0) {
+  SetTargetRate(target_framerate_fps);
+}
+
+void FramerateControllerDeprecated::SetTargetRate(float target_framerate_fps) {
+  if (target_framerate_fps_ != target_framerate_fps) {
+    framerate_estimator_.Reset();
+    if (last_timestamp_ms_) {
+      framerate_estimator_.Update(1, *last_timestamp_ms_);
+    }
+
+    const size_t target_frame_interval_ms = 1000 / target_framerate_fps;
+    target_framerate_fps_ = target_framerate_fps;
+    min_frame_interval_ms_ = 85 * target_frame_interval_ms / 100;
+  }
+}
+
+float FramerateControllerDeprecated::GetTargetRate() {
+  return *target_framerate_fps_;
+}
+
+void FramerateControllerDeprecated::Reset() {
+  framerate_estimator_.Reset();
+  last_timestamp_ms_.reset();
+}
+
+bool FramerateControllerDeprecated::DropFrame(uint32_t timestamp_ms) const {
+  if (timestamp_ms < last_timestamp_ms_) {
+    // Timestamp jumps backward. We can't make adequate drop decision. Don't
+    // drop this frame. Stats will be reset in AddFrame().
+    return false;
+  }
+
+  if (Rate(timestamp_ms).value_or(*target_framerate_fps_) >
+      target_framerate_fps_) {
+    return true;
+  }
+
+  if (last_timestamp_ms_) {
+    const int64_t diff_ms =
+        static_cast<int64_t>(timestamp_ms) - *last_timestamp_ms_;
+    if (diff_ms < min_frame_interval_ms_) {
+      return true;
+    }
+  }
+
+  return false;
+}
+
+void FramerateControllerDeprecated::AddFrame(uint32_t timestamp_ms) {
+  if (timestamp_ms < last_timestamp_ms_) {
+    // Timestamp jumps backward.
+    Reset();
+  }
+
+  framerate_estimator_.Update(1, timestamp_ms);
+  last_timestamp_ms_ = timestamp_ms;
+}
+
+absl::optional<float> FramerateControllerDeprecated::Rate(
+    uint32_t timestamp_ms) const {
+  return framerate_estimator_.Rate(timestamp_ms);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated.h b/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated.h
new file mode 100644
index 0000000000..ca0cbea053
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_FRAMERATE_CONTROLLER_DEPRECATED_H_
+#define MODULES_VIDEO_CODING_UTILITY_FRAMERATE_CONTROLLER_DEPRECATED_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "rtc_base/rate_statistics.h"
+
+namespace webrtc {
+
+// Please use webrtc::FramerateController instead.
+class FramerateControllerDeprecated {
+ public:
+  explicit FramerateControllerDeprecated(float target_framerate_fps);
+
+  void SetTargetRate(float target_framerate_fps);
+  float GetTargetRate();
+
+  // Advices user to drop next frame in order to reach target framerate.
+  bool DropFrame(uint32_t timestamp_ms) const;
+
+  void AddFrame(uint32_t timestamp_ms);
+
+  void Reset();
+
+ private:
+  absl::optional<float> Rate(uint32_t timestamp_ms) const;
+
+  absl::optional<float> target_framerate_fps_;
+  absl::optional<uint32_t> last_timestamp_ms_;
+  uint32_t min_frame_interval_ms_;
+  RateStatistics framerate_estimator_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_FRAMERATE_CONTROLLER_DEPRECATED_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated_unittest.cc
new file mode 100644
index 0000000000..eabf0529db
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated_unittest.cc
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/framerate_controller_deprecated.h"
+
+#include <stddef.h>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(FramerateControllerDeprecated, KeepTargetFramerate) {
+  const float input_framerate_fps = 20;
+  const float target_framerate_fps = 5;
+  const float max_abs_framerate_error_fps = target_framerate_fps * 0.1f;
+  const size_t input_duration_secs = 3;
+  const size_t num_input_frames = input_duration_secs * input_framerate_fps;
+
+  FramerateControllerDeprecated framerate_controller(target_framerate_fps);
+  size_t num_dropped_frames = 0;
+  for (size_t frame_num = 0; frame_num < num_input_frames; ++frame_num) {
+    const uint32_t timestamp_ms =
+        static_cast<uint32_t>(1000 * frame_num / input_framerate_fps);
+    if (framerate_controller.DropFrame(timestamp_ms)) {
+      ++num_dropped_frames;
+    } else {
+      framerate_controller.AddFrame(timestamp_ms);
+    }
+  }
+
+  const float output_framerate_fps =
+      static_cast<float>(num_input_frames - num_dropped_frames) /
+      input_duration_secs;
+  EXPECT_NEAR(output_framerate_fps, target_framerate_fps,
+              max_abs_framerate_error_fps);
+}
+
+TEST(FramerateControllerDeprecated, DoNotDropAnyFramesIfTargerEqualsInput) {
+  const float input_framerate_fps = 30;
+  const size_t input_duration_secs = 3;
+  const size_t num_input_frames = input_duration_secs * input_framerate_fps;
+
+  FramerateControllerDeprecated framerate_controller(input_framerate_fps);
+  size_t num_dropped_frames = 0;
+  for (size_t frame_num = 0; frame_num < num_input_frames; ++frame_num) {
+    const uint32_t timestamp_ms =
+        static_cast<uint32_t>(1000 * frame_num / input_framerate_fps);
+    if (framerate_controller.DropFrame(timestamp_ms)) {
+      ++num_dropped_frames;
+    } else {
+      framerate_controller.AddFrame(timestamp_ms);
+    }
+  }
+
+  EXPECT_EQ(num_dropped_frames, 0U);
+}
+
+TEST(FramerateControllerDeprecated, DoNotDropFrameWhenTimestampJumpsBackward) {
+  FramerateControllerDeprecated framerate_controller(30);
+  ASSERT_FALSE(framerate_controller.DropFrame(66));
+  framerate_controller.AddFrame(66);
+  EXPECT_FALSE(framerate_controller.DropFrame(33));
+}
+
+TEST(FramerateControllerDeprecated, DropFrameIfItIsTooCloseToPreviousFrame) {
+  FramerateControllerDeprecated framerate_controller(30);
+  ASSERT_FALSE(framerate_controller.DropFrame(33));
+  framerate_controller.AddFrame(33);
+  EXPECT_TRUE(framerate_controller.DropFrame(34));
+}
+
+TEST(FramerateControllerDeprecated, FrameDroppingStartsFromSecondInputFrame) {
+  const float input_framerate_fps = 23;
+  const float target_framerate_fps = 19;
+  const uint32_t input_frame_duration_ms =
+      static_cast<uint32_t>(1000 / input_framerate_fps);
+  FramerateControllerDeprecated framerate_controller(target_framerate_fps);
+  ASSERT_FALSE(framerate_controller.DropFrame(1 * input_frame_duration_ms));
+  framerate_controller.AddFrame(1 * input_frame_duration_ms);
+  EXPECT_TRUE(framerate_controller.DropFrame(2 * input_frame_duration_ms));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/ivf_defines.h b/third_party/libwebrtc/modules/video_coding/utility/ivf_defines.h
new file mode 100644
index 0000000000..212d381e70
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/ivf_defines.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * This file contains definitions that are common to the IvfFileReader and
+ * IvfFileWriter classes.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_IVF_DEFINES_H_
+#define MODULES_VIDEO_CODING_UTILITY_IVF_DEFINES_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+constexpr size_t kIvfHeaderSize = 32;
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_IVF_DEFINES_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader.cc b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader.cc
new file mode 100644
index 0000000000..13092b5e24
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader.cc
@@ -0,0 +1,238 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/ivf_file_reader.h"
+
+#include <string>
+#include <vector>
+
+#include "api/video_codecs/video_codec.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/video_coding/utility/ivf_defines.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kIvfFrameHeaderSize = 12;
+constexpr int kCodecTypeBytesCount = 4;
+
+constexpr uint8_t kFileHeaderStart[kCodecTypeBytesCount] = {'D', 'K', 'I', 'F'};
+constexpr uint8_t kVp8Header[kCodecTypeBytesCount] = {'V', 'P', '8', '0'};
+constexpr uint8_t kVp9Header[kCodecTypeBytesCount] = {'V', 'P', '9', '0'};
+constexpr uint8_t kAv1Header[kCodecTypeBytesCount] = {'A', 'V', '0', '1'};
+constexpr uint8_t kH264Header[kCodecTypeBytesCount] = {'H', '2', '6', '4'};
+
+// RTP standard required 90kHz clock rate.
+constexpr int32_t kRtpClockRateHz = 90000;
+
+}  // namespace
+
+std::unique_ptr<IvfFileReader> IvfFileReader::Create(FileWrapper file) {
+  auto reader =
+      std::unique_ptr<IvfFileReader>(new IvfFileReader(std::move(file)));
+  if (!reader->Reset()) {
+    return nullptr;
+  }
+  return reader;
+}
+IvfFileReader::~IvfFileReader() {
+  Close();
+}
+
+bool IvfFileReader::Reset() {
+  // Set error to true while initialization.
+  has_error_ = true;
+  if (!file_.Rewind()) {
+    RTC_LOG(LS_ERROR) << "Failed to rewind IVF file";
+    return false;
+  }
+
+  uint8_t ivf_header[kIvfHeaderSize] = {0};
+  size_t read = file_.Read(&ivf_header, kIvfHeaderSize);
+  if (read != kIvfHeaderSize) {
+    RTC_LOG(LS_ERROR) << "Failed to read IVF header";
+    return false;
+  }
+
+  if (memcmp(&ivf_header[0], kFileHeaderStart, 4) != 0) {
+    RTC_LOG(LS_ERROR) << "File is not in IVF format: DKIF header expected";
+    return false;
+  }
+
+  absl::optional<VideoCodecType> codec_type = ParseCodecType(ivf_header, 8);
+  if (!codec_type) {
+    return false;
+  }
+  codec_type_ = *codec_type;
+
+  width_ = ByteReader<uint16_t>::ReadLittleEndian(&ivf_header[12]);
+  height_ = ByteReader<uint16_t>::ReadLittleEndian(&ivf_header[14]);
+  if (width_ == 0 || height_ == 0) {
+    RTC_LOG(LS_ERROR) << "Invalid IVF header: width or height is 0";
+    return false;
+  }
+
+  time_scale_ = ByteReader<uint32_t>::ReadLittleEndian(&ivf_header[16]);
+  if (time_scale_ == 0) {
+    RTC_LOG(LS_ERROR) << "Invalid IVF header: time scale can't be 0";
+    return false;
+  }
+
+  num_frames_ = static_cast<size_t>(
+      ByteReader<uint32_t>::ReadLittleEndian(&ivf_header[24]));
+  if (num_frames_ <= 0) {
+    RTC_LOG(LS_ERROR) << "Invalid IVF header: number of frames 0 or negative";
+    return false;
+  }
+
+  num_read_frames_ = 0;
+  next_frame_header_ = ReadNextFrameHeader();
+  if (!next_frame_header_) {
+    RTC_LOG(LS_ERROR) << "Failed to read 1st frame header";
+    return false;
+  }
+  // Initialization succeed: reset error.
+  has_error_ = false;
+
+  const char* codec_name = CodecTypeToPayloadString(codec_type_);
+  RTC_LOG(LS_INFO) << "Opened IVF file with codec data of type " << codec_name
+                   << " at resolution " << width_ << " x " << height_
+                   << ", using " << time_scale_ << "Hz clock resolution.";
+
+  return true;
+}
+
+absl::optional<EncodedImage> IvfFileReader::NextFrame() {
+  if (has_error_ || !HasMoreFrames()) {
+    return absl::nullopt;
+  }
+
+  rtc::scoped_refptr<EncodedImageBuffer> payload = EncodedImageBuffer::Create();
+  std::vector<size_t> layer_sizes;
+  // next_frame_header_ have to be presented by the way how it was loaded. If it
+  // is missing it means there is a bug in error handling.
+  RTC_DCHECK(next_frame_header_);
+  int64_t current_timestamp = next_frame_header_->timestamp;
+  // The first frame from the file should be marked as Key frame.
+  bool is_first_frame = num_read_frames_ == 0;
+  while (next_frame_header_ &&
+         current_timestamp == next_frame_header_->timestamp) {
+    // Resize payload to fit next spatial layer.
+    size_t current_layer_size = next_frame_header_->frame_size;
+    size_t current_layer_start_pos = payload->size();
+    payload->Realloc(payload->size() + current_layer_size);
+    layer_sizes.push_back(current_layer_size);
+
+    // Read next layer into payload
+    size_t read = file_.Read(&payload->data()[current_layer_start_pos],
+                             current_layer_size);
+    if (read != current_layer_size) {
+      RTC_LOG(LS_ERROR) << "Frame #" << num_read_frames_
+                        << ": failed to read frame payload";
+      has_error_ = true;
+      return absl::nullopt;
+    }
+    num_read_frames_++;
+
+    current_timestamp = next_frame_header_->timestamp;
+    next_frame_header_ = ReadNextFrameHeader();
+  }
+  if (!next_frame_header_) {
+    // If EOF was reached, we need to check that all frames were met.
+    if (!has_error_ && num_read_frames_ != num_frames_) {
+      RTC_LOG(LS_ERROR) << "Unexpected EOF";
+      has_error_ = true;
+      return absl::nullopt;
+    }
+  }
+
+  EncodedImage image;
+  image.capture_time_ms_ = current_timestamp;
+  image.SetTimestamp(
+      static_cast<uint32_t>(current_timestamp * kRtpClockRateHz / time_scale_));
+  image.SetEncodedData(payload);
+  image.SetSpatialIndex(static_cast<int>(layer_sizes.size()) - 1);
+  for (size_t i = 0; i < layer_sizes.size(); ++i) {
+    image.SetSpatialLayerFrameSize(static_cast<int>(i), layer_sizes[i]);
+  }
+  if (is_first_frame) {
+    image._frameType = VideoFrameType::kVideoFrameKey;
+  }
+
+  return image;
+}
+
+bool IvfFileReader::Close() {
+  if (!file_.is_open())
+    return false;
+
+  file_.Close();
+  return true;
+}
+
+absl::optional<VideoCodecType> IvfFileReader::ParseCodecType(uint8_t* buffer,
+                                                             size_t start_pos) {
+  if (memcmp(&buffer[start_pos], kVp8Header, kCodecTypeBytesCount) == 0) {
+    return VideoCodecType::kVideoCodecVP8;
+  }
+  if (memcmp(&buffer[start_pos], kVp9Header, kCodecTypeBytesCount) == 0) {
+    return VideoCodecType::kVideoCodecVP9;
+  }
+  if (memcmp(&buffer[start_pos], kAv1Header, kCodecTypeBytesCount) == 0) {
+    return VideoCodecType::kVideoCodecAV1;
+  }
+  if (memcmp(&buffer[start_pos], kH264Header, kCodecTypeBytesCount) == 0) {
+    return VideoCodecType::kVideoCodecH264;
+  }
+  has_error_ = true;
+  RTC_LOG(LS_ERROR) << "Unknown codec type: "
+                    << std::string(
+                           reinterpret_cast<char const*>(&buffer[start_pos]),
+                           kCodecTypeBytesCount);
+  return absl::nullopt;
+}
+
+absl::optional<IvfFileReader::FrameHeader>
+IvfFileReader::ReadNextFrameHeader() {
+  uint8_t ivf_frame_header[kIvfFrameHeaderSize] = {0};
+  size_t read = file_.Read(&ivf_frame_header, kIvfFrameHeaderSize);
+  if (read != kIvfFrameHeaderSize) {
+    if (read != 0 || !file_.ReadEof()) {
+      has_error_ = true;
+      RTC_LOG(LS_ERROR) << "Frame #" << num_read_frames_
+                        << ": failed to read IVF frame header";
+    }
+    return absl::nullopt;
+  }
+  FrameHeader header;
+  header.frame_size = static_cast<size_t>(
+      ByteReader<uint32_t>::ReadLittleEndian(&ivf_frame_header[0]));
+  header.timestamp =
+      ByteReader<uint64_t>::ReadLittleEndian(&ivf_frame_header[4]);
+
+  if (header.frame_size == 0) {
+    has_error_ = true;
+    RTC_LOG(LS_ERROR) << "Frame #" << num_read_frames_
+                      << ": invalid frame size";
+    return absl::nullopt;
+  }
+
+  if (header.timestamp < 0) {
+    has_error_ = true;
+    RTC_LOG(LS_ERROR) << "Frame #" << num_read_frames_
+                      << ": negative timestamp";
+    return absl::nullopt;
+  }
+
+  return header;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader.h b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader.h
new file mode 100644
index 0000000000..db4fc25575
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_IVF_FILE_READER_H_
+#define MODULES_VIDEO_CODING_UTILITY_IVF_FILE_READER_H_
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_codec.h"
+#include "rtc_base/system/file_wrapper.h"
+
+namespace webrtc {
+
+class IvfFileReader {
+ public:
+  // Creates IvfFileReader. Returns nullptr if error acquired.
+  static std::unique_ptr<IvfFileReader> Create(FileWrapper file);
+  ~IvfFileReader();
+
+  IvfFileReader(const IvfFileReader&) = delete;
+  IvfFileReader& operator=(const IvfFileReader&) = delete;
+
+  // Reinitializes reader. Returns false if any error acquired.
+  bool Reset();
+
+  // Returns codec type which was used to create this IVF file and which should
+  // be used to decode EncodedImages from this file.
+  VideoCodecType GetVideoCodecType() const { return codec_type_; }
+  // Returns count of frames in this file.
+  size_t GetFramesCount() const { return num_frames_; }
+
+  // Returns next frame or absl::nullopt if any error acquired. Always returns
+  // absl::nullopt after first error was spotted.
+  absl::optional<EncodedImage> NextFrame();
+  bool HasMoreFrames() const { return num_read_frames_ < num_frames_; }
+  bool HasError() const { return has_error_; }
+
+  uint16_t GetFrameWidth() const { return width_; }
+  uint16_t GetFrameHeight() const { return height_; }
+
+  bool Close();
+
+ private:
+  struct FrameHeader {
+    size_t frame_size;
+    int64_t timestamp;
+  };
+
+  explicit IvfFileReader(FileWrapper file) : file_(std::move(file)) {}
+
+  // Parses codec type from specified position of the buffer. Codec type
+  // contains kCodecTypeBytesCount bytes and caller has to ensure that buffer
+  // won't overflow.
+  absl::optional<VideoCodecType> ParseCodecType(uint8_t* buffer,
+                                                size_t start_pos);
+  absl::optional<FrameHeader> ReadNextFrameHeader();
+
+  VideoCodecType codec_type_;
+  size_t num_frames_;
+  size_t num_read_frames_;
+  uint16_t width_;
+  uint16_t height_;
+  uint32_t time_scale_;
+  FileWrapper file_;
+
+  absl::optional<FrameHeader> next_frame_header_;
+  bool has_error_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_IVF_FILE_READER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader_unittest.cc
new file mode 100644
index 0000000000..c9cf14674b
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader_unittest.cc
@@ -0,0 +1,188 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/ivf_file_reader.h"
+#include "modules/video_coding/utility/ivf_file_writer.h"
+
+#include <memory>
+#include <string>
+
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kWidth = 320;
+constexpr int kHeight = 240;
+constexpr int kNumFrames = 3;
+constexpr uint8_t kDummyPayload[4] = {'0', '1', '2', '3'};
+
+}  // namespace
+
+class IvfFileReaderTest : public ::testing::Test {
+ protected:
+  void SetUp() override {
+    file_name_ =
+        webrtc::test::TempFilename(webrtc::test::OutputPath(), "test_file.ivf");
+  }
+  void TearDown() override { webrtc::test::RemoveFile(file_name_); }
+
+  bool WriteDummyTestFrames(IvfFileWriter* file_writer,
+                            VideoCodecType codec_type,
+                            int width,
+                            int height,
+                            int num_frames,
+                            bool use_capture_tims_ms,
+                            int spatial_layers_count) {
+    EncodedImage frame;
+    frame.SetSpatialIndex(spatial_layers_count);
+    rtc::scoped_refptr<EncodedImageBuffer> payload = EncodedImageBuffer::Create(
+        sizeof(kDummyPayload) * spatial_layers_count);
+    for (int i = 0; i < spatial_layers_count; ++i) {
+      memcpy(&payload->data()[i * sizeof(kDummyPayload)], kDummyPayload,
+             sizeof(kDummyPayload));
+      frame.SetSpatialLayerFrameSize(i, sizeof(kDummyPayload));
+    }
+    frame.SetEncodedData(payload);
+    frame._encodedWidth = width;
+    frame._encodedHeight = height;
+    for (int i = 1; i <= num_frames; ++i) {
+      if (use_capture_tims_ms) {
+        frame.capture_time_ms_ = i;
+      } else {
+        frame.SetTimestamp(i);
+      }
+      if (!file_writer->WriteFrame(frame, codec_type))
+        return false;
+    }
+    return true;
+  }
+
+  void CreateTestFile(VideoCodecType codec_type,
+                      bool use_capture_tims_ms,
+                      int spatial_layers_count) {
+    std::unique_ptr<IvfFileWriter> file_writer =
+        IvfFileWriter::Wrap(FileWrapper::OpenWriteOnly(file_name_), 0);
+    ASSERT_TRUE(file_writer.get());
+    ASSERT_TRUE(WriteDummyTestFrames(file_writer.get(), codec_type, kWidth,
+                                     kHeight, kNumFrames, use_capture_tims_ms,
+                                     spatial_layers_count));
+    ASSERT_TRUE(file_writer->Close());
+  }
+
+  void ValidateFrame(absl::optional<EncodedImage> frame,
+                     int frame_index,
+                     bool use_capture_tims_ms,
+                     int spatial_layers_count) {
+    ASSERT_TRUE(frame);
+    EXPECT_EQ(frame->SpatialIndex(), spatial_layers_count - 1);
+    if (use_capture_tims_ms) {
+      EXPECT_EQ(frame->capture_time_ms_, static_cast<int64_t>(frame_index));
+      EXPECT_EQ(frame->Timestamp(), static_cast<int64_t>(90 * frame_index));
+    } else {
+      EXPECT_EQ(frame->Timestamp(), static_cast<int64_t>(frame_index));
+    }
+    ASSERT_EQ(frame->size(), sizeof(kDummyPayload) * spatial_layers_count);
+    for (int i = 0; i < spatial_layers_count; ++i) {
+      EXPECT_EQ(memcmp(&frame->data()[i * sizeof(kDummyPayload)], kDummyPayload,
+                       sizeof(kDummyPayload)),
+                0)
+          << std::string(reinterpret_cast<char const*>(
+                             &frame->data()[i * sizeof(kDummyPayload)]),
+                         sizeof(kDummyPayload));
+    }
+  }
+
+  void ValidateContent(VideoCodecType codec_type,
+                       bool use_capture_tims_ms,
+                       int spatial_layers_count) {
+    std::unique_ptr<IvfFileReader> reader =
+        IvfFileReader::Create(FileWrapper::OpenReadOnly(file_name_));
+    ASSERT_TRUE(reader.get());
+    EXPECT_EQ(reader->GetVideoCodecType(), codec_type);
+    EXPECT_EQ(reader->GetFramesCount(),
+              spatial_layers_count * static_cast<size_t>(kNumFrames));
+    for (int i = 1; i <= kNumFrames; ++i) {
+      ASSERT_TRUE(reader->HasMoreFrames());
+      ValidateFrame(reader->NextFrame(), i, use_capture_tims_ms,
+                    spatial_layers_count);
+      EXPECT_FALSE(reader->HasError());
+    }
+    EXPECT_FALSE(reader->HasMoreFrames());
+    EXPECT_FALSE(reader->NextFrame());
+    EXPECT_FALSE(reader->HasError());
+    ASSERT_TRUE(reader->Close());
+  }
+
+  std::string file_name_;
+};
+
+TEST_F(IvfFileReaderTest, BasicVp8FileNtpTimestamp) {
+  CreateTestFile(kVideoCodecVP8, false, 1);
+  ValidateContent(kVideoCodecVP8, false, 1);
+}
+
+TEST_F(IvfFileReaderTest, BasicVP8FileMsTimestamp) {
+  CreateTestFile(kVideoCodecVP8, true, 1);
+  ValidateContent(kVideoCodecVP8, true, 1);
+}
+
+TEST_F(IvfFileReaderTest, BasicVP9FileNtpTimestamp) {
+  CreateTestFile(kVideoCodecVP9, false, 1);
+  ValidateContent(kVideoCodecVP9, false, 1);
+}
+
+TEST_F(IvfFileReaderTest, BasicVP9FileMsTimestamp) {
+  CreateTestFile(kVideoCodecVP9, true, 1);
+  ValidateContent(kVideoCodecVP9, true, 1);
+}
+
+TEST_F(IvfFileReaderTest, BasicAv1FileNtpTimestamp) {
+  CreateTestFile(kVideoCodecAV1, false, 1);
+  ValidateContent(kVideoCodecAV1, false, 1);
+}
+
+TEST_F(IvfFileReaderTest, BasicAv1FileMsTimestamp) {
+  CreateTestFile(kVideoCodecAV1, true, 1);
+  ValidateContent(kVideoCodecAV1, true, 1);
+}
+
+TEST_F(IvfFileReaderTest, BasicH264FileNtpTimestamp) {
+  CreateTestFile(kVideoCodecH264, false, 1);
+  ValidateContent(kVideoCodecH264, false, 1);
+}
+
+TEST_F(IvfFileReaderTest, BasicH264FileMsTimestamp) {
+  CreateTestFile(kVideoCodecH264, true, 1);
+  ValidateContent(kVideoCodecH264, true, 1);
+}
+
+TEST_F(IvfFileReaderTest, MultilayerVp8FileNtpTimestamp) {
+  CreateTestFile(kVideoCodecVP8, false, 3);
+  ValidateContent(kVideoCodecVP8, false, 3);
+}
+
+TEST_F(IvfFileReaderTest, MultilayerVP9FileNtpTimestamp) {
+  CreateTestFile(kVideoCodecVP9, false, 3);
+  ValidateContent(kVideoCodecVP9, false, 3);
+}
+
+TEST_F(IvfFileReaderTest, MultilayerAv1FileNtpTimestamp) {
+  CreateTestFile(kVideoCodecAV1, false, 3);
+  ValidateContent(kVideoCodecAV1, false, 3);
+}
+
+TEST_F(IvfFileReaderTest, MultilayerH264FileNtpTimestamp) {
+  CreateTestFile(kVideoCodecH264, false, 3);
+  ValidateContent(kVideoCodecH264, false, 3);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer.cc b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer.cc
new file mode 100644
index 0000000000..5b27ef3ef7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer.cc
@@ -0,0 +1,245 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/ivf_file_writer.h"
+
+#include <utility>
+
+#include "api/video_codecs/video_codec.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/video_coding/utility/ivf_defines.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+// TODO(palmkvist): make logging more informative in the absence of a file name
+// (or get one)
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDefaultWidth = 1280;
+constexpr int kDefaultHeight = 720;
+}  // namespace
+
+IvfFileWriter::IvfFileWriter(FileWrapper file, size_t byte_limit)
+    : codec_type_(kVideoCodecGeneric),
+      bytes_written_(0),
+      byte_limit_(byte_limit),
+      num_frames_(0),
+      width_(0),
+      height_(0),
+      last_timestamp_(-1),
+      using_capture_timestamps_(false),
+      file_(std::move(file)) {
+  RTC_DCHECK(byte_limit == 0 || webrtc::kIvfHeaderSize <= byte_limit)
+      << "The byte_limit is too low, not even the header will fit.";
+}
+
+IvfFileWriter::~IvfFileWriter() {
+  Close();
+}
+
+std::unique_ptr<IvfFileWriter> IvfFileWriter::Wrap(FileWrapper file,
+                                                   size_t byte_limit) {
+  return std::unique_ptr<IvfFileWriter>(
+      new IvfFileWriter(std::move(file), byte_limit));
+}
+
+bool IvfFileWriter::WriteHeader() {
+  if (!file_.Rewind()) {
+    RTC_LOG(LS_WARNING) << "Unable to rewind ivf output file.";
+    return false;
+  }
+
+  uint8_t ivf_header[webrtc::kIvfHeaderSize] = {0};
+  ivf_header[0] = 'D';
+  ivf_header[1] = 'K';
+  ivf_header[2] = 'I';
+  ivf_header[3] = 'F';
+  ByteWriter<uint16_t>::WriteLittleEndian(&ivf_header[4], 0);   // Version.
+  ByteWriter<uint16_t>::WriteLittleEndian(&ivf_header[6], 32);  // Header size.
+
+  switch (codec_type_) {
+    case kVideoCodecVP8:
+      ivf_header[8] = 'V';
+      ivf_header[9] = 'P';
+      ivf_header[10] = '8';
+      ivf_header[11] = '0';
+      break;
+    case kVideoCodecVP9:
+      ivf_header[8] = 'V';
+      ivf_header[9] = 'P';
+      ivf_header[10] = '9';
+      ivf_header[11] = '0';
+      break;
+    case kVideoCodecAV1:
+      ivf_header[8] = 'A';
+      ivf_header[9] = 'V';
+      ivf_header[10] = '0';
+      ivf_header[11] = '1';
+      break;
+    case kVideoCodecH264:
+      ivf_header[8] = 'H';
+      ivf_header[9] = '2';
+      ivf_header[10] = '6';
+      ivf_header[11] = '4';
+      break;
+    default:
+      // For unknown codec type use **** code. You can specify actual payload
+      // format when playing the video with ffplay: ffplay -f H263 file.ivf
+      ivf_header[8] = '*';
+      ivf_header[9] = '*';
+      ivf_header[10] = '*';
+      ivf_header[11] = '*';
+      break;
+  }
+
+  ByteWriter<uint16_t>::WriteLittleEndian(&ivf_header[12], width_);
+  ByteWriter<uint16_t>::WriteLittleEndian(&ivf_header[14], height_);
+  // Render timestamps are in ms (1/1000 scale), while RTP timestamps use a
+  // 90kHz clock.
+  ByteWriter<uint32_t>::WriteLittleEndian(
+      &ivf_header[16], using_capture_timestamps_ ? 1000 : 90000);
+  ByteWriter<uint32_t>::WriteLittleEndian(&ivf_header[20], 1);
+  ByteWriter<uint32_t>::WriteLittleEndian(&ivf_header[24],
+                                          static_cast<uint32_t>(num_frames_));
+  ByteWriter<uint32_t>::WriteLittleEndian(&ivf_header[28], 0);  // Reserved.
+
+  if (!file_.Write(ivf_header, webrtc::kIvfHeaderSize)) {
+    RTC_LOG(LS_ERROR) << "Unable to write IVF header for ivf output file.";
+    return false;
+  }
+
+  if (bytes_written_ < webrtc::kIvfHeaderSize) {
+    bytes_written_ = webrtc::kIvfHeaderSize;
+  }
+
+  return true;
+}
+
+bool IvfFileWriter::InitFromFirstFrame(const EncodedImage& encoded_image,
+                                       VideoCodecType codec_type) {
+  if (encoded_image._encodedWidth == 0 || encoded_image._encodedHeight == 0) {
+    width_ = kDefaultWidth;
+    height_ = kDefaultHeight;
+  } else {
+    width_ = encoded_image._encodedWidth;
+    height_ = encoded_image._encodedHeight;
+  }
+
+  using_capture_timestamps_ = encoded_image.Timestamp() == 0;
+
+  codec_type_ = codec_type;
+
+  if (!WriteHeader())
+    return false;
+
+  const char* codec_name = CodecTypeToPayloadString(codec_type_);
+  RTC_LOG(LS_WARNING) << "Created IVF file for codec data of type "
+                      << codec_name << " at resolution " << width_ << " x "
+                      << height_ << ", using "
+                      << (using_capture_timestamps_ ? "1" : "90")
+                      << "kHz clock resolution.";
+  return true;
+}
+
+bool IvfFileWriter::WriteFrame(const EncodedImage& encoded_image,
+                               VideoCodecType codec_type) {
+  if (!file_.is_open())
+    return false;
+
+  if (num_frames_ == 0 && !InitFromFirstFrame(encoded_image, codec_type))
+    return false;
+  RTC_DCHECK_EQ(codec_type_, codec_type);
+
+  if ((encoded_image._encodedWidth > 0 || encoded_image._encodedHeight > 0) &&
+      (encoded_image._encodedHeight != height_ ||
+       encoded_image._encodedWidth != width_)) {
+    RTC_LOG(LS_WARNING)
+        << "Incoming frame has resolution different from previous: (" << width_
+        << "x" << height_ << ") -> (" << encoded_image._encodedWidth << "x"
+        << encoded_image._encodedHeight << ")";
+  }
+
+  int64_t timestamp = using_capture_timestamps_
+                          ? encoded_image.capture_time_ms_
+                          : wrap_handler_.Unwrap(encoded_image.Timestamp());
+  if (last_timestamp_ != -1 && timestamp <= last_timestamp_) {
+    RTC_LOG(LS_WARNING) << "Timestamp no increasing: " << last_timestamp_
+                        << " -> " << timestamp;
+  }
+  last_timestamp_ = timestamp;
+
+  bool written_frames = false;
+  size_t max_sl_index = encoded_image.SpatialIndex().value_or(0);
+  const uint8_t* data = encoded_image.data();
+  for (size_t sl_idx = 0; sl_idx <= max_sl_index; ++sl_idx) {
+    size_t cur_size = encoded_image.SpatialLayerFrameSize(sl_idx).value_or(0);
+    if (cur_size > 0) {
+      written_frames = true;
+      if (!WriteOneSpatialLayer(timestamp, data, cur_size)) {
+        return false;
+      }
+      data += cur_size;
+    }
+  }
+
+  // If frame has only one spatial layer it won't have any spatial layers'
+  // sizes. Therefore this case should be addressed separately.
+  if (!written_frames) {
+    return WriteOneSpatialLayer(timestamp, data, encoded_image.size());
+  } else {
+    return true;
+  }
+}
+
+bool IvfFileWriter::WriteOneSpatialLayer(int64_t timestamp,
+                                         const uint8_t* data,
+                                         size_t size) {
+  const size_t kFrameHeaderSize = 12;
+  if (byte_limit_ != 0 &&
+      bytes_written_ + kFrameHeaderSize + size > byte_limit_) {
+    RTC_LOG(LS_WARNING) << "Closing IVF file due to reaching size limit: "
+                        << byte_limit_ << " bytes.";
+    Close();
+    return false;
+  }
+  uint8_t frame_header[kFrameHeaderSize] = {};
+  ByteWriter<uint32_t>::WriteLittleEndian(&frame_header[0],
+                                          static_cast<uint32_t>(size));
+  ByteWriter<uint64_t>::WriteLittleEndian(&frame_header[4], timestamp);
+  if (!file_.Write(frame_header, kFrameHeaderSize) ||
+      !file_.Write(data, size)) {
+    RTC_LOG(LS_ERROR) << "Unable to write frame to file.";
+    return false;
+  }
+
+  bytes_written_ += kFrameHeaderSize + size;
+
+  ++num_frames_;
+  return true;
+}
+
+bool IvfFileWriter::Close() {
+  if (!file_.is_open())
+    return false;
+
+  if (num_frames_ == 0) {
+    file_.Close();
+    return true;
+  }
+
+  bool ret = WriteHeader();
+  file_.Close();
+  return ret;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer.h b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer.h
new file mode 100644
index 0000000000..ec8a7bf9e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_IVF_FILE_WRITER_H_
+#define MODULES_VIDEO_CODING_UTILITY_IVF_FILE_WRITER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/video/encoded_image.h"
+#include "api/video/video_codec_type.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/system/file_wrapper.h"
+
+namespace webrtc {
+
+class IvfFileWriter {
+ public:
+  // Takes ownership of the file, which will be closed either through
+  // Close or ~IvfFileWriter. If writing a frame would take the file above the
+  // `byte_limit` the file will be closed, the write (and all future writes)
+  // will fail. A `byte_limit` of 0 is equivalent to no limit.
+  static std::unique_ptr<IvfFileWriter> Wrap(FileWrapper file,
+                                             size_t byte_limit);
+  ~IvfFileWriter();
+
+  IvfFileWriter(const IvfFileWriter&) = delete;
+  IvfFileWriter& operator=(const IvfFileWriter&) = delete;
+
+  bool WriteFrame(const EncodedImage& encoded_image, VideoCodecType codec_type);
+  bool Close();
+
+ private:
+  explicit IvfFileWriter(FileWrapper file, size_t byte_limit);
+
+  bool WriteHeader();
+  bool InitFromFirstFrame(const EncodedImage& encoded_image,
+                          VideoCodecType codec_type);
+  bool WriteOneSpatialLayer(int64_t timestamp,
+                            const uint8_t* data,
+                            size_t size);
+
+  VideoCodecType codec_type_;
+  size_t bytes_written_;
+  size_t byte_limit_;
+  size_t num_frames_;
+  uint16_t width_;
+  uint16_t height_;
+  int64_t last_timestamp_;
+  bool using_capture_timestamps_;
+  RtpTimestampUnwrapper wrap_handler_;
+  FileWrapper file_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_IVF_FILE_WRITER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer_unittest.cc
new file mode 100644
index 0000000000..c5d30a1286
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer_unittest.cc
@@ -0,0 +1,311 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/ivf_file_writer.h"
+
+#include <string.h>
+
+#include <memory>
+#include <string>
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+static const int kHeaderSize = 32;
+static const int kFrameHeaderSize = 12;
+static uint8_t dummy_payload[4] = {0, 1, 2, 3};
+// As the default parameter when the width and height of encodedImage are 0,
+// the values are copied from ivf_file_writer.cc
+constexpr int kDefaultWidth = 1280;
+constexpr int kDefaultHeight = 720;
+}  // namespace
+
+class IvfFileWriterTest : public ::testing::Test {
+ protected:
+  void SetUp() override {
+    file_name_ =
+        webrtc::test::TempFilename(webrtc::test::OutputPath(), "test_file");
+  }
+  void TearDown() override { webrtc::test::RemoveFile(file_name_); }
+
+  bool WriteDummyTestFrames(VideoCodecType codec_type,
+                            int width,
+                            int height,
+                            int num_frames,
+                            bool use_capture_tims_ms) {
+    EncodedImage frame;
+    frame.SetEncodedData(
+        EncodedImageBuffer::Create(dummy_payload, sizeof(dummy_payload)));
+    frame._encodedWidth = width;
+    frame._encodedHeight = height;
+    for (int i = 1; i <= num_frames; ++i) {
+      frame.set_size(i % sizeof(dummy_payload));
+      if (use_capture_tims_ms) {
+        frame.capture_time_ms_ = i;
+      } else {
+        frame.SetTimestamp(i);
+      }
+      if (!file_writer_->WriteFrame(frame, codec_type))
+        return false;
+    }
+    return true;
+  }
+
+  void VerifyIvfHeader(FileWrapper* file,
+                       const uint8_t fourcc[4],
+                       int width,
+                       int height,
+                       uint32_t num_frames,
+                       bool use_capture_tims_ms) {
+    ASSERT_TRUE(file->is_open());
+    uint8_t data[kHeaderSize];
+    ASSERT_EQ(static_cast<size_t>(kHeaderSize), file->Read(data, kHeaderSize));
+
+    uint8_t dkif[4] = {'D', 'K', 'I', 'F'};
+    EXPECT_EQ(0, memcmp(dkif, data, 4));
+    EXPECT_EQ(0u, ByteReader<uint16_t>::ReadLittleEndian(&data[4]));
+    EXPECT_EQ(32u, ByteReader<uint16_t>::ReadLittleEndian(&data[6]));
+    EXPECT_EQ(0, memcmp(fourcc, &data[8], 4));
+    EXPECT_EQ(width, ByteReader<uint16_t>::ReadLittleEndian(&data[12]));
+    EXPECT_EQ(height, ByteReader<uint16_t>::ReadLittleEndian(&data[14]));
+    EXPECT_EQ(use_capture_tims_ms ? 1000u : 90000u,
+              ByteReader<uint32_t>::ReadLittleEndian(&data[16]));
+    EXPECT_EQ(1u, ByteReader<uint32_t>::ReadLittleEndian(&data[20]));
+    EXPECT_EQ(num_frames, ByteReader<uint32_t>::ReadLittleEndian(&data[24]));
+    EXPECT_EQ(0u, ByteReader<uint32_t>::ReadLittleEndian(&data[28]));
+  }
+
+  void VerifyDummyTestFrames(FileWrapper* file, uint32_t num_frames) {
+    const int kMaxFrameSize = 4;
+    for (uint32_t i = 1; i <= num_frames; ++i) {
+      uint8_t frame_header[kFrameHeaderSize];
+      ASSERT_EQ(static_cast<unsigned int>(kFrameHeaderSize),
+                file->Read(frame_header, kFrameHeaderSize));
+      uint32_t frame_length =
+          ByteReader<uint32_t>::ReadLittleEndian(&frame_header[0]);
+      EXPECT_EQ(i % 4, frame_length);
+      uint64_t timestamp =
+          ByteReader<uint64_t>::ReadLittleEndian(&frame_header[4]);
+      EXPECT_EQ(i, timestamp);
+
+      uint8_t data[kMaxFrameSize] = {};
+      ASSERT_EQ(frame_length,
+                static_cast<uint32_t>(file->Read(data, frame_length)));
+      EXPECT_EQ(0, memcmp(data, dummy_payload, frame_length));
+    }
+  }
+
+  void RunBasicFileStructureTest(VideoCodecType codec_type,
+                                 const uint8_t fourcc[4],
+                                 bool use_capture_tims_ms) {
+    file_writer_ =
+        IvfFileWriter::Wrap(FileWrapper::OpenWriteOnly(file_name_), 0);
+    ASSERT_TRUE(file_writer_.get());
+    const int kWidth = 320;
+    const int kHeight = 240;
+    const int kNumFrames = 257;
+    ASSERT_TRUE(WriteDummyTestFrames(codec_type, kWidth, kHeight, kNumFrames,
+                                     use_capture_tims_ms));
+    EXPECT_TRUE(file_writer_->Close());
+
+    FileWrapper out_file = FileWrapper::OpenReadOnly(file_name_);
+    VerifyIvfHeader(&out_file, fourcc, kWidth, kHeight, kNumFrames,
+                    use_capture_tims_ms);
+    VerifyDummyTestFrames(&out_file, kNumFrames);
+
+    out_file.Close();
+  }
+
+  std::string file_name_;
+  std::unique_ptr<IvfFileWriter> file_writer_;
+};
+
+TEST_F(IvfFileWriterTest, WritesBasicVP8FileNtpTimestamp) {
+  const uint8_t fourcc[4] = {'V', 'P', '8', '0'};
+  RunBasicFileStructureTest(kVideoCodecVP8, fourcc, false);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicVP8FileMsTimestamp) {
+  const uint8_t fourcc[4] = {'V', 'P', '8', '0'};
+  RunBasicFileStructureTest(kVideoCodecVP8, fourcc, true);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicVP9FileNtpTimestamp) {
+  const uint8_t fourcc[4] = {'V', 'P', '9', '0'};
+  RunBasicFileStructureTest(kVideoCodecVP9, fourcc, false);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicVP9FileMsTimestamp) {
+  const uint8_t fourcc[4] = {'V', 'P', '9', '0'};
+  RunBasicFileStructureTest(kVideoCodecVP9, fourcc, true);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicAv1FileNtpTimestamp) {
+  const uint8_t fourcc[4] = {'A', 'V', '0', '1'};
+  RunBasicFileStructureTest(kVideoCodecAV1, fourcc, false);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicAv1FileMsTimestamp) {
+  const uint8_t fourcc[4] = {'A', 'V', '0', '1'};
+  RunBasicFileStructureTest(kVideoCodecAV1, fourcc, true);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicH264FileNtpTimestamp) {
+  const uint8_t fourcc[4] = {'H', '2', '6', '4'};
+  RunBasicFileStructureTest(kVideoCodecH264, fourcc, false);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicH264FileMsTimestamp) {
+  const uint8_t fourcc[4] = {'H', '2', '6', '4'};
+  RunBasicFileStructureTest(kVideoCodecH264, fourcc, true);
+}
+
+TEST_F(IvfFileWriterTest, WritesBasicUnknownCodecFileMsTimestamp) {
+  const uint8_t fourcc[4] = {'*', '*', '*', '*'};
+  RunBasicFileStructureTest(kVideoCodecGeneric, fourcc, true);
+}
+
+TEST_F(IvfFileWriterTest, ClosesWhenReachesLimit) {
+  const uint8_t fourcc[4] = {'V', 'P', '8', '0'};
+  const int kWidth = 320;
+  const int kHeight = 240;
+  const int kNumFramesToWrite = 2;
+  const int kNumFramesToFit = 1;
+
+  file_writer_ = IvfFileWriter::Wrap(
+      FileWrapper::OpenWriteOnly(file_name_),
+      kHeaderSize +
+          kNumFramesToFit * (kFrameHeaderSize + sizeof(dummy_payload)));
+  ASSERT_TRUE(file_writer_.get());
+
+  ASSERT_FALSE(WriteDummyTestFrames(kVideoCodecVP8, kWidth, kHeight,
+                                    kNumFramesToWrite, true));
+  ASSERT_FALSE(file_writer_->Close());
+
+  FileWrapper out_file = FileWrapper::OpenReadOnly(file_name_);
+  VerifyIvfHeader(&out_file, fourcc, kWidth, kHeight, kNumFramesToFit, true);
+  VerifyDummyTestFrames(&out_file, kNumFramesToFit);
+
+  out_file.Close();
+}
+
+TEST_F(IvfFileWriterTest, UseDefaultValueWhenWidthAndHeightAreZero) {
+  const uint8_t fourcc[4] = {'V', 'P', '8', '0'};
+  const int kWidth = 0;
+  const int kHeight = 0;
+  const int kNumFramesToWrite = 2;
+  const int kNumFramesToFit = 1;
+
+  file_writer_ = IvfFileWriter::Wrap(
+      FileWrapper::OpenWriteOnly(file_name_),
+      kHeaderSize +
+          kNumFramesToFit * (kFrameHeaderSize + sizeof(dummy_payload)));
+  ASSERT_TRUE(file_writer_.get());
+
+  ASSERT_FALSE(WriteDummyTestFrames(kVideoCodecVP8, kWidth, kHeight,
+                                    kNumFramesToWrite, true));
+  ASSERT_FALSE(file_writer_->Close());
+
+  FileWrapper out_file = FileWrapper::OpenReadOnly(file_name_);
+  // When the width and height are zero, we should expect the width and height
+  // in IvfHeader to be kDefaultWidth and kDefaultHeight instead of kWidth and
+  // kHeight.
+  VerifyIvfHeader(&out_file, fourcc, kDefaultWidth, kDefaultHeight,
+                  kNumFramesToFit, true);
+  VerifyDummyTestFrames(&out_file, kNumFramesToFit);
+
+  out_file.Close();
+}
+
+TEST_F(IvfFileWriterTest, UseDefaultValueWhenOnlyWidthIsZero) {
+  const uint8_t fourcc[4] = {'V', 'P', '8', '0'};
+  const int kWidth = 0;
+  const int kHeight = 360;
+  const int kNumFramesToWrite = 2;
+  const int kNumFramesToFit = 1;
+
+  file_writer_ = IvfFileWriter::Wrap(
+      FileWrapper::OpenWriteOnly(file_name_),
+      kHeaderSize +
+          kNumFramesToFit * (kFrameHeaderSize + sizeof(dummy_payload)));
+  ASSERT_TRUE(file_writer_.get());
+
+  ASSERT_FALSE(WriteDummyTestFrames(kVideoCodecVP8, kWidth, kHeight,
+                                    kNumFramesToWrite, true));
+  ASSERT_FALSE(file_writer_->Close());
+
+  FileWrapper out_file = FileWrapper::OpenReadOnly(file_name_);
+  // When the width and height are zero, we should expect the width and height
+  // in IvfHeader to be kDefaultWidth and kDefaultHeight instead of kWidth and
+  // kHeight.
+  VerifyIvfHeader(&out_file, fourcc, kDefaultWidth, kDefaultHeight,
+                  kNumFramesToFit, true);
+  VerifyDummyTestFrames(&out_file, kNumFramesToFit);
+
+  out_file.Close();
+}
+
+TEST_F(IvfFileWriterTest, UseDefaultValueWhenOnlyHeightIsZero) {
+  const uint8_t fourcc[4] = {'V', 'P', '8', '0'};
+  const int kWidth = 240;
+  const int kHeight = 0;
+  const int kNumFramesToWrite = 2;
+  const int kNumFramesToFit = 1;
+
+  file_writer_ = IvfFileWriter::Wrap(
+      FileWrapper::OpenWriteOnly(file_name_),
+      kHeaderSize +
+          kNumFramesToFit * (kFrameHeaderSize + sizeof(dummy_payload)));
+  ASSERT_TRUE(file_writer_.get());
+
+  ASSERT_FALSE(WriteDummyTestFrames(kVideoCodecVP8, kWidth, kHeight,
+                                    kNumFramesToWrite, true));
+  ASSERT_FALSE(file_writer_->Close());
+
+  FileWrapper out_file = FileWrapper::OpenReadOnly(file_name_);
+  // When the width and height are zero, we should expect the width and height
+  // in IvfHeader to be kDefaultWidth and kDefaultHeight instead of kWidth and
+  // kHeight.
+  VerifyIvfHeader(&out_file, fourcc, kDefaultWidth, kDefaultHeight,
+                  kNumFramesToFit, true);
+  VerifyDummyTestFrames(&out_file, kNumFramesToFit);
+
+  out_file.Close();
+}
+
+TEST_F(IvfFileWriterTest, UseDefaultValueWhenHeightAndWidthAreNotZero) {
+  const uint8_t fourcc[4] = {'V', 'P', '8', '0'};
+  const int kWidth = 360;
+  const int kHeight = 240;
+  const int kNumFramesToWrite = 2;
+  const int kNumFramesToFit = 1;
+
+  file_writer_ = IvfFileWriter::Wrap(
+      FileWrapper::OpenWriteOnly(file_name_),
+      kHeaderSize +
+          kNumFramesToFit * (kFrameHeaderSize + sizeof(dummy_payload)));
+  ASSERT_TRUE(file_writer_.get());
+
+  ASSERT_FALSE(WriteDummyTestFrames(kVideoCodecVP8, kWidth, kHeight,
+                                    kNumFramesToWrite, true));
+  ASSERT_FALSE(file_writer_->Close());
+
+  FileWrapper out_file = FileWrapper::OpenReadOnly(file_name_);
+  VerifyIvfHeader(&out_file, fourcc, kWidth, kHeight, kNumFramesToFit, true);
+  VerifyDummyTestFrames(&out_file, kNumFramesToFit);
+
+  out_file.Close();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/qp_parser.cc b/third_party/libwebrtc/modules/video_coding/utility/qp_parser.cc
new file mode 100644
index 0000000000..18f225447d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/qp_parser.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/qp_parser.h"
+
+#include "modules/video_coding/utility/vp8_header_parser.h"
+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+
+namespace webrtc {
+
+absl::optional<uint32_t> QpParser::Parse(VideoCodecType codec_type,
+                                         size_t spatial_idx,
+                                         const uint8_t* frame_data,
+                                         size_t frame_size) {
+  if (frame_data == nullptr || frame_size == 0 ||
+      spatial_idx >= kMaxSimulcastStreams) {
+    return absl::nullopt;
+  }
+
+  if (codec_type == kVideoCodecVP8) {
+    int qp = -1;
+    if (vp8::GetQp(frame_data, frame_size, &qp)) {
+      return qp;
+    }
+  } else if (codec_type == kVideoCodecVP9) {
+    int qp = -1;
+    if (vp9::GetQp(frame_data, frame_size, &qp)) {
+      return qp;
+    }
+  } else if (codec_type == kVideoCodecH264) {
+    return h264_parsers_[spatial_idx].Parse(frame_data, frame_size);
+  }
+
+  return absl::nullopt;
+}
+
+absl::optional<uint32_t> QpParser::H264QpParser::Parse(
+    const uint8_t* frame_data,
+    size_t frame_size) {
+  MutexLock lock(&mutex_);
+  bitstream_parser_.ParseBitstream(
+      rtc::ArrayView<const uint8_t>(frame_data, frame_size));
+  return bitstream_parser_.GetLastSliceQp();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/qp_parser.h b/third_party/libwebrtc/modules/video_coding/utility/qp_parser.h
new file mode 100644
index 0000000000..f132ff9337
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/qp_parser.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
+#define MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
+
+#include "absl/types/optional.h"
+#include "api/video/video_codec_constants.h"
+#include "api/video/video_codec_type.h"
+#include "common_video/h264/h264_bitstream_parser.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+class QpParser {
+ public:
+  absl::optional<uint32_t> Parse(VideoCodecType codec_type,
+                                 size_t spatial_idx,
+                                 const uint8_t* frame_data,
+                                 size_t frame_size);
+
+ private:
+  // A thread safe wrapper for H264 bitstream parser.
+  class H264QpParser {
+   public:
+    absl::optional<uint32_t> Parse(const uint8_t* frame_data,
+                                   size_t frame_size);
+
+   private:
+    Mutex mutex_;
+    H264BitstreamParser bitstream_parser_ RTC_GUARDED_BY(mutex_);
+  };
+
+  H264QpParser h264_parsers_[kMaxSimulcastStreams];
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_QP_PARSER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/qp_parser_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/qp_parser_unittest.cc
new file mode 100644
index 0000000000..1131288f26
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/qp_parser_unittest.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/qp_parser.h"
+
+#include <stddef.h>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+// ffmpeg -s 16x16 -f rawvideo -pix_fmt rgb24 -r 30 -i /dev/zero -c:v libvpx
+// -qmin 20 -qmax 20 -crf 20 -frames:v 1 -y out.ivf
+const uint8_t kCodedFrameVp8Qp25[] = {
+    0x10, 0x02, 0x00, 0x9d, 0x01, 0x2a, 0x10, 0x00, 0x10, 0x00,
+    0x02, 0x47, 0x08, 0x85, 0x85, 0x88, 0x85, 0x84, 0x88, 0x0c,
+    0x82, 0x00, 0x0c, 0x0d, 0x60, 0x00, 0xfe, 0xfc, 0x5c, 0xd0};
+
+// ffmpeg -s 16x16 -f rawvideo -pix_fmt rgb24 -r 30 -i /dev/zero -c:v libvpx-vp9
+// -qmin 24 -qmax 24 -crf 24 -frames:v 1 -y out.ivf
+const uint8_t kCodedFrameVp9Qp96[] = {
+    0xa2, 0x49, 0x83, 0x42, 0xe0, 0x00, 0xf0, 0x00, 0xf6, 0x00,
+    0x38, 0x24, 0x1c, 0x18, 0xc0, 0x00, 0x00, 0x30, 0x70, 0x00,
+    0x00, 0x4a, 0xa7, 0xff, 0xfc, 0xb9, 0x01, 0xbf, 0xff, 0xff,
+    0x97, 0x20, 0xdb, 0xff, 0xff, 0xcb, 0x90, 0x5d, 0x40};
+
+// ffmpeg -s 16x16 -f rawvideo -pix_fmt yuv420p -r 30 -i /dev/zero -c:v libx264
+// -qmin 38 -qmax 38 -crf 38 -profile:v baseline -frames:v 2 -y out.264
+const uint8_t kCodedFrameH264SpsPpsIdrQp38[] = {
+    0x00, 0x00, 0x00, 0x01, 0x67, 0x42, 0xc0, 0x0a, 0xd9, 0x1e, 0x84,
+    0x00, 0x00, 0x03, 0x00, 0x04, 0x00, 0x00, 0x03, 0x00, 0xf0, 0x3c,
+    0x48, 0x99, 0x20, 0x00, 0x00, 0x00, 0x01, 0x68, 0xcb, 0x80, 0xc4,
+    0xb2, 0x00, 0x00, 0x01, 0x65, 0x88, 0x84, 0xf1, 0x18, 0xa0, 0x00,
+    0x20, 0x5b, 0x1c, 0x00, 0x04, 0x07, 0xe3, 0x80, 0x00, 0x80, 0xfe};
+
+const uint8_t kCodedFrameH264SpsPpsIdrQp49[] = {
+    0x00, 0x00, 0x00, 0x01, 0x67, 0x42, 0xc0, 0x0a, 0xd9, 0x1e, 0x84,
+    0x00, 0x00, 0x03, 0x00, 0x04, 0x00, 0x00, 0x03, 0x00, 0xf0, 0x3c,
+    0x48, 0x99, 0x20, 0x00, 0x00, 0x00, 0x01, 0x68, 0xcb, 0x80, 0x5d,
+    0x2c, 0x80, 0x00, 0x00, 0x01, 0x65, 0x88, 0x84, 0xf1, 0x18, 0xa0,
+    0x00, 0x5e, 0x38, 0x00, 0x08, 0x03, 0xc7, 0x00, 0x01, 0x00, 0x7c};
+
+const uint8_t kCodedFrameH264InterSliceQpDelta0[] = {0x00, 0x00, 0x00, 0x01,
+                                                     0x41, 0x9a, 0x39, 0xea};
+
+}  // namespace
+
+TEST(QpParserTest, ParseQpVp8) {
+  QpParser parser;
+  absl::optional<uint32_t> qp = parser.Parse(
+      kVideoCodecVP8, 0, kCodedFrameVp8Qp25, sizeof(kCodedFrameVp8Qp25));
+  EXPECT_EQ(qp, 25u);
+}
+
+TEST(QpParserTest, ParseQpVp9) {
+  QpParser parser;
+  absl::optional<uint32_t> qp = parser.Parse(
+      kVideoCodecVP9, 0, kCodedFrameVp9Qp96, sizeof(kCodedFrameVp9Qp96));
+  EXPECT_EQ(qp, 96u);
+}
+
+TEST(QpParserTest, ParseQpH264) {
+  QpParser parser;
+  absl::optional<uint32_t> qp = parser.Parse(
+      VideoCodecType::kVideoCodecH264, 0, kCodedFrameH264SpsPpsIdrQp38,
+      sizeof(kCodedFrameH264SpsPpsIdrQp38));
+  EXPECT_EQ(qp, 38u);
+
+  qp = parser.Parse(kVideoCodecH264, 1, kCodedFrameH264SpsPpsIdrQp49,
+                    sizeof(kCodedFrameH264SpsPpsIdrQp49));
+  EXPECT_EQ(qp, 49u);
+
+  qp = parser.Parse(kVideoCodecH264, 0, kCodedFrameH264InterSliceQpDelta0,
+                    sizeof(kCodedFrameH264InterSliceQpDelta0));
+  EXPECT_EQ(qp, 38u);
+
+  qp = parser.Parse(kVideoCodecH264, 1, kCodedFrameH264InterSliceQpDelta0,
+                    sizeof(kCodedFrameH264InterSliceQpDelta0));
+  EXPECT_EQ(qp, 49u);
+}
+
+TEST(QpParserTest, ParseQpUnsupportedCodecType) {
+  QpParser parser;
+  absl::optional<uint32_t> qp = parser.Parse(
+      kVideoCodecGeneric, 0, kCodedFrameVp8Qp25, sizeof(kCodedFrameVp8Qp25));
+  EXPECT_FALSE(qp.has_value());
+}
+
+TEST(QpParserTest, ParseQpNullData) {
+  QpParser parser;
+  absl::optional<uint32_t> qp = parser.Parse(kVideoCodecVP8, 0, nullptr, 100);
+  EXPECT_FALSE(qp.has_value());
+}
+
+TEST(QpParserTest, ParseQpEmptyData) {
+  QpParser parser;
+  absl::optional<uint32_t> qp =
+      parser.Parse(kVideoCodecVP8, 0, kCodedFrameVp8Qp25, 0);
+  EXPECT_FALSE(qp.has_value());
+}
+
+TEST(QpParserTest, ParseQpSpatialIdxExceedsMax) {
+  QpParser parser;
+  absl::optional<uint32_t> qp =
+      parser.Parse(kVideoCodecVP8, kMaxSimulcastStreams, kCodedFrameVp8Qp25,
+                   sizeof(kCodedFrameVp8Qp25));
+  EXPECT_FALSE(qp.has_value());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/quality_scaler.cc b/third_party/libwebrtc/modules/video_coding/utility/quality_scaler.cc
new file mode 100644
index 0000000000..9fb41a0ad7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/quality_scaler.cc
@@ -0,0 +1,334 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/quality_scaler.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/units/time_delta.h"
+#include "api/video/video_adaptation_reason.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/quality_scaler_settings.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/exp_filter.h"
+#include "rtc_base/weak_ptr.h"
+
+namespace webrtc {
+
+namespace {
+// Threshold constant used until first downscale (to permit fast rampup).
+static const int kMeasureMs = 2000;
+static const float kSamplePeriodScaleFactor = 2.5;
+static const int kFramedropPercentThreshold = 60;
+static const size_t kMinFramesNeededToScale = 2 * 30;
+
+}  // namespace
+
+class QualityScaler::QpSmoother {
+ public:
+  explicit QpSmoother(float alpha)
+      : alpha_(alpha),
+        // The initial value of last_sample_ms doesn't matter since the smoother
+        // will ignore the time delta for the first update.
+        last_sample_ms_(0),
+        smoother_(alpha) {}
+
+  absl::optional<int> GetAvg() const {
+    float value = smoother_.filtered();
+    if (value == rtc::ExpFilter::kValueUndefined) {
+      return absl::nullopt;
+    }
+    return static_cast<int>(value);
+  }
+
+  void Add(float sample, int64_t time_sent_us) {
+    int64_t now_ms = time_sent_us / 1000;
+    smoother_.Apply(static_cast<float>(now_ms - last_sample_ms_), sample);
+    last_sample_ms_ = now_ms;
+  }
+
+  void Reset() { smoother_.Reset(alpha_); }
+
+ private:
+  const float alpha_;
+  int64_t last_sample_ms_;
+  rtc::ExpFilter smoother_;
+};
+
+// The QualityScaler checks for QP periodically by queuing CheckQpTasks. The
+// task will either run to completion and trigger a new task being queued, or it
+// will be destroyed because the QualityScaler is destroyed.
+//
+// When high or low QP is reported, the task will be pending until a callback is
+// invoked. This lets the QualityScalerQpUsageHandlerInterface react to QP usage
+// asynchronously and prevents checking for QP until the stream has potentially
+// been reconfigured.
+class QualityScaler::CheckQpTask {
+ public:
+  // The result of one CheckQpTask may influence the delay of the next
+  // CheckQpTask.
+  struct Result {
+    bool observed_enough_frames = false;
+    bool qp_usage_reported = false;
+  };
+
+  CheckQpTask(QualityScaler* quality_scaler, Result previous_task_result)
+      : quality_scaler_(quality_scaler),
+        state_(State::kNotStarted),
+        previous_task_result_(previous_task_result),
+        weak_ptr_factory_(this) {}
+
+  void StartDelayedTask() {
+    RTC_DCHECK_EQ(state_, State::kNotStarted);
+    state_ = State::kCheckingQp;
+    TaskQueueBase::Current()->PostDelayedTask(
+        [this_weak_ptr = weak_ptr_factory_.GetWeakPtr(), this] {
+          if (!this_weak_ptr) {
+            // The task has been cancelled through destruction.
+            return;
+          }
+          RTC_DCHECK_EQ(state_, State::kCheckingQp);
+          RTC_DCHECK_RUN_ON(&quality_scaler_->task_checker_);
+          switch (quality_scaler_->CheckQp()) {
+            case QualityScaler::CheckQpResult::kInsufficientSamples: {
+              result_.observed_enough_frames = false;
+              // After this line, `this` may be deleted.
+              break;
+            }
+            case QualityScaler::CheckQpResult::kNormalQp: {
+              result_.observed_enough_frames = true;
+              break;
+            }
+            case QualityScaler::CheckQpResult::kHighQp: {
+              result_.observed_enough_frames = true;
+              result_.qp_usage_reported = true;
+              quality_scaler_->fast_rampup_ = false;
+              quality_scaler_->handler_->OnReportQpUsageHigh();
+              quality_scaler_->ClearSamples();
+              break;
+            }
+            case QualityScaler::CheckQpResult::kLowQp: {
+              result_.observed_enough_frames = true;
+              result_.qp_usage_reported = true;
+              quality_scaler_->handler_->OnReportQpUsageLow();
+              quality_scaler_->ClearSamples();
+              break;
+            }
+          }
+          state_ = State::kCompleted;
+          // Starting the next task deletes the pending task. After this line,
+          // `this` has been deleted.
+          quality_scaler_->StartNextCheckQpTask();
+        },
+        TimeDelta::Millis(GetCheckingQpDelayMs()));
+  }
+
+  bool HasCompletedTask() const { return state_ == State::kCompleted; }
+
+  Result result() const {
+    RTC_DCHECK(HasCompletedTask());
+    return result_;
+  }
+
+ private:
+  enum class State {
+    kNotStarted,
+    kCheckingQp,
+    kCompleted,
+  };
+
+  // Determines the sampling period of CheckQpTasks.
+  int64_t GetCheckingQpDelayMs() const {
+    RTC_DCHECK_RUN_ON(&quality_scaler_->task_checker_);
+    if (quality_scaler_->fast_rampup_) {
+      return quality_scaler_->sampling_period_ms_;
+    }
+    if (quality_scaler_->experiment_enabled_ &&
+        !previous_task_result_.observed_enough_frames) {
+      // Use half the interval while waiting for enough frames.
+      return quality_scaler_->sampling_period_ms_ / 2;
+    }
+    if (quality_scaler_->scale_factor_ &&
+        !previous_task_result_.qp_usage_reported) {
+      // Last CheckQp did not call AdaptDown/Up, possibly reduce interval.
+      return quality_scaler_->sampling_period_ms_ *
+             quality_scaler_->scale_factor_.value();
+    }
+    return quality_scaler_->sampling_period_ms_ *
+           quality_scaler_->initial_scale_factor_;
+  }
+
+  QualityScaler* const quality_scaler_;
+  State state_;
+  const Result previous_task_result_;
+  Result result_;
+
+  rtc::WeakPtrFactory<CheckQpTask> weak_ptr_factory_;
+};
+
+QualityScaler::QualityScaler(QualityScalerQpUsageHandlerInterface* handler,
+                             VideoEncoder::QpThresholds thresholds)
+    : QualityScaler(handler, thresholds, kMeasureMs) {}
+
+// Protected ctor, should not be called directly.
+QualityScaler::QualityScaler(QualityScalerQpUsageHandlerInterface* handler,
+                             VideoEncoder::QpThresholds thresholds,
+                             int64_t default_sampling_period_ms)
+    : handler_(handler),
+      thresholds_(thresholds),
+      sampling_period_ms_(QualityScalerSettings::ParseFromFieldTrials()
+                              .SamplingPeriodMs()
+                              .value_or(default_sampling_period_ms)),
+      fast_rampup_(true),
+      // Arbitrarily choose size based on 30 fps for 5 seconds.
+      average_qp_(QualityScalerSettings::ParseFromFieldTrials()
+                      .AverageQpWindow()
+                      .value_or(5 * 30)),
+      framedrop_percent_media_opt_(5 * 30),
+      framedrop_percent_all_(5 * 30),
+      experiment_enabled_(QualityScalingExperiment::Enabled()),
+      min_frames_needed_(
+          QualityScalerSettings::ParseFromFieldTrials().MinFrames().value_or(
+              kMinFramesNeededToScale)),
+      initial_scale_factor_(QualityScalerSettings::ParseFromFieldTrials()
+                                .InitialScaleFactor()
+                                .value_or(kSamplePeriodScaleFactor)),
+      scale_factor_(
+          QualityScalerSettings::ParseFromFieldTrials().ScaleFactor()) {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  if (experiment_enabled_) {
+    config_ = QualityScalingExperiment::GetConfig();
+    qp_smoother_high_.reset(new QpSmoother(config_.alpha_high));
+    qp_smoother_low_.reset(new QpSmoother(config_.alpha_low));
+  }
+  RTC_DCHECK(handler_ != nullptr);
+  StartNextCheckQpTask();
+  RTC_LOG(LS_INFO) << "QP thresholds: low: " << thresholds_.low
+                   << ", high: " << thresholds_.high;
+}
+
+QualityScaler::~QualityScaler() {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+}
+
+void QualityScaler::StartNextCheckQpTask() {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  RTC_DCHECK(!pending_qp_task_ || pending_qp_task_->HasCompletedTask())
+      << "A previous CheckQpTask has not completed yet!";
+  CheckQpTask::Result previous_task_result;
+  if (pending_qp_task_) {
+    previous_task_result = pending_qp_task_->result();
+  }
+  pending_qp_task_ = std::make_unique<CheckQpTask>(this, previous_task_result);
+  pending_qp_task_->StartDelayedTask();
+}
+
+void QualityScaler::SetQpThresholds(VideoEncoder::QpThresholds thresholds) {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  thresholds_ = thresholds;
+}
+
+void QualityScaler::ReportDroppedFrameByMediaOpt() {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  framedrop_percent_media_opt_.AddSample(100);
+  framedrop_percent_all_.AddSample(100);
+}
+
+void QualityScaler::ReportDroppedFrameByEncoder() {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  framedrop_percent_all_.AddSample(100);
+}
+
+void QualityScaler::ReportQp(int qp, int64_t time_sent_us) {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  framedrop_percent_media_opt_.AddSample(0);
+  framedrop_percent_all_.AddSample(0);
+  average_qp_.AddSample(qp);
+  if (qp_smoother_high_)
+    qp_smoother_high_->Add(qp, time_sent_us);
+  if (qp_smoother_low_)
+    qp_smoother_low_->Add(qp, time_sent_us);
+}
+
+bool QualityScaler::QpFastFilterLow() const {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  size_t num_frames = config_.use_all_drop_reasons
+                          ? framedrop_percent_all_.Size()
+                          : framedrop_percent_media_opt_.Size();
+  const size_t kMinNumFrames = 10;
+  if (num_frames < kMinNumFrames) {
+    return false;  // Wait for more frames before making a decision.
+  }
+  absl::optional<int> avg_qp_high = qp_smoother_high_
+                                        ? qp_smoother_high_->GetAvg()
+                                        : average_qp_.GetAverageRoundedDown();
+  return (avg_qp_high) ? (avg_qp_high.value() <= thresholds_.low) : false;
+}
+
+QualityScaler::CheckQpResult QualityScaler::CheckQp() const {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  // Should be set through InitEncode -> Should be set by now.
+  RTC_DCHECK_GE(thresholds_.low, 0);
+
+  // If we have not observed at least this many frames we can't make a good
+  // scaling decision.
+  const size_t frames = config_.use_all_drop_reasons
+                            ? framedrop_percent_all_.Size()
+                            : framedrop_percent_media_opt_.Size();
+  if (frames < min_frames_needed_) {
+    return CheckQpResult::kInsufficientSamples;
+  }
+
+  // Check if we should scale down due to high frame drop.
+  const absl::optional<int> drop_rate =
+      config_.use_all_drop_reasons
+          ? framedrop_percent_all_.GetAverageRoundedDown()
+          : framedrop_percent_media_opt_.GetAverageRoundedDown();
+  if (drop_rate && *drop_rate >= kFramedropPercentThreshold) {
+    RTC_LOG(LS_INFO) << "Reporting high QP, framedrop percent " << *drop_rate;
+    return CheckQpResult::kHighQp;
+  }
+
+  // Check if we should scale up or down based on QP.
+  const absl::optional<int> avg_qp_high =
+      qp_smoother_high_ ? qp_smoother_high_->GetAvg()
+                        : average_qp_.GetAverageRoundedDown();
+  const absl::optional<int> avg_qp_low =
+      qp_smoother_low_ ? qp_smoother_low_->GetAvg()
+                       : average_qp_.GetAverageRoundedDown();
+  if (avg_qp_high && avg_qp_low) {
+    RTC_LOG(LS_INFO) << "Checking average QP " << *avg_qp_high << " ("
+                     << *avg_qp_low << ").";
+    if (*avg_qp_high > thresholds_.high) {
+      return CheckQpResult::kHighQp;
+    }
+    if (*avg_qp_low <= thresholds_.low) {
+      // QP has been low. We want to try a higher resolution.
+      return CheckQpResult::kLowQp;
+    }
+  }
+  return CheckQpResult::kNormalQp;
+}
+
+void QualityScaler::ClearSamples() {
+  RTC_DCHECK_RUN_ON(&task_checker_);
+  framedrop_percent_media_opt_.Reset();
+  framedrop_percent_all_.Reset();
+  average_qp_.Reset();
+  if (qp_smoother_high_)
+    qp_smoother_high_->Reset();
+  if (qp_smoother_low_)
+    qp_smoother_low_->Reset();
+}
+
+QualityScalerQpUsageHandlerInterface::~QualityScalerQpUsageHandlerInterface() {}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/quality_scaler.h b/third_party/libwebrtc/modules/video_coding/utility/quality_scaler.h
new file mode 100644
index 0000000000..93014e36a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/quality_scaler.h
@@ -0,0 +1,120 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
+#define MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/video_codecs/video_encoder.h"
+#include "rtc_base/experiments/quality_scaling_experiment.h"
+#include "rtc_base/numerics/moving_average.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace webrtc {
+
+class QualityScalerQpUsageHandlerCallbackInterface;
+class QualityScalerQpUsageHandlerInterface;
+
+// QualityScaler runs asynchronously and monitors QP values of encoded frames.
+// It holds a reference to a QualityScalerQpUsageHandlerInterface implementation
+// to signal an overuse or underuse of QP (which indicate a desire to scale the
+// video stream down or up).
+class QualityScaler {
+ public:
+  // Construct a QualityScaler with given `thresholds` and `handler`.
+  // This starts the quality scaler periodically checking what the average QP
+  // has been recently.
+  QualityScaler(QualityScalerQpUsageHandlerInterface* handler,
+                VideoEncoder::QpThresholds thresholds);
+  virtual ~QualityScaler();
+  // Should be called each time a frame is dropped at encoding.
+  void ReportDroppedFrameByMediaOpt();
+  void ReportDroppedFrameByEncoder();
+  // Inform the QualityScaler of the last seen QP.
+  void ReportQp(int qp, int64_t time_sent_us);
+
+  void SetQpThresholds(VideoEncoder::QpThresholds thresholds);
+  bool QpFastFilterLow() const;
+
+  // The following members declared protected for testing purposes.
+ protected:
+  QualityScaler(QualityScalerQpUsageHandlerInterface* handler,
+                VideoEncoder::QpThresholds thresholds,
+                int64_t sampling_period_ms);
+
+ private:
+  class QpSmoother;
+  class CheckQpTask;
+  class CheckQpTaskHandlerCallback;
+
+  enum class CheckQpResult {
+    kInsufficientSamples,
+    kNormalQp,
+    kHighQp,
+    kLowQp,
+  };
+
+  // Starts checking for QP in a delayed task. When the resulting CheckQpTask
+  // completes, it will invoke this method again, ensuring that we always
+  // periodically check for QP. See CheckQpTask for more details. We never run
+  // more than one CheckQpTask at a time.
+  void StartNextCheckQpTask();
+
+  CheckQpResult CheckQp() const;
+  void ClearSamples();
+
+  std::unique_ptr<CheckQpTask> pending_qp_task_ RTC_GUARDED_BY(&task_checker_);
+  QualityScalerQpUsageHandlerInterface* const handler_
+      RTC_GUARDED_BY(&task_checker_);
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker task_checker_;
+
+  VideoEncoder::QpThresholds thresholds_ RTC_GUARDED_BY(&task_checker_);
+  const int64_t sampling_period_ms_;
+  bool fast_rampup_ RTC_GUARDED_BY(&task_checker_);
+  rtc::MovingAverage average_qp_ RTC_GUARDED_BY(&task_checker_);
+  rtc::MovingAverage framedrop_percent_media_opt_
+      RTC_GUARDED_BY(&task_checker_);
+  rtc::MovingAverage framedrop_percent_all_ RTC_GUARDED_BY(&task_checker_);
+
+  // Used by QualityScalingExperiment.
+  const bool experiment_enabled_;
+  QualityScalingExperiment::Config config_ RTC_GUARDED_BY(&task_checker_);
+  std::unique_ptr<QpSmoother> qp_smoother_high_ RTC_GUARDED_BY(&task_checker_);
+  std::unique_ptr<QpSmoother> qp_smoother_low_ RTC_GUARDED_BY(&task_checker_);
+
+  const size_t min_frames_needed_;
+  const double initial_scale_factor_;
+  const absl::optional<double> scale_factor_;
+};
+
+// Reacts to QP being too high or too low. For best quality, when QP is high it
+// is desired to decrease the resolution or frame rate of the stream and when QP
+// is low it is desired to increase the resolution or frame rate of the stream.
+// Whether to reconfigure the stream is ultimately up to the handler, which is
+// able to respond asynchronously.
+class QualityScalerQpUsageHandlerInterface {
+ public:
+  virtual ~QualityScalerQpUsageHandlerInterface();
+
+  virtual void OnReportQpUsageHigh() = 0;
+  virtual void OnReportQpUsageLow() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/quality_scaler_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/quality_scaler_unittest.cc
new file mode 100644
index 0000000000..c17159fb64
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/quality_scaler_unittest.cc
@@ -0,0 +1,254 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/quality_scaler.h"
+
+#include <memory>
+#include <string>
+
+#include "api/units/time_delta.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+static const int kFramerate = 30;
+static const int kLowQp = 15;
+static const int kHighQp = 40;
+static const int kMinFramesNeededToScale = 60;  // From quality_scaler.cc.
+static constexpr TimeDelta kDefaultTimeout = TimeDelta::Millis(150);
+}  // namespace
+
+class FakeQpUsageHandler : public QualityScalerQpUsageHandlerInterface {
+ public:
+  ~FakeQpUsageHandler() override = default;
+
+  // QualityScalerQpUsageHandlerInterface implementation.
+  void OnReportQpUsageHigh() override {
+    adapt_down_events_++;
+    event.Set();
+  }
+
+  void OnReportQpUsageLow() override {
+    adapt_up_events_++;
+    event.Set();
+  }
+
+  rtc::Event event;
+  int adapt_up_events_ = 0;
+  int adapt_down_events_ = 0;
+};
+
+// Pass a lower sampling period to speed up the tests.
+class QualityScalerUnderTest : public QualityScaler {
+ public:
+  explicit QualityScalerUnderTest(QualityScalerQpUsageHandlerInterface* handler,
+                                  VideoEncoder::QpThresholds thresholds)
+      : QualityScaler(handler, thresholds, 5) {}
+};
+
+class QualityScalerTest : public ::testing::Test,
+                          public ::testing::WithParamInterface<std::string> {
+ protected:
+  enum ScaleDirection {
+    kKeepScaleAboveLowQp,
+    kKeepScaleAtHighQp,
+    kScaleDown,
+    kScaleDownAboveHighQp,
+    kScaleUp
+  };
+
+  QualityScalerTest()
+      : scoped_field_trial_(GetParam()),
+        task_queue_("QualityScalerTestQueue"),
+        handler_(std::make_unique<FakeQpUsageHandler>()) {
+    task_queue_.SendTask(
+        [this] {
+          qs_ = std::unique_ptr<QualityScaler>(new QualityScalerUnderTest(
+              handler_.get(), VideoEncoder::QpThresholds(kLowQp, kHighQp)));
+        });
+  }
+
+  ~QualityScalerTest() override {
+    task_queue_.SendTask([this] { qs_ = nullptr; });
+  }
+
+  void TriggerScale(ScaleDirection scale_direction) {
+    for (int i = 0; i < kFramerate * 5; ++i) {
+      switch (scale_direction) {
+        case kKeepScaleAboveLowQp:
+          qs_->ReportQp(kLowQp + 1, 0);
+          break;
+        case kScaleUp:
+          qs_->ReportQp(kLowQp, 0);
+          break;
+        case kScaleDown:
+          qs_->ReportDroppedFrameByMediaOpt();
+          break;
+        case kKeepScaleAtHighQp:
+          qs_->ReportQp(kHighQp, 0);
+          break;
+        case kScaleDownAboveHighQp:
+          qs_->ReportQp(kHighQp + 1, 0);
+          break;
+      }
+    }
+  }
+
+  test::ScopedFieldTrials scoped_field_trial_;
+  TaskQueueForTest task_queue_;
+  std::unique_ptr<QualityScaler> qs_;
+  std::unique_ptr<FakeQpUsageHandler> handler_;
+};
+
+INSTANTIATE_TEST_SUITE_P(
+    FieldTrials,
+    QualityScalerTest,
+    ::testing::Values(
+        "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,7,8,0.9,0.99,1/",
+        "WebRTC-Video-QualityScaling/Disabled/"));
+
+TEST_P(QualityScalerTest, DownscalesAfterContinuousFramedrop) {
+  task_queue_.SendTask([this] { TriggerScale(kScaleDown); });
+  EXPECT_TRUE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(1, handler_->adapt_down_events_);
+  EXPECT_EQ(0, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, KeepsScaleAtHighQp) {
+  task_queue_.SendTask([this] { TriggerScale(kKeepScaleAtHighQp); });
+  EXPECT_FALSE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(0, handler_->adapt_down_events_);
+  EXPECT_EQ(0, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, DownscalesAboveHighQp) {
+  task_queue_.SendTask([this] { TriggerScale(kScaleDownAboveHighQp); });
+  EXPECT_TRUE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(1, handler_->adapt_down_events_);
+  EXPECT_EQ(0, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, DownscalesAfterTwoThirdsFramedrop) {
+  task_queue_.SendTask([this] {
+    for (int i = 0; i < kFramerate * 5; ++i) {
+      qs_->ReportDroppedFrameByMediaOpt();
+      qs_->ReportDroppedFrameByMediaOpt();
+      qs_->ReportQp(kHighQp, 0);
+    }
+  });
+  EXPECT_TRUE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(1, handler_->adapt_down_events_);
+  EXPECT_EQ(0, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, DoesNotDownscaleAfterHalfFramedrop) {
+  task_queue_.SendTask([this] {
+    for (int i = 0; i < kFramerate * 5; ++i) {
+      qs_->ReportDroppedFrameByMediaOpt();
+      qs_->ReportQp(kHighQp, 0);
+    }
+  });
+  EXPECT_FALSE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(0, handler_->adapt_down_events_);
+  EXPECT_EQ(0, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, DownscalesAfterTwoThirdsIfFieldTrialEnabled) {
+  const bool kDownScaleExpected =
+      GetParam().find("Enabled") != std::string::npos;
+  task_queue_.SendTask([this] {
+    for (int i = 0; i < kFramerate * 5; ++i) {
+      qs_->ReportDroppedFrameByMediaOpt();
+      qs_->ReportDroppedFrameByEncoder();
+      qs_->ReportQp(kHighQp, 0);
+    }
+  });
+  EXPECT_EQ(kDownScaleExpected, handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(kDownScaleExpected ? 1 : 0, handler_->adapt_down_events_);
+  EXPECT_EQ(0, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, KeepsScaleOnNormalQp) {
+  task_queue_.SendTask([this] { TriggerScale(kKeepScaleAboveLowQp); });
+  EXPECT_FALSE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(0, handler_->adapt_down_events_);
+  EXPECT_EQ(0, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, UpscalesAfterLowQp) {
+  task_queue_.SendTask([this] { TriggerScale(kScaleUp); });
+  EXPECT_TRUE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(0, handler_->adapt_down_events_);
+  EXPECT_EQ(1, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, ScalesDownAndBackUp) {
+  task_queue_.SendTask([this] { TriggerScale(kScaleDown); });
+  EXPECT_TRUE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(1, handler_->adapt_down_events_);
+  EXPECT_EQ(0, handler_->adapt_up_events_);
+  task_queue_.SendTask([this] { TriggerScale(kScaleUp); });
+  EXPECT_TRUE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(1, handler_->adapt_down_events_);
+  EXPECT_EQ(1, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, DoesNotScaleUntilEnoughFramesObserved) {
+  task_queue_.SendTask([this] {
+    // Not enough frames to make a decision.
+    for (int i = 0; i < kMinFramesNeededToScale - 1; ++i) {
+      qs_->ReportQp(kLowQp, 0);
+    }
+  });
+  EXPECT_FALSE(handler_->event.Wait(kDefaultTimeout));
+  task_queue_.SendTask([this] {
+    // Send 1 more. Enough frames observed, should result in an adapt
+    // request.
+    qs_->ReportQp(kLowQp, 0);
+  });
+  EXPECT_TRUE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(0, handler_->adapt_down_events_);
+  EXPECT_EQ(1, handler_->adapt_up_events_);
+
+  // Samples should be cleared after an adapt request.
+  task_queue_.SendTask([this] {
+    // Not enough frames to make a decision.
+    qs_->ReportQp(kLowQp, 0);
+  });
+  EXPECT_FALSE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(0, handler_->adapt_down_events_);
+  EXPECT_EQ(1, handler_->adapt_up_events_);
+}
+
+TEST_P(QualityScalerTest, ScalesDownAndBackUpWithMinFramesNeeded) {
+  task_queue_.SendTask([this] {
+    for (int i = 0; i < kMinFramesNeededToScale; ++i) {
+      qs_->ReportQp(kHighQp + 1, 0);
+    }
+  });
+  EXPECT_TRUE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(1, handler_->adapt_down_events_);
+  EXPECT_EQ(0, handler_->adapt_up_events_);
+  // Samples cleared.
+  task_queue_.SendTask([this] {
+    for (int i = 0; i < kMinFramesNeededToScale; ++i) {
+      qs_->ReportQp(kLowQp, 0);
+    }
+  });
+  EXPECT_TRUE(handler_->event.Wait(kDefaultTimeout));
+  EXPECT_EQ(1, handler_->adapt_down_events_);
+  EXPECT_EQ(1, handler_->adapt_up_events_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator.cc b/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator.cc
new file mode 100644
index 0000000000..1496934e1c
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator.cc
@@ -0,0 +1,343 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <cmath>
+#include <cstdint>
+#include <numeric>
+#include <string>
+#include <tuple>
+#include <vector>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+// Ratio allocation between temporal streams:
+// Values as required for the VP8 codec (accumulating).
+static const float
+    kLayerRateAllocation[kMaxTemporalStreams][kMaxTemporalStreams] = {
+        {1.0f, 1.0f, 1.0f, 1.0f},  // 1 layer
+        {0.6f, 1.0f, 1.0f, 1.0f},  // 2 layers {60%, 40%}
+        {0.4f, 0.6f, 1.0f, 1.0f},  // 3 layers {40%, 20%, 40%}
+        {0.25f, 0.4f, 0.6f, 1.0f}  // 4 layers {25%, 15%, 20%, 40%}
+};
+
+static const float kBaseHeavy3TlRateAllocation[kMaxTemporalStreams] = {
+    0.6f, 0.8f, 1.0f, 1.0f  // 3 layers {60%, 20%, 20%}
+};
+
+const uint32_t kLegacyScreenshareTl0BitrateKbps = 200;
+const uint32_t kLegacyScreenshareTl1BitrateKbps = 1000;
+}  // namespace
+
+float SimulcastRateAllocator::GetTemporalRateAllocation(
+    int num_layers,
+    int temporal_id,
+    bool base_heavy_tl3_alloc) {
+  RTC_CHECK_GT(num_layers, 0);
+  RTC_CHECK_LE(num_layers, kMaxTemporalStreams);
+  RTC_CHECK_GE(temporal_id, 0);
+  RTC_CHECK_LT(temporal_id, num_layers);
+  if (num_layers == 3 && base_heavy_tl3_alloc) {
+    return kBaseHeavy3TlRateAllocation[temporal_id];
+  }
+  return kLayerRateAllocation[num_layers - 1][temporal_id];
+}
+
+SimulcastRateAllocator::SimulcastRateAllocator(const VideoCodec& codec)
+    : codec_(codec),
+      stable_rate_settings_(StableTargetRateExperiment::ParseFromFieldTrials()),
+      rate_control_settings_(RateControlSettings::ParseFromFieldTrials()),
+      legacy_conference_mode_(false) {}
+
+SimulcastRateAllocator::~SimulcastRateAllocator() = default;
+
+VideoBitrateAllocation SimulcastRateAllocator::Allocate(
+    VideoBitrateAllocationParameters parameters) {
+  VideoBitrateAllocation allocated_bitrates;
+  DataRate stable_rate = parameters.total_bitrate;
+  if (stable_rate_settings_.IsEnabled() &&
+      parameters.stable_bitrate > DataRate::Zero()) {
+    stable_rate = std::min(parameters.stable_bitrate, parameters.total_bitrate);
+  }
+  DistributeAllocationToSimulcastLayers(parameters.total_bitrate, stable_rate,
+                                        &allocated_bitrates);
+  DistributeAllocationToTemporalLayers(&allocated_bitrates);
+  return allocated_bitrates;
+}
+
+void SimulcastRateAllocator::DistributeAllocationToSimulcastLayers(
+    DataRate total_bitrate,
+    DataRate stable_bitrate,
+    VideoBitrateAllocation* allocated_bitrates) {
+  DataRate left_in_total_allocation = total_bitrate;
+  DataRate left_in_stable_allocation = stable_bitrate;
+
+  if (codec_.maxBitrate) {
+    DataRate max_rate = DataRate::KilobitsPerSec(codec_.maxBitrate);
+    left_in_total_allocation = std::min(left_in_total_allocation, max_rate);
+    left_in_stable_allocation = std::min(left_in_stable_allocation, max_rate);
+  }
+
+  if (codec_.numberOfSimulcastStreams == 0) {
+    // No simulcast, just set the target as this has been capped already.
+    if (codec_.active) {
+      allocated_bitrates->SetBitrate(
+          0, 0,
+          std::max(DataRate::KilobitsPerSec(codec_.minBitrate),
+                   left_in_total_allocation)
+              .bps());
+    }
+    return;
+  }
+
+  // Sort the layers by maxFramerate, they might not always be from smallest
+  // to biggest
+  std::vector<size_t> layer_index(codec_.numberOfSimulcastStreams);
+  std::iota(layer_index.begin(), layer_index.end(), 0);
+  std::stable_sort(layer_index.begin(), layer_index.end(),
+                   [this](size_t a, size_t b) {
+                     return std::tie(codec_.simulcastStream[a].maxBitrate) <
+                            std::tie(codec_.simulcastStream[b].maxBitrate);
+                   });
+
+  // Find the first active layer. We don't allocate to inactive layers.
+  size_t active_layer = 0;
+  for (; active_layer < codec_.numberOfSimulcastStreams; ++active_layer) {
+    if (codec_.simulcastStream[layer_index[active_layer]].active) {
+      // Found the first active layer.
+      break;
+    }
+  }
+  // All streams could be inactive, and nothing more to do.
+  if (active_layer == codec_.numberOfSimulcastStreams) {
+    return;
+  }
+
+  // Always allocate enough bitrate for the minimum bitrate of the first
+  // active layer. Suspending below min bitrate is controlled outside the
+  // codec implementation and is not overridden by this.
+  DataRate min_rate = DataRate::KilobitsPerSec(
+      codec_.simulcastStream[layer_index[active_layer]].minBitrate);
+  left_in_total_allocation = std::max(left_in_total_allocation, min_rate);
+  left_in_stable_allocation = std::max(left_in_stable_allocation, min_rate);
+
+  // Begin by allocating bitrate to simulcast streams, putting all bitrate in
+  // temporal layer 0. We'll then distribute this bitrate, across potential
+  // temporal layers, when stream allocation is done.
+
+  bool first_allocation = false;
+  if (stream_enabled_.empty()) {
+    // First time allocating, this means we should not include hysteresis in
+    // case this is a reconfiguration of an existing enabled stream.
+    first_allocation = true;
+    stream_enabled_.resize(codec_.numberOfSimulcastStreams, false);
+  }
+
+  size_t top_active_layer = active_layer;
+  // Allocate up to the target bitrate for each active simulcast layer.
+  for (; active_layer < codec_.numberOfSimulcastStreams; ++active_layer) {
+    const SimulcastStream& stream =
+        codec_.simulcastStream[layer_index[active_layer]];
+    if (!stream.active) {
+      stream_enabled_[layer_index[active_layer]] = false;
+      continue;
+    }
+    // If we can't allocate to the current layer we can't allocate to higher
+    // layers because they require a higher minimum bitrate.
+    DataRate min_bitrate = DataRate::KilobitsPerSec(stream.minBitrate);
+    DataRate target_bitrate = DataRate::KilobitsPerSec(stream.targetBitrate);
+    double hysteresis_factor =
+        codec_.mode == VideoCodecMode::kRealtimeVideo
+            ? stable_rate_settings_.GetVideoHysteresisFactor()
+            : stable_rate_settings_.GetScreenshareHysteresisFactor();
+    if (!first_allocation && !stream_enabled_[layer_index[active_layer]]) {
+      min_bitrate = std::min(hysteresis_factor * min_bitrate, target_bitrate);
+    }
+    if (left_in_stable_allocation < min_bitrate) {
+      allocated_bitrates->set_bw_limited(true);
+      break;
+    }
+
+    // We are allocating to this layer so it is the current active allocation.
+    top_active_layer = layer_index[active_layer];
+    stream_enabled_[layer_index[active_layer]] = true;
+    DataRate layer_rate = std::min(left_in_total_allocation, target_bitrate);
+    allocated_bitrates->SetBitrate(layer_index[active_layer], 0,
+                                   layer_rate.bps());
+    left_in_total_allocation -= layer_rate;
+    left_in_stable_allocation -=
+        std::min(left_in_stable_allocation, target_bitrate);
+  }
+
+  // All layers above this one are not active.
+  for (; active_layer < codec_.numberOfSimulcastStreams; ++active_layer) {
+    stream_enabled_[layer_index[active_layer]] = false;
+  }
+
+  // Next, try allocate remaining bitrate, up to max bitrate, in top active
+  // stream.
+  // TODO(sprang): Allocate up to max bitrate for all layers once we have a
+  //               better idea of possible performance implications.
+  if (left_in_total_allocation > DataRate::Zero()) {
+    const SimulcastStream& stream = codec_.simulcastStream[top_active_layer];
+    DataRate initial_layer_rate = DataRate::BitsPerSec(
+        allocated_bitrates->GetSpatialLayerSum(top_active_layer));
+    DataRate additional_allocation = std::min(
+        left_in_total_allocation,
+        DataRate::KilobitsPerSec(stream.maxBitrate) - initial_layer_rate);
+    allocated_bitrates->SetBitrate(
+        top_active_layer, 0,
+        (initial_layer_rate + additional_allocation).bps());
+  }
+}
+
+void SimulcastRateAllocator::DistributeAllocationToTemporalLayers(
+    VideoBitrateAllocation* allocated_bitrates_bps) const {
+  const int num_spatial_streams =
+      std::max(1, static_cast<int>(codec_.numberOfSimulcastStreams));
+
+  // Finally, distribute the bitrate for the simulcast streams across the
+  // available temporal layers.
+  for (int simulcast_id = 0; simulcast_id < num_spatial_streams;
+       ++simulcast_id) {
+    uint32_t target_bitrate_kbps =
+        allocated_bitrates_bps->GetBitrate(simulcast_id, 0) / 1000;
+    if (target_bitrate_kbps == 0) {
+      continue;
+    }
+
+    const uint32_t expected_allocated_bitrate_kbps = target_bitrate_kbps;
+    RTC_DCHECK_EQ(
+        target_bitrate_kbps,
+        allocated_bitrates_bps->GetSpatialLayerSum(simulcast_id) / 1000);
+    const int num_temporal_streams = NumTemporalStreams(simulcast_id);
+    uint32_t max_bitrate_kbps;
+    // Legacy temporal-layered only screenshare, or simulcast screenshare
+    // with legacy mode for simulcast stream 0.
+    if (codec_.mode == VideoCodecMode::kScreensharing &&
+        legacy_conference_mode_ && simulcast_id == 0) {
+      // TODO(holmer): This is a "temporary" hack for screensharing, where we
+      // interpret the startBitrate as the encoder target bitrate. This is
+      // to allow for a different max bitrate, so if the codec can't meet
+      // the target we still allow it to overshoot up to the max before dropping
+      // frames. This hack should be improved.
+      max_bitrate_kbps =
+          std::min(kLegacyScreenshareTl1BitrateKbps, target_bitrate_kbps);
+      target_bitrate_kbps =
+          std::min(kLegacyScreenshareTl0BitrateKbps, target_bitrate_kbps);
+    } else if (num_spatial_streams == 1) {
+      max_bitrate_kbps = codec_.maxBitrate;
+    } else {
+      max_bitrate_kbps = codec_.simulcastStream[simulcast_id].maxBitrate;
+    }
+
+    std::vector<uint32_t> tl_allocation;
+    if (num_temporal_streams == 1) {
+      tl_allocation.push_back(target_bitrate_kbps);
+    } else {
+      if (codec_.mode == VideoCodecMode::kScreensharing &&
+          legacy_conference_mode_ && simulcast_id == 0) {
+        tl_allocation = ScreenshareTemporalLayerAllocation(
+            target_bitrate_kbps, max_bitrate_kbps, simulcast_id);
+      } else {
+        tl_allocation = DefaultTemporalLayerAllocation(
+            target_bitrate_kbps, max_bitrate_kbps, simulcast_id);
+      }
+    }
+    RTC_DCHECK_GT(tl_allocation.size(), 0);
+    RTC_DCHECK_LE(tl_allocation.size(), num_temporal_streams);
+
+    uint64_t tl_allocation_sum_kbps = 0;
+    for (size_t tl_index = 0; tl_index < tl_allocation.size(); ++tl_index) {
+      uint32_t layer_rate_kbps = tl_allocation[tl_index];
+      if (layer_rate_kbps > 0) {
+        allocated_bitrates_bps->SetBitrate(simulcast_id, tl_index,
+                                           layer_rate_kbps * 1000);
+      }
+      tl_allocation_sum_kbps += layer_rate_kbps;
+    }
+    RTC_DCHECK_LE(tl_allocation_sum_kbps, expected_allocated_bitrate_kbps);
+  }
+}
+
+std::vector<uint32_t> SimulcastRateAllocator::DefaultTemporalLayerAllocation(
+    int bitrate_kbps,
+    int max_bitrate_kbps,
+    int simulcast_id) const {
+  const size_t num_temporal_layers = NumTemporalStreams(simulcast_id);
+  std::vector<uint32_t> bitrates;
+  for (size_t i = 0; i < num_temporal_layers; ++i) {
+    float layer_bitrate =
+        bitrate_kbps *
+        GetTemporalRateAllocation(
+            num_temporal_layers, i,
+            rate_control_settings_.Vp8BaseHeavyTl3RateAllocation());
+    bitrates.push_back(static_cast<uint32_t>(layer_bitrate + 0.5));
+  }
+
+  // Allocation table is of aggregates, transform to individual rates.
+  uint32_t sum = 0;
+  for (size_t i = 0; i < num_temporal_layers; ++i) {
+    uint32_t layer_bitrate = bitrates[i];
+    RTC_DCHECK_LE(sum, bitrates[i]);
+    bitrates[i] -= sum;
+    sum = layer_bitrate;
+
+    if (sum >= static_cast<uint32_t>(bitrate_kbps)) {
+      // Sum adds up; any subsequent layers will be 0.
+      bitrates.resize(i + 1);
+      break;
+    }
+  }
+
+  return bitrates;
+}
+
+std::vector<uint32_t>
+SimulcastRateAllocator::ScreenshareTemporalLayerAllocation(
+    int bitrate_kbps,
+    int max_bitrate_kbps,
+    int simulcast_id) const {
+  if (simulcast_id > 0) {
+    return DefaultTemporalLayerAllocation(bitrate_kbps, max_bitrate_kbps,
+                                          simulcast_id);
+  }
+  std::vector<uint32_t> allocation;
+  allocation.push_back(bitrate_kbps);
+  if (max_bitrate_kbps > bitrate_kbps)
+    allocation.push_back(max_bitrate_kbps - bitrate_kbps);
+  return allocation;
+}
+
+const VideoCodec& webrtc::SimulcastRateAllocator::GetCodec() const {
+  return codec_;
+}
+
+int SimulcastRateAllocator::NumTemporalStreams(size_t simulcast_id) const {
+  return std::max<uint8_t>(
+      1,
+      codec_.codecType == kVideoCodecVP8 && codec_.numberOfSimulcastStreams == 0
+          ? codec_.VP8().numberOfTemporalLayers
+          : codec_.simulcastStream[simulcast_id].numberOfTemporalLayers);
+}
+
+void SimulcastRateAllocator::SetLegacyConferenceMode(bool enabled) {
+  legacy_conference_mode_ = enabled;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator.h b/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator.h
new file mode 100644
index 0000000000..6f93dbde74
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_SIMULCAST_RATE_ALLOCATOR_H_
+#define MODULES_VIDEO_CODING_UTILITY_SIMULCAST_RATE_ALLOCATOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_bitrate_allocator.h"
+#include "api/video_codecs/video_codec.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+#include "rtc_base/experiments/stable_target_rate_experiment.h"
+
+namespace webrtc {
+
+class SimulcastRateAllocator : public VideoBitrateAllocator {
+ public:
+  explicit SimulcastRateAllocator(const VideoCodec& codec);
+  ~SimulcastRateAllocator() override;
+
+  SimulcastRateAllocator(const SimulcastRateAllocator&) = delete;
+  SimulcastRateAllocator& operator=(const SimulcastRateAllocator&) = delete;
+
+  VideoBitrateAllocation Allocate(
+      VideoBitrateAllocationParameters parameters) override;
+  const VideoCodec& GetCodec() const;
+
+  static float GetTemporalRateAllocation(int num_layers,
+                                         int temporal_id,
+                                         bool base_heavy_tl3_alloc);
+
+  void SetLegacyConferenceMode(bool mode) override;
+
+ private:
+  void DistributeAllocationToSimulcastLayers(
+      DataRate total_bitrate,
+      DataRate stable_bitrate,
+      VideoBitrateAllocation* allocated_bitrates);
+  void DistributeAllocationToTemporalLayers(
+      VideoBitrateAllocation* allocated_bitrates) const;
+  std::vector<uint32_t> DefaultTemporalLayerAllocation(int bitrate_kbps,
+                                                       int max_bitrate_kbps,
+                                                       int simulcast_id) const;
+  std::vector<uint32_t> ScreenshareTemporalLayerAllocation(
+      int bitrate_kbps,
+      int max_bitrate_kbps,
+      int simulcast_id) const;
+  int NumTemporalStreams(size_t simulcast_id) const;
+
+  const VideoCodec codec_;
+  const StableTargetRateExperiment stable_rate_settings_;
+  const RateControlSettings rate_control_settings_;
+  std::vector<bool> stream_enabled_;
+  bool legacy_conference_mode_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_SIMULCAST_RATE_ALLOCATOR_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc
new file mode 100644
index 0000000000..24d7c58bcd
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc
@@ -0,0 +1,824 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
+
+#include <limits>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/video_codecs/vp8_frame_buffer_controller.h"
+#include "api/video_codecs/vp8_frame_config.h"
+#include "api/video_codecs/vp8_temporal_layers.h"
+#include "rtc_base/checks.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+using ::testing::_;
+
+constexpr uint32_t kFramerateFps = 5;
+constexpr uint32_t kMinBitrateKbps = 50;
+// These correspond to kLegacyScreenshareTl(0|1)BitrateKbps in cc.
+constexpr uint32_t kLegacyScreenshareTargetBitrateKbps = 200;
+constexpr uint32_t kLegacyScreenshareMaxBitrateKbps = 1000;
+// Bitrates for upper simulcast screenshare layer.
+constexpr uint32_t kSimulcastScreenshareMinBitrateKbps = 600;
+constexpr uint32_t kSimulcastScreenshareMaxBitrateKbps = 1250;
+// Default video hysteresis factor: allocatable bitrate for next layer must
+// exceed 20% of min setting in order to be initially turned on.
+const double kDefaultHysteresis = 1.2;
+
+class MockTemporalLayers : public Vp8FrameBufferController {
+ public:
+  MOCK_METHOD(Vp8FrameConfig, NextFrameConfig, (size_t, uint32_t), (override));
+  MOCK_METHOD(void,
+              OnRatesUpdated,
+              (size_t, const std::vector<uint32_t>&, int),
+              (override));
+  MOCK_METHOD(Vp8EncoderConfig, UpdateConfiguration, (size_t), (override));
+  MOCK_METHOD(void,
+              OnEncodeDone,
+              (size_t, uint32_t, size_t, bool, int, CodecSpecificInfo*),
+              (override));
+};
+}  // namespace
+
+class SimulcastRateAllocatorTest : public ::testing::TestWithParam<bool> {
+ public:
+  SimulcastRateAllocatorTest() {
+    codec_.codecType = kVideoCodecVP8;
+    codec_.minBitrate = kMinBitrateKbps;
+    codec_.maxBitrate = kLegacyScreenshareMaxBitrateKbps;
+    codec_.active = true;
+    CreateAllocator();
+  }
+  virtual ~SimulcastRateAllocatorTest() {}
+
+  template <size_t S>
+  void ExpectEqual(uint32_t (&expected)[S],
+                   const std::vector<uint32_t>& actual) {
+    EXPECT_EQ(S, actual.size());
+    for (size_t i = 0; i < S; ++i)
+      EXPECT_EQ(expected[i], actual[i]) << "Mismatch at index " << i;
+  }
+
+  template <size_t S>
+  void ExpectEqual(uint32_t (&expected)[S],
+                   const VideoBitrateAllocation& actual) {
+    // EXPECT_EQ(S, actual.size());
+    uint32_t sum = 0;
+    for (size_t i = 0; i < S; ++i) {
+      uint32_t layer_bitrate = actual.GetSpatialLayerSum(i);
+      if (layer_bitrate == 0) {
+        EXPECT_FALSE(actual.IsSpatialLayerUsed(i));
+      }
+      EXPECT_EQ(expected[i] * 1000U, layer_bitrate)
+          << "Mismatch at index " << i;
+      sum += layer_bitrate;
+    }
+    EXPECT_EQ(sum, actual.get_sum_bps());
+  }
+
+  void CreateAllocator(bool legacy_conference_mode = false) {
+    allocator_.reset(new SimulcastRateAllocator(codec_));
+    allocator_->SetLegacyConferenceMode(legacy_conference_mode);
+  }
+
+  void SetupCodec3SL3TL(const std::vector<bool>& active_streams) {
+    const size_t num_simulcast_layers = 3;
+    RTC_DCHECK_GE(active_streams.size(), num_simulcast_layers);
+    SetupCodec2SL3TL(active_streams);
+    codec_.numberOfSimulcastStreams = num_simulcast_layers;
+    codec_.simulcastStream[2].numberOfTemporalLayers = 3;
+    codec_.simulcastStream[2].maxBitrate = 4000;
+    codec_.simulcastStream[2].targetBitrate = 3000;
+    codec_.simulcastStream[2].minBitrate = 2000;
+    codec_.simulcastStream[2].active = active_streams[2];
+  }
+
+  void SetupCodec2SL3TL(const std::vector<bool>& active_streams) {
+    const size_t num_simulcast_layers = 2;
+    RTC_DCHECK_GE(active_streams.size(), num_simulcast_layers);
+    SetupCodec1SL3TL(active_streams);
+    codec_.numberOfSimulcastStreams = num_simulcast_layers;
+    codec_.simulcastStream[1].numberOfTemporalLayers = 3;
+    codec_.simulcastStream[1].maxBitrate = 1000;
+    codec_.simulcastStream[1].targetBitrate = 500;
+    codec_.simulcastStream[1].minBitrate = 50;
+    codec_.simulcastStream[1].active = active_streams[1];
+  }
+
+  void SetupCodec1SL3TL(const std::vector<bool>& active_streams) {
+    const size_t num_simulcast_layers = 2;
+    RTC_DCHECK_GE(active_streams.size(), num_simulcast_layers);
+    SetupCodec3TL();
+    codec_.numberOfSimulcastStreams = num_simulcast_layers;
+    codec_.simulcastStream[0].numberOfTemporalLayers = 3;
+    codec_.simulcastStream[0].maxBitrate = 500;
+    codec_.simulcastStream[0].targetBitrate = 100;
+    codec_.simulcastStream[0].minBitrate = 10;
+    codec_.simulcastStream[0].active = active_streams[0];
+  }
+
+  void SetupCodec3TL() {
+    codec_.maxBitrate = 0;
+    codec_.VP8()->numberOfTemporalLayers = 3;
+  }
+
+  VideoBitrateAllocation GetAllocation(uint32_t target_bitrate) {
+    return allocator_->Allocate(VideoBitrateAllocationParameters(
+        DataRate::KilobitsPerSec(target_bitrate), kDefaultFrameRate));
+  }
+
+  VideoBitrateAllocation GetAllocation(DataRate target_rate,
+                                       DataRate stable_rate) {
+    return allocator_->Allocate(VideoBitrateAllocationParameters(
+        target_rate, stable_rate, kDefaultFrameRate));
+  }
+
+  DataRate MinRate(size_t layer_index) const {
+    return DataRate::KilobitsPerSec(
+        codec_.simulcastStream[layer_index].minBitrate);
+  }
+
+  DataRate TargetRate(size_t layer_index) const {
+    return DataRate::KilobitsPerSec(
+        codec_.simulcastStream[layer_index].targetBitrate);
+  }
+
+  DataRate MaxRate(size_t layer_index) const {
+    return DataRate::KilobitsPerSec(
+        codec_.simulcastStream[layer_index].maxBitrate);
+  }
+
+ protected:
+  static const int kDefaultFrameRate = 30;
+  VideoCodec codec_;
+  std::unique_ptr<SimulcastRateAllocator> allocator_;
+};
+
+TEST_F(SimulcastRateAllocatorTest, NoSimulcastBelowMin) {
+  uint32_t expected[] = {codec_.minBitrate};
+  codec_.active = true;
+  ExpectEqual(expected, GetAllocation(codec_.minBitrate - 1));
+  ExpectEqual(expected, GetAllocation(1));
+  ExpectEqual(expected, GetAllocation(0));
+}
+
+TEST_F(SimulcastRateAllocatorTest, NoSimulcastAboveMax) {
+  uint32_t expected[] = {codec_.maxBitrate};
+  codec_.active = true;
+  ExpectEqual(expected, GetAllocation(codec_.maxBitrate + 1));
+  ExpectEqual(expected, GetAllocation(std::numeric_limits<uint32_t>::max()));
+}
+
+TEST_F(SimulcastRateAllocatorTest, NoSimulcastNoMax) {
+  const uint32_t kMax = VideoBitrateAllocation::kMaxBitrateBps / 1000;
+  codec_.active = true;
+  codec_.maxBitrate = 0;
+  CreateAllocator();
+
+  uint32_t expected[] = {kMax};
+  ExpectEqual(expected, GetAllocation(kMax));
+}
+
+TEST_F(SimulcastRateAllocatorTest, NoSimulcastWithinLimits) {
+  codec_.active = true;
+  for (uint32_t bitrate = codec_.minBitrate; bitrate <= codec_.maxBitrate;
+       ++bitrate) {
+    uint32_t expected[] = {bitrate};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+}
+
+// Tests that when we aren't using simulcast and the codec is marked inactive no
+// bitrate will be allocated.
+TEST_F(SimulcastRateAllocatorTest, NoSimulcastInactive) {
+  codec_.active = false;
+  uint32_t expected[] = {0};
+  CreateAllocator();
+
+  ExpectEqual(expected, GetAllocation(kMinBitrateKbps - 10));
+  ExpectEqual(expected, GetAllocation(kLegacyScreenshareTargetBitrateKbps));
+  ExpectEqual(expected, GetAllocation(kLegacyScreenshareMaxBitrateKbps + 10));
+}
+
+TEST_F(SimulcastRateAllocatorTest, SingleSimulcastBelowMin) {
+  // With simulcast, use the min bitrate from the ss spec instead of the global.
+  codec_.numberOfSimulcastStreams = 1;
+  const uint32_t kMin = codec_.minBitrate - 10;
+  codec_.simulcastStream[0].minBitrate = kMin;
+  codec_.simulcastStream[0].targetBitrate = kLegacyScreenshareTargetBitrateKbps;
+  codec_.simulcastStream[0].active = true;
+  CreateAllocator();
+
+  uint32_t expected[] = {kMin};
+  ExpectEqual(expected, GetAllocation(kMin - 1));
+  ExpectEqual(expected, GetAllocation(1));
+  ExpectEqual(expected, GetAllocation(0));
+}
+
+TEST_F(SimulcastRateAllocatorTest, SignalsBwLimited) {
+  // Enough to enable all layers.
+  const int kVeryBigBitrate = 100000;
+
+  // With simulcast, use the min bitrate from the ss spec instead of the global.
+  SetupCodec3SL3TL({true, true, true});
+  CreateAllocator();
+
+  EXPECT_TRUE(
+      GetAllocation(codec_.simulcastStream[0].minBitrate - 10).is_bw_limited());
+  EXPECT_TRUE(
+      GetAllocation(codec_.simulcastStream[0].targetBitrate).is_bw_limited());
+  EXPECT_TRUE(GetAllocation(codec_.simulcastStream[0].targetBitrate +
+                            codec_.simulcastStream[1].minBitrate)
+                  .is_bw_limited());
+  EXPECT_FALSE(
+      GetAllocation(
+          codec_.simulcastStream[0].targetBitrate +
+          codec_.simulcastStream[1].targetBitrate +
+          static_cast<uint32_t>(
+              codec_.simulcastStream[2].minBitrate * kDefaultHysteresis + 0.5))
+          .is_bw_limited());
+  EXPECT_FALSE(GetAllocation(kVeryBigBitrate).is_bw_limited());
+}
+
+TEST_F(SimulcastRateAllocatorTest, SingleSimulcastAboveMax) {
+  codec_.numberOfSimulcastStreams = 1;
+  codec_.simulcastStream[0].minBitrate = kMinBitrateKbps;
+  const uint32_t kMax = codec_.simulcastStream[0].maxBitrate + 1000;
+  codec_.simulcastStream[0].maxBitrate = kMax;
+  codec_.simulcastStream[0].active = true;
+  CreateAllocator();
+
+  uint32_t expected[] = {kMax};
+  ExpectEqual(expected, GetAllocation(kMax));
+  ExpectEqual(expected, GetAllocation(kMax + 1));
+  ExpectEqual(expected, GetAllocation(std::numeric_limits<uint32_t>::max()));
+}
+
+TEST_F(SimulcastRateAllocatorTest, SingleSimulcastWithinLimits) {
+  codec_.numberOfSimulcastStreams = 1;
+  codec_.simulcastStream[0].minBitrate = kMinBitrateKbps;
+  codec_.simulcastStream[0].targetBitrate = kLegacyScreenshareTargetBitrateKbps;
+  codec_.simulcastStream[0].maxBitrate = kLegacyScreenshareMaxBitrateKbps;
+  codec_.simulcastStream[0].active = true;
+  CreateAllocator();
+
+  for (uint32_t bitrate = kMinBitrateKbps;
+       bitrate <= kLegacyScreenshareMaxBitrateKbps; ++bitrate) {
+    uint32_t expected[] = {bitrate};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+}
+
+TEST_F(SimulcastRateAllocatorTest, Regular3TLTemporalRateAllocation) {
+  SetupCodec3SL3TL({true, true, true});
+  CreateAllocator();
+
+  const VideoBitrateAllocation alloc = GetAllocation(kMinBitrateKbps);
+  // 40/20/40.
+  EXPECT_EQ(static_cast<uint32_t>(0.4 * kMinBitrateKbps),
+            alloc.GetBitrate(0, 0) / 1000);
+  EXPECT_EQ(static_cast<uint32_t>(0.2 * kMinBitrateKbps),
+            alloc.GetBitrate(0, 1) / 1000);
+  EXPECT_EQ(static_cast<uint32_t>(0.4 * kMinBitrateKbps),
+            alloc.GetBitrate(0, 2) / 1000);
+}
+
+TEST_F(SimulcastRateAllocatorTest, BaseHeavy3TLTemporalRateAllocation) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-UseBaseHeavyVP8TL3RateAllocation/Enabled/");
+
+  SetupCodec3SL3TL({true, true, true});
+  CreateAllocator();
+
+  const VideoBitrateAllocation alloc = GetAllocation(kMinBitrateKbps);
+  // 60/20/20.
+  EXPECT_EQ(static_cast<uint32_t>(0.6 * kMinBitrateKbps),
+            alloc.GetBitrate(0, 0) / 1000);
+  EXPECT_EQ(static_cast<uint32_t>(0.2 * kMinBitrateKbps),
+            alloc.GetBitrate(0, 1) / 1000);
+  EXPECT_EQ(static_cast<uint32_t>(0.2 * kMinBitrateKbps),
+            alloc.GetBitrate(0, 2) / 1000);
+}
+
+TEST_F(SimulcastRateAllocatorTest, SingleSimulcastInactive) {
+  codec_.numberOfSimulcastStreams = 1;
+  codec_.simulcastStream[0].minBitrate = kMinBitrateKbps;
+  codec_.simulcastStream[0].targetBitrate = kLegacyScreenshareTargetBitrateKbps;
+  codec_.simulcastStream[0].maxBitrate = kLegacyScreenshareMaxBitrateKbps;
+  codec_.simulcastStream[0].active = false;
+  CreateAllocator();
+
+  uint32_t expected[] = {0};
+  ExpectEqual(expected, GetAllocation(kMinBitrateKbps - 10));
+  ExpectEqual(expected, GetAllocation(kLegacyScreenshareTargetBitrateKbps));
+  ExpectEqual(expected, GetAllocation(kLegacyScreenshareMaxBitrateKbps + 10));
+}
+
+TEST_F(SimulcastRateAllocatorTest, OneToThreeStreams) {
+  SetupCodec3SL3TL({true, true, true});
+  CreateAllocator();
+
+  {
+    // Single stream, min bitrate.
+    const uint32_t bitrate = codec_.simulcastStream[0].minBitrate;
+    uint32_t expected[] = {bitrate, 0, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Single stream at target bitrate.
+    const uint32_t bitrate = codec_.simulcastStream[0].targetBitrate;
+    uint32_t expected[] = {bitrate, 0, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  uint32_t kMinInitialRateTwoLayers =
+      codec_.simulcastStream[0].targetBitrate +
+      static_cast<uint32_t>(codec_.simulcastStream[1].minBitrate *
+                            kDefaultHysteresis);
+  {
+    // Bitrate above target for first stream, but below min for the next one.
+    const uint32_t bitrate = kMinInitialRateTwoLayers - 1;
+    uint32_t expected[] = {bitrate, 0, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Just enough for two streams.
+    const uint32_t bitrate = kMinInitialRateTwoLayers;
+    uint32_t expected[] = {
+        codec_.simulcastStream[0].targetBitrate,
+        kMinInitialRateTwoLayers - codec_.simulcastStream[0].targetBitrate, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Second stream maxed out, but not enough for third.
+    const uint32_t bitrate = codec_.simulcastStream[0].targetBitrate +
+                             codec_.simulcastStream[1].maxBitrate;
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate,
+                           codec_.simulcastStream[1].maxBitrate, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  uint32_t kMinInitialRateThreeLayers =
+      codec_.simulcastStream[0].targetBitrate +
+      codec_.simulcastStream[1].targetBitrate +
+      static_cast<uint32_t>(codec_.simulcastStream[2].minBitrate *
+                            kDefaultHysteresis);
+  {
+    // First two streams maxed out, but not enough for third. Nowhere to put
+    // remaining bits.
+    const uint32_t bitrate = kMinInitialRateThreeLayers - 1;
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate,
+                           codec_.simulcastStream[1].maxBitrate, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Just enough for all three streams.
+    const uint32_t bitrate = kMinInitialRateThreeLayers;
+    uint32_t expected[] = {
+        codec_.simulcastStream[0].targetBitrate,
+        codec_.simulcastStream[1].targetBitrate,
+        static_cast<uint32_t>(codec_.simulcastStream[2].minBitrate *
+                              kDefaultHysteresis)};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Third maxed out.
+    const uint32_t bitrate = codec_.simulcastStream[0].targetBitrate +
+                             codec_.simulcastStream[1].targetBitrate +
+                             codec_.simulcastStream[2].maxBitrate;
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate,
+                           codec_.simulcastStream[1].targetBitrate,
+                           codec_.simulcastStream[2].maxBitrate};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Enough to max out all streams which will allocate the target amount to
+    // the lower streams.
+    const uint32_t bitrate = codec_.simulcastStream[0].maxBitrate +
+                             codec_.simulcastStream[1].maxBitrate +
+                             codec_.simulcastStream[2].maxBitrate;
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate,
+                           codec_.simulcastStream[1].targetBitrate,
+                           codec_.simulcastStream[2].maxBitrate};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+}
+
+// If three simulcast streams that are all inactive, none of them should be
+// allocated bitrate.
+TEST_F(SimulcastRateAllocatorTest, ThreeStreamsInactive) {
+  SetupCodec3SL3TL({false, false, false});
+  CreateAllocator();
+
+  // Just enough to allocate the min.
+  const uint32_t min_bitrate = codec_.simulcastStream[0].minBitrate +
+                               codec_.simulcastStream[1].minBitrate +
+                               codec_.simulcastStream[2].minBitrate;
+  // Enough bitrate to allocate target to all streams.
+  const uint32_t target_bitrate = codec_.simulcastStream[0].targetBitrate +
+                                  codec_.simulcastStream[1].targetBitrate +
+                                  codec_.simulcastStream[2].targetBitrate;
+  // Enough bitrate to allocate max to all streams.
+  const uint32_t max_bitrate = codec_.simulcastStream[0].maxBitrate +
+                               codec_.simulcastStream[1].maxBitrate +
+                               codec_.simulcastStream[2].maxBitrate;
+  uint32_t expected[] = {0, 0, 0};
+  ExpectEqual(expected, GetAllocation(0));
+  ExpectEqual(expected, GetAllocation(min_bitrate));
+  ExpectEqual(expected, GetAllocation(target_bitrate));
+  ExpectEqual(expected, GetAllocation(max_bitrate));
+}
+
+// If there are two simulcast streams, we expect the high active stream to be
+// allocated as if it is a single active stream.
+TEST_F(SimulcastRateAllocatorTest, TwoStreamsLowInactive) {
+  SetupCodec2SL3TL({false, true});
+  CreateAllocator();
+
+  const uint32_t kActiveStreamMinBitrate = codec_.simulcastStream[1].minBitrate;
+  const uint32_t kActiveStreamTargetBitrate =
+      codec_.simulcastStream[1].targetBitrate;
+  const uint32_t kActiveStreamMaxBitrate = codec_.simulcastStream[1].maxBitrate;
+  {
+    // Expect that the stream is always allocated its min bitrate.
+    uint32_t expected[] = {0, kActiveStreamMinBitrate};
+    ExpectEqual(expected, GetAllocation(0));
+    ExpectEqual(expected, GetAllocation(kActiveStreamMinBitrate - 10));
+    ExpectEqual(expected, GetAllocation(kActiveStreamMinBitrate));
+  }
+
+  {
+    // The stream should be allocated its target bitrate.
+    uint32_t expected[] = {0, kActiveStreamTargetBitrate};
+    ExpectEqual(expected, GetAllocation(kActiveStreamTargetBitrate));
+  }
+
+  {
+    // The stream should be allocated its max if the target input is sufficient.
+    uint32_t expected[] = {0, kActiveStreamMaxBitrate};
+    ExpectEqual(expected, GetAllocation(kActiveStreamMaxBitrate));
+    ExpectEqual(expected, GetAllocation(std::numeric_limits<uint32_t>::max()));
+  }
+}
+
+// If there are two simulcast streams, we expect the low active stream to be
+// allocated as if it is a single active stream.
+TEST_F(SimulcastRateAllocatorTest, TwoStreamsHighInactive) {
+  SetupCodec2SL3TL({true, false});
+  CreateAllocator();
+
+  const uint32_t kActiveStreamMinBitrate = codec_.simulcastStream[0].minBitrate;
+  const uint32_t kActiveStreamTargetBitrate =
+      codec_.simulcastStream[0].targetBitrate;
+  const uint32_t kActiveStreamMaxBitrate = codec_.simulcastStream[0].maxBitrate;
+  {
+    // Expect that the stream is always allocated its min bitrate.
+    uint32_t expected[] = {kActiveStreamMinBitrate, 0};
+    ExpectEqual(expected, GetAllocation(0));
+    ExpectEqual(expected, GetAllocation(kActiveStreamMinBitrate - 10));
+    ExpectEqual(expected, GetAllocation(kActiveStreamMinBitrate));
+  }
+
+  {
+    // The stream should be allocated its target bitrate.
+    uint32_t expected[] = {kActiveStreamTargetBitrate, 0};
+    ExpectEqual(expected, GetAllocation(kActiveStreamTargetBitrate));
+  }
+
+  {
+    // The stream should be allocated its max if the target input is sufficent.
+    uint32_t expected[] = {kActiveStreamMaxBitrate, 0};
+    ExpectEqual(expected, GetAllocation(kActiveStreamMaxBitrate));
+    ExpectEqual(expected, GetAllocation(std::numeric_limits<uint32_t>::max()));
+  }
+}
+
+// If there are three simulcast streams and the middle stream is inactive, the
+// other two streams should be allocated bitrate the same as if they are two
+// active simulcast streams.
+TEST_F(SimulcastRateAllocatorTest, ThreeStreamsMiddleInactive) {
+  SetupCodec3SL3TL({true, false, true});
+  CreateAllocator();
+
+  {
+    const uint32_t kLowStreamMinBitrate = codec_.simulcastStream[0].minBitrate;
+    // The lowest stream should always be allocated its minimum bitrate.
+    uint32_t expected[] = {kLowStreamMinBitrate, 0, 0};
+    ExpectEqual(expected, GetAllocation(0));
+    ExpectEqual(expected, GetAllocation(kLowStreamMinBitrate - 10));
+    ExpectEqual(expected, GetAllocation(kLowStreamMinBitrate));
+  }
+
+  {
+    // The lowest stream gets its target bitrate.
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate, 0, 0};
+    ExpectEqual(expected,
+                GetAllocation(codec_.simulcastStream[0].targetBitrate));
+  }
+
+  {
+    // The lowest stream gets its max bitrate, but not enough for the high
+    // stream.
+    const uint32_t bitrate = codec_.simulcastStream[0].targetBitrate +
+                             codec_.simulcastStream[2].minBitrate - 1;
+    uint32_t expected[] = {codec_.simulcastStream[0].maxBitrate, 0, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Both active streams get allocated target bitrate.
+    const uint32_t bitrate = codec_.simulcastStream[0].targetBitrate +
+                             codec_.simulcastStream[2].targetBitrate;
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate, 0,
+                           codec_.simulcastStream[2].targetBitrate};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Lowest stream gets its target bitrate, high stream gets its max bitrate.
+    uint32_t bitrate = codec_.simulcastStream[0].targetBitrate +
+                       codec_.simulcastStream[2].maxBitrate;
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate, 0,
+                           codec_.simulcastStream[2].maxBitrate};
+    ExpectEqual(expected, GetAllocation(bitrate));
+    ExpectEqual(expected, GetAllocation(bitrate + 10));
+    ExpectEqual(expected, GetAllocation(std::numeric_limits<uint32_t>::max()));
+  }
+}
+
+TEST_F(SimulcastRateAllocatorTest, NonConferenceModeScreenshare) {
+  codec_.mode = VideoCodecMode::kScreensharing;
+  SetupCodec3SL3TL({true, true, true});
+  CreateAllocator();
+
+  // Make sure we have enough bitrate for all 3 simulcast layers
+  const uint32_t bitrate = codec_.simulcastStream[0].maxBitrate +
+                           codec_.simulcastStream[1].maxBitrate +
+                           codec_.simulcastStream[2].maxBitrate;
+  const VideoBitrateAllocation alloc = GetAllocation(bitrate);
+
+  EXPECT_EQ(alloc.GetTemporalLayerAllocation(0).size(), 3u);
+  EXPECT_EQ(alloc.GetTemporalLayerAllocation(1).size(), 3u);
+  EXPECT_EQ(alloc.GetTemporalLayerAllocation(2).size(), 3u);
+}
+
+TEST_F(SimulcastRateAllocatorTest, StableRate) {
+  webrtc::test::ScopedFieldTrials field_trials(
+      "WebRTC-StableTargetRate/"
+      "enabled:true,"
+      "video_hysteresis_factor:1.1/");
+
+  SetupCodec3SL3TL({true, true, true});
+  CreateAllocator();
+
+  // Let the volatile rate always be be enough for all streams, in this test we
+  // are only interested in how the stable rate affects enablement.
+  const DataRate volatile_rate =
+      (TargetRate(0) + TargetRate(1) + MinRate(2)) * 1.1;
+
+  {
+    // On the first call to a new SimulcastRateAllocator instance, hysteresis
+    // is disabled, but stable rate still caps layers.
+    uint32_t expected[] = {TargetRate(0).kbps<uint32_t>(),
+                           MaxRate(1).kbps<uint32_t>()};
+    ExpectEqual(expected,
+                GetAllocation(volatile_rate, TargetRate(0) + MinRate(1)));
+  }
+
+  {
+    // Let stable rate go to a bitrate below what is needed for two streams.
+    uint32_t expected[] = {MaxRate(0).kbps<uint32_t>(), 0};
+    ExpectEqual(expected,
+                GetAllocation(volatile_rate, TargetRate(0) + MinRate(1) -
+                                                 DataRate::BitsPerSec(1)));
+  }
+
+  {
+    // Don't enable stream as we need to get up above hysteresis threshold.
+    uint32_t expected[] = {MaxRate(0).kbps<uint32_t>(), 0};
+    ExpectEqual(expected,
+                GetAllocation(volatile_rate, TargetRate(0) + MinRate(1)));
+  }
+
+  {
+    // Above threshold with hysteresis, enable second stream.
+    uint32_t expected[] = {TargetRate(0).kbps<uint32_t>(),
+                           MaxRate(1).kbps<uint32_t>()};
+    ExpectEqual(expected, GetAllocation(volatile_rate,
+                                        (TargetRate(0) + MinRate(1)) * 1.1));
+  }
+
+  {
+    // Enough to enable all thee layers.
+    uint32_t expected[] = {
+        TargetRate(0).kbps<uint32_t>(), TargetRate(1).kbps<uint32_t>(),
+        (volatile_rate - TargetRate(0) - TargetRate(1)).kbps<uint32_t>()};
+    ExpectEqual(expected, GetAllocation(volatile_rate, volatile_rate));
+  }
+
+  {
+    // Drop hysteresis, all three still on.
+    uint32_t expected[] = {
+        TargetRate(0).kbps<uint32_t>(), TargetRate(1).kbps<uint32_t>(),
+        (volatile_rate - TargetRate(0) - TargetRate(1)).kbps<uint32_t>()};
+    ExpectEqual(expected,
+                GetAllocation(volatile_rate,
+                              TargetRate(0) + TargetRate(1) + MinRate(2)));
+  }
+}
+
+class ScreenshareRateAllocationTest : public SimulcastRateAllocatorTest {
+ public:
+  void SetupConferenceScreenshare(bool use_simulcast, bool active = true) {
+    codec_.mode = VideoCodecMode::kScreensharing;
+    codec_.minBitrate = kMinBitrateKbps;
+    codec_.maxBitrate =
+        kLegacyScreenshareMaxBitrateKbps + kSimulcastScreenshareMaxBitrateKbps;
+    if (use_simulcast) {
+      codec_.numberOfSimulcastStreams = 2;
+      codec_.simulcastStream[0].minBitrate = kMinBitrateKbps;
+      codec_.simulcastStream[0].targetBitrate =
+          kLegacyScreenshareTargetBitrateKbps;
+      codec_.simulcastStream[0].maxBitrate = kLegacyScreenshareMaxBitrateKbps;
+      codec_.simulcastStream[0].numberOfTemporalLayers = 2;
+      codec_.simulcastStream[0].active = active;
+
+      codec_.simulcastStream[1].minBitrate =
+          kSimulcastScreenshareMinBitrateKbps;
+      codec_.simulcastStream[1].targetBitrate =
+          kSimulcastScreenshareMaxBitrateKbps;
+      codec_.simulcastStream[1].maxBitrate =
+          kSimulcastScreenshareMaxBitrateKbps;
+      codec_.simulcastStream[1].numberOfTemporalLayers = 2;
+      codec_.simulcastStream[1].active = active;
+    } else {
+      codec_.numberOfSimulcastStreams = 0;
+      codec_.VP8()->numberOfTemporalLayers = 2;
+      codec_.active = active;
+    }
+  }
+};
+
+INSTANTIATE_TEST_SUITE_P(ScreenshareTest,
+                         ScreenshareRateAllocationTest,
+                         ::testing::Bool());
+
+TEST_P(ScreenshareRateAllocationTest, ConferenceBitrateBelowTl0) {
+  SetupConferenceScreenshare(GetParam());
+  CreateAllocator(true);
+
+  VideoBitrateAllocation allocation =
+      allocator_->Allocate(VideoBitrateAllocationParameters(
+          kLegacyScreenshareTargetBitrateKbps * 1000, kFramerateFps));
+
+  // All allocation should go in TL0.
+  EXPECT_EQ(kLegacyScreenshareTargetBitrateKbps, allocation.get_sum_kbps());
+  EXPECT_EQ(kLegacyScreenshareTargetBitrateKbps,
+            allocation.GetBitrate(0, 0) / 1000);
+  EXPECT_EQ(allocation.is_bw_limited(), GetParam());
+}
+
+TEST_P(ScreenshareRateAllocationTest, ConferenceBitrateAboveTl0) {
+  SetupConferenceScreenshare(GetParam());
+  CreateAllocator(true);
+
+  uint32_t target_bitrate_kbps =
+      (kLegacyScreenshareTargetBitrateKbps + kLegacyScreenshareMaxBitrateKbps) /
+      2;
+  VideoBitrateAllocation allocation =
+      allocator_->Allocate(VideoBitrateAllocationParameters(
+          target_bitrate_kbps * 1000, kFramerateFps));
+
+  // Fill TL0, then put the rest in TL1.
+  EXPECT_EQ(target_bitrate_kbps, allocation.get_sum_kbps());
+  EXPECT_EQ(kLegacyScreenshareTargetBitrateKbps,
+            allocation.GetBitrate(0, 0) / 1000);
+  EXPECT_EQ(target_bitrate_kbps - kLegacyScreenshareTargetBitrateKbps,
+            allocation.GetBitrate(0, 1) / 1000);
+  EXPECT_EQ(allocation.is_bw_limited(), GetParam());
+}
+
+TEST_F(ScreenshareRateAllocationTest, ConferenceBitrateAboveTl1) {
+  // This test is only for the non-simulcast case.
+  SetupConferenceScreenshare(false);
+  CreateAllocator(true);
+
+  VideoBitrateAllocation allocation =
+      allocator_->Allocate(VideoBitrateAllocationParameters(
+          kLegacyScreenshareMaxBitrateKbps * 2000, kFramerateFps));
+
+  // Fill both TL0 and TL1, but no more.
+  EXPECT_EQ(kLegacyScreenshareMaxBitrateKbps, allocation.get_sum_kbps());
+  EXPECT_EQ(kLegacyScreenshareTargetBitrateKbps,
+            allocation.GetBitrate(0, 0) / 1000);
+  EXPECT_EQ(
+      kLegacyScreenshareMaxBitrateKbps - kLegacyScreenshareTargetBitrateKbps,
+      allocation.GetBitrate(0, 1) / 1000);
+  EXPECT_FALSE(allocation.is_bw_limited());
+}
+
+// This tests when the screenshare is inactive it should be allocated 0 bitrate
+// for all layers.
+TEST_P(ScreenshareRateAllocationTest, InactiveScreenshare) {
+  SetupConferenceScreenshare(GetParam(), false);
+  CreateAllocator();
+
+  // Enough bitrate for TL0 and TL1.
+  uint32_t target_bitrate_kbps =
+      (kLegacyScreenshareTargetBitrateKbps + kLegacyScreenshareMaxBitrateKbps) /
+      2;
+  VideoBitrateAllocation allocation =
+      allocator_->Allocate(VideoBitrateAllocationParameters(
+          target_bitrate_kbps * 1000, kFramerateFps));
+
+  EXPECT_EQ(0U, allocation.get_sum_kbps());
+}
+
+TEST_F(ScreenshareRateAllocationTest, Hysteresis) {
+  // This test is only for the simulcast case.
+  SetupConferenceScreenshare(true);
+  CreateAllocator();
+
+  // The bitrate at which we would normally enable the upper simulcast stream.
+  const uint32_t default_enable_rate_bps =
+      codec_.simulcastStream[0].targetBitrate +
+      codec_.simulcastStream[1].minBitrate;
+  const uint32_t enable_rate_with_hysteresis_bps =
+      (default_enable_rate_bps * 135) / 100;
+
+  {
+    // On the first call to a new SimulcastRateAllocator instance, hysteresis
+    // is disabled.
+    const uint32_t bitrate = default_enable_rate_bps;
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate,
+                           codec_.simulcastStream[1].minBitrate};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Go down to a bitrate below what is needed for two streams.
+    const uint32_t bitrate = default_enable_rate_bps - 1;
+    uint32_t expected[] = {bitrate, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Don't enable stream as we need to get up above hysteresis threshold.
+    const uint32_t bitrate = default_enable_rate_bps;
+    uint32_t expected[] = {bitrate, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Above threshold, enable second stream.
+    const uint32_t bitrate = enable_rate_with_hysteresis_bps;
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate,
+                           enable_rate_with_hysteresis_bps -
+                               codec_.simulcastStream[0].targetBitrate};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Go down again, still keep the second stream alive.
+    const uint32_t bitrate = default_enable_rate_bps;
+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate,
+                           codec_.simulcastStream[1].minBitrate};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Go down below default enable, second stream is shut down again.
+    const uint32_t bitrate = default_enable_rate_bps - 1;
+    uint32_t expected[] = {bitrate, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+
+  {
+    // Go up, hysteresis is blocking us again.
+    const uint32_t bitrate = default_enable_rate_bps;
+    uint32_t expected[] = {bitrate, 0};
+    ExpectEqual(expected, GetAllocation(bitrate));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc b/third_party/libwebrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc
new file mode 100644
index 0000000000..35224b17ed
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc
@@ -0,0 +1,967 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/simulcast_test_fixture_impl.h"
+
+#include <algorithm>
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_encoder.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::AllOf;
+using ::testing::Field;
+using ::testing::Return;
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+const int kDefaultWidth = 1280;
+const int kDefaultHeight = 720;
+const int kNumberOfSimulcastStreams = 3;
+const int kColorY = 66;
+const int kColorU = 22;
+const int kColorV = 33;
+const int kMaxBitrates[kNumberOfSimulcastStreams] = {150, 600, 1200};
+const int kMinBitrates[kNumberOfSimulcastStreams] = {50, 150, 600};
+const int kTargetBitrates[kNumberOfSimulcastStreams] = {100, 450, 1000};
+const float kMaxFramerates[kNumberOfSimulcastStreams] = {30, 30, 30};
+const int kScaleResolutionDownBy[kNumberOfSimulcastStreams] = {4, 2, 1};
+const int kDefaultTemporalLayerProfile[3] = {3, 3, 3};
+const int kNoTemporalLayerProfile[3] = {0, 0, 0};
+
+const VideoEncoder::Capabilities kCapabilities(false);
+const VideoEncoder::Settings kSettings(kCapabilities, 1, 1200);
+
+template <typename T>
+void SetExpectedValues3(T value0, T value1, T value2, T* expected_values) {
+  expected_values[0] = value0;
+  expected_values[1] = value1;
+  expected_values[2] = value2;
+}
+
+enum PlaneType {
+  kYPlane = 0,
+  kUPlane = 1,
+  kVPlane = 2,
+  kNumOfPlanes = 3,
+};
+
+}  // namespace
+
+class SimulcastTestFixtureImpl::TestEncodedImageCallback
+    : public EncodedImageCallback {
+ public:
+  TestEncodedImageCallback() {
+    memset(temporal_layer_, -1, sizeof(temporal_layer_));
+    memset(layer_sync_, false, sizeof(layer_sync_));
+  }
+
+  Result OnEncodedImage(const EncodedImage& encoded_image,
+                        const CodecSpecificInfo* codec_specific_info) override {
+    bool is_vp8 = (codec_specific_info->codecType == kVideoCodecVP8);
+    bool is_h264 = (codec_specific_info->codecType == kVideoCodecH264);
+    // Only store the base layer.
+    if (encoded_image.SpatialIndex().value_or(0) == 0) {
+      if (encoded_image._frameType == VideoFrameType::kVideoFrameKey) {
+        encoded_key_frame_.SetEncodedData(EncodedImageBuffer::Create(
+            encoded_image.data(), encoded_image.size()));
+        encoded_key_frame_._frameType = VideoFrameType::kVideoFrameKey;
+      } else {
+        encoded_frame_.SetEncodedData(EncodedImageBuffer::Create(
+            encoded_image.data(), encoded_image.size()));
+      }
+    }
+    if (is_vp8) {
+      layer_sync_[encoded_image.SpatialIndex().value_or(0)] =
+          codec_specific_info->codecSpecific.VP8.layerSync;
+      temporal_layer_[encoded_image.SpatialIndex().value_or(0)] =
+          codec_specific_info->codecSpecific.VP8.temporalIdx;
+    } else if (is_h264) {
+      layer_sync_[encoded_image.SpatialIndex().value_or(0)] =
+          codec_specific_info->codecSpecific.H264.base_layer_sync;
+      temporal_layer_[encoded_image.SpatialIndex().value_or(0)] =
+          codec_specific_info->codecSpecific.H264.temporal_idx;
+    }
+    return Result(Result::OK, encoded_image.Timestamp());
+  }
+  // This method only makes sense for VP8.
+  void GetLastEncodedFrameInfo(int* temporal_layer,
+                               bool* layer_sync,
+                               int stream) {
+    *temporal_layer = temporal_layer_[stream];
+    *layer_sync = layer_sync_[stream];
+  }
+  void GetLastEncodedKeyFrame(EncodedImage* encoded_key_frame) {
+    *encoded_key_frame = encoded_key_frame_;
+  }
+  void GetLastEncodedFrame(EncodedImage* encoded_frame) {
+    *encoded_frame = encoded_frame_;
+  }
+
+ private:
+  EncodedImage encoded_key_frame_;
+  EncodedImage encoded_frame_;
+  int temporal_layer_[kNumberOfSimulcastStreams];
+  bool layer_sync_[kNumberOfSimulcastStreams];
+};
+
+class SimulcastTestFixtureImpl::TestDecodedImageCallback
+    : public DecodedImageCallback {
+ public:
+  TestDecodedImageCallback() : decoded_frames_(0) {}
+  int32_t Decoded(VideoFrame& decoded_image) override {
+    rtc::scoped_refptr<I420BufferInterface> i420_buffer =
+        decoded_image.video_frame_buffer()->ToI420();
+    for (int i = 0; i < decoded_image.width(); ++i) {
+      EXPECT_NEAR(kColorY, i420_buffer->DataY()[i], 1);
+    }
+
+    // TODO(mikhal): Verify the difference between U,V and the original.
+    for (int i = 0; i < i420_buffer->ChromaWidth(); ++i) {
+      EXPECT_NEAR(kColorU, i420_buffer->DataU()[i], 4);
+      EXPECT_NEAR(kColorV, i420_buffer->DataV()[i], 4);
+    }
+    decoded_frames_++;
+    return 0;
+  }
+  int32_t Decoded(VideoFrame& decoded_image, int64_t decode_time_ms) override {
+    RTC_DCHECK_NOTREACHED();
+    return -1;
+  }
+  void Decoded(VideoFrame& decoded_image,
+               absl::optional<int32_t> decode_time_ms,
+               absl::optional<uint8_t> qp) override {
+    Decoded(decoded_image);
+  }
+  int DecodedFrames() { return decoded_frames_; }
+
+ private:
+  int decoded_frames_;
+};
+
+namespace {
+
+void SetPlane(uint8_t* data, uint8_t value, int width, int height, int stride) {
+  for (int i = 0; i < height; i++, data += stride) {
+    // Setting allocated area to zero - setting only image size to
+    // requested values - will make it easier to distinguish between image
+    // size and frame size (accounting for stride).
+    memset(data, value, width);
+    memset(data + width, 0, stride - width);
+  }
+}
+
+// Fills in an I420Buffer from `plane_colors`.
+void CreateImage(const rtc::scoped_refptr<I420Buffer>& buffer,
+                 int plane_colors[kNumOfPlanes]) {
+  SetPlane(buffer->MutableDataY(), plane_colors[0], buffer->width(),
+           buffer->height(), buffer->StrideY());
+
+  SetPlane(buffer->MutableDataU(), plane_colors[1], buffer->ChromaWidth(),
+           buffer->ChromaHeight(), buffer->StrideU());
+
+  SetPlane(buffer->MutableDataV(), plane_colors[2], buffer->ChromaWidth(),
+           buffer->ChromaHeight(), buffer->StrideV());
+}
+
+void ConfigureStream(int width,
+                     int height,
+                     int max_bitrate,
+                     int min_bitrate,
+                     int target_bitrate,
+                     float max_framerate,
+                     SimulcastStream* stream,
+                     int num_temporal_layers) {
+  RTC_DCHECK(stream);
+  stream->width = width;
+  stream->height = height;
+  stream->maxBitrate = max_bitrate;
+  stream->minBitrate = min_bitrate;
+  stream->targetBitrate = target_bitrate;
+  stream->maxFramerate = max_framerate;
+  if (num_temporal_layers >= 0) {
+    stream->numberOfTemporalLayers = num_temporal_layers;
+  }
+  stream->qpMax = 45;
+  stream->active = true;
+}
+
+}  // namespace
+
+void SimulcastTestFixtureImpl::DefaultSettings(
+    VideoCodec* settings,
+    const int* temporal_layer_profile,
+    VideoCodecType codec_type,
+    bool reverse_layer_order) {
+  RTC_CHECK(settings);
+  *settings = {};
+  settings->codecType = codec_type;
+  settings->startBitrate = 300;
+  settings->minBitrate = 30;
+  settings->maxBitrate = 0;
+  settings->maxFramerate = 30;
+  settings->width = kDefaultWidth;
+  settings->height = kDefaultHeight;
+  settings->numberOfSimulcastStreams = kNumberOfSimulcastStreams;
+  settings->active = true;
+  ASSERT_EQ(3, kNumberOfSimulcastStreams);
+  int layer_order[3] = {0, 1, 2};
+  if (reverse_layer_order) {
+    layer_order[0] = 2;
+    layer_order[2] = 0;
+  }
+  settings->timing_frame_thresholds = {kDefaultTimingFramesDelayMs,
+                                       kDefaultOutlierFrameSizePercent};
+  ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4, kMaxBitrates[0],
+                  kMinBitrates[0], kTargetBitrates[0], kMaxFramerates[0],
+                  &settings->simulcastStream[layer_order[0]],
+                  temporal_layer_profile[0]);
+  ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2, kMaxBitrates[1],
+                  kMinBitrates[1], kTargetBitrates[1], kMaxFramerates[1],
+                  &settings->simulcastStream[layer_order[1]],
+                  temporal_layer_profile[1]);
+  ConfigureStream(kDefaultWidth, kDefaultHeight, kMaxBitrates[2],
+                  kMinBitrates[2], kTargetBitrates[2], kMaxFramerates[2],
+                  &settings->simulcastStream[layer_order[2]],
+                  temporal_layer_profile[2]);
+  settings->SetFrameDropEnabled(true);
+  if (codec_type == kVideoCodecVP8) {
+    settings->VP8()->denoisingOn = true;
+    settings->VP8()->automaticResizeOn = false;
+    settings->VP8()->keyFrameInterval = 3000;
+  } else {
+    settings->H264()->keyFrameInterval = 3000;
+  }
+}
+
+SimulcastTestFixtureImpl::SimulcastTestFixtureImpl(
+    std::unique_ptr<VideoEncoderFactory> encoder_factory,
+    std::unique_ptr<VideoDecoderFactory> decoder_factory,
+    SdpVideoFormat video_format)
+    : codec_type_(PayloadStringToCodecType(video_format.name)) {
+  encoder_ = encoder_factory->CreateVideoEncoder(video_format);
+  decoder_ = decoder_factory->CreateVideoDecoder(video_format);
+  SetUpCodec((codec_type_ == kVideoCodecVP8 || codec_type_ == kVideoCodecH264)
+                 ? kDefaultTemporalLayerProfile
+                 : kNoTemporalLayerProfile);
+}
+
+SimulcastTestFixtureImpl::~SimulcastTestFixtureImpl() {
+  encoder_->Release();
+  decoder_->Release();
+}
+
+void SimulcastTestFixtureImpl::SetUpCodec(const int* temporal_layer_profile) {
+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback_);
+  decoder_->RegisterDecodeCompleteCallback(&decoder_callback_);
+  DefaultSettings(&settings_, temporal_layer_profile, codec_type_);
+  SetUpRateAllocator();
+  EXPECT_EQ(0, encoder_->InitEncode(&settings_, kSettings));
+  VideoDecoder::Settings decoder_settings;
+  decoder_settings.set_max_render_resolution({kDefaultWidth, kDefaultHeight});
+  decoder_settings.set_codec_type(codec_type_);
+  EXPECT_TRUE(decoder_->Configure(decoder_settings));
+  input_buffer_ = I420Buffer::Create(kDefaultWidth, kDefaultHeight);
+  input_buffer_->InitializeData();
+  input_frame_ = std::make_unique<webrtc::VideoFrame>(
+      webrtc::VideoFrame::Builder()
+          .set_video_frame_buffer(input_buffer_)
+          .set_rotation(webrtc::kVideoRotation_0)
+          .set_timestamp_us(0)
+          .build());
+}
+
+void SimulcastTestFixtureImpl::SetUpRateAllocator() {
+  rate_allocator_.reset(new SimulcastRateAllocator(settings_));
+}
+
+void SimulcastTestFixtureImpl::SetRates(uint32_t bitrate_kbps, uint32_t fps) {
+  encoder_->SetRates(VideoEncoder::RateControlParameters(
+      rate_allocator_->Allocate(
+          VideoBitrateAllocationParameters(bitrate_kbps * 1000, fps)),
+      static_cast<double>(fps)));
+}
+
+void SimulcastTestFixtureImpl::RunActiveStreamsTest(
+    const std::vector<bool> active_streams) {
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  UpdateActiveStreams(active_streams);
+  // Set sufficient bitrate for all streams so we can test active without
+  // bitrate being an issue.
+  SetRates(kMaxBitrates[0] + kMaxBitrates[1] + kMaxBitrates[2], 30);
+
+  ExpectStreams(VideoFrameType::kVideoFrameKey, active_streams);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, active_streams);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::UpdateActiveStreams(
+    const std::vector<bool> active_streams) {
+  ASSERT_EQ(static_cast<int>(active_streams.size()), kNumberOfSimulcastStreams);
+  for (size_t i = 0; i < active_streams.size(); ++i) {
+    settings_.simulcastStream[i].active = active_streams[i];
+  }
+  // Re initialize the allocator and encoder with the new settings.
+  // TODO(bugs.webrtc.org/8807): Currently, we do a full "hard"
+  // reconfiguration of the allocator and encoder. When the video bitrate
+  // allocator has support for updating active streams without a
+  // reinitialization, we can just call that here instead.
+  SetUpRateAllocator();
+  EXPECT_EQ(0, encoder_->InitEncode(&settings_, kSettings));
+}
+
+void SimulcastTestFixtureImpl::ExpectStream(VideoFrameType frame_type,
+                                            int scaleResolutionDownBy) {
+  EXPECT_CALL(
+      encoder_callback_,
+      OnEncodedImage(AllOf(Field(&EncodedImage::_frameType, frame_type),
+                           Field(&EncodedImage::_encodedWidth,
+                                 kDefaultWidth / scaleResolutionDownBy),
+                           Field(&EncodedImage::_encodedHeight,
+                                 kDefaultHeight / scaleResolutionDownBy)),
+                     _))
+      .Times(1)
+      .WillRepeatedly(Return(
+          EncodedImageCallback::Result(EncodedImageCallback::Result::OK, 0)));
+}
+
+void SimulcastTestFixtureImpl::ExpectStreams(
+    VideoFrameType frame_type,
+    const std::vector<bool> expected_streams_active) {
+  ASSERT_EQ(static_cast<int>(expected_streams_active.size()),
+            kNumberOfSimulcastStreams);
+  for (size_t i = 0; i < kNumberOfSimulcastStreams; i++) {
+    if (expected_streams_active[i]) {
+      ExpectStream(frame_type, kScaleResolutionDownBy[i]);
+    }
+  }
+}
+
+void SimulcastTestFixtureImpl::ExpectStreams(VideoFrameType frame_type,
+                                             int expected_video_streams) {
+  ASSERT_GE(expected_video_streams, 0);
+  ASSERT_LE(expected_video_streams, kNumberOfSimulcastStreams);
+  std::vector<bool> expected_streams_active(kNumberOfSimulcastStreams, false);
+  for (int i = 0; i < expected_video_streams; ++i) {
+    expected_streams_active[i] = true;
+  }
+  ExpectStreams(frame_type, expected_streams_active);
+}
+
+void SimulcastTestFixtureImpl::VerifyTemporalIdxAndSyncForAllSpatialLayers(
+    TestEncodedImageCallback* encoder_callback,
+    const int* expected_temporal_idx,
+    const bool* expected_layer_sync,
+    int num_spatial_layers) {
+  int temporal_layer = -1;
+  bool layer_sync = false;
+  for (int i = 0; i < num_spatial_layers; i++) {
+    encoder_callback->GetLastEncodedFrameInfo(&temporal_layer, &layer_sync, i);
+    EXPECT_EQ(expected_temporal_idx[i], temporal_layer);
+    EXPECT_EQ(expected_layer_sync[i], layer_sync);
+  }
+}
+
+// For some codecs (VP8) expect all active streams to generate a key frame even
+// though a key frame was only requested for some of them.
+void SimulcastTestFixtureImpl::TestKeyFrameRequestsOnAllStreams() {
+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, kNumberOfSimulcastStreams);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, kNumberOfSimulcastStreams);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  frame_types[0] = VideoFrameType::kVideoFrameKey;
+  ExpectStreams(VideoFrameType::kVideoFrameKey, kNumberOfSimulcastStreams);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  std::fill(frame_types.begin(), frame_types.end(),
+            VideoFrameType::kVideoFrameDelta);
+  frame_types[1] = VideoFrameType::kVideoFrameKey;
+  ExpectStreams(VideoFrameType::kVideoFrameKey, kNumberOfSimulcastStreams);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  std::fill(frame_types.begin(), frame_types.end(),
+            VideoFrameType::kVideoFrameDelta);
+  frame_types[2] = VideoFrameType::kVideoFrameKey;
+  ExpectStreams(VideoFrameType::kVideoFrameKey, kNumberOfSimulcastStreams);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  std::fill(frame_types.begin(), frame_types.end(),
+            VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, kNumberOfSimulcastStreams);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+// For some codecs (H264) expect only particular active streams to generate a
+// key frame when a key frame was only requested for some of them.
+void SimulcastTestFixtureImpl::TestKeyFrameRequestsOnSpecificStreams() {
+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, kNumberOfSimulcastStreams);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, kNumberOfSimulcastStreams);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  frame_types[0] = VideoFrameType::kVideoFrameKey;
+  ExpectStream(VideoFrameType::kVideoFrameKey, kScaleResolutionDownBy[0]);
+  ExpectStream(VideoFrameType::kVideoFrameDelta, kScaleResolutionDownBy[1]);
+  ExpectStream(VideoFrameType::kVideoFrameDelta, kScaleResolutionDownBy[2]);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  std::fill(frame_types.begin(), frame_types.end(),
+            VideoFrameType::kVideoFrameDelta);
+  frame_types[1] = VideoFrameType::kVideoFrameKey;
+  ExpectStream(VideoFrameType::kVideoFrameDelta, kScaleResolutionDownBy[0]);
+  ExpectStream(VideoFrameType::kVideoFrameKey, kScaleResolutionDownBy[1]);
+  ExpectStream(VideoFrameType::kVideoFrameDelta, kScaleResolutionDownBy[2]);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  std::fill(frame_types.begin(), frame_types.end(),
+            VideoFrameType::kVideoFrameDelta);
+  frame_types[2] = VideoFrameType::kVideoFrameKey;
+  ExpectStream(VideoFrameType::kVideoFrameDelta, kScaleResolutionDownBy[0]);
+  ExpectStream(VideoFrameType::kVideoFrameDelta, kScaleResolutionDownBy[1]);
+  ExpectStream(VideoFrameType::kVideoFrameKey, kScaleResolutionDownBy[2]);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  std::fill(frame_types.begin(), frame_types.end(),
+            VideoFrameType::kVideoFrameDelta);
+  frame_types[0] = VideoFrameType::kVideoFrameKey;
+  frame_types[2] = VideoFrameType::kVideoFrameKey;
+  ExpectStream(VideoFrameType::kVideoFrameKey, kScaleResolutionDownBy[0]);
+  ExpectStream(VideoFrameType::kVideoFrameDelta, kScaleResolutionDownBy[1]);
+  ExpectStream(VideoFrameType::kVideoFrameKey, kScaleResolutionDownBy[2]);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  std::fill(frame_types.begin(), frame_types.end(),
+            VideoFrameType::kVideoFrameKey);
+  ExpectStream(VideoFrameType::kVideoFrameKey, kScaleResolutionDownBy[0]);
+  ExpectStream(VideoFrameType::kVideoFrameKey, kScaleResolutionDownBy[1]);
+  ExpectStream(VideoFrameType::kVideoFrameKey, kScaleResolutionDownBy[2]);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  std::fill(frame_types.begin(), frame_types.end(),
+            VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, kNumberOfSimulcastStreams);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::TestPaddingAllStreams() {
+  // We should always encode the base layer.
+  SetRates(kMinBitrates[0] - 1, 30);
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 1);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 1);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::TestPaddingTwoStreams() {
+  // We have just enough to get only the first stream and padding for two.
+  SetRates(kMinBitrates[0], 30);
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 1);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 1);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::TestPaddingTwoStreamsOneMaxedOut() {
+  // We are just below limit of sending second stream, so we should get
+  // the first stream maxed out (at `maxBitrate`), and padding for two.
+  SetRates(kTargetBitrates[0] + kMinBitrates[1] - 1, 30);
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 1);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 1);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::TestPaddingOneStream() {
+  // We have just enough to send two streams, so padding for one stream.
+  SetRates(kTargetBitrates[0] + kMinBitrates[1], 30);
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 2);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 2);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::TestPaddingOneStreamTwoMaxedOut() {
+  // We are just below limit of sending third stream, so we should get
+  // first stream's rate maxed out at `targetBitrate`, second at `maxBitrate`.
+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] - 1, 30);
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 2);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 2);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::TestSendAllStreams() {
+  // We have just enough to send all streams.
+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2], 30);
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 3);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 3);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::TestDisablingStreams() {
+  // We should get three media streams.
+  SetRates(kMaxBitrates[0] + kMaxBitrates[1] + kMaxBitrates[2], 30);
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 3);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 3);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  // We should only get two streams and padding for one.
+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 2);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  // We should only get the first stream and padding for two.
+  SetRates(kTargetBitrates[0] + kMinBitrates[1] / 2, 30);
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 1);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  // We don't have enough bitrate for the thumbnail stream, but we should get
+  // it anyway with current configuration.
+  SetRates(kTargetBitrates[0] - 1, 30);
+  ExpectStreams(VideoFrameType::kVideoFrameDelta, 1);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  // We should only get two streams and padding for one.
+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
+  // We get a key frame because a new stream is being enabled.
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 2);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  // We should get all three streams.
+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kTargetBitrates[2], 30);
+  // We get a key frame because a new stream is being enabled.
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 3);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::TestActiveStreams() {
+  // All streams on.
+  RunActiveStreamsTest({true, true, true});
+  // All streams off.
+  RunActiveStreamsTest({false, false, false});
+  // Low stream off.
+  RunActiveStreamsTest({false, true, true});
+  // Middle stream off.
+  RunActiveStreamsTest({true, false, true});
+  // High stream off.
+  RunActiveStreamsTest({true, true, false});
+  // Only low stream turned on.
+  RunActiveStreamsTest({true, false, false});
+  // Only middle stream turned on.
+  RunActiveStreamsTest({false, true, false});
+  // Only high stream turned on.
+  RunActiveStreamsTest({false, false, true});
+}
+
+void SimulcastTestFixtureImpl::SwitchingToOneStream(int width, int height) {
+  const int* temporal_layer_profile = nullptr;
+  // Disable all streams except the last and set the bitrate of the last to
+  // 100 kbps. This verifies the way GTP switches to screenshare mode.
+  if (codec_type_ == kVideoCodecVP8) {
+    settings_.VP8()->numberOfTemporalLayers = 1;
+    temporal_layer_profile = kDefaultTemporalLayerProfile;
+  } else {
+    settings_.H264()->numberOfTemporalLayers = 1;
+    temporal_layer_profile = kNoTemporalLayerProfile;
+  }
+  settings_.maxBitrate = 100;
+  settings_.startBitrate = 100;
+  settings_.width = width;
+  settings_.height = height;
+  for (int i = 0; i < settings_.numberOfSimulcastStreams - 1; ++i) {
+    settings_.simulcastStream[i].maxBitrate = 0;
+    settings_.simulcastStream[i].width = settings_.width;
+    settings_.simulcastStream[i].height = settings_.height;
+    settings_.simulcastStream[i].numberOfTemporalLayers = 1;
+  }
+  // Setting input image to new resolution.
+  input_buffer_ = I420Buffer::Create(settings_.width, settings_.height);
+  input_buffer_->InitializeData();
+
+  input_frame_ = std::make_unique<webrtc::VideoFrame>(
+      webrtc::VideoFrame::Builder()
+          .set_video_frame_buffer(input_buffer_)
+          .set_rotation(webrtc::kVideoRotation_0)
+          .set_timestamp_us(0)
+          .build());
+
+  // The for loop above did not set the bitrate of the highest layer.
+  settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].maxBitrate =
+      0;
+  // The highest layer has to correspond to the non-simulcast resolution.
+  settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].width =
+      settings_.width;
+  settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].height =
+      settings_.height;
+  SetUpRateAllocator();
+  EXPECT_EQ(0, encoder_->InitEncode(&settings_, kSettings));
+
+  // Encode one frame and verify.
+  SetRates(kMaxBitrates[0] + kMaxBitrates[1], 30);
+  std::vector<VideoFrameType> frame_types(kNumberOfSimulcastStreams,
+                                          VideoFrameType::kVideoFrameDelta);
+  EXPECT_CALL(
+      encoder_callback_,
+      OnEncodedImage(AllOf(Field(&EncodedImage::_frameType,
+                                 VideoFrameType::kVideoFrameKey),
+                           Field(&EncodedImage::_encodedWidth, width),
+                           Field(&EncodedImage::_encodedHeight, height)),
+                     _))
+      .Times(1)
+      .WillRepeatedly(Return(
+          EncodedImageCallback::Result(EncodedImageCallback::Result::OK, 0)));
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+
+  // Switch back.
+  DefaultSettings(&settings_, temporal_layer_profile, codec_type_);
+  // Start at the lowest bitrate for enabling base stream.
+  settings_.startBitrate = kMinBitrates[0];
+  SetUpRateAllocator();
+  EXPECT_EQ(0, encoder_->InitEncode(&settings_, kSettings));
+  SetRates(settings_.startBitrate, 30);
+  ExpectStreams(VideoFrameType::kVideoFrameKey, 1);
+  // Resize `input_frame_` to the new resolution.
+  input_buffer_ = I420Buffer::Create(settings_.width, settings_.height);
+  input_buffer_->InitializeData();
+  input_frame_ = std::make_unique<webrtc::VideoFrame>(
+      webrtc::VideoFrame::Builder()
+          .set_video_frame_buffer(input_buffer_)
+          .set_rotation(webrtc::kVideoRotation_0)
+          .set_timestamp_us(0)
+          .build());
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, &frame_types));
+}
+
+void SimulcastTestFixtureImpl::TestSwitchingToOneStream() {
+  SwitchingToOneStream(1024, 768);
+}
+
+void SimulcastTestFixtureImpl::TestSwitchingToOneOddStream() {
+  SwitchingToOneStream(1023, 769);
+}
+
+void SimulcastTestFixtureImpl::TestSwitchingToOneSmallStream() {
+  SwitchingToOneStream(4, 4);
+}
+
+// Test the layer pattern and sync flag for various spatial-temporal patterns.
+// 3-3-3 pattern: 3 temporal layers for all spatial streams, so same
+// temporal_layer id and layer_sync is expected for all streams.
+void SimulcastTestFixtureImpl::TestSpatioTemporalLayers333PatternEncoder() {
+  bool is_h264 = codec_type_ == kVideoCodecH264;
+  TestEncodedImageCallback encoder_callback;
+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
+
+  int expected_temporal_idx[3] = {-1, -1, -1};
+  bool expected_layer_sync[3] = {false, false, false};
+
+  // First frame: #0.
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
+  SetExpectedValues3<bool>(!is_h264, !is_h264, !is_h264, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #1.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
+  SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #2.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(1, 1, 1, expected_temporal_idx);
+  SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #3.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
+  SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #4.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
+  SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #5.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
+  SetExpectedValues3<bool>(is_h264, is_h264, is_h264, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+}
+
+// Test the layer pattern and sync flag for various spatial-temporal patterns.
+// 3-2-1 pattern: 3 temporal layers for lowest resolution, 2 for middle, and
+// 1 temporal layer for highest resolution.
+// For this profile, we expect the temporal index pattern to be:
+// 1st stream: 0, 2, 1, 2, ....
+// 2nd stream: 0, 1, 0, 1, ...
+// 3rd stream: -1, -1, -1, -1, ....
+// Regarding the 3rd stream, note that a stream/encoder with 1 temporal layer
+// should always have temporal layer idx set to kNoTemporalIdx = -1.
+// Since CodecSpecificInfoVP8.temporalIdx is uint8_t, this will wrap to 255.
+// TODO(marpan): Although this seems safe for now, we should fix this.
+void SimulcastTestFixtureImpl::TestSpatioTemporalLayers321PatternEncoder() {
+  EXPECT_EQ(codec_type_, kVideoCodecVP8);
+  int temporal_layer_profile[3] = {3, 2, 1};
+  SetUpCodec(temporal_layer_profile);
+  TestEncodedImageCallback encoder_callback;
+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
+
+  int expected_temporal_idx[3] = {-1, -1, -1};
+  bool expected_layer_sync[3] = {false, false, false};
+
+  // First frame: #0.
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
+  SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #1.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
+  SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #2.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(1, 0, 255, expected_temporal_idx);
+  SetExpectedValues3<bool>(true, false, false, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #3.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
+  SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #4.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
+  SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+
+  // Next frame: #5.
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+  SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
+  SetExpectedValues3<bool>(false, true, false, expected_layer_sync);
+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
+}
+
+void SimulcastTestFixtureImpl::TestStrideEncodeDecode() {
+  TestEncodedImageCallback encoder_callback;
+  TestDecodedImageCallback decoder_callback;
+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
+  decoder_->RegisterDecodeCompleteCallback(&decoder_callback);
+
+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
+  // Setting two (possibly) problematic use cases for stride:
+  // 1. stride > width 2. stride_y != stride_uv/2
+  int stride_y = kDefaultWidth + 20;
+  int stride_uv = ((kDefaultWidth + 1) / 2) + 5;
+  input_buffer_ = I420Buffer::Create(kDefaultWidth, kDefaultHeight, stride_y,
+                                     stride_uv, stride_uv);
+  input_frame_ = std::make_unique<webrtc::VideoFrame>(
+      webrtc::VideoFrame::Builder()
+          .set_video_frame_buffer(input_buffer_)
+          .set_rotation(webrtc::kVideoRotation_0)
+          .set_timestamp_us(0)
+          .build());
+
+  // Set color.
+  int plane_offset[kNumOfPlanes];
+  plane_offset[kYPlane] = kColorY;
+  plane_offset[kUPlane] = kColorU;
+  plane_offset[kVPlane] = kColorV;
+  CreateImage(input_buffer_, plane_offset);
+
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+
+  // Change color.
+  plane_offset[kYPlane] += 1;
+  plane_offset[kUPlane] += 1;
+  plane_offset[kVPlane] += 1;
+  CreateImage(input_buffer_, plane_offset);
+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+
+  EncodedImage encoded_frame;
+  // Only encoding one frame - so will be a key frame.
+  encoder_callback.GetLastEncodedKeyFrame(&encoded_frame);
+  EXPECT_EQ(0, decoder_->Decode(encoded_frame, false, 0));
+  encoder_callback.GetLastEncodedFrame(&encoded_frame);
+  decoder_->Decode(encoded_frame, false, 0);
+  EXPECT_EQ(2, decoder_callback.DecodedFrames());
+}
+
+void SimulcastTestFixtureImpl::TestDecodeWidthHeightSet() {
+  MockEncodedImageCallback encoder_callback;
+  MockDecodedImageCallback decoder_callback;
+
+  EncodedImage encoded_frame[3];
+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
+  decoder_->RegisterDecodeCompleteCallback(&decoder_callback);
+
+  EXPECT_CALL(encoder_callback, OnEncodedImage(_, _))
+      .Times(3)
+      .WillRepeatedly(
+          ::testing::Invoke([&](const EncodedImage& encoded_image,
+                                const CodecSpecificInfo* codec_specific_info) {
+            EXPECT_EQ(encoded_image._frameType, VideoFrameType::kVideoFrameKey);
+
+            size_t index = encoded_image.SpatialIndex().value_or(0);
+            encoded_frame[index].SetEncodedData(EncodedImageBuffer::Create(
+                encoded_image.data(), encoded_image.size()));
+            encoded_frame[index]._frameType = encoded_image._frameType;
+            return EncodedImageCallback::Result(
+                EncodedImageCallback::Result::OK, 0);
+          }));
+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL));
+
+  EXPECT_CALL(decoder_callback, Decoded(_, _, _))
+      .WillOnce(::testing::Invoke([](VideoFrame& decodedImage,
+                                     absl::optional<int32_t> decode_time_ms,
+                                     absl::optional<uint8_t> qp) {
+        EXPECT_EQ(decodedImage.width(), kDefaultWidth / 4);
+        EXPECT_EQ(decodedImage.height(), kDefaultHeight / 4);
+      }));
+  EXPECT_EQ(0, decoder_->Decode(encoded_frame[0], false, 0));
+
+  EXPECT_CALL(decoder_callback, Decoded(_, _, _))
+      .WillOnce(::testing::Invoke([](VideoFrame& decodedImage,
+                                     absl::optional<int32_t> decode_time_ms,
+                                     absl::optional<uint8_t> qp) {
+        EXPECT_EQ(decodedImage.width(), kDefaultWidth / 2);
+        EXPECT_EQ(decodedImage.height(), kDefaultHeight / 2);
+      }));
+  EXPECT_EQ(0, decoder_->Decode(encoded_frame[1], false, 0));
+
+  EXPECT_CALL(decoder_callback, Decoded(_, _, _))
+      .WillOnce(::testing::Invoke([](VideoFrame& decodedImage,
+                                     absl::optional<int32_t> decode_time_ms,
+                                     absl::optional<uint8_t> qp) {
+        EXPECT_EQ(decodedImage.width(), kDefaultWidth);
+        EXPECT_EQ(decodedImage.height(), kDefaultHeight);
+      }));
+  EXPECT_EQ(0, decoder_->Decode(encoded_frame[2], false, 0));
+}
+
+void SimulcastTestFixtureImpl::
+    TestEncoderInfoForDefaultTemporalLayerProfileHasFpsAllocation() {
+  VideoEncoder::EncoderInfo encoder_info = encoder_->GetEncoderInfo();
+  EXPECT_EQ(encoder_info.fps_allocation[0].size(),
+            static_cast<size_t>(kDefaultTemporalLayerProfile[0]));
+  EXPECT_EQ(encoder_info.fps_allocation[1].size(),
+            static_cast<size_t>(kDefaultTemporalLayerProfile[1]));
+  EXPECT_EQ(encoder_info.fps_allocation[2].size(),
+            static_cast<size_t>(kDefaultTemporalLayerProfile[2]));
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h b/third_party/libwebrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h
new file mode 100644
index 0000000000..f142ab4813
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_SIMULCAST_TEST_FIXTURE_IMPL_H_
+#define MODULES_VIDEO_CODING_UTILITY_SIMULCAST_TEST_FIXTURE_IMPL_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/test/mock_video_decoder.h"
+#include "api/test/mock_video_encoder.h"
+#include "api/test/simulcast_test_fixture.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
+
+namespace webrtc {
+namespace test {
+
+class SimulcastTestFixtureImpl final : public SimulcastTestFixture {
+ public:
+  SimulcastTestFixtureImpl(std::unique_ptr<VideoEncoderFactory> encoder_factory,
+                           std::unique_ptr<VideoDecoderFactory> decoder_factory,
+                           SdpVideoFormat video_format);
+  ~SimulcastTestFixtureImpl() final;
+
+  // Implements SimulcastTestFixture.
+  void TestKeyFrameRequestsOnAllStreams() override;
+  void TestKeyFrameRequestsOnSpecificStreams() override;
+  void TestPaddingAllStreams() override;
+  void TestPaddingTwoStreams() override;
+  void TestPaddingTwoStreamsOneMaxedOut() override;
+  void TestPaddingOneStream() override;
+  void TestPaddingOneStreamTwoMaxedOut() override;
+  void TestSendAllStreams() override;
+  void TestDisablingStreams() override;
+  void TestActiveStreams() override;
+  void TestSwitchingToOneStream() override;
+  void TestSwitchingToOneOddStream() override;
+  void TestSwitchingToOneSmallStream() override;
+  void TestSpatioTemporalLayers333PatternEncoder() override;
+  void TestSpatioTemporalLayers321PatternEncoder() override;
+  void TestStrideEncodeDecode() override;
+  void TestDecodeWidthHeightSet() override;
+  void TestEncoderInfoForDefaultTemporalLayerProfileHasFpsAllocation() override;
+
+  static void DefaultSettings(VideoCodec* settings,
+                              const int* temporal_layer_profile,
+                              VideoCodecType codec_type,
+                              bool reverse_layer_order = false);
+
+ private:
+  class TestEncodedImageCallback;
+  class TestDecodedImageCallback;
+
+  void SetUpCodec(const int* temporal_layer_profile);
+  void SetUpRateAllocator();
+  void SetRates(uint32_t bitrate_kbps, uint32_t fps);
+  void RunActiveStreamsTest(std::vector<bool> active_streams);
+  void UpdateActiveStreams(std::vector<bool> active_streams);
+  void ExpectStream(VideoFrameType frame_type, int scaleResolutionDownBy);
+  void ExpectStreams(VideoFrameType frame_type,
+                     std::vector<bool> expected_streams_active);
+  void ExpectStreams(VideoFrameType frame_type, int expected_video_streams);
+  void VerifyTemporalIdxAndSyncForAllSpatialLayers(
+      TestEncodedImageCallback* encoder_callback,
+      const int* expected_temporal_idx,
+      const bool* expected_layer_sync,
+      int num_spatial_layers);
+  void SwitchingToOneStream(int width, int height);
+
+  std::unique_ptr<VideoEncoder> encoder_;
+  MockEncodedImageCallback encoder_callback_;
+  std::unique_ptr<VideoDecoder> decoder_;
+  MockDecodedImageCallback decoder_callback_;
+  VideoCodec settings_;
+  rtc::scoped_refptr<I420Buffer> input_buffer_;
+  std::unique_ptr<VideoFrame> input_frame_;
+  std::unique_ptr<SimulcastRateAllocator> rate_allocator_;
+  VideoCodecType codec_type_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_SIMULCAST_TEST_FIXTURE_IMPL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/simulcast_utility.cc b/third_party/libwebrtc/modules/video_coding/utility/simulcast_utility.cc
new file mode 100644
index 0000000000..65061ed32a
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/simulcast_utility.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/simulcast_utility.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+uint32_t SimulcastUtility::SumStreamMaxBitrate(int streams,
+                                               const VideoCodec& codec) {
+  uint32_t bitrate_sum = 0;
+  for (int i = 0; i < streams; ++i) {
+    bitrate_sum += codec.simulcastStream[i].maxBitrate;
+  }
+  return bitrate_sum;
+}
+
+int SimulcastUtility::NumberOfSimulcastStreams(const VideoCodec& codec) {
+  int streams =
+      codec.numberOfSimulcastStreams < 1 ? 1 : codec.numberOfSimulcastStreams;
+  uint32_t simulcast_max_bitrate = SumStreamMaxBitrate(streams, codec);
+  if (simulcast_max_bitrate == 0) {
+    streams = 1;
+  }
+  return streams;
+}
+
+bool SimulcastUtility::ValidSimulcastParameters(const VideoCodec& codec,
+                                                int num_streams) {
+  // Check resolution.
+  if (codec.width != codec.simulcastStream[num_streams - 1].width ||
+      codec.height != codec.simulcastStream[num_streams - 1].height) {
+    return false;
+  }
+  for (int i = 0; i < num_streams; ++i) {
+    if (codec.width * codec.simulcastStream[i].height !=
+        codec.height * codec.simulcastStream[i].width) {
+      return false;
+    }
+  }
+  for (int i = 1; i < num_streams; ++i) {
+    if (codec.simulcastStream[i].width < codec.simulcastStream[i - 1].width) {
+      return false;
+    }
+  }
+
+  // Check frame-rate.
+  for (int i = 1; i < num_streams; ++i) {
+    if (fabs(codec.simulcastStream[i].maxFramerate -
+             codec.simulcastStream[i - 1].maxFramerate) > 1e-9) {
+      return false;
+    }
+  }
+
+  // Check temporal layers.
+  for (int i = 0; i < num_streams - 1; ++i) {
+    if (codec.simulcastStream[i].numberOfTemporalLayers !=
+        codec.simulcastStream[i + 1].numberOfTemporalLayers)
+      return false;
+  }
+  return true;
+}
+
+bool SimulcastUtility::IsConferenceModeScreenshare(const VideoCodec& codec) {
+  return codec.mode == VideoCodecMode::kScreensharing &&
+         codec.legacy_conference_mode;
+}
+
+int SimulcastUtility::NumberOfTemporalLayers(const VideoCodec& codec,
+                                             int spatial_id) {
+  uint8_t num_temporal_layers =
+      std::max<uint8_t>(1, codec.VP8().numberOfTemporalLayers);
+  if (codec.numberOfSimulcastStreams > 0) {
+    RTC_DCHECK_LT(spatial_id, codec.numberOfSimulcastStreams);
+    num_temporal_layers =
+        std::max(num_temporal_layers,
+                 codec.simulcastStream[spatial_id].numberOfTemporalLayers);
+  }
+  return num_temporal_layers;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/simulcast_utility.h b/third_party/libwebrtc/modules/video_coding/utility/simulcast_utility.h
new file mode 100644
index 0000000000..e25a594360
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/simulcast_utility.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_SIMULCAST_UTILITY_H_
+#define MODULES_VIDEO_CODING_UTILITY_SIMULCAST_UTILITY_H_
+
+#include <stdint.h>
+
+#include "api/video_codecs/video_codec.h"
+
+namespace webrtc {
+
+class SimulcastUtility {
+ public:
+  static uint32_t SumStreamMaxBitrate(int streams, const VideoCodec& codec);
+  static int NumberOfSimulcastStreams(const VideoCodec& codec);
+  static bool ValidSimulcastParameters(const VideoCodec& codec,
+                                       int num_streams);
+  static int NumberOfTemporalLayers(const VideoCodec& codec, int spatial_id);
+  // TODO(sprang): Remove this hack when ScreenshareLayers is gone.
+  static bool IsConferenceModeScreenshare(const VideoCodec& codec);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_SIMULCAST_UTILITY_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/vp8_constants.h b/third_party/libwebrtc/modules/video_coding/utility/vp8_constants.h
new file mode 100644
index 0000000000..9321864dbc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/vp8_constants.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_VP8_CONSTANTS_H_
+#define MODULES_VIDEO_CODING_UTILITY_VP8_CONSTANTS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+
+namespace webrtc {
+
+// QP level below which VP8 variable framerate and zero hertz screencast reduces
+// framerate due to diminishing quality enhancement returns.
+constexpr int kVp8SteadyStateQpThreshold = 15;
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_VP8_CONSTANTS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/vp8_header_parser.cc b/third_party/libwebrtc/modules/video_coding/utility/vp8_header_parser.cc
new file mode 100644
index 0000000000..80026f9a0f
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/vp8_header_parser.cc
@@ -0,0 +1,200 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/utility/vp8_header_parser.h"
+
+#include "rtc_base/logging.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+
+namespace vp8 {
+namespace {
+const size_t kCommonPayloadHeaderLength = 3;
+const size_t kKeyPayloadHeaderLength = 10;
+const int kMbFeatureTreeProbs = 3;
+const int kNumMbSegments = 4;
+const int kNumRefLfDeltas = 4;
+const int kNumModeLfDeltas = 4;
+
+}  // namespace
+
+// Bitstream parser according to
+// https://tools.ietf.org/html/rfc6386#section-7.3
+void VP8InitBitReader(VP8BitReader* const br,
+                      const uint8_t* start,
+                      const uint8_t* end) {
+  br->range_ = 255;
+  br->buf_ = start;
+  br->buf_end_ = end;
+  br->value_ = 0;
+  br->bits_ = 0;
+
+  // Read 2 bytes.
+  int i = 0;
+  while (++i <= 2) {
+    if (br->buf_ != br->buf_end_) {
+      br->value_ = br->value_ << 8 | *br->buf_++;
+    } else {
+      br->value_ = br->value_ << 8;
+    }
+  }
+}
+
+// Bit decoder according to https://tools.ietf.org/html/rfc6386#section-7.3
+// Reads one bit from the bitstream, given that it has probability prob/256 to
+// be 1.
+int Vp8BitReaderGetBool(VP8BitReader* br, int prob) {
+  uint32_t split = 1 + (((br->range_ - 1) * prob) >> 8);
+  uint32_t split_hi = split << 8;
+  int retval = 0;
+  if (br->value_ >= split_hi) {
+    retval = 1;
+    br->range_ -= split;
+    br->value_ -= split_hi;
+  } else {
+    retval = 0;
+    br->range_ = split;
+  }
+
+  while (br->range_ < 128) {
+    br->value_ <<= 1;
+    br->range_ <<= 1;
+    if (++br->bits_ == 8) {
+      br->bits_ = 0;
+      if (br->buf_ != br->buf_end_) {
+        br->value_ |= *br->buf_++;
+      }
+    }
+  }
+  return retval;
+}
+
+uint32_t VP8GetValue(VP8BitReader* br, int num_bits) {
+  uint32_t v = 0;
+  while (num_bits--) {
+    // According to https://tools.ietf.org/html/rfc6386
+    // Probability 128/256 is used to encode header fields.
+    v = (v << 1) | Vp8BitReaderGetBool(br, 128);
+  }
+  return v;
+}
+
+// Not a read_signed_literal() from RFC 6386!
+// This one is used to read e.g. quantizer_update, which is written as:
+// L(num_bits), sign-bit.
+int32_t VP8GetSignedValue(VP8BitReader* br, int num_bits) {
+  int v = VP8GetValue(br, num_bits);
+  int sign = VP8GetValue(br, 1);
+  return sign ? -v : v;
+}
+
+static void ParseSegmentHeader(VP8BitReader* br) {
+  int use_segment = VP8GetValue(br, 1);
+  if (use_segment) {
+    int update_map = VP8GetValue(br, 1);
+    if (VP8GetValue(br, 1)) {  // update_segment_feature_data.
+      VP8GetValue(br, 1);      // segment_feature_mode.
+      int s;
+      for (s = 0; s < kNumMbSegments; ++s) {
+        bool quantizer_update = VP8GetValue(br, 1);
+        if (quantizer_update) {
+          VP8GetSignedValue(br, 7);
+        }
+      }
+      for (s = 0; s < kNumMbSegments; ++s) {
+        bool loop_filter_update = VP8GetValue(br, 1);
+        if (loop_filter_update) {
+          VP8GetSignedValue(br, 6);
+        }
+      }
+    }
+    if (update_map) {
+      int s;
+      for (s = 0; s < kMbFeatureTreeProbs; ++s) {
+        bool segment_prob_update = VP8GetValue(br, 1);
+        if (segment_prob_update) {
+          VP8GetValue(br, 8);
+        }
+      }
+    }
+  }
+}
+
+static void ParseFilterHeader(VP8BitReader* br) {
+  VP8GetValue(br, 1);  // filter_type.
+  VP8GetValue(br, 6);  // loop_filter_level.
+  VP8GetValue(br, 3);  // sharpness_level.
+
+  // mb_lf_adjustments.
+  int loop_filter_adj_enable = VP8GetValue(br, 1);
+  if (loop_filter_adj_enable) {
+    int mode_ref_lf_delta_update = VP8GetValue(br, 1);
+    if (mode_ref_lf_delta_update) {
+      int i;
+      for (i = 0; i < kNumRefLfDeltas; ++i) {
+        int ref_frame_delta_update_flag = VP8GetValue(br, 1);
+        if (ref_frame_delta_update_flag) {
+          VP8GetSignedValue(br, 6);  // delta_magnitude.
+        }
+      }
+      for (i = 0; i < kNumModeLfDeltas; ++i) {
+        int mb_mode_delta_update_flag = VP8GetValue(br, 1);
+        if (mb_mode_delta_update_flag) {
+          VP8GetSignedValue(br, 6);  // delta_magnitude.
+        }
+      }
+    }
+  }
+}
+
+bool GetQp(const uint8_t* buf, size_t length, int* qp) {
+  if (length < kCommonPayloadHeaderLength) {
+    RTC_LOG(LS_WARNING) << "Failed to get QP, invalid length.";
+    return false;
+  }
+  VP8BitReader br;
+  const uint32_t bits = buf[0] | (buf[1] << 8) | (buf[2] << 16);
+  int key_frame = !(bits & 1);
+  // Size of first partition in bytes.
+  uint32_t partition_length = (bits >> 5);
+  size_t header_length = kCommonPayloadHeaderLength;
+  if (key_frame) {
+    header_length = kKeyPayloadHeaderLength;
+  }
+  if (header_length + partition_length > length) {
+    RTC_LOG(LS_WARNING) << "Failed to get QP, invalid length: " << length;
+    return false;
+  }
+  buf += header_length;
+
+  VP8InitBitReader(&br, buf, buf + partition_length);
+  if (key_frame) {
+    // Color space and pixel type.
+    VP8GetValue(&br, 1);
+    VP8GetValue(&br, 1);
+  }
+  ParseSegmentHeader(&br);
+  ParseFilterHeader(&br);
+  // Parse log2_nbr_of_dct_partitions value.
+  VP8GetValue(&br, 2);
+  // Base QP.
+  const int base_q0 = VP8GetValue(&br, 7);
+  if (br.buf_ == br.buf_end_) {
+    RTC_LOG(LS_WARNING) << "Failed to get QP, bitstream is truncated or"
+                           " corrupted.";
+    return false;
+  }
+  *qp = base_q0;
+  return true;
+}
+
+}  // namespace vp8
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/vp8_header_parser.h b/third_party/libwebrtc/modules/video_coding/utility/vp8_header_parser.h
new file mode 100644
index 0000000000..dbad999dc8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/vp8_header_parser.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_VP8_HEADER_PARSER_H_
+#define MODULES_VIDEO_CODING_UTILITY_VP8_HEADER_PARSER_H_
+
+#include <stdint.h>
+#include <stdio.h>
+
+namespace webrtc {
+
+namespace vp8 {
+
+typedef struct VP8BitReader VP8BitReader;
+struct VP8BitReader {
+  // Boolean decoder.
+  uint32_t value_;  // Current value (2 bytes).
+  uint32_t range_;  // Current range (always in [128..255] interval).
+  int bits_;        // Number of bits shifted out of value, at most 7.
+  // Read buffer.
+  const uint8_t* buf_;      // Next byte to be read.
+  const uint8_t* buf_end_;  // End of read buffer.
+};
+
+// Gets the QP, QP range: [0, 127].
+// Returns true on success, false otherwise.
+bool GetQp(const uint8_t* buf, size_t length, int* qp);
+
+}  // namespace vp8
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_VP8_HEADER_PARSER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/vp9_constants.h b/third_party/libwebrtc/modules/video_coding/utility/vp9_constants.h
new file mode 100644
index 0000000000..af2c701b82
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/vp9_constants.h
@@ -0,0 +1,198 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_VP9_CONSTANTS_H_
+#define MODULES_VIDEO_CODING_UTILITY_VP9_CONSTANTS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+
+namespace webrtc {
+
+// Number of frames that can be stored for future reference.
+constexpr size_t kVp9NumRefFrames = 8;
+// Number of frame contexts that can be store for future reference.
+constexpr size_t kVp9NumFrameContexts = 4;
+// Each inter frame can use up to 3 frames for reference.
+constexpr size_t kVp9RefsPerFrame = 3;
+// Number of values that can be decoded for mv_fr.
+constexpr size_t kVp9MvFrSize = 4;
+// Number of positions to search in motion vector prediction.
+constexpr size_t kVp9MvrefNeighbours = 8;
+// Number of contexts when decoding intra_mode .
+constexpr size_t kVp9BlockSizeGroups = 4;
+// Number of different block sizes used.
+constexpr size_t kVp9BlockSizes = 13;
+// Sentinel value to mark partition choices that are illegal.
+constexpr size_t kVp9BlockInvalid = 14;
+// Number of contexts when decoding partition.
+constexpr size_t kVp9PartitionContexts = 16;
+// Smallest size of a mode info block.
+constexpr size_t kVp9MiSize = 8;
+// Minimum  width  of a  tile  in  units  of  superblocks  (although tiles on
+// the right hand edge can be narrower).
+constexpr size_t kVp9MinTileWidth_B64 = 4;
+// Maximum width of a tile in units of superblocks.
+constexpr size_t kVp9MaxTileWidth_B64 = 64;
+// Number of motion vectors returned by find_mv_refs process.
+constexpr size_t kVp9MaxMvRefCandidates = 2;
+// Number of values that can be derived for ref_frame.
+constexpr size_t kVp9MaxRefFrames = 4;
+// Number of contexts for is_inter.
+constexpr size_t kVp9IsInterContexts = 4;
+// Number of contexts for comp_mode.
+constexpr size_t kVp9CompModeContexts = 5;
+// Number of contexts for single_ref and comp_ref.
+constexpr size_t kVp9RefContexts = 5;
+// Number of segments allowed in segmentation map.
+constexpr size_t kVp9MaxSegments = 8;
+// Index for quantizer segment feature.
+constexpr size_t kVp9SegLvlAlt_Q = 0;
+// Index for loop filter segment feature.
+constexpr size_t kVp9SegLvlAlt_L = 1;
+// Index for reference frame segment feature.
+constexpr size_t kVp9SegLvlRefFrame = 2;
+// Index for skip segment feature.
+constexpr size_t kVp9SegLvlSkip = 3;
+// Number of segment features.
+constexpr size_t kVp9SegLvlMax = 4;
+// Number of different plane types (Y or UV).
+constexpr size_t kVp9BlockTypes = 2;
+// Number of different prediction types (intra or inter).
+constexpr size_t kVp9RefTypes = 2;
+// Number of coefficient bands.
+constexpr size_t kVp9CoefBands = 6;
+// Number of contexts for decoding coefficients.
+constexpr size_t kVp9PrevCoefContexts = 6;
+// Number  of  coefficient  probabilities  that  are  directly transmitted.
+constexpr size_t kVp9UnconstrainedNodes = 3;
+// Number of contexts for transform size.
+constexpr size_t kVp9TxSizeContexts = 2;
+// Number of values for interp_filter.
+constexpr size_t kVp9SwitchableFilters = 3;
+// Number of contexts for interp_filter.
+constexpr size_t kVp9InterpFilterContexts = 4;
+// Number of contexts for decoding skip.
+constexpr size_t kVp9SkipContexts = 3;
+// Number of values for partition.
+constexpr size_t kVp9PartitionTypes = 4;
+// Number of values for tx_size.
+constexpr size_t kVp9TxSizes = 4;
+// Number of values for tx_mode.
+constexpr size_t kVp9TxModes = 5;
+// Inverse transform rows with DCT and columns with DCT.
+constexpr size_t kVp9DctDct = 0;
+// Inverse transform rows with DCT and columns with ADST.
+constexpr size_t kVp9AdstDct = 1;
+// Inverse transform rows with ADST and columns with DCT.
+constexpr size_t kVp9DctAdst = 2;
+// Inverse transform rows with ADST and columns with ADST.
+constexpr size_t kVp9AdstAdst = 3;
+// Number of values for y_mode.
+constexpr size_t kVp9MbModeCount = 14;
+// Number of values for intra_mode.
+constexpr size_t kVp9IntraModes = 10;
+// Number of values for inter_mode.
+constexpr size_t kVp9InterModes = 4;
+// Number of contexts for inter_mode.
+constexpr size_t kVp9InterModeContexts = 7;
+// Number of values for mv_joint.
+constexpr size_t kVp9MvJoints = 4;
+// Number of values for mv_class.
+constexpr size_t kVp9MvClasses = 11;
+// Number of values for mv_class0_bit.
+constexpr size_t kVp9Class0Size = 2;
+// Maximum number of bits for decoding motion vectors.
+constexpr size_t kVp9MvOffsetBits = 10;
+// Number of values allowed for a probability adjustment.
+constexpr size_t kVp9MaxProb = 255;
+// Number of different mode types for loop filtering.
+constexpr size_t kVp9MaxModeLfDeltas = 2;
+// Threshold at which motion vectors are considered large.
+constexpr size_t kVp9CompandedMvrefThresh = 8;
+// Maximum value used for loop filtering.
+constexpr size_t kVp9MaxLoopFilter = 63;
+// Number of bits of precision when scaling reference frames.
+constexpr size_t kVp9RefScaleShift = 14;
+// Number of bits of precision when performing inter prediction.
+constexpr size_t kVp9SubpelBits = 4;
+// 1 << kVp9SubpelBits.
+constexpr size_t kVp9SubpelShifts = 16;
+// kVp9SubpelShifts - 1.
+constexpr size_t kVp9SubpelMask = 15;
+// Value used when clipping motion vectors.
+constexpr size_t kVp9MvBorder = 128;
+// Value used when clipping motion vectors.
+constexpr size_t kVp9InterpExtend = 4;
+// Value used when clipping motion vectors.
+constexpr size_t kVp9Borderinpixels = 160;
+// Value used in adapting probabilities.
+constexpr size_t kVp9MaxUpdateFactor = 128;
+// Value used in adapting probabilities.
+constexpr size_t kVp9CountSat = 20;
+// Both candidates use ZEROMV.
+constexpr size_t kVp9BothZero = 0;
+// One  candidate uses ZEROMV, one uses NEARMV or NEARESTMV.
+constexpr size_t kVp9ZeroPlusPredicted = 1;
+// Both candidates use NEARMV or NEARESTMV.
+constexpr size_t kVp9BothPredicted = 2;
+// One candidate uses NEWMV, one uses ZEROMV.
+constexpr size_t kVp9NewPlusNonIntra = 3;
+// Both candidates use NEWMV.
+constexpr size_t kVp9BothNew = 4;
+// One candidate uses intra prediction, one uses inter prediction.
+constexpr size_t kVp9IntraPlusNonIntra = 5;
+// Both candidates use intra prediction.
+constexpr size_t kVp9BothIntra = 6;
+// Sentinel value marking a case that can never occur.
+constexpr size_t kVp9InvalidCase = 9;
+
+enum class Vp9TxMode : uint8_t {
+  kOnly4X4 = 0,
+  kAllow8X8 = 1,
+  kAllow16x16 = 2,
+  kAllow32x32 = 3,
+  kTxModeSelect = 4
+};
+
+enum Vp9BlockSize : uint8_t {
+  kBlock4X4 = 0,
+  kBlock4X8 = 1,
+  kBlock8X4 = 2,
+  kBlock8X8 = 3,
+  kBlock8X16 = 4,
+  kBlock16X8 = 5,
+  kBlock16X16 = 6,
+  kBlock16X32 = 7,
+  kBlock32X16 = 8,
+  kBlock32X32 = 9,
+  kBlock32X64 = 10,
+  kBlock64X32 = 11,
+  kBlock64X64 = 12
+};
+
+enum Vp9Partition : uint8_t {
+  kPartitionNone = 0,
+  kPartitionHorizontal = 1,
+  kPartitionVertical = 2,
+  kPartitionSplit = 3
+};
+
+enum class Vp9ReferenceMode : uint8_t {
+  kSingleReference = 0,
+  kCompoundReference = 1,
+  kReferenceModeSelect = 2,
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_VP9_CONSTANTS_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.cc b/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
new file mode 100644
index 0000000000..bf9d51f692
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
@@ -0,0 +1,533 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+
+#include "absl/numeric/bits.h"
+#include "absl/strings/string_view.h"
+#include "rtc_base/bitstream_reader.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace {
+const size_t kVp9NumRefsPerFrame = 3;
+const size_t kVp9MaxRefLFDeltas = 4;
+const size_t kVp9MaxModeLFDeltas = 2;
+const size_t kVp9MinTileWidthB64 = 4;
+const size_t kVp9MaxTileWidthB64 = 64;
+
+void Vp9ReadColorConfig(BitstreamReader& br,
+                        Vp9UncompressedHeader* frame_info) {
+  if (frame_info->profile == 2 || frame_info->profile == 3) {
+    frame_info->bit_detph =
+        br.Read<bool>() ? Vp9BitDept::k12Bit : Vp9BitDept::k10Bit;
+  } else {
+    frame_info->bit_detph = Vp9BitDept::k8Bit;
+  }
+
+  frame_info->color_space = static_cast<Vp9ColorSpace>(br.ReadBits(3));
+
+  if (frame_info->color_space != Vp9ColorSpace::CS_RGB) {
+    frame_info->color_range =
+        br.Read<bool>() ? Vp9ColorRange::kFull : Vp9ColorRange::kStudio;
+
+    if (frame_info->profile == 1 || frame_info->profile == 3) {
+      static constexpr Vp9YuvSubsampling kSubSamplings[] = {
+          Vp9YuvSubsampling::k444, Vp9YuvSubsampling::k440,
+          Vp9YuvSubsampling::k422, Vp9YuvSubsampling::k420};
+      frame_info->sub_sampling = kSubSamplings[br.ReadBits(2)];
+
+      if (br.Read<bool>()) {
+        RTC_LOG(LS_WARNING) << "Failed to parse header. Reserved bit set.";
+        br.Invalidate();
+        return;
+      }
+    } else {
+      // Profile 0 or 2.
+      frame_info->sub_sampling = Vp9YuvSubsampling::k420;
+    }
+  } else {
+    // SRGB
+    frame_info->color_range = Vp9ColorRange::kFull;
+    if (frame_info->profile == 1 || frame_info->profile == 3) {
+      frame_info->sub_sampling = Vp9YuvSubsampling::k444;
+      if (br.Read<bool>()) {
+        RTC_LOG(LS_WARNING) << "Failed to parse header. Reserved bit set.";
+        br.Invalidate();
+      }
+    } else {
+      RTC_LOG(LS_WARNING) << "Failed to parse header. 4:4:4 color not supported"
+                             " in profile 0 or 2.";
+      br.Invalidate();
+    }
+  }
+}
+
+void ReadRefreshFrameFlags(BitstreamReader& br,
+                           Vp9UncompressedHeader* frame_info) {
+  // Refresh frame flags.
+  uint8_t flags = br.Read<uint8_t>();
+  for (int i = 0; i < 8; ++i) {
+    frame_info->updated_buffers.set(i, (flags & (0x01 << (7 - i))) != 0);
+  }
+}
+
+void Vp9ReadFrameSize(BitstreamReader& br, Vp9UncompressedHeader* frame_info) {
+  // 16 bits: frame (width|height) - 1.
+  frame_info->frame_width = br.Read<uint16_t>() + 1;
+  frame_info->frame_height = br.Read<uint16_t>() + 1;
+}
+
+void Vp9ReadRenderSize(size_t total_buffer_size_bits,
+                       BitstreamReader& br,
+                       Vp9UncompressedHeader* frame_info) {
+  // render_and_frame_size_different
+  if (br.Read<bool>()) {
+    frame_info->render_size_offset_bits =
+        total_buffer_size_bits - br.RemainingBitCount();
+    // 16 bits: render (width|height) - 1.
+    frame_info->render_width = br.Read<uint16_t>() + 1;
+    frame_info->render_height = br.Read<uint16_t>() + 1;
+  } else {
+    frame_info->render_height = frame_info->frame_height;
+    frame_info->render_width = frame_info->frame_width;
+  }
+}
+
+void Vp9ReadFrameSizeFromRefs(BitstreamReader& br,
+                              Vp9UncompressedHeader* frame_info) {
+  for (size_t i = 0; i < kVp9NumRefsPerFrame; i++) {
+    // Size in refs.
+    if (br.Read<bool>()) {
+      frame_info->infer_size_from_reference = frame_info->reference_buffers[i];
+      return;
+    }
+  }
+
+  Vp9ReadFrameSize(br, frame_info);
+}
+
+void Vp9ReadLoopfilter(BitstreamReader& br) {
+  // 6 bits: filter level.
+  // 3 bits: sharpness level.
+  br.ConsumeBits(9);
+
+  if (!br.Read<bool>()) {  // mode_ref_delta_enabled
+    return;
+  }
+  if (!br.Read<bool>()) {  // mode_ref_delta_update
+    return;
+  }
+
+  for (size_t i = 0; i < kVp9MaxRefLFDeltas; i++) {
+    if (br.Read<bool>()) {  // update_ref_delta
+      br.ConsumeBits(7);
+    }
+  }
+  for (size_t i = 0; i < kVp9MaxModeLFDeltas; i++) {
+    if (br.Read<bool>()) {  // update_mode_delta
+      br.ConsumeBits(7);
+    }
+  }
+}
+
+void Vp9ReadQp(BitstreamReader& br, Vp9UncompressedHeader* frame_info) {
+  frame_info->base_qp = br.Read<uint8_t>();
+
+  // yuv offsets
+  frame_info->is_lossless = frame_info->base_qp == 0;
+  for (int i = 0; i < 3; ++i) {
+    if (br.Read<bool>()) {  // if delta_coded
+      // delta_q is a signed integer with leading 4 bits containing absolute
+      // value and last bit containing sign. There are are two ways to represent
+      // zero with such encoding.
+      if ((br.ReadBits(5) & 0b1111'0) != 0) {  // delta_q
+        frame_info->is_lossless = false;
+      }
+    }
+  }
+}
+
+void Vp9ReadSegmentationParams(BitstreamReader& br,
+                               Vp9UncompressedHeader* frame_info) {
+  constexpr int kSegmentationFeatureBits[kVp9SegLvlMax] = {8, 6, 2, 0};
+  constexpr bool kSegmentationFeatureSigned[kVp9SegLvlMax] = {true, true, false,
+                                                              false};
+
+  frame_info->segmentation_enabled = br.Read<bool>();
+  if (!frame_info->segmentation_enabled) {
+    return;
+  }
+
+  if (br.Read<bool>()) {  // update_map
+    frame_info->segmentation_tree_probs.emplace();
+    for (int i = 0; i < 7; ++i) {
+      if (br.Read<bool>()) {
+        (*frame_info->segmentation_tree_probs)[i] = br.Read<uint8_t>();
+      } else {
+        (*frame_info->segmentation_tree_probs)[i] = 255;
+      }
+    }
+
+    // temporal_update
+    frame_info->segmentation_pred_prob.emplace();
+    if (br.Read<bool>()) {
+      for (int i = 0; i < 3; ++i) {
+        if (br.Read<bool>()) {
+          (*frame_info->segmentation_pred_prob)[i] = br.Read<uint8_t>();
+        } else {
+          (*frame_info->segmentation_pred_prob)[i] = 255;
+        }
+      }
+    } else {
+      frame_info->segmentation_pred_prob->fill(255);
+    }
+  }
+
+  if (br.Read<bool>()) {  // segmentation_update_data
+    frame_info->segmentation_is_delta = br.Read<bool>();
+    for (size_t i = 0; i < kVp9MaxSegments; ++i) {
+      for (size_t j = 0; j < kVp9SegLvlMax; ++j) {
+        if (!br.Read<bool>()) {  // feature_enabled
+          continue;
+        }
+        if (kSegmentationFeatureBits[j] == 0) {
+          // No feature bits used and no sign, just mark it and return.
+          frame_info->segmentation_features[i][j] = 1;
+          continue;
+        }
+        frame_info->segmentation_features[i][j] =
+            br.ReadBits(kSegmentationFeatureBits[j]);
+        if (kSegmentationFeatureSigned[j] && br.Read<bool>()) {
+          (*frame_info->segmentation_features[i][j]) *= -1;
+        }
+      }
+    }
+  }
+}
+
+void Vp9ReadTileInfo(BitstreamReader& br, Vp9UncompressedHeader* frame_info) {
+  size_t mi_cols = (frame_info->frame_width + 7) >> 3;
+  size_t sb64_cols = (mi_cols + 7) >> 3;
+
+  size_t min_log2 = 0;
+  while ((kVp9MaxTileWidthB64 << min_log2) < sb64_cols) {
+    ++min_log2;
+  }
+
+  size_t max_log2 = 1;
+  while ((sb64_cols >> max_log2) >= kVp9MinTileWidthB64) {
+    ++max_log2;
+  }
+  --max_log2;
+
+  frame_info->tile_cols_log2 = min_log2;
+  while (frame_info->tile_cols_log2 < max_log2) {
+    if (br.Read<bool>()) {
+      ++frame_info->tile_cols_log2;
+    } else {
+      break;
+    }
+  }
+  frame_info->tile_rows_log2 = 0;
+  if (br.Read<bool>()) {
+    ++frame_info->tile_rows_log2;
+    if (br.Read<bool>()) {
+      ++frame_info->tile_rows_log2;
+    }
+  }
+}
+
+const Vp9InterpolationFilter kLiteralToType[4] = {
+    Vp9InterpolationFilter::kEightTapSmooth, Vp9InterpolationFilter::kEightTap,
+    Vp9InterpolationFilter::kEightTapSharp, Vp9InterpolationFilter::kBilinear};
+}  // namespace
+
+std::string Vp9UncompressedHeader::ToString() const {
+  char buf[1024];
+  rtc::SimpleStringBuilder oss(buf);
+
+  oss << "Vp9UncompressedHeader { "
+      << "profile = " << profile;
+
+  if (show_existing_frame) {
+    oss << ", show_existing_frame = " << *show_existing_frame << " }";
+    return oss.str();
+  }
+
+  oss << ", frame type = " << (is_keyframe ? "key" : "delta")
+      << ", show_frame = " << (show_frame ? "true" : "false")
+      << ", error_resilient = " << (error_resilient ? "true" : "false");
+
+  oss << ", bit_depth = ";
+  switch (bit_detph) {
+    case Vp9BitDept::k8Bit:
+      oss << "8bit";
+      break;
+    case Vp9BitDept::k10Bit:
+      oss << "10bit";
+      break;
+    case Vp9BitDept::k12Bit:
+      oss << "12bit";
+      break;
+  }
+
+  if (color_space) {
+    oss << ", color_space = ";
+    switch (*color_space) {
+      case Vp9ColorSpace::CS_UNKNOWN:
+        oss << "unknown";
+        break;
+      case Vp9ColorSpace::CS_BT_601:
+        oss << "CS_BT_601 Rec. ITU-R BT.601-7";
+        break;
+      case Vp9ColorSpace::CS_BT_709:
+        oss << "Rec. ITU-R BT.709-6";
+        break;
+      case Vp9ColorSpace::CS_SMPTE_170:
+        oss << "SMPTE-170";
+        break;
+      case Vp9ColorSpace::CS_SMPTE_240:
+        oss << "SMPTE-240";
+        break;
+      case Vp9ColorSpace::CS_BT_2020:
+        oss << "Rec. ITU-R BT.2020-2";
+        break;
+      case Vp9ColorSpace::CS_RESERVED:
+        oss << "Reserved";
+        break;
+      case Vp9ColorSpace::CS_RGB:
+        oss << "sRGB (IEC 61966-2-1)";
+        break;
+    }
+  }
+
+  if (color_range) {
+    oss << ", color_range = ";
+    switch (*color_range) {
+      case Vp9ColorRange::kFull:
+        oss << "full";
+        break;
+      case Vp9ColorRange::kStudio:
+        oss << "studio";
+        break;
+    }
+  }
+
+  if (sub_sampling) {
+    oss << ", sub_sampling = ";
+    switch (*sub_sampling) {
+      case Vp9YuvSubsampling::k444:
+        oss << "444";
+        break;
+      case Vp9YuvSubsampling::k440:
+        oss << "440";
+        break;
+      case Vp9YuvSubsampling::k422:
+        oss << "422";
+        break;
+      case Vp9YuvSubsampling::k420:
+        oss << "420";
+        break;
+    }
+  }
+
+  if (infer_size_from_reference) {
+    oss << ", infer_frame_resolution_from = " << *infer_size_from_reference;
+  } else {
+    oss << ", frame_width = " << frame_width
+        << ", frame_height = " << frame_height;
+  }
+  if (render_width != 0 && render_height != 0) {
+    oss << ", render_width = " << render_width
+        << ", render_height = " << render_height;
+  }
+
+  oss << ", base qp = " << base_qp;
+  if (reference_buffers[0] != -1) {
+    oss << ", last_buffer = " << reference_buffers[0];
+  }
+  if (reference_buffers[1] != -1) {
+    oss << ", golden_buffer = " << reference_buffers[1];
+  }
+  if (reference_buffers[2] != -1) {
+    oss << ", altref_buffer = " << reference_buffers[2];
+  }
+
+  oss << ", updated buffers = { ";
+  bool first = true;
+  for (int i = 0; i < 8; ++i) {
+    if (updated_buffers.test(i)) {
+      if (first) {
+        first = false;
+      } else {
+        oss << ", ";
+      }
+      oss << i;
+    }
+  }
+  oss << " }";
+
+  oss << ", compressed_header_size_bytes = " << compressed_header_size;
+
+  oss << " }";
+  return oss.str();
+}
+
+void Parse(BitstreamReader& br,
+           Vp9UncompressedHeader* frame_info,
+           bool qp_only) {
+  const size_t total_buffer_size_bits = br.RemainingBitCount();
+
+  // Frame marker.
+  if (br.ReadBits(2) != 0b10) {
+    RTC_LOG(LS_WARNING) << "Failed to parse header. Frame marker should be 2.";
+    br.Invalidate();
+    return;
+  }
+
+  // Profile has low bit first.
+  frame_info->profile = br.ReadBit();
+  frame_info->profile |= br.ReadBit() << 1;
+  if (frame_info->profile > 2 && br.Read<bool>()) {
+    RTC_LOG(LS_WARNING)
+        << "Failed to parse header. Unsupported bitstream profile.";
+    br.Invalidate();
+    return;
+  }
+
+  // Show existing frame.
+  if (br.Read<bool>()) {
+    frame_info->show_existing_frame = br.ReadBits(3);
+    return;
+  }
+
+  // Frame type: KEY_FRAME(0), INTER_FRAME(1).
+  frame_info->is_keyframe = !br.Read<bool>();
+  frame_info->show_frame = br.Read<bool>();
+  frame_info->error_resilient = br.Read<bool>();
+
+  if (frame_info->is_keyframe) {
+    if (br.ReadBits(24) != 0x498342) {
+      RTC_LOG(LS_WARNING) << "Failed to parse header. Invalid sync code.";
+      br.Invalidate();
+      return;
+    }
+
+    Vp9ReadColorConfig(br, frame_info);
+    Vp9ReadFrameSize(br, frame_info);
+    Vp9ReadRenderSize(total_buffer_size_bits, br, frame_info);
+
+    // Key-frames implicitly update all buffers.
+    frame_info->updated_buffers.set();
+  } else {
+    // Non-keyframe.
+    bool is_intra_only = false;
+    if (!frame_info->show_frame) {
+      is_intra_only = br.Read<bool>();
+    }
+    if (!frame_info->error_resilient) {
+      br.ConsumeBits(2);  // Reset frame context.
+    }
+
+    if (is_intra_only) {
+      if (br.ReadBits(24) != 0x498342) {
+        RTC_LOG(LS_WARNING) << "Failed to parse header. Invalid sync code.";
+        br.Invalidate();
+        return;
+      }
+
+      if (frame_info->profile > 0) {
+        Vp9ReadColorConfig(br, frame_info);
+      } else {
+        frame_info->color_space = Vp9ColorSpace::CS_BT_601;
+        frame_info->sub_sampling = Vp9YuvSubsampling::k420;
+        frame_info->bit_detph = Vp9BitDept::k8Bit;
+      }
+      frame_info->reference_buffers.fill(-1);
+      ReadRefreshFrameFlags(br, frame_info);
+      Vp9ReadFrameSize(br, frame_info);
+      Vp9ReadRenderSize(total_buffer_size_bits, br, frame_info);
+    } else {
+      ReadRefreshFrameFlags(br, frame_info);
+
+      frame_info->reference_buffers_sign_bias[0] = false;
+      for (size_t i = 0; i < kVp9NumRefsPerFrame; i++) {
+        frame_info->reference_buffers[i] = br.ReadBits(3);
+        frame_info->reference_buffers_sign_bias[Vp9ReferenceFrame::kLast + i] =
+            br.Read<bool>();
+      }
+
+      Vp9ReadFrameSizeFromRefs(br, frame_info);
+      Vp9ReadRenderSize(total_buffer_size_bits, br, frame_info);
+
+      frame_info->allow_high_precision_mv = br.Read<bool>();
+
+      // Interpolation filter.
+      if (br.Read<bool>()) {
+        frame_info->interpolation_filter = Vp9InterpolationFilter::kSwitchable;
+      } else {
+        frame_info->interpolation_filter = kLiteralToType[br.ReadBits(2)];
+      }
+    }
+  }
+
+  if (!frame_info->error_resilient) {
+    // 1 bit: Refresh frame context.
+    // 1 bit: Frame parallel decoding mode.
+    br.ConsumeBits(2);
+  }
+
+  // Frame context index.
+  frame_info->frame_context_idx = br.ReadBits(2);
+
+  Vp9ReadLoopfilter(br);
+
+  // Read base QP.
+  Vp9ReadQp(br, frame_info);
+
+  if (qp_only) {
+    // Not interested in the rest of the header, return early.
+    return;
+  }
+
+  Vp9ReadSegmentationParams(br, frame_info);
+  Vp9ReadTileInfo(br, frame_info);
+  frame_info->compressed_header_size = br.Read<uint16_t>();
+  frame_info->uncompressed_header_size =
+      (total_buffer_size_bits / 8) - (br.RemainingBitCount() / 8);
+}
+
+absl::optional<Vp9UncompressedHeader> ParseUncompressedVp9Header(
+    rtc::ArrayView<const uint8_t> buf) {
+  BitstreamReader reader(buf);
+  Vp9UncompressedHeader frame_info;
+  Parse(reader, &frame_info, /*qp_only=*/false);
+  if (reader.Ok() && frame_info.frame_width > 0) {
+    return frame_info;
+  }
+  return absl::nullopt;
+}
+
+namespace vp9 {
+
+bool GetQp(const uint8_t* buf, size_t length, int* qp) {
+  BitstreamReader reader(rtc::MakeArrayView(buf, length));
+  Vp9UncompressedHeader frame_info;
+  Parse(reader, &frame_info, /*qp_only=*/true);
+  if (!reader.Ok()) {
+    return false;
+  }
+  *qp = frame_info.base_qp;
+  return true;
+}
+
+}  // namespace vp9
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.h b/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.h
new file mode 100644
index 0000000000..8d1b88c3d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.h
@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_UTILITY_VP9_UNCOMPRESSED_HEADER_PARSER_H_
+#define MODULES_VIDEO_CODING_UTILITY_VP9_UNCOMPRESSED_HEADER_PARSER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <array>
+#include <bitset>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/video_coding/utility/vp9_constants.h"
+
+namespace webrtc {
+
+namespace vp9 {
+
+// Gets the QP, QP range: [0, 255].
+// Returns true on success, false otherwise.
+bool GetQp(const uint8_t* buf, size_t length, int* qp);
+
+}  // namespace vp9
+
+// Bit depth per channel. Support varies by profile.
+enum class Vp9BitDept : uint8_t {
+  k8Bit = 8,
+  k10Bit = 10,
+  k12Bit = 12,
+};
+
+enum class Vp9ColorSpace : uint8_t {
+  CS_UNKNOWN = 0,    // Unknown (in this case the color space must be signaled
+                     // outside the VP9 bitstream).
+  CS_BT_601 = 1,     // CS_BT_601 Rec. ITU-R BT.601-7
+  CS_BT_709 = 2,     // Rec. ITU-R BT.709-6
+  CS_SMPTE_170 = 3,  // SMPTE-170
+  CS_SMPTE_240 = 4,  // SMPTE-240
+  CS_BT_2020 = 5,    // Rec. ITU-R BT.2020-2
+  CS_RESERVED = 6,   // Reserved
+  CS_RGB = 7,        // sRGB (IEC 61966-2-1)
+};
+
+enum class Vp9ColorRange {
+  kStudio,  // Studio swing:
+            // For BitDepth equals 8:
+            //     Y is between 16 and 235 inclusive.
+            //     U and V are between 16 and 240 inclusive.
+            // For BitDepth equals 10:
+            //     Y is between 64 and 940 inclusive.
+            //     U and V are between 64 and 960 inclusive.
+            // For BitDepth equals 12:
+            //     Y is between 256 and 3760.
+            //     U and V are between 256 and 3840 inclusive.
+  kFull     // Full swing; no restriction on Y, U, V values.
+};
+
+enum class Vp9YuvSubsampling {
+  k444,
+  k440,
+  k422,
+  k420,
+};
+
+enum Vp9ReferenceFrame : int {
+  kNone = -1,
+  kIntra = 0,
+  kLast = 1,
+  kGolden = 2,
+  kAltref = 3,
+};
+
+enum class Vp9InterpolationFilter : uint8_t {
+  kEightTap = 0,
+  kEightTapSmooth = 1,
+  kEightTapSharp = 2,
+  kBilinear = 3,
+  kSwitchable = 4
+};
+
+struct Vp9UncompressedHeader {
+  int profile = 0;  // Profiles 0-3 are valid.
+  absl::optional<uint8_t> show_existing_frame;
+  bool is_keyframe = false;
+  bool show_frame = false;
+  bool error_resilient = false;
+  Vp9BitDept bit_detph = Vp9BitDept::k8Bit;
+  absl::optional<Vp9ColorSpace> color_space;
+  absl::optional<Vp9ColorRange> color_range;
+  absl::optional<Vp9YuvSubsampling> sub_sampling;
+  int frame_width = 0;
+  int frame_height = 0;
+  int render_width = 0;
+  int render_height = 0;
+  // Width/height of the tiles used (in units of 8x8 blocks).
+  size_t tile_cols_log2 = 0;  // tile_cols = 1 << tile_cols_log2
+  size_t tile_rows_log2 = 0;  // tile_rows = 1 << tile_rows_log2
+  absl::optional<size_t> render_size_offset_bits;
+  Vp9InterpolationFilter interpolation_filter =
+      Vp9InterpolationFilter::kEightTap;
+  bool allow_high_precision_mv = false;
+  int base_qp = 0;
+  bool is_lossless = false;
+  uint8_t frame_context_idx = 0;
+
+  bool segmentation_enabled = false;
+  absl::optional<std::array<uint8_t, 7>> segmentation_tree_probs;
+  absl::optional<std::array<uint8_t, 3>> segmentation_pred_prob;
+  bool segmentation_is_delta = false;
+  std::array<std::array<absl::optional<int>, kVp9SegLvlMax>, kVp9MaxSegments>
+      segmentation_features;
+
+  // Which of the 8 reference buffers may be used as references for this frame.
+  // -1 indicates not used (e.g. {-1, -1, -1} for intra-only frames).
+  std::array<int, kVp9RefsPerFrame> reference_buffers = {-1, -1, -1};
+  // Sign bias corresponding to reference buffers, where the index is a
+  // ReferenceFrame.
+  // false/0 indidate backwards reference, true/1 indicate forwards reference).
+  std::bitset<kVp9MaxRefFrames> reference_buffers_sign_bias = 0;
+
+  // Indicates which reference buffer [0,7] to infer the frame size from.
+  absl::optional<int> infer_size_from_reference;
+  // Which of the 8 reference buffers are updated by this frame.
+  std::bitset<kVp9NumRefFrames> updated_buffers = 0;
+
+  // Header sizes, in bytes.
+  uint32_t uncompressed_header_size = 0;
+  uint32_t compressed_header_size = 0;
+
+  bool is_intra_only() const {
+    return reference_buffers[0] == -1 && reference_buffers[1] == -1 &&
+           reference_buffers[2] == -1;
+  }
+
+  std::string ToString() const;
+};
+
+// Parses the uncompressed header and populates (most) values in a
+// UncompressedHeader struct. Returns nullopt on failure.
+absl::optional<Vp9UncompressedHeader> ParseUncompressedVp9Header(
+    rtc::ArrayView<const uint8_t> buf);
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_UTILITY_VP9_UNCOMPRESSED_HEADER_PARSER_H_
diff --git a/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser_unittest.cc b/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser_unittest.cc
new file mode 100644
index 0000000000..d8cc738e07
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser_unittest.cc
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace vp9 {
+using ::testing::AllOf;
+using ::testing::ElementsAre;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::Optional;
+
+TEST(Vp9UncompressedHeaderParserTest, FrameWithSegmentation) {
+  // Uncompressed header from a frame generated with libvpx.
+  // Encoded QVGA frame (SL0 of a VGA frame) that includes a segmentation.
+  const uint8_t kHeader[] = {
+      0x87, 0x01, 0x00, 0x00, 0x02, 0x7e, 0x01, 0xdf, 0x02, 0x7f, 0x01, 0xdf,
+      0xc6, 0x87, 0x04, 0x83, 0x83, 0x2e, 0x46, 0x60, 0x20, 0x38, 0x0c, 0x06,
+      0x03, 0xcd, 0x80, 0xc0, 0x60, 0x9f, 0xc5, 0x46, 0x00, 0x00, 0x00, 0x00,
+      0x2e, 0x73, 0xb7, 0xee, 0x22, 0x06, 0x81, 0x82, 0xd4, 0xef, 0xc3, 0x58,
+      0x1f, 0x12, 0xd2, 0x7b, 0x28, 0x1f, 0x80, 0xfc, 0x07, 0xe0, 0x00, 0x00};
+
+  absl::optional<Vp9UncompressedHeader> frame_info =
+      ParseUncompressedVp9Header(kHeader);
+  ASSERT_TRUE(frame_info.has_value());
+
+  EXPECT_FALSE(frame_info->is_keyframe);
+  EXPECT_TRUE(frame_info->error_resilient);
+  EXPECT_TRUE(frame_info->show_frame);
+  EXPECT_FALSE(frame_info->show_existing_frame);
+  EXPECT_EQ(frame_info->base_qp, 185);
+  EXPECT_EQ(frame_info->frame_width, 320);
+  EXPECT_EQ(frame_info->frame_height, 240);
+  EXPECT_EQ(frame_info->render_width, 640);
+  EXPECT_EQ(frame_info->render_height, 480);
+  EXPECT_TRUE(frame_info->allow_high_precision_mv);
+  EXPECT_EQ(frame_info->frame_context_idx, 0u);
+  EXPECT_EQ(frame_info->interpolation_filter,
+            Vp9InterpolationFilter::kSwitchable);
+  EXPECT_EQ(frame_info->is_lossless, false);
+  EXPECT_EQ(frame_info->profile, 0);
+  EXPECT_THAT(frame_info->reference_buffers, ElementsAre(0, 0, 0));
+  EXPECT_THAT(frame_info->reference_buffers_sign_bias, 0b0000);
+  EXPECT_EQ(frame_info->updated_buffers, 0b10000000);
+  EXPECT_EQ(frame_info->tile_cols_log2, 0u);
+  EXPECT_EQ(frame_info->tile_rows_log2, 0u);
+  EXPECT_EQ(frame_info->render_size_offset_bits, 64u);
+  EXPECT_EQ(frame_info->compressed_header_size, 23u);
+  EXPECT_EQ(frame_info->uncompressed_header_size, 37u);
+
+  EXPECT_TRUE(frame_info->segmentation_enabled);
+  EXPECT_FALSE(frame_info->segmentation_is_delta);
+  EXPECT_THAT(frame_info->segmentation_pred_prob,
+              Optional(ElementsAre(205, 1, 1)));
+  EXPECT_THAT(frame_info->segmentation_tree_probs,
+              Optional(ElementsAre(255, 255, 128, 1, 128, 128, 128)));
+  EXPECT_THAT(frame_info->segmentation_features[1][kVp9SegLvlAlt_Q], Eq(-63));
+  EXPECT_THAT(frame_info->segmentation_features[2][kVp9SegLvlAlt_Q], Eq(-81));
+}
+
+TEST(Vp9UncompressedHeaderParserTest, SegmentationWithDefaultPredProbs) {
+  const uint8_t kHeader[] = {0x90, 0x49, 0x83, 0x42, 0x80, 0x2e,
+                             0x30, 0x0,  0xb0, 0x0,  0x37, 0xff,
+                             0x06, 0x80, 0x0,  0x0,  0x0,  0x0};
+  absl::optional<Vp9UncompressedHeader> frame_info =
+      ParseUncompressedVp9Header(kHeader);
+  ASSERT_TRUE(frame_info.has_value());
+  EXPECT_THAT(frame_info->segmentation_pred_prob,
+              Optional(ElementsAre(255, 255, 255)));
+}
+
+TEST(Vp9UncompressedHeaderParserTest, SegmentationWithSkipLevel) {
+  const uint8_t kHeader[] = {0x90, 0x49, 0x83, 0x42, 0x80, 0x2e, 0x30, 0x00,
+                             0xb0, 0x00, 0x37, 0xff, 0x06, 0x80, 0x01, 0x08,
+                             0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
+  absl::optional<Vp9UncompressedHeader> frame_info =
+      ParseUncompressedVp9Header(kHeader);
+  ASSERT_TRUE(frame_info.has_value());
+  EXPECT_THAT(frame_info->segmentation_features[0][kVp9SegLvlSkip], Eq(1));
+}
+
+}  // namespace vp9
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/video_codec_initializer.cc b/third_party/libwebrtc/modules/video_coding/video_codec_initializer.cc
new file mode 100644
index 0000000000..e1885d74c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_codec_initializer.cc
@@ -0,0 +1,352 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/include/video_codec_initializer.h"
+
+#include <stdint.h>
+#include <string.h>
+
+#include <algorithm>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/units/data_rate.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/codecs/av1/av1_svc_config.h"
+#include "modules/video_coding/codecs/vp8/vp8_scalability.h"
+#include "modules/video_coding/codecs/vp9/svc_config.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/min_video_bitrate_experiment.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+bool VideoCodecInitializer::SetupCodec(const VideoEncoderConfig& config,
+                                       const std::vector<VideoStream>& streams,
+                                       VideoCodec* codec) {
+  if (config.codec_type == kVideoCodecMultiplex) {
+    VideoEncoderConfig associated_config = config.Copy();
+    associated_config.codec_type = kVideoCodecVP9;
+    if (!SetupCodec(associated_config, streams, codec)) {
+      RTC_LOG(LS_ERROR) << "Failed to create stereo encoder configuration.";
+      return false;
+    }
+    codec->codecType = kVideoCodecMultiplex;
+    return true;
+  }
+
+  *codec = VideoEncoderConfigToVideoCodec(config, streams);
+  return true;
+}
+
+// TODO(sprang): Split this up and separate the codec specific parts.
+VideoCodec VideoCodecInitializer::VideoEncoderConfigToVideoCodec(
+    const VideoEncoderConfig& config,
+    const std::vector<VideoStream>& streams) {
+  static const int kEncoderMinBitrateKbps = 30;
+  RTC_DCHECK(!streams.empty());
+  RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
+
+  VideoCodec video_codec;
+  video_codec.codecType = config.codec_type;
+
+  switch (config.content_type) {
+    case VideoEncoderConfig::ContentType::kRealtimeVideo:
+      video_codec.mode = VideoCodecMode::kRealtimeVideo;
+      break;
+    case VideoEncoderConfig::ContentType::kScreen:
+      video_codec.mode = VideoCodecMode::kScreensharing;
+      break;
+  }
+
+  video_codec.legacy_conference_mode =
+      config.content_type == VideoEncoderConfig::ContentType::kScreen &&
+      config.legacy_conference_mode;
+
+  video_codec.SetFrameDropEnabled(config.frame_drop_enabled);
+  video_codec.numberOfSimulcastStreams =
+      static_cast<unsigned char>(streams.size());
+  video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
+  bool codec_active = false;
+  // Active configuration might not be fully copied to `streams` for SVC yet.
+  // Therefore the `config` is checked here.
+  for (const VideoStream& stream : config.simulcast_layers) {
+    if (stream.active) {
+      codec_active = true;
+      break;
+    }
+  }
+  // Set active for the entire video codec for the non simulcast case.
+  video_codec.active = codec_active;
+  if (video_codec.minBitrate < kEncoderMinBitrateKbps)
+    video_codec.minBitrate = kEncoderMinBitrateKbps;
+  video_codec.timing_frame_thresholds = {kDefaultTimingFramesDelayMs,
+                                         kDefaultOutlierFrameSizePercent};
+  RTC_DCHECK_LE(streams.size(), kMaxSimulcastStreams);
+
+  int max_framerate = 0;
+
+  absl::optional<ScalabilityMode> scalability_mode =
+      streams[0].scalability_mode;
+  for (size_t i = 0; i < streams.size(); ++i) {
+    SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
+    RTC_DCHECK_GT(streams[i].width, 0);
+    RTC_DCHECK_GT(streams[i].height, 0);
+    RTC_DCHECK_GT(streams[i].max_framerate, 0);
+    RTC_DCHECK_GE(streams[i].min_bitrate_bps, 0);
+    RTC_DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps);
+    RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
+    RTC_DCHECK_GE(streams[i].max_qp, 0);
+
+    sim_stream->width = static_cast<uint16_t>(streams[i].width);
+    sim_stream->height = static_cast<uint16_t>(streams[i].height);
+    sim_stream->maxFramerate = streams[i].max_framerate;
+    sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000;
+    sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
+    sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
+    sim_stream->qpMax = streams[i].max_qp;
+
+    int num_temporal_layers =
+        streams[i].scalability_mode.has_value()
+            ? ScalabilityModeToNumTemporalLayers(*streams[i].scalability_mode)
+            : streams[i].num_temporal_layers.value_or(1);
+
+    sim_stream->numberOfTemporalLayers =
+        static_cast<unsigned char>(num_temporal_layers);
+    sim_stream->active = streams[i].active;
+
+    video_codec.width =
+        std::max(video_codec.width, static_cast<uint16_t>(streams[i].width));
+    video_codec.height =
+        std::max(video_codec.height, static_cast<uint16_t>(streams[i].height));
+    video_codec.minBitrate =
+        std::min(static_cast<uint16_t>(video_codec.minBitrate),
+                 static_cast<uint16_t>(streams[i].min_bitrate_bps / 1000));
+    video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000;
+    video_codec.qpMax = std::max(video_codec.qpMax,
+                                 static_cast<unsigned int>(streams[i].max_qp));
+    max_framerate = std::max(max_framerate, streams[i].max_framerate);
+
+    // TODO(bugs.webrtc.org/11607): Since scalability mode is a top-level
+    // setting on VideoCodec, setting it makes sense only if it is the same for
+    // all simulcast streams.
+    if (streams[0].scalability_mode != streams[i].scalability_mode) {
+      scalability_mode.reset();
+      // For VP8, top-level scalability mode doesn't matter, since configuration
+      // is based on the per-simulcast stream configuration of temporal layers.
+      if (video_codec.codecType != kVideoCodecVP8) {
+        RTC_LOG(LS_WARNING) << "Inconsistent scalability modes configured.";
+      }
+    }
+  }
+
+  if (scalability_mode.has_value()) {
+    video_codec.SetScalabilityMode(*scalability_mode);
+  }
+
+  if (video_codec.maxBitrate == 0) {
+    // Unset max bitrate -> cap to one bit per pixel.
+    video_codec.maxBitrate =
+        (video_codec.width * video_codec.height * video_codec.maxFramerate) /
+        1000;
+  }
+  if (video_codec.maxBitrate < kEncoderMinBitrateKbps)
+    video_codec.maxBitrate = kEncoderMinBitrateKbps;
+
+  video_codec.maxFramerate = max_framerate;
+  video_codec.spatialLayers[0] = {0};
+  video_codec.spatialLayers[0].width = video_codec.width;
+  video_codec.spatialLayers[0].height = video_codec.height;
+  video_codec.spatialLayers[0].maxFramerate = max_framerate;
+  video_codec.spatialLayers[0].numberOfTemporalLayers =
+      streams[0].scalability_mode.has_value()
+          ? ScalabilityModeToNumTemporalLayers(*streams[0].scalability_mode)
+          : streams[0].num_temporal_layers.value_or(1);
+
+  // Set codec specific options
+  if (config.encoder_specific_settings)
+    config.encoder_specific_settings->FillEncoderSpecificSettings(&video_codec);
+
+  switch (video_codec.codecType) {
+    case kVideoCodecVP8: {
+      if (!config.encoder_specific_settings) {
+        *video_codec.VP8() = VideoEncoder::GetDefaultVp8Settings();
+      }
+
+      // Validate specified scalability modes. If some layer has an unsupported
+      // mode, store it as the top-level scalability mode, which will make
+      // InitEncode fail with an appropriate error.
+      for (const auto& stream : streams) {
+        if (stream.scalability_mode.has_value() &&
+            !VP8SupportsScalabilityMode(*stream.scalability_mode)) {
+          RTC_LOG(LS_WARNING)
+              << "Invalid scalability mode for VP8: "
+              << ScalabilityModeToString(*stream.scalability_mode);
+          video_codec.SetScalabilityMode(*stream.scalability_mode);
+          break;
+        }
+      }
+      video_codec.VP8()->numberOfTemporalLayers =
+          streams.back().scalability_mode.has_value()
+              ? ScalabilityModeToNumTemporalLayers(
+                    *streams.back().scalability_mode)
+              : streams.back().num_temporal_layers.value_or(
+                    video_codec.VP8()->numberOfTemporalLayers);
+
+      RTC_DCHECK_GE(video_codec.VP8()->numberOfTemporalLayers, 1);
+      RTC_DCHECK_LE(video_codec.VP8()->numberOfTemporalLayers,
+                    kMaxTemporalStreams);
+
+      break;
+    }
+    case kVideoCodecVP9: {
+      // Force the first stream to always be active.
+      video_codec.simulcastStream[0].active = codec_active;
+
+      if (!config.encoder_specific_settings) {
+        *video_codec.VP9() = VideoEncoder::GetDefaultVp9Settings();
+      }
+
+      video_codec.VP9()->numberOfTemporalLayers = static_cast<unsigned char>(
+          streams.back().num_temporal_layers.value_or(
+              video_codec.VP9()->numberOfTemporalLayers));
+      RTC_DCHECK_GE(video_codec.VP9()->numberOfTemporalLayers, 1);
+      RTC_DCHECK_LE(video_codec.VP9()->numberOfTemporalLayers,
+                    kMaxTemporalStreams);
+
+      RTC_DCHECK(config.spatial_layers.empty() ||
+                 config.spatial_layers.size() ==
+                     video_codec.VP9()->numberOfSpatialLayers);
+
+      std::vector<SpatialLayer> spatial_layers;
+      if (!config.spatial_layers.empty()) {
+        // Layering is set explicitly.
+        spatial_layers = config.spatial_layers;
+      } else if (scalability_mode.has_value()) {
+        // Layering is set via scalability mode.
+        spatial_layers = GetVp9SvcConfig(video_codec);
+        if (spatial_layers.empty())
+          break;
+      } else {
+        size_t first_active_layer = 0;
+        for (size_t spatial_idx = 0;
+             spatial_idx < config.simulcast_layers.size(); ++spatial_idx) {
+          if (config.simulcast_layers[spatial_idx].active) {
+            first_active_layer = spatial_idx;
+            break;
+          }
+        }
+
+        spatial_layers = GetSvcConfig(
+            video_codec.width, video_codec.height, video_codec.maxFramerate,
+            first_active_layer, video_codec.VP9()->numberOfSpatialLayers,
+            video_codec.VP9()->numberOfTemporalLayers,
+            video_codec.mode == VideoCodecMode::kScreensharing);
+
+        // If there was no request for spatial layering, don't limit bitrate
+        // of single spatial layer.
+        const bool no_spatial_layering =
+            video_codec.VP9()->numberOfSpatialLayers <= 1;
+        if (no_spatial_layering) {
+          // Use codec's bitrate limits.
+          spatial_layers.back().minBitrate = video_codec.minBitrate;
+          spatial_layers.back().targetBitrate = video_codec.maxBitrate;
+          spatial_layers.back().maxBitrate = video_codec.maxBitrate;
+        }
+
+        for (size_t spatial_idx = first_active_layer;
+             spatial_idx < config.simulcast_layers.size() &&
+             spatial_idx < spatial_layers.size() + first_active_layer;
+             ++spatial_idx) {
+          spatial_layers[spatial_idx - first_active_layer].active =
+              config.simulcast_layers[spatial_idx].active;
+        }
+      }
+
+      RTC_DCHECK(!spatial_layers.empty());
+      for (size_t i = 0; i < spatial_layers.size(); ++i) {
+        video_codec.spatialLayers[i] = spatial_layers[i];
+      }
+
+      // The top spatial layer dimensions may not be equal to the input
+      // resolution because of the rounding or explicit configuration.
+      // This difference must be propagated to the stream configuration.
+      video_codec.width = spatial_layers.back().width;
+      video_codec.height = spatial_layers.back().height;
+      video_codec.simulcastStream[0].width = spatial_layers.back().width;
+      video_codec.simulcastStream[0].height = spatial_layers.back().height;
+
+      // Update layering settings.
+      video_codec.VP9()->numberOfSpatialLayers =
+          static_cast<unsigned char>(spatial_layers.size());
+      RTC_DCHECK_GE(video_codec.VP9()->numberOfSpatialLayers, 1);
+      RTC_DCHECK_LE(video_codec.VP9()->numberOfSpatialLayers,
+                    kMaxSpatialLayers);
+
+      video_codec.VP9()->numberOfTemporalLayers = static_cast<unsigned char>(
+          spatial_layers.back().numberOfTemporalLayers);
+      RTC_DCHECK_GE(video_codec.VP9()->numberOfTemporalLayers, 1);
+      RTC_DCHECK_LE(video_codec.VP9()->numberOfTemporalLayers,
+                    kMaxTemporalStreams);
+
+      break;
+    }
+    case kVideoCodecAV1:
+      if (SetAv1SvcConfig(video_codec,
+                          /*num_temporal_layers=*/
+                          streams.back().num_temporal_layers.value_or(1),
+                          /*num_spatial_layers=*/
+                          std::max<int>(config.spatial_layers.size(), 1))) {
+        for (size_t i = 0; i < config.spatial_layers.size(); ++i) {
+          video_codec.spatialLayers[i].active = config.spatial_layers[i].active;
+        }
+      } else {
+        RTC_LOG(LS_WARNING) << "Failed to configure svc bitrates for av1.";
+      }
+      break;
+    case kVideoCodecH264: {
+      RTC_CHECK(!config.encoder_specific_settings);
+
+      *video_codec.H264() = VideoEncoder::GetDefaultH264Settings();
+      video_codec.H264()->numberOfTemporalLayers = static_cast<unsigned char>(
+          streams.back().num_temporal_layers.value_or(
+              video_codec.H264()->numberOfTemporalLayers));
+      RTC_DCHECK_GE(video_codec.H264()->numberOfTemporalLayers, 1);
+      RTC_DCHECK_LE(video_codec.H264()->numberOfTemporalLayers,
+                    kMaxTemporalStreams);
+      break;
+    }
+    default:
+      // TODO(pbos): Support encoder_settings codec-agnostically.
+      RTC_DCHECK(!config.encoder_specific_settings)
+          << "Encoder-specific settings for codec type not wired up.";
+      break;
+  }
+
+  const absl::optional<DataRate> experimental_min_bitrate =
+      GetExperimentalMinVideoBitrate(video_codec.codecType);
+  if (experimental_min_bitrate) {
+    const int experimental_min_bitrate_kbps =
+        rtc::saturated_cast<int>(experimental_min_bitrate->kbps());
+    video_codec.minBitrate = experimental_min_bitrate_kbps;
+    video_codec.simulcastStream[0].minBitrate = experimental_min_bitrate_kbps;
+    if (video_codec.codecType == kVideoCodecVP9) {
+      video_codec.spatialLayers[0].minBitrate = experimental_min_bitrate_kbps;
+    }
+  }
+
+  return video_codec;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/video_codec_initializer_unittest.cc b/third_party/libwebrtc/modules/video_coding/video_codec_initializer_unittest.cc
new file mode 100644
index 0000000000..0e6f2dfca2
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_codec_initializer_unittest.cc
@@ -0,0 +1,493 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/include/video_codec_initializer.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/test/mock_fec_controller_override.h"
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_bitrate_allocator.h"
+#include "api/video/video_bitrate_allocator_factory.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/vp8_temporal_layers.h"
+#include "api/video_codecs/vp8_temporal_layers_factory.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "rtc_base/checks.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+static const int kDefaultWidth = 1280;
+static const int kDefaultHeight = 720;
+static const int kDefaultFrameRate = 30;
+static const uint32_t kDefaultMinBitrateBps = 60000;
+static const uint32_t kDefaultTargetBitrateBps = 2000000;
+static const uint32_t kDefaultMaxBitrateBps = 2000000;
+static const uint32_t kDefaultMinTransmitBitrateBps = 400000;
+static const int kDefaultMaxQp = 48;
+static const uint32_t kScreenshareTl0BitrateBps = 120000;
+static const uint32_t kScreenshareConferenceTl0BitrateBps = 200000;
+static const uint32_t kScreenshareCodecTargetBitrateBps = 200000;
+static const uint32_t kScreenshareDefaultFramerate = 5;
+// Bitrates for the temporal layers of the higher screenshare simulcast stream.
+static const uint32_t kHighScreenshareTl0Bps = 800000;
+static const uint32_t kHighScreenshareTl1Bps = 1200000;
+}  // namespace
+
+// TODO(sprang): Extend coverage to handle the rest of the codec initializer.
+class VideoCodecInitializerTest : public ::testing::Test {
+ public:
+  VideoCodecInitializerTest() {}
+  virtual ~VideoCodecInitializerTest() {}
+
+ protected:
+  void SetUpFor(VideoCodecType type,
+                int num_spatial_streams,
+                int num_temporal_streams,
+                bool screenshare) {
+    config_ = VideoEncoderConfig();
+    config_.codec_type = type;
+
+    if (screenshare) {
+      config_.min_transmit_bitrate_bps = kDefaultMinTransmitBitrateBps;
+      config_.content_type = VideoEncoderConfig::ContentType::kScreen;
+    }
+
+    if (type == VideoCodecType::kVideoCodecVP8) {
+      config_.number_of_streams = num_spatial_streams;
+      VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings();
+      vp8_settings.numberOfTemporalLayers = num_temporal_streams;
+      config_.encoder_specific_settings = rtc::make_ref_counted<
+          webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings);
+    } else if (type == VideoCodecType::kVideoCodecVP9) {
+      VideoCodecVP9 vp9_settings = VideoEncoder::GetDefaultVp9Settings();
+      vp9_settings.numberOfSpatialLayers = num_spatial_streams;
+      vp9_settings.numberOfTemporalLayers = num_temporal_streams;
+      config_.encoder_specific_settings = rtc::make_ref_counted<
+          webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
+    } else if (type != VideoCodecType::kVideoCodecMultiplex) {
+      ADD_FAILURE() << "Unexpected codec type: " << type;
+    }
+  }
+
+  bool InitializeCodec() {
+    codec_out_ = VideoCodec();
+    frame_buffer_controller_.reset();
+    if (!VideoCodecInitializer::SetupCodec(config_, streams_, &codec_out_)) {
+      return false;
+    }
+    bitrate_allocator_ = CreateBuiltinVideoBitrateAllocatorFactory()
+                             ->CreateVideoBitrateAllocator(codec_out_);
+    RTC_CHECK(bitrate_allocator_);
+    if (codec_out_.codecType == VideoCodecType::kVideoCodecMultiplex)
+      return true;
+
+    // Make sure temporal layers instances have been created.
+    if (codec_out_.codecType == VideoCodecType::kVideoCodecVP8) {
+      Vp8TemporalLayersFactory factory;
+      const VideoEncoder::Settings settings(VideoEncoder::Capabilities(false),
+                                            1, 1000);
+      frame_buffer_controller_ =
+          factory.Create(codec_out_, settings, &fec_controller_override_);
+    }
+    return true;
+  }
+
+  VideoStream DefaultStream() {
+    VideoStream stream;
+    stream.width = kDefaultWidth;
+    stream.height = kDefaultHeight;
+    stream.max_framerate = kDefaultFrameRate;
+    stream.min_bitrate_bps = kDefaultMinBitrateBps;
+    stream.target_bitrate_bps = kDefaultTargetBitrateBps;
+    stream.max_bitrate_bps = kDefaultMaxBitrateBps;
+    stream.max_qp = kDefaultMaxQp;
+    stream.num_temporal_layers = 1;
+    stream.active = true;
+    return stream;
+  }
+
+  VideoStream DefaultScreenshareStream() {
+    VideoStream stream = DefaultStream();
+    stream.min_bitrate_bps = 30000;
+    stream.target_bitrate_bps = kScreenshareCodecTargetBitrateBps;
+    stream.max_bitrate_bps = 1000000;
+    stream.max_framerate = kScreenshareDefaultFramerate;
+    stream.num_temporal_layers = 2;
+    stream.active = true;
+    return stream;
+  }
+
+  MockFecControllerOverride fec_controller_override_;
+
+  // Input settings.
+  VideoEncoderConfig config_;
+  std::vector<VideoStream> streams_;
+
+  // Output.
+  VideoCodec codec_out_;
+  std::unique_ptr<VideoBitrateAllocator> bitrate_allocator_;
+  std::unique_ptr<Vp8FrameBufferController> frame_buffer_controller_;
+};
+
+TEST_F(VideoCodecInitializerTest, SingleStreamVp8Screenshare) {
+  SetUpFor(VideoCodecType::kVideoCodecVP8, 1, 1, true);
+  streams_.push_back(DefaultStream());
+  EXPECT_TRUE(InitializeCodec());
+
+  VideoBitrateAllocation bitrate_allocation =
+      bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
+          kDefaultTargetBitrateBps, kDefaultFrameRate));
+  EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
+  EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
+  EXPECT_EQ(kDefaultTargetBitrateBps, bitrate_allocation.get_sum_bps());
+}
+
+TEST_F(VideoCodecInitializerTest, SingleStreamVp8ScreenshareInactive) {
+  SetUpFor(VideoCodecType::kVideoCodecVP8, 1, 1, true);
+  VideoStream inactive_stream = DefaultStream();
+  inactive_stream.active = false;
+  streams_.push_back(inactive_stream);
+  EXPECT_TRUE(InitializeCodec());
+
+  VideoBitrateAllocation bitrate_allocation =
+      bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
+          kDefaultTargetBitrateBps, kDefaultFrameRate));
+  EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
+  EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
+  EXPECT_EQ(0U, bitrate_allocation.get_sum_bps());
+}
+
+TEST_F(VideoCodecInitializerTest, TemporalLayeredVp8ScreenshareConference) {
+  SetUpFor(VideoCodecType::kVideoCodecVP8, 1, 2, true);
+  streams_.push_back(DefaultScreenshareStream());
+  EXPECT_TRUE(InitializeCodec());
+  bitrate_allocator_->SetLegacyConferenceMode(true);
+
+  EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
+  EXPECT_EQ(2u, codec_out_.VP8()->numberOfTemporalLayers);
+  VideoBitrateAllocation bitrate_allocation =
+      bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
+          kScreenshareCodecTargetBitrateBps, kScreenshareDefaultFramerate));
+  EXPECT_EQ(kScreenshareCodecTargetBitrateBps,
+            bitrate_allocation.get_sum_bps());
+  EXPECT_EQ(kScreenshareConferenceTl0BitrateBps,
+            bitrate_allocation.GetBitrate(0, 0));
+}
+
+TEST_F(VideoCodecInitializerTest, TemporalLayeredVp8Screenshare) {
+  SetUpFor(VideoCodecType::kVideoCodecVP8, 1, 2, true);
+  streams_.push_back(DefaultScreenshareStream());
+  EXPECT_TRUE(InitializeCodec());
+
+  EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
+  EXPECT_EQ(2u, codec_out_.VP8()->numberOfTemporalLayers);
+  VideoBitrateAllocation bitrate_allocation =
+      bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
+          kScreenshareCodecTargetBitrateBps, kScreenshareDefaultFramerate));
+  EXPECT_EQ(kScreenshareCodecTargetBitrateBps,
+            bitrate_allocation.get_sum_bps());
+  EXPECT_EQ(kScreenshareTl0BitrateBps, bitrate_allocation.GetBitrate(0, 0));
+}
+
+TEST_F(VideoCodecInitializerTest, SimulcastVp8Screenshare) {
+  SetUpFor(VideoCodecType::kVideoCodecVP8, 2, 1, true);
+  streams_.push_back(DefaultScreenshareStream());
+  VideoStream video_stream = DefaultStream();
+  video_stream.max_framerate = kScreenshareDefaultFramerate;
+  streams_.push_back(video_stream);
+  EXPECT_TRUE(InitializeCodec());
+
+  EXPECT_EQ(2u, codec_out_.numberOfSimulcastStreams);
+  EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
+  const uint32_t max_bitrate_bps =
+      streams_[0].target_bitrate_bps + streams_[1].max_bitrate_bps;
+  VideoBitrateAllocation bitrate_allocation =
+      bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
+          max_bitrate_bps, kScreenshareDefaultFramerate));
+  EXPECT_EQ(max_bitrate_bps, bitrate_allocation.get_sum_bps());
+  EXPECT_EQ(static_cast<uint32_t>(streams_[0].target_bitrate_bps),
+            bitrate_allocation.GetSpatialLayerSum(0));
+  EXPECT_EQ(static_cast<uint32_t>(streams_[1].max_bitrate_bps),
+            bitrate_allocation.GetSpatialLayerSum(1));
+}
+
+// Tests that when a video stream is inactive, then the bitrate allocation will
+// be 0 for that stream.
+TEST_F(VideoCodecInitializerTest, SimulcastVp8ScreenshareInactive) {
+  SetUpFor(VideoCodecType::kVideoCodecVP8, 2, 1, true);
+  streams_.push_back(DefaultScreenshareStream());
+  VideoStream inactive_video_stream = DefaultStream();
+  inactive_video_stream.active = false;
+  inactive_video_stream.max_framerate = kScreenshareDefaultFramerate;
+  streams_.push_back(inactive_video_stream);
+  EXPECT_TRUE(InitializeCodec());
+
+  EXPECT_EQ(2u, codec_out_.numberOfSimulcastStreams);
+  EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
+  const uint32_t target_bitrate =
+      streams_[0].target_bitrate_bps + streams_[1].target_bitrate_bps;
+  VideoBitrateAllocation bitrate_allocation =
+      bitrate_allocator_->Allocate(VideoBitrateAllocationParameters(
+          target_bitrate, kScreenshareDefaultFramerate));
+  EXPECT_EQ(static_cast<uint32_t>(streams_[0].max_bitrate_bps),
+            bitrate_allocation.get_sum_bps());
+  EXPECT_EQ(static_cast<uint32_t>(streams_[0].max_bitrate_bps),
+            bitrate_allocation.GetSpatialLayerSum(0));
+  EXPECT_EQ(0U, bitrate_allocation.GetSpatialLayerSum(1));
+}
+
+TEST_F(VideoCodecInitializerTest, HighFpsSimulcastVp8Screenshare) {
+  // Two simulcast streams, the lower one using legacy settings (two temporal
+  // streams, 5fps), the higher one using 3 temporal streams and 30fps.
+  SetUpFor(VideoCodecType::kVideoCodecVP8, 2, 3, true);
+  streams_.push_back(DefaultScreenshareStream());
+  VideoStream video_stream = DefaultStream();
+  video_stream.num_temporal_layers = 3;
+  streams_.push_back(video_stream);
+  EXPECT_TRUE(InitializeCodec());
+
+  EXPECT_EQ(2u, codec_out_.numberOfSimulcastStreams);
+  EXPECT_EQ(3u, codec_out_.VP8()->numberOfTemporalLayers);
+  const uint32_t max_bitrate_bps =
+      streams_[0].target_bitrate_bps + streams_[1].max_bitrate_bps;
+  VideoBitrateAllocation bitrate_allocation = bitrate_allocator_->Allocate(
+      VideoBitrateAllocationParameters(max_bitrate_bps, kDefaultFrameRate));
+  EXPECT_EQ(max_bitrate_bps, bitrate_allocation.get_sum_bps());
+  EXPECT_EQ(static_cast<uint32_t>(streams_[0].target_bitrate_bps),
+            bitrate_allocation.GetSpatialLayerSum(0));
+  EXPECT_EQ(static_cast<uint32_t>(streams_[1].max_bitrate_bps),
+            bitrate_allocation.GetSpatialLayerSum(1));
+  EXPECT_EQ(kHighScreenshareTl0Bps, bitrate_allocation.GetBitrate(1, 0));
+  EXPECT_EQ(kHighScreenshareTl1Bps - kHighScreenshareTl0Bps,
+            bitrate_allocation.GetBitrate(1, 1));
+}
+
+TEST_F(VideoCodecInitializerTest, SingleStreamMultiplexCodec) {
+  SetUpFor(VideoCodecType::kVideoCodecMultiplex, 1, 1, true);
+  streams_.push_back(DefaultStream());
+  EXPECT_TRUE(InitializeCodec());
+}
+
+TEST_F(VideoCodecInitializerTest, Vp9SvcDefaultLayering) {
+  SetUpFor(VideoCodecType::kVideoCodecVP9, 3, 3, false);
+  VideoStream stream = DefaultStream();
+  stream.num_temporal_layers = 3;
+  streams_.push_back(stream);
+
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 3u);
+  EXPECT_EQ(codec_out_.VP9()->numberOfTemporalLayers, 3u);
+}
+
+TEST_F(VideoCodecInitializerTest, Vp9SvcAdjustedLayering) {
+  SetUpFor(VideoCodecType::kVideoCodecVP9, 3, 3, false);
+  VideoStream stream = DefaultStream();
+  stream.num_temporal_layers = 3;
+  // Set resolution which is only enough to produce 2 spatial layers.
+  stream.width = kMinVp9SpatialLayerLongSideLength * 2;
+  stream.height = kMinVp9SpatialLayerShortSideLength * 2;
+
+  streams_.push_back(stream);
+
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 2u);
+}
+
+TEST_F(VideoCodecInitializerTest,
+       Vp9SingleSpatialLayerMaxBitrateIsEqualToCodecMaxBitrate) {
+  SetUpFor(VideoCodecType::kVideoCodecVP9, 1, 3, false);
+  VideoStream stream = DefaultStream();
+  stream.num_temporal_layers = 3;
+  streams_.push_back(stream);
+
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.spatialLayers[0].maxBitrate,
+            kDefaultMaxBitrateBps / 1000);
+}
+
+TEST_F(VideoCodecInitializerTest,
+       Vp9SingleSpatialLayerTargetBitrateIsEqualToCodecMaxBitrate) {
+  SetUpFor(VideoCodecType::kVideoCodecVP9, 1, 1, true);
+  VideoStream stream = DefaultStream();
+  stream.num_temporal_layers = 1;
+  streams_.push_back(stream);
+
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.spatialLayers[0].targetBitrate,
+            kDefaultMaxBitrateBps / 1000);
+}
+
+TEST_F(VideoCodecInitializerTest,
+       Vp9KeepBitrateLimitsIfNumberOfSpatialLayersIsReducedToOne) {
+  // Request 3 spatial layers for 320x180 input. Actual number of layers will be
+  // reduced to 1 due to low input resolution but SVC bitrate limits should be
+  // applied.
+  SetUpFor(VideoCodecType::kVideoCodecVP9, 3, 3, false);
+  VideoStream stream = DefaultStream();
+  stream.width = 320;
+  stream.height = 180;
+  stream.num_temporal_layers = 3;
+  streams_.push_back(stream);
+
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_LT(codec_out_.spatialLayers[0].maxBitrate,
+            kDefaultMaxBitrateBps / 1000);
+}
+
+TEST_F(VideoCodecInitializerTest, Vp9DeactivateLayers) {
+  SetUpFor(VideoCodecType::kVideoCodecVP9, 3, 1, false);
+  VideoStream stream = DefaultStream();
+  streams_.push_back(stream);
+
+  config_.simulcast_layers.resize(3);
+
+  // Activate all layers.
+  config_.simulcast_layers[0].active = true;
+  config_.simulcast_layers[1].active = true;
+  config_.simulcast_layers[2].active = true;
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 3);
+  EXPECT_TRUE(codec_out_.spatialLayers[0].active);
+  EXPECT_TRUE(codec_out_.spatialLayers[1].active);
+  EXPECT_TRUE(codec_out_.spatialLayers[2].active);
+
+  // Deactivate top layer.
+  config_.simulcast_layers[0].active = true;
+  config_.simulcast_layers[1].active = true;
+  config_.simulcast_layers[2].active = false;
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 3);
+  EXPECT_TRUE(codec_out_.spatialLayers[0].active);
+  EXPECT_TRUE(codec_out_.spatialLayers[1].active);
+  EXPECT_FALSE(codec_out_.spatialLayers[2].active);
+
+  // Deactivate middle layer.
+  config_.simulcast_layers[0].active = true;
+  config_.simulcast_layers[1].active = false;
+  config_.simulcast_layers[2].active = true;
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 3);
+  EXPECT_TRUE(codec_out_.spatialLayers[0].active);
+  EXPECT_FALSE(codec_out_.spatialLayers[1].active);
+  EXPECT_TRUE(codec_out_.spatialLayers[2].active);
+
+  // Deactivate first layer.
+  config_.simulcast_layers[0].active = false;
+  config_.simulcast_layers[1].active = true;
+  config_.simulcast_layers[2].active = true;
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 2);
+  EXPECT_TRUE(codec_out_.spatialLayers[0].active);
+  EXPECT_TRUE(codec_out_.spatialLayers[1].active);
+
+  // HD singlecast.
+  config_.simulcast_layers[0].active = false;
+  config_.simulcast_layers[1].active = false;
+  config_.simulcast_layers[2].active = true;
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 1);
+  EXPECT_TRUE(codec_out_.spatialLayers[0].active);
+
+  // VGA singlecast.
+  config_.simulcast_layers[0].active = false;
+  config_.simulcast_layers[1].active = true;
+  config_.simulcast_layers[2].active = false;
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 2);
+  EXPECT_TRUE(codec_out_.spatialLayers[0].active);
+  EXPECT_FALSE(codec_out_.spatialLayers[1].active);
+
+  // QVGA singlecast.
+  config_.simulcast_layers[0].active = true;
+  config_.simulcast_layers[1].active = false;
+  config_.simulcast_layers[2].active = false;
+  EXPECT_TRUE(InitializeCodec());
+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 3);
+  EXPECT_TRUE(codec_out_.spatialLayers[0].active);
+  EXPECT_FALSE(codec_out_.spatialLayers[1].active);
+  EXPECT_FALSE(codec_out_.spatialLayers[2].active);
+}
+
+TEST_F(VideoCodecInitializerTest, Av1SingleSpatialLayerBitratesAreConsistent) {
+  VideoEncoderConfig config;
+  config.codec_type = VideoCodecType::kVideoCodecAV1;
+  std::vector<VideoStream> streams = {DefaultStream()};
+  streams[0].scalability_mode = ScalabilityMode::kL1T2;
+
+  VideoCodec codec;
+  EXPECT_TRUE(VideoCodecInitializer::SetupCodec(config, streams, &codec));
+
+  EXPECT_GE(codec.spatialLayers[0].targetBitrate,
+            codec.spatialLayers[0].minBitrate);
+  EXPECT_LE(codec.spatialLayers[0].targetBitrate,
+            codec.spatialLayers[0].maxBitrate);
+}
+
+TEST_F(VideoCodecInitializerTest, Av1TwoSpatialLayersBitratesAreConsistent) {
+  VideoEncoderConfig config;
+  config.codec_type = VideoCodecType::kVideoCodecAV1;
+  std::vector<VideoStream> streams = {DefaultStream()};
+  streams[0].scalability_mode = ScalabilityMode::kL2T2;
+
+  VideoCodec codec;
+  EXPECT_TRUE(VideoCodecInitializer::SetupCodec(config, streams, &codec));
+
+  EXPECT_GE(codec.spatialLayers[0].targetBitrate,
+            codec.spatialLayers[0].minBitrate);
+  EXPECT_LE(codec.spatialLayers[0].targetBitrate,
+            codec.spatialLayers[0].maxBitrate);
+
+  EXPECT_GE(codec.spatialLayers[1].targetBitrate,
+            codec.spatialLayers[1].minBitrate);
+  EXPECT_LE(codec.spatialLayers[1].targetBitrate,
+            codec.spatialLayers[1].maxBitrate);
+}
+
+TEST_F(VideoCodecInitializerTest, Av1TwoSpatialLayersActiveByDefault) {
+  VideoEncoderConfig config;
+  config.codec_type = VideoCodecType::kVideoCodecAV1;
+  std::vector<VideoStream> streams = {DefaultStream()};
+  streams[0].scalability_mode = ScalabilityMode::kL2T2;
+  config.spatial_layers = {};
+
+  VideoCodec codec;
+  EXPECT_TRUE(VideoCodecInitializer::SetupCodec(config, streams, &codec));
+
+  EXPECT_TRUE(codec.spatialLayers[0].active);
+  EXPECT_TRUE(codec.spatialLayers[1].active);
+}
+
+TEST_F(VideoCodecInitializerTest, Av1TwoSpatialLayersOneDeactivated) {
+  VideoEncoderConfig config;
+  config.codec_type = VideoCodecType::kVideoCodecAV1;
+  std::vector<VideoStream> streams = {DefaultStream()};
+  streams[0].scalability_mode = ScalabilityMode::kL2T2;
+  config.spatial_layers.resize(2);
+  config.spatial_layers[0].active = true;
+  config.spatial_layers[1].active = false;
+
+  VideoCodec codec;
+  EXPECT_TRUE(VideoCodecInitializer::SetupCodec(config, streams, &codec));
+
+  EXPECT_TRUE(codec.spatialLayers[0].active);
+  EXPECT_FALSE(codec.spatialLayers[1].active);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/video_codec_interface_gn/moz.build b/third_party/libwebrtc/modules/video_coding/video_codec_interface_gn/moz.build
new file mode 100644
index 0000000000..f01783ebd1
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_codec_interface_gn/moz.build
@@ -0,0 +1,226 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/include/video_codec_interface.cc",
+    "/third_party/libwebrtc/modules/video_coding/video_coding_defines.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("video_codec_interface_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/video_coding_defines.cc b/third_party/libwebrtc/modules/video_coding/video_coding_defines.cc
new file mode 100644
index 0000000000..436b1a6490
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_coding_defines.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/include/video_coding_defines.h"
+
+namespace webrtc {
+
+void VCMReceiveCallback::OnDroppedFrames(uint32_t frames_dropped) {}
+void VCMReceiveCallback::OnIncomingPayloadType(int payload_type) {}
+void VCMReceiveCallback::OnDecoderInfoChanged(
+    const VideoDecoder::DecoderInfo&) {}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/video_coding_gn/moz.build b/third_party/libwebrtc/modules/video_coding/video_coding_gn/moz.build
new file mode 100644
index 0000000000..a6034ad476
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_coding_gn/moz.build
@@ -0,0 +1,249 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/decoder_database.cc",
+    "/third_party/libwebrtc/modules/video_coding/event_wrapper.cc",
+    "/third_party/libwebrtc/modules/video_coding/fec_controller_default.cc",
+    "/third_party/libwebrtc/modules/video_coding/frame_object.cc",
+    "/third_party/libwebrtc/modules/video_coding/generic_decoder.cc",
+    "/third_party/libwebrtc/modules/video_coding/h264_sprop_parameter_sets.cc",
+    "/third_party/libwebrtc/modules/video_coding/h264_sps_pps_tracker.cc",
+    "/third_party/libwebrtc/modules/video_coding/loss_notification_controller.cc",
+    "/third_party/libwebrtc/modules/video_coding/media_opt_util.cc",
+    "/third_party/libwebrtc/modules/video_coding/rtp_frame_id_only_ref_finder.cc",
+    "/third_party/libwebrtc/modules/video_coding/rtp_frame_reference_finder.cc",
+    "/third_party/libwebrtc/modules/video_coding/rtp_generic_ref_finder.cc",
+    "/third_party/libwebrtc/modules/video_coding/rtp_seq_num_only_ref_finder.cc",
+    "/third_party/libwebrtc/modules/video_coding/rtp_vp8_ref_finder.cc",
+    "/third_party/libwebrtc/modules/video_coding/rtp_vp9_ref_finder.cc",
+    "/third_party/libwebrtc/modules/video_coding/video_codec_initializer.cc",
+    "/third_party/libwebrtc/modules/video_coding/video_receiver2.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "GLESv2",
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("video_coding_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/video_coding_impl.cc b/third_party/libwebrtc/modules/video_coding/video_coding_impl.cc
new file mode 100644
index 0000000000..2eaecd5011
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_coding_impl.cc
@@ -0,0 +1,254 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/video_coding_impl.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/video/encoded_image.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/timing/timing.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/memory/always_valid_pointer.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace vcm {
+
+int64_t VCMProcessTimer::Period() const {
+  return _periodMs;
+}
+
+int64_t VCMProcessTimer::TimeUntilProcess() const {
+  const int64_t time_since_process = _clock->TimeInMilliseconds() - _latestMs;
+  const int64_t time_until_process = _periodMs - time_since_process;
+  return std::max<int64_t>(time_until_process, 0);
+}
+
+void VCMProcessTimer::Processed() {
+  _latestMs = _clock->TimeInMilliseconds();
+}
+
+DEPRECATED_VCMDecoderDataBase::DEPRECATED_VCMDecoderDataBase() {
+  decoder_sequence_checker_.Detach();
+}
+
+VideoDecoder* DEPRECATED_VCMDecoderDataBase::DeregisterExternalDecoder(
+    uint8_t payload_type) {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  auto it = decoders_.find(payload_type);
+  if (it == decoders_.end()) {
+    return nullptr;
+  }
+
+  // We can't use payload_type to check if the decoder is currently in use,
+  // because payload type may be out of date (e.g. before we decode the first
+  // frame after RegisterReceiveCodec).
+  if (current_decoder_ && current_decoder_->IsSameDecoder(it->second)) {
+    // Release it if it was registered and in use.
+    current_decoder_ = absl::nullopt;
+  }
+  VideoDecoder* ret = it->second;
+  decoders_.erase(it);
+  return ret;
+}
+
+// Add the external decoder object to the list of external decoders.
+// Won't be registered as a receive codec until RegisterReceiveCodec is called.
+void DEPRECATED_VCMDecoderDataBase::RegisterExternalDecoder(
+    uint8_t payload_type,
+    VideoDecoder* external_decoder) {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  // If payload value already exists, erase old and insert new.
+  DeregisterExternalDecoder(payload_type);
+  decoders_[payload_type] = external_decoder;
+}
+
+bool DEPRECATED_VCMDecoderDataBase::IsExternalDecoderRegistered(
+    uint8_t payload_type) const {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  return payload_type == current_payload_type_ ||
+         decoders_.find(payload_type) != decoders_.end();
+}
+
+void DEPRECATED_VCMDecoderDataBase::RegisterReceiveCodec(
+    uint8_t payload_type,
+    const VideoDecoder::Settings& settings) {
+  // If payload value already exists, erase old and insert new.
+  if (payload_type == current_payload_type_) {
+    current_payload_type_ = absl::nullopt;
+  }
+  decoder_settings_[payload_type] = settings;
+}
+
+bool DEPRECATED_VCMDecoderDataBase::DeregisterReceiveCodec(
+    uint8_t payload_type) {
+  if (decoder_settings_.erase(payload_type) == 0) {
+    return false;
+  }
+  if (payload_type == current_payload_type_) {
+    // This codec is currently in use.
+    current_payload_type_ = absl::nullopt;
+  }
+  return true;
+}
+
+VCMGenericDecoder* DEPRECATED_VCMDecoderDataBase::GetDecoder(
+    const VCMEncodedFrame& frame,
+    VCMDecodedFrameCallback* decoded_frame_callback) {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  RTC_DCHECK(decoded_frame_callback->UserReceiveCallback());
+  uint8_t payload_type = frame.PayloadType();
+  if (payload_type == current_payload_type_ || payload_type == 0) {
+    return current_decoder_.has_value() ? &*current_decoder_ : nullptr;
+  }
+  // If decoder exists - delete.
+  if (current_decoder_.has_value()) {
+    current_decoder_ = absl::nullopt;
+    current_payload_type_ = absl::nullopt;
+  }
+
+  CreateAndInitDecoder(frame);
+  if (current_decoder_ == absl::nullopt) {
+    return nullptr;
+  }
+
+  VCMReceiveCallback* callback = decoded_frame_callback->UserReceiveCallback();
+  callback->OnIncomingPayloadType(payload_type);
+  if (current_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback) <
+      0) {
+    current_decoder_ = absl::nullopt;
+    return nullptr;
+  }
+
+  current_payload_type_ = payload_type;
+  return &*current_decoder_;
+}
+
+void DEPRECATED_VCMDecoderDataBase::CreateAndInitDecoder(
+    const VCMEncodedFrame& frame) {
+  uint8_t payload_type = frame.PayloadType();
+  RTC_LOG(LS_INFO) << "Initializing decoder with payload type '"
+                   << int{payload_type} << "'.";
+  auto decoder_item = decoder_settings_.find(payload_type);
+  if (decoder_item == decoder_settings_.end()) {
+    RTC_LOG(LS_ERROR) << "Can't find a decoder associated with payload type: "
+                      << int{payload_type};
+    return;
+  }
+  auto external_dec_item = decoders_.find(payload_type);
+  if (external_dec_item == decoders_.end()) {
+    RTC_LOG(LS_ERROR) << "No decoder of this type exists.";
+    return;
+  }
+  current_decoder_.emplace(external_dec_item->second);
+
+  // Copy over input resolutions to prevent codec reinitialization due to
+  // the first frame being of a different resolution than the database values.
+  // This is best effort, since there's no guarantee that width/height have been
+  // parsed yet (and may be zero).
+  RenderResolution frame_resolution(frame.EncodedImage()._encodedWidth,
+                                    frame.EncodedImage()._encodedHeight);
+  if (frame_resolution.Valid()) {
+    decoder_item->second.set_max_render_resolution(frame_resolution);
+  }
+  if (!current_decoder_->Configure(decoder_item->second)) {
+    current_decoder_ = absl::nullopt;
+    RTC_LOG(LS_ERROR) << "Failed to initialize decoder.";
+  }
+}
+
+}  // namespace vcm
+
+namespace {
+
+class VideoCodingModuleImpl : public VideoCodingModule {
+ public:
+  explicit VideoCodingModuleImpl(Clock* clock,
+                                 const FieldTrialsView* field_trials)
+      : VideoCodingModule(),
+        field_trials_(field_trials),
+        timing_(new VCMTiming(clock, *field_trials_)),
+        receiver_(clock, timing_.get(), *field_trials_) {}
+
+  ~VideoCodingModuleImpl() override = default;
+
+  void Process() override { receiver_.Process(); }
+
+  void RegisterReceiveCodec(
+      uint8_t payload_type,
+      const VideoDecoder::Settings& decoder_settings) override {
+    receiver_.RegisterReceiveCodec(payload_type, decoder_settings);
+  }
+
+  void RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                               uint8_t payloadType) override {
+    receiver_.RegisterExternalDecoder(externalDecoder, payloadType);
+  }
+
+  int32_t RegisterReceiveCallback(
+      VCMReceiveCallback* receiveCallback) override {
+    RTC_DCHECK(construction_thread_.IsCurrent());
+    return receiver_.RegisterReceiveCallback(receiveCallback);
+  }
+
+  int32_t RegisterFrameTypeCallback(
+      VCMFrameTypeCallback* frameTypeCallback) override {
+    return receiver_.RegisterFrameTypeCallback(frameTypeCallback);
+  }
+
+  int32_t RegisterPacketRequestCallback(
+      VCMPacketRequestCallback* callback) override {
+    RTC_DCHECK(construction_thread_.IsCurrent());
+    return receiver_.RegisterPacketRequestCallback(callback);
+  }
+
+  int32_t Decode(uint16_t maxWaitTimeMs) override {
+    return receiver_.Decode(maxWaitTimeMs);
+  }
+
+  int32_t IncomingPacket(const uint8_t* incomingPayload,
+                         size_t payloadLength,
+                         const RTPHeader& rtp_header,
+                         const RTPVideoHeader& video_header) override {
+    return receiver_.IncomingPacket(incomingPayload, payloadLength, rtp_header,
+                                    video_header);
+  }
+
+  void SetNackSettings(size_t max_nack_list_size,
+                       int max_packet_age_to_nack,
+                       int max_incomplete_time_ms) override {
+    return receiver_.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
+                                     max_incomplete_time_ms);
+  }
+
+ private:
+  AlwaysValidPointer<const FieldTrialsView, FieldTrialBasedConfig>
+      field_trials_;
+  SequenceChecker construction_thread_;
+  const std::unique_ptr<VCMTiming> timing_;
+  vcm::VideoReceiver receiver_;
+};
+}  // namespace
+
+// DEPRECATED.  Create method for current interface, will be removed when the
+// new jitter buffer is in place.
+VideoCodingModule* VideoCodingModule::Create(
+    Clock* clock,
+    const FieldTrialsView* field_trials) {
+  RTC_DCHECK(clock);
+  return new VideoCodingModuleImpl(clock, field_trials);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/video_coding_impl.h b/third_party/libwebrtc/modules/video_coding/video_coding_impl.h
new file mode 100644
index 0000000000..927b2da4b8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_coding_impl.h
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
+#define MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "modules/video_coding/frame_buffer.h"
+#include "modules/video_coding/generic_decoder.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/jitter_buffer.h"
+#include "modules/video_coding/receiver.h"
+#include "modules/video_coding/timing/timing.h"
+#include "rtc_base/one_time_event.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class VideoBitrateAllocator;
+class VideoBitrateAllocationObserver;
+
+namespace vcm {
+
+class VCMProcessTimer {
+ public:
+  static const int64_t kDefaultProcessIntervalMs = 1000;
+
+  VCMProcessTimer(int64_t periodMs, Clock* clock)
+      : _clock(clock),
+        _periodMs(periodMs),
+        _latestMs(_clock->TimeInMilliseconds()) {}
+  int64_t Period() const;
+  int64_t TimeUntilProcess() const;
+  void Processed();
+
+ private:
+  Clock* _clock;
+  int64_t _periodMs;
+  int64_t _latestMs;
+};
+
+class DEPRECATED_VCMDecoderDataBase {
+ public:
+  DEPRECATED_VCMDecoderDataBase();
+  DEPRECATED_VCMDecoderDataBase(const DEPRECATED_VCMDecoderDataBase&) = delete;
+  DEPRECATED_VCMDecoderDataBase& operator=(
+      const DEPRECATED_VCMDecoderDataBase&) = delete;
+  ~DEPRECATED_VCMDecoderDataBase() = default;
+
+  // Returns a pointer to the previously registered decoder or nullptr if none
+  // was registered for the `payload_type`.
+  VideoDecoder* DeregisterExternalDecoder(uint8_t payload_type);
+  void RegisterExternalDecoder(uint8_t payload_type,
+                               VideoDecoder* external_decoder);
+  bool IsExternalDecoderRegistered(uint8_t payload_type) const;
+
+  void RegisterReceiveCodec(uint8_t payload_type,
+                            const VideoDecoder::Settings& settings);
+  bool DeregisterReceiveCodec(uint8_t payload_type);
+
+  // Returns a decoder specified by frame.PayloadType. The decoded frame
+  // callback of the decoder is set to `decoded_frame_callback`. If no such
+  // decoder already exists an instance will be created and initialized.
+  // nullptr is returned if no decoder with the specified payload type was found
+  // and the function failed to create one.
+  VCMGenericDecoder* GetDecoder(
+      const VCMEncodedFrame& frame,
+      VCMDecodedFrameCallback* decoded_frame_callback);
+
+ private:
+  void CreateAndInitDecoder(const VCMEncodedFrame& frame)
+      RTC_RUN_ON(decoder_sequence_checker_);
+
+  SequenceChecker decoder_sequence_checker_;
+
+  absl::optional<uint8_t> current_payload_type_;
+  absl::optional<VCMGenericDecoder> current_decoder_
+      RTC_GUARDED_BY(decoder_sequence_checker_);
+  // Initialization paramaters for decoders keyed by payload type.
+  std::map<uint8_t, VideoDecoder::Settings> decoder_settings_;
+  // Decoders keyed by payload type.
+  std::map<uint8_t, VideoDecoder*> decoders_
+      RTC_GUARDED_BY(decoder_sequence_checker_);
+};
+
+class VideoReceiver {
+ public:
+  VideoReceiver(Clock* clock,
+                VCMTiming* timing,
+                const FieldTrialsView& field_trials);
+  ~VideoReceiver();
+
+  void RegisterReceiveCodec(uint8_t payload_type,
+                            const VideoDecoder::Settings& settings);
+
+  void RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                               uint8_t payloadType);
+  int32_t RegisterReceiveCallback(VCMReceiveCallback* receiveCallback);
+  int32_t RegisterFrameTypeCallback(VCMFrameTypeCallback* frameTypeCallback);
+  int32_t RegisterPacketRequestCallback(VCMPacketRequestCallback* callback);
+
+  int32_t Decode(uint16_t maxWaitTimeMs);
+
+  int32_t IncomingPacket(const uint8_t* incomingPayload,
+                         size_t payloadLength,
+                         const RTPHeader& rtp_header,
+                         const RTPVideoHeader& video_header);
+
+  void SetNackSettings(size_t max_nack_list_size,
+                       int max_packet_age_to_nack,
+                       int max_incomplete_time_ms);
+
+  void Process();
+
+ protected:
+  int32_t Decode(const webrtc::VCMEncodedFrame& frame);
+  int32_t RequestKeyFrame();
+
+ private:
+  // Used for DCHECKing thread correctness.
+  // In build where DCHECKs are enabled, will return false before
+  // DecoderThreadStarting is called, then true until DecoderThreadStopped
+  // is called.
+  // In builds where DCHECKs aren't enabled, it will return true.
+  bool IsDecoderThreadRunning();
+
+  SequenceChecker construction_thread_checker_;
+  SequenceChecker decoder_thread_checker_;
+  SequenceChecker module_thread_checker_;
+  Clock* const clock_;
+  Mutex process_mutex_;
+  VCMTiming* _timing;
+  VCMReceiver _receiver;
+  VCMDecodedFrameCallback _decodedFrameCallback;
+
+  // These callbacks are set on the construction thread before being attached
+  // to the module thread or decoding started, so a lock is not required.
+  VCMFrameTypeCallback* _frameTypeCallback;
+  VCMPacketRequestCallback* _packetRequestCallback;
+
+  // Used on both the module and decoder thread.
+  bool _scheduleKeyRequest RTC_GUARDED_BY(process_mutex_);
+  bool drop_frames_until_keyframe_ RTC_GUARDED_BY(process_mutex_);
+
+  // Modified on the construction thread while not attached to the process
+  // thread.  Once attached to the process thread, its value is only read
+  // so a lock is not required.
+  size_t max_nack_list_size_;
+
+  // Callbacks are set before the decoder thread starts.
+  // Once the decoder thread has been started, usage of `_codecDataBase` moves
+  // over to the decoder thread.
+  DEPRECATED_VCMDecoderDataBase _codecDataBase;
+
+  VCMProcessTimer _retransmissionTimer RTC_GUARDED_BY(module_thread_checker_);
+  VCMProcessTimer _keyRequestTimer RTC_GUARDED_BY(module_thread_checker_);
+  ThreadUnsafeOneTimeEvent first_frame_received_
+      RTC_GUARDED_BY(decoder_thread_checker_);
+};
+
+}  // namespace vcm
+}  // namespace webrtc
+#endif  // MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
diff --git a/third_party/libwebrtc/modules/video_coding/video_coding_utility_gn/moz.build b/third_party/libwebrtc/modules/video_coding/video_coding_utility_gn/moz.build
new file mode 100644
index 0000000000..1de983a786
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_coding_utility_gn/moz.build
@@ -0,0 +1,243 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/utility/bandwidth_quality_scaler.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/decoded_frames_history.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/frame_dropper.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/framerate_controller_deprecated.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/ivf_file_reader.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/ivf_file_writer.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/qp_parser.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/quality_scaler.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/simulcast_rate_allocator.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/simulcast_utility.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/vp8_header_parser.cc",
+    "/third_party/libwebrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("video_coding_utility_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/video_receiver.cc b/third_party/libwebrtc/modules/video_coding/video_receiver.cc
new file mode 100644
index 0000000000..38b70f87cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_receiver.cc
@@ -0,0 +1,278 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <vector>
+
+#include "api/rtp_headers.h"
+#include "api/sequence_checker.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/decoder_database.h"
+#include "modules/video_coding/encoded_frame.h"
+#include "modules/video_coding/generic_decoder.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "modules/video_coding/internal_defines.h"
+#include "modules/video_coding/jitter_buffer.h"
+#include "modules/video_coding/media_opt_util.h"
+#include "modules/video_coding/packet.h"
+#include "modules/video_coding/receiver.h"
+#include "modules/video_coding/timing/timing.h"
+#include "modules/video_coding/video_coding_impl.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/one_time_event.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace vcm {
+
+VideoReceiver::VideoReceiver(Clock* clock,
+                             VCMTiming* timing,
+                             const FieldTrialsView& field_trials)
+    : clock_(clock),
+      _timing(timing),
+      _receiver(_timing, clock_, field_trials),
+      _decodedFrameCallback(_timing, clock_, field_trials),
+      _frameTypeCallback(nullptr),
+      _packetRequestCallback(nullptr),
+      _scheduleKeyRequest(false),
+      drop_frames_until_keyframe_(false),
+      max_nack_list_size_(0),
+      _codecDataBase(),
+      _retransmissionTimer(10, clock_),
+      _keyRequestTimer(500, clock_) {
+  decoder_thread_checker_.Detach();
+  module_thread_checker_.Detach();
+}
+
+VideoReceiver::~VideoReceiver() {
+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
+}
+
+void VideoReceiver::Process() {
+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
+
+  // Key frame requests
+  if (_keyRequestTimer.TimeUntilProcess() == 0) {
+    _keyRequestTimer.Processed();
+    bool request_key_frame = _frameTypeCallback != nullptr;
+    if (request_key_frame) {
+      MutexLock lock(&process_mutex_);
+      request_key_frame = _scheduleKeyRequest;
+    }
+    if (request_key_frame)
+      RequestKeyFrame();
+  }
+
+  // Packet retransmission requests
+  // TODO(holmer): Add API for changing Process interval and make sure it's
+  // disabled when NACK is off.
+  if (_retransmissionTimer.TimeUntilProcess() == 0) {
+    _retransmissionTimer.Processed();
+    bool callback_registered = _packetRequestCallback != nullptr;
+    uint16_t length = max_nack_list_size_;
+    if (callback_registered && length > 0) {
+      // Collect sequence numbers from the default receiver.
+      bool request_key_frame = false;
+      std::vector<uint16_t> nackList = _receiver.NackList(&request_key_frame);
+      int32_t ret = VCM_OK;
+      if (request_key_frame) {
+        ret = RequestKeyFrame();
+      }
+      if (ret == VCM_OK && !nackList.empty()) {
+        MutexLock lock(&process_mutex_);
+        if (_packetRequestCallback != nullptr) {
+          _packetRequestCallback->ResendPackets(&nackList[0], nackList.size());
+        }
+      }
+    }
+  }
+}
+
+// Register a receive callback. Will be called whenever there is a new frame
+// ready for rendering.
+int32_t VideoReceiver::RegisterReceiveCallback(
+    VCMReceiveCallback* receiveCallback) {
+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
+  // This value is set before the decoder thread starts and unset after
+  // the decoder thread has been stopped.
+  _decodedFrameCallback.SetUserReceiveCallback(receiveCallback);
+  return VCM_OK;
+}
+
+// Register an externally defined decoder object.
+void VideoReceiver::RegisterExternalDecoder(VideoDecoder* externalDecoder,
+                                            uint8_t payloadType) {
+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
+  if (externalDecoder == nullptr) {
+    RTC_CHECK(_codecDataBase.DeregisterExternalDecoder(payloadType));
+    return;
+  }
+  _codecDataBase.RegisterExternalDecoder(payloadType, externalDecoder);
+}
+
+// Register a frame type request callback.
+int32_t VideoReceiver::RegisterFrameTypeCallback(
+    VCMFrameTypeCallback* frameTypeCallback) {
+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
+  // This callback is used on the module thread, but since we don't get
+  // callbacks on the module thread while the decoder thread isn't running
+  // (and this function must not be called when the decoder is running),
+  // we don't need a lock here.
+  _frameTypeCallback = frameTypeCallback;
+  return VCM_OK;
+}
+
+int32_t VideoReceiver::RegisterPacketRequestCallback(
+    VCMPacketRequestCallback* callback) {
+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
+  // This callback is used on the module thread, but since we don't get
+  // callbacks on the module thread while the decoder thread isn't running
+  // (and this function must not be called when the decoder is running),
+  // we don't need a lock here.
+  _packetRequestCallback = callback;
+  return VCM_OK;
+}
+
+// Decode next frame, blocking.
+// Should be called as often as possible to get the most out of the decoder.
+int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
+  RTC_DCHECK_RUN_ON(&decoder_thread_checker_);
+  VCMEncodedFrame* frame = _receiver.FrameForDecoding(maxWaitTimeMs, true);
+
+  if (!frame)
+    return VCM_FRAME_NOT_READY;
+
+  bool drop_frame = false;
+  {
+    MutexLock lock(&process_mutex_);
+    if (drop_frames_until_keyframe_) {
+      // Still getting delta frames, schedule another keyframe request as if
+      // decode failed.
+      if (frame->FrameType() != VideoFrameType::kVideoFrameKey) {
+        drop_frame = true;
+        _scheduleKeyRequest = true;
+      } else {
+        drop_frames_until_keyframe_ = false;
+      }
+    }
+  }
+
+  if (drop_frame) {
+    _receiver.ReleaseFrame(frame);
+    return VCM_FRAME_NOT_READY;
+  }
+
+  // If this frame was too late, we should adjust the delay accordingly
+  if (frame->RenderTimeMs() > 0)
+    _timing->UpdateCurrentDelay(Timestamp::Millis(frame->RenderTimeMs()),
+                                clock_->CurrentTime());
+
+  if (first_frame_received_()) {
+    RTC_LOG(LS_INFO) << "Received first complete decodable video frame";
+  }
+
+  const int32_t ret = Decode(*frame);
+  _receiver.ReleaseFrame(frame);
+  return ret;
+}
+
+int32_t VideoReceiver::RequestKeyFrame() {
+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
+
+  TRACE_EVENT0("webrtc", "RequestKeyFrame");
+  if (_frameTypeCallback != nullptr) {
+    const int32_t ret = _frameTypeCallback->RequestKeyFrame();
+    if (ret < 0) {
+      return ret;
+    }
+    MutexLock lock(&process_mutex_);
+    _scheduleKeyRequest = false;
+  } else {
+    return VCM_MISSING_CALLBACK;
+  }
+  return VCM_OK;
+}
+
+// Must be called from inside the receive side critical section.
+int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) {
+  RTC_DCHECK_RUN_ON(&decoder_thread_checker_);
+  TRACE_EVENT0("webrtc", "VideoReceiver::Decode");
+  // Change decoder if payload type has changed
+  VCMGenericDecoder* decoder =
+      _codecDataBase.GetDecoder(frame, &_decodedFrameCallback);
+  if (decoder == nullptr) {
+    return VCM_NO_CODEC_REGISTERED;
+  }
+  return decoder->Decode(frame, clock_->CurrentTime());
+}
+
+// Register possible receive codecs, can be called multiple times
+void VideoReceiver::RegisterReceiveCodec(
+    uint8_t payload_type,
+    const VideoDecoder::Settings& settings) {
+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
+  _codecDataBase.RegisterReceiveCodec(payload_type, settings);
+}
+
+// Incoming packet from network parsed and ready for decode, non blocking.
+int32_t VideoReceiver::IncomingPacket(const uint8_t* incomingPayload,
+                                      size_t payloadLength,
+                                      const RTPHeader& rtp_header,
+                                      const RTPVideoHeader& video_header) {
+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
+  if (video_header.frame_type == VideoFrameType::kVideoFrameKey) {
+    TRACE_EVENT1("webrtc", "VCM::PacketKeyFrame", "seqnum",
+                 rtp_header.sequenceNumber);
+  }
+  if (incomingPayload == nullptr) {
+    // The jitter buffer doesn't handle non-zero payload lengths for packets
+    // without payload.
+    // TODO(holmer): We should fix this in the jitter buffer.
+    payloadLength = 0;
+  }
+  // Callers don't provide any ntp time.
+  const VCMPacket packet(incomingPayload, payloadLength, rtp_header,
+                         video_header, /*ntp_time_ms=*/0,
+                         clock_->CurrentTime());
+  int32_t ret = _receiver.InsertPacket(packet);
+
+  // TODO(holmer): Investigate if this somehow should use the key frame
+  // request scheduling to throttle the requests.
+  if (ret == VCM_FLUSH_INDICATOR) {
+    {
+      MutexLock lock(&process_mutex_);
+      drop_frames_until_keyframe_ = true;
+    }
+    RequestKeyFrame();
+  } else if (ret < 0) {
+    return ret;
+  }
+  return VCM_OK;
+}
+
+void VideoReceiver::SetNackSettings(size_t max_nack_list_size,
+                                    int max_packet_age_to_nack,
+                                    int max_incomplete_time_ms) {
+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
+  if (max_nack_list_size != 0) {
+    max_nack_list_size_ = max_nack_list_size;
+  }
+  _receiver.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
+                            max_incomplete_time_ms);
+}
+
+}  // namespace vcm
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/video_receiver2.cc b/third_party/libwebrtc/modules/video_coding/video_receiver2.cc
new file mode 100644
index 0000000000..0751869a98
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_receiver2.cc
@@ -0,0 +1,108 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/video_receiver2.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/decoder_database.h"
+#include "modules/video_coding/encoded_frame.h"
+#include "modules/video_coding/generic_decoder.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "modules/video_coding/timing/timing.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+VideoReceiver2::VideoReceiver2(Clock* clock,
+                               VCMTiming* timing,
+                               const FieldTrialsView& field_trials)
+    : clock_(clock),
+      decoded_frame_callback_(timing, clock_, field_trials),
+      codec_database_() {
+  decoder_sequence_checker_.Detach();
+}
+
+VideoReceiver2::~VideoReceiver2() {
+  RTC_DCHECK_RUN_ON(&construction_sequence_checker_);
+}
+
+// Register a receive callback. Will be called whenever there is a new frame
+// ready for rendering.
+int32_t VideoReceiver2::RegisterReceiveCallback(
+    VCMReceiveCallback* receive_callback) {
+  RTC_DCHECK_RUN_ON(&construction_sequence_checker_);
+  // This value is set before the decoder thread starts and unset after
+  // the decoder thread has been stopped.
+  decoded_frame_callback_.SetUserReceiveCallback(receive_callback);
+  return VCM_OK;
+}
+
+void VideoReceiver2::RegisterExternalDecoder(
+    std::unique_ptr<VideoDecoder> decoder,
+    uint8_t payload_type) {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  RTC_DCHECK(decoded_frame_callback_.UserReceiveCallback());
+
+  if (decoder) {
+    RTC_DCHECK(!codec_database_.IsExternalDecoderRegistered(payload_type));
+    codec_database_.RegisterExternalDecoder(payload_type, std::move(decoder));
+  } else {
+    codec_database_.DeregisterExternalDecoder(payload_type);
+  }
+}
+
+bool VideoReceiver2::IsExternalDecoderRegistered(uint8_t payload_type) const {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  return codec_database_.IsExternalDecoderRegistered(payload_type);
+}
+
+// Must be called from inside the receive side critical section.
+int32_t VideoReceiver2::Decode(const VCMEncodedFrame* frame) {
+  RTC_DCHECK_RUN_ON(&decoder_sequence_checker_);
+  TRACE_EVENT0("webrtc", "VideoReceiver2::Decode");
+  // Change decoder if payload type has changed.
+  VCMGenericDecoder* decoder =
+      codec_database_.GetDecoder(*frame, &decoded_frame_callback_);
+  if (decoder == nullptr) {
+    return VCM_NO_CODEC_REGISTERED;
+  }
+  return decoder->Decode(*frame, clock_->CurrentTime());
+}
+
+// Register possible receive codecs, can be called multiple times.
+// Called before decoder thread is started.
+void VideoReceiver2::RegisterReceiveCodec(
+    uint8_t payload_type,
+    const VideoDecoder::Settings& settings) {
+  RTC_DCHECK_RUN_ON(&construction_sequence_checker_);
+  codec_database_.RegisterReceiveCodec(payload_type, settings);
+}
+
+void VideoReceiver2::DeregisterReceiveCodec(uint8_t payload_type) {
+  RTC_DCHECK_RUN_ON(&construction_sequence_checker_);
+  codec_database_.DeregisterReceiveCodec(payload_type);
+}
+
+void VideoReceiver2::DeregisterReceiveCodecs() {
+  RTC_DCHECK_RUN_ON(&construction_sequence_checker_);
+  codec_database_.DeregisterReceiveCodecs();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/video_receiver2.h b/third_party/libwebrtc/modules/video_coding/video_receiver2.h
new file mode 100644
index 0000000000..4457a5b5b3
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_receiver2.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_VIDEO_CODING_VIDEO_RECEIVER2_H_
+#define MODULES_VIDEO_CODING_VIDEO_RECEIVER2_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/decoder_database.h"
+#include "modules/video_coding/encoded_frame.h"
+#include "modules/video_coding/generic_decoder.h"
+#include "modules/video_coding/timing/timing.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+// This class is a copy of vcm::VideoReceiver, trimmed down to what's used by
+// VideoReceive stream, with the aim to incrementally trim it down further and
+// ultimately delete it. It's difficult to do this incrementally with the
+// original VideoReceiver class, since it is used by the legacy
+// VideoCodingModule api.
+class VideoReceiver2 {
+ public:
+  VideoReceiver2(Clock* clock,
+                 VCMTiming* timing,
+                 const FieldTrialsView& field_trials);
+  ~VideoReceiver2();
+
+  void RegisterReceiveCodec(uint8_t payload_type,
+                            const VideoDecoder::Settings& decoder_settings);
+  void DeregisterReceiveCodec(uint8_t payload_type);
+  void DeregisterReceiveCodecs();
+
+  void RegisterExternalDecoder(std::unique_ptr<VideoDecoder> decoder,
+                               uint8_t payload_type);
+
+  bool IsExternalDecoderRegistered(uint8_t payload_type) const;
+  int32_t RegisterReceiveCallback(VCMReceiveCallback* receive_callback);
+
+  int32_t Decode(const VCMEncodedFrame* frame);
+
+ private:
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker construction_sequence_checker_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker decoder_sequence_checker_;
+  Clock* const clock_;
+  VCMDecodedFrameCallback decoded_frame_callback_;
+  // Callbacks are set before the decoder thread starts.
+  // Once the decoder thread has been started, usage of `_codecDataBase` moves
+  // over to the decoder thread.
+  VCMDecoderDatabase codec_database_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_VIDEO_RECEIVER2_H_
diff --git a/third_party/libwebrtc/modules/video_coding/video_receiver2_unittest.cc b/third_party/libwebrtc/modules/video_coding/video_receiver2_unittest.cc
new file mode 100644
index 0000000000..6edf1230d8
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_receiver2_unittest.cc
@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/video_receiver2.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/test/mock_video_decoder.h"
+#include "api/units/timestamp.h"
+#include "api/video/encoded_frame.h"
+#include "common_video/test/utilities.h"
+#include "modules/video_coding/decoder_database.h"
+#include "modules/video_coding/timing/timing.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+class MockVCMReceiveCallback : public VCMReceiveCallback {
+ public:
+  MockVCMReceiveCallback() = default;
+
+  MOCK_METHOD(
+      int32_t,
+      FrameToRender,
+      (VideoFrame&, absl::optional<uint8_t>, TimeDelta, VideoContentType),
+      (override));
+  MOCK_METHOD(void, OnIncomingPayloadType, (int), (override));
+  MOCK_METHOD(void,
+              OnDecoderInfoChanged,
+              (const VideoDecoder::DecoderInfo&),
+              (override));
+};
+
+class TestEncodedFrame : public EncodedFrame {
+ public:
+  explicit TestEncodedFrame(int payload_type) {
+    _payloadType = payload_type;
+    SetPacketInfos(CreatePacketInfos(3));
+  }
+
+  void SetReceivedTime(webrtc::Timestamp received_time) {
+    received_time_ = received_time;
+  }
+
+  int64_t ReceivedTime() const override { return received_time_.ms(); }
+
+  int64_t RenderTime() const override { return _renderTimeMs; }
+
+ private:
+  webrtc::Timestamp received_time_ = webrtc::Timestamp::Millis(0);
+};
+
+class VideoReceiver2Test : public ::testing::Test {
+ protected:
+  VideoReceiver2Test() {
+    receiver_.RegisterReceiveCallback(&receive_callback_);
+  }
+
+  void RegisterReceiveCodecSettings(
+      int payload_type,
+      VideoCodecType codec_type = kVideoCodecVP8) {
+    VideoDecoder::Settings settings;
+    settings.set_codec_type(codec_type);
+    settings.set_max_render_resolution({10, 10});
+    settings.set_number_of_cores(4);
+    receiver_.RegisterReceiveCodec(payload_type, settings);
+  }
+
+  test::ScopedKeyValueConfig field_trials_;
+  SimulatedClock clock_{Timestamp::Millis(1337)};
+  VCMTiming timing_{&clock_, field_trials_};
+  NiceMock<MockVCMReceiveCallback> receive_callback_;
+  VideoReceiver2 receiver_{&clock_, &timing_, field_trials_};
+};
+
+TEST_F(VideoReceiver2Test, RegisterExternalDecoder) {
+  constexpr int kPayloadType = 1;
+  ASSERT_FALSE(receiver_.IsExternalDecoderRegistered(kPayloadType));
+
+  // Register a decoder, check for correctness, then unregister and check again.
+  auto decoder = std::make_unique<NiceMock<MockVideoDecoder>>();
+  bool decoder_deleted = false;
+  EXPECT_CALL(*decoder, Destruct).WillOnce([&decoder_deleted] {
+    decoder_deleted = true;
+  });
+  receiver_.RegisterExternalDecoder(std::move(decoder), kPayloadType);
+  EXPECT_TRUE(receiver_.IsExternalDecoderRegistered(kPayloadType));
+  receiver_.RegisterExternalDecoder(nullptr, kPayloadType);
+  EXPECT_TRUE(decoder_deleted);
+  EXPECT_FALSE(receiver_.IsExternalDecoderRegistered(kPayloadType));
+}
+
+TEST_F(VideoReceiver2Test, RegisterReceiveCodecs) {
+  constexpr int kPayloadType = 1;
+
+  RegisterReceiveCodecSettings(kPayloadType);
+
+  TestEncodedFrame frame(kPayloadType);
+
+  // A decoder has not been registered yet, so an attempt to decode should fail.
+  EXPECT_EQ(receiver_.Decode(&frame), VCM_NO_CODEC_REGISTERED);
+
+  // Register a decoder that will accept the Decode operation.
+  auto decoder = std::make_unique<NiceMock<MockVideoDecoder>>();
+  EXPECT_CALL(*decoder, RegisterDecodeCompleteCallback)
+      .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
+  EXPECT_CALL(*decoder, Decode).WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
+  EXPECT_CALL(*decoder, Release).WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
+
+  // Register the decoder. Note that this moves ownership of the mock object
+  // to the `receiver_`.
+  receiver_.RegisterExternalDecoder(std::move(decoder), kPayloadType);
+  EXPECT_TRUE(receiver_.IsExternalDecoderRegistered(kPayloadType));
+
+  EXPECT_CALL(receive_callback_, OnIncomingPayloadType(kPayloadType));
+  EXPECT_CALL(receive_callback_, OnDecoderInfoChanged);
+
+  // Call `Decode`. This triggers the above call expectations.
+  EXPECT_EQ(receiver_.Decode(&frame), VCM_OK);
+
+  // Unregister the decoder and verify.
+  receiver_.RegisterExternalDecoder(nullptr, kPayloadType);
+  EXPECT_FALSE(receiver_.IsExternalDecoderRegistered(kPayloadType));
+
+  receiver_.DeregisterReceiveCodec(kPayloadType);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/video_receiver_unittest.cc b/third_party/libwebrtc/modules/video_coding/video_receiver_unittest.cc
new file mode 100644
index 0000000000..fe9674e521
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/video_receiver_unittest.cc
@@ -0,0 +1,236 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/test/mock_video_decoder.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/timing/timing.h"
+#include "modules/video_coding/video_coding_impl.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/scoped_key_value_config.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::NiceMock;
+
+namespace webrtc {
+namespace vcm {
+namespace {
+
+class MockPacketRequestCallback : public VCMPacketRequestCallback {
+ public:
+  MOCK_METHOD(int32_t,
+              ResendPackets,
+              (const uint16_t* sequenceNumbers, uint16_t length),
+              (override));
+};
+
+class MockVCMReceiveCallback : public VCMReceiveCallback {
+ public:
+  MockVCMReceiveCallback() {}
+  virtual ~MockVCMReceiveCallback() {}
+
+  MOCK_METHOD(
+      int32_t,
+      FrameToRender,
+      (VideoFrame&, absl::optional<uint8_t>, TimeDelta, VideoContentType),
+      (override));
+  MOCK_METHOD(void, OnIncomingPayloadType, (int), (override));
+  MOCK_METHOD(void,
+              OnDecoderInfoChanged,
+              (const VideoDecoder::DecoderInfo&),
+              (override));
+};
+
+class TestVideoReceiver : public ::testing::Test {
+ protected:
+  static const int kUnusedPayloadType = 10;
+  static const uint16_t kMaxWaitTimeMs = 100;
+
+  TestVideoReceiver()
+      : clock_(0),
+        timing_(&clock_, field_trials_),
+        receiver_(&clock_, &timing_, field_trials_) {}
+
+  virtual void SetUp() {
+    // Register decoder.
+    receiver_.RegisterExternalDecoder(&decoder_, kUnusedPayloadType);
+    VideoDecoder::Settings settings;
+    settings.set_codec_type(kVideoCodecVP8);
+    receiver_.RegisterReceiveCodec(kUnusedPayloadType, settings);
+
+    // Set protection mode.
+    const size_t kMaxNackListSize = 250;
+    const int kMaxPacketAgeToNack = 450;
+    receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack, 0);
+    EXPECT_EQ(
+        0, receiver_.RegisterPacketRequestCallback(&packet_request_callback_));
+
+    // Since we call Decode, we need to provide a valid receive callback.
+    // However, for the purposes of these tests, we ignore the callbacks.
+    EXPECT_CALL(receive_callback_, OnIncomingPayloadType(_)).Times(AnyNumber());
+    EXPECT_CALL(receive_callback_, OnDecoderInfoChanged).Times(AnyNumber());
+    receiver_.RegisterReceiveCallback(&receive_callback_);
+  }
+
+  RTPHeader GetDefaultRTPHeader() const {
+    RTPHeader header;
+    header.markerBit = false;
+    header.payloadType = kUnusedPayloadType;
+    header.ssrc = 1;
+    header.headerLength = 12;
+    return header;
+  }
+
+  RTPVideoHeader GetDefaultVp8Header() const {
+    RTPVideoHeader video_header = {};
+    video_header.frame_type = VideoFrameType::kEmptyFrame;
+    video_header.codec = kVideoCodecVP8;
+    return video_header;
+  }
+
+  void InsertAndVerifyPaddingFrame(const uint8_t* payload,
+                                   RTPHeader* header,
+                                   const RTPVideoHeader& video_header) {
+    for (int j = 0; j < 5; ++j) {
+      // Padding only packets are passed to the VCM with payload size 0.
+      EXPECT_EQ(0, receiver_.IncomingPacket(payload, 0, *header, video_header));
+      ++header->sequenceNumber;
+    }
+    receiver_.Process();
+    EXPECT_CALL(decoder_, Decode(_, _, _)).Times(0);
+    EXPECT_EQ(VCM_FRAME_NOT_READY, receiver_.Decode(kMaxWaitTimeMs));
+  }
+
+  void InsertAndVerifyDecodableFrame(const uint8_t* payload,
+                                     size_t length,
+                                     RTPHeader* header,
+                                     const RTPVideoHeader& video_header) {
+    EXPECT_EQ(0,
+              receiver_.IncomingPacket(payload, length, *header, video_header));
+    ++header->sequenceNumber;
+    EXPECT_CALL(packet_request_callback_, ResendPackets(_, _)).Times(0);
+
+    receiver_.Process();
+    EXPECT_CALL(decoder_, Decode(_, _, _)).Times(1);
+    EXPECT_EQ(0, receiver_.Decode(kMaxWaitTimeMs));
+  }
+
+  test::ScopedKeyValueConfig field_trials_;
+  SimulatedClock clock_;
+  NiceMock<MockVideoDecoder> decoder_;
+  NiceMock<MockPacketRequestCallback> packet_request_callback_;
+  VCMTiming timing_;
+  MockVCMReceiveCallback receive_callback_;
+  VideoReceiver receiver_;
+};
+
+TEST_F(TestVideoReceiver, PaddingOnlyFrames) {
+  const size_t kPaddingSize = 220;
+  const uint8_t kPayload[kPaddingSize] = {0};
+  RTPHeader header = GetDefaultRTPHeader();
+  RTPVideoHeader video_header = GetDefaultVp8Header();
+  header.paddingLength = kPaddingSize;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_CALL(packet_request_callback_, ResendPackets(_, _)).Times(0);
+    InsertAndVerifyPaddingFrame(kPayload, &header, video_header);
+    clock_.AdvanceTimeMilliseconds(33);
+    header.timestamp += 3000;
+  }
+}
+
+TEST_F(TestVideoReceiver, PaddingOnlyFramesWithLosses) {
+  const size_t kFrameSize = 1200;
+  const size_t kPaddingSize = 220;
+  const uint8_t kPayload[kFrameSize] = {0};
+  RTPHeader header = GetDefaultRTPHeader();
+  RTPVideoHeader video_header = GetDefaultVp8Header();
+  header.paddingLength = kPaddingSize;
+  video_header.video_type_header.emplace<RTPVideoHeaderVP8>();
+
+  // Insert one video frame to get one frame decoded.
+  video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  video_header.is_first_packet_in_frame = true;
+  header.markerBit = true;
+  InsertAndVerifyDecodableFrame(kPayload, kFrameSize, &header, video_header);
+
+  clock_.AdvanceTimeMilliseconds(33);
+  header.timestamp += 3000;
+  video_header.frame_type = VideoFrameType::kEmptyFrame;
+  video_header.is_first_packet_in_frame = false;
+  header.markerBit = false;
+  // Insert padding frames.
+  for (int i = 0; i < 10; ++i) {
+    // Lose one packet from the 6th frame.
+    if (i == 5) {
+      ++header.sequenceNumber;
+    }
+    // Lose the 4th frame.
+    if (i == 3) {
+      header.sequenceNumber += 5;
+    } else {
+      if (i > 3 && i < 5) {
+        EXPECT_CALL(packet_request_callback_, ResendPackets(_, 5)).Times(1);
+      } else if (i >= 5) {
+        EXPECT_CALL(packet_request_callback_, ResendPackets(_, 6)).Times(1);
+      } else {
+        EXPECT_CALL(packet_request_callback_, ResendPackets(_, _)).Times(0);
+      }
+      InsertAndVerifyPaddingFrame(kPayload, &header, video_header);
+    }
+    clock_.AdvanceTimeMilliseconds(33);
+    header.timestamp += 3000;
+  }
+}
+
+TEST_F(TestVideoReceiver, PaddingOnlyAndVideo) {
+  const size_t kFrameSize = 1200;
+  const size_t kPaddingSize = 220;
+  const uint8_t kPayload[kFrameSize] = {0};
+  RTPHeader header = GetDefaultRTPHeader();
+  RTPVideoHeader video_header = GetDefaultVp8Header();
+  video_header.is_first_packet_in_frame = false;
+  header.paddingLength = kPaddingSize;
+  auto& vp8_header =
+      video_header.video_type_header.emplace<RTPVideoHeaderVP8>();
+  vp8_header.pictureId = -1;
+  vp8_header.tl0PicIdx = -1;
+
+  for (int i = 0; i < 3; ++i) {
+    // Insert 2 video frames.
+    for (int j = 0; j < 2; ++j) {
+      if (i == 0 && j == 0)  // First frame should be a key frame.
+        video_header.frame_type = VideoFrameType::kVideoFrameKey;
+      else
+        video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+      video_header.is_first_packet_in_frame = true;
+      header.markerBit = true;
+      InsertAndVerifyDecodableFrame(kPayload, kFrameSize, &header,
+                                    video_header);
+      clock_.AdvanceTimeMilliseconds(33);
+      header.timestamp += 3000;
+    }
+
+    // Insert 2 padding only frames.
+    video_header.frame_type = VideoFrameType::kEmptyFrame;
+    video_header.is_first_packet_in_frame = false;
+    header.markerBit = false;
+    for (int j = 0; j < 2; ++j) {
+      // InsertAndVerifyPaddingFrame(kPayload, &header);
+      clock_.AdvanceTimeMilliseconds(33);
+      header.timestamp += 3000;
+    }
+  }
+}
+
+}  // namespace
+}  // namespace vcm
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_libvpx_interface_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_libvpx_interface_gn/moz.build
new file mode 100644
index 0000000000..03437c5d57
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/webrtc_libvpx_interface_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/codecs/interface/libvpx_interface.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_libvpx_interface_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp8_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_gn/moz.build
new file mode 100644
index 0000000000..db366355de
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_gn/moz.build
@@ -0,0 +1,235 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/media/libyuv/",
+    "/media/libyuv/libyuv/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc",
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_vp8_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp8_scalability_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_scalability_gn/moz.build
new file mode 100644
index 0000000000..3d5642df00
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_scalability_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp8/vp8_scalability.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_vp8_scalability_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp8_temporal_layers_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_temporal_layers_gn/moz.build
new file mode 100644
index 0000000000..db3848fb57
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp8_temporal_layers_gn/moz.build
@@ -0,0 +1,237 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc",
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_vp8_temporal_layers_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp9_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp9_gn/moz.build
new file mode 100644
index 0000000000..f470652e6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp9_gn/moz.build
@@ -0,0 +1,238 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/media/libyuv/",
+    "/media/libyuv/libyuv/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc",
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc",
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9.cc",
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "GLESv2",
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_vp9_gn")
diff --git a/third_party/libwebrtc/modules/video_coding/webrtc_vp9_helpers_gn/moz.build b/third_party/libwebrtc/modules/video_coding/webrtc_vp9_helpers_gn/moz.build
new file mode 100644
index 0000000000..77a0c38f99
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/webrtc_vp9_helpers_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/video_coding/codecs/vp9/svc_config.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "GLESv2",
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("webrtc_vp9_helpers_gn")